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authorLinus Torvalds <torvalds@linux-foundation.org>2012-07-24 13:37:37 -0700
committerLinus Torvalds <torvalds@linux-foundation.org>2012-07-24 13:37:37 -0700
commitdbf7b5915b39bfff548e4c6a3a753fc291a60e25 (patch)
tree55c457a22aa869d2ab558317877138369ae5f9bb /sound/soc
parentd14b7a419a664cd7c1c585c9e7fffee9e9051d53 (diff)
parentc1b623d9e4117d18d244e9b7fb30d2c27aeaf074 (diff)
downloadop-kernel-dev-dbf7b5915b39bfff548e4c6a3a753fc291a60e25.zip
op-kernel-dev-dbf7b5915b39bfff548e4c6a3a753fc291a60e25.tar.gz
Merge tag 'sound-3.6' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound update from Takashi Iwai: "This is a fairly quiet release in all sound area. Only a little bit of changes in the core side while most of changes are seen in the drivers. HD-audio: - A few new codec additions for Nvidia, Realtek and VIA - Intel Haswell audio support - Support for "phantom" jacks for consistent jack reporting - Major clean-ups in HDMI/DP driver codes - A workaround for inverted digital-mic pins with Realtek codecs - Removal of beep_mode=2 option ASoC: - Added the ability to add and remove DAPM paths dynamically, mostly for reparenting on clock changes - New machine drivers for Marvell Brownstone, ST-Ericsson Ux500 reference platform and ttc-dkp - New CPU drivers for Blackfin BF6xx SPORTs in I2S mode, Marvell MMP, Synopsis Designware I2S controllers, and SPEAr DMA and S/PDIF - New CODEC drivers for Dialog DA732x, ST STA529, ST-Ericsson AB8500, TI Isabelle and Wolfson Microelectronics WM5102 and WM5110 - DAPM fixes for the recent locking changes - Fix for _PRE and _POST widgets (which have been broken for a few releases now) - A couple of minor driver updates Misc - Conversion to new dev_pm_ops in platform and PCI drivers - LTC support and some fixes in PCXHR driver - A few fixes and PM support for ISA OPti9xx and WSS cards - Some TLV code cleanup - Move driver-specific headers from include/sound to local dirs" * tag 'sound-3.6' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (212 commits) ASoC: dapm: Fix _PRE and _POST events for DAPM performance improvements ALSA: hda - add dock support for Thinkpad X230 Tablet ALSA: hda - Turn on PIN_OUT from hdmi playback prepare. ASoC imx-audmux: add MX31_AUDMUX_PORT7_SSI_PINS_7 define ASoC: littlemill: Add userspace control of the WM1250 I/O ASoC: wm8994: Update micdet for irqdomain conversion ALSA: hda - make sure alc268 does not OOPS on codec parse ALSA: hda - Add support for Realtek ALC282 ALSA: hda - Fix index number conflicts of phantom jacks ALSA: opti9xx: Fix section mismatch by PM support ALSA: snd-opti9xx: Implement suspend/resume ALSA: hda - Add new GPU codec ID to snd-hda ALSA: hda - Fix driver type of Haswell controller to AZX_DRIVER_SCH ALSA: hda - add Haswell HDMI codec id ALSA: hda - Add DeviceID for Haswell HDA ALSA: wss_lib: Fix resume on Yamaha OPL3-SAx ALSA: wss_lib: fix suspend/resume ALSA: es1938: replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ALSA: tlv: add DECLARE_TLV_DB_RANGE() ALSA: tlv: add DECLARE_TLV_CONTAINER() ...
Diffstat (limited to 'sound/soc')
-rw-r--r--sound/soc/Kconfig1
-rw-r--r--sound/soc/Makefile1
-rw-r--r--sound/soc/blackfin/Kconfig21
-rw-r--r--sound/soc/blackfin/Makefile4
-rw-r--r--sound/soc/blackfin/bf6xx-i2s.c234
-rw-r--r--sound/soc/blackfin/bf6xx-sport.c422
-rw-r--r--sound/soc/blackfin/bf6xx-sport.h82
-rw-r--r--sound/soc/codecs/Kconfig31
-rw-r--r--sound/soc/codecs/Makefile19
-rw-r--r--sound/soc/codecs/ab8500-codec.c2522
-rw-r--r--sound/soc/codecs/ab8500-codec.h590
-rw-r--r--sound/soc/codecs/ac97.c6
-rw-r--r--sound/soc/codecs/arizona.c937
-rw-r--r--sound/soc/codecs/arizona.h159
-rw-r--r--sound/soc/codecs/cs42l52.c19
-rw-r--r--sound/soc/codecs/cs42l73.c20
-rw-r--r--sound/soc/codecs/da732x.c1627
-rw-r--r--sound/soc/codecs/da732x.h133
-rw-r--r--sound/soc/codecs/da732x_reg.h654
-rw-r--r--sound/soc/codecs/isabelle.c1176
-rw-r--r--sound/soc/codecs/isabelle.h143
-rw-r--r--sound/soc/codecs/lm49453.c3
-rw-r--r--sound/soc/codecs/max98095.c5
-rw-r--r--sound/soc/codecs/ml26124.c5
-rw-r--r--sound/soc/codecs/spdif_receiver.c67
-rw-r--r--sound/soc/codecs/sta529.c442
-rw-r--r--sound/soc/codecs/tlv320aic3x.c40
-rw-r--r--sound/soc/codecs/tlv320aic3x.h27
-rw-r--r--sound/soc/codecs/twl6040.c2
-rw-r--r--sound/soc/codecs/wm1250-ev1.c7
-rw-r--r--sound/soc/codecs/wm2000.c32
-rw-r--r--sound/soc/codecs/wm5100-tables.c2
-rw-r--r--sound/soc/codecs/wm5100.c11
-rw-r--r--sound/soc/codecs/wm5102.c903
-rw-r--r--sound/soc/codecs/wm5102.h21
-rw-r--r--sound/soc/codecs/wm5110.c950
-rw-r--r--sound/soc/codecs/wm5110.h21
-rw-r--r--sound/soc/codecs/wm8350.c22
-rw-r--r--sound/soc/codecs/wm8400.c2
-rw-r--r--sound/soc/codecs/wm8580.c2
-rw-r--r--sound/soc/codecs/wm8731.c1
-rw-r--r--sound/soc/codecs/wm8741.c2
-rw-r--r--sound/soc/codecs/wm8753.c2
-rw-r--r--sound/soc/codecs/wm8776.c2
-rw-r--r--sound/soc/codecs/wm8804.c2
-rw-r--r--sound/soc/codecs/wm8903.c316
-rw-r--r--sound/soc/codecs/wm8904.c272
-rw-r--r--sound/soc/codecs/wm8960.c2
-rw-r--r--sound/soc/codecs/wm8961.c2
-rw-r--r--sound/soc/codecs/wm8962.c8
-rw-r--r--sound/soc/codecs/wm8993.c2
-rw-r--r--sound/soc/codecs/wm8994.c46
-rw-r--r--sound/soc/codecs/wm8996.c587
-rw-r--r--sound/soc/codecs/wm9081.c2
-rw-r--r--sound/soc/codecs/wm9090.c2
-rw-r--r--sound/soc/codecs/wm9712.c2
-rw-r--r--sound/soc/codecs/wm9713.c2
-rw-r--r--sound/soc/codecs/wm_hubs.c2
-rw-r--r--sound/soc/dwc/Kconfig9
-rw-r--r--sound/soc/dwc/Makefile3
-rw-r--r--sound/soc/dwc/designware_i2s.c455
-rw-r--r--sound/soc/ep93xx/ep93xx-pcm.c2
-rw-r--r--sound/soc/fsl/imx-audmux.c2
-rw-r--r--sound/soc/fsl/imx-audmux.h1
-rw-r--r--sound/soc/fsl/imx-mc13783.c49
-rw-r--r--sound/soc/fsl/imx-pcm-dma.c2
-rw-r--r--sound/soc/fsl/imx-sgtl5000.c3
-rw-r--r--sound/soc/mxs/mxs-pcm.c2
-rw-r--r--sound/soc/mxs/mxs-sgtl5000.c2
-rw-r--r--sound/soc/omap/omap-mcpdm.c1
-rw-r--r--sound/soc/pxa/Kconfig42
-rw-r--r--sound/soc/pxa/Makefile8
-rw-r--r--sound/soc/pxa/brownstone.c174
-rw-r--r--sound/soc/pxa/mioa701_wm9713.c33
-rw-r--r--sound/soc/pxa/mmp-pcm.c297
-rw-r--r--sound/soc/pxa/mmp-sspa.c480
-rw-r--r--sound/soc/pxa/mmp-sspa.h92
-rw-r--r--sound/soc/pxa/ttc-dkb.c173
-rw-r--r--sound/soc/samsung/littlemill.c7
-rw-r--r--sound/soc/samsung/s3c2412-i2s.c10
-rw-r--r--sound/soc/samsung/s3c24xx-i2s.c10
-rw-r--r--sound/soc/samsung/smdk_wm8994.c36
-rw-r--r--sound/soc/sh/fsi.c48
-rw-r--r--sound/soc/soc-core.c328
-rw-r--r--sound/soc/soc-dapm.c192
-rw-r--r--sound/soc/soc-dmaengine-pcm.c33
-rw-r--r--sound/soc/soc-io.c15
-rw-r--r--sound/soc/soc-pcm.c12
-rw-r--r--sound/soc/spear/spdif_in.c297
-rw-r--r--sound/soc/spear/spdif_in_regs.h60
-rw-r--r--sound/soc/spear/spdif_out.c389
-rw-r--r--sound/soc/spear/spdif_out_regs.h79
-rw-r--r--sound/soc/spear/spear_pcm.c214
-rw-r--r--sound/soc/tegra/Kconfig3
-rw-r--r--sound/soc/tegra/tegra20_i2s.c94
-rw-r--r--sound/soc/tegra/tegra20_i2s.h1
-rw-r--r--sound/soc/tegra/tegra20_spdif.c36
-rw-r--r--sound/soc/tegra/tegra20_spdif.h1
-rw-r--r--sound/soc/tegra/tegra30_i2s.c85
-rw-r--r--sound/soc/tegra/tegra30_i2s.h1
-rw-r--r--sound/soc/tegra/tegra_alc5632.c32
-rw-r--r--sound/soc/tegra/tegra_pcm.c115
-rw-r--r--sound/soc/tegra/tegra_pcm.h2
-rw-r--r--sound/soc/tegra/tegra_wm8753.c8
-rw-r--r--sound/soc/tegra/tegra_wm8903.c259
-rw-r--r--sound/soc/tegra/trimslice.c30
-rw-r--r--sound/soc/ux500/Kconfig18
-rw-r--r--sound/soc/ux500/Makefile6
-rw-r--r--sound/soc/ux500/mop500.c113
-rw-r--r--sound/soc/ux500/mop500_ab8500.c431
-rw-r--r--sound/soc/ux500/mop500_ab8500.h22
-rw-r--r--sound/soc/ux500/ux500_msp_dai.c2
-rw-r--r--sound/soc/ux500/ux500_msp_i2s.c2
-rw-r--r--sound/soc/ux500/ux500_pcm.c318
-rw-r--r--sound/soc/ux500/ux500_pcm.h35
115 files changed, 16425 insertions, 1368 deletions
diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig
index 40b2ad1..c5de0a8 100644
--- a/sound/soc/Kconfig
+++ b/sound/soc/Kconfig
@@ -33,6 +33,7 @@ source "sound/soc/atmel/Kconfig"
source "sound/soc/au1x/Kconfig"
source "sound/soc/blackfin/Kconfig"
source "sound/soc/davinci/Kconfig"
+source "sound/soc/dwc/Kconfig"
source "sound/soc/ep93xx/Kconfig"
source "sound/soc/fsl/Kconfig"
source "sound/soc/jz4740/Kconfig"
diff --git a/sound/soc/Makefile b/sound/soc/Makefile
index 70990f4..00a555a 100644
--- a/sound/soc/Makefile
+++ b/sound/soc/Makefile
@@ -11,6 +11,7 @@ obj-$(CONFIG_SND_SOC) += atmel/
obj-$(CONFIG_SND_SOC) += au1x/
obj-$(CONFIG_SND_SOC) += blackfin/
obj-$(CONFIG_SND_SOC) += davinci/
+obj-$(CONFIG_SND_SOC) += dwc/
obj-$(CONFIG_SND_SOC) += ep93xx/
obj-$(CONFIG_SND_SOC) += fsl/
obj-$(CONFIG_SND_SOC) += jz4740/
diff --git a/sound/soc/blackfin/Kconfig b/sound/soc/blackfin/Kconfig
index 9f6bc55..16b88f5 100644
--- a/sound/soc/blackfin/Kconfig
+++ b/sound/soc/blackfin/Kconfig
@@ -1,7 +1,8 @@
config SND_BF5XX_I2S
- tristate "SoC I2S Audio for the ADI BF5xx chip"
+ tristate "SoC I2S Audio for the ADI Blackfin chip"
depends on BLACKFIN
- select SND_BF5XX_SOC_SPORT
+ select SND_BF5XX_SOC_SPORT if !BF60x
+ select SND_BF6XX_SOC_SPORT if BF60x
help
Say Y or M if you want to add support for codecs attached to
the Blackfin SPORT (synchronous serial ports) interface in I2S
@@ -9,12 +10,14 @@ config SND_BF5XX_I2S
You will also need to select the audio interfaces to support below.
config SND_BF5XX_SOC_SSM2602
- tristate "SoC SSM2602 Audio support for BF52x ezkit"
+ tristate "SoC SSM2602 Audio Codec Add-On Card support"
depends on SND_BF5XX_I2S && (SPI_MASTER || I2C)
- select SND_BF5XX_SOC_I2S
+ select SND_BF5XX_SOC_I2S if !BF60x
+ select SND_BF6XX_SOC_I2S if BF60x
select SND_SOC_SSM2602
help
- Say Y if you want to add support for SoC audio on BF527-EZKIT.
+ Say Y if you want to add support for the Analog Devices
+ SSM2602 Audio Codec Add-On Card.
config SND_SOC_BFIN_EVAL_ADAU1701
tristate "Support for the EVAL-ADAU1701MINIZ board on Blackfin eval boards"
@@ -162,9 +165,15 @@ config SND_BF5XX_SOC_AD1980
config SND_BF5XX_SOC_SPORT
tristate
+config SND_BF6XX_SOC_SPORT
+ tristate
+
config SND_BF5XX_SOC_I2S
tristate
+config SND_BF6XX_SOC_I2S
+ tristate
+
config SND_BF5XX_SOC_TDM
tristate
@@ -173,7 +182,7 @@ config SND_BF5XX_SOC_AC97
config SND_BF5XX_SPORT_NUM
int "Set a SPORT for Sound chip"
- depends on (SND_BF5XX_I2S || SND_BF5XX_AC97 || SND_BF5XX_TDM)
+ depends on (SND_BF5XX_SOC_SPORT || SND_BF6XX_SOC_SPORT)
range 0 3 if BF54x
range 0 1 if !BF54x
default 0
diff --git a/sound/soc/blackfin/Makefile b/sound/soc/blackfin/Makefile
index 1bf86cc..6fea1f4 100644
--- a/sound/soc/blackfin/Makefile
+++ b/sound/soc/blackfin/Makefile
@@ -3,16 +3,20 @@ snd-bf5xx-ac97-objs := bf5xx-ac97-pcm.o
snd-bf5xx-i2s-objs := bf5xx-i2s-pcm.o
snd-bf5xx-tdm-objs := bf5xx-tdm-pcm.o
snd-soc-bf5xx-sport-objs := bf5xx-sport.o
+snd-soc-bf6xx-sport-objs := bf6xx-sport.o
snd-soc-bf5xx-ac97-objs := bf5xx-ac97.o
snd-soc-bf5xx-i2s-objs := bf5xx-i2s.o
+snd-soc-bf6xx-i2s-objs := bf6xx-i2s.o
snd-soc-bf5xx-tdm-objs := bf5xx-tdm.o
obj-$(CONFIG_SND_BF5XX_AC97) += snd-bf5xx-ac97.o
obj-$(CONFIG_SND_BF5XX_I2S) += snd-bf5xx-i2s.o
obj-$(CONFIG_SND_BF5XX_TDM) += snd-bf5xx-tdm.o
obj-$(CONFIG_SND_BF5XX_SOC_SPORT) += snd-soc-bf5xx-sport.o
+obj-$(CONFIG_SND_BF6XX_SOC_SPORT) += snd-soc-bf6xx-sport.o
obj-$(CONFIG_SND_BF5XX_SOC_AC97) += snd-soc-bf5xx-ac97.o
obj-$(CONFIG_SND_BF5XX_SOC_I2S) += snd-soc-bf5xx-i2s.o
+obj-$(CONFIG_SND_BF6XX_SOC_I2S) += snd-soc-bf6xx-i2s.o
obj-$(CONFIG_SND_BF5XX_SOC_TDM) += snd-soc-bf5xx-tdm.o
# Blackfin Machine Support
diff --git a/sound/soc/blackfin/bf6xx-i2s.c b/sound/soc/blackfin/bf6xx-i2s.c
new file mode 100644
index 0000000..c3c2466
--- /dev/null
+++ b/sound/soc/blackfin/bf6xx-i2s.c
@@ -0,0 +1,234 @@
+/*
+ * bf6xx-i2s.c - Analog Devices BF6XX i2s interface driver
+ *
+ * Copyright (c) 2012 Analog Devices Inc.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ */
+
+#include <linux/device.h>
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dai.h>
+
+#include "bf6xx-sport.h"
+
+struct sport_params param;
+
+static int bfin_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai,
+ unsigned int fmt)
+{
+ struct sport_device *sport = snd_soc_dai_get_drvdata(cpu_dai);
+ struct device *dev = &sport->pdev->dev;
+ int ret = 0;
+
+ param.spctl &= ~(SPORT_CTL_OPMODE | SPORT_CTL_CKRE | SPORT_CTL_FSR
+ | SPORT_CTL_LFS | SPORT_CTL_LAFS);
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ param.spctl |= SPORT_CTL_OPMODE | SPORT_CTL_CKRE
+ | SPORT_CTL_LFS;
+ break;
+ case SND_SOC_DAIFMT_DSP_A:
+ param.spctl |= SPORT_CTL_FSR;
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ param.spctl |= SPORT_CTL_OPMODE | SPORT_CTL_LFS
+ | SPORT_CTL_LAFS;
+ break;
+ default:
+ dev_err(dev, "%s: Unknown DAI format type\n", __func__);
+ ret = -EINVAL;
+ break;
+ }
+
+ param.spctl &= ~(SPORT_CTL_ICLK | SPORT_CTL_IFS);
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBM_CFM:
+ break;
+ case SND_SOC_DAIFMT_CBS_CFS:
+ case SND_SOC_DAIFMT_CBM_CFS:
+ case SND_SOC_DAIFMT_CBS_CFM:
+ ret = -EINVAL;
+ break;
+ default:
+ dev_err(dev, "%s: Unknown DAI master type\n", __func__);
+ ret = -EINVAL;
+ break;
+ }
+
+ return ret;
+}
+
+static int bfin_i2s_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct sport_device *sport = snd_soc_dai_get_drvdata(dai);
+ struct device *dev = &sport->pdev->dev;
+ int ret = 0;
+
+ param.spctl &= ~SPORT_CTL_SLEN;
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S8:
+ param.spctl |= 0x70;
+ sport->wdsize = 1;
+ case SNDRV_PCM_FORMAT_S16_LE:
+ param.spctl |= 0xf0;
+ sport->wdsize = 2;
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ param.spctl |= 0x170;
+ sport->wdsize = 3;
+ break;
+ case SNDRV_PCM_FORMAT_S32_LE:
+ param.spctl |= 0x1f0;
+ sport->wdsize = 4;
+ break;
+ }
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ ret = sport_set_tx_params(sport, &param);
+ if (ret) {
+ dev_err(dev, "SPORT tx is busy!\n");
+ return ret;
+ }
+ } else {
+ ret = sport_set_rx_params(sport, &param);
+ if (ret) {
+ dev_err(dev, "SPORT rx is busy!\n");
+ return ret;
+ }
+ }
+ return 0;
+}
+
+#ifdef CONFIG_PM
+static int bfin_i2s_suspend(struct snd_soc_dai *dai)
+{
+ struct sport_device *sport = snd_soc_dai_get_drvdata(dai);
+
+ if (dai->capture_active)
+ sport_rx_stop(sport);
+ if (dai->playback_active)
+ sport_tx_stop(sport);
+ return 0;
+}
+
+static int bfin_i2s_resume(struct snd_soc_dai *dai)
+{
+ struct sport_device *sport = snd_soc_dai_get_drvdata(dai);
+ struct device *dev = &sport->pdev->dev;
+ int ret;
+
+ ret = sport_set_tx_params(sport, &param);
+ if (ret) {
+ dev_err(dev, "SPORT tx is busy!\n");
+ return ret;
+ }
+ ret = sport_set_rx_params(sport, &param);
+ if (ret) {
+ dev_err(dev, "SPORT rx is busy!\n");
+ return ret;
+ }
+
+ return 0;
+}
+
+#else
+#define bfin_i2s_suspend NULL
+#define bfin_i2s_resume NULL
+#endif
+
+#define BFIN_I2S_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
+ SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | \
+ SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | \
+ SNDRV_PCM_RATE_96000)
+
+#define BFIN_I2S_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE | \
+ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
+
+static struct snd_soc_dai_ops bfin_i2s_dai_ops = {
+ .hw_params = bfin_i2s_hw_params,
+ .set_fmt = bfin_i2s_set_dai_fmt,
+};
+
+static struct snd_soc_dai_driver bfin_i2s_dai = {
+ .suspend = bfin_i2s_suspend,
+ .resume = bfin_i2s_resume,
+ .playback = {
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = BFIN_I2S_RATES,
+ .formats = BFIN_I2S_FORMATS,
+ },
+ .capture = {
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = BFIN_I2S_RATES,
+ .formats = BFIN_I2S_FORMATS,
+ },
+ .ops = &bfin_i2s_dai_ops,
+};
+
+static int __devinit bfin_i2s_probe(struct platform_device *pdev)
+{
+ struct sport_device *sport;
+ struct device *dev = &pdev->dev;
+ int ret;
+
+ sport = sport_create(pdev);
+ if (!sport)
+ return -ENODEV;
+
+ /* register with the ASoC layers */
+ ret = snd_soc_register_dai(dev, &bfin_i2s_dai);
+ if (ret) {
+ dev_err(dev, "Failed to register DAI: %d\n", ret);
+ sport_delete(sport);
+ return ret;
+ }
+ platform_set_drvdata(pdev, sport);
+
+ return 0;
+}
+
+static int __devexit bfin_i2s_remove(struct platform_device *pdev)
+{
+ struct sport_device *sport = platform_get_drvdata(pdev);
+
+ snd_soc_unregister_dai(&pdev->dev);
+ sport_delete(sport);
+
+ return 0;
+}
+
+static struct platform_driver bfin_i2s_driver = {
+ .probe = bfin_i2s_probe,
+ .remove = __devexit_p(bfin_i2s_remove),
+ .driver = {
+ .name = "bfin-i2s",
+ .owner = THIS_MODULE,
+ },
+};
+
+module_platform_driver(bfin_i2s_driver);
+
+MODULE_DESCRIPTION("Analog Devices BF6XX i2s interface driver");
+MODULE_AUTHOR("Scott Jiang <Scott.Jiang.Linux@gmail.com>");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/blackfin/bf6xx-sport.c b/sound/soc/blackfin/bf6xx-sport.c
new file mode 100644
index 0000000..318c5ba5
--- /dev/null
+++ b/sound/soc/blackfin/bf6xx-sport.c
@@ -0,0 +1,422 @@
+/*
+ * bf6xx_sport.c Analog Devices BF6XX SPORT driver
+ *
+ * Copyright (c) 2012 Analog Devices Inc.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ */
+
+#include <linux/device.h>
+#include <linux/dma-mapping.h>
+#include <linux/interrupt.h>
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+
+#include <asm/blackfin.h>
+#include <asm/dma.h>
+#include <asm/portmux.h>
+
+#include "bf6xx-sport.h"
+
+int sport_set_tx_params(struct sport_device *sport,
+ struct sport_params *params)
+{
+ if (sport->tx_regs->spctl & SPORT_CTL_SPENPRI)
+ return -EBUSY;
+ sport->tx_regs->spctl = params->spctl | SPORT_CTL_SPTRAN;
+ sport->tx_regs->div = params->div;
+ SSYNC();
+ return 0;
+}
+EXPORT_SYMBOL(sport_set_tx_params);
+
+int sport_set_rx_params(struct sport_device *sport,
+ struct sport_params *params)
+{
+ if (sport->rx_regs->spctl & SPORT_CTL_SPENPRI)
+ return -EBUSY;
+ sport->rx_regs->spctl = params->spctl & ~SPORT_CTL_SPTRAN;
+ sport->rx_regs->div = params->div;
+ SSYNC();
+ return 0;
+}
+EXPORT_SYMBOL(sport_set_rx_params);
+
+static int compute_wdsize(size_t wdsize)
+{
+ switch (wdsize) {
+ case 1:
+ return WDSIZE_8 | PSIZE_8;
+ case 2:
+ return WDSIZE_16 | PSIZE_16;
+ default:
+ return WDSIZE_32 | PSIZE_32;
+ }
+}
+
+void sport_tx_start(struct sport_device *sport)
+{
+ set_dma_next_desc_addr(sport->tx_dma_chan, sport->tx_desc);
+ set_dma_config(sport->tx_dma_chan, DMAFLOW_LIST | DI_EN
+ | compute_wdsize(sport->wdsize) | NDSIZE_6);
+ enable_dma(sport->tx_dma_chan);
+ sport->tx_regs->spctl |= SPORT_CTL_SPENPRI;
+ SSYNC();
+}
+EXPORT_SYMBOL(sport_tx_start);
+
+void sport_rx_start(struct sport_device *sport)
+{
+ set_dma_next_desc_addr(sport->rx_dma_chan, sport->rx_desc);
+ set_dma_config(sport->rx_dma_chan, DMAFLOW_LIST | DI_EN | WNR
+ | compute_wdsize(sport->wdsize) | NDSIZE_6);
+ enable_dma(sport->rx_dma_chan);
+ sport->rx_regs->spctl |= SPORT_CTL_SPENPRI;
+ SSYNC();
+}
+EXPORT_SYMBOL(sport_rx_start);
+
+void sport_tx_stop(struct sport_device *sport)
+{
+ sport->tx_regs->spctl &= ~SPORT_CTL_SPENPRI;
+ SSYNC();
+ disable_dma(sport->tx_dma_chan);
+}
+EXPORT_SYMBOL(sport_tx_stop);
+
+void sport_rx_stop(struct sport_device *sport)
+{
+ sport->rx_regs->spctl &= ~SPORT_CTL_SPENPRI;
+ SSYNC();
+ disable_dma(sport->rx_dma_chan);
+}
+EXPORT_SYMBOL(sport_rx_stop);
+
+void sport_set_tx_callback(struct sport_device *sport,
+ void (*tx_callback)(void *), void *tx_data)
+{
+ sport->tx_callback = tx_callback;
+ sport->tx_data = tx_data;
+}
+EXPORT_SYMBOL(sport_set_tx_callback);
+
+void sport_set_rx_callback(struct sport_device *sport,
+ void (*rx_callback)(void *), void *rx_data)
+{
+ sport->rx_callback = rx_callback;
+ sport->rx_data = rx_data;
+}
+EXPORT_SYMBOL(sport_set_rx_callback);
+
+static void setup_desc(struct dmasg *desc, void *buf, int fragcount,
+ size_t fragsize, unsigned int cfg,
+ unsigned int count, size_t wdsize)
+{
+
+ int i;
+
+ for (i = 0; i < fragcount; ++i) {
+ desc[i].next_desc_addr = &(desc[i + 1]);
+ desc[i].start_addr = (unsigned long)buf + i*fragsize;
+ desc[i].cfg = cfg;
+ desc[i].x_count = count;
+ desc[i].x_modify = wdsize;
+ desc[i].y_count = 0;
+ desc[i].y_modify = 0;
+ }
+
+ /* make circular */
+ desc[fragcount-1].next_desc_addr = desc;
+}
+
+int sport_config_tx_dma(struct sport_device *sport, void *buf,
+ int fragcount, size_t fragsize)
+{
+ unsigned int count;
+ unsigned int cfg;
+ dma_addr_t addr;
+
+ count = fragsize/sport->wdsize;
+
+ if (sport->tx_desc)
+ dma_free_coherent(NULL, sport->tx_desc_size,
+ sport->tx_desc, 0);
+
+ sport->tx_desc = dma_alloc_coherent(NULL,
+ fragcount * sizeof(struct dmasg), &addr, 0);
+ sport->tx_desc_size = fragcount * sizeof(struct dmasg);
+ if (!sport->tx_desc)
+ return -ENOMEM;
+
+ sport->tx_buf = buf;
+ sport->tx_fragsize = fragsize;
+ sport->tx_frags = fragcount;
+ cfg = DMAFLOW_LIST | DI_EN | compute_wdsize(sport->wdsize) | NDSIZE_6;
+
+ setup_desc(sport->tx_desc, buf, fragcount, fragsize,
+ cfg|DMAEN, count, sport->wdsize);
+
+ return 0;
+}
+EXPORT_SYMBOL(sport_config_tx_dma);
+
+int sport_config_rx_dma(struct sport_device *sport, void *buf,
+ int fragcount, size_t fragsize)
+{
+ unsigned int count;
+ unsigned int cfg;
+ dma_addr_t addr;
+
+ count = fragsize/sport->wdsize;
+
+ if (sport->rx_desc)
+ dma_free_coherent(NULL, sport->rx_desc_size,
+ sport->rx_desc, 0);
+
+ sport->rx_desc = dma_alloc_coherent(NULL,
+ fragcount * sizeof(struct dmasg), &addr, 0);
+ sport->rx_desc_size = fragcount * sizeof(struct dmasg);
+ if (!sport->rx_desc)
+ return -ENOMEM;
+
+ sport->rx_buf = buf;
+ sport->rx_fragsize = fragsize;
+ sport->rx_frags = fragcount;
+ cfg = DMAFLOW_LIST | DI_EN | compute_wdsize(sport->wdsize)
+ | WNR | NDSIZE_6;
+
+ setup_desc(sport->rx_desc, buf, fragcount, fragsize,
+ cfg|DMAEN, count, sport->wdsize);
+
+ return 0;
+}
+EXPORT_SYMBOL(sport_config_rx_dma);
+
+unsigned long sport_curr_offset_tx(struct sport_device *sport)
+{
+ unsigned long curr = get_dma_curr_addr(sport->tx_dma_chan);
+
+ return (unsigned char *)curr - sport->tx_buf;
+}
+EXPORT_SYMBOL(sport_curr_offset_tx);
+
+unsigned long sport_curr_offset_rx(struct sport_device *sport)
+{
+ unsigned long curr = get_dma_curr_addr(sport->rx_dma_chan);
+
+ return (unsigned char *)curr - sport->rx_buf;
+}
+EXPORT_SYMBOL(sport_curr_offset_rx);
+
+static irqreturn_t sport_tx_irq(int irq, void *dev_id)
+{
+ struct sport_device *sport = dev_id;
+ static unsigned long status;
+
+ status = get_dma_curr_irqstat(sport->tx_dma_chan);
+ if (status & (DMA_DONE|DMA_ERR)) {
+ clear_dma_irqstat(sport->tx_dma_chan);
+ SSYNC();
+ }
+ if (sport->tx_callback)
+ sport->tx_callback(sport->tx_data);
+ return IRQ_HANDLED;
+}
+
+static irqreturn_t sport_rx_irq(int irq, void *dev_id)
+{
+ struct sport_device *sport = dev_id;
+ unsigned long status;
+
+ status = get_dma_curr_irqstat(sport->rx_dma_chan);
+ if (status & (DMA_DONE|DMA_ERR)) {
+ clear_dma_irqstat(sport->rx_dma_chan);
+ SSYNC();
+ }
+ if (sport->rx_callback)
+ sport->rx_callback(sport->rx_data);
+ return IRQ_HANDLED;
+}
+
+static irqreturn_t sport_err_irq(int irq, void *dev_id)
+{
+ struct sport_device *sport = dev_id;
+ struct device *dev = &sport->pdev->dev;
+
+ if (sport->tx_regs->spctl & SPORT_CTL_DERRPRI)
+ dev_err(dev, "sport error: TUVF\n");
+ if (sport->rx_regs->spctl & SPORT_CTL_DERRPRI)
+ dev_err(dev, "sport error: ROVF\n");
+
+ return IRQ_HANDLED;
+}
+
+static int sport_get_resource(struct sport_device *sport)
+{
+ struct platform_device *pdev = sport->pdev;
+ struct device *dev = &pdev->dev;
+ struct bfin_snd_platform_data *pdata = dev->platform_data;
+ struct resource *res;
+
+ if (!pdata) {
+ dev_err(dev, "No platform data\n");
+ return -ENODEV;
+ }
+ sport->pin_req = pdata->pin_req;
+
+ res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+ if (!res) {
+ dev_err(dev, "No tx MEM resource\n");
+ return -ENODEV;
+ }
+ sport->tx_regs = (struct sport_register *)res->start;
+
+ res = platform_get_resource(pdev, IORESOURCE_MEM, 1);
+ if (!res) {
+ dev_err(dev, "No rx MEM resource\n");
+ return -ENODEV;
+ }
+ sport->rx_regs = (struct sport_register *)res->start;
+
+ res = platform_get_resource(pdev, IORESOURCE_DMA, 0);
+ if (!res) {
+ dev_err(dev, "No tx DMA resource\n");
+ return -ENODEV;
+ }
+ sport->tx_dma_chan = res->start;
+
+ res = platform_get_resource(pdev, IORESOURCE_DMA, 1);
+ if (!res) {
+ dev_err(dev, "No rx DMA resource\n");
+ return -ENODEV;
+ }
+ sport->rx_dma_chan = res->start;
+
+ res = platform_get_resource(pdev, IORESOURCE_IRQ, 0);
+ if (!res) {
+ dev_err(dev, "No tx error irq resource\n");
+ return -ENODEV;
+ }
+ sport->tx_err_irq = res->start;
+
+ res = platform_get_resource(pdev, IORESOURCE_IRQ, 1);
+ if (!res) {
+ dev_err(dev, "No rx error irq resource\n");
+ return -ENODEV;
+ }
+ sport->rx_err_irq = res->start;
+
+ return 0;
+}
+
+static int sport_request_resource(struct sport_device *sport)
+{
+ struct device *dev = &sport->pdev->dev;
+ int ret;
+
+ ret = peripheral_request_list(sport->pin_req, "soc-audio");
+ if (ret) {
+ dev_err(dev, "Unable to request sport pin\n");
+ return ret;
+ }
+
+ ret = request_dma(sport->tx_dma_chan, "SPORT TX Data");
+ if (ret) {
+ dev_err(dev, "Unable to allocate DMA channel for sport tx\n");
+ goto err_tx_dma;
+ }
+ set_dma_callback(sport->tx_dma_chan, sport_tx_irq, sport);
+
+ ret = request_dma(sport->rx_dma_chan, "SPORT RX Data");
+ if (ret) {
+ dev_err(dev, "Unable to allocate DMA channel for sport rx\n");
+ goto err_rx_dma;
+ }
+ set_dma_callback(sport->rx_dma_chan, sport_rx_irq, sport);
+
+ ret = request_irq(sport->tx_err_irq, sport_err_irq,
+ 0, "SPORT TX ERROR", sport);
+ if (ret) {
+ dev_err(dev, "Unable to allocate tx error IRQ for sport\n");
+ goto err_tx_irq;
+ }
+
+ ret = request_irq(sport->rx_err_irq, sport_err_irq,
+ 0, "SPORT RX ERROR", sport);
+ if (ret) {
+ dev_err(dev, "Unable to allocate rx error IRQ for sport\n");
+ goto err_rx_irq;
+ }
+
+ return 0;
+err_rx_irq:
+ free_irq(sport->tx_err_irq, sport);
+err_tx_irq:
+ free_dma(sport->rx_dma_chan);
+err_rx_dma:
+ free_dma(sport->tx_dma_chan);
+err_tx_dma:
+ peripheral_free_list(sport->pin_req);
+ return ret;
+}
+
+static void sport_free_resource(struct sport_device *sport)
+{
+ free_irq(sport->rx_err_irq, sport);
+ free_irq(sport->tx_err_irq, sport);
+ free_dma(sport->rx_dma_chan);
+ free_dma(sport->tx_dma_chan);
+ peripheral_free_list(sport->pin_req);
+}
+
+struct sport_device *sport_create(struct platform_device *pdev)
+{
+ struct device *dev = &pdev->dev;
+ struct sport_device *sport;
+ int ret;
+
+ sport = kzalloc(sizeof(*sport), GFP_KERNEL);
+ if (!sport) {
+ dev_err(dev, "Unable to allocate memory for sport device\n");
+ return NULL;
+ }
+ sport->pdev = pdev;
+
+ ret = sport_get_resource(sport);
+ if (ret) {
+ kfree(sport);
+ return NULL;
+ }
+
+ ret = sport_request_resource(sport);
+ if (ret) {
+ kfree(sport);
+ return NULL;
+ }
+
+ dev_dbg(dev, "SPORT create success\n");
+ return sport;
+}
+EXPORT_SYMBOL(sport_create);
+
+void sport_delete(struct sport_device *sport)
+{
+ sport_free_resource(sport);
+}
+EXPORT_SYMBOL(sport_delete);
+
+MODULE_DESCRIPTION("Analog Devices BF6XX SPORT driver");
+MODULE_AUTHOR("Scott Jiang <Scott.Jiang.Linux@gmail.com>");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/blackfin/bf6xx-sport.h b/sound/soc/blackfin/bf6xx-sport.h
new file mode 100644
index 0000000..307d193
--- /dev/null
+++ b/sound/soc/blackfin/bf6xx-sport.h
@@ -0,0 +1,82 @@
+/*
+ * bf6xx_sport - Analog Devices BF6XX SPORT driver
+ *
+ * Copyright (c) 2012 Analog Devices Inc.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ */
+
+#ifndef _BF6XX_SPORT_H_
+#define _BF6XX_SPORT_H_
+
+#include <linux/platform_device.h>
+#include <asm/bfin_sport3.h>
+
+struct sport_device {
+ struct platform_device *pdev;
+ const unsigned short *pin_req;
+ struct sport_register *tx_regs;
+ struct sport_register *rx_regs;
+ int tx_dma_chan;
+ int rx_dma_chan;
+ int tx_err_irq;
+ int rx_err_irq;
+
+ void (*tx_callback)(void *data);
+ void *tx_data;
+ void (*rx_callback)(void *data);
+ void *rx_data;
+
+ struct dmasg *tx_desc;
+ struct dmasg *rx_desc;
+ unsigned int tx_desc_size;
+ unsigned int rx_desc_size;
+ unsigned char *tx_buf;
+ unsigned char *rx_buf;
+ unsigned int tx_fragsize;
+ unsigned int rx_fragsize;
+ unsigned int tx_frags;
+ unsigned int rx_frags;
+ unsigned int wdsize;
+};
+
+struct sport_params {
+ u32 spctl;
+ u32 div;
+};
+
+struct sport_device *sport_create(struct platform_device *pdev);
+void sport_delete(struct sport_device *sport);
+int sport_set_tx_params(struct sport_device *sport,
+ struct sport_params *params);
+int sport_set_rx_params(struct sport_device *sport,
+ struct sport_params *params);
+void sport_tx_start(struct sport_device *sport);
+void sport_rx_start(struct sport_device *sport);
+void sport_tx_stop(struct sport_device *sport);
+void sport_rx_stop(struct sport_device *sport);
+void sport_set_tx_callback(struct sport_device *sport,
+ void (*tx_callback)(void *), void *tx_data);
+void sport_set_rx_callback(struct sport_device *sport,
+ void (*rx_callback)(void *), void *rx_data);
+int sport_config_tx_dma(struct sport_device *sport, void *buf,
+ int fragcount, size_t fragsize);
+int sport_config_rx_dma(struct sport_device *sport, void *buf,
+ int fragcount, size_t fragsize);
+unsigned long sport_curr_offset_tx(struct sport_device *sport);
+unsigned long sport_curr_offset_rx(struct sport_device *sport);
+
+
+
+#endif
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 1e1613a..9f8e859 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -12,6 +12,7 @@ config SND_SOC_ALL_CODECS
tristate "Build all ASoC CODEC drivers"
select SND_SOC_88PM860X if MFD_88PM860X
select SND_SOC_L3
+ select SND_SOC_AB8500_CODEC if ABX500_CORE
select SND_SOC_AC97_CODEC if SND_SOC_AC97_BUS
select SND_SOC_AD1836 if SPI_MASTER
select SND_SOC_AD193X if SND_SOC_I2C_AND_SPI
@@ -35,7 +36,9 @@ config SND_SOC_ALL_CODECS
select SND_SOC_CS4271 if SND_SOC_I2C_AND_SPI
select SND_SOC_CX20442
select SND_SOC_DA7210 if I2C
+ select SND_SOC_DA732X if I2C
select SND_SOC_DFBMCS320
+ select SND_SOC_ISABELLE if I2C
select SND_SOC_JZ4740_CODEC
select SND_SOC_LM4857 if I2C
select SND_SOC_LM49453 if I2C
@@ -54,6 +57,7 @@ config SND_SOC_ALL_CODECS
select SND_SOC_SPDIF
select SND_SOC_SSM2602 if SND_SOC_I2C_AND_SPI
select SND_SOC_STA32X if I2C
+ select SND_SOC_STA529 if I2C
select SND_SOC_STAC9766 if SND_SOC_AC97_BUS
select SND_SOC_TLV320AIC23 if I2C
select SND_SOC_TLV320AIC26 if SPI_MASTER
@@ -70,6 +74,8 @@ config SND_SOC_ALL_CODECS
select SND_SOC_WM2000 if I2C
select SND_SOC_WM2200 if I2C
select SND_SOC_WM5100 if I2C
+ select SND_SOC_WM5102 if MFD_WM5102
+ select SND_SOC_WM5110 if MFD_WM5110
select SND_SOC_WM8350 if MFD_WM8350
select SND_SOC_WM8400 if MFD_WM8400
select SND_SOC_WM8510 if SND_SOC_I2C_AND_SPI
@@ -126,11 +132,21 @@ config SND_SOC_ALL_CODECS
config SND_SOC_88PM860X
tristate
+config SND_SOC_ARIZONA
+ tristate
+ default y if SND_SOC_WM5102=y
+ default y if SND_SOC_WM5110=y
+ default m if SND_SOC_WM5102=m
+ default m if SND_SOC_WM5110=m
+
config SND_SOC_WM_HUBS
tristate
default y if SND_SOC_WM8993=y || SND_SOC_WM8994=y
default m if SND_SOC_WM8993=m || SND_SOC_WM8994=m
+config SND_SOC_AB8500_CODEC
+ tristate
+
config SND_SOC_AC97_CODEC
tristate
select SND_AC97_CODEC
@@ -219,12 +235,18 @@ config SND_SOC_L3
config SND_SOC_DA7210
tristate
+config SND_SOC_DA732X
+ tristate
+
config SND_SOC_DFBMCS320
tristate
config SND_SOC_DMIC
tristate
+config SND_SOC_ISABELLE
+ tristate
+
config SND_SOC_LM49453
tristate
@@ -266,6 +288,9 @@ config SND_SOC_SSM2602
config SND_SOC_STA32X
tristate
+config SND_SOC_STA529
+ tristate
+
config SND_SOC_STAC9766
tristate
@@ -313,6 +338,12 @@ config SND_SOC_WM2200
config SND_SOC_WM5100
tristate
+config SND_SOC_WM5102
+ tristate
+
+config SND_SOC_WM5110
+ tristate
+
config SND_SOC_WM8350
tristate
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index fc27fec..34148bb 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -1,4 +1,5 @@
snd-soc-88pm860x-objs := 88pm860x-codec.o
+snd-soc-ab8500-codec-objs := ab8500-codec.o
snd-soc-ac97-objs := ac97.o
snd-soc-ad1836-objs := ad1836.o
snd-soc-ad193x-objs := ad193x.o
@@ -13,6 +14,7 @@ snd-soc-ak4535-objs := ak4535.o
snd-soc-ak4641-objs := ak4641.o
snd-soc-ak4642-objs := ak4642.o
snd-soc-ak4671-objs := ak4671.o
+snd-soc-arizona-objs := arizona.o
snd-soc-cq93vc-objs := cq93vc.o
snd-soc-cs42l51-objs := cs42l51.o
snd-soc-cs42l52-objs := cs42l52.o
@@ -21,8 +23,10 @@ snd-soc-cs4270-objs := cs4270.o
snd-soc-cs4271-objs := cs4271.o
snd-soc-cx20442-objs := cx20442.o
snd-soc-da7210-objs := da7210.o
+snd-soc-da732x-objs := da732x.o
snd-soc-dfbmcs320-objs := dfbmcs320.o
snd-soc-dmic-objs := dmic.o
+snd-soc-isabelle-objs := isabelle.o
snd-soc-jz4740-codec-objs := jz4740.o
snd-soc-l3-objs := l3.o
snd-soc-lm4857-objs := lm4857.o
@@ -41,9 +45,11 @@ snd-soc-alc5623-objs := alc5623.o
snd-soc-alc5632-objs := alc5632.o
snd-soc-sigmadsp-objs := sigmadsp.o
snd-soc-sn95031-objs := sn95031.o
-snd-soc-spdif-objs := spdif_transciever.o
+snd-soc-spdif-tx-objs := spdif_transciever.o
+snd-soc-spdif-rx-objs := spdif_receiver.o
snd-soc-ssm2602-objs := ssm2602.o
snd-soc-sta32x-objs := sta32x.o
+snd-soc-sta529-objs := sta529.o
snd-soc-stac9766-objs := stac9766.o
snd-soc-tlv320aic23-objs := tlv320aic23.o
snd-soc-tlv320aic26-objs := tlv320aic26.o
@@ -59,6 +65,8 @@ snd-soc-wm1250-ev1-objs := wm1250-ev1.o
snd-soc-wm2000-objs := wm2000.o
snd-soc-wm2200-objs := wm2200.o
snd-soc-wm5100-objs := wm5100.o wm5100-tables.o
+snd-soc-wm5102-objs := wm5102.o
+snd-soc-wm5110-objs := wm5110.o
snd-soc-wm8350-objs := wm8350.o
snd-soc-wm8400-objs := wm8400.o
snd-soc-wm8510-objs := wm8510.o
@@ -108,6 +116,7 @@ snd-soc-max9877-objs := max9877.o
snd-soc-tpa6130a2-objs := tpa6130a2.o
obj-$(CONFIG_SND_SOC_88PM860X) += snd-soc-88pm860x.o
+obj-$(CONFIG_SND_SOC_AB8500_CODEC) += snd-soc-ab8500-codec.o
obj-$(CONFIG_SND_SOC_AC97_CODEC) += snd-soc-ac97.o
obj-$(CONFIG_SND_SOC_AD1836) += snd-soc-ad1836.o
obj-$(CONFIG_SND_SOC_AD193X) += snd-soc-ad193x.o
@@ -124,6 +133,7 @@ obj-$(CONFIG_SND_SOC_AK4642) += snd-soc-ak4642.o
obj-$(CONFIG_SND_SOC_AK4671) += snd-soc-ak4671.o
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-$(CONFIG_SND_SOC_CS42L51) += snd-soc-cs42l51.o
obj-$(CONFIG_SND_SOC_CS42L52) += snd-soc-cs42l52.o
@@ -132,8 +142,10 @@ obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o
obj-$(CONFIG_SND_SOC_CS4271) += snd-soc-cs4271.o
obj-$(CONFIG_SND_SOC_CX20442) += snd-soc-cx20442.o
obj-$(CONFIG_SND_SOC_DA7210) += snd-soc-da7210.o
+obj-$(CONFIG_SND_SOC_DA732X) += snd-soc-da732x.o
obj-$(CONFIG_SND_SOC_DFBMCS320) += snd-soc-dfbmcs320.o
obj-$(CONFIG_SND_SOC_DMIC) += snd-soc-dmic.o
+obj-$(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
obj-$(CONFIG_SND_SOC_LM4857) += snd-soc-lm4857.o
@@ -150,9 +162,10 @@ obj-$(CONFIG_SND_SOC_RT5631) += snd-soc-rt5631.o
obj-$(CONFIG_SND_SOC_SGTL5000) += snd-soc-sgtl5000.o
obj-$(CONFIG_SND_SOC_SIGMADSP) += snd-soc-sigmadsp.o
obj-$(CONFIG_SND_SOC_SN95031) +=snd-soc-sn95031.o
-obj-$(CONFIG_SND_SOC_SPDIF) += snd-soc-spdif.o
+obj-$(CONFIG_SND_SOC_SPDIF) += snd-soc-spdif-rx.o snd-soc-spdif-tx.o
obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o
obj-$(CONFIG_SND_SOC_STA32X) += snd-soc-sta32x.o
+obj-$(CONFIG_SND_SOC_STA529) += snd-soc-sta529.o
obj-$(CONFIG_SND_SOC_STAC9766) += snd-soc-stac9766.o
obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o
obj-$(CONFIG_SND_SOC_TLV320AIC26) += snd-soc-tlv320aic26.o
@@ -168,6 +181,8 @@ obj-$(CONFIG_SND_SOC_WM1250_EV1) += snd-soc-wm1250-ev1.o
obj-$(CONFIG_SND_SOC_WM2000) += snd-soc-wm2000.o
obj-$(CONFIG_SND_SOC_WM2200) += snd-soc-wm2200.o
obj-$(CONFIG_SND_SOC_WM5100) += snd-soc-wm5100.o
+obj-$(CONFIG_SND_SOC_WM5102) += snd-soc-wm5102.o
+obj-$(CONFIG_SND_SOC_WM5110) += snd-soc-wm5110.o
obj-$(CONFIG_SND_SOC_WM8350) += snd-soc-wm8350.o
obj-$(CONFIG_SND_SOC_WM8400) += snd-soc-wm8400.o
obj-$(CONFIG_SND_SOC_WM8510) += snd-soc-wm8510.o
diff --git a/sound/soc/codecs/ab8500-codec.c b/sound/soc/codecs/ab8500-codec.c
new file mode 100644
index 0000000..3c79592
--- /dev/null
+++ b/sound/soc/codecs/ab8500-codec.c
@@ -0,0 +1,2522 @@
+/*
+ * Copyright (C) ST-Ericsson SA 2012
+ *
+ * Author: Ola Lilja <ola.o.lilja@stericsson.com>,
+ * Kristoffer Karlsson <kristoffer.karlsson@stericsson.com>,
+ * Roger Nilsson <roger.xr.nilsson@stericsson.com>,
+ * for ST-Ericsson.
+ *
+ * Based on the early work done by:
+ * Mikko J. Lehto <mikko.lehto@symbio.com>,
+ * Mikko Sarmanne <mikko.sarmanne@symbio.com>,
+ * Jarmo K. Kuronen <jarmo.kuronen@symbio.com>,
+ * for ST-Ericsson.
+ *
+ * License terms:
+ *
+ * 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 <linux/kernel.h>
+#include <linux/module.h>
+#include <linux/device.h>
+#include <linux/slab.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/platform_device.h>
+#include <linux/mutex.h>
+#include <linux/mfd/abx500/ab8500.h>
+#include <linux/mfd/abx500.h>
+#include <linux/mfd/abx500/ab8500-sysctrl.h>
+#include <linux/mfd/abx500/ab8500-codec.h>
+#include <linux/regulator/consumer.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/tlv.h>
+
+#include "ab8500-codec.h"
+
+/* Macrocell value definitions */
+#define CLK_32K_OUT2_DISABLE 0x01
+#define INACTIVE_RESET_AUDIO 0x02
+#define ENABLE_AUDIO_CLK_TO_AUDIO_BLK 0x10
+#define ENABLE_VINTCORE12_SUPPLY 0x04
+#define GPIO27_DIR_OUTPUT 0x04
+#define GPIO29_DIR_OUTPUT 0x10
+#define GPIO31_DIR_OUTPUT 0x40
+
+/* Macrocell register definitions */
+#define AB8500_CTRL3_REG 0x0200
+#define AB8500_GPIO_DIR4_REG 0x1013
+
+/* Nr of FIR/IIR-coeff banks in ANC-block */
+#define AB8500_NR_OF_ANC_COEFF_BANKS 2
+
+/* Minimum duration to keep ANC IIR Init bit high or
+low before proceeding with the configuration sequence */
+#define AB8500_ANC_SM_DELAY 2000
+
+#define AB8500_FILTER_CONTROL(xname, xcount, xmin, xmax) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \
+ .info = filter_control_info, \
+ .get = filter_control_get, .put = filter_control_put, \
+ .private_value = (unsigned long)&(struct filter_control) \
+ {.count = xcount, .min = xmin, .max = xmax} }
+
+struct filter_control {
+ long min, max;
+ unsigned int count;
+ long value[128];
+};
+
+/* Sidetone states */
+static const char * const enum_sid_state[] = {
+ "Unconfigured",
+ "Apply FIR",
+ "FIR is configured",
+};
+enum sid_state {
+ SID_UNCONFIGURED = 0,
+ SID_APPLY_FIR = 1,
+ SID_FIR_CONFIGURED = 2,
+};
+
+static const char * const enum_anc_state[] = {
+ "Unconfigured",
+ "Apply FIR and IIR",
+ "FIR and IIR are configured",
+ "Apply FIR",
+ "FIR is configured",
+ "Apply IIR",
+ "IIR is configured"
+};
+enum anc_state {
+ ANC_UNCONFIGURED = 0,
+ ANC_APPLY_FIR_IIR = 1,
+ ANC_FIR_IIR_CONFIGURED = 2,
+ ANC_APPLY_FIR = 3,
+ ANC_FIR_CONFIGURED = 4,
+ ANC_APPLY_IIR = 5,
+ ANC_IIR_CONFIGURED = 6
+};
+
+/* Analog microphones */
+enum amic_idx {
+ AMIC_IDX_1A,
+ AMIC_IDX_1B,
+ AMIC_IDX_2
+};
+
+struct ab8500_codec_drvdata_dbg {
+ struct regulator *vaud;
+ struct regulator *vamic1;
+ struct regulator *vamic2;
+ struct regulator *vdmic;
+};
+
+/* Private data for AB8500 device-driver */
+struct ab8500_codec_drvdata {
+ /* Sidetone */
+ long *sid_fir_values;
+ enum sid_state sid_status;
+
+ /* ANC */
+ struct mutex anc_lock;
+ long *anc_fir_values;
+ long *anc_iir_values;
+ enum anc_state anc_status;
+};
+
+static inline const char *amic_micbias_str(enum amic_micbias micbias)
+{
+ switch (micbias) {
+ case AMIC_MICBIAS_VAMIC1:
+ return "VAMIC1";
+ case AMIC_MICBIAS_VAMIC2:
+ return "VAMIC2";
+ default:
+ return "Unknown";
+ }
+}
+
+static inline const char *amic_type_str(enum amic_type type)
+{
+ switch (type) {
+ case AMIC_TYPE_DIFFERENTIAL:
+ return "DIFFERENTIAL";
+ case AMIC_TYPE_SINGLE_ENDED:
+ return "SINGLE ENDED";
+ default:
+ return "Unknown";
+ }
+}
+
+/*
+ * Read'n'write functions
+ */
+
+/* Read a register from the audio-bank of AB8500 */
+static unsigned int ab8500_codec_read_reg(struct snd_soc_codec *codec,
+ unsigned int reg)
+{
+ 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;
+ }
+
+ return value;
+}
+
+/* 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)
+{
+ 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);
+
+ return status;
+}
+
+/*
+ * Controls - DAPM
+ */
+
+/* Earpiece */
+
+/* Earpiece source selector */
+static const char * const enum_ear_lineout_source[] = {"Headset Left",
+ "Speaker Left"};
+static SOC_ENUM_SINGLE_DECL(dapm_enum_ear_lineout_source, AB8500_DMICFILTCONF,
+ AB8500_DMICFILTCONF_DA3TOEAR, enum_ear_lineout_source);
+static const struct snd_kcontrol_new dapm_ear_lineout_source =
+ SOC_DAPM_ENUM("Earpiece or LineOut Mono Source",
+ dapm_enum_ear_lineout_source);
+
+/* LineOut */
+
+/* LineOut source selector */
+static const char * const enum_lineout_source[] = {"Mono Path", "Stereo Path"};
+static SOC_ENUM_DOUBLE_DECL(dapm_enum_lineout_source, AB8500_ANACONF5,
+ AB8500_ANACONF5_HSLDACTOLOL,
+ AB8500_ANACONF5_HSRDACTOLOR, enum_lineout_source);
+static const struct snd_kcontrol_new dapm_lineout_source[] = {
+ SOC_DAPM_ENUM("LineOut Source", dapm_enum_lineout_source),
+};
+
+/* Handsfree */
+
+/* Speaker Left - ANC selector */
+static const char * const enum_HFx_sel[] = {"Audio Path", "ANC"};
+static SOC_ENUM_SINGLE_DECL(dapm_enum_HFl_sel, AB8500_DIGMULTCONF2,
+ AB8500_DIGMULTCONF2_HFLSEL, enum_HFx_sel);
+static const struct snd_kcontrol_new dapm_HFl_select[] = {
+ SOC_DAPM_ENUM("Speaker Left Source", dapm_enum_HFl_sel),
+};
+
+/* Speaker Right - ANC selector */
+static SOC_ENUM_SINGLE_DECL(dapm_enum_HFr_sel, AB8500_DIGMULTCONF2,
+ AB8500_DIGMULTCONF2_HFRSEL, enum_HFx_sel);
+static const struct snd_kcontrol_new dapm_HFr_select[] = {
+ SOC_DAPM_ENUM("Speaker Right Source", dapm_enum_HFr_sel),
+};
+
+/* Mic 1 */
+
+/* Mic 1 - Mic 1a or 1b selector */
+static const char * const enum_mic1ab_sel[] = {"Mic 1b", "Mic 1a"};
+static SOC_ENUM_SINGLE_DECL(dapm_enum_mic1ab_sel, AB8500_ANACONF3,
+ AB8500_ANACONF3_MIC1SEL, enum_mic1ab_sel);
+static const struct snd_kcontrol_new dapm_mic1ab_mux[] = {
+ SOC_DAPM_ENUM("Mic 1a or 1b Select", dapm_enum_mic1ab_sel),
+};
+
+/* Mic 1 - AD3 - Mic 1 or DMic 3 selector */
+static const char * const enum_ad3_sel[] = {"Mic 1", "DMic 3"};
+static SOC_ENUM_SINGLE_DECL(dapm_enum_ad3_sel, AB8500_DIGMULTCONF1,
+ AB8500_DIGMULTCONF1_AD3SEL, enum_ad3_sel);
+static const struct snd_kcontrol_new dapm_ad3_select[] = {
+ SOC_DAPM_ENUM("AD3 Source Select", dapm_enum_ad3_sel),
+};
+
+/* Mic 1 - AD6 - Mic 1 or DMic 6 selector */
+static const char * const enum_ad6_sel[] = {"Mic 1", "DMic 6"};
+static SOC_ENUM_SINGLE_DECL(dapm_enum_ad6_sel, AB8500_DIGMULTCONF1,
+ AB8500_DIGMULTCONF1_AD6SEL, enum_ad6_sel);
+static const struct snd_kcontrol_new dapm_ad6_select[] = {
+ SOC_DAPM_ENUM("AD6 Source Select", dapm_enum_ad6_sel),
+};
+
+/* Mic 2 */
+
+/* Mic 2 - AD5 - Mic 2 or DMic 5 selector */
+static const char * const enum_ad5_sel[] = {"Mic 2", "DMic 5"};
+static SOC_ENUM_SINGLE_DECL(dapm_enum_ad5_sel, AB8500_DIGMULTCONF1,
+ AB8500_DIGMULTCONF1_AD5SEL, enum_ad5_sel);
+static const struct snd_kcontrol_new dapm_ad5_select[] = {
+ SOC_DAPM_ENUM("AD5 Source Select", dapm_enum_ad5_sel),
+};
+
+/* LineIn */
+
+/* LineIn left - AD1 - LineIn Left or DMic 1 selector */
+static const char * const enum_ad1_sel[] = {"LineIn Left", "DMic 1"};
+static SOC_ENUM_SINGLE_DECL(dapm_enum_ad1_sel, AB8500_DIGMULTCONF1,
+ AB8500_DIGMULTCONF1_AD1SEL, enum_ad1_sel);
+static const struct snd_kcontrol_new dapm_ad1_select[] = {
+ SOC_DAPM_ENUM("AD1 Source Select", dapm_enum_ad1_sel),
+};
+
+/* LineIn right - Mic 2 or LineIn Right selector */
+static const char * const enum_mic2lr_sel[] = {"Mic 2", "LineIn Right"};
+static SOC_ENUM_SINGLE_DECL(dapm_enum_mic2lr_sel, AB8500_ANACONF3,
+ AB8500_ANACONF3_LINRSEL, enum_mic2lr_sel);
+static const struct snd_kcontrol_new dapm_mic2lr_select[] = {
+ SOC_DAPM_ENUM("Mic 2 or LINR Select", dapm_enum_mic2lr_sel),
+};
+
+/* LineIn right - AD2 - LineIn Right or DMic2 selector */
+static const char * const enum_ad2_sel[] = {"LineIn Right", "DMic 2"};
+static SOC_ENUM_SINGLE_DECL(dapm_enum_ad2_sel, AB8500_DIGMULTCONF1,
+ AB8500_DIGMULTCONF1_AD2SEL, enum_ad2_sel);
+static const struct snd_kcontrol_new dapm_ad2_select[] = {
+ SOC_DAPM_ENUM("AD2 Source Select", dapm_enum_ad2_sel),
+};
+
+
+/* ANC */
+
+static const char * const enum_anc_in_sel[] = {"Mic 1 / DMic 6",
+ "Mic 2 / DMic 5"};
+static SOC_ENUM_SINGLE_DECL(dapm_enum_anc_in_sel, AB8500_DMICFILTCONF,
+ AB8500_DMICFILTCONF_ANCINSEL, enum_anc_in_sel);
+static const struct snd_kcontrol_new dapm_anc_in_select[] = {
+ SOC_DAPM_ENUM("ANC Source", dapm_enum_anc_in_sel),
+};
+
+/* ANC - Enable/Disable */
+static const struct snd_kcontrol_new dapm_anc_enable[] = {
+ SOC_DAPM_SINGLE("Switch", AB8500_ANCCONF1,
+ AB8500_ANCCONF1_ENANC, 0, 0),
+};
+
+/* ANC to Earpiece - Mute */
+static const struct snd_kcontrol_new dapm_anc_ear_mute[] = {
+ SOC_DAPM_SINGLE("Switch", AB8500_DIGMULTCONF1,
+ AB8500_DIGMULTCONF1_ANCSEL, 1, 0),
+};
+
+
+
+/* Sidetone left */
+
+/* Sidetone left - Input selector */
+static const char * const enum_stfir1_in_sel[] = {
+ "LineIn Left", "LineIn Right", "Mic 1", "Headset Left"
+};
+static SOC_ENUM_SINGLE_DECL(dapm_enum_stfir1_in_sel, AB8500_DIGMULTCONF2,
+ AB8500_DIGMULTCONF2_FIRSID1SEL, enum_stfir1_in_sel);
+static const struct snd_kcontrol_new dapm_stfir1_in_select[] = {
+ SOC_DAPM_ENUM("Sidetone Left Source", dapm_enum_stfir1_in_sel),
+};
+
+/* Sidetone right path */
+
+/* Sidetone right - Input selector */
+static const char * const enum_stfir2_in_sel[] = {
+ "LineIn Right", "Mic 1", "DMic 4", "Headset Right"
+};
+static SOC_ENUM_SINGLE_DECL(dapm_enum_stfir2_in_sel, AB8500_DIGMULTCONF2,
+ AB8500_DIGMULTCONF2_FIRSID2SEL, enum_stfir2_in_sel);
+static const struct snd_kcontrol_new dapm_stfir2_in_select[] = {
+ SOC_DAPM_ENUM("Sidetone Right Source", dapm_enum_stfir2_in_sel),
+};
+
+/* Vibra */
+
+static const char * const enum_pwm2vibx[] = {"Audio Path", "PWM Generator"};
+
+static SOC_ENUM_SINGLE_DECL(dapm_enum_pwm2vib1, AB8500_PWMGENCONF1,
+ AB8500_PWMGENCONF1_PWMTOVIB1, enum_pwm2vibx);
+
+static const struct snd_kcontrol_new dapm_pwm2vib1[] = {
+ SOC_DAPM_ENUM("Vibra 1 Controller", dapm_enum_pwm2vib1),
+};
+
+static SOC_ENUM_SINGLE_DECL(dapm_enum_pwm2vib2, AB8500_PWMGENCONF1,
+ AB8500_PWMGENCONF1_PWMTOVIB2, enum_pwm2vibx);
+
+static const struct snd_kcontrol_new dapm_pwm2vib2[] = {
+ SOC_DAPM_ENUM("Vibra 2 Controller", dapm_enum_pwm2vib2),
+};
+
+/*
+ * DAPM-widgets
+ */
+
+static const struct snd_soc_dapm_widget ab8500_dapm_widgets[] = {
+
+ /* Clocks */
+ SND_SOC_DAPM_CLOCK_SUPPLY("audioclk"),
+
+ /* Regulators */
+ SND_SOC_DAPM_REGULATOR_SUPPLY("V-AUD", 0),
+ SND_SOC_DAPM_REGULATOR_SUPPLY("V-AMIC1", 0),
+ SND_SOC_DAPM_REGULATOR_SUPPLY("V-AMIC2", 0),
+ SND_SOC_DAPM_REGULATOR_SUPPLY("V-DMIC", 0),
+
+ /* Power */
+ SND_SOC_DAPM_SUPPLY("Audio Power",
+ AB8500_POWERUP, AB8500_POWERUP_POWERUP, 0,
+ NULL, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_SUPPLY("Audio Analog Power",
+ AB8500_POWERUP, AB8500_POWERUP_ENANA, 0,
+ NULL, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+
+ /* Main supply node */
+ SND_SOC_DAPM_SUPPLY("Main Supply", SND_SOC_NOPM, 0, 0,
+ NULL, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+
+ /* DA/AD */
+
+ SND_SOC_DAPM_INPUT("ADC Input"),
+ SND_SOC_DAPM_ADC("ADC", "ab8500_0c", SND_SOC_NOPM, 0, 0),
+
+ SND_SOC_DAPM_DAC("DAC", NULL, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_OUTPUT("DAC Output"),
+
+ SND_SOC_DAPM_AIF_IN("DA_IN1", NULL, 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_IN("DA_IN2", NULL, 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_IN("DA_IN3", NULL, 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_IN("DA_IN4", NULL, 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_IN("DA_IN5", NULL, 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_IN("DA_IN6", NULL, 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("AD_OUT1", NULL, 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("AD_OUT2", NULL, 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("AD_OUT3", NULL, 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("AD_OUT4", NULL, 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("AD_OUT57", NULL, 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("AD_OUT68", NULL, 0, SND_SOC_NOPM, 0, 0),
+
+ /* Headset path */
+
+ SND_SOC_DAPM_SUPPLY("Charge Pump", AB8500_ANACONF5,
+ AB8500_ANACONF5_ENCPHS, 0, NULL, 0),
+
+ SND_SOC_DAPM_DAC("DA1 Enable", "ab8500_0p",
+ AB8500_DAPATHENA, AB8500_DAPATHENA_ENDA1, 0),
+ SND_SOC_DAPM_DAC("DA2 Enable", "ab8500_0p",
+ AB8500_DAPATHENA, AB8500_DAPATHENA_ENDA2, 0),
+
+ SND_SOC_DAPM_PGA("HSL Digital Volume", SND_SOC_NOPM, 0, 0,
+ NULL, 0),
+ SND_SOC_DAPM_PGA("HSR Digital Volume", SND_SOC_NOPM, 0, 0,
+ NULL, 0),
+
+ SND_SOC_DAPM_DAC("HSL DAC", "ab8500_0p",
+ AB8500_DAPATHCONF, AB8500_DAPATHCONF_ENDACHSL, 0),
+ SND_SOC_DAPM_DAC("HSR DAC", "ab8500_0p",
+ AB8500_DAPATHCONF, AB8500_DAPATHCONF_ENDACHSR, 0),
+ SND_SOC_DAPM_MIXER("HSL DAC Mute", AB8500_MUTECONF,
+ AB8500_MUTECONF_MUTDACHSL, 1,
+ NULL, 0),
+ SND_SOC_DAPM_MIXER("HSR DAC Mute", AB8500_MUTECONF,
+ AB8500_MUTECONF_MUTDACHSR, 1,
+ NULL, 0),
+ SND_SOC_DAPM_DAC("HSL DAC Driver", "ab8500_0p",
+ AB8500_ANACONF3, AB8500_ANACONF3_ENDRVHSL, 0),
+ SND_SOC_DAPM_DAC("HSR DAC Driver", "ab8500_0p",
+ AB8500_ANACONF3, AB8500_ANACONF3_ENDRVHSR, 0),
+
+ SND_SOC_DAPM_MIXER("HSL Mute",
+ AB8500_MUTECONF, AB8500_MUTECONF_MUTHSL, 1,
+ NULL, 0),
+ SND_SOC_DAPM_MIXER("HSR Mute",
+ AB8500_MUTECONF, AB8500_MUTECONF_MUTHSR, 1,
+ NULL, 0),
+ SND_SOC_DAPM_MIXER("HSL Enable",
+ AB8500_ANACONF4, AB8500_ANACONF4_ENHSL, 0,
+ NULL, 0),
+ SND_SOC_DAPM_MIXER("HSR Enable",
+ AB8500_ANACONF4, AB8500_ANACONF4_ENHSR, 0,
+ NULL, 0),
+ SND_SOC_DAPM_PGA("HSL Volume",
+ SND_SOC_NOPM, 0, 0,
+ NULL, 0),
+ SND_SOC_DAPM_PGA("HSR Volume",
+ SND_SOC_NOPM, 0, 0,
+ NULL, 0),
+
+ SND_SOC_DAPM_OUTPUT("Headset Left"),
+ SND_SOC_DAPM_OUTPUT("Headset Right"),
+
+ /* LineOut path */
+
+ SND_SOC_DAPM_MUX("LineOut Source",
+ SND_SOC_NOPM, 0, 0, dapm_lineout_source),
+
+ SND_SOC_DAPM_MIXER("LOL Disable HFL",
+ AB8500_ANACONF4, AB8500_ANACONF4_ENHFL, 1,
+ NULL, 0),
+ SND_SOC_DAPM_MIXER("LOR Disable HFR",
+ AB8500_ANACONF4, AB8500_ANACONF4_ENHFR, 1,
+ NULL, 0),
+
+ SND_SOC_DAPM_MIXER("LOL Enable",
+ AB8500_ANACONF5, AB8500_ANACONF5_ENLOL, 0,
+ NULL, 0),
+ SND_SOC_DAPM_MIXER("LOR Enable",
+ AB8500_ANACONF5, AB8500_ANACONF5_ENLOR, 0,
+ NULL, 0),
+
+ SND_SOC_DAPM_OUTPUT("LineOut Left"),
+ SND_SOC_DAPM_OUTPUT("LineOut Right"),
+
+ /* Earpiece path */
+
+ SND_SOC_DAPM_MUX("Earpiece or LineOut Mono Source",
+ SND_SOC_NOPM, 0, 0, &dapm_ear_lineout_source),
+ SND_SOC_DAPM_MIXER("EAR DAC",
+ AB8500_DAPATHCONF, AB8500_DAPATHCONF_ENDACEAR, 0,
+ NULL, 0),
+ SND_SOC_DAPM_MIXER("EAR Mute",
+ AB8500_MUTECONF, AB8500_MUTECONF_MUTEAR, 1,
+ NULL, 0),
+ SND_SOC_DAPM_MIXER("EAR Enable",
+ AB8500_ANACONF4, AB8500_ANACONF4_ENEAR, 0,
+ NULL, 0),
+
+ SND_SOC_DAPM_OUTPUT("Earpiece"),
+
+ /* Handsfree path */
+
+ SND_SOC_DAPM_MIXER("DA3 Channel Volume",
+ AB8500_DAPATHENA, AB8500_DAPATHENA_ENDA3, 0,
+ NULL, 0),
+ SND_SOC_DAPM_MIXER("DA4 Channel Volume",
+ AB8500_DAPATHENA, AB8500_DAPATHENA_ENDA4, 0,
+ NULL, 0),
+ SND_SOC_DAPM_MUX("Speaker Left Source",
+ SND_SOC_NOPM, 0, 0, dapm_HFl_select),
+ SND_SOC_DAPM_MUX("Speaker Right Source",
+ SND_SOC_NOPM, 0, 0, dapm_HFr_select),
+ SND_SOC_DAPM_MIXER("HFL DAC", AB8500_DAPATHCONF,
+ AB8500_DAPATHCONF_ENDACHFL, 0,
+ NULL, 0),
+ SND_SOC_DAPM_MIXER("HFR DAC",
+ AB8500_DAPATHCONF, AB8500_DAPATHCONF_ENDACHFR, 0,
+ NULL, 0),
+ SND_SOC_DAPM_MIXER("DA4 or ANC path to HfR",
+ AB8500_DIGMULTCONF2, AB8500_DIGMULTCONF2_DATOHFREN, 0,
+ NULL, 0),
+ SND_SOC_DAPM_MIXER("DA3 or ANC path to HfL",
+ AB8500_DIGMULTCONF2, AB8500_DIGMULTCONF2_DATOHFLEN, 0,
+ NULL, 0),
+ SND_SOC_DAPM_MIXER("HFL Enable",
+ AB8500_ANACONF4, AB8500_ANACONF4_ENHFL, 0,
+ NULL, 0),
+ SND_SOC_DAPM_MIXER("HFR Enable",
+ AB8500_ANACONF4, AB8500_ANACONF4_ENHFR, 0,
+ NULL, 0),
+
+ SND_SOC_DAPM_OUTPUT("Speaker Left"),
+ SND_SOC_DAPM_OUTPUT("Speaker Right"),
+
+ /* Vibrator path */
+
+ SND_SOC_DAPM_INPUT("PWMGEN1"),
+ SND_SOC_DAPM_INPUT("PWMGEN2"),
+
+ SND_SOC_DAPM_MIXER("DA5 Channel Volume",
+ AB8500_DAPATHENA, AB8500_DAPATHENA_ENDA5, 0,
+ NULL, 0),
+ SND_SOC_DAPM_MIXER("DA6 Channel Volume",
+ AB8500_DAPATHENA, AB8500_DAPATHENA_ENDA6, 0,
+ NULL, 0),
+ SND_SOC_DAPM_MIXER("VIB1 DAC",
+ AB8500_DAPATHCONF, AB8500_DAPATHCONF_ENDACVIB1, 0,
+ NULL, 0),
+ SND_SOC_DAPM_MIXER("VIB2 DAC",
+ AB8500_DAPATHCONF, AB8500_DAPATHCONF_ENDACVIB2, 0,
+ NULL, 0),
+ SND_SOC_DAPM_MUX("Vibra 1 Controller",
+ SND_SOC_NOPM, 0, 0, dapm_pwm2vib1),
+ SND_SOC_DAPM_MUX("Vibra 2 Controller",
+ SND_SOC_NOPM, 0, 0, dapm_pwm2vib2),
+ SND_SOC_DAPM_MIXER("VIB1 Enable",
+ AB8500_ANACONF4, AB8500_ANACONF4_ENVIB1, 0,
+ NULL, 0),
+ SND_SOC_DAPM_MIXER("VIB2 Enable",
+ AB8500_ANACONF4, AB8500_ANACONF4_ENVIB2, 0,
+ NULL, 0),
+
+ SND_SOC_DAPM_OUTPUT("Vibra 1"),
+ SND_SOC_DAPM_OUTPUT("Vibra 2"),
+
+ /* Mic 1 */
+
+ SND_SOC_DAPM_INPUT("Mic 1"),
+
+ SND_SOC_DAPM_MUX("Mic 1a or 1b Select",
+ SND_SOC_NOPM, 0, 0, dapm_mic1ab_mux),
+ SND_SOC_DAPM_MIXER("MIC1 Mute",
+ AB8500_ANACONF2, AB8500_ANACONF2_MUTMIC1, 1,
+ NULL, 0),
+ SND_SOC_DAPM_MIXER("MIC1A V-AMICx Enable",
+ AB8500_ANACONF2, AB8500_ANACONF2_ENMIC1, 0,
+ NULL, 0),
+ SND_SOC_DAPM_MIXER("MIC1B V-AMICx Enable",
+ AB8500_ANACONF2, AB8500_ANACONF2_ENMIC1, 0,
+ NULL, 0),
+ SND_SOC_DAPM_MIXER("MIC1 ADC",
+ AB8500_ANACONF3, AB8500_ANACONF3_ENADCMIC, 0,
+ NULL, 0),
+ SND_SOC_DAPM_MUX("AD3 Source Select",
+ SND_SOC_NOPM, 0, 0, dapm_ad3_select),
+ SND_SOC_DAPM_MIXER("AD3 Channel Volume",
+ SND_SOC_NOPM, 0, 0,
+ NULL, 0),
+ SND_SOC_DAPM_MIXER("AD3 Enable",
+ AB8500_ADPATHENA, AB8500_ADPATHENA_ENAD34, 0,
+ NULL, 0),
+
+ /* Mic 2 */
+
+ SND_SOC_DAPM_INPUT("Mic 2"),
+
+ SND_SOC_DAPM_MIXER("MIC2 Mute",
+ AB8500_ANACONF2, AB8500_ANACONF2_MUTMIC2, 1,
+ NULL, 0),
+ SND_SOC_DAPM_MIXER("MIC2 V-AMICx Enable", AB8500_ANACONF2,
+ AB8500_ANACONF2_ENMIC2, 0,
+ NULL, 0),
+
+ /* LineIn */
+
+ SND_SOC_DAPM_INPUT("LineIn Left"),
+ SND_SOC_DAPM_INPUT("LineIn Right"),
+
+ SND_SOC_DAPM_MIXER("LINL Mute",
+ AB8500_ANACONF2, AB8500_ANACONF2_MUTLINL, 1,
+ NULL, 0),
+ SND_SOC_DAPM_MIXER("LINR Mute",
+ AB8500_ANACONF2, AB8500_ANACONF2_MUTLINR, 1,
+ NULL, 0),
+ SND_SOC_DAPM_MIXER("LINL Enable", AB8500_ANACONF2,
+ AB8500_ANACONF2_ENLINL, 0,
+ NULL, 0),
+ SND_SOC_DAPM_MIXER("LINR Enable", AB8500_ANACONF2,
+ AB8500_ANACONF2_ENLINR, 0,
+ NULL, 0),
+
+ /* LineIn Bypass path */
+ SND_SOC_DAPM_MIXER("LINL to HSL Volume",
+ SND_SOC_NOPM, 0, 0,
+ NULL, 0),
+ SND_SOC_DAPM_MIXER("LINR to HSR Volume",
+ SND_SOC_NOPM, 0, 0,
+ NULL, 0),
+
+ /* LineIn, Mic 2 */
+ SND_SOC_DAPM_MUX("Mic 2 or LINR Select",
+ SND_SOC_NOPM, 0, 0, dapm_mic2lr_select),
+ SND_SOC_DAPM_MIXER("LINL ADC", AB8500_ANACONF3,
+ AB8500_ANACONF3_ENADCLINL, 0,
+ NULL, 0),
+ SND_SOC_DAPM_MIXER("LINR ADC", AB8500_ANACONF3,
+ AB8500_ANACONF3_ENADCLINR, 0,
+ NULL, 0),
+ SND_SOC_DAPM_MUX("AD1 Source Select",
+ SND_SOC_NOPM, 0, 0, dapm_ad1_select),
+ SND_SOC_DAPM_MUX("AD2 Source Select",
+ SND_SOC_NOPM, 0, 0, dapm_ad2_select),
+ SND_SOC_DAPM_MIXER("AD1 Channel Volume",
+ SND_SOC_NOPM, 0, 0,
+ NULL, 0),
+ SND_SOC_DAPM_MIXER("AD2 Channel Volume",
+ SND_SOC_NOPM, 0, 0,
+ NULL, 0),
+
+ SND_SOC_DAPM_MIXER("AD12 Enable",
+ AB8500_ADPATHENA, AB8500_ADPATHENA_ENAD12, 0,
+ NULL, 0),
+
+ /* HD Capture path */
+
+ SND_SOC_DAPM_MUX("AD5 Source Select",
+ SND_SOC_NOPM, 0, 0, dapm_ad5_select),
+ SND_SOC_DAPM_MUX("AD6 Source Select",
+ SND_SOC_NOPM, 0, 0, dapm_ad6_select),
+ SND_SOC_DAPM_MIXER("AD5 Channel Volume",
+ SND_SOC_NOPM, 0, 0,
+ NULL, 0),
+ SND_SOC_DAPM_MIXER("AD6 Channel Volume",
+ SND_SOC_NOPM, 0, 0,
+ NULL, 0),
+ SND_SOC_DAPM_MIXER("AD57 Enable",
+ AB8500_ADPATHENA, AB8500_ADPATHENA_ENAD5768, 0,
+ NULL, 0),
+ SND_SOC_DAPM_MIXER("AD68 Enable",
+ AB8500_ADPATHENA, AB8500_ADPATHENA_ENAD5768, 0,
+ NULL, 0),
+
+ /* Digital Microphone path */
+
+ SND_SOC_DAPM_INPUT("DMic 1"),
+ SND_SOC_DAPM_INPUT("DMic 2"),
+ SND_SOC_DAPM_INPUT("DMic 3"),
+ SND_SOC_DAPM_INPUT("DMic 4"),
+ SND_SOC_DAPM_INPUT("DMic 5"),
+ SND_SOC_DAPM_INPUT("DMic 6"),
+
+ SND_SOC_DAPM_MIXER("DMIC1",
+ AB8500_DIGMICCONF, AB8500_DIGMICCONF_ENDMIC1, 0,
+ NULL, 0),
+ SND_SOC_DAPM_MIXER("DMIC2",
+ AB8500_DIGMICCONF, AB8500_DIGMICCONF_ENDMIC2, 0,
+ NULL, 0),
+ SND_SOC_DAPM_MIXER("DMIC3",
+ AB8500_DIGMICCONF, AB8500_DIGMICCONF_ENDMIC3, 0,
+ NULL, 0),
+ SND_SOC_DAPM_MIXER("DMIC4",
+ AB8500_DIGMICCONF, AB8500_DIGMICCONF_ENDMIC4, 0,
+ NULL, 0),
+ SND_SOC_DAPM_MIXER("DMIC5",
+ AB8500_DIGMICCONF, AB8500_DIGMICCONF_ENDMIC5, 0,
+ NULL, 0),
+ SND_SOC_DAPM_MIXER("DMIC6",
+ AB8500_DIGMICCONF, AB8500_DIGMICCONF_ENDMIC6, 0,
+ NULL, 0),
+ SND_SOC_DAPM_MIXER("AD4 Channel Volume",
+ SND_SOC_NOPM, 0, 0,
+ NULL, 0),
+ SND_SOC_DAPM_MIXER("AD4 Enable",
+ AB8500_ADPATHENA, AB8500_ADPATHENA_ENAD34,
+ 0, NULL, 0),
+
+ /* Acoustical Noise Cancellation path */
+
+ SND_SOC_DAPM_INPUT("ANC Configure Input"),
+ SND_SOC_DAPM_OUTPUT("ANC Configure Output"),
+
+ SND_SOC_DAPM_MUX("ANC Source",
+ SND_SOC_NOPM, 0, 0,
+ dapm_anc_in_select),
+ SND_SOC_DAPM_SWITCH("ANC",
+ SND_SOC_NOPM, 0, 0,
+ dapm_anc_enable),
+ SND_SOC_DAPM_SWITCH("ANC to Earpiece",
+ SND_SOC_NOPM, 0, 0,
+ dapm_anc_ear_mute),
+
+ /* Sidetone Filter path */
+
+ SND_SOC_DAPM_MUX("Sidetone Left Source",
+ SND_SOC_NOPM, 0, 0,
+ dapm_stfir1_in_select),
+ SND_SOC_DAPM_MUX("Sidetone Right Source",
+ SND_SOC_NOPM, 0, 0,
+ dapm_stfir2_in_select),
+ SND_SOC_DAPM_MIXER("STFIR1 Control",
+ SND_SOC_NOPM, 0, 0,
+ NULL, 0),
+ SND_SOC_DAPM_MIXER("STFIR2 Control",
+ SND_SOC_NOPM, 0, 0,
+ NULL, 0),
+ SND_SOC_DAPM_MIXER("STFIR1 Volume",
+ SND_SOC_NOPM, 0, 0,
+ NULL, 0),
+ SND_SOC_DAPM_MIXER("STFIR2 Volume",
+ SND_SOC_NOPM, 0, 0,
+ NULL, 0),
+};
+
+/*
+ * DAPM-routes
+ */
+static const struct snd_soc_dapm_route ab8500_dapm_routes[] = {
+ /* Power AB8500 audio-block when AD/DA is active */
+ {"Main Supply", NULL, "V-AUD"},
+ {"Main Supply", NULL, "audioclk"},
+ {"Main Supply", NULL, "Audio Power"},
+ {"Main Supply", NULL, "Audio Analog Power"},
+
+ {"DAC", NULL, "ab8500_0p"},
+ {"DAC", NULL, "Main Supply"},
+ {"ADC", NULL, "ab8500_0c"},
+ {"ADC", NULL, "Main Supply"},
+
+ /* ANC Configure */
+ {"ANC Configure Input", NULL, "Main Supply"},
+ {"ANC Configure Output", NULL, "ANC Configure Input"},
+
+ /* AD/DA */
+ {"ADC", NULL, "ADC Input"},
+ {"DAC Output", NULL, "DAC"},
+
+ /* Powerup charge pump if DA1/2 is in use */
+
+ {"DA_IN1", NULL, "ab8500_0p"},
+ {"DA_IN1", NULL, "Charge Pump"},
+ {"DA_IN2", NULL, "ab8500_0p"},
+ {"DA_IN2", NULL, "Charge Pump"},
+
+ /* Headset path */
+
+ {"DA1 Enable", NULL, "DA_IN1"},
+ {"DA2 Enable", NULL, "DA_IN2"},
+
+ {"HSL Digital Volume", NULL, "DA1 Enable"},
+ {"HSR Digital Volume", NULL, "DA2 Enable"},
+
+ {"HSL DAC", NULL, "HSL Digital Volume"},
+ {"HSR DAC", NULL, "HSR Digital Volume"},
+
+ {"HSL DAC Mute", NULL, "HSL DAC"},
+ {"HSR DAC Mute", NULL, "HSR DAC"},
+
+ {"HSL DAC Driver", NULL, "HSL DAC Mute"},
+ {"HSR DAC Driver", NULL, "HSR DAC Mute"},
+
+ {"HSL Mute", NULL, "HSL DAC Driver"},
+ {"HSR Mute", NULL, "HSR DAC Driver"},
+
+ {"HSL Enable", NULL, "HSL Mute"},
+ {"HSR Enable", NULL, "HSR Mute"},
+
+ {"HSL Volume", NULL, "HSL Enable"},
+ {"HSR Volume", NULL, "HSR Enable"},
+
+ {"Headset Left", NULL, "HSL Volume"},
+ {"Headset Right", NULL, "HSR Volume"},
+
+ /* HF or LineOut path */
+
+ {"DA_IN3", NULL, "ab8500_0p"},
+ {"DA3 Channel Volume", NULL, "DA_IN3"},
+ {"DA_IN4", NULL, "ab8500_0p"},
+ {"DA4 Channel Volume", NULL, "DA_IN4"},
+
+ {"Speaker Left Source", "Audio Path", "DA3 Channel Volume"},
+ {"Speaker Right Source", "Audio Path", "DA4 Channel Volume"},
+
+ {"DA3 or ANC path to HfL", NULL, "Speaker Left Source"},
+ {"DA4 or ANC path to HfR", NULL, "Speaker Right Source"},
+
+ /* HF path */
+
+ {"HFL DAC", NULL, "DA3 or ANC path to HfL"},
+ {"HFR DAC", NULL, "DA4 or ANC path to HfR"},
+
+ {"HFL Enable", NULL, "HFL DAC"},
+ {"HFR Enable", NULL, "HFR DAC"},
+
+ {"Speaker Left", NULL, "HFL Enable"},
+ {"Speaker Right", NULL, "HFR Enable"},
+
+ /* Earpiece path */
+
+ {"Earpiece or LineOut Mono Source", "Headset Left",
+ "HSL Digital Volume"},
+ {"Earpiece or LineOut Mono Source", "Speaker Left",
+ "DA3 or ANC path to HfL"},
+
+ {"EAR DAC", NULL, "Earpiece or LineOut Mono Source"},
+
+ {"EAR Mute", NULL, "EAR DAC"},
+
+ {"EAR Enable", NULL, "EAR Mute"},
+
+ {"Earpiece", NULL, "EAR Enable"},
+
+ /* LineOut path stereo */
+
+ {"LineOut Source", "Stereo Path", "HSL DAC Driver"},
+ {"LineOut Source", "Stereo Path", "HSR DAC Driver"},
+
+ /* LineOut path mono */
+
+ {"LineOut Source", "Mono Path", "EAR DAC"},
+
+ /* LineOut path */
+
+ {"LOL Disable HFL", NULL, "LineOut Source"},
+ {"LOR Disable HFR", NULL, "LineOut Source"},
+
+ {"LOL Enable", NULL, "LOL Disable HFL"},
+ {"LOR Enable", NULL, "LOR Disable HFR"},
+
+ {"LineOut Left", NULL, "LOL Enable"},
+ {"LineOut Right", NULL, "LOR Enable"},
+
+ /* Vibrator path */
+
+ {"DA_IN5", NULL, "ab8500_0p"},
+ {"DA5 Channel Volume", NULL, "DA_IN5"},
+ {"DA_IN6", NULL, "ab8500_0p"},
+ {"DA6 Channel Volume", NULL, "DA_IN6"},
+
+ {"VIB1 DAC", NULL, "DA5 Channel Volume"},
+ {"VIB2 DAC", NULL, "DA6 Channel Volume"},
+
+ {"Vibra 1 Controller", "Audio Path", "VIB1 DAC"},
+ {"Vibra 2 Controller", "Audio Path", "VIB2 DAC"},
+ {"Vibra 1 Controller", "PWM Generator", "PWMGEN1"},
+ {"Vibra 2 Controller", "PWM Generator", "PWMGEN2"},
+
+ {"VIB1 Enable", NULL, "Vibra 1 Controller"},
+ {"VIB2 Enable", NULL, "Vibra 2 Controller"},
+
+ {"Vibra 1", NULL, "VIB1 Enable"},
+ {"Vibra 2", NULL, "VIB2 Enable"},
+
+
+ /* Mic 2 */
+
+ {"MIC2 V-AMICx Enable", NULL, "Mic 2"},
+
+ /* LineIn */
+ {"LINL Mute", NULL, "LineIn Left"},
+ {"LINR Mute", NULL, "LineIn Right"},
+
+ {"LINL Enable", NULL, "LINL Mute"},
+ {"LINR Enable", NULL, "LINR Mute"},
+
+ /* LineIn, Mic 2 */
+ {"Mic 2 or LINR Select", "LineIn Right", "LINR Enable"},
+ {"Mic 2 or LINR Select", "Mic 2", "MIC2 V-AMICx Enable"},
+
+ {"LINL ADC", NULL, "LINL Enable"},
+ {"LINR ADC", NULL, "Mic 2 or LINR Select"},
+
+ {"AD1 Source Select", "LineIn Left", "LINL ADC"},
+ {"AD2 Source Select", "LineIn Right", "LINR ADC"},
+
+ {"AD1 Channel Volume", NULL, "AD1 Source Select"},
+ {"AD2 Channel Volume", NULL, "AD2 Source Select"},
+
+ {"AD12 Enable", NULL, "AD1 Channel Volume"},
+ {"AD12 Enable", NULL, "AD2 Channel Volume"},
+
+ {"AD_OUT1", NULL, "ab8500_0c"},
+ {"AD_OUT1", NULL, "AD12 Enable"},
+ {"AD_OUT2", NULL, "ab8500_0c"},
+ {"AD_OUT2", NULL, "AD12 Enable"},
+
+ /* Mic 1 */
+
+ {"MIC1 Mute", NULL, "Mic 1"},
+
+ {"MIC1A V-AMICx Enable", NULL, "MIC1 Mute"},
+ {"MIC1B V-AMICx Enable", NULL, "MIC1 Mute"},
+
+ {"Mic 1a or 1b Select", "Mic 1a", "MIC1A V-AMICx Enable"},
+ {"Mic 1a or 1b Select", "Mic 1b", "MIC1B V-AMICx Enable"},
+
+ {"MIC1 ADC", NULL, "Mic 1a or 1b Select"},
+
+ {"AD3 Source Select", "Mic 1", "MIC1 ADC"},
+
+ {"AD3 Channel Volume", NULL, "AD3 Source Select"},
+
+ {"AD3 Enable", NULL, "AD3 Channel Volume"},
+
+ {"AD_OUT3", NULL, "ab8500_0c"},
+ {"AD_OUT3", NULL, "AD3 Enable"},
+
+ /* HD Capture path */
+
+ {"AD5 Source Select", "Mic 2", "LINR ADC"},
+ {"AD6 Source Select", "Mic 1", "MIC1 ADC"},
+
+ {"AD5 Channel Volume", NULL, "AD5 Source Select"},
+ {"AD6 Channel Volume", NULL, "AD6 Source Select"},
+
+ {"AD57 Enable", NULL, "AD5 Channel Volume"},
+ {"AD68 Enable", NULL, "AD6 Channel Volume"},
+
+ {"AD_OUT57", NULL, "ab8500_0c"},
+ {"AD_OUT57", NULL, "AD57 Enable"},
+ {"AD_OUT68", NULL, "ab8500_0c"},
+ {"AD_OUT68", NULL, "AD68 Enable"},
+
+ /* Digital Microphone path */
+
+ {"DMic 1", NULL, "V-DMIC"},
+ {"DMic 2", NULL, "V-DMIC"},
+ {"DMic 3", NULL, "V-DMIC"},
+ {"DMic 4", NULL, "V-DMIC"},
+ {"DMic 5", NULL, "V-DMIC"},
+ {"DMic 6", NULL, "V-DMIC"},
+
+ {"AD1 Source Select", NULL, "DMic 1"},
+ {"AD2 Source Select", NULL, "DMic 2"},
+ {"AD3 Source Select", NULL, "DMic 3"},
+ {"AD5 Source Select", NULL, "DMic 5"},
+ {"AD6 Source Select", NULL, "DMic 6"},
+
+ {"AD4 Channel Volume", NULL, "DMic 4"},
+ {"AD4 Enable", NULL, "AD4 Channel Volume"},
+
+ {"AD_OUT4", NULL, "ab8500_0c"},
+ {"AD_OUT4", NULL, "AD4 Enable"},
+
+ /* LineIn Bypass path */
+
+ {"LINL to HSL Volume", NULL, "LINL Enable"},
+ {"LINR to HSR Volume", NULL, "LINR Enable"},
+
+ {"HSL DAC Driver", NULL, "LINL to HSL Volume"},
+ {"HSR DAC Driver", NULL, "LINR to HSR Volume"},
+
+ /* ANC path (Acoustic Noise Cancellation) */
+
+ {"ANC Source", "Mic 2 / DMic 5", "AD5 Channel Volume"},
+ {"ANC Source", "Mic 1 / DMic 6", "AD6 Channel Volume"},
+
+ {"ANC", "Switch", "ANC Source"},
+
+ {"Speaker Left Source", "ANC", "ANC"},
+ {"Speaker Right Source", "ANC", "ANC"},
+ {"ANC to Earpiece", "Switch", "ANC"},
+
+ {"HSL Digital Volume", NULL, "ANC to Earpiece"},
+
+ /* Sidetone Filter path */
+
+ {"Sidetone Left Source", "LineIn Left", "AD12 Enable"},
+ {"Sidetone Left Source", "LineIn Right", "AD12 Enable"},
+ {"Sidetone Left Source", "Mic 1", "AD3 Enable"},
+ {"Sidetone Left Source", "Headset Left", "DA_IN1"},
+ {"Sidetone Right Source", "LineIn Right", "AD12 Enable"},
+ {"Sidetone Right Source", "Mic 1", "AD3 Enable"},
+ {"Sidetone Right Source", "DMic 4", "AD4 Enable"},
+ {"Sidetone Right Source", "Headset Right", "DA_IN2"},
+
+ {"STFIR1 Control", NULL, "Sidetone Left Source"},
+ {"STFIR2 Control", NULL, "Sidetone Right Source"},
+
+ {"STFIR1 Volume", NULL, "STFIR1 Control"},
+ {"STFIR2 Volume", NULL, "STFIR2 Control"},
+
+ {"DA1 Enable", NULL, "STFIR1 Volume"},
+ {"DA2 Enable", NULL, "STFIR2 Volume"},
+};
+
+static const struct snd_soc_dapm_route ab8500_dapm_routes_mic1a_vamicx[] = {
+ {"MIC1A V-AMICx Enable", NULL, "V-AMIC1"},
+ {"MIC1A V-AMICx Enable", NULL, "V-AMIC2"},
+};
+
+static const struct snd_soc_dapm_route ab8500_dapm_routes_mic1b_vamicx[] = {
+ {"MIC1B V-AMICx Enable", NULL, "V-AMIC1"},
+ {"MIC1B V-AMICx Enable", NULL, "V-AMIC2"},
+};
+
+static const struct snd_soc_dapm_route ab8500_dapm_routes_mic2_vamicx[] = {
+ {"MIC2 V-AMICx Enable", NULL, "V-AMIC1"},
+ {"MIC2 V-AMICx Enable", NULL, "V-AMIC2"},
+};
+
+/* ANC FIR-coefficients configuration sequence */
+static void anc_fir(struct snd_soc_codec *codec,
+ unsigned int bnk, unsigned int par, unsigned int val)
+{
+ if (par == 0 && bnk == 0)
+ snd_soc_update_bits(codec, AB8500_ANCCONF1,
+ BIT(AB8500_ANCCONF1_ANCFIRUPDATE),
+ BIT(AB8500_ANCCONF1_ANCFIRUPDATE));
+
+ snd_soc_write(codec, AB8500_ANCCONF5, val >> 8 & 0xff);
+ snd_soc_write(codec, AB8500_ANCCONF6, val & 0xff);
+
+ if (par == AB8500_ANC_FIR_COEFFS - 1 && bnk == 1)
+ snd_soc_update_bits(codec, AB8500_ANCCONF1,
+ BIT(AB8500_ANCCONF1_ANCFIRUPDATE), 0);
+}
+
+/* ANC IIR-coefficients configuration sequence */
+static void anc_iir(struct snd_soc_codec *codec, unsigned int bnk,
+ unsigned int par, unsigned int val)
+{
+ if (par == 0) {
+ if (bnk == 0) {
+ snd_soc_update_bits(codec, AB8500_ANCCONF1,
+ BIT(AB8500_ANCCONF1_ANCIIRINIT),
+ BIT(AB8500_ANCCONF1_ANCIIRINIT));
+ usleep_range(AB8500_ANC_SM_DELAY, AB8500_ANC_SM_DELAY);
+ snd_soc_update_bits(codec, AB8500_ANCCONF1,
+ BIT(AB8500_ANCCONF1_ANCIIRINIT), 0);
+ usleep_range(AB8500_ANC_SM_DELAY, AB8500_ANC_SM_DELAY);
+ } else {
+ snd_soc_update_bits(codec, AB8500_ANCCONF1,
+ BIT(AB8500_ANCCONF1_ANCIIRUPDATE),
+ BIT(AB8500_ANCCONF1_ANCIIRUPDATE));
+ }
+ } else if (par > 3) {
+ snd_soc_write(codec, AB8500_ANCCONF7, 0);
+ snd_soc_write(codec, AB8500_ANCCONF8, val >> 16 & 0xff);
+ }
+
+ snd_soc_write(codec, AB8500_ANCCONF7, val >> 8 & 0xff);
+ snd_soc_write(codec, AB8500_ANCCONF8, val & 0xff);
+
+ if (par == AB8500_ANC_IIR_COEFFS - 1 && bnk == 1)
+ snd_soc_update_bits(codec, AB8500_ANCCONF1,
+ BIT(AB8500_ANCCONF1_ANCIIRUPDATE), 0);
+}
+
+/* ANC IIR-/FIR-coefficients configuration sequence */
+static void anc_configure(struct snd_soc_codec *codec,
+ bool apply_fir, bool apply_iir)
+{
+ struct ab8500_codec_drvdata *drvdata = dev_get_drvdata(codec->dev);
+ unsigned int bnk, par, val;
+
+ dev_dbg(codec->dev, "%s: Enter.\n", __func__);
+
+ if (apply_fir)
+ snd_soc_update_bits(codec, AB8500_ANCCONF1,
+ BIT(AB8500_ANCCONF1_ENANC), 0);
+
+ snd_soc_update_bits(codec, AB8500_ANCCONF1,
+ BIT(AB8500_ANCCONF1_ENANC), BIT(AB8500_ANCCONF1_ENANC));
+
+ if (apply_fir)
+ for (bnk = 0; bnk < AB8500_NR_OF_ANC_COEFF_BANKS; bnk++)
+ for (par = 0; par < AB8500_ANC_FIR_COEFFS; par++) {
+ val = snd_soc_read(codec,
+ drvdata->anc_fir_values[par]);
+ anc_fir(codec, bnk, par, val);
+ }
+
+ if (apply_iir)
+ for (bnk = 0; bnk < AB8500_NR_OF_ANC_COEFF_BANKS; bnk++)
+ for (par = 0; par < AB8500_ANC_IIR_COEFFS; par++) {
+ val = snd_soc_read(codec,
+ drvdata->anc_iir_values[par]);
+ anc_iir(codec, bnk, par, val);
+ }
+
+ dev_dbg(codec->dev, "%s: Exit.\n", __func__);
+}
+
+/*
+ * Control-events
+ */
+
+static int sid_status_control_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct ab8500_codec_drvdata *drvdata = dev_get_drvdata(codec->dev);
+
+ mutex_lock(&codec->mutex);
+ ucontrol->value.integer.value[0] = drvdata->sid_status;
+ mutex_unlock(&codec->mutex);
+
+ return 0;
+}
+
+/* Write sidetone FIR-coefficients configuration sequence */
+static int sid_status_control_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct ab8500_codec_drvdata *drvdata = dev_get_drvdata(codec->dev);
+ unsigned int param, sidconf, val;
+ int status = 1;
+
+ dev_dbg(codec->dev, "%s: Enter\n", __func__);
+
+ if (ucontrol->value.integer.value[0] != SID_APPLY_FIR) {
+ dev_err(codec->dev,
+ "%s: ERROR: This control supports '%s' only!\n",
+ __func__, enum_sid_state[SID_APPLY_FIR]);
+ return -EIO;
+ }
+
+ mutex_lock(&codec->mutex);
+
+ sidconf = snd_soc_read(codec, AB8500_SIDFIRCONF);
+ if (((sidconf & BIT(AB8500_SIDFIRCONF_FIRSIDBUSY)) != 0)) {
+ if ((sidconf & BIT(AB8500_SIDFIRCONF_ENFIRSIDS)) == 0) {
+ dev_err(codec->dev, "%s: Sidetone busy while off!\n",
+ __func__);
+ status = -EPERM;
+ } else {
+ status = -EBUSY;
+ }
+ goto out;
+ }
+
+ snd_soc_write(codec, AB8500_SIDFIRADR, 0);
+
+ for (param = 0; param < AB8500_SID_FIR_COEFFS; param++) {
+ val = snd_soc_read(codec, drvdata->sid_fir_values[param]);
+ snd_soc_write(codec, AB8500_SIDFIRCOEF1, val >> 8 & 0xff);
+ snd_soc_write(codec, AB8500_SIDFIRCOEF2, val & 0xff);
+ }
+
+ snd_soc_update_bits(codec, AB8500_SIDFIRADR,
+ BIT(AB8500_SIDFIRADR_FIRSIDSET),
+ BIT(AB8500_SIDFIRADR_FIRSIDSET));
+ snd_soc_update_bits(codec, AB8500_SIDFIRADR,
+ BIT(AB8500_SIDFIRADR_FIRSIDSET), 0);
+
+ drvdata->sid_status = SID_FIR_CONFIGURED;
+
+out:
+ mutex_unlock(&codec->mutex);
+
+ dev_dbg(codec->dev, "%s: Exit\n", __func__);
+
+ return status;
+}
+
+static int anc_status_control_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct ab8500_codec_drvdata *drvdata = dev_get_drvdata(codec->dev);
+
+ mutex_lock(&codec->mutex);
+ ucontrol->value.integer.value[0] = drvdata->anc_status;
+ mutex_unlock(&codec->mutex);
+
+ return 0;
+}
+
+static int anc_status_control_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct ab8500_codec_drvdata *drvdata = dev_get_drvdata(codec->dev);
+ struct device *dev = codec->dev;
+ bool apply_fir, apply_iir;
+ int req, status;
+
+ dev_dbg(dev, "%s: Enter.\n", __func__);
+
+ mutex_lock(&drvdata->anc_lock);
+
+ req = ucontrol->value.integer.value[0];
+ if (req != ANC_APPLY_FIR_IIR && req != ANC_APPLY_FIR &&
+ req != ANC_APPLY_IIR) {
+ dev_err(dev, "%s: ERROR: Unsupported status to set '%s'!\n",
+ __func__, enum_anc_state[req]);
+ status = -EINVAL;
+ goto cleanup;
+ }
+ apply_fir = req == ANC_APPLY_FIR || req == ANC_APPLY_FIR_IIR;
+ apply_iir = req == ANC_APPLY_IIR || req == ANC_APPLY_FIR_IIR;
+
+ status = snd_soc_dapm_force_enable_pin(&codec->dapm,
+ "ANC Configure Input");
+ if (status < 0) {
+ dev_err(dev,
+ "%s: ERROR: Failed to enable power (status = %d)!\n",
+ __func__, status);
+ goto cleanup;
+ }
+ snd_soc_dapm_sync(&codec->dapm);
+
+ mutex_lock(&codec->mutex);
+ anc_configure(codec, apply_fir, apply_iir);
+ mutex_unlock(&codec->mutex);
+
