From cf4f7fc3e7336e2e946880890e60ed36178889ea Mon Sep 17 00:00:00 2001 From: Fabio Falzoi Date: Mon, 4 Aug 2014 17:08:07 +0200 Subject: ASoC: fsl-ssi: Support for SND_SOC_DAIFMT_CBM_CFS Add SND_SOC_DAIFMT_CBM_CFS support for Freescale architecture. Successfully tested on i.MX 6Quad Wandboard and UDOO boards connected to the pcm1792a codec. In CBM_CFS mode, when using a sample size of 16 bits, we cannot use CCSR_SSI_SCR_I2S_MODE_MASTER since we get a frame sync every 16 bits. Signed-off-by: Michael Trimarchi Signed-off-by: Fabio Falzoi Tested-by: Angelo Adamo Acked-by: Timur Tabi Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_ssi.c | 33 ++++++++++++++++++++++++++++----- 1 file changed, 28 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 87eb577..2fc3e66 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -259,6 +259,11 @@ static bool fsl_ssi_is_i2s_master(struct fsl_ssi_private *ssi_private) SND_SOC_DAIFMT_CBS_CFS; } +static bool fsl_ssi_is_i2s_cbm_cfs(struct fsl_ssi_private *ssi_private) +{ + return (ssi_private->dai_fmt & SND_SOC_DAIFMT_MASTER_MASK) == + SND_SOC_DAIFMT_CBM_CFS; +} /** * fsl_ssi_isr: SSI interrupt handler * @@ -705,6 +710,23 @@ static int fsl_ssi_hw_params(struct snd_pcm_substream *substream, } } + if (!fsl_ssi_is_ac97(ssi_private)) { + u8 i2smode; + /* + * Switch to normal net mode in order to have a frame sync + * signal every 32 bits instead of 16 bits + */ + if (fsl_ssi_is_i2s_cbm_cfs(ssi_private) && sample_size == 16) + i2smode = CCSR_SSI_SCR_I2S_MODE_NORMAL | + CCSR_SSI_SCR_NET; + else + i2smode = ssi_private->i2s_mode; + + regmap_update_bits(regs, CCSR_SSI_SCR, + CCSR_SSI_SCR_NET | CCSR_SSI_SCR_I2S_MODE_MASK, + channels == 1 ? 0 : i2smode); + } + /* * FIXME: The documentation says that SxCCR[WL] should not be * modified while the SSI is enabled. The only time this can @@ -724,11 +746,6 @@ static int fsl_ssi_hw_params(struct snd_pcm_substream *substream, regmap_update_bits(regs, CCSR_SSI_SRCCR, CCSR_SSI_SxCCR_WL_MASK, wl); - if (!fsl_ssi_is_ac97(ssi_private)) - regmap_update_bits(regs, CCSR_SSI_SCR, - CCSR_SSI_SCR_NET | CCSR_SSI_SCR_I2S_MODE_MASK, - channels == 1 ? 0 : ssi_private->i2s_mode); - return 0; } @@ -780,6 +797,7 @@ static int _fsl_ssi_set_dai_fmt(struct fsl_ssi_private *ssi_private, switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFS: case SND_SOC_DAIFMT_CBS_CFS: ssi_private->i2s_mode |= CCSR_SSI_SCR_I2S_MODE_MASTER; regmap_update_bits(regs, CCSR_SSI_STCCR, @@ -853,6 +871,11 @@ static int _fsl_ssi_set_dai_fmt(struct fsl_ssi_private *ssi_private, case SND_SOC_DAIFMT_CBM_CFM: scr &= ~CCSR_SSI_SCR_SYS_CLK_EN; break; + case SND_SOC_DAIFMT_CBM_CFS: + strcr &= ~CCSR_SSI_STCR_TXDIR; + strcr |= CCSR_SSI_STCR_TFDIR; + scr &= ~CCSR_SSI_SCR_SYS_CLK_EN; + break; default: return -EINVAL; } -- cgit v1.1 From d177143c3670aa57ee08c73880beb55ee9d8ab7c Mon Sep 17 00:00:00 2001 From: Shengjiu Wang Date: Fri, 8 Aug 2014 14:47:21 +0800 Subject: ASoC: fsl_esai: refine esai for TDM support Original driver didn't store the number of slots, just fix the slot number to 2, use this default number to calculate bclk and pins for TX/RX. In this patch, add one parameter for slots, and update the calculation of bclk and pins of TX/RX. Then driver will be compatible with slots > 2 in TDM mode. Signed-off-by: Shengjiu Wang Acked-by: Nicolin Chen Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_esai.c | 14 +++++++++++--- sound/soc/fsl/fsl_esai.h | 8 ++++---- 2 files changed, 15 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index 72d154e..f252370 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -38,6 +38,7 @@ * @fsysclk: system clock source to derive HCK, SCK and FS * @fifo_depth: depth of tx/rx FIFO * @slot_width: width of each DAI slot + * @slots: number of slots * @hck_rate: clock rate of desired HCKx clock * @sck_rate: clock rate of desired SCKx clock * @hck_dir: the direction of HCKx pads @@ -56,6 +57,7 @@ struct fsl_esai { struct clk *fsysclk; u32 fifo_depth; u32 slot_width; + u32 slots; u32 hck_rate[2]; u32 sck_rate[2]; bool hck_dir[2]; @@ -363,6 +365,7 @@ static int fsl_esai_set_dai_tdm_slot(struct snd_soc_dai *dai, u32 tx_mask, ESAI_xSMB_xS_MASK, ESAI_xSMB_xS(rx_mask)); esai_priv->slot_width = slot_width; + esai_priv->slots = slots; return 0; } @@ -510,10 +513,11 @@ static int fsl_esai_hw_params(struct snd_pcm_substream *substream, bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; u32 width = snd_pcm_format_width(params_format(params)); u32 channels = params_channels(params); + u32 pins = DIV_ROUND_UP(channels, esai_priv->slots); u32 bclk, mask, val; int ret; - bclk = params_rate(params) * esai_priv->slot_width * 2; + bclk = params_rate(params) * esai_priv->slot_width * esai_priv->slots; ret = fsl_esai_set_bclk(dai, tx, bclk); if (ret) @@ -530,7 +534,7 @@ static int fsl_esai_hw_params(struct snd_pcm_substream *substream, mask = ESAI_xFCR_xFR_MASK | ESAI_xFCR_xWA_MASK | ESAI_xFCR_xFWM_MASK | (tx ? ESAI_xFCR_TE_MASK | ESAI_xFCR_TIEN : ESAI_xFCR_RE_MASK); val = ESAI_xFCR_xWA(width) | ESAI_xFCR_xFWM(esai_priv->fifo_depth) | - (tx ? ESAI_xFCR_TE(channels) | ESAI_xFCR_TIEN : ESAI_xFCR_RE(channels)); + (tx ? ESAI_xFCR_TE(pins) | ESAI_xFCR_TIEN : ESAI_xFCR_RE(pins)); regmap_update_bits(esai_priv->regmap, REG_ESAI_xFCR(tx), mask, val); @@ -565,6 +569,7 @@ static int fsl_esai_trigger(struct snd_pcm_substream *substream, int cmd, struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai); bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; u8 i, channels = substream->runtime->channels; + u32 pins = DIV_ROUND_UP(channels, esai_priv->slots); switch (cmd) { case SNDRV_PCM_TRIGGER_START: @@ -579,7 +584,7 @@ static int fsl_esai_trigger(struct snd_pcm_substream *substream, int cmd, regmap_update_bits(esai_priv->regmap, REG_ESAI_xCR(tx), tx ? ESAI_xCR_TE_MASK : ESAI_xCR_RE_MASK, - tx ? ESAI_xCR_TE(channels) : ESAI_xCR_RE(channels)); + tx ? ESAI_xCR_TE(pins) : ESAI_xCR_RE(pins)); break; case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_STOP: @@ -783,6 +788,9 @@ static int fsl_esai_probe(struct platform_device *pdev) /* Set a default slot size */ esai_priv->slot_width = 32; + /* Set a default slot number */ + esai_priv->slots = 2; + /* Set a default master/slave state */ esai_priv->slave_mode = true; diff --git a/sound/soc/fsl/fsl_esai.h b/sound/soc/fsl/fsl_esai.h index 75e1403..91a550f 100644 --- a/sound/soc/fsl/fsl_esai.h +++ b/sound/soc/fsl/fsl_esai.h @@ -130,8 +130,8 @@ #define ESAI_xFCR_RE_WIDTH 4 #define ESAI_xFCR_TE_MASK (((1 << ESAI_xFCR_TE_WIDTH) - 1) << ESAI_xFCR_xE_SHIFT) #define ESAI_xFCR_RE_MASK (((1 << ESAI_xFCR_RE_WIDTH) - 1) << ESAI_xFCR_xE_SHIFT) -#define ESAI_xFCR_TE(x) ((ESAI_xFCR_TE_MASK >> (ESAI_xFCR_TE_WIDTH - ((x + 1) >> 1))) & ESAI_xFCR_TE_MASK) -#define ESAI_xFCR_RE(x) ((ESAI_xFCR_RE_MASK >> (ESAI_xFCR_RE_WIDTH - ((x + 1) >> 1))) & ESAI_xFCR_RE_MASK) +#define ESAI_xFCR_TE(x) ((ESAI_xFCR_TE_MASK >> (ESAI_xFCR_TE_WIDTH - x)) & ESAI_xFCR_TE_MASK) +#define ESAI_xFCR_RE(x) ((ESAI_xFCR_RE_MASK >> (ESAI_xFCR_RE_WIDTH - x)) & ESAI_xFCR_RE_MASK) #define ESAI_xFCR_xFR_SHIFT 1 #define ESAI_xFCR_xFR_MASK (1 << ESAI_xFCR_xFR_SHIFT) #define ESAI_xFCR_xFR (1 << ESAI_xFCR_xFR_SHIFT) @@ -272,8 +272,8 @@ #define ESAI_xCR_RE_WIDTH 4 #define ESAI_xCR_TE_MASK (((1 << ESAI_xCR_TE_WIDTH) - 1) << ESAI_xCR_xE_SHIFT) #define ESAI_xCR_RE_MASK (((1 << ESAI_xCR_RE_WIDTH) - 1) << ESAI_xCR_xE_SHIFT) -#define ESAI_xCR_TE(x) ((ESAI_xCR_TE_MASK >> (ESAI_xCR_TE_WIDTH - ((x + 1) >> 1))) & ESAI_xCR_TE_MASK) -#define ESAI_xCR_RE(x) ((ESAI_xCR_RE_MASK >> (ESAI_xCR_RE_WIDTH - ((x + 1) >> 1))) & ESAI_xCR_RE_MASK) +#define ESAI_xCR_TE(x) ((ESAI_xCR_TE_MASK >> (ESAI_xCR_TE_WIDTH - x)) & ESAI_xCR_TE_MASK) +#define ESAI_xCR_RE(x) ((ESAI_xCR_RE_MASK >> (ESAI_xCR_RE_WIDTH - x)) & ESAI_xCR_RE_MASK) /* * Transmit Clock Control Register -- REG_ESAI_TCCR 0xD8 -- cgit v1.1 From eef5bb2445ca49911c93c08ed0fb2ea7363ea945 Mon Sep 17 00:00:00 2001 From: Brian Austin Date: Mon, 4 Aug 2014 15:11:16 -0500 Subject: ASoC: cs35l32: Add support for CS35L32 Boosted Amplifier This patch adds support for the Cirrus Logic CS35L32 Boosted Amplifier I2S output provides monitor data to the SOC/CODEC/DSP for speaker protection/enhancement algorithms Signed-off-by: Brian Austin Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cs35l32.c | 647 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/cs35l32.h | 93 +++++++ 4 files changed, 747 insertions(+) create mode 100644 sound/soc/codecs/cs35l32.c create mode 100644 sound/soc/codecs/cs35l32.h (limited to 'sound') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 8838838e..77e5383 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -43,6 +43,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_ALC5623 if I2C select SND_SOC_ALC5632 if I2C select SND_SOC_CQ0093VC if MFD_DAVINCI_VOICECODEC + select SND_SOC_CS35L32 if I2C select SND_SOC_CS42L51_I2C if I2C select SND_SOC_CS42L52 if I2C && INPUT select SND_SOC_CS42L56 if I2C && INPUT @@ -323,6 +324,10 @@ config SND_SOC_ALC5632 config SND_SOC_CQ0093VC tristate +config SND_SOC_CS35L32 + tristate "Cirrus Logic CS35L32 CODEC" + depends on I2C + config SND_SOC_CS42L51 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 20afe0f..1dacefb 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -32,6 +32,7 @@ snd-soc-ak4671-objs := ak4671.o snd-soc-ak5386-objs := ak5386.o snd-soc-arizona-objs := arizona.o snd-soc-cq93vc-objs := cq93vc.o +snd-soc-cs35l32-objs := cs35l32.o snd-soc-cs42l51-objs := cs42l51.o snd-soc-cs42l51-i2c-objs := cs42l51-i2c.o snd-soc-cs42l52-objs := cs42l52.o @@ -203,6 +204,7 @@ obj-$(CONFIG_SND_SOC_ALC5623) += snd-soc-alc5623.o obj-$(CONFIG_SND_SOC_ALC5632) += snd-soc-alc5632.o obj-$(CONFIG_SND_SOC_ARIZONA) += snd-soc-arizona.o obj-$(CONFIG_SND_SOC_CQ0093VC) += snd-soc-cq93vc.o +obj-$(CONFIG_SND_SOC_CS35L32) += snd-soc-cs35l32.o obj-$(CONFIG_SND_SOC_CS42L51) += snd-soc-cs42l51.o obj-$(CONFIG_SND_SOC_CS42L51_I2C) += snd-soc-cs42l51-i2c.o obj-$(CONFIG_SND_SOC_CS42L52) += snd-soc-cs42l52.o diff --git a/sound/soc/codecs/cs35l32.c b/sound/soc/codecs/cs35l32.c new file mode 100644 index 0000000..90565d5 --- /dev/null +++ b/sound/soc/codecs/cs35l32.c @@ -0,0 +1,647 @@ +/* + * cs35l32.c -- CS35L32 ALSA SoC audio driver + * + * Copyright 2014 CirrusLogic, Inc. + * + * Author: Brian Austin + * + * 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 +#include +#include +#include + +#include "cs35l32.h" + +#define CS35L32_NUM_SUPPLIES 2 +static const char *const cs35l32_supply_names[CS35L32_NUM_SUPPLIES] = { + "VA", + "VP", +}; + +struct cs35l32_private { + struct regmap *regmap; + struct snd_soc_codec *codec; + struct regulator_bulk_data supplies[CS35L32_NUM_SUPPLIES]; + struct cs35l32_platform_data pdata; + struct gpio_desc *reset_gpio; +}; + +static const struct reg_default cs35l32_reg_defaults[] = { + + { 0x06, 0x04 }, /* Power Ctl 1 */ + { 0x07, 0xE8 }, /* Power Ctl 2 */ + { 0x08, 0x40 }, /* Clock Ctl */ + { 0x09, 0x20 }, /* Low Battery Threshold */ + { 0x0A, 0x00 }, /* Voltage Monitor [RO] */ + { 0x0B, 0x40 }, /* Conv Peak Curr Protection CTL */ + { 0x0C, 0x07 }, /* IMON Scaling */ + { 0x0D, 0x03 }, /* Audio/LED Pwr Manager */ + { 0x0F, 0x20 }, /* Serial Port Control */ + { 0x10, 0x14 }, /* Class D Amp CTL */ + { 0x11, 0x00 }, /* Protection Release CTL */ + { 0x12, 0xFF }, /* Interrupt Mask 1 */ + { 0x13, 0xFF }, /* Interrupt Mask 2 */ + { 0x14, 0xFF }, /* Interrupt Mask 3 */ + { 0x19, 0x00 }, /* LED Flash Mode Current */ + { 0x1A, 0x00 }, /* LED Movie Mode Current */ + { 0x1B, 0x20 }, /* LED Flash Timer */ + { 0x1C, 0x00 }, /* LED Flash Inhibit Current */ +}; + +static bool cs35l32_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS35L32_DEVID_AB: + case CS35L32_DEVID_CD: + case CS35L32_DEVID_E: + case CS35L32_FAB_ID: + case CS35L32_REV_ID: + case CS35L32_PWRCTL1: + case CS35L32_PWRCTL2: + case CS35L32_CLK_CTL: + case CS35L32_BATT_THRESHOLD: + case CS35L32_VMON: + case CS35L32_BST_CPCP_CTL: + case CS35L32_IMON_SCALING: + case CS35L32_AUDIO_LED_MNGR: + case CS35L32_ADSP_CTL: + case CS35L32_CLASSD_CTL: + case CS35L32_PROTECT_CTL: + case CS35L32_INT_MASK_1: + case CS35L32_INT_MASK_2: + case CS35L32_INT_MASK_3: + case CS35L32_INT_STATUS_1: + case CS35L32_INT_STATUS_2: + case CS35L32_INT_STATUS_3: + case CS35L32_LED_STATUS: + case CS35L32_FLASH_MODE: + case CS35L32_MOVIE_MODE: + case CS35L32_FLASH_TIMER: + case CS35L32_FLASH_INHIBIT: + return true; + default: + return false; + } +} + +static bool cs35l32_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS35L32_DEVID_AB: + case CS35L32_DEVID_CD: + case CS35L32_DEVID_E: + case CS35L32_FAB_ID: + case CS35L32_REV_ID: + case CS35L32_INT_STATUS_1: + case CS35L32_INT_STATUS_2: + case CS35L32_INT_STATUS_3: + case CS35L32_LED_STATUS: + return 1; + default: + return 0; + } +} + +static bool cs35l32_precious_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS35L32_INT_STATUS_1: + case CS35L32_INT_STATUS_2: + case CS35L32_INT_STATUS_3: + case CS35L32_LED_STATUS: + return 1; + default: + return 0; + } +} + +static DECLARE_TLV_DB_SCALE(classd_ctl_tlv, 900, 300, 0); + +static const struct snd_kcontrol_new imon_ctl = + SOC_DAPM_SINGLE("Switch", CS35L32_PWRCTL2, 6, 1, 1); + +static const struct snd_kcontrol_new vmon_ctl = + SOC_DAPM_SINGLE("Switch", CS35L32_PWRCTL2, 7, 1, 1); + +static const struct snd_kcontrol_new vpmon_ctl = + SOC_DAPM_SINGLE("Switch", CS35L32_PWRCTL2, 5, 1, 1); + +static const struct snd_kcontrol_new cs35l32_snd_controls[] = { + SOC_SINGLE_TLV("Speaker Volume", CS35L32_CLASSD_CTL, + 3, 0x04, 1, classd_ctl_tlv), + SOC_SINGLE("Zero Cross Switch", CS35L32_CLASSD_CTL, 2, 1, 0), + SOC_SINGLE("Gain Manager Switch", CS35L32_AUDIO_LED_MNGR, 3, 1, 0), +}; + +static const struct snd_soc_dapm_widget cs35l32_dapm_widgets[] = { + + SND_SOC_DAPM_SUPPLY("BOOST", CS35L32_PWRCTL1, 2, 1, NULL, 0), + SND_SOC_DAPM_OUT_DRV("Speaker", CS35L32_PWRCTL1, 7, 1, NULL, 0), + + SND_SOC_DAPM_AIF_OUT("SDOUT", NULL, 0, CS35L32_PWRCTL2, 3, 1), + + SND_SOC_DAPM_INPUT("VP"), + SND_SOC_DAPM_INPUT("ISENSE"), + SND_SOC_DAPM_INPUT("VSENSE"), + + SND_SOC_DAPM_SWITCH("VMON ADC", CS35L32_PWRCTL2, 7, 1, &vmon_ctl), + SND_SOC_DAPM_SWITCH("IMON ADC", CS35L32_PWRCTL2, 6, 1, &imon_ctl), + SND_SOC_DAPM_SWITCH("VPMON ADC", CS35L32_PWRCTL2, 5, 1, &vpmon_ctl), +}; + +static const struct snd_soc_dapm_route cs35l32_audio_map[] = { + + {"Speaker", NULL, "BOOST"}, + + {"VMON ADC", NULL, "VSENSE"}, + {"IMON ADC", NULL, "ISENSE"}, + {"VPMON ADC", NULL, "VP"}, + + {"SDOUT", "Switch", "VMON ADC"}, + {"SDOUT", "Switch", "IMON ADC"}, + {"SDOUT", "Switch", "VPMON ADC"}, + + {"Capture", NULL, "SDOUT"}, +}; + +static int cs35l32_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + snd_soc_update_bits(codec, CS35L32_ADSP_CTL, + CS35L32_ADSP_MASTER_MASK, + CS35L32_ADSP_MASTER_MASK); + break; + case SND_SOC_DAIFMT_CBS_CFS: + snd_soc_update_bits(codec, CS35L32_ADSP_CTL, + CS35L32_ADSP_MASTER_MASK, 0); + break; + default: + return -EINVAL; + } + + return 0; +} + +static int cs35l32_set_tristate(struct snd_soc_dai *dai, int tristate) +{ + struct snd_soc_codec *codec = dai->codec; + + return snd_soc_update_bits(codec, CS35L32_PWRCTL2, + CS35L32_SDOUT_3ST, tristate << 3); +} + +static const struct snd_soc_dai_ops cs35l32_ops = { + .set_fmt = cs35l32_set_dai_fmt, + .set_tristate = cs35l32_set_tristate, +}; + +static struct snd_soc_dai_driver cs35l32_dai[] = { + { + .name = "cs35l32-monitor", + .id = 0, + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 2, + .rates = CS35L32_RATES, + .formats = CS35L32_FORMATS, + }, + .ops = &cs35l32_ops, + .symmetric_rates = 1, + } +}; + +static int cs35l32_codec_set_sysclk(struct snd_soc_codec *codec, + int clk_id, int source, unsigned int freq, int dir) +{ + + switch (freq) { + case 6000000: + snd_soc_update_bits(codec, CS35L32_CLK_CTL, + CS35L32_MCLK_DIV2_MASK, 0); + snd_soc_update_bits(codec, CS35L32_CLK_CTL, + CS35L32_MCLK_RATIO_MASK, + CS35L32_MCLK_RATIO); + break; + case 12000000: + snd_soc_update_bits(codec, CS35L32_CLK_CTL, + CS35L32_MCLK_DIV2_MASK, + CS35L32_MCLK_DIV2_MASK); + snd_soc_update_bits(codec, CS35L32_CLK_CTL, + CS35L32_MCLK_RATIO_MASK, + CS35L32_MCLK_RATIO); + break; + case 6144000: + snd_soc_update_bits(codec, CS35L32_CLK_CTL, + CS35L32_MCLK_DIV2_MASK, 0); + snd_soc_update_bits(codec, CS35L32_CLK_CTL, + CS35L32_MCLK_RATIO_MASK, 0); + break; + case 12288000: + snd_soc_update_bits(codec, CS35L32_CLK_CTL, + CS35L32_MCLK_DIV2_MASK, + CS35L32_MCLK_DIV2_MASK); + snd_soc_update_bits(codec, CS35L32_CLK_CTL, + CS35L32_MCLK_RATIO_MASK, 0); + break; + default: + return -EINVAL; + } + + return 0; +} + +static struct snd_soc_codec_driver soc_codec_dev_cs35l32 = { + .set_sysclk = cs35l32_codec_set_sysclk, + + .dapm_widgets = cs35l32_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cs35l32_dapm_widgets), + .dapm_routes = cs35l32_audio_map, + .num_dapm_routes = ARRAY_SIZE(cs35l32_audio_map), + + .controls = cs35l32_snd_controls, + .num_controls = ARRAY_SIZE(cs35l32_snd_controls), +}; + +/* Current and threshold powerup sequence Pg37 in datasheet */ +static const struct reg_default cs35l32_monitor_patch[] = { + + { 0x00, 0x99 }, + { 0x48, 0x17 }, + { 0x49, 0x56 }, + { 0x43, 0x01 }, + { 0x3B, 0x62 }, + { 0x3C, 0x80 }, + { 0x00, 0x00 }, +}; + +static struct regmap_config cs35l32_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = CS35L32_MAX_REGISTER, + .reg_defaults = cs35l32_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(cs35l32_reg_defaults), + .volatile_reg = cs35l32_volatile_register, + .readable_reg = cs35l32_readable_register, + .precious_reg = cs35l32_precious_register, + .cache_type = REGCACHE_RBTREE, +}; + +static int cs35l32_handle_of_data(struct i2c_client *i2c_client, + struct cs35l32_platform_data *pdata) +{ + struct device_node *np = i2c_client->dev.of_node; + unsigned int val; + + if (of_property_read_u32(np, "cirrus,sdout-share", &val) >= 0) + pdata->sdout_share = val; + + of_property_read_u32(np, "cirrus,boost-manager", &val); + switch (val) { + case CS35L32_BOOST_MGR_AUTO: + case CS35L32_BOOST_MGR_AUTO_AUDIO: + case CS35L32_BOOST_MGR_BYPASS: + case CS35L32_BOOST_MGR_FIXED: + pdata->boost_mng = val; + break; + default: + dev_err(&i2c_client->dev, + "Wrong cirrus,boost-manager DT value %d\n", val); + pdata->boost_mng = CS35L32_BOOST_MGR_BYPASS; + } + + of_property_read_u32(np, "cirrus,sdout-datacfg", &val); + switch (val) { + case CS35L32_DATA_CFG_LR_VP: + case CS35L32_DATA_CFG_LR_STAT: + case CS35L32_DATA_CFG_LR: + case CS35L32_DATA_CFG_LR_VPSTAT: + pdata->sdout_datacfg = val; + break; + default: + dev_err(&i2c_client->dev, + "Wrong cirrus,sdout-datacfg DT value %d\n", val); + pdata->sdout_datacfg = CS35L32_DATA_CFG_LR; + } + + of_property_read_u32(np, "cirrus,battery-threshold", &val); + switch (val) { + case CS35L32_BATT_THRESH_3_1V: + case CS35L32_BATT_THRESH_3_2V: + case CS35L32_BATT_THRESH_3_3V: + case CS35L32_BATT_THRESH_3_4V: + pdata->batt_thresh = val; + break; + default: + dev_err(&i2c_client->dev, + "Wrong cirrus,battery-threshold DT value %d\n", val); + pdata->batt_thresh = CS35L32_BATT_THRESH_3_3V; + } + + of_property_read_u32(np, "cirrus,battery-recovery", &val); + switch (val) { + case CS35L32_BATT_RECOV_3_1V: + case CS35L32_BATT_RECOV_3_2V: + case CS35L32_BATT_RECOV_3_3V: + case CS35L32_BATT_RECOV_3_4V: + case CS35L32_BATT_RECOV_3_5V: + case CS35L32_BATT_RECOV_3_6V: + pdata->batt_recov = val; + break; + default: + dev_err(&i2c_client->dev, + "Wrong cirrus,battery-recovery DT value %d\n", val); + pdata->batt_recov = CS35L32_BATT_RECOV_3_4V; + } + + return 0; +} + +static int cs35l32_i2c_probe(struct i2c_client *i2c_client, + const struct i2c_device_id *id) +{ + struct cs35l32_private *cs35l32; + struct cs35l32_platform_data *pdata = + dev_get_platdata(&i2c_client->dev); + int ret, i; + unsigned int devid = 0; + unsigned int reg; + + + cs35l32 = devm_kzalloc(&i2c_client->dev, sizeof(struct cs35l32_private), + GFP_KERNEL); + if (!cs35l32) { + dev_err(&i2c_client->dev, "could not allocate codec\n"); + return -ENOMEM; + } + + i2c_set_clientdata(i2c_client, cs35l32); + + cs35l32->regmap = devm_regmap_init_i2c(i2c_client, &cs35l32_regmap); + if (IS_ERR(cs35l32->regmap)) { + ret = PTR_ERR(cs35l32->regmap); + dev_err(&i2c_client->dev, "regmap_init() failed: %d\n", ret); + return ret; + } + + if (pdata) { + cs35l32->pdata = *pdata; + } else { + pdata = devm_kzalloc(&i2c_client->dev, + sizeof(struct cs35l32_platform_data), + GFP_KERNEL); + if (!pdata) { + dev_err(&i2c_client->dev, "could not allocate pdata\n"); + return -ENOMEM; + } + if (i2c_client->dev.of_node) { + ret = cs35l32_handle_of_data(i2c_client, + &cs35l32->pdata); + if (ret != 0) + return ret; + } + } + + for (i = 0; i < ARRAY_SIZE(cs35l32->supplies); i++) + cs35l32->supplies[i].supply = cs35l32_supply_names[i]; + + ret = devm_regulator_bulk_get(&i2c_client->dev, + ARRAY_SIZE(cs35l32->supplies), + cs35l32->supplies); + if (ret != 0) { + dev_err(&i2c_client->dev, + "Failed to request supplies: %d\n", ret); + return ret; + } + + ret = regulator_bulk_enable(ARRAY_SIZE(cs35l32->supplies), + cs35l32->supplies); + if (ret != 0) { + dev_err(&i2c_client->dev, + "Failed to enable supplies: %d\n", ret); + return ret; + } + + /* Reset the Device */ + cs35l32->reset_gpio = devm_gpiod_get(&i2c_client->dev, + "reset-gpios"); + if (IS_ERR(cs35l32->reset_gpio)) { + ret = PTR_ERR(cs35l32->reset_gpio); + if (ret != -ENOENT && ret != -ENOSYS) + return ret; + + cs35l32->reset_gpio = NULL; + } else { + ret = gpiod_direction_output(cs35l32->reset_gpio, 0); + if (ret) + return ret; + gpiod_set_value_cansleep(cs35l32->reset_gpio, 1); + } + + /* initialize codec */ + ret = regmap_read(cs35l32->regmap, CS35L32_DEVID_AB, ®); + devid = (reg & 0xFF) << 12; + + ret = regmap_read(cs35l32->regmap, CS35L32_DEVID_CD, ®); + devid |= (reg & 0xFF) << 4; + + ret = regmap_read(cs35l32->regmap, CS35L32_DEVID_E, ®); + devid |= (reg & 0xF0) >> 4; + + if (devid != CS35L32_CHIP_ID) { + ret = -ENODEV; + dev_err(&i2c_client->dev, + "CS35L32 Device ID (%X). Expected %X\n", + devid, CS35L32_CHIP_ID); + return ret; + } + + ret = regmap_read(cs35l32->regmap, CS35L32_REV_ID, ®); + if (ret < 0) { + dev_err(&i2c_client->dev, "Get Revision ID failed\n"); + return ret; + } + + ret = regmap_register_patch(cs35l32->regmap, cs35l32_monitor_patch, + ARRAY_SIZE(cs35l32_monitor_patch)); + if (ret < 0) { + dev_err(&i2c_client->dev, "Failed to apply errata patch\n"); + return ret; + } + + dev_info(&i2c_client->dev, + "Cirrus Logic CS35L32, Revision: %02X\n", reg & 0xFF); + + /* Setup VBOOST Management */ + if (cs35l32->pdata.boost_mng) + regmap_update_bits(cs35l32->regmap, CS35L32_AUDIO_LED_MNGR, + CS35L32_BOOST_MASK, + cs35l32->pdata.boost_mng); + + /* Setup ADSP Format Config */ + if (cs35l32->pdata.sdout_share) + regmap_update_bits(cs35l32->regmap, CS35L32_ADSP_CTL, + CS35L32_ADSP_SHARE_MASK, + cs35l32->pdata.sdout_share << 3); + + /* Setup ADSP Data Configuration */ + if (cs35l32->pdata.sdout_datacfg) + regmap_update_bits(cs35l32->regmap, CS35L32_ADSP_CTL, + CS35L32_ADSP_DATACFG_MASK, + cs35l32->pdata.sdout_datacfg << 4); + + /* Setup Low Battery Recovery */ + if (cs35l32->pdata.batt_recov) + regmap_update_bits(cs35l32->regmap, CS35L32_BATT_THRESHOLD, + CS35L32_BATT_REC_MASK, + cs35l32->pdata.batt_recov << 1); + + /* Setup Low Battery Threshold */ + if (cs35l32->pdata.batt_thresh) + regmap_update_bits(cs35l32->regmap, CS35L32_BATT_THRESHOLD, + CS35L32_BATT_THRESH_MASK, + cs35l32->pdata.batt_thresh << 4); + + /* Power down the AMP */ + regmap_update_bits(cs35l32->regmap, CS35L32_PWRCTL1, CS35L32_PDN_AMP, + CS35L32_PDN_AMP); + + /* Clear MCLK Error Bit since we don't have the clock yet */ + ret = regmap_read(cs35l32->regmap, CS35L32_INT_STATUS_1, ®); + + ret = snd_soc_register_codec(&i2c_client->dev, + &soc_codec_dev_cs35l32, cs35l32_dai, + ARRAY_SIZE(cs35l32_dai)); + if (ret < 0) + goto err_disable; + + return 0; + +err_disable: + regulator_bulk_disable(ARRAY_SIZE(cs35l32->supplies), + cs35l32->supplies); +} + +static int cs35l32_i2c_remove(struct i2c_client *i2c_client) +{ + struct cs35l32_private *cs35l32 = i2c_get_clientdata(i2c_client); + + snd_soc_unregister_codec(&i2c_client->dev); + + /* Hold down reset */ + if (cs35l32->reset_gpio) + gpiod_set_value_cansleep(cs35l32->reset_gpio, 0); + + regulator_bulk_free(ARRAY_SIZE(cs35l32->supplies), cs35l32->supplies); + + return 0; +} + +#ifdef CONFIG_PM_RUNTIME +static int cs35l32_runtime_suspend(struct device *dev) +{ + struct cs35l32_private *cs35l32 = dev_get_drvdata(dev); + + regcache_cache_only(cs35l32->regmap, true); + regcache_mark_dirty(cs35l32->regmap); + + /* Hold down reset */ + if (cs35l32->reset_gpio) + gpiod_set_value_cansleep(cs35l32->reset_gpio, 0); + + /* remove power */ + regulator_bulk_disable(ARRAY_SIZE(cs35l32->supplies), + cs35l32->supplies); + + return 0; +} + +static int cs35l32_runtime_resume(struct device *dev) +{ + struct cs35l32_private *cs35l32 = dev_get_drvdata(dev); + int ret; + + /* Enable power */ + ret = regulator_bulk_enable(ARRAY_SIZE(cs35l32->supplies), + cs35l32->supplies); + if (ret != 0) { + dev_err(dev, "Failed to enable supplies: %d\n", + ret); + return ret; + } + + if (cs35l32->reset_gpio) + gpiod_set_value_cansleep(cs35l32->reset_gpio, 1); + + regcache_cache_only(cs35l32->regmap, false); + regcache_sync(cs35l32->regmap); + + return 0; +} +#endif + +static const struct dev_pm_ops cs35l32_runtime_pm = { + SET_RUNTIME_PM_OPS(cs35l32_runtime_suspend, cs35l32_runtime_resume, + NULL) +}; + +static const struct of_device_id cs35l32_of_match[] = { + { .compatible = "cirrus,cs35l32", }, + {}, +}; +MODULE_DEVICE_TABLE(of, cs35l32_of_match); + + +static const struct i2c_device_id cs35l32_id[] = { + {"cs35l32", 0}, + {} +}; + +MODULE_DEVICE_TABLE(i2c, cs35l32_id); + +static struct i2c_driver cs35l32_i2c_driver = { + .driver = { + .name = "cs35l32", + .owner = THIS_MODULE, + .pm = &cs35l32_runtime_pm, + .of_match_table = cs35l32_of_match, + }, + .id_table = cs35l32_id, + .probe = cs35l32_i2c_probe, + .remove = cs35l32_i2c_remove, +}; + +module_i2c_driver(cs35l32_i2c_driver); + +MODULE_DESCRIPTION("ASoC CS35L32 driver"); +MODULE_AUTHOR("Brian Austin, Cirrus Logic Inc, "); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs35l32.h b/sound/soc/codecs/cs35l32.h new file mode 100644 index 0000000..31ab804 --- /dev/null +++ b/sound/soc/codecs/cs35l32.h @@ -0,0 +1,93 @@ +/* + * cs35l32.h -- CS35L32 ALSA SoC audio driver + * + * Copyright 2014 CirrusLogic, Inc. + * + * Author: Brian Austin + * + * 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 __CS35L32_H__ +#define __CS35L32_H__ + +struct cs35l32_platform_data { + /* Low Battery Threshold */ + unsigned int batt_thresh; + /* Low Battery Recovery */ + unsigned int batt_recov; + /* LED Current Management*/ + unsigned int led_mng; + /* Audio Gain w/ LED */ + unsigned int audiogain_mng; + /* Boost Management */ + unsigned int boost_mng; + /* Data CFG for DUAL device */ + unsigned int sdout_datacfg; + /* SDOUT Sharing */ + unsigned int sdout_share; +}; + +#define CS35L32_CHIP_ID 0x00035A32 +#define CS35L32_DEVID_AB 0x01 /* Device ID A & B [RO] */ +#define CS35L32_DEVID_CD 0x02 /* Device ID C & D [RO] */ +#define CS35L32_DEVID_E 0x03 /* Device ID E [RO] */ +#define CS35L32_FAB_ID 0x04 /* Fab ID [RO] */ +#define CS35L32_REV_ID 0x05 /* Revision ID [RO] */ +#define CS35L32_PWRCTL1 0x06 /* Power Ctl 1 */ +#define CS35L32_PWRCTL2 0x07 /* Power Ctl 2 */ +#define CS35L32_CLK_CTL 0x08 /* Clock Ctl */ +#define CS35L32_BATT_THRESHOLD 0x09 /* Low Battery Threshold */ +#define CS35L32_VMON 0x0A /* Voltage Monitor [RO] */ +#define CS35L32_BST_CPCP_CTL 0x0B /* Conv Peak Curr Protection CTL */ +#define CS35L32_IMON_SCALING 0x0C /* IMON Scaling */ +#define CS35L32_AUDIO_LED_MNGR 0x0D /* Audio/LED Pwr Manager */ +#define CS35L32_ADSP_CTL 0x0F /* Serial Port Control */ +#define CS35L32_CLASSD_CTL 0x10 /* Class D Amp CTL */ +#define CS35L32_PROTECT_CTL 0x11 /* Protection Release CTL */ +#define CS35L32_INT_MASK_1 0x12 /* Interrupt Mask 1 */ +#define CS35L32_INT_MASK_2 0x13 /* Interrupt Mask 2 */ +#define CS35L32_INT_MASK_3 0x14 /* Interrupt Mask 3 */ +#define CS35L32_INT_STATUS_1 0x15 /* Interrupt Status 1 [RO] */ +#define CS35L32_INT_STATUS_2 0x16 /* Interrupt Status 2 [RO] */ +#define CS35L32_INT_STATUS_3 0x17 /* Interrupt Status 3 [RO] */ +#define CS35L32_LED_STATUS 0x18 /* LED Lighting Status [RO] */ +#define CS35L32_FLASH_MODE 0x19 /* LED Flash Mode Current */ +#define CS35L32_MOVIE_MODE 0x1A /* LED Movie Mode Current */ +#define CS35L32_FLASH_TIMER 0x1B /* LED Flash Timer */ +#define CS35L32_FLASH_INHIBIT 0x1C /* LED Flash Inhibit Current */ +#define CS35L32_MAX_REGISTER 0x1C + +#define CS35L32_MCLK_DIV2 0x01 +#define CS35L32_MCLK_RATIO 0x01 +#define CS35L32_MCLKDIS 0x80 +#define CS35L32_PDN_ALL 0x01 +#define CS35L32_PDN_AMP 0x80 +#define CS35L32_PDN_BOOST 0x04 +#define CS35L32_PDN_IMON 0x40 +#define CS35L32_PDN_VMON 0x80 +#define CS35L32_PDN_VPMON 0x20 +#define CS35L32_PDN_ADSP 0x08 + +#define CS35L32_MCLK_DIV2_MASK 0x40 +#define CS35L32_MCLK_RATIO_MASK 0x01 +#define CS35L32_MCLK_MASK 0x41 +#define CS35L32_ADSP_MASTER_MASK 0x40 +#define CS35L32_BOOST_MASK 0x03 +#define CS35L32_GAIN_MGR_MASK 0x08 +#define CS35L32_ADSP_SHARE_MASK 0x08 +#define CS35L32_ADSP_DATACFG_MASK 0x30 +#define CS35L32_SDOUT_3ST 0x80 +#define CS35L32_BATT_REC_MASK 0x0E +#define CS35L32_BATT_THRESH_MASK 0x30 + +#define CS35L32_RATES (SNDRV_PCM_RATE_48000) +#define CS35L32_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + + +#endif -- cgit v1.1 From 38f57532ede565a3c71da7b7727369f374c51acb Mon Sep 17 00:00:00 2001 From: Brian Austin Date: Thu, 7 Aug 2014 09:34:38 -0500 Subject: ASoC: cs35l32: fix compile warning for i2c_probe Forgot to add a return for err_disable goto statement. Causes compile warning of control reaching end of non-void Signed-off-by: Brian Austin Signed-off-by: Mark Brown --- sound/soc/codecs/cs35l32.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/codecs/cs35l32.c b/sound/soc/codecs/cs35l32.c index 90565d5..9c6b272 100644 --- a/sound/soc/codecs/cs35l32.c +++ b/sound/soc/codecs/cs35l32.c @@ -549,6 +549,7 @@ static int cs35l32_i2c_probe(struct i2c_client *i2c_client, err_disable: regulator_bulk_disable(ARRAY_SIZE(cs35l32->supplies), cs35l32->supplies); + return ret; } static int cs35l32_i2c_remove(struct i2c_client *i2c_client) -- cgit v1.1 From 708b4351f08c08ea93f773fb9197bdd3f3b08273 Mon Sep 17 00:00:00 2001 From: Nicolin Chen Date: Wed, 30 Jul 2014 19:27:38 +0800 Subject: ASoC: fsl: Add Freescale Generic ASoC Sound Card with ASRC support The Freescale Generic ASoC Sound Card is a general ASoC DAI Link driver that can be used, ideally, for all Freescale CPU DAI drivers and external CODECs. The idea of this generic sound card is a bit like ASoC Simple Card. However, for Freescale SoCs (especially those released in recent years), most of them have ASRC (Documentation/devicetree/bindings/sound/fsl,asrc.txt) inside. And this is a specific feature that might be painstakingly controlled and merged into the Simple Card driver. So having this driver will allow all Freescale SoC users to benefit from the simplification to support a new card and the capability of wide sample rates support through ASRC. The driver is initially designed for sound card using I2S or PCM DAI formats. However, it's also possible to merge those non-I2S/PCM type sound cards, such as S/PDIF audio and HDMI audio, into this card as long as the merge will not break the original function and as long as there is something redundant that can be abstracted along with I2S type sound cards. As an initial version, it only supports three cards that I can test: imx-audio-cs42888, a new card that links ESAI with CS42888 CODEC imx-audio-sgtl5000, just like the old imx-sgtl5000.c driver imx-audio-wm8962, just like the old imx-wm8962.c driver The driver is also compatible with the old Device Tree bindings of WM8962 and SGTL5000. So we may consider to remove those two drivers after this driver is totally enabled. (It needs to be added into defconfig) Signed-off-by: Nicolin Chen Signed-off-by: Mark Brown --- sound/soc/fsl/Kconfig | 16 ++ sound/soc/fsl/Makefile | 2 + sound/soc/fsl/fsl-asoc-card.c | 573 ++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 591 insertions(+) create mode 100644 sound/soc/fsl/fsl-asoc-card.c (limited to 'sound') diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index f54a8fc..2b99a9e 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -59,6 +59,22 @@ config SND_SOC_FSL_ESAI config SND_SOC_FSL_UTILS tristate +config SND_SOC_FSL_ASOC_CARD + tristate "Generic ASoC Sound Card with ASRC support" + depends on OF && I2C + select SND_SOC_IMX_PCM_DMA + select SND_SOC_FSL_ESAI + select SND_SOC_FSL_SAI + select SND_SOC_FSL_SSI + select SND_SOC_CS42XX8_I2C + select SND_SOC_SGTL5000 + select SND_SOC_WM8962 + help + ALSA SoC Audio support with ASRC feature for Freescale SoCs that have + ESAI/SAI/SSI and connect with external CODECs such as WM8962, CS42888 + and SGTL5000. + Say Y if you want to add support for Freescale Generic ASoC Sound Card. + config SND_SOC_IMX_PCM_DMA tristate select SND_SOC_GENERIC_DMAENGINE_PCM diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index 9ff5926..8f6d84e 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -11,6 +11,7 @@ snd-soc-p1022-rdk-objs := p1022_rdk.o obj-$(CONFIG_SND_SOC_P1022_RDK) += snd-soc-p1022-rdk.o # Freescale SSI/DMA/SAI/SPDIF Support +snd-soc-fsl-asoc-card-objs := fsl-asoc-card.o snd-soc-fsl-asrc-objs := fsl_asrc.o fsl_asrc_dma.o snd-soc-fsl-sai-objs := fsl_sai.o snd-soc-fsl-ssi-y := fsl_ssi.o @@ -19,6 +20,7 @@ snd-soc-fsl-spdif-objs := fsl_spdif.o snd-soc-fsl-esai-objs := fsl_esai.o snd-soc-fsl-utils-objs := fsl_utils.o snd-soc-fsl-dma-objs := fsl_dma.o +obj-$(CONFIG_SND_SOC_FSL_ASOC_CARD) += snd-soc-fsl-asoc-card.o obj-$(CONFIG_SND_SOC_FSL_ASRC) += snd-soc-fsl-asrc.o obj-$(CONFIG_SND_SOC_FSL_SAI) += snd-soc-fsl-sai.o obj-$(CONFIG_SND_SOC_FSL_SSI) += snd-soc-fsl-ssi.o diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c new file mode 100644 index 0000000..cf3f1f4 --- /dev/null +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -0,0 +1,573 @@ +/* + * Freescale Generic ASoC Sound Card driver with ASRC + * + * Copyright (C) 2014 Freescale Semiconductor, Inc. + * + * Author: Nicolin Chen + * + * This file is licensed under the terms of the GNU General Public License + * version 2. This program is licensed "as is" without any warranty of any + * kind, whether express or implied. + */ + +#include +#include +#include +#include +#include +#include + +#include "fsl_esai.h" +#include "fsl_sai.h" +#include "imx-audmux.h" + +#include "../codecs/sgtl5000.h" +#include "../codecs/wm8962.h" + +#define RX 0 +#define TX 1 + +/* Default DAI format without Master and Slave flag */ +#define DAI_FMT_BASE (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF) + +/** + * CODEC private data + * + * @mclk_freq: Clock rate of MCLK + * @mclk_id: MCLK (or main clock) id for set_sysclk() + * @fll_id: FLL (or secordary clock) id for set_sysclk() + * @pll_id: PLL id for set_pll() + */ +struct codec_priv { + unsigned long mclk_freq; + u32 mclk_id; + u32 fll_id; + u32 pll_id; +}; + +/** + * CPU private data + * + * @sysclk_freq[2]: SYSCLK rates for set_sysclk() + * @sysclk_dir[2]: SYSCLK directions for set_sysclk() + * @sysclk_id[2]: SYSCLK ids for set_sysclk() + * + * Note: [1] for tx and [0] for rx + */ +struct cpu_priv { + unsigned long sysclk_freq[2]; + u32 sysclk_dir[2]; + u32 sysclk_id[2]; +}; + +/** + * Freescale Generic ASOC card private data + * + * @dai_link[3]: DAI link structure including normal one and DPCM link + * @pdev: platform device pointer + * @codec_priv: CODEC private data + * @cpu_priv: CPU private data + * @card: ASoC card structure + * @sample_rate: Current sample rate + * @sample_format: Current sample format + * @asrc_rate: ASRC sample rate used by Back-Ends + * @asrc_format: ASRC sample format used by Back-Ends + * @dai_fmt: DAI format between CPU and CODEC + * @name: Card name + */ + +struct fsl_asoc_card_priv { + struct snd_soc_dai_link dai_link[3]; + struct platform_device *pdev; + struct codec_priv codec_priv; + struct cpu_priv cpu_priv; + struct snd_soc_card card; + u32 sample_rate; + u32 sample_format; + u32 asrc_rate; + u32 asrc_format; + u32 dai_fmt; + char name[32]; +}; + +/** + * This dapm route map exsits for DPCM link only. + * The other routes shall go through Device Tree. + */ +static const struct snd_soc_dapm_route audio_map[] = { + {"CPU-Playback", NULL, "ASRC-Playback"}, + {"Playback", NULL, "CPU-Playback"}, + {"ASRC-Capture", NULL, "CPU-Capture"}, + {"CPU-Capture", NULL, "Capture"}, +}; + +/* Add all possible widgets into here without being redundant */ +static const struct snd_soc_dapm_widget fsl_asoc_card_dapm_widgets[] = { + SND_SOC_DAPM_LINE("Line Out Jack", NULL), + SND_SOC_DAPM_LINE("Line In Jack", NULL), + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), + SND_SOC_DAPM_MIC("Mic Jack", NULL), + SND_SOC_DAPM_MIC("AMIC", NULL), + SND_SOC_DAPM_MIC("DMIC", NULL), +}; + +static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(rtd->card); + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + struct cpu_priv *cpu_priv = &priv->cpu_priv; + struct device *dev = rtd->card->dev; + int ret; + + priv->sample_rate = params_rate(params); + priv->sample_format = params_format(params); + + if (priv->card.set_bias_level) + return 0; + + /* Specific configurations of DAIs starts from here */ + ret = snd_soc_dai_set_sysclk(rtd->cpu_dai, cpu_priv->sysclk_id[tx], + cpu_priv->sysclk_freq[tx], + cpu_priv->sysclk_dir[tx]); + if (ret) { + dev_err(dev, "failed to set sysclk for cpu dai\n"); + return ret; + } + + return 0; +} + +static struct snd_soc_ops fsl_asoc_card_ops = { + .hw_params = fsl_asoc_card_hw_params, +}; + +static int be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(rtd->card); + struct snd_interval *rate; + struct snd_mask *mask; + + rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); + rate->max = rate->min = priv->asrc_rate; + + mask = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); + snd_mask_none(mask); + snd_mask_set(mask, priv->asrc_format); + + return 0; +} + +static struct snd_soc_dai_link fsl_asoc_card_dai[] = { + /* Default ASoC DAI Link*/ + { + .name = "HiFi", + .stream_name = "HiFi", + .ops = &fsl_asoc_card_ops, + }, + /* DPCM Link between Front-End and Back-End (Optional) */ + { + .name = "HiFi-ASRC-FE", + .stream_name = "HiFi-ASRC-FE", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .dpcm_playback = 1, + .dpcm_capture = 1, + .dynamic = 1, + }, + { + .name = "HiFi-ASRC-BE", + .stream_name = "HiFi-ASRC-BE", + .platform_name = "snd-soc-dummy", + .be_hw_params_fixup = be_hw_params_fixup, + .ops = &fsl_asoc_card_ops, + .dpcm_playback = 1, + .dpcm_capture = 1, + .no_pcm = 1, + }, +}; + +static int fsl_asoc_card_set_bias_level(struct snd_soc_card *card, + struct snd_soc_dapm_context *dapm, + enum snd_soc_bias_level level) +{ + struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(card); + struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; + struct codec_priv *codec_priv = &priv->codec_priv; + struct device *dev = card->dev; + unsigned int pll_out; + int ret; + + if (dapm->dev != codec_dai->dev) + return 0; + + switch (level) { + case SND_SOC_BIAS_PREPARE: + if (dapm->bias_level != SND_SOC_BIAS_STANDBY) + break; + + if (priv->sample_format == SNDRV_PCM_FORMAT_S24_LE) + pll_out = priv->sample_rate * 384; + else + pll_out = priv->sample_rate * 256; + + ret = snd_soc_dai_set_pll(codec_dai, codec_priv->pll_id, + codec_priv->mclk_id, + codec_priv->mclk_freq, pll_out); + if (ret) { + dev_err(dev, "failed to start FLL: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(codec_dai, codec_priv->fll_id, + pll_out, SND_SOC_CLOCK_IN); + if (ret) { + dev_err(dev, "failed to set SYSCLK: %d\n", ret); + return ret; + } + break; + + case SND_SOC_BIAS_STANDBY: + if (dapm->bias_level != SND_SOC_BIAS_PREPARE) + break; + + ret = snd_soc_dai_set_sysclk(codec_dai, codec_priv->mclk_id, + codec_priv->mclk_freq, + SND_SOC_CLOCK_IN); + if (ret) { + dev_err(dev, "failed to switch away from FLL: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_pll(codec_dai, codec_priv->pll_id, 0, 0, 0); + if (ret) { + dev_err(dev, "failed to stop FLL: %d\n", ret); + return ret; + } + break; + + default: + break; + } + + return 0; +} + +static int fsl_asoc_card_audmux_init(struct device_node *np, + struct fsl_asoc_card_priv *priv) +{ + struct device *dev = &priv->pdev->dev; + u32 int_ptcr = 0, ext_ptcr = 0; + int int_port, ext_port; + int ret; + + ret = of_property_read_u32(np, "mux-int-port", &int_port); + if (ret) { + dev_err(dev, "mux-int-port missing or invalid\n"); + return ret; + } + ret = of_property_read_u32(np, "mux-ext-port", &ext_port); + if (ret) { + dev_err(dev, "mux-ext-port missing or invalid\n"); + return ret; + } + + /* + * The port numbering in the hardware manual starts at 1, while + * the AUDMUX API expects it starts at 0. + */ + int_port--; + ext_port--; + + /* + * Use asynchronous mode (6 wires) for all cases. + * If only 4 wires are needed, just set SSI into + * synchronous mode and enable 4 PADs in IOMUX. + */ + switch (priv->dai_fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + int_ptcr = IMX_AUDMUX_V2_PTCR_RFSEL(8 | ext_port) | + IMX_AUDMUX_V2_PTCR_RCSEL(8 | ext_port) | + IMX_AUDMUX_V2_PTCR_TFSEL(ext_port) | + IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) | + IMX_AUDMUX_V2_PTCR_RFSDIR | + IMX_AUDMUX_V2_PTCR_RCLKDIR | + IMX_AUDMUX_V2_PTCR_TFSDIR | + IMX_AUDMUX_V2_PTCR_TCLKDIR; + break; + case SND_SOC_DAIFMT_CBM_CFS: + int_ptcr = IMX_AUDMUX_V2_PTCR_RCSEL(8 | ext_port) | + IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) | + IMX_AUDMUX_V2_PTCR_RCLKDIR | + IMX_AUDMUX_V2_PTCR_TCLKDIR; + ext_ptcr = IMX_AUDMUX_V2_PTCR_RFSEL(8 | int_port) | + IMX_AUDMUX_V2_PTCR_TFSEL(int_port) | + IMX_AUDMUX_V2_PTCR_RFSDIR | + IMX_AUDMUX_V2_PTCR_TFSDIR; + break; + case SND_SOC_DAIFMT_CBS_CFM: + int_ptcr = IMX_AUDMUX_V2_PTCR_RFSEL(8 | ext_port) | + IMX_AUDMUX_V2_PTCR_TFSEL(ext_port) | + IMX_AUDMUX_V2_PTCR_RFSDIR | + IMX_AUDMUX_V2_PTCR_TFSDIR; + ext_ptcr = IMX_AUDMUX_V2_PTCR_RCSEL(8 | int_port) | + IMX_AUDMUX_V2_PTCR_TCSEL(int_port) | + IMX_AUDMUX_V2_PTCR_RCLKDIR | + IMX_AUDMUX_V2_PTCR_TCLKDIR; + break; + case SND_SOC_DAIFMT_CBS_CFS: + ext_ptcr = IMX_AUDMUX_V2_PTCR_RFSEL(8 | int_port) | + IMX_AUDMUX_V2_PTCR_RCSEL(8 | int_port) | + IMX_AUDMUX_V2_PTCR_TFSEL(int_port) | + IMX_AUDMUX_V2_PTCR_TCSEL(int_port) | + IMX_AUDMUX_V2_PTCR_RFSDIR | + IMX_AUDMUX_V2_PTCR_RCLKDIR | + IMX_AUDMUX_V2_PTCR_TFSDIR | + IMX_AUDMUX_V2_PTCR_TCLKDIR; + break; + default: + return -EINVAL; + } + + /* Asynchronous mode can not be set along with RCLKDIR */ + ret = imx_audmux_v2_configure_port(int_port, 0, + IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port)); + if (ret) { + dev_err(dev, "audmux internal port setup failed\n"); + return ret; + } + + ret = imx_audmux_v2_configure_port(int_port, int_ptcr, + IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port)); + if (ret) { + dev_err(dev, "audmux internal port setup failed\n"); + return ret; + } + + ret = imx_audmux_v2_configure_port(ext_port, 0, + IMX_AUDMUX_V2_PDCR_RXDSEL(int_port)); + if (ret) { + dev_err(dev, "audmux external port setup failed\n"); + return ret; + } + + ret = imx_audmux_v2_configure_port(ext_port, ext_ptcr, + IMX_AUDMUX_V2_PDCR_RXDSEL(int_port)); + if (ret) { + dev_err(dev, "audmux external port setup failed\n"); + return ret; + } + + return 0; +} + +static int fsl_asoc_card_late_probe(struct snd_soc_card *card) +{ + struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(card); + struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; + struct codec_priv *codec_priv = &priv->codec_priv; + struct device *dev = card->dev; + int ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, codec_priv->mclk_id, + codec_priv->mclk_freq, SND_SOC_CLOCK_IN); + if (ret) { + dev_err(dev, "failed to set sysclk in %s\n", __func__); + return ret; + } + + return 0; +} + +static int fsl_asoc_card_probe(struct platform_device *pdev) +{ + struct device_node *cpu_np, *codec_np, *asrc_np; + struct device_node *np = pdev->dev.of_node; + struct platform_device *asrc_pdev = NULL; + struct platform_device *cpu_pdev; + struct fsl_asoc_card_priv *priv; + struct i2c_client *codec_dev; + struct clk *codec_clk; + u32 width; + int ret; + + priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + cpu_np = of_parse_phandle(np, "audio-cpu", 0); + /* Give a chance to old DT binding */ + if (!cpu_np) + cpu_np = of_parse_phandle(np, "ssi-controller", 0); + codec_np = of_parse_phandle(np, "audio-codec", 0); + if (!cpu_np || !codec_np) { + dev_err(&pdev->dev, "phandle missing or invalid\n"); + ret = -EINVAL; + goto fail; + } + + cpu_pdev = of_find_device_by_node(cpu_np); + if (!cpu_pdev) { + dev_err(&pdev->dev, "failed to find CPU DAI device\n"); + ret = -EINVAL; + goto fail; + } + + codec_dev = of_find_i2c_device_by_node(codec_np); + if (!codec_dev) { + dev_err(&pdev->dev, "failed to find codec platform device\n"); + ret = -EINVAL; + goto fail; + } + + asrc_np = of_parse_phandle(np, "audio-asrc", 0); + if (asrc_np) + asrc_pdev = of_find_device_by_node(asrc_np); + + /* Get the MCLK rate only, and leave it controlled by CODEC drivers */ + codec_clk = clk_get(&codec_dev->dev, NULL); + if (!IS_ERR(codec_clk)) { + priv->codec_priv.mclk_freq = clk_get_rate(codec_clk); + clk_put(codec_clk); + } + + /* Default sample rate and format, will be updated in hw_params() */ + priv->sample_rate = 44100; + priv->sample_format = SNDRV_PCM_FORMAT_S16_LE; + + /* Assign a default DAI format, and allow each card to overwrite it */ + priv->dai_fmt = DAI_FMT_BASE; + + /* Diversify the card configurations */ + if (of_device_is_compatible(np, "fsl,imx-audio-cs42888")) { + priv->card.set_bias_level = NULL; + priv->cpu_priv.sysclk_freq[TX] = priv->codec_priv.mclk_freq; + priv->cpu_priv.sysclk_freq[RX] = priv->codec_priv.mclk_freq; + priv->cpu_priv.sysclk_dir[TX] = SND_SOC_CLOCK_OUT; + priv->cpu_priv.sysclk_dir[RX] = SND_SOC_CLOCK_OUT; + priv->dai_fmt |= SND_SOC_DAIFMT_CBS_CFS; + } else if (of_device_is_compatible(np, "fsl,imx-audio-sgtl5000")) { + priv->codec_priv.mclk_id = SGTL5000_SYSCLK; + priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; + } else if (of_device_is_compatible(np, "fsl,imx-audio-wm8962")) { + priv->card.set_bias_level = fsl_asoc_card_set_bias_level; + priv->codec_priv.mclk_id = WM8962_SYSCLK_MCLK; + priv->codec_priv.fll_id = WM8962_SYSCLK_FLL; + priv->codec_priv.pll_id = WM8962_FLL; + priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; + } else { + dev_err(&pdev->dev, "unknown Device Tree compatible\n"); + return -EINVAL; + } + + /* Common settings for corresponding Freescale CPU DAI driver */ + if (strstr(cpu_np->name, "ssi")) { + /* Only SSI needs to configure AUDMUX */ + ret = fsl_asoc_card_audmux_init(np, priv); + if (ret) { + dev_err(&pdev->dev, "failed to init audmux\n"); + goto fail; + } + } else if (strstr(cpu_np->name, "esai")) { + priv->cpu_priv.sysclk_id[1] = ESAI_HCKT_EXTAL; + priv->cpu_priv.sysclk_id[0] = ESAI_HCKR_EXTAL; + } else if (strstr(cpu_np->name, "sai")) { + priv->cpu_priv.sysclk_id[1] = FSL_SAI_CLK_MAST1; + priv->cpu_priv.sysclk_id[0] = FSL_SAI_CLK_MAST1; + } + + sprintf(priv->name, "%s-audio", codec_dev->name); + + /* Initialize sound card */ + priv->pdev = pdev; + priv->card.dev = &pdev->dev; + priv->card.name = priv->name; + priv->card.dai_link = priv->dai_link; + priv->card.dapm_routes = audio_map; + priv->card.late_probe = fsl_asoc_card_late_probe; + priv->card.num_dapm_routes = ARRAY_SIZE(audio_map); + priv->card.dapm_widgets = fsl_asoc_card_dapm_widgets; + priv->card.num_dapm_widgets = ARRAY_SIZE(fsl_asoc_card_dapm_widgets); + + memcpy(priv->dai_link, fsl_asoc_card_dai, + sizeof(struct snd_soc_dai_link) * ARRAY_SIZE(priv->dai_link)); + + /* Normal DAI Link */ + priv->dai_link[0].cpu_of_node = cpu_np; + priv->dai_link[0].codec_of_node = codec_np; + priv->dai_link[0].codec_dai_name = codec_dev->name; + priv->dai_link[0].platform_of_node = cpu_np; + priv->dai_link[0].dai_fmt = priv->dai_fmt; + priv->card.num_links = 1; + + if (asrc_pdev) { + /* DPCM DAI Links only if ASRC exsits */ + priv->dai_link[1].cpu_of_node = asrc_np; + priv->dai_link[1].platform_of_node = asrc_np; + priv->dai_link[2].codec_dai_name = codec_dev->name; + priv->dai_link[2].codec_of_node = codec_np; + priv->dai_link[2].cpu_of_node = cpu_np; + priv->dai_link[2].dai_fmt = priv->dai_fmt; + priv->card.num_links = 3; + + ret = of_property_read_u32(asrc_np, "fsl,asrc-rate", + &priv->asrc_rate); + if (ret) { + dev_err(&pdev->dev, "failed to get output rate\n"); + ret = -EINVAL; + goto fail; + } + + ret = of_property_read_u32(asrc_np, "fsl,asrc-width", &width); + if (ret) { + dev_err(&pdev->dev, "failed to get output rate\n"); + ret = -EINVAL; + goto fail; + } + + if (width == 24) + priv->asrc_format = SNDRV_PCM_FORMAT_S24_LE; + else + priv->asrc_format = SNDRV_PCM_FORMAT_S16_LE; + } + + /* Finish card registering */ + platform_set_drvdata(pdev, priv); + snd_soc_card_set_drvdata(&priv->card, priv); + + ret = devm_snd_soc_register_card(&pdev->dev, &priv->card); + if (ret) + dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); + +fail: + of_node_put(codec_np); + of_node_put(asrc_np); + of_node_put(cpu_np); + + return ret; +} + +static const struct of_device_id fsl_asoc_card_dt_ids[] = { + { .compatible = "fsl,imx-audio-cs42888", }, + { .compatible = "fsl,imx-audio-sgtl5000", }, + { .compatible = "fsl,imx-audio-wm8962", }, + {} +}; + +static struct platform_driver fsl_asoc_card_driver = { + .probe = fsl_asoc_card_probe, + .driver = { + .name = "fsl-asoc-card", + .pm = &snd_soc_pm_ops, + .of_match_table = fsl_asoc_card_dt_ids, + }, +}; +module_platform_driver(fsl_asoc_card_driver); + +MODULE_DESCRIPTION("Freescale Generic ASoC Sound Card driver with ASRC"); +MODULE_AUTHOR("Nicolin Chen "); +MODULE_ALIAS("platform:fsl-asoc-card"); +MODULE_LICENSE("GPL"); -- cgit v1.1 From de0d712a6dd1eed097dc6aa4f97ee461949414fe Mon Sep 17 00:00:00 2001 From: Shengjiu Wang Date: Fri, 8 Aug 2014 14:47:21 +0800 Subject: ASoC: fsl_esai: refine esai for TDM support Original driver didn't store the number of slots, just fix the slot number to 2, use this default number to calculate bclk and pins for TX/RX. In this patch, add one parameter for slots, and update the calculation of bclk and pins of TX/RX. Then driver will be compatible with slots > 2 in TDM mode. Signed-off-by: Shengjiu Wang Acked-by: Nicolin Chen Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_esai.c | 14 +++++++++++--- sound/soc/fsl/fsl_esai.h | 8 ++++---- 2 files changed, 15 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index 72d154e..f252370 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -38,6 +38,7 @@ * @fsysclk: system clock source to derive HCK, SCK and FS * @fifo_depth: depth of tx/rx FIFO * @slot_width: width of each DAI slot + * @slots: number of slots * @hck_rate: clock rate of desired HCKx clock * @sck_rate: clock rate of desired SCKx clock * @hck_dir: the direction of HCKx pads @@ -56,6 +57,7 @@ struct fsl_esai { struct clk *fsysclk; u32 fifo_depth; u32 slot_width; + u32 slots; u32 hck_rate[2]; u32 sck_rate[2]; bool hck_dir[2]; @@ -363,6 +365,7 @@ static int fsl_esai_set_dai_tdm_slot(struct snd_soc_dai *dai, u32 tx_mask, ESAI_xSMB_xS_MASK, ESAI_xSMB_xS(rx_mask)); esai_priv->slot_width = slot_width; + esai_priv->slots = slots; return 0; } @@ -510,10 +513,11 @@ static int fsl_esai_hw_params(struct snd_pcm_substream *substream, bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; u32 width = snd_pcm_format_width(params_format(params)); u32 channels = params_channels(params); + u32 pins = DIV_ROUND_UP(channels, esai_priv->slots); u32 bclk, mask, val; int ret; - bclk = params_rate(params) * esai_priv->slot_width * 2; + bclk = params_rate(params) * esai_priv->slot_width * esai_priv->slots; ret = fsl_esai_set_bclk(dai, tx, bclk); if (ret) @@ -530,7 +534,7 @@ static int fsl_esai_hw_params(struct snd_pcm_substream *substream, mask = ESAI_xFCR_xFR_MASK | ESAI_xFCR_xWA_MASK | ESAI_xFCR_xFWM_MASK | (tx ? ESAI_xFCR_TE_MASK | ESAI_xFCR_TIEN : ESAI_xFCR_RE_MASK); val = ESAI_xFCR_xWA(width) | ESAI_xFCR_xFWM(esai_priv->fifo_depth) | - (tx ? ESAI_xFCR_TE(channels) | ESAI_xFCR_TIEN : ESAI_xFCR_RE(channels)); + (tx ? ESAI_xFCR_TE(pins) | ESAI_xFCR_TIEN : ESAI_xFCR_RE(pins)); regmap_update_bits(esai_priv->regmap, REG_ESAI_xFCR(tx), mask, val); @@ -565,6 +569,7 @@ static int fsl_esai_trigger(struct snd_pcm_substream *substream, int cmd, struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai); bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; u8 i, channels = substream->runtime->channels; + u32 pins = DIV_ROUND_UP(channels, esai_priv->slots); switch (cmd) { case SNDRV_PCM_TRIGGER_START: @@ -579,7 +584,7 @@ static int fsl_esai_trigger(struct snd_pcm_substream *substream, int cmd, regmap_update_bits(esai_priv->regmap, REG_ESAI_xCR(tx), tx ? ESAI_xCR_TE_MASK : ESAI_xCR_RE_MASK, - tx ? ESAI_xCR_TE(channels) : ESAI_xCR_RE(channels)); + tx ? ESAI_xCR_TE(pins) : ESAI_xCR_RE(pins)); break; case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_STOP: @@ -783,6 +788,9 @@ static int fsl_esai_probe(struct platform_device *pdev) /* Set a default slot size */ esai_priv->slot_width = 32; + /* Set a default slot number */ + esai_priv->slots = 2; + /* Set a default master/slave state */ esai_priv->slave_mode = true; diff --git a/sound/soc/fsl/fsl_esai.h b/sound/soc/fsl/fsl_esai.h index 75e1403..91a550f 100644 --- a/sound/soc/fsl/fsl_esai.h +++ b/sound/soc/fsl/fsl_esai.h @@ -130,8 +130,8 @@ #define ESAI_xFCR_RE_WIDTH 4 #define ESAI_xFCR_TE_MASK (((1 << ESAI_xFCR_TE_WIDTH) - 1) << ESAI_xFCR_xE_SHIFT) #define ESAI_xFCR_RE_MASK (((1 << ESAI_xFCR_RE_WIDTH) - 1) << ESAI_xFCR_xE_SHIFT) -#define ESAI_xFCR_TE(x) ((ESAI_xFCR_TE_MASK >> (ESAI_xFCR_TE_WIDTH - ((x + 1) >> 1))) & ESAI_xFCR_TE_MASK) -#define ESAI_xFCR_RE(x) ((ESAI_xFCR_RE_MASK >> (ESAI_xFCR_RE_WIDTH - ((x + 1) >> 1))) & ESAI_xFCR_RE_MASK) +#define ESAI_xFCR_TE(x) ((ESAI_xFCR_TE_MASK >> (ESAI_xFCR_TE_WIDTH - x)) & ESAI_xFCR_TE_MASK) +#define ESAI_xFCR_RE(x) ((ESAI_xFCR_RE_MASK >> (ESAI_xFCR_RE_WIDTH - x)) & ESAI_xFCR_RE_MASK) #define ESAI_xFCR_xFR_SHIFT 1 #define ESAI_xFCR_xFR_MASK (1 << ESAI_xFCR_xFR_SHIFT) #define ESAI_xFCR_xFR (1 << ESAI_xFCR_xFR_SHIFT) @@ -272,8 +272,8 @@ #define ESAI_xCR_RE_WIDTH 4 #define ESAI_xCR_TE_MASK (((1 << ESAI_xCR_TE_WIDTH) - 1) << ESAI_xCR_xE_SHIFT) #define ESAI_xCR_RE_MASK (((1 << ESAI_xCR_RE_WIDTH) - 1) << ESAI_xCR_xE_SHIFT) -#define ESAI_xCR_TE(x) ((ESAI_xCR_TE_MASK >> (ESAI_xCR_TE_WIDTH - ((x + 1) >> 1))) & ESAI_xCR_TE_MASK) -#define ESAI_xCR_RE(x) ((ESAI_xCR_RE_MASK >> (ESAI_xCR_RE_WIDTH - ((x + 1) >> 1))) & ESAI_xCR_RE_MASK) +#define ESAI_xCR_TE(x) ((ESAI_xCR_TE_MASK >> (ESAI_xCR_TE_WIDTH - x)) & ESAI_xCR_TE_MASK) +#define ESAI_xCR_RE(x) ((ESAI_xCR_RE_MASK >> (ESAI_xCR_RE_WIDTH - x)) & ESAI_xCR_RE_MASK) /* * Transmit Clock Control Register -- REG_ESAI_TCCR 0xD8 -- cgit v1.1 From 376d1a92ca587d3974d4791cdb99baa8b8e7f0dd Mon Sep 17 00:00:00 2001 From: Nicolin Chen Date: Tue, 5 Aug 2014 17:20:21 +0800 Subject: ASoC: fsl_sai: Initialize with software reset This patch adds software reset code in dai_probe() so as to make a true init by clearing SAI's internal logic, including the bit clock generation, status flags, and FIFO pointers. Signed-off-by: Nicolin Chen Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_sai.c | 9 +++++++-- sound/soc/fsl/fsl_sai.h | 1 + 2 files changed, 8 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index faa0497..7b1eecb 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -437,8 +437,13 @@ static int fsl_sai_dai_probe(struct snd_soc_dai *cpu_dai) { struct fsl_sai *sai = dev_get_drvdata(cpu_dai->dev); - regmap_update_bits(sai->regmap, FSL_SAI_TCSR, 0xffffffff, 0x0); - regmap_update_bits(sai->regmap, FSL_SAI_RCSR, 0xffffffff, 0x0); + /* Software Reset for both Tx and Rx */ + regmap_write(sai->regmap, FSL_SAI_TCSR, FSL_SAI_CSR_SR); + regmap_write(sai->regmap, FSL_SAI_RCSR, FSL_SAI_CSR_SR); + /* Clear SR bit to finish the reset */ + regmap_write(sai->regmap, FSL_SAI_TCSR, 0); + regmap_write(sai->regmap, FSL_SAI_RCSR, 0); + regmap_update_bits(sai->regmap, FSL_SAI_TCR1, FSL_SAI_CR1_RFW_MASK, FSL_SAI_MAXBURST_TX * 2); regmap_update_bits(sai->regmap, FSL_SAI_RCR1, FSL_SAI_CR1_RFW_MASK, diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h index 0e6c9f5..8e1feab 100644 --- a/sound/soc/fsl/fsl_sai.h +++ b/sound/soc/fsl/fsl_sai.h @@ -48,6 +48,7 @@ /* SAI Transmit/Recieve Control Register */ #define FSL_SAI_CSR_TERE BIT(31) #define FSL_SAI_CSR_FR BIT(25) +#define FSL_SAI_CSR_SR BIT(24) #define FSL_SAI_CSR_xF_SHIFT 16 #define FSL_SAI_CSR_xF_W_SHIFT 18 #define FSL_SAI_CSR_xF_MASK (0x1f << FSL_SAI_CSR_xF_SHIFT) -- cgit v1.1 From af96ff5b7448dc776dc24a5c4313c6ec1ee94e53 Mon Sep 17 00:00:00 2001 From: Nicolin Chen Date: Mon, 4 Aug 2014 15:07:25 +0800 Subject: ASoC: fsl_sai: Set SYNC bit of TCR2 to Asynchronous Mode There is one design rule according to SAI's reference manual: If the transmitter bit clock and frame sync are to be used by both transmitter and receiver, the transmitter must be configured for asynchronous operation and the receiver for synchronous operation. And SYNC of TCR2 is a 2-width control bit: 00 Asynchronous mode. 01 Synchronous with receiver. 10 Synchronous with another SAI transmitter. 11 Synchronous with another SAI receiver. So the driver should have set SYNC bit of TCR2 to 0x0, and meanwhile set SYNC bit of RCR2 to 0x1 (Synchronous with transmitter). Signed-off-by: Nicolin Chen Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_sai.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 7b1eecb..3d865ad 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -333,8 +333,7 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd, * The transmitter bit clock and frame sync are to be * used by both the transmitter and receiver. */ - regmap_update_bits(sai->regmap, FSL_SAI_TCR2, FSL_SAI_CR2_SYNC, - ~FSL_SAI_CR2_SYNC); + regmap_update_bits(sai->regmap, FSL_SAI_TCR2, FSL_SAI_CR2_SYNC, 0); regmap_update_bits(sai->regmap, FSL_SAI_RCR2, FSL_SAI_CR2_SYNC, FSL_SAI_CR2_SYNC); -- cgit v1.1 From 08fdf65e37d560581233e06a659f73deeb3766f9 Mon Sep 17 00:00:00 2001 From: Nicolin Chen Date: Tue, 5 Aug 2014 15:32:05 +0800 Subject: ASoC: fsl_sai: Add asynchronous mode support SAI supports these operation modes: 1) asynchronous mode Both Tx and Rx are set to be asynchronous. 2) synchronous mode (Rx sync with Tx) Tx is set to be asynchronous, Rx is set to be synchronous. 3) synchronous mode (Tx sync with Rx) Rx is set to be asynchronous, Tx is set to be synchronous. 