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authorAnton Arapov <anton@redhat.com>2012-08-07 11:21:50 +0200
committerAnton Arapov <anton@redhat.com>2012-08-07 12:52:25 +0200
commit1d44b6f3fcf6058fb7c960b7558766967e8028f7 (patch)
tree53d88547c973ba048d233091a3f91f3173ad01df /sound/soc
parentd91eda5d7b0383e6a0c83e0146ff141ff3b1355b (diff)
downloadkernel-uprobes-1d44b6f3fcf6058fb7c960b7558766967e8028f7.tar.gz
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fedora kernel: 222b075b3ff0d9e88aa9353e3c80667756ed7361v3.5.0-4
Signed-off-by: Anton Arapov <anton@redhat.com>
Diffstat (limited to 'sound/soc')
-rw-r--r--sound/soc/Kconfig5
-rw-r--r--sound/soc/Makefile3
-rw-r--r--sound/soc/blackfin/bf5xx-ssm2602.c37
-rw-r--r--sound/soc/codecs/Kconfig20
-rw-r--r--sound/soc/codecs/Makefile12
-rw-r--r--sound/soc/codecs/ac97.c6
-rw-r--r--sound/soc/codecs/ad1836.c4
-rw-r--r--sound/soc/codecs/ad193x.c4
-rw-r--r--sound/soc/codecs/adau1701.c3
-rw-r--r--sound/soc/codecs/ak4104.c3
-rw-r--r--sound/soc/codecs/ak4535.c3
-rw-r--r--sound/soc/codecs/ak4641.c113
-rw-r--r--sound/soc/codecs/alc5623.c23
-rw-r--r--sound/soc/codecs/alc5632.c31
-rw-r--r--sound/soc/codecs/cs4270.c11
-rw-r--r--sound/soc/codecs/cs4271.c3
-rw-r--r--sound/soc/codecs/cs42l51.c9
-rw-r--r--sound/soc/codecs/cs42l52.c1295
-rw-r--r--sound/soc/codecs/cs42l52.h274
-rw-r--r--sound/soc/codecs/cs42l73.c93
-rw-r--r--sound/soc/codecs/da7210.c379
-rw-r--r--sound/soc/codecs/jz4740.c3
-rw-r--r--sound/soc/codecs/lm49453.c1550
-rw-r--r--sound/soc/codecs/lm49453.h380
-rw-r--r--sound/soc/codecs/max98095.c158
-rw-r--r--sound/soc/codecs/max98095.h22
-rw-r--r--sound/soc/codecs/mc13783.c786
-rw-r--r--sound/soc/codecs/mc13783.h28
-rw-r--r--sound/soc/codecs/ml26124.c681
-rw-r--r--sound/soc/codecs/ml26124.h184
-rw-r--r--sound/soc/codecs/omap-hdmi.c69
-rw-r--r--sound/soc/codecs/rt5631.c110
-rw-r--r--sound/soc/codecs/sgtl5000.c33
-rw-r--r--sound/soc/codecs/ssm2602.c138
-rw-r--r--sound/soc/codecs/sta32x.c3
-rw-r--r--sound/soc/codecs/tlv320aic23.c13
-rw-r--r--sound/soc/codecs/tlv320aic26.c3
-rw-r--r--sound/soc/codecs/tlv320aic3x.c25
-rw-r--r--sound/soc/codecs/tlv320aic3x.h1
-rw-r--r--sound/soc/codecs/tlv320dac33.c35
-rw-r--r--sound/soc/codecs/twl4030.c18
-rw-r--r--sound/soc/codecs/twl6040.c450
-rw-r--r--sound/soc/codecs/uda134x.c6
-rw-r--r--sound/soc/codecs/uda1380.c6
-rw-r--r--sound/soc/codecs/wl1273.c6
-rw-r--r--sound/soc/codecs/wm1250-ev1.c65
-rw-r--r--sound/soc/codecs/wm2000.c59
-rw-r--r--sound/soc/codecs/wm2200.c1
-rw-r--r--sound/soc/codecs/wm5100-tables.c125
-rw-r--r--sound/soc/codecs/wm5100.c47
-rw-r--r--sound/soc/codecs/wm5100.h159