+ if (apply_fir) {
+ if (drvdata->anc_status == ANC_IIR_CONFIGURED)
+ drvdata->anc_status = ANC_FIR_IIR_CONFIGURED;
+ else if (drvdata->anc_status != ANC_FIR_IIR_CONFIGURED)
+ drvdata->anc_status = ANC_FIR_CONFIGURED;
+ }
+ if (apply_iir) {
+ if (drvdata->anc_status == ANC_FIR_CONFIGURED)
+ drvdata->anc_status = ANC_FIR_IIR_CONFIGURED;
+ else if (drvdata->anc_status != ANC_FIR_IIR_CONFIGURED)
+ drvdata->anc_status = ANC_IIR_CONFIGURED;
+ }
+
+ status = snd_soc_dapm_disable_pin(&codec->dapm, "ANC Configure Input");
+ snd_soc_dapm_sync(&codec->dapm);
+
+cleanup:
+ mutex_unlock(&drvdata->anc_lock);
+
+ if (status < 0)
+ dev_err(dev, "%s: Unable to configure ANC! (status = %d)\n",
+ __func__, status);
+
+ dev_dbg(dev, "%s: Exit.\n", __func__);
+
+ return (status < 0) ? status : 1;
+}
+
+static int filter_control_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ struct filter_control *fc =
+ (struct filter_control *)kcontrol->private_value;
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = fc->count;
+ uinfo->value.integer.min = fc->min;
+ uinfo->value.integer.max = fc->max;
+
+ return 0;
+}
+
+static int filter_control_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct filter_control *fc =
+ (struct filter_control *)kcontrol->private_value;
+ unsigned int i;
+
+ mutex_lock(&codec->mutex);
+ for (i = 0; i < fc->count; i++)
+ ucontrol->value.integer.value[i] = fc->value[i];
+ mutex_unlock(&codec->mutex);
+
+ return 0;
+}
+
+static int filter_control_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct filter_control *fc =
+ (struct filter_control *)kcontrol->private_value;
+ unsigned int i;
+
+ mutex_lock(&codec->mutex);
+ for (i = 0; i < fc->count; i++)
+ fc->value[i] = ucontrol->value.integer.value[i];
+ mutex_unlock(&codec->mutex);
+
+ return 0;
+}
+
+/*
+ * Controls - Non-DAPM ASoC
+ */
+
+static DECLARE_TLV_DB_SCALE(adx_dig_gain_tlv, -3200, 100, 1);
+/* -32dB = Mute */
+
+static DECLARE_TLV_DB_SCALE(dax_dig_gain_tlv, -6300, 100, 1);
+/* -63dB = Mute */
+
+static DECLARE_TLV_DB_SCALE(hs_ear_dig_gain_tlv, -100, 100, 1);
+/* -1dB = Mute */
+
+static const unsigned int hs_gain_tlv[] = {
+ TLV_DB_RANGE_HEAD(2),
+ 0, 3, TLV_DB_SCALE_ITEM(-3200, 400, 0),
+ 4, 15, TLV_DB_SCALE_ITEM(-1800, 200, 0),
+};
+
+static DECLARE_TLV_DB_SCALE(mic_gain_tlv, 0, 100, 0);
+
+static DECLARE_TLV_DB_SCALE(lin_gain_tlv, -1000, 200, 0);
+
+static DECLARE_TLV_DB_SCALE(lin2hs_gain_tlv, -3800, 200, 1);
+/* -38dB = Mute */
+
+static const char * const enum_hsfadspeed[] = {"2ms", "0.5ms", "10.6ms",
+ "5ms"};
+static SOC_ENUM_SINGLE_DECL(soc_enum_hsfadspeed,
+ AB8500_DIGMICCONF, AB8500_DIGMICCONF_HSFADSPEED, enum_hsfadspeed);
+
+static const char * const enum_envdetthre[] = {
+ "250mV", "300mV", "350mV", "400mV",
+ "450mV", "500mV", "550mV", "600mV",
+ "650mV", "700mV", "750mV", "800mV",
+ "850mV", "900mV", "950mV", "1.00V" };
+static SOC_ENUM_SINGLE_DECL(soc_enum_envdeththre,
+ AB8500_ENVCPCONF, AB8500_ENVCPCONF_ENVDETHTHRE, enum_envdetthre);
+static SOC_ENUM_SINGLE_DECL(soc_enum_envdetlthre,
+ AB8500_ENVCPCONF, AB8500_ENVCPCONF_ENVDETLTHRE, enum_envdetthre);
+static const char * const enum_envdettime[] = {
+ "26.6us", "53.2us", "106us", "213us",
+ "426us", "851us", "1.70ms", "3.40ms",
+ "6.81ms", "13.6ms", "27.2ms", "54.5ms",
+ "109ms", "218ms", "436ms", "872ms" };
+static SOC_ENUM_SINGLE_DECL(soc_enum_envdettime,
+ AB8500_SIGENVCONF, AB8500_SIGENVCONF_ENVDETTIME, enum_envdettime);
+
+static const char * const enum_sinc31[] = {"Sinc 3", "Sinc 1"};
+static SOC_ENUM_SINGLE_DECL(soc_enum_hsesinc, AB8500_HSLEARDIGGAIN,
+ AB8500_HSLEARDIGGAIN_HSSINC1, enum_sinc31);
+
+static const char * const enum_fadespeed[] = {"1ms", "4ms", "8ms", "16ms"};
+static SOC_ENUM_SINGLE_DECL(soc_enum_fadespeed, AB8500_HSRDIGGAIN,
+ AB8500_HSRDIGGAIN_FADESPEED, enum_fadespeed);
+
+/* Earpiece */
+
+static const char * const enum_lowpow[] = {"Normal", "Low Power"};
+static SOC_ENUM_SINGLE_DECL(soc_enum_eardaclowpow, AB8500_ANACONF1,
+ AB8500_ANACONF1_EARDACLOWPOW, enum_lowpow);
+static SOC_ENUM_SINGLE_DECL(soc_enum_eardrvlowpow, AB8500_ANACONF1,
+ AB8500_ANACONF1_EARDRVLOWPOW, enum_lowpow);
+
+static const char * const enum_av_mode[] = {"Audio", "Voice"};
+static SOC_ENUM_DOUBLE_DECL(soc_enum_ad12voice, AB8500_ADFILTCONF,
+ AB8500_ADFILTCONF_AD1VOICE, AB8500_ADFILTCONF_AD2VOICE, enum_av_mode);
+static SOC_ENUM_DOUBLE_DECL(soc_enum_ad34voice, AB8500_ADFILTCONF,
+ AB8500_ADFILTCONF_AD3VOICE, AB8500_ADFILTCONF_AD4VOICE, enum_av_mode);
+
+/* DA */
+
+static SOC_ENUM_SINGLE_DECL(soc_enum_da12voice,
+ AB8500_DASLOTCONF1, AB8500_DASLOTCONF1_DA12VOICE,
+ enum_av_mode);
+static SOC_ENUM_SINGLE_DECL(soc_enum_da34voice,
+ AB8500_DASLOTCONF3, AB8500_DASLOTCONF3_DA34VOICE,
+ enum_av_mode);
+static SOC_ENUM_SINGLE_DECL(soc_enum_da56voice,
+ AB8500_DASLOTCONF5, AB8500_DASLOTCONF5_DA56VOICE,
+ enum_av_mode);
+
+static const char * const enum_da2hslr[] = {"Sidetone", "Audio Path"};
+static SOC_ENUM_DOUBLE_DECL(soc_enum_da2hslr, AB8500_DIGMULTCONF1,
+ AB8500_DIGMULTCONF1_DATOHSLEN,
+ AB8500_DIGMULTCONF1_DATOHSREN, enum_da2hslr);
+
+static const char * const enum_sinc53[] = {"Sinc 5", "Sinc 3"};
+static SOC_ENUM_DOUBLE_DECL(soc_enum_dmic12sinc, AB8500_DMICFILTCONF,
+ AB8500_DMICFILTCONF_DMIC1SINC3,
+ AB8500_DMICFILTCONF_DMIC2SINC3, enum_sinc53);
+static SOC_ENUM_DOUBLE_DECL(soc_enum_dmic34sinc, AB8500_DMICFILTCONF,
+ AB8500_DMICFILTCONF_DMIC3SINC3,
+ AB8500_DMICFILTCONF_DMIC4SINC3, enum_sinc53);
+static SOC_ENUM_DOUBLE_DECL(soc_enum_dmic56sinc, AB8500_DMICFILTCONF,
+ AB8500_DMICFILTCONF_DMIC5SINC3,
+ AB8500_DMICFILTCONF_DMIC6SINC3, enum_sinc53);
+
+/* Digital interface - DA from slot mapping */
+static const char * const enum_da_from_slot_map[] = {"SLOT0",
+ "SLOT1",
+ "SLOT2",
+ "SLOT3",
+ "SLOT4",
+ "SLOT5",
+ "SLOT6",
+ "SLOT7",
+ "SLOT8",
+ "SLOT9",
+ "SLOT10",
+ "SLOT11",
+ "SLOT12",
+ "SLOT13",
+ "SLOT14",
+ "SLOT15",
+ "SLOT16",
+ "SLOT17",
+ "SLOT18",
+ "SLOT19",
+ "SLOT20",
+ "SLOT21",
+ "SLOT22",
+ "SLOT23",
+ "SLOT24",
+ "SLOT25",
+ "SLOT26",
+ "SLOT27",
+ "SLOT28",
+ "SLOT29",
+ "SLOT30",
+ "SLOT31"};
+static SOC_ENUM_SINGLE_DECL(soc_enum_da1slotmap,
+ AB8500_DASLOTCONF1, AB8500_DASLOTCONFX_SLTODAX_SHIFT,
+ enum_da_from_slot_map);
+static SOC_ENUM_SINGLE_DECL(soc_enum_da2slotmap,
+ AB8500_DASLOTCONF2, AB8500_DASLOTCONFX_SLTODAX_SHIFT,
+ enum_da_from_slot_map);
+static SOC_ENUM_SINGLE_DECL(soc_enum_da3slotmap,
+ AB8500_DASLOTCONF3, AB8500_DASLOTCONFX_SLTODAX_SHIFT,
+ enum_da_from_slot_map);
+static SOC_ENUM_SINGLE_DECL(soc_enum_da4slotmap,
+ AB8500_DASLOTCONF4, AB8500_DASLOTCONFX_SLTODAX_SHIFT,
+ enum_da_from_slot_map);
+static SOC_ENUM_SINGLE_DECL(soc_enum_da5slotmap,
+ AB8500_DASLOTCONF5, AB8500_DASLOTCONFX_SLTODAX_SHIFT,
+ enum_da_from_slot_map);
+static SOC_ENUM_SINGLE_DECL(soc_enum_da6slotmap,
+ AB8500_DASLOTCONF6, AB8500_DASLOTCONFX_SLTODAX_SHIFT,
+ enum_da_from_slot_map);
+static SOC_ENUM_SINGLE_DECL(soc_enum_da7slotmap,
+ AB8500_DASLOTCONF7, AB8500_DASLOTCONFX_SLTODAX_SHIFT,
+ enum_da_from_slot_map);
+static SOC_ENUM_SINGLE_DECL(soc_enum_da8slotmap,
+ AB8500_DASLOTCONF8, AB8500_DASLOTCONFX_SLTODAX_SHIFT,
+ enum_da_from_slot_map);
+
+/* Digital interface - AD to slot mapping */
+static const char * const enum_ad_to_slot_map[] = {"AD_OUT1",
+ "AD_OUT2",
+ "AD_OUT3",
+ "AD_OUT4",
+ "AD_OUT5",
+ "AD_OUT6",
+ "AD_OUT7",
+ "AD_OUT8",
+ "zeroes",
+ "tristate"};
+static SOC_ENUM_SINGLE_DECL(soc_enum_adslot0map,
+ AB8500_ADSLOTSEL1, AB8500_ADSLOTSELX_EVEN_SHIFT,
+ enum_ad_to_slot_map);
+static SOC_ENUM_SINGLE_DECL(soc_enum_adslot1map,
+ AB8500_ADSLOTSEL1, AB8500_ADSLOTSELX_ODD_SHIFT,
+ enum_ad_to_slot_map);
+static SOC_ENUM_SINGLE_DECL(soc_enum_adslot2map,
+ AB8500_ADSLOTSEL2, AB8500_ADSLOTSELX_EVEN_SHIFT,
+ enum_ad_to_slot_map);
+static SOC_ENUM_SINGLE_DECL(soc_enum_adslot3map,
+ AB8500_ADSLOTSEL2, AB8500_ADSLOTSELX_ODD_SHIFT,
+ enum_ad_to_slot_map);
+static SOC_ENUM_SINGLE_DECL(soc_enum_adslot4map,
+ AB8500_ADSLOTSEL3, AB8500_ADSLOTSELX_EVEN_SHIFT,
+ enum_ad_to_slot_map);
+static SOC_ENUM_SINGLE_DECL(soc_enum_adslot5map,
+ AB8500_ADSLOTSEL3, AB8500_ADSLOTSELX_ODD_SHIFT,
+ enum_ad_to_slot_map);
+static SOC_ENUM_SINGLE_DECL(soc_enum_adslot6map,
+ AB8500_ADSLOTSEL4, AB8500_ADSLOTSELX_EVEN_SHIFT,
+ enum_ad_to_slot_map);
+static SOC_ENUM_SINGLE_DECL(soc_enum_adslot7map,
+ AB8500_ADSLOTSEL4, AB8500_ADSLOTSELX_ODD_SHIFT,
+ enum_ad_to_slot_map);
+static SOC_ENUM_SINGLE_DECL(soc_enum_adslot8map,
+ AB8500_ADSLOTSEL5, AB8500_ADSLOTSELX_EVEN_SHIFT,
+ enum_ad_to_slot_map);
+static SOC_ENUM_SINGLE_DECL(soc_enum_adslot9map,
+ AB8500_ADSLOTSEL5, AB8500_ADSLOTSELX_ODD_SHIFT,
+ enum_ad_to_slot_map);
+static SOC_ENUM_SINGLE_DECL(soc_enum_adslot10map,
+ AB8500_ADSLOTSEL6, AB8500_ADSLOTSELX_EVEN_SHIFT,
+ enum_ad_to_slot_map);
+static SOC_ENUM_SINGLE_DECL(soc_enum_adslot11map,
+ AB8500_ADSLOTSEL6, AB8500_ADSLOTSELX_ODD_SHIFT,
+ enum_ad_to_slot_map);
+static SOC_ENUM_SINGLE_DECL(soc_enum_adslot12map,
+ AB8500_ADSLOTSEL7, AB8500_ADSLOTSELX_EVEN_SHIFT,
+ enum_ad_to_slot_map);
+static SOC_ENUM_SINGLE_DECL(soc_enum_adslot13map,
+ AB8500_ADSLOTSEL7, AB8500_ADSLOTSELX_ODD_SHIFT,
+ enum_ad_to_slot_map);
+static SOC_ENUM_SINGLE_DECL(soc_enum_adslot14map,
+ AB8500_ADSLOTSEL8, AB8500_ADSLOTSELX_EVEN_SHIFT,
+ enum_ad_to_slot_map);
+static SOC_ENUM_SINGLE_DECL(soc_enum_adslot15map,
+ AB8500_ADSLOTSEL8, AB8500_ADSLOTSELX_ODD_SHIFT,
+ enum_ad_to_slot_map);
+static SOC_ENUM_SINGLE_DECL(soc_enum_adslot16map,
+ AB8500_ADSLOTSEL9, AB8500_ADSLOTSELX_EVEN_SHIFT,
+ enum_ad_to_slot_map);
+static SOC_ENUM_SINGLE_DECL(soc_enum_adslot17map,
+ AB8500_ADSLOTSEL9, AB8500_ADSLOTSELX_ODD_SHIFT,
+ enum_ad_to_slot_map);
+static SOC_ENUM_SINGLE_DECL(soc_enum_adslot18map,
+ AB8500_ADSLOTSEL10, AB8500_ADSLOTSELX_EVEN_SHIFT,
+ enum_ad_to_slot_map);
+static SOC_ENUM_SINGLE_DECL(soc_enum_adslot19map,
+ AB8500_ADSLOTSEL10, AB8500_ADSLOTSELX_ODD_SHIFT,
+ enum_ad_to_slot_map);
+static SOC_ENUM_SINGLE_DECL(soc_enum_adslot20map,
+ AB8500_ADSLOTSEL11, AB8500_ADSLOTSELX_EVEN_SHIFT,
+ enum_ad_to_slot_map);
+static SOC_ENUM_SINGLE_DECL(soc_enum_adslot21map,
+ AB8500_ADSLOTSEL11, AB8500_ADSLOTSELX_ODD_SHIFT,
+ enum_ad_to_slot_map);
+static SOC_ENUM_SINGLE_DECL(soc_enum_adslot22map,
+ AB8500_ADSLOTSEL12, AB8500_ADSLOTSELX_EVEN_SHIFT,
+ enum_ad_to_slot_map);
+static SOC_ENUM_SINGLE_DECL(soc_enum_adslot23map,
+ AB8500_ADSLOTSEL12, AB8500_ADSLOTSELX_ODD_SHIFT,
+ enum_ad_to_slot_map);
+static SOC_ENUM_SINGLE_DECL(soc_enum_adslot24map,
+ AB8500_ADSLOTSEL13, AB8500_ADSLOTSELX_EVEN_SHIFT,
+ enum_ad_to_slot_map);
+static SOC_ENUM_SINGLE_DECL(soc_enum_adslot25map,
+ AB8500_ADSLOTSEL13, AB8500_ADSLOTSELX_ODD_SHIFT,
+ enum_ad_to_slot_map);
+static SOC_ENUM_SINGLE_DECL(soc_enum_adslot26map,
+ AB8500_ADSLOTSEL14, AB8500_ADSLOTSELX_EVEN_SHIFT,
+ enum_ad_to_slot_map);
+static SOC_ENUM_SINGLE_DECL(soc_enum_adslot27map,
+ AB8500_ADSLOTSEL14, AB8500_ADSLOTSELX_ODD_SHIFT,
+ enum_ad_to_slot_map);
+static SOC_ENUM_SINGLE_DECL(soc_enum_adslot28map,
+ AB8500_ADSLOTSEL15, AB8500_ADSLOTSELX_EVEN_SHIFT,
+ enum_ad_to_slot_map);
+static SOC_ENUM_SINGLE_DECL(soc_enum_adslot29map,
+ AB8500_ADSLOTSEL15, AB8500_ADSLOTSELX_ODD_SHIFT,
+ enum_ad_to_slot_map);
+static SOC_ENUM_SINGLE_DECL(soc_enum_adslot30map,
+ AB8500_ADSLOTSEL16, AB8500_ADSLOTSELX_EVEN_SHIFT,
+ enum_ad_to_slot_map);
+static SOC_ENUM_SINGLE_DECL(soc_enum_adslot31map,
+ AB8500_ADSLOTSEL16, AB8500_ADSLOTSELX_ODD_SHIFT,
+ enum_ad_to_slot_map);
+
+/* Digital interface - Burst mode */
+static const char * const enum_mask[] = {"Unmasked", "Masked"};
+static SOC_ENUM_SINGLE_DECL(soc_enum_bfifomask,
+ AB8500_FIFOCONF1, AB8500_FIFOCONF1_BFIFOMASK,
+ enum_mask);
+static const char * const enum_bitclk0[] = {"19_2_MHz", "38_4_MHz"};
+static SOC_ENUM_SINGLE_DECL(soc_enum_bfifo19m2,
+ AB8500_FIFOCONF1, AB8500_FIFOCONF1_BFIFO19M2,
+ enum_bitclk0);
+static const char * const enum_slavemaster[] = {"Slave", "Master"};
+static SOC_ENUM_SINGLE_DECL(soc_enum_bfifomast,
+ AB8500_FIFOCONF3, AB8500_FIFOCONF3_BFIFOMAST_SHIFT,
+ enum_slavemaster);
+
+/* Sidetone */
+static SOC_ENUM_SINGLE_EXT_DECL(soc_enum_sidstate, enum_sid_state);
+
+/* ANC */
+static SOC_ENUM_SINGLE_EXT_DECL(soc_enum_ancstate, enum_anc_state);
+
+static struct snd_kcontrol_new ab8500_ctrls[] = {
+ /* Charge pump */
+ SOC_ENUM("Charge Pump High Threshold For Low Voltage",
+ soc_enum_envdeththre),
+ SOC_ENUM("Charge Pump Low Threshold For Low Voltage",
+ soc_enum_envdetlthre),
+ SOC_SINGLE("Charge Pump Envelope Detection Switch",
+ AB8500_SIGENVCONF, AB8500_SIGENVCONF_ENVDETCPEN,
+ 1, 0),
+ SOC_ENUM("Charge Pump Envelope Detection Decay Time",
+ soc_enum_envdettime),
+
+ /* Headset */
+ SOC_ENUM("Headset Mode", soc_enum_da12voice),
+ SOC_SINGLE("Headset High Pass Switch",
+ AB8500_ANACONF1, AB8500_ANACONF1_HSHPEN,
+ 1, 0),
+ SOC_SINGLE("Headset Low Power Switch",
+ AB8500_ANACONF1, AB8500_ANACONF1_HSLOWPOW,
+ 1, 0),
+ SOC_SINGLE("Headset DAC Low Power Switch",
+ AB8500_ANACONF1, AB8500_ANACONF1_DACLOWPOW1,
+ 1, 0),
+ SOC_SINGLE("Headset DAC Drv Low Power Switch",
+ AB8500_ANACONF1, AB8500_ANACONF1_DACLOWPOW0,
+ 1, 0),
+ SOC_ENUM("Headset Fade Speed", soc_enum_hsfadspeed),
+ SOC_ENUM("Headset Source", soc_enum_da2hslr),
+ SOC_ENUM("Headset Filter", soc_enum_hsesinc),
+ SOC_DOUBLE_R_TLV("Headset Master Volume",
+ AB8500_DADIGGAIN1, AB8500_DADIGGAIN2,
+ 0, AB8500_DADIGGAINX_DAXGAIN_MAX, 1, dax_dig_gain_tlv),
+ SOC_DOUBLE_R_TLV("Headset Digital Volume",
+ AB8500_HSLEARDIGGAIN, AB8500_HSRDIGGAIN,
+ 0, AB8500_HSLEARDIGGAIN_HSLDGAIN_MAX, 1, hs_ear_dig_gain_tlv),
+ SOC_DOUBLE_TLV("Headset Volume",
+ AB8500_ANAGAIN3,
+ AB8500_ANAGAIN3_HSLGAIN, AB8500_ANAGAIN3_HSRGAIN,
+ AB8500_ANAGAIN3_HSXGAIN_MAX, 1, hs_gain_tlv),
+
+ /* Earpiece */
+ SOC_ENUM("Earpiece DAC Mode",
+ soc_enum_eardaclowpow),
+ SOC_ENUM("Earpiece DAC Drv Mode",
+ soc_enum_eardrvlowpow),
+
+ /* HandsFree */
+ SOC_ENUM("HF Mode", soc_enum_da34voice),
+ SOC_SINGLE("HF and Headset Swap Switch",
+ AB8500_DASLOTCONF1, AB8500_DASLOTCONF1_SWAPDA12_34,
+ 1, 0),
+ SOC_DOUBLE("HF Low EMI Mode Switch",
+ AB8500_CLASSDCONF1,
+ AB8500_CLASSDCONF1_HFLSWAPEN, AB8500_CLASSDCONF1_HFRSWAPEN,
+ 1, 0),
+ SOC_DOUBLE("HF FIR Bypass Switch",
+ AB8500_CLASSDCONF2,
+ AB8500_CLASSDCONF2_FIRBYP0, AB8500_CLASSDCONF2_FIRBYP1,
+ 1, 0),
+ SOC_DOUBLE("HF High Volume Switch",
+ AB8500_CLASSDCONF2,
+ AB8500_CLASSDCONF2_HIGHVOLEN0, AB8500_CLASSDCONF2_HIGHVOLEN1,
+ 1, 0),
+ SOC_SINGLE("HF L and R Bridge Switch",
+ AB8500_CLASSDCONF1, AB8500_CLASSDCONF1_PARLHF,
+ 1, 0),
+ SOC_DOUBLE_R_TLV("HF Master Volume",
+ AB8500_DADIGGAIN3, AB8500_DADIGGAIN4,
+ 0, AB8500_DADIGGAINX_DAXGAIN_MAX, 1, dax_dig_gain_tlv),
+
+ /* Vibra */
+ SOC_DOUBLE("Vibra High Volume Switch",
+ AB8500_CLASSDCONF2,
+ AB8500_CLASSDCONF2_HIGHVOLEN2, AB8500_CLASSDCONF2_HIGHVOLEN3,
+ 1, 0),
+ SOC_DOUBLE("Vibra Low EMI Mode Switch",
+ AB8500_CLASSDCONF1,
+ AB8500_CLASSDCONF1_VIB1SWAPEN, AB8500_CLASSDCONF1_VIB2SWAPEN,
+ 1, 0),
+ SOC_DOUBLE("Vibra FIR Bypass Switch",
+ AB8500_CLASSDCONF2,
+ AB8500_CLASSDCONF2_FIRBYP2, AB8500_CLASSDCONF2_FIRBYP3,
+ 1, 0),
+ SOC_ENUM("Vibra Mode", soc_enum_da56voice),
+ SOC_DOUBLE_R("Vibra PWM Duty Cycle N",
+ AB8500_PWMGENCONF3, AB8500_PWMGENCONF5,
+ AB8500_PWMGENCONFX_PWMVIBXDUTCYC,
+ AB8500_PWMGENCONFX_PWMVIBXDUTCYC_MAX, 0),
+ SOC_DOUBLE_R("Vibra PWM Duty Cycle P",
+ AB8500_PWMGENCONF2, AB8500_PWMGENCONF4,
+ AB8500_PWMGENCONFX_PWMVIBXDUTCYC,
+ AB8500_PWMGENCONFX_PWMVIBXDUTCYC_MAX, 0),
+ SOC_SINGLE("Vibra 1 and 2 Bridge Switch",
+ AB8500_CLASSDCONF1, AB8500_CLASSDCONF1_PARLVIB,
+ 1, 0),
+ SOC_DOUBLE_R_TLV("Vibra Master Volume",
+ AB8500_DADIGGAIN5, AB8500_DADIGGAIN6,
+ 0, AB8500_DADIGGAINX_DAXGAIN_MAX, 1, dax_dig_gain_tlv),
+
+ /* HandsFree, Vibra */
+ SOC_SINGLE("ClassD High Pass Volume",
+ AB8500_CLASSDCONF3, AB8500_CLASSDCONF3_DITHHPGAIN,
+ AB8500_CLASSDCONF3_DITHHPGAIN_MAX, 0),
+ SOC_SINGLE("ClassD White Volume",
+ AB8500_CLASSDCONF3, AB8500_CLASSDCONF3_DITHWGAIN,
+ AB8500_CLASSDCONF3_DITHWGAIN_MAX, 0),
+
+ /* Mic 1, Mic 2, LineIn */
+ SOC_DOUBLE_R_TLV("Mic Master Volume",
+ AB8500_ADDIGGAIN3, AB8500_ADDIGGAIN4,
+ 0, AB8500_ADDIGGAINX_ADXGAIN_MAX, 1, adx_dig_gain_tlv),
+
+ /* Mic 1 */
+ SOC_SINGLE_TLV("Mic 1",
+ AB8500_ANAGAIN1,
+ AB8500_ANAGAINX_MICXGAIN,
+ AB8500_ANAGAINX_MICXGAIN_MAX, 0, mic_gain_tlv),
+ SOC_SINGLE("Mic 1 Low Power Switch",
+ AB8500_ANAGAIN1, AB8500_ANAGAINX_LOWPOWMICX,
+ 1, 0),
+
+ /* Mic 2 */
+ SOC_DOUBLE("Mic High Pass Switch",
+ AB8500_ADFILTCONF,
+ AB8500_ADFILTCONF_AD3NH, AB8500_ADFILTCONF_AD4NH,
+ 1, 1),
+ SOC_ENUM("Mic Mode", soc_enum_ad34voice),
+ SOC_ENUM("Mic Filter", soc_enum_dmic34sinc),
+ SOC_SINGLE_TLV("Mic 2",
+ AB8500_ANAGAIN2,
+ AB8500_ANAGAINX_MICXGAIN,
+ AB8500_ANAGAINX_MICXGAIN_MAX, 0, mic_gain_tlv),
+ SOC_SINGLE("Mic 2 Low Power Switch",
+ AB8500_ANAGAIN2, AB8500_ANAGAINX_LOWPOWMICX,
+ 1, 0),
+
+ /* LineIn */
+ SOC_DOUBLE("LineIn High Pass Switch",
+ AB8500_ADFILTCONF,
+ AB8500_ADFILTCONF_AD1NH, AB8500_ADFILTCONF_AD2NH,
+ 1, 1),
+ SOC_ENUM("LineIn Filter", soc_enum_dmic12sinc),
+ SOC_ENUM("LineIn Mode", soc_enum_ad12voice),
+ SOC_DOUBLE_R_TLV("LineIn Master Volume",
+ AB8500_ADDIGGAIN1, AB8500_ADDIGGAIN2,
+ 0, AB8500_ADDIGGAINX_ADXGAIN_MAX, 1, adx_dig_gain_tlv),
+ SOC_DOUBLE_TLV("LineIn",
+ AB8500_ANAGAIN4,
+ AB8500_ANAGAIN4_LINLGAIN, AB8500_ANAGAIN4_LINRGAIN,
+ AB8500_ANAGAIN4_LINXGAIN_MAX, 0, lin_gain_tlv),
+ SOC_DOUBLE_R_TLV("LineIn to Headset Volume",
+ AB8500_DIGLINHSLGAIN, AB8500_DIGLINHSRGAIN,
+ AB8500_DIGLINHSXGAIN_LINTOHSXGAIN,
+ AB8500_DIGLINHSXGAIN_LINTOHSXGAIN_MAX,
+ 1, lin2hs_gain_tlv),
+
+ /* DMic */
+ SOC_ENUM("DMic Filter", soc_enum_dmic56sinc),
+ SOC_DOUBLE_R_TLV("DMic Master Volume",
+ AB8500_ADDIGGAIN5, AB8500_ADDIGGAIN6,
+ 0, AB8500_ADDIGGAINX_ADXGAIN_MAX, 1, adx_dig_gain_tlv),
+
+ /* Digital gains */
+ SOC_ENUM("Digital Gain Fade Speed", soc_enum_fadespeed),
+
+ /* Analog loopback */
+ SOC_DOUBLE_R_TLV("Analog Loopback Volume",
+ AB8500_ADDIGLOOPGAIN1, AB8500_ADDIGLOOPGAIN2,
+ 0, AB8500_ADDIGLOOPGAINX_ADXLBGAIN_MAX, 1, dax_dig_gain_tlv),
+
+ /* Digital interface - DA from slot mapping */
+ SOC_ENUM("Digital Interface DA 1 From Slot Map", soc_enum_da1slotmap),
+ SOC_ENUM("Digital Interface DA 2 From Slot Map", soc_enum_da2slotmap),
+ SOC_ENUM("Digital Interface DA 3 From Slot Map", soc_enum_da3slotmap),
+ SOC_ENUM("Digital Interface DA 4 From Slot Map", soc_enum_da4slotmap),
+ SOC_ENUM("Digital Interface DA 5 From Slot Map", soc_enum_da5slotmap),
+ SOC_ENUM("Digital Interface DA 6 From Slot Map", soc_enum_da6slotmap),
+ SOC_ENUM("Digital Interface DA 7 From Slot Map", soc_enum_da7slotmap),
+ SOC_ENUM("Digital Interface DA 8 From Slot Map", soc_enum_da8slotmap),
+
+ /* Digital interface - AD to slot mapping */
+ SOC_ENUM("Digital Interface AD To Slot 0 Map", soc_enum_adslot0map),
+ SOC_ENUM("Digital Interface AD To Slot 1 Map", soc_enum_adslot1map),
+ SOC_ENUM("Digital Interface AD To Slot 2 Map", soc_enum_adslot2map),
+ SOC_ENUM("Digital Interface AD To Slot 3 Map", soc_enum_adslot3map),
+ SOC_ENUM("Digital Interface AD To Slot 4 Map", soc_enum_adslot4map),
+ SOC_ENUM("Digital Interface AD To Slot 5 Map", soc_enum_adslot5map),
+ SOC_ENUM("Digital Interface AD To Slot 6 Map", soc_enum_adslot6map),
+ SOC_ENUM("Digital Interface AD To Slot 7 Map", soc_enum_adslot7map),
+ SOC_ENUM("Digital Interface AD To Slot 8 Map", soc_enum_adslot8map),
+ SOC_ENUM("Digital Interface AD To Slot 9 Map", soc_enum_adslot9map),
+ SOC_ENUM("Digital Interface AD To Slot 10 Map", soc_enum_adslot10map),
+ SOC_ENUM("Digital Interface AD To Slot 11 Map", soc_enum_adslot11map),
+ SOC_ENUM("Digital Interface AD To Slot 12 Map", soc_enum_adslot12map),
+ SOC_ENUM("Digital Interface AD To Slot 13 Map", soc_enum_adslot13map),
+ SOC_ENUM("Digital Interface AD To Slot 14 Map", soc_enum_adslot14map),
+ SOC_ENUM("Digital Interface AD To Slot 15 Map", soc_enum_adslot15map),
+ SOC_ENUM("Digital Interface AD To Slot 16 Map", soc_enum_adslot16map),
+ SOC_ENUM("Digital Interface AD To Slot 17 Map", soc_enum_adslot17map),
+ SOC_ENUM("Digital Interface AD To Slot 18 Map", soc_enum_adslot18map),
+ SOC_ENUM("Digital Interface AD To Slot 19 Map", soc_enum_adslot19map),
+ SOC_ENUM("Digital Interface AD To Slot 20 Map", soc_enum_adslot20map),
+ SOC_ENUM("Digital Interface AD To Slot 21 Map", soc_enum_adslot21map),
+ SOC_ENUM("Digital Interface AD To Slot 22 Map", soc_enum_adslot22map),
+ SOC_ENUM("Digital Interface AD To Slot 23 Map", soc_enum_adslot23map),
+ SOC_ENUM("Digital Interface AD To Slot 24 Map", soc_enum_adslot24map),
+ SOC_ENUM("Digital Interface AD To Slot 25 Map", soc_enum_adslot25map),
+ SOC_ENUM("Digital Interface AD To Slot 26 Map", soc_enum_adslot26map),
+ SOC_ENUM("Digital Interface AD To Slot 27 Map", soc_enum_adslot27map),
+ SOC_ENUM("Digital Interface AD To Slot 28 Map", soc_enum_adslot28map),
+ SOC_ENUM("Digital Interface AD To Slot 29 Map", soc_enum_adslot29map),
+ SOC_ENUM("Digital Interface AD To Slot 30 Map", soc_enum_adslot30map),
+ SOC_ENUM("Digital Interface AD To Slot 31 Map", soc_enum_adslot31map),
+
+ /* Digital interface - Loopback */
+ SOC_SINGLE("Digital Interface AD 1 Loopback Switch",
+ AB8500_DASLOTCONF1, AB8500_DASLOTCONF1_DAI7TOADO1,
+ 1, 0),
+ SOC_SINGLE("Digital Interface AD 2 Loopback Switch",
+ AB8500_DASLOTCONF2, AB8500_DASLOTCONF2_DAI8TOADO2,
+ 1, 0),
+ SOC_SINGLE("Digital Interface AD 3 Loopback Switch",
+ AB8500_DASLOTCONF3, AB8500_DASLOTCONF3_DAI7TOADO3,
+ 1, 0),
+ SOC_SINGLE("Digital Interface AD 4 Loopback Switch",
+ AB8500_DASLOTCONF4, AB8500_DASLOTCONF4_DAI8TOADO4,
+ 1, 0),
+ SOC_SINGLE("Digital Interface AD 5 Loopback Switch",
+ AB8500_DASLOTCONF5, AB8500_DASLOTCONF5_DAI7TOADO5,
+ 1, 0),
+ SOC_SINGLE("Digital Interface AD 6 Loopback Switch",
+ AB8500_DASLOTCONF6, AB8500_DASLOTCONF6_DAI8TOADO6,
+ 1, 0),
+ SOC_SINGLE("Digital Interface AD 7 Loopback Switch",
+ AB8500_DASLOTCONF7, AB8500_DASLOTCONF7_DAI8TOADO7,
+ 1, 0),
+ SOC_SINGLE("Digital Interface AD 8 Loopback Switch",
+ AB8500_DASLOTCONF8, AB8500_DASLOTCONF8_DAI7TOADO8,
+ 1, 0),
+
+ /* Digital interface - Burst FIFO */
+ SOC_SINGLE("Digital Interface 0 FIFO Enable Switch",
+ AB8500_DIGIFCONF3, AB8500_DIGIFCONF3_IF0BFIFOEN,
+ 1, 0),
+ SOC_ENUM("Burst FIFO Mask", soc_enum_bfifomask),
+ SOC_ENUM("Burst FIFO Bit-clock Frequency", soc_enum_bfifo19m2),
+ SOC_SINGLE("Burst FIFO Threshold",
+ AB8500_FIFOCONF1, AB8500_FIFOCONF1_BFIFOINT_SHIFT,
+ AB8500_FIFOCONF1_BFIFOINT_MAX, 0),
+ SOC_SINGLE("Burst FIFO Length",
+ AB8500_FIFOCONF2, AB8500_FIFOCONF2_BFIFOTX_SHIFT,
+ AB8500_FIFOCONF2_BFIFOTX_MAX, 0),
+ SOC_SINGLE("Burst FIFO EOS Extra Slots",
+ AB8500_FIFOCONF3, AB8500_FIFOCONF3_BFIFOEXSL_SHIFT,
+ AB8500_FIFOCONF3_BFIFOEXSL_MAX, 0),
+ SOC_SINGLE("Burst FIFO FS Extra Bit-clocks",
+ AB8500_FIFOCONF3, AB8500_FIFOCONF3_PREBITCLK0_SHIFT,
+ AB8500_FIFOCONF3_PREBITCLK0_MAX, 0),
+ SOC_ENUM("Burst FIFO Interface Mode", soc_enum_bfifomast),
+
+ SOC_SINGLE("Burst FIFO Interface Switch",
+ AB8500_FIFOCONF3, AB8500_FIFOCONF3_BFIFORUN_SHIFT,
+ 1, 0),
+ SOC_SINGLE("Burst FIFO Switch Frame Number",
+ AB8500_FIFOCONF4, AB8500_FIFOCONF4_BFIFOFRAMSW_SHIFT,
+ AB8500_FIFOCONF4_BFIFOFRAMSW_MAX, 0),
+ SOC_SINGLE("Burst FIFO Wake Up Delay",
+ AB8500_FIFOCONF5, AB8500_FIFOCONF5_BFIFOWAKEUP_SHIFT,
+ AB8500_FIFOCONF5_BFIFOWAKEUP_MAX, 0),
+ SOC_SINGLE("Burst FIFO Samples In FIFO",
+ AB8500_FIFOCONF6, AB8500_FIFOCONF6_BFIFOSAMPLE_SHIFT,
+ AB8500_FIFOCONF6_BFIFOSAMPLE_MAX, 0),
+
+ /* ANC */
+ SOC_ENUM_EXT("ANC Status", soc_enum_ancstate,
+ anc_status_control_get, anc_status_control_put),
+ SOC_SINGLE_XR_SX("ANC Warp Delay Shift",
+ AB8500_ANCCONF2, 1, AB8500_ANCCONF2_SHIFT,
+ AB8500_ANCCONF2_MIN, AB8500_ANCCONF2_MAX, 0),
+ SOC_SINGLE_XR_SX("ANC FIR Output Shift",
+ AB8500_ANCCONF3, 1, AB8500_ANCCONF3_SHIFT,
+ AB8500_ANCCONF3_MIN, AB8500_ANCCONF3_MAX, 0),
+ SOC_SINGLE_XR_SX("ANC IIR Output Shift",
+ AB8500_ANCCONF4, 1, AB8500_ANCCONF4_SHIFT,
+ AB8500_ANCCONF4_MIN, AB8500_ANCCONF4_MAX, 0),
+ SOC_SINGLE_XR_SX("ANC Warp Delay",
+ AB8500_ANCCONF9, 2, AB8500_ANC_WARP_DELAY_SHIFT,
+ AB8500_ANC_WARP_DELAY_MIN, AB8500_ANC_WARP_DELAY_MAX, 0),
+
+ /* Sidetone */
+ SOC_ENUM_EXT("Sidetone Status", soc_enum_sidstate,
+ sid_status_control_get, sid_status_control_put),
+ SOC_SINGLE_STROBE("Sidetone Reset",
+ AB8500_SIDFIRADR, AB8500_SIDFIRADR_FIRSIDSET, 0),
+};
+
+static struct snd_kcontrol_new ab8500_filter_controls[] = {
+ AB8500_FILTER_CONTROL("ANC FIR Coefficients", AB8500_ANC_FIR_COEFFS,
+ AB8500_ANC_FIR_COEFF_MIN, AB8500_ANC_FIR_COEFF_MAX),
+ AB8500_FILTER_CONTROL("ANC IIR Coefficients", AB8500_ANC_IIR_COEFFS,
+ AB8500_ANC_IIR_COEFF_MIN, AB8500_ANC_IIR_COEFF_MAX),
+ AB8500_FILTER_CONTROL("Sidetone FIR Coefficients",
+ AB8500_SID_FIR_COEFFS, AB8500_SID_FIR_COEFF_MIN,
+ AB8500_SID_FIR_COEFF_MAX)
+};
+enum ab8500_filter {
+ AB8500_FILTER_ANC_FIR = 0,
+ AB8500_FILTER_ANC_IIR = 1,
+ AB8500_FILTER_SID_FIR = 2,
+};
+
+/*
+ * Extended interface for codec-driver
+ */
+
+static int ab8500_audio_init_audioblock(struct snd_soc_codec *codec)
+{
+ int status;
+
+ dev_dbg(codec->dev, "%s: Enter.\n", __func__);
+
+ /* Reset audio-registers and disable 32kHz-clock output 2 */
+ status = ab8500_sysctrl_write(AB8500_STW4500CTRL3,
+ AB8500_STW4500CTRL3_CLK32KOUT2DIS |
+ AB8500_STW4500CTRL3_RESETAUDN,
+ AB8500_STW4500CTRL3_RESETAUDN);
+ if (status < 0)
+ return status;
+
+ return 0;
+}
+
+static int ab8500_audio_setup_mics(struct snd_soc_codec *codec,
+ struct amic_settings *amics)
+{
+ u8 value8;
+ unsigned int value;
+ int status;
+ const struct snd_soc_dapm_route *route;
+
+ 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,
+ &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,
+ value);
+ if (status < 0)
+ return status;
+
+ /* Attach regulators to AMic DAPM-paths */
+ dev_dbg(codec->dev, "%s: Mic 1a regulator: %s\n", __func__,
+ amic_micbias_str(amics->mic1a_micbias));
+ route = &ab8500_dapm_routes_mic1a_vamicx[amics->mic1a_micbias];
+ status = snd_soc_dapm_add_routes(&codec->dapm, route, 1);
+ dev_dbg(codec->dev, "%s: Mic 1b regulator: %s\n", __func__,
+ amic_micbias_str(amics->mic1b_micbias));
+ route = &ab8500_dapm_routes_mic1b_vamicx[amics->mic1b_micbias];
+ status |= snd_soc_dapm_add_routes(&codec->dapm, route, 1);
+ dev_dbg(codec->dev, "%s: Mic 2 regulator: %s\n", __func__,
+ amic_micbias_str(amics->mic2_micbias));
+ route = &ab8500_dapm_routes_mic2_vamicx[amics->mic2_micbias];
+ status |= snd_soc_dapm_add_routes(&codec->dapm, route, 1);
+ if (status < 0) {
+ dev_err(codec->dev,
+ "%s: Failed to add AMic-regulator DAPM-routes (%d).\n",
+ __func__, status);
+ return status;
+ }
+
+ /* Set AMic-configuration */
+ dev_dbg(codec->dev, "%s: Mic 1 mic-type: %s\n", __func__,
+ amic_type_str(amics->mic1_type));
+ snd_soc_update_bits(codec, AB8500_ANAGAIN1, AB8500_ANAGAINX_ENSEMICX,
+ amics->mic1_type == AMIC_TYPE_DIFFERENTIAL ?
+ 0 : AB8500_ANAGAINX_ENSEMICX);
+ dev_dbg(codec->dev, "%s: Mic 2 mic-type: %s\n", __func__,
+ amic_type_str(amics->mic2_type));
+ snd_soc_update_bits(codec, AB8500_ANAGAIN2, AB8500_ANAGAINX_ENSEMICX,
+ amics->mic2_type == AMIC_TYPE_DIFFERENTIAL ?
+ 0 : AB8500_ANAGAINX_ENSEMICX);
+
+ return 0;
+}
+EXPORT_SYMBOL_GPL(ab8500_audio_setup_mics);
+
+static int ab8500_audio_set_ear_cmv(struct snd_soc_codec *codec,
+ enum ear_cm_voltage ear_cmv)
+{
+ char *cmv_str;
+
+ switch (ear_cmv) {
+ case EAR_CMV_0_95V:
+ cmv_str = "0.95V";
+ break;
+ case EAR_CMV_1_10V:
+ cmv_str = "1.10V";
+ break;
+ case EAR_CMV_1_27V:
+ cmv_str = "1.27V";
+ break;
+ case EAR_CMV_1_58V:
+ cmv_str = "1.58V";
+ break;
+ default:
+ dev_err(codec->dev,
+ "%s: Unknown earpiece CM-voltage (%d)!\n",
+ __func__, (int)ear_cmv);
+ return -EINVAL;
+ }
+ dev_dbg(codec->dev, "%s: Earpiece CM-voltage: %s\n", __func__,
+ cmv_str);
+ snd_soc_update_bits(codec, AB8500_ANACONF1, AB8500_ANACONF1_EARSELCM,
+ ear_cmv);
+
+ return 0;
+}
+EXPORT_SYMBOL_GPL(ab8500_audio_set_ear_cmv);
+
+static int ab8500_audio_set_bit_delay(struct snd_soc_dai *dai,
+ unsigned int delay)
+{
+ unsigned int mask, val;
+ struct snd_soc_codec *codec = dai->codec;
+
+ mask = BIT(AB8500_DIGIFCONF2_IF0DEL);
+ val = 0;
+
+ switch (delay) {
+ case 0:
+ break;
+ case 1:
+ val |= BIT(AB8500_DIGIFCONF2_IF0DEL);
+ break;
+ default:
+ dev_err(dai->codec->dev,
+ "%s: ERROR: Unsupported bit-delay (0x%x)!\n",
+ __func__, delay);
+ return -EINVAL;
+ }
+
+ dev_dbg(dai->codec->dev, "%s: IF0 Bit-delay: %d bits.\n",
+ __func__, delay);
+ snd_soc_update_bits(codec, AB8500_DIGIFCONF2, mask, val);
+
+ return 0;
+}
+
+/* Gates clocking according format mask */
+static int ab8500_codec_set_dai_clock_gate(struct snd_soc_codec *codec,
+ unsigned int fmt)
+{
+ unsigned int mask;
+ unsigned int val;
+
+ mask = BIT(AB8500_DIGIFCONF1_ENMASTGEN) |
+ BIT(AB8500_DIGIFCONF1_ENFSBITCLK0);
+
+ val = BIT(AB8500_DIGIFCONF1_ENMASTGEN);
+
+ switch (fmt & SND_SOC_DAIFMT_CLOCK_MASK) {
+ case SND_SOC_DAIFMT_CONT: /* continuous clock */
+ dev_dbg(codec->dev, "%s: IF0 Clock is continuous.\n",
+ __func__);
+ val |= BIT(AB8500_DIGIFCONF1_ENFSBITCLK0);
+ break;
+ case SND_SOC_DAIFMT_GATED: /* clock is gated */
+ dev_dbg(codec->dev, "%s: IF0 Clock is gated.\n",
+ __func__);
+ break;
+ default:
+ dev_err(codec->dev,
+ "%s: ERROR: Unsupported clock mask (0x%x)!\n",
+ __func__, fmt & SND_SOC_DAIFMT_CLOCK_MASK);
+ return -EINVAL;
+ }
+
+ snd_soc_update_bits(codec, AB8500_DIGIFCONF1, mask, val);
+
+ return 0;
+}
+
+static int ab8500_codec_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt)
+{
+ unsigned int mask;
+ unsigned int val;
+ struct snd_soc_codec *codec = dai->codec;
+ int status;
+
+ dev_dbg(codec->dev, "%s: Enter (fmt = 0x%x)\n", __func__, fmt);
+
+ mask = BIT(AB8500_DIGIFCONF3_IF1DATOIF0AD) |
+ BIT(AB8500_DIGIFCONF3_IF1CLKTOIF0CLK) |
+ BIT(AB8500_DIGIFCONF3_IF0BFIFOEN) |
+ BIT(AB8500_DIGIFCONF3_IF0MASTER);
+ val = 0;
+
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBM_CFM: /* codec clk & FRM master */
+ dev_dbg(dai->codec->dev,
+ "%s: IF0 Master-mode: AB8500 master.\n", __func__);
+ val |= BIT(AB8500_DIGIFCONF3_IF0MASTER);
+ break;
+ case SND_SOC_DAIFMT_CBS_CFS: /* codec clk & FRM slave */
+ dev_dbg(dai->codec->dev,
+ "%s: IF0 Master-mode: AB8500 slave.\n", __func__);
+ break;
+ case SND_SOC_DAIFMT_CBS_CFM: /* codec clk slave & FRM master */
+ case SND_SOC_DAIFMT_CBM_CFS: /* codec clk master & frame slave */
+ dev_err(dai->codec->dev,
+ "%s: ERROR: The device is either a master or a slave.\n",
+ __func__);
+ default:
+ dev_err(dai->codec->dev,
+ "%s: ERROR: Unsupporter master mask 0x%x\n",
+ __func__, fmt & SND_SOC_DAIFMT_MASTER_MASK);
+ return -EINVAL;
+ break;
+ }
+
+ snd_soc_update_bits(codec, AB8500_DIGIFCONF3, mask, val);
+
+ /* Set clock gating */
+ status = ab8500_codec_set_dai_clock_gate(codec, fmt);
+ if (status) {
+ dev_err(dai->codec->dev,
+ "%s: ERRROR: Failed to set clock gate (%d).\n",
+ __func__, status);
+ return status;
+ }
+
+ /* Setting data transfer format */
+
+ mask = BIT(AB8500_DIGIFCONF2_IF0FORMAT0) |
+ BIT(AB8500_DIGIFCONF2_IF0FORMAT1) |
+ BIT(AB8500_DIGIFCONF2_FSYNC0P) |
+ BIT(AB8500_DIGIFCONF2_BITCLK0P);
+ val = 0;
+
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S: /* I2S mode */
+ dev_dbg(dai->codec->dev, "%s: IF0 Protocol: I2S\n", __func__);
+ val |= BIT(AB8500_DIGIFCONF2_IF0FORMAT1);
+ ab8500_audio_set_bit_delay(dai, 0);
+ break;
+
+ case SND_SOC_DAIFMT_DSP_A: /* L data MSB after FRM LRC */
+ dev_dbg(dai->codec->dev,
+ "%s: IF0 Protocol: DSP A (TDM)\n", __func__);
+ val |= BIT(AB8500_DIGIFCONF2_IF0FORMAT0);
+ ab8500_audio_set_bit_delay(dai, 1);
+ break;
+
+ case SND_SOC_DAIFMT_DSP_B: /* L data MSB during FRM LRC */
+ dev_dbg(dai->codec->dev,
+ "%s: IF0 Protocol: DSP B (TDM)\n", __func__);
+ val |= BIT(AB8500_DIGIFCONF2_IF0FORMAT0);
+ ab8500_audio_set_bit_delay(dai, 0);
+ break;
+
+ default:
+ dev_err(dai->codec->dev,
+ "%s: ERROR: Unsupported format (0x%x)!\n",
+ __func__, fmt & SND_SOC_DAIFMT_FORMAT_MASK);
+ return -EINVAL;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_NB_NF: /* normal bit clock + frame */
+ dev_dbg(dai->codec->dev,
+ "%s: IF0: Normal bit clock, normal frame\n",
+ __func__);
+ break;
+ case SND_SOC_DAIFMT_NB_IF: /* normal BCLK + inv FRM */
+ dev_dbg(dai->codec->dev,
+ "%s: IF0: Normal bit clock, inverted frame\n",
+ __func__);
+ val |= BIT(AB8500_DIGIFCONF2_FSYNC0P);
+ break;
+ case SND_SOC_DAIFMT_IB_NF: /* invert BCLK + nor FRM */
+ dev_dbg(dai->codec->dev,
+ "%s: IF0: Inverted bit clock, normal frame\n",
+ __func__);
+ val |= BIT(AB8500_DIGIFCONF2_BITCLK0P);
+ break;
+ case SND_SOC_DAIFMT_IB_IF: /* invert BCLK + FRM */
+ dev_dbg(dai->codec->dev,
+ "%s: IF0: Inverted bit clock, inverted frame\n",
+ __func__);
+ val |= BIT(AB8500_DIGIFCONF2_FSYNC0P);
+ val |= BIT(AB8500_DIGIFCONF2_BITCLK0P);
+ break;
+ default:
+ dev_err(dai->codec->dev,
+ "%s: ERROR: Unsupported INV mask 0x%x\n",
+ __func__, fmt & SND_SOC_DAIFMT_INV_MASK);
+ return -EINVAL;
+ }
+
+ snd_soc_update_bits(codec, AB8500_DIGIFCONF2, mask, val);
+
+ return 0;
+}
+
+static int ab8500_codec_set_dai_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, mask, slots_active;
+
+ mask = BIT(AB8500_DIGIFCONF2_IF0WL0) |
+ BIT(AB8500_DIGIFCONF2_IF0WL1);
+ val = 0;
+
+ switch (slot_width) {
+ case 16:
+ break;
+ case 20:
+ val |= BIT(AB8500_DIGIFCONF2_IF0WL0);
+ break;
+ case 24:
+ val |= BIT(AB8500_DIGIFCONF2_IF0WL1);
+ break;
+ case 32:
+ val |= BIT(AB8500_DIGIFCONF2_IF0WL1) |
+ BIT(AB8500_DIGIFCONF2_IF0WL0);
+ break;
+ default:
+ dev_err(dai->codec->dev, "%s: Unsupported slot-width 0x%x\n",
+ __func__, slot_width);
+ return -EINVAL;
+ }
+
+ dev_dbg(dai->codec->dev, "%s: IF0 slot-width: %d bits.\n",
+ __func__, slot_width);
+ snd_soc_update_bits(codec, AB8500_DIGIFCONF2, mask, val);
+
+ /* Setup TDM clocking according to slot count */
+ dev_dbg(dai->codec->dev, "%s: Slots, total: %d\n", __func__, slots);
+ mask = BIT(AB8500_DIGIFCONF1_IF0BITCLKOS0) |
+ BIT(AB8500_DIGIFCONF1_IF0BITCLKOS1);
+ switch (slots) {
+ case 2:
+ val = AB8500_MASK_NONE;
+ break;
+ case 4:
+ val = BIT(AB8500_DIGIFCONF1_IF0BITCLKOS0);
+ break;
+ case 8:
+ val = BIT(AB8500_DIGIFCONF1_IF0BITCLKOS1);
+ break;
+ case 16:
+ val = BIT(AB8500_DIGIFCONF1_IF0BITCLKOS0) |
+ BIT(AB8500_DIGIFCONF1_IF0BITCLKOS1);
+ break;
+ default:
+ dev_err(dai->codec->dev,
+ "%s: ERROR: Unsupported number of slots (%d)!\n",
+ __func__, slots);
+ return -EINVAL;
+ }
+ snd_soc_update_bits(codec, AB8500_DIGIFCONF1, mask, val);
+
+ /* Setup TDM DA according to active tx slots */
+ mask = AB8500_DASLOTCONFX_SLTODAX_MASK;
+ slots_active = hweight32(tx_mask);
+ dev_dbg(dai->codec->dev, "%s: Slots, active, TX: %d\n", __func__,
+ slots_active);
+ switch (slots_active) {
+ case 0:
+ break;
+ case 1:
+ /* Slot 9 -> DA_IN1 & DA_IN3 */
+ snd_soc_update_bits(codec, AB8500_DASLOTCONF1, mask, 11);
+ snd_soc_update_bits(codec, AB8500_DASLOTCONF3, mask, 11);
+ snd_soc_update_bits(codec, AB8500_DASLOTCONF2, mask, 11);
+ snd_soc_update_bits(codec, AB8500_DASLOTCONF4, mask, 11);
+ break;
+ case 2:
+ /* Slot 9 -> DA_IN1 & DA_IN3, Slot 11 -> DA_IN2 & DA_IN4 */
+ snd_soc_update_bits(codec, AB8500_DASLOTCONF1, mask, 9);
+ snd_soc_update_bits(codec, AB8500_DASLOTCONF3, mask, 9);
+ snd_soc_update_bits(codec, AB8500_DASLOTCONF2, mask, 11);
+ snd_soc_update_bits(codec, AB8500_DASLOTCONF4, mask, 11);
+
+ break;
+ case 8:
+ dev_dbg(dai->codec->dev,
+ "%s: In 8-channel mode DA-from-slot mapping is set manually.",
+ __func__);
+ break;
+ default:
+ dev_err(dai->codec->dev,
+ "%s: Unsupported number of active TX-slots (%d)!\n",
+ __func__, slots_active);
+ return -EINVAL;
+ }
+
+ /* Setup TDM AD according to active RX-slots */
+ slots_active = hweight32(rx_mask);
+ dev_dbg(dai->codec->dev, "%s: Slots, active, RX: %d\n", __func__,
+ slots_active);
+ switch (slots_active) {
+ case 0:
+ break;
+ case 1:
+ /* AD_OUT3 -> slot 0 & 1 */
+ snd_soc_update_bits(codec, AB8500_ADSLOTSEL1, AB8500_MASK_ALL,
+ AB8500_ADSLOTSELX_AD_OUT3_TO_SLOT_EVEN |
+ AB8500_ADSLOTSELX_AD_OUT3_TO_SLOT_ODD);
+ break;
+ case 2:
+ /* AD_OUT3 -> slot 0, AD_OUT2 -> slot 1 */
+ snd_soc_update_bits(codec,
+ AB8500_ADSLOTSEL1,
+ AB8500_MASK_ALL,
+ AB8500_ADSLOTSELX_AD_OUT3_TO_SLOT_EVEN |
+ AB8500_ADSLOTSELX_AD_OUT2_TO_SLOT_ODD);
+ break;
+ case 8:
+ dev_dbg(dai->codec->dev,
+ "%s: In 8-channel mode AD-to-slot mapping is set manually.",
+ __func__);
+ break;
+ default:
+ dev_err(dai->codec->dev,
+ "%s: Unsupported number of active RX-slots (%d)!\n",
+ __func__, slots_active);
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+struct snd_soc_dai_driver ab8500_codec_dai[] = {
+ {
+ .name = "ab8500-codec-dai.0",
+ .id = 0,
+ .playback = {
+ .stream_name = "ab8500_0p",
+ .channels_min = 1,
+ .channels_max = 8,
+ .rates = AB8500_SUPPORTED_RATE,
+ .formats = AB8500_SUPPORTED_FMT,
+ },
+ .ops = (struct snd_soc_dai_ops[]) {
+ {
+ .set_tdm_slot = ab8500_codec_set_dai_tdm_slot,
+ .set_fmt = ab8500_codec_set_dai_fmt,
+ }
+ },
+ .symmetric_rates = 1
+ },
+ {
+ .name = "ab8500-codec-dai.1",
+ .id = 1,
+ .capture = {
+ .stream_name = "ab8500_0c",
+ .channels_min = 1,
+ .channels_max = 8,
+ .rates = AB8500_SUPPORTED_RATE,
+ .formats = AB8500_SUPPORTED_FMT,
+ },
+ .ops = (struct snd_soc_dai_ops[]) {
+ {
+ .set_tdm_slot = ab8500_codec_set_dai_tdm_slot,
+ .set_fmt = ab8500_codec_set_dai_fmt,
+ }
+ },
+ .symmetric_rates = 1
+ }
+};
+
+static int ab8500_codec_probe(struct snd_soc_codec *codec)
+{
+ struct device *dev = codec->dev;
+ struct ab8500_codec_drvdata *drvdata = dev_get_drvdata(dev);
+ struct ab8500_platform_data *pdata;
+ struct filter_control *fc;
+ int status;
+
+ dev_dbg(dev, "%s: Enter.\n", __func__);
+
+ /* Setup AB8500 according to board-settings */
+ pdata = (struct ab8500_platform_data *)dev_get_platdata(dev->parent);
+ status = ab8500_audio_setup_mics(codec, &pdata->codec->amics);
+ if (status < 0) {
+ pr_err("%s: Failed to setup mics (%d)!\n", __func__, status);
+ return status;
+ }
+ status = ab8500_audio_set_ear_cmv(codec, pdata->codec->ear_cmv);
+ if (status < 0) {
+ pr_err("%s: Failed to set earpiece CM-voltage (%d)!\n",
+ __func__, status);
+ return status;
+ }
+
+ status = ab8500_audio_init_audioblock(codec);
+ if (status < 0) {
+ dev_err(dev, "%s: failed to init audio-block (%d)!\n",
+ __func__, status);
+ return status;
+ }
+
+ /* Override HW-defaults */
+ ab8500_codec_write_reg(codec,
+ AB8500_ANACONF5,
+ BIT(AB8500_ANACONF5_HSAUTOEN));
+ ab8500_codec_write_reg(codec,
+ AB8500_SHORTCIRCONF,
+ BIT(AB8500_SHORTCIRCONF_HSZCDDIS));
+
+ /* Add filter controls */
+ status = snd_soc_add_codec_controls(codec, ab8500_filter_controls,
+ ARRAY_SIZE(ab8500_filter_controls));
+ if (status < 0) {
+ dev_err(dev,
+ "%s: failed to add ab8500 filter controls (%d).\n",
+ __func__, status);
+ return status;
+ }
+ fc = (struct filter_control *)
+ &ab8500_filter_controls[AB8500_FILTER_ANC_FIR].private_value;
+ drvdata->anc_fir_values = (long *)fc->value;
+ fc = (struct filter_control *)
+ &ab8500_filter_controls[AB8500_FILTER_ANC_IIR].private_value;
+ drvdata->anc_iir_values = (long *)fc->value;
+ fc = (struct filter_control *)
+ &ab8500_filter_controls[AB8500_FILTER_SID_FIR].private_value;
+ drvdata->sid_fir_values = (long *)fc->value;
+
+ (void)snd_soc_dapm_disable_pin(&codec->dapm, "ANC Configure Input");
+
+ mutex_init(&drvdata->anc_lock);
+
+ return status;
+}
+
+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,
+ .num_dapm_widgets = ARRAY_SIZE(ab8500_dapm_widgets),
+ .dapm_routes = ab8500_dapm_routes,
+ .num_dapm_routes = ARRAY_SIZE(ab8500_dapm_routes),
+};
+
+static int __devinit ab8500_codec_driver_probe(struct platform_device *pdev)
+{
+ int status;
+ struct ab8500_codec_drvdata *drvdata;
+
+ dev_dbg(&pdev->dev, "%s: Enter.\n", __func__);
+
+ /* Create driver private-data struct */
+ drvdata = devm_kzalloc(&pdev->dev, sizeof(struct ab8500_codec_drvdata),
+ GFP_KERNEL);
+ drvdata->sid_status = SID_UNCONFIGURED;
+ drvdata->anc_status = ANC_UNCONFIGURED;
+ dev_set_drvdata(&pdev->dev, drvdata);
+
+ dev_dbg(&pdev->dev, "%s: Register codec.\n", __func__);
+ status = snd_soc_register_codec(&pdev->dev, &ab8500_codec_driver,
+ ab8500_codec_dai,
+ ARRAY_SIZE(ab8500_codec_dai));
+ if (status < 0)
+ dev_err(&pdev->dev,
+ "%s: Error: Failed to register codec (%d).\n",
+ __func__, status);
+
+ return status;
+}
+
+static int __devexit ab8500_codec_driver_remove(struct platform_device *pdev)
+{
+ dev_info(&pdev->dev, "%s Enter.\n", __func__);
+
+ snd_soc_unregister_codec(&pdev->dev);
+
+ return 0;
+}
+
+static struct platform_driver ab8500_codec_platform_driver = {
+ .driver = {
+ .name = "ab8500-codec",
+ .owner = THIS_MODULE,
+ },
+ .probe = ab8500_codec_driver_probe,
+ .remove = __devexit_p(ab8500_codec_driver_remove),
+ .suspend = NULL,
+ .resume = NULL,
+};
+module_platform_driver(ab8500_codec_platform_driver);
+
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/codecs/ab8500-codec.h b/sound/soc/codecs/ab8500-codec.h
new file mode 100644
index 0000000..114f69a
--- /dev/null
+++ b/sound/soc/codecs/ab8500-codec.h
@@ -0,0 +1,590 @@
+/*
+ * Copyright (C) ST-Ericsson SA 2012
+ *
+ * Author: Ola Lilja <ola.o.lilja@stericsson.com>,
+ * Kristoffer Karlsson <kristoffer.karlsson@stericsson.com>,
+ * Roger Nilsson <roger.xr.nilsson@stericsson.com>,
+ * for ST-Ericsson.
+ *
+ * Based on the early work done by:
+ * Mikko J. Lehto <mikko.lehto@symbio.com>,
+ * Mikko Sarmanne <mikko.sarmanne@symbio.com>,
+ * for ST-Ericsson.
+ *
+ * License terms:
+ *
+ * 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 AB8500_CODEC_REGISTERS_H
+#define AB8500_CODEC_REGISTERS_H
+
+#define AB8500_SUPPORTED_RATE (SNDRV_PCM_RATE_48000)
+#define AB8500_SUPPORTED_FMT (SNDRV_PCM_FMTBIT_S16_LE)
+
+/* AB8500 audio bank (0x0d) register definitions */
+
+#define AB8500_POWERUP 0x00
+#define AB8500_AUDSWRESET 0x01
+#define AB8500_ADPATHENA 0x02
+#define AB8500_DAPATHENA 0x03
+#define AB8500_ANACONF1 0x04
+#define AB8500_ANACONF2 0x05
+#define AB8500_DIGMICCONF 0x06
+#define AB8500_ANACONF3 0x07
+#define AB8500_ANACONF4 0x08
+#define AB8500_DAPATHCONF 0x09
+#define AB8500_MUTECONF 0x0A
+#define AB8500_SHORTCIRCONF 0x0B
+#define AB8500_ANACONF5 0x0C
+#define AB8500_ENVCPCONF 0x0D
+#define AB8500_SIGENVCONF 0x0E
+#define AB8500_PWMGENCONF1 0x0F
+#define AB8500_PWMGENCONF2 0x10
+#define AB8500_PWMGENCONF3 0x11
+#define AB8500_PWMGENCONF4 0x12
+#define AB8500_PWMGENCONF5 0x13
+#define AB8500_ANAGAIN1 0x14
+#define AB8500_ANAGAIN2 0x15
+#define AB8500_ANAGAIN3 0x16
+#define AB8500_ANAGAIN4 0x17
+#define AB8500_DIGLINHSLGAIN 0x18
+#define AB8500_DIGLINHSRGAIN 0x19
+#define AB8500_ADFILTCONF 0x1A
+#define AB8500_DIGIFCONF1 0x1B
+#define AB8500_DIGIFCONF2 0x1C
+#define AB8500_DIGIFCONF3 0x1D
+#define AB8500_DIGIFCONF4 0x1E
+#define AB8500_ADSLOTSEL1 0x1F
+#define AB8500_ADSLOTSEL2 0x20
+#define AB8500_ADSLOTSEL3 0x21
+#define AB8500_ADSLOTSEL4 0x22
+#define AB8500_ADSLOTSEL5 0x23
+#define AB8500_ADSLOTSEL6 0x24
+#define AB8500_ADSLOTSEL7 0x25
+#define AB8500_ADSLOTSEL8 0x26
+#define AB8500_ADSLOTSEL9 0x27
+#define AB8500_ADSLOTSEL10 0x28
+#define AB8500_ADSLOTSEL11 0x29
+#define AB8500_ADSLOTSEL12 0x2A
+#define AB8500_ADSLOTSEL13 0x2B
+#define AB8500_ADSLOTSEL14 0x2C
+#define AB8500_ADSLOTSEL15 0x2D
+#define AB8500_ADSLOTSEL16 0x2E
+#define AB8500_ADSLOTHIZCTRL1 0x2F
+#define AB8500_ADSLOTHIZCTRL2 0x30
+#define AB8500_ADSLOTHIZCTRL3 0x31
+#define AB8500_ADSLOTHIZCTRL4 0x32
+#define AB8500_DASLOTCONF1 0x33
+#define AB8500_DASLOTCONF2 0x34
+#define AB8500_DASLOTCONF3 0x35
+#define AB8500_DASLOTCONF4 0x36
+#define AB8500_DASLOTCONF5 0x37
+#define AB8500_DASLOTCONF6 0x38
+#define AB8500_DASLOTCONF7 0x39
+#define AB8500_DASLOTCONF8 0x3A
+#define AB8500_CLASSDCONF1 0x3B
+#define AB8500_CLASSDCONF2 0x3C
+#define AB8500_CLASSDCONF3 0x3D
+#define AB8500_DMICFILTCONF 0x3E
+#define AB8500_DIGMULTCONF1 0x3F
+#define AB8500_DIGMULTCONF2 0x40
+#define AB8500_ADDIGGAIN1 0x41
+#define AB8500_ADDIGGAIN2 0x42
+#define AB8500_ADDIGGAIN3 0x43
+#define AB8500_ADDIGGAIN4 0x44
+#define AB8500_ADDIGGAIN5 0x45
+#define AB8500_ADDIGGAIN6 0x46
+#define AB8500_DADIGGAIN1 0x47
+#define AB8500_DADIGGAIN2 0x48
+#define AB8500_DADIGGAIN3 0x49
+#define AB8500_DADIGGAIN4 0x4A
+#define AB8500_DADIGGAIN5 0x4B
+#define AB8500_DADIGGAIN6 0x4C
+#define AB8500_ADDIGLOOPGAIN1 0x4D
+#define AB8500_ADDIGLOOPGAIN2 0x4E
+#define AB8500_HSLEARDIGGAIN 0x4F
+#define AB8500_HSRDIGGAIN 0x50
+#define AB8500_SIDFIRGAIN1 0x51
+#define AB8500_SIDFIRGAIN2 0x52
+#define AB8500_ANCCONF1 0x53
+#define AB8500_ANCCONF2 0x54
+#define AB8500_ANCCONF3 0x55
+#define AB8500_ANCCONF4 0x56
+#define AB8500_ANCCONF5 0x57
+#define AB8500_ANCCONF6 0x58
+#define AB8500_ANCCONF7 0x59
+#define AB8500_ANCCONF8 0x5A
+#define AB8500_ANCCONF9 0x5B
+#define AB8500_ANCCONF10 0x5C
+#define AB8500_ANCCONF11 0x5D
+#define AB8500_ANCCONF12 0x5E
+#define AB8500_ANCCONF13 0x5F
+#define AB8500_ANCCONF14 0x60
+#define AB8500_SIDFIRADR 0x61
+#define AB8500_SIDFIRCOEF1 0x62
+#define AB8500_SIDFIRCOEF2 0x63
+#define AB8500_SIDFIRCONF 0x64
+#define AB8500_AUDINTMASK1 0x65
+#define AB8500_AUDINTSOURCE1 0x66
+#define AB8500_AUDINTMASK2 0x67
+#define AB8500_AUDINTSOURCE2 0x68
+#define AB8500_FIFOCONF1 0x69
+#define AB8500_FIFOCONF2 0x6A
+#define AB8500_FIFOCONF3 0x6B
+#define AB8500_FIFOCONF4 0x6C
+#define AB8500_FIFOCONF5 0x6D
+#define AB8500_FIFOCONF6 0x6E
+#define AB8500_AUDREV 0x6F
+
+#define AB8500_FIRST_REG AB8500_POWERUP
+#define AB8500_LAST_REG AB8500_AUDREV
+#define AB8500_CACHEREGNUM (AB8500_LAST_REG + 1)
+
+#define AB8500_MASK_ALL 0xFF
+#define AB8500_MASK_NONE 0x00
+
+/* AB8500_POWERUP */
+#define AB8500_POWERUP_POWERUP 7
+#define AB8500_POWERUP_ENANA 3
+
+/* AB8500_AUDSWRESET */
+#define AB8500_AUDSWRESET_SWRESET 7
+
+/* AB8500_ADPATHENA */
+#define AB8500_ADPATHENA_ENAD12 7
+#define AB8500_ADPATHENA_ENAD34 5
+#define AB8500_ADPATHENA_ENAD5768 3
+
+/* AB8500_DAPATHENA */
+#define AB8500_DAPATHENA_ENDA1 7
+#define AB8500_DAPATHENA_ENDA2 6
+#define AB8500_DAPATHENA_ENDA3 5
+#define AB8500_DAPATHENA_ENDA4 4
+#define AB8500_DAPATHENA_ENDA5 3
+#define AB8500_DAPATHENA_ENDA6 2
+
+/* AB8500_ANACONF1 */
+#define AB8500_ANACONF1_HSLOWPOW 7
+#define AB8500_ANACONF1_DACLOWPOW1 6
+#define AB8500_ANACONF1_DACLOWPOW0 5
+#define AB8500_ANACONF1_EARDACLOWPOW 4
+#define AB8500_ANACONF1_EARSELCM 2
+#define AB8500_ANACONF1_HSHPEN 1
+#define AB8500_ANACONF1_EARDRVLOWPOW 0
+
+/* AB8500_ANACONF2 */
+#define AB8500_ANACONF2_ENMIC1 7
+#define AB8500_ANACONF2_ENMIC2 6
+#define AB8500_ANACONF2_ENLINL 5
+#define AB8500_ANACONF2_ENLINR 4
+#define AB8500_ANACONF2_MUTMIC1 3
+#define AB8500_ANACONF2_MUTMIC2 2
+#define AB8500_ANACONF2_MUTLINL 1
+#define AB8500_ANACONF2_MUTLINR 0
+
+/* AB8500_DIGMICCONF */
+#define AB8500_DIGMICCONF_ENDMIC1 7
+#define AB8500_DIGMICCONF_ENDMIC2 6
+#define AB8500_DIGMICCONF_ENDMIC3 5
+#define AB8500_DIGMICCONF_ENDMIC4 4
+#define AB8500_DIGMICCONF_ENDMIC5 3
+#define AB8500_DIGMICCONF_ENDMIC6 2
+#define AB8500_DIGMICCONF_HSFADSPEED 0
+
+/* AB8500_ANACONF3 */
+#define AB8500_ANACONF3_MIC1SEL 7
+#define AB8500_ANACONF3_LINRSEL 6
+#define AB8500_ANACONF3_ENDRVHSL 5
+#define AB8500_ANACONF3_ENDRVHSR 4
+#define AB8500_ANACONF3_ENADCMIC 2
+#define AB8500_ANACONF3_ENADCLINL 1
+#define AB8500_ANACONF3_ENADCLINR 0
+
+/* AB8500_ANACONF4 */
+#define AB8500_ANACONF4_DISPDVSS 7
+#define AB8500_ANACONF4_ENEAR 6
+#define AB8500_ANACONF4_ENHSL 5
+#define AB8500_ANACONF4_ENHSR 4
+#define AB8500_ANACONF4_ENHFL 3
+#define AB8500_ANACONF4_ENHFR 2
+#define AB8500_ANACONF4_ENVIB1 1
+#define AB8500_ANACONF4_ENVIB2 0
+
+/* AB8500_DAPATHCONF */
+#define AB8500_DAPATHCONF_ENDACEAR 6
+#define AB8500_DAPATHCONF_ENDACHSL 5
+#define AB8500_DAPATHCONF_ENDACHSR 4
+#define AB8500_DAPATHCONF_ENDACHFL 3
+#define AB8500_DAPATHCONF_ENDACHFR 2
+#define AB8500_DAPATHCONF_ENDACVIB1 1
+#define AB8500_DAPATHCONF_ENDACVIB2 0
+
+/* AB8500_MUTECONF */
+#define AB8500_MUTECONF_MUTEAR 6
+#define AB8500_MUTECONF_MUTHSL 5
+#define AB8500_MUTECONF_MUTHSR 4
+#define AB8500_MUTECONF_MUTDACEAR 2
+#define AB8500_MUTECONF_MUTDACHSL 1
+#define AB8500_MUTECONF_MUTDACHSR 0
+
+/* AB8500_SHORTCIRCONF */
+#define AB8500_SHORTCIRCONF_ENSHORTPWD 7
+#define AB8500_SHORTCIRCONF_EARSHORTDIS 6
+#define AB8500_SHORTCIRCONF_HSSHORTDIS 5
+#define AB8500_SHORTCIRCONF_HSPULLDEN 4
+#define AB8500_SHORTCIRCONF_HSOSCEN 2
+#define AB8500_SHORTCIRCONF_HSFADDIS 1
+#define AB8500_SHORTCIRCONF_HSZCDDIS 0
+/* Zero cross should be disabled */
+
+/* AB8500_ANACONF5 */
+#define AB8500_ANACONF5_ENCPHS 7
+#define AB8500_ANACONF5_HSLDACTOLOL 5
+#define AB8500_ANACONF5_HSRDACTOLOR 4
+#define AB8500_ANACONF5_ENLOL 3
+#define AB8500_ANACONF5_ENLOR 2
+#define AB8500_ANACONF5_HSAUTOEN 0
+
+/* AB8500_ENVCPCONF */
+#define AB8500_ENVCPCONF_ENVDETHTHRE 4
+#define AB8500_ENVCPCONF_ENVDETLTHRE 0
+#define AB8500_ENVCPCONF_ENVDETHTHRE_MAX 0x0F
+#define AB8500_ENVCPCONF_ENVDETLTHRE_MAX 0x0F
+
+/* AB8500_SIGENVCONF */
+#define AB8500_SIGENVCONF_CPLVEN 5
+#define AB8500_SIGENVCONF_ENVDETCPEN 4
+#define AB8500_SIGENVCONF_ENVDETTIME 0
+#define AB8500_SIGENVCONF_ENVDETTIME_MAX 0x0F
+
+/* AB8500_PWMGENCONF1 */
+#define AB8500_PWMGENCONF1_PWMTOVIB1 7
+#define AB8500_PWMGENCONF1_PWMTOVIB2 6
+#define AB8500_PWMGENCONF1_PWM1CTRL 5
+#define AB8500_PWMGENCONF1_PWM2CTRL 4
+#define AB8500_PWMGENCONF1_PWM1NCTRL 3
+#define AB8500_PWMGENCONF1_PWM1PCTRL 2
+#define AB8500_PWMGENCONF1_PWM2NCTRL 1
+#define AB8500_PWMGENCONF1_PWM2PCTRL 0
+
+/* AB8500_PWMGENCONF2 */
+/* AB8500_PWMGENCONF3 */
+/* AB8500_PWMGENCONF4 */
+/* AB8500_PWMGENCONF5 */
+#define AB8500_PWMGENCONFX_PWMVIBXPOL 7
+#define AB8500_PWMGENCONFX_PWMVIBXDUTCYC 0
+#define AB8500_PWMGENCONFX_PWMVIBXDUTCYC_MAX 0x64
+
+/* AB8500_ANAGAIN1 */
+/* AB8500_ANAGAIN2 */
+#define AB8500_ANAGAINX_ENSEMICX 7
+#define AB8500_ANAGAINX_LOWPOWMICX 6
+#define AB8500_ANAGAINX_MICXGAIN 0
+#define AB8500_ANAGAINX_MICXGAIN_MAX 0x1F
+
+/* AB8500_ANAGAIN3 */
+#define AB8500_ANAGAIN3_HSLGAIN 4
+#define AB8500_ANAGAIN3_HSRGAIN 0
+#define AB8500_ANAGAIN3_HSXGAIN_MAX 0x0F
+
+/* AB8500_ANAGAIN4 */
+#define AB8500_ANAGAIN4_LINLGAIN 4
+#define AB8500_ANAGAIN4_LINRGAIN 0
+#define AB8500_ANAGAIN4_LINXGAIN_MAX 0x0F
+
+/* AB8500_DIGLINHSLGAIN */
+/* AB8500_DIGLINHSRGAIN */
+#define AB8500_DIGLINHSXGAIN_LINTOHSXGAIN 0
+#define AB8500_DIGLINHSXGAIN_LINTOHSXGAIN_MAX 0x13
+
+/* AB8500_ADFILTCONF */
+#define AB8500_ADFILTCONF_AD1NH 7
+#define AB8500_ADFILTCONF_AD2NH 6
+#define AB8500_ADFILTCONF_AD3NH 5
+#define AB8500_ADFILTCONF_AD4NH 4
+#define AB8500_ADFILTCONF_AD1VOICE 3
+#define AB8500_ADFILTCONF_AD2VOICE 2
+#define AB8500_ADFILTCONF_AD3VOICE 1
+#define AB8500_ADFILTCONF_AD4VOICE 0
+
+/* AB8500_DIGIFCONF1 */
+#define AB8500_DIGIFCONF1_ENMASTGEN 7
+#define AB8500_DIGIFCONF1_IF1BITCLKOS1 6
+#define AB8500_DIGIFCONF1_IF1BITCLKOS0 5
+#define AB8500_DIGIFCONF1_ENFSBITCLK1 4
+#define AB8500_DIGIFCONF1_IF0BITCLKOS1 2
+#define AB8500_DIGIFCONF1_IF0BITCLKOS0 1
+#define AB8500_DIGIFCONF1_ENFSBITCLK0 0
+
+/* AB8500_DIGIFCONF2 */
+#define AB8500_DIGIFCONF2_FSYNC0P 6
+#define AB8500_DIGIFCONF2_BITCLK0P 5
+#define AB8500_DIGIFCONF2_IF0DEL 4
+#define AB8500_DIGIFCONF2_IF0FORMAT1 3
+#define AB8500_DIGIFCONF2_IF0FORMAT0 2
+#define AB8500_DIGIFCONF2_IF0WL1 1
+#define AB8500_DIGIFCONF2_IF0WL0 0
+
+/* AB8500_DIGIFCONF3 */
+#define AB8500_DIGIFCONF3_IF0DATOIF1AD 7
+#define AB8500_DIGIFCONF3_IF0CLKTOIF1CLK 6
+#define AB8500_DIGIFCONF3_IF1MASTER 5
+#define AB8500_DIGIFCONF3_IF1DATOIF0AD 3
+#define AB8500_DIGIFCONF3_IF1CLKTOIF0CLK 2
+#define AB8500_DIGIFCONF3_IF0MASTER 1
+#define AB8500_DIGIFCONF3_IF0BFIFOEN 0
+
+/* AB8500_DIGIFCONF4 */
+#define AB8500_DIGIFCONF4_FSYNC1P 6
+#define AB8500_DIGIFCONF4_BITCLK1P 5
+#define AB8500_DIGIFCONF4_IF1DEL 4
+#define AB8500_DIGIFCONF4_IF1FORMAT1 3
+#define AB8500_DIGIFCONF4_IF1FORMAT0 2
+#define AB8500_DIGIFCONF4_IF1WL1 1
+#define AB8500_DIGIFCONF4_IF1WL0 0
+
+/* AB8500_ADSLOTSELX */
+#define AB8500_ADSLOTSELX_AD_OUT1_TO_SLOT_ODD 0x00
+#define AB8500_ADSLOTSELX_AD_OUT2_TO_SLOT_ODD 0x01
+#define AB8500_ADSLOTSELX_AD_OUT3_TO_SLOT_ODD 0x02
+#define AB8500_ADSLOTSELX_AD_OUT4_TO_SLOT_ODD 0x03
+#define AB8500_ADSLOTSELX_AD_OUT5_TO_SLOT_ODD 0x04
+#define AB8500_ADSLOTSELX_AD_OUT6_TO_SLOT_ODD 0x05
+#define AB8500_ADSLOTSELX_AD_OUT7_TO_SLOT_ODD 0x06
+#define AB8500_ADSLOTSELX_AD_OUT8_TO_SLOT_ODD 0x07
+#define AB8500_ADSLOTSELX_ZEROES_TO_SLOT_ODD 0x08
+#define AB8500_ADSLOTSELX_TRISTATE_TO_SLOT_ODD 0x0F
+#define AB8500_ADSLOTSELX_AD_OUT1_TO_SLOT_EVEN 0x00
+#define AB8500_ADSLOTSELX_AD_OUT2_TO_SLOT_EVEN 0x10
+#define AB8500_ADSLOTSELX_AD_OUT3_TO_SLOT_EVEN 0x20
+#define AB8500_ADSLOTSELX_AD_OUT4_TO_SLOT_EVEN 0x30
+#define AB8500_ADSLOTSELX_AD_OUT5_TO_SLOT_EVEN 0x40
+#define AB8500_ADSLOTSELX_AD_OUT6_TO_SLOT_EVEN 0x50
+#define AB8500_ADSLOTSELX_AD_OUT7_TO_SLOT_EVEN 0x60
+#define AB8500_ADSLOTSELX_AD_OUT8_TO_SLOT_EVEN 0x70
+#define AB8500_ADSLOTSELX_ZEROES_TO_SLOT_EVEN 0x80
+#define AB8500_ADSLOTSELX_TRISTATE_TO_SLOT_EVEN 0xF0
+#define AB8500_ADSLOTSELX_EVEN_SHIFT 0
+#define AB8500_ADSLOTSELX_ODD_SHIFT 4
+
+/* AB8500_ADSLOTHIZCTRL1 */
+/* AB8500_ADSLOTHIZCTRL2 */
+/* AB8500_ADSLOTHIZCTRL3 */
+/* AB8500_ADSLOTHIZCTRL4 */
+/* AB8500_DASLOTCONF1 */
+#define AB8500_DASLOTCONF1_DA12VOICE 7
+#define AB8500_DASLOTCONF1_SWAPDA12_34 6
+#define AB8500_DASLOTCONF1_DAI7TOADO1 5
+
+/* AB8500_DASLOTCONF2 */
+#define AB8500_DASLOTCONF2_DAI8TOADO2 5
+
+/* AB8500_DASLOTCONF3 */
+#define AB8500_DASLOTCONF3_DA34VOICE 7
+#define AB8500_DASLOTCONF3_DAI7TOADO3 5
+
+/* AB8500_DASLOTCONF4 */
+#define AB8500_DASLOTCONF4_DAI8TOADO4 5
+
+/* AB8500_DASLOTCONF5 */
+#define AB8500_DASLOTCONF5_DA56VOICE 7
+#define AB8500_DASLOTCONF5_DAI7TOADO5 5
+
+/* AB8500_DASLOTCONF6 */
+#define AB8500_DASLOTCONF6_DAI8TOADO6 5
+
+/* AB8500_DASLOTCONF7 */
+#define AB8500_DASLOTCONF7_DAI8TOADO7 5
+
+/* AB8500_DASLOTCONF8 */
+#define AB8500_DASLOTCONF8_DAI7TOADO8 5
+
+#define AB8500_DASLOTCONFX_SLTODAX_SHIFT 0
+#define AB8500_DASLOTCONFX_SLTODAX_MASK 0x1F
+
+/* AB8500_CLASSDCONF1 */
+#define AB8500_CLASSDCONF1_PARLHF 7
+#define AB8500_CLASSDCONF1_PARLVIB 6
+#define AB8500_CLASSDCONF1_VIB1SWAPEN 3
+#define AB8500_CLASSDCONF1_VIB2SWAPEN 2
+#define AB8500_CLASSDCONF1_HFLSWAPEN 1
+#define AB8500_CLASSDCONF1_HFRSWAPEN 0
+
+/* AB8500_CLASSDCONF2 */
+#define AB8500_CLASSDCONF2_FIRBYP3 7
+#define AB8500_CLASSDCONF2_FIRBYP2 6
+#define AB8500_CLASSDCONF2_FIRBYP1 5
+#define AB8500_CLASSDCONF2_FIRBYP0 4
+#define AB8500_CLASSDCONF2_HIGHVOLEN3 3
+#define AB8500_CLASSDCONF2_HIGHVOLEN2 2
+#define AB8500_CLASSDCONF2_HIGHVOLEN1 1
+#define AB8500_CLASSDCONF2_HIGHVOLEN0 0
+
+/* AB8500_CLASSDCONF3 */
+#define AB8500_CLASSDCONF3_DITHHPGAIN 4
+#define AB8500_CLASSDCONF3_DITHHPGAIN_MAX 0x0A
+#define AB8500_CLASSDCONF3_DITHWGAIN 0
+#define AB8500_CLASSDCONF3_DITHWGAIN_MAX 0x0A
+
+/* AB8500_DMICFILTCONF */
+#define AB8500_DMICFILTCONF_ANCINSEL 7
+#define AB8500_DMICFILTCONF_DA3TOEAR 6
+#define AB8500_DMICFILTCONF_DMIC1SINC3 5
+#define AB8500_DMICFILTCONF_DMIC2SINC3 4
+#define AB8500_DMICFILTCONF_DMIC3SINC3 3
+#define AB8500_DMICFILTCONF_DMIC4SINC3 2
+#define AB8500_DMICFILTCONF_DMIC5SINC3 1
+#define AB8500_DMICFILTCONF_DMIC6SINC3 0
+
+/* AB8500_DIGMULTCONF1 */
+#define AB8500_DIGMULTCONF1_DATOHSLEN 7
+#define AB8500_DIGMULTCONF1_DATOHSREN 6
+#define AB8500_DIGMULTCONF1_AD1SEL 5
+#define AB8500_DIGMULTCONF1_AD2SEL 4
+#define AB8500_DIGMULTCONF1_AD3SEL 3
+#define AB8500_DIGMULTCONF1_AD5SEL 2
+#define AB8500_DIGMULTCONF1_AD6SEL 1
+#define AB8500_DIGMULTCONF1_ANCSEL 0
+
+/* AB8500_DIGMULTCONF2 */
+#define AB8500_DIGMULTCONF2_DATOHFREN 7
+#define AB8500_DIGMULTCONF2_DATOHFLEN 6
+#define AB8500_DIGMULTCONF2_HFRSEL 5
+#define AB8500_DIGMULTCONF2_HFLSEL 4
+#define AB8500_DIGMULTCONF2_FIRSID1SEL 2
+#define AB8500_DIGMULTCONF2_FIRSID2SEL 0
+
+/* AB8500_ADDIGGAIN1 */
+/* AB8500_ADDIGGAIN2 */
+/* AB8500_ADDIGGAIN3 */
+/* AB8500_ADDIGGAIN4 */
+/* AB8500_ADDIGGAIN5 */
+/* AB8500_ADDIGGAIN6 */
+#define AB8500_ADDIGGAINX_FADEDISADX 6
+#define AB8500_ADDIGGAINX_ADXGAIN_MAX 0x3F
+
+/* AB8500_DADIGGAIN1 */
+/* AB8500_DADIGGAIN2 */
+/* AB8500_DADIGGAIN3 */
+/* AB8500_DADIGGAIN4 */
+/* AB8500_DADIGGAIN5 */
+/* AB8500_DADIGGAIN6 */
+#define AB8500_DADIGGAINX_FADEDISDAX 6
+#define AB8500_DADIGGAINX_DAXGAIN_MAX 0x3F
+
+/* AB8500_ADDIGLOOPGAIN1 */
+/* AB8500_ADDIGLOOPGAIN2 */
+#define AB8500_ADDIGLOOPGAINX_FADEDISADXL 6
+#define AB8500_ADDIGLOOPGAINX_ADXLBGAIN_MAX 0x3F
+
+/* AB8500_HSLEARDIGGAIN */
+#define AB8500_HSLEARDIGGAIN_HSSINC1 7
+#define AB8500_HSLEARDIGGAIN_FADEDISHSL 4
+#define AB8500_HSLEARDIGGAIN_HSLDGAIN_MAX 0x09
+
+/* AB8500_HSRDIGGAIN */
+#define AB8500_HSRDIGGAIN_FADESPEED 6
+#define AB8500_HSRDIGGAIN_FADEDISHSR 4
+#define AB8500_HSRDIGGAIN_HSRDGAIN_MAX 0x09
+
+/* AB8500_SIDFIRGAIN1 */
+/* AB8500_SIDFIRGAIN2 */
+#define AB8500_SIDFIRGAINX_FIRSIDXGAIN_MAX 0x1F
+
+/* AB8500_ANCCONF1 */
+#define AB8500_ANCCONF1_ANCIIRUPDATE 3
+#define AB8500_ANCCONF1_ENANC 2
+#define AB8500_ANCCONF1_ANCIIRINIT 1
+#define AB8500_ANCCONF1_ANCFIRUPDATE 0
+
+/* AB8500_ANCCONF2 */
+#define AB8500_ANCCONF2_SHIFT 5
+#define AB8500_ANCCONF2_MIN -0x10
+#define AB8500_ANCCONF2_MAX 0xF
+
+/* AB8500_ANCCONF3 */
+#define AB8500_ANCCONF3_SHIFT 5
+#define AB8500_ANCCONF3_MIN -0x10
+#define AB8500_ANCCONF3_MAX 0xF
+
+/* AB8500_ANCCONF4 */
+#define AB8500_ANCCONF4_SHIFT 5
+#define AB8500_ANCCONF4_MIN -0x10
+#define AB8500_ANCCONF4_MAX 0xF
+
+/* AB8500_ANC_FIR_COEFFS */
+#define AB8500_ANC_FIR_COEFF_MIN -0x8000
+#define AB8500_ANC_FIR_COEFF_MAX 0x7FFF
+#define AB8500_ANC_FIR_COEFFS 15
+
+/* AB8500_ANC_IIR_COEFFS */
+#define AB8500_ANC_IIR_COEFF_MIN -0x800000
+#define AB8500_ANC_IIR_COEFF_MAX 0x7FFFFF
+#define AB8500_ANC_IIR_COEFFS 24
+/* AB8500_ANC_WARP_DELAY */
+#define AB8500_ANC_WARP_DELAY_SHIFT 16
+#define AB8500_ANC_WARP_DELAY_MIN 0x0000
+#define AB8500_ANC_WARP_DELAY_MAX 0xFFFF
+
+/* AB8500_ANCCONF11 */
+/* AB8500_ANCCONF12 */
+/* AB8500_ANCCONF13 */
+/* AB8500_ANCCONF14 */
+
+/* AB8500_SIDFIRADR */
+#define AB8500_SIDFIRADR_FIRSIDSET 7
+#define AB8500_SIDFIRADR_ADDRESS_SHIFT 0
+#define AB8500_SIDFIRADR_ADDRESS_MAX 0x7F
+
+/* AB8500_SIDFIRCOEF1 */
+/* AB8500_SIDFIRCOEF2 */
+#define AB8500_SID_FIR_COEFF_MIN 0
+#define AB8500_SID_FIR_COEFF_MAX 0xFFFF
+#define AB8500_SID_FIR_COEFFS 128
+
+/* AB8500_SIDFIRCONF */
+#define AB8500_SIDFIRCONF_ENFIRSIDS 2
+#define AB8500_SIDFIRCONF_FIRSIDSTOIF1 1
+#define AB8500_SIDFIRCONF_FIRSIDBUSY 0
+
+/* AB8500_AUDINTMASK1 */
+/* AB8500_AUDINTSOURCE1 */
+/* AB8500_AUDINTMASK2 */
+/* AB8500_AUDINTSOURCE2 */
+
+/* AB8500_FIFOCONF1 */
+#define AB8500_FIFOCONF1_BFIFOMASK 0x80
+#define AB8500_FIFOCONF1_BFIFO19M2 0x40
+#define AB8500_FIFOCONF1_BFIFOINT_SHIFT 0
+#define AB8500_FIFOCONF1_BFIFOINT_MAX 0x3F
+
+/* AB8500_FIFOCONF2 */
+#define AB8500_FIFOCONF2_BFIFOTX_SHIFT 0
+#define AB8500_FIFOCONF2_BFIFOTX_MAX 0xFF
+
+/* AB8500_FIFOCONF3 */
+#define AB8500_FIFOCONF3_BFIFOEXSL_SHIFT 5
+#define AB8500_FIFOCONF3_BFIFOEXSL_MAX 0x5
+#define AB8500_FIFOCONF3_PREBITCLK0_SHIFT 2
+#define AB8500_FIFOCONF3_PREBITCLK0_MAX 0x7
+#define AB8500_FIFOCONF3_BFIFOMAST_SHIFT 1
+#define AB8500_FIFOCONF3_BFIFORUN_SHIFT 0
+
+/* AB8500_FIFOCONF4 */
+#define AB8500_FIFOCONF4_BFIFOFRAMSW_SHIFT 0
+#define AB8500_FIFOCONF4_BFIFOFRAMSW_MAX 0xFF
+
+/* AB8500_FIFOCONF5 */
+#define AB8500_FIFOCONF5_BFIFOWAKEUP_SHIFT 0
+#define AB8500_FIFOCONF5_BFIFOWAKEUP_MAX 0xFF
+
+/* AB8500_FIFOCONF6 */
+#define AB8500_FIFOCONF6_BFIFOSAMPLE_SHIFT 0
+#define AB8500_FIFOCONF6_BFIFOSAMPLE_MAX 0xFF
+
+/* AB8500_AUDREV */
+
+#endif
diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c
index 2023c74..ea06b83 100644
--- a/sound/soc/codecs/ac97.c
+++ b/sound/soc/codecs/ac97.c
@@ -91,11 +91,6 @@ static int ac97_soc_probe(struct snd_soc_codec *codec)
return 0;
}
-static int ac97_soc_remove(struct snd_soc_codec *codec)
-{
- return 0;
-}
-
#ifdef CONFIG_PM
static int ac97_soc_suspend(struct snd_soc_codec *codec)
{
@@ -119,7 +114,6 @@ static struct snd_soc_codec_driver soc_codec_dev_ac97 = {
.write = ac97_write,
.read = ac97_read,
.probe = ac97_soc_probe,
- .remove = ac97_soc_remove,
.suspend = ac97_soc_suspend,
.resume = ac97_soc_resume,
};
diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c
new file mode 100644
index 0000000..5c9caca
--- /dev/null
+++ b/sound/soc/codecs/arizona.c
@@ -0,0 +1,937 @@
+/*
+ * arizona.c - Wolfson Arizona class device shared support
+ *
+ * Copyright 2012 Wolfson Microelectronics plc
+ *
+ * Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
+ *
+ * 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 <linux/gcd.h>
+#include <linux/module.h>
+#include <linux/pm_runtime.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/tlv.h>
+
+#include <linux/mfd/arizona/core.h>
+#include <linux/mfd/arizona/registers.h>
+
+#include "arizona.h"
+
+#define ARIZONA_AIF_BCLK_CTRL 0x00
+#define ARIZONA_AIF_TX_PIN_CTRL 0x01
+#define ARIZONA_AIF_RX_PIN_CTRL 0x02
+#define ARIZONA_AIF_RATE_CTRL 0x03
+#define ARIZONA_AIF_FORMAT 0x04
+#define ARIZONA_AIF_TX_BCLK_RATE 0x05
+#define ARIZONA_AIF_RX_BCLK_RATE 0x06
+#define ARIZONA_AIF_FRAME_CTRL_1 0x07
+#define ARIZONA_AIF_FRAME_CTRL_2 0x08
+#define ARIZONA_AIF_FRAME_CTRL_3 0x09
+#define ARIZONA_AIF_FRAME_CTRL_4 0x0A
+#define ARIZONA_AIF_FRAME_CTRL_5 0x0B
+#define ARIZONA_AIF_FRAME_CTRL_6 0x0C
+#define ARIZONA_AIF_FRAME_CTRL_7 0x0D
+#define ARIZONA_AIF_FRAME_CTRL_8 0x0E
+#define ARIZONA_AIF_FRAME_CTRL_9 0x0F
+#define ARIZONA_AIF_FRAME_CTRL_10 0x10
+#define ARIZONA_AIF_FRAME_CTRL_11 0x11
+#define ARIZONA_AIF_FRAME_CTRL_12 0x12
+#define ARIZONA_AIF_FRAME_CTRL_13 0x13
+#define ARIZONA_AIF_FRAME_CTRL_14 0x14
+#define ARIZONA_AIF_FRAME_CTRL_15 0x15
+#define ARIZONA_AIF_FRAME_CTRL_16 0x16
+#define ARIZONA_AIF_FRAME_CTRL_17 0x17
+#define ARIZONA_AIF_FRAME_CTRL_18 0x18
+#define ARIZONA_AIF_TX_ENABLES 0x19
+#define ARIZONA_AIF_RX_ENABLES 0x1A
+#define ARIZONA_AIF_FORCE_WRITE 0x1B
+
+#define arizona_fll_err(_fll, fmt, ...) \
+ dev_err(_fll->arizona->dev, "FLL%d: " fmt, _fll->id, ##__VA_ARGS__)
+#define arizona_fll_warn(_fll, fmt, ...) \
+ dev_warn(_fll->arizona->dev, "FLL%d: " fmt, _fll->id, ##__VA_ARGS__)
+#define arizona_fll_dbg(_fll, fmt, ...) \
+ dev_err(_fll->arizona->dev, "FLL%d: " fmt, _fll->id, ##__VA_ARGS__)
+
+#define arizona_aif_err(_dai, fmt, ...) \
+ dev_err(_dai->dev, "AIF%d: " fmt, _dai->id, ##__VA_ARGS__)
+#define arizona_aif_warn(_dai, fmt, ...) \
+ dev_warn(_dai->dev, "AIF%d: " fmt, _dai->id, ##__VA_ARGS__)
+#define arizona_aif_dbg(_dai, fmt, ...) \
+ dev_err(_dai->dev, "AIF%d: " fmt, _dai->id, ##__VA_ARGS__)
+
+const char *arizona_mixer_texts[ARIZONA_NUM_MIXER_INPUTS] = {
+ "None",
+ "Tone Generator 1",
+ "Tone Generator 2",
+ "Haptics",
+ "AEC",
+ "Mic Mute Mixer",
+ "Noise Generator",
+ "IN1L",
+ "IN1R",
+ "IN2L",
+ "IN2R",
+ "IN3L",
+ "IN3R",
+ "IN4L",
+ "IN4R",
+ "AIF1RX1",
+ "AIF1RX2",
+ "AIF1RX3",
+ "AIF1RX4",
+ "AIF1RX5",
+ "AIF1RX6",
+ "AIF1RX7",
+ "AIF1RX8",
+ "AIF2RX1",
+ "AIF2RX2",
+ "AIF3RX1",
+ "AIF3RX2",
+ "SLIMRX1",
+ "SLIMRX2",
+ "SLIMRX3",
+ "SLIMRX4",
+ "SLIMRX5",
+ "SLIMRX6",
+ "SLIMRX7",
+ "SLIMRX8",
+ "EQ1",
+ "EQ2",
+ "EQ3",
+ "EQ4",
+ "DRC1L",
+ "DRC1R",
+ "DRC2L",
+ "DRC2R",
+ "LHPF1",
+ "LHPF2",
+ "LHPF3",
+ "LHPF4",
+ "DSP1.1",
+ "DSP1.2",
+ "DSP1.3",
+ "DSP1.4",
+ "DSP1.5",
+ "DSP1.6",
+ "ASRC1L",
+ "ASRC1R",
+ "ASRC2L",
+ "ASRC2R",
+};
+EXPORT_SYMBOL_GPL(arizona_mixer_texts);
+
+int arizona_mixer_values[ARIZONA_NUM_MIXER_INPUTS] = {
+ 0x00, /* None */
+ 0x04, /* Tone */
+ 0x05,
+ 0x06, /* Haptics */
+ 0x08, /* AEC */
+ 0x0c, /* Noise mixer */
+ 0x0d, /* Comfort noise */
+ 0x10, /* IN1L */
+ 0x11,
+ 0x12,
+ 0x13,
+ 0x14,
+ 0x15,
+ 0x16,
+ 0x17,
+ 0x20, /* AIF1RX1 */
+ 0x21,
+ 0x22,
+ 0x23,
+ 0x24,
+ 0x25,
+ 0x26,
+ 0x27,
+ 0x28, /* AIF2RX1 */
+ 0x29,
+ 0x30, /* AIF3RX1 */
+ 0x31,
+ 0x38, /* SLIMRX1 */
+ 0x39,
+ 0x3a,
+ 0x3b,
+ 0x3c,
+ 0x3d,
+ 0x3e,
+ 0x3f,
+ 0x50, /* EQ1 */
+ 0x51,
+ 0x52,
+ 0x53,
+ 0x58, /* DRC1L */
+ 0x59,
+ 0x5a,
+ 0x5b,
+ 0x60, /* LHPF1 */
+ 0x61,
+ 0x62,
+ 0x63,
+ 0x68, /* DSP1.1 */
+ 0x69,
+ 0x6a,
+ 0x6b,
+ 0x6c,
+ 0x6d,
+ 0x90, /* ASRC1L */
+ 0x91,
+ 0x92,
+ 0x93,
+};
+EXPORT_SYMBOL_GPL(arizona_mixer_values);
+
+const DECLARE_TLV_DB_SCALE(arizona_mixer_tlv, -3200, 100, 0);
+EXPORT_SYMBOL_GPL(arizona_mixer_tlv);
+
+static const char *arizona_lhpf_mode_text[] = {
+ "Low-pass", "High-pass"
+};
+
+const struct soc_enum arizona_lhpf1_mode =
+ SOC_ENUM_SINGLE(ARIZONA_HPLPF1_1, ARIZONA_LHPF1_MODE_SHIFT, 2,
+ arizona_lhpf_mode_text);
+EXPORT_SYMBOL_GPL(arizona_lhpf1_mode);
+
+const struct soc_enum arizona_lhpf2_mode =
+ SOC_ENUM_SINGLE(ARIZONA_HPLPF2_1, ARIZONA_LHPF2_MODE_SHIFT, 2,
+ arizona_lhpf_mode_text);
+EXPORT_SYMBOL_GPL(arizona_lhpf2_mode);
+
+const struct soc_enum arizona_lhpf3_mode =
+ SOC_ENUM_SINGLE(ARIZONA_HPLPF3_1, ARIZONA_LHPF3_MODE_SHIFT, 2,
+ arizona_lhpf_mode_text);
+EXPORT_SYMBOL_GPL(arizona_lhpf3_mode);
+
+const struct soc_enum arizona_lhpf4_mode =
+ SOC_ENUM_SINGLE(ARIZONA_HPLPF4_1, ARIZONA_LHPF4_MODE_SHIFT, 2,
+ arizona_lhpf_mode_text);
+EXPORT_SYMBOL_GPL(arizona_lhpf4_mode);
+
+int arizona_in_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol,
+ int event)
+{
+ return 0;
+}
+EXPORT_SYMBOL_GPL(arizona_in_ev);
+
+int arizona_out_ev(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol,
+ int event)
+{
+ return 0;
+}
+EXPORT_SYMBOL_GPL(arizona_out_ev);
+
+int arizona_set_sysclk(struct snd_soc_codec *codec, int clk_id,
+ int source, unsigned int freq, int dir)
+{
+ struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec);
+ struct arizona *arizona = priv->arizona;
+ char *name;
+ unsigned int reg;
+ unsigned int mask = ARIZONA_SYSCLK_FREQ_MASK | ARIZONA_SYSCLK_SRC_MASK;
+ unsigned int val = source << ARIZONA_SYSCLK_SRC_SHIFT;
+ unsigned int *clk;
+
+ switch (clk_id) {
+ case ARIZONA_CLK_SYSCLK:
+ name = "SYSCLK";
+ reg = ARIZONA_SYSTEM_CLOCK_1;
+ clk = &priv->sysclk;
+ mask |= ARIZONA_SYSCLK_FRAC;
+ break;
+ case ARIZONA_CLK_ASYNCCLK:
+ name = "ASYNCCLK";
+ reg = ARIZONA_ASYNC_CLOCK_1;
+ clk = &priv->asyncclk;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ switch (freq) {
+ case 5644800:
+ case 6144000:
+ break;
+ case 11289600:
+ case 12288000:
+ val |= 1 << ARIZONA_SYSCLK_FREQ_SHIFT;
+ break;
+ case 22579200:
+ case 24576000:
+ val |= 2 << ARIZONA_SYSCLK_FREQ_SHIFT;
+ break;
+ case 45158400:
+ case 49152000:
+ val |= 3 << ARIZONA_SYSCLK_FREQ_SHIFT;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ *clk = freq;
+
+ if (freq % 6144000)
+ val |= ARIZONA_SYSCLK_FRAC;
+
+ dev_dbg(arizona->dev, "%s set to %uHz", name, freq);
+
+ return regmap_update_bits(arizona->regmap, reg, mask, val);
+}
+EXPORT_SYMBOL_GPL(arizona_set_sysclk);
+
+static int arizona_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ int lrclk, bclk, mode, base;
+
+ base = dai->driver->base;
+
+ lrclk = 0;
+ bclk = 0;
+
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_DSP_A:
+ mode = 0;
+ break;
+ case SND_SOC_DAIFMT_DSP_B:
+ mode = 1;
+ break;
+ case SND_SOC_DAIFMT_I2S:
+ mode = 2;
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ mode = 3;
+ break;
+ default:
+ arizona_aif_err(dai, "Unsupported DAI format %d\n",
+ fmt & SND_SOC_DAIFMT_FORMAT_MASK);
+ return -EINVAL;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBS_CFS:
+ break;
+ case SND_SOC_DAIFMT_CBS_CFM:
+ lrclk |= ARIZONA_AIF1TX_LRCLK_MSTR;
+ break;
+ case SND_SOC_DAIFMT_CBM_CFS:
+ bclk |= ARIZONA_AIF1_BCLK_MSTR;
+ break;
+ case SND_SOC_DAIFMT_CBM_CFM:
+ bclk |= ARIZONA_AIF1_BCLK_MSTR;
+ lrclk |= ARIZONA_AIF1TX_LRCLK_MSTR;
+ break;
+ default:
+ arizona_aif_err(dai, "Unsupported master mode %d\n",
+ fmt & SND_SOC_DAIFMT_MASTER_MASK);
+ return -EINVAL;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_NB_NF:
+ break;
+ case SND_SOC_DAIFMT_IB_IF:
+ bclk |= ARIZONA_AIF1_BCLK_INV;
+ lrclk |= ARIZONA_AIF1TX_LRCLK_INV;
+ break;
+ case SND_SOC_DAIFMT_IB_NF:
+ bclk |= ARIZONA_AIF1_BCLK_INV;
+ break;
+ case SND_SOC_DAIFMT_NB_IF:
+ lrclk |= ARIZONA_AIF1TX_LRCLK_INV;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ snd_soc_update_bits(codec, base + ARIZONA_AIF_BCLK_CTRL,
+ ARIZONA_AIF1_BCLK_INV | ARIZONA_AIF1_BCLK_MSTR,
+ bclk);
+ snd_soc_update_bits(codec, base + ARIZONA_AIF_TX_PIN_CTRL,
+ ARIZONA_AIF1TX_LRCLK_INV |
+ ARIZONA_AIF1TX_LRCLK_MSTR, lrclk);
+ snd_soc_update_bits(codec, base + ARIZONA_AIF_RX_PIN_CTRL,
+ ARIZONA_AIF1RX_LRCLK_INV |
+ ARIZONA_AIF1RX_LRCLK_MSTR, lrclk);
+ snd_soc_update_bits(codec, base + ARIZONA_AIF_FORMAT,
+ ARIZONA_AIF1_FMT_MASK, mode);
+
+ return 0;
+}
+
+static const int arizona_48k_bclk_rates[] = {
+ -1,
+ 48000,
+ 64000,
+ 96000,
+ 128000,
+ 192000,
+ 256000,
+ 384000,
+ 512000,
+ 768000,
+ 1024000,
+ 1536000,
+ 2048000,
+ 3072000,
+ 4096000,
+ 6144000,
+ 8192000,
+ 12288000,
+ 24576000,
+};
+
+static const unsigned int arizona_48k_rates[] = {
+ 12000,
+ 24000,
+ 48000,
+ 96000,
+ 192000,
+ 384000,
+ 768000,
+ 4000,
+ 8000,
+ 16000,
+ 32000,
+ 64000,
+ 128000,
+ 256000,
+ 512000,
+};
+
+static const struct snd_pcm_hw_constraint_list arizona_48k_constraint = {
+ .count = ARRAY_SIZE(arizona_48k_rates),
+ .list = arizona_48k_rates,
+};
+
+static const int arizona_44k1_bclk_rates[] = {
+ -1,
+ 44100,
+ 58800,
+ 88200,
+ 117600,
+ 177640,
+ 235200,
+ 352800,
+ 470400,
+ 705600,
+ 940800,
+ 1411200,
+ 1881600,
+ 2882400,
+ 3763200,
+ 5644800,
+ 7526400,
+ 11289600,
+ 22579200,
+};
+
+static const unsigned int arizona_44k1_rates[] = {
+ 11025,
+ 22050,
+ 44100,
+ 88200,
+ 176400,
+ 352800,
+ 705600,
+};
+
+static const struct snd_pcm_hw_constraint_list arizona_44k1_constraint = {
+ .count = ARRAY_SIZE(arizona_44k1_rates),
+ .list = arizona_44k1_rates,
+};
+
+static int arizona_sr_vals[] = {
+ 0,
+ 12000,
+ 24000,
+ 48000,
+ 96000,
+ 192000,
+ 384000,
+ 768000,
+ 0,
+ 11025,
+ 22050,
+ 44100,
+ 88200,
+ 176400,
+ 352800,
+ 705600,
+ 4000,
+ 8000,
+ 16000,
+ 32000,
+ 64000,
+ 128000,
+ 256000,
+ 512000,
+};
+
+static int arizona_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec);
+ struct arizona_dai_priv *dai_priv = &priv->dai[dai->id - 1];
+ const struct snd_pcm_hw_constraint_list *constraint;
+ unsigned int base_rate;
+
+ switch (dai_priv->clk) {
+ case ARIZONA_CLK_SYSCLK:
+ base_rate = priv->sysclk;
+ break;
+ case ARIZONA_CLK_ASYNCCLK:
+ base_rate = priv->asyncclk;
+ break;
+ default:
+ return 0;
+ }
+
+ if (base_rate % 8000)
+ constraint = &arizona_44k1_constraint;
+ else
+ constraint = &arizona_48k_constraint;
+
+ return snd_pcm_hw_constraint_list(substream->runtime, 0,
+ SNDRV_PCM_HW_PARAM_RATE,
+ constraint);
+}
+
+static int arizona_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 arizona_priv *priv = snd_soc_codec_get_drvdata(codec);
+ struct arizona_dai_priv *dai_priv = &priv->dai[dai->id - 1];
+ int base = dai->driver->base;
+ const int *rates;
+ int i;
+ int bclk, lrclk, wl, frame, sr_val;
+
+ if (params_rate(params) % 8000)
+ rates = &arizona_44k1_bclk_rates[0];
+ else
+ rates = &arizona_48k_bclk_rates[0];
+
+ for (i = 0; i < ARRAY_SIZE(arizona_44k1_bclk_rates); i++) {
+ if (rates[i] >= snd_soc_params_to_bclk(params) &&
+ rates[i] % params_rate(params) == 0) {
+ bclk = i;
+ break;
+ }
+ }
+ if (i == ARRAY_SIZE(arizona_44k1_bclk_rates)) {
+ arizona_aif_err(dai, "Unsupported sample rate %dHz\n",
+ params_rate(params));
+ return -EINVAL;
+ }
+
+ for (i = 0; i < ARRAY_SIZE(arizona_sr_vals); i++)
+ if (arizona_sr_vals[i] == params_rate(params))
+ break;
+ if (i == ARRAY_SIZE(arizona_sr_vals)) {
+ arizona_aif_err(dai, "Unsupported sample rate %dHz\n",
+ params_rate(params));
+ return -EINVAL;
+ }
+ sr_val = i;
+
+ lrclk = snd_soc_params_to_bclk(params) / params_rate(params);
+
+ arizona_aif_dbg(dai, "BCLK %dHz LRCLK %dHz\n",
+ rates[bclk], rates[bclk] / lrclk);
+
+ wl = snd_pcm_format_width(params_format(params));
+ frame = wl << ARIZONA_AIF1TX_WL_SHIFT | wl;
+
+ /*
+ * We will need to be more flexible than this in future,
+ * currently we use a single sample rate for SYSCLK.