4) synchronous mode (Tx/Rx sync with another SAI's Tx) 5) synchronous mode (Tx/Rx sync with another SAI's Rx) * 4) and 5) are beyond this patch because they are related with another SAI. As the initial version of this SAI driver, it supported 2) as default while the others were totally missing. So this patch just adds supports for 1) and 3). Signed-off-by: Nicolin Chen Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_sai.c | 30 ++++++++++++++++++++++++++---- sound/soc/fsl/fsl_sai.h | 4 ++++ 2 files changed, 30 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 3d865ad..ef7c758 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -330,12 +330,14 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd, u32 xcsr, count = 100; /* - * The transmitter bit clock and frame sync are to be - * used by both the transmitter and receiver. + * Asynchronous mode: Clear SYNC for both Tx and Rx. + * Rx sync with Tx clocks: Clear SYNC for Tx, set it for Rx. + * Tx sync with Rx clocks: Clear SYNC for Rx, set it for Tx. */ - regmap_update_bits(sai->regmap, FSL_SAI_TCR2, FSL_SAI_CR2_SYNC, 0); + regmap_update_bits(sai->regmap, FSL_SAI_TCR2, FSL_SAI_CR2_SYNC, + sai->synchronous[TX] ? FSL_SAI_CR2_SYNC : 0); regmap_update_bits(sai->regmap, FSL_SAI_RCR2, FSL_SAI_CR2_SYNC, - FSL_SAI_CR2_SYNC); + sai->synchronous[RX] ? FSL_SAI_CR2_SYNC : 0); /* * It is recommended that the transmitter is the last enabled @@ -625,6 +627,26 @@ static int fsl_sai_probe(struct platform_device *pdev) return ret; } + /* Sync Tx with Rx as default by following old DT binding */ + sai->synchronous[RX] = true; + sai->synchronous[TX] = false; + fsl_sai_dai.symmetric_rates = 1; + fsl_sai_dai.symmetric_channels = 1; + fsl_sai_dai.symmetric_samplebits = 1; + + if (of_find_property(np, "fsl,sai-synchronous-rx", NULL)) { + /* Sync Rx with Tx */ + sai->synchronous[RX] = false; + sai->synchronous[TX] = true; + } else if (of_find_property(np, "fsl,sai-asynchronous", NULL)) { + /* Discard all settings for asynchronous mode */ + sai->synchronous[RX] = false; + sai->synchronous[TX] = false; + fsl_sai_dai.symmetric_rates = 0; + fsl_sai_dai.symmetric_channels = 0; + fsl_sai_dai.symmetric_samplebits = 0; + } + sai->dma_params_rx.addr = res->start + FSL_SAI_RDR; sai->dma_params_tx.addr = res->start + FSL_SAI_TDR; sai->dma_params_rx.maxburst = FSL_SAI_MAXBURST_RX; diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h index 8e1feab..b3d8864 100644 --- a/sound/soc/fsl/fsl_sai.h +++ b/sound/soc/fsl/fsl_sai.h @@ -136,9 +136,13 @@ struct fsl_sai { bool big_endian_data; bool is_dsp_mode; bool sai_on_imx; + bool synchronous[2]; struct snd_dmaengine_dai_dma_data dma_params_rx; struct snd_dmaengine_dai_dma_data dma_params_tx; }; +#define TX 1 +#define RX 0 + #endif /* __FSL_SAI_H */ -- cgit v1.1 From ce7344a4ebabe90e064d3e087727f45624cdc942 Mon Sep 17 00:00:00 2001 From: Nicolin Chen Date: Fri, 8 Aug 2014 18:41:19 +0800 Subject: ASoC: fsl_sai: Make Synchronous and Asynchronous modes exclusive The previous patch (ASoC: fsl_sai: Add asynchronous mode support) added new Device Tree bindings for Asynchronous and Synchronous modes support. However, these two shall not be present at the same time. So this patch just simply makes them exclusive so as to avoid incorrect Device Tree binding usage. Signed-off-by: Nicolin Chen Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_sai.c | 7 +++++++ 1 file changed, 7 insertions(+) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index ef7c758..4c9e71c 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -634,6 +634,13 @@ static int fsl_sai_probe(struct platform_device *pdev) fsl_sai_dai.symmetric_channels = 1; fsl_sai_dai.symmetric_samplebits = 1; + if (of_find_property(np, "fsl,sai-synchronous-rx", NULL) && + of_find_property(np, "fsl,sai-asynchronous", NULL)) { + /* error out if both synchronous and asynchronous are present */ + dev_err(&pdev->dev, "invalid binding for synchronous mode\n"); + return -EINVAL; + } + if (of_find_property(np, "fsl,sai-synchronous-rx", NULL)) { /* Sync Rx with Tx */ sai->synchronous[RX] = false; -- cgit v1.1 From ea5edfe2f1ce5b2254a5ec4c1bb224fac48c3153 Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Mon, 4 Aug 2014 15:04:19 +0530 Subject: ASoC: Intel: Fix to use byte control interface Using a byte control interface instead of generic_params ioctl. Signed-off-by: Subhransu S. Prusty Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/sst-mfld-platform.h | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/sst-mfld-platform.h b/sound/soc/intel/sst-mfld-platform.h index 6c6a42c..cc3a088 100644 --- a/sound/soc/intel/sst-mfld-platform.h +++ b/sound/soc/intel/sst-mfld-platform.h @@ -63,9 +63,7 @@ enum sst_controls { SST_SND_BUFFER_POINTER = 0x05, SST_SND_STREAM_INIT = 0x06, SST_SND_START = 0x07, - SST_SET_BYTE_STREAM = 0x100A, - SST_GET_BYTE_STREAM = 0x100B, - SST_MAX_CONTROLS = SST_GET_BYTE_STREAM, + SST_MAX_CONTROLS = 0x07, }; enum sst_stream_ops { @@ -129,7 +127,7 @@ struct compress_sst_ops { struct sst_ops { int (*open) (struct snd_sst_params *str_param); int (*device_control) (int cmd, void *arg); - int (*set_generic_params)(enum sst_controls cmd, void *arg); + int (*send_byte_stream)(struct snd_sst_bytes_v2 *bytes); int (*close) (unsigned int str_id); }; -- cgit v1.1 From 5981c2d6db2ef16d96ee4d1c4d3ddff4ad9d8ebc Mon Sep 17 00:00:00 2001 From: "Subhransu S. Prusty" Date: Mon, 4 Aug 2014 15:04:20 +0530 Subject: ASoC: Intel: mfld-pcm: Use function instead of ioctl Signed-off-by: Subhransu S. Prusty Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/sst-mfld-platform-pcm.c | 21 +++++++++------------ sound/soc/intel/sst-mfld-platform.h | 19 ++++++------------- 2 files changed, 15 insertions(+), 25 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/sst-mfld-platform-pcm.c b/sound/soc/intel/sst-mfld-platform-pcm.c index 706212a..42766a5 100644 --- a/sound/soc/intel/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/sst-mfld-platform-pcm.c @@ -314,8 +314,7 @@ static int sst_platform_init_stream(struct snd_pcm_substream *substream) stream->stream_info.arg = substream; stream->stream_info.buffer_ptr = 0; stream->stream_info.sfreq = substream->runtime->rate; - ret_val = stream->ops->device_control( - SST_SND_STREAM_INIT, &stream->stream_info); + ret_val = stream->ops->stream_init(&stream->stream_info); if (ret_val) pr_err("control_set ret error %d\n", ret_val); return ret_val; @@ -403,8 +402,7 @@ static int sst_media_prepare(struct snd_pcm_substream *substream, stream = substream->runtime->private_data; str_id = stream->stream_info.str_id; if (stream->stream_info.str_id) { - ret_val = stream->ops->device_control( - SST_SND_DROP, &str_id); + ret_val = stream->ops->stream_drop(str_id); return ret_val; } @@ -461,7 +459,7 @@ static int sst_platform_pcm_trigger(struct snd_pcm_substream *substream, { int ret_val = 0, str_id; struct sst_runtime_stream *stream; - int str_cmd, status; + int status; pr_debug("sst_platform_pcm_trigger called\n"); stream = substream->runtime->private_data; @@ -469,29 +467,29 @@ static int sst_platform_pcm_trigger(struct snd_pcm_substream *substream, switch (cmd) { case SNDRV_PCM_TRIGGER_START: pr_debug("sst: Trigger Start\n"); - str_cmd = SST_SND_START; status = SST_PLATFORM_RUNNING; stream->stream_info.arg = substream; + ret_val = stream->ops->stream_start(str_id); break; case SNDRV_PCM_TRIGGER_STOP: pr_debug("sst: in stop\n"); - str_cmd = SST_SND_DROP; status = SST_PLATFORM_DROPPED; + ret_val = stream->ops->stream_drop(str_id); break; case SNDRV_PCM_TRIGGER_PAUSE_PUSH: pr_debug("sst: in pause\n"); - str_cmd = SST_SND_PAUSE; status = SST_PLATFORM_PAUSED; + ret_val = stream->ops->stream_pause(str_id); break; case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: pr_debug("sst: in pause release\n"); - str_cmd = SST_SND_RESUME; status = SST_PLATFORM_RUNNING; + ret_val = stream->ops->stream_pause_release(str_id); break; default: return -EINVAL; } - ret_val = stream->ops->device_control(str_cmd, &str_id); + if (!ret_val) sst_set_stream_status(stream, status); @@ -511,8 +509,7 @@ static snd_pcm_uframes_t sst_platform_pcm_pointer if (status == SST_PLATFORM_INIT) return 0; str_info = &stream->stream_info; - ret_val = stream->ops->device_control( - SST_SND_BUFFER_POINTER, str_info); + ret_val = stream->ops->stream_read_tstamp(str_info); if (ret_val) { pr_err("sst: error code = %d\n", ret_val); return ret_val; diff --git a/sound/soc/intel/sst-mfld-platform.h b/sound/soc/intel/sst-mfld-platform.h index cc3a088..2d6e65b 100644 --- a/sound/soc/intel/sst-mfld-platform.h +++ b/sound/soc/intel/sst-mfld-platform.h @@ -54,18 +54,6 @@ enum sst_drv_status { SST_PLATFORM_DROPPED, }; -enum sst_controls { - SST_SND_ALLOC = 0x00, - SST_SND_PAUSE = 0x01, - SST_SND_RESUME = 0x02, - SST_SND_DROP = 0x03, - SST_SND_FREE = 0x04, - SST_SND_BUFFER_POINTER = 0x05, - SST_SND_STREAM_INIT = 0x06, - SST_SND_START = 0x07, - SST_MAX_CONTROLS = 0x07, -}; - enum sst_stream_ops { STREAM_OPS_PLAYBACK = 0, STREAM_OPS_CAPTURE, @@ -126,7 +114,12 @@ struct compress_sst_ops { struct sst_ops { int (*open) (struct snd_sst_params *str_param); - int (*device_control) (int cmd, void *arg); + int (*stream_init) (struct pcm_stream_info *str_info); + int (*stream_start) (int str_id); + int (*stream_drop) (int str_id); + int (*stream_pause) (int str_id); + int (*stream_pause_release) (int str_id); + int (*stream_read_tstamp) (struct pcm_stream_info *str_info); int (*send_byte_stream)(struct snd_sst_bytes_v2 *bytes); int (*close) (unsigned int str_id); }; -- cgit v1.1 From b12b087c8715286b8759016f1d5c36cac0bb37f6 Mon Sep 17 00:00:00 2001 From: "Subhransu S. Prusty" Date: Mon, 4 Aug 2014 15:04:21 +0530 Subject: ASoC: Intel: mfld-pcm: Change sst_ops prototypes to take dev parameter sst_ops need to use the sst driver context. So pass sst device as argument, which can be used to retrieve sst context. Signed-off-by: Subhransu S. Prusty Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/sst-mfld-platform-pcm.c | 19 +++++++++---------- sound/soc/intel/sst-mfld-platform.h | 18 +++++++++--------- 2 files changed, 18 insertions(+), 19 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/sst-mfld-platform-pcm.c b/sound/soc/intel/sst-mfld-platform-pcm.c index 42766a5..a89ff7e 100644 --- a/sound/soc/intel/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/sst-mfld-platform-pcm.c @@ -277,7 +277,7 @@ static int sst_platform_alloc_stream(struct snd_pcm_substream *substream, stream->stream_info.str_id = str_params.stream_id; - ret_val = stream->ops->open(&str_params); + ret_val = stream->ops->open(sst->dev, &str_params); if (ret_val <= 0) return ret_val; @@ -314,13 +314,12 @@ static int sst_platform_init_stream(struct snd_pcm_substream *substream) stream->stream_info.arg = substream; stream->stream_info.buffer_ptr = 0; stream->stream_info.sfreq = substream->runtime->rate; - ret_val = stream->ops->stream_init(&stream->stream_info); + ret_val = stream->ops->stream_init(sst->dev, &stream->stream_info); if (ret_val) pr_err("control_set ret error %d\n", ret_val); return ret_val; } -/* end -- helper functions */ static int sst_media_open(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) @@ -372,7 +371,7 @@ static void sst_media_close(struct snd_pcm_substream *substream, stream = substream->runtime->private_data; str_id = stream->stream_info.str_id; if (str_id) - ret_val = stream->ops->close(str_id); + ret_val = stream->ops->close(sst->dev, str_id); module_put(sst->dev->driver->owner); kfree(stream); } @@ -402,7 +401,7 @@ static int sst_media_prepare(struct snd_pcm_substream *substream, stream = substream->runtime->private_data; str_id = stream->stream_info.str_id; if (stream->stream_info.str_id) { - ret_val = stream->ops->stream_drop(str_id); + ret_val = stream->ops->stream_drop(sst->dev, str_id); return ret_val; } @@ -469,22 +468,22 @@ static int sst_platform_pcm_trigger(struct snd_pcm_substream *substream, pr_debug("sst: Trigger Start\n"); status = SST_PLATFORM_RUNNING; stream->stream_info.arg = substream; - ret_val = stream->ops->stream_start(str_id); + ret_val = stream->ops->stream_start(sst->dev, str_id); break; case SNDRV_PCM_TRIGGER_STOP: pr_debug("sst: in stop\n"); status = SST_PLATFORM_DROPPED; - ret_val = stream->ops->stream_drop(str_id); + ret_val = stream->ops->stream_drop(sst->dev, str_id); break; case SNDRV_PCM_TRIGGER_PAUSE_PUSH: pr_debug("sst: in pause\n"); status = SST_PLATFORM_PAUSED; - ret_val = stream->ops->stream_pause(str_id); + ret_val = stream->ops->stream_pause(sst->dev, str_id); break; case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: pr_debug("sst: in pause release\n"); status = SST_PLATFORM_RUNNING; - ret_val = stream->ops->stream_pause_release(str_id); + ret_val = stream->ops->stream_pause_release(sst->dev, str_id); break; default: return -EINVAL; @@ -509,7 +508,7 @@ static snd_pcm_uframes_t sst_platform_pcm_pointer if (status == SST_PLATFORM_INIT) return 0; str_info = &stream->stream_info; - ret_val = stream->ops->stream_read_tstamp(str_info); + ret_val = stream->ops->stream_read_tstamp(sst->dev, str_info); if (ret_val) { pr_err("sst: error code = %d\n", ret_val); return ret_val; diff --git a/sound/soc/intel/sst-mfld-platform.h b/sound/soc/intel/sst-mfld-platform.h index 2d6e65b..d4c28b8 100644 --- a/sound/soc/intel/sst-mfld-platform.h +++ b/sound/soc/intel/sst-mfld-platform.h @@ -113,15 +113,15 @@ struct compress_sst_ops { }; struct sst_ops { - int (*open) (struct snd_sst_params *str_param); - int (*stream_init) (struct pcm_stream_info *str_info); - int (*stream_start) (int str_id); - int (*stream_drop) (int str_id); - int (*stream_pause) (int str_id); - int (*stream_pause_release) (int str_id); - int (*stream_read_tstamp) (struct pcm_stream_info *str_info); - int (*send_byte_stream)(struct snd_sst_bytes_v2 *bytes); - int (*close) (unsigned int str_id); + int (*open) (struct device *dev, struct snd_sst_params *str_param); + int (*stream_init) (struct device *dev, struct pcm_stream_info *str_info); + int (*stream_start) (struct device *dev, int str_id); + int (*stream_drop) (struct device *dev, int str_id); + int (*stream_pause) (struct device *dev, int str_id); + int (*stream_pause_release) (struct device *dev, int str_id); + int (*stream_read_tstamp) (struct device *dev, struct pcm_stream_info *str_info); + int (*send_byte_stream)(struct device *dev, struct snd_sst_bytes_v2 *bytes); + int (*close) (struct device *dev, unsigned int str_id); }; struct sst_runtime_stream { -- cgit v1.1 From d8499c9b4b03ca88d7c7b4094cb09471658df7c2 Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Mon, 4 Aug 2014 15:15:55 +0530 Subject: ASoC: Intel: add mrfld DSP defines We define the DSP commands,structures here which will be used to send the IPCs Signed-off-by: Vinod Koul Signed-off-by: Subhransu S. Prusty Signed-off-by: Mark Brown --- sound/soc/intel/Makefile | 3 +- sound/soc/intel/sst-atom-controls.c | 39 +++++ sound/soc/intel/sst-atom-controls.h | 286 +++++++++++++++++++++++++++++++- sound/soc/intel/sst-mfld-platform-pcm.c | 8 +- sound/soc/intel/sst-mfld-platform.h | 3 + 5 files changed, 335 insertions(+), 4 deletions(-) create mode 100644 sound/soc/intel/sst-atom-controls.c (limited to 'sound') diff --git a/sound/soc/intel/Makefile b/sound/soc/intel/Makefile index 7acbfc4..f841786 100644 --- a/sound/soc/intel/Makefile +++ b/sound/soc/intel/Makefile @@ -2,7 +2,8 @@ snd-soc-sst-dsp-objs := sst-dsp.o sst-firmware.o snd-soc-sst-acpi-objs := sst-acpi.o -snd-soc-sst-mfld-platform-objs := sst-mfld-platform-pcm.o sst-mfld-platform-compress.o +snd-soc-sst-mfld-platform-objs := sst-mfld-platform-pcm.o \ + sst-mfld-platform-compress.o sst-atom-controls.o snd-soc-mfld-machine-objs := mfld_machine.o obj-$(CONFIG_SND_SST_MFLD_PLATFORM) += snd-soc-sst-mfld-platform.o diff --git a/sound/soc/intel/sst-atom-controls.c b/sound/soc/intel/sst-atom-controls.c new file mode 100644 index 0000000..ace3c4a --- /dev/null +++ b/sound/soc/intel/sst-atom-controls.c @@ -0,0 +1,39 @@ +/* + * sst-atom-controls.c - Intel MID Platform driver DPCM ALSA controls for Mrfld + * + * Copyright (C) 2013-14 Intel Corp + * Author: Omair Mohammed Abdullah + * Vinod Koul + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * 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; version 2 of the License. + * + * 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. + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + */ +#define pr_fmt(fmt) KBUILD_MODNAME ": " fmt + +#include +#include +#include +#include "sst-mfld-platform.h" +#include "sst-atom-controls.h" + +int sst_dsp_init_v2_dpcm(struct snd_soc_platform *platform) +{ + int ret = 0; + struct sst_data *drv = snd_soc_platform_get_drvdata(platform); + + drv->byte_stream = devm_kzalloc(platform->dev, + SST_MAX_BIN_BYTES, GFP_KERNEL); + if (!drv->byte_stream) + return -ENOMEM; + + return ret; +} diff --git a/sound/soc/intel/sst-atom-controls.h b/sound/soc/intel/sst-atom-controls.h index 14063ab..8554889 100644 --- a/sound/soc/intel/sst-atom-controls.h +++ b/sound/soc/intel/sst-atom-controls.h @@ -1,4 +1,6 @@ /* + * sst-atom-controls.h - Intel MID Platform driver header file + * * Copyright (C) 2013-14 Intel Corp * Author: Ramesh Babu * Omair M Abdullah @@ -18,13 +20,293 @@ * */ -#ifndef __SST_CONTROLS_V2_H__ -#define __SST_CONTROLS_V2_H__ +#ifndef __SST_ATOM_CONTROLS_H__ +#define __SST_ATOM_CONTROLS_H__ enum { MERR_DPCM_AUDIO = 0, MERR_DPCM_COMPR, }; +/* define a bit for each mixer input */ +#define SST_MIX_IP(x) (x) + +#define SST_IP_CODEC0 SST_MIX_IP(2) +#define SST_IP_CODEC1 SST_MIX_IP(3) +#define SST_IP_LOOP0 SST_MIX_IP(4) +#define SST_IP_LOOP1 SST_MIX_IP(5) +#define SST_IP_LOOP2 SST_MIX_IP(6) +#define SST_IP_PROBE SST_MIX_IP(7) +#define SST_IP_VOIP SST_MIX_IP(12) +#define SST_IP_PCM0 SST_MIX_IP(13) +#define SST_IP_PCM1 SST_MIX_IP(14) +#define SST_IP_MEDIA0 SST_MIX_IP(17) +#define SST_IP_MEDIA1 SST_MIX_IP(18) +#define SST_IP_MEDIA2 SST_MIX_IP(19) +#define SST_IP_MEDIA3 SST_MIX_IP(20) + +#define SST_IP_LAST SST_IP_MEDIA3 + +#define SST_SWM_INPUT_COUNT (SST_IP_LAST + 1) +#define SST_CMD_SWM_MAX_INPUTS 6 + +#define SST_PATH_ID_SHIFT 8 +#define SST_DEFAULT_LOCATION_ID 0xFFFF +#define SST_DEFAULT_CELL_NBR 0xFF +#define SST_DEFAULT_MODULE_ID 0xFFFF + +/* + * Audio DSP Path Ids. Specified by the audio DSP FW + */ +enum sst_path_index { + SST_PATH_INDEX_CODEC_OUT0 = (0x02 << SST_PATH_ID_SHIFT), + SST_PATH_INDEX_CODEC_OUT1 = (0x03 << SST_PATH_ID_SHIFT), + + SST_PATH_INDEX_SPROT_LOOP_OUT = (0x04 << SST_PATH_ID_SHIFT), + SST_PATH_INDEX_MEDIA_LOOP1_OUT = (0x05 << SST_PATH_ID_SHIFT), + SST_PATH_INDEX_MEDIA_LOOP2_OUT = (0x06 << SST_PATH_ID_SHIFT), + + SST_PATH_INDEX_VOIP_OUT = (0x0C << SST_PATH_ID_SHIFT), + SST_PATH_INDEX_PCM0_OUT = (0x0D << SST_PATH_ID_SHIFT), + SST_PATH_INDEX_PCM1_OUT = (0x0E << SST_PATH_ID_SHIFT), + SST_PATH_INDEX_PCM2_OUT = (0x0F << SST_PATH_ID_SHIFT), + + SST_PATH_INDEX_MEDIA0_OUT = (0x12 << SST_PATH_ID_SHIFT), + SST_PATH_INDEX_MEDIA1_OUT = (0x13 << SST_PATH_ID_SHIFT), + + + /* Start of input paths */ + SST_PATH_INDEX_CODEC_IN0 = (0x82 << SST_PATH_ID_SHIFT), + SST_PATH_INDEX_CODEC_IN1 = (0x83 << SST_PATH_ID_SHIFT), + + SST_PATH_INDEX_SPROT_LOOP_IN = (0x84 << SST_PATH_ID_SHIFT), + SST_PATH_INDEX_MEDIA_LOOP1_IN = (0x85 << SST_PATH_ID_SHIFT), + SST_PATH_INDEX_MEDIA_LOOP2_IN = (0x86 << SST_PATH_ID_SHIFT), + + SST_PATH_INDEX_VOIP_IN = (0x8C << SST_PATH_ID_SHIFT), + + SST_PATH_INDEX_PCM0_IN = (0x8D << SST_PATH_ID_SHIFT), + SST_PATH_INDEX_PCM1_IN = (0x8E << SST_PATH_ID_SHIFT), + + SST_PATH_INDEX_MEDIA0_IN = (0x8F << SST_PATH_ID_SHIFT), + SST_PATH_INDEX_MEDIA1_IN = (0x90 << SST_PATH_ID_SHIFT), + SST_PATH_INDEX_MEDIA2_IN = (0x91 << SST_PATH_ID_SHIFT), + + SST_PATH_INDEX_MEDIA3_IN = (0x9C << SST_PATH_ID_SHIFT), + + SST_PATH_INDEX_RESERVED = (0xFF << SST_PATH_ID_SHIFT), +}; + +/* + * path IDs + */ +enum sst_swm_inputs { + SST_SWM_IN_CODEC0 = (SST_PATH_INDEX_CODEC_IN0 | SST_DEFAULT_CELL_NBR), + SST_SWM_IN_CODEC1 = (SST_PATH_INDEX_CODEC_IN1 | SST_DEFAULT_CELL_NBR), + SST_SWM_IN_SPROT_LOOP = (SST_PATH_INDEX_SPROT_LOOP_IN | SST_DEFAULT_CELL_NBR), + SST_SWM_IN_MEDIA_LOOP1 = (SST_PATH_INDEX_MEDIA_LOOP1_IN | SST_DEFAULT_CELL_NBR), + SST_SWM_IN_MEDIA_LOOP2 = (SST_PATH_INDEX_MEDIA_LOOP2_IN | SST_DEFAULT_CELL_NBR), + SST_SWM_IN_VOIP = (SST_PATH_INDEX_VOIP_IN | SST_DEFAULT_CELL_NBR), + SST_SWM_IN_PCM0 = (SST_PATH_INDEX_PCM0_IN | SST_DEFAULT_CELL_NBR), + SST_SWM_IN_PCM1 = (SST_PATH_INDEX_PCM1_IN | SST_DEFAULT_CELL_NBR), + SST_SWM_IN_MEDIA0 = (SST_PATH_INDEX_MEDIA0_IN | SST_DEFAULT_CELL_NBR), /* Part of Media Mixer */ + SST_SWM_IN_MEDIA1 = (SST_PATH_INDEX_MEDIA1_IN | SST_DEFAULT_CELL_NBR), /* Part of Media Mixer */ + SST_SWM_IN_MEDIA2 = (SST_PATH_INDEX_MEDIA2_IN | SST_DEFAULT_CELL_NBR), /* Part of Media Mixer */ + SST_SWM_IN_MEDIA3 = (SST_PATH_INDEX_MEDIA3_IN | SST_DEFAULT_CELL_NBR), /* Part of Media Mixer */ + SST_SWM_IN_END = (SST_PATH_INDEX_RESERVED | SST_DEFAULT_CELL_NBR) +}; + +/* + * path IDs + */ +enum sst_swm_outputs { + SST_SWM_OUT_CODEC0 = (SST_PATH_INDEX_CODEC_OUT0 | SST_DEFAULT_CELL_NBR), + SST_SWM_OUT_CODEC1 = (SST_PATH_INDEX_CODEC_OUT1 | SST_DEFAULT_CELL_NBR), + SST_SWM_OUT_SPROT_LOOP = (SST_PATH_INDEX_SPROT_LOOP_OUT | SST_DEFAULT_CELL_NBR), + SST_SWM_OUT_MEDIA_LOOP1 = (SST_PATH_INDEX_MEDIA_LOOP1_OUT | SST_DEFAULT_CELL_NBR), + SST_SWM_OUT_MEDIA_LOOP2 = (SST_PATH_INDEX_MEDIA_LOOP2_OUT | SST_DEFAULT_CELL_NBR), + SST_SWM_OUT_VOIP = (SST_PATH_INDEX_VOIP_OUT | SST_DEFAULT_CELL_NBR), + SST_SWM_OUT_PCM0 = (SST_PATH_INDEX_PCM0_OUT | SST_DEFAULT_CELL_NBR), + SST_SWM_OUT_PCM1 = (SST_PATH_INDEX_PCM1_OUT | SST_DEFAULT_CELL_NBR), + SST_SWM_OUT_PCM2 = (SST_PATH_INDEX_PCM2_OUT | SST_DEFAULT_CELL_NBR), + SST_SWM_OUT_MEDIA0 = (SST_PATH_INDEX_MEDIA0_OUT | SST_DEFAULT_CELL_NBR), /* Part of Media Mixer */ + SST_SWM_OUT_MEDIA1 = (SST_PATH_INDEX_MEDIA1_OUT | SST_DEFAULT_CELL_NBR), /* Part of Media Mixer */ + SST_SWM_OUT_END = (SST_PATH_INDEX_RESERVED | SST_DEFAULT_CELL_NBR), +}; + +enum sst_ipc_msg { + SST_IPC_IA_CMD = 1, + SST_IPC_IA_SET_PARAMS, + SST_IPC_IA_GET_PARAMS, +}; + +enum sst_cmd_type { + SST_CMD_BYTES_SET = 1, + SST_CMD_BYTES_GET = 2, +}; + +enum sst_task { + SST_TASK_SBA = 1, + SST_TASK_MMX, +}; + +enum sst_type { + SST_TYPE_CMD = 1, + SST_TYPE_PARAMS, +}; + +enum sst_flag { + SST_FLAG_BLOCKED = 1, + SST_FLAG_NONBLOCK, +}; + +/* + * Enumeration for indexing the gain cells in VB_SET_GAIN DSP command + */ +enum sst_gain_index { + /* GAIN IDs for SB task start here */ + SST_GAIN_INDEX_CODEC_OUT0, + SST_GAIN_INDEX_CODEC_OUT1, + SST_GAIN_INDEX_CODEC_IN0, + SST_GAIN_INDEX_CODEC_IN1, + + SST_GAIN_INDEX_SPROT_LOOP_OUT, + SST_GAIN_INDEX_MEDIA_LOOP1_OUT, + SST_GAIN_INDEX_MEDIA_LOOP2_OUT, + + SST_GAIN_INDEX_PCM0_IN_LEFT, + SST_GAIN_INDEX_PCM0_IN_RIGHT, + + SST_GAIN_INDEX_PCM1_OUT_LEFT, + SST_GAIN_INDEX_PCM1_OUT_RIGHT, + SST_GAIN_INDEX_PCM1_IN_LEFT, + SST_GAIN_INDEX_PCM1_IN_RIGHT, + SST_GAIN_INDEX_PCM2_OUT_LEFT, + + SST_GAIN_INDEX_PCM2_OUT_RIGHT, + SST_GAIN_INDEX_VOIP_OUT, + SST_GAIN_INDEX_VOIP_IN, + + /* Gain IDs for MMX task start here */ + SST_GAIN_INDEX_MEDIA0_IN_LEFT, + SST_GAIN_INDEX_MEDIA0_IN_RIGHT, + SST_GAIN_INDEX_MEDIA1_IN_LEFT, + SST_GAIN_INDEX_MEDIA1_IN_RIGHT, + + SST_GAIN_INDEX_MEDIA2_IN_LEFT, + SST_GAIN_INDEX_MEDIA2_IN_RIGHT, + + SST_GAIN_INDEX_GAIN_END +}; + +/* + * Audio DSP module IDs specified by FW spec + * TODO: Update with all modules + */ +enum sst_module_id { + SST_MODULE_ID_PCM = 0x0001, + SST_MODULE_ID_MP3 = 0x0002, + SST_MODULE_ID_MP24 = 0x0003, + SST_MODULE_ID_AAC = 0x0004, + SST_MODULE_ID_AACP = 0x0005, + SST_MODULE_ID_EAACP = 0x0006, + SST_MODULE_ID_WMA9 = 0x0007, + SST_MODULE_ID_WMA10 = 0x0008, + SST_MODULE_ID_WMA10P = 0x0009, + SST_MODULE_ID_RA = 0x000A, + SST_MODULE_ID_DDAC3 = 0x000B, + SST_MODULE_ID_TRUE_HD = 0x000C, + SST_MODULE_ID_HD_PLUS = 0x000D, + + SST_MODULE_ID_SRC = 0x0064, + SST_MODULE_ID_DOWNMIX = 0x0066, + SST_MODULE_ID_GAIN_CELL = 0x0067, + SST_MODULE_ID_SPROT = 0x006D, + SST_MODULE_ID_BASS_BOOST = 0x006E, + SST_MODULE_ID_STEREO_WDNG = 0x006F, + SST_MODULE_ID_AV_REMOVAL = 0x0070, + SST_MODULE_ID_MIC_EQ = 0x0071, + SST_MODULE_ID_SPL = 0x0072, + SST_MODULE_ID_ALGO_VTSV = 0x0073, + SST_MODULE_ID_NR = 0x0076, + SST_MODULE_ID_BWX = 0x0077, + SST_MODULE_ID_DRP = 0x0078, + SST_MODULE_ID_MDRP = 0x0079, + + SST_MODULE_ID_ANA = 0x007A, + SST_MODULE_ID_AEC = 0x007B, + SST_MODULE_ID_NR_SNS = 0x007C, + SST_MODULE_ID_SER = 0x007D, + SST_MODULE_ID_AGC = 0x007E, + + SST_MODULE_ID_CNI = 0x007F, + SST_MODULE_ID_CONTEXT_ALGO_AWARE = 0x0080, + SST_MODULE_ID_FIR_24 = 0x0081, + SST_MODULE_ID_IIR_24 = 0x0082, + + SST_MODULE_ID_ASRC = 0x0083, + SST_MODULE_ID_TONE_GEN = 0x0084, + SST_MODULE_ID_BMF = 0x0086, + SST_MODULE_ID_EDL = 0x0087, + SST_MODULE_ID_GLC = 0x0088, + + SST_MODULE_ID_FIR_16 = 0x0089, + SST_MODULE_ID_IIR_16 = 0x008A, + SST_MODULE_ID_DNR = 0x008B, + + SST_MODULE_ID_VIRTUALIZER = 0x008C, + SST_MODULE_ID_VISUALIZATION = 0x008D, + SST_MODULE_ID_LOUDNESS_OPTIMIZER = 0x008E, + SST_MODULE_ID_REVERBERATION = 0x008F, + + SST_MODULE_ID_CNI_TX = 0x0090, + SST_MODULE_ID_REF_LINE = 0x0091, + SST_MODULE_ID_VOLUME = 0x0092, + SST_MODULE_ID_FILT_DCR = 0x0094, + SST_MODULE_ID_SLV = 0x009A, + SST_MODULE_ID_NLF = 0x009B, + SST_MODULE_ID_TNR = 0x009C, + SST_MODULE_ID_WNR = 0x009D, + + SST_MODULE_ID_LOG = 0xFF00, + + SST_MODULE_ID_TASK = 0xFFFF, +}; + +enum sst_cmd { + SBA_IDLE = 14, + SBA_VB_SET_SPEECH_PATH = 26, + MMX_SET_GAIN = 33, + SBA_VB_SET_GAIN = 33, + FBA_VB_RX_CNI = 35, + MMX_SET_GAIN_TIMECONST = 36, + SBA_VB_SET_TIMECONST = 36, + SBA_VB_START = 85, + SBA_SET_SWM = 114, + SBA_SET_MDRP = 116, + SBA_HW_SET_SSP = 117, + SBA_SET_MEDIA_LOOP_MAP = 118, + SBA_SET_MEDIA_PATH = 119, + MMX_SET_MEDIA_PATH = 119, + SBA_VB_LPRO = 126, + SBA_VB_SET_FIR = 128, + SBA_VB_SET_IIR = 129, + SBA_SET_SSP_SLOT_MAP = 130, +}; + +enum sst_dsp_switch { + SST_SWITCH_OFF = 0, + SST_SWITCH_ON = 3, +}; + +enum sst_path_switch { + SST_PATH_OFF = 0, + SST_PATH_ON = 1, +}; + +enum sst_swm_state { + SST_SWM_OFF = 0, + SST_SWM_ON = 3, +}; #endif diff --git a/sound/soc/intel/sst-mfld-platform-pcm.c b/sound/soc/intel/sst-mfld-platform-pcm.c index a89ff7e..8e1e9bc 100644 --- a/sound/soc/intel/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/sst-mfld-platform-pcm.c @@ -550,7 +550,13 @@ static int sst_pcm_new(struct snd_soc_pcm_runtime *rtd) return retval; } -static struct snd_soc_platform_driver sst_soc_platform_drv = { +static int sst_soc_probe(struct snd_soc_platform *platform) +{ + return sst_dsp_init_v2_dpcm(platform); +} + +static struct snd_soc_platform_driver sst_soc_platform_drv = { + .probe = sst_soc_probe, .ops = &sst_platform_ops, .compr_ops = &sst_platform_compr_ops, .pcm_new = sst_pcm_new, diff --git a/sound/soc/intel/sst-mfld-platform.h b/sound/soc/intel/sst-mfld-platform.h index d4c28b8..faaba10 100644 --- a/sound/soc/intel/sst-mfld-platform.h +++ b/sound/soc/intel/sst-mfld-platform.h @@ -143,6 +143,8 @@ struct sst_device { }; struct sst_data; + +int sst_dsp_init_v2_dpcm(struct snd_soc_platform *platform); void sst_set_stream_status(struct sst_runtime_stream *stream, int state); int sst_fill_stream_params(void *substream, const struct sst_data *ctx, struct snd_sst_params *str_params, bool is_compress); @@ -157,6 +159,7 @@ struct sst_algo_int_control_v2 { struct sst_data { struct platform_device *pdev; struct sst_platform_data *pdata; + char *byte_stream; struct mutex lock; }; int sst_register_dsp(struct sst_device *sst); -- cgit v1.1 From a493b6a637e9d8e828d7ed4be4bdf24dfd1f9250 Mon Sep 17 00:00:00 2001 From: Julia Lawall Date: Fri, 8 Aug 2014 12:07:49 +0200 Subject: ASoC: rsnd: delete unneeded test before of_node_put MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Of_node_put supports NULL as its argument, so the initial test is not necessary. Suggested by Uwe Kleine-König. The semantic patch that fixes this problem is as follows: (http://coccinelle.lip6.fr/) // @@ expression e; @@ -if (e) of_node_put(e); // Signed-off-by: Julia Lawall Signed-off-by: Mark Brown --- sound/soc/sh/rcar/core.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 19f7896..1922ec5 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -798,10 +798,8 @@ if (name##_node) { \ mod_parse(src); mod_parse(dvc); - if (playback) - of_node_put(playback); - if (capture) - of_node_put(capture); + of_node_put(playback); + of_node_put(capture); } dai_i++; -- cgit v1.1 From 0d985b1c76623747107dbab1052044d6bac3866d Mon Sep 17 00:00:00 2001 From: Rongjun Ying Date: Wed, 13 Aug 2014 16:31:40 +0800 Subject: ASoC: sirf: usp: Add bitclock inversion support Signed-off-by: Rongjun Ying Signed-off-by: Mark Brown --- sound/soc/sirf/sirf-usp.c | 24 +++++++++++++++++++++++- 1 file changed, 23 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/sirf/sirf-usp.c b/sound/soc/sirf/sirf-usp.c index 3a73037..186dc7f 100644 --- a/sound/soc/sirf/sirf-usp.c +++ b/sound/soc/sirf/sirf-usp.c @@ -100,6 +100,16 @@ static int sirf_usp_pcm_set_dai_fmt(struct snd_soc_dai *dai, return -EINVAL; } + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_NF: + usp->daifmt_format |= (fmt & SND_SOC_DAIFMT_INV_MASK); + break; + default: + return -EINVAL; + } + return 0; } @@ -177,7 +187,7 @@ static int sirf_usp_pcm_hw_params(struct snd_pcm_substream *substream, shifter_len = data_len; - switch (usp->daifmt_format) { + switch (usp->daifmt_format & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: regmap_update_bits(usp->regmap, USP_RX_FRAME_CTRL, USP_I2S_SYNC_CHG, USP_I2S_SYNC_CHG); @@ -193,6 +203,18 @@ static int sirf_usp_pcm_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } + switch (usp->daifmt_format & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_NF: + regmap_update_bits(usp->regmap, USP_MODE1, + USP_RXD_ACT_EDGE_FALLING | USP_TXD_ACT_EDGE_FALLING, + USP_RXD_ACT_EDGE_FALLING); + break; + default: + return -EINVAL; + } + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) regmap_update_bits(usp->regmap, USP_TX_FRAME_CTRL, USP_TXC_DATA_LEN_MASK | USP_TXC_FRAME_LEN_MASK -- cgit v1.1 From a7a8e994ddd004fbabfcf04c26c204297b5f826d Mon Sep 17 00:00:00 2001 From: Dan Murphy Date: Fri, 1 Aug 2014 10:57:04 -0500 Subject: ASoC: tas2552: Add DAPM calls for amp and PLL Add DAPM calls to enable/disable the Class D amp. Also add a DAPM call to turn off the PLL upon the stream completing. Signed-off-by: Dan Murphy Signed-off-by: Mark Brown --- sound/soc/codecs/tas2552.c | 68 ++++++++++++++++++++++++++++++++-------------- 1 file changed, 48 insertions(+), 20 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tas2552.c b/sound/soc/codecs/tas2552.c index 23b3296..1ed57a7 100644 --- a/sound/soc/codecs/tas2552.c +++ b/sound/soc/codecs/tas2552.c @@ -78,6 +78,43 @@ struct tas2552_data { unsigned int mclk; }; +/* Input mux controls */ +static const char *tas2552_input_texts[] = { + "Digital", "Analog" +}; + +static SOC_ENUM_SINGLE_DECL(tas2552_input_mux_enum, TAS2552_CFG_3, 7, + tas2552_input_texts); + +static const struct snd_kcontrol_new tas2552_input_mux_control[] = { + SOC_DAPM_ENUM("Input selection", tas2552_input_mux_enum) +}; + +static const struct snd_soc_dapm_widget tas2552_dapm_widgets[] = +{ + SND_SOC_DAPM_INPUT("IN"), + + /* MUX Controls */ + SND_SOC_DAPM_MUX("Input selection", SND_SOC_NOPM, 0, 0, + tas2552_input_mux_control), + + SND_SOC_DAPM_AIF_IN("DAC IN", "DAC Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("DAC", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_OUT_DRV("ClassD", TAS2552_CFG_2, 7, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("PLL", TAS2552_CFG_2, 3, 0, NULL, 0), + + SND_SOC_DAPM_OUTPUT("OUT") +}; + +static const struct snd_soc_dapm_route tas2552_audio_map[] = { + {"DAC", NULL, "DAC IN"}, + {"Input selection", "Digital", "DAC"}, + {"Input selection", "Analog", "IN"}, + {"ClassD", NULL, "Input selection"}, + {"OUT", NULL, "ClassD"}, + {"ClassD", NULL, "PLL"}, +}; + static void tas2552_sw_shutdown(struct tas2552_data *tas_data, int sw_shutdown) { u8 cfg1_reg; @@ -101,10 +138,6 @@ static int tas2552_hw_params(struct snd_pcm_substream *substream, int d; u8 p, j; - /* Turn on Class D amplifier */ - snd_soc_update_bits(codec, TAS2552_CFG_2, TAS2552_CLASSD_EN_MASK, - TAS2552_CLASSD_EN); - if (!tas2552->mclk) return -EINVAL; @@ -147,9 +180,6 @@ static int tas2552_hw_params(struct snd_pcm_substream *substream, } - snd_soc_update_bits(codec, TAS2552_CFG_2, TAS2552_PLL_ENABLE, - TAS2552_PLL_ENABLE); - return 0; } @@ -269,19 +299,10 @@ static const struct dev_pm_ops tas2552_pm = { NULL) }; -static void tas2552_shutdown(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct snd_soc_codec *codec = dai->codec; - - snd_soc_update_bits(codec, TAS2552_CFG_2, TAS2552_PLL_ENABLE, 0); -} - static struct snd_soc_dai_ops tas2552_speaker_dai_ops = { .hw_params = tas2552_hw_params, .set_sysclk = tas2552_set_dai_sysclk, .set_fmt = tas2552_set_dai_fmt, - .shutdown = tas2552_shutdown, .digital_mute = tas2552_mute, }; @@ -294,7 +315,7 @@ static struct snd_soc_dai_driver tas2552_dai[] = { { .name = "tas2552-amplifier", .playback = { - .stream_name = "Speaker", + .stream_name = "Playback", .channels_min = 2, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_192000, @@ -312,6 +333,7 @@ static DECLARE_TLV_DB_SCALE(dac_tlv, -7, 100, 24); static const struct snd_kcontrol_new tas2552_snd_controls[] = { SOC_SINGLE_TLV("Speaker Driver Playback Volume", TAS2552_PGA_GAIN, 0, 0x1f, 1, dac_tlv), + SOC_DAPM_SINGLE("Playback AMP", SND_SOC_NOPM, 0, 1, 0), }; static const struct reg_default tas2552_init_regs[] = { @@ -321,6 +343,7 @@ static const struct reg_default tas2552_init_regs[] = { static int tas2552_codec_probe(struct snd_soc_codec *codec) { struct tas2552_data *tas2552 = snd_soc_codec_get_drvdata(codec); + struct snd_soc_dapm_context *dapm = &codec->dapm; int ret; tas2552->codec = codec; @@ -362,9 +385,14 @@ static int tas2552_codec_probe(struct snd_soc_codec *codec) goto patch_fail; } - snd_soc_write(codec, TAS2552_CFG_2, TAS2552_CLASSD_EN | - TAS2552_BOOST_EN | TAS2552_APT_EN | - TAS2552_LIM_EN); + snd_soc_write(codec, TAS2552_CFG_2, TAS2552_BOOST_EN | + TAS2552_APT_EN | TAS2552_LIM_EN); + + snd_soc_dapm_new_controls(dapm, tas2552_dapm_widgets, + ARRAY_SIZE(tas2552_dapm_widgets)); + snd_soc_dapm_add_routes(dapm, tas2552_audio_map, + ARRAY_SIZE(tas2552_audio_map)); + return 0; patch_fail: -- cgit v1.1 From dfe8f1f3f22f9922e773ae64f5621f290cb26023 Mon Sep 17 00:00:00 2001 From: Nikesh Oswal Date: Wed, 13 Aug 2014 10:05:45 +0100 Subject: ASoC: wm8994: Demux the microphone detection IRQ Current code only allows direct routing of the WM8994 microphone detection signal to a GPIO this change adds support to demux the interrupt from the main interrupt line of the codec. Signed-off-by: Nikesh Oswal Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 18 ++++++++++++------ 1 file changed, 12 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 6cc0566..1fcb9f3 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -4082,17 +4082,23 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) switch (control->type) { case WM8994: - if (wm8994->micdet_irq) { + if (wm8994->micdet_irq) ret = request_threaded_irq(wm8994->micdet_irq, NULL, wm8994_mic_irq, IRQF_TRIGGER_RISING, "Mic1 detect", wm8994); - if (ret != 0) - dev_warn(codec->dev, - "Failed to request Mic1 detect IRQ: %d\n", - ret); - } + else + ret = wm8994_request_irq(wm8994->wm8994, + WM8994_IRQ_MIC1_DET, + wm8994_mic_irq, "Mic 1 detect", + wm8994); + + if (ret != 0) + dev_warn(codec->dev, + "Failed to request Mic1 detect IRQ: %d\n", + ret); + ret = wm8994_request_irq(wm8994->wm8994, WM8994_IRQ_MIC1_SHRT, -- cgit v1.1 From 567e4f98922ce5542f8c2aa469a0c6ddf182b6ea Mon Sep 17 00:00:00 2001 From: Sean Cross Date: Thu, 31 Jul 2014 10:43:36 +0800 Subject: ASoC: add es8328 codec driver Add a codec driver for the Everest ES8328. It supports two separate audio outputs and two separate audio inputs. Signed-off-by: Sean Cross Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 13 + sound/soc/codecs/Makefile | 6 + sound/soc/codecs/es8328-i2c.c | 60 ++++ sound/soc/codecs/es8328-spi.c | 49 +++ sound/soc/codecs/es8328.c | 756 ++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/es8328.h | 314 ++++++++++++++++++ 6 files changed, 1198 insertions(+) create mode 100644 sound/soc/codecs/es8328-i2c.c create mode 100644 sound/soc/codecs/es8328-spi.c create mode 100644 sound/soc/codecs/es8328.c create mode 100644 sound/soc/codecs/es8328.h (limited to 'sound') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 8838838e..8bca634 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -57,6 +57,8 @@ config SND_SOC_ALL_CODECS select SND_SOC_DA732X if I2C select SND_SOC_DA9055 if I2C select SND_SOC_BT_SCO + select SND_SOC_ES8328_SPI if SPI_MASTER + select SND_SOC_ES8328_I2C if I2C select SND_SOC_ISABELLE if I2C select SND_SOC_JZ4740_CODEC select SND_SOC_LM4857 if I2C @@ -405,6 +407,17 @@ config SND_SOC_DMIC config SND_SOC_HDMI_CODEC tristate "HDMI stub CODEC" +config SND_SOC_ES8328 + tristate "Everest Semi ES8328 CODEC" + +config SND_SOC_ES8328_I2C + tristate + select SND_SOC_ES8328 + +config SND_SOC_ES8328_SPI + tristate + select SND_SOC_ES8328 + config SND_SOC_ISABELLE tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 20afe0f..31a8283 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -49,6 +49,9 @@ snd-soc-da732x-objs := da732x.o snd-soc-da9055-objs := da9055.o snd-soc-bt-sco-objs := bt-sco.o snd-soc-dmic-objs := dmic.o +snd-soc-es8328-objs := es8328.o +snd-soc-es8328-i2c-objs := es8328-i2c.o +snd-soc-es8328-spi-objs := es8328-spi.o snd-soc-isabelle-objs := isabelle.o snd-soc-jz4740-codec-objs := jz4740.o snd-soc-l3-objs := l3.o @@ -220,6 +223,9 @@ obj-$(CONFIG_SND_SOC_DA732X) += snd-soc-da732x.o obj-$(CONFIG_SND_SOC_DA9055) += snd-soc-da9055.o obj-$(CONFIG_SND_SOC_BT_SCO) += snd-soc-bt-sco.o obj-$(CONFIG_SND_SOC_DMIC) += snd-soc-dmic.o +obj-$(CONFIG_SND_SOC_ES8328) += snd-soc-es8328.o +obj-$(CONFIG_SND_SOC_ES8328_I2C)+= snd-soc-es8328-i2c.o +obj-$(CONFIG_SND_SOC_ES8328_SPI)+= snd-soc-es8328-spi.o obj-$(CONFIG_SND_SOC_ISABELLE) += snd-soc-isabelle.o obj-$(CONFIG_SND_SOC_JZ4740_CODEC) += snd-soc-jz4740-codec.o obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o diff --git a/sound/soc/codecs/es8328-i2c.c b/sound/soc/codecs/es8328-i2c.c new file mode 100644 index 0000000..aae410d --- /dev/null +++ b/sound/soc/codecs/es8328-i2c.c @@ -0,0 +1,60 @@ +/* + * es8328-i2c.c -- ES8328 ALSA SoC I2C Audio driver + * + * Copyright 2014 Sutajio Ko-Usagi PTE LTD + * + * Author: Sean Cross + * + * 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 "es8328.h" + +static const struct i2c_device_id es8328_id[] = { + { "everest,es8328", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, es8328_id); + +static const struct of_device_id es8328_of_match[] = { + { .compatible = "everest,es8328", }, + { } +}; +MODULE_DEVICE_TABLE(of, es8328_of_match); + +static int es8328_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + return es8328_probe(&i2c->dev, + devm_regmap_init_i2c(i2c, &es8328_regmap_config)); +} + +static int es8328_i2c_remove(struct i2c_client *i2c) +{ + snd_soc_unregister_codec(&i2c->dev); + return 0; +} + +static struct i2c_driver es8328_i2c_driver = { + .driver = { + .name = "es8328", + .of_match_table = es8328_of_match, + }, + .probe = es8328_i2c_probe, + .remove = es8328_i2c_remove, + .id_table = es8328_id, +}; + +module_i2c_driver(es8328_i2c_driver); + +MODULE_DESCRIPTION("ASoC ES8328 audio CODEC I2C driver"); +MODULE_AUTHOR("Sean Cross "); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/es8328-spi.c b/sound/soc/codecs/es8328-spi.c new file mode 100644 index 0000000..8fbd935 --- /dev/null +++ b/sound/soc/codecs/es8328-spi.c @@ -0,0 +1,49 @@ +/* + * es8328.c -- ES8328 ALSA SoC SPI Audio driver + * + * Copyright 2014 Sutajio Ko-Usagi PTE LTD + * + * Author: Sean Cross + * + * 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 "es8328.h" + +static const struct of_device_id es8328_of_match[] = { + { .compatible = "everest,es8328", }, + { } +}; +MODULE_DEVICE_TABLE(of, es8328_of_match); + +static int es8328_spi_probe(struct spi_device *spi) +{ + return es8328_probe(&spi->dev, + devm_regmap_init_spi(spi, &es8328_regmap_config)); +} + +static int es8328_spi_remove(struct spi_device *spi) +{ + snd_soc_unregister_codec(&spi->dev); + return 0; +} + +static struct spi_driver es8328_spi_driver = { + .driver = { + .name = "es8328", + .of_match_table = es8328_of_match, + }, + .probe = es8328_spi_probe, + .remove = es8328_spi_remove, +}; + +module_spi_driver(es8328_spi_driver); +MODULE_DESCRIPTION("ASoC ES8328 audio CODEC SPI driver"); +MODULE_AUTHOR("Sean Cross "); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/es8328.c b/sound/soc/codecs/es8328.c new file mode 100644 index 0000000..7a9f65a --- /dev/null +++ b/sound/soc/codecs/es8328.c @@ -0,0 +1,756 @@ +/* + * es8328.c -- ES8328 ALSA SoC Audio driver + * + * Copyright 2014 Sutajio Ko-Usagi PTE LTD + * + * Author: Sean Cross + * + * 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 "es8328.h" + +#define ES8328_SYSCLK_RATE_1X 11289600 +#define ES8328_SYSCLK_RATE_2X 22579200 + +/* Run the codec at 22.5792 or 11.2896 MHz to support these rates */ +static struct { + int rate; + u8 ratio; +} mclk_ratios[] = { + { 8000, 9 }, + {11025, 7 }, + {22050, 4 }, + {44100, 2 }, +}; + +/* regulator supplies for sgtl5000, VDDD is an optional external supply */ +enum sgtl5000_regulator_supplies { + DVDD, + AVDD, + PVDD, + HPVDD, + ES8328_SUPPLY_NUM +}; + +/* vddd is optional supply */ +static const char * const supply_names[ES8328_SUPPLY_NUM] = { + "DVDD", + "AVDD", + "PVDD", + "HPVDD", +}; + +#define ES8328_RATES (SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_22050 | \ + SNDRV_PCM_RATE_11025) +#define ES8328_FORMATS (SNDRV_PCM_FMTBIT_S16_LE) + +struct es8328_priv { + struct regmap *regmap; + struct clk *clk; + int playback_fs; + bool deemph; + struct regulator_bulk_data supplies[ES8328_SUPPLY_NUM]; +}; + +/* + * ES8328 Controls + */ + +static const char * const adcpol_txt[] = {"Normal", "L Invert", "R Invert", + "L + R Invert"}; +static SOC_ENUM_SINGLE_DECL(adcpol, + ES8328_ADCCONTROL6, 6, adcpol_txt); + +static const DECLARE_TLV_DB_SCALE(play_tlv, -3000, 100, 0); +static const DECLARE_TLV_DB_SCALE(dac_adc_tlv, -9600, 50, 0); +static const DECLARE_TLV_DB_SCALE(pga_tlv, 0, 300, 0); +static const DECLARE_TLV_DB_SCALE(bypass_tlv, -1500, 300, 0); +static const DECLARE_TLV_DB_SCALE(mic_tlv, 0, 300, 0); + +static const int deemph_settings[] = { 0, 32000, 44100, 48000 }; + +static int es8328_set_deemph(struct snd_soc_codec *codec) +{ + struct es8328_priv *es8328 = snd_soc_codec_get_drvdata(codec); + int val, i, best; + + /* + * If we're using deemphasis select the nearest available sample + * rate. + */ + if (es8328->deemph) { + best = 1; + for (i = 2; i < ARRAY_SIZE(deemph_settings); i++) { + if (abs(deemph_settings[i] - es8328->playback_fs) < + abs(deemph_settings[best] - es8328->playback_fs)) + best = i; + } + + val = best << 1; + } else { + val = 0; + } + + dev_dbg(codec->dev, "Set deemphasis %d\n", val); + + return snd_soc_update_bits(codec, ES8328_DACCONTROL6, 0x6, val); +} + +static int es8328_get_deemph(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct es8328_priv *es8328 = snd_soc_codec_get_drvdata(codec); + + ucontrol->value.enumerated.item[0] = es8328->deemph; + return 0; +} + +static int es8328_put_deemph(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct es8328_priv *es8328 = snd_soc_codec_get_drvdata(codec); + int deemph = ucontrol->value.enumerated.item[0]; + int ret; + + if (deemph > 1) + return -EINVAL; + + ret = es8328_set_deemph(codec); + if (ret < 0) + return ret; + + es8328->deemph = deemph; + + return 0; +} + + + +static const struct snd_kcontrol_new es8328_snd_controls[] = { + SOC_DOUBLE_R_TLV("Capture Digital Volume", + ES8328_ADCCONTROL8, ES8328_ADCCONTROL9, + 0, 0xc0, 1, dac_adc_tlv), + SOC_SINGLE("Capture ZC Switch", ES8328_ADCCONTROL7, 6, 1, 0), + + SOC_SINGLE_BOOL_EXT("DAC Deemphasis Switch", 0, + es8328_get_deemph, es8328_put_deemph), + + SOC_ENUM("Capture Polarity", adcpol), + + SOC_SINGLE_TLV("Left Mixer Left Bypass Volume", + ES8328_DACCONTROL17, 3, 7, 1, bypass_tlv), + SOC_SINGLE_TLV("Left Mixer Right Bypass Volume", + ES8328_DACCONTROL19, 3, 7, 1, bypass_tlv), + SOC_SINGLE_TLV("Right Mixer Left Bypass Volume", + ES8328_DACCONTROL18, 3, 7, 1, bypass_tlv), + SOC_SINGLE_TLV("Right Mixer Right Bypass Volume", + ES8328_DACCONTROL20, 3, 7, 1, bypass_tlv), + + SOC_DOUBLE_R_TLV("PCM Volume", + ES8328_LDACVOL, ES8328_RDACVOL, + 0, ES8328_DACVOL_MAX, 1, dac_adc_tlv), + + SOC_DOUBLE_R_TLV("Output 1 Playback Volume", + ES8328_LOUT1VOL, ES8328_ROUT1VOL, + 0, ES8328_OUT1VOL_MAX, 0, play_tlv), + + SOC_DOUBLE_R_TLV("Output 2 Playback Volume", + ES8328_LOUT2VOL, ES8328_ROUT2VOL, + 0, ES8328_OUT2VOL_MAX, 0, play_tlv), + + SOC_DOUBLE_TLV("Mic PGA Volume", ES8328_ADCCONTROL1, + 4, 0, 8, 0, mic_tlv), +}; + +/* + * DAPM Controls + */ + +static const char * const es8328_line_texts[] = { + "Line 1", "Line 2", "PGA", "Differential"}; + +static const struct soc_enum es8328_lline_enum = + SOC_ENUM_SINGLE(ES8328_DACCONTROL16, 3, + ARRAY_SIZE(es8328_line_texts), + es8328_line_texts); +static const struct snd_kcontrol_new es8328_left_line_controls = + SOC_DAPM_ENUM("Route", es8328_lline_enum); + +static const struct soc_enum es8328_rline_enum = + SOC_ENUM_SINGLE(ES8328_DACCONTROL16, 0, + ARRAY_SIZE(es8328_line_texts), + es8328_line_texts); +static const struct snd_kcontrol_new es8328_right_line_controls = + SOC_DAPM_ENUM("Route", es8328_lline_enum); + +/* Left Mixer */ +static const struct snd_kcontrol_new es8328_left_mixer_controls[] = { + SOC_DAPM_SINGLE("Playback Switch", ES8328_DACCONTROL17, 8, 1, 0), + SOC_DAPM_SINGLE("Left Bypass Switch", ES8328_DACCONTROL17, 7, 1, 0), + SOC_DAPM_SINGLE("Right Playback Switch", ES8328_DACCONTROL18, 8, 1, 0), + SOC_DAPM_SINGLE("Right Bypass Switch", ES8328_DACCONTROL18, 7, 1, 0), +}; + +/* Right Mixer */ +static const struct snd_kcontrol_new es8328_right_mixer_controls[] = { + SOC_DAPM_SINGLE("Left Playback Switch", ES8328_DACCONTROL19, 8, 1, 0), + SOC_DAPM_SINGLE("Left Bypass Switch", ES8328_DACCONTROL19, 7, 1, 0), + SOC_DAPM_SINGLE("Playback Switch", ES8328_DACCONTROL20, 8, 1, 0), + SOC_DAPM_SINGLE("Right Bypass Switch", ES8328_DACCONTROL20, 7, 1, 0), +}; + +static const char * const es8328_pga_sel[] = { + "Line 1", "Line 2", "Line 3", "Differential"}; + +/* Left PGA Mux */ +static const struct soc_enum es8328_lpga_enum = + SOC_ENUM_SINGLE(ES8328_ADCCONTROL2, 6, + ARRAY_SIZE(es8328_pga_sel), + es8328_pga_sel); +static const struct snd_kcontrol_new es8328_left_pga_controls = + SOC_DAPM_ENUM("Route", es8328_lpga_enum); + +/* Right PGA Mux */ +static const struct soc_enum es8328_rpga_enum = + SOC_ENUM_SINGLE(ES8328_ADCCONTROL2, 4, + ARRAY_SIZE(es8328_pga_sel), + es8328_pga_sel); +static const struct snd_kcontrol_new es8328_right_pga_controls = + SOC_DAPM_ENUM("Route", es8328_rpga_enum); + +/* Differential Mux */ +static const char * const es8328_diff_sel[] = {"Line 1", "Line 2"}; +static SOC_ENUM_SINGLE_DECL(diffmux, + ES8328_ADCCONTROL3, 7, es8328_diff_sel); +static const struct snd_kcontrol_new es8328_diffmux_controls = + SOC_DAPM_ENUM("Route", diffmux); + +/* Mono ADC Mux */ +static const char * const es8328_mono_mux[] = {"Stereo", "Mono (Left)", + "Mono (Right)", "Digital Mono"}; +static SOC_ENUM_SINGLE_DECL(monomux, + ES8328_ADCCONTROL3, 3, es8328_mono_mux); +static const struct snd_kcontrol_new es8328_monomux_controls = + SOC_DAPM_ENUM("Route", monomux); + +static const struct snd_soc_dapm_widget es8328_dapm_widgets[] = { + SND_SOC_DAPM_MUX("Differential Mux", SND_SOC_NOPM, 0, 0, + &es8328_diffmux_controls), + SND_SOC_DAPM_MUX("Left ADC Mux", SND_SOC_NOPM, 0, 0, + &es8328_monomux_controls), + SND_SOC_DAPM_MUX("Right ADC Mux", SND_SOC_NOPM, 0, 0, + &es8328_monomux_controls), + + SND_SOC_DAPM_MUX("Left PGA Mux", ES8328_ADCPOWER, + ES8328_ADCPOWER_AINL_OFF, 1, + &es8328_left_pga_controls), + SND_SOC_DAPM_MUX("Right PGA Mux", ES8328_ADCPOWER, + ES8328_ADCPOWER_AINR_OFF, 1, + &es8328_right_pga_controls), + + SND_SOC_DAPM_MUX("Left Line Mux", SND_SOC_NOPM, 0, 0, + &es8328_left_line_controls), + SND_SOC_DAPM_MUX("Right Line Mux", SND_SOC_NOPM, 0, 0, + &es8328_right_line_controls), + + SND_SOC_DAPM_ADC("Right ADC", "Right Capture", ES8328_ADCPOWER, + ES8328_ADCPOWER_ADCR_OFF, 1), + SND_SOC_DAPM_ADC("Left ADC", "Left Capture", ES8328_ADCPOWER, + ES8328_ADCPOWER_ADCL_OFF, 1), + + SND_SOC_DAPM_SUPPLY("Mic Bias", ES8328_ADCPOWER, + ES8328_ADCPOWER_MIC_BIAS_OFF, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("Mic Bias Gen", ES8328_ADCPOWER, + ES8328_ADCPOWER_ADC_BIAS_GEN_OFF, 1, NULL, 0), + + SND_SOC_DAPM_SUPPLY("DAC STM", ES8328_CHIPPOWER, + ES8328_CHIPPOWER_DACSTM_RESET, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC STM", ES8328_CHIPPOWER, + ES8328_CHIPPOWER_ADCSTM_RESET, 1, NULL, 0), + + SND_SOC_DAPM_SUPPLY("DAC DIG", ES8328_CHIPPOWER, + ES8328_CHIPPOWER_DACDIG_OFF, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC DIG", ES8328_CHIPPOWER, + ES8328_CHIPPOWER_ADCDIG_OFF, 1, NULL, 0), + + SND_SOC_DAPM_SUPPLY("DAC DLL", ES8328_CHIPPOWER, + ES8328_CHIPPOWER_DACDLL_OFF, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC DLL", ES8328_CHIPPOWER, + ES8328_CHIPPOWER_ADCDLL_OFF, 1, NULL, 0), + + SND_SOC_DAPM_SUPPLY("ADC Vref", ES8328_CHIPPOWER, + ES8328_CHIPPOWER_ADCVREF_OFF, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("DAC Vref", ES8328_CHIPPOWER, + ES8328_CHIPPOWER_DACVREF_OFF, 1, NULL, 0), + + SND_SOC_DAPM_DAC("Right DAC", "Right Playback", ES8328_DACPOWER, + ES8328_DACPOWER_RDAC_OFF, 1), + SND_SOC_DAPM_DAC("Left DAC", "Left Playback", ES8328_DACPOWER, + ES8328_DACPOWER_LDAC_OFF, 1), + + SND_SOC_DAPM_MIXER("Left Mixer", SND_SOC_NOPM, 0, 0, + &es8328_left_mixer_controls[0], + ARRAY_SIZE(es8328_left_mixer_controls)), + SND_SOC_DAPM_MIXER("Right Mixer", SND_SOC_NOPM, 0, 0, + &es8328_right_mixer_controls[0], + ARRAY_SIZE(es8328_right_mixer_controls)), + + SND_SOC_DAPM_PGA("Right Out 2", ES8328_DACPOWER, + ES8328_DACPOWER_ROUT2_ON, 0, NULL, 0), + SND_SOC_DAPM_PGA("Left Out 2", ES8328_DACPOWER, + ES8328_DACPOWER_LOUT2_ON, 0, NULL, 0), + SND_SOC_DAPM_PGA("Right Out 1", ES8328_DACPOWER, + ES8328_DACPOWER_ROUT1_ON, 0, NULL, 0), + SND_SOC_DAPM_PGA("Left Out 1", ES8328_DACPOWER, + ES8328_DACPOWER_LOUT1_ON, 0, NULL, 0), + + SND_SOC_DAPM_OUTPUT("LOUT1"), + SND_SOC_DAPM_OUTPUT("ROUT1"), + SND_SOC_DAPM_OUTPUT("LOUT2"), + SND_SOC_DAPM_OUTPUT("ROUT2"), + + SND_SOC_DAPM_INPUT("LINPUT1"), + SND_SOC_DAPM_INPUT("LINPUT2"), + SND_SOC_DAPM_INPUT("RINPUT1"), + SND_SOC_DAPM_INPUT("RINPUT2"), +}; + +static const struct snd_soc_dapm_route es8328_dapm_routes[] = { + + { "Left Line Mux", "Line 1", "LINPUT1" }, + { "Left Line Mux", "Line 2", "LINPUT2" }, + { "Left Line Mux", "PGA", "Left PGA Mux" }, + { "Left Line Mux", "Differential", "Differential Mux" }, + + { "Right Line Mux", "Line 1", "RINPUT1" }, + { "Right Line Mux", "Line 2", "RINPUT2" }, + { "Right Line Mux", "PGA", "Right PGA Mux" }, + { "Right Line Mux", "Differential", "Differential Mux" }, + + { "Left PGA Mux", "Line 1", "LINPUT1" }, + { "Left PGA Mux", "Line 2", "LINPUT2" }, + { "Left PGA Mux", "Differential", "Differential Mux" }, + + { "Right PGA Mux", "Line 1", "RINPUT1" }, + { "Right PGA Mux", "Line 2", "RINPUT2" }, + { "Right PGA Mux", "Differential", "Differential Mux" }, + + { "Differential Mux", "Line 1", "LINPUT1" }, + { "Differential Mux", "Line 1", "RINPUT1" }, + { "Differential Mux", "Line 2", "LINPUT2" }, + { "Differential Mux", "Line 2", "RINPUT2" }, + + { "Left ADC Mux", "Stereo", "Left PGA Mux" }, + { "Left ADC Mux", "Mono (Left)", "Left PGA Mux" }, + { "Left ADC Mux", "Digital Mono", "Left PGA Mux" }, + + { "Right ADC Mux", "Stereo", "Right PGA Mux" }, + { "Right ADC Mux", "Mono (Right)", "Right PGA Mux" }, + { "Right ADC Mux", "Digital Mono", "Right PGA Mux" }, + + { "Left ADC", NULL, "Left ADC Mux" }, + { "Right ADC", NULL, "Right ADC Mux" }, + + { "ADC DIG", NULL, "ADC STM" }, + { "ADC DIG", NULL, "ADC Vref" }, + { "ADC DIG", NULL, "ADC DLL" }, + + { "Left ADC", NULL, "ADC DIG" }, + { "Right ADC", NULL, "ADC DIG" }, + + { "Mic Bias", NULL, "Mic Bias Gen" }, + + { "Left Line Mux", "Line 1", "LINPUT1" }, + { "Left Line Mux", "Line 2", "LINPUT2" }, + { "Left Line Mux", "PGA", "Left PGA Mux" }, + { "Left Line Mux", "Differential", "Differential Mux" }, + + { "Right Line Mux", "Line 1", "RINPUT1" }, + { "Right Line Mux", "Line 2", "RINPUT2" }, + { "Right Line Mux", "PGA", "Right PGA Mux" }, + { "Right Line Mux", "Differential", "Differential Mux" }, + + { "Left Out 1", NULL, "Left DAC" }, + { "Right Out 1", NULL, "Right DAC" }, + { "Left Out 2", NULL, "Left DAC" }, + { "Right Out 2", NULL, "Right DAC" }, + + { "Left Mixer", "Playback Switch", "Left DAC" }, + { "Left Mixer", "Left Bypass Switch", "Left Line Mux" }, + { "Left Mixer", "Right Playback Switch", "Right DAC" }, + { "Left Mixer", "Right Bypass Switch", "Right Line Mux" }, + + { "Right Mixer", "Left Playback Switch", "Left DAC" }, + { "Right Mixer", "Left Bypass Switch", "Left Line Mux" }, + { "Right Mixer", "Playback Switch", "Right DAC" }, + { "Right Mixer", "Right Bypass Switch", "Right Line Mux" }, + + { "DAC DIG", NULL, "DAC STM" }, + { "DAC DIG", NULL, "DAC Vref" }, + { "DAC DIG", NULL, "DAC DLL" }, + + { "Left DAC", NULL, "DAC DIG" }, + { "Right DAC", NULL, "DAC DIG" }, + + { "Left Out 1", NULL, "Left Mixer" }, + { "LOUT1", NULL, "Left Out 1" }, + { "Right Out 1", NULL, "Right Mixer" }, + { "ROUT1", NULL, "Right Out 1" }, + + { "Left Out 2", NULL, "Left Mixer" }, + { "LOUT2", NULL, "Left Out 2" }, + { "Right Out 2", NULL, "Right Mixer" }, + { "ROUT2", NULL, "Right Out 2" }, +}; + +static int es8328_mute(struct snd_soc_dai *dai, int mute) +{ + return snd_soc_update_bits(dai->codec, ES8328_DACCONTROL3, + ES8328_DACCONTROL3_DACMUTE, + mute ? ES8328_DACCONTROL3_DACMUTE : 0); +} + +static int es8328_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 es8328_priv *es8328 = snd_soc_codec_get_drvdata(codec); + int clk_rate; + int i; + int reg; + u8 ratio; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + reg = ES8328_DACCONTROL2; + else + reg = ES8328_ADCCONTROL5; + + clk_rate = clk_get_rate(es8328->clk); + + if ((clk_rate != ES8328_SYSCLK_RATE_1X) && + (clk_rate != ES8328_SYSCLK_RATE_2X)) { + dev_err(codec->dev, + "%s: clock is running at %d Hz, not %d or %d Hz\n", + __func__, clk_rate, + ES8328_SYSCLK_RATE_1X, ES8328_SYSCLK_RATE_2X); + return -EINVAL; + } + + /* find master mode MCLK to sampling frequency ratio */ + ratio = mclk_ratios[0].rate; + for (i = 1; i < ARRAY_SIZE(mclk_ratios); i++) + if (params_rate(params) <= mclk_ratios[i].rate) + ratio = mclk_ratios[i].ratio; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + es8328->playback_fs = params_rate(params); + es8328_set_deemph(codec); + } + + return snd_soc_update_bits(codec, reg, ES8328_RATEMASK, ratio); +} + +static int es8328_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct es8328_priv *es8328 = snd_soc_codec_get_drvdata(codec); + int clk_rate; + u8 mode = ES8328_DACCONTROL1_DACWL_16; + + /* set master/slave audio interface */ + if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBM_CFM) + return -EINVAL; + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + mode |= ES8328_DACCONTROL1_DACFORMAT_I2S; + break; + case SND_SOC_DAIFMT_RIGHT_J: + mode |= ES8328_DACCONTROL1_DACFORMAT_RJUST; + break; + case SND_SOC_DAIFMT_LEFT_J: + mode |= ES8328_DACCONTROL1_DACFORMAT_LJUST; + break; + default: + return -EINVAL; + } + + /* clock inversion */ + if ((fmt & SND_SOC_DAIFMT_INV_MASK) != SND_SOC_DAIFMT_NB_NF) + return -EINVAL; + + snd_soc_write(codec, ES8328_DACCONTROL1, mode); + snd_soc_write(codec, ES8328_ADCCONTROL4, mode); + + /* Master serial port mode, with BCLK generated automatically */ + clk_rate = clk_get_rate(es8328->clk); + if (clk_rate == ES8328_SYSCLK_RATE_1X) + snd_soc_write(codec, ES8328_MASTERMODE, + ES8328_MASTERMODE_MSC); + else + snd_soc_write(codec, ES8328_MASTERMODE, + ES8328_MASTERMODE_MCLKDIV2 | + ES8328_MASTERMODE_MSC); + + return 0; +} + +static int es8328_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_ON: + break; + + case SND_SOC_BIAS_PREPARE: + /* VREF, VMID=2x50k, digital enabled */ + snd_soc_write(codec, ES8328_CHIPPOWER, 0); + snd_soc_update_bits(codec, ES8328_CONTROL1, + ES8328_CONTROL1_VMIDSEL_MASK | + ES8328_CONTROL1_ENREF, + ES8328_CONTROL1_VMIDSEL_50k | + ES8328_CONTROL1_ENREF); + break; + + case SND_SOC_BIAS_STANDBY: + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { + snd_soc_update_bits(codec, ES8328_CONTROL1, + ES8328_CONTROL1_VMIDSEL_MASK | + ES8328_CONTROL1_ENREF, + ES8328_CONTROL1_VMIDSEL_5k | + ES8328_CONTROL1_ENREF); + + /* Charge caps */ + msleep(100); + } + + snd_soc_write(codec, ES8328_CONTROL2, + ES8328_CONTROL2_OVERCURRENT_ON | + ES8328_CONTROL2_THERMAL_SHUTDOWN_ON); + + /* VREF, VMID=2*500k, digital stopped */ + snd_soc_update_bits(codec, ES8328_CONTROL1, + ES8328_CONTROL1_VMIDSEL_MASK | + ES8328_CONTROL1_ENREF, + ES8328_CONTROL1_VMIDSEL_500k | + ES8328_CONTROL1_ENREF); + break; + + case SND_SOC_BIAS_OFF: + snd_soc_update_bits(codec, ES8328_CONTROL1, + ES8328_CONTROL1_VMIDSEL_MASK | + ES8328_CONTROL1_ENREF, + 0); + break; + } + codec->dapm.bias_level = level; + return 0; +} + +static const struct snd_soc_dai_ops es8328_dai_ops = { + .hw_params = es8328_hw_params, + .digital_mute = es8328_mute, + .set_fmt = es8328_set_dai_fmt, +}; + +static struct snd_soc_dai_driver es8328_dai = { + .name = "es8328-hifi-analog", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = ES8328_RATES, + .formats = ES8328_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 2, + .rates = ES8328_RATES, + .formats = ES8328_FORMATS, + }, + .ops = &es8328_dai_ops, +}; + +static int es8328_suspend(struct snd_soc_codec *codec) +{ + struct es8328_priv *es8328; + int ret; + + es8328 = snd_soc_codec_get_drvdata(codec); + + es8328_set_bias_level(codec, SND_SOC_BIAS_OFF); + + clk_disable_unprepare(es8328->clk); + + ret = regulator_bulk_disable(ARRAY_SIZE(es8328->supplies), + es8328->supplies); + if (ret) { + dev_err(codec->dev, "unable to disable regulators\n"); + return ret; + } + return 0; +} + +static int es8328_resume(struct snd_soc_codec *codec) +{ + struct regmap *regmap = dev_get_regmap(codec->dev, NULL); + struct es8328_priv *es8328; + int ret; + + es8328 = snd_soc_codec_get_drvdata(codec); + + ret = clk_prepare_enable(es8328->clk); + if (ret) { + dev_err(codec->dev, "unable to enable clock\n"); + return ret; + } + + ret = regulator_bulk_enable(ARRAY_SIZE(es8328->supplies), + es8328->supplies); + if (ret) { + dev_err(codec->dev, "unable to enable regulators\n"); + return ret; + } + + regcache_mark_dirty(regmap); + ret = regcache_sync(regmap); + if (ret) { + dev_err(codec->dev, "unable to sync regcache\n"); + return ret; + } + + es8328_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + return 0; +} + +static int es8328_codec_probe(struct snd_soc_codec *codec) +{ + struct es8328_priv *es8328; + int ret; + + es8328 = snd_soc_codec_get_drvdata(codec); + + ret = regulator_bulk_enable(ARRAY_SIZE(es8328->supplies), + es8328->supplies); + if (ret) { + dev_err(codec->dev, "unable to enable regulators\n"); + return ret; + } + + /* Setup clocks */ + es8328->clk = devm_clk_get(codec->dev, NULL); + if (IS_ERR(es8328->clk)) { + dev_err(codec->dev, "codec clock missing or invalid\n"); + goto clk_fail; + } + + ret = clk_prepare_enable(es8328->clk); + if (ret) { + dev_err(codec->dev, "unable to prepare codec clk\n"); + goto clk_fail; + } + + return 0; + +clk_fail: + regulator_bulk_disable(ARRAY_SIZE(es8328->supplies), + es8328->supplies); + return ret; +} + +static int es8328_remove(struct snd_soc_codec *codec) +{ + struct es8328_priv *es8328; + + es8328 = snd_soc_codec_get_drvdata(codec); + + if (es8328->clk) + clk_disable_unprepare(es8328->clk); + + regulator_bulk_disable(ARRAY_SIZE(es8328->supplies), + es8328->supplies); + + return 0; +} + +const struct regmap_config es8328_regmap_config = { + .reg_bits = 8, + .val_bits = 8, + .max_register = ES8328_REG_MAX, + .cache_type = REGCACHE_RBTREE, +}; +EXPORT_SYMBOL_GPL(es8328_regmap_config); + +static struct snd_soc_codec_driver es8328_codec_driver = { + .probe = es8328_codec_probe, + .suspend = es8328_suspend, + .resume = es8328_resume, + .remove = es8328_remove, + .set_bias_level = es8328_set_bias_level, + .controls = es8328_snd_controls, + .num_controls = ARRAY_SIZE(es8328_snd_controls), + .dapm_widgets = es8328_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(es8328_dapm_widgets), + .dapm_routes = es8328_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(es8328_dapm_routes), +}; + +int es8328_probe(struct device *dev, struct regmap *regmap) +{ + struct es8328_priv *es8328; + int ret; + int i; + + if (IS_ERR(regmap)) + return PTR_ERR(regmap); + + es8328 = devm_kzalloc(dev, sizeof(*es8328), GFP_KERNEL); + if (es8328 == NULL) + return -ENOMEM; + + es8328->regmap = regmap; + + for (i = 0; i < ARRAY_SIZE(es8328->supplies); i++) + es8328->supplies[i].supply = supply_names[i]; + + ret = devm_regulator_bulk_get(dev, ARRAY_SIZE(es8328->supplies), + es8328->supplies); + if (ret) { + dev_err(dev, "unable to get regulators\n"); + return ret; + } + + dev_set_drvdata(dev, es8328); + + return snd_soc_register_codec(dev, + &es8328_codec_driver, &es8328_dai, 1); +} +EXPORT_SYMBOL_GPL(es8328_probe); + +MODULE_DESCRIPTION("ASoC ES8328 driver"); +MODULE_AUTHOR("Sean Cross "); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/es8328.h b/sound/soc/codecs/es8328.h new file mode 100644 index 0000000..cb36afe --- /dev/null +++ b/sound/soc/codecs/es8328.h @@ -0,0 +1,314 @@ +/* + * es8328.h -- ES8328 ALSA SoC Audio driver + */ + +#ifndef _ES8328_H +#define _ES8328_H + +#include + +struct device; + +extern const struct regmap_config es8328_regmap_config; +int es8328_probe(struct device *dev, struct regmap *regmap); + +#define ES8328_DACLVOL 46 +#define ES8328_DACRVOL 47 +#define ES8328_DACCTL 28 +#define ES8328_RATEMASK (0x1f << 0) + +#define ES8328_CONTROL1 0x00 +#define ES8328_CONTROL1_VMIDSEL_OFF (0 << 0) +#define ES8328_CONTROL1_VMIDSEL_50k (1 << 0) +#define ES8328_CONTROL1_VMIDSEL_500k (2 << 0) +#define ES8328_CONTROL1_VMIDSEL_5k (3 << 0) +#define ES8328_CONTROL1_VMIDSEL_MASK (7 << 0) +#define ES8328_CONTROL1_ENREF (1 << 2) +#define ES8328_CONTROL1_SEQEN (1 << 3) +#define ES8328_CONTROL1_SAMEFS (1 << 4) +#define ES8328_CONTROL1_DACMCLK_ADC (0 << 5) +#define ES8328_CONTROL1_DACMCLK_DAC (1 << 5) +#define ES8328_CONTROL1_LRCM (1 << 6) +#define ES8328_CONTROL1_SCP_RESET (1 << 7) + +#define ES8328_CONTROL2 0x01 +#define ES8328_CONTROL2_VREF_BUF_OFF (1 << 0) +#define ES8328_CONTROL2_VREF_LOWPOWER (1 << 1) +#define ES8328_CONTROL2_IBIASGEN_OFF (1 << 2) +#define ES8328_CONTROL2_ANALOG_OFF (1 << 3) +#define ES8328_CONTROL2_VREF_BUF_LOWPOWER (1 << 4) +#define ES8328_CONTROL2_VCM_MOD_LOWPOWER (1 << 5) +#define ES8328_CONTROL2_OVERCURRENT_ON (1 << 6) +#define ES8328_CONTROL2_THERMAL_SHUTDOWN_ON (1 << 7) + +#define ES8328_CHIPPOWER 0x02 +#define ES8328_CHIPPOWER_DACVREF_OFF 0 +#define ES8328_CHIPPOWER_ADCVREF_OFF 1 +#define ES8328_CHIPPOWER_DACDLL_OFF 2 +#define ES8328_CHIPPOWER_ADCDLL_OFF 3 +#define ES8328_CHIPPOWER_DACSTM_RESET 4 +#define ES8328_CHIPPOWER_ADCSTM_RESET 5 +#define ES8328_CHIPPOWER_DACDIG_OFF 6 +#define ES8328_CHIPPOWER_ADCDIG_OFF 7 + +#define ES8328_ADCPOWER 0x03 +#define ES8328_ADCPOWER_INT1_LOWPOWER 0 +#define ES8328_ADCPOWER_FLASH_ADC_LOWPOWER 1 +#define ES8328_ADCPOWER_ADC_BIAS_GEN_OFF 2 +#define ES8328_ADCPOWER_MIC_BIAS_OFF 3 +#define ES8328_ADCPOWER_ADCR_OFF 4 +#define ES8328_ADCPOWER_ADCL_OFF 5 +#define ES8328_ADCPOWER_AINR_OFF 6 +#define ES8328_ADCPOWER_AINL_OFF 7 + +#define ES8328_DACPOWER 0x04 +#define ES8328_DACPOWER_OUT3_ON 0 +#define ES8328_DACPOWER_MONO_ON 1 +#define ES8328_DACPOWER_ROUT2_ON 2 +#define ES8328_DACPOWER_LOUT2_ON 3 +#define ES8328_DACPOWER_ROUT1_ON 4 +#define ES8328_DACPOWER_LOUT1_ON 5 +#define ES8328_DACPOWER_RDAC_OFF 6 +#define ES8328_DACPOWER_LDAC_OFF 7 + +#define ES8328_CHIPLOPOW1 0x05 +#define ES8328_CHIPLOPOW2 0x06 +#define ES8328_ANAVOLMANAG 0x07 + +#define ES8328_MASTERMODE 0x08 +#define ES8328_MASTERMODE_BCLKDIV (0 << 0) +#define ES8328_MASTERMODE_BCLK_INV (1 << 5) +#define ES8328_MASTERMODE_MCLKDIV2 (1 << 6) +#define ES8328_MASTERMODE_MSC (1 << 7) + +#define ES8328_ADCCONTROL1 0x09 +#define ES8328_ADCCONTROL2 0x0a +#define ES8328_ADCCONTROL3 0x0b +#define ES8328_ADCCONTROL4 0x0c +#define ES8328_ADCCONTROL5 0x0d +#define ES8328_ADCCONTROL5_RATEMASK (0x1f << 0) + +#define ES8328_ADCCONTROL6 0x0e + +#define ES8328_ADCCONTROL7 0x0f +#define ES8328_ADCCONTROL7_ADC_MUTE (1 << 2) +#define ES8328_ADCCONTROL7_ADC_LER (1 << 3) +#define ES8328_ADCCONTROL7_ADC_ZERO_CROSS (1 << 4) +#define ES8328_ADCCONTROL7_ADC_SOFT_RAMP (1 << 5) +#define ES8328_ADCCONTROL7_ADC_RAMP_RATE_4 (0 << 6) +#define ES8328_ADCCONTROL7_ADC_RAMP_RATE_8 (1 << 6) +#define ES8328_ADCCONTROL7_ADC_RAMP_RATE_16 (2 << 6) +#define ES8328_ADCCONTROL7_ADC_RAMP_RATE_32 (3 << 6) + +#define ES8328_ADCCONTROL8 0x10 +#define ES8328_ADCCONTROL9 0x11 +#define ES8328_ADCCONTROL10 0x12 +#define ES8328_ADCCONTROL11 0x13 +#define ES8328_ADCCONTROL12 0x14 +#define ES8328_ADCCONTROL13 0x15 +#define ES8328_ADCCONTROL14 0x16 + +#define ES8328_DACCONTROL1 0x17 +#define ES8328_DACCONTROL1_DACFORMAT_I2S (0 << 1) +#define ES8328_DACCONTROL1_DACFORMAT_LJUST (1 << 1) +#define ES8328_DACCONTROL1_DACFORMAT_RJUST (2 << 1) +#define ES8328_DACCONTROL1_DACFORMAT_PCM (3 << 1) +#define ES8328_DACCONTROL1_DACWL_24 (0 << 3) +#define ES8328_DACCONTROL1_DACWL_20 (1 << 3) +#define ES8328_DACCONTROL1_DACWL_18 (2 << 3) +#define ES8328_DACCONTROL1_DACWL_16 (3 << 3) +#define ES8328_DACCONTROL1_DACWL_32 (4 << 3) +#define ES8328_DACCONTROL1_DACLRP_I2S_POL_NORMAL (0 << 6) +#define ES8328_DACCONTROL1_DACLRP_I2S_POL_INV (1 << 6) +#define ES8328_DACCONTROL1_DACLRP_PCM_MSB_CLK2 (0 << 6) +#define ES8328_DACCONTROL1_DACLRP_PCM_MSB_CLK1 (1 << 6) +#define ES8328_DACCONTROL1_LRSWAP (1 << 7) + +#define ES8328_DACCONTROL2 0x18 +#define ES8328_DACCONTROL2_RATEMASK (0x1f << 0) +#define ES8328_DACCONTROL2_DOUBLESPEED (1 << 5) + +#define ES8328_DACCONTROL3 0x19 +#define ES8328_DACCONTROL3_AUTOMUTE (1 << 2) +#define ES8328_DACCONTROL3_DACMUTE (1 << 2) +#define ES8328_DACCONTROL3_LEFTGAINVOL (1 << 3) +#define ES8328_DACCONTROL3_DACZEROCROSS (1 << 4) +#define ES8328_DACCONTROL3_DACSOFTRAMP (1 << 5) +#define ES8328_DACCONTROL3_DACRAMPRATE (3 << 6) + +#define ES8328_LDACVOL 0x1a +#define ES8328_LDACVOL_MASK (0 << 0) +#define ES8328_LDACVOL_MAX (0xc0) + +#define ES8328_RDACVOL 0x1b +#define ES8328_RDACVOL_MASK (0 << 0) +#define ES8328_RDACVOL_MAX (0xc0) + +#define ES8328_DACVOL_MAX (0xc0) + +#define ES8328_DACCONTROL4 0x1a +#define ES8328_DACCONTROL5 0x1b + +#define ES8328_DACCONTROL6 0x1c +#define ES8328_DACCONTROL6_CLICKFREE (1 << 3) +#define ES8328_DACCONTROL6_DAC_INVR (1 << 4) +#define ES8328_DACCONTROL6_DAC_INVL (1 << 5) +#define ES8328_DACCONTROL6_DEEMPH_OFF (0 << 6) +#define ES8328_DACCONTROL6_DEEMPH_32k (1 << 6) +#define ES8328_DACCONTROL6_DEEMPH_44_1k (2 << 6) +#define ES8328_DACCONTROL6_DEEMPH_48k (3 << 6) + +#define ES8328_DACCONTROL7 0x1d +#define ES8328_DACCONTROL7_VPP_SCALE_3p5 (0 << 0) +#define ES8328_DACCONTROL7_VPP_SCALE_4p0 (1 << 0) +#define ES8328_DACCONTROL7_VPP_SCALE_3p0 (2 << 0) +#define ES8328_DACCONTROL7_VPP_SCALE_2p5 (3 << 0) +#define ES8328_DACCONTROL7_SHELVING_STRENGTH (1 << 2) /* In eights */ +#define ES8328_DACCONTROL7_MONO (1 << 5) +#define ES8328_DACCONTROL7_ZEROR (1 << 6) +#define ES8328_DACCONTROL7_ZEROL (1 << 7) + +/* Shelving filter */ +#define ES8328_DACCONTROL8 0x1e +#define ES8328_DACCONTROL9 0x1f +#define ES8328_DACCONTROL10 0x20 +#define ES8328_DACCONTROL11 0x21 +#define ES8328_DACCONTROL12 0x22 +#define ES8328_DACCONTROL13 0x23 +#define ES8328_DACCONTROL14 0x24 +#define ES8328_DACCONTROL15 0x25 + +#define ES8328_DACCONTROL16 0x26 +#define ES8328_DACCONTROL16_RMIXSEL_RIN1 (0 << 0) +#define ES8328_DACCONTROL16_RMIXSEL_RIN2 (1 << 0) +#define ES8328_DACCONTROL16_RMIXSEL_RIN3 (2 << 0) +#define ES8328_DACCONTROL16_RMIXSEL_RADC (3 << 0) +#define ES8328_DACCONTROL16_LMIXSEL_LIN1 (0 << 3) +#define ES8328_DACCONTROL16_LMIXSEL_LIN2 (1 << 3) +#define ES8328_DACCONTROL16_LMIXSEL_LIN3 (2 << 3) +#define ES8328_DACCONTROL16_LMIXSEL_LADC (3 << 3) + +#define ES8328_DACCONTROL17 0x27 +#define ES8328_DACCONTROL17_LI2LOVOL (7 << 3) +#define ES8328_DACCONTROL17_LI2LO (1 << 6) +#define ES8328_DACCONTROL17_LD2LO (1 << 7) + +#define ES8328_DACCONTROL18 0x28 +#define ES8328_DACCONTROL18_RI2LOVOL (7 << 3) +#define ES8328_DACCONTROL18_RI2LO (1 << 6) +#define ES8328_DACCONTROL18_RD2LO (1 << 7) + +#define ES8328_DACCONTROL19 0x29 +#define ES8328_DACCONTROL19_LI2ROVOL (7 << 3) +#define ES8328_DACCONTROL19_LI2RO (1 << 6) +#define ES8328_DACCONTROL19_LD2RO (1 << 7) + +#define ES8328_DACCONTROL20 0x2a +#define ES8328_DACCONTROL20_RI2ROVOL (7 << 3) +#define ES8328_DACCONTROL20_RI2RO (1 << 6) +#define ES8328_DACCONTROL20_RD2RO (1 << 7) + +#define ES8328_DACCONTROL21 0x2b +#define ES8328_DACCONTROL21_LI2MOVOL (7 << 3) +#define ES8328_DACCONTROL21_LI2MO (1 << 6) +#define ES8328_DACCONTROL21_LD2MO (1 << 7) + +#define ES8328_DACCONTROL22 0x2c +#define ES8328_DACCONTROL22_RI2MOVOL (7 << 3) +#define ES8328_DACCONTROL22_RI2MO (1 << 6) +#define ES8328_DACCONTROL22_RD2MO (1 << 7) + +#define ES8328_DACCONTROL23 0x2d +#define ES8328_DACCONTROL23_MOUTINV (1 << 1) +#define ES8328_DACCONTROL23_HPSWPOL (1 << 2) +#define ES8328_DACCONTROL23_HPSWEN (1 << 3) +#define ES8328_DACCONTROL23_VROI_1p5k (0 << 4) +#define ES8328_DACCONTROL23_VROI_40k (1 << 4) +#define ES8328_DACCONTROL23_OUT3_VREF (0 << 5) +#define ES8328_DACCONTROL23_OUT3_ROUT1 (1 << 5) +#define ES8328_DACCONTROL23_OUT3_MONOOUT (2 << 5) +#define ES8328_DACCONTROL23_OUT3_RIGHT_MIXER (3 << 5) +#define ES8328_DACCONTROL23_ROUT2INV (1 << 7) + +/* LOUT1 Amplifier */ +#define ES8328_LOUT1VOL 0x2e +#define ES8328_LOUT1VOL_MASK (0 << 5) +#define ES8328_LOUT1VOL_MAX (0x24) + +/* ROUT1 Amplifier */ +#define ES8328_ROUT1VOL 0x2f +#define ES8328_ROUT1VOL_MASK (0 << 5) +#define ES8328_ROUT1VOL_MAX (0x24) + +#define ES8328_OUT1VOL_MAX (0x24) + +/* LOUT2 Amplifier */ +#define ES8328_LOUT2VOL 0x30 +#define ES8328_LOUT2VOL_MASK (0 << 5) +#define ES8328_LOUT2VOL_MAX (0x24) + +/* ROUT2 Amplifier */ +#define ES8328_ROUT2VOL 0x31 +#define ES8328_ROUT2VOL_MASK (0 << 5) +#define ES8328_ROUT2VOL_MAX (0x24) + +#define ES8328_OUT2VOL_MAX (0x24) + +/* Mono Out Amplifier */ +#define ES8328_MONOOUTVOL 0x32 +#define ES8328_MONOOUTVOL_MASK (0 << 5) +#define ES8328_MONOOUTVOL_MAX (0x24) + +#define ES8328_DACCONTROL29 0x33 +#define ES8328_DACCONTROL30 0x34 + +#define ES8328_SYSCLK 0 + +#define ES8328_REG_MAX 0x35 + +#define ES8328_PLL1 0 +#define ES8328_PLL2 1 + +/* clock inputs */ +#define ES8328_MCLK 0 +#define ES8328_PCMCLK 1 + +/* clock divider id's */ +#define ES8328_PCMDIV 0 +#define ES8328_BCLKDIV 1 +#define ES8328_VXCLKDIV 2 + +/* PCM clock dividers */ +#define ES8328_PCM_DIV_1 (0 << 6) +#define ES8328_PCM_DIV_3 (2 << 6) +#define ES8328_PCM_DIV_5_5 (3 << 6) +#define ES8328_PCM_DIV_2 (4 << 6) +#define ES8328_PCM_DIV_4 (5 << 6) +#define ES8328_PCM_DIV_6 (6 << 6) +#define ES8328_PCM_DIV_8 (7 << 6) + +/* BCLK clock dividers */ +#define ES8328_BCLK_DIV_1 (0 << 7) +#define ES8328_BCLK_DIV_2 (1 << 7) +#define ES8328_BCLK_DIV_4 (2 << 7) +#define ES8328_BCLK_DIV_8 (3 << 7) + +/* VXCLK clock dividers */ +#define ES8328_VXCLK_DIV_1 (0 << 6) +#define ES8328_VXCLK_DIV_2 (1 << 6) +#define ES8328_VXCLK_DIV_4 (2 << 6) +#define ES8328_VXCLK_DIV_8 (3 << 6) +#define ES8328_VXCLK_DIV_16 (4 << 6) + +#define ES8328_DAI_HIFI 0 +#define ES8328_DAI_VOICE 1 + +#define ES8328_1536FS 1536 +#define ES8328_1024FS 1024 +#define ES8328_768FS 768 +#define ES8328_512FS 512 +#define ES8328_384FS 384 +#define ES8328_256FS 256 +#define ES8328_128FS 128 + +#endif -- cgit v1.1 From 7e7292dba2155c1433ce9f9a819f1acb9090747b Mon Sep 17 00:00:00 2001 From: Sean Cross Date: Thu, 31 Jul 2014 10:43:37 +0800 Subject: ASoC: fsl: add imx-es8328 machine driver This adds an initial machine driver for the ES8328 audio codec on Freescale boards. The driver supports headphones and an audio regulator for an onboard speaker amp. Signed-off-by: Sean Cross Signed-off-by: Mark Brown --- sound/soc/fsl/Kconfig | 14 +++ sound/soc/fsl/Makefile | 2 + sound/soc/fsl/imx-es8328.c | 232 +++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 248 insertions(+) create mode 100644 sound/soc/fsl/imx-es8328.c (limited to 'sound') diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 2b99a9e..fa90340 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -257,6 +257,20 @@ config SND_SOC_IMX_WM8962 Say Y if you want to add support for SoC audio on an i.MX board with a wm8962 codec. +config SND_SOC_IMX_ES8328 + tristate "SoC Audio support for i.MX boards with the ES8328 codec" + depends on OF && (I2C || SPI) + select SND_SOC_ES8328_I2C if I2C + select SND_SOC_ES8328_SPI if SPI_MASTER + select SND_SOC_IMX_PCM_DMA + select SND_SOC_IMX_AUDMUX + select SND_SOC_FSL_SSI + select SND_SOC_FSL_UTILS + select SND_SOC_IMX_PCM_FIQ + help + Say Y if you want to add support for the ES8328 audio codec connected + via SSI/I2S over either SPI or I2C. + config SND_SOC_IMX_SGTL5000 tristate "SoC Audio support for i.MX boards with sgtl5000" depends on OF && I2C diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index 8f6d84e..d28dc25 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -52,6 +52,7 @@ snd-soc-eukrea-tlv320-objs := eukrea-tlv320.o snd-soc-phycore-ac97-objs := phycore-ac97.o snd-soc-mx27vis-aic32x4-objs := mx27vis-aic32x4.o snd-soc-wm1133-ev1-objs := wm1133-ev1.o +snd-soc-imx-es8328-objs := imx-es8328.o snd-soc-imx-sgtl5000-objs := imx-sgtl5000.o snd-soc-imx-wm8962-objs := imx-wm8962.o snd-soc-imx-spdif-objs := imx-spdif.o @@ -61,6 +62,7 @@ obj-$(CONFIG_SND_SOC_EUKREA_TLV320) += snd-soc-eukrea-tlv320.o obj-$(CONFIG_SND_SOC_PHYCORE_AC97) += snd-soc-phycore-ac97.o obj-$(CONFIG_SND_SOC_MX27VIS_AIC32X4) += snd-soc-mx27vis-aic32x4.o obj-$(CONFIG_SND_MXC_SOC_WM1133_EV1) += snd-soc-wm1133-ev1.o +obj-$(CONFIG_SND_SOC_IMX_ES8328) += snd-soc-imx-es8328.o obj-$(CONFIG_SND_SOC_IMX_SGTL5000) += snd-soc-imx-sgtl5000.o obj-$(CONFIG_SND_SOC_IMX_WM8962) += snd-soc-imx-wm8962.o obj-$(CONFIG_SND_SOC_IMX_SPDIF) += snd-soc-imx-spdif.o diff --git a/sound/soc/fsl/imx-es8328.c b/sound/soc/fsl/imx-es8328.c new file mode 100644 index 0000000..653e66d --- /dev/null +++ b/sound/soc/fsl/imx-es8328.c @@ -0,0 +1,232 @@ +/* + * Copyright 2012 Freescale Semiconductor, Inc. + * Copyright 2012 Linaro Ltd. + * + * The code contained herein is licensed under the GNU General Public + * License. You may obtain a copy of the GNU General Public License + * Version 2 or later at the following locations: + * + * http://www.opensource.org/licenses/gpl-license.html + * http://www.gnu.org/copyleft/gpl.html + */ + +#include +#include +#include +#include +#include +#include +#include +#include + +#include "imx-audmux.h" + +#define DAI_NAME_SIZE 32 +#define MUX_PORT_MAX 7 + +struct imx_es8328_data { + struct device *dev; + struct snd_soc_dai_link dai; + struct snd_soc_card card; + char codec_dai_name[DAI_NAME_SIZE]; + char platform_name[DAI_NAME_SIZE]; + int jack_gpio; +}; + +static struct snd_soc_jack_gpio headset_jack_gpios[] = { + { + .gpio = -1, + .name = "headset-gpio", + .report = SND_JACK_HEADSET, + .invert = 0, + .debounce_time = 200, + }, +}; + +static struct snd_soc_jack headset_jack; + +static int imx_es8328_dai_init(struct snd_soc_pcm_runtime *rtd) +{ + struct imx_es8328_data *data = container_of(rtd->card, + struct imx_es8328_data, card); + int ret = 0; + + /* Headphone jack detection */ + if (gpio_is_valid(data->jack_gpio)) { + ret = snd_soc_jack_new(rtd->codec, "Headphone", + SND_JACK_HEADPHONE | SND_JACK_BTN_0, + &headset_jack); + if (ret) + return ret; + + headset_jack_gpios[0].gpio = data->jack_gpio; + ret = snd_soc_jack_add_gpios(&headset_jack, + ARRAY_SIZE(headset_jack_gpios), + headset_jack_gpios); + } + + return ret; +} + +static const struct snd_soc_dapm_widget imx_es8328_dapm_widgets[] = { + SND_SOC_DAPM_MIC("Mic Jack", NULL), + SND_SOC_DAPM_HP("Headphone", NULL), + SND_SOC_DAPM_SPK("Speaker", NULL), + SND_SOC_DAPM_REGULATOR_SUPPLY("audio-amp", 1, 0), +}; + +static int imx_es8328_probe(struct platform_device *pdev) +{ + struct device_node *np = pdev->dev.of_node; + struct device_node *ssi_np, *codec_np; + struct platform_device *ssi_pdev; + struct imx_es8328_data *data; + u32 int_port, ext_port; + int ret; + struct device *dev = &pdev->dev; + + ret = of_property_read_u32(np, "mux-int-port", &int_port); + if (ret) { + dev_err(dev, "mux-int-port missing or invalid\n"); + goto fail; + } + if (int_port > MUX_PORT_MAX || int_port == 0) { + dev_err(dev, "mux-int-port: hardware only has %d mux ports\n", + MUX_PORT_MAX); + goto fail; + } + + ret = of_property_read_u32(np, "mux-ext-port", &ext_port); + if (ret) { + dev_err(dev, "mux-ext-port missing or invalid\n"); + goto fail; + } + if (ext_port > MUX_PORT_MAX || ext_port == 0) { + dev_err(dev, "mux-ext-port: hardware only has %d mux ports\n", + MUX_PORT_MAX); + goto fail; + } + + /* + * The port numbering in the hardware manual starts at 1, while + * the audmux API expects it starts at 0. + */ + int_port--; + ext_port--; + ret = imx_audmux_v2_configure_port(int_port, + IMX_AUDMUX_V2_PTCR_SYN | + IMX_AUDMUX_V2_PTCR_TFSEL(ext_port) | + IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) | + IMX_AUDMUX_V2_PTCR_TFSDIR | + IMX_AUDMUX_V2_PTCR_TCLKDIR, + IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port)); + if (ret) { + dev_err(dev, "audmux internal port setup failed\n"); + return ret; + } + ret = imx_audmux_v2_configure_port(ext_port, + IMX_AUDMUX_V2_PTCR_SYN, + IMX_AUDMUX_V2_PDCR_RXDSEL(int_port)); + if (ret) { + dev_err(dev, "audmux external port setup failed\n"); + return ret; + } + + ssi_np = of_parse_phandle(pdev->dev.of_node, "ssi-controller", 0); + codec_np = of_parse_phandle(pdev->dev.of_node, "audio-codec", 0); + if (!ssi_np || !codec_np) { + dev_err(dev, "phandle missing or invalid\n"); + ret = -EINVAL; + goto fail; + } + + ssi_pdev = of_find_device_by_node(ssi_np); + if (!ssi_pdev) { + dev_err(dev, "failed to find SSI platform device\n"); + ret = -EINVAL; + goto fail; + } + + data = devm_kzalloc(dev, sizeof(*data), GFP_KERNEL); + if (!data) { + ret = -ENOMEM; + goto fail; + } + + data->dev = dev; + + data->jack_gpio = of_get_named_gpio(pdev->dev.of_node, "jack-gpio", 0); + + data->dai.name = "hifi"; + data->dai.stream_name = "hifi"; + data->dai.codec_dai_name = "es8328-hifi-analog"; + data->dai.codec_of_node = codec_np; + data->dai.cpu_of_node = ssi_np; + data->dai.platform_of_node = ssi_np; + data->dai.init = &imx_es8328_dai_init; + data->dai.