-rw-r--r--sound/soc/codecs/wm8350.c187
-rw-r--r--sound/soc/codecs/wm8400.c135
-rw-r--r--sound/soc/codecs/wm8510.c3
-rw-r--r--sound/soc/codecs/wm8523.c3
-rw-r--r--sound/soc/codecs/wm8728.c3
-rw-r--r--sound/soc/codecs/wm8731.c37
-rw-r--r--sound/soc/codecs/wm8737.c3
-rw-r--r--sound/soc/codecs/wm8741.c3
-rw-r--r--sound/soc/codecs/wm8750.c3
-rw-r--r--sound/soc/codecs/wm8753.c6
-rw-r--r--sound/soc/codecs/wm8900.c3
-rw-r--r--sound/soc/codecs/wm8903.c3
-rw-r--r--sound/soc/codecs/wm8904.c13
-rw-r--r--sound/soc/codecs/wm8940.c3
-rw-r--r--sound/soc/codecs/wm8960.c3
-rw-r--r--sound/soc/codecs/wm8962.c21
-rw-r--r--sound/soc/codecs/wm8971.c3
-rw-r--r--sound/soc/codecs/wm8978.c3
-rw-r--r--sound/soc/codecs/wm8988.c3
-rw-r--r--sound/soc/codecs/wm8990.c3
-rw-r--r--sound/soc/codecs/wm8993.c86
-rw-r--r--sound/soc/codecs/wm8994.c287
-rw-r--r--sound/soc/codecs/wm8994.h3
-rw-r--r--sound/soc/codecs/wm8996.c20
-rw-r--r--sound/soc/codecs/wm9081.c5
-rw-r--r--sound/soc/codecs/wm9705.c6
-rw-r--r--sound/soc/codecs/wm9712.c10
-rw-r--r--sound/soc/codecs/wm_hubs.c220
-rw-r--r--sound/soc/codecs/wm_hubs.h12
-rw-r--r--sound/soc/ep93xx/ep93xx-ac97.c74
-rw-r--r--sound/soc/ep93xx/ep93xx-i2s.c49
-rw-r--r--sound/soc/fsl/Kconfig129
-rw-r--r--sound/soc/fsl/Makefile31
-rw-r--r--sound/soc/fsl/eukrea-tlv320.c (renamed from sound/soc/imx/eukrea-tlv320.c)2
-rw-r--r--sound/soc/fsl/fsl_ssi.c167
-rw-r--r--sound/soc/fsl/fsl_utils.c91
-rw-r--r--sound/soc/fsl/fsl_utils.h26
-rw-r--r--sound/soc/fsl/imx-audmux.c (renamed from sound/soc/imx/imx-audmux.c)8
-rw-r--r--sound/soc/fsl/imx-audmux.h (renamed from sound/soc/imx/imx-audmux.h)0
-rw-r--r--sound/soc/fsl/imx-mc13783.c156
-rw-r--r--sound/soc/fsl/imx-pcm-dma.c (renamed from sound/soc/imx/imx-pcm-dma-mx2.c)3
-rw-r--r--sound/soc/fsl/imx-pcm-fiq.c (renamed from sound/soc/imx/imx-pcm-fiq.c)0
-rw-r--r--sound/soc/fsl/imx-pcm.c (renamed from sound/soc/imx/imx-pcm.c)0
-rw-r--r--sound/soc/fsl/imx-pcm.h (renamed from sound/soc/imx/imx-pcm.h)1
-rw-r--r--sound/soc/fsl/imx-sgtl5000.c221
-rw-r--r--sound/soc/fsl/imx-ssi.c (renamed from sound/soc/imx/imx-ssi.c)8
-rw-r--r--sound/soc/fsl/imx-ssi.h (renamed from sound/soc/imx/imx-ssi.h)0
-rw-r--r--sound/soc/fsl/mpc8610_hpcd.c166
-rw-r--r--sound/soc/fsl/mx27vis-aic32x4.c (renamed from sound/soc/imx/mx27vis-aic32x4.c)0
-rw-r--r--sound/soc/fsl/p1022_ds.c158
-rw-r--r--sound/soc/fsl/phycore-ac97.c (renamed from sound/soc/imx/phycore-ac97.c)0
-rw-r--r--sound/soc/fsl/wm1133-ev1.c (renamed from sound/soc/imx/wm1133-ev1.c)0
-rw-r--r--sound/soc/generic/Kconfig4
-rw-r--r--sound/soc/generic/Makefile3
-rw-r--r--sound/soc/generic/simple-card.c114
-rw-r--r--sound/soc/imx/Kconfig79