+ */
+ switch (dai_priv->clk) {
+ case ARIZONA_CLK_SYSCLK:
+ snd_soc_update_bits(codec, ARIZONA_SAMPLE_RATE_1,
+ ARIZONA_SAMPLE_RATE_1_MASK, sr_val);
+ snd_soc_update_bits(codec, base + ARIZONA_AIF_RATE_CTRL,
+ ARIZONA_AIF1_RATE_MASK, 0);
+ break;
+ case ARIZONA_CLK_ASYNCCLK:
+ snd_soc_update_bits(codec, ARIZONA_ASYNC_SAMPLE_RATE_1,
+ ARIZONA_ASYNC_SAMPLE_RATE_MASK, sr_val);
+ snd_soc_update_bits(codec, base + ARIZONA_AIF_RATE_CTRL,
+ ARIZONA_AIF1_RATE_MASK, 8);
+ break;
+ default:
+ arizona_aif_err(dai, "Invalid clock %d\n", dai_priv->clk);
+ return -EINVAL;
+ }
+
+ snd_soc_update_bits(codec, base + ARIZONA_AIF_BCLK_CTRL,
+ ARIZONA_AIF1_BCLK_FREQ_MASK, bclk);
+ snd_soc_update_bits(codec, base + ARIZONA_AIF_TX_BCLK_RATE,
+ ARIZONA_AIF1TX_BCPF_MASK, lrclk);
+ snd_soc_update_bits(codec, base + ARIZONA_AIF_RX_BCLK_RATE,
+ ARIZONA_AIF1RX_BCPF_MASK, lrclk);
+ snd_soc_update_bits(codec, base + ARIZONA_AIF_FRAME_CTRL_1,
+ ARIZONA_AIF1TX_WL_MASK |
+ ARIZONA_AIF1TX_SLOT_LEN_MASK, frame);
+ snd_soc_update_bits(codec, base + ARIZONA_AIF_FRAME_CTRL_2,
+ ARIZONA_AIF1RX_WL_MASK |
+ ARIZONA_AIF1RX_SLOT_LEN_MASK, frame);
+
+ return 0;
+}
+
+static const char *arizona_dai_clk_str(int clk_id)
+{
+ switch (clk_id) {
+ case ARIZONA_CLK_SYSCLK:
+ return "SYSCLK";
+ case ARIZONA_CLK_ASYNCCLK:
+ return "ASYNCCLK";
+ default:
+ return "Unknown clock";
+ }
+}
+
+static int arizona_dai_set_sysclk(struct snd_soc_dai *dai,
+ int clk_id, unsigned int freq, int dir)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec);
+ struct arizona_dai_priv *dai_priv = &priv->dai[dai->id - 1];
+ struct snd_soc_dapm_route routes[2];
+
+ switch (clk_id) {
+ case ARIZONA_CLK_SYSCLK:
+ case ARIZONA_CLK_ASYNCCLK:
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ if (clk_id == dai_priv->clk)
+ return 0;
+
+ if (dai->active) {
+ dev_err(codec->dev, "Can't change clock on active DAI %d\n",
+ dai->id);
+ return -EBUSY;
+ }
+
+ memset(&routes, 0, sizeof(routes));
+ routes[0].sink = dai->driver->capture.stream_name;
+ routes[1].sink = dai->driver->playback.stream_name;
+
+ routes[0].source = arizona_dai_clk_str(dai_priv->clk);
+ routes[1].source = arizona_dai_clk_str(dai_priv->clk);
+ snd_soc_dapm_del_routes(&codec->dapm, routes, ARRAY_SIZE(routes));
+
+ routes[0].source = arizona_dai_clk_str(clk_id);
+ routes[1].source = arizona_dai_clk_str(clk_id);
+ snd_soc_dapm_add_routes(&codec->dapm, routes, ARRAY_SIZE(routes));
+
+ return snd_soc_dapm_sync(&codec->dapm);
+}
+
+const struct snd_soc_dai_ops arizona_dai_ops = {
+ .startup = arizona_startup,
+ .set_fmt = arizona_set_fmt,
+ .hw_params = arizona_hw_params,
+ .set_sysclk = arizona_dai_set_sysclk,
+};
+EXPORT_SYMBOL_GPL(arizona_dai_ops);
+
+int arizona_init_dai(struct arizona_priv *priv, int id)
+{
+ struct arizona_dai_priv *dai_priv = &priv->dai[id];
+
+ dai_priv->clk = ARIZONA_CLK_SYSCLK;
+
+ return 0;
+}
+EXPORT_SYMBOL_GPL(arizona_init_dai);
+
+static irqreturn_t arizona_fll_lock(int irq, void *data)
+{
+ struct arizona_fll *fll = data;
+
+ arizona_fll_dbg(fll, "Locked\n");
+
+ complete(&fll->lock);
+
+ return IRQ_HANDLED;
+}
+
+static irqreturn_t arizona_fll_clock_ok(int irq, void *data)
+{
+ struct arizona_fll *fll = data;
+
+ arizona_fll_dbg(fll, "clock OK\n");
+
+ complete(&fll->ok);
+
+ return IRQ_HANDLED;
+}
+
+static struct {
+ unsigned int min;
+ unsigned int max;
+ u16 fratio;
+ int ratio;
+} fll_fratios[] = {
+ { 0, 64000, 4, 16 },
+ { 64000, 128000, 3, 8 },
+ { 128000, 256000, 2, 4 },
+ { 256000, 1000000, 1, 2 },
+ { 1000000, 13500000, 0, 1 },
+};
+
+struct arizona_fll_cfg {
+ int n;
+ int theta;
+ int lambda;
+ int refdiv;
+ int outdiv;
+ int fratio;
+};
+
+static int arizona_calc_fll(struct arizona_fll *fll,
+ struct arizona_fll_cfg *cfg,
+ unsigned int Fref,
+ unsigned int Fout)
+{
+ unsigned int target, div, gcd_fll;
+ int i, ratio;
+
+ arizona_fll_dbg(fll, "Fref=%u Fout=%u\n", Fref, Fout);
+
+ /* Fref must be <=13.5MHz */
+ div = 1;
+ cfg->refdiv = 0;
+ while ((Fref / div) > 13500000) {
+ div *= 2;
+ cfg->refdiv++;
+
+ if (div > 8) {
+ arizona_fll_err(fll,
+ "Can't scale %dMHz in to <=13.5MHz\n",
+ Fref);
+ return -EINVAL;
+ }
+ }
+
+ /* Apply the division for our remaining calculations */
+ Fref /= div;
+
+ /* Fvco should be over the targt; don't check the upper bound */
+ div = 1;
+ while (Fout * div < 90000000 * fll->vco_mult) {
+ div++;
+ if (div > 7) {
+ arizona_fll_err(fll, "No FLL_OUTDIV for Fout=%uHz\n",
+ Fout);
+ return -EINVAL;
+ }
+ }
+ target = Fout * div / fll->vco_mult;
+ cfg->outdiv = div;
+
+ arizona_fll_dbg(fll, "Fvco=%dHz\n", target);
+
+ /* Find an appropraite FLL_FRATIO and factor it out of the target */
+ for (i = 0; i < ARRAY_SIZE(fll_fratios); i++) {
+ if (fll_fratios[i].min <= Fref && Fref <= fll_fratios[i].max) {
+ cfg->fratio = fll_fratios[i].fratio;
+ ratio = fll_fratios[i].ratio;
+ break;
+ }
+ }
+ if (i == ARRAY_SIZE(fll_fratios)) {
+ arizona_fll_err(fll, "Unable to find FRATIO for Fref=%uHz\n",
+ Fref);
+ return -EINVAL;
+ }
+
+ cfg->n = target / (ratio * Fref);
+
+ if (target % Fref) {
+ gcd_fll = gcd(target, ratio * Fref);
+ arizona_fll_dbg(fll, "GCD=%u\n", gcd_fll);
+
+ cfg->theta = (target - (cfg->n * ratio * Fref))
+ / gcd_fll;
+ cfg->lambda = (ratio * Fref) / gcd_fll;
+ } else {
+ cfg->theta = 0;
+ cfg->lambda = 0;
+ }
+
+ arizona_fll_dbg(fll, "N=%x THETA=%x LAMBDA=%x\n",
+ cfg->n, cfg->theta, cfg->lambda);
+ arizona_fll_dbg(fll, "FRATIO=%x(%d) OUTDIV=%x REFCLK_DIV=%x\n",
+ cfg->fratio, cfg->fratio, cfg->outdiv, cfg->refdiv);
+
+ return 0;
+
+}
+
+static void arizona_apply_fll(struct arizona *arizona, unsigned int base,
+ struct arizona_fll_cfg *cfg, int source)
+{
+ regmap_update_bits(arizona->regmap, base + 3,
+ ARIZONA_FLL1_THETA_MASK, cfg->theta);
+ regmap_update_bits(arizona->regmap, base + 4,
+ ARIZONA_FLL1_LAMBDA_MASK, cfg->lambda);
+ regmap_update_bits(arizona->regmap, base + 5,
+ ARIZONA_FLL1_FRATIO_MASK,
+ cfg->fratio << ARIZONA_FLL1_FRATIO_SHIFT);
+ regmap_update_bits(arizona->regmap, base + 6,
+ ARIZONA_FLL1_CLK_REF_DIV_MASK |
+ ARIZONA_FLL1_CLK_REF_SRC_MASK,
+ cfg->refdiv << ARIZONA_FLL1_CLK_REF_DIV_SHIFT |
+ source << ARIZONA_FLL1_CLK_REF_SRC_SHIFT);
+
+ regmap_update_bits(arizona->regmap, base + 2,
+ ARIZONA_FLL1_CTRL_UPD | ARIZONA_FLL1_N_MASK,
+ ARIZONA_FLL1_CTRL_UPD | cfg->n);
+}
+
+int arizona_set_fll(struct arizona_fll *fll, int source,
+ unsigned int Fref, unsigned int Fout)
+{
+ struct arizona *arizona = fll->arizona;
+ struct arizona_fll_cfg cfg, sync;
+ unsigned int reg, val;
+ int syncsrc;
+ bool ena;
+ int ret;
+
+ ret = regmap_read(arizona->regmap, fll->base + 1, &reg);
+ if (ret != 0) {
+ arizona_fll_err(fll, "Failed to read current state: %d\n",
+ ret);
+ return ret;
+ }
+ ena = reg & ARIZONA_FLL1_ENA;
+
+ if (Fout) {
+ /* Do we have a 32kHz reference? */
+ regmap_read(arizona->regmap, ARIZONA_CLOCK_32K_1, &val);
+ switch (val & ARIZONA_CLK_32K_SRC_MASK) {
+ case ARIZONA_CLK_SRC_MCLK1:
+ case ARIZONA_CLK_SRC_MCLK2:
+ syncsrc = val & ARIZONA_CLK_32K_SRC_MASK;
+ break;
+ default:
+ syncsrc = -1;
+ }
+
+ if (source == syncsrc)
+ syncsrc = -1;
+
+ if (syncsrc >= 0) {
+ ret = arizona_calc_fll(fll, &sync, Fref, Fout);
+ if (ret != 0)
+ return ret;
+
+ ret = arizona_calc_fll(fll, &cfg, 32768, Fout);
+ if (ret != 0)
+ return ret;
+ } else {
+ ret = arizona_calc_fll(fll, &cfg, Fref, Fout);
+ if (ret != 0)
+ return ret;
+ }
+ } else {
+ regmap_update_bits(arizona->regmap, fll->base + 1,
+ ARIZONA_FLL1_ENA, 0);
+ regmap_update_bits(arizona->regmap, fll->base + 0x11,
+ ARIZONA_FLL1_SYNC_ENA, 0);
+
+ if (ena)
+ pm_runtime_put_autosuspend(arizona->dev);
+
+ return 0;
+ }
+
+ regmap_update_bits(arizona->regmap, fll->base + 5,
+ ARIZONA_FLL1_OUTDIV_MASK,
+ cfg.outdiv << ARIZONA_FLL1_OUTDIV_SHIFT);
+
+ if (syncsrc >= 0) {
+ arizona_apply_fll(arizona, fll->base, &cfg, syncsrc);
+ arizona_apply_fll(arizona, fll->base + 0x10, &sync, source);
+ } else {
+ arizona_apply_fll(arizona, fll->base, &cfg, source);
+ }
+
+ if (!ena)
+ pm_runtime_get(arizona->dev);
+
+ /* Clear any pending completions */
+ try_wait_for_completion(&fll->ok);
+
+ regmap_update_bits(arizona->regmap, fll->base + 1,
+ ARIZONA_FLL1_ENA, ARIZONA_FLL1_ENA);
+ if (syncsrc >= 0)
+ regmap_update_bits(arizona->regmap, fll->base + 0x11,
+ ARIZONA_FLL1_SYNC_ENA,
+ ARIZONA_FLL1_SYNC_ENA);
+
+ ret = wait_for_completion_timeout(&fll->ok,
+ msecs_to_jiffies(25));
+ if (ret == 0)
+ arizona_fll_warn(fll, "Timed out waiting for lock\n");
+
+ return 0;
+}
+EXPORT_SYMBOL_GPL(arizona_set_fll);
+
+int arizona_init_fll(struct arizona *arizona, int id, int base, int lock_irq,
+ int ok_irq, struct arizona_fll *fll)
+{
+ int ret;
+
+ init_completion(&fll->lock);
+ init_completion(&fll->ok);
+
+ fll->id = id;
+ fll->base = base;
+ fll->arizona = arizona;
+
+ snprintf(fll->lock_name, sizeof(fll->lock_name), "FLL%d lock", id);
+ snprintf(fll->clock_ok_name, sizeof(fll->clock_ok_name),
+ "FLL%d clock OK", id);
+
+ ret = arizona_request_irq(arizona, lock_irq, fll->lock_name,
+ arizona_fll_lock, fll);
+ if (ret != 0) {
+ dev_err(arizona->dev, "Failed to get FLL%d lock IRQ: %d\n",
+ id, ret);
+ }
+
+ ret = arizona_request_irq(arizona, ok_irq, fll->clock_ok_name,
+ arizona_fll_clock_ok, fll);
+ if (ret != 0) {
+ dev_err(arizona->dev, "Failed to get FLL%d clock OK IRQ: %d\n",
+ id, ret);
+ }
+
+ return 0;
+}
+EXPORT_SYMBOL_GPL(arizona_init_fll);
+
+MODULE_DESCRIPTION("ASoC Wolfson Arizona class device support");
+MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h
new file mode 100644
index 0000000..59caca8
--- /dev/null
+++ b/sound/soc/codecs/arizona.h
@@ -0,0 +1,159 @@
+/*
+ * arizona.h - Wolfson Arizona class device shared support
+ *
+ * Copyright 2012 Wolfson Microelectronics plc
+ *
+ * Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
+ *
+ * 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 _ASOC_ARIZONA_H
+#define _ASOC_ARIZONA_H
+
+#include <linux/completion.h>
+
+#include <sound/soc.h>
+
+#define ARIZONA_CLK_SYSCLK 1
+#define ARIZONA_CLK_ASYNCCLK 2
+
+#define ARIZONA_CLK_SRC_MCLK1 0x0
+#define ARIZONA_CLK_SRC_MCLK2 0x1
+#define ARIZONA_CLK_SRC_FLL1 0x4
+#define ARIZONA_CLK_SRC_FLL2 0x5
+#define ARIZONA_CLK_SRC_AIF1BCLK 0x8
+#define ARIZONA_CLK_SRC_AIF2BCLK 0x9
+#define ARIZONA_CLK_SRC_AIF3BCLK 0xa
+
+#define ARIZONA_FLL_SRC_MCLK1 0
+#define ARIZONA_FLL_SRC_MCLK2 1
+#define ARIZONA_FLL_SRC_SLIMCLK 2
+#define ARIZONA_FLL_SRC_FLL1 3
+#define ARIZONA_FLL_SRC_FLL2 4
+#define ARIZONA_FLL_SRC_AIF1BCLK 5
+#define ARIZONA_FLL_SRC_AIF2BCLK 6
+#define ARIZONA_FLL_SRC_AIF3BCLK 7
+#define ARIZONA_FLL_SRC_AIF1LRCLK 8
+#define ARIZONA_FLL_SRC_AIF2LRCLK 9
+#define ARIZONA_FLL_SRC_AIF3LRCLK 10
+
+#define ARIZONA_MIXER_VOL_MASK 0x00FE
+#define ARIZONA_MIXER_VOL_SHIFT 1
+#define ARIZONA_MIXER_VOL_WIDTH 7
+
+#define ARIZONA_MAX_DAI 3
+
+struct arizona;
+
+struct arizona_dai_priv {
+ int clk;
+};
+
+struct arizona_priv {
+ struct arizona *arizona;
+ int sysclk;
+ int asyncclk;
+ struct arizona_dai_priv dai[ARIZONA_MAX_DAI];
+};
+
+#define ARIZONA_NUM_MIXER_INPUTS 57
+
+extern const unsigned int arizona_mixer_tlv[];
+extern const char *arizona_mixer_texts[ARIZONA_NUM_MIXER_INPUTS];
+extern int arizona_mixer_values[ARIZONA_NUM_MIXER_INPUTS];
+
+#define ARIZONA_MIXER_CONTROLS(name, base) \
+ SOC_SINGLE_RANGE_TLV(name " Input 1 Volume", base + 1, \
+ ARIZONA_MIXER_VOL_SHIFT, 0x20, 0x50, 0, \
+ arizona_mixer_tlv), \
+ SOC_SINGLE_RANGE_TLV(name " Input 2 Volume", base + 3, \
+ ARIZONA_MIXER_VOL_SHIFT, 0x20, 0x50, 0, \
+ arizona_mixer_tlv), \
+ SOC_SINGLE_RANGE_TLV(name " Input 3 Volume", base + 5, \
+ ARIZONA_MIXER_VOL_SHIFT, 0x20, 0x50, 0, \
+ arizona_mixer_tlv), \
+ SOC_SINGLE_RANGE_TLV(name " Input 4 Volume", base + 7, \
+ ARIZONA_MIXER_VOL_SHIFT, 0x20, 0x50, 0, \
+ arizona_mixer_tlv)
+
+#define ARIZONA_MUX_ENUM_DECL(name, reg) \
+ SOC_VALUE_ENUM_SINGLE_DECL(name, reg, 0, 0xff, \
+ arizona_mixer_texts, arizona_mixer_values)
+
+#define ARIZONA_MUX_CTL_DECL(name) \
+ const struct snd_kcontrol_new name##_mux = \
+ SOC_DAPM_VALUE_ENUM("Route", name##_enum)
+
+#define ARIZONA_MIXER_ENUMS(name, base_reg) \
+ static ARIZONA_MUX_ENUM_DECL(name##_in1_enum, base_reg); \
+ static ARIZONA_MUX_ENUM_DECL(name##_in2_enum, base_reg + 2); \
+ static ARIZONA_MUX_ENUM_DECL(name##_in3_enum, base_reg + 4); \
+ static ARIZONA_MUX_ENUM_DECL(name##_in4_enum, base_reg + 6); \
+ static ARIZONA_MUX_CTL_DECL(name##_in1); \
+ static ARIZONA_MUX_CTL_DECL(name##_in2); \
+ static ARIZONA_MUX_CTL_DECL(name##_in3); \
+ static ARIZONA_MUX_CTL_DECL(name##_in4)
+
+#define ARIZONA_MUX(name, ctrl) \
+ SND_SOC_DAPM_VALUE_MUX(name, SND_SOC_NOPM, 0, 0, ctrl)
+
+#define ARIZONA_MIXER_WIDGETS(name, name_str) \
+ ARIZONA_MUX(name_str " Input 1", &name##_in1_mux), \
+ ARIZONA_MUX(name_str " Input 2", &name##_in2_mux), \
+ ARIZONA_MUX(name_str " Input 3", &name##_in3_mux), \
+ ARIZONA_MUX(name_str " Input 4", &name##_in4_mux), \
+ SND_SOC_DAPM_MIXER(name_str " Mixer", SND_SOC_NOPM, 0, 0, NULL, 0)
+
+#define ARIZONA_MIXER_ROUTES(widget, name) \
+ { widget, NULL, name " Mixer" }, \
+ { name " Mixer", NULL, name " Input 1" }, \
+ { name " Mixer", NULL, name " Input 2" }, \
+ { name " Mixer", NULL, name " Input 3" }, \
+ { name " Mixer", NULL, name " Input 4" }, \
+ ARIZONA_MIXER_INPUT_ROUTES(name " Input 1"), \
+ ARIZONA_MIXER_INPUT_ROUTES(name " Input 2"), \
+ ARIZONA_MIXER_INPUT_ROUTES(name " Input 3"), \
+ ARIZONA_MIXER_INPUT_ROUTES(name " Input 4")
+
+extern const struct soc_enum arizona_lhpf1_mode;
+extern const struct soc_enum arizona_lhpf2_mode;
+extern const struct soc_enum arizona_lhpf3_mode;
+extern const struct soc_enum arizona_lhpf4_mode;
+
+extern int arizona_in_ev(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol,
+ int event);
+extern int arizona_out_ev(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol,
+ int event);
+
+extern int arizona_set_sysclk(struct snd_soc_codec *codec, int clk_id,
+ int source, unsigned int freq, int dir);
+
+extern const struct snd_soc_dai_ops arizona_dai_ops;
+
+#define ARIZONA_FLL_NAME_LEN 20
+
+struct arizona_fll {
+ struct arizona *arizona;
+ int id;
+ unsigned int base;
+ unsigned int vco_mult;
+ struct completion lock;
+ struct completion ok;
+
+ char lock_name[ARIZONA_FLL_NAME_LEN];
+ char clock_ok_name[ARIZONA_FLL_NAME_LEN];
+};
+
+extern int arizona_init_fll(struct arizona *arizona, int id, int base,
+ int lock_irq, int ok_irq, struct arizona_fll *fll);
+extern int arizona_set_fll(struct arizona_fll *fll, int source,
+ unsigned int Fref, unsigned int Fout);
+
+extern int arizona_init_dai(struct arizona_priv *priv, int dai);
+
+#endif
diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c
index a710941..628daf6 100644
--- a/sound/soc/codecs/cs42l52.c
+++ b/sound/soc/codecs/cs42l52.c
@@ -14,7 +14,6 @@
#include <linux/module.h>
#include <linux/moduleparam.h>
-#include <linux/version.h>
#include <linux/kernel.h>
#include <linux/init.h>
#include <linux/delay.h>
@@ -1217,11 +1216,11 @@ static int cs42l52_i2c_probe(struct i2c_client *i2c_client,
return -ENOMEM;
cs42l52->dev = &i2c_client->dev;
- cs42l52->regmap = regmap_init_i2c(i2c_client, &cs42l52_regmap);
+ cs42l52->regmap = devm_regmap_init_i2c(i2c_client, &cs42l52_regmap);
if (IS_ERR(cs42l52->regmap)) {
ret = PTR_ERR(cs42l52->regmap);
dev_err(&i2c_client->dev, "regmap_init() failed: %d\n", ret);
- goto err;
+ return ret;
}
i2c_set_clientdata(i2c_client, cs42l52);
@@ -1243,7 +1242,7 @@ static int cs42l52_i2c_probe(struct i2c_client *i2c_client,
dev_err(&i2c_client->dev,
"CS42L52 Device ID (%X). Expected %X\n",
devid, CS42L52_CHIP_ID);
- goto err_regmap;
+ return ret;
}
regcache_cache_only(cs42l52->regmap, true);
@@ -1251,23 +1250,13 @@ static int cs42l52_i2c_probe(struct i2c_client *i2c_client,
ret = snd_soc_register_codec(&i2c_client->dev,
&soc_codec_dev_cs42l52, &cs42l52_dai, 1);
if (ret < 0)
- goto err_regmap;
+ return ret;
return 0;
-
-err_regmap:
- regmap_exit(cs42l52->regmap);
-
-err:
- return ret;
}
static int cs42l52_i2c_remove(struct i2c_client *client)
{
- struct cs42l52_private *cs42l52 = i2c_get_clientdata(client);
-
snd_soc_unregister_codec(&client->dev);
- regmap_exit(cs42l52->regmap);
-
return 0;
}
diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c
index e0d45fd..2c08c4c 100644
--- a/sound/soc/codecs/cs42l73.c
+++ b/sound/soc/codecs/cs42l73.c
@@ -1362,11 +1362,11 @@ static __devinit int cs42l73_i2c_probe(struct i2c_client *i2c_client,
i2c_set_clientdata(i2c_client, cs42l73);
- cs42l73->regmap = regmap_init_i2c(i2c_client, &cs42l73_regmap);
+ cs42l73->regmap = devm_regmap_init_i2c(i2c_client, &cs42l73_regmap);
if (IS_ERR(cs42l73->regmap)) {
ret = PTR_ERR(cs42l73->regmap);
dev_err(&i2c_client->dev, "regmap_init() failed: %d\n", ret);
- goto err;
+ return ret;
}
/* initialize codec */
ret = regmap_read(cs42l73->regmap, CS42L73_DEVID_AB, &reg);
@@ -1384,13 +1384,13 @@ static __devinit int cs42l73_i2c_probe(struct i2c_client *i2c_client,
dev_err(&i2c_client->dev,
"CS42L73 Device ID (%X). Expected %X\n",
devid, CS42L73_DEVID);
- goto err_regmap;
+ return ret;
}
ret = regmap_read(cs42l73->regmap, CS42L73_REVID, &reg);
if (ret < 0) {
dev_err(&i2c_client->dev, "Get Revision ID failed\n");
- goto err_regmap;
+ return ret;;
}
dev_info(&i2c_client->dev,
@@ -1402,23 +1402,13 @@ static __devinit int cs42l73_i2c_probe(struct i2c_client *i2c_client,
&soc_codec_dev_cs42l73, cs42l73_dai,
ARRAY_SIZE(cs42l73_dai));
if (ret < 0)
- goto err_regmap;
+ return ret;
return 0;
-
-err_regmap:
- regmap_exit(cs42l73->regmap);
-
-err:
- return ret;
}
static __devexit int cs42l73_i2c_remove(struct i2c_client *client)
{
- struct cs42l73_private *cs42l73 = i2c_get_clientdata(client);
-
snd_soc_unregister_codec(&client->dev);
- regmap_exit(cs42l73->regmap);
-
return 0;
}
diff --git a/sound/soc/codecs/da732x.c b/sound/soc/codecs/da732x.c
new file mode 100644
index 0000000..01be2a3
--- /dev/null
+++ b/sound/soc/codecs/da732x.c
@@ -0,0 +1,1627 @@
+/*
+ * da732x.c --- Dialog DA732X ALSA SoC Audio Driver
+ *
+ * Copyright (C) 2012 Dialog Semiconductor GmbH
+ *
+ * Author: Michal Hajduk <Michal.Hajduk@diasemi.com>
+ *
+ * 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 <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/i2c.h>
+#include <linux/regmap.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+#include <linux/sysfs.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/initval.h>
+#include <sound/tlv.h>
+#include <asm/div64.h>
+
+#include "da732x.h"
+#include "da732x_reg.h"
+
+
+struct da732x_priv {
+ struct regmap *regmap;
+ struct snd_soc_codec *codec;
+
+ unsigned int sysclk;
+ bool pll_en;
+};
+
+/*
+ * da732x register cache - default settings
+ */
+static struct reg_default da732x_reg_cache[] = {
+ { DA732X_REG_REF1 , 0x02 },
+ { DA732X_REG_BIAS_EN , 0x80 },
+ { DA732X_REG_BIAS1 , 0x00 },
+ { DA732X_REG_BIAS2 , 0x00 },
+ { DA732X_REG_BIAS3 , 0x00 },
+ { DA732X_REG_BIAS4 , 0x00 },
+ { DA732X_REG_MICBIAS2 , 0x00 },
+ { DA732X_REG_MICBIAS1 , 0x00 },
+ { DA732X_REG_MICDET , 0x00 },
+ { DA732X_REG_MIC1_PRE , 0x01 },
+ { DA732X_REG_MIC1 , 0x40 },
+ { DA732X_REG_MIC2_PRE , 0x01 },
+ { DA732X_REG_MIC2 , 0x40 },
+ { DA732X_REG_AUX1L , 0x75 },
+ { DA732X_REG_AUX1R , 0x75 },
+ { DA732X_REG_MIC3_PRE , 0x01 },
+ { DA732X_REG_MIC3 , 0x40 },
+ { DA732X_REG_INP_PINBIAS , 0x00 },
+ { DA732X_REG_INP_ZC_EN , 0x00 },
+ { DA732X_REG_INP_MUX , 0x50 },
+ { DA732X_REG_HP_DET , 0x00 },
+ { DA732X_REG_HPL_DAC_OFFSET , 0x00 },
+ { DA732X_REG_HPL_DAC_OFF_CNTL , 0x00 },
+ { DA732X_REG_HPL_OUT_OFFSET , 0x00 },
+ { DA732X_REG_HPL , 0x40 },
+ { DA732X_REG_HPL_VOL , 0x0F },
+ { DA732X_REG_HPR_DAC_OFFSET , 0x00 },
+ { DA732X_REG_HPR_DAC_OFF_CNTL , 0x00 },
+ { DA732X_REG_HPR_OUT_OFFSET , 0x00 },
+ { DA732X_REG_HPR , 0x40 },
+ { DA732X_REG_HPR_VOL , 0x0F },
+ { DA732X_REG_LIN2 , 0x4F },
+ { DA732X_REG_LIN3 , 0x4F },
+ { DA732X_REG_LIN4 , 0x4F },
+ { DA732X_REG_OUT_ZC_EN , 0x00 },
+ { DA732X_REG_HP_LIN1_GNDSEL , 0x00 },
+ { DA732X_REG_CP_HP1 , 0x0C },
+ { DA732X_REG_CP_HP2 , 0x03 },
+ { DA732X_REG_CP_CTRL1 , 0x00 },
+ { DA732X_REG_CP_CTRL2 , 0x99 },
+ { DA732X_REG_CP_CTRL3 , 0x25 },
+ { DA732X_REG_CP_LEVEL_MASK , 0x3F },
+ { DA732X_REG_CP_DET , 0x00 },
+ { DA732X_REG_CP_STATUS , 0x00 },
+ { DA732X_REG_CP_THRESH1 , 0x00 },
+ { DA732X_REG_CP_THRESH2 , 0x00 },
+ { DA732X_REG_CP_THRESH3 , 0x00 },
+ { DA732X_REG_CP_THRESH4 , 0x00 },
+ { DA732X_REG_CP_THRESH5 , 0x00 },
+ { DA732X_REG_CP_THRESH6 , 0x00 },
+ { DA732X_REG_CP_THRESH7 , 0x00 },
+ { DA732X_REG_CP_THRESH8 , 0x00 },
+ { DA732X_REG_PLL_DIV_LO , 0x00 },
+ { DA732X_REG_PLL_DIV_MID , 0x00 },
+ { DA732X_REG_PLL_DIV_HI , 0x00 },
+ { DA732X_REG_PLL_CTRL , 0x02 },
+ { DA732X_REG_CLK_CTRL , 0xaa },
+ { DA732X_REG_CLK_DSP , 0x07 },
+ { DA732X_REG_CLK_EN1 , 0x00 },
+ { DA732X_REG_CLK_EN2 , 0x00 },
+ { DA732X_REG_CLK_EN3 , 0x00 },
+ { DA732X_REG_CLK_EN4 , 0x00 },
+ { DA732X_REG_CLK_EN5 , 0x00 },
+ { DA732X_REG_AIF_MCLK , 0x00 },
+ { DA732X_REG_AIFA1 , 0x02 },
+ { DA732X_REG_AIFA2 , 0x00 },
+ { DA732X_REG_AIFA3 , 0x08 },
+ { DA732X_REG_AIFB1 , 0x02 },
+ { DA732X_REG_AIFB2 , 0x00 },
+ { DA732X_REG_AIFB3 , 0x08 },
+ { DA732X_REG_PC_CTRL , 0xC0 },
+ { DA732X_REG_DATA_ROUTE , 0x00 },
+ { DA732X_REG_DSP_CTRL , 0x00 },
+ { DA732X_REG_CIF_CTRL2 , 0x00 },
+ { DA732X_REG_HANDSHAKE , 0x00 },
+ { DA732X_REG_SPARE1_OUT , 0x00 },
+ { DA732X_REG_SPARE2_OUT , 0x00 },
+ { DA732X_REG_SPARE1_IN , 0x00 },
+ { DA732X_REG_ADC1_PD , 0x00 },
+ { DA732X_REG_ADC1_HPF , 0x00 },
+ { DA732X_REG_ADC1_SEL , 0x00 },
+ { DA732X_REG_ADC1_EQ12 , 0x00 },
+ { DA732X_REG_ADC1_EQ34 , 0x00 },
+ { DA732X_REG_ADC1_EQ5 , 0x00 },
+ { DA732X_REG_ADC2_PD , 0x00 },
+ { DA732X_REG_ADC2_HPF , 0x00 },
+ { DA732X_REG_ADC2_SEL , 0x00 },
+ { DA732X_REG_ADC2_EQ12 , 0x00 },
+ { DA732X_REG_ADC2_EQ34 , 0x00 },
+ { DA732X_REG_ADC2_EQ5 , 0x00 },
+ { DA732X_REG_DAC1_HPF , 0x00 },
+ { DA732X_REG_DAC1_L_VOL , 0x00 },
+ { DA732X_REG_DAC1_R_VOL , 0x00 },
+ { DA732X_REG_DAC1_SEL , 0x00 },
+ { DA732X_REG_DAC1_SOFTMUTE , 0x00 },
+ { DA732X_REG_DAC1_EQ12 , 0x00 },
+ { DA732X_REG_DAC1_EQ34 , 0x00 },
+ { DA732X_REG_DAC1_EQ5 , 0x00 },
+ { DA732X_REG_DAC2_HPF , 0x00 },
+ { DA732X_REG_DAC2_L_VOL , 0x00 },
+ { DA732X_REG_DAC2_R_VOL , 0x00 },
+ { DA732X_REG_DAC2_SEL , 0x00 },
+ { DA732X_REG_DAC2_SOFTMUTE , 0x00 },
+ { DA732X_REG_DAC2_EQ12 , 0x00 },
+ { DA732X_REG_DAC2_EQ34 , 0x00 },
+ { DA732X_REG_DAC2_EQ5 , 0x00 },
+ { DA732X_REG_DAC3_HPF , 0x00 },
+ { DA732X_REG_DAC3_VOL , 0x00 },
+ { DA732X_REG_DAC3_SEL , 0x00 },
+ { DA732X_REG_DAC3_SOFTMUTE , 0x00 },
+ { DA732X_REG_DAC3_EQ12 , 0x00 },
+ { DA732X_REG_DAC3_EQ34 , 0x00 },
+ { DA732X_REG_DAC3_EQ5 , 0x00 },
+ { DA732X_REG_BIQ_BYP , 0x00 },
+ { DA732X_REG_DMA_CMD , 0x00 },
+ { DA732X_REG_DMA_ADDR0 , 0x00 },
+ { DA732X_REG_DMA_ADDR1 , 0x00 },
+ { DA732X_REG_DMA_DATA0 , 0x00 },
+ { DA732X_REG_DMA_DATA1 , 0x00 },
+ { DA732X_REG_DMA_DATA2 , 0x00 },
+ { DA732X_REG_DMA_DATA3 , 0x00 },
+ { DA732X_REG_UNLOCK , 0x00 },
+};
+
+static inline int da732x_get_input_div(struct snd_soc_codec *codec, int sysclk)
+{
+ int val;
+ int ret;
+
+ if (sysclk < DA732X_MCLK_10MHZ) {
+ val = DA732X_MCLK_RET_0_10MHZ;
+ ret = DA732X_MCLK_VAL_0_10MHZ;
+ } else if ((sysclk >= DA732X_MCLK_10MHZ) &&
+ (sysclk < DA732X_MCLK_20MHZ)) {
+ val = DA732X_MCLK_RET_10_20MHZ;
+ ret = DA732X_MCLK_VAL_10_20MHZ;
+ } else if ((sysclk >= DA732X_MCLK_20MHZ) &&
+ (sysclk < DA732X_MCLK_40MHZ)) {
+ val = DA732X_MCLK_RET_20_40MHZ;
+ ret = DA732X_MCLK_VAL_20_40MHZ;
+ } else if ((sysclk >= DA732X_MCLK_40MHZ) &&
+ (sysclk <= DA732X_MCLK_54MHZ)) {
+ val = DA732X_MCLK_RET_40_54MHZ;
+ ret = DA732X_MCLK_VAL_40_54MHZ;
+ } else {
+ return -EINVAL;
+ }
+
+ snd_soc_write(codec, DA732X_REG_PLL_CTRL, val);
+
+ return ret;
+}
+
+static void da732x_set_charge_pump(struct snd_soc_codec *codec, int state)
+{
+ switch (state) {
+ case DA732X_ENABLE_CP:
+ snd_soc_write(codec, DA732X_REG_CLK_EN2, DA732X_CP_CLK_EN);
+ snd_soc_write(codec, DA732X_REG_CP_HP2, DA732X_HP_CP_EN |
+ DA732X_HP_CP_REG | DA732X_HP_CP_PULSESKIP);
+ snd_soc_write(codec, DA732X_REG_CP_CTRL1, DA732X_CP_EN |
+ DA732X_CP_CTRL_CPVDD1);
+ snd_soc_write(codec, DA732X_REG_CP_CTRL2,
+ DA732X_CP_MANAGE_MAGNITUDE | DA732X_CP_BOOST);
+ snd_soc_write(codec, DA732X_REG_CP_CTRL3, DA732X_CP_1MHZ);
+ break;
+ case DA732X_DISABLE_CP:
+ snd_soc_write(codec, DA732X_REG_CLK_EN2, DA732X_CP_CLK_DIS);
+ snd_soc_write(codec, DA732X_REG_CP_HP2, DA732X_HP_CP_DIS);
+ snd_soc_write(codec, DA732X_REG_CP_CTRL1, DA723X_CP_DIS);
+ break;
+ default:
+ pr_err(KERN_ERR "Wrong charge pump state\n");
+ break;
+ }
+}
+
+static const DECLARE_TLV_DB_SCALE(mic_boost_tlv, DA732X_MIC_PRE_VOL_DB_MIN,
+ DA732X_MIC_PRE_VOL_DB_INC, 0);
+
+static const DECLARE_TLV_DB_SCALE(mic_pga_tlv, DA732X_MIC_VOL_DB_MIN,
+ DA732X_MIC_VOL_DB_INC, 0);
+
+static const DECLARE_TLV_DB_SCALE(aux_pga_tlv, DA732X_AUX_VOL_DB_MIN,
+ DA732X_AUX_VOL_DB_INC, 0);
+
+static const DECLARE_TLV_DB_SCALE(hp_pga_tlv, DA732X_HP_VOL_DB_MIN,
+ DA732X_AUX_VOL_DB_INC, 0);
+
+static const DECLARE_TLV_DB_SCALE(lin2_pga_tlv, DA732X_LIN2_VOL_DB_MIN,
+ DA732X_LIN2_VOL_DB_INC, 0);
+
+static const DECLARE_TLV_DB_SCALE(lin3_pga_tlv, DA732X_LIN3_VOL_DB_MIN,
+ DA732X_LIN3_VOL_DB_INC, 0);
+
+static const DECLARE_TLV_DB_SCALE(lin4_pga_tlv, DA732X_LIN4_VOL_DB_MIN,
+ DA732X_LIN4_VOL_DB_INC, 0);
+
+static const DECLARE_TLV_DB_SCALE(adc_pga_tlv, DA732X_ADC_VOL_DB_MIN,
+ DA732X_ADC_VOL_DB_INC, 0);
+
+static const DECLARE_TLV_DB_SCALE(dac_pga_tlv, DA732X_DAC_VOL_DB_MIN,
+ DA732X_DAC_VOL_DB_INC, 0);
+
+static const DECLARE_TLV_DB_SCALE(eq_band_pga_tlv, DA732X_EQ_BAND_VOL_DB_MIN,
+ DA732X_EQ_BAND_VOL_DB_INC, 0);
+
+static const DECLARE_TLV_DB_SCALE(eq_overall_tlv, DA732X_EQ_OVERALL_VOL_DB_MIN,
+ DA732X_EQ_OVERALL_VOL_DB_INC, 0);
+
+/* High Pass Filter */
+static const char *da732x_hpf_mode[] = {
+ "Disable", "Music", "Voice",
+};
+
+static const char *da732x_hpf_music[] = {
+ "1.8Hz", "3.75Hz", "7.5Hz", "15Hz",
+};
+
+static const char *da732x_hpf_voice[] = {
+ "2.5Hz", "25Hz", "50Hz", "100Hz",
+ "150Hz", "200Hz", "300Hz", "400Hz"
+};
+
+static const struct soc_enum da732x_dac1_hpf_mode_enum[] = {
+ SOC_ENUM_SINGLE(DA732X_REG_DAC1_HPF, DA732X_HPF_MODE_SHIFT,
+ DA732X_HPF_MODE_MAX, da732x_hpf_mode)
+};
+
+static const struct soc_enum da732x_dac2_hpf_mode_enum[] = {
+ SOC_ENUM_SINGLE(DA732X_REG_DAC2_HPF, DA732X_HPF_MODE_SHIFT,
+ DA732X_HPF_MODE_MAX, da732x_hpf_mode)
+};
+
+static const struct soc_enum da732x_dac3_hpf_mode_enum[] = {
+ SOC_ENUM_SINGLE(DA732X_REG_DAC3_HPF, DA732X_HPF_MODE_SHIFT,
+ DA732X_HPF_MODE_MAX, da732x_hpf_mode)
+};
+
+static const struct soc_enum da732x_adc1_hpf_mode_enum[] = {
+ SOC_ENUM_SINGLE(DA732X_REG_ADC1_HPF, DA732X_HPF_MODE_SHIFT,
+ DA732X_HPF_MODE_MAX, da732x_hpf_mode)
+};
+
+static const struct soc_enum da732x_adc2_hpf_mode_enum[] = {
+ SOC_ENUM_SINGLE(DA732X_REG_ADC2_HPF, DA732X_HPF_MODE_SHIFT,
+ DA732X_HPF_MODE_MAX, da732x_hpf_mode)
+};
+
+static const struct soc_enum da732x_dac1_hp_filter_enum[] = {
+ SOC_ENUM_SINGLE(DA732X_REG_DAC1_HPF, DA732X_HPF_MUSIC_SHIFT,
+ DA732X_HPF_MUSIC_MAX, da732x_hpf_music)
+};
+
+static const struct soc_enum da732x_dac2_hp_filter_enum[] = {
+ SOC_ENUM_SINGLE(DA732X_REG_DAC2_HPF, DA732X_HPF_MUSIC_SHIFT,
+ DA732X_HPF_MUSIC_MAX, da732x_hpf_music)
+};
+
+static const struct soc_enum da732x_dac3_hp_filter_enum[] = {
+ SOC_ENUM_SINGLE(DA732X_REG_DAC3_HPF, DA732X_HPF_MUSIC_SHIFT,
+ DA732X_HPF_MUSIC_MAX, da732x_hpf_music)
+};
+
+static const struct soc_enum da732x_adc1_hp_filter_enum[] = {
+ SOC_ENUM_SINGLE(DA732X_REG_ADC1_HPF, DA732X_HPF_MUSIC_SHIFT,
+ DA732X_HPF_MUSIC_MAX, da732x_hpf_music)
+};
+
+static const struct soc_enum da732x_adc2_hp_filter_enum[] = {
+ SOC_ENUM_SINGLE(DA732X_REG_ADC2_HPF, DA732X_HPF_MUSIC_SHIFT,
+ DA732X_HPF_MUSIC_MAX, da732x_hpf_music)
+};
+
+static const struct soc_enum da732x_dac1_voice_filter_enum[] = {
+ SOC_ENUM_SINGLE(DA732X_REG_DAC1_HPF, DA732X_HPF_VOICE_SHIFT,
+ DA732X_HPF_VOICE_MAX, da732x_hpf_voice)
+};
+
+static const struct soc_enum da732x_dac2_voice_filter_enum[] = {
+ SOC_ENUM_SINGLE(DA732X_REG_DAC2_HPF, DA732X_HPF_VOICE_SHIFT,
+ DA732X_HPF_VOICE_MAX, da732x_hpf_voice)
+};
+
+static const struct soc_enum da732x_dac3_voice_filter_enum[] = {
+ SOC_ENUM_SINGLE(DA732X_REG_DAC3_HPF, DA732X_HPF_VOICE_SHIFT,
+ DA732X_HPF_VOICE_MAX, da732x_hpf_voice)
+};
+
+static const struct soc_enum da732x_adc1_voice_filter_enum[] = {
+ SOC_ENUM_SINGLE(DA732X_REG_ADC1_HPF, DA732X_HPF_VOICE_SHIFT,
+ DA732X_HPF_VOICE_MAX, da732x_hpf_voice)
+};
+
+static const struct soc_enum da732x_adc2_voice_filter_enum[] = {
+ SOC_ENUM_SINGLE(DA732X_REG_ADC2_HPF, DA732X_HPF_VOICE_SHIFT,
+ DA732X_HPF_VOICE_MAX, da732x_hpf_voice)
+};
+
+
+static int da732x_hpf_set(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct soc_enum *enum_ctrl = (struct soc_enum *)kcontrol->private_value;
+ unsigned int reg = enum_ctrl->reg;
+ unsigned int sel = ucontrol->value.integer.value[0];
+ unsigned int bits;
+
+ switch (sel) {
+ case DA732X_HPF_DISABLED:
+ bits = DA732X_HPF_DIS;
+ break;
+ case DA732X_HPF_VOICE:
+ bits = DA732X_HPF_VOICE_EN;
+ break;
+ case DA732X_HPF_MUSIC:
+ bits = DA732X_HPF_MUSIC_EN;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ snd_soc_update_bits(codec, reg, DA732X_HPF_MASK, bits);
+
+ return 0;
+}
+
+static int da732x_hpf_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct soc_enum *enum_ctrl = (struct soc_enum *)kcontrol->private_value;
+ unsigned int reg = enum_ctrl->reg;
+ int val;
+
+ val = snd_soc_read(codec, reg) & DA732X_HPF_MASK;
+
+ switch (val) {
+ case DA732X_HPF_VOICE_EN:
+ ucontrol->value.integer.value[0] = DA732X_HPF_VOICE;
+ break;
+ case DA732X_HPF_MUSIC_EN:
+ ucontrol->value.integer.value[0] = DA732X_HPF_MUSIC;
+ break;
+ default:
+ ucontrol->value.integer.value[0] = DA732X_HPF_DISABLED;
+ break;
+ }
+
+ return 0;
+}
+
+static const struct snd_kcontrol_new da732x_snd_controls[] = {
+ /* Input PGAs */
+ SOC_SINGLE_RANGE_TLV("MIC1 Boost Volume", DA732X_REG_MIC1_PRE,
+ DA732X_MICBOOST_SHIFT, DA732X_MICBOOST_MIN,
+ DA732X_MICBOOST_MAX, 0, mic_boost_tlv),
+ SOC_SINGLE_RANGE_TLV("MIC2 Boost Volume", DA732X_REG_MIC2_PRE,
+ DA732X_MICBOOST_SHIFT, DA732X_MICBOOST_MIN,
+ DA732X_MICBOOST_MAX, 0, mic_boost_tlv),
+ SOC_SINGLE_RANGE_TLV("MIC3 Boost Volume", DA732X_REG_MIC3_PRE,
+ DA732X_MICBOOST_SHIFT, DA732X_MICBOOST_MIN,
+ DA732X_MICBOOST_MAX, 0, mic_boost_tlv),
+
+ /* MICs */
+ SOC_SINGLE("MIC1 Switch", DA732X_REG_MIC1, DA732X_MIC_MUTE_SHIFT,
+ DA732X_SWITCH_MAX, DA732X_INVERT),
+ SOC_SINGLE_RANGE_TLV("MIC1 Volume", DA732X_REG_MIC1,
+ DA732X_MIC_VOL_SHIFT, DA732X_MIC_VOL_VAL_MIN,
+ DA732X_MIC_VOL_VAL_MAX, 0, mic_pga_tlv),
+ SOC_SINGLE("MIC2 Switch", DA732X_REG_MIC2, DA732X_MIC_MUTE_SHIFT,
+ DA732X_SWITCH_MAX, DA732X_INVERT),
+ SOC_SINGLE_RANGE_TLV("MIC2 Volume", DA732X_REG_MIC2,
+ DA732X_MIC_VOL_SHIFT, DA732X_MIC_VOL_VAL_MIN,
+ DA732X_MIC_VOL_VAL_MAX, 0, mic_pga_tlv),
+ SOC_SINGLE("MIC3 Switch", DA732X_REG_MIC3, DA732X_MIC_MUTE_SHIFT,
+ DA732X_SWITCH_MAX, DA732X_INVERT),
+ SOC_SINGLE_RANGE_TLV("MIC3 Volume", DA732X_REG_MIC3,
+ DA732X_MIC_VOL_SHIFT, DA732X_MIC_VOL_VAL_MIN,
+ DA732X_MIC_VOL_VAL_MAX, 0, mic_pga_tlv),
+
+ /* AUXs */
+ SOC_SINGLE("AUX1L Switch", DA732X_REG_AUX1L, DA732X_AUX_MUTE_SHIFT,
+ DA732X_SWITCH_MAX, DA732X_INVERT),
+ SOC_SINGLE_TLV("AUX1L Volume", DA732X_REG_AUX1L,
+ DA732X_AUX_VOL_SHIFT, DA732X_AUX_VOL_VAL_MAX,
+ DA732X_NO_INVERT, aux_pga_tlv),
+ SOC_SINGLE("AUX1R Switch", DA732X_REG_AUX1R, DA732X_AUX_MUTE_SHIFT,
+ DA732X_SWITCH_MAX, DA732X_INVERT),
+ SOC_SINGLE_TLV("AUX1R Volume", DA732X_REG_AUX1R,
+ DA732X_AUX_VOL_SHIFT, DA732X_AUX_VOL_VAL_MAX,
+ DA732X_NO_INVERT, aux_pga_tlv),
+
+ /* ADCs */
+ SOC_DOUBLE_TLV("ADC1 Volume", DA732X_REG_ADC1_SEL,
+ DA732X_ADCL_VOL_SHIFT, DA732X_ADCR_VOL_SHIFT,
+ DA732X_ADC_VOL_VAL_MAX, DA732X_INVERT, adc_pga_tlv),
+
+ SOC_DOUBLE_TLV("ADC2 Volume", DA732X_REG_ADC2_SEL,
+ DA732X_ADCL_VOL_SHIFT, DA732X_ADCR_VOL_SHIFT,
+ DA732X_ADC_VOL_VAL_MAX, DA732X_INVERT, adc_pga_tlv),
+
+ /* DACs */
+ SOC_DOUBLE("Digital Playback DAC12 Switch", DA732X_REG_DAC1_SEL,
+ DA732X_DACL_MUTE_SHIFT, DA732X_DACR_MUTE_SHIFT,
+ DA732X_SWITCH_MAX, DA732X_INVERT),
+ SOC_DOUBLE_R_TLV("Digital Playback DAC12 Volume", DA732X_REG_DAC1_L_VOL,
+ DA732X_REG_DAC1_R_VOL, DA732X_DAC_VOL_SHIFT,
+ DA732X_DAC_VOL_VAL_MAX, DA732X_INVERT, dac_pga_tlv),
+ SOC_SINGLE("Digital Playback DAC3 Switch", DA732X_REG_DAC2_SEL,
+ DA732X_DACL_MUTE_SHIFT, DA732X_SWITCH_MAX, DA732X_INVERT),
+ SOC_SINGLE_TLV("Digital Playback DAC3 Volume", DA732X_REG_DAC2_L_VOL,
+ DA732X_DAC_VOL_SHIFT, DA732X_DAC_VOL_VAL_MAX,
+ DA732X_INVERT, dac_pga_tlv),
+ SOC_SINGLE("Digital Playback DAC4 Switch", DA732X_REG_DAC2_SEL,
+ DA732X_DACR_MUTE_SHIFT, DA732X_SWITCH_MAX, DA732X_INVERT),
+ SOC_SINGLE_TLV("Digital Playback DAC4 Volume", DA732X_REG_DAC2_R_VOL,
+ DA732X_DAC_VOL_SHIFT, DA732X_DAC_VOL_VAL_MAX,
+ DA732X_INVERT, dac_pga_tlv),
+ SOC_SINGLE("Digital Playback DAC5 Switch", DA732X_REG_DAC3_SEL,
+ DA732X_DACL_MUTE_SHIFT, DA732X_SWITCH_MAX, DA732X_INVERT),
+ SOC_SINGLE_TLV("Digital Playback DAC5 Volume", DA732X_REG_DAC3_VOL,
+ DA732X_DAC_VOL_SHIFT, DA732X_DAC_VOL_VAL_MAX,
+ DA732X_INVERT, dac_pga_tlv),
+
+ /* High Pass Filters */
+ SOC_ENUM_EXT("DAC1 High Pass Filter Mode",
+ da732x_dac1_hpf_mode_enum, da732x_hpf_get, da732x_hpf_set),
+ SOC_ENUM("DAC1 High Pass Filter", da732x_dac1_hp_filter_enum),
+ SOC_ENUM("DAC1 Voice Filter", da732x_dac1_voice_filter_enum),
+
+ SOC_ENUM_EXT("DAC2 High Pass Filter Mode",
+ da732x_dac2_hpf_mode_enum, da732x_hpf_get, da732x_hpf_set),
+ SOC_ENUM("DAC2 High Pass Filter", da732x_dac2_hp_filter_enum),
+ SOC_ENUM("DAC2 Voice Filter", da732x_dac2_voice_filter_enum),
+
+ SOC_ENUM_EXT("DAC3 High Pass Filter Mode",
+ da732x_dac3_hpf_mode_enum, da732x_hpf_get, da732x_hpf_set),
+ SOC_ENUM("DAC3 High Pass Filter", da732x_dac3_hp_filter_enum),
+ SOC_ENUM("DAC3 Filter Mode", da732x_dac3_voice_filter_enum),
+
+ SOC_ENUM_EXT("ADC1 High Pass Filter Mode",
+ da732x_adc1_hpf_mode_enum, da732x_hpf_get, da732x_hpf_set),
+ SOC_ENUM("ADC1 High Pass Filter", da732x_adc1_hp_filter_enum),
+ SOC_ENUM("ADC1 Voice Filter", da732x_adc1_voice_filter_enum),
+
+ SOC_ENUM_EXT("ADC2 High Pass Filter Mode",
+ da732x_adc2_hpf_mode_enum, da732x_hpf_get, da732x_hpf_set),
+ SOC_ENUM("ADC2 High Pass Filter", da732x_adc2_hp_filter_enum),
+ SOC_ENUM("ADC2 Voice Filter", da732x_adc2_voice_filter_enum),
+
+ /* Equalizers */
+ SOC_SINGLE("ADC1 EQ Switch", DA732X_REG_ADC1_EQ5,
+ DA732X_EQ_EN_SHIFT, DA732X_EQ_EN_MAX, DA732X_NO_INVERT),
+ SOC_SINGLE_TLV("ADC1 EQ Band 1 Volume", DA732X_REG_ADC1_EQ12,
+ DA732X_EQ_BAND1_SHIFT, DA732X_EQ_VOL_VAL_MAX,
+ DA732X_INVERT, eq_band_pga_tlv),
+ SOC_SINGLE_TLV("ADC1 EQ Band 2 Volume", DA732X_REG_ADC1_EQ12,
+ DA732X_EQ_BAND2_SHIFT, DA732X_EQ_VOL_VAL_MAX,
+ DA732X_INVERT, eq_band_pga_tlv),
+ SOC_SINGLE_TLV("ADC1 EQ Band 3 Volume", DA732X_REG_ADC1_EQ34,
+ DA732X_EQ_BAND3_SHIFT, DA732X_EQ_VOL_VAL_MAX,
+ DA732X_INVERT, eq_band_pga_tlv),
+ SOC_SINGLE_TLV("ADC1 EQ Band 4 Volume", DA732X_REG_ADC1_EQ34,
+ DA732X_EQ_BAND4_SHIFT, DA732X_EQ_VOL_VAL_MAX,
+ DA732X_INVERT, eq_band_pga_tlv),
+ SOC_SINGLE_TLV("ADC1 EQ Band 5 Volume", DA732X_REG_ADC1_EQ5,
+ DA732X_EQ_BAND5_SHIFT, DA732X_EQ_VOL_VAL_MAX,
+ DA732X_INVERT, eq_band_pga_tlv),
+ SOC_SINGLE_TLV("ADC1 EQ Overall Volume", DA732X_REG_ADC1_EQ5,
+ DA732X_EQ_OVERALL_SHIFT, DA732X_EQ_OVERALL_VOL_VAL_MAX,
+ DA732X_INVERT, eq_overall_tlv),
+
+ SOC_SINGLE("ADC2 EQ Switch", DA732X_REG_ADC2_EQ5,
+ DA732X_EQ_EN_SHIFT, DA732X_EQ_EN_MAX, DA732X_NO_INVERT),
+ SOC_SINGLE_TLV("ADC2 EQ Band 1 Volume", DA732X_REG_ADC2_EQ12,
+ DA732X_EQ_BAND1_SHIFT, DA732X_EQ_VOL_VAL_MAX,
+ DA732X_INVERT, eq_band_pga_tlv),
+ SOC_SINGLE_TLV("ADC2 EQ Band 2 Volume", DA732X_REG_ADC2_EQ12,
+ DA732X_EQ_BAND2_SHIFT, DA732X_EQ_VOL_VAL_MAX,
+ DA732X_INVERT, eq_band_pga_tlv),
+ SOC_SINGLE_TLV("ADC2 EQ Band 3 Volume", DA732X_REG_ADC2_EQ34,
+ DA732X_EQ_BAND3_SHIFT, DA732X_EQ_VOL_VAL_MAX,
+ DA732X_INVERT, eq_band_pga_tlv),
+ SOC_SINGLE_TLV("ACD2 EQ Band 4 Volume", DA732X_REG_ADC2_EQ34,
+ DA732X_EQ_BAND4_SHIFT, DA732X_EQ_VOL_VAL_MAX,
+ DA732X_INVERT, eq_band_pga_tlv),
+ SOC_SINGLE_TLV("ACD2 EQ Band 5 Volume", DA732X_REG_ADC2_EQ5,
+ DA732X_EQ_BAND5_SHIFT, DA732X_EQ_VOL_VAL_MAX,
+ DA732X_INVERT, eq_band_pga_tlv),
+ SOC_SINGLE_TLV("ADC2 EQ Overall Volume", DA732X_REG_ADC1_EQ5,
+ DA732X_EQ_OVERALL_SHIFT, DA732X_EQ_OVERALL_VOL_VAL_MAX,
+ DA732X_INVERT, eq_overall_tlv),
+
+ SOC_SINGLE("DAC1 EQ Switch", DA732X_REG_DAC1_EQ5,
+ DA732X_EQ_EN_SHIFT, DA732X_EQ_EN_MAX, DA732X_NO_INVERT),
+ SOC_SINGLE_TLV("DAC1 EQ Band 1 Volume", DA732X_REG_DAC1_EQ12,
+ DA732X_EQ_BAND1_SHIFT, DA732X_EQ_VOL_VAL_MAX,
+ DA732X_INVERT, eq_band_pga_tlv),
+ SOC_SINGLE_TLV("DAC1 EQ Band 2 Volume", DA732X_REG_DAC1_EQ12,
+ DA732X_EQ_BAND2_SHIFT, DA732X_EQ_VOL_VAL_MAX,
+ DA732X_INVERT, eq_band_pga_tlv),
+ SOC_SINGLE_TLV("DAC1 EQ Band 3 Volume", DA732X_REG_DAC1_EQ34,
+ DA732X_EQ_BAND3_SHIFT, DA732X_EQ_VOL_VAL_MAX,
+ DA732X_INVERT, eq_band_pga_tlv),
+ SOC_SINGLE_TLV("DAC1 EQ Band 4 Volume", DA732X_REG_DAC1_EQ34,
+ DA732X_EQ_BAND4_SHIFT, DA732X_EQ_VOL_VAL_MAX,
+ DA732X_INVERT, eq_band_pga_tlv),
+ SOC_SINGLE_TLV("DAC1 EQ Band 5 Volume", DA732X_REG_DAC1_EQ5,
+ DA732X_EQ_BAND5_SHIFT, DA732X_EQ_VOL_VAL_MAX,
+ DA732X_INVERT, eq_band_pga_tlv),
+
+ SOC_SINGLE("DAC2 EQ Switch", DA732X_REG_DAC2_EQ5,
+ DA732X_EQ_EN_SHIFT, DA732X_EQ_EN_MAX, DA732X_NO_INVERT),
+ SOC_SINGLE_TLV("DAC2 EQ Band 1 Volume", DA732X_REG_DAC2_EQ12,
+ DA732X_EQ_BAND1_SHIFT, DA732X_EQ_VOL_VAL_MAX,
+ DA732X_INVERT, eq_band_pga_tlv),
+ SOC_SINGLE_TLV("DAC2 EQ Band 2 Volume", DA732X_REG_DAC2_EQ12,
+ DA732X_EQ_BAND2_SHIFT, DA732X_EQ_VOL_VAL_MAX,
+ DA732X_INVERT, eq_band_pga_tlv),
+ SOC_SINGLE_TLV("DAC2 EQ Band 3 Volume", DA732X_REG_DAC2_EQ34,
+ DA732X_EQ_BAND3_SHIFT, DA732X_EQ_VOL_VAL_MAX,
+ DA732X_INVERT, eq_band_pga_tlv),
+ SOC_SINGLE_TLV("DAC2 EQ Band 4 Volume", DA732X_REG_DAC2_EQ34,
+ DA732X_EQ_BAND4_SHIFT, DA732X_EQ_VOL_VAL_MAX,
+ DA732X_INVERT, eq_band_pga_tlv),
+ SOC_SINGLE_TLV("DAC2 EQ Band 5 Volume", DA732X_REG_DAC2_EQ5,
+ DA732X_EQ_BAND5_SHIFT, DA732X_EQ_VOL_VAL_MAX,
+ DA732X_INVERT, eq_band_pga_tlv),
+
+ SOC_SINGLE("DAC3 EQ Switch", DA732X_REG_DAC3_EQ5,
+ DA732X_EQ_EN_SHIFT, DA732X_EQ_EN_MAX, DA732X_NO_INVERT),
+ SOC_SINGLE_TLV("DAC3 EQ Band 1 Volume", DA732X_REG_DAC3_EQ12,
+ DA732X_EQ_BAND1_SHIFT, DA732X_EQ_VOL_VAL_MAX,
+ DA732X_INVERT, eq_band_pga_tlv),
+ SOC_SINGLE_TLV("DAC3 EQ Band 2 Volume", DA732X_REG_DAC3_EQ12,
+ DA732X_EQ_BAND2_SHIFT, DA732X_EQ_VOL_VAL_MAX,
+ DA732X_INVERT, eq_band_pga_tlv),
+ SOC_SINGLE_TLV("DAC3 EQ Band 3 Volume", DA732X_REG_DAC3_EQ34,
+ DA732X_EQ_BAND3_SHIFT, DA732X_EQ_VOL_VAL_MAX,
+ DA732X_INVERT, eq_band_pga_tlv),
+ SOC_SINGLE_TLV("DAC3 EQ Band 4 Volume", DA732X_REG_DAC3_EQ34,
+ DA732X_EQ_BAND4_SHIFT, DA732X_EQ_VOL_VAL_MAX,
+ DA732X_INVERT, eq_band_pga_tlv),
+ SOC_SINGLE_TLV("DAC3 EQ Band 5 Volume", DA732X_REG_DAC3_EQ5,
+ DA732X_EQ_BAND5_SHIFT, DA732X_EQ_VOL_VAL_MAX,
+ DA732X_INVERT, eq_band_pga_tlv),
+
+ /* Lineout 2 Reciever*/
+ SOC_SINGLE("Lineout 2 Switch", DA732X_REG_LIN2, DA732X_LOUT_MUTE_SHIFT,
+ DA732X_SWITCH_MAX, DA732X_INVERT),
+ SOC_SINGLE_TLV("Lineout 2 Volume", DA732X_REG_LIN2,
+ DA732X_LOUT_VOL_SHIFT, DA732X_LOUT_VOL_VAL_MAX,
+ DA732X_NO_INVERT, lin2_pga_tlv),
+
+ /* Lineout 3 SPEAKER*/
+ SOC_SINGLE("Lineout 3 Switch", DA732X_REG_LIN3, DA732X_LOUT_MUTE_SHIFT,
+ DA732X_SWITCH_MAX, DA732X_INVERT),
+ SOC_SINGLE_TLV("Lineout 3 Volume", DA732X_REG_LIN3,
+ DA732X_LOUT_VOL_SHIFT, DA732X_LOUT_VOL_VAL_MAX,
+ DA732X_NO_INVERT, lin3_pga_tlv),
+
+ /* Lineout 4 */
+ SOC_SINGLE("Lineout 4 Switch", DA732X_REG_LIN4, DA732X_LOUT_MUTE_SHIFT,
+ DA732X_SWITCH_MAX, DA732X_INVERT),
+ SOC_SINGLE_TLV("Lineout 4 Volume", DA732X_REG_LIN4,
+ DA732X_LOUT_VOL_SHIFT, DA732X_LOUT_VOL_VAL_MAX,
+ DA732X_NO_INVERT, lin4_pga_tlv),
+
+ /* Headphones */
+ SOC_DOUBLE_R("Headphone Switch", DA732X_REG_HPR, DA732X_REG_HPL,
+ DA732X_HP_MUTE_SHIFT, DA732X_SWITCH_MAX, DA732X_INVERT),
+ SOC_DOUBLE_R_TLV("Headphone Volume", DA732X_REG_HPL_VOL,
+ DA732X_REG_HPR_VOL, DA732X_HP_VOL_SHIFT,
+ DA732X_HP_VOL_VAL_MAX, DA732X_NO_INVERT, hp_pga_tlv),
+};
+
+static int da732x_adc_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_codec *codec = w->codec;
+
+ switch (event) {
+ case SND_SOC_DAPM_POST_PMU:
+ switch (w->reg) {
+ case DA732X_REG_ADC1_PD:
+ snd_soc_update_bits(codec, DA732X_REG_CLK_EN3,
+ DA732X_ADCA_BB_CLK_EN,
+ DA732X_ADCA_BB_CLK_EN);
+ break;
+ case DA732X_REG_ADC2_PD:
+ snd_soc_update_bits(codec, DA732X_REG_CLK_EN3,
+ DA732X_ADCC_BB_CLK_EN,
+ DA732X_ADCC_BB_CLK_EN);
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ snd_soc_update_bits(codec, w->reg, DA732X_ADC_RST_MASK,
+ DA732X_ADC_SET_ACT);
+ snd_soc_update_bits(codec, w->reg, DA732X_ADC_PD_MASK,
+ DA732X_ADC_ON);
+ break;
+ case SND_SOC_DAPM_POST_PMD:
+ snd_soc_update_bits(codec, w->reg, DA732X_ADC_PD_MASK,
+ DA732X_ADC_OFF);
+ snd_soc_update_bits(codec, w->reg, DA732X_ADC_RST_MASK,
+ DA732X_ADC_SET_RST);
+
+ switch (w->reg) {
+ case DA732X_REG_ADC1_PD:
+ snd_soc_update_bits(codec, DA732X_REG_CLK_EN3,
+ DA732X_ADCA_BB_CLK_EN, 0);
+ break;
+ case DA732X_REG_ADC2_PD:
+ snd_soc_update_bits(codec, DA732X_REG_CLK_EN3,
+ DA732X_ADCC_BB_CLK_EN, 0);
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static int da732x_out_pga_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_codec *codec = w->codec;
+
+ switch (event) {
+ case SND_SOC_DAPM_POST_PMU:
+ snd_soc_update_bits(codec, w->reg,
+ (1 << w->shift) | DA732X_OUT_HIZ_EN,
+ (1 << w->shift) | DA732X_OUT_HIZ_EN);
+ break;
+ case SND_SOC_DAPM_POST_PMD:
+ snd_soc_update_bits(codec, w->reg,
+ (1 << w->shift) | DA732X_OUT_HIZ_EN,
+ (1 << w->shift) | DA732X_OUT_HIZ_DIS);
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static const char *adcl_text[] = {
+ "AUX1L", "MIC1"
+};
+
+static const char *adcr_text[] = {
+ "AUX1R", "MIC2", "MIC3"
+};
+
+static const char *enable_text[] = {
+ "Disabled",
+ "Enabled"
+};
+
+/* ADC1LMUX */
+static const struct soc_enum adc1l_enum =
+ SOC_ENUM_SINGLE(DA732X_REG_INP_MUX, DA732X_ADC1L_MUX_SEL_SHIFT,
+ DA732X_ADCL_MUX_MAX, adcl_text);
+static const struct snd_kcontrol_new adc1l_mux =
+ SOC_DAPM_ENUM("ADC Route", adc1l_enum);
+
+/* ADC1RMUX */
+static const struct soc_enum adc1r_enum =
+ SOC_ENUM_SINGLE(DA732X_REG_INP_MUX, DA732X_ADC1R_MUX_SEL_SHIFT,
+ DA732X_ADCR_MUX_MAX, adcr_text);
+static const struct snd_kcontrol_new adc1r_mux =
+ SOC_DAPM_ENUM("ADC Route", adc1r_enum);
+
+/* ADC2LMUX */
+static const struct soc_enum adc2l_enum =
+ SOC_ENUM_SINGLE(DA732X_REG_INP_MUX, DA732X_ADC2L_MUX_SEL_SHIFT,
+ DA732X_ADCL_MUX_MAX, adcl_text);
+static const struct snd_kcontrol_new adc2l_mux =
+ SOC_DAPM_ENUM("ADC Route", adc2l_enum);
+
+/* ADC2RMUX */
+static const struct soc_enum adc2r_enum =
+ SOC_ENUM_SINGLE(DA732X_REG_INP_MUX, DA732X_ADC2R_MUX_SEL_SHIFT,
+ DA732X_ADCR_MUX_MAX, adcr_text);
+
+static const struct snd_kcontrol_new adc2r_mux =
+ SOC_DAPM_ENUM("ADC Route", adc2r_enum);
+
+static const struct soc_enum da732x_hp_left_output =
+ SOC_ENUM_SINGLE(DA732X_REG_HPL, DA732X_HP_OUT_DAC_EN_SHIFT,
+ DA732X_DAC_EN_MAX, enable_text);
+
+static const struct snd_kcontrol_new hpl_mux =
+ SOC_DAPM_ENUM("HPL Switch", da732x_hp_left_output);
+
+static const struct soc_enum da732x_hp_right_output =
+ SOC_ENUM_SINGLE(DA732X_REG_HPR, DA732X_HP_OUT_DAC_EN_SHIFT,
+ DA732X_DAC_EN_MAX, enable_text);
+
+static const struct snd_kcontrol_new hpr_mux =
+ SOC_DAPM_ENUM("HPR Switch", da732x_hp_right_output);
+
+static const struct soc_enum da732x_speaker_output =
+ SOC_ENUM_SINGLE(DA732X_REG_LIN3, DA732X_LOUT_DAC_EN_SHIFT,
+ DA732X_DAC_EN_MAX, enable_text);
+
+static const struct snd_kcontrol_new spk_mux =
+ SOC_DAPM_ENUM("SPK Switch", da732x_speaker_output);
+
+static const struct soc_enum da732x_lout4_output =
+ SOC_ENUM_SINGLE(DA732X_REG_LIN4, DA732X_LOUT_DAC_EN_SHIFT,
+ DA732X_DAC_EN_MAX, enable_text);
+
+static const struct snd_kcontrol_new lout4_mux =
+ SOC_DAPM_ENUM("LOUT4 Switch", da732x_lout4_output);
+
+static const struct soc_enum da732x_lout2_output =
+ SOC_ENUM_SINGLE(DA732X_REG_LIN2, DA732X_LOUT_DAC_EN_SHIFT,
+ DA732X_DAC_EN_MAX, enable_text);
+
+static const struct snd_kcontrol_new lout2_mux =
+ SOC_DAPM_ENUM("LOUT2 Switch", da732x_lout2_output);
+
+static const struct snd_soc_dapm_widget da732x_dapm_widgets[] = {
+ /* Supplies */
+ SND_SOC_DAPM_SUPPLY("ADC1 Supply", DA732X_REG_ADC1_PD, 0,
+ DA732X_NO_INVERT, da732x_adc_event,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_SUPPLY("ADC2 Supply", DA732X_REG_ADC2_PD, 0,
+ DA732X_NO_INVERT, da732x_adc_event,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_SUPPLY("DAC1 CLK", DA732X_REG_CLK_EN4,
+ DA732X_DACA_BB_CLK_SHIFT, DA732X_NO_INVERT,
+ NULL, 0),
+ SND_SOC_DAPM_SUPPLY("DAC2 CLK", DA732X_REG_CLK_EN4,
+ DA732X_DACC_BB_CLK_SHIFT, DA732X_NO_INVERT,
+ NULL, 0),
+ SND_SOC_DAPM_SUPPLY("DAC3 CLK", DA732X_REG_CLK_EN5,
+ DA732X_DACE_BB_CLK_SHIFT, DA732X_NO_INVERT,
+ NULL, 0),
+
+ /* Micbias */
+ SND_SOC_DAPM_SUPPLY("MICBIAS1", DA732X_REG_MICBIAS1,
+ DA732X_MICBIAS_EN_SHIFT,
+ DA732X_NO_INVERT, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("MICBIAS2", DA732X_REG_MICBIAS2,
+ DA732X_MICBIAS_EN_SHIFT,
+ DA732X_NO_INVERT, NULL, 0),
+
+ /* Inputs */
+ SND_SOC_DAPM_INPUT("MIC1"),
+ SND_SOC_DAPM_INPUT("MIC2"),
+ SND_SOC_DAPM_INPUT("MIC3"),
+ SND_SOC_DAPM_INPUT("AUX1L"),
+ SND_SOC_DAPM_INPUT("AUX1R"),
+
+ /* Outputs */
+ SND_SOC_DAPM_OUTPUT("HPL"),
+ SND_SOC_DAPM_OUTPUT("HPR"),
+ SND_SOC_DAPM_OUTPUT("LOUTL"),
+ SND_SOC_DAPM_OUTPUT("LOUTR"),
+ SND_SOC_DAPM_OUTPUT("ClassD"),
+
+ /* ADCs */
+ SND_SOC_DAPM_ADC("ADC1L", NULL, DA732X_REG_ADC1_SEL,
+ DA732X_ADCL_EN_SHIFT, DA732X_NO_INVERT),
+ SND_SOC_DAPM_ADC("ADC1R", NULL, DA732X_REG_ADC1_SEL,
+ DA732X_ADCR_EN_SHIFT, DA732X_NO_INVERT),
+ SND_SOC_DAPM_ADC("ADC2L", NULL, DA732X_REG_ADC2_SEL,
+ DA732X_ADCL_EN_SHIFT, DA732X_NO_INVERT),
+ SND_SOC_DAPM_ADC("ADC2R", NULL, DA732X_REG_ADC2_SEL,
+ DA732X_ADCR_EN_SHIFT, DA732X_NO_INVERT),
+
+ /* DACs */
+ SND_SOC_DAPM_DAC("DAC1L", NULL, DA732X_REG_DAC1_SEL,
+ DA732X_DACL_EN_SHIFT, DA732X_NO_INVERT),
+ SND_SOC_DAPM_DAC("DAC1R", NULL, DA732X_REG_DAC1_SEL,
+ DA732X_DACR_EN_SHIFT, DA732X_NO_INVERT),
+ SND_SOC_DAPM_DAC("DAC2L", NULL, DA732X_REG_DAC2_SEL,
+ DA732X_DACL_EN_SHIFT, DA732X_NO_INVERT),
+ SND_SOC_DAPM_DAC("DAC2R", NULL, DA732X_REG_DAC2_SEL,
+ DA732X_DACR_EN_SHIFT, DA732X_NO_INVERT),
+ SND_SOC_DAPM_DAC("DAC3", NULL, DA732X_REG_DAC3_SEL,
+ DA732X_DACL_EN_SHIFT, DA732X_NO_INVERT),
+
+ /* Input Pgas */
+ SND_SOC_DAPM_PGA("MIC1 PGA", DA732X_REG_MIC1, DA732X_MIC_EN_SHIFT,
+ 0, NULL, 0),
+ SND_SOC_DAPM_PGA("MIC2 PGA", DA732X_REG_MIC2, DA732X_MIC_EN_SHIFT,
+ 0, NULL, 0),
+ SND_SOC_DAPM_PGA("MIC3 PGA", DA732X_REG_MIC3, DA732X_MIC_EN_SHIFT,
+ 0, NULL, 0),
+ SND_SOC_DAPM_PGA("AUX1L PGA", DA732X_REG_AUX1L, DA732X_AUX_EN_SHIFT,
+ 0, NULL, 0),
+ SND_SOC_DAPM_PGA("AUX1R PGA", DA732X_REG_AUX1R, DA732X_AUX_EN_SHIFT,
+ 0, NULL, 0),
+
+ SND_SOC_DAPM_PGA_E("HP Left", DA732X_REG_HPL, DA732X_HP_OUT_EN_SHIFT,
+ 0, NULL, 0, da732x_out_pga_event,
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_PGA_E("HP Right", DA732X_REG_HPR, DA732X_HP_OUT_EN_SHIFT,
+ 0, NULL, 0, da732x_out_pga_event,
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_PGA_E("LIN2", DA732X_REG_LIN2, DA732X_LIN_OUT_EN_SHIFT,
+ 0, NULL, 0, da732x_out_pga_event,
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_PGA_E("LIN3", DA732X_REG_LIN3, DA732X_LIN_OUT_EN_SHIFT,
+ 0, NULL, 0, da732x_out_pga_event,
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_PGA_E("LIN4", DA732X_REG_LIN4, DA732X_LIN_OUT_EN_SHIFT,
+ 0, NULL, 0, da732x_out_pga_event,
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+
+ /* MUXs */
+ SND_SOC_DAPM_MUX("ADC1 Left MUX", SND_SOC_NOPM, 0, 0, &adc1l_mux),
+ SND_SOC_DAPM_MUX("ADC1 Right MUX", SND_SOC_NOPM, 0, 0, &adc1r_mux),
+ SND_SOC_DAPM_MUX("ADC2 Left MUX", SND_SOC_NOPM, 0, 0, &adc2l_mux),
+ SND_SOC_DAPM_MUX("ADC2 Right MUX", SND_SOC_NOPM, 0, 0, &adc2r_mux),
+
+ SND_SOC_DAPM_MUX("HP Left MUX", SND_SOC_NOPM, 0, 0, &hpl_mux),
+ SND_SOC_DAPM_MUX("HP Right MUX", SND_SOC_NOPM, 0, 0, &hpr_mux),
+ SND_SOC_DAPM_MUX("Speaker MUX", SND_SOC_NOPM, 0, 0, &spk_mux),
+ SND_SOC_DAPM_MUX("LOUT2 MUX", SND_SOC_NOPM, 0, 0, &lout2_mux),
+ SND_SOC_DAPM_MUX("LOUT4 MUX", SND_SOC_NOPM, 0, 0, &lout4_mux),
+
+ /* AIF interfaces */
+ SND_SOC_DAPM_AIF_OUT("AIFA Output", "AIFA Capture", 0, DA732X_REG_AIFA3,
+ DA732X_AIF_EN_SHIFT, 0),
+ SND_SOC_DAPM_AIF_IN("AIFA Input", "AIFA Playback", 0, DA732X_REG_AIFA3,
+ DA732X_AIF_EN_SHIFT, 0),
+
+ SND_SOC_DAPM_AIF_OUT("AIFB Output", "AIFB Capture", 0, DA732X_REG_AIFB3,
+ DA732X_AIF_EN_SHIFT, 0),
+ SND_SOC_DAPM_AIF_IN("AIFB Input", "AIFB Playback", 0, DA732X_REG_AIFB3,
+ DA732X_AIF_EN_SHIFT, 0),
+};
+
+static const struct snd_soc_dapm_route da732x_dapm_routes[] = {
+ /* Inputs */
+ {"AUX1L PGA", "NULL", "AUX1L"},
+ {"AUX1R PGA", "NULL", "AUX1R"},
+ {"MIC1 PGA", NULL, "MIC1"},
+ {"MIC2 PGA", "NULL", "MIC2"},
+ {"MIC3 PGA", "NULL", "MIC3"},
+
+ /* Capture Path */
+ {"ADC1 Left MUX", "MIC1", "MIC1 PGA"},
+ {"ADC1 Left MUX", "AUX1L", "AUX1L PGA"},
+
+ {"ADC1 Right MUX", "AUX1R", "AUX1R PGA"},
+ {"ADC1 Right MUX", "MIC2", "MIC2 PGA"},
+ {"ADC1 Right MUX", "MIC3", "MIC3 PGA"},
+
+ {"ADC2 Left MUX", "AUX1L", "AUX1L PGA"},
+ {"ADC2 Left MUX", "MIC1", "MIC1 PGA"},
+
+ {"ADC2 Right MUX", "AUX1R", "AUX1R PGA"},
+ {"ADC2 Right MUX", "MIC2", "MIC2 PGA"},
+ {"ADC2 Right MUX", "MIC3", "MIC3 PGA"},
+
+ {"ADC1L", NULL, "ADC1 Supply"},
+ {"ADC1R", NULL, "ADC1 Supply"},
+ {"ADC2L", NULL, "ADC2 Supply"},
+ {"ADC2R", NULL, "ADC2 Supply"},
+
+ {"ADC1L", NULL, "ADC1 Left MUX"},
+ {"ADC1R", NULL, "ADC1 Right MUX"},
+ {"ADC2L", NULL, "ADC2 Left MUX"},
+ {"ADC2R", NULL, "ADC2 Right MUX"},
+
+ {"AIFA Output", NULL, "ADC1L"},
+ {"AIFA Output", NULL, "ADC1R"},
+ {"AIFB Output", NULL, "ADC2L"},
+ {"AIFB Output", NULL, "ADC2R"},
+
+ {"HP Left MUX", "Enabled", "AIFA Input"},
+ {"HP Right MUX", "Enabled", "AIFA Input"},
+ {"Speaker MUX", "Enabled", "AIFB Input"},
+ {"LOUT2 MUX", "Enabled", "AIFB Input"},
+ {"LOUT4 MUX", "Enabled", "AIFB Input"},
+
+ {"DAC1L", NULL, "DAC1 CLK"},
+ {"DAC1R", NULL, "DAC1 CLK"},
+ {"DAC2L", NULL, "DAC2 CLK"},
+ {"DAC2R", NULL, "DAC2 CLK"},
+ {"DAC3", NULL, "DAC3 CLK"},
+
+ {"DAC1L", NULL, "HP Left MUX"},
+ {"DAC1R", NULL, "HP Right MUX"},
+ {"DAC2L", NULL, "Speaker MUX"},
+ {"DAC2R", NULL, "LOUT4 MUX"},
+ {"DAC3", NULL, "LOUT2 MUX"},
+
+ /* Output Pgas */
+ {"HP Left", NULL, "DAC1L"},
+ {"HP Right", NULL, "DAC1R"},
+ {"LIN3", NULL, "DAC2L"},
+ {"LIN4", NULL, "DAC2R"},
+ {"LIN2", NULL, "DAC3"},
+
+ /* Outputs */
+ {"ClassD", NULL, "LIN3"},
+ {"LOUTL", NULL, "LIN2"},
+ {"LOUTR", NULL, "LIN4"},
+ {"HPL", NULL, "HP Left"},
+ {"HPR", NULL, "HP Right"},
+};
+
+static int da732x_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;
+ u32 aif = 0;
+ u32 reg_aif;
+ u32 fs;
+
+ reg_aif = dai->driver->base;
+
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ aif |= DA732X_AIF_WORD_16;
+ break;
+ case SNDRV_PCM_FORMAT_S20_3LE:
+ aif |= DA732X_AIF_WORD_20;
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ aif |= DA732X_AIF_WORD_24;
+ break;
+ case SNDRV_PCM_FORMAT_S32_LE:
+ aif |= DA732X_AIF_WORD_32;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ switch (params_rate(params)) {
+ case 8000:
+ fs = DA732X_SR_8KHZ;
+ break;
+ case 11025:
+ fs = DA732X_SR_11_025KHZ;
+ break;
+ case 12000:
+ fs = DA732X_SR_12KHZ;
+ break;
+ case 16000:
+ fs = DA732X_SR_16KHZ;
+ break;
+ case 22050:
+ fs = DA732X_SR_22_05KHZ;
+ break;
+ case 24000:
+ fs = DA732X_SR_24KHZ;
+ break;
+ case 32000:
+ fs = DA732X_SR_32KHZ;
+ break;
+ case 44100:
+ fs = DA732X_SR_44_1KHZ;
+ break;
+ case 48000:
+ fs = DA732X_SR_48KHZ;
+ break;
+ case 88100:
+ fs = DA732X_SR_88_1KHZ;
+ break;
+ case 96000:
+ fs = DA732X_SR_96KHZ;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ snd_soc_update_bits(codec, reg_aif, DA732X_AIF_WORD_MASK, aif);
+ snd_soc_update_bits(codec, DA732X_REG_CLK_CTRL, DA732X_SR1_MASK, fs);
+
+ return 0;
+}
+
+static int da732x_set_dai_fmt(struct snd_soc_dai *dai, u32 fmt)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ u32 aif_mclk, pc_count;
+ u32 reg_aif1, aif1;
+ u32 reg_aif3, aif3;
+
+ switch (dai->id) {
+ case DA732X_DAI_ID1:
+ reg_aif1 = DA732X_REG_AIFA1;
+ reg_aif3 = DA732X_REG_AIFA3;
+ pc_count = DA732X_PC_PULSE_AIFA | DA732X_PC_RESYNC_NOT_AUT |
+ DA732X_PC_SAME;
+ break;
+ case DA732X_DAI_ID2:
+ reg_aif1 = DA732X_REG_AIFB1;
+ reg_aif3 = DA732X_REG_AIFB3;
+ pc_count = DA732X_PC_PULSE_AIFB | DA732X_PC_RESYNC_NOT_AUT |
+ DA732X_PC_SAME;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBS_CFS:
+ aif1 = DA732X_AIF_SLAVE;
+ aif_mclk = DA732X_AIFM_FRAME_64 | DA732X_AIFM_SRC_SEL_AIFA;
+ break;
+ case SND_SOC_DAIFMT_CBM_CFM:
+ aif1 = DA732X_AIF_CLK_FROM_SRC;
+ aif_mclk = DA732X_CLK_GENERATION_AIF_A;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ aif3 = DA732X_AIF_I2S_MODE;
+ break;
+ case SND_SOC_DAIFMT_RIGHT_J:
+ aif3 = DA732X_AIF_RIGHT_J_MODE;
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ aif3 = DA732X_AIF_LEFT_J_MODE;
+ break;
+ case SND_SOC_DAIFMT_DSP_B:
+ aif3 = DA732X_AIF_DSP_MODE;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ /* Clock inversion */
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_DSP_B:
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_NB_NF:
+ break;
+ case SND_SOC_DAIFMT_IB_NF:
+ aif3 |= DA732X_AIF_BCLK_INV;
+ break;
+ default:
+ return -EINVAL;
+ }
+ break;
+ case SND_SOC_DAIFMT_I2S:
+ case SND_SOC_DAIFMT_RIGHT_J:
+ case SND_SOC_DAIFMT_LEFT_J:
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_NB_NF:
+ break;
+ case SND_SOC_DAIFMT_IB_IF:
+ aif3 |= DA732X_AIF_BCLK_INV | DA732X_AIF_WCLK_INV;
+ break;
+ case SND_SOC_DAIFMT_IB_NF:
+ aif3 |= DA732X_AIF_BCLK_INV;
+ break;
+ case SND_SOC_DAIFMT_NB_IF:
+ aif3 |= DA732X_AIF_WCLK_INV;
+ break;
+ default:
+ return -EINVAL;
+ }
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ snd_soc_write(codec, DA732X_REG_AIF_MCLK, aif_mclk);
+ snd_soc_update_bits(codec, reg_aif1, DA732X_AIF1_CLK_MASK, aif1);
+ snd_soc_update_bits(codec, reg_aif3, DA732X_AIF_BCLK_INV |
+ DA732X_AIF_WCLK_INV | DA732X_AIF_MODE_MASK, aif3);
+ snd_soc_write(codec, DA732X_REG_PC_CTRL, pc_count);
+
+ return 0;
+}
+
+
+
+static int da732x_set_dai_pll(struct snd_soc_codec *codec, int pll_id,
+ int source, unsigned int freq_in,
+ unsigned int freq_out)
+{
+ struct da732x_priv *da732x = snd_soc_codec_get_drvdata(codec);
+ int fref, indiv;
+ u8 div_lo, div_mid, div_hi;
+ u64 frac_div;
+
+ /* Disable PLL */
+ if (freq_out == 0) {
+ snd_soc_update_bits(codec, DA732X_REG_PLL_CTRL,
+ DA732X_PLL_EN, 0);
+ da732x->pll_en = false;
+ return 0;
+ }
+
+ if (da732x->pll_en)
+ return -EBUSY;
+
+ if (source == DA732X_SRCCLK_MCLK) {
+ /* Validate Sysclk rate */
+ switch (da732x->sysclk) {
+ case 11290000:
+ case 12288000:
+ case 22580000:
+ case 24576000:
+ case 45160000:
+ case 49152000:
+ snd_soc_write(codec, DA732X_REG_PLL_CTRL,
+ DA732X_PLL_BYPASS);
+ return 0;
+ default:
+ dev_err(codec->dev,
+ "Cannot use PLL Bypass, invalid SYSCLK rate\n");
+ return -EINVAL;
+ }
+ }
+
+ indiv = da732x_get_input_div(codec, da732x->sysclk);
+ if (indiv < 0)
+ return indiv;
+
+ fref = (da732x->sysclk / indiv);
+ div_hi = freq_out / fref;
+ frac_div = (u64)(freq_out % fref) * 8192ULL;
+ do_div(frac_div, fref);
+ div_mid = (frac_div >> DA732X_1BYTE_SHIFT) & DA732X_U8_MASK;
+ div_lo = (frac_div) & DA732X_U8_MASK;
+
+ snd_soc_write(codec, DA732X_REG_PLL_DIV_LO, div_lo);
+ snd_soc_write(codec, DA732X_REG_PLL_DIV_MID, div_mid);
+ snd_soc_write(codec, DA732X_REG_PLL_DIV_HI, div_hi);
+
+ snd_soc_update_bits(codec, DA732X_REG_PLL_CTRL, DA732X_PLL_EN,
+ DA732X_PLL_EN);
+
+ da732x->pll_en = true;
+
+ return 0;
+}
+
+static int da732x_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id,
+ unsigned int freq, int dir)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ struct da732x_priv *da732x = snd_soc_codec_get_drvdata(codec);
+
+ da732x->sysclk = freq;
+
+ return 0;
+}
+
+#define DA732X_RATES SNDRV_PCM_RATE_8000_96000
+
+#define DA732X_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
+ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
+
+static struct snd_soc_dai_ops da732x_dai1_ops = {
+ .hw_params = da732x_hw_params,
+ .set_fmt = da732x_set_dai_fmt,
+ .set_sysclk = da732x_set_dai_sysclk,
+};
+
+static struct snd_soc_dai_ops da732x_dai2_ops = {
+ .hw_params = da732x_hw_params,
+ .set_fmt = da732x_set_dai_fmt,
+ .set_sysclk = da732x_set_dai_sysclk,
+};
+
+static struct snd_soc_dai_driver da732x_dai[] = {
+ {
+ .name = "DA732X_AIFA",
+ .id = DA732X_DAI_ID1,
+ .base = DA732X_REG_AIFA1,
+ .playback = {
+ .stream_name = "AIFA Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = DA732X_RATES,
+ .formats = DA732X_FORMATS,
+ },
+ .capture = {
+ .stream_name = "AIFA Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = DA732X_RATES,
+ .formats = DA732X_FORMATS,
+ },
+ .ops = &da732x_dai1_ops,
+ },
+ {
+ .name = "DA732X_AIFB",
+ .id = DA732X_DAI_ID2,
+ .base = DA732X_REG_AIFB1,
+ .playback = {
+ .stream_name = "AIFB Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = DA732X_RATES,
+ .formats = DA732X_FORMATS,
+ },
+ .capture = {
+ .stream_name = "AIFB Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = DA732X_RATES,
+ .formats = DA732X_FORMATS,
+ },
+ .ops = &da732x_dai2_ops,
+ },
+};
+
+static const struct regmap_config da732x_regmap = {
+ .reg_bits = 8,
+ .val_bits = 8,
+
+ .max_register = DA732X_MAX_REG,
+ .reg_defaults = da732x_reg_cache,
+ .num_reg_defaults = ARRAY_SIZE(da732x_reg_cache),
+ .cache_type = REGCACHE_RBTREE,
+};
+
+
+static void da732x_dac_offset_adjust(struct snd_soc_codec *codec)
+{
+ u8 offset[DA732X_HP_DACS];
+ u8 sign[DA732X_HP_DACS];
+ u8 step = DA732X_DAC_OFFSET_STEP;
+
+ /* Initialize DAC offset calibration circuits and registers */
+ snd_soc_write(codec, DA732X_REG_HPL_DAC_OFFSET,
+ DA732X_HP_DAC_OFFSET_TRIM_VAL);
+ snd_soc_write(codec, DA732X_REG_HPR_DAC_OFFSET,
+ DA732X_HP_DAC_OFFSET_TRIM_VAL);
+ snd_soc_write(codec, DA732X_REG_HPL_DAC_OFF_CNTL,
+ DA732X_HP_DAC_OFF_CALIBRATION |
+ DA732X_HP_DAC_OFF_SCALE_STEPS);
+ snd_soc_write(codec, DA732X_REG_HPR_DAC_OFF_CNTL,
+ DA732X_HP_DAC_OFF_CALIBRATION |
+ DA732X_HP_DAC_OFF_SCALE_STEPS);
+
+ /* Wait for voltage stabilization */
+ msleep(DA732X_WAIT_FOR_STABILIZATION);
+
+ /* Check DAC offset sign */
+ sign[DA732X_HPL_DAC] = (codec->hw_read(codec, DA732X_REG_HPL_DAC_OFF_CNTL) &
+ DA732X_HP_DAC_OFF_CNTL_COMPO);
+ sign[DA732X_HPR_DAC] = (codec->hw_read(codec, DA732X_REG_HPR_DAC_OFF_CNTL) &
+ DA732X_HP_DAC_OFF_CNTL_COMPO);
+
+ /* Binary search DAC offset values (both channels at once) */
+ offset[DA732X_HPL_DAC] = sign[DA732X_HPL_DAC] << DA732X_HP_DAC_COMPO_SHIFT;
+ offset[DA732X_HPR_DAC] = sign[DA732X_HPR_DAC] << DA732X_HP_DAC_COMPO_SHIFT;
+
+ do {
+ offset[DA732X_HPL_DAC] |= step;
+ offset[DA732X_HPR_DAC] |= step;
+ snd_soc_write(codec, DA732X_REG_HPL_DAC_OFFSET,
+ ~offset[DA732X_HPL_DAC] & DA732X_HP_DAC_OFF_MASK);
+ snd_soc_write(codec, DA732X_REG_HPR_DAC_OFFSET,
+ ~offset[DA732X_HPR_DAC] & DA732X_HP_DAC_OFF_MASK);
+
+ msleep(DA732X_WAIT_FOR_STABILIZATION);
+
+ if ((codec->hw_read(codec, DA732X_REG_HPL_DAC_OFF_CNTL) &
+ DA732X_HP_DAC_OFF_CNTL_COMPO) ^ sign[DA732X_HPL_DAC])
+ offset[DA732X_HPL_DAC] &= ~step;
+ if ((codec->hw_read(codec, DA732X_REG_HPR_DAC_OFF_CNTL) &
+ DA732X_HP_DAC_OFF_CNTL_COMPO) ^ sign[DA732X_HPR_DAC])
+ offset[DA732X_HPR_DAC] &= ~step;
+
+ step >>= 1;
+ } while (step);
+
+ /* Write final DAC offsets to registers */
+ snd_soc_write(codec, DA732X_REG_HPL_DAC_OFFSET,
+ ~offset[DA732X_HPL_DAC] & DA732X_HP_DAC_OFF_MASK);
+ snd_soc_write(codec, DA732X_REG_HPR_DAC_OFFSET,
+ ~offset[DA732X_HPR_DAC] & DA732X_HP_DAC_OFF_MASK);
+
+ /* End DAC calibration mode */
+ snd_soc_write(codec, DA732X_REG_HPL_DAC_OFF_CNTL,
+ DA732X_HP_DAC_OFF_SCALE_STEPS);
+ snd_soc_write(codec, DA732X_REG_HPR_DAC_OFF_CNTL,
+ DA732X_HP_DAC_OFF_SCALE_STEPS);
+}
+
+static void da732x_output_offset_adjust(struct snd_soc_codec *codec)
+{
+ u8 offset[DA732X_HP_AMPS];
+ u8 sign[DA732X_HP_AMPS];
+ u8 step = DA732X_OUTPUT_OFFSET_STEP;
+
+ offset[DA732X_HPL_AMP] = DA732X_HP_OUT_TRIM_VAL;
+ offset[DA732X_HPR_AMP] = DA732X_HP_OUT_TRIM_VAL;
+
+ /* Initialize output offset calibration circuits and registers */
+ snd_soc_write(codec, DA732X_REG_HPL_OUT_OFFSET, DA732X_HP_OUT_TRIM_VAL);
+ snd_soc_write(codec, DA732X_REG_HPR_OUT_OFFSET, DA732X_HP_OUT_TRIM_VAL);
+ snd_soc_write(codec, DA732X_REG_HPL,
+ DA732X_HP_OUT_COMP | DA732X_HP_OUT_EN);
+ snd_soc_write(codec, DA732X_REG_HPR,
+ DA732X_HP_OUT_COMP | DA732X_HP_OUT_EN);
+
+ /* Wait for voltage stabilization */
+ msleep(DA732X_WAIT_FOR_STABILIZATION);
+
+ /* Check output offset sign */
+ sign[DA732X_HPL_AMP] = codec->hw_read(codec, DA732X_REG_HPL) &
+ DA732X_HP_OUT_COMPO;
+ sign[DA732X_HPR_AMP] = codec->hw_read(codec, DA732X_REG_HPR) &
+ DA732X_HP_OUT_COMPO;
+
+ snd_soc_write(codec, DA732X_REG_HPL, DA732X_HP_OUT_COMP |
+ (sign[DA732X_HPL_AMP] >> DA732X_HP_OUT_COMPO_SHIFT) |
+ DA732X_HP_OUT_EN);
+ snd_soc_write(codec, DA732X_REG_HPR, DA732X_HP_OUT_COMP |
+ (sign[DA732X_HPR_AMP] >> DA732X_HP_OUT_COMPO_SHIFT) |
+ DA732X_HP_OUT_EN);
+
+ /* Binary search output offset values (both channels at once) */
+ do {
+ offset[DA732X_HPL_AMP] |= step;
+ offset[DA732X_HPR_AMP] |= step;
+ snd_soc_write(codec, DA732X_REG_HPL_OUT_OFFSET,
+ offset[DA732X_HPL_AMP]);
+ snd_soc_write(codec, DA732X_REG_HPR_OUT_OFFSET,
+ offset[DA732X_HPR_AMP]);
+
+ msleep(DA732X_WAIT_FOR_STABILIZATION);
+
+ if ((codec->hw_read(codec, DA732X_REG_HPL) &
+ DA732X_HP_OUT_COMPO) ^ sign[DA732X_HPL_AMP])
+ offset[DA732X_HPL_AMP] &= ~step;
+ if ((codec->hw_read(codec, DA732X_REG_HPR) &
+ DA732X_HP_OUT_COMPO) ^ sign[DA732X_HPR_AMP])
+ offset[DA732X_HPR_AMP] &= ~step;
+
+ step >>= 1;
+ } while (step);
+
+ /* Write final DAC offsets to registers */
+ snd_soc_write(codec, DA732X_REG_HPL_OUT_OFFSET, offset[DA732X_HPL_AMP]);
+ snd_soc_write(codec, DA732X_REG_HPR_OUT_OFFSET, offset[DA732X_HPR_AMP]);
+}
+
+static void da732x_hp_dc_offset_cancellation(struct snd_soc_codec *codec)
+{
+ /* Make sure that we have Soft Mute enabled */
+ snd_soc_write(codec, DA732X_REG_DAC1_SOFTMUTE, DA732X_SOFTMUTE_EN |
+ DA732X_GAIN_RAMPED | DA732X_16_SAMPLES);
+ snd_soc_write(codec, DA732X_REG_DAC1_SEL, DA732X_DACL_EN |
+ DA732X_DACR_EN | DA732X_DACL_SDM | DA732X_DACR_SDM |
+ DA732X_DACL_MUTE | DA732X_DACR_MUTE);
+ snd_soc_write(codec, DA732X_REG_HPL, DA732X_HP_OUT_DAC_EN |
+ DA732X_HP_OUT_MUTE | DA732X_HP_OUT_EN);
+ snd_soc_write(codec, DA732X_REG_HPR, DA732X_HP_OUT_EN |
+ DA732X_HP_OUT_MUTE | DA732X_HP_OUT_DAC_EN);
+
+ da732x_dac_offset_adjust(codec);
+ da732x_output_offset_adjust(codec);
+
+ snd_soc_write(codec, DA732X_REG_DAC1_SEL, DA732X_DACS_DIS);
+ snd_soc_write(codec, DA732X_REG_HPL, DA732X_HP_DIS);
+ snd_soc_write(codec, DA732X_REG_HPR, DA732X_HP_DIS);
+}
+
+static int da732x_set_bias_level(struct snd_soc_codec *codec,
+ enum snd_soc_bias_level level)
+{
+ struct da732x_priv *da732x = snd_soc_codec_get_drvdata(codec);
+
+ switch (level) {
+ case SND_SOC_BIAS_ON:
+ snd_soc_update_bits(codec, DA732X_REG_BIAS_EN,
+ DA732X_BIAS_BOOST_MASK,
+ DA732X_BIAS_BOOST_100PC);
+ break;
+ case SND_SOC_BIAS_PREPARE:
+ break;
+ case SND_SOC_BIAS_STANDBY:
+ if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
+ /* Init Codec */
+ snd_soc_write(codec, DA732X_REG_REF1,
+ DA732X_VMID_FASTCHG);
+ snd_soc_write(codec, DA732X_REG_BIAS_EN,
+ DA732X_BIAS_EN);
+
+ mdelay(DA732X_STARTUP_DELAY);
+
+ /* Disable Fast Charge and enable DAC ref voltage */
+ snd_soc_write(codec, DA732X_REG_REF1,
+ DA732X_REFBUFX2_EN);
+
+ /* Enable bypass DSP routing */
+ snd_soc_write(codec, DA732X_REG_DATA_ROUTE,
+ DA732X_BYPASS_DSP);
+
+ /* Enable Digital subsystem */
+ snd_soc_write(codec, DA732X_REG_DSP_CTRL,
+ DA732X_DIGITAL_EN);
+
+ snd_soc_write(codec, DA732X_REG_SPARE1_OUT,
+ DA732X_HP_DRIVER_EN |
+ DA732X_HP_GATE_LOW |
+ DA732X_HP_LOOP_GAIN_CTRL);
+ snd_soc_write(codec, DA732X_REG_HP_LIN1_GNDSEL,
+ DA732X_HP_OUT_GNDSEL);
+
+ da732x_set_charge_pump(codec, DA732X_ENABLE_CP);
+
+ snd_soc_write(codec, DA732X_REG_CLK_EN1,
+ DA732X_SYS3_CLK_EN | DA732X_PC_CLK_EN);
+
+ /* Enable Zero Crossing */
+ snd_soc_write(codec, DA732X_REG_INP_ZC_EN,
+ DA732X_MIC1_PRE_ZC_EN |
+ DA732X_MIC1_ZC_EN |
+ DA732X_MIC2_PRE_ZC_EN |
+ DA732X_MIC2_ZC_EN |
+ DA732X_AUXL_ZC_EN |
+ DA732X_AUXR_ZC_EN |
+ DA732X_MIC3_PRE_ZC_EN |
+ DA732X_MIC3_ZC_EN);
+ snd_soc_write(codec, DA732X_REG_OUT_ZC_EN,
+ DA732X_HPL_ZC_EN | DA732X_HPR_ZC_EN |
+ DA732X_LIN2_ZC_EN | DA732X_LIN3_ZC_EN |
+ DA732X_LIN4_ZC_EN);
+
+ da732x_hp_dc_offset_cancellation(codec);
+
+ regcache_cache_only(codec->control_data, false);
+ regcache_sync(codec->control_data);
+ } else {
+ snd_soc_update_bits(codec, DA732X_REG_BIAS_EN,
+ DA732X_BIAS_BOOST_MASK,
+ DA732X_BIAS_BOOST_50PC);
+ snd_soc_update_bits(codec, DA732X_REG_PLL_CTRL,
+ DA732X_PLL_EN, 0);
+ da732x->pll_en = false;
+ }
+ break;
+ case SND_SOC_BIAS_OFF:
+ regcache_cache_only(codec->control_data, true);
+ da732x_set_charge_pump(codec, DA732X_DISABLE_CP);
+ snd_soc_update_bits(codec, DA732X_REG_BIAS_EN, DA732X_BIAS_EN,
+ DA732X_BIAS_DIS);
+ da732x->pll_en = false;
+ break;
+ }
+
+ codec->dapm.bias_level = level;
+
+ return 0;
+}
+
+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;
+ int ret = 0;
+
+ da732x->codec = codec;
+
+ dapm->idle_bias_off = false;
+
+ codec->control_data = da732x->regmap;
+
+ ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP);
+ if (ret != 0) {
+ dev_err(codec->dev, "Failed to register codec.\n");
+ goto err;
+ }
+
+ da732x_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+err:
+ return ret;
+}
+
+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),
+ .dapm_widgets = da732x_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(da732x_dapm_widgets),
+ .dapm_routes = da732x_dapm_routes,
+ .num_dapm_routes = ARRAY_SIZE(da732x_dapm_routes),
+ .set_pll = da732x_set_dai_pll,
+ .reg_cache_size = ARRAY_SIZE(da732x_reg_cache),
+};
+
+static __devinit int da732x_i2c_probe(struct i2c_client *i2c,
+ const struct i2c_device_id *id)
+{
+ struct da732x_priv *da732x;
+ unsigned int reg;
+ int ret;
+
+ da732x = devm_kzalloc(&i2c->dev, sizeof(struct da732x_priv),
+ GFP_KERNEL);
+ if (!da732x)
+ return -ENOMEM;
+
+ i2c_set_clientdata(i2c, da732x);
+
+ da732x->regmap = devm_regmap_init_i2c(i2c, &da732x_regmap);
+ if (IS_ERR(da732x->regmap)) {
+ ret = PTR_ERR(da732x->regmap);
+ dev_err(&i2c->dev, "Failed to initialize regmap\n");
+ goto err;
+ }
+
+ ret = regmap_read(da732x->regmap, DA732X_REG_ID, &reg);
+ if (ret < 0) {
+ dev_err(&i2c->dev, "Failed to read ID register: %d\n", ret);
+ goto err;
+ }
+
+ dev_info(&i2c->dev, "Revision: %d.%d\n",
+ (reg & DA732X_ID_MAJOR_MASK), (reg & DA732X_ID_MINOR_MASK));
+
+ ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_da732x,
+ da732x_dai, ARRAY_SIZE(da732x_dai));
+ if (ret != 0)
+ dev_err(&i2c->dev, "Failed to register codec.\n");
+
+err:
+ return ret;
+}
+
+static __devexit int da732x_i2c_remove(struct i2c_client *client)
+{
+ snd_soc_unregister_codec(&client->dev);
+
+ return 0;
+}
+
+static const struct i2c_device_id da732x_i2c_id[] = {
+ { "da7320", 0},
+ { }
+};
+MODULE_DEVICE_TABLE(i2c, da732x_i2c_id);
+
+static struct i2c_driver da732x_i2c_driver = {
+ .driver = {
+ .name = "da7320",
+ .owner = THIS_MODULE,
+ },
+ .probe = da732x_i2c_probe,
+ .remove = __devexit_p(da732x_i2c_remove),
+ .id_table = da732x_i2c_id,
+};
+
+module_i2c_driver(da732x_i2c_driver);
+
+
+MODULE_DESCRIPTION("ASoC DA732X driver");
+MODULE_AUTHOR("Michal Hajduk <michal.hajduk@diasemi.com>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/da732x.h b/sound/soc/codecs/da732x.h
new file mode 100644
index 0000000..c8ce547
--- /dev/null
+++ b/sound/soc/codecs/da732x.h
@@ -0,0 +1,133 @@
+/*
+ * da732x.h -- Dialog DA732X ALSA SoC Audio Driver Header File
+ *
+ * Copyright (C) 2012 Dialog Semiconductor GmbH
+ *
+ * Author: Michal Hajduk <Michal.Hajduk@diasemi.com>
+ *
+ * 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 __DA732X_H_
+#define __DA732X_H
+
+#include <sound/soc.h>
+
+/* General */
+#define DA732X_U8_MASK 0xFF
+#define DA732X_4BYTES 4
+#define DA732X_3BYTES 3
+#define DA732X_2BYTES 2
+#define DA732X_1BYTE 1
+#define DA732X_1BYTE_SHIFT 8
+#define DA732X_2BYTES_SHIFT 16
+#define DA732X_3BYTES_SHIFT 24
+#define DA732X_4BYTES_SHIFT 32
+
+#define DA732X_DACS_DIS 0x0
+#define DA732X_HP_DIS 0x0
+#define DA732X_CLEAR_REG 0x0
+
+/* Calibration */
+#define DA732X_DAC_OFFSET_STEP 0x20
+#define DA732X_OUTPUT_OFFSET_STEP 0x80
+#define DA732X_HP_OUT_TRIM_VAL 0x0
+#define DA732X_WAIT_FOR_STABILIZATION 1
+#define DA732X_HPL_DAC 0
+#define DA732X_HPR_DAC 1
+#define DA732X_HP_DACS 2
+#define DA732X_HPL_AMP 0
+#define DA732X_HPR_AMP 1
+#define DA732X_HP_AMPS 2
+
+/* Clock settings */
+#define DA732X_STARTUP_DELAY 100
+#define DA732X_PLL_OUT_196608 196608000
+#define DA732X_PLL_OUT_180634 180633600
+#define DA732X_PLL_OUT_SRM 188620800
+#define DA732X_MCLK_10MHZ 10000000
+#define DA732X_MCLK_20MHZ 20000000
+#define DA732X_MCLK_40MHZ 40000000
+#define DA732X_MCLK_54MHZ 54000000
+#define DA732X_MCLK_RET_0_10MHZ 0
+#define DA732X_MCLK_VAL_0_10MHZ 1
+#define DA732X_MCLK_RET_10_20MHZ 1
+#define DA732X_MCLK_VAL_10_20MHZ 2
+#define DA732X_MCLK_RET_20_40MHZ 2
+#define DA732X_MCLK_VAL_20_40MHZ 4
+#define DA732X_MCLK_RET_40_54MHZ 3
+#define DA732X_MCLK_VAL_40_54MHZ 8
+#define DA732X_DAI_ID1 0
+#define DA732X_DAI_ID2 1
+#define DA732X_SRCCLK_PLL 0
+#define DA732X_SRCCLK_MCLK 1
+
+#define DA732X_LIN_LP_VOL 0x4F
+#define DA732X_LP_VOL 0x40
+
+/* Kcontrols */
+#define DA732X_DAC_EN_MAX 2
+#define DA732X_ADCL_MUX_MAX 2
+#define DA732X_ADCR_MUX_MAX 3
+#define DA732X_HPF_MODE_MAX 3
+#define DA732X_HPF_MODE_SHIFT 4
+#define DA732X_HPF_MUSIC_SHIFT 0
+#define DA732X_HPF_MUSIC_MAX 4
+#define DA732X_HPF_VOICE_SHIFT 4
+#define DA732X_HPF_VOICE_MAX 8
+#define DA732X_EQ_EN_MAX 1
+#define DA732X_HPF_VOICE 1
+#define DA732X_HPF_MUSIC 2
+#define DA732X_HPF_DISABLED 0
+#define DA732X_NO_INVERT 0
+#define DA732X_INVERT 1
+#define DA732X_SWITCH_MAX 1
+#define DA732X_ENABLE_CP 1
+#define DA732X_DISABLE_CP 0
+#define DA732X_DISABLE_ALL_CLKS 0
+#define DA732X_RESET_ADCS 0
+
+/* dB values */
+#define DA732X_MIC_VOL_DB_MIN 0
+#define DA732X_MIC_VOL_DB_INC 50
+#define DA732X_MIC_PRE_VOL_DB_MIN 0
+#define DA732X_MIC_PRE_VOL_DB_INC 600
+#define DA732X_AUX_VOL_DB_MIN -6000
+#define DA732X_AUX_VOL_DB_INC 150
+#define DA732X_HP_VOL_DB_MIN -2250
+#define DA732X_HP_VOL_DB_INC 150
+#define DA732X_LIN2_VOL_DB_MIN -1650
+#define DA732X_LIN2_VOL_DB_INC 150
+#define DA732X_LIN3_VOL_DB_MIN -1650
+#define DA732X_LIN3_VOL_DB_INC 150
+#define DA732X_LIN4_VOL_DB_MIN -2250
+#define DA732X_LIN4_VOL_DB_INC 150
+#define DA732X_EQ_BAND_VOL_DB_MIN -1050
+#define DA732X_EQ_BAND_VOL_DB_INC 150
+#define DA732X_DAC_VOL_DB_MIN -7725
+#define DA732X_DAC_VOL_DB_INC 75
+#define DA732X_ADC_VOL_DB_MIN 0
+#define DA732X_ADC_VOL_DB_INC -1
+#define DA732X_EQ_OVERALL_VOL_DB_MIN -1800
+#define DA732X_EQ_OVERALL_VOL_DB_INC 600
+
+#define DA732X_SOC_ENUM_DOUBLE_R(xreg, xrreg, xmax, xtext) \
+ {.reg = xreg, .reg2 = xrreg, .max = xmax, .texts = xtext}
+
+enum da732x_sysctl {
+ DA732X_SR_8KHZ = 0x1,
+ DA732X_SR_11_025KHZ = 0x2,
+ DA732X_SR_12KHZ = 0x3,
+ DA732X_SR_16KHZ = 0x5,
+ DA732X_SR_22_05KHZ = 0x6,
+ DA732X_SR_24KHZ = 0x7,
+ DA732X_SR_32KHZ = 0x9,
+ DA732X_SR_44_1KHZ = 0xA,
+ DA732X_SR_48KHZ = 0xB,
+ DA732X_SR_88_1KHZ = 0xE,
+ DA732X_SR_96KHZ = 0xF,
+};
+
+#endif /* __DA732X_H_ */
diff --git a/sound/soc/codecs/da732x_reg.h b/sound/soc/codecs/da732x_reg.h
new file mode 100644
index 0000000..bdd03ca
--- /dev/null
+++ b/sound/soc/codecs/da732x_reg.h
@@ -0,0 +1,654 @@
+/*
+ * da732x_reg.h --- Dialog DA732X ALSA SoC Audio Registers Header File
+ *
+ * Copyright (C) 2012 Dialog Semiconductor GmbH
+ *
+ * Author: Michal Hajduk <Michal.Hajduk@diasemi.com>
+ *
+ * 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 __DA732X_REG_H_
+#define __DA732X_REG_H_
+
+/* DA732X registers */
+#define DA732X_REG_STATUS_EXT 0x00
+#define DA732X_REG_STATUS 0x01
+#define DA732X_REG_REF1 0x02
+#define DA732X_REG_BIAS_EN 0x03
+#define DA732X_REG_BIAS1 0x04
+#define DA732X_REG_BIAS2 0x05
+#define DA732X_REG_BIAS3 0x06
+#define DA732X_REG_BIAS4 0x07
+#define DA732X_REG_MICBIAS2 0x0F
+#define DA732X_REG_MICBIAS1 0x10
+#define DA732X_REG_MICDET 0x11
+#define DA732X_REG_MIC1_PRE 0x12
+#define DA732X_REG_MIC1 0x13
+#define DA732X_REG_MIC2_PRE 0x14
+#define DA732X_REG_MIC2 0x15
+#define DA732X_REG_AUX1L 0x16
+#define DA732X_REG_AUX1R 0x17
+#define DA732X_REG_MIC3_PRE 0x18
+#define DA732X_REG_MIC3 0x19
+#define DA732X_REG_INP_PINBIAS 0x1A
+#define DA732X_REG_INP_ZC_EN 0x1B
+#define DA732X_REG_INP_MUX 0x1D
+#define DA732X_REG_HP_DET 0x20
+#define DA732X_REG_HPL_DAC_OFFSET 0x21
+#define DA732X_REG_HPL_DAC_OFF_CNTL 0x22
+#define DA732X_REG_HPL_OUT_OFFSET 0x23
+#define DA732X_REG_HPL 0x24
+#define DA732X_REG_HPL_VOL 0x25
+#define DA732X_REG_HPR_DAC_OFFSET 0x26
+#define DA732X_REG_HPR_DAC_OFF_CNTL 0x27
+#define DA732X_REG_HPR_OUT_OFFSET 0x28
+#define DA732X_REG_HPR 0x29
+#define DA732X_REG_HPR_VOL 0x2A
+#define DA732X_REG_LIN2 0x2B
+#define DA732X_REG_LIN3 0x2C
+#define DA732X_REG_LIN4 0x2D
+#define DA732X_REG_OUT_ZC_EN 0x2E
+#define DA732X_REG_HP_LIN1_GNDSEL 0x37
+#define DA732X_REG_CP_HP1 0x3A
+#define DA732X_REG_CP_HP2 0x3B
+#define DA732X_REG_CP_CTRL1 0x40
+#define DA732X_REG_CP_CTRL2 0x41
+#define DA732X_REG_CP_CTRL3 0x42
+#define DA732X_REG_CP_LEVEL_MASK 0x43
+#define DA732X_REG_CP_DET 0x44
+#define DA732X_REG_CP_STATUS 0x45
+#define DA732X_REG_CP_THRESH1 0x46
+#define DA732X_REG_CP_THRESH2 0x47
+#define DA732X_REG_CP_THRESH3 0x48
+#define DA732X_REG_CP_THRESH4 0x49
+#define DA732X_REG_CP_THRESH5 0x4A
+#define DA732X_REG_CP_THRESH6 0x4B
+#define DA732X_REG_CP_THRESH7 0x4C
+#define DA732X_REG_CP_THRESH8 0x4D
+#define DA732X_REG_PLL_DIV_LO 0x50
+#define DA732X_REG_PLL_DIV_MID 0x51
+#define DA732X_REG_PLL_DIV_HI 0x52
+#define DA732X_REG_PLL_CTRL 0x53
+#define DA732X_REG_CLK_CTRL 0x54
+#define DA732X_REG_CLK_DSP 0x5A
+#define DA732X_REG_CLK_EN1 0x5B
+#define DA732X_REG_CLK_EN2 0x5C
+#define DA732X_REG_CLK_EN3 0x5D
+#define DA732X_REG_CLK_EN4 0x5E
+#define DA732X_REG_CLK_EN5 0x5F
+#define DA732X_REG_AIF_MCLK 0x60
+#define DA732X_REG_AIFA1 0x61
+#define DA732X_REG_AIFA2 0x62
+#define DA732X_REG_AIFA3 0x63
+#define DA732X_REG_AIFB1 0x64
+#define DA732X_REG_AIFB2 0x65
+#define DA732X_REG_AIFB3 0x66
+#define DA732X_REG_PC_CTRL 0x6A
+#define DA732X_REG_DATA_ROUTE 0x70
+#define DA732X_REG_DSP_CTRL 0x71
+#define DA732X_REG_CIF_CTRL2 0x74
+#define DA732X_REG_HANDSHAKE 0x75
+#define DA732X_REG_MBOX0 0x76
+#define DA732X_REG_MBOX1 0x77
+#define DA732X_REG_MBOX2 0x78
+#define DA732X_REG_MBOX_STATUS 0x79
+#define DA732X_REG_SPARE1_OUT 0x7D
+#define DA732X_REG_SPARE2_OUT 0x7E
+#define DA732X_REG_SPARE1_IN 0x7F
+#define DA732X_REG_ID 0x81
+#define DA732X_REG_ADC1_PD 0x90
+#define DA732X_REG_ADC1_HPF 0x93
+#define DA732X_REG_ADC1_SEL 0x94
+#define DA732X_REG_ADC1_EQ12 0x95
+#define DA732X_REG_ADC1_EQ34 0x96
+#define DA732X_REG_ADC1_EQ5 0x97
+#define DA732X_REG_ADC2_PD 0x98
+#define DA732X_REG_ADC2_HPF 0x9B
+#define DA732X_REG_ADC2_SEL 0x9C
+#define DA732X_REG_ADC2_EQ12 0x9D
+#define DA732X_REG_ADC2_EQ34 0x9E
+#define DA732X_REG_ADC2_EQ5 0x9F
+#define DA732X_REG_DAC1_HPF 0xA0
+#define DA732X_REG_DAC1_L_VOL 0xA1
+#define DA732X_REG_DAC1_R_VOL 0xA2
+#define DA732X_REG_DAC1_SEL 0xA3
+#define DA732X_REG_DAC1_SOFTMUTE 0xA4
+#define DA732X_REG_DAC1_EQ12 0xA5
+#define DA732X_REG_DAC1_EQ34 0xA6
+#define DA732X_REG_DAC1_EQ5 0xA7
+#define DA732X_REG_DAC2_HPF 0xB0
+#define DA732X_REG_DAC2_L_VOL 0xB1
+#define DA732X_REG_DAC2_R_VOL 0xB2
+#define DA732X_REG_DAC2_SEL 0xB3
+#define DA732X_REG_DAC2_SOFTMUTE 0xB4
+#define DA732X_REG_DAC2_EQ12 0xB5
+#define DA732X_REG_DAC2_EQ34 0xB6
+#define DA732X_REG_DAC2_EQ5 0xB7
+#define DA732X_REG_DAC3_HPF 0xC0
+#define DA732X_REG_DAC3_VOL 0xC1
+#define DA732X_REG_DAC3_SEL 0xC3
+#define DA732X_REG_DAC3_SOFTMUTE 0xC4
+#define DA732X_REG_DAC3_EQ12 0xC5
+#define DA732X_REG_DAC3_EQ34 0xC6
+#define DA732X_REG_DAC3_EQ5 0xC7
+#define DA732X_REG_BIQ_BYP 0xD2
+#define DA732X_REG_DMA_CMD 0xD3
+#define DA732X_REG_DMA_ADDR0 0xD4
+#define DA732X_REG_DMA_ADDR1 0xD5
+#define DA732X_REG_DMA_DATA0 0xD6
+#define DA732X_REG_DMA_DATA1 0xD7
+#define DA732X_REG_DMA_DATA2 0xD8
+#define DA732X_REG_DMA_DATA3 0xD9
+#define DA732X_REG_DMA_STATUS 0xDA
+#define DA732X_REG_BROWNOUT 0xDF
+#define DA732X_REG_UNLOCK 0xE0
+
+#define DA732X_MAX_REG DA732X_REG_UNLOCK
+/*
+ * Bits
+ */
+
+/* DA732X_REG_STATUS_EXT (addr=0x00) */
+#define DA732X_STATUS_EXT_DSP (1 << 4)
+#define DA732X_STATUS_EXT_CLEAR (0 << 0)
+
+/* DA732X_REG_STATUS (addr=0x01) */
+#define DA732X_STATUS_PLL_LOCK (1 << 0)
+#define DA732X_STATUS_PLL_MCLK_DET (1 << 1)
+#define DA732X_STATUS_HPDET_OUT (1 << 2)
+#define DA732X_STATUS_INP_MIXDET_1 (1 << 3)
+#define DA732X_STATUS_INP_MIXDET_2 (1 << 4)
+#define DA732X_STATUS_BO_STATUS (1 << 5)
+
+/* DA732X_REG_REF1 (addr=0x02) */
+#define DA732X_VMID_FASTCHG (1 << 1)
+#define DA732X_VMID_FASTDISCHG (1 << 2)
+#define DA732X_REFBUFX2_EN (1 << 6)
+#define DA732X_REFBUFX2_DIS (0 << 6)
+
+/* DA732X_REG_BIAS_EN (addr=0x03) */
+#define DA732X_BIAS_BOOST_MASK (3 << 0)
+#define DA732X_BIAS_BOOST_100PC (0 << 0)
+#define DA732X_BIAS_BOOST_133PC (1 << 0)
+#define DA732X_BIAS_BOOST_88PC (2 << 0)
+#define DA732X_BIAS_BOOST_50PC (3 << 0)
+#define DA732X_BIAS_EN (1 << 7)
+#define DA732X_BIAS_DIS (0 << 7)
+
+/* DA732X_REG_BIAS1 (addr=0x04) */
+#define DA732X_BIAS1_HP_DAC_BIAS_MASK (3 << 0)
+#define DA732X_BIAS1_HP_DAC_BIAS_100PC (0 << 0)
+#define DA732X_BIAS1_HP_DAC_BIAS_150PC (1 << 0)
+#define DA732X_BIAS1_HP_DAC_BIAS_50PC (2 << 0)
+#define DA732X_BIAS1_HP_DAC_BIAS_75PC (3 << 0)
+#define DA732X_BIAS1_HP_OUT_BIAS_MASK (7 << 4)
+#define DA732X_BIAS1_HP_OUT_BIAS_100PC (0 << 4)
+#define DA732X_BIAS1_HP_OUT_BIAS_125PC (1 << 4)
+#define DA732X_BIAS1_HP_OUT_BIAS_150PC (2 << 4)
+#define DA732X_BIAS1_HP_OUT_BIAS_175PC (3 << 4)
+#define DA732X_BIAS1_HP_OUT_BIAS_200PC (4 << 4)
+#define DA732X_BIAS1_HP_OUT_BIAS_250PC (5 << 4)
+#define DA732X_BIAS1_HP_OUT_BIAS_300PC (6 << 4)
+#define DA732X_BIAS1_HP_OUT_BIAS_350PC (7 << 4)
+
+/* DA732X_REG_BIAS2 (addr=0x05) */
+#define DA732X_BIAS2_LINE2_DAC_BIAS_MASK (3 << 0)
+#define DA732X_BIAS2_LINE2_DAC_BIAS_100PC (0 << 0)
+#define DA732X_BIAS2_LINE2_DAC_BIAS_150PC (1 << 0)
+#define DA732X_BIAS2_LINE2_DAC_BIAS_50PC (2 << 0)
+#define DA732X_BIAS2_LINE2_DAC_BIAS_75PC (3 << 0)
+#define DA732X_BIAS2_LINE2_OUT_BIAS_MASK (7 << 4)
+#define DA732X_BIAS2_LINE2_OUT_BIAS_100PC (0 << 4)
+#define DA732X_BIAS2_LINE2_OUT_BIAS_125PC (1 << 4)
+#define DA732X_BIAS2_LINE2_OUT_BIAS_150PC (2 << 4)
+#define DA732X_BIAS2_LINE2_OUT_BIAS_175PC (3 << 4)
+#define DA732X_BIAS2_LINE2_OUT_BIAS_200PC (4 << 4)
+#define DA732X_BIAS2_LINE2_OUT_BIAS_250PC (5 << 4)
+#define DA732X_BIAS2_LINE2_OUT_BIAS_300PC (6 << 4)
+#define DA732X_BIAS2_LINE2_OUT_BIAS_350PC (7 << 4)
+
+/* DA732X_REG_BIAS3 (addr=0x06) */
+#define DA732X_BIAS3_LINE3_DAC_BIAS_MASK (3 << 0)
+#define DA732X_BIAS3_LINE3_DAC_BIAS_100PC (0 << 0)
+#define DA732X_BIAS3_LINE3_DAC_BIAS_150PC (1 << 0)
+#define DA732X_BIAS3_LINE3_DAC_BIAS_50PC (2 << 0)
+#define DA732X_BIAS3_LINE3_DAC_BIAS_75PC (3 << 0)
+#define DA732X_BIAS3_LINE3_OUT_BIAS_MASK (7 << 4)
+#define DA732X_BIAS3_LINE3_OUT_BIAS_100PC (0 << 4)
+#define DA732X_BIAS3_LINE3_OUT_BIAS_125PC (1 << 4)
+#define DA732X_BIAS3_LINE3_OUT_BIAS_150PC (2 << 4)
+#define DA732X_BIAS3_LINE3_OUT_BIAS_175PC (3 << 4)
+#define DA732X_BIAS3_LINE3_OUT_BIAS_200PC (4 << 4)
+#define DA732X_BIAS3_LINE3_OUT_BIAS_250PC (5 << 4)
+#define DA732X_BIAS3_LINE3_OUT_BIAS_300PC (6 << 4)
+#define DA732X_BIAS3_LINE3_OUT_BIAS_350PC (7 << 4)
+
+/* DA732X_REG_BIAS4 (addr=0x07) */
+#define DA732X_BIAS4_LINE4_DAC_BIAS_MASK (3 << 0)
+#define DA732X_BIAS4_LINE4_DAC_BIAS_100PC (0 << 0)
+#define DA732X_BIAS4_LINE4_DAC_BIAS_150PC (1 << 0)
+#define DA732X_BIAS4_LINE4_DAC_BIAS_50PC (2 << 0)
+#define DA732X_BIAS4_LINE4_DAC_BIAS_75PC (3 << 0)
+#define DA732X_BIAS4_LINE4_OUT_BIAS_MASK (7 << 4)
+#define DA732X_BIAS4_LINE4_OUT_BIAS_100PC (0 << 4)
+#define DA732X_BIAS4_LINE4_OUT_BIAS_125PC (1 << 4)
+#define DA732X_BIAS4_LINE4_OUT_BIAS_150PC (2 << 4)
+#define DA732X_BIAS4_LINE4_OUT_BIAS_175PC (3 << 4)
+#define DA732X_BIAS4_LINE4_OUT_BIAS_200PC (4 << 4)
+#define DA732X_BIAS4_LINE4_OUT_BIAS_250PC (5 << 4)
+#define DA732X_BIAS4_LINE4_OUT_BIAS_300PC (6 << 4)
+#define DA732X_BIAS4_LINE4_OUT_BIAS_350PC (7 << 4)
+
+/* DA732X_REG_SIF_VDD_SEL (addr=0x08) */
+#define DA732X_SIF_VDD_SEL_AIFA_VDD2 (1 << 0)
+#define DA732X_SIF_VDD_SEL_AIFB_VDD2 (1 << 1)
+#define DA732X_SIF_VDD_SEL_CIFA_VDD2 (1 << 4)
+
+/* DA732X_REG_MICBIAS2/1 (addr=0x0F/0x10) */
+#define DA732X_MICBIAS_VOLTAGE_MASK (0x0F << 0)
+#define DA732X_MICBIAS_VOLTAGE_2V (0x00 << 0)
+#define DA732X_MICBIAS_VOLTAGE_2V05 (0x01 << 0)
+#define DA732X_MICBIAS_VOLTAGE_2V1 (0x02 << 0)
+#define DA732X_MICBIAS_VOLTAGE_2V15 (0x03 << 0)
+#define DA732X_MICBIAS_VOLTAGE_2V2 (0x04 << 0)
+#define DA732X_MICBIAS_VOLTAGE_2V25 (0x05 << 0)
+#define DA732X_MICBIAS_VOLTAGE_2V3 (0x06 << 0)
+#define DA732X_MICBIAS_VOLTAGE_2V35 (0x07 << 0)
+#define DA732X_MICBIAS_VOLTAGE_2V4 (0x08 << 0)
+#define DA732X_MICBIAS_VOLTAGE_2V45 (0x09 << 0)
+#define DA732X_MICBIAS_VOLTAGE_2V5 (0x0A << 0)
+#define DA732X_MICBIAS_EN (1 << 7)
+#define DA732X_MICBIAS_EN_SHIFT 7
+#define DA732X_MICBIAS_VOLTAGE_SHIFT 0
+#define DA732X_MICBIAS_VOLTAGE_MAX 0x0B
+
+/* DA732X_REG_MICDET (addr=0x11) */
+#define DA732X_MICDET_INP_MICRES (1 << 0)
+#define DA732X_MICDET_INP_MICHOOK (1 << 1)
+#define DA732X_MICDET_INP_DEBOUNCE_PRD_8MS (0 << 0)
+#define DA732X_MICDET_INP_DEBOUNCE_PRD_16MS (1 << 0)
+#define DA732X_MICDET_INP_DEBOUNCE_PRD_32MS (2 << 0)
+#define DA732X_MICDET_INP_DEBOUNCE_PRD_64MS (3 << 0)
+#define DA732X_MICDET_INP_MICDET_EN (1 << 7)
+
+/* DA732X_REG_MIC1/2/3_PRE (addr=0x11/0x14/0x18) */
+#define DA732X_MICBOOST_MASK 0x7
+#define DA732X_MICBOOST_SHIFT 0
+#define DA732X_MICBOOST_MIN 0x1
+#define DA732X_MICBOOST_MAX DA732X_MICBOOST_MASK
+
+/* DA732X_REG_MIC1/2/3 (addr=0x13/0x15/0x19) */
+#define DA732X_MIC_VOL_SHIFT 0
+#define DA732X_MIC_VOL_VAL_MASK 0x1F
+#define DA732X_MIC_MUTE_SHIFT 6
+#define DA732X_MIC_EN_SHIFT 7
+#define DA732X_MIC_VOL_VAL_MIN 0x7
+#define DA732X_MIC_VOL_VAL_MAX DA732X_MIC_VOL_VAL_MASK
+
+/* DA732X_REG_AUX1L/R (addr=0x16/0x17) */
+#define DA732X_AUX_VOL_SHIFT 0
+#define DA732X_AUX_VOL_MASK 0x7
+#define DA732X_AUX_MUTE_SHIFT 6
+#define DA732X_AUX_EN_SHIFT 7
+#define DA732X_AUX_VOL_VAL_MAX DA732X_AUX_VOL_MASK
+
+/* DA732X_REG_INP_PINBIAS (addr=0x1A) */
+#define DA732X_INP_MICL_PINBIAS_EN (1 << 0)
+#define DA732X_INP_MICR_PINBIAS_EN (1 << 1)
+#define DA732X_INP_AUX1L_PINBIAS_EN (1 << 2)
+#define DA732X_INP_AUX1R_PINBIAS_EN (1 << 3)
+#define DA732X_INP_AUX2_PINBIAS_EN (1 << 4)
+
+/* DA732X_REG_INP_ZC_EN (addr=0x1B) */
+#define DA732X_MIC1_PRE_ZC_EN (1 << 0)
+#define DA732X_MIC1_ZC_EN (1 << 1)
+#define DA732X_MIC2_PRE_ZC_EN (1 << 2)
+#define DA732X_MIC2_ZC_EN (1 << 3)
+#define DA732X_AUXL_ZC_EN (1 << 4)
+#define DA732X_AUXR_ZC_EN (1 << 5)
+#define DA732X_MIC3_PRE_ZC_EN (1 << 6)
+#define DA732X_MIC3_ZC_EN (1 << 7)
+
+/* DA732X_REG_INP_MUX (addr=0x1D) */
+#define DA732X_INP_ADC1L_MUX_SEL_AUX1L (0 << 0)
+#define DA732X_INP_ADC1L_MUX_SEL_MIC1 (1 << 0)
+#define DA732X_INP_ADC1R_MUX_SEL_MASK (3 << 2)
+#define DA732X_INP_ADC1R_MUX_SEL_AUX1R (0 << 2)
+#define DA732X_INP_ADC1R_MUX_SEL_MIC2 (1 << 2)
+#define DA732X_INP_ADC1R_MUX_SEL_MIC3 (2 << 2)
+#define DA732X_INP_ADC2L_MUX_SEL_AUX1L (0 << 4)
+#define DA732X_INP_ADC2L_MUX_SEL_MICL (1 << 4)
+#define DA732X_INP_ADC2R_MUX_SEL_MASK (3 << 6)
+#define DA732X_INP_ADC2R_MUX_SEL_AUX1R (0 << 6)
+#define DA732X_INP_ADC2R_MUX_SEL_MICR (1 << 6)
+#define DA732X_INP_ADC2R_MUX_SEL_AUX2 (2 << 6)
+#define DA732X_ADC1L_MUX_SEL_SHIFT 0
+#define DA732X_ADC1R_MUX_SEL_SHIFT 2
+#define DA732X_ADC2L_MUX_SEL_SHIFT 4
+#define DA732X_ADC2R_MUX_SEL_SHIFT 6
+
+/* DA732X_REG_HP_DET (addr=0x20) */
+#define DA732X_HP_DET_AZ (1 << 0)
+#define DA732X_HP_DET_SEL1 (1 << 1)
+#define DA732X_HP_DET_IS_MASK (3 << 2)
+#define DA732X_HP_DET_IS_0_5UA (0 << 2)
+#define DA732X_HP_DET_IS_1UA (1 << 2)
+#define DA732X_HP_DET_IS_2UA (2 << 2)
+#define DA732X_HP_DET_IS_4UA (3 << 2)
+#define DA732X_HP_DET_RS_MASK (3 << 4)
+#define DA732X_HP_DET_RS_INFINITE (0 << 4)
+#define DA732X_HP_DET_RS_100KOHM (1 << 4)
+#define DA732X_HP_DET_RS_10KOHM (2 << 4)
+#define DA732X_HP_DET_RS_1KOHM (3 << 4)
+#define DA732X_HP_DET_EN (1 << 7)
+
+/* DA732X_REG_HPL_DAC_OFFSET (addr=0x21/0x26) */
+#define DA732X_HP_DAC_OFFSET_TRIM_MASK (0x3F << 0)
+#define DA732X_HP_DAC_OFFSET_DAC_SIGN (1 << 6)
+
+/* DA732X_REG_HPL_DAC_OFF_CNTL (addr=0x22/0x27) */
+#define DA732X_HP_DAC_OFF_CNTL_CONT_MASK (7 << 0)
+#define DA732X_HP_DAC_OFF_CNTL_COMPO (1 << 3)
+#define DA732X_HP_DAC_OFF_CALIBRATION (1 << 0)
+#define DA732X_HP_DAC_OFF_SCALE_STEPS (1 << 1)
+#define DA732X_HP_DAC_OFF_MASK 0x7F
+#define DA732X_HP_DAC_COMPO_SHIFT 3
+
+/* DA732X_REG_HPL_OUT_OFFSET (addr=0x23/0x28) */
+#define DA732X_HP_OUT_OFFSET_MASK (0xFF << 0)
+#define DA732X_HP_DAC_OFFSET_TRIM_VAL 0x7F
+
+/* DA732X_REG_HPL/R (addr=0x24/0x29) */
+#define DA732X_HP_OUT_SIGN (1 << 0)
+#define DA732X_HP_OUT_COMP (1 << 1)
+#define DA732X_HP_OUT_RESERVED (1 << 2)
+#define DA732X_HP_OUT_COMPO (1 << 3)
+#define DA732X_HP_OUT_DAC_EN (1 << 4)
+#define DA732X_HP_OUT_HIZ_EN (1 << 5)
+#define DA732X_HP_OUT_HIZ_DIS (0 << 5)
+#define DA732X_HP_OUT_MUTE (1 << 6)
+#define DA732X_HP_OUT_EN (1 << 7)
+#define DA732X_HP_OUT_COMPO_SHIFT 3
+#define DA732X_HP_OUT_DAC_EN_SHIFT 4
+#define DA732X_HP_HIZ_SHIFT 5
+#define DA732X_HP_MUTE_SHIFT 6
+#define DA732X_HP_OUT_EN_SHIFT 7
+
+#define DA732X_OUT_HIZ_EN (1 << 5)
+#define DA732X_OUT_HIZ_DIS (0 << 5)
+
+/* DA732X_REG_HPL/R_VOL (addr=0x25/0x2A) */
+#define DA732X_HP_VOL_VAL_MASK 0xF
+#define DA732X_HP_VOL_SHIFT 0
+#define DA732X_HP_VOL_VAL_MAX DA732X_HP_VOL_VAL_MASK
+
+/* DA732X_REG_LIN2/3/4 (addr=0x2B/0x2C/0x2D) */
+#define DA732X_LOUT_VOL_SHIFT 0
+#define DA732X_LOUT_VOL_MASK 0x0F
+#define DA732X_LOUT_DAC_OFF (0 << 4)
+#define DA732X_LOUT_DAC_EN (1 << 4)
+#define DA732X_LOUT_HIZ_N_DIS (0 << 5)
+#define DA732X_LOUT_HIZ_N_EN (1 << 5)
+#define DA732X_LOUT_UNMUTED (0 << 6)
+#define DA732X_LOUT_MUTED (1 << 6)
+#define DA732X_LOUT_EN (0 << 7)
+#define DA732X_LOUT_DIS (1 << 7)
+#define DA732X_LOUT_DAC_EN_SHIFT 4
+#define DA732X_LOUT_MUTE_SHIFT 6
+#define DA732X_LIN_OUT_EN_SHIFT 7
+#define DA732X_LOUT_VOL_VAL_MAX DA732X_LOUT_VOL_MASK
+
+/* DA732X_REG_OUT_ZC_EN (addr=0x2E) */
+#define DA732X_HPL_ZC_EN_SHIFT 0
+#define DA732X_HPR_ZC_EN_SHIFT 1
+#define DA732X_HPL_ZC_EN (1 << 0)
+#define DA732X_HPL_ZC_DIS (0 << 0)
+#define DA732X_HPR_ZC_EN (1 << 1)
+#define DA732X_HPR_ZC_DIS (0 << 1)
+#define DA732X_LIN2_ZC_EN (1 << 2)
+#define DA732X_LIN2_ZC_DIS (0 << 2)
+#define DA732X_LIN3_ZC_EN (1 << 3)
+#define DA732X_LIN3_ZC_DIS (0 << 3)
+#define DA732X_LIN4_ZC_EN (1 << 4)
+#define DA732X_LIN4_ZC_DIS (0 << 4)
+
+/* DA732X_REG_HP_LIN1_GNDSEL (addr=0x37) */
+#define DA732X_HP_OUT_GNDSEL (1 << 0)
+
+/* DA732X_REG_CP_HP2 (addr=0x3a) */
+#define DA732X_HP_CP_PULSESKIP (1 << 0)
+#define DA732X_HP_CP_REG (1 << 1)
+#define DA732X_HP_CP_EN (1 << 3)
+#define DA732X_HP_CP_DIS (0 << 3)
+
+/* DA732X_REG_CP_CTRL1 (addr=0x40) */
+#define DA732X_CP_MODE_MASK (7 << 1)
+#define DA732X_CP_CTRL_STANDBY (0 << 1)
+#define DA732X_CP_CTRL_CPVDD6 (2 << 1)
+#define DA732X_CP_CTRL_CPVDD5 (3 << 1)
+#define DA732X_CP_CTRL_CPVDD4 (4 << 1)
+#define DA732X_CP_CTRL_CPVDD3 (5 << 1)
+#define DA732X_CP_CTRL_CPVDD2 (6 << 1)
+#define DA732X_CP_CTRL_CPVDD1 (7 << 1)
+#define DA723X_CP_DIS (0 << 7)
+#define DA732X_CP_EN (1 << 7)
+
+/* DA732X_REG_CP_CTRL2 (addr=0x41) */
+#define DA732X_CP_BOOST (1 << 0)
+#define DA732X_CP_MANAGE_MAGNITUDE (2 << 2)
+
+/* DA732X_REG_CP_CTRL3 (addr=0x42) */
+#define DA732X_CP_1MHZ (0 << 0)
+#define DA732X_CP_500KHZ (1 << 0)
+#define DA732X_CP_250KHZ (2 << 0)
+#define DA732X_CP_125KHZ (3 << 0)
+#define DA732X_CP_63KHZ (4 << 0)
+#define DA732X_CP_0KHZ (5 << 0)
+
+/* DA732X_REG_PLL_CTRL (addr=0x53) */
+#define DA732X_PLL_INDIV_MASK (3 << 0)
+#define DA732X_PLL_SRM_EN (1 << 2)
+#define DA732X_PLL_EN (1 << 7)
+#define DA732X_PLL_BYPASS (0 << 0)
+
+/* DA732X_REG_CLK_CTRL (addr=0x54) */
+#define DA732X_SR1_MASK (0xF)
+#define DA732X_SR2_MASK (0xF0)
+
+/* DA732X_REG_CLK_DSP (addr=0x5A) */
+#define DA732X_DSP_FREQ_MASK (7 << 0)
+#define DA732X_DSP_FREQ_12MHZ (0 << 0)
+#define DA732X_DSP_FREQ_24MHZ (1 << 0)
+#define DA732X_DSP_FREQ_36MHZ (2 << 0)
+#define DA732X_DSP_FREQ_48MHZ (3 << 0)
+#define DA732X_DSP_FREQ_60MHZ (4 << 0)
+#define DA732X_DSP_FREQ_72MHZ (5 << 0)
+#define DA732X_DSP_FREQ_84MHZ (6 << 0)
+#define DA732X_DSP_FREQ_96MHZ (7 << 0)
+
+/* DA732X_REG_CLK_EN1 (addr=0x5B) */
+#define DA732X_DSP_CLK_EN (1 << 0)
+#define DA732X_SYS3_CLK_EN (1 << 1)
+#define DA732X_DSP12_CLK_EN (1 << 2)
+#define DA732X_PC_CLK_EN (1 << 3)
+#define DA732X_MCLK_SQR_EN (1 << 7)
+
+/* DA732X_REG_CLK_EN2 (addr=0x5C) */
+#define DA732X_UART_CLK_EN (1 << 1)
+#define DA732X_CP_CLK_EN (1 << 2)
+#define DA732X_CP_CLK_DIS (0 << 2)
+
+/* DA732X_REG_CLK_EN3 (addr=0x5D) */
+#define DA732X_ADCA_BB_CLK_EN (1 << 0)
+#define DA732X_ADCC_BB_CLK_EN (1 << 4)
+
+/* DA732X_REG_CLK_EN4 (addr=0x5E) */
+#define DA732X_DACA_BB_CLK_EN (1 << 0)
+#define DA732X_DACC_BB_CLK_EN (1 << 4)
+#define DA732X_DACA_BB_CLK_SHIFT 0
+#define DA732X_DACC_BB_CLK_SHIFT 4
+
+/* DA732X_REG_CLK_EN5 (addr=0x5F) */
+#define DA732X_DACE_BB_CLK_EN (1 << 0)
+#define DA732X_DACE_BB_CLK_SHIFT 0
+
+/* DA732X_REG_AIF_MCLK (addr=0x60) */
+#define DA732X_AIFM_FRAME_64 (1 << 2)
+#define DA732X_AIFM_SRC_SEL_AIFA (1 << 6)
+#define DA732X_CLK_GENERATION_AIF_A (1 << 4)
+#define DA732X_NO_CLK_GENERATION 0x0
+
+/* DA732X_REG_AIFA1 (addr=0x61) */
+#define DA732X_AIF_WORD_MASK (0x3 << 0)
+#define DA732X_AIF_WORD_16 (0 << 0)
+#define DA732X_AIF_WORD_20 (1 << 0)
+#define DA732X_AIF_WORD_24 (2 << 0)
+#define DA732X_AIF_WORD_32 (3 << 0)
+#define DA732X_AIF_TDM_MONO_SHIFT (1 << 6)
+#define DA732X_AIF1_CLK_MASK (1 << 7)
+#define DA732X_AIF_SLAVE (0 << 7)
+#define DA732X_AIF_CLK_FROM_SRC (1 << 7)
+
+/* DA732X_REG_AIFA3 (addr=0x63) */
+#define DA732X_AIF_MODE_SHIFT 0
+#define DA732X_AIF_MODE_MASK 0x3
+#define DA732X_AIF_I2S_MODE (0 << 0)
+#define DA732X_AIF_LEFT_J_MODE (1 << 0)
+#define DA732X_AIF_RIGHT_J_MODE (2 << 0)
+#define DA732X_AIF_DSP_MODE (3 << 0)
+#define DA732X_AIF_WCLK_INV (1 << 4)
+#define DA732X_AIF_BCLK_INV (1 << 5)
+#define DA732X_AIF_EN (1 << 7)
+#define DA732X_AIF_EN_SHIFT 7
+
+/* DA732X_REG_PC_CTRL (addr=0x6a) */
+#define DA732X_PC_PULSE_AIFA (0 << 0)
+#define DA732X_PC_PULSE_AIFB (1 << 0)
+#define DA732X_PC_RESYNC_AUT (1 << 6)
+#define DA732X_PC_RESYNC_NOT_AUT (0 << 6)
+#define DA732X_PC_SAME (1 << 7)
+
+/* DA732X_REG_DATA_ROUTE (addr=0x70) */
+#define DA732X_ADC1_TO_AIFA (0 << 0)
+#define DA732X_DSP_TO_AIFA (1 << 0)
+#define DA732X_ADC2_TO_AIFB (0 << 1)
+#define DA732X_DSP_TO_AIFB (1 << 1)
+#define DA732X_AIFA_TO_DAC1L (0 << 2)
+#define DA732X_DSP_TO_DAC1L (1 << 2)
+#define DA732X_AIFA_TO_DAC1R (0 << 3)
+#define DA732X_DSP_TO_DAC1R (1 << 3)
+#define DA732X_AIFB_TO_DAC2L (0 << 4)
+#define DA732X_DSP_TO_DAC2L (1 << 4)
+#define DA732X_AIFB_TO_DAC2R (0 << 5)
+#define DA732X_DSP_TO_DAC2R (1 << 5)
+#define DA732X_AIFB_TO_DAC3 (0 << 6)
+#define DA732X_DSP_TO_DAC3 (1 << 6)
+#define DA732X_BYPASS_DSP (0 << 0)
+#define DA732X_ALL_TO_DSP (0x7F << 0)
+
+/* DA732X_REG_DSP_CTRL (addr=0x71) */
+#define DA732X_DIGITAL_EN (1 << 0)
+#define DA732X_DIGITAL_RESET (0 << 0)
+#define DA732X_DSP_CORE_EN (1 << 1)
+#define DA732X_DSP_CORE_RESET (0 << 1)
+
+/* DA732X_REG_SPARE1_OUT (addr=0x7D)*/
+#define DA732X_HP_DRIVER_EN (1 << 0)
+#define DA732X_HP_GATE_LOW (1 << 2)
+#define DA732X_HP_LOOP_GAIN_CTRL (1 << 3)
+
+/* DA732X_REG_ID (addr=0x81)*/
+#define DA732X_ID_MINOR_MASK (0xF << 0)
+#define DA732X_ID_MAJOR_MASK (0xF << 4)
+
+/* DA732X_REG_ADC1/2_PD (addr=0x90/0x98) */
+#define DA732X_ADC_RST_MASK (0x3 << 0)
+#define DA732X_ADC_PD_MASK (0x3 << 2)
+#define DA732X_ADC_SET_ACT (0x3 << 0)
+#define DA732X_ADC_SET_RST (0x0 << 0)
+#define DA732X_ADC_ON (0x3 << 2)
+#define DA732X_ADC_OFF (0x0 << 2)
+
+/* DA732X_REG_ADC1/2_SEL (addr=0x94/0x9C) */
+#define DA732X_ADC_VOL_VAL_MASK 0x7
+#define DA732X_ADCL_VOL_SHIFT 0
+#define DA732X_ADCR_VOL_SHIFT 4
+#define DA732X_ADCL_EN_SHIFT 2
+#define DA732X_ADCR_EN_SHIFT 3
+#define DA732X_ADCL_EN (1 << 2)
+#define DA732X_ADCR_EN (1 << 3)
+#define DA732X_ADC_VOL_VAL_MAX DA732X_ADC_VOL_VAL_MASK
+
+/*
+ * DA732X_REG_ADC1/2_HPF (addr=0x93/0x9b)
+ * DA732x_REG_DAC1/2/3_HPG (addr=0xA5/0xB5/0xC5)
+ */
+#define DA732X_HPF_MUSIC_EN (1 << 3)
+#define DA732X_HPF_VOICE_EN ((1 << 3) | (1 << 7))
+#define DA732X_HPF_MASK ((1 << 3) | (1 << 7))
+#define DA732X_HPF_DIS ((0 << 3) | (0 << 7))
+
+/* DA732X_REG_DAC1/2/3_VOL */
+#define DA732X_DAC_VOL_VAL_MASK 0x7F
+#define DA732X_DAC_VOL_SHIFT 0
+#define DA732X_DAC_VOL_VAL_MAX DA732X_DAC_VOL_VAL_MASK
+
+/* DA732X_REG_DAC1/2/3_SEL (addr=0xA3/0xB3/0xC3) */
+#define DA732X_DACL_EN_SHIFT 3
+#define DA732X_DACR_EN_SHIFT 7
+#define DA732X_DACL_MUTE_SHIFT 2
+#define DA732X_DACR_MUTE_SHIFT 6
+#define DA732X_DACL_EN (1 << 3)
+#define DA732X_DACR_EN (1 << 7)
+#define DA732X_DACL_SDM (1 << 0)
+#define DA732X_DACR_SDM (1 << 4)
+#define DA732X_DACL_MUTE (1 << 2)
+#define DA732X_DACR_MUTE (1 << 6)
+
+/* DA732X_REG_DAC_SOFTMUTE (addr=0xA4/0xB4/0xC4) */
+#define DA732X_SOFTMUTE_EN (1 << 7)
+#define DA732X_GAIN_RAMPED (1 << 6)
+#define DA732X_16_SAMPLES (4 << 0)
+#define DA732X_SOFTMUTE_MASK (1 << 7)
+#define DA732X_SOFTMUTE_SHIFT 7
+
+/*
+ * DA732x_REG_ADC1/2_EQ12 (addr=0x95/0x9D)
+ * DA732x_REG_ADC1/2_EQ34 (addr=0x96/0x9E)
+ * DA732x_REG_ADC1/2_EQ5 (addr=0x97/0x9F)
+ * DA732x_REG_DAC1/2/3_EQ12 (addr=0xA5/0xB5/0xC5)
+ * DA732x_REG_DAC1/2/3_EQ34 (addr=0xA6/0xB6/0xC6)
+ * DA732x_REG_DAC1/2/3_EQ5 (addr=0xA7/0xB7/0xB7)
+ */
+#define DA732X_EQ_VOL_VAL_MASK 0xF
+#define DA732X_EQ_BAND1_SHIFT 0
+#define DA732X_EQ_BAND2_SHIFT 4
+#define DA732X_EQ_BAND3_SHIFT 0
+#define DA732X_EQ_BAND4_SHIFT 4
+#define DA732X_EQ_BAND5_SHIFT 0
+#define DA732X_EQ_OVERALL_SHIFT 4
+#define DA732X_EQ_OVERALL_VOL_VAL_MASK 0x3
+#define DA732X_EQ_DIS (0 << 7)
+#define DA732X_EQ_EN (1 << 7)
+#define DA732X_EQ_EN_SHIFT 7
+#define DA732X_EQ_VOL_VAL_MAX DA732X_EQ_VOL_VAL_MASK
+#define DA732X_EQ_OVERALL_VOL_VAL_MAX DA732X_EQ_OVERALL_VOL_VAL_MASK
+
+/* DA732X_REG_DMA_CMD (addr=0xD3) */
+#define DA732X_SEL_DSP_DMA_MASK (3 << 0)
+#define DA732X_SEL_DSP_DMA_DIS (0 << 0)
+#define DA732X_SEL_DSP_DMA_PMEM (1 << 0)
+#define DA732X_SEL_DSP_DMA_XMEM (2 << 0)
+#define DA732X_SEL_DSP_DMA_YMEM (3 << 0)
+#define DA732X_DSP_RW_MASK (1 << 4)
+#define DA732X_DSP_DMA_WRITE (0 << 4)
+#define DA732X_DSP_DMA_READ (1 << 4)
+
+/* DA732X_REG_DMA_STATUS (addr=0xDA) */
+#define DA732X_DSP_DMA_FREE (0 << 0)
+#define DA732X_DSP_DMA_BUSY (1 << 0)
+
+#endif /* __DA732X_REG_H_ */
diff --git a/sound/soc/codecs/isabelle.c b/sound/soc/codecs/isabelle.c
new file mode 100644
index 0000000..5d8f39e
--- /dev/null
+++ b/sound/soc/codecs/isabelle.c
@@ -0,0 +1,1176 @@
+/*
+ * isabelle.c - Low power high fidelity audio codec driver
+ *
+ * Copyright (c) 2012 Texas Instruments, Inc
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; version 2 of the License.