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM; + + data->card.dev = dev; + data->card.dapm_widgets = imx_es8328_dapm_widgets; + data->card.num_dapm_widgets = ARRAY_SIZE(imx_es8328_dapm_widgets); + ret = snd_soc_of_parse_card_name(&data->card, "model"); + if (ret) { + dev_err(dev, "Unable to parse card name\n"); + goto fail; + } + ret = snd_soc_of_parse_audio_routing(&data->card, "audio-routing"); + if (ret) { + dev_err(dev, "Unable to parse routing: %d\n", ret); + goto fail; + } + data->card.num_links = 1; + data->card.owner = THIS_MODULE; + data->card.dai_link = &data->dai; + + ret = snd_soc_register_card(&data->card); + if (ret) { + dev_err(dev, "Unable to register: %d\n", ret); + goto fail; + } + + platform_set_drvdata(pdev, data); +fail: + of_node_put(ssi_np); + of_node_put(codec_np); + + return ret; +} + +static int imx_es8328_remove(struct platform_device *pdev) +{ + struct imx_es8328_data *data = platform_get_drvdata(pdev); + + snd_soc_jack_free_gpios(&headset_jack, ARRAY_SIZE(headset_jack_gpios), + headset_jack_gpios); + + snd_soc_unregister_card(&data->card); + + return 0; +} + +static const struct of_device_id imx_es8328_dt_ids[] = { + { .compatible = "fsl,imx-audio-es8328", }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, imx_es8328_dt_ids); + +static struct platform_driver imx_es8328_driver = { + .driver = { + .name = "imx-es8328", + .of_match_table = imx_es8328_dt_ids, + }, + .probe = imx_es8328_probe, + .remove = imx_es8328_remove, +}; +module_platform_driver(imx_es8328_driver); + +MODULE_AUTHOR("Sean Cross "); +MODULE_DESCRIPTION("Kosagi i.MX6 ES8328 ASoC machine driver"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:imx-audio-es8328"); -- cgit v1.1 From 855675f6e6a65688a7f4cf45b9b5a98cf6c6f5c3 Mon Sep 17 00:00:00 2001 From: Nicolin Chen Date: Mon, 4 Aug 2014 15:07:25 +0800 Subject: ASoC: fsl_sai: Set SYNC bit of TCR2 to Asynchronous Mode There is one design rule according to SAI's reference manual: If the transmitter bit clock and frame sync are to be used by both transmitter and receiver, the transmitter must be configured for asynchronous operation and the receiver for synchronous operation. And SYNC of TCR2 is a 2-width control bit: 00 Asynchronous mode. 01 Synchronous with receiver. 10 Synchronous with another SAI transmitter. 11 Synchronous with another SAI receiver. So the driver should have set SYNC bit of TCR2 to 0x0, and meanwhile set SYNC bit of RCR2 to 0x1 (Synchronous with transmitter). Signed-off-by: Nicolin Chen Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_sai.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 4c9e71c..60fe7c7 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -334,8 +334,7 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd, * Rx sync with Tx clocks: Clear SYNC for Tx, set it for Rx. * Tx sync with Rx clocks: Clear SYNC for Rx, set it for Tx. */ - regmap_update_bits(sai->regmap, FSL_SAI_TCR2, FSL_SAI_CR2_SYNC, - sai->synchronous[TX] ? FSL_SAI_CR2_SYNC : 0); + regmap_update_bits(sai->regmap, FSL_SAI_TCR2, FSL_SAI_CR2_SYNC, 0); regmap_update_bits(sai->regmap, FSL_SAI_RCR2, FSL_SAI_CR2_SYNC, sai->synchronous[RX] ? FSL_SAI_CR2_SYNC : 0); -- cgit v1.1 From 8a36eaa2ff4a9452a78d799503b920b4e1a0ec31 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sun, 17 Aug 2014 12:03:05 +0200 Subject: ASoC: dmic: Add to SND_SOC_ALL_CODECS Improve build coverage of the dmic driver. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 8838838e..e514e98 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -56,6 +56,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_DA7213 if I2C select SND_SOC_DA732X if I2C select SND_SOC_DA9055 if I2C + select SND_SOC_DMIC select SND_SOC_BT_SCO select SND_SOC_ISABELLE if I2C select SND_SOC_JZ4740_CODEC -- cgit v1.1 From 371e07ec837464375fe4d7ef3bd13e13cdfbb458 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sun, 17 Aug 2014 16:18:17 +0200 Subject: ASoC: edma-pcm: Include edma-pcm.h edma_pcm_platform_register() is declared in edma-pcm.h and defined in edma-pcm.c. To make sure that the function signature matches for both edma-pcm.c should include edma-pcm.h Fixes the following sparse warning: sound/soc/davinci/edma-pcm.c:48:5: warning: symbol 'edma_pcm_platform_register' was not declared. Should it be static? Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/davinci/edma-pcm.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/soc/davinci/edma-pcm.c b/sound/soc/davinci/edma-pcm.c index 605e643..59e588a 100644 --- a/sound/soc/davinci/edma-pcm.c +++ b/sound/soc/davinci/edma-pcm.c @@ -25,6 +25,8 @@ #include #include +#include "edma-pcm.h" + static const struct snd_pcm_hardware edma_pcm_hardware = { .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | -- cgit v1.1 From d80a12f92466d0bc4fd244c9052a8a88518c868e Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sun, 17 Aug 2014 16:18:18 +0200 Subject: ASoC: odrodix2_max98090: Make non exported symbols static odroidx2_drvdata and odroidu3_drvdata are not used outside this module so make them static (and also const while we are at it). Fixes the following warnings from sparse: sound/soc/samsung/odroidx2_max98090.c:69:26: warning: symbol 'odroidx2_drvdata' was not declared. Should it be static? sound/soc/samsung/odroidx2_max98090.c:74:26: warning: symbol 'odroidu3_drvdata' was not declared. Should it be static? Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/samsung/odroidx2_max98090.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/odroidx2_max98090.c b/sound/soc/samsung/odroidx2_max98090.c index 278edf9..3c8f604 100644 --- a/sound/soc/samsung/odroidx2_max98090.c +++ b/sound/soc/samsung/odroidx2_max98090.c @@ -66,12 +66,12 @@ static struct snd_soc_card odroidx2 = { .late_probe = odroidx2_late_probe, }; -struct odroidx2_drv_data odroidx2_drvdata = { +static const struct odroidx2_drv_data odroidx2_drvdata = { .dapm_widgets = odroidx2_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(odroidx2_dapm_widgets), }; -struct odroidx2_drv_data odroidu3_drvdata = { +static const struct odroidx2_drv_data odroidu3_drvdata = { .dapm_widgets = odroidu3_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(odroidu3_dapm_widgets), }; -- cgit v1.1 From 6c7d1dfca999f58c65ed7b10c2f0945dd92db103 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sun, 17 Aug 2014 16:18:20 +0200 Subject: ASoC: sh: Fix dma direction type dmaengine_prep_slave_single() expects a enum dma_transfer_direction and not a enum dma_data_direction. Since the integer representations of both DMA_TO_DEVICE and DMA_MEM_TO_DEV aswell as DMA_FROM_DEVICE and DMA_DEV_TO_MEM have the same value the code worked fine even though it was using the wrong type. Fixes the following warnings from sparse: sound/soc/sh/fsi.c:1307:42: warning: mixing different enum types sound/soc/sh/fsi.c:1307:42: int enum dma_data_direction versus sound/soc/sh/fsi.c:1307:42: int enum dma_transfer_direction Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/sh/fsi.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index c763443..66fddec 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -1297,9 +1297,14 @@ static int fsi_dma_transfer(struct fsi_priv *fsi, struct fsi_stream *io) struct snd_pcm_substream *substream = io->substream; struct dma_async_tx_descriptor *desc; int is_play = fsi_stream_is_play(fsi, io); - enum dma_data_direction dir = is_play ? DMA_TO_DEVICE : DMA_FROM_DEVICE; + enum dma_transfer_direction dir; int ret = -EIO; + if (is_play) + dir = DMA_MEM_TO_DEV; + else + dir = DMA_DEV_TO_MEM; + desc = dmaengine_prep_dma_cyclic(io->chan, substream->runtime->dma_addr, snd_pcm_lib_buffer_bytes(substream), -- cgit v1.1 From e8a70c25b809367fc314743e1ba1dbf0159398a7 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sun, 17 Aug 2014 16:18:21 +0200 Subject: ASoC: samsung idma: Add proper annotation for casting iomem pointers It is not always possible to interchange iomem pointers with normal pointers, which why we have annotations for iomem pointers and warn when casting them to a normal pointer or vice versa. In this case the casting is fine and unfortunately necessary so add the proper annotations to tell code checkers that it is intentional. This silences the following warnings from sparse: sound/soc/samsung/idma.c:354:20: warning: incorrect type in argument 1 (different address spaces) expected void volatile [noderef] *addr got unsigned char *area sound/soc/samsung/idma.c:372:22: warning: cast removes address space of expression Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/samsung/idma.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/idma.c b/sound/soc/samsung/idma.c index db6cefa..0e8dd98 100644 --- a/sound/soc/samsung/idma.c +++ b/sound/soc/samsung/idma.c @@ -351,7 +351,7 @@ static void idma_free(struct snd_pcm *pcm) if (!buf->area) return; - iounmap(buf->area); + iounmap((void __iomem *)buf->area); buf->area = NULL; buf->addr = 0; @@ -369,7 +369,7 @@ static int preallocate_idma_buffer(struct snd_pcm *pcm, int stream) buf->dev.type = SNDRV_DMA_TYPE_CONTINUOUS; buf->addr = idma.lp_tx_addr; buf->bytes = idma_hardware.buffer_bytes_max; - buf->area = (unsigned char *)ioremap(buf->addr, buf->bytes); + buf->area = (unsigned char * __force)ioremap(buf->addr, buf->bytes); return 0; } -- cgit v1.1 From 6391fffb7b6099fae0e869229279d147c47f617a Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sun, 17 Aug 2014 16:18:22 +0200 Subject: ASoC: ab8500-codec: Drop bank prefix from AB8500_GPIO_DIR4_REG register define The AB8500_GPIO_DIR4_REG register define has the bank for the register in the upper 8 bits and the register itself in the lower 8 bits. When passing it to abx500_{set,get}_register_interruptible() the upper bits get truncated which generates the following warning from sparse: sound/soc/codecs/ab8500-codec.c:1972:53: warning: cast truncates bits from constant value (1013 becomes 13) sound/soc/codecs/ab8500-codec.c:1980:53: warning: cast truncates bits from constant value (1013 becomes 13) The bank is passed separately to abx500_{set,get}_register_interruptible() so the code works fine as it is. Given that all users of AB8500_GPIO_DIR4_REG always truncate the upper 8 bits just remove them from the define. Also remove the unnecessary casts to u8. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/ab8500-codec.c | 11 +++++------ 1 file changed, 5 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/ab8500-codec.c b/sound/soc/codecs/ab8500-codec.c index 1fb4402..62cf231 100644 --- a/sound/soc/codecs/ab8500-codec.c +++ b/sound/soc/codecs/ab8500-codec.c @@ -56,8 +56,7 @@ #define GPIO31_DIR_OUTPUT 0x40 /* Macrocell register definitions */ -#define AB8500_CTRL3_REG 0x0200 -#define AB8500_GPIO_DIR4_REG 0x1013 +#define AB8500_GPIO_DIR4_REG 0x13 /* Bank AB8500_MISC */ /* Nr of FIR/IIR-coeff banks in ANC-block */ #define AB8500_NR_OF_ANC_COEFF_BANKS 2 @@ -1968,16 +1967,16 @@ static int ab8500_audio_setup_mics(struct snd_soc_codec *codec, dev_dbg(codec->dev, "%s: Enter.\n", __func__); /* Set DMic-clocks to outputs */ - status = abx500_get_register_interruptible(codec->dev, (u8)AB8500_MISC, - (u8)AB8500_GPIO_DIR4_REG, + status = abx500_get_register_interruptible(codec->dev, AB8500_MISC, + AB8500_GPIO_DIR4_REG, &value8); if (status < 0) return status; value = value8 | GPIO27_DIR_OUTPUT | GPIO29_DIR_OUTPUT | GPIO31_DIR_OUTPUT; status = abx500_set_register_interruptible(codec->dev, - (u8)AB8500_MISC, - (u8)AB8500_GPIO_DIR4_REG, + AB8500_MISC, + AB8500_GPIO_DIR4_REG, value); if (status < 0) return status; -- cgit v1.1 From 5f37671e004eeca017b93f6b26f2425acbb8d411 Mon Sep 17 00:00:00 2001 From: Shengjiu Wang Date: Mon, 18 Aug 2014 16:38:39 +0800 Subject: ASoC: fsl-asoc-card: Fix build warning for maybe-uninitialized When build fsl-asoc-card as module, there is following error: sound/soc/fsl/fsl-asoc-card.c: In function 'fsl_asoc_card_probe': >> sound/soc/fsl/fsl-asoc-card.c:547:13: warning: 'asrc_np' may be used uninitialized in this function [-Wmaybe-uninitialized] of_node_put(asrc_np); ^ vim +/asrc_np +547 sound/soc/fsl/fsl-asoc-card.c 531 if (width == 24) 532 priv->asrc_format = SNDRV_PCM_FORMAT_S24_LE; 533 else 534 priv->asrc_format = SNDRV_PCM_FORMAT_S16_LE; 535 } 536 537 /* Finish card registering */ 538 platform_set_drvdata(pdev, priv); 539 snd_soc_card_set_drvdata(&priv->card, priv); 540 541 ret = devm_snd_soc_register_card(&pdev->dev, &priv->card); 542 if (ret) 543 dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); 544 545 fail: 546 of_node_put(codec_np); > 547 of_node_put(asrc_np); 548 of_node_put(cpu_np); 549 550 return ret; 551 } 552 553 static const struct of_device_id fsl_asoc_card_dt_ids[] = { 554 { .compatible = "fsl,imx-audio-cs42888", }, 555 { .compatible = "fsl,imx-audio-sgtl5000", }, Add 'asrc_fail' branch for error jump after asrc_np initialized. Reported-by: kbuild test robot Signed-off-by: Shengjiu Wang Signed-off-by: Mark Brown --- sound/soc/fsl/fsl-asoc-card.c | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index cf3f1f4..007c772 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -469,7 +469,7 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) ret = fsl_asoc_card_audmux_init(np, priv); if (ret) { dev_err(&pdev->dev, "failed to init audmux\n"); - goto fail; + goto asrc_fail; } } else if (strstr(cpu_np->name, "esai")) { priv->cpu_priv.sysclk_id[1] = ESAI_HCKT_EXTAL; @@ -518,14 +518,14 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) if (ret) { dev_err(&pdev->dev, "failed to get output rate\n"); ret = -EINVAL; - goto fail; + goto asrc_fail; } ret = of_property_read_u32(asrc_np, "fsl,asrc-width", &width); if (ret) { dev_err(&pdev->dev, "failed to get output rate\n"); ret = -EINVAL; - goto fail; + goto asrc_fail; } if (width == 24) @@ -542,9 +542,10 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) if (ret) dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); +asrc_fail: + of_node_put(asrc_np); fail: of_node_put(codec_np); - of_node_put(asrc_np); of_node_put(cpu_np); return ret; -- cgit v1.1 From 499898d66d88cc626a2e01b02c3b819536bdf169 Mon Sep 17 00:00:00 2001 From: Shengjiu Wang Date: Mon, 18 Aug 2014 16:38:40 +0800 Subject: ASoC: fsl: fsl-asoc-card: Select SND_SOC_IMX_AUDMUX Building kernel with SND_SOC_IMX_AUDMUX=n leads to the following error: sound/built-in.o: In function `fsl_asoc_card_probe': >> fsl-asoc-card.c:(.text+0x1467b5): undefined reference to `imx_audmux_v2_configure_port' >> fsl-asoc-card.c:(.text+0x1467d0): undefined reference to `imx_audmux_v2_configure_port' >> fsl-asoc-card.c:(.text+0x1467ed): undefined reference to `imx_audmux_v2_configure_port' >> fsl-asoc-card.c:(.text+0x146807): undefined reference to `imx_audmux_v2_configure_port' Update Kconfig to select SND_SOC_IMX_AUDMUX when SND_SOC_FSL_ASOC_CARD=y. Reported-by: kbuild test robot Signed-off-by: Shengjiu Wang Signed-off-by: Mark Brown --- sound/soc/fsl/Kconfig | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index fa90340..4698c01 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -62,6 +62,7 @@ config SND_SOC_FSL_UTILS config SND_SOC_FSL_ASOC_CARD tristate "Generic ASoC Sound Card with ASRC support" depends on OF && I2C + select SND_SOC_IMX_AUDMUX select SND_SOC_IMX_PCM_DMA select SND_SOC_FSL_ESAI select SND_SOC_FSL_SAI -- cgit v1.1 From 8ea21348868f37f5b2e6ebbaf336d2a415b2b9ff Mon Sep 17 00:00:00 2001 From: Xiubo Li Date: Mon, 18 Aug 2014 15:00:15 +0800 Subject: ASoC: simple-card: Fix the compile warning. sound/soc/generic/simple-card.c: In function simple_card_dai_link_of: sound/soc/generic/simple-card.c:198:10: warning: passing argument 3 of asoc_simple_card_sub_parse_of from incompatible pointer type [enabled by default] &dai_link->cpu_dai_name); ^ sound/soc/generic/simple-card.c:112:1: note: expected const struct device_node ** but argument is of type struct device_node ** asoc_simple_card_sub_parse_of(struct device_node *np, ^ sound/soc/generic/simple-card.c:229:10: warning: passing argument 3 of asoc_simple_card_sub_parse_of from incompatible pointer type [enabled by default] &dai_link->codec_dai_name); ^ sound/soc/generic/simple-card.c:112:1: note: expected const struct device_node ** but argument is of type struct device_node ** asoc_simple_card_sub_parse_of(struct device_node *np, ^ Since the asoc_simple_card_sub_parse_of() is used in simple-card module only, and the third argument is just used to get the node ponters address, so there is no need it must to be 'const' type. Signed-off-by: Xiubo Li Signed-off-by: Mark Brown --- sound/soc/generic/simple-card.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 159e517f..21b0ea2 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -111,7 +111,7 @@ static int asoc_simple_card_dai_init(struct snd_soc_pcm_runtime *rtd) static int asoc_simple_card_sub_parse_of(struct device_node *np, struct asoc_simple_dai *dai, - const struct device_node **p_node, + struct device_node **p_node, const char **name) { struct device_node *node; -- cgit v1.1 From cdec729765659adafba983d6b6760ad52c71d5d8 Mon Sep 17 00:00:00 2001 From: Sean Cross Date: Tue, 19 Aug 2014 12:49:34 +0800 Subject: ASoC: fsl: Fix building of imx-es8328 on PPC The imx-es8328 driver fails to build on PPC because it explicitly depends on SND_SOC_IMX_PCM_FIQ, which itself doesn't build on PPC. Instead, rely on the SND_SOC_FSL_SSI config option to pull in the necessary libraries. While we're at it, remove SND_SOC_FSL_UTILS, which also is not needed. Signed-off-by: Sean Cross Signed-off-by: Mark Brown --- sound/soc/fsl/Kconfig | 2 -- 1 file changed, 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 4698c01..3154f43 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -266,8 +266,6 @@ config SND_SOC_IMX_ES8328 select SND_SOC_IMX_PCM_DMA select SND_SOC_IMX_AUDMUX select SND_SOC_FSL_SSI - select SND_SOC_FSL_UTILS - select SND_SOC_IMX_PCM_FIQ help Say Y if you want to add support for the ES8328 audio codec connected via SSI/I2S over either SPI or I2C. -- cgit v1.1 From 81c7cfd1b22a0ee5e40efef72ec2cd17dbf12e6d Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 19 Aug 2014 15:51:18 +0200 Subject: ASoC: Move debugfs registration to the component level The debugfs registration is mostly identical between platforms and CODECs. This patches consolidates the two implementations at the component level. Unfortunately there are still a couple of CODEC specific debugfs files that are related to legacy ASoC IO that need to be registered. For this a new callback is added to the component struct that will be initialized when a CODEC is registered and will be used to register the CODEC specific files. Once there are no drivers left using legacy IO this can be removed again. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 122 ++++++++++++++++++++++----------------------------- 1 file changed, 52 insertions(+), 70 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index d4bfd4a..79371a7 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -270,79 +270,56 @@ static const struct file_operations codec_reg_fops = { .llseek = default_llseek, }; -static struct dentry *soc_debugfs_create_dir(struct dentry *parent, - const char *fmt, ...) +static void soc_init_component_debugfs(struct snd_soc_component *component) { - struct dentry *de; - va_list ap; - char *s; + if (component->debugfs_prefix) { + char *name; - va_start(ap, fmt); - s = kvasprintf(GFP_KERNEL, fmt, ap); - va_end(ap); + name = kasprintf(GFP_KERNEL, "%s:%s", + component->debugfs_prefix, component->name); + if (name) { + component->debugfs_root = debugfs_create_dir(name, + component->card->debugfs_card_root); + kfree(name); + } + } else { + component->debugfs_root = debugfs_create_dir(component->name, + component->card->debugfs_card_root); + } - if (!s) - return NULL; + if (!component->debugfs_root) { + dev_warn(component->dev, + "ASoC: Failed to create component debugfs directory\n"); + return; + } - de = debugfs_create_dir(s, parent); - kfree(s); + snd_soc_dapm_debugfs_init(snd_soc_component_get_dapm(component), + component->debugfs_root); - return de; + if (component->init_debugfs) + component->init_debugfs(component); } -static void soc_init_codec_debugfs(struct snd_soc_codec *codec) +static void soc_cleanup_component_debugfs(struct snd_soc_component *component) { - struct dentry *debugfs_card_root = codec->component.card->debugfs_card_root; + debugfs_remove_recursive(component->debugfs_root); +} - codec->debugfs_codec_root = soc_debugfs_create_dir(debugfs_card_root, - "codec:%s", - codec->component.name); - if (!codec->debugfs_codec_root) { - dev_warn(codec->dev, - "ASoC: Failed to create codec debugfs directory\n"); - return; - } +static void soc_init_codec_debugfs(struct snd_soc_component *component) +{ + struct snd_soc_codec *codec = snd_soc_component_to_codec(component); - debugfs_create_bool("cache_sync", 0444, codec->debugfs_codec_root, + debugfs_create_bool("cache_sync", 0444, codec->component.debugfs_root, &codec->cache_sync); - debugfs_create_bool("cache_only", 0444, codec->debugfs_codec_root, + debugfs_create_bool("cache_only", 0444, codec->component.debugfs_root, &codec->cache_only); codec->debugfs_reg = debugfs_create_file("codec_reg", 0644, - codec->debugfs_codec_root, + codec->component.debugfs_root, codec, &codec_reg_fops); if (!codec->debugfs_reg) dev_warn(codec->dev, "ASoC: Failed to create codec register debugfs file\n"); - - snd_soc_dapm_debugfs_init(&codec->dapm, codec->debugfs_codec_root); -} - -static void soc_cleanup_codec_debugfs(struct snd_soc_codec *codec) -{ - debugfs_remove_recursive(codec->debugfs_codec_root); -} - -static void soc_init_platform_debugfs(struct snd_soc_platform *platform) -{ - struct dentry *debugfs_card_root = platform->component.card->debugfs_card_root; - - platform->debugfs_platform_root = soc_debugfs_create_dir(debugfs_card_root, - "platform:%s", - platform->component.name); - if (!platform->debugfs_platform_root) { - dev_warn(platform->dev, - "ASoC: Failed to create platform debugfs directory\n"); - return; - } - - snd_soc_dapm_debugfs_init(&platform->component.dapm, - platform->debugfs_platform_root); -} - -static void soc_cleanup_platform_debugfs(struct snd_soc_platform *platform) -{ - debugfs_remove_recursive(platform->debugfs_platform_root); } static ssize_t codec_list_read_file(struct file *file, char __user *user_buf, @@ -474,19 +451,15 @@ static void soc_cleanup_card_debugfs(struct snd_soc_card *card) #else -static inline void soc_init_codec_debugfs(struct snd_soc_codec *codec) -{ -} +#define soc_init_codec_debugfs NULL -static inline void soc_cleanup_codec_debugfs(struct snd_soc_codec *codec) +static inline void soc_init_component_debugfs( + struct snd_soc_component *component) { } -static inline void soc_init_platform_debugfs(struct snd_soc_platform *platform) -{ -} - -static inline void soc_cleanup_platform_debugfs(struct snd_soc_platform *platform) +static inline void soc_cleanup_component_debugfs( + struct snd_soc_component *component) { } @@ -1026,7 +999,7 @@ static int soc_remove_platform(struct snd_soc_platform *platform) /* Make sure all DAPM widgets are freed */ snd_soc_dapm_free(&platform->component.dapm); - soc_cleanup_platform_debugfs(platform); + soc_cleanup_component_debugfs(&platform->component); platform->probed = 0; module_put(platform->dev->driver->owner); @@ -1046,7 +1019,7 @@ static void soc_remove_codec(struct snd_soc_codec *codec) /* Make sure all DAPM widgets are freed */ snd_soc_dapm_free(&codec->dapm); - soc_cleanup_codec_debugfs(codec); + soc_cleanup_component_debugfs(&codec->component); codec->probed = 0; list_del(&codec->card_list); module_put(codec->dev->driver->owner); @@ -1187,7 +1160,7 @@ static int soc_probe_codec(struct snd_soc_card *card, if (!try_module_get(codec->dev->driver->owner)) return -ENODEV; - soc_init_codec_debugfs(codec); + soc_init_component_debugfs(&codec->component); if (driver->dapm_widgets) { ret = snd_soc_dapm_new_controls(&codec->dapm, @@ -1242,7 +1215,7 @@ static int soc_probe_codec(struct snd_soc_card *card, return 0; err_probe: - soc_cleanup_codec_debugfs(codec); + soc_cleanup_component_debugfs(&codec->component); module_put(codec->dev->driver->owner); return ret; @@ -1262,7 +1235,7 @@ static int soc_probe_platform(struct snd_soc_card *card, if (!try_module_get(platform->dev->driver->owner)) return -ENODEV; - soc_init_platform_debugfs(platform); + soc_init_component_debugfs(&platform->component); if (driver->dapm_widgets) snd_soc_dapm_new_controls(&platform->component.dapm, @@ -1302,7 +1275,7 @@ static int soc_probe_platform(struct snd_soc_card *card, return 0; err_probe: - soc_cleanup_platform_debugfs(platform); + soc_cleanup_component_debugfs(&platform->component); module_put(platform->dev->driver->owner); return ret; @@ -4266,6 +4239,10 @@ int snd_soc_add_platform(struct device *dev, struct snd_soc_platform *platform, if (platform_drv->read) platform->component.read = snd_soc_platform_drv_read; +#ifdef CONFIG_DEBUG_FS + platform->component.debugfs_prefix = "platform"; +#endif + mutex_lock(&client_mutex); snd_soc_component_add_unlocked(&platform->component); list_add(&platform->list, &platform_list); @@ -4455,6 +4432,11 @@ int snd_soc_register_codec(struct device *dev, codec->component.val_bytes = codec_drv->reg_word_size; mutex_init(&codec->mutex); +#ifdef CONFIG_DEBUG_FS + codec->component.init_debugfs = soc_init_codec_debugfs; + codec->component.debugfs_prefix = "codec"; +#endif + if (!codec->component.write) { if (codec_drv->get_regmap) regmap = codec_drv->get_regmap(dev); -- cgit v1.1 From f1d45cc3ae96a6173129b2c164c216272faa5fc0 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 19 Aug 2014 15:51:19 +0200 Subject: ASoC: Consolidate platform and CODEC probe/remove The platform and CODEC probe and remove code is now largely identical. This patch consolidates it at the component level. The resulting code is slightly larger due to all the boiler plate code setting up the indirection for the table based control and DAPM registration. Once all drivers have been update to no longer use the snd_soc_codec_driver and snd_soc_platform_driver specific fields for this the indirection can be removed again. This patch contains two noteworthy hacks that are only meant to be temporary to be able to update drivers and the core in separate incremental patches. The first hack is related to that some DPCM platforms expect that the DAPM widgets for the DAIs of a snd_soc_component are created in the DAPM context of the snd_soc_platform that has the same parent device. For handling this the steal_sibling_dai_widgets attribute is introduced. It gets set for snd_soc_platforms that register DAPM elements. When creating the DAI widgets for a component this flag is checked and if it is found on one of the siblings the component will not create any DAI widgets in its own DAPM context. If the attribute is set on a platform it will look for siblings components and create DAI widgets for them in its own context. The fix for this will be to update the offending drivers to only register a single component rather than two. The second hack deals with the fact that the ASoC card suspend and resume code still needs a list of CODECs that have been registered for the card. To handle this the generic probe and remove path have a check to see if the component is CODEC and if yes add/remove it to the card's CODEC list. While it is possible to clean up the suspend/resume code to not need the CODEC list anymore this is a bit of a chicken and egg problem since it will become easier to clean up the suspend/resume code once there is a unified component layer. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 335 ++++++++++++++++++---------------- sound/soc/soc-generic-dmaengine-pcm.c | 4 +- 2 files changed, 177 insertions(+), 162 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 79371a7..b833cc6 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -985,44 +985,20 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num) return 0; } -static int soc_remove_platform(struct snd_soc_platform *platform) +static void soc_remove_component(struct snd_soc_component *component) { - int ret; - - if (platform->driver->remove) { - ret = platform->driver->remove(platform); - if (ret < 0) - dev_err(platform->dev, "ASoC: failed to remove %d\n", - ret); - } - - /* Make sure all DAPM widgets are freed */ - snd_soc_dapm_free(&platform->component.dapm); - - soc_cleanup_component_debugfs(&platform->component); - platform->probed = 0; - module_put(platform->dev->driver->owner); - - return 0; -} - -static void soc_remove_codec(struct snd_soc_codec *codec) -{ - int err; + /* This is a HACK and will be removed soon */ + if (component->codec) + list_del(&component->codec->card_list); - if (codec->driver->remove) { - err = codec->driver->remove(codec); - if (err < 0) - dev_err(codec->dev, "ASoC: failed to remove %d\n", err); - } + if (component->remove) + component->remove(component); - /* Make sure all DAPM widgets are freed */ - snd_soc_dapm_free(&codec->dapm); + snd_soc_dapm_free(snd_soc_component_get_dapm(component)); - soc_cleanup_component_debugfs(&codec->component); - codec->probed = 0; - list_del(&codec->card_list); - module_put(codec->dev->driver->owner); + soc_cleanup_component_debugfs(component); + component->probed = 0; + module_put(component->dev->driver->owner); } static void soc_remove_codec_dai(struct snd_soc_dai *codec_dai, int order) @@ -1086,25 +1062,24 @@ static void soc_remove_link_components(struct snd_soc_card *card, int num, int i; /* remove the platform */ - if (platform && platform->probed && - platform->driver->remove_order == order) { - soc_remove_platform(platform); - } + if (platform && platform->component.probed && + platform->component.driver->remove_order == order) + soc_remove_component(&platform->component); /* remove the CODEC-side CODEC */ for (i = 0; i < rtd->num_codecs; i++) { codec = rtd->codec_dais[i]->codec; - if (codec && codec->probed && - codec->driver->remove_order == order) - soc_remove_codec(codec); + if (codec && codec->component.probed && + codec->component.driver->remove_order == order) + soc_remove_component(&codec->component); } /* remove any CPU-side CODEC */ if (cpu_dai) { codec = cpu_dai->codec; - if (codec && codec->probed && - codec->driver->remove_order == order) - soc_remove_codec(codec); + if (codec && codec->component.probed && + codec->component.driver->remove_order == order) + soc_remove_component(&codec->component); } } @@ -1146,137 +1121,108 @@ static void soc_set_name_prefix(struct snd_soc_card *card, } } -static int soc_probe_codec(struct snd_soc_card *card, - struct snd_soc_codec *codec) +static int soc_probe_component(struct snd_soc_card *card, + struct snd_soc_component *component) { - int ret = 0; - const struct snd_soc_codec_driver *driver = codec->driver; + struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); + struct snd_soc_component *dai_component, *component2; struct snd_soc_dai *dai; + int ret; - codec->component.card = card; - codec->dapm.card = card; - soc_set_name_prefix(card, &codec->component); + component->card = card; + dapm->card = card; + soc_set_name_prefix(card, component); - if (!try_module_get(codec->dev->driver->owner)) + if (!try_module_get(component->dev->driver->owner)) return -ENODEV; - soc_init_component_debugfs(&codec->component); + soc_init_component_debugfs(component); - if (driver->dapm_widgets) { - ret = snd_soc_dapm_new_controls(&codec->dapm, - driver->dapm_widgets, - driver->num_dapm_widgets); + if (component->dapm_widgets) { + ret = snd_soc_dapm_new_controls(dapm, component->dapm_widgets, + component->num_dapm_widgets); if (ret != 0) { - dev_err(codec->dev, + dev_err(component->dev, "Failed to create new controls %d\n", ret); goto err_probe; } } - /* Create DAPM widgets for each DAI stream */ - list_for_each_entry(dai, &codec->component.dai_list, list) { - ret = snd_soc_dapm_new_dai_widgets(&codec->dapm, dai); + /* + * This is rather ugly, but certain platforms expect that the DAPM + * widgets for the DAIs for components with the same parent device are + * created in the platforms DAPM context. Until that is fixed we need to + * keep this. + */ + if (component->steal_sibling_dai_widgets) { + dai_component = NULL; + list_for_each_entry(component2, &component_list, list) { + if (component == component2) + continue; - if (ret != 0) { - dev_err(codec->dev, - "Failed to create DAI widgets %d\n", ret); - goto err_probe; + if (component2->dev == component->dev && + !list_empty(&component2->dai_list)) { + dai_component = component2; + break; + } } - } - - codec->dapm.idle_bias_off = driver->idle_bias_off; - - if (driver->probe) { - ret = driver->probe(codec); - if (ret < 0) { - dev_err(codec->dev, - "ASoC: failed to probe CODEC %d\n", ret); - goto err_probe; + } else { + dai_component = component; + list_for_each_entry(component2, &component_list, list) { + if (component2->dev == component->dev && + component2->steal_sibling_dai_widgets) { + dai_component = NULL; + break; + } } - WARN(codec->dapm.idle_bias_off && - codec->dapm.bias_level != SND_SOC_BIAS_OFF, - "codec %s can not start from non-off bias with idle_bias_off==1\n", - codec->component.name); } - if (driver->controls) - snd_soc_add_codec_controls(codec, driver->controls, - driver->num_controls); - if (driver->dapm_routes) - snd_soc_dapm_add_routes(&codec->dapm, driver->dapm_routes, - driver->num_dapm_routes); - - /* mark codec as probed and add to card codec list */ - codec->probed = 1; - list_add(&codec->card_list, &card->codec_dev_list); - list_add(&codec->dapm.list, &card->dapm_list); - - return 0; - -err_probe: - soc_cleanup_component_debugfs(&codec->component); - module_put(codec->dev->driver->owner); - - return ret; -} - -static int soc_probe_platform(struct snd_soc_card *card, - struct snd_soc_platform *platform) -{ - int ret = 0; - const struct snd_soc_platform_driver *driver = platform->driver; - struct snd_soc_component *component; - struct snd_soc_dai *dai; - - platform->component.card = card; - platform->component.dapm.card = card; - - if (!try_module_get(platform->dev->driver->owner)) - return -ENODEV; - - soc_init_component_debugfs(&platform->component); - - if (driver->dapm_widgets) - snd_soc_dapm_new_controls(&platform->component.dapm, - driver->dapm_widgets, driver->num_dapm_widgets); - - /* Create DAPM widgets for each DAI stream */ - list_for_each_entry(component, &component_list, list) { - if (component->dev != platform->dev) - continue; - list_for_each_entry(dai, &component->dai_list, list) - snd_soc_dapm_new_dai_widgets(&platform->component.dapm, - dai); + if (dai_component) { + list_for_each_entry(dai, &dai_component->dai_list, list) { + snd_soc_dapm_new_dai_widgets(dapm, dai); + if (ret != 0) { + dev_err(component->dev, + "Failed to create DAI widgets %d\n", + ret); + goto err_probe; + } + } } - platform->component.dapm.idle_bias_off = 1; - - if (driver->probe) { - ret = driver->probe(platform); + if (component->probe) { + ret = component->probe(component); if (ret < 0) { - dev_err(platform->dev, - "ASoC: failed to probe platform %d\n", ret); + dev_err(component->dev, + "ASoC: failed to probe component %d\n", ret); goto err_probe; } + + WARN(dapm->idle_bias_off && + dapm->bias_level != SND_SOC_BIAS_OFF, + "codec %s can not start from non-off bias with idle_bias_off==1\n", + component->name); } - if (driver->controls) - snd_soc_add_platform_controls(platform, driver->controls, - driver->num_controls); - if (driver->dapm_routes) - snd_soc_dapm_add_routes(&platform->component.dapm, - driver->dapm_routes, driver->num_dapm_routes); + if (component->controls) + snd_soc_add_component_controls(component, component->controls, + component->num_controls); + if (component->dapm_routes) + snd_soc_dapm_add_routes(dapm, component->dapm_routes, + component->num_dapm_routes); - /* mark platform as probed and add to card platform list */ - platform->probed = 1; - list_add(&platform->component.dapm.list, &card->dapm_list); + component->probed = 1; + list_add(&dapm->list, &card->dapm_list); + + /* This is a HACK and will be removed soon */ + if (component->codec) + list_add(&component->codec->card_list, &card->codec_dev_list); return 0; err_probe: - soc_cleanup_component_debugfs(&platform->component); - module_put(platform->dev->driver->owner); + soc_cleanup_component_debugfs(component); + module_put(component->dev->driver->owner); return ret; } @@ -1334,33 +1280,36 @@ static int soc_probe_link_components(struct snd_soc_card *card, int num, int order) { struct snd_soc_pcm_runtime *rtd = &card->rtd[num]; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct snd_soc_platform *platform = rtd->platform; + struct snd_soc_component *component; int i, ret; /* probe the CPU-side component, if it is a CODEC */ - if (cpu_dai->codec && - !cpu_dai->codec->probed && - cpu_dai->codec->driver->probe_order == order) { - ret = soc_probe_codec(card, cpu_dai->codec); - if (ret < 0) - return ret; + if (rtd->cpu_dai->codec) { + component = &rtd->cpu_dai->codec->component; + if (!component->probed && + component->driver->probe_order == order) { + ret = soc_probe_component(card, component); + if (ret < 0) + return ret; + } } /* probe the CODEC-side components */ for (i = 0; i < rtd->num_codecs; i++) { - if (!rtd->codec_dais[i]->codec->probed && - rtd->codec_dais[i]->codec->driver->probe_order == order) { - ret = soc_probe_codec(card, rtd->codec_dais[i]->codec); + component = &rtd->codec_dais[i]->codec->component; + if (!component->probed && + component->driver->probe_order == order) { + ret = soc_probe_component(card, component); if (ret < 0) return ret; } } /* probe the platform */ - if (!platform->probed && - platform->driver->probe_order == order) { - ret = soc_probe_platform(card, platform); + if (!platform->component.probed && + platform->component.driver->probe_order == order) { + ret = soc_probe_component(card, &platform->component); if (ret < 0) return ret; } @@ -1647,12 +1596,12 @@ static int soc_probe_aux_dev(struct snd_soc_card *card, int num) struct snd_soc_aux_dev *aux_dev = &card->aux_dev[num]; int ret; - if (rtd->codec->probed) { + if (rtd->codec->component.probed) { dev_err(rtd->codec->dev, "ASoC: codec already probed\n"); return -EBUSY; } - ret = soc_probe_codec(card, rtd->codec); + ret = soc_probe_component(card, &rtd->codec->component); if (ret < 0) return ret; @@ -1681,8 +1630,8 @@ static void soc_remove_aux_dev(struct snd_soc_card *card, int num) rtd->dev_registered = 0; } - if (codec && codec->probed) - soc_remove_codec(codec); + if (codec && codec->component.probed) + soc_remove_component(&codec->component); } static int snd_soc_init_codec_cache(struct snd_soc_codec *codec) @@ -4198,6 +4147,20 @@ found: } EXPORT_SYMBOL_GPL(snd_soc_unregister_component); +static int snd_soc_platform_drv_probe(struct snd_soc_component *component) +{ + struct snd_soc_platform *platform = snd_soc_component_to_platform(component); + + return platform->driver->probe(platform); +} + +static void snd_soc_platform_drv_remove(struct snd_soc_component *component) +{ + struct snd_soc_platform *platform = snd_soc_component_to_platform(component); + + platform->driver->remove(platform); +} + static int snd_soc_platform_drv_write(struct snd_soc_component *component, unsigned int reg, unsigned int val) { @@ -4234,6 +4197,24 @@ int snd_soc_add_platform(struct device *dev, struct snd_soc_platform *platform, platform->dev = dev; platform->driver = platform_drv; + if (platform_drv->controls) { + platform->component.controls = platform_drv->controls; + platform->component.num_controls = platform_drv->num_controls; + } + if (platform_drv->dapm_widgets) { + platform->component.dapm_widgets = platform_drv->dapm_widgets; + platform->component.num_dapm_widgets = platform_drv->num_dapm_widgets; + platform->component.steal_sibling_dai_widgets = true; + } + if (platform_drv->dapm_routes) { + platform->component.dapm_routes = platform_drv->dapm_routes; + platform->component.num_dapm_routes = platform_drv->num_dapm_routes; + } + + if (platform_drv->probe) + platform->component.probe = snd_soc_platform_drv_probe; + if (platform_drv->remove) + platform->component.remove = snd_soc_platform_drv_remove; if (platform_drv->write) platform->component.write = snd_soc_platform_drv_write; if (platform_drv->read) @@ -4363,6 +4344,20 @@ static void fixup_codec_formats(struct snd_soc_pcm_stream *stream) stream->formats |= codec_format_map[i]; } +static int snd_soc_codec_drv_probe(struct snd_soc_component *component) +{ + struct snd_soc_codec *codec = snd_soc_component_to_codec(component); + + return codec->driver->probe(codec); +} + +static void snd_soc_codec_drv_remove(struct snd_soc_component *component) +{ + struct snd_soc_codec *codec = snd_soc_component_to_codec(component); + + codec->driver->remove(codec); +} + static int snd_soc_codec_drv_write(struct snd_soc_component *component, unsigned int reg, unsigned int val) { @@ -4411,12 +4406,30 @@ int snd_soc_register_codec(struct device *dev, return -ENOMEM; codec->component.dapm_ptr = &codec->dapm; + codec->component.codec = codec; ret = snd_soc_component_initialize(&codec->component, &codec_drv->component_driver, dev); if (ret) goto err_free; + if (codec_drv->controls) { + codec->component.controls = codec_drv->controls; + codec->component.num_controls = codec_drv->num_controls; + } + if (codec_drv->dapm_widgets) { + codec->component.dapm_widgets = codec_drv->dapm_widgets; + codec->component.num_dapm_widgets = codec_drv->num_dapm_widgets; + } + if (codec_drv->dapm_routes) { + codec->component.dapm_routes = codec_drv->dapm_routes; + codec->component.num_dapm_routes = codec_drv->num_dapm_routes; + } + + if (codec_drv->probe) + codec->component.probe = snd_soc_codec_drv_probe; + if (codec_drv->remove) + codec->component.remove = snd_soc_codec_drv_remove; if (codec_drv->write) codec->component.write = snd_soc_codec_drv_write; if (codec_drv->read) diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index 6307f85..b329b84 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -336,10 +336,12 @@ static const struct snd_pcm_ops dmaengine_pcm_ops = { }; static const struct snd_soc_platform_driver dmaengine_pcm_platform = { + .component_driver = { + .probe_order = SND_SOC_COMP_ORDER_LATE, + }, .ops = &dmaengine_pcm_ops, .pcm_new = dmaengine_pcm_new, .pcm_free = dmaengine_pcm_free, - .probe_order = SND_SOC_COMP_ORDER_LATE, }; static const char * const dmaengine_pcm_dma_channel_names[] = { -- cgit v1.1 From 93c3ce76ccced3a8718149e8734ccaa931e9a1f1 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 19 Aug 2014 15:51:20 +0200 Subject: ASoC: Make rtd->codec optional There are some place in the ASoC core that expect rtd->codec to be non NULL (mainly CODEC specific sysfs files). With componentization going forward rtd->codec might be NULL in some cases. This patch prepares the core for this by not registering CODEC specific sysfs files if rtd->codec is NULL. sysfs file removal does not need to be conditionalized as it handles the removal of non-existing files just fine. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 24 ++++++++++++++---------- 1 file changed, 14 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index b833cc6..1c705c2 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1261,17 +1261,21 @@ static int soc_post_component_init(struct snd_soc_pcm_runtime *rtd, } rtd->dev_registered = 1; - /* add DAPM sysfs entries for this codec */ - ret = snd_soc_dapm_sys_add(rtd->dev); - if (ret < 0) - dev_err(rtd->dev, - "ASoC: failed to add codec dapm sysfs entries: %d\n", ret); + if (rtd->codec) { + /* add DAPM sysfs entries for this codec */ + ret = snd_soc_dapm_sys_add(rtd->dev); + if (ret < 0) + dev_err(rtd->dev, + "ASoC: failed to add codec dapm sysfs entries: %d\n", + ret); - /* add codec sysfs entries */ - ret = device_create_file(rtd->dev, &dev_attr_codec_reg); - if (ret < 0) - dev_err(rtd->dev, - "ASoC: failed to add codec sysfs files: %d\n", ret); + /* add codec sysfs entries */ + ret = device_create_file(rtd->dev, &dev_attr_codec_reg); + if (ret < 0) + dev_err(rtd->dev, + "ASoC: failed to add codec sysfs files: %d\n", + ret); + } return 0; } -- cgit v1.1 From 61aca5646b736a794d40de29a197144db3f0c5ba Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 19 Aug 2014 15:51:21 +0200 Subject: ASoC: Add component level probe/remove support Now that we have a unified probe and remove path make sure to call them for all components. soc_{probe,remove}_component are responsible for setting up the DAPM context for the component, initialize the component prefix, manage the debugfs entries as well as do the registration of table based controls and DAPM elements. They also call the component drivers probe and remove callbacks. This patch makes these things available for generic snd_soc_component drivers rather than only having them for snd_soc_codec and snd_soc_platform drivers. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 42 ++++++++++++++++++++++++------------------ 1 file changed, 24 insertions(+), 18 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 1c705c2..08fd85e 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1058,7 +1058,7 @@ static void soc_remove_link_components(struct snd_soc_card *card, int num, struct snd_soc_pcm_runtime *rtd = &card->rtd[num]; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct snd_soc_platform *platform = rtd->platform; - struct snd_soc_codec *codec; + struct snd_soc_component *component; int i; /* remove the platform */ @@ -1068,18 +1068,17 @@ static void soc_remove_link_components(struct snd_soc_card *card, int num, /* remove the CODEC-side CODEC */ for (i = 0; i < rtd->num_codecs; i++) { - codec = rtd->codec_dais[i]->codec; - if (codec && codec->component.probed && - codec->component.driver->remove_order == order) - soc_remove_component(&codec->component); + component = rtd->codec_dais[i]->component; + if (component->probed && + component->driver->remove_order == order) + soc_remove_component(component); } /* remove any CPU-side CODEC */ if (cpu_dai) { - codec = cpu_dai->codec; - if (codec && codec->component.probed && - codec->component.driver->remove_order == order) - soc_remove_component(&codec->component); + if (cpu_dai->component->probed && + cpu_dai->component->driver->remove_order == order) + soc_remove_component(cpu_dai->component); } } @@ -1289,19 +1288,17 @@ static int soc_probe_link_components(struct snd_soc_card *card, int num, int i, ret; /* probe the CPU-side component, if it is a CODEC */ - if (rtd->cpu_dai->codec) { - component = &rtd->cpu_dai->codec->component; - if (!component->probed && - component->driver->probe_order == order) { - ret = soc_probe_component(card, component); - if (ret < 0) - return ret; - } + component = rtd->cpu_dai->component; + if (!component->probed && + component->driver->probe_order == order) { + ret = soc_probe_component(card, component); + if (ret < 0) + return ret; } /* probe the CODEC-side components */ for (i = 0; i < rtd->num_codecs; i++) { - component = &rtd->codec_dais[i]->codec->component; + component = rtd->codec_dais[i]->component; if (!component->probed && component->driver->probe_order == order) { ret = soc_probe_component(card, component); @@ -4042,6 +4039,8 @@ static int snd_soc_component_initialize(struct snd_soc_component *component, component->dev = dev; component->driver = driver; + component->probe = component->driver->probe; + component->remove = component->driver->remove; if (!component->dapm_ptr) component->dapm_ptr = &component->dapm; @@ -4055,6 +4054,13 @@ static int snd_soc_component_initialize(struct snd_soc_component *component, if (driver->stream_event) dapm->stream_event = snd_soc_component_stream_event; + component->controls = driver->controls; + component->num_controls = driver->num_controls; + component->dapm_widgets = driver->dapm_widgets; + component->num_dapm_widgets = driver->num_dapm_widgets; + component->dapm_routes = driver->dapm_routes; + component->num_dapm_routes = driver->num_dapm_routes; + INIT_LIST_HEAD(&component->dai_list); mutex_init(&component->io_mutex); -- cgit v1.1 From 65d9361f0cb50a20641802ee3075145d72e4409c Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 19 Aug 2014 15:51:22 +0200 Subject: ASoC: Move AUX dev support to the component level This patch makes it possible to register arbitrary components as a AUX dev for a card. This was previously only possible for CODEC components. With componentization having made it possible for components to have DAPM contexts and controls there is no reason why AUX devs should be artificially limited to snd_soc_codec devices. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 48 ++++++++++++++++++++++++++++++++++++------------ 1 file changed, 36 insertions(+), 12 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 08fd85e..08c04f4 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -860,6 +860,23 @@ EXPORT_SYMBOL_GPL(snd_soc_resume); static const struct snd_soc_dai_ops null_dai_ops = { }; +static struct snd_soc_component *soc_find_component( + const struct device_node *of_node, const char *name) +{ + struct snd_soc_component *component; + + list_for_each_entry(component, &component_list, list) { + if (of_node) { + if (component->dev->of_node == of_node) + return component; + } else if (strcmp(component->name, name) == 0) { + return component; + } + } + + return NULL; +} + static struct snd_soc_codec *soc_find_codec( const struct device_node *codec_of_node, const char *codec_name) @@ -1577,17 +1594,24 @@ static int soc_bind_aux_dev(struct snd_soc_card *card, int num) { struct snd_soc_pcm_runtime *rtd = &card->rtd_aux[num]; struct snd_soc_aux_dev *aux_dev = &card->aux_dev[num]; - const char *codecname = aux_dev->codec_name; + const char *name = aux_dev->codec_name; - rtd->codec = soc_find_codec(aux_dev->codec_of_node, codecname); - if (!rtd->codec) { + rtd->component = soc_find_component(aux_dev->codec_of_node, name); + if (!rtd->component) { if (aux_dev->codec_of_node) - codecname = of_node_full_name(aux_dev->codec_of_node); + name = of_node_full_name(aux_dev->codec_of_node); - dev_err(card->dev, "ASoC: %s not registered\n", codecname); + dev_err(card->dev, "ASoC: %s not registered\n", name); return -EPROBE_DEFER; } + /* + * Some places still reference rtd->codec, so we have to keep that + * initialized if the component is a CODEC. Once all those references + * have been removed, this code can be removed as well. + */ + rtd->codec = rtd->component->codec; + return 0; } @@ -1597,18 +1621,18 @@ static int soc_probe_aux_dev(struct snd_soc_card *card, int num) struct snd_soc_aux_dev *aux_dev = &card->aux_dev[num]; int ret; - if (rtd->codec->component.probed) { - dev_err(rtd->codec->dev, "ASoC: codec already probed\n"); + if (rtd->component->probed) { + dev_err(rtd->dev, "ASoC: codec already probed\n"); return -EBUSY; } - ret = soc_probe_component(card, &rtd->codec->component); + ret = soc_probe_component(card, rtd->component); if (ret < 0) return ret; /* do machine specific initialization */ if (aux_dev->init) { - ret = aux_dev->init(&rtd->codec->dapm); + ret = aux_dev->init(snd_soc_component_get_dapm(rtd->component)); if (ret < 0) { dev_err(card->dev, "ASoC: failed to init %s: %d\n", aux_dev->name, ret); @@ -1622,7 +1646,7 @@ static int soc_probe_aux_dev(struct snd_soc_card *card, int num) static void soc_remove_aux_dev(struct snd_soc_card *card, int num) { struct snd_soc_pcm_runtime *rtd = &card->rtd_aux[num]; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_component *component = rtd->component; /* unregister the rtd device */ if (rtd->dev_registered) { @@ -1631,8 +1655,8 @@ static void soc_remove_aux_dev(struct snd_soc_card *card, int num) rtd->dev_registered = 0; } - if (codec && codec->component.probed) - soc_remove_component(&codec->component); + if (component && component->probed) + soc_remove_component(component); } static int snd_soc_init_codec_cache(struct snd_soc_codec *codec) -- cgit v1.1 From 57bf772687700e206c760ba2e4097f78bde97887 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 19 Aug 2014 15:51:23 +0200 Subject: ASoC: Pass component instead of DAPM context to AUX dev init callback Given that the component is the containing structure it makes more sense to pass the component rather than the DAPM context to the AUX dev init callback. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/samsung/speyside.c | 6 ++++-- sound/soc/soc-core.c | 2 +- 2 files changed, 5 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/speyside.c b/sound/soc/samsung/speyside.c index 9902efc..a054826 100644 --- a/sound/soc/samsung/speyside.c +++ b/sound/soc/samsung/speyside.c @@ -228,10 +228,12 @@ static struct snd_soc_dai_link speyside_dai[] = { }, }; -static int speyside_wm9081_init(struct snd_soc_dapm_context *dapm) +static int speyside_wm9081_init(struct snd_soc_component *component) { + struct snd_soc_codec *codec = snd_soc_component_to_codec(component); + /* At any time the WM9081 is active it will have this clock */ - return snd_soc_codec_set_sysclk(dapm->codec, WM9081_SYSCLK_MCLK, 0, + return snd_soc_codec_set_sysclk(codec, WM9081_SYSCLK_MCLK, 0, MCLK_AUDIO_RATE, 0); } diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 08c04f4..4393bc3 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1632,7 +1632,7 @@ static int soc_probe_aux_dev(struct snd_soc_card *card, int num) /* do machine specific initialization */ if (aux_dev->init) { - ret = aux_dev->init(snd_soc_component_get_dapm(rtd->component)); + ret = aux_dev->init(rtd->component); if (ret < 0) { dev_err(card->dev, "ASoC: failed to init %s: %d\n", aux_dev->name, ret); -- cgit v1.1 From 70090bbb8b7d7da7a6f64969b43a61c493c560ff Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 19 Aug 2014 15:51:24 +0200 Subject: ASoC: Move component->probed check into soc_{remove,probe}_component() Having the check in a centralized place makes the code a bit cleaner and shorter. Note: There is a slight semantic change in this patch. soc_probe_aux_dev() will no longer return -EBUSY if the AUX dev has already been probed before. This is fine though since it will simply do nothing in that case and return success. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 29 ++++++++++++----------------- 1 file changed, 12 insertions(+), 17 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 4393bc3..2fbfbfc 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1004,6 +1004,9 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num) static void soc_remove_component(struct snd_soc_component *component) { + if (!component->probed) + return; + /* This is a HACK and will be removed soon */ if (component->codec) list_del(&component->codec->card_list); @@ -1079,22 +1082,19 @@ static void soc_remove_link_components(struct snd_soc_card *card, int num, int i; /* remove the platform */ - if (platform && platform->component.probed && - platform->component.driver->remove_order == order) + if (platform && platform->component.driver->remove_order == order) soc_remove_component(&platform->component); /* remove the CODEC-side CODEC */ for (i = 0; i < rtd->num_codecs; i++) { component = rtd->codec_dais[i]->component; - if (component->probed && - component->driver->remove_order == order) + if (component->driver->remove_order == order) soc_remove_component(component); } /* remove any CPU-side CODEC */ if (cpu_dai) { - if (cpu_dai->component->probed && - cpu_dai->component->driver->remove_order == order) + if (cpu_dai->component->driver->remove_order == order) soc_remove_component(cpu_dai->component); } } @@ -1145,6 +1145,9 @@ static int soc_probe_component(struct snd_soc_card *card, struct snd_soc_dai *dai; int ret; + if (component->probed) + return 0; + component->card = card; dapm->card = card; soc_set_name_prefix(card, component); @@ -1306,8 +1309,7 @@ static int soc_probe_link_components(struct snd_soc_card *card, int num, /* probe the CPU-side component, if it is a CODEC */ component = rtd->cpu_dai->component; - if (!component->probed && - component->driver->probe_order == order) { + if (component->driver->probe_order == order) { ret = soc_probe_component(card, component); if (ret < 0) return ret; @@ -1316,8 +1318,7 @@ static int soc_probe_link_components(struct snd_soc_card *card, int num, /* probe the CODEC-side components */ for (i = 0; i < rtd->num_codecs; i++) { component = rtd->codec_dais[i]->component; - if (!component->probed && - component->driver->probe_order == order) { + if (component->driver->probe_order == order) { ret = soc_probe_component(card, component); if (ret < 0) return ret; @@ -1325,8 +1326,7 @@ static int soc_probe_link_components(struct snd_soc_card *card, int num, } /* probe the platform */ - if (!platform->component.probed && - platform->component.driver->probe_order == order) { + if (platform->component.driver->probe_order == order) { ret = soc_probe_component(card, &platform->component); if (ret < 0) return ret; @@ -1621,11 +1621,6 @@ static int soc_probe_aux_dev(struct snd_soc_card *card, int num) struct snd_soc_aux_dev *aux_dev = &card->aux_dev[num]; int ret; - if (rtd->component->probed) { - dev_err(rtd->dev, "ASoC: codec already probed\n"); - return -EBUSY; - } - ret = soc_probe_component(card, rtd->component); if (ret < 0) return ret; -- cgit v1.1 From ffbd7dd72bd3ad9bcae9190788c858e57f1e8e4e Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 19 Aug 2014 15:51:25 +0200 Subject: ASoC: Cleanup DAI module reference counting Currently when a DAI has no CODEC associated to it the reference on the module containing the DAI driver is increased when the DAI is probed and decrease when the DAI is removed. For DAIs with CODECs the module reference count was already incremented when the CODEC is probed. Now that all components have their module reference count incremented when they are probed and all DAIs do have a component it is possible to remove the module reference counting on DAI probe and removal. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 8 -------- 1 file changed, 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 2fbfbfc..4dc2876 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1067,8 +1067,6 @@ static void soc_remove_link_dais(struct snd_soc_card *card, int num, int order) cpu_dai->name, err); } cpu_dai->probed = 0; - if (!cpu_dai->codec) - module_put(cpu_dai->dev->driver->owner); } } @@ -1422,18 +1420,12 @@ static int soc_probe_link_dais(struct snd_soc_card *card, int num, int order) /* probe the cpu_dai */ if (!cpu_dai->probed && cpu_dai->driver->probe_order == order) { - if (!cpu_dai->codec) { - if (!try_module_get(cpu_dai->dev->driver->owner)) - return -ENODEV; - } - if (cpu_dai->driver->probe) { ret = cpu_dai->driver->probe(cpu_dai); if (ret < 0) { dev_err(cpu_dai->dev, "ASoC: failed to probe CPU DAI %s: %d\n", cpu_dai->name, ret); - module_put(cpu_dai->dev->driver->owner); return ret; } } -- cgit v1.1 From e60cd14f0bf6c004cd7032a24a036ba32d56e08a Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 19 Aug 2014 15:51:26 +0200 Subject: ASoC: Consolidate CPU and CODEC DAI removal CPU and CODEC DAI works exactly the same way. There is already a helper function for CODEC DAI removal, use that one as well for CPU DAI removal. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 34 +++++++++++----------------------- 1 file changed, 11 insertions(+), 23 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 4dc2876..5f6f978 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1021,28 +1021,27 @@ static void soc_remove_component(struct snd_soc_component *component) module_put(component->dev->driver->owner); } -static void soc_remove_codec_dai(struct snd_soc_dai *codec_dai, int order) +static void soc_remove_dai(struct snd_soc_dai *dai, int order) { int err; - if (codec_dai && codec_dai->probed && - codec_dai->driver->remove_order == order) { - if (codec_dai->driver->remove) { - err = codec_dai->driver->remove(codec_dai); + if (dai && dai->probed && + dai->driver->remove_order == order) { + if (dai->driver->remove) { + err = dai->driver->remove(dai); if (err < 0) - dev_err(codec_dai->dev, + dev_err(dai->dev, "ASoC: failed to remove %s: %d\n", - codec_dai->name, err); + dai->name, err); } - codec_dai->probed = 0; + dai->probed = 0; } } static void soc_remove_link_dais(struct snd_soc_card *card, int num, int order) { struct snd_soc_pcm_runtime *rtd = &card->rtd[num]; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - int i, err; + int i; /* unregister the rtd device */ if (rtd->dev_registered) { @@ -1054,20 +1053,9 @@ static void soc_remove_link_dais(struct snd_soc_card *card, int num, int order) /* remove the CODEC DAI */ for (i = 0; i < rtd->num_codecs; i++) - soc_remove_codec_dai(rtd->codec_dais[i], order); + soc_remove_dai(rtd->codec_dais[i], order); - /* remove the cpu_dai */ - if (cpu_dai && cpu_dai->probed && - cpu_dai->driver->remove_order == order) { - if (cpu_dai->driver->remove) { - err = cpu_dai->driver->remove(cpu_dai); - if (err < 0) - dev_err(cpu_dai->dev, - "ASoC: failed to remove %s: %d\n", - cpu_dai->name, err); - } - cpu_dai->probed = 0; - } + soc_remove_dai(rtd->cpu_dai, order); } static void soc_remove_link_components(struct snd_soc_card *card, int num, -- cgit v1.1 From 14621c7e5e72200ec021a7580121130ce7f2ff22 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 19 Aug 2014 15:51:27 +0200 Subject: ASoC: Consolidate CPU and CODEC DAI lookup The lookup of CPU and CODEC DAIs is fairly similar and can easily be consolidated into a single helper function. There are two main differences in the current implementation of the CPU and CODEC DAI lookup: 1) CPU DAIs can be looked up by the DAI name alone and do not necessarily require a component name/of_node. 2) The CODEC DAI search only considers DAIs from CODEC components. For 1) the new helper function will allow to lookup DAIs without providing a component name or of_node, but since snd_soc_register_card() already rejects CODEC DAI link components without neither a of_node or a name we'll never get into the situation where we try to lookup a CODEC DAI without a name/of_node. For 2) the new helper function just always considers all components. Componentization is now at a point where it is possible to register a CODEC as a snd_soc_component rather than a snd_soc_codec, by considering DAIs from all components it is possible to use such a CODEC in a DAI link. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 72 ++++++++++++++-------------------------------------- 1 file changed, 19 insertions(+), 53 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 5f6f978..140f43f 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -877,35 +877,23 @@ static struct snd_soc_component *soc_find_component( return NULL; } -static struct snd_soc_codec *soc_find_codec( - const struct device_node *codec_of_node, - const char *codec_name) +static struct snd_soc_dai *snd_soc_find_dai( + const struct snd_soc_dai_link_component *dlc) { - struct snd_soc_codec *codec; + struct snd_soc_component *component; + struct snd_soc_dai *dai; - list_for_each_entry(codec, &codec_list, list) { - if (codec_of_node) { - if (codec->dev->of_node != codec_of_node) - continue; - } else { - if (strcmp(codec->component.name, codec_name)) + /* Find CPU DAI from registered DAIs*/ + list_for_each_entry(component, &component_list, list) { + if (dlc->of_node && component->dev->of_node != dlc->of_node) + continue; + if (dlc->name && strcmp(dev_name(component->dev), dlc->name)) + continue; + list_for_each_entry(dai, &component->dai_list, list) { + if (dlc->dai_name && strcmp(dai->name, dlc->dai_name)) continue; - } - - return codec; - } - - return NULL; -} - -static struct snd_soc_dai *soc_find_codec_dai(struct snd_soc_codec *codec, - const char *codec_dai_name) -{ - struct snd_soc_dai *codec_dai; - list_for_each_entry(codec_dai, &codec->component.dai_list, list) { - if (!strcmp(codec_dai->name, codec_dai_name)) { - return codec_dai; + return dai; } } @@ -916,33 +904,19 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num) { struct snd_soc_dai_link *dai_link = &card->dai_link[num]; struct snd_soc_pcm_runtime *rtd = &card->rtd[num]; - struct snd_soc_component *component; struct snd_soc_dai_link_component *codecs = dai_link->codecs; + struct snd_soc_dai_link_component cpu_dai_component; struct snd_soc_dai **codec_dais = rtd->codec_dais; struct snd_soc_platform *platform; - struct snd_soc_dai *cpu_dai; const char *platform_name; int i; dev_dbg(card->dev, "ASoC: binding %s at idx %d\n", dai_link->name, num); - /* Find CPU DAI from registered DAIs*/ - list_for_each_entry(component, &component_list, list) { - if (dai_link->cpu_of_node && - component->dev->of_node != dai_link->cpu_of_node) - continue; - if (dai_link->cpu_name && - strcmp(dev_name(component->dev), dai_link->cpu_name)) - continue; - list_for_each_entry(cpu_dai, &component->dai_list, list) { - if (dai_link->cpu_dai_name && - strcmp(cpu_dai->name, dai_link->cpu_dai_name)) - continue; - - rtd->cpu_dai = cpu_dai; - } - } - + cpu_dai_component.name = dai_link->cpu_name; + cpu_dai_component.of_node = dai_link->cpu_of_node; + cpu_dai_component.dai_name = dai_link->cpu_dai_name; + rtd->cpu_dai = snd_soc_find_dai(&cpu_dai_component); if (!rtd->cpu_dai) { dev_err(card->dev, "ASoC: CPU DAI %s not registered\n", dai_link->cpu_dai_name); @@ -953,15 +927,7 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num) /* Find CODEC from registered CODECs */ for (i = 0; i < rtd->num_codecs; i++) { - struct snd_soc_codec *codec; - codec = soc_find_codec(codecs[i].of_node, codecs[i].name); - if (!codec) { - dev_err(card->dev, "ASoC: CODEC %s not registered\n", - codecs[i].name); - return -EPROBE_DEFER; - } - - codec_dais[i] = soc_find_codec_dai(codec, codecs[i].dai_name); + codec_dais[i] = snd_soc_find_dai(&codecs[i]); if (!codec_dais[i]) { dev_err(card->dev, "ASoC: CODEC DAI %s not registered\n", codecs[i].dai_name); -- cgit v1.1 From 886f5692253de1a9509f5cb708432b2157afb57c Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 19 Aug 2014 15:51:28 +0200 Subject: ASoC: Automatically initialize regmap for all components So far regmap is only automatically initialized for CODECs. Now that we have the infrastructure in place to let components have DAPM widgets and controls that want to use the generic regmap based IO also make sure to automatically initialize regmap for all components. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 35 +++++++++++++++++------------------ sound/soc/soc-io.c | 28 ---------------------------- 2 files changed, 17 insertions(+), 46 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 140f43f..96f2866 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -4032,8 +4032,23 @@ static int snd_soc_component_initialize(struct snd_soc_component *component, return 0; } +static void snd_soc_component_init_regmap(struct snd_soc_component *component) +{ + if (!component->regmap) + component->regmap = dev_get_regmap(component->dev, NULL); + if (component->regmap) { + int val_bytes = regmap_get_val_bytes(component->regmap); + /* Errors are legitimate for non-integer byte multiples */ + if (val_bytes > 0) + component->val_bytes = val_bytes; + } +} + static void snd_soc_component_add_unlocked(struct snd_soc_component *component) { + if (!component->write && !component->read) + snd_soc_component_init_regmap(component); + list_add(&component->list, &component_list); } @@ -4371,7 +4386,6 @@ int snd_soc_register_codec(struct device *dev, { struct snd_soc_codec *codec; struct snd_soc_dai *dai; - struct regmap *regmap; int ret, i; dev_dbg(dev, "codec register %s\n", dev_name(dev)); @@ -4425,23 +4439,8 @@ int snd_soc_register_codec(struct device *dev, codec->component.debugfs_prefix = "codec"; #endif - if (!codec->component.write) { - if (codec_drv->get_regmap) - regmap = codec_drv->get_regmap(dev); - else - regmap = dev_get_regmap(dev, NULL); - - if (regmap) { - ret = snd_soc_component_init_io(&codec->component, - regmap); - if (ret) { - dev_err(codec->dev, - "Failed to set cache I/O:%d\n", - ret); - goto err_cleanup; - } - } - } + if (codec_drv->get_regmap) + codec->component.regmap = codec_drv->get_regmap(dev); for (i = 0; i < num_dai; i++) { fixup_codec_formats(&dai_drv[i].playback); diff --git a/sound/soc/soc-io.c b/sound/soc/soc-io.c index 7767fbd..9b39390 100644 --- a/sound/soc/soc-io.c +++ b/sound/soc/soc-io.c @@ -271,31 +271,3 @@ int snd_soc_platform_write(struct snd_soc_platform *platform, return snd_soc_component_write(&platform->component, reg, val); } EXPORT_SYMBOL_GPL(snd_soc_platform_write); - -/** - * snd_soc_component_init_io() - Initialize regmap IO - * - * @component: component to initialize - * @regmap: regmap instance to use for IO operations - * - * Return: 0 on success, a negative error code otherwise - */ -int snd_soc_component_init_io(struct snd_soc_component *component, - struct regmap *regmap) -{ - int ret; - - if (!regmap) - return -EINVAL; - - ret = regmap_get_val_bytes(regmap); - /* Errors are legitimate for non-integer byte - * multiples */ - if (ret > 0) - component->val_bytes = ret; - - component->regmap = regmap; - - return 0; -} -EXPORT_SYMBOL_GPL(snd_soc_component_init_io); -- cgit v1.1 From 75af7c081982d76cef0daf26e96b5d1e8cb9d631 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 19 Aug 2014 15:51:29 +0200 Subject: ASoC: Remove support for legacy snd_soc_platform IO There were never any actual users of this in upstream and by we have with regmap a replacement in place, which should be used by new drivers. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 22 ---------------------- 1 file changed, 22 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 96f2866..2d7a9ec 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -4151,24 +4151,6 @@ static void snd_soc_platform_drv_remove(struct snd_soc_component *component) platform->driver->remove(platform); } -static int snd_soc_platform_drv_write(struct snd_soc_component *component, - unsigned int reg, unsigned int val) -{ - struct snd_soc_platform *platform = snd_soc_component_to_platform(component); - - return platform->driver->write(platform, reg, val); -} - -static int snd_soc_platform_drv_read(struct snd_soc_component *component, - unsigned int reg, unsigned int *val) -{ - struct snd_soc_platform *platform = snd_soc_component_to_platform(component); - - *val = platform->driver->read(platform, reg); - - return 0; -} - /** * snd_soc_add_platform - Add a platform to the ASoC core * @dev: The parent device for the platform @@ -4205,10 +4187,6 @@ int snd_soc_add_platform(struct device *dev, struct snd_soc_platform *platform, platform->component.probe = snd_soc_platform_drv_probe; if (platform_drv->remove) platform->component.remove = snd_soc_platform_drv_remove; - if (platform_drv->write) - platform->component.write = snd_soc_platform_drv_write; - if (platform_drv->read) - platform->component.read = snd_soc_platform_drv_read; #ifdef CONFIG_DEBUG_FS platform->component.debugfs_prefix = "platform"; -- cgit v1.1 From c5599b87a8317738a541d8893cb327df5d04b007 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 19 Aug 2014 15:51:30 +0200 Subject: ASoC: Replace list_empty(&card->codec_dev_list) with !card->instantiated With componentization we no longer necessarily need a snd_soc_codec struct for a card. Instead of checking if the card's CODEC list is empty just use card->instantiated to check if the card has been instantiated yet. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 12 ++++-------- 1 file changed, 4 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 2d7a9ec..c36983a 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -552,10 +552,8 @@ int snd_soc_suspend(struct device *dev) struct snd_soc_codec *codec; int i, j; - /* If the initialization of this soc device failed, there is no codec - * associated with it. Just bail out in this case. - */ - if (list_empty(&card->codec_dev_list)) + /* If the card is not initialized yet there is nothing to do */ + if (!card->instantiated) return 0; /* Due to the resume being scheduled into a workqueue we could @@ -808,10 +806,8 @@ int snd_soc_resume(struct device *dev) struct snd_soc_card *card = dev_get_drvdata(dev); int i, ac97_control = 0; - /* If the initialization of this soc device failed, there is no codec - * associated with it. Just bail out in this case. - */ - if (list_empty(&card->codec_dev_list)) + /* If the card is not initialized yet there is nothing to do */ + if (!card->instantiated) return 0; /* activate pins from sleep state */ -- cgit v1.1 From 38c6e4bb67760db1392b9c5ee0082af07c0db20d Mon Sep 17 00:00:00 2001 From: Shengjiu Wang Date: Tue, 19 Aug 2014 17:36:41 +0800 Subject: ASoC: fsl-asoc-card: move 'config SND_SOC_FSL_ASOC_CARD' to 'if SND_IMX_SOC' Build kernel with SND_SOC_FSL_ASOC_CARD=m && SND_SOC_FSL_{SSI,SAI,ESAI}=y leads the following error: sound/built-in.o: In function `fsl_sai_probe': >> fsl_sai.c:(.text+0x5f662): undefined reference to `imx_pcm_dma_init' sound/built-in.o: In function `fsl_esai_probe': >> fsl_esai.c:(.text+0x6044b): undefined reference to `imx_pcm_dma_init' The config SND_SOC_FSL_ASOC_CARD is for IMX SOC, So move it under condition of 'if SND_IMX_SOC'. Reported-by: kbuild test robot Signed-off-by: Shengjiu Wang Signed-off-by: Mark Brown --- sound/soc/fsl/Kconfig | 34 +++++++++++++++++----------------- 1 file changed, 17 insertions(+), 17 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 3154f43..7c1da8e 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -59,23 +59,6 @@ config SND_SOC_FSL_ESAI config SND_SOC_FSL_UTILS tristate -config SND_SOC_FSL_ASOC_CARD - tristate "Generic ASoC Sound Card with ASRC support" - depends on OF && I2C - select SND_SOC_IMX_AUDMUX - select SND_SOC_IMX_PCM_DMA - select SND_SOC_FSL_ESAI - select SND_SOC_FSL_SAI - select SND_SOC_FSL_SSI - select SND_SOC_CS42XX8_I2C - select SND_SOC_SGTL5000 - select SND_SOC_WM8962 - help - ALSA SoC Audio support with ASRC feature for Freescale SoCs that have - ESAI/SAI/SSI and connect with external CODECs such as WM8962, CS42888 - and SGTL5000. - Say Y if you want to add support for Freescale Generic ASoC Sound Card. - config SND_SOC_IMX_PCM_DMA tristate select SND_SOC_GENERIC_DMAENGINE_PCM @@ -298,6 +281,23 @@ config SND_SOC_IMX_MC13783 select SND_SOC_MC13783 select SND_SOC_IMX_PCM_DMA +config SND_SOC_FSL_ASOC_CARD + tristate "Generic ASoC Sound Card with ASRC support" + depends on OF && I2C + select SND_SOC_IMX_AUDMUX + select SND_SOC_IMX_PCM_DMA + select SND_SOC_FSL_ESAI + select SND_SOC_FSL_SAI + select SND_SOC_FSL_SSI + select SND_SOC_CS42XX8_I2C + select SND_SOC_SGTL5000 + select SND_SOC_WM8962 + help + ALSA SoC Audio support with ASRC feature for Freescale SoCs that have + ESAI/SAI/SSI and connect with external CODECs such as WM8962, CS42888 + and SGTL5000. + Say Y if you want to add support for Freescale Generic ASoC Sound Card. + endif # SND_IMX_SOC endmenu -- cgit v1.1 From 5d0ecb0e7dd53e61e034bac8508d7601b04e679d Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 19 Aug 2014 17:48:06 +0200 Subject: ASoC: sh: Don't opencode DMAengine API calls Use the proper wrapper functions instead of directly calling the DMAengine callback functions. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/sh/siu_pcm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/sh/siu_pcm.c b/sound/soc/sh/siu_pcm.c index 488f9be..32eb6da 100644 --- a/sound/soc/sh/siu_pcm.c +++ b/sound/soc/sh/siu_pcm.c @@ -139,7 +139,7 @@ static int siu_pcm_wr_set(struct siu_port *port_info, desc->callback = siu_dma_tx_complete; desc->callback_param = siu_stream; - cookie = desc->tx_submit(desc); + cookie = dmaengine_submit(desc); if (cookie < 0) { dev_err(dev, "Failed to submit a dma transfer\n"); return cookie; @@ -189,7 +189,7 @@ static int siu_pcm_rd_set(struct siu_port *port_info, desc->callback = siu_dma_tx_complete; desc->callback_param = siu_stream; - cookie = desc->tx_submit(desc); + cookie = dmaengine_submit(desc); if (cookie < 0) { dev_err(dev, "Failed to submit dma descriptor\n"); return cookie; -- cgit v1.1 From ff495d3a8ea4d46d237096e6521b24b7ba612e53 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 19 Aug 2014 17:48:07 +0200 Subject: ASoC: txx9: Don't opencode DMAengine API calls Use the proper wrapper functions instead of directly calling the DMAengine callback functions. Also add the missing include to linux/dmaengine.h. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/txx9/txx9aclc.c | 14 +++++++------- 1 file changed, 7 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/soc/txx9/txx9aclc.c b/sound/soc/txx9/txx9aclc.c index f0829de..cd71fd8 100644 --- a/sound/soc/txx9/txx9aclc.c +++ b/sound/soc/txx9/txx9aclc.c @@ -16,6 +16,7 @@ #include #include #include +#include #include #include #include @@ -137,7 +138,7 @@ txx9aclc_dma_submit(struct txx9aclc_dmadata *dmadata, dma_addr_t buf_dma_addr) } desc->callback = txx9aclc_dma_complete; desc->callback_param = dmadata; - desc->tx_submit(desc); + dmaengine_submit(desc); return desc; } @@ -160,7 +161,7 @@ static void txx9aclc_dma_tasklet(unsigned long data) void __iomem *base = drvdata->base; spin_unlock_irqrestore(&dmadata->dma_lock, flags); - chan->device->device_control(chan, DMA_TERMINATE_ALL, 0); + dmaengine_terminate_all(chan); /* first time */ for (i = 0; i < NR_DMA_CHAIN; i++) { desc = txx9aclc_dma_submit(dmadata, @@ -169,7 +170,7 @@ static void txx9aclc_dma_tasklet(unsigned long data) return; } dmadata->dmacount = NR_DMA_CHAIN; - chan->device->device_issue_pending(chan); + dma_async_issue_pending(chan); spin_lock_irqsave(&dmadata->dma_lock, flags); __raw_writel(ctlbit, base + ACCTLEN); dmadata->frag_count = NR_DMA_CHAIN % dmadata->frags; @@ -188,7 +189,7 @@ static void txx9aclc_dma_tasklet(unsigned long data) dmadata->frag_count * dmadata->frag_bytes); if (!desc) return; - chan->device->device_issue_pending(chan); + dma_async_issue_pending(chan); spin_lock_irqsave(&dmadata->dma_lock, flags); dmadata->frag_count++; @@ -266,7 +267,7 @@ static int txx9aclc_pcm_close(struct snd_pcm_substream *substream) struct dma_chan *chan = dmadata->dma_chan; dmadata->frag_count = -1; - chan->device->device_control(chan, DMA_TERMINATE_ALL, 0); + dmaengine_terminate_all(chan); return 0; } @@ -398,8 +399,7 @@ static int txx9aclc_pcm_remove(struct snd_soc_platform *platform) struct dma_chan *chan = dmadata->dma_chan; if (chan) { dmadata->frag_count = -1; - chan->device->device_control(chan, - DMA_TERMINATE_ALL, 0); + dmaengine_terminate_all(chan); dma_release_channel(chan); } dev->dmadata[i].dma_chan = NULL; -- cgit v1.1 From a18a32ce22d8b0e3174c0633fa61e46aac39e81e Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sat, 23 Aug 2014 11:05:21 +0200 Subject: ASoC: ac97-codec: Remove ASoC level IO support This driver doesn't use any ASoC level IO nor does it register any controls or DAPM elements that require it. This means it can safely be removed. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/ac97.c | 15 --------------- 1 file changed, 15 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c index e889e1b..bd9b183 100644 --- a/sound/soc/codecs/ac97.c +++ b/sound/soc/codecs/ac97.c @@ -69,19 +69,6 @@ static struct snd_soc_dai_driver ac97_dai = { .ops = &ac97_dai_ops, }; -static unsigned int ac97_read(struct snd_soc_codec *codec, - unsigned int reg) -{ - return soc_ac97_ops->read(codec->ac97, reg); -} - -static int ac97_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int val) -{ - soc_ac97_ops->write(codec->ac97, reg, val); - return 0; -} - static int ac97_soc_probe(struct snd_soc_codec *codec) { struct snd_ac97_bus *ac97_bus; @@ -122,8 +109,6 @@ static int ac97_soc_resume(struct snd_soc_codec *codec) #endif static struct snd_soc_codec_driver soc_codec_dev_ac97 = { - .write = ac97_write, - .read = ac97_read, .probe = ac97_soc_probe, .suspend = ac97_soc_suspend, .resume = ac97_soc_resume, -- cgit v1.1 From 5819c2fa55d4a6eaf7fe025a393dce98fc4b2116 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sun, 24 Aug 2014 15:36:55 +0200 Subject: ASoC: Restore idle_bias_off initialization This was accidentally lost in commit f1d45cc3ae96 ("ASoC: Consolidate platform and CODEC probe/remove"). Signed-off-by: Lars-Peter Clausen 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 c36983a..4196826 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -4010,6 +4010,7 @@ static int snd_soc_component_initialize(struct snd_soc_component *component, dapm->dev = dev; dapm->component = component; dapm->bias_level = SND_SOC_BIAS_OFF; + dapm->idle_bias_off = true; if (driver->seq_notifier) dapm->seq_notifier = snd_soc_component_seq_notifier; if (driver->stream_event) @@ -4399,6 +4400,7 @@ int snd_soc_register_codec(struct device *dev, codec->component.read = snd_soc_codec_drv_read; codec->component.ignore_pmdown_time = codec_drv->ignore_pmdown_time; codec->dapm.codec = codec; + codec->dapm.idle_bias_off = codec_drv->idle_bias_off; if (codec_drv->seq_notifier) codec->dapm.seq_notifier = codec_drv->seq_notifier; if (codec_drv->set_bias_level) -- cgit v1.1 From 2d15d974618db4ed3adafe9b9fe092db0f5076a0 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Wed, 27 Aug 2014 19:50:34 +0800 Subject: ASoC: rt5677: Add DMIC2 clock selection There are two pins can be used for rt5677's DMIC2 clock. This patch add the select options for it. Signed-off-by: Bard Liao Signed-off-by: Mark Brown --- sound/soc/codecs/rt5677.c | 57 ++++++++++++++++++++++++++++++++++++++++------- sound/soc/codecs/rt5677.h | 10 +++++++++ 2 files changed, 59 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 67f1455..f0b751b 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -1700,14 +1700,19 @@ static const struct snd_soc_dapm_widget rt5677_dapm_widgets[] = { SND_SOC_DAPM_INPUT("Haptic Generator"), - SND_SOC_DAPM_PGA("DMIC1", RT5677_DMIC_CTRL1, RT5677_DMIC_1_EN_SFT, 0, - NULL, 0), - SND_SOC_DAPM_PGA("DMIC2", RT5677_DMIC_CTRL1, RT5677_DMIC_2_EN_SFT, 0, - NULL, 0), - SND_SOC_DAPM_PGA("DMIC3", RT5677_DMIC_CTRL1, RT5677_DMIC_3_EN_SFT, 0, - NULL, 0), - SND_SOC_DAPM_PGA("DMIC4", RT5677_DMIC_CTRL2, RT5677_DMIC_4_EN_SFT, 0, - NULL, 0), + SND_SOC_DAPM_PGA("DMIC1", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("DMIC2", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("DMIC3", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("DMIC4", SND_SOC_NOPM, 0, 0, NULL, 0), + + SND_SOC_DAPM_SUPPLY("DMIC1 power", RT5677_DMIC_CTRL1, + RT5677_DMIC_1_EN_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DMIC2 power", RT5677_DMIC_CTRL1, + RT5677_DMIC_2_EN_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DMIC3 power", RT5677_DMIC_CTRL1, + RT5677_DMIC_3_EN_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DMIC4 power", RT5677_DMIC_CTRL2, + RT5677_DMIC_4_EN_SFT, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("DMIC CLK", SND_SOC_NOPM, 0, 0, set_dmic_clk, SND_SOC_DAPM_PRE_PMU), @@ -2130,6 +2135,13 @@ static const struct snd_soc_dapm_route rt5677_dapm_routes[] = { { "DMIC L4", NULL, "DMIC CLK" }, { "DMIC R4", NULL, "DMIC CLK" }, + { "DMIC L1", NULL, "DMIC1 power" }, + { "DMIC R1", NULL, "DMIC1 power" }, + { "DMIC L3", NULL, "DMIC3 power" }, + { "DMIC R3", NULL, "DMIC3 power" }, + { "DMIC L4", NULL, "DMIC4 power" }, + { "DMIC R4", NULL, "DMIC4 power" }, + { "BST1", NULL, "IN1P" }, { "BST1", NULL, "IN1N" }, { "BST2", NULL, "IN2P" }, @@ -2793,6 +2805,16 @@ static const struct snd_soc_dapm_route rt5677_dapm_routes[] = { { "PDM2R", NULL, "PDM2 R Mux" }, }; +static const struct snd_soc_dapm_route rt5677_dmic2_clk_1[] = { + { "DMIC L2", NULL, "DMIC1 power" }, + { "DMIC R2", NULL, "DMIC1 power" }, +}; + +static const struct snd_soc_dapm_route rt5677_dmic2_clk_2[] = { + { "DMIC L2", NULL, "DMIC2 power" }, + { "DMIC R2", NULL, "DMIC2 power" }, +}; + static int rt5677_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { @@ -3144,6 +3166,16 @@ static int rt5677_probe(struct snd_soc_codec *codec) rt5677->codec = codec; + if (rt5677->pdata.dmic2_clk_pin == RT5677_DMIC_CLK2) { + snd_soc_dapm_add_routes(&codec->dapm, + rt5677_dmic2_clk_2, + ARRAY_SIZE(rt5677_dmic2_clk_2)); + } else { /*use dmic1 clock by default*/ + snd_soc_dapm_add_routes(&codec->dapm, + rt5677_dmic2_clk_1, + ARRAY_SIZE(rt5677_dmic2_clk_1)); + } + rt5677_set_bias_level(codec, SND_SOC_BIAS_OFF); regmap_write(rt5677->regmap, RT5677_DIG_MISC, 0x0020); @@ -3381,6 +3413,15 @@ static int rt5677_i2c_probe(struct i2c_client *i2c, regmap_update_bits(rt5677->regmap, RT5677_IN1, RT5677_IN_DF2, RT5677_IN_DF2); + if (rt5677->pdata.dmic2_clk_pin == RT5677_DMIC_CLK2) { + regmap_update_bits(rt5677->regmap, RT5677_GEN_CTRL2, + RT5677_GPIO5_FUNC_MASK, + RT5677_GPIO5_FUNC_DMIC); + regmap_update_bits(rt5677->regmap, RT5677_GPIO_CTRL2, + RT5677_GPIO5_DIR_MASK, + RT5677_GPIO5_DIR_OUT); + } + return snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5677, rt5677_dai, ARRAY_SIZE(rt5677_dai)); } diff --git a/sound/soc/codecs/rt5677.h b/sound/soc/codecs/rt5677.h index 863393e..8791ab9 100644 --- a/sound/soc/codecs/rt5677.h +++ b/sound/soc/codecs/rt5677.h @@ -1363,6 +1363,11 @@ #define RT5677_SEL_SRC_IB01 (0x1 << 0) #define RT5677_SEL_SRC_IB01_SFT 0 +/* GPIO Control 2 (0xc1) */ +#define RT5677_GPIO5_DIR_MASK (0x1 << 14) +#define RT5677_GPIO5_DIR_IN (0x0 << 14) +#define RT5677_GPIO5_DIR_OUT (0x1 << 14) + /* Virtual DSP Mixer Control (0xf7 0xf8 0xf9) */ #define RT5677_DSP_IB_01_H (0x1 << 15) #define RT5677_DSP_IB_01_H_SFT 15 @@ -1393,6 +1398,11 @@ #define RT5677_DSP_IB_9_L (0x1 << 1) #define RT5677_DSP_IB_9_L_SFT 1 +/* General Control2 (0xfc)*/ +#define RT5677_GPIO5_FUNC_MASK (0x1 << 9) +#define RT5677_GPIO5_FUNC_GPIO (0x0 << 9) +#define RT5677_GPIO5_FUNC_DMIC (0x1 << 9) + /* System Clock Source */ enum { RT5677_SCLK_S_MCLK, -- cgit v1.1 From bf16d883263dedefb6149916e41b3e2779bb1573 Mon Sep 17 00:00:00 2001 From: Xiubo Li Date: Mon, 25 Aug 2014 11:30:59 +0800 Subject: ASoC: fsl-asrc: Convert to use regmap framework's endianness method. Signed-off-by: Xiubo Li Acked-by: Nicolin Chen Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_asrc.c | 6 +----- 1 file changed, 1 insertion(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_asrc.c b/sound/soc/fsl/fsl_asrc.c index 8221104..3b14531 100644 --- a/sound/soc/fsl/fsl_asrc.c +++ b/sound/soc/fsl/fsl_asrc.c @@ -684,7 +684,7 @@ static bool fsl_asrc_writeable_reg(struct device *dev, unsigned int reg) } } -static struct regmap_config fsl_asrc_regmap_config = { +static const struct regmap_config fsl_asrc_regmap_config = { .reg_bits = 32, .reg_stride = 4, .val_bits = 32, @@ -802,10 +802,6 @@ static int fsl_asrc_probe(struct platform_device *pdev) asrc_priv->paddr = res->start; - /* Register regmap and let it prepare core clock */ - if (of_property_read_bool(np, "big-endian")) - fsl_asrc_regmap_config.val_format_endian = REGMAP_ENDIAN_BIG; - asrc_priv->regmap = devm_regmap_init_mmio_clk(&pdev->dev, "mem", regs, &fsl_asrc_regmap_config); if (IS_ERR(asrc_priv->regmap)) { -- cgit v1.1 From 92bd0334b27845f250f1fadb091242140391c99b Mon Sep 17 00:00:00 2001 From: Xiubo Li Date: Mon, 25 Aug 2014 11:31:00 +0800 Subject: ASoC: fsl-esai: Convert to use regmap framework's endianness method. Signed-off-by: Xiubo Li Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_esai.c | 5 +---- 1 file changed, 1 insertion(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index 72d154e..2882fc6 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -707,7 +707,7 @@ static bool fsl_esai_writeable_reg(struct device *dev, unsigned int reg) } } -static struct regmap_config fsl_esai_regmap_config = { +static const struct regmap_config fsl_esai_regmap_config = { .reg_bits = 32, .reg_stride = 4, .val_bits = 32, @@ -733,9 +733,6 @@ static int fsl_esai_probe(struct platform_device *pdev) esai_priv->pdev = pdev; strcpy(esai_priv->name, np->name); - if (of_property_read_bool(np, "big-endian")) - fsl_esai_regmap_config.val_format_endian = REGMAP_ENDIAN_BIG; - /* Get the addresses and IRQ */ res = platform_get_resource(pdev, IORESOURCE_MEM, 0); regs = devm_ioremap_resource(&pdev->dev, res); -- cgit v1.1 From 664915074e750614c5d140093d5098a165a24e3d Mon Sep 17 00:00:00 2001 From: Xiubo Li Date: Mon, 25 Aug 2014 11:31:01 +0800 Subject: ASoC: fsl-spdif: Convert to use regmap framework's endianness method. Signed-off-by: Xiubo Li Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_spdif.c | 5 +---- 1 file changed, 1 insertion(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index 70acfe4..ae4e4088 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -1040,7 +1040,7 @@ static bool fsl_spdif_writeable_reg(struct device *dev, unsigned int reg) } } -static struct regmap_config fsl_spdif_regmap_config = { +static const struct regmap_config fsl_spdif_regmap_config = { .reg_bits = 32, .reg_stride = 4, .val_bits = 32, @@ -1184,9 +1184,6 @@ static int fsl_spdif_probe(struct platform_device *pdev) memcpy(&spdif_priv->cpu_dai_drv, &fsl_spdif_dai, sizeof(fsl_spdif_dai)); spdif_priv->cpu_dai_drv.name = spdif_priv->name; - if (of_property_read_bool(np, "big-endian")) - fsl_spdif_regmap_config.val_format_endian = REGMAP_ENDIAN_BIG; - /* Get the addresses and IRQ */ res = platform_get_resource(pdev, IORESOURCE_MEM, 0); regs = devm_ioremap_resource(&pdev->dev, res); -- cgit v1.1 From 014fd22ef9c6a7e9536b7e16635714a1a34810a8 Mon Sep 17 00:00:00 2001 From: Xiubo Li Date: Mon, 25 Aug 2014 11:31:02 +0800 Subject: ASoC: fsl-sai: Convert to use regmap framework's endianness method. Signed-off-by: Xiubo Li Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_sai.c | 6 +----- sound/soc/fsl/fsl_sai.h | 1 - 2 files changed, 1 insertion(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index faa0497..52d1e99 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -539,7 +539,7 @@ static bool fsl_sai_writeable_reg(struct device *dev, unsigned int reg) } } -static struct regmap_config fsl_sai_regmap_config = { +static const struct regmap_config fsl_sai_regmap_config = { .reg_bits = 32, .reg_stride = 4, .val_bits = 32, @@ -568,10 +568,6 @@ static int fsl_sai_probe(struct platform_device *pdev) if (of_device_is_compatible(pdev->dev.of_node, "fsl,imx6sx-sai")) sai->sai_on_imx = true; - sai->big_endian_regs = of_property_read_bool(np, "big-endian-regs"); - if (sai->big_endian_regs) - fsl_sai_regmap_config.val_format_endian = REGMAP_ENDIAN_BIG; - sai->big_endian_data = of_property_read_bool(np, "big-endian-data"); res = platform_get_resource(pdev, IORESOURCE_MEM, 0); diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h index 0e6c9f5..20e3e53 100644 --- a/sound/soc/fsl/fsl_sai.h +++ b/sound/soc/fsl/fsl_sai.h @@ -131,7 +131,6 @@ struct fsl_sai { struct clk *bus_clk; struct clk *mclk_clk[FSL_SAI_MCLK_MAX]; - bool big_endian_regs; bool big_endian_data; bool is_dsp_mode; bool sai_on_imx; -- cgit v1.1 From 06cb1eb3de5c905da60ab91dbf99aaf96a43d043 Mon Sep 17 00:00:00 2001 From: "Subhransu S. Prusty" Date: Thu, 21 Aug 2014 18:20:49 +0530 Subject: ASoC: mfld-compress: Use dedicated function instead of ioctl Also pass sst device as an argument to function pointer prototypes of compr_ops. This will be used to derive sst driver context. Signed-off-by: Subhransu S. Prusty Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/sst-mfld-platform-compress.c | 38 +++++++++++++++++++++------- sound/soc/intel/sst-mfld-platform.h | 27 ++++++++++++-------- 2 files changed, 46 insertions(+), 19 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/sst-mfld-platform-compress.c b/sound/soc/intel/sst-mfld-platform-compress.c index 29c059c..59467775 100644 --- a/sound/soc/intel/sst-mfld-platform-compress.c +++ b/sound/soc/intel/sst-mfld-platform-compress.c @@ -86,7 +86,7 @@ static int sst_platform_compr_free(struct snd_compr_stream *cstream) /*need to check*/ str_id = stream->id; if (str_id) - ret_val = stream->compr_ops->close(str_id); + ret_val = stream->compr_ops->close(sst->dev, str_id); module_put(sst->dev->driver->owner); kfree(stream); pr_debug("%s: %d\n", __func__, ret_val); @@ -158,7 +158,7 @@ static int sst_platform_compr_set_params(struct snd_compr_stream *cstream, cb.drain_cb_param = cstream; cb.drain_notify = sst_drain_notify; - retval = stream->compr_ops->open(&str_params, &cb); + retval = stream->compr_ops->open(sst->dev, &str_params, &cb); if (retval < 0) { pr_err("stream allocation failed %d\n", retval); return retval; @@ -170,10 +170,30 @@ static int sst_platform_compr_set_params(struct snd_compr_stream *cstream, static int sst_platform_compr_trigger(struct snd_compr_stream *cstream, int cmd) { - struct sst_runtime_stream *stream = - cstream->runtime->private_data; - - return stream->compr_ops->control(cmd, stream->id); + struct sst_runtime_stream *stream = cstream->runtime->private_data; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + if (stream->compr_ops->stream_start) + return stream->compr_ops->stream_start(sst->dev, stream->id); + case SNDRV_PCM_TRIGGER_STOP: + if (stream->compr_ops->stream_drop) + return stream->compr_ops->stream_drop(sst->dev, stream->id); + case SND_COMPR_TRIGGER_DRAIN: + if (stream->compr_ops->stream_drain) + return stream->compr_ops->stream_drain(sst->dev, stream->id); + case SND_COMPR_TRIGGER_PARTIAL_DRAIN: + if (stream->compr_ops->stream_partial_drain) + return stream->compr_ops->stream_partial_drain(sst->dev, stream->id); + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + if (stream->compr_ops->stream_pause) + return stream->compr_ops->stream_pause(sst->dev, stream->id); + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + if (stream->compr_ops->stream_pause_release) + return stream->compr_ops->stream_pause_release(sst->dev, stream->id); + default: + return -EINVAL; + } } static int sst_platform_compr_pointer(struct snd_compr_stream *cstream, @@ -182,7 +202,7 @@ static int sst_platform_compr_pointer(struct snd_compr_stream *cstream, struct sst_runtime_stream *stream; stream = cstream->runtime->private_data; - stream->compr_ops->tstamp(stream->id, tstamp); + stream->compr_ops->tstamp(sst->dev, stream->id, tstamp); tstamp->byte_offset = tstamp->copied_total % (u32)cstream->runtime->buffer_size; pr_debug("calc bytes offset/copied bytes as %d\n", tstamp->byte_offset); @@ -195,7 +215,7 @@ static int sst_platform_compr_ack(struct snd_compr_stream *cstream, struct sst_runtime_stream *stream; stream = cstream->runtime->private_data; - stream->compr_ops->ack(stream->id, (unsigned long)bytes); + stream->compr_ops->ack(sst->dev, stream->id, (unsigned long)bytes); stream->bytes_written += bytes; return 0; @@ -225,7 +245,7 @@ static int sst_platform_compr_set_metadata(struct snd_compr_stream *cstream, struct sst_runtime_stream *stream = cstream->runtime->private_data; - return stream->compr_ops->set_metadata(stream->id, metadata); + return stream->compr_ops->set_metadata(sst->dev, stream->id, metadata); } struct snd_compr_ops sst_platform_compr_ops = { diff --git a/sound/soc/intel/sst-mfld-platform.h b/sound/soc/intel/sst-mfld-platform.h index faaba10..0c5b943d 100644 --- a/sound/soc/intel/sst-mfld-platform.h +++ b/sound/soc/intel/sst-mfld-platform.h @@ -99,17 +99,24 @@ struct sst_compress_cb { struct compress_sst_ops { const char *name; - int (*open) (struct snd_sst_params *str_params, - struct sst_compress_cb *cb); - int (*control) (unsigned int cmd, unsigned int str_id); - int (*tstamp) (unsigned int str_id, struct snd_compr_tstamp *tstamp); - int (*ack) (unsigned int str_id, unsigned long bytes); - int (*close) (unsigned int str_id); - int (*get_caps) (struct snd_compr_caps *caps); - int (*get_codec_caps) (struct snd_compr_codec_caps *codec); - int (*set_metadata) (unsigned int str_id, + int (*open)(struct device *dev, + struct snd_sst_params *str_params, struct sst_compress_cb *cb); + int (*stream_start)(struct device *dev, unsigned int str_id); + int (*stream_drop)(struct device *dev, unsigned int str_id); + int (*stream_drain)(struct device *dev, unsigned int str_id); + int (*stream_partial_drain)(struct device *dev, unsigned int str_id); + int (*stream_pause)(struct device *dev, unsigned int str_id); + int (*stream_pause_release)(struct device *dev, unsigned int str_id); + + int (*tstamp)(struct device *dev, unsigned int str_id, + struct snd_compr_tstamp *tstamp); + int (*ack)(struct device *dev, unsigned int str_id, + unsigned long bytes); + int (*close)(struct device *dev, unsigned int str_id); + int (*get_caps)(struct snd_compr_caps *caps); + int (*get_codec_caps)(struct snd_compr_codec_caps *codec); + int (*set_metadata)(struct device *dev, unsigned int str_id, struct snd_compr_metadata *mdata); - }; struct sst_ops { -- cgit v1.1 From 77c545398e33a0263a68142fcfbd4b11b0f06294 Mon Sep 17 00:00:00 2001 From: Geert Uytterhoeven Date: Thu, 28 Aug 2014 14:48:24 +0200 Subject: ASoC: Allow SND_SOC_WM8978 to be selected manually When using a DT-based multi-platform kernel, there's not always Kconfig logic that selects the right codec driver. Allow the user to manually select WM8978. This is needed for Armadillo 800 EVA using a generic r8a7740 multi-platform kernel. Signed-off-by: Geert Uytterhoeven Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 8838838e..9c400a2 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -712,7 +712,8 @@ config SND_SOC_WM8974 tristate config SND_SOC_WM8978 - tristate + tristate "Wolfson Microelectronics WM8978 codec" + depends on I2C config SND_SOC_WM8983 tristate -- cgit v1.1 From 98c5d36240e10c2e0e06e2bb10496291626d1d43 Mon Sep 17 00:00:00 2001 From: Paul Handrigan Date: Thu, 28 Aug 2014 10:54:08 -0500 Subject: ASoC: cs4265: Add CHIP_ID as a readable register MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Reported-by: Zoltán Szenczi Signed-off-by: Paul Handrigan Signed-off-by: Mark Brown --- sound/soc/codecs/cs4265.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/codecs/cs4265.c b/sound/soc/codecs/cs4265.c index a20b30c..2dad15a 100644 --- a/sound/soc/codecs/cs4265.c +++ b/sound/soc/codecs/cs4265.c @@ -77,6 +77,7 @@ static bool cs4265_readable_register(struct device *dev, unsigned int reg) case CS4265_INT_MASK: case CS4265_STATUS_MODE_MSB: case CS4265_STATUS_MODE_LSB: + case CS4265_CHIP_ID: return true; default: return false; -- cgit v1.1 From 7eef08554ca35454e6da0de8a74f7c96bc2e58e0 Mon Sep 17 00:00:00 2001 From: Brian Austin Date: Thu, 28 Aug 2014 10:02:40 -0500 Subject: ASoC: cs35l32: use true/false returns for bool functions Return true or false instead of 1 and 0 Reported-by: Fengguang Wu Signed-off-by: Brian Austin Signed-off-by: Mark Brown --- sound/soc/codecs/cs35l32.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/cs35l32.c b/sound/soc/codecs/cs35l32.c index 9c6b272..ca897c4 100644 --- a/sound/soc/codecs/cs35l32.c +++ b/sound/soc/codecs/cs35l32.c @@ -121,9 +121,9 @@ static bool cs35l32_volatile_register(struct device *dev, unsigned int reg) case CS35L32_INT_STATUS_2: case CS35L32_INT_STATUS_3: case CS35L32_LED_STATUS: - return 1; + return true; default: - return 0; + return false; } } @@ -134,9 +134,9 @@ static bool cs35l32_precious_register(struct device *dev, unsigned int reg) case CS35L32_INT_STATUS_2: case CS35L32_INT_STATUS_3: case CS35L32_LED_STATUS: - return 1; + return true; default: - return 0; + return false; } } -- cgit v1.1 From 5c216cc3f37a6eecb4e12ab0248b66e6386da0fe Mon Sep 17 00:00:00 2001 From: Brian Austin Date: Thu, 28 Aug 2014 10:02:41 -0500 Subject: ASoC: cs42l52: use true/false returns for bool functions Return true or false instead of 1 and 0 Signed-off-by: Brian Austin Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l52.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 969167d..da4f758 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -176,9 +176,9 @@ static bool cs42l52_volatile_register(struct device *dev, unsigned int reg) case CS42L52_BATT_LEVEL: case CS42L52_SPK_STATUS: case CS42L52_CHARGE_PUMP: - return 1; + return true; default: - return 0; + return false; } } -- cgit v1.1 From c2b49ae678b8bd1fd4ea3e3ae106020d663e8969 Mon Sep 17 00:00:00 2001 From: Brian Austin Date: Thu, 28 Aug 2014 10:02:42 -0500 Subject: ASoC: cs42l56: use true/false returns for bool functions Return true or false instead of 1 and 0 Signed-off-by: Brian Austin Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l56.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/cs42l56.c b/sound/soc/codecs/cs42l56.c index c766a5a..b1c7396 100644 --- a/sound/soc/codecs/cs42l56.c +++ b/sound/soc/codecs/cs42l56.c @@ -171,9 +171,9 @@ static bool cs42l56_volatile_register(struct device *dev, unsigned int reg) { switch (reg) { case CS42L56_INT_STATUS: - return 1; + return true; default: - return 0; + return false; } } -- cgit v1.1 From b792346fa8660a22a06f118cebe47709f507914f Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Thu, 28 Aug 2014 14:07:11 +0300 Subject: ASoC: Remove unused cache_only from struct snd_soc_codec There are no real users for cache_only in "struct snd_soc_codec" so remove it and needless debugfs node. Signed-off-by: Jarkko Nikula Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 2 -- 1 file changed, 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 4196826..1b422c5 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -311,8 +311,6 @@ static void soc_init_codec_debugfs(struct snd_soc_component *component) debugfs_create_bool("cache_sync", 0444, codec->component.debugfs_root, &codec->cache_sync); - debugfs_create_bool("cache_only", 0444, codec->component.debugfs_root, - &codec->cache_only); codec->debugfs_reg = debugfs_create_file("codec_reg", 0644, codec->component.debugfs_root, -- cgit v1.1 From 1a83269d5c41b77f2a4bbb3828c668c96832742e Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Thu, 28 Aug 2014 17:54:38 +0800 Subject: ASoC: cs35l32: Remove unneeded regulator_bulk_free call in cs35l32_i2c_remove The regulator_bulk_free() call is not required because current code is using devm_regulator_bulk_get(). Signed-off-by: Axel Lin Acked-by: Brian Austin Signed-off-by: Mark Brown --- sound/soc/codecs/cs35l32.c | 2 -- 1 file changed, 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/cs35l32.c b/sound/soc/codecs/cs35l32.c index ca897c4..b32d7a9 100644 --- a/sound/soc/codecs/cs35l32.c +++ b/sound/soc/codecs/cs35l32.c @@ -562,8 +562,6 @@ static int cs35l32_i2c_remove(struct i2c_client *i2c_client) if (cs35l32->reset_gpio) gpiod_set_value_cansleep(cs35l32->reset_gpio, 0); - regulator_bulk_free(ARRAY_SIZE(cs35l32->supplies), cs35l32->supplies); - return 0; } -- cgit v1.1 From a4f87cea72d78f80c0bda1b4d8a821278eb1e4e2 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Thu, 28 Aug 2014 17:55:20 +0800 Subject: ASoC: cs42l56: Remove unneeded regulator_bulk_free call in cs42l56_remove The regulator_bulk_free() call is not required because current code is using devm_regulator_bulk_get(). Signed-off-by: Axel Lin Acked-by: Brian Austin Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l56.c | 3 --- 1 file changed, 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/cs42l56.c b/sound/soc/codecs/cs42l56.c index b1c7396..bb74dd1 100644 --- a/sound/soc/codecs/cs42l56.c +++ b/sound/soc/codecs/cs42l56.c @@ -1175,11 +1175,8 @@ static int cs42l56_probe(struct snd_soc_codec *codec) static int cs42l56_remove(struct snd_soc_codec *codec) { - struct cs42l56_private *cs42l56 = snd_soc_codec_get_drvdata(codec); - cs42l56_free_beep(codec); cs42l56_set_bias_level(codec, SND_SOC_BIAS_OFF); - regulator_bulk_free(ARRAY_SIZE(cs42l56->supplies), cs42l56->supplies); return 0; } -- cgit v1.1 From 5f609f282b59f111840e755bac8da980387e044e Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Thu, 28 Aug 2014 16:27:56 +0800 Subject: ASoC: cs35l32: Simplify implementation of cs35l32_codec_set_sysclk Use single snd_soc_update_bits() call to update the register bits. Signed-off-by: Axel Lin Tested-by: Brian Austin Acked-by: Brian Austin Signed-off-by: Mark Brown --- sound/soc/codecs/cs35l32.c | 28 +++++++--------------------- 1 file changed, 7 insertions(+), 21 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/cs35l32.c b/sound/soc/codecs/cs35l32.c index b32d7a9..76f628b 100644 --- a/sound/soc/codecs/cs35l32.c +++ b/sound/soc/codecs/cs35l32.c @@ -242,41 +242,27 @@ static struct snd_soc_dai_driver cs35l32_dai[] = { static int cs35l32_codec_set_sysclk(struct snd_soc_codec *codec, int clk_id, int source, unsigned int freq, int dir) { + unsigned int val; switch (freq) { case 6000000: - snd_soc_update_bits(codec, CS35L32_CLK_CTL, - CS35L32_MCLK_DIV2_MASK, 0); - snd_soc_update_bits(codec, CS35L32_CLK_CTL, - CS35L32_MCLK_RATIO_MASK, - CS35L32_MCLK_RATIO); + val = CS35L32_MCLK_RATIO; break; case 12000000: - snd_soc_update_bits(codec, CS35L32_CLK_CTL, - CS35L32_MCLK_DIV2_MASK, - CS35L32_MCLK_DIV2_MASK); - snd_soc_update_bits(codec, CS35L32_CLK_CTL, - CS35L32_MCLK_RATIO_MASK, - CS35L32_MCLK_RATIO); + val = CS35L32_MCLK_DIV2_MASK | CS35L32_MCLK_RATIO; break; case 6144000: - snd_soc_update_bits(codec, CS35L32_CLK_CTL, - CS35L32_MCLK_DIV2_MASK, 0); - snd_soc_update_bits(codec, CS35L32_CLK_CTL, - CS35L32_MCLK_RATIO_MASK, 0); + val = 0; break; case 12288000: - snd_soc_update_bits(codec, CS35L32_CLK_CTL, - CS35L32_MCLK_DIV2_MASK, - CS35L32_MCLK_DIV2_MASK); - snd_soc_update_bits(codec, CS35L32_CLK_CTL, - CS35L32_MCLK_RATIO_MASK, 0); + val = CS35L32_MCLK_DIV2_MASK; break; default: return -EINVAL; } - return 0; + return snd_soc_update_bits(codec, CS35L32_CLK_CTL, + CS35L32_MCLK_DIV2_MASK | CS35L32_MCLK_RATIO_MASK, val); } static struct snd_soc_codec_driver soc_codec_dev_cs35l32 = { -- cgit v1.1 From 2d82eeb02655e32358efd42598d8276284c23364 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Wed, 27 Aug 2014 20:07:46 -0700 Subject: ASoC: simple-card: use asoc_simple_xxx prefix simple-card driver is using asoc_simple_xxx() prefix. simple_card_dai_link_of() should be asoc_simple_card_dai_link_of(). Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/generic/simple-card.c | 20 +++++++++++--------- 1 file changed, 11 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 21b0ea2..c5445b0 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -163,11 +163,11 @@ asoc_simple_card_sub_parse_of(struct device_node *np, return 0; } -static int simple_card_dai_link_of(struct device_node *node, - struct device *dev, - struct snd_soc_dai_link *dai_link, - struct simple_dai_props *dai_props, - bool is_top_level_node) +static int asoc_simple_card_dai_link_of(struct device_node *node, + struct device *dev, + struct snd_soc_dai_link *dai_link, + struct simple_dai_props *dai_props, + bool is_top_level_node) { struct device_node *np = NULL; struct device_node *bitclkmaster = NULL; @@ -337,16 +337,18 @@ static int asoc_simple_card_parse_of(struct device_node *node, int i; for (i = 0; (np = of_get_next_child(node, np)); i++) { dev_dbg(dev, "\tlink %d:\n", i); - ret = simple_card_dai_link_of(np, dev, dai_link + i, - dai_props + i, false); + ret = asoc_simple_card_dai_link_of(np, dev, + dai_link + i, + dai_props + i, + false); if (ret < 0) { of_node_put(np); return ret; } } } else { - ret = simple_card_dai_link_of(node, dev, dai_link, dai_props, - true); + ret = asoc_simple_card_dai_link_of(node, dev, + dai_link, dai_props, true); if (ret < 0) return ret; } -- cgit v1.1 From 179949bc04c7157a4b2279f62a842638b61f78f9 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Wed, 27 Aug 2014 20:08:06 -0700 Subject: ASoC: simple-card: remove dai_link->cpu_dai_name when DT f687d900d30a61dda38db2a99239f5284a86a309 (ASoC: simple-card: cpu_dai_name creates confusion when DT case) removed dai_link->cpu_dai_name when DT case, since it uses DT phand in soc_bind_dai_link(). This binding will fail if it has cpu_dai_name. 6a91a17bd7b92b2d2aa9ece85457f52a62fd7708 (ASoC: simple-card: Handle many DAI links) added multi DAI link support to simple-card driver. Then, removing cpu_dai_name was cared only single DAI. But, it is needed in all DT cases. This patch moves it to asoc_simple_card_dai_link_of() so that care about all DAIs. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/generic/simple-card.c | 23 +++++++++++------------ 1 file changed, 11 insertions(+), 12 deletions(-) (limited to 'sound') diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index c5445b0..e8185a0 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -285,6 +285,17 @@ static int asoc_simple_card_dai_link_of(struct device_node *node, dai_props->codec_dai.fmt, dai_props->codec_dai.sysclk); + /* + * soc_bind_dai_link() will check cpu name + * after of_node matching if dai_link has cpu_dai_name. + * but, it will never match if name was created by fmt_single_name() + * remove cpu_dai_name to escape name matching. + * see + * fmt_single_name() + * fmt_multiple_name() + */ + dai_link->cpu_dai_name = NULL; + dai_link_of_err: if (np) of_node_put(np); @@ -429,18 +440,6 @@ static int asoc_simple_card_probe(struct platform_device *pdev) goto err; } - /* - * soc_bind_dai_link() will check cpu name - * after of_node matching if dai_link has cpu_dai_name. - * but, it will never match if name was created by fmt_single_name() - * remove cpu_dai_name to escape name matching. - * see - * fmt_single_name() - * fmt_multiple_name() - */ - if (num_links == 1) - dai_link->cpu_dai_name = NULL; - } else { struct asoc_simple_card_info *cinfo; -- cgit v1.1 From a5960bd5984c808cdf7aa528e162e9e20e61b923 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Wed, 27 Aug 2014 20:08:27 -0700 Subject: ASoC: simple-card: dai_link->init should be cared when multi DAI 6a91a17bd7b92b2d2aa9ece85457f52a62fd7708 (ASoC: simple-card: Handle many DAI links) added multi DAI support on simple-card. This means priv->dai_link might be pointer of multi DAI. dai_link->init is needed for all DAI. This patch cares it for all DAIs on DT/non-DT Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/generic/simple-card.c | 7 ++----- 1 file changed, 2 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index e8185a0..8902704 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -274,6 +274,7 @@ static int asoc_simple_card_dai_link_of(struct device_node *node, dai_link->codec_dai_name); dai_link->name = dai_link->stream_name = name; dai_link->ops = &asoc_simple_card_ops; + dai_link->init = asoc_simple_card_dai_init; dev_dbg(dev, "\tname : %s\n", dai_link->stream_name); dev_dbg(dev, "\tcpu : %s / %04x / %d\n", @@ -465,6 +466,7 @@ static int asoc_simple_card_probe(struct platform_device *pdev) dai_link->codec_name = cinfo->codec; dai_link->cpu_dai_name = cinfo->cpu_dai.name; dai_link->codec_dai_name = cinfo->codec_dai.name; + dai_link->init = asoc_simple_card_dai_init; memcpy(&priv->dai_props->cpu_dai, &cinfo->cpu_dai, sizeof(priv->dai_props->cpu_dai)); memcpy(&priv->dai_props->codec_dai, &cinfo->codec_dai, @@ -474,11 +476,6 @@ static int asoc_simple_card_probe(struct platform_device *pdev) priv->dai_props->codec_dai.fmt |= cinfo->daifmt; } - /* - * init snd_soc_dai_link - */ - dai_link->init = asoc_simple_card_dai_init; - snd_soc_card_set_drvdata(&priv->snd_card, priv); ret = devm_snd_soc_register_card(&pdev->dev, &priv->snd_card); -- cgit v1.1 From a44a750e5299fe2ece5aa68e8562dd6e2c2b16f4 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Wed, 27 Aug 2014 20:08:47 -0700 Subject: ASoC: simple-card: use common for_each_child_of_node() for loop Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/generic/simple-card.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 8902704..fd8b045 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -346,8 +346,9 @@ static int asoc_simple_card_parse_of(struct device_node *node, if (multi) { struct device_node *np = NULL; - int i; - for (i = 0; (np = of_get_next_child(node, np)); i++) { + int i = 0; + + for_each_child_of_node(node, np) { dev_dbg(dev, "\tlink %d:\n", i); ret = asoc_simple_card_dai_link_of(np, dev, dai_link + i, @@ -357,6 +358,7 @@ static int asoc_simple_card_parse_of(struct device_node *node, of_node_put(np); return ret; } + i++; } } else { ret = asoc_simple_card_dai_link_of(node, dev, -- cgit v1.1 From 085f3ec6fd6c87907c4a19481dc13f02ecfcd316 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Mon, 1 Sep 2014 12:46:37 +0300 Subject: ASoC: tlv320aic31xx: Correct interface register 2 variable name Rename iface_reg3 to iface_reg2 since this variable is actually used for interface register 2. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic31xx.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index 0f64c78..9f9d23b 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -813,7 +813,7 @@ static int aic31xx_set_dai_fmt(struct snd_soc_dai *codec_dai, { struct snd_soc_codec *codec = codec_dai->codec; u8 iface_reg1 = 0; - u8 iface_reg3 = 0; + u8 iface_reg2 = 0; u8 dsp_a_val = 0; dev_dbg(codec->dev, "## %s: fmt = 0x%x\n", __func__, fmt); @@ -838,7 +838,7 @@ static int aic31xx_set_dai_fmt(struct snd_soc_dai *codec_dai, /* NOTE: BCLKINV bit value 1 equas NB and 0 equals IB */ switch (fmt & SND_SOC_DAIFMT_INV_MASK) { case SND_SOC_DAIFMT_NB_NF: - iface_reg3 |= AIC31XX_BCLKINV_MASK; + iface_reg2 |= AIC31XX_BCLKINV_MASK; break; case SND_SOC_DAIFMT_IB_NF: break; @@ -870,7 +870,7 @@ static int aic31xx_set_dai_fmt(struct snd_soc_dai *codec_dai, dsp_a_val); snd_soc_update_bits(codec, AIC31XX_IFACE2, AIC31XX_BCLKINV_MASK, - iface_reg3); + iface_reg2); return 0; } -- cgit v1.1 From 75c3daaad5a2f791e0fbad732690130ce1bc55d2 Mon Sep 17 00:00:00 2001 From: Wei Yongjun Date: Mon, 1 Sep 2014 08:47:50 +0800 Subject: ASoC: es8328: fix error return code in es8328_codec_probe() Fix to return a negative error code from the error handling case instead of 0, as done elsewhere in this function. Signed-off-by: Wei Yongjun Signed-off-by: Mark Brown --- sound/soc/codecs/es8328.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/codecs/es8328.c b/sound/soc/codecs/es8328.c index 7a9f65a..3ff7870 100644 --- a/sound/soc/codecs/es8328.c +++ b/sound/soc/codecs/es8328.c @@ -665,6 +665,7 @@ static int es8328_codec_probe(struct snd_soc_codec *codec) es8328->clk = devm_clk_get(codec->dev, NULL); if (IS_ERR(es8328->clk)) { dev_err(codec->dev, "codec clock missing or invalid\n"); + ret = PTR_ERR(es8328->clk); goto clk_fail; } -- cgit v1.1 From eadb0019d206591e34e864b62059b292e157d8fc Mon Sep 17 00:00:00 2001 From: Xiubo Li Date: Fri, 29 Aug 2014 15:12:12 +0800 Subject: ASoC: fsl-sai: using 'lsb-first' property instead of 'big-endian-data'. The 'big-endian-data' property is originally used to indicate whether the LSB firstly or MSB firstly will be transmitted to the CODEC or received from the CODEC, and there has nothing relation to the memory data. Generally, if the audio data in big endian format, which will be using the bytes reversion, Here this can only be used to bits reversion. So using the 'lsb-first' instead of 'big-endian-data' can make the code to be readable easier and more easy to understand what this property is used to do. This property used for configuring whether the LSB or the MSB is transmitted first for the fifo data. Signed-off-by: Xiubo Li Acked-by: Nicolin Chen Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_sai.c | 6 +++--- sound/soc/fsl/fsl_sai.h | 2 +- 2 files changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index a6eb784..7eeb1dd 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -175,7 +175,7 @@ static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai, bool tx = fsl_dir == FSL_FMT_TRANSMITTER; u32 val_cr2 = 0, val_cr4 = 0; - if (!sai->big_endian_data) + if (!sai->is_lsb_first) val_cr4 |= FSL_SAI_CR4_MF; /* DAI mode */ @@ -304,7 +304,7 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, val_cr5 |= FSL_SAI_CR5_WNW(word_width); val_cr5 |= FSL_SAI_CR5_W0W(word_width); - if (sai->big_endian_data) + if (sai->is_lsb_first) val_cr5 |= FSL_SAI_CR5_FBT(0); else val_cr5 |= FSL_SAI_CR5_FBT(word_width - 1); @@ -573,7 +573,7 @@ static int fsl_sai_probe(struct platform_device *pdev) if (of_device_is_compatible(pdev->dev.of_node, "fsl,imx6sx-sai")) sai->sai_on_imx = true; - sai->big_endian_data = of_property_read_bool(np, "big-endian-data"); + sai->is_lsb_first = of_property_read_bool(np, "lsb-first"); res = platform_get_resource(pdev, IORESOURCE_MEM, 0); base = devm_ioremap_resource(&pdev->dev, res); diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h index 2cded44..3466720 100644 --- a/sound/soc/fsl/fsl_sai.h +++ b/sound/soc/fsl/fsl_sai.h @@ -132,7 +132,7 @@ struct fsl_sai { struct clk *bus_clk; struct clk *mclk_clk[FSL_SAI_MCLK_MAX]; - bool big_endian_data; + bool is_lsb_first; bool is_dsp_mode; bool sai_on_imx; bool synchronous[2]; -- cgit v1.1 From ae70b190fce4a09a969dd69d0bd1c33441e24e60 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 25 Aug 2014 10:20:44 +0200 Subject: ASoC: ab8500-codec: Revert back to regmap Commit ff795d614bfa ("ASoC: ab8500: Convert register I/O to regmap") initially converted the ab8500 CODEC driver to use regmap rather than legacy ASoC IO. This was reverted though in commit 63e6d43bf80d ("ASoC: ab8500: Revert to using custom I/O functions") since the inital conversion was not working properly. This was presumebly because the SOC_SINGLE_XR_SX controls, which are used by this driver, did not properly support regmap at that point. This has since been fixed in commit 6137a5ca326d ("ASoC: Prepare SOC_SINGLE_XR_SX controls for regmap"). So revert back to regmap again. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/ab8500-codec.c | 62 +++++++++++++++++++---------------------- 1 file changed, 28 insertions(+), 34 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/ab8500-codec.c b/sound/soc/codecs/ab8500-codec.c index 62cf231..fd43827 100644 --- a/sound/soc/codecs/ab8500-codec.c +++ b/sound/soc/codecs/ab8500-codec.c @@ -125,6 +125,8 @@ struct ab8500_codec_drvdata_dbg { /* Private data for AB8500 device-driver */ struct ab8500_codec_drvdata { + struct regmap *regmap; + /* Sidetone */ long *sid_fir_values; enum sid_state sid_status; @@ -165,49 +167,35 @@ static inline const char *amic_type_str(enum amic_type type) */ /* Read a register from the audio-bank of AB8500 */ -static unsigned int ab8500_codec_read_reg(struct snd_soc_codec *codec, - unsigned int reg) +static int ab8500_codec_read_reg(void *context, unsigned int reg, + unsigned int *value) { + struct device *dev = context; int status; - unsigned int value = 0; u8 value8; - status = abx500_get_register_interruptible(codec->dev, AB8500_AUDIO, - reg, &value8); - if (status < 0) { - dev_err(codec->dev, - "%s: ERROR: Register (0x%02x:0x%02x) read failed (%d).\n", - __func__, (u8)AB8500_AUDIO, (u8)reg, status); - } else { - dev_dbg(codec->dev, - "%s: Read 0x%02x from register 0x%02x:0x%02x\n", - __func__, value8, (u8)AB8500_AUDIO, (u8)reg); - value = (unsigned int)value8; - } + status = abx500_get_register_interruptible(dev, AB8500_AUDIO, + reg, &value8); + *value = (unsigned int)value8; - return value; + return status; } /* Write to a register in the audio-bank of AB8500 */ -static int ab8500_codec_write_reg(struct snd_soc_codec *codec, - unsigned int reg, unsigned int value) +static int ab8500_codec_write_reg(void *context, unsigned int reg, + unsigned int value) { - int status; - - status = abx500_set_register_interruptible(codec->dev, AB8500_AUDIO, - reg, value); - if (status < 0) - dev_err(codec->dev, - "%s: ERROR: Register (%02x:%02x) write failed (%d).\n", - __func__, (u8)AB8500_AUDIO, (u8)reg, status); - else - dev_dbg(codec->dev, - "%s: Wrote 0x%02x into register %02x:%02x\n", - __func__, (u8)value, (u8)AB8500_AUDIO, (u8)reg); + struct device *dev = context; - return status; + return abx500_set_register_interruptible(dev, AB8500_AUDIO, + reg, value); } +static const struct regmap_config ab8500_codec_regmap = { + .reg_read = ab8500_codec_read_reg, + .reg_write = ab8500_codec_write_reg, +}; + /* * Controls - DAPM */ @@ -2564,9 +2552,6 @@ static int ab8500_codec_probe(struct snd_soc_codec *codec) static struct snd_soc_codec_driver ab8500_codec_driver = { .probe = ab8500_codec_probe, - .read = ab8500_codec_read_reg, - .write = ab8500_codec_write_reg, - .reg_word_size = sizeof(u8), .controls = ab8500_ctrls, .num_controls = ARRAY_SIZE(ab8500_ctrls), .dapm_widgets = ab8500_dapm_widgets, @@ -2591,6 +2576,15 @@ static int ab8500_codec_driver_probe(struct platform_device *pdev) drvdata->anc_status = ANC_UNCONFIGURED; dev_set_drvdata(&pdev->dev, drvdata); + drvdata->regmap = devm_regmap_init(&pdev->dev, NULL, &pdev->dev, + &ab8500_codec_regmap); + if (IS_ERR(drvdata->regmap)) { + status = PTR_ERR(drvdata->regmap); + dev_err(&pdev->dev, "%s: Failed to allocate regmap: %d\n", + __func__, status); + return status; + } + dev_dbg(&pdev->dev, "%s: Register codec.