From 6d4baf084f4d8dc43cf5d5a3c182018604afa80c Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 20 Sep 2011 15:44:21 +0100 Subject: ASoC: Add WM5100 driver The WM5100 is a highly integrated low power audio subsystem with advanced digital signal processing capabilities including effects, speech clarity enhancement and active noise cancellation. This initial driver provides support for basic audio paths, further patches will provide more complete functionality. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wm5100-tables.c | 1530 ++++++++++++ sound/soc/codecs/wm5100.c | 2560 +++++++++++++++++++ sound/soc/codecs/wm5100.h | 5146 ++++++++++++++++++++++++++++++++++++++ 5 files changed, 9242 insertions(+) create mode 100644 sound/soc/codecs/wm5100-tables.c create mode 100644 sound/soc/codecs/wm5100.c create mode 100644 sound/soc/codecs/wm5100.h (limited to 'sound') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 71b46c8..45c966c 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -59,6 +59,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_WL1273 if MFD_WL1273_CORE select SND_SOC_WM1250_EV1 if I2C select SND_SOC_WM2000 if I2C + select SND_SOC_WM5100 if I2C select SND_SOC_WM8350 if MFD_WM8350 select SND_SOC_WM8400 if MFD_WM8400 select SND_SOC_WM8510 if SND_SOC_I2C_AND_SPI @@ -273,6 +274,9 @@ config SND_SOC_WL1273 config SND_SOC_WM1250_EV1 tristate +config SND_SOC_WM5100 + tristate + config SND_SOC_WM8350 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 70c1769..4f3ff24 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -44,6 +44,7 @@ snd-soc-uda134x-objs := uda134x.o snd-soc-uda1380-objs := uda1380.o snd-soc-wl1273-objs := wl1273.o snd-soc-wm1250-ev1-objs := wm1250-ev1.o +snd-soc-wm5100-objs := wm5100.o wm5100-tables.o snd-soc-wm8350-objs := wm8350.o snd-soc-wm8400-objs := wm8400.o snd-soc-wm8510-objs := wm8510.o @@ -142,6 +143,7 @@ obj-$(CONFIG_SND_SOC_UDA134X) += snd-soc-uda134x.o obj-$(CONFIG_SND_SOC_UDA1380) += snd-soc-uda1380.o obj-$(CONFIG_SND_SOC_WL1273) += snd-soc-wl1273.o obj-$(CONFIG_SND_SOC_WM1250_EV1) += snd-soc-wm1250-ev1.o +obj-$(CONFIG_SND_SOC_WM5100) += snd-soc-wm5100.o obj-$(CONFIG_SND_SOC_WM8350) += snd-soc-wm8350.o obj-$(CONFIG_SND_SOC_WM8400) += snd-soc-wm8400.o obj-$(CONFIG_SND_SOC_WM8510) += snd-soc-wm8510.o diff --git a/sound/soc/codecs/wm5100-tables.c b/sound/soc/codecs/wm5100-tables.c new file mode 100644 index 0000000..960617b --- /dev/null +++ b/sound/soc/codecs/wm5100-tables.c @@ -0,0 +1,1530 @@ +/* + * wm5100-tables.c -- WM5100 ALSA SoC Audio driver data + * + * Copyright 2011 Wolfson Microelectronics plc + * + * Author: Mark Brown + * + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include "wm5100.h" + +int wm5100_volatile_register(struct snd_soc_codec *codec, unsigned int reg) +{ + switch (reg) { + case WM5100_SOFTWARE_RESET: + case WM5100_DEVICE_REVISION: + case WM5100_FX_CTRL: + case WM5100_INTERRUPT_STATUS_1: + case WM5100_INTERRUPT_STATUS_2: + case WM5100_INTERRUPT_STATUS_3: + case WM5100_INTERRUPT_STATUS_4: + case WM5100_INTERRUPT_RAW_STATUS_2: + case WM5100_INTERRUPT_RAW_STATUS_3: + case WM5100_INTERRUPT_RAW_STATUS_4: + case WM5100_OUTPUT_STATUS_1: + case WM5100_OUTPUT_STATUS_2: + case WM5100_INPUT_ENABLES_STATUS: + case WM5100_MIC_DETECT_3: + return 1; + default: + return 0; + } +} + +int wm5100_readable_register(struct snd_soc_codec *codec, unsigned int reg) +{ + switch (reg) { + case WM5100_SOFTWARE_RESET: + case WM5100_DEVICE_REVISION: + case WM5100_CTRL_IF_1: + case WM5100_TONE_GENERATOR_1: + case WM5100_PWM_DRIVE_1: + case WM5100_PWM_DRIVE_2: + case WM5100_PWM_DRIVE_3: + case WM5100_CLOCKING_1: + case WM5100_CLOCKING_3: + case WM5100_CLOCKING_4: + case WM5100_CLOCKING_5: + case WM5100_CLOCKING_6: + case WM5100_CLOCKING_7: + case WM5100_CLOCKING_8: + case WM5100_ASRC_ENABLE: + case WM5100_ASRC_STATUS: + case WM5100_ASRC_RATE1: + case WM5100_ISRC_1_CTRL_1: + case WM5100_ISRC_1_CTRL_2: + case WM5100_ISRC_2_CTRL1: + case WM5100_ISRC_2_CTRL_2: + case WM5100_FLL1_CONTROL_1: + case WM5100_FLL1_CONTROL_2: + case WM5100_FLL1_CONTROL_3: + case WM5100_FLL1_CONTROL_5: + case WM5100_FLL1_CONTROL_6: + case WM5100_FLL1_EFS_1: + case WM5100_FLL2_CONTROL_1: + case WM5100_FLL2_CONTROL_2: + case WM5100_FLL2_CONTROL_3: + case WM5100_FLL2_CONTROL_5: + case WM5100_FLL2_CONTROL_6: + case WM5100_FLL2_EFS_1: + case WM5100_MIC_CHARGE_PUMP_1: + case WM5100_MIC_CHARGE_PUMP_2: + case WM5100_HP_CHARGE_PUMP_1: + case WM5100_LDO1_CONTROL: + case WM5100_MIC_BIAS_CTRL_1: + case WM5100_MIC_BIAS_CTRL_2: + case WM5100_MIC_BIAS_CTRL_3: + case WM5100_ACCESSORY_DETECT_MODE_1: + case WM5100_HEADPHONE_DETECT_1: + case WM5100_HEADPHONE_DETECT_2: + case WM5100_MIC_DETECT_1: + case WM5100_MIC_DETECT_2: + case WM5100_MIC_DETECT_3: + case WM5100_INPUT_ENABLES: + case WM5100_INPUT_ENABLES_STATUS: + case WM5100_IN1L_CONTROL: + case WM5100_IN1R_CONTROL: + case WM5100_IN2L_CONTROL: + case WM5100_IN2R_CONTROL: + case WM5100_IN3L_CONTROL: + case WM5100_IN3R_CONTROL: + case WM5100_IN4L_CONTROL: + case WM5100_IN4R_CONTROL: + case WM5100_RXANC_SRC: + case WM5100_INPUT_VOLUME_RAMP: + case WM5100_ADC_DIGITAL_VOLUME_1L: + case WM5100_ADC_DIGITAL_VOLUME_1R: + case WM5100_ADC_DIGITAL_VOLUME_2L: + case WM5100_ADC_DIGITAL_VOLUME_2R: + case WM5100_ADC_DIGITAL_VOLUME_3L: + case WM5100_ADC_DIGITAL_VOLUME_3R: + case WM5100_ADC_DIGITAL_VOLUME_4L: + case WM5100_ADC_DIGITAL_VOLUME_4R: + case WM5100_OUTPUT_ENABLES_2: + case WM5100_OUTPUT_STATUS_1: + case WM5100_OUTPUT_STATUS_2: + case WM5100_CHANNEL_ENABLES_1: + case WM5100_OUT_VOLUME_1L: + case WM5100_OUT_VOLUME_1R: + case WM5100_DAC_VOLUME_LIMIT_1L: + case WM5100_DAC_VOLUME_LIMIT_1R: + case WM5100_OUT_VOLUME_2L: + case WM5100_OUT_VOLUME_2R: + case WM5100_DAC_VOLUME_LIMIT_2L: + case WM5100_DAC_VOLUME_LIMIT_2R: + case WM5100_OUT_VOLUME_3L: + case WM5100_OUT_VOLUME_3R: + case WM5100_DAC_VOLUME_LIMIT_3L: + case WM5100_DAC_VOLUME_LIMIT_3R: + case WM5100_OUT_VOLUME_4L: + case WM5100_OUT_VOLUME_4R: + case WM5100_DAC_VOLUME_LIMIT_5L: + case WM5100_DAC_VOLUME_LIMIT_5R: + case WM5100_DAC_VOLUME_LIMIT_6L: + case WM5100_DAC_VOLUME_LIMIT_6R: + case WM5100_DAC_AEC_CONTROL_1: + case WM5100_OUTPUT_VOLUME_RAMP: + case WM5100_DAC_DIGITAL_VOLUME_1L: + case WM5100_DAC_DIGITAL_VOLUME_1R: + case WM5100_DAC_DIGITAL_VOLUME_2L: + case WM5100_DAC_DIGITAL_VOLUME_2R: + case WM5100_DAC_DIGITAL_VOLUME_3L: + case WM5100_DAC_DIGITAL_VOLUME_3R: + case WM5100_DAC_DIGITAL_VOLUME_4L: + case WM5100_DAC_DIGITAL_VOLUME_4R: + case WM5100_DAC_DIGITAL_VOLUME_5L: + case WM5100_DAC_DIGITAL_VOLUME_5R: + case WM5100_DAC_DIGITAL_VOLUME_6L: + case WM5100_DAC_DIGITAL_VOLUME_6R: + case WM5100_PDM_SPK1_CTRL_1: + case WM5100_PDM_SPK1_CTRL_2: + case WM5100_PDM_SPK2_CTRL_1: + case WM5100_PDM_SPK2_CTRL_2: + case WM5100_AUDIO_IF_1_1: + case WM5100_AUDIO_IF_1_2: + case WM5100_AUDIO_IF_1_3: + case WM5100_AUDIO_IF_1_4: + case WM5100_AUDIO_IF_1_5: + case WM5100_AUDIO_IF_1_6: + case WM5100_AUDIO_IF_1_7: + case WM5100_AUDIO_IF_1_8: + case WM5100_AUDIO_IF_1_9: + case WM5100_AUDIO_IF_1_10: + case WM5100_AUDIO_IF_1_11: + case WM5100_AUDIO_IF_1_12: + case WM5100_AUDIO_IF_1_13: + case WM5100_AUDIO_IF_1_14: + case WM5100_AUDIO_IF_1_15: + case WM5100_AUDIO_IF_1_16: + case WM5100_AUDIO_IF_1_17: + case WM5100_AUDIO_IF_1_18: + case WM5100_AUDIO_IF_1_19: + case WM5100_AUDIO_IF_1_20: + case WM5100_AUDIO_IF_1_21: + case WM5100_AUDIO_IF_1_22: + case WM5100_AUDIO_IF_1_23: + case WM5100_AUDIO_IF_1_24: + case WM5100_AUDIO_IF_1_25: + case WM5100_AUDIO_IF_1_26: + case WM5100_AUDIO_IF_1_27: + case WM5100_AUDIO_IF_2_1: + case WM5100_AUDIO_IF_2_2: + case WM5100_AUDIO_IF_2_3: + case WM5100_AUDIO_IF_2_4: + case WM5100_AUDIO_IF_2_5: + case WM5100_AUDIO_IF_2_6: + case WM5100_AUDIO_IF_2_7: + case WM5100_AUDIO_IF_2_8: + case WM5100_AUDIO_IF_2_9: + case WM5100_AUDIO_IF_2_10: + case WM5100_AUDIO_IF_2_11: + case WM5100_AUDIO_IF_2_18: + case WM5100_AUDIO_IF_2_19: + case WM5100_AUDIO_IF_2_26: + case WM5100_AUDIO_IF_2_27: + case WM5100_AUDIO_IF_3_1: + case WM5100_AUDIO_IF_3_2: + case WM5100_AUDIO_IF_3_3: + case WM5100_AUDIO_IF_3_4: + case WM5100_AUDIO_IF_3_5: + case WM5100_AUDIO_IF_3_6: + case WM5100_AUDIO_IF_3_7: + case WM5100_AUDIO_IF_3_8: + case WM5100_AUDIO_IF_3_9: + case WM5100_AUDIO_IF_3_10: + case WM5100_AUDIO_IF_3_11: + case WM5100_AUDIO_IF_3_18: + case WM5100_AUDIO_IF_3_19: + case WM5100_AUDIO_IF_3_26: + case WM5100_AUDIO_IF_3_27: + case WM5100_PWM1MIX_INPUT_1_SOURCE: + case WM5100_PWM1MIX_INPUT_1_VOLUME: + case WM5100_PWM1MIX_INPUT_2_SOURCE: + case WM5100_PWM1MIX_INPUT_2_VOLUME: + case WM5100_PWM1MIX_INPUT_3_SOURCE: + case WM5100_PWM1MIX_INPUT_3_VOLUME: + case WM5100_PWM1MIX_INPUT_4_SOURCE: + case WM5100_PWM1MIX_INPUT_4_VOLUME: + case WM5100_PWM2MIX_INPUT_1_SOURCE: + case WM5100_PWM2MIX_INPUT_1_VOLUME: + case WM5100_PWM2MIX_INPUT_2_SOURCE: + case WM5100_PWM2MIX_INPUT_2_VOLUME: + case WM5100_PWM2MIX_INPUT_3_SOURCE: + case WM5100_PWM2MIX_INPUT_3_VOLUME: + case WM5100_PWM2MIX_INPUT_4_SOURCE: + case WM5100_PWM2MIX_INPUT_4_VOLUME: + case WM5100_OUT1LMIX_INPUT_1_SOURCE: + case WM5100_OUT1LMIX_INPUT_1_VOLUME: + case WM5100_OUT1LMIX_INPUT_2_SOURCE: + case WM5100_OUT1LMIX_INPUT_2_VOLUME: + case WM5100_OUT1LMIX_INPUT_3_SOURCE: + case WM5100_OUT1LMIX_INPUT_3_VOLUME: + case WM5100_OUT1LMIX_INPUT_4_SOURCE: + case WM5100_OUT1LMIX_INPUT_4_VOLUME: + case WM5100_OUT1RMIX_INPUT_1_SOURCE: + case WM5100_OUT1RMIX_INPUT_1_VOLUME: + case WM5100_OUT1RMIX_INPUT_2_SOURCE: + case WM5100_OUT1RMIX_INPUT_2_VOLUME: + case WM5100_OUT1RMIX_INPUT_3_SOURCE: + case WM5100_OUT1RMIX_INPUT_3_VOLUME: + case WM5100_OUT1RMIX_INPUT_4_SOURCE: + case WM5100_OUT1RMIX_INPUT_4_VOLUME: + case WM5100_OUT2LMIX_INPUT_1_SOURCE: + case WM5100_OUT2LMIX_INPUT_1_VOLUME: + case WM5100_OUT2LMIX_INPUT_2_SOURCE: + case WM5100_OUT2LMIX_INPUT_2_VOLUME: + case WM5100_OUT2LMIX_INPUT_3_SOURCE: + case WM5100_OUT2LMIX_INPUT_3_VOLUME: + case WM5100_OUT2LMIX_INPUT_4_SOURCE: + case WM5100_OUT2LMIX_INPUT_4_VOLUME: + case WM5100_OUT2RMIX_INPUT_1_SOURCE: + case WM5100_OUT2RMIX_INPUT_1_VOLUME: + case WM5100_OUT2RMIX_INPUT_2_SOURCE: + case WM5100_OUT2RMIX_INPUT_2_VOLUME: + case WM5100_OUT2RMIX_INPUT_3_SOURCE: + case WM5100_OUT2RMIX_INPUT_3_VOLUME: + case WM5100_OUT2RMIX_INPUT_4_SOURCE: + case WM5100_OUT2RMIX_INPUT_4_VOLUME: + case WM5100_OUT3LMIX_INPUT_1_SOURCE: + case WM5100_OUT3LMIX_INPUT_1_VOLUME: + case WM5100_OUT3LMIX_INPUT_2_SOURCE: + case WM5100_OUT3LMIX_INPUT_2_VOLUME: + case WM5100_OUT3LMIX_INPUT_3_SOURCE: + case WM5100_OUT3LMIX_INPUT_3_VOLUME: + case WM5100_OUT3LMIX_INPUT_4_SOURCE: + case WM5100_OUT3LMIX_INPUT_4_VOLUME: + case WM5100_OUT3RMIX_INPUT_1_SOURCE: + case WM5100_OUT3RMIX_INPUT_1_VOLUME: + case WM5100_OUT3RMIX_INPUT_2_SOURCE: + case WM5100_OUT3RMIX_INPUT_2_VOLUME: + case WM5100_OUT3RMIX_INPUT_3_SOURCE: + case WM5100_OUT3RMIX_INPUT_3_VOLUME: + case WM5100_OUT3RMIX_INPUT_4_SOURCE: + case WM5100_OUT3RMIX_INPUT_4_VOLUME: + case WM5100_OUT4LMIX_INPUT_1_SOURCE: + case WM5100_OUT4LMIX_INPUT_1_VOLUME: + case WM5100_OUT4LMIX_INPUT_2_SOURCE: + case WM5100_OUT4LMIX_INPUT_2_VOLUME: + case WM5100_OUT4LMIX_INPUT_3_SOURCE: + case WM5100_OUT4LMIX_INPUT_3_VOLUME: + case WM5100_OUT4LMIX_INPUT_4_SOURCE: + case WM5100_OUT4LMIX_INPUT_4_VOLUME: + case WM5100_OUT4RMIX_INPUT_1_SOURCE: + case WM5100_OUT4RMIX_INPUT_1_VOLUME: + case WM5100_OUT4RMIX_INPUT_2_SOURCE: + case WM5100_OUT4RMIX_INPUT_2_VOLUME: + case WM5100_OUT4RMIX_INPUT_3_SOURCE: + case WM5100_OUT4RMIX_INPUT_3_VOLUME: + case WM5100_OUT4RMIX_INPUT_4_SOURCE: + case WM5100_OUT4RMIX_INPUT_4_VOLUME: + case WM5100_OUT5LMIX_INPUT_1_SOURCE: + case WM5100_OUT5LMIX_INPUT_1_VOLUME: + case WM5100_OUT5LMIX_INPUT_2_SOURCE: + case WM5100_OUT5LMIX_INPUT_2_VOLUME: + case WM5100_OUT5LMIX_INPUT_3_SOURCE: + case WM5100_OUT5LMIX_INPUT_3_VOLUME: + case WM5100_OUT5LMIX_INPUT_4_SOURCE: + case WM5100_OUT5LMIX_INPUT_4_VOLUME: + case WM5100_OUT5RMIX_INPUT_1_SOURCE: + case WM5100_OUT5RMIX_INPUT_1_VOLUME: + case WM5100_OUT5RMIX_INPUT_2_SOURCE: + case WM5100_OUT5RMIX_INPUT_2_VOLUME: + case WM5100_OUT5RMIX_INPUT_3_SOURCE: + case WM5100_OUT5RMIX_INPUT_3_VOLUME: + case WM5100_OUT5RMIX_INPUT_4_SOURCE: + case WM5100_OUT5RMIX_INPUT_4_VOLUME: + case WM5100_OUT6LMIX_INPUT_1_SOURCE: + case WM5100_OUT6LMIX_INPUT_1_VOLUME: + case WM5100_OUT6LMIX_INPUT_2_SOURCE: + case WM5100_OUT6LMIX_INPUT_2_VOLUME: + case WM5100_OUT6LMIX_INPUT_3_SOURCE: + case WM5100_OUT6LMIX_INPUT_3_VOLUME: + case WM5100_OUT6LMIX_INPUT_4_SOURCE: + case WM5100_OUT6LMIX_INPUT_4_VOLUME: + case WM5100_OUT6RMIX_INPUT_1_SOURCE: + case WM5100_OUT6RMIX_INPUT_1_VOLUME: + case WM5100_OUT6RMIX_INPUT_2_SOURCE: + case WM5100_OUT6RMIX_INPUT_2_VOLUME: + case WM5100_OUT6RMIX_INPUT_3_SOURCE: + case WM5100_OUT6RMIX_INPUT_3_VOLUME: + case WM5100_OUT6RMIX_INPUT_4_SOURCE: + case WM5100_OUT6RMIX_INPUT_4_VOLUME: + case WM5100_AIF1TX1MIX_INPUT_1_SOURCE: + case WM5100_AIF1TX1MIX_INPUT_1_VOLUME: + case WM5100_AIF1TX1MIX_INPUT_2_SOURCE: + case WM5100_AIF1TX1MIX_INPUT_2_VOLUME: + case WM5100_AIF1TX1MIX_INPUT_3_SOURCE: + case WM5100_AIF1TX1MIX_INPUT_3_VOLUME: + case WM5100_AIF1TX1MIX_INPUT_4_SOURCE: + case WM5100_AIF1TX1MIX_INPUT_4_VOLUME: + case WM5100_AIF1TX2MIX_INPUT_1_SOURCE: + case WM5100_AIF1TX2MIX_INPUT_1_VOLUME: + case WM5100_AIF1TX2MIX_INPUT_2_SOURCE: + case WM5100_AIF1TX2MIX_INPUT_2_VOLUME: + case WM5100_AIF1TX2MIX_INPUT_3_SOURCE: + case WM5100_AIF1TX2MIX_INPUT_3_VOLUME: + case WM5100_AIF1TX2MIX_INPUT_4_SOURCE: + case WM5100_AIF1TX2MIX_INPUT_4_VOLUME: + case WM5100_AIF1TX3MIX_INPUT_1_SOURCE: + case WM5100_AIF1TX3MIX_INPUT_1_VOLUME: + case WM5100_AIF1TX3MIX_INPUT_2_SOURCE: + case WM5100_AIF1TX3MIX_INPUT_2_VOLUME: + case WM5100_AIF1TX3MIX_INPUT_3_SOURCE: + case WM5100_AIF1TX3MIX_INPUT_3_VOLUME: + case WM5100_AIF1TX3MIX_INPUT_4_SOURCE: + case WM5100_AIF1TX3MIX_INPUT_4_VOLUME: + case WM5100_AIF1TX4MIX_INPUT_1_SOURCE: + case WM5100_AIF1TX4MIX_INPUT_1_VOLUME: + case WM5100_AIF1TX4MIX_INPUT_2_SOURCE: + case WM5100_AIF1TX4MIX_INPUT_2_VOLUME: + case WM5100_AIF1TX4MIX_INPUT_3_SOURCE: + case WM5100_AIF1TX4MIX_INPUT_3_VOLUME: + case WM5100_AIF1TX4MIX_INPUT_4_SOURCE: + case WM5100_AIF1TX4MIX_INPUT_4_VOLUME: + case WM5100_AIF1TX5MIX_INPUT_1_SOURCE: + case WM5100_AIF1TX5MIX_INPUT_1_VOLUME: + case WM5100_AIF1TX5MIX_INPUT_2_SOURCE: + case WM5100_AIF1TX5MIX_INPUT_2_VOLUME: + case WM5100_AIF1TX5MIX_INPUT_3_SOURCE: + case WM5100_AIF1TX5MIX_INPUT_3_VOLUME: + case WM5100_AIF1TX5MIX_INPUT_4_SOURCE: + case WM5100_AIF1TX5MIX_INPUT_4_VOLUME: + case WM5100_AIF1TX6MIX_INPUT_1_SOURCE: + case WM5100_AIF1TX6MIX_INPUT_1_VOLUME: + case WM5100_AIF1TX6MIX_INPUT_2_SOURCE: + case WM5100_AIF1TX6MIX_INPUT_2_VOLUME: + case WM5100_AIF1TX6MIX_INPUT_3_SOURCE: + case WM5100_AIF1TX6MIX_INPUT_3_VOLUME: + case WM5100_AIF1TX6MIX_INPUT_4_SOURCE: + case WM5100_AIF1TX6MIX_INPUT_4_VOLUME: + case WM5100_AIF1TX7MIX_INPUT_1_SOURCE: + case WM5100_AIF1TX7MIX_INPUT_1_VOLUME: + case WM5100_AIF1TX7MIX_INPUT_2_SOURCE: + case WM5100_AIF1TX7MIX_INPUT_2_VOLUME: + case WM5100_AIF1TX7MIX_INPUT_3_SOURCE: + case WM5100_AIF1TX7MIX_INPUT_3_VOLUME: + case WM5100_AIF1TX7MIX_INPUT_4_SOURCE: + case WM5100_AIF1TX7MIX_INPUT_4_VOLUME: + case WM5100_AIF1TX8MIX_INPUT_1_SOURCE: + case WM5100_AIF1TX8MIX_INPUT_1_VOLUME: + case WM5100_AIF1TX8MIX_INPUT_2_SOURCE: + case WM5100_AIF1TX8MIX_INPUT_2_VOLUME: + case WM5100_AIF1TX8MIX_INPUT_3_SOURCE: + case WM5100_AIF1TX8MIX_INPUT_3_VOLUME: + case WM5100_AIF1TX8MIX_INPUT_4_SOURCE: + case WM5100_AIF1TX8MIX_INPUT_4_VOLUME: + case WM5100_AIF2TX1MIX_INPUT_1_SOURCE: + case WM5100_AIF2TX1MIX_INPUT_1_VOLUME: + case WM5100_AIF2TX1MIX_INPUT_2_SOURCE: + case WM5100_AIF2TX1MIX_INPUT_2_VOLUME: + case WM5100_AIF2TX1MIX_INPUT_3_SOURCE: + case WM5100_AIF2TX1MIX_INPUT_3_VOLUME: + case WM5100_AIF2TX1MIX_INPUT_4_SOURCE: + case WM5100_AIF2TX1MIX_INPUT_4_VOLUME: + case WM5100_AIF2TX2MIX_INPUT_1_SOURCE: + case WM5100_AIF2TX2MIX_INPUT_1_VOLUME: + case WM5100_AIF2TX2MIX_INPUT_2_SOURCE: + case WM5100_AIF2TX2MIX_INPUT_2_VOLUME: + case WM5100_AIF2TX2MIX_INPUT_3_SOURCE: + case WM5100_AIF2TX2MIX_INPUT_3_VOLUME: + case WM5100_AIF2TX2MIX_INPUT_4_SOURCE: + case WM5100_AIF2TX2MIX_INPUT_4_VOLUME: + case WM5100_AIF3TX1MIX_INPUT_1_SOURCE: + case WM5100_AIF3TX1MIX_INPUT_1_VOLUME: + case WM5100_AIF3TX1MIX_INPUT_2_SOURCE: + case WM5100_AIF3TX1MIX_INPUT_2_VOLUME: + case WM5100_AIF3TX1MIX_INPUT_3_SOURCE: + case WM5100_AIF3TX1MIX_INPUT_3_VOLUME: + case WM5100_AIF3TX1MIX_INPUT_4_SOURCE: + case WM5100_AIF3TX1MIX_INPUT_4_VOLUME: + case WM5100_AIF3TX2MIX_INPUT_1_SOURCE: + case WM5100_AIF3TX2MIX_INPUT_1_VOLUME: + case WM5100_AIF3TX2MIX_INPUT_2_SOURCE: + case WM5100_AIF3TX2MIX_INPUT_2_VOLUME: + case WM5100_AIF3TX2MIX_INPUT_3_SOURCE: + case WM5100_AIF3TX2MIX_INPUT_3_VOLUME: + case WM5100_AIF3TX2MIX_INPUT_4_SOURCE: + case WM5100_AIF3TX2MIX_INPUT_4_VOLUME: + case WM5100_EQ1MIX_INPUT_1_SOURCE: + case WM5100_EQ1MIX_INPUT_1_VOLUME: + case WM5100_EQ1MIX_INPUT_2_SOURCE: + case WM5100_EQ1MIX_INPUT_2_VOLUME: + case WM5100_EQ1MIX_INPUT_3_SOURCE: + case WM5100_EQ1MIX_INPUT_3_VOLUME: + case WM5100_EQ1MIX_INPUT_4_SOURCE: + case WM5100_EQ1MIX_INPUT_4_VOLUME: + case WM5100_EQ2MIX_INPUT_1_SOURCE: + case WM5100_EQ2MIX_INPUT_1_VOLUME: + case WM5100_EQ2MIX_INPUT_2_SOURCE: + case WM5100_EQ2MIX_INPUT_2_VOLUME: + case WM5100_EQ2MIX_INPUT_3_SOURCE: + case WM5100_EQ2MIX_INPUT_3_VOLUME: + case WM5100_EQ2MIX_INPUT_4_SOURCE: + case WM5100_EQ2MIX_INPUT_4_VOLUME: + case WM5100_EQ3MIX_INPUT_1_SOURCE: + case WM5100_EQ3MIX_INPUT_1_VOLUME: + case WM5100_EQ3MIX_INPUT_2_SOURCE: + case WM5100_EQ3MIX_INPUT_2_VOLUME: + case WM5100_EQ3MIX_INPUT_3_SOURCE: + case WM5100_EQ3MIX_INPUT_3_VOLUME: + case WM5100_EQ3MIX_INPUT_4_SOURCE: + case WM5100_EQ3MIX_INPUT_4_VOLUME: + case WM5100_EQ4MIX_INPUT_1_SOURCE: + case WM5100_EQ4MIX_INPUT_1_VOLUME: + case WM5100_EQ4MIX_INPUT_2_SOURCE: + case WM5100_EQ4MIX_INPUT_2_VOLUME: + case WM5100_EQ4MIX_INPUT_3_SOURCE: + case WM5100_EQ4MIX_INPUT_3_VOLUME: + case WM5100_EQ4MIX_INPUT_4_SOURCE: + case WM5100_EQ4MIX_INPUT_4_VOLUME: + case WM5100_DRC1LMIX_INPUT_1_SOURCE: + case WM5100_DRC1LMIX_INPUT_1_VOLUME: + case WM5100_DRC1LMIX_INPUT_2_SOURCE: + case WM5100_DRC1LMIX_INPUT_2_VOLUME: + case WM5100_DRC1LMIX_INPUT_3_SOURCE: + case WM5100_DRC1LMIX_INPUT_3_VOLUME: + case WM5100_DRC1LMIX_INPUT_4_SOURCE: + case WM5100_DRC1LMIX_INPUT_4_VOLUME: + case WM5100_DRC1RMIX_INPUT_1_SOURCE: + case WM5100_DRC1RMIX_INPUT_1_VOLUME: + case WM5100_DRC1RMIX_INPUT_2_SOURCE: + case WM5100_DRC1RMIX_INPUT_2_VOLUME: + case WM5100_DRC1RMIX_INPUT_3_SOURCE: + case WM5100_DRC1RMIX_INPUT_3_VOLUME: + case WM5100_DRC1RMIX_INPUT_4_SOURCE: + case WM5100_DRC1RMIX_INPUT_4_VOLUME: + case WM5100_HPLP1MIX_INPUT_1_SOURCE: + case WM5100_HPLP1MIX_INPUT_1_VOLUME: + case WM5100_HPLP1MIX_INPUT_2_SOURCE: + case WM5100_HPLP1MIX_INPUT_2_VOLUME: + case WM5100_HPLP1MIX_INPUT_3_SOURCE: + case WM5100_HPLP1MIX_INPUT_3_VOLUME: + case WM5100_HPLP1MIX_INPUT_4_SOURCE: + case WM5100_HPLP1MIX_INPUT_4_VOLUME: + case WM5100_HPLP2MIX_INPUT_1_SOURCE: + case WM5100_HPLP2MIX_INPUT_1_VOLUME: + case WM5100_HPLP2MIX_INPUT_2_SOURCE: + case WM5100_HPLP2MIX_INPUT_2_VOLUME: + case WM5100_HPLP2MIX_INPUT_3_SOURCE: + case WM5100_HPLP2MIX_INPUT_3_VOLUME: + case WM5100_HPLP2MIX_INPUT_4_SOURCE: + case WM5100_HPLP2MIX_INPUT_4_VOLUME: + case WM5100_HPLP3MIX_INPUT_1_SOURCE: + case WM5100_HPLP3MIX_INPUT_1_VOLUME: + case WM5100_HPLP3MIX_INPUT_2_SOURCE: + case WM5100_HPLP3MIX_INPUT_2_VOLUME: + case WM5100_HPLP3MIX_INPUT_3_SOURCE: + case WM5100_HPLP3MIX_INPUT_3_VOLUME: + case WM5100_HPLP3MIX_INPUT_4_SOURCE: + case WM5100_HPLP3MIX_INPUT_4_VOLUME: + case WM5100_HPLP4MIX_INPUT_1_SOURCE: + case WM5100_HPLP4MIX_INPUT_1_VOLUME: + case WM5100_HPLP4MIX_INPUT_2_SOURCE: + case WM5100_HPLP4MIX_INPUT_2_VOLUME: + case WM5100_HPLP4MIX_INPUT_3_SOURCE: + case WM5100_HPLP4MIX_INPUT_3_VOLUME: + case WM5100_HPLP4MIX_INPUT_4_SOURCE: + case WM5100_HPLP4MIX_INPUT_4_VOLUME: + case WM5100_DSP1LMIX_INPUT_1_SOURCE: + case WM5100_DSP1LMIX_INPUT_1_VOLUME: + case WM5100_DSP1LMIX_INPUT_2_SOURCE: + case WM5100_DSP1LMIX_INPUT_2_VOLUME: + case WM5100_DSP1LMIX_INPUT_3_SOURCE: + case WM5100_DSP1LMIX_INPUT_3_VOLUME: + case WM5100_DSP1LMIX_INPUT_4_SOURCE: + case WM5100_DSP1LMIX_INPUT_4_VOLUME: + case WM5100_DSP1RMIX_INPUT_1_SOURCE: + case WM5100_DSP1RMIX_INPUT_1_VOLUME: + case WM5100_DSP1RMIX_INPUT_2_SOURCE: + case WM5100_DSP1RMIX_INPUT_2_VOLUME: + case WM5100_DSP1RMIX_INPUT_3_SOURCE: + case WM5100_DSP1RMIX_INPUT_3_VOLUME: + case WM5100_DSP1RMIX_INPUT_4_SOURCE: + case WM5100_DSP1RMIX_INPUT_4_VOLUME: + case WM5100_DSP1AUX1MIX_INPUT_1_SOURCE: + case WM5100_DSP1AUX2MIX_INPUT_1_SOURCE: + case WM5100_DSP1AUX3MIX_INPUT_1_SOURCE: + case WM5100_DSP1AUX4MIX_INPUT_1_SOURCE: + case WM5100_DSP1AUX5MIX_INPUT_1_SOURCE: + case WM5100_DSP1AUX6MIX_INPUT_1_SOURCE: + case WM5100_DSP2LMIX_INPUT_1_SOURCE: + case WM5100_DSP2LMIX_INPUT_1_VOLUME: + case WM5100_DSP2LMIX_INPUT_2_SOURCE: + case WM5100_DSP2LMIX_INPUT_2_VOLUME: + case WM5100_DSP2LMIX_INPUT_3_SOURCE: + case WM5100_DSP2LMIX_INPUT_3_VOLUME: + case WM5100_DSP2LMIX_INPUT_4_SOURCE: + case WM5100_DSP2LMIX_INPUT_4_VOLUME: + case WM5100_DSP2RMIX_INPUT_1_SOURCE: + case WM5100_DSP2RMIX_INPUT_1_VOLUME: + case WM5100_DSP2RMIX_INPUT_2_SOURCE: + case WM5100_DSP2RMIX_INPUT_2_VOLUME: + case WM5100_DSP2RMIX_INPUT_3_SOURCE: + case WM5100_DSP2RMIX_INPUT_3_VOLUME: + case WM5100_DSP2RMIX_INPUT_4_SOURCE: + case WM5100_DSP2RMIX_INPUT_4_VOLUME: + case WM5100_DSP2AUX1MIX_INPUT_1_SOURCE: + case WM5100_DSP2AUX2MIX_INPUT_1_SOURCE: + case WM5100_DSP2AUX3MIX_INPUT_1_SOURCE: + case WM5100_DSP2AUX4MIX_INPUT_1_SOURCE: + case WM5100_DSP2AUX5MIX_INPUT_1_SOURCE: + case WM5100_DSP2AUX6MIX_INPUT_1_SOURCE: + case WM5100_DSP3LMIX_INPUT_1_SOURCE: + case WM5100_DSP3LMIX_INPUT_1_VOLUME: + case WM5100_DSP3LMIX_INPUT_2_SOURCE: + case WM5100_DSP3LMIX_INPUT_2_VOLUME: + case WM5100_DSP3LMIX_INPUT_3_SOURCE: + case WM5100_DSP3LMIX_INPUT_3_VOLUME: + case WM5100_DSP3LMIX_INPUT_4_SOURCE: + case WM5100_DSP3LMIX_INPUT_4_VOLUME: + case WM5100_DSP3RMIX_INPUT_1_SOURCE: + case WM5100_DSP3RMIX_INPUT_1_VOLUME: + case WM5100_DSP3RMIX_INPUT_2_SOURCE: + case WM5100_DSP3RMIX_INPUT_2_VOLUME: + case WM5100_DSP3RMIX_INPUT_3_SOURCE: + case WM5100_DSP3RMIX_INPUT_3_VOLUME: + case WM5100_DSP3RMIX_INPUT_4_SOURCE: + case WM5100_DSP3RMIX_INPUT_4_VOLUME: + case WM5100_DSP3AUX1MIX_INPUT_1_SOURCE: + case WM5100_DSP3AUX2MIX_INPUT_1_SOURCE: + case WM5100_DSP3AUX3MIX_INPUT_1_SOURCE: + case WM5100_DSP3AUX4MIX_INPUT_1_SOURCE: + case WM5100_DSP3AUX5MIX_INPUT_1_SOURCE: + case WM5100_DSP3AUX6MIX_INPUT_1_SOURCE: + case WM5100_ASRC1LMIX_INPUT_1_SOURCE: + case WM5100_ASRC1RMIX_INPUT_1_SOURCE: + case WM5100_ASRC2LMIX_INPUT_1_SOURCE: + case WM5100_ASRC2RMIX_INPUT_1_SOURCE: + case WM5100_ISRC1DEC1MIX_INPUT_1_SOURCE: + case WM5100_ISRC1DEC2MIX_INPUT_1_SOURCE: + case WM5100_ISRC1DEC3MIX_INPUT_1_SOURCE: + case WM5100_ISRC1DEC4MIX_INPUT_1_SOURCE: + case WM5100_ISRC1INT1MIX_INPUT_1_SOURCE: + case WM5100_ISRC1INT2MIX_INPUT_1_SOURCE: + case WM5100_ISRC1INT3MIX_INPUT_1_SOURCE: + case WM5100_ISRC1INT4MIX_INPUT_1_SOURCE: + case WM5100_ISRC2DEC1MIX_INPUT_1_SOURCE: + case WM5100_ISRC2DEC2MIX_INPUT_1_SOURCE: + case WM5100_ISRC2DEC3MIX_INPUT_1_SOURCE: + case WM5100_ISRC2DEC4MIX_INPUT_1_SOURCE: + case WM5100_ISRC2INT1MIX_INPUT_1_SOURCE: + case WM5100_ISRC2INT2MIX_INPUT_1_SOURCE: + case WM5100_ISRC2INT3MIX_INPUT_1_SOURCE: + case WM5100_ISRC2INT4MIX_INPUT_1_SOURCE: + case WM5100_GPIO_CTRL_1: + case WM5100_GPIO_CTRL_2: + case WM5100_GPIO_CTRL_3: + case WM5100_GPIO_CTRL_4: + case WM5100_GPIO_CTRL_5: + case WM5100_GPIO_CTRL_6: + case WM5100_MISC_PAD_CTRL_1: + case WM5100_MISC_PAD_CTRL_2: + case WM5100_MISC_PAD_CTRL_3: + case WM5100_MISC_PAD_CTRL_4: + case WM5100_MISC_PAD_CTRL_5: + case WM5100_MISC_GPIO_1: + case WM5100_INTERRUPT_STATUS_1: + case WM5100_INTERRUPT_STATUS_2: + case WM5100_INTERRUPT_STATUS_3: + case WM5100_INTERRUPT_STATUS_4: + case WM5100_INTERRUPT_RAW_STATUS_2: + case WM5100_INTERRUPT_RAW_STATUS_3: + case WM5100_INTERRUPT_RAW_STATUS_4: + case WM5100_INTERRUPT_STATUS_1_MASK: + case WM5100_INTERRUPT_STATUS_2_MASK: + case WM5100_INTERRUPT_STATUS_3_MASK: + case WM5100_INTERRUPT_STATUS_4_MASK: + case WM5100_INTERRUPT_CONTROL: + case WM5100_IRQ_DEBOUNCE_1: + case WM5100_IRQ_DEBOUNCE_2: + case WM5100_FX_CTRL: + case WM5100_EQ1_1: + case WM5100_EQ1_2: + case WM5100_EQ1_3: + case WM5100_EQ1_4: + case WM5100_EQ1_5: + case WM5100_EQ1_6: + case WM5100_EQ1_7: + case WM5100_EQ1_8: + case WM5100_EQ1_9: + case WM5100_EQ1_10: + case WM5100_EQ1_11: + case WM5100_EQ1_12: + case WM5100_EQ1_13: + case WM5100_EQ1_14: + case WM5100_EQ1_15: + case WM5100_EQ1_16: + case WM5100_EQ1_17: + case WM5100_EQ1_18: + case WM5100_EQ1_19: + case WM5100_EQ1_20: + case WM5100_EQ2_1: + case WM5100_EQ2_2: + case WM5100_EQ2_3: + case WM5100_EQ2_4: + case WM5100_EQ2_5: + case WM5100_EQ2_6: + case WM5100_EQ2_7: + case WM5100_EQ2_8: + case WM5100_EQ2_9: + case WM5100_EQ2_10: + case WM5100_EQ2_11: + case WM5100_EQ2_12: + case WM5100_EQ2_13: + case WM5100_EQ2_14: + case WM5100_EQ2_15: + case WM5100_EQ2_16: + case WM5100_EQ2_17: + case WM5100_EQ2_18: + case WM5100_EQ2_19: + case WM5100_EQ2_20: + case WM5100_EQ3_1: + case WM5100_EQ3_2: + case WM5100_EQ3_3: + case WM5100_EQ3_4: + case WM5100_EQ3_5: + case WM5100_EQ3_6: + case WM5100_EQ3_7: + case WM5100_EQ3_8: + case WM5100_EQ3_9: + case WM5100_EQ3_10: + case WM5100_EQ3_11: + case WM5100_EQ3_12: + case WM5100_EQ3_13: + case WM5100_EQ3_14: + case WM5100_EQ3_15: + case WM5100_EQ3_16: + case WM5100_EQ3_17: + case WM5100_EQ3_18: + case WM5100_EQ3_19: + case WM5100_EQ3_20: + case WM5100_EQ4_1: + case WM5100_EQ4_2: + case WM5100_EQ4_3: + case WM5100_EQ4_4: + case WM5100_EQ4_5: + case WM5100_EQ4_6: + case WM5100_EQ4_7: + case WM5100_EQ4_8: + case WM5100_EQ4_9: + case WM5100_EQ4_10: + case WM5100_EQ4_11: + case WM5100_EQ4_12: + case WM5100_EQ4_13: + case WM5100_EQ4_14: + case WM5100_EQ4_15: + case WM5100_EQ4_16: + case WM5100_EQ4_17: + case WM5100_EQ4_18: + case WM5100_EQ4_19: + case WM5100_EQ4_20: + case WM5100_DRC1_CTRL1: + case WM5100_DRC1_CTRL2: + case WM5100_DRC1_CTRL3: + case WM5100_DRC1_CTRL4: + case WM5100_DRC1_CTRL5: + case WM5100_HPLPF1_1: + case WM5100_HPLPF1_2: + case WM5100_HPLPF2_1: + case WM5100_HPLPF2_2: + case WM5100_HPLPF3_1: + case WM5100_HPLPF3_2: + case WM5100_HPLPF4_1: + case WM5100_HPLPF4_2: + case WM5100_DSP1_DM_0: + case WM5100_DSP1_DM_1: + case WM5100_DSP1_DM_2: + case WM5100_DSP1_DM_3: + case WM5100_DSP1_DM_508: + case WM5100_DSP1_DM_509: + case WM5100_DSP1_DM_510: + case WM5100_DSP1_DM_511: + case WM5100_DSP1_PM_0: + case WM5100_DSP1_PM_1: + case WM5100_DSP1_PM_2: + case WM5100_DSP1_PM_3: + case WM5100_DSP1_PM_4: + case WM5100_DSP1_PM_5: + case WM5100_DSP1_PM_1530: + case WM5100_DSP1_PM_1531: + case WM5100_DSP1_PM_1532: + case WM5100_DSP1_PM_1533: + case WM5100_DSP1_PM_1534: + case WM5100_DSP1_PM_1535: + case WM5100_DSP1_ZM_0: + case WM5100_DSP1_ZM_1: + case WM5100_DSP1_ZM_2: + case WM5100_DSP1_ZM_3: + case WM5100_DSP1_ZM_2044: + case WM5100_DSP1_ZM_2045: + case WM5100_DSP1_ZM_2046: + case WM5100_DSP1_ZM_2047: + case WM5100_DSP2_DM_0: + case WM5100_DSP2_DM_1: + case WM5100_DSP2_DM_2: + case WM5100_DSP2_DM_3: + case WM5100_DSP2_DM_508: + case WM5100_DSP2_DM_509: + case WM5100_DSP2_DM_510: + case WM5100_DSP2_DM_511: + case WM5100_DSP2_PM_0: + case WM5100_DSP2_PM_1: + case WM5100_DSP2_PM_2: + case WM5100_DSP2_PM_3: + case WM5100_DSP2_PM_4: + case WM5100_DSP2_PM_5: + case WM5100_DSP2_PM_1530: + case WM5100_DSP2_PM_1531: + case WM5100_DSP2_PM_1532: + case WM5100_DSP2_PM_1533: + case WM5100_DSP2_PM_1534: + case WM5100_DSP2_PM_1535: + case WM5100_DSP2_ZM_0: + case WM5100_DSP2_ZM_1: + case WM5100_DSP2_ZM_2: + case WM5100_DSP2_ZM_3: + case WM5100_DSP2_ZM_2044: + case WM5100_DSP2_ZM_2045: + case WM5100_DSP2_ZM_2046: + case WM5100_DSP2_ZM_2047: + case WM5100_DSP3_DM_0: + case WM5100_DSP3_DM_1: + case WM5100_DSP3_DM_2: + case WM5100_DSP3_DM_3: + case WM5100_DSP3_DM_508: + case WM5100_DSP3_DM_509: + case WM5100_DSP3_DM_510: + case WM5100_DSP3_DM_511: + case WM5100_DSP3_PM_0: + case WM5100_DSP3_PM_1: + case WM5100_DSP3_PM_2: + case WM5100_DSP3_PM_3: + case WM5100_DSP3_PM_4: + case WM5100_DSP3_PM_5: + case WM5100_DSP3_PM_1530: + case WM5100_DSP3_PM_1531: + case WM5100_DSP3_PM_1532: + case WM5100_DSP3_PM_1533: + case WM5100_DSP3_PM_1534: + case WM5100_DSP3_PM_1535: + case WM5100_DSP3_ZM_0: + case WM5100_DSP3_ZM_1: + case WM5100_DSP3_ZM_2: + case WM5100_DSP3_ZM_3: + case WM5100_DSP3_ZM_2044: + case WM5100_DSP3_ZM_2045: + case WM5100_DSP3_ZM_2046: + case WM5100_DSP3_ZM_2047: + return 1; + default: + return 0; + } +} + +u16 wm5100_reg_defaults[WM5100_MAX_REGISTER + 1] = { + [0x0000] = 0x0000, /* R0 - software reset */ + [0x0001] = 0x0000, /* R1 - Device Revision */ + [0x0010] = 0x0801, /* R16 - Ctrl IF 1 */ + [0x0020] = 0x0000, /* R32 - Tone Generator 1 */ + [0x0030] = 0x0000, /* R48 - PWM Drive 1 */ + [0x0031] = 0x0100, /* R49 - PWM Drive 2 */ + [0x0032] = 0x0100, /* R50 - PWM Drive 3 */ + [0x0101] = 0x0000, /* R257 - Clocking 3 */ + [0x0102] = 0x0011, /* R258 - Clocking 4 */ + [0x0103] = 0x0011, /* R259 - Clocking 5 */ + [0x0104] = 0x0011, /* R260 - Clocking 6 */ + [0x0107] = 0x0000, /* R263 - Clocking 7 */ + [0x0108] = 0x0000, /* R264 - Clocking 8 */ + [0x0120] = 0x0000, /* R288 - ASRC_ENABLE */ + [0x0121] = 0x0000, /* R289 - ASRC_STATUS */ + [0x0122] = 0x0000, /* R290 - ASRC_RATE1 */ + [0x0141] = 0x8000, /* R321 - ISRC 1 CTRL 1 */ + [0x0142] = 0x0000, /* R322 - ISRC 1 CTRL 2 */ + [0x0143] = 0x8000, /* R323 - ISRC 2 CTRL1 */ + [0x0144] = 0x0000, /* R324 - ISRC 2 CTRL 2 */ + [0x0182] = 0x0000, /* R386 - FLL1 Control 1 */ + [0x0183] = 0x0000, /* R387 - FLL1 Control 2 */ + [0x0184] = 0x0000, /* R388 - FLL1 Control 3 */ + [0x0186] = 0x0177, /* R390 - FLL1 Control 5 */ + [0x0187] = 0x0001, /* R391 - FLL1 Control 6 */ + [0x0188] = 0x0000, /* R392 - FLL1 EFS 1 */ + [0x01A2] = 0x0000, /* R418 - FLL2 Control 1 */ + [0x01A3] = 0x0000, /* R419 - FLL2 Control 2 */ + [0x01A4] = 0x0000, /* R420 - FLL2 Control 3 */ + [0x01A6] = 0x0177, /* R422 - FLL2 Control 5 */ + [0x01A7] = 0x0001, /* R423 - FLL2 Control 6 */ + [0x01A8] = 0x0000, /* R424 - FLL2 EFS 1 */ + [0x0200] = 0x0020, /* R512 - Mic Charge Pump 1 */ + [0x0201] = 0xB084, /* R513 - Mic Charge Pump 2 */ + [0x0202] = 0xBBDE, /* R514 - HP Charge Pump 1 */ + [0x0211] = 0x20D4, /* R529 - LDO1 Control */ + [0x0215] = 0x0062, /* R533 - Mic Bias Ctrl 1 */ + [0x0216] = 0x0062, /* R534 - Mic Bias Ctrl 2 */ + [0x0217] = 0x0062, /* R535 - Mic Bias Ctrl 3 */ + [0x0280] = 0x0004, /* R640 - Accessory Detect Mode 1 */ + [0x0288] = 0x0020, /* R648 - Headphone Detect 1 */ + [0x0289] = 0x0000, /* R649 - Headphone Detect 2 */ + [0x0290] = 0x1100, /* R656 - Mic Detect 1 */ + [0x0291] = 0x009F, /* R657 - Mic Detect 2 */ + [0x0292] = 0x0000, /* R658 - Mic Detect 3 */ + [0x0301] = 0x0000, /* R769 - Input Enables */ + [0x0302] = 0x0000, /* R770 - Input Enables Status */ + [0x0310] = 0x2280, /* R784 - Status */ + [0x0311] = 0x0080, /* R785 - IN1R Control */ + [0x0312] = 0x2280, /* R786 - IN2L Control */ + [0x0313] = 0x0080, /* R787 - IN2R Control */ + [0x0314] = 0x2280, /* R788 - IN3L Control */ + [0x0315] = 0x0080, /* R789 - IN3R Control */ + [0x0316] = 0x2280, /* R790 - IN4L Control */ + [0x0317] = 0x0080, /* R791 - IN4R Control */ + [0x0318] = 0x0000, /* R792 - RXANC_SRC */ + [0x0319] = 0x0022, /* R793 - Input Volume Ramp */ + [0x0320] = 0x0180, /* R800 - ADC Digital Volume 1L */ + [0x0321] = 0x0180, /* R801 - ADC Digital Volume 1R */ + [0x0322] = 0x0180, /* R802 - ADC Digital Volume 2L */ + [0x0323] = 0x0180, /* R803 - ADC Digital Volume 2R */ + [0x0324] = 0x0180, /* R804 - ADC Digital Volume 3L */ + [0x0325] = 0x0180, /* R805 - ADC Digital Volume 3R */ + [0x0326] = 0x0180, /* R806 - ADC Digital Volume 4L */ + [0x0327] = 0x0180, /* R807 - ADC Digital Volume 4R */ + [0x0401] = 0x0000, /* R1025 - Output Enables 2 */ + [0x0402] = 0x0000, /* R1026 - Output Status 1 */ + [0x0403] = 0x0000, /* R1027 - Output Status 2 */ + [0x0408] = 0x0000, /* R1032 - Channel Enables 1 */ + [0x0410] = 0x0080, /* R1040 - Out Volume 1L */ + [0x0411] = 0x0080, /* R1041 - Out Volume 1R */ + [0x0412] = 0x0080, /* R1042 - DAC Volume Limit 1L */ + [0x0413] = 0x0080, /* R1043 - DAC Volume Limit 1R */ + [0x0414] = 0x0080, /* R1044 - Out Volume 2L */ + [0x0415] = 0x0080, /* R1045 - Out Volume 2R */ + [0x0416] = 0x0080, /* R1046 - DAC Volume Limit 2L */ + [0x0417] = 0x0080, /* R1047 - DAC Volume Limit 2R */ + [0x0418] = 0x0080, /* R1048 - Out Volume 3L */ + [0x0419] = 0x0080, /* R1049 - Out Volume 3R */ + [0x041A] = 0x0080, /* R1050 - DAC Volume Limit 3L */ + [0x041B] = 0x0080, /* R1051 - DAC Volume Limit 3R */ + [0x041C] = 0x0080, /* R1052 - Out Volume 4L */ + [0x041D] = 0x0080, /* R1053 - Out Volume 4R */ + [0x041E] = 0x0080, /* R1054 - DAC Volume Limit 5L */ + [0x041F] = 0x0080, /* R1055 - DAC Volume Limit 5R */ + [0x0420] = 0x0080, /* R1056 - DAC Volume Limit 6L */ + [0x0421] = 0x0080, /* R1057 - DAC Volume Limit 6R */ + [0x0440] = 0x0000, /* R1088 - DAC AEC Control 1 */ + [0x0441] = 0x0022, /* R1089 - Output Volume Ramp */ + [0x0480] = 0x0180, /* R1152 - DAC Digital Volume 1L */ + [0x0481] = 0x0180, /* R1153 - DAC Digital Volume 1R */ + [0x0482] = 0x0180, /* R1154 - DAC Digital Volume 2L */ + [0x0483] = 0x0180, /* R1155 - DAC Digital Volume 2R */ + [0x0484] = 0x0180, /* R1156 - DAC Digital Volume 3L */ + [0x0485] = 0x0180, /* R1157 - DAC Digital Volume 3R */ + [0x0486] = 0x0180, /* R1158 - DAC Digital Volume 4L */ + [0x0487] = 0x0180, /* R1159 - DAC Digital Volume 4R */ + [0x0488] = 0x0180, /* R1160 - DAC Digital Volume 5L */ + [0x0489] = 0x0180, /* R1161 - DAC Digital Volume 5R */ + [0x048A] = 0x0180, /* R1162 - DAC Digital Volume 6L */ + [0x048B] = 0x0180, /* R1163 - DAC Digital Volume 6R */ + [0x04C0] = 0x0069, /* R1216 - PDM SPK1 CTRL 1 */ + [0x04C1] = 0x0000, /* R1217 - PDM SPK1 CTRL 2 */ + [0x04C2] = 0x0069, /* R1218 - PDM SPK2 CTRL 1 */ + [0x04C3] = 0x0000, /* R1219 - PDM SPK2 CTRL 2 */ + [0x0500] = 0x000C, /* R1280 - Audio IF 1_1 */ + [0x0501] = 0x0008, /* R1281 - Audio IF 1_2 */ + [0x0502] = 0x0000, /* R1282 - Audio IF 1_3 */ + [0x0503] = 0x0000, /* R1283 - Audio IF 1_4 */ + [0x0504] = 0x0000, /* R1284 - Audio IF 1_5 */ + [0x0505] = 0x0300, /* R1285 - Audio IF 1_6 */ + [0x0506] = 0x0300, /* R1286 - Audio IF 1_7 */ + [0x0507] = 0x1820, /* R1287 - Audio IF 1_8 */ + [0x0508] = 0x1820, /* R1288 - Audio IF 1_9 */ + [0x0509] = 0x0000, /* R1289 - Audio IF 1_10 */ + [0x050A] = 0x0001, /* R1290 - Audio IF 1_11 */ + [0x050B] = 0x0002, /* R1291 - Audio IF 1_12 */ + [0x050C] = 0x0003, /* R1292 - Audio IF 1_13 */ + [0x050D] = 0x0004, /* R1293 - Audio IF 1_14 */ + [0x050E] = 0x0005, /* R1294 - Audio IF 1_15 */ + [0x050F] = 0x0006, /* R1295 - Audio IF 1_16 */ + [0x0510] = 0x0007, /* R1296 - Audio IF 1_17 */ + [0x0511] = 0x0000, /* R1297 - Audio IF 1_18 */ + [0x0512] = 0x0001, /* R1298 - Audio IF 1_19 */ + [0x0513] = 0x0002, /* R1299 - Audio IF 1_20 */ + [0x0514] = 0x0003, /* R1300 - Audio IF 1_21 */ + [0x0515] = 0x0004, /* R1301 - Audio IF 1_22 */ + [0x0516] = 0x0005, /* R1302 - Audio IF 1_23 */ + [0x0517] = 0x0006, /* R1303 - Audio IF 1_24 */ + [0x0518] = 0x0007, /* R1304 - Audio IF 1_25 */ + [0x0519] = 0x0000, /* R1305 - Audio IF 1_26 */ + [0x051A] = 0x0000, /* R1306 - Audio IF 1_27 */ + [0x0540] = 0x000C, /* R1344 - Audio IF 2_1 */ + [0x0541] = 0x0008, /* R1345 - Audio IF 2_2 */ + [0x0542] = 0x0000, /* R1346 - Audio IF 2_3 */ + [0x0543] = 0x0000, /* R1347 - Audio IF 2_4 */ + [0x0544] = 0x0000, /* R1348 - Audio IF 2_5 */ + [0x0545] = 0x0300, /* R1349 - Audio IF 2_6 */ + [0x0546] = 0x0300, /* R1350 - Audio IF 2_7 */ + [0x0547] = 0x1820, /* R1351 - Audio IF 2_8 */ + [0x0548] = 0x1820, /* R1352 - Audio IF 2_9 */ + [0x0549] = 0x0000, /* R1353 - Audio IF 2_10 */ + [0x054A] = 0x0001, /* R1354 - Audio IF 2_11 */ + [0x0551] = 0x0000, /* R1361 - Audio IF 2_18 */ + [0x0552] = 0x0001, /* R1362 - Audio IF 2_19 */ + [0x0559] = 0x0000, /* R1369 - Audio IF 2_26 */ + [0x055A] = 0x0000, /* R1370 - Audio IF 2_27 */ + [0x0580] = 0x000C, /* R1408 - Audio IF 3_1 */ + [0x0581] = 0x0008, /* R1409 - Audio IF 3_2 */ + [0x0582] = 0x0000, /* R1410 - Audio IF 3_3 */ + [0x0583] = 0x0000, /* R1411 - Audio IF 3_4 */ + [0x0584] = 0x0000, /* R1412 - Audio IF 3_5 */ + [0x0585] = 0x0300, /* R1413 - Audio IF 3_6 */ + [0x0586] = 0x0300, /* R1414 - Audio IF 3_7 */ + [0x0587] = 0x1820, /* R1415 - Audio IF 3_8 */ + [0x0588] = 0x1820, /* R1416 - Audio IF 3_9 */ + [0x0589] = 0x0000, /* R1417 - Audio IF 3_10 */ + [0x058A] = 0x0001, /* R1418 - Audio IF 3_11 */ + [0x0591] = 0x0000, /* R1425 - Audio IF 3_18 */ + [0x0592] = 0x0001, /* R1426 - Audio IF 3_19 */ + [0x0599] = 0x0000, /* R1433 - Audio IF 3_26 */ + [0x059A] = 0x0000, /* R1434 - Audio IF 3_27 */ + [0x0640] = 0x0000, /* R1600 - PWM1MIX Input 1 Source */ + [0x0641] = 0x0080, /* R1601 - PWM1MIX Input 1 Volume */ + [0x0642] = 0x0000, /* R1602 - PWM1MIX Input 2 Source */ + [0x0643] = 0x0080, /* R1603 - PWM1MIX Input 2 Volume */ + [0x0644] = 0x0000, /* R1604 - PWM1MIX Input 3 Source */ + [0x0645] = 0x0080, /* R1605 - PWM1MIX Input 3 Volume */ + [0x0646] = 0x0000, /* R1606 - PWM1MIX Input 4 Source */ + [0x0647] = 0x0080, /* R1607 - PWM1MIX Input 4 Volume */ + [0x0648] = 0x0000, /* R1608 - PWM2MIX Input 1 Source */ + [0x0649] = 0x0080, /* R1609 - PWM2MIX Input 1 Volume */ + [0x064A] = 0x0000, /* R1610 - PWM2MIX Input 2 Source */ + [0x064B] = 0x0080, /* R1611 - PWM2MIX Input 2 Volume */ + [0x064C] = 0x0000, /* R1612 - PWM2MIX Input 3 Source */ + [0x064D] = 0x0080, /* R1613 - PWM2MIX Input 3 Volume */ + [0x064E] = 0x0000, /* R1614 - PWM2MIX Input 4 Source */ + [0x064F] = 0x0080, /* R1615 - PWM2MIX Input 4 Volume */ + [0x0680] = 0x0000, /* R1664 - OUT1LMIX Input 1 Source */ + [0x0681] = 0x0080, /* R1665 - OUT1LMIX Input 1 Volume */ + [0x0682] = 0x0000, /* R1666 - OUT1LMIX Input 2 Source */ + [0x0683] = 0x0080, /* R1667 - OUT1LMIX Input 2 Volume */ + [0x0684] = 0x0000, /* R1668 - OUT1LMIX Input 3 Source */ + [0x0685] = 0x0080, /* R1669 - OUT1LMIX Input 3 Volume */ + [0x0686] = 0x0000, /* R1670 - OUT1LMIX Input 4 Source */ + [0x0687] = 0x0080, /* R1671 - OUT1LMIX Input 4 Volume */ + [0x0688] = 0x0000, /* R1672 - OUT1RMIX Input 1 Source */ + [0x0689] = 0x0080, /* R1673 - OUT1RMIX Input 1 Volume */ + [0x068A] = 0x0000, /* R1674 - OUT1RMIX Input 2 Source */ + [0x068B] = 0x0080, /* R1675 - OUT1RMIX Input 2 Volume */ + [0x068C] = 0x0000, /* R1676 - OUT1RMIX Input 3 Source */ + [0x068D] = 0x0080, /* R1677 - OUT1RMIX Input 3 Volume */ + [0x068E] = 0x0000, /* R1678 - OUT1RMIX Input 4 Source */ + [0x068F] = 0x0080, /* R1679 - OUT1RMIX Input 4 Volume */ + [0x0690] = 0x0000, /* R1680 - OUT2LMIX Input 1 Source */ + [0x0691] = 0x0080, /* R1681 - OUT2LMIX Input 1 Volume */ + [0x0692] = 0x0000, /* R1682 - OUT2LMIX Input 2 Source */ + [0x0693] = 0x0080, /* R1683 - OUT2LMIX Input 2 Volume */ + [0x0694] = 0x0000, /* R1684 - OUT2LMIX Input 3 Source */ + [0x0695] = 0x0080, /* R1685 - OUT2LMIX Input 3 Volume */ + [0x0696] = 0x0000, /* R1686 - OUT2LMIX Input 4 Source */ + [0x0697] = 0x0080, /* R1687 - OUT2LMIX Input 4 Volume */ + [0x0698] = 0x0000, /* R1688 - OUT2RMIX Input 1 Source */ + [0x0699] = 0x0080, /* R1689 - OUT2RMIX Input 1 Volume */ + [0x069A] = 0x0000, /* R1690 - OUT2RMIX Input 2 Source */ + [0x069B] = 0x0080, /* R1691 - OUT2RMIX Input 2 Volume */ + [0x069C] = 0x0000, /* R1692 - OUT2RMIX Input 3 Source */ + [0x069D] = 0x0080, /* R1693 - OUT2RMIX Input 3 Volume */ + [0x069E] = 0x0000, /* R1694 - OUT2RMIX Input 4 Source */ + [0x069F] = 0x0080, /* R1695 - OUT2RMIX Input 4 Volume */ + [0x06A0] = 0x0000, /* R1696 - OUT3LMIX Input 1 Source */ + [0x06A1] = 0x0080, /* R1697 - OUT3LMIX Input 1 Volume */ + [0x06A2] = 0x0000, /* R1698 - OUT3LMIX Input 2 Source */ + [0x06A3] = 0x0080, /* R1699 - OUT3LMIX Input 2 Volume */ + [0x06A4] = 0x0000, /* R1700 - OUT3LMIX Input 3 Source */ + [0x06A5] = 0x0080, /* R1701 - OUT3LMIX Input 3 Volume */ + [0x06A6] = 0x0000, /* R1702 - OUT3LMIX Input 4 Source */ + [0x06A7] = 0x0080, /* R1703 - OUT3LMIX Input 4 Volume */ + [0x06A8] = 0x0000, /* R1704 - OUT3RMIX Input 1 Source */ + [0x06A9] = 0x0080, /* R1705 - OUT3RMIX Input 1 Volume */ + [0x06AA] = 0x0000, /* R1706 - OUT3RMIX Input 2 Source */ + [0x06AB] = 0x0080, /* R1707 - OUT3RMIX Input 2 Volume */ + [0x06AC] = 0x0000, /* R1708 - OUT3RMIX Input 3 Source */ + [0x06AD] = 0x0080, /* R1709 - OUT3RMIX Input 3 Volume */ + [0x06AE] = 0x0000, /* R1710 - OUT3RMIX Input 4 Source */ + [0x06AF] = 0x0080, /* R1711 - OUT3RMIX Input 4 Volume */ + [0x06B0] = 0x0000, /* R1712 - OUT4LMIX Input 1 Source */ + [0x06B1] = 0x0080, /* R1713 - OUT4LMIX Input 1 Volume */ + [0x06B2] = 0x0000, /* R1714 - OUT4LMIX Input 2 Source */ + [0x06B3] = 0x0080, /* R1715 - OUT4LMIX Input 2 Volume */ + [0x06B4] = 0x0000, /* R1716 - OUT4LMIX Input 3 Source */ + [0x06B5] = 0x0080, /* R1717 - OUT4LMIX Input 3 Volume */ + [0x06B6] = 0x0000, /* R1718 - OUT4LMIX Input 4 Source */ + [0x06B7] = 0x0080, /* R1719 - OUT4LMIX Input 4 Volume */ + [0x06B8] = 0x0000, /* R1720 - OUT4RMIX Input 1 Source */ + [0x06B9] = 0x0080, /* R1721 - OUT4RMIX Input 1 Volume */ + [0x06BA] = 0x0000, /* R1722 - OUT4RMIX Input 2 Source */ + [0x06BB] = 0x0080, /* R1723 - OUT4RMIX Input 2 Volume */ + [0x06BC] = 0x0000, /* R1724 - OUT4RMIX Input 3 Source */ + [0x06BD] = 0x0080, /* R1725 - OUT4RMIX Input 3 Volume */ + [0x06BE] = 0x0000, /* R1726 - OUT4RMIX Input 4 Source */ + [0x06BF] = 0x0080, /* R1727 - OUT4RMIX Input 4 Volume */ + [0x06C0] = 0x0000, /* R1728 - OUT5LMIX Input 1 Source */ + [0x06C1] = 0x0080, /* R1729 - OUT5LMIX Input 1 Volume */ + [0x06C2] = 0x0000, /* R1730 - OUT5LMIX Input 2 Source */ + [0x06C3] = 0x0080, /* R1731 - OUT5LMIX Input 2 Volume */ + [0x06C4] = 0x0000, /* R1732 - OUT5LMIX Input 3 Source */ + [0x06C5] = 0x0080, /* R1733 - OUT5LMIX Input 3 Volume */ + [0x06C6] = 0x0000, /* R1734 - OUT5LMIX Input 4 Source */ + [0x06C7] = 0x0080, /* R1735 - OUT5LMIX Input 4 Volume */ + [0x06C8] = 0x0000, /* R1736 - OUT5RMIX Input 1 Source */ + [0x06C9] = 0x0080, /* R1737 - OUT5RMIX Input 1 Volume */ + [0x06CA] = 0x0000, /* R1738 - OUT5RMIX Input 2 Source */ + [0x06CB] = 0x0080, /* R1739 - OUT5RMIX Input 2 Volume */ + [0x06CC] = 0x0000, /* R1740 - OUT5RMIX Input 3 Source */ + [0x06CD] = 0x0080, /* R1741 - OUT5RMIX Input 3 Volume */ + [0x06CE] = 0x0000, /* R1742 - OUT5RMIX Input 4 Source */ + [0x06CF] = 0x0080, /* R1743 - OUT5RMIX Input 4 Volume */ + [0x06D0] = 0x0000, /* R1744 - OUT6LMIX Input 1 Source */ + [0x06D1] = 0x0080, /* R1745 - OUT6LMIX Input 1 Volume */ + [0x06D2] = 0x0000, /* R1746 - OUT6LMIX Input 2 Source */ + [0x06D3] = 0x0080, /* R1747 - OUT6LMIX Input 2 Volume */ + [0x06D4] = 0x0000, /* R1748 - OUT6LMIX Input 3 Source */ + [0x06D5] = 0x0080, /* R1749 - OUT6LMIX Input 3 Volume */ + [0x06D6] = 0x0000, /* R1750 - OUT6LMIX Input 4 Source */ + [0x06D7] = 0x0080, /* R1751 - OUT6LMIX Input 4 Volume */ + [0x06D8] = 0x0000, /* R1752 - OUT6RMIX Input 1 Source */ + [0x06D9] = 0x0080, /* R1753 - OUT6RMIX Input 1 Volume */ + [0x06DA] = 0x0000, /* R1754 - OUT6RMIX Input 2 Source */ + [0x06DB] = 0x0080, /* R1755 - OUT6RMIX Input 2 Volume */ + [0x06DC] = 0x0000, /* R1756 - OUT6RMIX Input 3 Source */ + [0x06DD] = 0x0080, /* R1757 - OUT6RMIX Input 3 Volume */ + [0x06DE] = 0x0000, /* R1758 - OUT6RMIX Input 4 Source */ + [0x06DF] = 0x0080, /* R1759 - OUT6RMIX Input 4 Volume */ + [0x0700] = 0x0000, /* R1792 - AIF1TX1MIX Input 1 Source */ + [0x0701] = 0x0080, /* R1793 - AIF1TX1MIX Input 1 Volume */ + [0x0702] = 0x0000, /* R1794 - AIF1TX1MIX Input 2 Source */ + [0x0703] = 0x0080, /* R1795 - AIF1TX1MIX Input 2 Volume */ + [0x0704] = 0x0000, /* R1796 - AIF1TX1MIX Input 3 Source */ + [0x0705] = 0x0080, /* R1797 - AIF1TX1MIX Input 3 Volume */ + [0x0706] = 0x0000, /* R1798 - AIF1TX1MIX Input 4 Source */ + [0x0707] = 0x0080, /* R1799 - AIF1TX1MIX Input 4 Volume */ + [0x0708] = 0x0000, /* R1800 - AIF1TX2MIX Input 1 Source */ + [0x0709] = 0x0080, /* R1801 - AIF1TX2MIX Input 1 Volume */ + [0x070A] = 0x0000, /* R1802 - AIF1TX2MIX Input 2 Source */ + [0x070B] = 0x0080, /* R1803 - AIF1TX2MIX Input 2 Volume */ + [0x070C] = 0x0000, /* R1804 - AIF1TX2MIX Input 3 Source */ + [0x070D] = 0x0080, /* R1805 - AIF1TX2MIX Input 3 Volume */ + [0x070E] = 0x0000, /* R1806 - AIF1TX2MIX Input 4 Source */ + [0x070F] = 0x0080, /* R1807 - AIF1TX2MIX Input 4 Volume */ + [0x0710] = 0x0000, /* R1808 - AIF1TX3MIX Input 1 Source */ + [0x0711] = 0x0080, /* R1809 - AIF1TX3MIX Input 1 Volume */ + [0x0712] = 0x0000, /* R1810 - AIF1TX3MIX Input 2 Source */ + [0x0713] = 0x0080, /* R1811 - AIF1TX3MIX Input 2 Volume */ + [0x0714] = 0x0000, /* R1812 - AIF1TX3MIX Input 3 Source */ + [0x0715] = 0x0080, /* R1813 - AIF1TX3MIX Input 3 Volume */ + [0x0716] = 0x0000, /* R1814 - AIF1TX3MIX Input 4 Source */ + [0x0717] = 0x0080, /* R1815 - AIF1TX3MIX Input 4 Volume */ + [0x0718] = 0x0000, /* R1816 - AIF1TX4MIX Input 1 Source */ + [0x0719] = 0x0080, /* R1817 - AIF1TX4MIX Input 1 Volume */ + [0x071A] = 0x0000, /* R1818 - AIF1TX4MIX Input 2 Source */ + [0x071B] = 0x0080, /* R1819 - AIF1TX4MIX Input 2 Volume */ + [0x071C] = 0x0000, /* R1820 - AIF1TX4MIX Input 3 Source */ + [0x071D] = 0x0080, /* R1821 - AIF1TX4MIX Input 3 Volume */ + [0x071E] = 0x0000, /* R1822 - AIF1TX4MIX Input 4 Source */ + [0x071F] = 0x0080, /* R1823 - AIF1TX4MIX Input 4 Volume */ + [0x0720] = 0x0000, /* R1824 - AIF1TX5MIX Input 1 Source */ + [0x0721] = 0x0080, /* R1825 - AIF1TX5MIX Input 1 Volume */ + [0x0722] = 0x0000, /* R1826 - AIF1TX5MIX Input 2 Source */ + [0x0723] = 0x0080, /* R1827 - AIF1TX5MIX Input 2 Volume */ + [0x0724] = 0x0000, /* R1828 - AIF1TX5MIX Input 3 Source */ + [0x0725] = 0x0080, /* R1829 - AIF1TX5MIX Input 3 Volume */ + [0x0726] = 0x0000, /* R1830 - AIF1TX5MIX Input 4 Source */ + [0x0727] = 0x0080, /* R1831 - AIF1TX5MIX Input 4 Volume */ + [0x0728] = 0x0000, /* R1832 - AIF1TX6MIX Input 1 Source */ + [0x0729] = 0x0080, /* R1833 - AIF1TX6MIX Input 1 Volume */ + [0x072A] = 0x0000, /* R1834 - AIF1TX6MIX Input 2 Source */ + [0x072B] = 0x0080, /* R1835 - AIF1TX6MIX Input 2 Volume */ + [0x072C] = 0x0000, /* R1836 - AIF1TX6MIX Input 3 Source */ + [0x072D] = 0x0080, /* R1837 - AIF1TX6MIX Input 3 Volume */ + [0x072E] = 0x0000, /* R1838 - AIF1TX6MIX Input 4 Source */ + [0x072F] = 0x0080, /* R1839 - AIF1TX6MIX Input 4 Volume */ + [0x0730] = 0x0000, /* R1840 - AIF1TX7MIX Input 1 Source */ + [0x0731] = 0x0080, /* R1841 - AIF1TX7MIX Input 1 Volume */ + [0x0732] = 0x0000, /* R1842 - AIF1TX7MIX Input 2 Source */ + [0x0733] = 0x0080, /* R1843 - AIF1TX7MIX Input 2 Volume */ + [0x0734] = 0x0000, /* R1844 - AIF1TX7MIX Input 3 Source */ + [0x0735] = 0x0080, /* R1845 - AIF1TX7MIX Input 3 Volume */ + [0x0736] = 0x0000, /* R1846 - AIF1TX7MIX Input 4 Source */ + [0x0737] = 0x0080, /* R1847 - AIF1TX7MIX Input 4 Volume */ + [0x0738] = 0x0000, /* R1848 - AIF1TX8MIX Input 1 Source */ + [0x0739] = 0x0080, /* R1849 - AIF1TX8MIX Input 1 Volume */ + [0x073A] = 0x0000, /* R1850 - AIF1TX8MIX Input 2 Source */ + [0x073B] = 0x0080, /* R1851 - AIF1TX8MIX Input 2 Volume */ + [0x073C] = 0x0000, /* R1852 - AIF1TX8MIX Input 3 Source */ + [0x073D] = 0x0080, /* R1853 - AIF1TX8MIX Input 3 Volume */ + [0x073E] = 0x0000, /* R1854 - AIF1TX8MIX Input 4 Source */ + [0x073F] = 0x0080, /* R1855 - AIF1TX8MIX Input 4 Volume */ + [0x0740] = 0x0000, /* R1856 - AIF2TX1MIX Input 1 Source */ + [0x0741] = 0x0080, /* R1857 - AIF2TX1MIX Input 1 Volume */ + [0x0742] = 0x0000, /* R1858 - AIF2TX1MIX Input 2 Source */ + [0x0743] = 0x0080, /* R1859 - AIF2TX1MIX Input 2 Volume */ + [0x0744] = 0x0000, /* R1860 - AIF2TX1MIX Input 3 Source */ + [0x0745] = 0x0080, /* R1861 - AIF2TX1MIX Input 3 Volume */ + [0x0746] = 0x0000, /* R1862 - AIF2TX1MIX Input 4 Source */ + [0x0747] = 0x0080, /* R1863 - AIF2TX1MIX Input 4 Volume */ + [0x0748] = 0x0000, /* R1864 - AIF2TX2MIX Input 1 Source */ + [0x0749] = 0x0080, /* R1865 - AIF2TX2MIX Input 1 Volume */ + [0x074A] = 0x0000, /* R1866 - AIF2TX2MIX Input 2 Source */ + [0x074B] = 0x0080, /* R1867 - AIF2TX2MIX Input 2 Volume */ + [0x074C] = 0x0000, /* R1868 - AIF2TX2MIX Input 3 Source */ + [0x074D] = 0x0080, /* R1869 - AIF2TX2MIX Input 3 Volume */ + [0x074E] = 0x0000, /* R1870 - AIF2TX2MIX Input 4 Source */ + [0x074F] = 0x0080, /* R1871 - AIF2TX2MIX Input 4 Volume */ + [0x0780] = 0x0000, /* R1920 - AIF3TX1MIX Input 1 Source */ + [0x0781] = 0x0080, /* R1921 - AIF3TX1MIX Input 1 Volume */ + [0x0782] = 0x0000, /* R1922 - AIF3TX1MIX Input 2 Source */ + [0x0783] = 0x0080, /* R1923 - AIF3TX1MIX Input 2 Volume */ + [0x0784] = 0x0000, /* R1924 - AIF3TX1MIX Input 3 Source */ + [0x0785] = 0x0080, /* R1925 - AIF3TX1MIX Input 3 Volume */ + [0x0786] = 0x0000, /* R1926 - AIF3TX1MIX Input 4 Source */ + [0x0787] = 0x0080, /* R1927 - AIF3TX1MIX Input 4 Volume */ + [0x0788] = 0x0000, /* R1928 - AIF3TX2MIX Input 1 Source */ + [0x0789] = 0x0080, /* R1929 - AIF3TX2MIX Input 1 Volume */ + [0x078A] = 0x0000, /* R1930 - AIF3TX2MIX Input 2 Source */ + [0x078B] = 0x0080, /* R1931 - AIF3TX2MIX Input 2 Volume */ + [0x078C] = 0x0000, /* R1932 - AIF3TX2MIX Input 3 Source */ + [0x078D] = 0x0080, /* R1933 - AIF3TX2MIX Input 3 Volume */ + [0x078E] = 0x0000, /* R1934 - AIF3TX2MIX Input 4 Source */ + [0x078F] = 0x0080, /* R1935 - AIF3TX2MIX Input 4 Volume */ + [0x0880] = 0x0000, /* R2176 - EQ1MIX Input 1 Source */ + [0x0881] = 0x0080, /* R2177 - EQ1MIX Input 1 Volume */ + [0x0882] = 0x0000, /* R2178 - EQ1MIX Input 2 Source */ + [0x0883] = 0x0080, /* R2179 - EQ1MIX Input 2 Volume */ + [0x0884] = 0x0000, /* R2180 - EQ1MIX Input 3 Source */ + [0x0885] = 0x0080, /* R2181 - EQ1MIX Input 3 Volume */ + [0x0886] = 0x0000, /* R2182 - EQ1MIX Input 4 Source */ + [0x0887] = 0x0080, /* R2183 - EQ1MIX Input 4 Volume */ + [0x0888] = 0x0000, /* R2184 - EQ2MIX Input 1 Source */ + [0x0889] = 0x0080, /* R2185 - EQ2MIX Input 1 Volume */ + [0x088A] = 0x0000, /* R2186 - EQ2MIX Input 2 Source */ + [0x088B] = 0x0080, /* R2187 - EQ2MIX Input 2 Volume */ + [0x088C] = 0x0000, /* R2188 - EQ2MIX Input 3 Source */ + [0x088D] = 0x0080, /* R2189 - EQ2MIX Input 3 Volume */ + [0x088E] = 0x0000, /* R2190 - EQ2MIX Input 4 Source */ + [0x088F] = 0x0080, /* R2191 - EQ2MIX Input 4 Volume */ + [0x0890] = 0x0000, /* R2192 - EQ3MIX Input 1 Source */ + [0x0891] = 0x0080, /* R2193 - EQ3MIX Input 1 Volume */ + [0x0892] = 0x0000, /* R2194 - EQ3MIX Input 2 Source */ + [0x0893] = 0x0080, /* R2195 - EQ3MIX Input 2 Volume */ + [0x0894] = 0x0000, /* R2196 - EQ3MIX Input 3 Source */ + [0x0895] = 0x0080, /* R2197 - EQ3MIX Input 3 Volume */ + [0x0896] = 0x0000, /* R2198 - EQ3MIX Input 4 Source */ + [0x0897] = 0x0080, /* R2199 - EQ3MIX Input 4 Volume */ + [0x0898] = 0x0000, /* R2200 - EQ4MIX Input 1 Source */ + [0x0899] = 0x0080, /* R2201 - EQ4MIX Input 1 Volume */ + [0x089A] = 0x0000, /* R2202 - EQ4MIX Input 2 Source */ + [0x089B] = 0x0080, /* R2203 - EQ4MIX Input 2 Volume */ + [0x089C] = 0x0000, /* R2204 - EQ4MIX Input 3 Source */ + [0x089D] = 0x0080, /* R2205 - EQ4MIX Input 3 Volume */ + [0x089E] = 0x0000, /* R2206 - EQ4MIX Input 4 Source */ + [0x089F] = 0x0080, /* R2207 - EQ4MIX Input 4 Volume */ + [0x08C0] = 0x0000, /* R2240 - DRC1LMIX Input 1 Source */ + [0x08C1] = 0x0080, /* R2241 - DRC1LMIX Input 1 Volume */ + [0x08C2] = 0x0000, /* R2242 - DRC1LMIX Input 2 Source */ + [0x08C3] = 0x0080, /* R2243 - DRC1LMIX Input 2 Volume */ + [0x08C4] = 0x0000, /* R2244 - DRC1LMIX Input 3 Source */ + [0x08C5] = 0x0080, /* R2245 - DRC1LMIX Input 3 Volume */ + [0x08C6] = 0x0000, /* R2246 - DRC1LMIX Input 4 Source */ + [0x08C7] = 0x0080, /* R2247 - DRC1LMIX Input 4 Volume */ + [0x08C8] = 0x0000, /* R2248 - DRC1RMIX Input 1 Source */ + [0x08C9] = 0x0080, /* R2249 - DRC1RMIX Input 1 Volume */ + [0x08CA] = 0x0000, /* R2250 - DRC1RMIX Input 2 Source */ + [0x08CB] = 0x0080, /* R2251 - DRC1RMIX Input 2 Volume */ + [0x08CC] = 0x0000, /* R2252 - DRC1RMIX Input 3 Source */ + [0x08CD] = 0x0080, /* R2253 - DRC1RMIX Input 3 Volume */ + [0x08CE] = 0x0000, /* R2254 - DRC1RMIX Input 4 Source */ + [0x08CF] = 0x0080, /* R2255 - DRC1RMIX Input 4 Volume */ + [0x0900] = 0x0000, /* R2304 - HPLP1MIX Input 1 Source */ + [0x0901] = 0x0080, /* R2305 - HPLP1MIX Input 1 Volume */ + [0x0902] = 0x0000, /* R2306 - HPLP1MIX Input 2 Source */ + [0x0903] = 0x0080, /* R2307 - HPLP1MIX Input 2 Volume */ + [0x0904] = 0x0000, /* R2308 - HPLP1MIX Input 3 Source */ + [0x0905] = 0x0080, /* R2309 - HPLP1MIX Input 3 Volume */ + [0x0906] = 0x0000, /* R2310 - HPLP1MIX Input 4 Source */ + [0x0907] = 0x0080, /* R2311 - HPLP1MIX Input 4 Volume */ + [0x0908] = 0x0000, /* R2312 - HPLP2MIX Input 1 Source */ + [0x0909] = 0x0080, /* R2313 - HPLP2MIX Input 1 Volume */ + [0x090A] = 0x0000, /* R2314 - HPLP2MIX Input 2 Source */ + [0x090B] = 0x0080, /* R2315 - HPLP2MIX Input 2 Volume */ + [0x090C] = 0x0000, /* R2316 - HPLP2MIX Input 3 Source */ + [0x090D] = 0x0080, /* R2317 - HPLP2MIX Input 3 Volume */ + [0x090E] = 0x0000, /* R2318 - HPLP2MIX Input 4 Source */ + [0x090F] = 0x0080, /* R2319 - HPLP2MIX Input 4 Volume */ + [0x0910] = 0x0000, /* R2320 - HPLP3MIX Input 1 Source */ + [0x0911] = 0x0080, /* R2321 - HPLP3MIX Input 1 Volume */ + [0x0912] = 0x0000, /* R2322 - HPLP3MIX Input 2 Source */ + [0x0913] = 0x0080, /* R2323 - HPLP3MIX Input 2 Volume */ + [0x0914] = 0x0000, /* R2324 - HPLP3MIX Input 3 Source */ + [0x0915] = 0x0080, /* R2325 - HPLP3MIX Input 3 Volume */ + [0x0916] = 0x0000, /* R2326 - HPLP3MIX Input 4 Source */ + [0x0917] = 0x0080, /* R2327 - HPLP3MIX Input 4 Volume */ + [0x0918] = 0x0000, /* R2328 - HPLP4MIX Input 1 Source */ + [0x0919] = 0x0080, /* R2329 - HPLP4MIX Input 1 Volume */ + [0x091A] = 0x0000, /* R2330 - HPLP4MIX Input 2 Source */ + [0x091B] = 0x0080, /* R2331 - HPLP4MIX Input 2 Volume */ + [0x091C] = 0x0000, /* R2332 - HPLP4MIX Input 3 Source */ + [0x091D] = 0x0080, /* R2333 - HPLP4MIX Input 3 Volume */ + [0x091E] = 0x0000, /* R2334 - HPLP4MIX Input 4 Source */ + [0x091F] = 0x0080, /* R2335 - HPLP4MIX Input 4 Volume */ + [0x0940] = 0x0000, /* R2368 - DSP1LMIX Input 1 Source */ + [0x0941] = 0x0080, /* R2369 - DSP1LMIX Input 1 Volume */ + [0x0942] = 0x0000, /* R2370 - DSP1LMIX Input 2 Source */ + [0x0943] = 0x0080, /* R2371 - DSP1LMIX Input 2 Volume */ + [0x0944] = 0x0000, /* R2372 - DSP1LMIX Input 3 Source */ + [0x0945] = 0x0080, /* R2373 - DSP1LMIX Input 3 Volume */ + [0x0946] = 0x0000, /* R2374 - DSP1LMIX Input 4 Source */ + [0x0947] = 0x0080, /* R2375 - DSP1LMIX Input 4 Volume */ + [0x0948] = 0x0000, /* R2376 - DSP1RMIX Input 1 Source */ + [0x0949] = 0x0080, /* R2377 - DSP1RMIX Input 1 Volume */ + [0x094A] = 0x0000, /* R2378 - DSP1RMIX Input 2 Source */ + [0x094B] = 0x0080, /* R2379 - DSP1RMIX Input 2 Volume */ + [0x094C] = 0x0000, /* R2380 - DSP1RMIX Input 3 Source */ + [0x094D] = 0x0080, /* R2381 - DSP1RMIX Input 3 Volume */ + [0x094E] = 0x0000, /* R2382 - DSP1RMIX Input 4 Source */ + [0x094F] = 0x0080, /* R2383 - DSP1RMIX Input 4 Volume */ + [0x0950] = 0x0000, /* R2384 - DSP1AUX1MIX Input 1 Source */ + [0x0958] = 0x0000, /* R2392 - DSP1AUX2MIX Input 1 Source */ + [0x0960] = 0x0000, /* R2400 - DSP1AUX3MIX Input 1 Source */ + [0x0968] = 0x0000, /* R2408 - DSP1AUX4MIX Input 1 Source */ + [0x0970] = 0x0000, /* R2416 - DSP1AUX5MIX Input 1 Source */ + [0x0978] = 0x0000, /* R2424 - DSP1AUX6MIX Input 1 Source */ + [0x0980] = 0x0000, /* R2432 - DSP2LMIX Input 1 Source */ + [0x0981] = 0x0080, /* R2433 - DSP2LMIX Input 1 Volume */ + [0x0982] = 0x0000, /* R2434 - DSP2LMIX Input 2 Source */ + [0x0983] = 0x0080, /* R2435 - DSP2LMIX Input 2 Volume */ + [0x0984] = 0x0000, /* R2436 - DSP2LMIX Input 3 Source */ + [0x0985] = 0x0080, /* R2437 - DSP2LMIX Input 3 Volume */ + [0x0986] = 0x0000, /* R2438 - DSP2LMIX Input 4 Source */ + [0x0987] = 0x0080, /* R2439 - DSP2LMIX Input 4 Volume */ + [0x0988] = 0x0000, /* R2440 - DSP2RMIX Input 1 Source */ + [0x0989] = 0x0080, /* R2441 - DSP2RMIX Input 1 Volume */ + [0x098A] = 0x0000, /* R2442 - DSP2RMIX Input 2 Source */ + [0x098B] = 0x0080, /* R2443 - DSP2RMIX Input 2 Volume */ + [0x098C] = 0x0000, /* R2444 - DSP2RMIX Input 3 Source */ + [0x098D] = 0x0080, /* R2445 - DSP2RMIX Input 3 Volume */ + [0x098E] = 0x0000, /* R2446 - DSP2RMIX Input 4 Source */ + [0x098F] = 0x0080, /* R2447 - DSP2RMIX Input 4 Volume */ + [0x0990] = 0x0000, /* R2448 - DSP2AUX1MIX Input 1 Source */ + [0x0998] = 0x0000, /* R2456 - DSP2AUX2MIX Input 1 Source */ + [0x09A0] = 0x0000, /* R2464 - DSP2AUX3MIX Input 1 Source */ + [0x09A8] = 0x0000, /* R2472 - DSP2AUX4MIX Input 1 Source */ + [0x09B0] = 0x0000, /* R2480 - DSP2AUX5MIX Input 1 Source */ + [0x09B8] = 0x0000, /* R2488 - DSP2AUX6MIX Input 1 Source */ + [0x09C0] = 0x0000, /* R2496 - DSP3LMIX Input 1 Source */ + [0x09C1] = 0x0080, /* R2497 - DSP3LMIX Input 1 Volume */ + [0x09C2] = 0x0000, /* R2498 - DSP3LMIX Input 2 Source */ + [0x09C3] = 0x0080, /* R2499 - DSP3LMIX Input 2 Volume */ + [0x09C4] = 0x0000, /* R2500 - DSP3LMIX Input 3 Source */ + [0x09C5] = 0x0080, /* R2501 - DSP3LMIX Input 3 Volume */ + [0x09C6] = 0x0000, /* R2502 - DSP3LMIX Input 4 Source */ + [0x09C7] = 0x0080, /* R2503 - DSP3LMIX Input 4 Volume */ + [0x09C8] = 0x0000, /* R2504 - DSP3RMIX Input 1 Source */ + [0x09C9] = 0x0080, /* R2505 - DSP3RMIX Input 1 Volume */ + [0x09CA] = 0x0000, /* R2506 - DSP3RMIX Input 2 Source */ + [0x09CB] = 0x0080, /* R2507 - DSP3RMIX Input 2 Volume */ + [0x09CC] = 0x0000, /* R2508 - DSP3RMIX Input 3 Source */ + [0x09CD] = 0x0080, /* R2509 - DSP3RMIX Input 3 Volume */ + [0x09CE] = 0x0000, /* R2510 - DSP3RMIX Input 4 Source */ + [0x09CF] = 0x0080, /* R2511 - DSP3RMIX Input 4 Volume */ + [0x09D0] = 0x0000, /* R2512 - DSP3AUX1MIX Input 1 Source */ + [0x09D8] = 0x0000, /* R2520 - DSP3AUX2MIX Input 1 Source */ + [0x09E0] = 0x0000, /* R2528 - DSP3AUX3MIX Input 1 Source */ + [0x09E8] = 0x0000, /* R2536 - DSP3AUX4MIX Input 1 Source */ + [0x09F0] = 0x0000, /* R2544 - DSP3AUX5MIX Input 1 Source */ + [0x09F8] = 0x0000, /* R2552 - DSP3AUX6MIX Input 1 Source */ + [0x0A80] = 0x0000, /* R2688 - ASRC1LMIX Input 1 Source */ + [0x0A88] = 0x0000, /* R2696 - ASRC1RMIX Input 1 Source */ + [0x0A90] = 0x0000, /* R2704 - ASRC2LMIX Input 1 Source */ + [0x0A98] = 0x0000, /* R2712 - ASRC2RMIX Input 1 Source */ + [0x0B00] = 0x0000, /* R2816 - ISRC1DEC1MIX Input 1 Source */ + [0x0B08] = 0x0000, /* R2824 - ISRC1DEC2MIX Input 1 Source */ + [0x0B10] = 0x0000, /* R2832 - ISRC1DEC3MIX Input 1 Source */ + [0x0B18] = 0x0000, /* R2840 - ISRC1DEC4MIX Input 1 Source */ + [0x0B20] = 0x0000, /* R2848 - ISRC1INT1MIX Input 1 Source */ + [0x0B28] = 0x0000, /* R2856 - ISRC1INT2MIX Input 1 Source */ + [0x0B30] = 0x0000, /* R2864 - ISRC1INT3MIX Input 1 Source */ + [0x0B38] = 0x0000, /* R2872 - ISRC1INT4MIX Input 1 Source */ + [0x0B40] = 0x0000, /* R2880 - ISRC2DEC1MIX Input 1 Source */ + [0x0B48] = 0x0000, /* R2888 - ISRC2DEC2MIX Input 1 Source */ + [0x0B50] = 0x0000, /* R2896 - ISRC2DEC3MIX Input 1 Source */ + [0x0B58] = 0x0000, /* R2904 - ISRC2DEC4MIX Input 1 Source */ + [0x0B60] = 0x0000, /* R2912 - ISRC2INT1MIX Input 1 Source */ + [0x0B68] = 0x0000, /* R2920 - ISRC2INT2MIX Input 1 Source */ + [0x0B70] = 0x0000, /* R2928 - ISRC2INT3MIX Input 1 Source */ + [0x0B78] = 0x0000, /* R2936 - ISRC2INT4MIX Input 1 Source */ + [0x0C00] = 0xA001, /* R3072 - GPIO CTRL 1 */ + [0x0C01] = 0xA001, /* R3073 - GPIO CTRL 2 */ + [0x0C02] = 0xA001, /* R3074 - GPIO CTRL 3 */ + [0x0C03] = 0xA001, /* R3075 - GPIO CTRL 4 */ + [0x0C04] = 0xA001, /* R3076 - GPIO CTRL 5 */ + [0x0C05] = 0xA001, /* R3077 - GPIO CTRL 6 */ + [0x0C23] = 0x4003, /* R3107 - Misc Pad Ctrl 1 */ + [0x0C24] = 0x0000, /* R3108 - Misc Pad Ctrl 2 */ + [0x0C25] = 0x0000, /* R3109 - Misc Pad Ctrl 3 */ + [0x0C26] = 0x0000, /* R3110 - Misc Pad Ctrl 4 */ + [0x0C27] = 0x0000, /* R3111 - Misc Pad Ctrl 5 */ + [0x0C28] = 0x0000, /* R3112 - Misc GPIO 1 */ + [0x0D00] = 0x0000, /* R3328 - Interrupt Status 1 */ + [0x0D01] = 0x0000, /* R3329 - Interrupt Status 2 */ + [0x0D02] = 0x0000, /* R3330 - Interrupt Status 3 */ + [0x0D03] = 0x0000, /* R3331 - Interrupt Status 4 */ + [0x0D04] = 0x0000, /* R3332 - Interrupt Raw Status 2 */ + [0x0D05] = 0x0000, /* R3333 - Interrupt Raw Status 3 */ + [0x0D06] = 0x0000, /* R3334 - Interrupt Raw Status 4 */ + [0x0D07] = 0xFFFF, /* R3335 - Interrupt Status 1 Mask */ + [0x0D08] = 0xFFFF, /* R3336 - Interrupt Status 2 Mask */ + [0x0D09] = 0xFFFF, /* R3337 - Interrupt Status 3 Mask */ + [0x0D0A] = 0xFFFF, /* R3338 - Interrupt Status 4 Mask */ + [0x0D1F] = 0x0000, /* R3359 - Interrupt Control */ + [0x0D20] = 0xFFFF, /* R3360 - IRQ Debounce 1 */ + [0x0D21] = 0xFFFF, /* R3361 - IRQ Debounce 2 */ + [0x0E00] = 0x0000, /* R3584 - FX_Ctrl */ + [0x0E10] = 0x6318, /* R3600 - EQ1_1 */ + [0x0E11] = 0x6300, /* R3601 - EQ1_2 */ + [0x0E12] = 0x0FC8, /* R3602 - EQ1_3 */ + [0x0E13] = 0x03FE, /* R3603 - EQ1_4 */ + [0x0E14] = 0x00E0, /* R3604 - EQ1_5 */ + [0x0E15] = 0x1EC4, /* R3605 - EQ1_6 */ + [0x0E16] = 0xF136, /* R3606 - EQ1_7 */ + [0x0E17] = 0x0409, /* R3607 - EQ1_8 */ + [0x0E18] = 0x04CC, /* R3608 - EQ1_9 */ + [0x0E19] = 0x1C9B, /* R3609 - EQ1_10 */ + [0x0E1A] = 0xF337, /* R3610 - EQ1_11 */ + [0x0E1B] = 0x040B, /* R3611 - EQ1_12 */ + [0x0E1C] = 0x0CBB, /* R3612 - EQ1_13 */ + [0x0E1D] = 0x16F8, /* R3613 - EQ1_14 */ + [0x0E1E] = 0xF7D9, /* R3614 - EQ1_15 */ + [0x0E1F] = 0x040A, /* R3615 - EQ1_16 */ + [0x0E20] = 0x1F14, /* R3616 - EQ1_17 */ + [0x0E21] = 0x058C, /* R3617 - EQ1_18 */ + [0x0E22] = 0x0563, /* R3618 - EQ1_19 */ + [0x0E23] = 0x4000, /* R3619 - EQ1_20 */ + [0x0E26] = 0x6318, /* R3622 - EQ2_1 */ + [0x0E27] = 0x6300, /* R3623 - EQ2_2 */ + [0x0E28] = 0x0FC8, /* R3624 - EQ2_3 */ + [0x0E29] = 0x03FE, /* R3625 - EQ2_4 */ + [0x0E2A] = 0x00E0, /* R3626 - EQ2_5 */ + [0x0E2B] = 0x1EC4, /* R3627 - EQ2_6 */ + [0x0E2C] = 0xF136, /* R3628 - EQ2_7 */ + [0x0E2D] = 0x0409, /* R3629 - EQ2_8 */ + [0x0E2E] = 0x04CC, /* R3630 - EQ2_9 */ + [0x0E2F] = 0x1C9B, /* R3631 - EQ2_10 */ + [0x0E30] = 0xF337, /* R3632 - EQ2_11 */ + [0x0E31] = 0x040B, /* R3633 - EQ2_12 */ + [0x0E32] = 0x0CBB, /* R3634 - EQ2_13 */ + [0x0E33] = 0x16F8, /* R3635 - EQ2_14 */ + [0x0E34] = 0xF7D9, /* R3636 - EQ2_15 */ + [0x0E35] = 0x040A, /* R3637 - EQ2_16 */ + [0x0E36] = 0x1F14, /* R3638 - EQ2_17 */ + [0x0E37] = 0x058C, /* R3639 - EQ2_18 */ + [0x0E38] = 0x0563, /* R3640 - EQ2_19 */ + [0x0E39] = 0x4000, /* R3641 - EQ2_20 */ + [0x0E3C] = 0x6318, /* R3644 - EQ3_1 */ + [0x0E3D] = 0x6300, /* R3645 - EQ3_2 */ + [0x0E3E] = 0x0FC8, /* R3646 - EQ3_3 */ + [0x0E3F] = 0x03FE, /* R3647 - EQ3_4 */ + [0x0E40] = 0x00E0, /* R3648 - EQ3_5 */ + [0x0E41] = 0x1EC4, /* R3649 - EQ3_6 */ + [0x0E42] = 0xF136, /* R3650 - EQ3_7 */ + [0x0E43] = 0x0409, /* R3651 - EQ3_8 */ + [0x0E44] = 0x04CC, /* R3652 - EQ3_9 */ + [0x0E45] = 0x1C9B, /* R3653 - EQ3_10 */ + [0x0E46] = 0xF337, /* R3654 - EQ3_11 */ + [0x0E47] = 0x040B, /* R3655 - EQ3_12 */ + [0x0E48] = 0x0CBB, /* R3656 - EQ3_13 */ + [0x0E49] = 0x16F8, /* R3657 - EQ3_14 */ + [0x0E4A] = 0xF7D9, /* R3658 - EQ3_15 */ + [0x0E4B] = 0x040A, /* R3659 - EQ3_16 */ + [0x0E4C] = 0x1F14, /* R3660 - EQ3_17 */ + [0x0E4D] = 0x058C, /* R3661 - EQ3_18 */ + [0x0E4E] = 0x0563, /* R3662 - EQ3_19 */ + [0x0E4F] = 0x4000, /* R3663 - EQ3_20 */ + [0x0E52] = 0x6318, /* R3666 - EQ4_1 */ + [0x0E53] = 0x6300, /* R3667 - EQ4_2 */ + [0x0E54] = 0x0FC8, /* R3668 - EQ4_3 */ + [0x0E55] = 0x03FE, /* R3669 - EQ4_4 */ + [0x0E56] = 0x00E0, /* R3670 - EQ4_5 */ + [0x0E57] = 0x1EC4, /* R3671 - EQ4_6 */ + [0x0E58] = 0xF136, /* R3672 - EQ4_7 */ + [0x0E59] = 0x0409, /* R3673 - EQ4_8 */ + [0x0E5A] = 0x04CC, /* R3674 - EQ4_9 */ + [0x0E5B] = 0x1C9B, /* R3675 - EQ4_10 */ + [0x0E5C] = 0xF337, /* R3676 - EQ4_11 */ + [0x0E5D] = 0x040B, /* R3677 - EQ4_12 */ + [0x0E5E] = 0x0CBB, /* R3678 - EQ4_13 */ + [0x0E5F] = 0x16F8, /* R3679 - EQ4_14 */ + [0x0E60] = 0xF7D9, /* R3680 - EQ4_15 */ + [0x0E61] = 0x040A, /* R3681 - EQ4_16 */ + [0x0E62] = 0x1F14, /* R3682 - EQ4_17 */ + [0x0E63] = 0x058C, /* R3683 - EQ4_18 */ + [0x0E64] = 0x0563, /* R3684 - EQ4_19 */ + [0x0E65] = 0x4000, /* R3685 - EQ4_20 */ + [0x0E80] = 0x0018, /* R3712 - DRC1 ctrl1 */ + [0x0E81] = 0x0933, /* R3713 - DRC1 ctrl2 */ + [0x0E82] = 0x0018, /* R3714 - DRC1 ctrl3 */ + [0x0E83] = 0x0000, /* R3715 - DRC1 ctrl4 */ + [0x0E84] = 0x0000, /* R3716 - DRC1 ctrl5 */ + [0x0EC0] = 0x0000, /* R3776 - HPLPF1_1 */ + [0x0EC1] = 0x0000, /* R3777 - HPLPF1_2 */ + [0x0EC4] = 0x0000, /* R3780 - HPLPF2_1 */ + [0x0EC5] = 0x0000, /* R3781 - HPLPF2_2 */ + [0x0EC8] = 0x0000, /* R3784 - HPLPF3_1 */ + [0x0EC9] = 0x0000, /* R3785 - HPLPF3_2 */ + [0x0ECC] = 0x0000, /* R3788 - HPLPF4_1 */ + [0x0ECD] = 0x0000, /* R3789 - HPLPF4_2 */ + [0x4000] = 0x0000, /* R16384 - DSP1 DM 0 */ + [0x4001] = 0x0000, /* R16385 - DSP1 DM 1 */ + [0x4002] = 0x0000, /* R16386 - DSP1 DM 2 */ + [0x4003] = 0x0000, /* R16387 - DSP1 DM 3 */ + [0x41FC] = 0x0000, /* R16892 - DSP1 DM 508 */ + [0x41FD] = 0x0000, /* R16893 - DSP1 DM 509 */ + [0x41FE] = 0x0000, /* R16894 - DSP1 DM 510 */ + [0x41FF] = 0x0000, /* R16895 - DSP1 DM 511 */ + [0x4800] = 0x0000, /* R18432 - DSP1 PM 0 */ + [0x4801] = 0x0000, /* R18433 - DSP1 PM 1 */ + [0x4802] = 0x0000, /* R18434 - DSP1 PM 2 */ + [0x4803] = 0x0000, /* R18435 - DSP1 PM 3 */ + [0x4804] = 0x0000, /* R18436 - DSP1 PM 4 */ + [0x4805] = 0x0000, /* R18437 - DSP1 PM 5 */ + [0x4DFA] = 0x0000, /* R19962 - DSP1 PM 1530 */ + [0x4DFB] = 0x0000, /* R19963 - DSP1 PM 1531 */ + [0x4DFC] = 0x0000, /* R19964 - DSP1 PM 1532 */ + [0x4DFD] = 0x0000, /* R19965 - DSP1 PM 1533 */ + [0x4DFE] = 0x0000, /* R19966 - DSP1 PM 1534 */ + [0x4DFF] = 0x0000, /* R19967 - DSP1 PM 1535 */ + [0x5000] = 0x0000, /* R20480 - DSP1 ZM 0 */ + [0x5001] = 0x0000, /* R20481 - DSP1 ZM 1 */ + [0x5002] = 0x0000, /* R20482 - DSP1 ZM 2 */ + [0x5003] = 0x0000, /* R20483 - DSP1 ZM 3 */ + [0x57FC] = 0x0000, /* R22524 - DSP1 ZM 2044 */ + [0x57FD] = 0x0000, /* R22525 - DSP1 ZM 2045 */ + [0x57FE] = 0x0000, /* R22526 - DSP1 ZM 2046 */ + [0x57FF] = 0x0000, /* R22527 - DSP1 ZM 2047 */ + [0x6000] = 0x0000, /* R24576 - DSP2 DM 0 */ + [0x6001] = 0x0000, /* R24577 - DSP2 DM 1 */ + [0x6002] = 0x0000, /* R24578 - DSP2 DM 2 */ + [0x6003] = 0x0000, /* R24579 - DSP2 DM 3 */ + [0x61FC] = 0x0000, /* R25084 - DSP2 DM 508 */ + [0x61FD] = 0x0000, /* R25085 - DSP2 DM 509 */ + [0x61FE] = 0x0000, /* R25086 - DSP2 DM 510 */ + [0x61FF] = 0x0000, /* R25087 - DSP2 DM 511 */ + [0x6800] = 0x0000, /* R26624 - DSP2 PM 0 */ + [0x6801] = 0x0000, /* R26625 - DSP2 PM 1 */ + [0x6802] = 0x0000, /* R26626 - DSP2 PM 2 */ + [0x6803] = 0x0000, /* R26627 - DSP2 PM 3 */ + [0x6804] = 0x0000, /* R26628 - DSP2 PM 4 */ + [0x6805] = 0x0000, /* R26629 - DSP2 PM 5 */ + [0x6DFA] = 0x0000, /* R28154 - DSP2 PM 1530 */ + [0x6DFB] = 0x0000, /* R28155 - DSP2 PM 1531 */ + [0x6DFC] = 0x0000, /* R28156 - DSP2 PM 1532 */ + [0x6DFD] = 0x0000, /* R28157 - DSP2 PM 1533 */ + [0x6DFE] = 0x0000, /* R28158 - DSP2 PM 1534 */ + [0x6DFF] = 0x0000, /* R28159 - DSP2 PM 1535 */ + [0x7000] = 0x0000, /* R28672 - DSP2 ZM 0 */ + [0x7001] = 0x0000, /* R28673 - DSP2 ZM 1 */ + [0x7002] = 0x0000, /* R28674 - DSP2 ZM 2 */ + [0x7003] = 0x0000, /* R28675 - DSP2 ZM 3 */ + [0x77FC] = 0x0000, /* R30716 - DSP2 ZM 2044 */ + [0x77FD] = 0x0000, /* R30717 - DSP2 ZM 2045 */ + [0x77FE] = 0x0000, /* R30718 - DSP2 ZM 2046 */ + [0x77FF] = 0x0000, /* R30719 - DSP2 ZM 2047 */ + [0x8000] = 0x0000, /* R32768 - DSP3 DM 0 */ + [0x8001] = 0x0000, /* R32769 - DSP3 DM 1 */ + [0x8002] = 0x0000, /* R32770 - DSP3 DM 2 */ + [0x8003] = 0x0000, /* R32771 - DSP3 DM 3 */ + [0x81FC] = 0x0000, /* R33276 - DSP3 DM 508 */ + [0x81FD] = 0x0000, /* R33277 - DSP3 DM 509 */ + [0x81FE] = 0x0000, /* R33278 - DSP3 DM 510 */ + [0x81FF] = 0x0000, /* R33279 - DSP3 DM 511 */ + [0x8800] = 0x0000, /* R34816 - DSP3 PM 0 */ + [0x8801] = 0x0000, /* R34817 - DSP3 PM 1 */ + [0x8802] = 0x0000, /* R34818 - DSP3 PM 2 */ + [0x8803] = 0x0000, /* R34819 - DSP3 PM 3 */ + [0x8804] = 0x0000, /* R34820 - DSP3 PM 4 */ + [0x8805] = 0x0000, /* R34821 - DSP3 PM 5 */ + [0x8DFA] = 0x0000, /* R36346 - DSP3 PM 1530 */ + [0x8DFB] = 0x0000, /* R36347 - DSP3 PM 1531 */ + [0x8DFC] = 0x0000, /* R36348 - DSP3 PM 1532 */ + [0x8DFD] = 0x0000, /* R36349 - DSP3 PM 1533 */ + [0x8DFE] = 0x0000, /* R36350 - DSP3 PM 1534 */ + [0x8DFF] = 0x0000, /* R36351 - DSP3 PM 1535 */ + [0x9000] = 0x0000, /* R36864 - DSP3 ZM 0 */ + [0x9001] = 0x0000, /* R36865 - DSP3 ZM 1 */ + [0x9002] = 0x0000, /* R36866 - DSP3 ZM 2 */ + [0x9003] = 0x0000, /* R36867 - DSP3 ZM 3 */ + [0x97FC] = 0x0000, /* R38908 - DSP3 ZM 2044 */ + [0x97FD] = 0x0000, /* R38909 - DSP3 ZM 2045 */ + [0x97FE] = 0x0000, /* R38910 - DSP3 ZM 2046 */ + [0x97FF] = 0x0000 /* R38911 - DSP3 ZM 2047 */ +}; diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c new file mode 100644 index 0000000..576081a --- /dev/null +++ b/sound/soc/codecs/wm5100.c @@ -0,0 +1,2560 @@ +/* + * wm5100.c -- WM5100 ALSA SoC Audio driver + * + * Copyright 2011 Wolfson Microelectronics plc + * + * Author: Mark Brown + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "wm5100.h" + +#define WM5100_NUM_CORE_SUPPLIES 2 +static const char *wm5100_core_supply_names[WM5100_NUM_CORE_SUPPLIES] = { + "DBVDD1", + "LDOVDD", /* If DCVDD is supplied externally specify as LDOVDD */ +}; + +#define WM5100_AIFS 3 +#define WM5100_SYNC_SRS 3 + +struct wm5100_fll { + int fref; + int fout; + int src; + struct completion lock; +}; + +/* codec private data */ +struct wm5100_priv { + struct snd_soc_codec *codec; + + struct regulator_bulk_data core_supplies[WM5100_NUM_CORE_SUPPLIES]; + struct regulator *cpvdd; + + int rev; + + int sysclk; + int asyncclk; + + bool aif_async[WM5100_AIFS]; + bool aif_symmetric[WM5100_AIFS]; + int sr_ref[WM5100_SYNC_SRS]; + + bool out_ena[2]; + + struct wm5100_fll fll[2]; + + struct wm5100_pdata pdata; + +#ifdef CONFIG_GPIOLIB + struct gpio_chip gpio_chip; +#endif +}; + +static int wm5100_sr_code[] = { + 0, + 12000, + 24000, + 48000, + 96000, + 192000, + 384000, + 768000, + 0, + 11025, + 22050, + 44100, + 88200, + 176400, + 352800, + 705600, + 4000, + 8000, + 16000, + 32000, + 64000, + 128000, + 256000, + 512000, +}; + +static int wm5100_sr_regs[WM5100_SYNC_SRS] = { + WM5100_CLOCKING_4, + WM5100_CLOCKING_5, + WM5100_CLOCKING_6, +}; + +static int wm5100_alloc_sr(struct snd_soc_codec *codec, int rate) +{ + struct wm5100_priv *wm5100 = snd_soc_codec_get_drvdata(codec); + int sr_code, sr_free, i; + + for (i = 0; i < ARRAY_SIZE(wm5100_sr_code); i++) + if (wm5100_sr_code[i] == rate) + break; + if (i == ARRAY_SIZE(wm5100_sr_code)) { + dev_err(codec->dev, "Unsupported sample rate: %dHz\n", rate); + return -EINVAL; + } + sr_code = i; + + if ((wm5100->sysclk % rate) == 0) { + /* Is this rate already in use? */ + sr_free = -1; + for (i = 0; i < ARRAY_SIZE(wm5100_sr_regs); i++) { + if (!wm5100->sr_ref[i] && sr_free == -1) { + sr_free = i; + continue; + } + if ((snd_soc_read(codec, wm5100_sr_regs[i]) & + WM5100_SAMPLE_RATE_1_MASK) == sr_code) + break; + } + + if (i < ARRAY_SIZE(wm5100_sr_regs)) { + wm5100->sr_ref[i]++; + dev_dbg(codec->dev, "SR %dHz, slot %d, ref %d\n", + rate, i, wm5100->sr_ref[i]); + return i; + } + + if (sr_free == -1) { + dev_err(codec->dev, "All SR slots already in use\n"); + return -EBUSY; + } + + dev_dbg(codec->dev, "Allocating SR slot %d for %dHz\n", + sr_free, rate); + wm5100->sr_ref[sr_free]++; + snd_soc_update_bits(codec, wm5100_sr_regs[sr_free], + WM5100_SAMPLE_RATE_1_MASK, + sr_code); + + return sr_free; + + } else { + dev_err(codec->dev, + "SR %dHz incompatible with %dHz SYSCLK and %dHz ASYNCCLK\n", + rate, wm5100->sysclk, wm5100->asyncclk); + return -EINVAL; + } +} + +static void wm5100_free_sr(struct snd_soc_codec *codec, int rate) +{ + struct wm5100_priv *wm5100 = snd_soc_codec_get_drvdata(codec); + int i, sr_code; + + for (i = 0; i < ARRAY_SIZE(wm5100_sr_code); i++) + if (wm5100_sr_code[i] == rate) + break; + if (i == ARRAY_SIZE(wm5100_sr_code)) { + dev_err(codec->dev, "Unsupported sample rate: %dHz\n", rate); + return; + } + sr_code = wm5100_sr_code[i]; + + for (i = 0; i < ARRAY_SIZE(wm5100_sr_regs); i++) { + if (!wm5100->sr_ref[i]) + continue; + + if ((snd_soc_read(codec, wm5100_sr_regs[i]) & + WM5100_SAMPLE_RATE_1_MASK) == sr_code) + break; + } + if (i < ARRAY_SIZE(wm5100_sr_regs)) { + wm5100->sr_ref[i]--; + dev_dbg(codec->dev, "Dereference SR %dHz, count now %d\n", + rate, wm5100->sr_ref[i]); + } else { + dev_warn(codec->dev, "Freeing unreferenced sample rate %dHz\n", + rate); + } +} + +static int wm5100_reset(struct snd_soc_codec *codec) +{ + struct wm5100_priv *wm5100 = snd_soc_codec_get_drvdata(codec); + + if (wm5100->pdata.reset) { + gpio_set_value_cansleep(wm5100->pdata.reset, 0); + gpio_set_value_cansleep(wm5100->pdata.reset, 1); + + return 0; + } else { + return snd_soc_write(codec, WM5100_SOFTWARE_RESET, 0); + } +} + +static DECLARE_TLV_DB_SCALE(in_tlv, -6300, 100, 0); +static DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0); +static DECLARE_TLV_DB_SCALE(mixer_tlv, -3200, 100, 0); +static DECLARE_TLV_DB_SCALE(out_tlv, -6400, 100, 0); +static DECLARE_TLV_DB_SCALE(digital_tlv, -6400, 50, 0); + +static const char *wm5100_mixer_texts[] = { + "None", + "Tone Generator 1", + "Tone Generator 2", + "AEC loopback", + "IN1L", + "IN1R", + "IN2L", + "IN2R", + "IN3L", + "IN3R", + "IN4L", + "IN4R", + "AIF1RX1", + "AIF1RX2", + "AIF1RX3", + "AIF1RX4", + "AIF1RX5", + "AIF1RX6", + "AIF1RX7", + "AIF1RX8", + "AIF2RX1", + "AIF2RX2", + "AIF3RX1", + "AIF3RX2", + "EQ1", + "EQ2", + "EQ3", + "EQ4", + "DRC1L", + "DRC1R", + "LHPF1", + "LHPF2", + "LHPF3", + "LHPF4", + "DSP1.1", + "DSP1.2", + "DSP1.3", + "DSP1.4", + "DSP1.5", + "DSP1.6", + "DSP2.1", + "DSP2.2", + "DSP2.3", + "DSP2.4", + "DSP2.5", + "DSP2.6", + "DSP3.1", + "DSP3.2", + "DSP3.3", + "DSP3.4", + "DSP3.5", + "DSP3.6", + "ASRC1L", + "ASRC1R", + "ASRC2L", + "ASRC2R", + "ISRC1INT1", + "ISRC1INT2", + "ISRC1INT3", + "ISRC1INT4", + "ISRC2INT1", + "ISRC2INT2", + "ISRC2INT3", + "ISRC2INT4", + "ISRC1DEC1", + "ISRC1DEC2", + "ISRC1DEC3", + "ISRC1DEC4", + "ISRC2DEC1", + "ISRC2DEC2", + "ISRC2DEC3", + "ISRC2DEC4", +}; + +static int wm5100_mixer_values[] = { + 0x00, + 0x04, /* Tone */ + 0x05, + 0x08, /* AEC */ + 0x10, /* Input */ + 0x11, + 0x12, + 0x13, + 0x14, + 0x15, + 0x16, + 0x17, + 0x20, /* AIF */ + 0x21, + 0x22, + 0x23, + 0x24, + 0x25, + 0x26, + 0x27, + 0x28, + 0x29, + 0x30, /* AIF3 - check */ + 0x31, + 0x50, /* EQ */ + 0x51, + 0x52, + 0x53, + 0x54, + 0x58, /* DRC */ + 0x59, + 0x60, /* LHPF1 */ + 0x61, /* LHPF2 */ + 0x62, /* LHPF3 */ + 0x63, /* LHPF4 */ + 0x68, /* DSP1 */ + 0x69, + 0x6a, + 0x6b, + 0x6c, + 0x6d, + 0x70, /* DSP2 */ + 0x71, + 0x72, + 0x73, + 0x74, + 0x75, + 0x78, /* DSP3 */ + 0x79, + 0x7a, + 0x7b, + 0x7c, + 0x7d, + 0x90, /* ASRC1 */ + 0x91, + 0x92, /* ASRC2 */ + 0x93, + 0xa0, /* ISRC1DEC1 */ + 0xa1, + 0xa2, + 0xa3, + 0xa4, /* ISRC1INT1 */ + 0xa5, + 0xa6, + 0xa7, + 0xa8, /* ISRC2DEC1 */ + 0xa9, + 0xaa, + 0xab, + 0xac, /* ISRC2INT1 */ + 0xad, + 0xae, + 0xaf, +}; + +#define WM5100_MIXER_CONTROLS(name, base) \ + SOC_SINGLE_TLV(name " Input 1 Volume", base + 1 , \ + WM5100_MIXER_VOL_SHIFT, 80, 0, mixer_tlv), \ + SOC_SINGLE_TLV(name " Input 2 Volume", base + 3 , \ + WM5100_MIXER_VOL_SHIFT, 80, 0, mixer_tlv), \ + SOC_SINGLE_TLV(name " Input 3 Volume", base + 5 , \ + WM5100_MIXER_VOL_SHIFT, 80, 0, mixer_tlv), \ + SOC_SINGLE_TLV(name " Input 4 Volume", base + 7 , \ + WM5100_MIXER_VOL_SHIFT, 80, 0, mixer_tlv) + +#define WM5100_MUX_ENUM_DECL(name, reg) \ + SOC_VALUE_ENUM_SINGLE_DECL(name, reg, 0, 0xff, \ + wm5100_mixer_texts, wm5100_mixer_values) + +#define WM5100_MUX_CTL_DECL(name) \ + const struct snd_kcontrol_new name##_mux = \ + SOC_DAPM_VALUE_ENUM("Route", name##_enum) + +#define WM5100_MIXER_ENUMS(name, base_reg) \ + static WM5100_MUX_ENUM_DECL(name##_in1_enum, base_reg); \ + static WM5100_MUX_ENUM_DECL(name##_in2_enum, base_reg + 2); \ + static WM5100_MUX_ENUM_DECL(name##_in3_enum, base_reg + 4); \ + static WM5100_MUX_ENUM_DECL(name##_in4_enum, base_reg + 6); \ + static WM5100_MUX_CTL_DECL(name##_in1); \ + static WM5100_MUX_CTL_DECL(name##_in2); \ + static WM5100_MUX_CTL_DECL(name##_in3); \ + static WM5100_MUX_CTL_DECL(name##_in4) + +WM5100_MIXER_ENUMS(HPOUT1L, WM5100_OUT1LMIX_INPUT_1_SOURCE); +WM5100_MIXER_ENUMS(HPOUT1R, WM5100_OUT1RMIX_INPUT_1_SOURCE); +WM5100_MIXER_ENUMS(HPOUT2L, WM5100_OUT2LMIX_INPUT_1_SOURCE); +WM5100_MIXER_ENUMS(HPOUT2R, WM5100_OUT2RMIX_INPUT_1_SOURCE); +WM5100_MIXER_ENUMS(HPOUT3L, WM5100_OUT3LMIX_INPUT_1_SOURCE); +WM5100_MIXER_ENUMS(HPOUT3R, WM5100_OUT3RMIX_INPUT_1_SOURCE); + +WM5100_MIXER_ENUMS(SPKOUTL, WM5100_OUT4LMIX_INPUT_1_SOURCE); +WM5100_MIXER_ENUMS(SPKOUTR, WM5100_OUT4RMIX_INPUT_1_SOURCE); +WM5100_MIXER_ENUMS(SPKDAT1L, WM5100_OUT5LMIX_INPUT_1_SOURCE); +WM5100_MIXER_ENUMS(SPKDAT1R, WM5100_OUT5RMIX_INPUT_1_SOURCE); +WM5100_MIXER_ENUMS(SPKDAT2L, WM5100_OUT6LMIX_INPUT_1_SOURCE); +WM5100_MIXER_ENUMS(SPKDAT2R, WM5100_OUT6RMIX_INPUT_1_SOURCE); + +WM5100_MIXER_ENUMS(PWM1, WM5100_PWM1MIX_INPUT_1_SOURCE); +WM5100_MIXER_ENUMS(PWM2, WM5100_PWM1MIX_INPUT_1_SOURCE); + +WM5100_MIXER_ENUMS(AIF1TX1, WM5100_AIF1TX1MIX_INPUT_1_SOURCE); +WM5100_MIXER_ENUMS(AIF1TX2, WM5100_AIF1TX2MIX_INPUT_1_SOURCE); +WM5100_MIXER_ENUMS(AIF1TX3, WM5100_AIF1TX3MIX_INPUT_1_SOURCE); +WM5100_MIXER_ENUMS(AIF1TX4, WM5100_AIF1TX4MIX_INPUT_1_SOURCE); +WM5100_MIXER_ENUMS(AIF1TX5, WM5100_AIF1TX5MIX_INPUT_1_SOURCE); +WM5100_MIXER_ENUMS(AIF1TX6, WM5100_AIF1TX6MIX_INPUT_1_SOURCE); +WM5100_MIXER_ENUMS(AIF1TX7, WM5100_AIF1TX7MIX_INPUT_1_SOURCE); +WM5100_MIXER_ENUMS(AIF1TX8, WM5100_AIF1TX8MIX_INPUT_1_SOURCE); + +WM5100_MIXER_ENUMS(AIF2TX1, WM5100_AIF2TX1MIX_INPUT_1_SOURCE); +WM5100_MIXER_ENUMS(AIF2TX2, WM5100_AIF2TX2MIX_INPUT_1_SOURCE); + +WM5100_MIXER_ENUMS(AIF3TX1, WM5100_AIF1TX1MIX_INPUT_1_SOURCE); +WM5100_MIXER_ENUMS(AIF3TX2, WM5100_AIF1TX2MIX_INPUT_1_SOURCE); + +WM5100_MIXER_ENUMS(EQ1, WM5100_EQ1MIX_INPUT_1_SOURCE); +WM5100_MIXER_ENUMS(EQ2, WM5100_EQ2MIX_INPUT_1_SOURCE); +WM5100_MIXER_ENUMS(EQ3, WM5100_EQ3MIX_INPUT_1_SOURCE); +WM5100_MIXER_ENUMS(EQ4, WM5100_EQ4MIX_INPUT_1_SOURCE); + +WM5100_MIXER_ENUMS(DRC1L, WM5100_DRC1LMIX_INPUT_1_SOURCE); +WM5100_MIXER_ENUMS(DRC1R, WM5100_DRC1RMIX_INPUT_1_SOURCE); + +WM5100_MIXER_ENUMS(LHPF1, WM5100_HPLP1MIX_INPUT_1_SOURCE); +WM5100_MIXER_ENUMS(LHPF2, WM5100_HPLP2MIX_INPUT_1_SOURCE); +WM5100_MIXER_ENUMS(LHPF3, WM5100_HPLP3MIX_INPUT_1_SOURCE); +WM5100_MIXER_ENUMS(LHPF4, WM5100_HPLP4MIX_INPUT_1_SOURCE); + +#define WM5100_MUX(name, ctrl) \ + SND_SOC_DAPM_VALUE_MUX(name, SND_SOC_NOPM, 0, 0, ctrl) + +#define WM5100_MIXER_WIDGETS(name, name_str) \ + WM5100_MUX(name_str " Input 1", &name##_in1_mux), \ + WM5100_MUX(name_str " Input 2", &name##_in2_mux), \ + WM5100_MUX(name_str " Input 3", &name##_in3_mux), \ + WM5100_MUX(name_str " Input 4", &name##_in4_mux), \ + SND_SOC_DAPM_MIXER(name_str " Mixer", SND_SOC_NOPM, 0, 0, NULL, 0) + +#define WM5100_MIXER_INPUT_ROUTES(name) \ + { name, "Tone Generator 1", "Tone Generator 1" }, \ + { name, "Tone Generator 2", "Tone Generator 2" }, \ + { name, "IN1L", "IN1L PGA" }, \ + { name, "IN1R", "IN1R PGA" }, \ + { name, "IN2L", "IN2L PGA" }, \ + { name, "IN2R", "IN2R PGA" }, \ + { name, "IN3L", "IN3L PGA" }, \ + { name, "IN3R", "IN3R PGA" }, \ + { name, "IN4L", "IN4L PGA" }, \ + { name, "IN4R", "IN4R PGA" }, \ + { name, "AIF1RX1", "AIF1RX1" }, \ + { name, "AIF1RX2", "AIF1RX2" }, \ + { name, "AIF1RX3", "AIF1RX3" }, \ + { name, "AIF1RX4", "AIF1RX4" }, \ + { name, "AIF1RX5", "AIF1RX5" }, \ + { name, "AIF1RX6", "AIF1RX6" }, \ + { name, "AIF1RX7", "AIF1RX7" }, \ + { name, "AIF1RX8", "AIF1RX8" }, \ + { name, "AIF2RX1", "AIF2RX1" }, \ + { name, "AIF2RX2", "AIF2RX2" }, \ + { name, "AIF3RX1", "AIF3RX1" }, \ + { name, "AIF3RX2", "AIF3RX2" }, \ + { name, "EQ1", "EQ1" }, \ + { name, "EQ2", "EQ2" }, \ + { name, "EQ3", "EQ3" }, \ + { name, "EQ4", "EQ4" }, \ + { name, "DRC1L", "DRC1L" }, \ + { name, "DRC1R", "DRC1R" }, \ + { name, "LHPF1", "LHPF1" }, \ + { name, "LHPF2", "LHPF2" }, \ + { name, "LHPF3", "LHPF3" }, \ + { name, "LHPF4", "LHPF4" } + +#define WM5100_MIXER_ROUTES(widget, name) \ + { widget, NULL, name " Mixer" }, \ + { name " Mixer", NULL, name " Input 1" }, \ + { name " Mixer", NULL, name " Input 2" }, \ + { name " Mixer", NULL, name " Input 3" }, \ + { name " Mixer", NULL, name " Input 4" }, \ + WM5100_MIXER_INPUT_ROUTES(name " Input 1"), \ + WM5100_MIXER_INPUT_ROUTES(name " Input 2"), \ + WM5100_MIXER_INPUT_ROUTES(name " Input 3"), \ + WM5100_MIXER_INPUT_ROUTES(name " Input 4") + +static const char *wm5100_lhpf_mode_text[] = { + "Low-pass", "High-pass" +}; + +static const struct soc_enum wm5100_lhpf1_mode = + SOC_ENUM_SINGLE(WM5100_HPLPF1_1, WM5100_LHPF1_MODE_SHIFT, 2, + wm5100_lhpf_mode_text); + +static const struct soc_enum wm5100_lhpf2_mode = + SOC_ENUM_SINGLE(WM5100_HPLPF2_1, WM5100_LHPF2_MODE_SHIFT, 2, + wm5100_lhpf_mode_text); + +static const struct soc_enum wm5100_lhpf3_mode = + SOC_ENUM_SINGLE(WM5100_HPLPF3_1, WM5100_LHPF3_MODE_SHIFT, 2, + wm5100_lhpf_mode_text); + +static const struct soc_enum wm5100_lhpf4_mode = + SOC_ENUM_SINGLE(WM5100_HPLPF4_1, WM5100_LHPF4_MODE_SHIFT, 2, + wm5100_lhpf_mode_text); + +static const struct snd_kcontrol_new wm5100_snd_controls[] = { +SOC_SINGLE("IN1 High Performance Switch", WM5100_IN1L_CONTROL, + WM5100_IN1_OSR_SHIFT, 1, 0), +SOC_SINGLE("IN2 High Performance Switch", WM5100_IN2L_CONTROL, + WM5100_IN2_OSR_SHIFT, 1, 0), +SOC_SINGLE("IN3 High Performance Switch", WM5100_IN3L_CONTROL, + WM5100_IN3_OSR_SHIFT, 1, 0), +SOC_SINGLE("IN4 High Performance Switch", WM5100_IN4L_CONTROL, + WM5100_IN4_OSR_SHIFT, 1, 0), + +/* Only applicable for analogue inputs */ +SOC_DOUBLE_R_TLV("IN1 Volume", WM5100_IN1L_CONTROL, WM5100_IN1R_CONTROL, + WM5100_IN1L_PGA_VOL_SHIFT, 94, 0, in_tlv), +SOC_DOUBLE_R_TLV("IN2 Volume", WM5100_IN2L_CONTROL, WM5100_IN2R_CONTROL, + WM5100_IN2L_PGA_VOL_SHIFT, 94, 0, in_tlv), +SOC_DOUBLE_R_TLV("IN3 Volume", WM5100_IN3L_CONTROL, WM5100_IN3R_CONTROL, + WM5100_IN3L_PGA_VOL_SHIFT, 94, 0, in_tlv), +SOC_DOUBLE_R_TLV("IN4 Volume", WM5100_IN4L_CONTROL, WM5100_IN4R_CONTROL, + WM5100_IN4L_PGA_VOL_SHIFT, 94, 0, in_tlv), + +SOC_DOUBLE_R_TLV("IN1 Digital Volume", WM5100_ADC_DIGITAL_VOLUME_1L, + WM5100_ADC_DIGITAL_VOLUME_1R, WM5100_IN1L_VOL_SHIFT, 191, + 0, digital_tlv), +SOC_DOUBLE_R_TLV("IN2 Digital Volume", WM5100_ADC_DIGITAL_VOLUME_2L, + WM5100_ADC_DIGITAL_VOLUME_2R, WM5100_IN2L_VOL_SHIFT, 191, + 0, digital_tlv), +SOC_DOUBLE_R_TLV("IN3 Digital Volume", WM5100_ADC_DIGITAL_VOLUME_3L, + WM5100_ADC_DIGITAL_VOLUME_3R, WM5100_IN3L_VOL_SHIFT, 191, + 0, digital_tlv), +SOC_DOUBLE_R_TLV("IN4 Digital Volume", WM5100_ADC_DIGITAL_VOLUME_4L, + WM5100_ADC_DIGITAL_VOLUME_4R, WM5100_IN4L_VOL_SHIFT, 191, + 0, digital_tlv), + +SOC_DOUBLE_R("IN1 Switch", WM5100_ADC_DIGITAL_VOLUME_1L, + WM5100_ADC_DIGITAL_VOLUME_1R, WM5100_IN1L_MUTE_SHIFT, 1, 1), +SOC_DOUBLE_R("IN2 Switch", WM5100_ADC_DIGITAL_VOLUME_2L, + WM5100_ADC_DIGITAL_VOLUME_2R, WM5100_IN2L_MUTE_SHIFT, 1, 1), +SOC_DOUBLE_R("IN3 Switch", WM5100_ADC_DIGITAL_VOLUME_3L, + WM5100_ADC_DIGITAL_VOLUME_3R, WM5100_IN3L_MUTE_SHIFT, 1, 1), +SOC_DOUBLE_R("IN4 Switch", WM5100_ADC_DIGITAL_VOLUME_4L, + WM5100_ADC_DIGITAL_VOLUME_4R, WM5100_IN4L_MUTE_SHIFT, 1, 1), + +SOC_SINGLE("HPOUT1 High Performance Switch", WM5100_OUT_VOLUME_1L, + WM5100_OUT1_OSR_SHIFT, 1, 0), +SOC_SINGLE("HPOUT2 High Performance Switch", WM5100_OUT_VOLUME_2L, + WM5100_OUT2_OSR_SHIFT, 1, 0), +SOC_SINGLE("HPOUT3 High Performance Switch", WM5100_OUT_VOLUME_3L, + WM5100_OUT3_OSR_SHIFT, 1, 0), +SOC_SINGLE("SPKOUT High Performance Switch", WM5100_OUT_VOLUME_4L, + WM5100_OUT4_OSR_SHIFT, 1, 0), +SOC_SINGLE("SPKDAT1 High Performance Switch", WM5100_DAC_VOLUME_LIMIT_5L, + WM5100_OUT5_OSR_SHIFT, 1, 0), +SOC_SINGLE("SPKDAT2 High Performance Switch", WM5100_DAC_VOLUME_LIMIT_6L, + WM5100_OUT6_OSR_SHIFT, 1, 0), + +SOC_DOUBLE_R_TLV("HPOUT1 Digital Volume", WM5100_DAC_DIGITAL_VOLUME_1L, + WM5100_DAC_DIGITAL_VOLUME_1R, WM5100_OUT1L_VOL_SHIFT, 159, 0, + digital_tlv), +SOC_DOUBLE_R_TLV("HPOUT2 Digital Volume", WM5100_DAC_DIGITAL_VOLUME_2L, + WM5100_DAC_DIGITAL_VOLUME_2R, WM5100_OUT2L_VOL_SHIFT, 159, 0, + digital_tlv), +SOC_DOUBLE_R_TLV("HPOUT3 Digital Volume", WM5100_DAC_DIGITAL_VOLUME_3L, + WM5100_DAC_DIGITAL_VOLUME_3R, WM5100_OUT3L_VOL_SHIFT, 159, 0, + digital_tlv), +SOC_DOUBLE_R_TLV("SPKOUT Digital Volume", WM5100_DAC_DIGITAL_VOLUME_4L, + WM5100_DAC_DIGITAL_VOLUME_4R, WM5100_OUT4L_VOL_SHIFT, 159, 0, + digital_tlv), +SOC_DOUBLE_R_TLV("SPKDAT1 Digital Volume", WM5100_DAC_DIGITAL_VOLUME_5L, + WM5100_DAC_DIGITAL_VOLUME_5R, WM5100_OUT5L_VOL_SHIFT, 159, 0, + digital_tlv), +SOC_DOUBLE_R_TLV("SPKDAT2 Digital Volume", WM5100_DAC_DIGITAL_VOLUME_6L, + WM5100_DAC_DIGITAL_VOLUME_6R, WM5100_OUT6L_VOL_SHIFT, 159, 0, + digital_tlv), + +SOC_DOUBLE_R("HPOUT1 Digital Switch", WM5100_DAC_DIGITAL_VOLUME_1L, + WM5100_DAC_DIGITAL_VOLUME_1R, WM5100_OUT1L_MUTE_SHIFT, 1, 1), +SOC_DOUBLE_R("HPOUT2 Digital Switch", WM5100_DAC_DIGITAL_VOLUME_2L, + WM5100_DAC_DIGITAL_VOLUME_2R, WM5100_OUT2L_MUTE_SHIFT, 1, 1), +SOC_DOUBLE_R("HPOUT3 Digital Switch", WM5100_DAC_DIGITAL_VOLUME_3L, + WM5100_DAC_DIGITAL_VOLUME_3R, WM5100_OUT3L_MUTE_SHIFT, 1, 1), +SOC_DOUBLE_R("SPKOUT Digital Switch", WM5100_DAC_DIGITAL_VOLUME_4L, + WM5100_DAC_DIGITAL_VOLUME_4R, WM5100_OUT4L_MUTE_SHIFT, 1, 1), +SOC_DOUBLE_R("SPKDAT1 Digital Switch", WM5100_DAC_DIGITAL_VOLUME_5L, + WM5100_DAC_DIGITAL_VOLUME_5R, WM5100_OUT5L_MUTE_SHIFT, 1, 1), +SOC_DOUBLE_R("SPKDAT2 Digital Switch", WM5100_DAC_DIGITAL_VOLUME_6L, + WM5100_DAC_DIGITAL_VOLUME_6R, WM5100_OUT6L_MUTE_SHIFT, 1, 1), + +/* FIXME: Only valid from -12dB to 0dB (52-64) */ +SOC_DOUBLE_R_TLV("HPOUT1 Volume", WM5100_OUT_VOLUME_1L, WM5100_OUT_VOLUME_1R, + WM5100_OUT1L_PGA_VOL_SHIFT, 64, 0, out_tlv), +SOC_DOUBLE_R_TLV("HPOUT2 Volume", WM5100_OUT_VOLUME_2L, WM5100_OUT_VOLUME_2R, + WM5100_OUT2L_PGA_VOL_SHIFT, 64, 0, out_tlv), +SOC_DOUBLE_R_TLV("HPOUT3 Volume", WM5100_OUT_VOLUME_3L, WM5100_OUT_VOLUME_3R, + WM5100_OUT2L_PGA_VOL_SHIFT, 64, 0, out_tlv), + +SOC_DOUBLE("SPKDAT1 Switch", WM5100_PDM_SPK1_CTRL_1, WM5100_SPK1L_MUTE_SHIFT, + WM5100_SPK1R_MUTE_SHIFT, 1, 1), +SOC_DOUBLE("SPKDAT2 Switch", WM5100_PDM_SPK2_CTRL_1, WM5100_SPK2L_MUTE_SHIFT, + WM5100_SPK2R_MUTE_SHIFT, 1, 1), + +SOC_SINGLE_TLV("EQ1 Band 1 Volume", WM5100_EQ1_1, WM5100_EQ1_B1_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ1 Band 2 Volume", WM5100_EQ1_1, WM5100_EQ1_B2_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ1 Band 3 Volume", WM5100_EQ1_1, WM5100_EQ1_B3_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ1 Band 4 Volume", WM5100_EQ1_2, WM5100_EQ1_B4_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ1 Band 5 Volume", WM5100_EQ1_2, WM5100_EQ1_B5_GAIN_SHIFT, + 24, 0, eq_tlv), + +SOC_SINGLE_TLV("EQ2 Band 1 Volume", WM5100_EQ2_1, WM5100_EQ2_B1_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ2 Band 2 Volume", WM5100_EQ2_1, WM5100_EQ2_B2_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ2 Band 3 Volume", WM5100_EQ2_1, WM5100_EQ2_B3_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ2 Band 4 Volume", WM5100_EQ2_2, WM5100_EQ2_B4_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ2 Band 5 Volume", WM5100_EQ2_2, WM5100_EQ2_B5_GAIN_SHIFT, + 24, 0, eq_tlv), + +SOC_SINGLE_TLV("EQ3 Band 1 Volume", WM5100_EQ1_1, WM5100_EQ3_B1_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ3 Band 2 Volume", WM5100_EQ3_1, WM5100_EQ3_B2_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ3 Band 3 Volume", WM5100_EQ3_1, WM5100_EQ3_B3_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ3 Band 4 Volume", WM5100_EQ3_2, WM5100_EQ3_B4_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ3 Band 5 Volume", WM5100_EQ3_2, WM5100_EQ3_B5_GAIN_SHIFT, + 24, 0, eq_tlv), + +SOC_SINGLE_TLV("EQ4 Band 1 Volume", WM5100_EQ4_1, WM5100_EQ4_B1_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ4 Band 2 Volume", WM5100_EQ4_1, WM5100_EQ4_B2_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ4 Band 3 Volume", WM5100_EQ4_1, WM5100_EQ4_B3_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ4 Band 4 Volume", WM5100_EQ4_2, WM5100_EQ4_B4_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ4 Band 5 Volume", WM5100_EQ4_2, WM5100_EQ4_B5_GAIN_SHIFT, + 24, 0, eq_tlv), + +SOC_ENUM("LHPF1 Mode", wm5100_lhpf1_mode), +SOC_ENUM("LHPF2 Mode", wm5100_lhpf2_mode), +SOC_ENUM("LHPF3 Mode", wm5100_lhpf3_mode), +SOC_ENUM("LHPF4 Mode", wm5100_lhpf4_mode), + +WM5100_MIXER_CONTROLS("HPOUT1L", WM5100_OUT1LMIX_INPUT_1_SOURCE), +WM5100_MIXER_CONTROLS("HPOUT1R", WM5100_OUT1RMIX_INPUT_1_SOURCE), +WM5100_MIXER_CONTROLS("HPOUT2L", WM5100_OUT2LMIX_INPUT_1_SOURCE), +WM5100_MIXER_CONTROLS("HPOUT2R", WM5100_OUT2RMIX_INPUT_1_SOURCE), +WM5100_MIXER_CONTROLS("HPOUT3L", WM5100_OUT3LMIX_INPUT_1_SOURCE), +WM5100_MIXER_CONTROLS("HPOUT3R", WM5100_OUT3RMIX_INPUT_1_SOURCE), + +WM5100_MIXER_CONTROLS("SPKOUTL", WM5100_OUT4LMIX_INPUT_1_SOURCE), +WM5100_MIXER_CONTROLS("SPKOUTR", WM5100_OUT4RMIX_INPUT_1_SOURCE), +WM5100_MIXER_CONTROLS("SPKDAT1L", WM5100_OUT5LMIX_INPUT_1_SOURCE), +WM5100_MIXER_CONTROLS("SPKDAT1R", WM5100_OUT5RMIX_INPUT_1_SOURCE), +WM5100_MIXER_CONTROLS("SPKDAT2L", WM5100_OUT6LMIX_INPUT_1_SOURCE), +WM5100_MIXER_CONTROLS("SPKDAT2R", WM5100_OUT6RMIX_INPUT_1_SOURCE), + +WM5100_MIXER_CONTROLS("PWM1", WM5100_PWM1MIX_INPUT_1_SOURCE), +WM5100_MIXER_CONTROLS("PWM2", WM5100_PWM2MIX_INPUT_1_SOURCE), + +WM5100_MIXER_CONTROLS("AIF1TX1", WM5100_AIF1TX1MIX_INPUT_1_SOURCE), +WM5100_MIXER_CONTROLS("AIF1TX2", WM5100_AIF1TX2MIX_INPUT_1_SOURCE), +WM5100_MIXER_CONTROLS("AIF1TX3", WM5100_AIF1TX3MIX_INPUT_1_SOURCE), +WM5100_MIXER_CONTROLS("AIF1TX4", WM5100_AIF1TX4MIX_INPUT_1_SOURCE), +WM5100_MIXER_CONTROLS("AIF1TX5", WM5100_AIF1TX5MIX_INPUT_1_SOURCE), +WM5100_MIXER_CONTROLS("AIF1TX6", WM5100_AIF1TX6MIX_INPUT_1_SOURCE), +WM5100_MIXER_CONTROLS("AIF1TX7", WM5100_AIF1TX7MIX_INPUT_1_SOURCE), +WM5100_MIXER_CONTROLS("AIF1TX8", WM5100_AIF1TX8MIX_INPUT_1_SOURCE), + +WM5100_MIXER_CONTROLS("AIF2TX1", WM5100_AIF2TX1MIX_INPUT_1_SOURCE), +WM5100_MIXER_CONTROLS("AIF2TX2", WM5100_AIF2TX2MIX_INPUT_1_SOURCE), + +WM5100_MIXER_CONTROLS("AIF3TX1", WM5100_AIF3TX1MIX_INPUT_1_SOURCE), +WM5100_MIXER_CONTROLS("AIF3TX2", WM5100_AIF3TX2MIX_INPUT_1_SOURCE), + +WM5100_MIXER_CONTROLS("EQ1", WM5100_EQ1MIX_INPUT_1_SOURCE), +WM5100_MIXER_CONTROLS("EQ2", WM5100_EQ2MIX_INPUT_1_SOURCE), +WM5100_MIXER_CONTROLS("EQ3", WM5100_EQ3MIX_INPUT_1_SOURCE), +WM5100_MIXER_CONTROLS("EQ4", WM5100_EQ4MIX_INPUT_1_SOURCE), + +WM5100_MIXER_CONTROLS("DRC1L", WM5100_DRC1LMIX_INPUT_1_SOURCE), +WM5100_MIXER_CONTROLS("DRC1R", WM5100_DRC1RMIX_INPUT_1_SOURCE), + +WM5100_MIXER_CONTROLS("LHPF1", WM5100_HPLP1MIX_INPUT_1_SOURCE), +WM5100_MIXER_CONTROLS("LHPF2", WM5100_HPLP2MIX_INPUT_1_SOURCE), +WM5100_MIXER_CONTROLS("LHPF3", WM5100_HPLP3MIX_INPUT_1_SOURCE), +WM5100_MIXER_CONTROLS("LHPF4", WM5100_HPLP4MIX_INPUT_1_SOURCE), +}; + +static void wm5100_seq_notifier(struct snd_soc_dapm_context *dapm, + enum snd_soc_dapm_type event, int subseq) +{ + struct snd_soc_codec *codec = container_of(dapm, + struct snd_soc_codec, dapm); + struct wm5100_priv *wm5100 = snd_soc_codec_get_drvdata(codec); + u16 val, expect, i; + + /* Wait for the outputs to flag themselves as enabled */ + if (wm5100->out_ena[0]) { + expect = snd_soc_read(codec, WM5100_CHANNEL_ENABLES_1); + for (i = 0; i < 200; i++) { + val = snd_soc_read(codec, WM5100_OUTPUT_STATUS_1); + if (val == expect) { + wm5100->out_ena[0] = false; + break; + } + } + if (i == 200) { + dev_err(codec->dev, "Timeout waiting for OUTPUT1 %x\n", + expect); + } + } + + if (wm5100->out_ena[1]) { + expect = snd_soc_read(codec, WM5100_OUTPUT_ENABLES_2); + for (i = 0; i < 200; i++) { + val = snd_soc_read(codec, WM5100_OUTPUT_STATUS_2); + if (val == expect) { + wm5100->out_ena[1] = false; + break; + } + } + if (i == 200) { + dev_err(codec->dev, "Timeout waiting for OUTPUT2 %x\n", + expect); + } + } +} + +static int wm5100_out_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, + int event) +{ + struct wm5100_priv *wm5100 = snd_soc_codec_get_drvdata(w->codec); + + switch (w->reg) { + case WM5100_CHANNEL_ENABLES_1: + wm5100->out_ena[0] = true; + break; + case WM5100_OUTPUT_ENABLES_2: + wm5100->out_ena[0] = true; + break; + default: + break; + } + + return 0; +} + +static int wm5100_cp_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, + int event) +{ + struct snd_soc_codec *codec = w->codec; + struct wm5100_priv *wm5100 = snd_soc_codec_get_drvdata(codec); + int ret; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + ret = regulator_enable(wm5100->cpvdd); + if (ret != 0) { + dev_err(codec->dev, "Failed to enable CPVDD: %d\n", + ret); + return ret; + } + return ret; + + case SND_SOC_DAPM_POST_PMD: + ret = regulator_disable_deferred(wm5100->cpvdd, 20); + if (ret != 0) { + dev_err(codec->dev, "Failed to disable CPVDD: %d\n", + ret); + return ret; + } + return ret; + + default: + BUG(); + return 0; + } +} + +static void wm5100_log_status3(struct snd_soc_codec *codec, int val) +{ + if (val & WM5100_SPK_SHUTDOWN_WARN_EINT) + dev_crit(codec->dev, "Speaker shutdown warning\n"); + if (val & WM5100_SPK_SHUTDOWN_EINT) + dev_crit(codec->dev, "Speaker shutdown\n"); + if (val & WM5100_CLKGEN_ERR_EINT) + dev_crit(codec->dev, "SYSCLK underclocked\n"); + if (val & WM5100_CLKGEN_ERR_ASYNC_EINT) + dev_crit(codec->dev, "ASYNCCLK underclocked\n"); +} + +static void wm5100_log_status4(struct snd_soc_codec *codec, int val) +{ + if (val & WM5100_AIF3_ERR_EINT) + dev_err(codec->dev, "AIF3 configuration error\n"); + if (val & WM5100_AIF2_ERR_EINT) + dev_err(codec->dev, "AIF2 configuration error\n"); + if (val & WM5100_AIF1_ERR_EINT) + dev_err(codec->dev, "AIF1 configuration error\n"); + if (val & WM5100_CTRLIF_ERR_EINT) + dev_err(codec->dev, "Control interface error\n"); + if (val & WM5100_ISRC2_UNDERCLOCKED_EINT) + dev_err(codec->dev, "ISRC2 underclocked\n"); + if (val & WM5100_ISRC1_UNDERCLOCKED_EINT) + dev_err(codec->dev, "ISRC1 underclocked\n"); + if (val & WM5100_FX_UNDERCLOCKED_EINT) + dev_err(codec->dev, "FX underclocked\n"); + if (val & WM5100_AIF3_UNDERCLOCKED_EINT) + dev_err(codec->dev, "AIF3 underclocked\n"); + if (val & WM5100_AIF2_UNDERCLOCKED_EINT) + dev_err(codec->dev, "AIF2 underclocked\n"); + if (val & WM5100_AIF1_UNDERCLOCKED_EINT) + dev_err(codec->dev, "AIF1 underclocked\n"); + if (val & WM5100_ASRC_UNDERCLOCKED_EINT) + dev_err(codec->dev, "ASRC underclocked\n"); + if (val & WM5100_DAC_UNDERCLOCKED_EINT) + dev_err(codec->dev, "DAC underclocked\n"); + if (val & WM5100_ADC_UNDERCLOCKED_EINT) + dev_err(codec->dev, "ADC underclocked\n"); + if (val & WM5100_MIXER_UNDERCLOCKED_EINT) + dev_err(codec->dev, "Mixer underclocked\n"); +} + +static int wm5100_post_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, + int event) +{ + struct snd_soc_codec *codec = w->codec; + int ret; + + ret = snd_soc_read(codec, WM5100_INTERRUPT_RAW_STATUS_3); + ret &= WM5100_SPK_SHUTDOWN_WARN_STS | + WM5100_SPK_SHUTDOWN_STS | WM5100_CLKGEN_ERR_STS | + WM5100_CLKGEN_ERR_ASYNC_STS; + wm5100_log_status3(codec, ret); + + ret = snd_soc_read(codec, WM5100_INTERRUPT_RAW_STATUS_4); + wm5100_log_status4(codec, ret); + + return 0; +} + +static const struct snd_soc_dapm_widget wm5100_dapm_widgets[] = { +SND_SOC_DAPM_SUPPLY("SYSCLK", WM5100_CLOCKING_3, WM5100_SYSCLK_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_SUPPLY("ASYNCCLK", WM5100_CLOCKING_6, WM5100_ASYNC_CLK_ENA_SHIFT, + 0, NULL, 0), + +SND_SOC_DAPM_SUPPLY("CP1", WM5100_HP_CHARGE_PUMP_1, WM5100_CP1_ENA_SHIFT, 0, + wm5100_cp_ev, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), +SND_SOC_DAPM_SUPPLY("CP2", WM5100_MIC_CHARGE_PUMP_1, WM5100_CP2_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_SUPPLY("CP2 Active", WM5100_MIC_CHARGE_PUMP_1, + WM5100_CP2_BYPASS_SHIFT, 1, wm5100_cp_ev, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + +SND_SOC_DAPM_SUPPLY("MICBIAS1", WM5100_MIC_BIAS_CTRL_1, WM5100_MICB1_ENA_SHIFT, + 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("MICBIAS2", WM5100_MIC_BIAS_CTRL_2, WM5100_MICB2_ENA_SHIFT, + 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("MICBIAS3", WM5100_MIC_BIAS_CTRL_3, WM5100_MICB3_ENA_SHIFT, + 0, NULL, 0), + +SND_SOC_DAPM_INPUT("IN1L"), +SND_SOC_DAPM_INPUT("IN1R"), +SND_SOC_DAPM_INPUT("IN2L"), +SND_SOC_DAPM_INPUT("IN2R"), +SND_SOC_DAPM_INPUT("IN3L"), +SND_SOC_DAPM_INPUT("IN3R"), +SND_SOC_DAPM_INPUT("IN4L"), +SND_SOC_DAPM_INPUT("IN4R"), +SND_SOC_DAPM_INPUT("TONE"), + +SND_SOC_DAPM_PGA_E("IN1L PGA", WM5100_INPUT_ENABLES, WM5100_IN1L_ENA_SHIFT, 0, + NULL, 0, wm5100_out_ev, SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("IN1R PGA", WM5100_INPUT_ENABLES, WM5100_IN1R_ENA_SHIFT, 0, + NULL, 0, wm5100_out_ev, SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("IN2L PGA", WM5100_INPUT_ENABLES, WM5100_IN2L_ENA_SHIFT, 0, + NULL, 0, wm5100_out_ev, SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("IN2R PGA", WM5100_INPUT_ENABLES, WM5100_IN2R_ENA_SHIFT, 0, + NULL, 0, wm5100_out_ev, SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("IN3L PGA", WM5100_INPUT_ENABLES, WM5100_IN3L_ENA_SHIFT, 0, + NULL, 0, wm5100_out_ev, SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("IN3R PGA", WM5100_INPUT_ENABLES, WM5100_IN3R_ENA_SHIFT, 0, + NULL, 0, wm5100_out_ev, SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("IN4L PGA", WM5100_INPUT_ENABLES, WM5100_IN4L_ENA_SHIFT, 0, + NULL, 0, wm5100_out_ev, SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("IN4R PGA", WM5100_INPUT_ENABLES, WM5100_IN4R_ENA_SHIFT, 0, + NULL, 0, wm5100_out_ev, SND_SOC_DAPM_POST_PMU), + +SND_SOC_DAPM_PGA("Tone Generator 1", WM5100_TONE_GENERATOR_1, + WM5100_TONE1_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("Tone Generator 2", WM5100_TONE_GENERATOR_1, + WM5100_TONE2_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_AIF_IN("AIF1RX1", "AIF1 Playback", 0, + WM5100_AUDIO_IF_1_27, WM5100_AIF1RX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX2", "AIF1 Playback", 1, + WM5100_AUDIO_IF_1_27, WM5100_AIF1RX2_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX3", "AIF1 Playback", 2, + WM5100_AUDIO_IF_1_27, WM5100_AIF1RX3_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX4", "AIF1 Playback", 3, + WM5100_AUDIO_IF_1_27, WM5100_AIF1RX4_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX5", "AIF1 Playback", 4, + WM5100_AUDIO_IF_1_27, WM5100_AIF1RX5_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX6", "AIF1 Playback", 5, + WM5100_AUDIO_IF_1_27, WM5100_AIF1RX6_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX7", "AIF1 Playback", 6, + WM5100_AUDIO_IF_1_27, WM5100_AIF1RX7_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX8", "AIF1 Playback", 7, + WM5100_AUDIO_IF_1_27, WM5100_AIF1RX8_ENA_SHIFT, 0), + +SND_SOC_DAPM_AIF_IN("AIF2RX1", "AIF2 Playback", 0, + WM5100_AUDIO_IF_2_27, WM5100_AIF2RX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF2RX2", "AIF2 Playback", 1, + WM5100_AUDIO_IF_2_27, WM5100_AIF2RX2_ENA_SHIFT, 0), + +SND_SOC_DAPM_AIF_IN("AIF3RX1", "AIF3 Playback", 0, + WM5100_AUDIO_IF_3_27, WM5100_AIF3RX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF3RX2", "AIF3 Playback", 1, + WM5100_AUDIO_IF_3_27, WM5100_AIF3RX2_ENA_SHIFT, 0), + +SND_SOC_DAPM_AIF_OUT("AIF1TX1", "AIF1 Capture", 0, + WM5100_AUDIO_IF_1_26, WM5100_AIF1TX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX2", "AIF1 Capture", 1, + WM5100_AUDIO_IF_1_26, WM5100_AIF1TX2_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX3", "AIF1 Capture", 2, + WM5100_AUDIO_IF_1_26, WM5100_AIF1TX3_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX4", "AIF1 Capture", 3, + WM5100_AUDIO_IF_1_26, WM5100_AIF1TX4_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX5", "AIF1 Capture", 4, + WM5100_AUDIO_IF_1_26, WM5100_AIF1TX5_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX6", "AIF1 Capture", 5, + WM5100_AUDIO_IF_1_26, WM5100_AIF1TX6_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX7", "AIF1 Capture", 6, + WM5100_AUDIO_IF_1_26, WM5100_AIF1TX7_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX8", "AIF1 Capture", 7, + WM5100_AUDIO_IF_1_26, WM5100_AIF1TX8_ENA_SHIFT, 0), + +SND_SOC_DAPM_AIF_OUT("AIF2TX1", "AIF2 Capture", 0, + WM5100_AUDIO_IF_2_26, WM5100_AIF2TX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF2TX2", "AIF2 Capture", 1, + WM5100_AUDIO_IF_2_26, WM5100_AIF2TX2_ENA_SHIFT, 0), + +SND_SOC_DAPM_AIF_OUT("AIF3TX1", "AIF3 Capture", 0, + WM5100_AUDIO_IF_3_26, WM5100_AIF3TX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF3TX2", "AIF3 Capture", 1, + WM5100_AUDIO_IF_3_26, WM5100_AIF3TX2_ENA_SHIFT, 0), + +SND_SOC_DAPM_PGA_E("OUT6L", WM5100_OUTPUT_ENABLES_2, WM5100_OUT6L_ENA_SHIFT, 0, + NULL, 0, wm5100_out_ev, SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("OUT6R", WM5100_OUTPUT_ENABLES_2, WM5100_OUT6R_ENA_SHIFT, 0, + NULL, 0, wm5100_out_ev, SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("OUT5L", WM5100_OUTPUT_ENABLES_2, WM5100_OUT5L_ENA_SHIFT, 0, + NULL, 0, wm5100_out_ev, SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("OUT5R", WM5100_OUTPUT_ENABLES_2, WM5100_OUT5R_ENA_SHIFT, 0, + NULL, 0, wm5100_out_ev, SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("OUT4L", WM5100_OUTPUT_ENABLES_2, WM5100_OUT4L_ENA_SHIFT, 0, + NULL, 0, wm5100_out_ev, SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("OUT4R", WM5100_OUTPUT_ENABLES_2, WM5100_OUT4R_ENA_SHIFT, 0, + NULL, 0, wm5100_out_ev, SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("OUT3L", WM5100_CHANNEL_ENABLES_1, WM5100_HP3L_ENA_SHIFT, 0, + NULL, 0, wm5100_out_ev, SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("OUT3R", WM5100_CHANNEL_ENABLES_1, WM5100_HP3R_ENA_SHIFT, 0, + NULL, 0, wm5100_out_ev, SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("OUT2L", WM5100_CHANNEL_ENABLES_1, WM5100_HP2L_ENA_SHIFT, 0, + NULL, 0, wm5100_out_ev, SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("OUT2R", WM5100_CHANNEL_ENABLES_1, WM5100_HP2R_ENA_SHIFT, 0, + NULL, 0, wm5100_out_ev, SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("OUT1L", WM5100_CHANNEL_ENABLES_1, WM5100_HP1L_ENA_SHIFT, 0, + NULL, 0, wm5100_out_ev, SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("OUT1R", WM5100_CHANNEL_ENABLES_1, WM5100_HP1R_ENA_SHIFT, 0, + NULL, 0, wm5100_out_ev, SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("PWM1 Driver", WM5100_PWM_DRIVE_1, WM5100_PWM1_ENA_SHIFT, 0, + NULL, 0, wm5100_out_ev, SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("PWM2 Driver", WM5100_PWM_DRIVE_1, WM5100_PWM2_ENA_SHIFT, 0, + NULL, 0, wm5100_out_ev, SND_SOC_DAPM_POST_PMU), + +SND_SOC_DAPM_PGA("EQ1", WM5100_EQ1_1, WM5100_EQ1_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("EQ2", WM5100_EQ2_1, WM5100_EQ2_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("EQ3", WM5100_EQ3_1, WM5100_EQ3_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("EQ4", WM5100_EQ4_1, WM5100_EQ4_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("DRC1L", WM5100_DRC1_CTRL1, WM5100_DRCL_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_PGA("DRC1R", WM5100_DRC1_CTRL1, WM5100_DRCR_ENA_SHIFT, 0, + NULL, 0), + +SND_SOC_DAPM_PGA("LHPF1", WM5100_HPLPF1_1, WM5100_LHPF1_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_PGA("LHPF2", WM5100_HPLPF2_1, WM5100_LHPF2_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_PGA("LHPF3", WM5100_HPLPF3_1, WM5100_LHPF3_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_PGA("LHPF4", WM5100_HPLPF4_1, WM5100_LHPF4_ENA_SHIFT, 0, + NULL, 0), + +WM5100_MIXER_WIDGETS(EQ1, "EQ1"), +WM5100_MIXER_WIDGETS(EQ2, "EQ2"), +WM5100_MIXER_WIDGETS(EQ3, "EQ3"), +WM5100_MIXER_WIDGETS(EQ4, "EQ4"), + +WM5100_MIXER_WIDGETS(DRC1L, "DRC1L"), +WM5100_MIXER_WIDGETS(DRC1R, "DRC1R"), + +WM5100_MIXER_WIDGETS(LHPF1, "LHPF1"), +WM5100_MIXER_WIDGETS(LHPF2, "LHPF2"), +WM5100_MIXER_WIDGETS(LHPF3, "LHPF3"), +WM5100_MIXER_WIDGETS(LHPF4, "LHPF4"), + +WM5100_MIXER_WIDGETS(AIF1TX1, "AIF1TX1"), +WM5100_MIXER_WIDGETS(AIF1TX2, "AIF1TX2"), +WM5100_MIXER_WIDGETS(AIF1TX3, "AIF1TX3"), +WM5100_MIXER_WIDGETS(AIF1TX4, "AIF1TX4"), +WM5100_MIXER_WIDGETS(AIF1TX5, "AIF1TX5"), +WM5100_MIXER_WIDGETS(AIF1TX6, "AIF1TX6"), +WM5100_MIXER_WIDGETS(AIF1TX7, "AIF1TX7"), +WM5100_MIXER_WIDGETS(AIF1TX8, "AIF1TX8"), + +WM5100_MIXER_WIDGETS(AIF2TX1, "AIF2TX1"), +WM5100_MIXER_WIDGETS(AIF2TX2, "AIF2TX2"), + +WM5100_MIXER_WIDGETS(AIF3TX1, "AIF3TX1"), +WM5100_MIXER_WIDGETS(AIF3TX2, "AIF3TX2"), + +WM5100_MIXER_WIDGETS(HPOUT1L, "HPOUT1L"), +WM5100_MIXER_WIDGETS(HPOUT1R, "HPOUT1R"), +WM5100_MIXER_WIDGETS(HPOUT2L, "HPOUT2L"), +WM5100_MIXER_WIDGETS(HPOUT2R, "HPOUT2R"), +WM5100_MIXER_WIDGETS(HPOUT3L, "HPOUT3L"), +WM5100_MIXER_WIDGETS(HPOUT3R, "HPOUT3R"), + +WM5100_MIXER_WIDGETS(SPKOUTL, "SPKOUTL"), +WM5100_MIXER_WIDGETS(SPKOUTR, "SPKOUTR"), +WM5100_MIXER_WIDGETS(SPKDAT1L, "SPKDAT1L"), +WM5100_MIXER_WIDGETS(SPKDAT1R, "SPKDAT1R"), +WM5100_MIXER_WIDGETS(SPKDAT2L, "SPKDAT2L"), +WM5100_MIXER_WIDGETS(SPKDAT2R, "SPKDAT2R"), + +WM5100_MIXER_WIDGETS(PWM1, "PWM1"), +WM5100_MIXER_WIDGETS(PWM2, "PWM2"), + +SND_SOC_DAPM_OUTPUT("HPOUT1L"), +SND_SOC_DAPM_OUTPUT("HPOUT1R"), +SND_SOC_DAPM_OUTPUT("HPOUT2L"), +SND_SOC_DAPM_OUTPUT("HPOUT2R"), +SND_SOC_DAPM_OUTPUT("HPOUT3L"), +SND_SOC_DAPM_OUTPUT("HPOUT3R"), +SND_SOC_DAPM_OUTPUT("SPKOUTL"), +SND_SOC_DAPM_OUTPUT("SPKOUTR"), +SND_SOC_DAPM_OUTPUT("SPKDAT1"), +SND_SOC_DAPM_OUTPUT("SPKDAT2"), +SND_SOC_DAPM_OUTPUT("PWM1"), +SND_SOC_DAPM_OUTPUT("PWM2"), +}; + +/* We register a _POST event if we don't have IRQ support so we can + * look at the error status from the CODEC - if we've got the IRQ + * hooked up then we will get prompted to look by an interrupt. + */ +static const struct snd_soc_dapm_widget wm5100_dapm_widgets_noirq[] = { +SND_SOC_DAPM_POST("Post", wm5100_post_ev), +}; + +static const struct snd_soc_dapm_route wm5100_dapm_routes[] = { + { "IN1L", NULL, "SYSCLK" }, + { "IN1R", NULL, "SYSCLK" }, + { "IN2L", NULL, "SYSCLK" }, + { "IN2R", NULL, "SYSCLK" }, + { "IN3L", NULL, "SYSCLK" }, + { "IN3R", NULL, "SYSCLK" }, + { "IN4L", NULL, "SYSCLK" }, + { "IN4R", NULL, "SYSCLK" }, + + { "OUT1L", NULL, "SYSCLK" }, + { "OUT1R", NULL, "SYSCLK" }, + { "OUT2L", NULL, "SYSCLK" }, + { "OUT2R", NULL, "SYSCLK" }, + { "OUT3L", NULL, "SYSCLK" }, + { "OUT3R", NULL, "SYSCLK" }, + { "OUT4L", NULL, "SYSCLK" }, + { "OUT4R", NULL, "SYSCLK" }, + { "OUT5L", NULL, "SYSCLK" }, + { "OUT5R", NULL, "SYSCLK" }, + { "OUT6L", NULL, "SYSCLK" }, + { "OUT6R", NULL, "SYSCLK" }, + + { "AIF1RX1", NULL, "SYSCLK" }, + { "AIF1RX2", NULL, "SYSCLK" }, + { "AIF1RX3", NULL, "SYSCLK" }, + { "AIF1RX4", NULL, "SYSCLK" }, + { "AIF1RX5", NULL, "SYSCLK" }, + { "AIF1RX6", NULL, "SYSCLK" }, + { "AIF1RX7", NULL, "SYSCLK" }, + { "AIF1RX8", NULL, "SYSCLK" }, + + { "AIF2RX1", NULL, "SYSCLK" }, + { "AIF2RX2", NULL, "SYSCLK" }, + + { "AIF3RX1", NULL, "SYSCLK" }, + { "AIF3RX2", NULL, "SYSCLK" }, + + { "AIF1TX1", NULL, "SYSCLK" }, + { "AIF1TX2", NULL, "SYSCLK" }, + { "AIF1TX3", NULL, "SYSCLK" }, + { "AIF1TX4", NULL, "SYSCLK" }, + { "AIF1TX5", NULL, "SYSCLK" }, + { "AIF1TX6", NULL, "SYSCLK" }, + { "AIF1TX7", NULL, "SYSCLK" }, + { "AIF1TX8", NULL, "SYSCLK" }, + + { "AIF2TX1", NULL, "SYSCLK" }, + { "AIF2TX2", NULL, "SYSCLK" }, + + { "AIF3TX1", NULL, "SYSCLK" }, + { "AIF3TX2", NULL, "SYSCLK" }, + + { "MICBIAS1", NULL, "CP2" }, + { "MICBIAS2", NULL, "CP2" }, + { "MICBIAS3", NULL, "CP2" }, + + { "IN1L PGA", NULL, "CP2" }, + { "IN1R PGA", NULL, "CP2" }, + { "IN2L PGA", NULL, "CP2" }, + { "IN2R PGA", NULL, "CP2" }, + { "IN3L PGA", NULL, "CP2" }, + { "IN3R PGA", NULL, "CP2" }, + { "IN4L PGA", NULL, "CP2" }, + { "IN4R PGA", NULL, "CP2" }, + + { "IN1L PGA", NULL, "CP2 Active" }, + { "IN1R PGA", NULL, "CP2 Active" }, + { "IN2L PGA", NULL, "CP2 Active" }, + { "IN2R PGA", NULL, "CP2 Active" }, + { "IN3L PGA", NULL, "CP2 Active" }, + { "IN3R PGA", NULL, "CP2 Active" }, + { "IN4L PGA", NULL, "CP2 Active" }, + { "IN4R PGA", NULL, "CP2 Active" }, + + { "OUT1L", NULL, "CP1" }, + { "OUT1R", NULL, "CP1" }, + { "OUT2L", NULL, "CP1" }, + { "OUT2R", NULL, "CP1" }, + { "OUT3L", NULL, "CP1" }, + { "OUT3R", NULL, "CP1" }, + + { "Tone Generator 1", NULL, "TONE" }, + { "Tone Generator 2", NULL, "TONE" }, + + { "IN1L PGA", NULL, "IN1L" }, + { "IN1R PGA", NULL, "IN1R" }, + { "IN2L PGA", NULL, "IN2L" }, + { "IN2R PGA", NULL, "IN2R" }, + { "IN3L PGA", NULL, "IN3L" }, + { "IN3R PGA", NULL, "IN3R" }, + { "IN4L PGA", NULL, "IN4L" }, + { "IN4R PGA", NULL, "IN4R" }, + + WM5100_MIXER_ROUTES("OUT1L", "HPOUT1L"), + WM5100_MIXER_ROUTES("OUT1R", "HPOUT1R"), + WM5100_MIXER_ROUTES("OUT2L", "HPOUT2L"), + WM5100_MIXER_ROUTES("OUT2R", "HPOUT2R"), + WM5100_MIXER_ROUTES("OUT3L", "HPOUT3L"), + WM5100_MIXER_ROUTES("OUT3R", "HPOUT3R"), + + WM5100_MIXER_ROUTES("OUT4L", "SPKOUTL"), + WM5100_MIXER_ROUTES("OUT4R", "SPKOUTR"), + WM5100_MIXER_ROUTES("OUT5L", "SPKDAT1L"), + WM5100_MIXER_ROUTES("OUT5R", "SPKDAT1R"), + WM5100_MIXER_ROUTES("OUT6L", "SPKDAT2L"), + WM5100_MIXER_ROUTES("OUT6R", "SPKDAT2R"), + + WM5100_MIXER_ROUTES("PWM1 Driver", "PWM1"), + WM5100_MIXER_ROUTES("PWM2 Driver", "PWM2"), + + WM5100_MIXER_ROUTES("AIF1TX1", "AIF1TX1"), + WM5100_MIXER_ROUTES("AIF1TX2", "AIF1TX2"), + WM5100_MIXER_ROUTES("AIF1TX3", "AIF1TX3"), + WM5100_MIXER_ROUTES("AIF1TX4", "AIF1TX4"), + WM5100_MIXER_ROUTES("AIF1TX5", "AIF1TX5"), + WM5100_MIXER_ROUTES("AIF1TX6", "AIF1TX6"), + WM5100_MIXER_ROUTES("AIF1TX7", "AIF1TX7"), + WM5100_MIXER_ROUTES("AIF1TX8", "AIF1TX8"), + + WM5100_MIXER_ROUTES("AIF2TX1", "AIF2TX1"), + WM5100_MIXER_ROUTES("AIF2TX2", "AIF2TX2"), + + WM5100_MIXER_ROUTES("AIF3TX1", "AIF3TX1"), + WM5100_MIXER_ROUTES("AIF3TX2", "AIF3TX2"), + + WM5100_MIXER_ROUTES("EQ1", "EQ1"), + WM5100_MIXER_ROUTES("EQ2", "EQ2"), + WM5100_MIXER_ROUTES("EQ3", "EQ3"), + WM5100_MIXER_ROUTES("EQ4", "EQ4"), + + WM5100_MIXER_ROUTES("DRC1L", "DRC1L"), + WM5100_MIXER_ROUTES("DRC1R", "DRC1R"), + + WM5100_MIXER_ROUTES("LHPF1", "LHPF1"), + WM5100_MIXER_ROUTES("LHPF2", "LHPF2"), + WM5100_MIXER_ROUTES("LHPF3", "LHPF3"), + WM5100_MIXER_ROUTES("LHPF4", "LHPF4"), + + { "HPOUT1L", NULL, "OUT1L" }, + { "HPOUT1R", NULL, "OUT1R" }, + { "HPOUT2L", NULL, "OUT2L" }, + { "HPOUT2R", NULL, "OUT2R" }, + { "HPOUT3L", NULL, "OUT3L" }, + { "HPOUT3R", NULL, "OUT3R" }, + { "SPKOUTL", NULL, "OUT4L" }, + { "SPKOUTR", NULL, "OUT4R" }, + { "SPKDAT1", NULL, "OUT5L" }, + { "SPKDAT1", NULL, "OUT5R" }, + { "SPKDAT2", NULL, "OUT6L" }, + { "SPKDAT2", NULL, "OUT6R" }, + { "PWM1", NULL, "PWM1 Driver" }, + { "PWM2", NULL, "PWM2 Driver" }, +}; + +static struct { + int reg; + int val; +} wm5100_reva_patches[] = { + { WM5100_AUDIO_IF_1_10, 0 }, + { WM5100_AUDIO_IF_1_11, 1 }, + { WM5100_AUDIO_IF_1_12, 2 }, + { WM5100_AUDIO_IF_1_13, 3 }, + { WM5100_AUDIO_IF_1_14, 4 }, + { WM5100_AUDIO_IF_1_15, 5 }, + { WM5100_AUDIO_IF_1_16, 6 }, + { WM5100_AUDIO_IF_1_17, 7 }, + + { WM5100_AUDIO_IF_1_18, 0 }, + { WM5100_AUDIO_IF_1_19, 1 }, + { WM5100_AUDIO_IF_1_20, 2 }, + { WM5100_AUDIO_IF_1_21, 3 }, + { WM5100_AUDIO_IF_1_22, 4 }, + { WM5100_AUDIO_IF_1_23, 5 }, + { WM5100_AUDIO_IF_1_24, 6 }, + { WM5100_AUDIO_IF_1_25, 7 }, + + { WM5100_AUDIO_IF_2_10, 0 }, + { WM5100_AUDIO_IF_2_11, 1 }, + + { WM5100_AUDIO_IF_2_18, 0 }, + { WM5100_AUDIO_IF_2_19, 1 }, + + { WM5100_AUDIO_IF_3_10, 0 }, + { WM5100_AUDIO_IF_3_11, 1 }, + + { WM5100_AUDIO_IF_3_18, 0 }, + { WM5100_AUDIO_IF_3_19, 1 }, +}; + +static int wm5100_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct wm5100_priv *wm5100 = snd_soc_codec_get_drvdata(codec); + int ret, i; + + switch (level) { + case SND_SOC_BIAS_ON: + break; + + case SND_SOC_BIAS_PREPARE: + break; + + case SND_SOC_BIAS_STANDBY: + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { + ret = regulator_bulk_enable(ARRAY_SIZE(wm5100->core_supplies), + wm5100->core_supplies); + if (ret != 0) { + dev_err(codec->dev, + "Failed to enable supplies: %d\n", + ret); + return ret; + } + + if (wm5100->pdata.ldo_ena) { + gpio_set_value_cansleep(wm5100->pdata.ldo_ena, + 1); + msleep(2); + } + + codec->cache_only = false; + + switch (wm5100->rev) { + case 0: + snd_soc_write(codec, 0x11, 0x3); + snd_soc_write(codec, 0x203, 0xc); + snd_soc_write(codec, 0x206, 0); + snd_soc_write(codec, 0x207, 0xf0); + snd_soc_write(codec, 0x208, 0x3c); + snd_soc_write(codec, 0x209, 0); + snd_soc_write(codec, 0x211, 0x20d8); + snd_soc_write(codec, 0x11, 0); + + for (i = 0; + i < ARRAY_SIZE(wm5100_reva_patches); + i++) + snd_soc_write(codec, + wm5100_reva_patches[i].reg, + wm5100_reva_patches[i].val); + break; + default: + break; + } + + snd_soc_cache_sync(codec); + } + break; + + case SND_SOC_BIAS_OFF: + if (wm5100->pdata.ldo_ena) + gpio_set_value_cansleep(wm5100->pdata.ldo_ena, 0); + regulator_bulk_disable(ARRAY_SIZE(wm5100->core_supplies), + wm5100->core_supplies); + break; + } + codec->dapm.bias_level = level; + + return 0; +} + +static int wm5100_dai_to_base(struct snd_soc_dai *dai) +{ + switch (dai->id) { + case 0: + return WM5100_AUDIO_IF_1_1 - 1; + case 1: + return WM5100_AUDIO_IF_2_1 - 1; + case 2: + return WM5100_AUDIO_IF_3_1 - 1; + default: + BUG(); + return -EINVAL; + } +} + +static int wm5100_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + int lrclk, bclk, mask, base; + + base = wm5100_dai_to_base(dai); + if (base < 0) + return base; + + lrclk = 0; + bclk = 0; + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_DSP_A: + mask = 0; + break; + case SND_SOC_DAIFMT_DSP_B: + mask = 1; + break; + case SND_SOC_DAIFMT_I2S: + mask = 2; + break; + case SND_SOC_DAIFMT_LEFT_J: + mask = 3; + break; + default: + dev_err(codec->dev, "Unsupported DAI format %d\n", + fmt & SND_SOC_DAIFMT_FORMAT_MASK); + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + break; + case SND_SOC_DAIFMT_CBS_CFM: + lrclk |= WM5100_AIF1TX_LRCLK_MSTR; + break; + case SND_SOC_DAIFMT_CBM_CFS: + bclk |= WM5100_AIF1_BCLK_MSTR; + break; + case SND_SOC_DAIFMT_CBM_CFM: + lrclk |= WM5100_AIF1TX_LRCLK_MSTR; + bclk |= WM5100_AIF1_BCLK_MSTR; + break; + default: + dev_err(codec->dev, "Unsupported master mode %d\n", + fmt & SND_SOC_DAIFMT_MASTER_MASK); + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_IF: + bclk |= WM5100_AIF1_BCLK_INV; + lrclk |= WM5100_AIF1TX_LRCLK_INV; + break; + case SND_SOC_DAIFMT_IB_NF: + bclk |= WM5100_AIF1_BCLK_INV; + break; + case SND_SOC_DAIFMT_NB_IF: + lrclk |= WM5100_AIF1TX_LRCLK_INV; + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, base + 1, WM5100_AIF1_BCLK_MSTR | + WM5100_AIF1_BCLK_INV, bclk); + snd_soc_update_bits(codec, base + 2, WM5100_AIF1TX_LRCLK_MSTR | + WM5100_AIF1TX_LRCLK_INV, lrclk); + snd_soc_update_bits(codec, base + 3, WM5100_AIF1TX_LRCLK_MSTR | + WM5100_AIF1TX_LRCLK_INV, lrclk); + snd_soc_update_bits(codec, base + 5, WM5100_AIF1_FMT_MASK, mask); + + return 0; +} + +#define WM5100_NUM_BCLK_RATES 19 + +static int wm5100_bclk_rates_dat[WM5100_NUM_BCLK_RATES] = { + 32000, + 48000, + 64000, + 96000, + 128000, + 192000, + 384000, + 512000, + 768000, + 1024000, + 1536000, + 2048000, + 3072000, + 4096000, + 6144000, + 8192000, + 12288000, + 24576000, +}; + +static int wm5100_bclk_rates_cd[WM5100_NUM_BCLK_RATES] = { + 29400, + 44100, + 58800, + 88200, + 117600, + 176400, + 235200, + 352800, + 470400, + 705600, + 940800, + 1411200, + 1881600, + 2882400, + 3763200, + 5644800, + 7526400, + 11289600, + 22579600, +}; + +static int wm5100_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct wm5100_priv *wm5100 = snd_soc_codec_get_drvdata(codec); + bool async = wm5100->aif_async[dai->id]; + int i, base, bclk, aif_rate, lrclk, wl, fl, sr; + int *bclk_rates; + + base = wm5100_dai_to_base(dai); + if (base < 0) + return base; + + /* Data sizes if not using TDM */ + wl = snd_pcm_format_width(params_format(params)); + if (wl < 0) + return wl; + fl = snd_soc_params_to_frame_size(params); + if (fl < 0) + return fl; + + dev_dbg(codec->dev, "Word length %d bits, frame length %d bits\n", + wl, fl); + + /* Target BCLK rate */ + bclk = snd_soc_params_to_bclk(params); + if (bclk < 0) + return bclk; + + /* Root for BCLK depends on SYS/ASYNCCLK */ + if (!async) { + aif_rate = wm5100->sysclk; + sr = wm5100_alloc_sr(codec, params_rate(params)); + if (sr < 0) + return sr; + } else { + /* If we're in ASYNCCLK set the ASYNC sample rate */ + aif_rate = wm5100->asyncclk; + sr = 3; + + for (i = 0; i < ARRAY_SIZE(wm5100_sr_code); i++) + if (params_rate(params) == wm5100_sr_code[i]) + break; + if (i == ARRAY_SIZE(wm5100_sr_code)) { + dev_err(codec->dev, "Invalid rate %dHzn", + params_rate(params)); + return -EINVAL; + } + + /* TODO: We should really check for symmetry */ + snd_soc_update_bits(codec, WM5100_CLOCKING_8, + WM5100_ASYNC_SAMPLE_RATE_MASK, i); + } + + if (!aif_rate) { + dev_err(codec->dev, "%s has no rate set\n", + async ? "ASYNCCLK" : "SYSCLK"); + return -EINVAL; + } + + dev_dbg(codec->dev, "Target BCLK is %dHz, using %dHz %s\n", + bclk, aif_rate, async ? "ASYNCCLK" : "SYSCLK"); + + if (aif_rate % 4000) + bclk_rates = wm5100_bclk_rates_cd; + else + bclk_rates = wm5100_bclk_rates_dat; + + for (i = 0; i < WM5100_NUM_BCLK_RATES; i++) + if (bclk_rates[i] >= bclk && (bclk_rates[i] % bclk == 0)) + break; + if (i == WM5100_NUM_BCLK_RATES) { + dev_err(codec->dev, + "No valid BCLK for %dHz found from %dHz %s\n", + bclk, aif_rate, async ? "ASYNCCLK" : "SYSCLK"); + return -EINVAL; + } + + bclk = i; + dev_dbg(codec->dev, "Setting %dHz BCLK\n", bclk_rates[bclk]); + snd_soc_update_bits(codec, base + 1, WM5100_AIF1_BCLK_FREQ_MASK, bclk); + + lrclk = bclk_rates[bclk] / params_rate(params); + dev_dbg(codec->dev, "Setting %dHz LRCLK\n", bclk_rates[bclk] / lrclk); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK || + wm5100->aif_symmetric[dai->id]) + snd_soc_update_bits(codec, base + 7, + WM5100_AIF1RX_BCPF_MASK, lrclk); + else + snd_soc_update_bits(codec, base + 6, + WM5100_AIF1TX_BCPF_MASK, lrclk); + + i = (wl << WM5100_AIF1TX_WL_SHIFT) | fl; + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + snd_soc_update_bits(codec, base + 9, + WM5100_AIF1RX_WL_MASK | + WM5100_AIF1RX_SLOT_LEN_MASK, i); + else + snd_soc_update_bits(codec, base + 8, + WM5100_AIF1TX_WL_MASK | + WM5100_AIF1TX_SLOT_LEN_MASK, i); + + snd_soc_update_bits(codec, base + 4, WM5100_AIF1_RATE_MASK, sr); + + return 0; +} + +static struct snd_soc_dai_ops wm5100_dai_ops = { + .set_fmt = wm5100_set_fmt, + .hw_params = wm5100_hw_params, +}; + +static int wm5100_set_sysclk(struct snd_soc_codec *codec, int clk_id, + int source, unsigned int freq, int dir) +{ + struct wm5100_priv *wm5100 = snd_soc_codec_get_drvdata(codec); + int *rate_store; + int fval, audio_rate, ret, reg; + + switch (clk_id) { + case WM5100_CLK_SYSCLK: + reg = WM5100_CLOCKING_3; + rate_store = &wm5100->sysclk; + break; + case WM5100_CLK_ASYNCCLK: + reg = WM5100_CLOCKING_7; + rate_store = &wm5100->asyncclk; + break; + case WM5100_CLK_32KHZ: + /* The 32kHz clock is slightly different to the others */ + switch (source) { + case WM5100_CLKSRC_MCLK1: + case WM5100_CLKSRC_MCLK2: + case WM5100_CLKSRC_SYSCLK: + snd_soc_update_bits(codec, WM5100_CLOCKING_1, + WM5100_CLK_32K_SRC_MASK, + source); + break; + default: + return -EINVAL; + } + return 0; + + case WM5100_CLK_AIF1: + case WM5100_CLK_AIF2: + case WM5100_CLK_AIF3: + /* Not real clocks, record which clock domain they're in */ + switch (source) { + case WM5100_CLKSRC_SYSCLK: + wm5100->aif_async[clk_id - 1] = false; + break; + case WM5100_CLKSRC_ASYNCCLK: + wm5100->aif_async[clk_id - 1] = true; + break; + default: + dev_err(codec->dev, "Invalid source %d\n", source); + return -EINVAL; + } + return 0; + + case WM5100_CLK_OPCLK: + switch (freq) { + case 5644800: + case 6144000: + snd_soc_update_bits(codec, WM5100_MISC_GPIO_1, + WM5100_OPCLK_SEL_MASK, 0); + break; + case 11289600: + case 12288000: + snd_soc_update_bits(codec, WM5100_MISC_GPIO_1, + WM5100_OPCLK_SEL_MASK, 0); + break; + case 22579200: + case 24576000: + snd_soc_update_bits(codec, WM5100_MISC_GPIO_1, + WM5100_OPCLK_SEL_MASK, 0); + break; + default: + dev_err(codec->dev, "Unsupported OPCLK %dHz\n", + freq); + return -EINVAL; + } + return 0; + + default: + dev_err(codec->dev, "Unknown clock %d\n", clk_id); + return -EINVAL; + } + + switch (source) { + case WM5100_CLKSRC_SYSCLK: + case WM5100_CLKSRC_ASYNCCLK: + dev_err(codec->dev, "Invalid source %d\n", source); + return -EINVAL; + } + + switch (freq) { + case 5644800: + case 6144000: + fval = 0; + break; + case 11289600: + case 12288000: + fval = 1; + break; + case 22579200: + case 2457600: + fval = 2; + break; + default: + dev_err(codec->dev, "Invalid clock rate: %d\n", freq); + return -EINVAL; + } + + switch (freq) { + case 5644800: + case 11289600: + case 22579200: + audio_rate = 44100; + break; + + case 6144000: + case 12288000: + case 2457600: + audio_rate = 48000; + break; + + default: + BUG(); + audio_rate = 0; + break; + } + + /* TODO: Check if MCLKs are in use and enable/disable pulls to + * match. + */ + + snd_soc_update_bits(codec, reg, WM5100_SYSCLK_FREQ_MASK | + WM5100_SYSCLK_SRC_MASK, + fval << WM5100_SYSCLK_FREQ_SHIFT | source); + + /* If this is SYSCLK then configure the clock rate for the + * internal audio functions to the natural sample rate for + * this clock rate. + */ + if (clk_id == WM5100_CLK_SYSCLK) { + dev_dbg(codec->dev, "Setting primary audio rate to %dHz", + audio_rate); + if (0 && *rate_store) + wm5100_free_sr(codec, audio_rate); + ret = wm5100_alloc_sr(codec, audio_rate); + if (ret != 0) + dev_warn(codec->dev, "Primary audio slot is %d\n", + ret); + } + + *rate_store = freq; + + return 0; +} + +struct _fll_div { + u16 fll_fratio; + u16 fll_outdiv; + u16 fll_refclk_div; + u16 n; + u16 theta; + u16 lambda; +}; + +static struct { + unsigned int min; + unsigned int max; + u16 fll_fratio; + int ratio; +} fll_fratios[] = { + { 0, 64000, 4, 16 }, + { 64000, 128000, 3, 8 }, + { 128000, 256000, 2, 4 }, + { 256000, 1000000, 1, 2 }, + { 1000000, 13500000, 0, 1 }, +}; + +static int fll_factors(struct _fll_div *fll_div, unsigned int Fref, + unsigned int Fout) +{ + unsigned int target; + unsigned int div; + unsigned int fratio, gcd_fll; + int i; + + /* Fref must be <=13.5MHz */ + div = 1; + fll_div->fll_refclk_div = 0; + while ((Fref / div) > 13500000) { + div *= 2; + fll_div->fll_refclk_div++; + + if (div > 8) { + pr_err("Can't scale %dMHz input down to <=13.5MHz\n", + Fref); + return -EINVAL; + } + } + + pr_debug("FLL Fref=%u Fout=%u\n", Fref, Fout); + + /* Apply the division for our remaining calculations */ + Fref /= div; + + /* Fvco should be 90-100MHz; don't check the upper bound */ + div = 2; + while (Fout * div < 90000000) { + div++; + if (div > 64) { + pr_err("Unable to find FLL_OUTDIV for Fout=%uHz\n", + Fout); + return -EINVAL; + } + } + target = Fout * div; + fll_div->fll_outdiv = div - 1; + + pr_debug("FLL Fvco=%dHz\n", target); + + /* Find an appropraite FLL_FRATIO and factor it out of the target */ + for (i = 0; i < ARRAY_SIZE(fll_fratios); i++) { + if (fll_fratios[i].min <= Fref && Fref <= fll_fratios[i].max) { + fll_div->fll_fratio = fll_fratios[i].fll_fratio; + fratio = fll_fratios[i].ratio; + break; + } + } + if (i == ARRAY_SIZE(fll_fratios)) { + pr_err("Unable to find FLL_FRATIO for Fref=%uHz\n", Fref); + return -EINVAL; + } + + fll_div->n = target / (fratio * Fref); + + if (target % Fref == 0) { + fll_div->theta = 0; + fll_div->lambda = 0; + } else { + gcd_fll = gcd(target, fratio * Fref); + + fll_div->theta = (target - (fll_div->n * fratio * Fref)) + / gcd_fll; + fll_div->lambda = (fratio * Fref) / gcd_fll; + } + + pr_debug("FLL N=%x THETA=%x LAMBDA=%x\n", + fll_div->n, fll_div->theta, fll_div->lambda); + pr_debug("FLL_FRATIO=%x(%d) FLL_OUTDIV=%x FLL_REFCLK_DIV=%x\n", + fll_div->fll_fratio, fratio, fll_div->fll_outdiv, + fll_div->fll_refclk_div); + + return 0; +} + +static int wm5100_set_fll(struct snd_soc_codec *codec, int fll_id, int source, + unsigned int Fref, unsigned int Fout) +{ + struct i2c_client *i2c = to_i2c_client(codec->dev); + struct wm5100_priv *wm5100 = snd_soc_codec_get_drvdata(codec); + struct _fll_div factors; + struct wm5100_fll *fll; + int ret, base, lock, i, timeout; + + switch (fll_id) { + case WM5100_FLL1: + fll = &wm5100->fll[0]; + base = WM5100_FLL1_CONTROL_1 - 1; + lock = WM5100_FLL1_LOCK_STS; + break; + case WM5100_FLL2: + fll = &wm5100->fll[1]; + base = WM5100_FLL2_CONTROL_2 - 1; + lock = WM5100_FLL2_LOCK_STS; + break; + default: + dev_err(codec->dev, "Unknown FLL %d\n",fll_id); + return -EINVAL; + } + + if (!Fout) { + dev_dbg(codec->dev, "FLL%d disabled", fll_id); + fll->fout = 0; + snd_soc_update_bits(codec, base + 1, WM5100_FLL1_ENA, 0); + return 0; + } + + switch (source) { + case WM5100_FLL_SRC_MCLK1: + case WM5100_FLL_SRC_MCLK2: + case WM5100_FLL_SRC_FLL1: + case WM5100_FLL_SRC_FLL2: + case WM5100_FLL_SRC_AIF1BCLK: + case WM5100_FLL_SRC_AIF2BCLK: + case WM5100_FLL_SRC_AIF3BCLK: + break; + default: + dev_err(codec->dev, "Invalid FLL source %d\n", source); + return -EINVAL; + } + + ret = fll_factors(&factors, Fref, Fout); + if (ret < 0) + return ret; + + /* Disable the FLL while we reconfigure */ + snd_soc_update_bits(codec, base + 1, WM5100_FLL1_ENA, 0); + + snd_soc_update_bits(codec, base + 2, + WM5100_FLL1_OUTDIV_MASK | WM5100_FLL1_FRATIO_MASK, + (factors.fll_outdiv << WM5100_FLL1_OUTDIV_SHIFT) | + factors.fll_fratio); + snd_soc_update_bits(codec, base + 3, WM5100_FLL1_THETA_MASK, + factors.theta); + snd_soc_update_bits(codec, base + 5, WM5100_FLL1_N_MASK, factors.n); + snd_soc_update_bits(codec, base + 6, + WM5100_FLL1_REFCLK_DIV_MASK | + WM5100_FLL1_REFCLK_SRC_MASK, + (factors.fll_refclk_div + << WM5100_FLL1_REFCLK_DIV_SHIFT) | source); + snd_soc_update_bits(codec, base + 7, WM5100_FLL1_LAMBDA_MASK, + factors.lambda); + + /* Clear any pending completions */ + try_wait_for_completion(&fll->lock); + + snd_soc_update_bits(codec, base + 1, WM5100_FLL1_ENA, WM5100_FLL1_ENA); + + if (i2c->irq) + timeout = 2; + else + timeout = 50; + + /* Poll for the lock; will use interrupt when we can test */ + for (i = 0; i < timeout; i++) { + if (i2c->irq) { + ret = wait_for_completion_timeout(&fll->lock, + msecs_to_jiffies(25)); + if (ret > 0) + break; + } else { + msleep(1); + } + + ret = snd_soc_read(codec, + WM5100_INTERRUPT_RAW_STATUS_3); + if (ret < 0) { + dev_err(codec->dev, + "Failed to read FLL status: %d\n", + ret); + continue; + } + if (ret & lock) + break; + } + if (i == timeout) { + dev_err(codec->dev, "FLL%d lock timed out\n", fll_id); + return -ETIMEDOUT; + } + + fll->src = source; + fll->fref = Fref; + fll->fout = Fout; + + dev_dbg(codec->dev, "FLL%d running %dHz->%dHz\n", fll_id, + Fref, Fout); + + return 0; +} + +/* Actually go much higher */ +#define WM5100_RATES SNDRV_PCM_RATE_8000_192000 + +#define WM5100_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_driver wm5100_dai[] = { + { + .name = "wm5100-aif1", + .playback = { + .stream_name = "AIF1 Playback", + .channels_min = 2, + .channels_max = 2, + .rates = WM5100_RATES, + .formats = WM5100_FORMATS, + }, + .capture = { + .stream_name = "AIF1 Capture", + .channels_min = 2, + .channels_max = 2, + .rates = WM5100_RATES, + .formats = WM5100_FORMATS, + }, + .ops = &wm5100_dai_ops, + }, + { + .name = "wm5100-aif2", + .id = 1, + .playback = { + .stream_name = "AIF2 Playback", + .channels_min = 2, + .channels_max = 2, + .rates = WM5100_RATES, + .formats = WM5100_FORMATS, + }, + .capture = { + .stream_name = "AIF2 Capture", + .channels_min = 2, + .channels_max = 2, + .rates = WM5100_RATES, + .formats = WM5100_FORMATS, + }, + .ops = &wm5100_dai_ops, + }, + { + .name = "wm5100-aif3", + .id = 2, + .playback = { + .stream_name = "AIF3 Playback", + .channels_min = 2, + .channels_max = 2, + .rates = WM5100_RATES, + .formats = WM5100_FORMATS, + }, + .capture = { + .stream_name = "AIF3 Capture", + .channels_min = 2, + .channels_max = 2, + .rates = WM5100_RATES, + .formats = WM5100_FORMATS, + }, + .ops = &wm5100_dai_ops, + }, +}; + +static int wm5100_dig_vu[] = { + WM5100_ADC_DIGITAL_VOLUME_1L, + WM5100_ADC_DIGITAL_VOLUME_1R, + WM5100_ADC_DIGITAL_VOLUME_2L, + WM5100_ADC_DIGITAL_VOLUME_2R, + WM5100_ADC_DIGITAL_VOLUME_3L, + WM5100_ADC_DIGITAL_VOLUME_3R, + WM5100_ADC_DIGITAL_VOLUME_4L, + WM5100_ADC_DIGITAL_VOLUME_4R, + + WM5100_DAC_DIGITAL_VOLUME_1L, + WM5100_DAC_DIGITAL_VOLUME_1R, + WM5100_DAC_DIGITAL_VOLUME_2L, + WM5100_DAC_DIGITAL_VOLUME_2R, + WM5100_DAC_DIGITAL_VOLUME_3L, + WM5100_DAC_DIGITAL_VOLUME_3R, + WM5100_DAC_DIGITAL_VOLUME_4L, + WM5100_DAC_DIGITAL_VOLUME_4R, + WM5100_DAC_DIGITAL_VOLUME_5L, + WM5100_DAC_DIGITAL_VOLUME_5R, + WM5100_DAC_DIGITAL_VOLUME_6L, + WM5100_DAC_DIGITAL_VOLUME_6R, +}; + +static irqreturn_t wm5100_irq(int irq, void *data) +{ + struct snd_soc_codec *codec = data; + struct wm5100_priv *wm5100 = snd_soc_codec_get_drvdata(codec); + irqreturn_t status = IRQ_NONE; + int irq_val; + + irq_val = snd_soc_read(codec, WM5100_INTERRUPT_STATUS_3); + if (irq_val < 0) { + dev_err(codec->dev, "Failed to read IRQ status 3: %d\n", + irq_val); + irq_val = 0; + } + irq_val &= ~snd_soc_read(codec, WM5100_INTERRUPT_STATUS_3_MASK); + + snd_soc_write(codec, WM5100_INTERRUPT_STATUS_3, irq_val); + + if (irq_val) + status = IRQ_HANDLED; + + wm5100_log_status3(codec, irq_val); + + if (irq_val & WM5100_FLL1_LOCK_EINT) { + dev_dbg(codec->dev, "FLL1 locked\n"); + complete(&wm5100->fll[0].lock); + } + if (irq_val & WM5100_FLL2_LOCK_EINT) { + dev_dbg(codec->dev, "FLL2 locked\n"); + complete(&wm5100->fll[1].lock); + } + + irq_val = snd_soc_read(codec, WM5100_INTERRUPT_STATUS_4); + if (irq_val < 0) { + dev_err(codec->dev, "Failed to read IRQ status 4: %d\n", + irq_val); + irq_val = 0; + } + irq_val &= ~snd_soc_read(codec, WM5100_INTERRUPT_STATUS_4_MASK); + + if (irq_val) + status = IRQ_HANDLED; + + snd_soc_write(codec, WM5100_INTERRUPT_STATUS_4, irq_val); + + wm5100_log_status4(codec, irq_val); + + return status; +} + +static irqreturn_t wm5100_edge_irq(int irq, void *data) +{ + irqreturn_t ret = IRQ_NONE; + irqreturn_t val; + + do { + val = wm5100_irq(irq, data); + if (val != IRQ_NONE) + ret = val; + } while (val != IRQ_NONE); + + return ret; +} + +#ifdef CONFIG_GPIOLIB +static inline struct wm5100_priv *gpio_to_wm5100(struct gpio_chip *chip) +{ + return container_of(chip, struct wm5100_priv, gpio_chip); +} + +static void wm5100_gpio_set(struct gpio_chip *chip, unsigned offset, int value) +{ + struct wm5100_priv *wm5100 = gpio_to_wm5100(chip); + struct snd_soc_codec *codec = wm5100->codec; + + snd_soc_update_bits(codec, WM5100_GPIO_CTRL_1 + offset, + WM5100_GP1_LVL, !!value << WM5100_GP1_LVL_SHIFT); +} + +static int wm5100_gpio_direction_out(struct gpio_chip *chip, + unsigned offset, int value) +{ + struct wm5100_priv *wm5100 = gpio_to_wm5100(chip); + struct snd_soc_codec *codec = wm5100->codec; + int val; + + val = (1 << WM5100_GP1_FN_SHIFT) | (!!value << WM5100_GP1_LVL_SHIFT); + + return snd_soc_update_bits(codec, WM5100_GPIO_CTRL_1 + offset, + WM5100_GP1_FN_MASK | WM5100_GP1_DIR | + WM5100_GP1_LVL, val); +} + +static int wm5100_gpio_get(struct gpio_chip *chip, unsigned offset) +{ + struct wm5100_priv *wm5100 = gpio_to_wm5100(chip); + struct snd_soc_codec *codec = wm5100->codec; + int ret; + + ret = snd_soc_read(codec, WM5100_GPIO_CTRL_1 + offset); + if (ret < 0) + return ret; + + return (ret & WM5100_GP1_LVL) != 0; +} + +static int wm5100_gpio_direction_in(struct gpio_chip *chip, unsigned offset) +{ + struct wm5100_priv *wm5100 = gpio_to_wm5100(chip); + struct snd_soc_codec *codec = wm5100->codec; + + return snd_soc_update_bits(codec, WM5100_GPIO_CTRL_1 + offset, + WM5100_GP1_FN_MASK | WM5100_GP1_DIR, + (1 << WM5100_GP1_FN_SHIFT) | + (1 << WM5100_GP1_DIR_SHIFT)); +} + +static struct gpio_chip wm5100_template_chip = { + .label = "wm5100", + .owner = THIS_MODULE, + .direction_output = wm5100_gpio_direction_out, + .set = wm5100_gpio_set, + .direction_input = wm5100_gpio_direction_in, + .get = wm5100_gpio_get, + .can_sleep = 1, +}; + +static void wm5100_init_gpio(struct snd_soc_codec *codec) +{ + struct wm5100_priv *wm5100 = snd_soc_codec_get_drvdata(codec); + int ret; + + wm5100->gpio_chip = wm5100_template_chip; + wm5100->gpio_chip.ngpio = 6; + wm5100->gpio_chip.dev = codec->dev; + + if (wm5100->pdata.gpio_base) + wm5100->gpio_chip.base = wm5100->pdata.gpio_base; + else + wm5100->gpio_chip.base = -1; + + ret = gpiochip_add(&wm5100->gpio_chip); + if (ret != 0) + dev_err(codec->dev, "Failed to add GPIOs: %d\n", ret); +} + +static void wm5100_free_gpio(struct snd_soc_codec *codec) +{ + struct wm5100_priv *wm5100 = snd_soc_codec_get_drvdata(codec); + int ret; + + ret = gpiochip_remove(&wm5100->gpio_chip); + if (ret != 0) + dev_err(codec->dev, "Failed to remove GPIOs: %d\n", ret); +} +#else +static void wm5100_init_gpio(struct snd_soc_codec *codec) +{ +} + +static void wm5100_free_gpio(struct snd_soc_codec *codec) +{ +} +#endif + +static int wm5100_probe(struct snd_soc_codec *codec) +{ + struct i2c_client *i2c = to_i2c_client(codec->dev); + struct wm5100_priv *wm5100 = snd_soc_codec_get_drvdata(codec); + int ret, i, irq_flags; + + wm5100->codec = codec; + + codec->dapm.bias_level = SND_SOC_BIAS_OFF; + + ret = snd_soc_codec_set_cache_io(codec, 16, 16, SND_SOC_I2C); + if (ret != 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } + + for (i = 0; i < ARRAY_SIZE(wm5100->core_supplies); i++) + wm5100->core_supplies[i].supply = wm5100_core_supply_names[i]; + + ret = regulator_bulk_get(&i2c->dev, ARRAY_SIZE(wm5100->core_supplies), + wm5100->core_supplies); + if (ret != 0) { + dev_err(codec->dev, "Failed to request core supplies: %d\n", + ret); + return ret; + } + + wm5100->cpvdd = regulator_get(&i2c->dev, "CPVDD"); + if (IS_ERR(wm5100->cpvdd)) { + ret = PTR_ERR(wm5100->cpvdd); + dev_err(&i2c->dev, "Failed to get CPVDD: %d\n", ret); + goto err_core; + } + + ret = regulator_bulk_enable(ARRAY_SIZE(wm5100->core_supplies), + wm5100->core_supplies); + if (ret != 0) { + dev_err(codec->dev, "Failed to enable core supplies: %d\n", + ret); + goto err_cpvdd; + } + + if (wm5100->pdata.ldo_ena) { + ret = gpio_request_one(wm5100->pdata.ldo_ena, + GPIOF_OUT_INIT_HIGH, "WM5100 LDOENA"); + if (ret < 0) { + dev_err(&i2c->dev, "Failed to request LDOENA %d: %d\n", + wm5100->pdata.ldo_ena, ret); + goto err_enable; + } + msleep(2); + } + + if (wm5100->pdata.reset) { + ret = gpio_request_one(wm5100->pdata.reset, + GPIOF_OUT_INIT_HIGH, "WM5100 /RESET"); + if (ret < 0) { + dev_err(&i2c->dev, "Failed to request /RESET %d: %d\n", + wm5100->pdata.reset, ret); + goto err_ldo; + } + } + + ret = snd_soc_read(codec, WM5100_SOFTWARE_RESET); + if (ret < 0) { + dev_err(codec->dev, "Failed to read ID register\n"); + goto err_reset; + } + switch (ret) { + case 0x8997: + case 0x5100: + break; + + default: + dev_err(codec->dev, "Device is not a WM5100, ID is %x\n", ret); + ret = -EINVAL; + goto err_reset; + } + + ret = snd_soc_read(codec, WM5100_DEVICE_REVISION); + if (ret < 0) { + dev_err(codec->dev, "Failed to read revision register\n"); + goto err_reset; + } + wm5100->rev = ret & WM5100_DEVICE_REVISION_MASK; + + dev_info(codec->dev, "revision %c\n", wm5100->rev + 'A'); + + ret = wm5100_reset(codec); + if (ret < 0) { + dev_err(codec->dev, "Failed to issue reset\n"); + goto err_reset; + } + + codec->cache_only = true; + + wm5100_init_gpio(codec); + + for (i = 0; i < ARRAY_SIZE(wm5100_dig_vu); i++) + snd_soc_update_bits(codec, wm5100_dig_vu[i], WM5100_OUT_VU, + WM5100_OUT_VU); + + for (i = 0; i < ARRAY_SIZE(wm5100->pdata.in_mode); i++) { + snd_soc_update_bits(codec, WM5100_IN1L_CONTROL, + WM5100_IN1_MODE_MASK | + WM5100_IN1_DMIC_SUP_MASK, + (wm5100->pdata.in_mode[i] << + WM5100_IN1_MODE_SHIFT) | + (wm5100->pdata.dmic_sup[i] << + WM5100_IN1_DMIC_SUP_SHIFT)); + } + + for (i = 0; i < ARRAY_SIZE(wm5100->pdata.gpio_defaults); i++) { + if (!wm5100->pdata.gpio_defaults[i]) + continue; + + snd_soc_write(codec, WM5100_GPIO_CTRL_1 + i, + wm5100->pdata.gpio_defaults[i]); + } + + /* Don't debounce interrupts to support use of SYSCLK only */ + snd_soc_write(codec, WM5100_IRQ_DEBOUNCE_1, 0); + snd_soc_write(codec, WM5100_IRQ_DEBOUNCE_2, 0); + + /* TODO: check if we're symmetric */ + + if (i2c->irq) { + if (wm5100->pdata.irq_flags) + irq_flags = wm5100->pdata.irq_flags; + else + irq_flags = IRQF_TRIGGER_LOW; + + irq_flags |= IRQF_ONESHOT; + + if (irq_flags & (IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING)) + ret = request_threaded_irq(i2c->irq, NULL, + wm5100_edge_irq, + irq_flags, "wm5100", codec); + else + ret = request_threaded_irq(i2c->irq, NULL, wm5100_irq, + irq_flags, "wm5100", codec); + + if (ret != 0) { + dev_err(codec->dev, "Failed to request IRQ %d: %d\n", + i2c->irq, ret); + } else { + /* Enable default interrupts */ + snd_soc_update_bits(codec, + WM5100_INTERRUPT_STATUS_3_MASK, + WM5100_IM_SPK_SHUTDOWN_WARN_EINT | + WM5100_IM_SPK_SHUTDOWN_EINT | + WM5100_IM_ASRC2_LOCK_EINT | + WM5100_IM_ASRC1_LOCK_EINT | + WM5100_IM_FLL2_LOCK_EINT | + WM5100_IM_FLL1_LOCK_EINT | + WM5100_CLKGEN_ERR_EINT | + WM5100_CLKGEN_ERR_ASYNC_EINT, 0); + + snd_soc_update_bits(codec, + WM5100_INTERRUPT_STATUS_4_MASK, + WM5100_AIF3_ERR_EINT | + WM5100_AIF2_ERR_EINT | + WM5100_AIF1_ERR_EINT | + WM5100_CTRLIF_ERR_EINT | + WM5100_ISRC2_UNDERCLOCKED_EINT | + WM5100_ISRC1_UNDERCLOCKED_EINT | + WM5100_FX_UNDERCLOCKED_EINT | + WM5100_AIF3_UNDERCLOCKED_EINT | + WM5100_AIF2_UNDERCLOCKED_EINT | + WM5100_AIF1_UNDERCLOCKED_EINT | + WM5100_ASRC_UNDERCLOCKED_EINT | + WM5100_DAC_UNDERCLOCKED_EINT | + WM5100_ADC_UNDERCLOCKED_EINT | + WM5100_MIXER_UNDERCLOCKED_EINT, 0); + } + } else { + snd_soc_dapm_new_controls(&codec->dapm, + wm5100_dapm_widgets_noirq, + ARRAY_SIZE(wm5100_dapm_widgets_noirq)); + } + + if (wm5100->pdata.hp_pol) { + ret = gpio_request_one(wm5100->pdata.hp_pol, + GPIOF_OUT_INIT_HIGH, "WM5100 HP_POL"); + if (ret < 0) { + dev_err(&i2c->dev, "Failed to request HP_POL %d: %d\n", + wm5100->pdata.hp_pol, ret); + goto err_gpio; + } + } + + /* We'll get woken up again when the system has something useful + * for us to do. + */ + if (wm5100->pdata.ldo_ena) + gpio_set_value_cansleep(wm5100->pdata.ldo_ena, 0); + regulator_bulk_disable(ARRAY_SIZE(wm5100->core_supplies), + wm5100->core_supplies); + + return 0; + +err_gpio: + wm5100_free_gpio(codec); +err_reset: + if (wm5100->pdata.reset) { + gpio_set_value_cansleep(wm5100->pdata.reset, 1); + gpio_free(wm5100->pdata.reset); + } +err_ldo: + if (wm5100->pdata.ldo_ena) { + gpio_set_value_cansleep(wm5100->pdata.ldo_ena, 0); + gpio_free(wm5100->pdata.ldo_ena); + } +err_enable: + regulator_bulk_disable(ARRAY_SIZE(wm5100->core_supplies), + wm5100->core_supplies); +err_cpvdd: + regulator_put(wm5100->cpvdd); +err_core: + regulator_bulk_free(ARRAY_SIZE(wm5100->core_supplies), + wm5100->core_supplies); + + return ret; +} + +static int wm5100_remove(struct snd_soc_codec *codec) +{ + struct wm5100_priv *wm5100 = snd_soc_codec_get_drvdata(codec); + + wm5100_set_bias_level(codec, SND_SOC_BIAS_OFF); + if (wm5100->pdata.hp_pol) { + gpio_free(wm5100->pdata.hp_pol); + } + wm5100_free_gpio(codec); + if (wm5100->pdata.reset) { + gpio_set_value_cansleep(wm5100->pdata.reset, 1); + gpio_free(wm5100->pdata.reset); + } + if (wm5100->pdata.ldo_ena) { + gpio_set_value_cansleep(wm5100->pdata.ldo_ena, 0); + gpio_free(wm5100->pdata.ldo_ena); + } + regulator_put(wm5100->cpvdd); + regulator_bulk_free(ARRAY_SIZE(wm5100->core_supplies), + wm5100->core_supplies); + return 0; +} + +static struct snd_soc_codec_driver soc_codec_dev_wm5100 = { + .probe = wm5100_probe, + .remove = wm5100_remove, + + .set_sysclk = wm5100_set_sysclk, + .set_pll = wm5100_set_fll, + .set_bias_level = wm5100_set_bias_level, + .idle_bias_off = 1, + + .seq_notifier = wm5100_seq_notifier, + .controls = wm5100_snd_controls, + .num_controls = ARRAY_SIZE(wm5100_snd_controls), + .dapm_widgets = wm5100_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(wm5100_dapm_widgets), + .dapm_routes = wm5100_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(wm5100_dapm_routes), + + .reg_cache_size = ARRAY_SIZE(wm5100_reg_defaults), + .reg_word_size = sizeof(u16), + .compress_type = SND_SOC_RBTREE_COMPRESSION, + .reg_cache_default = wm5100_reg_defaults, + + .volatile_register = wm5100_volatile_register, + .readable_register = wm5100_readable_register, +}; + +static __devinit int wm5100_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct wm5100_pdata *pdata = dev_get_platdata(&i2c->dev); + struct wm5100_priv *wm5100; + int ret, i; + + wm5100 = kzalloc(sizeof(struct wm5100_priv), GFP_KERNEL); + if (wm5100 == NULL) + return -ENOMEM; + + for (i = 0; i < ARRAY_SIZE(wm5100->fll); i++) + init_completion(&wm5100->fll[i].lock); + + if (pdata) + wm5100->pdata = *pdata; + + i2c_set_clientdata(i2c, wm5100); + + ret = snd_soc_register_codec(&i2c->dev, + &soc_codec_dev_wm5100, wm5100_dai, + ARRAY_SIZE(wm5100_dai)); + if (ret < 0) { + dev_err(&i2c->dev, "Failed to register WM5100: %d\n", ret); + kfree(wm5100); + } + + return ret; +} + +static __devexit int wm5100_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + kfree(i2c_get_clientdata(client)); + return 0; +} + +static const struct i2c_device_id wm5100_i2c_id[] = { + { "wm5100", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, wm5100_i2c_id); + +static struct i2c_driver wm5100_i2c_driver = { + .driver = { + .name = "wm5100", + .owner = THIS_MODULE, + }, + .probe = wm5100_i2c_probe, + .remove = __devexit_p(wm5100_i2c_remove), + .id_table = wm5100_i2c_id, +}; + +static int __init wm5100_modinit(void) +{ + return i2c_add_driver(&wm5100_i2c_driver); +} +module_init(wm5100_modinit); + +static void __exit wm5100_exit(void) +{ + i2c_del_driver(&wm5100_i2c_driver); +} +module_exit(wm5100_exit); + +MODULE_DESCRIPTION("ASoC WM5100 driver"); +MODULE_AUTHOR("Mark Brown "); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/wm5100.h b/sound/soc/codecs/wm5100.h new file mode 100644 index 0000000..345b3cf --- /dev/null +++ b/sound/soc/codecs/wm5100.h @@ -0,0 +1,5146 @@ +/* + * wm5100.h -- WM5100 ALSA SoC Audio driver + * + * Copyright 2011 Wolfson Microelectronics plc + * + * Author: Mark Brown + * + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef WM5100_ASOC_H +#define WM5100_ASOC_H + +#include + +#define WM5100_CLK_AIF1 1 +#define WM5100_CLK_AIF2 2 +#define WM5100_CLK_AIF3 3 +#define WM5100_CLK_SYSCLK 4 +#define WM5100_CLK_ASYNCCLK 5 +#define WM5100_CLK_32KHZ 6 +#define WM5100_CLK_OPCLK 7 + +#define WM5100_CLKSRC_MCLK1 0 +#define WM5100_CLKSRC_MCLK2 1 +#define WM5100_CLKSRC_SYSCLK 2 +#define WM5100_CLKSRC_FLL1 4 +#define WM5100_CLKSRC_FLL2 5 +#define WM5100_CLKSRC_AIF1BCLK 8 +#define WM5100_CLKSRC_AIF2BCLK 9 +#define WM5100_CLKSRC_AIF3BCLK 10 +#define WM5100_CLKSRC_ASYNCCLK 0x100 + +#define WM5100_FLL1 1 +#define WM5100_FLL2 2 + +#define WM5100_FLL_SRC_MCLK1 0x0 +#define WM5100_FLL_SRC_MCLK2 0x1 +#define WM5100_FLL_SRC_FLL1 0x4 +#define WM5100_FLL_SRC_FLL2 0x5 +#define WM5100_FLL_SRC_AIF1BCLK 0x8 +#define WM5100_FLL_SRC_AIF2BCLK 0x9 +#define WM5100_FLL_SRC_AIF3BCLK 0xa + +/* + * Register values. + */ +#define WM5100_SOFTWARE_RESET 0x00 +#define WM5100_DEVICE_REVISION 0x01 +#define WM5100_CTRL_IF_1 0x10 +#define WM5100_TONE_GENERATOR_1 0x20 +#define WM5100_PWM_DRIVE_1 0x30 +#define WM5100_PWM_DRIVE_2 0x31 +#define WM5100_PWM_DRIVE_3 0x32 +#define WM5100_CLOCKING_1 0x100 +#define WM5100_CLOCKING_3 0x101 +#define WM5100_CLOCKING_4 0x102 +#define WM5100_CLOCKING_5 0x103 +#define WM5100_CLOCKING_6 0x104 +#define WM5100_CLOCKING_7 0x107 +#define WM5100_CLOCKING_8 0x108 +#define WM5100_ASRC_ENABLE 0x120 +#define WM5100_ASRC_STATUS 0x121 +#define WM5100_ASRC_RATE1 0x122 +#define WM5100_ISRC_1_CTRL_1 0x141 +#define WM5100_ISRC_1_CTRL_2 0x142 +#define WM5100_ISRC_2_CTRL1 0x143 +#define WM5100_ISRC_2_CTRL_2 0x144 +#define WM5100_FLL1_CONTROL_1 0x182 +#define WM5100_FLL1_CONTROL_2 0x183 +#define WM5100_FLL1_CONTROL_3 0x184 +#define WM5100_FLL1_CONTROL_5 0x186 +#define WM5100_FLL1_CONTROL_6 0x187 +#define WM5100_FLL1_EFS_1 0x188 +#define WM5100_FLL2_CONTROL_1 0x1A2 +#define WM5100_FLL2_CONTROL_2 0x1A3 +#define WM5100_FLL2_CONTROL_3 0x1A4 +#define WM5100_FLL2_CONTROL_5 0x1A6 +#define WM5100_FLL2_CONTROL_6 0x1A7 +#define WM5100_FLL2_EFS_1 0x1A8 +#define WM5100_MIC_CHARGE_PUMP_1 0x200 +#define WM5100_MIC_CHARGE_PUMP_2 0x201 +#define WM5100_HP_CHARGE_PUMP_1 0x202 +#define WM5100_LDO1_CONTROL 0x211 +#define WM5100_MIC_BIAS_CTRL_1 0x215 +#define WM5100_MIC_BIAS_CTRL_2 0x216 +#define WM5100_MIC_BIAS_CTRL_3 0x217 +#define WM5100_ACCESSORY_DETECT_MODE_1 0x280 +#define WM5100_HEADPHONE_DETECT_1 0x288 +#define WM5100_HEADPHONE_DETECT_2 0x289 +#define WM5100_MIC_DETECT_1 0x290 +#define WM5100_MIC_DETECT_2 0x291 +#define WM5100_MIC_DETECT_3 0x292 +#define WM5100_INPUT_ENABLES 0x301 +#define WM5100_INPUT_ENABLES_STATUS 0x302 +#define WM5100_IN1L_CONTROL 0x310 +#define WM5100_IN1R_CONTROL 0x311 +#define WM5100_IN2L_CONTROL 0x312 +#define WM5100_IN2R_CONTROL 0x313 +#define WM5100_IN3L_CONTROL 0x314 +#define WM5100_IN3R_CONTROL 0x315 +#define WM5100_IN4L_CONTROL 0x316 +#define WM5100_IN4R_CONTROL 0x317 +#define WM5100_RXANC_SRC 0x318 +#define WM5100_INPUT_VOLUME_RAMP 0x319 +#define WM5100_ADC_DIGITAL_VOLUME_1L 0x320 +#define WM5100_ADC_DIGITAL_VOLUME_1R 0x321 +#define WM5100_ADC_DIGITAL_VOLUME_2L 0x322 +#define WM5100_ADC_DIGITAL_VOLUME_2R 0x323 +#define WM5100_ADC_DIGITAL_VOLUME_3L 0x324 +#define WM5100_ADC_DIGITAL_VOLUME_3R 0x325 +#define WM5100_ADC_DIGITAL_VOLUME_4L 0x326 +#define WM5100_ADC_DIGITAL_VOLUME_4R 0x327 +#define WM5100_OUTPUT_ENABLES_2 0x401 +#define WM5100_OUTPUT_STATUS_1 0x402 +#define WM5100_OUTPUT_STATUS_2 0x403 +#define WM5100_CHANNEL_ENABLES_1 0x408 +#define WM5100_OUT_VOLUME_1L 0x410 +#define WM5100_OUT_VOLUME_1R 0x411 +#define WM5100_DAC_VOLUME_LIMIT_1L 0x412 +#define WM5100_DAC_VOLUME_LIMIT_1R 0x413 +#define WM5100_OUT_VOLUME_2L 0x414 +#define WM5100_OUT_VOLUME_2R 0x415 +#define WM5100_DAC_VOLUME_LIMIT_2L 0x416 +#define WM5100_DAC_VOLUME_LIMIT_2R 0x417 +#define WM5100_OUT_VOLUME_3L 0x418 +#define WM5100_OUT_VOLUME_3R 0x419 +#define WM5100_DAC_VOLUME_LIMIT_3L 0x41A +#define WM5100_DAC_VOLUME_LIMIT_3R 0x41B +#define WM5100_OUT_VOLUME_4L 0x41C +#define WM5100_OUT_VOLUME_4R 0x41D +#define WM5100_DAC_VOLUME_LIMIT_5L 0x41E +#define WM5100_DAC_VOLUME_LIMIT_5R 0x41F +#define WM5100_DAC_VOLUME_LIMIT_6L 0x420 +#define WM5100_DAC_VOLUME_LIMIT_6R 0x421 +#define WM5100_DAC_AEC_CONTROL_1 0x440 +#define WM5100_OUTPUT_VOLUME_RAMP 0x441 +#define WM5100_DAC_DIGITAL_VOLUME_1L 0x480 +#define WM5100_DAC_DIGITAL_VOLUME_1R 0x481 +#define WM5100_DAC_DIGITAL_VOLUME_2L 0x482 +#define WM5100_DAC_DIGITAL_VOLUME_2R 0x483 +#define WM5100_DAC_DIGITAL_VOLUME_3L 0x484 +#define WM5100_DAC_DIGITAL_VOLUME_3R 0x485 +#define WM5100_DAC_DIGITAL_VOLUME_4L 0x486 +#define WM5100_DAC_DIGITAL_VOLUME_4R 0x487 +#define WM5100_DAC_DIGITAL_VOLUME_5L 0x488 +#define WM5100_DAC_DIGITAL_VOLUME_5R 0x489 +#define WM5100_DAC_DIGITAL_VOLUME_6L 0x48A +#define WM5100_DAC_DIGITAL_VOLUME_6R 0x48B +#define WM5100_PDM_SPK1_CTRL_1 0x4C0 +#define WM5100_PDM_SPK1_CTRL_2 0x4C1 +#define WM5100_PDM_SPK2_CTRL_1 0x4C2 +#define WM5100_PDM_SPK2_CTRL_2 0x4C3 +#define WM5100_AUDIO_IF_1_1 0x500 +#define WM5100_AUDIO_IF_1_2 0x501 +#define WM5100_AUDIO_IF_1_3 0x502 +#define WM5100_AUDIO_IF_1_4 0x503 +#define WM5100_AUDIO_IF_1_5 0x504 +#define WM5100_AUDIO_IF_1_6 0x505 +#define WM5100_AUDIO_IF_1_7 0x506 +#define WM5100_AUDIO_IF_1_8 0x507 +#define WM5100_AUDIO_IF_1_9 0x508 +#define WM5100_AUDIO_IF_1_10 0x509 +#define WM5100_AUDIO_IF_1_11 0x50A +#define WM5100_AUDIO_IF_1_12 0x50B +#define WM5100_AUDIO_IF_1_13 0x50C +#define WM5100_AUDIO_IF_1_14 0x50D +#define WM5100_AUDIO_IF_1_15 0x50E +#define WM5100_AUDIO_IF_1_16 0x50F +#define WM5100_AUDIO_IF_1_17 0x510 +#define WM5100_AUDIO_IF_1_18 0x511 +#define WM5100_AUDIO_IF_1_19 0x512 +#define WM5100_AUDIO_IF_1_20 0x513 +#define WM5100_AUDIO_IF_1_21 0x514 +#define WM5100_AUDIO_IF_1_22 0x515 +#define WM5100_AUDIO_IF_1_23 0x516 +#define WM5100_AUDIO_IF_1_24 0x517 +#define WM5100_AUDIO_IF_1_25 0x518 +#define WM5100_AUDIO_IF_1_26 0x519 +#define WM5100_AUDIO_IF_1_27 0x51A +#define WM5100_AUDIO_IF_2_1 0x540 +#define WM5100_AUDIO_IF_2_2 0x541 +#define WM5100_AUDIO_IF_2_3 0x542 +#define WM5100_AUDIO_IF_2_4 0x543 +#define WM5100_AUDIO_IF_2_5 0x544 +#define WM5100_AUDIO_IF_2_6 0x545 +#define WM5100_AUDIO_IF_2_7 0x546 +#define WM5100_AUDIO_IF_2_8 0x547 +#define WM5100_AUDIO_IF_2_9 0x548 +#define WM5100_AUDIO_IF_2_10 0x549 +#define WM5100_AUDIO_IF_2_11 0x54A +#define WM5100_AUDIO_IF_2_18 0x551 +#define WM5100_AUDIO_IF_2_19 0x552 +#define WM5100_AUDIO_IF_2_26 0x559 +#define WM5100_AUDIO_IF_2_27 0x55A +#define WM5100_AUDIO_IF_3_1 0x580 +#define WM5100_AUDIO_IF_3_2 0x581 +#define WM5100_AUDIO_IF_3_3 0x582 +#define WM5100_AUDIO_IF_3_4 0x583 +#define WM5100_AUDIO_IF_3_5 0x584 +#define WM5100_AUDIO_IF_3_6 0x585 +#define WM5100_AUDIO_IF_3_7 0x586 +#define WM5100_AUDIO_IF_3_8 0x587 +#define WM5100_AUDIO_IF_3_9 0x588 +#define WM5100_AUDIO_IF_3_10 0x589 +#define WM5100_AUDIO_IF_3_11 0x58A +#define WM5100_AUDIO_IF_3_18 0x591 +#define WM5100_AUDIO_IF_3_19 0x592 +#define WM5100_AUDIO_IF_3_26 0x599 +#define WM5100_AUDIO_IF_3_27 0x59A +#define WM5100_PWM1MIX_INPUT_1_SOURCE 0x640 +#define WM5100_PWM1MIX_INPUT_1_VOLUME 0x641 +#define WM5100_PWM1MIX_INPUT_2_SOURCE 0x642 +#define WM5100_PWM1MIX_INPUT_2_VOLUME 0x643 +#define WM5100_PWM1MIX_INPUT_3_SOURCE 0x644 +#define WM5100_PWM1MIX_INPUT_3_VOLUME 0x645 +#define WM5100_PWM1MIX_INPUT_4_SOURCE 0x646 +#define WM5100_PWM1MIX_INPUT_4_VOLUME 0x647 +#define WM5100_PWM2MIX_INPUT_1_SOURCE 0x648 +#define WM5100_PWM2MIX_INPUT_1_VOLUME 0x649 +#define WM5100_PWM2MIX_INPUT_2_SOURCE 0x64A +#define WM5100_PWM2MIX_INPUT_2_VOLUME 0x64B +#define WM5100_PWM2MIX_INPUT_3_SOURCE 0x64C +#define WM5100_PWM2MIX_INPUT_3_VOLUME 0x64D +#define WM5100_PWM2MIX_INPUT_4_SOURCE 0x64E +#define WM5100_PWM2MIX_INPUT_4_VOLUME 0x64F +#define WM5100_OUT1LMIX_INPUT_1_SOURCE 0x680 +#define WM5100_OUT1LMIX_INPUT_1_VOLUME 0x681 +#define WM5100_OUT1LMIX_INPUT_2_SOURCE 0x682 +#define WM5100_OUT1LMIX_INPUT_2_VOLUME 0x683 +#define WM5100_OUT1LMIX_INPUT_3_SOURCE 0x684 +#define WM5100_OUT1LMIX_INPUT_3_VOLUME 0x685 +#define WM5100_OUT1LMIX_INPUT_4_SOURCE 0x686 +#define WM5100_OUT1LMIX_INPUT_4_VOLUME 0x687 +#define WM5100_OUT1RMIX_INPUT_1_SOURCE 0x688 +#define WM5100_OUT1RMIX_INPUT_1_VOLUME 0x689 +#define WM5100_OUT1RMIX_INPUT_2_SOURCE 0x68A +#define WM5100_OUT1RMIX_INPUT_2_VOLUME 0x68B +#define WM5100_OUT1RMIX_INPUT_3_SOURCE 0x68C +#define WM5100_OUT1RMIX_INPUT_3_VOLUME 0x68D +#define WM5100_OUT1RMIX_INPUT_4_SOURCE 0x68E +#define WM5100_OUT1RMIX_INPUT_4_VOLUME 0x68F +#define WM5100_OUT2LMIX_INPUT_1_SOURCE 0x690 +#define WM5100_OUT2LMIX_INPUT_1_VOLUME 0x691 +#define WM5100_OUT2LMIX_INPUT_2_SOURCE 0x692 +#define WM5100_OUT2LMIX_INPUT_2_VOLUME 0x693 +#define WM5100_OUT2LMIX_INPUT_3_SOURCE 0x694 +#define WM5100_OUT2LMIX_INPUT_3_VOLUME 0x695 +#define WM5100_OUT2LMIX_INPUT_4_SOURCE 0x696 +#define WM5100_OUT2LMIX_INPUT_4_VOLUME 0x697 +#define WM5100_OUT2RMIX_INPUT_1_SOURCE 0x698 +#define WM5100_OUT2RMIX_INPUT_1_VOLUME 0x699 +#define WM5100_OUT2RMIX_INPUT_2_SOURCE 0x69A +#define WM5100_OUT2RMIX_INPUT_2_VOLUME 0x69B +#define WM5100_OUT2RMIX_INPUT_3_SOURCE 0x69C +#define WM5100_OUT2RMIX_INPUT_3_VOLUME 0x69D +#define WM5100_OUT2RMIX_INPUT_4_SOURCE 0x69E +#define WM5100_OUT2RMIX_INPUT_4_VOLUME 0x69F +#define WM5100_OUT3LMIX_INPUT_1_SOURCE 0x6A0 +#define WM5100_OUT3LMIX_INPUT_1_VOLUME 0x6A1 +#define WM5100_OUT3LMIX_INPUT_2_SOURCE 0x6A2 +#define WM5100_OUT3LMIX_INPUT_2_VOLUME 0x6A3 +#define WM5100_OUT3LMIX_INPUT_3_SOURCE 0x6A4 +#define WM5100_OUT3LMIX_INPUT_3_VOLUME 0x6A5 +#define WM5100_OUT3LMIX_INPUT_4_SOURCE 0x6A6 +#define WM5100_OUT3LMIX_INPUT_4_VOLUME 0x6A7 +#define WM5100_OUT3RMIX_INPUT_1_SOURCE 0x6A8 +#define WM5100_OUT3RMIX_INPUT_1_VOLUME 0x6A9 +#define WM5100_OUT3RMIX_INPUT_2_SOURCE 0x6AA +#define WM5100_OUT3RMIX_INPUT_2_VOLUME 0x6AB +#define WM5100_OUT3RMIX_INPUT_3_SOURCE 0x6AC +#define WM5100_OUT3RMIX_INPUT_3_VOLUME 0x6AD +#define WM5100_OUT3RMIX_INPUT_4_SOURCE 0x6AE +#define WM5100_OUT3RMIX_INPUT_4_VOLUME 0x6AF +#define WM5100_OUT4LMIX_INPUT_1_SOURCE 0x6B0 +#define WM5100_OUT4LMIX_INPUT_1_VOLUME 0x6B1 +#define WM5100_OUT4LMIX_INPUT_2_SOURCE 0x6B2 +#define WM5100_OUT4LMIX_INPUT_2_VOLUME 0x6B3 +#define WM5100_OUT4LMIX_INPUT_3_SOURCE 0x6B4 +#define WM5100_OUT4LMIX_INPUT_3_VOLUME 0x6B5 +#define WM5100_OUT4LMIX_INPUT_4_SOURCE 0x6B6 +#define WM5100_OUT4LMIX_INPUT_4_VOLUME 0x6B7 +#define WM5100_OUT4RMIX_INPUT_1_SOURCE 0x6B8 +#define WM5100_OUT4RMIX_INPUT_1_VOLUME 0x6B9 +#define WM5100_OUT4RMIX_INPUT_2_SOURCE 0x6BA +#define WM5100_OUT4RMIX_INPUT_2_VOLUME 0x6BB +#define WM5100_OUT4RMIX_INPUT_3_SOURCE 0x6BC +#define WM5100_OUT4RMIX_INPUT_3_VOLUME 0x6BD +#define WM5100_OUT4RMIX_INPUT_4_SOURCE 0x6BE +#define WM5100_OUT4RMIX_INPUT_4_VOLUME 0x6BF +#define WM5100_OUT5LMIX_INPUT_1_SOURCE 0x6C0 +#define WM5100_OUT5LMIX_INPUT_1_VOLUME 0x6C1 +#define WM5100_OUT5LMIX_INPUT_2_SOURCE 0x6C2 +#define WM5100_OUT5LMIX_INPUT_2_VOLUME 0x6C3 +#define WM5100_OUT5LMIX_INPUT_3_SOURCE 0x6C4 +#define WM5100_OUT5LMIX_INPUT_3_VOLUME 0x6C5 +#define WM5100_OUT5LMIX_INPUT_4_SOURCE 0x6C6 +#define WM5100_OUT5LMIX_INPUT_4_VOLUME 0x6C7 +#define WM5100_OUT5RMIX_INPUT_1_SOURCE 0x6C8 +#define WM5100_OUT5RMIX_INPUT_1_VOLUME 0x6C9 +#define WM5100_OUT5RMIX_INPUT_2_SOURCE 0x6CA +#define WM5100_OUT5RMIX_INPUT_2_VOLUME 0x6CB +#define WM5100_OUT5RMIX_INPUT_3_SOURCE 0x6CC +#define WM5100_OUT5RMIX_INPUT_3_VOLUME 0x6CD +#define WM5100_OUT5RMIX_INPUT_4_SOURCE 0x6CE +#define WM5100_OUT5RMIX_INPUT_4_VOLUME 0x6CF +#define WM5100_OUT6LMIX_INPUT_1_SOURCE 0x6D0 +#define WM5100_OUT6LMIX_INPUT_1_VOLUME 0x6D1 +#define WM5100_OUT6LMIX_INPUT_2_SOURCE 0x6D2 +#define WM5100_OUT6LMIX_INPUT_2_VOLUME 0x6D3 +#define WM5100_OUT6LMIX_INPUT_3_SOURCE 0x6D4 +#define WM5100_OUT6LMIX_INPUT_3_VOLUME 0x6D5 +#define WM5100_OUT6LMIX_INPUT_4_SOURCE 0x6D6 +#define WM5100_OUT6LMIX_INPUT_4_VOLUME 0x6D7 +#define WM5100_OUT6RMIX_INPUT_1_SOURCE 0x6D8 +#define WM5100_OUT6RMIX_INPUT_1_VOLUME 0x6D9 +#define WM5100_OUT6RMIX_INPUT_2_SOURCE 0x6DA +#define WM5100_OUT6RMIX_INPUT_2_VOLUME 0x6DB +#define WM5100_OUT6RMIX_INPUT_3_SOURCE 0x6DC +#define WM5100_OUT6RMIX_INPUT_3_VOLUME 0x6DD +#define WM5100_OUT6RMIX_INPUT_4_SOURCE 0x6DE +#define WM5100_OUT6RMIX_INPUT_4_VOLUME 0x6DF +#define WM5100_AIF1TX1MIX_INPUT_1_SOURCE 0x700 +#define WM5100_AIF1TX1MIX_INPUT_1_VOLUME 0x701 +#define WM5100_AIF1TX1MIX_INPUT_2_SOURCE 0x702 +#define WM5100_AIF1TX1MIX_INPUT_2_VOLUME 0x703 +#define WM5100_AIF1TX1MIX_INPUT_3_SOURCE 0x704 +#define WM5100_AIF1TX1MIX_INPUT_3_VOLUME 0x705 +#define WM5100_AIF1TX1MIX_INPUT_4_SOURCE 0x706 +#define WM5100_AIF1TX1MIX_INPUT_4_VOLUME 0x707 +#define WM5100_AIF1TX2MIX_INPUT_1_SOURCE 0x708 +#define WM5100_AIF1TX2MIX_INPUT_1_VOLUME 0x709 +#define WM5100_AIF1TX2MIX_INPUT_2_SOURCE 0x70A +#define WM5100_AIF1TX2MIX_INPUT_2_VOLUME 0x70B +#define WM5100_AIF1TX2MIX_INPUT_3_SOURCE 0x70C +#define WM5100_AIF1TX2MIX_INPUT_3_VOLUME 0x70D +#define WM5100_AIF1TX2MIX_INPUT_4_SOURCE 0x70E +#define WM5100_AIF1TX2MIX_INPUT_4_VOLUME 0x70F +#define WM5100_AIF1TX3MIX_INPUT_1_SOURCE 0x710 +#define WM5100_AIF1TX3MIX_INPUT_1_VOLUME 0x711 +#define WM5100_AIF1TX3MIX_INPUT_2_SOURCE 0x712 +#define WM5100_AIF1TX3MIX_INPUT_2_VOLUME 0x713 +#define WM5100_AIF1TX3MIX_INPUT_3_SOURCE 0x714 +#define WM5100_AIF1TX3MIX_INPUT_3_VOLUME 0x715 +#define WM5100_AIF1TX3MIX_INPUT_4_SOURCE 0x716 +#define WM5100_AIF1TX3MIX_INPUT_4_VOLUME 0x717 +#define WM5100_AIF1TX4MIX_INPUT_1_SOURCE 0x718 +#define WM5100_AIF1TX4MIX_INPUT_1_VOLUME 0x719 +#define WM5100_AIF1TX4MIX_INPUT_2_SOURCE 0x71A +#define WM5100_AIF1TX4MIX_INPUT_2_VOLUME 0x71B +#define WM5100_AIF1TX4MIX_INPUT_3_SOURCE 0x71C +#define WM5100_AIF1TX4MIX_INPUT_3_VOLUME 0x71D +#define WM5100_AIF1TX4MIX_INPUT_4_SOURCE 0x71E +#define WM5100_AIF1TX4MIX_INPUT_4_VOLUME 0x71F +#define WM5100_AIF1TX5MIX_INPUT_1_SOURCE 0x720 +#define WM5100_AIF1TX5MIX_INPUT_1_VOLUME 0x721 +#define WM5100_AIF1TX5MIX_INPUT_2_SOURCE 0x722 +#define WM5100_AIF1TX5MIX_INPUT_2_VOLUME 0x723 +#define WM5100_AIF1TX5MIX_INPUT_3_SOURCE 0x724 +#define WM5100_AIF1TX5MIX_INPUT_3_VOLUME 0x725 +#define WM5100_AIF1TX5MIX_INPUT_4_SOURCE 0x726 +#define WM5100_AIF1TX5MIX_INPUT_4_VOLUME 0x727 +#define WM5100_AIF1TX6MIX_INPUT_1_SOURCE 0x728 +#define WM5100_AIF1TX6MIX_INPUT_1_VOLUME 0x729 +#define WM5100_AIF1TX6MIX_INPUT_2_SOURCE 0x72A +#define WM5100_AIF1TX6MIX_INPUT_2_VOLUME 0x72B +#define WM5100_AIF1TX6MIX_INPUT_3_SOURCE 0x72C +#define WM5100_AIF1TX6MIX_INPUT_3_VOLUME 0x72D +#define WM5100_AIF1TX6MIX_INPUT_4_SOURCE 0x72E +#define WM5100_AIF1TX6MIX_INPUT_4_VOLUME 0x72F +#define WM5100_AIF1TX7MIX_INPUT_1_SOURCE 0x730 +#define WM5100_AIF1TX7MIX_INPUT_1_VOLUME 0x731 +#define WM5100_AIF1TX7MIX_INPUT_2_SOURCE 0x732 +#define WM5100_AIF1TX7MIX_INPUT_2_VOLUME 0x733 +#define WM5100_AIF1TX7MIX_INPUT_3_SOURCE 0x734 +#define WM5100_AIF1TX7MIX_INPUT_3_VOLUME 0x735 +#define WM5100_AIF1TX7MIX_INPUT_4_SOURCE 0x736 +#define WM5100_AIF1TX7MIX_INPUT_4_VOLUME 0x737 +#define WM5100_AIF1TX8MIX_INPUT_1_SOURCE 0x738 +#define WM5100_AIF1TX8MIX_INPUT_1_VOLUME 0x739 +#define WM5100_AIF1TX8MIX_INPUT_2_SOURCE 0x73A +#define WM5100_AIF1TX8MIX_INPUT_2_VOLUME 0x73B +#define WM5100_AIF1TX8MIX_INPUT_3_SOURCE 0x73C +#define WM5100_AIF1TX8MIX_INPUT_3_VOLUME 0x73D +#define WM5100_AIF1TX8MIX_INPUT_4_SOURCE 0x73E +#define WM5100_AIF1TX8MIX_INPUT_4_VOLUME 0x73F +#define WM5100_AIF2TX1MIX_INPUT_1_SOURCE 0x740 +#define WM5100_AIF2TX1MIX_INPUT_1_VOLUME 0x741 +#define WM5100_AIF2TX1MIX_INPUT_2_SOURCE 0x742 +#define WM5100_AIF2TX1MIX_INPUT_2_VOLUME 0x743 +#define WM5100_AIF2TX1MIX_INPUT_3_SOURCE 0x744 +#define WM5100_AIF2TX1MIX_INPUT_3_VOLUME 0x745 +#define WM5100_AIF2TX1MIX_INPUT_4_SOURCE 0x746 +#define WM5100_AIF2TX1MIX_INPUT_4_VOLUME 0x747 +#define WM5100_AIF2TX2MIX_INPUT_1_SOURCE 0x748 +#define WM5100_AIF2TX2MIX_INPUT_1_VOLUME 0x749 +#define WM5100_AIF2TX2MIX_INPUT_2_SOURCE 0x74A +#define WM5100_AIF2TX2MIX_INPUT_2_VOLUME 0x74B +#define WM5100_AIF2TX2MIX_INPUT_3_SOURCE 0x74C +#define WM5100_AIF2TX2MIX_INPUT_3_VOLUME 0x74D +#define WM5100_AIF2TX2MIX_INPUT_4_SOURCE 0x74E +#define WM5100_AIF2TX2MIX_INPUT_4_VOLUME 0x74F +#define WM5100_AIF3TX1MIX_INPUT_1_SOURCE 0x780 +#define WM5100_AIF3TX1MIX_INPUT_1_VOLUME 0x781 +#define WM5100_AIF3TX1MIX_INPUT_2_SOURCE 0x782 +#define WM5100_AIF3TX1MIX_INPUT_2_VOLUME 0x783 +#define WM5100_AIF3TX1MIX_INPUT_3_SOURCE 0x784 +#define WM5100_AIF3TX1MIX_INPUT_3_VOLUME 0x785 +#define WM5100_AIF3TX1MIX_INPUT_4_SOURCE 0x786 +#define WM5100_AIF3TX1MIX_INPUT_4_VOLUME 0x787 +#define WM5100_AIF3TX2MIX_INPUT_1_SOURCE 0x788 +#define WM5100_AIF3TX2MIX_INPUT_1_VOLUME 0x789 +#define WM5100_AIF3TX2MIX_INPUT_2_SOURCE 0x78A +#define WM5100_AIF3TX2MIX_INPUT_2_VOLUME 0x78B +#define WM5100_AIF3TX2MIX_INPUT_3_SOURCE 0x78C +#define WM5100_AIF3TX2MIX_INPUT_3_VOLUME 0x78D +#define WM5100_AIF3TX2MIX_INPUT_4_SOURCE 0x78E +#define WM5100_AIF3TX2MIX_INPUT_4_VOLUME 0x78F +#define WM5100_EQ1MIX_INPUT_1_SOURCE 0x880 +#define WM5100_EQ1MIX_INPUT_1_VOLUME 0x881 +#define WM5100_EQ1MIX_INPUT_2_SOURCE 0x882 +#define WM5100_EQ1MIX_INPUT_2_VOLUME 0x883 +#define WM5100_EQ1MIX_INPUT_3_SOURCE 0x884 +#define WM5100_EQ1MIX_INPUT_3_VOLUME 0x885 +#define WM5100_EQ1MIX_INPUT_4_SOURCE 0x886 +#define WM5100_EQ1MIX_INPUT_4_VOLUME 0x887 +#define WM5100_EQ2MIX_INPUT_1_SOURCE 0x888 +#define WM5100_EQ2MIX_INPUT_1_VOLUME 0x889 +#define WM5100_EQ2MIX_INPUT_2_SOURCE 0x88A +#define WM5100_EQ2MIX_INPUT_2_VOLUME 0x88B +#define WM5100_EQ2MIX_INPUT_3_SOURCE 0x88C +#define WM5100_EQ2MIX_INPUT_3_VOLUME 0x88D +#define WM5100_EQ2MIX_INPUT_4_SOURCE 0x88E +#define WM5100_EQ2MIX_INPUT_4_VOLUME 0x88F +#define WM5100_EQ3MIX_INPUT_1_SOURCE 0x890 +#define WM5100_EQ3MIX_INPUT_1_VOLUME 0x891 +#define WM5100_EQ3MIX_INPUT_2_SOURCE 0x892 +#define WM5100_EQ3MIX_INPUT_2_VOLUME 0x893 +#define WM5100_EQ3MIX_INPUT_3_SOURCE 0x894 +#define WM5100_EQ3MIX_INPUT_3_VOLUME 0x895 +#define WM5100_EQ3MIX_INPUT_4_SOURCE 0x896 +#define WM5100_EQ3MIX_INPUT_4_VOLUME 0x897 +#define WM5100_EQ4MIX_INPUT_1_SOURCE 0x898 +#define WM5100_EQ4MIX_INPUT_1_VOLUME 0x899 +#define WM5100_EQ4MIX_INPUT_2_SOURCE 0x89A +#define WM5100_EQ4MIX_INPUT_2_VOLUME 0x89B +#define WM5100_EQ4MIX_INPUT_3_SOURCE 0x89C +#define WM5100_EQ4MIX_INPUT_3_VOLUME 0x89D +#define WM5100_EQ4MIX_INPUT_4_SOURCE 0x89E +#define WM5100_EQ4MIX_INPUT_4_VOLUME 0x89F +#define WM5100_DRC1LMIX_INPUT_1_SOURCE 0x8C0 +#define WM5100_DRC1LMIX_INPUT_1_VOLUME 0x8C1 +#define WM5100_DRC1LMIX_INPUT_2_SOURCE 0x8C2 +#define WM5100_DRC1LMIX_INPUT_2_VOLUME 0x8C3 +#define WM5100_DRC1LMIX_INPUT_3_SOURCE 0x8C4 +#define WM5100_DRC1LMIX_INPUT_3_VOLUME 0x8C5 +#define WM5100_DRC1LMIX_INPUT_4_SOURCE 0x8C6 +#define WM5100_DRC1LMIX_INPUT_4_VOLUME 0x8C7 +#define WM5100_DRC1RMIX_INPUT_1_SOURCE 0x8C8 +#define WM5100_DRC1RMIX_INPUT_1_VOLUME 0x8C9 +#define WM5100_DRC1RMIX_INPUT_2_SOURCE 0x8CA +#define WM5100_DRC1RMIX_INPUT_2_VOLUME 0x8CB +#define WM5100_DRC1RMIX_INPUT_3_SOURCE 0x8CC +#define WM5100_DRC1RMIX_INPUT_3_VOLUME 0x8CD +#define WM5100_DRC1RMIX_INPUT_4_SOURCE 0x8CE +#define WM5100_DRC1RMIX_INPUT_4_VOLUME 0x8CF +#define WM5100_HPLP1MIX_INPUT_1_SOURCE 0x900 +#define WM5100_HPLP1MIX_INPUT_1_VOLUME 0x901 +#define WM5100_HPLP1MIX_INPUT_2_SOURCE 0x902 +#define WM5100_HPLP1MIX_INPUT_2_VOLUME 0x903 +#define WM5100_HPLP1MIX_INPUT_3_SOURCE 0x904 +#define WM5100_HPLP1MIX_INPUT_3_VOLUME 0x905 +#define WM5100_HPLP1MIX_INPUT_4_SOURCE 0x906 +#define WM5100_HPLP1MIX_INPUT_4_VOLUME 0x907 +#define WM5100_HPLP2MIX_INPUT_1_SOURCE 0x908 +#define WM5100_HPLP2MIX_INPUT_1_VOLUME 0x909 +#define WM5100_HPLP2MIX_INPUT_2_SOURCE 0x90A +#define WM5100_HPLP2MIX_INPUT_2_VOLUME 0x90B +#define WM5100_HPLP2MIX_INPUT_3_SOURCE 0x90C +#define WM5100_HPLP2MIX_INPUT_3_VOLUME 0x90D +#define WM5100_HPLP2MIX_INPUT_4_SOURCE 0x90E +#define WM5100_HPLP2MIX_INPUT_4_VOLUME 0x90F +#define WM5100_HPLP3MIX_INPUT_1_SOURCE 0x910 +#define WM5100_HPLP3MIX_INPUT_1_VOLUME 0x911 +#define WM5100_HPLP3MIX_INPUT_2_SOURCE 0x912 +#define WM5100_HPLP3MIX_INPUT_2_VOLUME 0x913 +#define WM5100_HPLP3MIX_INPUT_3_SOURCE 0x914 +#define WM5100_HPLP3MIX_INPUT_3_VOLUME 0x915 +#define WM5100_HPLP3MIX_INPUT_4_SOURCE 0x916 +#define WM5100_HPLP3MIX_INPUT_4_VOLUME 0x917 +#define WM5100_HPLP4MIX_INPUT_1_SOURCE 0x918 +#define WM5100_HPLP4MIX_INPUT_1_VOLUME 0x919 +#define WM5100_HPLP4MIX_INPUT_2_SOURCE 0x91A +#define WM5100_HPLP4MIX_INPUT_2_VOLUME 0x91B +#define WM5100_HPLP4MIX_INPUT_3_SOURCE 0x91C +#define WM5100_HPLP4MIX_INPUT_3_VOLUME 0x91D +#define WM5100_HPLP4MIX_INPUT_4_SOURCE 0x91E +#define WM5100_HPLP4MIX_INPUT_4_VOLUME 0x91F +#define WM5100_DSP1LMIX_INPUT_1_SOURCE 0x940 +#define WM5100_DSP1LMIX_INPUT_1_VOLUME 0x941 +#define WM5100_DSP1LMIX_INPUT_2_SOURCE 0x942 +#define WM5100_DSP1LMIX_INPUT_2_VOLUME 0x943 +#define WM5100_DSP1LMIX_INPUT_3_SOURCE 0x944 +#define WM5100_DSP1LMIX_INPUT_3_VOLUME 0x945 +#define WM5100_DSP1LMIX_INPUT_4_SOURCE 0x946 +#define WM5100_DSP1LMIX_INPUT_4_VOLUME 0x947 +#define WM5100_DSP1RMIX_INPUT_1_SOURCE 0x948 +#define WM5100_DSP1RMIX_INPUT_1_VOLUME 0x949 +#define WM5100_DSP1RMIX_INPUT_2_SOURCE 0x94A +#define WM5100_DSP1RMIX_INPUT_2_VOLUME 0x94B +#define WM5100_DSP1RMIX_INPUT_3_SOURCE 0x94C +#define WM5100_DSP1RMIX_INPUT_3_VOLUME 0x94D +#define WM5100_DSP1RMIX_INPUT_4_SOURCE 0x94E +#define WM5100_DSP1RMIX_INPUT_4_VOLUME 0x94F +#define WM5100_DSP1AUX1MIX_INPUT_1_SOURCE 0x950 +#define WM5100_DSP1AUX2MIX_INPUT_1_SOURCE 0x958 +#define WM5100_DSP1AUX3MIX_INPUT_1_SOURCE 0x960 +#define WM5100_DSP1AUX4MIX_INPUT_1_SOURCE 0x968 +#define WM5100_DSP1AUX5MIX_INPUT_1_SOURCE 0x970 +#define WM5100_DSP1AUX6MIX_INPUT_1_SOURCE 0x978 +#define WM5100_DSP2LMIX_INPUT_1_SOURCE 0x980 +#define WM5100_DSP2LMIX_INPUT_1_VOLUME 0x981 +#define WM5100_DSP2LMIX_INPUT_2_SOURCE 0x982 +#define WM5100_DSP2LMIX_INPUT_2_VOLUME 0x983 +#define WM5100_DSP2LMIX_INPUT_3_SOURCE 0x984 +#define WM5100_DSP2LMIX_INPUT_3_VOLUME 0x985 +#define WM5100_DSP2LMIX_INPUT_4_SOURCE 0x986 +#define WM5100_DSP2LMIX_INPUT_4_VOLUME 0x987 +#define WM5100_DSP2RMIX_INPUT_1_SOURCE 0x988 +#define WM5100_DSP2RMIX_INPUT_1_VOLUME 0x989 +#define WM5100_DSP2RMIX_INPUT_2_SOURCE 0x98A +#define WM5100_DSP2RMIX_INPUT_2_VOLUME 0x98B +#define WM5100_DSP2RMIX_INPUT_3_SOURCE 0x98C +#define WM5100_DSP2RMIX_INPUT_3_VOLUME 0x98D +#define WM5100_DSP2RMIX_INPUT_4_SOURCE 0x98E +#define WM5100_DSP2RMIX_INPUT_4_VOLUME 0x98F +#define WM5100_DSP2AUX1MIX_INPUT_1_SOURCE 0x990 +#define WM5100_DSP2AUX2MIX_INPUT_1_SOURCE 0x998 +#define WM5100_DSP2AUX3MIX_INPUT_1_SOURCE 0x9A0 +#define WM5100_DSP2AUX4MIX_INPUT_1_SOURCE 0x9A8 +#define WM5100_DSP2AUX5MIX_INPUT_1_SOURCE 0x9B0 +#define WM5100_DSP2AUX6MIX_INPUT_1_SOURCE 0x9B8 +#define WM5100_DSP3LMIX_INPUT_1_SOURCE 0x9C0 +#define WM5100_DSP3LMIX_INPUT_1_VOLUME 0x9C1 +#define WM5100_DSP3LMIX_INPUT_2_SOURCE 0x9C2 +#define WM5100_DSP3LMIX_INPUT_2_VOLUME 0x9C3 +#define WM5100_DSP3LMIX_INPUT_3_SOURCE 0x9C4 +#define WM5100_DSP3LMIX_INPUT_3_VOLUME 0x9C5 +#define WM5100_DSP3LMIX_INPUT_4_SOURCE 0x9C6 +#define WM5100_DSP3LMIX_INPUT_4_VOLUME 0x9C7 +#define WM5100_DSP3RMIX_INPUT_1_SOURCE 0x9C8 +#define WM5100_DSP3RMIX_INPUT_1_VOLUME 0x9C9 +#define WM5100_DSP3RMIX_INPUT_2_SOURCE 0x9CA +#define WM5100_DSP3RMIX_INPUT_2_VOLUME 0x9CB +#define WM5100_DSP3RMIX_INPUT_3_SOURCE 0x9CC +#define WM5100_DSP3RMIX_INPUT_3_VOLUME 0x9CD +#define WM5100_DSP3RMIX_INPUT_4_SOURCE 0x9CE +#define WM5100_DSP3RMIX_INPUT_4_VOLUME 0x9CF +#define WM5100_DSP3AUX1MIX_INPUT_1_SOURCE 0x9D0 +#define WM5100_DSP3AUX2MIX_INPUT_1_SOURCE 0x9D8 +#define WM5100_DSP3AUX3MIX_INPUT_1_SOURCE 0x9E0 +#define WM5100_DSP3AUX4MIX_INPUT_1_SOURCE 0x9E8 +#define WM5100_DSP3AUX5MIX_INPUT_1_SOURCE 0x9F0 +#define WM5100_DSP3AUX6MIX_INPUT_1_SOURCE 0x9F8 +#define WM5100_ASRC1LMIX_INPUT_1_SOURCE 0xA80 +#define WM5100_ASRC1RMIX_INPUT_1_SOURCE 0xA88 +#define WM5100_ASRC2LMIX_INPUT_1_SOURCE 0xA90 +#define WM5100_ASRC2RMIX_INPUT_1_SOURCE 0xA98 +#define WM5100_ISRC1DEC1MIX_INPUT_1_SOURCE 0xB00 +#define WM5100_ISRC1DEC2MIX_INPUT_1_SOURCE 0xB08 +#define WM5100_ISRC1DEC3MIX_INPUT_1_SOURCE 0xB10 +#define WM5100_ISRC1DEC4MIX_INPUT_1_SOURCE 0xB18 +#define WM5100_ISRC1INT1MIX_INPUT_1_SOURCE 0xB20 +#define WM5100_ISRC1INT2MIX_INPUT_1_SOURCE 0xB28 +#define WM5100_ISRC1INT3MIX_INPUT_1_SOURCE 0xB30 +#define WM5100_ISRC1INT4MIX_INPUT_1_SOURCE 0xB38 +#define WM5100_ISRC2DEC1MIX_INPUT_1_SOURCE 0xB40 +#define WM5100_ISRC2DEC2MIX_INPUT_1_SOURCE 0xB48 +#define WM5100_ISRC2DEC3MIX_INPUT_1_SOURCE 0xB50 +#define WM5100_ISRC2DEC4MIX_INPUT_1_SOURCE 0xB58 +#define WM5100_ISRC2INT1MIX_INPUT_1_SOURCE 0xB60 +#define WM5100_ISRC2INT2MIX_INPUT_1_SOURCE 0xB68 +#define WM5100_ISRC2INT3MIX_INPUT_1_SOURCE 0xB70 +#define WM5100_ISRC2INT4MIX_INPUT_1_SOURCE 0xB78 +#define WM5100_GPIO_CTRL_1 0xC00 +#define WM5100_GPIO_CTRL_2 0xC01 +#define WM5100_GPIO_CTRL_3 0xC02 +#define WM5100_GPIO_CTRL_4 0xC03 +#define WM5100_GPIO_CTRL_5 0xC04 +#define WM5100_GPIO_CTRL_6 0xC05 +#define WM5100_MISC_PAD_CTRL_1 0xC23 +#define WM5100_MISC_PAD_CTRL_2 0xC24 +#define WM5100_MISC_PAD_CTRL_3 0xC25 +#define WM5100_MISC_PAD_CTRL_4 0xC26 +#define WM5100_MISC_PAD_CTRL_5 0xC27 +#define WM5100_MISC_GPIO_1 0xC28 +#define WM5100_INTERRUPT_STATUS_1 0xD00 +#define WM5100_INTERRUPT_STATUS_2 0xD01 +#define WM5100_INTERRUPT_STATUS_3 0xD02 +#define WM5100_INTERRUPT_STATUS_4 0xD03 +#define WM5100_INTERRUPT_RAW_STATUS_2 0xD04 +#define WM5100_INTERRUPT_RAW_STATUS_3 0xD05 +#define WM5100_INTERRUPT_RAW_STATUS_4 0xD06 +#define WM5100_INTERRUPT_STATUS_1_MASK 0xD07 +#define WM5100_INTERRUPT_STATUS_2_MASK 0xD08 +#define WM5100_INTERRUPT_STATUS_3_MASK 0xD09 +#define WM5100_INTERRUPT_STATUS_4_MASK 0xD0A +#define WM5100_INTERRUPT_CONTROL 0xD1F +#define WM5100_IRQ_DEBOUNCE_1 0xD20 +#define WM5100_IRQ_DEBOUNCE_2 0xD21 +#define WM5100_FX_CTRL 0xE00 +#define WM5100_EQ1_1 0xE10 +#define WM5100_EQ1_2 0xE11 +#define WM5100_EQ1_3 0xE12 +#define WM5100_EQ1_4 0xE13 +#define WM5100_EQ1_5 0xE14 +#define WM5100_EQ1_6 0xE15 +#define WM5100_EQ1_7 0xE16 +#define WM5100_EQ1_8 0xE17 +#define WM5100_EQ1_9 0xE18 +#define WM5100_EQ1_10 0xE19 +#define WM5100_EQ1_11 0xE1A +#define WM5100_EQ1_12 0xE1B +#define WM5100_EQ1_13 0xE1C +#define WM5100_EQ1_14 0xE1D +#define WM5100_EQ1_15 0xE1E +#define WM5100_EQ1_16 0xE1F +#define WM5100_EQ1_17 0xE20 +#define WM5100_EQ1_18 0xE21 +#define WM5100_EQ1_19 0xE22 +#define WM5100_EQ1_20 0xE23 +#define WM5100_EQ2_1 0xE26 +#define WM5100_EQ2_2 0xE27 +#define WM5100_EQ2_3 0xE28 +#define WM5100_EQ2_4 0xE29 +#define WM5100_EQ2_5 0xE2A +#define WM5100_EQ2_6 0xE2B +#define WM5100_EQ2_7 0xE2C +#define WM5100_EQ2_8 0xE2D +#define WM5100_EQ2_9 0xE2E +#define WM5100_EQ2_10 0xE2F +#define WM5100_EQ2_11 0xE30 +#define WM5100_EQ2_12 0xE31 +#define WM5100_EQ2_13 0xE32 +#define WM5100_EQ2_14 0xE33 +#define WM5100_EQ2_15 0xE34 +#define WM5100_EQ2_16 0xE35 +#define WM5100_EQ2_17 0xE36 +#define WM5100_EQ2_18 0xE37 +#define WM5100_EQ2_19 0xE38 +#define WM5100_EQ2_20 0xE39 +#define WM5100_EQ3_1 0xE3C +#define WM5100_EQ3_2 0xE3D +#define WM5100_EQ3_3 0xE3E +#define WM5100_EQ3_4 0xE3F +#define WM5100_EQ3_5 0xE40 +#define WM5100_EQ3_6 0xE41 +#define WM5100_EQ3_7 0xE42 +#define WM5100_EQ3_8 0xE43 +#define WM5100_EQ3_9 0xE44 +#define WM5100_EQ3_10 0xE45 +#define WM5100_EQ3_11 0xE46 +#define WM5100_EQ3_12 0xE47 +#define WM5100_EQ3_13 0xE48 +#define WM5100_EQ3_14 0xE49 +#define WM5100_EQ3_15 0xE4A +#define WM5100_EQ3_16 0xE4B +#define WM5100_EQ3_17 0xE4C +#define WM5100_EQ3_18 0xE4D +#define WM5100_EQ3_19 0xE4E +#define WM5100_EQ3_20 0xE4F +#define WM5100_EQ4_1 0xE52 +#define WM5100_EQ4_2 0xE53 +#define WM5100_EQ4_3 0xE54 +#define WM5100_EQ4_4 0xE55 +#define WM5100_EQ4_5 0xE56 +#define WM5100_EQ4_6 0xE57 +#define WM5100_EQ4_7 0xE58 +#define WM5100_EQ4_8 0xE59 +#define WM5100_EQ4_9 0xE5A +#define WM5100_EQ4_10 0xE5B +#define WM5100_EQ4_11 0xE5C +#define WM5100_EQ4_12 0xE5D +#define WM5100_EQ4_13 0xE5E +#define WM5100_EQ4_14 0xE5F +#define WM5100_EQ4_15 0xE60 +#define WM5100_EQ4_16 0xE61 +#define WM5100_EQ4_17 0xE62 +#define WM5100_EQ4_18 0xE63 +#define WM5100_EQ4_19 0xE64 +#define WM5100_EQ4_20 0xE65 +#define WM5100_DRC1_CTRL1 0xE80 +#define WM5100_DRC1_CTRL2 0xE81 +#define WM5100_DRC1_CTRL3 0xE82 +#define WM5100_DRC1_CTRL4 0xE83 +#define WM5100_DRC1_CTRL5 0xE84 +#define WM5100_HPLPF1_1 0xEC0 +#define WM5100_HPLPF1_2 0xEC1 +#define WM5100_HPLPF2_1 0xEC4 +#define WM5100_HPLPF2_2 0xEC5 +#define WM5100_HPLPF3_1 0xEC8 +#define WM5100_HPLPF3_2 0xEC9 +#define WM5100_HPLPF4_1 0xECC +#define WM5100_HPLPF4_2 0xECD +#define WM5100_DSP1_DM_0 0x4000 +#define WM5100_DSP1_DM_1 0x4001 +#define WM5100_DSP1_DM_2 0x4002 +#define WM5100_DSP1_DM_3 0x4003 +#define WM5100_DSP1_DM_508 0x41FC +#define WM5100_DSP1_DM_509 0x41FD +#define WM5100_DSP1_DM_510 0x41FE +#define WM5100_DSP1_DM_511 0x41FF +#define WM5100_DSP1_PM_0 0x4800 +#define WM5100_DSP1_PM_1 0x4801 +#define WM5100_DSP1_PM_2 0x4802 +#define WM5100_DSP1_PM_3 0x4803 +#define WM5100_DSP1_PM_4 0x4804 +#define WM5100_DSP1_PM_5 0x4805 +#define WM5100_DSP1_PM_1530 0x4DFA +#define WM5100_DSP1_PM_1531 0x4DFB +#define WM5100_DSP1_PM_1532 0x4DFC +#define WM5100_DSP1_PM_1533 0x4DFD +#define WM5100_DSP1_PM_1534 0x4DFE +#define WM5100_DSP1_PM_1535 0x4DFF +#define WM5100_DSP1_ZM_0 0x5000 +#define WM5100_DSP1_ZM_1 0x5001 +#define WM5100_DSP1_ZM_2 0x5002 +#define WM5100_DSP1_ZM_3 0x5003 +#define WM5100_DSP1_ZM_2044 0x57FC +#define WM5100_DSP1_ZM_2045 0x57FD +#define WM5100_DSP1_ZM_2046 0x57FE +#define WM5100_DSP1_ZM_2047 0x57FF +#define WM5100_DSP2_DM_0 0x6000 +#define WM5100_DSP2_DM_1 0x6001 +#define WM5100_DSP2_DM_2 0x6002 +#define WM5100_DSP2_DM_3 0x6003 +#define WM5100_DSP2_DM_508 0x61FC +#define WM5100_DSP2_DM_509 0x61FD +#define WM5100_DSP2_DM_510 0x61FE +#define WM5100_DSP2_DM_511 0x61FF +#define WM5100_DSP2_PM_0 0x6800 +#define WM5100_DSP2_PM_1 0x6801 +#define WM5100_DSP2_PM_2 0x6802 +#define WM5100_DSP2_PM_3 0x6803 +#define WM5100_DSP2_PM_4 0x6804 +#define WM5100_DSP2_PM_5 0x6805 +#define WM5100_DSP2_PM_1530 0x6DFA +#define WM5100_DSP2_PM_1531 0x6DFB +#define WM5100_DSP2_PM_1532 0x6DFC +#define WM5100_DSP2_PM_1533 0x6DFD +#define WM5100_DSP2_PM_1534 0x6DFE +#define WM5100_DSP2_PM_1535 0x6DFF +#define WM5100_DSP2_ZM_0 0x7000 +#define WM5100_DSP2_ZM_1 0x7001 +#define WM5100_DSP2_ZM_2 0x7002 +#define WM5100_DSP2_ZM_3 0x7003 +#define WM5100_DSP2_ZM_2044 0x77FC +#define WM5100_DSP2_ZM_2045 0x77FD +#define WM5100_DSP2_ZM_2046 0x77FE +#define WM5100_DSP2_ZM_2047 0x77FF +#define WM5100_DSP3_DM_0 0x8000 +#define WM5100_DSP3_DM_1 0x8001 +#define WM5100_DSP3_DM_2 0x8002 +#define WM5100_DSP3_DM_3 0x8003 +#define WM5100_DSP3_DM_508 0x81FC +#define WM5100_DSP3_DM_509 0x81FD +#define WM5100_DSP3_DM_510 0x81FE +#define WM5100_DSP3_DM_511 0x81FF +#define WM5100_DSP3_PM_0 0x8800 +#define WM5100_DSP3_PM_1 0x8801 +#define WM5100_DSP3_PM_2 0x8802 +#define WM5100_DSP3_PM_3 0x8803 +#define WM5100_DSP3_PM_4 0x8804 +#define WM5100_DSP3_PM_5 0x8805 +#define WM5100_DSP3_PM_1530 0x8DFA +#define WM5100_DSP3_PM_1531 0x8DFB +#define WM5100_DSP3_PM_1532 0x8DFC +#define WM5100_DSP3_PM_1533 0x8DFD +#define WM5100_DSP3_PM_1534 0x8DFE +#define WM5100_DSP3_PM_1535 0x8DFF +#define WM5100_DSP3_ZM_0 0x9000 +#define WM5100_DSP3_ZM_1 0x9001 +#define WM5100_DSP3_ZM_2 0x9002 +#define WM5100_DSP3_ZM_3 0x9003 +#define WM5100_DSP3_ZM_2044 0x97FC +#define WM5100_DSP3_ZM_2045 0x97FD +#define WM5100_DSP3_ZM_2046 0x97FE +#define WM5100_DSP3_ZM_2047 0x97FF + +#define WM5100_REGISTER_COUNT 1435 +#define WM5100_MAX_REGISTER 0x97FF + +/* + * Field Definitions. + */ + +/* + * R0 (0x00) - software reset + */ +#define WM5100_SW_RST_DEV_ID1_MASK 0xFFFF /* SW_RST_DEV_ID1 - [15:0] */ +#define WM5100_SW_RST_DEV_ID1_SHIFT 0 /* SW_RST_DEV_ID1 - [15:0] */ +#define WM5100_SW_RST_DEV_ID1_WIDTH 16 /* SW_RST_DEV_ID1 - [15:0] */ + +/* + * R1 (0x01) - Device Revision + */ +#define WM5100_DEVICE_REVISION_MASK 0x000F /* DEVICE_REVISION - [3:0] */ +#define WM5100_DEVICE_REVISION_SHIFT 0 /* DEVICE_REVISION - [3:0] */ +#define WM5100_DEVICE_REVISION_WIDTH 4 /* DEVICE_REVISION - [3:0] */ + +/* + * R16 (0x10) - Ctrl IF 1 + */ +#define WM5100_AUTO_INC 0x0001 /* AUTO_INC */ +#define WM5100_AUTO_INC_MASK 0x0001 /* AUTO_INC */ +#define WM5100_AUTO_INC_SHIFT 0 /* AUTO_INC */ +#define WM5100_AUTO_INC_WIDTH 1 /* AUTO_INC */ + +/* + * R32 (0x20) - Tone Generator 1 + */ +#define WM5100_TONE_RATE_MASK 0x3000 /* TONE_RATE - [13:12] */ +#define WM5100_TONE_RATE_SHIFT 12 /* TONE_RATE - [13:12] */ +#define WM5100_TONE_RATE_WIDTH 2 /* TONE_RATE - [13:12] */ +#define WM5100_TONE_OFFSET_MASK 0x0300 /* TONE_OFFSET - [9:8] */ +#define WM5100_TONE_OFFSET_SHIFT 8 /* TONE_OFFSET - [9:8] */ +#define WM5100_TONE_OFFSET_WIDTH 2 /* TONE_OFFSET - [9:8] */ +#define WM5100_TONE2_ENA 0x0002 /* TONE2_ENA */ +#define WM5100_TONE2_ENA_MASK 0x0002 /* TONE2_ENA */ +#define WM5100_TONE2_ENA_SHIFT 1 /* TONE2_ENA */ +#define WM5100_TONE2_ENA_WIDTH 1 /* TONE2_ENA */ +#define WM5100_TONE1_ENA 0x0001 /* TONE1_ENA */ +#define WM5100_TONE1_ENA_MASK 0x0001 /* TONE1_ENA */ +#define WM5100_TONE1_ENA_SHIFT 0 /* TONE1_ENA */ +#define WM5100_TONE1_ENA_WIDTH 1 /* TONE1_ENA */ + +/* + * R48 (0x30) - PWM Drive 1 + */ +#define WM5100_PWM_RATE_MASK 0x3000 /* PWM_RATE - [13:12] */ +#define WM5100_PWM_RATE_SHIFT 12 /* PWM_RATE - [13:12] */ +#define WM5100_PWM_RATE_WIDTH 2 /* PWM_RATE - [13:12] */ +#define WM5100_PWM_CLK_SEL_MASK 0x0300 /* PWM_CLK_SEL - [9:8] */ +#define WM5100_PWM_CLK_SEL_SHIFT 8 /* PWM_CLK_SEL - [9:8] */ +#define WM5100_PWM_CLK_SEL_WIDTH 2 /* PWM_CLK_SEL - [9:8] */ +#define WM5100_PWM2_OVD 0x0020 /* PWM2_OVD */ +#define WM5100_PWM2_OVD_MASK 0x0020 /* PWM2_OVD */ +#define WM5100_PWM2_OVD_SHIFT 5 /* PWM2_OVD */ +#define WM5100_PWM2_OVD_WIDTH 1 /* PWM2_OVD */ +#define WM5100_PWM1_OVD 0x0010 /* PWM1_OVD */ +#define WM5100_PWM1_OVD_MASK 0x0010 /* PWM1_OVD */ +#define WM5100_PWM1_OVD_SHIFT 4 /* PWM1_OVD */ +#define WM5100_PWM1_OVD_WIDTH 1 /* PWM1_OVD */ +#define WM5100_PWM2_ENA 0x0002 /* PWM2_ENA */ +#define WM5100_PWM2_ENA_MASK 0x0002 /* PWM2_ENA */ +#define WM5100_PWM2_ENA_SHIFT 1 /* PWM2_ENA */ +#define WM5100_PWM2_ENA_WIDTH 1 /* PWM2_ENA */ +#define WM5100_PWM1_ENA 0x0001 /* PWM1_ENA */ +#define WM5100_PWM1_ENA_MASK 0x0001 /* PWM1_ENA */ +#define WM5100_PWM1_ENA_SHIFT 0 /* PWM1_ENA */ +#define WM5100_PWM1_ENA_WIDTH 1 /* PWM1_ENA */ + +/* + * R49 (0x31) - PWM Drive 2 + */ +#define WM5100_PWM1_LVL_MASK 0x03FF /* PWM1_LVL - [9:0] */ +#define WM5100_PWM1_LVL_SHIFT 0 /* PWM1_LVL - [9:0] */ +#define WM5100_PWM1_LVL_WIDTH 10 /* PWM1_LVL - [9:0] */ + +/* + * R50 (0x32) - PWM Drive 3 + */ +#define WM5100_PWM2_LVL_MASK 0x03FF /* PWM2_LVL - [9:0] */ +#define WM5100_PWM2_LVL_SHIFT 0 /* PWM2_LVL - [9:0] */ +#define WM5100_PWM2_LVL_WIDTH 10 /* PWM2_LVL - [9:0] */ + +/* + * R256 (0x100) - Clocking 1 + */ +#define WM5100_CLK_32K_SRC_MASK 0x000F /* CLK_32K_SRC - [3:0] */ +#define WM5100_CLK_32K_SRC_SHIFT 0 /* CLK_32K_SRC - [3:0] */ +#define WM5100_CLK_32K_SRC_WIDTH 4 /* CLK_32K_SRC - [3:0] */ + +/* + * R257 (0x101) - Clocking 3 + */ +#define WM5100_SYSCLK_FREQ_MASK 0x0700 /* SYSCLK_FREQ - [10:8] */ +#define WM5100_SYSCLK_FREQ_SHIFT 8 /* SYSCLK_FREQ - [10:8] */ +#define WM5100_SYSCLK_FREQ_WIDTH 3 /* SYSCLK_FREQ - [10:8] */ +#define WM5100_SYSCLK_ENA 0x0040 /* SYSCLK_ENA */ +#define WM5100_SYSCLK_ENA_MASK 0x0040 /* SYSCLK_ENA */ +#define WM5100_SYSCLK_ENA_SHIFT 6 /* SYSCLK_ENA */ +#define WM5100_SYSCLK_ENA_WIDTH 1 /* SYSCLK_ENA */ +#define WM5100_SYSCLK_SRC_MASK 0x000F /* SYSCLK_SRC - [3:0] */ +#define WM5100_SYSCLK_SRC_SHIFT 0 /* SYSCLK_SRC - [3:0] */ +#define WM5100_SYSCLK_SRC_WIDTH 4 /* SYSCLK_SRC - [3:0] */ + +/* + * R258 (0x102) - Clocking 4 + */ +#define WM5100_SAMPLE_RATE_1_MASK 0x001F /* SAMPLE_RATE_1 - [4:0] */ +#define WM5100_SAMPLE_RATE_1_SHIFT 0 /* SAMPLE_RATE_1 - [4:0] */ +#define WM5100_SAMPLE_RATE_1_WIDTH 5 /* SAMPLE_RATE_1 - [4:0] */ + +/* + * R259 (0x103) - Clocking 5 + */ +#define WM5100_SAMPLE_RATE_2_MASK 0x001F /* SAMPLE_RATE_2 - [4:0] */ +#define WM5100_SAMPLE_RATE_2_SHIFT 0 /* SAMPLE_RATE_2 - [4:0] */ +#define WM5100_SAMPLE_RATE_2_WIDTH 5 /* SAMPLE_RATE_2 - [4:0] */ + +/* + * R260 (0x104) - Clocking 6 + */ +#define WM5100_SAMPLE_RATE_3_MASK 0x001F /* SAMPLE_RATE_3 - [4:0] */ +#define WM5100_SAMPLE_RATE_3_SHIFT 0 /* SAMPLE_RATE_3 - [4:0] */ +#define WM5100_SAMPLE_RATE_3_WIDTH 5 /* SAMPLE_RATE_3 - [4:0] */ + +/* + * R263 (0x107) - Clocking 7 + */ +#define WM5100_ASYNC_CLK_FREQ_MASK 0x0700 /* ASYNC_CLK_FREQ - [10:8] */ +#define WM5100_ASYNC_CLK_FREQ_SHIFT 8 /* ASYNC_CLK_FREQ - [10:8] */ +#define WM5100_ASYNC_CLK_FREQ_WIDTH 3 /* ASYNC_CLK_FREQ - [10:8] */ +#define WM5100_ASYNC_CLK_ENA 0x0040 /* ASYNC_CLK_ENA */ +#define WM5100_ASYNC_CLK_ENA_MASK 0x0040 /* ASYNC_CLK_ENA */ +#define WM5100_ASYNC_CLK_ENA_SHIFT 6 /* ASYNC_CLK_ENA */ +#define WM5100_ASYNC_CLK_ENA_WIDTH 1 /* ASYNC_CLK_ENA */ +#define WM5100_ASYNC_CLK_SRC_MASK 0x000F /* ASYNC_CLK_SRC - [3:0] */ +#define WM5100_ASYNC_CLK_SRC_SHIFT 0 /* ASYNC_CLK_SRC - [3:0] */ +#define WM5100_ASYNC_CLK_SRC_WIDTH 4 /* ASYNC_CLK_SRC - [3:0] */ + +/* + * R264 (0x108) - Clocking 8 + */ +#define WM5100_ASYNC_SAMPLE_RATE_MASK 0x001F /* ASYNC_SAMPLE_RATE - [4:0] */ +#define WM5100_ASYNC_SAMPLE_RATE_SHIFT 0 /* ASYNC_SAMPLE_RATE - [4:0] */ +#define WM5100_ASYNC_SAMPLE_RATE_WIDTH 5 /* ASYNC_SAMPLE_RATE - [4:0] */ + +/* + * R288 (0x120) - ASRC_ENABLE + */ +#define WM5100_ASRC2L_ENA 0x0008 /* ASRC2L_ENA */ +#define WM5100_ASRC2L_ENA_MASK 0x0008 /* ASRC2L_ENA */ +#define WM5100_ASRC2L_ENA_SHIFT 3 /* ASRC2L_ENA */ +#define WM5100_ASRC2L_ENA_WIDTH 1 /* ASRC2L_ENA */ +#define WM5100_ASRC2R_ENA 0x0004 /* ASRC2R_ENA */ +#define WM5100_ASRC2R_ENA_MASK 0x0004 /* ASRC2R_ENA */ +#define WM5100_ASRC2R_ENA_SHIFT 2 /* ASRC2R_ENA */ +#define WM5100_ASRC2R_ENA_WIDTH 1 /* ASRC2R_ENA */ +#define WM5100_ASRC1L_ENA 0x0002 /* ASRC1L_ENA */ +#define WM5100_ASRC1L_ENA_MASK 0x0002 /* ASRC1L_ENA */ +#define WM5100_ASRC1L_ENA_SHIFT 1 /* ASRC1L_ENA */ +#define WM5100_ASRC1L_ENA_WIDTH 1 /* ASRC1L_ENA */ +#define WM5100_ASRC1R_ENA 0x0001 /* ASRC1R_ENA */ +#define WM5100_ASRC1R_ENA_MASK 0x0001 /* ASRC1R_ENA */ +#define WM5100_ASRC1R_ENA_SHIFT 0 /* ASRC1R_ENA */ +#define WM5100_ASRC1R_ENA_WIDTH 1 /* ASRC1R_ENA */ + +/* + * R289 (0x121) - ASRC_STATUS + */ +#define WM5100_ASRC2L_ENA_STS 0x0008 /* ASRC2L_ENA_STS */ +#define WM5100_ASRC2L_ENA_STS_MASK 0x0008 /* ASRC2L_ENA_STS */ +#define WM5100_ASRC2L_ENA_STS_SHIFT 3 /* ASRC2L_ENA_STS */ +#define WM5100_ASRC2L_ENA_STS_WIDTH 1 /* ASRC2L_ENA_STS */ +#define WM5100_ASRC2R_ENA_STS 0x0004 /* ASRC2R_ENA_STS */ +#define WM5100_ASRC2R_ENA_STS_MASK 0x0004 /* ASRC2R_ENA_STS */ +#define WM5100_ASRC2R_ENA_STS_SHIFT 2 /* ASRC2R_ENA_STS */ +#define WM5100_ASRC2R_ENA_STS_WIDTH 1 /* ASRC2R_ENA_STS */ +#define WM5100_ASRC1L_ENA_STS 0x0002 /* ASRC1L_ENA_STS */ +#define WM5100_ASRC1L_ENA_STS_MASK 0x0002 /* ASRC1L_ENA_STS */ +#define WM5100_ASRC1L_ENA_STS_SHIFT 1 /* ASRC1L_ENA_STS */ +#define WM5100_ASRC1L_ENA_STS_WIDTH 1 /* ASRC1L_ENA_STS */ +#define WM5100_ASRC1R_ENA_STS 0x0001 /* ASRC1R_ENA_STS */ +#define WM5100_ASRC1R_ENA_STS_MASK 0x0001 /* ASRC1R_ENA_STS */ +#define WM5100_ASRC1R_ENA_STS_SHIFT 0 /* ASRC1R_ENA_STS */ +#define WM5100_ASRC1R_ENA_STS_WIDTH 1 /* ASRC1R_ENA_STS */ + +/* + * R290 (0x122) - ASRC_RATE1 + */ +#define WM5100_ASRC_RATE1_MASK 0x0006 /* ASRC_RATE1 - [2:1] */ +#define WM5100_ASRC_RATE1_SHIFT 1 /* ASRC_RATE1 - [2:1] */ +#define WM5100_ASRC_RATE1_WIDTH 2 /* ASRC_RATE1 - [2:1] */ + +/* + * R321 (0x141) - ISRC 1 CTRL 1 + */ +#define WM5100_ISRC1_DFS_ENA 0x2000 /* ISRC1_DFS_ENA */ +#define WM5100_ISRC1_DFS_ENA_MASK 0x2000 /* ISRC1_DFS_ENA */ +#define WM5100_ISRC1_DFS_ENA_SHIFT 13 /* ISRC1_DFS_ENA */ +#define WM5100_ISRC1_DFS_ENA_WIDTH 1 /* ISRC1_DFS_ENA */ +#define WM5100_ISRC1_CLK_SEL_MASK 0x0300 /* ISRC1_CLK_SEL - [9:8] */ +#define WM5100_ISRC1_CLK_SEL_SHIFT 8 /* ISRC1_CLK_SEL - [9:8] */ +#define WM5100_ISRC1_CLK_SEL_WIDTH 2 /* ISRC1_CLK_SEL - [9:8] */ +#define WM5100_ISRC1_FSH_MASK 0x000C /* ISRC1_FSH - [3:2] */ +#define WM5100_ISRC1_FSH_SHIFT 2 /* ISRC1_FSH - [3:2] */ +#define WM5100_ISRC1_FSH_WIDTH 2 /* ISRC1_FSH - [3:2] */ +#define WM5100_ISRC1_FSL_MASK 0x0003 /* ISRC1_FSL - [1:0] */ +#define WM5100_ISRC1_FSL_SHIFT 0 /* ISRC1_FSL - [1:0] */ +#define WM5100_ISRC1_FSL_WIDTH 2 /* ISRC1_FSL - [1:0] */ + +/* + * R322 (0x142) - ISRC 1 CTRL 2 + */ +#define WM5100_ISRC1_INT1_ENA 0x8000 /* ISRC1_INT1_ENA */ +#define WM5100_ISRC1_INT1_ENA_MASK 0x8000 /* ISRC1_INT1_ENA */ +#define WM5100_ISRC1_INT1_ENA_SHIFT 15 /* ISRC1_INT1_ENA */ +#define WM5100_ISRC1_INT1_ENA_WIDTH 1 /* ISRC1_INT1_ENA */ +#define WM5100_ISRC1_INT2_ENA 0x4000 /* ISRC1_INT2_ENA */ +#define WM5100_ISRC1_INT2_ENA_MASK 0x4000 /* ISRC1_INT2_ENA */ +#define WM5100_ISRC1_INT2_ENA_SHIFT 14 /* ISRC1_INT2_ENA */ +#define WM5100_ISRC1_INT2_ENA_WIDTH 1 /* ISRC1_INT2_ENA */ +#define WM5100_ISRC1_INT3_ENA 0x2000 /* ISRC1_INT3_ENA */ +#define WM5100_ISRC1_INT3_ENA_MASK 0x2000 /* ISRC1_INT3_ENA */ +#define WM5100_ISRC1_INT3_ENA_SHIFT 13 /* ISRC1_INT3_ENA */ +#define WM5100_ISRC1_INT3_ENA_WIDTH 1 /* ISRC1_INT3_ENA */ +#define WM5100_ISRC1_INT4_ENA 0x1000 /* ISRC1_INT4_ENA */ +#define WM5100_ISRC1_INT4_ENA_MASK 0x1000 /* ISRC1_INT4_ENA */ +#define WM5100_ISRC1_INT4_ENA_SHIFT 12 /* ISRC1_INT4_ENA */ +#define WM5100_ISRC1_INT4_ENA_WIDTH 1 /* ISRC1_INT4_ENA */ +#define WM5100_ISRC1_DEC1_ENA 0x0200 /* ISRC1_DEC1_ENA */ +#define WM5100_ISRC1_DEC1_ENA_MASK 0x0200 /* ISRC1_DEC1_ENA */ +#define WM5100_ISRC1_DEC1_ENA_SHIFT 9 /* ISRC1_DEC1_ENA */ +#define WM5100_ISRC1_DEC1_ENA_WIDTH 1 /* ISRC1_DEC1_ENA */ +#define WM5100_ISRC1_DEC2_ENA 0x0100 /* ISRC1_DEC2_ENA */ +#define WM5100_ISRC1_DEC2_ENA_MASK 0x0100 /* ISRC1_DEC2_ENA */ +#define WM5100_ISRC1_DEC2_ENA_SHIFT 8 /* ISRC1_DEC2_ENA */ +#define WM5100_ISRC1_DEC2_ENA_WIDTH 1 /* ISRC1_DEC2_ENA */ +#define WM5100_ISRC1_DEC3_ENA 0x0080 /* ISRC1_DEC3_ENA */ +#define WM5100_ISRC1_DEC3_ENA_MASK 0x0080 /* ISRC1_DEC3_ENA */ +#define WM5100_ISRC1_DEC3_ENA_SHIFT 7 /* ISRC1_DEC3_ENA */ +#define WM5100_ISRC1_DEC3_ENA_WIDTH 1 /* ISRC1_DEC3_ENA */ +#define WM5100_ISRC1_DEC4_ENA 0x0040 /* ISRC1_DEC4_ENA */ +#define WM5100_ISRC1_DEC4_ENA_MASK 0x0040 /* ISRC1_DEC4_ENA */ +#define WM5100_ISRC1_DEC4_ENA_SHIFT 6 /* ISRC1_DEC4_ENA */ +#define WM5100_ISRC1_DEC4_ENA_WIDTH 1 /* ISRC1_DEC4_ENA */ +#define WM5100_ISRC1_NOTCH_ENA 0x0001 /* ISRC1_NOTCH_ENA */ +#define WM5100_ISRC1_NOTCH_ENA_MASK 0x0001 /* ISRC1_NOTCH_ENA */ +#define WM5100_ISRC1_NOTCH_ENA_SHIFT 0 /* ISRC1_NOTCH_ENA */ +#define WM5100_ISRC1_NOTCH_ENA_WIDTH 1 /* ISRC1_NOTCH_ENA */ + +/* + * R323 (0x143) - ISRC 2 CTRL1 + */ +#define WM5100_ISRC2_DFS_ENA 0x2000 /* ISRC2_DFS_ENA */ +#define WM5100_ISRC2_DFS_ENA_MASK 0x2000 /* ISRC2_DFS_ENA */ +#define WM5100_ISRC2_DFS_ENA_SHIFT 13 /* ISRC2_DFS_ENA */ +#define WM5100_ISRC2_DFS_ENA_WIDTH 1 /* ISRC2_DFS_ENA */ +#define WM5100_ISRC2_CLK_SEL_MASK 0x0300 /* ISRC2_CLK_SEL - [9:8] */ +#define WM5100_ISRC2_CLK_SEL_SHIFT 8 /* ISRC2_CLK_SEL - [9:8] */ +#define WM5100_ISRC2_CLK_SEL_WIDTH 2 /* ISRC2_CLK_SEL - [9:8] */ +#define WM5100_ISRC2_FSH_MASK 0x000C /* ISRC2_FSH - [3:2] */ +#define WM5100_ISRC2_FSH_SHIFT 2 /* ISRC2_FSH - [3:2] */ +#define WM5100_ISRC2_FSH_WIDTH 2 /* ISRC2_FSH - [3:2] */ +#define WM5100_ISRC2_FSL_MASK 0x0003 /* ISRC2_FSL - [1:0] */ +#define WM5100_ISRC2_FSL_SHIFT 0 /* ISRC2_FSL - [1:0] */ +#define WM5100_ISRC2_FSL_WIDTH 2 /* ISRC2_FSL - [1:0] */ + +/* + * R324 (0x144) - ISRC 2 CTRL 2 + */ +#define WM5100_ISRC2_INT1_ENA 0x8000 /* ISRC2_INT1_ENA */ +#define WM5100_ISRC2_INT1_ENA_MASK 0x8000 /* ISRC2_INT1_ENA */ +#define WM5100_ISRC2_INT1_ENA_SHIFT 15 /* ISRC2_INT1_ENA */ +#define WM5100_ISRC2_INT1_ENA_WIDTH 1 /* ISRC2_INT1_ENA */ +#define WM5100_ISRC2_INT2_ENA 0x4000 /* ISRC2_INT2_ENA */ +#define WM5100_ISRC2_INT2_ENA_MASK 0x4000 /* ISRC2_INT2_ENA */ +#define WM5100_ISRC2_INT2_ENA_SHIFT 14 /* ISRC2_INT2_ENA */ +#define WM5100_ISRC2_INT2_ENA_WIDTH 1 /* ISRC2_INT2_ENA */ +#define WM5100_ISRC2_INT3_ENA 0x2000 /* ISRC2_INT3_ENA */ +#define WM5100_ISRC2_INT3_ENA_MASK 0x2000 /* ISRC2_INT3_ENA */ +#define WM5100_ISRC2_INT3_ENA_SHIFT 13 /* ISRC2_INT3_ENA */ +#define WM5100_ISRC2_INT3_ENA_WIDTH 1 /* ISRC2_INT3_ENA */ +#define WM5100_ISRC2_INT4_ENA 0x1000 /* ISRC2_INT4_ENA */ +#define WM5100_ISRC2_INT4_ENA_MASK 0x1000 /* ISRC2_INT4_ENA */ +#define WM5100_ISRC2_INT4_ENA_SHIFT 12 /* ISRC2_INT4_ENA */ +#define WM5100_ISRC2_INT4_ENA_WIDTH 1 /* ISRC2_INT4_ENA */ +#define WM5100_ISRC2_DEC1_ENA 0x0200 /* ISRC2_DEC1_ENA */ +#define WM5100_ISRC2_DEC1_ENA_MASK 0x0200 /* ISRC2_DEC1_ENA */ +#define WM5100_ISRC2_DEC1_ENA_SHIFT 9 /* ISRC2_DEC1_ENA */ +#define WM5100_ISRC2_DEC1_ENA_WIDTH 1 /* ISRC2_DEC1_ENA */ +#define WM5100_ISRC2_DEC2_ENA 0x0100 /* ISRC2_DEC2_ENA */ +#define WM5100_ISRC2_DEC2_ENA_MASK 0x0100 /* ISRC2_DEC2_ENA */ +#define WM5100_ISRC2_DEC2_ENA_SHIFT 8 /* ISRC2_DEC2_ENA */ +#define WM5100_ISRC2_DEC2_ENA_WIDTH 1 /* ISRC2_DEC2_ENA */ +#define WM5100_ISRC2_DEC3_ENA 0x0080 /* ISRC2_DEC3_ENA */ +#define WM5100_ISRC2_DEC3_ENA_MASK 0x0080 /* ISRC2_DEC3_ENA */ +#define WM5100_ISRC2_DEC3_ENA_SHIFT 7 /* ISRC2_DEC3_ENA */ +#define WM5100_ISRC2_DEC3_ENA_WIDTH 1 /* ISRC2_DEC3_ENA */ +#define WM5100_ISRC2_DEC4_ENA 0x0040 /* ISRC2_DEC4_ENA */ +#define WM5100_ISRC2_DEC4_ENA_MASK 0x0040 /* ISRC2_DEC4_ENA */ +#define WM5100_ISRC2_DEC4_ENA_SHIFT 6 /* ISRC2_DEC4_ENA */ +#define WM5100_ISRC2_DEC4_ENA_WIDTH 1 /* ISRC2_DEC4_ENA */ +#define WM5100_ISRC2_NOTCH_ENA 0x0001 /* ISRC2_NOTCH_ENA */ +#define WM5100_ISRC2_NOTCH_ENA_MASK 0x0001 /* ISRC2_NOTCH_ENA */ +#define WM5100_ISRC2_NOTCH_ENA_SHIFT 0 /* ISRC2_NOTCH_ENA */ +#define WM5100_ISRC2_NOTCH_ENA_WIDTH 1 /* ISRC2_NOTCH_ENA */ + +/* + * R386 (0x182) - FLL1 Control 1 + */ +#define WM5100_FLL1_ENA 0x0001 /* FLL1_ENA */ +#define WM5100_FLL1_ENA_MASK 0x0001 /* FLL1_ENA */ +#define WM5100_FLL1_ENA_SHIFT 0 /* FLL1_ENA */ +#define WM5100_FLL1_ENA_WIDTH 1 /* FLL1_ENA */ + +/* + * R387 (0x183) - FLL1 Control 2 + */ +#define WM5100_FLL1_OUTDIV_MASK 0x3F00 /* FLL1_OUTDIV - [13:8] */ +#define WM5100_FLL1_OUTDIV_SHIFT 8 /* FLL1_OUTDIV - [13:8] */ +#define WM5100_FLL1_OUTDIV_WIDTH 6 /* FLL1_OUTDIV - [13:8] */ +#define WM5100_FLL1_FRATIO_MASK 0x0007 /* FLL1_FRATIO - [2:0] */ +#define WM5100_FLL1_FRATIO_SHIFT 0 /* FLL1_FRATIO - [2:0] */ +#define WM5100_FLL1_FRATIO_WIDTH 3 /* FLL1_FRATIO - [2:0] */ + +/* + * R388 (0x184) - FLL1 Control 3 + */ +#define WM5100_FLL1_THETA_MASK 0xFFFF /* FLL1_THETA - [15:0] */ +#define WM5100_FLL1_THETA_SHIFT 0 /* FLL1_THETA - [15:0] */ +#define WM5100_FLL1_THETA_WIDTH 16 /* FLL1_THETA - [15:0] */ + +/* + * R390 (0x186) - FLL1 Control 5 + */ +#define WM5100_FLL1_N_MASK 0x03FF /* FLL1_N - [9:0] */ +#define WM5100_FLL1_N_SHIFT 0 /* FLL1_N - [9:0] */ +#define WM5100_FLL1_N_WIDTH 10 /* FLL1_N - [9:0] */ + +/* + * R391 (0x187) - FLL1 Control 6 + */ +#define WM5100_FLL1_REFCLK_DIV_MASK 0x00C0 /* FLL1_REFCLK_DIV - [7:6] */ +#define WM5100_FLL1_REFCLK_DIV_SHIFT 6 /* FLL1_REFCLK_DIV - [7:6] */ +#define WM5100_FLL1_REFCLK_DIV_WIDTH 2 /* FLL1_REFCLK_DIV - [7:6] */ +#define WM5100_FLL1_REFCLK_SRC_MASK 0x000F /* FLL1_REFCLK_SRC - [3:0] */ +#define WM5100_FLL1_REFCLK_SRC_SHIFT 0 /* FLL1_REFCLK_SRC - [3:0] */ +#define WM5100_FLL1_REFCLK_SRC_WIDTH 4 /* FLL1_REFCLK_SRC - [3:0] */ + +/* + * R392 (0x188) - FLL1 EFS 1 + */ +#define WM5100_FLL1_LAMBDA_MASK 0xFFFF /* FLL1_LAMBDA - [15:0] */ +#define WM5100_FLL1_LAMBDA_SHIFT 0 /* FLL1_LAMBDA - [15:0] */ +#define WM5100_FLL1_LAMBDA_WIDTH 16 /* FLL1_LAMBDA - [15:0] */ + +/* + * R418 (0x1A2) - FLL2 Control 1 + */ +#define WM5100_FLL2_ENA 0x0001 /* FLL2_ENA */ +#define WM5100_FLL2_ENA_MASK 0x0001 /* FLL2_ENA */ +#define WM5100_FLL2_ENA_SHIFT 0 /* FLL2_ENA */ +#define WM5100_FLL2_ENA_WIDTH 1 /* FLL2_ENA */ + +/* + * R419 (0x1A3) - FLL2 Control 2 + */ +#define WM5100_FLL2_OUTDIV_MASK 0x3F00 /* FLL2_OUTDIV - [13:8] */ +#define WM5100_FLL2_OUTDIV_SHIFT 8 /* FLL2_OUTDIV - [13:8] */ +#define WM5100_FLL2_OUTDIV_WIDTH 6 /* FLL2_OUTDIV - [13:8] */ +#define WM5100_FLL2_FRATIO_MASK 0x0007 /* FLL2_FRATIO - [2:0] */ +#define WM5100_FLL2_FRATIO_SHIFT 0 /* FLL2_FRATIO - [2:0] */ +#define WM5100_FLL2_FRATIO_WIDTH 3 /* FLL2_FRATIO - [2:0] */ + +/* + * R420 (0x1A4) - FLL2 Control 3 + */ +#define WM5100_FLL2_THETA_MASK 0xFFFF /* FLL2_THETA - [15:0] */ +#define WM5100_FLL2_THETA_SHIFT 0 /* FLL2_THETA - [15:0] */ +#define WM5100_FLL2_THETA_WIDTH 16 /* FLL2_THETA - [15:0] */ + +/* + * R422 (0x1A6) - FLL2 Control 5 + */ +#define WM5100_FLL2_N_MASK 0x03FF /* FLL2_N - [9:0] */ +#define WM5100_FLL2_N_SHIFT 0 /* FLL2_N - [9:0] */ +#define WM5100_FLL2_N_WIDTH 10 /* FLL2_N - [9:0] */ + +/* + * R423 (0x1A7) - FLL2 Control 6 + */ +#define WM5100_FLL2_REFCLK_DIV_MASK 0x00C0 /* FLL2_REFCLK_DIV - [7:6] */ +#define WM5100_FLL2_REFCLK_DIV_SHIFT 6 /* FLL2_REFCLK_DIV - [7:6] */ +#define WM5100_FLL2_REFCLK_DIV_WIDTH 2 /* FLL2_REFCLK_DIV - [7:6] */ +#define WM5100_FLL2_REFCLK_SRC_MASK 0x000F /* FLL2_REFCLK_SRC - [3:0] */ +#define WM5100_FLL2_REFCLK_SRC_SHIFT 0 /* FLL2_REFCLK_SRC - [3:0] */ +#define WM5100_FLL2_REFCLK_SRC_WIDTH 4 /* FLL2_REFCLK_SRC - [3:0] */ + +/* + * R424 (0x1A8) - FLL2 EFS 1 + */ +#define WM5100_FLL2_LAMBDA_MASK 0xFFFF /* FLL2_LAMBDA - [15:0] */ +#define WM5100_FLL2_LAMBDA_SHIFT 0 /* FLL2_LAMBDA - [15:0] */ +#define WM5100_FLL2_LAMBDA_WIDTH 16 /* FLL2_LAMBDA - [15:0] */ + +/* + * R512 (0x200) - Mic Charge Pump 1 + */ +#define WM5100_CP2_BYPASS 0x0020 /* CP2_BYPASS */ +#define WM5100_CP2_BYPASS_MASK 0x0020 /* CP2_BYPASS */ +#define WM5100_CP2_BYPASS_SHIFT 5 /* CP2_BYPASS */ +#define WM5100_CP2_BYPASS_WIDTH 1 /* CP2_BYPASS */ +#define WM5100_CP2_ENA 0x0001 /* CP2_ENA */ +#define WM5100_CP2_ENA_MASK 0x0001 /* CP2_ENA */ +#define WM5100_CP2_ENA_SHIFT 0 /* CP2_ENA */ +#define WM5100_CP2_ENA_WIDTH 1 /* CP2_ENA */ + +/* + * R513 (0x201) - Mic Charge Pump 2 + */ +#define WM5100_LDO2_VSEL_MASK 0xF800 /* LDO2_VSEL - [15:11] */ +#define WM5100_LDO2_VSEL_SHIFT 11 /* LDO2_VSEL - [15:11] */ +#define WM5100_LDO2_VSEL_WIDTH 5 /* LDO2_VSEL - [15:11] */ + +/* + * R514 (0x202) - HP Charge Pump 1 + */ +#define WM5100_CP1_ENA 0x0001 /* CP1_ENA */ +#define WM5100_CP1_ENA_MASK 0x0001 /* CP1_ENA */ +#define WM5100_CP1_ENA_SHIFT 0 /* CP1_ENA */ +#define WM5100_CP1_ENA_WIDTH 1 /* CP1_ENA */ + +/* + * R529 (0x211) - LDO1 Control + */ +#define WM5100_LDO1_BYPASS 0x0002 /* LDO1_BYPASS */ +#define WM5100_LDO1_BYPASS_MASK 0x0002 /* LDO1_BYPASS */ +#define WM5100_LDO1_BYPASS_SHIFT 1 /* LDO1_BYPASS */ +#define WM5100_LDO1_BYPASS_WIDTH 1 /* LDO1_BYPASS */ + +/* + * R533 (0x215) - Mic Bias Ctrl 1 + */ +#define WM5100_MICB1_DISCH 0x0040 /* MICB1_DISCH */ +#define WM5100_MICB1_DISCH_MASK 0x0040 /* MICB1_DISCH */ +#define WM5100_MICB1_DISCH_SHIFT 6 /* MICB1_DISCH */ +#define WM5100_MICB1_DISCH_WIDTH 1 /* MICB1_DISCH */ +#define WM5100_MICB1_RATE 0x0020 /* MICB1_RATE */ +#define WM5100_MICB1_RATE_MASK 0x0020 /* MICB1_RATE */ +#define WM5100_MICB1_RATE_SHIFT 5 /* MICB1_RATE */ +#define WM5100_MICB1_RATE_WIDTH 1 /* MICB1_RATE */ +#define WM5100_MICB1_LVL_MASK 0x001C /* MICB1_LVL - [4:2] */ +#define WM5100_MICB1_LVL_SHIFT 2 /* MICB1_LVL - [4:2] */ +#define WM5100_MICB1_LVL_WIDTH 3 /* MICB1_LVL - [4:2] */ +#define WM5100_MICB1_BYPASS 0x0002 /* MICB1_BYPASS */ +#define WM5100_MICB1_BYPASS_MASK 0x0002 /* MICB1_BYPASS */ +#define WM5100_MICB1_BYPASS_SHIFT 1 /* MICB1_BYPASS */ +#define WM5100_MICB1_BYPASS_WIDTH 1 /* MICB1_BYPASS */ +#define WM5100_MICB1_ENA 0x0001 /* MICB1_ENA */ +#define WM5100_MICB1_ENA_MASK 0x0001 /* MICB1_ENA */ +#define WM5100_MICB1_ENA_SHIFT 0 /* MICB1_ENA */ +#define WM5100_MICB1_ENA_WIDTH 1 /* MICB1_ENA */ + +/* + * R534 (0x216) - Mic Bias Ctrl 2 + */ +#define WM5100_MICB2_DISCH 0x0040 /* MICB2_DISCH */ +#define WM5100_MICB2_DISCH_MASK 0x0040 /* MICB2_DISCH */ +#define WM5100_MICB2_DISCH_SHIFT 6 /* MICB2_DISCH */ +#define WM5100_MICB2_DISCH_WIDTH 1 /* MICB2_DISCH */ +#define WM5100_MICB2_RATE 0x0020 /* MICB2_RATE */ +#define WM5100_MICB2_RATE_MASK 0x0020 /* MICB2_RATE */ +#define WM5100_MICB2_RATE_SHIFT 5 /* MICB2_RATE */ +#define WM5100_MICB2_RATE_WIDTH 1 /* MICB2_RATE */ +#define WM5100_MICB2_LVL_MASK 0x001C /* MICB2_LVL - [4:2] */ +#define WM5100_MICB2_LVL_SHIFT 2 /* MICB2_LVL - [4:2] */ +#define WM5100_MICB2_LVL_WIDTH 3 /* MICB2_LVL - [4:2] */ +#define WM5100_MICB2_BYPASS 0x0002 /* MICB2_BYPASS */ +#define WM5100_MICB2_BYPASS_MASK 0x0002 /* MICB2_BYPASS */ +#define WM5100_MICB2_BYPASS_SHIFT 1 /* MICB2_BYPASS */ +#define WM5100_MICB2_BYPASS_WIDTH 1 /* MICB2_BYPASS */ +#define WM5100_MICB2_ENA 0x0001 /* MICB2_ENA */ +#define WM5100_MICB2_ENA_MASK 0x0001 /* MICB2_ENA */ +#define WM5100_MICB2_ENA_SHIFT 0 /* MICB2_ENA */ +#define WM5100_MICB2_ENA_WIDTH 1 /* MICB2_ENA */ + +/* + * R535 (0x217) - Mic Bias Ctrl 3 + */ +#define WM5100_MICB3_DISCH 0x0040 /* MICB3_DISCH */ +#define WM5100_MICB3_DISCH_MASK 0x0040 /* MICB3_DISCH */ +#define WM5100_MICB3_DISCH_SHIFT 6 /* MICB3_DISCH */ +#define WM5100_MICB3_DISCH_WIDTH 1 /* MICB3_DISCH */ +#define WM5100_MICB3_RATE 0x0020 /* MICB3_RATE */ +#define WM5100_MICB3_RATE_MASK 0x0020 /* MICB3_RATE */ +#define WM5100_MICB3_RATE_SHIFT 5 /* MICB3_RATE */ +#define WM5100_MICB3_RATE_WIDTH 1 /* MICB3_RATE */ +#define WM5100_MICB3_LVL_MASK 0x001C /* MICB3_LVL - [4:2] */ +#define WM5100_MICB3_LVL_SHIFT 2 /* MICB3_LVL - [4:2] */ +#define WM5100_MICB3_LVL_WIDTH 3 /* MICB3_LVL - [4:2] */ +#define WM5100_MICB3_BYPASS 0x0002 /* MICB3_BYPASS */ +#define WM5100_MICB3_BYPASS_MASK 0x0002 /* MICB3_BYPASS */ +#define WM5100_MICB3_BYPASS_SHIFT 1 /* MICB3_BYPASS */ +#define WM5100_MICB3_BYPASS_WIDTH 1 /* MICB3_BYPASS */ +#define WM5100_MICB3_ENA 0x0001 /* MICB3_ENA */ +#define WM5100_MICB3_ENA_MASK 0x0001 /* MICB3_ENA */ +#define WM5100_MICB3_ENA_SHIFT 0 /* MICB3_ENA */ +#define WM5100_MICB3_ENA_WIDTH 1 /* MICB3_ENA */ + +/* + * R640 (0x280) - Accessory Detect Mode 1 + */ +#define WM5100_ACCDET_BIAS_SRC_MASK 0xC000 /* ACCDET_BIAS_SRC - [15:14] */ +#define WM5100_ACCDET_BIAS_SRC_SHIFT 14 /* ACCDET_BIAS_SRC - [15:14] */ +#define WM5100_ACCDET_BIAS_SRC_WIDTH 2 /* ACCDET_BIAS_SRC - [15:14] */ +#define WM5100_ACCDET_SRC 0x2000 /* ACCDET_SRC */ +#define WM5100_ACCDET_SRC_MASK 0x2000 /* ACCDET_SRC */ +#define WM5100_ACCDET_SRC_SHIFT 13 /* ACCDET_SRC */ +#define WM5100_ACCDET_SRC_WIDTH 1 /* ACCDET_SRC */ +#define WM5100_ACCDET_MODE_MASK 0x0003 /* ACCDET_MODE - [1:0] */ +#define WM5100_ACCDET_MODE_SHIFT 0 /* ACCDET_MODE - [1:0] */ +#define WM5100_ACCDET_MODE_WIDTH 2 /* ACCDET_MODE - [1:0] */ + +/* + * R648 (0x288) - Headphone Detect 1 + */ +#define WM5100_HP_HOLDTIME_MASK 0x00E0 /* HP_HOLDTIME - [7:5] */ +#define WM5100_HP_HOLDTIME_SHIFT 5 /* HP_HOLDTIME - [7:5] */ +#define WM5100_HP_HOLDTIME_WIDTH 3 /* HP_HOLDTIME - [7:5] */ +#define WM5100_HP_CLK_DIV_MASK 0x0018 /* HP_CLK_DIV - [4:3] */ +#define WM5100_HP_CLK_DIV_SHIFT 3 /* HP_CLK_DIV - [4:3] */ +#define WM5100_HP_CLK_DIV_WIDTH 2 /* HP_CLK_DIV - [4:3] */ +#define WM5100_HP_STEP_SIZE 0x0002 /* HP_STEP_SIZE */ +#define WM5100_HP_STEP_SIZE_MASK 0x0002 /* HP_STEP_SIZE */ +#define WM5100_HP_STEP_SIZE_SHIFT 1 /* HP_STEP_SIZE */ +#define WM5100_HP_STEP_SIZE_WIDTH 1 /* HP_STEP_SIZE */ +#define WM5100_HP_POLL 0x0001 /* HP_POLL */ +#define WM5100_HP_POLL_MASK 0x0001 /* HP_POLL */ +#define WM5100_HP_POLL_SHIFT 0 /* HP_POLL */ +#define WM5100_HP_POLL_WIDTH 1 /* HP_POLL */ + +/* + * R649 (0x289) - Headphone Detect 2 + */ +#define WM5100_HP_DONE 0x0080 /* HP_DONE */ +#define WM5100_HP_DONE_MASK 0x0080 /* HP_DONE */ +#define WM5100_HP_DONE_SHIFT 7 /* HP_DONE */ +#define WM5100_HP_DONE_WIDTH 1 /* HP_DONE */ +#define WM5100_HP_LVL_MASK 0x007F /* HP_LVL - [6:0] */ +#define WM5100_HP_LVL_SHIFT 0 /* HP_LVL - [6:0] */ +#define WM5100_HP_LVL_WIDTH 7 /* HP_LVL - [6:0] */ + +/* + * R656 (0x290) - Mic Detect 1 + */ +#define WM5100_ACCDET_BIAS_STARTTIME_MASK 0xF000 /* ACCDET_BIAS_STARTTIME - [15:12] */ +#define WM5100_ACCDET_BIAS_STARTTIME_SHIFT 12 /* ACCDET_BIAS_STARTTIME - [15:12] */ +#define WM5100_ACCDET_BIAS_STARTTIME_WIDTH 4 /* ACCDET_BIAS_STARTTIME - [15:12] */ +#define WM5100_ACCDET_RATE_MASK 0x0F00 /* ACCDET_RATE - [11:8] */ +#define WM5100_ACCDET_RATE_SHIFT 8 /* ACCDET_RATE - [11:8] */ +#define WM5100_ACCDET_RATE_WIDTH 4 /* ACCDET_RATE - [11:8] */ +#define WM5100_ACCDET_DBTIME 0x0002 /* ACCDET_DBTIME */ +#define WM5100_ACCDET_DBTIME_MASK 0x0002 /* ACCDET_DBTIME */ +#define WM5100_ACCDET_DBTIME_SHIFT 1 /* ACCDET_DBTIME */ +#define WM5100_ACCDET_DBTIME_WIDTH 1 /* ACCDET_DBTIME */ +#define WM5100_ACCDET_ENA 0x0001 /* ACCDET_ENA */ +#define WM5100_ACCDET_ENA_MASK 0x0001 /* ACCDET_ENA */ +#define WM5100_ACCDET_ENA_SHIFT 0 /* ACCDET_ENA */ +#define WM5100_ACCDET_ENA_WIDTH 1 /* ACCDET_ENA */ + +/* + * R657 (0x291) - Mic Detect 2 + */ +#define WM5100_ACCDET_LVL_SEL_MASK 0x00FF /* ACCDET_LVL_SEL - [7:0] */ +#define WM5100_ACCDET_LVL_SEL_SHIFT 0 /* ACCDET_LVL_SEL - [7:0] */ +#define WM5100_ACCDET_LVL_SEL_WIDTH 8 /* ACCDET_LVL_SEL - [7:0] */ + +/* + * R658 (0x292) - Mic Detect 3 + */ +#define WM5100_ACCDET_LVL_MASK 0x07FC /* ACCDET_LVL - [10:2] */ +#define WM5100_ACCDET_LVL_SHIFT 2 /* ACCDET_LVL - [10:2] */ +#define WM5100_ACCDET_LVL_WIDTH 9 /* ACCDET_LVL - [10:2] */ +#define WM5100_ACCDET_VALID 0x0002 /* ACCDET_VALID */ +#define WM5100_ACCDET_VALID_MASK 0x0002 /* ACCDET_VALID */ +#define WM5100_ACCDET_VALID_SHIFT 1 /* ACCDET_VALID */ +#define WM5100_ACCDET_VALID_WIDTH 1 /* ACCDET_VALID */ +#define WM5100_ACCDET_STS 0x0001 /* ACCDET_STS */ +#define WM5100_ACCDET_STS_MASK 0x0001 /* ACCDET_STS */ +#define WM5100_ACCDET_STS_SHIFT 0 /* ACCDET_STS */ +#define WM5100_ACCDET_STS_WIDTH 1 /* ACCDET_STS */ + +/* + * R769 (0x301) - Input Enables + */ +#define WM5100_IN4L_ENA 0x0080 /* IN4L_ENA */ +#define WM5100_IN4L_ENA_MASK 0x0080 /* IN4L_ENA */ +#define WM5100_IN4L_ENA_SHIFT 7 /* IN4L_ENA */ +#define WM5100_IN4L_ENA_WIDTH 1 /* IN4L_ENA */ +#define WM5100_IN4R_ENA 0x0040 /* IN4R_ENA */ +#define WM5100_IN4R_ENA_MASK 0x0040 /* IN4R_ENA */ +#define WM5100_IN4R_ENA_SHIFT 6 /* IN4R_ENA */ +#define WM5100_IN4R_ENA_WIDTH 1 /* IN4R_ENA */ +#define WM5100_IN3L_ENA 0x0020 /* IN3L_ENA */ +#define WM5100_IN3L_ENA_MASK 0x0020 /* IN3L_ENA */ +#define WM5100_IN3L_ENA_SHIFT 5 /* IN3L_ENA */ +#define WM5100_IN3L_ENA_WIDTH 1 /* IN3L_ENA */ +#define WM5100_IN3R_ENA 0x0010 /* IN3R_ENA */ +#define WM5100_IN3R_ENA_MASK 0x0010 /* IN3R_ENA */ +#define WM5100_IN3R_ENA_SHIFT 4 /* IN3R_ENA */ +#define WM5100_IN3R_ENA_WIDTH 1 /* IN3R_ENA */ +#define WM5100_IN2L_ENA 0x0008 /* IN2L_ENA */ +#define WM5100_IN2L_ENA_MASK 0x0008 /* IN2L_ENA */ +#define WM5100_IN2L_ENA_SHIFT 3 /* IN2L_ENA */ +#define WM5100_IN2L_ENA_WIDTH 1 /* IN2L_ENA */ +#define WM5100_IN2R_ENA 0x0004 /* IN2R_ENA */ +#define WM5100_IN2R_ENA_MASK 0x0004 /* IN2R_ENA */ +#define WM5100_IN2R_ENA_SHIFT 2 /* IN2R_ENA */ +#define WM5100_IN2R_ENA_WIDTH 1 /* IN2R_ENA */ +#define WM5100_IN1L_ENA 0x0002 /* IN1L_ENA */ +#define WM5100_IN1L_ENA_MASK 0x0002 /* IN1L_ENA */ +#define WM5100_IN1L_ENA_SHIFT 1 /* IN1L_ENA */ +#define WM5100_IN1L_ENA_WIDTH 1 /* IN1L_ENA */ +#define WM5100_IN1R_ENA 0x0001 /* IN1R_ENA */ +#define WM5100_IN1R_ENA_MASK 0x0001 /* IN1R_ENA */ +#define WM5100_IN1R_ENA_SHIFT 0 /* IN1R_ENA */ +#define WM5100_IN1R_ENA_WIDTH 1 /* IN1R_ENA */ + +/* + * R770 (0x302) - Input Enables Status + */ +#define WM5100_IN4L_ENA_STS 0x0080 /* IN4L_ENA_STS */ +#define WM5100_IN4L_ENA_STS_MASK 0x0080 /* IN4L_ENA_STS */ +#define WM5100_IN4L_ENA_STS_SHIFT 7 /* IN4L_ENA_STS */ +#define WM5100_IN4L_ENA_STS_WIDTH 1 /* IN4L_ENA_STS */ +#define WM5100_IN4R_ENA_STS 0x0040 /* IN4R_ENA_STS */ +#define WM5100_IN4R_ENA_STS_MASK 0x0040 /* IN4R_ENA_STS */ +#define WM5100_IN4R_ENA_STS_SHIFT 6 /* IN4R_ENA_STS */ +#define WM5100_IN4R_ENA_STS_WIDTH 1 /* IN4R_ENA_STS */ +#define WM5100_IN3L_ENA_STS 0x0020 /* IN3L_ENA_STS */ +#define WM5100_IN3L_ENA_STS_MASK 0x0020 /* IN3L_ENA_STS */ +#define WM5100_IN3L_ENA_STS_SHIFT 5 /* IN3L_ENA_STS */ +#define WM5100_IN3L_ENA_STS_WIDTH 1 /* IN3L_ENA_STS */ +#define WM5100_IN3R_ENA_STS 0x0010 /* IN3R_ENA_STS */ +#define WM5100_IN3R_ENA_STS_MASK 0x0010 /* IN3R_ENA_STS */ +#define WM5100_IN3R_ENA_STS_SHIFT 4 /* IN3R_ENA_STS */ +#define WM5100_IN3R_ENA_STS_WIDTH 1 /* IN3R_ENA_STS */ +#define WM5100_IN2L_ENA_STS 0x0008 /* IN2L_ENA_STS */ +#define WM5100_IN2L_ENA_STS_MASK 0x0008 /* IN2L_ENA_STS */ +#define WM5100_IN2L_ENA_STS_SHIFT 3 /* IN2L_ENA_STS */ +#define WM5100_IN2L_ENA_STS_WIDTH 1 /* IN2L_ENA_STS */ +#define WM5100_IN2R_ENA_STS 0x0004 /* IN2R_ENA_STS */ +#define WM5100_IN2R_ENA_STS_MASK 0x0004 /* IN2R_ENA_STS */ +#define WM5100_IN2R_ENA_STS_SHIFT 2 /* IN2R_ENA_STS */ +#define WM5100_IN2R_ENA_STS_WIDTH 1 /* IN2R_ENA_STS */ +#define WM5100_IN1L_ENA_STS 0x0002 /* IN1L_ENA_STS */ +#define WM5100_IN1L_ENA_STS_MASK 0x0002 /* IN1L_ENA_STS */ +#define WM5100_IN1L_ENA_STS_SHIFT 1 /* IN1L_ENA_STS */ +#define WM5100_IN1L_ENA_STS_WIDTH 1 /* IN1L_ENA_STS */ +#define WM5100_IN1R_ENA_STS 0x0001 /* IN1R_ENA_STS */ +#define WM5100_IN1R_ENA_STS_MASK 0x0001 /* IN1R_ENA_STS */ +#define WM5100_IN1R_ENA_STS_SHIFT 0 /* IN1R_ENA_STS */ +#define WM5100_IN1R_ENA_STS_WIDTH 1 /* IN1R_ENA_STS */ + +/* + * R784 (0x310) - IN1L Control + */ +#define WM5100_IN_RATE_MASK 0xC000 /* IN_RATE - [15:14] */ +#define WM5100_IN_RATE_SHIFT 14 /* IN_RATE - [15:14] */ +#define WM5100_IN_RATE_WIDTH 2 /* IN_RATE - [15:14] */ +#define WM5100_IN1_OSR 0x2000 /* IN1_OSR */ +#define WM5100_IN1_OSR_MASK 0x2000 /* IN1_OSR */ +#define WM5100_IN1_OSR_SHIFT 13 /* IN1_OSR */ +#define WM5100_IN1_OSR_WIDTH 1 /* IN1_OSR */ +#define WM5100_IN1_DMIC_SUP_MASK 0x1800 /* IN1_DMIC_SUP - [12:11] */ +#define WM5100_IN1_DMIC_SUP_SHIFT 11 /* IN1_DMIC_SUP - [12:11] */ +#define WM5100_IN1_DMIC_SUP_WIDTH 2 /* IN1_DMIC_SUP - [12:11] */ +#define WM5100_IN1_MODE_MASK 0x0600 /* IN1_MODE - [10:9] */ +#define WM5100_IN1_MODE_SHIFT 9 /* IN1_MODE - [10:9] */ +#define WM5100_IN1_MODE_WIDTH 2 /* IN1_MODE - [10:9] */ +#define WM5100_IN1L_PGA_VOL_MASK 0x00FE /* IN1L_PGA_VOL - [7:1] */ +#define WM5100_IN1L_PGA_VOL_SHIFT 1 /* IN1L_PGA_VOL - [7:1] */ +#define WM5100_IN1L_PGA_VOL_WIDTH 7 /* IN1L_PGA_VOL - [7:1] */ + +/* + * R785 (0x311) - IN1R Control + */ +#define WM5100_IN1R_PGA_VOL_MASK 0x00FE /* IN1R_PGA_VOL - [7:1] */ +#define WM5100_IN1R_PGA_VOL_SHIFT 1 /* IN1R_PGA_VOL - [7:1] */ +#define WM5100_IN1R_PGA_VOL_WIDTH 7 /* IN1R_PGA_VOL - [7:1] */ + +/* + * R786 (0x312) - IN2L Control + */ +#define WM5100_IN2_OSR 0x2000 /* IN2_OSR */ +#define WM5100_IN2_OSR_MASK 0x2000 /* IN2_OSR */ +#define WM5100_IN2_OSR_SHIFT 13 /* IN2_OSR */ +#define WM5100_IN2_OSR_WIDTH 1 /* IN2_OSR */ +#define WM5100_IN2_DMIC_SUP_MASK 0x1800 /* IN2_DMIC_SUP - [12:11] */ +#define WM5100_IN2_DMIC_SUP_SHIFT 11 /* IN2_DMIC_SUP - [12:11] */ +#define WM5100_IN2_DMIC_SUP_WIDTH 2 /* IN2_DMIC_SUP - [12:11] */ +#define WM5100_IN2_MODE_MASK 0x0600 /* IN2_MODE - [10:9] */ +#define WM5100_IN2_MODE_SHIFT 9 /* IN2_MODE - [10:9] */ +#define WM5100_IN2_MODE_WIDTH 2 /* IN2_MODE - [10:9] */ +#define WM5100_IN2L_PGA_VOL_MASK 0x00FE /* IN2L_PGA_VOL - [7:1] */ +#define WM5100_IN2L_PGA_VOL_SHIFT 1 /* IN2L_PGA_VOL - [7:1] */ +#define WM5100_IN2L_PGA_VOL_WIDTH 7 /* IN2L_PGA_VOL - [7:1] */ + +/* + * R787 (0x313) - IN2R Control + */ +#define WM5100_IN2R_PGA_VOL_MASK 0x00FE /* IN2R_PGA_VOL - [7:1] */ +#define WM5100_IN2R_PGA_VOL_SHIFT 1 /* IN2R_PGA_VOL - [7:1] */ +#define WM5100_IN2R_PGA_VOL_WIDTH 7 /* IN2R_PGA_VOL - [7:1] */ + +/* + * R788 (0x314) - IN3L Control + */ +#define WM5100_IN3_OSR 0x2000 /* IN3_OSR */ +#define WM5100_IN3_OSR_MASK 0x2000 /* IN3_OSR */ +#define WM5100_IN3_OSR_SHIFT 13 /* IN3_OSR */ +#define WM5100_IN3_OSR_WIDTH 1 /* IN3_OSR */ +#define WM5100_IN3_DMIC_SUP_MASK 0x1800 /* IN3_DMIC_SUP - [12:11] */ +#define WM5100_IN3_DMIC_SUP_SHIFT 11 /* IN3_DMIC_SUP - [12:11] */ +#define WM5100_IN3_DMIC_SUP_WIDTH 2 /* IN3_DMIC_SUP - [12:11] */ +#define WM5100_IN3_MODE_MASK 0x0600 /* IN3_MODE - [10:9] */ +#define WM5100_IN3_MODE_SHIFT 9 /* IN3_MODE - [10:9] */ +#define WM5100_IN3_MODE_WIDTH 2 /* IN3_MODE - [10:9] */ +#define WM5100_IN3L_PGA_VOL_MASK 0x00FE /* IN3L_PGA_VOL - [7:1] */ +#define WM5100_IN3L_PGA_VOL_SHIFT 1 /* IN3L_PGA_VOL - [7:1] */ +#define WM5100_IN3L_PGA_VOL_WIDTH 7 /* IN3L_PGA_VOL - [7:1] */ + +/* + * R789 (0x315) - IN3R Control + */ +#define WM5100_IN3R_PGA_VOL_MASK 0x00FE /* IN3R_PGA_VOL - [7:1] */ +#define WM5100_IN3R_PGA_VOL_SHIFT 1 /* IN3R_PGA_VOL - [7:1] */ +#define WM5100_IN3R_PGA_VOL_WIDTH 7 /* IN3R_PGA_VOL - [7:1] */ + +/* + * R790 (0x316) - IN4L Control + */ +#define WM5100_IN4_OSR 0x2000 /* IN4_OSR */ +#define WM5100_IN4_OSR_MASK 0x2000 /* IN4_OSR */ +#define WM5100_IN4_OSR_SHIFT 13 /* IN4_OSR */ +#define WM5100_IN4_OSR_WIDTH 1 /* IN4_OSR */ +#define WM5100_IN4_DMIC_SUP_MASK 0x1800 /* IN4_DMIC_SUP - [12:11] */ +#define WM5100_IN4_DMIC_SUP_SHIFT 11 /* IN4_DMIC_SUP - [12:11] */ +#define WM5100_IN4_DMIC_SUP_WIDTH 2 /* IN4_DMIC_SUP - [12:11] */ +#define WM5100_IN4_MODE_MASK 0x0600 /* IN4_MODE - [10:9] */ +#define WM5100_IN4_MODE_SHIFT 9 /* IN4_MODE - [10:9] */ +#define WM5100_IN4_MODE_WIDTH 2 /* IN4_MODE - [10:9] */ +#define WM5100_IN4L_PGA_VOL_MASK 0x00FE /* IN4L_PGA_VOL - [7:1] */ +#define WM5100_IN4L_PGA_VOL_SHIFT 1 /* IN4L_PGA_VOL - [7:1] */ +#define WM5100_IN4L_PGA_VOL_WIDTH 7 /* IN4L_PGA_VOL - [7:1] */ + +/* + * R791 (0x317) - IN4R Control + */ +#define WM5100_IN4R_PGA_VOL_MASK 0x00FE /* IN4R_PGA_VOL - [7:1] */ +#define WM5100_IN4R_PGA_VOL_SHIFT 1 /* IN4R_PGA_VOL - [7:1] */ +#define WM5100_IN4R_PGA_VOL_WIDTH 7 /* IN4R_PGA_VOL - [7:1] */ + +/* + * R792 (0x318) - RXANC_SRC + */ +#define WM5100_IN_RXANC_SEL_MASK 0x0007 /* IN_RXANC_SEL - [2:0] */ +#define WM5100_IN_RXANC_SEL_SHIFT 0 /* IN_RXANC_SEL - [2:0] */ +#define WM5100_IN_RXANC_SEL_WIDTH 3 /* IN_RXANC_SEL - [2:0] */ + +/* + * R793 (0x319) - Input Volume Ramp + */ +#define WM5100_IN_VD_RAMP_MASK 0x0070 /* IN_VD_RAMP - [6:4] */ +#define WM5100_IN_VD_RAMP_SHIFT 4 /* IN_VD_RAMP - [6:4] */ +#define WM5100_IN_VD_RAMP_WIDTH 3 /* IN_VD_RAMP - [6:4] */ +#define WM5100_IN_VI_RAMP_MASK 0x0007 /* IN_VI_RAMP - [2:0] */ +#define WM5100_IN_VI_RAMP_SHIFT 0 /* IN_VI_RAMP - [2:0] */ +#define WM5100_IN_VI_RAMP_WIDTH 3 /* IN_VI_RAMP - [2:0] */ + +/* + * R800 (0x320) - ADC Digital Volume 1L + */ +#define WM5100_IN_VU 0x0200 /* IN_VU */ +#define WM5100_IN_VU_MASK 0x0200 /* IN_VU */ +#define WM5100_IN_VU_SHIFT 9 /* IN_VU */ +#define WM5100_IN_VU_WIDTH 1 /* IN_VU */ +#define WM5100_IN1L_MUTE 0x0100 /* IN1L_MUTE */ +#define WM5100_IN1L_MUTE_MASK 0x0100 /* IN1L_MUTE */ +#define WM5100_IN1L_MUTE_SHIFT 8 /* IN1L_MUTE */ +#define WM5100_IN1L_MUTE_WIDTH 1 /* IN1L_MUTE */ +#define WM5100_IN1L_VOL_MASK 0x00FF /* IN1L_VOL - [7:0] */ +#define WM5100_IN1L_VOL_SHIFT 0 /* IN1L_VOL - [7:0] */ +#define WM5100_IN1L_VOL_WIDTH 8 /* IN1L_VOL - [7:0] */ + +/* + * R801 (0x321) - ADC Digital Volume 1R + */ +#define WM5100_IN_VU 0x0200 /* IN_VU */ +#define WM5100_IN_VU_MASK 0x0200 /* IN_VU */ +#define WM5100_IN_VU_SHIFT 9 /* IN_VU */ +#define WM5100_IN_VU_WIDTH 1 /* IN_VU */ +#define WM5100_IN1R_MUTE 0x0100 /* IN1R_MUTE */ +#define WM5100_IN1R_MUTE_MASK 0x0100 /* IN1R_MUTE */ +#define WM5100_IN1R_MUTE_SHIFT 8 /* IN1R_MUTE */ +#define WM5100_IN1R_MUTE_WIDTH 1 /* IN1R_MUTE */ +#define WM5100_IN1R_VOL_MASK 0x00FF /* IN1R_VOL - [7:0] */ +#define WM5100_IN1R_VOL_SHIFT 0 /* IN1R_VOL - [7:0] */ +#define WM5100_IN1R_VOL_WIDTH 8 /* IN1R_VOL - [7:0] */ + +/* + * R802 (0x322) - ADC Digital Volume 2L + */ +#define WM5100_IN_VU 0x0200 /* IN_VU */ +#define WM5100_IN_VU_MASK 0x0200 /* IN_VU */ +#define WM5100_IN_VU_SHIFT 9 /* IN_VU */ +#define WM5100_IN_VU_WIDTH 1 /* IN_VU */ +#define WM5100_IN2L_MUTE 0x0100 /* IN2L_MUTE */ +#define WM5100_IN2L_MUTE_MASK 0x0100 /* IN2L_MUTE */ +#define WM5100_IN2L_MUTE_SHIFT 8 /* IN2L_MUTE */ +#define WM5100_IN2L_MUTE_WIDTH 1 /* IN2L_MUTE */ +#define WM5100_IN2L_VOL_MASK 0x00FF /* IN2L_VOL - [7:0] */ +#define WM5100_IN2L_VOL_SHIFT 0 /* IN2L_VOL - [7:0] */ +#define WM5100_IN2L_VOL_WIDTH 8 /* IN2L_VOL - [7:0] */ + +/* + * R803 (0x323) - ADC Digital Volume 2R + */ +#define WM5100_IN_VU 0x0200 /* IN_VU */ +#define WM5100_IN_VU_MASK 0x0200 /* IN_VU */ +#define WM5100_IN_VU_SHIFT 9 /* IN_VU */ +#define WM5100_IN_VU_WIDTH 1 /* IN_VU */ +#define WM5100_IN2R_MUTE 0x0100 /* IN2R_MUTE */ +#define WM5100_IN2R_MUTE_MASK 0x0100 /* IN2R_MUTE */ +#define WM5100_IN2R_MUTE_SHIFT 8 /* IN2R_MUTE */ +#define WM5100_IN2R_MUTE_WIDTH 1 /* IN2R_MUTE */ +#define WM5100_IN2R_VOL_MASK 0x00FF /* IN2R_VOL - [7:0] */ +#define WM5100_IN2R_VOL_SHIFT 0 /* IN2R_VOL - [7:0] */ +#define WM5100_IN2R_VOL_WIDTH 8 /* IN2R_VOL - [7:0] */ + +/* + * R804 (0x324) - ADC Digital Volume 3L + */ +#define WM5100_IN_VU 0x0200 /* IN_VU */ +#define WM5100_IN_VU_MASK 0x0200 /* IN_VU */ +#define WM5100_IN_VU_SHIFT 9 /* IN_VU */ +#define WM5100_IN_VU_WIDTH 1 /* IN_VU */ +#define WM5100_IN3L_MUTE 0x0100 /* IN3L_MUTE */ +#define WM5100_IN3L_MUTE_MASK 0x0100 /* IN3L_MUTE */ +#define WM5100_IN3L_MUTE_SHIFT 8 /* IN3L_MUTE */ +#define WM5100_IN3L_MUTE_WIDTH 1 /* IN3L_MUTE */ +#define WM5100_IN3L_VOL_MASK 0x00FF /* IN3L_VOL - [7:0] */ +#define WM5100_IN3L_VOL_SHIFT 0 /* IN3L_VOL - [7:0] */ +#define WM5100_IN3L_VOL_WIDTH 8 /* IN3L_VOL - [7:0] */ + +/* + * R805 (0x325) - ADC Digital Volume 3R + */ +#define WM5100_IN_VU 0x0200 /* IN_VU */ +#define WM5100_IN_VU_MASK 0x0200 /* IN_VU */ +#define WM5100_IN_VU_SHIFT 9 /* IN_VU */ +#define WM5100_IN_VU_WIDTH 1 /* IN_VU */ +#define WM5100_IN3R_MUTE 0x0100 /* IN3R_MUTE */ +#define WM5100_IN3R_MUTE_MASK 0x0100 /* IN3R_MUTE */ +#define WM5100_IN3R_MUTE_SHIFT 8 /* IN3R_MUTE */ +#define WM5100_IN3R_MUTE_WIDTH 1 /* IN3R_MUTE */ +#define WM5100_IN3R_VOL_MASK 0x00FF /* IN3R_VOL - [7:0] */ +#define WM5100_IN3R_VOL_SHIFT 0 /* IN3R_VOL - [7:0] */ +#define WM5100_IN3R_VOL_WIDTH 8 /* IN3R_VOL - [7:0] */ + +/* + * R806 (0x326) - ADC Digital Volume 4L + */ +#define WM5100_IN_VU 0x0200 /* IN_VU */ +#define WM5100_IN_VU_MASK 0x0200 /* IN_VU */ +#define WM5100_IN_VU_SHIFT 9 /* IN_VU */ +#define WM5100_IN_VU_WIDTH 1 /* IN_VU */ +#define WM5100_IN4L_MUTE 0x0100 /* IN4L_MUTE */ +#define WM5100_IN4L_MUTE_MASK 0x0100 /* IN4L_MUTE */ +#define WM5100_IN4L_MUTE_SHIFT 8 /* IN4L_MUTE */ +#define WM5100_IN4L_MUTE_WIDTH 1 /* IN4L_MUTE */ +#define WM5100_IN4L_VOL_MASK 0x00FF /* IN4L_VOL - [7:0] */ +#define WM5100_IN4L_VOL_SHIFT 0 /* IN4L_VOL - [7:0] */ +#define WM5100_IN4L_VOL_WIDTH 8 /* IN4L_VOL - [7:0] */ + +/* + * R807 (0x327) - ADC Digital Volume 4R + */ +#define WM5100_IN_VU 0x0200 /* IN_VU */ +#define WM5100_IN_VU_MASK 0x0200 /* IN_VU */ +#define WM5100_IN_VU_SHIFT 9 /* IN_VU */ +#define WM5100_IN_VU_WIDTH 1 /* IN_VU */ +#define WM5100_IN4R_MUTE 0x0100 /* IN4R_MUTE */ +#define WM5100_IN4R_MUTE_MASK 0x0100 /* IN4R_MUTE */ +#define WM5100_IN4R_MUTE_SHIFT 8 /* IN4R_MUTE */ +#define WM5100_IN4R_MUTE_WIDTH 1 /* IN4R_MUTE */ +#define WM5100_IN4R_VOL_MASK 0x00FF /* IN4R_VOL - [7:0] */ +#define WM5100_IN4R_VOL_SHIFT 0 /* IN4R_VOL - [7:0] */ +#define WM5100_IN4R_VOL_WIDTH 8 /* IN4R_VOL - [7:0] */ + +/* + * R1025 (0x401) - Output Enables 2 + */ +#define WM5100_OUT6L_ENA 0x0800 /* OUT6L_ENA */ +#define WM5100_OUT6L_ENA_MASK 0x0800 /* OUT6L_ENA */ +#define WM5100_OUT6L_ENA_SHIFT 11 /* OUT6L_ENA */ +#define WM5100_OUT6L_ENA_WIDTH 1 /* OUT6L_ENA */ +#define WM5100_OUT6R_ENA 0x0400 /* OUT6R_ENA */ +#define WM5100_OUT6R_ENA_MASK 0x0400 /* OUT6R_ENA */ +#define WM5100_OUT6R_ENA_SHIFT 10 /* OUT6R_ENA */ +#define WM5100_OUT6R_ENA_WIDTH 1 /* OUT6R_ENA */ +#define WM5100_OUT5L_ENA 0x0200 /* OUT5L_ENA */ +#define WM5100_OUT5L_ENA_MASK 0x0200 /* OUT5L_ENA */ +#define WM5100_OUT5L_ENA_SHIFT 9 /* OUT5L_ENA */ +#define WM5100_OUT5L_ENA_WIDTH 1 /* OUT5L_ENA */ +#define WM5100_OUT5R_ENA 0x0100 /* OUT5R_ENA */ +#define WM5100_OUT5R_ENA_MASK 0x0100 /* OUT5R_ENA */ +#define WM5100_OUT5R_ENA_SHIFT 8 /* OUT5R_ENA */ +#define WM5100_OUT5R_ENA_WIDTH 1 /* OUT5R_ENA */ +#define WM5100_OUT4L_ENA 0x0080 /* OUT4L_ENA */ +#define WM5100_OUT4L_ENA_MASK 0x0080 /* OUT4L_ENA */ +#define WM5100_OUT4L_ENA_SHIFT 7 /* OUT4L_ENA */ +#define WM5100_OUT4L_ENA_WIDTH 1 /* OUT4L_ENA */ +#define WM5100_OUT4R_ENA 0x0040 /* OUT4R_ENA */ +#define WM5100_OUT4R_ENA_MASK 0x0040 /* OUT4R_ENA */ +#define WM5100_OUT4R_ENA_SHIFT 6 /* OUT4R_ENA */ +#define WM5100_OUT4R_ENA_WIDTH 1 /* OUT4R_ENA */ + +/* + * R1026 (0x402) - Output Status 1 + */ +#define WM5100_OUT3L_ENA_STS 0x0020 /* OUT3L_ENA_STS */ +#define WM5100_OUT3L_ENA_STS_MASK 0x0020 /* OUT3L_ENA_STS */ +#define WM5100_OUT3L_ENA_STS_SHIFT 5 /* OUT3L_ENA_STS */ +#define WM5100_OUT3L_ENA_STS_WIDTH 1 /* OUT3L_ENA_STS */ +#define WM5100_OUT3R_ENA_STS 0x0010 /* OUT3R_ENA_STS */ +#define WM5100_OUT3R_ENA_STS_MASK 0x0010 /* OUT3R_ENA_STS */ +#define WM5100_OUT3R_ENA_STS_SHIFT 4 /* OUT3R_ENA_STS */ +#define WM5100_OUT3R_ENA_STS_WIDTH 1 /* OUT3R_ENA_STS */ +#define WM5100_OUT2L_ENA_STS 0x0008 /* OUT2L_ENA_STS */ +#define WM5100_OUT2L_ENA_STS_MASK 0x0008 /* OUT2L_ENA_STS */ +#define WM5100_OUT2L_ENA_STS_SHIFT 3 /* OUT2L_ENA_STS */ +#define WM5100_OUT2L_ENA_STS_WIDTH 1 /* OUT2L_ENA_STS */ +#define WM5100_OUT2R_ENA_STS 0x0004 /* OUT2R_ENA_STS */ +#define WM5100_OUT2R_ENA_STS_MASK 0x0004 /* OUT2R_ENA_STS */ +#define WM5100_OUT2R_ENA_STS_SHIFT 2 /* OUT2R_ENA_STS */ +#define WM5100_OUT2R_ENA_STS_WIDTH 1 /* OUT2R_ENA_STS */ +#define WM5100_OUT1L_ENA_STS 0x0002 /* OUT1L_ENA_STS */ +#define WM5100_OUT1L_ENA_STS_MASK 0x0002 /* OUT1L_ENA_STS */ +#define WM5100_OUT1L_ENA_STS_SHIFT 1 /* OUT1L_ENA_STS */ +#define WM5100_OUT1L_ENA_STS_WIDTH 1 /* OUT1L_ENA_STS */ +#define WM5100_OUT1R_ENA_STS 0x0001 /* OUT1R_ENA_STS */ +#define WM5100_OUT1R_ENA_STS_MASK 0x0001 /* OUT1R_ENA_STS */ +#define WM5100_OUT1R_ENA_STS_SHIFT 0 /* OUT1R_ENA_STS */ +#define WM5100_OUT1R_ENA_STS_WIDTH 1 /* OUT1R_ENA_STS */ + +/* + * R1027 (0x403) - Output Status 2 + */ +#define WM5100_OUT6L_ENA_STS 0x0800 /* OUT6L_ENA_STS */ +#define WM5100_OUT6L_ENA_STS_MASK 0x0800 /* OUT6L_ENA_STS */ +#define WM5100_OUT6L_ENA_STS_SHIFT 11 /* OUT6L_ENA_STS */ +#define WM5100_OUT6L_ENA_STS_WIDTH 1 /* OUT6L_ENA_STS */ +#define WM5100_OUT6R_ENA_STS 0x0400 /* OUT6R_ENA_STS */ +#define WM5100_OUT6R_ENA_STS_MASK 0x0400 /* OUT6R_ENA_STS */ +#define WM5100_OUT6R_ENA_STS_SHIFT 10 /* OUT6R_ENA_STS */ +#define WM5100_OUT6R_ENA_STS_WIDTH 1 /* OUT6R_ENA_STS */ +#define WM5100_OUT5L_ENA_STS 0x0200 /* OUT5L_ENA_STS */ +#define WM5100_OUT5L_ENA_STS_MASK 0x0200 /* OUT5L_ENA_STS */ +#define WM5100_OUT5L_ENA_STS_SHIFT 9 /* OUT5L_ENA_STS */ +#define WM5100_OUT5L_ENA_STS_WIDTH 1 /* OUT5L_ENA_STS */ +#define WM5100_OUT5R_ENA_STS 0x0100 /* OUT5R_ENA_STS */ +#define WM5100_OUT5R_ENA_STS_MASK 0x0100 /* OUT5R_ENA_STS */ +#define WM5100_OUT5R_ENA_STS_SHIFT 8 /* OUT5R_ENA_STS */ +#define WM5100_OUT5R_ENA_STS_WIDTH 1 /* OUT5R_ENA_STS */ +#define WM5100_OUT4L_ENA_STS 0x0080 /* OUT4L_ENA_STS */ +#define WM5100_OUT4L_ENA_STS_MASK 0x0080 /* OUT4L_ENA_STS */ +#define WM5100_OUT4L_ENA_STS_SHIFT 7 /* OUT4L_ENA_STS */ +#define WM5100_OUT4L_ENA_STS_WIDTH 1 /* OUT4L_ENA_STS */ +#define WM5100_OUT4R_ENA_STS 0x0040 /* OUT4R_ENA_STS */ +#define WM5100_OUT4R_ENA_STS_MASK 0x0040 /* OUT4R_ENA_STS */ +#define WM5100_OUT4R_ENA_STS_SHIFT 6 /* OUT4R_ENA_STS */ +#define WM5100_OUT4R_ENA_STS_WIDTH 1 /* OUT4R_ENA_STS */ + +/* + * R1032 (0x408) - Channel Enables 1 + */ +#define WM5100_HP3L_ENA 0x0020 /* HP3L_ENA */ +#define WM5100_HP3L_ENA_MASK 0x0020 /* HP3L_ENA */ +#define WM5100_HP3L_ENA_SHIFT 5 /* HP3L_ENA */ +#define WM5100_HP3L_ENA_WIDTH 1 /* HP3L_ENA */ +#define WM5100_HP3R_ENA 0x0010 /* HP3R_ENA */ +#define WM5100_HP3R_ENA_MASK 0x0010 /* HP3R_ENA */ +#define WM5100_HP3R_ENA_SHIFT 4 /* HP3R_ENA */ +#define WM5100_HP3R_ENA_WIDTH 1 /* HP3R_ENA */ +#define WM5100_HP2L_ENA 0x0008 /* HP2L_ENA */ +#define WM5100_HP2L_ENA_MASK 0x0008 /* HP2L_ENA */ +#define WM5100_HP2L_ENA_SHIFT 3 /* HP2L_ENA */ +#define WM5100_HP2L_ENA_WIDTH 1 /* HP2L_ENA */ +#define WM5100_HP2R_ENA 0x0004 /* HP2R_ENA */ +#define WM5100_HP2R_ENA_MASK 0x0004 /* HP2R_ENA */ +#define WM5100_HP2R_ENA_SHIFT 2 /* HP2R_ENA */ +#define WM5100_HP2R_ENA_WIDTH 1 /* HP2R_ENA */ +#define WM5100_HP1L_ENA 0x0002 /* HP1L_ENA */ +#define WM5100_HP1L_ENA_MASK 0x0002 /* HP1L_ENA */ +#define WM5100_HP1L_ENA_SHIFT 1 /* HP1L_ENA */ +#define WM5100_HP1L_ENA_WIDTH 1 /* HP1L_ENA */ +#define WM5100_HP1R_ENA 0x0001 /* HP1R_ENA */ +#define WM5100_HP1R_ENA_MASK 0x0001 /* HP1R_ENA */ +#define WM5100_HP1R_ENA_SHIFT 0 /* HP1R_ENA */ +#define WM5100_HP1R_ENA_WIDTH 1 /* HP1R_ENA */ + +/* + * R1040 (0x410) - Out Volume 1L + */ +#define WM5100_OUT_RATE_MASK 0xC000 /* OUT_RATE - [15:14] */ +#define WM5100_OUT_RATE_SHIFT 14 /* OUT_RATE - [15:14] */ +#define WM5100_OUT_RATE_WIDTH 2 /* OUT_RATE - [15:14] */ +#define WM5100_OUT1_OSR 0x2000 /* OUT1_OSR */ +#define WM5100_OUT1_OSR_MASK 0x2000 /* OUT1_OSR */ +#define WM5100_OUT1_OSR_SHIFT 13 /* OUT1_OSR */ +#define WM5100_OUT1_OSR_WIDTH 1 /* OUT1_OSR */ +#define WM5100_OUT1_MONO 0x1000 /* OUT1_MONO */ +#define WM5100_OUT1_MONO_MASK 0x1000 /* OUT1_MONO */ +#define WM5100_OUT1_MONO_SHIFT 12 /* OUT1_MONO */ +#define WM5100_OUT1_MONO_WIDTH 1 /* OUT1_MONO */ +#define WM5100_OUT1L_ANC_SRC 0x0800 /* OUT1L_ANC_SRC */ +#define WM5100_OUT1L_ANC_SRC_MASK 0x0800 /* OUT1L_ANC_SRC */ +#define WM5100_OUT1L_ANC_SRC_SHIFT 11 /* OUT1L_ANC_SRC */ +#define WM5100_OUT1L_ANC_SRC_WIDTH 1 /* OUT1L_ANC_SRC */ +#define WM5100_OUT1L_PGA_VOL_MASK 0x00FE /* OUT1L_PGA_VOL - [7:1] */ +#define WM5100_OUT1L_PGA_VOL_SHIFT 1 /* OUT1L_PGA_VOL - [7:1] */ +#define WM5100_OUT1L_PGA_VOL_WIDTH 7 /* OUT1L_PGA_VOL - [7:1] */ + +/* + * R1041 (0x411) - Out Volume 1R + */ +#define WM5100_OUT1R_ANC_SRC 0x0800 /* OUT1R_ANC_SRC */ +#define WM5100_OUT1R_ANC_SRC_MASK 0x0800 /* OUT1R_ANC_SRC */ +#define WM5100_OUT1R_ANC_SRC_SHIFT 11 /* OUT1R_ANC_SRC */ +#define WM5100_OUT1R_ANC_SRC_WIDTH 1 /* OUT1R_ANC_SRC */ +#define WM5100_OUT1R_PGA_VOL_MASK 0x00FE /* OUT1R_PGA_VOL - [7:1] */ +#define WM5100_OUT1R_PGA_VOL_SHIFT 1 /* OUT1R_PGA_VOL - [7:1] */ +#define WM5100_OUT1R_PGA_VOL_WIDTH 7 /* OUT1R_PGA_VOL - [7:1] */ + +/* + * R1042 (0x412) - DAC Volume Limit 1L + */ +#define WM5100_OUT1L_VOL_LIM_MASK 0x00FF /* OUT1L_VOL_LIM - [7:0] */ +#define WM5100_OUT1L_VOL_LIM_SHIFT 0 /* OUT1L_VOL_LIM - [7:0] */ +#define WM5100_OUT1L_VOL_LIM_WIDTH 8 /* OUT1L_VOL_LIM - [7:0] */ + +/* + * R1043 (0x413) - DAC Volume Limit 1R + */ +#define WM5100_OUT1R_VOL_LIM_MASK 0x00FF /* OUT1R_VOL_LIM - [7:0] */ +#define WM5100_OUT1R_VOL_LIM_SHIFT 0 /* OUT1R_VOL_LIM - [7:0] */ +#define WM5100_OUT1R_VOL_LIM_WIDTH 8 /* OUT1R_VOL_LIM - [7:0] */ + +/* + * R1044 (0x414) - Out Volume 2L + */ +#define WM5100_OUT2_OSR 0x2000 /* OUT2_OSR */ +#define WM5100_OUT2_OSR_MASK 0x2000 /* OUT2_OSR */ +#define WM5100_OUT2_OSR_SHIFT 13 /* OUT2_OSR */ +#define WM5100_OUT2_OSR_WIDTH 1 /* OUT2_OSR */ +#define WM5100_OUT2_MONO 0x1000 /* OUT2_MONO */ +#define WM5100_OUT2_MONO_MASK 0x1000 /* OUT2_MONO */ +#define WM5100_OUT2_MONO_SHIFT 12 /* OUT2_MONO */ +#define WM5100_OUT2_MONO_WIDTH 1 /* OUT2_MONO */ +#define WM5100_OUT2L_ANC_SRC 0x0800 /* OUT2L_ANC_SRC */ +#define WM5100_OUT2L_ANC_SRC_MASK 0x0800 /* OUT2L_ANC_SRC */ +#define WM5100_OUT2L_ANC_SRC_SHIFT 11 /* OUT2L_ANC_SRC */ +#define WM5100_OUT2L_ANC_SRC_WIDTH 1 /* OUT2L_ANC_SRC */ +#define WM5100_OUT2L_PGA_VOL_MASK 0x00FE /* OUT2L_PGA_VOL - [7:1] */ +#define WM5100_OUT2L_PGA_VOL_SHIFT 1 /* OUT2L_PGA_VOL - [7:1] */ +#define WM5100_OUT2L_PGA_VOL_WIDTH 7 /* OUT2L_PGA_VOL - [7:1] */ + +/* + * R1045 (0x415) - Out Volume 2R + */ +#define WM5100_OUT2R_ANC_SRC 0x0800 /* OUT2R_ANC_SRC */ +#define WM5100_OUT2R_ANC_SRC_MASK 0x0800 /* OUT2R_ANC_SRC */ +#define WM5100_OUT2R_ANC_SRC_SHIFT 11 /* OUT2R_ANC_SRC */ +#define WM5100_OUT2R_ANC_SRC_WIDTH 1 /* OUT2R_ANC_SRC */ +#define WM5100_OUT2R_PGA_VOL_MASK 0x00FE /* OUT2R_PGA_VOL - [7:1] */ +#define WM5100_OUT2R_PGA_VOL_SHIFT 1 /* OUT2R_PGA_VOL - [7:1] */ +#define WM5100_OUT2R_PGA_VOL_WIDTH 7 /* OUT2R_PGA_VOL - [7:1] */ + +/* + * R1046 (0x416) - DAC Volume Limit 2L + */ +#define WM5100_OUT2L_VOL_LIM_MASK 0x00FF /* OUT2L_VOL_LIM - [7:0] */ +#define WM5100_OUT2L_VOL_LIM_SHIFT 0 /* OUT2L_VOL_LIM - [7:0] */ +#define WM5100_OUT2L_VOL_LIM_WIDTH 8 /* OUT2L_VOL_LIM - [7:0] */ + +/* + * R1047 (0x417) - DAC Volume Limit 2R + */ +#define WM5100_OUT2R_VOL_LIM_MASK 0x00FF /* OUT2R_VOL_LIM - [7:0] */ +#define WM5100_OUT2R_VOL_LIM_SHIFT 0 /* OUT2R_VOL_LIM - [7:0] */ +#define WM5100_OUT2R_VOL_LIM_WIDTH 8 /* OUT2R_VOL_LIM - [7:0] */ + +/* + * R1048 (0x418) - Out Volume 3L + */ +#define WM5100_OUT3_OSR 0x2000 /* OUT3_OSR */ +#define WM5100_OUT3_OSR_MASK 0x2000 /* OUT3_OSR */ +#define WM5100_OUT3_OSR_SHIFT 13 /* OUT3_OSR */ +#define WM5100_OUT3_OSR_WIDTH 1 /* OUT3_OSR */ +#define WM5100_OUT3_MONO 0x1000 /* OUT3_MONO */ +#define WM5100_OUT3_MONO_MASK 0x1000 /* OUT3_MONO */ +#define WM5100_OUT3_MONO_SHIFT 12 /* OUT3_MONO */ +#define WM5100_OUT3_MONO_WIDTH 1 /* OUT3_MONO */ +#define WM5100_OUT3L_ANC_SRC 0x0800 /* OUT3L_ANC_SRC */ +#define WM5100_OUT3L_ANC_SRC_MASK 0x0800 /* OUT3L_ANC_SRC */ +#define WM5100_OUT3L_ANC_SRC_SHIFT 11 /* OUT3L_ANC_SRC */ +#define WM5100_OUT3L_ANC_SRC_WIDTH 1 /* OUT3L_ANC_SRC */ +#define WM5100_OUT3L_PGA_VOL_MASK 0x00FE /* OUT3L_PGA_VOL - [7:1] */ +#define WM5100_OUT3L_PGA_VOL_SHIFT 1 /* OUT3L_PGA_VOL - [7:1] */ +#define WM5100_OUT3L_PGA_VOL_WIDTH 7 /* OUT3L_PGA_VOL - [7:1] */ + +/* + * R1049 (0x419) - Out Volume 3R + */ +#define WM5100_OUT3R_ANC_SRC 0x0800 /* OUT3R_ANC_SRC */ +#define WM5100_OUT3R_ANC_SRC_MASK 0x0800 /* OUT3R_ANC_SRC */ +#define WM5100_OUT3R_ANC_SRC_SHIFT 11 /* OUT3R_ANC_SRC */ +#define WM5100_OUT3R_ANC_SRC_WIDTH 1 /* OUT3R_ANC_SRC */ +#define WM5100_OUT3R_PGA_VOL_MASK 0x00FE /* OUT3R_PGA_VOL - [7:1] */ +#define WM5100_OUT3R_PGA_VOL_SHIFT 1 /* OUT3R_PGA_VOL - [7:1] */ +#define WM5100_OUT3R_PGA_VOL_WIDTH 7 /* OUT3R_PGA_VOL - [7:1] */ + +/* + * R1050 (0x41A) - DAC Volume Limit 3L + */ +#define WM5100_OUT3L_VOL_LIM_MASK 0x00FF /* OUT3L_VOL_LIM - [7:0] */ +#define WM5100_OUT3L_VOL_LIM_SHIFT 0 /* OUT3L_VOL_LIM - [7:0] */ +#define WM5100_OUT3L_VOL_LIM_WIDTH 8 /* OUT3L_VOL_LIM - [7:0] */ + +/* + * R1051 (0x41B) - DAC Volume Limit 3R + */ +#define WM5100_OUT3R_VOL_LIM_MASK 0x00FF /* OUT3R_VOL_LIM - [7:0] */ +#define WM5100_OUT3R_VOL_LIM_SHIFT 0 /* OUT3R_VOL_LIM - [7:0] */ +#define WM5100_OUT3R_VOL_LIM_WIDTH 8 /* OUT3R_VOL_LIM - [7:0] */ + +/* + * R1052 (0x41C) - Out Volume 4L + */ +#define WM5100_OUT4_OSR 0x2000 /* OUT4_OSR */ +#define WM5100_OUT4_OSR_MASK 0x2000 /* OUT4_OSR */ +#define WM5100_OUT4_OSR_SHIFT 13 /* OUT4_OSR */ +#define WM5100_OUT4_OSR_WIDTH 1 /* OUT4_OSR */ +#define WM5100_OUT4L_ANC_SRC 0x0800 /* OUT4L_ANC_SRC */ +#define WM5100_OUT4L_ANC_SRC_MASK 0x0800 /* OUT4L_ANC_SRC */ +#define WM5100_OUT4L_ANC_SRC_SHIFT 11 /* OUT4L_ANC_SRC */ +#define WM5100_OUT4L_ANC_SRC_WIDTH 1 /* OUT4L_ANC_SRC */ +#define WM5100_OUT4L_VOL_LIM_MASK 0x00FF /* OUT4L_VOL_LIM - [7:0] */ +#define WM5100_OUT4L_VOL_LIM_SHIFT 0 /* OUT4L_VOL_LIM - [7:0] */ +#define WM5100_OUT4L_VOL_LIM_WIDTH 8 /* OUT4L_VOL_LIM - [7:0] */ + +/* + * R1053 (0x41D) - Out Volume 4R + */ +#define WM5100_OUT4R_ANC_SRC 0x0800 /* OUT4R_ANC_SRC */ +#define WM5100_OUT4R_ANC_SRC_MASK 0x0800 /* OUT4R_ANC_SRC */ +#define WM5100_OUT4R_ANC_SRC_SHIFT 11 /* OUT4R_ANC_SRC */ +#define WM5100_OUT4R_ANC_SRC_WIDTH 1 /* OUT4R_ANC_SRC */ +#define WM5100_OUT4R_VOL_LIM_MASK 0x00FF /* OUT4R_VOL_LIM - [7:0] */ +#define WM5100_OUT4R_VOL_LIM_SHIFT 0 /* OUT4R_VOL_LIM - [7:0] */ +#define WM5100_OUT4R_VOL_LIM_WIDTH 8 /* OUT4R_VOL_LIM - [7:0] */ + +/* + * R1054 (0x41E) - DAC Volume Limit 5L + */ +#define WM5100_OUT5_OSR 0x2000 /* OUT5_OSR */ +#define WM5100_OUT5_OSR_MASK 0x2000 /* OUT5_OSR */ +#define WM5100_OUT5_OSR_SHIFT 13 /* OUT5_OSR */ +#define WM5100_OUT5_OSR_WIDTH 1 /* OUT5_OSR */ +#define WM5100_OUT5L_ANC_SRC 0x0800 /* OUT5L_ANC_SRC */ +#define WM5100_OUT5L_ANC_SRC_MASK 0x0800 /* OUT5L_ANC_SRC */ +#define WM5100_OUT5L_ANC_SRC_SHIFT 11 /* OUT5L_ANC_SRC */ +#define WM5100_OUT5L_ANC_SRC_WIDTH 1 /* OUT5L_ANC_SRC */ +#define WM5100_OUT5L_VOL_LIM_MASK 0x00FF /* OUT5L_VOL_LIM - [7:0] */ +#define WM5100_OUT5L_VOL_LIM_SHIFT 0 /* OUT5L_VOL_LIM - [7:0] */ +#define WM5100_OUT5L_VOL_LIM_WIDTH 8 /* OUT5L_VOL_LIM - [7:0] */ + +/* + * R1055 (0x41F) - DAC Volume Limit 5R + */ +#define WM5100_OUT5R_ANC_SRC 0x0800 /* OUT5R_ANC_SRC */ +#define WM5100_OUT5R_ANC_SRC_MASK 0x0800 /* OUT5R_ANC_SRC */ +#define WM5100_OUT5R_ANC_SRC_SHIFT 11 /* OUT5R_ANC_SRC */ +#define WM5100_OUT5R_ANC_SRC_WIDTH 1 /* OUT5R_ANC_SRC */ +#define WM5100_OUT5R_VOL_LIM_MASK 0x00FF /* OUT5R_VOL_LIM - [7:0] */ +#define WM5100_OUT5R_VOL_LIM_SHIFT 0 /* OUT5R_VOL_LIM - [7:0] */ +#define WM5100_OUT5R_VOL_LIM_WIDTH 8 /* OUT5R_VOL_LIM - [7:0] */ + +/* + * R1056 (0x420) - DAC Volume Limit 6L + */ +#define WM5100_OUT6_OSR 0x2000 /* OUT6_OSR */ +#define WM5100_OUT6_OSR_MASK 0x2000 /* OUT6_OSR */ +#define WM5100_OUT6_OSR_SHIFT 13 /* OUT6_OSR */ +#define WM5100_OUT6_OSR_WIDTH 1 /* OUT6_OSR */ +#define WM5100_OUT6L_ANC_SRC 0x0800 /* OUT6L_ANC_SRC */ +#define WM5100_OUT6L_ANC_SRC_MASK 0x0800 /* OUT6L_ANC_SRC */ +#define WM5100_OUT6L_ANC_SRC_SHIFT 11 /* OUT6L_ANC_SRC */ +#define WM5100_OUT6L_ANC_SRC_WIDTH 1 /* OUT6L_ANC_SRC */ +#define WM5100_OUT6L_VOL_LIM_MASK 0x00FF /* OUT6L_VOL_LIM - [7:0] */ +#define WM5100_OUT6L_VOL_LIM_SHIFT 0 /* OUT6L_VOL_LIM - [7:0] */ +#define WM5100_OUT6L_VOL_LIM_WIDTH 8 /* OUT6L_VOL_LIM - [7:0] */ + +/* + * R1057 (0x421) - DAC Volume Limit 6R + */ +#define WM5100_OUT6R_ANC_SRC 0x0800 /* OUT6R_ANC_SRC */ +#define WM5100_OUT6R_ANC_SRC_MASK 0x0800 /* OUT6R_ANC_SRC */ +#define WM5100_OUT6R_ANC_SRC_SHIFT 11 /* OUT6R_ANC_SRC */ +#define WM5100_OUT6R_ANC_SRC_WIDTH 1 /* OUT6R_ANC_SRC */ +#define WM5100_OUT6R_VOL_LIM_MASK 0x00FF /* OUT6R_VOL_LIM - [7:0] */ +#define WM5100_OUT6R_VOL_LIM_SHIFT 0 /* OUT6R_VOL_LIM - [7:0] */ +#define WM5100_OUT6R_VOL_LIM_WIDTH 8 /* OUT6R_VOL_LIM - [7:0] */ + +/* + * R1088 (0x440) - DAC AEC Control 1 + */ +#define WM5100_AEC_LOOPBACK_SRC_MASK 0x003C /* AEC_LOOPBACK_SRC - [5:2] */ +#define WM5100_AEC_LOOPBACK_SRC_SHIFT 2 /* AEC_LOOPBACK_SRC - [5:2] */ +#define WM5100_AEC_LOOPBACK_SRC_WIDTH 4 /* AEC_LOOPBACK_SRC - [5:2] */ +#define WM5100_AEC_ENA_STS 0x0002 /* AEC_ENA_STS */ +#define WM5100_AEC_ENA_STS_MASK 0x0002 /* AEC_ENA_STS */ +#define WM5100_AEC_ENA_STS_SHIFT 1 /* AEC_ENA_STS */ +#define WM5100_AEC_ENA_STS_WIDTH 1 /* AEC_ENA_STS */ +#define WM5100_AEC_LOOPBACK_ENA 0x0001 /* AEC_LOOPBACK_ENA */ +#define WM5100_AEC_LOOPBACK_ENA_MASK 0x0001 /* AEC_LOOPBACK_ENA */ +#define WM5100_AEC_LOOPBACK_ENA_SHIFT 0 /* AEC_LOOPBACK_ENA */ +#define WM5100_AEC_LOOPBACK_ENA_WIDTH 1 /* AEC_LOOPBACK_ENA */ + +/* + * R1089 (0x441) - Output Volume Ramp + */ +#define WM5100_OUT_VD_RAMP_MASK 0x0070 /* OUT_VD_RAMP - [6:4] */ +#define WM5100_OUT_VD_RAMP_SHIFT 4 /* OUT_VD_RAMP - [6:4] */ +#define WM5100_OUT_VD_RAMP_WIDTH 3 /* OUT_VD_RAMP - [6:4] */ +#define WM5100_OUT_VI_RAMP_MASK 0x0007 /* OUT_VI_RAMP - [2:0] */ +#define WM5100_OUT_VI_RAMP_SHIFT 0 /* OUT_VI_RAMP - [2:0] */ +#define WM5100_OUT_VI_RAMP_WIDTH 3 /* OUT_VI_RAMP - [2:0] */ + +/* + * R1152 (0x480) - DAC Digital Volume 1L + */ +#define WM5100_OUT_VU 0x0200 /* OUT_VU */ +#define WM5100_OUT_VU_MASK 0x0200 /* OUT_VU */ +#define WM5100_OUT_VU_SHIFT 9 /* OUT_VU */ +#define WM5100_OUT_VU_WIDTH 1 /* OUT_VU */ +#define WM5100_OUT1L_MUTE 0x0100 /* OUT1L_MUTE */ +#define WM5100_OUT1L_MUTE_MASK 0x0100 /* OUT1L_MUTE */ +#define WM5100_OUT1L_MUTE_SHIFT 8 /* OUT1L_MUTE */ +#define WM5100_OUT1L_MUTE_WIDTH 1 /* OUT1L_MUTE */ +#define WM5100_OUT1L_VOL_MASK 0x00FF /* OUT1L_VOL - [7:0] */ +#define WM5100_OUT1L_VOL_SHIFT 0 /* OUT1L_VOL - [7:0] */ +#define WM5100_OUT1L_VOL_WIDTH 8 /* OUT1L_VOL - [7:0] */ + +/* + * R1153 (0x481) - DAC Digital Volume 1R + */ +#define WM5100_OUT_VU 0x0200 /* OUT_VU */ +#define WM5100_OUT_VU_MASK 0x0200 /* OUT_VU */ +#define WM5100_OUT_VU_SHIFT 9 /* OUT_VU */ +#define WM5100_OUT_VU_WIDTH 1 /* OUT_VU */ +#define WM5100_OUT1R_MUTE 0x0100 /* OUT1R_MUTE */ +#define WM5100_OUT1R_MUTE_MASK 0x0100 /* OUT1R_MUTE */ +#define WM5100_OUT1R_MUTE_SHIFT 8 /* OUT1R_MUTE */ +#define WM5100_OUT1R_MUTE_WIDTH 1 /* OUT1R_MUTE */ +#define WM5100_OUT1R_VOL_MASK 0x00FF /* OUT1R_VOL - [7:0] */ +#define WM5100_OUT1R_VOL_SHIFT 0 /* OUT1R_VOL - [7:0] */ +#define WM5100_OUT1R_VOL_WIDTH 8 /* OUT1R_VOL - [7:0] */ + +/* + * R1154 (0x482) - DAC Digital Volume 2L + */ +#define WM5100_OUT_VU 0x0200 /* OUT_VU */ +#define WM5100_OUT_VU_MASK 0x0200 /* OUT_VU */ +#define WM5100_OUT_VU_SHIFT 9 /* OUT_VU */ +#define WM5100_OUT_VU_WIDTH 1 /* OUT_VU */ +#define WM5100_OUT2L_MUTE 0x0100 /* OUT2L_MUTE */ +#define WM5100_OUT2L_MUTE_MASK 0x0100 /* OUT2L_MUTE */ +#define WM5100_OUT2L_MUTE_SHIFT 8 /* OUT2L_MUTE */ +#define WM5100_OUT2L_MUTE_WIDTH 1 /* OUT2L_MUTE */ +#define WM5100_OUT2L_VOL_MASK 0x00FF /* OUT2L_VOL - [7:0] */ +#define WM5100_OUT2L_VOL_SHIFT 0 /* OUT2L_VOL - [7:0] */ +#define WM5100_OUT2L_VOL_WIDTH 8 /* OUT2L_VOL - [7:0] */ + +/* + * R1155 (0x483) - DAC Digital Volume 2R + */ +#define WM5100_OUT_VU 0x0200 /* OUT_VU */ +#define WM5100_OUT_VU_MASK 0x0200 /* OUT_VU */ +#define WM5100_OUT_VU_SHIFT 9 /* OUT_VU */ +#define WM5100_OUT_VU_WIDTH 1 /* OUT_VU */ +#define WM5100_OUT2R_MUTE 0x0100 /* OUT2R_MUTE */ +#define WM5100_OUT2R_MUTE_MASK 0x0100 /* OUT2R_MUTE */ +#define WM5100_OUT2R_MUTE_SHIFT 8 /* OUT2R_MUTE */ +#define WM5100_OUT2R_MUTE_WIDTH 1 /* OUT2R_MUTE */ +#define WM5100_OUT2R_VOL_MASK 0x00FF /* OUT2R_VOL - [7:0] */ +#define WM5100_OUT2R_VOL_SHIFT 0 /* OUT2R_VOL - [7:0] */ +#define WM5100_OUT2R_VOL_WIDTH 8 /* OUT2R_VOL - [7:0] */ + +/* + * R1156 (0x484) - DAC Digital Volume 3L + */ +#define WM5100_OUT_VU 0x0200 /* OUT_VU */ +#define WM5100_OUT_VU_MASK 0x0200 /* OUT_VU */ +#define WM5100_OUT_VU_SHIFT 9 /* OUT_VU */ +#define WM5100_OUT_VU_WIDTH 1 /* OUT_VU */ +#define WM5100_OUT3L_MUTE 0x0100 /* OUT3L_MUTE */ +#define WM5100_OUT3L_MUTE_MASK 0x0100 /* OUT3L_MUTE */ +#define WM5100_OUT3L_MUTE_SHIFT 8 /* OUT3L_MUTE */ +#define WM5100_OUT3L_MUTE_WIDTH 1 /* OUT3L_MUTE */ +#define WM5100_OUT3L_VOL_MASK 0x00FF /* OUT3L_VOL - [7:0] */ +#define WM5100_OUT3L_VOL_SHIFT 0 /* OUT3L_VOL - [7:0] */ +#define WM5100_OUT3L_VOL_WIDTH 8 /* OUT3L_VOL - [7:0] */ + +/* + * R1157 (0x485) - DAC Digital Volume 3R + */ +#define WM5100_OUT_VU 0x0200 /* OUT_VU */ +#define WM5100_OUT_VU_MASK 0x0200 /* OUT_VU */ +#define WM5100_OUT_VU_SHIFT 9 /* OUT_VU */ +#define WM5100_OUT_VU_WIDTH 1 /* OUT_VU */ +#define WM5100_OUT3R_MUTE 0x0100 /* OUT3R_MUTE */ +#define WM5100_OUT3R_MUTE_MASK 0x0100 /* OUT3R_MUTE */ +#define WM5100_OUT3R_MUTE_SHIFT 8 /* OUT3R_MUTE */ +#define WM5100_OUT3R_MUTE_WIDTH 1 /* OUT3R_MUTE */ +#define WM5100_OUT3R_VOL_MASK 0x00FF /* OUT3R_VOL - [7:0] */ +#define WM5100_OUT3R_VOL_SHIFT 0 /* OUT3R_VOL - [7:0] */ +#define WM5100_OUT3R_VOL_WIDTH 8 /* OUT3R_VOL - [7:0] */ + +/* + * R1158 (0x486) - DAC Digital Volume 4L + */ +#define WM5100_OUT_VU 0x0200 /* OUT_VU */ +#define WM5100_OUT_VU_MASK 0x0200 /* OUT_VU */ +#define WM5100_OUT_VU_SHIFT 9 /* OUT_VU */ +#define WM5100_OUT_VU_WIDTH 1 /* OUT_VU */ +#define WM5100_OUT4L_MUTE 0x0100 /* OUT4L_MUTE */ +#define WM5100_OUT4L_MUTE_MASK 0x0100 /* OUT4L_MUTE */ +#define WM5100_OUT4L_MUTE_SHIFT 8 /* OUT4L_MUTE */ +#define WM5100_OUT4L_MUTE_WIDTH 1 /* OUT4L_MUTE */ +#define WM5100_OUT4L_VOL_MASK 0x00FF /* OUT4L_VOL - [7:0] */ +#define WM5100_OUT4L_VOL_SHIFT 0 /* OUT4L_VOL - [7:0] */ +#define WM5100_OUT4L_VOL_WIDTH 8 /* OUT4L_VOL - [7:0] */ + +/* + * R1159 (0x487) - DAC Digital Volume 4R + */ +#define WM5100_OUT_VU 0x0200 /* OUT_VU */ +#define WM5100_OUT_VU_MASK 0x0200 /* OUT_VU */ +#define WM5100_OUT_VU_SHIFT 9 /* OUT_VU */ +#define WM5100_OUT_VU_WIDTH 1 /* OUT_VU */ +#define WM5100_OUT4R_MUTE 0x0100 /* OUT4R_MUTE */ +#define WM5100_OUT4R_MUTE_MASK 0x0100 /* OUT4R_MUTE */ +#define WM5100_OUT4R_MUTE_SHIFT 8 /* OUT4R_MUTE */ +#define WM5100_OUT4R_MUTE_WIDTH 1 /* OUT4R_MUTE */ +#define WM5100_OUT4R_VOL_MASK 0x00FF /* OUT4R_VOL - [7:0] */ +#define WM5100_OUT4R_VOL_SHIFT 0 /* OUT4R_VOL - [7:0] */ +#define WM5100_OUT4R_VOL_WIDTH 8 /* OUT4R_VOL - [7:0] */ + +/* + * R1160 (0x488) - DAC Digital Volume 5L + */ +#define WM5100_OUT_VU 0x0200 /* OUT_VU */ +#define WM5100_OUT_VU_MASK 0x0200 /* OUT_VU */ +#define WM5100_OUT_VU_SHIFT 9 /* OUT_VU */ +#define WM5100_OUT_VU_WIDTH 1 /* OUT_VU */ +#define WM5100_OUT5L_MUTE 0x0100 /* OUT5L_MUTE */ +#define WM5100_OUT5L_MUTE_MASK 0x0100 /* OUT5L_MUTE */ +#define WM5100_OUT5L_MUTE_SHIFT 8 /* OUT5L_MUTE */ +#define WM5100_OUT5L_MUTE_WIDTH 1 /* OUT5L_MUTE */ +#define WM5100_OUT5L_VOL_MASK 0x00FF /* OUT5L_VOL - [7:0] */ +#define WM5100_OUT5L_VOL_SHIFT 0 /* OUT5L_VOL - [7:0] */ +#define WM5100_OUT5L_VOL_WIDTH 8 /* OUT5L_VOL - [7:0] */ + +/* + * R1161 (0x489) - DAC Digital Volume 5R + */ +#define WM5100_OUT_VU 0x0200 /* OUT_VU */ +#define WM5100_OUT_VU_MASK 0x0200 /* OUT_VU */ +#define WM5100_OUT_VU_SHIFT 9 /* OUT_VU */ +#define WM5100_OUT_VU_WIDTH 1 /* OUT_VU */ +#define WM5100_OUT5R_MUTE 0x0100 /* OUT5R_MUTE */ +#define WM5100_OUT5R_MUTE_MASK 0x0100 /* OUT5R_MUTE */ +#define WM5100_OUT5R_MUTE_SHIFT 8 /* OUT5R_MUTE */ +#define WM5100_OUT5R_MUTE_WIDTH 1 /* OUT5R_MUTE */ +#define WM5100_OUT5R_VOL_MASK 0x00FF /* OUT5R_VOL - [7:0] */ +#define WM5100_OUT5R_VOL_SHIFT 0 /* OUT5R_VOL - [7:0] */ +#define WM5100_OUT5R_VOL_WIDTH 8 /* OUT5R_VOL - [7:0] */ + +/* + * R1162 (0x48A) - DAC Digital Volume 6L + */ +#define WM5100_OUT_VU 0x0200 /* OUT_VU */ +#define WM5100_OUT_VU_MASK 0x0200 /* OUT_VU */ +#define WM5100_OUT_VU_SHIFT 9 /* OUT_VU */ +#define WM5100_OUT_VU_WIDTH 1 /* OUT_VU */ +#define WM5100_OUT6L_MUTE 0x0100 /* OUT6L_MUTE */ +#define WM5100_OUT6L_MUTE_MASK 0x0100 /* OUT6L_MUTE */ +#define WM5100_OUT6L_MUTE_SHIFT 8 /* OUT6L_MUTE */ +#define WM5100_OUT6L_MUTE_WIDTH 1 /* OUT6L_MUTE */ +#define WM5100_OUT6L_VOL_MASK 0x00FF /* OUT6L_VOL - [7:0] */ +#define WM5100_OUT6L_VOL_SHIFT 0 /* OUT6L_VOL - [7:0] */ +#define WM5100_OUT6L_VOL_WIDTH 8 /* OUT6L_VOL - [7:0] */ + +/* + * R1163 (0x48B) - DAC Digital Volume 6R + */ +#define WM5100_OUT_VU 0x0200 /* OUT_VU */ +#define WM5100_OUT_VU_MASK 0x0200 /* OUT_VU */ +#define WM5100_OUT_VU_SHIFT 9 /* OUT_VU */ +#define WM5100_OUT_VU_WIDTH 1 /* OUT_VU */ +#define WM5100_OUT6R_MUTE 0x0100 /* OUT6R_MUTE */ +#define WM5100_OUT6R_MUTE_MASK 0x0100 /* OUT6R_MUTE */ +#define WM5100_OUT6R_MUTE_SHIFT 8 /* OUT6R_MUTE */ +#define WM5100_OUT6R_MUTE_WIDTH 1 /* OUT6R_MUTE */ +#define WM5100_OUT6R_VOL_MASK 0x00FF /* OUT6R_VOL - [7:0] */ +#define WM5100_OUT6R_VOL_SHIFT 0 /* OUT6R_VOL - [7:0] */ +#define WM5100_OUT6R_VOL_WIDTH 8 /* OUT6R_VOL - [7:0] */ + +/* + * R1216 (0x4C0) - PDM SPK1 CTRL 1 + */ +#define WM5100_SPK1R_MUTE 0x2000 /* SPK1R_MUTE */ +#define WM5100_SPK1R_MUTE_MASK 0x2000 /* SPK1R_MUTE */ +#define WM5100_SPK1R_MUTE_SHIFT 13 /* SPK1R_MUTE */ +#define WM5100_SPK1R_MUTE_WIDTH 1 /* SPK1R_MUTE */ +#define WM5100_SPK1L_MUTE 0x1000 /* SPK1L_MUTE */ +#define WM5100_SPK1L_MUTE_MASK 0x1000 /* SPK1L_MUTE */ +#define WM5100_SPK1L_MUTE_SHIFT 12 /* SPK1L_MUTE */ +#define WM5100_SPK1L_MUTE_WIDTH 1 /* SPK1L_MUTE */ +#define WM5100_SPK1_MUTE_ENDIAN 0x0100 /* SPK1_MUTE_ENDIAN */ +#define WM5100_SPK1_MUTE_ENDIAN_MASK 0x0100 /* SPK1_MUTE_ENDIAN */ +#define WM5100_SPK1_MUTE_ENDIAN_SHIFT 8 /* SPK1_MUTE_ENDIAN */ +#define WM5100_SPK1_MUTE_ENDIAN_WIDTH 1 /* SPK1_MUTE_ENDIAN */ +#define WM5100_SPK1_MUTE_SEQ1_MASK 0x00FF /* SPK1_MUTE_SEQ1 - [7:0] */ +#define WM5100_SPK1_MUTE_SEQ1_SHIFT 0 /* SPK1_MUTE_SEQ1 - [7:0] */ +#define WM5100_SPK1_MUTE_SEQ1_WIDTH 8 /* SPK1_MUTE_SEQ1 - [7:0] */ + +/* + * R1217 (0x4C1) - PDM SPK1 CTRL 2 + */ +#define WM5100_SPK1_FMT 0x0001 /* SPK1_FMT */ +#define WM5100_SPK1_FMT_MASK 0x0001 /* SPK1_FMT */ +#define WM5100_SPK1_FMT_SHIFT 0 /* SPK1_FMT */ +#define WM5100_SPK1_FMT_WIDTH 1 /* SPK1_FMT */ + +/* + * R1218 (0x4C2) - PDM SPK2 CTRL 1 + */ +#define WM5100_SPK2R_MUTE 0x2000 /* SPK2R_MUTE */ +#define WM5100_SPK2R_MUTE_MASK 0x2000 /* SPK2R_MUTE */ +#define WM5100_SPK2R_MUTE_SHIFT 13 /* SPK2R_MUTE */ +#define WM5100_SPK2R_MUTE_WIDTH 1 /* SPK2R_MUTE */ +#define WM5100_SPK2L_MUTE 0x1000 /* SPK2L_MUTE */ +#define WM5100_SPK2L_MUTE_MASK 0x1000 /* SPK2L_MUTE */ +#define WM5100_SPK2L_MUTE_SHIFT 12 /* SPK2L_MUTE */ +#define WM5100_SPK2L_MUTE_WIDTH 1 /* SPK2L_MUTE */ +#define WM5100_SPK2_MUTE_ENDIAN 0x0100 /* SPK2_MUTE_ENDIAN */ +#define WM5100_SPK2_MUTE_ENDIAN_MASK 0x0100 /* SPK2_MUTE_ENDIAN */ +#define WM5100_SPK2_MUTE_ENDIAN_SHIFT 8 /* SPK2_MUTE_ENDIAN */ +#define WM5100_SPK2_MUTE_ENDIAN_WIDTH 1 /* SPK2_MUTE_ENDIAN */ +#define WM5100_SPK2_MUTE_SEQ1_MASK 0x00FF /* SPK2_MUTE_SEQ1 - [7:0] */ +#define WM5100_SPK2_MUTE_SEQ1_SHIFT 0 /* SPK2_MUTE_SEQ1 - [7:0] */ +#define WM5100_SPK2_MUTE_SEQ1_WIDTH 8 /* SPK2_MUTE_SEQ1 - [7:0] */ + +/* + * R1219 (0x4C3) - PDM SPK2 CTRL 2 + */ +#define WM5100_SPK2_FMT 0x0001 /* SPK2_FMT */ +#define WM5100_SPK2_FMT_MASK 0x0001 /* SPK2_FMT */ +#define WM5100_SPK2_FMT_SHIFT 0 /* SPK2_FMT */ +#define WM5100_SPK2_FMT_WIDTH 1 /* SPK2_FMT */ + +/* + * R1280 (0x500) - Audio IF 1_1 + */ +#define WM5100_AIF1_BCLK_INV 0x0080 /* AIF1_BCLK_INV */ +#define WM5100_AIF1_BCLK_INV_MASK 0x0080 /* AIF1_BCLK_INV */ +#define WM5100_AIF1_BCLK_INV_SHIFT 7 /* AIF1_BCLK_INV */ +#define WM5100_AIF1_BCLK_INV_WIDTH 1 /* AIF1_BCLK_INV */ +#define WM5100_AIF1_BCLK_FRC 0x0040 /* AIF1_BCLK_FRC */ +#define WM5100_AIF1_BCLK_FRC_MASK 0x0040 /* AIF1_BCLK_FRC */ +#define WM5100_AIF1_BCLK_FRC_SHIFT 6 /* AIF1_BCLK_FRC */ +#define WM5100_AIF1_BCLK_FRC_WIDTH 1 /* AIF1_BCLK_FRC */ +#define WM5100_AIF1_BCLK_MSTR 0x0020 /* AIF1_BCLK_MSTR */ +#define WM5100_AIF1_BCLK_MSTR_MASK 0x0020 /* AIF1_BCLK_MSTR */ +#define WM5100_AIF1_BCLK_MSTR_SHIFT 5 /* AIF1_BCLK_MSTR */ +#define WM5100_AIF1_BCLK_MSTR_WIDTH 1 /* AIF1_BCLK_MSTR */ +#define WM5100_AIF1_BCLK_FREQ_MASK 0x001F /* AIF1_BCLK_FREQ - [4:0] */ +#define WM5100_AIF1_BCLK_FREQ_SHIFT 0 /* AIF1_BCLK_FREQ - [4:0] */ +#define WM5100_AIF1_BCLK_FREQ_WIDTH 5 /* AIF1_BCLK_FREQ - [4:0] */ + +/* + * R1281 (0x501) - Audio IF 1_2 + */ +#define WM5100_AIF1TX_DAT_TRI 0x0020 /* AIF1TX_DAT_TRI */ +#define WM5100_AIF1TX_DAT_TRI_MASK 0x0020 /* AIF1TX_DAT_TRI */ +#define WM5100_AIF1TX_DAT_TRI_SHIFT 5 /* AIF1TX_DAT_TRI */ +#define WM5100_AIF1TX_DAT_TRI_WIDTH 1 /* AIF1TX_DAT_TRI */ +#define WM5100_AIF1TX_LRCLK_SRC 0x0008 /* AIF1TX_LRCLK_SRC */ +#define WM5100_AIF1TX_LRCLK_SRC_MASK 0x0008 /* AIF1TX_LRCLK_SRC */ +#define WM5100_AIF1TX_LRCLK_SRC_SHIFT 3 /* AIF1TX_LRCLK_SRC */ +#define WM5100_AIF1TX_LRCLK_SRC_WIDTH 1 /* AIF1TX_LRCLK_SRC */ +#define WM5100_AIF1TX_LRCLK_INV 0x0004 /* AIF1TX_LRCLK_INV */ +#define WM5100_AIF1TX_LRCLK_INV_MASK 0x0004 /* AIF1TX_LRCLK_INV */ +#define WM5100_AIF1TX_LRCLK_INV_SHIFT 2 /* AIF1TX_LRCLK_INV */ +#define WM5100_AIF1TX_LRCLK_INV_WIDTH 1 /* AIF1TX_LRCLK_INV */ +#define WM5100_AIF1TX_LRCLK_FRC 0x0002 /* AIF1TX_LRCLK_FRC */ +#define WM5100_AIF1TX_LRCLK_FRC_MASK 0x0002 /* AIF1TX_LRCLK_FRC */ +#define WM5100_AIF1TX_LRCLK_FRC_SHIFT 1 /* AIF1TX_LRCLK_FRC */ +#define WM5100_AIF1TX_LRCLK_FRC_WIDTH 1 /* AIF1TX_LRCLK_FRC */ +#define WM5100_AIF1TX_LRCLK_MSTR 0x0001 /* AIF1TX_LRCLK_MSTR */ +#define WM5100_AIF1TX_LRCLK_MSTR_MASK 0x0001 /* AIF1TX_LRCLK_MSTR */ +#define WM5100_AIF1TX_LRCLK_MSTR_SHIFT 0 /* AIF1TX_LRCLK_MSTR */ +#define WM5100_AIF1TX_LRCLK_MSTR_WIDTH 1 /* AIF1TX_LRCLK_MSTR */ + +/* + * R1282 (0x502) - Audio IF 1_3 + */ +#define WM5100_AIF1RX_LRCLK_INV 0x0004 /* AIF1RX_LRCLK_INV */ +#define WM5100_AIF1RX_LRCLK_INV_MASK 0x0004 /* AIF1RX_LRCLK_INV */ +#define WM5100_AIF1RX_LRCLK_INV_SHIFT 2 /* AIF1RX_LRCLK_INV */ +#define WM5100_AIF1RX_LRCLK_INV_WIDTH 1 /* AIF1RX_LRCLK_INV */ +#define WM5100_AIF1RX_LRCLK_FRC 0x0002 /* AIF1RX_LRCLK_FRC */ +#define WM5100_AIF1RX_LRCLK_FRC_MASK 0x0002 /* AIF1RX_LRCLK_FRC */ +#define WM5100_AIF1RX_LRCLK_FRC_SHIFT 1 /* AIF1RX_LRCLK_FRC */ +#define WM5100_AIF1RX_LRCLK_FRC_WIDTH 1 /* AIF1RX_LRCLK_FRC */ +#define WM5100_AIF1RX_LRCLK_MSTR 0x0001 /* AIF1RX_LRCLK_MSTR */ +#define WM5100_AIF1RX_LRCLK_MSTR_MASK 0x0001 /* AIF1RX_LRCLK_MSTR */ +#define WM5100_AIF1RX_LRCLK_MSTR_SHIFT 0 /* AIF1RX_LRCLK_MSTR */ +#define WM5100_AIF1RX_LRCLK_MSTR_WIDTH 1 /* AIF1RX_LRCLK_MSTR */ + +/* + * R1283 (0x503) - Audio IF 1_4 + */ +#define WM5100_AIF1_TRI 0x0040 /* AIF1_TRI */ +#define WM5100_AIF1_TRI_MASK 0x0040 /* AIF1_TRI */ +#define WM5100_AIF1_TRI_SHIFT 6 /* AIF1_TRI */ +#define WM5100_AIF1_TRI_WIDTH 1 /* AIF1_TRI */ +#define WM5100_AIF1_RATE_MASK 0x0003 /* AIF1_RATE - [1:0] */ +#define WM5100_AIF1_RATE_SHIFT 0 /* AIF1_RATE - [1:0] */ +#define WM5100_AIF1_RATE_WIDTH 2 /* AIF1_RATE - [1:0] */ + +/* + * R1284 (0x504) - Audio IF 1_5 + */ +#define WM5100_AIF1_FMT_MASK 0x0007 /* AIF1_FMT - [2:0] */ +#define WM5100_AIF1_FMT_SHIFT 0 /* AIF1_FMT - [2:0] */ +#define WM5100_AIF1_FMT_WIDTH 3 /* AIF1_FMT - [2:0] */ + +/* + * R1285 (0x505) - Audio IF 1_6 + */ +#define WM5100_AIF1TX_BCPF_MASK 0x1FFF /* AIF1TX_BCPF - [12:0] */ +#define WM5100_AIF1TX_BCPF_SHIFT 0 /* AIF1TX_BCPF - [12:0] */ +#define WM5100_AIF1TX_BCPF_WIDTH 13 /* AIF1TX_BCPF - [12:0] */ + +/* + * R1286 (0x506) - Audio IF 1_7 + */ +#define WM5100_AIF1RX_BCPF_MASK 0x1FFF /* AIF1RX_BCPF - [12:0] */ +#define WM5100_AIF1RX_BCPF_SHIFT 0 /* AIF1RX_BCPF - [12:0] */ +#define WM5100_AIF1RX_BCPF_WIDTH 13 /* AIF1RX_BCPF - [12:0] */ + +/* + * R1287 (0x507) - Audio IF 1_8 + */ +#define WM5100_AIF1TX_WL_MASK 0x3F00 /* AIF1TX_WL - [13:8] */ +#define WM5100_AIF1TX_WL_SHIFT 8 /* AIF1TX_WL - [13:8] */ +#define WM5100_AIF1TX_WL_WIDTH 6 /* AIF1TX_WL - [13:8] */ +#define WM5100_AIF1TX_SLOT_LEN_MASK 0x00FF /* AIF1TX_SLOT_LEN - [7:0] */ +#define WM5100_AIF1TX_SLOT_LEN_SHIFT 0 /* AIF1TX_SLOT_LEN - [7:0] */ +#define WM5100_AIF1TX_SLOT_LEN_WIDTH 8 /* AIF1TX_SLOT_LEN - [7:0] */ + +/* + * R1288 (0x508) - Audio IF 1_9 + */ +#define WM5100_AIF1RX_WL_MASK 0x3F00 /* AIF1RX_WL - [13:8] */ +#define WM5100_AIF1RX_WL_SHIFT 8 /* AIF1RX_WL - [13:8] */ +#define WM5100_AIF1RX_WL_WIDTH 6 /* AIF1RX_WL - [13:8] */ +#define WM5100_AIF1RX_SLOT_LEN_MASK 0x00FF /* AIF1RX_SLOT_LEN - [7:0] */ +#define WM5100_AIF1RX_SLOT_LEN_SHIFT 0 /* AIF1RX_SLOT_LEN - [7:0] */ +#define WM5100_AIF1RX_SLOT_LEN_WIDTH 8 /* AIF1RX_SLOT_LEN - [7:0] */ + +/* + * R1289 (0x509) - Audio IF 1_10 + */ +#define WM5100_AIF1TX1_SLOT_MASK 0x003F /* AIF1TX1_SLOT - [5:0] */ +#define WM5100_AIF1TX1_SLOT_SHIFT 0 /* AIF1TX1_SLOT - [5:0] */ +#define WM5100_AIF1TX1_SLOT_WIDTH 6 /* AIF1TX1_SLOT - [5:0] */ + +/* + * R1290 (0x50A) - Audio IF 1_11 + */ +#define WM5100_AIF1TX2_SLOT_MASK 0x003F /* AIF1TX2_SLOT - [5:0] */ +#define WM5100_AIF1TX2_SLOT_SHIFT 0 /* AIF1TX2_SLOT - [5:0] */ +#define WM5100_AIF1TX2_SLOT_WIDTH 6 /* AIF1TX2_SLOT - [5:0] */ + +/* + * R1291 (0x50B) - Audio IF 1_12 + */ +#define WM5100_AIF1TX3_SLOT_MASK 0x003F /* AIF1TX3_SLOT - [5:0] */ +#define WM5100_AIF1TX3_SLOT_SHIFT 0 /* AIF1TX3_SLOT - [5:0] */ +#define WM5100_AIF1TX3_SLOT_WIDTH 6 /* AIF1TX3_SLOT - [5:0] */ + +/* + * R1292 (0x50C) - Audio IF 1_13 + */ +#define WM5100_AIF1TX4_SLOT_MASK 0x003F /* AIF1TX4_SLOT - [5:0] */ +#define WM5100_AIF1TX4_SLOT_SHIFT 0 /* AIF1TX4_SLOT - [5:0] */ +#define WM5100_AIF1TX4_SLOT_WIDTH 6 /* AIF1TX4_SLOT - [5:0] */ + +/* + * R1293 (0x50D) - Audio IF 1_14 + */ +#define WM5100_AIF1TX5_SLOT_MASK 0x003F /* AIF1TX5_SLOT - [5:0] */ +#define WM5100_AIF1TX5_SLOT_SHIFT 0 /* AIF1TX5_SLOT - [5:0] */ +#define WM5100_AIF1TX5_SLOT_WIDTH 6 /* AIF1TX5_SLOT - [5:0] */ + +/* + * R1294 (0x50E) - Audio IF 1_15 + */ +#define WM5100_AIF1TX6_SLOT_MASK 0x003F /* AIF1TX6_SLOT - [5:0] */ +#define WM5100_AIF1TX6_SLOT_SHIFT 0 /* AIF1TX6_SLOT - [5:0] */ +#define WM5100_AIF1TX6_SLOT_WIDTH 6 /* AIF1TX6_SLOT - [5:0] */ + +/* + * R1295 (0x50F) - Audio IF 1_16 + */ +#define WM5100_AIF1TX7_SLOT_MASK 0x003F /* AIF1TX7_SLOT - [5:0] */ +#define WM5100_AIF1TX7_SLOT_SHIFT 0 /* AIF1TX7_SLOT - [5:0] */ +#define WM5100_AIF1TX7_SLOT_WIDTH 6 /* AIF1TX7_SLOT - [5:0] */ + +/* + * R1296 (0x510) - Audio IF 1_17 + */ +#define WM5100_AIF1TX8_SLOT_MASK 0x003F /* AIF1TX8_SLOT - [5:0] */ +#define WM5100_AIF1TX8_SLOT_SHIFT 0 /* AIF1TX8_SLOT - [5:0] */ +#define WM5100_AIF1TX8_SLOT_WIDTH 6 /* AIF1TX8_SLOT - [5:0] */ + +/* + * R1297 (0x511) - Audio IF 1_18 + */ +#define WM5100_AIF1RX1_SLOT_MASK 0x003F /* AIF1RX1_SLOT - [5:0] */ +#define WM5100_AIF1RX1_SLOT_SHIFT 0 /* AIF1RX1_SLOT - [5:0] */ +#define WM5100_AIF1RX1_SLOT_WIDTH 6 /* AIF1RX1_SLOT - [5:0] */ + +/* + * R1298 (0x512) - Audio IF 1_19 + */ +#define WM5100_AIF1RX2_SLOT_MASK 0x003F /* AIF1RX2_SLOT - [5:0] */ +#define WM5100_AIF1RX2_SLOT_SHIFT 0 /* AIF1RX2_SLOT - [5:0] */ +#define WM5100_AIF1RX2_SLOT_WIDTH 6 /* AIF1RX2_SLOT - [5:0] */ + +/* + * R1299 (0x513) - Audio IF 1_20 + */ +#define WM5100_AIF1RX3_SLOT_MASK 0x003F /* AIF1RX3_SLOT - [5:0] */ +#define WM5100_AIF1RX3_SLOT_SHIFT 0 /* AIF1RX3_SLOT - [5:0] */ +#define WM5100_AIF1RX3_SLOT_WIDTH 6 /* AIF1RX3_SLOT - [5:0] */ + +/* + * R1300 (0x514) - Audio IF 1_21 + */ +#define WM5100_AIF1RX4_SLOT_MASK 0x003F /* AIF1RX4_SLOT - [5:0] */ +#define WM5100_AIF1RX4_SLOT_SHIFT 0 /* AIF1RX4_SLOT - [5:0] */ +#define WM5100_AIF1RX4_SLOT_WIDTH 6 /* AIF1RX4_SLOT - [5:0] */ + +/* + * R1301 (0x515) - Audio IF 1_22 + */ +#define WM5100_AIF1RX5_SLOT_MASK 0x003F /* AIF1RX5_SLOT - [5:0] */ +#define WM5100_AIF1RX5_SLOT_SHIFT 0 /* AIF1RX5_SLOT - [5:0] */ +#define WM5100_AIF1RX5_SLOT_WIDTH 6 /* AIF1RX5_SLOT - [5:0] */ + +/* + * R1302 (0x516) - Audio IF 1_23 + */ +#define WM5100_AIF1RX6_SLOT_MASK 0x003F /* AIF1RX6_SLOT - [5:0] */ +#define WM5100_AIF1RX6_SLOT_SHIFT 0 /* AIF1RX6_SLOT - [5:0] */ +#define WM5100_AIF1RX6_SLOT_WIDTH 6 /* AIF1RX6_SLOT - [5:0] */ + +/* + * R1303 (0x517) - Audio IF 1_24 + */ +#define WM5100_AIF1RX7_SLOT_MASK 0x003F /* AIF1RX7_SLOT - [5:0] */ +#define WM5100_AIF1RX7_SLOT_SHIFT 0 /* AIF1RX7_SLOT - [5:0] */ +#define WM5100_AIF1RX7_SLOT_WIDTH 6 /* AIF1RX7_SLOT - [5:0] */ + +/* + * R1304 (0x518) - Audio IF 1_25 + */ +#define WM5100_AIF1RX8_SLOT_MASK 0x003F /* AIF1RX8_SLOT - [5:0] */ +#define WM5100_AIF1RX8_SLOT_SHIFT 0 /* AIF1RX8_SLOT - [5:0] */ +#define WM5100_AIF1RX8_SLOT_WIDTH 6 /* AIF1RX8_SLOT - [5:0] */ + +/* + * R1305 (0x519) - Audio IF 1_26 + */ +#define WM5100_AIF1TX8_ENA 0x0080 /* AIF1TX8_ENA */ +#define WM5100_AIF1TX8_ENA_MASK 0x0080 /* AIF1TX8_ENA */ +#define WM5100_AIF1TX8_ENA_SHIFT 7 /* AIF1TX8_ENA */ +#define WM5100_AIF1TX8_ENA_WIDTH 1 /* AIF1TX8_ENA */ +#define WM5100_AIF1TX7_ENA 0x0040 /* AIF1TX7_ENA */ +#define WM5100_AIF1TX7_ENA_MASK 0x0040 /* AIF1TX7_ENA */ +#define WM5100_AIF1TX7_ENA_SHIFT 6 /* AIF1TX7_ENA */ +#define WM5100_AIF1TX7_ENA_WIDTH 1 /* AIF1TX7_ENA */ +#define WM5100_AIF1TX6_ENA 0x0020 /* AIF1TX6_ENA */ +#define WM5100_AIF1TX6_ENA_MASK 0x0020 /* AIF1TX6_ENA */ +#define WM5100_AIF1TX6_ENA_SHIFT 5 /* AIF1TX6_ENA */ +#define WM5100_AIF1TX6_ENA_WIDTH 1 /* AIF1TX6_ENA */ +#define WM5100_AIF1TX5_ENA 0x0010 /* AIF1TX5_ENA */ +#define WM5100_AIF1TX5_ENA_MASK 0x0010 /* AIF1TX5_ENA */ +#define WM5100_AIF1TX5_ENA_SHIFT 4 /* AIF1TX5_ENA */ +#define WM5100_AIF1TX5_ENA_WIDTH 1 /* AIF1TX5_ENA */ +#define WM5100_AIF1TX4_ENA 0x0008 /* AIF1TX4_ENA */ +#define WM5100_AIF1TX4_ENA_MASK 0x0008 /* AIF1TX4_ENA */ +#define WM5100_AIF1TX4_ENA_SHIFT 3 /* AIF1TX4_ENA */ +#define WM5100_AIF1TX4_ENA_WIDTH 1 /* AIF1TX4_ENA */ +#define WM5100_AIF1TX3_ENA 0x0004 /* AIF1TX3_ENA */ +#define WM5100_AIF1TX3_ENA_MASK 0x0004 /* AIF1TX3_ENA */ +#define WM5100_AIF1TX3_ENA_SHIFT 2 /* AIF1TX3_ENA */ +#define WM5100_AIF1TX3_ENA_WIDTH 1 /* AIF1TX3_ENA */ +#define WM5100_AIF1TX2_ENA 0x0002 /* AIF1TX2_ENA */ +#define WM5100_AIF1TX2_ENA_MASK 0x0002 /* AIF1TX2_ENA */ +#define WM5100_AIF1TX2_ENA_SHIFT 1 /* AIF1TX2_ENA */ +#define WM5100_AIF1TX2_ENA_WIDTH 1 /* AIF1TX2_ENA */ +#define WM5100_AIF1TX1_ENA 0x0001 /* AIF1TX1_ENA */ +#define WM5100_AIF1TX1_ENA_MASK 0x0001 /* AIF1TX1_ENA */ +#define WM5100_AIF1TX1_ENA_SHIFT 0 /* AIF1TX1_ENA */ +#define WM5100_AIF1TX1_ENA_WIDTH 1 /* AIF1TX1_ENA */ + +/* + * R1306 (0x51A) - Audio IF 1_27 + */ +#define WM5100_AIF1RX8_ENA 0x0080 /* AIF1RX8_ENA */ +#define WM5100_AIF1RX8_ENA_MASK 0x0080 /* AIF1RX8_ENA */ +#define WM5100_AIF1RX8_ENA_SHIFT 7 /* AIF1RX8_ENA */ +#define WM5100_AIF1RX8_ENA_WIDTH 1 /* AIF1RX8_ENA */ +#define WM5100_AIF1RX7_ENA 0x0040 /* AIF1RX7_ENA */ +#define WM5100_AIF1RX7_ENA_MASK 0x0040 /* AIF1RX7_ENA */ +#define WM5100_AIF1RX7_ENA_SHIFT 6 /* AIF1RX7_ENA */ +#define WM5100_AIF1RX7_ENA_WIDTH 1 /* AIF1RX7_ENA */ +#define WM5100_AIF1RX6_ENA 0x0020 /* AIF1RX6_ENA */ +#define WM5100_AIF1RX6_ENA_MASK 0x0020 /* AIF1RX6_ENA */ +#define WM5100_AIF1RX6_ENA_SHIFT 5 /* AIF1RX6_ENA */ +#define WM5100_AIF1RX6_ENA_WIDTH 1 /* AIF1RX6_ENA */ +#define WM5100_AIF1RX5_ENA 0x0010 /* AIF1RX5_ENA */ +#define WM5100_AIF1RX5_ENA_MASK 0x0010 /* AIF1RX5_ENA */ +#define WM5100_AIF1RX5_ENA_SHIFT 4 /* AIF1RX5_ENA */ +#define WM5100_AIF1RX5_ENA_WIDTH 1 /* AIF1RX5_ENA */ +#define WM5100_AIF1RX4_ENA 0x0008 /* AIF1RX4_ENA */ +#define WM5100_AIF1RX4_ENA_MASK 0x0008 /* AIF1RX4_ENA */ +#define WM5100_AIF1RX4_ENA_SHIFT 3 /* AIF1RX4_ENA */ +#define WM5100_AIF1RX4_ENA_WIDTH 1 /* AIF1RX4_ENA */ +#define WM5100_AIF1RX3_ENA 0x0004 /* AIF1RX3_ENA */ +#define WM5100_AIF1RX3_ENA_MASK 0x0004 /* AIF1RX3_ENA */ +#define WM5100_AIF1RX3_ENA_SHIFT 2 /* AIF1RX3_ENA */ +#define WM5100_AIF1RX3_ENA_WIDTH 1 /* AIF1RX3_ENA */ +#define WM5100_AIF1RX2_ENA 0x0002 /* AIF1RX2_ENA */ +#define WM5100_AIF1RX2_ENA_MASK 0x0002 /* AIF1RX2_ENA */ +#define WM5100_AIF1RX2_ENA_SHIFT 1 /* AIF1RX2_ENA */ +#define WM5100_AIF1RX2_ENA_WIDTH 1 /* AIF1RX2_ENA */ +#define WM5100_AIF1RX1_ENA 0x0001 /* AIF1RX1_ENA */ +#define WM5100_AIF1RX1_ENA_MASK 0x0001 /* AIF1RX1_ENA */ +#define WM5100_AIF1RX1_ENA_SHIFT 0 /* AIF1RX1_ENA */ +#define WM5100_AIF1RX1_ENA_WIDTH 1 /* AIF1RX1_ENA */ + +/* + * R1344 (0x540) - Audio IF 2_1 + */ +#define WM5100_AIF2_BCLK_INV 0x0080 /* AIF2_BCLK_INV */ +#define WM5100_AIF2_BCLK_INV_MASK 0x0080 /* AIF2_BCLK_INV */ +#define WM5100_AIF2_BCLK_INV_SHIFT 7 /* AIF2_BCLK_INV */ +#define WM5100_AIF2_BCLK_INV_WIDTH 1 /* AIF2_BCLK_INV */ +#define WM5100_AIF2_BCLK_FRC 0x0040 /* AIF2_BCLK_FRC */ +#define WM5100_AIF2_BCLK_FRC_MASK 0x0040 /* AIF2_BCLK_FRC */ +#define WM5100_AIF2_BCLK_FRC_SHIFT 6 /* AIF2_BCLK_FRC */ +#define WM5100_AIF2_BCLK_FRC_WIDTH 1 /* AIF2_BCLK_FRC */ +#define WM5100_AIF2_BCLK_MSTR 0x0020 /* AIF2_BCLK_MSTR */ +#define WM5100_AIF2_BCLK_MSTR_MASK 0x0020 /* AIF2_BCLK_MSTR */ +#define WM5100_AIF2_BCLK_MSTR_SHIFT 5 /* AIF2_BCLK_MSTR */ +#define WM5100_AIF2_BCLK_MSTR_WIDTH 1 /* AIF2_BCLK_MSTR */ +#define WM5100_AIF2_BCLK_FREQ_MASK 0x001F /* AIF2_BCLK_FREQ - [4:0] */ +#define WM5100_AIF2_BCLK_FREQ_SHIFT 0 /* AIF2_BCLK_FREQ - [4:0] */ +#define WM5100_AIF2_BCLK_FREQ_WIDTH 5 /* AIF2_BCLK_FREQ - [4:0] */ + +/* + * R1345 (0x541) - Audio IF 2_2 + */ +#define WM5100_AIF2TX_DAT_TRI 0x0020 /* AIF2TX_DAT_TRI */ +#define WM5100_AIF2TX_DAT_TRI_MASK 0x0020 /* AIF2TX_DAT_TRI */ +#define WM5100_AIF2TX_DAT_TRI_SHIFT 5 /* AIF2TX_DAT_TRI */ +#define WM5100_AIF2TX_DAT_TRI_WIDTH 1 /* AIF2TX_DAT_TRI */ +#define WM5100_AIF2TX_LRCLK_SRC 0x0008 /* AIF2TX_LRCLK_SRC */ +#define WM5100_AIF2TX_LRCLK_SRC_MASK 0x0008 /* AIF2TX_LRCLK_SRC */ +#define WM5100_AIF2TX_LRCLK_SRC_SHIFT 3 /* AIF2TX_LRCLK_SRC */ +#define WM5100_AIF2TX_LRCLK_SRC_WIDTH 1 /* AIF2TX_LRCLK_SRC */ +#define WM5100_AIF2TX_LRCLK_INV 0x0004 /* AIF2TX_LRCLK_INV */ +#define WM5100_AIF2TX_LRCLK_INV_MASK 0x0004 /* AIF2TX_LRCLK_INV */ +#define WM5100_AIF2TX_LRCLK_INV_SHIFT 2 /* AIF2TX_LRCLK_INV */ +#define WM5100_AIF2TX_LRCLK_INV_WIDTH 1 /* AIF2TX_LRCLK_INV */ +#define WM5100_AIF2TX_LRCLK_FRC 0x0002 /* AIF2TX_LRCLK_FRC */ +#define WM5100_AIF2TX_LRCLK_FRC_MASK 0x0002 /* AIF2TX_LRCLK_FRC */ +#define WM5100_AIF2TX_LRCLK_FRC_SHIFT 1 /* AIF2TX_LRCLK_FRC */ +#define WM5100_AIF2TX_LRCLK_FRC_WIDTH 1 /* AIF2TX_LRCLK_FRC */ +#define WM5100_AIF2TX_LRCLK_MSTR 0x0001 /* AIF2TX_LRCLK_MSTR */ +#define WM5100_AIF2TX_LRCLK_MSTR_MASK 0x0001 /* AIF2TX_LRCLK_MSTR */ +#define WM5100_AIF2TX_LRCLK_MSTR_SHIFT 0 /* AIF2TX_LRCLK_MSTR */ +#define WM5100_AIF2TX_LRCLK_MSTR_WIDTH 1 /* AIF2TX_LRCLK_MSTR */ + +/* + * R1346 (0x542) - Audio IF 2_3 + */ +#define WM5100_AIF2RX_LRCLK_INV 0x0004 /* AIF2RX_LRCLK_INV */ +#define WM5100_AIF2RX_LRCLK_INV_MASK 0x0004 /* AIF2RX_LRCLK_INV */ +#define WM5100_AIF2RX_LRCLK_INV_SHIFT 2 /* AIF2RX_LRCLK_INV */ +#define WM5100_AIF2RX_LRCLK_INV_WIDTH 1 /* AIF2RX_LRCLK_INV */ +#define WM5100_AIF2RX_LRCLK_FRC 0x0002 /* AIF2RX_LRCLK_FRC */ +#define WM5100_AIF2RX_LRCLK_FRC_MASK 0x0002 /* AIF2RX_LRCLK_FRC */ +#define WM5100_AIF2RX_LRCLK_FRC_SHIFT 1 /* AIF2RX_LRCLK_FRC */ +#define WM5100_AIF2RX_LRCLK_FRC_WIDTH 1 /* AIF2RX_LRCLK_FRC */ +#define WM5100_AIF2RX_LRCLK_MSTR 0x0001 /* AIF2RX_LRCLK_MSTR */ +#define WM5100_AIF2RX_LRCLK_MSTR_MASK 0x0001 /* AIF2RX_LRCLK_MSTR */ +#define WM5100_AIF2RX_LRCLK_MSTR_SHIFT 0 /* AIF2RX_LRCLK_MSTR */ +#define WM5100_AIF2RX_LRCLK_MSTR_WIDTH 1 /* AIF2RX_LRCLK_MSTR */ + +/* + * R1347 (0x543) - Audio IF 2_4 + */ +#define WM5100_AIF2_TRI 0x0040 /* AIF2_TRI */ +#define WM5100_AIF2_TRI_MASK 0x0040 /* AIF2_TRI */ +#define WM5100_AIF2_TRI_SHIFT 6 /* AIF2_TRI */ +#define WM5100_AIF2_TRI_WIDTH 1 /* AIF2_TRI */ +#define WM5100_AIF2_RATE_MASK 0x0003 /* AIF2_RATE - [1:0] */ +#define WM5100_AIF2_RATE_SHIFT 0 /* AIF2_RATE - [1:0] */ +#define WM5100_AIF2_RATE_WIDTH 2 /* AIF2_RATE - [1:0] */ + +/* + * R1348 (0x544) - Audio IF 2_5 + */ +#define WM5100_AIF2_FMT_MASK 0x0007 /* AIF2_FMT - [2:0] */ +#define WM5100_AIF2_FMT_SHIFT 0 /* AIF2_FMT - [2:0] */ +#define WM5100_AIF2_FMT_WIDTH 3 /* AIF2_FMT - [2:0] */ + +/* + * R1349 (0x545) - Audio IF 2_6 + */ +#define WM5100_AIF2TX_BCPF_MASK 0x1FFF /* AIF2TX_BCPF - [12:0] */ +#define WM5100_AIF2TX_BCPF_SHIFT 0 /* AIF2TX_BCPF - [12:0] */ +#define WM5100_AIF2TX_BCPF_WIDTH 13 /* AIF2TX_BCPF - [12:0] */ + +/* + * R1350 (0x546) - Audio IF 2_7 + */ +#define WM5100_AIF2RX_BCPF_MASK 0x1FFF /* AIF2RX_BCPF - [12:0] */ +#define WM5100_AIF2RX_BCPF_SHIFT 0 /* AIF2RX_BCPF - [12:0] */ +#define WM5100_AIF2RX_BCPF_WIDTH 13 /* AIF2RX_BCPF - [12:0] */ + +/* + * R1351 (0x547) - Audio IF 2_8 + */ +#define WM5100_AIF2TX_WL_MASK 0x3F00 /* AIF2TX_WL - [13:8] */ +#define WM5100_AIF2TX_WL_SHIFT 8 /* AIF2TX_WL - [13:8] */ +#define WM5100_AIF2TX_WL_WIDTH 6 /* AIF2TX_WL - [13:8] */ +#define WM5100_AIF2TX_SLOT_LEN_MASK 0x00FF /* AIF2TX_SLOT_LEN - [7:0] */ +#define WM5100_AIF2TX_SLOT_LEN_SHIFT 0 /* AIF2TX_SLOT_LEN - [7:0] */ +#define WM5100_AIF2TX_SLOT_LEN_WIDTH 8 /* AIF2TX_SLOT_LEN - [7:0] */ + +/* + * R1352 (0x548) - Audio IF 2_9 + */ +#define WM5100_AIF2RX_WL_MASK 0x3F00 /* AIF2RX_WL - [13:8] */ +#define WM5100_AIF2RX_WL_SHIFT 8 /* AIF2RX_WL - [13:8] */ +#define WM5100_AIF2RX_WL_WIDTH 6 /* AIF2RX_WL - [13:8] */ +#define WM5100_AIF2RX_SLOT_LEN_MASK 0x00FF /* AIF2RX_SLOT_LEN - [7:0] */ +#define WM5100_AIF2RX_SLOT_LEN_SHIFT 0 /* AIF2RX_SLOT_LEN - [7:0] */ +#define WM5100_AIF2RX_SLOT_LEN_WIDTH 8 /* AIF2RX_SLOT_LEN - [7:0] */ + +/* + * R1353 (0x549) - Audio IF 2_10 + */ +#define WM5100_AIF2TX1_SLOT_MASK 0x003F /* AIF2TX1_SLOT - [5:0] */ +#define WM5100_AIF2TX1_SLOT_SHIFT 0 /* AIF2TX1_SLOT - [5:0] */ +#define WM5100_AIF2TX1_SLOT_WIDTH 6 /* AIF2TX1_SLOT - [5:0] */ + +/* + * R1354 (0x54A) - Audio IF 2_11 + */ +#define WM5100_AIF2TX2_SLOT_MASK 0x003F /* AIF2TX2_SLOT - [5:0] */ +#define WM5100_AIF2TX2_SLOT_SHIFT 0 /* AIF2TX2_SLOT - [5:0] */ +#define WM5100_AIF2TX2_SLOT_WIDTH 6 /* AIF2TX2_SLOT - [5:0] */ + +/* + * R1361 (0x551) - Audio IF 2_18 + */ +#define WM5100_AIF2RX1_SLOT_MASK 0x003F /* AIF2RX1_SLOT - [5:0] */ +#define WM5100_AIF2RX1_SLOT_SHIFT 0 /* AIF2RX1_SLOT - [5:0] */ +#define WM5100_AIF2RX1_SLOT_WIDTH 6 /* AIF2RX1_SLOT - [5:0] */ + +/* + * R1362 (0x552) - Audio IF 2_19 + */ +#define WM5100_AIF2RX2_SLOT_MASK 0x003F /* AIF2RX2_SLOT - [5:0] */ +#define WM5100_AIF2RX2_SLOT_SHIFT 0 /* AIF2RX2_SLOT - [5:0] */ +#define WM5100_AIF2RX2_SLOT_WIDTH 6 /* AIF2RX2_SLOT - [5:0] */ + +/* + * R1369 (0x559) - Audio IF 2_26 + */ +#define WM5100_AIF2TX2_ENA 0x0002 /* AIF2TX2_ENA */ +#define WM5100_AIF2TX2_ENA_MASK 0x0002 /* AIF2TX2_ENA */ +#define WM5100_AIF2TX2_ENA_SHIFT 1 /* AIF2TX2_ENA */ +#define WM5100_AIF2TX2_ENA_WIDTH 1 /* AIF2TX2_ENA */ +#define WM5100_AIF2TX1_ENA 0x0001 /* AIF2TX1_ENA */ +#define WM5100_AIF2TX1_ENA_MASK 0x0001 /* AIF2TX1_ENA */ +#define WM5100_AIF2TX1_ENA_SHIFT 0 /* AIF2TX1_ENA */ +#define WM5100_AIF2TX1_ENA_WIDTH 1 /* AIF2TX1_ENA */ + +/* + * R1370 (0x55A) - Audio IF 2_27 + */ +#define WM5100_AIF2RX2_ENA 0x0002 /* AIF2RX2_ENA */ +#define WM5100_AIF2RX2_ENA_MASK 0x0002 /* AIF2RX2_ENA */ +#define WM5100_AIF2RX2_ENA_SHIFT 1 /* AIF2RX2_ENA */ +#define WM5100_AIF2RX2_ENA_WIDTH 1 /* AIF2RX2_ENA */ +#define WM5100_AIF2RX1_ENA 0x0001 /* AIF2RX1_ENA */ +#define WM5100_AIF2RX1_ENA_MASK 0x0001 /* AIF2RX1_ENA */ +#define WM5100_AIF2RX1_ENA_SHIFT 0 /* AIF2RX1_ENA */ +#define WM5100_AIF2RX1_ENA_WIDTH 1 /* AIF2RX1_ENA */ + +/* + * R1408 (0x580) - Audio IF 3_1 + */ +#define WM5100_AIF3_BCLK_INV 0x0080 /* AIF3_BCLK_INV */ +#define WM5100_AIF3_BCLK_INV_MASK 0x0080 /* AIF3_BCLK_INV */ +#define WM5100_AIF3_BCLK_INV_SHIFT 7 /* AIF3_BCLK_INV */ +#define WM5100_AIF3_BCLK_INV_WIDTH 1 /* AIF3_BCLK_INV */ +#define WM5100_AIF3_BCLK_FRC 0x0040 /* AIF3_BCLK_FRC */ +#define WM5100_AIF3_BCLK_FRC_MASK 0x0040 /* AIF3_BCLK_FRC */ +#define WM5100_AIF3_BCLK_FRC_SHIFT 6 /* AIF3_BCLK_FRC */ +#define WM5100_AIF3_BCLK_FRC_WIDTH 1 /* AIF3_BCLK_FRC */ +#define WM5100_AIF3_BCLK_MSTR 0x0020 /* AIF3_BCLK_MSTR */ +#define WM5100_AIF3_BCLK_MSTR_MASK 0x0020 /* AIF3_BCLK_MSTR */ +#define WM5100_AIF3_BCLK_MSTR_SHIFT 5 /* AIF3_BCLK_MSTR */ +#define WM5100_AIF3_BCLK_MSTR_WIDTH 1 /* AIF3_BCLK_MSTR */ +#define WM5100_AIF3_BCLK_FREQ_MASK 0x001F /* AIF3_BCLK_FREQ - [4:0] */ +#define WM5100_AIF3_BCLK_FREQ_SHIFT 0 /* AIF3_BCLK_FREQ - [4:0] */ +#define WM5100_AIF3_BCLK_FREQ_WIDTH 5 /* AIF3_BCLK_FREQ - [4:0] */ + +/* + * R1409 (0x581) - Audio IF 3_2 + */ +#define WM5100_AIF3TX_DAT_TRI 0x0020 /* AIF3TX_DAT_TRI */ +#define WM5100_AIF3TX_DAT_TRI_MASK 0x0020 /* AIF3TX_DAT_TRI */ +#define WM5100_AIF3TX_DAT_TRI_SHIFT 5 /* AIF3TX_DAT_TRI */ +#define WM5100_AIF3TX_DAT_TRI_WIDTH 1 /* AIF3TX_DAT_TRI */ +#define WM5100_AIF3TX_LRCLK_SRC 0x0008 /* AIF3TX_LRCLK_SRC */ +#define WM5100_AIF3TX_LRCLK_SRC_MASK 0x0008 /* AIF3TX_LRCLK_SRC */ +#define WM5100_AIF3TX_LRCLK_SRC_SHIFT 3 /* AIF3TX_LRCLK_SRC */ +#define WM5100_AIF3TX_LRCLK_SRC_WIDTH 1 /* AIF3TX_LRCLK_SRC */ +#define WM5100_AIF3TX_LRCLK_INV 0x0004 /* AIF3TX_LRCLK_INV */ +#define WM5100_AIF3TX_LRCLK_INV_MASK 0x0004 /* AIF3TX_LRCLK_INV */ +#define WM5100_AIF3TX_LRCLK_INV_SHIFT 2 /* AIF3TX_LRCLK_INV */ +#define WM5100_AIF3TX_LRCLK_INV_WIDTH 1 /* AIF3TX_LRCLK_INV */ +#define WM5100_AIF3TX_LRCLK_FRC 0x0002 /* AIF3TX_LRCLK_FRC */ +#define WM5100_AIF3TX_LRCLK_FRC_MASK 0x0002 /* AIF3TX_LRCLK_FRC */ +#define WM5100_AIF3TX_LRCLK_FRC_SHIFT 1 /* AIF3TX_LRCLK_FRC */ +#define WM5100_AIF3TX_LRCLK_FRC_WIDTH 1 /* AIF3TX_LRCLK_FRC */ +#define WM5100_AIF3TX_LRCLK_MSTR 0x0001 /* AIF3TX_LRCLK_MSTR */ +#define WM5100_AIF3TX_LRCLK_MSTR_MASK 0x0001 /* AIF3TX_LRCLK_MSTR */ +#define WM5100_AIF3TX_LRCLK_MSTR_SHIFT 0 /* AIF3TX_LRCLK_MSTR */ +#define WM5100_AIF3TX_LRCLK_MSTR_WIDTH 1 /* AIF3TX_LRCLK_MSTR */ + +/* + * R1410 (0x582) - Audio IF 3_3 + */ +#define WM5100_AIF3RX_LRCLK_INV 0x0004 /* AIF3RX_LRCLK_INV */ +#define WM5100_AIF3RX_LRCLK_INV_MASK 0x0004 /* AIF3RX_LRCLK_INV */ +#define WM5100_AIF3RX_LRCLK_INV_SHIFT 2 /* AIF3RX_LRCLK_INV */ +#define WM5100_AIF3RX_LRCLK_INV_WIDTH 1 /* AIF3RX_LRCLK_INV */ +#define WM5100_AIF3RX_LRCLK_FRC 0x0002 /* AIF3RX_LRCLK_FRC */ +#define WM5100_AIF3RX_LRCLK_FRC_MASK 0x0002 /* AIF3RX_LRCLK_FRC */ +#define WM5100_AIF3RX_LRCLK_FRC_SHIFT 1 /* AIF3RX_LRCLK_FRC */ +#define WM5100_AIF3RX_LRCLK_FRC_WIDTH 1 /* AIF3RX_LRCLK_FRC */ +#define WM5100_AIF3RX_LRCLK_MSTR 0x0001 /* AIF3RX_LRCLK_MSTR */ +#define WM5100_AIF3RX_LRCLK_MSTR_MASK 0x0001 /* AIF3RX_LRCLK_MSTR */ +#define WM5100_AIF3RX_LRCLK_MSTR_SHIFT 0 /* AIF3RX_LRCLK_MSTR */ +#define WM5100_AIF3RX_LRCLK_MSTR_WIDTH 1 /* AIF3RX_LRCLK_MSTR */ + +/* + * R1411 (0x583) - Audio IF 3_4 + */ +#define WM5100_AIF3_TRI 0x0040 /* AIF3_TRI */ +#define WM5100_AIF3_TRI_MASK 0x0040 /* AIF3_TRI */ +#define WM5100_AIF3_TRI_SHIFT 6 /* AIF3_TRI */ +#define WM5100_AIF3_TRI_WIDTH 1 /* AIF3_TRI */ +#define WM5100_AIF3_RATE_MASK 0x0003 /* AIF3_RATE - [1:0] */ +#define WM5100_AIF3_RATE_SHIFT 0 /* AIF3_RATE - [1:0] */ +#define WM5100_AIF3_RATE_WIDTH 2 /* AIF3_RATE - [1:0] */ + +/* + * R1412 (0x584) - Audio IF 3_5 + */ +#define WM5100_AIF3_FMT_MASK 0x0007 /* AIF3_FMT - [2:0] */ +#define WM5100_AIF3_FMT_SHIFT 0 /* AIF3_FMT - [2:0] */ +#define WM5100_AIF3_FMT_WIDTH 3 /* AIF3_FMT - [2:0] */ + +/* + * R1413 (0x585) - Audio IF 3_6 + */ +#define WM5100_AIF3TX_BCPF_MASK 0x1FFF /* AIF3TX_BCPF - [12:0] */ +#define WM5100_AIF3TX_BCPF_SHIFT 0 /* AIF3TX_BCPF - [12:0] */ +#define WM5100_AIF3TX_BCPF_WIDTH 13 /* AIF3TX_BCPF - [12:0] */ + +/* + * R1414 (0x586) - Audio IF 3_7 + */ +#define WM5100_AIF3RX_BCPF_MASK 0x1FFF /* AIF3RX_BCPF - [12:0] */ +#define WM5100_AIF3RX_BCPF_SHIFT 0 /* AIF3RX_BCPF - [12:0] */ +#define WM5100_AIF3RX_BCPF_WIDTH 13 /* AIF3RX_BCPF - [12:0] */ + +/* + * R1415 (0x587) - Audio IF 3_8 + */ +#define WM5100_AIF3TX_WL_MASK 0x3F00 /* AIF3TX_WL - [13:8] */ +#define WM5100_AIF3TX_WL_SHIFT 8 /* AIF3TX_WL - [13:8] */ +#define WM5100_AIF3TX_WL_WIDTH 6 /* AIF3TX_WL - [13:8] */ +#define WM5100_AIF3TX_SLOT_LEN_MASK 0x00FF /* AIF3TX_SLOT_LEN - [7:0] */ +#define WM5100_AIF3TX_SLOT_LEN_SHIFT 0 /* AIF3TX_SLOT_LEN - [7:0] */ +#define WM5100_AIF3TX_SLOT_LEN_WIDTH 8 /* AIF3TX_SLOT_LEN - [7:0] */ + +/* + * R1416 (0x588) - Audio IF 3_9 + */ +#define WM5100_AIF3RX_WL_MASK 0x3F00 /* AIF3RX_WL - [13:8] */ +#define WM5100_AIF3RX_WL_SHIFT 8 /* AIF3RX_WL - [13:8] */ +#define WM5100_AIF3RX_WL_WIDTH 6 /* AIF3RX_WL - [13:8] */ +#define WM5100_AIF3RX_SLOT_LEN_MASK 0x00FF /* AIF3RX_SLOT_LEN - [7:0] */ +#define WM5100_AIF3RX_SLOT_LEN_SHIFT 0 /* AIF3RX_SLOT_LEN - [7:0] */ +#define WM5100_AIF3RX_SLOT_LEN_WIDTH 8 /* AIF3RX_SLOT_LEN - [7:0] */ + +/* + * R1417 (0x589) - Audio IF 3_10 + */ +#define WM5100_AIF3TX1_SLOT_MASK 0x003F /* AIF3TX1_SLOT - [5:0] */ +#define WM5100_AIF3TX1_SLOT_SHIFT 0 /* AIF3TX1_SLOT - [5:0] */ +#define WM5100_AIF3TX1_SLOT_WIDTH 6 /* AIF3TX1_SLOT - [5:0] */ + +/* + * R1418 (0x58A) - Audio IF 3_11 + */ +#define WM5100_AIF3TX2_SLOT_MASK 0x003F /* AIF3TX2_SLOT - [5:0] */ +#define WM5100_AIF3TX2_SLOT_SHIFT 0 /* AIF3TX2_SLOT - [5:0] */ +#define WM5100_AIF3TX2_SLOT_WIDTH 6 /* AIF3TX2_SLOT - [5:0] */ + +/* + * R1425 (0x591) - Audio IF 3_18 + */ +#define WM5100_AIF3RX1_SLOT_MASK 0x003F /* AIF3RX1_SLOT - [5:0] */ +#define WM5100_AIF3RX1_SLOT_SHIFT 0 /* AIF3RX1_SLOT - [5:0] */ +#define WM5100_AIF3RX1_SLOT_WIDTH 6 /* AIF3RX1_SLOT - [5:0] */ + +/* + * R1426 (0x592) - Audio IF 3_19 + */ +#define WM5100_AIF3RX2_SLOT_MASK 0x003F /* AIF3RX2_SLOT - [5:0] */ +#define WM5100_AIF3RX2_SLOT_SHIFT 0 /* AIF3RX2_SLOT - [5:0] */ +#define WM5100_AIF3RX2_SLOT_WIDTH 6 /* AIF3RX2_SLOT - [5:0] */ + +/* + * R1433 (0x599) - Audio IF 3_26 + */ +#define WM5100_AIF3TX2_ENA 0x0002 /* AIF3TX2_ENA */ +#define WM5100_AIF3TX2_ENA_MASK 0x0002 /* AIF3TX2_ENA */ +#define WM5100_AIF3TX2_ENA_SHIFT 1 /* AIF3TX2_ENA */ +#define WM5100_AIF3TX2_ENA_WIDTH 1 /* AIF3TX2_ENA */ +#define WM5100_AIF3TX1_ENA 0x0001 /* AIF3TX1_ENA */ +#define WM5100_AIF3TX1_ENA_MASK 0x0001 /* AIF3TX1_ENA */ +#define WM5100_AIF3TX1_ENA_SHIFT 0 /* AIF3TX1_ENA */ +#define WM5100_AIF3TX1_ENA_WIDTH 1 /* AIF3TX1_ENA */ + +/* + * R1434 (0x59A) - Audio IF 3_27 + */ +#define WM5100_AIF3RX2_ENA 0x0002 /* AIF3RX2_ENA */ +#define WM5100_AIF3RX2_ENA_MASK 0x0002 /* AIF3RX2_ENA */ +#define WM5100_AIF3RX2_ENA_SHIFT 1 /* AIF3RX2_ENA */ +#define WM5100_AIF3RX2_ENA_WIDTH 1 /* AIF3RX2_ENA */ +#define WM5100_AIF3RX1_ENA 0x0001 /* AIF3RX1_ENA */ +#define WM5100_AIF3RX1_ENA_MASK 0x0001 /* AIF3RX1_ENA */ +#define WM5100_AIF3RX1_ENA_SHIFT 0 /* AIF3RX1_ENA */ +#define WM5100_AIF3RX1_ENA_WIDTH 1 /* AIF3RX1_ENA */ + +#define WM5100_MIXER_VOL_MASK 0x00FE /* MIXER_VOL - [7:1] */ +#define WM5100_MIXER_VOL_SHIFT 1 /* MIXER_VOL - [7:1] */ +#define WM5100_MIXER_VOL_WIDTH 7 /* MIXER_VOL - [7:1] */ + +/* + * R3072 (0xC00) - GPIO CTRL 1 + */ +#define WM5100_GP1_DIR 0x8000 /* GP1_DIR */ +#define WM5100_GP1_DIR_MASK 0x8000 /* GP1_DIR */ +#define WM5100_GP1_DIR_SHIFT 15 /* GP1_DIR */ +#define WM5100_GP1_DIR_WIDTH 1 /* GP1_DIR */ +#define WM5100_GP1_PU 0x4000 /* GP1_PU */ +#define WM5100_GP1_PU_MASK 0x4000 /* GP1_PU */ +#define WM5100_GP1_PU_SHIFT 14 /* GP1_PU */ +#define WM5100_GP1_PU_WIDTH 1 /* GP1_PU */ +#define WM5100_GP1_PD 0x2000 /* GP1_PD */ +#define WM5100_GP1_PD_MASK 0x2000 /* GP1_PD */ +#define WM5100_GP1_PD_SHIFT 13 /* GP1_PD */ +#define WM5100_GP1_PD_WIDTH 1 /* GP1_PD */ +#define WM5100_GP1_POL 0x0400 /* GP1_POL */ +#define WM5100_GP1_POL_MASK 0x0400 /* GP1_POL */ +#define WM5100_GP1_POL_SHIFT 10 /* GP1_POL */ +#define WM5100_GP1_POL_WIDTH 1 /* GP1_POL */ +#define WM5100_GP1_OP_CFG 0x0200 /* GP1_OP_CFG */ +#define WM5100_GP1_OP_CFG_MASK 0x0200 /* GP1_OP_CFG */ +#define WM5100_GP1_OP_CFG_SHIFT 9 /* GP1_OP_CFG */ +#define WM5100_GP1_OP_CFG_WIDTH 1 /* GP1_OP_CFG */ +#define WM5100_GP1_DB 0x0100 /* GP1_DB */ +#define WM5100_GP1_DB_MASK 0x0100 /* GP1_DB */ +#define WM5100_GP1_DB_SHIFT 8 /* GP1_DB */ +#define WM5100_GP1_DB_WIDTH 1 /* GP1_DB */ +#define WM5100_GP1_LVL 0x0040 /* GP1_LVL */ +#define WM5100_GP1_LVL_MASK 0x0040 /* GP1_LVL */ +#define WM5100_GP1_LVL_SHIFT 6 /* GP1_LVL */ +#define WM5100_GP1_LVL_WIDTH 1 /* GP1_LVL */ +#define WM5100_GP1_FN_MASK 0x003F /* GP1_FN - [5:0] */ +#define WM5100_GP1_FN_SHIFT 0 /* GP1_FN - [5:0] */ +#define WM5100_GP1_FN_WIDTH 6 /* GP1_FN - [5:0] */ + +/* + * R3073 (0xC01) - GPIO CTRL 2 + */ +#define WM5100_GP2_DIR 0x8000 /* GP2_DIR */ +#define WM5100_GP2_DIR_MASK 0x8000 /* GP2_DIR */ +#define WM5100_GP2_DIR_SHIFT 15 /* GP2_DIR */ +#define WM5100_GP2_DIR_WIDTH 1 /* GP2_DIR */ +#define WM5100_GP2_PU 0x4000 /* GP2_PU */ +#define WM5100_GP2_PU_MASK 0x4000 /* GP2_PU */ +#define WM5100_GP2_PU_SHIFT 14 /* GP2_PU */ +#define WM5100_GP2_PU_WIDTH 1 /* GP2_PU */ +#define WM5100_GP2_PD 0x2000 /* GP2_PD */ +#define WM5100_GP2_PD_MASK 0x2000 /* GP2_PD */ +#define WM5100_GP2_PD_SHIFT 13 /* GP2_PD */ +#define WM5100_GP2_PD_WIDTH 1 /* GP2_PD */ +#define WM5100_GP2_POL 0x0400 /* GP2_POL */ +#define WM5100_GP2_POL_MASK 0x0400 /* GP2_POL */ +#define WM5100_GP2_POL_SHIFT 10 /* GP2_POL */ +#define WM5100_GP2_POL_WIDTH 1 /* GP2_POL */ +#define WM5100_GP2_OP_CFG 0x0200 /* GP2_OP_CFG */ +#define WM5100_GP2_OP_CFG_MASK 0x0200 /* GP2_OP_CFG */ +#define WM5100_GP2_OP_CFG_SHIFT 9 /* GP2_OP_CFG */ +#define WM5100_GP2_OP_CFG_WIDTH 1 /* GP2_OP_CFG */ +#define WM5100_GP2_DB 0x0100 /* GP2_DB */ +#define WM5100_GP2_DB_MASK 0x0100 /* GP2_DB */ +#define WM5100_GP2_DB_SHIFT 8 /* GP2_DB */ +#define WM5100_GP2_DB_WIDTH 1 /* GP2_DB */ +#define WM5100_GP2_LVL 0x0040 /* GP2_LVL */ +#define WM5100_GP2_LVL_MASK 0x0040 /* GP2_LVL */ +#define WM5100_GP2_LVL_SHIFT 6 /* GP2_LVL */ +#define WM5100_GP2_LVL_WIDTH 1 /* GP2_LVL */ +#define WM5100_GP2_FN_MASK 0x003F /* GP2_FN - [5:0] */ +#define WM5100_GP2_FN_SHIFT 0 /* GP2_FN - [5:0] */ +#define WM5100_GP2_FN_WIDTH 6 /* GP2_FN - [5:0] */ + +/* + * R3074 (0xC02) - GPIO CTRL 3 + */ +#define WM5100_GP3_DIR 0x8000 /* GP3_DIR */ +#define WM5100_GP3_DIR_MASK 0x8000 /* GP3_DIR */ +#define WM5100_GP3_DIR_SHIFT 15 /* GP3_DIR */ +#define WM5100_GP3_DIR_WIDTH 1 /* GP3_DIR */ +#define WM5100_GP3_PU 0x4000 /* GP3_PU */ +#define WM5100_GP3_PU_MASK 0x4000 /* GP3_PU */ +#define WM5100_GP3_PU_SHIFT 14 /* GP3_PU */ +#define WM5100_GP3_PU_WIDTH 1 /* GP3_PU */ +#define WM5100_GP3_PD 0x2000 /* GP3_PD */ +#define WM5100_GP3_PD_MASK 0x2000 /* GP3_PD */ +#define WM5100_GP3_PD_SHIFT 13 /* GP3_PD */ +#define WM5100_GP3_PD_WIDTH 1 /* GP3_PD */ +#define WM5100_GP3_POL 0x0400 /* GP3_POL */ +#define WM5100_GP3_POL_MASK 0x0400 /* GP3_POL */ +#define WM5100_GP3_POL_SHIFT 10 /* GP3_POL */ +#define WM5100_GP3_POL_WIDTH 1 /* GP3_POL */ +#define WM5100_GP3_OP_CFG 0x0200 /* GP3_OP_CFG */ +#define WM5100_GP3_OP_CFG_MASK 0x0200 /* GP3_OP_CFG */ +#define WM5100_GP3_OP_CFG_SHIFT 9 /* GP3_OP_CFG */ +#define WM5100_GP3_OP_CFG_WIDTH 1 /* GP3_OP_CFG */ +#define WM5100_GP3_DB 0x0100 /* GP3_DB */ +#define WM5100_GP3_DB_MASK 0x0100 /* GP3_DB */ +#define WM5100_GP3_DB_SHIFT 8 /* GP3_DB */ +#define WM5100_GP3_DB_WIDTH 1 /* GP3_DB */ +#define WM5100_GP3_LVL 0x0040 /* GP3_LVL */ +#define WM5100_GP3_LVL_MASK 0x0040 /* GP3_LVL */ +#define WM5100_GP3_LVL_SHIFT 6 /* GP3_LVL */ +#define WM5100_GP3_LVL_WIDTH 1 /* GP3_LVL */ +#define WM5100_GP3_FN_MASK 0x003F /* GP3_FN - [5:0] */ +#define WM5100_GP3_FN_SHIFT 0 /* GP3_FN - [5:0] */ +#define WM5100_GP3_FN_WIDTH 6 /* GP3_FN - [5:0] */ + +/* + * R3075 (0xC03) - GPIO CTRL 4 + */ +#define WM5100_GP4_DIR 0x8000 /* GP4_DIR */ +#define WM5100_GP4_DIR_MASK 0x8000 /* GP4_DIR */ +#define WM5100_GP4_DIR_SHIFT 15 /* GP4_DIR */ +#define WM5100_GP4_DIR_WIDTH 1 /* GP4_DIR */ +#define WM5100_GP4_PU 0x4000 /* GP4_PU */ +#define WM5100_GP4_PU_MASK 0x4000 /* GP4_PU */ +#define WM5100_GP4_PU_SHIFT 14 /* GP4_PU */ +#define WM5100_GP4_PU_WIDTH 1 /* GP4_PU */ +#define WM5100_GP4_PD 0x2000 /* GP4_PD */ +#define WM5100_GP4_PD_MASK 0x2000 /* GP4_PD */ +#define WM5100_GP4_PD_SHIFT 13 /* GP4_PD */ +#define WM5100_GP4_PD_WIDTH 1 /* GP4_PD */ +#define WM5100_GP4_POL 0x0400 /* GP4_POL */ +#define WM5100_GP4_POL_MASK 0x0400 /* GP4_POL */ +#define WM5100_GP4_POL_SHIFT 10 /* GP4_POL */ +#define WM5100_GP4_POL_WIDTH 1 /* GP4_POL */ +#define WM5100_GP4_OP_CFG 0x0200 /* GP4_OP_CFG */ +#define WM5100_GP4_OP_CFG_MASK 0x0200 /* GP4_OP_CFG */ +#define WM5100_GP4_OP_CFG_SHIFT 9 /* GP4_OP_CFG */ +#define WM5100_GP4_OP_CFG_WIDTH 1 /* GP4_OP_CFG */ +#define WM5100_GP4_DB 0x0100 /* GP4_DB */ +#define WM5100_GP4_DB_MASK 0x0100 /* GP4_DB */ +#define WM5100_GP4_DB_SHIFT 8 /* GP4_DB */ +#define WM5100_GP4_DB_WIDTH 1 /* GP4_DB */ +#define WM5100_GP4_LVL 0x0040 /* GP4_LVL */ +#define WM5100_GP4_LVL_MASK 0x0040 /* GP4_LVL */ +#define WM5100_GP4_LVL_SHIFT 6 /* GP4_LVL */ +#define WM5100_GP4_LVL_WIDTH 1 /* GP4_LVL */ +#define WM5100_GP4_FN_MASK 0x003F /* GP4_FN - [5:0] */ +#define WM5100_GP4_FN_SHIFT 0 /* GP4_FN - [5:0] */ +#define WM5100_GP4_FN_WIDTH 6 /* GP4_FN - [5:0] */ + +/* + * R3076 (0xC04) - GPIO CTRL 5 + */ +#define WM5100_GP5_DIR 0x8000 /* GP5_DIR */ +#define WM5100_GP5_DIR_MASK 0x8000 /* GP5_DIR */ +#define WM5100_GP5_DIR_SHIFT 15 /* GP5_DIR */ +#define WM5100_GP5_DIR_WIDTH 1 /* GP5_DIR */ +#define WM5100_GP5_PU 0x4000 /* GP5_PU */ +#define WM5100_GP5_PU_MASK 0x4000 /* GP5_PU */ +#define WM5100_GP5_PU_SHIFT 14 /* GP5_PU */ +#define WM5100_GP5_PU_WIDTH 1 /* GP5_PU */ +#define WM5100_GP5_PD 0x2000 /* GP5_PD */ +#define WM5100_GP5_PD_MASK 0x2000 /* GP5_PD */ +#define WM5100_GP5_PD_SHIFT 13 /* GP5_PD */ +#define WM5100_GP5_PD_WIDTH 1 /* GP5_PD */ +#define WM5100_GP5_POL 0x0400 /* GP5_POL */ +#define WM5100_GP5_POL_MASK 0x0400 /* GP5_POL */ +#define WM5100_GP5_POL_SHIFT 10 /* GP5_POL */ +#define WM5100_GP5_POL_WIDTH 1 /* GP5_POL */ +#define WM5100_GP5_OP_CFG 0x0200 /* GP5_OP_CFG */ +#define WM5100_GP5_OP_CFG_MASK 0x0200 /* GP5_OP_CFG */ +#define WM5100_GP5_OP_CFG_SHIFT 9 /* GP5_OP_CFG */ +#define WM5100_GP5_OP_CFG_WIDTH 1 /* GP5_OP_CFG */ +#define WM5100_GP5_DB 0x0100 /* GP5_DB */ +#define WM5100_GP5_DB_MASK 0x0100 /* GP5_DB */ +#define WM5100_GP5_DB_SHIFT 8 /* GP5_DB */ +#define WM5100_GP5_DB_WIDTH 1 /* GP5_DB */ +#define WM5100_GP5_LVL 0x0040 /* GP5_LVL */ +#define WM5100_GP5_LVL_MASK 0x0040 /* GP5_LVL */ +#define WM5100_GP5_LVL_SHIFT 6 /* GP5_LVL */ +#define WM5100_GP5_LVL_WIDTH 1 /* GP5_LVL */ +#define WM5100_GP5_FN_MASK 0x003F /* GP5_FN - [5:0] */ +#define WM5100_GP5_FN_SHIFT 0 /* GP5_FN - [5:0] */ +#define WM5100_GP5_FN_WIDTH 6 /* GP5_FN - [5:0] */ + +/* + * R3077 (0xC05) - GPIO CTRL 6 + */ +#define WM5100_GP6_DIR 0x8000 /* GP6_DIR */ +#define WM5100_GP6_DIR_MASK 0x8000 /* GP6_DIR */ +#define WM5100_GP6_DIR_SHIFT 15 /* GP6_DIR */ +#define WM5100_GP6_DIR_WIDTH 1 /* GP6_DIR */ +#define WM5100_GP6_PU 0x4000 /* GP6_PU */ +#define WM5100_GP6_PU_MASK 0x4000 /* GP6_PU */ +#define WM5100_GP6_PU_SHIFT 14 /* GP6_PU */ +#define WM5100_GP6_PU_WIDTH 1 /* GP6_PU */ +#define WM5100_GP6_PD 0x2000 /* GP6_PD */ +#define WM5100_GP6_PD_MASK 0x2000 /* GP6_PD */ +#define WM5100_GP6_PD_SHIFT 13 /* GP6_PD */ +#define WM5100_GP6_PD_WIDTH 1 /* GP6_PD */ +#define WM5100_GP6_POL 0x0400 /* GP6_POL */ +#define WM5100_GP6_POL_MASK 0x0400 /* GP6_POL */ +#define WM5100_GP6_POL_SHIFT 10 /* GP6_POL */ +#define WM5100_GP6_POL_WIDTH 1 /* GP6_POL */ +#define WM5100_GP6_OP_CFG 0x0200 /* GP6_OP_CFG */ +#define WM5100_GP6_OP_CFG_MASK 0x0200 /* GP6_OP_CFG */ +#define WM5100_GP6_OP_CFG_SHIFT 9 /* GP6_OP_CFG */ +#define WM5100_GP6_OP_CFG_WIDTH 1 /* GP6_OP_CFG */ +#define WM5100_GP6_DB 0x0100 /* GP6_DB */ +#define WM5100_GP6_DB_MASK 0x0100 /* GP6_DB */ +#define WM5100_GP6_DB_SHIFT 8 /* GP6_DB */ +#define WM5100_GP6_DB_WIDTH 1 /* GP6_DB */ +#define WM5100_GP6_LVL 0x0040 /* GP6_LVL */ +#define WM5100_GP6_LVL_MASK 0x0040 /* GP6_LVL */ +#define WM5100_GP6_LVL_SHIFT 6 /* GP6_LVL */ +#define WM5100_GP6_LVL_WIDTH 1 /* GP6_LVL */ +#define WM5100_GP6_FN_MASK 0x003F /* GP6_FN - [5:0] */ +#define WM5100_GP6_FN_SHIFT 0 /* GP6_FN - [5:0] */ +#define WM5100_GP6_FN_WIDTH 6 /* GP6_FN - [5:0] */ + +/* + * R3107 (0xC23) - Misc Pad Ctrl 1 + */ +#define WM5100_LDO1ENA_PD 0x8000 /* LDO1ENA_PD */ +#define WM5100_LDO1ENA_PD_MASK 0x8000 /* LDO1ENA_PD */ +#define WM5100_LDO1ENA_PD_SHIFT 15 /* LDO1ENA_PD */ +#define WM5100_LDO1ENA_PD_WIDTH 1 /* LDO1ENA_PD */ +#define WM5100_MCLK2_PD 0x2000 /* MCLK2_PD */ +#define WM5100_MCLK2_PD_MASK 0x2000 /* MCLK2_PD */ +#define WM5100_MCLK2_PD_SHIFT 13 /* MCLK2_PD */ +#define WM5100_MCLK2_PD_WIDTH 1 /* MCLK2_PD */ +#define WM5100_MCLK1_PD 0x1000 /* MCLK1_PD */ +#define WM5100_MCLK1_PD_MASK 0x1000 /* MCLK1_PD */ +#define WM5100_MCLK1_PD_SHIFT 12 /* MCLK1_PD */ +#define WM5100_MCLK1_PD_WIDTH 1 /* MCLK1_PD */ +#define WM5100_RESET_PU 0x0002 /* RESET_PU */ +#define WM5100_RESET_PU_MASK 0x0002 /* RESET_PU */ +#define WM5100_RESET_PU_SHIFT 1 /* RESET_PU */ +#define WM5100_RESET_PU_WIDTH 1 /* RESET_PU */ +#define WM5100_ADDR_PD 0x0001 /* ADDR_PD */ +#define WM5100_ADDR_PD_MASK 0x0001 /* ADDR_PD */ +#define WM5100_ADDR_PD_SHIFT 0 /* ADDR_PD */ +#define WM5100_ADDR_PD_WIDTH 1 /* ADDR_PD */ + +/* + * R3108 (0xC24) - Misc Pad Ctrl 2 + */ +#define WM5100_DMICDAT4_PD 0x0008 /* DMICDAT4_PD */ +#define WM5100_DMICDAT4_PD_MASK 0x0008 /* DMICDAT4_PD */ +#define WM5100_DMICDAT4_PD_SHIFT 3 /* DMICDAT4_PD */ +#define WM5100_DMICDAT4_PD_WIDTH 1 /* DMICDAT4_PD */ +#define WM5100_DMICDAT3_PD 0x0004 /* DMICDAT3_PD */ +#define WM5100_DMICDAT3_PD_MASK 0x0004 /* DMICDAT3_PD */ +#define WM5100_DMICDAT3_PD_SHIFT 2 /* DMICDAT3_PD */ +#define WM5100_DMICDAT3_PD_WIDTH 1 /* DMICDAT3_PD */ +#define WM5100_DMICDAT2_PD 0x0002 /* DMICDAT2_PD */ +#define WM5100_DMICDAT2_PD_MASK 0x0002 /* DMICDAT2_PD */ +#define WM5100_DMICDAT2_PD_SHIFT 1 /* DMICDAT2_PD */ +#define WM5100_DMICDAT2_PD_WIDTH 1 /* DMICDAT2_PD */ +#define WM5100_DMICDAT1_PD 0x0001 /* DMICDAT1_PD */ +#define WM5100_DMICDAT1_PD_MASK 0x0001 /* DMICDAT1_PD */ +#define WM5100_DMICDAT1_PD_SHIFT 0 /* DMICDAT1_PD */ +#define WM5100_DMICDAT1_PD_WIDTH 1 /* DMICDAT1_PD */ + +/* + * R3109 (0xC25) - Misc Pad Ctrl 3 + */ +#define WM5100_AIF1RXLRCLK_PU 0x0020 /* AIF1RXLRCLK_PU */ +#define WM5100_AIF1RXLRCLK_PU_MASK 0x0020 /* AIF1RXLRCLK_PU */ +#define WM5100_AIF1RXLRCLK_PU_SHIFT 5 /* AIF1RXLRCLK_PU */ +#define WM5100_AIF1RXLRCLK_PU_WIDTH 1 /* AIF1RXLRCLK_PU */ +#define WM5100_AIF1RXLRCLK_PD 0x0010 /* AIF1RXLRCLK_PD */ +#define WM5100_AIF1RXLRCLK_PD_MASK 0x0010 /* AIF1RXLRCLK_PD */ +#define WM5100_AIF1RXLRCLK_PD_SHIFT 4 /* AIF1RXLRCLK_PD */ +#define WM5100_AIF1RXLRCLK_PD_WIDTH 1 /* AIF1RXLRCLK_PD */ +#define WM5100_AIF1BCLK_PU 0x0008 /* AIF1BCLK_PU */ +#define WM5100_AIF1BCLK_PU_MASK 0x0008 /* AIF1BCLK_PU */ +#define WM5100_AIF1BCLK_PU_SHIFT 3 /* AIF1BCLK_PU */ +#define WM5100_AIF1BCLK_PU_WIDTH 1 /* AIF1BCLK_PU */ +#define WM5100_AIF1BCLK_PD 0x0004 /* AIF1BCLK_PD */ +#define WM5100_AIF1BCLK_PD_MASK 0x0004 /* AIF1BCLK_PD */ +#define WM5100_AIF1BCLK_PD_SHIFT 2 /* AIF1BCLK_PD */ +#define WM5100_AIF1BCLK_PD_WIDTH 1 /* AIF1BCLK_PD */ +#define WM5100_AIF1RXDAT_PU 0x0002 /* AIF1RXDAT_PU */ +#define WM5100_AIF1RXDAT_PU_MASK 0x0002 /* AIF1RXDAT_PU */ +#define WM5100_AIF1RXDAT_PU_SHIFT 1 /* AIF1RXDAT_PU */ +#define WM5100_AIF1RXDAT_PU_WIDTH 1 /* AIF1RXDAT_PU */ +#define WM5100_AIF1RXDAT_PD 0x0001 /* AIF1RXDAT_PD */ +#define WM5100_AIF1RXDAT_PD_MASK 0x0001 /* AIF1RXDAT_PD */ +#define WM5100_AIF1RXDAT_PD_SHIFT 0 /* AIF1RXDAT_PD */ +#define WM5100_AIF1RXDAT_PD_WIDTH 1 /* AIF1RXDAT_PD */ + +/* + * R3110 (0xC26) - Misc Pad Ctrl 4 + */ +#define WM5100_AIF2RXLRCLK_PU 0x0020 /* AIF2RXLRCLK_PU */ +#define WM5100_AIF2RXLRCLK_PU_MASK 0x0020 /* AIF2RXLRCLK_PU */ +#define WM5100_AIF2RXLRCLK_PU_SHIFT 5 /* AIF2RXLRCLK_PU */ +#define WM5100_AIF2RXLRCLK_PU_WIDTH 1 /* AIF2RXLRCLK_PU */ +#define WM5100_AIF2RXLRCLK_PD 0x0010 /* AIF2RXLRCLK_PD */ +#define WM5100_AIF2RXLRCLK_PD_MASK 0x0010 /* AIF2RXLRCLK_PD */ +#define WM5100_AIF2RXLRCLK_PD_SHIFT 4 /* AIF2RXLRCLK_PD */ +#define WM5100_AIF2RXLRCLK_PD_WIDTH 1 /* AIF2RXLRCLK_PD */ +#define WM5100_AIF2BCLK_PU 0x0008 /* AIF2BCLK_PU */ +#define WM5100_AIF2BCLK_PU_MASK 0x0008 /* AIF2BCLK_PU */ +#define WM5100_AIF2BCLK_PU_SHIFT 3 /* AIF2BCLK_PU */ +#define WM5100_AIF2BCLK_PU_WIDTH 1 /* AIF2BCLK_PU */ +#define WM5100_AIF2BCLK_PD 0x0004 /* AIF2BCLK_PD */ +#define WM5100_AIF2BCLK_PD_MASK 0x0004 /* AIF2BCLK_PD */ +#define WM5100_AIF2BCLK_PD_SHIFT 2 /* AIF2BCLK_PD */ +#define WM5100_AIF2BCLK_PD_WIDTH 1 /* AIF2BCLK_PD */ +#define WM5100_AIF2RXDAT_PU 0x0002 /* AIF2RXDAT_PU */ +#define WM5100_AIF2RXDAT_PU_MASK 0x0002 /* AIF2RXDAT_PU */ +#define WM5100_AIF2RXDAT_PU_SHIFT 1 /* AIF2RXDAT_PU */ +#define WM5100_AIF2RXDAT_PU_WIDTH 1 /* AIF2RXDAT_PU */ +#define WM5100_AIF2RXDAT_PD 0x0001 /* AIF2RXDAT_PD */ +#define WM5100_AIF2RXDAT_PD_MASK 0x0001 /* AIF2RXDAT_PD */ +#define WM5100_AIF2RXDAT_PD_SHIFT 0 /* AIF2RXDAT_PD */ +#define WM5100_AIF2RXDAT_PD_WIDTH 1 /* AIF2RXDAT_PD */ + +/* + * R3111 (0xC27) - Misc Pad Ctrl 5 + */ +#define WM5100_AIF3RXLRCLK_PU 0x0020 /* AIF3RXLRCLK_PU */ +#define WM5100_AIF3RXLRCLK_PU_MASK 0x0020 /* AIF3RXLRCLK_PU */ +#define WM5100_AIF3RXLRCLK_PU_SHIFT 5 /* AIF3RXLRCLK_PU */ +#define WM5100_AIF3RXLRCLK_PU_WIDTH 1 /* AIF3RXLRCLK_PU */ +#define WM5100_AIF3RXLRCLK_PD 0x0010 /* AIF3RXLRCLK_PD */ +#define WM5100_AIF3RXLRCLK_PD_MASK 0x0010 /* AIF3RXLRCLK_PD */ +#define WM5100_AIF3RXLRCLK_PD_SHIFT 4 /* AIF3RXLRCLK_PD */ +#define WM5100_AIF3RXLRCLK_PD_WIDTH 1 /* AIF3RXLRCLK_PD */ +#define WM5100_AIF3BCLK_PU 0x0008 /* AIF3BCLK_PU */ +#define WM5100_AIF3BCLK_PU_MASK 0x0008 /* AIF3BCLK_PU */ +#define WM5100_AIF3BCLK_PU_SHIFT 3 /* AIF3BCLK_PU */ +#define WM5100_AIF3BCLK_PU_WIDTH 1 /* AIF3BCLK_PU */ +#define WM5100_AIF3BCLK_PD 0x0004 /* AIF3BCLK_PD */ +#define WM5100_AIF3BCLK_PD_MASK 0x0004 /* AIF3BCLK_PD */ +#define WM5100_AIF3BCLK_PD_SHIFT 2 /* AIF3BCLK_PD */ +#define WM5100_AIF3BCLK_PD_WIDTH 1 /* AIF3BCLK_PD */ +#define WM5100_AIF3RXDAT_PU 0x0002 /* AIF3RXDAT_PU */ +#define WM5100_AIF3RXDAT_PU_MASK 0x0002 /* AIF3RXDAT_PU */ +#define WM5100_AIF3RXDAT_PU_SHIFT 1 /* AIF3RXDAT_PU */ +#define WM5100_AIF3RXDAT_PU_WIDTH 1 /* AIF3RXDAT_PU */ +#define WM5100_AIF3RXDAT_PD 0x0001 /* AIF3RXDAT_PD */ +#define WM5100_AIF3RXDAT_PD_MASK 0x0001 /* AIF3RXDAT_PD */ +#define WM5100_AIF3RXDAT_PD_SHIFT 0 /* AIF3RXDAT_PD */ +#define WM5100_AIF3RXDAT_PD_WIDTH 1 /* AIF3RXDAT_PD */ + +/* + * R3112 (0xC28) - Misc GPIO 1 + */ +#define WM5100_OPCLK_SEL_MASK 0x0003 /* OPCLK_SEL - [1:0] */ +#define WM5100_OPCLK_SEL_SHIFT 0 /* OPCLK_SEL - [1:0] */ +#define WM5100_OPCLK_SEL_WIDTH 2 /* OPCLK_SEL - [1:0] */ + +/* + * R3328 (0xD00) - Interrupt Status 1 + */ +#define WM5100_GP6_EINT 0x0020 /* GP6_EINT */ +#define WM5100_GP6_EINT_MASK 0x0020 /* GP6_EINT */ +#define WM5100_GP6_EINT_SHIFT 5 /* GP6_EINT */ +#define WM5100_GP6_EINT_WIDTH 1 /* GP6_EINT */ +#define WM5100_GP5_EINT 0x0010 /* GP5_EINT */ +#define WM5100_GP5_EINT_MASK 0x0010 /* GP5_EINT */ +#define WM5100_GP5_EINT_SHIFT 4 /* GP5_EINT */ +#define WM5100_GP5_EINT_WIDTH 1 /* GP5_EINT */ +#define WM5100_GP4_EINT 0x0008 /* GP4_EINT */ +#define WM5100_GP4_EINT_MASK 0x0008 /* GP4_EINT */ +#define WM5100_GP4_EINT_SHIFT 3 /* GP4_EINT */ +#define WM5100_GP4_EINT_WIDTH 1 /* GP4_EINT */ +#define WM5100_GP3_EINT 0x0004 /* GP3_EINT */ +#define WM5100_GP3_EINT_MASK 0x0004 /* GP3_EINT */ +#define WM5100_GP3_EINT_SHIFT 2 /* GP3_EINT */ +#define WM5100_GP3_EINT_WIDTH 1 /* GP3_EINT */ +#define WM5100_GP2_EINT 0x0002 /* GP2_EINT */ +#define WM5100_GP2_EINT_MASK 0x0002 /* GP2_EINT */ +#define WM5100_GP2_EINT_SHIFT 1 /* GP2_EINT */ +#define WM5100_GP2_EINT_WIDTH 1 /* GP2_EINT */ +#define WM5100_GP1_EINT 0x0001 /* GP1_EINT */ +#define WM5100_GP1_EINT_MASK 0x0001 /* GP1_EINT */ +#define WM5100_GP1_EINT_SHIFT 0 /* GP1_EINT */ +#define WM5100_GP1_EINT_WIDTH 1 /* GP1_EINT */ + +/* + * R3329 (0xD01) - Interrupt Status 2 + */ +#define WM5100_DSP_IRQ6_EINT 0x0020 /* DSP_IRQ6_EINT */ +#define WM5100_DSP_IRQ6_EINT_MASK 0x0020 /* DSP_IRQ6_EINT */ +#define WM5100_DSP_IRQ6_EINT_SHIFT 5 /* DSP_IRQ6_EINT */ +#define WM5100_DSP_IRQ6_EINT_WIDTH 1 /* DSP_IRQ6_EINT */ +#define WM5100_DSP_IRQ5_EINT 0x0010 /* DSP_IRQ5_EINT */ +#define WM5100_DSP_IRQ5_EINT_MASK 0x0010 /* DSP_IRQ5_EINT */ +#define WM5100_DSP_IRQ5_EINT_SHIFT 4 /* DSP_IRQ5_EINT */ +#define WM5100_DSP_IRQ5_EINT_WIDTH 1 /* DSP_IRQ5_EINT */ +#define WM5100_DSP_IRQ4_EINT 0x0008 /* DSP_IRQ4_EINT */ +#define WM5100_DSP_IRQ4_EINT_MASK 0x0008 /* DSP_IRQ4_EINT */ +#define WM5100_DSP_IRQ4_EINT_SHIFT 3 /* DSP_IRQ4_EINT */ +#define WM5100_DSP_IRQ4_EINT_WIDTH 1 /* DSP_IRQ4_EINT */ +#define WM5100_DSP_IRQ3_EINT 0x0004 /* DSP_IRQ3_EINT */ +#define WM5100_DSP_IRQ3_EINT_MASK 0x0004 /* DSP_IRQ3_EINT */ +#define WM5100_DSP_IRQ3_EINT_SHIFT 2 /* DSP_IRQ3_EINT */ +#define WM5100_DSP_IRQ3_EINT_WIDTH 1 /* DSP_IRQ3_EINT */ +#define WM5100_DSP_IRQ2_EINT 0x0002 /* DSP_IRQ2_EINT */ +#define WM5100_DSP_IRQ2_EINT_MASK 0x0002 /* DSP_IRQ2_EINT */ +#define WM5100_DSP_IRQ2_EINT_SHIFT 1 /* DSP_IRQ2_EINT */ +#define WM5100_DSP_IRQ2_EINT_WIDTH 1 /* DSP_IRQ2_EINT */ +#define WM5100_DSP_IRQ1_EINT 0x0001 /* DSP_IRQ1_EINT */ +#define WM5100_DSP_IRQ1_EINT_MASK 0x0001 /* DSP_IRQ1_EINT */ +#define WM5100_DSP_IRQ1_EINT_SHIFT 0 /* DSP_IRQ1_EINT */ +#define WM5100_DSP_IRQ1_EINT_WIDTH 1 /* DSP_IRQ1_EINT */ + +/* + * R3330 (0xD02) - Interrupt Status 3 + */ +#define WM5100_SPK_SHUTDOWN_WARN_EINT 0x8000 /* SPK_SHUTDOWN_WARN_EINT */ +#define WM5100_SPK_SHUTDOWN_WARN_EINT_MASK 0x8000 /* SPK_SHUTDOWN_WARN_EINT */ +#define WM5100_SPK_SHUTDOWN_WARN_EINT_SHIFT 15 /* SPK_SHUTDOWN_WARN_EINT */ +#define WM5100_SPK_SHUTDOWN_WARN_EINT_WIDTH 1 /* SPK_SHUTDOWN_WARN_EINT */ +#define WM5100_SPK_SHUTDOWN_EINT 0x4000 /* SPK_SHUTDOWN_EINT */ +#define WM5100_SPK_SHUTDOWN_EINT_MASK 0x4000 /* SPK_SHUTDOWN_EINT */ +#define WM5100_SPK_SHUTDOWN_EINT_SHIFT 14 /* SPK_SHUTDOWN_EINT */ +#define WM5100_SPK_SHUTDOWN_EINT_WIDTH 1 /* SPK_SHUTDOWN_EINT */ +#define WM5100_HPDET_EINT 0x2000 /* HPDET_EINT */ +#define WM5100_HPDET_EINT_MASK 0x2000 /* HPDET_EINT */ +#define WM5100_HPDET_EINT_SHIFT 13 /* HPDET_EINT */ +#define WM5100_HPDET_EINT_WIDTH 1 /* HPDET_EINT */ +#define WM5100_ACCDET_EINT 0x1000 /* ACCDET_EINT */ +#define WM5100_ACCDET_EINT_MASK 0x1000 /* ACCDET_EINT */ +#define WM5100_ACCDET_EINT_SHIFT 12 /* ACCDET_EINT */ +#define WM5100_ACCDET_EINT_WIDTH 1 /* ACCDET_EINT */ +#define WM5100_DRC_SIG_DET_EINT 0x0200 /* DRC_SIG_DET_EINT */ +#define WM5100_DRC_SIG_DET_EINT_MASK 0x0200 /* DRC_SIG_DET_EINT */ +#define WM5100_DRC_SIG_DET_EINT_SHIFT 9 /* DRC_SIG_DET_EINT */ +#define WM5100_DRC_SIG_DET_EINT_WIDTH 1 /* DRC_SIG_DET_EINT */ +#define WM5100_ASRC2_LOCK_EINT 0x0100 /* ASRC2_LOCK_EINT */ +#define WM5100_ASRC2_LOCK_EINT_MASK 0x0100 /* ASRC2_LOCK_EINT */ +#define WM5100_ASRC2_LOCK_EINT_SHIFT 8 /* ASRC2_LOCK_EINT */ +#define WM5100_ASRC2_LOCK_EINT_WIDTH 1 /* ASRC2_LOCK_EINT */ +#define WM5100_ASRC1_LOCK_EINT 0x0080 /* ASRC1_LOCK_EINT */ +#define WM5100_ASRC1_LOCK_EINT_MASK 0x0080 /* ASRC1_LOCK_EINT */ +#define WM5100_ASRC1_LOCK_EINT_SHIFT 7 /* ASRC1_LOCK_EINT */ +#define WM5100_ASRC1_LOCK_EINT_WIDTH 1 /* ASRC1_LOCK_EINT */ +#define WM5100_FLL2_LOCK_EINT 0x0008 /* FLL2_LOCK_EINT */ +#define WM5100_FLL2_LOCK_EINT_MASK 0x0008 /* FLL2_LOCK_EINT */ +#define WM5100_FLL2_LOCK_EINT_SHIFT 3 /* FLL2_LOCK_EINT */ +#define WM5100_FLL2_LOCK_EINT_WIDTH 1 /* FLL2_LOCK_EINT */ +#define WM5100_FLL1_LOCK_EINT 0x0004 /* FLL1_LOCK_EINT */ +#define WM5100_FLL1_LOCK_EINT_MASK 0x0004 /* FLL1_LOCK_EINT */ +#define WM5100_FLL1_LOCK_EINT_SHIFT 2 /* FLL1_LOCK_EINT */ +#define WM5100_FLL1_LOCK_EINT_WIDTH 1 /* FLL1_LOCK_EINT */ +#define WM5100_CLKGEN_ERR_EINT 0x0002 /* CLKGEN_ERR_EINT */ +#define WM5100_CLKGEN_ERR_EINT_MASK 0x0002 /* CLKGEN_ERR_EINT */ +#define WM5100_CLKGEN_ERR_EINT_SHIFT 1 /* CLKGEN_ERR_EINT */ +#define WM5100_CLKGEN_ERR_EINT_WIDTH 1 /* CLKGEN_ERR_EINT */ +#define WM5100_CLKGEN_ERR_ASYNC_EINT 0x0001 /* CLKGEN_ERR_ASYNC_EINT */ +#define WM5100_CLKGEN_ERR_ASYNC_EINT_MASK 0x0001 /* CLKGEN_ERR_ASYNC_EINT */ +#define WM5100_CLKGEN_ERR_ASYNC_EINT_SHIFT 0 /* CLKGEN_ERR_ASYNC_EINT */ +#define WM5100_CLKGEN_ERR_ASYNC_EINT_WIDTH 1 /* CLKGEN_ERR_ASYNC_EINT */ + +/* + * R3331 (0xD03) - Interrupt Status 4 + */ +#define WM5100_AIF3_ERR_EINT 0x2000 /* AIF3_ERR_EINT */ +#define WM5100_AIF3_ERR_EINT_MASK 0x2000 /* AIF3_ERR_EINT */ +#define WM5100_AIF3_ERR_EINT_SHIFT 13 /* AIF3_ERR_EINT */ +#define WM5100_AIF3_ERR_EINT_WIDTH 1 /* AIF3_ERR_EINT */ +#define WM5100_AIF2_ERR_EINT 0x1000 /* AIF2_ERR_EINT */ +#define WM5100_AIF2_ERR_EINT_MASK 0x1000 /* AIF2_ERR_EINT */ +#define WM5100_AIF2_ERR_EINT_SHIFT 12 /* AIF2_ERR_EINT */ +#define WM5100_AIF2_ERR_EINT_WIDTH 1 /* AIF2_ERR_EINT */ +#define WM5100_AIF1_ERR_EINT 0x0800 /* AIF1_ERR_EINT */ +#define WM5100_AIF1_ERR_EINT_MASK 0x0800 /* AIF1_ERR_EINT */ +#define WM5100_AIF1_ERR_EINT_SHIFT 11 /* AIF1_ERR_EINT */ +#define WM5100_AIF1_ERR_EINT_WIDTH 1 /* AIF1_ERR_EINT */ +#define WM5100_CTRLIF_ERR_EINT 0x0400 /* CTRLIF_ERR_EINT */ +#define WM5100_CTRLIF_ERR_EINT_MASK 0x0400 /* CTRLIF_ERR_EINT */ +#define WM5100_CTRLIF_ERR_EINT_SHIFT 10 /* CTRLIF_ERR_EINT */ +#define WM5100_CTRLIF_ERR_EINT_WIDTH 1 /* CTRLIF_ERR_EINT */ +#define WM5100_ISRC2_UNDERCLOCKED_EINT 0x0200 /* ISRC2_UNDERCLOCKED_EINT */ +#define WM5100_ISRC2_UNDERCLOCKED_EINT_MASK 0x0200 /* ISRC2_UNDERCLOCKED_EINT */ +#define WM5100_ISRC2_UNDERCLOCKED_EINT_SHIFT 9 /* ISRC2_UNDERCLOCKED_EINT */ +#define WM5100_ISRC2_UNDERCLOCKED_EINT_WIDTH 1 /* ISRC2_UNDERCLOCKED_EINT */ +#define WM5100_ISRC1_UNDERCLOCKED_EINT 0x0100 /* ISRC1_UNDERCLOCKED_EINT */ +#define WM5100_ISRC1_UNDERCLOCKED_EINT_MASK 0x0100 /* ISRC1_UNDERCLOCKED_EINT */ +#define WM5100_ISRC1_UNDERCLOCKED_EINT_SHIFT 8 /* ISRC1_UNDERCLOCKED_EINT */ +#define WM5100_ISRC1_UNDERCLOCKED_EINT_WIDTH 1 /* ISRC1_UNDERCLOCKED_EINT */ +#define WM5100_FX_UNDERCLOCKED_EINT 0x0080 /* FX_UNDERCLOCKED_EINT */ +#define WM5100_FX_UNDERCLOCKED_EINT_MASK 0x0080 /* FX_UNDERCLOCKED_EINT */ +#define WM5100_FX_UNDERCLOCKED_EINT_SHIFT 7 /* FX_UNDERCLOCKED_EINT */ +#define WM5100_FX_UNDERCLOCKED_EINT_WIDTH 1 /* FX_UNDERCLOCKED_EINT */ +#define WM5100_AIF3_UNDERCLOCKED_EINT 0x0040 /* AIF3_UNDERCLOCKED_EINT */ +#define WM5100_AIF3_UNDERCLOCKED_EINT_MASK 0x0040 /* AIF3_UNDERCLOCKED_EINT */ +#define WM5100_AIF3_UNDERCLOCKED_EINT_SHIFT 6 /* AIF3_UNDERCLOCKED_EINT */ +#define WM5100_AIF3_UNDERCLOCKED_EINT_WIDTH 1 /* AIF3_UNDERCLOCKED_EINT */ +#define WM5100_AIF2_UNDERCLOCKED_EINT 0x0020 /* AIF2_UNDERCLOCKED_EINT */ +#define WM5100_AIF2_UNDERCLOCKED_EINT_MASK 0x0020 /* AIF2_UNDERCLOCKED_EINT */ +#define WM5100_AIF2_UNDERCLOCKED_EINT_SHIFT 5 /* AIF2_UNDERCLOCKED_EINT */ +#define WM5100_AIF2_UNDERCLOCKED_EINT_WIDTH 1 /* AIF2_UNDERCLOCKED_EINT */ +#define WM5100_AIF1_UNDERCLOCKED_EINT 0x0010 /* AIF1_UNDERCLOCKED_EINT */ +#define WM5100_AIF1_UNDERCLOCKED_EINT_MASK 0x0010 /* AIF1_UNDERCLOCKED_EINT */ +#define WM5100_AIF1_UNDERCLOCKED_EINT_SHIFT 4 /* AIF1_UNDERCLOCKED_EINT */ +#define WM5100_AIF1_UNDERCLOCKED_EINT_WIDTH 1 /* AIF1_UNDERCLOCKED_EINT */ +#define WM5100_ASRC_UNDERCLOCKED_EINT 0x0008 /* ASRC_UNDERCLOCKED_EINT */ +#define WM5100_ASRC_UNDERCLOCKED_EINT_MASK 0x0008 /* ASRC_UNDERCLOCKED_EINT */ +#define WM5100_ASRC_UNDERCLOCKED_EINT_SHIFT 3 /* ASRC_UNDERCLOCKED_EINT */ +#define WM5100_ASRC_UNDERCLOCKED_EINT_WIDTH 1 /* ASRC_UNDERCLOCKED_EINT */ +#define WM5100_DAC_UNDERCLOCKED_EINT 0x0004 /* DAC_UNDERCLOCKED_EINT */ +#define WM5100_DAC_UNDERCLOCKED_EINT_MASK 0x0004 /* DAC_UNDERCLOCKED_EINT */ +#define WM5100_DAC_UNDERCLOCKED_EINT_SHIFT 2 /* DAC_UNDERCLOCKED_EINT */ +#define WM5100_DAC_UNDERCLOCKED_EINT_WIDTH 1 /* DAC_UNDERCLOCKED_EINT */ +#define WM5100_ADC_UNDERCLOCKED_EINT 0x0002 /* ADC_UNDERCLOCKED_EINT */ +#define WM5100_ADC_UNDERCLOCKED_EINT_MASK 0x0002 /* ADC_UNDERCLOCKED_EINT */ +#define WM5100_ADC_UNDERCLOCKED_EINT_SHIFT 1 /* ADC_UNDERCLOCKED_EINT */ +#define WM5100_ADC_UNDERCLOCKED_EINT_WIDTH 1 /* ADC_UNDERCLOCKED_EINT */ +#define WM5100_MIXER_UNDERCLOCKED_EINT 0x0001 /* MIXER_UNDERCLOCKED_EINT */ +#define WM5100_MIXER_UNDERCLOCKED_EINT_MASK 0x0001 /* MIXER_UNDERCLOCKED_EINT */ +#define WM5100_MIXER_UNDERCLOCKED_EINT_SHIFT 0 /* MIXER_UNDERCLOCKED_EINT */ +#define WM5100_MIXER_UNDERCLOCKED_EINT_WIDTH 1 /* MIXER_UNDERCLOCKED_EINT */ + +/* + * R3332 (0xD04) - Interrupt Raw Status 2 + */ +#define WM5100_DSP_IRQ6_STS 0x0020 /* DSP_IRQ6_STS */ +#define WM5100_DSP_IRQ6_STS_MASK 0x0020 /* DSP_IRQ6_STS */ +#define WM5100_DSP_IRQ6_STS_SHIFT 5 /* DSP_IRQ6_STS */ +#define WM5100_DSP_IRQ6_STS_WIDTH 1 /* DSP_IRQ6_STS */ +#define WM5100_DSP_IRQ5_STS 0x0010 /* DSP_IRQ5_STS */ +#define WM5100_DSP_IRQ5_STS_MASK 0x0010 /* DSP_IRQ5_STS */ +#define WM5100_DSP_IRQ5_STS_SHIFT 4 /* DSP_IRQ5_STS */ +#define WM5100_DSP_IRQ5_STS_WIDTH 1 /* DSP_IRQ5_STS */ +#define WM5100_DSP_IRQ4_STS 0x0008 /* DSP_IRQ4_STS */ +#define WM5100_DSP_IRQ4_STS_MASK 0x0008 /* DSP_IRQ4_STS */ +#define WM5100_DSP_IRQ4_STS_SHIFT 3 /* DSP_IRQ4_STS */ +#define WM5100_DSP_IRQ4_STS_WIDTH 1 /* DSP_IRQ4_STS */ +#define WM5100_DSP_IRQ3_STS 0x0004 /* DSP_IRQ3_STS */ +#define WM5100_DSP_IRQ3_STS_MASK 0x0004 /* DSP_IRQ3_STS */ +#define WM5100_DSP_IRQ3_STS_SHIFT 2 /* DSP_IRQ3_STS */ +#define WM5100_DSP_IRQ3_STS_WIDTH 1 /* DSP_IRQ3_STS */ +#define WM5100_DSP_IRQ2_STS 0x0002 /* DSP_IRQ2_STS */ +#define WM5100_DSP_IRQ2_STS_MASK 0x0002 /* DSP_IRQ2_STS */ +#define WM5100_DSP_IRQ2_STS_SHIFT 1 /* DSP_IRQ2_STS */ +#define WM5100_DSP_IRQ2_STS_WIDTH 1 /* DSP_IRQ2_STS */ +#define WM5100_DSP_IRQ1_STS 0x0001 /* DSP_IRQ1_STS */ +#define WM5100_DSP_IRQ1_STS_MASK 0x0001 /* DSP_IRQ1_STS */ +#define WM5100_DSP_IRQ1_STS_SHIFT 0 /* DSP_IRQ1_STS */ +#define WM5100_DSP_IRQ1_STS_WIDTH 1 /* DSP_IRQ1_STS */ + +/* + * R3333 (0xD05) - Interrupt Raw Status 3 + */ +#define WM5100_SPK_SHUTDOWN_WARN_STS 0x8000 /* SPK_SHUTDOWN_WARN_STS */ +#define WM5100_SPK_SHUTDOWN_WARN_STS_MASK 0x8000 /* SPK_SHUTDOWN_WARN_STS */ +#define WM5100_SPK_SHUTDOWN_WARN_STS_SHIFT 15 /* SPK_SHUTDOWN_WARN_STS */ +#define WM5100_SPK_SHUTDOWN_WARN_STS_WIDTH 1 /* SPK_SHUTDOWN_WARN_STS */ +#define WM5100_SPK_SHUTDOWN_STS 0x4000 /* SPK_SHUTDOWN_STS */ +#define WM5100_SPK_SHUTDOWN_STS_MASK 0x4000 /* SPK_SHUTDOWN_STS */ +#define WM5100_SPK_SHUTDOWN_STS_SHIFT 14 /* SPK_SHUTDOWN_STS */ +#define WM5100_SPK_SHUTDOWN_STS_WIDTH 1 /* SPK_SHUTDOWN_STS */ +#define WM5100_HPDET_STS 0x2000 /* HPDET_STS */ +#define WM5100_HPDET_STS_MASK 0x2000 /* HPDET_STS */ +#define WM5100_HPDET_STS_SHIFT 13 /* HPDET_STS */ +#define WM5100_HPDET_STS_WIDTH 1 /* HPDET_STS */ +#define WM5100_DRC_SID_DET_STS 0x0200 /* DRC_SID_DET_STS */ +#define WM5100_DRC_SID_DET_STS_MASK 0x0200 /* DRC_SID_DET_STS */ +#define WM5100_DRC_SID_DET_STS_SHIFT 9 /* DRC_SID_DET_STS */ +#define WM5100_DRC_SID_DET_STS_WIDTH 1 /* DRC_SID_DET_STS */ +#define WM5100_ASRC2_LOCK_STS 0x0100 /* ASRC2_LOCK_STS */ +#define WM5100_ASRC2_LOCK_STS_MASK 0x0100 /* ASRC2_LOCK_STS */ +#define WM5100_ASRC2_LOCK_STS_SHIFT 8 /* ASRC2_LOCK_STS */ +#define WM5100_ASRC2_LOCK_STS_WIDTH 1 /* ASRC2_LOCK_STS */ +#define WM5100_ASRC1_LOCK_STS 0x0080 /* ASRC1_LOCK_STS */ +#define WM5100_ASRC1_LOCK_STS_MASK 0x0080 /* ASRC1_LOCK_STS */ +#define WM5100_ASRC1_LOCK_STS_SHIFT 7 /* ASRC1_LOCK_STS */ +#define WM5100_ASRC1_LOCK_STS_WIDTH 1 /* ASRC1_LOCK_STS */ +#define WM5100_FLL2_LOCK_STS 0x0008 /* FLL2_LOCK_STS */ +#define WM5100_FLL2_LOCK_STS_MASK 0x0008 /* FLL2_LOCK_STS */ +#define WM5100_FLL2_LOCK_STS_SHIFT 3 /* FLL2_LOCK_STS */ +#define WM5100_FLL2_LOCK_STS_WIDTH 1 /* FLL2_LOCK_STS */ +#define WM5100_FLL1_LOCK_STS 0x0004 /* FLL1_LOCK_STS */ +#define WM5100_FLL1_LOCK_STS_MASK 0x0004 /* FLL1_LOCK_STS */ +#define WM5100_FLL1_LOCK_STS_SHIFT 2 /* FLL1_LOCK_STS */ +#define WM5100_FLL1_LOCK_STS_WIDTH 1 /* FLL1_LOCK_STS */ +#define WM5100_CLKGEN_ERR_STS 0x0002 /* CLKGEN_ERR_STS */ +#define WM5100_CLKGEN_ERR_STS_MASK 0x0002 /* CLKGEN_ERR_STS */ +#define WM5100_CLKGEN_ERR_STS_SHIFT 1 /* CLKGEN_ERR_STS */ +#define WM5100_CLKGEN_ERR_STS_WIDTH 1 /* CLKGEN_ERR_STS */ +#define WM5100_CLKGEN_ERR_ASYNC_STS 0x0001 /* CLKGEN_ERR_ASYNC_STS */ +#define WM5100_CLKGEN_ERR_ASYNC_STS_MASK 0x0001 /* CLKGEN_ERR_ASYNC_STS */ +#define WM5100_CLKGEN_ERR_ASYNC_STS_SHIFT 0 /* CLKGEN_ERR_ASYNC_STS */ +#define WM5100_CLKGEN_ERR_ASYNC_STS_WIDTH 1 /* CLKGEN_ERR_ASYNC_STS */ + +/* + * R3334 (0xD06) - Interrupt Raw Status 4 + */ +#define WM5100_AIF3_ERR_STS 0x2000 /* AIF3_ERR_STS */ +#define WM5100_AIF3_ERR_STS_MASK 0x2000 /* AIF3_ERR_STS */ +#define WM5100_AIF3_ERR_STS_SHIFT 13 /* AIF3_ERR_STS */ +#define WM5100_AIF3_ERR_STS_WIDTH 1 /* AIF3_ERR_STS */ +#define WM5100_AIF2_ERR_STS 0x1000 /* AIF2_ERR_STS */ +#define WM5100_AIF2_ERR_STS_MASK 0x1000 /* AIF2_ERR_STS */ +#define WM5100_AIF2_ERR_STS_SHIFT 12 /* AIF2_ERR_STS */ +#define WM5100_AIF2_ERR_STS_WIDTH 1 /* AIF2_ERR_STS */ +#define WM5100_AIF1_ERR_STS 0x0800 /* AIF1_ERR_STS */ +#define WM5100_AIF1_ERR_STS_MASK 0x0800 /* AIF1_ERR_STS */ +#define WM5100_AIF1_ERR_STS_SHIFT 11 /* AIF1_ERR_STS */ +#define WM5100_AIF1_ERR_STS_WIDTH 1 /* AIF1_ERR_STS */ +#define WM5100_CTRLIF_ERR_STS 0x0400 /* CTRLIF_ERR_STS */ +#define WM5100_CTRLIF_ERR_STS_MASK 0x0400 /* CTRLIF_ERR_STS */ +#define WM5100_CTRLIF_ERR_STS_SHIFT 10 /* CTRLIF_ERR_STS */ +#define WM5100_CTRLIF_ERR_STS_WIDTH 1 /* CTRLIF_ERR_STS */ +#define WM5100_ISRC2_UNDERCLOCKED_STS 0x0200 /* ISRC2_UNDERCLOCKED_STS */ +#define WM5100_ISRC2_UNDERCLOCKED_STS_MASK 0x0200 /* ISRC2_UNDERCLOCKED_STS */ +#define WM5100_ISRC2_UNDERCLOCKED_STS_SHIFT 9 /* ISRC2_UNDERCLOCKED_STS */ +#define WM5100_ISRC2_UNDERCLOCKED_STS_WIDTH 1 /* ISRC2_UNDERCLOCKED_STS */ +#define WM5100_ISRC1_UNDERCLOCKED_STS 0x0100 /* ISRC1_UNDERCLOCKED_STS */ +#define WM5100_ISRC1_UNDERCLOCKED_STS_MASK 0x0100 /* ISRC1_UNDERCLOCKED_STS */ +#define WM5100_ISRC1_UNDERCLOCKED_STS_SHIFT 8 /* ISRC1_UNDERCLOCKED_STS */ +#define WM5100_ISRC1_UNDERCLOCKED_STS_WIDTH 1 /* ISRC1_UNDERCLOCKED_STS */ +#define WM5100_FX_UNDERCLOCKED_STS 0x0080 /* FX_UNDERCLOCKED_STS */ +#define WM5100_FX_UNDERCLOCKED_STS_MASK 0x0080 /* FX_UNDERCLOCKED_STS */ +#define WM5100_FX_UNDERCLOCKED_STS_SHIFT 7 /* FX_UNDERCLOCKED_STS */ +#define WM5100_FX_UNDERCLOCKED_STS_WIDTH 1 /* FX_UNDERCLOCKED_STS */ +#define WM5100_AIF3_UNDERCLOCKED_STS 0x0040 /* AIF3_UNDERCLOCKED_STS */ +#define WM5100_AIF3_UNDERCLOCKED_STS_MASK 0x0040 /* AIF3_UNDERCLOCKED_STS */ +#define WM5100_AIF3_UNDERCLOCKED_STS_SHIFT 6 /* AIF3_UNDERCLOCKED_STS */ +#define WM5100_AIF3_UNDERCLOCKED_STS_WIDTH 1 /* AIF3_UNDERCLOCKED_STS */ +#define WM5100_AIF2_UNDERCLOCKED_STS 0x0020 /* AIF2_UNDERCLOCKED_STS */ +#define WM5100_AIF2_UNDERCLOCKED_STS_MASK 0x0020 /* AIF2_UNDERCLOCKED_STS */ +#define WM5100_AIF2_UNDERCLOCKED_STS_SHIFT 5 /* AIF2_UNDERCLOCKED_STS */ +#define WM5100_AIF2_UNDERCLOCKED_STS_WIDTH 1 /* AIF2_UNDERCLOCKED_STS */ +#define WM5100_AIF1_UNDERCLOCKED_STS 0x0010 /* AIF1_UNDERCLOCKED_STS */ +#define WM5100_AIF1_UNDERCLOCKED_STS_MASK 0x0010 /* AIF1_UNDERCLOCKED_STS */ +#define WM5100_AIF1_UNDERCLOCKED_STS_SHIFT 4 /* AIF1_UNDERCLOCKED_STS */ +#define WM5100_AIF1_UNDERCLOCKED_STS_WIDTH 1 /* AIF1_UNDERCLOCKED_STS */ +#define WM5100_ASRC_UNDERCLOCKED_STS 0x0008 /* ASRC_UNDERCLOCKED_STS */ +#define WM5100_ASRC_UNDERCLOCKED_STS_MASK 0x0008 /* ASRC_UNDERCLOCKED_STS */ +#define WM5100_ASRC_UNDERCLOCKED_STS_SHIFT 3 /* ASRC_UNDERCLOCKED_STS */ +#define WM5100_ASRC_UNDERCLOCKED_STS_WIDTH 1 /* ASRC_UNDERCLOCKED_STS */ +#define WM5100_DAC_UNDERCLOCKED_STS 0x0004 /* DAC_UNDERCLOCKED_STS */ +#define WM5100_DAC_UNDERCLOCKED_STS_MASK 0x0004 /* DAC_UNDERCLOCKED_STS */ +#define WM5100_DAC_UNDERCLOCKED_STS_SHIFT 2 /* DAC_UNDERCLOCKED_STS */ +#define WM5100_DAC_UNDERCLOCKED_STS_WIDTH 1 /* DAC_UNDERCLOCKED_STS */ +#define WM5100_ADC_UNDERCLOCKED_STS 0x0002 /* ADC_UNDERCLOCKED_STS */ +#define WM5100_ADC_UNDERCLOCKED_STS_MASK 0x0002 /* ADC_UNDERCLOCKED_STS */ +#define WM5100_ADC_UNDERCLOCKED_STS_SHIFT 1 /* ADC_UNDERCLOCKED_STS */ +#define WM5100_ADC_UNDERCLOCKED_STS_WIDTH 1 /* ADC_UNDERCLOCKED_STS */ +#define WM5100_MIXER_UNDERCLOCKED_STS 0x0001 /* MIXER_UNDERCLOCKED_STS */ +#define WM5100_MIXER_UNDERCLOCKED_STS_MASK 0x0001 /* MIXER_UNDERCLOCKED_STS */ +#define WM5100_MIXER_UNDERCLOCKED_STS_SHIFT 0 /* MIXER_UNDERCLOCKED_STS */ +#define WM5100_MIXER_UNDERCLOCKED_STS_WIDTH 1 /* MIXER_UNDERCLOCKED_STS */ + +/* + * R3335 (0xD07) - Interrupt Status 1 Mask + */ +#define WM5100_IM_GP6_EINT 0x0020 /* IM_GP6_EINT */ +#define WM5100_IM_GP6_EINT_MASK 0x0020 /* IM_GP6_EINT */ +#define WM5100_IM_GP6_EINT_SHIFT 5 /* IM_GP6_EINT */ +#define WM5100_IM_GP6_EINT_WIDTH 1 /* IM_GP6_EINT */ +#define WM5100_IM_GP5_EINT 0x0010 /* IM_GP5_EINT */ +#define WM5100_IM_GP5_EINT_MASK 0x0010 /* IM_GP5_EINT */ +#define WM5100_IM_GP5_EINT_SHIFT 4 /* IM_GP5_EINT */ +#define WM5100_IM_GP5_EINT_WIDTH 1 /* IM_GP5_EINT */ +#define WM5100_IM_GP4_EINT 0x0008 /* IM_GP4_EINT */ +#define WM5100_IM_GP4_EINT_MASK 0x0008 /* IM_GP4_EINT */ +#define WM5100_IM_GP4_EINT_SHIFT 3 /* IM_GP4_EINT */ +#define WM5100_IM_GP4_EINT_WIDTH 1 /* IM_GP4_EINT */ +#define WM5100_IM_GP3_EINT 0x0004 /* IM_GP3_EINT */ +#define WM5100_IM_GP3_EINT_MASK 0x0004 /* IM_GP3_EINT */ +#define WM5100_IM_GP3_EINT_SHIFT 2 /* IM_GP3_EINT */ +#define WM5100_IM_GP3_EINT_WIDTH 1 /* IM_GP3_EINT */ +#define WM5100_IM_GP2_EINT 0x0002 /* IM_GP2_EINT */ +#define WM5100_IM_GP2_EINT_MASK 0x0002 /* IM_GP2_EINT */ +#define WM5100_IM_GP2_EINT_SHIFT 1 /* IM_GP2_EINT */ +#define WM5100_IM_GP2_EINT_WIDTH 1 /* IM_GP2_EINT */ +#define WM5100_IM_GP1_EINT 0x0001 /* IM_GP1_EINT */ +#define WM5100_IM_GP1_EINT_MASK 0x0001 /* IM_GP1_EINT */ +#define WM5100_IM_GP1_EINT_SHIFT 0 /* IM_GP1_EINT */ +#define WM5100_IM_GP1_EINT_WIDTH 1 /* IM_GP1_EINT */ + +/* + * R3336 (0xD08) - Interrupt Status 2 Mask + */ +#define WM5100_IM_DSP_IRQ6_EINT 0x0020 /* IM_DSP_IRQ6_EINT */ +#define WM5100_IM_DSP_IRQ6_EINT_MASK 0x0020 /* IM_DSP_IRQ6_EINT */ +#define WM5100_IM_DSP_IRQ6_EINT_SHIFT 5 /* IM_DSP_IRQ6_EINT */ +#define WM5100_IM_DSP_IRQ6_EINT_WIDTH 1 /* IM_DSP_IRQ6_EINT */ +#define WM5100_IM_DSP_IRQ5_EINT 0x0010 /* IM_DSP_IRQ5_EINT */ +#define WM5100_IM_DSP_IRQ5_EINT_MASK 0x0010 /* IM_DSP_IRQ5_EINT */ +#define WM5100_IM_DSP_IRQ5_EINT_SHIFT 4 /* IM_DSP_IRQ5_EINT */ +#define WM5100_IM_DSP_IRQ5_EINT_WIDTH 1 /* IM_DSP_IRQ5_EINT */ +#define WM5100_IM_DSP_IRQ4_EINT 0x0008 /* IM_DSP_IRQ4_EINT */ +#define WM5100_IM_DSP_IRQ4_EINT_MASK 0x0008 /* IM_DSP_IRQ4_EINT */ +#define WM5100_IM_DSP_IRQ4_EINT_SHIFT 3 /* IM_DSP_IRQ4_EINT */ +#define WM5100_IM_DSP_IRQ4_EINT_WIDTH 1 /* IM_DSP_IRQ4_EINT */ +#define WM5100_IM_DSP_IRQ3_EINT 0x0004 /* IM_DSP_IRQ3_EINT */ +#define WM5100_IM_DSP_IRQ3_EINT_MASK 0x0004 /* IM_DSP_IRQ3_EINT */ +#define WM5100_IM_DSP_IRQ3_EINT_SHIFT 2 /* IM_DSP_IRQ3_EINT */ +#define WM5100_IM_DSP_IRQ3_EINT_WIDTH 1 /* IM_DSP_IRQ3_EINT */ +#define WM5100_IM_DSP_IRQ2_EINT 0x0002 /* IM_DSP_IRQ2_EINT */ +#define WM5100_IM_DSP_IRQ2_EINT_MASK 0x0002 /* IM_DSP_IRQ2_EINT */ +#define WM5100_IM_DSP_IRQ2_EINT_SHIFT 1 /* IM_DSP_IRQ2_EINT */ +#define WM5100_IM_DSP_IRQ2_EINT_WIDTH 1 /* IM_DSP_IRQ2_EINT */ +#define WM5100_IM_DSP_IRQ1_EINT 0x0001 /* IM_DSP_IRQ1_EINT */ +#define WM5100_IM_DSP_IRQ1_EINT_MASK 0x0001 /* IM_DSP_IRQ1_EINT */ +#define WM5100_IM_DSP_IRQ1_EINT_SHIFT 0 /* IM_DSP_IRQ1_EINT */ +#define WM5100_IM_DSP_IRQ1_EINT_WIDTH 1 /* IM_DSP_IRQ1_EINT */ + +/* + * R3337 (0xD09) - Interrupt Status 3 Mask + */ +#define WM5100_IM_SPK_SHUTDOWN_WARN_EINT 0x8000 /* IM_SPK_SHUTDOWN_WARN_EINT */ +#define WM5100_IM_SPK_SHUTDOWN_WARN_EINT_MASK 0x8000 /* IM_SPK_SHUTDOWN_WARN_EINT */ +#define WM5100_IM_SPK_SHUTDOWN_WARN_EINT_SHIFT 15 /* IM_SPK_SHUTDOWN_WARN_EINT */ +#define WM5100_IM_SPK_SHUTDOWN_WARN_EINT_WIDTH 1 /* IM_SPK_SHUTDOWN_WARN_EINT */ +#define WM5100_IM_SPK_SHUTDOWN_EINT 0x4000 /* IM_SPK_SHUTDOWN_EINT */ +#define WM5100_IM_SPK_SHUTDOWN_EINT_MASK 0x4000 /* IM_SPK_SHUTDOWN_EINT */ +#define WM5100_IM_SPK_SHUTDOWN_EINT_SHIFT 14 /* IM_SPK_SHUTDOWN_EINT */ +#define WM5100_IM_SPK_SHUTDOWN_EINT_WIDTH 1 /* IM_SPK_SHUTDOWN_EINT */ +#define WM5100_IM_HPDET_EINT 0x2000 /* IM_HPDET_EINT */ +#define WM5100_IM_HPDET_EINT_MASK 0x2000 /* IM_HPDET_EINT */ +#define WM5100_IM_HPDET_EINT_SHIFT 13 /* IM_HPDET_EINT */ +#define WM5100_IM_HPDET_EINT_WIDTH 1 /* IM_HPDET_EINT */ +#define WM5100_IM_ACCDET_EINT 0x1000 /* IM_ACCDET_EINT */ +#define WM5100_IM_ACCDET_EINT_MASK 0x1000 /* IM_ACCDET_EINT */ +#define WM5100_IM_ACCDET_EINT_SHIFT 12 /* IM_ACCDET_EINT */ +#define WM5100_IM_ACCDET_EINT_WIDTH 1 /* IM_ACCDET_EINT */ +#define WM5100_IM_DRC_SIG_DET_EINT 0x0200 /* IM_DRC_SIG_DET_EINT */ +#define WM5100_IM_DRC_SIG_DET_EINT_MASK 0x0200 /* IM_DRC_SIG_DET_EINT */ +#define WM5100_IM_DRC_SIG_DET_EINT_SHIFT 9 /* IM_DRC_SIG_DET_EINT */ +#define WM5100_IM_DRC_SIG_DET_EINT_WIDTH 1 /* IM_DRC_SIG_DET_EINT */ +#define WM5100_IM_ASRC2_LOCK_EINT 0x0100 /* IM_ASRC2_LOCK_EINT */ +#define WM5100_IM_ASRC2_LOCK_EINT_MASK 0x0100 /* IM_ASRC2_LOCK_EINT */ +#define WM5100_IM_ASRC2_LOCK_EINT_SHIFT 8 /* IM_ASRC2_LOCK_EINT */ +#define WM5100_IM_ASRC2_LOCK_EINT_WIDTH 1 /* IM_ASRC2_LOCK_EINT */ +#define WM5100_IM_ASRC1_LOCK_EINT 0x0080 /* IM_ASRC1_LOCK_EINT */ +#define WM5100_IM_ASRC1_LOCK_EINT_MASK 0x0080 /* IM_ASRC1_LOCK_EINT */ +#define WM5100_IM_ASRC1_LOCK_EINT_SHIFT 7 /* IM_ASRC1_LOCK_EINT */ +#define WM5100_IM_ASRC1_LOCK_EINT_WIDTH 1 /* IM_ASRC1_LOCK_EINT */ +#define WM5100_IM_FLL2_LOCK_EINT 0x0008 /* IM_FLL2_LOCK_EINT */ +#define WM5100_IM_FLL2_LOCK_EINT_MASK 0x0008 /* IM_FLL2_LOCK_EINT */ +#define WM5100_IM_FLL2_LOCK_EINT_SHIFT 3 /* IM_FLL2_LOCK_EINT */ +#define WM5100_IM_FLL2_LOCK_EINT_WIDTH 1 /* IM_FLL2_LOCK_EINT */ +#define WM5100_IM_FLL1_LOCK_EINT 0x0004 /* IM_FLL1_LOCK_EINT */ +#define WM5100_IM_FLL1_LOCK_EINT_MASK 0x0004 /* IM_FLL1_LOCK_EINT */ +#define WM5100_IM_FLL1_LOCK_EINT_SHIFT 2 /* IM_FLL1_LOCK_EINT */ +#define WM5100_IM_FLL1_LOCK_EINT_WIDTH 1 /* IM_FLL1_LOCK_EINT */ +#define WM5100_IM_CLKGEN_ERR_EINT 0x0002 /* IM_CLKGEN_ERR_EINT */ +#define WM5100_IM_CLKGEN_ERR_EINT_MASK 0x0002 /* IM_CLKGEN_ERR_EINT */ +#define WM5100_IM_CLKGEN_ERR_EINT_SHIFT 1 /* IM_CLKGEN_ERR_EINT */ +#define WM5100_IM_CLKGEN_ERR_EINT_WIDTH 1 /* IM_CLKGEN_ERR_EINT */ +#define WM5100_IM_CLKGEN_ERR_ASYNC_EINT 0x0001 /* IM_CLKGEN_ERR_ASYNC_EINT */ +#define WM5100_IM_CLKGEN_ERR_ASYNC_EINT_MASK 0x0001 /* IM_CLKGEN_ERR_ASYNC_EINT */ +#define WM5100_IM_CLKGEN_ERR_ASYNC_EINT_SHIFT 0 /* IM_CLKGEN_ERR_ASYNC_EINT */ +#define WM5100_IM_CLKGEN_ERR_ASYNC_EINT_WIDTH 1 /* IM_CLKGEN_ERR_ASYNC_EINT */ + +/* + * R3338 (0xD0A) - Interrupt Status 4 Mask + */ +#define WM5100_IM_AIF3_ERR_EINT 0x2000 /* IM_AIF3_ERR_EINT */ +#define WM5100_IM_AIF3_ERR_EINT_MASK 0x2000 /* IM_AIF3_ERR_EINT */ +#define WM5100_IM_AIF3_ERR_EINT_SHIFT 13 /* IM_AIF3_ERR_EINT */ +#define WM5100_IM_AIF3_ERR_EINT_WIDTH 1 /* IM_AIF3_ERR_EINT */ +#define WM5100_IM_AIF2_ERR_EINT 0x1000 /* IM_AIF2_ERR_EINT */ +#define WM5100_IM_AIF2_ERR_EINT_MASK 0x1000 /* IM_AIF2_ERR_EINT */ +#define WM5100_IM_AIF2_ERR_EINT_SHIFT 12 /* IM_AIF2_ERR_EINT */ +#define WM5100_IM_AIF2_ERR_EINT_WIDTH 1 /* IM_AIF2_ERR_EINT */ +#define WM5100_IM_AIF1_ERR_EINT 0x0800 /* IM_AIF1_ERR_EINT */ +#define WM5100_IM_AIF1_ERR_EINT_MASK 0x0800 /* IM_AIF1_ERR_EINT */ +#define WM5100_IM_AIF1_ERR_EINT_SHIFT 11 /* IM_AIF1_ERR_EINT */ +#define WM5100_IM_AIF1_ERR_EINT_WIDTH 1 /* IM_AIF1_ERR_EINT */ +#define WM5100_IM_CTRLIF_ERR_EINT 0x0400 /* IM_CTRLIF_ERR_EINT */ +#define WM5100_IM_CTRLIF_ERR_EINT_MASK 0x0400 /* IM_CTRLIF_ERR_EINT */ +#define WM5100_IM_CTRLIF_ERR_EINT_SHIFT 10 /* IM_CTRLIF_ERR_EINT */ +#define WM5100_IM_CTRLIF_ERR_EINT_WIDTH 1 /* IM_CTRLIF_ERR_EINT */ +#define WM5100_IM_ISRC2_UNDERCLOCKED_EINT 0x0200 /* IM_ISRC2_UNDERCLOCKED_EINT */ +#define WM5100_IM_ISRC2_UNDERCLOCKED_EINT_MASK 0x0200 /* IM_ISRC2_UNDERCLOCKED_EINT */ +#define WM5100_IM_ISRC2_UNDERCLOCKED_EINT_SHIFT 9 /* IM_ISRC2_UNDERCLOCKED_EINT */ +#define WM5100_IM_ISRC2_UNDERCLOCKED_EINT_WIDTH 1 /* IM_ISRC2_UNDERCLOCKED_EINT */ +#define WM5100_IM_ISRC1_UNDERCLOCKED_EINT 0x0100 /* IM_ISRC1_UNDERCLOCKED_EINT */ +#define WM5100_IM_ISRC1_UNDERCLOCKED_EINT_MASK 0x0100 /* IM_ISRC1_UNDERCLOCKED_EINT */ +#define WM5100_IM_ISRC1_UNDERCLOCKED_EINT_SHIFT 8 /* IM_ISRC1_UNDERCLOCKED_EINT */ +#define WM5100_IM_ISRC1_UNDERCLOCKED_EINT_WIDTH 1 /* IM_ISRC1_UNDERCLOCKED_EINT */ +#define WM5100_IM_FX_UNDERCLOCKED_EINT 0x0080 /* IM_FX_UNDERCLOCKED_EINT */ +#define WM5100_IM_FX_UNDERCLOCKED_EINT_MASK 0x0080 /* IM_FX_UNDERCLOCKED_EINT */ +#define WM5100_IM_FX_UNDERCLOCKED_EINT_SHIFT 7 /* IM_FX_UNDERCLOCKED_EINT */ +#define WM5100_IM_FX_UNDERCLOCKED_EINT_WIDTH 1 /* IM_FX_UNDERCLOCKED_EINT */ +#define WM5100_IM_AIF3_UNDERCLOCKED_EINT 0x0040 /* IM_AIF3_UNDERCLOCKED_EINT */ +#define WM5100_IM_AIF3_UNDERCLOCKED_EINT_MASK 0x0040 /* IM_AIF3_UNDERCLOCKED_EINT */ +#define WM5100_IM_AIF3_UNDERCLOCKED_EINT_SHIFT 6 /* IM_AIF3_UNDERCLOCKED_EINT */ +#define WM5100_IM_AIF3_UNDERCLOCKED_EINT_WIDTH 1 /* IM_AIF3_UNDERCLOCKED_EINT */ +#define WM5100_IM_AIF2_UNDERCLOCKED_EINT 0x0020 /* IM_AIF2_UNDERCLOCKED_EINT */ +#define WM5100_IM_AIF2_UNDERCLOCKED_EINT_MASK 0x0020 /* IM_AIF2_UNDERCLOCKED_EINT */ +#define WM5100_IM_AIF2_UNDERCLOCKED_EINT_SHIFT 5 /* IM_AIF2_UNDERCLOCKED_EINT */ +#define WM5100_IM_AIF2_UNDERCLOCKED_EINT_WIDTH 1 /* IM_AIF2_UNDERCLOCKED_EINT */ +#define WM5100_IM_AIF1_UNDERCLOCKED_EINT 0x0010 /* IM_AIF1_UNDERCLOCKED_EINT */ +#define WM5100_IM_AIF1_UNDERCLOCKED_EINT_MASK 0x0010 /* IM_AIF1_UNDERCLOCKED_EINT */ +#define WM5100_IM_AIF1_UNDERCLOCKED_EINT_SHIFT 4 /* IM_AIF1_UNDERCLOCKED_EINT */ +#define WM5100_IM_AIF1_UNDERCLOCKED_EINT_WIDTH 1 /* IM_AIF1_UNDERCLOCKED_EINT */ +#define WM5100_IM_ASRC_UNDERCLOCKED_EINT 0x0008 /* IM_ASRC_UNDERCLOCKED_EINT */ +#define WM5100_IM_ASRC_UNDERCLOCKED_EINT_MASK 0x0008 /* IM_ASRC_UNDERCLOCKED_EINT */ +#define WM5100_IM_ASRC_UNDERCLOCKED_EINT_SHIFT 3 /* IM_ASRC_UNDERCLOCKED_EINT */ +#define WM5100_IM_ASRC_UNDERCLOCKED_EINT_WIDTH 1 /* IM_ASRC_UNDERCLOCKED_EINT */ +#define WM5100_IM_DAC_UNDERCLOCKED_EINT 0x0004 /* IM_DAC_UNDERCLOCKED_EINT */ +#define WM5100_IM_DAC_UNDERCLOCKED_EINT_MASK 0x0004 /* IM_DAC_UNDERCLOCKED_EINT */ +#define WM5100_IM_DAC_UNDERCLOCKED_EINT_SHIFT 2 /* IM_DAC_UNDERCLOCKED_EINT */ +#define WM5100_IM_DAC_UNDERCLOCKED_EINT_WIDTH 1 /* IM_DAC_UNDERCLOCKED_EINT */ +#define WM5100_IM_ADC_UNDERCLOCKED_EINT 0x0002 /* IM_ADC_UNDERCLOCKED_EINT */ +#define WM5100_IM_ADC_UNDERCLOCKED_EINT_MASK 0x0002 /* IM_ADC_UNDERCLOCKED_EINT */ +#define WM5100_IM_ADC_UNDERCLOCKED_EINT_SHIFT 1 /* IM_ADC_UNDERCLOCKED_EINT */ +#define WM5100_IM_ADC_UNDERCLOCKED_EINT_WIDTH 1 /* IM_ADC_UNDERCLOCKED_EINT */ +#define WM5100_IM_MIXER_UNDERCLOCKED_EINT 0x0001 /* IM_MIXER_UNDERCLOCKED_EINT */ +#define WM5100_IM_MIXER_UNDERCLOCKED_EINT_MASK 0x0001 /* IM_MIXER_UNDERCLOCKED_EINT */ +#define WM5100_IM_MIXER_UNDERCLOCKED_EINT_SHIFT 0 /* IM_MIXER_UNDERCLOCKED_EINT */ +#define WM5100_IM_MIXER_UNDERCLOCKED_EINT_WIDTH 1 /* IM_MIXER_UNDERCLOCKED_EINT */ + +/* + * R3359 (0xD1F) - Interrupt Control + */ +#define WM5100_IM_IRQ 0x0001 /* IM_IRQ */ +#define WM5100_IM_IRQ_MASK 0x0001 /* IM_IRQ */ +#define WM5100_IM_IRQ_SHIFT 0 /* IM_IRQ */ +#define WM5100_IM_IRQ_WIDTH 1 /* IM_IRQ */ + +/* + * R3360 (0xD20) - IRQ Debounce 1 + */ +#define WM5100_SPK_SHUTDOWN_WARN_DB 0x0200 /* SPK_SHUTDOWN_WARN_DB */ +#define WM5100_SPK_SHUTDOWN_WARN_DB_MASK 0x0200 /* SPK_SHUTDOWN_WARN_DB */ +#define WM5100_SPK_SHUTDOWN_WARN_DB_SHIFT 9 /* SPK_SHUTDOWN_WARN_DB */ +#define WM5100_SPK_SHUTDOWN_WARN_DB_WIDTH 1 /* SPK_SHUTDOWN_WARN_DB */ +#define WM5100_SPK_SHUTDOWN_DB 0x0100 /* SPK_SHUTDOWN_DB */ +#define WM5100_SPK_SHUTDOWN_DB_MASK 0x0100 /* SPK_SHUTDOWN_DB */ +#define WM5100_SPK_SHUTDOWN_DB_SHIFT 8 /* SPK_SHUTDOWN_DB */ +#define WM5100_SPK_SHUTDOWN_DB_WIDTH 1 /* SPK_SHUTDOWN_DB */ +#define WM5100_FLL1_LOCK_IRQ_DB 0x0008 /* FLL1_LOCK_IRQ_DB */ +#define WM5100_FLL1_LOCK_IRQ_DB_MASK 0x0008 /* FLL1_LOCK_IRQ_DB */ +#define WM5100_FLL1_LOCK_IRQ_DB_SHIFT 3 /* FLL1_LOCK_IRQ_DB */ +#define WM5100_FLL1_LOCK_IRQ_DB_WIDTH 1 /* FLL1_LOCK_IRQ_DB */ +#define WM5100_FLL2_LOCK_IRQ_DB 0x0004 /* FLL2_LOCK_IRQ_DB */ +#define WM5100_FLL2_LOCK_IRQ_DB_MASK 0x0004 /* FLL2_LOCK_IRQ_DB */ +#define WM5100_FLL2_LOCK_IRQ_DB_SHIFT 2 /* FLL2_LOCK_IRQ_DB */ +#define WM5100_FLL2_LOCK_IRQ_DB_WIDTH 1 /* FLL2_LOCK_IRQ_DB */ +#define WM5100_CLKGEN_ERR_IRQ_DB 0x0002 /* CLKGEN_ERR_IRQ_DB */ +#define WM5100_CLKGEN_ERR_IRQ_DB_MASK 0x0002 /* CLKGEN_ERR_IRQ_DB */ +#define WM5100_CLKGEN_ERR_IRQ_DB_SHIFT 1 /* CLKGEN_ERR_IRQ_DB */ +#define WM5100_CLKGEN_ERR_IRQ_DB_WIDTH 1 /* CLKGEN_ERR_IRQ_DB */ +#define WM5100_CLKGEN_ERR_ASYNC_IRQ_DB 0x0001 /* CLKGEN_ERR_ASYNC_IRQ_DB */ +#define WM5100_CLKGEN_ERR_ASYNC_IRQ_DB_MASK 0x0001 /* CLKGEN_ERR_ASYNC_IRQ_DB */ +#define WM5100_CLKGEN_ERR_ASYNC_IRQ_DB_SHIFT 0 /* CLKGEN_ERR_ASYNC_IRQ_DB */ +#define WM5100_CLKGEN_ERR_ASYNC_IRQ_DB_WIDTH 1 /* CLKGEN_ERR_ASYNC_IRQ_DB */ + +/* + * R3361 (0xD21) - IRQ Debounce 2 + */ +#define WM5100_AIF_ERR_DB 0x0001 /* AIF_ERR_DB */ +#define WM5100_AIF_ERR_DB_MASK 0x0001 /* AIF_ERR_DB */ +#define WM5100_AIF_ERR_DB_SHIFT 0 /* AIF_ERR_DB */ +#define WM5100_AIF_ERR_DB_WIDTH 1 /* AIF_ERR_DB */ + +/* + * R3584 (0xE00) - FX_Ctrl + */ +#define WM5100_FX_STS_MASK 0xFFC0 /* FX_STS - [15:6] */ +#define WM5100_FX_STS_SHIFT 6 /* FX_STS - [15:6] */ +#define WM5100_FX_STS_WIDTH 10 /* FX_STS - [15:6] */ +#define WM5100_FX_RATE_MASK 0x0003 /* FX_RATE - [1:0] */ +#define WM5100_FX_RATE_SHIFT 0 /* FX_RATE - [1:0] */ +#define WM5100_FX_RATE_WIDTH 2 /* FX_RATE - [1:0] */ + +/* + * R3600 (0xE10) - EQ1_1 + */ +#define WM5100_EQ1_B1_GAIN_MASK 0xF800 /* EQ1_B1_GAIN - [15:11] */ +#define WM5100_EQ1_B1_GAIN_SHIFT 11 /* EQ1_B1_GAIN - [15:11] */ +#define WM5100_EQ1_B1_GAIN_WIDTH 5 /* EQ1_B1_GAIN - [15:11] */ +#define WM5100_EQ1_B2_GAIN_MASK 0x07C0 /* EQ1_B2_GAIN - [10:6] */ +#define WM5100_EQ1_B2_GAIN_SHIFT 6 /* EQ1_B2_GAIN - [10:6] */ +#define WM5100_EQ1_B2_GAIN_WIDTH 5 /* EQ1_B2_GAIN - [10:6] */ +#define WM5100_EQ1_B3_GAIN_MASK 0x003E /* EQ1_B3_GAIN - [5:1] */ +#define WM5100_EQ1_B3_GAIN_SHIFT 1 /* EQ1_B3_GAIN - [5:1] */ +#define WM5100_EQ1_B3_GAIN_WIDTH 5 /* EQ1_B3_GAIN - [5:1] */ +#define WM5100_EQ1_ENA 0x0001 /* EQ1_ENA */ +#define WM5100_EQ1_ENA_MASK 0x0001 /* EQ1_ENA */ +#define WM5100_EQ1_ENA_SHIFT 0 /* EQ1_ENA */ +#define WM5100_EQ1_ENA_WIDTH 1 /* EQ1_ENA */ + +/* + * R3601 (0xE11) - EQ1_2 + */ +#define WM5100_EQ1_B4_GAIN_MASK 0xF800 /* EQ1_B4_GAIN - [15:11] */ +#define WM5100_EQ1_B4_GAIN_SHIFT 11 /* EQ1_B4_GAIN - [15:11] */ +#define WM5100_EQ1_B4_GAIN_WIDTH 5 /* EQ1_B4_GAIN - [15:11] */ +#define WM5100_EQ1_B5_GAIN_MASK 0x07C0 /* EQ1_B5_GAIN - [10:6] */ +#define WM5100_EQ1_B5_GAIN_SHIFT 6 /* EQ1_B5_GAIN - [10:6] */ +#define WM5100_EQ1_B5_GAIN_WIDTH 5 /* EQ1_B5_GAIN - [10:6] */ + +/* + * R3602 (0xE12) - EQ1_3 + */ +#define WM5100_EQ1_B1_A_MASK 0xFFFF /* EQ1_B1_A - [15:0] */ +#define WM5100_EQ1_B1_A_SHIFT 0 /* EQ1_B1_A - [15:0] */ +#define WM5100_EQ1_B1_A_WIDTH 16 /* EQ1_B1_A - [15:0] */ + +/* + * R3603 (0xE13) - EQ1_4 + */ +#define WM5100_EQ1_B1_B_MASK 0xFFFF /* EQ1_B1_B - [15:0] */ +#define WM5100_EQ1_B1_B_SHIFT 0 /* EQ1_B1_B - [15:0] */ +#define WM5100_EQ1_B1_B_WIDTH 16 /* EQ1_B1_B - [15:0] */ + +/* + * R3604 (0xE14) - EQ1_5 + */ +#define WM5100_EQ1_B1_PG_MASK 0xFFFF /* EQ1_B1_PG - [15:0] */ +#define WM5100_EQ1_B1_PG_SHIFT 0 /* EQ1_B1_PG - [15:0] */ +#define WM5100_EQ1_B1_PG_WIDTH 16 /* EQ1_B1_PG - [15:0] */ + +/* + * R3605 (0xE15) - EQ1_6 + */ +#define WM5100_EQ1_B2_A_MASK 0xFFFF /* EQ1_B2_A - [15:0] */ +#define WM5100_EQ1_B2_A_SHIFT 0 /* EQ1_B2_A - [15:0] */ +#define WM5100_EQ1_B2_A_WIDTH 16 /* EQ1_B2_A - [15:0] */ + +/* + * R3606 (0xE16) - EQ1_7 + */ +#define WM5100_EQ1_B2_B_MASK 0xFFFF /* EQ1_B2_B - [15:0] */ +#define WM5100_EQ1_B2_B_SHIFT 0 /* EQ1_B2_B - [15:0] */ +#define WM5100_EQ1_B2_B_WIDTH 16 /* EQ1_B2_B - [15:0] */ + +/* + * R3607 (0xE17) - EQ1_8 + */ +#define WM5100_EQ1_B2_C_MASK 0xFFFF /* EQ1_B2_C - [15:0] */ +#define WM5100_EQ1_B2_C_SHIFT 0 /* EQ1_B2_C - [15:0] */ +#define WM5100_EQ1_B2_C_WIDTH 16 /* EQ1_B2_C - [15:0] */ + +/* + * R3608 (0xE18) - EQ1_9 + */ +#define WM5100_EQ1_B2_PG_MASK 0xFFFF /* EQ1_B2_PG - [15:0] */ +#define WM5100_EQ1_B2_PG_SHIFT 0 /* EQ1_B2_PG - [15:0] */ +#define WM5100_EQ1_B2_PG_WIDTH 16 /* EQ1_B2_PG - [15:0] */ + +/* + * R3609 (0xE19) - EQ1_10 + */ +#define WM5100_EQ1_B3_A_MASK 0xFFFF /* EQ1_B3_A - [15:0] */ +#define WM5100_EQ1_B3_A_SHIFT 0 /* EQ1_B3_A - [15:0] */ +#define WM5100_EQ1_B3_A_WIDTH 16 /* EQ1_B3_A - [15:0] */ + +/* + * R3610 (0xE1A) - EQ1_11 + */ +#define WM5100_EQ1_B3_B_MASK 0xFFFF /* EQ1_B3_B - [15:0] */ +#define WM5100_EQ1_B3_B_SHIFT 0 /* EQ1_B3_B - [15:0] */ +#define WM5100_EQ1_B3_B_WIDTH 16 /* EQ1_B3_B - [15:0] */ + +/* + * R3611 (0xE1B) - EQ1_12 + */ +#define WM5100_EQ1_B3_C_MASK 0xFFFF /* EQ1_B3_C - [15:0] */ +#define WM5100_EQ1_B3_C_SHIFT 0 /* EQ1_B3_C - [15:0] */ +#define WM5100_EQ1_B3_C_WIDTH 16 /* EQ1_B3_C - [15:0] */ + +/* + * R3612 (0xE1C) - EQ1_13 + */ +#define WM5100_EQ1_B3_PG_MASK 0xFFFF /* EQ1_B3_PG - [15:0] */ +#define WM5100_EQ1_B3_PG_SHIFT 0 /* EQ1_B3_PG - [15:0] */ +#define WM5100_EQ1_B3_PG_WIDTH 16 /* EQ1_B3_PG - [15:0] */ + +/* + * R3613 (0xE1D) - EQ1_14 + */ +#define WM5100_EQ1_B4_A_MASK 0xFFFF /* EQ1_B4_A - [15:0] */ +#define WM5100_EQ1_B4_A_SHIFT 0 /* EQ1_B4_A - [15:0] */ +#define WM5100_EQ1_B4_A_WIDTH 16 /* EQ1_B4_A - [15:0] */ + +/* + * R3614 (0xE1E) - EQ1_15 + */ +#define WM5100_EQ1_B4_B_MASK 0xFFFF /* EQ1_B4_B - [15:0] */ +#define WM5100_EQ1_B4_B_SHIFT 0 /* EQ1_B4_B - [15:0] */ +#define WM5100_EQ1_B4_B_WIDTH 16 /* EQ1_B4_B - [15:0] */ + +/* + * R3615 (0xE1F) - EQ1_16 + */ +#define WM5100_EQ1_B4_C_MASK 0xFFFF /* EQ1_B4_C - [15:0] */ +#define WM5100_EQ1_B4_C_SHIFT 0 /* EQ1_B4_C - [15:0] */ +#define WM5100_EQ1_B4_C_WIDTH 16 /* EQ1_B4_C - [15:0] */ + +/* + * R3616 (0xE20) - EQ1_17 + */ +#define WM5100_EQ1_B4_PG_MASK 0xFFFF /* EQ1_B4_PG - [15:0] */ +#define WM5100_EQ1_B4_PG_SHIFT 0 /* EQ1_B4_PG - [15:0] */ +#define WM5100_EQ1_B4_PG_WIDTH 16 /* EQ1_B4_PG - [15:0] */ + +/* + * R3617 (0xE21) - EQ1_18 + */ +#define WM5100_EQ1_B5_A_MASK 0xFFFF /* EQ1_B5_A - [15:0] */ +#define WM5100_EQ1_B5_A_SHIFT 0 /* EQ1_B5_A - [15:0] */ +#define WM5100_EQ1_B5_A_WIDTH 16 /* EQ1_B5_A - [15:0] */ + +/* + * R3618 (0xE22) - EQ1_19 + */ +#define WM5100_EQ1_B5_B_MASK 0xFFFF /* EQ1_B5_B - [15:0] */ +#define WM5100_EQ1_B5_B_SHIFT 0 /* EQ1_B5_B - [15:0] */ +#define WM5100_EQ1_B5_B_WIDTH 16 /* EQ1_B5_B - [15:0] */ + +/* + * R3619 (0xE23) - EQ1_20 + */ +#define WM5100_EQ1_B5_PG_MASK 0xFFFF /* EQ1_B5_PG - [15:0] */ +#define WM5100_EQ1_B5_PG_SHIFT 0 /* EQ1_B5_PG - [15:0] */ +#define WM5100_EQ1_B5_PG_WIDTH 16 /* EQ1_B5_PG - [15:0] */ + +/* + * R3622 (0xE26) - EQ2_1 + */ +#define WM5100_EQ2_B1_GAIN_MASK 0xF800 /* EQ2_B1_GAIN - [15:11] */ +#define WM5100_EQ2_B1_GAIN_SHIFT 11 /* EQ2_B1_GAIN - [15:11] */ +#define WM5100_EQ2_B1_GAIN_WIDTH 5 /* EQ2_B1_GAIN - [15:11] */ +#define WM5100_EQ2_B2_GAIN_MASK 0x07C0 /* EQ2_B2_GAIN - [10:6] */ +#define WM5100_EQ2_B2_GAIN_SHIFT 6 /* EQ2_B2_GAIN - [10:6] */ +#define WM5100_EQ2_B2_GAIN_WIDTH 5 /* EQ2_B2_GAIN - [10:6] */ +#define WM5100_EQ2_B3_GAIN_MASK 0x003E /* EQ2_B3_GAIN - [5:1] */ +#define WM5100_EQ2_B3_GAIN_SHIFT 1 /* EQ2_B3_GAIN - [5:1] */ +#define WM5100_EQ2_B3_GAIN_WIDTH 5 /* EQ2_B3_GAIN - [5:1] */ +#define WM5100_EQ2_ENA 0x0001 /* EQ2_ENA */ +#define WM5100_EQ2_ENA_MASK 0x0001 /* EQ2_ENA */ +#define WM5100_EQ2_ENA_SHIFT 0 /* EQ2_ENA */ +#define WM5100_EQ2_ENA_WIDTH 1 /* EQ2_ENA */ + +/* + * R3623 (0xE27) - EQ2_2 + */ +#define WM5100_EQ2_B4_GAIN_MASK 0xF800 /* EQ2_B4_GAIN - [15:11] */ +#define WM5100_EQ2_B4_GAIN_SHIFT 11 /* EQ2_B4_GAIN - [15:11] */ +#define WM5100_EQ2_B4_GAIN_WIDTH 5 /* EQ2_B4_GAIN - [15:11] */ +#define WM5100_EQ2_B5_GAIN_MASK 0x07C0 /* EQ2_B5_GAIN - [10:6] */ +#define WM5100_EQ2_B5_GAIN_SHIFT 6 /* EQ2_B5_GAIN - [10:6] */ +#define WM5100_EQ2_B5_GAIN_WIDTH 5 /* EQ2_B5_GAIN - [10:6] */ + +/* + * R3624 (0xE28) - EQ2_3 + */ +#define WM5100_EQ2_B1_A_MASK 0xFFFF /* EQ2_B1_A - [15:0] */ +#define WM5100_EQ2_B1_A_SHIFT 0 /* EQ2_B1_A - [15:0] */ +#define WM5100_EQ2_B1_A_WIDTH 16 /* EQ2_B1_A - [15:0] */ + +/* + * R3625 (0xE29) - EQ2_4 + */ +#define WM5100_EQ2_B1_B_MASK 0xFFFF /* EQ2_B1_B - [15:0] */ +#define WM5100_EQ2_B1_B_SHIFT 0 /* EQ2_B1_B - [15:0] */ +#define WM5100_EQ2_B1_B_WIDTH 16 /* EQ2_B1_B - [15:0] */ + +/* + * R3626 (0xE2A) - EQ2_5 + */ +#define WM5100_EQ2_B1_PG_MASK 0xFFFF /* EQ2_B1_PG - [15:0] */ +#define WM5100_EQ2_B1_PG_SHIFT 0 /* EQ2_B1_PG - [15:0] */ +#define WM5100_EQ2_B1_PG_WIDTH 16 /* EQ2_B1_PG - [15:0] */ + +/* + * R3627 (0xE2B) - EQ2_6 + */ +#define WM5100_EQ2_B2_A_MASK 0xFFFF /* EQ2_B2_A - [15:0] */ +#define WM5100_EQ2_B2_A_SHIFT 0 /* EQ2_B2_A - [15:0] */ +#define WM5100_EQ2_B2_A_WIDTH 16 /* EQ2_B2_A - [15:0] */ + +/* + * R3628 (0xE2C) - EQ2_7 + */ +#define WM5100_EQ2_B2_B_MASK 0xFFFF /* EQ2_B2_B - [15:0] */ +#define WM5100_EQ2_B2_B_SHIFT 0 /* EQ2_B2_B - [15:0] */ +#define WM5100_EQ2_B2_B_WIDTH 16 /* EQ2_B2_B - [15:0] */ + +/* + * R3629 (0xE2D) - EQ2_8 + */ +#define WM5100_EQ2_B2_C_MASK 0xFFFF /* EQ2_B2_C - [15:0] */ +#define WM5100_EQ2_B2_C_SHIFT 0 /* EQ2_B2_C - [15:0] */ +#define WM5100_EQ2_B2_C_WIDTH 16 /* EQ2_B2_C - [15:0] */ + +/* + * R3630 (0xE2E) - EQ2_9 + */ +#define WM5100_EQ2_B2_PG_MASK 0xFFFF /* EQ2_B2_PG - [15:0] */ +#define WM5100_EQ2_B2_PG_SHIFT 0 /* EQ2_B2_PG - [15:0] */ +#define WM5100_EQ2_B2_PG_WIDTH 16 /* EQ2_B2_PG - [15:0] */ + +/* + * R3631 (0xE2F) - EQ2_10 + */ +#define WM5100_EQ2_B3_A_MASK 0xFFFF /* EQ2_B3_A - [15:0] */ +#define WM5100_EQ2_B3_A_SHIFT 0 /* EQ2_B3_A - [15:0] */ +#define WM5100_EQ2_B3_A_WIDTH 16 /* EQ2_B3_A - [15:0] */ + +/* + * R3632 (0xE30) - EQ2_11 + */ +#define WM5100_EQ2_B3_B_MASK 0xFFFF /* EQ2_B3_B - [15:0] */ +#define WM5100_EQ2_B3_B_SHIFT 0 /* EQ2_B3_B - [15:0] */ +#define WM5100_EQ2_B3_B_WIDTH 16 /* EQ2_B3_B - [15:0] */ + +/* + * R3633 (0xE31) - EQ2_12 + */ +#define WM5100_EQ2_B3_C_MASK 0xFFFF /* EQ2_B3_C - [15:0] */ +#define WM5100_EQ2_B3_C_SHIFT 0 /* EQ2_B3_C - [15:0] */ +#define WM5100_EQ2_B3_C_WIDTH 16 /* EQ2_B3_C - [15:0] */ + +/* + * R3634 (0xE32) - EQ2_13 + */ +#define WM5100_EQ2_B3_PG_MASK 0xFFFF /* EQ2_B3_PG - [15:0] */ +#define WM5100_EQ2_B3_PG_SHIFT 0 /* EQ2_B3_PG - [15:0] */ +#define WM5100_EQ2_B3_PG_WIDTH 16 /* EQ2_B3_PG - [15:0] */ + +/* + * R3635 (0xE33) - EQ2_14 + */ +#define WM5100_EQ2_B4_A_MASK 0xFFFF /* EQ2_B4_A - [15:0] */ +#define WM5100_EQ2_B4_A_SHIFT 0 /* EQ2_B4_A - [15:0] */ +#define WM5100_EQ2_B4_A_WIDTH 16 /* EQ2_B4_A - [15:0] */ + +/* + * R3636 (0xE34) - EQ2_15 + */ +#define WM5100_EQ2_B4_B_MASK 0xFFFF /* EQ2_B4_B - [15:0] */ +#define WM5100_EQ2_B4_B_SHIFT 0 /* EQ2_B4_B - [15:0] */ +#define WM5100_EQ2_B4_B_WIDTH 16 /* EQ2_B4_B - [15:0] */ + +/* + * R3637 (0xE35) - EQ2_16 + */ +#define WM5100_EQ2_B4_C_MASK 0xFFFF /* EQ2_B4_C - [15:0] */ +#define WM5100_EQ2_B4_C_SHIFT 0 /* EQ2_B4_C - [15:0] */ +#define WM5100_EQ2_B4_C_WIDTH 16 /* EQ2_B4_C - [15:0] */ + +/* + * R3638 (0xE36) - EQ2_17 + */ +#define WM5100_EQ2_B4_PG_MASK 0xFFFF /* EQ2_B4_PG - [15:0] */ +#define WM5100_EQ2_B4_PG_SHIFT 0 /* EQ2_B4_PG - [15:0] */ +#define WM5100_EQ2_B4_PG_WIDTH 16 /* EQ2_B4_PG - [15:0] */ + +/* + * R3639 (0xE37) - EQ2_18 + */ +#define WM5100_EQ2_B5_A_MASK 0xFFFF /* EQ2_B5_A - [15:0] */ +#define WM5100_EQ2_B5_A_SHIFT 0 /* EQ2_B5_A - [15:0] */ +#define WM5100_EQ2_B5_A_WIDTH 16 /* EQ2_B5_A - [15:0] */ + +/* + * R3640 (0xE38) - EQ2_19 + */ +#define WM5100_EQ2_B5_B_MASK 0xFFFF /* EQ2_B5_B - [15:0] */ +#define WM5100_EQ2_B5_B_SHIFT 0 /* EQ2_B5_B - [15:0] */ +#define WM5100_EQ2_B5_B_WIDTH 16 /* EQ2_B5_B - [15:0] */ + +/* + * R3641 (0xE39) - EQ2_20 + */ +#define WM5100_EQ2_B5_PG_MASK 0xFFFF /* EQ2_B5_PG - [15:0] */ +#define WM5100_EQ2_B5_PG_SHIFT 0 /* EQ2_B5_PG - [15:0] */ +#define WM5100_EQ2_B5_PG_WIDTH 16 /* EQ2_B5_PG - [15:0] */ + +/* + * R3644 (0xE3C) - EQ3_1 + */ +#define WM5100_EQ3_B1_GAIN_MASK 0xF800 /* EQ3_B1_GAIN - [15:11] */ +#define WM5100_EQ3_B1_GAIN_SHIFT 11 /* EQ3_B1_GAIN - [15:11] */ +#define WM5100_EQ3_B1_GAIN_WIDTH 5 /* EQ3_B1_GAIN - [15:11] */ +#define WM5100_EQ3_B2_GAIN_MASK 0x07C0 /* EQ3_B2_GAIN - [10:6] */ +#define WM5100_EQ3_B2_GAIN_SHIFT 6 /* EQ3_B2_GAIN - [10:6] */ +#define WM5100_EQ3_B2_GAIN_WIDTH 5 /* EQ3_B2_GAIN - [10:6] */ +#define WM5100_EQ3_B3_GAIN_MASK 0x003E /* EQ3_B3_GAIN - [5:1] */ +#define WM5100_EQ3_B3_GAIN_SHIFT 1 /* EQ3_B3_GAIN - [5:1] */ +#define WM5100_EQ3_B3_GAIN_WIDTH 5 /* EQ3_B3_GAIN - [5:1] */ +#define WM5100_EQ3_ENA 0x0001 /* EQ3_ENA */ +#define WM5100_EQ3_ENA_MASK 0x0001 /* EQ3_ENA */ +#define WM5100_EQ3_ENA_SHIFT 0 /* EQ3_ENA */ +#define WM5100_EQ3_ENA_WIDTH 1 /* EQ3_ENA */ + +/* + * R3645 (0xE3D) - EQ3_2 + */ +#define WM5100_EQ3_B4_GAIN_MASK 0xF800 /* EQ3_B4_GAIN - [15:11] */ +#define WM5100_EQ3_B4_GAIN_SHIFT 11 /* EQ3_B4_GAIN - [15:11] */ +#define WM5100_EQ3_B4_GAIN_WIDTH 5 /* EQ3_B4_GAIN - [15:11] */ +#define WM5100_EQ3_B5_GAIN_MASK 0x07C0 /* EQ3_B5_GAIN - [10:6] */ +#define WM5100_EQ3_B5_GAIN_SHIFT 6 /* EQ3_B5_GAIN - [10:6] */ +#define WM5100_EQ3_B5_GAIN_WIDTH 5 /* EQ3_B5_GAIN - [10:6] */ + +/* + * R3646 (0xE3E) - EQ3_3 + */ +#define WM5100_EQ3_B1_A_MASK 0xFFFF /* EQ3_B1_A - [15:0] */ +#define WM5100_EQ3_B1_A_SHIFT 0 /* EQ3_B1_A - [15:0] */ +#define WM5100_EQ3_B1_A_WIDTH 16 /* EQ3_B1_A - [15:0] */ + +/* + * R3647 (0xE3F) - EQ3_4 + */ +#define WM5100_EQ3_B1_B_MASK 0xFFFF /* EQ3_B1_B - [15:0] */ +#define WM5100_EQ3_B1_B_SHIFT 0 /* EQ3_B1_B - [15:0] */ +#define WM5100_EQ3_B1_B_WIDTH 16 /* EQ3_B1_B - [15:0] */ + +/* + * R3648 (0xE40) - EQ3_5 + */ +#define WM5100_EQ3_B1_PG_MASK 0xFFFF /* EQ3_B1_PG - [15:0] */ +#define WM5100_EQ3_B1_PG_SHIFT 0 /* EQ3_B1_PG - [15:0] */ +#define WM5100_EQ3_B1_PG_WIDTH 16 /* EQ3_B1_PG - [15:0] */ + +/* + * R3649 (0xE41) - EQ3_6 + */ +#define WM5100_EQ3_B2_A_MASK 0xFFFF /* EQ3_B2_A - [15:0] */ +#define WM5100_EQ3_B2_A_SHIFT 0 /* EQ3_B2_A - [15:0] */ +#define WM5100_EQ3_B2_A_WIDTH 16 /* EQ3_B2_A - [15:0] */ + +/* + * R3650 (0xE42) - EQ3_7 + */ +#define WM5100_EQ3_B2_B_MASK 0xFFFF /* EQ3_B2_B - [15:0] */ +#define WM5100_EQ3_B2_B_SHIFT 0 /* EQ3_B2_B - [15:0] */ +#define WM5100_EQ3_B2_B_WIDTH 16 /* EQ3_B2_B - [15:0] */ + +/* + * R3651 (0xE43) - EQ3_8 + */ +#define WM5100_EQ3_B2_C_MASK 0xFFFF /* EQ3_B2_C - [15:0] */ +#define WM5100_EQ3_B2_C_SHIFT 0 /* EQ3_B2_C - [15:0] */ +#define WM5100_EQ3_B2_C_WIDTH 16 /* EQ3_B2_C - [15:0] */ + +/* + * R3652 (0xE44) - EQ3_9 + */ +#define WM5100_EQ3_B2_PG_MASK 0xFFFF /* EQ3_B2_PG - [15:0] */ +#define WM5100_EQ3_B2_PG_SHIFT 0 /* EQ3_B2_PG - [15:0] */ +#define WM5100_EQ3_B2_PG_WIDTH 16 /* EQ3_B2_PG - [15:0] */ + +/* + * R3653 (0xE45) - EQ3_10 + */ +#define WM5100_EQ3_B3_A_MASK 0xFFFF /* EQ3_B3_A - [15:0] */ +#define WM5100_EQ3_B3_A_SHIFT 0 /* EQ3_B3_A - [15:0] */ +#define WM5100_EQ3_B3_A_WIDTH 16 /* EQ3_B3_A - [15:0] */ + +/* + * R3654 (0xE46) - EQ3_11 + */ +#define WM5100_EQ3_B3_B_MASK 0xFFFF /* EQ3_B3_B - [15:0] */ +#define WM5100_EQ3_B3_B_SHIFT 0 /* EQ3_B3_B - [15:0] */ +#define WM5100_EQ3_B3_B_WIDTH 16 /* EQ3_B3_B - [15:0] */ + +/* + * R3655 (0xE47) - EQ3_12 + */ +#define WM5100_EQ3_B3_C_MASK 0xFFFF /* EQ3_B3_C - [15:0] */ +#define WM5100_EQ3_B3_C_SHIFT 0 /* EQ3_B3_C - [15:0] */ +#define WM5100_EQ3_B3_C_WIDTH 16 /* EQ3_B3_C - [15:0] */ + +/* + * R3656 (0xE48) - EQ3_13 + */ +#define WM5100_EQ3_B3_PG_MASK 0xFFFF /* EQ3_B3_PG - [15:0] */ +#define WM5100_EQ3_B3_PG_SHIFT 0 /* EQ3_B3_PG - [15:0] */ +#define WM5100_EQ3_B3_PG_WIDTH 16 /* EQ3_B3_PG - [15:0] */ + +/* + * R3657 (0xE49) - EQ3_14 + */ +#define WM5100_EQ3_B4_A_MASK 0xFFFF /* EQ3_B4_A - [15:0] */ +#define WM5100_EQ3_B4_A_SHIFT 0 /* EQ3_B4_A - [15:0] */ +#define WM5100_EQ3_B4_A_WIDTH 16 /* EQ3_B4_A - [15:0] */ + +/* + * R3658 (0xE4A) - EQ3_15 + */ +#define WM5100_EQ3_B4_B_MASK 0xFFFF /* EQ3_B4_B - [15:0] */ +#define WM5100_EQ3_B4_B_SHIFT 0 /* EQ3_B4_B - [15:0] */ +#define WM5100_EQ3_B4_B_WIDTH 16 /* EQ3_B4_B - [15:0] */ + +/* + * R3659 (0xE4B) - EQ3_16 + */ +#define WM5100_EQ3_B4_C_MASK 0xFFFF /* EQ3_B4_C - [15:0] */ +#define WM5100_EQ3_B4_C_SHIFT 0 /* EQ3_B4_C - [15:0] */ +#define WM5100_EQ3_B4_C_WIDTH 16 /* EQ3_B4_C - [15:0] */ + +/* + * R3660 (0xE4C) - EQ3_17 + */ +#define WM5100_EQ3_B4_PG_MASK 0xFFFF /* EQ3_B4_PG - [15:0] */ +#define WM5100_EQ3_B4_PG_SHIFT 0 /* EQ3_B4_PG - [15:0] */ +#define WM5100_EQ3_B4_PG_WIDTH 16 /* EQ3_B4_PG - [15:0] */ + +/* + * R3661 (0xE4D) - EQ3_18 + */ +#define WM5100_EQ3_B5_A_MASK 0xFFFF /* EQ3_B5_A - [15:0] */ +#define WM5100_EQ3_B5_A_SHIFT 0 /* EQ3_B5_A - [15:0] */ +#define WM5100_EQ3_B5_A_WIDTH 16 /* EQ3_B5_A - [15:0] */ + +/* + * R3662 (0xE4E) - EQ3_19 + */ +#define WM5100_EQ3_B5_B_MASK 0xFFFF /* EQ3_B5_B - [15:0] */ +#define WM5100_EQ3_B5_B_SHIFT 0 /* EQ3_B5_B - [15:0] */ +#define WM5100_EQ3_B5_B_WIDTH 16 /* EQ3_B5_B - [15:0] */ + +/* + * R3663 (0xE4F) - EQ3_20 + */ +#define WM5100_EQ3_B5_PG_MASK 0xFFFF /* EQ3_B5_PG - [15:0] */ +#define WM5100_EQ3_B5_PG_SHIFT 0 /* EQ3_B5_PG - [15:0] */ +#define WM5100_EQ3_B5_PG_WIDTH 16 /* EQ3_B5_PG - [15:0] */ + +/* + * R3666 (0xE52) - EQ4_1 + */ +#define WM5100_EQ4_B1_GAIN_MASK 0xF800 /* EQ4_B1_GAIN - [15:11] */ +#define WM5100_EQ4_B1_GAIN_SHIFT 11 /* EQ4_B1_GAIN - [15:11] */ +#define WM5100_EQ4_B1_GAIN_WIDTH 5 /* EQ4_B1_GAIN - [15:11] */ +#define WM5100_EQ4_B2_GAIN_MASK 0x07C0 /* EQ4_B2_GAIN - [10:6] */ +#define WM5100_EQ4_B2_GAIN_SHIFT 6 /* EQ4_B2_GAIN - [10:6] */ +#define WM5100_EQ4_B2_GAIN_WIDTH 5 /* EQ4_B2_GAIN - [10:6] */ +#define WM5100_EQ4_B3_GAIN_MASK 0x003E /* EQ4_B3_GAIN - [5:1] */ +#define WM5100_EQ4_B3_GAIN_SHIFT 1 /* EQ4_B3_GAIN - [5:1] */ +#define WM5100_EQ4_B3_GAIN_WIDTH 5 /* EQ4_B3_GAIN - [5:1] */ +#define WM5100_EQ4_ENA 0x0001 /* EQ4_ENA */ +#define WM5100_EQ4_ENA_MASK 0x0001 /* EQ4_ENA */ +#define WM5100_EQ4_ENA_SHIFT 0 /* EQ4_ENA */ +#define WM5100_EQ4_ENA_WIDTH 1 /* EQ4_ENA */ + +/* + * R3667 (0xE53) - EQ4_2 + */ +#define WM5100_EQ4_B4_GAIN_MASK 0xF800 /* EQ4_B4_GAIN - [15:11] */ +#define WM5100_EQ4_B4_GAIN_SHIFT 11 /* EQ4_B4_GAIN - [15:11] */ +#define WM5100_EQ4_B4_GAIN_WIDTH 5 /* EQ4_B4_GAIN - [15:11] */ +#define WM5100_EQ4_B5_GAIN_MASK 0x07C0 /* EQ4_B5_GAIN - [10:6] */ +#define WM5100_EQ4_B5_GAIN_SHIFT 6 /* EQ4_B5_GAIN - [10:6] */ +#define WM5100_EQ4_B5_GAIN_WIDTH 5 /* EQ4_B5_GAIN - [10:6] */ + +/* + * R3668 (0xE54) - EQ4_3 + */ +#define WM5100_EQ4_B1_A_MASK 0xFFFF /* EQ4_B1_A - [15:0] */ +#define WM5100_EQ4_B1_A_SHIFT 0 /* EQ4_B1_A - [15:0] */ +#define WM5100_EQ4_B1_A_WIDTH 16 /* EQ4_B1_A - [15:0] */ + +/* + * R3669 (0xE55) - EQ4_4 + */ +#define WM5100_EQ4_B1_B_MASK 0xFFFF /* EQ4_B1_B - [15:0] */ +#define WM5100_EQ4_B1_B_SHIFT 0 /* EQ4_B1_B - [15:0] */ +#define WM5100_EQ4_B1_B_WIDTH 16 /* EQ4_B1_B - [15:0] */ + +/* + * R3670 (0xE56) - EQ4_5 + */ +#define WM5100_EQ4_B1_PG_MASK 0xFFFF /* EQ4_B1_PG - [15:0] */ +#define WM5100_EQ4_B1_PG_SHIFT 0 /* EQ4_B1_PG - [15:0] */ +#define WM5100_EQ4_B1_PG_WIDTH 16 /* EQ4_B1_PG - [15:0] */ + +/* + * R3671 (0xE57) - EQ4_6 + */ +#define WM5100_EQ4_B2_A_MASK 0xFFFF /* EQ4_B2_A - [15:0] */ +#define WM5100_EQ4_B2_A_SHIFT 0 /* EQ4_B2_A - [15:0] */ +#define WM5100_EQ4_B2_A_WIDTH 16 /* EQ4_B2_A - [15:0] */ + +/* + * R3672 (0xE58) - EQ4_7 + */ +#define WM5100_EQ4_B2_B_MASK 0xFFFF /* EQ4_B2_B - [15:0] */ +#define WM5100_EQ4_B2_B_SHIFT 0 /* EQ4_B2_B - [15:0] */ +#define WM5100_EQ4_B2_B_WIDTH 16 /* EQ4_B2_B - [15:0] */ + +/* + * R3673 (0xE59) - EQ4_8 + */ +#define WM5100_EQ4_B2_C_MASK 0xFFFF /* EQ4_B2_C - [15:0] */ +#define WM5100_EQ4_B2_C_SHIFT 0 /* EQ4_B2_C - [15:0] */ +#define WM5100_EQ4_B2_C_WIDTH 16 /* EQ4_B2_C - [15:0] */ + +/* + * R3674 (0xE5A) - EQ4_9 + */ +#define WM5100_EQ4_B2_PG_MASK 0xFFFF /* EQ4_B2_PG - [15:0] */ +#define WM5100_EQ4_B2_PG_SHIFT 0 /* EQ4_B2_PG - [15:0] */ +#define WM5100_EQ4_B2_PG_WIDTH 16 /* EQ4_B2_PG - [15:0] */ + +/* + * R3675 (0xE5B) - EQ4_10 + */ +#define WM5100_EQ4_B3_A_MASK 0xFFFF /* EQ4_B3_A - [15:0] */ +#define WM5100_EQ4_B3_A_SHIFT 0 /* EQ4_B3_A - [15:0] */ +#define WM5100_EQ4_B3_A_WIDTH 16 /* EQ4_B3_A - [15:0] */ + +/* + * R3676 (0xE5C) - EQ4_11 + */ +#define WM5100_EQ4_B3_B_MASK 0xFFFF /* EQ4_B3_B - [15:0] */ +#define WM5100_EQ4_B3_B_SHIFT 0 /* EQ4_B3_B - [15:0] */ +#define WM5100_EQ4_B3_B_WIDTH 16 /* EQ4_B3_B - [15:0] */ + +/* + * R3677 (0xE5D) - EQ4_12 + */ +#define WM5100_EQ4_B3_C_MASK 0xFFFF /* EQ4_B3_C - [15:0] */ +#define WM5100_EQ4_B3_C_SHIFT 0 /* EQ4_B3_C - [15:0] */ +#define WM5100_EQ4_B3_C_WIDTH 16 /* EQ4_B3_C - [15:0] */ + +/* + * R3678 (0xE5E) - EQ4_13 + */ +#define WM5100_EQ4_B3_PG_MASK 0xFFFF /* EQ4_B3_PG - [15:0] */ +#define WM5100_EQ4_B3_PG_SHIFT 0 /* EQ4_B3_PG - [15:0] */ +#define WM5100_EQ4_B3_PG_WIDTH 16 /* EQ4_B3_PG - [15:0] */ + +/* + * R3679 (0xE5F) - EQ4_14 + */ +#define WM5100_EQ4_B4_A_MASK 0xFFFF /* EQ4_B4_A - [15:0] */ +#define WM5100_EQ4_B4_A_SHIFT 0 /* EQ4_B4_A - [15:0] */ +#define WM5100_EQ4_B4_A_WIDTH 16 /* EQ4_B4_A - [15:0] */ + +/* + * R3680 (0xE60) - EQ4_15 + */ +#define WM5100_EQ4_B4_B_MASK 0xFFFF /* EQ4_B4_B - [15:0] */ +#define WM5100_EQ4_B4_B_SHIFT 0 /* EQ4_B4_B - [15:0] */ +#define WM5100_EQ4_B4_B_WIDTH 16 /* EQ4_B4_B - [15:0] */ + +/* + * R3681 (0xE61) - EQ4_16 + */ +#define WM5100_EQ4_B4_C_MASK 0xFFFF /* EQ4_B4_C - [15:0] */ +#define WM5100_EQ4_B4_C_SHIFT 0 /* EQ4_B4_C - [15:0] */ +#define WM5100_EQ4_B4_C_WIDTH 16 /* EQ4_B4_C - [15:0] */ + +/* + * R3682 (0xE62) - EQ4_17 + */ +#define WM5100_EQ4_B4_PG_MASK 0xFFFF /* EQ4_B4_PG - [15:0] */ +#define WM5100_EQ4_B4_PG_SHIFT 0 /* EQ4_B4_PG - [15:0] */ +#define WM5100_EQ4_B4_PG_WIDTH 16 /* EQ4_B4_PG - [15:0] */ + +/* + * R3683 (0xE63) - EQ4_18 + */ +#define WM5100_EQ4_B5_A_MASK 0xFFFF /* EQ4_B5_A - [15:0] */ +#define WM5100_EQ4_B5_A_SHIFT 0 /* EQ4_B5_A - [15:0] */ +#define WM5100_EQ4_B5_A_WIDTH 16 /* EQ4_B5_A - [15:0] */ + +/* + * R3684 (0xE64) - EQ4_19 + */ +#define WM5100_EQ4_B5_B_MASK 0xFFFF /* EQ4_B5_B - [15:0] */ +#define WM5100_EQ4_B5_B_SHIFT 0 /* EQ4_B5_B - [15:0] */ +#define WM5100_EQ4_B5_B_WIDTH 16 /* EQ4_B5_B - [15:0] */ + +/* + * R3685 (0xE65) - EQ4_20 + */ +#define WM5100_EQ4_B5_PG_MASK 0xFFFF /* EQ4_B5_PG - [15:0] */ +#define WM5100_EQ4_B5_PG_SHIFT 0 /* EQ4_B5_PG - [15:0] */ +#define WM5100_EQ4_B5_PG_WIDTH 16 /* EQ4_B5_PG - [15:0] */ + +/* + * R3712 (0xE80) - DRC1 ctrl1 + */ +#define WM5100_DRC_SIG_DET_RMS_MASK 0xF800 /* DRC_SIG_DET_RMS - [15:11] */ +#define WM5100_DRC_SIG_DET_RMS_SHIFT 11 /* DRC_SIG_DET_RMS - [15:11] */ +#define WM5100_DRC_SIG_DET_RMS_WIDTH 5 /* DRC_SIG_DET_RMS - [15:11] */ +#define WM5100_DRC_SIG_DET_PK_MASK 0x0600 /* DRC_SIG_DET_PK - [10:9] */ +#define WM5100_DRC_SIG_DET_PK_SHIFT 9 /* DRC_SIG_DET_PK - [10:9] */ +#define WM5100_DRC_SIG_DET_PK_WIDTH 2 /* DRC_SIG_DET_PK - [10:9] */ +#define WM5100_DRC_NG_ENA 0x0100 /* DRC_NG_ENA */ +#define WM5100_DRC_NG_ENA_MASK 0x0100 /* DRC_NG_ENA */ +#define WM5100_DRC_NG_ENA_SHIFT 8 /* DRC_NG_ENA */ +#define WM5100_DRC_NG_ENA_WIDTH 1 /* DRC_NG_ENA */ +#define WM5100_DRC_SIG_DET_MODE 0x0080 /* DRC_SIG_DET_MODE */ +#define WM5100_DRC_SIG_DET_MODE_MASK 0x0080 /* DRC_SIG_DET_MODE */ +#define WM5100_DRC_SIG_DET_MODE_SHIFT 7 /* DRC_SIG_DET_MODE */ +#define WM5100_DRC_SIG_DET_MODE_WIDTH 1 /* DRC_SIG_DET_MODE */ +#define WM5100_DRC_SIG_DET 0x0040 /* DRC_SIG_DET */ +#define WM5100_DRC_SIG_DET_MASK 0x0040 /* DRC_SIG_DET */ +#define WM5100_DRC_SIG_DET_SHIFT 6 /* DRC_SIG_DET */ +#define WM5100_DRC_SIG_DET_WIDTH 1 /* DRC_SIG_DET */ +#define WM5100_DRC_KNEE2_OP_ENA 0x0020 /* DRC_KNEE2_OP_ENA */ +#define WM5100_DRC_KNEE2_OP_ENA_MASK 0x0020 /* DRC_KNEE2_OP_ENA */ +#define WM5100_DRC_KNEE2_OP_ENA_SHIFT 5 /* DRC_KNEE2_OP_ENA */ +#define WM5100_DRC_KNEE2_OP_ENA_WIDTH 1 /* DRC_KNEE2_OP_ENA */ +#define WM5100_DRC_QR 0x0010 /* DRC_QR */ +#define WM5100_DRC_QR_MASK 0x0010 /* DRC_QR */ +#define WM5100_DRC_QR_SHIFT 4 /* DRC_QR */ +#define WM5100_DRC_QR_WIDTH 1 /* DRC_QR */ +#define WM5100_DRC_ANTICLIP 0x0008 /* DRC_ANTICLIP */ +#define WM5100_DRC_ANTICLIP_MASK 0x0008 /* DRC_ANTICLIP */ +#define WM5100_DRC_ANTICLIP_SHIFT 3 /* DRC_ANTICLIP */ +#define WM5100_DRC_ANTICLIP_WIDTH 1 /* DRC_ANTICLIP */ +#define WM5100_DRCL_ENA 0x0002 /* DRCL_ENA */ +#define WM5100_DRCL_ENA_MASK 0x0002 /* DRCL_ENA */ +#define WM5100_DRCL_ENA_SHIFT 1 /* DRCL_ENA */ +#define WM5100_DRCL_ENA_WIDTH 1 /* DRCL_ENA */ +#define WM5100_DRCR_ENA 0x0001 /* DRCR_ENA */ +#define WM5100_DRCR_ENA_MASK 0x0001 /* DRCR_ENA */ +#define WM5100_DRCR_ENA_SHIFT 0 /* DRCR_ENA */ +#define WM5100_DRCR_ENA_WIDTH 1 /* DRCR_ENA */ + +/* + * R3713 (0xE81) - DRC1 ctrl2 + */ +#define WM5100_DRC_ATK_MASK 0x1E00 /* DRC_ATK - [12:9] */ +#define WM5100_DRC_ATK_SHIFT 9 /* DRC_ATK - [12:9] */ +#define WM5100_DRC_ATK_WIDTH 4 /* DRC_ATK - [12:9] */ +#define WM5100_DRC_DCY_MASK 0x01E0 /* DRC_DCY - [8:5] */ +#define WM5100_DRC_DCY_SHIFT 5 /* DRC_DCY - [8:5] */ +#define WM5100_DRC_DCY_WIDTH 4 /* DRC_DCY - [8:5] */ +#define WM5100_DRC_MINGAIN_MASK 0x001C /* DRC_MINGAIN - [4:2] */ +#define WM5100_DRC_MINGAIN_SHIFT 2 /* DRC_MINGAIN - [4:2] */ +#define WM5100_DRC_MINGAIN_WIDTH 3 /* DRC_MINGAIN - [4:2] */ +#define WM5100_DRC_MAXGAIN_MASK 0x0003 /* DRC_MAXGAIN - [1:0] */ +#define WM5100_DRC_MAXGAIN_SHIFT 0 /* DRC_MAXGAIN - [1:0] */ +#define WM5100_DRC_MAXGAIN_WIDTH 2 /* DRC_MAXGAIN - [1:0] */ + +/* + * R3714 (0xE82) - DRC1 ctrl3 + */ +#define WM5100_DRC_NG_MINGAIN_MASK 0xF000 /* DRC_NG_MINGAIN - [15:12] */ +#define WM5100_DRC_NG_MINGAIN_SHIFT 12 /* DRC_NG_MINGAIN - [15:12] */ +#define WM5100_DRC_NG_MINGAIN_WIDTH 4 /* DRC_NG_MINGAIN - [15:12] */ +#define WM5100_DRC_NG_EXP_MASK 0x0C00 /* DRC_NG_EXP - [11:10] */ +#define WM5100_DRC_NG_EXP_SHIFT 10 /* DRC_NG_EXP - [11:10] */ +#define WM5100_DRC_NG_EXP_WIDTH 2 /* DRC_NG_EXP - [11:10] */ +#define WM5100_DRC_QR_THR_MASK 0x0300 /* DRC_QR_THR - [9:8] */ +#define WM5100_DRC_QR_THR_SHIFT 8 /* DRC_QR_THR - [9:8] */ +#define WM5100_DRC_QR_THR_WIDTH 2 /* DRC_QR_THR - [9:8] */ +#define WM5100_DRC_QR_DCY_MASK 0x00C0 /* DRC_QR_DCY - [7:6] */ +#define WM5100_DRC_QR_DCY_SHIFT 6 /* DRC_QR_DCY - [7:6] */ +#define WM5100_DRC_QR_DCY_WIDTH 2 /* DRC_QR_DCY - [7:6] */ +#define WM5100_DRC_HI_COMP_MASK 0x0038 /* DRC_HI_COMP - [5:3] */ +#define WM5100_DRC_HI_COMP_SHIFT 3 /* DRC_HI_COMP - [5:3] */ +#define WM5100_DRC_HI_COMP_WIDTH 3 /* DRC_HI_COMP - [5:3] */ +#define WM5100_DRC_LO_COMP_MASK 0x0007 /* DRC_LO_COMP - [2:0] */ +#define WM5100_DRC_LO_COMP_SHIFT 0 /* DRC_LO_COMP - [2:0] */ +#define WM5100_DRC_LO_COMP_WIDTH 3 /* DRC_LO_COMP - [2:0] */ + +/* + * R3715 (0xE83) - DRC1 ctrl4 + */ +#define WM5100_DRC_KNEE_IP_MASK 0x07E0 /* DRC_KNEE_IP - [10:5] */ +#define WM5100_DRC_KNEE_IP_SHIFT 5 /* DRC_KNEE_IP - [10:5] */ +#define WM5100_DRC_KNEE_IP_WIDTH 6 /* DRC_KNEE_IP - [10:5] */ +#define WM5100_DRC_KNEE_OP_MASK 0x001F /* DRC_KNEE_OP - [4:0] */ +#define WM5100_DRC_KNEE_OP_SHIFT 0 /* DRC_KNEE_OP - [4:0] */ +#define WM5100_DRC_KNEE_OP_WIDTH 5 /* DRC_KNEE_OP - [4:0] */ + +/* + * R3716 (0xE84) - DRC1 ctrl5 + */ +#define WM5100_DRC_KNEE2_IP_MASK 0x03E0 /* DRC_KNEE2_IP - [9:5] */ +#define WM5100_DRC_KNEE2_IP_SHIFT 5 /* DRC_KNEE2_IP - [9:5] */ +#define WM5100_DRC_KNEE2_IP_WIDTH 5 /* DRC_KNEE2_IP - [9:5] */ +#define WM5100_DRC_KNEE2_OP_MASK 0x001F /* DRC_KNEE2_OP - [4:0] */ +#define WM5100_DRC_KNEE2_OP_SHIFT 0 /* DRC_KNEE2_OP - [4:0] */ +#define WM5100_DRC_KNEE2_OP_WIDTH 5 /* DRC_KNEE2_OP - [4:0] */ + +/* + * R3776 (0xEC0) - HPLPF1_1 + */ +#define WM5100_LHPF1_MODE 0x0002 /* LHPF1_MODE */ +#define WM5100_LHPF1_MODE_MASK 0x0002 /* LHPF1_MODE */ +#define WM5100_LHPF1_MODE_SHIFT 1 /* LHPF1_MODE */ +#define WM5100_LHPF1_MODE_WIDTH 1 /* LHPF1_MODE */ +#define WM5100_LHPF1_ENA 0x0001 /* LHPF1_ENA */ +#define WM5100_LHPF1_ENA_MASK 0x0001 /* LHPF1_ENA */ +#define WM5100_LHPF1_ENA_SHIFT 0 /* LHPF1_ENA */ +#define WM5100_LHPF1_ENA_WIDTH 1 /* LHPF1_ENA */ + +/* + * R3777 (0xEC1) - HPLPF1_2 + */ +#define WM5100_LHPF1_COEFF_MASK 0xFFFF /* LHPF1_COEFF - [15:0] */ +#define WM5100_LHPF1_COEFF_SHIFT 0 /* LHPF1_COEFF - [15:0] */ +#define WM5100_LHPF1_COEFF_WIDTH 16 /* LHPF1_COEFF - [15:0] */ + +/* + * R3780 (0xEC4) - HPLPF2_1 + */ +#define WM5100_LHPF2_MODE 0x0002 /* LHPF2_MODE */ +#define WM5100_LHPF2_MODE_MASK 0x0002 /* LHPF2_MODE */ +#define WM5100_LHPF2_MODE_SHIFT 1 /* LHPF2_MODE */ +#define WM5100_LHPF2_MODE_WIDTH 1 /* LHPF2_MODE */ +#define WM5100_LHPF2_ENA 0x0001 /* LHPF2_ENA */ +#define WM5100_LHPF2_ENA_MASK 0x0001 /* LHPF2_ENA */ +#define WM5100_LHPF2_ENA_SHIFT 0 /* LHPF2_ENA */ +#define WM5100_LHPF2_ENA_WIDTH 1 /* LHPF2_ENA */ + +/* + * R3781 (0xEC5) - HPLPF2_2 + */ +#define WM5100_LHPF2_COEFF_MASK 0xFFFF /* LHPF2_COEFF - [15:0] */ +#define WM5100_LHPF2_COEFF_SHIFT 0 /* LHPF2_COEFF - [15:0] */ +#define WM5100_LHPF2_COEFF_WIDTH 16 /* LHPF2_COEFF - [15:0] */ + +/* + * R3784 (0xEC8) - HPLPF3_1 + */ +#define WM5100_LHPF3_MODE 0x0002 /* LHPF3_MODE */ +#define WM5100_LHPF3_MODE_MASK 0x0002 /* LHPF3_MODE */ +#define WM5100_LHPF3_MODE_SHIFT 1 /* LHPF3_MODE */ +#define WM5100_LHPF3_MODE_WIDTH 1 /* LHPF3_MODE */ +#define WM5100_LHPF3_ENA 0x0001 /* LHPF3_ENA */ +#define WM5100_LHPF3_ENA_MASK 0x0001 /* LHPF3_ENA */ +#define WM5100_LHPF3_ENA_SHIFT 0 /* LHPF3_ENA */ +#define WM5100_LHPF3_ENA_WIDTH 1 /* LHPF3_ENA */ + +/* + * R3785 (0xEC9) - HPLPF3_2 + */ +#define WM5100_LHPF3_COEFF_MASK 0xFFFF /* LHPF3_COEFF - [15:0] */ +#define WM5100_LHPF3_COEFF_SHIFT 0 /* LHPF3_COEFF - [15:0] */ +#define WM5100_LHPF3_COEFF_WIDTH 16 /* LHPF3_COEFF - [15:0] */ + +/* + * R3788 (0xECC) - HPLPF4_1 + */ +#define WM5100_LHPF4_MODE 0x0002 /* LHPF4_MODE */ +#define WM5100_LHPF4_MODE_MASK 0x0002 /* LHPF4_MODE */ +#define WM5100_LHPF4_MODE_SHIFT 1 /* LHPF4_MODE */ +#define WM5100_LHPF4_MODE_WIDTH 1 /* LHPF4_MODE */ +#define WM5100_LHPF4_ENA 0x0001 /* LHPF4_ENA */ +#define WM5100_LHPF4_ENA_MASK 0x0001 /* LHPF4_ENA */ +#define WM5100_LHPF4_ENA_SHIFT 0 /* LHPF4_ENA */ +#define WM5100_LHPF4_ENA_WIDTH 1 /* LHPF4_ENA */ + +/* + * R3789 (0xECD) - HPLPF4_2 + */ +#define WM5100_LHPF4_COEFF_MASK 0xFFFF /* LHPF4_COEFF - [15:0] */ +#define WM5100_LHPF4_COEFF_SHIFT 0 /* LHPF4_COEFF - [15:0] */ +#define WM5100_LHPF4_COEFF_WIDTH 16 /* LHPF4_COEFF - [15:0] */ + +/* + * R16384 (0x4000) - DSP1 DM 0 + */ +#define WM5100_DSP1_DM_START_1_MASK 0x00FF /* DSP1_DM_START - [7:0] */ +#define WM5100_DSP1_DM_START_1_SHIFT 0 /* DSP1_DM_START - [7:0] */ +#define WM5100_DSP1_DM_START_1_WIDTH 8 /* DSP1_DM_START - [7:0] */ + +/* + * R16385 (0x4001) - DSP1 DM 1 + */ +#define WM5100_DSP1_DM_START_MASK 0xFFFF /* DSP1_DM_START - [15:0] */ +#define WM5100_DSP1_DM_START_SHIFT 0 /* DSP1_DM_START - [15:0] */ +#define WM5100_DSP1_DM_START_WIDTH 16 /* DSP1_DM_START - [15:0] */ + +/* + * R16386 (0x4002) - DSP1 DM 2 + */ +#define WM5100_DSP1_DM_1_1_MASK 0x00FF /* DSP1_DM_1 - [7:0] */ +#define WM5100_DSP1_DM_1_1_SHIFT 0 /* DSP1_DM_1 - [7:0] */ +#define WM5100_DSP1_DM_1_1_WIDTH 8 /* DSP1_DM_1 - [7:0] */ + +/* + * R16387 (0x4003) - DSP1 DM 3 + */ +#define WM5100_DSP1_DM_1_MASK 0xFFFF /* DSP1_DM_1 - [15:0] */ +#define WM5100_DSP1_DM_1_SHIFT 0 /* DSP1_DM_1 - [15:0] */ +#define WM5100_DSP1_DM_1_WIDTH 16 /* DSP1_DM_1 - [15:0] */ + +/* + * R16892 (0x41FC) - DSP1 DM 508 + */ +#define WM5100_DSP1_DM_254_1_MASK 0x00FF /* DSP1_DM_254 - [7:0] */ +#define WM5100_DSP1_DM_254_1_SHIFT 0 /* DSP1_DM_254 - [7:0] */ +#define WM5100_DSP1_DM_254_1_WIDTH 8 /* DSP1_DM_254 - [7:0] */ + +/* + * R16893 (0x41FD) - DSP1 DM 509 + */ +#define WM5100_DSP1_DM_254_MASK 0xFFFF /* DSP1_DM_254 - [15:0] */ +#define WM5100_DSP1_DM_254_SHIFT 0 /* DSP1_DM_254 - [15:0] */ +#define WM5100_DSP1_DM_254_WIDTH 16 /* DSP1_DM_254 - [15:0] */ + +/* + * R16894 (0x41FE) - DSP1 DM 510 + */ +#define WM5100_DSP1_DM_END_1_MASK 0x00FF /* DSP1_DM_END - [7:0] */ +#define WM5100_DSP1_DM_END_1_SHIFT 0 /* DSP1_DM_END - [7:0] */ +#define WM5100_DSP1_DM_END_1_WIDTH 8 /* DSP1_DM_END - [7:0] */ + +/* + * R16895 (0x41FF) - DSP1 DM 511 + */ +#define WM5100_DSP1_DM_END_MASK 0xFFFF /* DSP1_DM_END - [15:0] */ +#define WM5100_DSP1_DM_END_SHIFT 0 /* DSP1_DM_END - [15:0] */ +#define WM5100_DSP1_DM_END_WIDTH 16 /* DSP1_DM_END - [15:0] */ + +/* + * R18432 (0x4800) - DSP1 PM 0 + */ +#define WM5100_DSP1_PM_START_2_MASK 0x00FF /* DSP1_PM_START - [7:0] */ +#define WM5100_DSP1_PM_START_2_SHIFT 0 /* DSP1_PM_START - [7:0] */ +#define WM5100_DSP1_PM_START_2_WIDTH 8 /* DSP1_PM_START - [7:0] */ + +/* + * R18433 (0x4801) - DSP1 PM 1 + */ +#define WM5100_DSP1_PM_START_1_MASK 0xFFFF /* DSP1_PM_START - [15:0] */ +#define WM5100_DSP1_PM_START_1_SHIFT 0 /* DSP1_PM_START - [15:0] */ +#define WM5100_DSP1_PM_START_1_WIDTH 16 /* DSP1_PM_START - [15:0] */ + +/* + * R18434 (0x4802) - DSP1 PM 2 + */ +#define WM5100_DSP1_PM_START_MASK 0xFFFF /* DSP1_PM_START - [15:0] */ +#define WM5100_DSP1_PM_START_SHIFT 0 /* DSP1_PM_START - [15:0] */ +#define WM5100_DSP1_PM_START_WIDTH 16 /* DSP1_PM_START - [15:0] */ + +/* + * R18435 (0x4803) - DSP1 PM 3 + */ +#define WM5100_DSP1_PM_1_2_MASK 0x00FF /* DSP1_PM_1 - [7:0] */ +#define WM5100_DSP1_PM_1_2_SHIFT 0 /* DSP1_PM_1 - [7:0] */ +#define WM5100_DSP1_PM_1_2_WIDTH 8 /* DSP1_PM_1 - [7:0] */ + +/* + * R18436 (0x4804) - DSP1 PM 4 + */ +#define WM5100_DSP1_PM_1_1_MASK 0xFFFF /* DSP1_PM_1 - [15:0] */ +#define WM5100_DSP1_PM_1_1_SHIFT 0 /* DSP1_PM_1 - [15:0] */ +#define WM5100_DSP1_PM_1_1_WIDTH 16 /* DSP1_PM_1 - [15:0] */ + +/* + * R18437 (0x4805) - DSP1 PM 5 + */ +#define WM5100_DSP1_PM_1_MASK 0xFFFF /* DSP1_PM_1 - [15:0] */ +#define WM5100_DSP1_PM_1_SHIFT 0 /* DSP1_PM_1 - [15:0] */ +#define WM5100_DSP1_PM_1_WIDTH 16 /* DSP1_PM_1 - [15:0] */ + +/* + * R19962 (0x4DFA) - DSP1 PM 1530 + */ +#define WM5100_DSP1_PM_510_2_MASK 0x00FF /* DSP1_PM_510 - [7:0] */ +#define WM5100_DSP1_PM_510_2_SHIFT 0 /* DSP1_PM_510 - [7:0] */ +#define WM5100_DSP1_PM_510_2_WIDTH 8 /* DSP1_PM_510 - [7:0] */ + +/* + * R19963 (0x4DFB) - DSP1 PM 1531 + */ +#define WM5100_DSP1_PM_510_1_MASK 0xFFFF /* DSP1_PM_510 - [15:0] */ +#define WM5100_DSP1_PM_510_1_SHIFT 0 /* DSP1_PM_510 - [15:0] */ +#define WM5100_DSP1_PM_510_1_WIDTH 16 /* DSP1_PM_510 - [15:0] */ + +/* + * R19964 (0x4DFC) - DSP1 PM 1532 + */ +#define WM5100_DSP1_PM_510_MASK 0xFFFF /* DSP1_PM_510 - [15:0] */ +#define WM5100_DSP1_PM_510_SHIFT 0 /* DSP1_PM_510 - [15:0] */ +#define WM5100_DSP1_PM_510_WIDTH 16 /* DSP1_PM_510 - [15:0] */ + +/* + * R19965 (0x4DFD) - DSP1 PM 1533 + */ +#define WM5100_DSP1_PM_END_2_MASK 0x00FF /* DSP1_PM_END - [7:0] */ +#define WM5100_DSP1_PM_END_2_SHIFT 0 /* DSP1_PM_END - [7:0] */ +#define WM5100_DSP1_PM_END_2_WIDTH 8 /* DSP1_PM_END - [7:0] */ + +/* + * R19966 (0x4DFE) - DSP1 PM 1534 + */ +#define WM5100_DSP1_PM_END_1_MASK 0xFFFF /* DSP1_PM_END - [15:0] */ +#define WM5100_DSP1_PM_END_1_SHIFT 0 /* DSP1_PM_END - [15:0] */ +#define WM5100_DSP1_PM_END_1_WIDTH 16 /* DSP1_PM_END - [15:0] */ + +/* + * R19967 (0x4DFF) - DSP1 PM 1535 + */ +#define WM5100_DSP1_PM_END_MASK 0xFFFF /* DSP1_PM_END - [15:0] */ +#define WM5100_DSP1_PM_END_SHIFT 0 /* DSP1_PM_END - [15:0] */ +#define WM5100_DSP1_PM_END_WIDTH 16 /* DSP1_PM_END - [15:0] */ + +/* + * R20480 (0x5000) - DSP1 ZM 0 + */ +#define WM5100_DSP1_ZM_START_1_MASK 0x00FF /* DSP1_ZM_START - [7:0] */ +#define WM5100_DSP1_ZM_START_1_SHIFT 0 /* DSP1_ZM_START - [7:0] */ +#define WM5100_DSP1_ZM_START_1_WIDTH 8 /* DSP1_ZM_START - [7:0] */ + +/* + * R20481 (0x5001) - DSP1 ZM 1 + */ +#define WM5100_DSP1_ZM_START_MASK 0xFFFF /* DSP1_ZM_START - [15:0] */ +#define WM5100_DSP1_ZM_START_SHIFT 0 /* DSP1_ZM_START - [15:0] */ +#define WM5100_DSP1_ZM_START_WIDTH 16 /* DSP1_ZM_START - [15:0] */ + +/* + * R20482 (0x5002) - DSP1 ZM 2 + */ +#define WM5100_DSP1_ZM_1_1_MASK 0x00FF /* DSP1_ZM_1 - [7:0] */ +#define WM5100_DSP1_ZM_1_1_SHIFT 0 /* DSP1_ZM_1 - [7:0] */ +#define WM5100_DSP1_ZM_1_1_WIDTH 8 /* DSP1_ZM_1 - [7:0] */ + +/* + * R20483 (0x5003) - DSP1 ZM 3 + */ +#define WM5100_DSP1_ZM_1_MASK 0xFFFF /* DSP1_ZM_1 - [15:0] */ +#define WM5100_DSP1_ZM_1_SHIFT 0 /* DSP1_ZM_1 - [15:0] */ +#define WM5100_DSP1_ZM_1_WIDTH 16 /* DSP1_ZM_1 - [15:0] */ + +/* + * R22524 (0x57FC) - DSP1 ZM 2044 + */ +#define WM5100_DSP1_ZM_1022_1_MASK 0x00FF /* DSP1_ZM_1022 - [7:0] */ +#define WM5100_DSP1_ZM_1022_1_SHIFT 0 /* DSP1_ZM_1022 - [7:0] */ +#define WM5100_DSP1_ZM_1022_1_WIDTH 8 /* DSP1_ZM_1022 - [7:0] */ + +/* + * R22525 (0x57FD) - DSP1 ZM 2045 + */ +#define WM5100_DSP1_ZM_1022_MASK 0xFFFF /* DSP1_ZM_1022 - [15:0] */ +#define WM5100_DSP1_ZM_1022_SHIFT 0 /* DSP1_ZM_1022 - [15:0] */ +#define WM5100_DSP1_ZM_1022_WIDTH 16 /* DSP1_ZM_1022 - [15:0] */ + +/* + * R22526 (0x57FE) - DSP1 ZM 2046 + */ +#define WM5100_DSP1_ZM_END_1_MASK 0x00FF /* DSP1_ZM_END - [7:0] */ +#define WM5100_DSP1_ZM_END_1_SHIFT 0 /* DSP1_ZM_END - [7:0] */ +#define WM5100_DSP1_ZM_END_1_WIDTH 8 /* DSP1_ZM_END - [7:0] */ + +/* + * R22527 (0x57FF) - DSP1 ZM 2047 + */ +#define WM5100_DSP1_ZM_END_MASK 0xFFFF /* DSP1_ZM_END - [15:0] */ +#define WM5100_DSP1_ZM_END_SHIFT 0 /* DSP1_ZM_END - [15:0] */ +#define WM5100_DSP1_ZM_END_WIDTH 16 /* DSP1_ZM_END - [15:0] */ + +/* + * R24576 (0x6000) - DSP2 DM 0 + */ +#define WM5100_DSP2_DM_START_1_MASK 0x00FF /* DSP2_DM_START - [7:0] */ +#define WM5100_DSP2_DM_START_1_SHIFT 0 /* DSP2_DM_START - [7:0] */ +#define WM5100_DSP2_DM_START_1_WIDTH 8 /* DSP2_DM_START - [7:0] */ + +/* + * R24577 (0x6001) - DSP2 DM 1 + */ +#define WM5100_DSP2_DM_START_MASK 0xFFFF /* DSP2_DM_START - [15:0] */ +#define WM5100_DSP2_DM_START_SHIFT 0 /* DSP2_DM_START - [15:0] */ +#define WM5100_DSP2_DM_START_WIDTH 16 /* DSP2_DM_START - [15:0] */ + +/* + * R24578 (0x6002) - DSP2 DM 2 + */ +#define WM5100_DSP2_DM_1_1_MASK 0x00FF /* DSP2_DM_1 - [7:0] */ +#define WM5100_DSP2_DM_1_1_SHIFT 0 /* DSP2_DM_1 - [7:0] */ +#define WM5100_DSP2_DM_1_1_WIDTH 8 /* DSP2_DM_1 - [7:0] */ + +/* + * R24579 (0x6003) - DSP2 DM 3 + */ +#define WM5100_DSP2_DM_1_MASK 0xFFFF /* DSP2_DM_1 - [15:0] */ +#define WM5100_DSP2_DM_1_SHIFT 0 /* DSP2_DM_1 - [15:0] */ +#define WM5100_DSP2_DM_1_WIDTH 16 /* DSP2_DM_1 - [15:0] */ + +/* + * R25084 (0x61FC) - DSP2 DM 508 + */ +#define WM5100_DSP2_DM_254_1_MASK 0x00FF /* DSP2_DM_254 - [7:0] */ +#define WM5100_DSP2_DM_254_1_SHIFT 0 /* DSP2_DM_254 - [7:0] */ +#define WM5100_DSP2_DM_254_1_WIDTH 8 /* DSP2_DM_254 - [7:0] */ + +/* + * R25085 (0x61FD) - DSP2 DM 509 + */ +#define WM5100_DSP2_DM_254_MASK 0xFFFF /* DSP2_DM_254 - [15:0] */ +#define WM5100_DSP2_DM_254_SHIFT 0 /* DSP2_DM_254 - [15:0] */ +#define WM5100_DSP2_DM_254_WIDTH 16 /* DSP2_DM_254 - [15:0] */ + +/* + * R25086 (0x61FE) - DSP2 DM 510 + */ +#define WM5100_DSP2_DM_END_1_MASK 0x00FF /* DSP2_DM_END - [7:0] */ +#define WM5100_DSP2_DM_END_1_SHIFT 0 /* DSP2_DM_END - [7:0] */ +#define WM5100_DSP2_DM_END_1_WIDTH 8 /* DSP2_DM_END - [7:0] */ + +/* + * R25087 (0x61FF) - DSP2 DM 511 + */ +#define WM5100_DSP2_DM_END_MASK 0xFFFF /* DSP2_DM_END - [15:0] */ +#define WM5100_DSP2_DM_END_SHIFT 0 /* DSP2_DM_END - [15:0] */ +#define WM5100_DSP2_DM_END_WIDTH 16 /* DSP2_DM_END - [15:0] */ + +/* + * R26624 (0x6800) - DSP2 PM 0 + */ +#define WM5100_DSP2_PM_START_2_MASK 0x00FF /* DSP2_PM_START - [7:0] */ +#define WM5100_DSP2_PM_START_2_SHIFT 0 /* DSP2_PM_START - [7:0] */ +#define WM5100_DSP2_PM_START_2_WIDTH 8 /* DSP2_PM_START - [7:0] */ + +/* + * R26625 (0x6801) - DSP2 PM 1 + */ +#define WM5100_DSP2_PM_START_1_MASK 0xFFFF /* DSP2_PM_START - [15:0] */ +#define WM5100_DSP2_PM_START_1_SHIFT 0 /* DSP2_PM_START - [15:0] */ +#define WM5100_DSP2_PM_START_1_WIDTH 16 /* DSP2_PM_START - [15:0] */ + +/* + * R26626 (0x6802) - DSP2 PM 2 + */ +#define WM5100_DSP2_PM_START_MASK 0xFFFF /* DSP2_PM_START - [15:0] */ +#define WM5100_DSP2_PM_START_SHIFT 0 /* DSP2_PM_START - [15:0] */ +#define WM5100_DSP2_PM_START_WIDTH 16 /* DSP2_PM_START - [15:0] */ + +/* + * R26627 (0x6803) - DSP2 PM 3 + */ +#define WM5100_DSP2_PM_1_2_MASK 0x00FF /* DSP2_PM_1 - [7:0] */ +#define WM5100_DSP2_PM_1_2_SHIFT 0 /* DSP2_PM_1 - [7:0] */ +#define WM5100_DSP2_PM_1_2_WIDTH 8 /* DSP2_PM_1 - [7:0] */ + +/* + * R26628 (0x6804) - DSP2 PM 4 + */ +#define WM5100_DSP2_PM_1_1_MASK 0xFFFF /* DSP2_PM_1 - [15:0] */ +#define WM5100_DSP2_PM_1_1_SHIFT 0 /* DSP2_PM_1 - [15:0] */ +#define WM5100_DSP2_PM_1_1_WIDTH 16 /* DSP2_PM_1 - [15:0] */ + +/* + * R26629 (0x6805) - DSP2 PM 5 + */ +#define WM5100_DSP2_PM_1_MASK 0xFFFF /* DSP2_PM_1 - [15:0] */ +#define WM5100_DSP2_PM_1_SHIFT 0 /* DSP2_PM_1 - [15:0] */ +#define WM5100_DSP2_PM_1_WIDTH 16 /* DSP2_PM_1 - [15:0] */ + +/* + * R28154 (0x6DFA) - DSP2 PM 1530 + */ +#define WM5100_DSP2_PM_510_2_MASK 0x00FF /* DSP2_PM_510 - [7:0] */ +#define WM5100_DSP2_PM_510_2_SHIFT 0 /* DSP2_PM_510 - [7:0] */ +#define WM5100_DSP2_PM_510_2_WIDTH 8 /* DSP2_PM_510 - [7:0] */ + +/* + * R28155 (0x6DFB) - DSP2 PM 1531 + */ +#define WM5100_DSP2_PM_510_1_MASK 0xFFFF /* DSP2_PM_510 - [15:0] */ +#define WM5100_DSP2_PM_510_1_SHIFT 0 /* DSP2_PM_510 - [15:0] */ +#define WM5100_DSP2_PM_510_1_WIDTH 16 /* DSP2_PM_510 - [15:0] */ + +/* + * R28156 (0x6DFC) - DSP2 PM 1532 + */ +#define WM5100_DSP2_PM_510_MASK 0xFFFF /* DSP2_PM_510 - [15:0] */ +#define WM5100_DSP2_PM_510_SHIFT 0 /* DSP2_PM_510 - [15:0] */ +#define WM5100_DSP2_PM_510_WIDTH 16 /* DSP2_PM_510 - [15:0] */ + +/* + * R28157 (0x6DFD) - DSP2 PM 1533 + */ +#define WM5100_DSP2_PM_END_2_MASK 0x00FF /* DSP2_PM_END - [7:0] */ +#define WM5100_DSP2_PM_END_2_SHIFT 0 /* DSP2_PM_END - [7:0] */ +#define WM5100_DSP2_PM_END_2_WIDTH 8 /* DSP2_PM_END - [7:0] */ + +/* + * R28158 (0x6DFE) - DSP2 PM 1534 + */ +#define WM5100_DSP2_PM_END_1_MASK 0xFFFF /* DSP2_PM_END - [15:0] */ +#define WM5100_DSP2_PM_END_1_SHIFT 0 /* DSP2_PM_END - [15:0] */ +#define WM5100_DSP2_PM_END_1_WIDTH 16 /* DSP2_PM_END - [15:0] */ + +/* + * R28159 (0x6DFF) - DSP2 PM 1535 + */ +#define WM5100_DSP2_PM_END_MASK 0xFFFF /* DSP2_PM_END - [15:0] */ +#define WM5100_DSP2_PM_END_SHIFT 0 /* DSP2_PM_END - [15:0] */ +#define WM5100_DSP2_PM_END_WIDTH 16 /* DSP2_PM_END - [15:0] */ + +/* + * R28672 (0x7000) - DSP2 ZM 0 + */ +#define WM5100_DSP2_ZM_START_1_MASK 0x00FF /* DSP2_ZM_START - [7:0] */ +#define WM5100_DSP2_ZM_START_1_SHIFT 0 /* DSP2_ZM_START - [7:0] */ +#define WM5100_DSP2_ZM_START_1_WIDTH 8 /* DSP2_ZM_START - [7:0] */ + +/* + * R28673 (0x7001) - DSP2 ZM 1 + */ +#define WM5100_DSP2_ZM_START_MASK 0xFFFF /* DSP2_ZM_START - [15:0] */ +#define WM5100_DSP2_ZM_START_SHIFT 0 /* DSP2_ZM_START - [15:0] */ +#define WM5100_DSP2_ZM_START_WIDTH 16 /* DSP2_ZM_START - [15:0] */ + +/* + * R28674 (0x7002) - DSP2 ZM 2 + */ +#define WM5100_DSP2_ZM_1_1_MASK 0x00FF /* DSP2_ZM_1 - [7:0] */ +#define WM5100_DSP2_ZM_1_1_SHIFT 0 /* DSP2_ZM_1 - [7:0] */ +#define WM5100_DSP2_ZM_1_1_WIDTH 8 /* DSP2_ZM_1 - [7:0] */ + +/* + * R28675 (0x7003) - DSP2 ZM 3 + */ +#define WM5100_DSP2_ZM_1_MASK 0xFFFF /* DSP2_ZM_1 - [15:0] */ +#define WM5100_DSP2_ZM_1_SHIFT 0 /* DSP2_ZM_1 - [15:0] */ +#define WM5100_DSP2_ZM_1_WIDTH 16 /* DSP2_ZM_1 - [15:0] */ + +/* + * R30716 (0x77FC) - DSP2 ZM 2044 + */ +#define WM5100_DSP2_ZM_1022_1_MASK 0x00FF /* DSP2_ZM_1022 - [7:0] */ +#define WM5100_DSP2_ZM_1022_1_SHIFT 0 /* DSP2_ZM_1022 - [7:0] */ +#define WM5100_DSP2_ZM_1022_1_WIDTH 8 /* DSP2_ZM_1022 - [7:0] */ + +/* + * R30717 (0x77FD) - DSP2 ZM 2045 + */ +#define WM5100_DSP2_ZM_1022_MASK 0xFFFF /* DSP2_ZM_1022 - [15:0] */ +#define WM5100_DSP2_ZM_1022_SHIFT 0 /* DSP2_ZM_1022 - [15:0] */ +#define WM5100_DSP2_ZM_1022_WIDTH 16 /* DSP2_ZM_1022 - [15:0] */ + +/* + * R30718 (0x77FE) - DSP2 ZM 2046 + */ +#define WM5100_DSP2_ZM_END_1_MASK 0x00FF /* DSP2_ZM_END - [7:0] */ +#define WM5100_DSP2_ZM_END_1_SHIFT 0 /* DSP2_ZM_END - [7:0] */ +#define WM5100_DSP2_ZM_END_1_WIDTH 8 /* DSP2_ZM_END - [7:0] */ + +/* + * R30719 (0x77FF) - DSP2 ZM 2047 + */ +#define WM5100_DSP2_ZM_END_MASK 0xFFFF /* DSP2_ZM_END - [15:0] */ +#define WM5100_DSP2_ZM_END_SHIFT 0 /* DSP2_ZM_END - [15:0] */ +#define WM5100_DSP2_ZM_END_WIDTH 16 /* DSP2_ZM_END - [15:0] */ + +/* + * R32768 (0x8000) - DSP3 DM 0 + */ +#define WM5100_DSP3_DM_START_1_MASK 0x00FF /* DSP3_DM_START - [7:0] */ +#define WM5100_DSP3_DM_START_1_SHIFT 0 /* DSP3_DM_START - [7:0] */ +#define WM5100_DSP3_DM_START_1_WIDTH 8 /* DSP3_DM_START - [7:0] */ + +/* + * R32769 (0x8001) - DSP3 DM 1 + */ +#define WM5100_DSP3_DM_START_MASK 0xFFFF /* DSP3_DM_START - [15:0] */ +#define WM5100_DSP3_DM_START_SHIFT 0 /* DSP3_DM_START - [15:0] */ +#define WM5100_DSP3_DM_START_WIDTH 16 /* DSP3_DM_START - [15:0] */ + +/* + * R32770 (0x8002) - DSP3 DM 2 + */ +#define WM5100_DSP3_DM_1_1_MASK 0x00FF /* DSP3_DM_1 - [7:0] */ +#define WM5100_DSP3_DM_1_1_SHIFT 0 /* DSP3_DM_1 - [7:0] */ +#define WM5100_DSP3_DM_1_1_WIDTH 8 /* DSP3_DM_1 - [7:0] */ + +/* + * R32771 (0x8003) - DSP3 DM 3 + */ +#define WM5100_DSP3_DM_1_MASK 0xFFFF /* DSP3_DM_1 - [15:0] */ +#define WM5100_DSP3_DM_1_SHIFT 0 /* DSP3_DM_1 - [15:0] */ +#define WM5100_DSP3_DM_1_WIDTH 16 /* DSP3_DM_1 - [15:0] */ + +/* + * R33276 (0x81FC) - DSP3 DM 508 + */ +#define WM5100_DSP3_DM_254_1_MASK 0x00FF /* DSP3_DM_254 - [7:0] */ +#define WM5100_DSP3_DM_254_1_SHIFT 0 /* DSP3_DM_254 - [7:0] */ +#define WM5100_DSP3_DM_254_1_WIDTH 8 /* DSP3_DM_254 - [7:0] */ + +/* + * R33277 (0x81FD) - DSP3 DM 509 + */ +#define WM5100_DSP3_DM_254_MASK 0xFFFF /* DSP3_DM_254 - [15:0] */ +#define WM5100_DSP3_DM_254_SHIFT 0 /* DSP3_DM_254 - [15:0] */ +#define WM5100_DSP3_DM_254_WIDTH 16 /* DSP3_DM_254 - [15:0] */ + +/* + * R33278 (0x81FE) - DSP3 DM 510 + */ +#define WM5100_DSP3_DM_END_1_MASK 0x00FF /* DSP3_DM_END - [7:0] */ +#define WM5100_DSP3_DM_END_1_SHIFT 0 /* DSP3_DM_END - [7:0] */ +#define WM5100_DSP3_DM_END_1_WIDTH 8 /* DSP3_DM_END - [7:0] */ + +/* + * R33279 (0x81FF) - DSP3 DM 511 + */ +#define WM5100_DSP3_DM_END_MASK 0xFFFF /* DSP3_DM_END - [15:0] */ +#define WM5100_DSP3_DM_END_SHIFT 0 /* DSP3_DM_END - [15:0] */ +#define WM5100_DSP3_DM_END_WIDTH 16 /* DSP3_DM_END - [15:0] */ + +/* + * R34816 (0x8800) - DSP3 PM 0 + */ +#define WM5100_DSP3_PM_START_2_MASK 0x00FF /* DSP3_PM_START - [7:0] */ +#define WM5100_DSP3_PM_START_2_SHIFT 0 /* DSP3_PM_START - [7:0] */ +#define WM5100_DSP3_PM_START_2_WIDTH 8 /* DSP3_PM_START - [7:0] */ + +/* + * R34817 (0x8801) - DSP3 PM 1 + */ +#define WM5100_DSP3_PM_START_1_MASK 0xFFFF /* DSP3_PM_START - [15:0] */ +#define WM5100_DSP3_PM_START_1_SHIFT 0 /* DSP3_PM_START - [15:0] */ +#define WM5100_DSP3_PM_START_1_WIDTH 16 /* DSP3_PM_START - [15:0] */ + +/* + * R34818 (0x8802) - DSP3 PM 2 + */ +#define WM5100_DSP3_PM_START_MASK 0xFFFF /* DSP3_PM_START - [15:0] */ +#define WM5100_DSP3_PM_START_SHIFT 0 /* DSP3_PM_START - [15:0] */ +#define WM5100_DSP3_PM_START_WIDTH 16 /* DSP3_PM_START - [15:0] */ + +/* + * R34819 (0x8803) - DSP3 PM 3 + */ +#define WM5100_DSP3_PM_1_2_MASK 0x00FF /* DSP3_PM_1 - [7:0] */ +#define WM5100_DSP3_PM_1_2_SHIFT 0 /* DSP3_PM_1 - [7:0] */ +#define WM5100_DSP3_PM_1_2_WIDTH 8 /* DSP3_PM_1 - [7:0] */ + +/* + * R34820 (0x8804) - DSP3 PM 4 + */ +#define WM5100_DSP3_PM_1_1_MASK 0xFFFF /* DSP3_PM_1 - [15:0] */ +#define WM5100_DSP3_PM_1_1_SHIFT 0 /* DSP3_PM_1 - [15:0] */ +#define WM5100_DSP3_PM_1_1_WIDTH 16 /* DSP3_PM_1 - [15:0] */ + +/* + * R34821 (0x8805) - DSP3 PM 5 + */ +#define WM5100_DSP3_PM_1_MASK 0xFFFF /* DSP3_PM_1 - [15:0] */ +#define WM5100_DSP3_PM_1_SHIFT 0 /* DSP3_PM_1 - [15:0] */ +#define WM5100_DSP3_PM_1_WIDTH 16 /* DSP3_PM_1 - [15:0] */ + +/* + * R36346 (0x8DFA) - DSP3 PM 1530 + */ +#define WM5100_DSP3_PM_510_2_MASK 0x00FF /* DSP3_PM_510 - [7:0] */ +#define WM5100_DSP3_PM_510_2_SHIFT 0 /* DSP3_PM_510 - [7:0] */ +#define WM5100_DSP3_PM_510_2_WIDTH 8 /* DSP3_PM_510 - [7:0] */ + +/* + * R36347 (0x8DFB) - DSP3 PM 1531 + */ +#define WM5100_DSP3_PM_510_1_MASK 0xFFFF /* DSP3_PM_510 - [15:0] */ +#define WM5100_DSP3_PM_510_1_SHIFT 0 /* DSP3_PM_510 - [15:0] */ +#define WM5100_DSP3_PM_510_1_WIDTH 16 /* DSP3_PM_510 - [15:0] */ + +/* + * R36348 (0x8DFC) - DSP3 PM 1532 + */ +#define WM5100_DSP3_PM_510_MASK 0xFFFF /* DSP3_PM_510 - [15:0] */ +#define WM5100_DSP3_PM_510_SHIFT 0 /* DSP3_PM_510 - [15:0] */ +#define WM5100_DSP3_PM_510_WIDTH 16 /* DSP3_PM_510 - [15:0] */ + +/* + * R36349 (0x8DFD) - DSP3 PM 1533 + */ +#define WM5100_DSP3_PM_END_2_MASK 0x00FF /* DSP3_PM_END - [7:0] */ +#define WM5100_DSP3_PM_END_2_SHIFT 0 /* DSP3_PM_END - [7:0] */ +#define WM5100_DSP3_PM_END_2_WIDTH 8 /* DSP3_PM_END - [7:0] */ + +/* + * R36350 (0x8DFE) - DSP3 PM 1534 + */ +#define WM5100_DSP3_PM_END_1_MASK 0xFFFF /* DSP3_PM_END - [15:0] */ +#define WM5100_DSP3_PM_END_1_SHIFT 0 /* DSP3_PM_END - [15:0] */ +#define WM5100_DSP3_PM_END_1_WIDTH 16 /* DSP3_PM_END - [15:0] */ + +/* + * R36351 (0x8DFF) - DSP3 PM 1535 + */ +#define WM5100_DSP3_PM_END_MASK 0xFFFF /* DSP3_PM_END - [15:0] */ +#define WM5100_DSP3_PM_END_SHIFT 0 /* DSP3_PM_END - [15:0] */ +#define WM5100_DSP3_PM_END_WIDTH 16 /* DSP3_PM_END - [15:0] */ + +/* + * R36864 (0x9000) - DSP3 ZM 0 + */ +#define WM5100_DSP3_ZM_START_1_MASK 0x00FF /* DSP3_ZM_START - [7:0] */ +#define WM5100_DSP3_ZM_START_1_SHIFT 0 /* DSP3_ZM_START - [7:0] */ +#define WM5100_DSP3_ZM_START_1_WIDTH 8 /* DSP3_ZM_START - [7:0] */ + +/* + * R36865 (0x9001) - DSP3 ZM 1 + */ +#define WM5100_DSP3_ZM_START_MASK 0xFFFF /* DSP3_ZM_START - [15:0] */ +#define WM5100_DSP3_ZM_START_SHIFT 0 /* DSP3_ZM_START - [15:0] */ +#define WM5100_DSP3_ZM_START_WIDTH 16 /* DSP3_ZM_START - [15:0] */ + +/* + * R36866 (0x9002) - DSP3 ZM 2 + */ +#define WM5100_DSP3_ZM_1_1_MASK 0x00FF /* DSP3_ZM_1 - [7:0] */ +#define WM5100_DSP3_ZM_1_1_SHIFT 0 /* DSP3_ZM_1 - [7:0] */ +#define WM5100_DSP3_ZM_1_1_WIDTH 8 /* DSP3_ZM_1 - [7:0] */ + +/* + * R36867 (0x9003) - DSP3 ZM 3 + */ +#define WM5100_DSP3_ZM_1_MASK 0xFFFF /* DSP3_ZM_1 - [15:0] */ +#define WM5100_DSP3_ZM_1_SHIFT 0 /* DSP3_ZM_1 - [15:0] */ +#define WM5100_DSP3_ZM_1_WIDTH 16 /* DSP3_ZM_1 - [15:0] */ + +/* + * R38908 (0x97FC) - DSP3 ZM 2044 + */ +#define WM5100_DSP3_ZM_1022_1_MASK 0x00FF /* DSP3_ZM_1022 - [7:0] */ +#define WM5100_DSP3_ZM_1022_1_SHIFT 0 /* DSP3_ZM_1022 - [7:0] */ +#define WM5100_DSP3_ZM_1022_1_WIDTH 8 /* DSP3_ZM_1022 - [7:0] */ + +/* + * R38909 (0x97FD) - DSP3 ZM 2045 + */ +#define WM5100_DSP3_ZM_1022_MASK 0xFFFF /* DSP3_ZM_1022 - [15:0] */ +#define WM5100_DSP3_ZM_1022_SHIFT 0 /* DSP3_ZM_1022 - [15:0] */ +#define WM5100_DSP3_ZM_1022_WIDTH 16 /* DSP3_ZM_1022 - [15:0] */ + +/* + * R38910 (0x97FE) - DSP3 ZM 2046 + */ +#define WM5100_DSP3_ZM_END_1_MASK 0x00FF /* DSP3_ZM_END - [7:0] */ +#define WM5100_DSP3_ZM_END_1_SHIFT 0 /* DSP3_ZM_END - [7:0] */ +#define WM5100_DSP3_ZM_END_1_WIDTH 8 /* DSP3_ZM_END - [7:0] */ + +/* + * R38911 (0x97FF) - DSP3 ZM 2047 + */ +#define WM5100_DSP3_ZM_END_MASK 0xFFFF /* DSP3_ZM_END - [15:0] */ +#define WM5100_DSP3_ZM_END_SHIFT 0 /* DSP3_ZM_END - [15:0] */ +#define WM5100_DSP3_ZM_END_WIDTH 16 /* DSP3_ZM_END - [15:0] */ + +int wm5100_readable_register(struct snd_soc_codec *codec, unsigned int reg); +int wm5100_volatile_register(struct snd_soc_codec *codec, unsigned int reg); + +extern u16 wm5100_reg_defaults[WM5100_MAX_REGISTER + 1]; + +#endif -- cgit v1.1 From de02d0786d4075091f5b1860474cd21d85ff5862 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 20 Sep 2011 21:43:24 +0100 Subject: ASoC: Trace and collect statistics for DAPM graph walking One of the longest standing areas for improvement in ASoC has been the DAPM algorithm - it repeats the same checks many times whenever it is run and makes no effort to limit the areas of the graph it checks meaning we do an awful lot of walks over the full graph. This has never mattered too much as the size of the graph has generally been small in relation to the size of the devices supported and the speed of CPUs but it is annoying. In preparation for work on improving this insert a trace point after the graph walk has been done. This gives us specific timing information for the walk, and in order to give quantifiable (non-benchmark) numbers also count every time we check a link or check the power for a widget and report those numbers. Substantial changes in the algorithm may require tweaks to the stats but they should be useful for simpler things. Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 17 +++++++++++++++++ 1 file changed, 17 insertions(+) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 4a440b5..6a1e13e 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -48,6 +48,8 @@ #include +#define DAPM_UPDATE_STAT(widget, val) widget->dapm->card->dapm_stats.val++; + /* dapm power sequences - make this per codec in the future */ static int dapm_up_seq[] = { [snd_soc_dapm_pre] = 0, @@ -649,6 +651,8 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget) struct snd_soc_dapm_path *path; int con = 0; + DAPM_UPDATE_STAT(widget, path_checks); + if (widget->id == snd_soc_dapm_supply) return 0; @@ -697,6 +701,8 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget) struct snd_soc_dapm_path *path; int con = 0; + DAPM_UPDATE_STAT(widget, path_checks); + if (widget->id == snd_soc_dapm_supply) return 0; @@ -767,6 +773,8 @@ static int dapm_generic_check_power(struct snd_soc_dapm_widget *w) { int in, out; + DAPM_UPDATE_STAT(w, power_checks); + in = is_connected_input_ep(w); dapm_clear_walk(w->dapm); out = is_connected_output_ep(w); @@ -779,6 +787,8 @@ static int dapm_adc_check_power(struct snd_soc_dapm_widget *w) { int in; + DAPM_UPDATE_STAT(w, power_checks); + if (w->active) { in = is_connected_input_ep(w); dapm_clear_walk(w->dapm); @@ -793,6 +803,8 @@ static int dapm_dac_check_power(struct snd_soc_dapm_widget *w) { int out; + DAPM_UPDATE_STAT(w, power_checks); + if (w->active) { out = is_connected_output_ep(w); dapm_clear_walk(w->dapm); @@ -808,6 +820,8 @@ static int dapm_supply_check_power(struct snd_soc_dapm_widget *w) struct snd_soc_dapm_path *path; int power = 0; + DAPM_UPDATE_STAT(w, power_checks); + /* Check if one of our outputs is connected */ list_for_each_entry(path, &w->sinks, list_source) { if (path->weak) @@ -1208,6 +1222,8 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event) } } + memset(&card->dapm_stats, 0, sizeof(card->dapm_stats)); + /* Check which widgets we need to power and store them in * lists indicating if they should be powered up or down. */ @@ -1299,6 +1315,7 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event) list_for_each_entry(d, &card->dapm_list, list) d->target_bias_level = bias; + trace_snd_soc_dapm_walk_done(card); /* Run all the bias changes in parallel */ list_for_each_entry(d, &dapm->card->dapm_list, list) -- cgit v1.1 From 7c81beb048b49a9fe73254c6e6396e4b1937cdb9 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 20 Sep 2011 22:22:32 +0100 Subject: ASoC: Factor out per-widget DAPM power checks The indentation is getting a little deep. Should be straight code motion, no functional changes. Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 112 +++++++++++++++++++++++++++------------------------ 1 file changed, 60 insertions(+), 52 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 6a1e13e..84d1d79 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1191,6 +1191,65 @@ static void dapm_post_sequence_async(void *data, async_cookie_t cookie) } } +static void dapm_power_one_widget(struct snd_soc_dapm_widget *w, + struct list_head *up_list, + struct list_head *down_list) +{ + struct snd_soc_dapm_context *d; + int power; + + switch (w->id) { + case snd_soc_dapm_pre: + dapm_seq_insert(w, down_list, false); + break; + case snd_soc_dapm_post: + dapm_seq_insert(w, up_list, true); + break; + + default: + if (!w->power_check) + break; + + if (!w->force) + power = w->power_check(w); + else + power = 1; + + if (power) { + d = w->dapm; + + /* Supplies and micbiases only bring the + * context up to STANDBY as unless something + * else is active and passing audio they + * generally don't require full power. + */ + switch (w->id) { + case snd_soc_dapm_supply: + case snd_soc_dapm_micbias: + if (d->target_bias_level < SND_SOC_BIAS_STANDBY) + d->target_bias_level = SND_SOC_BIAS_STANDBY; + break; + default: + d->target_bias_level = SND_SOC_BIAS_ON; + break; + } + } + + if (w->power == power) + break; + + trace_snd_soc_dapm_widget_power(w, power); + + if (power) + dapm_seq_insert(w, up_list, true); + else + dapm_seq_insert(w, down_list, false); + + w->power = power; + break; + } +} + /* * Scan each dapm widget for complete audio path. * A complete path is a route that has valid endpoints i.e.:- @@ -1209,7 +1268,6 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event) LIST_HEAD(down_list); LIST_HEAD(async_domain); enum snd_soc_bias_level bias; - int power; trace_snd_soc_dapm_start(card); @@ -1228,57 +1286,7 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event) * lists indicating if they should be powered up or down. */ list_for_each_entry(w, &card->widgets, list) { - switch (w->id) { - case snd_soc_dapm_pre: - dapm_seq_insert(w, &down_list, false); - break; - case snd_soc_dapm_post: - dapm_seq_insert(w, &up_list, true); - break; - - default: - if (!w->power_check) - continue; - - if (!w->force) - power = w->power_check(w); - else - power = 1; - - if (power) { - d = w->dapm; - - /* Supplies and micbiases only bring - * the context up to STANDBY as unless - * something else is active and - * passing audio they generally don't - * require full power. - */ - switch (w->id) { - case snd_soc_dapm_supply: - case snd_soc_dapm_micbias: - if (d->target_bias_level < SND_SOC_BIAS_STANDBY) - d->target_bias_level = SND_SOC_BIAS_STANDBY; - break; - default: - d->target_bias_level = SND_SOC_BIAS_ON; - break; - } - } - - if (w->power == power) - continue; - - trace_snd_soc_dapm_widget_power(w, power); - - if (power) - dapm_seq_insert(w, &up_list, true); - else - dapm_seq_insert(w, &down_list, false); - - w->power = power; - break; - } + dapm_power_one_widget(w, &up_list, &down_list); } /* If there are no DAPM widgets then try to figure out power from the -- cgit v1.1 From 3156cf73bdde4903bcd6db3fb43943b399813ba7 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Wed, 21 Sep 2011 14:40:05 +0800 Subject: ASoC: Staticise bf5xx_pcm_ac97_new() It is not used outside this driver so no need to make the symbol global. Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/blackfin/bf5xx-ac97-pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/blackfin/bf5xx-ac97-pcm.c b/sound/soc/blackfin/bf5xx-ac97-pcm.c index 9e59f68..56815c1 100644 --- a/sound/soc/blackfin/bf5xx-ac97-pcm.c +++ b/sound/soc/blackfin/bf5xx-ac97-pcm.c @@ -418,7 +418,7 @@ static void bf5xx_pcm_free_dma_buffers(struct snd_pcm *pcm) static u64 bf5xx_pcm_dmamask = DMA_BIT_MASK(32); -int bf5xx_pcm_ac97_new(struct snd_soc_pcm_runtime *rtd) +static int bf5xx_pcm_ac97_new(struct snd_soc_pcm_runtime *rtd) { struct snd_card *card = rtd->card->snd_card; struct snd_soc_dai *dai = rtd->cpu_dai; -- cgit v1.1 From 3b70e3b55dad3b5f552fd03714d551a905794b04 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Wed, 21 Sep 2011 14:40:58 +0800 Subject: ASoC: Staticise bf5xx_pcm_i2s_new() It is not used outside this driver so no need to make the symbol global. Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/blackfin/bf5xx-i2s-pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/blackfin/bf5xx-i2s-pcm.c b/sound/soc/blackfin/bf5xx-i2s-pcm.c index 61ddf94..7565e15 100644 --- a/sound/soc/blackfin/bf5xx-i2s-pcm.c +++ b/sound/soc/blackfin/bf5xx-i2s-pcm.c @@ -257,7 +257,7 @@ static void bf5xx_pcm_free_dma_buffers(struct snd_pcm *pcm) static u64 bf5xx_pcm_dmamask = DMA_BIT_MASK(32); -int bf5xx_pcm_i2s_new(struct snd_soc_pcm_runtime *rtd) +static int bf5xx_pcm_i2s_new(struct snd_soc_pcm_runtime *rtd) { struct snd_card *card = rtd->card->snd_card; struct snd_soc_dai *dai = rtd->cpu_dai; -- cgit v1.1 From bb754636844eae69dad8c714640ac8ad564ec62a Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Wed, 21 Sep 2011 14:41:50 +0800 Subject: ASoC: Staticise jz4740_pcm_new() It is not used outside this driver so no need to make the symbol global. Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/jz4740/jz4740-pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/jz4740/jz4740-pcm.c b/sound/soc/jz4740/jz4740-pcm.c index a7c9578..d1989cd 100644 --- a/sound/soc/jz4740/jz4740-pcm.c +++ b/sound/soc/jz4740/jz4740-pcm.c @@ -299,7 +299,7 @@ static void jz4740_pcm_free(struct snd_pcm *pcm) static u64 jz4740_pcm_dmamask = DMA_BIT_MASK(32); -int jz4740_pcm_new(struct snd_soc_pcm_runtime *rtd) +static int jz4740_pcm_new(struct snd_soc_pcm_runtime *rtd) { struct snd_card *card = rtd->card->snd_card; struct snd_soc_dai *dai = rtd->cpu_dai; -- cgit v1.1 From 49acf73bd03a22d1a0e866b244a4003c222aabee Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Wed, 21 Sep 2011 14:42:49 +0800 Subject: ASoC: Staticise nuc900_dma_getposition() It is not used outside this driver so no need to make the symbol global. Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/nuc900/nuc900-pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/nuc900/nuc900-pcm.c b/sound/soc/nuc900/nuc900-pcm.c index e46d551..4e3626b 100644 --- a/sound/soc/nuc900/nuc900-pcm.c +++ b/sound/soc/nuc900/nuc900-pcm.c @@ -227,7 +227,7 @@ static int nuc900_dma_trigger(struct snd_pcm_substream *substream, int cmd) return ret; } -int nuc900_dma_getposition(struct snd_pcm_substream *substream, +static int nuc900_dma_getposition(struct snd_pcm_substream *substream, dma_addr_t *src, dma_addr_t *dst) { struct snd_pcm_runtime *runtime = substream->runtime; -- cgit v1.1 From d80852223ecabd1ab433a9c71436d81b697ef1fc Mon Sep 17 00:00:00 2001 From: "johnnyhsu@realtek.com" Date: Wed, 7 Sep 2011 11:16:35 +0800 Subject: ASoC: Add driver for rt5631 Signed-off-by: Johnny Hsu Signed-off-by: Mark Brown --- sound/soc/codecs/rt5631.c | 1789 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/rt5631.h | 701 ++++++++++++++++++ 2 files changed, 2490 insertions(+) create mode 100644 sound/soc/codecs/rt5631.c create mode 100644 sound/soc/codecs/rt5631.h (limited to 'sound') diff --git a/sound/soc/codecs/rt5631.c b/sound/soc/codecs/rt5631.c new file mode 100644 index 0000000..110b3d8 --- /dev/null +++ b/sound/soc/codecs/rt5631.c @@ -0,0 +1,1789 @@ +/* + * rt5631.c -- RT5631 ALSA Soc Audio driver + * + * Copyright 2011 Realtek Microelectronics + * + * Author: flove + * + * Based on WM8753.c + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + */ +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "rt5631.h" + +struct rt5631_priv { + struct snd_soc_codec *codec; + int codec_version; + int master; + int sysclk; + int rx_rate; + int bclk_rate; + int dmic_used_flag; +}; + +static const u16 rt5631_reg[RT5631_VENDOR_ID2 + 1] = { + [RT5631_SPK_OUT_VOL] = 0x8888, + [RT5631_HP_OUT_VOL] = 0x8080, + [RT5631_MONO_AXO_1_2_VOL] = 0xa080, + [RT5631_AUX_IN_VOL] = 0x0808, + [RT5631_ADC_REC_MIXER] = 0xf0f0, + [RT5631_VDAC_DIG_VOL] = 0x0010, + [RT5631_OUTMIXER_L_CTRL] = 0xffc0, + [RT5631_OUTMIXER_R_CTRL] = 0xffc0, + [RT5631_AXO1MIXER_CTRL] = 0x88c0, + [RT5631_AXO2MIXER_CTRL] = 0x88c0, + [RT5631_DIG_MIC_CTRL] = 0x3000, + [RT5631_MONO_INPUT_VOL] = 0x8808, + [RT5631_SPK_MIXER_CTRL] = 0xf8f8, + [RT5631_SPK_MONO_OUT_CTRL] = 0xfc00, + [RT5631_SPK_MONO_HP_OUT_CTRL] = 0x4440, + [RT5631_SDP_CTRL] = 0x8000, + [RT5631_MONO_SDP_CTRL] = 0x8000, + [RT5631_STEREO_AD_DA_CLK_CTRL] = 0x2010, + [RT5631_GEN_PUR_CTRL_REG] = 0x0e00, + [RT5631_INT_ST_IRQ_CTRL_2] = 0x071a, + [RT5631_MISC_CTRL] = 0x2040, + [RT5631_DEPOP_FUN_CTRL_2] = 0x8000, + [RT5631_SOFT_VOL_CTRL] = 0x07e0, + [RT5631_ALC_CTRL_1] = 0x0206, + [RT5631_ALC_CTRL_3] = 0x2000, + [RT5631_PSEUDO_SPATL_CTRL] = 0x0553, +}; + +/** + * rt5631_write_index - write index register of 2nd layer + */ +static void rt5631_write_index(struct snd_soc_codec *codec, + unsigned int reg, unsigned int value) +{ + snd_soc_write(codec, RT5631_INDEX_ADD, reg); + snd_soc_write(codec, RT5631_INDEX_DATA, value); +} + +/** + * rt5631_read_index - read index register of 2nd layer + */ +static unsigned int rt5631_read_index(struct snd_soc_codec *codec, + unsigned int reg) +{ + unsigned int value; + + snd_soc_write(codec, RT5631_INDEX_ADD, reg); + value = snd_soc_read(codec, RT5631_INDEX_DATA); + + return value; +} + +static int rt5631_reset(struct snd_soc_codec *codec) +{ + return snd_soc_write(codec, RT5631_RESET, 0); +} + +static int rt5631_volatile_register(struct snd_soc_codec *codec, + unsigned int reg) +{ + switch (reg) { + case RT5631_RESET: + case RT5631_INT_ST_IRQ_CTRL_2: + case RT5631_INDEX_ADD: + case RT5631_INDEX_DATA: + case RT5631_EQ_CTRL: + return 1; + default: + return 0; + } +} + +static int rt5631_readable_register(struct snd_soc_codec *codec, + unsigned int reg) +{ + switch (reg) { + case RT5631_RESET: + case RT5631_SPK_OUT_VOL: + case RT5631_HP_OUT_VOL: + case RT5631_MONO_AXO_1_2_VOL: + case RT5631_AUX_IN_VOL: + case RT5631_STEREO_DAC_VOL_1: + case RT5631_MIC_CTRL_1: + case RT5631_STEREO_DAC_VOL_2: + case RT5631_ADC_CTRL_1: + case RT5631_ADC_REC_MIXER: + case RT5631_ADC_CTRL_2: + case RT5631_VDAC_DIG_VOL: + case RT5631_OUTMIXER_L_CTRL: + case RT5631_OUTMIXER_R_CTRL: + case RT5631_AXO1MIXER_CTRL: + case RT5631_AXO2MIXER_CTRL: + case RT5631_MIC_CTRL_2: + case RT5631_DIG_MIC_CTRL: + case RT5631_MONO_INPUT_VOL: + case RT5631_SPK_MIXER_CTRL: + case RT5631_SPK_MONO_OUT_CTRL: + case RT5631_SPK_MONO_HP_OUT_CTRL: + case RT5631_SDP_CTRL: + case RT5631_MONO_SDP_CTRL: + case RT5631_STEREO_AD_DA_CLK_CTRL: + case RT5631_PWR_MANAG_ADD1: + case RT5631_PWR_MANAG_ADD2: + case RT5631_PWR_MANAG_ADD3: + case RT5631_PWR_MANAG_ADD4: + case RT5631_GEN_PUR_CTRL_REG: + case RT5631_GLOBAL_CLK_CTRL: + case RT5631_PLL_CTRL: + case RT5631_INT_ST_IRQ_CTRL_1: + case RT5631_INT_ST_IRQ_CTRL_2: + case RT5631_GPIO_CTRL: + case RT5631_MISC_CTRL: + case RT5631_DEPOP_FUN_CTRL_1: + case RT5631_DEPOP_FUN_CTRL_2: + case RT5631_JACK_DET_CTRL: + case RT5631_SOFT_VOL_CTRL: + case RT5631_ALC_CTRL_1: + case RT5631_ALC_CTRL_2: + case RT5631_ALC_CTRL_3: + case RT5631_PSEUDO_SPATL_CTRL: + case RT5631_INDEX_ADD: + case RT5631_INDEX_DATA: + case RT5631_EQ_CTRL: + case RT5631_VENDOR_ID: + case RT5631_VENDOR_ID1: + case RT5631_VENDOR_ID2: + return 1; + default: + return 0; + } +} + +static const DECLARE_TLV_DB_SCALE(out_vol_tlv, -4650, 150, 0); +static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -95625, 375, 0); +static const DECLARE_TLV_DB_SCALE(in_vol_tlv, -3450, 150, 0); +/* {0, +20, +24, +30, +35, +40, +44, +50, +52}dB */ +static unsigned int mic_bst_tlv[] = { + TLV_DB_RANGE_HEAD(6), + 0, 0, TLV_DB_SCALE_ITEM(0, 0, 0), + 1, 1, TLV_DB_SCALE_ITEM(2000, 0, 0), + 2, 2, TLV_DB_SCALE_ITEM(2400, 0, 0), + 3, 5, TLV_DB_SCALE_ITEM(3000, 500, 0), + 6, 6, TLV_DB_SCALE_ITEM(4400, 0, 0), + 7, 7, TLV_DB_SCALE_ITEM(5000, 0, 0), + 8, 8, TLV_DB_SCALE_ITEM(5200, 0, 0), +}; + +static int rt5631_dmic_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct rt5631_priv *rt5631 = snd_soc_codec_get_drvdata(codec); + + ucontrol->value.integer.value[0] = rt5631->dmic_used_flag; + + return 0; +} + +static int rt5631_dmic_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct rt5631_priv *rt5631 = snd_soc_codec_get_drvdata(codec); + + rt5631->dmic_used_flag = ucontrol->value.integer.value[0]; + return 0; +} + +/* MIC Input Type */ +static const char *rt5631_input_mode[] = { + "Single ended", "Differential"}; + +static const SOC_ENUM_SINGLE_DECL( + rt5631_mic1_mode_enum, RT5631_MIC_CTRL_1, + RT5631_MIC1_DIFF_INPUT_SHIFT, rt5631_input_mode); + +static const SOC_ENUM_SINGLE_DECL( + rt5631_mic2_mode_enum, RT5631_MIC_CTRL_1, + RT5631_MIC2_DIFF_INPUT_SHIFT, rt5631_input_mode); + +/* MONO Input Type */ +static const SOC_ENUM_SINGLE_DECL( + rt5631_monoin_mode_enum, RT5631_MONO_INPUT_VOL, + RT5631_MONO_DIFF_INPUT_SHIFT, rt5631_input_mode); + +/* SPK Ratio Gain Control */ +static const char *rt5631_spk_ratio[] = {"1.00x", "1.09x", "1.27x", "1.44x", + "1.56x", "1.68x", "1.99x", "2.34x"}; + +static const SOC_ENUM_SINGLE_DECL( + rt5631_spk_ratio_enum, RT5631_GEN_PUR_CTRL_REG, + RT5631_SPK_AMP_RATIO_CTRL_SHIFT, rt5631_spk_ratio); + +static const struct snd_kcontrol_new rt5631_snd_controls[] = { + /* MIC */ + SOC_ENUM("MIC1 Mode Control", rt5631_mic1_mode_enum), + SOC_SINGLE_TLV("MIC1 Boost", RT5631_MIC_CTRL_2, + RT5631_MIC1_BOOST_SHIFT, 8, 0, mic_bst_tlv), + SOC_ENUM("MIC2 Mode Control", rt5631_mic2_mode_enum), + SOC_SINGLE_TLV("MIC2 Boost", RT5631_MIC_CTRL_2, + RT5631_MIC2_BOOST_SHIFT, 8, 0, mic_bst_tlv), + /* MONO IN */ + SOC_ENUM("MONOIN Mode Control", rt5631_monoin_mode_enum), + SOC_DOUBLE_TLV("MONOIN_RX Capture Volume", RT5631_MONO_INPUT_VOL, + RT5631_L_VOL_SHIFT, RT5631_R_VOL_SHIFT, + RT5631_VOL_MASK, 1, in_vol_tlv), + /* AXI */ + SOC_DOUBLE_TLV("AXI Capture Volume", RT5631_AUX_IN_VOL, + RT5631_L_VOL_SHIFT, RT5631_R_VOL_SHIFT, + RT5631_VOL_MASK, 1, in_vol_tlv), + /* DAC */ + SOC_DOUBLE_TLV("PCM Playback Volume", RT5631_STEREO_DAC_VOL_2, + RT5631_L_VOL_SHIFT, RT5631_R_VOL_SHIFT, + RT5631_DAC_VOL_MASK, 1, dac_vol_tlv), + SOC_DOUBLE("PCM Playback Switch", RT5631_STEREO_DAC_VOL_1, + RT5631_L_MUTE_SHIFT, RT5631_R_MUTE_SHIFT, 1, 1), + /* AXO */ + SOC_SINGLE("AXO1 Playback Switch", RT5631_MONO_AXO_1_2_VOL, + RT5631_L_MUTE_SHIFT, 1, 1), + SOC_SINGLE("AXO2 Playback Switch", RT5631_MONO_AXO_1_2_VOL, + RT5631_R_VOL_SHIFT, 1, 1), + /* OUTVOL */ + SOC_DOUBLE("OUTVOL Channel Switch", RT5631_SPK_OUT_VOL, + RT5631_L_EN_SHIFT, RT5631_R_EN_SHIFT, 1, 0), + + /* SPK */ + SOC_DOUBLE("Speaker Playback Switch", RT5631_SPK_OUT_VOL, + RT5631_L_MUTE_SHIFT, RT5631_R_MUTE_SHIFT, 1, 1), + SOC_DOUBLE_TLV("Speaker Playback Volume", RT5631_SPK_OUT_VOL, + RT5631_L_VOL_SHIFT, RT5631_R_VOL_SHIFT, 39, 1, out_vol_tlv), + /* MONO OUT */ + SOC_SINGLE("MONO Playback Switch", RT5631_MONO_AXO_1_2_VOL, + RT5631_MUTE_MONO_SHIFT, 1, 1), + /* HP */ + SOC_DOUBLE("HP Playback Switch", RT5631_HP_OUT_VOL, + RT5631_L_MUTE_SHIFT, RT5631_R_MUTE_SHIFT, 1, 1), + SOC_DOUBLE_TLV("HP Playback Volume", RT5631_HP_OUT_VOL, + RT5631_L_VOL_SHIFT, RT5631_R_VOL_SHIFT, + RT5631_VOL_MASK, 1, out_vol_tlv), + /* DMIC */ + SOC_SINGLE_EXT("DMIC Switch", 0, 0, 1, 0, + rt5631_dmic_get, rt5631_dmic_put), + SOC_DOUBLE("DMIC Capture Switch", RT5631_DIG_MIC_CTRL, + RT5631_DMIC_L_CH_MUTE_SHIFT, + RT5631_DMIC_R_CH_MUTE_SHIFT, 1, 1), + + /* SPK Ratio Gain Control */ + SOC_ENUM("SPK Ratio Control", rt5631_spk_ratio_enum), +}; + +static int check_sysclk1_source(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + unsigned int reg; + + reg = snd_soc_read(source->codec, RT5631_GLOBAL_CLK_CTRL); + return reg & RT5631_SYSCLK_SOUR_SEL_PLL; +} + +static int check_dmic_used(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + struct rt5631_priv *rt5631 = snd_soc_codec_get_drvdata(source->codec); + return rt5631->dmic_used_flag; +} + +static int check_dacl_to_outmixl(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + unsigned int reg; + + reg = snd_soc_read(source->codec, RT5631_OUTMIXER_L_CTRL); + return !(reg & RT5631_M_DAC_L_TO_OUTMIXER_L); +} + +static int check_dacr_to_outmixr(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + unsigned int reg; + + reg = snd_soc_read(source->codec, RT5631_OUTMIXER_R_CTRL); + return !(reg & RT5631_M_DAC_R_TO_OUTMIXER_R); +} + +static int check_dacl_to_spkmixl(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + unsigned int reg; + + reg = snd_soc_read(source->codec, RT5631_SPK_MIXER_CTRL); + return !(reg & RT5631_M_DAC_L_TO_SPKMIXER_L); +} + +static int check_dacr_to_spkmixr(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + unsigned int reg; + + reg = snd_soc_read(source->codec, RT5631_SPK_MIXER_CTRL); + return !(reg & RT5631_M_DAC_R_TO_SPKMIXER_R); +} + +static int check_vdac_to_outmix(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + unsigned int reg, ret = 1; + + reg = snd_soc_read(source->codec, RT5631_OUTMIXER_L_CTRL); + if (reg & RT5631_M_VDAC_TO_OUTMIXER_L) { + reg = snd_soc_read(source->codec, RT5631_OUTMIXER_R_CTRL); + if (reg & RT5631_M_VDAC_TO_OUTMIXER_R) + ret = 0; + } + return ret; +} + +static int check_adcl_select(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + unsigned int reg; + + reg = snd_soc_read(source->codec, RT5631_ADC_REC_MIXER); + return !(reg & RT5631_M_MIC1_TO_RECMIXER_L); +} + +static int check_adcr_select(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + unsigned int reg; + + reg = snd_soc_read(source->codec, RT5631_ADC_REC_MIXER); + return !