+ *
+ *
+ * Initially based on sound/soc/codecs/twl6040.c
+ *
+ */
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/version.h>
+#include <linux/kernel.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/regmap.h>
+#include <linux/i2c.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/tlv.h>
+#include <sound/jack.h>
+#include <sound/initval.h>
+#include <asm/div64.h>
+#include "isabelle.h"
+
+
+/* Register default values for ISABELLE driver. */
+static struct reg_default isabelle_reg_defs[] = {
+ { 0, 0x00 },
+ { 1, 0x00 },
+ { 2, 0x00 },
+ { 3, 0x00 },
+ { 4, 0x00 },
+ { 5, 0x00 },
+ { 6, 0x00 },
+ { 7, 0x00 },
+ { 8, 0x00 },
+ { 9, 0x00 },
+ { 10, 0x00 },
+ { 11, 0x00 },
+ { 12, 0x00 },
+ { 13, 0x00 },
+ { 14, 0x00 },
+ { 15, 0x00 },
+ { 16, 0x00 },
+ { 17, 0x00 },
+ { 18, 0x00 },
+ { 19, 0x00 },
+ { 20, 0x00 },
+ { 21, 0x02 },
+ { 22, 0x02 },
+ { 23, 0x02 },
+ { 24, 0x02 },
+ { 25, 0x0F },
+ { 26, 0x8F },
+ { 27, 0x0F },
+ { 28, 0x8F },
+ { 29, 0x00 },
+ { 30, 0x00 },
+ { 31, 0x00 },
+ { 32, 0x00 },
+ { 33, 0x00 },
+ { 34, 0x00 },
+ { 35, 0x00 },
+ { 36, 0x00 },
+ { 37, 0x00 },
+ { 38, 0x00 },
+ { 39, 0x00 },
+ { 40, 0x00 },
+ { 41, 0x00 },
+ { 42, 0x00 },
+ { 43, 0x00 },
+ { 44, 0x00 },
+ { 45, 0x00 },
+ { 46, 0x00 },
+ { 47, 0x00 },
+ { 48, 0x00 },
+ { 49, 0x00 },
+ { 50, 0x00 },
+ { 51, 0x00 },
+ { 52, 0x00 },
+ { 53, 0x00 },
+ { 54, 0x00 },
+ { 55, 0x00 },
+ { 56, 0x00 },
+ { 57, 0x00 },
+ { 58, 0x00 },
+ { 59, 0x00 },
+ { 60, 0x00 },
+ { 61, 0x00 },
+ { 62, 0x00 },
+ { 63, 0x00 },
+ { 64, 0x00 },
+ { 65, 0x00 },
+ { 66, 0x00 },
+ { 67, 0x00 },
+ { 68, 0x00 },
+ { 69, 0x90 },
+ { 70, 0x90 },
+ { 71, 0x90 },
+ { 72, 0x00 },
+ { 73, 0x00 },
+ { 74, 0x00 },
+ { 75, 0x00 },
+ { 76, 0x00 },
+ { 77, 0x00 },
+ { 78, 0x00 },
+ { 79, 0x00 },
+ { 80, 0x00 },
+ { 81, 0x00 },
+ { 82, 0x00 },
+ { 83, 0x00 },
+ { 84, 0x00 },
+ { 85, 0x07 },
+ { 86, 0x00 },
+ { 87, 0x00 },
+ { 88, 0x00 },
+ { 89, 0x07 },
+ { 90, 0x80 },
+ { 91, 0x07 },
+ { 92, 0x07 },
+ { 93, 0x00 },
+ { 94, 0x00 },
+ { 95, 0x00 },
+ { 96, 0x00 },
+ { 97, 0x00 },
+ { 98, 0x00 },
+ { 99, 0x00 },
+};
+
+static const char *isabelle_rx1_texts[] = {"VRX1", "ARX1"};
+static const char *isabelle_rx2_texts[] = {"VRX2", "ARX2"};
+
+static const struct soc_enum isabelle_rx1_enum[] = {
+ SOC_ENUM_SINGLE(ISABELLE_VOICE_HPF_CFG_REG, 3, 1, isabelle_rx1_texts),
+ SOC_ENUM_SINGLE(ISABELLE_AUDIO_HPF_CFG_REG, 5, 1, isabelle_rx1_texts),
+};
+
+static const struct soc_enum isabelle_rx2_enum[] = {
+ SOC_ENUM_SINGLE(ISABELLE_VOICE_HPF_CFG_REG, 2, 1, isabelle_rx2_texts),
+ SOC_ENUM_SINGLE(ISABELLE_AUDIO_HPF_CFG_REG, 4, 1, isabelle_rx2_texts),
+};
+
+/* Headset DAC playback switches */
+static const struct snd_kcontrol_new rx1_mux_controls =
+ SOC_DAPM_ENUM("Route", isabelle_rx1_enum);
+
+static const struct snd_kcontrol_new rx2_mux_controls =
+ SOC_DAPM_ENUM("Route", isabelle_rx2_enum);
+
+/* TX input selection */
+static const char *isabelle_atx_texts[] = {"AMIC1", "DMIC"};
+static const char *isabelle_vtx_texts[] = {"AMIC2", "DMIC"};
+
+static const struct soc_enum isabelle_atx_enum[] = {
+ SOC_ENUM_SINGLE(ISABELLE_AMIC_CFG_REG, 7, 1, isabelle_atx_texts),
+ SOC_ENUM_SINGLE(ISABELLE_DMIC_CFG_REG, 0, 1, isabelle_atx_texts),
+};
+
+static const struct soc_enum isabelle_vtx_enum[] = {
+ SOC_ENUM_SINGLE(ISABELLE_AMIC_CFG_REG, 6, 1, isabelle_vtx_texts),
+ SOC_ENUM_SINGLE(ISABELLE_DMIC_CFG_REG, 0, 1, isabelle_vtx_texts),
+};
+
+static const struct snd_kcontrol_new atx_mux_controls =
+ SOC_DAPM_ENUM("Route", isabelle_atx_enum);
+
+static const struct snd_kcontrol_new vtx_mux_controls =
+ SOC_DAPM_ENUM("Route", isabelle_vtx_enum);
+
+/* Left analog microphone selection */
+static const char *isabelle_amic1_texts[] = {
+ "Main Mic", "Headset Mic", "Aux/FM Left"};
+
+/* Left analog microphone selection */
+static const char *isabelle_amic2_texts[] = {"Sub Mic", "Aux/FM Right"};
+
+static const struct soc_enum isabelle_amic1_enum[] = {
+ SOC_ENUM_SINGLE(ISABELLE_AMIC_CFG_REG, 5,
+ ARRAY_SIZE(isabelle_amic1_texts),
+ isabelle_amic1_texts),
+};
+
+static const struct soc_enum isabelle_amic2_enum[] = {
+ SOC_ENUM_SINGLE(ISABELLE_AMIC_CFG_REG, 4,
+ ARRAY_SIZE(isabelle_amic2_texts),
+ isabelle_amic2_texts),
+};
+
+static const struct snd_kcontrol_new amic1_control =
+ SOC_DAPM_ENUM("Route", isabelle_amic1_enum);
+
+static const struct snd_kcontrol_new amic2_control =
+ SOC_DAPM_ENUM("Route", isabelle_amic2_enum);
+
+static const char *isabelle_st_audio_texts[] = {"ATX1", "ATX2"};
+
+static const char *isabelle_st_voice_texts[] = {"VTX1", "VTX2"};
+
+static const struct soc_enum isabelle_st_audio_enum[] = {
+ SOC_ENUM_SINGLE(ISABELLE_ATX_STPGA1_CFG_REG, 7, 1,
+ isabelle_st_audio_texts),
+ SOC_ENUM_SINGLE(ISABELLE_ATX_STPGA2_CFG_REG, 7, 1,
+ isabelle_st_audio_texts),
+};
+
+static const struct soc_enum isabelle_st_voice_enum[] = {
+ SOC_ENUM_SINGLE(ISABELLE_VTX_STPGA1_CFG_REG, 7, 1,
+ isabelle_st_voice_texts),
+ SOC_ENUM_SINGLE(ISABELLE_VTX2_STPGA2_CFG_REG, 7, 1,
+ isabelle_st_voice_texts),
+};
+
+static const struct snd_kcontrol_new st_audio_control =
+ SOC_DAPM_ENUM("Route", isabelle_st_audio_enum);
+
+static const struct snd_kcontrol_new st_voice_control =
+ SOC_DAPM_ENUM("Route", isabelle_st_voice_enum);
+
+/* Mixer controls */
+static const struct snd_kcontrol_new isabelle_hs_left_mixer_controls[] = {
+SOC_DAPM_SINGLE("DAC1L Playback Switch", ISABELLE_HSDRV_CFG1_REG, 7, 1, 0),
+SOC_DAPM_SINGLE("APGA1 Playback Switch", ISABELLE_HSDRV_CFG1_REG, 6, 1, 0),
+};
+
+static const struct snd_kcontrol_new isabelle_hs_right_mixer_controls[] = {
+SOC_DAPM_SINGLE("DAC1R Playback Switch", ISABELLE_HSDRV_CFG1_REG, 5, 1, 0),
+SOC_DAPM_SINGLE("APGA2 Playback Switch", ISABELLE_HSDRV_CFG1_REG, 4, 1, 0),
+};
+
+static const struct snd_kcontrol_new isabelle_hf_left_mixer_controls[] = {
+SOC_DAPM_SINGLE("DAC2L Playback Switch", ISABELLE_HFLPGA_CFG_REG, 7, 1, 0),
+SOC_DAPM_SINGLE("APGA1 Playback Switch", ISABELLE_HFLPGA_CFG_REG, 6, 1, 0),
+};
+
+static const struct snd_kcontrol_new isabelle_hf_right_mixer_controls[] = {
+SOC_DAPM_SINGLE("DAC2R Playback Switch", ISABELLE_HFRPGA_CFG_REG, 7, 1, 0),
+SOC_DAPM_SINGLE("APGA2 Playback Switch", ISABELLE_HFRPGA_CFG_REG, 6, 1, 0),
+};
+
+static const struct snd_kcontrol_new isabelle_ep_mixer_controls[] = {
+SOC_DAPM_SINGLE("DAC2L Playback Switch", ISABELLE_EARDRV_CFG1_REG, 7, 1, 0),
+SOC_DAPM_SINGLE("APGA1 Playback Switch", ISABELLE_EARDRV_CFG1_REG, 6, 1, 0),
+};
+
+static const struct snd_kcontrol_new isabelle_aux_left_mixer_controls[] = {
+SOC_DAPM_SINGLE("DAC3L Playback Switch", ISABELLE_LINEAMP_CFG_REG, 7, 1, 0),
+SOC_DAPM_SINGLE("APGA1 Playback Switch", ISABELLE_LINEAMP_CFG_REG, 6, 1, 0),
+};
+
+static const struct snd_kcontrol_new isabelle_aux_right_mixer_controls[] = {
+SOC_DAPM_SINGLE("DAC3R Playback Switch", ISABELLE_LINEAMP_CFG_REG, 5, 1, 0),
+SOC_DAPM_SINGLE("APGA2 Playback Switch", ISABELLE_LINEAMP_CFG_REG, 4, 1, 0),
+};
+
+static const struct snd_kcontrol_new isabelle_dpga1_left_mixer_controls[] = {
+SOC_DAPM_SINGLE("RX1 Playback Switch", ISABELLE_DPGA1LR_IN_SEL_REG, 7, 1, 0),
+SOC_DAPM_SINGLE("RX3 Playback Switch", ISABELLE_DPGA1LR_IN_SEL_REG, 6, 1, 0),
+SOC_DAPM_SINGLE("RX5 Playback Switch", ISABELLE_DPGA1LR_IN_SEL_REG, 5, 1, 0),
+};
+
+static const struct snd_kcontrol_new isabelle_dpga1_right_mixer_controls[] = {
+SOC_DAPM_SINGLE("RX2 Playback Switch", ISABELLE_DPGA1LR_IN_SEL_REG, 3, 1, 0),
+SOC_DAPM_SINGLE("RX4 Playback Switch", ISABELLE_DPGA1LR_IN_SEL_REG, 2, 1, 0),
+SOC_DAPM_SINGLE("RX6 Playback Switch", ISABELLE_DPGA1LR_IN_SEL_REG, 1, 1, 0),
+};
+
+static const struct snd_kcontrol_new isabelle_dpga2_left_mixer_controls[] = {
+SOC_DAPM_SINGLE("RX1 Playback Switch", ISABELLE_DPGA2L_IN_SEL_REG, 7, 1, 0),
+SOC_DAPM_SINGLE("RX2 Playback Switch", ISABELLE_DPGA2L_IN_SEL_REG, 6, 1, 0),
+SOC_DAPM_SINGLE("RX3 Playback Switch", ISABELLE_DPGA2L_IN_SEL_REG, 5, 1, 0),
+SOC_DAPM_SINGLE("RX4 Playback Switch", ISABELLE_DPGA2L_IN_SEL_REG, 4, 1, 0),
+SOC_DAPM_SINGLE("RX5 Playback Switch", ISABELLE_DPGA2L_IN_SEL_REG, 3, 1, 0),
+SOC_DAPM_SINGLE("RX6 Playback Switch", ISABELLE_DPGA2L_IN_SEL_REG, 2, 1, 0),
+};
+
+static const struct snd_kcontrol_new isabelle_dpga2_right_mixer_controls[] = {
+SOC_DAPM_SINGLE("USNC Playback Switch", ISABELLE_DPGA2R_IN_SEL_REG, 7, 1, 0),
+SOC_DAPM_SINGLE("RX2 Playback Switch", ISABELLE_DPGA2R_IN_SEL_REG, 3, 1, 0),
+SOC_DAPM_SINGLE("RX4 Playback Switch", ISABELLE_DPGA2R_IN_SEL_REG, 2, 1, 0),
+SOC_DAPM_SINGLE("RX6 Playback Switch", ISABELLE_DPGA2R_IN_SEL_REG, 1, 1, 0),
+};
+
+static const struct snd_kcontrol_new isabelle_dpga3_left_mixer_controls[] = {
+SOC_DAPM_SINGLE("RX1 Playback Switch", ISABELLE_DPGA3LR_IN_SEL_REG, 7, 1, 0),
+SOC_DAPM_SINGLE("RX3 Playback Switch", ISABELLE_DPGA3LR_IN_SEL_REG, 6, 1, 0),
+SOC_DAPM_SINGLE("RX5 Playback Switch", ISABELLE_DPGA3LR_IN_SEL_REG, 5, 1, 0),
+};
+
+static const struct snd_kcontrol_new isabelle_dpga3_right_mixer_controls[] = {
+SOC_DAPM_SINGLE("RX2 Playback Switch", ISABELLE_DPGA3LR_IN_SEL_REG, 3, 1, 0),
+SOC_DAPM_SINGLE("RX4 Playback Switch", ISABELLE_DPGA3LR_IN_SEL_REG, 2, 1, 0),
+SOC_DAPM_SINGLE("RX6 Playback Switch", ISABELLE_DPGA3LR_IN_SEL_REG, 1, 1, 0),
+};
+
+static const struct snd_kcontrol_new isabelle_rx1_mixer_controls[] = {
+SOC_DAPM_SINGLE("ST1 Playback Switch", ISABELLE_RX_INPUT_CFG_REG, 7, 1, 0),
+SOC_DAPM_SINGLE("DL1 Playback Switch", ISABELLE_RX_INPUT_CFG_REG, 6, 1, 0),
+};
+
+static const struct snd_kcontrol_new isabelle_rx2_mixer_controls[] = {
+SOC_DAPM_SINGLE("ST2 Playback Switch", ISABELLE_RX_INPUT_CFG_REG, 5, 1, 0),
+SOC_DAPM_SINGLE("DL2 Playback Switch", ISABELLE_RX_INPUT_CFG_REG, 4, 1, 0),
+};
+
+static const struct snd_kcontrol_new isabelle_rx3_mixer_controls[] = {
+SOC_DAPM_SINGLE("ST1 Playback Switch", ISABELLE_RX_INPUT_CFG_REG, 3, 1, 0),
+SOC_DAPM_SINGLE("DL3 Playback Switch", ISABELLE_RX_INPUT_CFG_REG, 2, 1, 0),
+};
+
+static const struct snd_kcontrol_new isabelle_rx4_mixer_controls[] = {
+SOC_DAPM_SINGLE("ST2 Playback Switch", ISABELLE_RX_INPUT_CFG_REG, 1, 1, 0),
+SOC_DAPM_SINGLE("DL4 Playback Switch", ISABELLE_RX_INPUT_CFG_REG, 0, 1, 0),
+};
+
+static const struct snd_kcontrol_new isabelle_rx5_mixer_controls[] = {
+SOC_DAPM_SINGLE("ST1 Playback Switch", ISABELLE_RX_INPUT_CFG2_REG, 7, 1, 0),
+SOC_DAPM_SINGLE("DL5 Playback Switch", ISABELLE_RX_INPUT_CFG2_REG, 6, 1, 0),
+};
+
+static const struct snd_kcontrol_new isabelle_rx6_mixer_controls[] = {
+SOC_DAPM_SINGLE("ST2 Playback Switch", ISABELLE_RX_INPUT_CFG2_REG, 5, 1, 0),
+SOC_DAPM_SINGLE("DL6 Playback Switch", ISABELLE_RX_INPUT_CFG2_REG, 4, 1, 0),
+};
+
+static const struct snd_kcontrol_new ep_path_enable_control =
+ SOC_DAPM_SINGLE("Switch", ISABELLE_EARDRV_CFG2_REG, 0, 1, 0);
+
+/* TLV Declarations */
+static const DECLARE_TLV_DB_SCALE(mic_amp_tlv, 0, 100, 0);
+static const DECLARE_TLV_DB_SCALE(afm_amp_tlv, -3300, 300, 0);
+static const DECLARE_TLV_DB_SCALE(dac_tlv, -1200, 200, 0);
+static const DECLARE_TLV_DB_SCALE(hf_tlv, -5000, 200, 0);
+
+/* from -63 to 0 dB in 1 dB steps */
+static const DECLARE_TLV_DB_SCALE(dpga_tlv, -6300, 100, 1);
+
+/* from -63 to 9 dB in 1 dB steps */
+static const DECLARE_TLV_DB_SCALE(rx_tlv, -6300, 100, 1);
+
+static const DECLARE_TLV_DB_SCALE(st_tlv, -2700, 300, 1);
+static const DECLARE_TLV_DB_SCALE(tx_tlv, -600, 100, 0);
+
+static const struct snd_kcontrol_new isabelle_snd_controls[] = {
+ SOC_DOUBLE_TLV("Headset Playback Volume", ISABELLE_HSDRV_GAIN_REG,
+ 4, 0, 0xF, 0, dac_tlv),
+ SOC_DOUBLE_R_TLV("Handsfree Playback Volume",
+ ISABELLE_HFLPGA_CFG_REG, ISABELLE_HFRPGA_CFG_REG,
+ 0, 0x1F, 0, hf_tlv),
+ SOC_DOUBLE_TLV("Aux Playback Volume", ISABELLE_LINEAMP_GAIN_REG,
+ 4, 0, 0xF, 0, dac_tlv),
+ SOC_SINGLE_TLV("Earpiece Playback Volume", ISABELLE_EARDRV_CFG1_REG,
+ 0, 0xF, 0, dac_tlv),
+
+ SOC_DOUBLE_TLV("Aux FM Volume", ISABELLE_APGA_GAIN_REG, 4, 0, 0xF, 0,
+ afm_amp_tlv),
+ SOC_SINGLE_TLV("Mic1 Capture Volume", ISABELLE_MIC1_GAIN_REG, 3, 0x1F,
+ 0, mic_amp_tlv),
+ SOC_SINGLE_TLV("Mic2 Capture Volume", ISABELLE_MIC2_GAIN_REG, 3, 0x1F,
+ 0, mic_amp_tlv),
+
+ SOC_DOUBLE_R_TLV("DPGA1 Volume", ISABELLE_DPGA1L_GAIN_REG,
+ ISABELLE_DPGA1R_GAIN_REG, 0, 0x3F, 0, dpga_tlv),
+ SOC_DOUBLE_R_TLV("DPGA2 Volume", ISABELLE_DPGA2L_GAIN_REG,
+ ISABELLE_DPGA2R_GAIN_REG, 0, 0x3F, 0, dpga_tlv),
+ SOC_DOUBLE_R_TLV("DPGA3 Volume", ISABELLE_DPGA3L_GAIN_REG,
+ ISABELLE_DPGA3R_GAIN_REG, 0, 0x3F, 0, dpga_tlv),
+
+ SOC_SINGLE_TLV("Sidetone Audio TX1 Volume",
+ ISABELLE_ATX_STPGA1_CFG_REG, 0, 0xF, 0, st_tlv),
+ SOC_SINGLE_TLV("Sidetone Audio TX2 Volume",
+ ISABELLE_ATX_STPGA2_CFG_REG, 0, 0xF, 0, st_tlv),
+ SOC_SINGLE_TLV("Sidetone Voice TX1 Volume",
+ ISABELLE_VTX_STPGA1_CFG_REG, 0, 0xF, 0, st_tlv),
+ SOC_SINGLE_TLV("Sidetone Voice TX2 Volume",
+ ISABELLE_VTX2_STPGA2_CFG_REG, 0, 0xF, 0, st_tlv),
+
+ SOC_SINGLE_TLV("Audio TX1 Volume", ISABELLE_ATX1_DPGA_REG, 4, 0xF, 0,
+ tx_tlv),
+ SOC_SINGLE_TLV("Audio TX2 Volume", ISABELLE_ATX2_DPGA_REG, 4, 0xF, 0,
+ tx_tlv),
+ SOC_SINGLE_TLV("Voice TX1 Volume", ISABELLE_VTX1_DPGA_REG, 4, 0xF, 0,
+ tx_tlv),
+ SOC_SINGLE_TLV("Voice TX2 Volume", ISABELLE_VTX2_DPGA_REG, 4, 0xF, 0,
+ tx_tlv),
+
+ SOC_SINGLE_TLV("RX1 DPGA Volume", ISABELLE_RX1_DPGA_REG, 0, 0x3F, 0,
+ rx_tlv),
+ SOC_SINGLE_TLV("RX2 DPGA Volume", ISABELLE_RX2_DPGA_REG, 0, 0x3F, 0,
+ rx_tlv),
+ SOC_SINGLE_TLV("RX3 DPGA Volume", ISABELLE_RX3_DPGA_REG, 0, 0x3F, 0,
+ rx_tlv),
+ SOC_SINGLE_TLV("RX4 DPGA Volume", ISABELLE_RX4_DPGA_REG, 0, 0x3F, 0,
+ rx_tlv),
+ SOC_SINGLE_TLV("RX5 DPGA Volume", ISABELLE_RX5_DPGA_REG, 0, 0x3F, 0,
+ rx_tlv),
+ SOC_SINGLE_TLV("RX6 DPGA Volume", ISABELLE_RX6_DPGA_REG, 0, 0x3F, 0,
+ rx_tlv),
+
+ SOC_SINGLE("Headset Noise Gate", ISABELLE_HS_NG_CFG1_REG, 7, 1, 0),
+ SOC_SINGLE("Handsfree Noise Gate", ISABELLE_HF_NG_CFG1_REG, 7, 1, 0),
+
+ SOC_SINGLE("ATX1 Filter Bypass Switch", ISABELLE_AUDIO_HPF_CFG_REG,
+ 7, 1, 0),
+ SOC_SINGLE("ATX2 Filter Bypass Switch", ISABELLE_AUDIO_HPF_CFG_REG,
+ 6, 1, 0),
+ SOC_SINGLE("ARX1 Filter Bypass Switch", ISABELLE_AUDIO_HPF_CFG_REG,
+ 5, 1, 0),
+ SOC_SINGLE("ARX2 Filter Bypass Switch", ISABELLE_AUDIO_HPF_CFG_REG,
+ 4, 1, 0),
+ SOC_SINGLE("ARX3 Filter Bypass Switch", ISABELLE_AUDIO_HPF_CFG_REG,
+ 3, 1, 0),
+ SOC_SINGLE("ARX4 Filter Bypass Switch", ISABELLE_AUDIO_HPF_CFG_REG,
+ 2, 1, 0),
+ SOC_SINGLE("ARX5 Filter Bypass Switch", ISABELLE_AUDIO_HPF_CFG_REG,
+ 1, 1, 0),
+ SOC_SINGLE("ARX6 Filter Bypass Switch", ISABELLE_AUDIO_HPF_CFG_REG,
+ 0, 1, 0),
+ SOC_SINGLE("VRX1 Filter Bypass Switch", ISABELLE_AUDIO_HPF_CFG_REG,
+ 3, 1, 0),
+ SOC_SINGLE("VRX2 Filter Bypass Switch", ISABELLE_AUDIO_HPF_CFG_REG,
+ 2, 1, 0),
+
+ SOC_SINGLE("ATX1 Filter Enable Switch", ISABELLE_ALU_TX_EN_REG,
+ 7, 1, 0),
+ SOC_SINGLE("ATX2 Filter Enable Switch", ISABELLE_ALU_TX_EN_REG,
+ 6, 1, 0),
+ SOC_SINGLE("VTX1 Filter Enable Switch", ISABELLE_ALU_TX_EN_REG,
+ 5, 1, 0),
+ SOC_SINGLE("VTX2 Filter Enable Switch", ISABELLE_ALU_TX_EN_REG,
+ 4, 1, 0),
+ SOC_SINGLE("RX1 Filter Enable Switch", ISABELLE_ALU_RX_EN_REG,
+ 5, 1, 0),
+ SOC_SINGLE("RX2 Filter Enable Switch", ISABELLE_ALU_RX_EN_REG,
+ 4, 1, 0),
+ SOC_SINGLE("RX3 Filter Enable Switch", ISABELLE_ALU_RX_EN_REG,
+ 3, 1, 0),
+ SOC_SINGLE("RX4 Filter Enable Switch", ISABELLE_ALU_RX_EN_REG,
+ 2, 1, 0),
+ SOC_SINGLE("RX5 Filter Enable Switch", ISABELLE_ALU_RX_EN_REG,
+ 1, 1, 0),
+ SOC_SINGLE("RX6 Filter Enable Switch", ISABELLE_ALU_RX_EN_REG,
+ 0, 1, 0),
+
+ SOC_SINGLE("ULATX12 Capture Switch", ISABELLE_ULATX12_INTF_CFG_REG,
+ 7, 1, 0),
+
+ SOC_SINGLE("DL12 Playback Switch", ISABELLE_DL12_INTF_CFG_REG,
+ 7, 1, 0),
+ SOC_SINGLE("DL34 Playback Switch", ISABELLE_DL34_INTF_CFG_REG,
+ 7, 1, 0),
+ SOC_SINGLE("DL56 Playback Switch", ISABELLE_DL56_INTF_CFG_REG,
+ 7, 1, 0),
+
+ /* DMIC Switch */
+ SOC_SINGLE("DMIC Switch", ISABELLE_DMIC_CFG_REG, 0, 1, 0),
+};
+
+static const struct snd_soc_dapm_widget isabelle_dapm_widgets[] = {
+ /* Inputs */
+ SND_SOC_DAPM_INPUT("MAINMIC"),
+ SND_SOC_DAPM_INPUT("HSMIC"),
+ SND_SOC_DAPM_INPUT("SUBMIC"),
+ SND_SOC_DAPM_INPUT("LINEIN1"),
+ SND_SOC_DAPM_INPUT("LINEIN2"),
+ SND_SOC_DAPM_INPUT("DMICDAT"),
+
+ /* Outputs */
+ SND_SOC_DAPM_OUTPUT("HSOL"),
+ SND_SOC_DAPM_OUTPUT("HSOR"),
+ SND_SOC_DAPM_OUTPUT("HFL"),
+ SND_SOC_DAPM_OUTPUT("HFR"),
+ SND_SOC_DAPM_OUTPUT("EP"),
+ SND_SOC_DAPM_OUTPUT("LINEOUT1"),
+ SND_SOC_DAPM_OUTPUT("LINEOUT2"),
+
+ SND_SOC_DAPM_PGA("DL1", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("DL2", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("DL3", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("DL4", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("DL5", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("DL6", SND_SOC_NOPM, 0, 0, NULL, 0),
+
+ /* Analog input muxes for the capture amplifiers */
+ SND_SOC_DAPM_MUX("Analog Left Capture Route",
+ SND_SOC_NOPM, 0, 0, &amic1_control),
+ SND_SOC_DAPM_MUX("Analog Right Capture Route",
+ SND_SOC_NOPM, 0, 0, &amic2_control),
+
+ SND_SOC_DAPM_MUX("Sidetone Audio Playback", SND_SOC_NOPM, 0, 0,
+ &st_audio_control),
+ SND_SOC_DAPM_MUX("Sidetone Voice Playback", SND_SOC_NOPM, 0, 0,
+ &st_voice_control),
+
+ /* AIF */
+ SND_SOC_DAPM_AIF_IN("INTF1_SDI", NULL, 0, ISABELLE_INTF_EN_REG, 7, 0),
+ SND_SOC_DAPM_AIF_IN("INTF2_SDI", NULL, 0, ISABELLE_INTF_EN_REG, 6, 0),
+
+ SND_SOC_DAPM_AIF_OUT("INTF1_SDO", NULL, 0, ISABELLE_INTF_EN_REG, 5, 0),
+ SND_SOC_DAPM_AIF_OUT("INTF2_SDO", NULL, 0, ISABELLE_INTF_EN_REG, 4, 0),
+
+ SND_SOC_DAPM_OUT_DRV("ULATX1", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_OUT_DRV("ULATX2", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_OUT_DRV("ULVTX1", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_OUT_DRV("ULVTX2", SND_SOC_NOPM, 0, 0, NULL, 0),
+
+ /* Analog Capture PGAs */
+ SND_SOC_DAPM_PGA("MicAmp1", ISABELLE_AMIC_CFG_REG, 5, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("MicAmp2", ISABELLE_AMIC_CFG_REG, 4, 0, NULL, 0),
+
+ /* Auxiliary FM PGAs */
+ SND_SOC_DAPM_PGA("APGA1", ISABELLE_APGA_CFG_REG, 7, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("APGA2", ISABELLE_APGA_CFG_REG, 6, 0, NULL, 0),
+
+ /* ADCs */
+ SND_SOC_DAPM_ADC("ADC1", "Left Front Capture",
+ ISABELLE_AMIC_CFG_REG, 7, 0),
+ SND_SOC_DAPM_ADC("ADC2", "Right Front Capture",
+ ISABELLE_AMIC_CFG_REG, 6, 0),
+
+ /* Microphone Bias */
+ SND_SOC_DAPM_SUPPLY("Headset Mic Bias", ISABELLE_ABIAS_CFG_REG,
+ 3, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("Main Mic Bias", ISABELLE_ABIAS_CFG_REG,
+ 2, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("Digital Mic1 Bias",
+ ISABELLE_DBIAS_CFG_REG, 3, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("Digital Mic2 Bias",
+ ISABELLE_DBIAS_CFG_REG, 2, 0, NULL, 0),
+
+ /* Mixers */
+ SND_SOC_DAPM_MIXER("Headset Left Mixer", SND_SOC_NOPM, 0, 0,
+ isabelle_hs_left_mixer_controls,
+ ARRAY_SIZE(isabelle_hs_left_mixer_controls)),
+ SND_SOC_DAPM_MIXER("Headset Right Mixer", SND_SOC_NOPM, 0, 0,
+ isabelle_hs_right_mixer_controls,
+ ARRAY_SIZE(isabelle_hs_right_mixer_controls)),
+ SND_SOC_DAPM_MIXER("Handsfree Left Mixer", SND_SOC_NOPM, 0, 0,
+ isabelle_hf_left_mixer_controls,
+ ARRAY_SIZE(isabelle_hf_left_mixer_controls)),
+ SND_SOC_DAPM_MIXER("Handsfree Right Mixer", SND_SOC_NOPM, 0, 0,
+ isabelle_hf_right_mixer_controls,
+ ARRAY_SIZE(isabelle_hf_right_mixer_controls)),
+ SND_SOC_DAPM_MIXER("LINEOUT1 Mixer", SND_SOC_NOPM, 0, 0,
+ isabelle_aux_left_mixer_controls,
+ ARRAY_SIZE(isabelle_aux_left_mixer_controls)),
+ SND_SOC_DAPM_MIXER("LINEOUT2 Mixer", SND_SOC_NOPM, 0, 0,
+ isabelle_aux_right_mixer_controls,
+ ARRAY_SIZE(isabelle_aux_right_mixer_controls)),
+ SND_SOC_DAPM_MIXER("Earphone Mixer", SND_SOC_NOPM, 0, 0,
+ isabelle_ep_mixer_controls,
+ ARRAY_SIZE(isabelle_ep_mixer_controls)),
+
+ SND_SOC_DAPM_MIXER("DPGA1L Mixer", SND_SOC_NOPM, 0, 0,
+ isabelle_dpga1_left_mixer_controls,
+ ARRAY_SIZE(isabelle_dpga1_left_mixer_controls)),
+ SND_SOC_DAPM_MIXER("DPGA1R Mixer", SND_SOC_NOPM, 0, 0,
+ isabelle_dpga1_right_mixer_controls,
+ ARRAY_SIZE(isabelle_dpga1_right_mixer_controls)),
+ SND_SOC_DAPM_MIXER("DPGA2L Mixer", SND_SOC_NOPM, 0, 0,
+ isabelle_dpga2_left_mixer_controls,
+ ARRAY_SIZE(isabelle_dpga2_left_mixer_controls)),
+ SND_SOC_DAPM_MIXER("DPGA2R Mixer", SND_SOC_NOPM, 0, 0,
+ isabelle_dpga2_right_mixer_controls,
+ ARRAY_SIZE(isabelle_dpga2_right_mixer_controls)),
+ SND_SOC_DAPM_MIXER("DPGA3L Mixer", SND_SOC_NOPM, 0, 0,
+ isabelle_dpga3_left_mixer_controls,
+ ARRAY_SIZE(isabelle_dpga3_left_mixer_controls)),
+ SND_SOC_DAPM_MIXER("DPGA3R Mixer", SND_SOC_NOPM, 0, 0,
+ isabelle_dpga3_right_mixer_controls,
+ ARRAY_SIZE(isabelle_dpga3_right_mixer_controls)),
+
+ SND_SOC_DAPM_MIXER("RX1 Mixer", SND_SOC_NOPM, 0, 0,
+ isabelle_rx1_mixer_controls,
+ ARRAY_SIZE(isabelle_rx1_mixer_controls)),
+ SND_SOC_DAPM_MIXER("RX2 Mixer", SND_SOC_NOPM, 0, 0,
+ isabelle_rx2_mixer_controls,
+ ARRAY_SIZE(isabelle_rx2_mixer_controls)),
+ SND_SOC_DAPM_MIXER("RX3 Mixer", SND_SOC_NOPM, 0, 0,
+ isabelle_rx3_mixer_controls,
+ ARRAY_SIZE(isabelle_rx3_mixer_controls)),
+ SND_SOC_DAPM_MIXER("RX4 Mixer", SND_SOC_NOPM, 0, 0,
+ isabelle_rx4_mixer_controls,
+ ARRAY_SIZE(isabelle_rx4_mixer_controls)),
+ SND_SOC_DAPM_MIXER("RX5 Mixer", SND_SOC_NOPM, 0, 0,
+ isabelle_rx5_mixer_controls,
+ ARRAY_SIZE(isabelle_rx5_mixer_controls)),
+ SND_SOC_DAPM_MIXER("RX6 Mixer", SND_SOC_NOPM, 0, 0,
+ isabelle_rx6_mixer_controls,
+ ARRAY_SIZE(isabelle_rx6_mixer_controls)),
+
+ /* DACs */
+ SND_SOC_DAPM_DAC("DAC1L", "Headset Playback", ISABELLE_DAC_CFG_REG,
+ 5, 0),
+ SND_SOC_DAPM_DAC("DAC1R", "Headset Playback", ISABELLE_DAC_CFG_REG,
+ 4, 0),
+ SND_SOC_DAPM_DAC("DAC2L", "Handsfree Playback", ISABELLE_DAC_CFG_REG,
+ 3, 0),
+ SND_SOC_DAPM_DAC("DAC2R", "Handsfree Playback", ISABELLE_DAC_CFG_REG,
+ 2, 0),
+ SND_SOC_DAPM_DAC("DAC3L", "Lineout Playback", ISABELLE_DAC_CFG_REG,
+ 1, 0),
+ SND_SOC_DAPM_DAC("DAC3R", "Lineout Playback", ISABELLE_DAC_CFG_REG,
+ 0, 0),
+
+ /* Analog Playback PGAs */
+ SND_SOC_DAPM_PGA("Sidetone Audio PGA", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("Sidetone Voice PGA", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("HF Left PGA", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("HF Right PGA", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("DPGA1L", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("DPGA1R", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("DPGA2L", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("DPGA2R", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("DPGA3L", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("DPGA3R", SND_SOC_NOPM, 0, 0, NULL, 0),
+
+ /* Analog Playback Mux */
+ SND_SOC_DAPM_MUX("RX1 Playback", ISABELLE_ALU_RX_EN_REG, 5, 0,
+ &rx1_mux_controls),
+ SND_SOC_DAPM_MUX("RX2 Playback", ISABELLE_ALU_RX_EN_REG, 4, 0,
+ &rx2_mux_controls),
+
+ /* TX Select */
+ SND_SOC_DAPM_MUX("ATX Select", ISABELLE_TX_INPUT_CFG_REG,
+ 7, 0, &atx_mux_controls),
+ SND_SOC_DAPM_MUX("VTX Select", ISABELLE_TX_INPUT_CFG_REG,
+ 6, 0, &vtx_mux_controls),
+
+ SND_SOC_DAPM_SWITCH("Earphone Playback", SND_SOC_NOPM, 0, 0,
+ &ep_path_enable_control),
+
+ /* Output Drivers */
+ SND_SOC_DAPM_OUT_DRV("HS Left Driver", ISABELLE_HSDRV_CFG2_REG,
+ 1, 0, NULL, 0),
+ SND_SOC_DAPM_OUT_DRV("HS Right Driver", ISABELLE_HSDRV_CFG2_REG,
+ 0, 0, NULL, 0),
+ SND_SOC_DAPM_OUT_DRV("LINEOUT1 Left Driver", ISABELLE_LINEAMP_CFG_REG,
+ 1, 0, NULL, 0),
+ SND_SOC_DAPM_OUT_DRV("LINEOUT2 Right Driver", ISABELLE_LINEAMP_CFG_REG,
+ 0, 0, NULL, 0),
+ SND_SOC_DAPM_OUT_DRV("Earphone Driver", ISABELLE_EARDRV_CFG2_REG,
+ 1, 0, NULL, 0),
+
+ SND_SOC_DAPM_OUT_DRV("HF Left Driver", ISABELLE_HFDRV_CFG_REG,
+ 1, 0, NULL, 0),
+ SND_SOC_DAPM_OUT_DRV("HF Right Driver", ISABELLE_HFDRV_CFG_REG,
+ 0, 0, NULL, 0),
+};
+
+static const struct snd_soc_dapm_route isabelle_intercon[] = {
+ /* Interface mapping */
+ { "DL1", "DL12 Playback Switch", "INTF1_SDI" },
+ { "DL2", "DL12 Playback Switch", "INTF1_SDI" },
+ { "DL3", "DL34 Playback Switch", "INTF1_SDI" },
+ { "DL4", "DL34 Playback Switch", "INTF1_SDI" },
+ { "DL5", "DL56 Playback Switch", "INTF1_SDI" },
+ { "DL6", "DL56 Playback Switch", "INTF1_SDI" },
+
+ { "DL1", "DL12 Playback Switch", "INTF2_SDI" },
+ { "DL2", "DL12 Playback Switch", "INTF2_SDI" },
+ { "DL3", "DL34 Playback Switch", "INTF2_SDI" },
+ { "DL4", "DL34 Playback Switch", "INTF2_SDI" },
+ { "DL5", "DL56 Playback Switch", "INTF2_SDI" },
+ { "DL6", "DL56 Playback Switch", "INTF2_SDI" },
+
+ /* Input side mapping */
+ { "Sidetone Audio PGA", NULL, "Sidetone Audio Playback" },
+ { "Sidetone Voice PGA", NULL, "Sidetone Voice Playback" },
+
+ { "RX1 Mixer", "ST1 Playback Switch", "Sidetone Audio PGA" },
+
+ { "RX1 Mixer", "ST1 Playback Switch", "Sidetone Voice PGA" },
+ { "RX1 Mixer", "DL1 Playback Switch", "DL1" },
+
+ { "RX2 Mixer", "ST2 Playback Switch", "Sidetone Audio PGA" },
+
+ { "RX2 Mixer", "ST2 Playback Switch", "Sidetone Voice PGA" },
+ { "RX2 Mixer", "DL2 Playback Switch", "DL2" },
+
+ { "RX3 Mixer", "ST1 Playback Switch", "Sidetone Voice PGA" },
+ { "RX3 Mixer", "DL3 Playback Switch", "DL3" },
+
+ { "RX4 Mixer", "ST2 Playback Switch", "Sidetone Voice PGA" },
+ { "RX4 Mixer", "DL4 Playback Switch", "DL4" },
+
+ { "RX5 Mixer", "ST1 Playback Switch", "Sidetone Voice PGA" },
+ { "RX5 Mixer", "DL5 Playback Switch", "DL5" },
+
+ { "RX6 Mixer", "ST2 Playback Switch", "Sidetone Voice PGA" },
+ { "RX6 Mixer", "DL6 Playback Switch", "DL6" },
+
+ /* Capture path */
+ { "Analog Left Capture Route", "Headset Mic", "HSMIC" },
+ { "Analog Left Capture Route", "Main Mic", "MAINMIC" },
+ { "Analog Left Capture Route", "Aux/FM Left", "LINEIN1" },
+
+ { "Analog Right Capture Route", "Sub Mic", "SUBMIC" },
+ { "Analog Right Capture Route", "Aux/FM Right", "LINEIN2" },
+
+ { "MicAmp1", NULL, "Analog Left Capture Route" },
+ { "MicAmp2", NULL, "Analog Right Capture Route" },
+
+ { "ADC1", NULL, "MicAmp1" },
+ { "ADC2", NULL, "MicAmp2" },
+
+ { "ATX Select", "AMIC1", "ADC1" },
+ { "ATX Select", "DMIC", "DMICDAT" },
+ { "ATX Select", "AMIC2", "ADC2" },
+
+ { "VTX Select", "AMIC1", "ADC1" },
+ { "VTX Select", "DMIC", "DMICDAT" },
+ { "VTX Select", "AMIC2", "ADC2" },
+
+ { "ULATX1", "ATX1 Filter Enable Switch", "ATX Select" },
+ { "ULATX1", "ATX1 Filter Bypass Switch", "ATX Select" },
+ { "ULATX2", "ATX2 Filter Enable Switch", "ATX Select" },
+ { "ULATX2", "ATX2 Filter Bypass Switch", "ATX Select" },
+
+ { "ULVTX1", "VTX1 Filter Enable Switch", "VTX Select" },
+ { "ULVTX1", "VTX1 Filter Bypass Switch", "VTX Select" },
+ { "ULVTX2", "VTX2 Filter Enable Switch", "VTX Select" },
+ { "ULVTX2", "VTX2 Filter Bypass Switch", "VTX Select" },
+
+ { "INTF1_SDO", "ULATX12 Capture Switch", "ULATX1" },
+ { "INTF1_SDO", "ULATX12 Capture Switch", "ULATX2" },
+ { "INTF2_SDO", "ULATX12 Capture Switch", "ULATX1" },
+ { "INTF2_SDO", "ULATX12 Capture Switch", "ULATX2" },
+
+ { "INTF1_SDO", NULL, "ULVTX1" },
+ { "INTF1_SDO", NULL, "ULVTX2" },
+ { "INTF2_SDO", NULL, "ULVTX1" },
+ { "INTF2_SDO", NULL, "ULVTX2" },
+
+ /* AFM Path */
+ { "APGA1", NULL, "LINEIN1" },
+ { "APGA2", NULL, "LINEIN2" },
+
+ { "RX1 Playback", "VRX1 Filter Bypass Switch", "RX1 Mixer" },
+ { "RX1 Playback", "ARX1 Filter Bypass Switch", "RX1 Mixer" },
+ { "RX1 Playback", "RX1 Filter Enable Switch", "RX1 Mixer" },
+
+ { "RX2 Playback", "VRX2 Filter Bypass Switch", "RX2 Mixer" },
+ { "RX2 Playback", "ARX2 Filter Bypass Switch", "RX2 Mixer" },
+ { "RX2 Playback", "RX2 Filter Enable Switch", "RX2 Mixer" },
+
+ { "RX3 Playback", "ARX3 Filter Bypass Switch", "RX3 Mixer" },
+ { "RX3 Playback", "RX3 Filter Enable Switch", "RX3 Mixer" },
+
+ { "RX4 Playback", "ARX4 Filter Bypass Switch", "RX4 Mixer" },
+ { "RX4 Playback", "RX4 Filter Enable Switch", "RX4 Mixer" },
+
+ { "RX5 Playback", "ARX5 Filter Bypass Switch", "RX5 Mixer" },
+ { "RX5 Playback", "RX5 Filter Enable Switch", "RX5 Mixer" },
+
+ { "RX6 Playback", "ARX6 Filter Bypass Switch", "RX6 Mixer" },
+ { "RX6 Playback", "RX6 Filter Enable Switch", "RX6 Mixer" },
+
+ { "DPGA1L Mixer", "RX1 Playback Switch", "RX1 Playback" },
+ { "DPGA1L Mixer", "RX3 Playback Switch", "RX3 Playback" },
+ { "DPGA1L Mixer", "RX5 Playback Switch", "RX5 Playback" },
+
+ { "DPGA1R Mixer", "RX2 Playback Switch", "RX2 Playback" },
+ { "DPGA1R Mixer", "RX4 Playback Switch", "RX4 Playback" },
+ { "DPGA1R Mixer", "RX6 Playback Switch", "RX6 Playback" },
+
+ { "DPGA1L", NULL, "DPGA1L Mixer" },
+ { "DPGA1R", NULL, "DPGA1R Mixer" },
+
+ { "DAC1L", NULL, "DPGA1L" },
+ { "DAC1R", NULL, "DPGA1R" },
+
+ { "DPGA2L Mixer", "RX1 Playback Switch", "RX1 Playback" },
+ { "DPGA2L Mixer", "RX2 Playback Switch", "RX2 Playback" },
+ { "DPGA2L Mixer", "RX3 Playback Switch", "RX3 Playback" },
+ { "DPGA2L Mixer", "RX4 Playback Switch", "RX4 Playback" },
+ { "DPGA2L Mixer", "RX5 Playback Switch", "RX5 Playback" },
+ { "DPGA2L Mixer", "RX6 Playback Switch", "RX6 Playback" },
+
+ { "DPGA2R Mixer", "RX2 Playback Switch", "RX2 Playback" },
+ { "DPGA2R Mixer", "RX4 Playback Switch", "RX4 Playback" },
+ { "DPGA2R Mixer", "RX6 Playback Switch", "RX6 Playback" },
+
+ { "DPGA2L", NULL, "DPGA2L Mixer" },
+ { "DPGA2R", NULL, "DPGA2R Mixer" },
+
+ { "DAC2L", NULL, "DPGA2L" },
+ { "DAC2R", NULL, "DPGA2R" },
+
+ { "DPGA3L Mixer", "RX1 Playback Switch", "RX1 Playback" },
+ { "DPGA3L Mixer", "RX3 Playback Switch", "RX3 Playback" },
+ { "DPGA3L Mixer", "RX5 Playback Switch", "RX5 Playback" },
+
+ { "DPGA3R Mixer", "RX2 Playback Switch", "RX2 Playback" },
+ { "DPGA3R Mixer", "RX4 Playback Switch", "RX4 Playback" },
+ { "DPGA3R Mixer", "RX6 Playback Switch", "RX6 Playback" },
+
+ { "DPGA3L", NULL, "DPGA3L Mixer" },
+ { "DPGA3R", NULL, "DPGA3R Mixer" },
+
+ { "DAC3L", NULL, "DPGA3L" },
+ { "DAC3R", NULL, "DPGA3R" },
+
+ { "Headset Left Mixer", "DAC1L Playback Switch", "DAC1L" },
+ { "Headset Left Mixer", "APGA1 Playback Switch", "APGA1" },
+
+ { "Headset Right Mixer", "DAC1R Playback Switch", "DAC1R" },
+ { "Headset Right Mixer", "APGA2 Playback Switch", "APGA2" },
+
+ { "HS Left Driver", NULL, "Headset Left Mixer" },
+ { "HS Right Driver", NULL, "Headset Right Mixer" },
+
+ { "HSOL", NULL, "HS Left Driver" },
+ { "HSOR", NULL, "HS Right Driver" },
+
+ /* Earphone playback path */
+ { "Earphone Mixer", "DAC2L Playback Switch", "DAC2L" },
+ { "Earphone Mixer", "APGA1 Playback Switch", "APGA1" },
+
+ { "Earphone Playback", "Switch", "Earphone Mixer" },
+ { "Earphone Driver", NULL, "Earphone Playback" },
+ { "EP", NULL, "Earphone Driver" },
+
+ { "Handsfree Left Mixer", "DAC2L Playback Switch", "DAC2L" },
+ { "Handsfree Left Mixer", "APGA1 Playback Switch", "APGA1" },
+
+ { "Handsfree Right Mixer", "DAC2R Playback Switch", "DAC2R" },
+ { "Handsfree Right Mixer", "APGA2 Playback Switch", "APGA2" },
+
+ { "HF Left PGA", NULL, "Handsfree Left Mixer" },
+ { "HF Right PGA", NULL, "Handsfree Right Mixer" },
+
+ { "HF Left Driver", NULL, "HF Left PGA" },
+ { "HF Right Driver", NULL, "HF Right PGA" },
+
+ { "HFL", NULL, "HF Left Driver" },
+ { "HFR", NULL, "HF Right Driver" },
+
+ { "LINEOUT1 Mixer", "DAC3L Playback Switch", "DAC3L" },
+ { "LINEOUT1 Mixer", "APGA1 Playback Switch", "APGA1" },
+
+ { "LINEOUT2 Mixer", "DAC3R Playback Switch", "DAC3R" },
+ { "LINEOUT2 Mixer", "APGA2 Playback Switch", "APGA2" },
+
+ { "LINEOUT1 Driver", NULL, "LINEOUT1 Mixer" },
+ { "LINEOUT2 Driver", NULL, "LINEOUT2 Mixer" },
+
+ { "LINEOUT1", NULL, "LINEOUT1 Driver" },
+ { "LINEOUT2", NULL, "LINEOUT2 Driver" },
+};
+
+static int isabelle_hs_mute(struct snd_soc_dai *dai, int mute)
+{
+ snd_soc_update_bits(dai->codec, ISABELLE_DAC1_SOFTRAMP_REG,
+ BIT(4), (mute ? BIT(4) : 0));
+
+ return 0;
+}
+
+static int isabelle_hf_mute(struct snd_soc_dai *dai, int mute)
+{
+ snd_soc_update_bits(dai->codec, ISABELLE_DAC2_SOFTRAMP_REG,
+ BIT(4), (mute ? BIT(4) : 0));
+
+ return 0;
+}
+
+static int isabelle_line_mute(struct snd_soc_dai *dai, int mute)
+{
+ snd_soc_update_bits(dai->codec, ISABELLE_DAC3_SOFTRAMP_REG,
+ BIT(4), (mute ? BIT(4) : 0));
+
+ return 0;
+}
+
+static int isabelle_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:
+ break;
+
+ case SND_SOC_BIAS_STANDBY:
+ snd_soc_update_bits(codec, ISABELLE_PWR_EN_REG,
+ ISABELLE_CHIP_EN, BIT(0));
+ break;
+
+ case SND_SOC_BIAS_OFF:
+ snd_soc_update_bits(codec, ISABELLE_PWR_EN_REG,
+ ISABELLE_CHIP_EN, 0);
+ break;
+ }
+
+ codec->dapm.bias_level = level;
+
+ return 0;
+}
+
+static int isabelle_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_codec *codec = rtd->codec;
+ u16 aif = 0;
+ unsigned int fs_val = 0;
+
+ switch (params_rate(params)) {
+ case 8000:
+ fs_val = ISABELLE_FS_RATE_8;
+ break;
+ case 11025:
+ fs_val = ISABELLE_FS_RATE_11;
+ break;
+ case 12000:
+ fs_val = ISABELLE_FS_RATE_12;
+ break;
+ case 16000:
+ fs_val = ISABELLE_FS_RATE_16;
+ break;
+ case 22050:
+ fs_val = ISABELLE_FS_RATE_22;
+ break;
+ case 24000:
+ fs_val = ISABELLE_FS_RATE_24;
+ break;
+ case 32000:
+ fs_val = ISABELLE_FS_RATE_32;
+ break;
+ case 44100:
+ fs_val = ISABELLE_FS_RATE_44;
+ break;
+ case 48000:
+ fs_val = ISABELLE_FS_RATE_48;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ snd_soc_update_bits(codec, ISABELLE_FS_RATE_CFG_REG,
+ ISABELLE_FS_RATE_MASK, fs_val);
+
+ /* bit size */
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S20_3LE:
+ aif |= ISABELLE_AIF_LENGTH_20;
+ break;
+ case SNDRV_PCM_FORMAT_S32_LE:
+ aif |= ISABELLE_AIF_LENGTH_32;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ snd_soc_update_bits(codec, ISABELLE_INTF_CFG_REG,
+ ISABELLE_AIF_LENGTH_MASK, aif);
+
+ return 0;
+}
+
+static int isabelle_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ unsigned int aif_val = 0;
+
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBS_CFS:
+ aif_val &= ~ISABELLE_AIF_MS;
+ break;
+ case SND_SOC_DAIFMT_CBM_CFM:
+ aif_val |= ISABELLE_AIF_MS;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ aif_val |= ISABELLE_I2S_MODE;
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ aif_val |= ISABELLE_LEFT_J_MODE;
+ break;
+ case SND_SOC_DAIFMT_PDM:
+ aif_val |= ISABELLE_PDM_MODE;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ snd_soc_update_bits(codec, ISABELLE_INTF_CFG_REG,
+ (ISABELLE_AIF_MS | ISABELLE_AIF_FMT_MASK), aif_val);
+
+ return 0;
+}
+
+/* Rates supported by Isabelle driver */
+#define ISABELLE_RATES SNDRV_PCM_RATE_8000_48000
+
+/* Formates supported by Isabelle driver. */
+#define ISABELLE_FORMATS (SNDRV_PCM_FMTBIT_S20_3LE |\
+ SNDRV_PCM_FMTBIT_S32_LE)
+
+static struct snd_soc_dai_ops isabelle_hs_dai_ops = {
+ .hw_params = isabelle_hw_params,
+ .set_fmt = isabelle_set_dai_fmt,
+ .digital_mute = isabelle_hs_mute,
+};
+
+static struct snd_soc_dai_ops isabelle_hf_dai_ops = {
+ .hw_params = isabelle_hw_params,
+ .set_fmt = isabelle_set_dai_fmt,
+ .digital_mute = isabelle_hf_mute,
+};
+
+static struct snd_soc_dai_ops isabelle_line_dai_ops = {
+ .hw_params = isabelle_hw_params,
+ .set_fmt = isabelle_set_dai_fmt,
+ .digital_mute = isabelle_line_mute,
+};
+
+static struct snd_soc_dai_ops isabelle_ul_dai_ops = {
+ .hw_params = isabelle_hw_params,
+ .set_fmt = isabelle_set_dai_fmt,
+};
+
+/* ISABELLE dai structure */
+static struct snd_soc_dai_driver isabelle_dai[] = {
+ {
+ .name = "isabelle-dl1",
+ .playback = {
+ .stream_name = "Headset Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = ISABELLE_RATES,
+ .formats = ISABELLE_FORMATS,
+ },
+ .ops = &isabelle_hs_dai_ops,
+ },
+ {
+ .name = "isabelle-dl2",
+ .playback = {
+ .stream_name = "Handsfree Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = ISABELLE_RATES,
+ .formats = ISABELLE_FORMATS,
+ },
+ .ops = &isabelle_hf_dai_ops,
+ },
+ {
+ .name = "isabelle-lineout",
+ .playback = {
+ .stream_name = "Lineout Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = ISABELLE_RATES,
+ .formats = ISABELLE_FORMATS,
+ },
+ .ops = &isabelle_line_dai_ops,
+ },
+ {
+ .name = "isabelle-ul",
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = ISABELLE_RATES,
+ .formats = ISABELLE_FORMATS,
+ },
+ .ops = &isabelle_ul_dai_ops,
+ },
+};
+
+static int isabelle_probe(struct snd_soc_codec *codec)
+{
+ int ret = 0;
+
+ codec->control_data = dev_get_regmap(codec->dev, NULL);
+
+ ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP);
+ if (ret < 0) {
+ dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
+ return ret;
+ }
+
+ return 0;
+}
+
+static struct snd_soc_codec_driver soc_codec_dev_isabelle = {
+ .probe = isabelle_probe,
+ .set_bias_level = isabelle_set_bias_level,
+ .controls = isabelle_snd_controls,
+ .num_controls = ARRAY_SIZE(isabelle_snd_controls),
+ .dapm_widgets = isabelle_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(isabelle_dapm_widgets),
+ .dapm_routes = isabelle_intercon,
+ .num_dapm_routes = ARRAY_SIZE(isabelle_intercon),
+ .idle_bias_off = true,
+};
+
+static const struct regmap_config isabelle_regmap_config = {
+ .reg_bits = 8,
+ .val_bits = 8,
+
+ .max_register = ISABELLE_MAX_REGISTER,
+ .reg_defaults = isabelle_reg_defs,
+ .num_reg_defaults = ARRAY_SIZE(isabelle_reg_defs),
+ .cache_type = REGCACHE_RBTREE,
+};
+
+static int __devinit isabelle_i2c_probe(struct i2c_client *i2c,
+ const struct i2c_device_id *id)
+{
+ struct regmap *isabelle_regmap;
+ int ret = 0;
+
+ isabelle_regmap = devm_regmap_init_i2c(i2c, &isabelle_regmap_config);
+ if (IS_ERR(isabelle_regmap)) {
+ ret = PTR_ERR(isabelle_regmap);
+ dev_err(&i2c->dev, "Failed to allocate register map: %d\n",
+ ret);
+ return ret;
+ }
+ i2c_set_clientdata(i2c, isabelle_regmap);
+
+ ret = snd_soc_register_codec(&i2c->dev,
+ &soc_codec_dev_isabelle, isabelle_dai,
+ ARRAY_SIZE(isabelle_dai));
+ if (ret < 0) {
+ dev_err(&i2c->dev, "Failed to register codec: %d\n", ret);
+ return ret;
+ }
+
+ return ret;
+}
+
+static int __devexit isabelle_i2c_remove(struct i2c_client *client)
+{
+ snd_soc_unregister_codec(&client->dev);
+ return 0;
+}
+
+static const struct i2c_device_id isabelle_i2c_id[] = {
+ { "isabelle", 0 },
+ { }
+};
+MODULE_DEVICE_TABLE(i2c, isabelle_i2c_id);
+
+static struct i2c_driver isabelle_i2c_driver = {
+ .driver = {
+ .name = "isabelle",
+ .owner = THIS_MODULE,
+ },
+ .probe = isabelle_i2c_probe,
+ .remove = __devexit_p(isabelle_i2c_remove),
+ .id_table = isabelle_i2c_id,
+};
+
+module_i2c_driver(isabelle_i2c_driver);
+
+MODULE_DESCRIPTION("ASoC ISABELLE driver");
+MODULE_AUTHOR("Vishwas A Deshpande <vishwas.a.deshpande@ti.com>");
+MODULE_AUTHOR("M R Swami Reddy <MR.Swami.Reddy@ti.com>");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/codecs/isabelle.h b/sound/soc/codecs/isabelle.h
new file mode 100644
index 0000000..96d839a
--- /dev/null
+++ b/sound/soc/codecs/isabelle.h
@@ -0,0 +1,143 @@
+/*
+ * isabelle.h - Low power high fidelity audio codec driver header file
+ *
+ * Copyright (c) 2012 Texas Instruments, Inc
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; version 2 of the License.
+ *
+ */
+
+#ifndef _ISABELLE_H
+#define _ISABELLE_H
+
+#include <linux/bitops.h>
+
+/* ISABELLE REGISTERS */
+
+#define ISABELLE_PWR_CFG_REG 0x01
+#define ISABELLE_PWR_EN_REG 0x02
+#define ISABELLE_PS_EN1_REG 0x03
+#define ISABELLE_INT1_STATUS_REG 0x04
+#define ISABELLE_INT1_MASK_REG 0x05
+#define ISABELLE_INT2_STATUS_REG 0x06
+#define ISABELLE_INT2_MASK_REG 0x07
+#define ISABELLE_HKCTL1_REG 0x08
+#define ISABELLE_HKCTL2_REG 0x09
+#define ISABELLE_HKCTL3_REG 0x0A
+#define ISABELLE_ACCDET_STATUS_REG 0x0B
+#define ISABELLE_BUTTON_ID_REG 0x0C
+#define ISABELLE_PLL_CFG_REG 0x10
+#define ISABELLE_PLL_EN_REG 0x11
+#define ISABELLE_FS_RATE_CFG_REG 0x12
+#define ISABELLE_INTF_CFG_REG 0x13
+#define ISABELLE_INTF_EN_REG 0x14
+#define ISABELLE_ULATX12_INTF_CFG_REG 0x15
+#define ISABELLE_DL12_INTF_CFG_REG 0x16
+#define ISABELLE_DL34_INTF_CFG_REG 0x17
+#define ISABELLE_DL56_INTF_CFG_REG 0x18
+#define ISABELLE_ATX_STPGA1_CFG_REG 0x19
+#define ISABELLE_ATX_STPGA2_CFG_REG 0x1A
+#define ISABELLE_VTX_STPGA1_CFG_REG 0x1B
+#define ISABELLE_VTX2_STPGA2_CFG_REG 0x1C
+#define ISABELLE_ATX1_DPGA_REG 0x1D
+#define ISABELLE_ATX2_DPGA_REG 0x1E
+#define ISABELLE_VTX1_DPGA_REG 0x1F
+#define ISABELLE_VTX2_DPGA_REG 0x20
+#define ISABELLE_TX_INPUT_CFG_REG 0x21
+#define ISABELLE_RX_INPUT_CFG_REG 0x22
+#define ISABELLE_RX_INPUT_CFG2_REG 0x23
+#define ISABELLE_VOICE_HPF_CFG_REG 0x24
+#define ISABELLE_AUDIO_HPF_CFG_REG 0x25
+#define ISABELLE_RX1_DPGA_REG 0x26
+#define ISABELLE_RX2_DPGA_REG 0x27
+#define ISABELLE_RX3_DPGA_REG 0x28
+#define ISABELLE_RX4_DPGA_REG 0x29
+#define ISABELLE_RX5_DPGA_REG 0x2A
+#define ISABELLE_RX6_DPGA_REG 0x2B
+#define ISABELLE_ALU_TX_EN_REG 0x2C
+#define ISABELLE_ALU_RX_EN_REG 0x2D
+#define ISABELLE_IIR_RESYNC_REG 0x2E
+#define ISABELLE_ABIAS_CFG_REG 0x30
+#define ISABELLE_DBIAS_CFG_REG 0x31
+#define ISABELLE_MIC1_GAIN_REG 0x32
+#define ISABELLE_MIC2_GAIN_REG 0x33
+#define ISABELLE_AMIC_CFG_REG 0x34
+#define ISABELLE_DMIC_CFG_REG 0x35
+#define ISABELLE_APGA_GAIN_REG 0x36
+#define ISABELLE_APGA_CFG_REG 0x37
+#define ISABELLE_TX_GAIN_DLY_REG 0x38
+#define ISABELLE_RX_GAIN_DLY_REG 0x39
+#define ISABELLE_RX_PWR_CTRL_REG 0x3A
+#define ISABELLE_DPGA1LR_IN_SEL_REG 0x3B
+#define ISABELLE_DPGA1L_GAIN_REG 0x3C
+#define ISABELLE_DPGA1R_GAIN_REG 0x3D
+#define ISABELLE_DPGA2L_IN_SEL_REG 0x3E
+#define ISABELLE_DPGA2R_IN_SEL_REG 0x3F
+#define ISABELLE_DPGA2L_GAIN_REG 0x40
+#define ISABELLE_DPGA2R_GAIN_REG 0x41
+#define ISABELLE_DPGA3LR_IN_SEL_REG 0x42
+#define ISABELLE_DPGA3L_GAIN_REG 0x43
+#define ISABELLE_DPGA3R_GAIN_REG 0x44
+#define ISABELLE_DAC1_SOFTRAMP_REG 0x45
+#define ISABELLE_DAC2_SOFTRAMP_REG 0x46
+#define ISABELLE_DAC3_SOFTRAMP_REG 0x47
+#define ISABELLE_DAC_CFG_REG 0x48
+#define ISABELLE_EARDRV_CFG1_REG 0x49
+#define ISABELLE_EARDRV_CFG2_REG 0x4A
+#define ISABELLE_HSDRV_GAIN_REG 0x4B
+#define ISABELLE_HSDRV_CFG1_REG 0x4C
+#define ISABELLE_HSDRV_CFG2_REG 0x4D
+#define ISABELLE_HS_NG_CFG1_REG 0x4E
+#define ISABELLE_HS_NG_CFG2_REG 0x4F
+#define ISABELLE_LINEAMP_GAIN_REG 0x50
+#define ISABELLE_LINEAMP_CFG_REG 0x51
+#define ISABELLE_HFL_VOL_CTRL_REG 0x52
+#define ISABELLE_HFL_SFTVOL_CTRL_REG 0x53
+#define ISABELLE_HFL_LIM_CTRL_1_REG 0x54
+#define ISABELLE_HFL_LIM_CTRL_2_REG 0x55
+#define ISABELLE_HFR_VOL_CTRL_REG 0x56
+#define ISABELLE_HFR_SFTVOL_CTRL_REG 0x57
+#define ISABELLE_HFR_LIM_CTRL_1_REG 0x58
+#define ISABELLE_HFR_LIM_CTRL_2_REG 0x59
+#define ISABELLE_HF_MODE_REG 0x5A
+#define ISABELLE_HFLPGA_CFG_REG 0x5B
+#define ISABELLE_HFRPGA_CFG_REG 0x5C
+#define ISABELLE_HFDRV_CFG_REG 0x5D
+#define ISABELLE_PDMOUT_CFG1_REG 0x5E
+#define ISABELLE_PDMOUT_CFG2_REG 0x5F
+#define ISABELLE_PDMOUT_L_WM_REG 0x60
+#define ISABELLE_PDMOUT_R_WM_REG 0x61
+#define ISABELLE_HF_NG_CFG1_REG 0x62
+#define ISABELLE_HF_NG_CFG2_REG 0x63
+
+/* ISABELLE_PWR_EN_REG (0x02h) */
+#define ISABELLE_CHIP_EN BIT(0)
+
+/* ISABELLE DAI FORMATS */
+#define ISABELLE_AIF_FMT_MASK 0x70
+#define ISABELLE_I2S_MODE 0x0
+#define ISABELLE_LEFT_J_MODE 0x1
+#define ISABELLE_PDM_MODE 0x2
+
+#define ISABELLE_AIF_LENGTH_MASK 0x30
+#define ISABELLE_AIF_LENGTH_20 0x00
+#define ISABELLE_AIF_LENGTH_32 0x10
+
+#define ISABELLE_AIF_MS 0x80
+
+#define ISABELLE_FS_RATE_MASK 0xF
+#define ISABELLE_FS_RATE_8 0x0
+#define ISABELLE_FS_RATE_11 0x1
+#define ISABELLE_FS_RATE_12 0x2
+#define ISABELLE_FS_RATE_16 0x4
+#define ISABELLE_FS_RATE_22 0x5
+#define ISABELLE_FS_RATE_24 0x6
+#define ISABELLE_FS_RATE_32 0x8
+#define ISABELLE_FS_RATE_44 0x9
+#define ISABELLE_FS_RATE_48 0xA
+
+#define ISABELLE_MAX_REGISTER 0xFF
+
+#endif
diff --git a/sound/soc/codecs/lm49453.c b/sound/soc/codecs/lm49453.c
index 802b9f1..99b0a9d 100644
--- a/sound/soc/codecs/lm49453.c
+++ b/sound/soc/codecs/lm49453.c
@@ -12,7 +12,6 @@
#include <linux/module.h>
#include <linux/moduleparam.h>
-#include <linux/version.h>
#include <linux/kernel.h>
#include <linux/init.h>
#include <linux/delay.h>
@@ -1358,7 +1357,7 @@ static struct snd_soc_dai_ops lm49453_lineout_dai_ops = {
};
/* LM49453 dai structure. */
-static const struct snd_soc_dai_driver lm49453_dai[] = {
+static struct snd_soc_dai_driver lm49453_dai[] = {
{
.name = "LM49453 Headset",
.playback = {
diff --git a/sound/soc/codecs/max98095.c b/sound/soc/codecs/max98095.c
index 35179e2..7cd508e 100644
--- a/sound/soc/codecs/max98095.c
+++ b/sound/soc/codecs/max98095.c
@@ -2216,7 +2216,7 @@ static irqreturn_t max98095_report_jack(int irq, void *data)
return IRQ_HANDLED;
}
-int max98095_jack_detect_enable(struct snd_soc_codec *codec)
+static int max98095_jack_detect_enable(struct snd_soc_codec *codec)
{
struct max98095_priv *max98095 = snd_soc_codec_get_drvdata(codec);
int ret = 0;
@@ -2245,7 +2245,7 @@ int max98095_jack_detect_enable(struct snd_soc_codec *codec)
return ret;
}
-int max98095_jack_detect_disable(struct snd_soc_codec *codec)
+static int max98095_jack_detect_disable(struct snd_soc_codec *codec)
{
int ret = 0;
@@ -2286,6 +2286,7 @@ int max98095_jack_detect(struct snd_soc_codec *codec,
max98095_report_jack(client->irq, codec);
return 0;
}
+EXPORT_SYMBOL_GPL(max98095_jack_detect);
#ifdef CONFIG_PM
static int max98095_suspend(struct snd_soc_codec *codec)
diff --git a/sound/soc/codecs/ml26124.c b/sound/soc/codecs/ml26124.c
index 22cb5bf..96aa5fa 100644
--- a/sound/soc/codecs/ml26124.c
+++ b/sound/soc/codecs/ml26124.c
@@ -638,7 +638,7 @@ static __devinit int ml26124_i2c_probe(struct i2c_client *i2c,
i2c_set_clientdata(i2c, priv);
- priv->regmap = regmap_init_i2c(i2c, &ml26124_i2c_regmap);
+ priv->regmap = devm_regmap_init_i2c(i2c, &ml26124_i2c_regmap);
if (IS_ERR(priv->regmap)) {
ret = PTR_ERR(priv->regmap);
dev_err(&i2c->dev, "regmap_init_i2c() failed: %d\n", ret);
@@ -651,10 +651,7 @@ static __devinit int ml26124_i2c_probe(struct i2c_client *i2c,
static __devexit int ml26124_i2c_remove(struct i2c_client *client)
{
- struct ml26124_priv *priv = i2c_get_clientdata(client);
-
snd_soc_unregister_codec(&client->dev);
- regmap_exit(priv->regmap);
return 0;
}
diff --git a/sound/soc/codecs/spdif_receiver.c b/sound/soc/codecs/spdif_receiver.c
new file mode 100644
index 0000000..dd8d856
--- /dev/null
+++ b/sound/soc/codecs/spdif_receiver.c
@@ -0,0 +1,67 @@
+/*
+ * ALSA SoC SPDIF DIR (Digital Interface Reciever) driver
+ *
+ * Based on ALSA SoC SPDIF DIT driver
+ *
+ * This driver is used by controllers which can operate in DIR (SPDI/F) where
+ * no codec is needed. This file provides stub codec that can be used
+ * in these configurations. SPEAr SPDIF IN Audio controller uses this driver.
+ *
+ * Author: Vipin Kumar, <vipin.kumar@st.com>
+ * Copyright: (C) 2012 ST Microelectronics
+ *
+ * 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 <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/slab.h>
+#include <sound/soc.h>
+#include <sound/pcm.h>
+#include <sound/initval.h>
+
+#define STUB_RATES SNDRV_PCM_RATE_8000_192000
+#define STUB_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \
+ SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_LE)
+
+static struct snd_soc_codec_driver soc_codec_spdif_dir;
+
+static struct snd_soc_dai_driver dir_stub_dai = {
+ .name = "dir-hifi",
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 1,
+ .channels_max = 384,
+ .rates = STUB_RATES,
+ .formats = STUB_FORMATS,
+ },
+};
+
+static int spdif_dir_probe(struct platform_device *pdev)
+{
+ return snd_soc_register_codec(&pdev->dev, &soc_codec_spdif_dir,
+ &dir_stub_dai, 1);
+}
+
+static int spdif_dir_remove(struct platform_device *pdev)
+{
+ snd_soc_unregister_codec(&pdev->dev);
+ return 0;
+}
+
+static struct platform_driver spdif_dir_driver = {
+ .probe = spdif_dir_probe,
+ .remove = spdif_dir_remove,
+ .driver = {
+ .name = "spdif-dir",
+ .owner = THIS_MODULE,
+ },
+};
+
+module_platform_driver(spdif_dir_driver);
+
+MODULE_DESCRIPTION("ASoC SPDIF DIR driver");
+MODULE_AUTHOR("Vipin Kumar <vipin.kumar@st.com>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/sta529.c b/sound/soc/codecs/sta529.c
new file mode 100644
index 0000000..0c225cd
--- /dev/null
+++ b/sound/soc/codecs/sta529.c
@@ -0,0 +1,442 @@
+/*
+ * ASoC codec driver for spear platform
+ *
+ * sound/soc/codecs/sta529.c -- spear ALSA Soc codec driver
+ *
+ * Copyright (C) 2012 ST Microelectronics
+ * Rajeev Kumar <rajeev-dlh.kumar@st.com>
+ *
+ * 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 <linux/clk.h>
+#include <linux/init.h>
+#include <linux/i2c.h>
+#include <linux/io.h>
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/pm.h>
+#include <linux/regmap.h>
+#include <linux/slab.h>
+
+#include <sound/core.h>
+#include <sound/initval.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/tlv.h>
+
+/* STA529 Register offsets */
+#define STA529_FFXCFG0 0x00
+#define STA529_FFXCFG1 0x01
+#define STA529_MVOL 0x02
+#define STA529_LVOL 0x03
+#define STA529_RVOL 0x04
+#define STA529_TTF0 0x05
+#define STA529_TTF1 0x06
+#define STA529_TTP0 0x07
+#define STA529_TTP1 0x08
+#define STA529_S2PCFG0 0x0A
+#define STA529_S2PCFG1 0x0B
+#define STA529_P2SCFG0 0x0C
+#define STA529_P2SCFG1 0x0D
+#define STA529_PLLCFG0 0x14
+#define STA529_PLLCFG1 0x15
+#define STA529_PLLCFG2 0x16
+#define STA529_PLLCFG3 0x17
+#define STA529_PLLPFE 0x18
+#define STA529_PLLST 0x19
+#define STA529_ADCCFG 0x1E /*mic_select*/
+#define STA529_CKOCFG 0x1F
+#define STA529_MISC 0x20
+#define STA529_PADST0 0x21
+#define STA529_PADST1 0x22
+#define STA529_FFXST 0x23
+#define STA529_PWMIN1 0x2D
+#define STA529_PWMIN2 0x2E
+#define STA529_POWST 0x32
+
+#define STA529_MAX_REGISTER 0x32
+
+#define STA529_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)
+
+#define STA529_FORMAT (SNDRV_PCM_FMTBIT_S16_LE | \
+ SNDRV_PCM_FMTBIT_S24_LE | \
+ SNDRV_PCM_FMTBIT_S32_LE)
+#define S2PC_VALUE 0x98
+#define CLOCK_OUT 0x60
+#define LEFT_J_DATA_FORMAT 0x10
+#define I2S_DATA_FORMAT 0x12
+#define RIGHT_J_DATA_FORMAT 0x14
+#define CODEC_MUTE_VAL 0x80
+
+#define POWER_CNTLMSAK 0x40
+#define POWER_STDBY 0x40
+#define FFX_MASK 0x80
+#define FFX_OFF 0x80
+#define POWER_UP 0x00
+#define FFX_CLK_ENB 0x01
+#define FFX_CLK_DIS 0x00
+#define FFX_CLK_MSK 0x01
+#define PLAY_FREQ_RANGE_MSK 0x70
+#define CAP_FREQ_RANGE_MSK 0x0C
+#define PDATA_LEN_MSK 0xC0
+#define BCLK_TO_FS_MSK 0x30
+#define AUDIO_MUTE_MSK 0x80
+
+static const struct reg_default sta529_reg_defaults[] = {
+ { 0, 0x35 }, /* R0 - FFX Configuration reg 0 */
+ { 1, 0xc8 }, /* R1 - FFX Configuration reg 1 */
+ { 2, 0x50 }, /* R2 - Master Volume */
+ { 3, 0x00 }, /* R3 - Left Volume */
+ { 4, 0x00 }, /* R4 - Right Volume */
+ { 10, 0xb2 }, /* R10 - S2P Config Reg 0 */
+ { 11, 0x41 }, /* R11 - S2P Config Reg 1 */
+ { 12, 0x92 }, /* R12 - P2S Config Reg 0 */
+ { 13, 0x41 }, /* R13 - P2S Config Reg 1 */
+ { 30, 0xd2 }, /* R30 - ADC Config Reg */
+ { 31, 0x40 }, /* R31 - clock Out Reg */
+ { 32, 0x21 }, /* R32 - Misc Register */
+};
+
+struct sta529 {
+ struct regmap *regmap;
+};
+
+static bool sta529_readable(struct device *dev, unsigned int reg)
+{
+ switch (reg) {
+
+ case STA529_FFXCFG0:
+ case STA529_FFXCFG1:
+ case STA529_MVOL:
+ case STA529_LVOL:
+ case STA529_RVOL:
+ case STA529_S2PCFG0:
+ case STA529_S2PCFG1:
+ case STA529_P2SCFG0:
+ case STA529_P2SCFG1:
+ case STA529_ADCCFG:
+ case STA529_CKOCFG:
+ case STA529_MISC:
+ return true;
+ default:
+ return false;
+ }
+}
+
+
+static const char *pwm_mode_text[] = { "Binary", "Headphone", "Ternary",
+ "Phase-shift"};
+
+static const DECLARE_TLV_DB_SCALE(out_gain_tlv, -9150, 50, 0);
+static const DECLARE_TLV_DB_SCALE(master_vol_tlv, -12750, 50, 0);
+static const SOC_ENUM_SINGLE_DECL(pwm_src, STA529_FFXCFG1, 4, pwm_mode_text);
+
+static const struct snd_kcontrol_new sta529_snd_controls[] = {
+ SOC_DOUBLE_R_TLV("Digital Playback Volume", STA529_LVOL, STA529_RVOL, 0,
+ 127, 0, out_gain_tlv),
+ SOC_SINGLE_TLV("Master Playback Volume", STA529_MVOL, 0, 127, 1,
+ master_vol_tlv),
+ SOC_ENUM("PWM Select", pwm_src),
+};
+
+static int sta529_set_bias_level(struct snd_soc_codec *codec, enum
+ snd_soc_bias_level level)
+{
+ struct sta529 *sta529 = snd_soc_codec_get_drvdata(codec);
+
+ switch (level) {
+ case SND_SOC_BIAS_ON:
+ case SND_SOC_BIAS_PREPARE:
+ snd_soc_update_bits(codec, STA529_FFXCFG0, POWER_CNTLMSAK,
+ POWER_UP);
+ snd_soc_update_bits(codec, STA529_MISC, FFX_CLK_MSK,
+ FFX_CLK_ENB);
+ break;
+ case SND_SOC_BIAS_STANDBY:
+ if (codec->dapm.bias_level == SND_SOC_BIAS_OFF)
+ regcache_sync(sta529->regmap);
+ snd_soc_update_bits(codec, STA529_FFXCFG0,
+ POWER_CNTLMSAK, POWER_STDBY);
+ /* Making FFX output to zero */
+ snd_soc_update_bits(codec, STA529_FFXCFG0, FFX_MASK,
+ FFX_OFF);
+ snd_soc_update_bits(codec, STA529_MISC, FFX_CLK_MSK,
+ FFX_CLK_DIS);
+ break;
+ case SND_SOC_BIAS_OFF:
+ break;
+ }
+
+ /*
+ * store the label for powers down audio subsystem for suspend.This is
+ * used by soc core layer
+ */
+ codec->dapm.bias_level = level;
+
+ return 0;
+
+}
+
+static int sta529_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_codec *codec = rtd->codec;
+ int pdata, play_freq_val, record_freq_val;
+ int bclk_to_fs_ratio;
+
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ pdata = 1;
+ bclk_to_fs_ratio = 0;
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ pdata = 2;
+ bclk_to_fs_ratio = 1;
+ break;
+ case SNDRV_PCM_FORMAT_S32_LE:
+ pdata = 3;
+ bclk_to_fs_ratio = 2;
+ break;
+ default:
+ dev_err(codec->dev, "Unsupported format\n");
+ return -EINVAL;
+ }
+
+ switch (params_rate(params)) {
+ case 8000:
+ case 11025:
+ play_freq_val = 0;
+ record_freq_val = 2;
+ break;
+ case 16000:
+ case 22050:
+ play_freq_val = 1;
+ record_freq_val = 0;
+ break;
+
+ case 32000:
+ case 44100:
+ case 48000:
+ play_freq_val = 2;
+ record_freq_val = 0;
+ break;
+ default:
+ dev_err(codec->dev, "Unsupported rate\n");
+ return -EINVAL;
+ }
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ snd_soc_update_bits(codec, STA529_S2PCFG1, PDATA_LEN_MSK,
+ pdata << 6);
+ snd_soc_update_bits(codec, STA529_S2PCFG1, BCLK_TO_FS_MSK,
+ bclk_to_fs_ratio << 4);
+ snd_soc_update_bits(codec, STA529_MISC, PLAY_FREQ_RANGE_MSK,
+ play_freq_val << 4);
+ } else {
+ snd_soc_update_bits(codec, STA529_P2SCFG1, PDATA_LEN_MSK,
+ pdata << 6);
+ snd_soc_update_bits(codec, STA529_P2SCFG1, BCLK_TO_FS_MSK,
+ bclk_to_fs_ratio << 4);
+ snd_soc_update_bits(codec, STA529_MISC, CAP_FREQ_RANGE_MSK,
+ record_freq_val << 2);
+ }
+
+ return 0;
+}
+
+static int sta529_mute(struct snd_soc_dai *dai, int mute)
+{
+ u8 val = 0;
+
+ if (mute)
+ val |= CODEC_MUTE_VAL;
+
+ snd_soc_update_bits(dai->codec, STA529_FFXCFG0, AUDIO_MUTE_MSK, val);
+
+ return 0;
+}
+
+static int sta529_set_dai_fmt(struct snd_soc_dai *codec_dai, u32 fmt)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ u8 mode = 0;
+
+ /* interface format */
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_LEFT_J:
+ mode = LEFT_J_DATA_FORMAT;
+ break;
+ case SND_SOC_DAIFMT_I2S:
+ mode = I2S_DATA_FORMAT;
+ break;
+ case SND_SOC_DAIFMT_RIGHT_J:
+ mode = RIGHT_J_DATA_FORMAT;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ snd_soc_update_bits(codec, STA529_S2PCFG0, 0x0D, mode);
+
+ return 0;
+}
+
+static const struct snd_soc_dai_ops sta529_dai_ops = {
+ .hw_params = sta529_hw_params,
+ .set_fmt = sta529_set_dai_fmt,
+ .digital_mute = sta529_mute,
+};
+
+static struct snd_soc_dai_driver sta529_dai = {
+ .name = "sta529-audio",
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = STA529_RATES,
+ .formats = STA529_FORMAT,
+ },
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = STA529_RATES,
+ .formats = STA529_FORMAT,
+ },
+ .ops = &sta529_dai_ops,
+};
+
+static int sta529_probe(struct snd_soc_codec *codec)
+{
+ struct sta529 *sta529 = snd_soc_codec_get_drvdata(codec);
+ int ret;
+
+ codec->control_data = sta529->regmap;
+ ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP);
+
+ if (ret < 0) {
+ dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
+ return ret;
+ }
+ sta529_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+ return 0;
+}
+
+/* power down chip */
+static int sta529_remove(struct snd_soc_codec *codec)
+{
+ sta529_set_bias_level(codec, SND_SOC_BIAS_OFF);
+
+ return 0;
+}
+
+static int sta529_suspend(struct snd_soc_codec *codec)
+{
+ sta529_set_bias_level(codec, SND_SOC_BIAS_OFF);
+
+ return 0;
+}
+
+static int sta529_resume(struct snd_soc_codec *codec)
+{
+ sta529_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+ return 0;
+}
+
+struct snd_soc_codec_driver sta529_codec_driver = {
+ .probe = sta529_probe,
+ .remove = sta529_remove,
+ .set_bias_level = sta529_set_bias_level,
+ .suspend = sta529_suspend,
+ .resume = sta529_resume,
+ .controls = sta529_snd_controls,
+ .num_controls = ARRAY_SIZE(sta529_snd_controls),
+};
+
+static const struct regmap_config sta529_regmap = {
+ .reg_bits = 8,
+ .val_bits = 8,
+
+ .max_register = STA529_MAX_REGISTER,
+ .readable_reg = sta529_readable,
+
+ .cache_type = REGCACHE_RBTREE,
+ .reg_defaults = sta529_reg_defaults,
+ .num_reg_defaults = ARRAY_SIZE(sta529_reg_defaults),
+};
+
+static __devinit int sta529_i2c_probe(struct i2c_client *i2c,
+ const struct i2c_device_id *id)
+{
+ struct sta529 *sta529;
+ int ret;
+
+ if (!i2c_check_functionality(i2c->adapter, I2C_FUNC_SMBUS_BYTE_DATA))
+ return -EINVAL;
+
+ sta529 = devm_kzalloc(&i2c->dev, sizeof(struct sta529), GFP_KERNEL);
+ if (sta529 == NULL) {
+ dev_err(&i2c->dev, "Can not allocate memory\n");
+ return -ENOMEM;
+ }
+
+ sta529->regmap = devm_regmap_init_i2c(i2c, &sta529_regmap);
+ if (IS_ERR(sta529->regmap)) {
+ ret = PTR_ERR(sta529->regmap);
+ dev_err(&i2c->dev, "Failed to allocate regmap: %d\n", ret);
+ return ret;
+ }
+
+ i2c_set_clientdata(i2c, sta529);
+
+ ret = snd_soc_register_codec(&i2c->dev,
+ &sta529_codec_driver, &sta529_dai, 1);
+ if (ret != 0)
+ dev_err(&i2c->dev, "Failed to register CODEC: %d\n", ret);
+
+ return ret;
+}
+
+static int __devexit sta529_i2c_remove(struct i2c_client *client)
+{
+ snd_soc_unregister_codec(&client->dev);
+
+ return 0;
+}
+
+static const struct i2c_device_id sta529_i2c_id[] = {
+ { "sta529", 0 },
+ { }
+};
+MODULE_DEVICE_TABLE(i2c, sta529_i2c_id);
+
+static struct i2c_driver sta529_i2c_driver = {
+ .driver = {
+ .name = "sta529",
+ .owner = THIS_MODULE,
+ },
+ .probe = sta529_i2c_probe,
+ .remove = __devexit_p(sta529_i2c_remove),
+ .id_table = sta529_i2c_id,
+};
+
+module_i2c_driver(sta529_i2c_driver);
+
+MODULE_DESCRIPTION("ASoC STA529 codec driver");
+MODULE_AUTHOR("Rajeev Kumar <rajeev-dlh.kumar@st.com>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index e9b62b5..dc78f5a 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -118,7 +118,9 @@ static const u8 aic3x_reg[AIC3X_CACHEREGNUM] = {
0x00, 0x00, 0x00, 0x00, /* 88 */
0x00, 0x00, 0x00, 0x00, /* 92 */
0x00, 0x00, 0x00, 0x00, /* 96 */
- 0x00, 0x00, 0x02, /* 100 */
+ 0x00, 0x00, 0x02, 0x00, /* 100 */
+ 0x00, 0x00, 0x00, 0x00, /* 104 */
+ 0x00, 0x00, /* 108 */
};
#define SOC_DAPM_SINGLE_AIC3X(xname, reg, shift, mask, invert) \
@@ -229,6 +231,25 @@ static const struct soc_enum aic3x_enum[] = {
SOC_ENUM_DOUBLE(AIC3X_CODEC_DFILT_CTRL, 6, 4, 4, aic3x_adc_hpf),
};
+static const char *aic3x_agc_level[] =
+ { "-5.5dB", "-8dB", "-10dB", "-12dB", "-14dB", "-17dB", "-20dB", "-24dB" };
+static const struct soc_enum aic3x_agc_level_enum[] = {
+ SOC_ENUM_SINGLE(LAGC_CTRL_A, 4, 8, aic3x_agc_level),
+ SOC_ENUM_SINGLE(RAGC_CTRL_A, 4, 8, aic3x_agc_level),
+};
+
+static const char *aic3x_agc_attack[] = { "8ms", "11ms", "16ms", "20ms" };
+static const struct soc_enum aic3x_agc_attack_enum[] = {
+ SOC_ENUM_SINGLE(LAGC_CTRL_A, 2, 4, aic3x_agc_attack),
+ SOC_ENUM_SINGLE(RAGC_CTRL_A, 2, 4, aic3x_agc_attack),
+};
+
+static const char *aic3x_agc_decay[] = { "100ms", "200ms", "400ms", "500ms" };
+static const struct soc_enum aic3x_agc_decay_enum[] = {
+ SOC_ENUM_SINGLE(LAGC_CTRL_A, 0, 4, aic3x_agc_decay),
+ SOC_ENUM_SINGLE(RAGC_CTRL_A, 0, 4, aic3x_agc_decay),
+};
+
/*
* DAC digital volumes. From -63.5 to 0 dB in 0.5 dB steps
*/
@@ -353,6 +374,15 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = {
* adjust PGA to max value when ADC is on and will never go back.