\n", __func__); status = snd_soc_register_codec(&pdev->dev, &ab8500_codec_driver, ab8500_codec_dai, -- cgit v1.1 From 7c7b9cf53d284fe12eeab6e13d3098b18cff4692 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 2 Sep 2014 04:05:30 -0700 Subject: ASoC: simple-card: fixup cpu_dai_name clear case f687d900d30a61dda38db2a99239f5284a86a309 (ASoC: simple-card: cpu_dai_name creates confusion when DT case) cleared cpu_dai_name for caring fmt_single_name case, and 179949bc04c7157a4b2279f62a842638b61f78f9 (ASoC: simple-card: remove dai_link->cpu_dai_name when DT) cared multi dai-link case. but, cpu_dai_name matching is required when fmt_multiple_name was used Signed-off-by: Kuninori Morimoto Tested-by: Jean-Francois Moine Signed-off-by: Mark Brown --- sound/soc/generic/simple-card.c | 29 +++++++++++++++++++---------- 1 file changed, 19 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index fd8b045..b63860d 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -112,8 +112,10 @@ static int asoc_simple_card_sub_parse_of(struct device_node *np, struct asoc_simple_dai *dai, struct device_node **p_node, - const char **name) + const char **name, + int *args_count) { + struct of_phandle_args args; struct device_node *node; struct clk *clk; u32 val; @@ -123,10 +125,15 @@ asoc_simple_card_sub_parse_of(struct device_node *np, * get node via "sound-dai = <&phandle port>" * it will be used as xxx_of_node on soc_bind_dai_link() */ - node = of_parse_phandle(np, "sound-dai", 0); - if (!node) - return -ENODEV; - *p_node = node; + ret = of_parse_phandle_with_args(np, "sound-dai", + "#sound-dai-cells", 0, &args); + if (ret) + return ret; + + *p_node = args.np; + + if (args_count) + *args_count = args.args_count; /* get dai->name */ ret = snd_soc_of_get_dai_name(np, name); @@ -176,7 +183,7 @@ static int asoc_simple_card_dai_link_of(struct device_node *node, char *name; char prop[128]; char *prefix = ""; - int ret; + int ret, cpu_args; if (is_top_level_node) prefix = "simple-audio-card,"; @@ -195,7 +202,8 @@ static int asoc_simple_card_dai_link_of(struct device_node *node, ret = asoc_simple_card_sub_parse_of(np, &dai_props->cpu_dai, &dai_link->cpu_of_node, - &dai_link->cpu_dai_name); + &dai_link->cpu_dai_name, + &cpu_args); if (ret < 0) goto dai_link_of_err; @@ -226,7 +234,7 @@ static int asoc_simple_card_dai_link_of(struct device_node *node, ret = asoc_simple_card_sub_parse_of(np, &dai_props->codec_dai, &dai_link->codec_of_node, - &dai_link->codec_dai_name); + &dai_link->codec_dai_name, NULL); if (ret < 0) goto dai_link_of_err; @@ -290,12 +298,13 @@ static int asoc_simple_card_dai_link_of(struct device_node *node, * soc_bind_dai_link() will check cpu name * after of_node matching if dai_link has cpu_dai_name. * but, it will never match if name was created by fmt_single_name() - * remove cpu_dai_name to escape name matching. + * remove cpu_dai_name if cpu_args was 0. * see * fmt_single_name() * fmt_multiple_name() */ - dai_link->cpu_dai_name = NULL; + if (!cpu_args) + dai_link->cpu_dai_name = NULL; dai_link_of_err: if (np) -- cgit v1.1 From 03be88ee4ab3acceddca43f11f4d01bcd6edcb93 Mon Sep 17 00:00:00 2001 From: Jyri Sarha Date: Wed, 3 Sep 2014 15:52:33 +0300 Subject: ASoC: tlv320aic31xx: Fix 24bit samples with I2S format and 12MHz mclk I2S format requires bitclock to have an exact amount of cycles in a frame for audio to work cleanly. With dsp formats that is not so important. Updates aic31xx_setup_pll() to look for a line in aic31xx_divs table that produces the best match for the bitclock and adds lines to aic31xx_divs for 12MHz mclk and 24bit samples. Signed-off-by: Jyri Sarha Tested-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic31xx.c | 51 ++++++++++++++++++++++++++++++---------- 1 file changed, 39 insertions(+), 12 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index 0f64c78..aea9e1f 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -189,46 +189,57 @@ static const struct aic31xx_rate_divs aic31xx_divs[] = { /* mclk rate pll: p j d dosr ndac mdac aors nadc madc */ /* 8k rate */ {12000000, 8000, 1, 8, 1920, 128, 48, 2, 128, 48, 2}, + {12000000, 8000, 1, 8, 1920, 128, 32, 3, 128, 32, 3}, {24000000, 8000, 2, 8, 1920, 128, 48, 2, 128, 48, 2}, {25000000, 8000, 2, 7, 8643, 128, 48, 2, 128, 48, 2}, /* 11.025k rate */ {12000000, 11025, 1, 7, 5264, 128, 32, 2, 128, 32, 2}, + {12000000, 11025, 1, 8, 4672, 128, 24, 3, 128, 24, 3}, {24000000, 11025, 2, 7, 5264, 128, 32, 2, 128, 32, 2}, {25000000, 11025, 2, 7, 2253, 128, 32, 2, 128, 32, 2}, /* 16k rate */ {12000000, 16000, 1, 8, 1920, 128, 24, 2, 128, 24, 2}, + {12000000, 16000, 1, 8, 1920, 128, 16, 3, 128, 16, 3}, {24000000, 16000, 2, 8, 1920, 128, 24, 2, 128, 24, 2}, {25000000, 16000, 2, 7, 8643, 128, 24, 2, 128, 24, 2}, /* 22.05k rate */ {12000000, 22050, 1, 7, 5264, 128, 16, 2, 128, 16, 2}, + {12000000, 22050, 1, 8, 4672, 128, 12, 3, 128, 12, 3}, {24000000, 22050, 2, 7, 5264, 128, 16, 2, 128, 16, 2}, {25000000, 22050, 2, 7, 2253, 128, 16, 2, 128, 16, 2}, /* 32k rate */ {12000000, 32000, 1, 8, 1920, 128, 12, 2, 128, 12, 2}, + {12000000, 32000, 1, 8, 1920, 128, 8, 3, 128, 8, 3}, {24000000, 32000, 2, 8, 1920, 128, 12, 2, 128, 12, 2}, {25000000, 32000, 2, 7, 8643, 128, 12, 2, 128, 12, 2}, /* 44.1k rate */ {12000000, 44100, 1, 7, 5264, 128, 8, 2, 128, 8, 2}, + {12000000, 44100, 1, 8, 4672, 128, 6, 3, 128, 6, 3}, {24000000, 44100, 2, 7, 5264, 128, 8, 2, 128, 8, 2}, {25000000, 44100, 2, 7, 2253, 128, 8, 2, 128, 8, 2}, /* 48k rate */ {12000000, 48000, 1, 8, 1920, 128, 8, 2, 128, 8, 2}, + {12000000, 48000, 1, 7, 6800, 96, 5, 4, 96, 5, 4}, {24000000, 48000, 2, 8, 1920, 128, 8, 2, 128, 8, 2}, {25000000, 48000, 2, 7, 8643, 128, 8, 2, 128, 8, 2}, /* 88.2k rate */ {12000000, 88200, 1, 7, 5264, 64, 8, 2, 64, 8, 2}, + {12000000, 88200, 1, 8, 4672, 64, 6, 3, 64, 6, 3}, {24000000, 88200, 2, 7, 5264, 64, 8, 2, 64, 8, 2}, {25000000, 88200, 2, 7, 2253, 64, 8, 2, 64, 8, 2}, /* 96k rate */ {12000000, 96000, 1, 8, 1920, 64, 8, 2, 64, 8, 2}, + {12000000, 96000, 1, 7, 6800, 48, 5, 4, 48, 5, 4}, {24000000, 96000, 2, 8, 1920, 64, 8, 2, 64, 8, 2}, {25000000, 96000, 2, 7, 8643, 64, 8, 2, 64, 8, 2}, /* 176.4k rate */ {12000000, 176400, 1, 7, 5264, 32, 8, 2, 32, 8, 2}, + {12000000, 176400, 1, 8, 4672, 32, 6, 3, 32, 6, 3}, {24000000, 176400, 2, 7, 5264, 32, 8, 2, 32, 8, 2}, {25000000, 176400, 2, 7, 2253, 32, 8, 2, 32, 8, 2}, /* 192k rate */ {12000000, 192000, 1, 8, 1920, 32, 8, 2, 32, 8, 2}, + {12000000, 192000, 1, 7, 6800, 24, 5, 4, 24, 5, 4}, {24000000, 192000, 2, 8, 1920, 32, 8, 2, 32, 8, 2}, {25000000, 192000, 2, 7, 8643, 32, 8, 2, 32, 8, 2}, }; @@ -680,7 +691,9 @@ static int aic31xx_setup_pll(struct snd_soc_codec *codec, struct snd_pcm_hw_params *params) { struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec); + int bclk_score = snd_soc_params_to_frame_size(params); int bclk_n = 0; + int match = -1; int i; /* Use PLL as CODEC_CLKIN and DAC_CLK as BDIV_CLKIN */ @@ -691,15 +704,37 @@ static int aic31xx_setup_pll(struct snd_soc_codec *codec, for (i = 0; i < ARRAY_SIZE(aic31xx_divs); i++) { if (aic31xx_divs[i].rate == params_rate(params) && - aic31xx_divs[i].mclk == aic31xx->sysclk) - break; + aic31xx_divs[i].mclk == aic31xx->sysclk) { + int s = (aic31xx_divs[i].dosr * aic31xx_divs[i].mdac) % + snd_soc_params_to_frame_size(params); + int bn = (aic31xx_divs[i].dosr * aic31xx_divs[i].mdac) / + snd_soc_params_to_frame_size(params); + if (s < bclk_score && bn > 0) { + match = i; + bclk_n = bn; + bclk_score = s; + } + } } - if (i == ARRAY_SIZE(aic31xx_divs)) { - dev_err(codec->dev, "%s: Sampling rate %u not supported\n", + if (match == -1) { + dev_err(codec->dev, + "%s: Sample rate (%u) and format not supported\n", __func__, params_rate(params)); + /* See bellow for details how fix this. */ return -EINVAL; } + if (bclk_score != 0) { + dev_warn(codec->dev, "Can not produce exact bitclock"); + /* This is fine if using dsp format, but if using i2s + there may be trouble. To fix the issue edit the + aic31xx_divs table for your mclk and sample + rate. Details can be found from: + http://www.ti.com/lit/ds/symlink/tlv320aic3100.pdf + Section: 5.6 CLOCK Generation and PLL + */ + } + i = match; /* PLL configuration */ snd_soc_update_bits(codec, AIC31XX_PLLPR, AIC31XX_PLL_MASK, @@ -729,14 +764,6 @@ static int aic31xx_setup_pll(struct snd_soc_codec *codec, snd_soc_write(codec, AIC31XX_AOSR, aic31xx_divs[i].aosr); /* Bit clock divider configuration. */ - bclk_n = (aic31xx_divs[i].dosr * aic31xx_divs[i].mdac) - / snd_soc_params_to_frame_size(params); - if (bclk_n == 0) { - dev_err(codec->dev, "%s: Not enough BLCK bandwidth\n", - __func__); - return -EINVAL; - } - snd_soc_update_bits(codec, AIC31XX_BCLKN, AIC31XX_PLL_MASK, bclk_n); -- cgit v1.1 From 7ed36e96fd05470e98e7daf648f9cf7f38609670 Mon Sep 17 00:00:00 2001 From: Jyri Sarha Date: Wed, 3 Sep 2014 15:52:34 +0300 Subject: ASoC: tlv320aic31xx: Choose PLL p divider automatically This simplifies aic31xx_divs table. There is no more need for p_val or separate lines for 12 and 24 MHz mclks. Signed-off-by: Jyri Sarha Tested-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic31xx.c | 101 +++++++++++++++++++-------------------- 1 file changed, 50 insertions(+), 51 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index 40a636f..145fe5b 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -167,13 +167,13 @@ struct aic31xx_priv { struct regulator_bulk_data supplies[AIC31XX_NUM_SUPPLIES]; struct aic31xx_disable_nb disable_nb[AIC31XX_NUM_SUPPLIES]; unsigned int sysclk; + u8 p_div; int rate_div_line; }; struct aic31xx_rate_divs { - u32 mclk; + u32 mclk_p; u32 rate; - u8 p_val; u8 pll_j; u16 pll_d; u16 dosr; @@ -186,62 +186,51 @@ struct aic31xx_rate_divs { /* ADC dividers can be disabled by cofiguring them to 0 */ static const struct aic31xx_rate_divs aic31xx_divs[] = { - /* mclk rate pll: p j d dosr ndac mdac aors nadc madc */ + /* mclk/p rate pll: j d dosr ndac mdac aors nadc madc */ /* 8k rate */ - {12000000, 8000, 1, 8, 1920, 128, 48, 2, 128, 48, 2}, - {12000000, 8000, 1, 8, 1920, 128, 32, 3, 128, 32, 3}, - {24000000, 8000, 2, 8, 1920, 128, 48, 2, 128, 48, 2}, - {25000000, 8000, 2, 7, 8643, 128, 48, 2, 128, 48, 2}, + {12000000, 8000, 8, 1920, 128, 48, 2, 128, 48, 2}, + {12000000, 8000, 8, 1920, 128, 32, 3, 128, 32, 3}, + {12500000, 8000, 7, 8643, 128, 48, 2, 128, 48, 2}, /* 11.025k rate */ - {12000000, 11025, 1, 7, 5264, 128, 32, 2, 128, 32, 2}, - {12000000, 11025, 1, 8, 4672, 128, 24, 3, 128, 24, 3}, - {24000000, 11025, 2, 7, 5264, 128, 32, 2, 128, 32, 2}, - {25000000, 11025, 2, 7, 2253, 128, 32, 2, 128, 32, 2}, + {12000000, 11025, 7, 5264, 128, 32, 2, 128, 32, 2}, + {12000000, 11025, 8, 4672, 128, 24, 3, 128, 24, 3}, + {12500000, 11025, 7, 2253, 128, 32, 2, 128, 32, 2}, /* 16k rate */ - {12000000, 16000, 1, 8, 1920, 128, 24, 2, 128, 24, 2}, - {12000000, 16000, 1, 8, 1920, 128, 16, 3, 128, 16, 3}, - {24000000, 16000, 2, 8, 1920, 128, 24, 2, 128, 24, 2}, - {25000000, 16000, 2, 7, 8643, 128, 24, 2, 128, 24, 2}, + {12000000, 16000, 8, 1920, 128, 24, 2, 128, 24, 2}, + {12000000, 16000, 8, 1920, 128, 16, 3, 128, 16, 3}, + {12500000, 16000, 7, 8643, 128, 24, 2, 128, 24, 2}, /* 22.05k rate */ - {12000000, 22050, 1, 7, 5264, 128, 16, 2, 128, 16, 2}, - {12000000, 22050, 1, 8, 4672, 128, 12, 3, 128, 12, 3}, - {24000000, 22050, 2, 7, 5264, 128, 16, 2, 128, 16, 2}, - {25000000, 22050, 2, 7, 2253, 128, 16, 2, 128, 16, 2}, + {12000000, 22050, 7, 5264, 128, 16, 2, 128, 16, 2}, + {12000000, 22050, 8, 4672, 128, 12, 3, 128, 12, 3}, + {12500000, 22050, 7, 2253, 128, 16, 2, 128, 16, 2}, /* 32k rate */ - {12000000, 32000, 1, 8, 1920, 128, 12, 2, 128, 12, 2}, - {12000000, 32000, 1, 8, 1920, 128, 8, 3, 128, 8, 3}, - {24000000, 32000, 2, 8, 1920, 128, 12, 2, 128, 12, 2}, - {25000000, 32000, 2, 7, 8643, 128, 12, 2, 128, 12, 2}, + {12000000, 32000, 8, 1920, 128, 12, 2, 128, 12, 2}, + {12000000, 32000, 8, 1920, 128, 8, 3, 128, 8, 3}, + {12500000, 32000, 7, 8643, 128, 12, 2, 128, 12, 2}, /* 44.1k rate */ - {12000000, 44100, 1, 7, 5264, 128, 8, 2, 128, 8, 2}, - {12000000, 44100, 1, 8, 4672, 128, 6, 3, 128, 6, 3}, - {24000000, 44100, 2, 7, 5264, 128, 8, 2, 128, 8, 2}, - {25000000, 44100, 2, 7, 2253, 128, 8, 2, 128, 8, 2}, + {12000000, 44100, 7, 5264, 128, 8, 2, 128, 8, 2}, + {12000000, 44100, 8, 4672, 128, 6, 3, 128, 6, 3}, + {12500000, 44100, 7, 2253, 128, 8, 2, 128, 8, 2}, /* 48k rate */ - {12000000, 48000, 1, 8, 1920, 128, 8, 2, 128, 8, 2}, - {12000000, 48000, 1, 7, 6800, 96, 5, 4, 96, 5, 4}, - {24000000, 48000, 2, 8, 1920, 128, 8, 2, 128, 8, 2}, - {25000000, 48000, 2, 7, 8643, 128, 8, 2, 128, 8, 2}, + {12000000, 48000, 8, 1920, 128, 8, 2, 128, 8, 2}, + {12000000, 48000, 7, 6800, 96, 5, 4, 96, 5, 4}, + {12500000, 48000, 7, 8643, 128, 8, 2, 128, 8, 2}, /* 88.2k rate */ - {12000000, 88200, 1, 7, 5264, 64, 8, 2, 64, 8, 2}, - {12000000, 88200, 1, 8, 4672, 64, 6, 3, 64, 6, 3}, - {24000000, 88200, 2, 7, 5264, 64, 8, 2, 64, 8, 2}, - {25000000, 88200, 2, 7, 2253, 64, 8, 2, 64, 8, 2}, + {12000000, 88200, 7, 5264, 64, 8, 2, 64, 8, 2}, + {12000000, 88200, 8, 4672, 64, 6, 3, 64, 6, 3}, + {12500000, 88200, 7, 2253, 64, 8, 2, 64, 8, 2}, /* 96k rate */ - {12000000, 96000, 1, 8, 1920, 64, 8, 2, 64, 8, 2}, - {12000000, 96000, 1, 7, 6800, 48, 5, 4, 48, 5, 4}, - {24000000, 96000, 2, 8, 1920, 64, 8, 2, 64, 8, 2}, - {25000000, 96000, 2, 7, 8643, 64, 8, 2, 64, 8, 2}, + {12000000, 96000, 8, 1920, 64, 8, 2, 64, 8, 2}, + {12000000, 96000, 7, 6800, 48, 5, 4, 48, 5, 4}, + {12500000, 96000, 7, 8643, 64, 8, 2, 64, 8, 2}, /* 176.4k rate */ - {12000000, 176400, 1, 7, 5264, 32, 8, 2, 32, 8, 2}, - {12000000, 176400, 1, 8, 4672, 32, 6, 3, 32, 6, 3}, - {24000000, 176400, 2, 7, 5264, 32, 8, 2, 32, 8, 2}, - {25000000, 176400, 2, 7, 2253, 32, 8, 2, 32, 8, 2}, + {12000000, 176400, 7, 5264, 32, 8, 2, 32, 8, 2}, + {12000000, 176400, 8, 4672, 32, 6, 3, 32, 6, 3}, + {12500000, 176400, 7, 2253, 32, 8, 2, 32, 8, 2}, /* 192k rate */ - {12000000, 192000, 1, 8, 1920, 32, 8, 2, 32, 8, 2}, - {12000000, 192000, 1, 7, 6800, 24, 5, 4, 24, 5, 4}, - {24000000, 192000, 2, 8, 1920, 32, 8, 2, 32, 8, 2}, - {25000000, 192000, 2, 7, 8643, 32, 8, 2, 32, 8, 2}, + {12000000, 192000, 8, 1920, 32, 8, 2, 32, 8, 2}, + {12000000, 192000, 7, 6800, 24, 5, 4, 24, 5, 4}, + {12500000, 192000, 7, 8643, 32, 8, 2, 32, 8, 2}, }; static const char * const ldac_in_text[] = { @@ -692,6 +681,7 @@ static int aic31xx_setup_pll(struct snd_soc_codec *codec, { struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec); int bclk_score = snd_soc_params_to_frame_size(params); + int mclk_p = aic31xx->sysclk / aic31xx->p_div; int bclk_n = 0; int match = -1; int i; @@ -704,7 +694,7 @@ static int aic31xx_setup_pll(struct snd_soc_codec *codec, for (i = 0; i < ARRAY_SIZE(aic31xx_divs); i++) { if (aic31xx_divs[i].rate == params_rate(params) && - aic31xx_divs[i].mclk == aic31xx->sysclk) { + aic31xx_divs[i].mclk_p == mclk_p) { int s = (aic31xx_divs[i].dosr * aic31xx_divs[i].mdac) % snd_soc_params_to_frame_size(params); int bn = (aic31xx_divs[i].dosr * aic31xx_divs[i].mdac) / @@ -738,7 +728,7 @@ static int aic31xx_setup_pll(struct snd_soc_codec *codec, /* PLL configuration */ snd_soc_update_bits(codec, AIC31XX_PLLPR, AIC31XX_PLL_MASK, - (aic31xx_divs[i].p_val << 4) | 0x01); + (aic31xx->p_div << 4) | 0x01); snd_soc_write(codec, AIC31XX_PLLJ, aic31xx_divs[i].pll_j); snd_soc_write(codec, AIC31XX_PLLDMSB, @@ -772,7 +762,7 @@ static int aic31xx_setup_pll(struct snd_soc_codec *codec, dev_dbg(codec->dev, "pll %d.%04d/%d dosr %d n %d m %d aosr %d n %d m %d bclk_n %d\n", aic31xx_divs[i].pll_j, aic31xx_divs[i].pll_d, - aic31xx_divs[i].p_val, aic31xx_divs[i].dosr, + aic31xx->p_div, aic31xx_divs[i].dosr, aic31xx_divs[i].ndac, aic31xx_divs[i].mdac, aic31xx_divs[i].aosr, aic31xx_divs[i].nadc, aic31xx_divs[i].madc, bclk_n); @@ -912,7 +902,16 @@ static int aic31xx_set_dai_sysclk(struct snd_soc_dai *codec_dai, dev_dbg(codec->dev, "## %s: clk_id = %d, freq = %d, dir = %d\n", __func__, clk_id, freq, dir); - for (i = 0; aic31xx_divs[i].mclk != freq; i++) { + for (i = 1; freq/i > 20000000 && i < 8; i++) + ; + if (freq/i > 20000000) { + dev_err(aic31xx->dev, "%s: Too high mclk frequency %u\n", + __func__, freq); + return -EINVAL; + } + aic31xx->p_div = i; + + for (i = 0; aic31xx_divs[i].mclk_p != freq/aic31xx->p_div; i++) { if (i == ARRAY_SIZE(aic31xx_divs)) { dev_err(aic31xx->dev, "%s: Unsupported frequency %d\n", __func__, freq); -- cgit v1.1 From b8a3ee820f7b0802c9b90a9f3426dbda54e93d09 Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Wed, 3 Sep 2014 15:42:48 +0300 Subject: ASoC: max98090: Add recovery for PLL lock failure All MAX98090 input clocks MCLK, LRCLK and BCLK must be running and stable before powering on the codec in slave mode. Otherwise the PLL may not lock to LRCLK causing silence in playback and capture. How often that happens is somewhat hardware and clock configuration specific. Now if wanting to follow strictly this clocks must be active before powering the codec on requirement we should have a notification from DAI driver to codec driver when clocks are activated and take codec out of shutdown only after that. Plus take care of possible active bypass paths. However, when PLL unlock occurs, MAX98090 asserts the PLL Unlock Flag which can be configured as an IRQ source. This allows to workaround around the issue by toggling the codec power shortly in case of PLL lock failure. In order to prevent needlessly toggling codec power in case of short PLL unlocks at the beginning of stream this patch implements delayed activation for PLL unlock interrupt. Then workaround is run only when the PLL doesn't lock at all. Power toggling workaround for PLL unlock comes originally from Liam Girdwood and delayed activation from me. Signed-off-by: Jarkko Nikula Signed-off-by: Mark Brown --- sound/soc/codecs/max98090.c | 111 +++++++++++++++++++++++++++++++++++++++++++- sound/soc/codecs/max98090.h | 3 ++ 2 files changed, 112 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index 4a063fa..f154365 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -1972,6 +1972,102 @@ static int max98090_dai_digital_mute(struct snd_soc_dai *codec_dai, int mute) return 0; } +static int max98090_dai_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + if (!max98090->master && dai->active == 1) + queue_delayed_work(system_power_efficient_wq, + &max98090->pll_det_enable_work, + msecs_to_jiffies(10)); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + if (!max98090->master && dai->active == 1) + schedule_work(&max98090->pll_det_disable_work); + break; + default: + break; + } + + return 0; +} + +static void max98090_pll_det_enable_work(struct work_struct *work) +{ + struct max98090_priv *max98090 = + container_of(work, struct max98090_priv, + pll_det_enable_work.work); + struct snd_soc_codec *codec = max98090->codec; + unsigned int status, mask; + + /* + * Clear status register in order to clear possibly already occurred + * PLL unlock. If PLL hasn't still locked, the status will be set + * again and PLL unlock interrupt will occur. + * Note this will clear all status bits + */ + regmap_read(max98090->regmap, M98090_REG_DEVICE_STATUS, &status); + + /* + * Queue jack work in case jack state has just changed but handler + * hasn't run yet + */ + regmap_read(max98090->regmap, M98090_REG_INTERRUPT_S, &mask); + status &= mask; + if (status & M98090_JDET_MASK) + queue_delayed_work(system_power_efficient_wq, + &max98090->jack_work, + msecs_to_jiffies(100)); + + /* Enable PLL unlock interrupt */ + snd_soc_update_bits(codec, M98090_REG_INTERRUPT_S, + M98090_IULK_MASK, + 1 << M98090_IULK_SHIFT); +} + +static void max98090_pll_det_disable_work(struct work_struct *work) +{ + struct max98090_priv *max98090 = + container_of(work, struct max98090_priv, pll_det_disable_work); + struct snd_soc_codec *codec = max98090->codec; + + cancel_delayed_work_sync(&max98090->pll_det_enable_work); + + /* Disable PLL unlock interrupt */ + snd_soc_update_bits(codec, M98090_REG_INTERRUPT_S, + M98090_IULK_MASK, 0); +} + +static void max98090_pll_work(struct work_struct *work) +{ + struct max98090_priv *max98090 = + container_of(work, struct max98090_priv, pll_work); + struct snd_soc_codec *codec = max98090->codec; + + if (!snd_soc_codec_is_active(codec)) + return; + + dev_info(codec->dev, "PLL unlocked\n"); + + /* Toggle shutdown OFF then ON */ + snd_soc_update_bits(codec, M98090_REG_DEVICE_SHUTDOWN, + M98090_SHDNN_MASK, 0); + msleep(10); + snd_soc_update_bits(codec, M98090_REG_DEVICE_SHUTDOWN, + M98090_SHDNN_MASK, M98090_SHDNN_MASK); + + /* Give PLL time to lock */ + msleep(10); +} + static void max98090_jack_work(struct work_struct *work) { struct max98090_priv *max98090 = container_of(work, @@ -2103,8 +2199,10 @@ static irqreturn_t max98090_interrupt(int irq, void *data) if (active & M98090_SLD_MASK) dev_dbg(codec->dev, "M98090_SLD_MASK\n"); - if (active & M98090_ULK_MASK) - dev_err(codec->dev, "M98090_ULK_MASK\n"); + if (active & M98090_ULK_MASK) { + dev_dbg(codec->dev, "M98090_ULK_MASK\n"); + schedule_work(&max98090->pll_work); + } if (active & M98090_JDET_MASK) { dev_dbg(codec->dev, "M98090_JDET_MASK\n"); @@ -2177,6 +2275,7 @@ static struct snd_soc_dai_ops max98090_dai_ops = { .set_tdm_slot = max98090_set_tdm_slot, .hw_params = max98090_dai_hw_params, .digital_mute = max98090_dai_digital_mute, + .trigger = max98090_dai_trigger, }; static struct snd_soc_dai_driver max98090_dai[] = { @@ -2258,6 +2357,11 @@ static int max98090_probe(struct snd_soc_codec *codec) max98090->jack_state = M98090_JACK_STATE_NO_HEADSET; INIT_DELAYED_WORK(&max98090->jack_work, max98090_jack_work); + INIT_DELAYED_WORK(&max98090->pll_det_enable_work, + max98090_pll_det_enable_work); + INIT_WORK(&max98090->pll_det_disable_work, + max98090_pll_det_disable_work); + INIT_WORK(&max98090->pll_work, max98090_pll_work); /* Enable jack detection */ snd_soc_write(codec, M98090_REG_JACK_DETECT, @@ -2310,6 +2414,9 @@ static int max98090_remove(struct snd_soc_codec *codec) struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); cancel_delayed_work_sync(&max98090->jack_work); + cancel_delayed_work_sync(&max98090->pll_det_enable_work); + cancel_work_sync(&max98090->pll_det_disable_work); + cancel_work_sync(&max98090->pll_work); return 0; } diff --git a/sound/soc/codecs/max98090.h b/sound/soc/codecs/max98090.h index cf1b606..14427a5 100644 --- a/sound/soc/codecs/max98090.h +++ b/sound/soc/codecs/max98090.h @@ -1532,6 +1532,9 @@ struct max98090_priv { int irq; int jack_state; struct delayed_work jack_work; + struct delayed_work pll_det_enable_work; + struct work_struct pll_det_disable_work; + struct work_struct pll_work; struct snd_soc_jack *jack; unsigned int dai_fmt; int tdm_slots; -- cgit v1.1 From b43cfb245f7346cbb25c1919577d9607d2adb974 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 2 Sep 2014 22:20:30 +0200 Subject: ASoC: adau1373: Remove unnecessary suspend/resume bias level changes The ASoC core will only call the suspend/resume callbacks when the device's DAPM context is idle. Since this driver sets idle_bias_off to true this means that the device is already in SND_SOC_BIAS_OFF when the suspend callback is called, so there is no need to manually set this state again. There is also no need to go to SND_SOC_BIAS_STANDBY in the resume callback since the core will go right back to SND_SOC_BIAS_OFF. Also drop the regcache_cache_only() calls from the suspend and resume handlers. There shouldn't be any IO happening after suspend and before resume. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/adau1373.c | 14 -------------- 1 file changed, 14 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/adau1373.c b/sound/soc/codecs/adau1373.c index 1ff7d4d..1947565 100644 --- a/sound/soc/codecs/adau1373.c +++ b/sound/soc/codecs/adau1373.c @@ -1454,23 +1454,10 @@ static int adau1373_remove(struct snd_soc_codec *codec) return 0; } -static int adau1373_suspend(struct snd_soc_codec *codec) -{ - struct adau1373 *adau1373 = snd_soc_codec_get_drvdata(codec); - int ret; - - ret = adau1373_set_bias_level(codec, SND_SOC_BIAS_OFF); - regcache_cache_only(adau1373->regmap, true); - - return ret; -} - static int adau1373_resume(struct snd_soc_codec *codec) { struct adau1373 *adau1373 = snd_soc_codec_get_drvdata(codec); - regcache_cache_only(adau1373->regmap, false); - adau1373_set_bias_level(codec, SND_SOC_BIAS_STANDBY); regcache_sync(adau1373->regmap); return 0; @@ -1502,7 +1489,6 @@ static const struct regmap_config adau1373_regmap_config = { static struct snd_soc_codec_driver adau1373_codec_driver = { .probe = adau1373_probe, .remove = adau1373_remove, - .suspend = adau1373_suspend, .resume = adau1373_resume, .set_bias_level = adau1373_set_bias_level, .idle_bias_off = true, -- cgit v1.1 From 8e6fe35eabc64f35eff5844a2e542c403a00db15 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 2 Sep 2014 22:20:31 +0200 Subject: ASoC: lm49453: Remove unnecessary suspend/resume bias level changes The ASoC core will only call the suspend/resume callbacks when the device's DAPM context is idle. Since this driver sets idle_bias_off to true this means that the device is already in SND_SOC_BIAS_OFF when the suspend callback is called, so there is no need to manually set this state again. There is also no need to go to SND_SOC_BIAS_STANDBY in the resume callback since the core will go right back to SND_SOC_BIAS_OFF. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/lm49453.c | 14 -------------- 1 file changed, 14 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/lm49453.c b/sound/soc/codecs/lm49453.c index 275b3f7..c1ae576 100644 --- a/sound/soc/codecs/lm49453.c +++ b/sound/soc/codecs/lm49453.c @@ -1395,18 +1395,6 @@ static struct snd_soc_dai_driver lm49453_dai[] = { }, }; -static int lm49453_suspend(struct snd_soc_codec *codec) -{ - lm49453_set_bias_level(codec, SND_SOC_BIAS_OFF); - return 0; -} - -static int lm49453_resume(struct snd_soc_codec *codec) -{ - lm49453_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - return 0; -} - /* power down chip */ static int lm49453_remove(struct snd_soc_codec *codec) { @@ -1416,8 +1404,6 @@ static int lm49453_remove(struct snd_soc_codec *codec) static struct snd_soc_codec_driver soc_codec_dev_lm49453 = { .remove = lm49453_remove, - .suspend = lm49453_suspend, - .resume = lm49453_resume, .set_bias_level = lm49453_set_bias_level, .controls = lm49453_snd_controls, .num_controls = ARRAY_SIZE(lm49453_snd_controls), -- cgit v1.1 From 7d1a99da0861330f02de5c0f59df1d338477cb54 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 2 Sep 2014 22:20:32 +0200 Subject: ASoC: tlv320aic3x: Remove unnecessary suspend/resume bias level changes The ASoC core will only call the suspend/resume callbacks when the device's DAPM context is idle. Since this driver sets idle_bias_off to true this means that the device is already in SND_SOC_BIAS_OFF when the suspend callback is called, so there is no need to manually set this state again. There is also no need to go to SND_SOC_BIAS_STANDBY in the resume callback since the core will go right back to SND_SOC_BIAS_OFF. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic3x.c | 16 ---------------- 1 file changed, 16 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 64f179e..f2c416d 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1222,20 +1222,6 @@ static struct snd_soc_dai_driver aic3x_dai = { .symmetric_rates = 1, }; -static int aic3x_suspend(struct snd_soc_codec *codec) -{ - aic3x_set_bias_level(codec, SND_SOC_BIAS_OFF); - - return 0; -} - -static int aic3x_resume(struct snd_soc_codec *codec) -{ - aic3x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - - return 0; -} - static void aic3x_mono_init(struct snd_soc_codec *codec) { /* DAC to Mono Line Out default volume and route to Output mixer */ @@ -1429,8 +1415,6 @@ static struct snd_soc_codec_driver soc_codec_dev_aic3x = { .idle_bias_off = true, .probe = aic3x_probe, .remove = aic3x_remove, - .suspend = aic3x_suspend, - .resume = aic3x_resume, .controls = aic3x_snd_controls, .num_controls = ARRAY_SIZE(aic3x_snd_controls), .dapm_widgets = aic3x_dapm_widgets, -- cgit v1.1 From a7edeba4cbbd0f3d22d6d54da7c507bda29b2658 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 2 Sep 2014 22:20:33 +0200 Subject: ASoC: wm8804: Remove unnecessary suspend/resume bias level changes The ASoC core will only call the suspend/resume callbacks when the device's DAPM context is idle. Since this driver sets idle_bias_off to true this means that the device is already in SND_SOC_BIAS_OFF when the suspend callback is called, so there is no need to manually set this state again. There is also no need to go to SND_SOC_BIAS_STANDBY in the resume callback since the core will go right back to SND_SOC_BIAS_OFF. Signed-off-by: Lars-Peter Clausen Acked-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm8804.c | 19 ------------------- 1 file changed, 19 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8804.c b/sound/soc/codecs/wm8804.c index 0ea01df..3addc5f 100644 --- a/sound/soc/codecs/wm8804.c +++ b/sound/soc/codecs/wm8804.c @@ -518,23 +518,6 @@ static int wm8804_set_bias_level(struct snd_soc_codec *codec, return 0; } -#ifdef CONFIG_PM -static int wm8804_suspend(struct snd_soc_codec *codec) -{ - wm8804_set_bias_level(codec, SND_SOC_BIAS_OFF); - return 0; -} - -static int wm8804_resume(struct snd_soc_codec *codec) -{ - wm8804_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - return 0; -} -#else -#define wm8804_suspend NULL -#define wm8804_resume NULL -#endif - static int wm8804_remove(struct snd_soc_codec *codec) { struct wm8804_priv *wm8804; @@ -671,8 +654,6 @@ static struct snd_soc_dai_driver wm8804_dai = { static struct snd_soc_codec_driver soc_codec_dev_wm8804 = { .probe = wm8804_probe, .remove = wm8804_remove, - .suspend = wm8804_suspend, - .resume = wm8804_resume, .set_bias_level = wm8804_set_bias_level, .idle_bias_off = true, -- cgit v1.1 From e02c716d2ec065fd58c2fc8100fd5f359ab61e7e Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 2 Sep 2014 22:20:34 +0200 Subject: ASoC: wm8995: Remove unnecessary suspend/resume bias level changes The ASoC core will only call the suspend/resume callbacks when the device's DAPM context is idle. Since this driver sets idle_bias_off to true this means that the device is already in SND_SOC_BIAS_OFF when the suspend callback is called, so there is no need to manually set this state again. There is also no need to go to SND_SOC_BIAS_STANDBY in the resume callback since the core will go right back to SND_SOC_BIAS_OFF. Signed-off-by: Lars-Peter Clausen Acked-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm8995.c | 19 ------------------- 1 file changed, 19 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8995.c b/sound/soc/codecs/wm8995.c index cae4ac5..1288ede 100644 --- a/sound/soc/codecs/wm8995.c +++ b/sound/soc/codecs/wm8995.c @@ -1998,23 +1998,6 @@ static int wm8995_set_bias_level(struct snd_soc_codec *codec, return 0; } -#ifdef CONFIG_PM -static int wm8995_suspend(struct snd_soc_codec *codec) -{ - wm8995_set_bias_level(codec, SND_SOC_BIAS_OFF); - return 0; -} - -static int wm8995_resume(struct snd_soc_codec *codec) -{ - wm8995_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - return 0; -} -#else -#define wm8995_suspend NULL -#define wm8995_resume NULL -#endif - static int wm8995_remove(struct snd_soc_codec *codec) { struct wm8995_priv *wm8995; @@ -2220,8 +2203,6 @@ static struct snd_soc_dai_driver wm8995_dai[] = { static struct snd_soc_codec_driver soc_codec_dev_wm8995 = { .probe = wm8995_probe, .remove = wm8995_remove, - .suspend = wm8995_suspend, - .resume = wm8995_resume, .set_bias_level = wm8995_set_bias_level, .idle_bias_off = true, }; -- cgit v1.1 From 9cfb76905da525579d0d43c1205c86033d0ae3e5 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 4 Sep 2014 10:59:41 +0300 Subject: ASoC: tlv320aic31xx: Enable support for S24_LE format S24_LE is the same on the bus as S24_3LE, which means the codec can support it. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic31xx.h | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320aic31xx.h b/sound/soc/codecs/tlv320aic31xx.h index 52ed57c..fe16c34 100644 --- a/sound/soc/codecs/tlv320aic31xx.h +++ b/sound/soc/codecs/tlv320aic31xx.h @@ -18,7 +18,8 @@ #define AIC31XX_RATES SNDRV_PCM_RATE_8000_192000 #define AIC31XX_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE \ - | SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE) + | SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_LE \ + | SNDRV_PCM_FMTBIT_S32_LE) #define AIC31XX_STEREO_CLASS_D_BIT 0x1 -- cgit v1.1 From fe0a29e163a5d045c73faab682a8dac71c2f8012 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 4 Sep 2014 10:52:53 +0300 Subject: ASoC: davinci-mcasp: Correct rx format unit configuration In case of capture we should not use rotation. The reverse and mask is enough to get the data align correctly from the bus to MCU: Format data from bus after reverse (XRBUF) S16_LE: |LSB|MSB|xxx|xxx| |xxx|xxx|MSB|LSB| S24_3LE: |LSB|DAT|MSB|xxx| |xxx|MSB|DAT|LSB| S24_LE: |LSB|DAT|MSB|xxx| |xxx|MSB|DAT|LSB| S32_LE: |LSB|DAT|DAT|MSB| |MSB|DAT|DAT|LSB| With this patch all supported formats will work for playback and capture. Reported-by: Jyri Sarha (broken S24_3LE capture) Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/davinci/davinci-mcasp.c | 11 ++++++++++- 1 file changed, 10 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index c28508d..0062601 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -459,8 +459,17 @@ static int davinci_config_channel_size(struct davinci_mcasp *mcasp, { u32 fmt; u32 tx_rotate = (word_length / 4) & 0x7; - u32 rx_rotate = (32 - word_length) / 4; u32 mask = (1ULL << word_length) - 1; + /* + * For captured data we should not rotate, inversion and masking is + * enoguh to get the data to the right position: + * Format data from bus after reverse (XRBUF) + * S16_LE: |LSB|MSB|xxx|xxx| |xxx|xxx|MSB|LSB| + * S24_3LE: |LSB|DAT|MSB|xxx| |xxx|MSB|DAT|LSB| + * S24_LE: |LSB|DAT|MSB|xxx| |xxx|MSB|DAT|LSB| + * S32_LE: |LSB|DAT|DAT|MSB| |MSB|DAT|DAT|LSB| + */ + u32 rx_rotate = 0; /* * if s BCLK-to-LRCLK ratio has been configured via the set_clkdiv() -- cgit v1.1 From 01e0df6647e713469466c7bb6d7157c2e3046192 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 4 Sep 2014 19:44:04 +0200 Subject: ASoC: Set card->instantiated to false when removing the card Set card->instantiated to false when the card is removed to make sure that operations that expect the card to be fully instantiated do not run anymore during card removal. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 1b422c5..ff9d289 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3810,8 +3810,10 @@ EXPORT_SYMBOL_GPL(snd_soc_register_card); */ int snd_soc_unregister_card(struct snd_soc_card *card) { - if (card->instantiated) + if (card->instantiated) { + card->instantiated = false; soc_cleanup_card_resources(card); + } dev_dbg(card->dev, "ASoC: Unregistered card '%s'\n", card->name); return 0; -- cgit v1.1 From 1c325f771a88579f227fe017e4ee77d852cf5435 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 4 Sep 2014 19:44:05 +0200 Subject: ASoC: Shutdown DAPM contexts when removing a card Currently when a ASoC sound card is unregistered we leave the individual components in their current state, just call the remove() callback and leave it to the drivers to do the proper shutdown/cleanup. This patch introduces a call to snd_soc_dapm_shutdown() when removing the card. This will make sure that all DAPM widgets are properly powered down and all DAPM contexts are put at the SND_SOC_BIAS_OFF level. This will ensure that all components are properly powered down when the card is removed. Since a lot of drivers manually go to SND_SOC_BIAS_OFF in their remove callback this will also allow us to remove a bit of duplicated code. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index ff9d289..068785f 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3812,6 +3812,7 @@ int snd_soc_unregister_card(struct snd_soc_card *card) { if (card->instantiated) { card->instantiated = false; + snd_soc_dapm_shutdown(card); soc_cleanup_card_resources(card); } dev_dbg(card->dev, "ASoC: Unregistered card '%s'\n", card->name); -- cgit v1.1 From 86dbf2ac6fcb2d2932d4610f2dfe0954aa0633f7 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 4 Sep 2014 19:44:06 +0200 Subject: ASoC: Add support for automatically going to BIAS_OFF on suspend There is a substantial amount of drivers that in go to SND_SOC_BIAS_OFF on suspend and go back to SND_SOC_BIAS_SUSPEND on resume (Often this is even the only thing done in the suspend and resume handlers). This patch introduces a new suspend_bias_off flag, which when set by a driver will let the ASoC core automatically put the device's DAPM context at the SND_SOC_BIAS_OFF level during suspend. Once the device is resumed the DAPM context will go back to SND_SOC_BIAS_STANDBY (if the context is idle, otherwise to SND_SOC_BIAS_ON). This will allow us to remove a fair bit of duplicated code from the drivers. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 1 + sound/soc/soc-dapm.c | 20 ++++++++++++++++++-- 2 files changed, 19 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 068785f..2bdf9a4 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -4402,6 +4402,7 @@ int snd_soc_register_codec(struct device *dev, codec->component.ignore_pmdown_time = codec_drv->ignore_pmdown_time; codec->dapm.codec = codec; codec->dapm.idle_bias_off = codec_drv->idle_bias_off; + codec->dapm.suspend_bias_off = codec_drv->suspend_bias_off; if (codec_drv->seq_notifier) codec->dapm.seq_notifier = codec_drv->seq_notifier; if (codec_drv->set_bias_level) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 8348352..a2025a6 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1683,6 +1683,22 @@ static void dapm_power_one_widget(struct snd_soc_dapm_widget *w, } } +static bool dapm_idle_bias_off(struct snd_soc_dapm_context *dapm) +{ + if (dapm->idle_bias_off) + return true; + + switch (snd_power_get_state(dapm->card->snd_card)) { + case SNDRV_CTL_POWER_D3hot: + case SNDRV_CTL_POWER_D3cold: + return dapm->suspend_bias_off; + default: + break; + } + + return false; +} + /* * Scan each dapm widget for complete audio path. * A complete path is a route that has valid endpoints i.e.:- @@ -1706,7 +1722,7 @@ static int dapm_power_widgets(struct snd_soc_card *card, int event) trace_snd_soc_dapm_start(card); list_for_each_entry(d, &card->dapm_list, list) { - if (d->idle_bias_off) + if (dapm_idle_bias_off(d)) d->target_bias_level = SND_SOC_BIAS_OFF; else d->target_bias_level = SND_SOC_BIAS_STANDBY; @@ -1772,7 +1788,7 @@ static int dapm_power_widgets(struct snd_soc_card *card, int event) if (d->target_bias_level > bias) bias = d->target_bias_level; list_for_each_entry(d, &card->dapm_list, list) - if (!d->idle_bias_off) + if (!dapm_idle_bias_off(d)) d->target_bias_level = bias; trace_snd_soc_dapm_walk_done(card); -- cgit v1.1 From a80932979a72ef9d4e66a69520c7588cc6de5699 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 4 Sep 2014 19:44:07 +0200 Subject: ASoC: Always run default suspend/resume code We do a bit more than just running the callbacks during suspend and resume these days (e.g. call regcache_mark_dirty() during suspend). But this is only when suspend and resume callbacks are specified for the driver, otherwise nothing is done. This means that drivers which don't want to do anything special during suspend and resume, but still want the standard operations to run, need to provide empty suspend and resume callback functions (rather than no callbacks). This patch updates the suspend and resume code to always run standard sequence regardless of whether suspend and resume handlers are provided. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 11 +++++++---- 1 file changed, 7 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 2bdf9a4..c612900 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -637,7 +637,7 @@ int snd_soc_suspend(struct device *dev) list_for_each_entry(codec, &card->codec_dev_list, card_list) { /* If there are paths active then the CODEC will be held with * bias _ON and should not be suspended. */ - if (!codec->suspended && codec->driver->suspend) { + if (!codec->suspended) { switch (codec->dapm.bias_level) { case SND_SOC_BIAS_STANDBY: /* @@ -651,8 +651,10 @@ int snd_soc_suspend(struct device *dev) "ASoC: idle_bias_off CODEC on over suspend\n"); break; } + case SND_SOC_BIAS_OFF: - codec->driver->suspend(codec); + if (codec->driver->suspend) + codec->driver->suspend(codec); codec->suspended = 1; codec->cache_sync = 1; if (codec->component.regmap) @@ -726,11 +728,12 @@ static void soc_resume_deferred(struct work_struct *work) * left with bias OFF or STANDBY and suspended so we must now * resume. Otherwise the suspend was suppressed. */ - if (codec->driver->resume && codec->suspended) { + if (codec->suspended) { switch (codec->dapm.bias_level) { case SND_SOC_BIAS_STANDBY: case SND_SOC_BIAS_OFF: - codec->driver->resume(codec); + if (codec->driver->resume) + codec->driver->resume(codec); codec->suspended = 0; break; default: -- cgit v1.1 From d7858bd647cda68bf832997a280a2f44aec01f1b Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 4 Sep 2014 19:44:08 +0200 Subject: ASoC: adau1373: Cleanup manual bias level transitions The ASoC core now takes care of setting the bias level to SND_SOC_BIAS_OFF when removing the CODEC, no need to do it manually anymore. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/adau1373.c | 7 ------- 1 file changed, 7 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/adau1373.c b/sound/soc/codecs/adau1373.c index 1947565..7c784ad 100644 --- a/sound/soc/codecs/adau1373.c +++ b/sound/soc/codecs/adau1373.c @@ -1448,12 +1448,6 @@ static int adau1373_set_bias_level(struct snd_soc_codec *codec, return 0; } -static int adau1373_remove(struct snd_soc_codec *codec) -{ - adau1373_set_bias_level(codec, SND_SOC_BIAS_OFF); - return 0; -} - static int adau1373_resume(struct snd_soc_codec *codec) { struct adau1373 *adau1373 = snd_soc_codec_get_drvdata(codec); @@ -1488,7 +1482,6 @@ static const struct regmap_config adau1373_regmap_config = { static struct snd_soc_codec_driver adau1373_codec_driver = { .probe = adau1373_probe, - .remove = adau1373_remove, .resume = adau1373_resume, .set_bias_level = adau1373_set_bias_level, .idle_bias_off = true, -- cgit v1.1 From 0e0f9b960a011a9e3815004f37cc475229170dfd Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 4 Sep 2014 19:44:09 +0200 Subject: ASoC: adau17x1: Cleanup manual bias level transitions Set the CODEC driver's suspend_bias_off flag rather than manually going to SND_SOC_BIAS_OFF in suspend and SND_SOC_BIAS_STANDBY in resume. This makes the code a bit shorter and cleaner. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/adau1761.c | 2 +- sound/soc/codecs/adau1781.c | 2 +- sound/soc/codecs/adau17x1.c | 8 -------- sound/soc/codecs/adau17x1.h | 1 - 4 files changed, 2 insertions(+), 11 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/adau1761.c b/sound/soc/codecs/adau1761.c index 848cab8..5518ebd 100644 --- a/sound/soc/codecs/adau1761.c +++ b/sound/soc/codecs/adau1761.c @@ -714,9 +714,9 @@ static int adau1761_codec_probe(struct snd_soc_codec *codec) static const struct snd_soc_codec_driver adau1761_codec_driver = { .probe = adau1761_codec_probe, - .suspend = adau17x1_suspend, .resume = adau17x1_resume, .set_bias_level = adau1761_set_bias_level, + .suspend_bias_off = true, .controls = adau1761_controls, .num_controls = ARRAY_SIZE(adau1761_controls), diff --git a/sound/soc/codecs/adau1781.c b/sound/soc/codecs/adau1781.c index 045a614..e9fc00f 100644 --- a/sound/soc/codecs/adau1781.c +++ b/sound/soc/codecs/adau1781.c @@ -446,9 +446,9 @@ static int adau1781_codec_probe(struct snd_soc_codec *codec) static const struct snd_soc_codec_driver adau1781_codec_driver = { .probe = adau1781_codec_probe, - .suspend = adau17x1_suspend, .resume = adau17x1_resume, .set_bias_level = adau1781_set_bias_level, + .suspend_bias_off = true, .controls = adau1781_controls, .num_controls = ARRAY_SIZE(adau1781_controls), diff --git a/sound/soc/codecs/adau17x1.c b/sound/soc/codecs/adau17x1.c index 0b65970..3e16c1c 100644 --- a/sound/soc/codecs/adau17x1.c +++ b/sound/soc/codecs/adau17x1.c @@ -815,13 +815,6 @@ int adau17x1_add_routes(struct snd_soc_codec *codec) } EXPORT_SYMBOL_GPL(adau17x1_add_routes); -int adau17x1_suspend(struct snd_soc_codec *codec) -{ - codec->driver->set_bias_level(codec, SND_SOC_BIAS_OFF); - return 0; -} -EXPORT_SYMBOL_GPL(adau17x1_suspend); - int adau17x1_resume(struct snd_soc_codec *codec) { struct adau *adau = snd_soc_codec_get_drvdata(codec); @@ -829,7 +822,6 @@ int adau17x1_resume(struct snd_soc_codec *codec) if (adau->switch_mode) adau->switch_mode(codec->dev); - codec->driver->set_bias_level(codec, SND_SOC_BIAS_STANDBY); regcache_sync(adau->regmap); return 0; diff --git a/sound/soc/codecs/adau17x1.h b/sound/soc/codecs/adau17x1.h index 3ffabaf..e4a557f 100644 --- a/sound/soc/codecs/adau17x1.h +++ b/sound/soc/codecs/adau17x1.h @@ -52,7 +52,6 @@ int adau17x1_set_micbias_voltage(struct snd_soc_codec *codec, enum adau17x1_micbias_voltage micbias); bool adau17x1_readable_register(struct device *dev, unsigned int reg); bool adau17x1_volatile_register(struct device *dev, unsigned int reg); -int adau17x1_suspend(struct snd_soc_codec *codec); int adau17x1_resume(struct snd_soc_codec *codec); extern const struct snd_soc_dai_ops adau17x1_dai_ops; -- cgit v1.1 From cd5d3a151118cd815be15970db099bcdb3f0ad12 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 4 Sep 2014 19:44:10 +0200 Subject: ASoC: adav80x: Cleanup manual bias level transitions Set the CODEC driver's suspend_bias_off flag rather than manually going to SND_SOC_BIAS_OFF in suspend and SND_SOC_BIAS_STANDBY in resume. This makes the code a bit shorter and cleaner. While we are at it also remove the regcache_cache_only() calls from suspend/resume as there shouldn't be any IO between suspend and resume. Since the ASoC core now takes care of setting the bias level to SND_SOC_BIAS_OFF when removing the CODEC there is no need to do it manually anymore either. The manual transition to SND_SOC_BIAS_STANDBY at the end of CODEC probe() can also be removed as the core will automatically do this after the CODEC has been probed. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/adav80x.c | 23 ++--------------------- 1 file changed, 2 insertions(+), 21 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/adav80x.c b/sound/soc/codecs/adav80x.c index c43b93f..ce3cdca 100644 --- a/sound/soc/codecs/adav80x.c +++ b/sound/soc/codecs/adav80x.c @@ -812,42 +812,23 @@ static int adav80x_probe(struct snd_soc_codec *codec) /* Disable DAC zero flag */ regmap_write(adav80x->regmap, ADAV80X_DAC_CTRL3, 0x6); - return adav80x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); -} - -static int adav80x_suspend(struct snd_soc_codec *codec) -{ - struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); - int ret; - - ret = adav80x_set_bias_level(codec, SND_SOC_BIAS_OFF); - regcache_cache_only(adav80x->regmap, true); - - return ret; + return 0; } static int adav80x_resume(struct snd_soc_codec *codec) { struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); - regcache_cache_only(adav80x->regmap, false); - adav80x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); regcache_sync(adav80x->regmap); return 0; } -static int adav80x_remove(struct snd_soc_codec *codec) -{ - return adav80x_set_bias_level(codec, SND_SOC_BIAS_OFF); -} - static struct snd_soc_codec_driver adav80x_codec_driver = { .probe = adav80x_probe, - .remove = adav80x_remove, - .suspend = adav80x_suspend, .resume = adav80x_resume, .set_bias_level = adav80x_set_bias_level, + .suspend_bias_off = true, .set_pll = adav80x_set_pll, .set_sysclk = adav80x_set_sysclk, -- cgit v1.1 From 0f0cc5a775ebe88d9be12489874bd2799b42e242 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 4 Sep 2014 19:44:11 +0200 Subject: ASoC: ssm2518: Cleanup manual bias level transitions Since the ASoC core now takes care of setting the bias level to SND_SOC_BIAS_OFF when removing the CODEC there is no need to do it manually anymore either. The manual transition to SND_SOC_BIAS_OFF at the end of CODEC probe() can also be removed as the CODEC is already in OFF state at this point. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/ssm2518.c | 13 ------------- 1 file changed, 13 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/ssm2518.c b/sound/soc/codecs/ssm2518.c index e8680be..67ea55a 100644 --- a/sound/soc/codecs/ssm2518.c +++ b/sound/soc/codecs/ssm2518.c @@ -646,17 +646,6 @@ static struct snd_soc_dai_driver ssm2518_dai = { .ops = &ssm2518_dai_ops, }; -static int ssm2518_probe(struct snd_soc_codec *codec) -{ - return ssm2518_set_bias_level(codec, SND_SOC_BIAS_OFF); -} - -static int ssm2518_remove(struct snd_soc_codec *codec) -{ - ssm2518_set_bias_level(codec, SND_SOC_BIAS_OFF); - return 0; -} - static int ssm2518_set_sysclk(struct snd_soc_codec *codec, int clk_id, int source, unsigned int freq, int dir) { @@ -727,8 +716,6 @@ static int ssm2518_set_sysclk(struct snd_soc_codec *codec, int clk_id, } static struct snd_soc_codec_driver ssm2518_codec_driver = { - .probe = ssm2518_probe, - .remove = ssm2518_remove, .set_bias_level = ssm2518_set_bias_level, .set_sysclk = ssm2518_set_sysclk, .idle_bias_off = true, -- cgit v1.1 From 85362efb80070bed890602483f71cd103be303c2 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 4 Sep 2014 19:44:12 +0200 Subject: ASoC: ssm2602: Cleanup manual bias level transitions Set the CODEC driver's suspend_bias_off flag rather than manually going to SND_SOC_BIAS_OFF in suspend and SND_SOC_BIAS_STANDBY in resume. This makes the code a bit shorter and cleaner. While we are at it also remove the regcache_cache_only() calls from suspend/resume as there shouldn't be any IO between suspend and resume. Since the ASoC core now takes care of setting the bias level to SND_SOC_BIAS_OFF when removing the CODEC there is no need to do it manually anymore either. The manual transition to SND_SOC_BIAS_STANDBY at the end of CODEC probe() can also be removed as the core will automatically do this after the CODEC has been probed. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/ssm2602.c | 24 ++---------------------- 1 file changed, 2 insertions(+), 22 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index 484b3bb..0dec136 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -502,18 +502,11 @@ static struct snd_soc_dai_driver ssm2602_dai = { .symmetric_samplebits = 1, }; -static int ssm2602_suspend(struct snd_soc_codec *codec) -{ - ssm2602_set_bias_level(codec, SND_SOC_BIAS_OFF); - return 0; -} - static int ssm2602_resume(struct snd_soc_codec *codec) { struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec); regcache_sync(ssm2602->regmap); - ssm2602_set_bias_level(codec, SND_SOC_BIAS_STANDBY); return 0; } @@ -586,27 +579,14 @@ static int ssm260x_codec_probe(struct snd_soc_codec *codec) break; } - if (ret) - return ret; - - ssm2602_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - - return 0; -} - -/* remove everything here */ -static int ssm2602_remove(struct snd_soc_codec *codec) -{ - ssm2602_set_bias_level(codec, SND_SOC_BIAS_OFF); - return 0; + return ret; } static struct snd_soc_codec_driver soc_codec_dev_ssm2602 = { .probe = ssm260x_codec_probe, - .remove = ssm2602_remove, - .suspend = ssm2602_suspend, .resume = ssm2602_resume, .set_bias_level = ssm2602_set_bias_level, + .suspend_bias_off = true, .controls = ssm260x_snd_controls, .num_controls = ARRAY_SIZE(ssm260x_snd_controls), -- cgit v1.1 From 9a302c32f363e420b6aa6e42c0cd686a495771f6 Mon Sep 17 00:00:00 2001 From: Rajeev Kumar Date: Fri, 5 Sep 2014 16:47:04 +0530 Subject: ASoC: dwc: Update email id of the author I moved from ST Microelectronics and the email-id no longer exists. Update email-id to personal one, Signed-off-by: Rajeev Kumar Signed-off-by: Mark Brown --- sound/soc/dwc/designware_i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c index 25c31f1..a97d27f 100644 --- a/sound/soc/dwc/designware_i2s.c +++ b/sound/soc/dwc/designware_i2s.c @@ -4,7 +4,7 @@ * sound/soc/dwc/designware_i2s.c * * Copyright (C) 2010 ST Microelectronics - * Rajeev Kumar + * Rajeev Kumar * * This file is licensed under the terms of the GNU General Public * License version 2. This program is licensed "as is" without any -- cgit v1.1 From 3d2c42d191a89ab35e3002309882e3b70fe12112 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sat, 6 Sep 2014 14:29:31 +0200 Subject: ASoC: 88pm860x-codec: Cleanup manual bias level transitions Since the ASoC core now takes care of setting the bias level to SND_SOC_BIAS_OFF when removing the CODEC there is no need to do it manually anymore either. The manual transition to SND_SOC_BIAS_STANDBY at the end of CODEC probe() can also be removed as the core will automatically do this after the CODEC has been probed. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/88pm860x-codec.c | 3 --- 1 file changed, 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/88pm860x-codec.c b/sound/soc/codecs/88pm860x-codec.c index 922006d..4c3b0af 100644 --- a/sound/soc/codecs/88pm860x-codec.c +++ b/sound/soc/codecs/88pm860x-codec.c @@ -1337,8 +1337,6 @@ static int pm860x_probe(struct snd_soc_codec *codec) } } - pm860x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - return 0; out: @@ -1354,7 +1352,6 @@ static int pm860x_remove(struct snd_soc_codec *codec) for (i = 3; i >= 0; i--) free_irq(pm860x->irq[i], pm860x); - pm860x_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; } -- cgit v1.1 From 2a93f70925a56ae1629be8b46c3c6d502f98dded Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sat, 6 Sep 2014 14:29:33 +0200 Subject: ASoC: jz4740: Cleanup manual bias level transitions Set the CODEC driver's suspend_bias_off flag rather than manually going to SND_SOC_BIAS_OFF in suspend and SND_SOC_BIAS_STANDBY in resume. This makes the code a bit shorter and cleaner. Since the ASoC core now takes care of setting the bias level to SND_SOC_BIAS_OFF when removing the CODEC there is no need to do it manually anymore either. The manual transition to SND_SOC_BIAS_STANDBY at the end of CODEC probe() can also be removed as the core will automatically do this after the CODEC has been probed. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/jz4740.c | 30 +----------------------------- 1 file changed, 1 insertion(+), 29 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/jz4740.c b/sound/soc/codecs/jz4740.c index bcebd1a..df7c01c 100644 --- a/sound/soc/codecs/jz4740.c +++ b/sound/soc/codecs/jz4740.c @@ -293,41 +293,13 @@ static int jz4740_codec_dev_probe(struct snd_soc_codec *codec) regmap_update_bits(jz4740_codec->regmap, JZ4740_REG_CODEC_1, JZ4740_CODEC_1_SW2_ENABLE, JZ4740_CODEC_1_SW2_ENABLE); - jz4740_codec_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - return 0; } -static int jz4740_codec_dev_remove(struct snd_soc_codec *codec) -{ - jz4740_codec_set_bias_level(codec, SND_SOC_BIAS_OFF); - - return 0; -} - -#ifdef CONFIG_PM_SLEEP - -static int jz4740_codec_suspend(struct snd_soc_codec *codec) -{ - return jz4740_codec_set_bias_level(codec, SND_SOC_BIAS_OFF); -} - -static int jz4740_codec_resume(struct snd_soc_codec *codec) -{ - return jz4740_codec_set_bias_level(codec, SND_SOC_BIAS_STANDBY); -} - -#else -#define jz4740_codec_suspend NULL -#define jz4740_codec_resume NULL -#endif - static struct snd_soc_codec_driver soc_codec_dev_jz4740_codec = { .probe = jz4740_codec_dev_probe, - .remove = jz4740_codec_dev_remove, - .suspend = jz4740_codec_suspend, - .resume = jz4740_codec_resume, .set_bias_level = jz4740_codec_set_bias_level, + .suspend_bias_off = true, .controls = jz4740_codec_controls, .num_controls = ARRAY_SIZE(jz4740_codec_controls), -- cgit v1.1 From 35199a7c11d5f6a87a5b35dfd69fde3f65d37fac Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sat, 6 Sep 2014 14:29:34 +0200 Subject: ASoC: ml26124: Cleanup manual bias level transitions Set the CODEC driver's suspend_bias_off flag rather than manually going to SND_SOC_BIAS_OFF in suspend and SND_SOC_BIAS_STANDBY in resume. This makes the code a bit shorter and cleaner. The manual transition to SND_SOC_BIAS_STANDBY at the end of CODEC probe() can also be removed as the core will automatically do this after the CODEC has been probed. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/ml26124.c | 24 +----------------------- 1 file changed, 1 insertion(+), 23 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/ml26124.c b/sound/soc/codecs/ml26124.c index e661e84..711f550 100644 --- a/sound/soc/codecs/ml26124.c +++ b/sound/soc/codecs/ml26124.c @@ -565,41 +565,19 @@ static struct snd_soc_dai_driver ml26124_dai = { .symmetric_rates = 1, }; -#ifdef CONFIG_PM -static int ml26124_suspend(struct snd_soc_codec *codec) -{ - ml26124_set_bias_level(codec, SND_SOC_BIAS_OFF); - - return 0; -} - -static int ml26124_resume(struct snd_soc_codec *codec) -{ - ml26124_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - - return 0; -} -#else -#define ml26124_suspend NULL -#define ml26124_resume NULL -#endif - static int ml26124_probe(struct snd_soc_codec *codec) { /* Software Reset */ snd_soc_update_bits(codec, ML26124_SW_RST, 0x01, 1); snd_soc_update_bits(codec, ML26124_SW_RST, 0x01, 0); - ml26124_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - return 0; } static struct snd_soc_codec_driver soc_codec_dev_ml26124 = { .probe = ml26124_probe, - .suspend = ml26124_suspend, - .resume = ml26124_resume, .set_bias_level = ml26124_set_bias_level, + .suspend_bias_off = true, .dapm_widgets = ml26124_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(ml26124_dapm_widgets), .dapm_routes = ml26124_intercon, -- cgit v1.1 From e649057a41c24b4122e976746649e471709d4b16 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sat, 6 Sep 2014 14:29:35 +0200 Subject: ASoC: sgtl5000: Cleanup bias level transitions Set the CODEC driver's suspend_bias_off flag rather than manually going to SND_SOC_BIAS_OFF in suspend and SND_SOC_BIAS_STANDBY in resume. This makes the code a bit shorter and cleaner. Since the ASoC core now takes care of setting the bias level to SND_SOC_BIAS_OFF when removing the CODEC there is no need to do it manually anymore either. The manual transition to SND_SOC_BIAS_STANDBY at the end of CODEC probe() can also be removed as the core will automatically do this after the CODEC has been probed. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/sgtl5000.c | 30 +----------------------------- 1 file changed, 1 insertion(+), 29 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index e997d27..a604a22 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -1073,26 +1073,6 @@ static bool sgtl5000_readable(struct device *dev, unsigned int reg) } } -#ifdef CONFIG_SUSPEND -static int sgtl5000_suspend(struct snd_soc_codec *codec) -{ - sgtl5000_set_bias_level(codec, SND_SOC_BIAS_OFF); - - return 0; -} - -static int sgtl5000_resume(struct snd_soc_codec *codec) -{ - /* Bring the codec back up to standby to enable regulators */ - sgtl5000_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - - return 0; -} -#else -#define sgtl5000_suspend NULL -#define sgtl5000_resume NULL -#endif /* CONFIG_SUSPEND */ - /* * sgtl5000 has 3 internal power supplies: * 1. VAG, normally set to vdda/2 @@ -1352,11 +1332,6 @@ static int sgtl5000_probe(struct snd_soc_codec *codec) */ snd_soc_write(codec, SGTL5000_DAP_CTRL, 0); - /* leading to standby state */ - ret = sgtl5000_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - if (ret) - goto err; - return 0; err: @@ -1373,8 +1348,6 @@ static int sgtl5000_remove(struct snd_soc_codec *codec) { struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); - sgtl5000_set_bias_level(codec, SND_SOC_BIAS_OFF); - regulator_bulk_disable(ARRAY_SIZE(sgtl5000->supplies), sgtl5000->supplies); regulator_bulk_free(ARRAY_SIZE(sgtl5000->supplies), @@ -1387,9 +1360,8 @@ static int sgtl5000_remove(struct snd_soc_codec *codec) static struct snd_soc_codec_driver sgtl5000_driver = { .probe = sgtl5000_probe, .remove = sgtl5000_remove, - .suspend = sgtl5000_suspend, - .resume = sgtl5000_resume, .set_bias_level = sgtl5000_set_bias_level, + .suspend_bias_off = true, .controls = sgtl5000_snd_controls, .num_controls = ARRAY_SIZE(sgtl5000_snd_controls), .dapm_widgets = sgtl5000_dapm_widgets, -- cgit v1.1 From 8d01370f59856a0ac5b222878667d52477b589f0 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sat, 6 Sep 2014 14:29:32 +0200 Subject: ASoC: es8328: Cleanup manual bias level transitions Set the CODEC driver's suspend_bias_off flag rather than manually going to SND_SOC_BIAS_OFF in suspend and SND_SOC_BIAS_STANDBY in resume. This makes the code a bit shorter and cleaner. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/es8328.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/es8328.c b/sound/soc/codecs/es8328.c index 3ff7870..f273251 100644 --- a/sound/soc/codecs/es8328.c +++ b/sound/soc/codecs/es8328.c @@ -602,8 +602,6 @@ static int es8328_suspend(struct snd_soc_codec *codec) es8328 = snd_soc_codec_get_drvdata(codec); - es8328_set_bias_level(codec, SND_SOC_BIAS_OFF); - clk_disable_unprepare(es8328->clk); ret = regulator_bulk_disable(ARRAY_SIZE(es8328->supplies), @@ -643,7 +641,6 @@ static int es8328_resume(struct snd_soc_codec *codec) return ret; } - es8328_set_bias_level(codec, SND_SOC_BIAS_STANDBY); return 0; } @@ -712,6 +709,8 @@ static struct snd_soc_codec_driver es8328_codec_driver = { .resume = es8328_resume, .remove = es8328_remove, .set_bias_level = es8328_set_bias_level, + .suspend_bias_off = true, + .controls = es8328_snd_controls, .num_controls = ARRAY_SIZE(es8328_snd_controls), .dapm_widgets = es8328_dapm_widgets, -- cgit v1.1 From bd033808e2b160bab61cfe18b0ecb4ccc7809516 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 20 Aug 2014 13:08:47 +0200 Subject: ASoC: sst-haswell-pcm: Alloc state struct in driver probe() Resource allocations should happen in driver probe callback rather than in snd_soc_platform probe functions. Especially if the resource is device managed. The snd_soc_* probe/remove functions are mainly intended to be used for things that require the component to be already bound to a card. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/intel/sst-haswell-pcm.c | 24 +++++++++++++----------- 1 file changed, 13 insertions(+), 11 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/sst-haswell-pcm.c b/sound/soc/intel/sst-haswell-pcm.c index 61bf6da..1de0958 100644 --- a/sound/soc/intel/sst-haswell-pcm.c +++ b/sound/soc/intel/sst-haswell-pcm.c @@ -778,20 +778,11 @@ static const struct snd_soc_dapm_route graph[] = { static int hsw_pcm_probe(struct snd_soc_platform *platform) { + struct hsw_priv_data *priv_data = snd_soc_platform_get_drvdata(platform); struct sst_pdata *pdata = dev_get_platdata(platform->dev); - struct hsw_priv_data *priv_data; - struct device *dma_dev; + struct device *dma_dev = pdata->dma_dev; int i, ret = 0; - if (!pdata) - return -ENODEV; - - dma_dev = pdata->dma_dev; - - priv_data = devm_kzalloc(platform->dev, sizeof(*priv_data), GFP_KERNEL); - priv_data->hsw = pdata->dsp; - snd_soc_platform_set_drvdata(platform, priv_data); - /* allocate DSP buffer page tables */ for (i = 0; i < ARRAY_SIZE(hsw_dais); i++) { @@ -863,12 +854,23 @@ static const struct snd_soc_component_driver hsw_dai_component = { static int hsw_pcm_dev_probe(struct platform_device *pdev) { struct sst_pdata *sst_pdata = dev_get_platdata(&pdev->dev); + struct hsw_priv_data *priv_data; int ret; + if (!sst_pdata) + return -EINVAL; + + priv_data = devm_kzalloc(&pdev->dev, sizeof(*priv_data), GFP_KERNEL); + if (!priv_data) + return -ENOMEM; + ret = sst_hsw_dsp_init(&pdev->dev, sst_pdata); if (ret < 0) return -ENODEV; + priv_data->hsw = sst_pdata->dsp; + platform_set_drvdata(pdev, priv_data); + ret = snd_soc_register_platform(&pdev->dev, &hsw_soc_platform); if (ret < 0) goto err_plat; -- cgit v1.1 From 923976a30b36ce0970e88f53ed2f2b5b61aeeb73 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 20 Aug 2014 13:08:48 +0200 Subject: ASoC: sst-haswell-pcm: Move controls and DAPM elements to component The sst-haswell-pcm driver registers both a snd_soc_component and a snd_soc_platform and expects that the DAPM widgets for the DAIs registered by component are added to the DAPM context of the platform. This requires us to have a hack in the ASoC core which does so. Moving the DAPM elements over to the component allows us to remove this hack. While we are at it also move the controls over to the component. The controls don't need the platform for anything other than snd_soc_platform_get_drvdata(), this can easily be replaced by snd_soc_component_get_drvdata(). As the long term goal is to register only a single component this is a step in the right direction. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/intel/sst-haswell-pcm.c | 32 +++++++++++++++----------------- 1 file changed, 15 insertions(+), 17 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/sst-haswell-pcm.c b/sound/soc/intel/sst-haswell-pcm.c index 1de0958..33fc5c3 100644 --- a/sound/soc/intel/sst-haswell-pcm.c +++ b/sound/soc/intel/sst-haswell-pcm.c @@ -138,11 +138,10 @@ static inline unsigned int hsw_ipc_to_mixer(u32 value) static int hsw_stream_volume_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_platform *platform = snd_soc_kcontrol_platform(kcontrol); + struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol); + struct hsw_priv_data *pdata = snd_soc_component_get_drvdata(cmpnt); struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; - struct hsw_priv_data *pdata = - snd_soc_platform_get_drvdata(platform); struct hsw_pcm_data *pcm_data = &pdata->pcm[mc->reg]; struct sst_hsw *hsw = pdata->hsw; u32 volume; @@ -176,11 +175,10 @@ static int hsw_stream_volume_put(struct snd_kcontrol *kcontrol, static int hsw_stream_volume_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_platform *platform = snd_soc_kcontrol_platform(kcontrol); + struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol); + struct hsw_priv_data *pdata = snd_soc_component_get_drvdata(cmpnt); struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; - struct hsw_priv_data *pdata = - snd_soc_platform_get_drvdata(platform); struct hsw_pcm_data *pcm_data = &pdata->pcm[mc->reg]; struct sst_hsw *hsw = pdata->hsw; u32 volume; @@ -208,8 +206,8 @@ static int hsw_stream_volume_get(struct snd_kcontrol *kcontrol, static int hsw_volume_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_platform *platform = snd_soc_kcontrol_platform(kcontrol); - struct hsw_priv_data *pdata = snd_soc_platform_get_drvdata(platform); + struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol); + struct hsw_priv_data *pdata = snd_soc_component_get_drvdata(cmpnt); struct sst_hsw *hsw = pdata->hsw; u32 volume; @@ -233,8 +231,8 @@ static int hsw_volume_put(struct snd_kcontrol *kcontrol, static int hsw_volume_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_platform *platform = snd_soc_kcontrol_platform(kcontrol); - struct hsw_priv_data *pdata = snd_soc_platform_get_drvdata(platform); + struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol); + struct hsw_priv_data *pdata = snd_soc_component_get_drvdata(cmpnt); struct sst_hsw *hsw = pdata->hsw; unsigned int volume = 0; @@ -839,16 +837,16 @@ static struct snd_soc_platform_driver hsw_soc_platform = { .ops = &hsw_pcm_ops, .pcm_new = hsw_pcm_new, .pcm_free = hsw_pcm_free, - .controls = hsw_volume_controls, - .num_controls = ARRAY_SIZE(hsw_volume_controls), - .dapm_widgets = widgets, - .num_dapm_widgets = ARRAY_SIZE(widgets), - .dapm_routes = graph, - .num_dapm_routes = ARRAY_SIZE(graph), }; static const struct snd_soc_component_driver hsw_dai_component = { - .name = "haswell-dai", + .name = "haswell-dai", + .controls = hsw_volume_controls, + .num_controls = ARRAY_SIZE(hsw_volume_controls), + .dapm_widgets = widgets, + .num_dapm_widgets = ARRAY_SIZE(widgets), + .dapm_routes = graph, + .num_dapm_routes = ARRAY_SIZE(graph), }; static int hsw_pcm_dev_probe(struct platform_device *pdev) -- cgit v1.1 From 0634814fe0f29a46c44386a03f259f99c983bf7e Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 20 Aug 2014 13:08:49 +0200 Subject: ASoC: Remove table based DAPM/control setup support from snd_soc_platform_driver There are no users left and new users should rather use the component_driver struct embedded in the snd_soc_platform_driver struct to do this. E.g.: static const struct snd_soc_platform_driver foobar_driver = { .component_driver = { .dapm_widgets = ..., .num_dapm_widgets = ..., ..., }, ... }; instead of static const struct snd_soc_platform_driver foobar_driver = { .dapm_widgets = ..., .num_dapm_widgets = ..., ... }; This also allows us to remove the steal_sibling_dai_widgets hack. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 58 ++++++---------------------------------------------- 1 file changed, 6 insertions(+), 52 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 1b422c5..052f59c 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1087,7 +1087,6 @@ static int soc_probe_component(struct snd_soc_card *card, struct snd_soc_component *component) { struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); - struct snd_soc_component *dai_component, *component2; struct snd_soc_dai *dai; int ret; @@ -1114,44 +1113,12 @@ static int soc_probe_component(struct snd_soc_card *card, } } - /* - * This is rather ugly, but certain platforms expect that the DAPM - * widgets for the DAIs for components with the same parent device are - * created in the platforms DAPM context. Until that is fixed we need to - * keep this. - */ - if (component->steal_sibling_dai_widgets) { - dai_component = NULL; - list_for_each_entry(component2, &component_list, list) { - if (component == component2) - continue; - - if (component2->dev == component->dev && - !list_empty(&component2->dai_list)) { - dai_component = component2; - break; - } - } - } else { - dai_component = component; - list_for_each_entry(component2, &component_list, list) { - if (component2->dev == component->dev && - component2->steal_sibling_dai_widgets) { - dai_component = NULL; - break; - } - } - } - - if (dai_component) { - list_for_each_entry(dai, &dai_component->dai_list, list) { - snd_soc_dapm_new_dai_widgets(dapm, dai); - if (ret != 0) { - dev_err(component->dev, - "Failed to create DAI widgets %d\n", - ret); - goto err_probe; - } + list_for_each_entry(dai, &component->dai_list, list) { + ret = snd_soc_dapm_new_dai_widgets(dapm, dai); + if (ret != 0) { + dev_err(component->dev, + "Failed to create DAI widgets %d\n", ret); + goto err_probe; } } @@ -4164,19 +4131,6 @@ int snd_soc_add_platform(struct device *dev, struct snd_soc_platform *platform, platform->dev = dev; platform->driver = platform_drv; - if (platform_drv->controls) { - platform->component.controls = platform_drv->controls; - platform->component.num_controls = platform_drv->num_controls; - } - if (platform_drv->dapm_widgets) { - platform->component.dapm_widgets = platform_drv->dapm_widgets; - platform->component.num_dapm_widgets = platform_drv->num_dapm_widgets; - platform->component.steal_sibling_dai_widgets = true; - } - if (platform_drv->dapm_routes) { - platform->component.dapm_routes = platform_drv->dapm_routes; - platform->component.num_dapm_routes = platform_drv->num_dapm_routes; - } if (platform_drv->probe) platform->component.probe = snd_soc_platform_drv_probe; -- cgit v1.1 From 02024756e6ab3a3fcdc3b203552b16b345ebd97d Mon Sep 17 00:00:00 2001 From: "Subhransu S. Prusty" Date: Tue, 2 Sep 2014 18:05:56 +0530 Subject: ASoC: mfld: pcm: Replace pr_ with dev_ Signed-off-by: Subhransu S. Prusty Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/sst-mfld-platform-pcm.c | 44 +++++++++++++++++---------------- 1 file changed, 23 insertions(+), 21 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/sst-mfld-platform-pcm.c b/sound/soc/intel/sst-mfld-platform-pcm.c index 8e1e9bc..85deecd 100644 --- a/sound/soc/intel/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/sst-mfld-platform-pcm.c @@ -43,12 +43,12 @@ int sst_register_dsp(struct sst_device *dev) return -ENODEV; mutex_lock(&sst_lock); if (sst) { - pr_err("we already have a device %s\n", sst->name); + dev_err(dev->dev, "we already have a device %s\n", sst->name); module_put(dev->dev->driver->owner); mutex_unlock(&sst_lock); return -EEXIST; } - pr_debug("registering device %s\n", dev->name); + dev_dbg(dev->dev, "registering device %s\n", dev->name); sst = dev; mutex_unlock(&sst_lock); return 0; @@ -70,7 +70,7 @@ int sst_unregister_dsp(struct sst_device *dev) } module_put(sst->dev->driver->owner); - pr_debug("unreg %s\n", sst->name); + dev_dbg(dev->dev, "unreg %s\n", sst->name); sst = NULL; mutex_unlock(&sst_lock); return 0; @@ -306,9 +306,10 @@ static int sst_platform_init_stream(struct snd_pcm_substream *substream) { struct sst_runtime_stream *stream = substream->runtime->private_data; + struct snd_soc_pcm_runtime *rtd = substream->private_data; int ret_val; - pr_debug("setting buffer ptr param\n"); + dev_dbg(rtd->dev, "setting buffer ptr param\n"); sst_set_stream_status(stream, SST_PLATFORM_INIT); stream->stream_info.period_elapsed = sst_period_elapsed; stream->stream_info.arg = substream; @@ -316,7 +317,7 @@ static int sst_platform_init_stream(struct snd_pcm_substream *substream) stream->stream_info.sfreq = substream->runtime->rate; ret_val = stream->ops->stream_init(sst->dev, &stream->stream_info); if (ret_val) - pr_err("control_set ret error %d\n", ret_val); + dev_err(rtd->dev, "control_set ret error %d\n", ret_val); return ret_val; } @@ -337,7 +338,7 @@ static int sst_media_open(struct snd_pcm_substream *substream, mutex_lock(&sst_lock); if (!sst || !try_module_get(sst->dev->driver->owner)) { - pr_err("no device available to run\n"); + dev_err(dai->dev, "no device available to run\n"); ret_val = -ENODEV; goto out_ops; } @@ -385,10 +386,11 @@ static inline unsigned int get_current_pipe_id(struct snd_soc_platform *platform substream->runtime->private_data; u32 str_id = stream->stream_info.str_id; unsigned int pipe_id; + pipe_id = map[str_id].device_id; - pr_debug("%s: got pipe_id = %#x for str_id = %d\n", - __func__, pipe_id, str_id); + dev_dbg(platform->dev, "got pipe_id = %#x for str_id = %d\n", + pipe_id, str_id); return pipe_id; } @@ -459,29 +461,30 @@ static int sst_platform_pcm_trigger(struct snd_pcm_substream *substream, int ret_val = 0, str_id; struct sst_runtime_stream *stream; int status; + struct snd_soc_pcm_runtime *rtd = substream->private_data; - pr_debug("sst_platform_pcm_trigger called\n"); + dev_dbg(rtd->dev, "sst_platform_pcm_trigger called\n"); stream = substream->runtime->private_data; str_id = stream->stream_info.str_id; switch (cmd) { case SNDRV_PCM_TRIGGER_START: - pr_debug("sst: Trigger Start\n"); + dev_dbg(rtd->dev, "sst: Trigger Start\n"); status = SST_PLATFORM_RUNNING; stream->stream_info.arg = substream; ret_val = stream->ops->stream_start(sst->dev, str_id); break; case SNDRV_PCM_TRIGGER_STOP: - pr_debug("sst: in stop\n"); + dev_dbg(rtd->dev, "sst: in stop\n"); status = SST_PLATFORM_DROPPED; ret_val = stream->ops->stream_drop(sst->dev, str_id); break; case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - pr_debug("sst: in pause\n"); + dev_dbg(rtd->dev, "sst: in pause\n"); status = SST_PLATFORM_PAUSED; ret_val = stream->ops->stream_pause(sst->dev, str_id); break; case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - pr_debug("sst: in pause release\n"); + dev_dbg(rtd->dev, "sst: in pause release\n"); status = SST_PLATFORM_RUNNING; ret_val = stream->ops->stream_pause_release(sst->dev, str_id); break; @@ -502,6 +505,7 @@ static snd_pcm_uframes_t sst_platform_pcm_pointer struct sst_runtime_stream *stream; int ret_val, status; struct pcm_stream_info *str_info; + struct snd_soc_pcm_runtime *rtd = substream->private_data; stream = substream->runtime->private_data; status = sst_get_stream_status(stream); @@ -510,7 +514,7 @@ static snd_pcm_uframes_t sst_platform_pcm_pointer str_info = &stream->stream_info; ret_val = stream->ops->stream_read_tstamp(sst->dev, str_info); if (ret_val) { - pr_err("sst: error code = %d\n", ret_val); + dev_err(rtd->dev, "sst: error code = %d\n", ret_val); return ret_val; } substream->runtime->delay = str_info->pcm_delay; @@ -526,7 +530,7 @@ static struct snd_pcm_ops sst_platform_ops = { static void sst_pcm_free(struct snd_pcm *pcm) { - pr_debug("sst_pcm_free called\n"); + dev_dbg(pcm->dev, "sst_pcm_free called\n"); snd_pcm_lib_preallocate_free_for_all(pcm); } @@ -543,7 +547,7 @@ static int sst_pcm_new(struct snd_soc_pcm_runtime *rtd) snd_dma_continuous_data(GFP_DMA), SST_MIN_BUFFER, SST_MAX_BUFFER); if (retval) { - pr_err("dma buffer allocationf fail\n"); + dev_err(rtd->dev, "dma buffer allocationf fail\n"); return retval; } } @@ -576,13 +580,11 @@ static int sst_platform_probe(struct platform_device *pdev) drv = devm_kzalloc(&pdev->dev, sizeof(*drv), GFP_KERNEL); if (drv == NULL) { - pr_err("kzalloc failed\n"); return -ENOMEM; } pdata = devm_kzalloc(&pdev->dev, sizeof(*pdata), GFP_KERNEL); if (pdata == NULL) { - pr_err("kzalloc failed for pdata\n"); return -ENOMEM; } @@ -594,14 +596,14 @@ static int sst_platform_probe(struct platform_device *pdev) ret = snd_soc_register_platform(&pdev->dev, &sst_soc_platform_drv); if (ret) { - pr_err("registering soc platform failed\n"); + dev_err(&pdev->dev, "registering soc platform failed\n"); return ret; } ret = snd_soc_register_component(&pdev->dev, &sst_component, sst_platform_dai, ARRAY_SIZE(sst_platform_dai)); if (ret) { - pr_err("registering cpu dais failed\n"); + dev_err(&pdev->dev, "registering cpu dais failed\n"); snd_soc_unregister_platform(&pdev->dev); } return ret; @@ -612,7 +614,7 @@ static int sst_platform_remove(struct platform_device *pdev) snd_soc_unregister_component(&pdev->dev); snd_soc_unregister_platform(&pdev->dev); - pr_debug("sst_platform_remove success\n"); + dev_dbg(&pdev->dev, "sst_platform_remove success\n"); return 0; } -- cgit v1.1 From b794dbcd31c8b19118b7d02a39453576aa8e78fb Mon Sep 17 00:00:00 2001 From: Rajeev Kumar Date: Tue, 9 Sep 2014 12:27:19 +0530 Subject: ASoC: Update email id of the author I moved from ST Microelectronics and so updating email-id to personal one. Signed-off-by: Rajeev Kumar Signed-off-by: Mark Brown --- sound/soc/codecs/sta529.c | 4 ++-- sound/soc/dwc/designware_i2s.c | 2 +- sound/soc/spear/spear_pcm.c | 4 ++-- 3 files changed, 5 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/sta529.c b/sound/soc/codecs/sta529.c index 9aa1323..89c748d 100644 --- a/sound/soc/codecs/sta529.c +++ b/sound/soc/codecs/sta529.c @@ -4,7 +4,7 @@ * sound/soc/codecs/sta529.c -- spear ALSA Soc codec driver * * Copyright (C) 2012 ST Microelectronics - * Rajeev Kumar + * Rajeev Kumar * * This file is licensed under the terms of the GNU General Public * License version 2. This program is licensed "as is" without any @@ -426,5 +426,5 @@ static struct i2c_driver sta529_i2c_driver = { module_i2c_driver(sta529_i2c_driver); MODULE_DESCRIPTION("ASoC STA529 codec driver"); -MODULE_AUTHOR("Rajeev Kumar "); +MODULE_AUTHOR("Rajeev Kumar "); MODULE_LICENSE("GPL"); diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c index a97d27f..e961388 100644 --- a/sound/soc/dwc/designware_i2s.c +++ b/sound/soc/dwc/designware_i2s.c @@ -455,7 +455,7 @@ static struct platform_driver dw_i2s_driver = { module_platform_driver(dw_i2s_driver); -MODULE_AUTHOR("Rajeev Kumar "); +MODULE_AUTHOR("Rajeev Kumar "); MODULE_DESCRIPTION("DESIGNWARE I2S SoC Interface"); MODULE_LICENSE("GPL"); MODULE_ALIAS("platform:designware_i2s"); diff --git a/sound/soc/spear/spear_pcm.c b/sound/soc/spear/spear_pcm.c index 0e5a8f3..a7dc3c5 100644 --- a/sound/soc/spear/spear_pcm.c +++ b/sound/soc/spear/spear_pcm.c @@ -4,7 +4,7 @@ * sound/soc/spear/spear_pcm.c * * Copyright (C) 2012 ST Microelectronics - * Rajeev Kumar + * Rajeev Kumar * * This file is licensed under the terms of the GNU General Public * License version 2. This program is licensed "as is" without any @@ -50,6 +50,6 @@ int devm_spear_pcm_platform_register(struct device *dev, } EXPORT_SYMBOL_GPL(devm_spear_pcm_platform_register); -MODULE_AUTHOR("Rajeev Kumar "); +MODULE_AUTHOR("Rajeev Kumar "); MODULE_DESCRIPTION("SPEAr PCM DMA module"); MODULE_LICENSE("GPL"); -- cgit v1.1 From 2080437d375f4d8ba2fe37254199427f3f5e7bc2 Mon Sep 17 00:00:00 2001 From: Xiubo Li Date: Wed, 3 Sep 2014 10:23:39 +0800 Subject: ASoC: simple-card: Merge single and muti DAI link(s) code. This patch will split the DT node into old style and new style: The new style will merge the single DAI link and muti DAI links code together, the new style will be easier to add muti DAI links from old single DAI link DTs. This patch will maintian compatibility with the old DTs. Signed-off-by: Xiubo Li Signed-off-by: Mark Brown --- sound/soc/generic/simple-card.c | 22 ++++++++++++---------- 1 file changed, 12 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index b63860d..e0abe77 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -185,6 +185,7 @@ static int asoc_simple_card_dai_link_of(struct device_node *node, char *prefix = ""; int ret, cpu_args; + /* For single DAI link & old style of DT node */ if (is_top_level_node) prefix = "simple-audio-card,"; @@ -318,14 +319,16 @@ dai_link_of_err: static int asoc_simple_card_parse_of(struct device_node *node, struct simple_card_data *priv, - struct device *dev, - int multi) + struct device *dev) { struct snd_soc_dai_link *dai_link = priv->snd_card.dai_link; struct simple_dai_props *dai_props = priv->dai_props; u32 val; int ret; + if (!node) + return -EINVAL; + /* parsing the card name from DT */ snd_soc_of_parse_card_name(&priv->snd_card, "simple-audio-card,name"); @@ -353,7 +356,8 @@ static int asoc_simple_card_parse_of(struct device_node *node, dev_dbg(dev, "New simple-card: %s\n", priv->snd_card.name ? priv->snd_card.name : ""); - if (multi) { + /* Single/Muti DAI link(s) & New style of DT node */ + if (of_get_child_by_name(node, "simple-audio-card,dai-link")) { struct device_node *np = NULL; int i = 0; @@ -370,6 +374,7 @@ static int asoc_simple_card_parse_of(struct device_node *node, i++; } } else { + /* For single DAI link & old style of DT node */ ret = asoc_simple_card_dai_link_of(node, dev, dai_link, dai_props, true); if (ret < 0) @@ -409,16 +414,13 @@ static int asoc_simple_card_probe(struct platform_device *pdev) struct snd_soc_dai_link *dai_link; struct device_node *np = pdev->dev.of_node; struct device *dev = &pdev->dev; - int num_links, multi, ret; + int num_links, ret; /* get the number of DAI links */ - if (np && of_get_child_by_name(np, "simple-audio-card,dai-link")) { + if (np && of_get_child_by_name(np, "simple-audio-card,dai-link")) num_links = of_get_child_count(np); - multi = 1; - } else { + else num_links = 1; - multi = 0; - } /* allocate the private data and the DAI link array */ priv = devm_kzalloc(dev, @@ -445,7 +447,7 @@ static int asoc_simple_card_probe(struct platform_device *pdev) if (np && of_device_is_available(np)) { - ret = asoc_simple_card_parse_of(np, priv, dev, multi); + ret = asoc_simple_card_parse_of(np, priv, dev); if (ret < 0) { if (ret != -EPROBE_DEFER) dev_err(dev, "parse error %d\n", ret); -- cgit v1.1 From 133c2681c4a0c1b589d138c2fdd0f131bdce20ed Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Tue, 9 Sep 2014 16:51:49 +0100 Subject: ASoC: samsung-i2s: Check secondary DAI exists before referencing In a couple of places the driver is missing a check to ensure there is a secondary DAI before it de-references the pointer to it, causing a null pointer de-reference. This patch adds a check to avoid this. Signed-off-by: Charles Keepax Acked-by: Sylwester Nawrocki Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/samsung/i2s.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 03eec22..9d51347 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -462,7 +462,7 @@ static int i2s_set_sysclk(struct snd_soc_dai *dai, if (dir == SND_SOC_CLOCK_IN) rfs = 0; - if ((rfs && other->rfs && (other->rfs != rfs)) || + if ((rfs && other && other->rfs && (other->rfs != rfs)) || (any_active(i2s) && (((dir == SND_SOC_CLOCK_IN) && !(mod & MOD_CDCLKCON)) || @@ -762,7 +762,8 @@ static void i2s_shutdown(struct snd_pcm_substream *substream, } else { u32 mod = readl(i2s->addr + I2SMOD); i2s->cdclk_out = !(mod & MOD_CDCLKCON); - other->cdclk_out = i2s->cdclk_out; + if (other) + other->cdclk_out = i2s->cdclk_out; } /* Reset any constraint on RFS and BFS */ i2s->rfs = 0; -- cgit v1.1 From 417c60e8f248a84e8e768c55d191689d1e27e05f Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 9 Sep 2014 20:42:40 +0200 Subject: ASoC: cs42l52: Cleanup manual bias level transitions Set the CODEC driver's suspend_bias_off flag rather than manually going to SND_SOC_BIAS_OFF in suspend and SND_SOC_BIAS_STANDBY in resume. This makes the code a bit shorter and cleaner. Since the ASoC core now takes care of setting the bias level to SND_SOC_BIAS_OFF when removing the CODEC there is no need to do it manually anymore either. The manual transition to SND_SOC_BIAS_STANDBY at the end of CODEC probe() can also be removed as the core will automatically do this after the CODEC has been probed. Signed-off-by: Lars-Peter Clausen Acked-by: Brian Austin Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l52.c | 20 +------------------- 1 file changed, 1 insertion(+), 19 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 969167d..6efff71 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -946,20 +946,6 @@ static struct snd_soc_dai_driver cs42l52_dai = { .ops = &cs42l52_ops, }; -static int cs42l52_suspend(struct snd_soc_codec *codec) -{ - cs42l52_set_bias_level(codec, SND_SOC_BIAS_OFF); - - return 0; -} - -static int cs42l52_resume(struct snd_soc_codec *codec) -{ - cs42l52_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - - return 0; -} - static int beep_rates[] = { 261, 522, 585, 667, 706, 774, 889, 1000, 1043, 1200, 1333, 1412, 1600, 1714, 2000, 2182 @@ -1104,8 +1090,6 @@ static int cs42l52_probe(struct snd_soc_codec *codec) cs42l52_init_beep(codec); - cs42l52_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - cs42l52->sysclk = CS42L52_DEFAULT_CLK; cs42l52->config.format = CS42L52_DEFAULT_FORMAT; @@ -1115,7 +1099,6 @@ static int cs42l52_probe(struct snd_soc_codec *codec) static int cs42l52_remove(struct snd_soc_codec *codec) { cs42l52_free_beep(codec); - cs42l52_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; } @@ -1123,9 +1106,8 @@ static int cs42l52_remove(struct snd_soc_codec *codec) static struct snd_soc_codec_driver soc_codec_dev_cs42l52 = { .probe = cs42l52_probe, .remove = cs42l52_remove, - .suspend = cs42l52_suspend, - .resume = cs42l52_resume, .set_bias_level = cs42l52_set_bias_level, + .suspend_bias_off = true, .dapm_widgets = cs42l52_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(cs42l52_dapm_widgets), -- cgit v1.1 From 2a4bc751fcc50c15bd4782cfc2ea513bef92a20f Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 9 Sep 2014 20:42:41 +0200 Subject: ASoC: cs42l56: Cleanup manual bias level transitions Set the CODEC driver's suspend_bias_off flag rather than manually going to SND_SOC_BIAS_OFF in suspend and SND_SOC_BIAS_STANDBY in resume. This makes the code a bit shorter and cleaner. Since the ASoC core now takes care of setting the bias level to SND_SOC_BIAS_OFF when removing the CODEC there is no need to do it manually anymore either. The manual transition to SND_SOC_BIAS_STANDBY at the end of CODEC probe() can also be removed as the core will automatically do this after the CODEC has been probed. Signed-off-by: Lars-Peter Clausen Acked-by: Brian Austin Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l56.c | 20 +------------------- 1 file changed, 1 insertion(+), 19 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/cs42l56.c b/sound/soc/codecs/cs42l56.c index bb74dd1..2ddc7ac 100644 --- a/sound/soc/codecs/cs42l56.c +++ b/sound/soc/codecs/cs42l56.c @@ -1016,20 +1016,6 @@ static struct snd_soc_dai_driver cs42l56_dai = { .ops = &cs42l56_ops, }; -static int cs42l56_suspend(struct snd_soc_codec *codec) -{ - cs42l56_set_bias_level(codec, SND_SOC_BIAS_OFF); - - return 0; -} - -static int cs42l56_resume(struct snd_soc_codec *codec) -{ - cs42l56_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - - return 0; -} - static int beep_freq[] = { 261, 522, 585, 667, 706, 774, 889, 1000, 1043, 1200, 1333, 1412, 1600, 1714, 2000, 2182 @@ -1168,15 +1154,12 @@ static int cs42l56_probe(struct snd_soc_codec *codec) { cs42l56_init_beep(codec); - cs42l56_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - return 0; } static int cs42l56_remove(struct snd_soc_codec *codec) { cs42l56_free_beep(codec); - cs42l56_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; } @@ -1184,9 +1167,8 @@ static int cs42l56_remove(struct snd_soc_codec *codec) static struct snd_soc_codec_driver soc_codec_dev_cs42l56 = { .probe = cs42l56_probe, .remove = cs42l56_remove, - .suspend = cs42l56_suspend, - .resume = cs42l56_resume, .set_bias_level = cs42l56_set_bias_level, + .suspend_bias_off = true, .dapm_widgets = cs42l56_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(cs42l56_dapm_widgets), -- cgit v1.1 From 02bf34f4b8793a23dd0dbc4fda09d611a70ca0c9 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 9 Sep 2014 20:42:42 +0200 Subject: ASoC: cs42l73: Cleanup manual bias level transitions Set the CODEC driver's suspend_bias_off flag rather than manually going to SND_SOC_BIAS_OFF in suspend and SND_SOC_BIAS_STANDBY in resume. This makes the code a bit shorter and cleaner. Since the ASoC core now takes care of setting the bias level to SND_SOC_BIAS_OFF when removing the CODEC there is no need to do it manually anymore either. The manual transition to SND_SOC_BIAS_STANDBY at the end of CODEC probe() can also be removed as the core will automatically do this after the CODEC has been probed. Signed-off-by: Lars-Peter Clausen Acked-by: Brian Austin Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l73.c | 25 +------------------------ 1 file changed, 1 insertion(+), 24 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c index 0e7b9eb..2f8b946 100644 --- a/sound/soc/codecs/cs42l73.c +++ b/sound/soc/codecs/cs42l73.c @@ -1330,25 +1330,10 @@ static struct snd_soc_dai_driver cs42l73_dai[] = { } }; -static int cs42l73_suspend(struct snd_soc_codec *codec) -{ - cs42l73_set_bias_level(codec, SND_SOC_BIAS_OFF); - - return 0; -} - -static int cs42l73_resume(struct snd_soc_codec *codec) -{ - cs42l73_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - return 0; -} - static int cs42l73_probe(struct snd_soc_codec *codec) { struct cs42l73_private *cs42l73 = snd_soc_codec_get_drvdata(codec); - cs42l73_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - /* Set Charge Pump Frequency */ if (cs42l73->pdata.chgfreq) snd_soc_update_bits(codec, CS42L73_CPFCHC, @@ -1362,18 +1347,10 @@ static int cs42l73_probe(struct snd_soc_codec *codec) return 0; } -static int cs42l73_remove(struct snd_soc_codec *codec) -{ - cs42l73_set_bias_level(codec, SND_SOC_BIAS_OFF); - return 0; -} - static struct snd_soc_codec_driver soc_codec_dev_cs42l73 = { .probe = cs42l73_probe, - .remove = cs42l73_remove, - .suspend = cs42l73_suspend, - .resume = cs42l73_resume, .set_bias_level = cs42l73_set_bias_level, + .suspend_bias_off = true, .dapm_widgets = cs42l73_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(cs42l73_dapm_widgets), -- cgit v1.1 From f66a91ff8e83e95c822691270d883cbcb3244302 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 9 Sep 2014 20:42:43 +0200 Subject: ASoC: da732x: Remove unnecessary idle_bias_off initialization idle_bias_off is false by default, no need to set it explicitly. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/da732x.c | 3 --- 1 file changed, 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/da732x.c b/sound/soc/codecs/da732x.c index 2fae31c..edcbfea 100644 --- a/sound/soc/codecs/da732x.c +++ b/sound/soc/codecs/da732x.c @@ -1511,12 +1511,9 @@ static int da732x_set_bias_level(struct snd_soc_codec *codec, static int da732x_probe(struct snd_soc_codec *codec) { struct da732x_priv *da732x = snd_soc_codec_get_drvdata(codec); - struct snd_soc_dapm_context *dapm = &codec->dapm; da732x->codec = codec; - dapm->idle_bias_off = false; - da732x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); return 0; -- cgit v1.1 From ee6b42ee21b16aa322758fdab0d57082761b09fd Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 9 Sep 2014 20:42:44 +0200 Subject: ASoC: da732x: Remove unused codec field form da732x_priv struct The field is initialized in the probe callback, but never used again. So it can be removed. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/da732x.c | 3 --- 1 file changed, 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/da732x.c b/sound/soc/codecs/da732x.c index edcbfea..c28cf33 100644 --- a/sound/soc/codecs/da732x.c +++ b/sound/soc/codecs/da732x.c @@ -35,7 +35,6 @@ struct da732x_priv { struct regmap *regmap; - struct snd_soc_codec *codec; unsigned int sysclk; bool pll_en; @@ -1512,8 +1511,6 @@ static int da732x_probe(struct snd_soc_codec *codec) { struct da732x_priv *da732x = snd_soc_codec_get_drvdata(codec); - da732x->codec = codec; - da732x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); return 0; -- cgit v1.1 From f0b99ca041258ed0eb27dc724de22d84dab78a7c Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 9 Sep 2014 20:42:45 +0200 Subject: ASoC: da732x: Cleanup manual bias level transitions Since the ASoC core now takes care of setting the bias level to SND_SOC_BIAS_OFF when removing the CODEC there is no need to do it manually anymore either. The manual transition to SND_SOC_BIAS_STANDBY at the end of CODEC probe() can also be removed as the core will automatically do this after the CODEC has been probed. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/da732x.c | 19 ------------------- 1 file changed, 19 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/da732x.c b/sound/soc/codecs/da732x.c index c28cf33..f35e83e 100644 --- a/sound/soc/codecs/da732x.c +++ b/sound/soc/codecs/da732x.c @@ -1507,26 +1507,7 @@ static int da732x_set_bias_level(struct snd_soc_codec *codec, return 0; } -static int da732x_probe(struct snd_soc_codec *codec) -{ - struct da732x_priv *da732x = snd_soc_codec_get_drvdata(codec); - - da732x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - - return 0; -} - -static int da732x_remove(struct snd_soc_codec *codec) -{ - - da732x_set_bias_level(codec, SND_SOC_BIAS_OFF); - - return 0; -} - static struct snd_soc_codec_driver soc_codec_dev_da732x = { - .probe = da732x_probe, - .remove = da732x_remove, .set_bias_level = da732x_set_bias_level, .controls = da732x_snd_controls, .num_controls = ARRAY_SIZE(da732x_snd_controls), -- cgit v1.1 From 8f70e515a8bb6a1908b40b786cb43f6491e8da04 Mon Sep 17 00:00:00 2001 From: Qiao Zhou Date: Wed, 10 Sep 2014 17:54:07 +0800 Subject: ASoC: soc-pcm: fix dpcm_path_get error handling dpcm_path_get may return -ENOMEM when allocating memory for list fails. We should not keep processing path or start up dpcm dai in this case. Signed-off-by: Qiao Zhou Acked-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/soc-compress.c | 6 +++++- sound/soc/soc-pcm.c | 6 +++++- 2 files changed, 10 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 27c06ac..3092b58f 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -101,7 +101,11 @@ static int soc_compr_open_fe(struct snd_compr_stream *cstream) fe->dpcm[stream].runtime = fe_substream->runtime; - if (dpcm_path_get(fe, stream, &list) <= 0) { + ret = dpcm_path_get(fe, stream, &list); + if (ret < 0) { + mutex_unlock(&fe->card->mutex); + goto fe_err; + } else if (ret == 0) { dev_dbg(fe->dev, "ASoC: %s no valid %s route\n", fe->dai_link->name, stream ? "capture" : "playback"); } diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 731fdb5..642c862 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2352,7 +2352,11 @@ static int dpcm_fe_dai_open(struct snd_pcm_substream *fe_substream) mutex_lock_nested(&fe->card->mutex, SND_SOC_CARD_CLASS_RUNTIME); fe->dpcm[stream].runtime = fe_substream->runtime; - if (dpcm_path_get(fe, stream, &list) <= 0) { + ret = dpcm_path_get(fe, stream, &list); + if (ret < 0) { + mutex_unlock(&fe->card->mutex); + return ret; + } else if (ret == 0) { dev_dbg(fe->dev, "ASoC: %s no valid %s route\n", fe->dai_link->name, stream ? "capture" : "playback"); } -- cgit v1.1 From 0dd4fc3c2f663b9124855daf3fd841d70b4dbeea Mon Sep 17 00:00:00 2001 From: Xiubo Li Date: Wed, 10 Sep 2014 09:59:55 +0800 Subject: ASoC: simple-card: Adjust the comments of simple card. Signed-off-by: Xiubo Li Signed-off-by: Mark Brown --- sound/soc/generic/simple-card.c | 48 ++++++++++++++++++++--------------------- 1 file changed, 24 insertions(+), 24 deletions(-) (limited to 'sound') diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index e0abe77..f79347c 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -122,7 +122,7 @@ asoc_simple_card_sub_parse_of(struct device_node *np, int ret; /* - * get node via "sound-dai = <&phandle port>" + * Get node via "sound-dai = <&phandle port>" * it will be used as xxx_of_node on soc_bind_dai_link() */ ret = of_parse_phandle_with_args(np, "sound-dai", @@ -135,19 +135,19 @@ asoc_simple_card_sub_parse_of(struct device_node *np, if (args_count) *args_count = args.args_count; - /* get dai->name */ + /* Get dai->name */ ret = snd_soc_of_get_dai_name(np, name); if (ret < 0) return ret; - /* parse TDM slot */ + /* Parse TDM slot */ ret = snd_soc_of_parse_tdm_slot(np, &dai->slots, &dai->slot_width); if (ret) return ret; /* - * dai->sysclk come from - * "clocks = <&xxx>" (if system has common clock) + * Parse dai->sysclk come from "clocks = <&xxx>" + * (if system has common clock) * or "system-clock-frequency = " * or device's module clock. */ @@ -240,9 +240,11 @@ static int asoc_simple_card_dai_link_of(struct device_node *node, goto dai_link_of_err; if (strlen(prefix) && !bitclkmaster && !framemaster) { - /* No dai-link level and master setting was not found from - sound node level, revert back to legacy DT parsing and - take the settings from codec node. */ + /* + * No DAI link level and master setting was found + * from sound node level, revert back to legacy DT + * parsing and take the settings from codec node. + */ dev_dbg(dev, "%s: Revert to legacy daifmt parsing\n", __func__); dai_props->cpu_dai.fmt = dai_props->codec_dai.fmt = @@ -271,10 +273,10 @@ static int asoc_simple_card_dai_link_of(struct device_node *node, goto dai_link_of_err; } - /* simple-card assumes platform == cpu */ + /* Simple Card assumes platform == cpu */ dai_link->platform_of_node = dai_link->cpu_of_node; - /* Link name is created from CPU/CODEC dai name */ + /* DAI link name is created from CPU/CODEC dai name */ name = devm_kzalloc(dev, strlen(dai_link->cpu_dai_name) + strlen(dai_link->codec_dai_name) + 2, @@ -296,11 +298,11 @@ static int asoc_simple_card_dai_link_of(struct device_node *node, dai_props->codec_dai.sysclk); /* - * soc_bind_dai_link() will check cpu name - * after of_node matching if dai_link has cpu_dai_name. - * but, it will never match if name was created by fmt_single_name() - * remove cpu_dai_name if cpu_args was 0. - * see + * In soc_bind_dai_link() will check cpu name after + * of_node matching if dai_link has cpu_dai_name. + * but, it will never match if name was created by + * fmt_single_name() remove cpu_dai_name if cpu_args + * was 0. See: * fmt_single_name() * fmt_multiple_name() */ @@ -329,10 +331,10 @@ static int asoc_simple_card_parse_of(struct device_node *node, if (!node) return -EINVAL; - /* parsing the card name from DT */ + /* Parse the card name from DT */ snd_soc_of_parse_card_name(&priv->snd_card, "simple-audio-card,name"); - /* off-codec widgets */ + /* The off-codec widgets */ if (of_property_read_bool(node, "simple-audio-card,widgets")) { ret = snd_soc_of_parse_audio_simple_widgets(&priv->snd_card, "simple-audio-card,widgets"); @@ -387,7 +389,7 @@ static int asoc_simple_card_parse_of(struct device_node *node, return 0; } -/* update the reference count of the devices nodes at end of probe */ +/* Decrease the reference count of the device nodes */ static int asoc_simple_card_unref(struct platform_device *pdev) { struct snd_soc_card *card = platform_get_drvdata(pdev); @@ -416,29 +418,27 @@ static int asoc_simple_card_probe(struct platform_device *pdev) struct device *dev = &pdev->dev; int num_links, ret; - /* get the number of DAI links */ + /* Get the number of DAI links */ if (np && of_get_child_by_name(np, "simple-audio-card,dai-link")) num_links = of_get_child_count(np); else num_links = 1; - /* allocate the private data and the DAI link array */ + /* Allocate the private data and the DAI link array */ priv = devm_kzalloc(dev, sizeof(*priv) + sizeof(*dai_link) * num_links, GFP_KERNEL); if (!priv) return -ENOMEM; - /* - * init snd_soc_card - */ + /* Init snd_soc_card */ priv->snd_card.owner = THIS_MODULE; priv->snd_card.dev = dev; dai_link = priv->dai_link; priv->snd_card.dai_link = dai_link; priv->snd_card.num_links = num_links; - /* get room for the other properties */ + /* Get room for the other properties */ priv->dai_props = devm_kzalloc(dev, sizeof(*priv->dai_props) * num_links, GFP_KERNEL); -- cgit v1.1 From 88a60e552f114ae34796604575239fb196658067 Mon Sep 17 00:00:00 2001 From: Arnd Bergmann Date: Mon, 8 Sep 2014 13:14:05 +0200 Subject: ASoC: simple-card: fix regression in clock rate lookup Commit 7c7b9cf53d284f ("ASoC: simple-card: fixup cpu_dai_name clear case") changed the way that "sound-dai" properties are handled, which leads to the clock frequency not being picked up from the node that the phandle points to, as correctly identified by gcc with this warning: sound/soc/generic/simple-card.c: In function 'asoc_simple_card_sub_parse_of': sound/soc/generic/simple-card.c:165:7: warning: 'node' may be used uninitialized in this function [-Wmaybe-uninitialized] This restores the previous behavior by using the node from of_parse_phandle_with_args() that was previously being returned from of_parse_phandle(). Signed-off-by: Arnd Bergmann Signed-off-by: Mark Brown --- sound/soc/generic/simple-card.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index f79347c..106fdad 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -116,7 +116,6 @@ asoc_simple_card_sub_parse_of(struct device_node *np, int *args_count) { struct of_phandle_args args; - struct device_node *node; struct clk *clk; u32 val; int ret; @@ -162,7 +161,7 @@ asoc_simple_card_sub_parse_of(struct device_node *np, } else if (!of_property_read_u32(np, "system-clock-frequency", &val)) { dai->sysclk = val; } else { - clk = of_clk_get(node, 0); + clk = of_clk_get(args.np, 0); if (!IS_ERR(clk)) dai->sysclk = clk_get_rate(clk); } -- cgit v1.1 From 774418253e0ec226ad220c6237bba80fd3f4fbc0 Mon Sep 17 00:00:00 2001 From: Paul Handrigan Date: Thu, 11 Sep 2014 09:52:46 -0500 Subject: ASoC: cs4265: Fix register address to set the proper data type. The SPDIF control register must be written to set the data type in hw_params not the ADC control register. Signed-off-by: Paul Handrigan Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/cs4265.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/cs4265.c b/sound/soc/codecs/cs4265.c index a20b30c..367242a 100644 --- a/sound/soc/codecs/cs4265.c +++ b/sound/soc/codecs/cs4265.c @@ -458,12 +458,12 @@ static int cs4265_pcm_hw_params(struct snd_pcm_substream *substream, if (params_width(params) == 16) { snd_soc_update_bits(codec, CS4265_DAC_CTL, CS4265_DAC_CTL_DIF, (1 << 5)); - snd_soc_update_bits(codec, CS4265_ADC_CTL, + snd_soc_update_bits(codec, CS4265_SPDIF_CTL2, CS4265_SPDIF_CTL2_DIF, (1 << 7)); } else { snd_soc_update_bits(codec, CS4265_DAC_CTL, CS4265_DAC_CTL_DIF, (3 << 5)); - snd_soc_update_bits(codec, CS4265_ADC_CTL, + snd_soc_update_bits(codec, CS4265_SPDIF_CTL2, CS4265_SPDIF_CTL2_DIF, (1 << 7)); } break; @@ -472,7 +472,7 @@ static int cs4265_pcm_hw_params(struct snd_pcm_substream *substream, CS4265_DAC_CTL_DIF, 0); snd_soc_update_bits(codec, CS4265_ADC_CTL, CS4265_ADC_DIF, 0); - snd_soc_update_bits(codec, CS4265_ADC_CTL, + snd_soc_update_bits(codec, CS4265_SPDIF_CTL2, CS4265_SPDIF_CTL2_DIF, (1 << 6)); break; -- cgit v1.1 From f531913f01a07253d013a9c67a80df11154e7ae2 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 9 Sep 2014 21:37:57 -0700 Subject: ASoC: simple-card: tidyup use priv in parameter priv has many information about simple-card driver. Using it becomes easy to extend feature. This patch gets dev from priv as 1st step Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/generic/simple-card.c | 15 +++++++++------ 1 file changed, 9 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 106fdad..28aa5e2 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -28,6 +28,8 @@ struct simple_card_data { struct snd_soc_dai_link dai_link[]; /* dynamically allocated */ }; +#define simple_priv_to_dev(priv) ((priv)->snd_card.dev) + static int asoc_simple_card_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -170,11 +172,12 @@ asoc_simple_card_sub_parse_of(struct device_node *np, } static int asoc_simple_card_dai_link_of(struct device_node *node, - struct device *dev, + struct simple_card_data *priv, struct snd_soc_dai_link *dai_link, struct simple_dai_props *dai_props, bool is_top_level_node) { + struct device *dev = simple_priv_to_dev(priv); struct device_node *np = NULL; struct device_node *bitclkmaster = NULL; struct device_node *framemaster = NULL; @@ -319,9 +322,9 @@ dai_link_of_err: } static int asoc_simple_card_parse_of(struct device_node *node, - struct simple_card_data *priv, - struct device *dev) + struct simple_card_data *priv) { + struct device *dev = simple_priv_to_dev(priv); struct snd_soc_dai_link *dai_link = priv->snd_card.dai_link; struct simple_dai_props *dai_props = priv->dai_props; u32 val; @@ -364,7 +367,7 @@ static int asoc_simple_card_parse_of(struct device_node *node, for_each_child_of_node(node, np) { dev_dbg(dev, "\tlink %d:\n", i); - ret = asoc_simple_card_dai_link_of(np, dev, + ret = asoc_simple_card_dai_link_of(np, priv, dai_link + i, dai_props + i, false); @@ -376,7 +379,7 @@ static int asoc_simple_card_parse_of(struct device_node *node, } } else { /* For single DAI link & old style of DT node */ - ret = asoc_simple_card_dai_link_of(node, dev, + ret = asoc_simple_card_dai_link_of(node, priv, dai_link, dai_props, true); if (ret < 0) return ret; @@ -446,7 +449,7 @@ static int asoc_simple_card_probe(struct platform_device *pdev) if (np && of_device_is_available(np)) { - ret = asoc_simple_card_parse_of(np, priv, dev); + ret = asoc_simple_card_parse_of(np, priv); if (ret < 0) { if (ret != -EPROBE_DEFER) dev_err(dev, "parse error %d\n", ret); -- cgit v1.1 From 9810f5370b6e60c4b564f294feb51761f0e741f6 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 9 Sep 2014 21:38:24 -0700 Subject: ASoC: simple-card: tidyup get dai_link/dai_props from priv It can get dai_link/dai_props pointer from priv + index Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/generic/simple-card.c | 16 +++++++--------- 1 file changed, 7 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 28aa5e2..a887707 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -29,6 +29,8 @@ struct simple_card_data { }; #define simple_priv_to_dev(priv) ((priv)->snd_card.dev) +#define simple_priv_to_link(priv, i) ((priv)->snd_card.dai_link + i) +#define simple_priv_to_props(priv, i) ((priv)->dai_props + i) static int asoc_simple_card_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) @@ -173,11 +175,12 @@ asoc_simple_card_sub_parse_of(struct device_node *np, static int asoc_simple_card_dai_link_of(struct device_node *node, struct simple_card_data *priv, - struct snd_soc_dai_link *dai_link, - struct simple_dai_props *dai_props, + int idx, bool is_top_level_node) { struct device *dev = simple_priv_to_dev(priv); + struct snd_soc_dai_link *dai_link = simple_priv_to_link(priv, idx); + struct simple_dai_props *dai_props = simple_priv_to_props(priv, idx); struct device_node *np = NULL; struct device_node *bitclkmaster = NULL; struct device_node *framemaster = NULL; @@ -325,8 +328,6 @@ static int asoc_simple_card_parse_of(struct device_node *node, struct simple_card_data *priv) { struct device *dev = simple_priv_to_dev(priv); - struct snd_soc_dai_link *dai_link = priv->snd_card.dai_link; - struct simple_dai_props *dai_props = priv->dai_props; u32 val; int ret; @@ -368,9 +369,7 @@ static int asoc_simple_card_parse_of(struct device_node *node, for_each_child_of_node(node, np) { dev_dbg(dev, "\tlink %d:\n", i); ret = asoc_simple_card_dai_link_of(np, priv, - dai_link + i, - dai_props + i, - false); + i, false); if (ret < 0) { of_node_put(np); return ret; @@ -379,8 +378,7 @@ static int asoc_simple_card_parse_of(struct device_node *node, } } else { /* For single DAI link & old style of DT node */ - ret = asoc_simple_card_dai_link_of(node, priv, - dai_link, dai_props, true); + ret = asoc_simple_card_dai_link_of(node, priv, 0, true); if (ret < 0) return ret; } -- cgit v1.1 From 07833d88314c496f8a136c6e4b4729c69e65b878 Mon Sep 17 00:00:00 2001 From: Jianqun Date: Sat, 13 Sep 2014 08:41:03 +0800 Subject: ASoC: rockchip-i2s: fix master mode set bit error Fix error format set to I2S master or slave mode. Test on RK3288 board with max98090. Signed-off-by: Jianqun Xu Signed-off-by: Mark Brown --- sound/soc/rockchip/rockchip_i2s.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index 8d8e4b5..870a664 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -165,13 +165,14 @@ static int rockchip_i2s_set_fmt(struct snd_soc_dai *cpu_dai, struct rk_i2s_dev *i2s = to_info(cpu_dai); unsigned int mask = 0, val = 0; - mask = I2S_CKR_MSS_SLAVE; + mask = I2S_CKR_MSS_MASK; switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBS_CFS: - val = I2S_CKR_MSS_SLAVE; + /* Set source clock in Master mode */ + val = I2S_CKR_MSS_MASTER; break; case SND_SOC_DAIFMT_CBM_CFM: - val = I2S_CKR_MSS_MASTER; + val = I2S_CKR_MSS_SLAVE; break; default: return -EINVAL; -- cgit v1.1 From 2f1e93f81cebfa99b668f27cdb14992ff23480a4 Mon Sep 17 00:00:00 2001 From: Jianqun Date: Sat, 13 Sep 2014 08:42:12 +0800 Subject: ASoC: rockchip-i2s: fix registers' property of rockchip i2s controller Reference rockchip I2S controller TRM, modify some registers' property I2S_FIFOLR: read / write, but not volatile, not precious I2S_INTSR: read / write I2S_CLR: volatile, register value will be cleared by read Test on RK3288 with max98090. Signed-off-by: Jianqun Xu Signed-off-by: Mark Brown --- sound/soc/rockchip/rockchip_i2s.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index 870a664..fb9e05c 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -362,6 +362,8 @@ static bool rockchip_i2s_rd_reg(struct device *dev, unsigned int reg) case I2S_XFER: case I2S_CLR: case I2S_RXDR: + case I2S_FIFOLR: + case I2S_INTSR: return true; default: return false; @@ -371,8 +373,8 @@ static bool rockchip_i2s_rd_reg(struct device *dev, unsigned int reg) static bool rockchip_i2s_volatile_reg(struct device *dev, unsigned int reg) { switch (reg) { - case I2S_FIFOLR: case I2S_INTSR: + case I2S_CLR: return true; default: return false; @@ -382,8 +384,6 @@ static bool rockchip_i2s_volatile_reg(struct device *dev, unsigned int reg) static bool rockchip_i2s_precious_reg(struct device *dev, unsigned int reg) { switch (reg) { - case I2S_FIFOLR: - return true; default: return false; } -- cgit v1.1 From 01605ad12875c7b5ed71b486f9badb338f4f8c21 Mon Sep 17 00:00:00 2001 From: Jianqun Date: Sat, 13 Sep 2014 08:43:13 +0800 Subject: ASoC: rockchip-i2s: enable "hclk" for rockchip I2S controller As "hclk" is used for rockchip I2S controller, driver must to enable it in probe. Tested on RK3288 with max98090. Signed-off-by: Jianqun Xu Signed-off-by: Mark Brown --- sound/soc/rockchip/rockchip_i2s.c | 5 +++++ 1 file changed, 5 insertions(+) (limited to 'sound') diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index 8d8e4b5..dd423c6 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -419,6 +419,11 @@ static int rockchip_i2s_probe(struct platform_device *pdev) dev_err(&pdev->dev, "Can't retrieve i2s bus clock\n"); return PTR_ERR(i2s->hclk); } + ret = clk_prepare_enable(i2s->hclk); + if (ret) { + dev_err(i2s->dev, "hclock enable failed %d\n", ret); + return ret; + } i2s->mclk = devm_clk_get(&pdev->dev, "i2s_clk"); if (IS_ERR(i2s->mclk)) { -- cgit v1.1 From 38306afc107c53c379757e7f3146a6418328ebc9 Mon Sep 17 00:00:00 2001 From: Jianqun Date: Sat, 13 Sep 2014 08:40:19 +0800 Subject: ASoC: rockchip-i2s: fix rockchip i2s defination more reasonable Fix SND_ROCKCHIP_I2S to be more reasonable - SND_SOC_ROCKCHIP_I2S, SND_SOC_ROCKCHIP_I2S should select by audio driver, instead of SND_SOC_ROCKCHIP. Signed-off-by: Jianqun Xu Signed-off-by: Mark Brown --- sound/soc/rockchip/Kconfig | 3 +-- sound/soc/rockchip/Makefile | 2 +- 2 files changed, 2 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/rockchip/Kconfig b/sound/soc/rockchip/Kconfig index c196a46..78fc159 100644 --- a/sound/soc/rockchip/Kconfig +++ b/sound/soc/rockchip/Kconfig @@ -2,11 +2,10 @@ config SND_SOC_ROCKCHIP tristate "ASoC support for Rockchip" depends on COMPILE_TEST || ARCH_ROCKCHIP select SND_SOC_GENERIC_DMAENGINE_PCM - select SND_ROCKCHIP_I2S help Say Y or M if you want to add support for codecs attached to the Rockchip SoCs' Audio interfaces. You will also need to select the audio interfaces to support below. -config SND_ROCKCHIP_I2S +config SND_SOC_ROCKCHIP_I2S tristate diff --git a/sound/soc/rockchip/Makefile b/sound/soc/rockchip/Makefile index 1006418..b921909 100644 --- a/sound/soc/rockchip/Makefile +++ b/sound/soc/rockchip/Makefile @@ -1,4 +1,4 @@ # ROCKCHIP Platform Support snd-soc-i2s-objs := rockchip_i2s.o -obj-$(CONFIG_SND_ROCKCHIP_I2S) += snd-soc-i2s.o +obj-$(CONFIG_SND_SOC_ROCKCHIP_I2S) += snd-soc-i2s.o -- cgit v1.1 From 3b40a80216e941c518426f7b86705e52acbd413f Mon Sep 17 00:00:00 2001 From: Jianqun Date: Sat, 13 Sep 2014 08:41:38 +0800 Subject: ASoC: rockchip-i2s: add dma data to snd_soc_dai Add playback/capture dma data to snd_soc_dai. Test on RK3288 with max98090. Signed-off-by: Jianqun Xu Signed-off-by: Mark Brown --- sound/soc/rockchip/rockchip_i2s.c | 23 +++++++++++++---------- 1 file changed, 13 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index dd423c6..c8172dd 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -243,16 +243,6 @@ static int rockchip_i2s_hw_params(struct snd_pcm_substream *substream, regmap_update_bits(i2s->regmap, I2S_TXCR, I2S_TXCR_VDW_MASK, val); regmap_update_bits(i2s->regmap, I2S_RXCR, I2S_RXCR_VDW_MASK, val); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - dai->playback_dma_data = &i2s->playback_dma_data; - regmap_update_bits(i2s->regmap, I2S_DMACR, I2S_DMACR_TDL_MASK, - I2S_DMACR_TDL(1) | I2S_DMACR_TDE_ENABLE); - } else { - dai->capture_dma_data = &i2s->capture_dma_data; - regmap_update_bits(i2s->regmap, I2S_DMACR, I2S_DMACR_RDL_MASK, - I2S_DMACR_RDL(1) | I2S_DMACR_RDE_ENABLE); - } - return 0; } @@ -300,6 +290,16 @@ static int rockchip_i2s_set_sysclk(struct snd_soc_dai *cpu_dai, int clk_id, return ret; } +static int rockchip_i2s_dai_probe(struct snd_soc_dai *dai) +{ + struct rk_i2s_dev *i2s = snd_soc_dai_get_drvdata(dai); + + dai->capture_dma_data = &i2s->capture_dma_data; + dai->playback_dma_data = &i2s->playback_dma_data; + + return 0; +} + static const struct snd_soc_dai_ops rockchip_i2s_dai_ops = { .hw_params = rockchip_i2s_hw_params, .set_sysclk = rockchip_i2s_set_sysclk, @@ -308,7 +308,9 @@ static const struct snd_soc_dai_ops rockchip_i2s_dai_ops = { }; static struct snd_soc_dai_driver rockchip_i2s_dai = { + .probe = rockchip_i2s_dai_probe, .playback = { + .stream_name = "Playback", .channels_min = 2, .channels_max = 8, .rates = SNDRV_PCM_RATE_8000_192000, @@ -318,6 +320,7 @@ static struct snd_soc_dai_driver rockchip_i2s_dai = { SNDRV_PCM_FMTBIT_S24_LE), }, .capture = { + .stream_name = "Capture", .channels_min = 2, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_192000, -- cgit v1.1 From e5b2791d2a57e9da369bd75ae2a209bcce2ad4d3 Mon Sep 17 00:00:00 2001 From: Oder Chiou Date: Mon, 15 Sep 2014 19:58:44 +0800 Subject: ASoC: rt5677: Revise the wrong name in the header file The patch revises the wrong name in the header file. Signed-off-by: Oder Chiou Signed-off-by: Mark Brown --- sound/soc/codecs/rt5677.h | 44 ++++++++++++++++++++++---------------------- 1 file changed, 22 insertions(+), 22 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt5677.h b/sound/soc/codecs/rt5677.h index 8791ab9..a334eb6 100644 --- a/sound/soc/codecs/rt5677.h +++ b/sound/soc/codecs/rt5677.h @@ -1287,16 +1287,16 @@ #define RT5677_PLL1_PD_SFT 8 #define RT5677_PLL1_PD_1 (0x0 << 8) #define RT5677_PLL1_PD_2 (0x1 << 8) -#define RT5671_DAC_OSR_MASK (0x3 << 6) -#define RT5671_DAC_OSR_SFT 6 -#define RT5671_DAC_OSR_128 (0x0 << 6) -#define RT5671_DAC_OSR_64 (0x1 << 6) -#define RT5671_DAC_OSR_32 (0x2 << 6) -#define RT5671_ADC_OSR_MASK (0x3 << 4) -#define RT5671_ADC_OSR_SFT 4 -#define RT5671_ADC_OSR_128 (0x0 << 4) -#define RT5671_ADC_OSR_64 (0x1 << 4) -#define RT5671_ADC_OSR_32 (0x2 << 4) +#define RT5677_DAC_OSR_MASK (0x3 << 6) +#define RT5677_DAC_OSR_SFT 6 +#define RT5677_DAC_OSR_128 (0x0 << 6) +#define RT5677_DAC_OSR_64 (0x1 << 6) +#define RT5677_DAC_OSR_32 (0x2 << 6) +#define RT5677_ADC_OSR_MASK (0x3 << 4) +#define RT5677_ADC_OSR_SFT 4 +#define RT5677_ADC_OSR_128 (0x0 << 4) +#define RT5677_ADC_OSR_64 (0x1 << 4) +#define RT5677_ADC_OSR_32 (0x2 << 4) /* Global Clock Control 2 (0x81) */ #define RT5677_PLL2_PR_SRC_MASK (0x1 << 15) @@ -1312,18 +1312,18 @@ #define RT5677_PLL2_SRC_BCLK4 (0x4 << 12) #define RT5677_PLL2_SRC_RCCLK (0x5 << 12) #define RT5677_PLL2_SRC_SLIM (0x6 << 12) -#define RT5671_DSP_ASRC_O_SRC (0x3 << 10) -#define RT5671_DSP_ASRC_O_SRC_SFT 10 -#define RT5671_DSP_ASRC_O_MCLK (0x0 << 10) -#define RT5671_DSP_ASRC_O_PLL1 (0x1 << 10) -#define RT5671_DSP_ASRC_O_SLIM (0x2 << 10) -#define RT5671_DSP_ASRC_O_RCCLK (0x3 << 10) -#define RT5671_DSP_ASRC_I_SRC (0x3 << 8) -#define RT5671_DSP_ASRC_I_SRC_SFT 8 -#define RT5671_DSP_ASRC_I_MCLK (0x0 << 8) -#define RT5671_DSP_ASRC_I_PLL1 (0x1 << 8) -#define RT5671_DSP_ASRC_I_SLIM (0x2 << 8) -#define RT5671_DSP_ASRC_I_RCCLK (0x3 << 8) +#define RT5677_DSP_ASRC_O_SRC (0x3 << 10) +#define RT5677_DSP_ASRC_O_SRC_SFT 10 +#define RT5677_DSP_ASRC_O_MCLK (0x0 << 10) +#define RT5677_DSP_ASRC_O_PLL1 (0x1 << 10) +#define RT5677_DSP_ASRC_O_SLIM (0x2 << 10) +#define RT5677_DSP_ASRC_O_RCCLK (0x3 << 10) +#define RT5677_DSP_ASRC_I_SRC (0x3 << 8) +#define RT5677_DSP_ASRC_I_SRC_SFT 8 +#define RT5677_DSP_ASRC_I_MCLK (0x0 << 8) +#define RT5677_DSP_ASRC_I_PLL1 (0x1 << 8) +#define RT5677_DSP_ASRC_I_SLIM (0x2 << 8) +#define RT5677_DSP_ASRC_I_RCCLK (0x3 << 8) #define RT5677_DSP_CLK_SRC_MASK (0x1 << 7) #define RT5677_DSP_CLK_SRC_SFT 7 #define RT5677_DSP_CLK_SRC_PLL2 (0x0 << 7) -- cgit v1.1 From 44caf7648064502fd1d37d18443ae92c064ebadd Mon Sep 17 00:00:00 2001 From: Oder Chiou Date: Tue, 16 Sep 2014 11:37:39 +0800 Subject: ASoC: rt5677: Add the GPIO function The patch adds the GPIO function. Signed-off-by: Oder Chiou Signed-off-by: Mark Brown --- sound/soc/codecs/rt5677.c | 133 ++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/rt5677.h | 112 ++++++++++++++++++++++++++++++++++++++ 2 files changed, 245 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index f0b751b..02bc8bd 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -19,6 +19,7 @@ #include #include #include +#include #include #include #include @@ -3160,6 +3161,135 @@ static int rt5677_set_bias_level(struct snd_soc_codec *codec, return 0; } +#ifdef CONFIG_GPIOLIB +static inline struct rt5677_priv *gpio_to_rt5677(struct gpio_chip *chip) +{ + return container_of(chip, struct rt5677_priv, gpio_chip); +} + +static void rt5677_gpio_set(struct gpio_chip *chip, unsigned offset, int value) +{ + struct rt5677_priv *rt5677 = gpio_to_rt5677(chip); + + switch (offset) { + case RT5677_GPIO1 ... RT5677_GPIO5: + regmap_update_bits(rt5677->regmap, RT5677_GPIO_CTRL2, + 0x1 << (offset * 3 + 1), !!value << (offset * 3 + 1)); + break; + + case RT5677_GPIO6: + regmap_update_bits(rt5677->regmap, RT5677_GPIO_CTRL3, + RT5677_GPIO6_OUT_MASK, !!value << RT5677_GPIO6_OUT_SFT); + break; + + default: + break; + } +} + +static int rt5677_gpio_direction_out(struct gpio_chip *chip, + unsigned offset, int value) +{ + struct rt5677_priv *rt5677 = gpio_to_rt5677(chip); + + switch (offset) { + case RT5677_GPIO1 ... RT5677_GPIO5: + regmap_update_bits(rt5677->regmap, RT5677_GPIO_CTRL2, + 0x3 << (offset * 3 + 1), + (0x2 | !!value) << (offset * 3 + 1)); + break; + + case RT5677_GPIO6: + regmap_update_bits(rt5677->regmap, RT5677_GPIO_CTRL3, + RT5677_GPIO6_DIR_MASK | RT5677_GPIO6_OUT_MASK, + RT5677_GPIO6_DIR_OUT | !!value << RT5677_GPIO6_OUT_SFT); + break; + + default: + break; + } + + return 0; +} + +static int rt5677_gpio_get(struct gpio_chip *chip, unsigned offset) +{ + struct rt5677_priv *rt5677 = gpio_to_rt5677(chip); + int value, ret; + + ret = regmap_read(rt5677->regmap, RT5677_GPIO_ST, &value); + if (ret < 0) + return ret; + + return (value & (0x1 << offset)) >> offset; +} + +static int rt5677_gpio_direction_in(struct gpio_chip *chip, unsigned offset) +{ + struct rt5677_priv *rt5677 = gpio_to_rt5677(chip); + + switch (offset) { + case RT5677_GPIO1 ... RT5677_GPIO5: + regmap_update_bits(rt5677->regmap, RT5677_GPIO_CTRL2, + 0x1 << (offset * 3 + 2), 0x0); + break; + + case RT5677_GPIO6: + regmap_update_bits(rt5677->regmap, RT5677_GPIO_CTRL3, + RT5677_GPIO6_DIR_MASK, RT5677_GPIO6_DIR_IN); + break; + + default: + break; + } + + return 0; +} + +static struct gpio_chip rt5677_template_chip = { + .label = "rt5677", + .owner = THIS_MODULE, + .direction_output = rt5677_gpio_direction_out, + .set = rt5677_gpio_set, + .direction_input = rt5677_gpio_direction_in, + .get = rt5677_gpio_get, + .can_sleep = 1, +}; + +static void rt5677_init_gpio(struct i2c_client *i2c) +{ + struct rt5677_priv *rt5677 = i2c_get_clientdata(i2c); + int ret; + + rt5677->gpio_chip = rt5677_template_chip; + rt5677->gpio_chip.ngpio = RT5677_GPIO_NUM; + rt5677->gpio_chip.dev = &i2c->dev; + rt5677->gpio_chip.base = -1; + + ret = gpiochip_add(&rt5677->gpio_chip); + if (ret != 0) + dev_err(&i2c->dev, "Failed to add GPIOs: %d\n", ret); +} + +static void rt5677_free_gpio(struct i2c_client *i2c) +{ + struct rt5677_priv *rt5677 = i2c_get_clientdata(i2c); + int ret; + + ret = gpiochip_remove(&rt5677->gpio_chip); + if (ret != 0) + dev_err(&i2c->dev, "Failed to remove GPIOs: %d\n", ret); +} +#else +static void rt5677_init_gpio(struct i2c_client *i2c) +{ +} + +static void rt5677_free_gpio(struct i2c_client *i2c) +{ +} +#endif + static int rt5677_probe(struct snd_soc_codec *codec) { struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); @@ -3422,6 +3552,8 @@ static int rt5677_i2c_probe(struct i2c_client *i2c, RT5677_GPIO5_DIR_OUT); } + rt5677_init_gpio(i2c); + return snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5677, rt5677_dai, ARRAY_SIZE(rt5677_dai)); } @@ -3429,6 +3561,7 @@ static int rt5677_i2c_probe(struct i2c_client *i2c, static int rt5677_i2c_remove(struct i2c_client *i2c) { snd_soc_unregister_codec(&i2c->dev); + rt5677_free_gpio(i2c); return 0; } diff --git a/sound/soc/codecs/rt5677.h b/sound/soc/codecs/rt5677.h index a334eb6..b61b72c 100644 --- a/sound/soc/codecs/rt5677.h +++ b/sound/soc/codecs/rt5677.h @@ -1363,10 +1363,109 @@ #define RT5677_SEL_SRC_IB01 (0x1 << 0) #define RT5677_SEL_SRC_IB01_SFT 0 +/* GPIO status (0xbf) */ +#define RT5677_GPIO6_STATUS_MASK (0x1 << 5) +#define RT5677_GPIO6_STATUS_SFT 5 +#define RT5677_GPIO5_STATUS_MASK (0x1 << 4) +#define RT5677_GPIO5_STATUS_SFT 4 +#define RT5677_GPIO4_STATUS_MASK (0x1 << 3) +#define RT5677_GPIO4_STATUS_SFT 3 +#define RT5677_GPIO3_STATUS_MASK (0x1 << 2) +#define RT5677_GPIO3_STATUS_SFT 2 +#define RT5677_GPIO2_STATUS_MASK (0x1 << 1) +#define RT5677_GPIO2_STATUS_SFT 1 +#define RT5677_GPIO1_STATUS_MASK (0x1 << 0) +#define RT5677_GPIO1_STATUS_SFT 0 + +/* GPIO Control 1 (0xc0) */ +#define RT5677_GPIO1_PIN_MASK (0x1 << 15) +#define RT5677_GPIO1_PIN_SFT 15 +#define RT5677_GPIO1_PIN_GPIO1 (0x0 << 15) +#define RT5677_GPIO1_PIN_IRQ (0x1 << 15) +#define RT5677_IPTV_MODE_MASK (0x1 << 14) +#define RT5677_IPTV_MODE_SFT 14 +#define RT5677_IPTV_MODE_GPIO (0x0 << 14) +#define RT5677_IPTV_MODE_IPTV (0x1 << 14) +#define RT5677_FUNC_MODE_MASK (0x1 << 13) +#define RT5677_FUNC_MODE_SFT 13 +#define RT5677_FUNC_MODE_DMIC_GPIO (0x0 << 13) +#define RT5677_FUNC_MODE_JTAG (0x1 << 13) + /* GPIO Control 2 (0xc1) */ #define RT5677_GPIO5_DIR_MASK (0x1 << 14) +#define RT5677_GPIO5_DIR_SFT 14 #define RT5677_GPIO5_DIR_IN (0x0 << 14) #define RT5677_GPIO5_DIR_OUT (0x1 << 14) +#define RT5677_GPIO5_OUT_MASK (0x1 << 13) +#define RT5677_GPIO5_OUT_SFT 13 +#define RT5677_GPIO5_OUT_LO (0x0 << 13) +#define RT5677_GPIO5_OUT_HI (0x1 << 13) +#define RT5677_GPIO5_P_MASK (0x1 << 12) +#define RT5677_GPIO5_P_SFT 12 +#define RT5677_GPIO5_P_NOR (0x0 << 12) +#define RT5677_GPIO5_P_INV (0x1 << 12) +#define RT5677_GPIO4_DIR_MASK (0x1 << 11) +#define RT5677_GPIO4_DIR_SFT 11 +#define RT5677_GPIO4_DIR_IN (0x0 << 11) +#define RT5677_GPIO4_DIR_OUT (0x1 << 11) +#define RT5677_GPIO4_OUT_MASK (0x1 << 10) +#define RT5677_GPIO4_OUT_SFT 10 +#define RT5677_GPIO4_OUT_LO (0x0 << 10) +#define RT5677_GPIO4_OUT_HI (0x1 << 10) +#define RT5677_GPIO4_P_MASK (0x1 << 9) +#define RT5677_GPIO4_P_SFT 9 +#define RT5677_GPIO4_P_NOR (0x0 << 9) +#define RT5677_GPIO4_P_INV (0x1 << 9) +#define RT5677_GPIO3_DIR_MASK (0x1 << 8) +#define RT5677_GPIO3_DIR_SFT 8 +#define RT5677_GPIO3_DIR_IN (0x0 << 8) +#define RT5677_GPIO3_DIR_OUT (0x1 << 8) +#define RT5677_GPIO3_OUT_MASK (0x1 << 7) +#define RT5677_GPIO3_OUT_SFT 7 +#define RT5677_GPIO3_OUT_LO (0x0 << 7) +#define RT5677_GPIO3_OUT_HI (0x1 << 7) +#define RT5677_GPIO3_P_MASK (0x1 << 6) +#define RT5677_GPIO3_P_SFT 6 +#define RT5677_GPIO3_P_NOR (0x0 << 6) +#define RT5677_GPIO3_P_INV (0x1 << 6) +#define RT5677_GPIO2_DIR_MASK (0x1 << 5) +#define RT5677_GPIO2_DIR_SFT 5 +#define RT5677_GPIO2_DIR_IN (0x0 << 5) +#define RT5677_GPIO2_DIR_OUT (0x1 << 5) +#define RT5677_GPIO2_OUT_MASK (0x1 << 4) +#define RT5677_GPIO2_OUT_SFT 4 +#define RT5677_GPIO2_OUT_LO (0x0 << 4) +#define RT5677_GPIO2_OUT_HI (0x1 << 4) +#define RT5677_GPIO2_P_MASK (0x1 << 3) +#define RT5677_GPIO2_P_SFT 3 +#define RT5677_GPIO2_P_NOR (0x0 << 3) +#define RT5677_GPIO2_P_INV (0x1 << 3) +#define RT5677_GPIO1_DIR_MASK (0x1 << 2) +#define RT5677_GPIO1_DIR_SFT 2 +#define RT5677_GPIO1_DIR_IN (0x0 << 2) +#define RT5677_GPIO1_DIR_OUT (0x1 << 2) +#define RT5677_GPIO1_OUT_MASK (0x1 << 1) +#define RT5677_GPIO1_OUT_SFT 1 +#define RT5677_GPIO1_OUT_LO (0x0 << 1) +#define RT5677_GPIO1_OUT_HI (0x1 << 1) +#define RT5677_GPIO1_P_MASK (0x1 << 0) +#define RT5677_GPIO1_P_SFT 0 +#define RT5677_GPIO1_P_NOR (0x0 << 0) +#define RT5677_GPIO1_P_INV (0x1 << 0) + +/* GPIO Control 3 (0xc2) */ +#define RT5677_GPIO6_DIR_MASK (0x1 << 2) +#define RT5677_GPIO6_DIR_SFT 2 +#define RT5677_GPIO6_DIR_IN (0x0 << 2) +#define RT5677_GPIO6_DIR_OUT (0x1 << 2) +#define RT5677_GPIO6_OUT_MASK (0x1 << 1) +#define RT5677_GPIO6_OUT_SFT 1 +#define RT5677_GPIO6_OUT_LO (0x0 << 1) +#define RT5677_GPIO6_OUT_HI (0x1 << 1) +#define RT5677_GPIO6_P_MASK (0x1 << 0) +#define RT5677_GPIO6_P_SFT 0 +#define RT5677_GPIO6_P_NOR (0x0 << 0) +#define RT5677_GPIO6_P_INV (0x1 << 0) /* Virtual DSP Mixer Control (0xf7 0xf8 0xf9) */ #define RT5677_DSP_IB_01_H (0x1 << 15) @@ -1428,6 +1527,16 @@ enum { RT5677_AIFS, }; +enum { + RT5677_GPIO1, + RT5677_GPIO2, + RT5677_GPIO3, + RT5677_GPIO4, + RT5677_GPIO5, + RT5677_GPIO6, + RT5677_GPIO_NUM, +}; + struct rt5677_priv { struct snd_soc_codec *codec; struct rt5677_platform_data pdata; @@ -1441,6 +1550,9 @@ struct rt5677_priv { int pll_src; int pll_in; int pll_out; +#ifdef CONFIG_GPIOLIB + struct gpio_chip gpio_chip; +#endif }; #endif /* __RT5677_H__ */ -- cgit v1.1 From d2b16b8fa1b6352757cd0a58234591e1496a82ad Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Tue, 9 Sep 2014 15:11:24 +0530 Subject: ASoC: Intel: mfld-pcm: don't call trigger ops to DSP for internal streams For internal stream i.e. BE we have don't need trigger ops as that would be handled by DAPM for us in subsequent patches Signed-off-by: Vinod Koul Signed-off-by: Subhransu S. Prusty Signed-off-by: Mark Brown --- sound/soc/intel/sst-mfld-platform-pcm.