-rw-r--r--sound/soc/imx/Makefile22
-rw-r--r--sound/soc/jz4740/jz4740-i2s.c4
-rw-r--r--sound/soc/kirkwood/kirkwood-i2s.c13
-rw-r--r--sound/soc/kirkwood/kirkwood.h1
-rw-r--r--sound/soc/mxs/mxs-pcm.c24
-rw-r--r--sound/soc/mxs/mxs-pcm.h3
-rw-r--r--sound/soc/mxs/mxs-saif.c100
-rw-r--r--sound/soc/mxs/mxs-saif.h1
-rw-r--r--sound/soc/mxs/mxs-sgtl5000.c50
-rw-r--r--sound/soc/omap/Kconfig8
-rw-r--r--sound/soc/omap/Makefile4
-rw-r--r--sound/soc/omap/mcbsp.c115
-rw-r--r--sound/soc/omap/mcbsp.h8
-rw-r--r--sound/soc/omap/omap-abe-twl6040.c68
-rw-r--r--sound/soc/omap/omap-dmic.c8
-rw-r--r--sound/soc/omap/omap-hdmi-card.c87
-rw-r--r--sound/soc/omap/omap-hdmi.c238
-rw-r--r--sound/soc/omap/omap-hdmi.h4
-rw-r--r--sound/soc/omap/omap-mcbsp.c45
-rw-r--r--sound/soc/omap/omap-mcpdm.c8
-rw-r--r--sound/soc/omap/omap4-hdmi-card.c121
-rw-r--r--sound/soc/pxa/pxa-ssp.c66
-rw-r--r--sound/soc/pxa/pxa2xx-i2s.c4
-rw-r--r--sound/soc/samsung/littlemill.c102
-rw-r--r--sound/soc/samsung/lowland.c75
-rw-r--r--sound/soc/samsung/speyside.c33
-rw-r--r--sound/soc/sh/Kconfig24
-rw-r--r--sound/soc/sh/Makefile6
-rw-r--r--sound/soc/sh/fsi-ak4642.c108
-rw-r--r--sound/soc/sh/fsi-da7210.c81
-rw-r--r--sound/soc/sh/fsi-hdmi.c118
-rw-r--r--sound/soc/sh/fsi.c247
-rw-r--r--sound/soc/soc-core.c690
-rw-r--r--sound/soc/soc-dapm.c567
-rw-r--r--sound/soc/soc-jack.c5
-rw-r--r--sound/soc/soc-pcm.c1724
-rw-r--r--sound/soc/tegra/Kconfig68
-rw-r--r--sound/soc/tegra/Makefile20
-rw-r--r--sound/soc/tegra/tegra20_das.c233
-rw-r--r--sound/soc/tegra/tegra20_das.h134
-rw-r--r--sound/soc/tegra/tegra20_i2s.c494
-rw-r--r--sound/soc/tegra/tegra20_i2s.h164
-rw-r--r--sound/soc/tegra/tegra20_spdif.c404
-rw-r--r--sound/soc/tegra/tegra20_spdif.h471
-rw-r--r--sound/soc/tegra/tegra30_ahub.c632
-rw-r--r--sound/soc/tegra/tegra30_ahub.h483
-rw-r--r--sound/soc/tegra/tegra30_i2s.c536
-rw-r--r--sound/soc/tegra/tegra30_i2s.h242
-rw-r--r--sound/soc/tegra/tegra_alc5632.c48
-rw-r--r--sound/soc/tegra/tegra_asoc_utils.c37
-rw-r--r--sound/soc/tegra/tegra_asoc_utils.h9
-rw-r--r--sound/soc/tegra/tegra_das.c261
-rw-r--r--sound/soc/tegra/tegra_das.h135
-rw-r--r--sound/soc/tegra/tegra_i2s.c459
-rw-r--r--sound/soc/tegra/tegra_i2s.h166
-rw-r--r--sound/soc/tegra/tegra_pcm.c28
-rw-r--r--sound/soc/tegra/tegra_pcm.h5
-rw-r--r--sound/soc/tegra/tegra_spdif.c364
-rw-r--r--sound/soc/tegra/tegra_spdif.h473
-rw-r--r--sound/soc/tegra/tegra_wm8753.c224
-rw-r--r--sound/soc/tegra/tegra_wm8903.c42
-rw-r--r--sound/soc/tegra/trimslice.c41
-rw-r--r--sound/soc/ux500/Kconfig14
-rw-r--r--sound/soc/ux500/Makefile4
-rw-r--r--sound/soc/ux500/ux500_msp_dai.c843
-rw-r--r--sound/soc/ux500/ux500_msp_dai.h79