*/
SOC_DOUBLE_R("AGC Switch", LAGC_CTRL_A, RAGC_CTRL_A, 7, 0x01, 0),
+ SOC_ENUM("Left AGC Target level", aic3x_agc_level_enum[0]),
+ SOC_ENUM("Right AGC Target level", aic3x_agc_level_enum[1]),
+ SOC_ENUM("Left AGC Attack time", aic3x_agc_attack_enum[0]),
+ SOC_ENUM("Right AGC Attack time", aic3x_agc_attack_enum[1]),
+ SOC_ENUM("Left AGC Decay time", aic3x_agc_decay_enum[0]),
+ SOC_ENUM("Right AGC Decay time", aic3x_agc_decay_enum[1]),
+
+ /* De-emphasis */
+ SOC_DOUBLE("De-emphasis Switch", AIC3X_CODEC_DFILT_CTRL, 2, 0, 0x01, 0),
/* Input */
SOC_DOUBLE_R_TLV("PGA Capture Volume", LADC_VOL, RADC_VOL,
@@ -368,7 +398,7 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = {
static DECLARE_TLV_DB_SCALE(classd_amp_tlv, 0, 600, 0);
static const struct snd_kcontrol_new aic3x_classd_amp_gain_ctrl =
- SOC_DOUBLE_TLV("Class-D Amplifier Gain", CLASSD_CTRL, 6, 4, 3, 0, classd_amp_tlv);
+ SOC_DOUBLE_TLV("Class-D Playback Volume", CLASSD_CTRL, 6, 4, 3, 0, classd_amp_tlv);
/* Left DAC Mux */
static const struct snd_kcontrol_new aic3x_left_dac_mux_controls =
@@ -970,6 +1000,12 @@ static int aic3x_set_dai_sysclk(struct snd_soc_dai *codec_dai,
struct snd_soc_codec *codec = codec_dai->codec;
struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec);
+ /* set clock on MCLK or GPIO2 or BCLK */
+ snd_soc_update_bits(codec, AIC3X_CLKGEN_CTRL_REG, PLLCLK_IN_MASK,
+ clk_id << PLLCLK_IN_SHIFT);
+ snd_soc_update_bits(codec, AIC3X_CLKGEN_CTRL_REG, CLKDIV_IN_MASK,
+ clk_id << CLKDIV_IN_SHIFT);
+
aic3x->sysclk = freq;
return 0;
}
diff --git a/sound/soc/codecs/tlv320aic3x.h b/sound/soc/codecs/tlv320aic3x.h
index 08c7f66..6db3c41 100644
--- a/sound/soc/codecs/tlv320aic3x.h
+++ b/sound/soc/codecs/tlv320aic3x.h
@@ -13,7 +13,7 @@
#define _AIC3X_H
/* AIC3X register space */
-#define AIC3X_CACHEREGNUM 103
+#define AIC3X_CACHEREGNUM 110
/* Page select register */
#define AIC3X_PAGE_SELECT 0
@@ -74,6 +74,8 @@
#define HPLCOM_CFG 37
/* Right High Power Output control registers */
#define HPRCOM_CFG 38
+/* High Power Output Stage Control Register */
+#define HPOUT_SC 40
/* DAC Output Switching control registers */
#define DAC_LINE_MUX 41
/* High Power Output Driver Pop Reduction registers */
@@ -148,6 +150,17 @@
#define AIC3X_GPIOB_REG 101
/* Clock generation control register */
#define AIC3X_CLKGEN_CTRL_REG 102
+/* New AGC registers */
+#define LAGCN_ATTACK 103
+#define LAGCN_DECAY 104
+#define RAGCN_ATTACK 105
+#define RAGCN_DECAY 106
+/* New Programmable ADC Digital Path and I2C Bus Condition Register */
+#define NEW_ADC_DIGITALPATH 107
+/* Passive Analog Signal Bypass Selection During Powerdown Register */
+#define PASSIVE_BYPASS 108
+/* DAC Quiescent Current Adjustment Register */
+#define DAC_ICC_ADJ 109
/* Page select register bits */
#define PAGE0_SELECT 0
@@ -163,6 +176,10 @@
#define DUAL_RATE_MODE ((1 << 5) | (1 << 6))
#define LDAC2LCH (0x1 << 3)
#define RDAC2RCH (0x1 << 1)
+#define LDAC2RCH (0x2 << 3)
+#define RDAC2LCH (0x2 << 1)
+#define LDAC2MONOMIX (0x3 << 3)
+#define RDAC2MONOMIX (0x3 << 1)
/* PLL registers bitfields */
#define PLLP_SHIFT 0
@@ -179,6 +196,14 @@
#define PLL_CLKIN_SHIFT 4
#define MCLK_SOURCE 0x0
#define PLL_CLKDIV_SHIFT 0
+#define PLLCLK_IN_MASK 0x30
+#define PLLCLK_IN_SHIFT 4
+#define CLKDIV_IN_MASK 0xc0
+#define CLKDIV_IN_SHIFT 6
+/* clock in source */
+#define CLKIN_MCLK 0
+#define CLKIN_GPIO2 1
+#define CLKIN_BCLK 2
/* Software reset register bits */
#define SOFT_RESET 0x80
diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c
index a36e9fc..0ff1e70 100644
--- a/sound/soc/codecs/twl6040.c
+++ b/sound/soc/codecs/twl6040.c
@@ -553,7 +553,7 @@ static const struct snd_kcontrol_new vibrar_mux_controls =
/* Headset power mode */
static const char *twl6040_power_mode_texts[] = {
- "Low-Power", "High-Perfomance",
+ "Low-Power", "High-Performance",
};
static const struct soc_enum twl6040_power_mode_enum =
diff --git a/sound/soc/codecs/wm1250-ev1.c b/sound/soc/codecs/wm1250-ev1.c
index e0b51e9..951d7b4 100644
--- a/sound/soc/codecs/wm1250-ev1.c
+++ b/sound/soc/codecs/wm1250-ev1.c
@@ -121,20 +121,23 @@ static const struct snd_soc_dai_ops wm1250_ev1_ops = {
.hw_params = wm1250_ev1_hw_params,
};
+#define WM1250_EV1_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |\
+ SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_64000)
+
static struct snd_soc_dai_driver wm1250_ev1_dai = {
.name = "wm1250-ev1",
.playback = {
.stream_name = "Playback",
.channels_min = 1,
.channels_max = 2,
- .rates = SNDRV_PCM_RATE_8000,
+ .rates = WM1250_EV1_RATES,
.formats = SNDRV_PCM_FMTBIT_S16_LE,
},
.capture = {
.stream_name = "Capture",
.channels_min = 1,
.channels_max = 2,
- .rates = SNDRV_PCM_RATE_8000,
+ .rates = WM1250_EV1_RATES,
.formats = SNDRV_PCM_FMTBIT_S16_LE,
},
.ops = &wm1250_ev1_ops,
diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c
index 0418fa1..3fd5b29 100644
--- a/sound/soc/codecs/wm2000.c
+++ b/sound/soc/codecs/wm2000.c
@@ -1,7 +1,7 @@
/*
* wm2000.c -- WM2000 ALSA Soc Audio driver
*
- * Copyright 2008-2010 Wolfson Microelectronics PLC.
+ * Copyright 2008-2011 Wolfson Microelectronics PLC.
*
* Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
*
@@ -674,9 +674,39 @@ static int wm2000_resume(struct snd_soc_codec *codec)
#define wm2000_resume NULL
#endif
+static bool wm2000_readable_reg(struct device *dev, unsigned int reg)
+{
+ switch (reg) {
+ case WM2000_REG_SYS_START:
+ case WM2000_REG_SPEECH_CLARITY:
+ case WM2000_REG_SYS_WATCHDOG:
+ case WM2000_REG_ANA_VMID_PD_TIME:
+ case WM2000_REG_ANA_VMID_PU_TIME:
+ case WM2000_REG_CAT_FLTR_INDX:
+ case WM2000_REG_CAT_GAIN_0:
+ case WM2000_REG_SYS_STATUS:
+ case WM2000_REG_SYS_MODE_CNTRL:
+ case WM2000_REG_SYS_START0:
+ case WM2000_REG_SYS_START1:
+ case WM2000_REG_ID1:
+ case WM2000_REG_ID2:
+ case WM2000_REG_REVISON:
+ case WM2000_REG_SYS_CTL1:
+ case WM2000_REG_SYS_CTL2:
+ case WM2000_REG_ANC_STAT:
+ case WM2000_REG_IF_CTL:
+ return true;
+ default:
+ return false;
+ }
+}
+
static const struct regmap_config wm2000_regmap = {
.reg_bits = 8,
.val_bits = 8,
+
+ .max_register = WM2000_REG_IF_CTL,
+ .readable_reg = wm2000_readable_reg,
};
static int wm2000_probe(struct snd_soc_codec *codec)
diff --git a/sound/soc/codecs/wm5100-tables.c b/sound/soc/codecs/wm5100-tables.c
index e167207..e239f4b 100644
--- a/sound/soc/codecs/wm5100-tables.c
+++ b/sound/soc/codecs/wm5100-tables.c
@@ -1,7 +1,7 @@
/*
* wm5100-tables.c -- WM5100 ALSA SoC Audio driver data
*
- * Copyright 2011 Wolfson Microelectronics plc
+ * Copyright 2011-2 Wolfson Microelectronics plc
*
* Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
*
diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c
index cb6d537..f481729 100644
--- a/sound/soc/codecs/wm5100.c
+++ b/sound/soc/codecs/wm5100.c
@@ -1,7 +1,7 @@
/*
* wm5100.c -- WM5100 ALSA SoC Audio driver
*
- * Copyright 2011 Wolfson Microelectronics plc
+ * Copyright 2011-2 Wolfson Microelectronics plc
*
* Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
*
@@ -2378,13 +2378,6 @@ static int wm5100_remove(struct snd_soc_codec *codec)
return 0;
}
-static int wm5100_soc_volatile(struct snd_soc_codec *codec,
- unsigned int reg)
-{
- return true;
-}
-
-
static struct snd_soc_codec_driver soc_codec_dev_wm5100 = {
.probe = wm5100_probe,
.remove = wm5100_remove,
@@ -2392,8 +2385,6 @@ static struct snd_soc_codec_driver soc_codec_dev_wm5100 = {
.set_sysclk = wm5100_set_sysclk,
.set_pll = wm5100_set_fll,
.idle_bias_off = 1,
- .reg_cache_size = WM5100_MAX_REGISTER,
- .volatile_register = wm5100_soc_volatile,
.seq_notifier = wm5100_seq_notifier,
.controls = wm5100_snd_controls,
diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c
new file mode 100644
index 0000000..6537f16
--- /dev/null
+++ b/sound/soc/codecs/wm5102.c
@@ -0,0 +1,903 @@
+/*
+ * wm5102.c -- WM5102 ALSA SoC Audio driver
+ *
+ * Copyright 2012 Wolfson Microelectronics plc
+ *
+ * Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
+ *
+ * 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 <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/pm_runtime.h>
+#include <linux/regmap.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/jack.h>
+#include <sound/initval.h>
+#include <sound/tlv.h>
+
+#include <linux/mfd/arizona/core.h>
+#include <linux/mfd/arizona/registers.h>
+
+#include "arizona.h"
+#include "wm5102.h"
+
+struct wm5102_priv {
+ struct arizona_priv core;
+ struct arizona_fll fll[2];
+};
+
+static DECLARE_TLV_DB_SCALE(ana_tlv, 0, 100, 0);
+static DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0);
+static DECLARE_TLV_DB_SCALE(digital_tlv, -6400, 50, 0);
+static DECLARE_TLV_DB_SCALE(noise_tlv, 0, 600, 0);
+
+static const struct snd_kcontrol_new wm5102_snd_controls[] = {
+SOC_SINGLE("IN1 High Performance Switch", ARIZONA_IN1L_CONTROL,
+ ARIZONA_IN1_OSR_SHIFT, 1, 0),
+SOC_SINGLE("IN2 High Performance Switch", ARIZONA_IN2L_CONTROL,
+ ARIZONA_IN2_OSR_SHIFT, 1, 0),
+SOC_SINGLE("IN3 High Performance Switch", ARIZONA_IN3L_CONTROL,
+ ARIZONA_IN3_OSR_SHIFT, 1, 0),
+
+SOC_DOUBLE_R_RANGE_TLV("IN1 Volume", ARIZONA_IN1L_CONTROL,
+ ARIZONA_IN1R_CONTROL,
+ ARIZONA_IN1L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv),
+SOC_DOUBLE_R_RANGE_TLV("IN2 Volume", ARIZONA_IN2L_CONTROL,
+ ARIZONA_IN2R_CONTROL,
+ ARIZONA_IN2L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv),
+SOC_DOUBLE_R_RANGE_TLV("IN3 Volume", ARIZONA_IN3L_CONTROL,
+ ARIZONA_IN3R_CONTROL,
+ ARIZONA_IN3L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv),
+
+SOC_DOUBLE_R("IN1 Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_1L,
+ ARIZONA_ADC_DIGITAL_VOLUME_1R, ARIZONA_IN1L_MUTE_SHIFT, 1, 1),
+SOC_DOUBLE_R("IN2 Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_2L,
+ ARIZONA_ADC_DIGITAL_VOLUME_2R, ARIZONA_IN2L_MUTE_SHIFT, 1, 1),
+SOC_DOUBLE_R("IN3 Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_3L,
+ ARIZONA_ADC_DIGITAL_VOLUME_3R, ARIZONA_IN3L_MUTE_SHIFT, 1, 1),
+
+SOC_DOUBLE_R_TLV("IN1 Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_1L,
+ ARIZONA_ADC_DIGITAL_VOLUME_1R, ARIZONA_IN1L_DIG_VOL_SHIFT,
+ 0xbf, 0, digital_tlv),
+SOC_DOUBLE_R_TLV("IN2 Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_2L,
+ ARIZONA_ADC_DIGITAL_VOLUME_2R, ARIZONA_IN2L_DIG_VOL_SHIFT,
+ 0xbf, 0, digital_tlv),
+SOC_DOUBLE_R_TLV("IN3 Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_3L,
+ ARIZONA_ADC_DIGITAL_VOLUME_3R, ARIZONA_IN3L_DIG_VOL_SHIFT,
+ 0xbf, 0, digital_tlv),
+
+ARIZONA_MIXER_CONTROLS("EQ1", ARIZONA_EQ1MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("EQ2", ARIZONA_EQ2MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("EQ3", ARIZONA_EQ3MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("EQ4", ARIZONA_EQ4MIX_INPUT_1_SOURCE),
+
+SOC_SINGLE_TLV("EQ1 B1 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B1_GAIN_SHIFT,
+ 24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ1 B2 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B2_GAIN_SHIFT,
+ 24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ1 B3 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B3_GAIN_SHIFT,
+ 24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ1 B4 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B4_GAIN_SHIFT,
+ 24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ1 B5 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B5_GAIN_SHIFT,
+ 24, 0, eq_tlv),
+
+SOC_SINGLE_TLV("EQ2 B1 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B1_GAIN_SHIFT,
+ 24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ2 B2 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B2_GAIN_SHIFT,
+ 24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ2 B3 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B3_GAIN_SHIFT,
+ 24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ2 B4 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B4_GAIN_SHIFT,
+ 24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ2 B5 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B5_GAIN_SHIFT,
+ 24, 0, eq_tlv),
+
+SOC_SINGLE_TLV("EQ3 B1 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B1_GAIN_SHIFT,
+ 24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ3 B2 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B2_GAIN_SHIFT,
+ 24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ3 B3 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B3_GAIN_SHIFT,
+ 24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ3 B4 Volume", ARIZONA_EQ3_2, ARIZONA_EQ3_B4_GAIN_SHIFT,
+ 24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ3 B5 Volume", ARIZONA_EQ3_2, ARIZONA_EQ3_B5_GAIN_SHIFT,
+ 24, 0, eq_tlv),
+
+SOC_SINGLE_TLV("EQ4 B1 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B1_GAIN_SHIFT,
+ 24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ4 B2 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B2_GAIN_SHIFT,
+ 24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ4 B3 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B3_GAIN_SHIFT,
+ 24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ4 B4 Volume", ARIZONA_EQ4_2, ARIZONA_EQ4_B4_GAIN_SHIFT,
+ 24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ4 B5 Volume", ARIZONA_EQ4_2, ARIZONA_EQ4_B5_GAIN_SHIFT,
+ 24, 0, eq_tlv),
+
+ARIZONA_MIXER_CONTROLS("DRC1L", ARIZONA_DRC1LMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("DRC1R", ARIZONA_DRC1RMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("DRC2L", ARIZONA_DRC2LMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("DRC2R", ARIZONA_DRC2RMIX_INPUT_1_SOURCE),
+
+SND_SOC_BYTES_MASK("DRC1", ARIZONA_DRC1_CTRL1, 5,
+ ARIZONA_DRC1R_ENA | ARIZONA_DRC1L_ENA),
+SND_SOC_BYTES_MASK("DRC2", ARIZONA_DRC2_CTRL1, 5,
+ ARIZONA_DRC2R_ENA | ARIZONA_DRC2L_ENA),
+
+ARIZONA_MIXER_CONTROLS("LHPF1", ARIZONA_HPLP1MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("LHPF2", ARIZONA_HPLP2MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("LHPF3", ARIZONA_HPLP3MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("LHPF4", ARIZONA_HPLP4MIX_INPUT_1_SOURCE),
+
+SOC_ENUM("LHPF1 Mode", arizona_lhpf1_mode),
+SOC_ENUM("LHPF2 Mode", arizona_lhpf2_mode),
+SOC_ENUM("LHPF3 Mode", arizona_lhpf3_mode),
+SOC_ENUM("LHPF4 Mode", arizona_lhpf4_mode),
+
+ARIZONA_MIXER_CONTROLS("Mic", ARIZONA_MICMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("Noise", ARIZONA_NOISEMIX_INPUT_1_SOURCE),
+
+SOC_SINGLE_TLV("Noise Generator Volume", ARIZONA_COMFORT_NOISE_GENERATOR,
+ ARIZONA_NOISE_GEN_GAIN_SHIFT, 0x16, 0, noise_tlv),
+
+ARIZONA_MIXER_CONTROLS("HPOUT1L", ARIZONA_OUT1LMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("HPOUT1R", ARIZONA_OUT1RMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("HPOUT2L", ARIZONA_OUT2LMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("HPOUT2R", ARIZONA_OUT2RMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("EPOUT", ARIZONA_OUT3LMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("SPKOUTL", ARIZONA_OUT4LMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("SPKOUTR", ARIZONA_OUT4RMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("SPKDAT1L", ARIZONA_OUT5LMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("SPKDAT1R", ARIZONA_OUT5RMIX_INPUT_1_SOURCE),
+
+SOC_SINGLE("HPOUT1 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_1L,
+ ARIZONA_OUT1_OSR_SHIFT, 1, 0),
+SOC_SINGLE("OUT2 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_2L,
+ ARIZONA_OUT2_OSR_SHIFT, 1, 0),
+SOC_SINGLE("EPOUT High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_3L,
+ ARIZONA_OUT3_OSR_SHIFT, 1, 0),
+SOC_SINGLE("Speaker High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_4L,
+ ARIZONA_OUT4_OSR_SHIFT, 1, 0),
+SOC_SINGLE("SPKDAT1 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_5L,
+ ARIZONA_OUT5_OSR_SHIFT, 1, 0),
+
+SOC_DOUBLE_R("HPOUT1 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_1L,
+ ARIZONA_DAC_DIGITAL_VOLUME_1R, ARIZONA_OUT1L_MUTE_SHIFT, 1, 1),
+SOC_DOUBLE_R("OUT2 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_2L,
+ ARIZONA_DAC_DIGITAL_VOLUME_2R, ARIZONA_OUT2L_MUTE_SHIFT, 1, 1),
+SOC_SINGLE("EPOUT Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_3L,
+ ARIZONA_OUT3L_MUTE_SHIFT, 1, 1),
+SOC_DOUBLE_R("Speaker Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_4L,
+ ARIZONA_DAC_DIGITAL_VOLUME_4R, ARIZONA_OUT4L_MUTE_SHIFT, 1, 1),
+SOC_DOUBLE_R("SPKDAT1 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_5L,
+ ARIZONA_DAC_DIGITAL_VOLUME_5R, ARIZONA_OUT5L_MUTE_SHIFT, 1, 1),
+
+SOC_DOUBLE_R_TLV("HPOUT1 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_1L,
+ ARIZONA_DAC_DIGITAL_VOLUME_1R, ARIZONA_OUT1L_VOL_SHIFT,
+ 0xbf, 0, digital_tlv),
+SOC_DOUBLE_R_TLV("OUT2 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_2L,
+ ARIZONA_DAC_DIGITAL_VOLUME_2R, ARIZONA_OUT2L_VOL_SHIFT,
+ 0xbf, 0, digital_tlv),
+SOC_SINGLE_TLV("EPOUT Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_3L,
+ ARIZONA_OUT3L_VOL_SHIFT, 0xbf, 0, digital_tlv),
+SOC_DOUBLE_R_TLV("Speaker Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_4L,
+ ARIZONA_DAC_DIGITAL_VOLUME_4R, ARIZONA_OUT4L_VOL_SHIFT,
+ 0xbf, 0, digital_tlv),
+SOC_DOUBLE_R_TLV("SPKDAT1 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_5L,
+ ARIZONA_DAC_DIGITAL_VOLUME_5R, ARIZONA_OUT5L_VOL_SHIFT,
+ 0xbf, 0, digital_tlv),
+
+SOC_DOUBLE_R_RANGE_TLV("HPOUT1 Volume", ARIZONA_OUTPUT_PATH_CONFIG_1L,
+ ARIZONA_OUTPUT_PATH_CONFIG_1R,
+ ARIZONA_OUT1L_PGA_VOL_SHIFT,
+ 0x34, 0x40, 0, ana_tlv),
+SOC_DOUBLE_R_RANGE_TLV("OUT2 Volume", ARIZONA_OUTPUT_PATH_CONFIG_2L,
+ ARIZONA_OUTPUT_PATH_CONFIG_2R,
+ ARIZONA_OUT2L_PGA_VOL_SHIFT,
+ 0x34, 0x40, 0, ana_tlv),
+SOC_SINGLE_RANGE_TLV("EPOUT Volume", ARIZONA_OUTPUT_PATH_CONFIG_3L,
+ ARIZONA_OUT3L_PGA_VOL_SHIFT, 0x34, 0x40, 0, ana_tlv),
+
+SOC_DOUBLE("SPKDAT1 Switch", ARIZONA_PDM_SPK1_CTRL_1, ARIZONA_SPK1L_MUTE_SHIFT,
+ ARIZONA_SPK1R_MUTE_SHIFT, 1, 1),
+
+ARIZONA_MIXER_CONTROLS("AIF1TX1", ARIZONA_AIF1TX1MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("AIF1TX2", ARIZONA_AIF1TX2MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("AIF1TX3", ARIZONA_AIF1TX3MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("AIF1TX4", ARIZONA_AIF1TX4MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("AIF1TX5", ARIZONA_AIF1TX5MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("AIF1TX6", ARIZONA_AIF1TX6MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("AIF1TX7", ARIZONA_AIF1TX7MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("AIF1TX8", ARIZONA_AIF1TX8MIX_INPUT_1_SOURCE),
+
+ARIZONA_MIXER_CONTROLS("AIF2TX1", ARIZONA_AIF2TX1MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("AIF2TX2", ARIZONA_AIF2TX2MIX_INPUT_1_SOURCE),
+
+ARIZONA_MIXER_CONTROLS("AIF3TX1", ARIZONA_AIF3TX1MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("AIF3TX2", ARIZONA_AIF3TX2MIX_INPUT_1_SOURCE),
+};
+
+ARIZONA_MIXER_ENUMS(EQ1, ARIZONA_EQ1MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(EQ2, ARIZONA_EQ2MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(EQ3, ARIZONA_EQ3MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(EQ4, ARIZONA_EQ4MIX_INPUT_1_SOURCE);
+
+ARIZONA_MIXER_ENUMS(DRC1L, ARIZONA_DRC1LMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(DRC1R, ARIZONA_DRC1RMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(DRC2L, ARIZONA_DRC2LMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(DRC2R, ARIZONA_DRC2RMIX_INPUT_1_SOURCE);
+
+ARIZONA_MIXER_ENUMS(LHPF1, ARIZONA_HPLP1MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(LHPF2, ARIZONA_HPLP2MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(LHPF3, ARIZONA_HPLP3MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(LHPF4, ARIZONA_HPLP4MIX_INPUT_1_SOURCE);
+
+ARIZONA_MIXER_ENUMS(Mic, ARIZONA_MICMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(Noise, ARIZONA_NOISEMIX_INPUT_1_SOURCE);
+
+ARIZONA_MIXER_ENUMS(PWM1, ARIZONA_PWM1MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(PWM2, ARIZONA_PWM2MIX_INPUT_1_SOURCE);
+
+ARIZONA_MIXER_ENUMS(OUT1L, ARIZONA_OUT1LMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(OUT1R, ARIZONA_OUT1RMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(OUT2L, ARIZONA_OUT2LMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(OUT2R, ARIZONA_OUT2RMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(OUT3, ARIZONA_OUT3LMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(SPKOUTL, ARIZONA_OUT4LMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(SPKOUTR, ARIZONA_OUT4RMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(SPKDAT1L, ARIZONA_OUT5LMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(SPKDAT1R, ARIZONA_OUT5RMIX_INPUT_1_SOURCE);
+
+ARIZONA_MIXER_ENUMS(AIF1TX1, ARIZONA_AIF1TX1MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(AIF1TX2, ARIZONA_AIF1TX2MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(AIF1TX3, ARIZONA_AIF1TX3MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(AIF1TX4, ARIZONA_AIF1TX4MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(AIF1TX5, ARIZONA_AIF1TX5MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(AIF1TX6, ARIZONA_AIF1TX6MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(AIF1TX7, ARIZONA_AIF1TX7MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(AIF1TX8, ARIZONA_AIF1TX8MIX_INPUT_1_SOURCE);
+
+ARIZONA_MIXER_ENUMS(AIF2TX1, ARIZONA_AIF2TX1MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(AIF2TX2, ARIZONA_AIF2TX2MIX_INPUT_1_SOURCE);
+
+ARIZONA_MIXER_ENUMS(AIF3TX1, ARIZONA_AIF3TX1MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(AIF3TX2, ARIZONA_AIF3TX2MIX_INPUT_1_SOURCE);
+
+ARIZONA_MIXER_ENUMS(ASRC1L, ARIZONA_ASRC1LMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(ASRC1R, ARIZONA_ASRC1RMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(ASRC2L, ARIZONA_ASRC2LMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(ASRC2R, ARIZONA_ASRC2RMIX_INPUT_1_SOURCE);
+
+static const struct snd_soc_dapm_widget wm5102_dapm_widgets[] = {
+SND_SOC_DAPM_SUPPLY("SYSCLK", ARIZONA_SYSTEM_CLOCK_1, ARIZONA_SYSCLK_ENA_SHIFT,
+ 0, NULL, 0),
+SND_SOC_DAPM_SUPPLY("ASYNCCLK", ARIZONA_ASYNC_CLOCK_1,
+ ARIZONA_ASYNC_CLK_ENA_SHIFT, 0, NULL, 0),
+
+SND_SOC_DAPM_REGULATOR_SUPPLY("DBVDD2", 0),
+SND_SOC_DAPM_REGULATOR_SUPPLY("DBVDD3", 0),
+SND_SOC_DAPM_REGULATOR_SUPPLY("CPVDD", 20),
+SND_SOC_DAPM_REGULATOR_SUPPLY("MICVDD", 0),
+SND_SOC_DAPM_REGULATOR_SUPPLY("SPKVDDL", 0),
+SND_SOC_DAPM_REGULATOR_SUPPLY("SPKVDDR", 0),
+
+SND_SOC_DAPM_SIGGEN("TONE"),
+SND_SOC_DAPM_SIGGEN("NOISE"),
+
+SND_SOC_DAPM_INPUT("IN1L"),
+SND_SOC_DAPM_INPUT("IN1R"),
+SND_SOC_DAPM_INPUT("IN2L"),
+SND_SOC_DAPM_INPUT("IN2R"),
+SND_SOC_DAPM_INPUT("IN3L"),
+SND_SOC_DAPM_INPUT("IN3R"),
+
+SND_SOC_DAPM_PGA_E("IN1L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN1L_ENA_SHIFT,
+ 0, NULL, 0, arizona_in_ev,
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("IN1R PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN1R_ENA_SHIFT,
+ 0, NULL, 0, arizona_in_ev,
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("IN2L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN2L_ENA_SHIFT,
+ 0, NULL, 0, arizona_in_ev,
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("IN2R PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN2R_ENA_SHIFT,
+ 0, NULL, 0, arizona_in_ev,
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("IN3L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN3L_ENA_SHIFT,
+ 0, NULL, 0, arizona_in_ev,
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("IN3R PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN3R_ENA_SHIFT,
+ 0, NULL, 0, arizona_in_ev,
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+
+SND_SOC_DAPM_SUPPLY("MICBIAS1", ARIZONA_MIC_BIAS_CTRL_1,
+ ARIZONA_MICB1_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_SUPPLY("MICBIAS2", ARIZONA_MIC_BIAS_CTRL_2,
+ ARIZONA_MICB1_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_SUPPLY("MICBIAS3", ARIZONA_MIC_BIAS_CTRL_3,
+ ARIZONA_MICB1_ENA_SHIFT, 0, NULL, 0),
+
+SND_SOC_DAPM_PGA("Noise Generator", ARIZONA_COMFORT_NOISE_GENERATOR,
+ ARIZONA_NOISE_GEN_ENA_SHIFT, 0, NULL, 0),
+
+SND_SOC_DAPM_PGA("Tone Generator 1", ARIZONA_TONE_GENERATOR_1,
+ ARIZONA_TONE1_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA("Tone Generator 2", ARIZONA_TONE_GENERATOR_1,
+ ARIZONA_TONE2_ENA_SHIFT, 0, NULL, 0),
+
+SND_SOC_DAPM_PGA("Mic Mute Mixer", ARIZONA_MIC_NOISE_MIX_CONTROL_1,
+ ARIZONA_MICMUTE_MIX_ENA_SHIFT, 0, NULL, 0),
+
+SND_SOC_DAPM_PGA("EQ1", ARIZONA_EQ1_1, ARIZONA_EQ1_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA("EQ2", ARIZONA_EQ2_1, ARIZONA_EQ2_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA("EQ3", ARIZONA_EQ3_1, ARIZONA_EQ3_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA("EQ4", ARIZONA_EQ4_1, ARIZONA_EQ4_ENA_SHIFT, 0, NULL, 0),
+
+SND_SOC_DAPM_PGA("DRC1L", ARIZONA_DRC1_CTRL1, ARIZONA_DRC1L_ENA_SHIFT, 0,
+ NULL, 0),
+SND_SOC_DAPM_PGA("DRC1R", ARIZONA_DRC1_CTRL1, ARIZONA_DRC1R_ENA_SHIFT, 0,
+ NULL, 0),
+SND_SOC_DAPM_PGA("DRC2L", ARIZONA_DRC2_CTRL1, ARIZONA_DRC2L_ENA_SHIFT, 0,
+ NULL, 0),
+SND_SOC_DAPM_PGA("DRC2R", ARIZONA_DRC2_CTRL1, ARIZONA_DRC2R_ENA_SHIFT, 0,
+ NULL, 0),
+
+SND_SOC_DAPM_PGA("LHPF1", ARIZONA_HPLPF1_1, ARIZONA_LHPF1_ENA_SHIFT, 0,
+ NULL, 0),
+SND_SOC_DAPM_PGA("LHPF2", ARIZONA_HPLPF2_1, ARIZONA_LHPF2_ENA_SHIFT, 0,
+ NULL, 0),
+SND_SOC_DAPM_PGA("LHPF3", ARIZONA_HPLPF3_1, ARIZONA_LHPF3_ENA_SHIFT, 0,
+ NULL, 0),
+SND_SOC_DAPM_PGA("LHPF4", ARIZONA_HPLPF4_1, ARIZONA_LHPF4_ENA_SHIFT, 0,
+ NULL, 0),
+
+SND_SOC_DAPM_PGA("PWM1 Driver", ARIZONA_PWM_DRIVE_1, ARIZONA_PWM1_ENA_SHIFT,
+ 0, NULL, 0),
+SND_SOC_DAPM_PGA("PWM2 Driver", ARIZONA_PWM_DRIVE_1, ARIZONA_PWM2_ENA_SHIFT,
+ 0, NULL, 0),
+
+SND_SOC_DAPM_PGA("ASRC1L", ARIZONA_ASRC_ENABLE, ARIZONA_ASRC1L_ENA_SHIFT, 0,
+ NULL, 0),
+SND_SOC_DAPM_PGA("ASRC1R", ARIZONA_ASRC_ENABLE, ARIZONA_ASRC1R_ENA_SHIFT, 0,
+ NULL, 0),
+SND_SOC_DAPM_PGA("ASRC2L", ARIZONA_ASRC_ENABLE, ARIZONA_ASRC2L_ENA_SHIFT, 0,
+ NULL, 0),
+SND_SOC_DAPM_PGA("ASRC2R", ARIZONA_ASRC_ENABLE, ARIZONA_ASRC2R_ENA_SHIFT, 0,
+ NULL, 0),
+
+SND_SOC_DAPM_AIF_OUT("AIF1TX1", NULL, 0,
+ ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX1_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF1TX2", NULL, 0,
+ ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX2_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF1TX3", NULL, 0,
+ ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX3_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF1TX4", NULL, 0,
+ ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX4_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF1TX5", NULL, 0,
+ ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX5_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF1TX6", NULL, 0,
+ ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX6_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF1TX7", NULL, 0,
+ ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX7_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF1TX8", NULL, 0,
+ ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX8_ENA_SHIFT, 0),
+
+SND_SOC_DAPM_AIF_IN("AIF1RX1", NULL, 0,
+ ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX1_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF1RX2", NULL, 0,
+ ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX2_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF1RX3", NULL, 0,
+ ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX3_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF1RX4", NULL, 0,
+ ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX4_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF1RX5", NULL, 0,
+ ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX5_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF1RX6", NULL, 0,
+ ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX6_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF1RX7", NULL, 0,
+ ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX7_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF1RX8", NULL, 0,
+ ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX8_ENA_SHIFT, 0),
+
+SND_SOC_DAPM_AIF_OUT("AIF2TX1", NULL, 0,
+ ARIZONA_AIF2_TX_ENABLES, ARIZONA_AIF2TX1_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF2TX2", NULL, 0,
+ ARIZONA_AIF2_TX_ENABLES, ARIZONA_AIF2TX2_ENA_SHIFT, 0),
+
+SND_SOC_DAPM_AIF_IN("AIF2RX1", NULL, 0,
+ ARIZONA_AIF2_RX_ENABLES, ARIZONA_AIF2RX1_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF2RX2", NULL, 0,
+ ARIZONA_AIF2_RX_ENABLES, ARIZONA_AIF2RX2_ENA_SHIFT, 0),
+
+SND_SOC_DAPM_AIF_OUT("AIF3TX1", NULL, 0,
+ ARIZONA_AIF3_TX_ENABLES, ARIZONA_AIF3TX1_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF3TX2", NULL, 0,
+ ARIZONA_AIF3_TX_ENABLES, ARIZONA_AIF3TX2_ENA_SHIFT, 0),
+
+SND_SOC_DAPM_AIF_IN("AIF3RX1", NULL, 0,
+ ARIZONA_AIF3_RX_ENABLES, ARIZONA_AIF3RX1_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF3RX2", NULL, 0,
+ ARIZONA_AIF3_RX_ENABLES, ARIZONA_AIF3RX2_ENA_SHIFT, 0),
+
+SND_SOC_DAPM_PGA_E("OUT1L", ARIZONA_OUTPUT_ENABLES_1,
+ ARIZONA_OUT1L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("OUT1R", ARIZONA_OUTPUT_ENABLES_1,
+ ARIZONA_OUT1R_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("OUT2L", ARIZONA_OUTPUT_ENABLES_1,
+ ARIZONA_OUT2L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("OUT2R", ARIZONA_OUTPUT_ENABLES_1,
+ ARIZONA_OUT2R_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("OUT3L", ARIZONA_OUTPUT_ENABLES_1,
+ ARIZONA_OUT3L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("OUT4L", ARIZONA_OUTPUT_ENABLES_1,
+ ARIZONA_OUT4L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("OUT4R", ARIZONA_OUTPUT_ENABLES_1,
+ ARIZONA_OUT4R_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("OUT5L", ARIZONA_OUTPUT_ENABLES_1,
+ ARIZONA_OUT5L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("OUT5R", ARIZONA_OUTPUT_ENABLES_1,
+ ARIZONA_OUT5R_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+
+ARIZONA_MIXER_WIDGETS(EQ1, "EQ1"),
+ARIZONA_MIXER_WIDGETS(EQ2, "EQ2"),
+ARIZONA_MIXER_WIDGETS(EQ3, "EQ3"),
+ARIZONA_MIXER_WIDGETS(EQ4, "EQ4"),
+
+ARIZONA_MIXER_WIDGETS(DRC1L, "DRC1L"),
+ARIZONA_MIXER_WIDGETS(DRC1R, "DRC1R"),
+ARIZONA_MIXER_WIDGETS(DRC2L, "DRC2L"),
+ARIZONA_MIXER_WIDGETS(DRC2R, "DRC2R"),
+
+ARIZONA_MIXER_WIDGETS(LHPF1, "LHPF1"),
+ARIZONA_MIXER_WIDGETS(LHPF2, "LHPF2"),
+ARIZONA_MIXER_WIDGETS(LHPF3, "LHPF3"),
+ARIZONA_MIXER_WIDGETS(LHPF4, "LHPF4"),
+
+ARIZONA_MIXER_WIDGETS(Mic, "Mic"),
+ARIZONA_MIXER_WIDGETS(Noise, "Noise"),
+
+ARIZONA_MIXER_WIDGETS(PWM1, "PWM1"),
+ARIZONA_MIXER_WIDGETS(PWM2, "PWM2"),
+
+ARIZONA_MIXER_WIDGETS(OUT1L, "HPOUT1L"),
+ARIZONA_MIXER_WIDGETS(OUT1R, "HPOUT1R"),
+ARIZONA_MIXER_WIDGETS(OUT2L, "HPOUT2L"),
+ARIZONA_MIXER_WIDGETS(OUT2R, "HPOUT2R"),
+ARIZONA_MIXER_WIDGETS(OUT3, "EPOUT"),
+ARIZONA_MIXER_WIDGETS(SPKOUTL, "SPKOUTL"),
+ARIZONA_MIXER_WIDGETS(SPKOUTR, "SPKOUTR"),
+ARIZONA_MIXER_WIDGETS(SPKDAT1L, "SPKDAT1L"),
+ARIZONA_MIXER_WIDGETS(SPKDAT1R, "SPKDAT1R"),
+
+ARIZONA_MIXER_WIDGETS(AIF1TX1, "AIF1TX1"),
+ARIZONA_MIXER_WIDGETS(AIF1TX2, "AIF1TX2"),
+ARIZONA_MIXER_WIDGETS(AIF1TX3, "AIF1TX3"),
+ARIZONA_MIXER_WIDGETS(AIF1TX4, "AIF1TX4"),
+ARIZONA_MIXER_WIDGETS(AIF1TX5, "AIF1TX5"),
+ARIZONA_MIXER_WIDGETS(AIF1TX6, "AIF1TX6"),
+ARIZONA_MIXER_WIDGETS(AIF1TX7, "AIF1TX7"),
+ARIZONA_MIXER_WIDGETS(AIF1TX8, "AIF1TX8"),
+
+ARIZONA_MIXER_WIDGETS(AIF2TX1, "AIF2TX1"),
+ARIZONA_MIXER_WIDGETS(AIF2TX2, "AIF2TX2"),
+
+ARIZONA_MIXER_WIDGETS(AIF3TX1, "AIF3TX1"),
+ARIZONA_MIXER_WIDGETS(AIF3TX2, "AIF3TX2"),
+
+ARIZONA_MIXER_WIDGETS(ASRC1L, "ASRC1L"),
+ARIZONA_MIXER_WIDGETS(ASRC1R, "ASRC1R"),
+ARIZONA_MIXER_WIDGETS(ASRC2L, "ASRC2L"),
+ARIZONA_MIXER_WIDGETS(ASRC2R, "ASRC2R"),
+
+SND_SOC_DAPM_OUTPUT("HPOUT1L"),
+SND_SOC_DAPM_OUTPUT("HPOUT1R"),
+SND_SOC_DAPM_OUTPUT("HPOUT2L"),
+SND_SOC_DAPM_OUTPUT("HPOUT2R"),
+SND_SOC_DAPM_OUTPUT("EPOUTN"),
+SND_SOC_DAPM_OUTPUT("EPOUTP"),
+SND_SOC_DAPM_OUTPUT("SPKOUTLN"),
+SND_SOC_DAPM_OUTPUT("SPKOUTLP"),
+SND_SOC_DAPM_OUTPUT("SPKOUTRN"),
+SND_SOC_DAPM_OUTPUT("SPKOUTRP"),
+SND_SOC_DAPM_OUTPUT("SPKDAT1L"),
+SND_SOC_DAPM_OUTPUT("SPKDAT1R"),
+};
+
+#define ARIZONA_MIXER_INPUT_ROUTES(name) \
+ { name, "Noise Generator", "Noise Generator" }, \
+ { name, "Tone Generator 1", "Tone Generator 1" }, \
+ { name, "Tone Generator 2", "Tone Generator 2" }, \
+ { name, "IN1L", "IN1L PGA" }, \
+ { name, "IN1R", "IN1R PGA" }, \
+ { name, "IN2L", "IN2L PGA" }, \
+ { name, "IN2R", "IN2R PGA" }, \
+ { name, "IN3L", "IN3L PGA" }, \
+ { name, "IN3R", "IN3R PGA" }, \
+ { name, "Mic Mute Mixer", "Mic Mute Mixer" }, \
+ { name, "AIF1RX1", "AIF1RX1" }, \
+ { name, "AIF1RX2", "AIF1RX2" }, \
+ { name, "AIF1RX3", "AIF1RX3" }, \
+ { name, "AIF1RX4", "AIF1RX4" }, \
+ { name, "AIF1RX5", "AIF1RX5" }, \
+ { name, "AIF1RX6", "AIF1RX6" }, \
+ { name, "AIF1RX7", "AIF1RX7" }, \
+ { name, "AIF1RX8", "AIF1RX8" }, \
+ { name, "AIF2RX1", "AIF2RX1" }, \
+ { name, "AIF2RX2", "AIF2RX2" }, \
+ { name, "AIF3RX1", "AIF3RX1" }, \
+ { name, "AIF3RX2", "AIF3RX2" }, \
+ { name, "EQ1", "EQ1" }, \
+ { name, "EQ2", "EQ2" }, \
+ { name, "EQ3", "EQ3" }, \
+ { name, "EQ4", "EQ4" }, \
+ { name, "DRC1L", "DRC1L" }, \
+ { name, "DRC1R", "DRC1R" }, \
+ { name, "DRC2L", "DRC2L" }, \
+ { name, "DRC2R", "DRC2R" }, \
+ { name, "LHPF1", "LHPF1" }, \
+ { name, "LHPF2", "LHPF2" }, \
+ { name, "LHPF3", "LHPF3" }, \
+ { name, "LHPF4", "LHPF4" }, \
+ { name, "ASRC1L", "ASRC1L" }, \
+ { name, "ASRC1R", "ASRC1R" }, \
+ { name, "ASRC2L", "ASRC2L" }, \
+ { name, "ASRC2R", "ASRC2R" }
+
+static const struct snd_soc_dapm_route wm5102_dapm_routes[] = {
+ { "AIF2 Capture", NULL, "DBVDD2" },
+ { "AIF2 Playback", NULL, "DBVDD2" },
+
+ { "AIF3 Capture", NULL, "DBVDD3" },
+ { "AIF3 Playback", NULL, "DBVDD3" },
+
+ { "OUT1L", NULL, "CPVDD" },
+ { "OUT1R", NULL, "CPVDD" },
+ { "OUT2L", NULL, "CPVDD" },
+ { "OUT2R", NULL, "CPVDD" },
+ { "OUT3L", NULL, "CPVDD" },
+
+ { "OUT4L", NULL, "SPKVDDL" },
+ { "OUT4R", NULL, "SPKVDDR" },
+
+ { "OUT1L", NULL, "SYSCLK" },
+ { "OUT1R", NULL, "SYSCLK" },
+ { "OUT2L", NULL, "SYSCLK" },
+ { "OUT2R", NULL, "SYSCLK" },
+ { "OUT3L", NULL, "SYSCLK" },
+ { "OUT4L", NULL, "SYSCLK" },
+ { "OUT4R", NULL, "SYSCLK" },
+ { "OUT5L", NULL, "SYSCLK" },
+ { "OUT5R", NULL, "SYSCLK" },
+
+ { "MICBIAS1", NULL, "MICVDD" },
+ { "MICBIAS2", NULL, "MICVDD" },
+ { "MICBIAS3", NULL, "MICVDD" },
+
+ { "Noise Generator", NULL, "NOISE" },
+ { "Tone Generator 1", NULL, "TONE" },
+ { "Tone Generator 2", NULL, "TONE" },
+
+ { "Mic Mute Mixer", NULL, "Noise Mixer" },
+ { "Mic Mute Mixer", NULL, "Mic Mixer" },
+
+ { "AIF1 Capture", NULL, "AIF1TX1" },
+ { "AIF1 Capture", NULL, "AIF1TX2" },
+ { "AIF1 Capture", NULL, "AIF1TX3" },
+ { "AIF1 Capture", NULL, "AIF1TX4" },
+ { "AIF1 Capture", NULL, "AIF1TX5" },
+ { "AIF1 Capture", NULL, "AIF1TX6" },
+ { "AIF1 Capture", NULL, "AIF1TX7" },
+ { "AIF1 Capture", NULL, "AIF1TX8" },
+
+ { "AIF1RX1", NULL, "AIF1 Playback" },
+ { "AIF1RX2", NULL, "AIF1 Playback" },
+ { "AIF1RX3", NULL, "AIF1 Playback" },
+ { "AIF1RX4", NULL, "AIF1 Playback" },
+ { "AIF1RX5", NULL, "AIF1 Playback" },
+ { "AIF1RX6", NULL, "AIF1 Playback" },
+ { "AIF1RX7", NULL, "AIF1 Playback" },
+ { "AIF1RX8", NULL, "AIF1 Playback" },
+
+ { "AIF2 Capture", NULL, "AIF2TX1" },
+ { "AIF2 Capture", NULL, "AIF2TX2" },
+
+ { "AIF2RX1", NULL, "AIF2 Playback" },
+ { "AIF2RX2", NULL, "AIF2 Playback" },
+
+ { "AIF3 Capture", NULL, "AIF3TX1" },
+ { "AIF3 Capture", NULL, "AIF3TX2" },
+
+ { "AIF3RX1", NULL, "AIF3 Playback" },
+ { "AIF3RX2", NULL, "AIF3 Playback" },
+
+ { "AIF1 Playback", NULL, "SYSCLK" },
+ { "AIF2 Playback", NULL, "SYSCLK" },
+ { "AIF3 Playback", NULL, "SYSCLK" },
+
+ { "AIF1 Capture", NULL, "SYSCLK" },
+ { "AIF2 Capture", NULL, "SYSCLK" },
+ { "AIF3 Capture", NULL, "SYSCLK" },
+
+ ARIZONA_MIXER_ROUTES("OUT1L", "HPOUT1L"),
+ ARIZONA_MIXER_ROUTES("OUT1R", "HPOUT1R"),
+ ARIZONA_MIXER_ROUTES("OUT2L", "HPOUT2L"),
+ ARIZONA_MIXER_ROUTES("OUT2R", "HPOUT2R"),
+ ARIZONA_MIXER_ROUTES("OUT3L", "EPOUT"),
+
+ ARIZONA_MIXER_ROUTES("OUT4L", "SPKOUTL"),
+ ARIZONA_MIXER_ROUTES("OUT4R", "SPKOUTR"),
+ ARIZONA_MIXER_ROUTES("OUT5L", "SPKDAT1L"),
+ ARIZONA_MIXER_ROUTES("OUT5R", "SPKDAT1R"),
+
+ ARIZONA_MIXER_ROUTES("PWM1 Driver", "PWM1"),
+ ARIZONA_MIXER_ROUTES("PWM2 Driver", "PWM2"),
+
+ ARIZONA_MIXER_ROUTES("AIF1TX1", "AIF1TX1"),
+ ARIZONA_MIXER_ROUTES("AIF1TX2", "AIF1TX2"),
+ ARIZONA_MIXER_ROUTES("AIF1TX3", "AIF1TX3"),
+ ARIZONA_MIXER_ROUTES("AIF1TX4", "AIF1TX4"),
+ ARIZONA_MIXER_ROUTES("AIF1TX5", "AIF1TX5"),
+ ARIZONA_MIXER_ROUTES("AIF1TX6", "AIF1TX6"),
+ ARIZONA_MIXER_ROUTES("AIF1TX7", "AIF1TX7"),
+ ARIZONA_MIXER_ROUTES("AIF1TX8", "AIF1TX8"),
+
+ ARIZONA_MIXER_ROUTES("AIF2TX1", "AIF2TX1"),
+ ARIZONA_MIXER_ROUTES("AIF2TX2", "AIF2TX2"),
+
+ ARIZONA_MIXER_ROUTES("AIF3TX1", "AIF3TX1"),
+ ARIZONA_MIXER_ROUTES("AIF3TX2", "AIF3TX2"),
+
+ ARIZONA_MIXER_ROUTES("EQ1", "EQ1"),
+ ARIZONA_MIXER_ROUTES("EQ2", "EQ2"),
+ ARIZONA_MIXER_ROUTES("EQ3", "EQ3"),
+ ARIZONA_MIXER_ROUTES("EQ4", "EQ4"),
+
+ ARIZONA_MIXER_ROUTES("DRC1L", "DRC1L"),
+ ARIZONA_MIXER_ROUTES("DRC1R", "DRC1R"),
+ ARIZONA_MIXER_ROUTES("DRC2L", "DRC2L"),
+ ARIZONA_MIXER_ROUTES("DRC2R", "DRC2R"),
+
+ ARIZONA_MIXER_ROUTES("LHPF1", "LHPF1"),
+ ARIZONA_MIXER_ROUTES("LHPF2", "LHPF2"),
+ ARIZONA_MIXER_ROUTES("LHPF3", "LHPF3"),
+ ARIZONA_MIXER_ROUTES("LHPF4", "LHPF4"),
+
+ ARIZONA_MIXER_ROUTES("ASRC1L", "ASRC1L"),
+ ARIZONA_MIXER_ROUTES("ASRC1R", "ASRC1R"),
+ ARIZONA_MIXER_ROUTES("ASRC2L", "ASRC2L"),
+ ARIZONA_MIXER_ROUTES("ASRC2R", "ASRC2R"),
+
+ { "HPOUT1L", NULL, "OUT1L" },
+ { "HPOUT1R", NULL, "OUT1R" },
+
+ { "HPOUT2L", NULL, "OUT2L" },
+ { "HPOUT2R", NULL, "OUT2R" },
+
+ { "EPOUTN", NULL, "OUT3L" },
+ { "EPOUTP", NULL, "OUT3L" },
+
+ { "SPKOUTLN", NULL, "OUT4L" },
+ { "SPKOUTLP", NULL, "OUT4L" },
+
+ { "SPKOUTRN", NULL, "OUT4R" },
+ { "SPKOUTRP", NULL, "OUT4R" },
+
+ { "SPKDAT1L", NULL, "OUT5L" },
+ { "SPKDAT1R", NULL, "OUT5R" },
+};
+
+static int wm5102_set_fll(struct snd_soc_codec *codec, int fll_id, int source,
+ unsigned int Fref, unsigned int Fout)
+{
+ struct wm5102_priv *wm5102 = snd_soc_codec_get_drvdata(codec);
+
+ switch (fll_id) {
+ case WM5102_FLL1:
+ return arizona_set_fll(&wm5102->fll[0], source, Fref, Fout);
+ case WM5102_FLL2:
+ return arizona_set_fll(&wm5102->fll[1], source, Fref, Fout);
+ default:
+ return -EINVAL;
+ }
+}
+
+#define WM5102_RATES SNDRV_PCM_RATE_8000_192000
+
+#define WM5102_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
+ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
+
+static struct snd_soc_dai_driver wm5102_dai[] = {
+ {
+ .name = "wm5102-aif1",
+ .id = 1,
+ .base = ARIZONA_AIF1_BCLK_CTRL,
+ .playback = {
+ .stream_name = "AIF1 Playback",
+ .channels_min = 1,
+ .channels_max = 8,
+ .rates = WM5102_RATES,
+ .formats = WM5102_FORMATS,
+ },
+ .capture = {
+ .stream_name = "AIF1 Capture",
+ .channels_min = 1,
+ .channels_max = 8,
+ .rates = WM5102_RATES,
+ .formats = WM5102_FORMATS,
+ },
+ .ops = &arizona_dai_ops,
+ .symmetric_rates = 1,
+ },
+ {
+ .name = "wm5102-aif2",
+ .id = 2,
+ .base = ARIZONA_AIF2_BCLK_CTRL,
+ .playback = {
+ .stream_name = "AIF2 Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = WM5102_RATES,
+ .formats = WM5102_FORMATS,
+ },
+ .capture = {
+ .stream_name = "AIF2 Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = WM5102_RATES,
+ .formats = WM5102_FORMATS,
+ },
+ .ops = &arizona_dai_ops,
+ .symmetric_rates = 1,
+ },
+ {
+ .name = "wm5102-aif3",
+ .id = 3,
+ .base = ARIZONA_AIF3_BCLK_CTRL,
+ .playback = {
+ .stream_name = "AIF3 Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = WM5102_RATES,
+ .formats = WM5102_FORMATS,
+ },
+ .capture = {
+ .stream_name = "AIF3 Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = WM5102_RATES,
+ .formats = WM5102_FORMATS,
+ },
+ .ops = &arizona_dai_ops,
+ .symmetric_rates = 1,
+ },
+};
+
+static int wm5102_codec_probe(struct snd_soc_codec *codec)
+{
+ struct wm5102_priv *priv = snd_soc_codec_get_drvdata(codec);
+
+ codec->control_data = priv->core.arizona->regmap;
+ return snd_soc_codec_set_cache_io(codec, 32, 16, SND_SOC_REGMAP);
+}
+
+#define WM5102_DIG_VU 0x0200
+
+static unsigned int wm5102_digital_vu[] = {
+ ARIZONA_ADC_DIGITAL_VOLUME_1L,
+ ARIZONA_ADC_DIGITAL_VOLUME_1R,
+ ARIZONA_ADC_DIGITAL_VOLUME_2L,
+ ARIZONA_ADC_DIGITAL_VOLUME_2R,
+ ARIZONA_ADC_DIGITAL_VOLUME_3L,
+ ARIZONA_ADC_DIGITAL_VOLUME_3R,
+
+ ARIZONA_DAC_DIGITAL_VOLUME_1L,
+ ARIZONA_DAC_DIGITAL_VOLUME_1R,
+ ARIZONA_DAC_DIGITAL_VOLUME_2L,
+ ARIZONA_DAC_DIGITAL_VOLUME_2R,
+ ARIZONA_DAC_DIGITAL_VOLUME_3L,
+ ARIZONA_DAC_DIGITAL_VOLUME_3R,
+ ARIZONA_DAC_DIGITAL_VOLUME_4L,
+ ARIZONA_DAC_DIGITAL_VOLUME_4R,
+ ARIZONA_DAC_DIGITAL_VOLUME_5L,
+ ARIZONA_DAC_DIGITAL_VOLUME_5R,
+};
+
+static struct snd_soc_codec_driver soc_codec_dev_wm5102 = {
+ .probe = wm5102_codec_probe,
+
+ .idle_bias_off = true,
+
+ .set_sysclk = arizona_set_sysclk,
+ .set_pll = wm5102_set_fll,
+
+ .controls = wm5102_snd_controls,
+ .num_controls = ARRAY_SIZE(wm5102_snd_controls),
+ .dapm_widgets = wm5102_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(wm5102_dapm_widgets),
+ .dapm_routes = wm5102_dapm_routes,
+ .num_dapm_routes = ARRAY_SIZE(wm5102_dapm_routes),
+};
+
+static int __devinit wm5102_probe(struct platform_device *pdev)
+{
+ struct arizona *arizona = dev_get_drvdata(pdev->dev.parent);
+ struct wm5102_priv *wm5102;
+ int i;
+
+ wm5102 = devm_kzalloc(&pdev->dev, sizeof(struct wm5102_priv),
+ GFP_KERNEL);
+ if (wm5102 == NULL)
+ return -ENOMEM;
+ platform_set_drvdata(pdev, wm5102);
+
+ wm5102->core.arizona = arizona;
+
+ for (i = 0; i < ARRAY_SIZE(wm5102->fll); i++)
+ wm5102->fll[i].vco_mult = 1;
+
+ arizona_init_fll(arizona, 1, ARIZONA_FLL1_CONTROL_1 - 1,
+ ARIZONA_IRQ_FLL1_LOCK, ARIZONA_IRQ_FLL1_CLOCK_OK,
+ &wm5102->fll[0]);
+ arizona_init_fll(arizona, 2, ARIZONA_FLL2_CONTROL_1 - 1,
+ ARIZONA_IRQ_FLL2_LOCK, ARIZONA_IRQ_FLL2_CLOCK_OK,
+ &wm5102->fll[1]);
+
+ for (i = 0; i < ARRAY_SIZE(wm5102_dai); i++)
+ arizona_init_dai(&wm5102->core, i);
+
+ /* Latch volume update bits */
+ for (i = 0; i < ARRAY_SIZE(wm5102_digital_vu); i++)
+ regmap_update_bits(arizona->regmap, wm5102_digital_vu[i],
+ WM5102_DIG_VU, WM5102_DIG_VU);
+
+ pm_runtime_enable(&pdev->dev);
+ pm_runtime_idle(&pdev->dev);
+
+ return snd_soc_register_codec(&pdev->dev, &soc_codec_dev_wm5102,
+ wm5102_dai, ARRAY_SIZE(wm5102_dai));
+}
+
+static int __devexit wm5102_remove(struct platform_device *pdev)
+{
+ snd_soc_unregister_codec(&pdev->dev);
+ pm_runtime_disable(&pdev->dev);
+
+ return 0;
+}
+
+static struct platform_driver wm5102_codec_driver = {
+ .driver = {
+ .name = "wm5102-codec",
+ .owner = THIS_MODULE,
+ },
+ .probe = wm5102_probe,
+ .remove = __devexit_p(wm5102_remove),
+};
+
+module_platform_driver(wm5102_codec_driver);
+
+MODULE_DESCRIPTION("ASoC WM5102 driver");
+MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:wm5102-codec");
diff --git a/sound/soc/codecs/wm5102.h b/sound/soc/codecs/wm5102.h
new file mode 100644
index 0000000..d30477f
--- /dev/null
+++ b/sound/soc/codecs/wm5102.h
@@ -0,0 +1,21 @@
+/*
+ * wm5102.h -- WM5102 ALSA SoC Audio driver
+ *
+ * Copyright 2012 Wolfson Microelectronics plc
+ *
+ * Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
+ *
+ * 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 _WM5102_H
+#define _WM5102_H
+
+#include "arizona.h"
+
+#define WM5102_FLL1 1
+#define WM5102_FLL2 2
+
+#endif
diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c
new file mode 100644
index 0000000..8033f70
--- /dev/null
+++ b/sound/soc/codecs/wm5110.c
@@ -0,0 +1,950 @@
+/*
+ * wm5110.c -- WM5110 ALSA SoC Audio driver
+ *
+ * Copyright 2012 Wolfson Microelectronics plc
+ *
+ * Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
+ *
+ * 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 <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/pm_runtime.h>
+#include <linux/regmap.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/jack.h>
+#include <sound/initval.h>
+#include <sound/tlv.h>
+
+#include <linux/mfd/arizona/core.h>
+#include <linux/mfd/arizona/registers.h>
+
+#include "arizona.h"
+#include "wm5110.h"
+
+struct wm5110_priv {
+ struct arizona_priv core;
+ struct arizona_fll fll[2];
+};
+
+static DECLARE_TLV_DB_SCALE(ana_tlv, 0, 100, 0);
+static DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0);
+static DECLARE_TLV_DB_SCALE(digital_tlv, -6400, 50, 0);
+static DECLARE_TLV_DB_SCALE(noise_tlv, 0, 600, 0);
+
+static const struct snd_kcontrol_new wm5110_snd_controls[] = {
+SOC_SINGLE("IN1 High Performance Switch", ARIZONA_IN1L_CONTROL,
+ ARIZONA_IN1_OSR_SHIFT, 1, 0),
+SOC_SINGLE("IN2 High Performance Switch", ARIZONA_IN2L_CONTROL,
+ ARIZONA_IN2_OSR_SHIFT, 1, 0),
+SOC_SINGLE("IN3 High Performance Switch", ARIZONA_IN3L_CONTROL,
+ ARIZONA_IN3_OSR_SHIFT, 1, 0),
+SOC_SINGLE("IN4 High Performance Switch", ARIZONA_IN4L_CONTROL,
+ ARIZONA_IN4_OSR_SHIFT, 1, 0),
+
+SOC_DOUBLE_R_RANGE_TLV("IN1 Volume", ARIZONA_IN1L_CONTROL,
+ ARIZONA_IN1R_CONTROL,
+ ARIZONA_IN1L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv),
+SOC_DOUBLE_R_RANGE_TLV("IN2 Volume", ARIZONA_IN2L_CONTROL,
+ ARIZONA_IN2R_CONTROL,
+ ARIZONA_IN2L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv),
+SOC_DOUBLE_R_RANGE_TLV("IN3 Volume", ARIZONA_IN3L_CONTROL,
+ ARIZONA_IN3R_CONTROL,
+ ARIZONA_IN3L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv),
+
+SOC_DOUBLE_R("IN1 Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_1L,
+ ARIZONA_ADC_DIGITAL_VOLUME_1R, ARIZONA_IN1L_MUTE_SHIFT, 1, 1),
+SOC_DOUBLE_R("IN2 Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_2L,
+ ARIZONA_ADC_DIGITAL_VOLUME_2R, ARIZONA_IN2L_MUTE_SHIFT, 1, 1),
+SOC_DOUBLE_R("IN3 Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_3L,
+ ARIZONA_ADC_DIGITAL_VOLUME_3R, ARIZONA_IN3L_MUTE_SHIFT, 1, 1),
+SOC_DOUBLE_R("IN4 Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_4L,
+ ARIZONA_ADC_DIGITAL_VOLUME_4R, ARIZONA_IN4L_MUTE_SHIFT, 1, 1),
+
+SOC_DOUBLE_R_TLV("IN1 Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_1L,
+ ARIZONA_ADC_DIGITAL_VOLUME_1R, ARIZONA_IN1L_DIG_VOL_SHIFT,
+ 0xbf, 0, digital_tlv),
+SOC_DOUBLE_R_TLV("IN2 Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_2L,
+ ARIZONA_ADC_DIGITAL_VOLUME_2R, ARIZONA_IN2L_DIG_VOL_SHIFT,
+ 0xbf, 0, digital_tlv),
+SOC_DOUBLE_R_TLV("IN3 Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_3L,
+ ARIZONA_ADC_DIGITAL_VOLUME_3R, ARIZONA_IN3L_DIG_VOL_SHIFT,
+ 0xbf, 0, digital_tlv),
+SOC_DOUBLE_R_TLV("IN4 Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_4L,
+ ARIZONA_ADC_DIGITAL_VOLUME_4R, ARIZONA_IN4L_DIG_VOL_SHIFT,
+ 0xbf, 0, digital_tlv),
+
+ARIZONA_MIXER_CONTROLS("EQ1", ARIZONA_EQ1MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("EQ2", ARIZONA_EQ2MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("EQ3", ARIZONA_EQ3MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("EQ4", ARIZONA_EQ4MIX_INPUT_1_SOURCE),
+
+SOC_SINGLE_TLV("EQ1 B1 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B1_GAIN_SHIFT,
+ 24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ1 B2 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B2_GAIN_SHIFT,
+ 24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ1 B3 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B3_GAIN_SHIFT,
+ 24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ1 B4 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B4_GAIN_SHIFT,
+ 24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ1 B5 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B5_GAIN_SHIFT,
+ 24, 0, eq_tlv),
+
+SOC_SINGLE_TLV("EQ2 B1 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B1_GAIN_SHIFT,
+ 24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ2 B2 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B2_GAIN_SHIFT,
+ 24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ2 B3 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B3_GAIN_SHIFT,
+ 24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ2 B4 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B4_GAIN_SHIFT,
+ 24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ2 B5 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B5_GAIN_SHIFT,
+ 24, 0, eq_tlv),
+
+SOC_SINGLE_TLV("EQ3 B1 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B1_GAIN_SHIFT,
+ 24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ3 B2 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B2_GAIN_SHIFT,
+ 24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ3 B3 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B3_GAIN_SHIFT,
+ 24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ3 B4 Volume", ARIZONA_EQ3_2, ARIZONA_EQ3_B4_GAIN_SHIFT,
+ 24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ3 B5 Volume", ARIZONA_EQ3_2, ARIZONA_EQ3_B5_GAIN_SHIFT,
+ 24, 0, eq_tlv),
+
+SOC_SINGLE_TLV("EQ4 B1 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B1_GAIN_SHIFT,
+ 24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ4 B2 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B2_GAIN_SHIFT,
+ 24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ4 B3 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B3_GAIN_SHIFT,
+ 24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ4 B4 Volume", ARIZONA_EQ4_2, ARIZONA_EQ4_B4_GAIN_SHIFT,
+ 24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ4 B5 Volume", ARIZONA_EQ4_2, ARIZONA_EQ4_B5_GAIN_SHIFT,
+ 24, 0, eq_tlv),
+
+ARIZONA_MIXER_CONTROLS("DRC1L", ARIZONA_DRC1LMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("DRC1R", ARIZONA_DRC1RMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("DRC2L", ARIZONA_DRC2LMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("DRC2R", ARIZONA_DRC2RMIX_INPUT_1_SOURCE),
+
+SND_SOC_BYTES_MASK("DRC1", ARIZONA_DRC1_CTRL1, 5,
+ ARIZONA_DRC1R_ENA | ARIZONA_DRC1L_ENA),
+SND_SOC_BYTES_MASK("DRC2", ARIZONA_DRC2_CTRL1, 5,
+ ARIZONA_DRC2R_ENA | ARIZONA_DRC2L_ENA),
+
+ARIZONA_MIXER_CONTROLS("LHPF1", ARIZONA_HPLP1MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("LHPF2", ARIZONA_HPLP2MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("LHPF3", ARIZONA_HPLP3MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("LHPF4", ARIZONA_HPLP4MIX_INPUT_1_SOURCE),
+
+SOC_ENUM("LHPF1 Mode", arizona_lhpf1_mode),
+SOC_ENUM("LHPF2 Mode", arizona_lhpf2_mode),
+SOC_ENUM("LHPF3 Mode", arizona_lhpf3_mode),
+SOC_ENUM("LHPF4 Mode", arizona_lhpf4_mode),
+
+ARIZONA_MIXER_CONTROLS("Mic", ARIZONA_MICMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("Noise", ARIZONA_NOISEMIX_INPUT_1_SOURCE),
+
+SOC_SINGLE_TLV("Noise Generator Volume", ARIZONA_COMFORT_NOISE_GENERATOR,
+ ARIZONA_NOISE_GEN_GAIN_SHIFT, 0x16, 0, noise_tlv),
+
+ARIZONA_MIXER_CONTROLS("HPOUT1L", ARIZONA_OUT1LMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("HPOUT1R", ARIZONA_OUT1RMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("HPOUT2L", ARIZONA_OUT2LMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("HPOUT2R", ARIZONA_OUT2RMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("EPOUT", ARIZONA_OUT3LMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("SPKOUTL", ARIZONA_OUT4LMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("SPKOUTR", ARIZONA_OUT4RMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("SPKDAT1L", ARIZONA_OUT5LMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("SPKDAT1R", ARIZONA_OUT5RMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("SPKDAT2L", ARIZONA_OUT6LMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("SPKDAT2R", ARIZONA_OUT6RMIX_INPUT_1_SOURCE),
+
+SOC_SINGLE("HPOUT1 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_1L,
+ ARIZONA_OUT1_OSR_SHIFT, 1, 0),
+SOC_SINGLE("OUT2 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_2L,
+ ARIZONA_OUT2_OSR_SHIFT, 1, 0),
+SOC_SINGLE("EPOUT High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_3L,
+ ARIZONA_OUT3_OSR_SHIFT, 1, 0),
+SOC_SINGLE("Speaker High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_4L,
+ ARIZONA_OUT4_OSR_SHIFT, 1, 0),
+SOC_SINGLE("SPKDAT1 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_5L,
+ ARIZONA_OUT5_OSR_SHIFT, 1, 0),
+SOC_SINGLE("SPKDAT2 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_6L,
+ ARIZONA_OUT6_OSR_SHIFT, 1, 0),
+
+SOC_DOUBLE_R("HPOUT1 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_1L,
+ ARIZONA_DAC_DIGITAL_VOLUME_1R, ARIZONA_OUT1L_MUTE_SHIFT, 1, 1),
+SOC_DOUBLE_R("OUT2 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_2L,
+ ARIZONA_DAC_DIGITAL_VOLUME_2R, ARIZONA_OUT2L_MUTE_SHIFT, 1, 1),
+SOC_SINGLE("EPOUT Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_3L,
+ ARIZONA_OUT3L_MUTE_SHIFT, 1, 1),
+SOC_DOUBLE_R("Speaker Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_4L,
+ ARIZONA_DAC_DIGITAL_VOLUME_4R, ARIZONA_OUT4L_MUTE_SHIFT, 1, 1),
+SOC_DOUBLE_R("SPKDAT1 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_5L,
+ ARIZONA_DAC_DIGITAL_VOLUME_5R, ARIZONA_OUT5L_MUTE_SHIFT, 1, 1),
+SOC_DOUBLE_R("SPKDAT2 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_6L,
+ ARIZONA_DAC_DIGITAL_VOLUME_6R, ARIZONA_OUT6L_MUTE_SHIFT, 1, 1),
+
+SOC_DOUBLE_R_TLV("HPOUT1 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_1L,
+ ARIZONA_DAC_DIGITAL_VOLUME_1R, ARIZONA_OUT1L_VOL_SHIFT,
+ 0xbf, 0, digital_tlv),
+SOC_DOUBLE_R_TLV("OUT2 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_2L,
+ ARIZONA_DAC_DIGITAL_VOLUME_2R, ARIZONA_OUT2L_VOL_SHIFT,
+ 0xbf, 0, digital_tlv),
+SOC_SINGLE_TLV("EPOUT Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_3L,
+ ARIZONA_OUT3L_VOL_SHIFT, 0xbf, 0, digital_tlv),
+SOC_DOUBLE_R_TLV("Speaker Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_4L,
+ ARIZONA_DAC_DIGITAL_VOLUME_4R, ARIZONA_OUT4L_VOL_SHIFT,
+ 0xbf, 0, digital_tlv),
+SOC_DOUBLE_R_TLV("SPKDAT1 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_5L,
+ ARIZONA_DAC_DIGITAL_VOLUME_5R, ARIZONA_OUT5L_VOL_SHIFT,
+ 0xbf, 0, digital_tlv),
+SOC_DOUBLE_R_TLV("SPKDAT2 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_6L,
+ ARIZONA_DAC_DIGITAL_VOLUME_6R, ARIZONA_OUT6L_VOL_SHIFT,
+ 0xbf, 0, digital_tlv),
+
+SOC_DOUBLE_R_RANGE_TLV("HPOUT1 Volume", ARIZONA_OUTPUT_PATH_CONFIG_1L,
+ ARIZONA_OUTPUT_PATH_CONFIG_1R,
+ ARIZONA_OUT1L_PGA_VOL_SHIFT,
+ 0x34, 0x40, 0, ana_tlv),
+SOC_DOUBLE_R_RANGE_TLV("OUT2 Volume", ARIZONA_OUTPUT_PATH_CONFIG_2L,
+ ARIZONA_OUTPUT_PATH_CONFIG_2R,
+ ARIZONA_OUT2L_PGA_VOL_SHIFT,
+ 0x34, 0x40, 0, ana_tlv),
+SOC_SINGLE_RANGE_TLV("EPOUT Volume", ARIZONA_OUTPUT_PATH_CONFIG_3L,
+ ARIZONA_OUT3L_PGA_VOL_SHIFT, 0x34, 0x40, 0, ana_tlv),
+
+SOC_DOUBLE("SPKDAT1 Switch", ARIZONA_PDM_SPK1_CTRL_1, ARIZONA_SPK1L_MUTE_SHIFT,
+ ARIZONA_SPK1R_MUTE_SHIFT, 1, 1),
+SOC_DOUBLE("SPKDAT2 Switch", ARIZONA_PDM_SPK2_CTRL_1, ARIZONA_SPK2L_MUTE_SHIFT,
+ ARIZONA_SPK2R_MUTE_SHIFT, 1, 1),
+
+ARIZONA_MIXER_CONTROLS("AIF1TX1", ARIZONA_AIF1TX1MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("AIF1TX2", ARIZONA_AIF1TX2MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("AIF1TX3", ARIZONA_AIF1TX3MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("AIF1TX4", ARIZONA_AIF1TX4MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("AIF1TX5", ARIZONA_AIF1TX5MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("AIF1TX6", ARIZONA_AIF1TX6MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("AIF1TX7", ARIZONA_AIF1TX7MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("AIF1TX8", ARIZONA_AIF1TX8MIX_INPUT_1_SOURCE),
+
+ARIZONA_MIXER_CONTROLS("AIF2TX1", ARIZONA_AIF2TX1MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("AIF2TX2", ARIZONA_AIF2TX2MIX_INPUT_1_SOURCE),
+
+ARIZONA_MIXER_CONTROLS("AIF3TX1", ARIZONA_AIF3TX1MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("AIF3TX2", ARIZONA_AIF3TX2MIX_INPUT_1_SOURCE),
+};
+
+ARIZONA_MIXER_ENUMS(EQ1, ARIZONA_EQ1MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(EQ2, ARIZONA_EQ2MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(EQ3, ARIZONA_EQ3MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(EQ4, ARIZONA_EQ4MIX_INPUT_1_SOURCE);
+
+ARIZONA_MIXER_ENUMS(DRC1L, ARIZONA_DRC1LMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(DRC1R, ARIZONA_DRC1RMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(DRC2L, ARIZONA_DRC2LMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(DRC2R, ARIZONA_DRC2RMIX_INPUT_1_SOURCE);
+
+ARIZONA_MIXER_ENUMS(LHPF1, ARIZONA_HPLP1MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(LHPF2, ARIZONA_HPLP2MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(LHPF3, ARIZONA_HPLP3MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(LHPF4, ARIZONA_HPLP4MIX_INPUT_1_SOURCE);
+
+ARIZONA_MIXER_ENUMS(Mic, ARIZONA_MICMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(Noise, ARIZONA_NOISEMIX_INPUT_1_SOURCE);
+
+ARIZONA_MIXER_ENUMS(PWM1, ARIZONA_PWM1MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(PWM2, ARIZONA_PWM2MIX_INPUT_1_SOURCE);
+
+ARIZONA_MIXER_ENUMS(OUT1L, ARIZONA_OUT1LMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(OUT1R, ARIZONA_OUT1RMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(OUT2L, ARIZONA_OUT2LMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(OUT2R, ARIZONA_OUT2RMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(OUT3, ARIZONA_OUT3LMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(SPKOUTL, ARIZONA_OUT4LMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(SPKOUTR, ARIZONA_OUT4RMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(SPKDAT1L, ARIZONA_OUT5LMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(SPKDAT1R, ARIZONA_OUT5RMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(SPKDAT2L, ARIZONA_OUT6LMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(SPKDAT2R, ARIZONA_OUT6RMIX_INPUT_1_SOURCE);
+
+ARIZONA_MIXER_ENUMS(AIF1TX1, ARIZONA_AIF1TX1MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(AIF1TX2, ARIZONA_AIF1TX2MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(AIF1TX3, ARIZONA_AIF1TX3MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(AIF1TX4, ARIZONA_AIF1TX4MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(AIF1TX5, ARIZONA_AIF1TX5MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(AIF1TX6, ARIZONA_AIF1TX6MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(AIF1TX7, ARIZONA_AIF1TX7MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(AIF1TX8, ARIZONA_AIF1TX8MIX_INPUT_1_SOURCE);
+
+ARIZONA_MIXER_ENUMS(AIF2TX1, ARIZONA_AIF2TX1MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(AIF2TX2, ARIZONA_AIF2TX2MIX_INPUT_1_SOURCE);
+
+ARIZONA_MIXER_ENUMS(AIF3TX1, ARIZONA_AIF3TX1MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(AIF3TX2, ARIZONA_AIF3TX2MIX_INPUT_1_SOURCE);
+
+ARIZONA_MIXER_ENUMS(ASRC1L, ARIZONA_ASRC1LMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(ASRC1R, ARIZONA_ASRC1RMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(ASRC2L, ARIZONA_ASRC2LMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(ASRC2R, ARIZONA_ASRC2RMIX_INPUT_1_SOURCE);
+
+static const struct snd_soc_dapm_widget wm5110_dapm_widgets[] = {
+SND_SOC_DAPM_SUPPLY("SYSCLK", ARIZONA_SYSTEM_CLOCK_1, ARIZONA_SYSCLK_ENA_SHIFT,
+ 0, NULL, 0),
+SND_SOC_DAPM_SUPPLY("ASYNCCLK", ARIZONA_ASYNC_CLOCK_1,
+ ARIZONA_ASYNC_CLK_ENA_SHIFT, 0, NULL, 0),
+
+SND_SOC_DAPM_REGULATOR_SUPPLY("DBVDD2", 0),
+SND_SOC_DAPM_REGULATOR_SUPPLY("DBVDD3", 0),
+SND_SOC_DAPM_REGULATOR_SUPPLY("CPVDD", 20),
+SND_SOC_DAPM_REGULATOR_SUPPLY("MICVDD", 0),
+SND_SOC_DAPM_REGULATOR_SUPPLY("SPKVDDL", 0),
+SND_SOC_DAPM_REGULATOR_SUPPLY("SPKVDDR", 0),
+
+SND_SOC_DAPM_SIGGEN("TONE"),
+SND_SOC_DAPM_SIGGEN("NOISE"),
+
+SND_SOC_DAPM_INPUT("IN1L"),
+SND_SOC_DAPM_INPUT("IN1R"),
+SND_SOC_DAPM_INPUT("IN2L"),
+SND_SOC_DAPM_INPUT("IN2R"),
+SND_SOC_DAPM_INPUT("IN3L"),
+SND_SOC_DAPM_INPUT("IN3R"),
+SND_SOC_DAPM_INPUT("IN4L"),
+SND_SOC_DAPM_INPUT("IN4R"),
+
+SND_SOC_DAPM_PGA_E("IN1L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN1L_ENA_SHIFT,
+ 0, NULL, 0, arizona_in_ev,
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("IN1R PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN1R_ENA_SHIFT,
+ 0, NULL, 0, arizona_in_ev,
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("IN2L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN2L_ENA_SHIFT,
+ 0, NULL, 0, arizona_in_ev,
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("IN2R PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN2R_ENA_SHIFT,
+ 0, NULL, 0, arizona_in_ev,
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("IN3L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN3L_ENA_SHIFT,
+ 0, NULL, 0, arizona_in_ev,
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("IN3R PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN3R_ENA_SHIFT,
+ 0, NULL, 0, arizona_in_ev,
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("IN4L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN4L_ENA_SHIFT,
+ 0, NULL, 0, arizona_in_ev,
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("IN4R PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN4R_ENA_SHIFT,
+ 0, NULL, 0, arizona_in_ev,
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+
+SND_SOC_DAPM_SUPPLY("MICBIAS1", ARIZONA_MIC_BIAS_CTRL_1,
+ ARIZONA_MICB1_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_SUPPLY("MICBIAS2", ARIZONA_MIC_BIAS_CTRL_2,
+ ARIZONA_MICB1_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_SUPPLY("MICBIAS3", ARIZONA_MIC_BIAS_CTRL_3,
+ ARIZONA_MICB1_ENA_SHIFT, 0, NULL, 0),
+
+SND_SOC_DAPM_PGA("Noise Generator", ARIZONA_COMFORT_NOISE_GENERATOR,
+ ARIZONA_NOISE_GEN_ENA_SHIFT, 0, NULL, 0),
+
+SND_SOC_DAPM_PGA("Tone Generator 1", ARIZONA_TONE_GENERATOR_1,
+ ARIZONA_TONE1_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA("Tone Generator 2", ARIZONA_TONE_GENERATOR_1,
+ ARIZONA_TONE2_ENA_SHIFT, 0, NULL, 0),
+
+SND_SOC_DAPM_PGA("Mic Mute Mixer", ARIZONA_MIC_NOISE_MIX_CONTROL_1,
+ ARIZONA_MICMUTE_MIX_ENA_SHIFT, 0, NULL, 0),
+
+SND_SOC_DAPM_PGA("EQ1", ARIZONA_EQ1_1, ARIZONA_EQ1_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA("EQ2", ARIZONA_EQ2_1, ARIZONA_EQ2_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA("EQ3", ARIZONA_EQ3_1, ARIZONA_EQ3_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA("EQ4", ARIZONA_EQ4_1, ARIZONA_EQ4_ENA_SHIFT, 0, NULL, 0),
+
+SND_SOC_DAPM_PGA("DRC1L", ARIZONA_DRC1_CTRL1, ARIZONA_DRC1L_ENA_SHIFT, 0,
+ NULL, 0),
+SND_SOC_DAPM_PGA("DRC1R", ARIZONA_DRC1_CTRL1, ARIZONA_DRC1R_ENA_SHIFT, 0,
+ NULL, 0),
+SND_SOC_DAPM_PGA("DRC2L", ARIZONA_DRC2_CTRL1, ARIZONA_DRC2L_ENA_SHIFT, 0,
+ NULL, 0),
+SND_SOC_DAPM_PGA("DRC2R", ARIZONA_DRC2_CTRL1, ARIZONA_DRC2R_ENA_SHIFT, 0,
+ NULL, 0),
+
+SND_SOC_DAPM_PGA("LHPF1", ARIZONA_HPLPF1_1, ARIZONA_LHPF1_ENA_SHIFT, 0,
+ NULL, 0),
+SND_SOC_DAPM_PGA("LHPF2", ARIZONA_HPLPF2_1, ARIZONA_LHPF2_ENA_SHIFT, 0,
+ NULL, 0),
+SND_SOC_DAPM_PGA("LHPF3", ARIZONA_HPLPF3_1, ARIZONA_LHPF3_ENA_SHIFT, 0,
+ NULL, 0),
+SND_SOC_DAPM_PGA("LHPF4", ARIZONA_HPLPF4_1, ARIZONA_LHPF4_ENA_SHIFT, 0,
+ NULL, 0),
+
+SND_SOC_DAPM_PGA("PWM1 Driver", ARIZONA_PWM_DRIVE_1, ARIZONA_PWM1_ENA_SHIFT,
+ 0, NULL, 0),
+SND_SOC_DAPM_PGA("PWM2 Driver", ARIZONA_PWM_DRIVE_1, ARIZONA_PWM2_ENA_SHIFT,
+ 0, NULL, 0),
+
+SND_SOC_DAPM_PGA("ASRC1L", ARIZONA_ASRC_ENABLE, ARIZONA_ASRC1L_ENA_SHIFT, 0,
+ NULL, 0),
+SND_SOC_DAPM_PGA("ASRC1R", ARIZONA_ASRC_ENABLE, ARIZONA_ASRC1R_ENA_SHIFT, 0,
+ NULL, 0),
+SND_SOC_DAPM_PGA("ASRC2L", ARIZONA_ASRC_ENABLE, ARIZONA_ASRC2L_ENA_SHIFT, 0,
+ NULL, 0),
+SND_SOC_DAPM_PGA("ASRC2R", ARIZONA_ASRC_ENABLE, ARIZONA_ASRC2R_ENA_SHIFT, 0,
+ NULL, 0),
+
+SND_SOC_DAPM_AIF_OUT("AIF1TX1", NULL, 0,
+ ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX1_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF1TX2", NULL, 0,
+ ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX2_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF1TX3", NULL, 0,
+ ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX3_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF1TX4", NULL, 0,
+ ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX4_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF1TX5", NULL, 0,
+ ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX5_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF1TX6", NULL, 0,
+ ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX6_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF1TX7", NULL, 0,
+ ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX7_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF1TX8", NULL, 0,
+ ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX8_ENA_SHIFT, 0),
+
+SND_SOC_DAPM_AIF_IN("AIF1RX1", NULL, 0,
+ ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX1_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF1RX2", NULL, 0,
+ ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX2_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF1RX3", NULL, 0,
+ ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX3_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF1RX4", NULL, 0,
+ ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX4_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF1RX5", NULL, 0,
+ ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX5_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF1RX6", NULL, 0,
+ ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX6_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF1RX7", NULL, 0,
+ ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX7_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF1RX8", NULL, 0,
+ ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX8_ENA_SHIFT, 0),
+
+SND_SOC_DAPM_AIF_OUT("AIF2TX1", NULL, 0,
+ ARIZONA_AIF2_TX_ENABLES, ARIZONA_AIF2TX1_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF2TX2", NULL, 0,
+ ARIZONA_AIF2_TX_ENABLES, ARIZONA_AIF2TX2_ENA_SHIFT, 0),
+
+SND_SOC_DAPM_AIF_IN("AIF2RX1", NULL, 0,
+ ARIZONA_AIF2_RX_ENABLES, ARIZONA_AIF2RX1_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF2RX2", NULL, 0,
+ ARIZONA_AIF2_RX_ENABLES, ARIZONA_AIF2RX2_ENA_SHIFT, 0),
+
+SND_SOC_DAPM_AIF_OUT("AIF3TX1", NULL, 0,
+ ARIZONA_AIF3_TX_ENABLES, ARIZONA_AIF3TX1_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF3TX2", NULL, 0,
+ ARIZONA_AIF3_TX_ENABLES, ARIZONA_AIF3TX2_ENA_SHIFT, 0),
+
+SND_SOC_DAPM_AIF_IN("AIF3RX1", NULL, 0,
+ ARIZONA_AIF3_RX_ENABLES, ARIZONA_AIF3RX1_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF3RX2", NULL, 0,
+ ARIZONA_AIF3_RX_ENABLES, ARIZONA_AIF3RX2_ENA_SHIFT, 0),
+
+SND_SOC_DAPM_PGA_E("OUT1L", ARIZONA_OUTPUT_ENABLES_1,
+ ARIZONA_OUT1L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("OUT1R", ARIZONA_OUTPUT_ENABLES_1,
+ ARIZONA_OUT1R_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("OUT2L", ARIZONA_OUTPUT_ENABLES_1,
+ ARIZONA_OUT2L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("OUT2R", ARIZONA_OUTPUT_ENABLES_1,
+ ARIZONA_OUT2R_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("OUT3L", ARIZONA_OUTPUT_ENABLES_1,
+ ARIZONA_OUT3L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("OUT4L", ARIZONA_OUTPUT_ENABLES_1,
+ ARIZONA_OUT4L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("OUT4R", ARIZONA_OUTPUT_ENABLES_1,
+ ARIZONA_OUT4R_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("OUT5L", ARIZONA_OUTPUT_ENABLES_1,
+ ARIZONA_OUT5L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("OUT5R", ARIZONA_OUTPUT_ENABLES_1,
+ ARIZONA_OUT5R_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("OUT6L", ARIZONA_OUTPUT_ENABLES_1,
+ ARIZONA_OUT6L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("OUT6R", ARIZONA_OUTPUT_ENABLES_1,
+ ARIZONA_OUT6R_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+
+ARIZONA_MIXER_WIDGETS(EQ1, "EQ1"),
+ARIZONA_MIXER_WIDGETS(EQ2, "EQ2"),
+ARIZONA_MIXER_WIDGETS(EQ3, "EQ3"),
+ARIZONA_MIXER_WIDGETS(EQ4, "EQ4"),
+
+ARIZONA_MIXER_WIDGETS(DRC1L, "DRC1L"),
+ARIZONA_MIXER_WIDGETS(DRC1R, "DRC1R"),
+ARIZONA_MIXER_WIDGETS(DRC2L, "DRC2L"),
+ARIZONA_MIXER_WIDGETS(DRC2R, "DRC2R"),
+
+ARIZONA_MIXER_WIDGETS(LHPF1, "LHPF1"),
+ARIZONA_MIXER_WIDGETS(LHPF2, "LHPF2"),
+ARIZONA_MIXER_WIDGETS(LHPF3, "LHPF3"),
+ARIZONA_MIXER_WIDGETS(LHPF4, "LHPF4"),
+
+ARIZONA_MIXER_WIDGETS(Mic, "Mic"),
+ARIZONA_MIXER_WIDGETS(Noise, "Noise"),
+
+ARIZONA_MIXER_WIDGETS(PWM1, "PWM1"),
+ARIZONA_MIXER_WIDGETS(PWM2, "PWM2"),
+
+ARIZONA_MIXER_WIDGETS(OUT1L, "HPOUT1L"),
+ARIZONA_MIXER_WIDGETS(OUT1R, "HPOUT1R"),
+ARIZONA_MIXER_WIDGETS(OUT2L, "HPOUT2L"),
+ARIZONA_MIXER_WIDGETS(OUT2R, "HPOUT2R"),
+ARIZONA_MIXER_WIDGETS(OUT3, "EPOUT"),
+ARIZONA_MIXER_WIDGETS(SPKOUTL, "SPKOUTL"),
+ARIZONA_MIXER_WIDGETS(SPKOUTR, "SPKOUTR"),
+ARIZONA_MIXER_WIDGETS(SPKDAT1L, "SPKDAT1L"),
+ARIZONA_MIXER_WIDGETS(SPKDAT1R, "SPKDAT1R"),
+ARIZONA_MIXER_WIDGETS(SPKDAT2L, "SPKDAT2L"),
+ARIZONA_MIXER_WIDGETS(SPKDAT2R, "SPKDAT2R"),
+
+ARIZONA_MIXER_WIDGETS(AIF1TX1, "AIF1TX1"),
+ARIZONA_MIXER_WIDGETS(AIF1TX2, "AIF1TX2"),
+ARIZONA_MIXER_WIDGETS(AIF1TX3, "AIF1TX3"),
+ARIZONA_MIXER_WIDGETS(AIF1TX4, "AIF1TX4"),
+ARIZONA_MIXER_WIDGETS(AIF1TX5, "AIF1TX5"),
+ARIZONA_MIXER_WIDGETS(AIF1TX6, "AIF1TX6"),
+ARIZONA_MIXER_WIDGETS(AIF1TX7, "AIF1TX7"),
+ARIZONA_MIXER_WIDGETS(AIF1TX8, "AIF1TX8"),
+
+ARIZONA_MIXER_WIDGETS(AIF2TX1, "AIF2TX1"),
+ARIZONA_MIXER_WIDGETS(AIF2TX2, "AIF2TX2"),
+
+ARIZONA_MIXER_WIDGETS(AIF3TX1, "AIF3TX1"),
+ARIZONA_MIXER_WIDGETS(AIF3TX2, "AIF3TX2"),
+
+ARIZONA_MIXER_WIDGETS(ASRC1L, "ASRC1L"),
+ARIZONA_MIXER_WIDGETS(ASRC1R, "ASRC1R"),
+ARIZONA_MIXER_WIDGETS(ASRC2L, "ASRC2L"),
+ARIZONA_MIXER_WIDGETS(ASRC2R, "ASRC2R"),
+
+SND_SOC_DAPM_OUTPUT("HPOUT1L"),
+SND_SOC_DAPM_OUTPUT("HPOUT1R"),
+SND_SOC_DAPM_OUTPUT("HPOUT2L"),
+SND_SOC_DAPM_OUTPUT("HPOUT2R"),
+SND_SOC_DAPM_OUTPUT("EPOUTN"),
+SND_SOC_DAPM_OUTPUT("EPOUTP"),
+SND_SOC_DAPM_OUTPUT("SPKOUTLN"),
+SND_SOC_DAPM_OUTPUT("SPKOUTLP"),
+SND_SOC_DAPM_OUTPUT("SPKOUTRN"),
+SND_SOC_DAPM_OUTPUT("SPKOUTRP"),
+SND_SOC_DAPM_OUTPUT("SPKDAT1L"),
+SND_SOC_DAPM_OUTPUT("SPKDAT1R"),
+SND_SOC_DAPM_OUTPUT("SPKDAT2L"),
+SND_SOC_DAPM_OUTPUT("SPKDAT2R"),
+};
+
+#define ARIZONA_MIXER_INPUT_ROUTES(name) \
+ { name, "Noise Generator", "Noise Generator" }, \
+ { name, "Tone Generator 1", "Tone Generator 1" }, \
+ { name, "Tone Generator 2", "Tone Generator 2" }, \
+ { name, "IN1L", "IN1L PGA" }, \
+ { name, "IN1R", "IN1R PGA" }, \
+ { name, "IN2L", "IN2L PGA" }, \
+ { name, "IN2R", "IN2R PGA" }, \
+ { name, "IN3L", "IN3L PGA" }, \
+ { name, "IN3R", "IN3R PGA" }, \
+ { name, "IN4L", "IN4L PGA" }, \
+ { name, "IN4R", "IN4R PGA" }, \
+ { name, "Mic Mute Mixer", "Mic Mute Mixer" }, \
+ { name, "AIF1RX1", "AIF1RX1" }, \
+ { name, "AIF1RX2", "AIF1RX2" }, \
+ { name, "AIF1RX3", "AIF1RX3" }, \
+ { name, "AIF1RX4", "AIF1RX4" }, \
+ { name, "AIF1RX5", "AIF1RX5" }, \
+ { name, "AIF1RX6", "AIF1RX6" }, \
+ { name, "AIF1RX7", "AIF1RX7" }, \
+ { name, "AIF1RX8", "AIF1RX8" }, \
+ { name, "AIF2RX1", "AIF2RX1" }, \
+ { name, "AIF2RX2", "AIF2RX2" }, \
+ { name, "AIF3RX1", "AIF3RX1" }, \
+ { name, "AIF3RX2", "AIF3RX2" }, \
+ { name, "EQ1", "EQ1" }, \
+ { name, "EQ2", "EQ2" }, \
+ { name, "EQ3", "EQ3" }, \
+ { name, "EQ4", "EQ4" }, \
+ { name, "DRC1L", "DRC1L" }, \
+ { name, "DRC1R", "DRC1R" }, \
+ { name, "DRC2L", "DRC2L" }, \
+ { name, "DRC2R", "DRC2R" }, \
+ { name, "LHPF1", "LHPF1" }, \
+ { name, "LHPF2", "LHPF2" }, \
+ { name, "LHPF3", "LHPF3" }, \
+ { name, "LHPF4", "LHPF4" }, \
+ { name, "ASRC1L", "ASRC1L" }, \
+ { name, "ASRC1R", "ASRC1R" }, \
+ { name, "ASRC2L", "ASRC2L" }, \
+ { name, "ASRC2R", "ASRC2R" }
+
+static const struct snd_soc_dapm_route wm5110_dapm_routes[] = {
+ { "AIF2 Capture", NULL, "DBVDD2" },
+ { "AIF2 Playback", NULL, "DBVDD2" },
+
+ { "AIF3 Capture", NULL, "DBVDD3" },
+ { "AIF3 Playback", NULL, "DBVDD3" },
+
+ { "OUT1L", NULL, "CPVDD" },
+ { "OUT1R", NULL, "CPVDD" },
+ { "OUT2L", NULL, "CPVDD" },
+ { "OUT2R", NULL, "CPVDD" },
+ { "OUT3L", NULL, "CPVDD" },
+
+ { "OUT4L", NULL, "SPKVDDL" },
+ { "OUT4R", NULL, "SPKVDDR" },
+
+ { "OUT1L", NULL, "SYSCLK" },
+ { "OUT1R", NULL, "SYSCLK" },
+ { "OUT2L", NULL, "SYSCLK" },
+ { "OUT2R", NULL, "SYSCLK" },
+ { "OUT3L", NULL, "SYSCLK" },
+ { "OUT4L", NULL, "SYSCLK" },
+ { "OUT4R", NULL, "SYSCLK" },
+ { "OUT5L", NULL, "SYSCLK" },
+ { "OUT5R", NULL, "SYSCLK" },
+ { "OUT6L", NULL, "SYSCLK" },
+ { "OUT6R", NULL, "SYSCLK" },
+
+ { "MICBIAS1", NULL, "MICVDD" },
+ { "MICBIAS2", NULL, "MICVDD" },
+ { "MICBIAS3", NULL, "MICVDD" },
+
+ { "Noise Generator", NULL, "NOISE" },
+ { "Tone Generator 1", NULL, "TONE" },
+ { "Tone Generator 2", NULL, "TONE" },
+
+ { "Mic Mute Mixer", NULL, "Noise Mixer" },
+ { "Mic Mute Mixer", NULL, "Mic Mixer" },
+
+ { "AIF1 Capture", NULL, "AIF1TX1" },
+ { "AIF1 Capture", NULL, "AIF1TX2" },
+ { "AIF1 Capture", NULL, "AIF1TX3" },
+ { "AIF1 Capture", NULL, "AIF1TX4" },
+ { "AIF1 Capture", NULL, "AIF1TX5" },
+ { "AIF1 Capture", NULL, "AIF1TX6" },
+ { "AIF1 Capture", NULL, "AIF1TX7" },
+ { "AIF1 Capture", NULL, "AIF1TX8" },
+
+ { "AIF1RX1", NULL, "AIF1 Playback" },
+ { "AIF1RX2", NULL, "AIF1 Playback" },
+ { "AIF1RX3", NULL, "AIF1 Playback" },
+ { "AIF1RX4", NULL, "AIF1 Playback" },
+ { "AIF1RX5", NULL, "AIF1 Playback" },
+ { "AIF1RX6", NULL, "AIF1 Playback" },
+ { "AIF1RX7", NULL, "AIF1 Playback" },
+ { "AIF1RX8", NULL, "AIF1 Playback" },
+
+ { "AIF2 Capture", NULL, "AIF2TX1" },
+ { "AIF2 Capture", NULL, "AIF2TX2" },
+
+ { "AIF2RX1", NULL, "AIF2 Playback" },
+ { "AIF2RX2", NULL, "AIF2 Playback" },
+
+ { "AIF3 Capture", NULL, "AIF3TX1" },
+ { "AIF3 Capture", NULL, "AIF3TX2" },
+
+ { "AIF3RX1", NULL, "AIF3 Playback" },
+ { "AIF3RX2", NULL, "AIF3 Playback" },
+
+ { "AIF1 Playback", NULL, "SYSCLK" },
+ { "AIF2 Playback", NULL, "SYSCLK" },
+ { "AIF3 Playback", NULL, "SYSCLK" },
+
+ { "AIF1 Capture", NULL, "SYSCLK" },
+ { "AIF2 Capture", NULL, "SYSCLK" },
+ { "AIF3 Capture", NULL, "SYSCLK" },
+
+ ARIZONA_MIXER_ROUTES("OUT1L", "HPOUT1L"),
+ ARIZONA_MIXER_ROUTES("OUT1R", "HPOUT1R"),
+ ARIZONA_MIXER_ROUTES("OUT2L", "HPOUT2L"),
+ ARIZONA_MIXER_ROUTES("OUT2R", "HPOUT2R"),
+ ARIZONA_MIXER_ROUTES("OUT3L", "EPOUT"),
+
+ ARIZONA_MIXER_ROUTES("OUT4L", "SPKOUTL"),
+ ARIZONA_MIXER_ROUTES("OUT4R", "SPKOUTR"),
+ ARIZONA_MIXER_ROUTES("OUT5L", "SPKDAT1L"),
+ ARIZONA_MIXER_ROUTES("OUT5R", "SPKDAT1R"),
+ ARIZONA_MIXER_ROUTES("OUT6L", "SPKDAT2L"),
+ ARIZONA_MIXER_ROUTES("OUT6R", "SPKDAT2R"),
+
+ ARIZONA_MIXER_ROUTES("PWM1 Driver", "PWM1"),
+ ARIZONA_MIXER_ROUTES("PWM2 Driver", "PWM2"),
+
+ ARIZONA_MIXER_ROUTES("AIF1TX1", "AIF1TX1"),
+ ARIZONA_MIXER_ROUTES("AIF1TX2", "AIF1TX2"),
+ ARIZONA_MIXER_ROUTES("AIF1TX3", "AIF1TX3"),
+ ARIZONA_MIXER_ROUTES("AIF1TX4", "AIF1TX4"),
+ ARIZONA_MIXER_ROUTES("AIF1TX5", "AIF1TX5"),
+ ARIZONA_MIXER_ROUTES("AIF1TX6", "AIF1TX6"),
+ ARIZONA_MIXER_ROUTES("AIF1TX7", "AIF1TX7"),
+ ARIZONA_MIXER_ROUTES("AIF1TX8", "AIF1TX8"),
+
+ ARIZONA_MIXER_ROUTES("AIF2TX1", "AIF2TX1"),
+ ARIZONA_MIXER_ROUTES("AIF2TX2", "AIF2TX2"),
+
+ ARIZONA_MIXER_ROUTES("AIF3TX1", "AIF3TX1"),
+ ARIZONA_MIXER_ROUTES("AIF3TX2", "AIF3TX2"),
+
+ ARIZONA_MIXER_ROUTES("EQ1", "EQ1"),
+ ARIZONA_MIXER_ROUTES("EQ2", "EQ2"),
+ ARIZONA_MIXER_ROUTES("EQ3", "EQ3"),
+ ARIZONA_MIXER_ROUTES("EQ4", "EQ4"),
+
+ ARIZONA_MIXER_ROUTES("DRC1L", "DRC1L"),
+ ARIZONA_MIXER_ROUTES("DRC1R", "DRC1R"),
+ ARIZONA_MIXER_ROUTES("DRC2L", "DRC2L"),
+ ARIZONA_MIXER_ROUTES("DRC2R", "DRC2R"),
+
+ ARIZONA_MIXER_ROUTES("LHPF1", "LHPF1"),
+ ARIZONA_MIXER_ROUTES("LHPF2", "LHPF2"),
+ ARIZONA_MIXER_ROUTES("LHPF3", "LHPF3"),
+ ARIZONA_MIXER_ROUTES("LHPF4", "LHPF4"),
+
+ ARIZONA_MIXER_ROUTES("ASRC1L", "ASRC1L"),
+ ARIZONA_MIXER_ROUTES("ASRC1R", "ASRC1R"),
+ ARIZONA_MIXER_ROUTES("ASRC2L", "ASRC2L"),
+ ARIZONA_MIXER_ROUTES("ASRC2R", "ASRC2R"),
+
+ { "HPOUT1L", NULL, "OUT1L" },
+ { "HPOUT1R", NULL, "OUT1R" },
+
+ { "HPOUT2L", NULL, "OUT2L" },
+ { "HPOUT2R", NULL, "OUT2R" },
+
+ { "EPOUTN", NULL, "OUT3L" },
+ { "EPOUTP", NULL, "OUT3L" },
+
+ { "SPKOUTLN", NULL, "OUT4L" },
+ { "SPKOUTLP", NULL, "OUT4L" },
+
+ { "SPKOUTRN", NULL, "OUT4R" },
+ { "SPKOUTRP", NULL, "OUT4R" },
+
+ { "SPKDAT1L", NULL, "OUT5L" },
+ { "SPKDAT1R", NULL, "OUT5R" },
+
+ { "SPKDAT2L", NULL, "OUT6L" },
+ { "SPKDAT2R", NULL, "OUT6R" },
+};
+
+static int wm5110_set_fll(struct snd_soc_codec *codec, int fll_id, int source,
+ unsigned int Fref, unsigned int Fout)
+{
+ struct wm5110_priv *wm5110 = snd_soc_codec_get_drvdata(codec);
+
+ switch (fll_id) {
+ case WM5110_FLL1:
+ return arizona_set_fll(&wm5110->fll[0], source, Fref, Fout);
+ case WM5110_FLL2:
+ return arizona_set_fll(&wm5110->fll[1], source, Fref, Fout);
+ default:
+ return -EINVAL;
+ }
+}
+
+#define WM5110_RATES SNDRV_PCM_RATE_8000_192000
+
+#define WM5110_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
+ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
+
+static struct snd_soc_dai_driver wm5110_dai[] = {
+ {
+ .name = "wm5110-aif1",
+ .id = 1,
+ .base = ARIZONA_AIF1_BCLK_CTRL,
+ .playback = {
+ .stream_name = "AIF1 Playback",
+ .channels_min = 1,
+ .channels_max = 8,
+ .rates = WM5110_RATES,
+ .formats = WM5110_FORMATS,
+ },
+ .capture = {
+ .stream_name = "AIF1 Capture",
+ .channels_min = 1,
+ .channels_max = 8,
+ .rates = WM5110_RATES,
+ .formats = WM5110_FORMATS,
+ },
+ .ops = &arizona_dai_ops,
+ .symmetric_rates = 1,
+ },
+ {
+ .name = "wm5110-aif2",
+ .id = 2,
+ .base = ARIZONA_AIF2_BCLK_CTRL,
+ .playback = {
+ .stream_name = "AIF2 Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = WM5110_RATES,
+ .formats = WM5110_FORMATS,
+ },
+ .capture = {
+ .stream_name = "AIF2 Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = WM5110_RATES,
+ .formats = WM5110_FORMATS,
+ },
+ .ops = &arizona_dai_ops,
+ .symmetric_rates = 1,
+ },
+ {
+ .name = "wm5110-aif3",
+ .id = 3,
+ .base = ARIZONA_AIF3_BCLK_CTRL,
+ .playback = {
+ .stream_name = "AIF3 Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = WM5110_RATES,
+ .formats = WM5110_FORMATS,
+ },
+ .capture = {
+ .stream_name = "AIF3 Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = WM5110_RATES,
+ .formats = WM5110_FORMATS,
+ },
+ .ops = &arizona_dai_ops,
+ .symmetric_rates = 1,
+ },
+};
+
+static int wm5110_codec_probe(struct snd_soc_codec *codec)
+{
+ struct wm5110_priv *priv = snd_soc_codec_get_drvdata(codec);
+
+ codec->control_data = priv->core.arizona->regmap;
+ return snd_soc_codec_set_cache_io(codec, 32, 16, SND_SOC_REGMAP);
+}
+
+#define WM5110_DIG_VU 0x0200
+
+static unsigned int wm5110_digital_vu[] = {
+ ARIZONA_ADC_DIGITAL_VOLUME_1L,
+ ARIZONA_ADC_DIGITAL_VOLUME_1R,
+ ARIZONA_ADC_DIGITAL_VOLUME_2L,
+ ARIZONA_ADC_DIGITAL_VOLUME_2R,
+ ARIZONA_ADC_DIGITAL_VOLUME_3L,
+ ARIZONA_ADC_DIGITAL_VOLUME_3R,
+
+ ARIZONA_DAC_DIGITAL_VOLUME_1L,
+ ARIZONA_DAC_DIGITAL_VOLUME_1R,
+ ARIZONA_DAC_DIGITAL_VOLUME_2L,
+ ARIZONA_DAC_DIGITAL_VOLUME_2R,
+ ARIZONA_DAC_DIGITAL_VOLUME_3L,
+ ARIZONA_DAC_DIGITAL_VOLUME_3R,
+ ARIZONA_DAC_DIGITAL_VOLUME_4L,
+ ARIZONA_DAC_DIGITAL_VOLUME_4R,
+ ARIZONA_DAC_DIGITAL_VOLUME_5L,
+ ARIZONA_DAC_DIGITAL_VOLUME_5R,
+};
+
+static struct snd_soc_codec_driver soc_codec_dev_wm5110 = {
+ .probe = wm5110_codec_probe,
+
+ .idle_bias_off = true,
+
+ .set_sysclk = arizona_set_sysclk,
+ .set_pll = wm5110_set_fll,
+
+ .controls = wm5110_snd_controls,
+ .num_controls = ARRAY_SIZE(wm5110_snd_controls),
+ .dapm_widgets = wm5110_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(wm5110_dapm_widgets),
+ .dapm_routes = wm5110_dapm_routes,
+ .num_dapm_routes = ARRAY_SIZE(wm5110_dapm_routes),
+};
+
+static int __devinit wm5110_probe(struct platform_device *pdev)
+{
+ struct arizona *arizona = dev_get_drvdata(pdev->dev.parent);
+ struct wm5110_priv *wm5110;
+ int i;
+
+ wm5110 = devm_kzalloc(&pdev->dev, sizeof(struct wm5110_priv),
+ GFP_KERNEL);
+ if (wm5110 == NULL)
+ return -ENOMEM;
+ platform_set_drvdata(pdev, wm5110);
+
+ wm5110->core.arizona = arizona;
+
+ for (i = 0; i < ARRAY_SIZE(wm5110->fll); i++)
+ wm5110->fll[i].vco_mult = 3;
+
+ arizona_init_fll(arizona, 1, ARIZONA_FLL1_CONTROL_1 - 1,
+ ARIZONA_IRQ_FLL1_LOCK, ARIZONA_IRQ_FLL1_CLOCK_OK,
+ &wm5110->fll[0]);
+ arizona_init_fll(arizona, 2, ARIZONA_FLL2_CONTROL_1 - 1,
+ ARIZONA_IRQ_FLL2_LOCK, ARIZONA_IRQ_FLL2_CLOCK_OK,
+ &wm5110->fll[1]);
+
+ for (i = 0; i < ARRAY_SIZE(wm5110_dai); i++)
+ arizona_init_dai(&wm5110->core, i);
+
+ /* Latch volume update bits */
+ for (i = 0; i < ARRAY_SIZE(wm5110_digital_vu); i++)
+ regmap_update_bits(arizona->regmap, wm5110_digital_vu[i],
+ WM5110_DIG_VU, WM5110_DIG_VU);
+
+ pm_runtime_enable(&pdev->dev);
+ pm_runtime_idle(&pdev->dev);
+
+ return snd_soc_register_codec(&pdev->dev, &soc_codec_dev_wm5110,
+ wm5110_dai, ARRAY_SIZE(wm5110_dai));
+}
+
+static int __devexit wm5110_remove(struct platform_device *pdev)
+{
+ snd_soc_unregister_codec(&pdev->dev);
+ pm_runtime_disable(&pdev->dev);
+
+ return 0;
+}
+
+static struct platform_driver wm5110_codec_driver = {
+ .driver = {
+ .name = "wm5110-codec",
+ .owner = THIS_MODULE,
+ },
+ .probe = wm5110_probe,
+ .remove = __devexit_p(wm5110_remove),
+};
+
+module_platform_driver(wm5110_codec_driver);
+
+MODULE_DESCRIPTION("ASoC WM5110 driver");
+MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:wm5110-codec");
diff --git a/sound/soc/codecs/wm5110.h b/sound/soc/codecs/wm5110.h
new file mode 100644
index 0000000..75e9351
--- /dev/null
+++ b/sound/soc/codecs/wm5110.h
@@ -0,0 +1,21 @@
+/*
+ * wm5110.h -- WM5110 ALSA SoC Audio driver
+ *
+ * Copyright 2012 Wolfson Microelectronics plc
+ *
+ * Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
+ *
+ * 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 _WM5110_H
+#define _WM5110_H
+
+#include "arizona.h"
+
+#define WM5110_FLL1 1
+#define WM5110_FLL2 2
+
+#endif
diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c
index 555ee14..d26c8ae 100644
--- a/sound/soc/codecs/wm8350.c
+++ b/sound/soc/codecs/wm8350.c
@@ -1,7 +1,7 @@
/*
* wm8350.c -- WM8350 ALSA SoC audio driver
*
- * Copyright (C) 2007, 2008 Wolfson Microelectronics PLC.