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/soc/intel/sst-mfld-platform-pcm.c b/sound/soc/intel/sst-mfld-platform-pcm.c index 85deecd..9906b7c 100644 --- a/sound/soc/intel/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/sst-mfld-platform-pcm.c @@ -464,6 +464,8 @@ static int sst_platform_pcm_trigger(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; dev_dbg(rtd->dev, "sst_platform_pcm_trigger called\n"); + if (substream->pcm->internal) + return 0; stream = substream->runtime->private_data; str_id = stream->stream_info.str_id; switch (cmd) { -- cgit v1.1 From 10615a5c49721803ed258316280858142a24e72a Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Tue, 9 Sep 2014 15:11:25 +0530 Subject: ASoC: Intel: mrfld: add bytes control for modules This patch add support for various modules like eq etc for mrfld DSP. All these modules will be exposed to usermode as bytes controls. Signed-off-by: Vinod Koul Signed-off-by: Subhransu S. Prusty Signed-off-by: Mark Brown --- sound/soc/intel/sst-atom-controls.c | 179 ++++++++++++++++++++++++++++++++++++ sound/soc/intel/sst-atom-controls.h | 130 ++++++++++++++++++++++++++ sound/soc/intel/sst-mfld-platform.h | 2 +- 3 files changed, 310 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/intel/sst-atom-controls.c b/sound/soc/intel/sst-atom-controls.c index ace3c4a..7104a34 100644 --- a/sound/soc/intel/sst-atom-controls.c +++ b/sound/soc/intel/sst-atom-controls.c @@ -25,6 +25,179 @@ #include "sst-mfld-platform.h" #include "sst-atom-controls.h" +static int sst_fill_byte_control(struct sst_data *drv, + u8 ipc_msg, u8 block, + u8 task_id, u8 pipe_id, + u16 len, void *cmd_data) +{ + struct snd_sst_bytes_v2 *byte_data = drv->byte_stream; + + byte_data->type = SST_CMD_BYTES_SET; + byte_data->ipc_msg = ipc_msg; + byte_data->block = block; + byte_data->task_id = task_id; + byte_data->pipe_id = pipe_id; + + if (len > SST_MAX_BIN_BYTES - sizeof(*byte_data)) { + dev_err(&drv->pdev->dev, "command length too big (%u)", len); + return -EINVAL; + } + byte_data->len = len; + memcpy(byte_data->bytes, cmd_data, len); + print_hex_dump_bytes("writing to lpe: ", DUMP_PREFIX_OFFSET, + byte_data, len + sizeof(*byte_data)); + return 0; +} + +static int sst_fill_and_send_cmd_unlocked(struct sst_data *drv, + u8 ipc_msg, u8 block, u8 task_id, u8 pipe_id, + void *cmd_data, u16 len) +{ + int ret = 0; + + ret = sst_fill_byte_control(drv, ipc_msg, + block, task_id, pipe_id, len, cmd_data); + if (ret < 0) + return ret; + return sst->ops->send_byte_stream(sst->dev, drv->byte_stream); +} + +/** + * sst_fill_and_send_cmd - generate the IPC message and send it to the FW + * @ipc_msg: type of IPC (CMD, SET_PARAMS, GET_PARAMS) + * @cmd_data: the IPC payload + */ +static int sst_fill_and_send_cmd(struct sst_data *drv, + u8 ipc_msg, u8 block, u8 task_id, u8 pipe_id, + void *cmd_data, u16 len) +{ + int ret; + + mutex_lock(&drv->lock); + ret = sst_fill_and_send_cmd_unlocked(drv, ipc_msg, block, + task_id, pipe_id, cmd_data, len); + mutex_unlock(&drv->lock); + + return ret; +} + +static int sst_send_algo_cmd(struct sst_data *drv, + struct sst_algo_control *bc) +{ + int len, ret = 0; + struct sst_cmd_set_params *cmd; + + /*bc->max includes sizeof algos + length field*/ + len = sizeof(cmd->dst) + sizeof(cmd->command_id) + bc->max; + + cmd = kzalloc(len, GFP_KERNEL); + if (cmd == NULL) + return -ENOMEM; + + SST_FILL_DESTINATION(2, cmd->dst, bc->pipe_id, bc->module_id); + cmd->command_id = bc->cmd_id; + memcpy(cmd->params, bc->params, bc->max); + + ret = sst_fill_and_send_cmd_unlocked(drv, SST_IPC_IA_SET_PARAMS, + SST_FLAG_BLOCKED, bc->task_id, 0, cmd, len); + kfree(cmd); + return ret; +} + +static int sst_algo_bytes_ctl_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct sst_algo_control *bc = (void *)kcontrol->private_value; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_BYTES; + uinfo->count = bc->max; + + return 0; +} + +static int sst_algo_control_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct sst_algo_control *bc = (void *)kcontrol->private_value; + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + + switch (bc->type) { + case SST_ALGO_PARAMS: + memcpy(ucontrol->value.bytes.data, bc->params, bc->max); + break; + default: + dev_err(component->dev, "Invalid Input- algo type:%d\n", + bc->type); + return -EINVAL; + + } + return 0; +} + +static int sst_algo_control_set(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + int ret = 0; + struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol); + struct sst_data *drv = snd_soc_component_get_drvdata(cmpnt); + struct sst_algo_control *bc = (void *)kcontrol->private_value; + + dev_dbg(cmpnt->dev, "control_name=%s\n", kcontrol->id.name); + mutex_lock(&drv->lock); + switch (bc->type) { + case SST_ALGO_PARAMS: + memcpy(bc->params, ucontrol->value.bytes.data, bc->max); + break; + default: + mutex_unlock(&drv->lock); + dev_err(cmpnt->dev, "Invalid Input- algo type:%d\n", + bc->type); + return -EINVAL; + } + /*if pipe is enabled, need to send the algo params from here*/ + if (bc->w && bc->w->power) + ret = sst_send_algo_cmd(drv, bc); + mutex_unlock(&drv->lock); + + return ret; +} + +static const struct snd_kcontrol_new sst_algo_controls[] = { + SST_ALGO_KCONTROL_BYTES("media_loop1_out", "fir", 272, SST_MODULE_ID_FIR_24, + SST_PATH_INDEX_MEDIA_LOOP1_OUT, 0, SST_TASK_SBA, SBA_VB_SET_FIR), + SST_ALGO_KCONTROL_BYTES("media_loop1_out", "iir", 300, SST_MODULE_ID_IIR_24, + SST_PATH_INDEX_MEDIA_LOOP1_OUT, 0, SST_TASK_SBA, SBA_VB_SET_IIR), + SST_ALGO_KCONTROL_BYTES("media_loop1_out", "mdrp", 286, SST_MODULE_ID_MDRP, + SST_PATH_INDEX_MEDIA_LOOP1_OUT, 0, SST_TASK_SBA, SBA_SET_MDRP), + SST_ALGO_KCONTROL_BYTES("media_loop2_out", "fir", 272, SST_MODULE_ID_FIR_24, + SST_PATH_INDEX_MEDIA_LOOP2_OUT, 0, SST_TASK_SBA, SBA_VB_SET_FIR), + SST_ALGO_KCONTROL_BYTES("media_loop2_out", "iir", 300, SST_MODULE_ID_IIR_24, + SST_PATH_INDEX_MEDIA_LOOP2_OUT, 0, SST_TASK_SBA, SBA_VB_SET_IIR), + SST_ALGO_KCONTROL_BYTES("media_loop2_out", "mdrp", 286, SST_MODULE_ID_MDRP, + SST_PATH_INDEX_MEDIA_LOOP2_OUT, 0, SST_TASK_SBA, SBA_SET_MDRP), + SST_ALGO_KCONTROL_BYTES("sprot_loop_out", "lpro", 192, SST_MODULE_ID_SPROT, + SST_PATH_INDEX_SPROT_LOOP_OUT, 0, SST_TASK_SBA, SBA_VB_LPRO), + SST_ALGO_KCONTROL_BYTES("codec_in0", "dcr", 52, SST_MODULE_ID_FILT_DCR, + SST_PATH_INDEX_CODEC_IN0, 0, SST_TASK_SBA, SBA_VB_SET_IIR), + SST_ALGO_KCONTROL_BYTES("codec_in1", "dcr", 52, SST_MODULE_ID_FILT_DCR, + SST_PATH_INDEX_CODEC_IN1, 0, SST_TASK_SBA, SBA_VB_SET_IIR), + +}; + +static int sst_algo_control_init(struct device *dev) +{ + int i = 0; + struct sst_algo_control *bc; + /*allocate space to cache the algo parameters in the driver*/ + for (i = 0; i < ARRAY_SIZE(sst_algo_controls); i++) { + bc = (struct sst_algo_control *)sst_algo_controls[i].private_value; + bc->params = devm_kzalloc(dev, bc->max, GFP_KERNEL); + if (bc->params == NULL) + return -ENOMEM; + } + return 0; +} + int sst_dsp_init_v2_dpcm(struct snd_soc_platform *platform) { int ret = 0; @@ -35,5 +208,11 @@ int sst_dsp_init_v2_dpcm(struct snd_soc_platform *platform) if (!drv->byte_stream) return -ENOMEM; + /*Initialize algo control params*/ + ret = sst_algo_control_init(platform->dev); + if (ret) + return ret; + ret = snd_soc_add_platform_controls(platform, sst_algo_controls, + ARRAY_SIZE(sst_algo_controls)); return ret; } diff --git a/sound/soc/intel/sst-atom-controls.h b/sound/soc/intel/sst-atom-controls.h index 8554889..a73e894 100644 --- a/sound/soc/intel/sst-atom-controls.h +++ b/sound/soc/intel/sst-atom-controls.h @@ -309,4 +309,134 @@ enum sst_swm_state { SST_SWM_ON = 3, }; +#define SST_FILL_LOCATION_IDS(dst, cell_idx, pipe_id) do { \ + dst.location_id.p.cell_nbr_idx = (cell_idx); \ + dst.location_id.p.path_id = (pipe_id); \ + } while (0) +#define SST_FILL_LOCATION_ID(dst, loc_id) (\ + dst.location_id.f = (loc_id)) +#define SST_FILL_MODULE_ID(dst, mod_id) (\ + dst.module_id = (mod_id)) + +#define SST_FILL_DESTINATION1(dst, id) do { \ + SST_FILL_LOCATION_ID(dst, (id) & 0xFFFF); \ + SST_FILL_MODULE_ID(dst, ((id) & 0xFFFF0000) >> 16); \ + } while (0) +#define SST_FILL_DESTINATION2(dst, loc_id, mod_id) do { \ + SST_FILL_LOCATION_ID(dst, loc_id); \ + SST_FILL_MODULE_ID(dst, mod_id); \ + } while (0) +#define SST_FILL_DESTINATION3(dst, cell_idx, path_id, mod_id) do { \ + SST_FILL_LOCATION_IDS(dst, cell_idx, path_id); \ + SST_FILL_MODULE_ID(dst, mod_id); \ + } while (0) + +#define SST_FILL_DESTINATION(level, dst, ...) \ + SST_FILL_DESTINATION##level(dst, __VA_ARGS__) +#define SST_FILL_DEFAULT_DESTINATION(dst) \ + SST_FILL_DESTINATION(2, dst, SST_DEFAULT_LOCATION_ID, SST_DEFAULT_MODULE_ID) + +struct sst_destination_id { + union sst_location_id { + struct { + u8 cell_nbr_idx; /* module index */ + u8 path_id; /* pipe_id */ + } __packed p; /* part */ + u16 f; /* full */ + } __packed location_id; + u16 module_id; +} __packed; +struct sst_dsp_header { + struct sst_destination_id dst; + u16 command_id; + u16 length; +} __packed; + +/* + * + * Common Commands + * + */ +struct sst_cmd_generic { + struct sst_dsp_header header; +} __packed; +struct sst_cmd_set_params { + struct sst_destination_id dst; + u16 command_id; + char params[0]; +} __packed; +#define SST_CONTROL_NAME(xpname, xmname, xinstance, xtype) \ + xpname " " xmname " " #xinstance " " xtype + +#define SST_COMBO_CONTROL_NAME(xpname, xmname, xinstance, xtype, xsubmodule) \ + xpname " " xmname " " #xinstance " " xtype " " xsubmodule +enum sst_algo_kcontrol_type { + SST_ALGO_PARAMS, + SST_ALGO_BYPASS, +}; + +struct sst_algo_control { + enum sst_algo_kcontrol_type type; + int max; + u16 module_id; + u16 pipe_id; + u16 task_id; + u16 cmd_id; + bool bypass; + unsigned char *params; + struct snd_soc_dapm_widget *w; +}; + +/* size of the control = size of params + size of length field */ +#define SST_ALGO_CTL_VALUE(xcount, xtype, xpipe, xmod, xtask, xcmd) \ + (struct sst_algo_control){ \ + .max = xcount + sizeof(u16), .type = xtype, .module_id = xmod, \ + .pipe_id = xpipe, .task_id = xtask, .cmd_id = xcmd, \ + } + +#define SST_ALGO_KCONTROL(xname, xcount, xmod, xpipe, \ + xtask, xcmd, xtype, xinfo, xget, xput) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .name = xname, \ + .info = xinfo, .get = xget, .put = xput, \ + .private_value = (unsigned long)& \ + SST_ALGO_CTL_VALUE(xcount, xtype, xpipe, \ + xmod, xtask, xcmd), \ +} + +#define SST_ALGO_KCONTROL_BYTES(xpname, xmname, xcount, xmod, \ + xpipe, xinstance, xtask, xcmd) \ + SST_ALGO_KCONTROL(SST_CONTROL_NAME(xpname, xmname, xinstance, "params"), \ + xcount, xmod, xpipe, xtask, xcmd, SST_ALGO_PARAMS, \ + sst_algo_bytes_ctl_info, \ + sst_algo_control_get, sst_algo_control_set) + +#define SST_ALGO_KCONTROL_BOOL(xpname, xmname, xmod, xpipe, xinstance, xtask) \ + SST_ALGO_KCONTROL(SST_CONTROL_NAME(xpname, xmname, xinstance, "bypass"), \ + 0, xmod, xpipe, xtask, 0, SST_ALGO_BYPASS, \ + snd_soc_info_bool_ext, \ + sst_algo_control_get, sst_algo_control_set) + +#define SST_ALGO_BYPASS_PARAMS(xpname, xmname, xcount, xmod, xpipe, \ + xinstance, xtask, xcmd) \ + SST_ALGO_KCONTROL_BOOL(xpname, xmname, xmod, xpipe, xinstance, xtask), \ + SST_ALGO_KCONTROL_BYTES(xpname, xmname, xcount, xmod, xpipe, xinstance, xtask, xcmd) + +#define SST_COMBO_ALGO_KCONTROL_BYTES(xpname, xmname, xsubmod, xcount, xmod, \ + xpipe, xinstance, xtask, xcmd) \ + SST_ALGO_KCONTROL(SST_COMBO_CONTROL_NAME(xpname, xmname, xinstance, "params", \ + xsubmod), \ + xcount, xmod, xpipe, xtask, xcmd, SST_ALGO_PARAMS, \ + sst_algo_bytes_ctl_info, \ + sst_algo_control_get, sst_algo_control_set) + + +struct sst_enum { + bool tx; + unsigned short reg; + unsigned int max; + const char * const *texts; + struct snd_soc_dapm_widget *w; +}; + #endif diff --git a/sound/soc/intel/sst-mfld-platform.h b/sound/soc/intel/sst-mfld-platform.h index 0c5b943d..7092ee3 100644 --- a/sound/soc/intel/sst-mfld-platform.h +++ b/sound/soc/intel/sst-mfld-platform.h @@ -166,7 +166,7 @@ struct sst_algo_int_control_v2 { struct sst_data { struct platform_device *pdev; struct sst_platform_data *pdata; - char *byte_stream; + struct snd_sst_bytes_v2 *byte_stream; struct mutex lock; }; int sst_register_dsp(struct sst_device *sst); -- cgit v1.1 From e306b6ee4d7ed7632765165749a36b8c8b4aeff2 Mon Sep 17 00:00:00 2001 From: Fabian Frederick Date: Tue, 16 Sep 2014 21:02:31 +0200 Subject: ASoC: cs35l32: remove second linux/slab.h inclusion linux/slab.h was included twice. Signed-off-by: Fabian Frederick Acked-by: Brian Austin Signed-off-by: Mark Brown --- sound/soc/codecs/cs35l32.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/cs35l32.c b/sound/soc/codecs/cs35l32.c index 76f628b..c125925 100644 --- a/sound/soc/codecs/cs35l32.c +++ b/sound/soc/codecs/cs35l32.c @@ -25,7 +25,6 @@ #include #include #include -#include #include #include #include -- cgit v1.1 From 6df5d768050f31d810dd3ba0ad8210922c3e9b6d Mon Sep 17 00:00:00 2001 From: "Subhransu S. Prusty" Date: Tue, 9 Sep 2014 15:11:32 +0530 Subject: ASoC: Intel: mrfld: Use snd_soc_dai_get_drvdata to derive drv data Signed-off-by: Subhransu S. Prusty Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/sst-mfld-platform-pcm.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/sst-mfld-platform-pcm.c b/sound/soc/intel/sst-mfld-platform-pcm.c index 9906b7c..8e1b2c1 100644 --- a/sound/soc/intel/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/sst-mfld-platform-pcm.c @@ -252,7 +252,7 @@ int sst_fill_stream_params(void *substream, } static int sst_platform_alloc_stream(struct snd_pcm_substream *substream, - struct snd_soc_platform *platform) + struct snd_soc_dai *dai) { struct sst_runtime_stream *stream = substream->runtime->private_data; @@ -260,7 +260,7 @@ static int sst_platform_alloc_stream(struct snd_pcm_substream *substream, struct snd_sst_params str_params = {0}; struct snd_sst_alloc_params_ext alloc_params = {0}; int ret_val = 0; - struct sst_data *ctx = snd_soc_platform_get_drvdata(platform); + struct sst_data *ctx = snd_soc_dai_get_drvdata(dai); /* set codec params and inform SST driver the same */ sst_fill_pcm_params(substream, ¶m); @@ -377,10 +377,10 @@ static void sst_media_close(struct snd_pcm_substream *substream, kfree(stream); } -static inline unsigned int get_current_pipe_id(struct snd_soc_platform *platform, +static inline unsigned int get_current_pipe_id(struct snd_soc_dai *dai, struct snd_pcm_substream *substream) { - struct sst_data *sst = snd_soc_platform_get_drvdata(platform); + struct sst_data *sst = snd_soc_dai_get_drvdata(dai); struct sst_dev_stream_map *map = sst->pdata->pdev_strm_map; struct sst_runtime_stream *stream = substream->runtime->private_data; @@ -389,7 +389,7 @@ static inline unsigned int get_current_pipe_id(struct snd_soc_platform *platform pipe_id = map[str_id].device_id; - dev_dbg(platform->dev, "got pipe_id = %#x for str_id = %d\n", + dev_dbg(dai->dev, "got pipe_id = %#x for str_id = %d\n", pipe_id, str_id); return pipe_id; } @@ -407,7 +407,7 @@ static int sst_media_prepare(struct snd_pcm_substream *substream, return ret_val; } - ret_val = sst_platform_alloc_stream(substream, dai->platform); + ret_val = sst_platform_alloc_stream(substream, dai); if (ret_val <= 0) return ret_val; snprintf(substream->pcm->id, sizeof(substream->pcm->id), -- cgit v1.1 From 5d5e63af998026f0340d1081fb15ad3c26d80c81 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Wed, 17 Sep 2014 20:58:02 +0800 Subject: ASoC: Remove return value checking for gpiochip_remove() gpiochip_remove() will return void eventually. Thus this patch removes return value checking for gpiochip_remove(). Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/rt5677.c | 5 +---- sound/soc/codecs/wm5100.c | 5 +---- sound/soc/codecs/wm8903.c | 6 +----- sound/soc/codecs/wm8962.c | 5 +---- sound/soc/codecs/wm8996.c | 6 +----- 5 files changed, 5 insertions(+), 22 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 02bc8bd..991409f 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -3274,11 +3274,8 @@ static void rt5677_init_gpio(struct i2c_client *i2c) static void rt5677_free_gpio(struct i2c_client *i2c) { struct rt5677_priv *rt5677 = i2c_get_clientdata(i2c); - int ret; - ret = gpiochip_remove(&rt5677->gpio_chip); - if (ret != 0) - dev_err(&i2c->dev, "Failed to remove GPIOs: %d\n", ret); + gpiochip_remove(&rt5677->gpio_chip); } #else static void rt5677_init_gpio(struct i2c_client *i2c) diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index 7bb0d36..a01ad62 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -2319,11 +2319,8 @@ static void wm5100_init_gpio(struct i2c_client *i2c) static void wm5100_free_gpio(struct i2c_client *i2c) { struct wm5100_priv *wm5100 = i2c_get_clientdata(i2c); - int ret; - ret = gpiochip_remove(&wm5100->gpio_chip); - if (ret != 0) - dev_err(&i2c->dev, "Failed to remove GPIOs: %d\n", ret); + gpiochip_remove(&wm5100->gpio_chip); } #else static void wm5100_init_gpio(struct i2c_client *i2c) diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index aa09848..c038b3e 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -1877,11 +1877,7 @@ static void wm8903_init_gpio(struct wm8903_priv *wm8903) static void wm8903_free_gpio(struct wm8903_priv *wm8903) { - int ret; - - ret = gpiochip_remove(&wm8903->gpio_chip); - if (ret != 0) - dev_err(wm8903->dev, "Failed to remove GPIOs: %d\n", ret); + gpiochip_remove(&wm8903->gpio_chip); } #else static void wm8903_init_gpio(struct wm8903_priv *wm8903) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 1098ae3..9077411 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3398,11 +3398,8 @@ static void wm8962_init_gpio(struct snd_soc_codec *codec) static void wm8962_free_gpio(struct snd_soc_codec *codec) { struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec); - int ret; - ret = gpiochip_remove(&wm8962->gpio_chip); - if (ret != 0) - dev_err(codec->dev, "Failed to remove GPIOs: %d\n", ret); + gpiochip_remove(&wm8962->gpio_chip); } #else static void wm8962_init_gpio(struct snd_soc_codec *codec) diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index f16ff4f..b1dcc11 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -2216,11 +2216,7 @@ static void wm8996_init_gpio(struct wm8996_priv *wm8996) static void wm8996_free_gpio(struct wm8996_priv *wm8996) { - int ret; - - ret = gpiochip_remove(&wm8996->gpio_chip); - if (ret != 0) - dev_err(wm8996->dev, "Failed to remove GPIOs: %d\n", ret); + gpiochip_remove(&wm8996->gpio_chip); } #else static void wm8996_init_gpio(struct wm8996_priv *wm8996) -- cgit v1.1 From 48561afef401876b4b0e35a303d89884c10fe468 Mon Sep 17 00:00:00 2001 From: Oder Chiou Date: Wed, 17 Sep 2014 15:12:33 +0800 Subject: ASoC: rt5677: Add the TDM function The patch adds the TDM function. Signed-off-by: Oder Chiou Signed-off-by: Mark Brown --- sound/soc/codecs/rt5677.c | 54 +++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 54 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 02bc8bd..1d4719f 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -3107,6 +3107,59 @@ static int rt5677_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source, return 0; } +static int rt5677_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, + unsigned int rx_mask, int slots, int slot_width) +{ + struct snd_soc_codec *codec = dai->codec; + unsigned int val = 0; + + if (rx_mask || tx_mask) + val |= (1 << 12); + + switch (slots) { + case 4: + val |= (1 << 10); + break; + case 6: + val |= (2 << 10); + break; + case 8: + val |= (3 << 10); + break; + case 2: + default: + break; + } + + switch (slot_width) { + case 20: + val |= (1 << 8); + break; + case 24: + val |= (2 << 8); + break; + case 32: + val |= (3 << 8); + break; + case 16: + default: + break; + } + + switch (dai->id) { + case RT5677_AIF1: + snd_soc_update_bits(codec, RT5677_TDM1_CTRL1, 0x1f00, val); + break; + case RT5677_AIF2: + snd_soc_update_bits(codec, RT5677_TDM2_CTRL1, 0x1f00, val); + break; + default: + break; + } + + return 0; +} + static int rt5677_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { @@ -3357,6 +3410,7 @@ static struct snd_soc_dai_ops rt5677_aif_dai_ops = { .set_fmt = rt5677_set_dai_fmt, .set_sysclk = rt5677_set_dai_sysclk, .set_pll = rt5677_set_dai_pll, + .set_tdm_slot = rt5677_set_tdm_slot, }; static struct snd_soc_dai_driver rt5677_dai[] = { -- cgit v1.1 From f4a43caba7d495699f98532b4faee90fd9980732 Mon Sep 17 00:00:00 2001 From: Shengjiu Wang Date: Tue, 16 Sep 2014 10:13:16 +0800 Subject: ASoC: fsl_ssi: refine ipg clock usage in this module Check if ipg clock is in clock-names property, then we can move the ipg clock enable and disable operation to startup and shutdown, that is only enable ipg clock when ssi is working and keep clock is disabled when ssi is in idle. But when the checking is failed, remain the clock control as before. Tested-by: Markus Pargmann Signed-off-by: Shengjiu Wang Acked-by: Nicolin Chen Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_ssi.c | 53 +++++++++++++++++++++++++++++++++++++++++-------- 1 file changed, 45 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 2fc3e66..16a1361 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -169,6 +169,7 @@ struct fsl_ssi_private { u8 i2s_mode; bool use_dma; bool use_dual_fifo; + bool has_ipg_clk_name; unsigned int fifo_depth; struct fsl_ssi_rxtx_reg_val rxtx_reg_val; @@ -530,6 +531,11 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(rtd->cpu_dai); + int ret; + + ret = clk_prepare_enable(ssi_private->clk); + if (ret) + return ret; /* When using dual fifo mode, it is safer to ensure an even period * size. If appearing to an odd number while DMA always starts its @@ -544,6 +550,21 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream, } /** + * fsl_ssi_shutdown: shutdown the SSI + * + */ +static void fsl_ssi_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct fsl_ssi_private *ssi_private = + snd_soc_dai_get_drvdata(rtd->cpu_dai); + + clk_disable_unprepare(ssi_private->clk); + +} + +/** * fsl_ssi_set_bclk - configure Digital Audio Interface bit clock * * Note: This function can be only called when using SSI as DAI master @@ -1043,6 +1064,7 @@ static int fsl_ssi_dai_probe(struct snd_soc_dai *dai) static const struct snd_soc_dai_ops fsl_ssi_dai_ops = { .startup = fsl_ssi_startup, + .shutdown = fsl_ssi_shutdown, .hw_params = fsl_ssi_hw_params, .hw_free = fsl_ssi_hw_free, .set_fmt = fsl_ssi_set_dai_fmt, @@ -1168,17 +1190,22 @@ static int fsl_ssi_imx_probe(struct platform_device *pdev, u32 dmas[4]; int ret; - ssi_private->clk = devm_clk_get(&pdev->dev, NULL); + if (ssi_private->has_ipg_clk_name) + ssi_private->clk = devm_clk_get(&pdev->dev, "ipg"); + else + ssi_private->clk = devm_clk_get(&pdev->dev, NULL); if (IS_ERR(ssi_private->clk)) { ret = PTR_ERR(ssi_private->clk); dev_err(&pdev->dev, "could not get clock: %d\n", ret); return ret; } - ret = clk_prepare_enable(ssi_private->clk); - if (ret) { - dev_err(&pdev->dev, "clk_prepare_enable failed: %d\n", ret); - return ret; + if (!ssi_private->has_ipg_clk_name) { + ret = clk_prepare_enable(ssi_private->clk); + if (ret) { + dev_err(&pdev->dev, "clk_prepare_enable failed: %d\n", ret); + return ret; + } } /* For those SLAVE implementations, we ingore non-baudclk cases @@ -1236,8 +1263,9 @@ static int fsl_ssi_imx_probe(struct platform_device *pdev, return 0; error_pcm: - clk_disable_unprepare(ssi_private->clk); + if (!ssi_private->has_ipg_clk_name) + clk_disable_unprepare(ssi_private->clk); return ret; } @@ -1246,7 +1274,8 @@ static void fsl_ssi_imx_clean(struct platform_device *pdev, { if (!ssi_private->use_dma) imx_pcm_fiq_exit(pdev); - clk_disable_unprepare(ssi_private->clk); + if (!ssi_private->has_ipg_clk_name) + clk_disable_unprepare(ssi_private->clk); } static int fsl_ssi_probe(struct platform_device *pdev) @@ -1321,8 +1350,16 @@ static int fsl_ssi_probe(struct platform_device *pdev) return -ENOMEM; } - ssi_private->regs = devm_regmap_init_mmio(&pdev->dev, iomem, + ret = of_property_match_string(np, "clock-names", "ipg"); + if (ret < 0) { + ssi_private->has_ipg_clk_name = false; + ssi_private->regs = devm_regmap_init_mmio(&pdev->dev, iomem, &fsl_ssi_regconfig); + } else { + ssi_private->has_ipg_clk_name = true; + ssi_private->regs = devm_regmap_init_mmio_clk(&pdev->dev, + "ipg", iomem, &fsl_ssi_regconfig); + } if (IS_ERR(ssi_private->regs)) { dev_err(&pdev->dev, "Failed to init register map\n"); return PTR_ERR(ssi_private->regs); -- cgit v1.1 From 8245b3634516e6b7eb1c94594c0fd41d233502aa Mon Sep 17 00:00:00 2001 From: Huacai Chen Date: Fri, 19 Sep 2014 14:57:02 +0800 Subject: ALSA: hda - Add fixup model name lookup for Lemote A1205 Lemote A1004 is already added in commit a2dd933d01f (ALSA: hda - Add fixup name lookup for CX5051 and 5066 codecs), but Lemote A1205 has missing. Signed-off-by: Huacai Chen Cc: # 3.15+ Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 6e5d0cb..47ccb8f 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -777,6 +777,7 @@ static const struct hda_model_fixup cxt5066_fixup_models[] = { { .id = CXT_PINCFG_LENOVO_TP410, .name = "tp410" }, { .id = CXT_FIXUP_THINKPAD_ACPI, .name = "thinkpad" }, { .id = CXT_PINCFG_LEMOTE_A1004, .name = "lemote-a1004" }, + { .id = CXT_PINCFG_LEMOTE_A1205, .name = "lemote-a1205" }, { .id = CXT_FIXUP_OLPC_XO, .name = "olpc-xo" }, {} }; -- cgit v1.1 From a9960e6a293e6fc3ed414643bb4e4106272e4d0a Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Sun, 21 Sep 2014 22:50:57 +0200 Subject: ALSA: pcm: fix fifo_size frame calculation The calculated frame size was wrong because snd_pcm_format_physical_width() actually returns the number of bits, not bytes. Use snd_pcm_format_size() instead, which not only returns bytes, but also simplifies the calculation. Fixes: 8bea869c5e56 ("ALSA: PCM midlevel: improve fifo_size handling") Signed-off-by: Clemens Ladisch Cc: Signed-off-by: Takashi Iwai --- sound/core/pcm_lib.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 9acc77e..0032278 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -1782,14 +1782,16 @@ static int snd_pcm_lib_ioctl_fifo_size(struct snd_pcm_substream *substream, { struct snd_pcm_hw_params *params = arg; snd_pcm_format_t format; - int channels, width; + int channels; + ssize_t frame_size; params->fifo_size = substream->runtime->hw.fifo_size; if (!(substream->runtime->hw.info & SNDRV_PCM_INFO_FIFO_IN_FRAMES)) { format = params_format(params); channels = params_channels(params); - width = snd_pcm_format_physical_width(format); - params->fifo_size /= width * channels; + frame_size = snd_pcm_format_size(format, channels); + if (frame_size > 0) + params->fifo_size /= (unsigned)frame_size; } return 0; } -- cgit v1.1 From e76bf634870e3c5e3a767ad575f1d404c9f1cab8 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Sun, 21 Sep 2014 23:55:38 +0200 Subject: ALSA: snd-usb-caiaq: Fix LED commands for Kore controller KoreController and KoreController2 need an EP1_CMD_DIMM_LEDS command to set their LEDs, not EP1_CMD_WRITE_IO. Signed-off-by: Daniel Mack Reported-and-tested-by: Brad Wilson Signed-off-by: Takashi Iwai --- sound/usb/caiaq/control.c | 18 +++++++++++------- 1 file changed, 11 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/usb/caiaq/control.c b/sound/usb/caiaq/control.c index f65fc09..b7a7c80 100644 --- a/sound/usb/caiaq/control.c +++ b/sound/usb/caiaq/control.c @@ -100,15 +100,19 @@ static int control_put(struct snd_kcontrol *kcontrol, struct snd_usb_caiaqdev *cdev = caiaqdev(chip->card); int pos = kcontrol->private_value; int v = ucontrol->value.integer.value[0]; - unsigned char cmd = EP1_CMD_WRITE_IO; + unsigned char cmd; - if (cdev->chip.usb_id == - USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_TRAKTORKONTROLX1)) - cmd = EP1_CMD_DIMM_LEDS; - - if (cdev->chip.usb_id == - USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_MASCHINECONTROLLER)) + switch (cdev->chip.usb_id) { + case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_MASCHINECONTROLLER): + case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_TRAKTORKONTROLX1): + case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_KORECONTROLLER2): + case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_KORECONTROLLER): cmd = EP1_CMD_DIMM_LEDS; + break; + default: + cmd = EP1_CMD_WRITE_IO; + break; + } if (pos & CNT_INTVAL) { int i = pos & ~CNT_INTVAL; -- cgit v1.1 From 90bdbb46f41c9fa670d7b0709e0c8a92ad82bdfe Mon Sep 17 00:00:00 2001 From: Oder Chiou Date: Thu, 18 Sep 2014 14:45:59 +0800 Subject: ASoC: rt5677: Add sidetone function Add sidetone function Signed-off-by: Anatol Pomozov Signed-off-by: Oder Chiou Signed-off-by: Mark Brown --- sound/soc/codecs/rt5677.c | 9 +++++++++ sound/soc/codecs/rt5677.h | 4 ++++ 2 files changed, 13 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 1d4719f..4a0f3df 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -541,6 +541,7 @@ static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -65625, 375, 0); static const DECLARE_TLV_DB_SCALE(in_vol_tlv, -3450, 150, 0); static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -17625, 375, 0); static const DECLARE_TLV_DB_SCALE(adc_bst_tlv, 0, 1200, 0); +static const DECLARE_TLV_DB_SCALE(st_vol_tlv, -4650, 150, 0); /* {0, +20, +24, +30, +35, +40, +44, +50, +52} dB */ static unsigned int bst_tlv[] = { @@ -605,6 +606,10 @@ static const struct snd_kcontrol_new rt5677_snd_controls[] = { RT5677_MONO_ADC_L_VOL_SFT, RT5677_MONO_ADC_R_VOL_SFT, 127, 0, adc_vol_tlv), + /* Sidetone Control */ + SOC_SINGLE_TLV("Sidetone Volume", RT5677_SIDETONE_CTRL, + RT5677_ST_VOL_SFT, 31, 0, st_vol_tlv), + /* ADC Boost Volume Control */ SOC_DOUBLE_TLV("STO1 ADC Boost Volume", RT5677_STO1_2_ADC_BST, RT5677_STO1_ADC_L_BST_SFT, RT5677_STO1_ADC_R_BST_SFT, 3, 0, @@ -1993,6 +1998,9 @@ static const struct snd_soc_dapm_widget rt5677_dapm_widgets[] = { /* Sidetone Mux */ SND_SOC_DAPM_MUX("Sidetone Mux", SND_SOC_NOPM, 0, 0, &rt5677_sidetone_mux), + SND_SOC_DAPM_SUPPLY("Sidetone Power", RT5677_SIDETONE_CTRL, + RT5677_ST_EN_SFT, 0, NULL, 0), + /* VAD Mux*/ SND_SOC_DAPM_MUX("VAD ADC Mux", SND_SOC_NOPM, 0, 0, &rt5677_vad_src_mux), @@ -2704,6 +2712,7 @@ static const struct snd_soc_dapm_route rt5677_dapm_routes[] = { { "Sidetone Mux", "DMIC4 L", "DMIC L4" }, { "Sidetone Mux", "ADC1", "ADC 1" }, { "Sidetone Mux", "ADC2", "ADC 2" }, + { "Sidetone Mux", NULL, "Sidetone Power" }, { "Stereo DAC MIXL", "ST L Switch", "Sidetone Mux" }, { "Stereo DAC MIXL", "DAC1 L Switch", "DAC1 MIXL" }, diff --git a/sound/soc/codecs/rt5677.h b/sound/soc/codecs/rt5677.h index b61b72c..1fe8872 100644 --- a/sound/soc/codecs/rt5677.h +++ b/sound/soc/codecs/rt5677.h @@ -382,6 +382,10 @@ #define RT5677_ST_SEL_SFT 9 #define RT5677_ST_EN (0x1 << 6) #define RT5677_ST_EN_SFT 6 +#define RT5677_ST_GAIN (0x1 << 5) +#define RT5677_ST_GAIN_SFT 5 +#define RT5677_ST_VOL_MASK (0x1f << 0) +#define RT5677_ST_VOL_SFT 0 /* Analog DAC1/2/3 Source Control (0x15) */ #define RT5677_ANA_DAC3_SRC_SEL_MASK (0x3 << 4) -- cgit v1.1 From e3f205a72c4554b58f51d5afd98195c4ff54d215 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 23 Sep 2014 00:56:28 +0200 Subject: ASoC: Remove locking in snd_soc_{new,free}_ac97_codec() snd_soc_new_ac97_codec() and snd_soc_free_ac97_codec() are called from within a CODEC's probe() and remove() callbacks. Those will not run concurrently against each other for the same CODEC instance, hence it is not necessary to protect the two functions with a mutex. This removes the last user in the ASoC core of the snd_soc_codec mutex field and will allow us to eventually remove it. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 9 +-------- 1 file changed, 1 insertion(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index d4bfd4a..a504cf4 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2107,13 +2107,9 @@ static struct platform_driver soc_driver = { int snd_soc_new_ac97_codec(struct snd_soc_codec *codec, struct snd_ac97_bus_ops *ops, int num) { - mutex_lock(&codec->mutex); - codec->ac97 = kzalloc(sizeof(struct snd_ac97), GFP_KERNEL); - if (codec->ac97 == NULL) { - mutex_unlock(&codec->mutex); + if (codec->ac97 == NULL) return -ENOMEM; - } codec->ac97->bus = kzalloc(sizeof(struct snd_ac97_bus), GFP_KERNEL); if (codec->ac97->bus == NULL) { @@ -2132,7 +2128,6 @@ int snd_soc_new_ac97_codec(struct snd_soc_codec *codec, */ codec->ac97_created = 1; - mutex_unlock(&codec->mutex); return 0; } EXPORT_SYMBOL_GPL(snd_soc_new_ac97_codec); @@ -2302,7 +2297,6 @@ EXPORT_SYMBOL_GPL(snd_soc_set_ac97_ops_of_reset); */ void snd_soc_free_ac97_codec(struct snd_soc_codec *codec) { - mutex_lock(&codec->mutex); #ifdef CONFIG_SND_SOC_AC97_BUS soc_unregister_ac97_codec(codec); #endif @@ -2310,7 +2304,6 @@ void snd_soc_free_ac97_codec(struct snd_soc_codec *codec) kfree(codec->ac97); codec->ac97 = NULL; codec->ac97_created = 0; - mutex_unlock(&codec->mutex); } EXPORT_SYMBOL_GPL(snd_soc_free_ac97_codec); -- cgit v1.1 From 85151461f114f2fca386bb8ae6de185461d35d87 Mon Sep 17 00:00:00 2001 From: Michael Trimarchi Date: Thu, 18 Sep 2014 20:38:09 +0200 Subject: ASoC: fsl_ssi: fix kernel panic in probe function code can raise a panic when the ssi_private->pdev is null [...] /* * If codec-handle property is missing from SSI node, we assume * that the machine driver uses new binding which does not require * SSI driver to trigger machine driver's probe. */ if (!of_get_property(np, "codec-handle", NULL)) goto done; [...] ssi_private->pdev = platform_device_register_data(&pdev->dev, name, 0, NULL, 0); [...] done: if (ssi_private->dai_fmt) _fsl_ssi_set_dai_fmt(ssi_private, ssi_private->dai_fmt); Proposal was to not use ssi_private->pdev->dev here but adding a new parameter of *dev pointer to this _set_dai_fmt() -- passing pdev->dev in probe() and cpu_dai->dev in fsl_ssi_set_dai_fmt(). Signed-off-by: Michael Trimarchi Reported-by: Jean-Michel Hautbois Acked-by: Nicolin Chen Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_ssi.c | 12 +++++++----- 1 file changed, 7 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 87eb577..de6ab06 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -748,8 +748,9 @@ static int fsl_ssi_hw_free(struct snd_pcm_substream *substream, return 0; } -static int _fsl_ssi_set_dai_fmt(struct fsl_ssi_private *ssi_private, - unsigned int fmt) +static int _fsl_ssi_set_dai_fmt(struct device *dev, + struct fsl_ssi_private *ssi_private, + unsigned int fmt) { struct regmap *regs = ssi_private->regs; u32 strcr = 0, stcr, srcr, scr, mask; @@ -758,7 +759,7 @@ static int _fsl_ssi_set_dai_fmt(struct fsl_ssi_private *ssi_private, ssi_private->dai_fmt = fmt; if (fsl_ssi_is_i2s_master(ssi_private) && IS_ERR(ssi_private->baudclk)) { - dev_err(&ssi_private->pdev->dev, "baudclk is missing which is necessary for master mode\n"); + dev_err(dev, "baudclk is missing which is necessary for master mode\n"); return -EINVAL; } @@ -913,7 +914,7 @@ static int fsl_ssi_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) { struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(cpu_dai); - return _fsl_ssi_set_dai_fmt(ssi_private, fmt); + return _fsl_ssi_set_dai_fmt(cpu_dai->dev, ssi_private, fmt); } /** @@ -1387,7 +1388,8 @@ static int fsl_ssi_probe(struct platform_device *pdev) done: if (ssi_private->dai_fmt) - _fsl_ssi_set_dai_fmt(ssi_private, ssi_private->dai_fmt); + _fsl_ssi_set_dai_fmt(&pdev->dev, ssi_private, + ssi_private->dai_fmt); return 0; -- cgit v1.1 From 8c8f2f6fc1c8eec9e14810f21386fe295a42a40f Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 23 Sep 2014 04:15:48 +0200 Subject: ASoC: Fix snd_soc_{new,free}_ac97_codec() locking removal Commit e3f205a72c45 ("ASoC: Remove locking in snd_soc_{new,free}_ac97_codec()") overlooked a unlock on one of the error paths. Fixes: e3f205a72c45 ("ASoC: Remove locking in snd_soc_{new,free}_ac97_codec()") Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index a504cf4..3c57f5c 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2115,7 +2115,6 @@ int snd_soc_new_ac97_codec(struct snd_soc_codec *codec, if (codec->ac97->bus == NULL) { kfree(codec->ac97); codec->ac97 = NULL; - mutex_unlock(&codec->mutex); return -ENOMEM; } -- cgit v1.1 From 5c7c343a1159d1cb7604b6137cf547b2c1e2375d Mon Sep 17 00:00:00 2001 From: Howard Mitchell Date: Fri, 19 Sep 2014 12:50:31 +0100 Subject: ASoC: core: Fix volsw_range funcs so SOC_DOUBLE_R_RANGE_TLV works. This fixes a bug when using the SOC_DOUBLE_R_RANGE_TLV macro in the invert mode. In the non-invert case, e.g. SOC_DOUBLE_R_RANGE_TLV("", , , 0, 40, 255, 0, ) the range sent to the hardware is 40..255, but in the invert case: SOC_DOUBLE_R_RANGE_TLV("", , , 0, 40, 255, 1, ) the range 215..0 was being sent to the hardware. This commit corrects this to 255..40 so it is consistent with the non-invert case. Signed-off-by: Howard Mitchell Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 20 ++++++++++++-------- 1 file changed, 12 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 3c57f5c..dde4b82 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3019,9 +3019,10 @@ int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol, unsigned int val, val_mask; int ret; - val = ((ucontrol->value.integer.value[0] + min) & mask); if (invert) - val = max - val; + val = (max - ucontrol->value.integer.value[0]) & mask; + else + val = ((ucontrol->value.integer.value[0] + min) & mask); val_mask = mask << shift; val = val << shift; @@ -3030,9 +3031,10 @@ int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol, return ret; if (snd_soc_volsw_is_stereo(mc)) { - val = ((ucontrol->value.integer.value[1] + min) & mask); if (invert) - val = max - val; + val = (max - ucontrol->value.integer.value[1]) & mask; + else + val = ((ucontrol->value.integer.value[1] + min) & mask); val_mask = mask << shift; val = val << shift; @@ -3077,8 +3079,9 @@ int snd_soc_get_volsw_range(struct snd_kcontrol *kcontrol, if (invert) ucontrol->value.integer.value[0] = max - ucontrol->value.integer.value[0]; - ucontrol->value.integer.value[0] = - ucontrol->value.integer.value[0] - min; + else + ucontrol->value.integer.value[0] = + ucontrol->value.integer.value[0] - min; if (snd_soc_volsw_is_stereo(mc)) { ret = snd_soc_component_read(component, rreg, &val); @@ -3089,8 +3092,9 @@ int snd_soc_get_volsw_range(struct snd_kcontrol *kcontrol, if (invert) ucontrol->value.integer.value[1] = max - ucontrol->value.integer.value[1]; - ucontrol->value.integer.value[1] = - ucontrol->value.integer.value[1] - min; + else + ucontrol->value.integer.value[1] = + ucontrol->value.integer.value[1] - min; } return 0; -- cgit v1.1 From 7a7f0ba03d521ac2d36c9015278bc35657b3dcc9 Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Fri, 19 Sep 2014 14:48:17 +0300 Subject: ASoC: max98090: Move interrupt request from codec probe to i2c probe Keep MAX98090 interrupt requested after i2c device probing as long as the driver is loaded. This fixes the issue where subsequent codec probe max98090_probe() call fails in interrupt request since interrupt wasn't freed over codec remove-reprobe cycle. Signed-off-by: Jarkko Nikula Signed-off-by: Mark Brown --- sound/soc/codecs/max98090.c | 29 ++++++++++++++++------------- 1 file changed, 16 insertions(+), 13 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index f154365..fe77df6 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -2159,12 +2159,16 @@ static void max98090_jack_work(struct work_struct *work) static irqreturn_t max98090_interrupt(int irq, void *data) { - struct snd_soc_codec *codec = data; - struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); + struct max98090_priv *max98090 = data; + struct snd_soc_codec *codec = max98090->codec; int ret; unsigned int mask; unsigned int active; + /* Treat interrupt before codec is initialized as spurious */ + if (codec == NULL) + return IRQ_NONE; + dev_dbg(codec->dev, "***** max98090_interrupt *****\n"); ret = regmap_read(max98090->regmap, M98090_REG_INTERRUPT_S, &mask); @@ -2367,17 +2371,6 @@ static int max98090_probe(struct snd_soc_codec *codec) snd_soc_write(codec, M98090_REG_JACK_DETECT, M98090_JDETEN_MASK | M98090_JDEB_25MS); - /* Register for interrupts */ - dev_dbg(codec->dev, "irq = %d\n", max98090->irq); - - ret = devm_request_threaded_irq(codec->dev, max98090->irq, NULL, - max98090_interrupt, IRQF_TRIGGER_FALLING | IRQF_ONESHOT, - "max98090_interrupt", codec); - if (ret < 0) { - dev_err(codec->dev, "request_irq failed: %d\n", - ret); - } - /* * Clear any old interrupts. * An old interrupt ocurring prior to installing the ISR @@ -2417,6 +2410,7 @@ static int max98090_remove(struct snd_soc_codec *codec) cancel_delayed_work_sync(&max98090->pll_det_enable_work); cancel_work_sync(&max98090->pll_det_disable_work); cancel_work_sync(&max98090->pll_work); + max98090->codec = NULL; return 0; } @@ -2478,6 +2472,15 @@ static int max98090_i2c_probe(struct i2c_client *i2c, goto err_enable; } + ret = devm_request_threaded_irq(&i2c->dev, max98090->irq, NULL, + max98090_interrupt, IRQF_TRIGGER_FALLING | IRQF_ONESHOT, + "max98090_interrupt", max98090); + if (ret < 0) { + dev_err(&i2c->dev, "request_irq failed: %d\n", + ret); + return ret; + } + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_max98090, max98090_dai, ARRAY_SIZE(max98090_dai)); -- cgit v1.1 From ced1933db67087554abf22bcb285eb6873380b10 Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Fri, 19 Sep 2014 14:48:18 +0300 Subject: ASoC: max98090: Remove structure member irq from private data struct max98090_priv member irq is now used only locally in max98090_i2c_probe() and can be removed. Signed-off-by: Jarkko Nikula Signed-off-by: Mark Brown --- sound/soc/codecs/max98090.c | 3 +-- sound/soc/codecs/max98090.h | 1 - 2 files changed, 1 insertion(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index fe77df6..3e27de1 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -2463,7 +2463,6 @@ static int max98090_i2c_probe(struct i2c_client *i2c, max98090->devtype = driver_data; i2c_set_clientdata(i2c, max98090); max98090->pdata = i2c->dev.platform_data; - max98090->irq = i2c->irq; max98090->regmap = devm_regmap_init_i2c(i2c, &max98090_regmap); if (IS_ERR(max98090->regmap)) { @@ -2472,7 +2471,7 @@ static int max98090_i2c_probe(struct i2c_client *i2c, goto err_enable; } - ret = devm_request_threaded_irq(&i2c->dev, max98090->irq, NULL, + ret = devm_request_threaded_irq(&i2c->dev, i2c->irq, NULL, max98090_interrupt, IRQF_TRIGGER_FALLING | IRQF_ONESHOT, "max98090_interrupt", max98090); if (ret < 0) { diff --git a/sound/soc/codecs/max98090.h b/sound/soc/codecs/max98090.h index 14427a5..a163195 100644 --- a/sound/soc/codecs/max98090.h +++ b/sound/soc/codecs/max98090.h @@ -1529,7 +1529,6 @@ struct max98090_priv { unsigned int bclk; unsigned int lrclk; struct max98090_cdata dai[1]; - int irq; int jack_state; struct delayed_work jack_work; struct delayed_work pll_det_enable_work; -- cgit v1.1 From 3256ff6e5117c493ec20e96aad9f0a20d656d561 Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Fri, 19 Sep 2014 14:48:19 +0300 Subject: ASoC: max98090: Remove structure member extmic_mux from private data There is no other use for extmic_mux than setting it to zero so remove it. Signed-off-by: Jarkko Nikula Signed-off-by: Mark Brown --- sound/soc/codecs/max98090.c | 1 - sound/soc/codecs/max98090.h | 1 - 2 files changed, 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index 3e27de1..f2a3f30 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -2333,7 +2333,6 @@ static int max98090_probe(struct snd_soc_codec *codec) max98090->lin_state = 0; max98090->pa1en = 0; max98090->pa2en = 0; - max98090->extmic_mux = 0; ret = snd_soc_read(codec, M98090_REG_REVISION_ID); if (ret < 0) { diff --git a/sound/soc/codecs/max98090.h b/sound/soc/codecs/max98090.h index a163195..84ca3f4 100644 --- a/sound/soc/codecs/max98090.h +++ b/sound/soc/codecs/max98090.h @@ -1541,7 +1541,6 @@ struct max98090_priv { u8 lin_state; unsigned int pa1en; unsigned int pa2en; - unsigned int extmic_mux; unsigned int sidetone; bool master; }; -- cgit v1.1 From 0e2cadf39a37f633d3b6d286318506ea3bd0b286 Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Fri, 19 Sep 2014 14:48:20 +0300 Subject: ASoC: max98090: Remove unused version define Signed-off-by: Jarkko Nikula Signed-off-by: Mark Brown --- sound/soc/codecs/max98090.h | 5 ----- 1 file changed, 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/max98090.h b/sound/soc/codecs/max98090.h index 84ca3f4..2613fdb 100644 --- a/sound/soc/codecs/max98090.h +++ b/sound/soc/codecs/max98090.h @@ -11,11 +11,6 @@ #ifndef _MAX98090_H #define _MAX98090_H -#include - -/* One can override the Linux version here with an explicit version number */ -#define M98090_LINUX_VERSION LINUX_VERSION_CODE - /* * MAX98090 Register Definitions */ -- cgit v1.1 From 99632d1077853c2030bec3530011b9d9f423cc89 Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Fri, 19 Sep 2014 14:48:21 +0300 Subject: ASoC: max98090: Remove unused byte access macros Signed-off-by: Jarkko Nikula Signed-off-by: Mark Brown --- sound/soc/codecs/max98090.h | 3 --- 1 file changed, 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/max98090.h b/sound/soc/codecs/max98090.h index 2613fdb..a5f6bad 100644 --- a/sound/soc/codecs/max98090.h +++ b/sound/soc/codecs/max98090.h @@ -1497,9 +1497,6 @@ #define M98090_REVID_WIDTH 8 #define M98090_REVID_NUM (1<> 8) & 0xff) -#define M98090_BYTE0(w) (w & 0xff) - /* Silicon revision number */ #define M98090_REVA 0x40 #define M98091_REVA 0x50 -- cgit v1.1 From f9f6a592cf4f35e7b614f1fb2e8d73969ee39a6d Mon Sep 17 00:00:00 2001 From: Anatol Pomozov Date: Wed, 17 Sep 2014 13:14:20 -0700 Subject: ASoC: rt5677: Add a configuration option for LDO2_POW pin Some boards have this pin statically tied and do not require any configuration, some other boards allow to enable chip using GPIO. Add an option that tells which GPIO is used to power the audio codec. Signed-off-by: Anatol Pomozov Signed-off-by: Mark Brown --- sound/soc/codecs/rt5677.c | 54 +++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/rt5677.h | 1 + 2 files changed, 55 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 4a0f3df..d2c6abf 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -15,6 +15,7 @@ #include #include #include +#include #include #include #include @@ -3381,6 +3382,8 @@ static int rt5677_remove(struct snd_soc_codec *codec) struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); regmap_write(rt5677->regmap, RT5677_RESET, 0x10ec); + if (gpio_is_valid(rt5677->pow_ldo2)) + gpio_set_value_cansleep(rt5677->pow_ldo2, 0); return 0; } @@ -3392,6 +3395,8 @@ static int rt5677_suspend(struct snd_soc_codec *codec) regcache_cache_only(rt5677->regmap, true); regcache_mark_dirty(rt5677->regmap); + if (gpio_is_valid(rt5677->pow_ldo2)) + gpio_set_value_cansleep(rt5677->pow_ldo2, 0); return 0; } @@ -3400,6 +3405,10 @@ static int rt5677_resume(struct snd_soc_codec *codec) { struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); + if (gpio_is_valid(rt5677->pow_ldo2)) { + gpio_set_value_cansleep(rt5677->pow_ldo2, 1); + msleep(10); + } regcache_cache_only(rt5677->regmap, false); regcache_sync(rt5677->regmap); @@ -3558,6 +3567,24 @@ static const struct i2c_device_id rt5677_i2c_id[] = { }; MODULE_DEVICE_TABLE(i2c, rt5677_i2c_id); +static int rt5677_parse_dt(struct rt5677_priv *rt5677, struct device_node *np) +{ + rt5677->pow_ldo2 = of_get_named_gpio(np, + "realtek,pow-ldo2-gpio", 0); + + /* + * POW_LDO2 is optional (it may be statically tied on the board). + * -ENOENT means that the property doesn't exist, i.e. there is no + * GPIO, so is not an error. Any other error code means the property + * exists, but could not be parsed. + */ + if (!gpio_is_valid(rt5677->pow_ldo2) && + (rt5677->pow_ldo2 != -ENOENT)) + return rt5677->pow_ldo2; + + return 0; +} + static int rt5677_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -3576,6 +3603,33 @@ static int rt5677_i2c_probe(struct i2c_client *i2c, if (pdata) rt5677->pdata = *pdata; + if (i2c->dev.of_node) { + ret = rt5677_parse_dt(rt5677, i2c->dev.of_node); + if (ret) { + dev_err(&i2c->dev, "Failed to parse device tree: %d\n", + ret); + return ret; + } + } else { + rt5677->pow_ldo2 = -EINVAL; + } + + if (gpio_is_valid(rt5677->pow_ldo2)) { + ret = devm_gpio_request_one(&i2c->dev, rt5677->pow_ldo2, + GPIOF_OUT_INIT_HIGH, + "RT5677 POW_LDO2"); + if (ret < 0) { + dev_err(&i2c->dev, "Failed to request POW_LDO2 %d: %d\n", + rt5677->pow_ldo2, ret); + return ret; + } + /* Wait a while until I2C bus becomes available. The datasheet + * does not specify the exact we should wait but startup + * sequence mentiones at least a few milliseconds. + */ + msleep(10); + } + rt5677->regmap = devm_regmap_init_i2c(i2c, &rt5677_regmap); if (IS_ERR(rt5677->regmap)) { ret = PTR_ERR(rt5677->regmap); diff --git a/sound/soc/codecs/rt5677.h b/sound/soc/codecs/rt5677.h index 1fe8872..d4eb6d5 100644 --- a/sound/soc/codecs/rt5677.h +++ b/sound/soc/codecs/rt5677.h @@ -1554,6 +1554,7 @@ struct rt5677_priv { int pll_src; int pll_in; int pll_out; + int pow_ldo2; /* POW_LDO2 pin */ #ifdef CONFIG_GPIOLIB struct gpio_chip gpio_chip; #endif -- cgit v1.1 From 2e4ec1c0b8791f3556576a19cf56941c3d2a90fc Mon Sep 17 00:00:00 2001 From: Qiao Zhou Date: Fri, 12 Sep 2014 09:41:31 +0800 Subject: ASoC: soc-compress: fix double unlock of fe card mutex Fix double unlock of fe card mutex introduced by patch 8f70e515a8bb "ASoC: soc-pcm: fix dpcm_path_get error handling" The first unlock is at line 106, and the unlock is at line 149. we should remove the first unlock. Reported-by: Dan Carpenter Signed-off-by: Qiao Zhou Signed-off-by: Mark Brown --- sound/soc/soc-compress.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 3092b58f..cecfab3c 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -102,13 +102,11 @@ static int soc_compr_open_fe(struct snd_compr_stream *cstream) fe->dpcm[stream].runtime = fe_substream->runtime; ret = dpcm_path_get(fe, stream, &list); - if (ret < 0) { - mutex_unlock(&fe->card->mutex); + if (ret < 0) goto fe_err; - } else if (ret == 0) { + else if (ret == 0) dev_dbg(fe->dev, "ASoC: %s no valid %s route\n", fe->dai_link->name, stream ? "capture" : "playback"); - } /* calculate valid and active FE <-> BE dpcms */ dpcm_process_paths(fe, stream, &list, 1); -- cgit v1.1 From 0121327c1a68bc8c80f240c2794e682722b69051 Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Fri, 19 Sep 2014 16:46:03 +0530 Subject: ASoC: Intel: mfld-pcm: add control for powering up/down dsp When we have PCM (FE/BE) opened or DAPM widgets triggered we need power up/down DSP accordingly. The DSP will do ref count of these requests i.e. link these runtime_get/put calls of DSP Also fix some preexisting spacing error. Signed-off-by: Vinod Koul Signed-off-by: Subhransu S. Prusty Signed-off-by: Mark Brown --- sound/soc/intel/sst-mfld-platform-pcm.c | 16 ++++++++++++++++ sound/soc/intel/sst-mfld-platform.h | 17 +++++++++-------- 2 files changed, 25 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/sst-mfld-platform-pcm.c b/sound/soc/intel/sst-mfld-platform-pcm.c index 8e1b2c1..aa9b600 100644 --- a/sound/soc/intel/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/sst-mfld-platform-pcm.c @@ -322,6 +322,16 @@ static int sst_platform_init_stream(struct snd_pcm_substream *substream) } +static int power_up_sst(struct sst_runtime_stream *stream) +{ + return stream->ops->power(sst->dev, true); +} + +static void power_down_sst(struct sst_runtime_stream *stream) +{ + stream->ops->power(sst->dev, false); +} + static int sst_media_open(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { @@ -351,6 +361,10 @@ static int sst_media_open(struct snd_pcm_substream *substream, /* allocate memory for SST API set */ runtime->private_data = stream; + ret_val = power_up_sst(stream); + if (ret_val < 0) + return ret_val; + /* Make sure, that the period size is always even */ snd_pcm_hw_constraint_step(substream->runtime, 0, SNDRV_PCM_HW_PARAM_PERIODS, 2); @@ -370,6 +384,8 @@ static void sst_media_close(struct snd_pcm_substream *substream, int ret_val = 0, str_id; stream = substream->runtime->private_data; + power_down_sst(stream); + str_id = stream->stream_info.str_id; if (str_id) ret_val = stream->ops->close(sst->dev, str_id); diff --git a/sound/soc/intel/sst-mfld-platform.h b/sound/soc/intel/sst-mfld-platform.h index 7092ee3..19f83ec 100644 --- a/sound/soc/intel/sst-mfld-platform.h +++ b/sound/soc/intel/sst-mfld-platform.h @@ -120,15 +120,16 @@ struct compress_sst_ops { }; struct sst_ops { - int (*open) (struct device *dev, struct snd_sst_params *str_param); - int (*stream_init) (struct device *dev, struct pcm_stream_info *str_info); - int (*stream_start) (struct device *dev, int str_id); - int (*stream_drop) (struct device *dev, int str_id); - int (*stream_pause) (struct device *dev, int str_id); - int (*stream_pause_release) (struct device *dev, int str_id); - int (*stream_read_tstamp) (struct device *dev, struct pcm_stream_info *str_info); + int (*open)(struct device *dev, struct snd_sst_params *str_param); + int (*stream_init)(struct device *dev, struct pcm_stream_info *str_info); + int (*stream_start)(struct device *dev, int str_id); + int (*stream_drop)(struct device *dev, int str_id); + int (*stream_pause)(struct device *dev, int str_id); + int (*stream_pause_release)(struct device *dev, int str_id); + int (*stream_read_tstamp)(struct device *dev, struct pcm_stream_info *str_info); int (*send_byte_stream)(struct device *dev, struct snd_sst_bytes_v2 *bytes); - int (*close) (struct device *dev, unsigned int str_id); + int (*close)(struct device *dev, unsigned int str_id); + int (*power)(struct device *dev, bool state); }; struct sst_runtime_stream { -- cgit v1.1 From 83a7fc98dc9c29c5d2d66c80fb50725303a78192 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 25 Sep 2014 16:19:30 +0200 Subject: ASoC: wm8741: Remove unused wm8741_suspend define This driver has no suspend callback. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/wm8741.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c index a237f16..31bb480 100644 --- a/sound/soc/codecs/wm8741.c +++ b/sound/soc/codecs/wm8741.c @@ -413,7 +413,6 @@ static int wm8741_resume(struct snd_soc_codec *codec) return 0; } #else -#define wm8741_suspend NULL #define wm8741_resume NULL #endif -- cgit v1.1 From b7a297677540789b8fb35a6ce66c500739fb4bf9 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Fri, 26 Sep 2014 11:06:39 +0800 Subject: ASoC: rt286: Correct default value This patch corrects some incorrect default value in the cache. Signed-off-by: Bard Liao Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/rt286.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index e4f6102..7a66084 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -51,7 +51,7 @@ static struct reg_default rt286_index_def[] = { { 0x04, 0xaf01 }, { 0x08, 0x000d }, { 0x09, 0xd810 }, - { 0x0a, 0x0060 }, + { 0x0a, 0x0120 }, { 0x0b, 0x0000 }, { 0x0d, 0x2800 }, { 0x0f, 0x0000 }, @@ -60,7 +60,7 @@ static struct reg_default rt286_index_def[] = { { 0x33, 0x0208 }, { 0x49, 0x0004 }, { 0x4f, 0x50e9 }, - { 0x50, 0x2c00 }, + { 0x50, 0x2000 }, { 0x63, 0x2902 }, { 0x67, 0x1111 }, { 0x68, 0x1016 }, @@ -104,7 +104,6 @@ static const struct reg_default rt286_reg[] = { { 0x02170700, 0x00000000 }, { 0x02270100, 0x00000000 }, { 0x02370100, 0x00000000 }, - { 0x02040000, 0x00004002 }, { 0x01870700, 0x00000020 }, { 0x00830000, 0x000000c3 }, { 0x00930000, 0x000000c3 }, -- cgit v1.1 From 66d627d554a4284dad00b2039efd18e1c129cc2f Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Fri, 26 Sep 2014 11:06:40 +0800 Subject: ASoC: rt286: Fix sync function We try to write index registers into cache when we write an index register, but we change the reg value before updating the cache. As a result, the cache is never be updated. This patch will fix this issue. Signed-off-by: Bard Liao Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/rt286.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index 7a66084..b86b426 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -191,7 +191,6 @@ static int rt286_hw_write(void *context, unsigned int reg, unsigned int value) /*handle index registers*/ if (reg <= 0xff) { rt286_hw_write(client, RT286_COEF_INDEX, reg); - reg = RT286_PROC_COEF; for (i = 0; i < INDEX_CACHE_SIZE; i++) { if (reg == rt286->index_cache[i].reg) { rt286->index_cache[i].def = value; @@ -199,6 +198,7 @@ static int rt286_hw_write(void *context, unsigned int reg, unsigned int value) } } + reg = RT286_PROC_COEF; } data[0] = (reg >> 24) & 0xff; -- cgit v1.1 From 1ee44ce03011bab025949e7636416912185f4122 Mon Sep 17 00:00:00 2001 From: Anatol Pomozov Date: Fri, 26 Sep 2014 13:31:06 -0700 Subject: ASoC: ssm4567: Add driver for Analog Devices SSM4567 amplifier Analog Devices SSM4567 is a boost class-D audio amplifier. Signed-off-by: Anatol Pomozov Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ssm4567.c | 345 +++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 352 insertions(+) create mode 100644 sound/soc/codecs/ssm4567.c (limited to 'sound') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 8838838e..bc1fe4e 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -90,6 +90,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_SSM2518 if I2C select SND_SOC_SSM2602_SPI if SPI_MASTER select SND_SOC_SSM2602_I2C if I2C + select SND_SOC_SSM4567 if I2C select SND_SOC_STA32X if I2C select SND_SOC_STA350 if I2C select SND_SOC_STA529 if I2C @@ -527,6 +528,10 @@ config SND_SOC_SSM2602_I2C select SND_SOC_SSM2602 tristate +config SND_SOC_SSM4567 + tristate "Analog Devices ssm4567 amplifier driver support" + depends on I2C + config SND_SOC_STA32X tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 20afe0f..bebad36 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -91,6 +91,7 @@ snd-soc-ssm2518-objs := ssm2518.o snd-soc-ssm2602-objs := ssm2602.o snd-soc-ssm2602-spi-objs := ssm2602-spi.o snd-soc-ssm2602-i2c-objs := ssm2602-i2c.o +snd-soc-ssm4567-objs := ssm4567.o snd-soc-sta32x-objs := sta32x.o snd-soc-sta350-objs := sta350.o snd-soc-sta529-objs := sta529.o @@ -258,6 +259,7 @@ obj-$(CONFIG_SND_SOC_SSM2518) += snd-soc-ssm2518.o obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o obj-$(CONFIG_SND_SOC_SSM2602_SPI) += snd-soc-ssm2602-spi.o obj-$(CONFIG_SND_SOC_SSM2602_I2C) += snd-soc-ssm2602-i2c.o +obj-$(CONFIG_SND_SOC_SSM4567) += snd-soc-ssm4567.o obj-$(CONFIG_SND_SOC_STA32X) += snd-soc-sta32x.o obj-$(CONFIG_SND_SOC_STA350) += snd-soc-sta350.o obj-$(CONFIG_SND_SOC_STA529) += snd-soc-sta529.o diff --git a/sound/soc/codecs/ssm4567.c b/sound/soc/codecs/ssm4567.c new file mode 100644 index 0000000..1dadacb --- /dev/null +++ b/sound/soc/codecs/ssm4567.c @@ -0,0 +1,345 @@ +/* + * SSM4567 amplifier audio driver + * + * Copyright 2014 Google Chromium project. + * Author: Anatol Pomozov + * + * Based on code copyright/by: + * Copyright 2013 Analog Devices Inc. + * + * Licensed under the GPL-2. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#define SSM4567_REG_POWER_CTRL 0x00 +#define SSM4567_REG_AMP_SNS_CTRL 0x01 +#define SSM4567_REG_DAC_CTRL 0x02 +#define SSM4567_REG_DAC_VOLUME 0x03 +#define SSM4567_REG_SAI_CTRL_1 0x04 +#define SSM4567_REG_SAI_CTRL_2 0x05 +#define SSM4567_REG_SAI_PLACEMENT_1 0x06 +#define SSM4567_REG_SAI_PLACEMENT_2 0x07 +#define SSM4567_REG_SAI_PLACEMENT_3 0x08 +#define SSM4567_REG_SAI_PLACEMENT_4 0x09 +#define SSM4567_REG_SAI_PLACEMENT_5 0x0a +#define SSM4567_REG_SAI_PLACEMENT_6 0x0b +#define SSM4567_REG_BATTERY_V_OUT 0x0c +#define SSM4567_REG_LIMITER_CTRL_1 0x0d +#define SSM4567_REG_LIMITER_CTRL_2 0x0e +#define SSM4567_REG_LIMITER_CTRL_3 0x0f +#define SSM4567_REG_STATUS_1 0x10 +#define SSM4567_REG_STATUS_2 0x11 +#define SSM4567_REG_FAULT_CTRL 0x12 +#define SSM4567_REG_PDM_CTRL 0x13 +#define SSM4567_REG_MCLK_RATIO 0x14 +#define SSM4567_REG_BOOST_CTRL_1 0x15 +#define SSM4567_REG_BOOST_CTRL_2 0x16 +#define SSM4567_REG_SOFT_RESET 0xff + +/* POWER_CTRL */ +#define SSM4567_POWER_APWDN_EN BIT(7) +#define SSM4567_POWER_BSNS_PWDN BIT(6) +#define SSM4567_POWER_VSNS_PWDN BIT(5) +#define SSM4567_POWER_ISNS_PWDN BIT(4) +#define SSM4567_POWER_BOOST_PWDN BIT(3) +#define SSM4567_POWER_AMP_PWDN BIT(2) +#define SSM4567_POWER_VBAT_ONLY BIT(1) +#define SSM4567_POWER_SPWDN BIT(0) + +/* DAC_CTRL */ +#define SSM4567_DAC_HV BIT(7) +#define SSM4567_DAC_MUTE BIT(6) +#define SSM4567_DAC_HPF BIT(5) +#define SSM4567_DAC_LPM BIT(4) +#define SSM4567_DAC_FS_MASK 0x7 +#define SSM4567_DAC_FS_8000_12000 0x0 +#define SSM4567_DAC_FS_16000_24000 0x1 +#define SSM4567_DAC_FS_32000_48000 0x2 +#define SSM4567_DAC_FS_64000_96000 0x3 +#define SSM4567_DAC_FS_128000_192000 0x4 + +struct ssm4567 { + struct regmap *regmap; +}; + +static const struct reg_default ssm4567_reg_defaults[] = { + { SSM4567_REG_POWER_CTRL, 0x81 }, + { SSM4567_REG_AMP_SNS_CTRL, 0x09 }, + { SSM4567_REG_DAC_CTRL, 0x32 }, + { SSM4567_REG_DAC_VOLUME, 0x40 }, + { SSM4567_REG_SAI_CTRL_1, 0x00 }, + { SSM4567_REG_SAI_CTRL_2, 0x08 }, + { SSM4567_REG_SAI_PLACEMENT_1, 0x01 }, + { SSM4567_REG_SAI_PLACEMENT_2, 0x20 }, + { SSM4567_REG_SAI_PLACEMENT_3, 0x32 }, + { SSM4567_REG_SAI_PLACEMENT_4, 0x07 }, + { SSM4567_REG_SAI_PLACEMENT_5, 0x07 }, + { SSM4567_REG_SAI_PLACEMENT_6, 0x07 }, + { SSM4567_REG_BATTERY_V_OUT, 0x00 }, + { SSM4567_REG_LIMITER_CTRL_1, 0xa4 }, + { SSM4567_REG_LIMITER_CTRL_2, 0x73 }, + { SSM4567_REG_LIMITER_CTRL_3, 0x00 }, + { SSM4567_REG_STATUS_1, 0x00 }, + { SSM4567_REG_STATUS_2, 0x00 }, + { SSM4567_REG_FAULT_CTRL, 0x30 }, + { SSM4567_REG_PDM_CTRL, 0x40 }, + { SSM4567_REG_MCLK_RATIO, 0x11 }, + { SSM4567_REG_BOOST_CTRL_1, 0x03 }, + { SSM4567_REG_BOOST_CTRL_2, 0x00 }, + { SSM4567_REG_SOFT_RESET, 0x00 }, +}; + + +static bool ssm4567_readable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case SSM4567_REG_POWER_CTRL ... SSM4567_REG_BOOST_CTRL_2: + return true; + default: + return false; + } + +} + +static bool ssm4567_writeable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case SSM4567_REG_POWER_CTRL ... SSM4567_REG_SAI_PLACEMENT_6: + case SSM4567_REG_LIMITER_CTRL_1 ... SSM4567_REG_LIMITER_CTRL_3: + case SSM4567_REG_FAULT_CTRL ... SSM4567_REG_BOOST_CTRL_2: + /* The datasheet states that soft reset register is read-only, + * but logically it is write-only. */ + case SSM4567_REG_SOFT_RESET: + return true; + default: + return false; + } +} + +static bool ssm4567_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case SSM4567_REG_BATTERY_V_OUT: + case SSM4567_REG_STATUS_1 ... SSM4567_REG_STATUS_2: + case SSM4567_REG_SOFT_RESET: + return true; + default: + return false; + } +} + +static const DECLARE_TLV_DB_MINMAX_MUTE(ssm4567_vol_tlv, -7125, 2400); + +static const struct snd_kcontrol_new ssm4567_snd_controls[] = { + SOC_SINGLE_TLV("Master Playback Volume", SSM4567_REG_DAC_VOLUME, 0, + 0xff, 1, ssm4567_vol_tlv), + SOC_SINGLE("DAC Low Power Mode Switch", SSM4567_REG_DAC_CTRL, 4, 1, 0), +}; + +static const struct snd_soc_dapm_widget ssm4567_dapm_widgets[] = { + SND_SOC_DAPM_DAC("DAC", "HiFi Playback", SSM4567_REG_POWER_CTRL, 2, 1), + + SND_SOC_DAPM_OUTPUT("OUT"), +}; + +static const struct snd_soc_dapm_route ssm4567_routes[] = { + { "OUT", NULL, "DAC" }, +}; + +static int ssm4567_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 ssm4567 *ssm4567 = snd_soc_codec_get_drvdata(codec); + unsigned int rate = params_rate(params); + unsigned int dacfs; + + if (rate >= 8000 && rate <= 12000) + dacfs = SSM4567_DAC_FS_8000_12000; + else if (rate >= 16000 && rate <= 24000) + dacfs = SSM4567_DAC_FS_16000_24000; + else if (rate >= 32000 && rate <= 48000) + dacfs = SSM4567_DAC_FS_32000_48000; + else if (rate >= 64000 && rate <= 96000) + dacfs = SSM4567_DAC_FS_64000_96000; + else if (rate >= 64000 && rate <= 96000) + dacfs = SSM4567_DAC_FS_64000_96000; + else if (rate >= 128000 && rate <= 192000) + dacfs = SSM4567_DAC_FS_128000_192000; + else + return -EINVAL; + + return regmap_update_bits(ssm4567->regmap, SSM4567_REG_DAC_CTRL, + SSM4567_DAC_FS_MASK, dacfs); +} + +static int ssm4567_mute(struct snd_soc_dai *dai, int mute) +{ + struct ssm4567 *ssm4567 = snd_soc_codec_get_drvdata(dai->codec); + unsigned int val; + + val = mute ? SSM4567_DAC_MUTE : 0; + return regmap_update_bits(ssm4567->regmap, SSM4567_REG_DAC_CTRL, + SSM4567_DAC_MUTE, val); +} + +static int ssm4567_set_power(struct ssm4567 *ssm4567, bool enable) +{ + int ret = 0; + + if (!enable) { + ret = regmap_update_bits(ssm4567->regmap, + SSM4567_REG_POWER_CTRL, + SSM4567_POWER_SPWDN, SSM4567_POWER_SPWDN); + regcache_mark_dirty(ssm4567->regmap); + } + + regcache_cache_only(ssm4567->regmap, !enable); + + if (enable) { + ret = regmap_update_bits(ssm4567->regmap, + SSM4567_REG_POWER_CTRL, + SSM4567_POWER_SPWDN, 0x00); + regcache_sync(ssm4567->regmap); + } + + return ret; +} + +static int ssm4567_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct ssm4567 *ssm4567 = snd_soc_codec_get_drvdata(codec); + int ret = 0; + + 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 = ssm4567_set_power(ssm4567, true); + break; + case SND_SOC_BIAS_OFF: + ret = ssm4567_set_power(ssm4567, false); + break; + } + + if (ret) + return ret; + + codec->dapm.bias_level = level; + + return 0; +} + +static const struct snd_soc_dai_ops ssm4567_dai_ops = { + .hw_params = ssm4567_hw_params, + .digital_mute = ssm4567_mute, +}; + +static struct snd_soc_dai_driver ssm4567_dai = { + .name = "ssm4567-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 1, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32, + }, + .ops = &ssm4567_dai_ops, +}; + +static struct snd_soc_codec_driver ssm4567_codec_driver = { + .set_bias_level = ssm4567_set_bias_level, + .idle_bias_off = true, + + .controls = ssm4567_snd_controls, + .num_controls = ARRAY_SIZE(ssm4567_snd_controls), + .dapm_widgets = ssm4567_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(ssm4567_dapm_widgets), + .dapm_routes = ssm4567_routes, + .num_dapm_routes = ARRAY_SIZE(ssm4567_routes), +}; + +static const struct regmap_config ssm4567_regmap_config = { + .val_bits = 8, + .reg_bits = 8, + + .max_register = SSM4567_REG_SOFT_RESET, + .readable_reg = ssm4567_readable_reg, + .writeable_reg = ssm4567_writeable_reg, + .volatile_reg = ssm4567_volatile_reg, + + .cache_type = REGCACHE_RBTREE, + .reg_defaults = ssm4567_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(ssm4567_reg_defaults), +}; + +static int ssm4567_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct ssm4567 *ssm4567; + int ret; + + ssm4567 = devm_kzalloc(&i2c->dev, sizeof(*ssm4567), GFP_KERNEL); + if (ssm4567 == NULL) + return -ENOMEM; + + i2c_set_clientdata(i2c, ssm4567); + + ssm4567->regmap = devm_regmap_init_i2c(i2c, &ssm4567_regmap_config); + if (IS_ERR(ssm4567->regmap)) + return PTR_ERR(ssm4567->regmap); + + ret = regmap_write(ssm4567->regmap, SSM4567_REG_SOFT_RESET, 0x00); + if (ret) + return ret; + + ret = ssm4567_set_power(ssm4567, false); + if (ret) + return ret; + + return snd_soc_register_codec(&i2c->dev, &ssm4567_codec_driver, + &ssm4567_dai, 1); +} + +static int ssm4567_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static const struct i2c_device_id ssm4567_i2c_ids[] = { + { "ssm4567", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, ssm4567_i2c_ids); + +static struct i2c_driver ssm4567_driver = { + .driver = { + .name = "ssm4567", + .owner = THIS_MODULE, + }, + .probe = ssm4567_i2c_probe, + .remove = ssm4567_i2c_remove, + .id_table = ssm4567_i2c_ids, +}; +module_i2c_driver(ssm4567_driver); + +MODULE_DESCRIPTION("ASoC SSM4567 driver"); +MODULE_AUTHOR("Anatol Pomozov "); +MODULE_LICENSE("GPL"); -- cgit v1.1 From 6596aa047b624aeec2ea321962cfdecf9953a383 Mon Sep 17 00:00:00 2001 From: Xiubo Li Date: Sun, 28 Sep 2014 17:29:37 +0800 Subject: ASoC: core: fix possible ZERO_SIZE_PTR pointer dereferencing error. Since we cannot make sure the 'params->num_regs' will always be none zero here, and then if it equals to zero, the kmemdup() will return ZERO_SIZE_PTR, which equals to ((void *)16). So this patch fix this with just doing the zero check before calling kmemdup(). Signed-off-by: Xiubo Li Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/soc-core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index d4bfd4a..ae48f10 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3203,7 +3203,7 @@ int snd_soc_bytes_put(struct snd_kcontrol *kcontrol, unsigned int val, mask; void *data; - if (!component->regmap) + if (!component->regmap || !params->num_regs) return -EINVAL; len = params->num_regs * component->val_bytes; -- cgit v1.1 From f69e3caa9e1855737bf1e99e1fe4488e33d74bfe Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Fri, 26 Sep 2014 16:25:37 +0300 Subject: ASoC: max98090: Enable both DMIC channels also when using mono configuration According to MAX98090 specification "Digital microphone clock (DMC) is enabled once both data channels are enabled.". Therefore both digital microphone data channels must be enabled also when using mono microphone configuration. Fix this by moving "DMICL_ENA" and "DMICR_ENA" supply widgets from "DMICL" and "DMICR" inputs to "DMIC Mux" in order to enable both data channels whenever there is active mono or stereo digital microphone input path. Use of "DMICL" and "DMICR" inputs are retained for informative source and in case the driver would find use for exact digital microphone configuration in the future. Signed-off-by: Jarkko Nikula Signed-off-by: Mark Brown --- sound/soc/codecs/max98090.