(reg & RT5631_M_MIC2_TO_RECMIXER_R); +} + +/** + * onebit_depop_power_stage - auto depop in power stage. + * @enable: power on/off + * + * When power on/off headphone, the depop sequence is done by hardware. + */ +static void onebit_depop_power_stage(struct snd_soc_codec *codec, int enable) +{ + unsigned int soft_vol, hp_zc; + + /* enable one-bit depop function */ + snd_soc_update_bits(codec, RT5631_DEPOP_FUN_CTRL_2, + RT5631_EN_ONE_BIT_DEPOP, 0); + + /* keep soft volume and zero crossing setting */ + soft_vol = snd_soc_read(codec, RT5631_SOFT_VOL_CTRL); + snd_soc_write(codec, RT5631_SOFT_VOL_CTRL, 0); + hp_zc = snd_soc_read(codec, RT5631_INT_ST_IRQ_CTRL_2); + snd_soc_write(codec, RT5631_INT_ST_IRQ_CTRL_2, hp_zc & 0xf7ff); + if (enable) { + /* config one-bit depop parameter */ + rt5631_write_index(codec, RT5631_TEST_MODE_CTRL, 0x84c0); + rt5631_write_index(codec, RT5631_SPK_INTL_CTRL, 0x309f); + rt5631_write_index(codec, RT5631_CP_INTL_REG2, 0x6530); + /* power on capless block */ + snd_soc_write(codec, RT5631_DEPOP_FUN_CTRL_2, + RT5631_EN_CAP_FREE_DEPOP); + } else { + /* power off capless block */ + snd_soc_write(codec, RT5631_DEPOP_FUN_CTRL_2, 0); + msleep(100); + } + + /* recover soft volume and zero crossing setting */ + snd_soc_write(codec, RT5631_SOFT_VOL_CTRL, soft_vol); + snd_soc_write(codec, RT5631_INT_ST_IRQ_CTRL_2, hp_zc); +} + +/** + * onebit_depop_mute_stage - auto depop in mute stage. + * @enable: mute/unmute + * + * When mute/unmute headphone, the depop sequence is done by hardware. + */ +static void onebit_depop_mute_stage(struct snd_soc_codec *codec, int enable) +{ + unsigned int soft_vol, hp_zc; + + /* enable one-bit depop function */ + snd_soc_update_bits(codec, RT5631_DEPOP_FUN_CTRL_2, + RT5631_EN_ONE_BIT_DEPOP, 0); + + /* keep soft volume and zero crossing setting */ + soft_vol = snd_soc_read(codec, RT5631_SOFT_VOL_CTRL); + snd_soc_write(codec, RT5631_SOFT_VOL_CTRL, 0); + hp_zc = snd_soc_read(codec, RT5631_INT_ST_IRQ_CTRL_2); + snd_soc_write(codec, RT5631_INT_ST_IRQ_CTRL_2, hp_zc & 0xf7ff); + if (enable) { + schedule_timeout_uninterruptible(msecs_to_jiffies(10)); + /* config one-bit depop parameter */ + rt5631_write_index(codec, RT5631_SPK_INTL_CTRL, 0x307f); + snd_soc_update_bits(codec, RT5631_HP_OUT_VOL, + RT5631_L_MUTE | RT5631_R_MUTE, 0); + msleep(300); + } else { + snd_soc_update_bits(codec, RT5631_HP_OUT_VOL, + RT5631_L_MUTE | RT5631_R_MUTE, + RT5631_L_MUTE | RT5631_R_MUTE); + msleep(100); + } + + /* recover soft volume and zero crossing setting */ + snd_soc_write(codec, RT5631_SOFT_VOL_CTRL, soft_vol); + snd_soc_write(codec, RT5631_INT_ST_IRQ_CTRL_2, hp_zc); +} + +/** + * onebit_depop_power_stage - step by step depop sequence in power stage. + * @enable: power on/off + * + * When power on/off headphone, the depop sequence is done in step by step. + */ +static void depop_seq_power_stage(struct snd_soc_codec *codec, int enable) +{ + unsigned int soft_vol, hp_zc; + + /* depop control by register */ + snd_soc_update_bits(codec, RT5631_DEPOP_FUN_CTRL_2, + RT5631_EN_ONE_BIT_DEPOP, RT5631_EN_ONE_BIT_DEPOP); + + /* keep soft volume and zero crossing setting */ + soft_vol = snd_soc_read(codec, RT5631_SOFT_VOL_CTRL); + snd_soc_write(codec, RT5631_SOFT_VOL_CTRL, 0); + hp_zc = snd_soc_read(codec, RT5631_INT_ST_IRQ_CTRL_2); + snd_soc_write(codec, RT5631_INT_ST_IRQ_CTRL_2, hp_zc & 0xf7ff); + if (enable) { + /* config depop sequence parameter */ + rt5631_write_index(codec, RT5631_SPK_INTL_CTRL, 0x303e); + + /* power on headphone and charge pump */ + snd_soc_update_bits(codec, RT5631_PWR_MANAG_ADD3, + RT5631_PWR_CHARGE_PUMP | RT5631_PWR_HP_L_AMP | + RT5631_PWR_HP_R_AMP, + RT5631_PWR_CHARGE_PUMP | RT5631_PWR_HP_L_AMP | + RT5631_PWR_HP_R_AMP); + + /* power on soft generator and depop mode2 */ + snd_soc_write(codec, RT5631_DEPOP_FUN_CTRL_1, + RT5631_POW_ON_SOFT_GEN | RT5631_EN_DEPOP2_FOR_HP); + msleep(100); + + /* stop depop mode */ + snd_soc_update_bits(codec, RT5631_PWR_MANAG_ADD3, + RT5631_PWR_HP_DEPOP_DIS, RT5631_PWR_HP_DEPOP_DIS); + } else { + /* config depop sequence parameter */ + rt5631_write_index(codec, RT5631_SPK_INTL_CTRL, 0x303F); + snd_soc_write(codec, RT5631_DEPOP_FUN_CTRL_1, + RT5631_POW_ON_SOFT_GEN | RT5631_EN_MUTE_UNMUTE_DEPOP | + RT5631_PD_HPAMP_L_ST_UP | RT5631_PD_HPAMP_R_ST_UP); + msleep(75); + snd_soc_write(codec, RT5631_DEPOP_FUN_CTRL_1, + RT5631_POW_ON_SOFT_GEN | RT5631_PD_HPAMP_L_ST_UP | + RT5631_PD_HPAMP_R_ST_UP); + + /* start depop mode */ + snd_soc_update_bits(codec, RT5631_PWR_MANAG_ADD3, + RT5631_PWR_HP_DEPOP_DIS, 0); + + /* config depop sequence parameter */ + snd_soc_write(codec, RT5631_DEPOP_FUN_CTRL_1, + RT5631_POW_ON_SOFT_GEN | RT5631_EN_DEPOP2_FOR_HP | + RT5631_PD_HPAMP_L_ST_UP | RT5631_PD_HPAMP_R_ST_UP); + msleep(80); + snd_soc_write(codec, RT5631_DEPOP_FUN_CTRL_1, + RT5631_POW_ON_SOFT_GEN); + + /* power down headphone and charge pump */ + snd_soc_update_bits(codec, RT5631_PWR_MANAG_ADD3, + RT5631_PWR_CHARGE_PUMP | RT5631_PWR_HP_L_AMP | + RT5631_PWR_HP_R_AMP, 0); + } + + /* recover soft volume and zero crossing setting */ + snd_soc_write(codec, RT5631_SOFT_VOL_CTRL, soft_vol); + snd_soc_write(codec, RT5631_INT_ST_IRQ_CTRL_2, hp_zc); +} + +/** + * depop_seq_mute_stage - step by step depop sequence in mute stage. + * @enable: mute/unmute + * + * When mute/unmute headphone, the depop sequence is done in step by step. + */ +static void depop_seq_mute_stage(struct snd_soc_codec *codec, int enable) +{ + unsigned int soft_vol, hp_zc; + + /* depop control by register */ + snd_soc_update_bits(codec, RT5631_DEPOP_FUN_CTRL_2, + RT5631_EN_ONE_BIT_DEPOP, RT5631_EN_ONE_BIT_DEPOP); + + /* keep soft volume and zero crossing setting */ + soft_vol = snd_soc_read(codec, RT5631_SOFT_VOL_CTRL); + snd_soc_write(codec, RT5631_SOFT_VOL_CTRL, 0); + hp_zc = snd_soc_read(codec, RT5631_INT_ST_IRQ_CTRL_2); + snd_soc_write(codec, RT5631_INT_ST_IRQ_CTRL_2, hp_zc & 0xf7ff); + if (enable) { + schedule_timeout_uninterruptible(msecs_to_jiffies(10)); + + /* config depop sequence parameter */ + rt5631_write_index(codec, RT5631_SPK_INTL_CTRL, 0x302f); + snd_soc_write(codec, RT5631_DEPOP_FUN_CTRL_1, + RT5631_POW_ON_SOFT_GEN | RT5631_EN_MUTE_UNMUTE_DEPOP | + RT5631_EN_HP_R_M_UN_MUTE_DEPOP | + RT5631_EN_HP_L_M_UN_MUTE_DEPOP); + + snd_soc_update_bits(codec, RT5631_HP_OUT_VOL, + RT5631_L_MUTE | RT5631_R_MUTE, 0); + msleep(160); + } else { + /* config depop sequence parameter */ + rt5631_write_index(codec, RT5631_SPK_INTL_CTRL, 0x302f); + snd_soc_write(codec, RT5631_DEPOP_FUN_CTRL_1, + RT5631_POW_ON_SOFT_GEN | RT5631_EN_MUTE_UNMUTE_DEPOP | + RT5631_EN_HP_R_M_UN_MUTE_DEPOP | + RT5631_EN_HP_L_M_UN_MUTE_DEPOP); + + snd_soc_update_bits(codec, RT5631_HP_OUT_VOL, + RT5631_L_MUTE | RT5631_R_MUTE, + RT5631_L_MUTE | RT5631_R_MUTE); + msleep(150); + } + + /* recover soft volume and zero crossing setting */ + snd_soc_write(codec, RT5631_SOFT_VOL_CTRL, soft_vol); + snd_soc_write(codec, RT5631_INT_ST_IRQ_CTRL_2, hp_zc); +} + +static int hp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct rt5631_priv *rt5631 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_PRE_PMD: + if (rt5631->codec_version) { + onebit_depop_mute_stage(codec, 0); + onebit_depop_power_stage(codec, 0); + } else { + depop_seq_mute_stage(codec, 0); + depop_seq_power_stage(codec, 0); + } + break; + + case SND_SOC_DAPM_POST_PMU: + if (rt5631->codec_version) { + onebit_depop_power_stage(codec, 1); + onebit_depop_mute_stage(codec, 1); + } else { + depop_seq_power_stage(codec, 1); + depop_seq_mute_stage(codec, 1); + } + break; + + default: + break; + } + + return 0; +} + +static int set_dmic_params(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct rt5631_priv *rt5631 = snd_soc_codec_get_drvdata(codec); + + switch (rt5631->rx_rate) { + case 44100: + case 48000: + snd_soc_update_bits(codec, RT5631_DIG_MIC_CTRL, + RT5631_DMIC_CLK_CTRL_MASK, + RT5631_DMIC_CLK_CTRL_TO_32FS); + break; + + case 32000: + case 22050: + snd_soc_update_bits(codec, RT5631_DIG_MIC_CTRL, + RT5631_DMIC_CLK_CTRL_MASK, + RT5631_DMIC_CLK_CTRL_TO_64FS); + break; + + case 16000: + case 11025: + case 8000: + snd_soc_update_bits(codec, RT5631_DIG_MIC_CTRL, + RT5631_DMIC_CLK_CTRL_MASK, + RT5631_DMIC_CLK_CTRL_TO_128FS); + break; + + default: + return -EINVAL; + } + + return 0; +} + +static const struct snd_kcontrol_new rt5631_recmixl_mixer_controls[] = { + SOC_DAPM_SINGLE("OUTMIXL Capture Switch", RT5631_ADC_REC_MIXER, + RT5631_M_OUTMIXL_RECMIXL_BIT, 1, 1), + SOC_DAPM_SINGLE("MIC1_BST1 Capture Switch", RT5631_ADC_REC_MIXER, + RT5631_M_MIC1_RECMIXL_BIT, 1, 1), + SOC_DAPM_SINGLE("AXILVOL Capture Switch", RT5631_ADC_REC_MIXER, + RT5631_M_AXIL_RECMIXL_BIT, 1, 1), + SOC_DAPM_SINGLE("MONOIN_RX Capture Switch", RT5631_ADC_REC_MIXER, + RT5631_M_MONO_IN_RECMIXL_BIT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5631_recmixr_mixer_controls[] = { + SOC_DAPM_SINGLE("MONOIN_RX Capture Switch", RT5631_ADC_REC_MIXER, + RT5631_M_MONO_IN_RECMIXR_BIT, 1, 1), + SOC_DAPM_SINGLE("AXIRVOL Capture Switch", RT5631_ADC_REC_MIXER, + RT5631_M_AXIR_RECMIXR_BIT, 1, 1), + SOC_DAPM_SINGLE("MIC2_BST2 Capture Switch", RT5631_ADC_REC_MIXER, + RT5631_M_MIC2_RECMIXR_BIT, 1, 1), + SOC_DAPM_SINGLE("OUTMIXR Capture Switch", RT5631_ADC_REC_MIXER, + RT5631_M_OUTMIXR_RECMIXR_BIT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5631_spkmixl_mixer_controls[] = { + SOC_DAPM_SINGLE("RECMIXL Playback Switch", RT5631_SPK_MIXER_CTRL, + RT5631_M_RECMIXL_SPKMIXL_BIT, 1, 1), + SOC_DAPM_SINGLE("MIC1_P Playback Switch", RT5631_SPK_MIXER_CTRL, + RT5631_M_MIC1P_SPKMIXL_BIT, 1, 1), + SOC_DAPM_SINGLE("DACL Playback Switch", RT5631_SPK_MIXER_CTRL, + RT5631_M_DACL_SPKMIXL_BIT, 1, 1), + SOC_DAPM_SINGLE("OUTMIXL Playback Switch", RT5631_SPK_MIXER_CTRL, + RT5631_M_OUTMIXL_SPKMIXL_BIT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5631_spkmixr_mixer_controls[] = { + SOC_DAPM_SINGLE("OUTMIXR Playback Switch", RT5631_SPK_MIXER_CTRL, + RT5631_M_OUTMIXR_SPKMIXR_BIT, 1, 1), + SOC_DAPM_SINGLE("DACR Playback Switch", RT5631_SPK_MIXER_CTRL, + RT5631_M_DACR_SPKMIXR_BIT, 1, 1), + SOC_DAPM_SINGLE("MIC2_P Playback Switch", RT5631_SPK_MIXER_CTRL, + RT5631_M_MIC2P_SPKMIXR_BIT, 1, 1), + SOC_DAPM_SINGLE("RECMIXR Playback Switch", RT5631_SPK_MIXER_CTRL, + RT5631_M_RECMIXR_SPKMIXR_BIT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5631_outmixl_mixer_controls[] = { + SOC_DAPM_SINGLE("RECMIXL Playback Switch", RT5631_OUTMIXER_L_CTRL, + RT5631_M_RECMIXL_OUTMIXL_BIT, 1, 1), + SOC_DAPM_SINGLE("RECMIXR Playback Switch", RT5631_OUTMIXER_L_CTRL, + RT5631_M_RECMIXR_OUTMIXL_BIT, 1, 1), + SOC_DAPM_SINGLE("DACL Playback Switch", RT5631_OUTMIXER_L_CTRL, + RT5631_M_DACL_OUTMIXL_BIT, 1, 1), + SOC_DAPM_SINGLE("MIC1_BST1 Playback Switch", RT5631_OUTMIXER_L_CTRL, + RT5631_M_MIC1_OUTMIXL_BIT, 1, 1), + SOC_DAPM_SINGLE("MIC2_BST2 Playback Switch", RT5631_OUTMIXER_L_CTRL, + RT5631_M_MIC2_OUTMIXL_BIT, 1, 1), + SOC_DAPM_SINGLE("MONOIN_RXP Playback Switch", RT5631_OUTMIXER_L_CTRL, + RT5631_M_MONO_INP_OUTMIXL_BIT, 1, 1), + SOC_DAPM_SINGLE("AXILVOL Playback Switch", RT5631_OUTMIXER_L_CTRL, + RT5631_M_AXIL_OUTMIXL_BIT, 1, 1), + SOC_DAPM_SINGLE("AXIRVOL Playback Switch", RT5631_OUTMIXER_L_CTRL, + RT5631_M_AXIR_OUTMIXL_BIT, 1, 1), + SOC_DAPM_SINGLE("VDAC Playback Switch", RT5631_OUTMIXER_L_CTRL, + RT5631_M_VDAC_OUTMIXL_BIT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5631_outmixr_mixer_controls[] = { + SOC_DAPM_SINGLE("VDAC Playback Switch", RT5631_OUTMIXER_R_CTRL, + RT5631_M_VDAC_OUTMIXR_BIT, 1, 1), + SOC_DAPM_SINGLE("AXIRVOL Playback Switch", RT5631_OUTMIXER_R_CTRL, + RT5631_M_AXIR_OUTMIXR_BIT, 1, 1), + SOC_DAPM_SINGLE("AXILVOL Playback Switch", RT5631_OUTMIXER_R_CTRL, + RT5631_M_AXIL_OUTMIXR_BIT, 1, 1), + SOC_DAPM_SINGLE("MONOIN_RXN Playback Switch", RT5631_OUTMIXER_R_CTRL, + RT5631_M_MONO_INN_OUTMIXR_BIT, 1, 1), + SOC_DAPM_SINGLE("MIC2_BST2 Playback Switch", RT5631_OUTMIXER_R_CTRL, + RT5631_M_MIC2_OUTMIXR_BIT, 1, 1), + SOC_DAPM_SINGLE("MIC1_BST1 Playback Switch", RT5631_OUTMIXER_R_CTRL, + RT5631_M_MIC1_OUTMIXR_BIT, 1, 1), + SOC_DAPM_SINGLE("DACR Playback Switch", RT5631_OUTMIXER_R_CTRL, + RT5631_M_DACR_OUTMIXR_BIT, 1, 1), + SOC_DAPM_SINGLE("RECMIXR Playback Switch", RT5631_OUTMIXER_R_CTRL, + RT5631_M_RECMIXR_OUTMIXR_BIT, 1, 1), + SOC_DAPM_SINGLE("RECMIXL Playback Switch", RT5631_OUTMIXER_R_CTRL, + RT5631_M_RECMIXL_OUTMIXR_BIT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5631_AXO1MIX_mixer_controls[] = { + SOC_DAPM_SINGLE("MIC1_BST1 Playback Switch", RT5631_AXO1MIXER_CTRL, + RT5631_M_MIC1_AXO1MIX_BIT , 1, 1), + SOC_DAPM_SINGLE("MIC2_BST2 Playback Switch", RT5631_AXO1MIXER_CTRL, + RT5631_M_MIC2_AXO1MIX_BIT, 1, 1), + SOC_DAPM_SINGLE("OUTVOLL Playback Switch", RT5631_AXO1MIXER_CTRL, + RT5631_M_OUTMIXL_AXO1MIX_BIT , 1 , 1), + SOC_DAPM_SINGLE("OUTVOLR Playback Switch", RT5631_AXO1MIXER_CTRL, + RT5631_M_OUTMIXR_AXO1MIX_BIT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5631_AXO2MIX_mixer_controls[] = { + SOC_DAPM_SINGLE("MIC1_BST1 Playback Switch", RT5631_AXO2MIXER_CTRL, + RT5631_M_MIC1_AXO2MIX_BIT, 1, 1), + SOC_DAPM_SINGLE("MIC2_BST2 Playback Switch", RT5631_AXO2MIXER_CTRL, + RT5631_M_MIC2_AXO2MIX_BIT, 1, 1), + SOC_DAPM_SINGLE("OUTVOLL Playback Switch", RT5631_AXO2MIXER_CTRL, + RT5631_M_OUTMIXL_AXO2MIX_BIT, 1, 1), + SOC_DAPM_SINGLE("OUTVOLR Playback Switch", RT5631_AXO2MIXER_CTRL, + RT5631_M_OUTMIXR_AXO2MIX_BIT, 1 , 1), +}; + +static const struct snd_kcontrol_new rt5631_spolmix_mixer_controls[] = { + SOC_DAPM_SINGLE("SPKVOLL Playback Switch", RT5631_SPK_MONO_OUT_CTRL, + RT5631_M_SPKVOLL_SPOLMIX_BIT, 1, 1), + SOC_DAPM_SINGLE("SPKVOLR Playback Switch", RT5631_SPK_MONO_OUT_CTRL, + RT5631_M_SPKVOLR_SPOLMIX_BIT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5631_spormix_mixer_controls[] = { + SOC_DAPM_SINGLE("SPKVOLL Playback Switch", RT5631_SPK_MONO_OUT_CTRL, + RT5631_M_SPKVOLL_SPORMIX_BIT, 1, 1), + SOC_DAPM_SINGLE("SPKVOLR Playback Switch", RT5631_SPK_MONO_OUT_CTRL, + RT5631_M_SPKVOLR_SPORMIX_BIT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5631_monomix_mixer_controls[] = { + SOC_DAPM_SINGLE("OUTVOLL Playback Switch", RT5631_SPK_MONO_OUT_CTRL, + RT5631_M_OUTVOLL_MONOMIX_BIT, 1, 1), + SOC_DAPM_SINGLE("OUTVOLR Playback Switch", RT5631_SPK_MONO_OUT_CTRL, + RT5631_M_OUTVOLR_MONOMIX_BIT, 1, 1), +}; + +/* Left SPK Volume Input */ +static const char *rt5631_spkvoll_sel[] = {"Vmid", "SPKMIXL"}; + +static const SOC_ENUM_SINGLE_DECL( + rt5631_spkvoll_enum, RT5631_SPK_OUT_VOL, + RT5631_L_EN_SHIFT, rt5631_spkvoll_sel); + +static const struct snd_kcontrol_new rt5631_spkvoll_mux_control = + SOC_DAPM_ENUM("Left SPKVOL SRC", rt5631_spkvoll_enum); + +/* Left HP Volume Input */ +static const char *rt5631_hpvoll_sel[] = {"Vmid", "OUTMIXL"}; + +static const SOC_ENUM_SINGLE_DECL( + rt5631_hpvoll_enum, RT5631_HP_OUT_VOL, + RT5631_L_EN_SHIFT, rt5631_hpvoll_sel); + +static const struct snd_kcontrol_new rt5631_hpvoll_mux_control = + SOC_DAPM_ENUM("Left HPVOL SRC", rt5631_hpvoll_enum); + +/* Left Out Volume Input */ +static const char *rt5631_outvoll_sel[] = {"Vmid", "OUTMIXL"}; + +static const SOC_ENUM_SINGLE_DECL( + rt5631_outvoll_enum, RT5631_MONO_AXO_1_2_VOL, + RT5631_L_EN_SHIFT, rt5631_outvoll_sel); + +static const struct snd_kcontrol_new rt5631_outvoll_mux_control = + SOC_DAPM_ENUM("Left OUTVOL SRC", rt5631_outvoll_enum); + +/* Right Out Volume Input */ +static const char *rt5631_outvolr_sel[] = {"Vmid", "OUTMIXR"}; + +static const SOC_ENUM_SINGLE_DECL( + rt5631_outvolr_enum, RT5631_MONO_AXO_1_2_VOL, + RT5631_R_EN_SHIFT, rt5631_outvolr_sel); + +static const struct snd_kcontrol_new rt5631_outvolr_mux_control = + SOC_DAPM_ENUM("Right OUTVOL SRC", rt5631_outvolr_enum); + +/* Right HP Volume Input */ +static const char *rt5631_hpvolr_sel[] = {"Vmid", "OUTMIXR"}; + +static const SOC_ENUM_SINGLE_DECL( + rt5631_hpvolr_enum, RT5631_HP_OUT_VOL, + RT5631_R_EN_SHIFT, rt5631_hpvolr_sel); + +static const struct snd_kcontrol_new rt5631_hpvolr_mux_control = + SOC_DAPM_ENUM("Right HPVOL SRC", rt5631_hpvolr_enum); + +/* Right SPK Volume Input */ +static const char *rt5631_spkvolr_sel[] = {"Vmid", "SPKMIXR"}; + +static const SOC_ENUM_SINGLE_DECL( + rt5631_spkvolr_enum, RT5631_SPK_OUT_VOL, + RT5631_R_EN_SHIFT, rt5631_spkvolr_sel); + +static const struct snd_kcontrol_new rt5631_spkvolr_mux_control = + SOC_DAPM_ENUM("Right SPKVOL SRC", rt5631_spkvolr_enum); + +/* SPO Left Channel Input */ +static const char *rt5631_spol_src_sel[] = { + "SPOLMIX", "MONOIN_RX", "VDAC", "DACL"}; + +static const SOC_ENUM_SINGLE_DECL( + rt5631_spol_src_enum, RT5631_SPK_MONO_HP_OUT_CTRL, + RT5631_SPK_L_MUX_SEL_SHIFT, rt5631_spol_src_sel); + +static const struct snd_kcontrol_new rt5631_spol_mux_control = + SOC_DAPM_ENUM("SPOL SRC", rt5631_spol_src_enum); + +/* SPO Right Channel Input */ +static const char *rt5631_spor_src_sel[] = { + "SPORMIX", "MONOIN_RX", "VDAC", "DACR"}; + +static const SOC_ENUM_SINGLE_DECL( + rt5631_spor_src_enum, RT5631_SPK_MONO_HP_OUT_CTRL, + RT5631_SPK_R_MUX_SEL_SHIFT, rt5631_spor_src_sel); + +static const struct snd_kcontrol_new rt5631_spor_mux_control = + SOC_DAPM_ENUM("SPOR SRC", rt5631_spor_src_enum); + +/* MONO Input */ +static const char *rt5631_mono_src_sel[] = {"MONOMIX", "MONOIN_RX", "VDAC"}; + +static const SOC_ENUM_SINGLE_DECL( + rt5631_mono_src_enum, RT5631_SPK_MONO_HP_OUT_CTRL, + RT5631_MONO_MUX_SEL_SHIFT, rt5631_mono_src_sel); + +static const struct snd_kcontrol_new rt5631_mono_mux_control = + SOC_DAPM_ENUM("MONO SRC", rt5631_mono_src_enum); + +/* Left HPO Input */ +static const char *rt5631_hpl_src_sel[] = {"Left HPVOL", "Left DAC"}; + +static const SOC_ENUM_SINGLE_DECL( + rt5631_hpl_src_enum, RT5631_SPK_MONO_HP_OUT_CTRL, + RT5631_HP_L_MUX_SEL_SHIFT, rt5631_hpl_src_sel); + +static const struct snd_kcontrol_new rt5631_hpl_mux_control = + SOC_DAPM_ENUM("HPL SRC", rt5631_hpl_src_enum); + +/* Right HPO Input */ +static const char *rt5631_hpr_src_sel[] = {"Right HPVOL", "Right DAC"}; + +static const SOC_ENUM_SINGLE_DECL( + rt5631_hpr_src_enum, RT5631_SPK_MONO_HP_OUT_CTRL, + RT5631_HP_R_MUX_SEL_SHIFT, rt5631_hpr_src_sel); + +static const struct snd_kcontrol_new rt5631_hpr_mux_control = + SOC_DAPM_ENUM("HPR SRC", rt5631_hpr_src_enum); + +static const struct snd_soc_dapm_widget rt5631_dapm_widgets[] = { + /* Vmid */ + SND_SOC_DAPM_VMID("Vmid"), + /* PLL1 */ + SND_SOC_DAPM_SUPPLY("PLL1", RT5631_PWR_MANAG_ADD2, + RT5631_PWR_PLL1_BIT, 0, NULL, 0), + + /* Input Side */ + /* Input Lines */ + SND_SOC_DAPM_INPUT("MIC1"), + SND_SOC_DAPM_INPUT("MIC2"), + SND_SOC_DAPM_INPUT("AXIL"), + SND_SOC_DAPM_INPUT("AXIR"), + SND_SOC_DAPM_INPUT("MONOIN_RXN"), + SND_SOC_DAPM_INPUT("MONOIN_RXP"), + SND_SOC_DAPM_INPUT("DMIC"), + + /* MICBIAS */ + SND_SOC_DAPM_MICBIAS("MIC Bias1", RT5631_PWR_MANAG_ADD2, + RT5631_PWR_MICBIAS1_VOL_BIT, 0), + SND_SOC_DAPM_MICBIAS("MIC Bias2", RT5631_PWR_MANAG_ADD2, + RT5631_PWR_MICBIAS2_VOL_BIT, 0), + + /* Boost */ + SND_SOC_DAPM_PGA("MIC1 Boost", RT5631_PWR_MANAG_ADD2, + RT5631_PWR_MIC1_BOOT_GAIN_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA("MIC2 Boost", RT5631_PWR_MANAG_ADD2, + RT5631_PWR_MIC2_BOOT_GAIN_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA("MONOIN_RXP Boost", RT5631_PWR_MANAG_ADD4, + RT5631_PWR_MONO_IN_P_VOL_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA("MONOIN_RXN Boost", RT5631_PWR_MANAG_ADD4, + RT5631_PWR_MONO_IN_N_VOL_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA("AXIL Boost", RT5631_PWR_MANAG_ADD4, + RT5631_PWR_AXIL_IN_VOL_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA("AXIR Boost", RT5631_PWR_MANAG_ADD4, + RT5631_PWR_AXIR_IN_VOL_BIT, 0, NULL, 0), + + /* MONO In */ + SND_SOC_DAPM_MIXER("MONO_IN", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* REC Mixer */ + SND_SOC_DAPM_MIXER("RECMIXL Mixer", RT5631_PWR_MANAG_ADD2, + RT5631_PWR_RECMIXER_L_BIT, 0, + &rt5631_recmixl_mixer_controls[0], + ARRAY_SIZE(rt5631_recmixl_mixer_controls)), + SND_SOC_DAPM_MIXER("RECMIXR Mixer", RT5631_PWR_MANAG_ADD2, + RT5631_PWR_RECMIXER_R_BIT, 0, + &rt5631_recmixr_mixer_controls[0], + ARRAY_SIZE(rt5631_recmixr_mixer_controls)), + /* Because of record duplication for L/R channel, + * L/R ADCs need power up at the same time */ + SND_SOC_DAPM_MIXER("ADC Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* DMIC */ + SND_SOC_DAPM_SUPPLY("DMIC Supply", RT5631_DIG_MIC_CTRL, + RT5631_DMIC_ENA_SHIFT, 0, + set_dmic_params, SND_SOC_DAPM_PRE_PMU), + /* ADC Data Srouce */ + SND_SOC_DAPM_SUPPLY("Left ADC Select", RT5631_INT_ST_IRQ_CTRL_2, + RT5631_ADC_DATA_SEL_MIC1_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Right ADC Select", RT5631_INT_ST_IRQ_CTRL_2, + RT5631_ADC_DATA_SEL_MIC2_SHIFT, 0, NULL, 0), + + /* ADCs */ + SND_SOC_DAPM_ADC("Left ADC", "HIFI Capture", + RT5631_PWR_MANAG_ADD1, RT5631_PWR_ADC_L_CLK_BIT, 0), + SND_SOC_DAPM_ADC("Right ADC", "HIFI Capture", + RT5631_PWR_MANAG_ADD1, RT5631_PWR_ADC_R_CLK_BIT, 0), + + /* DAC and ADC supply power */ + SND_SOC_DAPM_SUPPLY("I2S", RT5631_PWR_MANAG_ADD1, + RT5631_PWR_MAIN_I2S_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DAC REF", RT5631_PWR_MANAG_ADD1, + RT5631_PWR_DAC_REF_BIT, 0, NULL, 0), + + /* Output Side */ + /* DACs */ + SND_SOC_DAPM_DAC("Left DAC", "HIFI Playback", + RT5631_PWR_MANAG_ADD1, RT5631_PWR_DAC_L_CLK_BIT, 0), + SND_SOC_DAPM_DAC("Right DAC", "HIFI Playback", + RT5631_PWR_MANAG_ADD1, RT5631_PWR_DAC_R_CLK_BIT, 0), + SND_SOC_DAPM_DAC("Voice DAC", "Voice DAC Mono Playback", + SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_PGA("Voice DAC Boost", SND_SOC_NOPM, 0, 0, NULL, 0), + /* DAC supply power */ + SND_SOC_DAPM_SUPPLY("Left DAC To Mixer", RT5631_PWR_MANAG_ADD1, + RT5631_PWR_DAC_L_TO_MIXER_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Right DAC To Mixer", RT5631_PWR_MANAG_ADD1, + RT5631_PWR_DAC_R_TO_MIXER_BIT, 0, NULL, 0), + + /* Left SPK Mixer */ + SND_SOC_DAPM_MIXER("SPKMIXL Mixer", RT5631_PWR_MANAG_ADD2, + RT5631_PWR_SPKMIXER_L_BIT, 0, + &rt5631_spkmixl_mixer_controls[0], + ARRAY_SIZE(rt5631_spkmixl_mixer_controls)), + /* Left Out Mixer */ + SND_SOC_DAPM_MIXER("OUTMIXL Mixer", RT5631_PWR_MANAG_ADD2, + RT5631_PWR_OUTMIXER_L_BIT, 0, + &rt5631_outmixl_mixer_controls[0], + ARRAY_SIZE(rt5631_outmixl_mixer_controls)), + /* Right Out Mixer */ + SND_SOC_DAPM_MIXER("OUTMIXR Mixer", RT5631_PWR_MANAG_ADD2, + RT5631_PWR_OUTMIXER_R_BIT, 0, + &rt5631_outmixr_mixer_controls[0], + ARRAY_SIZE(rt5631_outmixr_mixer_controls)), + /* Right SPK Mixer */ + SND_SOC_DAPM_MIXER("SPKMIXR Mixer", RT5631_PWR_MANAG_ADD2, + RT5631_PWR_SPKMIXER_R_BIT, 0, + &rt5631_spkmixr_mixer_controls[0], + ARRAY_SIZE(rt5631_spkmixr_mixer_controls)), + + /* Volume Mux */ + SND_SOC_DAPM_MUX("Left SPKVOL Mux", RT5631_PWR_MANAG_ADD4, + RT5631_PWR_SPK_L_VOL_BIT, 0, + &rt5631_spkvoll_mux_control), + SND_SOC_DAPM_MUX("Left HPVOL Mux", RT5631_PWR_MANAG_ADD4, + RT5631_PWR_HP_L_OUT_VOL_BIT, 0, + &rt5631_hpvoll_mux_control), + SND_SOC_DAPM_MUX("Left OUTVOL Mux", RT5631_PWR_MANAG_ADD4, + RT5631_PWR_LOUT_VOL_BIT, 0, + &rt5631_outvoll_mux_control), + SND_SOC_DAPM_MUX("Right OUTVOL Mux", RT5631_PWR_MANAG_ADD4, + RT5631_PWR_ROUT_VOL_BIT, 0, + &rt5631_outvolr_mux_control), + SND_SOC_DAPM_MUX("Right HPVOL Mux", RT5631_PWR_MANAG_ADD4, + RT5631_PWR_HP_R_OUT_VOL_BIT, 0, + &rt5631_hpvolr_mux_control), + SND_SOC_DAPM_MUX("Right SPKVOL Mux", RT5631_PWR_MANAG_ADD4, + RT5631_PWR_SPK_R_VOL_BIT, 0, + &rt5631_spkvolr_mux_control), + + /* DAC To HP */ + SND_SOC_DAPM_PGA_S("Left DAC_HP", 0, SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA_S("Right DAC_HP", 0, SND_SOC_NOPM, 0, 0, NULL, 0), + + /* HP Depop */ + SND_SOC_DAPM_PGA_S("HP Depop", 1, SND_SOC_NOPM, 0, 0, + hp_event, SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + + /* AXO1 Mixer */ + SND_SOC_DAPM_MIXER("AXO1MIX Mixer", RT5631_PWR_MANAG_ADD3, + RT5631_PWR_AXO1MIXER_BIT, 0, + &rt5631_AXO1MIX_mixer_controls[0], + ARRAY_SIZE(rt5631_AXO1MIX_mixer_controls)), + /* SPOL Mixer */ + SND_SOC_DAPM_MIXER("SPOLMIX Mixer", SND_SOC_NOPM, 0, 0, + &rt5631_spolmix_mixer_controls[0], + ARRAY_SIZE(rt5631_spolmix_mixer_controls)), + /* MONO Mixer */ + SND_SOC_DAPM_MIXER("MONOMIX Mixer", RT5631_PWR_MANAG_ADD3, + RT5631_PWR_MONOMIXER_BIT, 0, + &rt5631_monomix_mixer_controls[0], + ARRAY_SIZE(rt5631_monomix_mixer_controls)), + /* SPOR Mixer */ + SND_SOC_DAPM_MIXER("SPORMIX Mixer", SND_SOC_NOPM, 0, 0, + &rt5631_spormix_mixer_controls[0], + ARRAY_SIZE(rt5631_spormix_mixer_controls)), + /* AXO2 Mixer */ + SND_SOC_DAPM_MIXER("AXO2MIX Mixer", RT5631_PWR_MANAG_ADD3, + RT5631_PWR_AXO2MIXER_BIT, 0, + &rt5631_AXO2MIX_mixer_controls[0], + ARRAY_SIZE(rt5631_AXO2MIX_mixer_controls)), + + /* Mux */ + SND_SOC_DAPM_MUX("SPOL Mux", SND_SOC_NOPM, 0, 0, + &rt5631_spol_mux_control), + SND_SOC_DAPM_MUX("SPOR Mux", SND_SOC_NOPM, 0, 0, + &rt5631_spor_mux_control), + SND_SOC_DAPM_MUX("MONO Mux", SND_SOC_NOPM, 0, 0, + &rt5631_mono_mux_control), + SND_SOC_DAPM_MUX("HPL Mux", SND_SOC_NOPM, 0, 0, + &rt5631_hpl_mux_control), + SND_SOC_DAPM_MUX("HPR Mux", SND_SOC_NOPM, 0, 0, + &rt5631_hpr_mux_control), + + /* AMP supply */ + SND_SOC_DAPM_SUPPLY("MONO Depop", RT5631_PWR_MANAG_ADD3, + RT5631_PWR_MONO_DEPOP_DIS_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Class D", RT5631_PWR_MANAG_ADD1, + RT5631_PWR_CLASS_D_BIT, 0, NULL, 0), + + /* Output Lines */ + SND_SOC_DAPM_OUTPUT("AUXO1"), + SND_SOC_DAPM_OUTPUT("AUXO2"), + SND_SOC_DAPM_OUTPUT("SPOL"), + SND_SOC_DAPM_OUTPUT("SPOR"), + SND_SOC_DAPM_OUTPUT("HPOL"), + SND_SOC_DAPM_OUTPUT("HPOR"), + SND_SOC_DAPM_OUTPUT("MONO"), +}; + +static const struct snd_soc_dapm_route rt5631_dapm_routes[] = { + {"MIC1 Boost", NULL, "MIC1"}, + {"MIC2 Boost", NULL, "MIC2"}, + {"MONOIN_RXP Boost", NULL, "MONOIN_RXP"}, + {"MONOIN_RXN Boost", NULL, "MONOIN_RXN"}, + {"AXIL Boost", NULL, "AXIL"}, + {"AXIR Boost", NULL, "AXIR"}, + + {"MONO_IN", NULL, "MONOIN_RXP Boost"}, + {"MONO_IN", NULL, "MONOIN_RXN Boost"}, + + {"RECMIXL Mixer", "OUTMIXL Capture Switch", "OUTMIXL Mixer"}, + {"RECMIXL Mixer", "MIC1_BST1 Capture Switch", "MIC1 Boost"}, + {"RECMIXL Mixer", "AXILVOL Capture Switch", "AXIL Boost"}, + {"RECMIXL Mixer", "MONOIN_RX Capture Switch", "MONO_IN"}, + + {"RECMIXR Mixer", "OUTMIXR Capture Switch", "OUTMIXR Mixer"}, + {"RECMIXR Mixer", "MIC2_BST2 Capture Switch", "MIC2 Boost"}, + {"RECMIXR Mixer", "AXIRVOL Capture Switch", "AXIR Boost"}, + {"RECMIXR Mixer", "MONOIN_RX Capture Switch", "MONO_IN"}, + + {"ADC Mixer", NULL, "RECMIXL Mixer"}, + {"ADC Mixer", NULL, "RECMIXR Mixer"}, + + {"Left ADC", NULL, "ADC Mixer"}, + {"Left ADC", NULL, "Left ADC Select", check_adcl_select}, + {"Left ADC", NULL, "PLL1", check_sysclk1_source}, + {"Left ADC", NULL, "I2S"}, + {"Left ADC", NULL, "DAC REF"}, + + {"Right ADC", NULL, "ADC Mixer"}, + {"Right ADC", NULL, "Right ADC Select", check_adcr_select}, + {"Right ADC", NULL, "PLL1", check_sysclk1_source}, + {"Right ADC", NULL, "I2S"}, + {"Right ADC", NULL, "DAC REF"}, + + {"DMIC", NULL, "DMIC Supply", check_dmic_used}, + {"Left ADC", NULL, "DMIC"}, + {"Right ADC", NULL, "DMIC"}, + + {"Left DAC", NULL, "PLL1", check_sysclk1_source}, + {"Left DAC", NULL, "I2S"}, + {"Left DAC", NULL, "DAC REF"}, + {"Right DAC", NULL, "PLL1", check_sysclk1_source}, + {"Right DAC", NULL, "I2S"}, + {"Right DAC", NULL, "DAC REF"}, + + {"Voice DAC Boost", NULL, "Voice DAC"}, + + {"SPKMIXL Mixer", NULL, "Left DAC To Mixer", check_dacl_to_spkmixl}, + {"SPKMIXL Mixer", "RECMIXL Playback Switch", "RECMIXL Mixer"}, + {"SPKMIXL Mixer", "MIC1_P Playback Switch", "MIC1"}, + {"SPKMIXL Mixer", "DACL Playback Switch", "Left DAC"}, + {"SPKMIXL Mixer", "OUTMIXL Playback Switch", "OUTMIXL Mixer"}, + + {"SPKMIXR Mixer", NULL, "Right DAC To Mixer", check_dacr_to_spkmixr}, + {"SPKMIXR Mixer", "OUTMIXR Playback Switch", "OUTMIXR Mixer"}, + {"SPKMIXR Mixer", "DACR Playback Switch", "Right DAC"}, + {"SPKMIXR Mixer", "MIC2_P Playback Switch", "MIC2"}, + {"SPKMIXR Mixer", "RECMIXR Playback Switch", "RECMIXR Mixer"}, + + {"OUTMIXL Mixer", NULL, "Left DAC To Mixer", check_dacl_to_outmixl}, + {"OUTMIXL Mixer", "RECMIXL Playback Switch", "RECMIXL Mixer"}, + {"OUTMIXL Mixer", "RECMIXR Playback Switch", "RECMIXR Mixer"}, + {"OUTMIXL Mixer", "DACL Playback Switch", "Left DAC"}, + {"OUTMIXL Mixer", "MIC1_BST1 Playback Switch", "MIC1 Boost"}, + {"OUTMIXL Mixer", "MIC2_BST2 Playback Switch", "MIC2 Boost"}, + {"OUTMIXL Mixer", "MONOIN_RXP Playback Switch", "MONOIN_RXP Boost"}, + {"OUTMIXL Mixer", "AXILVOL Playback Switch", "AXIL Boost"}, + {"OUTMIXL Mixer", "AXIRVOL Playback Switch", "AXIR Boost"}, + {"OUTMIXL Mixer", "VDAC Playback Switch", "Voice DAC Boost"}, + + {"OUTMIXR Mixer", NULL, "Right DAC To Mixer", check_dacr_to_outmixr}, + {"OUTMIXR Mixer", "RECMIXL Playback Switch", "RECMIXL Mixer"}, + {"OUTMIXR Mixer", "RECMIXR Playback Switch", "RECMIXR Mixer"}, + {"OUTMIXR Mixer", "DACR Playback Switch", "Right DAC"}, + {"OUTMIXR Mixer", "MIC1_BST1 Playback Switch", "MIC1 Boost"}, + {"OUTMIXR Mixer", "MIC2_BST2 Playback Switch", "MIC2 Boost"}, + {"OUTMIXR Mixer", "MONOIN_RXN Playback Switch", "MONOIN_RXN Boost"}, + {"OUTMIXR Mixer", "AXILVOL Playback Switch", "AXIL Boost"}, + {"OUTMIXR Mixer", "AXIRVOL Playback Switch", "AXIR Boost"}, + {"OUTMIXR Mixer", "VDAC Playback Switch", "Voice DAC Boost"}, + + {"Left SPKVOL Mux", "SPKMIXL", "SPKMIXL Mixer"}, + {"Left SPKVOL Mux", "Vmid", "Vmid"}, + {"Left HPVOL Mux", "OUTMIXL", "OUTMIXL Mixer"}, + {"Left HPVOL Mux", "Vmid", "Vmid"}, + {"Left OUTVOL Mux", "OUTMIXL", "OUTMIXL Mixer"}, + {"Left OUTVOL Mux", "Vmid", "Vmid"}, + {"Right OUTVOL Mux", "OUTMIXR", "OUTMIXR Mixer"}, + {"Right OUTVOL Mux", "Vmid", "Vmid"}, + {"Right HPVOL Mux", "OUTMIXR", "OUTMIXR Mixer"}, + {"Right HPVOL Mux", "Vmid", "Vmid"}, + {"Right SPKVOL Mux", "SPKMIXR", "SPKMIXR Mixer"}, + {"Right SPKVOL Mux", "Vmid", "Vmid"}, + + {"AXO1MIX Mixer", "MIC1_BST1 Playback Switch", "MIC1 Boost"}, + {"AXO1MIX Mixer", "OUTVOLL Playback Switch", "Left OUTVOL Mux"}, + {"AXO1MIX Mixer", "OUTVOLR Playback Switch", "Right OUTVOL Mux"}, + {"AXO1MIX Mixer", "MIC2_BST2 Playback Switch", "MIC2 Boost"}, + + {"AXO2MIX Mixer", "MIC1_BST1 Playback Switch", "MIC1 Boost"}, + {"AXO2MIX Mixer", "OUTVOLL Playback Switch", "Left OUTVOL Mux"}, + {"AXO2MIX Mixer", "OUTVOLR Playback Switch", "Right OUTVOL Mux"}, + {"AXO2MIX Mixer", "MIC2_BST2 Playback Switch", "MIC2 Boost"}, + + {"SPOLMIX Mixer", "SPKVOLL Playback Switch", "Left SPKVOL Mux"}, + {"SPOLMIX Mixer", "SPKVOLR Playback Switch", "Right SPKVOL Mux"}, + + {"SPORMIX Mixer", "SPKVOLL Playback Switch", "Left SPKVOL Mux"}, + {"SPORMIX Mixer", "SPKVOLR Playback Switch", "Right SPKVOL Mux"}, + + {"MONOMIX Mixer", "OUTVOLL Playback Switch", "Left OUTVOL Mux"}, + {"MONOMIX Mixer", "OUTVOLR Playback Switch", "Right OUTVOL Mux"}, + + {"SPOL Mux", "SPOLMIX", "SPOLMIX Mixer"}, + {"SPOL Mux", "MONOIN_RX", "MONO_IN"}, + {"SPOL Mux", "VDAC", "Voice DAC Boost"}, + {"SPOL Mux", "DACL", "Left DAC"}, + + {"SPOR Mux", "SPORMIX", "SPORMIX Mixer"}, + {"SPOR Mux", "MONOIN_RX", "MONO_IN"}, + {"SPOR Mux", "VDAC", "Voice DAC Boost"}, + {"SPOR Mux", "DACR", "Right DAC"}, + + {"MONO Mux", "MONOMIX", "MONOMIX Mixer"}, + {"MONO Mux", "MONOIN_RX", "MONO_IN"}, + {"MONO Mux", "VDAC", "Voice DAC Boost"}, + + {"Right DAC_HP", NULL, "Right DAC"}, + {"Left DAC_HP", NULL, "Left DAC"}, + + {"HPL Mux", "Left HPVOL", "Left HPVOL Mux"}, + {"HPL Mux", "Left DAC", "Left DAC_HP"}, + {"HPR Mux", "Right HPVOL", "Right HPVOL Mux"}, + {"HPR Mux", "Right DAC", "Right DAC_HP"}, + + {"HP Depop", NULL, "HPL Mux"}, + {"HP Depop", NULL, "HPR Mux"}, + + {"AUXO1", NULL, "AXO1MIX Mixer"}, + {"AUXO2", NULL, "AXO2MIX Mixer"}, + + {"SPOL", NULL, "Class D"}, + {"SPOL", NULL, "SPOL Mux"}, + {"SPOR", NULL, "Class D"}, + {"SPOR", NULL, "SPOR Mux"}, + + {"HPOL", NULL, "HP Depop"}, + {"HPOR", NULL, "HP Depop"}, + + {"MONO", NULL, "MONO Depop"}, + {"MONO", NULL, "MONO Mux"}, +}; + +struct coeff_clk_div { + u32 mclk; + u32 bclk; + u32 rate; + u16 reg_val; +}; + +/* PLL divisors */ +struct pll_div { + u32 pll_in; + u32 pll_out; + u16 reg_val; +}; + +static const struct pll_div codec_master_pll_div[] = { + {2048000, 8192000, 0x0ea0}, + {3686400, 8192000, 0x4e27}, + {12000000, 8192000, 0x456b}, + {13000000, 8192000, 0x495f}, + {13100000, 8192000, 0x0320}, + {2048000, 11289600, 0xf637}, + {3686400, 11289600, 0x2f22}, + {12000000, 11289600, 0x3e2f}, + {13000000, 11289600, 0x4d5b}, + {13100000, 11289600, 0x363b}, + {2048000, 16384000, 0x1ea0}, + {3686400, 16384000, 0x9e27}, + {12000000, 16384000, 0x452b}, + {13000000, 16384000, 0x542f}, + {13100000, 16384000, 0x03a0}, + {2048000, 16934400, 0xe625}, + {3686400, 16934400, 0x9126}, + {12000000, 16934400, 0x4d2c}, + {13000000, 16934400, 0x742f}, + {13100000, 16934400, 0x3c27}, + {2048000, 22579200, 0x2aa0}, + {3686400, 22579200, 0x2f20}, + {12000000, 22579200, 0x7e2f}, + {13000000, 22579200, 0x742f}, + {13100000, 22579200, 0x3c27}, + {2048000, 24576000, 0x2ea0}, + {3686400, 24576000, 0xee27}, + {12000000, 24576000, 0x2915}, + {13000000, 24576000, 0x772e}, + {13100000, 24576000, 0x0d20}, + {26000000, 24576000, 0x2027}, + {26000000, 22579200, 0x392f}, + {24576000, 22579200, 0x0921}, + {24576000, 24576000, 0x02a0}, +}; + +static const struct pll_div codec_slave_pll_div[] = { + {256000, 2048000, 0x46f0}, + {256000, 4096000, 0x3ea0}, + {352800, 5644800, 0x3ea0}, + {512000, 8192000, 0x3ea0}, + {1024000, 8192000, 0x46f0}, + {705600, 11289600, 0x3ea0}, + {1024000, 16384000, 0x3ea0}, + {1411200, 22579200, 0x3ea0}, + {1536000, 24576000, 0x3ea0}, + {2048000, 16384000, 0x1ea0}, + {2822400, 22579200, 0x1ea0}, + {2822400, 45158400, 0x5ec0}, + {5644800, 45158400, 0x46f0}, + {3072000, 24576000, 0x1ea0}, + {3072000, 49152000, 0x5ec0}, + {6144000, 49152000, 0x46f0}, + {705600, 11289600, 0x3ea0}, + {705600, 8467200, 0x3ab0}, + {24576000, 24576000, 0x02a0}, + {1411200, 11289600, 0x1690}, + {2822400, 11289600, 0x0a90}, + {1536000, 12288000, 0x1690}, + {3072000, 12288000, 0x0a90}, +}; + +struct coeff_clk_div coeff_div[] = { + /* sysclk is 256fs */ + {2048000, 8000 * 32, 8000, 0x1000}, + {2048000, 8000 * 64, 8000, 0x0000}, + {2822400, 11025 * 32, 11025, 0x1000}, + {2822400, 11025 * 64, 11025, 0x0000}, + {4096000, 16000 * 32, 16000, 0x1000}, + {4096000, 16000 * 64, 16000, 0x0000}, + {5644800, 22050 * 32, 22050, 0x1000}, + {5644800, 22050 * 64, 22050, 0x0000}, + {8192000, 32000 * 32, 32000, 0x1000}, + {8192000, 32000 * 64, 32000, 0x0000}, + {11289600, 44100 * 32, 44100, 0x1000}, + {11289600, 44100 * 64, 44100, 0x0000}, + {12288000, 48000 * 32, 48000, 0x1000}, + {12288000, 48000 * 64, 48000, 0x0000}, + {22579200, 88200 * 32, 88200, 0x1000}, + {22579200, 88200 * 64, 88200, 0x0000}, + {24576000, 96000 * 32, 96000, 0x1000}, + {24576000, 96000 * 64, 96000, 0x0000}, + /* sysclk is 512fs */ + {4096000, 8000 * 32, 8000, 0x3000}, + {4096000, 8000 * 64, 8000, 0x2000}, + {5644800, 11025 * 32, 11025, 0x3000}, + {5644800, 11025 * 64, 11025, 0x2000}, + {8192000, 16000 * 32, 16000, 0x3000}, + {8192000, 16000 * 64, 16000, 0x2000}, + {11289600, 22050 * 32, 22050, 0x3000}, + {11289600, 22050 * 64, 22050, 0x2000}, + {16384000, 32000 * 32, 32000, 0x3000}, + {16384000, 32000 * 64, 32000, 0x2000}, + {22579200, 44100 * 32, 44100, 0x3000}, + {22579200, 44100 * 64, 44100, 0x2000}, + {24576000, 48000 * 32, 48000, 0x3000}, + {24576000, 48000 * 64, 48000, 0x2000}, + {45158400, 88200 * 32, 88200, 0x3000}, + {45158400, 88200 * 64, 88200, 0x2000}, + {49152000, 96000 * 32, 96000, 0x3000}, + {49152000, 96000 * 64, 96000, 0x2000}, + /* sysclk is 24.576Mhz or 22.5792Mhz */ + {24576000, 8000 * 32, 8000, 0x7080}, + {24576000, 8000 * 64, 8000, 0x6080}, + {24576000, 16000 * 32, 16000, 0x5080}, + {24576000, 16000 * 64, 16000, 0x4080}, + {24576000, 24000 * 32, 24000, 0x5000}, + {24576000, 24000 * 64, 24000, 0x4000}, + {24576000, 32000 * 32, 32000, 0x3080}, + {24576000, 32000 * 64, 32000, 0x2080}, + {22579200, 11025 * 32, 11025, 0x7000}, + {22579200, 11025 * 64, 11025, 0x6000}, + {22579200, 22050 * 32, 22050, 0x5000}, + {22579200, 22050 * 64, 22050, 0x4000}, +}; + +static int get_coeff(int mclk, int rate, int timesofbclk) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(coeff_div); i++) { + if (coeff_div[i].mclk == mclk && coeff_div[i].rate == rate && + (coeff_div[i].bclk / coeff_div[i].rate) == timesofbclk) + return i; + } + return -EINVAL; +} + +static int rt5631_hifi_pcm_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->codec; + struct rt5631_priv *rt5631 = snd_soc_codec_get_drvdata(codec); + int timesofbclk = 32, coeff; + unsigned int iface = 0; + + dev_dbg(codec->dev, "enter %s\n", __func__); + + rt5631->bclk_rate = snd_soc_params_to_bclk(params); + if (rt5631->bclk_rate < 0) { + dev_err(codec->dev, "Fail to get BCLK rate\n"); + return rt5631->bclk_rate; + } + rt5631->rx_rate = params_rate(params); + + if (rt5631->master) + coeff = get_coeff(rt5631->sysclk, rt5631->rx_rate, + rt5631->bclk_rate / rt5631->rx_rate); + else + coeff = get_coeff(rt5631->sysclk, rt5631->rx_rate, + timesofbclk); + if (coeff < 0) { + dev_err(codec->dev, "Fail to get coeff\n"); + return -EINVAL; + } + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S20_3LE: + iface |= RT5631_SDP_I2S_DL_20; + break; + case SNDRV_PCM_FORMAT_S24_LE: + iface |= RT5631_SDP_I2S_DL_24; + break; + case SNDRV_PCM_FORMAT_S8: + iface |= RT5631_SDP_I2S_DL_8; + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, RT5631_SDP_CTRL, + RT5631_SDP_I2S_DL_MASK, iface); + snd_soc_write(codec, RT5631_STEREO_AD_DA_CLK_CTRL, + coeff_div[coeff].reg_val); + + return 0; +} + +static int rt5631_hifi_codec_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct rt5631_priv *rt5631 = snd_soc_codec_get_drvdata(codec); + unsigned int iface = 0; + + dev_dbg(codec->dev, "enter %s\n", __func__); + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + rt5631->master = 1; + break; + case SND_SOC_DAIFMT_CBS_CFS: + iface |= RT5631_SDP_MODE_SEL_SLAVE; + rt5631->master = 0; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + break; + case SND_SOC_DAIFMT_LEFT_J: + iface |= RT5631_SDP_I2S_DF_LEFT; + break; + case SND_SOC_DAIFMT_DSP_A: + iface |= RT5631_SDP_I2S_DF_PCM_A; + break; + case SND_SOC_DAIFMT_DSP_B: + iface |= RT5631_SDP_I2S_DF_PCM_B; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_NF: + iface |= RT5631_SDP_I2S_BCLK_POL_CTRL; + break; + default: + return -EINVAL; + } + + snd_soc_write(codec, RT5631_SDP_CTRL, iface); + + return 0; +} + +static int rt5631_hifi_codec_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct rt5631_priv *rt5631 = snd_soc_codec_get_drvdata(codec); + + dev_dbg(codec->dev, "enter %s, syclk=%d\n", __func__, freq); + + if ((freq >= (256 * 8000)) && (freq <= (512 * 96000))) { + rt5631->sysclk = freq; + return 0; + } + + return -EINVAL; +} + +static int rt5631_codec_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, + int source, unsigned int freq_in, unsigned int freq_out) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct rt5631_priv *rt5631 = snd_soc_codec_get_drvdata(codec); + int i, ret = -EINVAL; + + dev_dbg(codec->dev, "enter %s\n", __func__); + + if (!freq_in || !freq_out) { + dev_dbg(codec->dev, "PLL disabled\n"); + + snd_soc_update_bits(codec, RT5631_GLOBAL_CLK_CTRL, + RT5631_SYSCLK_SOUR_SEL_MASK, + RT5631_SYSCLK_SOUR_SEL_MCLK); + + return 0; + } + + if (rt5631->master) { + for (i = 0; i < ARRAY_SIZE(codec_master_pll_div); i++) + if (freq_in == codec_master_pll_div[i].pll_in && + freq_out == codec_master_pll_div[i].pll_out) { + dev_info(codec->dev, + "change PLL in master mode\n"); + snd_soc_write(codec, RT5631_PLL_CTRL, + codec_master_pll_div[i].reg_val); + schedule_timeout_uninterruptible( + msecs_to_jiffies(20)); + snd_soc_update_bits(codec, + RT5631_GLOBAL_CLK_CTRL, + RT5631_SYSCLK_SOUR_SEL_MASK | + RT5631_PLLCLK_SOUR_SEL_MASK, + RT5631_SYSCLK_SOUR_SEL_PLL | + RT5631_PLLCLK_SOUR_SEL_MCLK); + ret = 0; + break; + } + } else { + for (i = 0; i < ARRAY_SIZE(codec_slave_pll_div); i++) + if (freq_in == codec_slave_pll_div[i].pll_in && + freq_out == codec_slave_pll_div[i].pll_out) { + dev_info(codec->dev, + "change PLL in slave mode\n"); + snd_soc_write(codec, RT5631_PLL_CTRL, + codec_slave_pll_div[i].reg_val); + schedule_timeout_uninterruptible( + msecs_to_jiffies(20)); + snd_soc_update_bits(codec, + RT5631_GLOBAL_CLK_CTRL, + RT5631_SYSCLK_SOUR_SEL_MASK | + RT5631_PLLCLK_SOUR_SEL_MASK, + RT5631_SYSCLK_SOUR_SEL_PLL | + RT5631_PLLCLK_SOUR_SEL_BCLK); + ret = 0; + break; + } + } + + return ret; +} + +static int rt5631_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + snd_soc_update_bits(codec, RT5631_PWR_MANAG_ADD2, + RT5631_PWR_MICBIAS1_VOL | RT5631_PWR_MICBIAS2_VOL, + RT5631_PWR_MICBIAS1_VOL | RT5631_PWR_MICBIAS2_VOL); + break; + + case SND_SOC_BIAS_STANDBY: + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { + snd_soc_update_bits(codec, RT5631_PWR_MANAG_ADD3, + RT5631_PWR_VREF | RT5631_PWR_MAIN_BIAS, + RT5631_PWR_VREF | RT5631_PWR_MAIN_BIAS); + msleep(80); + snd_soc_update_bits(codec, RT5631_PWR_MANAG_ADD3, + RT5631_PWR_FAST_VREF_CTRL, + RT5631_PWR_FAST_VREF_CTRL); + codec->cache_only = false; + snd_soc_cache_sync(codec); + } + break; + + case SND_SOC_BIAS_OFF: + snd_soc_write(codec, RT5631_PWR_MANAG_ADD1, 0x0000); + snd_soc_write(codec, RT5631_PWR_MANAG_ADD2, 0x0000); + snd_soc_write(codec, RT5631_PWR_MANAG_ADD3, 0x0000); + snd_soc_write(codec, RT5631_PWR_MANAG_ADD4, 0x0000); + break; + + default: + break; + } + codec->dapm.bias_level = level; + + return 0; +} + +static int rt5631_probe(struct snd_soc_codec *codec) +{ + struct rt5631_priv *rt5631 = snd_soc_codec_get_drvdata(codec); + unsigned int val; + int ret; + + ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_I2C); + if (ret != 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } + + val = rt5631_read_index(codec, RT5631_ADDA_MIXER_INTL_REG3); + if (val & 0x0002) + rt5631->codec_version = 1; + else + rt5631->codec_version = 0; + + rt5631_reset(codec); + snd_soc_update_bits(codec, RT5631_PWR_MANAG_ADD3, + RT5631_PWR_VREF | RT5631_PWR_MAIN_BIAS, + RT5631_PWR_VREF | RT5631_PWR_MAIN_BIAS); + msleep(80); + snd_soc_update_bits(codec, RT5631_PWR_MANAG_ADD3, + RT5631_PWR_FAST_VREF_CTRL, RT5631_PWR_FAST_VREF_CTRL); + /* enable HP zero cross */ + snd_soc_write(codec, RT5631_INT_ST_IRQ_CTRL_2, 0x0f18); + /* power off ClassD auto Recovery */ + if (rt5631->codec_version) + snd_soc_update_bits(codec, RT5631_INT_ST_IRQ_CTRL_2, + 0x2000, 0x2000); + else + snd_soc_update_bits(codec, RT5631_INT_ST_IRQ_CTRL_2, + 0x2000, 0); + /* DMIC */ + if (rt5631->dmic_used_flag) { + snd_soc_update_bits(codec, RT5631_GPIO_CTRL, + RT5631_GPIO_PIN_FUN_SEL_MASK | + RT5631_GPIO_DMIC_FUN_SEL_MASK, + RT5631_GPIO_PIN_FUN_SEL_GPIO_DIMC | + RT5631_GPIO_DMIC_FUN_SEL_DIMC); + snd_soc_update_bits(codec, RT5631_DIG_MIC_CTRL, + RT5631_DMIC_L_CH_LATCH_MASK | + RT5631_DMIC_R_CH_LATCH_MASK, + RT5631_DMIC_L_CH_LATCH_FALLING | + RT5631_DMIC_R_CH_LATCH_RISING); + } + + codec->dapm.bias_level = SND_SOC_BIAS_STANDBY; + rt5631->codec = codec; + + return 0; +} + +static int rt5631_remove(struct snd_soc_codec *codec) +{ + rt5631_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +#ifdef CONFIG_PM +static int rt5631_suspend(struct snd_soc_codec *codec, pm_message_t state) +{ + rt5631_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int rt5631_resume(struct snd_soc_codec *codec) +{ + rt5631_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + return 0; +} +#else +#define rt5631_suspend NULL +#define rt5631_resume NULL +#endif + +#define RT5631_STEREO_RATES SNDRV_PCM_RATE_8000_96000 +#define RT5631_FORMAT (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S8) + +struct snd_soc_dai_ops rt5631_ops = { + .hw_params = rt5631_hifi_pcm_params, + .set_fmt = rt5631_hifi_codec_set_dai_fmt, + .set_sysclk = rt5631_hifi_codec_set_dai_sysclk, + .set_pll = rt5631_codec_set_dai_pll, +}; + +struct snd_soc_dai_driver rt5631_dai[] = { + { + .name = "rt5631-hifi", + .id = 1, + .playback = { + .stream_name = "HIFI Playback", + .channels_min = 1, + .channels_max = 2, + .rates = RT5631_STEREO_RATES, + .formats = RT5631_FORMAT, + }, + .capture = { + .stream_name = "HIFI Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RT5631_STEREO_RATES, + .formats = RT5631_FORMAT, + }, + .ops = &rt5631_ops, + }, +}; + +static struct snd_soc_codec_driver soc_codec_dev_rt5631 = { + .probe = rt5631_probe, + .remove = rt5631_remove, + .suspend = rt5631_suspend, + .resume = rt5631_resume, + .set_bias_level = rt5631_set_bias_level, + .reg_cache_size = RT5631_VENDOR_ID2 + 1, + .reg_word_size = sizeof(u16), + .reg_cache_default = rt5631_reg, + .volatile_register = rt5631_volatile_register, + .readable_register = rt5631_readable_register, + .reg_cache_step = 1, + .controls = rt5631_snd_controls, + .num_controls = ARRAY_SIZE(rt5631_snd_controls), + .dapm_widgets = rt5631_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(rt5631_dapm_widgets), + .dapm_routes = rt5631_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(rt5631_dapm_routes), +}; + +static const struct i2c_device_id rt5631_i2c_id[] = { + { "rt5631", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, rt5631_i2c_id); + +static int rt5631_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct rt5631_priv *rt5631; + int ret; + + rt5631 = kzalloc(sizeof(struct rt5631_priv), GFP_KERNEL); + if (NULL == rt5631) + return -ENOMEM; + + i2c_set_clientdata(i2c, rt5631); + + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5631, + rt5631_dai, ARRAY_SIZE(rt5631_dai)); + if (ret < 0) + kfree(rt5631); + + return ret; +} + +static __devexit int rt5631_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + kfree(i2c_get_clientdata(client)); + return 0; +} + +struct i2c_driver rt5631_i2c_driver = { + .driver = { + .name = "rt5631", + .owner = THIS_MODULE, + }, + .probe = rt5631_i2c_probe, + .remove = __devexit_p(rt5631_i2c_remove), + .id_table = rt5631_i2c_id, +}; + +static int __init rt5631_modinit(void) +{ + return i2c_add_driver(&rt5631_i2c_driver); +} +module_init(rt5631_modinit); + +static void __exit rt5631_modexit(void) +{ + i2c_del_driver(&rt5631_i2c_driver); +} +module_exit(rt5631_modexit); + +MODULE_DESCRIPTION("ASoC RT5631 driver"); +MODULE_AUTHOR("flove "); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/rt5631.h b/sound/soc/codecs/rt5631.h new file mode 100644 index 0000000..1340158 --- /dev/null +++ b/sound/soc/codecs/rt5631.h @@ -0,0 +1,701 @@ +#ifndef __RTCODEC5631_H__ +#define __RTCODEC5631_H__ + + +#define RT5631_RESET 0x00 +#define RT5631_SPK_OUT_VOL 0x02 +#define RT5631_HP_OUT_VOL 0x04 +#define RT5631_MONO_AXO_1_2_VOL 0x06 +#define RT5631_AUX_IN_VOL 0x0A +#define RT5631_STEREO_DAC_VOL_1 0x0C +#define RT5631_MIC_CTRL_1 0x0E +#define RT5631_STEREO_DAC_VOL_2 0x10 +#define RT5631_ADC_CTRL_1 0x12 +#define RT5631_ADC_REC_MIXER 0x14 +#define RT5631_ADC_CTRL_2 0x16 +#define RT5631_VDAC_DIG_VOL 0x18 +#define RT5631_OUTMIXER_L_CTRL 0x1A +#define RT5631_OUTMIXER_R_CTRL 0x1C +#define RT5631_AXO1MIXER_CTRL 0x1E +#define RT5631_AXO2MIXER_CTRL 0x20 +#define RT5631_MIC_CTRL_2 0x22 +#define RT5631_DIG_MIC_CTRL 0x24 +#define RT5631_MONO_INPUT_VOL 0x26 +#define RT5631_SPK_MIXER_CTRL 0x28 +#define RT5631_SPK_MONO_OUT_CTRL 0x2A +#define RT5631_SPK_MONO_HP_OUT_CTRL 0x2C +#define RT5631_SDP_CTRL 0x34 +#define RT5631_MONO_SDP_CTRL 0x36 +#define RT5631_STEREO_AD_DA_CLK_CTRL 0x38 +#define RT5631_PWR_MANAG_ADD1 0x3A +#define RT5631_PWR_MANAG_ADD2 0x3B +#define RT5631_PWR_MANAG_ADD3 0x3C +#define RT5631_PWR_MANAG_ADD4 0x3E +#define RT5631_GEN_PUR_CTRL_REG 0x40 +#define RT5631_GLOBAL_CLK_CTRL 0x42 +#define RT5631_PLL_CTRL 0x44 +#define RT5631_INT_ST_IRQ_CTRL_1 0x48 +#define RT5631_INT_ST_IRQ_CTRL_2 0x4A +#define RT5631_GPIO_CTRL 0x4C +#define RT5631_MISC_CTRL 0x52 +#define RT5631_DEPOP_FUN_CTRL_1 0x54 +#define RT5631_DEPOP_FUN_CTRL_2 0x56 +#define RT5631_JACK_DET_CTRL 0x5A +#define RT5631_SOFT_VOL_CTRL 0x5C +#define RT5631_ALC_CTRL_1 0x64 +#define RT5631_ALC_CTRL_2 0x65 +#define RT5631_ALC_CTRL_3 0x66 +#define RT5631_PSEUDO_SPATL_CTRL 0x68 +#define RT5631_INDEX_ADD 0x6A +#define RT5631_INDEX_DATA 0x6C +#define RT5631_EQ_CTRL 0x6E +#define RT5631_VENDOR_ID 0x7A +#define RT5631_VENDOR_ID1 0x7C +#define RT5631_VENDOR_ID2 0x7E + +/* Index of Codec Private Register definition */ +#define RT5631_EQ_BW_LOP 0x00 +#define RT5631_EQ_GAIN_LOP 0x01 +#define RT5631_EQ_FC_BP1 0x02 +#define RT5631_EQ_BW_BP1 0x03 +#define RT5631_EQ_GAIN_BP1 0x04 +#define RT5631_EQ_FC_BP2 0x05 +#define RT5631_EQ_BW_BP2 0x06 +#define RT5631_EQ_GAIN_BP2 0x07 +#define RT5631_EQ_FC_BP3 0x08 +#define RT5631_EQ_BW_BP3 0x09 +#define RT5631_EQ_GAIN_BP3 0x0a +#define RT5631_EQ_BW_HIP 0x0b +#define RT5631_EQ_GAIN_HIP 0x0c +#define RT5631_EQ_HPF_A1 0x0d +#define RT5631_EQ_HPF_A2 0x0e +#define RT5631_EQ_HPF_GAIN 0x0f +#define RT5631_EQ_PRE_VOL_CTRL 0x11 +#define RT5631_EQ_POST_VOL_CTRL 0x12 +#define RT5631_TEST_MODE_CTRL 0x39 +#define RT5631_CP_INTL_REG2 0x45 +#define RT5631_ADDA_MIXER_INTL_REG3 0x52 +#define RT5631_SPK_INTL_CTRL 0x56 + + +/* global definition */ +#define RT5631_L_MUTE (0x1 << 15) +#define RT5631_L_MUTE_SHIFT 15 +#define RT5631_L_EN (0x1 << 14) +#define RT5631_L_EN_SHIFT 14 +#define RT5631_R_MUTE (0x1 << 7) +#define RT5631_R_MUTE_SHIFT 7 +#define RT5631_R_EN (0x1 << 6) +#define RT5631_R_EN_SHIFT 6 +#define RT5631_VOL_MASK 0x1f +#define RT5631_L_VOL_SHIFT 8 +#define RT5631_R_VOL_SHIFT 0 + +/* Speaker Output Control(0x02) */ +#define RT5631_SPK_L_VOL_SEL_MASK (0x1 << 14) +#define RT5631_SPK_L_VOL_SEL_VMID (0x0 << 14) +#define RT5631_SPK_L_VOL_SEL_SPKMIX_L (0x1 << 14) +#define RT5631_SPK_R_VOL_SEL_MASK (0x1 << 6) +#define RT5631_SPK_R_VOL_SEL_VMID (0x0 << 6) +#define RT5631_SPK_R_VOL_SEL_SPKMIX_R (0x1 << 6) + +/* Headphone Output Control(0x04) */ +#define RT5631_HP_L_VOL_SEL_MASK (0x1 << 14) +#define RT5631_HP_L_VOL_SEL_VMID (0x0 << 14) +#define RT5631_HP_L_VOL_SEL_OUTMIX_L (0x1 << 14) +#define RT5631_HP_R_VOL_SEL_MASK (0x1 << 6) +#define RT5631_HP_R_VOL_SEL_VMID (0x0 << 6) +#define RT5631_HP_R_VOL_SEL_OUTMIX_R (0x1 << 6) + +/* Output Control for AUXOUT/MONO(0x06) */ +#define RT5631_AUXOUT_1_VOL_SEL_MASK (0x1 << 14) +#define RT5631_AUXOUT_1_VOL_SEL_VMID (0x0 << 14) +#define RT5631_AUXOUT_1_VOL_SEL_OUTMIX_L (0x1 << 14) +#define RT5631_MUTE_MONO (0x1 << 13) +#define RT5631_MUTE_MONO_SHIFT 13 +#define RT5631_AUXOUT_2_VOL_SEL_MASK (0x1 << 6) +#define RT5631_AUXOUT_2_VOL_SEL_VMID (0x0 << 6) +#define RT5631_AUXOUT_2_VOL_SEL_OUTMIX_R (0x1 << 6) + +/* Microphone Input Control 1(0x0E) */ +#define RT5631_MIC1_DIFF_INPUT_CTRL (0x1 << 15) +#define RT5631_MIC1_DIFF_INPUT_SHIFT 15 +#define RT5631_MIC2_DIFF_INPUT_CTRL (0x1 << 7) +#define RT5631_MIC2_DIFF_INPUT_SHIFT 7 + +/* Stereo DAC Digital Volume2(0x10) */ +#define RT5631_DAC_VOL_MASK 0xff + +/* ADC Recording Mixer Control(0x14) */ +#define RT5631_M_OUTMIXER_L_TO_RECMIXER_L (0x1 << 15) +#define RT5631_M_OUTMIXL_RECMIXL_BIT 15 +#define RT5631_M_MIC1_TO_RECMIXER_L (0x1 << 14) +#define RT5631_M_MIC1_RECMIXL_BIT 14 +#define RT5631_M_AXIL_TO_RECMIXER_L (0x1 << 13) +#define RT5631_M_AXIL_RECMIXL_BIT 13 +#define RT5631_M_MONO_IN_TO_RECMIXER_L (0x1 << 12) +#define RT5631_M_MONO_IN_RECMIXL_BIT 12 +#define RT5631_M_OUTMIXER_R_TO_RECMIXER_R (0x1 << 7) +#define RT5631_M_OUTMIXR_RECMIXR_BIT 7 +#define RT5631_M_MIC2_TO_RECMIXER_R (0x1 << 6) +#define RT5631_M_MIC2_RECMIXR_BIT 6 +#define RT5631_M_AXIR_TO_RECMIXER_R (0x1 << 5) +#define RT5631_M_AXIR_RECMIXR_BIT 5 +#define RT5631_M_MONO_IN_TO_RECMIXER_R (0x1 << 4) +#define RT5631_M_MONO_IN_RECMIXR_BIT 4 + +/* Left Output Mixer Control(0x1A) */ +#define RT5631_M_RECMIXER_L_TO_OUTMIXER_L (0x1 << 15) +#define RT5631_M_RECMIXL_OUTMIXL_BIT 15 +#define RT5631_M_RECMIXER_R_TO_OUTMIXER_L (0x1 << 14) +#define RT5631_M_RECMIXR_OUTMIXL_BIT 14 +#define RT5631_M_DAC_L_TO_OUTMIXER_L (0x1 << 13) +#define RT5631_M_DACL_OUTMIXL_BIT 13 +#define RT5631_M_MIC1_TO_OUTMIXER_L (0x1 << 12) +#define RT5631_M_MIC1_OUTMIXL_BIT 12 +#define RT5631_M_MIC2_TO_OUTMIXER_L (0x1 << 11) +#define RT5631_M_MIC2_OUTMIXL_BIT 11 +#define RT5631_M_MONO_IN_P_TO_OUTMIXER_L (0x1 << 10) +#define RT5631_M_MONO_INP_OUTMIXL_BIT 10 +#define RT5631_M_AXIL_TO_OUTMIXER_L (0x1 << 9) +#define RT5631_M_AXIL_OUTMIXL_BIT 9 +#define RT5631_M_AXIR_TO_OUTMIXER_L (0x1 << 8) +#define RT5631_M_AXIR_OUTMIXL_BIT 8 +#define RT5631_M_VDAC_TO_OUTMIXER_L (0x1 << 7) +#define RT5631_M_VDAC_OUTMIXL_BIT 7 + +/* Right Output Mixer Control(0x1C) */ +#define RT5631_M_RECMIXER_L_TO_OUTMIXER_R (0x1 << 15) +#define RT5631_M_RECMIXL_OUTMIXR_BIT 15 +#define RT5631_M_RECMIXER_R_TO_OUTMIXER_R (0x1 << 14) +#define RT5631_M_RECMIXR_OUTMIXR_BIT 14 +#define RT5631_M_DAC_R_TO_OUTMIXER_R (0x1 << 13) +#define RT5631_M_DACR_OUTMIXR_BIT 13 +#define RT5631_M_MIC1_TO_OUTMIXER_R (0x1 << 12) +#define RT5631_M_MIC1_OUTMIXR_BIT 12 +#define RT5631_M_MIC2_TO_OUTMIXER_R (0x1 << 11) +#define RT5631_M_MIC2_OUTMIXR_BIT 11 +#define RT5631_M_MONO_IN_N_TO_OUTMIXER_R (0x1 << 10) +#define RT5631_M_MONO_INN_OUTMIXR_BIT 10 +#define RT5631_M_AXIL_TO_OUTMIXER_R (0x1 << 9) +#define RT5631_M_AXIL_OUTMIXR_BIT 9 +#define RT5631_M_AXIR_TO_OUTMIXER_R (0x1 << 8) +#define RT5631_M_AXIR_OUTMIXR_BIT 8 +#define RT5631_M_VDAC_TO_OUTMIXER_R (0x1 << 7) +#define RT5631_M_VDAC_OUTMIXR_BIT 7 + +/* Lout Mixer Control(0x1E) */ +#define RT5631_M_MIC1_TO_AXO1MIXER (0x1 << 15) +#define RT5631_M_MIC1_AXO1MIX_BIT 15 +#define RT5631_M_MIC2_TO_AXO1MIXER (0x1 << 11) +#define RT5631_M_MIC2_AXO1MIX_BIT 11 +#define RT5631_M_OUTMIXER_L_TO_AXO1MIXER (0x1 << 7) +#define RT5631_M_OUTMIXL_AXO1MIX_BIT 7 +#define RT5631_M_OUTMIXER_R_TO_AXO1MIXER (0x1 << 6) +#define RT5631_M_OUTMIXR_AXO1MIX_BIT 6 + +/* Rout Mixer Control(0x20) */ +#define RT5631_M_MIC1_TO_AXO2MIXER (0x1 << 15) +#define RT5631_M_MIC1_AXO2MIX_BIT 15 +#define RT5631_M_MIC2_TO_AXO2MIXER (0x1 << 11) +#define RT5631_M_MIC2_AXO2MIX_BIT 11 +#define RT5631_M_OUTMIXER_L_TO_AXO2MIXER (0x1 << 7) +#define RT5631_M_OUTMIXL_AXO2MIX_BIT 7 +#define RT5631_M_OUTMIXER_R_TO_AXO2MIXER (0x1 << 6) +#define RT5631_M_OUTMIXR_AXO2MIX_BIT 6 + +/* Micphone Input Control 2(0x22) */ +#define RT5631_MIC_BIAS_90_PRECNET_AVDD 1 +#define RT5631_MIC_BIAS_75_PRECNET_AVDD 2 + +#define RT5631_MIC1_BOOST_CTRL_MASK (0xf << 12) +#define RT5631_MIC1_BOOST_CTRL_BYPASS (0x0 << 12) +#define RT5631_MIC1_BOOST_CTRL_20DB (0x1 << 12) +#define RT5631_MIC1_BOOST_CTRL_24DB (0x2 << 12) +#define RT5631_MIC1_BOOST_CTRL_30DB (0x3 << 12) +#define RT5631_MIC1_BOOST_CTRL_35DB (0x4 << 12) +#define RT5631_MIC1_BOOST_CTRL_40DB (0x5 << 12) +#define RT5631_MIC1_BOOST_CTRL_34DB (0x6 << 12) +#define RT5631_MIC1_BOOST_CTRL_50DB (0x7 << 12) +#define RT5631_MIC1_BOOST_CTRL_52DB (0x8 << 12) +#define RT5631_MIC1_BOOST_SHIFT 12 + +#define RT5631_MIC2_BOOST_CTRL_MASK (0xf << 8) +#define RT5631_MIC2_BOOST_CTRL_BYPASS (0x0 << 8) +#define RT5631_MIC2_BOOST_CTRL_20DB (0x1 << 8) +#define RT5631_MIC2_BOOST_CTRL_24DB (0x2 << 8) +#define RT5631_MIC2_BOOST_CTRL_30DB (0x3 << 8) +#define RT5631_MIC2_BOOST_CTRL_35DB (0x4 << 8) +#define RT5631_MIC2_BOOST_CTRL_40DB (0x5 << 8) +#define RT5631_MIC2_BOOST_CTRL_34DB (0x6 << 8) +#define RT5631_MIC2_BOOST_CTRL_50DB (0x7 << 8) +#define RT5631_MIC2_BOOST_CTRL_52DB (0x8 << 8) +#define RT5631_MIC2_BOOST_SHIFT 8 + +#define RT5631_MICBIAS1_VOLT_CTRL_MASK (0x1 << 7) +#define RT5631_MICBIAS1_VOLT_CTRL_90P (0x0 << 7) +#define RT5631_MICBIAS1_VOLT_CTRL_75P (0x1 << 7) + +#define RT5631_MICBIAS1_S_C_DET_MASK (0x1 << 6) +#define RT5631_MICBIAS1_S_C_DET_DIS (0x0 << 6) +#define RT5631_MICBIAS1_S_C_DET_ENA (0x1 << 6) + +#define RT5631_MICBIAS1_SHORT_CURR_DET_MASK (0x3 << 4) +#define RT5631_MICBIAS1_SHORT_CURR_DET_600UA (0x0 << 4) +#define RT5631_MICBIAS1_SHORT_CURR_DET_1500UA (0x1 << 4) +#define RT5631_MICBIAS1_SHORT_CURR_DET_2000UA (0x2 << 4) + +#define RT5631_MICBIAS2_VOLT_CTRL_MASK (0x1 << 3) +#define RT5631_MICBIAS2_VOLT_CTRL_90P (0x0 << 3) +#define RT5631_MICBIAS2_VOLT_CTRL_75P (0x1 << 3) + +#define RT5631_MICBIAS2_S_C_DET_MASK (0x1 << 2) +#define RT5631_MICBIAS2_S_C_DET_DIS (0x0 << 2) +#define RT5631_MICBIAS2_S_C_DET_ENA (0x1 << 2) + +#define RT5631_MICBIAS2_SHORT_CURR_DET_MASK (0x3) +#define RT5631_MICBIAS2_SHORT_CURR_DET_600UA (0x0) +#define RT5631_MICBIAS2_SHORT_CURR_DET_1500UA (0x1) +#define RT5631_MICBIAS2_SHORT_CURR_DET_2000UA (0x2) + + +/* Digital Microphone Control(0x24) */ +#define RT5631_DMIC_ENA_MASK (0x1 << 15) +#define RT5631_DMIC_ENA_SHIFT 15 +/* DMIC_ENA: DMIC to ADC Digital filter */ +#define RT5631_DMIC_ENA (0x1 << 15) +/* DMIC_DIS: ADC mixer to ADC Digital filter */ +#define RT5631_DMIC_DIS (0x0 << 15) +#define RT5631_DMIC_L_CH_MUTE (0x1 << 13) +#define RT5631_DMIC_L_CH_MUTE_SHIFT 13 +#define RT5631_DMIC_R_CH_MUTE (0x1 << 12) +#define RT5631_DMIC_R_CH_MUTE_SHIFT 12 +#define RT5631_DMIC_L_CH_LATCH_MASK (0x1 << 9) +#define RT5631_DMIC_L_CH_LATCH_RISING (0x1 << 9) +#define RT5631_DMIC_L_CH_LATCH_FALLING (0x0 << 9) +#define RT5631_DMIC_R_CH_LATCH_MASK (0x1 << 8) +#define RT5631_DMIC_R_CH_LATCH_RISING (0x1 << 8) +#define RT5631_DMIC_R_CH_LATCH_FALLING (0x0 << 8) +#define RT5631_DMIC_CLK_CTRL_MASK (0x3 << 4) +#define RT5631_DMIC_CLK_CTRL_TO_128FS (0x0 << 4) +#define RT5631_DMIC_CLK_CTRL_TO_64FS (0x1 << 4) +#define RT5631_DMIC_CLK_CTRL_TO_32FS (0x2 << 4) + +/* Microphone Input Volume(0x26) */ +#define RT5631_MONO_DIFF_INPUT_SHIFT 15 + +/* Speaker Mixer Control(0x28) */ +#define RT5631_M_RECMIXER_L_TO_SPKMIXER_L (0x1 << 15) +#define RT5631_M_RECMIXL_SPKMIXL_BIT 15 +#define RT5631_M_MIC1_P_TO_SPKMIXER_L (0x1 << 14) +#define RT5631_M_MIC1P_SPKMIXL_BIT 14 +#define RT5631_M_DAC_L_TO_SPKMIXER_L (0x1 << 13) +#define RT5631_M_DACL_SPKMIXL_BIT 13 +#define RT5631_M_OUTMIXER_L_TO_SPKMIXER_L (0x1 << 12) +#define RT5631_M_OUTMIXL_SPKMIXL_BIT 12 + +#define RT5631_M_RECMIXER_R_TO_SPKMIXER_R (0x1 << 7) +#define RT5631_M_RECMIXR_SPKMIXR_BIT 7 +#define RT5631_M_MIC2_P_TO_SPKMIXER_R (0x1 << 6) +#define RT5631_M_MIC2P_SPKMIXR_BIT 6 +#define RT5631_M_DAC_R_TO_SPKMIXER_R (0x1 << 5) +#define RT5631_M_DACR_SPKMIXR_BIT 5 +#define RT5631_M_OUTMIXER_R_TO_SPKMIXER_R (0x1 << 4) +#define RT5631_M_OUTMIXR_SPKMIXR_BIT 4 + +/* Speaker/Mono Output Control(0x2A) */ +#define RT5631_M_SPKVOL_L_TO_SPOL_MIXER (0x1 << 15) +#define RT5631_M_SPKVOLL_SPOLMIX_BIT 15 +#define RT5631_M_SPKVOL_R_TO_SPOL_MIXER (0x1 << 14) +#define RT5631_M_SPKVOLR_SPOLMIX_BIT 14 +#define RT5631_M_SPKVOL_L_TO_SPOR_MIXER (0x1 << 13) +#define RT5631_M_SPKVOLL_SPORMIX_BIT 13 +#define RT5631_M_SPKVOL_R_TO_SPOR_MIXER (0x1 << 12) +#define RT5631_M_SPKVOLR_SPORMIX_BIT 12 +#define RT5631_M_OUTVOL_L_TO_MONOMIXER (0x1 << 11) +#define RT5631_M_OUTVOLL_MONOMIX_BIT 11 +#define RT5631_M_OUTVOL_R_TO_MONOMIXER (0x1 << 10) +#define RT5631_M_OUTVOLR_MONOMIX_BIT 10 + +/* Speaker/Mono/HP Output Control(0x2C) */ +#define RT5631_SPK_L_MUX_SEL_MASK (0x3 << 14) +#define RT5631_SPK_L_MUX_SEL_SPKMIXER_L (0x0 << 14) +#define RT5631_SPK_L_MUX_SEL_MONO_IN (0x1 << 14) +#define RT5631_SPK_L_MUX_SEL_DAC_L (0x3 << 14) +#define RT5631_SPK_L_MUX_SEL_SHIFT 14 + +#define RT5631_SPK_R_MUX_SEL_MASK (0x3 << 10) +#define RT5631_SPK_R_MUX_SEL_SPKMIXER_R (0x0 << 10) +#define RT5631_SPK_R_MUX_SEL_MONO_IN (0x1 << 10) +#define RT5631_SPK_R_MUX_SEL_DAC_R (0x3 << 10) +#define RT5631_SPK_R_MUX_SEL_SHIFT 10 + +#define RT5631_MONO_MUX_SEL_MASK (0x3 << 6) +#define RT5631_MONO_MUX_SEL_MONOMIXER (0x0 << 6) +#define RT5631_MONO_MUX_SEL_MONO_IN (0x1 << 6) +#define RT5631_MONO_MUX_SEL_SHIFT 6 + +#define RT5631_HP_L_MUX_SEL_MASK (0x1 << 3) +#define RT5631_HP_L_MUX_SEL_HPVOL_L (0x0 << 3) +#define RT5631_HP_L_MUX_SEL_DAC_L (0x1 << 3) +#define RT5631_HP_L_MUX_SEL_SHIFT 3 + +#define RT5631_HP_R_MUX_SEL_MASK (0x1 << 2) +#define RT5631_HP_R_MUX_SEL_HPVOL_R (0x0 << 2) +#define RT5631_HP_R_MUX_SEL_DAC_R (0x1 << 2) +#define RT5631_HP_R_MUX_SEL_SHIFT 2 + +/* Stereo I2S Serial Data Port Control(0x34) */ +#define RT5631_SDP_MODE_SEL_MASK (0x1 << 15) +#define RT5631_SDP_MODE_SEL_MASTER (0x0 << 15) +#define RT5631_SDP_MODE_SEL_SLAVE (0x1 << 15) + +#define RT5631_SDP_ADC_CPS_SEL_MASK (0x3 << 10) +#define RT5631_SDP_ADC_CPS_SEL_OFF (0x0 << 10) +#define RT5631_SDP_ADC_CPS_SEL_U_LAW (0x1 << 10) +#define RT5631_SDP_ADC_CPS_SEL_A_LAW (0x2 << 10) + +#define RT5631_SDP_DAC_CPS_SEL_MASK (0x3 << 8) +#define RT5631_SDP_DAC_CPS_SEL_OFF (0x0 << 8) +#define RT5631_SDP_DAC_CPS_SEL_U_LAW (0x1 << 8) +#define RT5631_SDP_DAC_CPS_SEL_A_LAW (0x2 << 8) +/* 0:Normal 1:Invert */ +#define RT5631_SDP_I2S_BCLK_POL_CTRL (0x1 << 7) +/* 0:Normal 1:Invert */ +#define RT5631_SDP_DAC_R_INV (0x1 << 6) +/* 0:ADC data appear at left phase of LRCK + * 1:ADC data appear at right phase of LRCK + */ +#define RT5631_SDP_ADC_DATA_L_R_SWAP (0x1 << 5) +/* 0:DAC data appear at left phase of LRCK + * 1:DAC data appear at right phase of LRCK + */ +#define RT5631_SDP_DAC_DATA_L_R_SWAP (0x1 << 4) + +/* Data Length Slection */ +#define RT5631_SDP_I2S_DL_MASK (0x3 << 2) +#define RT5631_SDP_I2S_DL_16 (0x0 << 2) +#define RT5631_SDP_I2S_DL_20 (0x1 << 2) +#define RT5631_SDP_I2S_DL_24 (0x2 << 2) +#define RT5631_SDP_I2S_DL_8 (0x3 << 2) + +/* PCM Data Format Selection */ +#define RT5631_SDP_I2S_DF_MASK (0x3) +#define RT5631_SDP_I2S_DF_I2S (0x0) +#define RT5631_SDP_I2S_DF_LEFT (0x1) +#define RT5631_SDP_I2S_DF_PCM_A (0x2) +#define RT5631_SDP_I2S_DF_PCM_B (0x3) + +/* Stereo AD/DA Clock Control(0x38h) */ +#define RT5631_I2S_PRE_DIV_MASK (0x7 << 13) +#define RT5631_I2S_PRE_DIV_1 (0x0 << 13) +#define RT5631_I2S_PRE_DIV_2 (0x1 << 13) +#define RT5631_I2S_PRE_DIV_4 (0x2 << 13) +#define RT5631_I2S_PRE_DIV_8 (0x3 << 13) +#define RT5631_I2S_PRE_DIV_16 (0x4 << 13) +#define RT5631_I2S_PRE_DIV_32 (0x5 << 13) +/* CLOCK RELATIVE OF BCLK AND LCRK */ +#define RT5631_I2S_LRCK_SEL_N_BCLK_MASK (0x1 << 12) +#define RT5631_I2S_LRCK_SEL_64_BCLK (0x0 << 12) /* 64FS */ +#define RT5631_I2S_LRCK_SEL_32_BCLK (0x1 << 12) /* 32FS */ + +#define RT5631_DAC_OSR_SEL_MASK (0x3 << 10) +#define RT5631_DAC_OSR_SEL_128FS (0x3 << 10) +#define RT5631_DAC_OSR_SEL_64FS (0x3 << 10) +#define RT5631_DAC_OSR_SEL_32FS (0x3 << 10) +#define RT5631_DAC_OSR_SEL_16FS (0x3 << 10) + +#define RT5631_ADC_OSR_SEL_MASK (0x3 << 8) +#define RT5631_ADC_OSR_SEL_128FS (0x3 << 8) +#define RT5631_ADC_OSR_SEL_64FS (0x3 << 8) +#define RT5631_ADC_OSR_SEL_32FS (0x3 << 8) +#define RT5631_ADC_OSR_SEL_16FS (0x3 << 8) + +#define RT5631_ADDA_FILTER_CLK_SEL_256FS (0 << 7) /* 256FS */ +#define RT5631_ADDA_FILTER_CLK_SEL_384FS (1 << 7) /* 384FS */ + +/* Power managment addition 1 (0x3A) */ +#define RT5631_PWR_MAIN_I2S_EN (0x1 << 15) +#define RT5631_PWR_MAIN_I2S_BIT 15 +#define RT5631_PWR_CLASS_D (0x1 << 12) +#define RT5631_PWR_CLASS_D_BIT 12 +#define RT5631_PWR_ADC_L_CLK (0x1 << 11) +#define RT5631_PWR_ADC_L_CLK_BIT 11 +#define RT5631_PWR_ADC_R_CLK (0x1 << 10) +#define RT5631_PWR_ADC_R_CLK_BIT 10 +#define RT5631_PWR_DAC_L_CLK (0x1 << 9) +#define RT5631_PWR_DAC_L_CLK_BIT 9 +#define RT5631_PWR_DAC_R_CLK (0x1 << 8) +#define RT5631_PWR_DAC_R_CLK_BIT 8 +#define RT5631_PWR_DAC_REF (0x1 << 7) +#define RT5631_PWR_DAC_REF_BIT 7 +#define RT5631_PWR_DAC_L_TO_MIXER (0x1 << 6) +#define RT5631_PWR_DAC_L_TO_MIXER_BIT 6 +#define RT5631_PWR_DAC_R_TO_MIXER (0x1 << 5) +#define RT5631_PWR_DAC_R_TO_MIXER_BIT 5 + +/* Power managment addition 2 (0x3B) */ +#define RT5631_PWR_OUTMIXER_L (0x1 << 15) +#define RT5631_PWR_OUTMIXER_L_BIT 15 +#define RT5631_PWR_OUTMIXER_R (0x1 << 14) +#define RT5631_PWR_OUTMIXER_R_BIT 14 +#define RT5631_PWR_SPKMIXER_L (0x1 << 13) +#define RT5631_PWR_SPKMIXER_L_BIT 13 +#define RT5631_PWR_SPKMIXER_R (0x1 << 12) +#define RT5631_PWR_SPKMIXER_R_BIT 12 +#define RT5631_PWR_RECMIXER_L (0x1 << 11) +#define RT5631_PWR_RECMIXER_L_BIT 11 +#define RT5631_PWR_RECMIXER_R (0x1 << 10) +#define RT5631_PWR_RECMIXER_R_BIT 10 +#define RT5631_PWR_MIC1_BOOT_GAIN (0x1 << 5) +#define RT5631_PWR_MIC1_BOOT_GAIN_BIT 5 +#define RT5631_PWR_MIC2_BOOT_GAIN (0x1 << 4) +#define RT5631_PWR_MIC2_BOOT_GAIN_BIT 4 +#define RT5631_PWR_MICBIAS1_VOL (0x1 << 3) +#define RT5631_PWR_MICBIAS1_VOL_BIT 3 +#define RT5631_PWR_MICBIAS2_VOL (0x1 << 2) +#define RT5631_PWR_MICBIAS2_VOL_BIT 2 +#define RT5631_PWR_PLL1 (0x1 << 1) +#define RT5631_PWR_PLL1_BIT 1 +#define RT5631_PWR_PLL2 (0x1 << 0) +#define RT5631_PWR_PLL2_BIT 0 + +/* Power managment addition 3(0x3C) */ +#define RT5631_PWR_VREF (0x1 << 15) +#define RT5631_PWR_VREF_BIT 15 +#define RT5631_PWR_FAST_VREF_CTRL (0x1 << 14) +#define RT5631_PWR_FAST_VREF_CTRL_BIT 14 +#define RT5631_PWR_MAIN_BIAS (0x1 << 13) +#define RT5631_PWR_MAIN_BIAS_BIT 13 +#define RT5631_PWR_AXO1MIXER (0x1 << 11) +#define RT5631_PWR_AXO1MIXER_BIT 11 +#define RT5631_PWR_AXO2MIXER (0x1 << 10) +#define RT5631_PWR_AXO2MIXER_BIT 10 +#define RT5631_PWR_MONOMIXER (0x1 << 9) +#define RT5631_PWR_MONOMIXER_BIT 9 +#define RT5631_PWR_MONO_DEPOP_DIS (0x1 << 8) +#define RT5631_PWR_MONO_DEPOP_DIS_BIT 8 +#define RT5631_PWR_MONO_AMP_EN (0x1 << 7) +#define RT5631_PWR_MONO_AMP_EN_BIT 7 +#define RT5631_PWR_CHARGE_PUMP (0x1 << 4) +#define RT5631_PWR_CHARGE_PUMP_BIT 4 +#define RT5631_PWR_HP_L_AMP (0x1 << 3) +#define RT5631_PWR_HP_L_AMP_BIT 3 +#define RT5631_PWR_HP_R_AMP (0x1 << 2) +#define RT5631_PWR_HP_R_AMP_BIT 2 +#define RT5631_PWR_HP_DEPOP_DIS (0x1 << 1) +#define RT5631_PWR_HP_DEPOP_DIS_BIT 1 +#define RT5631_PWR_HP_AMP_DRIVING (0x1 << 0) +#define RT5631_PWR_HP_AMP_DRIVING_BIT 0 + +/* Power managment addition 4(0x3E) */ +#define RT5631_PWR_SPK_L_VOL (0x1 << 15) +#define RT5631_PWR_SPK_L_VOL_BIT 15 +#define RT5631_PWR_SPK_R_VOL (0x1 << 14) +#define RT5631_PWR_SPK_R_VOL_BIT 14 +#define RT5631_PWR_LOUT_VOL (0x1 << 13) +#define RT5631_PWR_LOUT_VOL_BIT 13 +#define RT5631_PWR_ROUT_VOL (0x1 << 12) +#define RT5631_PWR_ROUT_VOL_BIT 12 +#define RT5631_PWR_HP_L_OUT_VOL (0x1 << 11) +#define RT5631_PWR_HP_L_OUT_VOL_BIT 11 +#define RT5631_PWR_HP_R_OUT_VOL (0x1 << 10) +#define RT5631_PWR_HP_R_OUT_VOL_BIT 10 +#define RT5631_PWR_AXIL_IN_VOL (0x1 << 9) +#define RT5631_PWR_AXIL_IN_VOL_BIT 9 +#define RT5631_PWR_AXIR_IN_VOL (0x1 << 8) +#define RT5631_PWR_AXIR_IN_VOL_BIT 8 +#define RT5631_PWR_MONO_IN_P_VOL (0x1 << 7) +#define RT5631_PWR_MONO_IN_P_VOL_BIT 7 +#define RT5631_PWR_MONO_IN_N_VOL (0x1 << 6) +#define RT5631_PWR_MONO_IN_N_VOL_BIT 6 + +/* General Purpose Control Register(0x40) */ +#define RT5631_SPK_AMP_AUTO_RATIO_EN (0x1 << 15) + +#define RT5631_SPK_AMP_RATIO_CTRL_MASK (0x7 << 12) +#define RT5631_SPK_AMP_RATIO_CTRL_2_34 (0x0 << 12) /* 7.40DB */ +#define RT5631_SPK_AMP_RATIO_CTRL_1_99 (0x1 << 12) /* 5.99DB */ +#define RT5631_SPK_AMP_RATIO_CTRL_1_68 (0x2 << 12) /* 4.50DB */ +#define RT5631_SPK_AMP_RATIO_CTRL_1_56 (0x3 << 12) /* 3.86DB */ +#define RT5631_SPK_AMP_RATIO_CTRL_1_44 (0x4 << 12) /* 3.16DB */ +#define RT5631_SPK_AMP_RATIO_CTRL_1_27 (0x5 << 12) /* 2.10DB */ +#define RT5631_SPK_AMP_RATIO_CTRL_1_09 (0x6 << 12) /* 0.80DB */ +#define RT5631_SPK_AMP_RATIO_CTRL_1_00 (0x7 << 12) /* 0.00DB */ +#define RT5631_SPK_AMP_RATIO_CTRL_SHIFT 12 + +#define RT5631_STEREO_DAC_HI_PASS_FILT_EN (0x1 << 11) +#define RT5631_STEREO_ADC_HI_PASS_FILT_EN (0x1 << 10) +/* Select ADC Wind Filter Clock type */ +#define RT5631_ADC_WIND_FILT_MASK (0x3 << 4) +#define RT5631_ADC_WIND_FILT_8_16_32K (0x0 << 4) /*8/16/32k*/ +#define RT5631_ADC_WIND_FILT_11_22_44K (0x1 << 4) /*11/22/44k*/ +#define RT5631_ADC_WIND_FILT_12_24_48K (0x2 << 4) /*12/24/48k*/ +#define RT5631_ADC_WIND_FILT_EN (0x1 << 3) +/* SelectADC Wind Filter Corner Frequency */ +#define RT5631_ADC_WIND_CNR_FREQ_MASK (0x7 << 0) +#define RT5631_ADC_WIND_CNR_FREQ_82_113_122 (0x0 << 0) /* 82/113/122 Hz */ +#define RT5631_ADC_WIND_CNR_FREQ_102_141_153 (0x1 << 0) /* 102/141/153 Hz */ +#define RT5631_ADC_WIND_CNR_FREQ_131_180_156 (0x2 << 0) /* 131/180/156 Hz */ +#define RT5631_ADC_WIND_CNR_FREQ_163_225_245 (0x3 << 0) /* 163/225/245 Hz */ +#define RT5631_ADC_WIND_CNR_FREQ_204_281_306 (0x4 << 0) /* 204/281/306 Hz */ +#define RT5631_ADC_WIND_CNR_FREQ_261_360_392 (0x5 << 0) /* 261/360/392 Hz */ +#define RT5631_ADC_WIND_CNR_FREQ_327_450_490 (0x6 << 0) /* 327/450/490 Hz */ +#define RT5631_ADC_WIND_CNR_FREQ_408_563_612 (0x7 << 0) /* 408/563/612 Hz */ + +/* Global Clock Control Register(0x42) */ +#define RT5631_SYSCLK_SOUR_SEL_MASK (0x3 << 14) +#define RT5631_SYSCLK_SOUR_SEL_MCLK (0x0 << 14) +#define RT5631_SYSCLK_SOUR_SEL_PLL (0x1 << 14) +#define RT5631_SYSCLK_SOUR_SEL_PLL_TCK (0x2 << 14) + +#define RT5631_PLLCLK_SOUR_SEL_MASK (0x3 << 12) +#define RT5631_PLLCLK_SOUR_SEL_MCLK (0x0 << 12) +#define RT5631_PLLCLK_SOUR_SEL_BCLK (0x1 << 12) +#define RT5631_PLLCLK_SOUR_SEL_VBCLK (0x2 << 12) + +#define RT5631_PLLCLK_PRE_DIV1 (0x0 << 11) +#define RT5631_PLLCLK_PRE_DIV2 (0x1 << 11) + +/* PLL Control(0x44) */ +#define RT5631_PLL_CTRL_M_VAL(m) ((m)&0xf) +#define RT5631_PLL_CTRL_K_VAL(k) (((k)&0x7) << 4) +#define RT5631_PLL_CTRL_N_VAL(n) (((n)&0xff) << 8) + +/* Internal Status and IRQ Control2(0x4A) */ +#define RT5631_ADC_DATA_SEL_MASK (0x3 << 14) +#define RT5631_ADC_DATA_SEL_Disable (0x0 << 14) +#define RT5631_ADC_DATA_SEL_MIC1 (0x1 << 14) +#define RT5631_ADC_DATA_SEL_MIC1_SHIFT 14 +#define RT5631_ADC_DATA_SEL_MIC2 (0x2 << 14) +#define RT5631_ADC_DATA_SEL_MIC2_SHIFT 15 +#define RT5631_ADC_DATA_SEL_STO (0x3 << 14) +#define RT5631_ADC_DATA_SEL_SHIFT 14 + +/* GPIO Pin Configuration(0x4C) */ +#define RT5631_GPIO_PIN_FUN_SEL_MASK (0x1 << 15) +#define RT5631_GPIO_PIN_FUN_SEL_IRQ (0x1 << 15) +#define RT5631_GPIO_PIN_FUN_SEL_GPIO_DIMC (0x0 << 15) + +#define RT5631_GPIO_DMIC_FUN_SEL_MASK (0x1 << 3) +#define RT5631_GPIO_DMIC_FUN_SEL_DIMC (0x1 << 3) +#define RT5631_GPIO_DMIC_FUN_SEL_GPIO (0x0 << 3) + +#define RT5631_GPIO_PIN_CON_MASK (0x1 << 2) +#define RT5631_GPIO_PIN_SET_INPUT (0x0 << 2) +#define RT5631_GPIO_PIN_SET_OUTPUT (0x1 << 2) + +/* De-POP function Control 1(0x54) */ +#define RT5631_POW_ON_SOFT_GEN (0x1 << 15) +#define RT5631_EN_MUTE_UNMUTE_DEPOP (0x1 << 14) +#define RT5631_EN_DEPOP2_FOR_HP (0x1 << 7) +/* Power Down HPAMP_L Starts Up Signal */ +#define RT5631_PD_HPAMP_L_ST_UP (0x1 << 5) +/* Power Down HPAMP_R Starts Up Signal */ +#define RT5631_PD_HPAMP_R_ST_UP (0x1 << 4) +/* Enable left HP mute/unmute depop */ +#define RT5631_EN_HP_L_M_UN_MUTE_DEPOP (0x1 << 1) +/* Enable right HP mute/unmute depop */ +#define RT5631_EN_HP_R_M_UN_MUTE_DEPOP (0x1 << 0) + +/* De-POP Fnction Control(0x56) */ +#define RT5631_EN_ONE_BIT_DEPOP (0x1 << 15) +#define RT5631_EN_CAP_FREE_DEPOP (0x1 << 14) + +/* Jack Detect Control Register(0x5A) */ +#define RT5631_JD_USE_MASK (0x3 << 14) +#define RT5631_JD_USE_JD2 (0x3 << 14) +#define RT5631_JD_USE_JD1 (0x2 << 14) +#define RT5631_JD_USE_GPIO (0x1 << 14) +#define RT5631_JD_OFF (0x0 << 14) +/* JD trigger enable for HP */ +#define RT5631_JD_HP_EN (0x1 << 11) +#define RT5631_JD_HP_TRI_MASK (0x1 << 10) +#define RT5631_JD_HP_TRI_HI (0x1 << 10) +#define RT5631_JD_HP_TRI_LO (0x1 << 10) +/* JD trigger enable for speaker LP/LN */ +#define RT5631_JD_SPK_L_EN (0x1 << 9) +#define RT5631_JD_SPK_L_TRI_MASK (0x1 << 8) +#define RT5631_JD_SPK_L_TRI_HI (0x1 << 8) +#define RT5631_JD_SPK_L_TRI_LO (0x0 << 8) +/* JD trigger enable for speaker RP/RN */ +#define RT5631_JD_SPK_R_EN (0x1 << 7) +#define RT5631_JD_SPK_R_TRI_MASK (0x1 << 6) +#define RT5631_JD_SPK_R_TRI_HI (0x1 << 6) +#define RT5631_JD_SPK_R_TRI_LO (0x0 << 6) +/* JD trigger enable for monoout */ +#define RT5631_JD_MONO_EN (0x1 << 5) +#define RT5631_JD_MONO_TRI_MASK (0x1 << 4) +#define RT5631_JD_MONO_TRI_HI (0x1 << 4) +#define RT5631_JD_MONO_TRI_LO (0x0 << 4) +/* JD trigger enable for Lout */ +#define RT5631_JD_AUX_1_EN (0x1 << 3) +#define RT5631_JD_AUX_1_MASK (0x1 << 2) +#define RT5631_JD_AUX_1_TRI_HI (0x1 << 2) +#define RT5631_JD_AUX_1_TRI_LO (0x0 << 2) +/* JD trigger enable for Rout */ +#define RT5631_JD_AUX_2_EN (0x1 << 1) +#define RT5631_JD_AUX_2_MASK (0x1 << 0) +#define RT5631_JD_AUX_2_TRI_HI (0x1 << 0) +#define RT5631_JD_AUX_2_TRI_LO (0x0 << 0) + +/* ALC CONTROL 1(0x64) */ +#define RT5631_ALC_ATTACK_RATE_MASK (0x1F << 8) +#define RT5631_ALC_RECOVERY_RATE_MASK (0x1F << 0) + +/* ALC CONTROL 2(0x65) */ +/* select Compensation gain for Noise gate function */ +#define RT5631_ALC_COM_NOISE_GATE_MASK (0xF << 0) + +/* ALC CONTROL 3(0x66) */ +#define RT5631_ALC_FUN_MASK (0x3 << 14) +#define RT5631_ALC_FUN_DIS (0x0 << 14) +#define RT5631_ALC_ENA_DAC_PATH (0x1 << 14) +#define RT5631_ALC_ENA_ADC_PATH (0x3 << 14) +#define RT5631_ALC_PARA_UPDATE (0x1 << 13) +#define RT5631_ALC_LIMIT_LEVEL_MASK (0x1F << 8) +#define RT5631_ALC_NOISE_GATE_FUN_MASK (0x1 << 7) +#define RT5631_ALC_NOISE_GATE_FUN_DIS (0x0 << 7) +#define RT5631_ALC_NOISE_GATE_FUN_ENA (0x1 << 7) +/* ALC noise gate hold data function */ +#define RT5631_ALC_NOISE_GATE_H_D_MASK (0x1 << 6) +#define RT5631_ALC_NOISE_GATE_H_D_DIS (0x0 << 6) +#define RT5631_ALC_NOISE_GATE_H_D_ENA (0x1 << 6) + +/* Psedueo Stereo & Spatial Effect Block Control(0x68) */ +#define RT5631_SPATIAL_CTRL_EN (0x1 << 15) +#define RT5631_ALL_PASS_FILTER_EN (0x1 << 14) +#define RT5631_PSEUDO_STEREO_EN (0x1 << 13) +#define RT5631_STEREO_EXPENSION_EN (0x1 << 12) +/* 3D gain parameter */ +#define RT5631_GAIN_3D_PARA_MASK (0x3 << 6) +#define RT5631_GAIN_3D_PARA_1_00 (0x0 << 6) /* 3D gain 1.0 */ +#define RT5631_GAIN_3D_PARA_1_50 (0x1 << 6) /* 3D gain 1.5 */ +#define RT5631_GAIN_3D_PARA_2_00 (0x2 << 6) /* 3D gain 2.0 */ +/* 3D ratio parameter */ +#define RT5631_RATIO_3D_MASK (0x3 << 4) +#define RT5631_RATIO_3D_0_0 (0x0 << 4) /* 3D ratio 0.0 */ +#define RT5631_RATIO_3D_0_66 (0x1 << 4) /* 3D ratio 0.66 */ +#define RT5631_RATIO_3D_1_0 (0x2 << 4) /* 3D ratio 1.0 */ +/* select samplerate for all pass filter */ +#define RT5631_APF_FUN_SLE_MASK (0x3 << 0) +#define RT5631_APF_FUN_SEL_48K (0x3 << 0) +#define RT5631_APF_FUN_SEL_44_1K (0x2 << 0) +#define RT5631_APF_FUN_SEL_32K (0x1 << 0) +#define RT5631_APF_FUN_DIS (0x0 << 0) + +/* EQ CONTROL 1(0x6E) */ +#define RT5631_HW_EQ_PATH_SEL_MASK (0x1 << 15) +#define RT5631_HW_EQ_PATH_SEL_DAC (0x0 << 15) +#define RT5631_HW_EQ_PATH_SEL_ADC (0x1 << 15) +#define RT5631_HW_EQ_UPDATE_CTRL (0x1 << 14) + +#define RT5631_EN_HW_EQ_HPF2 (0x1 << 5) +#define RT5631_EN_HW_EQ_HPF1 (0x1 << 4) +#define RT5631_EN_HW_EQ_BP3 (0x1 << 3) +#define RT5631_EN_HW_EQ_BP2 (0x1 << 2) +#define RT5631_EN_HW_EQ_BP1 (0x1 << 1) +#define RT5631_EN_HW_EQ_LPF (0x1 << 0) + + +#endif /* __RTCODEC5631_H__ */ -- cgit v1.1 From 06c15baf90fc47eca1dc19e1f8ab26a7e159e173 Mon Sep 17 00:00:00 2001 From: Bas Vermeulen Date: Mon, 19 Sep 2011 12:57:09 +0200 Subject: ASoC: 88pm860x-codec - Allow independent use of both I2S playback and capture Introduce a I2S CLK supply so playback and capture can operate independently. Signed-off-by: Bas Vermeulen Signed-off-by: Mark Brown --- sound/soc/codecs/88pm860x-codec.c | 10 ++++++++-- 1 file changed, 8 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/88pm860x-codec.c b/sound/soc/codecs/88pm860x-codec.c index 1924157..0198dbb 100644 --- a/sound/soc/codecs/88pm860x-codec.c +++ b/sound/soc/codecs/88pm860x-codec.c @@ -772,11 +772,12 @@ static const struct snd_soc_dapm_widget pm860x_dapm_widgets[] = { SND_SOC_DAPM_AIF_IN("I2S DIN", "I2S Playback", 0, - PM860X_DAC_EN_2, 0, 0), + SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("I2S DIN1", "I2S Playback", 0, - PM860X_DAC_EN_2, 0, 0), + SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_OUT("I2S DOUT", "I2S Capture", 0, PM860X_I2S_IFACE_3, 5, 1), + SND_SOC_DAPM_SUPPLY("I2S CLK", PM860X_DAC_EN_2, 0, 0, NULL, 0), SND_SOC_DAPM_MUX("I2S Mic Mux", SND_SOC_NOPM, 0, 0, &i2s_mic_mux), SND_SOC_DAPM_MUX("ADC Left Mux", SND_SOC_NOPM, 0, 0, &adcl_mux), SND_SOC_DAPM_MUX("ADC Right Mux", SND_SOC_NOPM, 0, 0, &adcr_mux), @@ -868,6 +869,11 @@ static const struct snd_soc_dapm_route audio_map[] = { {"Left ADC", NULL, "Left ADC MOD"}, {"Right ADC", NULL, "Right ADC MOD"}, + /* I2S Clock */ + {"I2S DIN", NULL, "I2S CLK"}, + {"I2S DIN1", NULL, "I2S CLK"}, + {"I2S DOUT", NULL, "I2S CLK"}, + /* PCM/AIF1 Inputs */ {"PCM SDO", NULL, "ADC Left Mux"}, {"PCM SDO", NULL, "ADCR EC Mux"}, -- cgit v1.1 From 548aae8cc497397310c66c336ed9c4f7dd5be4f4 Mon Sep 17 00:00:00 2001 From: Bas Vermeulen Date: Mon, 19 Sep 2011 13:09:10 +0200 Subject: ASoC: 88pm860x-codec - reset the codec correctly Reset the codec according to the Audio power-up delay errata for the 88PM8607. Signed-off-by: Bas Vermeulen Signed-off-by: Mark Brown --- sound/soc/codecs/88pm860x-codec.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/88pm860x-codec.c b/sound/soc/codecs/88pm860x-codec.c index 0198dbb..df7b4a0 100644 --- a/sound/soc/codecs/88pm860x-codec.c +++ b/sound/soc/codecs/88pm860x-codec.c @@ -1179,6 +1179,9 @@ static int pm860x_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_STANDBY: if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { /* Enable Audio PLL & Audio section */ + data = AUDIO_PLL | AUDIO_SECTION_ON; + pm860x_reg_write(codec->control_data, REG_MISC2, data); + udelay(300); data = AUDIO_PLL | AUDIO_SECTION_RESET | AUDIO_SECTION_ON; pm860x_reg_write(codec->control_data, REG_MISC2, data); -- cgit v1.1 From 17841020e9d3dbd4e8114c2142c2bc6d45c01da1 Mon Sep 17 00:00:00 2001 From: Dong Aisheng Date: Mon, 29 Aug 2011 17:15:14 +0800 Subject: ASoC: soc-core: symmetry checking for each DAIs separately The orginal code does not cover the case that one DAI such as codec may be shared between other two DAIs(CPU). When do symmetry checking, altough the codec DAI requires symmetry, the two CPU DAIs may still be configured to run on different rates. We change to check each DAI's state separately instead of only checking the dai link to prevent this issue. Signed-off-by: Dong Aisheng Tested-by: Wolfram Sang Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/soc-pcm.c | 40 +++++++++++++++++++++++++--------------- 1 file changed, 25 insertions(+), 15 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 1aee9fc..8eb0f07 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -27,15 +27,13 @@ #include #include -static int soc_pcm_apply_symmetry(struct snd_pcm_substream *substream) +static int soc_pcm_apply_symmetry(struct snd_pcm_substream *substream, + struct snd_soc_dai *soc_dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; int ret; - if (!codec_dai->driver->symmetric_rates && - !cpu_dai->driver->symmetric_rates && + if (!soc_dai->driver->symmetric_rates && !rtd->dai_link->symmetric_rates) return 0; @@ -43,19 +41,19 @@ static int soc_pcm_apply_symmetry(struct snd_pcm_substream *substream) * the second can need to get its constraints before the first has * picked a rate. Complain and allow the application to carry on. */ - if (!rtd->rate) { - dev_warn(&rtd->dev, + if (!soc_dai->rate) { + dev_warn(soc_dai->dev, "Not enforcing symmetric_rates due to race\n"); return 0; } - dev_dbg(&rtd->dev, "Symmetry forces %dHz rate\n", rtd->rate); + dev_dbg(soc_dai->dev, "Symmetry forces %dHz rate\n", soc_dai->rate); ret = snd_pcm_hw_constraint_minmax(substream->runtime, SNDRV_PCM_HW_PARAM_RATE, - rtd->rate, rtd->rate); + soc_dai->rate, soc_dai->rate); if (ret < 0) { - dev_err(&rtd->dev, + dev_err(soc_dai->dev, "Unable to apply rate symmetry constraint: %d\n", ret); return ret; } @@ -185,8 +183,14 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) } /* Symmetry only applies if we've already got an active stream. */ - if (cpu_dai->active || codec_dai->active) { - ret = soc_pcm_apply_symmetry(substream); + if (cpu_dai->active) { + ret = soc_pcm_apply_symmetry(substream, cpu_dai); + if (ret != 0) + goto config_err; + } + + if (codec_dai->active) { + ret = soc_pcm_apply_symmetry(substream, codec_dai); if (ret != 0) goto config_err; } @@ -288,8 +292,12 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) codec_dai->active--; codec->active--; - if (!cpu_dai->active && !codec_dai->active) - rtd->rate = 0; + /* clear the corresponding DAIs rate when inactive */ + if (!cpu_dai->active) + cpu_dai->rate = 0; + + if (!codec_dai->active) + codec_dai->rate = 0; /* Muting the DAC suppresses artifacts caused during digital * shutdown, for example from stopping clocks. @@ -447,7 +455,9 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, } } - rtd->rate = params_rate(params); + /* store the rate for each DAIs */ + cpu_dai->rate = params_rate(params); + codec_dai->rate = params_rate(params); out: mutex_unlock(&rtd->pcm_mutex); -- cgit v1.1 From 07441006b2a1df0478bb7bdafd9dcd578898f2d4 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 30 Aug 2011 14:39:52 +0300 Subject: ASoC: tpa6130a2: Model support cleanup Use the device name and driver_data to identify the TPA model supported by the driver. Board files should use either "tpa6130a2" or "tpa6140a2" as device name to specify the model in used on the specific board. Signed-off-by: Peter Ujfalusi Tested-by: Jarkko Nikula Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/tpa6130a2.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tpa6130a2.c b/sound/soc/codecs/tpa6130a2.c index b2572c4..a14689b 100644 --- a/sound/soc/codecs/tpa6130a2.c +++ b/sound/soc/codecs/tpa6130a2.c @@ -383,7 +383,7 @@ static int __devinit tpa6130a2_probe(struct i2c_client *client, pdata = client->dev.platform_data; data->power_gpio = pdata->power_gpio; - data->id = pdata->id; + data->id = id->driver_data; mutex_init(&data->mutex); @@ -405,7 +405,7 @@ static int __devinit tpa6130a2_probe(struct i2c_client *client, switch (data->id) { default: dev_warn(dev, "Unknown TPA model (%d). Assuming 6130A2\n", - pdata->id); + data->id); case TPA6130A2: regulator = "Vdd"; break; @@ -469,7 +469,8 @@ static int __devexit tpa6130a2_remove(struct i2c_client *client) } static const struct i2c_device_id tpa6130a2_id[] = { - { "tpa6130a2", 0 }, + { "tpa6130a2", TPA6130A2 }, + { "tpa6140a2", TPA6140A2 }, { } }; MODULE_DEVICE_TABLE(i2c, tpa6130a2_id); -- cgit v1.1 From 0722d055ac2236da4e319d22a99c9f7e82dbdd5d Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 30 Aug 2011 14:39:54 +0300 Subject: ASoC: tpa6130a2: Remove model_id from platform data The model_id is no longer needed within the platform_data for the TPA driver since the model of TPA specified with the device name (tpa6130a2/tpa6140a2). Also update rx51 (the only affected user) to use the device name rather than platform data. Signed-off-by: Peter Ujfalusi Tested-by: Jarkko Nikula Acked-by: Liam Girdwood Acked-by: Tony Lindgren Signed-off-by: Mark Brown --- sound/soc/codecs/tpa6130a2.c | 5 +++++ 1 file changed, 5 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/tpa6130a2.c b/sound/soc/codecs/tpa6130a2.c index a14689b..7eeca79 100644 --- a/sound/soc/codecs/tpa6130a2.c +++ b/sound/soc/codecs/tpa6130a2.c @@ -33,6 +33,11 @@ #include "tpa6130a2.h" +enum tpa_model { + TPA6130A2, + TPA6140A2, +}; + static struct i2c_client *tpa6130a2_client; /* This struct is used to save the context */ -- cgit v1.1 From b199adfdff98092b16f67860013fb5263579c476 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 2 Aug 2011 13:35:30 +0300 Subject: ASoC: omap-mcpdm: Fix threshold and dma configuration DMA packet_size must be configured based on the McPDM FIFO threshold value, number of channels. Due to the FIFO operation the DMA muse be configured differently for playback, and capture. At the same time fix the McPDM threshold values used for playback, and capture to avoid broken code. Signed-off-by: Peter Ujfalusi Acked-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/omap/omap-mcpdm.c | 17 ++++++++++++----- 1 file changed, 12 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/omap-mcpdm.c b/sound/soc/omap/omap-mcpdm.c index bed09c2..7727de0 100644 --- a/sound/soc/omap/omap-mcpdm.c +++ b/sound/soc/omap/omap-mcpdm.c @@ -46,13 +46,13 @@ static struct omap_mcpdm_link omap_mcpdm_links[] = { /* downlink */ { .irq_mask = MCPDM_DN_IRQ_EMPTY | MCPDM_DN_IRQ_FULL, - .threshold = 1, + .threshold = 2, .format = PDMOUTFORMAT_LJUST, }, /* uplink */ { .irq_mask = MCPDM_UP_IRQ_EMPTY | MCPDM_UP_IRQ_FULL, - .threshold = 1, + .threshold = UP_THRES_MAX - 3, .format = PDMOUTFORMAT_LJUST, }, }; @@ -136,12 +136,11 @@ static int omap_mcpdm_dai_hw_params(struct snd_pcm_substream *substream, { struct omap_mcpdm_data *mcpdm_priv = snd_soc_dai_get_drvdata(dai); struct omap_mcpdm_link *mcpdm_links = mcpdm_priv->links; + struct omap_pcm_dma_data *dma_data; + int threshold; int stream = substream->stream; int channels, err, link_mask = 0; - snd_soc_dai_set_dma_data(dai, substream, - &omap_mcpdm_dai_dma_params[stream]); - channels = params_channels(params); switch (channels) { case 4: @@ -164,14 +163,22 @@ static int omap_mcpdm_dai_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } + dma_data = &omap_mcpdm_dai_dma_params[stream]; + threshold = mcpdm_links[stream].threshold; + if (stream == SNDRV_PCM_STREAM_PLAYBACK) { mcpdm_links[stream].channels = link_mask << 3; + dma_data->packet_size = (DN_THRES_MAX - threshold) * channels; + err = omap_mcpdm_playback_open(&mcpdm_links[stream]); } else { mcpdm_links[stream].channels = link_mask << 0; + dma_data->packet_size = threshold * channels; + err = omap_mcpdm_capture_open(&mcpdm_links[stream]); } + snd_soc_dai_set_dma_data(dai, substream, dma_data); return err; } -- cgit v1.1 From 3a98cd6b2b0459de4ebf85356fbcc34b9848b58a Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 2 Aug 2011 14:04:26 +0300 Subject: ASoC: OMAP4: McPDM: Convert to hwmod/omap_device In order to probe, and operate correctly, the OMAP McPDM driver needs to be converted to use hwmod. The device name has been changed to probe the driver. Replace the clk_* with pm_runtime_* calls to manage the clocks correctly. Missing request_mem_region/release_mem_region added. Signed-off-by: Peter Ujfalusi Acked-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/omap/mcpdm.c | 38 +++++++++++++++++++++----------------- sound/soc/omap/mcpdm.h | 1 - sound/soc/omap/omap-mcpdm.c | 2 +- sound/soc/omap/sdp4430.c | 2 +- 4 files changed, 23 insertions(+), 20 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/mcpdm.c b/sound/soc/omap/mcpdm.c index 928f037..d29cc98 100644 --- a/sound/soc/omap/mcpdm.c +++ b/sound/soc/omap/mcpdm.c @@ -28,7 +28,7 @@ #include #include #include -#include +#include #include #include #include @@ -322,11 +322,11 @@ static irqreturn_t omap_mcpdm_irq_handler(int irq, void *dev_id) return IRQ_HANDLED; } -int omap_mcpdm_request(void) + int omap_mcpdm_request(void) { int ret; - clk_enable(mcpdm->clk); + pm_runtime_get_sync(mcpdm->dev); spin_lock(&mcpdm->lock); @@ -353,7 +353,8 @@ int omap_mcpdm_request(void) return 0; err: - clk_disable(mcpdm->clk); + mcpdm->free = 1; + pm_runtime_put_sync(mcpdm->dev); return ret; } @@ -368,7 +369,7 @@ void omap_mcpdm_free(void) mcpdm->free = 1; spin_unlock(&mcpdm->lock); - clk_disable(mcpdm->clk); + pm_runtime_put_sync(mcpdm->dev); free_irq(mcpdm->irq, (void *)mcpdm); } @@ -421,28 +422,29 @@ int __devinit omap_mcpdm_probe(struct platform_device *pdev) spin_lock_init(&mcpdm->lock); mcpdm->free = 1; + + if (!request_mem_region(res->start, resource_size(res), "McPDM")) { + ret = -EBUSY; + goto err_resource; + } + mcpdm->io_base = ioremap(res->start, resource_size(res)); if (!mcpdm->io_base) { ret = -ENOMEM; - goto err_resource; + goto err_remap; } mcpdm->irq = platform_get_irq(pdev, 0); - mcpdm->clk = clk_get(&pdev->dev, "pdm_ck"); - if (IS_ERR(mcpdm->clk)) { - ret = PTR_ERR(mcpdm->clk); - dev_err(&pdev->dev, "unable to get pdm_ck: %d\n", ret); - goto err_clk; - } - mcpdm->dev = &pdev->dev; platform_set_drvdata(pdev, mcpdm); + pm_runtime_enable(mcpdm->dev); + return 0; -err_clk: - iounmap(mcpdm->io_base); +err_remap: + release_mem_region(res->start, resource_size(res)); err_resource: kfree(mcpdm); exit: @@ -452,14 +454,16 @@ exit: int __devexit omap_mcpdm_remove(struct platform_device *pdev) { struct omap_mcpdm *mcpdm_ptr = platform_get_drvdata(pdev); + struct resource *res; platform_set_drvdata(pdev, NULL); - clk_put(mcpdm_ptr->clk); + pm_runtime_disable(mcpdm_ptr->dev); iounmap(mcpdm_ptr->io_base); + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + release_mem_region(res->start, resource_size(res)); - mcpdm_ptr->clk = NULL; mcpdm_ptr->free = 0; mcpdm_ptr->dev = NULL; diff --git a/sound/soc/omap/mcpdm.h b/sound/soc/omap/mcpdm.h index df3e16f..b055ad1 100644 --- a/sound/soc/omap/mcpdm.h +++ b/sound/soc/omap/mcpdm.h @@ -131,7 +131,6 @@ struct omap_mcpdm { spinlock_t lock; struct omap_mcpdm_platform_data *pdata; - struct clk *clk; struct omap_mcpdm_link *downlink; struct omap_mcpdm_link *uplink; struct completion irq_completion; diff --git a/sound/soc/omap/omap-mcpdm.c b/sound/soc/omap/omap-mcpdm.c index 7727de0..0820b9e 100644 --- a/sound/soc/omap/omap-mcpdm.c +++ b/sound/soc/omap/omap-mcpdm.c @@ -254,7 +254,7 @@ static int __devexit asoc_mcpdm_remove(struct platform_device *pdev) static struct platform_driver asoc_mcpdm_driver = { .driver = { - .name = "omap-mcpdm-dai", + .name = "omap-mcpdm", .owner = THIS_MODULE, }, diff --git a/sound/soc/omap/sdp4430.c b/sound/soc/omap/sdp4430.c index b80efb0..32782b9 100644 --- a/sound/soc/omap/sdp4430.c +++ b/sound/soc/omap/sdp4430.c @@ -165,7 +165,7 @@ static int sdp4430_twl6040_init(struct snd_soc_pcm_runtime *rtd) static struct snd_soc_dai_link sdp4430_dai = { .name = "TWL6040", .stream_name = "TWL6040", - .cpu_dai_name ="omap-mcpdm-dai", + .cpu_dai_name = "omap-mcpdm", .codec_dai_name = "twl6040-hifi", .platform_name = "omap-pcm-audio", .codec_name = "twl6040-codec", -- cgit v1.1 From f5f9d7bf6eb2efc6e819550b70a3292d83acc6eb Mon Sep 17 00:00:00 2001 From: Misael Lopez Cruz Date: Tue, 5 Jul 2011 19:50:45 +0300 Subject: ASoC: omap-mcpdm: Replace legacy driver Reasons for the replacement: The current driver for McPDM was developed to support the legacy mode only. In preparation for the ABE support the current driver stack need the be replaced. The new driver is much simpler, easier to extend, and it also fixes some of the issues with the old stack. Main changes: - single file for omap-mcpdm (mcpdm.c/h removed) - Define names for registers, bits cleaned up, prefixed - Full-duplex audio operation (arecord | aplay) has been fixed - Less code McPDM need to be turned off after all streams has been stopped. This might cause pop noise on the output, if the codec's DAC is still powered at this time. Signed-off-by: Misael Lopez Cruz Signed-off-by: Liam Girdwood Signed-off-by: Sebastien Guiriec Signed-off-by: Peter Ujfalusi Acked-by: Mark Brown --- sound/soc/omap/Makefile | 2 +- sound/soc/omap/mcpdm.c | 474 -------------------------------------------- sound/soc/omap/mcpdm.h | 152 -------------- sound/soc/omap/omap-mcpdm.c | 443 +++++++++++++++++++++++++++++++---------- sound/soc/omap/omap-mcpdm.h | 95 +++++++++ sound/soc/omap/sdp4430.c | 2 +- 6 files changed, 434 insertions(+), 734 deletions(-) delete mode 100644 sound/soc/omap/mcpdm.c delete mode 100644 sound/soc/omap/mcpdm.h create mode 100644 sound/soc/omap/omap-mcpdm.h (limited to 'sound') diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile index 59e2c8d..052fd75 100644 --- a/sound/soc/omap/Makefile +++ b/sound/soc/omap/Makefile @@ -1,7 +1,7 @@ # OMAP Platform Support snd-soc-omap-objs := omap-pcm.o snd-soc-omap-mcbsp-objs := omap-mcbsp.o -snd-soc-omap-mcpdm-objs := omap-mcpdm.o mcpdm.o +snd-soc-omap-mcpdm-objs := omap-mcpdm.o snd-soc-omap-hdmi-objs := omap-hdmi.o obj-$(CONFIG_SND_OMAP_SOC) += snd-soc-omap.o diff --git a/sound/soc/omap/mcpdm.c b/sound/soc/omap/mcpdm.c deleted file mode 100644 index d29cc98..0000000 --- a/sound/soc/omap/mcpdm.c +++ /dev/null @@ -1,474 +0,0 @@ -/* - * mcpdm.c -- McPDM interface driver - * - * Author: Jorge Eduardo Candelaria - * Copyright (C) 2009 - Texas Instruments, Inc. - * - * This program is free software; you can redistribute it and/or - * modify it under the terms of the GNU General Public License - * version 2 as published by the Free Software Foundation. - * - * This program is distributed in the hope that it will be useful, but - * WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA - * 02110-1301 USA - * - */ - -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include - -#include "mcpdm.h" - -static struct omap_mcpdm *mcpdm; - -static inline void omap_mcpdm_write(u16 reg, u32 val) -{ - __raw_writel(val, mcpdm->io_base + reg); -} - -static inline int omap_mcpdm_read(u16 reg) -{ - return __raw_readl(mcpdm->io_base + reg); -} - -static void omap_mcpdm_reg_dump(void) -{ - dev_dbg(mcpdm->dev, "***********************\n"); - dev_dbg(mcpdm->dev, "IRQSTATUS_RAW: 0x%04x\n", - omap_mcpdm_read(MCPDM_IRQSTATUS_RAW)); - dev_dbg(mcpdm->dev, "IRQSTATUS: 0x%04x\n", - omap_mcpdm_read(MCPDM_IRQSTATUS)); - dev_dbg(mcpdm->dev, "IRQENABLE_SET: 0x%04x\n", - omap_mcpdm_read(MCPDM_IRQENABLE_SET)); - dev_dbg(mcpdm->dev, "IRQENABLE_CLR: 0x%04x\n", - omap_mcpdm_read(MCPDM_IRQENABLE_CLR)); - dev_dbg(mcpdm->dev, "IRQWAKE_EN: 0x%04x\n", - omap_mcpdm_read(MCPDM_IRQWAKE_EN)); - dev_dbg(mcpdm->dev, "DMAENABLE_SET: 0x%04x\n", - omap_mcpdm_read(MCPDM_DMAENABLE_SET)); - dev_dbg(mcpdm->dev, "DMAENABLE_CLR: 0x%04x\n", - omap_mcpdm_read(MCPDM_DMAENABLE_CLR)); - dev_dbg(mcpdm->dev, "DMAWAKEEN: 0x%04x\n", - omap_mcpdm_read(MCPDM_DMAWAKEEN)); - dev_dbg(mcpdm->dev, "CTRL: 0x%04x\n", - omap_mcpdm_read(MCPDM_CTRL)); - dev_dbg(mcpdm->dev, "DN_DATA: 0x%04x\n", - omap_mcpdm_read(MCPDM_DN_DATA)); - dev_dbg(mcpdm->dev, "UP_DATA: 0x%04x\n", - omap_mcpdm_read(MCPDM_UP_DATA)); - dev_dbg(mcpdm->dev, "FIFO_CTRL_DN: 0x%04x\n", - omap_mcpdm_read(MCPDM_FIFO_CTRL_DN)); - dev_dbg(mcpdm->dev, "FIFO_CTRL_UP: 0x%04x\n", - omap_mcpdm_read(MCPDM_FIFO_CTRL_UP)); - dev_dbg(mcpdm->dev, "DN_OFFSET: 0x%04x\n", - omap_mcpdm_read(MCPDM_DN_OFFSET)); - dev_dbg(mcpdm->dev, "***********************\n"); -} - -/* - * Takes the McPDM module in and out of reset state. - * Uplink and downlink can be reset individually. - */ -static void omap_mcpdm_reset_capture(int reset) -{ - int ctrl = omap_mcpdm_read(MCPDM_CTRL); - - if (reset) - ctrl |= SW_UP_RST; - else - ctrl &= ~SW_UP_RST; - - omap_mcpdm_write(MCPDM_CTRL, ctrl); -} - -static void omap_mcpdm_reset_playback(int reset) -{ - int ctrl = omap_mcpdm_read(MCPDM_CTRL); - - if (reset) - ctrl |= SW_DN_RST; - else - ctrl &= ~SW_DN_RST; - - omap_mcpdm_write(MCPDM_CTRL, ctrl); -} - -/* - * Enables the transfer through the PDM interface to/from the Phoenix - * codec by enabling the corresponding UP or DN channels. - */ -void omap_mcpdm_start(int stream) -{ - int ctrl = omap_mcpdm_read(MCPDM_CTRL); - - if (stream) - ctrl |= mcpdm->up_channels; - else - ctrl |= mcpdm->dn_channels; - - omap_mcpdm_write(MCPDM_CTRL, ctrl); -} - -/* - * Disables the transfer through the PDM interface to/from the Phoenix - * codec by disabling the corresponding UP or DN channels. - */ -void omap_mcpdm_stop(int stream) -{ - int ctrl = omap_mcpdm_read(MCPDM_CTRL); - - if (stream) - ctrl &= ~mcpdm->up_channels; - else - ctrl &= ~mcpdm->dn_channels; - - omap_mcpdm_write(MCPDM_CTRL, ctrl); -} - -/* - * Configures McPDM uplink for audio recording. - * This function should be called before omap_mcpdm_start. - */ -int omap_mcpdm_capture_open(struct omap_mcpdm_link *uplink) -{ - int irq_mask = 0; - int ctrl; - - if (!uplink) - return -EINVAL; - - mcpdm->uplink = uplink; - - /* Enable irq request generation */ - irq_mask |= uplink->irq_mask & MCPDM_UPLINK_IRQ_MASK; - omap_mcpdm_write(MCPDM_IRQENABLE_SET, irq_mask); - - /* Configure uplink threshold */ - if (uplink->threshold > UP_THRES_MAX) - uplink->threshold = UP_THRES_MAX; - - omap_mcpdm_write(MCPDM_FIFO_CTRL_UP, uplink->threshold); - - /* Configure DMA controller */ - omap_mcpdm_write(MCPDM_DMAENABLE_SET, DMA_UP_ENABLE); - - /* Set pdm out format */ - ctrl = omap_mcpdm_read(MCPDM_CTRL); - ctrl &= ~PDMOUTFORMAT; - ctrl |= uplink->format & PDMOUTFORMAT; - - /* Uplink channels */ - mcpdm->up_channels = uplink->channels & (PDM_UP_MASK | PDM_STATUS_MASK); - - omap_mcpdm_write(MCPDM_CTRL, ctrl); - - return 0; -} - -/* - * Configures McPDM downlink for audio playback. - * This function should be called before omap_mcpdm_start. - */ -int omap_mcpdm_playback_open(struct omap_mcpdm_link *downlink) -{ - int irq_mask = 0; - int ctrl; - - if (!downlink) - return -EINVAL; - - mcpdm->downlink = downlink; - - /* Enable irq request generation */ - irq_mask |= downlink->irq_mask & MCPDM_DOWNLINK_IRQ_MASK; - omap_mcpdm_write(MCPDM_IRQENABLE_SET, irq_mask); - - /* Configure uplink threshold */ - if (downlink->threshold > DN_THRES_MAX) - downlink->threshold = DN_THRES_MAX; - - omap_mcpdm_write(MCPDM_FIFO_CTRL_DN, downlink->threshold); - - /* Enable DMA request generation */ - omap_mcpdm_write(MCPDM_DMAENABLE_SET, DMA_DN_ENABLE); - - /* Set pdm out format */ - ctrl = omap_mcpdm_read(MCPDM_CTRL); - ctrl &= ~PDMOUTFORMAT; - ctrl |= downlink->format & PDMOUTFORMAT; - - /* Downlink channels */ - mcpdm->dn_channels = downlink->channels & (PDM_DN_MASK | PDM_CMD_MASK); - - omap_mcpdm_write(MCPDM_CTRL, ctrl); - - return 0; -} - -/* - * Cleans McPDM uplink configuration. - * This function should be called when the stream is closed. - */ -int omap_mcpdm_capture_close(struct omap_mcpdm_link *uplink) -{ - int irq_mask = 0; - - if (!uplink) - return -EINVAL; - - /* Disable irq request generation */ - irq_mask |= uplink->irq_mask & MCPDM_UPLINK_IRQ_MASK; - omap_mcpdm_write(MCPDM_IRQENABLE_CLR, irq_mask); - - /* Disable DMA request generation */ - omap_mcpdm_write(MCPDM_DMAENABLE_CLR, DMA_UP_ENABLE); - - /* Clear Downlink channels */ - mcpdm->up_channels = 0; - - mcpdm->uplink = NULL; - - return 0; -} - -/* - * Cleans McPDM downlink configuration. - * This function should be called when the stream is closed. - */ -int omap_mcpdm_playback_close(struct omap_mcpdm_link *downlink) -{ - int irq_mask = 0; - - if (!downlink) - return -EINVAL; - - /* Disable irq request generation */ - irq_mask |= downlink->irq_mask & MCPDM_DOWNLINK_IRQ_MASK; - omap_mcpdm_write(MCPDM_IRQENABLE_CLR, irq_mask); - - /* Disable DMA request generation */ - omap_mcpdm_write(MCPDM_DMAENABLE_CLR, DMA_DN_ENABLE); - - /* clear Downlink channels */ - mcpdm->dn_channels = 0; - - mcpdm->downlink = NULL; - - return 0; -} - -static irqreturn_t omap_mcpdm_irq_handler(int irq, void *dev_id) -{ - struct omap_mcpdm *mcpdm_irq = dev_id; - int irq_status; - - irq_status = omap_mcpdm_read(MCPDM_IRQSTATUS); - - /* Acknowledge irq event */ - omap_mcpdm_write(MCPDM_IRQSTATUS, irq_status); - - if (irq & MCPDM_DN_IRQ_FULL) { - dev_err(mcpdm_irq->dev, "DN FIFO error %x\n", irq_status); - omap_mcpdm_reset_playback(1); - omap_mcpdm_playback_open(mcpdm_irq->downlink); - omap_mcpdm_reset_playback(0); - } - - if (irq & MCPDM_DN_IRQ_EMPTY) { - dev_err(mcpdm_irq->dev, "DN FIFO error %x\n", irq_status); - omap_mcpdm_reset_playback(1); - omap_mcpdm_playback_open(mcpdm_irq->downlink); - omap_mcpdm_reset_playback(0); - } - - if (irq & MCPDM_DN_IRQ) { - dev_dbg(mcpdm_irq->dev, "DN write request\n"); - } - - if (irq & MCPDM_UP_IRQ_FULL) { - dev_err(mcpdm_irq->dev, "UP FIFO error %x\n", irq_status); - omap_mcpdm_reset_capture(1); - omap_mcpdm_capture_open(mcpdm_irq->uplink); - omap_mcpdm_reset_capture(0); - } - - if (irq & MCPDM_UP_IRQ_EMPTY) { - dev_err(mcpdm_irq->dev, "UP FIFO error %x\n", irq_status); - omap_mcpdm_reset_capture(1); - omap_mcpdm_capture_open(mcpdm_irq->uplink); - omap_mcpdm_reset_capture(0); - } - - if (irq & MCPDM_UP_IRQ) { - dev_dbg(mcpdm_irq->dev, "UP write request\n"); - } - - return IRQ_HANDLED; -} - - int omap_mcpdm_request(void) -{ - int ret; - - pm_runtime_get_sync(mcpdm->dev); - - spin_lock(&mcpdm->lock); - - if (!mcpdm->free) { - dev_err(mcpdm->dev, "McPDM interface is in use\n"); - spin_unlock(&mcpdm->lock); - ret = -EBUSY; - goto err; - } - mcpdm->free = 0; - - spin_unlock(&mcpdm->lock); - - /* Disable lines while request is ongoing */ - omap_mcpdm_write(MCPDM_CTRL, 0x00); - - ret = request_irq(mcpdm->irq, omap_mcpdm_irq_handler, - 0, "McPDM", (void *)mcpdm); - if (ret) { - dev_err(mcpdm->dev, "Request for McPDM IRQ failed\n"); - goto err; - } - - return 0; - -err: - mcpdm->free = 1; - pm_runtime_put_sync(mcpdm->dev); - return ret; -} - -void omap_mcpdm_free(void) -{ - spin_lock(&mcpdm->lock); - if (mcpdm->free) { - dev_err(mcpdm->dev, "McPDM interface is already free\n"); - spin_unlock(&mcpdm->lock); - return; - } - mcpdm->free = 1; - spin_unlock(&mcpdm->lock); - - pm_runtime_put_sync(mcpdm->dev); - - free_irq(mcpdm->irq, (void *)mcpdm); -} - -/* Enable/disable DC offset cancelation for the analog - * headset path (PDM channels 1 and 2). - */ -int omap_mcpdm_set_offset(int offset1, int offset2) -{ - int offset; - - if ((offset1 > DN_OFST_MAX) || (offset2 > DN_OFST_MAX)) - return -EINVAL; - - offset = (offset1 << DN_OFST_RX1) | (offset2 << DN_OFST_RX2); - - /* offset cancellation for channel 1 */ - if (offset1) - offset |= DN_OFST_RX1_EN; - else - offset &= ~DN_OFST_RX1_EN; - - /* offset cancellation for channel 2 */ - if (offset2) - offset |= DN_OFST_RX2_EN; - else - offset &= ~DN_OFST_RX2_EN; - - omap_mcpdm_write(MCPDM_DN_OFFSET, offset); - - return 0; -} - -int __devinit omap_mcpdm_probe(struct platform_device *pdev) -{ - struct resource *res; - int ret = 0; - - mcpdm = kzalloc(sizeof(struct omap_mcpdm), GFP_KERNEL); - if (!mcpdm) { - ret = -ENOMEM; - goto exit; - } - - res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (res == NULL) { - dev_err(&pdev->dev, "no resource\n"); - goto err_resource; - } - - spin_lock_init(&mcpdm->lock); - mcpdm->free = 1; - - if (!request_mem_region(res->start, resource_size(res), "McPDM")) { - ret = -EBUSY; - goto err_resource; - } - - mcpdm->io_base = ioremap(res->start, resource_size(res)); - if (!mcpdm->io_base) { - ret = -ENOMEM; - goto err_remap; - } - - mcpdm->irq = platform_get_irq(pdev, 0); - - mcpdm->dev = &pdev->dev; - platform_set_drvdata(pdev, mcpdm); - - pm_runtime_enable(mcpdm->dev); - - return 0; - -err_remap: - release_mem_region(res->start, resource_size(res)); -err_resource: - kfree(mcpdm); -exit: - return ret; -} - -int __devexit omap_mcpdm_remove(struct platform_device *pdev) -{ - struct omap_mcpdm *mcpdm_ptr = platform_get_drvdata(pdev); - struct resource *res; - - platform_set_drvdata(pdev, NULL); - - pm_runtime_disable(mcpdm_ptr->dev); - - iounmap(mcpdm_ptr->io_base); - res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - release_mem_region(res->start, resource_size(res)); - - mcpdm_ptr->free = 0; - mcpdm_ptr->dev = NULL; - - kfree(mcpdm_ptr); - - return 0; -} - diff --git a/sound/soc/omap/mcpdm.h b/sound/soc/omap/mcpdm.h deleted file mode 100644 index b055ad1..0000000 --- a/sound/soc/omap/mcpdm.h +++ /dev/null @@ -1,152 +0,0 @@ -/* - * mcpdm.h -- Defines for McPDM driver - * - * Author: Jorge Eduardo Candelaria - * - * This program is free software; you can redistribute it and/or - * modify it under the terms of the GNU General Public License - * version 2 as published by the Free Software Foundation. - * - * This program is distributed in the hope that it will be useful, but - * WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA - * 02110-1301 USA - * - */ - -/* McPDM registers */ - -#define MCPDM_REVISION 0x00 -#define MCPDM_SYSCONFIG 0x10 -#define MCPDM_IRQSTATUS_RAW 0x24 -#define MCPDM_IRQSTATUS 0x28 -#define MCPDM_IRQENABLE_SET 0x2C -#define MCPDM_IRQENABLE_CLR 0x30 -#define MCPDM_IRQWAKE_EN 0x34 -#define MCPDM_DMAENABLE_SET 0x38 -#define MCPDM_DMAENABLE_CLR 0x3C -#define MCPDM_DMAWAKEEN 0x40 -#define MCPDM_CTRL 0x44 -#define MCPDM_DN_DATA 0x48 -#define MCPDM_UP_DATA 0x4C -#define MCPDM_FIFO_CTRL_DN 0x50 -#define MCPDM_FIFO_CTRL_UP 0x54 -#define MCPDM_DN_OFFSET 0x58 - -/* - * MCPDM_IRQ bit fields - * IRQSTATUS_RAW, IRQSTATUS, IRQENABLE_SET, IRQENABLE_CLR - */ - -#define MCPDM_DN_IRQ (1 << 0) -#define MCPDM_DN_IRQ_EMPTY (1 << 1) -#define MCPDM_DN_IRQ_ALMST_EMPTY (1 << 2) -#define MCPDM_DN_IRQ_FULL (1 << 3) - -#define MCPDM_UP_IRQ (1 << 8) -#define MCPDM_UP_IRQ_EMPTY (1 << 9) -#define MCPDM_UP_IRQ_ALMST_FULL (1 << 10) -#define MCPDM_UP_IRQ_FULL (1 << 11) - -#define MCPDM_DOWNLINK_IRQ_MASK 0x00F -#define MCPDM_UPLINK_IRQ_MASK 0xF00 - -/* - * MCPDM_DMAENABLE bit fields - */ - -#define DMA_DN_ENABLE 0x1 -#define DMA_UP_ENABLE 0x2 - -/* - * MCPDM_CTRL bit fields - */ - -#define PDM_UP1_EN 0x0001 -#define PDM_UP2_EN 0x0002 -#define PDM_UP3_EN 0x0004 -#define PDM_DN1_EN 0x0008 -#define PDM_DN2_EN 0x0010 -#define PDM_DN3_EN 0x0020 -#define PDM_DN4_EN 0x0040 -#define PDM_DN5_EN 0x0080 -#define PDMOUTFORMAT 0x0100 -#define CMD_INT 0x0200 -#define STATUS_INT 0x0400 -#define SW_UP_RST 0x0800 -#define SW_DN_RST 0x1000 -#define PDM_UP_MASK 0x007 -#define PDM_DN_MASK 0x0F8 -#define PDM_CMD_MASK 0x200 -#define PDM_STATUS_MASK 0x400 - - -#define PDMOUTFORMAT_LJUST (0 << 8) -#define PDMOUTFORMAT_RJUST (1 << 8) - -/* - * MCPDM_FIFO_CTRL bit fields - */ - -#define UP_THRES_MAX 0xF -#define DN_THRES_MAX 0xF - -/* - * MCPDM_DN_OFFSET bit fields - */ - -#define DN_OFST_RX1_EN 0x0001 -#define DN_OFST_RX2_EN 0x0100 - -#define DN_OFST_RX1 1 -#define DN_OFST_RX2 9 -#define DN_OFST_MAX 0x1F - -#define MCPDM_UPLINK 1 -#define MCPDM_DOWNLINK 2 - -struct omap_mcpdm_link { - int irq_mask; - int threshold; - int format; - int channels; -}; - -struct omap_mcpdm_platform_data { - unsigned long phys_base; - u16 irq; -}; - -struct omap_mcpdm { - struct device *dev; - unsigned long phys_base; - void __iomem *io_base; - u8 free; - int irq; - - spinlock_t lock; - struct omap_mcpdm_platform_data *pdata; - struct omap_mcpdm_link *downlink; - struct omap_mcpdm_link *uplink; - struct completion irq_completion; - - int dn_channels; - int up_channels; -}; - -extern void omap_mcpdm_start(int stream); -extern void omap_mcpdm_stop(int stream); -extern int omap_mcpdm_capture_open(struct omap_mcpdm_link *uplink); -extern int omap_mcpdm_playback_open(struct omap_mcpdm_link *downlink); -extern int omap_mcpdm_capture_close(struct omap_mcpdm_link *uplink); -extern int omap_mcpdm_playback_close(struct omap_mcpdm_link *downlink); -extern int omap_mcpdm_request(void); -extern void omap_mcpdm_free(void); -extern int omap_mcpdm_set_offset(int offset1, int offset2); -int __devinit omap_mcpdm_probe(struct platform_device *pdev); -int __devexit omap_mcpdm_remove(struct platform_device *pdev); diff --git a/sound/soc/omap/omap-mcpdm.c b/sound/soc/omap/omap-mcpdm.c index 0820b9e..159a5e98 100644 --- a/sound/soc/omap/omap-mcpdm.c +++ b/sound/soc/omap/omap-mcpdm.c @@ -1,11 +1,12 @@ /* * omap-mcpdm.c -- OMAP ALSA SoC DAI driver using McPDM port * - * Copyright (C) 2009 Texas Instruments + * Copyright (C) 2009 - 2011 Texas Instruments * - * Author: Misael Lopez Cruz + * Author: Misael Lopez Cruz * Contact: Jorge Eduardo Candelaria * Margarita Olaya + * Peter Ujfalusi * * This program is free software; you can redistribute it and/or * modify it under the terms of the GNU General Public License @@ -25,41 +26,39 @@ #include #include -#include +#include +#include +#include +#include +#include +#include +#include + #include #include #include -#include #include #include -#include -#include "mcpdm.h" +#include +#include "omap-mcpdm.h" #include "omap-pcm.h" -struct omap_mcpdm_data { - struct omap_mcpdm_link *links; - int active; -}; +struct omap_mcpdm { + struct device *dev; + unsigned long phys_base; + void __iomem *io_base; + int irq; -static struct omap_mcpdm_link omap_mcpdm_links[] = { - /* downlink */ - { - .irq_mask = MCPDM_DN_IRQ_EMPTY | MCPDM_DN_IRQ_FULL, - .threshold = 2, - .format = PDMOUTFORMAT_LJUST, - }, - /* uplink */ - { - .irq_mask = MCPDM_UP_IRQ_EMPTY | MCPDM_UP_IRQ_FULL, - .threshold = UP_THRES_MAX - 3, - .format = PDMOUTFORMAT_LJUST, - }, -}; + struct mutex mutex; + + /* channel data */ + u32 dn_channels; + u32 up_channels; -static struct omap_mcpdm_data mcpdm_data = { - .links = omap_mcpdm_links, - .active = 0, + /* McPDM FIFO thresholds */ + u32 dn_threshold; + u32 up_threshold; }; /* @@ -71,75 +70,232 @@ static struct omap_pcm_dma_data omap_mcpdm_dai_dma_params[] = { .dma_req = OMAP44XX_DMA_MCPDM_DL, .data_type = OMAP_DMA_DATA_TYPE_S32, .sync_mode = OMAP_DMA_SYNC_PACKET, - .packet_size = 16, - .port_addr = OMAP44XX_MCPDM_L3_BASE + MCPDM_DN_DATA, + .port_addr = OMAP44XX_MCPDM_L3_BASE + MCPDM_REG_DN_DATA, }, { .name = "Audio capture", .dma_req = OMAP44XX_DMA_MCPDM_UP, .data_type = OMAP_DMA_DATA_TYPE_S32, .sync_mode = OMAP_DMA_SYNC_PACKET, - .packet_size = 16, - .port_addr = OMAP44XX_MCPDM_L3_BASE + MCPDM_UP_DATA, + .port_addr = OMAP44XX_MCPDM_L3_BASE + MCPDM_REG_UP_DATA, }, }; -static int omap_mcpdm_dai_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) +static inline void omap_mcpdm_write(struct omap_mcpdm *mcpdm, u16 reg, u32 val) { - int err = 0; + __raw_writel(val, mcpdm->io_base + reg); +} - if (!dai->active) - err = omap_mcpdm_request(); +static inline int omap_mcpdm_read(struct omap_mcpdm *mcpdm, u16 reg) +{ + return __raw_readl(mcpdm->io_base + reg); +} - return err; +#ifdef DEBUG +static void omap_mcpdm_reg_dump(struct omap_mcpdm *mcpdm) +{ + dev_dbg(mcpdm->dev, "***********************\n"); + dev_dbg(mcpdm->dev, "IRQSTATUS_RAW: 0x%04x\n", + omap_mcpdm_read(mcpdm, MCPDM_REG_IRQSTATUS_RAW)); + dev_dbg(mcpdm->dev, "IRQSTATUS: 0x%04x\n", + omap_mcpdm_read(mcpdm, MCPDM_REG_IRQSTATUS)); + dev_dbg(mcpdm->dev, "IRQENABLE_SET: 0x%04x\n", + omap_mcpdm_read(mcpdm, MCPDM_REG_IRQENABLE_SET)); + dev_dbg(mcpdm->dev, "IRQENABLE_CLR: 0x%04x\n", + omap_mcpdm_read(mcpdm, MCPDM_REG_IRQENABLE_CLR)); + dev_dbg(mcpdm->dev, "IRQWAKE_EN: 0x%04x\n", + omap_mcpdm_read(mcpdm, MCPDM_REG_IRQWAKE_EN)); + dev_dbg(mcpdm->dev, "DMAENABLE_SET: 0x%04x\n", + omap_mcpdm_read(mcpdm, MCPDM_REG_DMAENABLE_SET)); + dev_dbg(mcpdm->dev, "DMAENABLE_CLR: 0x%04x\n", + omap_mcpdm_read(mcpdm, MCPDM_REG_DMAENABLE_CLR)); + dev_dbg(mcpdm->dev, "DMAWAKEEN: 0x%04x\n", + omap_mcpdm_read(mcpdm, MCPDM_REG_DMAWAKEEN)); + dev_dbg(mcpdm->dev, "CTRL: 0x%04x\n", + omap_mcpdm_read(mcpdm, MCPDM_REG_CTRL)); + dev_dbg(mcpdm->dev, "DN_DATA: 0x%04x\n", + omap_mcpdm_read(mcpdm, MCPDM_REG_DN_DATA)); + dev_dbg(mcpdm->dev, "UP_DATA: 0x%04x\n", + omap_mcpdm_read(mcpdm, MCPDM_REG_UP_DATA)); + dev_dbg(mcpdm->dev, "FIFO_CTRL_DN: 0x%04x\n", + omap_mcpdm_read(mcpdm, MCPDM_REG_FIFO_CTRL_DN)); + dev_dbg(mcpdm->dev, "FIFO_CTRL_UP: 0x%04x\n", + omap_mcpdm_read(mcpdm, MCPDM_REG_FIFO_CTRL_UP)); + dev_dbg(mcpdm->dev, "***********************\n"); } +#else +static void omap_mcpdm_reg_dump(struct omap_mcpdm *mcpdm) {} +#endif -static void omap_mcpdm_dai_shutdown(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) +/* + * Enables the transfer through the PDM interface to/from the Phoenix + * codec by enabling the corresponding UP or DN channels. + */ +static void omap_mcpdm_start(struct omap_mcpdm *mcpdm) +{ + u32 ctrl = omap_mcpdm_read(mcpdm, MCPDM_REG_CTRL); + + ctrl |= (MCPDM_SW_DN_RST | MCPDM_SW_UP_RST); + omap_mcpdm_write(mcpdm, MCPDM_REG_CTRL, ctrl); + + ctrl |= mcpdm->dn_channels | mcpdm->up_channels; + omap_mcpdm_write(mcpdm, MCPDM_REG_CTRL, ctrl); + + ctrl &= ~(MCPDM_SW_DN_RST | MCPDM_SW_UP_RST); + omap_mcpdm_write(mcpdm, MCPDM_REG_CTRL, ctrl); +} + +/* + * Disables the transfer through the PDM interface to/from the Phoenix + * codec by disabling the corresponding UP or DN channels. + */ +static void omap_mcpdm_stop(struct omap_mcpdm *mcpdm) +{ + u32 ctrl = omap_mcpdm_read(mcpdm, MCPDM_REG_CTRL); + + ctrl |= (MCPDM_SW_DN_RST | MCPDM_SW_UP_RST); + omap_mcpdm_write(mcpdm, MCPDM_REG_CTRL, ctrl); + + ctrl &= ~(mcpdm->dn_channels | mcpdm->up_channels); + omap_mcpdm_write(mcpdm, MCPDM_REG_CTRL, ctrl); + + ctrl &= ~(MCPDM_SW_DN_RST | MCPDM_SW_UP_RST); + omap_mcpdm_write(mcpdm, MCPDM_REG_CTRL, ctrl); + +} + +/* + * Is the physical McPDM interface active. + */ +static inline int omap_mcpdm_active(struct omap_mcpdm *mcpdm) +{ + return omap_mcpdm_read(mcpdm, MCPDM_REG_CTRL) & + (MCPDM_PDM_DN_MASK | MCPDM_PDM_UP_MASK); +} + +/* + * Configures McPDM uplink, and downlink for audio. + * This function should be called before omap_mcpdm_start. + */ +static void omap_mcpdm_open_streams(struct omap_mcpdm *mcpdm) +{ + omap_mcpdm_write(mcpdm, MCPDM_REG_IRQENABLE_SET, + MCPDM_DN_IRQ_EMPTY | MCPDM_DN_IRQ_FULL | + MCPDM_UP_IRQ_EMPTY | MCPDM_UP_IRQ_FULL); + + omap_mcpdm_write(mcpdm, MCPDM_REG_FIFO_CTRL_DN, mcpdm->dn_threshold); + omap_mcpdm_write(mcpdm, MCPDM_REG_FIFO_CTRL_UP, mcpdm->up_threshold); + + omap_mcpdm_write(mcpdm, MCPDM_REG_DMAENABLE_SET, + MCPDM_DMA_DN_ENABLE | MCPDM_DMA_UP_ENABLE); +} + +/* + * Cleans McPDM uplink, and downlink configuration. + * This function should be called when the stream is closed. + */ +static void omap_mcpdm_close_streams(struct omap_mcpdm *mcpdm) { - if (!dai->active) - omap_mcpdm_free(); + /* Disable irq request generation for downlink */ + omap_mcpdm_write(mcpdm, MCPDM_REG_IRQENABLE_CLR, + MCPDM_DN_IRQ_EMPTY | MCPDM_DN_IRQ_FULL); + + /* Disable DMA request generation for downlink */ + omap_mcpdm_write(mcpdm, MCPDM_REG_DMAENABLE_CLR, MCPDM_DMA_DN_ENABLE); + + /* Disable irq request generation for uplink */ + omap_mcpdm_write(mcpdm, MCPDM_REG_IRQENABLE_CLR, + MCPDM_UP_IRQ_EMPTY | MCPDM_UP_IRQ_FULL); + + /* Disable DMA request generation for uplink */ + omap_mcpdm_write(mcpdm, MCPDM_REG_DMAENABLE_CLR, MCPDM_DMA_UP_ENABLE); +} + +static irqreturn_t omap_mcpdm_irq_handler(int irq, void *dev_id) +{ + struct omap_mcpdm *mcpdm = dev_id; + int irq_status; + + irq_status = omap_mcpdm_read(mcpdm, MCPDM_REG_IRQSTATUS); + + /* Acknowledge irq event */ + omap_mcpdm_write(mcpdm, MCPDM_REG_IRQSTATUS, irq_status); + + if (irq_status & MCPDM_DN_IRQ_FULL) + dev_dbg(mcpdm->dev, "DN (playback) FIFO Full\n"); + + if (irq_status & MCPDM_DN_IRQ_EMPTY) + dev_dbg(mcpdm->dev, "DN (playback) FIFO Empty\n"); + + if (irq_status & MCPDM_DN_IRQ) + dev_dbg(mcpdm->dev, "DN (playback) write request\n"); + + if (irq_status & MCPDM_UP_IRQ_FULL) + dev_dbg(mcpdm->dev, "UP (capture) FIFO Full\n"); + + if (irq_status & MCPDM_UP_IRQ_EMPTY) + dev_dbg(mcpdm->dev, "UP (capture) FIFO Empty\n"); + + if (irq_status & MCPDM_UP_IRQ) + dev_dbg(mcpdm->dev, "UP (capture) write request\n"); + + return IRQ_HANDLED; } -static int omap_mcpdm_dai_trigger(struct snd_pcm_substream *substream, int cmd, +static int omap_mcpdm_dai_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct omap_mcpdm_data *mcpdm_priv = snd_soc_dai_get_drvdata(dai); - int stream = substream->stream; - int err = 0; - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - case SNDRV_PCM_TRIGGER_RESUME: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - if (!mcpdm_priv->active++) - omap_mcpdm_start(stream); - break; + struct omap_mcpdm *mcpdm = snd_soc_dai_get_drvdata(dai); - case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - if (!--mcpdm_priv->active) - omap_mcpdm_stop(stream); - break; - default: - err = -EINVAL; + mutex_lock(&mcpdm->mutex); + + if (!dai->active) { + pm_runtime_get_sync(mcpdm->dev); + + /* Enable watch dog for ES above ES 1.0 to avoid saturation */ + if (omap_rev() != OMAP4430_REV_ES1_0) { + u32 ctrl = omap_mcpdm_read(mcpdm, MCPDM_REG_CTRL); + + omap_mcpdm_write(mcpdm, MCPDM_REG_CTRL, + ctrl | MCPDM_WD_EN); + } + omap_mcpdm_open_streams(mcpdm); } - return err; + mutex_unlock(&mcpdm->mutex); + + return 0; +} + +static void omap_mcpdm_dai_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct omap_mcpdm *mcpdm = snd_soc_dai_get_drvdata(dai); + + mutex_lock(&mcpdm->mutex); + + if (!dai->active) { + if (omap_mcpdm_active(mcpdm)) { + omap_mcpdm_stop(mcpdm); + omap_mcpdm_close_streams(mcpdm); + } + + if (!omap_mcpdm_active(mcpdm)) + pm_runtime_put_sync(mcpdm->dev); + } + + mutex_unlock(&mcpdm->mutex); } static int omap_mcpdm_dai_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct omap_mcpdm_data *mcpdm_priv = snd_soc_dai_get_drvdata(dai); - struct omap_mcpdm_link *mcpdm_links = mcpdm_priv->links; - struct omap_pcm_dma_data *dma_data; - int threshold; + struct omap_mcpdm *mcpdm = snd_soc_dai_get_drvdata(dai); int stream = substream->stream; - int channels, err, link_mask = 0; + struct omap_pcm_dma_data *dma_data; + int channels; + int link_mask = 0; channels = params_channels(params); switch (channels) { @@ -164,59 +320,87 @@ static int omap_mcpdm_dai_hw_params(struct snd_pcm_substream *substream, } dma_data = &omap_mcpdm_dai_dma_params[stream]; - threshold = mcpdm_links[stream].threshold; + /* Configure McPDM channels, and DMA packet size */ if (stream == SNDRV_PCM_STREAM_PLAYBACK) { - mcpdm_links[stream].channels = link_mask << 3; - dma_data->packet_size = (DN_THRES_MAX - threshold) * channels; - - err = omap_mcpdm_playback_open(&mcpdm_links[stream]); + mcpdm->dn_channels = link_mask << 3; + dma_data->packet_size = + (MCPDM_DN_THRES_MAX - mcpdm->dn_threshold) * channels; } else { - mcpdm_links[stream].channels = link_mask << 0; - dma_data->packet_size = threshold * channels; - - err = omap_mcpdm_capture_open(&mcpdm_links[stream]); + mcpdm->up_channels = link_mask << 0; + dma_data->packet_size = mcpdm->up_threshold * channels; } snd_soc_dai_set_dma_data(dai, substream, dma_data); - return err; + + return 0; } -static int omap_mcpdm_dai_hw_free(struct snd_pcm_substream *substream, +static int omap_mcpdm_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct omap_mcpdm_data *mcpdm_priv = snd_soc_dai_get_drvdata(dai); - struct omap_mcpdm_link *mcpdm_links = mcpdm_priv->links; - int stream = substream->stream; - int err; + struct omap_mcpdm *mcpdm = snd_soc_dai_get_drvdata(dai); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - err = omap_mcpdm_playback_close(&mcpdm_links[stream]); - else - err = omap_mcpdm_capture_close(&mcpdm_links[stream]); + if (!omap_mcpdm_active(mcpdm)) { + omap_mcpdm_start(mcpdm); + omap_mcpdm_reg_dump(mcpdm); + } - return err; + return 0; } static struct snd_soc_dai_ops omap_mcpdm_dai_ops = { .startup = omap_mcpdm_dai_startup, .shutdown = omap_mcpdm_dai_shutdown, - .trigger = omap_mcpdm_dai_trigger, .hw_params = omap_mcpdm_dai_hw_params, - .hw_free = omap_mcpdm_dai_hw_free, + .prepare = omap_mcpdm_prepare, }; -#define OMAP_MCPDM_RATES (SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) -#define OMAP_MCPDM_FORMATS (SNDRV_PCM_FMTBIT_S32_LE) +static int omap_mcpdm_probe(struct snd_soc_dai *dai) +{ + struct omap_mcpdm *mcpdm = snd_soc_dai_get_drvdata(dai); + int ret; + + pm_runtime_enable(mcpdm->dev); + + /* Disable lines while request is ongoing */ + pm_runtime_get_sync(mcpdm->dev); + omap_mcpdm_write(mcpdm, MCPDM_REG_CTRL, 0x00); + + ret = request_irq(mcpdm->irq, omap_mcpdm_irq_handler, + 0, "McPDM", (void *)mcpdm); -static int omap_mcpdm_dai_probe(struct snd_soc_dai *dai) + pm_runtime_put_sync(mcpdm->dev); + + if (ret) { + dev_err(mcpdm->dev, "Request for IRQ failed\n"); + pm_runtime_disable(mcpdm->dev); + } + + /* Configure McPDM threshold values */ + mcpdm->dn_threshold = 2; + mcpdm->up_threshold = MCPDM_UP_THRES_MAX - 3; + return ret; +} + +static int omap_mcpdm_remove(struct snd_soc_dai *dai) { - snd_soc_dai_set_drvdata(dai, &mcpdm_data); + struct omap_mcpdm *mcpdm = snd_soc_dai_get_drvdata(dai); + + free_irq(mcpdm->irq, (void *)mcpdm); + pm_runtime_disable(mcpdm->dev); + return 0; } +#define OMAP_MCPDM_RATES (SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) +#define OMAP_MCPDM_FORMATS SNDRV_PCM_FMTBIT_S32_LE + static struct snd_soc_dai_driver omap_mcpdm_dai = { - .probe = omap_mcpdm_dai_probe, + .probe = omap_mcpdm_probe, + .remove = omap_mcpdm_remove, + .probe_order = SND_SOC_COMP_ORDER_LATE, + .remove_order = SND_SOC_COMP_ORDER_EARLY, .playback = { .channels_min = 1, .channels_max = 4, @@ -234,32 +418,79 @@ static struct snd_soc_dai_driver omap_mcpdm_dai = { static __devinit int asoc_mcpdm_probe(struct platform_device *pdev) { - int ret; + struct omap_mcpdm *mcpdm; + struct resource *res; + int ret = 0; + + mcpdm = kzalloc(sizeof(struct omap_mcpdm), GFP_KERNEL); + if (!mcpdm) + return -ENOMEM; + + platform_set_drvdata(pdev, mcpdm); + + mutex_init(&mcpdm->mutex); + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (res == NULL) { + dev_err(&pdev->dev, "no resource\n"); + goto err_res; + } + + if (!request_mem_region(res->start, resource_size(res), "McPDM")) { + ret = -EBUSY; + goto err_res; + } + + mcpdm->io_base = ioremap(res->start, resource_size(res)); + if (!mcpdm->io_base) { + ret = -ENOMEM; + goto err_iomap; + } + + mcpdm->irq = platform_get_irq(pdev, 0); + if (mcpdm->irq < 0) { + ret = mcpdm->irq; + goto err_irq; + } + + mcpdm->dev = &pdev->dev; - ret = omap_mcpdm_probe(pdev); - if (ret < 0) - return ret; ret = snd_soc_register_dai(&pdev->dev, &omap_mcpdm_dai); - if (ret < 0) - omap_mcpdm_remove(pdev); + if (!ret) + return 0; + +err_irq: + iounmap(mcpdm->io_base); +err_iomap: + release_mem_region(res->start, resource_size(res)); +err_res: + kfree(mcpdm); return ret; } static int __devexit asoc_mcpdm_remove(struct platform_device *pdev) { + struct omap_mcpdm *mcpdm = platform_get_drvdata(pdev); + struct resource *res; + snd_soc_unregister_dai(&pdev->dev); - omap_mcpdm_remove(pdev); + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + iounmap(mcpdm->io_base); + release_mem_region(res->start, resource_size(res)); + + kfree(mcpdm); return 0; } static struct platform_driver asoc_mcpdm_driver = { .driver = { - .name = "omap-mcpdm", - .owner = THIS_MODULE, + .name = "omap-mcpdm", + .owner = THIS_MODULE, }, - .probe = asoc_mcpdm_probe, - .remove = __devexit_p(asoc_mcpdm_remove), + .probe = asoc_mcpdm_probe, + .remove = __devexit_p(asoc_mcpdm_remove), }; static int __init snd_omap_mcpdm_init(void) @@ -274,6 +505,6 @@ static void __exit snd_omap_mcpdm_exit(void) } module_exit(snd_omap_mcpdm_exit); -MODULE_AUTHOR("Misael Lopez Cruz "); +MODULE_AUTHOR("Misael Lopez Cruz "); MODULE_DESCRIPTION("OMAP PDM SoC Interface"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/omap/omap-mcpdm.h b/sound/soc/omap/omap-mcpdm.h new file mode 100644 index 0000000..d23122a --- /dev/null +++ b/sound/soc/omap/omap-mcpdm.h @@ -0,0 +1,95 @@ +/* + * omap-mcpdm.h + * + * Copyright (C) 2009 - 2011 Texas Instruments + * + * Contact: Misael Lopez Cruz + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#ifndef __OMAP_MCPDM_H__ +#define __OMAP_MCPDM_H__ + +#define MCPDM_REG_REVISION 0x00 +#define MCPDM_REG_SYSCONFIG 0x10 +#define MCPDM_REG_IRQSTATUS_RAW 0x24 +#define MCPDM_REG_IRQSTATUS 0x28 +#define MCPDM_REG_IRQENABLE_SET 0x2C +#define MCPDM_REG_IRQENABLE_CLR 0x30 +#define MCPDM_REG_IRQWAKE_EN 0x34 +#define MCPDM_REG_DMAENABLE_SET 0x38 +#define MCPDM_REG_DMAENABLE_CLR 0x3C +#define MCPDM_REG_DMAWAKEEN 0x40 +#define MCPDM_REG_CTRL 0x44 +#define MCPDM_REG_DN_DATA 0x48 +#define MCPDM_REG_UP_DATA 0x4C +#define MCPDM_REG_FIFO_CTRL_DN 0x50 +#define MCPDM_REG_FIFO_CTRL_UP 0x54 +#define MCPDM_REG_DN_OFFSET 0x58 + +/* + * MCPDM_IRQ bit fields + * IRQSTATUS_RAW, IRQSTATUS, IRQENABLE_SET, IRQENABLE_CLR + */ + +#define MCPDM_DN_IRQ (1 << 0) +#define MCPDM_DN_IRQ_EMPTY (1 << 1) +#define MCPDM_DN_IRQ_ALMST_EMPTY (1 << 2) +#define MCPDM_DN_IRQ_FULL (1 << 3) + +#define MCPDM_UP_IRQ (1 << 8) +#define MCPDM_UP_IRQ_EMPTY (1 << 9) +#define MCPDM_UP_IRQ_ALMST_FULL (1 << 10) +#define MCPDM_UP_IRQ_FULL (1 << 11) + +#define MCPDM_DOWNLINK_IRQ_MASK 0x00F +#define MCPDM_UPLINK_IRQ_MASK 0xF00 + +/* + * MCPDM_DMAENABLE bit fields + */ + +#define MCPDM_DMA_DN_ENABLE (1 << 0) +#define MCPDM_DMA_UP_ENABLE (1 << 1) + +/* + * MCPDM_CTRL bit fields + */ + +#define MCPDM_PDM_UPLINK_EN(x) (1 << (x - 1)) /* ch1 is at bit 0 */ +#define MCPDM_PDM_DOWNLINK_EN(x) (1 << (x + 2)) /* ch1 is at bit 3 */ +#define MCPDM_PDMOUTFORMAT (1 << 8) +#define MCPDM_CMD_INT (1 << 9) +#define MCPDM_STATUS_INT (1 << 10) +#define MCPDM_SW_UP_RST (1 << 11) +#define MCPDM_SW_DN_RST (1 << 12) +#define MCPDM_WD_EN (1 << 14) +#define MCPDM_PDM_UP_MASK 0x7 +#define MCPDM_PDM_DN_MASK (0x1f << 3) + + +#define MCPDM_PDMOUTFORMAT_LJUST (0 << 8) +#define MCPDM_PDMOUTFORMAT_RJUST (1 << 8) + +/* + * MCPDM_FIFO_CTRL bit fields + */ + +#define MCPDM_UP_THRES_MAX 0xF +#define MCPDM_DN_THRES_MAX 0xF + +#endif /* End of __OMAP_MCPDM_H__ */ diff --git a/sound/soc/omap/sdp4430.c b/sound/soc/omap/sdp4430.c index 32782b9..a951774 100644 --- a/sound/soc/omap/sdp4430.c +++ b/sound/soc/omap/sdp4430.c @@ -32,7 +32,7 @@ #include #include -#include "mcpdm.h" +#include "omap-mcpdm.h" #include "omap-pcm.h" #include "../codecs/twl6040.h" -- cgit v1.1 From 8af0894546ef781bd613fa73a9df414c83c547a0 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 22 Sep 2011 11:16:10 +0100 Subject: ASoC: Include delay.h in 88pm860x Reported-by: Stephen Rothwell Signed-off-by: Mark Brown --- sound/soc/codecs/88pm860x-codec.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/codecs/88pm860x-codec.c b/sound/soc/codecs/88pm860x-codec.c index df7b4a0..5ca122e 100644 --- a/sound/soc/codecs/88pm860x-codec.c +++ b/sound/soc/codecs/88pm860x-codec.c @@ -15,6 +15,7 @@ #include #include #include +#include #include #include #include -- cgit v1.1 From 3acef6854c440b29f20d7ea0ec5f4707aad23923 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 22 Sep 2011 11:05:45 +0300 Subject: ASoC: twl6040: Lower the power on gain values at startup The default gains on outputs/inputs are set to 0dB. This is fixing the pop noise issue at the first playback, which caused by the wrong starting point of the ramp code. The ramp code for the outputs expects the gains to be in their lowest configuration in order to be effective. After the playback stops, the ramp code takes care of ramping down the gains to their minimum. Signed-off-by: Peter Ujfalusi Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/twl6040.c | 11 +++++++++++ 1 file changed, 11 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 81645c6..0694d65 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -269,6 +269,17 @@ static void twl6040_init_chip(struct snd_soc_codec *codec) /* No imput selected for microphone amplifiers */ twl6040_write_reg_cache(codec, TWL6040_REG_MICLCTL, 0x18); twl6040_write_reg_cache(codec, TWL6040_REG_MICRCTL, 0x18); + + /* + * We need to lower the default gain values, so the ramp code + * can work correctly for the first playback. + * This reduces the pop noise heard at the first playback. + */ + twl6040_write_reg_cache(codec, TWL6040_REG_HSGAIN, 0xff); + twl6040_write_reg_cache(codec, TWL6040_REG_EARCTL, 0x1e); + twl6040_write_reg_cache(codec, TWL6040_REG_HFLGAIN, 0x1d); + twl6040_write_reg_cache(codec, TWL6040_REG_HFRGAIN, 0x1d); + twl6040_write_reg_cache(codec, TWL6040_REG_LINEGAIN, 0); } static void twl6040_restore_regs(struct snd_soc_codec *codec) -- cgit v1.1 From 4548dc3c05d304cc94a550c2457a3cc3ad429a86 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 22 Sep 2011 11:05:46 +0300 Subject: ASoC: twl6040: Fix comments for register names Change the register name strings in the comments for the twl6040_reg table, so it is easier to search for specific register. This is cosmetic change. Before we had for example: TWL6040_REG_HSLCTL as register definition. At the register table we had: TWL6040_HSLCTL Searching for TWL6040_HSLCTL resulted no hits. While if we look for REG_HSLCTL, we can find the places the register has been used. Signed-off-by: Peter Ujfalusi Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/twl6040.c | 94 +++++++++++++++++++++++----------------------- 1 file changed, 47 insertions(+), 47 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 0694d65..6f6b180 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -106,53 +106,53 @@ struct twl6040_data { * twl6040 register cache & default register settings */ static const u8 twl6040_reg[TWL6040_CACHEREGNUM] = { - 0x00, /* not used 0x00 */ - 0x4B, /* TWL6040_ASICID (ro) 0x01 */ - 0x00, /* TWL6040_ASICREV (ro) 0x02 */ - 0x00, /* TWL6040_INTID 0x03 */ - 0x00, /* TWL6040_INTMR 0x04 */ - 0x00, /* TWL6040_NCPCTRL 0x05 */ - 0x00, /* TWL6040_LDOCTL 0x06 */ - 0x60, /* TWL6040_HPPLLCTL 0x07 */ - 0x00, /* TWL6040_LPPLLCTL 0x08 */ - 0x4A, /* TWL6040_LPPLLDIV 0x09 */ - 0x00, /* TWL6040_AMICBCTL 0x0A */ - 0x00, /* TWL6040_DMICBCTL 0x0B */ - 0x00, /* TWL6040_MICLCTL 0x0C */ - 0x00, /* TWL6040_MICRCTL 0x0D */ - 0x00, /* TWL6040_MICGAIN 0x0E */ - 0x1B, /* TWL6040_LINEGAIN 0x0F */ - 0x00, /* TWL6040_HSLCTL 0x10 */ - 0x00, /* TWL6040_HSRCTL 0x11 */ - 0x00, /* TWL6040_HSGAIN 0x12 */ - 0x00, /* TWL6040_EARCTL 0x13 */ - 0x00, /* TWL6040_HFLCTL 0x14 */ - 0x00, /* TWL6040_HFLGAIN 0x15 */ - 0x00, /* TWL6040_HFRCTL 0x16 */ - 0x00, /* TWL6040_HFRGAIN 0x17 */ - 0x00, /* TWL6040_VIBCTLL 0x18 */ - 0x00, /* TWL6040_VIBDATL 0x19 */ - 0x00, /* TWL6040_VIBCTLR 0x1A */ - 0x00, /* TWL6040_VIBDATR 0x1B */ - 0x00, /* TWL6040_HKCTL1 0x1C */ - 0x00, /* TWL6040_HKCTL2 0x1D */ - 0x00, /* TWL6040_GPOCTL 0x1E */ - 0x00, /* TWL6040_ALB 0x1F */ - 0x00, /* TWL6040_DLB 0x20 */ - 0x00, /* not used 0x21 */ - 0x00, /* not used 0x22 */ - 0x00, /* not used 0x23 */ - 0x00, /* not used 0x24 */ - 0x00, /* not used 0x25 */ - 0x00, /* not used 0x26 */ - 0x00, /* not used 0x27 */ - 0x00, /* TWL6040_TRIM1 0x28 */ - 0x00, /* TWL6040_TRIM2 0x29 */ - 0x00, /* TWL6040_TRIM3 0x2A */ - 0x00, /* TWL6040_HSOTRIM 0x2B */ - 0x00, /* TWL6040_HFOTRIM 0x2C */ - 0x09, /* TWL6040_ACCCTL 0x2D */ - 0x00, /* TWL6040_STATUS (ro) 0x2E */ + 0x00, /* not used 0x00 */ + 0x4B, /* REG_ASICID 0x01 (ro) */ + 0x00, /* REG_ASICREV 0x02 (ro) */ + 0x00, /* REG_INTID 0x03 */ + 0x00, /* REG_INTMR 0x04 */ + 0x00, /* REG_NCPCTRL 0x05 */ + 0x00, /* REG_LDOCTL 0x06 */ + 0x60, /* REG_HPPLLCTL 0x07 */ + 0x00, /* REG_LPPLLCTL 0x08 */ + 0x4A, /* REG_LPPLLDIV 0x09 */ + 0x00, /* REG_AMICBCTL 0x0A */ + 0x00, /* REG_DMICBCTL 0x0B */ + 0x00, /* REG_MICLCTL 0x0C */ + 0x00, /* REG_MICRCTL 0x0D */ + 0x00, /* REG_MICGAIN 0x0E */ + 0x1B, /* REG_LINEGAIN 0x0F */ + 0x00, /* REG_HSLCTL 0x10 */ + 0x00, /* REG_HSRCTL 0x11 */ + 0x00, /* REG_HSGAIN 0x12 */ + 0x00, /* REG_EARCTL 0x13 */ + 0x00, /* REG_HFLCTL 0x14 */ + 0x00, /* REG_HFLGAIN 0x15 */ + 0x00, /* REG_HFRCTL 0x16 */ + 0x00, /* REG_HFRGAIN 0x17 */ + 0x00, /* REG_VIBCTLL 0x18 */ + 0x00, /* REG_VIBDATL 0x19 */ + 0x00, /* REG_VIBCTLR 0x1A */ + 0x00, /* REG_VIBDATR 0x1B */ + 0x00, /* REG_HKCTL1 0x1C */ + 0x00, /* REG_HKCTL2 0x1D */ + 0x00, /* REG_GPOCTL 0x1E */ + 0x00, /* REG_ALB 0x1F */ + 0x00, /* REG_DLB 0x20 */ + 0x00, /* not used 0x21 */ + 0x00, /* not used 0x22 */ + 0x00, /* not used 0x23 */ + 0x00, /* not used 0x24 */ + 0x00, /* not used 0x25 */ + 0x00, /* not used 0x26 */ + 0x00, /* not used 0x27 */ + 0x00, /* REG_TRIM1 0x28 */ + 0x00, /* REG_TRIM2 0x29 */ + 0x00, /* REG_TRIM3 0x2A */ + 0x00, /* REG_HSOTRIM 0x2B */ + 0x00, /* REG_HFOTRIM 0x2C */ + 0x09, /* REG_ACCCTL 0x2D */ + 0x00, /* REG_STATUS 0x2E (ro) */ }; /* List of registers to be restored after power up */ -- cgit v1.1 From 5bf692d97225a1e714cfd40a9a67401ebd630a7b Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 22 Sep 2011 11:05:47 +0300 Subject: ASoC: twl6040: Remove strings "NULL" from DAPM route Replace the string with plain NULL. Signed-off-by: Peter Ujfalusi Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/twl6040.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 6f6b180..9fbfe0e 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -1199,8 +1199,8 @@ static const struct snd_soc_dapm_route intercon[] = { {"ADC Right", NULL, "MicAmpR"}, /* AFM path */ - {"AFMAmpL", "NULL", "AFML"}, - {"AFMAmpR", "NULL", "AFMR"}, + {"AFMAmpL", NULL, "AFML"}, + {"AFMAmpR", NULL, "AFMR"}, {"HS Left Playback", "HS DAC", "HSDAC Left"}, {"HS Left Playback", "Line-In amp", "AFMAmpL"}, @@ -1208,8 +1208,8 @@ static const struct snd_soc_dapm_route intercon[] = { {"HS Right Playback", "HS DAC", "HSDAC Right"}, {"HS Right Playback", "Line-In amp", "AFMAmpR"}, - {"Headset Left Driver", "NULL", "HS Left Playback"}, - {"Headset Right Driver", "NULL", "HS Right Playback"}, + {"Headset Left Driver", NULL, "HS Left Playback"}, + {"Headset Right Driver", NULL, "HS Right Playback"}, {"HSOL", NULL, "Headset Left Driver"}, {"HSOR", NULL, "Headset Right Driver"}, -- cgit v1.1 From d17bf31832d30b91225a84b53fae380dbdd07d3d Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 22 Sep 2011 11:05:48 +0300 Subject: ASoC: twl6040: Introduce SW only shadow register Software only shadow register to be used by the driver. For example Earpiece path will need this shadow register. Signed-off-by: Peter Ujfalusi Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/twl6040.c | 19 ++++++++++++++++--- 1 file changed, 16 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 9fbfe0e..9635466 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -57,6 +57,10 @@ #define TWL6040_HF_VOL_MASK 0x1F #define TWL6040_HF_VOL_SHIFT 0 +/* Shadow register used by the driver */ +#define TWL6040_REG_SW_SHADOW 0x2F +#define TWL6040_CACHEREGNUM (TWL6040_REG_SW_SHADOW + 1) + struct twl6040_output { u16 active; u16 left_vol; @@ -153,6 +157,8 @@ static const u8 twl6040_reg[TWL6040_CACHEREGNUM] = { 0x00, /* REG_HFOTRIM 0x2C */ 0x09, /* REG_ACCCTL 0x2D */ 0x00, /* REG_STATUS 0x2E (ro) */ + + 0x00, /* REG_SW_SHADOW 0x2F - Shadow, non HW register */ }; /* List of registers to be restored after power up */ @@ -236,8 +242,12 @@ static int twl6040_read_reg_volatile(struct snd_soc_codec *codec, if (reg >= TWL6040_CACHEREGNUM) return -EIO; - value = twl6040_reg_read(twl6040, reg); - twl6040_write_reg_cache(codec, reg, value); + if (likely(reg < TWL6040_REG_SW_SHADOW)) { + value = twl6040_reg_read(twl6040, reg); + twl6040_write_reg_cache(codec, reg, value); + } else { + value = twl6040_read_reg_cache(codec, reg); + } return value; } @@ -254,7 +264,10 @@ static int twl6040_write(struct snd_soc_codec *codec, return -EIO; twl6040_write_reg_cache(codec, reg, value); - return twl6040_reg_write(twl6040, reg, value); + if (likely(reg < TWL6040_REG_SW_SHADOW)) + return twl6040_reg_write(twl6040, reg, value); + else + return 0; } static void twl6040_init_chip(struct snd_soc_codec *codec) -- cgit v1.1 From 317596a69453772dcba2ab1e6e041de69e762794 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 22 Sep 2011 11:05:49 +0300 Subject: ASoC: twl6040: Earphone path correction Fix the DAPM routing for the earphone path. Convert the DAPM_SWITCH_E to DAPM_OUT_DRV_E, so we can have correct power up, and down sequence for EP. Introduce mute control (Earphone Playback Switch) for users to enable/disable the EP path. Note: the EP does not have it's own dedicated DAC. EP is connected to HSL DAC. Signed-off-by: Peter Ujfalusi Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/twl6040.c | 17 ++++++++++++----- 1 file changed, 12 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 9635466..3450b11 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -61,6 +61,9 @@ #define TWL6040_REG_SW_SHADOW 0x2F #define TWL6040_CACHEREGNUM (TWL6040_REG_SW_SHADOW + 1) +/* TWL6040_REG_SW_SHADOW (0x2F) fields */ +#define TWL6040_EAR_PATH_ENABLE 0x01 + struct twl6040_output { u16 active; u16 left_vol; @@ -991,8 +994,8 @@ static const struct snd_kcontrol_new hfl_mux_controls = static const struct snd_kcontrol_new hfr_mux_controls = SOC_DAPM_ENUM("Route", twl6040_hf_enum[1]); -static const struct snd_kcontrol_new ep_driver_switch_controls = - SOC_DAPM_SINGLE("Switch", TWL6040_REG_EARCTL, 0, 1, 0); +static const struct snd_kcontrol_new ep_path_enable_control = + SOC_DAPM_SINGLE("Switch", TWL6040_REG_SW_SHADOW, 0, 1, 0); /* Headset power mode */ static const char *twl6040_power_mode_texts[] = { @@ -1165,6 +1168,9 @@ static const struct snd_soc_dapm_widget twl6040_dapm_widgets[] = { SND_SOC_DAPM_MUX("HS Right Playback", SND_SOC_NOPM, 0, 0, &hsr_mux_controls), + SND_SOC_DAPM_SWITCH("Earphone Playback", SND_SOC_NOPM, 0, 0, + &ep_path_enable_control), + /* Analog playback drivers */ SND_SOC_DAPM_OUT_DRV_E("Handsfree Left Driver", TWL6040_REG_HFLCTL, 4, 0, NULL, 0, @@ -1182,8 +1188,8 @@ static const struct snd_soc_dapm_widget twl6040_dapm_widgets[] = { TWL6040_REG_HSRCTL, 2, 0, NULL, 0, pga_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), - SND_SOC_DAPM_SWITCH_E("Earphone Driver", - SND_SOC_NOPM, 0, 0, &ep_driver_switch_controls, + SND_SOC_DAPM_OUT_DRV_E("Earphone Driver", + TWL6040_REG_EARCTL, 0, 0, NULL, 0, twl6040_power_mode_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), @@ -1228,7 +1234,8 @@ static const struct snd_soc_dapm_route intercon[] = { {"HSOR", NULL, "Headset Right Driver"}, /* Earphone playback path */ - {"Earphone Driver", "Switch", "HSDAC Left"}, + {"Earphone Playback", "Switch", "HSDAC Left"}, + {"Earphone Driver", NULL, "Earphone Playback"}, {"EP", NULL, "Earphone Driver"}, {"HF Left Playback", "HF DAC", "HFDAC Left"}, -- cgit v1.1 From df11ce295a0390428121b799696095a0ed017db9 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 22 Sep 2011 11:05:50 +0300 Subject: ASoC: twl6040: Use consistent names for Handsfree path Use "Handsfree XYZ" for user visible controls, while the internal DAPM widgets can use "HF XYZ". In this way we can group the Handsfree related controls in UI (alsamixer for example). Signed-off-by: Peter Ujfalusi Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/twl6040.c | 32 ++++++++++++++++---------------- 1 file changed, 16 insertions(+), 16 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 3450b11..10f292d 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -1158,9 +1158,9 @@ static const struct snd_soc_dapm_widget twl6040_dapm_widgets[] = { twl6040_power_mode_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), - SND_SOC_DAPM_MUX("HF Left Playback", + SND_SOC_DAPM_MUX("Handsfree Left Playback", SND_SOC_NOPM, 0, 0, &hfl_mux_controls), - SND_SOC_DAPM_MUX("HF Right Playback", + SND_SOC_DAPM_MUX("Handsfree Right Playback", SND_SOC_NOPM, 0, 0, &hfr_mux_controls), /* Analog playback Muxes */ SND_SOC_DAPM_MUX("HS Left Playback", @@ -1172,11 +1172,11 @@ static const struct snd_soc_dapm_widget twl6040_dapm_widgets[] = { &ep_path_enable_control), /* Analog playback drivers */ - SND_SOC_DAPM_OUT_DRV_E("Handsfree Left Driver", + SND_SOC_DAPM_OUT_DRV_E("HF Left Driver", TWL6040_REG_HFLCTL, 4, 0, NULL, 0, pga_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), - SND_SOC_DAPM_OUT_DRV_E("Handsfree Right Driver", + SND_SOC_DAPM_OUT_DRV_E("HF Right Driver", TWL6040_REG_HFRCTL, 4, 0, NULL, 0, pga_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), @@ -1194,9 +1194,9 @@ static const struct snd_soc_dapm_widget twl6040_dapm_widgets[] = { SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), /* Analog playback PGAs */ - SND_SOC_DAPM_PGA("HFDAC Left PGA", + SND_SOC_DAPM_PGA("HF Left PGA", TWL6040_REG_HFLCTL, 1, 0, NULL, 0), - SND_SOC_DAPM_PGA("HFDAC Right PGA", + SND_SOC_DAPM_PGA("HF Right PGA", TWL6040_REG_HFRCTL, 1, 0, NULL, 0), }; @@ -1238,20 +1238,20 @@ static const struct snd_soc_dapm_route intercon[] = { {"Earphone Driver", NULL, "Earphone Playback"}, {"EP", NULL, "Earphone Driver"}, - {"HF Left Playback", "HF DAC", "HFDAC Left"}, - {"HF Left Playback", "Line-In amp", "AFMAmpL"}, + {"Handsfree Left Playback", "HF DAC", "HFDAC Left"}, + {"Handsfree Left Playback", "Line-In amp", "AFMAmpL"}, - {"HF Right Playback", "HF DAC", "HFDAC Right"}, - {"HF Right Playback", "Line-In amp", "AFMAmpR"}, + {"Handsfree Right Playback", "HF DAC", "HFDAC Right"}, + {"Handsfree Right Playback", "Line-In amp", "AFMAmpR"}, - {"HFDAC Left PGA", NULL, "HF Left Playback"}, - {"HFDAC Right PGA", NULL, "HF Right Playback"}, + {"HF Left PGA", NULL, "Handsfree Left Playback"}, + {"HF Right PGA", NULL, "Handsfree Right Playback"}, - {"Handsfree Left Driver", "Switch", "HFDAC Left PGA"}, - {"Handsfree Right Driver", "Switch", "HFDAC Right PGA"}, + {"HF Left Driver", NULL, "HF Left PGA"}, + {"HF Right Driver", NULL, "HF Right PGA"}, - {"HFL", NULL, "Handsfree Left Driver"}, - {"HFR", NULL, "Handsfree Right Driver"}, + {"HFL", NULL, "HF Left Driver"}, + {"HFR", NULL, "HF Right Driver"}, }; static int twl6040_add_widgets(struct snd_soc_codec *codec) -- cgit v1.1 From 45b0f60de2525dc29ee309eccdf3d9a64260d83d Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 22 Sep 2011 11:05:51 +0300 Subject: ASoC: twl6040: Use consistent names for Headset path Use "Headset XYZ" for user visible controls, while the internal DAPM widgets can use "HS XYZ". In this way we can group the Headset related controls in UI (alsamixer for example). Signed-off-by: Peter Ujfalusi Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/twl6040.c | 24 ++++++++++++------------ 1 file changed, 12 insertions(+), 12 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 10f292d..fffd7ff 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -1163,9 +1163,9 @@ static const struct snd_soc_dapm_widget twl6040_dapm_widgets[] = { SND_SOC_DAPM_MUX("Handsfree Right Playback", SND_SOC_NOPM, 0, 0, &hfr_mux_controls), /* Analog playback Muxes */ - SND_SOC_DAPM_MUX("HS Left Playback", + SND_SOC_DAPM_MUX("Headset Left Playback", SND_SOC_NOPM, 0, 0, &hsl_mux_controls), - SND_SOC_DAPM_MUX("HS Right Playback", + SND_SOC_DAPM_MUX("Headset Right Playback", SND_SOC_NOPM, 0, 0, &hsr_mux_controls), SND_SOC_DAPM_SWITCH("Earphone Playback", SND_SOC_NOPM, 0, 0, @@ -1180,11 +1180,11 @@ static const struct snd_soc_dapm_widget twl6040_dapm_widgets[] = { TWL6040_REG_HFRCTL, 4, 0, NULL, 0, pga_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), - SND_SOC_DAPM_OUT_DRV_E("Headset Left Driver", + SND_SOC_DAPM_OUT_DRV_E("HS Left Driver", TWL6040_REG_HSLCTL, 2, 0, NULL, 0, pga_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), - SND_SOC_DAPM_OUT_DRV_E("Headset Right Driver", + SND_SOC_DAPM_OUT_DRV_E("HS Right Driver", TWL6040_REG_HSRCTL, 2, 0, NULL, 0, pga_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), @@ -1221,17 +1221,17 @@ static const struct snd_soc_dapm_route intercon[] = { {"AFMAmpL", NULL, "AFML"}, {"AFMAmpR", NULL, "AFMR"}, - {"HS Left Playback", "HS DAC", "HSDAC Left"}, - {"HS Left Playback", "Line-In amp", "AFMAmpL"}, + {"Headset Left Playback", "HS DAC", "HSDAC Left"}, + {"Headset Left Playback", "Line-In amp", "AFMAmpL"}, - {"HS Right Playback", "HS DAC", "HSDAC Right"}, - {"HS Right Playback", "Line-In amp", "AFMAmpR"}, + {"Headset Right Playback", "HS DAC", "HSDAC Right"}, + {"Headset Right Playback", "Line-In amp", "AFMAmpR"}, - {"Headset Left Driver", NULL, "HS Left Playback"}, - {"Headset Right Driver", NULL, "HS Right Playback"}, + {"HS Left Driver", NULL, "Headset Left Playback"}, + {"HS Right Driver", NULL, "Headset Right Playback"}, - {"HSOL", NULL, "Headset Left Driver"}, - {"HSOR", NULL, "Headset Right Driver"}, + {"HSOL", NULL, "HS Left Driver"}, + {"HSOR", NULL, "HS Right Driver"}, /* Earphone playback path */ {"Earphone Playback", "Switch", "HSDAC Left"}, -- cgit v1.1 From fdb625ffd26cc3f6bd171fa61854083540bc28f8 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 22 Sep 2011 11:05:52 +0300 Subject: ASoC: twl6040: Support for AUX L/R output AUX L/R outputs can be driver from the Handsfree PGA output. Signed-off-by: Peter Ujfalusi Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/twl6040.c | 18 ++++++++++++++++++ 1 file changed, 18 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index fffd7ff..3908b88 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -997,6 +997,12 @@ static const struct snd_kcontrol_new hfr_mux_controls = static const struct snd_kcontrol_new ep_path_enable_control = SOC_DAPM_SINGLE("Switch", TWL6040_REG_SW_SHADOW, 0, 1, 0); +static const struct snd_kcontrol_new auxl_switch_control = + SOC_DAPM_SINGLE("Switch", TWL6040_REG_HFLCTL, 6, 1, 0); + +static const struct snd_kcontrol_new auxr_switch_control = + SOC_DAPM_SINGLE("Switch", TWL6040_REG_HFRCTL, 6, 1, 0); + /* Headset power mode */ static const char *twl6040_power_mode_texts[] = { "Low-Power", "High-Perfomance", @@ -1105,6 +1111,8 @@ static const struct snd_soc_dapm_widget twl6040_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("HFL"), SND_SOC_DAPM_OUTPUT("HFR"), SND_SOC_DAPM_OUTPUT("EP"), + SND_SOC_DAPM_OUTPUT("AUXL"), + SND_SOC_DAPM_OUTPUT("AUXR"), /* Analog input muxes for the capture amplifiers */ SND_SOC_DAPM_MUX("Analog Left Capture Route", @@ -1170,6 +1178,10 @@ static const struct snd_soc_dapm_widget twl6040_dapm_widgets[] = { SND_SOC_DAPM_SWITCH("Earphone Playback", SND_SOC_NOPM, 0, 0, &ep_path_enable_control), + SND_SOC_DAPM_SWITCH("AUXL Playback", SND_SOC_NOPM, 0, 0, + &auxl_switch_control), + SND_SOC_DAPM_SWITCH("AUXR Playback", SND_SOC_NOPM, 0, 0, + &auxr_switch_control), /* Analog playback drivers */ SND_SOC_DAPM_OUT_DRV_E("HF Left Driver", @@ -1252,6 +1264,12 @@ static const struct snd_soc_dapm_route intercon[] = { {"HFL", NULL, "HF Left Driver"}, {"HFR", NULL, "HF Right Driver"}, + + {"AUXL Playback", "Switch", "HF Left PGA"}, + {"AUXR Playback", "Switch", "HF Right PGA"}, + + {"AUXL", NULL, "AUXL Playback"}, + {"AUXR", NULL, "AUXR Playback"}, }; static int twl6040_add_widgets(struct snd_soc_codec *codec) -- cgit v1.1 From d13f1fe04412b2319a79ff456cf73cc59692f6fb Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 22 Sep 2011 11:05:53 +0300 Subject: ASoC: twl6040/sdp4430: Change legacy DAI name Change the legacy DAI name from "twl6040-hifi" to "twl6040-legacy" to be more intuitive. Signed-off-by: Peter Ujfalusi Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/twl6040.c | 2 +- sound/soc/omap/sdp4430.