+ * Copyright (C) 2007-12 Wolfson Microelectronics PLC.
*
* Author: Liam Girdwood <lrg@slimlogic.co.uk>
*
@@ -71,20 +71,6 @@ struct wm8350_data {
int fll_freq_in;
};
-static unsigned int wm8350_codec_read(struct snd_soc_codec *codec,
- unsigned int reg)
-{
- struct wm8350 *wm8350 = codec->control_data;
- return wm8350_reg_read(wm8350, reg);
-}
-
-static int wm8350_codec_write(struct snd_soc_codec *codec, unsigned int reg,
- unsigned int value)
-{
- struct wm8350 *wm8350 = codec->control_data;
- return wm8350_reg_write(wm8350, reg, value);
-}
-
/*
* Ramp OUT1 PGA volume to minimise pops at stream startup and shutdown.
*/
@@ -1519,7 +1505,9 @@ static int wm8350_codec_probe(struct snd_soc_codec *codec)
if (ret != 0)
return ret;
- codec->control_data = wm8350;
+ codec->control_data = wm8350->regmap;
+
+ snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP);
/* Put the codec into reset if it wasn't already */
wm8350_clear_bits(wm8350, WM8350_POWER_MGMT_5, WM8350_CODEC_ENA);
@@ -1629,8 +1617,6 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8350 = {
.remove = wm8350_codec_remove,
.suspend = wm8350_suspend,
.resume = wm8350_resume,
- .read = wm8350_codec_read,
- .write = wm8350_codec_write,
.set_bias_level = wm8350_set_bias_level,
.controls = wm8350_snd_controls,
diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c
index 5dc31eb..5d277a9 100644
--- a/sound/soc/codecs/wm8400.c
+++ b/sound/soc/codecs/wm8400.c
@@ -1,7 +1,7 @@
/*
* wm8400.c -- WM8400 ALSA Soc Audio driver
*
- * Copyright 2008, 2009 Wolfson Microelectronics PLC.
+ * Copyright 2008-11 Wolfson Microelectronics PLC.
* Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
*
* This program is free software; you can redistribute it and/or modify it
diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c
index 2112851..7c68226 100644
--- a/sound/soc/codecs/wm8580.c
+++ b/sound/soc/codecs/wm8580.c
@@ -1,7 +1,7 @@
/*
* wm8580.c -- WM8580 ALSA Soc Audio driver
*
- * Copyright 2008, 2009 Wolfson Microelectronics PLC.
+ * Copyright 2008-11 Wolfson Microelectronics PLC.
*
* 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
diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c
index 9d1b9b02..bb1d269 100644
--- a/sound/soc/codecs/wm8731.c
+++ b/sound/soc/codecs/wm8731.c
@@ -2,6 +2,7 @@
* wm8731.c -- WM8731 ALSA SoC Audio driver
*
* Copyright 2005 Openedhand Ltd.
+ * Copyright 2006-12 Wolfson Microelectronics, plc
*
* Author: Richard Purdie <richard@openedhand.com>
*
diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c
index 6e849cb..35f3d23 100644
--- a/sound/soc/codecs/wm8741.c
+++ b/sound/soc/codecs/wm8741.c
@@ -1,7 +1,7 @@
/*
* wm8741.c -- WM8741 ALSA SoC Audio driver
*
- * Copyright 2010 Wolfson Microelectronics plc
+ * Copyright 2010-1 Wolfson Microelectronics plc
*
* Author: Ian Lartey <ian@opensource.wolfsonmicro.com>
*
diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c
index a26482c..13bff87 100644
--- a/sound/soc/codecs/wm8753.c
+++ b/sound/soc/codecs/wm8753.c
@@ -1,7 +1,7 @@
/*
* wm8753.c -- WM8753 ALSA Soc Audio driver
*
- * Copyright 2003 Wolfson Microelectronics PLC.
+ * Copyright 2003-11 Wolfson Microelectronics PLC.
* Author: Liam Girdwood <lrg@slimlogic.co.uk>
*
* This program is free software; you can redistribute it and/or modify it
diff --git a/sound/soc/codecs/wm8776.c b/sound/soc/codecs/wm8776.c
index a19db5a..879c356 100644
--- a/sound/soc/codecs/wm8776.c
+++ b/sound/soc/codecs/wm8776.c
@@ -1,7 +1,7 @@
/*
* wm8776.c -- WM8776 ALSA SoC Audio driver
*
- * Copyright 2009 Wolfson Microelectronics plc
+ * Copyright 2009-12 Wolfson Microelectronics plc
*
* Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
*
diff --git a/sound/soc/codecs/wm8804.c b/sound/soc/codecs/wm8804.c
index 6bd1b76..c088020 100644
--- a/sound/soc/codecs/wm8804.c
+++ b/sound/soc/codecs/wm8804.c
@@ -1,7 +1,7 @@
/*
* wm8804.c -- WM8804 S/PDIF transceiver driver
*
- * Copyright 2010 Wolfson Microelectronics plc
+ * Copyright 2010-11 Wolfson Microelectronics plc
*
* Author: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
*
diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c
index 86b8a29..73f1c8d 100644
--- a/sound/soc/codecs/wm8903.c
+++ b/sound/soc/codecs/wm8903.c
@@ -1,8 +1,8 @@
/*
* wm8903.c -- WM8903 ALSA SoC Audio driver
*
- * Copyright 2008 Wolfson Microelectronics
- * Copyright 2011 NVIDIA, Inc.
+ * Copyright 2008-12 Wolfson Microelectronics
+ * Copyright 2011-2012 NVIDIA, Inc.
*
* Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
*
@@ -116,6 +116,7 @@ static const struct reg_default wm8903_reg_defaults[] = {
struct wm8903_priv {
struct wm8903_platform_data *pdata;
+ struct device *dev;
struct snd_soc_codec *codec;
struct regmap *regmap;
@@ -1635,17 +1636,27 @@ EXPORT_SYMBOL_GPL(wm8903_mic_detect);
static irqreturn_t wm8903_irq(int irq, void *data)
{
- struct snd_soc_codec *codec = data;
- struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec);
- int mic_report;
- int int_pol;
- int int_val = 0;
- int mask = ~snd_soc_read(codec, WM8903_INTERRUPT_STATUS_1_MASK);
+ struct wm8903_priv *wm8903 = data;
+ int mic_report, ret;
+ unsigned int int_val, mask, int_pol;
- int_val = snd_soc_read(codec, WM8903_INTERRUPT_STATUS_1) & mask;
+ ret = regmap_read(wm8903->regmap, WM8903_INTERRUPT_STATUS_1_MASK,
+ &mask);
+ if (ret != 0) {
+ dev_err(wm8903->dev, "Failed to read IRQ mask: %d\n", ret);
+ return IRQ_NONE;
+ }
+
+ ret = regmap_read(wm8903->regmap, WM8903_INTERRUPT_STATUS_1, &int_val);
+ if (ret != 0) {
+ dev_err(wm8903->dev, "Failed to read IRQ status: %d\n", ret);
+ return IRQ_NONE;
+ }
+
+ int_val &= ~mask;
if (int_val & WM8903_WSEQ_BUSY_EINT) {
- dev_warn(codec->dev, "Write sequencer done\n");
+ dev_warn(wm8903->dev, "Write sequencer done\n");
}
/*
@@ -1656,22 +1667,28 @@ static irqreturn_t wm8903_irq(int irq, void *data)
* the polarity register.
*/
mic_report = wm8903->mic_last_report;
- int_pol = snd_soc_read(codec, WM8903_INTERRUPT_POLARITY_1);
+ ret = regmap_read(wm8903->regmap, WM8903_INTERRUPT_POLARITY_1,
+ &int_pol);
+ if (ret != 0) {
+ dev_err(wm8903->dev, "Failed to read interrupt polarity: %d\n",
+ ret);
+ return IRQ_HANDLED;
+ }
#ifndef CONFIG_SND_SOC_WM8903_MODULE
if (int_val & (WM8903_MICSHRT_EINT | WM8903_MICDET_EINT))
- trace_snd_soc_jack_irq(dev_name(codec->dev));
+ trace_snd_soc_jack_irq(dev_name(wm8903->dev));
#endif
if (int_val & WM8903_MICSHRT_EINT) {
- dev_dbg(codec->dev, "Microphone short (pol=%x)\n", int_pol);
+ dev_dbg(wm8903->dev, "Microphone short (pol=%x)\n", int_pol);
mic_report ^= wm8903->mic_short;
int_pol ^= WM8903_MICSHRT_INV;
}
if (int_val & WM8903_MICDET_EINT) {
- dev_dbg(codec->dev, "Microphone detect (pol=%x)\n", int_pol);
+ dev_dbg(wm8903->dev, "Microphone detect (pol=%x)\n", int_pol);
mic_report ^= wm8903->mic_det;
int_pol ^= WM8903_MICDET_INV;
@@ -1679,8 +1696,8 @@ static irqreturn_t wm8903_irq(int irq, void *data)
msleep(wm8903->mic_delay);
}
- snd_soc_update_bits(codec, WM8903_INTERRUPT_POLARITY_1,
- WM8903_MICSHRT_INV | WM8903_MICDET_INV, int_pol);
+ regmap_update_bits(wm8903->regmap, WM8903_INTERRUPT_POLARITY_1,
+ WM8903_MICSHRT_INV | WM8903_MICDET_INV, int_pol);
snd_soc_jack_report(wm8903->mic_jack, mic_report,
wm8903->mic_short | wm8903->mic_det);
@@ -1774,7 +1791,6 @@ static int wm8903_gpio_request(struct gpio_chip *chip, unsigned offset)
static int wm8903_gpio_direction_in(struct gpio_chip *chip, unsigned offset)
{
struct wm8903_priv *wm8903 = gpio_to_wm8903(chip);
- struct snd_soc_codec *codec = wm8903->codec;
unsigned int mask, val;
int ret;
@@ -1782,8 +1798,8 @@ static int wm8903_gpio_direction_in(struct gpio_chip *chip, unsigned offset)
val = (WM8903_GPn_FN_GPIO_INPUT << WM8903_GP1_FN_SHIFT) |
WM8903_GP1_DIR;
- ret = snd_soc_update_bits(codec, WM8903_GPIO_CONTROL_1 + offset,
- mask, val);
+ ret = regmap_update_bits(wm8903->regmap,
+ WM8903_GPIO_CONTROL_1 + offset, mask, val);
if (ret < 0)
return ret;
@@ -1793,10 +1809,9 @@ static int wm8903_gpio_direction_in(struct gpio_chip *chip, unsigned offset)
static int wm8903_gpio_get(struct gpio_chip *chip, unsigned offset)
{
struct wm8903_priv *wm8903 = gpio_to_wm8903(chip);
- struct snd_soc_codec *codec = wm8903->codec;
- int reg;
+ unsigned int reg;
- reg = snd_soc_read(codec, WM8903_GPIO_CONTROL_1 + offset);
+ regmap_read(wm8903->regmap, WM8903_GPIO_CONTROL_1 + offset, &reg);
return (reg & WM8903_GP1_LVL_MASK) >> WM8903_GP1_LVL_SHIFT;
}
@@ -1805,7 +1820,6 @@ static int wm8903_gpio_direction_out(struct gpio_chip *chip,
unsigned offset, int value)
{
struct wm8903_priv *wm8903 = gpio_to_wm8903(chip);
- struct snd_soc_codec *codec = wm8903->codec;
unsigned int mask, val;
int ret;
@@ -1813,8 +1827,8 @@ static int wm8903_gpio_direction_out(struct gpio_chip *chip,
val = (WM8903_GPn_FN_GPIO_OUTPUT << WM8903_GP1_FN_SHIFT) |
(value << WM8903_GP2_LVL_SHIFT);
- ret = snd_soc_update_bits(codec, WM8903_GPIO_CONTROL_1 + offset,
- mask, val);
+ ret = regmap_update_bits(wm8903->regmap,
+ WM8903_GPIO_CONTROL_1 + offset, mask, val);
if (ret < 0)
return ret;
@@ -1824,11 +1838,10 @@ static int wm8903_gpio_direction_out(struct gpio_chip *chip,
static void wm8903_gpio_set(struct gpio_chip *chip, unsigned offset, int value)
{
struct wm8903_priv *wm8903 = gpio_to_wm8903(chip);
- struct snd_soc_codec *codec = wm8903->codec;
- snd_soc_update_bits(codec, WM8903_GPIO_CONTROL_1 + offset,
- WM8903_GP1_LVL_MASK,
- !!value << WM8903_GP1_LVL_SHIFT);
+ regmap_update_bits(wm8903->regmap, WM8903_GPIO_CONTROL_1 + offset,
+ WM8903_GP1_LVL_MASK,
+ !!value << WM8903_GP1_LVL_SHIFT);
}
static struct gpio_chip wm8903_template_chip = {
@@ -1842,15 +1855,14 @@ static struct gpio_chip wm8903_template_chip = {
.can_sleep = 1,
};
-static void wm8903_init_gpio(struct snd_soc_codec *codec)
+static void wm8903_init_gpio(struct wm8903_priv *wm8903)
{
- struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec);
struct wm8903_platform_data *pdata = wm8903->pdata;
int ret;
wm8903->gpio_chip = wm8903_template_chip;
wm8903->gpio_chip.ngpio = WM8903_NUM_GPIO;
- wm8903->gpio_chip.dev = codec->dev;
+ wm8903->gpio_chip.dev = wm8903->dev;
if (pdata->gpio_base)
wm8903->gpio_chip.base = pdata->gpio_base;
@@ -1859,24 +1871,23 @@ static void wm8903_init_gpio(struct snd_soc_codec *codec)
ret = gpiochip_add(&wm8903->gpio_chip);
if (ret != 0)
- dev_err(codec->dev, "Failed to add GPIOs: %d\n", ret);
+ dev_err(wm8903->dev, "Failed to add GPIOs: %d\n", ret);
}
-static void wm8903_free_gpio(struct snd_soc_codec *codec)
+static void wm8903_free_gpio(struct wm8903_priv *wm8903)
{
- struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec);
int ret;
ret = gpiochip_remove(&wm8903->gpio_chip);
if (ret != 0)
- dev_err(codec->dev, "Failed to remove GPIOs: %d\n", ret);
+ dev_err(wm8903->dev, "Failed to remove GPIOs: %d\n", ret);
}
#else
-static void wm8903_init_gpio(struct snd_soc_codec *codec)
+static void wm8903_init_gpio(struct wm8903_priv *wm8903)
{
}
-static void wm8903_free_gpio(struct snd_soc_codec *codec)
+static void wm8903_free_gpio(struct wm8903_priv *wm8903)
{
}
#endif
@@ -1884,11 +1895,7 @@ static void wm8903_free_gpio(struct snd_soc_codec *codec)
static int wm8903_probe(struct snd_soc_codec *codec)
{
struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec);
- struct wm8903_platform_data *pdata = wm8903->pdata;
- int ret, i;
- int trigger, irq_pol;
- u16 val;
- bool mic_gpio = false;
+ int ret;
wm8903->codec = codec;
codec->control_data = wm8903->regmap;
@@ -1899,121 +1906,16 @@ static int wm8903_probe(struct snd_soc_codec *codec)
return ret;
}
- /* Set up GPIOs, detect if any are MIC detect outputs */
- for (i = 0; i < ARRAY_SIZE(pdata->gpio_cfg); i++) {
- if ((!pdata->gpio_cfg[i]) ||
- (pdata->gpio_cfg[i] > WM8903_GPIO_CONFIG_ZERO))
- continue;
-
- snd_soc_write(codec, WM8903_GPIO_CONTROL_1 + i,
- pdata->gpio_cfg[i] & 0x7fff);
-
- val = (pdata->gpio_cfg[i] & WM8903_GP1_FN_MASK)
- >> WM8903_GP1_FN_SHIFT;
-
- switch (val) {
- case WM8903_GPn_FN_MICBIAS_CURRENT_DETECT:
- case WM8903_GPn_FN_MICBIAS_SHORT_DETECT:
- mic_gpio = true;
- break;
- default:
- break;
- }
- }
-
- /* Set up microphone detection */
- snd_soc_write(codec, WM8903_MIC_BIAS_CONTROL_0,
- pdata->micdet_cfg);
-
- /* Microphone detection needs the WSEQ clock */
- if (pdata->micdet_cfg)
- snd_soc_update_bits(codec, WM8903_WRITE_SEQUENCER_0,
- WM8903_WSEQ_ENA, WM8903_WSEQ_ENA);
-
- /* If microphone detection is enabled by pdata but
- * detected via IRQ then interrupts can be lost before
- * the machine driver has set up microphone detection
- * IRQs as the IRQs are clear on read. The detection
- * will be enabled when the machine driver configures.
- */
- WARN_ON(!mic_gpio && (pdata->micdet_cfg & WM8903_MICDET_ENA));
-
- wm8903->mic_delay = pdata->micdet_delay;
-
- if (wm8903->irq) {
- if (pdata->irq_active_low) {
- trigger = IRQF_TRIGGER_LOW;
- irq_pol = WM8903_IRQ_POL;
- } else {
- trigger = IRQF_TRIGGER_HIGH;
- irq_pol = 0;
- }
-
- snd_soc_update_bits(codec, WM8903_INTERRUPT_CONTROL,
- WM8903_IRQ_POL, irq_pol);
-
- ret = request_threaded_irq(wm8903->irq, NULL, wm8903_irq,
- trigger | IRQF_ONESHOT,
- "wm8903", codec);
- if (ret != 0) {
- dev_err(codec->dev, "Failed to request IRQ: %d\n",
- ret);
- return ret;
- }
-
- /* Enable write sequencer interrupts */
- snd_soc_update_bits(codec, WM8903_INTERRUPT_STATUS_1_MASK,
- WM8903_IM_WSEQ_BUSY_EINT, 0);
- }
-
/* power on device */
wm8903_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
- /* Latch volume update bits */
- val = snd_soc_read(codec, WM8903_ADC_DIGITAL_VOLUME_LEFT);
- val |= WM8903_ADCVU;
- snd_soc_write(codec, WM8903_ADC_DIGITAL_VOLUME_LEFT, val);
- snd_soc_write(codec, WM8903_ADC_DIGITAL_VOLUME_RIGHT, val);
-
- val = snd_soc_read(codec, WM8903_DAC_DIGITAL_VOLUME_LEFT);
- val |= WM8903_DACVU;
- snd_soc_write(codec, WM8903_DAC_DIGITAL_VOLUME_LEFT, val);
- snd_soc_write(codec, WM8903_DAC_DIGITAL_VOLUME_RIGHT, val);
-
- val = snd_soc_read(codec, WM8903_ANALOGUE_OUT1_LEFT);
- val |= WM8903_HPOUTVU;
- snd_soc_write(codec, WM8903_ANALOGUE_OUT1_LEFT, val);
- snd_soc_write(codec, WM8903_ANALOGUE_OUT1_RIGHT, val);
-
- val = snd_soc_read(codec, WM8903_ANALOGUE_OUT2_LEFT);
- val |= WM8903_LINEOUTVU;
- snd_soc_write(codec, WM8903_ANALOGUE_OUT2_LEFT, val);
- snd_soc_write(codec, WM8903_ANALOGUE_OUT2_RIGHT, val);
-
- val = snd_soc_read(codec, WM8903_ANALOGUE_OUT3_LEFT);
- val |= WM8903_SPKVU;
- snd_soc_write(codec, WM8903_ANALOGUE_OUT3_LEFT, val);
- snd_soc_write(codec, WM8903_ANALOGUE_OUT3_RIGHT, val);
-
- /* Enable DAC soft mute by default */
- snd_soc_update_bits(codec, WM8903_DAC_DIGITAL_1,
- WM8903_DAC_MUTEMODE | WM8903_DAC_MUTE,
- WM8903_DAC_MUTEMODE | WM8903_DAC_MUTE);
-
- wm8903_init_gpio(codec);
-
return ret;
}
/* power down chip */
static int wm8903_remove(struct snd_soc_codec *codec)
{
- struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec);
-
- wm8903_free_gpio(codec);
wm8903_set_bias_level(codec, SND_SOC_BIAS_OFF);
- if (wm8903->irq)
- free_irq(wm8903->irq, codec);
return 0;
}
@@ -2123,15 +2025,18 @@ static __devinit int wm8903_i2c_probe(struct i2c_client *i2c,
{
struct wm8903_platform_data *pdata = dev_get_platdata(&i2c->dev);
struct wm8903_priv *wm8903;
- unsigned int val;
- int ret;
+ int trigger;
+ bool mic_gpio = false;
+ unsigned int val, irq_pol;
+ int ret, i;
wm8903 = devm_kzalloc(&i2c->dev, sizeof(struct wm8903_priv),
GFP_KERNEL);
if (wm8903 == NULL)
return -ENOMEM;
+ wm8903->dev = &i2c->dev;
- wm8903->regmap = regmap_init_i2c(i2c, &wm8903_regmap);
+ wm8903->regmap = devm_regmap_init_i2c(i2c, &wm8903_regmap);
if (IS_ERR(wm8903->regmap)) {
ret = PTR_ERR(wm8903->regmap);
dev_err(&i2c->dev, "Failed to allocate register map: %d\n",
@@ -2140,7 +2045,6 @@ static __devinit int wm8903_i2c_probe(struct i2c_client *i2c,
}
i2c_set_clientdata(i2c, wm8903);
- wm8903->irq = i2c->irq;
/* If no platform data was supplied, create storage for defaults */
if (pdata) {
@@ -2167,6 +2071,8 @@ static __devinit int wm8903_i2c_probe(struct i2c_client *i2c,
}
}
+ pdata = wm8903->pdata;
+
ret = regmap_read(wm8903->regmap, WM8903_SW_RESET_AND_ID, &val);
if (ret != 0) {
dev_err(&i2c->dev, "Failed to read chip ID: %d\n", ret);
@@ -2189,6 +2095,107 @@ static __devinit int wm8903_i2c_probe(struct i2c_client *i2c,
/* Reset the device */
regmap_write(wm8903->regmap, WM8903_SW_RESET_AND_ID, 0x8903);
+ wm8903_init_gpio(wm8903);
+
+ /* Set up GPIO pin state, detect if any are MIC detect outputs */
+ for (i = 0; i < ARRAY_SIZE(pdata->gpio_cfg); i++) {
+ if ((!pdata->gpio_cfg[i]) ||
+ (pdata->gpio_cfg[i] > WM8903_GPIO_CONFIG_ZERO))
+ continue;
+
+ regmap_write(wm8903->regmap, WM8903_GPIO_CONTROL_1 + i,
+ pdata->gpio_cfg[i] & 0x7fff);
+
+ val = (pdata->gpio_cfg[i] & WM8903_GP1_FN_MASK)
+ >> WM8903_GP1_FN_SHIFT;
+
+ switch (val) {
+ case WM8903_GPn_FN_MICBIAS_CURRENT_DETECT:
+ case WM8903_GPn_FN_MICBIAS_SHORT_DETECT:
+ mic_gpio = true;
+ break;
+ default:
+ break;
+ }
+ }
+
+ /* Set up microphone detection */
+ regmap_write(wm8903->regmap, WM8903_MIC_BIAS_CONTROL_0,
+ pdata->micdet_cfg);
+
+ /* Microphone detection needs the WSEQ clock */
+ if (pdata->micdet_cfg)
+ regmap_update_bits(wm8903->regmap, WM8903_WRITE_SEQUENCER_0,
+ WM8903_WSEQ_ENA, WM8903_WSEQ_ENA);
+
+ /* If microphone detection is enabled by pdata but
+ * detected via IRQ then interrupts can be lost before
+ * the machine driver has set up microphone detection
+ * IRQs as the IRQs are clear on read. The detection
+ * will be enabled when the machine driver configures.
+ */
+ WARN_ON(!mic_gpio && (pdata->micdet_cfg & WM8903_MICDET_ENA));
+
+ wm8903->mic_delay = pdata->micdet_delay;
+
+ if (i2c->irq) {
+ if (pdata->irq_active_low) {
+ trigger = IRQF_TRIGGER_LOW;
+ irq_pol = WM8903_IRQ_POL;
+ } else {
+ trigger = IRQF_TRIGGER_HIGH;
+ irq_pol = 0;
+ }
+
+ regmap_update_bits(wm8903->regmap, WM8903_INTERRUPT_CONTROL,
+ WM8903_IRQ_POL, irq_pol);
+
+ ret = request_threaded_irq(i2c->irq, NULL, wm8903_irq,
+ trigger | IRQF_ONESHOT,
+ "wm8903", wm8903);
+ if (ret != 0) {
+ dev_err(wm8903->dev, "Failed to request IRQ: %d\n",
+ ret);
+ return ret;
+ }
+
+ /* Enable write sequencer interrupts */
+ regmap_update_bits(wm8903->regmap,
+ WM8903_INTERRUPT_STATUS_1_MASK,
+ WM8903_IM_WSEQ_BUSY_EINT, 0);
+ }
+
+ /* Latch volume update bits */
+ regmap_update_bits(wm8903->regmap, WM8903_ADC_DIGITAL_VOLUME_LEFT,
+ WM8903_ADCVU, WM8903_ADCVU);
+ regmap_update_bits(wm8903->regmap, WM8903_ADC_DIGITAL_VOLUME_RIGHT,
+ WM8903_ADCVU, WM8903_ADCVU);
+
+ regmap_update_bits(wm8903->regmap, WM8903_DAC_DIGITAL_VOLUME_LEFT,
+ WM8903_DACVU, WM8903_DACVU);
+ regmap_update_bits(wm8903->regmap, WM8903_DAC_DIGITAL_VOLUME_RIGHT,
+ WM8903_DACVU, WM8903_DACVU);
+
+ regmap_update_bits(wm8903->regmap, WM8903_ANALOGUE_OUT1_LEFT,
+ WM8903_HPOUTVU, WM8903_HPOUTVU);
+ regmap_update_bits(wm8903->regmap, WM8903_ANALOGUE_OUT1_RIGHT,
+ WM8903_HPOUTVU, WM8903_HPOUTVU);
+
+ regmap_update_bits(wm8903->regmap, WM8903_ANALOGUE_OUT2_LEFT,
+ WM8903_LINEOUTVU, WM8903_LINEOUTVU);
+ regmap_update_bits(wm8903->regmap, WM8903_ANALOGUE_OUT2_RIGHT,
+ WM8903_LINEOUTVU, WM8903_LINEOUTVU);
+
+ regmap_update_bits(wm8903->regmap, WM8903_ANALOGUE_OUT3_LEFT,
+ WM8903_SPKVU, WM8903_SPKVU);
+ regmap_update_bits(wm8903->regmap, WM8903_ANALOGUE_OUT3_RIGHT,
+ WM8903_SPKVU, WM8903_SPKVU);
+
+ /* Enable DAC soft mute by default */
+ regmap_update_bits(wm8903->regmap, WM8903_DAC_DIGITAL_1,
+ WM8903_DAC_MUTEMODE | WM8903_DAC_MUTE,
+ WM8903_DAC_MUTEMODE | WM8903_DAC_MUTE);
+
ret = snd_soc_register_codec(&i2c->dev,
&soc_codec_dev_wm8903, &wm8903_dai, 1);
if (ret != 0)
@@ -2196,7 +2203,6 @@ static __devinit int wm8903_i2c_probe(struct i2c_client *i2c,
return 0;
err:
- regmap_exit(wm8903->regmap);
return ret;
}
@@ -2204,7 +2210,9 @@ static __devexit int wm8903_i2c_remove(struct i2c_client *client)
{
struct wm8903_priv *wm8903 = i2c_get_clientdata(client);
- regmap_exit(wm8903->regmap);
+ if (client->irq)
+ free_irq(client->irq, wm8903);
+ wm8903_free_gpio(wm8903);
snd_soc_unregister_codec(&client->dev);
return 0;
diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c
index 812acd8..0013afe 100644
--- a/sound/soc/codecs/wm8904.c
+++ b/sound/soc/codecs/wm8904.c
@@ -1,7 +1,7 @@
/*
* wm8904.c -- WM8904 ALSA SoC Audio driver
*
- * Copyright 2009 Wolfson Microelectronics plc
+ * Copyright 2009-12 Wolfson Microelectronics plc
*
* Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
*
@@ -314,11 +314,6 @@ static bool wm8904_readable_register(struct device *dev, unsigned int reg)
}
}
-static int wm8904_reset(struct snd_soc_codec *codec)
-{
- return snd_soc_write(codec, WM8904_SW_RESET_AND_ID, 0);
-}
-
static int wm8904_configure_clocking(struct snd_soc_codec *codec)
{
struct wm8904_priv *wm8904 = snd_soc_codec_get_drvdata(codec);
@@ -1945,25 +1940,6 @@ static struct snd_soc_dai_driver wm8904_dai = {
.symmetric_rates = 1,
};
-#ifdef CONFIG_PM
-static int wm8904_suspend(struct snd_soc_codec *codec)
-{
- wm8904_set_bias_level(codec, SND_SOC_BIAS_OFF);
-
- return 0;
-}
-
-static int wm8904_resume(struct snd_soc_codec *codec)
-{
- wm8904_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-
- return 0;
-}
-#else
-#define wm8904_suspend NULL
-#define wm8904_resume NULL
-#endif
-
static void wm8904_handle_retune_mobile_pdata(struct snd_soc_codec *codec)
{
struct wm8904_priv *wm8904 = snd_soc_codec_get_drvdata(codec);
@@ -2078,8 +2054,7 @@ static void wm8904_handle_pdata(struct snd_soc_codec *codec)
static int wm8904_probe(struct snd_soc_codec *codec)
{
struct wm8904_priv *wm8904 = snd_soc_codec_get_drvdata(codec);
- struct wm8904_pdata *pdata = wm8904->pdata;
- int ret, i;
+ int ret;
codec->control_data = wm8904->regmap;
@@ -2101,127 +2076,17 @@ static int wm8904_probe(struct snd_soc_codec *codec)
return ret;
}
- for (i = 0; i < ARRAY_SIZE(wm8904->supplies); i++)
- wm8904->supplies[i].supply = wm8904_supply_names[i];
-
- ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(wm8904->supplies),
- wm8904->supplies);
- if (ret != 0) {
- dev_err(codec->dev, "Failed to request supplies: %d\n", ret);
- return ret;
- }
-
- ret = regulator_bulk_enable(ARRAY_SIZE(wm8904->supplies),
- wm8904->supplies);
- if (ret != 0) {
- dev_err(codec->dev, "Failed to enable supplies: %d\n", ret);
- goto err_get;
- }
-
- ret = snd_soc_read(codec, WM8904_SW_RESET_AND_ID);
- if (ret < 0) {
- dev_err(codec->dev, "Failed to read ID register\n");
- goto err_enable;
- }
- if (ret != 0x8904) {
- dev_err(codec->dev, "Device is not a WM8904, ID is %x\n", ret);
- ret = -EINVAL;
- goto err_enable;
- }
-
- ret = snd_soc_read(codec, WM8904_REVISION);
- if (ret < 0) {
- dev_err(codec->dev, "Failed to read device revision: %d\n",
- ret);
- goto err_enable;
- }
- dev_info(codec->dev, "revision %c\n", ret + 'A');
-
- ret = wm8904_reset(codec);
- if (ret < 0) {
- dev_err(codec->dev, "Failed to issue reset\n");
- goto err_enable;
- }
-
- regcache_cache_only(wm8904->regmap, true);
- /* Change some default settings - latch VU and enable ZC */
- snd_soc_update_bits(codec, WM8904_ADC_DIGITAL_VOLUME_LEFT,
- WM8904_ADC_VU, WM8904_ADC_VU);
- snd_soc_update_bits(codec, WM8904_ADC_DIGITAL_VOLUME_RIGHT,
- WM8904_ADC_VU, WM8904_ADC_VU);
- snd_soc_update_bits(codec, WM8904_DAC_DIGITAL_VOLUME_LEFT,
- WM8904_DAC_VU, WM8904_DAC_VU);
- snd_soc_update_bits(codec, WM8904_DAC_DIGITAL_VOLUME_RIGHT,
- WM8904_DAC_VU, WM8904_DAC_VU);
- snd_soc_update_bits(codec, WM8904_ANALOGUE_OUT1_LEFT,
- WM8904_HPOUT_VU | WM8904_HPOUTLZC,
- WM8904_HPOUT_VU | WM8904_HPOUTLZC);
- snd_soc_update_bits(codec, WM8904_ANALOGUE_OUT1_RIGHT,
- WM8904_HPOUT_VU | WM8904_HPOUTRZC,
- WM8904_HPOUT_VU | WM8904_HPOUTRZC);
- snd_soc_update_bits(codec, WM8904_ANALOGUE_OUT2_LEFT,
- WM8904_LINEOUT_VU | WM8904_LINEOUTLZC,
- WM8904_LINEOUT_VU | WM8904_LINEOUTLZC);
- snd_soc_update_bits(codec, WM8904_ANALOGUE_OUT2_RIGHT,
- WM8904_LINEOUT_VU | WM8904_LINEOUTRZC,
- WM8904_LINEOUT_VU | WM8904_LINEOUTRZC);
- snd_soc_update_bits(codec, WM8904_CLOCK_RATES_0,
- WM8904_SR_MODE, 0);
-
- /* Apply configuration from the platform data. */
- if (wm8904->pdata) {
- for (i = 0; i < WM8904_GPIO_REGS; i++) {
- if (!pdata->gpio_cfg[i])
- continue;
-
- regmap_update_bits(wm8904->regmap,
- WM8904_GPIO_CONTROL_1 + i,
- 0xffff,
- pdata->gpio_cfg[i]);
- }
-
- /* Zero is the default value for these anyway */
- for (i = 0; i < WM8904_MIC_REGS; i++)
- regmap_update_bits(wm8904->regmap,
- WM8904_MIC_BIAS_CONTROL_0 + i,
- 0xffff,
- pdata->mic_cfg[i]);
- }
-
- /* Set Class W by default - this will be managed by the Class
- * G widget at runtime where bypass paths are available.
- */
- snd_soc_update_bits(codec, WM8904_CLASS_W_0,
- WM8904_CP_DYN_PWR, WM8904_CP_DYN_PWR);
-
- /* Use normal bias source */
- snd_soc_update_bits(codec, WM8904_BIAS_CONTROL_0,
- WM8904_POBCTRL, 0);
-
- wm8904_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-
- /* Bias level configuration will have done an extra enable */
- regulator_bulk_disable(ARRAY_SIZE(wm8904->supplies), wm8904->supplies);
-
wm8904_handle_pdata(codec);
wm8904_add_widgets(codec);
return 0;
-
-err_enable:
- regulator_bulk_disable(ARRAY_SIZE(wm8904->supplies), wm8904->supplies);
-err_get:
- regulator_bulk_free(ARRAY_SIZE(wm8904->supplies), wm8904->supplies);
- return ret;
}
static int wm8904_remove(struct snd_soc_codec *codec)
{
struct wm8904_priv *wm8904 = snd_soc_codec_get_drvdata(codec);
- wm8904_set_bias_level(codec, SND_SOC_BIAS_OFF);
- regulator_bulk_free(ARRAY_SIZE(wm8904->supplies), wm8904->supplies);
kfree(wm8904->retune_mobile_texts);
kfree(wm8904->drc_texts);
@@ -2231,8 +2096,6 @@ static int wm8904_remove(struct snd_soc_codec *codec)
static struct snd_soc_codec_driver soc_codec_dev_wm8904 = {
.probe = wm8904_probe,
.remove = wm8904_remove,
- .suspend = wm8904_suspend,
- .resume = wm8904_resume,
.set_bias_level = wm8904_set_bias_level,
.idle_bias_off = true,
};
@@ -2254,14 +2117,15 @@ static __devinit int wm8904_i2c_probe(struct i2c_client *i2c,
const struct i2c_device_id *id)
{
struct wm8904_priv *wm8904;
- int ret;
+ unsigned int val;
+ int ret, i;
wm8904 = devm_kzalloc(&i2c->dev, sizeof(struct wm8904_priv),
GFP_KERNEL);
if (wm8904 == NULL)
return -ENOMEM;
- wm8904->regmap = regmap_init_i2c(i2c, &wm8904_regmap);
+ wm8904->regmap = devm_regmap_init_i2c(i2c, &wm8904_regmap);
if (IS_ERR(wm8904->regmap)) {
ret = PTR_ERR(wm8904->regmap);
dev_err(&i2c->dev, "Failed to allocate register map: %d\n",
@@ -2273,23 +2137,121 @@ static __devinit int wm8904_i2c_probe(struct i2c_client *i2c,
i2c_set_clientdata(i2c, wm8904);
wm8904->pdata = i2c->dev.platform_data;
+ for (i = 0; i < ARRAY_SIZE(wm8904->supplies); i++)
+ wm8904->supplies[i].supply = wm8904_supply_names[i];
+
+ ret = devm_regulator_bulk_get(&i2c->dev, ARRAY_SIZE(wm8904->supplies),
+ wm8904->supplies);
+ if (ret != 0) {
+ dev_err(&i2c->dev, "Failed to request supplies: %d\n", ret);
+ return ret;
+ }
+
+ ret = regulator_bulk_enable(ARRAY_SIZE(wm8904->supplies),
+ wm8904->supplies);
+ if (ret != 0) {
+ dev_err(&i2c->dev, "Failed to enable supplies: %d\n", ret);
+ return ret;
+ }
+
+ ret = regmap_read(wm8904->regmap, WM8904_SW_RESET_AND_ID, &val);
+ if (ret < 0) {
+ dev_err(&i2c->dev, "Failed to read ID register: %d\n", ret);
+ goto err_enable;
+ }
+ if (val != 0x8904) {
+ dev_err(&i2c->dev, "Device is not a WM8904, ID is %x\n", val);
+ ret = -EINVAL;
+ goto err_enable;
+ }
+
+ ret = regmap_read(wm8904->regmap, WM8904_REVISION, &val);
+ if (ret < 0) {
+ dev_err(&i2c->dev, "Failed to read device revision: %d\n",
+ ret);
+ goto err_enable;
+ }
+ dev_info(&i2c->dev, "revision %c\n", val + 'A');
+
+ ret = regmap_write(wm8904->regmap, WM8904_SW_RESET_AND_ID, 0);
+ if (ret < 0) {
+ dev_err(&i2c->dev, "Failed to issue reset: %d\n", ret);
+ goto err_enable;
+ }
+
+ /* Change some default settings - latch VU and enable ZC */
+ regmap_update_bits(wm8904->regmap, WM8904_ADC_DIGITAL_VOLUME_LEFT,
+ WM8904_ADC_VU, WM8904_ADC_VU);
+ regmap_update_bits(wm8904->regmap, WM8904_ADC_DIGITAL_VOLUME_RIGHT,
+ WM8904_ADC_VU, WM8904_ADC_VU);
+ regmap_update_bits(wm8904->regmap, WM8904_DAC_DIGITAL_VOLUME_LEFT,
+ WM8904_DAC_VU, WM8904_DAC_VU);
+ regmap_update_bits(wm8904->regmap, WM8904_DAC_DIGITAL_VOLUME_RIGHT,
+ WM8904_DAC_VU, WM8904_DAC_VU);
+ regmap_update_bits(wm8904->regmap, WM8904_ANALOGUE_OUT1_LEFT,
+ WM8904_HPOUT_VU | WM8904_HPOUTLZC,
+ WM8904_HPOUT_VU | WM8904_HPOUTLZC);
+ regmap_update_bits(wm8904->regmap, WM8904_ANALOGUE_OUT1_RIGHT,
+ WM8904_HPOUT_VU | WM8904_HPOUTRZC,
+ WM8904_HPOUT_VU | WM8904_HPOUTRZC);
+ regmap_update_bits(wm8904->regmap, WM8904_ANALOGUE_OUT2_LEFT,
+ WM8904_LINEOUT_VU | WM8904_LINEOUTLZC,
+ WM8904_LINEOUT_VU | WM8904_LINEOUTLZC);
+ regmap_update_bits(wm8904->regmap, WM8904_ANALOGUE_OUT2_RIGHT,
+ WM8904_LINEOUT_VU | WM8904_LINEOUTRZC,
+ WM8904_LINEOUT_VU | WM8904_LINEOUTRZC);
+ regmap_update_bits(wm8904->regmap, WM8904_CLOCK_RATES_0,
+ WM8904_SR_MODE, 0);
+
+ /* Apply configuration from the platform data. */
+ if (wm8904->pdata) {
+ for (i = 0; i < WM8904_GPIO_REGS; i++) {
+ if (!wm8904->pdata->gpio_cfg[i])
+ continue;
+
+ regmap_update_bits(wm8904->regmap,
+ WM8904_GPIO_CONTROL_1 + i,
+ 0xffff,
+ wm8904->pdata->gpio_cfg[i]);
+ }
+
+ /* Zero is the default value for these anyway */
+ for (i = 0; i < WM8904_MIC_REGS; i++)
+ regmap_update_bits(wm8904->regmap,
+ WM8904_MIC_BIAS_CONTROL_0 + i,
+ 0xffff,
+ wm8904->pdata->mic_cfg[i]);
+ }
+
+ /* Set Class W by default - this will be managed by the Class
+ * G widget at runtime where bypass paths are available.
+ */
+ regmap_update_bits(wm8904->regmap, WM8904_CLASS_W_0,
+ WM8904_CP_DYN_PWR, WM8904_CP_DYN_PWR);
+
+ /* Use normal bias source */
+ regmap_update_bits(wm8904->regmap, WM8904_BIAS_CONTROL_0,
+ WM8904_POBCTRL, 0);
+
+ /* Can leave the device powered off until we need it */
+ regcache_cache_only(wm8904->regmap, true);
+ regulator_bulk_disable(ARRAY_SIZE(wm8904->supplies), wm8904->supplies);
+
ret = snd_soc_register_codec(&i2c->dev,
&soc_codec_dev_wm8904, &wm8904_dai, 1);
if (ret != 0)
- goto err;
+ return ret;
return 0;
-err:
- regmap_exit(wm8904->regmap);
+err_enable:
+ regulator_bulk_disable(ARRAY_SIZE(wm8904->supplies), wm8904->supplies);
return ret;
}
static __devexit int wm8904_i2c_remove(struct i2c_client *client)
{
- struct wm8904_priv *wm8904 = i2c_get_clientdata(client);
snd_soc_unregister_codec(&client->dev);
- regmap_exit(wm8904->regmap);
return 0;
}
@@ -2311,23 +2273,7 @@ static struct i2c_driver wm8904_i2c_driver = {
.id_table = wm8904_i2c_id,
};
-static int __init wm8904_modinit(void)
-{
- int ret = 0;
- ret = i2c_add_driver(&wm8904_i2c_driver);
- if (ret != 0) {
- printk(KERN_ERR "Failed to register wm8904 I2C driver: %d\n",
- ret);
- }
- return ret;
-}
-module_init(wm8904_modinit);
-
-static void __exit wm8904_exit(void)
-{
- i2c_del_driver(&wm8904_i2c_driver);
-}
-module_exit(wm8904_exit);
+module_i2c_driver(wm8904_i2c_driver);
MODULE_DESCRIPTION("ASoC WM8904 driver");
MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>");
diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c
index 8bc659d..96518ac 100644
--- a/sound/soc/codecs/wm8960.c
+++ b/sound/soc/codecs/wm8960.c
@@ -1,6 +1,8 @@
/*
* wm8960.c -- WM8960 ALSA SoC Audio driver
*
+ * Copyright 2007-11 Wolfson Microelectronics, plc
+ *
* Author: Liam Girdwood
*
* This program is free software; you can redistribute it and/or modify
diff --git a/sound/soc/codecs/wm8961.c b/sound/soc/codecs/wm8961.c
index 05ea7c2..01edbcc 100644
--- a/sound/soc/codecs/wm8961.c
+++ b/sound/soc/codecs/wm8961.c
@@ -1,6 +1,8 @@
/*
* wm8961.c -- WM8961 ALSA SoC Audio driver
*
+ * Copyright 2009-10 Wolfson Microelectronics, plc
+ *
* Author: Mark Brown
*
* This program is free software; you can redistribute it and/or modify
diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index 0cfce99..eaf6586 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -1,7 +1,7 @@
/*
* wm8962.c -- WM8962 ALSA SoC Audio driver
*
- * Copyright 2010 Wolfson Microelectronics plc
+ * Copyright 2010-2 Wolfson Microelectronics plc
*
* Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
*
@@ -2580,6 +2580,9 @@ static int wm8962_hw_params(struct snd_pcm_substream *substream,
WM8962_SAMPLE_RATE_INT_MODE |
WM8962_SAMPLE_RATE_MASK, adctl3);
+ dev_dbg(codec->dev, "hw_params set BCLK %dHz LRCLK %dHz\n",
+ wm8962->bclk, wm8962->lrclk);
+
if (codec->dapm.bias_level == SND_SOC_BIAS_ON)
wm8962_configure_bclk(codec);
@@ -3722,6 +3725,9 @@ static int wm8962_runtime_resume(struct device *dev)
}
regcache_cache_only(wm8962->regmap, false);
+
+ wm8962_reset(wm8962);
+
regcache_sync(wm8962->regmap);
regmap_update_bits(wm8962->regmap, WM8962_ANTI_POP,
diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c
index 36acfcc..9fd80d6 100644
--- a/sound/soc/codecs/wm8993.c
+++ b/sound/soc/codecs/wm8993.c
@@ -1,7 +1,7 @@
/*
* wm8993.c -- WM8993 ALSA SoC audio driver
*
- * Copyright 2009, 2010 Wolfson Microelectronics plc
+ * Copyright 2009-12 Wolfson Microelectronics plc
*
* Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
*
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index 1436b6c..bb62f4b 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -1,7 +1,7 @@
/*
* wm8994.c -- WM8994 ALSA SoC Audio driver
*
- * Copyright 2009 Wolfson Microelectronics plc
+ * Copyright 2009-12 Wolfson Microelectronics plc
*
* Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
*
@@ -2967,23 +2967,8 @@ static struct snd_soc_dai_driver wm8994_dai[] = {
static int wm8994_codec_suspend(struct snd_soc_codec *codec)
{
struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
- struct wm8994 *control = wm8994->wm8994;
int i, ret;
- switch (control->type) {
- case WM8994:
- snd_soc_update_bits(codec, WM8994_MICBIAS, WM8994_MICD_ENA, 0);
- break;
- case WM1811:
- snd_soc_update_bits(codec, WM8994_ANTIPOP_2,
- WM1811_JACKDET_MODE_MASK, 0);
- /* Fall through */
- case WM8958:
- snd_soc_update_bits(codec, WM8958_MIC_DETECT_1,
- WM8958_MICD_ENA, 0);
- break;
- }
-
for (i = 0; i < ARRAY_SIZE(wm8994->fll); i++) {
memcpy(&wm8994->fll_suspend[i], &wm8994->fll[i],
sizeof(struct wm8994_fll_config));
@@ -3033,28 +3018,6 @@ static int wm8994_codec_resume(struct snd_soc_codec *codec)
i + 1, ret);
}
- switch (control->type) {
- case WM8994:
- if (wm8994->micdet[0].jack || wm8994->micdet[1].jack)
- snd_soc_update_bits(codec, WM8994_MICBIAS,
- WM8994_MICD_ENA, WM8994_MICD_ENA);
- break;
- case WM1811:
- if (wm8994->jackdet && wm8994->jack_cb) {
- /* Restart from idle */
- snd_soc_update_bits(codec, WM8994_ANTIPOP_2,
- WM1811_JACKDET_MODE_MASK,
- WM1811_JACKDET_MODE_JACK);
- break;
- }
- break;
- case WM8958:
- if (wm8994->jack_cb)
- snd_soc_update_bits(codec, WM8958_MIC_DETECT_1,
- WM8958_MICD_ENA, WM8958_MICD_ENA);
- break;
- }
-
return 0;
}
#else
@@ -3729,9 +3692,6 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec)
if (wm8994->pdata && wm8994->pdata->micdet_irq)
wm8994->micdet_irq = wm8994->pdata->micdet_irq;
- else if (wm8994->pdata && wm8994->pdata->irq_base)
- wm8994->micdet_irq = wm8994->pdata->irq_base +
- WM8994_IRQ_MIC1_DET;
pm_runtime_enable(codec->dev);
pm_runtime_idle(codec->dev);
@@ -3870,6 +3830,10 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec)
dev_warn(codec->dev,
"Failed to request Mic detect IRQ: %d\n",
ret);
+ } else {
+ wm8994_request_irq(wm8994->wm8994, WM8994_IRQ_MIC1_DET,
+ wm8958_mic_irq, "Mic detect",
+ wm8994);
}
}
diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c
index dc9b42b..00f183d 100644
--- a/sound/soc/codecs/wm8996.c
+++ b/sound/soc/codecs/wm8996.c
@@ -1,7 +1,7 @@
/*
* wm8996.c - WM8996 audio codec interface
*
- * Copyright 2011 Wolfson Microelectronics PLC.
+ * Copyright 2011-2 Wolfson Microelectronics PLC.
* Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
*
* This program is free software; you can redistribute it and/or modify it
@@ -296,184 +296,6 @@ static struct reg_default wm8996_reg[] = {
{ WM8996_RIGHT_PDM_SPEAKER, 0x1 },
{ WM8996_PDM_SPEAKER_MUTE_SEQUENCE, 0x69 },
{ WM8996_PDM_SPEAKER_VOLUME, 0x66 },
- { WM8996_WRITE_SEQUENCER_0, 0x1 },
- { WM8996_WRITE_SEQUENCER_1, 0x1 },
- { WM8996_WRITE_SEQUENCER_3, 0x6 },
- { WM8996_WRITE_SEQUENCER_4, 0x40 },
- { WM8996_WRITE_SEQUENCER_5, 0x1 },
- { WM8996_WRITE_SEQUENCER_6, 0xf },
- { WM8996_WRITE_SEQUENCER_7, 0x6 },
- { WM8996_WRITE_SEQUENCER_8, 0x1 },
- { WM8996_WRITE_SEQUENCER_9, 0x3 },
- { WM8996_WRITE_SEQUENCER_10, 0x104 },
- { WM8996_WRITE_SEQUENCER_12, 0x60 },
- { WM8996_WRITE_SEQUENCER_13, 0x11 },
- { WM8996_WRITE_SEQUENCER_14, 0x401 },
- { WM8996_WRITE_SEQUENCER_16, 0x50 },
- { WM8996_WRITE_SEQUENCER_17, 0x3 },
- { WM8996_WRITE_SEQUENCER_18, 0x100 },
- { WM8996_WRITE_SEQUENCER_20, 0x51 },
- { WM8996_WRITE_SEQUENCER_21, 0x3 },
- { WM8996_WRITE_SEQUENCER_22, 0x104 },
- { WM8996_WRITE_SEQUENCER_23, 0xa },
- { WM8996_WRITE_SEQUENCER_24, 0x60 },
- { WM8996_WRITE_SEQUENCER_25, 0x3b },
- { WM8996_WRITE_SEQUENCER_26, 0x502 },
- { WM8996_WRITE_SEQUENCER_27, 0x100 },
- { WM8996_WRITE_SEQUENCER_28, 0x2fff },
- { WM8996_WRITE_SEQUENCER_32, 0x2fff },
- { WM8996_WRITE_SEQUENCER_36, 0x2fff },
- { WM8996_WRITE_SEQUENCER_40, 0x2fff },
- { WM8996_WRITE_SEQUENCER_44, 0x2fff },
- { WM8996_WRITE_SEQUENCER_48, 0x2fff },
- { WM8996_WRITE_SEQUENCER_52, 0x2fff },
- { WM8996_WRITE_SEQUENCER_56, 0x2fff },
- { WM8996_WRITE_SEQUENCER_60, 0x2fff },
- { WM8996_WRITE_SEQUENCER_64, 0x1 },
- { WM8996_WRITE_SEQUENCER_65, 0x1 },
- { WM8996_WRITE_SEQUENCER_67, 0x6 },
- { WM8996_WRITE_SEQUENCER_68, 0x40 },
- { WM8996_WRITE_SEQUENCER_69, 0x1 },
- { WM8996_WRITE_SEQUENCER_70, 0xf },
- { WM8996_WRITE_SEQUENCER_71, 0x6 },
- { WM8996_WRITE_SEQUENCER_72, 0x1 },
- { WM8996_WRITE_SEQUENCER_73, 0x3 },
- { WM8996_WRITE_SEQUENCER_74, 0x104 },
- { WM8996_WRITE_SEQUENCER_76, 0x60 },
- { WM8996_WRITE_SEQUENCER_77, 0x11 },
- { WM8996_WRITE_SEQUENCER_78, 0x401 },
- { WM8996_WRITE_SEQUENCER_80, 0x50 },
- { WM8996_WRITE_SEQUENCER_81, 0x3 },
- { WM8996_WRITE_SEQUENCER_82, 0x100 },
- { WM8996_WRITE_SEQUENCER_84, 0x60 },
- { WM8996_WRITE_SEQUENCER_85, 0x3b },
- { WM8996_WRITE_SEQUENCER_86, 0x502 },
- { WM8996_WRITE_SEQUENCER_87, 0x100 },
- { WM8996_WRITE_SEQUENCER_88, 0x2fff },
- { WM8996_WRITE_SEQUENCER_92, 0x2fff },
- { WM8996_WRITE_SEQUENCER_96, 0x2fff },
- { WM8996_WRITE_SEQUENCER_100, 0x2fff },
- { WM8996_WRITE_SEQUENCER_104, 0x2fff },
- { WM8996_WRITE_SEQUENCER_108, 0x2fff },
- { WM8996_WRITE_SEQUENCER_112, 0x2fff },
- { WM8996_WRITE_SEQUENCER_116, 0x2fff },
- { WM8996_WRITE_SEQUENCER_120, 0x2fff },
- { WM8996_WRITE_SEQUENCER_124, 0x2fff },
- { WM8996_WRITE_SEQUENCER_128, 0x1 },
- { WM8996_WRITE_SEQUENCER_129, 0x1 },
- { WM8996_WRITE_SEQUENCER_131, 0x6 },
- { WM8996_WRITE_SEQUENCER_132, 0x40 },
- { WM8996_WRITE_SEQUENCER_133, 0x1 },
- { WM8996_WRITE_SEQUENCER_134, 0xf },
- { WM8996_WRITE_SEQUENCER_135, 0x6 },
- { WM8996_WRITE_SEQUENCER_136, 0x1 },
- { WM8996_WRITE_SEQUENCER_137, 0x3 },
- { WM8996_WRITE_SEQUENCER_138, 0x106 },
- { WM8996_WRITE_SEQUENCER_140, 0x61 },
- { WM8996_WRITE_SEQUENCER_141, 0x11 },
- { WM8996_WRITE_SEQUENCER_142, 0x401 },
- { WM8996_WRITE_SEQUENCER_144, 0x50 },
- { WM8996_WRITE_SEQUENCER_145, 0x3 },
- { WM8996_WRITE_SEQUENCER_146, 0x102 },
- { WM8996_WRITE_SEQUENCER_148, 0x51 },
- { WM8996_WRITE_SEQUENCER_149, 0x3 },
- { WM8996_WRITE_SEQUENCER_150, 0x106 },
- { WM8996_WRITE_SEQUENCER_151, 0xa },
- { WM8996_WRITE_SEQUENCER_152, 0x61 },
- { WM8996_WRITE_SEQUENCER_153, 0x3b },
- { WM8996_WRITE_SEQUENCER_154, 0x502 },
- { WM8996_WRITE_SEQUENCER_155, 0x100 },
- { WM8996_WRITE_SEQUENCER_156, 0x2fff },
- { WM8996_WRITE_SEQUENCER_160, 0x2fff },
- { WM8996_WRITE_SEQUENCER_164, 0x2fff },
- { WM8996_WRITE_SEQUENCER_168, 0x2fff },
- { WM8996_WRITE_SEQUENCER_172, 0x2fff },
- { WM8996_WRITE_SEQUENCER_176, 0x2fff },
- { WM8996_WRITE_SEQUENCER_180, 0x2fff },
- { WM8996_WRITE_SEQUENCER_184, 0x2fff },
- { WM8996_WRITE_SEQUENCER_188, 0x2fff },
- { WM8996_WRITE_SEQUENCER_192, 0x1 },
- { WM8996_WRITE_SEQUENCER_193, 0x1 },
- { WM8996_WRITE_SEQUENCER_195, 0x6 },
- { WM8996_WRITE_SEQUENCER_196, 0x40 },
- { WM8996_WRITE_SEQUENCER_197, 0x1 },
- { WM8996_WRITE_SEQUENCER_198, 0xf },
- { WM8996_WRITE_SEQUENCER_199, 0x6 },
- { WM8996_WRITE_SEQUENCER_200, 0x1 },
- { WM8996_WRITE_SEQUENCER_201, 0x3 },
- { WM8996_WRITE_SEQUENCER_202, 0x106 },
- { WM8996_WRITE_SEQUENCER_204, 0x61 },
- { WM8996_WRITE_SEQUENCER_205, 0x11 },
- { WM8996_WRITE_SEQUENCER_206, 0x401 },
- { WM8996_WRITE_SEQUENCER_208, 0x50 },
- { WM8996_WRITE_SEQUENCER_209, 0x3 },
- { WM8996_WRITE_SEQUENCER_210, 0x102 },
- { WM8996_WRITE_SEQUENCER_212, 0x61 },
- { WM8996_WRITE_SEQUENCER_213, 0x3b },
- { WM8996_WRITE_SEQUENCER_214, 0x502 },
- { WM8996_WRITE_SEQUENCER_215, 0x100 },
- { WM8996_WRITE_SEQUENCER_216, 0x2fff },
- { WM8996_WRITE_SEQUENCER_220, 0x2fff },
- { WM8996_WRITE_SEQUENCER_224, 0x2fff },
- { WM8996_WRITE_SEQUENCER_228, 0x2fff },
- { WM8996_WRITE_SEQUENCER_232, 0x2fff },
- { WM8996_WRITE_SEQUENCER_236, 0x2fff },
- { WM8996_WRITE_SEQUENCER_240, 0x2fff },
- { WM8996_WRITE_SEQUENCER_244, 0x2fff },
- { WM8996_WRITE_SEQUENCER_248, 0x2fff },
- { WM8996_WRITE_SEQUENCER_252, 0x2fff },
- { WM8996_WRITE_SEQUENCER_256, 0x60 },
- { WM8996_WRITE_SEQUENCER_258, 0x601 },
- { WM8996_WRITE_SEQUENCER_260, 0x50 },
- { WM8996_WRITE_SEQUENCER_262, 0x100 },
- { WM8996_WRITE_SEQUENCER_264, 0x1 },
- { WM8996_WRITE_SEQUENCER_266, 0x104 },
- { WM8996_WRITE_SEQUENCER_267, 0x100 },
- { WM8996_WRITE_SEQUENCER_268, 0x2fff },
- { WM8996_WRITE_SEQUENCER_272, 0x2fff },
- { WM8996_WRITE_SEQUENCER_276, 0x2fff },
- { WM8996_WRITE_SEQUENCER_280, 0x2fff },
- { WM8996_WRITE_SEQUENCER_284, 0x2fff },
- { WM8996_WRITE_SEQUENCER_288, 0x2fff },
- { WM8996_WRITE_SEQUENCER_292, 0x2fff },
- { WM8996_WRITE_SEQUENCER_296, 0x2fff },
- { WM8996_WRITE_SEQUENCER_300, 0x2fff },
- { WM8996_WRITE_SEQUENCER_304, 0x2fff },
- { WM8996_WRITE_SEQUENCER_308, 0x2fff },
- { WM8996_WRITE_SEQUENCER_312, 0x2fff },
- { WM8996_WRITE_SEQUENCER_316, 0x2fff },
- { WM8996_WRITE_SEQUENCER_320, 0x61 },
- { WM8996_WRITE_SEQUENCER_322, 0x601 },
- { WM8996_WRITE_SEQUENCER_324, 0x50 },
- { WM8996_WRITE_SEQUENCER_326, 0x102 },
- { WM8996_WRITE_SEQUENCER_328, 0x1 },
- { WM8996_WRITE_SEQUENCER_330, 0x106 },
- { WM8996_WRITE_SEQUENCER_331, 0x100 },
- { WM8996_WRITE_SEQUENCER_332, 0x2fff },
- { WM8996_WRITE_SEQUENCER_336, 0x2fff },
- { WM8996_WRITE_SEQUENCER_340, 0x2fff },
- { WM8996_WRITE_SEQUENCER_344, 0x2fff },
- { WM8996_WRITE_SEQUENCER_348, 0x2fff },
- { WM8996_WRITE_SEQUENCER_352, 0x2fff },
- { WM8996_WRITE_SEQUENCER_356, 0x2fff },
- { WM8996_WRITE_SEQUENCER_360, 0x2fff },
- { WM8996_WRITE_SEQUENCER_364, 0x2fff },
- { WM8996_WRITE_SEQUENCER_368, 0x2fff },
- { WM8996_WRITE_SEQUENCER_372, 0x2fff },
- { WM8996_WRITE_SEQUENCER_376, 0x2fff },
- { WM8996_WRITE_SEQUENCER_380, 0x2fff },
- { WM8996_WRITE_SEQUENCER_384, 0x60 },
- { WM8996_WRITE_SEQUENCER_386, 0x601 },
- { WM8996_WRITE_SEQUENCER_388, 0x61 },
- { WM8996_WRITE_SEQUENCER_390, 0x601 },
- { WM8996_WRITE_SEQUENCER_392, 0x50 },
- { WM8996_WRITE_SEQUENCER_394, 0x300 },
- { WM8996_WRITE_SEQUENCER_396, 0x1 },
- { WM8996_WRITE_SEQUENCER_398, 0x304 },
- { WM8996_WRITE_SEQUENCER_400, 0x40 },
- { WM8996_WRITE_SEQUENCER_402, 0xf },
- { WM8996_WRITE_SEQUENCER_404, 0x1 },
- { WM8996_WRITE_SEQUENCER_407, 0x100 },
};
static const DECLARE_TLV_DB_SCALE(inpga_tlv, 0, 100, 0);
@@ -1706,18 +1528,6 @@ static bool wm8996_volatile_register(struct device *dev, unsigned int reg)
}
}
-static int wm8996_reset(struct wm8996_priv *wm8996)
-{
- if (wm8996->pdata.ldo_ena > 0) {
- gpio_set_value_cansleep(wm8996->pdata.ldo_ena, 0);
- gpio_set_value_cansleep(wm8996->pdata.ldo_ena, 1);
- return 0;
- } else {
- return regmap_write(wm8996->regmap, WM8996_SOFTWARE_RESET,
- 0x8915);
- }
-}
-
static const int bclk_divs[] = {
1, 2, 3, 4, 6, 8, 12, 16, 24, 32, 48, 64, 96
};
@@ -1809,8 +1619,10 @@ static int wm8996_set_bias_level(struct snd_soc_codec *codec,
case SND_SOC_BIAS_OFF:
regcache_cache_only(codec->control_data, true);
- if (wm8996->pdata.ldo_ena >= 0)
+ if (wm8996->pdata.ldo_ena >= 0) {
gpio_set_value_cansleep(wm8996->pdata.ldo_ena, 0);
+ regcache_cache_only(codec->control_data, true);
+ }
regulator_bulk_disable(ARRAY_SIZE(wm8996->supplies),
wm8996->supplies);
break;
@@ -2807,7 +2619,7 @@ static int wm8996_probe(struct snd_soc_codec *codec)
int ret;
struct wm8996_priv *wm8996 = snd_soc_codec_get_drvdata(codec);
struct i2c_client *i2c = to_i2c_client(codec->dev);
- int i, irq_flags;
+ int irq_flags;
wm8996->codec = codec;
@@ -2822,177 +2634,12 @@ static int wm8996_probe(struct snd_soc_codec *codec)
goto err;
}
- wm8996->disable_nb[0].notifier_call = wm8996_regulator_event_0;
- wm8996->disable_nb[1].notifier_call = wm8996_regulator_event_1;
- wm8996->disable_nb[2].notifier_call = wm8996_regulator_event_2;
-
- /* This should really be moved into the regulator core */
- for (i = 0; i < ARRAY_SIZE(wm8996->supplies); i++) {
- ret = regulator_register_notifier(wm8996->supplies[i].consumer,
- &wm8996->disable_nb[i]);
- if (ret != 0) {
- dev_err(codec->dev,
- "Failed to register regulator notifier: %d\n",
- ret);
- }
- }
-
- /* Apply platform data settings */
- snd_soc_update_bits(codec, WM8996_LINE_INPUT_CONTROL,
- WM8996_INL_MODE_MASK | WM8996_INR_MODE_MASK,
- wm8996->pdata.inl_mode << WM8996_INL_MODE_SHIFT |
- wm8996->pdata.inr_mode);
-
- for (i = 0; i < ARRAY_SIZE(wm8996->pdata.gpio_default); i++) {
- if (!wm8996->pdata.gpio_default[i])
- continue;
-
- snd_soc_write(codec, WM8996_GPIO_1 + i,
- wm8996->pdata.gpio_default[i] & 0xffff);
- }
-
- if (wm8996->pdata.spkmute_seq)
- snd_soc_update_bits(codec, WM8996_PDM_SPEAKER_MUTE_SEQUENCE,
- WM8996_SPK_MUTE_ENDIAN |
- WM8996_SPK_MUTE_SEQ1_MASK,
- wm8996->pdata.spkmute_seq);
-
- snd_soc_update_bits(codec, WM8996_ACCESSORY_DETECT_MODE_2,
- WM8996_MICD_BIAS_SRC | WM8996_HPOUT1FB_SRC |
- WM8996_MICD_SRC, wm8996->pdata.micdet_def);
-
- /* Latch volume update bits */
- snd_soc_update_bits(codec, WM8996_LEFT_LINE_INPUT_VOLUME,
- WM8996_IN1_VU, WM8996_IN1_VU);
- snd_soc_update_bits(codec, WM8996_RIGHT_LINE_INPUT_VOLUME,
- WM8996_IN1_VU, WM8996_IN1_VU);
-
- snd_soc_update_bits(codec, WM8996_DAC1_LEFT_VOLUME,
- WM8996_DAC1_VU, WM8996_DAC1_VU);
- snd_soc_update_bits(codec, WM8996_DAC1_RIGHT_VOLUME,
- WM8996_DAC1_VU, WM8996_DAC1_VU);
- snd_soc_update_bits(codec, WM8996_DAC2_LEFT_VOLUME,
- WM8996_DAC2_VU, WM8996_DAC2_VU);
- snd_soc_update_bits(codec, WM8996_DAC2_RIGHT_VOLUME,
- WM8996_DAC2_VU, WM8996_DAC2_VU);
-
- snd_soc_update_bits(codec, WM8996_OUTPUT1_LEFT_VOLUME,
- WM8996_DAC1_VU, WM8996_DAC1_VU);
- snd_soc_update_bits(codec, WM8996_OUTPUT1_RIGHT_VOLUME,
- WM8996_DAC1_VU, WM8996_DAC1_VU);
- snd_soc_update_bits(codec, WM8996_OUTPUT2_LEFT_VOLUME,
- WM8996_DAC2_VU, WM8996_DAC2_VU);
- snd_soc_update_bits(codec, WM8996_OUTPUT2_RIGHT_VOLUME,
- WM8996_DAC2_VU, WM8996_DAC2_VU);
-
- snd_soc_update_bits(codec, WM8996_DSP1_TX_LEFT_VOLUME,
- WM8996_DSP1TX_VU, WM8996_DSP1TX_VU);
- snd_soc_update_bits(codec, WM8996_DSP1_TX_RIGHT_VOLUME,
- WM8996_DSP1TX_VU, WM8996_DSP1TX_VU);
- snd_soc_update_bits(codec, WM8996_DSP2_TX_LEFT_VOLUME,
- WM8996_DSP2TX_VU, WM8996_DSP2TX_VU);
- snd_soc_update_bits(codec, WM8996_DSP2_TX_RIGHT_VOLUME,
- WM8996_DSP2TX_VU, WM8996_DSP2TX_VU);
-
- snd_soc_update_bits(codec, WM8996_DSP1_RX_LEFT_VOLUME,
- WM8996_DSP1RX_VU, WM8996_DSP1RX_VU);
- snd_soc_update_bits(codec, WM8996_DSP1_RX_RIGHT_VOLUME,
- WM8996_DSP1RX_VU, WM8996_DSP1RX_VU);
- snd_soc_update_bits(codec, WM8996_DSP2_RX_LEFT_VOLUME,
- WM8996_DSP2RX_VU, WM8996_DSP2RX_VU);
- snd_soc_update_bits(codec, WM8996_DSP2_RX_RIGHT_VOLUME,
- WM8996_DSP2RX_VU, WM8996_DSP2RX_VU);
-
- /* No support currently for the underclocked TDM modes and
- * pick a default TDM layout with each channel pair working with
- * slots 0 and 1. */
- snd_soc_update_bits(codec, WM8996_AIF1RX_CHANNEL_0_CONFIGURATION,
- WM8996_AIF1RX_CHAN0_SLOTS_MASK |
- WM8996_AIF1RX_CHAN0_START_SLOT_MASK,
- 1 << WM8996_AIF1RX_CHAN0_SLOTS_SHIFT | 0);
- snd_soc_update_bits(codec, WM8996_AIF1RX_CHANNEL_1_CONFIGURATION,
- WM8996_AIF1RX_CHAN1_SLOTS_MASK |
- WM8996_AIF1RX_CHAN1_START_SLOT_MASK,
- 1 << WM8996_AIF1RX_CHAN1_SLOTS_SHIFT | 1);
- snd_soc_update_bits(codec, WM8996_AIF1RX_CHANNEL_2_CONFIGURATION,
- WM8996_AIF1RX_CHAN2_SLOTS_MASK |
- WM8996_AIF1RX_CHAN2_START_SLOT_MASK,
- 1 << WM8996_AIF1RX_CHAN2_SLOTS_SHIFT | 0);
- snd_soc_update_bits(codec, WM8996_AIF1RX_CHANNEL_3_CONFIGURATION,
- WM8996_AIF1RX_CHAN3_SLOTS_MASK |
- WM8996_AIF1RX_CHAN0_START_SLOT_MASK,
- 1 << WM8996_AIF1RX_CHAN3_SLOTS_SHIFT | 1);
- snd_soc_update_bits(codec, WM8996_AIF1RX_CHANNEL_4_CONFIGURATION,
- WM8996_AIF1RX_CHAN4_SLOTS_MASK |
- WM8996_AIF1RX_CHAN0_START_SLOT_MASK,
- 1 << WM8996_AIF1RX_CHAN4_SLOTS_SHIFT | 0);
- snd_soc_update_bits(codec, WM8996_AIF1RX_CHANNEL_5_CONFIGURATION,
- WM8996_AIF1RX_CHAN5_SLOTS_MASK |
- WM8996_AIF1RX_CHAN0_START_SLOT_MASK,
- 1 << WM8996_AIF1RX_CHAN5_SLOTS_SHIFT | 1);
-
- snd_soc_update_bits(codec, WM8996_AIF2RX_CHANNEL_0_CONFIGURATION,
- WM8996_AIF2RX_CHAN0_SLOTS_MASK |
- WM8996_AIF2RX_CHAN0_START_SLOT_MASK,
- 1 << WM8996_AIF2RX_CHAN0_SLOTS_SHIFT | 0);
- snd_soc_update_bits(codec, WM8996_AIF2RX_CHANNEL_1_CONFIGURATION,
- WM8996_AIF2RX_CHAN1_SLOTS_MASK |
- WM8996_AIF2RX_CHAN1_START_SLOT_MASK,
- 1 << WM8996_AIF2RX_CHAN1_SLOTS_SHIFT | 1);
-
- snd_soc_update_bits(codec, WM8996_AIF1TX_CHANNEL_0_CONFIGURATION,
- WM8996_AIF1TX_CHAN0_SLOTS_MASK |
- WM8996_AIF1TX_CHAN0_START_SLOT_MASK,
- 1 << WM8996_AIF1TX_CHAN0_SLOTS_SHIFT | 0);
- snd_soc_update_bits(codec, WM8996_AIF1TX_CHANNEL_1_CONFIGURATION,
- WM8996_AIF1TX_CHAN1_SLOTS_MASK |
- WM8996_AIF1TX_CHAN0_START_SLOT_MASK,
- 1 << WM8996_AIF1TX_CHAN1_SLOTS_SHIFT | 1);
- snd_soc_update_bits(codec, WM8996_AIF1TX_CHANNEL_2_CONFIGURATION,
- WM8996_AIF1TX_CHAN2_SLOTS_MASK |
- WM8996_AIF1TX_CHAN0_START_SLOT_MASK,
- 1 << WM8996_AIF1TX_CHAN2_SLOTS_SHIFT | 0);
- snd_soc_update_bits(codec, WM8996_AIF1TX_CHANNEL_3_CONFIGURATION,
- WM8996_AIF1TX_CHAN3_SLOTS_MASK |
- WM8996_AIF1TX_CHAN0_START_SLOT_MASK,
- 1 << WM8996_AIF1TX_CHAN3_SLOTS_SHIFT | 1);
- snd_soc_update_bits(codec, WM8996_AIF1TX_CHANNEL_4_CONFIGURATION,
- WM8996_AIF1TX_CHAN4_SLOTS_MASK |
- WM8996_AIF1TX_CHAN0_START_SLOT_MASK,
- 1 << WM8996_AIF1TX_CHAN4_SLOTS_SHIFT | 0);
- snd_soc_update_bits(codec, WM8996_AIF1TX_CHANNEL_5_CONFIGURATION,
- WM8996_AIF1TX_CHAN5_SLOTS_MASK |
- WM8996_AIF1TX_CHAN0_START_SLOT_MASK,
- 1 << WM8996_AIF1TX_CHAN5_SLOTS_SHIFT | 1);
-
- snd_soc_update_bits(codec, WM8996_AIF2TX_CHANNEL_0_CONFIGURATION,
- WM8996_AIF2TX_CHAN0_SLOTS_MASK |
- WM8996_AIF2TX_CHAN0_START_SLOT_MASK,
- 1 << WM8996_AIF2TX_CHAN0_SLOTS_SHIFT | 0);
- snd_soc_update_bits(codec, WM8996_AIF1TX_CHANNEL_1_CONFIGURATION,
- WM8996_AIF2TX_CHAN1_SLOTS_MASK |
- WM8996_AIF2TX_CHAN1_START_SLOT_MASK,
- 1 << WM8996_AIF1TX_CHAN1_SLOTS_SHIFT | 1);
-
if (wm8996->pdata.num_retune_mobile_cfgs)
wm8996_retune_mobile_pdata(codec);
else
snd_soc_add_codec_controls(codec, wm8996_eq_controls,
ARRAY_SIZE(wm8996_eq_controls));
- /* If the TX LRCLK pins are not in LRCLK mode configure the
- * AIFs to source their clocks from the RX LRCLKs.
- */
- if ((snd_soc_read(codec, WM8996_GPIO_1)))
- snd_soc_update_bits(codec, WM8996_AIF1_TX_LRCLK_2,
- WM8996_AIF1TX_LRCLK_MODE,
- WM8996_AIF1TX_LRCLK_MODE);
-
- if ((snd_soc_read(codec, WM8996_GPIO_2)))
- snd_soc_update_bits(codec, WM8996_AIF2_TX_LRCLK_2,
- WM8996_AIF2TX_LRCLK_MODE,
- WM8996_AIF2TX_LRCLK_MODE);
-
if (i2c->irq) {
if (wm8996->pdata.irq_flags)
irq_flags = wm8996->pdata.irq_flags;
@@ -3036,9 +2683,7 @@ err:
static int wm8996_remove(struct snd_soc_codec *codec)
{
- struct wm8996_priv *wm8996 = snd_soc_codec_get_drvdata(codec);
struct i2c_client *i2c = to_i2c_client(codec->dev);
- int i;
snd_soc_update_bits(codec, WM8996_INTERRUPT_CONTROL,
WM8996_IM_IRQ, WM8996_IM_IRQ);
@@ -3046,10 +2691,6 @@ static int wm8996_remove(struct snd_soc_codec *codec)
if (i2c->irq)
free_irq(i2c->irq, codec);
- for (i = 0; i < ARRAY_SIZE(wm8996->supplies); i++)
- regulator_unregister_notifier(wm8996->supplies[i].consumer,
- &wm8996->disable_nb[i]);
-
return 0;
}
@@ -3163,6 +2804,21 @@ static __devinit int wm8996_i2c_probe(struct i2c_client *i2c,
goto err_gpio;
}
+ wm8996->disable_nb[0].notifier_call = wm8996_regulator_event_0;
+ wm8996->disable_nb[1].notifier_call = wm8996_regulator_event_1;
+ wm8996->disable_nb[2].notifier_call = wm8996_regulator_event_2;
+
+ /* This should really be moved into the regulator core */
+ for (i = 0; i < ARRAY_SIZE(wm8996->supplies); i++) {
+ ret = regulator_register_notifier(wm8996->supplies[i].consumer,
+ &wm8996->disable_nb[i]);
+ if (ret != 0) {
+ dev_err(&i2c->dev,
+ "Failed to register regulator notifier: %d\n",
+ ret);
+ }
+ }
+
ret = regulator_bulk_enable(ARRAY_SIZE(wm8996->supplies),
wm8996->supplies);
if (ret != 0) {
@@ -3175,7 +2831,7 @@ static __devinit int wm8996_i2c_probe(struct i2c_client *i2c,
msleep(5);
}
- wm8996->regmap = regmap_init_i2c(i2c, &wm8996_regmap);
+ wm8996->regmap = devm_regmap_init_i2c(i2c, &wm8996_regmap);
if (IS_ERR(wm8996->regmap)) {
ret = PTR_ERR(wm8996->regmap);
dev_err(&i2c->dev, "regmap_init() failed: %d\n", ret);
@@ -3203,15 +2859,199 @@ static __devinit int wm8996_i2c_probe(struct i2c_client *i2c,
dev_info(&i2c->dev, "revision %c\n",
(reg & WM8996_CHIP_REV_MASK) + 'A');
- ret = wm8996_reset(wm8996);
- if (ret < 0) {
- dev_err(&i2c->dev, "Failed to issue reset\n");
- goto err_regmap;
+ if (wm8996->pdata.ldo_ena > 0) {
+ gpio_set_value_cansleep(wm8996->pdata.ldo_ena, 0);
+ regcache_cache_only(wm8996->regmap, true);
+ } else {
+ ret = regmap_write(wm8996->regmap, WM8996_SOFTWARE_RESET,
+ 0x8915);
+ if (ret != 0) {
+ dev_err(&i2c->dev, "Failed to issue reset: %d\n", ret);
+ goto err_regmap;
+ }
}
- regcache_cache_only(wm8996->regmap, true);
regulator_bulk_disable(ARRAY_SIZE(wm8996->supplies), wm8996->supplies);
+ /* Apply platform data settings */
+ regmap_update_bits(wm8996->regmap, WM8996_LINE_INPUT_CONTROL,
+ WM8996_INL_MODE_MASK | WM8996_INR_MODE_MASK,
+ wm8996->pdata.inl_mode << WM8996_INL_MODE_SHIFT |
+ wm8996->pdata.inr_mode);
+
+ for (i = 0; i < ARRAY_SIZE(wm8996->pdata.gpio_default); i++) {
+ if (!wm8996->pdata.gpio_default[i])
+ continue;
+
+ regmap_write(wm8996->regmap, WM8996_GPIO_1 + i,
+ wm8996->pdata.gpio_default[i] & 0xffff);
+ }
+
+ if (wm8996->pdata.spkmute_seq)
+ regmap_update_bits(wm8996->regmap,
+ WM8996_PDM_SPEAKER_MUTE_SEQUENCE,
+ WM8996_SPK_MUTE_ENDIAN |
+ WM8996_SPK_MUTE_SEQ1_MASK,
+ wm8996->pdata.spkmute_seq);
+
+ regmap_update_bits(wm8996->regmap, WM8996_ACCESSORY_DETECT_MODE_2,
+ WM8996_MICD_BIAS_SRC | WM8996_HPOUT1FB_SRC |
+ WM8996_MICD_SRC, wm8996->pdata.micdet_def);
+
+ /* Latch volume update bits */
+ regmap_update_bits(wm8996->regmap, WM8996_LEFT_LINE_INPUT_VOLUME,
+ WM8996_IN1_VU, WM8996_IN1_VU);
+ regmap_update_bits(wm8996->regmap, WM8996_RIGHT_LINE_INPUT_VOLUME,
+ WM8996_IN1_VU, WM8996_IN1_VU);
+
+ regmap_update_bits(wm8996->regmap, WM8996_DAC1_LEFT_VOLUME,
+ WM8996_DAC1_VU, WM8996_DAC1_VU);
+ regmap_update_bits(wm8996->regmap, WM8996_DAC1_RIGHT_VOLUME,
+ WM8996_DAC1_VU, WM8996_DAC1_VU);
+ regmap_update_bits(wm8996->regmap, WM8996_DAC2_LEFT_VOLUME,
+ WM8996_DAC2_VU, WM8996_DAC2_VU);
+ regmap_update_bits(wm8996->regmap, WM8996_DAC2_RIGHT_VOLUME,
+ WM8996_DAC2_VU, WM8996_DAC2_VU);
+
+ regmap_update_bits(wm8996->regmap, WM8996_OUTPUT1_LEFT_VOLUME,
+ WM8996_DAC1_VU, WM8996_DAC1_VU);
+ regmap_update_bits(wm8996->regmap, WM8996_OUTPUT1_RIGHT_VOLUME,
+ WM8996_DAC1_VU, WM8996_DAC1_VU);
+ regmap_update_bits(wm8996->regmap, WM8996_OUTPUT2_LEFT_VOLUME,
+ WM8996_DAC2_VU, WM8996_DAC2_VU);
+ regmap_update_bits(wm8996->regmap, WM8996_OUTPUT2_RIGHT_VOLUME,
+ WM8996_DAC2_VU, WM8996_DAC2_VU);
+
+ regmap_update_bits(wm8996->regmap, WM8996_DSP1_TX_LEFT_VOLUME,
+ WM8996_DSP1TX_VU, WM8996_DSP1TX_VU);
+ regmap_update_bits(wm8996->regmap, WM8996_DSP1_TX_RIGHT_VOLUME,
+ WM8996_DSP1TX_VU, WM8996_DSP1TX_VU);
+ regmap_update_bits(wm8996->regmap, WM8996_DSP2_TX_LEFT_VOLUME,
+ WM8996_DSP2TX_VU, WM8996_DSP2TX_VU);
+ regmap_update_bits(wm8996->regmap, WM8996_DSP2_TX_RIGHT_VOLUME,
+ WM8996_DSP2TX_VU, WM8996_DSP2TX_VU);
+
+ regmap_update_bits(wm8996->regmap, WM8996_DSP1_RX_LEFT_VOLUME,
+ WM8996_DSP1RX_VU, WM8996_DSP1RX_VU);
+ regmap_update_bits(wm8996->regmap, WM8996_DSP1_RX_RIGHT_VOLUME,
+ WM8996_DSP1RX_VU, WM8996_DSP1RX_VU);
+ regmap_update_bits(wm8996->regmap, WM8996_DSP2_RX_LEFT_VOLUME,
+ WM8996_DSP2RX_VU, WM8996_DSP2RX_VU);
+ regmap_update_bits(wm8996->regmap, WM8996_DSP2_RX_RIGHT_VOLUME,
+ WM8996_DSP2RX_VU, WM8996_DSP2RX_VU);
+
+ /* No support currently for the underclocked TDM modes and
+ * pick a default TDM layout with each channel pair working with
+ * slots 0 and 1. */
+ regmap_update_bits(wm8996->regmap,
+ WM8996_AIF1RX_CHANNEL_0_CONFIGURATION,
+ WM8996_AIF1RX_CHAN0_SLOTS_MASK |
+ WM8996_AIF1RX_CHAN0_START_SLOT_MASK,
+ 1 << WM8996_AIF1RX_CHAN0_SLOTS_SHIFT | 0);
+ regmap_update_bits(wm8996->regmap,
+ WM8996_AIF1RX_CHANNEL_1_CONFIGURATION,
+ WM8996_AIF1RX_CHAN1_SLOTS_MASK |
+ WM8996_AIF1RX_CHAN1_START_SLOT_MASK,
+ 1 << WM8996_AIF1RX_CHAN1_SLOTS_SHIFT | 1);
+ regmap_update_bits(wm8996->regmap,
+ WM8996_AIF1RX_CHANNEL_2_CONFIGURATION,
+ WM8996_AIF1RX_CHAN2_SLOTS_MASK |
+ WM8996_AIF1RX_CHAN2_START_SLOT_MASK,
+ 1 << WM8996_AIF1RX_CHAN2_SLOTS_SHIFT | 0);
+ regmap_update_bits(wm8996->regmap,
+ WM8996_AIF1RX_CHANNEL_3_CONFIGURATION,
+ WM8996_AIF1RX_CHAN3_SLOTS_MASK |
+ WM8996_AIF1RX_CHAN0_START_SLOT_MASK,
+ 1 << WM8996_AIF1RX_CHAN3_SLOTS_SHIFT | 1);
+ regmap_update_bits(wm8996->regmap,
+ WM8996_AIF1RX_CHANNEL_4_CONFIGURATION,
+ WM8996_AIF1RX_CHAN4_SLOTS_MASK |
+ WM8996_AIF1RX_CHAN0_START_SLOT_MASK,
+ 1 << WM8996_AIF1RX_CHAN4_SLOTS_SHIFT | 0);
+ regmap_update_bits(wm8996->regmap,
+ WM8996_AIF1RX_CHANNEL_5_CONFIGURATION,
+ WM8996_AIF1RX_CHAN5_SLOTS_MASK |
+ WM8996_AIF1RX_CHAN0_START_SLOT_MASK,
+ 1 << WM8996_AIF1RX_CHAN5_SLOTS_SHIFT | 1);
+
+ regmap_update_bits(wm8996->regmap,
+ WM8996_AIF2RX_CHANNEL_0_CONFIGURATION,
+ WM8996_AIF2RX_CHAN0_SLOTS_MASK |
+ WM8996_AIF2RX_CHAN0_START_SLOT_MASK,
+ 1 << WM8996_AIF2RX_CHAN0_SLOTS_SHIFT | 0);
+ regmap_update_bits(wm8996->regmap,
+ WM8996_AIF2RX_CHANNEL_1_CONFIGURATION,
+ WM8996_AIF2RX_CHAN1_SLOTS_MASK |
+ WM8996_AIF2RX_CHAN1_START_SLOT_MASK,
+ 1 << WM8996_AIF2RX_CHAN1_SLOTS_SHIFT | 1);
+
+ regmap_update_bits(wm8996->regmap,
+ WM8996_AIF1TX_CHANNEL_0_CONFIGURATION,
+ WM8996_AIF1TX_CHAN0_SLOTS_MASK |
+ WM8996_AIF1TX_CHAN0_START_SLOT_MASK,
+ 1 << WM8996_AIF1TX_CHAN0_SLOTS_SHIFT | 0);
+ regmap_update_bits(wm8996->regmap,
+ WM8996_AIF1TX_CHANNEL_1_CONFIGURATION,
+ WM8996_AIF1TX_CHAN1_SLOTS_MASK |
+ WM8996_AIF1TX_CHAN0_START_SLOT_MASK,
+ 1 << WM8996_AIF1TX_CHAN1_SLOTS_SHIFT | 1);
+ regmap_update_bits(wm8996->regmap,
+ WM8996_AIF1TX_CHANNEL_2_CONFIGURATION,
+ WM8996_AIF1TX_CHAN2_SLOTS_MASK |
+ WM8996_AIF1TX_CHAN0_START_SLOT_MASK,
+ 1 << WM8996_AIF1TX_CHAN2_SLOTS_SHIFT | 0);
+ regmap_update_bits(wm8996->regmap,
+ WM8996_AIF1TX_CHANNEL_3_CONFIGURATION,
+ WM8996_AIF1TX_CHAN3_SLOTS_MASK |
+ WM8996_AIF1TX_CHAN0_START_SLOT_MASK,
+ 1 << WM8996_AIF1TX_CHAN3_SLOTS_SHIFT | 1);
+ regmap_update_bits(wm8996->regmap,
+ WM8996_AIF1TX_CHANNEL_4_CONFIGURATION,
+ WM8996_AIF1TX_CHAN4_SLOTS_MASK |
+ WM8996_AIF1TX_CHAN0_START_SLOT_MASK,
+ 1 << WM8996_AIF1TX_CHAN4_SLOTS_SHIFT | 0);
+ regmap_update_bits(wm8996->regmap,
+ WM8996_AIF1TX_CHANNEL_5_CONFIGURATION,
+ WM8996_AIF1TX_CHAN5_SLOTS_MASK |
+ WM8996_AIF1TX_CHAN0_START_SLOT_MASK,
+ 1 << WM8996_AIF1TX_CHAN5_SLOTS_SHIFT | 1);
+
+ regmap_update_bits(wm8996->regmap,
+ WM8996_AIF2TX_CHANNEL_0_CONFIGURATION,
+ WM8996_AIF2TX_CHAN0_SLOTS_MASK |
+ WM8996_AIF2TX_CHAN0_START_SLOT_MASK,
+ 1 << WM8996_AIF2TX_CHAN0_SLOTS_SHIFT | 0);
+ regmap_update_bits(wm8996->regmap,
+ WM8996_AIF1TX_CHANNEL_1_CONFIGURATION,
+ WM8996_AIF2TX_CHAN1_SLOTS_MASK |
+ WM8996_AIF2TX_CHAN1_START_SLOT_MASK,
+ 1 << WM8996_AIF1TX_CHAN1_SLOTS_SHIFT | 1);
+
+ /* If the TX LRCLK pins are not in LRCLK mode configure the
+ * AIFs to source their clocks from the RX LRCLKs.
+ */
+ ret = regmap_read(wm8996->regmap, WM8996_GPIO_1, &reg);
+ if (ret != 0) {
+ dev_err(&i2c->dev, "Failed to read GPIO1: %d\n", ret);
+ goto err_regmap;
+ }
+
+ if (reg & WM8996_GP1_FN_MASK)
+ regmap_update_bits(wm8996->regmap, WM8996_AIF1_TX_LRCLK_2,
+ WM8996_AIF1TX_LRCLK_MODE,
+ WM8996_AIF1TX_LRCLK_MODE);
+
+ ret = regmap_read(wm8996->regmap, WM8996_GPIO_2, &reg);
+ if (ret != 0) {
+ dev_err(&i2c->dev, "Failed to read GPIO2: %d\n", ret);
+ goto err_regmap;
+ }
+
+ if (reg & WM8996_GP2_FN_MASK)
+ regmap_update_bits(wm8996->regmap, WM8996_AIF2_TX_LRCLK_2,
+ WM8996_AIF2TX_LRCLK_MODE,
+ WM8996_AIF2TX_LRCLK_MODE);
+
wm8996_init_gpio(wm8996);
ret = snd_soc_register_codec(&i2c->dev,
@@ -3225,7 +3065,6 @@ static __devinit int wm8996_i2c_probe(struct i2c_client *i2c,
err_gpiolib:
wm8996_free_gpio(wm8996);
err_regmap:
- regmap_exit(wm8996->regmap);
err_enable:
if (wm8996->pdata.ldo_ena > 0)
gpio_set_value_cansleep(wm8996->pdata.ldo_ena, 0);
@@ -3241,14 +3080,18 @@ err:
static __devexit int wm8996_i2c_remove(struct i2c_client *client)
{
struct wm8996_priv *wm8996 = i2c_get_clientdata(client);
+ int i;
snd_soc_unregister_codec(&client->dev);
wm8996_free_gpio(wm8996);
- regmap_exit(wm8996->regmap);
if (wm8996->pdata.ldo_ena > 0) {
gpio_set_value_cansleep(wm8996->pdata.ldo_ena, 0);
gpio_free(wm8996->pdata.ldo_ena);
}
+ for (i = 0; i < ARRAY_SIZE(wm8996->supplies); i++)
+ regulator_unregister_notifier(wm8996->supplies[i].consumer,
+ &wm8996->disable_nb[i]);
+
return 0;
}
diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c
index 9328270..2de74e1 100644
--- a/sound/soc/codecs/wm9081.c
+++ b/sound/soc/codecs/wm9081.c
@@ -3,7 +3,7 @@
*
* Author: Mark Brown
*
- * Copyright 2009 Wolfson Microelectronics plc
+ * Copyright 2009-12 Wolfson Microelectronics plc
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License version 2 as
diff --git a/sound/soc/codecs/wm9090.c b/sound/soc/codecs/wm9090.c
index 4b263b6..2c2346f 100644
--- a/sound/soc/codecs/wm9090.c
+++ b/sound/soc/codecs/wm9090.c
@@ -1,7 +1,7 @@
/*
* ALSA SoC WM9090 driver
*
- * Copyright 2009, 2010 Wolfson Microelectronics
+ * Copyright 2009-12 Wolfson Microelectronics
*
* Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
*
diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c
index a154141..099e6ec 100644
--- a/sound/soc/codecs/wm9712.c
+++ b/sound/soc/codecs/wm9712.c
@@ -1,7 +1,7 @@
/*
* wm9712.c -- ALSA Soc WM9712 codec support
*
- * Copyright 2006 Wolfson Microelectronics PLC.
+ * Copyright 2006-12 Wolfson Microelectronics PLC.
* Author: Liam Girdwood <lrg@slimlogic.co.uk>
*
* This program is free software; you can redistribute it and/or modify it
diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c
index 2d22cc7..3eb19fb 100644
--- a/sound/soc/codecs/wm9713.c
+++ b/sound/soc/codecs/wm9713.c
@@ -1,7 +1,7 @@
/*
* wm9713.c -- ALSA Soc WM9713 codec support
*
- * Copyright 2006 Wolfson Microelectronics PLC.
+ * Copyright 2006-10 Wolfson Microelectronics PLC.
* Author: Liam Girdwood <lrg@slimlogic.co.uk>
*
* This program is free software; you can redistribute it and/or modify it
diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c
index dfe957a..61baa48 100644
--- a/sound/soc/codecs/wm_hubs.c
+++ b/sound/soc/codecs/wm_hubs.c
@@ -1,7 +1,7 @@
/*
* wm_hubs.c -- WM8993/4 common code
*
- * Copyright 2009 Wolfson Microelectronics plc
+ * Copyright 2009-12 Wolfson Microelectronics plc
*
* Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
*
diff --git a/sound/soc/dwc/Kconfig b/sound/soc/dwc/Kconfig
new file mode 100644
index 0000000..e334900
--- /dev/null
+++ b/sound/soc/dwc/Kconfig
@@ -0,0 +1,9 @@
+config SND_DESIGNWARE_I2S
+ tristate "Synopsys I2S Device Driver"
+ depends on CLKDEV_LOOKUP
+ help
+ Say Y or M if you want to add support for I2S driver for
+ Synopsys desigwnware I2S device. The device supports upto
+ maximum of 8 channels each for play and record.
+
+
diff --git a/sound/soc/dwc/Makefile b/sound/soc/dwc/Makefile
new file mode 100644
index 0000000..319371f
--- /dev/null
+++ b/sound/soc/dwc/Makefile
@@ -0,0 +1,3 @@
+# SYNOPSYS Platform Support
+obj-$(CONFIG_SND_DESIGNWARE_I2S) += designware_i2s.o
+
diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c
new file mode 100644
index 0000000..1aa5130
--- /dev/null
+++ b/sound/soc/dwc/designware_i2s.c
@@ -0,0 +1,455 @@
+/*
+ * ALSA SoC Synopsys I2S Audio Layer
+ *
+ * sound/soc/spear/designware_i2s.c
+ *
+ * Copyright (C) 2010 ST Microelectronics
+ * Rajeev Kumar <rajeev-dlh.kumar@st.com>
+ *
+ * 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 <linux/clk.h>
+#include <linux/device.h>
+#include <linux/init.h>
+#include <linux/io.h>
+#include <linux/interrupt.h>
+#include <linux/module.h>
+#include <linux/slab.h>
+#include <sound/designware_i2s.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+
+/* common register for all channel */
+#define IER 0x000
+#define IRER 0x004
+#define ITER 0x008
+#define CER 0x00C
+#define CCR 0x010
+#define RXFFR 0x014
+#define TXFFR 0x018
+
+/* I2STxRxRegisters for all channels */
+#define LRBR_LTHR(x) (0x40 * x + 0x020)
+#define RRBR_RTHR(x) (0x40 * x + 0x024)
+#define RER(x) (0x40 * x + 0x028)
+#define TER(x) (0x40 * x + 0x02C)
+#define RCR(x) (0x40 * x + 0x030)
+#define TCR(x) (0x40 * x + 0x034)
+#define ISR(x) (0x40 * x + 0x038)
+#define IMR(x) (0x40 * x + 0x03C)
+#define ROR(x) (0x40 * x + 0x040)
+#define TOR(x) (0x40 * x + 0x044)
+#define RFCR(x) (0x40 * x + 0x048)
+#define TFCR(x) (0x40 * x + 0x04C)
+#define RFF(x) (0x40 * x + 0x050)
+#define TFF(x) (0x40 * x + 0x054)
+
+/* I2SCOMPRegisters */
+#define I2S_COMP_PARAM_2 0x01F0
+#define I2S_COMP_PARAM_1 0x01F4
+#define I2S_COMP_VERSION 0x01F8
+#define I2S_COMP_TYPE 0x01FC
+
+#define MAX_CHANNEL_NUM 8
+#define MIN_CHANNEL_NUM 2
+
+struct dw_i2s_dev {
+ void __iomem *i2s_base;
+ struct clk *clk;
+ int active;
+ unsigned int capability;
+ struct device *dev;
+
+ /* data related to DMA transfers b/w i2s and DMAC */
+ struct i2s_dma_data play_dma_data;
+ struct i2s_dma_data capture_dma_data;
+ struct i2s_clk_config_data config;
+ int (*i2s_clk_cfg)(struct i2s_clk_config_data *config);
+};
+
+static inline void i2s_write_reg(void __iomem *io_base, int reg, u32 val)
+{
+ writel(val, io_base + reg);
+}
+
+static inline u32 i2s_read_reg(void __iomem *io_base, int reg)
+{
+ return readl(io_base + reg);
+}
+
+static inline void i2s_disable_channels(struct dw_i2s_dev *dev, u32 stream)
+{
+ u32 i = 0;
+
+ if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ for (i = 0; i < 4; i++)
+ i2s_write_reg(dev->i2s_base, TER(i), 0);
+ } else {
+ for (i = 0; i < 4; i++)
+ i2s_write_reg(dev->i2s_base, RER(i), 0);
+ }
+}
+
+static inline void i2s_clear_irqs(struct dw_i2s_dev *dev, u32 stream)
+{
+ u32 i = 0;
+
+ if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ for (i = 0; i < 4; i++)
+ i2s_write_reg(dev->i2s_base, TOR(i), 0);
+ } else {
+ for (i = 0; i < 4; i++)
+ i2s_write_reg(dev->i2s_base, ROR(i), 0);
+ }
+}
+
+static void i2s_start(struct dw_i2s_dev *dev,
+ struct snd_pcm_substream *substream)
+{
+
+ i2s_write_reg(dev->i2s_base, IER, 1);
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ i2s_write_reg(dev->i2s_base, ITER, 1);
+ else
+ i2s_write_reg(dev->i2s_base, IRER, 1);
+
+ i2s_write_reg(dev->i2s_base, CER, 1);
+}
+
+static void i2s_stop(struct dw_i2s_dev *dev,
+ struct snd_pcm_substream *substream)
+{
+ u32 i = 0, irq;
+
+ i2s_clear_irqs(dev, substream->stream);
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ i2s_write_reg(dev->i2s_base, ITER, 0);
+
+ for (i = 0; i < 4; i++) {
+ irq = i2s_read_reg(dev->i2s_base, IMR(i));
+ i2s_write_reg(dev->i2s_base, IMR(i), irq | 0x30);
+ }
+ } else {
+ i2s_write_reg(dev->i2s_base, IRER, 0);
+
+ for (i = 0; i < 4; i++) {
+ irq = i2s_read_reg(dev->i2s_base, IMR(i));
+ i2s_write_reg(dev->i2s_base, IMR(i), irq | 0x03);
+ }
+ }
+
+ if (!dev->active) {
+ i2s_write_reg(dev->i2s_base, CER, 0);
+ i2s_write_reg(dev->i2s_base, IER, 0);
+ }
+}
+
+static int dw_i2s_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *cpu_dai)
+{
+ struct dw_i2s_dev *dev = snd_soc_dai_get_drvdata(cpu_dai);
+ struct i2s_dma_data *dma_data = NULL;
+
+ if (!(dev->capability & DWC_I2S_RECORD) &&
+ (substream->stream == SNDRV_PCM_STREAM_CAPTURE))
+ return -EINVAL;
+
+ if (!(dev->capability & DWC_I2S_PLAY) &&
+ (substream->stream == SNDRV_PCM_STREAM_PLAYBACK))
+ return -EINVAL;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ dma_data = &dev->play_dma_data;
+ else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+ dma_data = &dev->capture_dma_data;
+
+ snd_soc_dai_set_dma_data(cpu_dai, substream, (void *)dma_data);
+
+ return 0;
+}
+
+static int dw_i2s_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params, struct snd_soc_dai *dai)
+{
+ struct dw_i2s_dev *dev = snd_soc_dai_get_drvdata(dai);
+ struct i2s_clk_config_data *config = &dev->config;
+ u32 ccr, xfer_resolution, ch_reg, irq;
+ int ret;
+
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ config->data_width = 16;
+ ccr = 0x00;
+ xfer_resolution = 0x02;
+ break;
+
+ case SNDRV_PCM_FORMAT_S24_LE:
+ config->data_width = 24;
+ ccr = 0x08;
+ xfer_resolution = 0x04;
+ break;
+
+ case SNDRV_PCM_FORMAT_S32_LE:
+ config->data_width = 32;
+ ccr = 0x10;
+ xfer_resolution = 0x05;
+ break;
+
+ default:
+ dev_err(dev->dev, "designware-i2s: unsuppted PCM fmt");
+ return -EINVAL;
+ }
+
+ config->chan_nr = params_channels(params);
+
+ switch (config->chan_nr) {
+ case EIGHT_CHANNEL_SUPPORT:
+ ch_reg = 3;
+ case SIX_CHANNEL_SUPPORT:
+ ch_reg = 2;
+ case FOUR_CHANNEL_SUPPORT:
+ ch_reg = 1;
+ case TWO_CHANNEL_SUPPORT:
+ ch_reg = 0;
+ break;
+ default:
+ dev_err(dev->dev, "channel not supported\n");
+ }
+
+ i2s_disable_channels(dev, substream->stream);
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ i2s_write_reg(dev->i2s_base, TCR(ch_reg), xfer_resolution);
+ i2s_write_reg(dev->i2s_base, TFCR(ch_reg), 0x02);
+ irq = i2s_read_reg(dev->i2s_base, IMR(ch_reg));
+ i2s_write_reg(dev->i2s_base, IMR(ch_reg), irq & ~0x30);
+ i2s_write_reg(dev->i2s_base, TER(ch_reg), 1);
+ } else {
+ i2s_write_reg(dev->i2s_base, RCR(ch_reg), xfer_resolution);
+ i2s_write_reg(dev->i2s_base, RFCR(ch_reg), 0x07);
+ irq = i2s_read_reg(dev->i2s_base, IMR(ch_reg));
+ i2s_write_reg(dev->i2s_base, IMR(ch_reg), irq & ~0x03);
+ i2s_write_reg(dev->i2s_base, RER(ch_reg), 1);
+ }
+
+ i2s_write_reg(dev->i2s_base, CCR, ccr);
+
+ config->sample_rate = params_rate(params);
+
+ if (!dev->i2s_clk_cfg)
+ return -EINVAL;
+
+ ret = dev->i2s_clk_cfg(config);
+ if (ret < 0) {
+ dev_err(dev->dev, "runtime audio clk config fail\n");
+ return ret;
+ }
+
+ return 0;
+}
+
+static void dw_i2s_shutdown(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ snd_soc_dai_set_dma_data(dai, substream, NULL);
+}
+
+static int dw_i2s_trigger(struct snd_pcm_substream *substream,
+ int cmd, struct snd_soc_dai *dai)
+{
+ struct dw_i2s_dev *dev = snd_soc_dai_get_drvdata(dai);
+ int ret = 0;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ dev->active++;
+ i2s_start(dev, substream);
+ break;
+
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ dev->active--;
+ i2s_stop(dev, substream);
+ break;
+ default:
+ ret = -EINVAL;
+ break;
+ }
+ return ret;
+}
+
+static struct snd_soc_dai_ops dw_i2s_dai_ops = {
+ .startup = dw_i2s_startup,
+ .shutdown = dw_i2s_shutdown,
+ .hw_params = dw_i2s_hw_params,
+ .trigger = dw_i2s_trigger,
+};
+
+#ifdef CONFIG_PM
+
+static int dw_i2s_suspend(struct snd_soc_dai *dai)
+{
+ struct dw_i2s_dev *dev = snd_soc_dai_get_drvdata(dai);
+
+ clk_disable(dev->clk);
+ return 0;
+}
+
+static int dw_i2s_resume(struct snd_soc_dai *dai)
+{
+ struct dw_i2s_dev *dev = snd_soc_dai_get_drvdata(dai);
+
+ clk_enable(dev->clk);
+ return 0;
+}
+
+#else
+#define dw_i2s_suspend NULL
+#define dw_i2s_resume NULL
+#endif
+
+static int dw_i2s_probe(struct platform_device *pdev)
+{
+ const struct i2s_platform_data *pdata = pdev->dev.platform_data;
+ struct dw_i2s_dev *dev;
+ struct resource *res;
+ int ret;
+ unsigned int cap;
+ struct snd_soc_dai_driver *dw_i2s_dai;
+
+ if (!pdata) {
+ dev_err(&pdev->dev, "Invalid platform data\n");
+ return -EINVAL;
+ }
+
+ res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+ if (!res) {
+ dev_err(&pdev->dev, "no i2s resource defined\n");
+ return -ENODEV;
+ }
+
+ if (!devm_request_mem_region(&pdev->dev, res->start,
+ resource_size(res), pdev->name)) {
+ dev_err(&pdev->dev, "i2s region already claimed\n");
+ return -EBUSY;
+ }
+
+ dev = devm_kzalloc(&pdev->dev, sizeof(*dev), GFP_KERNEL);
+ if (!dev) {
+ dev_warn(&pdev->dev, "kzalloc fail\n");
+ return -ENOMEM;
+ }
+
+ dev->i2s_base = devm_ioremap(&pdev->dev, res->start,
+ resource_size(res));
+ if (!dev->i2s_base) {
+ dev_err(&pdev->dev, "ioremap fail for i2s_region\n");
+ return -ENOMEM;
+ }
+
+ cap = pdata->cap;
+ dev->capability = cap;
+ dev->i2s_clk_cfg = pdata->i2s_clk_cfg;
+
+ /* Set DMA slaves info */
+
+ dev->play_dma_data.data = pdata->play_dma_data;
+ dev->capture_dma_data.data = pdata->capture_dma_data;
+ dev->play_dma_data.addr = res->start + I2S_TXDMA;
+ dev->capture_dma_data.addr = res->start + I2S_RXDMA;
+ dev->play_dma_data.max_burst = 16;
+ dev->capture_dma_data.max_burst = 16;
+ dev->play_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES;
+ dev->capture_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES;
+ dev->play_dma_data.filter = pdata->filter;
+ dev->capture_dma_data.filter = pdata->filter;
+
+ dev->clk = clk_get(&pdev->dev, NULL);
+ if (IS_ERR(dev->clk))
+ return PTR_ERR(dev->clk);
+
+ ret = clk_enable(dev->clk);
+ if (ret < 0)
+ goto err_clk_put;
+
+ dw_i2s_dai = devm_kzalloc(&pdev->dev, sizeof(*dw_i2s_dai), GFP_KERNEL);
+ if (!dw_i2s_dai) {
+ dev_err(&pdev->dev, "mem allocation failed for dai driver\n");
+ ret = -ENOMEM;
+ goto err_clk_disable;
+ }
+
+ if (cap & DWC_I2S_PLAY) {
+ dev_dbg(&pdev->dev, " SPEAr: play supported\n");
+ dw_i2s_dai->playback.channels_min = MIN_CHANNEL_NUM;
+ dw_i2s_dai->playback.channels_max = pdata->channel;
+ dw_i2s_dai->playback.formats = pdata->snd_fmts;
+ dw_i2s_dai->playback.rates = pdata->snd_rates;
+ }
+
+ if (cap & DWC_I2S_RECORD) {
+ dev_dbg(&pdev->dev, "SPEAr: record supported\n");
+ dw_i2s_dai->capture.channels_min = MIN_CHANNEL_NUM;
+ dw_i2s_dai->capture.channels_max = pdata->channel;
+ dw_i2s_dai->capture.formats = pdata->snd_fmts;
+ dw_i2s_dai->capture.rates = pdata->snd_rates;
+ }
+
+ dw_i2s_dai->ops = &dw_i2s_dai_ops;
+ dw_i2s_dai->suspend = dw_i2s_suspend;
+ dw_i2s_dai->resume = dw_i2s_resume;
+
+ dev->dev = &pdev->dev;
+ dev_set_drvdata(&pdev->dev, dev);
+ ret = snd_soc_register_dai(&pdev->dev, dw_i2s_dai);
+ if (ret != 0) {
+ dev_err(&pdev->dev, "not able to register dai\n");
+ goto err_set_drvdata;
+ }
+
+ return 0;
+
+err_set_drvdata:
+ dev_set_drvdata(&pdev->dev, NULL);
+err_clk_disable:
+ clk_disable(dev->clk);
+err_clk_put:
+ clk_put(dev->clk);
+ return ret;
+}
+
+static int dw_i2s_remove(struct platform_device *pdev)
+{
+ struct dw_i2s_dev *dev = dev_get_drvdata(&pdev->dev);
+
+ snd_soc_unregister_dai(&pdev->dev);
+ dev_set_drvdata(&pdev->dev, NULL);
+
+ clk_put(dev->clk);
+
+ return 0;
+}
+
+static struct platform_driver dw_i2s_driver = {
+ .probe = dw_i2s_probe,
+ .remove = dw_i2s_remove,
+ .driver = {
+ .name = "designware-i2s",
+ .owner = THIS_MODULE,
+ },
+};
+
+module_platform_driver(dw_i2s_driver);
+
+MODULE_AUTHOR("Rajeev Kumar <rajeev-dlh.kumar@st.com>");
+MODULE_DESCRIPTION("DESIGNWARE I2S SoC Interface");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:designware_i2s");
diff --git a/sound/soc/ep93xx/ep93xx-pcm.c b/sound/soc/ep93xx/ep93xx-pcm.c
index 162dbb7..4eea98b 100644
--- a/sound/soc/ep93xx/ep93xx-pcm.c
+++ b/sound/soc/ep93xx/ep93xx-pcm.c
@@ -136,7 +136,7 @@ static struct snd_pcm_ops ep93xx_pcm_ops = {
.hw_params = ep93xx_pcm_hw_params,
.hw_free = ep93xx_pcm_hw_free,
.trigger = snd_dmaengine_pcm_trigger,
- .pointer = snd_dmaengine_pcm_pointer,
+ .pointer = snd_dmaengine_pcm_pointer_no_residue,
.mmap = ep93xx_pcm_mmap,
};
diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c
index 0803274..e7c800e 100644
--- a/sound/soc/fsl/imx-audmux.c
+++ b/sound/soc/fsl/imx-audmux.c
@@ -156,7 +156,7 @@ static void __init audmux_debugfs_init(void)
return;
}
- for (i = 0; i < MX31_AUDMUX_PORT6_SSI_PINS_6 + 1; i++) {
+ for (i = 0; i < MX31_AUDMUX_PORT7_SSI_PINS_7 + 1; i++) {
snprintf(buf, sizeof(buf), "ssi%d", i);
if (!debugfs_create_file(buf, 0444, audmux_debugfs_root,
(void *)i, &audmux_debugfs_fops))
diff --git a/sound/soc/fsl/imx-audmux.h b/sound/soc/fsl/imx-audmux.h
index 04ebbab..b8ff44b 100644
--- a/sound/soc/fsl/imx-audmux.h
+++ b/sound/soc/fsl/imx-audmux.h
@@ -14,6 +14,7 @@
#define MX31_AUDMUX_PORT4_SSI_PINS_4 3
#define MX31_AUDMUX_PORT5_SSI_PINS_5 4
#define MX31_AUDMUX_PORT6_SSI_PINS_6 5
+#define MX31_AUDMUX_PORT7_SSI_PINS_7 6
#define MX51_AUDMUX_PORT1_SSI0 0
#define MX51_AUDMUX_PORT2_SSI1 1
diff --git a/sound/soc/fsl/imx-mc13783.c b/sound/soc/fsl/imx-mc13783.c
index f59c349..549b31f 100644
--- a/sound/soc/fsl/imx-mc13783.c
+++ b/sound/soc/fsl/imx-mc13783.c
@@ -111,22 +111,39 @@ static int __devinit imx_mc13783_probe(struct platform_device *pdev)
return ret;
}
- imx_audmux_v2_configure_port(MX31_AUDMUX_PORT4_SSI_PINS_4,
- IMX_AUDMUX_V2_PTCR_SYN,
- IMX_AUDMUX_V2_PDCR_RXDSEL(MX31_AUDMUX_PORT1_SSI0) |
- IMX_AUDMUX_V2_PDCR_MODE(1) |
- IMX_AUDMUX_V2_PDCR_INMMASK(0xfc));
- imx_audmux_v2_configure_port(MX31_AUDMUX_PORT1_SSI0,
- IMX_AUDMUX_V2_PTCR_SYN |
- IMX_AUDMUX_V2_PTCR_TFSDIR |
- IMX_AUDMUX_V2_PTCR_TFSEL(MX31_AUDMUX_PORT4_SSI_PINS_4) |
- IMX_AUDMUX_V2_PTCR_TCLKDIR |
- IMX_AUDMUX_V2_PTCR_TCSEL(MX31_AUDMUX_PORT4_SSI_PINS_4) |
- IMX_AUDMUX_V2_PTCR_RFSDIR |
- IMX_AUDMUX_V2_PTCR_RFSEL(MX31_AUDMUX_PORT4_SSI_PINS_4) |
- IMX_AUDMUX_V2_PTCR_RCLKDIR |
- IMX_AUDMUX_V2_PTCR_RCSEL(MX31_AUDMUX_PORT4_SSI_PINS_4),
- IMX_AUDMUX_V2_PDCR_RXDSEL(MX31_AUDMUX_PORT4_SSI_PINS_4));
+ if (machine_is_mx31_3ds()) {
+ imx_audmux_v2_configure_port(MX31_AUDMUX_PORT4_SSI_PINS_4,
+ IMX_AUDMUX_V2_PTCR_SYN,
+ IMX_AUDMUX_V2_PDCR_RXDSEL(MX31_AUDMUX_PORT1_SSI0) |
+ IMX_AUDMUX_V2_PDCR_MODE(1) |
+ IMX_AUDMUX_V2_PDCR_INMMASK(0xfc));
+ imx_audmux_v2_configure_port(MX31_AUDMUX_PORT1_SSI0,
+ IMX_AUDMUX_V2_PTCR_SYN |
+ IMX_AUDMUX_V2_PTCR_TFSDIR |
+ IMX_AUDMUX_V2_PTCR_TFSEL(MX31_AUDMUX_PORT4_SSI_PINS_4) |
+ IMX_AUDMUX_V2_PTCR_TCLKDIR |
+ IMX_AUDMUX_V2_PTCR_TCSEL(MX31_AUDMUX_PORT4_SSI_PINS_4) |
+ IMX_AUDMUX_V2_PTCR_RFSDIR |
+ IMX_AUDMUX_V2_PTCR_RFSEL(MX31_AUDMUX_PORT4_SSI_PINS_4) |
+ IMX_AUDMUX_V2_PTCR_RCLKDIR |
+ IMX_AUDMUX_V2_PTCR_RCSEL(MX31_AUDMUX_PORT4_SSI_PINS_4),
+ IMX_AUDMUX_V2_PDCR_RXDSEL(MX31_AUDMUX_PORT4_SSI_PINS_4));
+ } else if (machine_is_mx27_3ds()) {
+ imx_audmux_v1_configure_port(MX27_AUDMUX_HPCR1_SSI0,
+ IMX_AUDMUX_V1_PCR_SYN |
+ IMX_AUDMUX_V1_PCR_TFSDIR |
+ IMX_AUDMUX_V1_PCR_TCLKDIR |
+ IMX_AUDMUX_V1_PCR_RFSDIR |
+ IMX_AUDMUX_V1_PCR_RCLKDIR |
+ IMX_AUDMUX_V1_PCR_TFCSEL(MX27_AUDMUX_HPCR3_SSI_PINS_4) |
+ IMX_AUDMUX_V1_PCR_RFCSEL(MX27_AUDMUX_HPCR3_SSI_PINS_4) |
+ IMX_AUDMUX_V1_PCR_RXDSEL(MX27_AUDMUX_HPCR3_SSI_PINS_4)
+ );
+ imx_audmux_v1_configure_port(MX27_AUDMUX_HPCR3_SSI_PINS_4,
+ IMX_AUDMUX_V1_PCR_SYN |
+ IMX_AUDMUX_V1_PCR_RXDSEL(MX27_AUDMUX_HPCR1_SSI0)
+ );
+ }
return ret;
}
diff --git a/sound/soc/fsl/imx-pcm-dma.c b/sound/soc/fsl/imx-pcm-dma.c
index f3c0a5e..48f9d88 100644
--- a/sound/soc/fsl/imx-pcm-dma.c
+++ b/sound/soc/fsl/imx-pcm-dma.c
@@ -141,7 +141,7 @@ static struct snd_pcm_ops imx_pcm_ops = {
.ioctl = snd_pcm_lib_ioctl,
.hw_params = snd_imx_pcm_hw_params,
.trigger = snd_dmaengine_pcm_trigger,
- .pointer = snd_dmaengine_pcm_pointer,
+ .pointer = snd_dmaengine_pcm_pointer_no_residue,
.mmap = snd_imx_pcm_mmap,
};
diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c
index 3a729ca..fb21b17 100644
--- a/sound/soc/fsl/imx-sgtl5000.c
+++ b/sound/soc/fsl/imx-sgtl5000.c
@@ -95,8 +95,7 @@ static int __devinit imx_sgtl5000_probe(struct platform_device *pdev)
return ret;
}
imx_audmux_v2_configure_port(ext_port,
- IMX_AUDMUX_V2_PTCR_SYN |
- IMX_AUDMUX_V2_PTCR_TCSEL(int_port),
+ IMX_AUDMUX_V2_PTCR_SYN,
IMX_AUDMUX_V2_PDCR_RXDSEL(int_port));
if (ret) {
dev_err(&pdev->dev, "audmux external port setup failed\n");
diff --git a/sound/soc/mxs/mxs-pcm.c b/sound/soc/mxs/mxs-pcm.c
index 373dec9..f82d766 100644
--- a/sound/soc/mxs/mxs-pcm.c
+++ b/sound/soc/mxs/mxs-pcm.c
@@ -141,7 +141,7 @@ static struct snd_pcm_ops mxs_pcm_ops = {
.ioctl = snd_pcm_lib_ioctl,
.hw_params = snd_mxs_pcm_hw_params,
.trigger = snd_dmaengine_pcm_trigger,
- .pointer = snd_dmaengine_pcm_pointer,
+ .pointer = snd_dmaengine_pcm_pointer_no_residue,
.mmap = snd_mxs_pcm_mmap,
};
diff --git a/sound/soc/mxs/mxs-sgtl5000.c b/sound/soc/mxs/mxs-sgtl5000.c
index 3e6e876..215113b 100644
--- a/sound/soc/mxs/mxs-sgtl5000.c
+++ b/sound/soc/mxs/mxs-sgtl5000.c
@@ -133,7 +133,7 @@ static int __devinit mxs_sgtl5000_probe_dt(struct platform_device *pdev)
mxs_sgtl5000_dai[i].codec_name = NULL;
mxs_sgtl5000_dai[i].codec_of_node = codec_np;
mxs_sgtl5000_dai[i].cpu_dai_name = NULL;
- mxs_sgtl5000_dai[i].cpu_dai_of_node = saif_np[i];
+ mxs_sgtl5000_dai[i].cpu_of_node = saif_np[i];
mxs_sgtl5000_dai[i].platform_name = NULL;
mxs_sgtl5000_dai[i].platform_of_node = saif_np[i];
}
diff --git a/sound/soc/omap/omap-mcpdm.c b/sound/soc/omap/omap-mcpdm.c
index 59d47ab5..2c66e249 100644
--- a/sound/soc/omap/omap-mcpdm.c
+++ b/sound/soc/omap/omap-mcpdm.c
@@ -527,6 +527,7 @@ static struct platform_driver asoc_mcpdm_driver = {
module_platform_driver(asoc_mcpdm_driver);
+MODULE_ALIAS("platform:omap-mcpdm");
MODULE_AUTHOR("Misael Lopez Cruz <misael.lopez@ti.com>");
MODULE_DESCRIPTION("OMAP PDM SoC Interface");
MODULE_LICENSE("GPL");
diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig
index a0f7d3c..4d2e46f 100644
--- a/sound/soc/pxa/Kconfig
+++ b/sound/soc/pxa/Kconfig
@@ -8,6 +8,15 @@ config SND_PXA2XX_SOC
the PXA2xx AC97, I2S or SSP interface. You will also need
to select the audio interfaces to support below.