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index f154365..7e11186 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -1311,8 +1311,6 @@ static const struct snd_soc_dapm_route max98090_dapm_routes[] = { {"MIC1 Input", NULL, "MIC1"}, {"MIC2 Input", NULL, "MIC2"}, - {"DMICL", NULL, "DMICL_ENA"}, - {"DMICR", NULL, "DMICR_ENA"}, {"DMICL", NULL, "AHPF"}, {"DMICR", NULL, "AHPF"}, @@ -1370,6 +1368,8 @@ static const struct snd_soc_dapm_route max98090_dapm_routes[] = { {"DMIC Mux", "ADC", "ADCR"}, {"DMIC Mux", "DMIC", "DMICL"}, {"DMIC Mux", "DMIC", "DMICR"}, + {"DMIC Mux", "DMIC", "DMICL_ENA"}, + {"DMIC Mux", "DMIC", "DMICR_ENA"}, {"LBENL Mux", "Normal", "DMIC Mux"}, {"LBENL Mux", "Loopback", "LTENL Mux"}, -- cgit v1.1 From 969168e2e9f4a5bfd6a49344f46b820437cd9163 Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Fri, 26 Sep 2014 16:25:38 +0300 Subject: ASoC: Intel: byt-max98090: Set card as fully routed All byt-max98090 audio connections are known and described in DAPM routing table. Set the fully_routed flag in order to disable unused codec pins. Signed-off-by: Jarkko Nikula Signed-off-by: Mark Brown --- sound/soc/intel/byt-max98090.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/intel/byt-max98090.c b/sound/soc/intel/byt-max98090.c index b8b8af5..d52681e 100644 --- a/sound/soc/intel/byt-max98090.c +++ b/sound/soc/intel/byt-max98090.c @@ -139,6 +139,7 @@ static struct snd_soc_card byt_max98090_card = { .num_dapm_routes = ARRAY_SIZE(byt_max98090_audio_map), .controls = byt_max98090_controls, .num_controls = ARRAY_SIZE(byt_max98090_controls), + .fully_routed = true, }; static int byt_max98090_probe(struct platform_device *pdev) -- cgit v1.1 From 19926c6de0c37f486f00b7531aec4ba5a09451ae Mon Sep 17 00:00:00 2001 From: Arnd Bergmann Date: Mon, 29 Sep 2014 17:32:17 +0200 Subject: ASoC: davinci: vcif must be a module if SND_DAVINCI_SOC is It is possible to configure a kernel with SND_DAVINCI_SOC=m and SND_DM365_VOICE_CODEC=y, which results in a link error: sound/built-in.o: In function `davinci_vcif_probe': sound/soc/davinci/davinci-vcif.c:223: undefined reference to `davinci_soc_platform_register' The best way to avoid this is to make SND_DM365_VOICE_CODEC a tristate option that depends on SND_DAVINCI_SOC, so it can only be a module or disabled when the base driver is a loadable module Signed-off-by: Arnd Bergmann Signed-off-by: Mark Brown --- sound/soc/davinci/Kconfig | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig index d69510c..8e948c6 100644 --- a/sound/soc/davinci/Kconfig +++ b/sound/soc/davinci/Kconfig @@ -63,7 +63,8 @@ config SND_DM365_AIC3X_CODEC Say Y if you want to add support for AIC3101 audio codec config SND_DM365_VOICE_CODEC - bool "Voice Codec - CQ93VC" + tristate "Voice Codec - CQ93VC" + depends on SND_DAVINCI_SOC select MFD_DAVINCI_VOICECODEC select SND_DAVINCI_SOC_VCIF select SND_SOC_CQ0093VC -- cgit v1.1 From 9cca023e5c5c13486d48d47a46564c359af9ae73 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 29 Sep 2014 11:40:40 +0200 Subject: ASoC: wm8{350,753,971}: Use snd_soc_dapm_to_codec() instead of dapm->codec The CODEC struct in the snd_soc_dapm_context struct is deprecated and scheduled for removal. Use the snd_soc_dapm_to_codec() function instead. Signed-off-by: Lars-Peter Clausen Acked-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm8350.c | 2 +- sound/soc/codecs/wm8753.c | 2 +- sound/soc/codecs/wm8971.c | 2 +- 3 files changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index 3dfdcc4..628ec77 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -212,7 +212,7 @@ static void wm8350_pga_work(struct work_struct *work) { struct snd_soc_dapm_context *dapm = container_of(work, struct snd_soc_dapm_context, delayed_work.work); - struct snd_soc_codec *codec = dapm->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(dapm); struct wm8350_data *wm8350_data = snd_soc_codec_get_drvdata(codec); struct wm8350_output *out1 = &wm8350_data->out1, *out2 = &wm8350_data->out2; diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index e54e097..21ca3a9 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -1433,7 +1433,7 @@ static void wm8753_work(struct work_struct *work) struct snd_soc_dapm_context *dapm = container_of(work, struct snd_soc_dapm_context, delayed_work.work); - struct snd_soc_codec *codec = dapm->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(dapm); wm8753_set_bias_level(codec, dapm->bias_level); } diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c index 0499cd4..39ddb9b 100644 --- a/sound/soc/codecs/wm8971.c +++ b/sound/soc/codecs/wm8971.c @@ -615,7 +615,7 @@ static void wm8971_work(struct work_struct *work) struct snd_soc_dapm_context *dapm = container_of(work, struct snd_soc_dapm_context, delayed_work.work); - struct snd_soc_codec *codec = dapm->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(dapm); wm8971_set_bias_level(codec, codec->dapm.bias_level); } -- cgit v1.1 From a761f87f367a2a172cbc62d0e88eabe175d349a8 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 29 Sep 2014 11:40:41 +0200 Subject: ASoC: rx51: Use snd_soc_dapm_to_codec() instead of dapm->codec The CODEC struct in the snd_soc_dapm_context struct is deprecated and scheduled for removal. Use the snd_soc_dapm_to_codec() function instead. Signed-off-by: Lars-Peter Clausen Acked-by: Jarkko Nikula Signed-off-by: Mark Brown --- sound/soc/omap/rx51.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/omap/rx51.c b/sound/soc/omap/rx51.c index 943922c..b10ae80 100644 --- a/sound/soc/omap/rx51.c +++ b/sound/soc/omap/rx51.c @@ -168,7 +168,7 @@ static int rx51_spk_event(struct snd_soc_dapm_widget *w, static int rx51_hp_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *k, int event) { - struct snd_soc_codec *codec = w->dapm->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); if (SND_SOC_DAPM_EVENT_ON(event)) tpa6130a2_stereo_enable(codec, 1); -- cgit v1.1 From 0bd2ac3dae74ee25c5ea171cb572731c7a89c248 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 29 Sep 2014 11:40:42 +0200 Subject: ASoC: Remove CODEC pointer from snd_soc_dapm_context The only remaining user of the CODEC pointer in the DAPM struct is to initialize the CODEC pointer in the widget struct. The later is scheduled for removal, but has still a few users left. For now use dapm->component->codec to initialize it. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 1 - sound/soc/soc-dapm.c | 2 +- 2 files changed, 1 insertion(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 052f59c..8d45eec 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -4351,7 +4351,6 @@ int snd_soc_register_codec(struct device *dev, if (codec_drv->read) codec->component.read = snd_soc_codec_drv_read; codec->component.ignore_pmdown_time = codec_drv->ignore_pmdown_time; - codec->dapm.codec = codec; codec->dapm.idle_bias_off = codec_drv->idle_bias_off; if (codec_drv->seq_notifier) codec->dapm.seq_notifier = codec_drv->seq_notifier; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 8348352..1f1e965 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3107,7 +3107,7 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, } w->dapm = dapm; - w->codec = dapm->codec; + w->codec = dapm->component->codec; INIT_LIST_HEAD(&w->sources); INIT_LIST_HEAD(&w->sinks); INIT_LIST_HEAD(&w->list); -- cgit v1.1 From ac06dd8df6e13591524f0e1bedf36af4ca0e967b Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Mon, 29 Sep 2014 16:58:15 +0300 Subject: ASoC: Intel: byt-rt5640: Remove IN2N pin from DAPM route table I tested couple byt-rt5640 based platforms and they have single-ended headset microphone connection to IN2P only. I guess IN2N was either defined by accident or some early platform had floating ground for headset. It's better to remove IN2N and add a custom route for such a platform if needed. Signed-off-by: Jarkko Nikula Signed-off-by: Mark Brown --- sound/soc/intel/byt-rt5640.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/intel/byt-rt5640.c b/sound/soc/intel/byt-rt5640.c index 234a58d..d6d8b19 100644 --- a/sound/soc/intel/byt-rt5640.c +++ b/sound/soc/intel/byt-rt5640.c @@ -36,7 +36,6 @@ static const struct snd_soc_dapm_widget byt_rt5640_widgets[] = { static const struct snd_soc_dapm_route byt_rt5640_audio_map[] = { {"Headset Mic", NULL, "MICBIAS1"}, {"IN2P", NULL, "Headset Mic"}, - {"IN2N", NULL, "Headset Mic"}, {"DMIC1", NULL, "Internal Mic"}, {"Headphone", NULL, "HPOL"}, {"Headphone", NULL, "HPOR"}, -- cgit v1.1 From f8a770c2c67f28956f8f4601feb99e9bd02a16c8 Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Mon, 29 Sep 2014 16:58:16 +0300 Subject: ASoC: Intel: byt-rt5640: Add quirk for Asus T100 Asus T100 internal microphone is not digital but analogue connected to IN1P pin of the RT564x codec with shared bias between internal and headset microphones. Because of this there is need to have machine specific DAPM routes in byt-rt5640. Add handling for them with the help of DMI quirk that is used to add custom routes in addition to common. Because "Internal Mic" connected to DMIC1 is not common to all move it as a default custom route when there is no match in quirk table. Custom "Internal Mic" -> "IN1P" with MICBIAS1 route is added for Asus T100. Signed-off-by: Jarkko Nikula Signed-off-by: Mark Brown --- sound/soc/intel/byt-rt5640.c | 53 +++++++++++++++++++++++++++++++++++++++++++- 1 file changed, 52 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/intel/byt-rt5640.c b/sound/soc/intel/byt-rt5640.c index d6d8b19..c323a10 100644 --- a/sound/soc/intel/byt-rt5640.c +++ b/sound/soc/intel/byt-rt5640.c @@ -17,6 +17,7 @@ #include #include #include +#include #include #include #include @@ -36,7 +37,6 @@ static const struct snd_soc_dapm_widget byt_rt5640_widgets[] = { static const struct snd_soc_dapm_route byt_rt5640_audio_map[] = { {"Headset Mic", NULL, "MICBIAS1"}, {"IN2P", NULL, "Headset Mic"}, - {"DMIC1", NULL, "Internal Mic"}, {"Headphone", NULL, "HPOL"}, {"Headphone", NULL, "HPOR"}, {"Speaker", NULL, "SPOLP"}, @@ -45,6 +45,22 @@ static const struct snd_soc_dapm_route byt_rt5640_audio_map[] = { {"Speaker", NULL, "SPORN"}, }; +static const struct snd_soc_dapm_route byt_rt5640_intmic_dmic1_map[] = { + {"DMIC1", NULL, "Internal Mic"}, +}; + +static const struct snd_soc_dapm_route byt_rt5640_intmic_in1_map[] = { + {"Internal Mic", NULL, "MICBIAS1"}, + {"IN1P", NULL, "Internal Mic"}, +}; + +enum { + BYT_RT5640_DMIC1_MAP, + BYT_RT5640_IN1_MAP, +}; + +static unsigned long byt_rt5640_custom_map = BYT_RT5640_DMIC1_MAP; + static const struct snd_kcontrol_new byt_rt5640_controls[] = { SOC_DAPM_PIN_SWITCH("Headphone"), SOC_DAPM_PIN_SWITCH("Headset Mic"), @@ -76,12 +92,32 @@ static int byt_rt5640_hw_params(struct snd_pcm_substream *substream, return 0; } +static int byt_rt5640_quirk_cb(const struct dmi_system_id *id) +{ + byt_rt5640_custom_map = (unsigned long)id->driver_data; + return 1; +} + +static const struct dmi_system_id byt_rt5640_quirk_table[] = { + { + .callback = byt_rt5640_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "ASUSTeK COMPUTER INC."), + DMI_MATCH(DMI_PRODUCT_NAME, "T100TA"), + }, + .driver_data = (unsigned long *)BYT_RT5640_IN1_MAP, + }, + {} +}; + static int byt_rt5640_init(struct snd_soc_pcm_runtime *runtime) { int ret; struct snd_soc_codec *codec = runtime->codec; struct snd_soc_dapm_context *dapm = &codec->dapm; struct snd_soc_card *card = runtime->card; + const struct snd_soc_dapm_route *custom_map; + int num_routes; card->dapm.idle_bias_off = true; @@ -92,6 +128,21 @@ static int byt_rt5640_init(struct snd_soc_pcm_runtime *runtime) return ret; } + dmi_check_system(byt_rt5640_quirk_table); + switch (byt_rt5640_custom_map) { + case BYT_RT5640_IN1_MAP: + custom_map = byt_rt5640_intmic_in1_map; + num_routes = ARRAY_SIZE(byt_rt5640_intmic_in1_map); + break; + default: + custom_map = byt_rt5640_intmic_dmic1_map; + num_routes = ARRAY_SIZE(byt_rt5640_intmic_dmic1_map); + }; + + ret = snd_soc_dapm_add_routes(dapm, custom_map, num_routes); + if (ret) + return ret; + snd_soc_dapm_ignore_suspend(dapm, "HPOL"); snd_soc_dapm_ignore_suspend(dapm, "HPOR"); -- cgit v1.1 From 6f67c380056ceaf5844f18d3a5d769d233247849 Mon Sep 17 00:00:00 2001 From: Anatol Pomozov Date: Fri, 26 Sep 2014 09:57:27 -0700 Subject: ASoC: rt5677: Add dts properties for input/output differential configuration Signed-off-by: Anatol Pomozov Signed-off-by: Mark Brown --- sound/soc/codecs/rt5677.c | 23 +++++++++++++++++++++++ 1 file changed, 23 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index d2c6abf..97dff71 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -3569,6 +3569,17 @@ MODULE_DEVICE_TABLE(i2c, rt5677_i2c_id); static int rt5677_parse_dt(struct rt5677_priv *rt5677, struct device_node *np) { + rt5677->pdata.in1_diff = of_property_read_bool(np, + "realtek,in1-differential"); + rt5677->pdata.in2_diff = of_property_read_bool(np, + "realtek,in2-differential"); + rt5677->pdata.lout1_diff = of_property_read_bool(np, + "realtek,lout1-differential"); + rt5677->pdata.lout2_diff = of_property_read_bool(np, + "realtek,lout2-differential"); + rt5677->pdata.lout3_diff = of_property_read_bool(np, + "realtek,lout3-differential"); + rt5677->pow_ldo2 = of_get_named_gpio(np, "realtek,pow-ldo2-gpio", 0); @@ -3660,6 +3671,18 @@ static int rt5677_i2c_probe(struct i2c_client *i2c, regmap_update_bits(rt5677->regmap, RT5677_IN1, RT5677_IN_DF2, RT5677_IN_DF2); + if (rt5677->pdata.lout1_diff) + regmap_update_bits(rt5677->regmap, RT5677_LOUT1, + RT5677_LOUT1_L_DF, RT5677_LOUT1_L_DF); + + if (rt5677->pdata.lout2_diff) + regmap_update_bits(rt5677->regmap, RT5677_LOUT1, + RT5677_LOUT2_L_DF, RT5677_LOUT2_L_DF); + + if (rt5677->pdata.lout3_diff) + regmap_update_bits(rt5677->regmap, RT5677_LOUT1, + RT5677_LOUT3_L_DF, RT5677_LOUT3_L_DF); + if (rt5677->pdata.dmic2_clk_pin == RT5677_DMIC_CLK2) { regmap_update_bits(rt5677->regmap, RT5677_GEN_CTRL2, RT5677_GPIO5_FUNC_MASK, -- cgit v1.1 From fe2a08b3bf1a6e35c00e18843bc19aa1778432c3 Mon Sep 17 00:00:00 2001 From: Stefan Kristiansson Date: Mon, 29 Sep 2014 22:39:13 +0300 Subject: ASoC: ssm2602: do not hardcode type to SSM2602 The correct type (SSM2602/SSM2603/SSM2604) is passed down from the ssm2602_spi_probe()/ssm2602_spi_probe() functions, so use that instead of hardcoding it to SSM2602 in ssm2602_probe(). Fixes: c924dc68f737 ("ASoC: ssm2602: Split SPI and I2C code into different modules") Signed-off-by: Stefan Kristiansson Signed-off-by: Mark Brown Acked-by: Lars-Peter Clausen Cc: stable@vger.kernel.org --- sound/soc/codecs/ssm2602.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index 484b3bb..4021cd4 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -647,7 +647,7 @@ int ssm2602_probe(struct device *dev, enum ssm2602_type type, return -ENOMEM; dev_set_drvdata(dev, ssm2602); - ssm2602->type = SSM2602; + ssm2602->type = type; ssm2602->regmap = regmap; return snd_soc_register_codec(dev, &soc_codec_dev_ssm2602, -- cgit v1.1 From 555b9ee1368a9ceddd5c963ad918db5120638674 Mon Sep 17 00:00:00 2001 From: Stefan Kristiansson Date: Mon, 29 Sep 2014 22:41:10 +0300 Subject: ASoC: ssm2602: add device tree bindings Allow the ssm2602/ssm2603/ssm2604 codec driver to be instantiated from the device tree. Also, add Kconfig prompts to allow manual selection of both the I2C and SPI configuration versions of the driver. Signed-off-by: Stefan Kristiansson Signed-off-by: Mark Brown Acked-by: Lars-Peter Clausen --- sound/soc/codecs/Kconfig | 8 ++++++-- sound/soc/codecs/ssm2602-i2c.c | 9 +++++++++ sound/soc/codecs/ssm2602-spi.c | 7 +++++++ 3 files changed, 22 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 8838838e..3649e73 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -520,12 +520,16 @@ config SND_SOC_SSM2602 tristate config SND_SOC_SSM2602_SPI + tristate "Analog Devices SSM2602 CODEC - SPI" + depends on SPI_MASTER select SND_SOC_SSM2602 - tristate + select REGMAP_SPI config SND_SOC_SSM2602_I2C + tristate "Analog Devices SSM2602 CODEC - I2C" + depends on I2C select SND_SOC_SSM2602 - tristate + select REGMAP_I2C config SND_SOC_STA32X tristate diff --git a/sound/soc/codecs/ssm2602-i2c.c b/sound/soc/codecs/ssm2602-i2c.c index abd63d5..0d9779d 100644 --- a/sound/soc/codecs/ssm2602-i2c.c +++ b/sound/soc/codecs/ssm2602-i2c.c @@ -41,10 +41,19 @@ static const struct i2c_device_id ssm2602_i2c_id[] = { }; MODULE_DEVICE_TABLE(i2c, ssm2602_i2c_id); +static const struct of_device_id ssm2602_of_match[] = { + { .compatible = "adi,ssm2602", }, + { .compatible = "adi,ssm2603", }, + { .compatible = "adi,ssm2604", }, + { } +}; +MODULE_DEVICE_TABLE(of, ssm2602_of_match); + static struct i2c_driver ssm2602_i2c_driver = { .driver = { .name = "ssm2602", .owner = THIS_MODULE, + .of_match_table = ssm2602_of_match, }, .probe = ssm2602_i2c_probe, .remove = ssm2602_i2c_remove, diff --git a/sound/soc/codecs/ssm2602-spi.c b/sound/soc/codecs/ssm2602-spi.c index 2bf55e2..b5df14f 100644 --- a/sound/soc/codecs/ssm2602-spi.c +++ b/sound/soc/codecs/ssm2602-spi.c @@ -26,10 +26,17 @@ static int ssm2602_spi_remove(struct spi_device *spi) return 0; } +static const struct of_device_id ssm2602_of_match[] = { + { .compatible = "adi,ssm2602", }, + { } +}; +MODULE_DEVICE_TABLE(of, ssm2602_of_match); + static struct spi_driver ssm2602_spi_driver = { .driver = { .name = "ssm2602", .owner = THIS_MODULE, + .of_match_table = ssm2602_of_match, }, .probe = ssm2602_spi_probe, .remove = ssm2602_spi_remove, -- cgit v1.1 From 3b2a0013c7d49783d5ac3df9178e9907cd6ebd73 Mon Sep 17 00:00:00 2001 From: Stefan Kristiansson Date: Mon, 29 Sep 2014 22:41:37 +0300 Subject: ASoC: ssm2602: add support for 11.025kHz and 22.5kHz sample rates This adds the necessary values to the constraint list and register values to the coefficient table in order to configure the device for 11.025kHz and 22.5kHz sample rates. Signed-off-by: Stefan Kristiansson Signed-off-by: Mark Brown Acked-by: Lars-Peter Clausen --- sound/soc/codecs/ssm2602.c | 15 +++++++++++++-- 1 file changed, 13 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index 4021cd4..7c41848 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -192,7 +192,7 @@ static const struct snd_pcm_hw_constraint_list ssm2602_constraints_12288000 = { }; static const unsigned int ssm2602_rates_11289600[] = { - 8000, 44100, 88200, + 8000, 11025, 22050, 44100, 88200, }; static const struct snd_pcm_hw_constraint_list ssm2602_constraints_11289600 = { @@ -237,6 +237,16 @@ static const struct ssm2602_coeff ssm2602_coeff_table[] = { {18432000, 96000, SSM2602_COEFF_SRATE(0x7, 0x1, 0x0)}, {12000000, 96000, SSM2602_COEFF_SRATE(0x7, 0x0, 0x1)}, + /* 11.025k */ + {11289600, 11025, SSM2602_COEFF_SRATE(0xc, 0x0, 0x0)}, + {16934400, 11025, SSM2602_COEFF_SRATE(0xc, 0x1, 0x0)}, + {12000000, 11025, SSM2602_COEFF_SRATE(0xc, 0x1, 0x1)}, + + /* 22.05k */ + {11289600, 22050, SSM2602_COEFF_SRATE(0xd, 0x0, 0x0)}, + {16934400, 22050, SSM2602_COEFF_SRATE(0xd, 0x1, 0x0)}, + {12000000, 22050, SSM2602_COEFF_SRATE(0xd, 0x1, 0x1)}, + /* 44.1k */ {11289600, 44100, SSM2602_COEFF_SRATE(0x8, 0x0, 0x0)}, {16934400, 44100, SSM2602_COEFF_SRATE(0x8, 0x1, 0x0)}, @@ -467,7 +477,8 @@ static int ssm2602_set_bias_level(struct snd_soc_codec *codec, return 0; } -#define SSM2602_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |\ +#define SSM2602_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ + SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\ SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |\ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 |\ SNDRV_PCM_RATE_96000) -- cgit v1.1 From ece1e4999606fc323aee96a1cdb9b7991c01dd09 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Mon, 29 Sep 2014 23:25:29 -0300 Subject: ASoC: fsl_ssi: Remove unneeded 'i2s-slave' property There is no need to use 'i2s-slave' property, since master/slave configuration are passed via machine layer. This change does not break existing users because they do check for slave mode inside sound/soc/fsl/mpc8610_hpcd.c/p1022_ds.c/p1022_rdk.c Signed-off-by: Fabio Estevam Acked-by: Timur Tabi Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_ssi.c | 3 --- 1 file changed, 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 16a1361..f19224e 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1314,9 +1314,6 @@ static int fsl_ssi_probe(struct platform_device *pdev) if (sprop) { if (!strcmp(sprop, "ac97-slave")) ssi_private->dai_fmt = SND_SOC_DAIFMT_AC97; - else if (!strcmp(sprop, "i2s-slave")) - ssi_private->dai_fmt = SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_CBM_CFM; } ssi_private->use_dma = !of_property_read_bool(np, -- cgit v1.1 From 1cc0c054f380c1c477642b5d9d9d9f697f641dbc Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Wed, 1 Oct 2014 16:02:11 +0300 Subject: ASoC: davinci-mcasp: Convert the context save/restore to use array Instead of individual values use an array to store the registers need to be saved on suspend and restored on resume. It is going to be easier to add more registers to save and restore. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-mcasp.c | 38 +++++++++++++++++--------------------- 1 file changed, 17 insertions(+), 21 deletions(-) (limited to 'sound') diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index c28508d..63e2444 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -42,14 +42,18 @@ #define MCASP_MAX_AFIFO_DEPTH 64 +static u32 context_regs[] = { + DAVINCI_MCASP_TXFMCTL_REG, + DAVINCI_MCASP_RXFMCTL_REG, + DAVINCI_MCASP_TXFMT_REG, + DAVINCI_MCASP_RXFMT_REG, + DAVINCI_MCASP_ACLKXCTL_REG, + DAVINCI_MCASP_ACLKRCTL_REG, + DAVINCI_MCASP_PDIR_REG, +}; + struct davinci_mcasp_context { - u32 txfmtctl; - u32 rxfmtctl; - u32 txfmt; - u32 rxfmt; - u32 aclkxctl; - u32 aclkrctl; - u32 pdir; + u32 config_regs[ARRAY_SIZE(context_regs)]; }; struct davinci_mcasp { @@ -857,14 +861,10 @@ static int davinci_mcasp_suspend(struct snd_soc_dai *dai) { struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(dai); struct davinci_mcasp_context *context = &mcasp->context; + int i; - context->txfmtctl = mcasp_get_reg(mcasp, DAVINCI_MCASP_TXFMCTL_REG); - context->rxfmtctl = mcasp_get_reg(mcasp, DAVINCI_MCASP_RXFMCTL_REG); - context->txfmt = mcasp_get_reg(mcasp, DAVINCI_MCASP_TXFMT_REG); - context->rxfmt = mcasp_get_reg(mcasp, DAVINCI_MCASP_RXFMT_REG); - context->aclkxctl = mcasp_get_reg(mcasp, DAVINCI_MCASP_ACLKXCTL_REG); - context->aclkrctl = mcasp_get_reg(mcasp, DAVINCI_MCASP_ACLKRCTL_REG); - context->pdir = mcasp_get_reg(mcasp, DAVINCI_MCASP_PDIR_REG); + for (i = 0; i < ARRAY_SIZE(context_regs); i++) + context->config_regs[i] = mcasp_get_reg(mcasp, context_regs[i]); return 0; } @@ -873,14 +873,10 @@ static int davinci_mcasp_resume(struct snd_soc_dai *dai) { struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(dai); struct davinci_mcasp_context *context = &mcasp->context; + int i; - mcasp_set_reg(mcasp, DAVINCI_MCASP_TXFMCTL_REG, context->txfmtctl); - mcasp_set_reg(mcasp, DAVINCI_MCASP_RXFMCTL_REG, context->rxfmtctl); - mcasp_set_reg(mcasp, DAVINCI_MCASP_TXFMT_REG, context->txfmt); - mcasp_set_reg(mcasp, DAVINCI_MCASP_RXFMT_REG, context->rxfmt); - mcasp_set_reg(mcasp, DAVINCI_MCASP_ACLKXCTL_REG, context->aclkxctl); - mcasp_set_reg(mcasp, DAVINCI_MCASP_ACLKRCTL_REG, context->aclkrctl); - mcasp_set_reg(mcasp, DAVINCI_MCASP_PDIR_REG, context->pdir); + for (i = 0; i < ARRAY_SIZE(context_regs); i++) + mcasp_set_reg(mcasp, context_regs[i], context->config_regs[i]); return 0; } -- cgit v1.1 From f114ce605daa1fb9d4efa253ea6d5bd4802902af Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Wed, 1 Oct 2014 16:02:12 +0300 Subject: ASoC: davinvi-mcasp: Proper suspend/resume support while audio is active If the board is sent to suspend (deep sleep) the McASP context will be lost. In case when suspend happens during active audio we need to save and restore more registers, which was configured during hw_param times as well. We need to add more config registers, AFIFO control registers and we also need to save and restore the serializer configuration as well. Since the number of serializers depends on the SoC we need to allocate the memory for it based on the num_serializer for the given McASP instance. With this patch the ongoing stream will resume after resuming from deep sleep. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-mcasp.c | 41 +++++++++++++++++++++++++++++++++++++++ 1 file changed, 41 insertions(+) (limited to 'sound') diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 63e2444..5dcacc4 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -49,11 +49,19 @@ static u32 context_regs[] = { DAVINCI_MCASP_RXFMT_REG, DAVINCI_MCASP_ACLKXCTL_REG, DAVINCI_MCASP_ACLKRCTL_REG, + DAVINCI_MCASP_AHCLKXCTL_REG, + DAVINCI_MCASP_AHCLKRCTL_REG, DAVINCI_MCASP_PDIR_REG, + DAVINCI_MCASP_RXMASK_REG, + DAVINCI_MCASP_TXMASK_REG, + DAVINCI_MCASP_RXTDM_REG, + DAVINCI_MCASP_TXTDM_REG, }; struct davinci_mcasp_context { u32 config_regs[ARRAY_SIZE(context_regs)]; + u32 afifo_regs[2]; /* for read/write fifo control registers */ + u32 *xrsr_regs; /* for serializer configuration */ }; struct davinci_mcasp { @@ -861,11 +869,25 @@ static int davinci_mcasp_suspend(struct snd_soc_dai *dai) { struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(dai); struct davinci_mcasp_context *context = &mcasp->context; + u32 reg; int i; for (i = 0; i < ARRAY_SIZE(context_regs); i++) context->config_regs[i] = mcasp_get_reg(mcasp, context_regs[i]); + if (mcasp->txnumevt) { + reg = mcasp->fifo_base + MCASP_WFIFOCTL_OFFSET; + context->afifo_regs[0] = mcasp_get_reg(mcasp, reg); + } + if (mcasp->rxnumevt) { + reg = mcasp->fifo_base + MCASP_RFIFOCTL_OFFSET; + context->afifo_regs[1] = mcasp_get_reg(mcasp, reg); + } + + for (i = 0; i < mcasp->num_serializer; i++) + context->xrsr_regs[i] = mcasp_get_reg(mcasp, + DAVINCI_MCASP_XRSRCTL_REG(i)); + return 0; } @@ -873,11 +895,25 @@ static int davinci_mcasp_resume(struct snd_soc_dai *dai) { struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(dai); struct davinci_mcasp_context *context = &mcasp->context; + u32 reg; int i; for (i = 0; i < ARRAY_SIZE(context_regs); i++) mcasp_set_reg(mcasp, context_regs[i], context->config_regs[i]); + if (mcasp->txnumevt) { + reg = mcasp->fifo_base + MCASP_WFIFOCTL_OFFSET; + mcasp_set_reg(mcasp, reg, context->afifo_regs[0]); + } + if (mcasp->rxnumevt) { + reg = mcasp->fifo_base + MCASP_RFIFOCTL_OFFSET; + mcasp_set_reg(mcasp, reg, context->afifo_regs[1]); + } + + for (i = 0; i < mcasp->num_serializer; i++) + mcasp_set_reg(mcasp, DAVINCI_MCASP_XRSRCTL_REG(i), + context->xrsr_regs[i]); + return 0; } #else @@ -1195,6 +1231,11 @@ static int davinci_mcasp_probe(struct platform_device *pdev) mcasp->op_mode = pdata->op_mode; mcasp->tdm_slots = pdata->tdm_slots; mcasp->num_serializer = pdata->num_serializer; +#ifdef CONFIG_PM_SLEEP + mcasp->context.xrsr_regs = devm_kzalloc(&pdev->dev, + sizeof(u32) * mcasp->num_serializer, + GFP_KERNEL); +#endif mcasp->serial_dir = pdata->serial_dir; mcasp->version = pdata->version; mcasp->txnumevt = pdata->txnumevt; -- cgit v1.1 From cd69dc8868d64cfa2993944607d9e97927d95987 Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Wed, 1 Oct 2014 15:08:14 +0300 Subject: ASoC: rt5640: Add function for enabling DMIC from ACPI probed machine There is no code enabling DMIC clock in systems that don't provide platform data for rt5640 after commit 71d97a794301 ("ASoC: rt5640: Use the platform data for DMIC settings"). I think it's worth to keep this static DMIC clock and alternative data pin setting during probe time. For making possible to use DMIC from ACPI probed machine (prior ACPI 5.1 with _DSD) this patch moves DMIC configuration to new exported rt5640_dmic_enable() that machine drivers can call. Please note, this patch moves DMIC configuration from i2c probe to codec probe in case platform data for rt5640 is set. Signed-off-by: Jarkko Nikula Cc: Oder Chiou Signed-off-by: Mark Brown --- sound/soc/codecs/rt5640.c | 49 +++++++++++++++++++++++++++++------------------ sound/soc/codecs/rt5640.h | 3 +++ 2 files changed, 33 insertions(+), 19 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 6bc6efd..2fdcbb8 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -1906,6 +1906,32 @@ static int rt5640_set_bias_level(struct snd_soc_codec *codec, return 0; } +int rt5640_dmic_enable(struct snd_soc_codec *codec, + bool dmic1_data_pin, bool dmic2_data_pin) +{ + struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); + + regmap_update_bits(rt5640->regmap, RT5640_GPIO_CTRL1, + RT5640_GP2_PIN_MASK, RT5640_GP2_PIN_DMIC1_SCL); + + if (dmic1_data_pin) { + regmap_update_bits(rt5640->regmap, RT5640_DMIC, + RT5640_DMIC_1_DP_MASK, RT5640_DMIC_1_DP_GPIO3); + regmap_update_bits(rt5640->regmap, RT5640_GPIO_CTRL1, + RT5640_GP3_PIN_MASK, RT5640_GP3_PIN_DMIC1_SDA); + } + + if (dmic2_data_pin) { + regmap_update_bits(rt5640->regmap, RT5640_DMIC, + RT5640_DMIC_2_DP_MASK, RT5640_DMIC_2_DP_GPIO4); + regmap_update_bits(rt5640->regmap, RT5640_GPIO_CTRL1, + RT5640_GP4_PIN_MASK, RT5640_GP4_PIN_DMIC2_SDA); + } + + return 0; +} +EXPORT_SYMBOL_GPL(rt5640_dmic_enable); + static int rt5640_probe(struct snd_soc_codec *codec) { struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); @@ -1945,6 +1971,10 @@ static int rt5640_probe(struct snd_soc_codec *codec) return -ENODEV; } + if (rt5640->pdata.dmic_en) + rt5640_dmic_enable(codec, rt5640->pdata.dmic1_data_pin, + rt5640->pdata.dmic2_data_pin); + return 0; } @@ -2194,25 +2224,6 @@ static int rt5640_i2c_probe(struct i2c_client *i2c, regmap_update_bits(rt5640->regmap, RT5640_IN3_IN4, RT5640_IN_DF2, RT5640_IN_DF2); - if (rt5640->pdata.dmic_en) { - regmap_update_bits(rt5640->regmap, RT5640_GPIO_CTRL1, - RT5640_GP2_PIN_MASK, RT5640_GP2_PIN_DMIC1_SCL); - - if (rt5640->pdata.dmic1_data_pin) { - regmap_update_bits(rt5640->regmap, RT5640_DMIC, - RT5640_DMIC_1_DP_MASK, RT5640_DMIC_1_DP_GPIO3); - regmap_update_bits(rt5640->regmap, RT5640_GPIO_CTRL1, - RT5640_GP3_PIN_MASK, RT5640_GP3_PIN_DMIC1_SDA); - } - - if (rt5640->pdata.dmic2_data_pin) { - regmap_update_bits(rt5640->regmap, RT5640_DMIC, - RT5640_DMIC_2_DP_MASK, RT5640_DMIC_2_DP_GPIO4); - regmap_update_bits(rt5640->regmap, RT5640_GPIO_CTRL1, - RT5640_GP4_PIN_MASK, RT5640_GP4_PIN_DMIC2_SDA); - } - } - rt5640->hp_mute = 1; return snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5640, diff --git a/sound/soc/codecs/rt5640.h b/sound/soc/codecs/rt5640.h index 58ebe96..3deb8ba 100644 --- a/sound/soc/codecs/rt5640.h +++ b/sound/soc/codecs/rt5640.h @@ -2097,4 +2097,7 @@ struct rt5640_priv { bool hp_mute; }; +int rt5640_dmic_enable(struct snd_soc_codec *codec, + bool dmic1_data_pin, bool dmic2_data_pin); + #endif -- cgit v1.1 From a5f0ab05b67213ef33107b716e8596a480b5875f Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Wed, 1 Oct 2014 15:08:15 +0300 Subject: ASoC: Intel: byt-rt5640: Enable DMIC interface for default DAPM route It turned out DMIC interface wasn't enabled/disabled runtime for active DMIC route in the rt5640 codec driver anymore after commit 71d97a794301 ("ASoC: rt5640: Use the platform data for DMIC settings"). Since DMIC interface must be enabled explicitly either by passing platform data to rt5640 codec driver or by calling new rt5640_dmic_enable() this patch adds a DMI quirk flag that is used to conditionally enable DMIC interface during sound card init time. Signed-off-by: Jarkko Nikula Signed-off-by: Mark Brown --- sound/soc/intel/byt-rt5640.c | 16 +++++++++++++--- 1 file changed, 13 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/byt-rt5640.c b/sound/soc/intel/byt-rt5640.c index c323a10..8392c16 100644 --- a/sound/soc/intel/byt-rt5640.c +++ b/sound/soc/intel/byt-rt5640.c @@ -59,7 +59,11 @@ enum { BYT_RT5640_IN1_MAP, }; -static unsigned long byt_rt5640_custom_map = BYT_RT5640_DMIC1_MAP; +#define BYT_RT5640_MAP(quirk) ((quirk) & 0xff) +#define BYT_RT5640_DMIC_EN BIT(16) + +static unsigned long byt_rt5640_quirk = BYT_RT5640_DMIC1_MAP | + BYT_RT5640_DMIC_EN; static const struct snd_kcontrol_new byt_rt5640_controls[] = { SOC_DAPM_PIN_SWITCH("Headphone"), @@ -94,7 +98,7 @@ static int byt_rt5640_hw_params(struct snd_pcm_substream *substream, static int byt_rt5640_quirk_cb(const struct dmi_system_id *id) { - byt_rt5640_custom_map = (unsigned long)id->driver_data; + byt_rt5640_quirk = (unsigned long)id->driver_data; return 1; } @@ -129,7 +133,7 @@ static int byt_rt5640_init(struct snd_soc_pcm_runtime *runtime) } dmi_check_system(byt_rt5640_quirk_table); - switch (byt_rt5640_custom_map) { + switch (BYT_RT5640_MAP(byt_rt5640_quirk)) { case BYT_RT5640_IN1_MAP: custom_map = byt_rt5640_intmic_in1_map; num_routes = ARRAY_SIZE(byt_rt5640_intmic_in1_map); @@ -143,6 +147,12 @@ static int byt_rt5640_init(struct snd_soc_pcm_runtime *runtime) if (ret) return ret; + if (byt_rt5640_quirk & BYT_RT5640_DMIC_EN) { + ret = rt5640_dmic_enable(codec, 0, 0); + if (ret) + return ret; + } + snd_soc_dapm_ignore_suspend(dapm, "HPOL"); snd_soc_dapm_ignore_suspend(dapm, "HPOR"); -- cgit v1.1 From c05a11f7b8b5bc67f2c9f726c52b59f67b1bfe7d Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Tue, 30 Sep 2014 16:52:15 -0300 Subject: ASoC: fsl: Do not force codecs selection by SND_SOC_FSL_ASOC_CARD The wm8962 driver uses the input subsystem, but it is selected by SND_SOC_FSL_ASOC_CARD, which can be built with CONFIG_INPUT disabled, resulting in this link error: ERROR: "input_event" [sound/soc/codecs/snd-soc-wm8962.ko] undefined! ERROR: "input_register_device" [sound/soc/codecs/snd-soc-wm8962.ko] undefined! ERROR: "devm_input_allocate_device" [sound/soc/codecs/snd-soc-wm8962.ko] undefined! Do not force the selection of the codecs by SND_SOC_FSL_ASOC_CARD to avoid such problem. Reported-by: Arnd Bergmann Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/fsl/Kconfig | 3 --- 1 file changed, 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 7c1da8e..0f23d1a 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -289,9 +289,6 @@ config SND_SOC_FSL_ASOC_CARD select SND_SOC_FSL_ESAI select SND_SOC_FSL_SAI select SND_SOC_FSL_SSI - select SND_SOC_CS42XX8_I2C - select SND_SOC_SGTL5000 - select SND_SOC_WM8962 help ALSA SoC Audio support with ASRC feature for Freescale SoCs that have ESAI/SAI/SSI and connect with external CODECs such as WM8962, CS42888 -- cgit v1.1 From f3fa1bbd836a7d6efb2abd506ed8e24096f39062 Mon Sep 17 00:00:00 2001 From: Oder Chiou Date: Fri, 19 Sep 2014 19:15:45 +0800 Subject: ASoC: rt5645: Add headset detect function Add headset detect function Signed-off-by: Oder Chiou Signed-off-by: Mark Brown --- sound/soc/codecs/rt5645.c | 99 +++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/rt5645.h | 5 +++ 2 files changed, 104 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index a7762d0..3fb83bf 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -17,6 +17,7 @@ #include #include #include +#include #include #include #include @@ -2103,6 +2104,77 @@ static int rt5645_set_bias_level(struct snd_soc_codec *codec, return 0; } +static int rt5645_jack_detect(struct snd_soc_codec *codec, + struct snd_soc_jack *jack) +{ + struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); + int gpio_state, jack_type = 0; + unsigned int val; + + gpio_state = gpio_get_value(rt5645->pdata.hp_det_gpio); + + dev_dbg(codec->dev, "gpio = %d(%d)\n", rt5645->pdata.hp_det_gpio, + gpio_state); + + if ((rt5645->pdata.gpio_hp_det_active_high && gpio_state) || + (!rt5645->pdata.gpio_hp_det_active_high && !gpio_state)) { + snd_soc_dapm_force_enable_pin(&codec->dapm, "micbias1"); + snd_soc_dapm_force_enable_pin(&codec->dapm, "micbias2"); + snd_soc_dapm_force_enable_pin(&codec->dapm, "LDO2"); + snd_soc_dapm_force_enable_pin(&codec->dapm, "Mic Det Power"); + snd_soc_dapm_sync(&codec->dapm); + + snd_soc_write(codec, RT5645_IN1_CTRL1, 0x0006); + snd_soc_write(codec, RT5645_JD_CTRL3, 0x00b0); + + snd_soc_update_bits(codec, RT5645_IN1_CTRL2, + RT5645_CBJ_MN_JD, 0); + snd_soc_update_bits(codec, RT5645_IN1_CTRL2, + RT5645_CBJ_MN_JD, RT5645_CBJ_MN_JD); + + msleep(400); + val = snd_soc_read(codec, RT5645_IN1_CTRL3) & 0x7; + dev_dbg(codec->dev, "val = %d\n", val); + + if (val == 1 || val == 2) + jack_type = SND_JACK_HEADSET; + else + jack_type = SND_JACK_HEADPHONE; + + snd_soc_dapm_disable_pin(&codec->dapm, "micbias1"); + snd_soc_dapm_disable_pin(&codec->dapm, "micbias2"); + snd_soc_dapm_disable_pin(&codec->dapm, "LDO2"); + snd_soc_dapm_disable_pin(&codec->dapm, "Mic Det Power"); + snd_soc_dapm_sync(&codec->dapm); + } + + snd_soc_jack_report(rt5645->jack, jack_type, SND_JACK_HEADSET); + + return 0; +} + +int rt5645_set_jack_detect(struct snd_soc_codec *codec, + struct snd_soc_jack *jack) +{ + struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); + + rt5645->jack = jack; + + rt5645_jack_detect(codec, rt5645->jack); + + return 0; +} +EXPORT_SYMBOL_GPL(rt5645_set_jack_detect); + +static irqreturn_t rt5645_irq(int irq, void *data) +{ + struct rt5645_priv *rt5645 = data; + + rt5645_jack_detect(rt5645->codec, rt5645->jack); + + return IRQ_HANDLED; +} + static int rt5645_probe(struct snd_soc_codec *codec) { struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); @@ -2250,6 +2322,7 @@ static int rt5645_i2c_probe(struct i2c_client *i2c, if (rt5645 == NULL) return -ENOMEM; + rt5645->i2c = i2c; i2c_set_clientdata(i2c, rt5645); if (pdata) @@ -2345,12 +2418,38 @@ static int rt5645_i2c_probe(struct i2c_client *i2c, } + if (rt5645->i2c->irq) { + ret = request_threaded_irq(rt5645->i2c->irq, NULL, rt5645_irq, + IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING + | IRQF_ONESHOT, "rt5645", rt5645); + if (ret) + dev_err(&i2c->dev, "Failed to reguest IRQ: %d\n", ret); + } + + if (gpio_is_valid(rt5645->pdata.hp_det_gpio)) { + ret = gpio_request(rt5645->pdata.hp_det_gpio, "rt5645"); + if (ret) + dev_err(&i2c->dev, "Fail gpio_request hp_det_gpio\n"); + + ret = gpio_direction_input(rt5645->pdata.hp_det_gpio); + if (ret) + dev_err(&i2c->dev, "Fail gpio_direction hp_det_gpio\n"); + } + return snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5645, rt5645_dai, ARRAY_SIZE(rt5645_dai)); } static int rt5645_i2c_remove(struct i2c_client *i2c) { + struct rt5645_priv *rt5645 = i2c_get_clientdata(i2c); + + if (i2c->irq) + free_irq(i2c->irq, rt5645); + + if (gpio_is_valid(rt5645->pdata.hp_det_gpio)) + gpio_free(rt5645->pdata.hp_det_gpio); + snd_soc_unregister_codec(&i2c->dev); return 0; diff --git a/sound/soc/codecs/rt5645.h b/sound/soc/codecs/rt5645.h index 355b7e9..50c62c5 100644 --- a/sound/soc/codecs/rt5645.h +++ b/sound/soc/codecs/rt5645.h @@ -2166,6 +2166,8 @@ struct rt5645_priv { struct snd_soc_codec *codec; struct rt5645_platform_data pdata; struct regmap *regmap; + struct i2c_client *i2c; + struct snd_soc_jack *jack; int sysclk; int sysclk_src; @@ -2178,4 +2180,7 @@ struct rt5645_priv { int pll_out; }; +int rt5645_set_jack_detect(struct snd_soc_codec *codec, + struct snd_soc_jack *jack); + #endif /* __RT5645_H__ */ -- cgit v1.1 From 24221dcc8be736a2b0b83ecaeb60b99bd7e9334c Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Thu, 2 Oct 2014 13:29:08 +0300 Subject: ASoC: Intel: byt-rt5640: Add quirk for Dell Venue 8 Pro tablet It was found with help of Jan-Michael Brummer that Dell Venue 8 Pro tablet has a digital microphone connected to DMIC2 interface of the RT564x. This patch adds a DAPM route to DMIC2 and a quirk using it for that tablet. Signed-off-by: Jarkko Nikula Reported-by: Jan-Michael Brummer Signed-off-by: Mark Brown --- sound/soc/intel/byt-rt5640.c | 18 ++++++++++++++++++ 1 file changed, 18 insertions(+) (limited to 'sound') diff --git a/sound/soc/intel/byt-rt5640.c b/sound/soc/intel/byt-rt5640.c index 8392c16..a9619b4 100644 --- a/sound/soc/intel/byt-rt5640.c +++ b/sound/soc/intel/byt-rt5640.c @@ -49,6 +49,10 @@ static const struct snd_soc_dapm_route byt_rt5640_intmic_dmic1_map[] = { {"DMIC1", NULL, "Internal Mic"}, }; +static const struct snd_soc_dapm_route byt_rt5640_intmic_dmic2_map[] = { + {"DMIC2", NULL, "Internal Mic"}, +}; + static const struct snd_soc_dapm_route byt_rt5640_intmic_in1_map[] = { {"Internal Mic", NULL, "MICBIAS1"}, {"IN1P", NULL, "Internal Mic"}, @@ -56,6 +60,7 @@ static const struct snd_soc_dapm_route byt_rt5640_intmic_in1_map[] = { enum { BYT_RT5640_DMIC1_MAP, + BYT_RT5640_DMIC2_MAP, BYT_RT5640_IN1_MAP, }; @@ -111,6 +116,15 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { }, .driver_data = (unsigned long *)BYT_RT5640_IN1_MAP, }, + { + .callback = byt_rt5640_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "DellInc."), + DMI_MATCH(DMI_PRODUCT_NAME, "Venue 8 Pro 5830"), + }, + .driver_data = (unsigned long *)(BYT_RT5640_DMIC2_MAP | + BYT_RT5640_DMIC_EN), + }, {} }; @@ -138,6 +152,10 @@ static int byt_rt5640_init(struct snd_soc_pcm_runtime *runtime) custom_map = byt_rt5640_intmic_in1_map; num_routes = ARRAY_SIZE(byt_rt5640_intmic_in1_map); break; + case BYT_RT5640_DMIC2_MAP: + custom_map = byt_rt5640_intmic_dmic2_map; + num_routes = ARRAY_SIZE(byt_rt5640_intmic_dmic2_map); + break; default: custom_map = byt_rt5640_intmic_dmic1_map; num_routes = ARRAY_SIZE(byt_rt5640_intmic_dmic1_map); -- cgit v1.1 From c47a39a6806d756c34eb01b1081866845fb76dc3 Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Thu, 2 Oct 2014 13:29:09 +0300 Subject: ASoC: Intel: byt-rt5640: Set card as fully routed Although it's not known does current version of byt-rt5640 cover all possible variants it is better to set the fully_routed flag on in order to disable unused codecs pins in known machines and get regression from machines that use different routing than the default one. Signed-off-by: Jarkko Nikula Signed-off-by: Mark Brown --- sound/soc/intel/byt-rt5640.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/intel/byt-rt5640.c b/sound/soc/intel/byt-rt5640.c index a9619b4..88ad57f 100644 --- a/sound/soc/intel/byt-rt5640.c +++ b/sound/soc/intel/byt-rt5640.c @@ -209,6 +209,7 @@ static struct snd_soc_card byt_rt5640_card = { .num_dapm_widgets = ARRAY_SIZE(byt_rt5640_widgets), .dapm_routes = byt_rt5640_audio_map, .num_dapm_routes = ARRAY_SIZE(byt_rt5640_audio_map), + .fully_routed = true, }; static int byt_rt5640_probe(struct platform_device *pdev) -- cgit v1.1 From be1aa3ea1f4179cbc84c57d3b1128c49515910ac Mon Sep 17 00:00:00 2001 From: Thierry Reding Date: Thu, 2 Oct 2014 09:28:00 +0200 Subject: ASoC: tas2552: Fix compilation warning for !PM_RUNTIME The tas2552_sw_shutdown() function is only used by runtime suspend support, so only build it when necessary. Signed-off-by: Thierry Reding Signed-off-by: Mark Brown --- sound/soc/codecs/tas2552.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/tas2552.c b/sound/soc/codecs/tas2552.c index 1ed57a7..f039dc8 100644 --- a/sound/soc/codecs/tas2552.c +++ b/sound/soc/codecs/tas2552.c @@ -115,6 +115,7 @@ static const struct snd_soc_dapm_route tas2552_audio_map[] = { {"ClassD", NULL, "PLL"}, }; +#ifdef CONFIG_PM_RUNTIME static void tas2552_sw_shutdown(struct tas2552_data *tas_data, int sw_shutdown) { u8 cfg1_reg; @@ -127,6 +128,7 @@ static void tas2552_sw_shutdown(struct tas2552_data *tas_data, int sw_shutdown) snd_soc_update_bits(tas_data->codec, TAS2552_CFG_1, TAS2552_SWS_MASK, cfg1_reg); } +#endif static int tas2552_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, -- cgit v1.1 From 81f3dfe1908011ee12760ce4d75451e7446dff80 Mon Sep 17 00:00:00 2001 From: Thierry Reding Date: Thu, 2 Oct 2014 09:27:03 +0200 Subject: ASoC: rt286: Fix compilation warning for !PM The rt286_index_sync() function is only called in the resume path. If PM is disabled it becomes unused and shouldn't be built either. Signed-off-by: Thierry Reding Signed-off-by: Mark Brown --- sound/soc/codecs/rt286.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index e4f6102..2bb5a27 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -270,6 +270,7 @@ static int rt286_hw_read(void *context, unsigned int reg, unsigned int *value) return 0; } +#ifdef CONFIG_PM static void rt286_index_sync(struct snd_soc_codec *codec) { struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); @@ -280,6 +281,7 @@ static void rt286_index_sync(struct snd_soc_codec *codec) rt286->index_cache[i].def); } } +#endif static int rt286_support_power_controls[] = { RT286_DAC_OUT1, -- cgit v1.1 From 7c168d5f8bda5716e1a49040b901f26a3002517d Mon Sep 17 00:00:00 2001 From: Anatol Pomozov Date: Wed, 1 Oct 2014 10:15:57 -0700 Subject: ASoC: ssm4567: Remove duplicated else-if branch Signed-off-by: Anatol Pomozov Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/ssm4567.c | 2 -- 1 file changed, 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/ssm4567.c b/sound/soc/codecs/ssm4567.c index 1dadacb..4b5c17f 100644 --- a/sound/soc/codecs/ssm4567.c +++ b/sound/soc/codecs/ssm4567.c @@ -173,8 +173,6 @@ static int ssm4567_hw_params(struct snd_pcm_substream *substream, dacfs = SSM4567_DAC_FS_32000_48000; else if (rate >= 64000 && rate <= 96000) dacfs = SSM4567_DAC_FS_64000_96000; - else if (rate >= 64000 && rate <= 96000) - dacfs = SSM4567_DAC_FS_64000_96000; else if (rate >= 128000 && rate <= 192000) dacfs = SSM4567_DAC_FS_128000_192000; else -- cgit v1.1 From 3fe240326cc395c66eda0518b1945ea505afd1fc Mon Sep 17 00:00:00 2001 From: Dylan Reid Date: Wed, 1 Oct 2014 14:25:20 -0700 Subject: ASoC: simple-card: Add mic and hp detect gpios. Allow Headphone and Microphone jack detect gpios to be specified in device tree. This will allow a few systems including rk3288_max98090 to use simple-card instead of having their own board file. Signed-off-by: Dylan Reid Signed-off-by: Mark Brown --- sound/soc/generic/simple-card.c | 73 +++++++++++++++++++++++++++++++++++++++++ 1 file changed, 73 insertions(+) (limited to 'sound') diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 709ce67..fcb431f 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -10,10 +10,13 @@ */ #include #include +#include #include #include +#include #include #include +#include #include #include #include @@ -25,6 +28,8 @@ struct simple_card_data { struct asoc_simple_dai codec_dai; } *dai_props; unsigned int mclk_fs; + int gpio_hp_det; + int gpio_mic_det; struct snd_soc_dai_link dai_link[]; /* dynamically allocated */ }; @@ -54,6 +59,32 @@ static struct snd_soc_ops asoc_simple_card_ops = { .hw_params = asoc_simple_card_hw_params, }; +static struct snd_soc_jack simple_card_hp_jack; +static struct snd_soc_jack_pin simple_card_hp_jack_pins[] = { + { + .pin = "Headphones", + .mask = SND_JACK_HEADPHONE, + }, +}; +static struct snd_soc_jack_gpio simple_card_hp_jack_gpio = { + .name = "Headphone detection", + .report = SND_JACK_HEADPHONE, + .debounce_time = 150, +}; + +static struct snd_soc_jack simple_card_mic_jack; +static struct snd_soc_jack_pin simple_card_mic_jack_pins[] = { + { + .pin = "Mic Jack", + .mask = SND_JACK_MICROPHONE, + }, +}; +static struct snd_soc_jack_gpio simple_card_mic_jack_gpio = { + .name = "Mic detection", + .report = SND_JACK_MICROPHONE, + .debounce_time = 150, +}; + static int __asoc_simple_card_dai_init(struct snd_soc_dai *dai, struct asoc_simple_dai *set) { @@ -109,6 +140,28 @@ static int asoc_simple_card_dai_init(struct snd_soc_pcm_runtime *rtd) if (ret < 0) return ret; + if (gpio_is_valid(priv->gpio_hp_det)) { + snd_soc_jack_new(codec->codec, "Headphones", SND_JACK_HEADPHONE, + &simple_card_hp_jack); + snd_soc_jack_add_pins(&simple_card_hp_jack, + ARRAY_SIZE(simple_card_hp_jack_pins), + simple_card_hp_jack_pins); + + simple_card_hp_jack_gpio.gpio = priv->gpio_hp_det; + snd_soc_jack_add_gpios(&simple_card_hp_jack, 1, + &simple_card_hp_jack_gpio); + } + + if (gpio_is_valid(priv->gpio_mic_det)) { + snd_soc_jack_new(codec->codec, "Mic Jack", SND_JACK_MICROPHONE, + &simple_card_mic_jack); + snd_soc_jack_add_pins(&simple_card_mic_jack, + ARRAY_SIZE(simple_card_mic_jack_pins), + simple_card_mic_jack_pins); + simple_card_mic_jack_gpio.gpio = priv->gpio_mic_det; + snd_soc_jack_add_gpios(&simple_card_mic_jack, 1, + &simple_card_mic_jack_gpio); + } return 0; } @@ -383,6 +436,16 @@ static int asoc_simple_card_parse_of(struct device_node *node, return ret; } + priv->gpio_hp_det = of_get_named_gpio(node, + "simple-audio-card,hp-det-gpio", 0); + if (priv->gpio_hp_det == -EPROBE_DEFER) + return -EPROBE_DEFER; + + priv->gpio_mic_det = of_get_named_gpio(node, + "simple-audio-card,mic-det-gpio", 0); + if (priv->gpio_mic_det == -EPROBE_DEFER) + return -EPROBE_DEFER; + if (!priv->snd_card.name) priv->snd_card.name = priv->snd_card.dai_link->name; @@ -502,6 +565,16 @@ err: static int asoc_simple_card_remove(struct platform_device *pdev) { + struct snd_soc_card *card = platform_get_drvdata(pdev); + struct simple_card_data *priv = snd_soc_card_get_drvdata(card); + + if (gpio_is_valid(priv->gpio_hp_det)) + snd_soc_jack_free_gpios(&simple_card_hp_jack, 1, + &simple_card_hp_jack_gpio); + if (gpio_is_valid(priv->gpio_mic_det)) + snd_soc_jack_free_gpios(&simple_card_mic_jack, 1, + &simple_card_mic_jack_gpio); + return asoc_simple_card_unref(pdev); } -- cgit v1.1 From 5dc0158a27f65e7efaa6e3cc496d93b4c4c65d19 Mon Sep 17 00:00:00 2001 From: "Subhransu S. Prusty" Date: Fri, 19 Sep 2014 16:46:05 +0530 Subject: ASoC: Export dapm_kcontrol_get_value The DSP driver needs to know widget control value in its event handler for widgets like mixers. This is required in the subsequent patches Signed-off-by: Subhransu S. Prusty Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 8348352..08c79f0 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -326,12 +326,13 @@ static struct list_head *dapm_kcontrol_get_path_list( list_for_each_entry(path, dapm_kcontrol_get_path_list(kcontrol), \ list_kcontrol) -static unsigned int dapm_kcontrol_get_value(const struct snd_kcontrol *kcontrol) +unsigned int dapm_kcontrol_get_value(const struct snd_kcontrol *kcontrol) { struct dapm_kcontrol_data *data = snd_kcontrol_chip(kcontrol); return data->value; } +EXPORT_SYMBOL_GPL(dapm_kcontrol_get_value); static bool dapm_kcontrol_set_value(const struct snd_kcontrol *kcontrol, unsigned int value) -- cgit v1.1 From a577483b6906b3d7aba9cc07e383682fc9b65318 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Fri, 3 Oct 2014 09:55:07 +0800 Subject: ASoC: rt286: Add depends on I2C rt286 use I2C as its I/O. So the driver can only available when I2C is selected. Signed-off-by: Bard Liao Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 8838838e..00ae24b 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -464,6 +464,7 @@ config SND_SOC_RL6231 config SND_SOC_RT286 tristate + depends on I2C config SND_SOC_RT5631 tristate -- cgit v1.1 From fa558d0130debf847b6b8cd95880a2d7556770ac Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Thu, 2 Oct 2014 16:16:50 -0300 Subject: ASoC: sgtl5000: Improve the error message on slave mode setting For sgtl5000 to operate in slave mode it can only work in "Synchronous SYS_MCLK input" mode. In this mode only the following rates can be supported: 256*Fs, 384*Fs, 512*Fs. Improve the error message to give a better indication as to why the clocking failed for slave mode: [ 12.515399] sgtl5000 1-000a: PLL not supported in slave mode [ 12.524124] sgtl5000 1-000a: 233 ratio is not supported. SYS_MCLK needs to be 256, 384 or 512 * fs [ 12.535938] sgtl5000 1-000a: ASoC: can't set sgtl5000 hw params: -22 Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/codecs/sgtl5000.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index e997d27..7ef2687 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -626,6 +626,9 @@ static int sgtl5000_set_clock(struct snd_soc_codec *codec, int frame_rate) } else { dev_err(codec->dev, "PLL not supported in slave mode\n"); + dev_err(codec->dev, "%d ratio is not supported. " + "SYS_MCLK needs to be 256, 384 or 512 * fs\n", + sgtl5000->sysclk / sys_fs); return -EINVAL; } } -- cgit v1.1 From 6f4d2b3177ee3352e70c90f327e2dea3809c263e Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Thu, 2 Oct 2014 17:36:05 -0300 Subject: ASoC: sgtl5000: Do a sanity check on SYS_MCLK According to the sgtl5000 datasheet the valid range for SYS_MCLK is from 8 to 27 MHz. Add a sanity check prior to enabling SYS_MCLK. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/codecs/sgtl5000.c | 9 +++++++++ 1 file changed, 9 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 7ef2687..3e9db43 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -1445,6 +1445,7 @@ static int sgtl5000_i2c_probe(struct i2c_client *client, { struct sgtl5000_priv *sgtl5000; int ret, reg, rev; + unsigned int mclk; sgtl5000 = devm_kzalloc(&client->dev, sizeof(struct sgtl5000_priv), GFP_KERNEL); @@ -1468,6 +1469,14 @@ static int sgtl5000_i2c_probe(struct i2c_client *client, return ret; } + /* SGTL5000 SYS_MCLK should be between 8 and 27 MHz */ + mclk = clk_get_rate(sgtl5000->mclk); + if (mclk < 8000000 || mclk > 27000000) { + dev_err(&client->dev, "Invalid SYS_CLK frequency: %u.%03uMHz\n", + mclk / 1000000, mclk / 1000 % 1000); + return -EINVAL; + } + ret = clk_prepare_enable(sgtl5000->mclk); if (ret) return ret; -- cgit v1.1 From 9766a1cfe5ef2042d1e604e2223629dc43307a21 Mon Sep 17 00:00:00 2001 From: Dylan Reid Date: Thu, 2 Oct 2014 09:42:44 -0700 Subject: ASoC: tegra: add mic detect gpio to tegra_max98090 Add an optional mic detect gpio property. If specified in device tree there will be a mic jack created for the given gpio. This will be used by the Tegra-based Chromebooks. Signed-off-by: Dylan Reid Reviewed-by: Stephen Warren Signed-off-by: Mark Brown --- sound/soc/tegra/tegra_max98090.c | 40 ++++++++++++++++++++++++++++++++++++++++ 1 file changed, 40 insertions(+) (limited to 'sound') diff --git a/sound/soc/tegra/tegra_max98090.c b/sound/soc/tegra/tegra_max98090.c index b86cd99..01921d7 100644 --- a/sound/soc/tegra/tegra_max98090.c +++ b/sound/soc/tegra/tegra_max98090.c @@ -42,6 +42,7 @@ struct tegra_max98090 { struct tegra_asoc_utils_data util_data; int gpio_hp_det; + int gpio_mic_det; }; static int tegra_max98090_asoc_hw_params(struct snd_pcm_substream *substream, @@ -112,6 +113,22 @@ static struct snd_soc_jack_gpio tegra_max98090_hp_jack_gpio = { .invert = 1, }; +static struct snd_soc_jack tegra_max98090_mic_jack; + +static struct snd_soc_jack_pin tegra_max98090_mic_jack_pins[] = { + { + .pin = "Mic Jack", + .mask = SND_JACK_MICROPHONE, + }, +}; + +static struct snd_soc_jack_gpio tegra_max98090_mic_jack_gpio = { + .name = "Mic detection", + .report = SND_JACK_MICROPHONE, + .debounce_time = 150, + .invert = 1, +}; + static const struct snd_soc_dapm_widget tegra_max98090_dapm_widgets[] = { SND_SOC_DAPM_HP("Headphones", NULL), SND_SOC_DAPM_SPK("Speakers", NULL), @@ -141,6 +158,19 @@ static int tegra_max98090_asoc_init(struct snd_soc_pcm_runtime *rtd) &tegra_max98090_hp_jack_gpio); } + if (gpio_is_valid(machine->gpio_mic_det)) { + snd_soc_jack_new(codec, "Mic Jack", SND_JACK_MICROPHONE, + &tegra_max98090_mic_jack); + snd_soc_jack_add_pins(&tegra_max98090_mic_jack, + ARRAY_SIZE(tegra_max98090_mic_jack_pins), + tegra_max98090_mic_jack_pins); + + tegra_max98090_mic_jack_gpio.gpio = machine->gpio_mic_det; + snd_soc_jack_add_gpios(&tegra_max98090_mic_jack, + 1, + &tegra_max98090_mic_jack_gpio); + } + return 0; } @@ -153,6 +183,11 @@ static int tegra_max98090_card_remove(struct snd_soc_card *card) &tegra_max98090_hp_jack_gpio); } + if (gpio_is_valid(machine->gpio_mic_det)) { + snd_soc_jack_free_gpios(&tegra_max98090_mic_jack, 1, + &tegra_max98090_mic_jack_gpio); + } + return 0; } @@ -201,6 +236,11 @@ static int tegra_max98090_probe(struct platform_device *pdev) if (machine->gpio_hp_det == -EPROBE_DEFER) return -EPROBE_DEFER; + machine->gpio_mic_det = + of_get_named_gpio(np, "nvidia,mic-det-gpios", 0); + if (machine->gpio_mic_det == -EPROBE_DEFER) + return -EPROBE_DEFER; + ret = snd_soc_of_parse_card_name(card, "nvidia,model"); if (ret) goto err; -- cgit v1.1 From b2d9de549c30170eed5691d369cf16680e0ce03a Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Fri, 3 Oct 2014 15:32:40 +0300 Subject: ASoC: dapm: Fix NULL pointer dereference when registering card with widgets Commit 0bd2ac3dae74 ("ASoC: Remove CODEC pointer from snd_soc_dapm_context") introduced regression to snd_soc_dapm_new_controls() when registering a card with card->dapm_widgets set. Call chain is: snd_soc_register_card() -> snd_soc_instantiate_card() -> snd_soc_dapm_new_controls() -> snd_soc_dapm_new_control() Null pointer dereference occurs since card->dapm context doesn't have associated component. Fix this by setting widget codec pointer conditionally. Signed-off-by: Jarkko Nikula Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 1f1e965..231deb2 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3107,7 +3107,8 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, } w->dapm = dapm; - w->codec = dapm->component->codec; + if (dapm->component) + w->codec = dapm->component->codec; INIT_LIST_HEAD(&w->sources); INIT_LIST_HEAD(&w->sinks); INIT_LIST_HEAD(&w->list); -- cgit v1.1 From 31d9f8faf9a54c851e835af489c82f45105a442f Mon Sep 17 00:00:00 2001 From: Dmitry Lavnikevich Date: Fri, 3 Oct 2014 16:18:56 +0300 Subject: ASoC: tlv320aic3x: fix PLL D configuration Current caching implementation during regcache_sync() call bypasses all register writes of values that are already known as default (regmap reg_defaults). Same time in TLV320AIC3x codecs register 5 (AIC3X_PLL_PROGC_REG) write should be immediately followed by register 6 write (AIC3X_PLL_PROGD_REG) even if it was not changed. Otherwise both registers will not be written. This brings to issue that appears particulary in case of 44.1kHz playback with 19.2MHz master clock. In this case AIC3X_PLL_PROGC_REG is 0x6e while AIC3X_PLL_PROGD_REG is 0x0 (same as register default). Thus AIC3X_PLL_PROGC_REG also remains not written and we get wrong playback speed. In this patch snd_soc_read() is used to get cached pll values and snd_soc_write() (unlike regcache_sync() this function doesn't bypasses hardware default values) to write them to registers. Signed-off-by: Dmitry Lavnikevich Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/tlv320aic3x.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 64f179e..5e8626a 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1121,6 +1121,7 @@ static int aic3x_regulator_event(struct notifier_block *nb, static int aic3x_set_power(struct snd_soc_codec *codec, int power) { struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec); + unsigned int pll_c, pll_d; int ret; if (power) { @@ -1138,6 +1139,18 @@ static int aic3x_set_power(struct snd_soc_codec *codec, int power) /* Sync reg_cache with the hardware */ regcache_cache_only(aic3x->regmap, false); regcache_sync(aic3x->regmap); + + /* Rewrite paired PLL D registers in case cached sync skipped + * writing one of them and thus caused other one also not + * being written + */ + pll_c = snd_soc_read(codec, AIC3X_PLL_PROGC_REG); + pll_d = snd_soc_read(codec, AIC3X_PLL_PROGD_REG); + if (pll_c == aic3x_reg[AIC3X_PLL_PROGC_REG].def || + pll_d == aic3x_reg[AIC3X_PLL_PROGD_REG].def) { + snd_soc_write(codec, AIC3X_PLL_PROGC_REG, pll_c); + snd_soc_write(codec, AIC3X_PLL_PROGD_REG, pll_d); + } } else { /* * Do soft reset to this codec instance in order to clear -- cgit v1.1 From 58a9014ae6422325f12d54b5dbb95531009ab70f Mon Sep 17 00:00:00 2001 From: Tomeu Vizoso Date: Fri, 3 Oct 2014 17:54:13 +0200 Subject: ASoC: fsl_spdif: Remove unused includes of linux/clk-private.h Signed-off-by: Tomeu Vizoso Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_spdif.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index 70acfe4..5bda323 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -15,7 +15,6 @@ #include #include -#include #include #include #include -- cgit v1.1 From cd9241e44af3d49977c39ddadbefbb719e2a4baf Mon Sep 17 00:00:00 2001 From: Masanari Iida Date: Sat, 4 Oct 2014 02:17:08 +0900 Subject: ASoC: da732x: Remove unnecessary KERN_ERR in pr_err() This patch remove unnecessary KERN_ERR in pr_err(). Signed-off-by: Masanari Iida Signed-off-by: Mark Brown --- sound/soc/codecs/da732x.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/da732x.c b/sound/soc/codecs/da732x.c index 2fae31c..fa15fa1 100644 --- a/sound/soc/codecs/da732x.c +++ b/sound/soc/codecs/da732x.c @@ -217,7 +217,7 @@ static void da732x_set_charge_pump(struct snd_soc_codec *codec, int state) snd_soc_write(codec, DA732X_REG_CP_CTRL1, DA723X_CP_DIS); break; default: - pr_err(KERN_ERR "Wrong charge pump state\n"); + pr_err("Wrong charge pump state\n"); break; } } -- cgit v1.1 From bb78cdd4914df22bdf233a9cd4b554a1f6e39804 Mon Sep 17 00:00:00 2001 From: Fengguang Wu Date: Sat, 4 Oct 2014 19:09:33 +0100 Subject: ASoC: Intel: byt-rt5640: fix coccinelle warnings sound/soc/intel/byt-rt5640.c:140:2-3: Unneeded semicolon Removes unneeded semicolon. Generated by: scripts/coccinelle/misc/semicolon.cocci Signed-off-by: Fengguang Wu Signed-off-by: Mark Brown --- sound/soc/intel/byt-rt5640.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/intel/byt-rt5640.c b/sound/soc/intel/byt-rt5640.c index 88ad57f..e03abdf 100644 --- a/sound/soc/intel/byt-rt5640.c +++ b/sound/soc/intel/byt-rt5640.c @@ -159,7 +159,7 @@ static int byt_rt5640_init(struct snd_soc_pcm_runtime *runtime) default: custom_map = byt_rt5640_intmic_dmic1_map; num_routes = ARRAY_SIZE(byt_rt5640_intmic_dmic1_map); - }; + } ret = snd_soc_dapm_add_routes(dapm, custom_map, num_routes); if (ret) -- cgit v1.1