-rw-r--r--sound/soc/ux500/ux500_msp_i2s.c742
-rw-r--r--sound/soc/ux500/ux500_msp_i2s.h553
175 files changed, 18016 insertions, 5249 deletions
diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig
index 91c985599d3..40b2ad1bb1c 100644
--- a/sound/soc/Kconfig
+++ b/sound/soc/Kconfig
@@ -35,7 +35,6 @@ source "sound/soc/blackfin/Kconfig"
source "sound/soc/davinci/Kconfig"
source "sound/soc/ep93xx/Kconfig"
source "sound/soc/fsl/Kconfig"
-source "sound/soc/imx/Kconfig"
source "sound/soc/jz4740/Kconfig"
source "sound/soc/nuc900/Kconfig"
source "sound/soc/omap/Kconfig"
@@ -48,9 +47,13 @@ source "sound/soc/s6000/Kconfig"
source "sound/soc/sh/Kconfig"
source "sound/soc/tegra/Kconfig"
source "sound/soc/txx9/Kconfig"
+source "sound/soc/ux500/Kconfig"
# Supported codecs
source "sound/soc/codecs/Kconfig"
+# generic frame-work
+source "sound/soc/generic/Kconfig"
+
endif # SND_SOC
diff --git a/sound/soc/Makefile b/sound/soc/Makefile
index 2feaf376e94..70990f4017f 100644
--- a/sound/soc/Makefile
+++ b/sound/soc/Makefile
@@ -6,13 +6,13 @@ obj-$(CONFIG_SND_SOC_DMAENGINE_PCM) += snd-soc-dmaengine-pcm.o
obj-$(CONFIG_SND_SOC) += snd-soc-core.o
obj-$(CONFIG_SND_SOC) += codecs/
+obj-$(CONFIG_SND_SOC) += generic/
obj-$(CONFIG_SND_SOC) += atmel/
obj-$(CONFIG_SND_SOC) += au1x/
obj-$(CONFIG_SND_SOC) += blackfin/
obj-$(CONFIG_SND_SOC) += davinci/
obj-$(CONFIG_SND_SOC) += ep93xx/
obj-$(CONFIG_SND_SOC) += fsl/
-obj-$(CONFIG_SND_SOC) += imx/
obj-$(CONFIG_SND_SOC) += jz4740/
obj-$(CONFIG_SND_SOC) += mid-x86/
obj-$(CONFIG_SND_SOC) += mxs/
@@ -25,3 +25,4 @@ obj-$(CONFIG_SND_SOC) += s6000/
obj-$(CONFIG_SND_SOC) += sh/
obj-$(CONFIG_SND_SOC) += tegra/
obj-$(CONFIG_SND_SOC) += txx9/
+obj-$(CONFIG_SND_SOC) += ux500/
diff --git a/sound/soc/blackfin/bf5xx-ssm2602.c b/sound/soc/blackfin/bf5xx-ssm2602.c
index b39ad356b92..7dbeef1099b 100644
--- a/sound/soc/blackfin/bf5xx-ssm2602.c
+++ b/sound/soc/blackfin/bf5xx-ssm2602.c
@@ -44,16 +44,8 @@
static struct snd_soc_card bf5xx_ssm2602;
-static int bf5xx_ssm2602_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
+static int bf5xx_ssm2602_dai_init(struct snd_soc_pcm_runtime *rtd)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
- unsigned int clk = 0;
- int ret = 0;
-
- pr_debug("%s rate %d format %x\n", __func__, params_rate(params),
- params_format(params));
/*
* If you are using a crystal source which frequency is not 12MHz
* then modify the below case statement with frequency of the crystal.
@@ -61,31 +53,10 @@ static int bf5xx_ssm2602_hw_params(struct snd_pcm_substream *substream,
* If you are using the SPORT to generate clocking then this is
* where to do it.