+config SND_MMP_SOC
+ bool "Soc Audio for Marvell MMP chips"
+ depends on ARCH_MMP
+ select SND_SOC_DMAENGINE_PCM
+ select SND_ARM
+ help
+ Say Y if you want to add support for codecs attached to
+ the MMP SSPA interface.
+
config SND_PXA2XX_AC97
tristate
select SND_AC97_CODEC
@@ -26,6 +35,9 @@ config SND_PXA_SOC_SSP
tristate
select PXA_SSP
+config SND_MMP_SOC_SSPA
+ tristate
+
config SND_PXA2XX_SOC_CORGI
tristate "SoC Audio support for Sharp Zaurus SL-C7x0"
depends on SND_PXA2XX_SOC && PXA_SHARP_C7xx
@@ -138,6 +150,26 @@ config SND_SOC_TAVOREVB3
Say Y if you want to add support for SoC audio on the
Marvell Saarb reference platform.
+config SND_PXA910_SOC
+ tristate "SoC Audio for Marvell PXA910 chip"
+ depends on ARCH_MMP && SND
+ select SND_PCM
+ help
+ Say Y if you want to add support for SoC audio on the
+ Marvell PXA910 reference platform.
+
+config SND_SOC_TTC_DKB
+ bool "SoC Audio support for TTC DKB"
+ depends on SND_PXA910_SOC && MACH_TTC_DKB
+ select PXA_SSP
+ select SND_PXA_SOC_SSP
+ select SND_MMP_SOC
+ select MFD_88PM860X
+ select SND_SOC_88PM860X
+ help
+ Say Y if you want to add support for SoC audio on TTC DKB
+
+
config SND_SOC_ZYLONITE
tristate "SoC Audio support for Marvell Zylonite"
depends on SND_PXA2XX_SOC && MACH_ZYLONITE
@@ -194,3 +226,13 @@ config SND_PXA2XX_SOC_IMOTE2
help
Say Y if you want to add support for SoC audio on the
IMote 2.
+
+config SND_MMP_SOC_BROWNSTONE
+ tristate "SoC Audio support for Marvell Brownstone"
+ depends on SND_MMP_SOC && MACH_BROWNSTONE
+ select SND_MMP_SOC_SSPA
+ select MFD_WM8994
+ select SND_SOC_WM8994
+ help
+ Say Y if you want to add support for SoC audio on the
+ Marvell Brownstone reference platform.
diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile
index af35762..d8a265d 100644
--- a/sound/soc/pxa/Makefile
+++ b/sound/soc/pxa/Makefile
@@ -3,11 +3,15 @@ snd-soc-pxa2xx-objs := pxa2xx-pcm.o
snd-soc-pxa2xx-ac97-objs := pxa2xx-ac97.o
snd-soc-pxa2xx-i2s-objs := pxa2xx-i2s.o
snd-soc-pxa-ssp-objs := pxa-ssp.o
+snd-soc-mmp-objs := mmp-pcm.o
+snd-soc-mmp-sspa-objs := mmp-sspa.o
obj-$(CONFIG_SND_PXA2XX_SOC) += snd-soc-pxa2xx.o
obj-$(CONFIG_SND_PXA2XX_SOC_AC97) += snd-soc-pxa2xx-ac97.o
obj-$(CONFIG_SND_PXA2XX_SOC_I2S) += snd-soc-pxa2xx-i2s.o
obj-$(CONFIG_SND_PXA_SOC_SSP) += snd-soc-pxa-ssp.o
+obj-$(CONFIG_SND_MMP_SOC) += snd-soc-mmp.o
+obj-$(CONFIG_SND_MMP_SOC_SSPA) += snd-soc-mmp-sspa.o
# PXA Machine Support
snd-soc-corgi-objs := corgi.o
@@ -28,6 +32,8 @@ snd-soc-mioa701-objs := mioa701_wm9713.o
snd-soc-z2-objs := z2.o
snd-soc-imote2-objs := imote2.o
snd-soc-raumfeld-objs := raumfeld.o
+snd-soc-brownstone-objs := brownstone.o
+snd-soc-ttc-dkb-objs := ttc-dkb.o
obj-$(CONFIG_SND_PXA2XX_SOC_CORGI) += snd-soc-corgi.o
obj-$(CONFIG_SND_PXA2XX_SOC_POODLE) += snd-soc-poodle.o
@@ -47,3 +53,5 @@ obj-$(CONFIG_SND_SOC_TAVOREVB3) += snd-soc-tavorevb3.o
obj-$(CONFIG_SND_SOC_ZYLONITE) += snd-soc-zylonite.o
obj-$(CONFIG_SND_PXA2XX_SOC_IMOTE2) += snd-soc-imote2.o
obj-$(CONFIG_SND_SOC_RAUMFELD) += snd-soc-raumfeld.o
+obj-$(CONFIG_SND_MMP_SOC_BROWNSTONE) += snd-soc-brownstone.o
+obj-$(CONFIG_SND_SOC_TTC_DKB) += snd-soc-ttc-dkb.o
diff --git a/sound/soc/pxa/brownstone.c b/sound/soc/pxa/brownstone.c
new file mode 100644
index 0000000..5e666e0
--- /dev/null
+++ b/sound/soc/pxa/brownstone.c
@@ -0,0 +1,174 @@
+/*
+ * linux/sound/soc/pxa/brownstone.c
+ *
+ * Copyright (C) 2011 Marvell International Ltd.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ */
+
+#include <linux/module.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/jack.h>
+
+#include "../codecs/wm8994.h"
+#include "mmp-sspa.h"
+
+static const struct snd_kcontrol_new brownstone_dapm_control[] = {
+ SOC_DAPM_PIN_SWITCH("Ext Spk"),
+};
+
+static const struct snd_soc_dapm_widget brownstone_dapm_widgets[] = {
+ SND_SOC_DAPM_SPK("Ext Spk", NULL),
+ SND_SOC_DAPM_HP("Headset Stereophone", NULL),
+ SND_SOC_DAPM_MIC("Headset Mic", NULL),
+ SND_SOC_DAPM_MIC("Main Mic", NULL),
+};
+
+static const struct snd_soc_dapm_route brownstone_audio_map[] = {
+ {"Ext Spk", NULL, "SPKOUTLP"},
+ {"Ext Spk", NULL, "SPKOUTLN"},
+ {"Ext Spk", NULL, "SPKOUTRP"},
+ {"Ext Spk", NULL, "SPKOUTRN"},
+
+ {"Headset Stereophone", NULL, "HPOUT1L"},
+ {"Headset Stereophone", NULL, "HPOUT1R"},
+
+ {"IN1RN", NULL, "Headset Mic"},
+
+ {"DMIC1DAT", NULL, "MICBIAS1"},
+ {"MICBIAS1", NULL, "Main Mic"},
+};
+
+static int brownstone_wm8994_init(struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
+
+ snd_soc_dapm_enable_pin(dapm, "Ext Spk");
+ snd_soc_dapm_enable_pin(dapm, "Headset Stereophone");
+ snd_soc_dapm_enable_pin(dapm, "Headset Mic");
+ snd_soc_dapm_enable_pin(dapm, "Main Mic");
+
+ /* set endpoints to not connected */
+ snd_soc_dapm_nc_pin(dapm, "HPOUT2P");
+ snd_soc_dapm_nc_pin(dapm, "HPOUT2N");
+ snd_soc_dapm_nc_pin(dapm, "LINEOUT1N");
+ snd_soc_dapm_nc_pin(dapm, "LINEOUT1P");
+ snd_soc_dapm_nc_pin(dapm, "LINEOUT2N");
+ snd_soc_dapm_nc_pin(dapm, "LINEOUT2P");
+ snd_soc_dapm_nc_pin(dapm, "IN1LN");
+ snd_soc_dapm_nc_pin(dapm, "IN1LP");
+ snd_soc_dapm_nc_pin(dapm, "IN1RP");
+ snd_soc_dapm_nc_pin(dapm, "IN2LP:VXRN");
+ snd_soc_dapm_nc_pin(dapm, "IN2RN");
+ snd_soc_dapm_nc_pin(dapm, "IN2RP:VXRP");
+ snd_soc_dapm_nc_pin(dapm, "IN2LN");
+
+ snd_soc_dapm_sync(dapm);
+
+ return 0;
+}
+
+static int brownstone_wm8994_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ int freq_out, sspa_mclk, sysclk;
+ int sspa_div;
+
+ if (params_rate(params) > 11025) {
+ freq_out = params_rate(params) * 512;
+ sysclk = params_rate(params) * 256;
+ sspa_mclk = params_rate(params) * 64;
+ } else {
+ freq_out = params_rate(params) * 1024;
+ sysclk = params_rate(params) * 512;
+ sspa_mclk = params_rate(params) * 64;
+ }
+ sspa_div = freq_out;
+ do_div(sspa_div, sspa_mclk);
+
+ snd_soc_dai_set_sysclk(cpu_dai, MMP_SSPA_CLK_AUDIO, freq_out, 0);
+ snd_soc_dai_set_pll(cpu_dai, MMP_SYSCLK, 0, freq_out, sysclk);
+ snd_soc_dai_set_pll(cpu_dai, MMP_SSPA_CLK, 0, freq_out, sspa_mclk);
+
+ /* set wm8994 sysclk */
+ snd_soc_dai_set_sysclk(codec_dai, WM8994_SYSCLK_MCLK1, sysclk, 0);
+
+ return 0;
+}
+
+/* machine stream operations */
+static struct snd_soc_ops brownstone_ops = {
+ .hw_params = brownstone_wm8994_hw_params,
+};
+
+static struct snd_soc_dai_link brownstone_wm8994_dai[] = {
+{
+ .name = "WM8994",
+ .stream_name = "WM8994 HiFi",
+ .cpu_dai_name = "mmp-sspa-dai.0",
+ .codec_dai_name = "wm8994-aif1",
+ .platform_name = "mmp-pcm-audio",
+ .codec_name = "wm8994-codec",
+ .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBS_CFS,
+ .ops = &brownstone_ops,
+ .init = brownstone_wm8994_init,
+},
+};
+
+/* audio machine driver */
+static struct snd_soc_card brownstone = {
+ .name = "brownstone",
+ .dai_link = brownstone_wm8994_dai,
+ .num_links = ARRAY_SIZE(brownstone_wm8994_dai),
+
+ .controls = brownstone_dapm_control,
+ .num_controls = ARRAY_SIZE(brownstone_dapm_control),
+ .dapm_widgets = brownstone_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(brownstone_dapm_widgets),
+ .dapm_routes = brownstone_audio_map,
+ .num_dapm_routes = ARRAY_SIZE(brownstone_audio_map),
+};
+
+static int __devinit brownstone_probe(struct platform_device *pdev)
+{
+ int ret;
+
+ brownstone.dev = &pdev->dev;
+ ret = snd_soc_register_card(&brownstone);
+ if (ret)
+ dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
+ ret);
+ return ret;
+}
+
+static int __devexit brownstone_remove(struct platform_device *pdev)
+{
+ snd_soc_unregister_card(&brownstone);
+ return 0;
+}
+
+static struct platform_driver mmp_driver = {
+ .driver = {
+ .name = "brownstone-audio",
+ .owner = THIS_MODULE,
+ },
+ .probe = brownstone_probe,
+ .remove = __devexit_p(brownstone_remove),
+};
+
+module_platform_driver(mmp_driver);
+
+MODULE_AUTHOR("Leo Yan <leoy@marvell.com>");
+MODULE_DESCRIPTION("ALSA SoC Brownstone");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/pxa/mioa701_wm9713.c b/sound/soc/pxa/mioa701_wm9713.c
index 9c585af..8687c1c 100644
--- a/sound/soc/pxa/mioa701_wm9713.c
+++ b/sound/soc/pxa/mioa701_wm9713.c
@@ -186,36 +186,27 @@ static struct snd_soc_card mioa701 = {
.num_links = ARRAY_SIZE(mioa701_dai),
};
-static struct platform_device *mioa701_snd_device;
-
-static int mioa701_wm9713_probe(struct platform_device *pdev)
+static int __devinit mioa701_wm9713_probe(struct platform_device *pdev)
{
- int ret;
+ int rc;
if (!machine_is_mioa701())
return -ENODEV;
- dev_warn(&pdev->dev, "Be warned that incorrect mixers/muxes setup will"
- "lead to overheating and possible destruction of your device."
- "Do not use without a good knowledge of mio's board design!\n");
-
- mioa701_snd_device = platform_device_alloc("soc-audio", -1);
- if (!mioa701_snd_device)
- return -ENOMEM;
-
- platform_set_drvdata(mioa701_snd_device, &mioa701);
-
- ret = platform_device_add(mioa701_snd_device);
- if (!ret)
- return 0;
-
- platform_device_put(mioa701_snd_device);
- return ret;
+ mioa701.dev = &pdev->dev;
+ rc = snd_soc_register_card(&mioa701);
+ if (!rc)
+ dev_warn(&pdev->dev, "Be warned that incorrect mixers/muxes setup will"
+ "lead to overheating and possible destruction of your device."
+ " Do not use without a good knowledge of mio's board design!\n");
+ return rc;
}
static int __devexit mioa701_wm9713_remove(struct platform_device *pdev)
{
- platform_device_unregister(mioa701_snd_device);
+ struct snd_soc_card *card = platform_get_drvdata(pdev);
+
+ snd_soc_unregister_card(card);
return 0;
}
diff --git a/sound/soc/pxa/mmp-pcm.c b/sound/soc/pxa/mmp-pcm.c
new file mode 100644
index 0000000..73ac546
--- /dev/null
+++ b/sound/soc/pxa/mmp-pcm.c
@@ -0,0 +1,297 @@
+/*
+ * linux/sound/soc/pxa/mmp-pcm.c
+ *
+ * Copyright (C) 2011 Marvell International Ltd.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ */
+#include <linux/module.h>
+#include <linux/init.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+#include <linux/dma-mapping.h>
+#include <linux/dmaengine.h>
+#include <linux/platform_data/mmp_audio.h>
+#include <sound/pxa2xx-lib.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <mach/sram.h>
+#include <sound/dmaengine_pcm.h>
+
+struct mmp_dma_data {
+ int ssp_id;
+ struct resource *dma_res;
+};
+
+#define MMP_PCM_INFO (SNDRV_PCM_INFO_MMAP | \
+ SNDRV_PCM_INFO_MMAP_VALID | \
+ SNDRV_PCM_INFO_INTERLEAVED | \
+ SNDRV_PCM_INFO_PAUSE | \
+ SNDRV_PCM_INFO_RESUME)
+
+#define MMP_PCM_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \
+ SNDRV_PCM_FMTBIT_S24_LE | \
+ SNDRV_PCM_FMTBIT_S32_LE)
+
+static struct snd_pcm_hardware mmp_pcm_hardware[] = {
+ {
+ .info = MMP_PCM_INFO,
+ .formats = MMP_PCM_FORMATS,
+ .period_bytes_min = 1024,
+ .period_bytes_max = 2048,
+ .periods_min = 2,
+ .periods_max = 32,
+ .buffer_bytes_max = 4096,
+ .fifo_size = 32,
+ },
+ {
+ .info = MMP_PCM_INFO,
+ .formats = MMP_PCM_FORMATS,
+ .period_bytes_min = 1024,
+ .period_bytes_max = 2048,
+ .periods_min = 2,
+ .periods_max = 32,
+ .buffer_bytes_max = 4096,
+ .fifo_size = 32,
+ },
+};
+
+static int mmp_pcm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct dma_chan *chan = snd_dmaengine_pcm_get_chan(substream);
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct pxa2xx_pcm_dma_params *dma_params;
+ struct dma_slave_config slave_config;
+ int ret;
+
+ dma_params = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
+ if (!dma_params)
+ return 0;
+
+ ret = snd_hwparams_to_dma_slave_config(substream, params, &slave_config);
+ if (ret)
+ return ret;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ slave_config.dst_addr = dma_params->dev_addr;
+ slave_config.dst_maxburst = 4;
+ } else {
+ slave_config.src_addr = dma_params->dev_addr;
+ slave_config.src_maxburst = 4;
+ }
+
+ ret = dmaengine_slave_config(chan, &slave_config);
+ if (ret)
+ return ret;
+
+ snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
+
+ return 0;
+}
+
+static bool filter(struct dma_chan *chan, void *param)
+{
+ struct mmp_dma_data *dma_data = param;
+ bool found = false;
+ char *devname;
+
+ devname = kasprintf(GFP_KERNEL, "%s.%d", dma_data->dma_res->name,
+ dma_data->ssp_id);
+ if ((strcmp(dev_name(chan->device->dev), devname) == 0) &&
+ (chan->chan_id == dma_data->dma_res->start)) {
+ found = true;
+ }
+
+ kfree(devname);
+ return found;
+}
+
+static int mmp_pcm_open(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct platform_device *pdev = to_platform_device(rtd->platform->dev);
+ struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct mmp_dma_data *dma_data;
+ struct resource *r;
+ int ret;
+
+ r = platform_get_resource(pdev, IORESOURCE_DMA, substream->stream);
+ if (!r)
+ return -EBUSY;
+
+ snd_soc_set_runtime_hwparams(substream,
+ &mmp_pcm_hardware[substream->stream]);
+ dma_data = devm_kzalloc(&pdev->dev,
+ sizeof(struct mmp_dma_data), GFP_KERNEL);
+ if (dma_data == NULL)
+ return -ENOMEM;
+
+ dma_data->dma_res = r;
+ dma_data->ssp_id = cpu_dai->id;
+
+ ret = snd_dmaengine_pcm_open(substream, filter, dma_data);
+ if (ret) {
+ devm_kfree(&pdev->dev, dma_data);
+ return ret;
+ }
+
+ snd_dmaengine_pcm_set_data(substream, dma_data);
+ return 0;
+}
+
+static int mmp_pcm_close(struct snd_pcm_substream *substream)
+{
+ struct mmp_dma_data *dma_data = snd_dmaengine_pcm_get_data(substream);
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct platform_device *pdev = to_platform_device(rtd->platform->dev);
+
+ snd_dmaengine_pcm_close(substream);
+ devm_kfree(&pdev->dev, dma_data);
+ return 0;
+}
+
+static int mmp_pcm_mmap(struct snd_pcm_substream *substream,
+ struct vm_area_struct *vma)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ unsigned long off = vma->vm_pgoff;
+
+ vma->vm_page_prot = pgprot_noncached(vma->vm_page_prot);
+ return remap_pfn_range(vma, vma->vm_start,
+ __phys_to_pfn(runtime->dma_addr) + off,
+ vma->vm_end - vma->vm_start, vma->vm_page_prot);
+}
+
+struct snd_pcm_ops mmp_pcm_ops = {
+ .open = mmp_pcm_open,
+ .close = mmp_pcm_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = mmp_pcm_hw_params,
+ .trigger = snd_dmaengine_pcm_trigger,
+ .pointer = snd_dmaengine_pcm_pointer,
+ .mmap = mmp_pcm_mmap,
+};
+
+static void mmp_pcm_free_dma_buffers(struct snd_pcm *pcm)
+{
+ struct snd_pcm_substream *substream;
+ struct snd_dma_buffer *buf;
+ int stream;
+ struct gen_pool *gpool;
+
+ gpool = sram_get_gpool("asram");
+ if (!gpool)
+ return;
+
+ for (stream = 0; stream < 2; stream++) {
+ size_t size = mmp_pcm_hardware[stream].buffer_bytes_max;
+
+ substream = pcm->streams[stream].substream;
+ if (!substream)
+ continue;
+
+ buf = &substream->dma_buffer;
+ if (!buf->area)
+ continue;
+ gen_pool_free(gpool, (unsigned long)buf->area, size);
+ buf->area = NULL;
+ }
+
+ return;
+}
+
+static int mmp_pcm_preallocate_dma_buffer(struct snd_pcm_substream *substream,
+ int stream)
+{
+ struct snd_dma_buffer *buf = &substream->dma_buffer;
+ size_t size = mmp_pcm_hardware[stream].buffer_bytes_max;
+ struct gen_pool *gpool;
+
+ buf->dev.type = SNDRV_DMA_TYPE_DEV;
+ buf->dev.dev = substream->pcm->card->dev;
+ buf->private_data = NULL;
+
+ gpool = sram_get_gpool("asram");
+ if (!gpool)
+ return -ENOMEM;
+
+ buf->area = (unsigned char *)gen_pool_alloc(gpool, size);
+ if (!buf->area)
+ return -ENOMEM;
+ buf->addr = gen_pool_virt_to_phys(gpool, (unsigned long)buf->area);
+ buf->bytes = size;
+ return 0;
+}
+
+int mmp_pcm_new(struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_pcm_substream *substream;
+ struct snd_pcm *pcm = rtd->pcm;
+ int ret = 0, stream;
+
+ for (stream = 0; stream < 2; stream++) {
+ substream = pcm->streams[stream].substream;
+
+ ret = mmp_pcm_preallocate_dma_buffer(substream, stream);
+ if (ret)
+ goto err;
+ }
+
+ return 0;
+
+err:
+ mmp_pcm_free_dma_buffers(pcm);
+ return ret;
+}
+
+struct snd_soc_platform_driver mmp_soc_platform = {
+ .ops = &mmp_pcm_ops,
+ .pcm_new = mmp_pcm_new,
+ .pcm_free = mmp_pcm_free_dma_buffers,
+};
+
+static __devinit int mmp_pcm_probe(struct platform_device *pdev)
+{
+ struct mmp_audio_platdata *pdata = pdev->dev.platform_data;
+
+ if (pdata) {
+ mmp_pcm_hardware[SNDRV_PCM_STREAM_PLAYBACK].buffer_bytes_max =
+ pdata->buffer_max_playback;
+ mmp_pcm_hardware[SNDRV_PCM_STREAM_PLAYBACK].period_bytes_max =
+ pdata->period_max_playback;
+ mmp_pcm_hardware[SNDRV_PCM_STREAM_CAPTURE].buffer_bytes_max =
+ pdata->buffer_max_capture;
+ mmp_pcm_hardware[SNDRV_PCM_STREAM_CAPTURE].period_bytes_max =
+ pdata->period_max_capture;
+ }
+ return snd_soc_register_platform(&pdev->dev, &mmp_soc_platform);
+}
+
+static int __devexit mmp_pcm_remove(struct platform_device *pdev)
+{
+ snd_soc_unregister_platform(&pdev->dev);
+ return 0;
+}
+
+static struct platform_driver mmp_pcm_driver = {
+ .driver = {
+ .name = "mmp-pcm-audio",
+ .owner = THIS_MODULE,
+ },
+
+ .probe = mmp_pcm_probe,
+ .remove = __devexit_p(mmp_pcm_remove),
+};
+
+module_platform_driver(mmp_pcm_driver);
+
+MODULE_AUTHOR("Leo Yan <leoy@marvell.com>");
+MODULE_DESCRIPTION("MMP Soc Audio DMA module");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/pxa/mmp-sspa.c b/sound/soc/pxa/mmp-sspa.c
new file mode 100644
index 0000000..4d6cb8a
--- /dev/null
+++ b/sound/soc/pxa/mmp-sspa.c
@@ -0,0 +1,480 @@
+/*
+ * linux/sound/soc/pxa/mmp-sspa.c
+ * Base on pxa2xx-ssp.c
+ *
+ * Copyright (C) 2011 Marvell International Ltd.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <linux/delay.h>
+#include <linux/clk.h>
+#include <linux/slab.h>
+#include <linux/pxa2xx_ssp.h>
+#include <linux/io.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/initval.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/pxa2xx-lib.h>
+#include "mmp-sspa.h"
+
+/*
+ * SSPA audio private data
+ */
+struct sspa_priv {
+ struct ssp_device *sspa;
+ struct pxa2xx_pcm_dma_params *dma_params;
+ struct clk *audio_clk;
+ struct clk *sysclk;
+ int dai_fmt;
+ int running_cnt;
+};
+
+static void mmp_sspa_write_reg(struct ssp_device *sspa, u32 reg, u32 val)
+{
+ __raw_writel(val, sspa->mmio_base + reg);
+}
+
+static u32 mmp_sspa_read_reg(struct ssp_device *sspa, u32 reg)
+{
+ return __raw_readl(sspa->mmio_base + reg);
+}
+
+static void mmp_sspa_tx_enable(struct ssp_device *sspa)
+{
+ unsigned int sspa_sp;
+
+ sspa_sp = mmp_sspa_read_reg(sspa, SSPA_TXSP);
+ sspa_sp |= SSPA_SP_S_EN;
+ sspa_sp |= SSPA_SP_WEN;
+ mmp_sspa_write_reg(sspa, SSPA_TXSP, sspa_sp);
+}
+
+static void mmp_sspa_tx_disable(struct ssp_device *sspa)
+{
+ unsigned int sspa_sp;
+
+ sspa_sp = mmp_sspa_read_reg(sspa, SSPA_TXSP);
+ sspa_sp &= ~SSPA_SP_S_EN;
+ sspa_sp |= SSPA_SP_WEN;
+ mmp_sspa_write_reg(sspa, SSPA_TXSP, sspa_sp);
+}
+
+static void mmp_sspa_rx_enable(struct ssp_device *sspa)
+{
+ unsigned int sspa_sp;
+
+ sspa_sp = mmp_sspa_read_reg(sspa, SSPA_RXSP);
+ sspa_sp |= SSPA_SP_S_EN;
+ sspa_sp |= SSPA_SP_WEN;
+ mmp_sspa_write_reg(sspa, SSPA_RXSP, sspa_sp);
+}
+
+static void mmp_sspa_rx_disable(struct ssp_device *sspa)
+{
+ unsigned int sspa_sp;
+
+ sspa_sp = mmp_sspa_read_reg(sspa, SSPA_RXSP);
+ sspa_sp &= ~SSPA_SP_S_EN;
+ sspa_sp |= SSPA_SP_WEN;
+ mmp_sspa_write_reg(sspa, SSPA_RXSP, sspa_sp);
+}
+
+static int mmp_sspa_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct sspa_priv *priv = snd_soc_dai_get_drvdata(dai);
+
+ clk_enable(priv->sysclk);
+ clk_enable(priv->sspa->clk);
+
+ return 0;
+}
+
+static void mmp_sspa_shutdown(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct sspa_priv *priv = snd_soc_dai_get_drvdata(dai);
+
+ clk_disable(priv->sspa->clk);
+ clk_disable(priv->sysclk);
+
+ return;
+}
+
+/*
+ * Set the SSP ports SYSCLK.
+ */
+static int mmp_sspa_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
+ int clk_id, unsigned int freq, int dir)
+{
+ struct sspa_priv *priv = snd_soc_dai_get_drvdata(cpu_dai);
+ int ret = 0;
+
+ switch (clk_id) {
+ case MMP_SSPA_CLK_AUDIO:
+ ret = clk_set_rate(priv->audio_clk, freq);
+ if (ret)
+ return ret;
+ break;
+ case MMP_SSPA_CLK_PLL:
+ case MMP_SSPA_CLK_VCXO:
+ /* not support yet */
+ return -EINVAL;
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static int mmp_sspa_set_dai_pll(struct snd_soc_dai *cpu_dai, int pll_id,
+ int source, unsigned int freq_in,
+ unsigned int freq_out)
+{
+ struct sspa_priv *priv = snd_soc_dai_get_drvdata(cpu_dai);
+ int ret = 0;
+
+ switch (pll_id) {
+ case MMP_SYSCLK:
+ ret = clk_set_rate(priv->sysclk, freq_out);
+ if (ret)
+ return ret;
+ break;
+ case MMP_SSPA_CLK:
+ ret = clk_set_rate(priv->sspa->clk, freq_out);
+ if (ret)
+ return ret;
+ break;
+ default:
+ return -ENODEV;
+ }
+
+ return 0;
+}
+
+/*
+ * Set up the sspa dai format. The sspa port must be inactive
+ * before calling this function as the physical
+ * interface format is changed.
+ */
+static int mmp_sspa_set_dai_fmt(struct snd_soc_dai *cpu_dai,
+ unsigned int fmt)
+{
+ struct sspa_priv *sspa_priv = snd_soc_dai_get_drvdata(cpu_dai);
+ struct ssp_device *sspa = sspa_priv->sspa;
+ u32 sspa_sp, sspa_ctrl;
+
+ /* check if we need to change anything at all */
+ if (sspa_priv->dai_fmt == fmt)
+ return 0;
+
+ /* we can only change the settings if the port is not in use */
+ if ((mmp_sspa_read_reg(sspa, SSPA_TXSP) & SSPA_SP_S_EN) ||
+ (mmp_sspa_read_reg(sspa, SSPA_RXSP) & SSPA_SP_S_EN)) {
+ dev_err(&sspa->pdev->dev,
+ "can't change hardware dai format: stream is in use\n");
+ return -EINVAL;
+ }
+
+ /* reset port settings */
+ sspa_sp = SSPA_SP_WEN | SSPA_SP_S_RST | SSPA_SP_FFLUSH;
+ sspa_ctrl = 0;
+
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBS_CFS:
+ sspa_sp |= SSPA_SP_MSL;
+ break;
+ case SND_SOC_DAIFMT_CBM_CFM:
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_NB_NF:
+ sspa_sp |= SSPA_SP_FSP;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ sspa_sp |= SSPA_TXSP_FPER(63);
+ sspa_sp |= SSPA_SP_FWID(31);
+ sspa_ctrl |= SSPA_CTL_XDATDLY(1);
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ mmp_sspa_write_reg(sspa, SSPA_TXSP, sspa_sp);
+ mmp_sspa_write_reg(sspa, SSPA_RXSP, sspa_sp);
+
+ sspa_sp &= ~(SSPA_SP_S_RST | SSPA_SP_FFLUSH);
+ mmp_sspa_write_reg(sspa, SSPA_TXSP, sspa_sp);
+ mmp_sspa_write_reg(sspa, SSPA_RXSP, sspa_sp);
+
+ /*
+ * FIXME: hw issue, for the tx serial port,
+ * can not config the master/slave mode;
+ * so must clean this bit.
+ * The master/slave mode has been set in the
+ * rx port.
+ */
+ sspa_sp &= ~SSPA_SP_MSL;
+ mmp_sspa_write_reg(sspa, SSPA_TXSP, sspa_sp);
+
+ mmp_sspa_write_reg(sspa, SSPA_TXCTL, sspa_ctrl);
+ mmp_sspa_write_reg(sspa, SSPA_RXCTL, sspa_ctrl);
+
+ /* Since we are configuring the timings for the format by hand
+ * we have to defer some things until hw_params() where we
+ * know parameters like the sample size.
+ */
+ sspa_priv->dai_fmt = fmt;
+ return 0;
+}
+
+/*
+ * Set the SSPA audio DMA parameters and sample size.
+ * Can be called multiple times by oss emulation.
+ */
+static int mmp_sspa_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct sspa_priv *sspa_priv = snd_soc_dai_get_drvdata(dai);
+ struct ssp_device *sspa = sspa_priv->sspa;
+ struct pxa2xx_pcm_dma_params *dma_params;
+ u32 sspa_ctrl;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ sspa_ctrl = mmp_sspa_read_reg(sspa, SSPA_TXCTL);
+ else
+ sspa_ctrl = mmp_sspa_read_reg(sspa, SSPA_RXCTL);
+
+ sspa_ctrl &= ~SSPA_CTL_XFRLEN1_MASK;
+ sspa_ctrl |= SSPA_CTL_XFRLEN1(params_channels(params) - 1);
+ sspa_ctrl &= ~SSPA_CTL_XWDLEN1_MASK;
+ sspa_ctrl |= SSPA_CTL_XWDLEN1(SSPA_CTL_32_BITS);
+ sspa_ctrl &= ~SSPA_CTL_XSSZ1_MASK;
+
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S8:
+ sspa_ctrl |= SSPA_CTL_XSSZ1(SSPA_CTL_8_BITS);
+ break;
+ case SNDRV_PCM_FORMAT_S16_LE:
+ sspa_ctrl |= SSPA_CTL_XSSZ1(SSPA_CTL_16_BITS);
+ break;
+ case SNDRV_PCM_FORMAT_S20_3LE:
+ sspa_ctrl |= SSPA_CTL_XSSZ1(SSPA_CTL_20_BITS);
+ break;
+ case SNDRV_PCM_FORMAT_S24_3LE:
+ sspa_ctrl |= SSPA_CTL_XSSZ1(SSPA_CTL_24_BITS);
+ break;
+ case SNDRV_PCM_FORMAT_S32_LE:
+ sspa_ctrl |= SSPA_CTL_XSSZ1(SSPA_CTL_32_BITS);
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ mmp_sspa_write_reg(sspa, SSPA_TXCTL, sspa_ctrl);
+ mmp_sspa_write_reg(sspa, SSPA_TXFIFO_LL, 0x1);
+ } else {
+ mmp_sspa_write_reg(sspa, SSPA_RXCTL, sspa_ctrl);
+ mmp_sspa_write_reg(sspa, SSPA_RXFIFO_UL, 0x0);
+ }
+
+ dma_params = &sspa_priv->dma_params[substream->stream];
+ dma_params->dev_addr = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ?
+ (sspa->phys_base + SSPA_TXD) :
+ (sspa->phys_base + SSPA_RXD);
+ snd_soc_dai_set_dma_data(cpu_dai, substream, dma_params);
+ return 0;
+}
+
+static int mmp_sspa_trigger(struct snd_pcm_substream *substream, int cmd,
+ struct snd_soc_dai *dai)
+{
+ struct sspa_priv *sspa_priv = snd_soc_dai_get_drvdata(dai);
+ struct ssp_device *sspa = sspa_priv->sspa;
+ int ret = 0;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ /*
+ * whatever playback or capture, must enable rx.
+ * this is a hw issue, so need check if rx has been
+ * enabled or not; if has been enabled by another
+ * stream, do not enable again.
+ */
+ if (!sspa_priv->running_cnt)
+ mmp_sspa_rx_enable(sspa);
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ mmp_sspa_tx_enable(sspa);
+
+ sspa_priv->running_cnt++;
+ break;
+
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ sspa_priv->running_cnt--;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ mmp_sspa_tx_disable(sspa);
+
+ /* have no capture stream, disable rx port */
+ if (!sspa_priv->running_cnt)
+ mmp_sspa_rx_disable(sspa);
+ break;
+
+ default:
+ ret = -EINVAL;
+ }
+
+ return ret;
+}
+
+static int mmp_sspa_probe(struct snd_soc_dai *dai)
+{
+ struct sspa_priv *priv = dev_get_drvdata(dai->dev);
+
+ snd_soc_dai_set_drvdata(dai, priv);
+ return 0;
+
+}
+
+#define MMP_SSPA_RATES SNDRV_PCM_RATE_8000_192000
+#define MMP_SSPA_FORMATS (SNDRV_PCM_FMTBIT_S8 | \
+ SNDRV_PCM_FMTBIT_S16_LE | \
+ SNDRV_PCM_FMTBIT_S24_LE | \
+ SNDRV_PCM_FMTBIT_S24_LE | \
+ SNDRV_PCM_FMTBIT_S32_LE)
+
+static struct snd_soc_dai_ops mmp_sspa_dai_ops = {
+ .startup = mmp_sspa_startup,
+ .shutdown = mmp_sspa_shutdown,
+ .trigger = mmp_sspa_trigger,
+ .hw_params = mmp_sspa_hw_params,
+ .set_sysclk = mmp_sspa_set_dai_sysclk,
+ .set_pll = mmp_sspa_set_dai_pll,
+ .set_fmt = mmp_sspa_set_dai_fmt,
+};
+
+struct snd_soc_dai_driver mmp_sspa_dai = {
+ .probe = mmp_sspa_probe,
+ .playback = {
+ .channels_min = 1,
+ .channels_max = 128,
+ .rates = MMP_SSPA_RATES,
+ .formats = MMP_SSPA_FORMATS,
+ },
+ .capture = {
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = MMP_SSPA_RATES,
+ .formats = MMP_SSPA_FORMATS,
+ },
+ .ops = &mmp_sspa_dai_ops,
+};
+
+static __devinit int asoc_mmp_sspa_probe(struct platform_device *pdev)
+{
+ struct sspa_priv *priv;
+ struct resource *res;
+
+ priv = devm_kzalloc(&pdev->dev,
+ sizeof(struct sspa_priv), GFP_KERNEL);
+ if (!priv)
+ return -ENOMEM;
+
+ priv->sspa = devm_kzalloc(&pdev->dev,
+ sizeof(struct ssp_device), GFP_KERNEL);
+ if (priv->sspa == NULL)
+ return -ENOMEM;
+
+ priv->dma_params = devm_kzalloc(&pdev->dev,
+ 2 * sizeof(struct pxa2xx_pcm_dma_params), GFP_KERNEL);
+ if (priv->dma_params == NULL)
+ return -ENOMEM;
+
+ res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+ if (res == NULL)
+ return -ENOMEM;
+
+ priv->sspa->mmio_base = devm_request_and_ioremap(&pdev->dev, res);
+ if (priv->sspa->mmio_base == NULL)
+ return -ENODEV;
+
+ priv->sspa->clk = devm_clk_get(&pdev->dev, NULL);
+ if (IS_ERR(priv->sspa->clk))
+ return PTR_ERR(priv->sspa->clk);
+
+ priv->audio_clk = clk_get(NULL, "mmp-audio");
+ if (IS_ERR(priv->audio_clk))
+ return PTR_ERR(priv->audio_clk);
+
+ priv->sysclk = clk_get(NULL, "mmp-sysclk");
+ if (IS_ERR(priv->sysclk)) {
+ clk_put(priv->audio_clk);
+ return PTR_ERR(priv->sysclk);
+ }
+ clk_enable(priv->audio_clk);
+ priv->dai_fmt = (unsigned int) -1;
+ platform_set_drvdata(pdev, priv);
+
+ return snd_soc_register_dai(&pdev->dev, &mmp_sspa_dai);
+}
+
+static int __devexit asoc_mmp_sspa_remove(struct platform_device *pdev)
+{
+ struct sspa_priv *priv = platform_get_drvdata(pdev);
+
+ clk_disable(priv->audio_clk);
+ clk_put(priv->audio_clk);
+ clk_put(priv->sysclk);
+ snd_soc_unregister_dai(&pdev->dev);
+ return 0;
+}
+
+static struct platform_driver asoc_mmp_sspa_driver = {
+ .driver = {
+ .name = "mmp-sspa-dai",
+ .owner = THIS_MODULE,
+ },
+ .probe = asoc_mmp_sspa_probe,
+ .remove = __devexit_p(asoc_mmp_sspa_remove),
+};
+
+module_platform_driver(asoc_mmp_sspa_driver);
+
+MODULE_AUTHOR("Leo Yan <leoy@marvell.com>");
+MODULE_DESCRIPTION("MMP SSPA SoC Interface");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/pxa/mmp-sspa.h b/sound/soc/pxa/mmp-sspa.h
new file mode 100644
index 0000000..ea365cb
--- /dev/null
+++ b/sound/soc/pxa/mmp-sspa.h
@@ -0,0 +1,92 @@
+/*
+ * linux/sound/soc/pxa/mmp-sspa.h
+ *
+ * Copyright (C) 2011 Marvell International Ltd.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+#ifndef _MMP_SSPA_H
+#define _MMP_SSPA_H
+
+/*
+ * SSPA Registers
+ */
+#define SSPA_RXD (0x00)
+#define SSPA_RXID (0x04)
+#define SSPA_RXCTL (0x08)
+#define SSPA_RXSP (0x0c)
+#define SSPA_RXFIFO_UL (0x10)
+#define SSPA_RXINT_MASK (0x14)
+#define SSPA_RXC (0x18)
+#define SSPA_RXFIFO_NOFS (0x1c)
+#define SSPA_RXFIFO_SIZE (0x20)
+
+#define SSPA_TXD (0x80)
+#define SSPA_TXID (0x84)
+#define SSPA_TXCTL (0x88)
+#define SSPA_TXSP (0x8c)
+#define SSPA_TXFIFO_LL (0x90)
+#define SSPA_TXINT_MASK (0x94)
+#define SSPA_TXC (0x98)
+#define SSPA_TXFIFO_NOFS (0x9c)
+#define SSPA_TXFIFO_SIZE (0xa0)
+
+/* SSPA Control Register */
+#define SSPA_CTL_XPH (1 << 31) /* Read Phase */
+#define SSPA_CTL_XFIG (1 << 15) /* Transmit Zeros when FIFO Empty */
+#define SSPA_CTL_JST (1 << 3) /* Audio Sample Justification */
+#define SSPA_CTL_XFRLEN2_MASK (7 << 24)
+#define SSPA_CTL_XFRLEN2(x) ((x) << 24) /* Transmit Frame Length in Phase 2 */
+#define SSPA_CTL_XWDLEN2_MASK (7 << 21)
+#define SSPA_CTL_XWDLEN2(x) ((x) << 21) /* Transmit Word Length in Phase 2 */
+#define SSPA_CTL_XDATDLY(x) ((x) << 19) /* Tansmit Data Delay */
+#define SSPA_CTL_XSSZ2_MASK (7 << 16)
+#define SSPA_CTL_XSSZ2(x) ((x) << 16) /* Transmit Sample Audio Size */
+#define SSPA_CTL_XFRLEN1_MASK (7 << 8)
+#define SSPA_CTL_XFRLEN1(x) ((x) << 8) /* Transmit Frame Length in Phase 1 */
+#define SSPA_CTL_XWDLEN1_MASK (7 << 5)
+#define SSPA_CTL_XWDLEN1(x) ((x) << 5) /* Transmit Word Length in Phase 1 */
+#define SSPA_CTL_XSSZ1_MASK (7 << 0)
+#define SSPA_CTL_XSSZ1(x) ((x) << 0) /* XSSZ1 */
+
+#define SSPA_CTL_8_BITS (0x0) /* Sample Size */
+#define SSPA_CTL_12_BITS (0x1)
+#define SSPA_CTL_16_BITS (0x2)
+#define SSPA_CTL_20_BITS (0x3)
+#define SSPA_CTL_24_BITS (0x4)
+#define SSPA_CTL_32_BITS (0x5)
+
+/* SSPA Serial Port Register */
+#define SSPA_SP_WEN (1 << 31) /* Write Configuration Enable */
+#define SSPA_SP_MSL (1 << 18) /* Master Slave Configuration */
+#define SSPA_SP_CLKP (1 << 17) /* CLKP Polarity Clock Edge Select */
+#define SSPA_SP_FSP (1 << 16) /* FSP Polarity Clock Edge Select */
+#define SSPA_SP_FFLUSH (1 << 2) /* FIFO Flush */
+#define SSPA_SP_S_RST (1 << 1) /* Active High Reset Signal */
+#define SSPA_SP_S_EN (1 << 0) /* Serial Clock Domain Enable */
+#define SSPA_SP_FWID(x) ((x) << 20) /* Frame-Sync Width */
+#define SSPA_TXSP_FPER(x) ((x) << 4) /* Frame-Sync Active */
+
+/* sspa clock sources */
+#define MMP_SSPA_CLK_PLL 0
+#define MMP_SSPA_CLK_VCXO 1
+#define MMP_SSPA_CLK_AUDIO 3
+
+/* sspa pll id */
+#define MMP_SYSCLK 0
+#define MMP_SSPA_CLK 1
+
+#endif /* _MMP_SSPA_H */
diff --git a/sound/soc/pxa/ttc-dkb.c b/sound/soc/pxa/ttc-dkb.c
new file mode 100644
index 0000000..935491a
--- /dev/null
+++ b/sound/soc/pxa/ttc-dkb.c
@@ -0,0 +1,173 @@
+/*
+ * linux/sound/soc/pxa/ttc_dkb.c
+ *
+ * Copyright (C) 2012 Marvell International Ltd.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/jack.h>
+#include <asm/mach-types.h>
+#include <sound/pcm_params.h>
+#include "../codecs/88pm860x-codec.h"
+
+static struct snd_soc_jack hs_jack, mic_jack;
+
+static struct snd_soc_jack_pin hs_jack_pins[] = {
+ { .pin = "Headset Stereophone", .mask = SND_JACK_HEADPHONE, },
+};
+
+static struct snd_soc_jack_pin mic_jack_pins[] = {
+ { .pin = "Headset Mic 2", .mask = SND_JACK_MICROPHONE, },
+};
+
+/* ttc machine dapm widgets */
+static const struct snd_soc_dapm_widget ttc_dapm_widgets[] = {
+ SND_SOC_DAPM_HP("Headset Stereophone", NULL),
+ SND_SOC_DAPM_LINE("Lineout Out 1", NULL),
+ SND_SOC_DAPM_LINE("Lineout Out 2", NULL),
+ SND_SOC_DAPM_SPK("Ext Speaker", NULL),
+ SND_SOC_DAPM_MIC("Ext Mic 1", NULL),
+ SND_SOC_DAPM_MIC("Headset Mic 2", NULL),
+ SND_SOC_DAPM_MIC("Ext Mic 3", NULL),
+};
+
+/* ttc machine audio map */
+static const struct snd_soc_dapm_route ttc_audio_map[] = {
+ {"Headset Stereophone", NULL, "HS1"},
+ {"Headset Stereophone", NULL, "HS2"},
+
+ {"Ext Speaker", NULL, "LSP"},
+ {"Ext Speaker", NULL, "LSN"},
+
+ {"Lineout Out 1", NULL, "LINEOUT1"},
+ {"Lineout Out 2", NULL, "LINEOUT2"},
+
+ {"MIC1P", NULL, "Mic1 Bias"},
+ {"MIC1N", NULL, "Mic1 Bias"},
+ {"Mic1 Bias", NULL, "Ext Mic 1"},
+
+ {"MIC2P", NULL, "Mic1 Bias"},
+ {"MIC2N", NULL, "Mic1 Bias"},
+ {"Mic1 Bias", NULL, "Headset Mic 2"},
+
+ {"MIC3P", NULL, "Mic3 Bias"},
+ {"MIC3N", NULL, "Mic3 Bias"},
+ {"Mic3 Bias", NULL, "Ext Mic 3"},
+};
+
+static int ttc_pm860x_init(struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
+
+ /* connected pins */
+ snd_soc_dapm_enable_pin(dapm, "Ext Speaker");
+ snd_soc_dapm_enable_pin(dapm, "Ext Mic 1");
+ snd_soc_dapm_enable_pin(dapm, "Ext Mic 3");
+ snd_soc_dapm_disable_pin(dapm, "Headset Mic 2");
+ snd_soc_dapm_disable_pin(dapm, "Headset Stereophone");
+
+ /* Headset jack detection */
+ snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE
+ | SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2,
+ &hs_jack);
+ snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pins),
+ hs_jack_pins);
+ snd_soc_jack_new(codec, "Microphone Jack", SND_JACK_MICROPHONE,
+ &mic_jack);
+ snd_soc_jack_add_pins(&mic_jack, ARRAY_SIZE(mic_jack_pins),
+ mic_jack_pins);
+
+ /* headphone, microphone detection & headset short detection */
+ pm860x_hs_jack_detect(codec, &hs_jack, SND_JACK_HEADPHONE,
+ SND_JACK_BTN_0, SND_JACK_BTN_1, SND_JACK_BTN_2);
+ pm860x_mic_jack_detect(codec, &hs_jack, SND_JACK_MICROPHONE);
+
+ return 0;
+}
+
+/* ttc/td-dkb digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link ttc_pm860x_hifi_dai[] = {
+{
+ .name = "88pm860x i2s",
+ .stream_name = "audio playback",
+ .codec_name = "88pm860x-codec",
+ .platform_name = "mmp-pcm-audio",
+ .cpu_dai_name = "pxa-ssp-dai.1",
+ .codec_dai_name = "88pm860x-i2s",
+ .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBM_CFM,
+ .init = ttc_pm860x_init,
+},
+};
+
+/* ttc/td audio machine driver */
+static struct snd_soc_card ttc_dkb_card = {
+ .name = "ttc-dkb-hifi",
+ .dai_link = ttc_pm860x_hifi_dai,
+ .num_links = ARRAY_SIZE(ttc_pm860x_hifi_dai),
+
+ .dapm_widgets = ttc_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(ttc_dapm_widgets),
+ .dapm_routes = ttc_audio_map,
+ .num_dapm_routes = ARRAY_SIZE(ttc_audio_map),
+};
+
+static int __devinit ttc_dkb_probe(struct platform_device *pdev)
+{
+ struct snd_soc_card *card = &ttc_dkb_card;
+ int ret;
+
+ card->dev = &pdev->dev;
+
+ ret = snd_soc_register_card(card);
+ if (ret)
+ dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
+ ret);
+
+ return ret;
+}
+
+static int __devexit ttc_dkb_remove(struct platform_device *pdev)
+{
+ struct snd_soc_card *card = platform_get_drvdata(pdev);
+
+ snd_soc_unregister_card(card);
+
+ return 0;
+}
+
+static struct platform_driver ttc_dkb_driver = {
+ .driver = {
+ .name = "ttc-dkb-audio",
+ .owner = THIS_MODULE,
+ },
+ .probe = ttc_dkb_probe,
+ .remove = __devexit_p(ttc_dkb_remove),
+};
+
+module_platform_driver(ttc_dkb_driver);
+
+/* Module information */
+MODULE_AUTHOR("Qiao Zhou, <zhouqiao@marvell.com>");
+MODULE_DESCRIPTION("ALSA SoC TTC DKB");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:ttc-dkb-audio");
diff --git a/sound/soc/samsung/littlemill.c b/sound/soc/samsung/littlemill.c
index c82c646..ee52c8a0 100644
--- a/sound/soc/samsung/littlemill.c
+++ b/sound/soc/samsung/littlemill.c
@@ -211,6 +211,11 @@ static int bbclk_ev(struct snd_soc_dapm_widget *w,
return 0;
}
+static const struct snd_kcontrol_new controls[] = {
+ SOC_DAPM_PIN_SWITCH("WM1250 Input"),
+ SOC_DAPM_PIN_SWITCH("WM1250 Output"),
+};
+
static struct snd_soc_dapm_widget widgets[] = {
SND_SOC_DAPM_HP("Headphone", NULL),
@@ -282,6 +287,8 @@ static struct snd_soc_card littlemill = {
.set_bias_level = littlemill_set_bias_level,
.set_bias_level_post = littlemill_set_bias_level_post,
+ .controls = controls,
+ .num_controls = ARRAY_SIZE(controls),
.dapm_widgets = widgets,
.num_dapm_widgets = ARRAY_SIZE(widgets),
.dapm_routes = audio_paths,
diff --git a/sound/soc/samsung/s3c2412-i2s.c b/sound/soc/samsung/s3c2412-i2s.c
index 79fbeea..ac7701b 100644
--- a/sound/soc/samsung/s3c2412-i2s.c
+++ b/sound/soc/samsung/s3c2412-i2s.c
@@ -25,7 +25,6 @@
#include <sound/soc.h>
#include <sound/pcm_params.h>
-#include <mach/regs-gpio.h>
#include <mach/dma.h>
#include "dma.h"
@@ -83,12 +82,9 @@ static int s3c2412_i2s_probe(struct snd_soc_dai *dai)
s3c2412_i2s.iis_cclk = s3c2412_i2s.iis_pclk;
- /* Configure the I2S pins in correct mode */
- s3c2410_gpio_cfgpin(S3C2410_GPE0, S3C2410_GPE0_I2SLRCK);
- s3c2410_gpio_cfgpin(S3C2410_GPE1, S3C2410_GPE1_I2SSCLK);
- s3c2410_gpio_cfgpin(S3C2410_GPE2, S3C2410_GPE2_CDCLK);
- s3c2410_gpio_cfgpin(S3C2410_GPE3, S3C2410_GPE3_I2SSDI);
- s3c2410_gpio_cfgpin(S3C2410_GPE4, S3C2410_GPE4_I2SSDO);
+ /* Configure the I2S pins (GPE0...GPE4) in correct mode */
+ s3c_gpio_cfgall_range(S3C2410_GPE(0), 5, S3C_GPIO_SFN(2),
+ S3C_GPIO_PULL_NONE);
return 0;
}
diff --git a/sound/soc/samsung/s3c24xx-i2s.c b/sound/soc/samsung/s3c24xx-i2s.c
index c4aa4d4..0aae3a3 100644
--- a/sound/soc/samsung/s3c24xx-i2s.c
+++ b/sound/soc/samsung/s3c24xx-i2s.c
@@ -23,7 +23,6 @@
#include <sound/soc.h>
#include <sound/pcm_params.h>
-#include <mach/regs-gpio.h>
#include <mach/dma.h>
#include <plat/regs-iis.h>
@@ -391,12 +390,9 @@ static int s3c24xx_i2s_probe(struct snd_soc_dai *dai)
}
clk_enable(s3c24xx_i2s.iis_clk);
- /* Configure the I2S pins in correct mode */
- s3c2410_gpio_cfgpin(S3C2410_GPE0, S3C2410_GPE0_I2SLRCK);
- s3c2410_gpio_cfgpin(S3C2410_GPE1, S3C2410_GPE1_I2SSCLK);
- s3c2410_gpio_cfgpin(S3C2410_GPE2, S3C2410_GPE2_CDCLK);
- s3c2410_gpio_cfgpin(S3C2410_GPE3, S3C2410_GPE3_I2SSDI);
- s3c2410_gpio_cfgpin(S3C2410_GPE4, S3C2410_GPE4_I2SSDO);
+ /* Configure the I2S pins (GPE0...GPE4) in correct mode */
+ s3c_gpio_cfgall_range(S3C2410_GPE(0), 5, S3C_GPIO_SFN(2),
+ S3C_GPIO_PULL_NONE);
writel(S3C2410_IISCON_IISEN, s3c24xx_i2s.regs + S3C2410_IISCON);
diff --git a/sound/soc/samsung/smdk_wm8994.c b/sound/soc/samsung/smdk_wm8994.c
index 8eb309f..48dd4dd 100644
--- a/sound/soc/samsung/smdk_wm8994.c
+++ b/sound/soc/samsung/smdk_wm8994.c
@@ -149,31 +149,41 @@ static struct snd_soc_card smdk = {
.num_links = ARRAY_SIZE(smdk_dai),
};
-static struct platform_device *smdk_snd_device;
-static int __init smdk_audio_init(void)
+static int __devinit smdk_audio_probe(struct platform_device *pdev)
{
int ret;
+ struct snd_soc_card *card = &smdk;
- smdk_snd_device = platform_device_alloc("soc-audio", -1);
- if (!smdk_snd_device)
- return -ENOMEM;
+ card->dev = &pdev->dev;
+ ret = snd_soc_register_card(card);
- platform_set_drvdata(smdk_snd_device, &smdk);
-
- ret = platform_device_add(smdk_snd_device);
if (ret)
- platform_device_put(smdk_snd_device);
+ dev_err(&pdev->dev, "snd_soc_register_card() failed:%d\n", ret);
return ret;
}
-module_init(smdk_audio_init);
-static void __exit smdk_audio_exit(void)
+static int __devexit smdk_audio_remove(struct platform_device *pdev)
{
- platform_device_unregister(smdk_snd_device);
+ struct snd_soc_card *card = platform_get_drvdata(pdev);
+
+ snd_soc_unregister_card(card);
+
+ return 0;
}
-module_exit(smdk_audio_exit);
+
+static struct platform_driver smdk_audio_driver = {
+ .driver = {
+ .name = "smdk-audio",
+ .owner = THIS_MODULE,
+ },
+ .probe = smdk_audio_probe,
+ .remove = __devexit_p(smdk_audio_remove),
+};
+
+module_platform_driver(smdk_audio_driver);
MODULE_DESCRIPTION("ALSA SoC SMDK WM8994");
MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:smdk-audio");
diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c
index 2ef9853..53486ff 100644
--- a/sound/soc/sh/fsi.c
+++ b/sound/soc/sh/fsi.c
@@ -247,7 +247,7 @@ struct fsi_priv {
struct fsi_stream_handler {
int (*init)(struct fsi_priv *fsi, struct fsi_stream *io);
int (*quit)(struct fsi_priv *fsi, struct fsi_stream *io);
- int (*probe)(struct fsi_priv *fsi, struct fsi_stream *io);
+ int (*probe)(struct fsi_priv *fsi, struct fsi_stream *io, struct device *dev);
int (*transfer)(struct fsi_priv *fsi, struct fsi_stream *io);
int (*remove)(struct fsi_priv *fsi, struct fsi_stream *io);
void (*start_stop)(struct fsi_priv *fsi, struct fsi_stream *io,
@@ -571,16 +571,16 @@ static int fsi_stream_transfer(struct fsi_stream *io)
#define fsi_stream_stop(fsi, io)\
fsi_stream_handler_call(io, start_stop, fsi, io, 0)
-static int fsi_stream_probe(struct fsi_priv *fsi)
+static int fsi_stream_probe(struct fsi_priv *fsi, struct device *dev)
{
struct fsi_stream *io;
int ret1, ret2;
io = &fsi->playback;
- ret1 = fsi_stream_handler_call(io, probe, fsi, io);
+ ret1 = fsi_stream_handler_call(io, probe, fsi, io, dev);
io = &fsi->capture;
- ret2 = fsi_stream_handler_call(io, probe, fsi, io);
+ ret2 = fsi_stream_handler_call(io, probe, fsi, io, dev);
if (ret1 < 0)
return ret1;
@@ -1089,13 +1089,10 @@ static void fsi_dma_do_tasklet(unsigned long data)
{
struct fsi_stream *io = (struct fsi_stream *)data;
struct fsi_priv *fsi = fsi_stream_to_priv(io);
- struct dma_chan *chan;
struct snd_soc_dai *dai;
struct dma_async_tx_descriptor *desc;
- struct scatterlist sg;
struct snd_pcm_runtime *runtime;
enum dma_data_direction dir;
- dma_cookie_t cookie;
int is_play = fsi_stream_is_play(fsi, io);
int len;
dma_addr_t buf;
@@ -1104,7 +1101,6 @@ static void fsi_dma_do_tasklet(unsigned long data)
return;
dai = fsi_get_dai(io->substream);
- chan = io->chan;
runtime = io->substream->runtime;
dir = is_play ? DMA_TO_DEVICE : DMA_FROM_DEVICE;
len = samples_to_bytes(runtime, io->period_samples);
@@ -1112,14 +1108,8 @@ static void fsi_dma_do_tasklet(unsigned long data)
dma_sync_single_for_device(dai->dev, buf, len, dir);
- sg_init_table(&sg, 1);
- sg_set_page(&sg, pfn_to_page(PFN_DOWN(buf)),
- len , offset_in_page(buf));
- sg_dma_address(&sg) = buf;
- sg_dma_len(&sg) = len;
-
- desc = dmaengine_prep_slave_sg(chan, &sg, 1, dir,
- DMA_PREP_INTERRUPT | DMA_CTRL_ACK);
+ desc = dmaengine_prep_slave_single(io->chan, buf, len, dir,
+ DMA_PREP_INTERRUPT | DMA_CTRL_ACK);
if (!desc) {
dev_err(dai->dev, "dmaengine_prep_slave_sg() fail\n");
return;
@@ -1128,13 +1118,12 @@ static void fsi_dma_do_tasklet(unsigned long data)
desc->callback = fsi_dma_complete;
desc->callback_param = io;
- cookie = desc->tx_submit(desc);
- if (cookie < 0) {
+ if (dmaengine_submit(desc) < 0) {
dev_err(dai->dev, "tx_submit() fail\n");
return;
}
- dma_async_issue_pending(chan);
+ dma_async_issue_pending(io->chan);
/*
* FIXME
@@ -1184,7 +1173,7 @@ static void fsi_dma_push_start_stop(struct fsi_priv *fsi, struct fsi_stream *io,
fsi_master_mask_set(master, CLK_RST, clk, (enable) ? clk : 0);
}
-static int fsi_dma_probe(struct fsi_priv *fsi, struct fsi_stream *io)
+static int fsi_dma_probe(struct fsi_priv *fsi, struct fsi_stream *io, struct device *dev)
{
dma_cap_mask_t mask;
@@ -1192,8 +1181,19 @@ static int fsi_dma_probe(struct fsi_priv *fsi, struct fsi_stream *io)
dma_cap_set(DMA_SLAVE, mask);
io->chan = dma_request_channel(mask, fsi_dma_filter, &io->slave);
- if (!io->chan)
- return -EIO;
+ if (!io->chan) {
+
+ /* switch to PIO handler */
+ if (fsi_stream_is_play(fsi, io))
+ fsi->playback.handler = &fsi_pio_push_handler;
+ else
+ fsi->capture.handler = &fsi_pio_pop_handler;
+
+ dev_info(dev, "switch handler (dma => pio)\n");
+
+ /* probe again */
+ return fsi_stream_probe(fsi, dev);
+ }
tasklet_init(&io->tasklet, fsi_dma_do_tasklet, (unsigned long)io);
@@ -1683,7 +1683,7 @@ static int fsi_probe(struct platform_device *pdev)
master->fsia.master = master;
master->fsia.info = &info->port_a;
fsi_handler_init(&master->fsia);
- ret = fsi_stream_probe(&master->fsia);
+ ret = fsi_stream_probe(&master->fsia, &pdev->dev);
if (ret < 0) {
dev_err(&pdev->dev, "FSIA stream probe failed\n");
goto exit_iounmap;
@@ -1694,7 +1694,7 @@ static int fsi_probe(struct platform_device *pdev)
master->fsib.master = master;
master->fsib.info = &info->port_b;
fsi_handler_init(&master->fsib);
- ret = fsi_stream_probe(&master->fsib);
+ ret = fsi_stream_probe(&master->fsib, &pdev->dev);
if (ret < 0) {
dev_err(&pdev->dev, "FSIB stream probe failed\n");
goto exit_fsia;
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index b37ee80..f219b2f 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -812,13 +812,15 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num)
/* Find CPU DAI from registered DAIs*/
list_for_each_entry(cpu_dai, &dai_list, list) {
- if (dai_link->cpu_dai_of_node) {
- if (cpu_dai->dev->of_node != dai_link->cpu_dai_of_node)
- continue;
- } else {
- if (strcmp(cpu_dai->name, dai_link->cpu_dai_name))
- continue;
- }
+ if (dai_link->cpu_of_node &&
+ (cpu_dai->dev->of_node != dai_link->cpu_of_node))
+ continue;
+ if (dai_link->cpu_name &&
+ strcmp(dev_name(cpu_dai->dev), dai_link->cpu_name))
+ continue;
+ if (dai_link->cpu_dai_name &&
+ strcmp(cpu_dai->name, dai_link->cpu_dai_name))
+ continue;
rtd->cpu_dai = cpu_dai;
}
@@ -896,6 +898,28 @@ 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)
+{
+ int ret;
+
+ if (platform->driver->remove) {
+ ret = platform->driver->remove(platform);
+ if (ret < 0)
+ pr_err("asoc: failed to remove %s: %d\n",
+ platform->name, ret);
+ }
+
+ /* Make sure all DAPM widgets are freed */
+ snd_soc_dapm_free(&platform->dapm);
+
+ soc_cleanup_platform_debugfs(platform);
+ platform->probed = 0;
+ list_del(&platform->card_list);
+ module_put(platform->dev->driver->owner);
+
+ return 0;
+}
+
static void soc_remove_codec(struct snd_soc_codec *codec)
{
int err;
@@ -917,11 +941,9 @@ static void soc_remove_codec(struct snd_soc_codec *codec)
module_put(codec->dev->driver->owner);
}
-static void soc_remove_dai_link(struct snd_soc_card *card, int num, int order)
+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_codec *codec = rtd->codec;
- struct snd_soc_platform *platform = rtd->platform;
struct snd_soc_dai *codec_dai = rtd->codec_dai, *cpu_dai = rtd->cpu_dai;
int err;
@@ -946,30 +968,6 @@ static void soc_remove_dai_link(struct snd_soc_card *card, int num, int order)
list_del(&codec_dai->card_list);
}
- /* remove the platform */
- if (platform && platform->probed &&
- platform->driver->remove_order == order) {
- if (platform->driver->remove) {
- err = platform->driver->remove(platform);
- if (err < 0)
- pr_err("asoc: failed to remove %s: %d\n",
- platform->name, err);
- }
-
- /* Make sure all DAPM widgets are freed */
- snd_soc_dapm_free(&platform->dapm);
-
- soc_cleanup_platform_debugfs(platform);
- platform->probed = 0;
- list_del(&platform->card_list);
- module_put(platform->dev->driver->owner);
- }
-
- /* remove the CODEC */
- if (codec && codec->probed &&
- codec->driver->remove_order == order)
- soc_remove_codec(codec);
-
/* remove the cpu_dai */
if (cpu_dai && cpu_dai->probed &&
cpu_dai->driver->remove_order == order) {
@@ -981,7 +979,43 @@ static void soc_remove_dai_link(struct snd_soc_card *card, int num, int order)
}
cpu_dai->probed = 0;
list_del(&cpu_dai->card_list);
- module_put(cpu_dai->dev->driver->owner);
+
+ if (!cpu_dai->codec) {
+ snd_soc_dapm_free(&cpu_dai->dapm);
+ module_put(cpu_dai->dev->driver->owner);
+ }
+ }
+}
+
+static void soc_remove_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_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_platform *platform = rtd->platform;
+ struct snd_soc_codec *codec;
+
+ /* remove the platform */
+ if (platform && platform->probed &&
+ platform->driver->remove_order == order) {
+ soc_remove_platform(platform);
+ }
+
+ /* remove the CODEC-side CODEC */
+ if (codec_dai) {
+ codec = codec_dai->codec;
+ if (codec && codec->probed &&
+ codec->driver->remove_order == order)
+ soc_remove_codec(codec);
+ }
+
+ /* 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);
}
}
@@ -992,8 +1026,15 @@ static void soc_remove_dai_links(struct snd_soc_card *card)
for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST;
order++) {
for (dai = 0; dai < card->num_rtd; dai++)
- soc_remove_dai_link(card, dai, order);
+ soc_remove_link_dais(card, dai, order);
}
+
+ for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST;
+ order++) {
+ for (dai = 0; dai < card->num_rtd; dai++)
+ soc_remove_link_components(card, dai, order);
+ }
+
card->num_rtd = 0;
}
@@ -1054,6 +1095,10 @@ static int soc_probe_codec(struct snd_soc_card *card,
}
}
+ /* If the driver didn't set I/O up try regmap */
+ if (!codec->control_data)
+ snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP);
+
if (driver->controls)
snd_soc_add_codec_controls(codec, driver->controls,
driver->num_controls);
@@ -1230,7 +1275,44 @@ out:
return 0;
}
-static int soc_probe_dai_link(struct snd_soc_card *card, int num, int order)
+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_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_platform *platform = rtd->platform;
+ int 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;
+ }
+
+ /* probe the CODEC-side component */
+ if (!codec_dai->codec->probed &&
+ codec_dai->codec->driver->probe_order == order) {
+ ret = soc_probe_codec(card, codec_dai->codec);
+ if (ret < 0)
+ return ret;
+ }
+
+ /* probe the platform */
+ if (!platform->probed &&
+ platform->driver->probe_order == order) {
+ ret = soc_probe_platform(card, platform);
+ if (ret < 0)
+ return ret;
+ }
+
+ return 0;
+}
+
+static int soc_probe_link_dais(struct snd_soc_card *card, int num, int order)
{
struct snd_soc_dai_link *dai_link = &card->dai_link[num];
struct snd_soc_pcm_runtime *rtd = &card->rtd[num];
@@ -1255,11 +1337,14 @@ static int soc_probe_dai_link(struct snd_soc_card *card, int num, int order)
/* probe the cpu_dai */
if (!cpu_dai->probed &&
cpu_dai->driver->probe_order == order) {
- cpu_dai->dapm.card = card;
- if (!try_module_get(cpu_dai->dev->driver->owner))
- return -ENODEV;
+ if (!cpu_dai->codec) {
+ cpu_dai->dapm.card = card;
+ if (!try_module_get(cpu_dai->dev->driver->owner))
+ return -ENODEV;
- snd_soc_dapm_new_dai_widgets(&cpu_dai->dapm, cpu_dai);
+ list_add(&cpu_dai->dapm.list, &card->dapm_list);
+ snd_soc_dapm_new_dai_widgets(&cpu_dai->dapm, cpu_dai);
+ }
if (cpu_dai->driver->probe) {
ret = cpu_dai->driver->probe(cpu_dai);
@@ -1275,22 +1360,6 @@ static int soc_probe_dai_link(struct snd_soc_card *card, int num, int order)
list_add(&cpu_dai->card_list, &card->dai_dev_list);
}
- /* probe the CODEC */
- if (!codec->probed &&
- codec->driver->probe_order == order) {
- ret = soc_probe_codec(card, codec);
- if (ret < 0)
- return ret;
- }
-
- /* probe the platform */
- if (!platform->probed &&
- platform->driver->probe_order == order) {
- ret = soc_probe_platform(card, platform);
- if (ret < 0)
- return ret;
- }
-
/* probe the CODEC DAI */
if (!codec_dai->probed && codec_dai->driver->probe_order == order) {
if (codec_dai->driver->probe) {
@@ -1565,14 +1634,27 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card)
goto card_probe_error;
}
- /* early DAI link probe */
+ /* probe all components used by DAI links on this card */
for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST;
order++) {
for (i = 0; i < card->num_links; i++) {
- ret = soc_probe_dai_link(card, i, order);
+ ret = soc_probe_link_components(card, i, order);
if (ret < 0) {
pr_err("asoc: failed to instantiate card %s: %d\n",
- card->name, ret);
+ card->name, ret);
+ goto probe_dai_err;
+ }
+ }
+ }
+
+ /* probe all DAI links on this card */
+ for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST;
+ order++) {
+ for (i = 0; i < card->num_links; i++) {
+ ret = soc_probe_link_dais(card, i, order);
+ if (ret < 0) {
+ pr_err("asoc: failed to instantiate card %s: %d\n",
+ card->name, ret);
goto probe_dai_err;
}
}
@@ -2790,6 +2872,104 @@ int snd_soc_put_volsw_s8(struct snd_kcontrol *kcontrol,
EXPORT_SYMBOL_GPL(snd_soc_put_volsw_s8);
/**
+ * snd_soc_info_volsw_range - single mixer info callback with range.
+ * @kcontrol: mixer control
+ * @uinfo: control element information
+ *
+ * Callback to provide information, within a range, about a single
+ * mixer control.
+ *
+ * returns 0 for success.
+ */
+int snd_soc_info_volsw_range(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ struct soc_mixer_control *mc =
+ (struct soc_mixer_control *)kcontrol->private_value;
+ int platform_max;
+ int min = mc->min;
+
+ if (!mc->platform_max)
+ mc->platform_max = mc->max;
+ platform_max = mc->platform_max;
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 1;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = platform_max - min;
+
+ return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_info_volsw_range);
+
+/**
+ * snd_soc_put_volsw_range - single mixer put value callback with range.
+ * @kcontrol: mixer control
+ * @ucontrol: control element information
+ *
+ * Callback to set the value, within a range, for a single mixer control.
+ *
+ * Returns 0 for success.
+ */
+int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct soc_mixer_control *mc =
+ (struct soc_mixer_control *)kcontrol->private_value;
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ unsigned int reg = mc->reg;
+ unsigned int shift = mc->shift;
+ int min = mc->min;
+ int max = mc->max;
+ unsigned int mask = (1 << fls(max)) - 1;
+ unsigned int invert = mc->invert;
+ unsigned int val, val_mask;
+
+ val = ((ucontrol->value.integer.value[0] + min) & mask);
+ if (invert)
+ val = max - val;
+ val_mask = mask << shift;
+ val = val << shift;
+
+ return snd_soc_update_bits_locked(codec, reg, val_mask, val);
+}
+EXPORT_SYMBOL_GPL(snd_soc_put_volsw_range);
+
+/**
+ * snd_soc_get_volsw_range - single mixer get callback with range
+ * @kcontrol: mixer control
+ * @ucontrol: control element information
+ *
+ * Callback to get the value, within a range, of a single mixer control.
+ *
+ * Returns 0 for success.
+ */
+int snd_soc_get_volsw_range(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct soc_mixer_control *mc =
+ (struct soc_mixer_control *)kcontrol->private_value;
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ unsigned int reg = mc->reg;
+ unsigned int shift = mc->shift;
+ int min = mc->min;
+ int max = mc->max;
+ unsigned int mask = (1 << fls(max)) - 1;
+ unsigned int invert = mc->invert;
+
+ ucontrol->value.integer.value[0] =
+ (snd_soc_read(codec, reg) >> shift) & mask;
+ if (invert)
+ ucontrol->value.integer.value[0] =
+ max - ucontrol->value.integer.value[0];
+ ucontrol->value.integer.value[0] =
+ ucontrol->value.integer.value[0] - min;
+
+ return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_get_volsw_range);
+
+/**
* snd_soc_limit_volume - Set new limit to an existing volume control.
*
* @codec: where to look for the control
@@ -3346,6 +3526,12 @@ int snd_soc_register_card(struct snd_soc_card *card)
link->name);
return -EINVAL;
}
+ /* Codec DAI name must be specified */
+ if (!link->codec_dai_name) {
+ dev_err(card->dev, "codec_dai_name not set for %s\n",
+ link->name);
+ return -EINVAL;
+ }
/*
* Platform may be specified by either name or OF node, but
@@ -3358,12 +3544,24 @@ int snd_soc_register_card(struct snd_soc_card *card)
}
/*
- * CPU DAI must be specified by 1 of name or OF node,
- * not both or neither.
+ * CPU device may be specified by either name or OF node, but
+ * can be left unspecified, and will be matched based on DAI
+ * name alone..
+ */
+ if (link->cpu_name && link->cpu_of_node) {
+ dev_err(card->dev,
+ "Neither/both cpu name/of_node are set for %s\n",
+ link->name);
+ return -EINVAL;
+ }
+ /*
+ * At least one of CPU DAI name or CPU device name/node must be
+ * specified
*/
- if (!!link->cpu_dai_name == !!link->cpu_dai_of_node) {
+ if (!link->cpu_dai_name &&
+ !(link->cpu_name || link->cpu_of_node)) {
dev_err(card->dev,
- "Neither/both cpu_dai name/of_node are set for %s\n",
+ "Neither cpu_dai_name nor cpu_name/of_node are set for %s\n",
link->name);
return -EINVAL;
}
@@ -3938,6 +4136,7 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card,
dev_err(card->dev,
"Property '%s' index %d could not be read: %d\n",
propname, 2 * i, ret);
+ kfree(routes);
return -EINVAL;
}
ret = of_property_read_string_index(np, propname,
@@ -3946,6 +4145,7 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card,
dev_err(card->dev,
"Property '%s' index %d could not be read: %d\n",
propname, (2 * i) + 1, ret);
+ kfree(routes);
return -EINVAL;
}
}
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 89eae93..4d181df9 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -35,6 +35,7 @@
#include <linux/debugfs.h>
#include <linux/pm_runtime.h>
#include <linux/regulator/consumer.h>
+#include <linux/clk.h>
#include <linux/slab.h>
#include <sound/core.h>
#include <sound/pcm.h>
@@ -51,6 +52,7 @@ static int dapm_up_seq[] = {
[snd_soc_dapm_pre] = 0,
[snd_soc_dapm_supply] = 1,
[snd_soc_dapm_regulator_supply] = 1,
+ [snd_soc_dapm_clock_supply] = 1,
[snd_soc_dapm_micbias] = 2,
[snd_soc_dapm_dai_link] = 2,
[snd_soc_dapm_dai] = 3,
@@ -92,6 +94,7 @@ static int dapm_down_seq[] = {
[snd_soc_dapm_aif_out] = 10,
[snd_soc_dapm_dai] = 10,
[snd_soc_dapm_dai_link] = 11,
+ [snd_soc_dapm_clock_supply] = 12,
[snd_soc_dapm_regulator_supply] = 12,
[snd_soc_dapm_supply] = 12,
[snd_soc_dapm_post] = 13,
@@ -288,9 +291,9 @@ static int snd_soc_dapm_set_bias_level(struct snd_soc_dapm_context *dapm,
if (dapm->codec->driver->set_bias_level)
ret = dapm->codec->driver->set_bias_level(dapm->codec,
level);
- else
- dapm->bias_level = level;
- }
+ } else
+ dapm->bias_level = level;
+
if (ret != 0)
goto out;
@@ -321,11 +324,10 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w,
val = soc_widget_read(w, reg);
val = (val >> shift) & mask;
+ if (invert)
+ val = max - val;
- if ((invert && !val) || (!invert && val))
- p->connect = 1;
- else
- p->connect = 0;
+ p->connect = !!val;
}
break;
case snd_soc_dapm_mux: {
@@ -391,6 +393,7 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w,
case snd_soc_dapm_vmid:
case snd_soc_dapm_supply:
case snd_soc_dapm_regulator_supply:
+ case snd_soc_dapm_clock_supply:
case snd_soc_dapm_aif_in:
case snd_soc_dapm_aif_out:
case snd_soc_dapm_dai:
@@ -764,6 +767,7 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget,
switch (widget->id) {
case snd_soc_dapm_supply:
case snd_soc_dapm_regulator_supply:
+ case snd_soc_dapm_clock_supply:
return 0;
default:
break;
@@ -850,6 +854,7 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget,
switch (widget->id) {
case snd_soc_dapm_supply:
case snd_soc_dapm_regulator_supply:
+ case snd_soc_dapm_clock_supply:
return 0;
default:
break;
@@ -996,6 +1001,27 @@ int dapm_regulator_event(struct snd_soc_dapm_widget *w,
}
EXPORT_SYMBOL_GPL(dapm_regulator_event);
+/*
+ * Handler for clock supply widget.
+ */
+int dapm_clock_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ if (!w->clk)
+ return -EIO;
+
+#ifdef CONFIG_HAVE_CLK
+ if (SND_SOC_DAPM_EVENT_ON(event)) {
+ return clk_enable(w->clk);
+ } else {
+ clk_disable(w->clk);
+ return 0;
+ }
+#endif
+ return 0;
+}
+EXPORT_SYMBOL_GPL(dapm_clock_event);
+
static int dapm_widget_power_check(struct snd_soc_dapm_widget *w)
{
if (w->power_checked)
@@ -1487,6 +1513,7 @@ static void dapm_widget_set_power(struct snd_soc_dapm_widget *w, bool power,
switch (w->id) {
case snd_soc_dapm_supply:
case snd_soc_dapm_regulator_supply:
+ case snd_soc_dapm_clock_supply:
/* Supplies can't affect their outputs, only their inputs */
break;
default:
@@ -1570,7 +1597,15 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event)
}
list_for_each_entry(w, &card->widgets, list) {
- list_del_init(&w->dirty);
+ switch (w->id) {
+ case snd_soc_dapm_pre:
+ case snd_soc_dapm_post:
+ /* These widgets always need to be powered */
+ break;
+ default:
+ list_del_init(&w->dirty);
+ break;
+ }
if (w->power) {
d = w->dapm;
@@ -1587,6 +1622,7 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event)
break;
case snd_soc_dapm_supply:
case snd_soc_dapm_regulator_supply:
+ case snd_soc_dapm_clock_supply:
case snd_soc_dapm_micbias:
if (d->target_bias_level < SND_SOC_BIAS_STANDBY)
d->target_bias_level = SND_SOC_BIAS_STANDBY;
@@ -1941,6 +1977,7 @@ static ssize_t dapm_widget_show(struct device *dev,
case snd_soc_dapm_mixer_named_ctl:
case snd_soc_dapm_supply:
case snd_soc_dapm_regulator_supply:
+ case snd_soc_dapm_clock_supply:
if (w->name)
count += sprintf(buf + count, "%s: %s\n",
w->name, w->power ? "On":"Off");
@@ -2187,6 +2224,7 @@ static int snd_soc_dapm_add_route(struct snd_soc_dapm_context *dapm,
case snd_soc_dapm_post:
case snd_soc_dapm_supply:
case snd_soc_dapm_regulator_supply:
+ case snd_soc_dapm_clock_supply:
case snd_soc_dapm_aif_in:
case snd_soc_dapm_aif_out:
case snd_soc_dapm_dai:
@@ -2221,6 +2259,10 @@ static int snd_soc_dapm_add_route(struct snd_soc_dapm_context *dapm,
path->connect = 0;
return 0;
}
+
+ dapm_mark_dirty(wsource, "Route added");
+ dapm_mark_dirty(wsink, "Route added");
+
return 0;
err:
@@ -2230,6 +2272,59 @@ err:
return ret;
}
+static int snd_soc_dapm_del_route(struct snd_soc_dapm_context *dapm,
+ const struct snd_soc_dapm_route *route)
+{
+ struct snd_soc_dapm_path *path, *p;
+ const char *sink;
+ const char *source;
+ char prefixed_sink[80];
+ char prefixed_source[80];
+
+ if (route->control) {
+ dev_err(dapm->dev,
+ "Removal of routes with controls not supported\n");
+ return -EINVAL;
+ }
+
+ if (dapm->codec && dapm->codec->name_prefix) {
+ snprintf(prefixed_sink, sizeof(prefixed_sink), "%s %s",
+ dapm->codec->name_prefix, route->sink);
+ sink = prefixed_sink;
+ snprintf(prefixed_source, sizeof(prefixed_source), "%s %s",
+ dapm->codec->name_prefix, route->source);
+ source = prefixed_source;
+ } else {
+ sink = route->sink;
+ source = route->source;
+ }
+
+ path = NULL;
+ list_for_each_entry(p, &dapm->card->paths, list) {
+ if (strcmp(p->source->name, source) != 0)
+ continue;
+ if (strcmp(p->sink->name, sink) != 0)
+ continue;
+ path = p;
+ break;
+ }
+
+ if (path) {
+ dapm_mark_dirty(path->source, "Route removed");
+ dapm_mark_dirty(path->sink, "Route removed");
+
+ list_del(&path->list);
+ list_del(&path->list_sink);
+ list_del(&path->list_source);
+ kfree(path);
+ } else {
+ dev_warn(dapm->dev, "Route %s->%s does not exist\n",
+ source, sink);
+ }
+
+ return 0;
+}
+
/**
* snd_soc_dapm_add_routes - Add routes between DAPM widgets
* @dapm: DAPM context
@@ -2246,15 +2341,15 @@ err:
int snd_soc_dapm_add_routes(struct snd_soc_dapm_context *dapm,
const struct snd_soc_dapm_route *route, int num)
{
- int i, ret = 0;
+ int i, r, ret = 0;
mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_INIT);
for (i = 0; i < num; i++) {
- ret = snd_soc_dapm_add_route(dapm, route);
- if (ret < 0) {
+ r = snd_soc_dapm_add_route(dapm, route);
+ if (r < 0) {
dev_err(dapm->dev, "Failed to add route %s->%s\n",
route->source, route->sink);
- break;
+ ret = r;
}
route++;
}
@@ -2264,6 +2359,30 @@ int snd_soc_dapm_add_routes(struct snd_soc_dapm_context *dapm,
}
EXPORT_SYMBOL_GPL(snd_soc_dapm_add_routes);
+/**
+ * snd_soc_dapm_del_routes - Remove routes between DAPM widgets
+ * @dapm: DAPM context
+ * @route: audio routes
+ * @num: number of routes
+ *
+ * Removes routes from the DAPM context.
+ */
+int snd_soc_dapm_del_routes(struct snd_soc_dapm_context *dapm,
+ const struct snd_soc_dapm_route *route, int num)
+{
+ int i, ret = 0;
+
+ mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_INIT);
+ for (i = 0; i < num; i++) {
+ snd_soc_dapm_del_route(dapm, route);
+ route++;
+ }
+ mutex_unlock(&dapm->card->dapm_mutex);
+
+ return ret;
+}
+EXPORT_SYMBOL_GPL(snd_soc_dapm_del_routes);
+
static int snd_soc_dapm_weak_route(struct snd_soc_dapm_context *dapm,
const struct snd_soc_dapm_route *route)
{
@@ -2434,23 +2553,20 @@ int snd_soc_dapm_get_volsw(struct snd_kcontrol *kcontrol,
(struct soc_mixer_control *)kcontrol->private_value;
unsigned int reg = mc->reg;
unsigned int shift = mc->shift;
- unsigned int rshift = mc->rshift;
int max = mc->max;
- unsigned int invert = mc->invert;
unsigned int mask = (1 << fls(max)) - 1;
+ unsigned int invert = mc->invert;
+
+ if (snd_soc_volsw_is_stereo(mc))
+ dev_warn(widget->dapm->dev,
+ "Control '%s' is stereo, which is not supported\n",
+ kcontrol->id.name);
ucontrol->value.integer.value[0] =
(snd_soc_read(widget->codec, reg) >> shift) & mask;
- if (shift != rshift)
- ucontrol->value.integer.value[1] =
- (snd_soc_read(widget->codec, reg) >> rshift) & mask;
- if (invert) {
+ if (invert)
ucontrol->value.integer.value[0] =
max - ucontrol->value.integer.value[0];
- if (shift != rshift)
- ucontrol->value.integer.value[1] =
- max - ucontrol->value.integer.value[1];
- }
return 0;
}
@@ -2484,20 +2600,19 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol,
struct snd_soc_dapm_update update;
int wi;
+ if (snd_soc_volsw_is_stereo(mc))
+ dev_warn(widget->dapm->dev,
+ "Control '%s' is stereo, which is not supported\n",
+ kcontrol->id.name);
+
val = (ucontrol->value.integer.value[0] & mask);
+ connect = !!val;
if (invert)
val = max - val;
mask = mask << shift;
val = val << shift;
- if (val)
- /* new connection */
- connect = invert ? 0 : 1;
- else
- /* old connection must be powered down */
- connect = invert ? 1 : 0;
-
mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME);
change = snd_soc_test_bits(widget->codec, reg, mask, val);
@@ -2873,6 +2988,19 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm,
return NULL;
}
break;
+ case snd_soc_dapm_clock_supply:
+#ifdef CONFIG_CLKDEV_LOOKUP
+ w->clk = devm_clk_get(dapm->dev, w->name);
+ if (IS_ERR(w->clk)) {
+ ret = PTR_ERR(w->clk);
+ dev_err(dapm->dev, "Failed to request %s: %d\n",
+ w->name, ret);
+ return NULL;
+ }
+#else
+ return NULL;
+#endif
+ break;
default:
break;
}
@@ -2924,6 +3052,7 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm,
break;
case snd_soc_dapm_supply:
case snd_soc_dapm_regulator_supply:
+ case snd_soc_dapm_clock_supply:
w->power_check = dapm_supply_check_power;
break;
case snd_soc_dapm_dai:
@@ -3538,10 +3667,13 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_free);
static void soc_dapm_shutdown_codec(struct snd_soc_dapm_context *dapm)
{
+ struct snd_soc_card *card = dapm->card;
struct snd_soc_dapm_widget *w;
LIST_HEAD(down_list);
int powerdown = 0;
+ mutex_lock(&card->dapm_mutex);
+
list_for_each_entry(w, &dapm->card->widgets, list) {
if (w->dapm != dapm)
continue;
@@ -3564,6 +3696,8 @@ static void soc_dapm_shutdown_codec(struct snd_soc_dapm_context *dapm)
snd_soc_dapm_set_bias_level(dapm,
SND_SOC_BIAS_STANDBY);
}
+
+ mutex_unlock(&card->dapm_mutex);
}
/*
diff --git a/sound/soc/soc-dmaengine-pcm.c b/sound/soc/soc-dmaengine-pcm.c
index 4756952..5df529e 100644
--- a/sound/soc/soc-dmaengine-pcm.c
+++ b/sound/soc/soc-dmaengine-pcm.c
@@ -30,6 +30,7 @@
struct dmaengine_pcm_runtime_data {
struct dma_chan *dma_chan;
+ dma_cookie_t cookie;
unsigned int pos;
@@ -153,7 +154,7 @@ static int dmaengine_pcm_prepare_and_submit(struct snd_pcm_substream *substream)
desc->callback = dmaengine_pcm_dma_complete;
desc->callback_param = substream;
- dmaengine_submit(desc);
+ prtd->cookie = dmaengine_submit(desc);
return 0;
}
@@ -200,6 +201,20 @@ int snd_dmaengine_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_trigger);
/**
+ * snd_dmaengine_pcm_pointer_no_residue - dmaengine based PCM pointer implementation
+ * @substream: PCM substream
+ *
+ * This function is deprecated and should not be used by new drivers, as its
+ * results may be unreliable.
+ */
+snd_pcm_uframes_t snd_dmaengine_pcm_pointer_no_residue(struct snd_pcm_substream *substream)
+{
+ struct dmaengine_pcm_runtime_data *prtd = substream_to_prtd(substream);
+ return bytes_to_frames(substream->runtime, prtd->pos);
+}
+EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_pointer_no_residue);
+
+/**
* snd_dmaengine_pcm_pointer - dmaengine based PCM pointer implementation
* @substream: PCM substream
*
@@ -209,7 +224,19 @@ EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_trigger);
snd_pcm_uframes_t snd_dmaengine_pcm_pointer(struct snd_pcm_substream *substream)
{
struct dmaengine_pcm_runtime_data *prtd = substream_to_prtd(substream);
- return bytes_to_frames(substream->runtime, prtd->pos);
+ struct dma_tx_state state;
+ enum dma_status status;
+ unsigned int buf_size;
+ unsigned int pos = 0;
+
+ status = dmaengine_tx_status(prtd->dma_chan, prtd->cookie, &state);
+ if (status == DMA_IN_PROGRESS || status == DMA_PAUSED) {
+ buf_size = snd_pcm_lib_buffer_bytes(substream);
+ if (state.residue > 0 && state.residue <= buf_size)
+ pos = buf_size - state.residue;
+ }
+
+ return bytes_to_frames(substream->runtime, pos);
}
EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_pointer);
@@ -243,7 +270,7 @@ static int dmaengine_pcm_request_channel(struct dmaengine_pcm_runtime_data *prtd
* Note that this function will use private_data field of the substream's
* runtime. So it is not availabe to your pcm driver implementation. If you need
* to keep additional data attached to a substream use
- * snd_dmaeinge_pcm_{set,get}_data.
+ * snd_dmaengine_pcm_{set,get}_data.