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 3908b88..760701e 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -1444,7 +1444,7 @@ static struct snd_soc_dai_ops twl6040_dai_ops = { static struct snd_soc_dai_driver twl6040_dai[] = { { - .name = "twl6040-hifi", + .name = "twl6040-legacy", .playback = { .stream_name = "Playback", .channels_min = 1, diff --git a/sound/soc/omap/sdp4430.c b/sound/soc/omap/sdp4430.c index a951774..4200eb4 100644 --- a/sound/soc/omap/sdp4430.c +++ b/sound/soc/omap/sdp4430.c @@ -166,7 +166,7 @@ static struct snd_soc_dai_link sdp4430_dai = { .name = "TWL6040", .stream_name = "TWL6040", .cpu_dai_name = "omap-mcpdm", - .codec_dai_name = "twl6040-hifi", + .codec_dai_name = "twl6040-legacy", .platform_name = "omap-pcm-audio", .codec_name = "twl6040-codec", .init = sdp4430_twl6040_init, -- cgit v1.1 From ab6cf13943303f865320407b17b0f86095d23ce3 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 22 Sep 2011 11:05:54 +0300 Subject: ASoC/MFD: twl6040: Combine bit definitions for Headset control registers Use one set of defines for the HS bits, since they are identical in both control register. Signed-off-by: Peter Ujfalusi Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/twl6040.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 760701e..68e52c9 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -642,7 +642,7 @@ static int pga_event(struct snd_soc_dapm_widget *w, static int headset_power_mode(struct snd_soc_codec *codec, int high_perf) { int hslctl, hsrctl; - int mask = TWL6040_HSDRVMODEL | TWL6040_HSDACMODEL; + int mask = TWL6040_HSDRVMODE | TWL6040_HSDACMODE; hslctl = twl6040_read_reg_cache(codec, TWL6040_REG_HSLCTL); hsrctl = twl6040_read_reg_cache(codec, TWL6040_REG_HSRCTL); -- cgit v1.1 From 7aefb086c15fc44066e705e479d012d46476d8c5 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 21 Sep 2011 17:59:02 +0100 Subject: ASoC: Dynamically manage DBVDD2 and DBVDD3 on WM5100 Allow the DBVDD2 and DBVDD3 rails to be powered down when idle, helping fully power down connected devices when idle. Signed-off-by: Mark Brown --- sound/soc/codecs/wm5100.c | 82 ++++++++++++++++++++++++++++++++++++++++++++++- 1 file changed, 81 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index 576081a..443d76d 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -54,6 +54,8 @@ struct wm5100_priv { struct regulator_bulk_data core_supplies[WM5100_NUM_CORE_SUPPLIES]; struct regulator *cpvdd; + struct regulator *dbvdd2; + struct regulator *dbvdd3; int rev; @@ -803,6 +805,52 @@ static int wm5100_cp_ev(struct snd_soc_dapm_widget *w, } } +static int wm5100_dbvdd_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, + int event) +{ + struct snd_soc_codec *codec = w->codec; + struct wm5100_priv *wm5100 = snd_soc_codec_get_drvdata(codec); + struct regulator *regulator; + int ret; + + switch (w->shift) { + case 2: + regulator = wm5100->dbvdd2; + break; + case 3: + regulator = wm5100->dbvdd3; + break; + default: + BUG(); + return 0; + } + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + ret = regulator_enable(regulator); + if (ret != 0) { + dev_err(codec->dev, "Failed to enable DBVDD%d: %d\n", + w->shift, ret); + return ret; + } + return ret; + + case SND_SOC_DAPM_POST_PMD: + ret = regulator_disable(regulator); + if (ret != 0) { + dev_err(codec->dev, "Failed to enable DBVDD%d: %d\n", + w->shift, ret); + return ret; + } + return ret; + + default: + BUG(); + return 0; + } +} + static void wm5100_log_status3(struct snd_soc_codec *codec, int val) { if (val & WM5100_SPK_SHUTDOWN_WARN_EINT) @@ -880,6 +928,10 @@ SND_SOC_DAPM_SUPPLY("CP2", WM5100_MIC_CHARGE_PUMP_1, WM5100_CP2_ENA_SHIFT, 0, SND_SOC_DAPM_SUPPLY("CP2 Active", WM5100_MIC_CHARGE_PUMP_1, WM5100_CP2_BYPASS_SHIFT, 1, wm5100_cp_ev, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), +SND_SOC_DAPM_SUPPLY("DBVDD2", SND_SOC_NOPM, 2, 0, wm5100_dbvdd_ev, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), +SND_SOC_DAPM_SUPPLY("DBVDD3", SND_SOC_NOPM, 3, 0, wm5100_dbvdd_ev, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_SUPPLY("MICBIAS1", WM5100_MIC_BIAS_CTRL_1, WM5100_MICB1_ENA_SHIFT, 0, NULL, 0), @@ -1122,10 +1174,14 @@ static const struct snd_soc_dapm_route wm5100_dapm_routes[] = { { "AIF1RX8", NULL, "SYSCLK" }, { "AIF2RX1", NULL, "SYSCLK" }, + { "AIF2RX1", NULL, "DBVDD2" }, { "AIF2RX2", NULL, "SYSCLK" }, + { "AIF2RX2", NULL, "DBVDD2" }, { "AIF3RX1", NULL, "SYSCLK" }, + { "AIF3RX1", NULL, "DBVDD3" }, { "AIF3RX2", NULL, "SYSCLK" }, + { "AIF3RX2", NULL, "DBVDD3" }, { "AIF1TX1", NULL, "SYSCLK" }, { "AIF1TX2", NULL, "SYSCLK" }, @@ -1137,10 +1193,14 @@ static const struct snd_soc_dapm_route wm5100_dapm_routes[] = { { "AIF1TX8", NULL, "SYSCLK" }, { "AIF2TX1", NULL, "SYSCLK" }, + { "AIF2TX1", NULL, "DBVDD2" }, { "AIF2TX2", NULL, "SYSCLK" }, + { "AIF2TX2", NULL, "DBVDD2" }, { "AIF3TX1", NULL, "SYSCLK" }, + { "AIF3TX1", NULL, "DBVDD3" }, { "AIF3TX2", NULL, "SYSCLK" }, + { "AIF3TX2", NULL, "DBVDD3" }, { "MICBIAS1", NULL, "CP2" }, { "MICBIAS2", NULL, "CP2" }, @@ -2250,12 +2310,26 @@ static int wm5100_probe(struct snd_soc_codec *codec) goto err_core; } + wm5100->dbvdd2 = regulator_get(&i2c->dev, "DBVDD2"); + if (IS_ERR(wm5100->dbvdd2)) { + ret = PTR_ERR(wm5100->dbvdd2); + dev_err(&i2c->dev, "Failed to get DBVDD2: %d\n", ret); + goto err_cpvdd; + } + + wm5100->dbvdd3 = regulator_get(&i2c->dev, "DBVDD3"); + if (IS_ERR(wm5100->dbvdd3)) { + ret = PTR_ERR(wm5100->dbvdd3); + dev_err(&i2c->dev, "Failed to get DBVDD2: %d\n", ret); + goto err_dbvdd2; + } + ret = regulator_bulk_enable(ARRAY_SIZE(wm5100->core_supplies), wm5100->core_supplies); if (ret != 0) { dev_err(codec->dev, "Failed to enable core supplies: %d\n", ret); - goto err_cpvdd; + goto err_dbvdd3; } if (wm5100->pdata.ldo_ena) { @@ -2432,6 +2506,10 @@ err_ldo: err_enable: regulator_bulk_disable(ARRAY_SIZE(wm5100->core_supplies), wm5100->core_supplies); +err_dbvdd3: + regulator_put(wm5100->dbvdd3); +err_dbvdd2: + regulator_put(wm5100->dbvdd2); err_cpvdd: regulator_put(wm5100->cpvdd); err_core: @@ -2458,6 +2536,8 @@ static int wm5100_remove(struct snd_soc_codec *codec) gpio_set_value_cansleep(wm5100->pdata.ldo_ena, 0); gpio_free(wm5100->pdata.ldo_ena); } + regulator_put(wm5100->dbvdd3); + regulator_put(wm5100->dbvdd2); regulator_put(wm5100->cpvdd); regulator_bulk_free(ARRAY_SIZE(wm5100->core_supplies), wm5100->core_supplies); -- cgit v1.1 From e56235e099d7290a2331b984a79f75bbe0865fe8 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 21 Sep 2011 18:19:14 +0100 Subject: ASoC: Add another DAPM stat for neighbour checks The number of times we look at a potentially connected neighbour is just as important as the number of times we actually recurse into looking at that neighbour so also collect that statistic. Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 84d1d79..6cac045 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -677,6 +677,8 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget) } list_for_each_entry(path, &widget->sinks, list_source) { + DAPM_UPDATE_STAT(widget, neighbour_checks); + if (path->weak) continue; @@ -732,6 +734,8 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget) } list_for_each_entry(path, &widget->sources, list_sink) { + DAPM_UPDATE_STAT(widget, neighbour_checks); + if (path->weak) continue; -- cgit v1.1 From d73ec75cc469107e182cf6210cb5f7b760cda339 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 22 Sep 2011 17:48:01 +0100 Subject: ASoC: Add missed BCLK rate to WM5100 driver Reported-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/wm5100.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index 443d76d..cb940a8 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -1511,6 +1511,7 @@ static int wm5100_bclk_rates_dat[WM5100_NUM_BCLK_RATES] = { 96000, 128000, 192000, + 256000, 384000, 512000, 768000, -- cgit v1.1 From 689b956e2cf87bf3f67163964d69bca97befafaa Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Thu, 22 Sep 2011 20:52:12 +0800 Subject: ASoC: Add Kconfig and Makefile entries for rt5631 codec Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 4 ++++ sound/soc/codecs/Makefile | 2 ++ 2 files changed, 6 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 45c966c..3449431 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -40,6 +40,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_MAX9850 if I2C select SND_SOC_MAX9877 if I2C select SND_SOC_PCM3008 + select SND_SOC_RT5631 if I2C select SND_SOC_SGTL5000 if I2C select SND_SOC_SN95031 if INTEL_SCU_IPC select SND_SOC_SPDIF @@ -219,6 +220,9 @@ config SND_SOC_MAX9850 config SND_SOC_PCM3008 tristate +config SND_SOC_RT5631 + tristate + #Freescale sgtl5000 codec config SND_SOC_SGTL5000 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 4f3ff24..787881b 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -26,6 +26,7 @@ snd-soc-max98088-objs := max98088.o snd-soc-max98095-objs := max98095.o snd-soc-max9850-objs := max9850.o snd-soc-pcm3008-objs := pcm3008.o +snd-soc-rt5631-objs := rt5631.o snd-soc-sgtl5000-objs := sgtl5000.o snd-soc-alc5623-objs := alc5623.o snd-soc-sn95031-objs := sn95031.o @@ -126,6 +127,7 @@ obj-$(CONFIG_SND_SOC_MAX98088) += snd-soc-max98088.o obj-$(CONFIG_SND_SOC_MAX98095) += snd-soc-max98095.o obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o +obj-$(CONFIG_SND_SOC_RT5631) += snd-soc-rt5631.o obj-$(CONFIG_SND_SOC_SGTL5000) += snd-soc-sgtl5000.o obj-$(CONFIG_SND_SOC_SN95031) +=snd-soc-sn95031.o obj-$(CONFIG_SND_SOC_SPDIF) += snd-soc-spdif.o -- cgit v1.1 From 3b5b516fbf7a057b6e2d115c301fec2e206eb6ea Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 23 Sep 2011 09:49:43 +0300 Subject: ASoC: omap-mcpdm: Correct the supported number of channels OMAP4 McPDM supports 5 downlink (playback), and 3 uplink (capture) channels. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/omap/omap-mcpdm.c | 14 ++++++++------ 1 file changed, 8 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/omap-mcpdm.c b/sound/soc/omap/omap-mcpdm.c index 159a5e98..2c9fa51 100644 --- a/sound/soc/omap/omap-mcpdm.c +++ b/sound/soc/omap/omap-mcpdm.c @@ -299,15 +299,17 @@ static int omap_mcpdm_dai_hw_params(struct snd_pcm_substream *substream, channels = params_channels(params); switch (channels) { + case 5: + if (stream == SNDRV_PCM_STREAM_CAPTURE) + /* up to 3 channels for capture */ + return -EINVAL; + link_mask |= 1 << 4; case 4: if (stream == SNDRV_PCM_STREAM_CAPTURE) - /* up to 2 channels for capture */ + /* up to 3 channels for capture */ return -EINVAL; link_mask |= 1 << 3; case 3: - if (stream == SNDRV_PCM_STREAM_CAPTURE) - /* up to 2 channels for capture */ - return -EINVAL; link_mask |= 1 << 2; case 2: link_mask |= 1 << 1; @@ -403,13 +405,13 @@ static struct snd_soc_dai_driver omap_mcpdm_dai = { .remove_order = SND_SOC_COMP_ORDER_EARLY, .playback = { .channels_min = 1, - .channels_max = 4, + .channels_max = 5, .rates = OMAP_MCPDM_RATES, .formats = OMAP_MCPDM_FORMATS, }, .capture = { .channels_min = 1, - .channels_max = 2, + .channels_max = 3, .rates = OMAP_MCPDM_RATES, .formats = OMAP_MCPDM_FORMATS, }, -- cgit v1.1 From f34c660662cc2b6e133083160bf6a3c77f11886e Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 23 Sep 2011 09:52:02 +0300 Subject: ASoC: twl6040: No need to read the INTID register Since our irq handler has been called, it is granted, that the reason was either PLUGINT, or UNPLUGINT. The INTID register has been checked in the MFD part of twl6040 driver (twl6040-irq.c). We have no reason to read from chip again here. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl6040.c | 9 ++------- 1 file changed, 2 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 68e52c9..91b9818 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -740,15 +740,10 @@ static void twl6040_accessory_work(struct work_struct *work) static irqreturn_t twl6040_audio_handler(int irq, void *data) { struct snd_soc_codec *codec = data; - struct twl6040 *twl6040 = codec->control_data; struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec); - u8 intid; - - intid = twl6040_reg_read(twl6040, TWL6040_REG_INTID); - if ((intid & TWL6040_PLUGINT) || (intid & TWL6040_UNPLUGINT)) - queue_delayed_work(priv->workqueue, &priv->delayed_work, - msecs_to_jiffies(200)); + queue_delayed_work(priv->workqueue, &priv->delayed_work, + msecs_to_jiffies(200)); return IRQ_HANDLED; } -- cgit v1.1 From 51e19fb385b6e424d1b21785744de1f40354b810 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Fri, 23 Sep 2011 16:22:07 +0800 Subject: ASoC: Staticize rt5631_dai Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/rt5631.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt5631.c b/sound/soc/codecs/rt5631.c index 110b3d8..889a7dd 100644 --- a/sound/soc/codecs/rt5631.c +++ b/sound/soc/codecs/rt5631.c @@ -1687,7 +1687,7 @@ struct snd_soc_dai_ops rt5631_ops = { .set_pll = rt5631_codec_set_dai_pll, }; -struct snd_soc_dai_driver rt5631_dai[] = { +static struct snd_soc_dai_driver rt5631_dai[] = { { .name = "rt5631-hifi", .id = 1, -- cgit v1.1 From a436089b77ff9e1ea4ee982a6b4b2fa411cd3016 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Fri, 23 Sep 2011 16:24:19 +0800 Subject: ASoC: Staticize sn95031_dais Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/sn95031.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c index 29945b0..d1781d1 100644 --- a/sound/soc/codecs/sn95031.c +++ b/sound/soc/codecs/sn95031.c @@ -716,7 +716,7 @@ static struct snd_soc_dai_ops sn95031_vib2_dai_ops = { .hw_params = sn95031_pcm_hw_params, }; -struct snd_soc_dai_driver sn95031_dais[] = { +static struct snd_soc_dai_driver sn95031_dais[] = { { .name = "SN95031 Headset", .playback = { -- cgit v1.1 From 0010bcc2260e3c139c8f54ac452a6d0f7aa45db1 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Fri, 23 Sep 2011 13:10:57 +0800 Subject: ASoC: Remove unneeded mutex_init in wl1273_probe() Since f0fba2ad "ASoC: multi-component - ASoC Multi-Component Support", snd_soc_register_codec() now does all the codec list and mutex init. Thus don't need to call mutex_init(&codec->mutex) in wl1273_probe() any more. Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/wl1273.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wl1273.c b/sound/soc/codecs/wl1273.c index 5836201..9fa1429 100644 --- a/sound/soc/codecs/wl1273.c +++ b/sound/soc/codecs/wl1273.c @@ -462,7 +462,6 @@ static int wl1273_probe(struct snd_soc_codec *codec) wl1273->core = *core; snd_soc_codec_set_drvdata(codec, wl1273); - mutex_init(&codec->mutex); r = snd_soc_add_controls(codec, wl1273_controls, ARRAY_SIZE(wl1273_controls)); -- cgit v1.1 From 0a742681e6072a71f30cfe6312f758f1cd185c21 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Fri, 23 Sep 2011 13:23:10 +0800 Subject: ASoC: Add missed free_irq in wm5100_remove and wm5100_probe error path Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/wm5100.c | 5 +++++ 1 file changed, 5 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index cb940a8..f603989 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -2493,6 +2493,8 @@ static int wm5100_probe(struct snd_soc_codec *codec) return 0; err_gpio: + if (i2c->irq) + free_irq(i2c->irq, codec); wm5100_free_gpio(codec); err_reset: if (wm5100->pdata.reset) { @@ -2523,11 +2525,14 @@ err_core: static int wm5100_remove(struct snd_soc_codec *codec) { struct wm5100_priv *wm5100 = snd_soc_codec_get_drvdata(codec); + struct i2c_client *i2c = to_i2c_client(codec->dev); wm5100_set_bias_level(codec, SND_SOC_BIAS_OFF); if (wm5100->pdata.hp_pol) { gpio_free(wm5100->pdata.hp_pol); } + if (i2c->irq) + free_irq(i2c->irq, codec); wm5100_free_gpio(codec); if (wm5100->pdata.reset) { gpio_set_value_cansleep(wm5100->pdata.reset, 1); -- cgit v1.1 From ef473fee67f88c0032eb5e55d940b81b08c0ee89 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 23 Sep 2011 16:05:00 +0100 Subject: ASoC: Support a wider range of sample rates on Speyside WM8962 As we've only got one audio interface and it is symmetric we can just set SYSCLK based on the sample rate requested by the application layer. Provide a default so bypass paths work before audio playback. Signed-off-by: Mark Brown --- sound/soc/samsung/speyside_wm8962.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/speyside_wm8962.c b/sound/soc/samsung/speyside_wm8962.c index 3820a6b..98b28ea 100644 --- a/sound/soc/samsung/speyside_wm8962.c +++ b/sound/soc/samsung/speyside_wm8962.c @@ -16,6 +16,8 @@ #include "../codecs/wm8962.h" +static int sample_rate = 44100; + static int speyside_wm8962_set_bias_level(struct snd_soc_card *card, struct snd_soc_dapm_context *dapm, enum snd_soc_bias_level level) @@ -31,13 +33,13 @@ static int speyside_wm8962_set_bias_level(struct snd_soc_card *card, if (dapm->bias_level == SND_SOC_BIAS_STANDBY) { ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL, WM8962_FLL_MCLK, 32768, - 44100 * 512); + sample_rate * 512); if (ret < 0) pr_err("Failed to start FLL: %d\n", ret); ret = snd_soc_dai_set_sysclk(codec_dai, WM8962_SYSCLK_FLL, - 44100 * 512, + sample_rate * 512, SND_SOC_CLOCK_IN); if (ret < 0) { pr_err("Failed to set SYSCLK: %d\n", ret); @@ -109,6 +111,8 @@ static int speyside_wm8962_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; + sample_rate = params_rate(params); + return 0; } -- cgit v1.1 From 3f7d55a19adbf37b5b91eea91b21f2209a1b9ca2 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 23 Sep 2011 16:39:31 +0100 Subject: ASoC: Rename WM8962 DMIC widget to DMIC_ENA Matches the register name and avoids confusion with board widgets. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8962.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 3676b38..d26ec6d 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2688,7 +2688,7 @@ SND_SOC_DAPM_MIXER("MIXINL", WM8962_PWR_MGMT_1, 5, 0, SND_SOC_DAPM_MIXER("MIXINR", WM8962_PWR_MGMT_1, 4, 0, mixinr, ARRAY_SIZE(mixinr)), -SND_SOC_DAPM_AIF_IN("DMIC", NULL, 0, WM8962_PWR_MGMT_1, 10, 0), +SND_SOC_DAPM_AIF_IN("DMIC_ENA", NULL, 0, WM8962_PWR_MGMT_1, 10, 0), SND_SOC_DAPM_ADC("ADCL", "Capture", WM8962_PWR_MGMT_1, 3, 0), SND_SOC_DAPM_ADC("ADCR", "Capture", WM8962_PWR_MGMT_1, 2, 0), @@ -2767,18 +2767,18 @@ static const struct snd_soc_dapm_route wm8962_intercon[] = { { "MICBIAS", NULL, "SYSCLK" }, - { "DMIC", NULL, "DMICDAT" }, + { "DMIC_ENA", NULL, "DMICDAT" }, { "ADCL", NULL, "SYSCLK" }, { "ADCL", NULL, "TOCLK" }, { "ADCL", NULL, "MIXINL" }, - { "ADCL", NULL, "DMIC" }, + { "ADCL", NULL, "DMIC_ENA" }, { "ADCL", NULL, "DSP2" }, { "ADCR", NULL, "SYSCLK" }, { "ADCR", NULL, "TOCLK" }, { "ADCR", NULL, "MIXINR" }, - { "ADCR", NULL, "DMIC" }, + { "ADCR", NULL, "DMIC_ENA" }, { "ADCR", NULL, "DSP2" }, { "STL", "Left", "ADCL" }, -- cgit v1.1 From 086d7f804e269454b4fffe757ed5517c3703baf8 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 23 Sep 2011 16:22:48 +0100 Subject: ASoC: Convert WM8962 MICBIAS to a supply widget A supply widget is generally clearer than a MICBIAS widget and a mic bias is just a type of supply so use a supply widget for the MICBIAS. This also avoids confusion with the routing when connected to multiple inputs. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8962.c | 2 +- sound/soc/samsung/speyside_wm8962.c | 10 +++++----- 2 files changed, 6 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index d26ec6d..bc6bdde 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2667,7 +2667,7 @@ SND_SOC_DAPM_INPUT("IN4R"), SND_SOC_DAPM_INPUT("Beep"), SND_SOC_DAPM_INPUT("DMICDAT"), -SND_SOC_DAPM_MICBIAS("MICBIAS", WM8962_PWR_MGMT_1, 1, 0), +SND_SOC_DAPM_SUPPLY("MICBIAS", WM8962_PWR_MGMT_1, 1, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("Class G", WM8962_CHARGE_PUMP_B, 0, 1, NULL, 0), SND_SOC_DAPM_SUPPLY("SYSCLK", WM8962_CLOCKING2, 5, 0, sysclk_event, diff --git a/sound/soc/samsung/speyside_wm8962.c b/sound/soc/samsung/speyside_wm8962.c index 98b28ea..35539eb 100644 --- a/sound/soc/samsung/speyside_wm8962.c +++ b/sound/soc/samsung/speyside_wm8962.c @@ -152,12 +152,12 @@ static struct snd_soc_dapm_route audio_paths[] = { { "Main Speaker", NULL, "SPKOUTL" }, { "Main Speaker", NULL, "SPKOUTR" }, - { "MICBIAS", NULL, "Headset Mic" }, - { "IN4L", NULL, "MICBIAS" }, - { "IN4R", NULL, "MICBIAS" }, + { "Headset Mic", NULL, "MICBIAS" }, + { "IN4L", NULL, "Headset Mic" }, + { "IN4R", NULL, "Headset Mic" }, - { "MICBIAS", NULL, "DMIC" }, - { "DMICDAT", NULL, "MICBIAS" }, + { "DMIC", NULL, "MICBIAS" }, + { "DMICDAT", NULL, "DMIC" }, }; static struct snd_soc_jack speyside_wm8962_headset; -- cgit v1.1 From 7564463c64e824a9cf545bcfbfbe6f159a5b9d8c Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 23 Sep 2011 16:23:11 +0100 Subject: ASoC: Add support for on-board analogue microphones on Speyside WM8962 Signed-off-by: Mark Brown --- sound/soc/samsung/speyside_wm8962.c | 5 +++++ 1 file changed, 5 insertions(+) (limited to 'sound') diff --git a/sound/soc/samsung/speyside_wm8962.c b/sound/soc/samsung/speyside_wm8962.c index 35539eb..10c06dc 100644 --- a/sound/soc/samsung/speyside_wm8962.c +++ b/sound/soc/samsung/speyside_wm8962.c @@ -141,6 +141,7 @@ static struct snd_soc_dapm_widget widgets[] = { SND_SOC_DAPM_MIC("Headset Mic", NULL), SND_SOC_DAPM_MIC("DMIC", NULL), + SND_SOC_DAPM_MIC("AMIC", NULL), SND_SOC_DAPM_SPK("Main Speaker", NULL), }; @@ -156,6 +157,10 @@ static struct snd_soc_dapm_route audio_paths[] = { { "IN4L", NULL, "Headset Mic" }, { "IN4R", NULL, "Headset Mic" }, + { "AMIC", NULL, "MICBIAS" }, + { "IN1L", NULL, "AMIC" }, + { "IN1R", NULL, "AMIC" }, + { "DMIC", NULL, "MICBIAS" }, { "DMICDAT", NULL, "DMIC" }, }; -- cgit v1.1 From 8c756158343029c046ade6e7076f25f99774e72f Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 23 Sep 2011 16:46:24 +0100 Subject: ASoC: Add DMIC control to Speyside WM8962 board Signed-off-by: Mark Brown --- sound/soc/samsung/speyside_wm8962.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/samsung/speyside_wm8962.c b/sound/soc/samsung/speyside_wm8962.c index 10c06dc..07b827e 100644 --- a/sound/soc/samsung/speyside_wm8962.c +++ b/sound/soc/samsung/speyside_wm8962.c @@ -134,6 +134,7 @@ static struct snd_soc_dai_link speyside_wm8962_dai[] = { static const struct snd_kcontrol_new controls[] = { SOC_DAPM_PIN_SWITCH("Main Speaker"), + SOC_DAPM_PIN_SWITCH("DMIC"), }; static struct snd_soc_dapm_widget widgets[] = { -- cgit v1.1 From 85a843c50ffb3597928968250a3f552a45b1b9de Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 21 Sep 2011 21:29:47 +0100 Subject: ASoC: Don't force bias on ground referenced devices Currently we force all devices in the system to be at the same bias level. This is due to concerns about power or pop/click impacts from either ramping VMID or mismatching VMID on the analogue I/O lines between connected devices but does mean we power devices up more often than we really need to. If a device flags idle_bias_off this will usually mean that it's either all digital or ground referenced (in which case the idle and powered bias levels are identical) so this concern does not apply and we can save some power by leaving it off when not needed itself. Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 6cac045..2bde6b0 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1319,13 +1319,16 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event) } } - /* Force all contexts in the card to the same bias state */ + /* Force all contexts in the card to the same bias state if + * they're not ground referenced. + */ bias = SND_SOC_BIAS_OFF; list_for_each_entry(d, &card->dapm_list, list) if (d->target_bias_level > bias) bias = d->target_bias_level; list_for_each_entry(d, &card->dapm_list, list) - d->target_bias_level = bias; + if (!d->idle_bias_off) + d->target_bias_level = bias; trace_snd_soc_dapm_walk_done(card); -- cgit v1.1 From 213eb0fb1e8e4ddfb8ffdb239c45ba2a1eef3dc2 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 21 Sep 2011 20:54:47 +0100 Subject: ASoC: Add platform data for WM1250 EV1 GPIOs The WM1250 EV1 has some GPIOs which can be used to control the behaviour at runtime. Request them all if supplied and add a set_bias_level() function to start and stop the clocks. Signed-off-by: Mark Brown --- sound/soc/codecs/wm1250-ev1.c | 121 ++++++++++++++++++++++++++++++++++++++++-- 1 file changed, 118 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm1250-ev1.c b/sound/soc/codecs/wm1250-ev1.c index 4523c4c..0554b88 100644 --- a/sound/soc/codecs/wm1250-ev1.c +++ b/sound/soc/codecs/wm1250-ev1.c @@ -12,10 +12,59 @@ #include #include +#include #include +#include #include #include +#include + +static const char *wm1250_gpio_names[WM1250_EV1_NUM_GPIOS] = { + "WM1250 CLK_ENA", + "WM1250 CLK_SEL0", + "WM1250 CLK_SEL1", + "WM1250 OSR", + "WM1250 MASTER", +}; + +struct wm1250_priv { + struct gpio gpios[WM1250_EV1_NUM_GPIOS]; +}; + +static int wm1250_ev1_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct wm1250_priv *wm1250 = dev_get_drvdata(codec->dev); + int ena; + + if (wm1250) + ena = wm1250->gpios[WM1250_EV1_GPIO_CLK_ENA].gpio; + else + ena = -1; + + switch (level) { + case SND_SOC_BIAS_ON: + break; + + case SND_SOC_BIAS_PREPARE: + break; + + case SND_SOC_BIAS_STANDBY: + if (ena >= 0) + gpio_set_value_cansleep(ena, 1); + break; + + case SND_SOC_BIAS_OFF: + if (ena >= 0) + gpio_set_value_cansleep(ena, 0); + break; + } + + codec->dapm.bias_level = level; + + return 0; +} static const struct snd_soc_dapm_widget wm1250_ev1_dapm_widgets[] = { SND_SOC_DAPM_ADC("ADC", "wm1250-ev1 Capture", SND_SOC_NOPM, 0, 0), @@ -53,12 +102,66 @@ static struct snd_soc_codec_driver soc_codec_dev_wm1250_ev1 = { .num_dapm_widgets = ARRAY_SIZE(wm1250_ev1_dapm_widgets), .dapm_routes = wm1250_ev1_dapm_routes, .num_dapm_routes = ARRAY_SIZE(wm1250_ev1_dapm_routes), + + .set_bias_level = wm1250_ev1_set_bias_level, }; +static int __devinit wm1250_ev1_pdata(struct i2c_client *i2c) +{ + struct wm1250_ev1_pdata *pdata = dev_get_platdata(&i2c->dev); + struct wm1250_priv *wm1250; + int i, ret; + + if (!pdata) + return 0; + + wm1250 = kzalloc(sizeof(*wm1250), GFP_KERNEL); + if (!wm1250) { + dev_err(&i2c->dev, "Unable to allocate private data\n"); + ret = -ENOMEM; + goto err; + } + + for (i = 0; i < ARRAY_SIZE(wm1250->gpios); i++) { + wm1250->gpios[i].gpio = pdata->gpios[i]; + wm1250->gpios[i].label = wm1250_gpio_names[i]; + wm1250->gpios[i].flags = GPIOF_OUT_INIT_LOW; + } + wm1250->gpios[WM1250_EV1_GPIO_CLK_SEL0].flags = GPIOF_OUT_INIT_HIGH; + wm1250->gpios[WM1250_EV1_GPIO_CLK_SEL1].flags = GPIOF_OUT_INIT_HIGH; + + ret = gpio_request_array(wm1250->gpios, ARRAY_SIZE(wm1250->gpios)); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to get GPIOs: %d\n", ret); + goto err_alloc; + } + + dev_set_drvdata(&i2c->dev, wm1250); + + return ret; + +err_alloc: + kfree(wm1250); +err: + return ret; +} + +static void wm1250_ev1_free(struct i2c_client *i2c) +{ + struct wm1250_priv *wm1250 = dev_get_drvdata(&i2c->dev); + + if (wm1250) { + gpio_free_array(wm1250->gpios, ARRAY_SIZE(wm1250->gpios)); + kfree(wm1250); + } +} + static int __devinit wm1250_ev1_probe(struct i2c_client *i2c, const struct i2c_device_id *i2c_id) { - int id, board, rev; + int id, board, rev, ret; + + dev_set_drvdata(&i2c->dev, NULL); board = i2c_smbus_read_byte_data(i2c, 0); if (board < 0) { @@ -76,13 +179,25 @@ static int __devinit wm1250_ev1_probe(struct i2c_client *i2c, dev_info(&i2c->dev, "revision %d\n", rev + 1); - return snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm1250_ev1, - &wm1250_ev1_dai, 1); + ret = wm1250_ev1_pdata(i2c); + if (ret != 0) + return ret; + + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm1250_ev1, + &wm1250_ev1_dai, 1); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to register CODEC: %d\n", ret); + wm1250_ev1_free(i2c); + return ret; + } + + return 0; } static int __devexit wm1250_ev1_remove(struct i2c_client *i2c) { snd_soc_unregister_codec(&i2c->dev); + wm1250_ev1_free(i2c); return 0; } -- cgit v1.1 From a850260e4722706805fda3a0f6e5bc1539e83bac Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 21 Sep 2011 21:33:40 +0100 Subject: ASoC: Set idle_bias_off for WM1250 EV1 The WM1250 EV1 is functionally digital in a system (the analogue I/O is either ground referenced or always powered) so flag it as idle_bias_off. Signed-off-by: Mark Brown --- sound/soc/codecs/wm1250-ev1.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm1250-ev1.c b/sound/soc/codecs/wm1250-ev1.c index 0554b88..cd0ec0f 100644 --- a/sound/soc/codecs/wm1250-ev1.c +++ b/sound/soc/codecs/wm1250-ev1.c @@ -104,6 +104,7 @@ static struct snd_soc_codec_driver soc_codec_dev_wm1250_ev1 = { .num_dapm_routes = ARRAY_SIZE(wm1250_ev1_dapm_routes), .set_bias_level = wm1250_ev1_set_bias_level, + .idle_bias_off = true, }; static int __devinit wm1250_ev1_pdata(struct i2c_client *i2c) -- cgit v1.1 From 141947e6ceb8f82fe8382b26f093bb379af9af15 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Mon, 26 Sep 2011 10:56:42 +0300 Subject: =?UTF-8?q?ASoC:=20omap-mcbsp:=20Fix=20compile=20time=20warning=20?= =?UTF-8?q?about=20ambiguous=20=E2=80=98else=E2=80=99?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Fixes the following compile time warning: omap-mcbsp.c:519: warning: suggest explicit braces to avoid ambiguous ‘else’ Signed-off-by: Peter Ujfalusi Acked-by: Jarkko Nikula Signed-off-by: Mark Brown --- sound/soc/omap/omap-mcbsp.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 478d607..1391ea0 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -516,11 +516,12 @@ static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai, struct omap_mcbsp_reg_cfg *regs = &mcbsp_data->regs; int err = 0; - if (mcbsp_data->active) + if (mcbsp_data->active) { if (freq == mcbsp_data->in_freq) return 0; else return -EBUSY; + } /* The McBSP signal muxing functions are only available on McBSP1 */ if (clk_id == OMAP_MCBSP_CLKR_SRC_CLKR || -- cgit v1.1 From 575e498ae1ac1db22761552079d6b28852eb2d3e Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Sat, 24 Sep 2011 18:16:15 +0800 Subject: ASoC: Drop exporting sn95031_get_mic_bias sn95031_get_mic_bias() is not used outside this driver and it is a static function now. Thus drop exporting sn95031_get_mic_bias. Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/sn95031.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c index d1781d1..8f8ce5f 100644 --- a/sound/soc/codecs/sn95031.c +++ b/sound/soc/codecs/sn95031.c @@ -161,7 +161,6 @@ static unsigned int sn95031_get_mic_bias(struct snd_soc_codec *codec) pr_debug("mic bias = %dmV\n", mic_bias); return mic_bias; } -EXPORT_SYMBOL_GPL(sn95031_get_mic_bias); /*end - adc helper functions */ static inline unsigned int sn95031_read(struct snd_soc_codec *codec, -- cgit v1.1 From c0fd9c9c420f0e980b8539c781494c11a82290ce Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Sat, 24 Sep 2011 18:43:43 +0800 Subject: ASoC: Drop exporting ad1980_dai ad1980_dai is not used outside this driver, thus drop exporting it. Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/ad1980.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c index 4c0fc30..e3931cc 100644 --- a/sound/soc/codecs/ad1980.c +++ b/sound/soc/codecs/ad1980.c @@ -148,7 +148,6 @@ static struct snd_soc_dai_driver ad1980_dai = { .rates = SNDRV_PCM_RATE_48000, .formats = SND_SOC_STD_AC97_FMTS, }, }; -EXPORT_SYMBOL_GPL(ad1980_dai); static int ad1980_reset(struct snd_soc_codec *codec, int try_warm) { -- cgit v1.1 From f97217f18e99235c374f5ce2cde07072e49b582f Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Mon, 26 Sep 2011 16:05:56 +0300 Subject: ASoC: twl6040: Read the TRIM values from the chip Update the reg_cache with values from chip regarding to TRIM. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl6040.c | 7 +++++++ 1 file changed, 7 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 91b9818..7226ae7 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -278,9 +278,16 @@ static void twl6040_init_chip(struct snd_soc_codec *codec) struct twl6040 *twl6040 = codec->control_data; u8 val; + /* Update reg_cache: ASICREV, and TRIM values */ val = twl6040_get_revid(twl6040); twl6040_write_reg_cache(codec, TWL6040_REG_ASICREV, val); + twl6040_read_reg_volatile(codec, TWL6040_REG_TRIM1); + twl6040_read_reg_volatile(codec, TWL6040_REG_TRIM2); + twl6040_read_reg_volatile(codec, TWL6040_REG_TRIM3); + twl6040_read_reg_volatile(codec, TWL6040_REG_HSOTRIM); + twl6040_read_reg_volatile(codec, TWL6040_REG_HFOTRIM); + /* Change chip defaults */ /* No imput selected for microphone amplifiers */ twl6040_write_reg_cache(codec, TWL6040_REG_MICLCTL, 0x18); -- cgit v1.1 From db4aabcc1f2ac32de290510bcc895a960886779d Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Mon, 26 Sep 2011 16:05:57 +0300 Subject: ASoC: twl6040: Function to fetch the TRIM values Provide API to fetch the TRIM values (for machine drivers) Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl6040.c | 9 +++++++++ sound/soc/codecs/twl6040.h | 13 +++++++++++++ 2 files changed, 22 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 7226ae7..0a8728e 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -1072,6 +1072,15 @@ int twl6040_get_clk_id(struct snd_soc_codec *codec) } EXPORT_SYMBOL_GPL(twl6040_get_clk_id); +int twl6040_get_trim_value(struct snd_soc_codec *codec, enum twl6040_trim trim) +{ + if (unlikely(trim >= TWL6040_TRIM_INVAL)) + return -EINVAL; + + return twl6040_read_reg_cache(codec, TWL6040_REG_TRIM1 + trim); +} +EXPORT_SYMBOL_GPL(twl6040_get_trim_value); + static const struct snd_kcontrol_new twl6040_snd_controls[] = { /* Capture gains */ SOC_DOUBLE_TLV("Capture Preamplifier Volume", diff --git a/sound/soc/codecs/twl6040.h b/sound/soc/codecs/twl6040.h index d8de678..a83277b 100644 --- a/sound/soc/codecs/twl6040.h +++ b/sound/soc/codecs/twl6040.h @@ -22,8 +22,21 @@ #ifndef __TWL6040_H__ #define __TWL6040_H__ +enum twl6040_trim { + TWL6040_TRIM_TRIM1 = 0, + TWL6040_TRIM_TRIM2, + TWL6040_TRIM_TRIM3, + TWL6040_TRIM_HSOTRIM, + TWL6040_TRIM_HFOTRIM, + TWL6040_TRIM_INVAL, +}; + +#define TWL6040_HSF_TRIM_LEFT(x) (x & 0x0f) +#define TWL6040_HSF_TRIM_RIGHT(x) ((x >> 4) & 0x0f) + void twl6040_hs_jack_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack, int report); int twl6040_get_clk_id(struct snd_soc_codec *codec); +int twl6040_get_trim_value(struct snd_soc_codec *codec, enum twl6040_trim trim); #endif /* End of __TWL6040_H__ */ -- cgit v1.1 From 89b0d550a6800793b917ce2290ddcd55374d7df6 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Mon, 26 Sep 2011 16:05:58 +0300 Subject: ASoC: omap-mcpdm: API to configure offset cancellation The offset cancellation values can be different from board to board, even on the same HW platform. Provide a way for the machine drivers to configure the McPDM offset cancellation. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/omap/omap-mcpdm.c | 25 +++++++++++++++++++++++++ sound/soc/omap/omap-mcpdm.h | 12 ++++++++++++ 2 files changed, 37 insertions(+) (limited to 'sound') diff --git a/sound/soc/omap/omap-mcpdm.c b/sound/soc/omap/omap-mcpdm.c index 2c9fa51..41d1706 100644 --- a/sound/soc/omap/omap-mcpdm.c +++ b/sound/soc/omap/omap-mcpdm.c @@ -59,6 +59,9 @@ struct omap_mcpdm { /* McPDM FIFO thresholds */ u32 dn_threshold; u32 up_threshold; + + /* McPDM dn offsets for rx1, and 2 channels */ + u32 dn_rx_offset; }; /* @@ -183,6 +186,15 @@ static void omap_mcpdm_open_streams(struct omap_mcpdm *mcpdm) MCPDM_DN_IRQ_EMPTY | MCPDM_DN_IRQ_FULL | MCPDM_UP_IRQ_EMPTY | MCPDM_UP_IRQ_FULL); + /* Enable DN RX1/2 offset cancellation feature, if configured */ + if (mcpdm->dn_rx_offset) { + u32 dn_offset = mcpdm->dn_rx_offset; + + omap_mcpdm_write(mcpdm, MCPDM_REG_DN_OFFSET, dn_offset); + dn_offset |= (MCPDM_DN_OFST_RX1_EN | MCPDM_DN_OFST_RX2_EN); + omap_mcpdm_write(mcpdm, MCPDM_REG_DN_OFFSET, dn_offset); + } + omap_mcpdm_write(mcpdm, MCPDM_REG_FIFO_CTRL_DN, mcpdm->dn_threshold); omap_mcpdm_write(mcpdm, MCPDM_REG_FIFO_CTRL_UP, mcpdm->up_threshold); @@ -209,6 +221,10 @@ static void omap_mcpdm_close_streams(struct omap_mcpdm *mcpdm) /* Disable DMA request generation for uplink */ omap_mcpdm_write(mcpdm, MCPDM_REG_DMAENABLE_CLR, MCPDM_DMA_UP_ENABLE); + + /* Disable RX1/2 offset cancellation */ + if (mcpdm->dn_rx_offset) + omap_mcpdm_write(mcpdm, MCPDM_REG_DN_OFFSET, 0); } static irqreturn_t omap_mcpdm_irq_handler(int irq, void *dev_id) @@ -418,6 +434,15 @@ static struct snd_soc_dai_driver omap_mcpdm_dai = { .ops = &omap_mcpdm_dai_ops, }; +void omap_mcpdm_configure_dn_offsets(struct snd_soc_pcm_runtime *rtd, + u8 rx1, u8 rx2) +{ + struct omap_mcpdm *mcpdm = snd_soc_dai_get_drvdata(rtd->cpu_dai); + + mcpdm->dn_rx_offset = MCPDM_DNOFST_RX1(rx1) | MCPDM_DNOFST_RX2(rx2); +} +EXPORT_SYMBOL_GPL(omap_mcpdm_configure_dn_offsets); + static __devinit int asoc_mcpdm_probe(struct platform_device *pdev) { struct omap_mcpdm *mcpdm; diff --git a/sound/soc/omap/omap-mcpdm.h b/sound/soc/omap/omap-mcpdm.h index d23122a..de8cf26 100644 --- a/sound/soc/omap/omap-mcpdm.h +++ b/sound/soc/omap/omap-mcpdm.h @@ -92,4 +92,16 @@ #define MCPDM_UP_THRES_MAX 0xF #define MCPDM_DN_THRES_MAX 0xF +/* + * MCPDM_DN_OFFSET bit fields + */ + +#define MCPDM_DN_OFST_RX1_EN (1 << 0) +#define MCPDM_DNOFST_RX1(x) ((x & 0x1f) << 1) +#define MCPDM_DN_OFST_RX2_EN (1 << 8) +#define MCPDM_DNOFST_RX2(x) ((x & 0x1f) << 9) + +void omap_mcpdm_configure_dn_offsets(struct snd_soc_pcm_runtime *rtd, + u8 rx1, u8 rx2); + #endif /* End of __OMAP_MCPDM_H__ */ -- cgit v1.1 From 7bf3d92cdd61e2d47fc5ae403ee5bc598c757f29 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Mon, 26 Sep 2011 16:05:59 +0300 Subject: ASoC: sdp4430: Configure McPDM offset cancellation Based on the values from twl6040 codec (HSOTRIM L/R) we can configure the McPDM offset cancellation. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/omap/sdp4430.c | 10 +++++++++- 1 file changed, 9 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/omap/sdp4430.c b/sound/soc/omap/sdp4430.c index 4200eb4..249d84b 100644 --- a/sound/soc/omap/sdp4430.c +++ b/sound/soc/omap/sdp4430.c @@ -121,7 +121,7 @@ static int sdp4430_twl6040_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; struct snd_soc_dapm_context *dapm = &codec->dapm; - int ret; + int ret, hs_trim; /* Add SDP4430 specific widgets */ ret = snd_soc_dapm_new_controls(dapm, sdp4430_twl6040_dapm_widgets, @@ -144,6 +144,14 @@ static int sdp4430_twl6040_init(struct snd_soc_pcm_runtime *rtd) if (ret) return ret; + /* + * Configure McPDM offset cancellation based on the HSOTRIM value from + * twl6040. + */ + hs_trim = twl6040_get_trim_value(codec, TWL6040_TRIM_HSOTRIM); + omap_mcpdm_configure_dn_offsets(rtd, TWL6040_HSF_TRIM_LEFT(hs_trim), + TWL6040_HSF_TRIM_RIGHT(hs_trim)); + /* Headset jack detection */ ret = snd_soc_jack_new(codec, "Headset Jack", SND_JACK_HEADSET, &hs_jack); -- cgit v1.1 From eb6b71e7d964ee4934c65a954dd5738a1bf3d0e8 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Mon, 26 Sep 2011 16:26:24 +0300 Subject: ASoC: twl6040: Rename pga_event to out_drv_event This event handler is used with the OUT_DRV widgets. The name pga_event was misleading. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl6040.c | 11 +++++------ 1 file changed, 5 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 0a8728e..ee4d1b4 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -72,7 +72,6 @@ struct twl6040_output { u16 right_step; unsigned int step_delay; u16 ramp; - u16 mute; struct completion ramp_done; }; @@ -573,7 +572,7 @@ static void twl6040_pga_hf_work(struct work_struct *work) handsfree->ramp = TWL6040_RAMP_NONE; } -static int pga_event(struct snd_soc_dapm_widget *w, +static int out_drv_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { struct snd_soc_codec *codec = w->codec; @@ -1197,19 +1196,19 @@ static const struct snd_soc_dapm_widget twl6040_dapm_widgets[] = { /* Analog playback drivers */ SND_SOC_DAPM_OUT_DRV_E("HF Left Driver", TWL6040_REG_HFLCTL, 4, 0, NULL, 0, - pga_event, + out_drv_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), SND_SOC_DAPM_OUT_DRV_E("HF Right Driver", TWL6040_REG_HFRCTL, 4, 0, NULL, 0, - pga_event, + out_drv_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), SND_SOC_DAPM_OUT_DRV_E("HS Left Driver", TWL6040_REG_HSLCTL, 2, 0, NULL, 0, - pga_event, + out_drv_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), SND_SOC_DAPM_OUT_DRV_E("HS Right Driver", TWL6040_REG_HSRCTL, 2, 0, NULL, 0, - pga_event, + out_drv_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), SND_SOC_DAPM_OUT_DRV_E("Earphone Driver", TWL6040_REG_EARCTL, 0, 0, NULL, 0, -- cgit v1.1 From a8cc7189cd1ff7856ef688af3a492668e30dda02 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Mon, 26 Sep 2011 16:26:25 +0300 Subject: ASoC: twl6040: Combine the custom volsw get, and put functions We can manage with one set of get, and put function for the gain controls we need to handle with custom code due to the shadowing of the register. For both get, and put function we can call decide based on the mc->rreg value, if we need to call the volsw, or the vlosw_2r variant (in 2r case rreg is not 0). Handling of the shadow values are the same for both type of controls. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl6040.c | 81 ++++++++++------------------------------------ 1 file changed, 17 insertions(+), 64 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index ee4d1b4..1068447 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -763,15 +763,17 @@ static int twl6040_put_volsw(struct snd_kcontrol *kcontrol, struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; int ret; - unsigned int reg = mc->reg; /* For HS and HF we shadow the values and only actually write * them out when active in order to ensure the amplifier comes on * as quietly as possible. */ - switch (reg) { + switch (mc->reg) { case TWL6040_REG_HSGAIN: out = &twl6040_priv->headset; break; + case TWL6040_REG_HFLGAIN: + out = &twl6040_priv->handsfree; + break; default: break; } @@ -783,7 +785,12 @@ static int twl6040_put_volsw(struct snd_kcontrol *kcontrol, return 1; } - ret = snd_soc_put_volsw(kcontrol, ucontrol); + /* call the appropriate handler depending on the rreg */ + if (mc->rreg) + ret = snd_soc_put_volsw_2r(kcontrol, ucontrol); + else + ret = snd_soc_put_volsw(kcontrol, ucontrol); + if (ret < 0) return ret; @@ -798,39 +805,12 @@ static int twl6040_get_volsw(struct snd_kcontrol *kcontrol, struct twl6040_output *out = &twl6040_priv->headset; struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; - unsigned int reg = mc->reg; - switch (reg) { + switch (mc->reg) { case TWL6040_REG_HSGAIN: out = &twl6040_priv->headset; - ucontrol->value.integer.value[0] = out->left_vol; - ucontrol->value.integer.value[1] = out->right_vol; - return 0; - - default: break; - } - - return snd_soc_get_volsw(kcontrol, ucontrol); -} - -static int twl6040_put_volsw_2r_vu(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - struct twl6040_data *twl6040_priv = snd_soc_codec_get_drvdata(codec); - struct twl6040_output *out = NULL; - struct soc_mixer_control *mc = - (struct soc_mixer_control *)kcontrol->private_value; - int ret; - unsigned int reg = mc->reg; - - /* For HS and HF we shadow the values and only actually write - * them out when active in order to ensure the amplifier comes on - * as quietly as possible. */ - switch (reg) { case TWL6040_REG_HFLGAIN: - case TWL6040_REG_HFRGAIN: out = &twl6040_priv->handsfree; break; default: @@ -838,43 +818,16 @@ static int twl6040_put_volsw_2r_vu(struct snd_kcontrol *kcontrol, } if (out) { - out->left_vol = ucontrol->value.integer.value[0]; - out->right_vol = ucontrol->value.integer.value[1]; - if (!out->active) - return 1; - } - - ret = snd_soc_put_volsw_2r(kcontrol, ucontrol); - if (ret < 0) - return ret; - - return 1; -} - -static int twl6040_get_volsw_2r(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - struct twl6040_data *twl6040_priv = snd_soc_codec_get_drvdata(codec); - struct twl6040_output *out = &twl6040_priv->handsfree; - struct soc_mixer_control *mc = - (struct soc_mixer_control *)kcontrol->private_value; - unsigned int reg = mc->reg; - - /* If these are cached registers use the cache */ - switch (reg) { - case TWL6040_REG_HFLGAIN: - case TWL6040_REG_HFRGAIN: - out = &twl6040_priv->handsfree; ucontrol->value.integer.value[0] = out->left_vol; ucontrol->value.integer.value[1] = out->right_vol; return 0; - - default: - break; } - return snd_soc_get_volsw_2r(kcontrol, ucontrol); + /* call the appropriate handler depending on the rreg */ + if (mc->rreg) + return snd_soc_get_volsw_2r(kcontrol, ucontrol); + else + return snd_soc_get_volsw(kcontrol, ucontrol); } /* double control with volume update */ @@ -899,7 +852,7 @@ static int twl6040_get_volsw_2r(struct snd_kcontrol *kcontrol, SNDRV_CTL_ELEM_ACCESS_VOLATILE, \ .tlv.p = (tlv_array), \ .info = snd_soc_info_volsw_2r, \ - .get = twl6040_get_volsw_2r, .put = twl6040_put_volsw_2r_vu, \ + .get = twl6040_get_volsw, .put = twl6040_put_volsw, \ .private_value = (unsigned long)&(struct soc_mixer_control) \ {.reg = reg_left, .rreg = reg_right, .shift = xshift, \ .rshift = xshift, .max = xmax, .invert = xinvert}, } -- cgit v1.1 From e71a5e5af69185f1c2e5c1bf4ee90d92dd1c1e8a Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Mon, 26 Sep 2011 16:26:26 +0300 Subject: ASoC: twl6040: Move delayed_work struct inside twl6040_output for HS/HF The delayed works for the output can be moved within the twl6040_output struct (from the twl6040_data) to be better organized. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl6040.c | 15 +++++++-------- 1 file changed, 7 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 1068447..7786520 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -72,6 +72,7 @@ struct twl6040_output { u16 right_step; unsigned int step_delay; u16 ramp; + struct delayed_work work; struct completion ramp_done; }; @@ -104,8 +105,6 @@ struct twl6040_data { struct twl6040_output handsfree; struct workqueue_struct *hf_workqueue; struct workqueue_struct *hs_workqueue; - struct delayed_work hs_delayed_work; - struct delayed_work hf_delayed_work; }; /* @@ -489,7 +488,7 @@ static inline int twl6040_hf_ramp_step(struct snd_soc_codec *codec, static void twl6040_pga_hs_work(struct work_struct *work) { struct twl6040_data *priv = - container_of(work, struct twl6040_data, hs_delayed_work.work); + container_of(work, struct twl6040_data, headset.work.work); struct snd_soc_codec *codec = priv->codec; struct twl6040_output *headset = &priv->headset; unsigned int delay = headset->step_delay; @@ -532,7 +531,7 @@ static void twl6040_pga_hs_work(struct work_struct *work) static void twl6040_pga_hf_work(struct work_struct *work) { struct twl6040_data *priv = - container_of(work, struct twl6040_data, hf_delayed_work.work); + container_of(work, struct twl6040_data, handsfree.work.work); struct snd_soc_codec *codec = priv->codec; struct twl6040_output *handsfree = &priv->handsfree; unsigned int delay = handsfree->step_delay; @@ -585,7 +584,6 @@ static int out_drv_event(struct snd_soc_dapm_widget *w, case 2: case 3: out = &priv->headset; - work = &priv->hs_delayed_work; queue = priv->hs_workqueue; out->left_step = priv->hs_left_step; out->right_step = priv->hs_right_step; @@ -593,7 +591,6 @@ static int out_drv_event(struct snd_soc_dapm_widget *w, break; case 4: out = &priv->handsfree; - work = &priv->hf_delayed_work; queue = priv->hf_workqueue; out->left_step = priv->hf_left_step; out->right_step = priv->hf_right_step; @@ -607,6 +604,8 @@ static int out_drv_event(struct snd_soc_dapm_widget *w, return -1; } + work = &out->work; + switch (event) { case SND_SOC_DAPM_POST_PMU: if (out->active) @@ -1553,8 +1552,8 @@ static int twl6040_probe(struct snd_soc_codec *codec) goto hswq_err; } - INIT_DELAYED_WORK(&priv->hs_delayed_work, twl6040_pga_hs_work); - INIT_DELAYED_WORK(&priv->hf_delayed_work, twl6040_pga_hf_work); + INIT_DELAYED_WORK(&priv->headset.work, twl6040_pga_hs_work); + INIT_DELAYED_WORK(&priv->handsfree.work, twl6040_pga_hf_work); ret = request_threaded_irq(priv->plug_irq, NULL, twl6040_audio_handler, 0, "twl6040_irq_plug", codec); -- cgit v1.1 From 46dd0b93a086b798a040c06479eabcb87cd29344 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Mon, 26 Sep 2011 16:26:27 +0300 Subject: ASoC: twl6040: Move the delayed_work for HS detection under twl6040_jack_data The delayed_work named 'delayed_work' is for the headset detection, so move it to the twl6040_jack_data struct. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl6040.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 7786520..7b543c0 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -78,6 +78,7 @@ struct twl6040_output { struct twl6040_jack_data { struct snd_soc_jack *jack; + struct delayed_work work; int report; }; @@ -99,7 +100,6 @@ struct twl6040_data { struct twl6040_jack_data hs_jack; struct snd_soc_codec *codec; struct workqueue_struct *workqueue; - struct delayed_work delayed_work; struct mutex mutex; struct twl6040_output headset; struct twl6040_output handsfree; @@ -734,7 +734,7 @@ EXPORT_SYMBOL_GPL(twl6040_hs_jack_detect); static void twl6040_accessory_work(struct work_struct *work) { struct twl6040_data *priv = container_of(work, - struct twl6040_data, delayed_work.work); + struct twl6040_data, hs_jack.work.work); struct snd_soc_codec *codec = priv->codec; struct twl6040_jack_data *hs_jack = &priv->hs_jack; @@ -747,7 +747,7 @@ static irqreturn_t twl6040_audio_handler(int irq, void *data) struct snd_soc_codec *codec = data; struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec); - queue_delayed_work(priv->workqueue, &priv->delayed_work, + queue_delayed_work(priv->workqueue, &priv->hs_jack.work, msecs_to_jiffies(200)); return IRQ_HANDLED; @@ -1534,7 +1534,7 @@ static int twl6040_probe(struct snd_soc_codec *codec) goto work_err; } - INIT_DELAYED_WORK(&priv->delayed_work, twl6040_accessory_work); + INIT_DELAYED_WORK(&priv->hs_jack.work, twl6040_accessory_work); mutex_init(&priv->mutex); -- cgit v1.1 From 8ff1e1709846c48d20a062293df013931d99585b Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Mon, 26 Sep 2011 16:26:30 +0300 Subject: ASoC: twl6040: No need to change delay during HS ramp The Headset gain have 2dB steps all the way, so there is no reason to have different delays as we approaching to the end of the scale. The comment was also wrong, since we have 0dB at 0x0 raw at one end of the range, and not in the middle. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl6040.c | 12 ++---------- 1 file changed, 2 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 7b543c0..0144e43 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -491,7 +491,6 @@ static void twl6040_pga_hs_work(struct work_struct *work) container_of(work, struct twl6040_data, headset.work.work); struct snd_soc_codec *codec = priv->codec; struct twl6040_output *headset = &priv->headset; - unsigned int delay = headset->step_delay; int i, headset_complete; /* do we need to ramp at all ? */ @@ -508,15 +507,8 @@ static void twl6040_pga_hs_work(struct work_struct *work) if (headset_complete) break; - /* - * TODO: tune: delay is longer over 0dB - * as increases are larger. - */ - if (i >= 8) - schedule_timeout_interruptible(msecs_to_jiffies(delay + - (delay >> 1))); - else - schedule_timeout_interruptible(msecs_to_jiffies(delay)); + schedule_timeout_interruptible( + msecs_to_jiffies(headset->step_delay)); } if (headset->ramp == TWL6040_RAMP_DOWN) { -- cgit v1.1 From 4d64bdca4485da8d2e604c2b02f3f32c9f468a28 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Mon, 26 Sep 2011 16:26:31 +0300 Subject: ASoC: twl6040: No need to change delay during HF ramp The Handsfree gain have 2dB steps all the way, so there is no reason to have different delays as we approaching to the end of the scale. The comment was also wrong, since we have 0dB at 0x3 raw, at 16 the gain is -26dB. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl6040.c | 12 ++---------- 1 file changed, 2 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 0144e43..c973347 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -526,7 +526,6 @@ static void twl6040_pga_hf_work(struct work_struct *work) container_of(work, struct twl6040_data, handsfree.work.work); struct snd_soc_codec *codec = priv->codec; struct twl6040_output *handsfree = &priv->handsfree; - unsigned int delay = handsfree->step_delay; int i, handsfree_complete; /* do we need to ramp at all ? */ @@ -543,15 +542,8 @@ static void twl6040_pga_hf_work(struct work_struct *work) if (handsfree_complete) break; - /* - * TODO: tune: delay is longer over 0dB - * as increases are larger. - */ - if (i >= 16) - schedule_timeout_interruptible(msecs_to_jiffies(delay + - (delay >> 1))); - else - schedule_timeout_interruptible(msecs_to_jiffies(delay)); + schedule_timeout_interruptible( + msecs_to_jiffies(handsfree->step_delay)); } -- cgit v1.1 From fe0cc75193cf7d06818b44b01b96c552111a43a6 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Tue, 27 Sep 2011 10:38:07 +0800 Subject: ASoC: Remove unused fields in struct mfld_mc_private Both *socdev and *codec of struct mfld_mc_private are not being used in this driver, remove it. Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/mid-x86/mfld_machine.c | 2 -- 1 file changed, 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/mid-x86/mfld_machine.c b/sound/soc/mid-x86/mfld_machine.c index 429aa1b..1d818dc 100644 --- a/sound/soc/mid-x86/mfld_machine.c +++ b/sound/soc/mid-x86/mfld_machine.c @@ -54,9 +54,7 @@ static unsigned int hs_switch; static unsigned int lo_dac; struct mfld_mc_private { - struct platform_device *socdev; void __iomem *int_base; - struct snd_soc_codec *codec; u8 interrupt_status; }; -- cgit v1.1 From d1b73287c2710f3b76f6980daa88cc719d11034f Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Tue, 27 Sep 2011 10:38:50 +0800 Subject: ASoC: Staticise sst_platform_dai It is not used outside this driver so no need to make the symbol global. Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/mid-x86/sst_platform.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/mid-x86/sst_platform.c b/sound/soc/mid-x86/sst_platform.c index 9925d20..7df8c58 100644 --- a/sound/soc/mid-x86/sst_platform.c +++ b/sound/soc/mid-x86/sst_platform.c @@ -63,7 +63,7 @@ static struct snd_pcm_hardware sst_platform_pcm_hw = { }; /* MFLD - MSIC */ -struct snd_soc_dai_driver sst_platform_dai[] = { +static struct snd_soc_dai_driver sst_platform_dai[] = { { .name = "Headset-cpu-dai", .id = 0, -- cgit v1.1 From a9d1974ea13b361bf60a9d493a6a05e5a42b0ba2 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 27 Sep 2011 11:08:47 +0200 Subject: ASoC: ssm2602: Set initial bias level to standby Set the initial bias level to standby during CODEC probe instead of leaving the CODEC powered off. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/ssm2602.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index 32d6c51..c9e0fdb 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -577,7 +577,12 @@ static int ssm260x_probe(struct snd_soc_codec *codec) break; } - return ret; + if (ret) + return ret; + + ssm2602_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + return 0; } /* remove everything here */ -- cgit v1.1 From 02890535269338a6d2034ad3ce8b22beb24b449a Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 27 Sep 2011 11:08:48 +0200 Subject: ASoC: ssm2602: Support setting the oscillator and the clock output state Currently the oscillator is always enabled and the clock output is always disabled. This patch adds support for controlling the oscillator and clock output state through snd_soc_dai_set_sysclk. Which makes it possible to disable or enable them dynamically according to the requirements of the board on which the CODEC is used. This patch also slightly modifies the behavior as to when the oscillator is going to be disabled in low-power states. Previously it would only be disabled in BIAS_OFF, now it is also going to be disabled in BIAS_STANDBY, since no components which depend on it should be active in this state. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/ssm2602.c | 67 +++++++++++++++++++++++++++++++++++----------- sound/soc/codecs/ssm2602.h | 6 ++++- 2 files changed, 56 insertions(+), 17 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index c9e0fdb..e149ec6 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -59,6 +59,7 @@ struct ssm2602_priv { struct snd_pcm_substream *slave_substream; enum ssm2602_type type; + unsigned int clk_out_pwr; }; /* @@ -356,16 +357,46 @@ static int ssm2602_set_dai_sysclk(struct snd_soc_dai *codec_dai, { struct snd_soc_codec *codec = codec_dai->codec; struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec); - switch (freq) { - case 11289600: - case 12000000: - case 12288000: - case 16934400: - case 18432000: - ssm2602->sysclk = freq; - return 0; + + if (dir == SND_SOC_CLOCK_IN) { + if (clk_id != SSM2602_SYSCLK) + return -EINVAL; + + switch (freq) { + case 11289600: + case 12000000: + case 12288000: + case 16934400: + case 18432000: + ssm2602->sysclk = freq; + break; + default: + return -EINVAL; + } + } else { + unsigned int mask; + + switch (clk_id) { + case SSM2602_CLK_CLKOUT: + mask = PWR_CLK_OUT_PDN; + break; + case SSM2602_CLK_XTO: + mask = PWR_OSC_PDN; + break; + default: + return -EINVAL; + } + + if (freq == 0) + ssm2602->clk_out_pwr |= mask; + else + ssm2602->clk_out_pwr &= ~mask; + + snd_soc_update_bits(codec, SSM2602_PWR, + PWR_CLK_OUT_PDN | PWR_OSC_PDN, ssm2602->clk_out_pwr); } - return -EINVAL; + + return 0; } static int ssm2602_set_dai_fmt(struct snd_soc_dai *codec_dai, @@ -430,23 +461,27 @@ static int ssm2602_set_dai_fmt(struct snd_soc_dai *codec_dai, static int ssm2602_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { - u16 reg = snd_soc_read(codec, SSM2602_PWR); - reg &= ~(PWR_POWER_OFF | PWR_OSC_PDN); + struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec); switch (level) { case SND_SOC_BIAS_ON: - /* vref/mid, osc on, dac unmute */ - snd_soc_write(codec, SSM2602_PWR, reg); + /* vref/mid on, osc and clkout on if enabled */ + snd_soc_update_bits(codec, SSM2602_PWR, + PWR_POWER_OFF | PWR_CLK_OUT_PDN | PWR_OSC_PDN, + ssm2602->clk_out_pwr); break; case SND_SOC_BIAS_PREPARE: break; case SND_SOC_BIAS_STANDBY: /* everything off except vref/vmid, */ - snd_soc_write(codec, SSM2602_PWR, reg | PWR_CLK_OUT_PDN); + snd_soc_update_bits(codec, SSM2602_PWR, + PWR_POWER_OFF | PWR_CLK_OUT_PDN | PWR_OSC_PDN, + PWR_CLK_OUT_PDN | PWR_OSC_PDN); break; case SND_SOC_BIAS_OFF: - /* everything off, dac mute, inactive */ - snd_soc_write(codec, SSM2602_PWR, 0xffff); + /* everything off */ + snd_soc_update_bits(codec, SSM2602_PWR, + PWR_POWER_OFF, PWR_POWER_OFF); break; } diff --git a/sound/soc/codecs/ssm2602.h b/sound/soc/codecs/ssm2602.h index b98c691..fbd07d7 100644 --- a/sound/soc/codecs/ssm2602.h +++ b/sound/soc/codecs/ssm2602.h @@ -116,6 +116,10 @@ #define SSM2602_CACHEREGNUM 10 -#define SSM2602_SYSCLK 0 +enum ssm2602_clk { + SSM2602_SYSCLK, + SSM2602_CLK_CLKOUT, + SSM2602_CLK_XTO +}; #endif -- cgit v1.1 From bcec267a176d72b779496ea2c7d63f8a89e4cdfe Mon Sep 17 00:00:00 2001 From: Karl Tsou Date: Wed, 28 Sep 2011 01:47:18 +0800 Subject: ASoC: Add DRC control for WM8996 Signed-off-by: Karl Tsou Signed-off-by: Mark Brown --- sound/soc/codecs/wm8996.c | 8 ++++++++ 1 file changed, 8 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index 833df74..b98a8f8 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -641,6 +641,14 @@ SOC_DOUBLE_R("Speaker ZC Switch", WM8996_LEFT_PDM_SPEAKER, SOC_SINGLE("DSP1 EQ Switch", WM8996_DSP1_RX_EQ_GAINS_1, 0, 1, 0), SOC_SINGLE("DSP2 EQ Switch", WM8996_DSP2_RX_EQ_GAINS_1, 0, 1, 0), + +SOC_SINGLE("DSP1 DRC TXL Switch", WM8996_DSP1_DRC_1, 0, 1, 0), +SOC_SINGLE("DSP1 DRC TXR Switch", WM8996_DSP1_DRC_1, 1, 1, 0), +SOC_SINGLE("DSP1 DRC RX Switch", WM8996_DSP1_DRC_1, 2, 1, 0), + +SOC_SINGLE("DSP2 DRC TXL Switch", WM8996_DSP2_DRC_1, 0, 1, 0), +SOC_SINGLE("DSP2 DRC TXR Switch", WM8996_DSP2_DRC_1, 1, 1, 0), +SOC_SINGLE("DSP2 DRC RX Switch", WM8996_DSP2_DRC_1, 2, 1, 0), }; static const struct snd_kcontrol_new wm8996_eq_controls[] = { -- cgit v1.1 From 644f1ff4ff5873124a2a19efb1ebd1878f97f5eb Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 27 Sep 2011 19:19:41 +0100 Subject: ASoC: Add device ID for WM9093 to WM9090 driver The WM9093 is an enhanced version of the WM9093. Add the device ID to the driver, further patches will add support for the additional features in the WM9093. Signed-off-by: Mark Brown --- sound/soc/codecs/wm9090.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm9090.c b/sound/soc/codecs/wm9090.c index f2f3077..bd27fc7 100644 --- a/sound/soc/codecs/wm9090.c +++ b/sound/soc/codecs/wm9090.c @@ -682,6 +682,7 @@ static int __devexit wm9090_i2c_remove(struct i2c_client *i2c) static const struct i2c_device_id wm9090_id[] = { { "wm9090", 0 }, + { "wm9093", 0 }, { } }; MODULE_DEVICE_TABLE(i2c, wm9090_id); -- cgit v1.1 From c9241ec6af54db453d03f3f4141462380372a2b8 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Tue, 27 Sep 2011 20:41:36 +0800 Subject: ASoC: Remove unused "control_data" field of struct wm8940_priv The control_data field is used to initialize the codec's control_data field, but since this is also done by the snd-soc-cache core, the redundant assignment can be removed and the field can be dropped. Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/wm8940.c | 3 --- 1 file changed, 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c index 056daa0..7e0f54c 100644 --- a/sound/soc/codecs/wm8940.c +++ b/sound/soc/codecs/wm8940.c @@ -43,7 +43,6 @@ struct wm8940_priv { unsigned int sysclk; enum snd_soc_control_type control_type; - void *control_data; }; static u16 wm8940_reg_defaults[] = { @@ -693,7 +692,6 @@ static int wm8940_probe(struct snd_soc_codec *codec) int ret; u16 reg; - codec->control_data = wm8940->control_data; ret = snd_soc_codec_set_cache_io(codec, 8, 16, wm8940->control_type); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); @@ -758,7 +756,6 @@ static __devinit int wm8940_i2c_probe(struct i2c_client *i2c, return -ENOMEM; i2c_set_clientdata(i2c, wm8940); - wm8940->control_data = i2c; wm8940->control_type = SND_SOC_I2C; ret = snd_soc_register_codec(&i2c->dev, -- cgit v1.1 From ec61bde573a9eec829a04822454ab4818f2f79b3 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Tue, 27 Sep 2011 20:42:34 +0800 Subject: ASoC: Remove unused "control_data" field of struct wm8960_priv The control_data field is used to initialize the codec's control_data field, but since this is also done by the snd-soc-cache core, the redundant assignment can be removed and the field can be dropped. Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/wm8960.c | 3 --- 1 file changed, 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 4393394..831c20f 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -72,7 +72,6 @@ static const u16 wm8960_reg[WM8960_CACHEREGNUM] = { struct wm8960_priv { enum snd_soc_control_type control_type; - void *control_data; int (*set_bias_level)(struct snd_soc_codec *, enum snd_soc_bias_level level); struct snd_soc_dapm_widget *lout1; @@ -925,7 +924,6 @@ static int wm8960_probe(struct snd_soc_codec *codec) u16 reg; wm8960->set_bias_level = wm8960_set_bias_level_out3; - codec->control_data = wm8960->control_data; if (!pdata) { dev_warn(codec->dev, "No platform data supplied\n"); @@ -1015,7 +1013,6 @@ static __devinit int wm8960_i2c_probe(struct i2c_client *i2c, i2c_set_clientdata(i2c, wm8960); wm8960->control_type = SND_SOC_I2C; - wm8960->control_data = i2c; ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8960, &wm8960_dai, 1); -- cgit v1.1 From 8c0c459ced458b19a589b3a31e5c1231bd1b887a Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Tue, 27 Sep 2011 20:43:24 +0800 Subject: ASoC: Remove unused "control_data" field of struct wm8978_priv The control_data field is used to initialize the codec's control_data field, but since this is also done by the snd-soc-cache core, the redundant assignment can be removed and the field can be dropped. Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/wm8978.c | 3 --- 1 file changed, 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c index 85e3e63..41ca4d9 100644 --- a/sound/soc/codecs/wm8978.c +++ b/sound/soc/codecs/wm8978.c @@ -52,7 +52,6 @@ static const u16 wm8978_reg[WM8978_CACHEREGNUM] = { /* codec private data */ struct wm8978_priv { enum snd_soc_control_type control_type; - void *control_data; unsigned int f_pllout; unsigned int f_mclk; unsigned int f_256fs; @@ -955,7 +954,6 @@ static int wm8978_probe(struct snd_soc_codec *codec) * default hardware setting */ wm8978->sysclk = WM8978_PLL; - codec->control_data = wm8978->control_data; ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_I2C); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); @@ -1016,7 +1014,6 @@ static __devinit int wm8978_i2c_probe(struct i2c_client *i2c, return -ENOMEM; i2c_set_clientdata(i2c, wm8978); - wm8978->control_data = i2c; ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8978, &wm8978_dai, 1); -- cgit v1.1 From 6e34216490d63a496af8db6f497dbfc251405397 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Tue, 27 Sep 2011 20:44:13 +0800 Subject: ASoC: Remove unused "control_data" field of struct wm9081_priv The control_data field is used to initialize the codec's control_data field, but since this is also done by the snd-soc-cache core, the redundant assignment can be removed and the field can be dropped. Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/wm9081.c | 3 --- 1 file changed, 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index f32ab1e..b2d3448 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -157,7 +157,6 @@ static struct { struct wm9081_priv { enum snd_soc_control_type control_type; - void *control_data; int sysclk_source; int mclk_rate; int sysclk_rate; @@ -1213,7 +1212,6 @@ static int wm9081_probe(struct snd_soc_codec *codec) int ret; u16 reg; - codec->control_data = wm9081->control_data; ret = snd_soc_codec_set_cache_io(codec, 8, 16, wm9081->control_type); if (ret != 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); @@ -1330,7 +1328,6 @@ static __devinit int wm9081_i2c_probe(struct i2c_client *i2c, i2c_set_clientdata(i2c, wm9081); wm9081->control_type = SND_SOC_I2C; - wm9081->control_data = i2c; if (dev_get_platdata(&i2c->dev)) memcpy(&wm9081->pdata, dev_get_platdata(&i2c->dev), -- cgit v1.1 From b6ba8cc287f7dde9302d8b152f8e60cd570ecbc8 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Tue, 27 Sep 2011 20:45:17 +0800 Subject: ASoC: Remove unused "control_data" field of struct wm9090_priv The control_data field is used to initialize the codec's control_data field, but since this is also done by the snd-soc-cache core, the redundant assignment can be removed and the field can be dropped. Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/wm9090.c | 3 --- 1 file changed, 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm9090.c b/sound/soc/codecs/wm9090.c index bd27fc7..228d782 100644 --- a/sound/soc/codecs/wm9090.c +++ b/sound/soc/codecs/wm9090.c @@ -140,7 +140,6 @@ static const u16 wm9090_reg_defaults[] = { /* This struct is used to save the context */ struct wm9090_priv { struct wm9090_platform_data pdata; - void *control_data; }; static int wm9090_volatile(struct snd_soc_codec *codec, unsigned int reg) @@ -552,7 +551,6 @@ static int wm9090_probe(struct snd_soc_codec *codec) struct wm9090_priv *wm9090 = snd_soc_codec_get_drvdata(codec); int ret; - codec->control_data = wm9090->control_data; ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_I2C); if (ret != 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); @@ -661,7 +659,6 @@ static int wm9090_i2c_probe(struct i2c_client *i2c, sizeof(wm9090->pdata)); i2c_set_clientdata(i2c, wm9090); - wm9090->control_data = i2c; ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm9090, NULL, 0); -- cgit v1.1 From 4addfd88ea6c5f6dba60aa43efa45551f4604b88 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Tue, 27 Sep 2011 20:40:22 +0800 Subject: ASoC: Remove unused "control_data" field of struct wm8904_priv The control_data field is used to initialize the codec's control_data field, but since this is also done by the snd-soc-cache core, the redundant assignment can be removed and the field can be dropped. Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/wm8904.c | 2 -- 1 file changed, 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index b085575..9fc8f4c 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -50,7 +50,6 @@ static const char *wm8904_supply_names[WM8904_NUM_SUPPLIES] = { struct wm8904_priv { enum wm8904_type devtype; - void *control_data; struct regulator_bulk_data supplies[WM8904_NUM_SUPPLIES]; @@ -2540,7 +2539,6 @@ static __devinit int wm8904_i2c_probe(struct i2c_client *i2c, wm8904->devtype = id->driver_data; i2c_set_clientdata(i2c, wm8904); - wm8904->control_data = i2c; wm8904->pdata = i2c->dev.platform_data; ret = snd_soc_register_codec(&i2c->dev, -- cgit v1.1 From 458f6f692105d4c08ef9e6b777022a829b1494b5 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Wed, 28 Sep 2011 15:14:56 +0800 Subject: ASoC: Fix setting adau1373_dai->master for SND_SOC_DAIFMT_CBS_CFS In the case of SND_SOC_DAIFMT_CBS_CFS, adau1373_dai->master should be false. Signed-off-by: Axel Lin Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/adau1373.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/adau1373.c b/sound/soc/codecs/adau1373.c index 2aa40c3..1ccf8dd 100644 --- a/sound/soc/codecs/adau1373.c +++ b/sound/soc/codecs/adau1373.c @@ -974,7 +974,7 @@ static int adau1373_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) break; case SND_SOC_DAIFMT_CBS_CFS: ctrl = 0; - adau1373_dai->master = true; + adau1373_dai->master = false; break; default: return -EINVAL; -- cgit v1.1 From 23d622b14b0209d243f5990cb4d369d4fffaf335 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Wed, 28 Sep 2011 15:21:28 +0800 Subject: ASoC: adau1701: Initialize codec->control_data before using it Currently codec->control_data is not initialized before calling process_sigma_firmware(codec->control_data, ADAU1701_FIRMWARE). Signed-off-by: Axel Lin Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/adau1701.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c index 2758d5f..b400afa 100644 --- a/sound/soc/codecs/adau1701.c +++ b/sound/soc/codecs/adau1701.c @@ -458,6 +458,7 @@ static int adau1701_probe(struct snd_soc_codec *codec) int ret; codec->dapm.idle_bias_off = 1; + codec->control_data = to_i2c_client(codec->dev); ret = adau1701_load_firmware(codec); if (ret) -- cgit v1.1 From 44cb209d33733790246afad6167c62a0a10ea9eb Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Wed, 28 Sep 2011 10:01:26 +0800 Subject: ASoC: Remove unused "control_data" field of struct alc5623_priv The control_data field is used to initialize the codec's control_data field, but since this is also done by the snd-soc-cache core, the redundant assignment can be removed and the field can be dropped. Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/alc5623.c | 2 -- 1 file changed, 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/alc5623.c b/sound/soc/codecs/alc5623.c index 0517315..557b3af 100644 --- a/sound/soc/codecs/alc5623.c +++ b/sound/soc/codecs/alc5623.c @@ -40,7 +40,6 @@ MODULE_PARM_DESC(caps_charge, "ALC5623 cap charge time (msecs)"); /* codec private data */ struct alc5623_priv { enum snd_soc_control_type control_type; - void *control_data; u8 id; unsigned int sysclk; u16 reg_cache[ALC5623_VENDOR_ID2+2]; @@ -1049,7 +1048,6 @@ static int alc5623_i2c_probe(struct i2c_client *client, } i2c_set_clientdata(client, alc5623); - alc5623->control_data = client; alc5623->control_type = SND_SOC_I2C; ret = snd_soc_register_codec(&client->dev, -- cgit v1.1 From 217069ea9a0ce579118f8a193f3534c8102d5ca8 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Wed, 28 Sep 2011 10:10:38 +0800 Subject: ASoC: Remove unused "control_data" field of struct cs4270_private The control_data field is used to initialize the codec's control_data field, but since this is also done by the snd-soc-cache core, the redundant assignment can be removed and the field can be dropped. Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/cs4270.c | 6 +----- 1 file changed, 1 insertion(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 6cc8678..5830c93 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -128,7 +128,6 @@ static const char *supply_names[] = { /* Private data for the CS4270 */ struct cs4270_private { enum snd_soc_control_type control_type; - void *control_data; unsigned int mclk; /* Input frequency of the MCLK pin */ unsigned int mode; /* The mode (I2S or left-justified) */ unsigned int slave_mode; @@ -490,8 +489,6 @@ static int cs4270_probe(struct snd_soc_codec *codec) struct cs4270_private *cs4270 = snd_soc_codec_get_drvdata(codec); int i, ret; - codec->control_data = cs4270->control_data; - /* Tell ASoC what kind of I/O to use to read the registers. ASoC will * then do the I2C transactions itself. */ @@ -604,7 +601,7 @@ static int cs4270_soc_suspend(struct snd_soc_codec *codec, pm_message_t mesg) static int cs4270_soc_resume(struct snd_soc_codec *codec) { struct cs4270_private *cs4270 = snd_soc_codec_get_drvdata(codec); - struct i2c_client *i2c_client = codec->control_data; + struct i2c_client *i2c_client = to_i2c_client(codec->dev); int reg; regulator_bulk_enable(ARRAY_SIZE(cs4270->supplies), @@ -690,7 +687,6 @@ static int cs4270_i2c_probe(struct i2c_client *i2c_client, } i2c_set_clientdata(i2c_client, cs4270); - cs4270->control_data = i2c_client; cs4270->control_type = SND_SOC_I2C; ret = snd_soc_register_codec(&i2c_client->dev, -- cgit v1.1 From 6d4f7097df481977d191cad85203fcf1ae3df8cf Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Wed, 28 Sep 2011 10:11:54 +0800 Subject: ASoC: Remove unused "control_data" field of struct cs42l51_private The control_data field is used to initialize the codec's control_data field, but since this is also done by the snd-soc-cache core, the redundant assignment can be removed and the field can be dropped. Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l51.c | 6 +----- 1 file changed, 1 insertion(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index 8fb7070..286878d 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -42,7 +42,6 @@ enum master_slave_mode { struct cs42l51_private { enum snd_soc_control_type control_type; - void *control_data; unsigned int mclk; unsigned int audio_mode; /* The mode (I2S or left-justified) */ enum master_slave_mode func; @@ -57,7 +56,7 @@ struct cs42l51_private { static int cs42l51_fill_cache(struct snd_soc_codec *codec) { u8 *cache = codec->reg_cache + 1; - struct i2c_client *i2c_client = codec->control_data; + struct i2c_client *i2c_client = to_i2c_client(codec->dev); s32 length; length = i2c_smbus_read_i2c_block_data(i2c_client, @@ -520,8 +519,6 @@ static int cs42l51_probe(struct snd_soc_codec *codec) struct snd_soc_dapm_context *dapm = &codec->dapm; int ret, reg; - codec->control_data = cs42l51->control_data; - ret = cs42l51_fill_cache(codec); if (ret < 0) { dev_err(codec->dev, "failed to fill register cache\n"); @@ -593,7 +590,6 @@ static int cs42l51_i2c_probe(struct i2c_client *i2c_client, } i2c_set_clientdata(i2c_client, cs42l51); - cs42l51->control_data = i2c_client; cs42l51->control_type = SND_SOC_I2C; ret = snd_soc_register_codec(&i2c_client->dev, -- cgit v1.1 From 72a921da070c9d7b5ac527ee20d80826100f0138 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Wed, 28 Sep 2011 10:12:48 +0800 Subject: ASoC: Remove unused "control_data" field of struct max98088_priv The control_data field is used to initialize the codec's control_data field, but since this is also done by the snd-soc-cache core, the redundant assignment can be removed and the field can be dropped. Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/max98088.c | 2 -- 1 file changed, 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index ac65a2d..587043b 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -40,7 +40,6 @@ struct max98088_cdata { struct max98088_priv { enum max98088_type devtype; - void *control_data; struct max98088_pdata *pdata; unsigned int sysclk; struct max98088_cdata dai[2]; @@ -2066,7 +2065,6 @@ static int max98088_i2c_probe(struct i2c_client *i2c, max98088->devtype = id->driver_data; i2c_set_clientdata(i2c, max98088); - max98088->control_data = i2c; max98088->pdata = i2c->dev.platform_data; ret = snd_soc_register_codec(&i2c->dev, -- cgit v1.1 From 63de012f35e4c48881ec14e9ec48ea92719fe3fb Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Wed, 28 Sep 2011 10:14:05 +0800 Subject: ASoC: Remove unused "control_data" field of struct max98095_priv The control_data field is used to initialize the codec's control_data field, but since this is also done by the snd-soc-cache core, the redundant assignment can be removed and the field can be dropped. Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/max98095.c | 2 -- 1 file changed, 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/max98095.c b/sound/soc/codecs/max98095.c index 668434d..8f8e255 100644 --- a/sound/soc/codecs/max98095.c +++ b/sound/soc/codecs/max98095.c @@ -40,7 +40,6 @@ struct max98095_cdata { struct max98095_priv { enum max98095_type devtype; - void *control_data; struct max98095_pdata *pdata; unsigned int sysclk; struct max98095_cdata dai[3]; @@ -2337,7 +2336,6 @@ static int max98095_i2c_probe(struct i2c_client *i2c, max98095->devtype = id->driver_data; i2c_set_clientdata(i2c, max98095); - max98095->control_data = i2c; max98095->pdata = i2c->dev.platform_data; ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_max98095, -- cgit v1.1 From 16b7a9aa9acfd4401f55731d39d63e8cb1665a45 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Wed, 28 Sep 2011 10:00:18 +0800 Subject: ASoC: Remove unused "control_data" field of struct ak4671_priv The control_data field is used to initialize the codec's control_data field, but since this is also done by the snd-soc-cache core, the redundant assignment can be removed and the field can be dropped. Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/ak4671.c | 2 -- 1 file changed, 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/ak4671.c b/sound/soc/codecs/ak4671.c index 88b29f8..2ecf128 100644 --- a/sound/soc/codecs/ak4671.c +++ b/sound/soc/codecs/ak4671.c @@ -26,7 +26,6 @@ /* codec private data */ struct ak4671_priv { enum snd_soc_control_type control_type; - void *control_data; }; /* ak4671 register cache & default register settings */ @@ -675,7 +674,6 @@ static int __devinit ak4671_i2c_probe(struct i2c_client *client, return -ENOMEM; i2c_set_clientdata(client, ak4671); - ak4671->control_data = client; ak4671->control_type = SND_SOC_I2C; ret = snd_soc_register_codec(&client->dev, -- cgit v1.1 From 21326db156b3d52983854c0071f17ef806f39156 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Wed, 28 Sep 2011 13:48:35 +0800 Subject: ASoC: adau1701: Fix prototype for adau1701_set_sysclk Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/adau1701.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c index b400afa..8b7e1c5 100644 --- a/sound/soc/codecs/adau1701.c +++ b/sound/soc/codecs/adau1701.c @@ -401,7 +401,7 @@ static int adau1701_digital_mute(struct snd_soc_dai *dai, int mute) } static int adau1701_set_sysclk(struct snd_soc_codec *codec, int clk_id, - unsigned int freq, int dir) + int source, unsigned int freq, int dir) { unsigned int val; -- cgit v1.1 From 75d9ac46b99280f5f381927ae75a9eaf21844d20 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 27 Sep 2011 16:41:01 +0100 Subject: ASoC: Allow DAI formats to be specified in the dai_link For almost all machines the DAI format is a constant, always set to the same thing. This means that not only should we normally set it on init rather than in hw_params() (where it has been for historical reasons) we should also allow users to configure this by setting a variable in the dai_link structure. The combination of these two will make many machine drivers even more data driven. Implement a new dai_fmt field in the dai_link doing just that. Since 0 is a valid value for many format flags and we need to be able to tell if the field is actually set also add one to all the values used to configure formats. Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 21 +++++++++++++++++++++ 1 file changed, 21 insertions(+) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index bd20154..a58c1fc 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1317,6 +1317,7 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) struct snd_soc_codec *codec; struct snd_soc_codec_conf *codec_conf; enum snd_soc_compress_type compress_type; + struct snd_soc_dai_link *dai_link; int ret, i, order; mutex_lock(&card->mutex); @@ -1429,6 +1430,26 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) snd_soc_dapm_add_routes(&card->dapm, card->dapm_routes, card->num_dapm_routes); + for (i = 0; i < card->num_links; i++) { + dai_link = &card->dai_link[i]; + + if (dai_link->dai_fmt) { + ret = snd_soc_dai_set_fmt(card->rtd[i].codec_dai, + dai_link->dai_fmt); + if (ret != 0) + dev_warn(card->rtd[i].codec_dai->dev, + "Failed to set DAI format: %d\n", + ret); + + ret = snd_soc_dai_set_fmt(card->rtd[i].cpu_dai, + dai_link->dai_fmt); + if (ret != 0) + dev_warn(card->rtd[i].cpu_dai->dev, + "Failed to set DAI format: %d\n", + ret); + } + } + snprintf(card->snd_card->shortname, sizeof(card->snd_card->shortname), "%s", card->name); snprintf(card->snd_card->longname, sizeof(card->snd_card->longname), -- cgit v1.1 From 44fdd4338749739692715f24907c450f6f410da3 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 27 Sep 2011 16:42:27 +0100 Subject: ASoC: Use dai_fmt in speyside_wm8962 Signed-off-by: Mark Brown --- sound/soc/samsung/speyside_wm8962.c | 19 ++----------------- 1 file changed, 2 insertions(+), 17 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/speyside_wm8962.c b/sound/soc/samsung/speyside_wm8962.c index 07b827e..8a08204 100644 --- a/sound/soc/samsung/speyside_wm8962.c +++ b/sound/soc/samsung/speyside_wm8962.c @@ -94,23 +94,6 @@ static int speyside_wm8962_set_bias_level_post(struct snd_soc_card *card, static int speyside_wm8962_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - int ret; - - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S - | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) - return ret; - - ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S - | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) - return ret; - sample_rate = params_rate(params); return 0; @@ -128,6 +111,8 @@ static struct snd_soc_dai_link speyside_wm8962_dai[] = { .codec_dai_name = "wm8962", .platform_name = "samsung-audio", .codec_name = "wm8962.1-001a", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM, .ops = &speyside_wm8962_ops, }, }; -- cgit v1.1 From a8fdac83a3703c7f35d4efe37a14e38aa256919b Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 28 Sep 2011 18:20:26 +0100 Subject: ASoC: Also count neighbour checks for supplies Missed when the stat was originally added. Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 2bde6b0..c277228 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -828,6 +828,8 @@ static int dapm_supply_check_power(struct snd_soc_dapm_widget *w) /* Check if one of our outputs is connected */ list_for_each_entry(path, &w->sinks, list_source) { + DAPM_UPDATE_STAT(w, neighbour_checks); + if (path->weak) continue; -- cgit v1.1 From c29429f3b72fe0b593f674378e99f22d5f8bea1f Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Thu, 29 Sep 2011 12:09:57 +0800 Subject: ASoC: tlv320dac33: Add guarding parentheses to macros Put parentheses around macro argument uses. This avoids pitfalls for the programmer, where the argument expansion does not give the expected result, for example: SAMPLES_TO_US(substream->runtime->rate, dac33->uthr - DAC33_MODE7_MARGIN + 1); Signed-off-by: Axel Lin Acked-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320dac33.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index faa5e9f..43ee3b1 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -55,13 +55,13 @@ #define BURST_BASEFREQ_HZ 49152000 #define SAMPLES_TO_US(rate, samples) \ - (1000000000 / ((rate * 1000) / samples)) + (1000000000 / (((rate) * 1000) / (samples))) #define US_TO_SAMPLES(rate, us) \ - (rate / (1000000 / (us < 1000000 ? us : 1000000))) + ((rate) / (1000000 / ((us) < 1000000 ? (us) : 1000000))) #define UTHR_FROM_PERIOD_SIZE(samples, playrate, burstrate) \ - ((samples * 5000) / ((burstrate * 5000) / (burstrate - playrate))) + (((samples)*5000) / (((burstrate)*5000) / ((burstrate) - (playrate)))) static void dac33_calculate_times(struct snd_pcm_substream *substream); static int dac33_prepare_chip(struct snd_pcm_substream *substream); -- cgit v1.1 From fbc7c62a3ff831aef24894b7982cd1adb2b7e070 Mon Sep 17 00:00:00 2001 From: Susan Gao Date: Thu, 29 Sep 2011 11:08:18 +0100 Subject: ASoC: Fix a bug in WM8962 DSP_A and DSP_B settings Signed-off-by: Susan Gao Signed-off-by: Mark Brown Cc: stable@kernel.org --- sound/soc/codecs/wm8962.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index bc6bdde..74ebbfa 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3223,9 +3223,9 @@ static int wm8962_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) int aif0 = 0; switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { - case SND_SOC_DAIFMT_DSP_A: - aif0 |= WM8962_LRCLK_INV; case SND_SOC_DAIFMT_DSP_B: + aif0 |= WM8962_LRCLK_INV | 3; + case SND_SOC_DAIFMT_DSP_A: aif0 |= 3; switch (fmt & SND_SOC_DAIFMT_INV_MASK) { -- cgit v1.1 From f79e5e8ce221c0c2e0754eb7076ba7611f209001 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 29 Sep 2011 17:30:06 +0100 Subject: ASoC: Staticise non-exported symbols in rt5631 Signed-off-by: Mark Brown --- sound/soc/codecs/rt5631.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt5631.c b/sound/soc/codecs/rt5631.c index 889a7dd..7e1f894 100644 --- a/sound/soc/codecs/rt5631.c +++ b/sound/soc/codecs/rt5631.c @@ -1308,7 +1308,7 @@ static const struct pll_div codec_slave_pll_div[] = { {3072000, 12288000, 0x0a90}, }; -struct coeff_clk_div coeff_div[] = { +static struct coeff_clk_div coeff_div[] = { /* sysclk is 256fs */ {2048000, 8000 * 32, 8000, 0x1000}, {2048000, 8000 * 64, 8000, 0x0000}, @@ -1680,7 +1680,7 @@ static int rt5631_resume(struct snd_soc_codec *codec) SNDRV_PCM_FMTBIT_S24_LE | \ SNDRV_PCM_FMTBIT_S8) -struct snd_soc_dai_ops rt5631_ops = { +static struct snd_soc_dai_ops rt5631_ops = { .hw_params = rt5631_hifi_pcm_params, .set_fmt = rt5631_hifi_codec_set_dai_fmt, .set_sysclk = rt5631_hifi_codec_set_dai_sysclk, @@ -1762,7 +1762,7 @@ static __devexit int rt5631_i2c_remove(struct i2c_client *client) return 0; } -struct i2c_driver rt5631_i2c_driver = { +static struct i2c_driver rt5631_i2c_driver = { .driver = { .name = "rt5631", .owner = THIS_MODULE, -- cgit v1.1 From 6d447be0141991d80433e098d6267f7498ba6071 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 29 Sep 2011 17:32:17 +0100 Subject: ASoC: Remove unused function check_vdac_to_outmix from rt5631 Signed-off-by: Mark Brown --- sound/soc/codecs/rt5631.c | 14 -------------- 1 file changed, 14 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt5631.c b/sound/soc/codecs/rt5631.c index 7e1f894..86e69f4 100644 --- a/sound/soc/codecs/rt5631.c +++ b/sound/soc/codecs/rt5631.c @@ -343,20 +343,6 @@ static int check_dacr_to_spkmixr(struct snd_soc_dapm_widget *source, return !(reg & RT5631_M_DAC_R_TO_SPKMIXER_R); } -static int check_vdac_to_outmix(struct snd_soc_dapm_widget *source, - struct snd_soc_dapm_widget *sink) -{ - unsigned int reg, ret = 1; - - reg = snd_soc_read(source->codec, RT5631_OUTMIXER_L_CTRL); - if (reg & RT5631_M_VDAC_TO_OUTMIXER_L) { - reg = snd_soc_read(source->codec, RT5631_OUTMIXER_R_CTRL); - if (reg & RT5631_M_VDAC_TO_OUTMIXER_R) - ret = 0; - } - return ret; -} - static int check_adcl_select(struct snd_soc_dapm_widget *source, struct snd_soc_dapm_widget *sink) { -- cgit v1.1 From 00e982a6a333a7749bfce51cbefa5cf4f48c64ee Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Fri, 30 Sep 2011 10:31:10 +0800 Subject: ASoC: Remove unused "control_data" field of struct cs4271_private The control_data field is used to initialize the codec's control_data field, but since this is also done by the snd-soc-cache core, the redundant assignment can be removed and the field can be dropped. Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/cs4271.c | 5 ----- 1 file changed, 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index 083aab9..23d1bd5 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -156,7 +156,6 @@ static const u8 cs4271_dflt_reg[CS4271_NR_REGS] = { struct cs4271_private { /* SND_SOC_I2C or SND_SOC_SPI */ enum snd_soc_control_type bus_type; - void *control_data; unsigned int mclk; bool master; bool deemph; @@ -466,8 +465,6 @@ static int cs4271_probe(struct snd_soc_codec *codec) int ret; int gpio_nreset = -EINVAL; - codec->control_data = cs4271->control_data; - if (cs4271plat && gpio_is_valid(cs4271plat->gpio_nreset)) gpio_nreset = cs4271plat->gpio_nreset; @@ -555,7 +552,6 @@ static int __devinit cs4271_spi_probe(struct spi_device *spi) return -ENOMEM; spi_set_drvdata(spi, cs4271); - cs4271->control_data = spi; cs4271->bus_type = SND_SOC_SPI; return snd_soc_register_codec(&spi->dev, &soc_codec_dev_cs4271, @@ -595,7 +591,6 @@ static int __devinit cs4271_i2c_probe(struct i2c_client *client, return -ENOMEM; i2c_set_clientdata(client, cs4271); - cs4271->control_data = client; cs4271->bus_type = SND_SOC_I2C; return snd_soc_register_codec(&client->dev, &soc_codec_dev_cs4271, -- cgit v1.1 From ad51f765447f55699e3280086b32502f9a391345 Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Fri, 30 Sep 2011 10:55:33 +0300 Subject: ASoC: Davinci: Fix FS polarity for I2S format Commit 75d9ac4 ("ASoC: Allow DAI formats to be specified in the dai_link") changed DAI format flag values and we cannot simply invert anymore e.g. frame-sync with ^= SND_SOC_DAIFMT_NB_IF (which was anyway misuse) as there is no anymore fixed bit position for bit-clock or frame-sync inversion. Fix this by relying only on DAI format flag values passed to us and by not making any assumption on individual bit positions Signed-off-by: Jarkko Nikula Cc: Vaibhav Bedia Cc: Sekhar Nori Cc: Kevin Hilman Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-i2s.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index d0d60b8..300e121 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -265,6 +265,7 @@ static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, struct davinci_mcbsp_dev *dev = snd_soc_dai_get_drvdata(cpu_dai); unsigned int pcr; unsigned int srgr; + bool inv_fs = false; /* Attention srgr is updated by hw_params! */ srgr = DAVINCI_MCBSP_SRGR_FSGM | DAVINCI_MCBSP_SRGR_FPER(DEFAULT_BITPERSAMPLE * 2 - 1) | @@ -330,7 +331,7 @@ static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, * more empty bit clock slots between channels as the sample * rate is lowered. */ - fmt ^= SND_SOC_DAIFMT_NB_IF; + inv_fs = true; case SND_SOC_DAIFMT_DSP_A: dev->mode = MOD_DSP_A; break; @@ -394,6 +395,8 @@ static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, default: return -EINVAL; } + if (inv_fs == true) + pcr ^= (DAVINCI_MCBSP_PCR_FSXP | DAVINCI_MCBSP_PCR_FSRP); davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SRGR_REG, srgr); dev->pcr = pcr; davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_PCR_REG, pcr); -- cgit v1.1 From 5382ffbb179459b81b36ef47e58b84e3e1eff862 Mon Sep 17 00:00:00 2001 From: "Ujfalusi, Peter" Date: Fri, 30 Sep 2011 11:39:50 +0300 Subject: ASoC: sdp4430: Fix string for FM input name The name contains invalid valid character (/), which causes problems when trying to create the debugfs directory structure: ASoC: Failed to create Aux/FM Stereo In debugfs file Signed-off-by: Peter Ujfalusi Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/omap/sdp4430.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/sdp4430.c b/sound/soc/omap/sdp4430.c index 249d84b..79ec76d 100644 --- a/sound/soc/omap/sdp4430.c +++ b/sound/soc/omap/sdp4430.c @@ -88,7 +88,7 @@ static const struct snd_soc_dapm_widget sdp4430_twl6040_dapm_widgets[] = { SND_SOC_DAPM_MIC("Headset Mic", NULL), SND_SOC_DAPM_HP("Headset Stereophone", NULL), SND_SOC_DAPM_SPK("Earphone Spk", NULL), - SND_SOC_DAPM_INPUT("Aux/FM Stereo In"), + SND_SOC_DAPM_INPUT("FM Stereo In"), }; static const struct snd_soc_dapm_route audio_map[] = { @@ -113,8 +113,8 @@ static const struct snd_soc_dapm_route audio_map[] = { {"Earphone Spk", NULL, "EP"}, /* Aux/FM Stereo In: AFML, AFMR */ - {"AFML", NULL, "Aux/FM Stereo In"}, - {"AFMR", NULL, "Aux/FM Stereo In"}, + {"AFML", NULL, "FM Stereo In"}, + {"AFMR", NULL, "FM Stereo In"}, }; static int sdp4430_twl6040_init(struct snd_soc_pcm_runtime *rtd) -- cgit v1.1 From 6423aa9154e247752e8894ad686959d39be659f9 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Fri, 30 Sep 2011 10:32:37 +0800 Subject: ASoC: Remove unused "control_data" field of struct aic3x_priv The control_data field is used to initialize the codec's control_data field, but since this is also done by the snd-soc-cache core, the redundant assignment can be removed and the field can be dropped. Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic3x.c | 3 --- 1 file changed, 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 0963c4c..4c5eab5 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -76,7 +76,6 @@ struct aic3x_priv { struct aic3x_disable_nb disable_nb[AIC3X_NUM_SUPPLIES]; enum snd_soc_control_type control_type; struct aic3x_setup_data *setup; - void *control_data; unsigned int sysclk; struct list_head list; int master; @@ -1383,7 +1382,6 @@ static int aic3x_probe(struct snd_soc_codec *codec) int ret, i; INIT_LIST_HEAD(&aic3x->list); - codec->control_data = aic3x->control_data; aic3x->codec = codec; codec->dapm.idle_bias_off = 1; @@ -1520,7 +1518,6 @@ static int aic3x_i2c_probe(struct i2c_client *i2c, return -ENOMEM; } - aic3x->control_data = i2c; aic3x->control_type = SND_SOC_I2C; i2c_set_clientdata(i2c, aic3x); -- cgit v1.1 From 5992c58781a38e193caf2fb6a5f4808d84af8591 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Fri, 30 Sep 2011 11:54:44 +0800 Subject: ASoC: Add missed regulator_unregister_notifier and regulator_bulk_free in wm8995_remove Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/wm8995.c | 7 +++++++ 1 file changed, 7 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm8995.c b/sound/soc/codecs/wm8995.c index 74ae599..e05ee79 100644 --- a/sound/soc/codecs/wm8995.c +++ b/sound/soc/codecs/wm8995.c @@ -1573,9 +1573,16 @@ static int wm8995_resume(struct snd_soc_codec *codec) static int wm8995_remove(struct snd_soc_codec *codec) { struct wm8995_priv *wm8995; + int i; wm8995 = snd_soc_codec_get_drvdata(codec); wm8995_set_bias_level(codec, SND_SOC_BIAS_OFF); + + for (i = 0; i < ARRAY_SIZE(wm8995->supplies); ++i) + regulator_unregister_notifier(wm8995->supplies[i].consumer, + &wm8995->disable_nb[i]); + + regulator_bulk_free(ARRAY_SIZE(wm8995->supplies), wm8995->supplies); return 0; } -- cgit v1.1 From f34dafb287a33ffda2f2a122daecedea474a4181 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Fri, 30 Sep 2011 13:56:59 +0800 Subject: ASoC: sn95031: Do not use static variable for channel_index No reason to use static variable for channel_index. Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/sn95031.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c index 8f8ce5f..5c5a4ab 100644 --- a/sound/soc/codecs/sn95031.c +++ b/sound/soc/codecs/sn95031.c @@ -102,7 +102,7 @@ static int sn95031_initialize_adc(struct snd_soc_codec *sn95031_codec) { int base_addr, chnl_addr; int value; - static int channel_index; + int channel_index; /* Index of the first channel in which the stop bit is set */ channel_index = find_free_channel(sn95031_codec); -- cgit v1.1 From 91a18ae8ffc05ffd445c5b40e5ca266888b360ce Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Fri, 30 Sep 2011 10:55:32 +0300 Subject: ASoC: omap-mcbsp: Fix FS polarity for LEFT_J, DSP_A and DSP_B formats Commit 75d9ac4 ("ASoC: Allow DAI formats to be specified in the dai_link") changed DAI format flag values and we cannot simply invert anymore e.g. frame-sync with ^= SND_SOC_DAIFMT_NB_IF (which was anyway misuse) as there is no anymore fixed bit position for bit-clock or frame-sync inversion. Fix this by relying only on DAI format flag values passed to us and by not making any assumption on individual bit positions. Signed-off-by: Jarkko Nikula Acked-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/omap/omap-mcbsp.c | 12 +++++++----- 1 file changed, 7 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 1391ea0..894f2f3 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -398,7 +398,7 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai, { struct omap_mcbsp_data *mcbsp_data = snd_soc_dai_get_drvdata(cpu_dai); struct omap_mcbsp_reg_cfg *regs = &mcbsp_data->regs; - unsigned int temp_fmt = fmt; + bool inv_fs = false; if (mcbsp_data->configured) return 0; @@ -430,21 +430,21 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai, regs->xcr2 |= XDATDLY(0); regs->spcr1 |= RJUST(2); /* Invert FS polarity configuration */ - temp_fmt ^= SND_SOC_DAIFMT_NB_IF; + inv_fs = true; break; case SND_SOC_DAIFMT_DSP_A: /* 1-bit data delay */ regs->rcr2 |= RDATDLY(1); regs->xcr2 |= XDATDLY(1); /* Invert FS polarity configuration */ - temp_fmt ^= SND_SOC_DAIFMT_NB_IF; + inv_fs = true; break; case SND_SOC_DAIFMT_DSP_B: /* 0-bit data delay */ regs->rcr2 |= RDATDLY(0); regs->xcr2 |= XDATDLY(0); /* Invert FS polarity configuration */ - temp_fmt ^= SND_SOC_DAIFMT_NB_IF; + inv_fs = true; break; default: /* Unsupported data format */ @@ -468,7 +468,7 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai, } /* Set bit clock (CLKX/CLKR) and FS polarities */ - switch (temp_fmt & SND_SOC_DAIFMT_INV_MASK) { + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { case SND_SOC_DAIFMT_NB_NF: /* * Normal BCLK + FS. @@ -489,6 +489,8 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai, default: return -EINVAL; } + if (inv_fs == true) + regs->pcr0 ^= FSXP | FSRP; return 0; } -- cgit v1.1 From a46737aee59e4e001106e1d3777e0801843361db Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 29 Sep 2011 15:22:34 +0300 Subject: ASoC: twl6040: One workqueue should be enough It is a bit overkill to have three (3) separate workqueue for a single driver. We can manage things with one workqueue nicely. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl6040.c | 33 +++++---------------------------- 1 file changed, 5 insertions(+), 28 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index c973347..1afc596 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -103,8 +103,6 @@ struct twl6040_data { struct mutex mutex; struct twl6040_output headset; struct twl6040_output handsfree; - struct workqueue_struct *hf_workqueue; - struct workqueue_struct *hs_workqueue; }; /* @@ -562,20 +560,17 @@ static int out_drv_event(struct snd_soc_dapm_widget *w, struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec); struct twl6040_output *out; struct delayed_work *work; - struct workqueue_struct *queue; switch (w->shift) { case 2: case 3: out = &priv->headset; - queue = priv->hs_workqueue; out->left_step = priv->hs_left_step; out->right_step = priv->hs_right_step; out->step_delay = 5; /* 5 ms between volume ramp steps */ break; case 4: out = &priv->handsfree; - queue = priv->hf_workqueue; out->left_step = priv->hf_left_step; out->right_step = priv->hf_right_step; out->step_delay = 5; /* 5 ms between volume ramp steps */ @@ -601,7 +596,7 @@ static int out_drv_event(struct snd_soc_dapm_widget *w, if (!delayed_work_pending(work)) { out->ramp = TWL6040_RAMP_UP; - queue_delayed_work(queue, work, + queue_delayed_work(priv->workqueue, work, msecs_to_jiffies(1)); } break; @@ -615,7 +610,7 @@ static int out_drv_event(struct snd_soc_dapm_widget *w, out->ramp = TWL6040_RAMP_DOWN; INIT_COMPLETION(out->ramp_done); - queue_delayed_work(queue, work, + queue_delayed_work(priv->workqueue, work, msecs_to_jiffies(1)); wait_for_completion_timeout(&out->ramp_done, @@ -1512,33 +1507,21 @@ static int twl6040_probe(struct snd_soc_codec *codec) goto work_err; } - priv->workqueue = create_singlethread_workqueue("twl6040-codec"); + priv->workqueue = alloc_workqueue("twl6040-codec", 0, 0); if (!priv->workqueue) { ret = -ENOMEM; goto work_err; } INIT_DELAYED_WORK(&priv->hs_jack.work, twl6040_accessory_work); + INIT_DELAYED_WORK(&priv->headset.work, twl6040_pga_hs_work); + INIT_DELAYED_WORK(&priv->handsfree.work, twl6040_pga_hf_work); mutex_init(&priv->mutex); init_completion(&priv->headset.ramp_done); init_completion(&priv->handsfree.ramp_done); - priv->hf_workqueue = create_singlethread_workqueue("twl6040-hf"); - if (priv->hf_workqueue == NULL) { - ret = -ENOMEM; - goto hfwq_err; - } - priv->hs_workqueue = create_singlethread_workqueue("twl6040-hs"); - if (priv->hs_workqueue == NULL) { - ret = -ENOMEM; - goto hswq_err; - } - - INIT_DELAYED_WORK(&priv->headset.work, twl6040_pga_hs_work); - INIT_DELAYED_WORK(&priv->handsfree.work, twl6040_pga_hf_work); - ret = request_threaded_irq(priv->plug_irq, NULL, twl6040_audio_handler, 0, "twl6040_irq_plug", codec); if (ret) { @@ -1562,10 +1545,6 @@ static int twl6040_probe(struct snd_soc_codec *codec) bias_err: free_irq(priv->plug_irq, codec); plugirq_err: - destroy_workqueue(priv->hs_workqueue); -hswq_err: - destroy_workqueue(priv->hf_workqueue); -hfwq_err: destroy_workqueue(priv->workqueue); work_err: kfree(priv); @@ -1579,8 +1558,6 @@ static int twl6040_remove(struct snd_soc_codec *codec) twl6040_set_bias_level(codec, SND_SOC_BIAS_OFF); free_irq(priv->plug_irq, codec); destroy_workqueue(priv->workqueue); - destroy_workqueue(priv->hf_workqueue); - destroy_workqueue(priv->hs_workqueue); kfree(priv); return 0; -- cgit v1.1 From 93eebc6982161f317c4a99118a4423bc3933fdfa Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 29 Sep 2011 15:22:35 +0300 Subject: ASoC: twl6040: correct loop counters for HS/HF ramp code The Headset gain range is 0 - 0xf (4 bit resolution) The Handsfree gain range is 0 - 0x1d (5 bit resolution, 0x1e, and 0x1f values are invalid) Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl6040.c | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 1afc596..d706bd00 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -495,8 +495,8 @@ static void twl6040_pga_hs_work(struct work_struct *work) if (headset->ramp == TWL6040_RAMP_NONE) return; - /* HS PGA volumes have 4 bits of resolution to ramp */ - for (i = 0; i <= 16; i++) { + /* HS PGA gain range: 0x0 - 0xf (0 - 15) */ + for (i = 0; i < 16; i++) { headset_complete = twl6040_hs_ramp_step(codec, headset->left_step, headset->right_step); @@ -530,8 +530,9 @@ static void twl6040_pga_hf_work(struct work_struct *work) if (handsfree->ramp == TWL6040_RAMP_NONE) return; - /* HF PGA volumes have 5 bits of resolution to ramp */ - for (i = 0; i <= 32; i++) { + /* + * HF PGA gain range: 0x00 - 0x1d (0 - 29) */ + for (i = 0; i < 30; i++) { handsfree_complete = twl6040_hf_ramp_step(codec, handsfree->left_step, handsfree->right_step); -- cgit v1.1 From 6fbb32d175368b6ab8fb827e65cd8d18ed04c1f3 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 29 Sep 2011 15:22:36 +0300 Subject: ASoC: twl6040: Shift 2 identifies the HS output in out_drv_event None of the driver handled by out_drv_event have it's power bit shifted by 3. Remove the case for shift 3, and also add comment for the cases. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl6040.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index d706bd00..738d102 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -563,14 +563,13 @@ static int out_drv_event(struct snd_soc_dapm_widget *w, struct delayed_work *work; switch (w->shift) { - case 2: - case 3: + case 2: /* Headset output driver */ out = &priv->headset; out->left_step = priv->hs_left_step; out->right_step = priv->hs_right_step; out->step_delay = 5; /* 5 ms between volume ramp steps */ break; - case 4: + case 4: /* Handsfree output driver */ out = &priv->handsfree; out->left_step = priv->hf_left_step; out->right_step = priv->hf_right_step; -- cgit v1.1 From 009d196b4755b42c02414b287889a337955f7e09 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 29 Sep 2011 15:22:37 +0300 Subject: ASoC: twl6040: Simplify code in out_drv_event for pending work check Instead of checking, if the work is pending, it is safer to cancel the pending work, or wait till the scheduled work finishes. This way we can avoid modifying the variables used by the work function. Since we know that no work is pending, we can remove the two additional checks in POST_PMU, and PRE_PMD for non pending works. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl6040.c | 38 ++++++++++++++++++++++---------------- 1 file changed, 22 insertions(+), 16 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 738d102..d040905 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -565,12 +565,26 @@ static int out_drv_event(struct snd_soc_dapm_widget *w, switch (w->shift) { case 2: /* Headset output driver */ out = &priv->headset; + work = &out->work; + /* + * Make sure, that we do not mess up variables for already + * executing work. + */ + cancel_delayed_work_sync(work); + out->left_step = priv->hs_left_step; out->right_step = priv->hs_right_step; out->step_delay = 5; /* 5 ms between volume ramp steps */ break; case 4: /* Handsfree output driver */ out = &priv->handsfree; + work = &out->work; + /* + * Make sure, that we do not mess up variables for already + * executing work. + */ + cancel_delayed_work_sync(work); + out->left_step = priv->hf_left_step; out->right_step = priv->hf_right_step; out->step_delay = 5; /* 5 ms between volume ramp steps */ @@ -583,39 +597,31 @@ static int out_drv_event(struct snd_soc_dapm_widget *w, return -1; } - work = &out->work; - switch (event) { case SND_SOC_DAPM_POST_PMU: if (out->active) break; /* don't use volume ramp for power-up */ + out->ramp = TWL6040_RAMP_UP; out->left_step = out->left_vol; out->right_step = out->right_vol; - if (!delayed_work_pending(work)) { - out->ramp = TWL6040_RAMP_UP; - queue_delayed_work(priv->workqueue, work, - msecs_to_jiffies(1)); - } + queue_delayed_work(priv->workqueue, work, msecs_to_jiffies(1)); break; case SND_SOC_DAPM_PRE_PMD: if (!out->active) break; - if (!delayed_work_pending(work)) { - /* use volume ramp for power-down */ - out->ramp = TWL6040_RAMP_DOWN; - INIT_COMPLETION(out->ramp_done); + /* use volume ramp for power-down */ + out->ramp = TWL6040_RAMP_DOWN; + INIT_COMPLETION(out->ramp_done); - queue_delayed_work(priv->workqueue, work, - msecs_to_jiffies(1)); + queue_delayed_work(priv->workqueue, work, msecs_to_jiffies(1)); - wait_for_completion_timeout(&out->ramp_done, - msecs_to_jiffies(2000)); - } + wait_for_completion_timeout(&out->ramp_done, + msecs_to_jiffies(2000)); break; } -- cgit v1.1 From 177fdd89f9c3f3f157c0b5e0f9c25a3a7c37ecf7 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Wed, 28 Sep 2011 21:56:48 +0800 Subject: ASoC: tlv320aic3x: Use driver_data field of struct i2c_device_id to identify models Save model information in driver_data so we can simplify the implementation. Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic3x.c | 13 ++++--------- 1 file changed, 4 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 4c5eab5..d877b39 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1493,9 +1493,9 @@ static struct snd_soc_codec_driver soc_codec_dev_aic3x = { */ static const struct i2c_device_id aic3x_i2c_id[] = { - [AIC3X_MODEL_3X] = { "tlv320aic3x", 0 }, - [AIC3X_MODEL_33] = { "tlv320aic33", 0 }, - [AIC3X_MODEL_3007] = { "tlv320aic3007", 0 }, + { "tlv320aic3x", AIC3X_MODEL_3X }, + { "tlv320aic33", AIC3X_MODEL_33 }, + { "tlv320aic3007", AIC3X_MODEL_3007 }, { } }; MODULE_DEVICE_TABLE(i2c, aic3x_i2c_id); @@ -1510,7 +1510,6 @@ static int aic3x_i2c_probe(struct i2c_client *i2c, struct aic3x_pdata *pdata = i2c->dev.platform_data; struct aic3x_priv *aic3x; int ret; - const struct i2c_device_id *tbl; aic3x = kzalloc(sizeof(struct aic3x_priv), GFP_KERNEL); if (aic3x == NULL) { @@ -1528,11 +1527,7 @@ static int aic3x_i2c_probe(struct i2c_client *i2c, aic3x->gpio_reset = -1; } - for (tbl = aic3x_i2c_id; tbl->name[0]; tbl++) { - if (!strcmp(tbl->name, id->name)) - break; - } - aic3x->model = tbl - aic3x_i2c_id; + aic3x->model = id->driver_data; ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_aic3x, &aic3x_dai, 1); -- cgit v1.1 From eff919ac0fc7565e71ffa35657c333dd8cdc0520 Mon Sep 17 00:00:00 2001 From: Arnd Bergmann Date: Sat, 1 Oct 2011 22:03:34 +0200 Subject: ASoC: use a valid device for dev_err() in Zylonite A recent conversion has introduced references to &pdev->dev, which does not actually exist in all the contexts it's used in. Replace this with card->dev where necessary, in order to let the driver build again. Signed-off-by: Arnd Bergmann Signed-off-by: Mark Brown Cc: stable@kernel.org --- sound/soc/pxa/zylonite.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/pxa/zylonite.c b/sound/soc/pxa/zylonite.c index b644575..2b8350b 100644 --- a/sound/soc/pxa/zylonite.c +++ b/sound/soc/pxa/zylonite.c @@ -196,20 +196,20 @@ static int zylonite_probe(struct snd_soc_card *card) if (clk_pout) { pout = clk_get(NULL, "CLK_POUT"); if (IS_ERR(pout)) { - dev_err(&pdev->dev, "Unable to obtain CLK_POUT: %ld\n", + dev_err(card->dev, "Unable to obtain CLK_POUT: %ld\n", PTR_ERR(pout)); return PTR_ERR(pout); } ret = clk_enable(pout); if (ret != 0) { - dev_err(&pdev->dev, "Unable to enable CLK_POUT: %d\n", + dev_err(card->dev, "Unable to enable CLK_POUT: %d\n", ret); clk_put(pout); return ret; } - dev_dbg(&pdev->dev, "MCLK enabled at %luHz\n", + dev_dbg(card->dev, "MCLK enabled at %luHz\n", clk_get_rate(pout)); } @@ -241,7 +241,7 @@ static int zylonite_resume_pre(struct snd_soc_card *card) if (clk_pout) { ret = clk_enable(pout); if (ret != 0) - dev_err(&pdev->dev, "Unable to enable CLK_POUT: %d\n", + dev_err(card->dev, "Unable to enable CLK_POUT: %d\n", ret); } -- cgit v1.1 From 21d17dd2a377ba894f26989915eb3c6e427a3656 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Sun, 2 Oct 2011 20:41:04 +0800 Subject: ASoC: Fix setting update bits for WM8753_LADC and WM8753_RADC Current code set update bits for WM8753_LDAC and WM8753_RDAC twice, but missed setting update bits for WM8753_LADC and WM8753_RADC. I think it is a copy-paste bug in commit 776065 "ASoC: codecs: wm8753: Fix register cache incoherency". Signed-off-by: Axel Lin Signed-off-by: Mark Brown Cc: stable@kernel.org --- sound/soc/codecs/wm8753.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index ffa2ffe..aa091a0 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -1454,8 +1454,8 @@ static int wm8753_probe(struct snd_soc_codec *codec) /* set the update bits */ snd_soc_update_bits(codec, WM8753_LDAC, 0x0100, 0x0100); snd_soc_update_bits(codec, WM8753_RDAC, 0x0100, 0x0100); - snd_soc_update_bits(codec, WM8753_LDAC, 0x0100, 0x0100); - snd_soc_update_bits(codec, WM8753_RDAC, 0x0100, 0x0100); + snd_soc_update_bits(codec, WM8753_LADC, 0x0100, 0x0100); + snd_soc_update_bits(codec, WM8753_RADC, 0x0100, 0x0100); snd_soc_update_bits(codec, WM8753_LOUT1V, 0x0100, 0x0100); snd_soc_update_bits(codec, WM8753_ROUT1V, 0x0100, 0x0100); snd_soc_update_bits(codec, WM8753_LOUT2V, 0x0100, 0x0100); -- cgit v1.1 From 4dd0417253be35bfbe368c40ec5a10732b24fd65 Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Fri, 30 Sep 2011 16:07:44 +0300 Subject: ASoC: omap-mcbsp: Prepare for init time DAI format setting Before commit 75d9ac4 ("ASoC: Allow DAI formats to be specified in the dai_link") expectation for omap-mcbsp was that snd_soc_dai_set_fmt is to be called first in machine hw_params callback before other CPU DAI functions. Thus it was enough that only omap_mcbsp_dai_set_dai_fmt cleared the mcbsp->regs structure. [Note that this was pure convention, it's always been OK to set things on init -- broonie] Now this doesn't hold anymore since machine drivers can set the DAI format only once on init time and thus mcbsp->regs may get out of sync when other CPU DAI functions are modifying them dynamically with different values between the calls. Therefore clear the accessed mcbsp->regs bits and bitfields in other functions too. Signed-off-by: Jarkko Nikula Cc: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/omap/omap-mcbsp.c | 9 +++++++++ 1 file changed, 9 insertions(+) (limited to 'sound') diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 894f2f3..7f70061 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -317,6 +317,10 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, return 0; } + regs->rcr2 &= ~(RPHASE | RFRLEN2(0x7f) | RWDLEN2(7)); + regs->xcr2 &= ~(RPHASE | XFRLEN2(0x7f) | XWDLEN2(7)); + regs->rcr1 &= ~(RFRLEN1(0x7f) | RWDLEN1(7)); + regs->xcr1 &= ~(XFRLEN1(0x7f) | XWDLEN1(7)); format = mcbsp_data->fmt & SND_SOC_DAIFMT_FORMAT_MASK; wpf = channels = params_channels(params); if (channels == 2 && (format == SND_SOC_DAIFMT_I2S || @@ -369,6 +373,8 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, framesize = wlen * channels; /* Set FS period and length in terms of bit clock periods */ + regs->srgr2 &= ~FPER(0xfff); + regs->srgr1 &= ~FWID(0xff); switch (format) { case SND_SOC_DAIFMT_I2S: case SND_SOC_DAIFMT_LEFT_J: @@ -505,6 +511,7 @@ static int omap_mcbsp_dai_set_clkdiv(struct snd_soc_dai *cpu_dai, return -ENODEV; mcbsp_data->clk_div = div; + regs->srgr1 &= ~CLKGDV(0xff); regs->srgr1 |= CLKGDV(div - 1); return 0; @@ -534,6 +541,8 @@ static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai, return -EINVAL; mcbsp_data->in_freq = freq; + regs->srgr2 &= ~CLKSM; + regs->pcr0 &= ~SCLKME; switch (clk_id) { case OMAP_MCBSP_SYSCLK_CLK: -- cgit v1.1 From cf9feff28fc1f00c82fb0cc016307d4c65da132a Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Fri, 30 Sep 2011 16:07:45 +0300 Subject: ASoC: omap: Convert bunch of machine drivers to use init time DAI format Signed-off-by: Jarkko Nikula Cc: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/omap/am3517evm.c | 22 ++------------------- sound/soc/omap/igep0020.c | 23 ++-------------------- sound/soc/omap/n810.c | 19 ++---------------- sound/soc/omap/omap3evm.c | 23 ++-------------------- sound/soc/omap/omap3pandora.c | 20 ++++--------------- sound/soc/omap/osk5912.c | 23 ++-------------------- sound/soc/omap/overo.c | 23 ++-------------------- sound/soc/omap/rx51.c | 20 ++----------------- sound/soc/omap/sdp3430.c | 46 ++++--------------------------------------- sound/soc/omap/zoom2.c | 46 ++++--------------------------------------- 10 files changed, 26 insertions(+), 239 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/am3517evm.c b/sound/soc/omap/am3517evm.c index 73dde4a..48af0f8 100644 --- a/sound/soc/omap/am3517evm.c +++ b/sound/soc/omap/am3517evm.c @@ -43,26 +43,6 @@ static int am3517evm_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai = rtd->cpu_dai; int ret; - /* Set codec DAI configuration */ - ret = snd_soc_dai_set_fmt(codec_dai, - SND_SOC_DAIFMT_DSP_B | - SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) { - printk(KERN_ERR "can't set codec DAI configuration\n"); - return ret; - } - - /* Set cpu DAI configuration */ - ret = snd_soc_dai_set_fmt(cpu_dai, - SND_SOC_DAIFMT_DSP_B | - SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) { - printk(KERN_ERR "can't set cpu DAI configuration\n"); - return ret; - } - /* Set the codec system clock for DAC and ADC */ ret = snd_soc_dai_set_sysclk(codec_dai, 0, CODEC_CLOCK, SND_SOC_CLOCK_IN); @@ -140,6 +120,8 @@ static struct snd_soc_dai_link am3517evm_dai = { .codec_dai_name = "tlv320aic23-hifi", .platform_name = "omap-pcm-audio", .codec_name = "tlv320aic23-codec.2-001a", + .dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM, .init = am3517evm_aic23_init, .ops = &am3517evm_ops, }; diff --git a/sound/soc/omap/igep0020.c b/sound/soc/omap/igep0020.c index 0ae3470..84615a7 100644 --- a/sound/soc/omap/igep0020.c +++ b/sound/soc/omap/igep0020.c @@ -38,29 +38,8 @@ static int igep2_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; int ret; - /* Set codec DAI configuration */ - ret = snd_soc_dai_set_fmt(codec_dai, - SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) { - printk(KERN_ERR "can't set codec DAI configuration\n"); - return ret; - } - - /* Set cpu DAI configuration */ - ret = snd_soc_dai_set_fmt(cpu_dai, - SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) { - printk(KERN_ERR "can't set cpu DAI configuration\n"); - return ret; - } - /* Set the codec system clock for DAC and ADC */ ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000, SND_SOC_CLOCK_IN); @@ -84,6 +63,8 @@ static struct snd_soc_dai_link igep2_dai = { .codec_dai_name = "twl4030-hifi", .platform_name = "omap-pcm-audio", .codec_name = "twl4030-codec", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM, .ops = &igep2_ops, }; diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c index 62e292f..c10d356 100644 --- a/sound/soc/omap/n810.c +++ b/sound/soc/omap/n810.c @@ -115,25 +115,8 @@ static int n810_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; int err; - /* Set codec DAI configuration */ - err = snd_soc_dai_set_fmt(codec_dai, - SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM); - if (err < 0) - return err; - - /* Set cpu DAI configuration */ - err = snd_soc_dai_set_fmt(cpu_dai, - SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM); - if (err < 0) - return err; - /* Set the codec system clock for DAC and ADC */ err = snd_soc_dai_set_sysclk(codec_dai, 0, 12000000, SND_SOC_CLOCK_IN); @@ -312,6 +295,8 @@ static struct snd_soc_dai_link n810_dai = { .platform_name = "omap-pcm-audio", .codec_name = "tlv320aic3x-codec.2-0018", .codec_dai_name = "tlv320aic3x-hifi", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM, .init = n810_aic33_init, .ops = &n810_ops, }; diff --git a/sound/soc/omap/omap3evm.c b/sound/soc/omap/omap3evm.c index 0daa044..bf9ae2a 100644 --- a/sound/soc/omap/omap3evm.c +++ b/sound/soc/omap/omap3evm.c @@ -36,29 +36,8 @@ static int omap3evm_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; int ret; - /* Set codec DAI configuration */ - ret = snd_soc_dai_set_fmt(codec_dai, - SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) { - printk(KERN_ERR "Can't set codec DAI configuration\n"); - return ret; - } - - /* Set cpu DAI configuration */ - ret = snd_soc_dai_set_fmt(cpu_dai, - SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) { - printk(KERN_ERR "Can't set cpu DAI configuration\n"); - return ret; - } - /* Set the codec system clock for DAC and ADC */ ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000, SND_SOC_CLOCK_IN); @@ -82,6 +61,8 @@ static struct snd_soc_dai_link omap3evm_dai = { .codec_dai_name = "twl4030-hifi", .platform_name = "omap-pcm-audio", .codec_name = "twl4030-codec", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM, .ops = &omap3evm_ops, }; diff --git a/sound/soc/omap/omap3pandora.c b/sound/soc/omap/omap3pandora.c index 8047c521e..3ae87fd 100644 --- a/sound/soc/omap/omap3pandora.c +++ b/sound/soc/omap/omap3pandora.c @@ -48,24 +48,8 @@ static int omap3pandora_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - int fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS; int ret; - /* Set codec DAI configuration */ - ret = snd_soc_dai_set_fmt(codec_dai, fmt); - if (ret < 0) { - pr_err(PREFIX "can't set codec DAI configuration\n"); - return ret; - } - - /* Set cpu DAI configuration */ - ret = snd_soc_dai_set_fmt(cpu_dai, fmt); - if (ret < 0) { - pr_err(PREFIX "can't set cpu DAI configuration\n"); - return ret; - } - /* Set the codec system clock for DAC and ADC */ ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000, SND_SOC_CLOCK_IN); @@ -231,6 +215,8 @@ static struct snd_soc_dai_link omap3pandora_dai[] = { .codec_dai_name = "twl4030-hifi", .platform_name = "omap-pcm-audio", .codec_name = "twl4030-codec", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, .ops = &omap3pandora_ops, .init = omap3pandora_out_init, }, { @@ -240,6 +226,8 @@ static struct snd_soc_dai_link omap3pandora_dai[] = { .codec_dai_name = "twl4030-hifi", .platform_name = "omap-pcm-audio", .codec_name = "twl4030-codec", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, .ops = &omap3pandora_ops, .init = omap3pandora_in_init, } diff --git a/sound/soc/omap/osk5912.c b/sound/soc/omap/osk5912.c index 7e75e77..5978332 100644 --- a/sound/soc/omap/osk5912.c +++ b/sound/soc/omap/osk5912.c @@ -55,29 +55,8 @@ static int osk_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; int err; - /* Set codec DAI configuration */ - err = snd_soc_dai_set_fmt(codec_dai, - SND_SOC_DAIFMT_DSP_B | - SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM); - if (err < 0) { - printk(KERN_ERR "can't set codec DAI configuration\n"); - return err; - } - - /* Set cpu DAI configuration */ - err = snd_soc_dai_set_fmt(cpu_dai, - SND_SOC_DAIFMT_DSP_B | - SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM); - if (err < 0) { - printk(KERN_ERR "can't set cpu DAI configuration\n"); - return err; - } - /* Set the codec system clock for DAC and ADC */ err = snd_soc_dai_set_sysclk(codec_dai, 0, CODEC_CLOCK, SND_SOC_CLOCK_IN); @@ -141,6 +120,8 @@ static struct snd_soc_dai_link osk_dai = { .codec_dai_name = "tlv320aic23-hifi", .platform_name = "omap-pcm-audio", .codec_name = "tlv320aic23-codec", + .dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM, .init = osk_tlv320aic23_init, .ops = &osk_ops, }; diff --git a/sound/soc/omap/overo.c b/sound/soc/omap/overo.c index bbcf380..739efe9 100644 --- a/sound/soc/omap/overo.c +++ b/sound/soc/omap/overo.c @@ -38,29 +38,8 @@ static int overo_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; int ret; - /* Set codec DAI configuration */ - ret = snd_soc_dai_set_fmt(codec_dai, - SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) { - printk(KERN_ERR "can't set codec DAI configuration\n"); - return ret; - } - - /* Set cpu DAI configuration */ - ret = snd_soc_dai_set_fmt(cpu_dai, - SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) { - printk(KERN_ERR "can't set cpu DAI configuration\n"); - return ret; - } - /* Set the codec system clock for DAC and ADC */ ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000, SND_SOC_CLOCK_IN); @@ -84,6 +63,8 @@ static struct snd_soc_dai_link overo_dai = { .codec_dai_name = "twl4030-hifi", .platform_name = "omap-pcm-audio", .codec_name = "twl4030-codec", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM, .ops = &overo_ops, }; diff --git a/sound/soc/omap/rx51.c b/sound/soc/omap/rx51.c index 893300a..7164db5 100644 --- a/sound/soc/omap/rx51.c +++ b/sound/soc/omap/rx51.c @@ -115,24 +115,6 @@ static int rx51_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - int err; - - /* Set codec DAI configuration */ - err = snd_soc_dai_set_fmt(codec_dai, - SND_SOC_DAIFMT_DSP_A | - SND_SOC_DAIFMT_IB_NF | - SND_SOC_DAIFMT_CBM_CFM); - if (err < 0) - return err; - - /* Set cpu DAI configuration */ - err = snd_soc_dai_set_fmt(cpu_dai, - SND_SOC_DAIFMT_DSP_A | - SND_SOC_DAIFMT_IB_NF | - SND_SOC_DAIFMT_CBM_CFM); - if (err < 0) - return err; /* Set the codec system clock for DAC and ADC */ return snd_soc_dai_set_sysclk(codec_dai, 0, 19200000, @@ -377,6 +359,8 @@ static struct snd_soc_dai_link rx51_dai[] = { .codec_dai_name = "tlv320aic3x-hifi", .platform_name = "omap-pcm-audio", .codec_name = "tlv320aic3x-codec.2-0018", + .dai_fmt = SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_IB_NF | + SND_SOC_DAIFMT_CBM_CFM, .init = rx51_aic34_init, .ops = &rx51_ops, }, diff --git a/sound/soc/omap/sdp3430.c b/sound/soc/omap/sdp3430.c index 9f6a758..2ff5f7b 100644 --- a/sound/soc/omap/sdp3430.c +++ b/sound/soc/omap/sdp3430.c @@ -53,29 +53,8 @@ static int sdp3430_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; int ret; - /* Set codec DAI configuration */ - ret = snd_soc_dai_set_fmt(codec_dai, - SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) { - printk(KERN_ERR "can't set codec DAI configuration\n"); - return ret; - } - - /* Set cpu DAI configuration */ - ret = snd_soc_dai_set_fmt(cpu_dai, - SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) { - printk(KERN_ERR "can't set cpu DAI configuration\n"); - return ret; - } - /* Set the codec system clock for DAC and ADC */ ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000, SND_SOC_CLOCK_IN); @@ -96,29 +75,8 @@ static int sdp3430_hw_voice_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; int ret; - /* Set codec DAI configuration */ - ret = snd_soc_dai_set_fmt(codec_dai, - SND_SOC_DAIFMT_DSP_A | - SND_SOC_DAIFMT_IB_NF | - SND_SOC_DAIFMT_CBM_CFM); - if (ret) { - printk(KERN_ERR "can't set codec DAI configuration\n"); - return ret; - } - - /* Set cpu DAI configuration */ - ret = snd_soc_dai_set_fmt(cpu_dai, - SND_SOC_DAIFMT_DSP_A | - SND_SOC_DAIFMT_IB_NF | - SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) { - printk(KERN_ERR "can't set cpu DAI configuration\n"); - return ret; - } - /* Set the codec system clock for DAC and ADC */ ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000, SND_SOC_CLOCK_IN); @@ -267,6 +225,8 @@ static struct snd_soc_dai_link sdp3430_dai[] = { .codec_dai_name = "twl4030-hifi", .platform_name = "omap-pcm-audio", .codec_name = "twl4030-codec", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM, .init = sdp3430_twl4030_init, .ops = &sdp3430_ops, }, @@ -277,6 +237,8 @@ static struct snd_soc_dai_link sdp3430_dai[] = { .codec_dai_name = "twl4030-voice", .platform_name = "omap-pcm-audio", .codec_name = "twl4030-codec", + .dai_fmt = SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_IB_NF | + SND_SOC_DAIFMT_CBM_CFM, .init = sdp3430_twl4030_voice_init, .ops = &sdp3430_voice_ops, }, diff --git a/sound/soc/omap/zoom2.c b/sound/soc/omap/zoom2.c index 9a2666f..c124562 100644 --- a/sound/soc/omap/zoom2.c +++ b/sound/soc/omap/zoom2.c @@ -44,29 +44,8 @@ static int zoom2_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; int ret; - /* Set codec DAI configuration */ - ret = snd_soc_dai_set_fmt(codec_dai, - SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) { - printk(KERN_ERR "can't set codec DAI configuration\n"); - return ret; - } - - /* Set cpu DAI configuration */ - ret = snd_soc_dai_set_fmt(cpu_dai, - SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) { - printk(KERN_ERR "can't set cpu DAI configuration\n"); - return ret; - } - /* Set the codec system clock for DAC and ADC */ ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000, SND_SOC_CLOCK_IN); @@ -87,29 +66,8 @@ static int zoom2_hw_voice_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; int ret; - /* Set codec DAI configuration */ - ret = snd_soc_dai_set_fmt(codec_dai, - SND_SOC_DAIFMT_DSP_A | - SND_SOC_DAIFMT_IB_NF | - SND_SOC_DAIFMT_CBM_CFM); - if (ret) { - printk(KERN_ERR "can't set codec DAI configuration\n"); - return ret; - } - - /* Set cpu DAI configuration */ - ret = snd_soc_dai_set_fmt(cpu_dai, - SND_SOC_DAIFMT_DSP_A | - SND_SOC_DAIFMT_IB_NF | - SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) { - printk(KERN_ERR "can't set cpu DAI configuration\n"); - return ret; - } - /* Set the codec system clock for DAC and ADC */ ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000, SND_SOC_CLOCK_IN); @@ -217,6 +175,8 @@ static struct snd_soc_dai_link zoom2_dai[] = { .codec_dai_name = "twl4030-hifi", .platform_name = "omap-pcm-audio", .codec_name = "twl4030-codec", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM, .init = zoom2_twl4030_init, .ops = &zoom2_ops, }, @@ -227,6 +187,8 @@ static struct snd_soc_dai_link zoom2_dai[] = { .codec_dai_name = "twl4030-voice", .platform_name = "omap-pcm-audio", .codec_name = "twl4030-codec", + .dai_fmt = SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_IB_NF | + SND_SOC_DAIFMT_CBM_CFM, .init = zoom2_twl4030_voice_init, .ops = &zoom2_voice_ops, }, -- cgit v1.1 From a3c6dac201005077f78ea5b18c89e6cd568e2d85 Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Fri, 30 Sep 2011 16:07:46 +0300 Subject: ASoC: omap: Use single hw_params callback in sdp3430 and zoom2 There is no need to use two hw_params callbacks in sdp3430 and zoom2 as thet are now identical. Use instead the same snd_soc_ops structure and hw_params callback for both DAI links. Signed-off-by: Jarkko Nikula Cc: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/omap/sdp3430.c | 24 +----------------------- sound/soc/omap/zoom2.c | 24 +----------------------- 2 files changed, 2 insertions(+), 46 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/sdp3430.c b/sound/soc/omap/sdp3430.c index 2ff5f7b..269aded 100644 --- a/sound/soc/omap/sdp3430.c +++ b/sound/soc/omap/sdp3430.c @@ -70,28 +70,6 @@ static struct snd_soc_ops sdp3430_ops = { .hw_params = sdp3430_hw_params, }; -static int sdp3430_hw_voice_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - int ret; - - /* Set the codec system clock for DAC and ADC */ - ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000, - SND_SOC_CLOCK_IN); - if (ret < 0) { - printk(KERN_ERR "can't set codec system clock\n"); - return ret; - } - - return 0; -} - -static struct snd_soc_ops sdp3430_voice_ops = { - .hw_params = sdp3430_hw_voice_params, -}; - /* Headset jack */ static struct snd_soc_jack hs_jack; @@ -240,7 +218,7 @@ static struct snd_soc_dai_link sdp3430_dai[] = { .dai_fmt = SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_IB_NF | SND_SOC_DAIFMT_CBM_CFM, .init = sdp3430_twl4030_voice_init, - .ops = &sdp3430_voice_ops, + .ops = &sdp3430_ops, }, }; diff --git a/sound/soc/omap/zoom2.c b/sound/soc/omap/zoom2.c index c124562..8b1ebbc 100644 --- a/sound/soc/omap/zoom2.c +++ b/sound/soc/omap/zoom2.c @@ -61,28 +61,6 @@ static struct snd_soc_ops zoom2_ops = { .hw_params = zoom2_hw_params, }; -static int zoom2_hw_voice_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - int ret; - - /* Set the codec system clock for DAC and ADC */ - ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000, - SND_SOC_CLOCK_IN); - if (ret < 0) { - printk(KERN_ERR "can't set codec system clock\n"); - return ret; - } - - return 0; -} - -static struct snd_soc_ops zoom2_voice_ops = { - .hw_params = zoom2_hw_voice_params, -}; - /* Zoom2 machine DAPM */ static const struct snd_soc_dapm_widget zoom2_twl4030_dapm_widgets[] = { SND_SOC_DAPM_MIC("Ext Mic", NULL), @@ -190,7 +168,7 @@ static struct snd_soc_dai_link zoom2_dai[] = { .dai_fmt = SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_IB_NF | SND_SOC_DAIFMT_CBM_CFM, .init = zoom2_twl4030_voice_init, - .ops = &zoom2_voice_ops, + .ops = &zoom2_ops, }, }; -- cgit v1.1 From 3f0456bfd7761efbd71e76db5606ecce81dc3d1e Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Sun, 2 Oct 2011 08:55:02 +0800 Subject: ASoC: wm8782: Add __devexit_p at necessary place According to the comments in include/linux/init.h: "Pointers to __devexit functions must use __devexit_p(function_name), the wrapper will insert either the function_name or NULL, depending on the config options." We have __devexit annotation for wm8782_remove(), thus add __devexit_p at necessary place. Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/wm8782.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8782.c b/sound/soc/codecs/wm8782.c index a2a09f8..f2ced71 100644 --- a/sound/soc/codecs/wm8782.c +++ b/sound/soc/codecs/wm8782.c @@ -60,7 +60,7 @@ static struct platform_driver wm8782_codec_driver = { .owner = THIS_MODULE, }, .probe = wm8782_probe, - .remove = wm8782_remove, + .remove = __devexit_p(wm8782_remove), }; static int __init wm8782_init(void) -- cgit v1.1 From 48860e409dd4d333a953ac871b538a80d0ef068b Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Sun, 2 Oct 2011 09:18:17 +0800 Subject: ASoC: kirkwood-i2s: Add __devexit_p at necessary place According to the comments in include/linux/init.h: "Pointers to __devexit functions must use __devexit_p(function_name), the wrapper will insert either the function_name or NULL, depending on the config options." We have __devexit annotation for kirkwood_i2s_dev_remove(), thus add __devexit_p at necessary place. Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/kirkwood/kirkwood-i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index d0bcf3f..715e841 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -476,7 +476,7 @@ static __devexit int kirkwood_i2s_dev_remove(struct platform_device *pdev) static struct platform_driver kirkwood_i2s_driver = { .probe = kirkwood_i2s_dev_probe, - .remove = kirkwood_i2s_dev_remove, + .remove = __devexit_p(kirkwood_i2s_dev_remove), .driver = { .name = DRV_NAME, .owner = THIS_MODULE, -- cgit v1.1 From c4c5839f9828de60682802367013c1dd375c46cf Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Sun, 2 Oct 2011 11:20:13 +0800 Subject: ASoC: samsung: Add __devexit_p at necessary places According to the comments in include/linux/init.h: "Pointers to __devexit functions must use __devexit_p(function_name), the wrapper will insert either the function_name or NULL, depending on the confi options." Signed-off-by: Axel Lin Cc: Jaswinder Singh Cc: Ben Dooks Cc: Seungwhan Youn Cc: Jassi Brar Signed-off-by: Mark Brown --- sound/soc/samsung/ac97.c | 2 +- sound/soc/samsung/i2s.c | 2 +- sound/soc/samsung/pcm.c | 2 +- sound/soc/samsung/s3c2412-i2s.c | 2 +- sound/soc/samsung/s3c24xx-i2s.c | 2 +- sound/soc/samsung/spdif.c | 2 +- 6 files changed, 6 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/ac97.c b/sound/soc/samsung/ac97.c index f97110e..65ea538 100644 --- a/sound/soc/samsung/ac97.c +++ b/sound/soc/samsung/ac97.c @@ -495,7 +495,7 @@ static __devexit int s3c_ac97_remove(struct platform_device *pdev) static struct platform_driver s3c_ac97_driver = { .probe = s3c_ac97_probe, - .remove = s3c_ac97_remove, + .remove = __devexit_p(s3c_ac97_remove), .driver = { .name = "samsung-ac97", .owner = THIS_MODULE, diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index c086b78..0c9ac20 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -1136,7 +1136,7 @@ static __devexit int samsung_i2s_remove(struct platform_device *pdev) static struct platform_driver samsung_i2s_driver = { .probe = samsung_i2s_probe, - .remove = samsung_i2s_remove, + .remove = __devexit_p(samsung_i2s_remove), .driver = { .name = "samsung-i2s", .owner = THIS_MODULE, diff --git a/sound/soc/samsung/pcm.c b/sound/soc/samsung/pcm.c index 9c7e8b4..e55d7a5 100644 --- a/sound/soc/samsung/pcm.c +++ b/sound/soc/samsung/pcm.c @@ -624,7 +624,7 @@ static __devexit int s3c_pcm_dev_remove(struct platform_device *pdev) static struct platform_driver s3c_pcm_driver = { .probe = s3c_pcm_dev_probe, - .remove = s3c_pcm_dev_remove, + .remove = __devexit_p(s3c_pcm_dev_remove), .driver = { .name = "samsung-pcm", .owner = THIS_MODULE, diff --git a/sound/soc/samsung/s3c2412-i2s.c b/sound/soc/samsung/s3c2412-i2s.c index 7ab8e2c..f26a8bf 100644 --- a/sound/soc/samsung/s3c2412-i2s.c +++ b/sound/soc/samsung/s3c2412-i2s.c @@ -176,7 +176,7 @@ static __devexit int s3c2412_iis_dev_remove(struct platform_device *pdev) static struct platform_driver s3c2412_iis_driver = { .probe = s3c2412_iis_dev_probe, - .remove = s3c2412_iis_dev_remove, + .remove = __devexit_p(s3c2412_iis_dev_remove), .driver = { .name = "s3c2412-iis", .owner = THIS_MODULE, diff --git a/sound/soc/samsung/s3c24xx-i2s.c b/sound/soc/samsung/s3c24xx-i2s.c index 21c92e2..c08117e 100644 --- a/sound/soc/samsung/s3c24xx-i2s.c +++ b/sound/soc/samsung/s3c24xx-i2s.c @@ -481,7 +481,7 @@ static __devexit int s3c24xx_iis_dev_remove(struct platform_device *pdev) static struct platform_driver s3c24xx_iis_driver = { .probe = s3c24xx_iis_dev_probe, - .remove = s3c24xx_iis_dev_remove, + .remove = __devexit_p(s3c24xx_iis_dev_remove), .driver = { .name = "s3c24xx-iis", .owner = THIS_MODULE, diff --git a/sound/soc/samsung/spdif.c b/sound/soc/samsung/spdif.c index 28c491d..c82b471 100644 --- a/sound/soc/samsung/spdif.c +++ b/sound/soc/samsung/spdif.c @@ -475,7 +475,7 @@ static __devexit int spdif_remove(struct platform_device *pdev) static struct platform_driver samsung_spdif_driver = { .probe = spdif_probe, - .remove = spdif_remove, + .remove = __devexit_p(spdif_remove), .driver = { .name = "samsung-spdif", .owner = THIS_MODULE, -- cgit v1.1 From 019cd3b25aafb3befefb2872c3a9900f37340172 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Sun, 2 Oct 2011 17:41:40 +0800 Subject: ASoC: tegra: Staticise tegra_i2s_dai and tegra_spdif_dai Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/tegra/tegra_i2s.c | 2 +- sound/soc/tegra/tegra_spdif.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/tegra/tegra_i2s.c b/sound/soc/tegra/tegra_i2s.c index f36b996..6728fab 100644 --- a/sound/soc/tegra/tegra_i2s.c +++ b/sound/soc/tegra/tegra_i2s.c @@ -312,7 +312,7 @@ static struct snd_soc_dai_ops tegra_i2s_dai_ops = { .trigger = tegra_i2s_trigger, }; -struct snd_soc_dai_driver tegra_i2s_dai[] = { +static struct snd_soc_dai_driver tegra_i2s_dai[] = { { .name = DRV_NAME ".0", .probe = tegra_i2s_probe, diff --git a/sound/soc/tegra/tegra_spdif.c b/sound/soc/tegra/tegra_spdif.c index abe606b..1bd07d1 100644 --- a/sound/soc/tegra/tegra_spdif.c +++ b/sound/soc/tegra/tegra_spdif.c @@ -232,7 +232,7 @@ static struct snd_soc_dai_ops tegra_spdif_dai_ops = { .trigger = tegra_spdif_trigger, }; -struct snd_soc_dai_driver tegra_spdif_dai = { +static struct snd_soc_dai_driver tegra_spdif_dai = { .name = DRV_NAME, .probe = tegra_spdif_probe, .playback = { -- cgit v1.1 From bc6bd90eb7bb7978e6e8d741071aaac14325e72f Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Sun, 2 Oct 2011 17:42:30 +0800 Subject: ASoC: Staticise samsung_spdif_dai Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/samsung/spdif.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/samsung/spdif.c b/sound/soc/samsung/spdif.c index c82b471..3122f31 100644 --- a/sound/soc/samsung/spdif.c +++ b/sound/soc/samsung/spdif.c @@ -340,7 +340,7 @@ static struct snd_soc_dai_ops spdif_dai_ops = { .shutdown = spdif_shutdown, }; -struct snd_soc_dai_driver samsung_spdif_dai = { +static struct snd_soc_dai_driver samsung_spdif_dai = { .name = "samsung-spdif", .playback = { .stream_name = "S/PDIF Playback", -- cgit v1.1 From 8fccf46905b05baa08f69516cabc229ef222f5d7 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Sun, 2 Oct 2011 17:43:16 +0800 Subject: ASoC: Staticise sh4_ssi_dai Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/sh/ssi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/sh/ssi.c b/sound/soc/sh/ssi.c index 05192d9..e0c621c 100644 --- a/sound/soc/sh/ssi.c +++ b/sound/soc/sh/ssi.c @@ -342,7 +342,7 @@ static struct snd_soc_dai_ops ssi_dai_ops = { .set_fmt = ssi_set_fmt, }; -struct snd_soc_dai_driver sh4_ssi_dai[] = { +static struct snd_soc_dai_driver sh4_ssi_dai[] = { { .name = "ssi-dai.0", .playback = { -- cgit v1.1 From c6add3f0e690e156d2e028e34916793497546152 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Sun, 2 Oct 2011 21:05:56 +0800 Subject: ASoC: Remove unused rate variable in magician_playback_hw_params Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/pxa/magician.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/pxa/magician.c b/sound/soc/pxa/magician.c index 67dcc36..098e5f4 100644 --- a/sound/soc/pxa/magician.c +++ b/sound/soc/pxa/magician.c @@ -92,11 +92,10 @@ static int magician_playback_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - unsigned int acps, acds, width, rate; + unsigned int acps, acds, width; unsigned int div4 = PXA_SSP_CLK_SCDB_4; int ret = 0; - rate = params_rate(params); width = snd_pcm_format_physical_width(params_format(params)); /* -- cgit v1.1 From 4b8713fd5459619ba8a5a43afcfe7b47a57a8d82 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Sun, 2 Oct 2011 21:07:02 +0800 Subject: ASoC: Remove unused srate variable in tegra_spdif_hw_params Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/tegra/tegra_spdif.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/tegra/tegra_spdif.c b/sound/soc/tegra/tegra_spdif.c index 1bd07d1..dd11d0c 100644 --- a/sound/soc/tegra/tegra_spdif.c +++ b/sound/soc/tegra/tegra_spdif.c @@ -127,7 +127,7 @@ static int tegra_spdif_hw_params(struct snd_pcm_substream *substream, { struct device *dev = substream->pcm->card->dev; struct tegra_spdif *spdif = snd_soc_dai_get_drvdata(dai); - int ret, srate, spdifclock; + int ret, spdifclock; spdif->reg_ctrl &= ~TEGRA_SPDIF_CTRL_PACK; spdif->reg_ctrl &= ~TEGRA_SPDIF_CTRL_BIT_MODE_MASK; @@ -140,7 +140,6 @@ static int tegra_spdif_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - srate = params_rate(params); switch (params_rate(params)) { case 32000: spdifclock = 4096000; -- cgit v1.1 From b5c49d49b9e175fd56cb4b5cf2c4fd972d15e013 Mon Sep 17 00:00:00 2001 From: Arnd Bergmann Date: Sun, 2 Oct 2011 16:45:31 +0200 Subject: ASoC: omap_mcpdm_remove cannot be __devexit omap_mcpdm_remove is used from asoc_mcpdm_probe, which is an initcall, and must not be discarded when HOTPLUG is disabled. Signed-off-by: Arnd Bergmann Signed-off-by: Mark Brown --- sound/soc/omap/mcpdm.c | 2 +- sound/soc/omap/mcpdm.h | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/mcpdm.c b/sound/soc/omap/mcpdm.c index 928f037..50e5919 100644 --- a/sound/soc/omap/mcpdm.c +++ b/sound/soc/omap/mcpdm.c @@ -449,7 +449,7 @@ exit: return ret; } -int __devexit omap_mcpdm_remove(struct platform_device *pdev) +int omap_mcpdm_remove(struct platform_device *pdev) { struct omap_mcpdm *mcpdm_ptr = platform_get_drvdata(pdev); diff --git a/sound/soc/omap/mcpdm.h b/sound/soc/omap/mcpdm.h index df3e16f..20c20a8 100644 --- a/sound/soc/omap/mcpdm.h +++ b/sound/soc/omap/mcpdm.h @@ -150,4 +150,4 @@ extern int omap_mcpdm_request(void); extern void omap_mcpdm_free(void); extern int omap_mcpdm_set_offset(int offset1, int offset2); int __devinit omap_mcpdm_probe(struct platform_device *pdev); -int __devexit omap_mcpdm_remove(struct platform_device *pdev); +int omap_mcpdm_remove(struct platform_device *pdev); -- cgit v1.1 From b381bc8ab8bb22ed1581033094c2c0af1a034fcc Mon Sep 17 00:00:00 2001 From: Arnd Bergmann Date: Sun, 2 Oct 2011 22:28:01 +0200 Subject: ASoC: imx: eukrea_tlv320 needs i2c Add a missing dependency that is required for random configurations. Signed-off-by: Arnd Bergmann Signed-off-by: Mark Brown --- sound/soc/imx/Kconfig | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/imx/Kconfig b/sound/soc/imx/Kconfig index bb699bb..dcd954b 100644 --- a/sound/soc/imx/Kconfig +++ b/sound/soc/imx/Kconfig @@ -50,6 +50,7 @@ config SND_SOC_EUKREA_TLV320 || MACH_EUKREA_MBIMXSD25_BASEBOARD \ || MACH_EUKREA_MBIMXSD35_BASEBOARD \ || MACH_EUKREA_MBIMXSD51_BASEBOARD + depends on I2C select SND_SOC_TLV320AIC23 select SND_MXC_SOC_FIQ help -- cgit v1.1 From ca7aceef21f2689d0b0e92aa4f316959f7931c25 Mon Sep 17 00:00:00 2001 From: Arnd Bergmann Date: Sun, 2 Oct 2011 22:28:02 +0200 Subject: ASoC: sh: use correct __iomem annotations This removes a few unnecessary type casts and avoids sparse warnings. Signed-off-by: Arnd Bergmann Signed-off-by: Mark Brown --- sound/soc/sh/fsi.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 8e112cc..916b9f9 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -210,7 +210,7 @@ struct fsi_master { * basic read write function */ -static void __fsi_reg_write(u32 reg, u32 data) +static void __fsi_reg_write(u32 __iomem *reg, u32 data) { /* valid data area is 24bit */ data &= 0x00ffffff; @@ -218,12 +218,12 @@ static void __fsi_reg_write(u32 reg, u32 data) __raw_writel(data, reg); } -static u32 __fsi_reg_read(u32 reg) +static u32 __fsi_reg_read(u32 __iomem *reg) { return __raw_readl(reg); } -static void __fsi_reg_mask_set(u32 reg, u32 mask, u32 data) +static void __fsi_reg_mask_set(u32 __iomem *reg, u32 mask, u32 data) { u32 val = __fsi_reg_read(reg); @@ -250,7 +250,7 @@ static u32 _fsi_master_read(struct fsi_master *master, u32 reg) unsigned long flags; spin_lock_irqsave(&master->lock, flags); - ret = __fsi_reg_read((u32)(master->base + reg)); + ret = __fsi_reg_read(master->base + reg); spin_unlock_irqrestore(&master->lock, flags); return ret; @@ -264,7 +264,7 @@ static void _fsi_master_mask_set(struct fsi_master *master, unsigned long flags; spin_lock_irqsave(&master->lock, flags); - __fsi_reg_mask_set((u32)(master->base + reg), mask, data); + __fsi_reg_mask_set(master->base + reg, mask, data); spin_unlock_irqrestore(&master->lock, flags); } -- cgit v1.1 From 43419b80fa46ee94d4b50ac6ebb1ee1ca5bbbcc7 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Mon, 3 Oct 2011 20:17:16 +0800 Subject: ASoC: Remove needless codec->dapm.bias_level assignment to SND_SOC_BIAS_OFF This assignment is done by the snd_soc_register_codec so there is no need to redo it in probe function of a codec driver. Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/sn95031.c | 1 - sound/soc/codecs/wm5100.c | 2 -- sound/soc/codecs/wm8996.c | 1 - 3 files changed, 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c index 5c5a4ab..f681e41 100644 --- a/sound/soc/codecs/sn95031.c +++ b/sound/soc/codecs/sn95031.c @@ -826,7 +826,6 @@ static int sn95031_codec_probe(struct snd_soc_codec *codec) { pr_debug("codec_probe called\n"); - codec->dapm.bias_level = SND_SOC_BIAS_OFF; codec->dapm.idle_bias_off = 1; /* PCM interface config diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index f603989..46afdf8 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -2285,8 +2285,6 @@ static int wm5100_probe(struct snd_soc_codec *codec) wm5100->codec = codec; - codec->dapm.bias_level = SND_SOC_BIAS_OFF; - ret = snd_soc_codec_set_cache_io(codec, 16, 16, SND_SOC_I2C); if (ret != 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index b98a8f8..43e46c7 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -2706,7 +2706,6 @@ static int wm8996_probe(struct snd_soc_codec *codec) init_completion(&wm8996->fll_lock); dapm->idle_bias_off = true; - dapm->bias_level = SND_SOC_BIAS_OFF; ret = snd_soc_codec_set_cache_io(codec, 16, 16, SND_SOC_I2C); if (ret != 0) { -- cgit v1.1 From 2e4891a5ec6e7c5147a6803bfdabe4f1c94d6bce Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Mon, 3 Oct 2011 16:11:52 +0800 Subject: ASoC: Staticise simtec_audio_resume() It is exported via resume callback of struct dev_pm_ops rather than referenced directly and so should be staticised. Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/samsung/s3c24xx_simtec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/samsung/s3c24xx_simtec.c b/sound/soc/samsung/s3c24xx_simtec.c index 349566f..c8d525b 100644 --- a/sound/soc/samsung/s3c24xx_simtec.c +++ b/sound/soc/samsung/s3c24xx_simtec.c @@ -300,7 +300,7 @@ static void detach_gpio_amp(struct s3c24xx_audio_simtec_pdata *pd) } #ifdef CONFIG_PM -int simtec_audio_resume(struct device *dev) +static int simtec_audio_resume(struct device *dev) { simtec_call_startup(pdata); return 0; -- cgit v1.1 From 1c6927f872990c43b73e0cd4ccdf1831725f3c4a Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Mon, 3 Oct 2011 11:33:05 +0800 Subject: ASoC: Staticise ep93xx_ac97_dai Signed-off-by: Axel Lin Acked-by: Mika Westerberg Signed-off-by: Mark Brown --- sound/soc/ep93xx/ep93xx-ac97.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/ep93xx/ep93xx-ac97.c b/sound/soc/ep93xx/ep93xx-ac97.c index c7417c7..3cd6158 100644 --- a/sound/soc/ep93xx/ep93xx-ac97.c +++ b/sound/soc/ep93xx/ep93xx-ac97.c @@ -335,7 +335,7 @@ static struct snd_soc_dai_ops ep93xx_ac97_dai_ops = { .trigger = ep93xx_ac97_trigger, }; -struct snd_soc_dai_driver ep93xx_ac97_dai = { +static struct snd_soc_dai_driver ep93xx_ac97_dai = { .name = "ep93xx-ac97", .id = 0, .ac97_control = 1, -- cgit v1.1 From f6a74ced462a3acb8fc222176e690b157dc9e181 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Mon, 3 Oct 2011 09:38:32 +0800 Subject: ASoC: txx9: Add __exit_p at necessary place We have __exit annotation for txx9aclc_generic_remove(), thus add __devexit_p to wrap it. Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/txx9/txx9aclc-generic.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/txx9/txx9aclc-generic.c b/sound/soc/txx9/txx9aclc-generic.c index 6770e71..9b5e283 100644 --- a/sound/soc/txx9/txx9aclc-generic.c +++ b/sound/soc/txx9/txx9aclc-generic.c @@ -62,7 +62,7 @@ static int __exit txx9aclc_generic_remove(struct platform_device *pdev) } static struct platform_driver txx9aclc_generic_driver = { - .remove = txx9aclc_generic_remove, + .remove = __exit_p(txx9aclc_generic_remove), .driver = { .name = "txx9aclc-generic", .owner = THIS_MODULE, -- cgit v1.1 From 9b5999b1bc74aea2a515701f937d7ad3e17f10e4 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Mon, 3 Oct 2011 11:09:01 +0800 Subject: ASoC: Fix setting update bits for WM8741_DACRMSB_ATTENUATION After checking the code and datasheet, I think what we want in the second snd_soc_update_bits call is to update WM8741_DACRMSB_ATTENUATION register instead of WM8741_DACRLSB_ATTENUATION. Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/wm8741.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c index 78c9e5a..a42b282 100644 --- a/sound/soc/codecs/wm8741.c +++ b/sound/soc/codecs/wm8741.c @@ -462,7 +462,7 @@ static int wm8741_probe(struct snd_soc_codec *codec) WM8741_UPDATELM, WM8741_UPDATELM); snd_soc_update_bits(codec, WM8741_DACRLSB_ATTENUATION, WM8741_UPDATERL, WM8741_UPDATERL); - snd_soc_update_bits(codec, WM8741_DACRLSB_ATTENUATION, + snd_soc_update_bits(codec, WM8741_DACRMSB_ATTENUATION, WM8741_UPDATERM, WM8741_UPDATERM); snd_soc_add_controls(codec, wm8741_snd_controls, -- cgit v1.1 From 61e49bf1941050e85035e139c6f9eace60c72e70 Mon Sep 17 00:00:00 2001 From: Arnd Bergmann Date: Mon, 3 Oct 2011 16:35:46 +0200 Subject: ASoC: samsung: WM8994 depends on MFD_WM8994 Any driver that selects SND_SOC_WM8994 should also make sure that MFD_WM8994 is set, since the codec relies on the mfd code: sound/built-in.o: In function `wm8994_read': last.c:(.text+0x20160): undefined reference to `wm8994_reg_read' sound/built-in.o: In function `wm8994_write': last.c:(.text+0x20e68): undefined reference to `wm8994_reg_write' This solves the problem by selecting the MFD driver directly and adding extra 'depends on' statements to make sure that we respect the dependencies of that driver. Signed-off-by: Arnd Bergmann Signed-off-by: Mark Brown --- sound/soc/samsung/Kconfig | 6 ++++++ 1 file changed, 6 insertions(+) (limited to 'sound') diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index dd3b3ea..53aaa69 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -64,6 +64,8 @@ config SND_SOC_SAMSUNG_SMDK_WM8580 config SND_SOC_SAMSUNG_SMDK_WM8994 tristate "SoC I2S Audio support for WM8994 on SMDK" depends on SND_SOC_SAMSUNG && (MACH_SMDKV310 || MACH_SMDKC210 || MACH_SMDK4212) + depends on I2C=y && GENERIC_HARDIRQS + select MFD_WM8994 select SND_SOC_WM8994 select SND_SAMSUNG_I2S help @@ -150,7 +152,9 @@ config SND_SOC_SMARTQ config SND_SOC_GONI_AQUILA_WM8994 tristate "SoC I2S Audio support for AQUILA/GONI - WM8994" depends on SND_SOC_SAMSUNG && (MACH_GONI || MACH_AQUILA) + depends on I2C=y && GENERIC_HARDIRQS select SND_SAMSUNG_I2S + select MFD_WM8994 select SND_SOC_WM8994 help Say Y if you want to add support for SoC audio on goni or aquila @@ -174,6 +178,8 @@ config SND_SOC_SMDK_WM8580_PCM config SND_SOC_SMDK_WM8994_PCM tristate "SoC PCM Audio support for WM8994 on SMDK" depends on SND_SOC_SAMSUNG && (MACH_SMDKC210 || MACH_SMDKV310 || MACH_SMDK4212) + depends on I2C=y && GENERIC_HARDIRQS + select MFD_WM8994 select SND_SOC_WM8994 select SND_SAMSUNG_PCM help -- cgit v1.1 From f3a54a283b58b932cd3d018ca4a1ba96cc43a895 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Sun, 2 Oct 2011 21:34:26 +0800 Subject: ASoC: samsung: s3c-i2s-v2.c needs module.h Include to fix below build error: CC sound/soc/samsung/s3c-i2s-v2.o sound/soc/samsung/s3c-i2s-v2.c:573: warning: data definition has no type or storage class sound/soc/samsung/s3c-i2s-v2.c:573: warning: type defaults to 'int' in declaration of 'EXPORT_SYMBOL_GPL' sound/soc/samsung/s3c-i2s-v2.c:573: warning: parameter names (without types) in function declaration sound/soc/samsung/s3c-i2s-v2.c:638: warning: data definition has no type or storage class sound/soc/samsung/s3c-i2s-v2.c:638: warning: type defaults to 'int' in declaration of 'EXPORT_SYMBOL_GPL' sound/soc/samsung/s3c-i2s-v2.c:638: warning: parameter names (without types) in function declaration sound/soc/samsung/s3c-i2s-v2.c:677: warning: data definition has no type or storage class sound/soc/samsung/s3c-i2s-v2.c:677: warning: type defaults to 'int' in declaration of 'EXPORT_SYMBOL_GPL' sound/soc/samsung/s3c-i2s-v2.c:677: warning: parameter names (without types) in function declaration sound/soc/samsung/s3c-i2s-v2.c: In function 's3c_i2sv2_register_dai': sound/soc/samsung/s3c-i2s-v2.c:736: warning: initialization discards qualifiers from pointer target type sound/soc/samsung/s3c-i2s-v2.c: At top level: sound/soc/samsung/s3c-i2s-v2.c:754: warning: data definition has no type or storage class sound/soc/samsung/s3c-i2s-v2.c:754: warning: type defaults to 'int' in declaration of 'EXPORT_SYMBOL_GPL' sound/soc/samsung/s3c-i2s-v2.c:754: warning: parameter names (without types) in function declaration sound/soc/samsung/s3c-i2s-v2.c:756: error: expected declaration specifiers or '...' before string constant sound/soc/samsung/s3c-i2s-v2.c:756: warning: data definition has no type or storage class sound/soc/samsung/s3c-i2s-v2.c:756: warning: type defaults to 'int' in declaration of 'MODULE_LICENSE' sound/soc/samsung/s3c-i2s-v2.c:756: warning: function declaration isn't a prototype make[3]: *** [sound/soc/samsung/s3c-i2s-v2.o] Error 1 make[2]: *** [sound/soc/samsung] Error 2 make[1]: *** [sound/soc] Error 2 make: *** [sound] Error 2 Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/samsung/s3c-i2s-v2.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/samsung/s3c-i2s-v2.c b/sound/soc/samsung/s3c-i2s-v2.c index 52074a2b..7a73380 100644 --- a/sound/soc/samsung/s3c-i2s-v2.c +++ b/sound/soc/samsung/s3c-i2s-v2.c @@ -16,6 +16,7 @@ * option) any later version. */ +#include #include #include #include -- cgit v1.1 From 0b07ab9244d34929b6e07d6a275137a229e9c839 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 28 Sep 2011 20:12:01 +0100 Subject: ASoC: Instantiate DAPM widgets before we do the DAI link init The DAI init function may want to do something that needs the widgets to be instantiated. Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index a58c1fc..1ed8093 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1059,6 +1059,9 @@ static int soc_post_component_init(struct snd_soc_card *card, temp = codec->name_prefix; codec->name_prefix = NULL; + /* Make sure all DAPM widgets are instantiated */ + snd_soc_dapm_new_widgets(&codec->dapm); + /* do machine specific initialization */ if (!dailess && dai_link->init) ret = dai_link->init(rtd); @@ -1070,9 +1073,6 @@ static int soc_post_component_init(struct snd_soc_card *card, } codec->name_prefix = temp; - /* Make sure all DAPM widgets are instantiated */ - snd_soc_dapm_new_widgets(&codec->dapm); - /* register the rtd device */ rtd->codec = codec; rtd->dev.parent = card->dev; -- cgit v1.1 From 11c2b5f2dc7ce42ddb779e1979d9defb02b70762 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 3 Oct 2011 21:07:06 +0100 Subject: ASoC: Fix typo in 24.576MHz rate in WM5100 Signed-off-by: Mark Brown --- sound/soc/codecs/wm5100.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index 46afdf8..8d90ba9 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -1755,7 +1755,7 @@ static int wm5100_set_sysclk(struct snd_soc_codec *codec, int clk_id, fval = 1; break; case 22579200: - case 2457600: + case 24576000: fval = 2; break; default: @@ -1772,7 +1772,7 @@ static int wm5100_set_sysclk(struct snd_soc_codec *codec, int clk_id, case 6144000: case 12288000: - case 2457600: + case 24576000: audio_rate = 48000; break; -- cgit v1.1 From 81204c84ca46604a04ab3d43ccfa1e464e6b1303 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 24 May 2011 17:35:53 +0800 Subject: ASoC: Add WM1811 support The WM1811 is mostly register compatible with the WM8994 and WM8958, providing a high performance audio hub CODEC in a small form factor suitable for ultra compact system designs. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 83 ++++++++++++++++++++++++++++++++++++++++++++--- 1 file changed, 78 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index e537267..5e8d66d 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -283,6 +283,7 @@ static const DECLARE_TLV_DB_SCALE(st_tlv, -3600, 300, 0); static const DECLARE_TLV_DB_SCALE(wm8994_3d_tlv, -1600, 183, 0); static const DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0); static const DECLARE_TLV_DB_SCALE(ng_tlv, -10200, 600, 0); +static const DECLARE_TLV_DB_SCALE(mixin_boost_tlv, 0, 900, 0); #define WM8994_DRC_SWITCH(xname, reg, shift) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ @@ -703,6 +704,13 @@ SOC_SINGLE_TLV("AIF2DAC Noise Gate Threshold Volume", 7, 1, ng_tlv), }; +static const struct snd_kcontrol_new wm1811_snd_controls[] = { +SOC_SINGLE_TLV("MIXINL IN1LP Boost Volume", WM8994_INPUT_MIXER_1, 7, 1, 0, + mixin_boost_tlv), +SOC_SINGLE_TLV("MIXINL IN1RP Boost Volume", WM8994_INPUT_MIXER_1, 8, 1, 0, + mixin_boost_tlv), +}; + static int clk_sys_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -2053,6 +2061,15 @@ static int wm8994_set_bias_level(struct snd_soc_codec *codec, WM8958_CP_DISCH); } break; + + case WM1811: + if (wm8994->revision < 2) { + snd_soc_write(codec, 0x102, 0x3); + snd_soc_write(codec, 0x5d, 0x7e); + snd_soc_write(codec, 0x5e, 0x0); + snd_soc_write(codec, 0x102, 0x0); + } + break; } /* Discharge LINEOUT1 & 2 */ @@ -2168,10 +2185,18 @@ static int wm8994_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) /* The AIF2 format configuration needs to be mirrored to AIF3 * on WM8958 if it's in use so just do it all the time. */ - if (control->type == WM8958 && dai->id == 2) - snd_soc_update_bits(codec, WM8958_AIF3_CONTROL_1, - WM8994_AIF1_LRCLK_INV | - WM8958_AIF3_FMT_MASK, aif1); + switch (control->type) { + case WM1811: + case WM8958: + if (dai->id == 2) + snd_soc_update_bits(codec, WM8958_AIF3_CONTROL_1, + WM8994_AIF1_LRCLK_INV | + WM8958_AIF3_FMT_MASK, aif1); + break; + + default: + break; + } snd_soc_update_bits(codec, aif1_reg, WM8994_AIF1_BCLK_INV | WM8994_AIF1_LRCLK_INV | @@ -2258,6 +2283,7 @@ static int wm8994_hw_params(struct snd_pcm_substream *substream, break; case 3: switch (control->type) { + case WM1811: case WM8958: aif1_reg = WM8958_AIF3_CONTROL_1; break; @@ -2384,6 +2410,7 @@ static int wm8994_aif3_hw_params(struct snd_pcm_substream *substream, switch (dai->id) { case 3: switch (control->type) { + case WM1811: case WM8958: aif1_reg = WM8958_AIF3_CONTROL_1; break; @@ -2614,6 +2641,7 @@ static int wm8994_suspend(struct snd_soc_codec *codec, pm_message_t state) case WM8994: snd_soc_update_bits(codec, WM8994_MICBIAS, WM8994_MICD_ENA, 0); break; + case WM1811: case WM8958: snd_soc_update_bits(codec, WM8958_MIC_DETECT_1, WM8958_MICD_ENA, 0); @@ -2682,6 +2710,7 @@ static int wm8994_resume(struct snd_soc_codec *codec) snd_soc_update_bits(codec, WM8994_MICBIAS, WM8994_MICD_ENA, WM8994_MICD_ENA); break; + case WM1811: case WM8958: if (wm8994->jack_cb) snd_soc_update_bits(codec, WM8958_MIC_DETECT_1, @@ -2980,8 +3009,13 @@ int wm8958_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack, struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); struct wm8994 *control = codec->control_data; - if (control->type != WM8958) + switch (control->type) { + case WM1811: + case WM8958: + break; + default: return -EINVAL; + } if (jack) { if (!cb) { @@ -3135,6 +3169,24 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) wm8994->hubs.dcs_readback_mode = 1; break; + case WM1811: + wm8994->hubs.dcs_readback_mode = 2; + wm8994->hubs.no_series_update = 1; + + switch (wm8994->revision) { + case 0: + case 1: + wm8994->hubs.dcs_codes_l = -7; + wm8994->hubs.dcs_codes_r = -4; + break; + default: + break; + } + + snd_soc_update_bits(codec, WM8994_ANALOGUE_HP_1, + WM1811_HPOUT1_ATTN, WM1811_HPOUT1_ATTN); + break; + default: break; } @@ -3195,6 +3247,7 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) break; case WM8958: + case WM1811: if (wm8994->micdet_irq) { ret = request_threaded_irq(wm8994->micdet_irq, NULL, wm8958_mic_irq, @@ -3357,6 +3410,19 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) ARRAY_SIZE(wm8994_dac_widgets)); } break; + + case WM1811: + snd_soc_add_controls(codec, wm8958_snd_controls, + ARRAY_SIZE(wm8958_snd_controls)); + snd_soc_dapm_new_controls(dapm, wm8958_dapm_widgets, + ARRAY_SIZE(wm8958_dapm_widgets)); + snd_soc_dapm_new_controls(dapm, wm8994_lateclk_widgets, + ARRAY_SIZE(wm8994_lateclk_widgets)); + snd_soc_dapm_new_controls(dapm, wm8994_adc_widgets, + ARRAY_SIZE(wm8994_adc_widgets)); + snd_soc_dapm_new_controls(dapm, wm8994_dac_widgets, + ARRAY_SIZE(wm8994_dac_widgets)); + break; } @@ -3393,6 +3459,12 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) wm8958_dsp2_init(codec); break; + case WM1811: + snd_soc_dapm_add_routes(dapm, wm8994_lateclk_intercon, + ARRAY_SIZE(wm8994_lateclk_intercon)); + snd_soc_dapm_add_routes(dapm, wm8958_intercon, + ARRAY_SIZE(wm8958_intercon)); + break; } return 0; @@ -3448,6 +3520,7 @@ static int wm8994_codec_remove(struct snd_soc_codec *codec) wm8994); break; + case WM1811: case WM8958: if (wm8994->micdet_irq) free_irq(wm8994->micdet_irq, wm8994); -- cgit v1.1 From 8754f2263fb0961c6dd26b4b4cbe73a4e632aa62 Mon Sep 17 00:00:00 2001 From: Ryan Mallon Date: Tue, 4 Oct 2011 09:55:40 +1100 Subject: ASoC: max98088 codec: Catch driver bugs for eq channel name Move the EQ channel names to a separate array and iterate over it in max98088_get_channel rather than duplicating the hardcoded channel names. Add an error message if an invalid channel is passed and check the error in the callers. Also added a BUILD_BUG_ON to ensure that the eq_mode_name and controls arrays are the same size. Signed-off-by: Ryan Mallon Signed-off-by: Mark Brown --- sound/soc/codecs/max98088.c | 34 +++++++++++++++++++++++----------- 1 file changed, 23 insertions(+), 11 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index 587043b..ebbf63c 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -1696,13 +1696,19 @@ static struct snd_soc_dai_driver max98088_dai[] = { } }; -static int max98088_get_channel(const char *name) +static const char *eq_mode_name[] = {"EQ1 Mode", "EQ2 Mode"}; + +static int max98088_get_channel(struct snd_soc_codec *codec, const char *name) { - if (strcmp(name, "EQ1 Mode") == 0) - return 0; - if (strcmp(name, "EQ2 Mode") == 0) - return 1; - return -EINVAL; + int i; + + for (i = 0; i < ARRAY_SIZE(eq_mode_name); i++) + if (strcmp(name, eq_mode_name[i]) == 0) + return i; + + /* Shouldn't happen */ + dev_err(codec->dev, "Bad EQ channel name '%s'\n", name); + return -EINVAL; } static void max98088_setup_eq1(struct snd_soc_codec *codec) @@ -1806,10 +1812,13 @@ static int max98088_put_eq_enum(struct snd_kcontrol *kcontrol, struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); struct max98088_priv *max98088 = snd_soc_codec_get_drvdata(codec); struct max98088_pdata *pdata = max98088->pdata; - int channel = max98088_get_channel(kcontrol->id.name); + int channel = max98088_get_channel(codec, kcontrol->id.name); struct max98088_cdata *cdata; int sel = ucontrol->value.integer.value[0]; + if (channel < 0) + return channel; + cdata = &max98088->dai[channel]; if (sel >= pdata->eq_cfgcnt) @@ -1834,9 +1843,12 @@ static int max98088_get_eq_enum(struct snd_kcontrol *kcontrol, { struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); struct max98088_priv *max98088 = snd_soc_codec_get_drvdata(codec); - int channel = max98088_get_channel(kcontrol->id.name); + int channel = max98088_get_channel(codec, kcontrol->id.name); struct max98088_cdata *cdata; + if (channel < 0) + return channel; + cdata = &max98088->dai[channel]; ucontrol->value.enumerated.item[0] = cdata->eq_sel; return 0; @@ -1851,17 +1863,17 @@ static void max98088_handle_eq_pdata(struct snd_soc_codec *codec) int i, j; const char **t; int ret; - struct snd_kcontrol_new controls[] = { - SOC_ENUM_EXT("EQ1 Mode", + SOC_ENUM_EXT((char *)eq_mode_name[0], max98088->eq_enum, max98088_get_eq_enum, max98088_put_eq_enum), - SOC_ENUM_EXT("EQ2 Mode", + SOC_ENUM_EXT((char *)eq_mode_name[1], max98088->eq_enum, max98088_get_eq_enum, max98088_put_eq_enum), }; + BUILD_BUG_ON(ARRAY_SIZE(controls) != ARRAY_SIZE(eq_mode_name)); cfg = pdata->eq_cfg; cfgcnt = pdata->eq_cfgcnt; -- cgit v1.1 From c855a1a7ff49a43e1e35571d504e89b4c670693d Mon Sep 17 00:00:00 2001 From: Ryan Mallon Date: Tue, 4 Oct 2011 09:55:41 +1100 Subject: ASoC: max98095 codec: Catch driver bugs for biquad channel name Move the biquad channel names to a separate array and iterate over it in max98095_get_bq_channel rather than duplicating the hardcoded channel names. Add an error message if an invalid channel is passed and check the error in the callers. Also added a BUILD_BUG_ON to ensure that the bq_mode_name and controls arrays are the same size. Signed-off-by: Ryan Mallon Signed-off-by: Mark Brown --- sound/soc/codecs/max98095.c | 32 ++++++++++++++++++++++---------- 1 file changed, 22 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/max98095.c b/sound/soc/codecs/max98095.c index 8f8e255..6982f74 100644 --- a/sound/soc/codecs/max98095.c +++ b/sound/soc/codecs/max98095.c @@ -1991,12 +1991,19 @@ static void max98095_handle_eq_pdata(struct snd_soc_codec *codec) dev_err(codec->dev, "Failed to add EQ control: %d\n", ret); } -static int max98095_get_bq_channel(const char *name) +static const char *bq_mode_name[] = {"Biquad1 Mode", "Biquad2 Mode"}; + +static int max98095_get_bq_channel(struct snd_soc_codec *codec, + const char *name) { - if (strcmp(name, "Biquad1 Mode") == 0) - return 0; - if (strcmp(name, "Biquad2 Mode") == 0) - return 1; + int i; + + for (i = 0; i < ARRAY_SIZE(bq_mode_name); i++) + if (strcmp(name, bq_mode_name[i]) == 0) + return i; + + /* Shouldn't happen */ + dev_err(codec->dev, "Bad biquad channel name '%s'\n", name); return -EINVAL; } @@ -2006,14 +2013,15 @@ static int max98095_put_bq_enum(struct snd_kcontrol *kcontrol, struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); struct max98095_priv *max98095 = snd_soc_codec_get_drvdata(codec); struct max98095_pdata *pdata = max98095->pdata; - int channel = max98095_get_bq_channel(kcontrol->id.name); + int channel = max98095_get_bq_channel(codec, kcontrol->id.name); struct max98095_cdata *cdata; int sel = ucontrol->value.integer.value[0]; struct max98095_biquad_cfg *coef_set; int fs, best, best_val, i; int regmask, regsave; - BUG_ON(channel > 1); + if (channel < 0) + return channel; if (!pdata || !max98095->bq_textcnt) return 0; @@ -2065,9 +2073,12 @@ static int max98095_get_bq_enum(struct snd_kcontrol *kcontrol, { struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); struct max98095_priv *max98095 = snd_soc_codec_get_drvdata(codec); - int channel = max98095_get_bq_channel(kcontrol->id.name); + int channel = max98095_get_bq_channel(codec, kcontrol->id.name); struct max98095_cdata *cdata; + if (channel < 0) + return channel; + cdata = &max98095->dai[channel]; ucontrol->value.enumerated.item[0] = cdata->bq_sel; @@ -2085,15 +2096,16 @@ static void max98095_handle_bq_pdata(struct snd_soc_codec *codec) int ret; struct snd_kcontrol_new controls[] = { - SOC_ENUM_EXT("Biquad1 Mode", + SOC_ENUM_EXT((char *)bq_mode_name[0], max98095->bq_enum, max98095_get_bq_enum, max98095_put_bq_enum), - SOC_ENUM_EXT("Biquad2 Mode", + SOC_ENUM_EXT((char *)bq_mode_name[1], max98095->bq_enum, max98095_get_bq_enum, max98095_put_bq_enum), }; + BUILD_BUG_ON(ARRAY_SIZE(controls) != ARRAY_SIZE(bq_mode_name)); cfg = pdata->bq_cfg; cfgcnt = pdata->bq_cfgcnt; -- cgit v1.1 From 11b9ce622a8c29740707e5fbb54cddf8d7892398 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Tue, 4 Oct 2011 09:55:45 +0800 Subject: ASoC: wm8711: Use snd_soc_update_bits for read-modify-write Use snd_soc_update_bits for read-modify-write register access instead of open-coding it using snd_soc_read and snd_soc_write Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/wm8711.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8711.c b/sound/soc/codecs/wm8711.c index 8457d3c..6ecf1ab 100644 --- a/sound/soc/codecs/wm8711.c +++ b/sound/soc/codecs/wm8711.c @@ -381,10 +381,8 @@ static int wm8711_probe(struct snd_soc_codec *codec) wm8711_set_bias_level(codec, SND_SOC_BIAS_STANDBY); /* Latch the update bits */ - reg = snd_soc_read(codec, WM8711_LOUT1V); - snd_soc_write(codec, WM8711_LOUT1V, reg | 0x0100); - reg = snd_soc_read(codec, WM8711_ROUT1V); - snd_soc_write(codec, WM8711_ROUT1V, reg | 0x0100); + snd_soc_update_bits(codec, WM8711_LOUT1V, 0x0100, 0x0100); + snd_soc_update_bits(codec, WM8711_ROUT1V, 0x0100, 0x0100); snd_soc_add_controls(codec, wm8711_snd_controls, ARRAY_SIZE(wm8711_snd_controls)); -- cgit v1.1 From c6d43417dd0ff6d4431f6f52e6eac1f48ff779b2 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Tue, 4 Oct 2011 09:58:28 +0800 Subject: ASoC: wm8971: Use snd_soc_update_bits for read-modify-write Use snd_soc_update_bits for read-modify-write register access instead of open-coding it using snd_soc_read and snd_soc_write Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/wm8971.c | 27 ++++++++------------------- 1 file changed, 8 insertions(+), 19 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c index 572bb80..ce33a94 100644 --- a/sound/soc/codecs/wm8971.c +++ b/sound/soc/codecs/wm8971.c @@ -660,25 +660,14 @@ static int wm8971_probe(struct snd_soc_codec *codec) msecs_to_jiffies(1000)); /* set the update bits */ - reg = snd_soc_read(codec, WM8971_LDAC); - snd_soc_write(codec, WM8971_LDAC, reg | 0x0100); - reg = snd_soc_read(codec, WM8971_RDAC); - snd_soc_write(codec, WM8971_RDAC, reg | 0x0100); - - reg = snd_soc_read(codec, WM8971_LOUT1V); - snd_soc_write(codec, WM8971_LOUT1V, reg | 0x0100); - reg = snd_soc_read(codec, WM8971_ROUT1V); - snd_soc_write(codec, WM8971_ROUT1V, reg | 0x0100); - - reg = snd_soc_read(codec, WM8971_LOUT2V); - snd_soc_write(codec, WM8971_LOUT2V, reg | 0x0100); - reg = snd_soc_read(codec, WM8971_ROUT2V); - snd_soc_write(codec, WM8971_ROUT2V, reg | 0x0100); - - reg = snd_soc_read(codec, WM8971_LINVOL); - snd_soc_write(codec, WM8971_LINVOL, reg | 0x0100); - reg = snd_soc_read(codec, WM8971_RINVOL); - snd_soc_write(codec, WM8971_RINVOL, reg | 0x0100); + snd_soc_update_bits(codec, WM8971_LDAC, 0x0100, 0x0100); + snd_soc_update_bits(codec, WM8971_RDAC, 0x0100, 0x0100); + snd_soc_update_bits(codec, WM8971_LOUT1V, 0x0100, 0x0100); + snd_soc_update_bits(codec, WM8971_ROUT1V, 0x0100, 0x0100); + snd_soc_update_bits(codec, WM8971_LOUT2V, 0x0100, 0x0100); + snd_soc_update_bits(codec, WM8971_ROUT2V, 0x0100, 0x0100); + snd_soc_update_bits(codec, WM8971_LINVOL, 0x0100, 0x0100); + snd_soc_update_bits(codec, WM8971_RINVOL, 0x0100, 0x0100); snd_soc_add_controls(codec, wm8971_snd_controls, ARRAY_SIZE(wm8971_snd_controls)); -- cgit v1.1 From 9cd113261b7d38b96e6edac688790bb965e23566 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Tue, 4 Oct 2011 09:59:26 +0800 Subject: ASoC: wm8988: Use snd_soc_update_bits for read-modify-write Use snd_soc_update_bits for read-modify-write register access instead of open-coding it using snd_soc_read and snd_soc_write Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/wm8988.c | 15 +++++---------- 1 file changed, 5 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8988.c b/sound/soc/codecs/wm8988.c index d7170f1..5099e62 100644 --- a/sound/soc/codecs/wm8988.c +++ b/sound/soc/codecs/wm8988.c @@ -774,16 +774,11 @@ static int wm8988_probe(struct snd_soc_codec *codec) } /* set the update bits (we always update left then right) */ - reg = snd_soc_read(codec, WM8988_RADC); - snd_soc_write(codec, WM8988_RADC, reg | 0x100); - reg = snd_soc_read(codec, WM8988_RDAC); - snd_soc_write(codec, WM8988_RDAC, reg | 0x0100); - reg = snd_soc_read(codec, WM8988_ROUT1V); - snd_soc_write(codec, WM8988_ROUT1V, reg | 0x0100); - reg = snd_soc_read(codec, WM8988_ROUT2V); - snd_soc_write(codec, WM8988_ROUT2V, reg | 0x0100); - reg = snd_soc_read(codec, WM8988_RINVOL); - snd_soc_write(codec, WM8988_RINVOL, reg | 0x0100); + snd_soc_update_bits(codec, WM8988_RADC, 0x0100, 0x0100); + snd_soc_update_bits(codec, WM8988_RDAC, 0x0100, 0x0100); + snd_soc_update_bits(codec, WM8988_ROUT1V, 0x0100, 0x0100); + snd_soc_update_bits(codec, WM8988_ROUT2V, 0x0100, 0x0100); + snd_soc_update_bits(codec, WM8988_RINVOL, 0x0100, 0x0100); wm8988_set_bias_level(codec, SND_SOC_BIAS_STANDBY); -- cgit v1.1 From b1b6cffeb71d363693d15444c79d828153a70865 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 4 Oct 2011 11:09:52 +0300 Subject: ASoC: omap-pcm: Fix the no period wakeup implementation After omap_request_dma the BLOCK_IRQ is enabled as default configuration for the channel. If we are requested for no period wakeup, we need to disable the BLOCK_IRQ in order to not receive any interrupts. Signed-off-by: Peter Ujfalusi Acked-by: Jarkko Nikula Signed-off-by: Mark Brown --- sound/soc/omap/omap-pcm.c | 8 ++++++++ 1 file changed, 8 insertions(+) (limited to 'sound') diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index 9b5c88a..5e37ec9 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -198,6 +198,14 @@ static int omap_pcm_prepare(struct snd_pcm_substream *substream) OMAP_DMA_LAST_IRQ | OMAP_DMA_BLOCK_IRQ); else if (!substream->runtime->no_period_wakeup) omap_enable_dma_irq(prtd->dma_ch, OMAP_DMA_FRAME_IRQ); + else { + /* + * No period wakeup: + * we need to disable BLOCK_IRQ, which is enabled by the omap + * dma core at request dma time. + */ + omap_disable_dma_irq(prtd->dma_ch, OMAP_DMA_BLOCK_IRQ); + } if (!(cpu_class_is_omap1())) { omap_set_dma_src_burst_mode(prtd->dma_ch, -- cgit v1.1 From 04f45c493ac6de7c3d1864c3193c225424c25b7d Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Tue, 4 Oct 2011 20:07:03 +0800 Subject: ASoC: wm8994: Slightly optimize configure_clock snd_soc_update_bits() will only write new register value if the old value is different from the new value. In additional, snd_soc_update_bits() returns 0 for no change. No need to read WM8994_CLOCKING_1 register before calling snd_soc_update_bits(). Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 11 ++++------- 1 file changed, 4 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 5e8d66d..546173f 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -208,7 +208,7 @@ static int configure_aif_clock(struct snd_soc_codec *codec, int aif) static int configure_clock(struct snd_soc_codec *codec) { struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); - int old, new; + int change, new; /* Bring up the AIF clocks first */ configure_aif_clock(codec, 0); @@ -229,14 +229,11 @@ static int configure_clock(struct snd_soc_codec *codec) else new = 0; - old = snd_soc_read(codec, WM8994_CLOCKING_1) & WM8994_SYSCLK_SRC; - - /* If there's no change then we're done. */ - if (old == new) + change = snd_soc_update_bits(codec, WM8994_CLOCKING_1, + WM8994_SYSCLK_SRC, new); + if (!change) return 0; - snd_soc_update_bits(codec, WM8994_CLOCKING_1, WM8994_SYSCLK_SRC, new); - snd_soc_dapm_sync(&codec->dapm); return 0; -- cgit v1.1 From 734787550a60b768b675c26f93d134f6dc370bb5 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Tue, 4 Oct 2011 20:08:04 +0800 Subject: ASoC: wm8995: Slightly optimize configure_clock snd_soc_update_bits() will only write new register value if the old value is different from the new value. In additional, snd_soc_update_bits() returns 0 for no change. No need to read WM8995_CLOCKING_1 register before calling snd_soc_update_bits(). Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/wm8995.c | 12 ++++-------- 1 file changed, 4 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8995.c b/sound/soc/codecs/wm8995.c index e05ee79..78eeb21 100644 --- a/sound/soc/codecs/wm8995.c +++ b/sound/soc/codecs/wm8995.c @@ -485,7 +485,7 @@ static int configure_aif_clock(struct snd_soc_codec *codec, int aif) static int configure_clock(struct snd_soc_codec *codec) { struct wm8995_priv *wm8995; - int old, new; + int change, new; wm8995 = snd_soc_codec_get_drvdata(codec); @@ -509,15 +509,11 @@ static int configure_clock(struct snd_soc_codec *codec) else new = 0; - old = snd_soc_read(codec, WM8995_CLOCKING_1) & WM8995_SYSCLK_SRC; - - /* If there's no change then we're done. */ - if (old == new) + change = snd_soc_update_bits(codec, WM8995_CLOCKING_1, + WM8995_SYSCLK_SRC_MASK, new); + if (!change) return 0; - snd_soc_update_bits(codec, WM8995_CLOCKING_1, - WM8995_SYSCLK_SRC_MASK, new); - snd_soc_dapm_sync(&codec->dapm); return 0; -- cgit v1.1 From c527e6aadc8f142ad388b6aa59a1ce6a4bfb1966 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Tue, 4 Oct 2011 22:07:18 +0800 Subject: ASoC: wm8994: Fix setting rate_reg for wm8994-aif2 For wm8994-aif2, the rate_reg should be WM8994_AIF2_RATE. Signed-off-by: Axel Lin Signed-off-by: Mark Brown Cc: stable@kernel.org --- sound/soc/codecs/wm8994.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index b393f9f..48ea611 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2311,7 +2311,7 @@ static void wm8994_aif_shutdown(struct snd_pcm_substream *substream, rate_reg = WM8994_AIF1_RATE; break; case 2: - rate_reg = WM8994_AIF1_RATE; + rate_reg = WM8994_AIF2_RATE; break; default: break; -- cgit v1.1 From 1a3bbb40da5c01e422309f52475e91886c573718 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Tue, 4 Oct 2011 07:44:22 +0800 Subject: ASoC: Avoid writing to WM8971_RESET in wm8971_resume Writing to WM8971_RESET resets all registers to the default state. Thus we should avoid writing to WM8971_RESET on resume. Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/wm8971.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c index ce33a94..08ea6f8 100644 --- a/sound/soc/codecs/wm8971.c +++ b/sound/soc/codecs/wm8971.c @@ -612,7 +612,7 @@ static int wm8971_resume(struct snd_soc_codec *codec) /* Sync reg_cache with the hardware */ for (i = 0; i < ARRAY_SIZE(wm8971_reg); i++) { - if (i + 1 == WM8971_RESET) + if (i == WM8971_RESET) continue; data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001); data[1] = cache[i] & 0x00ff; -- cgit v1.1 From f5b00d024fb3308a42610d23f9b8d5d5d9fad8eb Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Tue, 4 Oct 2011 11:17:24 +0800 Subject: ASoC: wm8750: Use snd_soc_update_bits for read-modify-write Use snd_soc_update_bits for read-modify-write register access instead of open-coding it using snd_soc_read and snd_soc_write Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/wm8750.c | 26 +++++++++----------------- 1 file changed, 9 insertions(+), 17 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index 15f0372..862c520 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -694,7 +694,7 @@ static int wm8750_resume(struct snd_soc_codec *codec) static int wm8750_probe(struct snd_soc_codec *codec) { struct wm8750_priv *wm8750 = snd_soc_codec_get_drvdata(codec); - int reg, ret; + int ret; ret = snd_soc_codec_set_cache_io(codec, 7, 9, wm8750->control_type); if (ret < 0) { @@ -712,22 +712,14 @@ static int wm8750_probe(struct snd_soc_codec *codec) wm8750_set_bias_level(codec, SND_SOC_BIAS_STANDBY); /* set the update bits */ - reg = snd_soc_read(codec, WM8750_LDAC); - snd_soc_write(codec, WM8750_LDAC, reg | 0x0100); - reg = snd_soc_read(codec, WM8750_RDAC); - snd_soc_write(codec, WM8750_RDAC, reg | 0x0100); - reg = snd_soc_read(codec, WM8750_LOUT1V); - snd_soc_write(codec, WM8750_LOUT1V, reg | 0x0100); - reg = snd_soc_read(codec, WM8750_ROUT1V); - snd_soc_write(codec, WM8750_ROUT1V, reg | 0x0100); - reg = snd_soc_read(codec, WM8750_LOUT2V); - snd_soc_write(codec, WM8750_LOUT2V, reg | 0x0100); - reg = snd_soc_read(codec, WM8750_ROUT2V); - snd_soc_write(codec, WM8750_ROUT2V, reg | 0x0100); - reg = snd_soc_read(codec, WM8750_LINVOL); - snd_soc_write(codec, WM8750_LINVOL, reg | 0x0100); - reg = snd_soc_read(codec, WM8750_RINVOL); - snd_soc_write(codec, WM8750_RINVOL, reg | 0x0100); + snd_soc_update_bits(codec, WM8750_LDAC, 0x0100, 0x0100); + snd_soc_update_bits(codec, WM8750_RDAC, 0x0100, 0x0100); + snd_soc_update_bits(codec, WM8750_LOUT1V, 0x0100, 0x0100); + snd_soc_update_bits(codec, WM8750_ROUT1V, 0x0100, 0x0100); + snd_soc_update_bits(codec, WM8750_LOUT2V, 0x0100, 0x0100); + snd_soc_update_bits(codec, WM8750_ROUT2V, 0x0100, 0x0100); + snd_soc_update_bits(codec, WM8750_LINVOL, 0x0100, 0x0100); + snd_soc_update_bits(codec, WM8750_RINVOL, 0x0100, 0x0100); snd_soc_add_controls(codec, wm8750_snd_controls, ARRAY_SIZE(wm8750_snd_controls)); -- cgit v1.1 From 696f9175fcb560aa45e99e6a1833f6f30d1f97da Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Tue, 4 Oct 2011 11:18:59 +0800 Subject: ASoC: wm8988: Use snd_soc_update_bits for read-modify-write Use snd_soc_update_bits for read-modify-write register access instead of open-coding it using snd_soc_read and snd_soc_write Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/wm8988.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8988.c b/sound/soc/codecs/wm8988.c index 5099e62..1c6f8bf 100644 --- a/sound/soc/codecs/wm8988.c +++ b/sound/soc/codecs/wm8988.c @@ -759,7 +759,6 @@ static int wm8988_probe(struct snd_soc_codec *codec) struct wm8988_priv *wm8988 = snd_soc_codec_get_drvdata(codec); struct snd_soc_dapm_context *dapm = &codec->dapm; int ret = 0; - u16 reg; ret = snd_soc_codec_set_cache_io(codec, 7, 9, wm8988->control_type); if (ret < 0) { -- cgit v1.1 From d434bc32d08435979514d437885cb9a7e216dd45 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Tue, 4 Oct 2011 11:15:30 +0800 Subject: ASoC: wm8711: Use snd_soc_update_bits for read-modify-write Use snd_soc_update_bits for read-modify-write register access instead of open-coding it using snd_soc_read and snd_soc_write Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/wm8711.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8711.c b/sound/soc/codecs/wm8711.c index 6ecf1ab..47c7fd5 100644 --- a/sound/soc/codecs/wm8711.c +++ b/sound/soc/codecs/wm8711.c @@ -364,7 +364,7 @@ static int wm8711_resume(struct snd_soc_codec *codec) static int wm8711_probe(struct snd_soc_codec *codec) { struct wm8711_priv *wm8711 = snd_soc_codec_get_drvdata(codec); - int ret, reg; + int ret; ret = snd_soc_codec_set_cache_io(codec, 7, 9, wm8711->bus_type); if (ret < 0) { -- cgit v1.1 From 499cb184e2a8923d71148a1f7c4d1813c1361f28 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Tue, 4 Oct 2011 11:44:42 +0800 Subject: ASoC: Remove unneeded hw_write initialisation in ak4671 It is not required now. Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/ak4671.c | 2 -- 1 file changed, 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/ak4671.c b/sound/soc/codecs/ak4671.c index 2ecf128..41e3d55 100644 --- a/sound/soc/codecs/ak4671.c +++ b/sound/soc/codecs/ak4671.c @@ -628,8 +628,6 @@ static int ak4671_probe(struct snd_soc_codec *codec) struct ak4671_priv *ak4671 = snd_soc_codec_get_drvdata(codec); int ret; - codec->hw_write = (hw_write_t)i2c_master_send; - ret = snd_soc_codec_set_cache_io(codec, 8, 8, ak4671->control_type); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); -- cgit v1.1 From 672f4c4d754273b4187e44f725ea418a97fa2a62 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Tue, 4 Oct 2011 11:45:41 +0800 Subject: ASoC: Remove unneeded hw_write initialisation in wm8523 It is not required after commit 8d50e447 "ASoC: Factor out I/O for Wolfson 8 bit data 16 bit register CODECs" Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/wm8523.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8523.c b/sound/soc/codecs/wm8523.c index 5355a7a..db7a681 100644 --- a/sound/soc/codecs/wm8523.c +++ b/sound/soc/codecs/wm8523.c @@ -406,7 +406,6 @@ static int wm8523_probe(struct snd_soc_codec *codec) struct wm8523_priv *wm8523 = snd_soc_codec_get_drvdata(codec); int ret, i; - codec->hw_write = (hw_write_t)i2c_master_send; wm8523->rate_constraint.list = &wm8523->rate_constraint_list[0]; wm8523->rate_constraint.count = ARRAY_SIZE(wm8523->rate_constraint_list); -- cgit v1.1 From bfd3d4e9fbb9705181b821b478cc044911e47320 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 4 Oct 2011 14:39:42 +0300 Subject: ASoC: twl6040: Simplify custom put_volsw callback Return -EINVAL in the unlikely event, if the function has been called for unhandled control. This way we can remove one check in the code. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl6040.c | 12 +++++------- 1 file changed, 5 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index d040905..8c740c1 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -759,15 +759,13 @@ static int twl6040_put_volsw(struct snd_kcontrol *kcontrol, out = &twl6040_priv->handsfree; break; default: - break; + return -EINVAL; } - if (out) { - out->left_vol = ucontrol->value.integer.value[0]; - out->right_vol = ucontrol->value.integer.value[1]; - if (!out->active) - return 1; - } + out->left_vol = ucontrol->value.integer.value[0]; + out->right_vol = ucontrol->value.integer.value[1]; + if (!out->active) + return 1; /* call the appropriate handler depending on the rreg */ if (mc->rreg) -- cgit v1.1 From 38f3f31a0a797bdbcc0cdb12553bbecc2f9a91c4 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 23 Sep 2011 21:26:33 +0100 Subject: ASoC: Remove direct register cache accesses from WM8962 driver Also fix return values for speaker switch updates. Signed-off-by: Mark Brown Cc: stable@kernel.org --- sound/soc/codecs/wm8962.c | 21 +++++++++++---------- 1 file changed, 11 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 74ebbfa..f60dfa1 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2139,7 +2139,6 @@ static int wm8962_put_spk_sw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - u16 *reg_cache = codec->reg_cache; int ret; /* Apply the update (if any) */ @@ -2148,16 +2147,19 @@ static int wm8962_put_spk_sw(struct snd_kcontrol *kcontrol, return 0; /* If the left PGA is enabled hit that VU bit... */ - if (reg_cache[WM8962_PWR_MGMT_2] & WM8962_SPKOUTL_PGA_ENA) - return snd_soc_write(codec, WM8962_SPKOUTL_VOLUME, - reg_cache[WM8962_SPKOUTL_VOLUME]); + ret = snd_soc_read(codec, WM8962_PWR_MGMT_2); + if (ret & WM8962_SPKOUTL_PGA_ENA) { + snd_soc_write(codec, WM8962_SPKOUTL_VOLUME, + snd_soc_read(codec, WM8962_SPKOUTL_VOLUME)); + return 1; + } /* ...otherwise the right. The VU is stereo. */ - if (reg_cache[WM8962_PWR_MGMT_2] & WM8962_SPKOUTR_PGA_ENA) - return snd_soc_write(codec, WM8962_SPKOUTR_VOLUME, - reg_cache[WM8962_SPKOUTR_VOLUME]); + if (ret & WM8962_SPKOUTR_PGA_ENA) + snd_soc_write(codec, WM8962_SPKOUTR_VOLUME, + snd_soc_read(codec, WM8962_SPKOUTR_VOLUME)); - return 0; + return 1; } static const char *cap_hpf_mode_text[] = { @@ -2498,7 +2500,6 @@ static int out_pga_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { struct snd_soc_codec *codec = w->codec; - u16 *reg_cache = codec->reg_cache; int reg; switch (w->shift) { @@ -2521,7 +2522,7 @@ static int out_pga_event(struct snd_soc_dapm_widget *w, switch (event) { case SND_SOC_DAPM_POST_PMU: - return snd_soc_write(codec, reg, reg_cache[reg]); + return snd_soc_write(codec, reg, snd_soc_read(codec, reg)); default: BUG(); return -EINVAL; -- cgit v1.1 From 05623c4314cba3971f8476151aff73126127925f Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 28 Sep 2011 17:02:31 +0100 Subject: ASoC: Factor write of widget power out into a separate function Split the decision about what the new power should be out from the implementation of that decision. Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 29 ++++++++++++++++++----------- 1 file changed, 18 insertions(+), 11 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index c277228..dcbd468 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1197,6 +1197,23 @@ static void dapm_post_sequence_async(void *data, async_cookie_t cookie) } } +static void dapm_widget_set_power(struct snd_soc_dapm_widget *w, bool power, + struct list_head *up_list, + struct list_head *down_list) +{ + if (w->power == power) + return; + + trace_snd_soc_dapm_widget_power(w, power); + + if (power) + dapm_seq_insert(w, up_list, true); + else + dapm_seq_insert(w, down_list, false); + + w->power = power; +} + static void dapm_power_one_widget(struct snd_soc_dapm_widget *w, struct list_head *up_list, struct list_head *down_list) @@ -1241,17 +1258,7 @@ static void dapm_power_one_widget(struct snd_soc_dapm_widget *w, } } - if (w->power == power) - break; - - trace_snd_soc_dapm_widget_power(w, power); - - if (power) - dapm_seq_insert(w, up_list, true); - else - dapm_seq_insert(w, down_list, false); - - w->power = power; + dapm_widget_set_power(w, power, up_list, down_list); break; } } -- cgit v1.1 From f9de6d741d246583a8fdcf212cf14456a1622ce1 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 28 Sep 2011 17:19:47 +0100 Subject: ASoC: Move bias level decision into main dapm_power_widgets() Future patches will try to reduce the number of widgets we check on each DAPM run but we're still going to need to look and see if the devices is on at all so we can manage the overall device bias. Move these checks out into the main dapm_power_widgets() function so we don't have to think about them for now. Once we're doing more incremental updates it'll probably be worth using refcounts for each bias level to avoid having to do the sweep over all widgets but that's not going to be where the big performance wins are. Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 44 +++++++++++++++++++++++--------------------- 1 file changed, 23 insertions(+), 21 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index dcbd468..12bd01a 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1218,7 +1218,6 @@ static void dapm_power_one_widget(struct snd_soc_dapm_widget *w, struct list_head *up_list, struct list_head *down_list) { - struct snd_soc_dapm_context *d; int power; switch (w->id) { @@ -1238,26 +1237,6 @@ static void dapm_power_one_widget(struct snd_soc_dapm_widget *w, else power = 1; - if (power) { - d = w->dapm; - - /* Supplies and micbiases only bring the - * context up to STANDBY as unless something - * else is active and passing audio they - * generally don't require full power. - */ - switch (w->id) { - case snd_soc_dapm_supply: - case snd_soc_dapm_micbias: - if (d->target_bias_level < SND_SOC_BIAS_STANDBY) - d->target_bias_level = SND_SOC_BIAS_STANDBY; - break; - default: - d->target_bias_level = SND_SOC_BIAS_ON; - break; - } - } - dapm_widget_set_power(w, power, up_list, down_list); break; } @@ -1302,6 +1281,29 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event) dapm_power_one_widget(w, &up_list, &down_list); } + list_for_each_entry(w, &card->widgets, list) { + if (w->power) { + d = w->dapm; + + /* Supplies and micbiases only bring the + * context up to STANDBY as unless something + * else is active and passing audio they + * generally don't require full power. + */ + switch (w->id) { + case snd_soc_dapm_supply: + case snd_soc_dapm_micbias: + if (d->target_bias_level < SND_SOC_BIAS_STANDBY) + d->target_bias_level = SND_SOC_BIAS_STANDBY; + break; + default: + d->target_bias_level = SND_SOC_BIAS_ON; + break; + } + } + + } + /* If there are no DAPM widgets then try to figure out power from the * event type. */ -- cgit v1.1 From 35c64bcad5c8244d973efbf7e58f6e0e09635504 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 28 Sep 2011 18:23:53 +0100 Subject: ASoC: Ensure all DAPM widgets have a power check callback Makes the code simpler. Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 11 ++++++++--- 1 file changed, 8 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 12bd01a..8d76044 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -857,6 +857,11 @@ static int dapm_supply_check_power(struct snd_soc_dapm_widget *w) return power; } +static int dapm_always_on_check_power(struct snd_soc_dapm_widget *w) +{ + return 1; +} + static int dapm_seq_compare(struct snd_soc_dapm_widget *a, struct snd_soc_dapm_widget *b, bool power_up) @@ -1229,9 +1234,6 @@ static void dapm_power_one_widget(struct snd_soc_dapm_widget *w, break; default: - if (!w->power_check) - break; - if (!w->force) power = w->power_check(w); else @@ -2090,6 +2092,9 @@ int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm) break; } + if (!w->power_check) + w->power_check = dapm_always_on_check_power; + /* Read the initial power state from the device */ if (w->reg >= 0) { val = soc_widget_read(w, w->reg); -- cgit v1.1 From d805002befc52a7edbfb0ec202a10a767e67515d Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 28 Sep 2011 18:28:23 +0100 Subject: ASoC: Factor out widget power check operation We've got the same code in two different places, let's have it in a single place instead. Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 21 ++++++++++----------- 1 file changed, 10 insertions(+), 11 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 8d76044..c1f3563 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -771,6 +771,14 @@ int dapm_reg_event(struct snd_soc_dapm_widget *w, } EXPORT_SYMBOL_GPL(dapm_reg_event); +static int dapm_widget_power_check(struct snd_soc_dapm_widget *w) +{ + if (w->force) + return 1; + else + return w->power_check(w); +} + /* Generic check to see if a widget should be powered. */ static int dapm_generic_check_power(struct snd_soc_dapm_widget *w) @@ -840,13 +848,7 @@ static int dapm_supply_check_power(struct snd_soc_dapm_widget *w) if (!path->sink) continue; - if (path->sink->force) { - power = 1; - break; - } - - if (path->sink->power_check && - path->sink->power_check(path->sink)) { + if (dapm_widget_power_check(path->sink)) { power = 1; break; } @@ -1234,10 +1236,7 @@ static void dapm_power_one_widget(struct snd_soc_dapm_widget *w, break; default: - if (!w->force) - power = w->power_check(w); - else - power = 1; + power = dapm_widget_power_check(w); dapm_widget_set_power(w, power, up_list, down_list); break; -- cgit v1.1 From 565631008f6dd27c3e975c2103141f344d80b84e Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 3 Oct 2011 22:41:09 +0100 Subject: ASoC: Mark headphone, mic, speaker and line widgets as always connected We're not actually doing any dynamic power management based on connection and output drivers (which are pretty much the same thing) are marked as unconditionally connected already. Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index c1f3563..cb00918 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -318,7 +318,7 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w, } } break; - /* does not effect routing - always connected */ + /* does not affect routing - always connected */ case snd_soc_dapm_pga: case snd_soc_dapm_out_drv: case snd_soc_dapm_output: @@ -330,13 +330,13 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w, case snd_soc_dapm_supply: case snd_soc_dapm_aif_in: case snd_soc_dapm_aif_out: - p->connect = 1; - break; - /* does effect routing - dynamically connected */ case snd_soc_dapm_hp: case snd_soc_dapm_mic: case snd_soc_dapm_spk: case snd_soc_dapm_line: + p->connect = 1; + break; + /* does affect routing - dynamically connected */ case snd_soc_dapm_pre: case snd_soc_dapm_post: p->connect = 0; -- cgit v1.1 From db432b414e20b7218bbd91654d7be9c524a4337a Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 3 Oct 2011 21:06:40 +0100 Subject: ASoC: Do DAPM power checks only for widgets changed since last run In order to reduce the number of DAPM power checks we run keep a list of widgets which have been changed since the last DAPM run and iterate over that rather than the full widget list. Whenever we change the power state for a widget we add all the source and sink widgets it has to the dirty list, ensuring that all widgets in the path are checked. This covers more widgets than we need to as some of the neighbour widgets won't be connected but it's simpler as a first step. On one system I tried this gave: Power Path Neighbour Before: 207 1939 2461 After: 114 1066 1327 which seems useful. Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 1 + sound/soc/soc-dapm.c | 61 ++++++++++++++++++++++++++++++++++++++++++++++------ 2 files changed, 56 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 1ed8093..778c177 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2916,6 +2916,7 @@ int snd_soc_register_card(struct snd_soc_card *card) card->rtd[i].dai_link = &card->dai_link[i]; INIT_LIST_HEAD(&card->list); + INIT_LIST_HEAD(&card->dapm_dirty); card->instantiated = 0; mutex_init(&card->mutex); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index cb00918..9d6bb33 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -119,6 +119,17 @@ static void pop_dbg(struct device *dev, u32 pop_time, const char *fmt, ...) kfree(buf); } +static bool dapm_dirty_widget(struct snd_soc_dapm_widget *w) +{ + return !list_empty(&w->dirty); +} + +static void dapm_mark_dirty(struct snd_soc_dapm_widget *w) +{ + if (!dapm_dirty_widget(w)) + list_add_tail(&w->dirty, &w->dapm->card->dapm_dirty); +} + /* create a new dapm widget */ static inline struct snd_soc_dapm_widget *dapm_cnew_widget( const struct snd_soc_dapm_widget *_widget) @@ -1208,11 +1219,30 @@ static void dapm_widget_set_power(struct snd_soc_dapm_widget *w, bool power, struct list_head *up_list, struct list_head *down_list) { + struct snd_soc_dapm_path *path; + if (w->power == power) return; trace_snd_soc_dapm_widget_power(w, power); + /* If we changed our power state perhaps our neigbours changed + * also. We're not yet smart enough to update relevant + * neighbours when we change the state of a widget, this acts + * as a proxy for that. It will notify more neighbours than + * is ideal. + */ + list_for_each_entry(path, &w->sources, list_sink) { + if (path->source) { + dapm_mark_dirty(path->source); + } + } + list_for_each_entry(path, &w->sinks, list_source) { + if (path->sink) { + dapm_mark_dirty(path->sink); + } + } + if (power) dapm_seq_insert(w, up_list, true); else @@ -1276,13 +1306,18 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event) memset(&card->dapm_stats, 0, sizeof(card->dapm_stats)); /* Check which widgets we need to power and store them in - * lists indicating if they should be powered up or down. + * lists indicating if they should be powered up or down. We + * only check widgets that have been flagged as dirty but note + * that new widgets may be added to the dirty list while we + * iterate. */ - list_for_each_entry(w, &card->widgets, list) { + list_for_each_entry(w, &card->dapm_dirty, dirty) { dapm_power_one_widget(w, &up_list, &down_list); } list_for_each_entry(w, &card->widgets, list) { + list_del_init(&w->dirty); + if (w->power) { d = w->dapm; @@ -1573,14 +1608,20 @@ static int dapm_mux_update_power(struct snd_soc_dapm_widget *widget, found = 1; /* we now need to match the string in the enum to the path */ - if (!(strcmp(path->name, e->texts[mux]))) + if (!(strcmp(path->name, e->texts[mux]))) { path->connect = 1; /* new connection */ - else + dapm_mark_dirty(path->source); + } else { + if (path->connect) + dapm_mark_dirty(path->source); path->connect = 0; /* old connection must be powered down */ + } } - if (found) + if (found) { + dapm_mark_dirty(widget); dapm_power_widgets(widget->dapm, SND_SOC_DAPM_STREAM_NOP); + } return 0; } @@ -1605,10 +1646,13 @@ static int dapm_mixer_update_power(struct snd_soc_dapm_widget *widget, /* found, now check type */ found = 1; path->connect = connect; + dapm_mark_dirty(path->source); } - if (found) + if (found) { + dapm_mark_dirty(widget); dapm_power_widgets(widget->dapm, SND_SOC_DAPM_STREAM_NOP); + } return 0; } @@ -1752,6 +1796,7 @@ static int snd_soc_dapm_set_pin(struct snd_soc_dapm_context *dapm, w->connected = status; if (status == 0) w->force = 0; + dapm_mark_dirty(w); return 0; } @@ -2107,6 +2152,7 @@ int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm) w->new = 1; + list_add(&w->dirty, &(w->dapm->card->dapm_dirty)); dapm_debugfs_add_widget(w); } @@ -2588,6 +2634,7 @@ int snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, INIT_LIST_HEAD(&w->sources); INIT_LIST_HEAD(&w->sinks); INIT_LIST_HEAD(&w->list); + INIT_LIST_HEAD(&w->dirty); list_add(&w->list, &dapm->card->widgets); /* machine layer set ups unconnected pins and insertions */ @@ -2638,6 +2685,7 @@ static void soc_dapm_stream_event(struct snd_soc_dapm_context *dapm, dev_vdbg(w->dapm->dev, "widget %s\n %s stream %s event %d\n", w->name, w->sname, stream, event); if (strstr(w->sname, stream)) { + dapm_mark_dirty(w); switch(event) { case SND_SOC_DAPM_STREAM_START: w->active = 1; @@ -2727,6 +2775,7 @@ int snd_soc_dapm_force_enable_pin(struct snd_soc_dapm_context *dapm, dev_dbg(w->dapm->dev, "dapm: force enable pin %s\n", pin); w->connected = 1; w->force = 1; + dapm_mark_dirty(w); return 0; } -- cgit v1.1 From fe4fda5d8f28d06ae8f1482f4bde8a83be16e44b Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 3 Oct 2011 22:36:57 +0100 Subject: ASoC: Reduce the number of neigbours we mark dirty when updating power If two widgets are not currently connected then there is no need to propagate a power state change between them as we mark the affected widgets when we change a connection. Similarly if a neighbour widget is already in the state being set for the current widget then there is no need to recheck. On one system I tested this gave: Power Path Neighbour Before: 114 1066 1327 After: 106 970 1186 which is an improvement, although relatively small. Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 26 ++++++++++++++++++++------ 1 file changed, 20 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 9d6bb33..214a709 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1215,6 +1215,21 @@ static void dapm_post_sequence_async(void *data, async_cookie_t cookie) } } +static void dapm_widget_set_peer_power(struct snd_soc_dapm_widget *peer, + bool power, bool connect) +{ + /* If a connection is being made or broken then that update + * will have marked the peer dirty, otherwise the widgets are + * not connected and this update has no impact. */ + if (!connect) + return; + + /* If the peer is already in the state we're moving to then we + * won't have an impact on it. */ + if (power != peer->power) + dapm_mark_dirty(peer); +} + static void dapm_widget_set_power(struct snd_soc_dapm_widget *w, bool power, struct list_head *up_list, struct list_head *down_list) @@ -1227,19 +1242,18 @@ static void dapm_widget_set_power(struct snd_soc_dapm_widget *w, bool power, trace_snd_soc_dapm_widget_power(w, power); /* If we changed our power state perhaps our neigbours changed - * also. We're not yet smart enough to update relevant - * neighbours when we change the state of a widget, this acts - * as a proxy for that. It will notify more neighbours than - * is ideal. + * also. */ list_for_each_entry(path, &w->sources, list_sink) { if (path->source) { - dapm_mark_dirty(path->source); + dapm_widget_set_peer_power(path->source, power, + path->connect); } } list_for_each_entry(path, &w->sinks, list_source) { if (path->sink) { - dapm_mark_dirty(path->sink); + dapm_widget_set_peer_power(path->sink, power, + path->connect); } } -- cgit v1.1 From 75c1f891b4c394c607532fdcea294c2556e410c4 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 4 Oct 2011 22:28:08 +0100 Subject: ASoC: Add verbose debugging showing why widgets get marked dirty Help diagnose why we're checking widgets by providing some logging when we first dirty them. This should possibly be a trace point if it's useful but can be absurdly verbose if enabled, we can always change it later if desired. Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 26 +++++++++++++++----------- 1 file changed, 15 insertions(+), 11 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 214a709..e6a0882 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -124,10 +124,13 @@ static bool dapm_dirty_widget(struct snd_soc_dapm_widget *w) return !list_empty(&w->dirty); } -static void dapm_mark_dirty(struct snd_soc_dapm_widget *w) +static void dapm_mark_dirty(struct snd_soc_dapm_widget *w, const char *reason) { - if (!dapm_dirty_widget(w)) + if (!dapm_dirty_widget(w)) { + dev_vdbg(w->dapm->dev, "Marking %s dirty due to %s\n", + w->name, reason); list_add_tail(&w->dirty, &w->dapm->card->dapm_dirty); + } } /* create a new dapm widget */ @@ -1227,7 +1230,7 @@ static void dapm_widget_set_peer_power(struct snd_soc_dapm_widget *peer, /* If the peer is already in the state we're moving to then we * won't have an impact on it. */ if (power != peer->power) - dapm_mark_dirty(peer); + dapm_mark_dirty(peer, "peer state change"); } static void dapm_widget_set_power(struct snd_soc_dapm_widget *w, bool power, @@ -1624,16 +1627,17 @@ static int dapm_mux_update_power(struct snd_soc_dapm_widget *widget, /* we now need to match the string in the enum to the path */ if (!(strcmp(path->name, e->texts[mux]))) { path->connect = 1; /* new connection */ - dapm_mark_dirty(path->source); + dapm_mark_dirty(path->source, "mux connection"); } else { if (path->connect) - dapm_mark_dirty(path->source); + dapm_mark_dirty(path->source, + "mux disconnection"); path->connect = 0; /* old connection must be powered down */ } } if (found) { - dapm_mark_dirty(widget); + dapm_mark_dirty(widget, "mux change"); dapm_power_widgets(widget->dapm, SND_SOC_DAPM_STREAM_NOP); } @@ -1660,11 +1664,11 @@ static int dapm_mixer_update_power(struct snd_soc_dapm_widget *widget, /* found, now check type */ found = 1; path->connect = connect; - dapm_mark_dirty(path->source); + dapm_mark_dirty(path->source, "mixer connection"); } if (found) { - dapm_mark_dirty(widget); + dapm_mark_dirty(widget, "mixer update"); dapm_power_widgets(widget->dapm, SND_SOC_DAPM_STREAM_NOP); } @@ -1810,7 +1814,7 @@ static int snd_soc_dapm_set_pin(struct snd_soc_dapm_context *dapm, w->connected = status; if (status == 0) w->force = 0; - dapm_mark_dirty(w); + dapm_mark_dirty(w, "pin configuration"); return 0; } @@ -2699,7 +2703,7 @@ static void soc_dapm_stream_event(struct snd_soc_dapm_context *dapm, dev_vdbg(w->dapm->dev, "widget %s\n %s stream %s event %d\n", w->name, w->sname, stream, event); if (strstr(w->sname, stream)) { - dapm_mark_dirty(w); + dapm_mark_dirty(w, "stream event"); switch(event) { case SND_SOC_DAPM_STREAM_START: w->active = 1; @@ -2789,7 +2793,7 @@ int snd_soc_dapm_force_enable_pin(struct snd_soc_dapm_context *dapm, dev_dbg(w->dapm->dev, "dapm: force enable pin %s\n", pin); w->connected = 1; w->force = 1; - dapm_mark_dirty(w); + dapm_mark_dirty(w, "force enable"); return 0; } -- cgit v1.1 From 9b8a83b205bd07b06784028effd94515fe9278c3 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 4 Oct 2011 22:15:59 +0100 Subject: ASoC: Only run power_check() on a widget once per run Some widgets will get power_check() run on them more than once during a DAPM run, most commonly due to supply widgets checking to see if their consumers are powered up. It's wasteful to do this so cache the result of power_check() during a run. For one system I tested this on I got an improvement of: Power Path Neighbour Before: 106 970 1186 After: 69 727 905 from this. Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 15 +++++++++++++-- 1 file changed, 13 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index e6a0882..c39146d 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -787,10 +787,17 @@ EXPORT_SYMBOL_GPL(dapm_reg_event); static int dapm_widget_power_check(struct snd_soc_dapm_widget *w) { + if (w->power_checked) + return w->new_power; + if (w->force) - return 1; + w->new_power = 1; else - return w->power_check(w); + w->new_power = w->power_check(w); + + w->power_checked = true; + + return w->new_power; } /* Generic check to see if a widget should be powered. @@ -1322,6 +1329,10 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event) memset(&card->dapm_stats, 0, sizeof(card->dapm_stats)); + list_for_each_entry(w, &card->widgets, list) { + w->power_checked = false; + } + /* Check which widgets we need to power and store them in * lists indicating if they should be powered up or down. We * only check widgets that have been flagged as dirty but note -- cgit v1.1 From f3bf3e456a8be9b359a8f4ff458ae1be4fc4c516 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 4 Oct 2011 22:43:31 +0100 Subject: ASoC: Don't mark the outputs of supplies as dirty on state changes The whole point of supply widgets is that they aren't inputs to their sinks so a state change in a supply should never affect the state of the widget being supplied and we don't need to mark them as dirty. Power Path Neighbour Before: 69 727 905 After: 63 607 731 This is particularly useful where supplies affect large portions of the chip (eg, a bandgap supplying the analogue sections). Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 15 +++++++++++---- 1 file changed, 11 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index c39146d..cbca1dd 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1260,11 +1260,18 @@ static void dapm_widget_set_power(struct snd_soc_dapm_widget *w, bool power, path->connect); } } - list_for_each_entry(path, &w->sinks, list_source) { - if (path->sink) { - dapm_widget_set_peer_power(path->sink, power, - path->connect); + switch (w->id) { + case snd_soc_dapm_supply: + /* Supplies can't affect their outputs, only their inputs */ + break; + default: + list_for_each_entry(path, &w->sinks, list_source) { + if (path->sink) { + dapm_widget_set_peer_power(path->sink, power, + path->connect); + } } + break; } if (power) -- cgit v1.1 From f68d7e168785a2e89f615863fb5fab22518c8eb8 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 4 Oct 2011 22:57:50 +0100 Subject: ASoC: Stop checking for supplied widgets after we find the first We don't really care how many widgets a supply is supplying, we just care if the number is non-zero. This didn't actually produce any improvement in the test cases I've been using but seems obviously sensible enough that I'm pushing it out anyway. We could do a similar thing for other widgets but this may be unhelpful for further refactorings Liam was working on aiming to allow us to identify connected audio paths. Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 9 +++------ 1 file changed, 3 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index cbca1dd..82d93bf 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -851,7 +851,6 @@ static int dapm_dac_check_power(struct snd_soc_dapm_widget *w) static int dapm_supply_check_power(struct snd_soc_dapm_widget *w) { struct snd_soc_dapm_path *path; - int power = 0; DAPM_UPDATE_STAT(w, power_checks); @@ -869,15 +868,13 @@ static int dapm_supply_check_power(struct snd_soc_dapm_widget *w) if (!path->sink) continue; - if (dapm_widget_power_check(path->sink)) { - power = 1; - break; - } + if (dapm_widget_power_check(path->sink)) + return 1; } dapm_clear_walk(w->dapm); - return power; + return 0; } static int dapm_always_on_check_power(struct snd_soc_dapm_widget *w) -- cgit v1.1 From 7508b12a8eb713436feb65893ae7ada57bf165ce Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 5 Oct 2011 12:09:12 +0100 Subject: ASoC: Use dapm_mark_dirty() for new DAPM widgets for consistency Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 82d93bf..8711aab 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2185,7 +2185,7 @@ int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm) w->new = 1; - list_add(&w->dirty, &(w->dapm->card->dapm_dirty)); + dapm_mark_dirty(w, "new widget"); dapm_debugfs_add_widget(w); } -- cgit v1.1 From 0f9887d11e7c59ebae5e464f30a6dde788ed9011 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Wed, 5 Oct 2011 10:29:19 +0300 Subject: ASoC: Consolidate use of controls with custom get/put function Use the macros for controls require custom get/put function. This is to make sure that the soc_mixer_control is used consistently among the drivers. Signed-off-by: Peter Ujfalusi Cc: Arun KS Cc: Misael Lopez Cruz Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic23.c | 14 +++--------- sound/soc/codecs/twl4030.c | 48 +++++++++++------------------------------- sound/soc/codecs/twl6040.c | 37 ++++++-------------------------- sound/soc/codecs/wm8350.c | 39 +++++++++++++--------------------- sound/soc/codecs/wm8580.c | 36 +++++++++++-------------------- sound/soc/codecs/wm_hubs.c | 18 +++++----------- 6 files changed, 53 insertions(+), 139 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index 33bb52f..c3a4bb2 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -159,15 +159,6 @@ static int snd_soc_tlv320aic23_get_volsw(struct snd_kcontrol *kcontrol, } -#define SOC_TLV320AIC23_SINGLE_TLV(xname, reg, shift, max, invert, tlv_array) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ - .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\ - SNDRV_CTL_ELEM_ACCESS_READWRITE,\ - .tlv.p = (tlv_array), \ - .info = snd_soc_info_volsw, .get = snd_soc_tlv320aic23_get_volsw,\ - .put = snd_soc_tlv320aic23_put_volsw, \ - .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert) } - static const struct snd_kcontrol_new tlv320aic23_snd_controls[] = { SOC_DOUBLE_R_TLV("Digital Playback Volume", TLV320AIC23_LCHNVOL, TLV320AIC23_RCHNVOL, 0, 127, 0, out_gain_tlv), @@ -178,8 +169,9 @@ static const struct snd_kcontrol_new tlv320aic23_snd_controls[] = { TLV320AIC23_RINVOL, 0, 31, 0, input_gain_tlv), SOC_SINGLE("Mic Input Switch", TLV320AIC23_ANLG, 1, 1, 1), SOC_SINGLE("Mic Booster Switch", TLV320AIC23_ANLG, 0, 1, 0), - SOC_TLV320AIC23_SINGLE_TLV("Sidetone Volume", TLV320AIC23_ANLG, - 6, 4, 0, sidetone_vol_tlv), + SOC_SINGLE_EXT_TLV("Sidetone Volume", TLV320AIC23_ANLG, 6, 4, 0, + snd_soc_tlv320aic23_get_volsw, + snd_soc_tlv320aic23_put_volsw, sidetone_vol_tlv), SOC_ENUM("Playback De-emphasis", tlv320aic23_deemph), }; diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 71674be..7c244cd 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -863,34 +863,6 @@ static int digimic_event(struct snd_soc_dapm_widget *w, * Inverting not going to help with these. * Custom volsw and volsw_2r get/put functions to handle these gain bits. */ -#define SOC_DOUBLE_TLV_TWL4030(xname, xreg, shift_left, shift_right, xmax,\ - xinvert, tlv_array) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\ - .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\ - SNDRV_CTL_ELEM_ACCESS_READWRITE,\ - .tlv.p = (tlv_array), \ - .info = snd_soc_info_volsw, \ - .get = snd_soc_get_volsw_twl4030, \ - .put = snd_soc_put_volsw_twl4030, \ - .private_value = (unsigned long)&(struct soc_mixer_control) \ - {.reg = xreg, .shift = shift_left, .rshift = shift_right,\ - .max = xmax, .invert = xinvert} } -#define SOC_DOUBLE_R_TLV_TWL4030(xname, reg_left, reg_right, xshift, xmax,\ - xinvert, tlv_array) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\ - .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\ - SNDRV_CTL_ELEM_ACCESS_READWRITE,\ - .tlv.p = (tlv_array), \ - .info = snd_soc_info_volsw_2r, \ - .get = snd_soc_get_volsw_r2_twl4030,\ - .put = snd_soc_put_volsw_r2_twl4030, \ - .private_value = (unsigned long)&(struct soc_mixer_control) \ - {.reg = reg_left, .rreg = reg_right, .shift = xshift, \ - .rshift = xshift, .max = xmax, .invert = xinvert} } -#define SOC_SINGLE_TLV_TWL4030(xname, xreg, xshift, xmax, xinvert, tlv_array) \ - SOC_DOUBLE_TLV_TWL4030(xname, xreg, xshift, xshift, xmax, \ - xinvert, tlv_array) - static int snd_soc_get_volsw_twl4030(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -1197,19 +1169,23 @@ static const struct snd_kcontrol_new twl4030_snd_controls[] = { TWL4030_REG_VDL_APGA_CTL, 1, 1, 0), /* Separate output gain controls */ - SOC_DOUBLE_R_TLV_TWL4030("PreDriv Playback Volume", + SOC_DOUBLE_R_EXT_TLV("PreDriv Playback Volume", TWL4030_REG_PREDL_CTL, TWL4030_REG_PREDR_CTL, - 4, 3, 0, output_tvl), + 4, 3, 0, snd_soc_get_volsw_r2_twl4030, + snd_soc_put_volsw_r2_twl4030, output_tvl), - SOC_DOUBLE_TLV_TWL4030("Headset Playback Volume", - TWL4030_REG_HS_GAIN_SET, 0, 2, 3, 0, output_tvl), + SOC_DOUBLE_EXT_TLV("Headset Playback Volume", + TWL4030_REG_HS_GAIN_SET, 0, 2, 3, 0, snd_soc_get_volsw_twl4030, + snd_soc_put_volsw_twl4030, output_tvl), - SOC_DOUBLE_R_TLV_TWL4030("Carkit Playback Volume", + SOC_DOUBLE_R_EXT_TLV("Carkit Playback Volume", TWL4030_REG_PRECKL_CTL, TWL4030_REG_PRECKR_CTL, - 4, 3, 0, output_tvl), + 4, 3, 0, snd_soc_get_volsw_r2_twl4030, + snd_soc_put_volsw_r2_twl4030, output_tvl), - SOC_SINGLE_TLV_TWL4030("Earpiece Playback Volume", - TWL4030_REG_EAR_CTL, 4, 3, 0, output_ear_tvl), + SOC_SINGLE_EXT_TLV("Earpiece Playback Volume", + TWL4030_REG_EAR_CTL, 4, 3, 0, snd_soc_get_volsw_twl4030, + snd_soc_put_volsw_twl4030, output_ear_tvl), /* Common capture gain controls */ SOC_DOUBLE_R_TLV("TX1 Digital Capture Volume", diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 8c740c1..11f681b 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -812,33 +812,6 @@ static int twl6040_get_volsw(struct snd_kcontrol *kcontrol, return snd_soc_get_volsw(kcontrol, ucontrol); } -/* double control with volume update */ -#define SOC_TWL6040_DOUBLE_TLV(xname, xreg, shift_left, shift_right, xmax,\ - xinvert, tlv_array)\ -{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\ - .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\ - SNDRV_CTL_ELEM_ACCESS_READWRITE,\ - .tlv.p = (tlv_array), \ - .info = snd_soc_info_volsw, .get = twl6040_get_volsw, \ - .put = twl6040_put_volsw, \ - .private_value = (unsigned long)&(struct soc_mixer_control) \ - {.reg = xreg, .shift = shift_left, .rshift = shift_right,\ - .max = xmax, .platform_max = xmax, .invert = xinvert} } - -/* double control with volume update */ -#define SOC_TWL6040_DOUBLE_R_TLV(xname, reg_left, reg_right, xshift, xmax,\ - xinvert, tlv_array)\ -{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\ - .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ - SNDRV_CTL_ELEM_ACCESS_READWRITE | \ - SNDRV_CTL_ELEM_ACCESS_VOLATILE, \ - .tlv.p = (tlv_array), \ - .info = snd_soc_info_volsw_2r, \ - .get = twl6040_get_volsw, .put = twl6040_put_volsw, \ - .private_value = (unsigned long)&(struct soc_mixer_control) \ - {.reg = reg_left, .rreg = reg_right, .shift = xshift, \ - .rshift = xshift, .max = xmax, .invert = xinvert}, } - /* * MICATT volume control: * from -6 to 0 dB in 6 dB steps @@ -1027,10 +1000,12 @@ static const struct snd_kcontrol_new twl6040_snd_controls[] = { TWL6040_REG_LINEGAIN, 0, 3, 7, 0, afm_amp_tlv), /* Playback gains */ - SOC_TWL6040_DOUBLE_TLV("Headset Playback Volume", - TWL6040_REG_HSGAIN, 0, 4, 0xF, 1, hs_tlv), - SOC_TWL6040_DOUBLE_R_TLV("Handsfree Playback Volume", - TWL6040_REG_HFLGAIN, TWL6040_REG_HFRGAIN, 0, 0x1D, 1, hf_tlv), + SOC_DOUBLE_EXT_TLV("Headset Playback Volume", + TWL6040_REG_HSGAIN, 0, 4, 0xF, 1, twl6040_get_volsw, + twl6040_put_volsw, hs_tlv), + SOC_DOUBLE_R_EXT_TLV("Handsfree Playback Volume", + TWL6040_REG_HFLGAIN, TWL6040_REG_HFRGAIN, 0, 0x1D, 1, + twl6040_get_volsw, twl6040_put_volsw, hf_tlv), SOC_SINGLE_TLV("Earphone Playback Volume", TWL6040_REG_EARCTL, 1, 0xF, 1, ep_tlv), diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index 6d6dc9e..50ea9d7 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -395,20 +395,6 @@ static int wm8350_get_volsw_2r(struct snd_kcontrol *kcontrol, return snd_soc_get_volsw_2r(kcontrol, ucontrol); } -/* double control with volume update */ -#define SOC_WM8350_DOUBLE_R_TLV(xname, reg_left, reg_right, xshift, xmax, \ - xinvert, tlv_array) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ - .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ - SNDRV_CTL_ELEM_ACCESS_READWRITE | \ - SNDRV_CTL_ELEM_ACCESS_VOLATILE, \ - .tlv.p = (tlv_array), \ - .info = snd_soc_info_volsw_2r, \ - .get = wm8350_get_volsw_2r, .put = wm8350_put_volsw_2r_vu, \ - .private_value = (unsigned long)&(struct soc_mixer_control) \ - {.reg = reg_left, .rreg = reg_right, .shift = xshift, \ - .rshift = xshift, .max = xmax, .invert = xinvert}, } - static const char *wm8350_deemp[] = { "None", "32kHz", "44.1kHz", "48kHz" }; static const char *wm8350_pol[] = { "Normal", "Inv R", "Inv L", "Inv L & R" }; static const char *wm8350_dacmutem[] = { "Normal", "Soft" }; @@ -443,26 +429,29 @@ static const unsigned int capture_sd_tlv[] = { static const struct snd_kcontrol_new wm8350_snd_controls[] = { SOC_ENUM("Playback Deemphasis", wm8350_enum[0]), SOC_ENUM("Playback DAC Inversion", wm8350_enum[1]), - SOC_WM8350_DOUBLE_R_TLV("Playback PCM Volume", + SOC_DOUBLE_R_EXT_TLV("Playback PCM Volume", WM8350_DAC_DIGITAL_VOLUME_L, WM8350_DAC_DIGITAL_VOLUME_R, - 0, 255, 0, dac_pcm_tlv), + 0, 255, 0, wm8350_get_volsw_2r, + wm8350_put_volsw_2r_vu, dac_pcm_tlv), SOC_ENUM("Playback PCM Mute Function", wm8350_enum[2]), SOC_ENUM("Playback PCM Mute Speed", wm8350_enum[3]), SOC_ENUM("Capture PCM Filter", wm8350_enum[4]), SOC_ENUM("Capture PCM HP Filter", wm8350_enum[5]), SOC_ENUM("Capture ADC Inversion", wm8350_enum[6]), - SOC_WM8350_DOUBLE_R_TLV("Capture PCM Volume", + SOC_DOUBLE_R_EXT_TLV("Capture PCM Volume", WM8350_ADC_DIGITAL_VOLUME_L, WM8350_ADC_DIGITAL_VOLUME_R, - 0, 255, 0, adc_pcm_tlv), + 0, 255, 0, wm8350_get_volsw_2r, + wm8350_put_volsw_2r_vu, adc_pcm_tlv), SOC_DOUBLE_TLV("Capture Sidetone Volume", WM8350_ADC_DIVIDER, 8, 4, 15, 1, capture_sd_tlv), - SOC_WM8350_DOUBLE_R_TLV("Capture Volume", + SOC_DOUBLE_R_EXT_TLV("Capture Volume", WM8350_LEFT_INPUT_VOLUME, WM8350_RIGHT_INPUT_VOLUME, - 2, 63, 0, pre_amp_tlv), + 2, 63, 0, wm8350_get_volsw_2r, + wm8350_put_volsw_2r_vu, pre_amp_tlv), SOC_DOUBLE_R("Capture ZC Switch", WM8350_LEFT_INPUT_VOLUME, WM8350_RIGHT_INPUT_VOLUME, 13, 1, 0), @@ -490,17 +479,19 @@ static const struct snd_kcontrol_new wm8350_snd_controls[] = { SOC_SINGLE_TLV("Out4 Capture Volume", WM8350_INPUT_MIXER_VOLUME, 1, 7, 0, out_mix_tlv), - SOC_WM8350_DOUBLE_R_TLV("Out1 Playback Volume", + SOC_DOUBLE_R_EXT_TLV("Out1 Playback Volume", WM8350_LOUT1_VOLUME, WM8350_ROUT1_VOLUME, - 2, 63, 0, out_pga_tlv), + 2, 63, 0, wm8350_get_volsw_2r, + wm8350_put_volsw_2r_vu, out_pga_tlv), SOC_DOUBLE_R("Out1 Playback ZC Switch", WM8350_LOUT1_VOLUME, WM8350_ROUT1_VOLUME, 13, 1, 0), - SOC_WM8350_DOUBLE_R_TLV("Out2 Playback Volume", + SOC_DOUBLE_R_EXT_TLV("Out2 Playback Volume", WM8350_LOUT2_VOLUME, WM8350_ROUT2_VOLUME, - 2, 63, 0, out_pga_tlv), + 2, 63, 0, wm8350_get_volsw_2r, + wm8350_put_volsw_2r_vu, out_pga_tlv), SOC_DOUBLE_R("Out2 Playback ZC Switch", WM8350_LOUT2_VOLUME, WM8350_ROUT2_VOLUME, 13, 1, 0), SOC_SINGLE("Out2 Right Invert Switch", WM8350_ROUT2_VOLUME, 10, 1, 0), diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index 4664c3a..02cbf13 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -224,31 +224,19 @@ static int wm8580_out_vu(struct snd_kcontrol *kcontrol, return 0; } -#define SOC_WM8580_OUT_DOUBLE_R_TLV(xname, reg_left, reg_right, xshift, xmax, \ - xinvert, tlv_array) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ - .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\ - SNDRV_CTL_ELEM_ACCESS_READWRITE, \ - .tlv.p = (tlv_array), \ - .info = snd_soc_info_volsw_2r, \ - .get = snd_soc_get_volsw_2r, .put = wm8580_out_vu, \ - .private_value = (unsigned long)&(struct soc_mixer_control) \ - {.reg = reg_left, .rreg = reg_right, .shift = xshift, \ - .max = xmax, .invert = xinvert} } - static const struct snd_kcontrol_new wm8580_snd_controls[] = { -SOC_WM8580_OUT_DOUBLE_R_TLV("DAC1 Playback Volume", - WM8580_DIGITAL_ATTENUATION_DACL1, - WM8580_DIGITAL_ATTENUATION_DACR1, - 0, 0xff, 0, dac_tlv), -SOC_WM8580_OUT_DOUBLE_R_TLV("DAC2 Playback Volume", - WM8580_DIGITAL_ATTENUATION_DACL2, - WM8580_DIGITAL_ATTENUATION_DACR2, - 0, 0xff, 0, dac_tlv), -SOC_WM8580_OUT_DOUBLE_R_TLV("DAC3 Playback Volume", - WM8580_DIGITAL_ATTENUATION_DACL3, - WM8580_DIGITAL_ATTENUATION_DACR3, - 0, 0xff, 0, dac_tlv), +SOC_DOUBLE_R_EXT_TLV("DAC1 Playback Volume", + WM8580_DIGITAL_ATTENUATION_DACL1, + WM8580_DIGITAL_ATTENUATION_DACR1, + 0, 0xff, 0, snd_soc_get_volsw_2r, wm8580_out_vu, dac_tlv), +SOC_DOUBLE_R_EXT_TLV("DAC2 Playback Volume", + WM8580_DIGITAL_ATTENUATION_DACL2, + WM8580_DIGITAL_ATTENUATION_DACR2, + 0, 0xff, 0, snd_soc_get_volsw_2r, wm8580_out_vu, dac_tlv), +SOC_DOUBLE_R_EXT_TLV("DAC3 Playback Volume", + WM8580_DIGITAL_ATTENUATION_DACL3, + WM8580_DIGITAL_ATTENUATION_DACR3, + 0, 0xff, 0, snd_soc_get_volsw_2r, wm8580_out_vu, dac_tlv), SOC_SINGLE("DAC1 Deemphasis Switch", WM8580_DAC_CONTROL3, 0, 1, 0), SOC_SINGLE("DAC2 Deemphasis Switch", WM8580_DAC_CONTROL3, 1, 1, 0), diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index ca8ce03..f3583a5 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -362,19 +362,11 @@ SOC_DOUBLE_TLV("Speaker Boost Volume", WM8993_SPKOUT_BOOST, 3, 0, 7, 0, SOC_ENUM("Speaker Reference", speaker_ref), SOC_ENUM("Speaker Mode", speaker_mode), -{ - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Headphone Volume", - .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | - SNDRV_CTL_ELEM_ACCESS_READWRITE, - .tlv.p = outpga_tlv, - .info = snd_soc_info_volsw_2r, - .get = snd_soc_get_volsw_2r, .put = wm8993_put_dc_servo, - .private_value = (unsigned long)&(struct soc_mixer_control) { - .reg = WM8993_LEFT_OUTPUT_VOLUME, - .rreg = WM8993_RIGHT_OUTPUT_VOLUME, - .shift = 0, .max = 63 - }, -}, +SOC_DOUBLE_R_EXT_TLV("Headphone Volume", + WM8993_LEFT_OUTPUT_VOLUME, WM8993_RIGHT_OUTPUT_VOLUME, + 0, 63, 0, snd_soc_get_volsw_2r, wm8993_put_dc_servo, + outpga_tlv), + SOC_DOUBLE_R("Headphone Switch", WM8993_LEFT_OUTPUT_VOLUME, WM8993_RIGHT_OUTPUT_VOLUME, 6, 1, 0), SOC_DOUBLE_R("Headphone ZC Switch", WM8993_LEFT_OUTPUT_VOLUME, -- cgit v1.1 From e49b68339ebc7d2e67dc1ae16a4ac6a35fcfc9d5 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Wed, 5 Oct 2011 10:29:20 +0300 Subject: ASoC: twl6040: Simplify custom get_volsw callback The custom get_volsw does not need to call any core get_volsw calls, since we are returning the shadow values for the gains. Return -EINVAL in the unlikely event, if the function has been called for unhandled control. This way we can remove one check in the code. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl6040.c | 18 ++++++------------ 1 file changed, 6 insertions(+), 12 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 11f681b..4ad04e3 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -796,20 +796,14 @@ static int twl6040_get_volsw(struct snd_kcontrol *kcontrol, out = &twl6040_priv->handsfree; break; default: - break; - } - - if (out) { - ucontrol->value.integer.value[0] = out->left_vol; - ucontrol->value.integer.value[1] = out->right_vol; - return 0; + dev_warn(codec->dev, "%s: Unexpected register: 0x%02x\n", + __func__, mc->reg); + return -EINVAL; } - /* call the appropriate handler depending on the rreg */ - if (mc->rreg) - return snd_soc_get_volsw_2r(kcontrol, ucontrol); - else - return snd_soc_get_volsw(kcontrol, ucontrol); + ucontrol->value.integer.value[0] = out->left_vol; + ucontrol->value.integer.value[1] = out->right_vol; + return 0; } /* -- cgit v1.1 From 08a1ed76f5cf94bef07cb370b079760553a87b4b Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Wed, 5 Oct 2011 10:29:21 +0300 Subject: ASoC: twl6040: Prepare for core put_volsw/volsw_2r merger Avoid using the mc->rreg to identify the 2r type of gain control. Introduce a variable to track this. This change is needed to avoid breakage with the upcoming volsw volsw_2r merger. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl6040.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 4ad04e3..c9a601d 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -746,7 +746,7 @@ static int twl6040_put_volsw(struct snd_kcontrol *kcontrol, struct twl6040_output *out = NULL; struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; - int ret; + int ret, type_2r; /* For HS and HF we shadow the values and only actually write * them out when active in order to ensure the amplifier comes on @@ -754,9 +754,11 @@ static int twl6040_put_volsw(struct snd_kcontrol *kcontrol, switch (mc->reg) { case TWL6040_REG_HSGAIN: out = &twl6040_priv->headset; + type_2r = 0; break; case TWL6040_REG_HFLGAIN: out = &twl6040_priv->handsfree; + type_2r = 1; break; default: return -EINVAL; @@ -768,7 +770,7 @@ static int twl6040_put_volsw(struct snd_kcontrol *kcontrol, return 1; /* call the appropriate handler depending on the rreg */ - if (mc->rreg) + if (type_2r) ret = snd_soc_put_volsw_2r(kcontrol, ucontrol); else ret = snd_soc_put_volsw(kcontrol, ucontrol); -- cgit v1.1 From e8f5a10307f7d224df91776033a0b8559a559844 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Wed, 5 Oct 2011 10:29:23 +0300 Subject: ASoC: core: Combine snd_soc_info_volsw/info_volsw_2r functions Handle the info_volsw/info_volsw_2r in one function. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 40 +++------------------------------------- 1 file changed, 3 insertions(+), 37 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 778c177..34e8550 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2248,7 +2248,8 @@ EXPORT_SYMBOL_GPL(snd_soc_info_volsw_ext); * @kcontrol: mixer control * @uinfo: control element information * - * Callback to provide information about a single mixer control. + * Callback to provide information about a single mixer control, or a double + * mixer control that spans 2 registers. * * Returns 0 for success. */ @@ -2258,8 +2259,6 @@ int snd_soc_info_volsw(struct snd_kcontrol *kcontrol, struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; int platform_max; - unsigned int shift = mc->shift; - unsigned int rshift = mc->rshift; if (!mc->platform_max) mc->platform_max = mc->max; @@ -2270,7 +2269,7 @@ int snd_soc_info_volsw(struct snd_kcontrol *kcontrol, else uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; - uinfo->count = shift == rshift ? 1 : 2; + uinfo->count = snd_soc_volsw_is_stereo(mc) ? 2 : 1; uinfo->value.integer.min = 0; uinfo->value.integer.max = platform_max; return 0; @@ -2356,39 +2355,6 @@ int snd_soc_put_volsw(struct snd_kcontrol *kcontrol, EXPORT_SYMBOL_GPL(snd_soc_put_volsw); /** - * snd_soc_info_volsw_2r - double mixer info callback - * @kcontrol: mixer control - * @uinfo: control element information - * - * Callback to provide information about a double mixer control that - * spans 2 codec registers. - * - * Returns 0 for success. - */ -int snd_soc_info_volsw_2r(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - struct soc_mixer_control *mc = - (struct soc_mixer_control *)kcontrol->private_value; - int platform_max; - - if (!mc->platform_max) - mc->platform_max = mc->max; - platform_max = mc->platform_max; - - if (platform_max == 1 && !strstr(kcontrol->id.name, " Volume")) - uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; - else - uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; - - uinfo->count = 2; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = platform_max; - return 0; -} -EXPORT_SYMBOL_GPL(snd_soc_info_volsw_2r); - -/** * snd_soc_get_volsw_2r - double mixer get callback * @kcontrol: mixer control * @ucontrol: control element information -- cgit v1.1 From f7915d997554d4e2ce123c7a4ddd28e12c2e034c Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Wed, 5 Oct 2011 10:29:24 +0300 Subject: ASoC: core: Combine snd_soc_get_volsw/get_volsw_2r functions Handle the get_volsw/get_volsw_2r in one function. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 56 ++++++++++++---------------------------------------- 1 file changed, 13 insertions(+), 43 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 34e8550..1a13d53 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2281,7 +2281,8 @@ EXPORT_SYMBOL_GPL(snd_soc_info_volsw); * @kcontrol: mixer control * @ucontrol: control element information * - * Callback to get the value of a single mixer control. + * Callback to get the value of a single mixer control, or a double mixer + * control that spans 2 registers. * * Returns 0 for success. */ @@ -2292,6 +2293,7 @@ int snd_soc_get_volsw(struct snd_kcontrol *kcontrol, (struct soc_mixer_control *)kcontrol->private_value; struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); unsigned int reg = mc->reg; + unsigned int reg2 = mc->rreg; unsigned int shift = mc->shift; unsigned int rshift = mc->rshift; int max = mc->max; @@ -2300,13 +2302,18 @@ int snd_soc_get_volsw(struct snd_kcontrol *kcontrol, ucontrol->value.integer.value[0] = (snd_soc_read(codec, reg) >> shift) & mask; - if (shift != rshift) - ucontrol->value.integer.value[1] = - (snd_soc_read(codec, reg) >> rshift) & mask; - if (invert) { + if (invert) ucontrol->value.integer.value[0] = max - ucontrol->value.integer.value[0]; - if (shift != rshift) + + if (snd_soc_volsw_is_stereo(mc)) { + if (reg == reg2) + ucontrol->value.integer.value[1] = + (snd_soc_read(codec, reg) >> rshift) & mask; + else + ucontrol->value.integer.value[1] = + (snd_soc_read(codec, reg2) >> shift) & mask; + if (invert) ucontrol->value.integer.value[1] = max - ucontrol->value.integer.value[1]; } @@ -2355,43 +2362,6 @@ int snd_soc_put_volsw(struct snd_kcontrol *kcontrol, EXPORT_SYMBOL_GPL(snd_soc_put_volsw); /** - * snd_soc_get_volsw_2r - double mixer get callback - * @kcontrol: mixer control - * @ucontrol: control element information - * - * Callback to get the value of a double mixer control that spans 2 registers. - * - * Returns 0 for success. - */ -int snd_soc_get_volsw_2r(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct soc_mixer_control *mc = - (struct soc_mixer_control *)kcontrol->private_value; - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - unsigned int reg = mc->reg; - unsigned int reg2 = mc->rreg; - unsigned int shift = mc->shift; - int max = mc->max; - unsigned int mask = (1 << fls(max)) - 1; - unsigned int invert = mc->invert; - - ucontrol->value.integer.value[0] = - (snd_soc_read(codec, reg) >> shift) & mask; - ucontrol->value.integer.value[1] = - (snd_soc_read(codec, reg2) >> shift) & mask; - if (invert) { - ucontrol->value.integer.value[0] = - max - ucontrol->value.integer.value[0]; - ucontrol->value.integer.value[1] = - max - ucontrol->value.integer.value[1]; - } - - return 0; -} -EXPORT_SYMBOL_GPL(snd_soc_get_volsw_2r); - -/** * snd_soc_put_volsw_2r - double mixer set callback * @kcontrol: mixer control * @ucontrol: control element information -- cgit v1.1 From 974815ba4f88f3f12f6f01384e822b23be058323 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Wed, 5 Oct 2011 10:29:25 +0300 Subject: ASoC: core: Combine snd_soc_put_volsw/put_volsw_2r functions Handle the put_volsw/put_volsw_2r in one function. To avoid build breakage in twl6040 keep the snd_soc_put_volsw_2r as define, and map it snd_soc_put_volsw. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 66 +++++++++++++++------------------------------------- 1 file changed, 19 insertions(+), 47 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 1a13d53..2a25076 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2327,7 +2327,8 @@ EXPORT_SYMBOL_GPL(snd_soc_get_volsw); * @kcontrol: mixer control * @ucontrol: control element information * - * Callback to set the value of a single mixer control. + * Callback to set the value of a single mixer control, or a double mixer + * control that spans 2 registers. * * Returns 0 for success. */ @@ -2338,73 +2339,44 @@ int snd_soc_put_volsw(struct snd_kcontrol *kcontrol, (struct soc_mixer_control *)kcontrol->private_value; struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); unsigned int reg = mc->reg; + unsigned int reg2 = mc->rreg; unsigned int shift = mc->shift; unsigned int rshift = mc->rshift; int max = mc->max; unsigned int mask = (1 << fls(max)) - 1; unsigned int invert = mc->invert; - unsigned int val, val2, val_mask; + int err; + bool type_2r = 0; + unsigned int val2 = 0; + unsigned int val, val_mask; val = (ucontrol->value.integer.value[0] & mask); if (invert) val = max - val; val_mask = mask << shift; val = val << shift; - if (shift != rshift) { + if (snd_soc_volsw_is_stereo(mc)) { val2 = (ucontrol->value.integer.value[1] & mask); if (invert) val2 = max - val2; - val_mask |= mask << rshift; - val |= val2 << rshift; - } - return snd_soc_update_bits_locked(codec, reg, val_mask, val); -} -EXPORT_SYMBOL_GPL(snd_soc_put_volsw); - -/** - * snd_soc_put_volsw_2r - double mixer set callback - * @kcontrol: mixer control - * @ucontrol: control element information - * - * Callback to set the value of a double mixer control that spans 2 registers. - * - * Returns 0 for success. - */ -int snd_soc_put_volsw_2r(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct soc_mixer_control *mc = - (struct soc_mixer_control *)kcontrol->private_value; - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - unsigned int reg = mc->reg; - unsigned int reg2 = mc->rreg; - unsigned int shift = mc->shift; - int max = mc->max; - unsigned int mask = (1 << fls(max)) - 1; - unsigned int invert = mc->invert; - int err; - unsigned int val, val2, val_mask; - - val_mask = mask << shift; - val = (ucontrol->value.integer.value[0] & mask); - val2 = (ucontrol->value.integer.value[1] & mask); - - if (invert) { - val = max - val; - val2 = max - val2; + if (reg == reg2) { + val_mask |= mask << rshift; + val |= val2 << rshift; + } else { + val2 = val2 << shift; + type_2r = 1; + } } - - val = val << shift; - val2 = val2 << shift; - err = snd_soc_update_bits_locked(codec, reg, val_mask, val); if (err < 0) return err; - err = snd_soc_update_bits_locked(codec, reg2, val_mask, val2); + if (type_2r) + err = snd_soc_update_bits_locked(codec, reg2, val_mask, val2); + return err; } -EXPORT_SYMBOL_GPL(snd_soc_put_volsw_2r); +EXPORT_SYMBOL_GPL(snd_soc_put_volsw); /** * snd_soc_info_volsw_s8 - signed mixer info callback -- cgit v1.1 From db382da5ff286b406c4819cc9ebd96bbb680884c Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Wed, 5 Oct 2011 10:29:26 +0300 Subject: ASoC: twl6040: Simply call snd_soc_put_volsw form the custom code The ASoC core now have one callback function, which can handle single, and double register mixer controls. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl6040.c | 11 ++--------- 1 file changed, 2 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index c9a601d..7450e1b 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -746,7 +746,7 @@ static int twl6040_put_volsw(struct snd_kcontrol *kcontrol, struct twl6040_output *out = NULL; struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; - int ret, type_2r; + int ret; /* For HS and HF we shadow the values and only actually write * them out when active in order to ensure the amplifier comes on @@ -754,11 +754,9 @@ static int twl6040_put_volsw(struct snd_kcontrol *kcontrol, switch (mc->reg) { case TWL6040_REG_HSGAIN: out = &twl6040_priv->headset; - type_2r = 0; break; case TWL6040_REG_HFLGAIN: out = &twl6040_priv->handsfree; - type_2r = 1; break; default: return -EINVAL; @@ -769,12 +767,7 @@ static int twl6040_put_volsw(struct snd_kcontrol *kcontrol, if (!out->active) return 1; - /* call the appropriate handler depending on the rreg */ - if (type_2r) - ret = snd_soc_put_volsw_2r(kcontrol, ucontrol); - else - ret = snd_soc_put_volsw(kcontrol, ucontrol); - + ret = snd_soc_put_volsw(kcontrol, ucontrol); if (ret < 0) return ret; -- cgit v1.1 From a0acf47f1b986a89026a26fc0365b4ed2f65cd85 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Wed, 5 Oct 2011 10:29:28 +0300 Subject: ASoC: twl6040: Warn user in twl6040_put_volsw for error case Let the user know, that the callback has been called with unexpected register parameter. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl6040.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 7450e1b..62edded 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -759,6 +759,8 @@ static int twl6040_put_volsw(struct snd_kcontrol *kcontrol, out = &twl6040_priv->handsfree; break; default: + dev_warn(codec->dev, "%s: Unexpected register: 0x%02x\n", + __func__, mc->reg); return -EINVAL; } -- cgit v1.1 From 089f3383c7f6f080a86d6d891c1474c4756b9ba2 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Wed, 5 Oct 2011 14:40:46 +0800 Subject: ASoC: Remove unused function declaration in imx-ssi.h These functions are removed in commit f0fba2ad "ASoC: multi-component - ASoC Multi-Component Support". Let's remove the leftover function declaration in header file. Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/imx/imx-ssi.h | 6 ------ 1 file changed, 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/imx/imx-ssi.h b/sound/soc/imx/imx-ssi.h index 0a84cec..1072dfb 100644 --- a/sound/soc/imx/imx-ssi.h +++ b/sound/soc/imx/imx-ssi.h @@ -218,12 +218,6 @@ struct imx_ssi { struct platform_device *soc_platform_pdev_fiq; }; -struct snd_soc_platform *imx_ssi_fiq_init(struct platform_device *pdev, - struct imx_ssi *ssi); -void imx_ssi_fiq_exit(struct platform_device *pdev, struct imx_ssi *ssi); -struct snd_soc_platform *imx_ssi_dma_mx2_init(struct platform_device *pdev, - struct imx_ssi *ssi); - int snd_imx_pcm_mmap(struct snd_pcm_substream *substream, struct vm_area_struct *vma); int imx_pcm_new(struct snd_soc_pcm_runtime *rtd); void imx_pcm_free(struct snd_pcm *pcm); -- cgit v1.1 From 0df2c594b92ebb18ed415f3b48ec2bd331b114a9 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Wed, 5 Oct 2011 14:41:35 +0800 Subject: ASoC: imx: Remove unused variable 'dai' Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/imx/imx-pcm-fiq.c | 1 - sound/soc/imx/imx-ssi.c | 1 - 2 files changed, 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/imx/imx-pcm-fiq.c b/sound/soc/imx/imx-pcm-fiq.c index c8527ea..8df0fae2 100644 --- a/sound/soc/imx/imx-pcm-fiq.c +++ b/sound/soc/imx/imx-pcm-fiq.c @@ -240,7 +240,6 @@ static int ssi_irq = 0; static int imx_pcm_fiq_new(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_dai *dai = rtd->cpu_dai; struct snd_pcm *pcm = rtd->pcm; struct snd_pcm_substream *substream; int ret; diff --git a/sound/soc/imx/imx-ssi.c b/sound/soc/imx/imx-ssi.c index 4297cb6..4c05e2b 100644 --- a/sound/soc/imx/imx-ssi.c +++ b/sound/soc/imx/imx-ssi.c @@ -391,7 +391,6 @@ static u64 imx_pcm_dmamask = DMA_BIT_MASK(32); int imx_pcm_new(struct snd_soc_pcm_runtime *rtd) { struct snd_card *card = rtd->card->snd_card; - struct snd_soc_dai *dai = rtd->cpu_dai; struct snd_pcm *pcm = rtd->pcm; int ret = 0; -- cgit v1.1 From 698570062d324e40d86294b585f2d08608caebde Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Wed, 5 Oct 2011 14:47:15 +0800 Subject: ASoC: Remove unused variable 'wm9090' in wm9090_probe Eliminate below build warning: CC sound/soc/codecs/wm9090.o sound/soc/codecs/wm9090.c: In function 'wm9090_probe': sound/soc/codecs/wm9090.c:550: warning: unused variable 'wm9090' Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/wm9090.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm9090.c b/sound/soc/codecs/wm9090.c index 228d782..2b5252c 100644 --- a/sound/soc/codecs/wm9090.c +++ b/sound/soc/codecs/wm9090.c @@ -548,7 +548,6 @@ static int wm9090_set_bias_level(struct snd_soc_codec *codec, static int wm9090_probe(struct snd_soc_codec *codec) { - struct wm9090_priv *wm9090 = snd_soc_codec_get_drvdata(codec); int ret; ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_I2C); -- cgit v1.1 From 9a185b9abacb7924b79e76a7a410de202aaf505b Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 6 Oct 2011 11:10:01 +0100 Subject: ASoC: Remove references to linux@wolfsonmicro.com Signed-off-by: Mark Brown --- sound/soc/codecs/lm4857.c | 2 +- sound/soc/codecs/wm8711.c | 2 +- sound/soc/codecs/wm8974.c | 2 +- sound/soc/codecs/wm8991.c | 2 +- 4 files changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/lm4857.c b/sound/soc/codecs/lm4857.c index 2c2a681..c387daf 100644 --- a/sound/soc/codecs/lm4857.c +++ b/sound/soc/codecs/lm4857.c @@ -3,7 +3,7 @@ * * Copyright 2007 Wolfson Microelectronics PLC. * Author: Graeme Gregory - * graeme.gregory@wolfsonmicro.com or linux@wolfsonmicro.com + * graeme.gregory@wolfsonmicro.com * Copyright 2011 Lars-Peter Clausen * * This program is free software; you can redistribute it and/or modify it diff --git a/sound/soc/codecs/wm8711.c b/sound/soc/codecs/wm8711.c index 47c7fd5..7475428 100644 --- a/sound/soc/codecs/wm8711.c +++ b/sound/soc/codecs/wm8711.c @@ -3,7 +3,7 @@ * * Copyright 2006 Wolfson Microelectronics * - * Author: Mike Arthur + * Author: Mike Arthur * * Based on wm8731.c by Richard Purdie * diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c index ca646a8..8e3bfc1 100644 --- a/sound/soc/codecs/wm8974.c +++ b/sound/soc/codecs/wm8974.c @@ -3,7 +3,7 @@ * * Copyright 2006-2009 Wolfson Microelectronics PLC. * - * Author: Liam Girdwood + * Author: Liam Girdwood * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License version 2 as diff --git a/sound/soc/codecs/wm8991.c b/sound/soc/codecs/wm8991.c index 6af23d0..08d64a6 100644 --- a/sound/soc/codecs/wm8991.c +++ b/sound/soc/codecs/wm8991.c @@ -3,7 +3,7 @@ * * Copyright 2007-2010 Wolfson Microelectronics PLC. * Author: Graeme Gregory - * linux@wolfsonmicro.com + * Graeme.Gregory@wolfsonmicro.com * * This program is free software; you can redistribute it and/or modify it * under the terms of the GNU General Public License as published by the -- cgit v1.1 From c4671a95857800941cb5aa6405170f3a91e448b4 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 6 Oct 2011 09:59:12 +0300 Subject: ASoC: Replace remaining use of *_volsw_2r with *_volsw The snd_soc_*_volsw_2r functionality has been merged to *volsw callbacks. Few places still used the get, or put variant of volsw_2r, replace those with the corresponding *_volsw. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/wm8350.c | 4 ++-- sound/soc/codecs/wm8580.c | 8 ++++---- sound/soc/codecs/wm_hubs.c | 4 ++-- 3 files changed, 8 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index 50ea9d7..35f3ad83 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -355,7 +355,7 @@ static int wm8350_put_volsw_2r_vu(struct snd_kcontrol *kcontrol, return 1; } - ret = snd_soc_put_volsw_2r(kcontrol, ucontrol); + ret = snd_soc_put_volsw(kcontrol, ucontrol); if (ret < 0) return ret; @@ -392,7 +392,7 @@ static int wm8350_get_volsw_2r(struct snd_kcontrol *kcontrol, break; } - return snd_soc_get_volsw_2r(kcontrol, ucontrol); + return snd_soc_get_volsw(kcontrol, ucontrol); } static const char *wm8350_deemp[] = { "None", "32kHz", "44.1kHz", "48kHz" }; diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index 02cbf13..b256727 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -213,7 +213,7 @@ static int wm8580_out_vu(struct snd_kcontrol *kcontrol, reg_cache[reg] = 0; reg_cache[reg2] = 0; - ret = snd_soc_put_volsw_2r(kcontrol, ucontrol); + ret = snd_soc_put_volsw(kcontrol, ucontrol); if (ret < 0) return ret; @@ -228,15 +228,15 @@ static const struct snd_kcontrol_new wm8580_snd_controls[] = { SOC_DOUBLE_R_EXT_TLV("DAC1 Playback Volume", WM8580_DIGITAL_ATTENUATION_DACL1, WM8580_DIGITAL_ATTENUATION_DACR1, - 0, 0xff, 0, snd_soc_get_volsw_2r, wm8580_out_vu, dac_tlv), + 0, 0xff, 0, snd_soc_get_volsw, wm8580_out_vu, dac_tlv), SOC_DOUBLE_R_EXT_TLV("DAC2 Playback Volume", WM8580_DIGITAL_ATTENUATION_DACL2, WM8580_DIGITAL_ATTENUATION_DACR2, - 0, 0xff, 0, snd_soc_get_volsw_2r, wm8580_out_vu, dac_tlv), + 0, 0xff, 0, snd_soc_get_volsw, wm8580_out_vu, dac_tlv), SOC_DOUBLE_R_EXT_TLV("DAC3 Playback Volume", WM8580_DIGITAL_ATTENUATION_DACL3, WM8580_DIGITAL_ATTENUATION_DACR3, - 0, 0xff, 0, snd_soc_get_volsw_2r, wm8580_out_vu, dac_tlv), + 0, 0xff, 0, snd_soc_get_volsw, wm8580_out_vu, dac_tlv), SOC_SINGLE("DAC1 Deemphasis Switch", WM8580_DAC_CONTROL3, 0, 1, 0), SOC_SINGLE("DAC2 Deemphasis Switch", WM8580_DAC_CONTROL3, 1, 1, 0), diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index f3583a5..84f33d4 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -222,7 +222,7 @@ static int wm8993_put_dc_servo(struct snd_kcontrol *kcontrol, struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec); int ret; - ret = snd_soc_put_volsw_2r(kcontrol, ucontrol); + ret = snd_soc_put_volsw(kcontrol, ucontrol); /* Updating the analogue gains invalidates the DC servo cache */ hubs->class_w_dcs = 0; @@ -364,7 +364,7 @@ SOC_ENUM("Speaker Mode", speaker_mode), SOC_DOUBLE_R_EXT_TLV("Headphone Volume", WM8993_LEFT_OUTPUT_VOLUME, WM8993_RIGHT_OUTPUT_VOLUME, - 0, 63, 0, snd_soc_get_volsw_2r, wm8993_put_dc_servo, + 0, 63, 0, snd_soc_get_volsw, wm8993_put_dc_servo, outpga_tlv), SOC_DOUBLE_R("Headphone Switch", WM8993_LEFT_OUTPUT_VOLUME, -- cgit v1.1 From 143d62a45b5f976067a8d705f7fae26a402651f9 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 6 Oct 2011 13:30:55 +0100 Subject: ASoC: Ensure DAPM widgets are set up before we sync jacks We synchronise jack state on startup - when we do that make sure that we have set up all the DAPM widgets first in case we end up touching any of the partially set up widgets when syncing the jack pins. Signed-off-by: Mark Brown Tested-by: Peter Ujfalusi --- sound/soc/soc-jack.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index fa31d9c..52db966 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -188,6 +188,8 @@ int snd_soc_jack_add_pins(struct snd_soc_jack *jack, int count, list_add(&(pins[i].list), &jack->pins); } + snd_soc_dapm_new_widgets(&jack->codec->card->dapm); + /* Update to reflect the last reported status; canned jack * implementations are likely to set their state before the * card has an opportunity to associate pins. -- cgit v1.1 From 23524eb16ace864c18a57ca035c76793a3c3eb65 Mon Sep 17 00:00:00 2001 From: Wolfram Sang Date: Thu, 6 Oct 2011 20:53:36 +0200 Subject: ASoC: tlv320aic32x4 fix initialization of micpga routing Checking the pdata-flags used 'or', so the check is always true. Use 'and' to correctly mask the flags. Signed-off-by: Wolfram Sang Cc: Javier Martin Cc: Liam Girdwood Cc: Mark Brown Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic32x4.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index e93b9d1..a68982e 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -685,10 +685,10 @@ static int aic32x4_probe(struct snd_soc_codec *codec) } /* Mic PGA routing */ - if (aic32x4->micpga_routing | AIC32X4_MICPGA_ROUTE_LMIC_IN2R_10K) { + if (aic32x4->micpga_routing & AIC32X4_MICPGA_ROUTE_LMIC_IN2R_10K) { snd_soc_write(codec, AIC32X4_LMICPGANIN, AIC32X4_LMICPGANIN_IN2R_10K); } - if (aic32x4->micpga_routing | AIC32X4_MICPGA_ROUTE_RMIC_IN1L_10K) { + if (aic32x4->micpga_routing & AIC32X4_MICPGA_ROUTE_RMIC_IN1L_10K) { snd_soc_write(codec, AIC32X4_RMICPGANIN, AIC32X4_RMICPGANIN_IN1L_10K); } -- cgit v1.1 From 416a0ce5f2338799f02fb41f6c56a6e490e4e8f0 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Thu, 6 Oct 2011 11:00:19 +0800 Subject: ASoC: wm8990: Convert to snd_soc_cache_sync for sync reg_cache with the hardware Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/wm8990.c | 32 +++++++++++++++++++------------- 1 file changed, 19 insertions(+), 13 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index 100aeee..48e9dd9 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -36,6 +36,17 @@ struct wm8990_priv { unsigned int pcmclk; }; +static int wm8990_volatile_register(struct snd_soc_codec *codec, + unsigned int reg) +{ + switch (reg) { + case WM8990_RESET: + return 1; + default: + return 0; + } +} + /* * wm8990 register cache. Note that register 0 is not included in the * cache. @@ -1156,6 +1167,7 @@ static int wm8990_mute(struct snd_soc_dai *dai, int mute) static int wm8990_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { + int ret; u16 val; switch (level) { @@ -1171,6 +1183,12 @@ static int wm8990_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_STANDBY: if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { + ret = snd_soc_cache_sync(codec); + if (ret < 0) { + dev_err(codec->dev, "Failed to sync cache: %d\n", ret); + return ret; + } + /* Enable all output discharge bits */ snd_soc_write(codec, WM8990_ANTIPOP1, WM8990_DIS_LLINE | WM8990_DIS_RLINE | WM8990_DIS_OUT3 | @@ -1319,19 +1337,6 @@ static int wm8990_suspend(struct snd_soc_codec *codec, pm_message_t state) static int wm8990_resume(struct snd_soc_codec *codec) { - int i; - u8 data[2]; - u16 *cache = codec->reg_cache; - - /* Sync reg_cache with the hardware */ - for (i = 0; i < ARRAY_SIZE(wm8990_reg); i++) { - if (i + 1 == WM8990_RESET) - continue; - data[0] = ((i + 1) << 1) | ((cache[i] >> 8) & 0x0001); - data[1] = cache[i] & 0x00ff; - codec->hw_write(codec->control_data, data, 2); - } - wm8990_set_bias_level(codec, SND_SOC_BIAS_STANDBY); return 0; } @@ -1392,6 +1397,7 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8990 = { .reg_cache_size = ARRAY_SIZE(wm8990_reg), .reg_word_size = sizeof(u16), .reg_cache_default = wm8990_reg, + .volatile_register = wm8990_volatile_register, }; #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) -- cgit v1.1 From 3c08600144f2a15fb3fba31b54cd6600371db6ef Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Thu, 6 Oct 2011 11:44:56 +0800 Subject: ASoC: wm8990: Remove incorrect comments Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/wm8990.c | 4 ---- 1 file changed, 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index 48e9dd9..ecdb8b2 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -47,10 +47,6 @@ static int wm8990_volatile_register(struct snd_soc_codec *codec, } } -/* - * wm8990 register cache. Note that register 0 is not included in the - * cache. - */ static const u16 wm8990_reg[] = { 0x8990, /* R0 - Reset */ 0x0000, /* R1 - Power Management (1) */ -- cgit v1.1 From ac60155f7afa3fd819befa35c2740a7a0d2f1a39 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Thu, 6 Oct 2011 07:29:56 +0800 Subject: ASoC: Return early with -EINVAL if invalid dai format is detected Signed-off-by: Axel Lin Acked-by: Timur Tabi Signed-off-by: Mark Brown --- sound/soc/codecs/cs4270.c | 8 ++++---- sound/soc/codecs/cs42l51.c | 9 ++++----- 2 files changed, 8 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 5830c93..f1f237e 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -261,7 +261,6 @@ static int cs4270_set_dai_fmt(struct snd_soc_dai *codec_dai, { struct snd_soc_codec *codec = codec_dai->codec; struct cs4270_private *cs4270 = snd_soc_codec_get_drvdata(codec); - int ret = 0; /* set DAI format */ switch (format & SND_SOC_DAIFMT_FORMAT_MASK) { @@ -271,7 +270,7 @@ static int cs4270_set_dai_fmt(struct snd_soc_dai *codec_dai, break; default: dev_err(codec->dev, "invalid dai format\n"); - ret = -EINVAL; + return -EINVAL; } /* set master/slave audio interface */ @@ -284,10 +283,11 @@ static int cs4270_set_dai_fmt(struct snd_soc_dai *codec_dai, break; default: /* all other modes are unsupported by the hardware */ - ret = -EINVAL; + dev_err(codec->dev, "Unknown master/slave configuration\n"); + return -EINVAL; } - return ret; + return 0; } /** diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index 286878d..8c3c820 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -288,7 +288,6 @@ static int cs42l51_set_dai_fmt(struct snd_soc_dai *codec_dai, { struct snd_soc_codec *codec = codec_dai->codec; struct cs42l51_private *cs42l51 = snd_soc_codec_get_drvdata(codec); - int ret = 0; switch (format & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: @@ -298,7 +297,7 @@ static int cs42l51_set_dai_fmt(struct snd_soc_dai *codec_dai, break; default: dev_err(codec->dev, "invalid DAI format\n"); - ret = -EINVAL; + return -EINVAL; } switch (format & SND_SOC_DAIFMT_MASTER_MASK) { @@ -309,11 +308,11 @@ static int cs42l51_set_dai_fmt(struct snd_soc_dai *codec_dai, cs42l51->func = MODE_SLAVE_AUTO; break; default: - ret = -EINVAL; - break; + dev_err(codec->dev, "Unknown master/slave configuration\n"); + return -EINVAL; } - return ret; + return 0; } struct cs42l51_ratios { -- cgit v1.1 From 4f4c0072228785179d35b2bd9e48081ce9fa51f6 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 7 Oct 2011 14:29:19 +0100 Subject: ASoC: Suppress early calls to snd_soc_dapm_sync() Ensure we only have one sync during the initial startup of the card by making snd_soc_dapm_sync() a noop on non-instantiated cards. This avoids any bounces due to things like jacks reporting their initial state on partially initialised cards. The callers that don't also get called at runtime should just be removed. Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 1 + sound/soc/soc-dapm.c | 7 +++++++ 2 files changed, 8 insertions(+) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 2a25076..b65e3d4 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1498,6 +1498,7 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) #endif card->instantiated = 1; + snd_soc_dapm_sync(&card->dapm); mutex_unlock(&card->mutex); return; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 8711aab..e49c56d 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1845,6 +1845,13 @@ static int snd_soc_dapm_set_pin(struct snd_soc_dapm_context *dapm, */ int snd_soc_dapm_sync(struct snd_soc_dapm_context *dapm) { + /* + * Suppress early reports (eg, jacks syncing their state) to avoid + * silly DAPM runs during card startup. + */ + if (!dapm->card || !dapm->card->instantiated) + return 0; + return dapm_power_widgets(dapm, SND_SOC_DAPM_STREAM_NOP); } EXPORT_SYMBOL_GPL(snd_soc_dapm_sync); -- cgit v1.1 From 87bea31c7b59a07fe5a1c827eb01db3b7c3ae672 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 8 Oct 2011 13:29:18 +0100 Subject: ASoC: Remove redundant snd_soc_dapm_sync() calls from machine drivers The core will sync DAPM as part of the card initialization, there is no need for machine drivers to do so during their setup. OMAP drivers are omitted as I know Peter already has patches for them. Signed-off-by: Mark Brown --- sound/soc/atmel/playpaq_wm8510.c | 1 - sound/soc/atmel/sam9g20_wm8731.c | 2 -- sound/soc/atmel/snd-soc-afeb9260.c | 2 -- sound/soc/davinci/davinci-evm.c | 2 -- sound/soc/kirkwood/kirkwood-t5325.c | 2 -- sound/soc/mid-x86/mfld_machine.c | 2 -- sound/soc/pxa/corgi.c | 1 - sound/soc/pxa/e740_wm9705.c | 2 -- sound/soc/pxa/e750_wm9705.c | 2 -- sound/soc/pxa/e800_wm9712.c | 1 - sound/soc/pxa/magician.c | 1 - sound/soc/pxa/mioa701_wm9713.c | 1 - sound/soc/pxa/palm27x.c | 4 ---- sound/soc/pxa/saarb.c | 4 ---- sound/soc/pxa/tavorevb3.c | 4 ---- sound/soc/pxa/tosa.c | 1 - sound/soc/pxa/z2.c | 4 ---- sound/soc/pxa/zylonite.c | 1 - sound/soc/samsung/goni_wm8994.c | 2 -- sound/soc/samsung/h1940_uda1380.c | 2 -- sound/soc/samsung/jive_wm8750.c | 1 - sound/soc/samsung/neo1973_wm8753.c | 4 ---- sound/soc/samsung/rx1950_uda1380.c | 2 -- sound/soc/samsung/s3c24xx_simtec_hermes.c | 1 - sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c | 1 - sound/soc/samsung/smartq_wm8987.c | 4 ---- sound/soc/samsung/smdk_wm8580.c | 6 ------ sound/soc/samsung/smdk_wm8994.c | 2 -- sound/soc/sh/sh7760-ac97.c | 7 ------- sound/soc/tegra/tegra_wm8903.c | 2 -- sound/soc/tegra/trimslice.c | 2 -- 31 files changed, 73 deletions(-) (limited to 'sound') diff --git a/sound/soc/atmel/playpaq_wm8510.c b/sound/soc/atmel/playpaq_wm8510.c index 2909bfa..73ae99a 100644 --- a/sound/soc/atmel/playpaq_wm8510.c +++ b/sound/soc/atmel/playpaq_wm8510.c @@ -338,7 +338,6 @@ static int playpaq_wm8510_init(struct snd_soc_pcm_runtime *rtd) /* always connected pins */ snd_soc_dapm_enable_pin(dapm, "Int Mic"); snd_soc_dapm_enable_pin(dapm, "Ext Spk"); - snd_soc_dapm_sync(dapm); diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c index bad3aa1..0377c54 100644 --- a/sound/soc/atmel/sam9g20_wm8731.c +++ b/sound/soc/atmel/sam9g20_wm8731.c @@ -173,8 +173,6 @@ static int at91sam9g20ek_wm8731_init(struct snd_soc_pcm_runtime *rtd) /* always connected */ snd_soc_dapm_enable_pin(dapm, "Ext Spk"); - snd_soc_dapm_sync(dapm); - return 0; } diff --git a/sound/soc/atmel/snd-soc-afeb9260.c b/sound/soc/atmel/snd-soc-afeb9260.c index 5e4d499..d427e92 100644 --- a/sound/soc/atmel/snd-soc-afeb9260.c +++ b/sound/soc/atmel/snd-soc-afeb9260.c @@ -117,8 +117,6 @@ static int afeb9260_tlv320aic23_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_enable_pin(dapm, "Line In"); snd_soc_dapm_enable_pin(dapm, "Mic Jack"); - snd_soc_dapm_sync(dapm); - return 0; } diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index fe79842..f78c3f0 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -150,8 +150,6 @@ static int evm_aic3x_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_enable_pin(dapm, "Mic Jack"); snd_soc_dapm_enable_pin(dapm, "Line In"); - snd_soc_dapm_sync(dapm); - return 0; } diff --git a/sound/soc/kirkwood/kirkwood-t5325.c b/sound/soc/kirkwood/kirkwood-t5325.c index c8d2195..c772b3c 100644 --- a/sound/soc/kirkwood/kirkwood-t5325.c +++ b/sound/soc/kirkwood/kirkwood-t5325.c @@ -79,8 +79,6 @@ static int t5325_dai_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_enable_pin(dapm, "Headphone Jack"); snd_soc_dapm_enable_pin(dapm, "Speaker"); - snd_soc_dapm_sync(dapm); - return 0; } diff --git a/sound/soc/mid-x86/mfld_machine.c b/sound/soc/mid-x86/mfld_machine.c index 1d818dc..598f48c 100644 --- a/sound/soc/mid-x86/mfld_machine.c +++ b/sound/soc/mid-x86/mfld_machine.c @@ -233,7 +233,6 @@ static int mfld_init(struct snd_soc_pcm_runtime *runtime) /* always connected */ snd_soc_dapm_enable_pin(dapm, "Headphones"); snd_soc_dapm_enable_pin(dapm, "Mic"); - snd_soc_dapm_sync(dapm); ret_val = snd_soc_add_controls(codec, mfld_snd_controls, ARRAY_SIZE(mfld_snd_controls)); @@ -251,7 +250,6 @@ static int mfld_init(struct snd_soc_pcm_runtime *runtime) /* we dont use linein in this so set to NC */ snd_soc_dapm_disable_pin(dapm, "LINEINL"); snd_soc_dapm_disable_pin(dapm, "LINEINR"); - snd_soc_dapm_sync(dapm); /* Headset and button jack detection */ ret_val = snd_soc_jack_new(codec, "Intel(R) MID Audio Jack", diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c index 28757fb..b0e2fb7 100644 --- a/sound/soc/pxa/corgi.c +++ b/sound/soc/pxa/corgi.c @@ -299,7 +299,6 @@ static int corgi_wm8731_init(struct snd_soc_pcm_runtime *rtd) /* Set up corgi specific audio path audio_map */ snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_sync(dapm); return 0; } diff --git a/sound/soc/pxa/e740_wm9705.c b/sound/soc/pxa/e740_wm9705.c index dc65650..35ed7eb 100644 --- a/sound/soc/pxa/e740_wm9705.c +++ b/sound/soc/pxa/e740_wm9705.c @@ -108,8 +108,6 @@ static int e740_ac97_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_sync(dapm); - return 0; } diff --git a/sound/soc/pxa/e750_wm9705.c b/sound/soc/pxa/e750_wm9705.c index 51897fc..ce5f0560 100644 --- a/sound/soc/pxa/e750_wm9705.c +++ b/sound/soc/pxa/e750_wm9705.c @@ -90,8 +90,6 @@ static int e750_ac97_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_sync(dapm); - return 0; } diff --git a/sound/soc/pxa/e800_wm9712.c b/sound/soc/pxa/e800_wm9712.c index 053ed20..6a8f38b 100644 --- a/sound/soc/pxa/e800_wm9712.c +++ b/sound/soc/pxa/e800_wm9712.c @@ -80,7 +80,6 @@ static int e800_ac97_init(struct snd_soc_pcm_runtime *rtd) ARRAY_SIZE(e800_dapm_widgets)); snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_sync(dapm); return 0; } diff --git a/sound/soc/pxa/magician.c b/sound/soc/pxa/magician.c index 098e5f4..e79f516 100644 --- a/sound/soc/pxa/magician.c +++ b/sound/soc/pxa/magician.c @@ -423,7 +423,6 @@ static int magician_uda1380_init(struct snd_soc_pcm_runtime *rtd) /* Set up magician specific audio path interconnects */ snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_sync(dapm); return 0; } diff --git a/sound/soc/pxa/mioa701_wm9713.c b/sound/soc/pxa/mioa701_wm9713.c index 38ca675..0b8d1ee 100644 --- a/sound/soc/pxa/mioa701_wm9713.c +++ b/sound/soc/pxa/mioa701_wm9713.c @@ -151,7 +151,6 @@ static int mioa701_wm9713_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_enable_pin(dapm, "Front Mic"); snd_soc_dapm_enable_pin(dapm, "GSM Line In"); snd_soc_dapm_enable_pin(dapm, "GSM Line Out"); - snd_soc_dapm_sync(dapm); return 0; } diff --git a/sound/soc/pxa/palm27x.c b/sound/soc/pxa/palm27x.c index 504e400..7edc1fb 100644 --- a/sound/soc/pxa/palm27x.c +++ b/sound/soc/pxa/palm27x.c @@ -107,10 +107,6 @@ static int palm27x_ac97_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_nc_pin(dapm, "PHONE"); snd_soc_dapm_nc_pin(dapm, "MIC2"); - err = snd_soc_dapm_sync(dapm); - if (err) - return err; - /* Jack detection API stuff */ err = snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE, &hs_jack); diff --git a/sound/soc/pxa/saarb.c b/sound/soc/pxa/saarb.c index 9595189..d9467a2 100644 --- a/sound/soc/pxa/saarb.c +++ b/sound/soc/pxa/saarb.c @@ -146,10 +146,6 @@ static int saarb_pm860x_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_disable_pin(dapm, "Headset Mic 2"); snd_soc_dapm_disable_pin(dapm, "Headset Stereophone"); - ret = snd_soc_dapm_sync(dapm); - if (ret) - return ret; - /* Headset jack detection */ snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE | SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2, diff --git a/sound/soc/pxa/tavorevb3.c b/sound/soc/pxa/tavorevb3.c index f881f65..eeec892 100644 --- a/sound/soc/pxa/tavorevb3.c +++ b/sound/soc/pxa/tavorevb3.c @@ -146,10 +146,6 @@ static int evb3_pm860x_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_disable_pin(dapm, "Headset Mic 2"); snd_soc_dapm_disable_pin(dapm, "Headset Stereophone"); - ret = snd_soc_dapm_sync(dapm); - if (ret) - return ret; - /* Headset jack detection */ snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE | SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2, diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c index 9a23513..620fc69 100644 --- a/sound/soc/pxa/tosa.c +++ b/sound/soc/pxa/tosa.c @@ -211,7 +211,6 @@ static int tosa_ac97_init(struct snd_soc_pcm_runtime *rtd) /* set up tosa specific audio path audio_map */ snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_sync(dapm); return 0; } diff --git a/sound/soc/pxa/z2.c b/sound/soc/pxa/z2.c index 4b81ffd..b311ffe 100644 --- a/sound/soc/pxa/z2.c +++ b/sound/soc/pxa/z2.c @@ -161,10 +161,6 @@ static int z2_wm8750_init(struct snd_soc_pcm_runtime *rtd) /* Set up z2 specific audio paths */ snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - ret = snd_soc_dapm_sync(dapm); - if (ret) - goto err; - /* Jack detection API stuff */ ret = snd_soc_jack_new(codec, "Headset Jack", SND_JACK_HEADSET, &hs_jack); diff --git a/sound/soc/pxa/zylonite.c b/sound/soc/pxa/zylonite.c index 2b8350b..580aae3 100644 --- a/sound/soc/pxa/zylonite.c +++ b/sound/soc/pxa/zylonite.c @@ -87,7 +87,6 @@ static int zylonite_wm9713_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_enable_pin(dapm, "Headphone"); snd_soc_dapm_enable_pin(dapm, "Headset Earpiece"); - snd_soc_dapm_sync(dapm); return 0; } diff --git a/sound/soc/samsung/goni_wm8994.c b/sound/soc/samsung/goni_wm8994.c index eb6d72e..86ece1a 100644 --- a/sound/soc/samsung/goni_wm8994.c +++ b/sound/soc/samsung/goni_wm8994.c @@ -120,8 +120,6 @@ static int goni_wm8994_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_nc_pin(dapm, "SPKOUTRP"); } - snd_soc_dapm_sync(dapm); - /* Headset jack detection */ ret = snd_soc_jack_new(codec, "Headset Jack", SND_JACK_HEADSET | SND_JACK_MECHANICAL | SND_JACK_AVOUT, diff --git a/sound/soc/samsung/h1940_uda1380.c b/sound/soc/samsung/h1940_uda1380.c index c6c6589..52025c9 100644 --- a/sound/soc/samsung/h1940_uda1380.c +++ b/sound/soc/samsung/h1940_uda1380.c @@ -198,8 +198,6 @@ static int h1940_uda1380_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_enable_pin(dapm, "Speaker"); snd_soc_dapm_enable_pin(dapm, "Mic Jack"); - snd_soc_dapm_sync(dapm); - snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE, &hp_jack); diff --git a/sound/soc/samsung/jive_wm8750.c b/sound/soc/samsung/jive_wm8750.c index ed8f13a..bb69e9f 100644 --- a/sound/soc/samsung/jive_wm8750.c +++ b/sound/soc/samsung/jive_wm8750.c @@ -120,7 +120,6 @@ static int jive_wm8750_init(struct snd_soc_pcm_runtime *rtd) } snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_sync(dapm); return 0; } diff --git a/sound/soc/samsung/neo1973_wm8753.c b/sound/soc/samsung/neo1973_wm8753.c index 16152ed..7207189 100644 --- a/sound/soc/samsung/neo1973_wm8753.c +++ b/sound/soc/samsung/neo1973_wm8753.c @@ -367,8 +367,6 @@ static int neo1973_wm8753_init(struct snd_soc_pcm_runtime *rtd) return ret; } - snd_soc_dapm_sync(dapm); - return 0; } @@ -409,8 +407,6 @@ static int neo1973_lm4857_init(struct snd_soc_dapm_context *dapm) snd_soc_dapm_ignore_suspend(dapm, "Handset Spk"); snd_soc_dapm_ignore_suspend(dapm, "Headphone"); - snd_soc_dapm_sync(dapm); - return 0; } diff --git a/sound/soc/samsung/rx1950_uda1380.c b/sound/soc/samsung/rx1950_uda1380.c index bc8c167..e5ad67f 100644 --- a/sound/soc/samsung/rx1950_uda1380.c +++ b/sound/soc/samsung/rx1950_uda1380.c @@ -238,8 +238,6 @@ static int rx1950_uda1380_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_enable_pin(dapm, "Speaker"); snd_soc_dapm_enable_pin(dapm, "Mic Jack"); - snd_soc_dapm_sync(dapm); - snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE, &hp_jack); diff --git a/sound/soc/samsung/s3c24xx_simtec_hermes.c b/sound/soc/samsung/s3c24xx_simtec_hermes.c index ce6aef6..359530b 100644 --- a/sound/soc/samsung/s3c24xx_simtec_hermes.c +++ b/sound/soc/samsung/s3c24xx_simtec_hermes.c @@ -76,7 +76,6 @@ static int simtec_hermes_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_enable_pin(dapm, "Mic Jack"); simtec_audio_init(rtd); - snd_soc_dapm_sync(dapm); return 0; } diff --git a/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c b/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c index a7ef7db..8bad349 100644 --- a/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c +++ b/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c @@ -65,7 +65,6 @@ static int simtec_tlv320aic23_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_enable_pin(dapm, "Mic Jack"); simtec_audio_init(rtd); - snd_soc_dapm_sync(dapm); return 0; } diff --git a/sound/soc/samsung/smartq_wm8987.c b/sound/soc/samsung/smartq_wm8987.c index bbd1476..6db4e55 100644 --- a/sound/soc/samsung/smartq_wm8987.c +++ b/sound/soc/samsung/smartq_wm8987.c @@ -178,10 +178,6 @@ static int smartq_wm8987_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_enable_pin(dapm, "Internal Mic"); snd_soc_dapm_disable_pin(dapm, "Headphone Jack"); - err = snd_soc_dapm_sync(dapm); - if (err) - return err; - /* Headphone jack detection */ err = snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE, &smartq_jack); diff --git a/sound/soc/samsung/smdk_wm8580.c b/sound/soc/samsung/smdk_wm8580.c index 20deecf..54c4cce 100644 --- a/sound/soc/samsung/smdk_wm8580.c +++ b/sound/soc/samsung/smdk_wm8580.c @@ -173,9 +173,6 @@ static int smdk_wm8580_init_paiftx(struct snd_soc_pcm_runtime *rtd) */ snd_soc_dapm_disable_pin(dapm, "MicIn"); - /* signal a DAPM event */ - snd_soc_dapm_sync(dapm); - return 0; } @@ -191,9 +188,6 @@ static int smdk_wm8580_init_paifrx(struct snd_soc_pcm_runtime *rtd) /* Set up PAIFRX audio path */ snd_soc_dapm_add_routes(dapm, audio_map_rx, ARRAY_SIZE(audio_map_rx)); - /* signal a DAPM event */ - snd_soc_dapm_sync(dapm); - return 0; } diff --git a/sound/soc/samsung/smdk_wm8994.c b/sound/soc/samsung/smdk_wm8994.c index 45fbe2b..f75e439 100644 --- a/sound/soc/samsung/smdk_wm8994.c +++ b/sound/soc/samsung/smdk_wm8994.c @@ -117,8 +117,6 @@ static int smdk_wm8994_init_paiftx(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_nc_pin(dapm, "IN1RP"); snd_soc_dapm_nc_pin(dapm, "IN2RP:VXRP"); - snd_soc_dapm_sync(dapm); - return 0; } diff --git a/sound/soc/sh/sh7760-ac97.c b/sound/soc/sh/sh7760-ac97.c index 917d3ce..c62ae68 100644 --- a/sound/soc/sh/sh7760-ac97.c +++ b/sound/soc/sh/sh7760-ac97.c @@ -20,12 +20,6 @@ extern struct snd_soc_dai_driver sh4_hac_dai[2]; extern struct snd_soc_platform_driver sh7760_soc_platform; -static int machine_init(struct snd_soc_pcm_runtime *rtd) -{ - snd_soc_dapm_sync(&rtd->codec->dapm); - return 0; -} - static struct snd_soc_dai_link sh7760_ac97_dai = { .name = "AC97", .stream_name = "AC97 HiFi", @@ -33,7 +27,6 @@ static struct snd_soc_dai_link sh7760_ac97_dai = { .codec_dai_name = "ac97-hifi", .platform_name = "sh7760-pcm-audio", .codec_name = "ac97-codec", - .init = machine_init, .ops = NULL, }; diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c index be27f1d..a81cf39 100644 --- a/sound/soc/tegra/tegra_wm8903.c +++ b/sound/soc/tegra/tegra_wm8903.c @@ -339,8 +339,6 @@ static int tegra_wm8903_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_nc_pin(dapm, "LINEOUTL"); } - snd_soc_dapm_sync(dapm); - return 0; } diff --git a/sound/soc/tegra/trimslice.c b/sound/soc/tegra/trimslice.c index 8fc07e9..b3a7efa 100644 --- a/sound/soc/tegra/trimslice.c +++ b/sound/soc/tegra/trimslice.c @@ -124,8 +124,6 @@ static int trimslice_asoc_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_nc_pin(dapm, "RHPOUT"); snd_soc_dapm_nc_pin(dapm, "MICIN"); - snd_soc_dapm_sync(dapm); - return 0; } -- cgit v1.1 From 2dc00213b03669010a67454d6448d91f3af06435 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 8 Oct 2011 13:59:44 +0100 Subject: ASoC: Ensure all DAPM widgets are instantiated with the card Specifically for the widgets added by machine driver late probe functions. Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index b65e3d4..8dc9aba 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1478,6 +1478,8 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) } } + snd_soc_dapm_new_widgets(&card->dapm); + ret = snd_card_register(card->snd_card); if (ret < 0) { printk(KERN_ERR "asoc: failed to register soundcard for %s\n", card->name); -- cgit v1.1 From 7ca3a18b055ac6667f4e7e34eae6637270002402 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 8 Oct 2011 14:04:50 +0100 Subject: ASoC: Assign power_check when we allocate DAPM widgets This ensures none of the rest of the code ever encounters a widget which does not have a power check function. Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 67 ++++++++++++++++++++++++++++++---------------------- 1 file changed, 39 insertions(+), 28 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index e49c56d..22fb735 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2137,48 +2137,21 @@ int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm) case snd_soc_dapm_switch: case snd_soc_dapm_mixer: case snd_soc_dapm_mixer_named_ctl: - w->power_check = dapm_generic_check_power; dapm_new_mixer(w); break; case snd_soc_dapm_mux: case snd_soc_dapm_virt_mux: case snd_soc_dapm_value_mux: - w->power_check = dapm_generic_check_power; dapm_new_mux(w); break; - case snd_soc_dapm_adc: - case snd_soc_dapm_aif_out: - w->power_check = dapm_adc_check_power; - break; - case snd_soc_dapm_dac: - case snd_soc_dapm_aif_in: - w->power_check = dapm_dac_check_power; - break; case snd_soc_dapm_pga: case snd_soc_dapm_out_drv: - w->power_check = dapm_generic_check_power; dapm_new_pga(w); break; - case snd_soc_dapm_input: - case snd_soc_dapm_output: - case snd_soc_dapm_micbias: - case snd_soc_dapm_spk: - case snd_soc_dapm_hp: - case snd_soc_dapm_mic: - case snd_soc_dapm_line: - w->power_check = dapm_generic_check_power; - break; - case snd_soc_dapm_supply: - w->power_check = dapm_supply_check_power; - case snd_soc_dapm_vmid: - case snd_soc_dapm_pre: - case snd_soc_dapm_post: + default: break; } - if (!w->power_check) - w->power_check = dapm_always_on_check_power; - /* Read the initial power state from the device */ if (w->reg >= 0) { val = soc_widget_read(w, w->reg); @@ -2667,6 +2640,44 @@ int snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, else snprintf(w->name, name_len, "%s", widget->name); + switch (w->id) { + case snd_soc_dapm_switch: + case snd_soc_dapm_mixer: + case snd_soc_dapm_mixer_named_ctl: + w->power_check = dapm_generic_check_power; + break; + case snd_soc_dapm_mux: + case snd_soc_dapm_virt_mux: + case snd_soc_dapm_value_mux: + w->power_check = dapm_generic_check_power; + break; + case snd_soc_dapm_adc: + case snd_soc_dapm_aif_out: + w->power_check = dapm_adc_check_power; + break; + case snd_soc_dapm_dac: + case snd_soc_dapm_aif_in: + w->power_check = dapm_dac_check_power; + break; + case snd_soc_dapm_pga: + case snd_soc_dapm_out_drv: + case snd_soc_dapm_input: + case snd_soc_dapm_output: + case snd_soc_dapm_micbias: + case snd_soc_dapm_spk: + case snd_soc_dapm_hp: + case snd_soc_dapm_mic: + case snd_soc_dapm_line: + w->power_check = dapm_generic_check_power; + break; + case snd_soc_dapm_supply: + w->power_check = dapm_supply_check_power; + break; + default: + w->power_check = dapm_always_on_check_power; + break; + } + dapm->n_widgets++; w->dapm = dapm; w->codec = dapm->codec; -- cgit v1.1 From 024dc078558e64e4cebc62c096285430a61dd10e Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sun, 9 Oct 2011 11:52:05 +0100 Subject: ASoC: Cache connected input and output recursions The number of connected input and output endpoints for a given widgets can't change during a DAPM run so there is no need to redo the recursion through branches of the tree we've already visited. Doing this on one of my test systems gives an improvement of: Power Path Neighbour Before: 63 607 731 After: 63 141 181 which scales up well as more widgets are involved in paths. Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 60 +++++++++++++++++++++++++++++++++++++++------------- 1 file changed, 45 insertions(+), 15 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 22fb735..258326b 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -665,6 +665,9 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget) struct snd_soc_dapm_path *path; int con = 0; + if (widget->outputs >= 0) + return widget->outputs; + DAPM_UPDATE_STAT(widget, path_checks); if (widget->id == snd_soc_dapm_supply) @@ -673,21 +676,29 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget) switch (widget->id) { case snd_soc_dapm_adc: case snd_soc_dapm_aif_out: - if (widget->active) - return snd_soc_dapm_suspend_check(widget); + if (widget->active) { + widget->outputs = snd_soc_dapm_suspend_check(widget); + return widget->outputs; + } default: break; } if (widget->connected) { /* connected pin ? */ - if (widget->id == snd_soc_dapm_output && !widget->ext) - return snd_soc_dapm_suspend_check(widget); + if (widget->id == snd_soc_dapm_output && !widget->ext) { + widget->outputs = snd_soc_dapm_suspend_check(widget); + return widget->outputs; + } /* connected jack or spk ? */ - if (widget->id == snd_soc_dapm_hp || widget->id == snd_soc_dapm_spk || - (widget->id == snd_soc_dapm_line && !list_empty(&widget->sources))) - return snd_soc_dapm_suspend_check(widget); + if (widget->id == snd_soc_dapm_hp || + widget->id == snd_soc_dapm_spk || + (widget->id == snd_soc_dapm_line && + !list_empty(&widget->sources))) { + widget->outputs = snd_soc_dapm_suspend_check(widget); + return widget->outputs; + } } list_for_each_entry(path, &widget->sinks, list_source) { @@ -705,6 +716,8 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget) } } + widget->outputs = con; + return con; } @@ -717,6 +730,9 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget) struct snd_soc_dapm_path *path; int con = 0; + if (widget->inputs >= 0) + return widget->inputs; + DAPM_UPDATE_STAT(widget, path_checks); if (widget->id == snd_soc_dapm_supply) @@ -726,25 +742,35 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget) switch (widget->id) { case snd_soc_dapm_dac: case snd_soc_dapm_aif_in: - if (widget->active) - return snd_soc_dapm_suspend_check(widget); + if (widget->active) { + widget->inputs = snd_soc_dapm_suspend_check(widget); + return widget->inputs; + } default: break; } if (widget->connected) { /* connected pin ? */ - if (widget->id == snd_soc_dapm_input && !widget->ext) - return snd_soc_dapm_suspend_check(widget); + if (widget->id == snd_soc_dapm_input && !widget->ext) { + widget->inputs = snd_soc_dapm_suspend_check(widget); + return widget->inputs; + } /* connected VMID/Bias for lower pops */ - if (widget->id == snd_soc_dapm_vmid) - return snd_soc_dapm_suspend_check(widget); + if (widget->id == snd_soc_dapm_vmid) { + widget->inputs = snd_soc_dapm_suspend_check(widget); + return widget->inputs; + } /* connected jack ? */ if (widget->id == snd_soc_dapm_mic || - (widget->id == snd_soc_dapm_line && !list_empty(&widget->sinks))) - return snd_soc_dapm_suspend_check(widget); + (widget->id == snd_soc_dapm_line && + !list_empty(&widget->sinks))) { + widget->inputs = snd_soc_dapm_suspend_check(widget); + return widget->inputs; + } + } list_for_each_entry(path, &widget->sources, list_sink) { @@ -762,6 +788,8 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget) } } + widget->inputs = con; + return con; } @@ -1335,6 +1363,8 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event) list_for_each_entry(w, &card->widgets, list) { w->power_checked = false; + w->inputs = -1; + w->outputs = -1; } /* Check which widgets we need to power and store them in -- cgit v1.1 From 3ebb5c9b1056b7eaae3e5dd11b97e2830797e51c Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sun, 9 Oct 2011 14:06:13 +0100 Subject: ASoC: Squash error codes from regmap down to -1 on read The ASoC code always uses -1 as the error code due to reporting errors in band with the value. Ensure we don't confuse anything by making sure we don't pass actual error codes back into the rest of the code on read. Signed-off-by: Mark Brown --- sound/soc/soc-io.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/soc-io.c b/sound/soc/soc-io.c index 66fcccd..dd89933 100644 --- a/sound/soc/soc-io.c +++ b/sound/soc/soc-io.c @@ -55,7 +55,7 @@ static unsigned int hw_read(struct snd_soc_codec *codec, unsigned int reg) if (ret == 0) return val; else - return ret; + return -1; } ret = snd_soc_cache_read(codec, reg, &val); -- cgit v1.1 From 25c77c5fae5e0ef43ab6381f89fc41e26d2ca0f4 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 8 Oct 2011 13:36:03 +0100 Subject: ASoC: Fix DAPM sync for TLV320AIC3x custom DAPM widget We really should be doing this in the core, not in a driver... Signed-off-by: Mark Brown Tested-by: Jarkko Nikula --- sound/soc/codecs/tlv320aic3x.c | 4 ++++ sound/soc/soc-dapm.c | 3 ++- 2 files changed, 6 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index d877b39..be55b7f 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -197,6 +197,10 @@ static int snd_soc_dapm_put_volsw_aic3x(struct snd_kcontrol *kcontrol, else /* old connection must be powered down */ path->connect = invert ? 1 : 0; + + dapm_mark_dirty(path->source, "tlv320aic3x source"); + dapm_mark_dirty(path->sink, "tlv320aic3x sink"); + break; } diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 258326b..f42e8b9 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -124,7 +124,7 @@ static bool dapm_dirty_widget(struct snd_soc_dapm_widget *w) return !list_empty(&w->dirty); } -static void dapm_mark_dirty(struct snd_soc_dapm_widget *w, const char *reason) +void dapm_mark_dirty(struct snd_soc_dapm_widget *w, const char *reason) { if (!dapm_dirty_widget(w)) { dev_vdbg(w->dapm->dev, "Marking %s dirty due to %s\n", @@ -132,6 +132,7 @@ static void dapm_mark_dirty(struct snd_soc_dapm_widget *w, const char *reason) list_add_tail(&w->dirty, &w->dapm->card->dapm_dirty); } } +EXPORT_SYMBOL_GPL(dapm_mark_dirty); /* create a new dapm widget */ static inline struct snd_soc_dapm_widget *dapm_cnew_widget( -- cgit v1.1 From 94f17e9cfa73d496c2289f02d2002465b79b0931 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Fri, 7 Oct 2011 21:36:27 +0800 Subject: ASoC: wm8510: Convert to snd_soc_cache_sync Convert to snd_soc_cache_sync for sync reg_cache with the hardware. Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/wm8510.c | 13 ++----------- 1 file changed, 2 insertions(+), 11 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c index 55a4c83..07c9cc7 100644 --- a/sound/soc/codecs/wm8510.c +++ b/sound/soc/codecs/wm8510.c @@ -480,6 +480,8 @@ static int wm8510_set_bias_level(struct snd_soc_codec *codec, power1 |= WM8510_POWER1_BIASEN | WM8510_POWER1_BUFIOEN; if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { + snd_soc_cache_sync(codec); + /* Initial cap charge at VMID 5k */ snd_soc_write(codec, WM8510_POWER1, power1 | 0x3); mdelay(100); @@ -541,18 +543,7 @@ static int wm8510_suspend(struct snd_soc_codec *codec, pm_message_t state) static int wm8510_resume(struct snd_soc_codec *codec) { - int i; - u8 data[2]; - u16 *cache = codec->reg_cache; - - /* Sync reg_cache with the hardware */ - for (i = 0; i < ARRAY_SIZE(wm8510_reg); i++) { - data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001); - data[1] = cache[i] & 0x00ff; - codec->hw_write(codec->control_data, data, 2); - } wm8510_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - return 0; } -- cgit v1.1 From 960622da0d0583637e5d2de85b4202cbfc0981c6 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Fri, 7 Oct 2011 21:37:54 +0800 Subject: ASoC: wm8711: Convert to snd_soc_cache_sync Convert to snd_soc_cache_sync for sync reg_cache with the hardware. Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/wm8711.c | 15 +++------------ 1 file changed, 3 insertions(+), 12 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8711.c b/sound/soc/codecs/wm8711.c index 7475428..8d0347c 100644 --- a/sound/soc/codecs/wm8711.c +++ b/sound/soc/codecs/wm8711.c @@ -287,7 +287,6 @@ static int wm8711_set_dai_fmt(struct snd_soc_dai *codec_dai, return 0; } - static int wm8711_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { @@ -300,6 +299,9 @@ static int wm8711_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_PREPARE: break; case SND_SOC_BIAS_STANDBY: + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) + snd_soc_cache_sync(codec); + snd_soc_write(codec, WM8711_PWR, reg | 0x0040); break; case SND_SOC_BIAS_OFF: @@ -346,18 +348,7 @@ static int wm8711_suspend(struct snd_soc_codec *codec, pm_message_t state) static int wm8711_resume(struct snd_soc_codec *codec) { - int i; - u8 data[2]; - u16 *cache = codec->reg_cache; - - /* Sync reg_cache with the hardware */ - for (i = 0; i < ARRAY_SIZE(wm8711_reg); i++) { - data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001); - data[1] = cache[i] & 0x00ff; - codec->hw_write(codec->control_data, data, 2); - } wm8711_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - return 0; } -- cgit v1.1 From 9bf311fe17f16effaf264af76cb3aaaf384556b3 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Fri, 7 Oct 2011 21:39:09 +0800 Subject: ASoC: wm8731: Convert to snd_soc_cache_sync Convert to snd_soc_cache_sync for sync reg_cache with the hardware. Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/wm8731.c | 15 ++------------- 1 file changed, 2 insertions(+), 13 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index f76b6fc..7e5ec03 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -427,9 +427,7 @@ static int wm8731_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { struct wm8731_priv *wm8731 = snd_soc_codec_get_drvdata(codec); - int i, ret; - u8 data[2]; - u16 *cache = codec->reg_cache; + int ret; u16 reg; switch (level) { @@ -444,16 +442,7 @@ static int wm8731_set_bias_level(struct snd_soc_codec *codec, if (ret != 0) return ret; - /* Sync reg_cache with the hardware */ - for (i = 0; i < ARRAY_SIZE(wm8731_reg); i++) { - if (cache[i] == wm8731_reg[i]) - continue; - - data[0] = (i << 1) | ((cache[i] >> 8) - & 0x0001); - data[1] = cache[i] & 0x00ff; - codec->hw_write(codec->control_data, data, 2); - } + snd_soc_cache_sync(codec); } /* Clear PWROFF, gate CLKOUT, everything else as-is */ -- cgit v1.1 From 4d4adfc9790da2a6f7382004451b73231c1d2ccf Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Fri, 7 Oct 2011 21:40:44 +0800 Subject: ASoC: wm8750: Convert to snd_soc_cache_sync Convert to snd_soc_cache_sync for sync reg_cache with the hardware. Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/wm8750.c | 16 ++-------------- 1 file changed, 2 insertions(+), 14 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index 862c520..ca75a81 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -616,6 +616,8 @@ static int wm8750_set_bias_level(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_STANDBY: if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { + snd_soc_cache_sync(codec); + /* Set VMID to 5k */ snd_soc_write(codec, WM8750_PWR1, pwr_reg | 0x01c1); @@ -673,21 +675,7 @@ static int wm8750_suspend(struct snd_soc_codec *codec, pm_message_t state) static int wm8750_resume(struct snd_soc_codec *codec) { - int i; - u8 data[2]; - u16 *cache = codec->reg_cache; - - /* Sync reg_cache with the hardware */ - for (i = 0; i < ARRAY_SIZE(wm8750_reg); i++) { - if (i == WM8750_RESET) - continue; - data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001); - data[1] = cache[i] & 0x00ff; - codec->hw_write(codec->control_data, data, 2); - } - wm8750_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - return 0; } -- cgit v1.1 From abe11d0aacc75eb400fc1c6e40b28703e481076e Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Fri, 7 Oct 2011 21:41:41 +0800 Subject: ASoC: wm8776: Convert to snd_soc_cache_sync Convert to snd_soc_cache_sync for sync reg_cache with the hardware. Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/wm8776.c | 16 ++-------------- 1 file changed, 2 insertions(+), 14 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8776.c b/sound/soc/codecs/wm8776.c index 00d8846..bfdc523 100644 --- a/sound/soc/codecs/wm8776.c +++ b/sound/soc/codecs/wm8776.c @@ -308,6 +308,8 @@ static int wm8776_set_bias_level(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_STANDBY: if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { + snd_soc_cache_sync(codec); + /* Disable the global powerdown; DAPM does the rest */ snd_soc_update_bits(codec, WM8776_PWRDOWN, 1, 0); } @@ -379,21 +381,7 @@ static int wm8776_suspend(struct snd_soc_codec *codec, pm_message_t state) static int wm8776_resume(struct snd_soc_codec *codec) { - int i; - u8 data[2]; - u16 *cache = codec->reg_cache; - - /* Sync reg_cache with the hardware */ - for (i = 0; i < ARRAY_SIZE(wm8776_reg); i++) { - if (cache[i] == wm8776_reg[i]) - continue; - data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001); - data[1] = cache[i] & 0x00ff; - codec->hw_write(codec->control_data, data, 2); - } - wm8776_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - return 0; } #else -- cgit v1.1 From 788b6e8efa052ab13fb6b9d957fbaf8e331008f9 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Fri, 7 Oct 2011 21:42:49 +0800 Subject: ASoC: wm8940: Convert to snd_soc_cache_sync Convert to snd_soc_cache_sync for sync reg_cache with the hardware. Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/wm8940.c | 46 ++++++++++++++++++++++------------------------ 1 file changed, 22 insertions(+), 24 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c index 7e0f54c..a4abfdf 100644 --- a/sound/soc/codecs/wm8940.c +++ b/sound/soc/codecs/wm8940.c @@ -45,6 +45,17 @@ struct wm8940_priv { enum snd_soc_control_type control_type; }; +static int wm8940_volatile_register(struct snd_soc_codec *codec, + unsigned int reg) +{ + switch (reg) { + case WM8940_SOFTRESET: + return 1; + default: + return 0; + } +} + static u16 wm8940_reg_defaults[] = { 0x8940, /* Soft Reset */ 0x0000, /* Power 1 */ @@ -459,6 +470,14 @@ static int wm8940_set_bias_level(struct snd_soc_codec *codec, ret = snd_soc_write(codec, WM8940_POWER1, pwr_reg | 0x1); break; case SND_SOC_BIAS_STANDBY: + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { + ret = snd_soc_cache_sync(codec); + if (ret < 0) { + dev_err(codec->dev, "Failed to sync cache: %d\n", ret); + return ret; + } + } + /* ensure bufioen and biasen */ pwr_reg |= (1 << 2) | (1 << 3); /* set vmid to 300k for standby */ @@ -659,30 +678,8 @@ static int wm8940_suspend(struct snd_soc_codec *codec, pm_message_t state) static int wm8940_resume(struct snd_soc_codec *codec) { - int i; - int ret; - u8 data[3]; - u16 *cache = codec->reg_cache; - - /* Sync reg_cache with the hardware - * Could use auto incremented writes to speed this up - */ - for (i = 0; i < ARRAY_SIZE(wm8940_reg_defaults); i++) { - data[0] = i; - data[1] = (cache[i] & 0xFF00) >> 8; - data[2] = cache[i] & 0x00FF; - ret = codec->hw_write(codec->control_data, data, 3); - if (ret < 0) - goto error_ret; - else if (ret != 3) { - ret = -EIO; - goto error_ret; - } - } - ret = wm8940_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - -error_ret: - return ret; + wm8940_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + return 0; } static int wm8940_probe(struct snd_soc_codec *codec) @@ -742,6 +739,7 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8940 = { .reg_cache_size = ARRAY_SIZE(wm8940_reg_defaults), .reg_word_size = sizeof(u16), .reg_cache_default = wm8940_reg_defaults, + .volatile_register = wm8940_volatile_register, }; #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) -- cgit v1.1 From bc45df2dd98cfd550f674c494965f0015d5f923e Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Fri, 7 Oct 2011 21:50:23 +0800 Subject: ASoC: wm8960: Convert to snd_soc_cache_sync Convert to snd_soc_cache_sync for sync reg_cache with the hardware. Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/wm8960.c | 15 +++++---------- 1 file changed, 5 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 831c20f..2df253c 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -574,6 +574,8 @@ static int wm8960_set_bias_level_out3(struct snd_soc_codec *codec, case SND_SOC_BIAS_STANDBY: if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { + snd_soc_cache_sync(codec); + /* Enable anti-pop features */ snd_soc_write(codec, WM8960_APOP1, WM8960_POBCTRL | WM8960_SOFT_ST | @@ -676,6 +678,9 @@ static int wm8960_set_bias_level_capless(struct snd_soc_codec *codec, WM8960_VREF | WM8960_VMID_MASK, 0); break; + case SND_SOC_BIAS_OFF: + snd_soc_cache_sync(codec); + break; default: break; } @@ -901,16 +906,6 @@ static int wm8960_suspend(struct snd_soc_codec *codec, pm_message_t state) static int wm8960_resume(struct snd_soc_codec *codec) { struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); - int i; - u8 data[2]; - u16 *cache = codec->reg_cache; - - /* Sync reg_cache with the hardware */ - for (i = 0; i < ARRAY_SIZE(wm8960_reg); i++) { - data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001); - data[1] = cache[i] & 0x00ff; - codec->hw_write(codec->control_data, data, 2); - } wm8960->set_bias_level(codec, SND_SOC_BIAS_STANDBY); return 0; -- cgit v1.1 From e46199ece4e0db886f90abe11d91b04601fd0300 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Fri, 7 Oct 2011 21:52:00 +0800 Subject: ASoC: wm8971: Convert to snd_soc_cache_sync Convert to snd_soc_cache_sync for sync reg_cache with the hardware. Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/wm8971.c | 15 +++------------ 1 file changed, 3 insertions(+), 12 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c index 08ea6f8..b444b29 100644 --- a/sound/soc/codecs/wm8971.c +++ b/sound/soc/codecs/wm8971.c @@ -546,6 +546,9 @@ static int wm8971_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_PREPARE: break; case SND_SOC_BIAS_STANDBY: + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) + snd_soc_cache_sync(codec); + /* mute dac and set vmid to 500k, enable VREF */ snd_soc_write(codec, WM8971_PWR1, pwr_reg | 0x0140); break; @@ -605,20 +608,8 @@ static int wm8971_suspend(struct snd_soc_codec *codec, pm_message_t state) static int wm8971_resume(struct snd_soc_codec *codec) { - int i; - u8 data[2]; - u16 *cache = codec->reg_cache; u16 reg; - /* Sync reg_cache with the hardware */ - for (i = 0; i < ARRAY_SIZE(wm8971_reg); i++) { - if (i == WM8971_RESET) - continue; - data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001); - data[1] = cache[i] & 0x00ff; - codec->hw_write(codec->control_data, data, 2); - } - wm8971_set_bias_level(codec, SND_SOC_BIAS_STANDBY); /* charge wm8971 caps */ -- cgit v1.1 From 0bad3d8453e60ed5093ffccc0dd906ffb9bfe62c Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Fri, 7 Oct 2011 21:52:42 +0800 Subject: ASoC: wm8974: Convert to snd_soc_cache_sync Convert to snd_soc_cache_sync for sync reg_cache with the hardware. Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/wm8974.c | 13 ++----------- 1 file changed, 2 insertions(+), 11 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c index 8e3bfc1..9352f1e 100644 --- a/sound/soc/codecs/wm8974.c +++ b/sound/soc/codecs/wm8974.c @@ -530,6 +530,8 @@ static int wm8974_set_bias_level(struct snd_soc_codec *codec, power1 |= WM8974_POWER1_BIASEN | WM8974_POWER1_BUFIOEN; if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { + snd_soc_cache_sync(codec); + /* Initial cap charge at VMID 5k */ snd_soc_write(codec, WM8974_POWER1, power1 | 0x3); mdelay(100); @@ -589,18 +591,7 @@ static int wm8974_suspend(struct snd_soc_codec *codec, pm_message_t state) static int wm8974_resume(struct snd_soc_codec *codec) { - int i; - u8 data[2]; - u16 *cache = codec->reg_cache; - - /* Sync reg_cache with the hardware */ - for (i = 0; i < ARRAY_SIZE(wm8974_reg); i++) { - data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001); - data[1] = cache[i] & 0x00ff; - codec->hw_write(codec->control_data, data, 2); - } wm8974_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - return 0; } -- cgit v1.1 From fa5fdb473e9c593525601d445124d563c6be150e Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Fri, 7 Oct 2011 21:53:39 +0800 Subject: ASoC: wm8988: Convert to snd_soc_cache_sync Convert to snd_soc_cache_sync for sync reg_cache with the hardware. Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/wm8988.c | 17 ++--------------- 1 file changed, 2 insertions(+), 15 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8988.c b/sound/soc/codecs/wm8988.c index 1c6f8bf..2e9eba7 100644 --- a/sound/soc/codecs/wm8988.c +++ b/sound/soc/codecs/wm8988.c @@ -55,7 +55,6 @@ struct wm8988_priv { struct snd_pcm_hw_constraint_list *sysclk_constraints; }; - #define wm8988_reset(c) snd_soc_write(c, WM8988_RESET, 0) /* @@ -676,6 +675,8 @@ static int wm8988_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_STANDBY: if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { + snd_soc_cache_sync(codec); + /* VREF, VMID=2x5k */ snd_soc_write(codec, WM8988_PWR1, pwr_reg | 0x1c1); @@ -736,21 +737,7 @@ static int wm8988_suspend(struct snd_soc_codec *codec, pm_message_t state) static int wm8988_resume(struct snd_soc_codec *codec) { - int i; - u8 data[2]; - u16 *cache = codec->reg_cache; - - /* Sync reg_cache with the hardware */ - for (i = 0; i < WM8988_NUM_REG; i++) { - if (i == WM8988_RESET) - continue; - data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001); - data[1] = cache[i] & 0x00ff; - codec->hw_write(codec->control_data, data, 2); - } - wm8988_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - return 0; } -- cgit v1.1 From 3b1845037182e06573945335961d3f4c7695f094 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Thu, 6 Oct 2011 10:11:51 +0800 Subject: ASoC: ak4535: convert to soc-cache Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/ak4535.c | 100 +++++++++------------------------------------- 1 file changed, 18 insertions(+), 82 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c index e1a214e..68df32d 100644 --- a/sound/soc/codecs/ak4535.c +++ b/sound/soc/codecs/ak4535.c @@ -34,7 +34,6 @@ struct ak4535_priv { unsigned int sysclk; enum snd_soc_control_type control_type; - void *control_data; }; /* @@ -47,63 +46,6 @@ static const u16 ak4535_reg[AK4535_CACHEREGNUM] = { 0x0000, 0x0000, 0x0057, 0x0000, }; -/* - * read ak4535 register cache - */ -static inline unsigned int ak4535_read_reg_cache(struct snd_soc_codec *codec, - unsigned int reg) -{ - u16 *cache = codec->reg_cache; - if (reg >= AK4535_CACHEREGNUM) - return -1; - return cache[reg]; -} - -/* - * write ak4535 register cache - */ -static inline void ak4535_write_reg_cache(struct snd_soc_codec *codec, - u16 reg, unsigned int value) -{ - u16 *cache = codec->reg_cache; - if (reg >= AK4535_CACHEREGNUM) - return; - cache[reg] = value; -} - -/* - * write to the AK4535 register space - */ -static int ak4535_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) -{ - u8 data[2]; - - /* data is - * D15..D8 AK4535 register offset - * D7...D0 register data - */ - data[0] = reg & 0xff; - data[1] = value & 0xff; - - ak4535_write_reg_cache(codec, reg, value); - if (codec->hw_write(codec->control_data, data, 2) == 2) - return 0; - else - return -EIO; -} - -static int ak4535_sync(struct snd_soc_codec *codec) -{ - u16 *cache = codec->reg_cache; - int i, r = 0; - - for (i = 0; i < AK4535_CACHEREGNUM; i++) - r |= ak4535_write(codec, i, cache[i]); - - return r; -}; - static const char *ak4535_mono_gain[] = {"+6dB", "-17dB"}; static const char *ak4535_mono_out[] = {"(L + R)/2", "Hi-Z"}; static const char *ak4535_hp_out[] = {"Stereo", "Mono"}; @@ -304,7 +246,7 @@ static int ak4535_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_codec *codec = rtd->codec; struct ak4535_priv *ak4535 = snd_soc_codec_get_drvdata(codec); - u8 mode2 = ak4535_read_reg_cache(codec, AK4535_MODE2) & ~(0x3 << 5); + u8 mode2 = snd_soc_read(codec, AK4535_MODE2) & ~(0x3 << 5); int rate = params_rate(params), fs = 256; if (rate) @@ -323,7 +265,7 @@ static int ak4535_hw_params(struct snd_pcm_substream *substream, } /* set rate */ - ak4535_write(codec, AK4535_MODE2, mode2); + snd_soc_write(codec, AK4535_MODE2, mode2); return 0; } @@ -348,44 +290,37 @@ static int ak4535_set_dai_fmt(struct snd_soc_dai *codec_dai, /* use 32 fs for BCLK to save power */ mode1 |= 0x4; - ak4535_write(codec, AK4535_MODE1, mode1); + snd_soc_write(codec, AK4535_MODE1, mode1); return 0; } static int ak4535_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; - u16 mute_reg = ak4535_read_reg_cache(codec, AK4535_DAC); + u16 mute_reg = snd_soc_read(codec, AK4535_DAC); if (!mute) - ak4535_write(codec, AK4535_DAC, mute_reg & ~0x20); + snd_soc_write(codec, AK4535_DAC, mute_reg & ~0x20); else - ak4535_write(codec, AK4535_DAC, mute_reg | 0x20); + snd_soc_write(codec, AK4535_DAC, mute_reg | 0x20); return 0; } static int ak4535_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { - u16 i, mute_reg; - switch (level) { case SND_SOC_BIAS_ON: - mute_reg = ak4535_read_reg_cache(codec, AK4535_DAC); - ak4535_write(codec, AK4535_DAC, mute_reg & ~0x20); + snd_soc_update_bits(codec, AK4535_DAC, 0x20, 0); break; case SND_SOC_BIAS_PREPARE: - mute_reg = ak4535_read_reg_cache(codec, AK4535_DAC); - ak4535_write(codec, AK4535_DAC, mute_reg | 0x20); + snd_soc_update_bits(codec, AK4535_DAC, 0x20, 0x20); break; case SND_SOC_BIAS_STANDBY: - i = ak4535_read_reg_cache(codec, AK4535_PM1); - ak4535_write(codec, AK4535_PM1, i | 0x80); - i = ak4535_read_reg_cache(codec, AK4535_PM2); - ak4535_write(codec, AK4535_PM2, i & (~0x80)); + snd_soc_update_bits(codec, AK4535_PM1, 0x80, 0x80); + snd_soc_update_bits(codec, AK4535_PM2, 0x80, 0); break; case SND_SOC_BIAS_OFF: - i = ak4535_read_reg_cache(codec, AK4535_PM1); - ak4535_write(codec, AK4535_PM1, i & (~0x80)); + snd_soc_update_bits(codec, AK4535_PM1, 0x80, 0); break; } codec->dapm.bias_level = level; @@ -428,7 +363,7 @@ static int ak4535_suspend(struct snd_soc_codec *codec, pm_message_t state) static int ak4535_resume(struct snd_soc_codec *codec) { - ak4535_sync(codec); + snd_soc_cache_sync(codec); ak4535_set_bias_level(codec, SND_SOC_BIAS_STANDBY); return 0; } @@ -436,11 +371,15 @@ static int ak4535_resume(struct snd_soc_codec *codec) static int ak4535_probe(struct snd_soc_codec *codec) { struct ak4535_priv *ak4535 = snd_soc_codec_get_drvdata(codec); + int ret; printk(KERN_INFO "AK4535 Audio Codec %s", AK4535_VERSION); - codec->control_data = ak4535->control_data; - + ret = snd_soc_codec_set_cache_io(codec, 8, 8, ak4535->control_type); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } /* power on device */ ak4535_set_bias_level(codec, SND_SOC_BIAS_STANDBY); @@ -461,8 +400,6 @@ static struct snd_soc_codec_driver soc_codec_dev_ak4535 = { .remove = ak4535_remove, .suspend = ak4535_suspend, .resume = ak4535_resume, - .read = ak4535_read_reg_cache, - .write = ak4535_write, .set_bias_level = ak4535_set_bias_level, .reg_cache_size = ARRAY_SIZE(ak4535_reg), .reg_word_size = sizeof(u8), @@ -485,7 +422,6 @@ static __devinit int ak4535_i2c_probe(struct i2c_client *i2c, return -ENOMEM; i2c_set_clientdata(i2c, ak4535); - ak4535->control_data = i2c; ak4535->control_type = SND_SOC_I2C; ret = snd_soc_register_codec(&i2c->dev, -- cgit v1.1 From 6179b772ac6ac2209096753e2aa91edb9c3988a5 Mon Sep 17 00:00:00 2001 From: Michael Opdenacker Date: Mon, 10 Oct 2011 07:07:08 +0200 Subject: ASoC: fix checkpatch.pl error in omap-mcbsp Signed-off-by: Michael Opdenacker Acked-by: Jarkko Nikula Signed-off-by: Mark Brown --- sound/soc/omap/omap-mcbsp.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 7f70061..4314647 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -617,8 +617,7 @@ static int mcbsp_dai_probe(struct snd_soc_dai *dai) return 0; } -static struct snd_soc_dai_driver omap_mcbsp_dai = -{ +static struct snd_soc_dai_driver omap_mcbsp_dai = { .probe = mcbsp_dai_probe, .playback = { .channels_min = 1, -- cgit v1.1 From 4f5448ae4b1b877c777a6f96af7bef31f505936d Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 8 Oct 2011 12:19:12 +0100 Subject: ASoC: Convert Simtec machines to table based DAPM init Signed-off-by: Mark Brown --- sound/soc/samsung/s3c24xx_simtec_hermes.c | 10 +++++----- sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c | 10 +++++----- 2 files changed, 10 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/s3c24xx_simtec_hermes.c b/sound/soc/samsung/s3c24xx_simtec_hermes.c index 359530b..6bc5a36 100644 --- a/sound/soc/samsung/s3c24xx_simtec_hermes.c +++ b/sound/soc/samsung/s3c24xx_simtec_hermes.c @@ -65,11 +65,6 @@ static int simtec_hermes_init(struct snd_soc_pcm_runtime *rtd) struct snd_soc_codec *codec = rtd->codec; struct snd_soc_dapm_context *dapm = &codec->dapm; - snd_soc_dapm_new_controls(dapm, dapm_widgets, - ARRAY_SIZE(dapm_widgets)); - - snd_soc_dapm_add_routes(dapm, base_map, ARRAY_SIZE(base_map)); - snd_soc_dapm_enable_pin(dapm, "Headphone Jack"); snd_soc_dapm_enable_pin(dapm, "Line In"); snd_soc_dapm_enable_pin(dapm, "Line Out"); @@ -95,6 +90,11 @@ static struct snd_soc_card snd_soc_machine_simtec_aic33 = { .name = "Simtec-Hermes", .dai_link = &simtec_dai_aic33, .num_links = 1, + + .dapm_widgets = dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(dapm_widgets), + .dapm_routes = base_map, + .num_dapm_routes = ARRAY_SIZE(base_map), }; static int __devinit simtec_audio_hermes_probe(struct platform_device *pd) diff --git a/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c b/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c index 8bad349..7bdda76 100644 --- a/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c +++ b/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c @@ -54,11 +54,6 @@ static int simtec_tlv320aic23_init(struct snd_soc_pcm_runtime *rtd) struct snd_soc_codec *codec = rtd->codec; struct snd_soc_dapm_context *dapm = &codec->dapm; - snd_soc_dapm_new_controls(dapm, dapm_widgets, - ARRAY_SIZE(dapm_widgets)); - - snd_soc_dapm_add_routes(dapm, base_map, ARRAY_SIZE(base_map)); - snd_soc_dapm_enable_pin(dapm, "Headphone Jack"); snd_soc_dapm_enable_pin(dapm, "Line In"); snd_soc_dapm_enable_pin(dapm, "Line Out"); @@ -84,6 +79,11 @@ static struct snd_soc_card snd_soc_machine_simtec_aic23 = { .name = "Simtec", .dai_link = &simtec_dai_aic23, .num_links = 1, + + .dapm_widgets = dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(dapm_widgets), + .dapm_routes = base_map, + .num_dapm_routes = ARRAY_SIZE(base_map), }; static int __devinit simtec_audio_tlv320aic23_probe(struct platform_device *pd) -- cgit v1.1 From 6119d016b1dd036d4dd7a8cfad4e4d113be32d91 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 8 Oct 2011 13:30:06 +0100 Subject: ASoC: Convert H1940 to table based init Signed-off-by: Mark Brown Acked-by: Sangbeom Kim --- sound/soc/samsung/h1940_uda1380.c | 17 +++++------------ 1 file changed, 5 insertions(+), 12 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/h1940_uda1380.c b/sound/soc/samsung/h1940_uda1380.c index 52025c9..f75a4b6 100644 --- a/sound/soc/samsung/h1940_uda1380.c +++ b/sound/soc/samsung/h1940_uda1380.c @@ -182,18 +182,6 @@ static int h1940_uda1380_init(struct snd_soc_pcm_runtime *rtd) struct snd_soc_dapm_context *dapm = &codec->dapm; int err; - /* Add h1940 specific widgets */ - err = snd_soc_dapm_new_controls(dapm, uda1380_dapm_widgets, - ARRAY_SIZE(uda1380_dapm_widgets)); - if (err) - return err; - - /* Set up h1940 specific audio path audio_mapnects */ - err = snd_soc_dapm_add_routes(dapm, audio_map, - ARRAY_SIZE(audio_map)); - if (err) - return err; - snd_soc_dapm_enable_pin(dapm, "Headphone Jack"); snd_soc_dapm_enable_pin(dapm, "Speaker"); snd_soc_dapm_enable_pin(dapm, "Mic Jack"); @@ -228,6 +216,11 @@ static struct snd_soc_card h1940_asoc = { .name = "h1940", .dai_link = h1940_uda1380_dai, .num_links = ARRAY_SIZE(h1940_uda1380_dai), + + .dapm_widgets = uda1380_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(uda1380_dapm_widgets), + .dapm_routes = audio_map, + .num_dapm_routes = ARRAY_SIZE(audio_map), }; static int __init h1940_init(void) -- cgit v1.1 From 8ae232299e0cc87671e27bd4446acee32c3ccc1d Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 8 Oct 2011 13:30:35 +0100 Subject: ASoC: Convert RX1950 to table based init Signed-off-by: Mark Brown Acked-by: Sangbeom Kim --- sound/soc/samsung/rx1950_uda1380.c | 31 +++++++++++-------------------- 1 file changed, 11 insertions(+), 20 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/rx1950_uda1380.c b/sound/soc/samsung/rx1950_uda1380.c index e5ad67f..aea7f1b 100644 --- a/sound/soc/samsung/rx1950_uda1380.c +++ b/sound/soc/samsung/rx1950_uda1380.c @@ -90,12 +90,6 @@ static struct snd_soc_dai_link rx1950_uda1380_dai[] = { }, }; -static struct snd_soc_card rx1950_asoc = { - .name = "rx1950", - .dai_link = rx1950_uda1380_dai, - .num_links = ARRAY_SIZE(rx1950_uda1380_dai), -}; - /* rx1950 machine dapm widgets */ static const struct snd_soc_dapm_widget uda1380_dapm_widgets[] = { SND_SOC_DAPM_HP("Headphone Jack", NULL), @@ -117,6 +111,17 @@ static const struct snd_soc_dapm_route audio_map[] = { {"VINM", NULL, "Mic Jack"}, }; +static struct snd_soc_card rx1950_asoc = { + .name = "rx1950", + .dai_link = rx1950_uda1380_dai, + .num_links = ARRAY_SIZE(rx1950_uda1380_dai), + + .dapm_widgets = uda1380_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(uda1380_dapm_widgets), + .dapm_routes = audio_map, + .num_dapm_routes = ARRAY_SIZE(audio_map), +}; + static struct platform_device *s3c24xx_snd_device; static int rx1950_startup(struct snd_pcm_substream *substream) @@ -220,20 +225,6 @@ static int rx1950_uda1380_init(struct snd_soc_pcm_runtime *rtd) struct snd_soc_dapm_context *dapm = &codec->dapm; int err; - /* Add rx1950 specific widgets */ - err = snd_soc_dapm_new_controls(dapm, uda1380_dapm_widgets, - ARRAY_SIZE(uda1380_dapm_widgets)); - - if (err) - return err; - - /* Set up rx1950 specific audio path audio_mapnects */ - err = snd_soc_dapm_add_routes(dapm, audio_map, - ARRAY_SIZE(audio_map)); - - if (err) - return err; - snd_soc_dapm_enable_pin(dapm, "Headphone Jack"); snd_soc_dapm_enable_pin(dapm, "Speaker"); snd_soc_dapm_enable_pin(dapm, "Mic Jack"); -- cgit v1.1 From 257fe5930dadc69e7da3e16a310e7fa22fbecdcf Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 8 Oct 2011 13:30:55 +0100 Subject: ASoC: Convert SmartQ to table based init Signed-off-by: Mark Brown Acked-by: Sangbeom Kim --- sound/soc/samsung/smartq_wm8987.c | 21 +++++++-------------- 1 file changed, 7 insertions(+), 14 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/smartq_wm8987.c b/sound/soc/samsung/smartq_wm8987.c index 6db4e55..6ac6bc2 100644 --- a/sound/soc/samsung/smartq_wm8987.c +++ b/sound/soc/samsung/smartq_wm8987.c @@ -153,20 +153,6 @@ static int smartq_wm8987_init(struct snd_soc_pcm_runtime *rtd) struct snd_soc_dapm_context *dapm = &codec->dapm; int err = 0; - /* Add SmartQ specific widgets */ - snd_soc_dapm_new_controls(dapm, wm8987_dapm_widgets, - ARRAY_SIZE(wm8987_dapm_widgets)); - - /* add SmartQ specific controls */ - err = snd_soc_add_controls(codec, wm8987_smartq_controls, - ARRAY_SIZE(wm8987_smartq_controls)); - - if (err < 0) - return err; - - /* setup SmartQ specific audio path */ - snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - /* set endpoints to not connected */ snd_soc_dapm_nc_pin(dapm, "LINPUT1"); snd_soc_dapm_nc_pin(dapm, "RINPUT1"); @@ -213,6 +199,13 @@ static struct snd_soc_card snd_soc_smartq = { .name = "SmartQ", .dai_link = smartq_dai, .num_links = ARRAY_SIZE(smartq_dai), + + .dapm_widgets = wm8987_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(wm8987_dapm_widgets), + .dapm_routes = audio_map, + .num_dapm_routes = ARRAY_SIZE(audio_map), + .controls = wm8987_smartq_controls, + .num_controls = ARRAY_SIZE(wm8987_smartq_controls), }; static struct platform_device *smartq_snd_device; -- cgit v1.1 From ce363f6d34e43dd860f1df10fa213ad1bbf6a4e0 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 8 Oct 2011 13:31:18 +0100 Subject: ASoC: Convert SMDK WM8580 to table based DAPM init Signed-off-by: Mark Brown Acked-by: Sangbeom Kim --- sound/soc/samsung/smdk_wm8580.c | 39 +++++++-------------------------------- 1 file changed, 7 insertions(+), 32 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/smdk_wm8580.c b/sound/soc/samsung/smdk_wm8580.c index 54c4cce..8f92ffc 100644 --- a/sound/soc/samsung/smdk_wm8580.c +++ b/sound/soc/samsung/smdk_wm8580.c @@ -119,30 +119,24 @@ static struct snd_soc_ops smdk_ops = { }; /* SMDK Playback widgets */ -static const struct snd_soc_dapm_widget wm8580_dapm_widgets_pbk[] = { +static const struct snd_soc_dapm_widget smdk_wm8580_dapm_widgets[] = { SND_SOC_DAPM_HP("Front", NULL), SND_SOC_DAPM_HP("Center+Sub", NULL), SND_SOC_DAPM_HP("Rear", NULL), -}; -/* SMDK Capture widgets */ -static const struct snd_soc_dapm_widget wm8580_dapm_widgets_cpt[] = { SND_SOC_DAPM_MIC("MicIn", NULL), SND_SOC_DAPM_LINE("LineIn", NULL), }; /* SMDK-PAIFTX connections */ -static const struct snd_soc_dapm_route audio_map_tx[] = { +static const struct snd_soc_dapm_route smdk_wm8580_audio_map[] = { /* MicIn feeds AINL */ {"AINL", NULL, "MicIn"}, /* LineIn feeds AINL/R */ {"AINL", NULL, "LineIn"}, {"AINR", NULL, "LineIn"}, -}; -/* SMDK-PAIFRX connections */ -static const struct snd_soc_dapm_route audio_map_rx[] = { /* Front Left/Right are fed VOUT1L/R */ {"Front", NULL, "VOUT1L"}, {"Front", NULL, "VOUT1R"}, @@ -161,13 +155,6 @@ static int smdk_wm8580_init_paiftx(struct snd_soc_pcm_runtime *rtd) struct snd_soc_codec *codec = rtd->codec; struct snd_soc_dapm_context *dapm = &codec->dapm; - /* Add smdk specific Capture widgets */ - snd_soc_dapm_new_controls(dapm, wm8580_dapm_widgets_cpt, - ARRAY_SIZE(wm8580_dapm_widgets_cpt)); - - /* Set up PAIFTX audio path */ - snd_soc_dapm_add_routes(dapm, audio_map_tx, ARRAY_SIZE(audio_map_tx)); - /* Enabling the microphone requires the fitting of a 0R * resistor to connect the line from the microphone jack. */ @@ -176,21 +163,6 @@ static int smdk_wm8580_init_paiftx(struct snd_soc_pcm_runtime *rtd) return 0; } -static int smdk_wm8580_init_paifrx(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_soc_codec *codec = rtd->codec; - struct snd_soc_dapm_context *dapm = &codec->dapm; - - /* Add smdk specific Playback widgets */ - snd_soc_dapm_new_controls(dapm, wm8580_dapm_widgets_pbk, - ARRAY_SIZE(wm8580_dapm_widgets_pbk)); - - /* Set up PAIFRX audio path */ - snd_soc_dapm_add_routes(dapm, audio_map_rx, ARRAY_SIZE(audio_map_rx)); - - return 0; -} - enum { PRI_PLAYBACK = 0, PRI_CAPTURE, @@ -205,7 +177,6 @@ static struct snd_soc_dai_link smdk_dai[] = { .codec_dai_name = "wm8580-hifi-playback", .platform_name = "samsung-audio", .codec_name = "wm8580.0-001b", - .init = smdk_wm8580_init_paifrx, .ops = &smdk_ops, }, [PRI_CAPTURE] = { /* Primary Capture i/f */ @@ -225,7 +196,6 @@ static struct snd_soc_dai_link smdk_dai[] = { .codec_dai_name = "wm8580-hifi-playback", .platform_name = "samsung-audio", .codec_name = "wm8580.0-001b", - .init = smdk_wm8580_init_paifrx, .ops = &smdk_ops, }, }; @@ -234,6 +204,11 @@ static struct snd_soc_card smdk = { .name = "SMDK-I2S", .dai_link = smdk_dai, .num_links = 2, + + .dapm_widgets = smdk_wm8580_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(smdk_wm8580_dapm_widgets), + .dapm_routes = smdk_wm8580_audio_map, + .num_dapm_routes = ARRAY_SIZE(smdk_wm8580_audio_map), }; static struct platform_device *smdk_snd_device; -- cgit v1.1 From 03b5362d2fef02cabbbb440d861d27e40ed3a929 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 8 Oct 2011 13:30:17 +0100 Subject: ASoC: Convert Jive to table based init Signed-off-by: Mark Brown Acked-by: Sangbeom Kim --- sound/soc/samsung/jive_wm8750.c | 16 +++++----------- 1 file changed, 5 insertions(+), 11 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/jive_wm8750.c b/sound/soc/samsung/jive_wm8750.c index bb69e9f..f5f7c6f 100644 --- a/sound/soc/samsung/jive_wm8750.c +++ b/sound/soc/samsung/jive_wm8750.c @@ -110,17 +110,6 @@ static int jive_wm8750_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_nc_pin(dapm, "OUT3"); snd_soc_dapm_nc_pin(dapm, "MONO"); - /* Add jive specific widgets */ - err = snd_soc_dapm_new_controls(dapm, wm8750_dapm_widgets, - ARRAY_SIZE(wm8750_dapm_widgets)); - if (err) { - printk(KERN_ERR "%s: failed to add widgets (%d)\n", - __func__, err); - return err; - } - - snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - return 0; } @@ -140,6 +129,11 @@ static struct snd_soc_card snd_soc_machine_jive = { .name = "Jive", .dai_link = &jive_dai, .num_links = 1, + + .dapm_widgtets = wm8750_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(wm8750_dapm_widgets), + .dapm_routes = audio_map, + .num_dapm_routes = ARRAY_SIZE(audio_map), }; static struct platform_device *jive_snd_device; -- cgit v1.1 From cb4248779d317eb5a0f554b298134689c395c4fd Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Mon, 10 Oct 2011 15:34:08 +0300 Subject: ASoC: OMAP machines: Remove soc_dapm_sync() call from init MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit No need to call soc_dapm_sync at init time. Signed-off-by: Peter Ujfalusi Cc: Anuj Aggarwal Cc: Janusz Krzysztofik Cc: Jarkko Nikula Cc: Gražvydas Ignotas Cc: Misael Lopez Cruz Signed-off-by: Mark Brown --- sound/soc/omap/am3517evm.c | 2 -- sound/soc/omap/ams-delta.c | 1 - sound/soc/omap/n810.c | 6 +----- sound/soc/omap/omap3pandora.c | 8 ++------ sound/soc/omap/osk5912.c | 2 -- sound/soc/omap/rx51.c | 2 -- sound/soc/omap/sdp3430.c | 4 ---- sound/soc/omap/sdp4430.c | 4 ---- sound/soc/omap/zoom2.c | 4 +--- 9 files changed, 4 insertions(+), 29 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/am3517evm.c b/sound/soc/omap/am3517evm.c index 48af0f8..f5a4d65 100644 --- a/sound/soc/omap/am3517evm.c +++ b/sound/soc/omap/am3517evm.c @@ -107,8 +107,6 @@ static int am3517evm_aic23_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_enable_pin(dapm, "Line In"); snd_soc_dapm_enable_pin(dapm, "Mic In"); - snd_soc_dapm_sync(dapm); - return 0; } diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c index 0aa475f..dcb7b689 100644 --- a/sound/soc/omap/ams-delta.c +++ b/sound/soc/omap/ams-delta.c @@ -569,7 +569,6 @@ static int ams_delta_cx20442_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_disable_pin(dapm, "Speaker"); snd_soc_dapm_disable_pin(dapm, "AGCIN"); snd_soc_dapm_disable_pin(dapm, "AGCOUT"); - snd_soc_dapm_sync(dapm); /* Add virtual switch */ ret = snd_soc_add_controls(codec, ams_delta_audio_controls, diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c index c10d356..4fa881b 100644 --- a/sound/soc/omap/n810.c +++ b/sound/soc/omap/n810.c @@ -280,11 +280,7 @@ static int n810_aic33_init(struct snd_soc_pcm_runtime *rtd) ARRAY_SIZE(aic33_dapm_widgets)); /* Set up N810 specific audio path audio_map */ - snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - - snd_soc_dapm_sync(dapm); - - return 0; + return snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); } /* Digital audio interface glue - connects codec <--> CPU */ diff --git a/sound/soc/omap/omap3pandora.c b/sound/soc/omap/omap3pandora.c index 3ae87fd..30a75b4 100644 --- a/sound/soc/omap/omap3pandora.c +++ b/sound/soc/omap/omap3pandora.c @@ -173,10 +173,8 @@ static int omap3pandora_out_init(struct snd_soc_pcm_runtime *rtd) if (ret < 0) return ret; - snd_soc_dapm_add_routes(dapm, omap3pandora_out_map, + return snd_soc_dapm_add_routes(dapm, omap3pandora_out_map, ARRAY_SIZE(omap3pandora_out_map)); - - return snd_soc_dapm_sync(dapm); } static int omap3pandora_in_init(struct snd_soc_pcm_runtime *rtd) @@ -196,10 +194,8 @@ static int omap3pandora_in_init(struct snd_soc_pcm_runtime *rtd) if (ret < 0) return ret; - snd_soc_dapm_add_routes(dapm, omap3pandora_in_map, + return snd_soc_dapm_add_routes(dapm, omap3pandora_in_map, ARRAY_SIZE(omap3pandora_in_map)); - - return snd_soc_dapm_sync(dapm); } static struct snd_soc_ops omap3pandora_ops = { diff --git a/sound/soc/omap/osk5912.c b/sound/soc/omap/osk5912.c index 5978332..18f8a2d 100644 --- a/sound/soc/omap/osk5912.c +++ b/sound/soc/omap/osk5912.c @@ -107,8 +107,6 @@ static int osk_tlv320aic23_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_enable_pin(dapm, "Line In"); snd_soc_dapm_enable_pin(dapm, "Mic Jack"); - snd_soc_dapm_sync(dapm); - return 0; } diff --git a/sound/soc/omap/rx51.c b/sound/soc/omap/rx51.c index 7164db5..a568423 100644 --- a/sound/soc/omap/rx51.c +++ b/sound/soc/omap/rx51.c @@ -317,8 +317,6 @@ static int rx51_aic34_init(struct snd_soc_pcm_runtime *rtd) if (err < 0) return err; - snd_soc_dapm_sync(dapm); - /* AV jack detection */ err = snd_soc_jack_new(codec, "AV Jack", SND_JACK_HEADSET | SND_JACK_VIDEOOUT, diff --git a/sound/soc/omap/sdp3430.c b/sound/soc/omap/sdp3430.c index 269aded..85e2e918 100644 --- a/sound/soc/omap/sdp3430.c +++ b/sound/soc/omap/sdp3430.c @@ -159,10 +159,6 @@ static int sdp3430_twl4030_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_nc_pin(dapm, "CARKITL"); snd_soc_dapm_nc_pin(dapm, "CARKITR"); - ret = snd_soc_dapm_sync(dapm); - if (ret) - return ret; - /* Headset jack detection */ ret = snd_soc_jack_new(codec, "Headset Jack", SND_JACK_HEADSET, &hs_jack); diff --git a/sound/soc/omap/sdp4430.c b/sound/soc/omap/sdp4430.c index 79ec76d..98f05dc 100644 --- a/sound/soc/omap/sdp4430.c +++ b/sound/soc/omap/sdp4430.c @@ -140,10 +140,6 @@ static int sdp4430_twl6040_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_enable_pin(dapm, "Headset Mic"); snd_soc_dapm_enable_pin(dapm, "Headset Stereophone"); - ret = snd_soc_dapm_sync(dapm); - if (ret) - return ret; - /* * Configure McPDM offset cancellation based on the HSOTRIM value from * twl6040. diff --git a/sound/soc/omap/zoom2.c b/sound/soc/omap/zoom2.c index 8b1ebbc..9a8288d 100644 --- a/sound/soc/omap/zoom2.c +++ b/sound/soc/omap/zoom2.c @@ -126,9 +126,7 @@ static int zoom2_twl4030_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_nc_pin(dapm, "CARKITL"); snd_soc_dapm_nc_pin(dapm, "CARKITR"); - ret = snd_soc_dapm_sync(dapm); - - return ret; + return 0; } static int zoom2_twl4030_voice_init(struct snd_soc_pcm_runtime *rtd) -- cgit v1.1 From 1083dbd1d75723960908f3d6bd32befb22944856 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Mon, 10 Oct 2011 15:34:13 +0300 Subject: ASoC: zoom2: Let core to deal with the DAPM widgets Pass the DAPM widgets/routes via the snd_soc_card struct to core. Signed-off-by: Peter Ujfalusi Cc: Misael Lopez Cruz Signed-off-by: Mark Brown --- sound/soc/omap/zoom2.c | 15 +++++---------- 1 file changed, 5 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/zoom2.c b/sound/soc/omap/zoom2.c index 9a8288d..4d01e16 100644 --- a/sound/soc/omap/zoom2.c +++ b/sound/soc/omap/zoom2.c @@ -98,16 +98,6 @@ static int zoom2_twl4030_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; struct snd_soc_dapm_context *dapm = &codec->dapm; - int ret; - - /* Add Zoom2 specific widgets */ - ret = snd_soc_dapm_new_controls(dapm, zoom2_twl4030_dapm_widgets, - ARRAY_SIZE(zoom2_twl4030_dapm_widgets)); - if (ret) - return ret; - - /* Set up Zoom2 specific audio path audio_map */ - snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); /* Zoom2 connected pins */ snd_soc_dapm_enable_pin(dapm, "Ext Mic"); @@ -175,6 +165,11 @@ static struct snd_soc_card snd_soc_zoom2 = { .name = "Zoom2", .dai_link = zoom2_dai, .num_links = ARRAY_SIZE(zoom2_dai), + + .dapm_widgets = zoom2_twl4030_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(zoom2_twl4030_dapm_widgets), + .dapm_routes = audio_map, + .num_dapm_routes = ARRAY_SIZE(audio_map), }; static struct platform_device *zoom2_snd_device; -- cgit v1.1 From cd4a3d5d0d1794ee08d716b5b88f46efa52f9a7d Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Mon, 10 Oct 2011 15:34:14 +0300 Subject: ASoC: zoom2: No need to call dapm_pin_enable at init time Widgets are connected by default. Signed-off-by: Peter Ujfalusi Cc: Misael Lopez Cruz Signed-off-by: Mark Brown --- sound/soc/omap/zoom2.c | 7 ------- 1 file changed, 7 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/zoom2.c b/sound/soc/omap/zoom2.c index 4d01e16..7cf35c8 100644 --- a/sound/soc/omap/zoom2.c +++ b/sound/soc/omap/zoom2.c @@ -99,13 +99,6 @@ static int zoom2_twl4030_init(struct snd_soc_pcm_runtime *rtd) struct snd_soc_codec *codec = rtd->codec; struct snd_soc_dapm_context *dapm = &codec->dapm; - /* Zoom2 connected pins */ - snd_soc_dapm_enable_pin(dapm, "Ext Mic"); - snd_soc_dapm_enable_pin(dapm, "Ext Spk"); - snd_soc_dapm_enable_pin(dapm, "Headset Mic"); - snd_soc_dapm_enable_pin(dapm, "Headset Stereophone"); - snd_soc_dapm_enable_pin(dapm, "Aux In"); - /* TWL4030 not connected pins */ snd_soc_dapm_nc_pin(dapm, "CARKITMIC"); snd_soc_dapm_nc_pin(dapm, "DIGIMIC0"); -- cgit v1.1 From fc8e5b4c6bc4194b27644142d51d466ecd9f1bd0 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Mon, 10 Oct 2011 15:34:15 +0300 Subject: ASoC: sdp4430: Let core to deal with the DAPM widgets Pass the DAPM widgets/routes via the snd_soc_card struct to core. Signed-off-by: Peter Ujfalusi Cc: Misael Lopez Cruz Signed-off-by: Mark Brown --- sound/soc/omap/sdp4430.c | 14 +++++--------- 1 file changed, 5 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/sdp4430.c b/sound/soc/omap/sdp4430.c index 98f05dc..ebe1bbb 100644 --- a/sound/soc/omap/sdp4430.c +++ b/sound/soc/omap/sdp4430.c @@ -123,15 +123,6 @@ static int sdp4430_twl6040_init(struct snd_soc_pcm_runtime *rtd) struct snd_soc_dapm_context *dapm = &codec->dapm; int ret, hs_trim; - /* Add SDP4430 specific widgets */ - ret = snd_soc_dapm_new_controls(dapm, sdp4430_twl6040_dapm_widgets, - ARRAY_SIZE(sdp4430_twl6040_dapm_widgets)); - if (ret) - return ret; - - /* Set up SDP4430 specific audio path audio_map */ - snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - /* SDP4430 connected pins */ snd_soc_dapm_enable_pin(dapm, "Ext Mic"); snd_soc_dapm_enable_pin(dapm, "Ext Spk"); @@ -182,6 +173,11 @@ static struct snd_soc_card snd_soc_sdp4430 = { .name = "SDP4430", .dai_link = &sdp4430_dai, .num_links = 1, + + .dapm_widgets = sdp4430_twl6040_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(sdp4430_twl6040_dapm_widgets), + .dapm_routes = audio_map, + .num_dapm_routes = ARRAY_SIZE(audio_map), }; static struct platform_device *sdp4430_snd_device; -- cgit v1.1 From d8058469606bfd1b0d6fca830a997ba06f83e5dc Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Mon, 10 Oct 2011 15:34:16 +0300 Subject: ASoC: sdp4430: No need to call dapm_pin_enable at init time Widgets are connected by default. Signed-off-by: Peter Ujfalusi Cc: Misael Lopez Cruz Signed-off-by: Mark Brown --- sound/soc/omap/sdp4430.c | 9 --------- 1 file changed, 9 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/sdp4430.c b/sound/soc/omap/sdp4430.c index ebe1bbb..cc3d792a 100644 --- a/sound/soc/omap/sdp4430.c +++ b/sound/soc/omap/sdp4430.c @@ -120,17 +120,8 @@ static const struct snd_soc_dapm_route audio_map[] = { static int sdp4430_twl6040_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; - struct snd_soc_dapm_context *dapm = &codec->dapm; int ret, hs_trim; - /* SDP4430 connected pins */ - snd_soc_dapm_enable_pin(dapm, "Ext Mic"); - snd_soc_dapm_enable_pin(dapm, "Ext Spk"); - snd_soc_dapm_enable_pin(dapm, "AFML"); - snd_soc_dapm_enable_pin(dapm, "AFMR"); - snd_soc_dapm_enable_pin(dapm, "Headset Mic"); - snd_soc_dapm_enable_pin(dapm, "Headset Stereophone"); - /* * Configure McPDM offset cancellation based on the HSOTRIM value from * twl6040. -- cgit v1.1 From d2266025ea26b6f3f8f93da6fad129541c82bd7c Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Mon, 10 Oct 2011 15:34:09 +0300 Subject: ASoC: am3517evm: Let core to deal with the DAPM widgets Pass the DAPM widgets/routes via the snd_soc_card struct to core. With this change we do not need the init function since the remaining snd_soc_dapm_enable_pin calls are not needed. Signed-off-by: Peter Ujfalusi Cc: Anuj Aggarwal Signed-off-by: Mark Brown --- sound/soc/omap/am3517evm.c | 26 +++++--------------------- 1 file changed, 5 insertions(+), 21 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/am3517evm.c b/sound/soc/omap/am3517evm.c index f5a4d65..8da55e9 100644 --- a/sound/soc/omap/am3517evm.c +++ b/sound/soc/omap/am3517evm.c @@ -90,26 +90,6 @@ static const struct snd_soc_dapm_route audio_map[] = { {"MICIN", NULL, "Mic In"}, }; -static int am3517evm_aic23_init(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_soc_codec *codec = rtd->codec; - struct snd_soc_dapm_context *dapm = &codec->dapm; - - /* Add am3517-evm specific widgets */ - snd_soc_dapm_new_controls(dapm, tlv320aic23_dapm_widgets, - ARRAY_SIZE(tlv320aic23_dapm_widgets)); - - /* Set up davinci-evm specific audio path audio_map */ - snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - - /* always connected */ - snd_soc_dapm_enable_pin(dapm, "Line Out"); - snd_soc_dapm_enable_pin(dapm, "Line In"); - snd_soc_dapm_enable_pin(dapm, "Mic In"); - - return 0; -} - /* Digital audio interface glue - connects codec <--> CPU */ static struct snd_soc_dai_link am3517evm_dai = { .name = "TLV320AIC23", @@ -120,7 +100,6 @@ static struct snd_soc_dai_link am3517evm_dai = { .codec_name = "tlv320aic23-codec.2-001a", .dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM, - .init = am3517evm_aic23_init, .ops = &am3517evm_ops, }; @@ -129,6 +108,11 @@ static struct snd_soc_card snd_soc_am3517evm = { .name = "am3517evm", .dai_link = &am3517evm_dai, .num_links = 1, + + .dapm_widgets = tlv320aic23_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(tlv320aic23_dapm_widgets), + .dapm_routes = audio_map, + .num_dapm_routes = ARRAY_SIZE(audio_map), }; static struct platform_device *am3517evm_snd_device; -- cgit v1.1 From 8966c2dd51909e6e3dcea099df9c501904085818 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Mon, 10 Oct 2011 15:34:10 +0300 Subject: ASoC: n810: Let the core to register DAPM widgets/routes and controls Pass the DAPM widgets/routes and static controls via the snd_soc_card struct to core. In this way the machine driver does not need to handle the DAPM widgets/routes. Signed-off-by: Peter Ujfalusi Cc: Jarkko Nikula Signed-off-by: Mark Brown --- sound/soc/omap/n810.c | 21 ++++++++------------- 1 file changed, 8 insertions(+), 13 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c index 4fa881b..7e3c20c 100644 --- a/sound/soc/omap/n810.c +++ b/sound/soc/omap/n810.c @@ -257,7 +257,6 @@ static int n810_aic33_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; struct snd_soc_dapm_context *dapm = &codec->dapm; - int err; /* Not connected */ snd_soc_dapm_nc_pin(dapm, "MONO_LOUT"); @@ -269,18 +268,7 @@ static int n810_aic33_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_nc_pin(dapm, "LINE2L"); snd_soc_dapm_nc_pin(dapm, "LINE2R"); - /* Add N810 specific controls */ - err = snd_soc_add_controls(codec, aic33_n810_controls, - ARRAY_SIZE(aic33_n810_controls)); - if (err < 0) - return err; - - /* Add N810 specific widgets */ - snd_soc_dapm_new_controls(dapm, aic33_dapm_widgets, - ARRAY_SIZE(aic33_dapm_widgets)); - - /* Set up N810 specific audio path audio_map */ - return snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); + return 0; } /* Digital audio interface glue - connects codec <--> CPU */ @@ -302,6 +290,13 @@ static struct snd_soc_card snd_soc_n810 = { .name = "N810", .dai_link = &n810_dai, .num_links = 1, + + .controls = aic33_n810_controls, + .num_controls = ARRAY_SIZE(aic33_n810_controls), + .dapm_widgets = aic33_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(aic33_dapm_widgets), + .dapm_routes = audio_map, + .num_dapm_routes = ARRAY_SIZE(audio_map), }; static struct platform_device *n810_snd_device; -- cgit v1.1 From 93b4d790b748a3475fec2ac962904b36874f0358 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Mon, 10 Oct 2011 15:34:11 +0300 Subject: ASoC: osk5912: Let core to deal with the DAPM widgets Pass the DAPM widgets/routes via the snd_soc_card struct to core. With this change we do not need the init function since the remaining snd_soc_dapm_enable_pin calls are not needed. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/omap/osk5912.c | 25 +++++-------------------- 1 file changed, 5 insertions(+), 20 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/osk5912.c b/sound/soc/omap/osk5912.c index 18f8a2d..db91cca 100644 --- a/sound/soc/omap/osk5912.c +++ b/sound/soc/omap/osk5912.c @@ -91,25 +91,6 @@ static const struct snd_soc_dapm_route audio_map[] = { {"MICIN", NULL, "Mic Jack"}, }; -static int osk_tlv320aic23_init(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_soc_codec *codec = rtd->codec; - struct snd_soc_dapm_context *dapm = &codec->dapm; - - /* Add osk5912 specific widgets */ - snd_soc_dapm_new_controls(dapm, tlv320aic23_dapm_widgets, - ARRAY_SIZE(tlv320aic23_dapm_widgets)); - - /* Set up osk5912 specific audio path audio_map */ - snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - - snd_soc_dapm_enable_pin(dapm, "Headphone Jack"); - snd_soc_dapm_enable_pin(dapm, "Line In"); - snd_soc_dapm_enable_pin(dapm, "Mic Jack"); - - return 0; -} - /* Digital audio interface glue - connects codec <--> CPU */ static struct snd_soc_dai_link osk_dai = { .name = "TLV320AIC23", @@ -120,7 +101,6 @@ static struct snd_soc_dai_link osk_dai = { .codec_name = "tlv320aic23-codec", .dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM, - .init = osk_tlv320aic23_init, .ops = &osk_ops, }; @@ -129,6 +109,11 @@ static struct snd_soc_card snd_soc_card_osk = { .name = "OSK5912", .dai_link = &osk_dai, .num_links = 1, + + .dapm_widgets = tlv320aic23_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(tlv320aic23_dapm_widgets), + .dapm_routes = audio_map, + .num_dapm_routes = ARRAY_SIZE(audio_map), }; static struct platform_device *osk_snd_device; -- cgit v1.1 From f8cf149f3e1568da7ea9ec5148d18ed83ac530de Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Mon, 10 Oct 2011 15:34:12 +0300 Subject: ASoC: sdp3430: Let core to deal with the DAPM widgets Pass the DAPM widgets/routes via the snd_soc_card struct to core. Signed-off-by: Peter Ujfalusi Cc: Misael Lopez Cruz Signed-off-by: Mark Brown --- sound/soc/omap/sdp3430.c | 14 +++++--------- 1 file changed, 5 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/sdp3430.c b/sound/soc/omap/sdp3430.c index 85e2e918..4f1969d 100644 --- a/sound/soc/omap/sdp3430.c +++ b/sound/soc/omap/sdp3430.c @@ -129,15 +129,6 @@ static int sdp3430_twl4030_init(struct snd_soc_pcm_runtime *rtd) struct snd_soc_dapm_context *dapm = &codec->dapm; int ret; - /* Add SDP3430 specific widgets */ - ret = snd_soc_dapm_new_controls(dapm, sdp3430_twl4030_dapm_widgets, - ARRAY_SIZE(sdp3430_twl4030_dapm_widgets)); - if (ret) - return ret; - - /* Set up SDP3430 specific audio path audio_map */ - snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - /* SDP3430 connected pins */ snd_soc_dapm_enable_pin(dapm, "Ext Mic"); snd_soc_dapm_enable_pin(dapm, "Ext Spk"); @@ -223,6 +214,11 @@ static struct snd_soc_card snd_soc_sdp3430 = { .name = "SDP3430", .dai_link = sdp3430_dai, .num_links = ARRAY_SIZE(sdp3430_dai), + + .dapm_widgets = sdp3430_twl4030_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(sdp3430_twl4030_dapm_widgets), + .dapm_routes = audio_map, + .num_dapm_routes = ARRAY_SIZE(audio_map), }; static struct platform_device *sdp3430_snd_device; -- cgit v1.1 From 35f0678ef7fc3a85d032bc870d1d11af6089e843 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 8 Oct 2011 12:02:13 +0100 Subject: ASoC: Convert Goni to data based DAPM init Signed-off-by: Mark Brown Acked-by: Kyungmin Park --- sound/soc/samsung/goni_wm8994.c | 13 +++++-------- 1 file changed, 5 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/goni_wm8994.c b/sound/soc/samsung/goni_wm8994.c index 86ece1a..4a34f60 100644 --- a/sound/soc/samsung/goni_wm8994.c +++ b/sound/soc/samsung/goni_wm8994.c @@ -99,14 +99,6 @@ static int goni_wm8994_init(struct snd_soc_pcm_runtime *rtd) struct snd_soc_dapm_context *dapm = &codec->dapm; int ret; - /* add goni specific widgets */ - snd_soc_dapm_new_controls(dapm, goni_dapm_widgets, - ARRAY_SIZE(goni_dapm_widgets)); - - /* set up goni specific audio routes */ - snd_soc_dapm_add_routes(dapm, goni_dapm_routes, - ARRAY_SIZE(goni_dapm_routes)); - /* set endpoints to not connected */ snd_soc_dapm_nc_pin(dapm, "IN2LP:VXRN"); snd_soc_dapm_nc_pin(dapm, "IN2RP:VXRP"); @@ -253,6 +245,11 @@ static struct snd_soc_card goni = { .name = "goni", .dai_link = goni_dai, .num_links = ARRAY_SIZE(goni_dai), + + .dapm_widgets = goni_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(goni_dapm_widgets), + .dapm_routes = goni_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(goni_dapm_routes), }; static int __init goni_init(void) -- cgit v1.1 From 6f25e4eed9751460ee5f0ae9ff26e3a201261f71 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Tue, 11 Oct 2011 17:55:00 +0800 Subject: ASoC: Writing register default value for the reset register The WM8983 can be reset by performing a write of any value to the software reset register. To avoid writing to the software reset register while resume, we should write the same value in wm8983_reg_defs to software reset register in wm8983_probe(). The write to the reset register is suppressed by the cache restore code when it skips writes of default registers. Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/wm8983.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8983.c b/sound/soc/codecs/wm8983.c index 17f04ec..93ee284 100644 --- a/sound/soc/codecs/wm8983.c +++ b/sound/soc/codecs/wm8983.c @@ -1007,7 +1007,7 @@ static int wm8983_probe(struct snd_soc_codec *codec) return ret; } - ret = snd_soc_write(codec, WM8983_SOFTWARE_RESET, 0x8983); + ret = snd_soc_write(codec, WM8983_SOFTWARE_RESET, 0); if (ret < 0) { dev_err(codec->dev, "Failed to issue reset: %d\n", ret); return ret; -- cgit v1.1 From a175fce01b963581e22b286f9a1f106581a29226 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 11 Oct 2011 13:11:12 +0300 Subject: ASoC: twl6040: Convert to table based init Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl6040.c | 31 ++++++++++--------------------- 1 file changed, 10 insertions(+), 21 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 62edded..93f8a59 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -1183,18 +1183,6 @@ static const struct snd_soc_dapm_route intercon[] = { {"AUXR", NULL, "AUXR Playback"}, }; -static int twl6040_add_widgets(struct snd_soc_codec *codec) -{ - struct snd_soc_dapm_context *dapm = &codec->dapm; - - snd_soc_dapm_new_controls(dapm, twl6040_dapm_widgets, - ARRAY_SIZE(twl6040_dapm_widgets)); - snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon)); - snd_soc_dapm_new_widgets(dapm); - - return 0; -} - static int twl6040_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { @@ -1503,16 +1491,10 @@ static int twl6040_probe(struct snd_soc_codec *codec) /* power on device */ ret = twl6040_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - if (ret) - goto bias_err; - - snd_soc_add_controls(codec, twl6040_snd_controls, - ARRAY_SIZE(twl6040_snd_controls)); - twl6040_add_widgets(codec); - - return 0; + if (!ret) + return 0; -bias_err: + /* Error path */ free_irq(priv->plug_irq, codec); plugirq_err: destroy_workqueue(priv->workqueue); @@ -1544,6 +1526,13 @@ static struct snd_soc_codec_driver soc_codec_dev_twl6040 = { .reg_cache_size = ARRAY_SIZE(twl6040_reg), .reg_word_size = sizeof(u8), .reg_cache_default = twl6040_reg, + + .controls = twl6040_snd_controls, + .num_controls = ARRAY_SIZE(twl6040_snd_controls), + .dapm_widgets = twl6040_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(twl6040_dapm_widgets), + .dapm_routes = intercon, + .num_dapm_routes = ARRAY_SIZE(intercon), }; static int __devinit twl6040_codec_probe(struct platform_device *pdev) -- cgit v1.1 From f7c93f018d21ed0d9218535497922a21066212dc Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 11 Oct 2011 13:11:32 +0300 Subject: ASoC: twl4030: Convert to table based init Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 21 +++++++-------------- 1 file changed, 7 insertions(+), 14 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 7c244cd..f798247 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -1609,17 +1609,6 @@ static const struct snd_soc_dapm_route intercon[] = { }; -static int twl4030_add_widgets(struct snd_soc_codec *codec) -{ - struct snd_soc_dapm_context *dapm = &codec->dapm; - - snd_soc_dapm_new_controls(dapm, twl4030_dapm_widgets, - ARRAY_SIZE(twl4030_dapm_widgets)); - snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon)); - - return 0; -} - static int twl4030_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { @@ -2241,9 +2230,6 @@ static int twl4030_soc_probe(struct snd_soc_codec *codec) twl4030_init_chip(codec); - snd_soc_add_controls(codec, twl4030_snd_controls, - ARRAY_SIZE(twl4030_snd_controls)); - twl4030_add_widgets(codec); return 0; } @@ -2269,6 +2255,13 @@ static struct snd_soc_codec_driver soc_codec_dev_twl4030 = { .reg_cache_size = sizeof(twl4030_reg), .reg_word_size = sizeof(u8), .reg_cache_default = twl4030_reg, + + .controls = twl4030_snd_controls, + .num_controls = ARRAY_SIZE(twl4030_snd_controls), + .dapm_widgets = twl4030_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(twl4030_dapm_widgets), + .dapm_routes = intercon, + .num_dapm_routes = ARRAY_SIZE(intercon), }; static int __devinit twl4030_codec_probe(struct platform_device *pdev) -- cgit v1.1 From 8066eb55b5ed15d6ec366fb6bad16ddd18eaf048 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 11 Oct 2011 13:11:55 +0300 Subject: ASoC: tlv320dac33: Convert to table based init Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320dac33.c | 23 +++++++---------------- 1 file changed, 7 insertions(+), 16 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index 43ee3b1..3f4920d 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -627,18 +627,6 @@ static const struct snd_soc_dapm_route audio_map[] = { {"RIGHT_LO", NULL, "Codec Power"}, }; -static int dac33_add_widgets(struct snd_soc_codec *codec) -{ - struct snd_soc_dapm_context *dapm = &codec->dapm; - - snd_soc_dapm_new_controls(dapm, dac33_dapm_widgets, - ARRAY_SIZE(dac33_dapm_widgets)); - /* set up audio path interconnects */ - snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - - return 0; -} - static int dac33_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { @@ -1451,15 +1439,11 @@ static int dac33_soc_probe(struct snd_soc_codec *codec) } } - snd_soc_add_controls(codec, dac33_snd_controls, - ARRAY_SIZE(dac33_snd_controls)); /* Only add the FIFO controls, if we have valid IRQ number */ if (dac33->irq >= 0) snd_soc_add_controls(codec, dac33_mode_snd_controls, ARRAY_SIZE(dac33_mode_snd_controls)); - dac33_add_widgets(codec); - err_power: return ret; } @@ -1502,6 +1486,13 @@ static struct snd_soc_codec_driver soc_codec_dev_tlv320dac33 = { .remove = dac33_soc_remove, .suspend = dac33_soc_suspend, .resume = dac33_soc_resume, + + .controls = dac33_snd_controls, + .num_controls = ARRAY_SIZE(dac33_snd_controls), + .dapm_widgets = dac33_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(dac33_dapm_widgets), + .dapm_routes = audio_map, + .num_dapm_routes = ARRAY_SIZE(audio_map), }; #define DAC33_RATES (SNDRV_PCM_RATE_44100 | \ -- cgit v1.1 From 684a65d4fba9099ed132a3a9a698390e17df9000 Mon Sep 17 00:00:00 2001 From: Wolfram Sang Date: Tue, 11 Oct 2011 12:43:02 +0200 Subject: ASoC: Fix typo in Kconfig symbol for tlv320aic32x4 It is currently named "TVL" instead of "TLV". Signed-off-by: Wolfram Sang Cc: Javier Martin Cc: Liam Girdwood Cc: Mark Brown Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 4 ++-- sound/soc/codecs/Makefile | 2 +- sound/soc/imx/Kconfig | 2 +- 3 files changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 3449431..4584514 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -49,7 +49,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_STAC9766 if SND_SOC_AC97_BUS select SND_SOC_TLV320AIC23 if I2C select SND_SOC_TLV320AIC26 if SPI_MASTER - select SND_SOC_TVL320AIC32X4 if I2C + select SND_SOC_TLV320AIC32X4 if I2C select SND_SOC_TLV320AIC3X if I2C select SND_SOC_TPA6130A2 if I2C select SND_SOC_TLV320DAC33 if I2C @@ -249,7 +249,7 @@ config SND_SOC_TLV320AIC26 tristate "TI TLV320AIC26 Codec support" if SND_SOC_OF_SIMPLE depends on SPI -config SND_SOC_TVL320AIC32X4 +config SND_SOC_TLV320AIC32X4 tristate config SND_SOC_TLV320AIC3X diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 787881b..a2c7842 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -137,7 +137,7 @@ obj-$(CONFIG_SND_SOC_STAC9766) += snd-soc-stac9766.o obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o obj-$(CONFIG_SND_SOC_TLV320AIC26) += snd-soc-tlv320aic26.o obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o -obj-$(CONFIG_SND_SOC_TVL320AIC32X4) += snd-soc-tlv320aic32x4.o +obj-$(CONFIG_SND_SOC_TLV320AIC32X4) += snd-soc-tlv320aic32x4.o obj-$(CONFIG_SND_SOC_TLV320DAC33) += snd-soc-tlv320dac33.o obj-$(CONFIG_SND_SOC_TWL4030) += snd-soc-twl4030.o obj-$(CONFIG_SND_SOC_TWL6040) += snd-soc-twl6040.o diff --git a/sound/soc/imx/Kconfig b/sound/soc/imx/Kconfig index dcd954b..b133bfc 100644 --- a/sound/soc/imx/Kconfig +++ b/sound/soc/imx/Kconfig @@ -29,7 +29,7 @@ config SND_MXC_SOC_WM1133_EV1 config SND_SOC_MX27VIS_AIC32X4 tristate "SoC audio support for Visstrim M10 boards" depends on MACH_IMX27_VISSTRIM_M10 - select SND_SOC_TVL320AIC32X4 + select SND_SOC_TLV320AIC32X4 select SND_MXC_SOC_MX2 help Say Y if you want to add support for SoC audio on Visstrim SM10 -- cgit v1.1 From 3a53d827292b657afcb73495cac139371cb157e1 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 11 Oct 2011 15:37:38 +0100 Subject: ASoC: Add missing default for WM5100 Clocking 1 Signed-off-by: Mark Brown --- sound/soc/codecs/wm5100-tables.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm5100-tables.c b/sound/soc/codecs/wm5100-tables.c index 960617b..e9ce81a 100644 --- a/sound/soc/codecs/wm5100-tables.c +++ b/sound/soc/codecs/wm5100-tables.c @@ -794,6 +794,7 @@ u16 wm5100_reg_defaults[WM5100_MAX_REGISTER + 1] = { [0x0030] = 0x0000, /* R48 - PWM Drive 1 */ [0x0031] = 0x0100, /* R49 - PWM Drive 2 */ [0x0032] = 0x0100, /* R50 - PWM Drive 3 */ + [0x0100] = 0x0002, /* R256 - Clocking 1 */ [0x0101] = 0x0000, /* R257 - Clocking 3 */ [0x0102] = 0x0011, /* R258 - Clocking 4 */ [0x0103] = 0x0011, /* R259 - Clocking 5 */ -- cgit v1.1 From ba896ede9a9a54a9114ee2a4fe534328078c6b02 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 27 Sep 2011 17:39:50 +0100 Subject: ASoC: Implement WM5100 accessory detection support The WM5100 includes an advanced, low power, accessory detect subsystem capable of detecting both accessory presence and button presses while the device is in an ultra low power mode. Implement initial support for this. Signed-off-by: Mark Brown --- sound/soc/codecs/wm5100.c | 162 ++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm5100.h | 2 + 2 files changed, 164 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index 8d90ba9..02c011d 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -26,6 +26,7 @@ #include #include #include +#include #include #include #include @@ -68,6 +69,11 @@ struct wm5100_priv { bool out_ena[2]; + struct snd_soc_jack *jack; + bool jack_detecting; + bool jack_mic; + int jack_mode; + struct wm5100_fll fll[2]; struct wm5100_pdata pdata; @@ -2113,6 +2119,159 @@ static int wm5100_dig_vu[] = { WM5100_DAC_DIGITAL_VOLUME_6R, }; +static void wm5100_set_detect_mode(struct snd_soc_codec *codec, int the_mode) +{ + struct wm5100_priv *wm5100 = snd_soc_codec_get_drvdata(codec); + struct wm5100_jack_mode *mode = &wm5100->pdata.jack_modes[the_mode]; + + BUG_ON(the_mode >= ARRAY_SIZE(wm5100->pdata.jack_modes)); + + gpio_set_value_cansleep(wm5100->pdata.hp_pol, mode->hp_pol); + snd_soc_update_bits(codec, WM5100_ACCESSORY_DETECT_MODE_1, + WM5100_ACCDET_BIAS_SRC_MASK | + WM5100_ACCDET_SRC, + (mode->bias << WM5100_ACCDET_BIAS_SRC_SHIFT) | + mode->micd_src << WM5100_ACCDET_SRC_SHIFT); + + wm5100->jack_mode = the_mode; + + dev_dbg(codec->dev, "Set microphone polarity to %d\n", + wm5100->jack_mode); +} + +static void wm5100_micd_irq(struct snd_soc_codec *codec) +{ + struct wm5100_priv *wm5100 = snd_soc_codec_get_drvdata(codec); + int val; + + val = snd_soc_read(codec, WM5100_MIC_DETECT_3); + + dev_dbg(codec->dev, "Microphone event: %x\n", val); + + if (!(val & WM5100_ACCDET_VALID)) { + dev_warn(codec->dev, "Microphone detection state invalid\n"); + return; + } + + /* No accessory, reset everything and report removal */ + if (!(val & WM5100_ACCDET_STS)) { + dev_dbg(codec->dev, "Jack removal detected\n"); + wm5100->jack_mic = false; + wm5100->jack_detecting = true; + snd_soc_jack_report(wm5100->jack, 0, + SND_JACK_LINEOUT | SND_JACK_HEADSET | + SND_JACK_BTN_0); + + snd_soc_update_bits(codec, WM5100_MIC_DETECT_1, + WM5100_ACCDET_RATE_MASK, + WM5100_ACCDET_RATE_MASK); + return; + } + + /* If the measurement is very high we've got a microphone, + * either we just detected one or if we already reported then + * we've got a button release event. + */ + if (val & 0x400) { + if (wm5100->jack_detecting) { + dev_dbg(codec->dev, "Microphone detected\n"); + wm5100->jack_mic = true; + snd_soc_jack_report(wm5100->jack, + SND_JACK_HEADSET, + SND_JACK_HEADSET | SND_JACK_BTN_0); + + /* Increase poll rate to give better responsiveness + * for buttons */ + snd_soc_update_bits(codec, WM5100_MIC_DETECT_1, + WM5100_ACCDET_RATE_MASK, + 5 << WM5100_ACCDET_RATE_SHIFT); + } else { + dev_dbg(codec->dev, "Mic button up\n"); + snd_soc_jack_report(wm5100->jack, 0, SND_JACK_BTN_0); + } + + return; + } + + /* If we detected a lower impedence during initial startup + * then we probably have the wrong polarity, flip it. Don't + * do this for the lowest impedences to speed up detection of + * plain headphones. + */ + if (wm5100->jack_detecting && (val & 0x3f8)) { + wm5100_set_detect_mode(codec, !wm5100->jack_mode); + + return; + } + + /* Don't distinguish between buttons, just report any low + * impedence as BTN_0. + */ + if (val & 0x3fc) { + if (wm5100->jack_mic) { + dev_dbg(codec->dev, "Mic button detected\n"); + snd_soc_jack_report(wm5100->jack, SND_JACK_BTN_0, + SND_JACK_BTN_0); + } else if (wm5100->jack_detecting) { + dev_dbg(codec->dev, "Headphone detected\n"); + snd_soc_jack_report(wm5100->jack, SND_JACK_HEADPHONE, + SND_JACK_HEADPHONE); + + /* Increase the detection rate a bit for + * responsiveness. + */ + snd_soc_update_bits(codec, WM5100_MIC_DETECT_1, + WM5100_ACCDET_RATE_MASK, + 7 << WM5100_ACCDET_RATE_SHIFT); + } + } +} + +int wm5100_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack) +{ + struct wm5100_priv *wm5100 = snd_soc_codec_get_drvdata(codec); + + if (jack) { + wm5100->jack = jack; + wm5100->jack_detecting = true; + + wm5100_set_detect_mode(codec, 0); + + /* Slowest detection rate, gives debounce for initial + * detection */ + snd_soc_update_bits(codec, WM5100_MIC_DETECT_1, + WM5100_ACCDET_BIAS_STARTTIME_MASK | + WM5100_ACCDET_RATE_MASK, + (7 << WM5100_ACCDET_BIAS_STARTTIME_SHIFT) | + WM5100_ACCDET_RATE_MASK); + + /* We need the charge pump to power MICBIAS */ + snd_soc_dapm_force_enable_pin(&codec->dapm, "CP2"); + snd_soc_dapm_force_enable_pin(&codec->dapm, "SYSCLK"); + snd_soc_dapm_sync(&codec->dapm); + + /* We start off just enabling microphone detection - even a + * plain headphone will trigger detection. + */ + snd_soc_update_bits(codec, WM5100_MIC_DETECT_1, + WM5100_ACCDET_ENA, WM5100_ACCDET_ENA); + + snd_soc_update_bits(codec, WM5100_INTERRUPT_STATUS_3_MASK, + WM5100_IM_ACCDET_EINT, 0); + } else { + snd_soc_update_bits(codec, WM5100_INTERRUPT_STATUS_3_MASK, + WM5100_IM_HPDET_EINT | + WM5100_IM_ACCDET_EINT, + WM5100_IM_HPDET_EINT | + WM5100_IM_ACCDET_EINT); + snd_soc_update_bits(codec, WM5100_MIC_DETECT_1, + WM5100_ACCDET_ENA, 0); + wm5100->jack = NULL; + } + + return 0; +} + static irqreturn_t wm5100_irq(int irq, void *data) { struct snd_soc_codec *codec = data; @@ -2144,6 +2303,9 @@ static irqreturn_t wm5100_irq(int irq, void *data) complete(&wm5100->fll[1].lock); } + if (irq_val & WM5100_ACCDET_EINT) + wm5100_micd_irq(codec); + irq_val = snd_soc_read(codec, WM5100_INTERRUPT_STATUS_4); if (irq_val < 0) { dev_err(codec->dev, "Failed to read IRQ status 4: %d\n", diff --git a/sound/soc/codecs/wm5100.h b/sound/soc/codecs/wm5100.h index 345b3cf..fa32b12 100644 --- a/sound/soc/codecs/wm5100.h +++ b/sound/soc/codecs/wm5100.h @@ -16,6 +16,8 @@ #include +int wm5100_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack); + #define WM5100_CLK_AIF1 1 #define WM5100_CLK_AIF2 2 #define WM5100_CLK_AIF3 3 -- cgit v1.1 From b90d2f908465c800388714180217620c4a5b2fe4 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 10 Oct 2011 13:38:06 +0100 Subject: ASoC: Instantiate card widgets immediately This ensures they are available prior to the card late_probe(). Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 8dc9aba..d6af52b 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1430,6 +1430,8 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) snd_soc_dapm_add_routes(&card->dapm, card->dapm_routes, card->num_dapm_routes); + snd_soc_dapm_new_widgets(&card->dapm); + for (i = 0; i < card->num_links; i++) { dai_link = &card->dai_link[i]; -- cgit v1.1 From b91470bb374ed7db0448696ec85a3ed4785da2ee Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Tue, 11 Oct 2011 20:20:53 +0800 Subject: ASoC: ak4642: convert to soc-cache Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/ak4642.c | 82 +++++++---------------------------------------- 1 file changed, 12 insertions(+), 70 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index 65f4604..5e25191 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -156,7 +156,6 @@ static const struct snd_kcontrol_new ak4642_snd_controls[] = { struct ak4642_priv { unsigned int sysclk; enum snd_soc_control_type control_type; - void *control_data; }; /* @@ -175,64 +174,6 @@ static const u16 ak4642_reg[AK4642_CACHEREGNUM] = { 0x0000, }; -/* - * read ak4642 register cache - */ -static inline unsigned int ak4642_read_reg_cache(struct snd_soc_codec *codec, - unsigned int reg) -{ - u16 *cache = codec->reg_cache; - if (reg >= AK4642_CACHEREGNUM) - return -1; - return cache[reg]; -} - -/* - * write ak4642 register cache - */ -static inline void ak4642_write_reg_cache(struct snd_soc_codec *codec, - u16 reg, unsigned int value) -{ - u16 *cache = codec->reg_cache; - if (reg >= AK4642_CACHEREGNUM) - return; - - cache[reg] = value; -} - -/* - * write to the AK4642 register space - */ -static int ak4642_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) -{ - u8 data[2]; - - /* data is - * D15..D8 AK4642 register offset - * D7...D0 register data - */ - data[0] = reg & 0xff; - data[1] = value & 0xff; - - if (codec->hw_write(codec->control_data, data, 2) == 2) { - ak4642_write_reg_cache(codec, reg, value); - return 0; - } else - return -EIO; -} - -static int ak4642_sync(struct snd_soc_codec *codec) -{ - u16 *cache = codec->reg_cache; - int i, r = 0; - - for (i = 0; i < AK4642_CACHEREGNUM; i++) - r |= ak4642_write(codec, i, cache[i]); - - return r; -}; - static int ak4642_dai_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { @@ -252,8 +193,8 @@ static int ak4642_dai_startup(struct snd_pcm_substream *substream, */ snd_soc_update_bits(codec, MD_CTL4, DACH, DACH); snd_soc_update_bits(codec, MD_CTL3, BST1, BST1); - ak4642_write(codec, L_IVC, 0x91); /* volume */ - ak4642_write(codec, R_IVC, 0x91); /* volume */ + snd_soc_write(codec, L_IVC, 0x91); /* volume */ + snd_soc_write(codec, R_IVC, 0x91); /* volume */ snd_soc_update_bits(codec, PW_MGMT1, PMVCM | PMMIN | PMDAC, PMVCM | PMMIN | PMDAC); snd_soc_update_bits(codec, PW_MGMT2, PMHP_MASK, PMHP); @@ -272,9 +213,9 @@ static int ak4642_dai_startup(struct snd_pcm_substream *substream, * This operation came from example code of * "ASAHI KASEI AK4642" (japanese) manual p94. */ - ak4642_write(codec, SG_SL1, PMMP | MGAIN0); - ak4642_write(codec, TIMER, ZTM(0x3) | WTM(0x3)); - ak4642_write(codec, ALC_CTL1, ALC | LMTH0); + snd_soc_write(codec, SG_SL1, PMMP | MGAIN0); + snd_soc_write(codec, TIMER, ZTM(0x3) | WTM(0x3)); + snd_soc_write(codec, ALC_CTL1, ALC | LMTH0); snd_soc_update_bits(codec, PW_MGMT1, PMVCM | PMADL, PMVCM | PMADL); snd_soc_update_bits(codec, PW_MGMT3, PMADR, PMADR); @@ -462,7 +403,7 @@ static struct snd_soc_dai_driver ak4642_dai = { static int ak4642_resume(struct snd_soc_codec *codec) { - ak4642_sync(codec); + snd_soc_cache_sync(codec); return 0; } @@ -470,11 +411,15 @@ static int ak4642_resume(struct snd_soc_codec *codec) static int ak4642_probe(struct snd_soc_codec *codec) { struct ak4642_priv *ak4642 = snd_soc_codec_get_drvdata(codec); + int ret; dev_info(codec->dev, "AK4642 Audio Codec %s", AK4642_VERSION); - codec->hw_write = (hw_write_t)i2c_master_send; - codec->control_data = ak4642->control_data; + ret = snd_soc_codec_set_cache_io(codec, 8, 8, ak4642->control_type); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } snd_soc_add_controls(codec, ak4642_snd_controls, ARRAY_SIZE(ak4642_snd_controls)); @@ -485,8 +430,6 @@ static int ak4642_probe(struct snd_soc_codec *codec) static struct snd_soc_codec_driver soc_codec_dev_ak4642 = { .probe = ak4642_probe, .resume = ak4642_resume, - .read = ak4642_read_reg_cache, - .write = ak4642_write, .reg_cache_size = ARRAY_SIZE(ak4642_reg), .reg_word_size = sizeof(u8), .reg_cache_default = ak4642_reg, @@ -504,7 +447,6 @@ static __devinit int ak4642_i2c_probe(struct i2c_client *i2c, return -ENOMEM; i2c_set_clientdata(i2c, ak4642); - ak4642->control_data = i2c; ak4642->control_type = SND_SOC_I2C; ret = snd_soc_register_codec(&i2c->dev, -- cgit v1.1 From 75b9a5782e2fcfb5f653bfc96d29d5e351b5b3b1 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Wed, 12 Oct 2011 06:57:25 +0800 Subject: ASoC: Delete ads117x.h This is not required after multi-component patch. Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/ads117x.h | 13 ------------- 1 file changed, 13 deletions(-) delete mode 100644 sound/soc/codecs/ads117x.h (limited to 'sound') diff --git a/sound/soc/codecs/ads117x.h b/sound/soc/codecs/ads117x.h deleted file mode 100644 index 3ce0286..0000000 --- a/sound/soc/codecs/ads117x.h +++ /dev/null @@ -1,13 +0,0 @@ -/* - * ads117x.h -- Driver for ads1174/8 ADC chips - * - * Copyright 2009 ShotSpotter Inc. - * Author: Graeme Gregory - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - */ -extern struct snd_soc_dai_driver ads117x_dai; -extern struct snd_soc_codec_driver soc_codec_dev_ads117x; -- cgit v1.1 From 48df93d4c73b95c3936beab4c69d4c522a29dca3 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Wed, 12 Oct 2011 07:03:09 +0800 Subject: ASoC: Remove impossible case from wm8994_hw_params We set hw_params callback for wm8994_aif3_dai_ops to wm8994_aif3_hw_params. Thus no need to check wm8994-aif3 in wm8994_hw_params. Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 10 ---------- 1 file changed, 10 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 16542de..68e769e 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2235,7 +2235,6 @@ static int wm8994_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct snd_soc_codec *codec = dai->codec; - struct wm8994 *control = codec->control_data; struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); int aif1_reg; int aif2_reg; @@ -2278,15 +2277,6 @@ static int wm8994_hw_params(struct snd_pcm_substream *substream, dev_dbg(codec->dev, "AIF2 using split LRCLK\n"); } break; - case 3: - switch (control->type) { - case WM1811: - case WM8958: - aif1_reg = WM8958_AIF3_CONTROL_1; - break; - default: - return 0; - } default: return -EINVAL; } -- cgit v1.1 From 40a49710107c237a2f4362c8b8bf07df3bac53dd Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Wed, 12 Oct 2011 07:16:25 +0800 Subject: ASoC: da7210: convert to soc-cache Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/da7210.c | 123 +++++++++++++++++----------------------------- 1 file changed, 45 insertions(+), 78 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index 92fd9d7..a9d9d39 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -161,7 +161,6 @@ static const struct snd_kcontrol_new da7210_snd_controls[] = { /* Codec private data */ struct da7210_priv { enum snd_soc_control_type control_type; - void *control_data; }; /* @@ -188,50 +187,16 @@ static const u8 da7210_reg[] = { 0x00, /* R88 */ }; -/* - * Read da7210 register cache - */ -static inline u32 da7210_read_reg_cache(struct snd_soc_codec *codec, u32 reg) -{ - u8 *cache = codec->reg_cache; - BUG_ON(reg >= ARRAY_SIZE(da7210_reg)); - return cache[reg]; -} - -/* - * Write to the da7210 register space - */ -static int da7210_write(struct snd_soc_codec *codec, u32 reg, u32 value) -{ - u8 *cache = codec->reg_cache; - u8 data[2]; - - BUG_ON(codec->driver->volatile_register); - - data[0] = reg & 0xff; - data[1] = value & 0xff; - - if (reg >= codec->driver->reg_cache_size) - return -EIO; - - if (2 != codec->hw_write(codec->control_data, data, 2)) - return -EIO; - - cache[reg] = value; - return 0; -} - -/* - * Read from the da7210 register space. - */ -static inline u32 da7210_read(struct snd_soc_codec *codec, u32 reg) +static int da7210_volatile_register(struct snd_soc_codec *codec, + unsigned int reg) { - if (DA7210_STATUS == reg) - return i2c_smbus_read_byte_data(codec->control_data, reg); - - return da7210_read_reg_cache(codec, reg); + switch (reg) { + case DA7210_STATUS: + return 1; + default: + return 0; + } } - static int da7210_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { @@ -270,13 +235,13 @@ static int da7210_hw_params(struct snd_pcm_substream *substream, u32 fs, bypass; /* set DAI source to Left and Right ADC */ - da7210_write(codec, DA7210_DAI_SRC_SEL, + snd_soc_write(codec, DA7210_DAI_SRC_SEL, DA7210_DAI_OUT_R_SRC | DA7210_DAI_OUT_L_SRC); /* Enable DAI */ - da7210_write(codec, DA7210_DAI_CFG3, DA7210_DAI_OE | DA7210_DAI_EN); + snd_soc_write(codec, DA7210_DAI_CFG3, DA7210_DAI_OE | DA7210_DAI_EN); - dai_cfg1 = 0xFC & da7210_read(codec, DA7210_DAI_CFG1); + dai_cfg1 = 0xFC & snd_soc_read(codec, DA7210_DAI_CFG1); switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: @@ -289,7 +254,7 @@ static int da7210_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - da7210_write(codec, DA7210_DAI_CFG1, dai_cfg1); + snd_soc_write(codec, DA7210_DAI_CFG1, dai_cfg1); hpf_reg = (SNDRV_PCM_STREAM_PLAYBACK == substream->stream) ? DA7210_DAC_HPF : DA7210_ADC_HPF; @@ -382,8 +347,8 @@ static int da7210_set_dai_fmt(struct snd_soc_dai *codec_dai, u32 fmt) u32 dai_cfg1; u32 dai_cfg3; - dai_cfg1 = 0x7f & da7210_read(codec, DA7210_DAI_CFG1); - dai_cfg3 = 0xfc & da7210_read(codec, DA7210_DAI_CFG3); + dai_cfg1 = 0x7f & snd_soc_read(codec, DA7210_DAI_CFG1); + dai_cfg3 = 0xfc & snd_soc_read(codec, DA7210_DAI_CFG3); switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBM_CFM: @@ -411,8 +376,8 @@ static int da7210_set_dai_fmt(struct snd_soc_dai *codec_dai, u32 fmt) */ dai_cfg1 |= DA7210_DAI_FLEN_64BIT; - da7210_write(codec, DA7210_DAI_CFG1, dai_cfg1); - da7210_write(codec, DA7210_DAI_CFG3, dai_cfg3); + snd_soc_write(codec, DA7210_DAI_CFG1, dai_cfg1); + snd_soc_write(codec, DA7210_DAI_CFG3, dai_cfg3); return 0; } @@ -451,11 +416,15 @@ static struct snd_soc_dai_driver da7210_dai = { static int da7210_probe(struct snd_soc_codec *codec) { struct da7210_priv *da7210 = snd_soc_codec_get_drvdata(codec); + int ret; dev_info(codec->dev, "DA7210 Audio Codec %s\n", DA7210_VERSION); - codec->control_data = da7210->control_data; - codec->hw_write = (hw_write_t)i2c_master_send; + ret = snd_soc_codec_set_cache_io(codec, 8, 8, da7210->control_type); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } /* FIXME * @@ -472,8 +441,8 @@ static int da7210_probe(struct snd_soc_codec *codec) /* * make sure that DA7210 use bypass mode before start up */ - da7210_write(codec, DA7210_STARTUP1, 0); - da7210_write(codec, DA7210_PLL_DIV3, + snd_soc_write(codec, DA7210_STARTUP1, 0); + snd_soc_write(codec, DA7210_PLL_DIV3, DA7210_MCLK_RANGE_10_20_MHZ | DA7210_PLL_BYP); /* @@ -481,36 +450,36 @@ static int da7210_probe(struct snd_soc_codec *codec) */ /* Enable Left & Right MIC PGA and Mic Bias */ - da7210_write(codec, DA7210_MIC_L, DA7210_MIC_L_EN | DA7210_MICBIAS_EN); - da7210_write(codec, DA7210_MIC_R, DA7210_MIC_R_EN); + snd_soc_write(codec, DA7210_MIC_L, DA7210_MIC_L_EN | DA7210_MICBIAS_EN); + snd_soc_write(codec, DA7210_MIC_R, DA7210_MIC_R_EN); /* Enable Left and Right input PGA */ - da7210_write(codec, DA7210_INMIX_L, DA7210_IN_L_EN); - da7210_write(codec, DA7210_INMIX_R, DA7210_IN_R_EN); + snd_soc_write(codec, DA7210_INMIX_L, DA7210_IN_L_EN); + snd_soc_write(codec, DA7210_INMIX_R, DA7210_IN_R_EN); /* Enable Left and Right ADC */ - da7210_write(codec, DA7210_ADC, DA7210_ADC_L_EN | DA7210_ADC_R_EN); + snd_soc_write(codec, DA7210_ADC, DA7210_ADC_L_EN | DA7210_ADC_R_EN); /* * DAC settings */ /* Enable Left and Right DAC */ - da7210_write(codec, DA7210_DAC_SEL, + snd_soc_write(codec, DA7210_DAC_SEL, DA7210_DAC_L_SRC_DAI_L | DA7210_DAC_L_EN | DA7210_DAC_R_SRC_DAI_R | DA7210_DAC_R_EN); /* Enable Left and Right out PGA */ - da7210_write(codec, DA7210_OUTMIX_L, DA7210_OUT_L_EN); - da7210_write(codec, DA7210_OUTMIX_R, DA7210_OUT_R_EN); + snd_soc_write(codec, DA7210_OUTMIX_L, DA7210_OUT_L_EN); + snd_soc_write(codec, DA7210_OUTMIX_R, DA7210_OUT_R_EN); /* Enable Left and Right HeadPhone PGA */ - da7210_write(codec, DA7210_HP_CFG, + snd_soc_write(codec, DA7210_HP_CFG, DA7210_HP_2CAP_MODE | DA7210_HP_SENSE_EN | DA7210_HP_L_EN | DA7210_HP_MODE | DA7210_HP_R_EN); /* Diable PLL and bypass it */ - da7210_write(codec, DA7210_PLL, DA7210_PLL_FS_48000); + snd_soc_write(codec, DA7210_PLL, DA7210_PLL_FS_48000); /* * If 48kHz sound came, it use bypass mode, @@ -521,22 +490,22 @@ static int da7210_probe(struct snd_soc_codec *codec) * DA7210_PLL_DIV3 :: DA7210_PLL_BYP bit. * see da7210_hw_params */ - da7210_write(codec, DA7210_PLL_DIV1, 0xE5); /* MCLK = 12.288MHz */ - da7210_write(codec, DA7210_PLL_DIV2, 0x99); - da7210_write(codec, DA7210_PLL_DIV3, 0x0A | + snd_soc_write(codec, DA7210_PLL_DIV1, 0xE5); /* MCLK = 12.288MHz */ + snd_soc_write(codec, DA7210_PLL_DIV2, 0x99); + snd_soc_write(codec, DA7210_PLL_DIV3, 0x0A | DA7210_MCLK_RANGE_10_20_MHZ | DA7210_PLL_BYP); snd_soc_update_bits(codec, DA7210_PLL, DA7210_PLL_EN, DA7210_PLL_EN); /* As suggested by Dialog */ - da7210_write(codec, DA7210_A_HID_UNLOCK, 0x8B); /* unlock */ - da7210_write(codec, DA7210_A_TEST_UNLOCK, 0xB4); - da7210_write(codec, DA7210_A_PLL1, 0x01); - da7210_write(codec, DA7210_A_CP_MODE, 0x7C); - da7210_write(codec, DA7210_A_HID_UNLOCK, 0x00); /* re-lock */ - da7210_write(codec, DA7210_A_TEST_UNLOCK, 0x00); + snd_soc_write(codec, DA7210_A_HID_UNLOCK, 0x8B); /* unlock */ + snd_soc_write(codec, DA7210_A_TEST_UNLOCK, 0xB4); + snd_soc_write(codec, DA7210_A_PLL1, 0x01); + snd_soc_write(codec, DA7210_A_CP_MODE, 0x7C); + snd_soc_write(codec, DA7210_A_HID_UNLOCK, 0x00); /* re-lock */ + snd_soc_write(codec, DA7210_A_TEST_UNLOCK, 0x00); /* Activate all enabled subsystem */ - da7210_write(codec, DA7210_STARTUP1, DA7210_SC_MST_EN); + snd_soc_write(codec, DA7210_STARTUP1, DA7210_SC_MST_EN); snd_soc_add_controls(codec, da7210_snd_controls, ARRAY_SIZE(da7210_snd_controls)); @@ -548,11 +517,10 @@ static int da7210_probe(struct snd_soc_codec *codec) static struct snd_soc_codec_driver soc_codec_dev_da7210 = { .probe = da7210_probe, - .read = da7210_read, - .write = da7210_write, .reg_cache_size = ARRAY_SIZE(da7210_reg), .reg_word_size = sizeof(u8), .reg_cache_default = da7210_reg, + .volatile_register = da7210_volatile_register, }; #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) @@ -567,7 +535,6 @@ static int __devinit da7210_i2c_probe(struct i2c_client *i2c, return -ENOMEM; i2c_set_clientdata(i2c, da7210); - da7210->control_data = i2c; da7210->control_type = SND_SOC_I2C; ret = snd_soc_register_codec(&i2c->dev, -- cgit v1.1 From 67c341302f5a401a405be758250bada39746c96b Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Wed, 12 Oct 2011 11:57:57 +0300 Subject: ASoC: twl6040: Support for vibra output paths twl6040 have two vibra output drivers. They can be operated with audio stream coming through the PDM interface (fifth channel). The vibra outputs can be controlled via the input/FF driver as well. Selection between the two mode is implemented within the codec driver, the input/FF driver can only operate if the routing is set to "Input FF". Changing from "Input FF" to "Audio PDM" mode is protected as well: The switchin can only be done, if there is no running effect from the input/FF. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl6040.c | 72 ++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 72 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 93f8a59..8648498 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -803,6 +803,23 @@ static int twl6040_get_volsw(struct snd_kcontrol *kcontrol, return 0; } +static int twl6040_soc_dapm_put_vibra_enum(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol); + struct snd_soc_dapm_widget *widget = wlist->widgets[0]; + struct snd_soc_codec *codec = widget->codec; + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + unsigned int val; + + /* Do not allow changes while Input/FF efect is running */ + val = twl6040_read_reg_volatile(codec, e->reg); + if (val & TWL6040_VIBENA && !(val & TWL6040_VIBSEL)) + return -EBUSY; + + return snd_soc_dapm_put_enum_double(kcontrol, ucontrol); +} + /* * MICATT volume control: * from -6 to 0 dB in 6 dB steps @@ -874,6 +891,19 @@ static const struct soc_enum twl6040_hf_enum[] = { twl6040_hf_texts), }; +static const char *twl6040_vibrapath_texts[] = { + "Input FF", "Audio PDM" +}; + +static const struct soc_enum twl6040_vibra_enum[] = { + SOC_ENUM_SINGLE(TWL6040_REG_VIBCTLL, 1, + ARRAY_SIZE(twl6040_vibrapath_texts), + twl6040_vibrapath_texts), + SOC_ENUM_SINGLE(TWL6040_REG_VIBCTLR, 1, + ARRAY_SIZE(twl6040_vibrapath_texts), + twl6040_vibrapath_texts), +}; + static const struct snd_kcontrol_new amicl_control = SOC_DAPM_ENUM("Route", twl6040_enum[0]); @@ -903,6 +933,17 @@ static const struct snd_kcontrol_new auxl_switch_control = static const struct snd_kcontrol_new auxr_switch_control = SOC_DAPM_SINGLE("Switch", TWL6040_REG_HFRCTL, 6, 1, 0); +/* Vibra playback switches */ +static const struct snd_kcontrol_new vibral_mux_controls = + SOC_DAPM_ENUM_EXT("Route", twl6040_vibra_enum[0], + snd_soc_dapm_get_enum_double, + twl6040_soc_dapm_put_vibra_enum); + +static const struct snd_kcontrol_new vibrar_mux_controls = + SOC_DAPM_ENUM_EXT("Route", twl6040_vibra_enum[1], + snd_soc_dapm_get_enum_double, + twl6040_soc_dapm_put_vibra_enum); + /* Headset power mode */ static const char *twl6040_power_mode_texts[] = { "Low-Power", "High-Perfomance", @@ -1024,6 +1065,8 @@ static const struct snd_soc_dapm_widget twl6040_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("EP"), SND_SOC_DAPM_OUTPUT("AUXL"), SND_SOC_DAPM_OUTPUT("AUXR"), + SND_SOC_DAPM_OUTPUT("VIBRAL"), + SND_SOC_DAPM_OUTPUT("VIBRAR"), /* Analog input muxes for the capture amplifiers */ SND_SOC_DAPM_MUX("Analog Left Capture Route", @@ -1076,6 +1119,9 @@ static const struct snd_soc_dapm_widget twl6040_dapm_widgets[] = { TWL6040_REG_HFRCTL, 0, 0, twl6040_power_mode_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + /* Virtual DAC for vibra path (DL4 channel) */ + SND_SOC_DAPM_DAC("VIBRA DAC", "Vibra Playback", + SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_MUX("Handsfree Left Playback", SND_SOC_NOPM, 0, 0, &hfl_mux_controls), @@ -1087,6 +1133,11 @@ static const struct snd_soc_dapm_widget twl6040_dapm_widgets[] = { SND_SOC_DAPM_MUX("Headset Right Playback", SND_SOC_NOPM, 0, 0, &hsr_mux_controls), + SND_SOC_DAPM_MUX("Vibra Left Playback", SND_SOC_NOPM, 0, 0, + &vibral_mux_controls), + SND_SOC_DAPM_MUX("Vibra Right Playback", SND_SOC_NOPM, 0, 0, + &vibrar_mux_controls), + SND_SOC_DAPM_SWITCH("Earphone Playback", SND_SOC_NOPM, 0, 0, &ep_path_enable_control), SND_SOC_DAPM_SWITCH("AUXL Playback", SND_SOC_NOPM, 0, 0, @@ -1115,6 +1166,15 @@ static const struct snd_soc_dapm_widget twl6040_dapm_widgets[] = { TWL6040_REG_EARCTL, 0, 0, NULL, 0, twl6040_power_mode_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_OUT_DRV("Vibra Left Driver", + TWL6040_REG_VIBCTLL, 0, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("Vibra Right Driver", + TWL6040_REG_VIBCTLR, 0, 0, NULL, 0), + + SND_SOC_DAPM_SUPPLY("Vibra Left Control", TWL6040_REG_VIBCTLL, 2, 0, + NULL, 0), + SND_SOC_DAPM_SUPPLY("Vibra Right Control", TWL6040_REG_VIBCTLR, 2, 0, + NULL, 0), /* Analog playback PGAs */ SND_SOC_DAPM_PGA("HF Left PGA", @@ -1181,6 +1241,18 @@ static const struct snd_soc_dapm_route intercon[] = { {"AUXL", NULL, "AUXL Playback"}, {"AUXR", NULL, "AUXR Playback"}, + + /* Vibrator paths */ + {"Vibra Left Playback", "Audio PDM", "VIBRA DAC"}, + {"Vibra Right Playback", "Audio PDM", "VIBRA DAC"}, + + {"Vibra Left Driver", NULL, "Vibra Left Playback"}, + {"Vibra Right Driver", NULL, "Vibra Right Playback"}, + {"Vibra Left Driver", NULL, "Vibra Left Control"}, + {"Vibra Right Driver", NULL, "Vibra Right Control"}, + + {"VIBRAL", NULL, "Vibra Left Driver"}, + {"VIBRAR", NULL, "Vibra Right Driver"}, }; static int twl6040_set_bias_level(struct snd_soc_codec *codec, -- cgit v1.1 From 33b6816ca3a4027a1b5444c83c1c24c0b1991262 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Wed, 12 Oct 2011 14:46:02 +0300 Subject: ASoC: twl6040: Workaround for headset DC offset caused pop noise Both Headset DAC need to be turned on/off at the same time before any of the output drivers are enabled (HS Left/Right, Earpiece). Move the HS DAC enable code to sequenced DAPM_SUPPLY, and attach it to the DACs. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl6040.c | 36 ++++++++++++++++++++++++++++-------- 1 file changed, 28 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 8648498..6369230 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -654,6 +654,26 @@ static int headset_power_mode(struct snd_soc_codec *codec, int high_perf) static int twl6040_hs_dac_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { + struct snd_soc_codec *codec = w->codec; + u8 hslctl, hsrctl; + + /* + * Workaround for Headset DC offset caused pop noise: + * Both HS DAC need to be turned on (before the HS driver) and off at + * the same time. + */ + hslctl = twl6040_read_reg_cache(codec, TWL6040_REG_HSLCTL); + hsrctl = twl6040_read_reg_cache(codec, TWL6040_REG_HSRCTL); + if (SND_SOC_DAPM_EVENT_ON(event)) { + hslctl |= TWL6040_HSDACENA; + hsrctl |= TWL6040_HSDACENA; + } else { + hslctl &= ~TWL6040_HSDACENA; + hsrctl &= ~TWL6040_HSDACENA; + } + twl6040_write(codec, TWL6040_REG_HSLCTL, hslctl); + twl6040_write(codec, TWL6040_REG_HSRCTL, hsrctl); + msleep(1); return 0; } @@ -1103,14 +1123,8 @@ static const struct snd_soc_dapm_widget twl6040_dapm_widgets[] = { TWL6040_REG_DMICBCTL, 4, 0), /* DACs */ - SND_SOC_DAPM_DAC_E("HSDAC Left", "Headset Playback", - TWL6040_REG_HSLCTL, 0, 0, - twl6040_hs_dac_event, - SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), - SND_SOC_DAPM_DAC_E("HSDAC Right", "Headset Playback", - TWL6040_REG_HSRCTL, 0, 0, - twl6040_hs_dac_event, - SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_DAC("HSDAC Left", "Headset Playback", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("HSDAC Right", "Headset Playback", SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_DAC_E("HFDAC Left", "Handsfree Playback", TWL6040_REG_HFLCTL, 0, 0, twl6040_power_mode_event, @@ -1175,6 +1189,9 @@ static const struct snd_soc_dapm_widget twl6040_dapm_widgets[] = { NULL, 0), SND_SOC_DAPM_SUPPLY("Vibra Right Control", TWL6040_REG_VIBCTLR, 2, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("HSDAC Power", 1, SND_SOC_NOPM, 0, 0, + twl6040_hs_dac_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), /* Analog playback PGAs */ SND_SOC_DAPM_PGA("HF Left PGA", @@ -1204,6 +1221,9 @@ static const struct snd_soc_dapm_route intercon[] = { {"AFMAmpL", NULL, "AFML"}, {"AFMAmpR", NULL, "AFMR"}, + {"HSDAC Left", NULL, "HSDAC Power"}, + {"HSDAC Right", NULL, "HSDAC Power"}, + {"Headset Left Playback", "HS DAC", "HSDAC Left"}, {"Headset Left Playback", "Line-In amp", "AFMAmpL"}, -- cgit v1.1 From 0f8ea586d7a9cfd6567b0cdfd73c5dfc2e8b9da8 Mon Sep 17 00:00:00 2001 From: Ashish Chavan Date: Wed, 12 Oct 2011 20:33:21 +0530 Subject: ASoC: da7210: Add support for other DAI word lengths, format and mode This patchs adds support for following, (1) DAI 20 and 32 bit word sizes (2) DAI left and right justified formats (3) DAI slave mode Signed-off-by: Ashish Chavan Signed-off-by: David Dajun Chen Signed-off-by: Mark Brown --- sound/soc/codecs/da7210.c | 23 ++++++++++++++++++++++- 1 file changed, 22 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index a9d9d39..ff68247 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -105,12 +105,17 @@ /* DAI_CFG1 bit fields */ #define DA7210_DAI_WORD_S16_LE (0 << 0) +#define DA7210_DAI_WORD_S20_3LE (1 << 0) #define DA7210_DAI_WORD_S24_LE (2 << 0) +#define DA7210_DAI_WORD_S32_LE (3 << 0) #define DA7210_DAI_FLEN_64BIT (1 << 2) +#define DA7210_DAI_MODE_SLAVE (0 << 7) #define DA7210_DAI_MODE_MASTER (1 << 7) /* DAI_CFG3 bit fields */ #define DA7210_DAI_FORMAT_I2SMODE (0 << 0) +#define DA7210_DAI_FORMAT_LEFT_J (1 << 0) +#define DA7210_DAI_FORMAT_RIGHT_J (2 << 0) #define DA7210_DAI_OE (1 << 3) #define DA7210_DAI_EN (1 << 7) @@ -247,9 +252,15 @@ static int da7210_hw_params(struct snd_pcm_substream *substream, case SNDRV_PCM_FORMAT_S16_LE: dai_cfg1 |= DA7210_DAI_WORD_S16_LE; break; + case SNDRV_PCM_FORMAT_S20_3LE: + dai_cfg1 |= DA7210_DAI_WORD_S20_3LE; + break; case SNDRV_PCM_FORMAT_S24_LE: dai_cfg1 |= DA7210_DAI_WORD_S24_LE; break; + case SNDRV_PCM_FORMAT_S32_LE: + dai_cfg1 |= DA7210_DAI_WORD_S32_LE; + break; default: return -EINVAL; } @@ -354,6 +365,9 @@ static int da7210_set_dai_fmt(struct snd_soc_dai *codec_dai, u32 fmt) case SND_SOC_DAIFMT_CBM_CFM: dai_cfg1 |= DA7210_DAI_MODE_MASTER; break; + case SND_SOC_DAIFMT_CBS_CFS: + dai_cfg1 |= DA7210_DAI_MODE_SLAVE; + break; default: return -EINVAL; } @@ -366,6 +380,12 @@ static int da7210_set_dai_fmt(struct snd_soc_dai *codec_dai, u32 fmt) case SND_SOC_DAIFMT_I2S: dai_cfg3 |= DA7210_DAI_FORMAT_I2SMODE; break; + case SND_SOC_DAIFMT_LEFT_J: + dai_cfg3 |= DA7210_DAI_FORMAT_LEFT_J; + break; + case SND_SOC_DAIFMT_RIGHT_J: + dai_cfg3 |= DA7210_DAI_FORMAT_RIGHT_J; + break; default: return -EINVAL; } @@ -382,7 +402,8 @@ static int da7210_set_dai_fmt(struct snd_soc_dai *codec_dai, u32 fmt) return 0; } -#define DA7210_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE) +#define DA7210_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) /* DAI operations */ static struct snd_soc_dai_ops da7210_dai_ops = { -- cgit v1.1 From b29a33a211a855e8e52c095af13e1d99ed60d07c Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Wed, 12 Oct 2011 21:43:08 +0800 Subject: ASoC: Make SND_SOC_SAARB and SND_SOC_TAVOREVB3 select MFD_88PM860X In saarb_pm860x_init() and evb3_pm860x_init(), we call pm860x_hs_jack_detect() and pm860x_mic_jack_detect() which in turn calls pm860x_set_bits(). Thus make SND_SOC_SAARB and SND_SOC_TAVOREVB3 select MFD_88PM860X. This patch fixes below build error if CONFIG_MFD_88PM860X is not configured. LD .tmp_vmlinux1 sound/built-in.o: In function `pm860x_write_reg_cache': last.c:(.text+0x29e9c): undefined reference to `pm860x_reg_write' sound/built-in.o: In function `pm860x_set_bias_level': last.c:(.text+0x29ecc): undefined reference to `pm860x_set_bits' last.c:(.text+0x29f00): undefined reference to `pm860x_reg_write' last.c:(.text+0x29f18): undefined reference to `pm860x_reg_write' sound/built-in.o: In function `pm860x_read_reg_cache': last.c:(.text+0x29f40): undefined reference to `pm860x_reg_read' sound/built-in.o: In function `pm860x_probe': last.c:(.text+0x2a034): undefined reference to `pm860x_bulk_read' sound/built-in.o: In function `pm860x_codec_handler': last.c:(.text+0x2a344): undefined reference to `pm860x_reg_read' last.c:(.text+0x2a354): undefined reference to `pm860x_reg_read' sound/built-in.o: In function `pm860x_mic_jack_detect': last.c:(.text+0x2a450): undefined reference to `pm860x_set_bits' sound/built-in.o: In function `pm860x_hs_jack_detect': last.c:(.text+0x2a4d0): undefined reference to `pm860x_set_bits' last.c:(.text+0x2a4f8): undefined reference to `pm860x_set_bits' last.c:(.text+0x2a510): undefined reference to `pm860x_set_bits' make: *** [.tmp_vmlinux1] Error 1 Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/pxa/Kconfig | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 33ebc46..ffd2242 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -121,6 +121,7 @@ config SND_PXA2XX_SOC_PALM27X config SND_SOC_SAARB tristate "SoC Audio support for Marvell Saarb" depends on SND_PXA2XX_SOC && MACH_SAARB + select MFD_88PM860X select SND_PXA_SOC_SSP select SND_SOC_88PM860X help @@ -130,6 +131,7 @@ config SND_SOC_SAARB config SND_SOC_TAVOREVB3 tristate "SoC Audio support for Marvell Tavor EVB3" depends on SND_PXA2XX_SOC && MACH_TAVOREVB3 + select MFD_88PM860X select SND_PXA_SOC_SSP select SND_SOC_88PM860X help -- cgit v1.1 From dbe37dbc1bf1629e517b7ef57429fb6af1f186af Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Thu, 13 Oct 2011 07:38:56 +0800 Subject: ASoC: pxa: Remove redundant snd_soc_dapm_sync() calls from machine drivers The core will sync DAPM as part of the card initialization, there is no need for machine drivers to do so during their setup. Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/pxa/poodle.c | 1 - sound/soc/pxa/spitz.c | 1 - 2 files changed, 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c index da3ae43..4c29bc1 100644 --- a/sound/soc/pxa/poodle.c +++ b/sound/soc/pxa/poodle.c @@ -265,7 +265,6 @@ static int poodle_wm8731_init(struct snd_soc_pcm_runtime *rtd) /* Set up poodle specific audio path audio_map */ snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_sync(dapm); return 0; } diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c index ce920e3..c2d6ff9 100644 --- a/sound/soc/pxa/spitz.c +++ b/sound/soc/pxa/spitz.c @@ -301,7 +301,6 @@ static int spitz_wm8750_init(struct snd_soc_pcm_runtime *rtd) /* Set up spitz specific audio paths */ snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_sync(dapm); return 0; } -- cgit v1.1 From f0bbc2b55f47f93286bb1b9ddbdb8ffed3572064 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Thu, 13 Oct 2011 14:40:08 +0800 Subject: ASoC: sta32x: Set reg_cache_default to sta32x_regs Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/sta32x.c | 5 +---- 1 file changed, 1 insertion(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c index 5c7def3..754b3ff 100644 --- a/sound/soc/codecs/sta32x.c +++ b/sound/soc/codecs/sta32x.c @@ -756,10 +756,6 @@ static int sta32x_probe(struct snd_soc_codec *codec) return ret; } - /* read reg reset values into cache */ - for (i = 0; i < STA32X_REGISTER_COUNT; i++) - snd_soc_cache_write(codec, i, sta32x_regs[i]); - /* preserve reset values of reserved register bits */ snd_soc_cache_write(codec, STA32X_CONFC, codec->hw_read(codec, STA32X_CONFC)); @@ -837,6 +833,7 @@ static const struct snd_soc_codec_driver sta32x_codec = { .resume = sta32x_resume, .reg_cache_size = STA32X_REGISTER_COUNT, .reg_word_size = sizeof(u8), + .reg_cache_default = sta32x_regs, .volatile_register = sta32x_reg_is_volatile, .set_bias_level = sta32x_set_bias_level, .controls = sta32x_snd_controls, -- cgit v1.1 From edf413f689e930011bf39ec726f704af99d7263b Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Thu, 13 Oct 2011 14:57:31 +0800 Subject: ASoC: sta32x: Write the register default value to cache for reserved registers Chip documentation explicitly requires that the reset values of reserved register bits are left untouched. codec->hw_read is broken now. Here we use below trick to avoid writing to reserved registers while resume. Write the register default value to cache for reserved registers, so the write to the these registers are suppressed by the cache restore code when it skips writes of default registers. Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/sta32x.c | 28 +++++++++++++--------------- 1 file changed, 13 insertions(+), 15 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c index 754b3ff..bb82408 100644 --- a/sound/soc/codecs/sta32x.c +++ b/sound/soc/codecs/sta32x.c @@ -756,21 +756,19 @@ static int sta32x_probe(struct snd_soc_codec *codec) return ret; } - /* preserve reset values of reserved register bits */ - snd_soc_cache_write(codec, STA32X_CONFC, - codec->hw_read(codec, STA32X_CONFC)); - snd_soc_cache_write(codec, STA32X_CONFE, - codec->hw_read(codec, STA32X_CONFE)); - snd_soc_cache_write(codec, STA32X_CONFF, - codec->hw_read(codec, STA32X_CONFF)); - snd_soc_cache_write(codec, STA32X_MMUTE, - codec->hw_read(codec, STA32X_MMUTE)); - snd_soc_cache_write(codec, STA32X_AUTO1, - codec->hw_read(codec, STA32X_AUTO1)); - snd_soc_cache_write(codec, STA32X_AUTO3, - codec->hw_read(codec, STA32X_AUTO3)); - snd_soc_cache_write(codec, STA32X_C3CFG, - codec->hw_read(codec, STA32X_C3CFG)); + /* Chip documentation explicitly requires that the reset values + * of reserved register bits are left untouched. + * Write the register default value to cache for reserved registers, + * so the write to the these registers are suppressed by the cache + * restore code when it skips writes of default registers. + */ + snd_soc_cache_write(codec, STA32X_CONFC, 0xc2); + snd_soc_cache_write(codec, STA32X_CONFE, 0xc2); + snd_soc_cache_write(codec, STA32X_CONFF, 0x5c); + snd_soc_cache_write(codec, STA32X_MMUTE, 0x10); + snd_soc_cache_write(codec, STA32X_AUTO1, 0x60); + snd_soc_cache_write(codec, STA32X_AUTO3, 0x00); + snd_soc_cache_write(codec, STA32X_C3CFG, 0x40); /* FIXME enable thermal warning adjustment and recovery */ snd_soc_update_bits(codec, STA32X_CONFA, -- cgit v1.1 From 19b115e523208a926813751aac8934cf3fc6085e Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 13 Oct 2011 02:03:54 -0700 Subject: ASoC: ak4642: fixup cache register table ak4642 register was 8bit, but cache table was defined as 16bit. ak4642 doesn't work correctry without this patch. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown Cc: stable@kernel.org --- sound/soc/codecs/ak4642.c | 22 +++++++++++----------- 1 file changed, 11 insertions(+), 11 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index 5e25191..d8fc044 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -161,17 +161,17 @@ struct ak4642_priv { /* * ak4642 register cache */ -static const u16 ak4642_reg[AK4642_CACHEREGNUM] = { - 0x0000, 0x0000, 0x0001, 0x0000, - 0x0002, 0x0000, 0x0000, 0x0000, - 0x00e1, 0x00e1, 0x0018, 0x0000, - 0x00e1, 0x0018, 0x0011, 0x0008, - 0x0000, 0x0000, 0x0000, 0x0000, - 0x0000, 0x0000, 0x0000, 0x0000, - 0x0000, 0x0000, 0x0000, 0x0000, - 0x0000, 0x0000, 0x0000, 0x0000, - 0x0000, 0x0000, 0x0000, 0x0000, - 0x0000, +static const u8 ak4642_reg[AK4642_CACHEREGNUM] = { + 0x00, 0x00, 0x01, 0x00, + 0x02, 0x00, 0x00, 0x00, + 0xe1, 0xe1, 0x18, 0x00, + 0xe1, 0x18, 0x11, 0x08, + 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, + 0x00, }; static int ak4642_dai_startup(struct snd_pcm_substream *substream, -- cgit v1.1 From 7c04241acbdaf97f1448dcccd27ea0fcd1a57684 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Thu, 13 Oct 2011 17:17:06 +0800 Subject: ASoC: ak4535: fixup cache register table ak4535_reg should be 8bit, but cache table is defined as 16bit. Signed-off-by: Axel Lin Signed-off-by: Mark Brown Cc: stable@kernel.org --- sound/soc/codecs/ak4535.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c index 68df32d..95d782d 100644 --- a/sound/soc/codecs/ak4535.c +++ b/sound/soc/codecs/ak4535.c @@ -39,11 +39,11 @@ struct ak4535_priv { /* * ak4535 register cache */ -static const u16 ak4535_reg[AK4535_CACHEREGNUM] = { - 0x0000, 0x0080, 0x0000, 0x0003, - 0x0002, 0x0000, 0x0011, 0x0001, - 0x0000, 0x0040, 0x0036, 0x0010, - 0x0000, 0x0000, 0x0057, 0x0000, +static const u8 ak4535_reg[AK4535_CACHEREGNUM] = { + 0x00, 0x80, 0x00, 0x03, + 0x02, 0x00, 0x11, 0x01, + 0x00, 0x40, 0x36, 0x10, + 0x00, 0x00, 0x57, 0x00, }; static const char *ak4535_mono_gain[] = {"+6dB", "-17dB"}; -- cgit v1.1 From 1cba77c16309e14565d4006bb4373a4866278663 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 12 Oct 2011 18:39:53 +0100 Subject: ASoC: Update WM5100 accessory detection for revision A Signed-off-by: Mark Brown --- sound/soc/codecs/wm5100.c | 3 +++ sound/soc/codecs/wm5100.h | 7 +++++++ 2 files changed, 10 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index 02c011d..5d88c99 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -2132,6 +2132,9 @@ static void wm5100_set_detect_mode(struct snd_soc_codec *codec, int the_mode) WM5100_ACCDET_SRC, (mode->bias << WM5100_ACCDET_BIAS_SRC_SHIFT) | mode->micd_src << WM5100_ACCDET_SRC_SHIFT); + snd_soc_update_bits(codec, WM5100_MISC_CONTROL, + WM5100_HPCOM_SRC, + mode->micd_src << WM5100_HPCOM_SRC_SHIFT); wm5100->jack_mode = the_mode; diff --git a/sound/soc/codecs/wm5100.h b/sound/soc/codecs/wm5100.h index fa32b12..9707596 100644 --- a/sound/soc/codecs/wm5100.h +++ b/sound/soc/codecs/wm5100.h @@ -96,6 +96,7 @@ int wm5100_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack); #define WM5100_MIC_DETECT_1 0x290 #define WM5100_MIC_DETECT_2 0x291 #define WM5100_MIC_DETECT_3 0x292 +#define WM5100_MISC_CONTROL 0x2BB #define WM5100_INPUT_ENABLES 0x301 #define WM5100_INPUT_ENABLES_STATUS 0x302 #define WM5100_IN1L_CONTROL 0x310 @@ -1389,6 +1390,12 @@ int wm5100_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack); #define WM5100_ACCDET_STS_WIDTH 1 /* ACCDET_STS */ /* + * R699 (0x2BB) - Misc Control + */ +#define WM5100_HPCOM_SRC 0x200 /* HPCOM_SRC */ +#define WM5100_HPCOM_SRC_SHIFT 9 /* HPCOM_SRC */ + +/* * R769 (0x301) - Input Enables */ #define WM5100_IN4L_ENA 0x0080 /* IN4L_ENA */ -- cgit v1.1 From f872826e940c0372221897981b0a60781c2212e5 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 13 Oct 2011 15:05:41 +0300 Subject: ASoC: twl6040: Remove Capture restriction for 17.64MHz sysclk Capture is supported in all PLL configuration. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl6040.c | 7 ------- 1 file changed, 7 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 6369230..d078099 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -1351,13 +1351,6 @@ static int twl6040_hw_params(struct snd_pcm_substream *substream, rate); return -EINVAL; } - /* Capture is not supported with 17.64MHz sysclk */ - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { - dev_err(codec->dev, - "capture mode is not supported at %dHz\n", - rate); - return -EINVAL; - } priv->sysclk = 17640000; break; case 8000: -- cgit v1.1 From fac2f3e4dccfd97f5146065540486dd3f6a2bca5 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 13 Oct 2011 15:05:42 +0300 Subject: ASoC: twl6040: Remove PLL usage restrictions There is no limitation dictated by outputs or inputs regarding to the selected PLL (LP/HP). Remove the checks for this, and allow all path with any PLL configuration. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl6040.c | 40 +++++++++------------------------------- 1 file changed, 9 insertions(+), 31 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index d078099..8f033f0 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -87,7 +87,6 @@ struct twl6040_data { int plug_irq; int codec_powered; int pll; - int non_lp; int pll_power_mode; int hs_power_mode; int hs_power_mode_locked; @@ -588,10 +587,6 @@ static int out_drv_event(struct snd_soc_dapm_widget *w, out->left_step = priv->hf_left_step; out->right_step = priv->hf_right_step; out->step_delay = 5; /* 5 ms between volume ramp steps */ - if (SND_SOC_DAPM_EVENT_ON(event)) - priv->non_lp++; - else - priv->non_lp--; break; default: return -1; @@ -686,18 +681,12 @@ static int twl6040_power_mode_event(struct snd_soc_dapm_widget *w, int ret = 0; if (SND_SOC_DAPM_EVENT_ON(event)) { - priv->non_lp++; - if (!strcmp(w->name, "Earphone Driver")) { - /* Earphone doesn't support low power mode */ - priv->hs_power_mode_locked = 1; - ret = headset_power_mode(codec, 1); - } + /* Earphone doesn't support low power mode */ + priv->hs_power_mode_locked = 1; + ret = headset_power_mode(codec, 1); } else { - priv->non_lp--; - if (!strcmp(w->name, "Earphone Driver")) { - priv->hs_power_mode_locked = 0; - ret = headset_power_mode(codec, priv->hs_power_mode); - } + priv->hs_power_mode_locked = 0; + ret = headset_power_mode(codec, priv->hs_power_mode); } msleep(1); @@ -1125,14 +1114,10 @@ static const struct snd_soc_dapm_widget twl6040_dapm_widgets[] = { /* DACs */ SND_SOC_DAPM_DAC("HSDAC Left", "Headset Playback", SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_DAC("HSDAC Right", "Headset Playback", SND_SOC_NOPM, 0, 0), - SND_SOC_DAPM_DAC_E("HFDAC Left", "Handsfree Playback", - TWL6040_REG_HFLCTL, 0, 0, - twl6040_power_mode_event, - SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), - SND_SOC_DAPM_DAC_E("HFDAC Right", "Handsfree Playback", - TWL6040_REG_HFRCTL, 0, 0, - twl6040_power_mode_event, - SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_DAC("HFDAC Left", "Handsfree Playback", + TWL6040_REG_HFLCTL, 0, 0), + SND_SOC_DAPM_DAC("HFDAC Right", "Handsfree Playback", + TWL6040_REG_HFRCTL, 0, 0), /* Virtual DAC for vibra path (DL4 channel) */ SND_SOC_DAPM_DAC("VIBRA DAC", "Vibra Playback", SND_SOC_NOPM, 0, 0), @@ -1383,13 +1368,6 @@ static int twl6040_prepare(struct snd_pcm_substream *substream, return -EINVAL; } - if ((priv->sysclk == 17640000) && priv->non_lp) { - dev_err(codec->dev, - "some enabled paths aren't supported at %dHz\n", - priv->sysclk); - return -EPERM; - } - ret = twl6040_set_pll(twl6040, priv->pll, priv->clk_in, priv->sysclk); if (ret) { dev_err(codec->dev, "Can not set PLL (%d)\n", ret); -- cgit v1.1 From aa1a41082fb8c47893186103bf53e96708041e1c Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 13 Oct 2011 15:05:43 +0300 Subject: ASoC: twl6040: Change event ordering for Earphone driver It is better to switch HS Power Mode (if it was in low power mode) before we enable the Earpiece driver. The switched off EP driver can filter out noise coming from the Low Power to High Performance transition on the HSL DAC. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl6040.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 8f033f0..eadece8 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -1164,7 +1164,7 @@ static const struct snd_soc_dapm_widget twl6040_dapm_widgets[] = { SND_SOC_DAPM_OUT_DRV_E("Earphone Driver", TWL6040_REG_EARCTL, 0, 0, NULL, 0, twl6040_power_mode_event, - SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_OUT_DRV("Vibra Left Driver", TWL6040_REG_VIBCTLL, 0, 0, NULL, 0), SND_SOC_DAPM_OUT_DRV("Vibra Right Driver", -- cgit v1.1 From 694b00010b0dfe727d485c3472cfe3ad7b91dcc2 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 13 Oct 2011 15:05:44 +0300 Subject: ASoC: twl6040: Rename the Earphone Driver event handler Since the event handler is only used by the Earphone Driver, it is better to rename it from twl6040_power_mode_event to twl6040_ep_drv_event. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl6040.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index eadece8..6c573c3 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -673,7 +673,7 @@ static int twl6040_hs_dac_event(struct snd_soc_dapm_widget *w, return 0; } -static int twl6040_power_mode_event(struct snd_soc_dapm_widget *w, +static int twl6040_ep_drv_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { struct snd_soc_codec *codec = w->codec; @@ -1163,7 +1163,7 @@ static const struct snd_soc_dapm_widget twl6040_dapm_widgets[] = { SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), SND_SOC_DAPM_OUT_DRV_E("Earphone Driver", TWL6040_REG_EARCTL, 0, 0, NULL, 0, - twl6040_power_mode_event, + twl6040_ep_drv_event, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_OUT_DRV("Vibra Left Driver", TWL6040_REG_VIBCTLL, 0, 0, NULL, 0), -- cgit v1.1 From bc6ae96a445fe527e32695135130ce4bd24b90d2 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Thu, 13 Oct 2011 22:56:34 +0800 Subject: ASoC: tlv320aic32x4: Use snd_soc_update_bits for read-modify-write Use snd_soc_update_bits for read-modify-write register access instead of open-coding it using snd_soc_read and snd_soc_write. Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic32x4.c | 61 ++++++++++++++++------------------------ 1 file changed, 24 insertions(+), 37 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index a68982e..b21c610 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -528,40 +528,33 @@ static int aic32x4_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec); - u8 value; switch (level) { case SND_SOC_BIAS_ON: if (aic32x4->master) { /* Switch on PLL */ - value = snd_soc_read(codec, AIC32X4_PLLPR); - snd_soc_write(codec, AIC32X4_PLLPR, - (value | AIC32X4_PLLEN)); + snd_soc_update_bits(codec, AIC32X4_PLLPR, + AIC32X4_PLLEN, AIC32X4_PLLEN); /* Switch on NDAC Divider */ - value = snd_soc_read(codec, AIC32X4_NDAC); - snd_soc_write(codec, AIC32X4_NDAC, - value | AIC32X4_NDACEN); + snd_soc_update_bits(codec, AIC32X4_NDAC, + AIC32X4_NDACEN, AIC32X4_NDACEN); /* Switch on MDAC Divider */ - value = snd_soc_read(codec, AIC32X4_MDAC); - snd_soc_write(codec, AIC32X4_MDAC, - value | AIC32X4_MDACEN); + snd_soc_update_bits(codec, AIC32X4_MDAC, + AIC32X4_MDACEN, AIC32X4_MDACEN); /* Switch on NADC Divider */ - value = snd_soc_read(codec, AIC32X4_NADC); - snd_soc_write(codec, AIC32X4_NADC, - value | AIC32X4_MDACEN); + snd_soc_update_bits(codec, AIC32X4_NADC, + AIC32X4_NADCEN, AIC32X4_NADCEN); /* Switch on MADC Divider */ - value = snd_soc_read(codec, AIC32X4_MADC); - snd_soc_write(codec, AIC32X4_MADC, - value | AIC32X4_MDACEN); + snd_soc_update_bits(codec, AIC32X4_MADC, + AIC32X4_MADCEN, AIC32X4_MADCEN); /* Switch on BCLK_N Divider */ - value = snd_soc_read(codec, AIC32X4_BCLKN); - snd_soc_write(codec, AIC32X4_BCLKN, - value | AIC32X4_BCLKEN); + snd_soc_update_bits(codec, AIC32X4_BCLKN, + AIC32X4_BCLKEN, AIC32X4_BCLKEN); } break; case SND_SOC_BIAS_PREPARE: @@ -569,34 +562,28 @@ static int aic32x4_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_STANDBY: if (aic32x4->master) { /* Switch off PLL */ - value = snd_soc_read(codec, AIC32X4_PLLPR); - snd_soc_write(codec, AIC32X4_PLLPR, - (value & ~AIC32X4_PLLEN)); + snd_soc_update_bits(codec, AIC32X4_PLLPR, + AIC32X4_PLLEN, 0); /* Switch off NDAC Divider */ - value = snd_soc_read(codec, AIC32X4_NDAC); - snd_soc_write(codec, AIC32X4_NDAC, - value & ~AIC32X4_NDACEN); + snd_soc_update_bits(codec, AIC32X4_NDAC, + AIC32X4_NDACEN, 0); /* Switch off MDAC Divider */ - value = snd_soc_read(codec, AIC32X4_MDAC); - snd_soc_write(codec, AIC32X4_MDAC, - value & ~AIC32X4_MDACEN); + snd_soc_update_bits(codec, AIC32X4_MDAC, + AIC32X4_MDACEN, 0); /* Switch off NADC Divider */ - value = snd_soc_read(codec, AIC32X4_NADC); - snd_soc_write(codec, AIC32X4_NADC, - value & ~AIC32X4_NDACEN); + snd_soc_update_bits(codec, AIC32X4_NADC, + AIC32X4_NADCEN, 0); /* Switch off MADC Divider */ - value = snd_soc_read(codec, AIC32X4_MADC); - snd_soc_write(codec, AIC32X4_MADC, - value & ~AIC32X4_MDACEN); - value = snd_soc_read(codec, AIC32X4_BCLKN); + snd_soc_update_bits(codec, AIC32X4_MADC, + AIC32X4_MADCEN, 0); /* Switch off BCLK_N Divider */ - snd_soc_write(codec, AIC32X4_BCLKN, - value & ~AIC32X4_BCLKEN); + snd_soc_update_bits(codec, AIC32X4_BCLKN, + AIC32X4_BCLKEN, 0); } break; case SND_SOC_BIAS_OFF: -- cgit v1.1 From a6f096f3b6effff9edc0f34a20eb0593406cc00a Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 14 Oct 2011 20:18:49 +0100 Subject: ASoC: Convert DA7210 to table based DAPM init Signed-off-by: Mark Brown --- sound/soc/codecs/da7210.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index ff68247..3b5dc0d 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -528,9 +528,6 @@ static int da7210_probe(struct snd_soc_codec *codec) /* Activate all enabled subsystem */ snd_soc_write(codec, DA7210_STARTUP1, DA7210_SC_MST_EN); - snd_soc_add_controls(codec, da7210_snd_controls, - ARRAY_SIZE(da7210_snd_controls)); - dev_info(codec->dev, "DA7210 Audio Codec %s\n", DA7210_VERSION); return 0; @@ -542,6 +539,9 @@ static struct snd_soc_codec_driver soc_codec_dev_da7210 = { .reg_word_size = sizeof(u8), .reg_cache_default = da7210_reg, .volatile_register = da7210_volatile_register, + + .controls = da7210_snd_controls, + .num_controls = ARRAY_SIZE(da7210_snd_controls), }; #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) -- cgit v1.1 From f9dfbf91cbf9a8875e955350c957f84e13557634 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Thu, 13 Oct 2011 15:06:43 +0800 Subject: ASoC: tlv320aic23: convert to soc-cache Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic23.c | 167 +++++++++++++---------------------------- 1 file changed, 52 insertions(+), 115 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index c3a4bb2..ab27dbc 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -47,63 +47,6 @@ static const u16 tlv320aic23_reg[] = { 0x0000, 0x0000, 0x0000, 0x0000, /* 12 */ }; -/* - * read tlv320aic23 register cache - */ -static inline unsigned int tlv320aic23_read_reg_cache(struct snd_soc_codec - *codec, unsigned int reg) -{ - u16 *cache = codec->reg_cache; - if (reg >= ARRAY_SIZE(tlv320aic23_reg)) - return -1; - return cache[reg]; -} - -/* - * write tlv320aic23 register cache - */ -static inline void tlv320aic23_write_reg_cache(struct snd_soc_codec *codec, - u8 reg, u16 value) -{ - u16 *cache = codec->reg_cache; - if (reg >= ARRAY_SIZE(tlv320aic23_reg)) - return; - cache[reg] = value; -} - -/* - * write to the tlv320aic23 register space - */ -static int tlv320aic23_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) -{ - - u8 data[2]; - - /* TLV320AIC23 has 7 bit address and 9 bits of data - * so we need to switch one data bit into reg and rest - * of data into val - */ - - if (reg > 9 && reg != 15) { - printk(KERN_WARNING "%s Invalid register R%u\n", __func__, reg); - return -1; - } - - data[0] = (reg << 1) | (value >> 8 & 0x01); - data[1] = value & 0xff; - - tlv320aic23_write_reg_cache(codec, reg, value); - - if (codec->hw_write(codec->control_data, data, 2) == 2) - return 0; - - printk(KERN_ERR "%s cannot write %03x to register R%u\n", __func__, - value, reg); - - return -EIO; -} - static const char *rec_src_text[] = { "Line", "Mic" }; static const char *deemph_text[] = {"None", "32Khz", "44.1Khz", "48Khz"}; @@ -139,8 +82,8 @@ static int snd_soc_tlv320aic23_put_volsw(struct snd_kcontrol *kcontrol, */ val = (val >= 4) ? 4 : (3 - val); - reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_ANLG) & (~0x1C0); - tlv320aic23_write(codec, TLV320AIC23_ANLG, reg | (val << 6)); + reg = snd_soc_read(codec, TLV320AIC23_ANLG) & (~0x1C0); + snd_soc_write(codec, TLV320AIC23_ANLG, reg | (val << 6)); return 0; } @@ -151,7 +94,7 @@ static int snd_soc_tlv320aic23_get_volsw(struct snd_kcontrol *kcontrol, struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); u16 val; - val = tlv320aic23_read_reg_cache(codec, TLV320AIC23_ANLG) & (0x1C0); + val = snd_soc_read(codec, TLV320AIC23_ANLG) & (0x1C0); val = val >> 6; val = (val >= 4) ? 4 : (3 - val); ucontrol->value.integer.value[0] = val; @@ -232,7 +175,6 @@ static const struct snd_soc_dapm_route tlv320aic23_intercon[] = { /* AIC23 driver data */ struct aic23 { enum snd_soc_control_type control_type; - void *control_data; int mclk; int requested_adc; int requested_dac; @@ -344,7 +286,7 @@ static int find_rate(int mclk, u32 need_adc, u32 need_dac) static void get_current_sample_rates(struct snd_soc_codec *codec, int mclk, u32 *sample_rate_adc, u32 *sample_rate_dac) { - int src = tlv320aic23_read_reg_cache(codec, TLV320AIC23_SRATE); + int src = snd_soc_read(codec, TLV320AIC23_SRATE); int sr = (src >> 2) & 0x0f; int val = (mclk / bosr_usb_divisor_table[src & 3]); int adc = (val * sr_adc_mult_table[sr]) / SR_MULT; @@ -368,7 +310,7 @@ static int set_sample_rate_control(struct snd_soc_codec *codec, int mclk, __func__, sample_rate_adc, sample_rate_dac); return -EINVAL; } - tlv320aic23_write(codec, TLV320AIC23_SRATE, data); + snd_soc_write(codec, TLV320AIC23_SRATE, data); #ifdef DEBUG { u32 adc, dac; @@ -407,9 +349,8 @@ static int tlv320aic23_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; - iface_reg = - tlv320aic23_read_reg_cache(codec, - TLV320AIC23_DIGT_FMT) & ~(0x03 << 2); + iface_reg = snd_soc_read(codec, TLV320AIC23_DIGT_FMT) & ~(0x03 << 2); + switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: break; @@ -423,7 +364,7 @@ static int tlv320aic23_hw_params(struct snd_pcm_substream *substream, iface_reg |= (0x03 << 2); break; } - tlv320aic23_write(codec, TLV320AIC23_DIGT_FMT, iface_reg); + snd_soc_write(codec, TLV320AIC23_DIGT_FMT, iface_reg); return 0; } @@ -435,7 +376,7 @@ static int tlv320aic23_pcm_prepare(struct snd_pcm_substream *substream, struct snd_soc_codec *codec = rtd->codec; /* set active */ - tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x0001); + snd_soc_write(codec, TLV320AIC23_ACTIVE, 0x0001); return 0; } @@ -450,7 +391,7 @@ static void tlv320aic23_shutdown(struct snd_pcm_substream *substream, /* deactivate */ if (!codec->active) { udelay(50); - tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x0); + snd_soc_write(codec, TLV320AIC23_ACTIVE, 0x0); } if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) aic23->requested_dac = 0; @@ -463,14 +404,14 @@ static int tlv320aic23_mute(struct snd_soc_dai *dai, int mute) struct snd_soc_codec *codec = dai->codec; u16 reg; - reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_DIGT); + reg = snd_soc_read(codec, TLV320AIC23_DIGT); if (mute) reg |= TLV320AIC23_DACM_MUTE; else reg &= ~TLV320AIC23_DACM_MUTE; - tlv320aic23_write(codec, TLV320AIC23_DIGT, reg); + snd_soc_write(codec, TLV320AIC23_DIGT, reg); return 0; } @@ -481,8 +422,7 @@ static int tlv320aic23_set_dai_fmt(struct snd_soc_dai *codec_dai, struct snd_soc_codec *codec = codec_dai->codec; u16 iface_reg; - iface_reg = - tlv320aic23_read_reg_cache(codec, TLV320AIC23_DIGT_FMT) & (~0x03); + iface_reg = snd_soc_read(codec, TLV320AIC23_DIGT_FMT) & (~0x03); /* set master/slave audio interface */ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { @@ -516,7 +456,7 @@ static int tlv320aic23_set_dai_fmt(struct snd_soc_dai *codec_dai, } - tlv320aic23_write(codec, TLV320AIC23_DIGT_FMT, iface_reg); + snd_soc_write(codec, TLV320AIC23_DIGT_FMT, iface_reg); return 0; } @@ -532,26 +472,26 @@ static int tlv320aic23_set_dai_sysclk(struct snd_soc_dai *codec_dai, static int tlv320aic23_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { - u16 reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_PWR) & 0xff7f; + u16 reg = snd_soc_read(codec, TLV320AIC23_PWR) & 0xff7f; switch (level) { case SND_SOC_BIAS_ON: /* vref/mid, osc on, dac unmute */ reg &= ~(TLV320AIC23_DEVICE_PWR_OFF | TLV320AIC23_OSC_OFF | \ TLV320AIC23_DAC_OFF); - tlv320aic23_write(codec, TLV320AIC23_PWR, reg); + snd_soc_write(codec, TLV320AIC23_PWR, reg); break; case SND_SOC_BIAS_PREPARE: break; case SND_SOC_BIAS_STANDBY: /* everything off except vref/vmid, */ - tlv320aic23_write(codec, TLV320AIC23_PWR, reg | \ - TLV320AIC23_CLK_OFF); + snd_soc_write(codec, TLV320AIC23_PWR, + reg | TLV320AIC23_CLK_OFF); break; case SND_SOC_BIAS_OFF: /* everything off, dac mute, inactive */ - tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x0); - tlv320aic23_write(codec, TLV320AIC23_PWR, 0xffff); + snd_soc_write(codec, TLV320AIC23_ACTIVE, 0x0); + snd_soc_write(codec, TLV320AIC23_PWR, 0xffff); break; } codec->dapm.bias_level = level; @@ -598,13 +538,7 @@ static int tlv320aic23_suspend(struct snd_soc_codec *codec, static int tlv320aic23_resume(struct snd_soc_codec *codec) { - u16 reg; - - /* Sync reg_cache with the hardware */ - for (reg = 0; reg <= TLV320AIC23_ACTIVE; reg++) { - u16 val = tlv320aic23_read_reg_cache(codec, reg); - tlv320aic23_write(codec, reg, val); - } + snd_soc_cache_sync(codec); tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_STANDBY); return 0; @@ -613,46 +547,52 @@ static int tlv320aic23_resume(struct snd_soc_codec *codec) static int tlv320aic23_probe(struct snd_soc_codec *codec) { struct aic23 *aic23 = snd_soc_codec_get_drvdata(codec); - int reg; + int ret; printk(KERN_INFO "AIC23 Audio Codec %s\n", AIC23_VERSION); - codec->control_data = aic23->control_data; - codec->hw_write = (hw_write_t)i2c_master_send; - codec->hw_read = NULL; + + ret = snd_soc_codec_set_cache_io(codec, 7, 9, aic23->control_type); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } /* Reset codec */ - tlv320aic23_write(codec, TLV320AIC23_RESET, 0); + snd_soc_write(codec, TLV320AIC23_RESET, 0); + + /* Write the register default value to cache for reserved registers, + * so the write to the these registers are suppressed by the cache + * restore code when it skips writes of default registers. + */ + snd_soc_cache_write(codec, 0x0A, 0); + snd_soc_cache_write(codec, 0x0B, 0); + snd_soc_cache_write(codec, 0x0C, 0); + snd_soc_cache_write(codec, 0x0D, 0); + snd_soc_cache_write(codec, 0x0E, 0); /* power on device */ tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - tlv320aic23_write(codec, TLV320AIC23_DIGT, TLV320AIC23_DEEMP_44K); + snd_soc_write(codec, TLV320AIC23_DIGT, TLV320AIC23_DEEMP_44K); /* Unmute input */ - reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_LINVOL); - tlv320aic23_write(codec, TLV320AIC23_LINVOL, - (reg & (~TLV320AIC23_LIM_MUTED)) | - (TLV320AIC23_LRS_ENABLED)); + snd_soc_update_bits(codec, TLV320AIC23_LINVOL, + TLV320AIC23_LIM_MUTED, TLV320AIC23_LRS_ENABLED); - reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_RINVOL); - tlv320aic23_write(codec, TLV320AIC23_RINVOL, - (reg & (~TLV320AIC23_LIM_MUTED)) | - TLV320AIC23_LRS_ENABLED); + snd_soc_update_bits(codec, TLV320AIC23_RINVOL, + TLV320AIC23_LIM_MUTED, TLV320AIC23_LRS_ENABLED); - reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_ANLG); - tlv320aic23_write(codec, TLV320AIC23_ANLG, - (reg) & (~TLV320AIC23_BYPASS_ON) & - (~TLV320AIC23_MICM_MUTED)); + snd_soc_update_bits(codec, TLV320AIC23_ANLG, + TLV320AIC23_BYPASS_ON | TLV320AIC23_MICM_MUTED, + 0); /* Default output volume */ - tlv320aic23_write(codec, TLV320AIC23_LCHNVOL, - TLV320AIC23_DEFAULT_OUT_VOL & - TLV320AIC23_OUT_VOL_MASK); - tlv320aic23_write(codec, TLV320AIC23_RCHNVOL, - TLV320AIC23_DEFAULT_OUT_VOL & - TLV320AIC23_OUT_VOL_MASK); + snd_soc_write(codec, TLV320AIC23_LCHNVOL, + TLV320AIC23_DEFAULT_OUT_VOL & TLV320AIC23_OUT_VOL_MASK); + snd_soc_write(codec, TLV320AIC23_RCHNVOL, + TLV320AIC23_DEFAULT_OUT_VOL & TLV320AIC23_OUT_VOL_MASK); - tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x1); + snd_soc_write(codec, TLV320AIC23_ACTIVE, 0x1); snd_soc_add_controls(codec, tlv320aic23_snd_controls, ARRAY_SIZE(tlv320aic23_snd_controls)); @@ -674,8 +614,6 @@ static struct snd_soc_codec_driver soc_codec_dev_tlv320aic23 = { .remove = tlv320aic23_remove, .suspend = tlv320aic23_suspend, .resume = tlv320aic23_resume, - .read = tlv320aic23_read_reg_cache, - .write = tlv320aic23_write, .set_bias_level = tlv320aic23_set_bias_level, .dapm_widgets = tlv320aic23_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(tlv320aic23_dapm_widgets), @@ -702,7 +640,6 @@ static int tlv320aic23_codec_probe(struct i2c_client *i2c, return -ENOMEM; i2c_set_clientdata(i2c, aic23); - aic23->control_data = i2c; aic23->control_type = SND_SOC_I2C; ret = snd_soc_register_codec(&i2c->dev, -- cgit v1.1 From 524205ce7182986c1961cbecd32a87953d4e18c3 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Fri, 14 Oct 2011 09:35:20 +0800 Subject: ASoC: alc5623: Convert codec->hw_read to snd_soc_read codec->hw_read is broken now, let's covert to snd_soc_read. Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/alc5623.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/alc5623.c b/sound/soc/codecs/alc5623.c index 557b3af..984b14b 100644 --- a/sound/soc/codecs/alc5623.c +++ b/sound/soc/codecs/alc5623.c @@ -53,8 +53,10 @@ static void alc5623_fill_cache(struct snd_soc_codec *codec) u16 *cache = codec->reg_cache; /* not really efficient ... */ + codec->cache_bypass = 1; for (i = 0 ; i < codec->driver->reg_cache_size ; i += step) - cache[i] = codec->hw_read(codec, i); + cache[i] = snd_soc_read(codec, i); + codec->cache_bypass = 0; } static inline int alc5623_reset(struct snd_soc_codec *codec) -- cgit v1.1 From 38c436aa9f7dc23ebe9e8f7ae88c586acc033d30 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Fri, 14 Oct 2011 09:37:00 +0800 Subject: ASoC: tlv320aic3x: Convert codec->hw_read to snd_soc_read codec->hw_read is broken now, let's covert to snd_soc_read. Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic3x.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index be55b7f..7a49390 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -137,7 +137,10 @@ static int aic3x_read(struct snd_soc_codec *codec, unsigned int reg, if (reg >= AIC3X_CACHEREGNUM) return -1; - *value = codec->hw_read(codec, reg); + codec->cache_bypass = 1; + *value = snd_soc_read(codec, reg); + codec->cache_bypass = 0; + cache[reg] = *value; return 0; -- cgit v1.1 From 370f464533c455864f2f5ce100eee361263e144f Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Fri, 14 Oct 2011 09:39:14 +0800 Subject: ASoC: wm8961: Convert codec->hw_read to snd_soc_read codec->hw_read is broken now, let's covert to snd_soc_read. Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/wm8961.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8961.c b/sound/soc/codecs/wm8961.c index cdee810..9568c8a 100644 --- a/sound/soc/codecs/wm8961.c +++ b/sound/soc/codecs/wm8961.c @@ -974,7 +974,9 @@ static int wm8961_probe(struct snd_soc_codec *codec) } /* This isn't volatile - readback doesn't correspond to write */ - reg = codec->hw_read(codec, WM8961_RIGHT_INPUT_VOLUME); + codec->cache_bypass = 1; + reg = snd_soc_read(codec, WM8961_RIGHT_INPUT_VOLUME); + codec->cache_bypass = 0; dev_info(codec->dev, "WM8961 family %d revision %c\n", (reg & WM8961_DEVICE_ID_MASK) >> WM8961_DEVICE_ID_SHIFT, ((reg & WM8961_CHIP_REV_MASK) >> WM8961_CHIP_REV_SHIFT) -- cgit v1.1 From 3f387a217044b3aa7785062aaa9113aa3cc729c0 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Fri, 14 Oct 2011 12:08:00 +0800 Subject: ASoC: wm8991: Fix wrong bit setting for WM8991_POWER_MANAGEMENT_2 If (fakepower & ((1 << WM8991_INMIXR_PWR_BIT)|(1 << WM8991_AINRMUX_PWR_BIT)))) is false, we should clear WM8991_AINR_ENA bits instead of WM8991_AINL_ENA. Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/wm8991.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8991.c b/sound/soc/codecs/wm8991.c index 08d64a6..708d251 100644 --- a/sound/soc/codecs/wm8991.c +++ b/sound/soc/codecs/wm8991.c @@ -393,7 +393,7 @@ static int inmixer_event(struct snd_soc_dapm_widget *w, (1 << WM8991_AINRMUX_PWR_BIT))) reg |= WM8991_AINR_ENA; else - reg &= ~WM8991_AINL_ENA; + reg &= ~WM8991_AINR_ENA; snd_soc_write(w->codec, WM8991_POWER_MANAGEMENT_2, reg); return 0; -- cgit v1.1 From c639adc68fc2b5b4899c7902d67fc095f42342e0 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Fri, 14 Oct 2011 12:09:48 +0800 Subject: ASoC: wm8991: Use snd_soc_update_bits for read-modify-write Use snd_soc_update_bits for read-modify-write register access instead of open-coding it using snd_soc_read and snd_soc_write Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/wm8991.c | 20 +++++++++----------- 1 file changed, 9 insertions(+), 11 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8991.c b/sound/soc/codecs/wm8991.c index 708d251..c9ab3ba 100644 --- a/sound/soc/codecs/wm8991.c +++ b/sound/soc/codecs/wm8991.c @@ -1264,7 +1264,6 @@ static int wm8991_probe(struct snd_soc_codec *codec) { struct wm8991_priv *wm8991; int ret; - unsigned int reg; wm8991 = snd_soc_codec_get_drvdata(codec); @@ -1282,19 +1281,18 @@ static int wm8991_probe(struct snd_soc_codec *codec) wm8991_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - reg = snd_soc_read(codec, WM8991_AUDIO_INTERFACE_4); - snd_soc_write(codec, WM8991_AUDIO_INTERFACE_4, reg | WM8991_ALRCGPIO1); + snd_soc_update_bits(codec, WM8991_AUDIO_INTERFACE_4, + WM8991_ALRCGPIO1, WM8991_ALRCGPIO1); - reg = snd_soc_read(codec, WM8991_GPIO1_GPIO2) & - ~WM8991_GPIO1_SEL_MASK; - snd_soc_write(codec, WM8991_GPIO1_GPIO2, reg | 1); + snd_soc_update_bits(codec, WM8991_GPIO1_GPIO2, + WM8991_GPIO1_SEL_MASK, 1); - reg = snd_soc_read(codec, WM8991_POWER_MANAGEMENT_1); - snd_soc_write(codec, WM8991_POWER_MANAGEMENT_1, reg | WM8991_VREF_ENA| - WM8991_VMID_MODE_MASK); + snd_soc_update_bits(codec, WM8991_POWER_MANAGEMENT_1, + WM8991_VREF_ENA | WM8991_VMID_MODE_MASK, + WM8991_VREF_ENA | WM8991_VMID_MODE_MASK); - reg = snd_soc_read(codec, WM8991_POWER_MANAGEMENT_2); - snd_soc_write(codec, WM8991_POWER_MANAGEMENT_2, reg | WM8991_OPCLK_ENA); + snd_soc_update_bits(codec, WM8991_POWER_MANAGEMENT_2, + WM8991_OPCLK_ENA, WM8991_OPCLK_ENA); snd_soc_write(codec, WM8991_DAC_CTRL, 0); snd_soc_write(codec, WM8991_LEFT_OUTPUT_VOLUME, 0x50 | (1<<8)); -- cgit v1.1 From 1a8e8d2234cfc89ee055205bd247b2184c6e5f2d Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Fri, 14 Oct 2011 13:56:49 +0800 Subject: ASoC: wm8400: Fix wrong bit setting for WM8400_POWER_MANAGEMENT_2 If (fakepower & ((1 << WM8400_INMIXR_PWR) | (1 << WM8400_AINRMUX_PWR))) is false, we should clear WM8400_AINR_ENA bits instead of WM8400_AINL_ENA. Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/wm8400.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c index fbee556c..dc13be2 100644 --- a/sound/soc/codecs/wm8400.c +++ b/sound/soc/codecs/wm8400.c @@ -383,7 +383,7 @@ static int inmixer_event (struct snd_soc_dapm_widget *w, (1 << WM8400_AINRMUX_PWR))) { reg |= WM8400_AINR_ENA; } else { - reg &= ~WM8400_AINL_ENA; + reg &= ~WM8400_AINR_ENA; } wm8400_write(w->codec, WM8400_POWER_MANAGEMENT_2, reg); -- cgit v1.1 From 790f932500061ce49c52ef9dbd48fbfbdeb631c5 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Fri, 14 Oct 2011 13:57:48 +0800 Subject: ASoC: wm8990: Fix wrong bit setting for WM8990_POWER_MANAGEMENT_2 If (fakepower & ((1 << WM8990_INMIXR_PWR_BIT) | (1 << WM8990_AINRMUX_PWR_BIT))) is false, we should clear WM8990_AINR_ENA bits instead of WM8990_AINL_ENA. Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/wm8990.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index ecdb8b2..b9c5ecc 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -401,7 +401,7 @@ static int inmixer_event(struct snd_soc_dapm_widget *w, (1 << WM8990_AINRMUX_PWR_BIT))) { reg |= WM8990_AINR_ENA; } else { - reg &= ~WM8990_AINL_ENA; + reg &= ~WM8990_AINR_ENA; } snd_soc_write(w->codec, WM8990_POWER_MANAGEMENT_2, reg); -- cgit v1.1 From 79d07265137c166cf298d74a29038a76458ec46a Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Fri, 14 Oct 2011 14:30:05 +0800 Subject: ASoC: wm8990: Use snd_soc_update_bits for read-modify-write Use snd_soc_update_bits for read-modify-write register access instead of open-coding it using snd_soc_read and snd_soc_write This patch also includes a comment fix in wm8990_set_dai_pll(), if freq_in and freq_out are 0, what we do is to clear WM8990_PLL_ENA bit. Thus the comment should be "Turn off PLL". Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/wm8990.c | 67 +++++++++++++++++++---------------------------- 1 file changed, 27 insertions(+), 40 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index b9c5ecc..d29a962 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -981,7 +981,6 @@ static void pll_factors(struct _pll_div *pll_div, unsigned int target, static int wm8990_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, int source, unsigned int freq_in, unsigned int freq_out) { - u16 reg; struct snd_soc_codec *codec = codec_dai->codec; struct _pll_div pll_div; @@ -989,13 +988,12 @@ static int wm8990_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, pll_factors(&pll_div, freq_out * 4, freq_in); /* Turn on PLL */ - reg = snd_soc_read(codec, WM8990_POWER_MANAGEMENT_2); - reg |= WM8990_PLL_ENA; - snd_soc_write(codec, WM8990_POWER_MANAGEMENT_2, reg); + snd_soc_update_bits(codec, WM8990_POWER_MANAGEMENT_2, + WM8990_PLL_ENA, WM8990_PLL_ENA); /* sysclk comes from PLL */ - reg = snd_soc_read(codec, WM8990_CLOCKING_2); - snd_soc_write(codec, WM8990_CLOCKING_2, reg | WM8990_SYSCLK_SRC); + snd_soc_update_bits(codec, WM8990_CLOCKING_2, + WM8990_SYSCLK_SRC, WM8990_SYSCLK_SRC); /* set up N , fractional mode and pre-divisor if necessary */ snd_soc_write(codec, WM8990_PLL1, pll_div.n | WM8990_SDM | @@ -1003,10 +1001,9 @@ static int wm8990_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, snd_soc_write(codec, WM8990_PLL2, (u8)(pll_div.k>>8)); snd_soc_write(codec, WM8990_PLL3, (u8)(pll_div.k & 0xFF)); } else { - /* Turn on PLL */ - reg = snd_soc_read(codec, WM8990_POWER_MANAGEMENT_2); - reg &= ~WM8990_PLL_ENA; - snd_soc_write(codec, WM8990_POWER_MANAGEMENT_2, reg); + /* Turn off PLL */ + snd_soc_update_bits(codec, WM8990_POWER_MANAGEMENT_2, + WM8990_PLL_ENA, 0); } return 0; } @@ -1084,28 +1081,23 @@ static int wm8990_set_dai_clkdiv(struct snd_soc_dai *codec_dai, int div_id, int div) { struct snd_soc_codec *codec = codec_dai->codec; - u16 reg; switch (div_id) { case WM8990_MCLK_DIV: - reg = snd_soc_read(codec, WM8990_CLOCKING_2) & - ~WM8990_MCLK_DIV_MASK; - snd_soc_write(codec, WM8990_CLOCKING_2, reg | div); + snd_soc_update_bits(codec, WM8990_CLOCKING_2, + WM8990_MCLK_DIV_MASK, div); break; case WM8990_DACCLK_DIV: - reg = snd_soc_read(codec, WM8990_CLOCKING_2) & - ~WM8990_DAC_CLKDIV_MASK; - snd_soc_write(codec, WM8990_CLOCKING_2, reg | div); + snd_soc_update_bits(codec, WM8990_CLOCKING_2, + WM8990_DAC_CLKDIV_MASK, div); break; case WM8990_ADCCLK_DIV: - reg = snd_soc_read(codec, WM8990_CLOCKING_2) & - ~WM8990_ADC_CLKDIV_MASK; - snd_soc_write(codec, WM8990_CLOCKING_2, reg | div); + snd_soc_update_bits(codec, WM8990_CLOCKING_2, + WM8990_ADC_CLKDIV_MASK, div); break; case WM8990_BCLK_DIV: - reg = snd_soc_read(codec, WM8990_CLOCKING_1) & - ~WM8990_BCLK_DIV_MASK; - snd_soc_write(codec, WM8990_CLOCKING_1, reg | div); + snd_soc_update_bits(codec, WM8990_CLOCKING_1, + WM8990_BCLK_DIV_MASK, div); break; default: return -EINVAL; @@ -1164,7 +1156,6 @@ static int wm8990_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { int ret; - u16 val; switch (level) { case SND_SOC_BIAS_ON: @@ -1172,9 +1163,8 @@ static int wm8990_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_PREPARE: /* VMID=2*50k */ - val = snd_soc_read(codec, WM8990_POWER_MANAGEMENT_1) & - ~WM8990_VMID_MODE_MASK; - snd_soc_write(codec, WM8990_POWER_MANAGEMENT_1, val | 0x2); + snd_soc_update_bits(codec, WM8990_POWER_MANAGEMENT_1, + WM8990_VMID_MODE_MASK, 0x2); break; case SND_SOC_BIAS_STANDBY: @@ -1239,9 +1229,8 @@ static int wm8990_set_bias_level(struct snd_soc_codec *codec, } /* VMID=2*250k */ - val = snd_soc_read(codec, WM8990_POWER_MANAGEMENT_1) & - ~WM8990_VMID_MODE_MASK; - snd_soc_write(codec, WM8990_POWER_MANAGEMENT_1, val | 0x4); + snd_soc_update_bits(codec, WM8990_POWER_MANAGEMENT_1, + WM8990_VMID_MODE_MASK, 0x4); break; case SND_SOC_BIAS_OFF: @@ -1255,8 +1244,8 @@ static int wm8990_set_bias_level(struct snd_soc_codec *codec, WM8990_BUFIOEN); /* mute DAC */ - val = snd_soc_read(codec, WM8990_DAC_CTRL); - snd_soc_write(codec, WM8990_DAC_CTRL, val | WM8990_DAC_MUTE); + snd_soc_update_bits(codec, WM8990_DAC_CTRL, + WM8990_DAC_MUTE, WM8990_DAC_MUTE); /* Enable any disabled outputs */ snd_soc_write(codec, WM8990_POWER_MANAGEMENT_1, 0x1f03); @@ -1344,7 +1333,6 @@ static int wm8990_resume(struct snd_soc_codec *codec) static int wm8990_probe(struct snd_soc_codec *codec) { int ret; - u16 reg; ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_I2C); if (ret < 0) { @@ -1357,15 +1345,14 @@ static int wm8990_probe(struct snd_soc_codec *codec) /* charge output caps */ wm8990_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - reg = snd_soc_read(codec, WM8990_AUDIO_INTERFACE_4); - snd_soc_write(codec, WM8990_AUDIO_INTERFACE_4, reg | WM8990_ALRCGPIO1); + snd_soc_update_bits(codec, WM8990_AUDIO_INTERFACE_4, + WM8990_ALRCGPIO1, WM8990_ALRCGPIO1); - reg = snd_soc_read(codec, WM8990_GPIO1_GPIO2) & - ~WM8990_GPIO1_SEL_MASK; - snd_soc_write(codec, WM8990_GPIO1_GPIO2, reg | 1); + snd_soc_update_bits(codec, WM8990_GPIO1_GPIO2, + WM8990_GPIO1_SEL_MASK, 1); - reg = snd_soc_read(codec, WM8990_POWER_MANAGEMENT_2); - snd_soc_write(codec, WM8990_POWER_MANAGEMENT_2, reg | WM8990_OPCLK_ENA); + snd_soc_update_bits(codec, WM8990_POWER_MANAGEMENT_2, + WM8990_OPCLK_ENA, WM8990_OPCLK_ENA); snd_soc_write(codec, WM8990_LEFT_OUTPUT_VOLUME, 0x50 | (1<<8)); snd_soc_write(codec, WM8990_RIGHT_OUTPUT_VOLUME, 0x50 | (1<<8)); -- cgit v1.1 From f3aa7219b15c140fece2ec6b9240fccc3b7a5afd Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Fri, 14 Oct 2011 17:01:59 +0800 Subject: ASoC: ad193x: Fix define of AD193X_PLL_INPUT_MASK Current code defines AD193X_PLL_INPUT_MASK as (~0x6) which is quite different from other MASK defines. To make it consistent with other mask defines, define AD193X_PLL_INPUT_MASK as 0x6 and change the code accordingly. I think this change improves the readability. Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/ad193x.c | 2 +- sound/soc/codecs/ad193x.h | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c index f934670..39056ce 100644 --- a/sound/soc/codecs/ad193x.c +++ b/sound/soc/codecs/ad193x.c @@ -298,7 +298,7 @@ static int ad193x_hw_params(struct snd_pcm_substream *substream, } reg = snd_soc_read(codec, AD193X_PLL_CLK_CTRL0); - reg = (reg & AD193X_PLL_INPUT_MASK) | master_rate; + reg = (reg & (~AD193X_PLL_INPUT_MASK)) | master_rate; snd_soc_write(codec, AD193X_PLL_CLK_CTRL0, reg); reg = snd_soc_read(codec, AD193X_DAC_CTRL2); diff --git a/sound/soc/codecs/ad193x.h b/sound/soc/codecs/ad193x.h index 536e5f2..1507eaa 100644 --- a/sound/soc/codecs/ad193x.h +++ b/sound/soc/codecs/ad193x.h @@ -11,7 +11,7 @@ #define AD193X_PLL_CLK_CTRL0 0x00 #define AD193X_PLL_POWERDOWN 0x01 -#define AD193X_PLL_INPUT_MASK (~0x6) +#define AD193X_PLL_INPUT_MASK 0x6 #define AD193X_PLL_INPUT_256 (0 << 1) #define AD193X_PLL_INPUT_384 (1 << 1) #define AD193X_PLL_INPUT_512 (2 << 1) -- cgit v1.1 From 7a0e67b68701d73b2252bd73f7fd49c54aea1e58 Mon Sep 17 00:00:00 2001 From: Ashish Chavan Date: Fri, 14 Oct 2011 16:25:25 +0530 Subject: ASoC: da7210: bugfix for head phone volume control This patch takes care of reserved bits of headphone volume register by using correct volume range. Signed-off-by: Ashish Chavan Signed-off-by: David Dajun Chen Signed-off-by: Mark Brown --- sound/soc/codecs/da7210.c | 10 ++++++---- 1 file changed, 6 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index 3b5dc0d..902fa58 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -149,12 +149,14 @@ * mute : 0x10 * reserved : 0x00 - 0x0F * - * ** FIXME ** - * * Reserved area are considered as "mute". - * -> min = -79.5 dB */ -static const DECLARE_TLV_DB_SCALE(hp_out_tlv, -7950, 150, 1); +static const unsigned int hp_out_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0x0, 0x10, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 1), + /* -54 dB to +15 dB */ + 0x11, 0x3f, TLV_DB_SCALE_ITEM(-5400, 150, 0), +}; static const struct snd_kcontrol_new da7210_snd_controls[] = { -- cgit v1.1 From 1d69c5c5de32c355667c105a5fac85c8043128e6 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 14 Oct 2011 14:43:33 +0300 Subject: ASoC: core: Add flag to ignore pmdown_time at pcm_close With this flag codec drivers can indicate that it is desired to ignore the pmdown_time for DAPM shutdown sequence when playback stream is stopped. The DAPM sequence will be executed without delay in this case. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/soc-pcm.c | 15 +++++++++++---- 1 file changed, 11 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 8eb0f07..ee15337 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -319,10 +319,17 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) cpu_dai->runtime = NULL; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - /* start delayed pop wq here for playback streams */ - codec_dai->pop_wait = 1; - schedule_delayed_work(&rtd->delayed_work, - msecs_to_jiffies(rtd->pmdown_time)); + if (unlikely(codec->ignore_pmdown_time)) { + /* powered down playback stream now */ + snd_soc_dapm_stream_event(rtd, + codec_dai->driver->playback.stream_name, + SND_SOC_DAPM_STREAM_STOP); + } else { + /* start delayed pop wq here for playback streams */ + codec_dai->pop_wait = 1; + schedule_delayed_work(&rtd->delayed_work, + msecs_to_jiffies(rtd->pmdown_time)); + } } else { /* capture streams can be powered down now */ snd_soc_dapm_stream_event(rtd, -- cgit v1.1 From 35dec697579459983e3471b622f57c18f6e9fd0a Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 14 Oct 2011 14:43:34 +0300 Subject: ASoC: twl6040: Request core to inline the DAPM sequence We need to have as less time between McPDM shutdown, and power down of the DAC on the twl6040 codec as possible. Request core to ignore the pmdown_time for the playback stream. Backround: with the McPDM protocol we are sendning not only the pure audio stream, but OMAP McPDM also transmits additional information (for example offset cancellation). If McPDM is stopped prior to the DAC this information will be not sent to the codec, which can result noise rendered by the twl6040 codec. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl6040.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 6c573c3..73e11f0 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -1504,6 +1504,7 @@ static int twl6040_probe(struct snd_soc_codec *codec) priv->codec = codec; codec->control_data = dev_get_drvdata(codec->dev->parent); + codec->ignore_pmdown_time = 1; if (pdata && pdata->hs_left_step && pdata->hs_right_step) { priv->hs_left_step = pdata->hs_left_step; -- cgit v1.1 From 01840bbe5f4406bf1d24590b96b0e3df43aaa81a Mon Sep 17 00:00:00 2001 From: Olof Johansson Date: Fri, 14 Oct 2011 15:54:19 -0700 Subject: ASoC: Tegra: sparse cleanup Fixes the following sparse warnings: sound/soc/tegra/tegra_das.c:215:8: warning: Using plain integer as NULL pointer sound/soc/tegra/tegra_das.c:237:8: warning: Using plain integer as NULL pointer sound/soc/tegra/tegra_pcm.c:370:32: warning: symbol 'tegra_pcm_platform' was not declared. Should it be static? Signed-off-by: Olof Johansson Acked-by: Stephen Warren Signed-off-by: Mark Brown --- sound/soc/tegra/tegra_das.c | 4 ++-- sound/soc/tegra/tegra_pcm.c | 2 +- 2 files changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/tegra/tegra_das.c b/sound/soc/tegra/tegra_das.c index 9f24ef7..3b55a44 100644 --- a/sound/soc/tegra/tegra_das.c +++ b/sound/soc/tegra/tegra_das.c @@ -212,7 +212,7 @@ err_release: release_mem_region(res->start, resource_size(res)); err_free: kfree(das); - das = 0; + das = NULL; exit: return ret; } @@ -234,7 +234,7 @@ static int __devexit tegra_das_remove(struct platform_device *pdev) release_mem_region(res->start, resource_size(res)); kfree(das); - das = 0; + das = NULL; return 0; } diff --git a/sound/soc/tegra/tegra_pcm.c b/sound/soc/tegra/tegra_pcm.c index c7cfd96..436def1 100644 --- a/sound/soc/tegra/tegra_pcm.c +++ b/sound/soc/tegra/tegra_pcm.c @@ -367,7 +367,7 @@ static void tegra_pcm_free(struct snd_pcm *pcm) tegra_pcm_deallocate_dma_buffer(pcm, SNDRV_PCM_STREAM_PLAYBACK); } -struct snd_soc_platform_driver tegra_pcm_platform = { +static struct snd_soc_platform_driver tegra_pcm_platform = { .ops = &tegra_pcm_ops, .pcm_new = tegra_pcm_new, .pcm_free = tegra_pcm_free, -- cgit v1.1 From 0ee6e9e721fc85e093e20e7a9ca848cfa71f80a9 Mon Sep 17 00:00:00 2001 From: Ashish Chavan Date: Sat, 15 Oct 2011 14:47:56 +0530 Subject: ASoC: da7210: Add support for ADC & DAC equalizers This patch adds support for ADC and DAC five band equalizers available on DA7210 codec. Signed-off-by: Ashish Chavan Signed-off-by: David Dajun Chen Signed-off-by: Mark Brown --- sound/soc/codecs/da7210.c | 37 +++++++++++++++++++++++++++++++++++++ 1 file changed, 37 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index 902fa58..7dc1259 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -34,10 +34,16 @@ #define DA7210_INMIX_R 0x0E #define DA7210_ADC_HPF 0x0F #define DA7210_ADC 0x10 +#define DA7210_ADC_EQ1_2 0X11 +#define DA7210_ADC_EQ3_4 0x12 +#define DA7210_ADC_EQ5 0x13 #define DA7210_DAC_HPF 0x14 #define DA7210_DAC_L 0x15 #define DA7210_DAC_R 0x16 #define DA7210_DAC_SEL 0x17 +#define DA7210_DAC_EQ1_2 0x19 +#define DA7210_DAC_EQ3_4 0x1A +#define DA7210_DAC_EQ5 0x1B #define DA7210_OUTMIX_L 0x1C #define DA7210_OUTMIX_R 0x1D #define DA7210_HP_L_VOL 0x21 @@ -158,11 +164,42 @@ static const unsigned int hp_out_tlv[] = { 0x11, 0x3f, TLV_DB_SCALE_ITEM(-5400, 150, 0), }; +static const DECLARE_TLV_DB_SCALE(eq_gain_tlv, -1050, 150, 0); +static const DECLARE_TLV_DB_SCALE(adc_eq_master_gain_tlv, -1800, 600, 1); + static const struct snd_kcontrol_new da7210_snd_controls[] = { SOC_DOUBLE_R_TLV("HeadPhone Playback Volume", DA7210_HP_L_VOL, DA7210_HP_R_VOL, 0, 0x3F, 0, hp_out_tlv), + + /* DAC Equalizer controls */ + SOC_SINGLE("DAC EQ Switch", DA7210_DAC_EQ5, 7, 1, 0), + SOC_SINGLE_TLV("DAC EQ1 Volume", DA7210_DAC_EQ1_2, 0, 0xf, 1, + eq_gain_tlv), + SOC_SINGLE_TLV("DAC EQ2 Volume", DA7210_DAC_EQ1_2, 4, 0xf, 1, + eq_gain_tlv), + SOC_SINGLE_TLV("DAC EQ3 Volume", DA7210_DAC_EQ3_4, 0, 0xf, 1, + eq_gain_tlv), + SOC_SINGLE_TLV("DAC EQ4 Volume", DA7210_DAC_EQ3_4, 4, 0xf, 1, + eq_gain_tlv), + SOC_SINGLE_TLV("DAC EQ5 Volume", DA7210_DAC_EQ5, 0, 0xf, 1, + eq_gain_tlv), + + /* ADC Equalizer controls */ + SOC_SINGLE("ADC EQ Switch", DA7210_ADC_EQ5, 7, 1, 0), + SOC_SINGLE_TLV("ADC EQ Master Volume", DA7210_ADC_EQ5, 4, 0x3, + 1, adc_eq_master_gain_tlv), + SOC_SINGLE_TLV("ADC EQ1 Volume", DA7210_ADC_EQ1_2, 0, 0xf, 1, + eq_gain_tlv), + SOC_SINGLE_TLV("ADC EQ2 Volume", DA7210_ADC_EQ1_2, 4, 0xf, 1, + eq_gain_tlv), + SOC_SINGLE_TLV("ADC EQ3 Volume", DA7210_ADC_EQ3_4, 0, 0xf, 1, + eq_gain_tlv), + SOC_SINGLE_TLV("ADC EQ4 Volume", DA7210_ADC_EQ3_4, 4, 0xf, 1, + eq_gain_tlv), + SOC_SINGLE_TLV("ADC EQ5 Volume", DA7210_ADC_EQ5, 0, 0xf, 1, + eq_gain_tlv), }; /* Codec private data */ -- cgit v1.1 From 4ced2b96f3d8b5944611e4e93b59b69ad440e10e Mon Sep 17 00:00:00 2001 From: Ashish Chavan Date: Sat, 15 Oct 2011 14:50:06 +0530 Subject: ASoC: da7210: Add support for High pass and Voice filters for ADC and DAC This patch add controls for setting cut-off for high pass and voice filters of ADC and DAC. There are also switches to enable/disable these filters. Also removed hard coded, fixed values of these parameters used by previous version of driver. Signed-off-by: Ashish Chavan Signed-off-by: David Dajun Chen Signed-off-by: Mark Brown --- sound/soc/codecs/da7210.c | 57 ++++++++++++++++++++++++++--------------------- 1 file changed, 32 insertions(+), 25 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index 7dc1259..fa0d512 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -167,6 +167,28 @@ static const unsigned int hp_out_tlv[] = { static const DECLARE_TLV_DB_SCALE(eq_gain_tlv, -1050, 150, 0); static const DECLARE_TLV_DB_SCALE(adc_eq_master_gain_tlv, -1800, 600, 1); +/* ADC and DAC high pass filter f0 value */ +static const char const *da7210_hpf_cutoff_txt[] = { + "Fs/8192*pi", "Fs/4096*pi", "Fs/2048*pi", "Fs/1024*pi" +}; + +static const struct soc_enum da7210_dac_hpf_cutoff = + SOC_ENUM_SINGLE(DA7210_DAC_HPF, 0, 4, da7210_hpf_cutoff_txt); + +static const struct soc_enum da7210_adc_hpf_cutoff = + SOC_ENUM_SINGLE(DA7210_ADC_HPF, 0, 4, da7210_hpf_cutoff_txt); + +/* ADC and DAC voice (8kHz) high pass cutoff value */ +static const char const *da7210_vf_cutoff_txt[] = { + "2.5Hz", "25Hz", "50Hz", "100Hz", "150Hz", "200Hz", "300Hz", "400Hz" +}; + +static const struct soc_enum da7210_dac_vf_cutoff = + SOC_ENUM_SINGLE(DA7210_DAC_HPF, 4, 8, da7210_vf_cutoff_txt); + +static const struct soc_enum da7210_adc_vf_cutoff = + SOC_ENUM_SINGLE(DA7210_ADC_HPF, 4, 8, da7210_vf_cutoff_txt); + static const struct snd_kcontrol_new da7210_snd_controls[] = { SOC_DOUBLE_R_TLV("HeadPhone Playback Volume", @@ -200,6 +222,16 @@ static const struct snd_kcontrol_new da7210_snd_controls[] = { eq_gain_tlv), SOC_SINGLE_TLV("ADC EQ5 Volume", DA7210_ADC_EQ5, 0, 0xf, 1, eq_gain_tlv), + + SOC_SINGLE("DAC HPF Switch", DA7210_DAC_HPF, 3, 1, 0), + SOC_ENUM("DAC HPF Cutoff", da7210_dac_hpf_cutoff), + SOC_SINGLE("DAC Voice Mode Switch", DA7210_DAC_HPF, 7, 1, 0), + SOC_ENUM("DAC Voice Cutoff", da7210_dac_vf_cutoff), + + SOC_SINGLE("ADC HPF Switch", DA7210_ADC_HPF, 3, 1, 0), + SOC_ENUM("ADC HPF Cutoff", da7210_adc_hpf_cutoff), + SOC_SINGLE("ADC Voice Mode Switch", DA7210_ADC_HPF, 7, 1, 0), + SOC_ENUM("ADC Voice Cutoff", da7210_adc_vf_cutoff), }; /* Codec private data */ @@ -275,7 +307,6 @@ static int da7210_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_codec *codec = rtd->codec; u32 dai_cfg1; - u32 hpf_reg, hpf_mask, hpf_value; u32 fs, bypass; /* set DAI source to Left and Right ADC */ @@ -306,68 +337,45 @@ static int da7210_hw_params(struct snd_pcm_substream *substream, snd_soc_write(codec, DA7210_DAI_CFG1, dai_cfg1); - hpf_reg = (SNDRV_PCM_STREAM_PLAYBACK == substream->stream) ? - DA7210_DAC_HPF : DA7210_ADC_HPF; - switch (params_rate(params)) { case 8000: fs = DA7210_PLL_FS_8000; - hpf_mask = DA7210_VOICE_F0_MASK | DA7210_VOICE_EN; - hpf_value = DA7210_VOICE_F0_25 | DA7210_VOICE_EN; bypass = DA7210_PLL_BYP; break; case 11025: fs = DA7210_PLL_FS_11025; - hpf_mask = DA7210_VOICE_F0_MASK | DA7210_VOICE_EN; - hpf_value = DA7210_VOICE_F0_25 | DA7210_VOICE_EN; bypass = 0; break; case 12000: fs = DA7210_PLL_FS_12000; - hpf_mask = DA7210_VOICE_F0_MASK | DA7210_VOICE_EN; - hpf_value = DA7210_VOICE_F0_25 | DA7210_VOICE_EN; bypass = DA7210_PLL_BYP; break; case 16000: fs = DA7210_PLL_FS_16000; - hpf_mask = DA7210_VOICE_F0_MASK | DA7210_VOICE_EN; - hpf_value = DA7210_VOICE_F0_25 | DA7210_VOICE_EN; bypass = DA7210_PLL_BYP; break; case 22050: fs = DA7210_PLL_FS_22050; - hpf_mask = DA7210_VOICE_EN; - hpf_value = 0; bypass = 0; break; case 32000: fs = DA7210_PLL_FS_32000; - hpf_mask = DA7210_VOICE_EN; - hpf_value = 0; bypass = DA7210_PLL_BYP; break; case 44100: fs = DA7210_PLL_FS_44100; - hpf_mask = DA7210_VOICE_EN; - hpf_value = 0; bypass = 0; break; case 48000: fs = DA7210_PLL_FS_48000; - hpf_mask = DA7210_VOICE_EN; - hpf_value = 0; bypass = DA7210_PLL_BYP; break; case 88200: fs = DA7210_PLL_FS_88200; - hpf_mask = DA7210_VOICE_EN; - hpf_value = 0; bypass = 0; break; case 96000: fs = DA7210_PLL_FS_96000; - hpf_mask = DA7210_VOICE_EN; - hpf_value = 0; bypass = DA7210_PLL_BYP; break; default: @@ -377,7 +385,6 @@ static int da7210_hw_params(struct snd_pcm_substream *substream, /* Disable active mode */ snd_soc_update_bits(codec, DA7210_STARTUP1, DA7210_SC_MST_EN, 0); - snd_soc_update_bits(codec, hpf_reg, hpf_mask, hpf_value); snd_soc_update_bits(codec, DA7210_PLL, DA7210_PLL_FS_MASK, fs); snd_soc_update_bits(codec, DA7210_PLL_DIV3, DA7210_PLL_BYP, bypass); -- cgit v1.1 From 3a340104fad6ecbea5ad6792a2ea855f0507a6e0 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Mon, 17 Oct 2011 20:14:56 +0800 Subject: ASoC: wm8741: Fix setting interface format for DSP modes According to the datasheet: Format Control (05h) BITS[3:2] FMT[1:0] Audio data format selection 00 = right justified mode 01 = left justified mode 10 = I2S mode 11 = DSP mode BIT[4] LRP Polarity selec for LRCLK/DSP mode select 0 = normal LRCLK poalrity/DSP mode A 1 = inverted LRCLK poarity/DSP mode B For SND_SOC_DAIFMT_DSP_A, we should set 0x000C instead of 0x0003. For SND_SOC_DAIFMT_DSP_B, we should set 0x001C instead of 0x0013. Signed-off-by: Axel Lin Acked-by: Liam Girdwood Signed-off-by: Mark Brown Cc: stable@kernel.org --- sound/soc/codecs/wm8741.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c index a42b282..85ebe02 100644 --- a/sound/soc/codecs/wm8741.c +++ b/sound/soc/codecs/wm8741.c @@ -339,10 +339,10 @@ static int wm8741_set_dai_fmt(struct snd_soc_dai *codec_dai, iface |= 0x0004; break; case SND_SOC_DAIFMT_DSP_A: - iface |= 0x0003; + iface |= 0x000C; break; case SND_SOC_DAIFMT_DSP_B: - iface |= 0x0013; + iface |= 0x001C; break; default: return -EINVAL; -- cgit v1.1 From df3431b74e72c73e8750bfe1b2a5c99eff958356 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Mon, 17 Oct 2011 20:16:37 +0800 Subject: ASoC: wm8741: Use snd_soc_cache_sync to sync reg_cache with the hardware Signed-off-by: Axel Lin Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/wm8741.c | 10 +--------- 1 file changed, 1 insertion(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c index 85ebe02..57ad22a 100644 --- a/sound/soc/codecs/wm8741.c +++ b/sound/soc/codecs/wm8741.c @@ -404,15 +404,7 @@ static struct snd_soc_dai_driver wm8741_dai = { #ifdef CONFIG_PM static int wm8741_resume(struct snd_soc_codec *codec) { - u16 *cache = codec->reg_cache; - int i; - - /* RESTORE REG Cache */ - for (i = 0; i < WM8741_REGISTER_COUNT; i++) { - if (cache[i] == wm8741_reg_defaults[i] || WM8741_RESET == i) - continue; - snd_soc_write(codec, i, cache[i]); - } + snd_soc_cache_sync(codec); return 0; } #else -- cgit v1.1 From 151b75995a5180834a0609dced3d76ab978cae3b Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Sun, 16 Oct 2011 23:27:55 +0800 Subject: ASoC: wm8900: Fix wrong mask for setting DAC_CLKDIV/ADC_CLKDIV/LRCLK_MODE After checking the datasheet, I think what we want to do here is to clear the WM8900_REG_CLOCKING2_DAC_CLKDIV/WM8900_REG_CLOCKING2_ADC_CLKDIV/ WM8900_REG_DACCTRL_AIF_LRCLKRATE bits and then OR with div value. Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/wm8900.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c index 082040e..b16522f 100644 --- a/sound/soc/codecs/wm8900.c +++ b/sound/soc/codecs/wm8900.c @@ -844,17 +844,17 @@ static int wm8900_set_dai_clkdiv(struct snd_soc_dai *codec_dai, case WM8900_DAC_CLKDIV: reg = snd_soc_read(codec, WM8900_REG_CLOCKING2); snd_soc_write(codec, WM8900_REG_CLOCKING2, - div | (reg & WM8900_REG_CLOCKING2_DAC_CLKDIV)); + div | (reg & ~WM8900_REG_CLOCKING2_DAC_CLKDIV)); break; case WM8900_ADC_CLKDIV: reg = snd_soc_read(codec, WM8900_REG_CLOCKING2); snd_soc_write(codec, WM8900_REG_CLOCKING2, - div | (reg & WM8900_REG_CLOCKING2_ADC_CLKDIV)); + div | (reg & ~WM8900_REG_CLOCKING2_ADC_CLKDIV)); break; case WM8900_LRCLK_MODE: reg = snd_soc_read(codec, WM8900_REG_DACCTRL); snd_soc_write(codec, WM8900_REG_DACCTRL, - div | (reg & WM8900_REG_DACCTRL_AIF_LRCLKRATE)); + div | (reg & ~WM8900_REG_DACCTRL_AIF_LRCLKRATE)); break; default: return -EINVAL; -- cgit v1.1 From de5035b1dda4993f432a796c1d1ddc7b8006b8fe Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Sun, 16 Oct 2011 23:29:12 +0800 Subject: ASoC: wm8900: Fix the mask defines Now we have done bitwise NOT against the mask bits for the defines of WM8900_REG_CLOCKING1_BCLK_MASK, WM8900_REG_CLOCKING1_OPCLK_MASK and WM8900_LRC_MASK. But we don't have the bitwise NOT against the mask bits for the defines of WM8900_REG_CLOCKING2_DAC_CLKDIV, WM8900_REG_CLOCKING2_ADC_CLKDIV and WM8900_REG_DACCTRL_AIF_LRCLKRATE. It is error prone to mix the inconsistent meaning for different mask defines. So lets make the defines for each mask to be corresponding to the bits defines in datasheet. Don't add extra "bitwise NOT" to the defines. Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/wm8900.c | 14 +++++++------- 1 file changed, 7 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c index b16522f..86de396 100644 --- a/sound/soc/codecs/wm8900.c +++ b/sound/soc/codecs/wm8900.c @@ -110,8 +110,8 @@ #define WM8900_REG_CLOCKING1_BCLK_DIR 0x1 #define WM8900_REG_CLOCKING1_MCLK_SRC 0x100 -#define WM8900_REG_CLOCKING1_BCLK_MASK (~0x01e) -#define WM8900_REG_CLOCKING1_OPCLK_MASK (~0x7000) +#define WM8900_REG_CLOCKING1_BCLK_MASK 0x01e +#define WM8900_REG_CLOCKING1_OPCLK_MASK 0x7000 #define WM8900_REG_CLOCKING2_ADC_CLKDIV 0xe0 #define WM8900_REG_CLOCKING2_DAC_CLKDIV 0x1c @@ -135,7 +135,7 @@ #define WM8900_REG_HPCTL1_HP_SHORT 0x08 #define WM8900_REG_HPCTL1_HP_SHORT2 0x04 -#define WM8900_LRC_MASK 0xfc00 +#define WM8900_LRC_MASK 0x03ff struct wm8900_priv { enum snd_soc_control_type control_type; @@ -824,22 +824,22 @@ static int wm8900_set_dai_clkdiv(struct snd_soc_dai *codec_dai, case WM8900_BCLK_DIV: reg = snd_soc_read(codec, WM8900_REG_CLOCKING1); snd_soc_write(codec, WM8900_REG_CLOCKING1, - div | (reg & WM8900_REG_CLOCKING1_BCLK_MASK)); + div | (reg & ~WM8900_REG_CLOCKING1_BCLK_MASK)); break; case WM8900_OPCLK_DIV: reg = snd_soc_read(codec, WM8900_REG_CLOCKING1); snd_soc_write(codec, WM8900_REG_CLOCKING1, - div | (reg & WM8900_REG_CLOCKING1_OPCLK_MASK)); + div | (reg & ~WM8900_REG_CLOCKING1_OPCLK_MASK)); break; case WM8900_DAC_LRCLK: reg = snd_soc_read(codec, WM8900_REG_AUDIO4); snd_soc_write(codec, WM8900_REG_AUDIO4, - div | (reg & WM8900_LRC_MASK)); + div | (reg & ~WM8900_LRC_MASK)); break; case WM8900_ADC_LRCLK: reg = snd_soc_read(codec, WM8900_REG_AUDIO3); snd_soc_write(codec, WM8900_REG_AUDIO3, - div | (reg & WM8900_LRC_MASK)); + div | (reg & ~WM8900_LRC_MASK)); break; case WM8900_DAC_CLKDIV: reg = snd_soc_read(codec, WM8900_REG_CLOCKING2); -- cgit v1.1 From 29c6a01df8de1329303da09ca9793e9f65608216 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Sun, 16 Oct 2011 23:30:21 +0800 Subject: ASoC: wm8900: Use snd_soc_update_bits for read-modify-write Use snd_soc_update_bits for read-modify-write register access instead of open-coding it using snd_soc_read and snd_soc_write Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/wm8900.c | 109 ++++++++++++++++++---------------------------- 1 file changed, 42 insertions(+), 67 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c index 86de396..3d0dc15 100644 --- a/sound/soc/codecs/wm8900.c +++ b/sound/soc/codecs/wm8900.c @@ -742,26 +742,20 @@ static int wm8900_set_fll(struct snd_soc_codec *codec, { struct wm8900_priv *wm8900 = snd_soc_codec_get_drvdata(codec); struct _fll_div fll_div; - unsigned int reg; if (wm8900->fll_in == freq_in && wm8900->fll_out == freq_out) return 0; /* The digital side should be disabled during any change. */ - reg = snd_soc_read(codec, WM8900_REG_POWER1); - snd_soc_write(codec, WM8900_REG_POWER1, - reg & (~WM8900_REG_POWER1_FLL_ENA)); + snd_soc_update_bits(codec, WM8900_REG_POWER1, + WM8900_REG_POWER1_FLL_ENA, 0); /* Disable the FLL? */ if (!freq_in || !freq_out) { - reg = snd_soc_read(codec, WM8900_REG_CLOCKING1); - snd_soc_write(codec, WM8900_REG_CLOCKING1, - reg & (~WM8900_REG_CLOCKING1_MCLK_SRC)); - - reg = snd_soc_read(codec, WM8900_REG_FLLCTL1); - snd_soc_write(codec, WM8900_REG_FLLCTL1, - reg & (~WM8900_REG_FLLCTL1_OSC_ENA)); - + snd_soc_update_bits(codec, WM8900_REG_CLOCKING1, + WM8900_REG_CLOCKING1_MCLK_SRC, 0); + snd_soc_update_bits(codec, WM8900_REG_FLLCTL1, + WM8900_REG_FLLCTL1_OSC_ENA, 0); wm8900->fll_in = freq_in; wm8900->fll_out = freq_out; @@ -796,15 +790,14 @@ static int wm8900_set_fll(struct snd_soc_codec *codec, else snd_soc_write(codec, WM8900_REG_FLLCTL6, 0); - reg = snd_soc_read(codec, WM8900_REG_POWER1); - snd_soc_write(codec, WM8900_REG_POWER1, - reg | WM8900_REG_POWER1_FLL_ENA); + snd_soc_update_bits(codec, WM8900_REG_POWER1, + WM8900_REG_POWER1_FLL_ENA, + WM8900_REG_POWER1_FLL_ENA); reenable: - reg = snd_soc_read(codec, WM8900_REG_CLOCKING1); - snd_soc_write(codec, WM8900_REG_CLOCKING1, - reg | WM8900_REG_CLOCKING1_MCLK_SRC); - + snd_soc_update_bits(codec, WM8900_REG_CLOCKING1, + WM8900_REG_CLOCKING1_MCLK_SRC, + WM8900_REG_CLOCKING1_MCLK_SRC); return 0; } @@ -818,43 +811,35 @@ static int wm8900_set_dai_clkdiv(struct snd_soc_dai *codec_dai, int div_id, int div) { struct snd_soc_codec *codec = codec_dai->codec; - unsigned int reg; switch (div_id) { case WM8900_BCLK_DIV: - reg = snd_soc_read(codec, WM8900_REG_CLOCKING1); - snd_soc_write(codec, WM8900_REG_CLOCKING1, - div | (reg & ~WM8900_REG_CLOCKING1_BCLK_MASK)); + snd_soc_update_bits(codec, WM8900_REG_CLOCKING1, + WM8900_REG_CLOCKING1_BCLK_MASK, div); break; case WM8900_OPCLK_DIV: - reg = snd_soc_read(codec, WM8900_REG_CLOCKING1); - snd_soc_write(codec, WM8900_REG_CLOCKING1, - div | (reg & ~WM8900_REG_CLOCKING1_OPCLK_MASK)); + snd_soc_update_bits(codec, WM8900_REG_CLOCKING1, + WM8900_REG_CLOCKING1_OPCLK_MASK, div); break; case WM8900_DAC_LRCLK: - reg = snd_soc_read(codec, WM8900_REG_AUDIO4); - snd_soc_write(codec, WM8900_REG_AUDIO4, - div | (reg & ~WM8900_LRC_MASK)); + snd_soc_update_bits(codec, WM8900_REG_AUDIO4, + WM8900_LRC_MASK, div); break; case WM8900_ADC_LRCLK: - reg = snd_soc_read(codec, WM8900_REG_AUDIO3); - snd_soc_write(codec, WM8900_REG_AUDIO3, - div | (reg & ~WM8900_LRC_MASK)); + snd_soc_update_bits(codec, WM8900_REG_AUDIO3, + WM8900_LRC_MASK, div); break; case WM8900_DAC_CLKDIV: - reg = snd_soc_read(codec, WM8900_REG_CLOCKING2); - snd_soc_write(codec, WM8900_REG_CLOCKING2, - div | (reg & ~WM8900_REG_CLOCKING2_DAC_CLKDIV)); + snd_soc_update_bits(codec, WM8900_REG_CLOCKING2, + WM8900_REG_CLOCKING2_DAC_CLKDIV, div); break; case WM8900_ADC_CLKDIV: - reg = snd_soc_read(codec, WM8900_REG_CLOCKING2); - snd_soc_write(codec, WM8900_REG_CLOCKING2, - div | (reg & ~WM8900_REG_CLOCKING2_ADC_CLKDIV)); + snd_soc_update_bits(codec, WM8900_REG_CLOCKING2, + WM8900_REG_CLOCKING2_ADC_CLKDIV, div); break; case WM8900_LRCLK_MODE: - reg = snd_soc_read(codec, WM8900_REG_DACCTRL); - snd_soc_write(codec, WM8900_REG_DACCTRL, - div | (reg & ~WM8900_REG_DACCTRL_AIF_LRCLKRATE)); + snd_soc_update_bits(codec, WM8900_REG_DACCTRL, + WM8900_REG_DACCTRL_AIF_LRCLKRATE, div); break; default: return -EINVAL; @@ -1037,12 +1022,12 @@ static int wm8900_set_bias_level(struct snd_soc_codec *codec, switch (level) { case SND_SOC_BIAS_ON: /* Enable thermal shutdown */ - reg = snd_soc_read(codec, WM8900_REG_GPIO); - snd_soc_write(codec, WM8900_REG_GPIO, - reg | WM8900_REG_GPIO_TEMP_ENA); - reg = snd_soc_read(codec, WM8900_REG_ADDCTL); - snd_soc_write(codec, WM8900_REG_ADDCTL, - reg | WM8900_REG_ADDCTL_TEMP_SD); + snd_soc_update_bits(codec, WM8900_REG_GPIO, + WM8900_REG_GPIO_TEMP_ENA, + WM8900_REG_GPIO_TEMP_ENA); + snd_soc_update_bits(codec, WM8900_REG_ADDCTL, + WM8900_REG_ADDCTL_TEMP_SD, + WM8900_REG_ADDCTL_TEMP_SD); break; case SND_SOC_BIAS_PREPARE: @@ -1205,26 +1190,16 @@ static int wm8900_probe(struct snd_soc_codec *codec) wm8900_set_bias_level(codec, SND_SOC_BIAS_STANDBY); /* Latch the volume update bits */ - snd_soc_write(codec, WM8900_REG_LINVOL, - snd_soc_read(codec, WM8900_REG_LINVOL) | 0x100); - snd_soc_write(codec, WM8900_REG_RINVOL, - snd_soc_read(codec, WM8900_REG_RINVOL) | 0x100); - snd_soc_write(codec, WM8900_REG_LOUT1CTL, - snd_soc_read(codec, WM8900_REG_LOUT1CTL) | 0x100); - snd_soc_write(codec, WM8900_REG_ROUT1CTL, - snd_soc_read(codec, WM8900_REG_ROUT1CTL) | 0x100); - snd_soc_write(codec, WM8900_REG_LOUT2CTL, - snd_soc_read(codec, WM8900_REG_LOUT2CTL) | 0x100); - snd_soc_write(codec, WM8900_REG_ROUT2CTL, - snd_soc_read(codec, WM8900_REG_ROUT2CTL) | 0x100); - snd_soc_write(codec, WM8900_REG_LDAC_DV, - snd_soc_read(codec, WM8900_REG_LDAC_DV) | 0x100); - snd_soc_write(codec, WM8900_REG_RDAC_DV, - snd_soc_read(codec, WM8900_REG_RDAC_DV) | 0x100); - snd_soc_write(codec, WM8900_REG_LADC_DV, - snd_soc_read(codec, WM8900_REG_LADC_DV) | 0x100); - snd_soc_write(codec, WM8900_REG_RADC_DV, - snd_soc_read(codec, WM8900_REG_RADC_DV) | 0x100); + snd_soc_update_bits(codec, WM8900_REG_LINVOL, 0x100, 0x100); + snd_soc_update_bits(codec, WM8900_REG_RINVOL, 0x100, 0x100); + snd_soc_update_bits(codec, WM8900_REG_LOUT1CTL, 0x100, 0x100); + snd_soc_update_bits(codec, WM8900_REG_ROUT1CTL, 0x100, 0x100); + snd_soc_update_bits(codec, WM8900_REG_LOUT2CTL, 0x100, 0x100); + snd_soc_update_bits(codec, WM8900_REG_ROUT2CTL, 0x100, 0x100); + snd_soc_update_bits(codec, WM8900_REG_LDAC_DV, 0x100, 0x100); + snd_soc_update_bits(codec, WM8900_REG_RDAC_DV, 0x100, 0x100); + snd_soc_update_bits(codec, WM8900_REG_LADC_DV, 0x100, 0x100); + snd_soc_update_bits(codec, WM8900_REG_RADC_DV, 0x100, 0x100); /* Set the DAC and mixer output bias */ snd_soc_write(codec, WM8900_REG_OUTBIASCTL, 0x81); -- cgit v1.1 From a6785d7df8d2790d99a4788a612764a92fb9f498 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Mon, 17 Oct 2011 11:50:46 +0800 Subject: ASoC: wm8580: Use snd_soc_update_bits for read-modify-write Use snd_soc_update_bits for read-modify-write register access instead of open-coding it using snd_soc_read and snd_soc_write Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/wm8580.c | 22 +++++++++------------- 1 file changed, 9 insertions(+), 13 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index b256727..8212b3c 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -430,8 +430,7 @@ static int wm8580_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, /* Always disable the PLL - it is not safe to leave it running * while reprogramming it. */ - reg = snd_soc_read(codec, WM8580_PWRDN2); - snd_soc_write(codec, WM8580_PWRDN2, reg | pwr_mask); + snd_soc_update_bits(codec, WM8580_PWRDN2, pwr_mask, pwr_mask); if (!freq_in || !freq_out) return 0; @@ -449,8 +448,7 @@ static int wm8580_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, snd_soc_write(codec, WM8580_PLLA4 + offset, reg); /* All done, turn it on */ - reg = snd_soc_read(codec, WM8580_PWRDN2); - snd_soc_write(codec, WM8580_PWRDN2, reg & ~pwr_mask); + snd_soc_update_bits(codec, WM8580_PWRDN2, pwr_mask, 0); return 0; } @@ -748,7 +746,6 @@ static int wm8580_digital_mute(struct snd_soc_dai *codec_dai, int mute) static int wm8580_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { - u16 reg; switch (level) { case SND_SOC_BIAS_ON: case SND_SOC_BIAS_PREPARE: @@ -757,20 +754,19 @@ static int wm8580_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_STANDBY: if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { /* Power up and get individual control of the DACs */ - reg = snd_soc_read(codec, WM8580_PWRDN1); - reg &= ~(WM8580_PWRDN1_PWDN | WM8580_PWRDN1_ALLDACPD); - snd_soc_write(codec, WM8580_PWRDN1, reg); + snd_soc_update_bits(codec, WM8580_PWRDN1, + WM8580_PWRDN1_PWDN | + WM8580_PWRDN1_ALLDACPD, 0); /* Make VMID high impedance */ - reg = snd_soc_read(codec, WM8580_ADC_CONTROL1); - reg &= ~0x100; - snd_soc_write(codec, WM8580_ADC_CONTROL1, reg); + snd_soc_update_bits(codec, WM8580_ADC_CONTROL1, + 0x100, 0); } break; case SND_SOC_BIAS_OFF: - reg = snd_soc_read(codec, WM8580_PWRDN1); - snd_soc_write(codec, WM8580_PWRDN1, reg | WM8580_PWRDN1_PWDN); + snd_soc_update_bits(codec, WM8580_PWRDN1, + WM8580_PWRDN1_PWDN, WM8580_PWRDN1_PWDN); break; } codec->dapm.bias_level = level; -- cgit v1.1 From 6473a148058f8d65fc013a57090354dc737f6143 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 17 Oct 2011 19:38:52 +0100 Subject: ASoC: Update WM1811 DCS codes for latest evaluation results Evaluation of larger quantities of material has provided new DCS codes values to be applied for WM1811. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8994.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 68e769e..f56c7c5 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3163,8 +3163,8 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) switch (wm8994->revision) { case 0: case 1: - wm8994->hubs.dcs_codes_l = -7; - wm8994->hubs.dcs_codes_r = -4; + wm8994->hubs.dcs_codes_l = -9; + wm8994->hubs.dcs_codes_r = -5; break; default: break; -- cgit v1.1 From 680fa1f807bc65ea147c6c3ea52751cab75bcd34 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 17 Oct 2011 23:53:37 +0100 Subject: ASoC: Convert WM9081 to table based control init At least for the core controls, the optionally selected controls are still added programatically. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm9081.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index b2d3448..2b6a75c 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -1248,8 +1248,6 @@ static int wm9081_probe(struct snd_soc_codec *codec) snd_soc_write(codec, WM9081_ANALOGUE_SPEAKER_PGA, reg | WM9081_SPKPGAZC); - snd_soc_add_controls(codec, wm9081_snd_controls, - ARRAY_SIZE(wm9081_snd_controls)); if (!wm9081->pdata.num_retune_configs) { dev_dbg(codec->dev, "No ReTune Mobile data, using normal EQ\n"); @@ -1309,6 +1307,8 @@ static struct snd_soc_codec_driver soc_codec_dev_wm9081 = { .reg_cache_default = wm9081_reg_defaults, .volatile_register = wm9081_volatile_register, + .controls = wm9081_snd_controls, + .num_controls = ARRAY_SIZE(wm9081_snd_controls), .dapm_widgets = wm9081_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(wm9081_dapm_widgets), .dapm_routes = wm9081_audio_paths, -- cgit v1.1 From 4b1cfcb4f36dca33b4552d9612537e6c0ca73639 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 18 Oct 2011 00:11:49 +0100 Subject: ASoC: Fix prefixing of DAPM controls We don't want to clear the prefix while we're creating the DAPM controls for the device as the prefix is applied during control creation. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/soc-core.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index d6af52b..a5d3685 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1055,13 +1055,13 @@ static int soc_post_component_init(struct snd_soc_card *card, } rtd->card = card; + /* Make sure all DAPM widgets are instantiated */ + snd_soc_dapm_new_widgets(&codec->dapm); + /* machine controls, routes and widgets are not prefixed */ temp = codec->name_prefix; codec->name_prefix = NULL; - /* Make sure all DAPM widgets are instantiated */ - snd_soc_dapm_new_widgets(&codec->dapm); - /* do machine specific initialization */ if (!dailess && dai_link->init) ret = dai_link->init(rtd); -- cgit v1.1 From 54c96cfd1ac815d278aa43f37d063b0c5972db1f Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Tue, 18 Oct 2011 06:25:08 +0800 Subject: ASoC: ad193x: Use snd_soc_update_bits for read-modify-write Use snd_soc_update_bits for read-modify-write register access instead of open-coding it using snd_soc_read and snd_soc_write Signed-off-by: Axel Lin Acked-by: Lars-Peter Clausen Acked-by: Barry Song <21cnbao@gmail.com> Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/ad193x.c | 31 +++++++++++++++---------------- 1 file changed, 15 insertions(+), 16 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c index 39056ce..1206021 100644 --- a/sound/soc/codecs/ad193x.c +++ b/sound/soc/codecs/ad193x.c @@ -103,12 +103,14 @@ static const struct snd_soc_dapm_route audio_paths[] = { static int ad193x_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; - int reg; - reg = snd_soc_read(codec, AD193X_DAC_CTRL2); - reg = (mute > 0) ? reg | AD193X_DAC_MASTER_MUTE : reg & - (~AD193X_DAC_MASTER_MUTE); - snd_soc_write(codec, AD193X_DAC_CTRL2, reg); + if (mute) + snd_soc_update_bits(codec, AD193X_DAC_CTRL2, + AD193X_DAC_MASTER_MUTE, + AD193X_DAC_MASTER_MUTE); + else + snd_soc_update_bits(codec, AD193X_DAC_CTRL2, + AD193X_DAC_MASTER_MUTE, 0); return 0; } @@ -262,7 +264,7 @@ static int ad193x_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - int word_len = 0, reg = 0, master_rate = 0; + int word_len = 0, master_rate = 0; struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_codec *codec = rtd->codec; @@ -297,18 +299,15 @@ static int ad193x_hw_params(struct snd_pcm_substream *substream, break; } - reg = snd_soc_read(codec, AD193X_PLL_CLK_CTRL0); - reg = (reg & (~AD193X_PLL_INPUT_MASK)) | master_rate; - snd_soc_write(codec, AD193X_PLL_CLK_CTRL0, reg); + snd_soc_update_bits(codec, AD193X_PLL_CLK_CTRL0, + AD193X_PLL_INPUT_MASK, master_rate); - reg = snd_soc_read(codec, AD193X_DAC_CTRL2); - reg = (reg & (~AD193X_DAC_WORD_LEN_MASK)) - | (word_len << AD193X_DAC_WORD_LEN_SHFT); - snd_soc_write(codec, AD193X_DAC_CTRL2, reg); + snd_soc_update_bits(codec, AD193X_DAC_CTRL2, + AD193X_DAC_WORD_LEN_MASK, + word_len << AD193X_DAC_WORD_LEN_SHFT); - reg = snd_soc_read(codec, AD193X_ADC_CTRL1); - reg = (reg & (~AD193X_ADC_WORD_LEN_MASK)) | word_len; - snd_soc_write(codec, AD193X_ADC_CTRL1, reg); + snd_soc_update_bits(codec, AD193X_ADC_CTRL1, + AD193X_ADC_WORD_LEN_MASK, word_len); return 0; } -- cgit v1.1 From 56c09aa520ad488c35c580d6f6fb1821bb4317b8 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Wed, 19 Oct 2011 10:54:56 +0800 Subject: ASoC: sgtl5000: Fix define for SGTL5000_BIAS_R_MASK According to the datasheet: CHIP_MIC_CTRL 0x002A BITS[9:8] BIAS_RESISTOR 0x0 = Powerd off 0x1 = 2.0 kohm 0x2 = 4.0 kohm 0x3 = 8.0 kohm Thus SGTL5000_BIAS_R_MASK should be defined as 0x0300 instead of 0x0200. Signed-off-by: Axel Lin Acked-by: Wolfram Sang Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/sgtl5000.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/sgtl5000.h b/sound/soc/codecs/sgtl5000.h index eec3ab3..8a9f435 100644 --- a/sound/soc/codecs/sgtl5000.h +++ b/sound/soc/codecs/sgtl5000.h @@ -280,7 +280,7 @@ /* * SGTL5000_CHIP_MIC_CTRL */ -#define SGTL5000_BIAS_R_MASK 0x0200 +#define SGTL5000_BIAS_R_MASK 0x0300 #define SGTL5000_BIAS_R_SHIFT 8 #define SGTL5000_BIAS_R_WIDTH 2 #define SGTL5000_BIAS_R_off 0x0 -- cgit v1.1 From dc56c5a862d1491dcdc561241371949cca6362e1 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Wed, 19 Oct 2011 11:00:42 +0800 Subject: ASoC: sgtl5000: Fix setting mic bias resistor According to the datasheet: CHIP_MIC_CTRL 0x002A BITS[9:8] BIAS_RESISTOR 0x0 = Powerd off 0x1 = 2.0 kohm 0x2 = 4.0 kohm 0x3 = 8.0 kohm To set mic bias resistor, we need to update bits[9:8] of SGTL5000_CHIP_MIC_CTRL register. Signed-off-by: Axel Lin Acked-by: Wolfram Sang Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/sgtl5000.c | 9 +++------ 1 file changed, 3 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 91130fb..3637a62 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -131,16 +131,13 @@ static int mic_bias_event(struct snd_soc_dapm_widget *w, case SND_SOC_DAPM_POST_PMU: /* change mic bias resistor to 4Kohm */ snd_soc_update_bits(w->codec, SGTL5000_CHIP_MIC_CTRL, - SGTL5000_BIAS_R_4k, SGTL5000_BIAS_R_4k); + SGTL5000_BIAS_R_MASK, + SGTL5000_BIAS_R_4k << SGTL5000_BIAS_R_SHIFT); break; case SND_SOC_DAPM_PRE_PMD: - /* - * SGTL5000_BIAS_R_8k as mask to clean the two bits - * of mic bias and output impedance - */ snd_soc_update_bits(w->codec, SGTL5000_CHIP_MIC_CTRL, - SGTL5000_BIAS_R_8k, 0); + SGTL5000_BIAS_R_MASK, 0); break; } return 0; -- cgit v1.1 From f8faadb6f204049252fe832d28df04640fa7e88e Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 18 Oct 2011 19:31:38 +0100 Subject: ASoC: WM9081 interrupt status register is volatile Not that we have interrupt handling in the driver at the minute. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm9081.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index 2b6a75c..8176138 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -173,6 +173,7 @@ static int wm9081_volatile_register(struct snd_soc_codec *codec, unsigned int re { switch (reg) { case WM9081_SOFTWARE_RESET: + case WM9081_INTERRUPT_STATUS: return 1; default: return 0; -- cgit v1.1 From 2ee9c183f39f6e77f65a9e3414ff469a4dc34a0b Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Wed, 19 Oct 2011 14:07:31 +0800 Subject: ASoC: ssm2602: Use snd_soc_update_bits for read-modify-write Use snd_soc_update_bits for read-modify-write register access instead of open-coding it using snd_soc_read and snd_soc_write Signed-off-by: Axel Lin Acked-by: Lars-Peter Clausen Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/ssm2602.c | 30 ++++++++++++++++-------------- 1 file changed, 16 insertions(+), 14 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index e149ec6..3cb3271 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -343,12 +343,14 @@ static void ssm2602_shutdown(struct snd_pcm_substream *substream, static int ssm2602_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; - u16 mute_reg = snd_soc_read(codec, SSM2602_APDIGI) & ~APDIGI_ENABLE_DAC_MUTE; + if (mute) - snd_soc_write(codec, SSM2602_APDIGI, - mute_reg | APDIGI_ENABLE_DAC_MUTE); + snd_soc_update_bits(codec, SSM2602_APDIGI, + APDIGI_ENABLE_DAC_MUTE, + APDIGI_ENABLE_DAC_MUTE); else - snd_soc_write(codec, SSM2602_APDIGI, mute_reg); + snd_soc_update_bits(codec, SSM2602_APDIGI, + APDIGI_ENABLE_DAC_MUTE, 0); return 0; } @@ -540,12 +542,12 @@ static int ssm2602_resume(struct snd_soc_codec *codec) static int ssm2602_probe(struct snd_soc_codec *codec) { struct snd_soc_dapm_context *dapm = &codec->dapm; - int ret, reg; + int ret; - reg = snd_soc_read(codec, SSM2602_LOUT1V); - snd_soc_write(codec, SSM2602_LOUT1V, reg | LOUT1V_LRHP_BOTH); - reg = snd_soc_read(codec, SSM2602_ROUT1V); - snd_soc_write(codec, SSM2602_ROUT1V, reg | ROUT1V_RLHP_BOTH); + snd_soc_update_bits(codec, SSM2602_LOUT1V, + LOUT1V_LRHP_BOTH, LOUT1V_LRHP_BOTH); + snd_soc_update_bits(codec, SSM2602_ROUT1V, + ROUT1V_RLHP_BOTH, ROUT1V_RLHP_BOTH); ret = snd_soc_add_controls(codec, ssm2602_snd_controls, ARRAY_SIZE(ssm2602_snd_controls)); @@ -578,7 +580,7 @@ static int ssm2604_probe(struct snd_soc_codec *codec) static int ssm260x_probe(struct snd_soc_codec *codec) { struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec); - int ret, reg; + int ret; pr_info("ssm2602 Audio Codec %s", SSM2602_VERSION); @@ -595,10 +597,10 @@ static int ssm260x_probe(struct snd_soc_codec *codec) } /* set the update bits */ - reg = snd_soc_read(codec, SSM2602_LINVOL); - snd_soc_write(codec, SSM2602_LINVOL, reg | LINVOL_LRIN_BOTH); - reg = snd_soc_read(codec, SSM2602_RINVOL); - snd_soc_write(codec, SSM2602_RINVOL, reg | RINVOL_RLIN_BOTH); + snd_soc_update_bits(codec, SSM2602_LINVOL, + LINVOL_LRIN_BOTH, LINVOL_LRIN_BOTH); + snd_soc_update_bits(codec, SSM2602_RINVOL, + RINVOL_RLIN_BOTH, RINVOL_RLIN_BOTH); /*select Line in as default input*/ snd_soc_write(codec, SSM2602_APANA, APANA_SELECT_DAC | APANA_ENABLE_MIC_BOOST); -- cgit v1.1 From 812b404c90d302e3f352568606c8c37c3ee1e4c7 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Wed, 19 Oct 2011 23:05:56 +0800 Subject: ASoC: ak4641: Remove unused codec field from struct ak4641_priv Signed-off-by: Axel Lin Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/ak4641.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/ak4641.c b/sound/soc/codecs/ak4641.c index 7a64e58..f5125ae 100644 --- a/sound/soc/codecs/ak4641.c +++ b/sound/soc/codecs/ak4641.c @@ -31,7 +31,6 @@ /* codec private data */ struct ak4641_priv { - struct snd_soc_codec *codec; unsigned int sysclk; int deemph; int playback_fs; -- cgit v1.1 From 5eda19497b0af2533a69f67b552cf7baae11f377 Mon Sep 17 00:00:00 2001 From: Ashish Chavan Date: Wed, 19 Oct 2011 14:19:06 +0530 Subject: ASoC: da7210: Add support for mute and zero cross controls This patch adds support for below set of controls, (1) Mute controls for MIC, AUX and ADC (2) Zero cross controls for head phone, AUX, INPGA and line out (3) Head phone mode selection - class H or G It also adds digital_mute() call back. Signed-off-by: Ashish Chavan Signed-off-by: David Dajun Chen Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/da7210.c | 44 ++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 44 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index fa0d512..4aad01c 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -30,6 +30,7 @@ #define DA7210_STARTUP1 0x03 #define DA7210_MIC_L 0x07 #define DA7210_MIC_R 0x08 +#define DA7210_AUX2 0x0B #define DA7210_INMIX_L 0x0D #define DA7210_INMIX_R 0x0E #define DA7210_ADC_HPF 0x0F @@ -41,6 +42,7 @@ #define DA7210_DAC_L 0x15 #define DA7210_DAC_R 0x16 #define DA7210_DAC_SEL 0x17 +#define DA7210_SOFTMUTE 0x18 #define DA7210_DAC_EQ1_2 0x19 #define DA7210_DAC_EQ3_4 0x1A #define DA7210_DAC_EQ5 0x1B @@ -49,6 +51,7 @@ #define DA7210_HP_L_VOL 0x21 #define DA7210_HP_R_VOL 0x22 #define DA7210_HP_CFG 0x23 +#define DA7210_ZERO_CROSS 0x24 #define DA7210_DAI_SRC_SEL 0x25 #define DA7210_DAI_CFG1 0x26 #define DA7210_DAI_CFG3 0x28 @@ -144,6 +147,9 @@ #define DA7210_PLL_FS_96000 (0xF << 0) #define DA7210_PLL_EN (0x1 << 7) +/* SOFTMUTE bit fields */ +#define DA7210_RAMP_EN (1 << 6) + #define DA7210_VERSION "0.0.1" /* @@ -189,6 +195,13 @@ static const struct soc_enum da7210_dac_vf_cutoff = static const struct soc_enum da7210_adc_vf_cutoff = SOC_ENUM_SINGLE(DA7210_ADC_HPF, 4, 8, da7210_vf_cutoff_txt); +static const char *da7210_hp_mode_txt[] = { + "Class H", "Class G" +}; + +static const struct soc_enum da7210_hp_mode_sel = + SOC_ENUM_SINGLE(DA7210_HP_CFG, 0, 2, da7210_hp_mode_txt); + static const struct snd_kcontrol_new da7210_snd_controls[] = { SOC_DOUBLE_R_TLV("HeadPhone Playback Volume", @@ -232,6 +245,21 @@ static const struct snd_kcontrol_new da7210_snd_controls[] = { SOC_ENUM("ADC HPF Cutoff", da7210_adc_hpf_cutoff), SOC_SINGLE("ADC Voice Mode Switch", DA7210_ADC_HPF, 7, 1, 0), SOC_ENUM("ADC Voice Cutoff", da7210_adc_vf_cutoff), + + /* Mute controls */ + SOC_DOUBLE_R("Mic Capture Switch", DA7210_MIC_L, DA7210_MIC_R, 3, 1, 0), + SOC_SINGLE("Aux2 Capture Switch", DA7210_AUX2, 2, 1, 0), + SOC_DOUBLE("ADC Capture Switch", DA7210_ADC, 2, 6, 1, 0), + SOC_SINGLE("Digital Soft Mute Switch", DA7210_SOFTMUTE, 7, 1, 0), + SOC_SINGLE("Digital Soft Mute Rate", DA7210_SOFTMUTE, 0, 0x7, 0), + + /* Zero cross controls */ + SOC_DOUBLE("Aux1 ZC Switch", DA7210_ZERO_CROSS, 0, 1, 1, 0), + SOC_DOUBLE("In PGA ZC Switch", DA7210_ZERO_CROSS, 2, 3, 1, 0), + SOC_DOUBLE("Lineout ZC Switch", DA7210_ZERO_CROSS, 4, 5, 1, 0), + SOC_DOUBLE("Headphone ZC Switch", DA7210_ZERO_CROSS, 6, 7, 1, 0), + + SOC_ENUM("Headphone Class", da7210_hp_mode_sel), }; /* Codec private data */ @@ -448,6 +476,18 @@ static int da7210_set_dai_fmt(struct snd_soc_dai *codec_dai, u32 fmt) return 0; } +static int da7210_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + u8 mute_reg = snd_soc_read(codec, DA7210_DAC_HPF) & 0xFB; + + if (mute) + snd_soc_write(codec, DA7210_DAC_HPF, mute_reg | 0x4); + else + snd_soc_write(codec, DA7210_DAC_HPF, mute_reg); + return 0; +} + #define DA7210_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) @@ -456,6 +496,7 @@ static struct snd_soc_dai_ops da7210_dai_ops = { .startup = da7210_startup, .hw_params = da7210_hw_params, .set_fmt = da7210_set_dai_fmt, + .digital_mute = da7210_mute, }; static struct snd_soc_dai_driver da7210_dai = { @@ -545,6 +586,9 @@ static int da7210_probe(struct snd_soc_codec *codec) DA7210_HP_2CAP_MODE | DA7210_HP_SENSE_EN | DA7210_HP_L_EN | DA7210_HP_MODE | DA7210_HP_R_EN); + /* Enable ramp mode for DAC gain update */ + snd_soc_write(codec, DA7210_SOFTMUTE, DA7210_RAMP_EN); + /* Diable PLL and bypass it */ snd_soc_write(codec, DA7210_PLL, DA7210_PLL_FS_48000); -- cgit v1.1 From de5eaf844e936cc80d9edde56eaa1025a1642210 Mon Sep 17 00:00:00 2001 From: Ashish Chavan Date: Wed, 19 Oct 2011 14:24:37 +0530 Subject: ASoC: da7210: Add support for ALC and Noise suppression This patch adds controls to set following ALC parameters, - Max gain, Min gain, Noise gain, Attack rate, Release rate and delay It also adds a switch to enable/disable noise suppression. As per DA7210 data sheet, ALC and noise suppression can be enabled only if certain conditions are met. This condition checks are handled by simply using "_EXT" version of controls to capture change events. Signed-off-by: Ashish Chavan Signed-off-by: David Dajun Chen Acked-by: Liam Girdwod Signed-off-by: Mark Brown --- sound/soc/codecs/da7210.c | 110 ++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 110 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index 4aad01c..e9ee6a4 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -26,11 +26,15 @@ #include /* DA7210 register space */ +#define DA7210_CONTROL 0x01 #define DA7210_STATUS 0x02 #define DA7210_STARTUP1 0x03 #define DA7210_MIC_L 0x07 #define DA7210_MIC_R 0x08 +#define DA7210_AUX1_L 0x09 +#define DA7210_AUX1_R 0x0A #define DA7210_AUX2 0x0B +#define DA7210_IN_GAIN 0x0C #define DA7210_INMIX_L 0x0D #define DA7210_INMIX_R 0x0E #define DA7210_ADC_HPF 0x0F @@ -59,6 +63,12 @@ #define DA7210_PLL_DIV2 0x2A #define DA7210_PLL_DIV3 0x2B #define DA7210_PLL 0x2C +#define DA7210_ALC_MAX 0x83 +#define DA7210_ALC_MIN 0x84 +#define DA7210_ALC_NOIS 0x85 +#define DA7210_ALC_ATT 0x86 +#define DA7210_ALC_REL 0x87 +#define DA7210_ALC_DEL 0x88 #define DA7210_A_HID_UNLOCK 0x8A #define DA7210_A_TEST_UNLOCK 0x8B #define DA7210_A_PLL1 0x90 @@ -81,6 +91,7 @@ #define DA7210_IN_R_EN (1 << 7) /* ADC bit fields */ +#define DA7210_ADC_ALC_EN (1 << 0) #define DA7210_ADC_L_EN (1 << 3) #define DA7210_ADC_R_EN (1 << 7) @@ -150,6 +161,29 @@ /* SOFTMUTE bit fields */ #define DA7210_RAMP_EN (1 << 6) +/* CONTROL bit fields */ +#define DA7210_NOISE_SUP_EN (1 << 3) + +/* IN_GAIN bit fields */ +#define DA7210_INPGA_L_VOL (0x0F << 0) +#define DA7210_INPGA_R_VOL (0xF0 << 0) + +/* ZERO_CROSS bit fields */ +#define DA7210_AUX1_L_ZC (1 << 0) +#define DA7210_AUX1_R_ZC (1 << 1) +#define DA7210_HP_L_ZC (1 << 6) +#define DA7210_HP_R_ZC (1 << 7) + +/* AUX1_L bit fields */ +#define DA7210_AUX1_L_VOL (0x3F << 0) + +/* AUX1_R bit fields */ +#define DA7210_AUX1_R_VOL (0x3F << 0) + +/* Minimum INPGA and AUX1 volume to enable noise suppression */ +#define DA7210_INPGA_MIN_VOL_NS 0x0A /* 10.5dB */ +#define DA7210_AUX1_MIN_VOL_NS 0x35 /* 6dB */ + #define DA7210_VERSION "0.0.1" /* @@ -202,6 +236,69 @@ static const char *da7210_hp_mode_txt[] = { static const struct soc_enum da7210_hp_mode_sel = SOC_ENUM_SINGLE(DA7210_HP_CFG, 0, 2, da7210_hp_mode_txt); +/* ALC can be enabled only if noise suppression is disabled */ +static int da7210_put_alc_sw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + + if (ucontrol->value.integer.value[0]) { + /* Check if noise suppression is enabled */ + if (snd_soc_read(codec, DA7210_CONTROL) & DA7210_NOISE_SUP_EN) { + dev_dbg(codec->dev, + "Disable noise suppression to enable ALC\n"); + return -EINVAL; + } + } + /* If all conditions are met or we are actually disabling ALC */ + return snd_soc_put_volsw(kcontrol, ucontrol); +} + +/* Noise suppression can be enabled only if following conditions are met + * ALC disabled + * ZC enabled for HP and AUX1 PGA + * INPGA_L_VOL and INPGA_R_VOL >= 10.5 dB + * AUX1_L_VOL and AUX1_R_VOL >= 6 dB + */ +static int da7210_put_noise_sup_sw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + u8 val; + + if (ucontrol->value.integer.value[0]) { + /* Check if ALC is enabled */ + if (snd_soc_read(codec, DA7210_ADC) & DA7210_ADC_ALC_EN) + goto err; + + /* Check ZC for HP and AUX1 PGA */ + if ((snd_soc_read(codec, DA7210_ZERO_CROSS) & + (DA7210_AUX1_L_ZC | DA7210_AUX1_R_ZC | DA7210_HP_L_ZC | + DA7210_HP_R_ZC)) != (DA7210_AUX1_L_ZC | + DA7210_AUX1_R_ZC | DA7210_HP_L_ZC | DA7210_HP_R_ZC)) + goto err; + + /* Check INPGA_L_VOL and INPGA_R_VOL */ + val = snd_soc_read(codec, DA7210_IN_GAIN); + if (((val & DA7210_INPGA_L_VOL) < DA7210_INPGA_MIN_VOL_NS) || + (((val & DA7210_INPGA_R_VOL) >> 4) < + DA7210_INPGA_MIN_VOL_NS)) + goto err; + + /* Check AUX1_L_VOL and AUX1_R_VOL */ + if (((snd_soc_read(codec, DA7210_AUX1_L) & DA7210_AUX1_L_VOL) < + DA7210_AUX1_MIN_VOL_NS) || + ((snd_soc_read(codec, DA7210_AUX1_R) & DA7210_AUX1_R_VOL) < + DA7210_AUX1_MIN_VOL_NS)) + goto err; + } + /* If all conditions are met or we are actually disabling Noise sup */ + return snd_soc_put_volsw(kcontrol, ucontrol); + +err: + return -EINVAL; +} + static const struct snd_kcontrol_new da7210_snd_controls[] = { SOC_DOUBLE_R_TLV("HeadPhone Playback Volume", @@ -260,6 +357,19 @@ static const struct snd_kcontrol_new da7210_snd_controls[] = { SOC_DOUBLE("Headphone ZC Switch", DA7210_ZERO_CROSS, 6, 7, 1, 0), SOC_ENUM("Headphone Class", da7210_hp_mode_sel), + + /* ALC controls */ + SOC_SINGLE_EXT("ALC Enable Switch", DA7210_ADC, 0, 1, 0, + snd_soc_get_volsw, da7210_put_alc_sw), + SOC_SINGLE("ALC Capture Max Volume", DA7210_ALC_MAX, 0, 0x3F, 0), + SOC_SINGLE("ALC Capture Min Volume", DA7210_ALC_MIN, 0, 0x3F, 0), + SOC_SINGLE("ALC Capture Noise Volume", DA7210_ALC_NOIS, 0, 0x3F, 0), + SOC_SINGLE("ALC Capture Attack Rate", DA7210_ALC_ATT, 0, 0xFF, 0), + SOC_SINGLE("ALC Capture Release Rate", DA7210_ALC_REL, 0, 0xFF, 0), + SOC_SINGLE("ALC Capture Release Delay", DA7210_ALC_DEL, 0, 0xFF, 0), + + SOC_SINGLE_EXT("Noise Suppression Enable Switch", DA7210_CONTROL, 3, 1, + 0, snd_soc_get_volsw, da7210_put_noise_sup_sw), }; /* Codec private data */ -- cgit v1.1 From 24441e17feb57668e4c5013750cef549bf3c4861 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Wed, 19 Oct 2011 23:24:54 +0800 Subject: ASoC: ak4641: Use SND_SOC_DAPM_DAC for Voice Playback stream widget Signed-off-by: Axel Lin Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/ak4641.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/ak4641.c b/sound/soc/codecs/ak4641.c index f5125ae..7783858 100644 --- a/sound/soc/codecs/ak4641.c +++ b/sound/soc/codecs/ak4641.c @@ -225,7 +225,7 @@ static const struct snd_soc_dapm_widget ak4641_dapm_widgets[] = { SND_SOC_DAPM_PGA("Mono Out 2", AK4641_PM2, 3, 0, NULL, 0), SND_SOC_DAPM_ADC("Voice ADC", "Voice Capture", AK4641_BTIF, 0, 0), - SND_SOC_DAPM_ADC("Voice DAC", "Voice Playback", AK4641_BTIF, 1, 0), + SND_SOC_DAPM_DAC("Voice DAC", "Voice Playback", AK4641_BTIF, 1, 0), SND_SOC_DAPM_MICBIAS("Mic Int Bias", AK4641_MIC, 3, 0), SND_SOC_DAPM_MICBIAS("Mic Ext Bias", AK4641_MIC, 4, 0), -- cgit v1.1 From 9f9619a0785f8eee42edf731fd18189faa5a7ce8 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Mon, 17 Oct 2011 12:34:31 +0800 Subject: ASoC: wm9081: Fix setting soft VMID ramp enable with VMID 2*240k According to the datasheet: BIT 2:1 VMID_SEL[1:0] VMID Divider Enable and Select 00 = VMID disabled 01 = 2x40k Omh divider 10 = 2x240k Omh divider 11 = 2x5k Omh divider To set VMID 2*240k, we should OR reg with 0x04 instead of 0x40. Signed-off-by: Axel Lin Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/wm9081.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index 8176138..3cd35a0 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -820,7 +820,7 @@ static int wm9081_set_bias_level(struct snd_soc_codec *codec, /* VMID 2*240k */ reg = snd_soc_read(codec, WM9081_BIAS_CONTROL_1); reg &= ~WM9081_VMID_SEL_MASK; - reg |= 0x40; + reg |= 0x04; snd_soc_write(codec, WM9081_VMID_CONTROL, reg); /* Standby bias current on */ -- cgit v1.1 From cf0feafbc306718292dcda499bf299fc60cc8cc6 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Thu, 20 Oct 2011 10:50:03 +0800 Subject: ASoC: Fix reg_word_size for ak4104 According to the register map in datasheet, the registers are 8 bit. Signed-off-by: Axel Lin Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/ak4104.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/ak4104.c b/sound/soc/codecs/ak4104.c index cbf0b6d..d3b29dc 100644 --- a/sound/soc/codecs/ak4104.c +++ b/sound/soc/codecs/ak4104.c @@ -247,7 +247,7 @@ static struct snd_soc_codec_driver soc_codec_device_ak4104 = { .probe = ak4104_probe, .remove = ak4104_remove, .reg_cache_size = AK4104_NUM_REGS, - .reg_word_size = sizeof(u16), + .reg_word_size = sizeof(u8), }; static int ak4104_spi_probe(struct spi_device *spi) -- cgit v1.1 From f96c255df75782c97dca8e2529bc09cb80425fe7 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Thu, 20 Oct 2011 10:54:13 +0800 Subject: ASoC: ak4671: Use snd_soc_update_bits for read-modify-write Use snd_soc_update_bits for read-modify-write register access instead of open-coding it using snd_soc_read and snd_soc_write Signed-off-by: Axel Lin Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/ak4671.c | 18 ++++++------------ 1 file changed, 6 insertions(+), 12 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/ak4671.c b/sound/soc/codecs/ak4671.c index 41e3d55..de9ff66 100644 --- a/sound/soc/codecs/ak4671.c +++ b/sound/soc/codecs/ak4671.c @@ -168,18 +168,15 @@ static int ak4671_out2_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { struct snd_soc_codec *codec = w->codec; - u8 reg; switch (event) { case SND_SOC_DAPM_POST_PMU: - reg = snd_soc_read(codec, AK4671_LOUT2_POWER_MANAGERMENT); - reg |= AK4671_MUTEN; - snd_soc_write(codec, AK4671_LOUT2_POWER_MANAGERMENT, reg); + snd_soc_update_bits(codec, AK4671_LOUT2_POWER_MANAGERMENT, + AK4671_MUTEN, AK4671_MUTEN); break; case SND_SOC_DAPM_PRE_PMD: - reg = snd_soc_read(codec, AK4671_LOUT2_POWER_MANAGERMENT); - reg &= ~AK4671_MUTEN; - snd_soc_write(codec, AK4671_LOUT2_POWER_MANAGERMENT, reg); + snd_soc_update_bits(codec, AK4671_LOUT2_POWER_MANAGERMENT, + AK4671_MUTEN, 0); break; } @@ -575,15 +572,12 @@ static int ak4671_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) static int ak4671_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { - u8 reg; - switch (level) { case SND_SOC_BIAS_ON: case SND_SOC_BIAS_PREPARE: case SND_SOC_BIAS_STANDBY: - reg = snd_soc_read(codec, AK4671_AD_DA_POWER_MANAGEMENT); - snd_soc_write(codec, AK4671_AD_DA_POWER_MANAGEMENT, - reg | AK4671_PMVCM); + snd_soc_update_bits(codec, AK4671_AD_DA_POWER_MANAGEMENT, + AK4671_PMVCM, AK4671_PMVCM); break; case SND_SOC_BIAS_OFF: snd_soc_write(codec, AK4671_AD_DA_POWER_MANAGEMENT, 0x00); -- cgit v1.1 From 6765ff778e8f887e518504bebfdd10b5db5c800d Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Thu, 20 Oct 2011 11:00:06 +0800 Subject: ASoC: rt5631: Remove unused codec field from struct rt5631_priv Signed-off-by: Axel Lin Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/rt5631.c | 2 -- 1 file changed, 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt5631.c b/sound/soc/codecs/rt5631.c index 86e69f4..27a078c 100644 --- a/sound/soc/codecs/rt5631.c +++ b/sound/soc/codecs/rt5631.c @@ -31,7 +31,6 @@ #include "rt5631.h" struct rt5631_priv { - struct snd_soc_codec *codec; int codec_version; int master; int sysclk; @@ -1632,7 +1631,6 @@ static int rt5631_probe(struct snd_soc_codec *codec) } codec->dapm.bias_level = SND_SOC_BIAS_STANDBY; - rt5631->codec = codec; return 0; } -- cgit v1.1 From 35024f4922f7b271e7529673413889aa3d51c5fc Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Thu, 20 Oct 2011 12:13:24 +0800 Subject: ASoC: wm8994: Use SND_SOC_DAPM_AIF_OUT for AIF3 Capture Signed-off-by: Axel Lin Acked-by: Liam Girdwood Signed-off-by: Mark Brown Cc: stable@kernel.org --- sound/soc/codecs/wm8994.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index f56c7c5..6b73efd 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -1419,7 +1419,7 @@ SND_SOC_DAPM_MUX("AIF2DAC Mux", SND_SOC_NOPM, 0, 0, &aif2dac_mux), SND_SOC_DAPM_MUX("AIF2ADC Mux", SND_SOC_NOPM, 0, 0, &aif2adc_mux), SND_SOC_DAPM_AIF_IN("AIF3DACDAT", "AIF3 Playback", 0, SND_SOC_NOPM, 0, 0), -SND_SOC_DAPM_AIF_IN("AIF3ADCDAT", "AIF3 Capture", 0, SND_SOC_NOPM, 0, 0), +SND_SOC_DAPM_AIF_OUT("AIF3ADCDAT", "AIF3 Capture", 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_SUPPLY("TOCLK", WM8994_CLOCKING_1, 4, 0, NULL, 0), -- cgit v1.1 From ff39dbe93543d5d4118fddf247db48431f984648 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Thu, 20 Oct 2011 12:16:31 +0800 Subject: ASoC: wm8996: Use SND_SOC_DAPM_AIF_OUT for AIF2 Capture Signed-off-by: Axel Lin Acked-by: Liam Girdwood Signed-off-by: Mark Brown Cc: stable@kernel.org --- sound/soc/codecs/wm8996.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index 43e46c7..1d7b174 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -1113,9 +1113,9 @@ SND_SOC_DAPM_AIF_IN("AIF2RX1", "AIF2 Playback", 0, SND_SOC_DAPM_AIF_IN("AIF2RX0", "AIF2 Playback", 1, WM8996_POWER_MANAGEMENT_4, 8, 0), -SND_SOC_DAPM_AIF_IN("AIF2TX1", "AIF2 Capture", 0, +SND_SOC_DAPM_AIF_OUT("AIF2TX1", "AIF2 Capture", 0, WM8996_POWER_MANAGEMENT_6, 9, 0), -SND_SOC_DAPM_AIF_IN("AIF2TX0", "AIF2 Capture", 1, +SND_SOC_DAPM_AIF_OUT("AIF2TX0", "AIF2 Capture", 1, WM8996_POWER_MANAGEMENT_6, 8, 0), SND_SOC_DAPM_AIF_IN("AIF1RX5", "AIF1 Playback", 5, -- cgit v1.1 From 5b13de7aa754eaa274fc9ab018191bcdcb21bc45 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Thu, 20 Oct 2011 18:32:59 +0800 Subject: ASoC: Set sgtl5000->ldo in ldo_regulator_register Otherwise calling ldo_regulator_remove() does not unregister regulator and free memories. Signed-off-by: Axel Lin Acked-by: Wolfram Sang Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/sgtl5000.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 3637a62..8395002 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -806,6 +806,7 @@ static int ldo_regulator_register(struct snd_soc_codec *codec, int voltage) { struct ldo_regulator *ldo; + struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); ldo = kzalloc(sizeof(struct ldo_regulator), GFP_KERNEL); @@ -840,6 +841,7 @@ static int ldo_regulator_register(struct snd_soc_codec *codec, return ret; } + sgtl5000->ldo = ldo; return 0; } -- cgit v1.1 From 064a4bcee4114e519ce22d56d2eb8e9dfa653804 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Thu, 20 Oct 2011 18:49:29 +0800 Subject: ASoC: Use SGTL5000_LINREG_VDDD_MASK instead of hardcoded mask value We have defined SGTL5000_LINREG_VDDD_MASK in sgtl5000.h, use it instead of hardcoded (0x1 << 4) - 1 for the mask. Signed-off-by: Axel Lin Acked-by: Wolfram Sang Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/sgtl5000.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 8395002..32b5bbd 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -754,7 +754,7 @@ static int ldo_regulator_enable(struct regulator_dev *dev) /* set voltage to register */ snd_soc_update_bits(codec, SGTL5000_CHIP_LINREG_CTRL, - (0x1 << 4) - 1, reg); + SGTL5000_LINREG_VDDD_MASK, reg); snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER, SGTL5000_LINEREG_D_POWERUP, @@ -780,7 +780,7 @@ static int ldo_regulator_disable(struct regulator_dev *dev) /* clear voltage info */ snd_soc_update_bits(codec, SGTL5000_CHIP_LINREG_CTRL, - (0x1 << 4) - 1, 0); + SGTL5000_LINREG_VDDD_MASK, 0); ldo->enabled = 0; @@ -1115,7 +1115,7 @@ static int sgtl5000_set_power_regs(struct snd_soc_codec *codec) /* set voltage to register */ snd_soc_update_bits(codec, SGTL5000_CHIP_LINREG_CTRL, - (0x1 << 4) - 1, 0x8); + SGTL5000_LINREG_VDDD_MASK, 0x8); /* * if vddd linear reg has been enabled, -- cgit v1.1 From 6950c60dc1a0981a6a99bece52437965be8e1be0 Mon Sep 17 00:00:00 2001 From: Ashish Chavan Date: Fri, 21 Oct 2011 18:16:08 +0530 Subject: ASoC: da7210: Add support for DAPM This patch adds support for DAPM covering all inputs and outputs as well as ADC and DAC. Signed-off-by: Ashish Chavan Signed-off-by: David Dajun Chen Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/da7210.c | 145 ++++++++++++++++++++++++++++++++++++++-------- 1 file changed, 121 insertions(+), 24 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index e9ee6a4..7a4b952 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -29,6 +29,8 @@ #define DA7210_CONTROL 0x01 #define DA7210_STATUS 0x02 #define DA7210_STARTUP1 0x03 +#define DA7210_STARTUP2 0x04 +#define DA7210_STARTUP3 0x05 #define DA7210_MIC_L 0x07 #define DA7210_MIC_R 0x08 #define DA7210_AUX1_L 0x09 @@ -372,6 +374,120 @@ static const struct snd_kcontrol_new da7210_snd_controls[] = { 0, snd_soc_get_volsw, da7210_put_noise_sup_sw), }; +/* + * DAPM Controls + * + * Current DAPM implementation covers almost all codec components e.g. IOs, + * mixers, PGAs,ADC and DAC. + */ +/* In Mixer Left */ +static const struct snd_kcontrol_new da7210_dapm_inmixl_controls[] = { + SOC_DAPM_SINGLE("Mic Left Switch", DA7210_INMIX_L, 0, 1, 0), + SOC_DAPM_SINGLE("Mic Right Switch", DA7210_INMIX_L, 1, 1, 0), +}; + +/* In Mixer Right */ +static const struct snd_kcontrol_new da7210_dapm_inmixr_controls[] = { + SOC_DAPM_SINGLE("Mic Right Switch", DA7210_INMIX_R, 0, 1, 0), + SOC_DAPM_SINGLE("Mic Left Switch", DA7210_INMIX_R, 1, 1, 0), +}; + +/* Out Mixer Left */ +static const struct snd_kcontrol_new da7210_dapm_outmixl_controls[] = { + SOC_DAPM_SINGLE("DAC Left Switch", DA7210_OUTMIX_L, 4, 1, 0), +}; + +/* Out Mixer Right */ +static const struct snd_kcontrol_new da7210_dapm_outmixr_controls[] = { + SOC_DAPM_SINGLE("DAC Right Switch", DA7210_OUTMIX_R, 4, 1, 0), +}; + +/* DAPM widgets */ +static const struct snd_soc_dapm_widget da7210_dapm_widgets[] = { + /* Input Side */ + /* Input Lines */ + SND_SOC_DAPM_INPUT("MICL"), + SND_SOC_DAPM_INPUT("MICR"), + + /* Input PGAs */ + SND_SOC_DAPM_PGA("Mic Left", DA7210_STARTUP3, 0, 1, NULL, 0), + SND_SOC_DAPM_PGA("Mic Right", DA7210_STARTUP3, 1, 1, NULL, 0), + + SND_SOC_DAPM_PGA("INPGA Left", DA7210_INMIX_L, 7, 0, NULL, 0), + SND_SOC_DAPM_PGA("INPGA Right", DA7210_INMIX_R, 7, 0, NULL, 0), + + /* Input Mixers */ + SND_SOC_DAPM_MIXER("In Mixer Left", SND_SOC_NOPM, 0, 0, + &da7210_dapm_inmixl_controls[0], + ARRAY_SIZE(da7210_dapm_inmixl_controls)), + + SND_SOC_DAPM_MIXER("In Mixer Right", SND_SOC_NOPM, 0, 0, + &da7210_dapm_inmixr_controls[0], + ARRAY_SIZE(da7210_dapm_inmixr_controls)), + + /* ADCs */ + SND_SOC_DAPM_ADC("ADC Left", "Capture", DA7210_STARTUP3, 5, 1), + SND_SOC_DAPM_ADC("ADC Right", "Capture", DA7210_STARTUP3, 6, 1), + + /* Output Side */ + /* DACs */ + SND_SOC_DAPM_DAC("DAC Left", "Playback", DA7210_STARTUP2, 5, 1), + SND_SOC_DAPM_DAC("DAC Right", "Playback", DA7210_STARTUP2, 6, 1), + + /* Output Mixers */ + SND_SOC_DAPM_MIXER("Out Mixer Left", SND_SOC_NOPM, 0, 0, + &da7210_dapm_outmixl_controls[0], + ARRAY_SIZE(da7210_dapm_outmixl_controls)), + + SND_SOC_DAPM_MIXER("Out Mixer Right", SND_SOC_NOPM, 0, 0, + &da7210_dapm_outmixr_controls[0], + ARRAY_SIZE(da7210_dapm_outmixr_controls)), + + /* Output PGAs */ + SND_SOC_DAPM_PGA("OUTPGA Left Enable", DA7210_OUTMIX_L, 7, 0, NULL, 0), + SND_SOC_DAPM_PGA("OUTPGA Right Enable", DA7210_OUTMIX_R, 7, 0, NULL, 0), + + SND_SOC_DAPM_PGA("Headphone Left", DA7210_STARTUP2, 3, 1, NULL, 0), + SND_SOC_DAPM_PGA("Headphone Right", DA7210_STARTUP2, 4, 1, NULL, 0), + + /* Output Lines */ + SND_SOC_DAPM_OUTPUT("HPL"), + SND_SOC_DAPM_OUTPUT("HPR"), +}; + +/* DAPM audio route definition */ +static const struct snd_soc_dapm_route da7210_audio_map[] = { + /* Dest Connecting Widget source */ + /* Input path */ + {"Mic Left", NULL, "MICL"}, + {"Mic Right", NULL, "MICR"}, + + {"In Mixer Left", "Mic Left Switch", "Mic Left"}, + {"In Mixer Left", "Mic Right Switch", "Mic Right"}, + + {"In Mixer Right", "Mic Right Switch", "Mic Right"}, + {"In Mixer Right", "Mic Left Switch", "Mic Left"}, + + {"INPGA Left", NULL, "In Mixer Left"}, + {"ADC Left", NULL, "INPGA Left"}, + + {"INPGA Right", NULL, "In Mixer Right"}, + {"ADC Right", NULL, "INPGA Right"}, + + /* Output path */ + {"Out Mixer Left", "DAC Left Switch", "DAC Left"}, + {"Out Mixer Right", "DAC Right Switch", "DAC Right"}, + + {"OUTPGA Left Enable", NULL, "Out Mixer Left"}, + {"OUTPGA Right Enable", NULL, "Out Mixer Right"}, + + {"Headphone Left", NULL, "OUTPGA Left Enable"}, + {"HPL", NULL, "Headphone Left"}, + + {"Headphone Right", NULL, "OUTPGA Right Enable"}, + {"HPR", NULL, "Headphone Right"}, +}; + /* Codec private data */ struct da7210_priv { enum snd_soc_control_type control_type; @@ -411,29 +527,6 @@ static int da7210_volatile_register(struct snd_soc_codec *codec, return 0; } } -static int da7210_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; - struct snd_soc_codec *codec = dai->codec; - - if (is_play) { - /* Enable Out */ - snd_soc_update_bits(codec, DA7210_OUTMIX_L, 0x1F, 0x10); - snd_soc_update_bits(codec, DA7210_OUTMIX_R, 0x1F, 0x10); - - } else { - /* Volume 7 */ - snd_soc_update_bits(codec, DA7210_MIC_L, 0x7, 0x7); - snd_soc_update_bits(codec, DA7210_MIC_R, 0x7, 0x7); - - /* Enable Mic */ - snd_soc_update_bits(codec, DA7210_INMIX_L, 0x1F, 0x1); - snd_soc_update_bits(codec, DA7210_INMIX_R, 0x1F, 0x1); - } - - return 0; -} /* * Set PCM DAI word length. @@ -603,7 +696,6 @@ static int da7210_mute(struct snd_soc_dai *dai, int mute) /* DAI operations */ static struct snd_soc_dai_ops da7210_dai_ops = { - .startup = da7210_startup, .hw_params = da7210_hw_params, .set_fmt = da7210_set_dai_fmt, .digital_mute = da7210_mute, @@ -742,6 +834,11 @@ static struct snd_soc_codec_driver soc_codec_dev_da7210 = { .controls = da7210_snd_controls, .num_controls = ARRAY_SIZE(da7210_snd_controls), + + .dapm_widgets = da7210_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(da7210_dapm_widgets), + .dapm_routes = da7210_audio_map, + .num_dapm_routes = ARRAY_SIZE(da7210_audio_map), }; #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) -- cgit v1.1 From 52082d8f562bb4ed4045ea691a3ec1f44d828eab Mon Sep 17 00:00:00 2001 From: Ashish Chavan Date: Fri, 21 Oct 2011 19:06:23 +0530 Subject: ASoC: da7210: Add support for line out and DAC DA7210 has three line outputs. OUT1 Left, OUT1 Right and OUT2 (mono). This patch adds support for gain controls for these three line outs. It also adds support for overall DAC gain control. Signed-off-by: Ashish Chavan Signed-off-by: David Dajun Chen Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/da7210.c | 96 +++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 96 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index 7a4b952..0ebcbd5 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -54,6 +54,9 @@ #define DA7210_DAC_EQ5 0x1B #define DA7210_OUTMIX_L 0x1C #define DA7210_OUTMIX_R 0x1D +#define DA7210_OUT1_L 0x1E +#define DA7210_OUT1_R 0x1F +#define DA7210_OUT2 0x20 #define DA7210_HP_L_VOL 0x21 #define DA7210_HP_R_VOL 0x22 #define DA7210_HP_CFG 0x23 @@ -186,6 +189,17 @@ #define DA7210_INPGA_MIN_VOL_NS 0x0A /* 10.5dB */ #define DA7210_AUX1_MIN_VOL_NS 0x35 /* 6dB */ +/* OUT1_L bit fields */ +#define DA7210_OUT1_L_EN (1 << 7) + +/* OUT1_R bit fields */ +#define DA7210_OUT1_R_EN (1 << 7) + +/* OUT2 bit fields */ +#define DA7210_OUT2_OUTMIX_R (1 << 5) +#define DA7210_OUT2_OUTMIX_L (1 << 6) +#define DA7210_OUT2_EN (1 << 7) + #define DA7210_VERSION "0.0.1" /* @@ -206,8 +220,23 @@ static const unsigned int hp_out_tlv[] = { 0x11, 0x3f, TLV_DB_SCALE_ITEM(-5400, 150, 0), }; +static const unsigned int lineout_vol_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0x0, 0x10, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 1), + /* -54dB to 15dB */ + 0x11, 0x3f, TLV_DB_SCALE_ITEM(-5400, 150, 0) +}; + +static const unsigned int mono_vol_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0x0, 0x2, TLV_DB_SCALE_ITEM(-1800, 0, 1), + /* -18dB to 6dB */ + 0x3, 0x7, TLV_DB_SCALE_ITEM(-1800, 600, 0) +}; + static const DECLARE_TLV_DB_SCALE(eq_gain_tlv, -1050, 150, 0); static const DECLARE_TLV_DB_SCALE(adc_eq_master_gain_tlv, -1800, 600, 1); +static const DECLARE_TLV_DB_SCALE(dac_gain_tlv, -7725, 75, 0); /* ADC and DAC high pass filter f0 value */ static const char const *da7210_hpf_cutoff_txt[] = { @@ -306,6 +335,14 @@ static const struct snd_kcontrol_new da7210_snd_controls[] = { SOC_DOUBLE_R_TLV("HeadPhone Playback Volume", DA7210_HP_L_VOL, DA7210_HP_R_VOL, 0, 0x3F, 0, hp_out_tlv), + SOC_DOUBLE_R_TLV("Digital Playback Volume", + DA7210_DAC_L, DA7210_DAC_R, + 0, 0x77, 1, dac_gain_tlv), + SOC_DOUBLE_R_TLV("Lineout Playback Volume", + DA7210_OUT1_L, DA7210_OUT1_R, + 0, 0x3f, 0, lineout_vol_tlv), + SOC_SINGLE_TLV("Mono Playback Volume", DA7210_OUT2, 0, 0x7, 0, + mono_vol_tlv), /* DAC Equalizer controls */ SOC_SINGLE("DAC EQ Switch", DA7210_DAC_EQ5, 7, 1, 0), @@ -402,6 +439,12 @@ static const struct snd_kcontrol_new da7210_dapm_outmixr_controls[] = { SOC_DAPM_SINGLE("DAC Right Switch", DA7210_OUTMIX_R, 4, 1, 0), }; +/* Mono Mixer */ +static const struct snd_kcontrol_new da7210_dapm_monomix_controls[] = { + SOC_DAPM_SINGLE("Outmix Right Switch", DA7210_OUT2, 5, 1, 0), + SOC_DAPM_SINGLE("Outmix Left Switch", DA7210_OUT2, 6, 1, 0), +}; + /* DAPM widgets */ static const struct snd_soc_dapm_widget da7210_dapm_widgets[] = { /* Input Side */ @@ -443,16 +486,26 @@ static const struct snd_soc_dapm_widget da7210_dapm_widgets[] = { &da7210_dapm_outmixr_controls[0], ARRAY_SIZE(da7210_dapm_outmixr_controls)), + SND_SOC_DAPM_MIXER("Mono Mixer", SND_SOC_NOPM, 0, 0, + &da7210_dapm_monomix_controls[0], + ARRAY_SIZE(da7210_dapm_monomix_controls)), + /* Output PGAs */ SND_SOC_DAPM_PGA("OUTPGA Left Enable", DA7210_OUTMIX_L, 7, 0, NULL, 0), SND_SOC_DAPM_PGA("OUTPGA Right Enable", DA7210_OUTMIX_R, 7, 0, NULL, 0), + SND_SOC_DAPM_PGA("Out1 Left", DA7210_STARTUP2, 0, 1, NULL, 0), + SND_SOC_DAPM_PGA("Out1 Right", DA7210_STARTUP2, 1, 1, NULL, 0), + SND_SOC_DAPM_PGA("Out2 Mono", DA7210_STARTUP2, 2, 1, NULL, 0), SND_SOC_DAPM_PGA("Headphone Left", DA7210_STARTUP2, 3, 1, NULL, 0), SND_SOC_DAPM_PGA("Headphone Right", DA7210_STARTUP2, 4, 1, NULL, 0), /* Output Lines */ + SND_SOC_DAPM_OUTPUT("OUT1L"), + SND_SOC_DAPM_OUTPUT("OUT1R"), SND_SOC_DAPM_OUTPUT("HPL"), SND_SOC_DAPM_OUTPUT("HPR"), + SND_SOC_DAPM_OUTPUT("OUT2"), }; /* DAPM audio route definition */ @@ -478,14 +531,26 @@ static const struct snd_soc_dapm_route da7210_audio_map[] = { {"Out Mixer Left", "DAC Left Switch", "DAC Left"}, {"Out Mixer Right", "DAC Right Switch", "DAC Right"}, + {"Mono Mixer", "Outmix Right Switch", "Out Mixer Right"}, + {"Mono Mixer", "Outmix Left Switch", "Out Mixer Left"}, + {"OUTPGA Left Enable", NULL, "Out Mixer Left"}, {"OUTPGA Right Enable", NULL, "Out Mixer Right"}, + {"Out1 Left", NULL, "OUTPGA Left Enable"}, + {"OUT1L", NULL, "Out1 Left"}, + + {"Out1 Right", NULL, "OUTPGA Right Enable"}, + {"OUT1R", NULL, "Out1 Right"}, + {"Headphone Left", NULL, "OUTPGA Left Enable"}, {"HPL", NULL, "Headphone Left"}, {"Headphone Right", NULL, "OUTPGA Right Enable"}, {"HPR", NULL, "Headphone Right"}, + + {"Out2 Mono", NULL, "Mono Mixer"}, + {"OUT2", NULL, "Out2 Mono"}, }; /* Codec private data */ @@ -791,6 +856,37 @@ static int da7210_probe(struct snd_soc_codec *codec) /* Enable ramp mode for DAC gain update */ snd_soc_write(codec, DA7210_SOFTMUTE, DA7210_RAMP_EN); + /* + * For DA7210 codec, there are two ways to enable/disable analog IOs + * and ADC/DAC, + * (1) Using "Enable Bit" of register associated with that IO + * (or ADC/DAC) + * e.g. Mic Left can be enabled using bit 7 of MIC_L(0x7) reg + * + * (2) Using "Standby Bit" of STARTUP2 or STARTUP3 register + * e.g. Mic left can be put to STANDBY using bit 0 of STARTUP3(0x5) + * + * Out of these two methods, the one using STANDBY bits is preferred + * way to enable/disable individual blocks. This is because STANDBY + * registers are part of system controller which allows system power + * up/down in a controlled, pop-free manner. Also, as per application + * note of DA7210, STANDBY register bits are only effective if a + * particular IO (or ADC/DAC) is already enabled using enable/disable + * register bits. Keeping these things in mind, current DAPM + * implementation manipulates only STANDBY bits. + * + * Overall implementation can be outlined as below, + * + * - "Enable bit" of an IO or ADC/DAC is used to enable it in probe() + * - "STANDBY bit" is controlled by DAPM + */ + + /* Enable Line out amplifiers */ + snd_soc_write(codec, DA7210_OUT1_L, DA7210_OUT1_L_EN); + snd_soc_write(codec, DA7210_OUT1_R, DA7210_OUT1_R_EN); + snd_soc_write(codec, DA7210_OUT2, DA7210_OUT2_EN | + DA7210_OUT2_OUTMIX_L | DA7210_OUT2_OUTMIX_R); + /* Diable PLL and bypass it */ snd_soc_write(codec, DA7210_PLL, DA7210_PLL_FS_48000); -- cgit v1.1 From 3205e6629bc0eb747fb7d1b4b8fec00b7b919e58 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Fri, 21 Oct 2011 10:44:07 +0800 Subject: ASoC: wm8996: Fix wrong mask for setting WM8996_AIF_CLOCKING_2 Signed-off-by: Axel Lin Acked-by: Liam Girdwood Signed-off-by: Mark Brown Cc: stable@kernel.org --- sound/soc/codecs/wm8996.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index 1d7b174..645c980 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -1920,7 +1920,7 @@ static int wm8996_hw_params(struct snd_pcm_substream *substream, snd_soc_update_bits(codec, lrclk_reg, WM8996_AIF1RX_RATE_MASK, lrclk); snd_soc_update_bits(codec, WM8996_AIF_CLOCKING_2, - WM8996_DSP1_DIV_SHIFT << dsp_shift, dsp); + WM8996_DSP1_DIV_MASK << dsp_shift, dsp); return 0; } -- cgit v1.1 From 33cb92cff9568dd9feb2825bd3605bf099bc6b63 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Fri, 21 Oct 2011 09:54:43 +0800 Subject: ASoC: sgtl5000: Fix wrong mask in some snd_soc_update_bits calls Ensure all mask bits are clear before setting new value. Signed-off-by: Axel Lin Acked-by: Dong Aisheng Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/sgtl5000.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 32b5bbd..d15695d 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -723,7 +723,9 @@ static int sgtl5000_pcm_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - snd_soc_update_bits(codec, SGTL5000_CHIP_I2S_CTRL, i2s_ctl, i2s_ctl); + snd_soc_update_bits(codec, SGTL5000_CHIP_I2S_CTRL, + SGTL5000_I2S_DLEN_MASK | SGTL5000_I2S_SCLKFREQ_MASK, + i2s_ctl); return 0; } @@ -1146,8 +1148,7 @@ static int sgtl5000_set_power_regs(struct snd_soc_codec *codec) vag = (vag - SGTL5000_ANA_GND_BASE) / SGTL5000_ANA_GND_STP; snd_soc_update_bits(codec, SGTL5000_CHIP_REF_CTRL, - vag << SGTL5000_ANA_GND_SHIFT, - vag << SGTL5000_ANA_GND_SHIFT); + SGTL5000_ANA_GND_MASK, vag << SGTL5000_ANA_GND_SHIFT); /* set line out VAG to vddio / 2, in range (0.8v, 1.675v) */ vag = vddio / 2; @@ -1161,9 +1162,8 @@ static int sgtl5000_set_power_regs(struct snd_soc_codec *codec) SGTL5000_LINE_OUT_GND_STP; snd_soc_update_bits(codec, SGTL5000_CHIP_LINE_OUT_CTRL, - vag << SGTL5000_LINE_OUT_GND_SHIFT | - SGTL5000_LINE_OUT_CURRENT_360u << - SGTL5000_LINE_OUT_CURRENT_SHIFT, + SGTL5000_LINE_OUT_CURRENT_MASK | + SGTL5000_LINE_OUT_GND_MASK, vag << SGTL5000_LINE_OUT_GND_SHIFT | SGTL5000_LINE_OUT_CURRENT_360u << SGTL5000_LINE_OUT_CURRENT_SHIFT); -- cgit v1.1 From 226d0f22d044f0151287bb7cf334b85182248f0e Mon Sep 17 00:00:00 2001 From: Julia Lawall Date: Tue, 18 Oct 2011 17:06:39 +0200 Subject: ASoC: keep pointer to resource so it can be freed Add a new variable for storing resources accessed subsequent to the one accessed using request_mem_region, so the one accessed using request_mem_region can be released if needed. The resource variable names are also changed to be more descriptive. This code is also missing some calls to iounmap. The semantic match that finds this problem is as follows: (http://coccinelle.lip6.fr/) // @r@ expression E, E1; identifier f; statement S1,S2,S3; @@ if (E == NULL) { ... when != if (E == NULL || ...) S1 else S2 when != E = E1 *E->f ... when any return ...; } else S3 // Signed-off-by: Julia Lawall Signed-off-by: Mark Brown --- sound/soc/au1x/ac97c.c | 33 ++++++++++++++++++--------------- sound/soc/au1x/i2sc.c | 33 ++++++++++++++++++--------------- sound/soc/au1x/psc-ac97.c | 25 +++++++++++++------------ sound/soc/au1x/psc-i2s.c | 25 +++++++++++++------------ sound/soc/mxs/mxs-saif.c | 19 ++++++++++--------- 5 files changed, 72 insertions(+), 63 deletions(-) (limited to 'sound') diff --git a/sound/soc/au1x/ac97c.c b/sound/soc/au1x/ac97c.c index 13802ff..726bd65 100644 --- a/sound/soc/au1x/ac97c.c +++ b/sound/soc/au1x/ac97c.c @@ -226,7 +226,7 @@ static struct snd_soc_dai_driver au1xac97c_dai_driver = { static int __devinit au1xac97c_drvprobe(struct platform_device *pdev) { int ret; - struct resource *r; + struct resource *iores, *dmares; struct au1xpsc_audio_data *ctx; ctx = kzalloc(sizeof(*ctx), GFP_KERNEL); @@ -235,29 +235,30 @@ static int __devinit au1xac97c_drvprobe(struct platform_device *pdev) mutex_init(&ctx->lock); - r = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!r) { + iores = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!iores) { ret = -ENODEV; goto out0; } ret = -EBUSY; - if (!request_mem_region(r->start, resource_size(r), pdev->name)) + if (!request_mem_region(iores->start, resource_size(iores), + pdev->name)) goto out0; - ctx->mmio = ioremap_nocache(r->start, resource_size(r)); + ctx->mmio = ioremap_nocache(iores->start, resource_size(iores)); if (!ctx->mmio) goto out1; - r = platform_get_resource(pdev, IORESOURCE_DMA, 0); - if (!r) - goto out1; - ctx->dmaids[SNDRV_PCM_STREAM_PLAYBACK] = r->start; + dmares = platform_get_resource(pdev, IORESOURCE_DMA, 0); + if (!dmares) + goto out2; + ctx->dmaids[SNDRV_PCM_STREAM_PLAYBACK] = dmares->start; - r = platform_get_resource(pdev, IORESOURCE_DMA, 1); - if (!r) - goto out1; - ctx->dmaids[SNDRV_PCM_STREAM_CAPTURE] = r->start; + dmares = platform_get_resource(pdev, IORESOURCE_DMA, 1); + if (!dmares) + goto out2; + ctx->dmaids[SNDRV_PCM_STREAM_CAPTURE] = dmares->start; /* switch it on */ WR(ctx, AC97_ENABLE, EN_D | EN_CE); @@ -270,13 +271,15 @@ static int __devinit au1xac97c_drvprobe(struct platform_device *pdev) ret = snd_soc_register_dai(&pdev->dev, &au1xac97c_dai_driver); if (ret) - goto out1; + goto out2; ac97c_workdata = ctx; return 0; +out2: + iounmap(ctx->mmio); out1: - release_mem_region(r->start, resource_size(r)); + release_mem_region(iores->start, resource_size(iores)); out0: kfree(ctx); return ret; diff --git a/sound/soc/au1x/i2sc.c b/sound/soc/au1x/i2sc.c index 19e0d2a..6bcf48f 100644 --- a/sound/soc/au1x/i2sc.c +++ b/sound/soc/au1x/i2sc.c @@ -228,47 +228,50 @@ static struct snd_soc_dai_driver au1xi2s_dai_driver = { static int __devinit au1xi2s_drvprobe(struct platform_device *pdev) { int ret; - struct resource *r; + struct resource *iores, *dmares; struct au1xpsc_audio_data *ctx; ctx = kzalloc(sizeof(*ctx), GFP_KERNEL); if (!ctx) return -ENOMEM; - r = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!r) { + iores = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!iores) { ret = -ENODEV; goto out0; } ret = -EBUSY; - if (!request_mem_region(r->start, resource_size(r), pdev->name)) + if (!request_mem_region(iores->start, resource_size(iores), + pdev->name)) goto out0; - ctx->mmio = ioremap_nocache(r->start, resource_size(r)); + ctx->mmio = ioremap_nocache(iores->start, resource_size(iores)); if (!ctx->mmio) goto out1; - r = platform_get_resource(pdev, IORESOURCE_DMA, 0); - if (!r) - goto out1; - ctx->dmaids[SNDRV_PCM_STREAM_PLAYBACK] = r->start; + dmares = platform_get_resource(pdev, IORESOURCE_DMA, 0); + if (!dmares) + goto out2; + ctx->dmaids[SNDRV_PCM_STREAM_PLAYBACK] = dmares->start; - r = platform_get_resource(pdev, IORESOURCE_DMA, 1); - if (!r) - goto out1; - ctx->dmaids[SNDRV_PCM_STREAM_CAPTURE] = r->start; + dmares = platform_get_resource(pdev, IORESOURCE_DMA, 1); + if (!dmares) + goto out2; + ctx->dmaids[SNDRV_PCM_STREAM_CAPTURE] = dmares->start; platform_set_drvdata(pdev, ctx); ret = snd_soc_register_dai(&pdev->dev, &au1xi2s_dai_driver); if (ret) - goto out1; + goto out2; return 0; +out2: + iounmap(ctx->mmio); out1: - release_mem_region(r->start, resource_size(r)); + release_mem_region(iores->start, resource_size(iores)); out0: kfree(ctx); return ret; diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c index 172eefd..0c6acd5 100644 --- a/sound/soc/au1x/psc-ac97.c +++ b/sound/soc/au1x/psc-ac97.c @@ -364,7 +364,7 @@ static const struct snd_soc_dai_driver au1xpsc_ac97_dai_template = { static int __devinit au1xpsc_ac97_drvprobe(struct platform_device *pdev) { int ret; - struct resource *r; + struct resource *iores, *dmares; unsigned long sel; struct au1xpsc_audio_data *wd; @@ -374,29 +374,30 @@ static int __devinit au1xpsc_ac97_drvprobe(struct platform_device *pdev) mutex_init(&wd->lock); - r = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!r) { + iores = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!iores) { ret = -ENODEV; goto out0; } ret = -EBUSY; - if (!request_mem_region(r->start, resource_size(r), pdev->name)) + if (!request_mem_region(iores->start, resource_size(iores), + pdev->name)) goto out0; - wd->mmio = ioremap(r->start, resource_size(r)); + wd->mmio = ioremap(iores->start, resource_size(iores)); if (!wd->mmio) goto out1; - r = platform_get_resource(pdev, IORESOURCE_DMA, 0); - if (!r) + dmares = platform_get_resource(pdev, IORESOURCE_DMA, 0); + if (!dmares) goto out2; - wd->dmaids[SNDRV_PCM_STREAM_PLAYBACK] = r->start; + wd->dmaids[SNDRV_PCM_STREAM_PLAYBACK] = dmares->start; - r = platform_get_resource(pdev, IORESOURCE_DMA, 1); - if (!r) + dmares = platform_get_resource(pdev, IORESOURCE_DMA, 1); + if (!dmares) goto out2; - wd->dmaids[SNDRV_PCM_STREAM_CAPTURE] = r->start; + wd->dmaids[SNDRV_PCM_STREAM_CAPTURE] = dmares->start; /* configuration: max dma trigger threshold, enable ac97 */ wd->cfg = PSC_AC97CFG_RT_FIFO8 | PSC_AC97CFG_TT_FIFO8 | @@ -428,7 +429,7 @@ static int __devinit au1xpsc_ac97_drvprobe(struct platform_device *pdev) out2: iounmap(wd->mmio); out1: - release_mem_region(r->start, resource_size(r)); + release_mem_region(iores->start, resource_size(iores)); out0: kfree(wd); return ret; diff --git a/sound/soc/au1x/psc-i2s.c b/sound/soc/au1x/psc-i2s.c index 7c5ae92..e03c5ce 100644 --- a/sound/soc/au1x/psc-i2s.c +++ b/sound/soc/au1x/psc-i2s.c @@ -290,7 +290,7 @@ static const struct snd_soc_dai_driver au1xpsc_i2s_dai_template = { static int __devinit au1xpsc_i2s_drvprobe(struct platform_device *pdev) { - struct resource *r; + struct resource *iores, *dmares; unsigned long sel; int ret; struct au1xpsc_audio_data *wd; @@ -299,29 +299,30 @@ static int __devinit au1xpsc_i2s_drvprobe(struct platform_device *pdev) if (!wd) return -ENOMEM; - r = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!r) { + iores = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!iores) { ret = -ENODEV; goto out0; } ret = -EBUSY; - if (!request_mem_region(r->start, resource_size(r), pdev->name)) + if (!request_mem_region(iores->start, resource_size(iores), + pdev->name)) goto out0; - wd->mmio = ioremap(r->start, resource_size(r)); + wd->mmio = ioremap(iores->start, resource_size(iores)); if (!wd->mmio) goto out1; - r = platform_get_resource(pdev, IORESOURCE_DMA, 0); - if (!r) + dmares = platform_get_resource(pdev, IORESOURCE_DMA, 0); + if (!dmares) goto out2; - wd->dmaids[SNDRV_PCM_STREAM_PLAYBACK] = r->start; + wd->dmaids[SNDRV_PCM_STREAM_PLAYBACK] = dmares->start; - r = platform_get_resource(pdev, IORESOURCE_DMA, 1); - if (!r) + dmares = platform_get_resource(pdev, IORESOURCE_DMA, 1); + if (!dmares) goto out2; - wd->dmaids[SNDRV_PCM_STREAM_CAPTURE] = r->start; + wd->dmaids[SNDRV_PCM_STREAM_CAPTURE] = dmares->start; /* preserve PSC clock source set up by platform (dev.platform_data * is already occupied by soc layer) @@ -355,7 +356,7 @@ static int __devinit au1xpsc_i2s_drvprobe(struct platform_device *pdev) out2: iounmap(wd->mmio); out1: - release_mem_region(r->start, resource_size(r)); + release_mem_region(iores->start, resource_size(iores)); out0: kfree(wd); return ret; diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index 401944c..76dc74d 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -617,7 +617,7 @@ static irqreturn_t mxs_saif_irq(int irq, void *dev_id) static int mxs_saif_probe(struct platform_device *pdev) { - struct resource *res; + struct resource *iores, *dmares; struct mxs_saif *saif; struct mxs_saif_platform_data *pdata; int ret = 0; @@ -655,35 +655,36 @@ static int mxs_saif_probe(struct platform_device *pdev) goto failed_clk; } - res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!res) { + iores = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!iores) { ret = -ENODEV; dev_err(&pdev->dev, "failed to get io resource: %d\n", ret); goto failed_get_resource; } - if (!request_mem_region(res->start, resource_size(res), "mxs-saif")) { + if (!request_mem_region(iores->start, resource_size(iores), + "mxs-saif")) { dev_err(&pdev->dev, "request_mem_region failed\n"); ret = -EBUSY; goto failed_get_resource; } - saif->base = ioremap(res->start, resource_size(res)); + saif->base = ioremap(iores->start, resource_size(iores)); if (!saif->base) { dev_err(&pdev->dev, "ioremap failed\n"); ret = -ENODEV; goto failed_ioremap; } - res = platform_get_resource(pdev, IORESOURCE_DMA, 0); - if (!res) { + dmares = platform_get_resource(pdev, IORESOURCE_DMA, 0); + if (!dmares) { ret = -ENODEV; dev_err(&pdev->dev, "failed to get dma resource: %d\n", ret); goto failed_ioremap; } - saif->dma_param.chan_num = res->start; + saif->dma_param.chan_num = dmares->start; saif->irq = platform_get_irq(pdev, 0); if (saif->irq < 0) { @@ -742,7 +743,7 @@ failed_get_irq2: failed_get_irq1: iounmap(saif->base); failed_ioremap: - release_mem_region(res->start, resource_size(res)); + release_mem_region(iores->start, resource_size(iores)); failed_get_resource: clk_put(saif->clk); failed_clk: -- cgit v1.1 From 0d8d293898ff0ea395840cdf2ac85fbd53c8d3ea Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Sat, 15 Oct 2011 11:46:02 +0800 Subject: ASoC: max98095: Convert codec->hw_write to snd_soc_write codec->hw_write is broken now, convert codec->hw_write to snd_soc_write. The hardware has 2 banks of registers sharing a section in I2C register space. The 1st bank is the primary one and is cached. The 2nd bank is for loading coefficients only and they do not need cache. These coefficients registers are therefore direct writes. Thus we set cache_bypass flag to deal with this before calling snd_soc_write. Signed-off-by: Axel Lin Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/max98095.c | 13 ++++++------- 1 file changed, 6 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/max98095.c b/sound/soc/codecs/max98095.c index 6982f74..26d7b08 100644 --- a/sound/soc/codecs/max98095.c +++ b/sound/soc/codecs/max98095.c @@ -617,14 +617,13 @@ static int max98095_volatile(struct snd_soc_codec *codec, unsigned int reg) static int max98095_hw_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int value) { - u8 data[2]; + int ret; - data[0] = reg; - data[1] = value; - if (codec->hw_write(codec->control_data, data, 2) == 2) - return 0; - else - return -EIO; + codec->cache_bypass = 1; + ret = snd_soc_write(codec, reg, value); + codec->cache_bypass = 0; + + return ret ? -EIO : 0; } /* -- cgit v1.1 From 5927f94700e860ae27ff24e7f3bc9e4f7b9922eb Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Wed, 26 Oct 2011 09:53:41 +0800 Subject: ASoC: wm8940: Properly set codec->dapm.bias_level Reported-by: Chris Paulson-Ellis Signed-off-by: Axel Lin Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/wm8940.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c index a4abfdf..dc5cb31 100644 --- a/sound/soc/codecs/wm8940.c +++ b/sound/soc/codecs/wm8940.c @@ -488,6 +488,8 @@ static int wm8940_set_bias_level(struct snd_soc_codec *codec, break; } + codec->dapm.bias_level = level; + return ret; } -- cgit v1.1