*/
-
- switch (params_rate(params)) {
- case 8000:
- case 16000:
- case 48000:
- case 96000:
- case 11025:
- case 22050:
- case 44100:
- clk = 12000000;
- break;
- }
-
- ret = snd_soc_dai_set_sysclk(codec_dai, SSM2602_SYSCLK, clk,
+ return snd_soc_dai_set_sysclk(rtd->codec_dai, SSM2602_SYSCLK, 12000000,
SND_SOC_CLOCK_IN);
- if (ret < 0)
- return ret;
-
- return 0;
}
-static struct snd_soc_ops bf5xx_ssm2602_ops = {
- .hw_params = bf5xx_ssm2602_hw_params,
-};
-
/* CODEC is master for BCLK and LRC in this configuration. */
#define BF5XX_SSM2602_DAIFMT (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | \
SND_SOC_DAIFMT_CBM_CFM)
@@ -98,7 +69,7 @@ static struct snd_soc_dai_link bf5xx_ssm2602_dai[] = {
.codec_dai_name = "ssm2602-hifi",
.platform_name = "bfin-i2s-pcm-audio",
.codec_name = "ssm2602.0-001b",
- .ops = &bf5xx_ssm2602_ops,
+ .init = bf5xx_ssm2602_dai_init,
.dai_fmt = BF5XX_SSM2602_DAIFMT,
},
{
@@ -108,7 +79,7 @@ static struct snd_soc_dai_link bf5xx_ssm2602_dai[] = {
.codec_dai_name = "ssm2602-hifi",
.platform_name = "bfin-i2s-pcm-audio",
.codec_name = "ssm2602.0-001b",
- .ops = &bf5xx_ssm2602_ops,
+ .init = bf5xx_ssm2602_dai_init,
.dai_fmt = BF5XX_SSM2602_DAIFMT,
},
};
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 59d8efaa17e..1e1613a438d 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -29,6 +29,7 @@ config SND_SOC_ALL_CODECS
select SND_SOC_ALC5632 if I2C
select SND_SOC_CQ0093VC if MFD_DAVINCI_VOICECODEC
select SND_SOC_CS42L51 if I2C
+ select SND_SOC_CS42L52 if I2C
select SND_SOC_CS42L73 if I2C
select SND_SOC_CS4270 if I2C
select SND_SOC_CS4271 if SND_SOC_I2C_AND_SPI
@@ -37,11 +38,15 @@ config SND_SOC_ALL_CODECS
select SND_SOC_DFBMCS320
select SND_SOC_JZ4740_CODEC
select SND_SOC_LM4857 if I2C
+ select SND_SOC_LM49453 if I2C
select SND_SOC_MAX98088 if I2C
select SND_SOC_MAX98095 if I2C
select SND_SOC_MAX9850 if I2C
select SND_SOC_MAX9768 if I2C
select SND_SOC_MAX9877 if I2C
+ select SND_SOC_MC13783 if MFD_MC13XXX
+ select SND_SOC_ML26124 if I2C
+ select SND_SOC_OMAP_HDMI_CODEC if OMAP4_DSS_HDMI
select SND_SOC_PCM3008
select SND_SOC_RT5631 if I2C
select SND_SOC_SGTL5000 if I2C
@@ -181,6 +186,9 @@ config SND_SOC_CQ0093VC
config SND_SOC_CS42L51
tristate
+config SND_SOC_CS42L52
+ tristate
+
config SND_SOC_CS42L73
tristate
@@ -217,6 +225,9 @@ config SND_SOC_DFBMCS320
config SND_SOC_DMIC
tristate
+config SND_SOC_LM49453
+ tristate
+
config SND_SOC_MAX98088
tristate
@@ -226,6 +237,9 @@ config SND_SOC_MAX98095
config SND_SOC_MAX9850
tristate
+config SND_SOC_OMAP_HDMI_CODEC
+ tristate
+
config SND_SOC_PCM3008
tristate
@@ -435,5 +449,11 @@ config SND_SOC_MAX9768
config SND_SOC_MAX9877
tristate
+config SND_SOC_MC13783
+ tristate
+
+config SND_SOC_ML26124
+ tristate