*/
int snd_dmaengine_pcm_open(struct snd_pcm_substream *substream,
dma_filter_fn filter_fn, void *filter_data)
diff --git a/sound/soc/soc-io.c b/sound/soc/soc-io.c
index 4d8dc6a..29183ef 100644
--- a/sound/soc/soc-io.c
+++ b/sound/soc/soc-io.c
@@ -142,11 +142,16 @@ int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec,
case SND_SOC_REGMAP:
/* Device has made its own regmap arrangements */
codec->using_regmap = true;
-
- ret = regmap_get_val_bytes(codec->control_data);
- /* Errors are legitimate for non-integer byte multiples */
- if (ret > 0)
- codec->val_bytes = ret;
+ if (!codec->control_data)
+ codec->control_data = dev_get_regmap(codec->dev, NULL);
+
+ if (codec->control_data) {
+ ret = regmap_get_val_bytes(codec->control_data);
+ /* Errors are legitimate for non-integer byte
+ * multiples */
+ if (ret > 0)
+ codec->val_bytes = ret;
+ }
break;
default:
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index 48fd15b..ef22d0b 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -1955,10 +1955,8 @@ static int dpcm_fe_dai_open(struct snd_pcm_substream *fe_substream)
fe->dpcm[stream].runtime = fe_substream->runtime;
if (dpcm_path_get(fe, stream, &list) <= 0) {
- dev_warn(fe->dev, "asoc: %s no valid %s route\n",
+ dev_dbg(fe->dev, "asoc: %s no valid %s route\n",
fe->dai_link->name, stream ? "capture" : "playback");
- mutex_unlock(&fe->card->mutex);
- return -EINVAL;
}
/* calculate valid and active FE <-> BE dpcms */
@@ -2003,7 +2001,6 @@ static int dpcm_fe_dai_close(struct snd_pcm_substream *fe_substream)
/* create a new pcm */
int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num)
{
- struct snd_soc_codec *codec = rtd->codec;
struct snd_soc_platform *platform = rtd->platform;
struct snd_soc_dai *codec_dai = rtd->codec_dai;
struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
@@ -2042,7 +2039,8 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num)
capture, &pcm);
}
if (ret < 0) {
- printk(KERN_ERR "asoc: can't create pcm for codec %s\n", codec->name);
+ dev_err(rtd->card->dev, "can't create pcm for %s\n",
+ rtd->dai_link->name);
return ret;
}
dev_dbg(rtd->card->dev, "registered pcm #%d %s\n",num, new_name);
@@ -2099,14 +2097,14 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num)
if (platform->driver->pcm_new) {
ret = platform->driver->pcm_new(rtd);
if (ret < 0) {
- pr_err("asoc: platform pcm constructor failed\n");
+ dev_err(platform->dev, "pcm constructor failed\n");
return ret;
}
}
pcm->private_free = platform->driver->pcm_free;
out:
- printk(KERN_INFO "asoc: %s <-> %s mapping ok\n", codec_dai->name,
+ dev_info(rtd->card->dev, " %s <-> %s mapping ok\n", codec_dai->name,
cpu_dai->name);
return ret;
}
diff --git a/sound/soc/spear/spdif_in.c b/sound/soc/spear/spdif_in.c
new file mode 100644
index 0000000..c7c4b20
--- /dev/null
+++ b/sound/soc/spear/spdif_in.c
@@ -0,0 +1,297 @@
+/*
+ * ALSA SoC SPDIF In Audio Layer for spear processors
+ *
+ * Copyright (C) 2012 ST Microelectronics
+ * Vipin Kumar <vipin.kumar@st.com>
+ *
+ * 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 <linux/clk.h>
+#include <linux/delay.h>
+#include <linux/device.h>
+#include <linux/kernel.h>
+#include <linux/init.h>
+#include <linux/io.h>
+#include <linux/ioport.h>
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/spear_dma.h>
+#include <sound/spear_spdif.h>
+#include "spdif_in_regs.h"
+
+struct spdif_in_params {
+ u32 format;
+};
+
+struct spdif_in_dev {
+ struct clk *clk;
+ struct spear_dma_data dma_params;
+ struct spdif_in_params saved_params;
+ void *io_base;
+ struct device *dev;
+ void (*reset_perip)(void);
+ int irq;
+};
+
+static void spdif_in_configure(struct spdif_in_dev *host)
+{
+ u32 ctrl = SPDIF_IN_PRTYEN | SPDIF_IN_STATEN | SPDIF_IN_USREN |
+ SPDIF_IN_VALEN | SPDIF_IN_BLKEN;
+ ctrl |= SPDIF_MODE_16BIT | SPDIF_FIFO_THRES_16;
+
+ writel(ctrl, host->io_base + SPDIF_IN_CTRL);
+ writel(0xF, host->io_base + SPDIF_IN_IRQ_MASK);
+}
+
+static int spdif_in_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *cpu_dai)
+{
+ struct spdif_in_dev *host = snd_soc_dai_get_drvdata(cpu_dai);
+
+ if (substream->stream != SNDRV_PCM_STREAM_CAPTURE)
+ return -EINVAL;
+
+ snd_soc_dai_set_dma_data(cpu_dai, substream, (void *)&host->dma_params);
+ return 0;
+}
+
+static void spdif_in_shutdown(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct spdif_in_dev *host = snd_soc_dai_get_drvdata(dai);
+
+ if (substream->stream != SNDRV_PCM_STREAM_CAPTURE)
+ return;
+
+ writel(0x0, host->io_base + SPDIF_IN_IRQ_MASK);
+ snd_soc_dai_set_dma_data(dai, substream, NULL);
+}
+
+static void spdif_in_format(struct spdif_in_dev *host, u32 format)
+{
+ u32 ctrl = readl(host->io_base + SPDIF_IN_CTRL);
+
+ switch (format) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ ctrl |= SPDIF_XTRACT_16BIT;
+ break;
+
+ case SNDRV_PCM_FORMAT_IEC958_SUBFRAME_LE:
+ ctrl &= ~SPDIF_XTRACT_16BIT;
+ break;
+ }
+
+ writel(ctrl, host->io_base + SPDIF_IN_CTRL);
+}
+
+static int spdif_in_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct spdif_in_dev *host = snd_soc_dai_get_drvdata(dai);
+ u32 format;
+
+ if (substream->stream != SNDRV_PCM_STREAM_CAPTURE)
+ return -EINVAL;
+
+ format = params_format(params);
+ host->saved_params.format = format;
+
+ return 0;
+}
+
+static int spdif_in_trigger(struct snd_pcm_substream *substream, int cmd,
+ struct snd_soc_dai *dai)
+{
+ struct spdif_in_dev *host = snd_soc_dai_get_drvdata(dai);
+ u32 ctrl;
+ int ret = 0;
+
+ if (substream->stream != SNDRV_PCM_STREAM_CAPTURE)
+ return -EINVAL;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ clk_enable(host->clk);
+ spdif_in_configure(host);
+ spdif_in_format(host, host->saved_params.format);
+
+ ctrl = readl(host->io_base + SPDIF_IN_CTRL);
+ ctrl |= SPDIF_IN_SAMPLE | SPDIF_IN_ENB;
+ writel(ctrl, host->io_base + SPDIF_IN_CTRL);
+ writel(0xF, host->io_base + SPDIF_IN_IRQ_MASK);
+ break;
+
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ ctrl = readl(host->io_base + SPDIF_IN_CTRL);
+ ctrl &= ~(SPDIF_IN_SAMPLE | SPDIF_IN_ENB);
+ writel(ctrl, host->io_base + SPDIF_IN_CTRL);
+ writel(0x0, host->io_base + SPDIF_IN_IRQ_MASK);
+
+ if (host->reset_perip)
+ host->reset_perip();
+ clk_disable(host->clk);
+ break;
+
+ default:
+ ret = -EINVAL;
+ break;
+ }
+ return ret;
+}
+
+static struct snd_soc_dai_ops spdif_in_dai_ops = {
+ .startup = spdif_in_startup,
+ .shutdown = spdif_in_shutdown,
+ .trigger = spdif_in_trigger,
+ .hw_params = spdif_in_hw_params,
+};
+
+struct snd_soc_dai_driver spdif_in_dai = {
+ .capture = {
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \
+ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000 | \
+ SNDRV_PCM_RATE_192000),
+ .formats = SNDRV_PCM_FMTBIT_S16_LE | \
+ SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_LE,
+ },
+ .ops = &spdif_in_dai_ops,
+};
+
+static irqreturn_t spdif_in_irq(int irq, void *arg)
+{
+ struct spdif_in_dev *host = (struct spdif_in_dev *)arg;
+
+ u32 irq_status = readl(host->io_base + SPDIF_IN_IRQ);
+
+ if (!irq_status)
+ return IRQ_NONE;
+
+ if (irq_status & SPDIF_IRQ_FIFOWRITE)
+ dev_err(host->dev, "spdif in: fifo write error");
+ if (irq_status & SPDIF_IRQ_EMPTYFIFOREAD)
+ dev_err(host->dev, "spdif in: empty fifo read error");
+ if (irq_status & SPDIF_IRQ_FIFOFULL)
+ dev_err(host->dev, "spdif in: fifo full error");
+ if (irq_status & SPDIF_IRQ_OUTOFRANGE)
+ dev_err(host->dev, "spdif in: out of range error");
+
+ writel(0, host->io_base + SPDIF_IN_IRQ);
+
+ return IRQ_HANDLED;
+}
+
+static int spdif_in_probe(struct platform_device *pdev)
+{
+ struct spdif_in_dev *host;
+ struct spear_spdif_platform_data *pdata;
+ struct resource *res, *res_fifo;
+ int ret;
+
+ res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+ if (!res)
+ return -EINVAL;
+
+ res_fifo = platform_get_resource(pdev, IORESOURCE_IO, 0);
+ if (!res_fifo)
+ return -EINVAL;
+
+ if (!devm_request_mem_region(&pdev->dev, res->start,
+ resource_size(res), pdev->name)) {
+ dev_warn(&pdev->dev, "Failed to get memory resourse\n");
+ return -ENOENT;
+ }
+
+ host = devm_kzalloc(&pdev->dev, sizeof(*host), GFP_KERNEL);
+ if (!host) {
+ dev_warn(&pdev->dev, "kzalloc fail\n");
+ return -ENOMEM;
+ }
+
+ host->io_base = devm_ioremap(&pdev->dev, res->start,
+ resource_size(res));
+ if (!host->io_base) {
+ dev_warn(&pdev->dev, "ioremap failed\n");
+ return -ENOMEM;
+ }
+
+ host->irq = platform_get_irq(pdev, 0);
+ if (host->irq < 0)
+ return -EINVAL;
+
+ host->clk = clk_get(&pdev->dev, NULL);
+ if (IS_ERR(host->clk))
+ return PTR_ERR(host->clk);
+
+ pdata = dev_get_platdata(&pdev->dev);
+
+ if (!pdata)
+ return -EINVAL;
+
+ host->dma_params.data = pdata->dma_params;
+ host->dma_params.addr = res_fifo->start;
+ host->dma_params.max_burst = 16;
+ host->dma_params.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES;
+ host->dma_params.filter = pdata->filter;
+ host->reset_perip = pdata->reset_perip;
+
+ host->dev = &pdev->dev;
+ dev_set_drvdata(&pdev->dev, host);
+
+ ret = devm_request_irq(&pdev->dev, host->irq, spdif_in_irq, 0,
+ "spdif-in", host);
+ if (ret) {
+ clk_put(host->clk);
+ dev_warn(&pdev->dev, "request_irq failed\n");
+ return ret;
+ }
+
+ ret = snd_soc_register_dai(&pdev->dev, &spdif_in_dai);
+ if (ret != 0) {
+ clk_put(host->clk);
+ return ret;
+ }
+
+ return 0;
+}
+
+static int spdif_in_remove(struct platform_device *pdev)
+{
+ struct spdif_in_dev *host = dev_get_drvdata(&pdev->dev);
+
+ snd_soc_unregister_dai(&pdev->dev);
+ dev_set_drvdata(&pdev->dev, NULL);
+
+ clk_put(host->clk);
+
+ return 0;
+}
+
+
+static struct platform_driver spdif_in_driver = {
+ .probe = spdif_in_probe,
+ .remove = spdif_in_remove,
+ .driver = {
+ .name = "spdif-in",
+ .owner = THIS_MODULE,
+ },
+};
+
+module_platform_driver(spdif_in_driver);
+
+MODULE_AUTHOR("Vipin Kumar <vipin.kumar@st.com>");
+MODULE_DESCRIPTION("SPEAr SPDIF IN SoC Interface");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:spdif_in");
diff --git a/sound/soc/spear/spdif_in_regs.h b/sound/soc/spear/spdif_in_regs.h
new file mode 100644
index 0000000..37af7bc
--- /dev/null
+++ b/sound/soc/spear/spdif_in_regs.h
@@ -0,0 +1,60 @@
+/*
+ * SPEAr SPDIF IN controller header file
+ *
+ * Copyright (ST) 2011 Vipin Kumar (vipin.kumar@st.com)
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#ifndef SPDIF_IN_REGS_H
+#define SPDIF_IN_REGS_H
+
+#define SPDIF_IN_CTRL 0x00
+ #define SPDIF_IN_PRTYEN (1 << 20)
+ #define SPDIF_IN_STATEN (1 << 19)
+ #define SPDIF_IN_USREN (1 << 18)
+ #define SPDIF_IN_VALEN (1 << 17)
+ #define SPDIF_IN_BLKEN (1 << 16)
+
+ #define SPDIF_MODE_24BIT (8 << 12)
+ #define SPDIF_MODE_23BIT (7 << 12)
+ #define SPDIF_MODE_22BIT (6 << 12)
+ #define SPDIF_MODE_21BIT (5 << 12)
+ #define SPDIF_MODE_20BIT (4 << 12)
+ #define SPDIF_MODE_19BIT (3 << 12)
+ #define SPDIF_MODE_18BIT (2 << 12)
+ #define SPDIF_MODE_17BIT (1 << 12)
+ #define SPDIF_MODE_16BIT (0 << 12)
+ #define SPDIF_MODE_MASK (0x0F << 12)
+
+ #define SPDIF_IN_VALID (1 << 11)
+ #define SPDIF_IN_SAMPLE (1 << 10)
+ #define SPDIF_DATA_SWAP (1 << 9)
+ #define SPDIF_IN_ENB (1 << 8)
+ #define SPDIF_DATA_REVERT (1 << 7)
+ #define SPDIF_XTRACT_16BIT (1 << 6)
+ #define SPDIF_FIFO_THRES_16 (16 << 0)
+
+#define SPDIF_IN_IRQ_MASK 0x04
+#define SPDIF_IN_IRQ 0x08
+ #define SPDIF_IRQ_FIFOWRITE (1 << 0)
+ #define SPDIF_IRQ_EMPTYFIFOREAD (1 << 1)
+ #define SPDIF_IRQ_FIFOFULL (1 << 2)
+ #define SPDIF_IRQ_OUTOFRANGE (1 << 3)
+
+#define SPDIF_IN_STA 0x0C
+ #define SPDIF_IN_LOCK (0x1 << 0)
+
+#endif /* SPDIF_IN_REGS_H */
diff --git a/sound/soc/spear/spdif_out.c b/sound/soc/spear/spdif_out.c
new file mode 100644
index 0000000..5eac4cd
--- /dev/null
+++ b/sound/soc/spear/spdif_out.c
@@ -0,0 +1,389 @@
+/*
+ * ALSA SoC SPDIF Out Audio Layer for spear processors
+ *
+ * Copyright (C) 2012 ST Microelectronics
+ * Vipin Kumar <vipin.kumar@st.com>
+ *
+ * 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 <linux/clk.h>
+#include <linux/delay.h>
+#include <linux/device.h>
+#include <linux/kernel.h>
+#include <linux/init.h>
+#include <linux/io.h>
+#include <linux/ioport.h>
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <sound/soc.h>
+#include <sound/spear_dma.h>
+#include <sound/spear_spdif.h>
+#include "spdif_out_regs.h"
+
+struct spdif_out_params {
+ u32 rate;
+ u32 core_freq;
+ u32 mute;
+};
+
+struct spdif_out_dev {
+ struct clk *clk;
+ struct spear_dma_data dma_params;
+ struct spdif_out_params saved_params;
+ u32 running;
+ void __iomem *io_base;
+};
+
+static void spdif_out_configure(struct spdif_out_dev *host)
+{
+ writel(SPDIF_OUT_RESET, host->io_base + SPDIF_OUT_SOFT_RST);
+ mdelay(1);
+ writel(readl(host->io_base + SPDIF_OUT_SOFT_RST) & ~SPDIF_OUT_RESET,
+ host->io_base + SPDIF_OUT_SOFT_RST);
+
+ writel(SPDIF_OUT_FDMA_TRIG_16 | SPDIF_OUT_MEMFMT_16_16 |
+ SPDIF_OUT_VALID_HW | SPDIF_OUT_USER_HW |
+ SPDIF_OUT_CHNLSTA_HW | SPDIF_OUT_PARITY_HW,
+ host->io_base + SPDIF_OUT_CFG);
+
+ writel(0x7F, host->io_base + SPDIF_OUT_INT_STA_CLR);
+ writel(0x7F, host->io_base + SPDIF_OUT_INT_EN_CLR);
+}
+
+static int spdif_out_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *cpu_dai)
+{
+ struct spdif_out_dev *host = snd_soc_dai_get_drvdata(cpu_dai);
+ int ret;
+
+ if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK)
+ return -EINVAL;
+
+ snd_soc_dai_set_dma_data(cpu_dai, substream, (void *)&host->dma_params);
+
+ ret = clk_enable(host->clk);
+ if (ret)
+ return ret;
+
+ host->running = true;
+ spdif_out_configure(host);
+
+ return 0;
+}
+
+static void spdif_out_shutdown(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct spdif_out_dev *host = snd_soc_dai_get_drvdata(dai);
+
+ if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK)
+ return;
+
+ clk_disable(host->clk);
+ host->running = false;
+ snd_soc_dai_set_dma_data(dai, substream, NULL);
+}
+
+static void spdif_out_clock(struct spdif_out_dev *host, u32 core_freq,
+ u32 rate)
+{
+ u32 divider, ctrl;
+
+ clk_set_rate(host->clk, core_freq);
+ divider = DIV_ROUND_CLOSEST(clk_get_rate(host->clk), (rate * 128));
+
+ ctrl = readl(host->io_base + SPDIF_OUT_CTRL);
+ ctrl &= ~SPDIF_DIVIDER_MASK;
+ ctrl |= (divider << SPDIF_DIVIDER_SHIFT) & SPDIF_DIVIDER_MASK;
+ writel(ctrl, host->io_base + SPDIF_OUT_CTRL);
+}
+
+static int spdif_out_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct spdif_out_dev *host = snd_soc_dai_get_drvdata(dai);
+ u32 rate, core_freq;
+
+ if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK)
+ return -EINVAL;
+
+ rate = params_rate(params);
+
+ switch (rate) {
+ case 8000:
+ case 16000:
+ case 32000:
+ case 64000:
+ /*
+ * The clock is multiplied by 10 to bring it to feasible range
+ * of frequencies for sscg
+ */
+ core_freq = 64000 * 128 * 10; /* 81.92 MHz */
+ break;
+ case 5512:
+ case 11025:
+ case 22050:
+ case 44100:
+ case 88200:
+ case 176400:
+ core_freq = 176400 * 128; /* 22.5792 MHz */
+ break;
+ case 48000:
+ case 96000:
+ case 192000:
+ default:
+ core_freq = 192000 * 128; /* 24.576 MHz */
+ break;
+ }
+
+ spdif_out_clock(host, core_freq, rate);
+ host->saved_params.core_freq = core_freq;
+ host->saved_params.rate = rate;
+
+ return 0;
+}
+
+static int spdif_out_trigger(struct snd_pcm_substream *substream, int cmd,
+ struct snd_soc_dai *dai)
+{
+ struct spdif_out_dev *host = snd_soc_dai_get_drvdata(dai);
+ u32 ctrl;
+ int ret = 0;
+
+ if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK)
+ return -EINVAL;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ ctrl = readl(host->io_base + SPDIF_OUT_CTRL);
+ ctrl &= ~SPDIF_OPMODE_MASK;
+ if (!host->saved_params.mute)
+ ctrl |= SPDIF_OPMODE_AUD_DATA |
+ SPDIF_STATE_NORMAL;
+ else
+ ctrl |= SPDIF_OPMODE_MUTE_PCM;
+ writel(ctrl, host->io_base + SPDIF_OUT_CTRL);
+ break;
+
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ ctrl = readl(host->io_base + SPDIF_OUT_CTRL);
+ ctrl &= ~SPDIF_OPMODE_MASK;
+ ctrl |= SPDIF_OPMODE_OFF;
+ writel(ctrl, host->io_base + SPDIF_OUT_CTRL);
+ break;
+
+ default:
+ ret = -EINVAL;
+ break;
+ }
+ return ret;
+}
+
+static int spdif_digital_mute(struct snd_soc_dai *dai, int mute)
+{
+ struct spdif_out_dev *host = snd_soc_dai_get_drvdata(dai);
+ u32 val;
+
+ host->saved_params.mute = mute;
+ val = readl(host->io_base + SPDIF_OUT_CTRL);
+ val &= ~SPDIF_OPMODE_MASK;
+
+ if (mute)
+ val |= SPDIF_OPMODE_MUTE_PCM;
+ else {
+ if (host->running)
+ val |= SPDIF_OPMODE_AUD_DATA | SPDIF_STATE_NORMAL;
+ else
+ val |= SPDIF_OPMODE_OFF;
+ }
+
+ writel(val, host->io_base + SPDIF_OUT_CTRL);
+ return 0;
+}
+
+static int spdif_mute_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct snd_soc_card *card = codec->card;
+ struct snd_soc_pcm_runtime *rtd = card->rtd;
+ struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct spdif_out_dev *host = snd_soc_dai_get_drvdata(cpu_dai);
+
+ ucontrol->value.integer.value[0] = host->saved_params.mute;
+ return 0;
+}
+
+static int spdif_mute_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct snd_soc_card *card = codec->card;
+ struct snd_soc_pcm_runtime *rtd = card->rtd;
+ struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct spdif_out_dev *host = snd_soc_dai_get_drvdata(cpu_dai);
+
+ if (host->saved_params.mute == ucontrol->value.integer.value[0])
+ return 0;
+
+ spdif_digital_mute(cpu_dai, ucontrol->value.integer.value[0]);
+
+ return 1;
+}
+static const struct snd_kcontrol_new spdif_out_controls[] = {
+ SOC_SINGLE_BOOL_EXT("IEC958 Playback Switch", 0,
+ spdif_mute_get, spdif_mute_put),
+};
+
+int spdif_soc_dai_probe(struct snd_soc_dai *dai)
+{
+ return snd_soc_add_dai_controls(dai, spdif_out_controls,
+ ARRAY_SIZE(spdif_out_controls));
+}
+
+static const struct snd_soc_dai_ops spdif_out_dai_ops = {
+ .digital_mute = spdif_digital_mute,
+ .startup = spdif_out_startup,
+ .shutdown = spdif_out_shutdown,
+ .trigger = spdif_out_trigger,
+ .hw_params = spdif_out_hw_params,
+};
+
+static struct snd_soc_dai_driver spdif_out_dai = {
+ .playback = {
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \
+ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000 | \
+ SNDRV_PCM_RATE_192000),
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ },
+ .probe = spdif_soc_dai_probe,
+ .ops = &spdif_out_dai_ops,
+};
+
+static int spdif_out_probe(struct platform_device *pdev)
+{
+ struct spdif_out_dev *host;
+ struct spear_spdif_platform_data *pdata;
+ struct resource *res;
+ int ret;
+
+ res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+ if (!res)
+ return -EINVAL;
+
+ if (!devm_request_mem_region(&pdev->dev, res->start,
+ resource_size(res), pdev->name)) {
+ dev_warn(&pdev->dev, "Failed to get memory resourse\n");
+ return -ENOENT;
+ }
+
+ host = devm_kzalloc(&pdev->dev, sizeof(*host), GFP_KERNEL);
+ if (!host) {
+ dev_warn(&pdev->dev, "kzalloc fail\n");
+ return -ENOMEM;
+ }
+
+ host->io_base = devm_ioremap(&pdev->dev, res->start,
+ resource_size(res));
+ if (!host->io_base) {
+ dev_warn(&pdev->dev, "ioremap failed\n");
+ return -ENOMEM;
+ }
+
+ host->clk = clk_get(&pdev->dev, NULL);
+ if (IS_ERR(host->clk))
+ return PTR_ERR(host->clk);
+
+ pdata = dev_get_platdata(&pdev->dev);
+
+ host->dma_params.data = pdata->dma_params;
+ host->dma_params.addr = res->start + SPDIF_OUT_FIFO_DATA;
+ host->dma_params.max_burst = 16;
+ host->dma_params.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES;
+ host->dma_params.filter = pdata->filter;
+
+ dev_set_drvdata(&pdev->dev, host);
+
+ ret = snd_soc_register_dai(&pdev->dev, &spdif_out_dai);
+ if (ret != 0) {
+ clk_put(host->clk);
+ return ret;
+ }
+
+ return 0;
+}
+
+static int spdif_out_remove(struct platform_device *pdev)
+{
+ struct spdif_out_dev *host = dev_get_drvdata(&pdev->dev);
+
+ snd_soc_unregister_dai(&pdev->dev);
+ dev_set_drvdata(&pdev->dev, NULL);
+
+ clk_put(host->clk);
+
+ return 0;
+}
+
+#ifdef CONFIG_PM
+static int spdif_out_suspend(struct device *dev)
+{
+ struct platform_device *pdev = to_platform_device(dev);
+ struct spdif_out_dev *host = dev_get_drvdata(&pdev->dev);
+
+ if (host->running)
+ clk_disable(host->clk);
+
+ return 0;
+}
+
+static int spdif_out_resume(struct device *dev)
+{
+ struct platform_device *pdev = to_platform_device(dev);
+ struct spdif_out_dev *host = dev_get_drvdata(&pdev->dev);
+
+ if (host->running) {
+ clk_enable(host->clk);
+ spdif_out_configure(host);
+ spdif_out_clock(host, host->saved_params.core_freq,
+ host->saved_params.rate);
+ }
+ return 0;
+}
+
+static SIMPLE_DEV_PM_OPS(spdif_out_dev_pm_ops, spdif_out_suspend, \
+ spdif_out_resume);
+
+#define SPDIF_OUT_DEV_PM_OPS (&spdif_out_dev_pm_ops)
+
+#else
+#define SPDIF_OUT_DEV_PM_OPS NULL
+
+#endif
+
+static struct platform_driver spdif_out_driver = {
+ .probe = spdif_out_probe,
+ .remove = spdif_out_remove,
+ .driver = {
+ .name = "spdif-out",
+ .owner = THIS_MODULE,
+ .pm = SPDIF_OUT_DEV_PM_OPS,
+ },
+};
+
+module_platform_driver(spdif_out_driver);
+
+MODULE_AUTHOR("Vipin Kumar <vipin.kumar@st.com>");
+MODULE_DESCRIPTION("SPEAr SPDIF OUT SoC Interface");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:spdif_out");
diff --git a/sound/soc/spear/spdif_out_regs.h b/sound/soc/spear/spdif_out_regs.h
new file mode 100644
index 0000000..a5e5332
--- /dev/null
+++ b/sound/soc/spear/spdif_out_regs.h
@@ -0,0 +1,79 @@
+/*
+ * SPEAr SPDIF OUT controller header file
+ *
+ * Copyright (ST) 2011 Vipin Kumar (vipin.kumar@st.com)
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#ifndef SPDIF_OUT_REGS_H
+#define SPDIF_OUT_REGS_H
+
+#define SPDIF_OUT_SOFT_RST 0x00
+ #define SPDIF_OUT_RESET (1 << 0)
+#define SPDIF_OUT_FIFO_DATA 0x04
+#define SPDIF_OUT_INT_STA 0x08
+#define SPDIF_OUT_INT_STA_CLR 0x0C
+ #define SPDIF_INT_UNDERFLOW (1 << 0)
+ #define SPDIF_INT_EODATA (1 << 1)
+ #define SPDIF_INT_EOBLOCK (1 << 2)
+ #define SPDIF_INT_EOLATENCY (1 << 3)
+ #define SPDIF_INT_EOPD_DATA (1 << 4)
+ #define SPDIF_INT_MEMFULLREAD (1 << 5)
+ #define SPDIF_INT_EOPD_PAUSE (1 << 6)
+
+#define SPDIF_OUT_INT_EN 0x10
+#define SPDIF_OUT_INT_EN_SET 0x14
+#define SPDIF_OUT_INT_EN_CLR 0x18
+#define SPDIF_OUT_CTRL 0x1C
+ #define SPDIF_OPMODE_MASK (7 << 0)
+ #define SPDIF_OPMODE_OFF (0 << 0)
+ #define SPDIF_OPMODE_MUTE_PCM (1 << 0)
+ #define SPDIF_OPMODE_MUTE_PAUSE (2 << 0)
+ #define SPDIF_OPMODE_AUD_DATA (3 << 0)
+ #define SPDIF_OPMODE_ENCODE (4 << 0)
+ #define SPDIF_STATE_NORMAL (1 << 3)
+ #define SPDIF_DIVIDER_MASK (0xff << 5)
+ #define SPDIF_DIVIDER_SHIFT (5)
+ #define SPDIF_SAMPLEREAD_MASK (0x1ffff << 15)
+ #define SPDIF_SAMPLEREAD_SHIFT (15)
+#define SPDIF_OUT_STA 0x20
+#define SPDIF_OUT_PA_PB 0x24
+#define SPDIF_OUT_PC_PD 0x28
+#define SPDIF_OUT_CL1 0x2C
+#define SPDIF_OUT_CR1 0x30
+#define SPDIF_OUT_CL2_CR2_UV 0x34
+#define SPDIF_OUT_PAUSE_LAT 0x38
+#define SPDIF_OUT_FRMLEN_BRST 0x3C
+#define SPDIF_OUT_CFG 0x40
+ #define SPDIF_OUT_MEMFMT_16_0 (0 << 5)
+ #define SPDIF_OUT_MEMFMT_16_16 (1 << 5)
+ #define SPDIF_OUT_VALID_DMA (0 << 3)
+ #define SPDIF_OUT_VALID_HW (1 << 3)
+ #define SPDIF_OUT_USER_DMA (0 << 2)
+ #define SPDIF_OUT_USER_HW (1 << 2)
+ #define SPDIF_OUT_CHNLSTA_DMA (0 << 1)
+ #define SPDIF_OUT_CHNLSTA_HW (1 << 1)
+ #define SPDIF_OUT_PARITY_HW (0 << 0)
+ #define SPDIF_OUT_PARITY_DMA (1 << 0)
+ #define SPDIF_OUT_FDMA_TRIG_2 (2 << 8)
+ #define SPDIF_OUT_FDMA_TRIG_6 (6 << 8)
+ #define SPDIF_OUT_FDMA_TRIG_8 (8 << 8)
+ #define SPDIF_OUT_FDMA_TRIG_10 (10 << 8)
+ #define SPDIF_OUT_FDMA_TRIG_12 (12 << 8)
+ #define SPDIF_OUT_FDMA_TRIG_16 (16 << 8)
+ #define SPDIF_OUT_FDMA_TRIG_18 (18 << 8)
+
+#endif /* SPDIF_OUT_REGS_H */
diff --git a/sound/soc/spear/spear_pcm.c b/sound/soc/spear/spear_pcm.c
new file mode 100644
index 0000000..97c2cac
--- /dev/null
+++ b/sound/soc/spear/spear_pcm.c
@@ -0,0 +1,214 @@
+/*
+ * ALSA PCM interface for ST SPEAr Processors
+ *
+ * sound/soc/spear/spear_pcm.c
+ *
+ * Copyright (C) 2012 ST Microelectronics
+ * Rajeev Kumar<rajeev-dlh.kumar@st.com>
+ *
+ * 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 <linux/module.h>
+#include <linux/dmaengine.h>
+#include <linux/dma-mapping.h>
+#include <linux/init.h>
+#include <linux/platform_device.h>
+#include <linux/scatterlist.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/dmaengine_pcm.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/spear_dma.h>
+
+struct snd_pcm_hardware spear_pcm_hardware = {
+ .info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME),
+ .buffer_bytes_max = 16 * 1024, /* max buffer size */
+ .period_bytes_min = 2 * 1024, /* 1 msec data minimum period size */
+ .period_bytes_max = 2 * 1024, /* maximum period size */
+ .periods_min = 1, /* min # periods */
+ .periods_max = 8, /* max # of periods */
+ .fifo_size = 0, /* fifo size in bytes */
+};
+
+static int spear_pcm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
+
+ return 0;
+}
+
+static int spear_pcm_hw_free(struct snd_pcm_substream *substream)
+{
+ snd_pcm_set_runtime_buffer(substream, NULL);
+
+ return 0;
+}
+
+static int spear_pcm_open(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+
+ struct spear_dma_data *dma_data = (struct spear_dma_data *)
+ snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
+ int ret;
+
+ ret = snd_soc_set_runtime_hwparams(substream, &spear_pcm_hardware);
+ if (ret)
+ return ret;
+
+ ret = snd_dmaengine_pcm_open(substream, dma_data->filter, dma_data);
+ if (ret)
+ return ret;
+
+ snd_dmaengine_pcm_set_data(substream, dma_data);
+
+ return 0;
+}
+
+static int spear_pcm_close(struct snd_pcm_substream *substream)
+{
+
+ snd_dmaengine_pcm_close(substream);
+
+ return 0;
+}
+
+static int spear_pcm_mmap(struct snd_pcm_substream *substream,
+ struct vm_area_struct *vma)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ return dma_mmap_writecombine(substream->pcm->card->dev, vma,
+ runtime->dma_area, runtime->dma_addr,
+ runtime->dma_bytes);
+}
+
+static struct snd_pcm_ops spear_pcm_ops = {
+ .open = spear_pcm_open,
+ .close = spear_pcm_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = spear_pcm_hw_params,
+ .hw_free = spear_pcm_hw_free,
+ .trigger = snd_dmaengine_pcm_trigger,
+ .pointer = snd_dmaengine_pcm_pointer,
+ .mmap = spear_pcm_mmap,
+};
+
+static int
+spear_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream,
+ size_t size)
+{
+ struct snd_pcm_substream *substream = pcm->streams[stream].substream;
+ struct snd_dma_buffer *buf = &substream->dma_buffer;
+
+ buf->dev.type = SNDRV_DMA_TYPE_DEV;
+ buf->dev.dev = pcm->card->dev;
+ buf->private_data = NULL;
+
+ buf->area = dma_alloc_writecombine(pcm->card->dev, size,
+ &buf->addr, GFP_KERNEL);
+ if (!buf->area)
+ return -ENOMEM;
+
+ dev_info(buf->dev.dev,
+ " preallocate_dma_buffer: area=%p, addr=%p, size=%d\n",
+ (void *)buf->area, (void *)buf->addr, size);
+
+ buf->bytes = size;
+ return 0;
+}
+
+static void spear_pcm_free(struct snd_pcm *pcm)
+{
+ struct snd_pcm_substream *substream;
+ struct snd_dma_buffer *buf;
+ int stream;
+
+ for (stream = 0; stream < 2; stream++) {
+ substream = pcm->streams[stream].substream;
+ if (!substream)
+ continue;
+
+ buf = &substream->dma_buffer;
+ if (!buf && !buf->area)
+ continue;
+
+ dma_free_writecombine(pcm->card->dev, buf->bytes,
+ buf->area, buf->addr);
+ buf->area = NULL;
+ }
+}
+
+static u64 spear_pcm_dmamask = DMA_BIT_MASK(32);
+
+static int spear_pcm_new(struct snd_card *card,
+ struct snd_soc_dai *dai, struct snd_pcm *pcm)
+{
+ int ret;
+
+ if (!card->dev->dma_mask)
+ card->dev->dma_mask = &spear_pcm_dmamask;
+ if (!card->dev->coherent_dma_mask)
+ card->dev->coherent_dma_mask = DMA_BIT_MASK(32);
+
+ if (dai->driver->playback.channels_min) {
+ ret = spear_pcm_preallocate_dma_buffer(pcm,
+ SNDRV_PCM_STREAM_PLAYBACK,
+ spear_pcm_hardware.buffer_bytes_max);
+ if (ret)
+ return ret;
+ }
+
+ if (dai->driver->capture.channels_min) {
+ ret = spear_pcm_preallocate_dma_buffer(pcm,
+ SNDRV_PCM_STREAM_CAPTURE,
+ spear_pcm_hardware.buffer_bytes_max);
+ if (ret)
+ return ret;
+ }
+
+ return 0;
+}
+
+struct snd_soc_platform_driver spear_soc_platform = {
+ .ops = &spear_pcm_ops,
+ .pcm_new = spear_pcm_new,
+ .pcm_free = spear_pcm_free,
+};
+
+static int __devinit spear_soc_platform_probe(struct platform_device *pdev)
+{
+ return snd_soc_register_platform(&pdev->dev, &spear_soc_platform);
+}
+
+static int __devexit spear_soc_platform_remove(struct platform_device *pdev)
+{
+ snd_soc_unregister_platform(&pdev->dev);
+
+ return 0;
+}
+
+static struct platform_driver spear_pcm_driver = {
+ .driver = {
+ .name = "spear-pcm-audio",
+ .owner = THIS_MODULE,
+ },
+
+ .probe = spear_soc_platform_probe,
+ .remove = __devexit_p(spear_soc_platform_remove),
+};
+
+module_platform_driver(spear_pcm_driver);
+
+MODULE_AUTHOR("Rajeev Kumar <rajeev-dlh.kumar@st.com>");
+MODULE_DESCRIPTION("SPEAr PCM DMA module");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:spear-pcm-audio");
diff --git a/sound/soc/tegra/Kconfig b/sound/soc/tegra/Kconfig
index 76dc230..02bcd30 100644
--- a/sound/soc/tegra/Kconfig
+++ b/sound/soc/tegra/Kconfig
@@ -1,7 +1,8 @@
config SND_SOC_TEGRA
tristate "SoC Audio for the Tegra System-on-Chip"
- depends on ARCH_TEGRA && TEGRA_SYSTEM_DMA
+ depends on ARCH_TEGRA && (TEGRA_SYSTEM_DMA || TEGRA20_APB_DMA)
select REGMAP_MMIO
+ select SND_SOC_DMAENGINE_PCM if TEGRA20_APB_DMA
help
Say Y or M here if you want support for SoC audio on Tegra.
diff --git a/sound/soc/tegra/tegra20_i2s.c b/sound/soc/tegra/tegra20_i2s.c
index 1647dbf..0832e8a 100644
--- a/sound/soc/tegra/tegra20_i2s.c
+++ b/sound/soc/tegra/tegra20_i2s.c
@@ -46,18 +46,6 @@
#define DRV_NAME "tegra20-i2s"
-static inline void tegra20_i2s_write(struct tegra20_i2s *i2s, u32 reg, u32 val)
-{
- regmap_write(i2s->regmap, reg, val);
-}
-
-static inline u32 tegra20_i2s_read(struct tegra20_i2s *i2s, u32 reg)
-{
- u32 val;
- regmap_read(i2s->regmap, reg, &val);
- return val;
-}
-
static int tegra20_i2s_runtime_suspend(struct device *dev)
{
struct tegra20_i2s *i2s = dev_get_drvdata(dev);
@@ -85,6 +73,7 @@ static int tegra20_i2s_set_fmt(struct snd_soc_dai *dai,
unsigned int fmt)
{
struct tegra20_i2s *i2s = snd_soc_dai_get_drvdata(dai);
+ unsigned int mask, val;
switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
case SND_SOC_DAIFMT_NB_NF:
@@ -93,10 +82,10 @@ static int tegra20_i2s_set_fmt(struct snd_soc_dai *dai,
return -EINVAL;
}
- i2s->reg_ctrl &= ~TEGRA20_I2S_CTRL_MASTER_ENABLE;
+ mask = TEGRA20_I2S_CTRL_MASTER_ENABLE;
switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
case SND_SOC_DAIFMT_CBS_CFS:
- i2s->reg_ctrl |= TEGRA20_I2S_CTRL_MASTER_ENABLE;
+ val = TEGRA20_I2S_CTRL_MASTER_ENABLE;
break;
case SND_SOC_DAIFMT_CBM_CFM:
break;
@@ -104,33 +93,35 @@ static int tegra20_i2s_set_fmt(struct snd_soc_dai *dai,
return -EINVAL;
}
- i2s->reg_ctrl &= ~(TEGRA20_I2S_CTRL_BIT_FORMAT_MASK |
- TEGRA20_I2S_CTRL_LRCK_MASK);
+ mask |= TEGRA20_I2S_CTRL_BIT_FORMAT_MASK |
+ TEGRA20_I2S_CTRL_LRCK_MASK;
switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
case SND_SOC_DAIFMT_DSP_A:
- i2s->reg_ctrl |= TEGRA20_I2S_CTRL_BIT_FORMAT_DSP;
- i2s->reg_ctrl |= TEGRA20_I2S_CTRL_LRCK_L_LOW;
+ val |= TEGRA20_I2S_CTRL_BIT_FORMAT_DSP;
+ val |= TEGRA20_I2S_CTRL_LRCK_L_LOW;
break;
case SND_SOC_DAIFMT_DSP_B:
- i2s->reg_ctrl |= TEGRA20_I2S_CTRL_BIT_FORMAT_DSP;
- i2s->reg_ctrl |= TEGRA20_I2S_CTRL_LRCK_R_LOW;
+ val |= TEGRA20_I2S_CTRL_BIT_FORMAT_DSP;
+ val |= TEGRA20_I2S_CTRL_LRCK_R_LOW;
break;
case SND_SOC_DAIFMT_I2S:
- i2s->reg_ctrl |= TEGRA20_I2S_CTRL_BIT_FORMAT_I2S;
- i2s->reg_ctrl |= TEGRA20_I2S_CTRL_LRCK_L_LOW;
+ val |= TEGRA20_I2S_CTRL_BIT_FORMAT_I2S;
+ val |= TEGRA20_I2S_CTRL_LRCK_L_LOW;
break;
case SND_SOC_DAIFMT_RIGHT_J:
- i2s->reg_ctrl |= TEGRA20_I2S_CTRL_BIT_FORMAT_RJM;
- i2s->reg_ctrl |= TEGRA20_I2S_CTRL_LRCK_L_LOW;
+ val |= TEGRA20_I2S_CTRL_BIT_FORMAT_RJM;
+ val |= TEGRA20_I2S_CTRL_LRCK_L_LOW;
break;
case SND_SOC_DAIFMT_LEFT_J:
- i2s->reg_ctrl |= TEGRA20_I2S_CTRL_BIT_FORMAT_LJM;
- i2s->reg_ctrl |= TEGRA20_I2S_CTRL_LRCK_L_LOW;
+ val |= TEGRA20_I2S_CTRL_BIT_FORMAT_LJM;
+ val |= TEGRA20_I2S_CTRL_LRCK_L_LOW;
break;
default:
return -EINVAL;
}
+ regmap_update_bits(i2s->regmap, TEGRA20_I2S_CTRL, mask, val);
+
return 0;
}
@@ -138,29 +129,34 @@ static int tegra20_i2s_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
{
- struct device *dev = substream->pcm->card->dev;
+ struct device *dev = dai->dev;
struct tegra20_i2s *i2s = snd_soc_dai_get_drvdata(dai);
- u32 reg;
+ unsigned int mask, val;
int ret, sample_size, srate, i2sclock, bitcnt;
- i2s->reg_ctrl &= ~TEGRA20_I2S_CTRL_BIT_SIZE_MASK;
+ mask = TEGRA20_I2S_CTRL_BIT_SIZE_MASK;
switch (params_format(params)) {
case SNDRV_PCM_FORMAT_S16_LE:
- i2s->reg_ctrl |= TEGRA20_I2S_CTRL_BIT_SIZE_16;
+ val = TEGRA20_I2S_CTRL_BIT_SIZE_16;
sample_size = 16;
break;
case SNDRV_PCM_FORMAT_S24_LE:
- i2s->reg_ctrl |= TEGRA20_I2S_CTRL_BIT_SIZE_24;
+ val = TEGRA20_I2S_CTRL_BIT_SIZE_24;
sample_size = 24;
break;
case SNDRV_PCM_FORMAT_S32_LE:
- i2s->reg_ctrl |= TEGRA20_I2S_CTRL_BIT_SIZE_32;
+ val = TEGRA20_I2S_CTRL_BIT_SIZE_32;
sample_size = 32;
break;
default:
return -EINVAL;
}
+ mask |= TEGRA20_I2S_CTRL_FIFO_FORMAT_MASK;
+ val |= TEGRA20_I2S_CTRL_FIFO_FORMAT_PACKED;
+
+ regmap_update_bits(i2s->regmap, TEGRA20_I2S_CTRL, mask, val);
+
srate = params_rate(params);
/* Final "* 2" required by Tegra hardware */
@@ -175,42 +171,44 @@ static int tegra20_i2s_hw_params(struct snd_pcm_substream *substream,
bitcnt = (i2sclock / (2 * srate)) - 1;
if (bitcnt < 0 || bitcnt > TEGRA20_I2S_TIMING_CHANNEL_BIT_COUNT_MASK_US)
return -EINVAL;
- reg = bitcnt << TEGRA20_I2S_TIMING_CHANNEL_BIT_COUNT_SHIFT;
+ val = bitcnt << TEGRA20_I2S_TIMING_CHANNEL_BIT_COUNT_SHIFT;
if (i2sclock % (2 * srate))
- reg |= TEGRA20_I2S_TIMING_NON_SYM_ENABLE;
+ val |= TEGRA20_I2S_TIMING_NON_SYM_ENABLE;
- tegra20_i2s_write(i2s, TEGRA20_I2S_TIMING, reg);
+ regmap_write(i2s->regmap, TEGRA20_I2S_TIMING, val);
- tegra20_i2s_write(i2s, TEGRA20_I2S_FIFO_SCR,
- TEGRA20_I2S_FIFO_SCR_FIFO2_ATN_LVL_FOUR_SLOTS |
- TEGRA20_I2S_FIFO_SCR_FIFO1_ATN_LVL_FOUR_SLOTS);
+ regmap_write(i2s->regmap, TEGRA20_I2S_FIFO_SCR,
+ TEGRA20_I2S_FIFO_SCR_FIFO2_ATN_LVL_FOUR_SLOTS |
+ TEGRA20_I2S_FIFO_SCR_FIFO1_ATN_LVL_FOUR_SLOTS);
return 0;
}
static void tegra20_i2s_start_playback(struct tegra20_i2s *i2s)
{
- i2s->reg_ctrl |= TEGRA20_I2S_CTRL_FIFO1_ENABLE;
- tegra20_i2s_write(i2s, TEGRA20_I2S_CTRL, i2s->reg_ctrl);
+ regmap_update_bits(i2s->regmap, TEGRA20_I2S_CTRL,
+ TEGRA20_I2S_CTRL_FIFO1_ENABLE,
+ TEGRA20_I2S_CTRL_FIFO1_ENABLE);
}
static void tegra20_i2s_stop_playback(struct tegra20_i2s *i2s)
{
- i2s->reg_ctrl &= ~TEGRA20_I2S_CTRL_FIFO1_ENABLE;
- tegra20_i2s_write(i2s, TEGRA20_I2S_CTRL, i2s->reg_ctrl);
+ regmap_update_bits(i2s->regmap, TEGRA20_I2S_CTRL,
+ TEGRA20_I2S_CTRL_FIFO1_ENABLE, 0);
}
static void tegra20_i2s_start_capture(struct tegra20_i2s *i2s)
{
- i2s->reg_ctrl |= TEGRA20_I2S_CTRL_FIFO2_ENABLE;
- tegra20_i2s_write(i2s, TEGRA20_I2S_CTRL, i2s->reg_ctrl);
+ regmap_update_bits(i2s->regmap, TEGRA20_I2S_CTRL,
+ TEGRA20_I2S_CTRL_FIFO2_ENABLE,
+ TEGRA20_I2S_CTRL_FIFO2_ENABLE);
}
static void tegra20_i2s_stop_capture(struct tegra20_i2s *i2s)
{
- i2s->reg_ctrl &= ~TEGRA20_I2S_CTRL_FIFO2_ENABLE;
- tegra20_i2s_write(i2s, TEGRA20_I2S_CTRL, i2s->reg_ctrl);
+ regmap_update_bits(i2s->regmap, TEGRA20_I2S_CTRL,
+ TEGRA20_I2S_CTRL_FIFO2_ENABLE, 0);
}
static int tegra20_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
@@ -261,12 +259,14 @@ static const struct snd_soc_dai_ops tegra20_i2s_dai_ops = {
static const struct snd_soc_dai_driver tegra20_i2s_dai_template = {
.probe = tegra20_i2s_probe,
.playback = {
+ .stream_name = "Playback",
.channels_min = 2,
.channels_max = 2,
.rates = SNDRV_PCM_RATE_8000_96000,
.formats = SNDRV_PCM_FMTBIT_S16_LE,
},
.capture = {
+ .stream_name = "Capture",
.channels_min = 2,
.channels_max = 2,
.rates = SNDRV_PCM_RATE_8000_96000,
@@ -412,8 +412,6 @@ static __devinit int tegra20_i2s_platform_probe(struct platform_device *pdev)
i2s->playback_dma_data.width = 32;
i2s->playback_dma_data.req_sel = dma_ch;
- i2s->reg_ctrl = TEGRA20_I2S_CTRL_FIFO_FORMAT_PACKED;
-
pm_runtime_enable(&pdev->dev);
if (!pm_runtime_enabled(&pdev->dev)) {
ret = tegra20_i2s_runtime_resume(&pdev->dev);
diff --git a/sound/soc/tegra/tegra20_i2s.h b/sound/soc/tegra/tegra20_i2s.h
index a57efc6..c27069d 100644
--- a/sound/soc/tegra/tegra20_i2s.h
+++ b/sound/soc/tegra/tegra20_i2s.h
@@ -158,7 +158,6 @@ struct tegra20_i2s {
struct tegra_pcm_dma_params capture_dma_data;
struct tegra_pcm_dma_params playback_dma_data;
struct regmap *regmap;
- u32 reg_ctrl;
};
#endif
diff --git a/sound/soc/tegra/tegra20_spdif.c b/sound/soc/tegra/tegra20_spdif.c
index 2262e4f..3ebc867 100644
--- a/sound/soc/tegra/tegra20_spdif.c
+++ b/sound/soc/tegra/tegra20_spdif.c
@@ -37,19 +37,6 @@
#define DRV_NAME "tegra20-spdif"
-static inline void tegra20_spdif_write(struct tegra20_spdif *spdif, u32 reg,
- u32 val)
-{
- regmap_write(spdif->regmap, reg, val);
-}
-
-static inline u32 tegra20_spdif_read(struct tegra20_spdif *spdif, u32 reg)
-{
- u32 val;
- regmap_read(spdif->regmap, reg, &val);
- return val;
-}
-
static int tegra20_spdif_runtime_suspend(struct device *dev)
{
struct tegra20_spdif *spdif = dev_get_drvdata(dev);
@@ -77,21 +64,24 @@ static int tegra20_spdif_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
{
- struct device *dev = substream->pcm->card->dev;
+ struct device *dev = dai->dev;
struct tegra20_spdif *spdif = snd_soc_dai_get_drvdata(dai);
+ unsigned int mask, val;
int ret, spdifclock;
- spdif->reg_ctrl &= ~TEGRA20_SPDIF_CTRL_PACK;
- spdif->reg_ctrl &= ~TEGRA20_SPDIF_CTRL_BIT_MODE_MASK;
+ mask = TEGRA20_SPDIF_CTRL_PACK |
+ TEGRA20_SPDIF_CTRL_BIT_MODE_MASK;
switch (params_format(params)) {
case SNDRV_PCM_FORMAT_S16_LE:
- spdif->reg_ctrl |= TEGRA20_SPDIF_CTRL_PACK;
- spdif->reg_ctrl |= TEGRA20_SPDIF_CTRL_BIT_MODE_16BIT;
+ val = TEGRA20_SPDIF_CTRL_PACK |
+ TEGRA20_SPDIF_CTRL_BIT_MODE_16BIT;
break;
default:
return -EINVAL;
}
+ regmap_update_bits(spdif->regmap, TEGRA20_SPDIF_CTRL, mask, val);
+
switch (params_rate(params)) {
case 32000:
spdifclock = 4096000;
@@ -129,14 +119,15 @@ static int tegra20_spdif_hw_params(struct snd_pcm_substream *substream,
static void tegra20_spdif_start_playback(struct tegra20_spdif *spdif)
{
- spdif->reg_ctrl |= TEGRA20_SPDIF_CTRL_TX_EN;
- tegra20_spdif_write(spdif, TEGRA20_SPDIF_CTRL, spdif->reg_ctrl);
+ regmap_update_bits(spdif->regmap, TEGRA20_SPDIF_CTRL,
+ TEGRA20_SPDIF_CTRL_TX_EN,
+ TEGRA20_SPDIF_CTRL_TX_EN);
}
static void tegra20_spdif_stop_playback(struct tegra20_spdif *spdif)
{
- spdif->reg_ctrl &= ~TEGRA20_SPDIF_CTRL_TX_EN;
- tegra20_spdif_write(spdif, TEGRA20_SPDIF_CTRL, spdif->reg_ctrl);
+ regmap_update_bits(spdif->regmap, TEGRA20_SPDIF_CTRL,
+ TEGRA20_SPDIF_CTRL_TX_EN, 0);
}
static int tegra20_spdif_trigger(struct snd_pcm_substream *substream, int cmd,
@@ -181,6 +172,7 @@ static struct snd_soc_dai_driver tegra20_spdif_dai = {
.name = DRV_NAME,
.probe = tegra20_spdif_probe,
.playback = {
+ .stream_name = "Playback",
.channels_min = 2,
.channels_max = 2,
.rates = SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |
diff --git a/sound/soc/tegra/tegra20_spdif.h b/sound/soc/tegra/tegra20_spdif.h
index ed75652..b48d699 100644
--- a/sound/soc/tegra/tegra20_spdif.h
+++ b/sound/soc/tegra/tegra20_spdif.h
@@ -465,7 +465,6 @@ struct tegra20_spdif {
struct tegra_pcm_dma_params capture_dma_data;
struct tegra_pcm_dma_params playback_dma_data;
struct regmap *regmap;
- u32 reg_ctrl;
};
#endif
diff --git a/sound/soc/tegra/tegra30_i2s.c b/sound/soc/tegra/tegra30_i2s.c
index d308faa..4418422 100644
--- a/sound/soc/tegra/tegra30_i2s.c
+++ b/sound/soc/tegra/tegra30_i2s.c
@@ -44,18 +44,6 @@
#define DRV_NAME "tegra30-i2s"
-static inline void tegra30_i2s_write(struct tegra30_i2s *i2s, u32 reg, u32 val)
-{
- regmap_write(i2s->regmap, reg, val);
-}
-
-static inline u32 tegra30_i2s_read(struct tegra30_i2s *i2s, u32 reg)
-{
- u32 val;
- regmap_read(i2s->regmap, reg, &val);
- return val;
-}
-
static int tegra30_i2s_runtime_suspend(struct device *dev)
{
struct tegra30_i2s *i2s = dev_get_drvdata(dev);
@@ -128,6 +116,7 @@ static int tegra30_i2s_set_fmt(struct snd_soc_dai *dai,
unsigned int fmt)
{
struct tegra30_i2s *i2s = snd_soc_dai_get_drvdata(dai);
+ unsigned int mask, val;
switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
case SND_SOC_DAIFMT_NB_NF:
@@ -136,10 +125,10 @@ static int tegra30_i2s_set_fmt(struct snd_soc_dai *dai,
return -EINVAL;
}
- i2s->reg_ctrl &= ~TEGRA30_I2S_CTRL_MASTER_ENABLE;
+ mask = TEGRA30_I2S_CTRL_MASTER_ENABLE;
switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
case SND_SOC_DAIFMT_CBS_CFS:
- i2s->reg_ctrl |= TEGRA30_I2S_CTRL_MASTER_ENABLE;
+ val = TEGRA30_I2S_CTRL_MASTER_ENABLE;
break;
case SND_SOC_DAIFMT_CBM_CFM:
break;
@@ -147,33 +136,37 @@ static int tegra30_i2s_set_fmt(struct snd_soc_dai *dai,
return -EINVAL;
}
- i2s->reg_ctrl &= ~(TEGRA30_I2S_CTRL_FRAME_FORMAT_MASK |
- TEGRA30_I2S_CTRL_LRCK_MASK);
+ mask |= TEGRA30_I2S_CTRL_FRAME_FORMAT_MASK |
+ TEGRA30_I2S_CTRL_LRCK_MASK;
switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
case SND_SOC_DAIFMT_DSP_A:
- i2s->reg_ctrl |= TEGRA30_I2S_CTRL_FRAME_FORMAT_FSYNC;
- i2s->reg_ctrl |= TEGRA30_I2S_CTRL_LRCK_L_LOW;
+ val |= TEGRA30_I2S_CTRL_FRAME_FORMAT_FSYNC;
+ val |= TEGRA30_I2S_CTRL_LRCK_L_LOW;
break;
case SND_SOC_DAIFMT_DSP_B:
- i2s->reg_ctrl |= TEGRA30_I2S_CTRL_FRAME_FORMAT_FSYNC;
- i2s->reg_ctrl |= TEGRA30_I2S_CTRL_LRCK_R_LOW;
+ val |= TEGRA30_I2S_CTRL_FRAME_FORMAT_FSYNC;
+ val |= TEGRA30_I2S_CTRL_LRCK_R_LOW;
break;
case SND_SOC_DAIFMT_I2S:
- i2s->reg_ctrl |= TEGRA30_I2S_CTRL_FRAME_FORMAT_LRCK;
- i2s->reg_ctrl |= TEGRA30_I2S_CTRL_LRCK_L_LOW;
+ val |= TEGRA30_I2S_CTRL_FRAME_FORMAT_LRCK;
+ val |= TEGRA30_I2S_CTRL_LRCK_L_LOW;
break;
case SND_SOC_DAIFMT_RIGHT_J:
- i2s->reg_ctrl |= TEGRA30_I2S_CTRL_FRAME_FORMAT_LRCK;
- i2s->reg_ctrl |= TEGRA30_I2S_CTRL_LRCK_L_LOW;
+ val |= TEGRA30_I2S_CTRL_FRAME_FORMAT_LRCK;
+ val |= TEGRA30_I2S_CTRL_LRCK_L_LOW;
break;
case SND_SOC_DAIFMT_LEFT_J:
- i2s->reg_ctrl |= TEGRA30_I2S_CTRL_FRAME_FORMAT_LRCK;
- i2s->reg_ctrl |= TEGRA30_I2S_CTRL_LRCK_L_LOW;
+ val |= TEGRA30_I2S_CTRL_FRAME_FORMAT_LRCK;
+ val |= TEGRA30_I2S_CTRL_LRCK_L_LOW;
break;
default:
return -EINVAL;
}
+ pm_runtime_get_sync(dai->dev);
+ regmap_update_bits(i2s->regmap, TEGRA30_I2S_CTRL, mask, val);
+ pm_runtime_put(dai->dev);
+
return 0;
}
@@ -181,24 +174,26 @@ static int tegra30_i2s_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
{
- struct device *dev = substream->pcm->card->dev;
+ struct device *dev = dai->dev;
struct tegra30_i2s *i2s = snd_soc_dai_get_drvdata(dai);
- u32 val;
+ unsigned int mask, val, reg;
int ret, sample_size, srate, i2sclock, bitcnt;
if (params_channels(params) != 2)
return -EINVAL;
- i2s->reg_ctrl &= ~TEGRA30_I2S_CTRL_BIT_SIZE_MASK;
+ mask = TEGRA30_I2S_CTRL_BIT_SIZE_MASK;
switch (params_format(params)) {
case SNDRV_PCM_FORMAT_S16_LE:
- i2s->reg_ctrl |= TEGRA30_I2S_CTRL_BIT_SIZE_16;
+ val = TEGRA30_I2S_CTRL_BIT_SIZE_16;
sample_size = 16;
break;
default:
return -EINVAL;
}
+ regmap_update_bits(i2s->regmap, TEGRA30_I2S_CTRL, mask, val);
+
srate = params_rate(params);
/* Final "* 2" required by Tegra hardware */
@@ -219,7 +214,7 @@ static int tegra30_i2s_hw_params(struct snd_pcm_substream *substream,
if (i2sclock % (2 * srate))
val |= TEGRA30_I2S_TIMING_NON_SYM_ENABLE;
- tegra30_i2s_write(i2s, TEGRA30_I2S_TIMING, val);
+ regmap_write(i2s->regmap, TEGRA30_I2S_TIMING, val);
val = (0 << TEGRA30_AUDIOCIF_CTRL_FIFO_THRESHOLD_SHIFT) |
(1 << TEGRA30_AUDIOCIF_CTRL_AUDIO_CHANNELS_SHIFT) |
@@ -229,15 +224,17 @@ static int tegra30_i2s_hw_params(struct snd_pcm_substream *substream,
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
val |= TEGRA30_AUDIOCIF_CTRL_DIRECTION_RX;
- tegra30_i2s_write(i2s, TEGRA30_I2S_CIF_RX_CTRL, val);
+ reg = TEGRA30_I2S_CIF_RX_CTRL;
} else {
val |= TEGRA30_AUDIOCIF_CTRL_DIRECTION_TX;
- tegra30_i2s_write(i2s, TEGRA30_I2S_CIF_TX_CTRL, val);
+ reg = TEGRA30_I2S_CIF_RX_CTRL;
}
+ regmap_write(i2s->regmap, reg, val);
+
val = (1 << TEGRA30_I2S_OFFSET_RX_DATA_OFFSET_SHIFT) |
(1 << TEGRA30_I2S_OFFSET_TX_DATA_OFFSET_SHIFT);
- tegra30_i2s_write(i2s, TEGRA30_I2S_OFFSET, val);
+ regmap_write(i2s->regmap, TEGRA30_I2S_OFFSET, val);
return 0;
}
@@ -245,29 +242,31 @@ static int tegra30_i2s_hw_params(struct snd_pcm_substream *substream,
static void tegra30_i2s_start_playback(struct tegra30_i2s *i2s)
{
tegra30_ahub_enable_tx_fifo(i2s->playback_fifo_cif);
- i2s->reg_ctrl |= TEGRA30_I2S_CTRL_XFER_EN_TX;
- tegra30_i2s_write(i2s, TEGRA30_I2S_CTRL, i2s->reg_ctrl);
+ regmap_update_bits(i2s->regmap, TEGRA30_I2S_CTRL,
+ TEGRA30_I2S_CTRL_XFER_EN_TX,
+ TEGRA30_I2S_CTRL_XFER_EN_TX);
}
static void tegra30_i2s_stop_playback(struct tegra30_i2s *i2s)
{
tegra30_ahub_disable_tx_fifo(i2s->playback_fifo_cif);
- i2s->reg_ctrl &= ~TEGRA30_I2S_CTRL_XFER_EN_TX;
- tegra30_i2s_write(i2s, TEGRA30_I2S_CTRL, i2s->reg_ctrl);
+ regmap_update_bits(i2s->regmap, TEGRA30_I2S_CTRL,
+ TEGRA30_I2S_CTRL_XFER_EN_TX, 0);
}
static void tegra30_i2s_start_capture(struct tegra30_i2s *i2s)
{
tegra30_ahub_enable_rx_fifo(i2s->capture_fifo_cif);
- i2s->reg_ctrl |= TEGRA30_I2S_CTRL_XFER_EN_RX;
- tegra30_i2s_write(i2s, TEGRA30_I2S_CTRL, i2s->reg_ctrl);
+ regmap_update_bits(i2s->regmap, TEGRA30_I2S_CTRL,
+ TEGRA30_I2S_CTRL_XFER_EN_RX,
+ TEGRA30_I2S_CTRL_XFER_EN_RX);
}
static void tegra30_i2s_stop_capture(struct tegra30_i2s *i2s)
{
tegra30_ahub_disable_rx_fifo(i2s->capture_fifo_cif);
- i2s->reg_ctrl &= ~TEGRA30_I2S_CTRL_XFER_EN_RX;
- tegra30_i2s_write(i2s, TEGRA30_I2S_CTRL, i2s->reg_ctrl);
+ regmap_update_bits(i2s->regmap, TEGRA30_I2S_CTRL,
+ TEGRA30_I2S_CTRL_XFER_EN_RX, 0);
}
static int tegra30_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
@@ -320,12 +319,14 @@ static struct snd_soc_dai_ops tegra30_i2s_dai_ops = {
static const struct snd_soc_dai_driver tegra30_i2s_dai_template = {
.probe = tegra30_i2s_probe,
.playback = {
+ .stream_name = "Playback",
.channels_min = 2,
.channels_max = 2,
.rates = SNDRV_PCM_RATE_8000_96000,
.formats = SNDRV_PCM_FMTBIT_S16_LE,
},
.capture = {
+ .stream_name = "Capture",
.channels_min = 2,
.channels_max = 2,
.rates = SNDRV_PCM_RATE_8000_96000,
diff --git a/sound/soc/tegra/tegra30_i2s.h b/sound/soc/tegra/tegra30_i2s.h
index 91adf29..34dc47b 100644
--- a/sound/soc/tegra/tegra30_i2s.h
+++ b/sound/soc/tegra/tegra30_i2s.h
@@ -236,7 +236,6 @@ struct tegra30_i2s {
enum tegra30_ahub_txcif playback_fifo_cif;
struct tegra_pcm_dma_params playback_dma_data;
struct regmap *regmap;
- u32 reg_ctrl;
};
#endif
diff --git a/sound/soc/tegra/tegra_alc5632.c b/sound/soc/tegra/tegra_alc5632.c
index 32de700..d684df2 100644
--- a/sound/soc/tegra/tegra_alc5632.c
+++ b/sound/soc/tegra/tegra_alc5632.c
@@ -1,5 +1,5 @@
/*
- * tegra_alc5632.c -- Toshiba AC100(PAZ00) machine ASoC driver
+* tegra_alc5632.c -- Toshiba AC100(PAZ00) machine ASoC driver
*
* Copyright (C) 2011 The AC100 Kernel Team <ac100@lists.lauchpad.net>
* Copyright (C) 2012 - NVIDIA, Inc.
@@ -33,11 +33,8 @@
#define DRV_NAME "tegra-alc5632"
-#define GPIO_HP_DET BIT(0)
-
struct tegra_alc5632 {
struct tegra_asoc_utils_data util_data;
- int gpio_requested;
int gpio_hp_det;
};
@@ -46,7 +43,7 @@ static int tegra_alc5632_asoc_hw_params(struct snd_pcm_substream *substream,
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_dai *codec_dai = rtd->codec_dai;
- struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_codec *codec = codec_dai->codec;
struct snd_soc_card *card = codec->card;
struct tegra_alc5632 *alc5632 = snd_soc_card_get_drvdata(card);
int srate, mclk;
@@ -108,9 +105,9 @@ static const struct snd_kcontrol_new tegra_alc5632_controls[] = {
static int tegra_alc5632_asoc_init(struct snd_soc_pcm_runtime *rtd)
{
- struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_codec *codec = codec_dai->codec;
struct snd_soc_dapm_context *dapm = &codec->dapm;
- struct device_node *np = codec->card->dev->of_node;
struct tegra_alc5632 *machine = snd_soc_card_get_drvdata(codec->card);
snd_soc_jack_new(codec, "Headset Jack", SND_JACK_HEADSET,
@@ -119,14 +116,11 @@ static int tegra_alc5632_asoc_init(struct snd_soc_pcm_runtime *rtd)
ARRAY_SIZE(tegra_alc5632_hs_jack_pins),
tegra_alc5632_hs_jack_pins);
- machine->gpio_hp_det = of_get_named_gpio(np, "nvidia,hp-det-gpios", 0);
-
if (gpio_is_valid(machine->gpio_hp_det)) {
tegra_alc5632_hp_jack_gpio.gpio = machine->gpio_hp_det;
snd_soc_jack_add_gpios(&tegra_alc5632_hs_jack,
1,
&tegra_alc5632_hp_jack_gpio);
- machine->gpio_requested |= GPIO_HP_DET;
}
snd_soc_dapm_force_enable_pin(dapm, "MICBIAS1");
@@ -159,6 +153,7 @@ static struct snd_soc_card snd_soc_tegra_alc5632 = {
static __devinit int tegra_alc5632_probe(struct platform_device *pdev)
{
+ struct device_node *np = pdev->dev.of_node;
struct snd_soc_card *card = &snd_soc_tegra_alc5632;
struct tegra_alc5632 *alc5632;
int ret;
@@ -181,6 +176,10 @@ static __devinit int tegra_alc5632_probe(struct platform_device *pdev)
goto err;
}
+ alc5632->gpio_hp_det = of_get_named_gpio(np, "nvidia,hp-det-gpios", 0);
+ if (alc5632->gpio_hp_det == -ENODEV)
+ return -EPROBE_DEFER;
+
ret = snd_soc_of_parse_card_name(card, "nvidia,model");
if (ret)
goto err;
@@ -199,16 +198,16 @@ static __devinit int tegra_alc5632_probe(struct platform_device *pdev)
goto err;
}
- tegra_alc5632_dai.cpu_dai_of_node = of_parse_phandle(
+ tegra_alc5632_dai.cpu_of_node = of_parse_phandle(
pdev->dev.of_node, "nvidia,i2s-controller", 0);
- if (!tegra_alc5632_dai.cpu_dai_of_node) {
+ if (!tegra_alc5632_dai.cpu_of_node) {
dev_err(&pdev->dev,
"Property 'nvidia,i2s-controller' missing or invalid\n");
ret = -EINVAL;
goto err;
}
- tegra_alc5632_dai.platform_of_node = tegra_alc5632_dai.cpu_dai_of_node;
+ tegra_alc5632_dai.platform_of_node = tegra_alc5632_dai.cpu_of_node;
ret = tegra_asoc_utils_init(&alc5632->util_data, &pdev->dev);
if (ret)
@@ -234,11 +233,8 @@ static int __devexit tegra_alc5632_remove(struct platform_device *pdev)
struct snd_soc_card *card = platform_get_drvdata(pdev);
struct tegra_alc5632 *machine = snd_soc_card_get_drvdata(card);
- if (machine->gpio_requested & GPIO_HP_DET)
- snd_soc_jack_free_gpios(&tegra_alc5632_hs_jack,
- 1,
- &tegra_alc5632_hp_jack_gpio);
- machine->gpio_requested = 0;
+ snd_soc_jack_free_gpios(&tegra_alc5632_hs_jack, 1,
+ &tegra_alc5632_hp_jack_gpio);
snd_soc_unregister_card(card);
diff --git a/sound/soc/tegra/tegra_pcm.c b/sound/soc/tegra/tegra_pcm.c
index 127348d..5658bce 100644
--- a/sound/soc/tegra/tegra_pcm.c
+++ b/sound/soc/tegra/tegra_pcm.c
@@ -36,6 +36,7 @@
#include <sound/pcm.h>
#include <sound/pcm_params.h>
#include <sound/soc.h>
+#include <sound/dmaengine_pcm.h>
#include "tegra_pcm.h"
@@ -56,6 +57,7 @@ static const struct snd_pcm_hardware tegra_pcm_hardware = {
.fifo_size = 4,
};
+#if defined(CONFIG_TEGRA_SYSTEM_DMA)
static void tegra_pcm_queue_dma(struct tegra_runtime_data *prtd)
{
struct snd_pcm_substream *substream = prtd->substream;
@@ -285,6 +287,119 @@ static struct snd_pcm_ops tegra_pcm_ops = {
.pointer = tegra_pcm_pointer,
.mmap = tegra_pcm_mmap,
};
+#else
+static int tegra_pcm_open(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct device *dev = rtd->platform->dev;
+ int ret;
+
+ /* Set HW params now that initialization is complete */
+ snd_soc_set_runtime_hwparams(substream, &tegra_pcm_hardware);
+
+ ret = snd_dmaengine_pcm_open(substream, NULL, NULL);
+ if (ret) {
+ dev_err(dev, "dmaengine pcm open failed with err %d\n", ret);
+ return ret;
+ }
+
+ return 0;
+}
+
+static int tegra_pcm_close(struct snd_pcm_substream *substream)
+{
+ snd_dmaengine_pcm_close(substream);
+ return 0;
+}
+
+static int tegra_pcm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct device *dev = rtd->platform->dev;
+ struct dma_chan *chan = snd_dmaengine_pcm_get_chan(substream);
+ struct tegra_pcm_dma_params *dmap;
+ struct dma_slave_config slave_config;
+ int ret;
+
+ dmap = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
+
+ ret = snd_hwparams_to_dma_slave_config(substream, params,
+ &slave_config);
+ if (ret) {
+ dev_err(dev, "hw params config failed with err %d\n", ret);
+ return ret;
+ }
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ slave_config.dst_addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES;
+ slave_config.dst_addr = dmap->addr;
+ slave_config.src_maxburst = 0;
+ } else {
+ slave_config.src_addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES;
+ slave_config.src_addr = dmap->addr;
+ slave_config.dst_maxburst = 0;
+ }
+ slave_config.slave_id = dmap->req_sel;
+
+ ret = dmaengine_slave_config(chan, &slave_config);
+ if (ret < 0) {
+ dev_err(dev, "dma slave config failed with err %d\n", ret);
+ return ret;
+ }
+
+ snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
+ return 0;
+}
+
+static int tegra_pcm_hw_free(struct snd_pcm_substream *substream)
+{
+ snd_pcm_set_runtime_buffer(substream, NULL);
+ return 0;
+}
+
+static int tegra_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ return snd_dmaengine_pcm_trigger(substream,
+ SNDRV_PCM_TRIGGER_START);
+
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ return snd_dmaengine_pcm_trigger(substream,
+ SNDRV_PCM_TRIGGER_STOP);
+ default:
+ return -EINVAL;
+ }
+ return 0;
+}
+
+static int tegra_pcm_mmap(struct snd_pcm_substream *substream,
+ struct vm_area_struct *vma)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ return dma_mmap_writecombine(substream->pcm->card->dev, vma,
+ runtime->dma_area,
+ runtime->dma_addr,
+ runtime->dma_bytes);
+}
+
+static struct snd_pcm_ops tegra_pcm_ops = {
+ .open = tegra_pcm_open,
+ .close = tegra_pcm_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = tegra_pcm_hw_params,
+ .hw_free = tegra_pcm_hw_free,
+ .trigger = tegra_pcm_trigger,
+ .pointer = snd_dmaengine_pcm_pointer,
+ .mmap = tegra_pcm_mmap,
+};
+#endif
static int tegra_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream)
{
diff --git a/sound/soc/tegra/tegra_pcm.h b/sound/soc/tegra/tegra_pcm.h
index 985d418..a3a4503 100644
--- a/sound/soc/tegra/tegra_pcm.h
+++ b/sound/soc/tegra/tegra_pcm.h
@@ -40,6 +40,7 @@ struct tegra_pcm_dma_params {
unsigned long req_sel;
};
+#if defined(CONFIG_TEGRA_SYSTEM_DMA)
struct tegra_runtime_data {
struct snd_pcm_substream *substream;
spinlock_t lock;
@@ -51,6 +52,7 @@ struct tegra_runtime_data {
struct tegra_dma_req dma_req[2];
struct tegra_dma_channel *dma_chan;
};
+#endif
int tegra_pcm_platform_register(struct device *dev);
void tegra_pcm_platform_unregister(struct device *dev);
diff --git a/sound/soc/tegra/tegra_wm8753.c b/sound/soc/tegra/tegra_wm8753.c
index 4e77026..ea9166d 100644
--- a/sound/soc/tegra/tegra_wm8753.c
+++ b/sound/soc/tegra/tegra_wm8753.c
@@ -57,7 +57,7 @@ static int tegra_wm8753_hw_params(struct snd_pcm_substream *substream,
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_dai *codec_dai = rtd->codec_dai;
- struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_codec *codec = codec_dai->codec;
struct snd_soc_card *card = codec->card;
struct tegra_wm8753 *machine = snd_soc_card_get_drvdata(card);
int srate, mclk;
@@ -157,9 +157,9 @@ static __devinit int tegra_wm8753_driver_probe(struct platform_device *pdev)
goto err;
}
- tegra_wm8753_dai.cpu_dai_of_node = of_parse_phandle(
+ tegra_wm8753_dai.cpu_of_node = of_parse_phandle(
pdev->dev.of_node, "nvidia,i2s-controller", 0);
- if (!tegra_wm8753_dai.cpu_dai_of_node) {
+ if (!tegra_wm8753_dai.cpu_of_node) {
dev_err(&pdev->dev,
"Property 'nvidia,i2s-controller' missing or invalid\n");
ret = -EINVAL;
@@ -167,7 +167,7 @@ static __devinit int tegra_wm8753_driver_probe(struct platform_device *pdev)
}
tegra_wm8753_dai.platform_of_node =
- tegra_wm8753_dai.cpu_dai_of_node;
+ tegra_wm8753_dai.cpu_of_node;
ret = tegra_asoc_utils_init(&machine->util_data, &pdev->dev);
if (ret)
diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c
index 3b6da91..0c5bb33 100644
--- a/sound/soc/tegra/tegra_wm8903.c
+++ b/sound/soc/tegra/tegra_wm8903.c
@@ -28,8 +28,6 @@
*
*/
-#include <asm/mach-types.h>
-
#include <linux/module.h>
#include <linux/platform_device.h>
#include <linux/slab.h>
@@ -50,16 +48,9 @@
#define DRV_NAME "tegra-snd-wm8903"
-#define GPIO_SPKR_EN BIT(0)
-#define GPIO_HP_MUTE BIT(1)
-#define GPIO_INT_MIC_EN BIT(2)
-#define GPIO_EXT_MIC_EN BIT(3)
-#define GPIO_HP_DET BIT(4)
-
struct tegra_wm8903 {
struct tegra_wm8903_platform_data pdata;
struct tegra_asoc_utils_data util_data;
- int gpio_requested;
};
static int tegra_wm8903_hw_params(struct snd_pcm_substream *substream,
@@ -67,8 +58,7 @@ static int tegra_wm8903_hw_params(struct snd_pcm_substream *substream,
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_dai *codec_dai = rtd->codec_dai;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
- struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_codec *codec = codec_dai->codec;
struct snd_soc_card *card = codec->card;
struct tegra_wm8903 *machine = snd_soc_card_get_drvdata(card);
int srate, mclk;
@@ -95,24 +85,6 @@ static int tegra_wm8903_hw_params(struct snd_pcm_substream *substream,
return err;
}
- err = snd_soc_dai_set_fmt(codec_dai,
- SND_SOC_DAIFMT_I2S |
- SND_SOC_DAIFMT_NB_NF |
- SND_SOC_DAIFMT_CBS_CFS);
- if (err < 0) {
- dev_err(card->dev, "codec_dai fmt not set\n");
- return err;
- }
-
- err = snd_soc_dai_set_fmt(cpu_dai,
- SND_SOC_DAIFMT_I2S |
- SND_SOC_DAIFMT_NB_NF |
- SND_SOC_DAIFMT_CBS_CFS);
- if (err < 0) {
- dev_err(card->dev, "cpu_dai fmt not set\n");
- return err;
- }
-
err = snd_soc_dai_set_sysclk(codec_dai, 0, mclk,
SND_SOC_CLOCK_IN);
if (err < 0) {
@@ -160,7 +132,7 @@ static int tegra_wm8903_event_int_spk(struct snd_soc_dapm_widget *w,
struct tegra_wm8903 *machine = snd_soc_card_get_drvdata(card);
struct tegra_wm8903_platform_data *pdata = &machine->pdata;
- if (!(machine->gpio_requested & GPIO_SPKR_EN))
+ if (!gpio_is_valid(pdata->gpio_spkr_en))
return 0;
gpio_set_value_cansleep(pdata->gpio_spkr_en,
@@ -177,7 +149,7 @@ static int tegra_wm8903_event_hp(struct snd_soc_dapm_widget *w,
struct tegra_wm8903 *machine = snd_soc_card_get_drvdata(card);
struct tegra_wm8903_platform_data *pdata = &machine->pdata;
- if (!(machine->gpio_requested & GPIO_HP_MUTE))
+ if (!gpio_is_valid(pdata->gpio_hp_mute))
return 0;
gpio_set_value_cansleep(pdata->gpio_hp_mute,
@@ -203,122 +175,18 @@ static const struct snd_soc_dapm_route harmony_audio_map[] = {
{"IN1L", NULL, "Mic Jack"},
};
-static const struct snd_soc_dapm_route seaboard_audio_map[] = {
- {"Headphone Jack", NULL, "HPOUTR"},
- {"Headphone Jack", NULL, "HPOUTL"},
- {"Int Spk", NULL, "ROP"},
- {"Int Spk", NULL, "RON"},
- {"Int Spk", NULL, "LOP"},
- {"Int Spk", NULL, "LON"},
- {"Mic Jack", NULL, "MICBIAS"},
- {"IN1R", NULL, "Mic Jack"},
-};
-
-static const struct snd_soc_dapm_route kaen_audio_map[] = {
- {"Headphone Jack", NULL, "HPOUTR"},
- {"Headphone Jack", NULL, "HPOUTL"},
- {"Int Spk", NULL, "ROP"},
- {"Int Spk", NULL, "RON"},
- {"Int Spk", NULL, "LOP"},
- {"Int Spk", NULL, "LON"},
- {"Mic Jack", NULL, "MICBIAS"},
- {"IN2R", NULL, "Mic Jack"},
-};
-
-static const struct snd_soc_dapm_route aebl_audio_map[] = {
- {"Headphone Jack", NULL, "HPOUTR"},
- {"Headphone Jack", NULL, "HPOUTL"},
- {"Int Spk", NULL, "LINEOUTR"},
- {"Int Spk", NULL, "LINEOUTL"},
- {"Mic Jack", NULL, "MICBIAS"},
- {"IN1R", NULL, "Mic Jack"},
-};
-
static const struct snd_kcontrol_new tegra_wm8903_controls[] = {
SOC_DAPM_PIN_SWITCH("Int Spk"),
};
static int tegra_wm8903_init(struct snd_soc_pcm_runtime *rtd)
{
- struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_codec *codec = codec_dai->codec;
struct snd_soc_dapm_context *dapm = &codec->dapm;
struct snd_soc_card *card = codec->card;
struct tegra_wm8903 *machine = snd_soc_card_get_drvdata(card);
struct tegra_wm8903_platform_data *pdata = &machine->pdata;
- struct device_node *np = card->dev->of_node;
- int ret;
-
- if (card->dev->platform_data) {
- memcpy(pdata, card->dev->platform_data, sizeof(*pdata));
- } else if (np) {
- /*
- * This part must be in init() rather than probe() in order to
- * guarantee that the WM8903 has been probed, and hence its
- * GPIO controller registered, which is a pre-condition for
- * of_get_named_gpio() to be able to map the phandles in the
- * properties to the controller node. Given this, all
- * pdata handling is in init() for consistency.
- */
- pdata->gpio_spkr_en = of_get_named_gpio(np,
- "nvidia,spkr-en-gpios", 0);
- pdata->gpio_hp_mute = of_get_named_gpio(np,
- "nvidia,hp-mute-gpios", 0);
- pdata->gpio_hp_det = of_get_named_gpio(np,
- "nvidia,hp-det-gpios", 0);
- pdata->gpio_int_mic_en = of_get_named_gpio(np,
- "nvidia,int-mic-en-gpios", 0);
- pdata->gpio_ext_mic_en = of_get_named_gpio(np,
- "nvidia,ext-mic-en-gpios", 0);
- } else {
- dev_err(card->dev, "No platform data supplied\n");
- return -EINVAL;
- }
-
- if (gpio_is_valid(pdata->gpio_spkr_en)) {
- ret = gpio_request(pdata->gpio_spkr_en, "spkr_en");
- if (ret) {
- dev_err(card->dev, "cannot get spkr_en gpio\n");
- return ret;
- }
- machine->gpio_requested |= GPIO_SPKR_EN;
-
- gpio_direction_output(pdata->gpio_spkr_en, 0);
- }
-
- if (gpio_is_valid(pdata->gpio_hp_mute)) {
- ret = gpio_request(pdata->gpio_hp_mute, "hp_mute");
- if (ret) {
- dev_err(card->dev, "cannot get hp_mute gpio\n");
- return ret;
- }
- machine->gpio_requested |= GPIO_HP_MUTE;
-
- gpio_direction_output(pdata->gpio_hp_mute, 1);
- }
-
- if (gpio_is_valid(pdata->gpio_int_mic_en)) {
- ret = gpio_request(pdata->gpio_int_mic_en, "int_mic_en");
- if (ret) {
- dev_err(card->dev, "cannot get int_mic_en gpio\n");
- return ret;
- }
- machine->gpio_requested |= GPIO_INT_MIC_EN;
-
- /* Disable int mic; enable signal is active-high */
- gpio_direction_output(pdata->gpio_int_mic_en, 0);
- }
-
- if (gpio_is_valid(pdata->gpio_ext_mic_en)) {
- ret = gpio_request(pdata->gpio_ext_mic_en, "ext_mic_en");
- if (ret) {
- dev_err(card->dev, "cannot get ext_mic_en gpio\n");
- return ret;
- }
- machine->gpio_requested |= GPIO_EXT_MIC_EN;
-
- /* Enable ext mic; enable signal is active-low */
- gpio_direction_output(pdata->gpio_ext_mic_en, 0);
- }
if (gpio_is_valid(pdata->gpio_hp_det)) {
tegra_wm8903_hp_jack_gpio.gpio = pdata->gpio_hp_det;
@@ -330,7 +198,6 @@ static int tegra_wm8903_init(struct snd_soc_pcm_runtime *rtd)
snd_soc_jack_add_gpios(&tegra_wm8903_hp_jack,
1,
&tegra_wm8903_hp_jack_gpio);
- machine->gpio_requested |= GPIO_HP_DET;
}
snd_soc_jack_new(codec, "Mic Jack", SND_JACK_MICROPHONE,
@@ -366,6 +233,9 @@ static struct snd_soc_dai_link tegra_wm8903_dai = {
.codec_dai_name = "wm8903-hifi",
.init = tegra_wm8903_init,
.ops = &tegra_wm8903_ops,
+ .dai_fmt = SND_SOC_DAIFMT_I2S |
+ SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBS_CFS,
};
static struct snd_soc_card snd_soc_tegra_wm8903 = {
@@ -385,8 +255,10 @@ static struct snd_soc_card snd_soc_tegra_wm8903 = {
static __devinit int tegra_wm8903_driver_probe(struct platform_device *pdev)
{
+ struct device_node *np = pdev->dev.of_node;
struct snd_soc_card *card = &snd_soc_tegra_wm8903;
struct tegra_wm8903 *machine;
+ struct tegra_wm8903_platform_data *pdata;
int ret;
if (!pdev->dev.platform_data && !pdev->dev.of_node) {
@@ -401,12 +273,42 @@ static __devinit int tegra_wm8903_driver_probe(struct platform_device *pdev)
ret = -ENOMEM;
goto err;
}
+ pdata = &machine->pdata;
card->dev = &pdev->dev;
platform_set_drvdata(pdev, card);
snd_soc_card_set_drvdata(card, machine);
- if (pdev->dev.of_node) {
+ if (pdev->dev.platform_data) {
+ memcpy(pdata, card->dev->platform_data, sizeof(*pdata));
+ } else if (np) {
+ pdata->gpio_spkr_en = of_get_named_gpio(np,
+ "nvidia,spkr-en-gpios", 0);
+ if (pdata->gpio_spkr_en == -ENODEV)
+ return -EPROBE_DEFER;
+
+ pdata->gpio_hp_mute = of_get_named_gpio(np,
+ "nvidia,hp-mute-gpios", 0);
+ if (pdata->gpio_hp_mute == -ENODEV)
+ return -EPROBE_DEFER;
+
+ pdata->gpio_hp_det = of_get_named_gpio(np,
+ "nvidia,hp-det-gpios", 0);
+ if (pdata->gpio_hp_det == -ENODEV)
+ return -EPROBE_DEFER;
+
+ pdata->gpio_int_mic_en = of_get_named_gpio(np,
+ "nvidia,int-mic-en-gpios", 0);
+ if (pdata->gpio_int_mic_en == -ENODEV)
+ return -EPROBE_DEFER;
+
+ pdata->gpio_ext_mic_en = of_get_named_gpio(np,
+ "nvidia,ext-mic-en-gpios", 0);
+ if (pdata->gpio_ext_mic_en == -ENODEV)
+ return -EPROBE_DEFER;
+ }
+
+ if (np) {
ret = snd_soc_of_parse_card_name(card, "nvidia,model");
if (ret)
goto err;
@@ -417,8 +319,8 @@ static __devinit int tegra_wm8903_driver_probe(struct platform_device *pdev)
goto err;
tegra_wm8903_dai.codec_name = NULL;
- tegra_wm8903_dai.codec_of_node = of_parse_phandle(
- pdev->dev.of_node, "nvidia,audio-codec", 0);
+ tegra_wm8903_dai.codec_of_node = of_parse_phandle(np,
+ "nvidia,audio-codec", 0);
if (!tegra_wm8903_dai.codec_of_node) {
dev_err(&pdev->dev,
"Property 'nvidia,audio-codec' missing or invalid\n");
@@ -427,9 +329,9 @@ static __devinit int tegra_wm8903_driver_probe(struct platform_device *pdev)
}
tegra_wm8903_dai.cpu_dai_name = NULL;
- tegra_wm8903_dai.cpu_dai_of_node = of_parse_phandle(
- pdev->dev.of_node, "nvidia,i2s-controller", 0);
- if (!tegra_wm8903_dai.cpu_dai_of_node) {
+ tegra_wm8903_dai.cpu_of_node = of_parse_phandle(np,
+ "nvidia,i2s-controller", 0);
+ if (!tegra_wm8903_dai.cpu_of_node) {
dev_err(&pdev->dev,
"Property 'nvidia,i2s-controller' missing or invalid\n");
ret = -EINVAL;
@@ -438,20 +340,47 @@ static __devinit int tegra_wm8903_driver_probe(struct platform_device *pdev)
tegra_wm8903_dai.platform_name = NULL;
tegra_wm8903_dai.platform_of_node =
- tegra_wm8903_dai.cpu_dai_of_node;
+ tegra_wm8903_dai.cpu_of_node;
} else {
- if (machine_is_harmony()) {
- card->dapm_routes = harmony_audio_map;
- card->num_dapm_routes = ARRAY_SIZE(harmony_audio_map);
- } else if (machine_is_seaboard()) {
- card->dapm_routes = seaboard_audio_map;
- card->num_dapm_routes = ARRAY_SIZE(seaboard_audio_map);
- } else if (machine_is_kaen()) {
- card->dapm_routes = kaen_audio_map;
- card->num_dapm_routes = ARRAY_SIZE(kaen_audio_map);
- } else {
- card->dapm_routes = aebl_audio_map;
- card->num_dapm_routes = ARRAY_SIZE(aebl_audio_map);
+ card->dapm_routes = harmony_audio_map;
+ card->num_dapm_routes = ARRAY_SIZE(harmony_audio_map);
+ }
+
+ if (gpio_is_valid(pdata->gpio_spkr_en)) {
+ ret = devm_gpio_request_one(&pdev->dev, pdata->gpio_spkr_en,
+ GPIOF_OUT_INIT_LOW, "spkr_en");
+ if (ret) {
+ dev_err(card->dev, "cannot get spkr_en gpio\n");
+ return ret;
+ }
+ }
+
+ if (gpio_is_valid(pdata->gpio_hp_mute)) {
+ ret = devm_gpio_request_one(&pdev->dev, pdata->gpio_hp_mute,
+ GPIOF_OUT_INIT_HIGH, "hp_mute");
+ if (ret) {
+ dev_err(card->dev, "cannot get hp_mute gpio\n");
+ return ret;
+ }
+ }
+
+ if (gpio_is_valid(pdata->gpio_int_mic_en)) {
+ /* Disable int mic; enable signal is active-high */
+ ret = devm_gpio_request_one(&pdev->dev, pdata->gpio_int_mic_en,
+ GPIOF_OUT_INIT_LOW, "int_mic_en");
+ if (ret) {
+ dev_err(card->dev, "cannot get int_mic_en gpio\n");
+ return ret;
+ }
+ }
+
+ if (gpio_is_valid(pdata->gpio_ext_mic_en)) {
+ /* Enable ext mic; enable signal is active-low */
+ ret = devm_gpio_request_one(&pdev->dev, pdata->gpio_ext_mic_en,
+ GPIOF_OUT_INIT_LOW, "ext_mic_en");
+ if (ret) {
+ dev_err(card->dev, "cannot get ext_mic_en gpio\n");
+ return ret;
}
}
@@ -478,21 +407,9 @@ static int __devexit tegra_wm8903_driver_remove(struct platform_device *pdev)
{
struct snd_soc_card *card = platform_get_drvdata(pdev);
struct tegra_wm8903 *machine = snd_soc_card_get_drvdata(card);
- struct tegra_wm8903_platform_data *pdata = &machine->pdata;
- if (machine->gpio_requested & GPIO_HP_DET)
- snd_soc_jack_free_gpios(&tegra_wm8903_hp_jack,
- 1,
- &tegra_wm8903_hp_jack_gpio);
- if (machine->gpio_requested & GPIO_EXT_MIC_EN)
- gpio_free(pdata->gpio_ext_mic_en);
- if (machine->gpio_requested & GPIO_INT_MIC_EN)
- gpio_free(pdata->gpio_int_mic_en);
- if (machine->gpio_requested & GPIO_HP_MUTE)
- gpio_free(pdata->gpio_hp_mute);
- if (machine->gpio_requested & GPIO_SPKR_EN)
- gpio_free(pdata->gpio_spkr_en);
- machine->gpio_requested = 0;
+ snd_soc_jack_free_gpios(&tegra_wm8903_hp_jack, 1,
+ &tegra_wm8903_hp_jack_gpio);
snd_soc_unregister_card(card);
diff --git a/sound/soc/tegra/trimslice.c b/sound/soc/tegra/trimslice.c
index 4a8d5b6..e69a4f7 100644
--- a/sound/soc/tegra/trimslice.c
+++ b/sound/soc/tegra/trimslice.c
@@ -52,8 +52,7 @@ static int trimslice_asoc_hw_params(struct snd_pcm_substream *substream,
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_dai *codec_dai = rtd->codec_dai;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
- struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_codec *codec = codec_dai->codec;
struct snd_soc_card *card = codec->card;
struct tegra_trimslice *trimslice = snd_soc_card_get_drvdata(card);
int srate, mclk;
@@ -68,24 +67,6 @@ static int trimslice_asoc_hw_params(struct snd_pcm_substream *substream,
return err;
}
- err = snd_soc_dai_set_fmt(codec_dai,
- SND_SOC_DAIFMT_I2S |
- SND_SOC_DAIFMT_NB_NF |
- SND_SOC_DAIFMT_CBS_CFS);
- if (err < 0) {
- dev_err(card->dev, "codec_dai fmt not set\n");
- return err;
- }
-
- err = snd_soc_dai_set_fmt(cpu_dai,
- SND_SOC_DAIFMT_I2S |
- SND_SOC_DAIFMT_NB_NF |
- SND_SOC_DAIFMT_CBS_CFS);
- if (err < 0) {
- dev_err(card->dev, "cpu_dai fmt not set\n");
- return err;
- }
-
err = snd_soc_dai_set_sysclk(codec_dai, 0, mclk,
SND_SOC_CLOCK_IN);
if (err < 0) {
@@ -121,6 +102,9 @@ static struct snd_soc_dai_link trimslice_tlv320aic23_dai = {
.cpu_dai_name = "tegra20-i2s.0",
.codec_dai_name = "tlv320aic23-hifi",
.ops = &trimslice_asoc_ops,
+ .dai_fmt = SND_SOC_DAIFMT_I2S |
+ SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBS_CFS,
};
static struct snd_soc_card snd_soc_trimslice = {
@@ -162,9 +146,9 @@ static __devinit int tegra_snd_trimslice_probe(struct platform_device *pdev)
}
trimslice_tlv320aic23_dai.cpu_dai_name = NULL;
- trimslice_tlv320aic23_dai.cpu_dai_of_node = of_parse_phandle(
+ trimslice_tlv320aic23_dai.cpu_of_node = of_parse_phandle(
pdev->dev.of_node, "nvidia,i2s-controller", 0);
- if (!trimslice_tlv320aic23_dai.cpu_dai_of_node) {
+ if (!trimslice_tlv320aic23_dai.cpu_of_node) {
dev_err(&pdev->dev,
"Property 'nvidia,i2s-controller' missing or invalid\n");
ret = -EINVAL;
@@ -173,7 +157,7 @@ static __devinit int tegra_snd_trimslice_probe(struct platform_device *pdev)
trimslice_tlv320aic23_dai.platform_name = NULL;
trimslice_tlv320aic23_dai.platform_of_node =
- trimslice_tlv320aic23_dai.cpu_dai_of_node;
+ trimslice_tlv320aic23_dai.cpu_of_node;
}
ret = tegra_asoc_utils_init(&trimslice->util_data, &pdev->dev);
diff --git a/sound/soc/ux500/Kconfig b/sound/soc/ux500/Kconfig
index 44cf434..069330d 100644
--- a/sound/soc/ux500/Kconfig
+++ b/sound/soc/ux500/Kconfig
@@ -12,3 +12,21 @@ menuconfig SND_SOC_UX500
config SND_SOC_UX500_PLAT_MSP_I2S
tristate
depends on SND_SOC_UX500
+
+config SND_SOC_UX500_PLAT_DMA
+ tristate "Platform - DB8500 (DMA)"
+ depends on SND_SOC_UX500
+ select SND_SOC_DMAENGINE_PCM
+ help
+ Say Y if you want to enable the Ux500 platform-driver.
+
++config SND_SOC_UX500_MACH_MOP500
++ tristate "Machine - MOP500 (Ux500 + AB8500)"
+ depends on AB8500_CORE && AB8500_GPADC && SND_SOC_UX500
+ select SND_SOC_AB8500_CODEC
+ select SND_SOC_UX500_PLAT_MSP_I2S
+ select SND_SOC_UX500_PLAT_DMA
+ help
+ Select this to enable the MOP500 machine-driver.
+ This will enable platform-drivers for: Ux500
+ This will enable codec-drivers for: AB8500
diff --git a/sound/soc/ux500/Makefile b/sound/soc/ux500/Makefile
index 19974c5..cce0c11 100644
--- a/sound/soc/ux500/Makefile
+++ b/sound/soc/ux500/Makefile
@@ -2,3 +2,9 @@
snd-soc-ux500-plat-msp-i2s-objs := ux500_msp_dai.o ux500_msp_i2s.o
obj-$(CONFIG_SND_SOC_UX500_PLAT_MSP_I2S) += snd-soc-ux500-plat-msp-i2s.o
+
+snd-soc-ux500-plat-dma-objs := ux500_pcm.o
+obj-$(CONFIG_SND_SOC_UX500_PLAT_DMA) += snd-soc-ux500-plat-dma.o
+
+snd-soc-ux500-mach-mop500-objs := mop500.o mop500_ab8500.o
+obj-$(CONFIG_SND_SOC_UX500_MACH_MOP500) += snd-soc-ux500-mach-mop500.o
diff --git a/sound/soc/ux500/mop500.c b/sound/soc/ux500/mop500.c
new file mode 100644
index 0000000..31c4d26
--- /dev/null
+++ b/sound/soc/ux500/mop500.c
@@ -0,0 +1,113 @@
+/*
+ * Copyright (C) ST-Ericsson SA 2012
+ *
+ * Author: Ola Lilja (ola.o.lilja@stericsson.com)
+ * for ST-Ericsson.
+ *
+ * License terms:
+ *
+ * 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 <asm/mach-types.h>
+
+#include <linux/module.h>
+#include <linux/io.h>
+#include <linux/spi/spi.h>
+
+#include <sound/soc.h>
+#include <sound/initval.h>
+
+#include "ux500_pcm.h"
+#include "ux500_msp_dai.h"
+
+#include <mop500_ab8500.h>
+
+/* Define the whole MOP500 soundcard, linking platform to the codec-drivers */
+struct snd_soc_dai_link mop500_dai_links[] = {
+ {
+ .name = "ab8500_0",
+ .stream_name = "ab8500_0",
+ .cpu_dai_name = "ux500-msp-i2s.1",
+ .codec_dai_name = "ab8500-codec-dai.0",
+ .platform_name = "ux500-pcm.0",
+ .codec_name = "ab8500-codec.0",
+ .init = mop500_ab8500_machine_init,
+ .ops = mop500_ab8500_ops,
+ },
+ {
+ .name = "ab8500_1",
+ .stream_name = "ab8500_1",
+ .cpu_dai_name = "ux500-msp-i2s.3",
+ .codec_dai_name = "ab8500-codec-dai.1",
+ .platform_name = "ux500-pcm.0",
+ .codec_name = "ab8500-codec.0",
+ .init = NULL,
+ .ops = mop500_ab8500_ops,
+ },
+};
+
+static struct snd_soc_card mop500_card = {
+ .name = "MOP500-card",
+ .probe = NULL,
+ .dai_link = mop500_dai_links,
+ .num_links = ARRAY_SIZE(mop500_dai_links),
+};
+
+static int __devinit mop500_probe(struct platform_device *pdev)
+{
+ int ret;
+
+ pr_debug("%s: Enter.\n", __func__);
+
+ dev_dbg(&pdev->dev, "%s: Enter.\n", __func__);
+
+ mop500_card.dev = &pdev->dev;
+
+ dev_dbg(&pdev->dev, "%s: Card %s: Set platform drvdata.\n",
+ __func__, mop500_card.name);
+ platform_set_drvdata(pdev, &mop500_card);
+
+ snd_soc_card_set_drvdata(&mop500_card, NULL);
+
+ dev_dbg(&pdev->dev, "%s: Card %s: num_links = %d\n",
+ __func__, mop500_card.name, mop500_card.num_links);
+ dev_dbg(&pdev->dev, "%s: Card %s: DAI-link 0: name = %s\n",
+ __func__, mop500_card.name, mop500_card.dai_link[0].name);
+ dev_dbg(&pdev->dev, "%s: Card %s: DAI-link 0: stream_name = %s\n",
+ __func__, mop500_card.name,
+ mop500_card.dai_link[0].stream_name);
+
+ ret = snd_soc_register_card(&mop500_card);
+ if (ret)
+ dev_err(&pdev->dev,
+ "Error: snd_soc_register_card failed (%d)!\n",
+ ret);
+
+ return ret;
+}
+
+static int __devexit mop500_remove(struct platform_device *pdev)
+{
+ struct snd_soc_card *mop500_card = platform_get_drvdata(pdev);
+
+ pr_debug("%s: Enter.\n", __func__);
+
+ snd_soc_unregister_card(mop500_card);
+ mop500_ab8500_remove(mop500_card);
+
+ return 0;
+}
+
+static struct platform_driver snd_soc_mop500_driver = {
+ .driver = {
+ .owner = THIS_MODULE,
+ .name = "snd-soc-mop500",
+ },
+ .probe = mop500_probe,
+ .remove = __devexit_p(mop500_remove),
+};
+
+module_platform_driver(snd_soc_mop500_driver);
diff --git a/sound/soc/ux500/mop500_ab8500.c b/sound/soc/ux500/mop500_ab8500.c
new file mode 100644
index 0000000..78cce23
--- /dev/null
+++ b/sound/soc/ux500/mop500_ab8500.c
@@ -0,0 +1,431 @@
+/*
+ * Copyright (C) ST-Ericsson SA 2012
+ *
+ * Author: Ola Lilja <ola.o.lilja@stericsson.com>,
+ * Kristoffer Karlsson <kristoffer.karlsson@stericsson.com>
+ * for ST-Ericsson.
+ *
+ * License terms:
+ *
+ * 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 <linux/module.h>
+#include <linux/device.h>
+#include <linux/io.h>
+#include <linux/clk.h>
+
+#include <mach/hardware.h>
+
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+
+#include "ux500_pcm.h"
+#include "ux500_msp_dai.h"
+#include "../codecs/ab8500-codec.h"
+
+#define TX_SLOT_MONO 0x0008
+#define TX_SLOT_STEREO 0x000a
+#define RX_SLOT_MONO 0x0001
+#define RX_SLOT_STEREO 0x0003
+#define TX_SLOT_8CH 0x00FF
+#define RX_SLOT_8CH 0x00FF
+
+#define DEF_TX_SLOTS TX_SLOT_STEREO
+#define DEF_RX_SLOTS RX_SLOT_MONO
+
+#define DRIVERMODE_NORMAL 0
+#define DRIVERMODE_CODEC_ONLY 1
+
+/* Slot configuration */
+static unsigned int tx_slots = DEF_TX_SLOTS;
+static unsigned int rx_slots = DEF_RX_SLOTS;
+
+/* Clocks */
+static const char * const enum_mclk[] = {
+ "SYSCLK",
+ "ULPCLK"
+};
+enum mclk {
+ MCLK_SYSCLK,
+ MCLK_ULPCLK,
+};
+
+static SOC_ENUM_SINGLE_EXT_DECL(soc_enum_mclk, enum_mclk);
+
+/* Private data for machine-part MOP500<->AB8500 */
+struct mop500_ab8500_drvdata {
+ /* Clocks */
+ enum mclk mclk_sel;
+ struct clk *clk_ptr_intclk;
+ struct clk *clk_ptr_sysclk;
+ struct clk *clk_ptr_ulpclk;
+};
+
+static inline const char *get_mclk_str(enum mclk mclk_sel)
+{
+ switch (mclk_sel) {
+ case MCLK_SYSCLK:
+ return "SYSCLK";
+ case MCLK_ULPCLK:
+ return "ULPCLK";
+ default:
+ return "Unknown";
+ }
+}
+
+static int mop500_ab8500_set_mclk(struct device *dev,
+ struct mop500_ab8500_drvdata *drvdata)
+{
+ int status;
+ struct clk *clk_ptr;
+
+ if (IS_ERR(drvdata->clk_ptr_intclk)) {
+ dev_err(dev,
+ "%s: ERROR: intclk not initialized!\n", __func__);
+ return -EIO;
+ }
+
+ switch (drvdata->mclk_sel) {
+ case MCLK_SYSCLK:
+ clk_ptr = drvdata->clk_ptr_sysclk;
+ break;
+ case MCLK_ULPCLK:
+ clk_ptr = drvdata->clk_ptr_ulpclk;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ if (IS_ERR(clk_ptr)) {
+ dev_err(dev, "%s: ERROR: %s not initialized!\n", __func__,
+ get_mclk_str(drvdata->mclk_sel));
+ return -EIO;
+ }
+
+ status = clk_set_parent(drvdata->clk_ptr_intclk, clk_ptr);
+ if (status)
+ dev_err(dev,
+ "%s: ERROR: Setting intclk parent to %s failed (ret = %d)!",
+ __func__, get_mclk_str(drvdata->mclk_sel), status);
+ else
+ dev_dbg(dev,
+ "%s: intclk parent changed to %s.\n",
+ __func__, get_mclk_str(drvdata->mclk_sel));
+
+ return status;
+}
+
+/*
+ * Control-events
+ */
+
+static int mclk_input_control_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct mop500_ab8500_drvdata *drvdata =
+ snd_soc_card_get_drvdata(codec->card);
+
+ ucontrol->value.enumerated.item[0] = drvdata->mclk_sel;
+
+ return 0;
+}
+
+static int mclk_input_control_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct mop500_ab8500_drvdata *drvdata =
+ snd_soc_card_get_drvdata(codec->card);
+ unsigned int val = ucontrol->value.enumerated.item[0];
+
+ if (val > (unsigned int)MCLK_ULPCLK)
+ return -EINVAL;
+ if (drvdata->mclk_sel == val)
+ return 0;
+
+ drvdata->mclk_sel = val;
+
+ return 1;
+}
+
+/*
+ * Controls
+ */
+
+static struct snd_kcontrol_new mop500_ab8500_ctrls[] = {
+ SOC_ENUM_EXT("Master Clock Select",
+ soc_enum_mclk,
+ mclk_input_control_get, mclk_input_control_put),
+ /* Digital interface - Clocks */
+ SOC_SINGLE("Digital Interface Master Generator Switch",
+ AB8500_DIGIFCONF1, AB8500_DIGIFCONF1_ENMASTGEN,
+ 1, 0),
+ SOC_SINGLE("Digital Interface 0 Bit-clock Switch",
+ AB8500_DIGIFCONF1, AB8500_DIGIFCONF1_ENFSBITCLK0,
+ 1, 0),
+ SOC_SINGLE("Digital Interface 1 Bit-clock Switch",
+ AB8500_DIGIFCONF1, AB8500_DIGIFCONF1_ENFSBITCLK1,
+ 1, 0),
+ SOC_DAPM_PIN_SWITCH("Headset Left"),
+ SOC_DAPM_PIN_SWITCH("Headset Right"),
+ SOC_DAPM_PIN_SWITCH("Earpiece"),
+ SOC_DAPM_PIN_SWITCH("Speaker Left"),
+ SOC_DAPM_PIN_SWITCH("Speaker Right"),
+ SOC_DAPM_PIN_SWITCH("LineOut Left"),
+ SOC_DAPM_PIN_SWITCH("LineOut Right"),
+ SOC_DAPM_PIN_SWITCH("Vibra 1"),
+ SOC_DAPM_PIN_SWITCH("Vibra 2"),
+ SOC_DAPM_PIN_SWITCH("Mic 1"),
+ SOC_DAPM_PIN_SWITCH("Mic 2"),
+ SOC_DAPM_PIN_SWITCH("LineIn Left"),
+ SOC_DAPM_PIN_SWITCH("LineIn Right"),
+ SOC_DAPM_PIN_SWITCH("DMic 1"),
+ SOC_DAPM_PIN_SWITCH("DMic 2"),
+ SOC_DAPM_PIN_SWITCH("DMic 3"),
+ SOC_DAPM_PIN_SWITCH("DMic 4"),
+ SOC_DAPM_PIN_SWITCH("DMic 5"),
+ SOC_DAPM_PIN_SWITCH("DMic 6"),
+};
+
+/* ASoC */
+
+int mop500_ab8500_startup(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+
+ /* Set audio-clock source */
+ return mop500_ab8500_set_mclk(rtd->card->dev,
+ snd_soc_card_get_drvdata(rtd->card));
+}
+
+void mop500_ab8500_shutdown(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct device *dev = rtd->card->dev;
+
+ dev_dbg(dev, "%s: Enter\n", __func__);
+
+ /* Reset slots configuration to default(s) */
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ tx_slots = DEF_TX_SLOTS;
+ else
+ rx_slots = DEF_RX_SLOTS;
+}
+
+int mop500_ab8500_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct device *dev = rtd->card->dev;
+ unsigned int fmt;
+ int channels, ret = 0, driver_mode, slots;
+ unsigned int sw_codec, sw_cpu;
+ bool is_playback;
+
+ dev_dbg(dev, "%s: Enter\n", __func__);
+
+ dev_dbg(dev, "%s: substream->pcm->name = %s\n"
+ "substream->pcm->id = %s.\n"
+ "substream->name = %s.\n"
+ "substream->number = %d.\n",
+ __func__,
+ substream->pcm->name,
+ substream->pcm->id,
+ substream->name,
+ substream->number);
+
+ channels = params_channels(params);
+
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S32_LE:
+ sw_cpu = 32;
+ break;
+
+ case SNDRV_PCM_FORMAT_S16_LE:
+ sw_cpu = 16;
+ break;
+
+ default:
+ return -EINVAL;
+ }
+
+ /* Setup codec depending on driver-mode */
+ if (channels == 8)
+ driver_mode = DRIVERMODE_CODEC_ONLY;
+ else
+ driver_mode = DRIVERMODE_NORMAL;
+ dev_dbg(dev, "%s: Driver-mode: %s.\n", __func__,
+ (driver_mode == DRIVERMODE_NORMAL) ? "NORMAL" : "CODEC_ONLY");
+
+ /* Setup format */
+
+ if (driver_mode == DRIVERMODE_NORMAL) {
+ fmt = SND_SOC_DAIFMT_DSP_A |
+ SND_SOC_DAIFMT_CBM_CFM |
+ SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CONT;
+ } else {
+ fmt = SND_SOC_DAIFMT_DSP_A |
+ SND_SOC_DAIFMT_CBM_CFM |
+ SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_GATED;
+ }
+
+ ret = snd_soc_dai_set_fmt(codec_dai, fmt);
+ if (ret < 0) {
+ dev_err(dev,
+ "%s: ERROR: snd_soc_dai_set_fmt failed for codec_dai (ret = %d)!\n",
+ __func__, ret);
+ return ret;
+ }
+
+ ret = snd_soc_dai_set_fmt(cpu_dai, fmt);
+ if (ret < 0) {
+ dev_err(dev,
+ "%s: ERROR: snd_soc_dai_set_fmt failed for cpu_dai (ret = %d)!\n",
+ __func__, ret);
+ return ret;
+ }
+
+ /* Setup TDM-slots */
+
+ is_playback = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK);
+ switch (channels) {
+ case 1:
+ slots = 16;
+ tx_slots = (is_playback) ? TX_SLOT_MONO : 0;
+ rx_slots = (is_playback) ? 0 : RX_SLOT_MONO;
+ break;
+ case 2:
+ slots = 16;
+ tx_slots = (is_playback) ? TX_SLOT_STEREO : 0;
+ rx_slots = (is_playback) ? 0 : RX_SLOT_STEREO;
+ break;
+ case 8:
+ slots = 16;
+ tx_slots = (is_playback) ? TX_SLOT_8CH : 0;
+ rx_slots = (is_playback) ? 0 : RX_SLOT_8CH;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ if (driver_mode == DRIVERMODE_NORMAL)
+ sw_codec = sw_cpu;
+ else
+ sw_codec = 20;
+
+ dev_dbg(dev, "%s: CPU-DAI TDM: TX=0x%04X RX=0x%04x\n", __func__,
+ tx_slots, rx_slots);
+ ret = snd_soc_dai_set_tdm_slot(cpu_dai, tx_slots, rx_slots, slots,
+ sw_cpu);
+ if (ret)
+ return ret;
+
+ dev_dbg(dev, "%s: CODEC-DAI TDM: TX=0x%04X RX=0x%04x\n", __func__,
+ tx_slots, rx_slots);
+ ret = snd_soc_dai_set_tdm_slot(codec_dai, tx_slots, rx_slots, slots,
+ sw_codec);
+ if (ret)
+ return ret;
+
+ return 0;
+}
+
+struct snd_soc_ops mop500_ab8500_ops[] = {
+ {
+ .hw_params = mop500_ab8500_hw_params,
+ .startup = mop500_ab8500_startup,
+ .shutdown = mop500_ab8500_shutdown,
+ }
+};
+
+int mop500_ab8500_machine_init(struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_soc_codec *codec = rtd->codec;
+ struct device *dev = rtd->card->dev;
+ struct mop500_ab8500_drvdata *drvdata;
+ int ret;
+
+ dev_dbg(dev, "%s Enter.\n", __func__);
+
+ /* Create driver private-data struct */
+ drvdata = devm_kzalloc(dev, sizeof(struct mop500_ab8500_drvdata),
+ GFP_KERNEL);
+ snd_soc_card_set_drvdata(rtd->card, drvdata);
+
+ /* Setup clocks */
+
+ drvdata->clk_ptr_sysclk = clk_get(dev, "sysclk");
+ if (IS_ERR(drvdata->clk_ptr_sysclk))
+ dev_warn(dev, "%s: WARNING: clk_get failed for 'sysclk'!\n",
+ __func__);
+ drvdata->clk_ptr_ulpclk = clk_get(dev, "ulpclk");
+ if (IS_ERR(drvdata->clk_ptr_ulpclk))
+ dev_warn(dev, "%s: WARNING: clk_get failed for 'ulpclk'!\n",
+ __func__);
+ drvdata->clk_ptr_intclk = clk_get(dev, "intclk");
+ if (IS_ERR(drvdata->clk_ptr_intclk))
+ dev_warn(dev, "%s: WARNING: clk_get failed for 'intclk'!\n",
+ __func__);
+
+ /* Set intclk default parent to ulpclk */
+ drvdata->mclk_sel = MCLK_ULPCLK;
+ ret = mop500_ab8500_set_mclk(dev, drvdata);
+ if (ret < 0)
+ dev_warn(dev, "%s: WARNING: mop500_ab8500_set_mclk!\n",
+ __func__);
+
+ drvdata->mclk_sel = MCLK_ULPCLK;
+
+ /* Add controls */
+ ret = snd_soc_add_codec_controls(codec, mop500_ab8500_ctrls,
+ ARRAY_SIZE(mop500_ab8500_ctrls));
+ if (ret < 0) {
+ pr_err("%s: Failed to add machine-controls (%d)!\n",
+ __func__, ret);
+ return ret;
+ }
+
+ ret = snd_soc_dapm_disable_pin(&codec->dapm, "Earpiece");
+ ret |= snd_soc_dapm_disable_pin(&codec->dapm, "Speaker Left");
+ ret |= snd_soc_dapm_disable_pin(&codec->dapm, "Speaker Right");
+ ret |= snd_soc_dapm_disable_pin(&codec->dapm, "LineOut Left");
+ ret |= snd_soc_dapm_disable_pin(&codec->dapm, "LineOut Right");
+ ret |= snd_soc_dapm_disable_pin(&codec->dapm, "Vibra 1");
+ ret |= snd_soc_dapm_disable_pin(&codec->dapm, "Vibra 2");
+ ret |= snd_soc_dapm_disable_pin(&codec->dapm, "Mic 1");
+ ret |= snd_soc_dapm_disable_pin(&codec->dapm, "Mic 2");
+ ret |= snd_soc_dapm_disable_pin(&codec->dapm, "LineIn Left");
+ ret |= snd_soc_dapm_disable_pin(&codec->dapm, "LineIn Right");
+ ret |= snd_soc_dapm_disable_pin(&codec->dapm, "DMic 1");
+ ret |= snd_soc_dapm_disable_pin(&codec->dapm, "DMic 2");
+ ret |= snd_soc_dapm_disable_pin(&codec->dapm, "DMic 3");
+ ret |= snd_soc_dapm_disable_pin(&codec->dapm, "DMic 4");
+ ret |= snd_soc_dapm_disable_pin(&codec->dapm, "DMic 5");
+ ret |= snd_soc_dapm_disable_pin(&codec->dapm, "DMic 6");
+
+ return ret;
+}
+
+void mop500_ab8500_remove(struct snd_soc_card *card)
+{
+ struct mop500_ab8500_drvdata *drvdata = snd_soc_card_get_drvdata(card);
+
+ if (drvdata->clk_ptr_sysclk != NULL)
+ clk_put(drvdata->clk_ptr_sysclk);
+ if (drvdata->clk_ptr_ulpclk != NULL)
+ clk_put(drvdata->clk_ptr_ulpclk);
+ if (drvdata->clk_ptr_intclk != NULL)
+ clk_put(drvdata->clk_ptr_intclk);
+
+ snd_soc_card_set_drvdata(card, drvdata);
+}
diff --git a/sound/soc/ux500/mop500_ab8500.h b/sound/soc/ux500/mop500_ab8500.h
new file mode 100644
index 0000000..cca5b33
--- /dev/null
+++ b/sound/soc/ux500/mop500_ab8500.h
@@ -0,0 +1,22 @@
+/*
+ * Copyright (C) ST-Ericsson SA 2012
+ *
+ * Author: Ola Lilja <ola.o.lilja@stericsson.com>
+ * for ST-Ericsson.
+ *
+ * License terms:
+ *
+ * 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 MOP500_AB8500_H
+#define MOP500_AB8500_H
+
+extern struct snd_soc_ops mop500_ab8500_ops[];
+
+int mop500_ab8500_machine_init(struct snd_soc_pcm_runtime *runtime);
+void mop500_ab8500_remove(struct snd_soc_card *card);
+
+#endif
diff --git a/sound/soc/ux500/ux500_msp_dai.c b/sound/soc/ux500/ux500_msp_dai.c
index 93c6c40..62ac028 100644
--- a/sound/soc/ux500/ux500_msp_dai.c
+++ b/sound/soc/ux500/ux500_msp_dai.c
@@ -840,4 +840,4 @@ static struct platform_driver msp_i2s_driver = {
};
module_platform_driver(msp_i2s_driver);
-MODULE_LICENSE("GPLv2");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/ux500/ux500_msp_i2s.c b/sound/soc/ux500/ux500_msp_i2s.c
index 496dec1..ee14d2d 100644
--- a/sound/soc/ux500/ux500_msp_i2s.c
+++ b/sound/soc/ux500/ux500_msp_i2s.c
@@ -739,4 +739,4 @@ void ux500_msp_i2s_cleanup_msp(struct platform_device *pdev,
devm_kfree(&pdev->dev, msp);
}
-MODULE_LICENSE("GPLv2");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/ux500/ux500_pcm.c b/sound/soc/ux500/ux500_pcm.c
new file mode 100644
index 0000000..1a04e24
--- /dev/null
+++ b/sound/soc/ux500/ux500_pcm.c
@@ -0,0 +1,318 @@
+/*
+ * Copyright (C) ST-Ericsson SA 2012
+ *
+ * Author: Ola Lilja <ola.o.lilja@stericsson.com>,
+ * Roger Nilsson <roger.xr.nilsson@stericsson.com>
+ * for ST-Ericsson.
+ *
+ * License terms:
+ *
+ * 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 <asm/page.h>
+
+#include <linux/module.h>
+#include <linux/dma-mapping.h>
+#include <linux/dmaengine.h>
+#include <linux/slab.h>
+
+#include <plat/ste_dma40.h>
+
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/dmaengine_pcm.h>
+
+#include "ux500_msp_i2s.h"
+#include "ux500_pcm.h"
+
+static struct snd_pcm_hardware ux500_pcm_hw_playback = {
+ .info = SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_RESUME |
+ SNDRV_PCM_INFO_PAUSE,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_U16_LE |
+ SNDRV_PCM_FMTBIT_S16_BE |
+ SNDRV_PCM_FMTBIT_U16_BE,
+ .rates = SNDRV_PCM_RATE_KNOT,
+ .rate_min = UX500_PLATFORM_MIN_RATE_PLAYBACK,
+ .rate_max = UX500_PLATFORM_MAX_RATE_PLAYBACK,
+ .channels_min = UX500_PLATFORM_MIN_CHANNELS,
+ .channels_max = UX500_PLATFORM_MAX_CHANNELS,
+ .buffer_bytes_max = UX500_PLATFORM_BUFFER_BYTES_MAX,
+ .period_bytes_min = UX500_PLATFORM_PERIODS_BYTES_MIN,
+ .period_bytes_max = UX500_PLATFORM_PERIODS_BYTES_MAX,
+ .periods_min = UX500_PLATFORM_PERIODS_MIN,
+ .periods_max = UX500_PLATFORM_PERIODS_MAX,
+};
+
+static struct snd_pcm_hardware ux500_pcm_hw_capture = {
+ .info = SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_RESUME |
+ SNDRV_PCM_INFO_PAUSE,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_U16_LE |
+ SNDRV_PCM_FMTBIT_S16_BE |
+ SNDRV_PCM_FMTBIT_U16_BE,
+ .rates = SNDRV_PCM_RATE_KNOT,
+ .rate_min = UX500_PLATFORM_MIN_RATE_CAPTURE,
+ .rate_max = UX500_PLATFORM_MAX_RATE_CAPTURE,
+ .channels_min = UX500_PLATFORM_MIN_CHANNELS,
+ .channels_max = UX500_PLATFORM_MAX_CHANNELS,
+ .buffer_bytes_max = UX500_PLATFORM_BUFFER_BYTES_MAX,
+ .period_bytes_min = UX500_PLATFORM_PERIODS_BYTES_MIN,
+ .period_bytes_max = UX500_PLATFORM_PERIODS_BYTES_MAX,
+ .periods_min = UX500_PLATFORM_PERIODS_MIN,
+ .periods_max = UX500_PLATFORM_PERIODS_MAX,
+};
+
+static void ux500_pcm_dma_hw_free(struct device *dev,
+ struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_dma_buffer *buf = runtime->dma_buffer_p;
+
+ if (runtime->dma_area == NULL)
+ return;
+
+ if (buf != &substream->dma_buffer) {
+ dma_free_coherent(buf->dev.dev, buf->bytes, buf->area,
+ buf->addr);
+ kfree(runtime->dma_buffer_p);
+ }
+
+ snd_pcm_set_runtime_buffer(substream, NULL);
+}
+
+static int ux500_pcm_open(struct snd_pcm_substream *substream)
+{
+ int stream_id = substream->pstr->stream;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *dai = rtd->cpu_dai;
+ struct device *dev = dai->dev;
+ int ret;
+ struct ux500_msp_dma_params *dma_params;
+ u16 per_data_width, mem_data_width;
+ struct stedma40_chan_cfg *dma_cfg;
+
+ dev_dbg(dev, "%s: MSP %d (%s): Enter.\n", __func__, dai->id,
+ snd_pcm_stream_str(substream));
+
+ dev_dbg(dev, "%s: Set runtime hwparams.\n", __func__);
+ if (stream_id == SNDRV_PCM_STREAM_PLAYBACK)
+ snd_soc_set_runtime_hwparams(substream,
+ &ux500_pcm_hw_playback);
+ else
+ snd_soc_set_runtime_hwparams(substream,
+ &ux500_pcm_hw_capture);
+
+ /* ensure that buffer size is a multiple of period size */
+ ret = snd_pcm_hw_constraint_integer(runtime,
+ SNDRV_PCM_HW_PARAM_PERIODS);
+ if (ret < 0) {
+ dev_err(dev, "%s: Error: snd_pcm_hw_constraints failed (%d)\n",
+ __func__, ret);
+ return ret;
+ }
+
+ dev_dbg(dev, "%s: Set hw-struct for %s.\n", __func__,
+ snd_pcm_stream_str(substream));
+ runtime->hw = (stream_id == SNDRV_PCM_STREAM_PLAYBACK) ?
+ ux500_pcm_hw_playback : ux500_pcm_hw_capture;
+
+ mem_data_width = STEDMA40_HALFWORD_WIDTH;
+
+ dma_params = snd_soc_dai_get_dma_data(dai, substream);
+ switch (dma_params->data_size) {
+ case 32:
+ per_data_width = STEDMA40_WORD_WIDTH;
+ break;
+ case 16:
+ per_data_width = STEDMA40_HALFWORD_WIDTH;
+ break;
+ case 8:
+ per_data_width = STEDMA40_BYTE_WIDTH;
+ break;
+ default:
+ per_data_width = STEDMA40_WORD_WIDTH;
+ dev_warn(rtd->platform->dev,
+ "%s: Unknown data-size (%d)! Assuming 32 bits.\n",
+ __func__, dma_params->data_size);
+ }
+
+ dma_cfg = dma_params->dma_cfg;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ dma_cfg->src_info.data_width = mem_data_width;
+ dma_cfg->dst_info.data_width = per_data_width;
+ } else {
+ dma_cfg->src_info.data_width = per_data_width;
+ dma_cfg->dst_info.data_width = mem_data_width;
+ }
+
+
+ ret = snd_dmaengine_pcm_open(substream, stedma40_filter, dma_cfg);
+ if (ret) {
+ dev_dbg(dai->dev,
+ "%s: ERROR: snd_dmaengine_pcm_open failed (%d)!\n",
+ __func__, ret);
+ return ret;
+ }
+
+ snd_dmaengine_pcm_set_data(substream, dma_cfg);
+
+ return 0;
+}
+
+static int ux500_pcm_close(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *dai = rtd->cpu_dai;
+
+ dev_dbg(dai->dev, "%s: Enter\n", __func__);
+
+ snd_dmaengine_pcm_close(substream);
+
+ return 0;
+}
+
+static int ux500_pcm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_dma_buffer *buf = runtime->dma_buffer_p;
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ int ret = 0;
+ int size;
+
+ dev_dbg(rtd->platform->dev, "%s: Enter\n", __func__);
+
+ size = params_buffer_bytes(hw_params);
+
+ if (buf) {
+ if (buf->bytes >= size)
+ goto out;
+ ux500_pcm_dma_hw_free(NULL, substream);
+ }
+
+ if (substream->dma_buffer.area != NULL &&
+ substream->dma_buffer.bytes >= size) {
+ buf = &substream->dma_buffer;
+ } else {
+ buf = kmalloc(sizeof(struct snd_dma_buffer), GFP_KERNEL);
+ if (!buf)
+ goto nomem;
+
+ buf->dev.type = SNDRV_DMA_TYPE_DEV;
+ buf->dev.dev = NULL;
+ buf->area = dma_alloc_coherent(NULL, size, &buf->addr,
+ GFP_KERNEL);
+ buf->bytes = size;
+ buf->private_data = NULL;
+
+ if (!buf->area)
+ goto free;
+ }
+ snd_pcm_set_runtime_buffer(substream, buf);
+ ret = 1;
+ out:
+ runtime->dma_bytes = size;
+ return ret;
+
+ free:
+ kfree(buf);
+ nomem:
+ return -ENOMEM;
+}
+
+static int ux500_pcm_hw_free(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+
+ dev_dbg(rtd->platform->dev, "%s: Enter\n", __func__);
+
+ ux500_pcm_dma_hw_free(NULL, substream);
+
+ return 0;
+}
+
+static int ux500_pcm_mmap(struct snd_pcm_substream *substream,
+ struct vm_area_struct *vma)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+
+ dev_dbg(rtd->platform->dev, "%s: Enter.\n", __func__);
+
+ return dma_mmap_coherent(NULL, vma, runtime->dma_area,
+ runtime->dma_addr, runtime->dma_bytes);
+}
+
+static struct snd_pcm_ops ux500_pcm_ops = {
+ .open = ux500_pcm_open,
+ .close = ux500_pcm_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = ux500_pcm_hw_params,
+ .hw_free = ux500_pcm_hw_free,
+ .trigger = snd_dmaengine_pcm_trigger,
+ .pointer = snd_dmaengine_pcm_pointer_no_residue,
+ .mmap = ux500_pcm_mmap
+};
+
+int ux500_pcm_new(struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_pcm *pcm = rtd->pcm;
+
+ dev_dbg(rtd->platform->dev, "%s: Enter (id = '%s').\n", __func__,
+ pcm->id);
+
+ pcm->info_flags = 0;
+
+ return 0;
+}
+
+static struct snd_soc_platform_driver ux500_pcm_soc_drv = {
+ .ops = &ux500_pcm_ops,
+ .pcm_new = ux500_pcm_new,
+};
+
+static int __devexit ux500_pcm_drv_probe(struct platform_device *pdev)
+{
+ int ret;
+
+ ret = snd_soc_register_platform(&pdev->dev, &ux500_pcm_soc_drv);
+ if (ret < 0) {
+ dev_err(&pdev->dev,
+ "%s: ERROR: Failed to register platform '%s' (%d)!\n",
+ __func__, pdev->name, ret);
+ return ret;
+ }
+
+ return 0;
+}
+
+static int __devinit ux500_pcm_drv_remove(struct platform_device *pdev)
+{
+ snd_soc_unregister_platform(&pdev->dev);
+
+ return 0;
+}
+
+static struct platform_driver ux500_pcm_driver = {
+ .driver = {
+ .name = "ux500-pcm",
+ .owner = THIS_MODULE,
+ },
+
+ .probe = ux500_pcm_drv_probe,
+ .remove = __devexit_p(ux500_pcm_drv_remove),
+};
+module_platform_driver(ux500_pcm_driver);
+
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/ux500/ux500_pcm.h b/sound/soc/ux500/ux500_pcm.h
new file mode 100644
index 0000000..77ed44d
--- /dev/null
+++ b/sound/soc/ux500/ux500_pcm.h
@@ -0,0 +1,35 @@
+/*
+ * Copyright (C) ST-Ericsson SA 2012
+ *
+ * Author: Ola Lilja <ola.o.lilja@stericsson.com>,
+ * Roger Nilsson <roger.xr.nilsson@stericsson.com>
+ * for ST-Ericsson.
+ *
+ * License terms:
+ *
+ * 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 UX500_PCM_H
+#define UX500_PCM_H
+
+#include <asm/page.h>
+
+#include <linux/workqueue.h>
+
+#define UX500_PLATFORM_MIN_RATE_PLAYBACK 8000
+#define UX500_PLATFORM_MAX_RATE_PLAYBACK 48000
+#define UX500_PLATFORM_MIN_RATE_CAPTURE 8000
+#define UX500_PLATFORM_MAX_RATE_CAPTURE 48000
+
+#define UX500_PLATFORM_MIN_CHANNELS 1
+#define UX500_PLATFORM_MAX_CHANNELS 8
+
+#define UX500_PLATFORM_PERIODS_BYTES_MIN 128
+#define UX500_PLATFORM_PERIODS_BYTES_MAX (64 * PAGE_SIZE)
+#define UX500_PLATFORM_PERIODS_MIN 2
+#define UX500_PLATFORM_PERIODS_MAX 48
+#define UX500_PLATFORM_BUFFER_BYTES_MAX (2048 * PAGE_SIZE)
+
+#endif
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