+
config SND_SOC_TPA6130A2
tristate
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index 6662eb0cdcc..fc27fec3948 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -15,6 +15,7 @@ snd-soc-ak4642-objs := ak4642.o
snd-soc-ak4671-objs := ak4671.o
snd-soc-cq93vc-objs := cq93vc.o
snd-soc-cs42l51-objs := cs42l51.o
+snd-soc-cs42l52-objs := cs42l52.o
snd-soc-cs42l73-objs := cs42l73.o
snd-soc-cs4270-objs := cs4270.o
snd-soc-cs4271-objs := cs4271.o
@@ -25,10 +26,14 @@ snd-soc-dmic-objs := dmic.o
snd-soc-jz4740-codec-objs := jz4740.o
snd-soc-l3-objs := l3.o
snd-soc-lm4857-objs := lm4857.o
+snd-soc-lm49453-objs := lm49453.o
snd-soc-max9768-objs := max9768.o
snd-soc-max98088-objs := max98088.o
snd-soc-max98095-objs := max98095.o
snd-soc-max9850-objs := max9850.o
+snd-soc-mc13783-objs := mc13783.o
+snd-soc-ml26124-objs := ml26124.o
+snd-soc-omap-hdmi-codec-objs := omap-hdmi.o
snd-soc-pcm3008-objs := pcm3008.o
snd-soc-rt5631-objs := rt5631.o
snd-soc-sgtl5000-objs := sgtl5000.o
@@ -121,6 +126,7 @@ obj-$(CONFIG_SND_SOC_ALC5623) += snd-soc-alc5623.o
obj-$(CONFIG_SND_SOC_ALC5632) += snd-soc-alc5632.o
obj-$(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
obj-$(CONFIG_SND_SOC_CS42L73) += snd-soc-cs42l73.o
obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o
obj-$(CONFIG_SND_SOC_CS4271) += snd-soc-cs4271.o
@@ -128,13 +134,17 @@ obj-$(CONFIG_SND_SOC_CX20442) += snd-soc-cx20442.o
obj-$(CONFIG_SND_SOC_DA7210) += snd-soc-da7210.o
obj-$(CONFIG_SND_SOC_DFBMCS320) += snd-soc-dfbmcs320.o
obj-$(CONFIG_SND_SOC_DMIC) += snd-soc-dmic.o
+obj-$(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
-obj-$(CONFIG_SND_SOC_JZ4740_CODEC) += snd-soc-jz4740-codec.o
+obj-$(CONFIG_SND_SOC_LM49453) += snd-soc-lm49453.o
obj-$(CONFIG_SND_SOC_MAX9768) += snd-soc-max9768.o
obj-$(CONFIG_SND_SOC_MAX98088) += snd-soc-max98088.o
obj-$(CONFIG_SND_SOC_MAX98095) += snd-soc-max98095.o
obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o
+obj-$(CONFIG_SND_SOC_MC13783) += snd-soc-mc13783.o
+obj-$(CONFIG_SND_SOC_ML26124) += snd-soc-ml26124.o
+obj-$(CONFIG_SND_SOC_OMAP_HDMI_CODEC) += snd-soc-omap-hdmi-codec.o
obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o
obj-$(CONFIG_SND_SOC_RT5631) += snd-soc-rt5631.o
obj-$(CONFIG_SND_SOC_SGTL5000) += snd-soc-sgtl5000.o
diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c
index 1bbad4c16d2..2023c749f23 100644
--- a/sound/soc/codecs/ac97.c
+++ b/sound/soc/codecs/ac97.c
@@ -26,13 +26,11 @@
static int ac97_prepare(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
- struct snd_pcm_runtime *runtime = substream->runtime;
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_codec *codec = dai->codec;
int reg = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ?
AC97_PCM_FRONT_DAC_RATE : AC97_PCM_LR_ADC_RATE;
- return snd_ac97_set_rate(codec->ac97, reg, runtime->rate);
+ return snd_ac97_set_rate(codec->ac97, reg, substream->runtime->rate);
}
#define STD_AC97_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
diff --git a/sound/soc/codecs/ad1836.c b/sound/soc/codecs/ad1836.c
index 12e3b411855..c67b50d8b31 100644
--- a/sound/soc/codecs/ad1836.c
+++ b/sound/soc/codecs/ad1836.c
@@ -162,9 +162,7 @@ static int ad1836_hw_params(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
int word_len = 0;
-
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_codec *codec = dai->codec;
/* bit size */
switch (params_format(params)) {
diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c
index a4a6bef2c0b..13e62be4f99 100644
--- a/sound/soc/codecs/ad193x.c
+++ b/sound/soc/codecs/ad193x.c
@@ -245,9 +245,7 @@ static int ad193x_hw_params(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
int word_len = 0, master_rate = 0;
-
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_codec *codec = dai->codec;
struct ad193x_priv *ad193x = snd_soc_codec_get_drvdata(codec);
/* bit size */
diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c
index 78e9ce48bb9..3d50fc8646b 100644
--- a/sound/soc/codecs/adau1701.c
+++ b/sound/soc/codecs/adau1701.c
@@ -258,8 +258,7 @@ static int adau1701_set_playback_pcm_format(struct snd_soc_codec *codec,
static int adau1701_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;
+ struct snd_soc_codec *codec = dai->codec;
snd_pcm_format_t format;
unsigned int val;
diff --git a/sound/soc/codecs/ak4104.c b/sound/soc/codecs/ak4104.c
index ceb96ecf558..31d4483245d 100644
--- a/sound/soc/codecs/ak4104.c
+++ b/sound/soc/codecs/ak4104.c
@@ -88,8 +88,7 @@ static int ak4104_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;
+ struct snd_soc_codec *codec = dai->codec;
int val = 0;
/* set the IEC958 bits: consumer mode, no copyright bit */
diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c
index 838ae8b22b5..618fdc30f73 100644
--- a/sound/soc/codecs/ak4535.c
+++ b/sound/soc/codecs/ak4535.c
@@ -262,8 +262,7 @@ static int ak4535_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;
+ struct snd_soc_codec *codec = dai->codec;
struct ak4535_priv *ak4535 = snd_soc_codec_get_drvdata(codec);
u8 mode2 = snd_soc_read(codec, AK4535_MODE2) & ~(0x3 << 5);
int rate = params_rate(params), fs = 256;
diff --git a/sound/soc/codecs/ak4641.c b/sound/soc/codecs/ak4641.c
index c4d165a4bdd..543a12f471b 100644
--- a/sound/soc/codecs/ak4641.c
+++ b/sound/soc/codecs/ak4641.c
@@ -296,8 +296,7 @@ static int ak4641_i2s_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;
+ struct snd_soc_codec *codec = dai->codec;
struct ak4641_priv *ak4641 = snd_soc_codec_get_drvdata(codec);
int rate = params_rate(params), fs = 256;
u8 mode2;
@@ -517,67 +516,24 @@ static int ak4641_resume(struct snd_soc_codec *codec)
static int ak4641_probe(struct snd_soc_codec *codec)
{
- struct ak4641_platform_data *pdata = codec->dev->platform_data;
int ret;
-
- if (pdata) {
- if (gpio_is_valid(pdata->gpio_power)) {
- ret = gpio_request_one(pdata->gpio_power,
- GPIOF_OUT_INIT_LOW, "ak4641 power");
- if (ret)
- goto err_out;
- }
- if (gpio_is_valid(pdata->gpio_npdn)) {
- ret = gpio_request_one(pdata->gpio_npdn,
- GPIOF_OUT_INIT_LOW, "ak4641 npdn");
- if (ret)
- goto err_gpio;
-
- udelay(1); /* > 150 ns */
- gpio_set_value(pdata->gpio_npdn, 1);
- }
- }
-
ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_I2C);
if (ret != 0) {
dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
- goto err_register;
+ return ret;
}
/* power on device */
ak4641_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
return 0;
-
-err_register:
- if (pdata) {
- if (gpio_is_valid(pdata->gpio_power))
- gpio_set_value(pdata->gpio_power, 0);
- if (gpio_is_valid(pdata->gpio_npdn))
- gpio_free(pdata->gpio_npdn);
- }
-err_gpio:
- if (pdata && gpio_is_valid(pdata->gpio_power))
- gpio_free(pdata->gpio_power);
-err_out:
- return ret;
}
static int ak4641_remove(struct snd_soc_codec *codec)
{
- struct ak4641_platform_data *pdata = codec->dev->platform_data;
-
ak4641_set_bias_level(codec, SND_SOC_BIAS_OFF);
- if (pdata) {
- if (gpio_is_valid(pdata->gpio_power)) {
- gpio_set_value(pdata->gpio_power, 0);
- gpio_free(pdata->gpio_power);
- }
- if (gpio_is_valid(pdata->gpio_npdn))
- gpio_free(pdata->gpio_npdn);
- }
return 0;
}
@@ -604,6 +560,7 @@ static struct snd_soc_codec_driver soc_codec_dev_ak4641 = {
static int __devinit ak4641_i2c_probe(struct i2c_client *i2c,
const struct i2c_device_id *id)
{
+ struct ak4641_platform_data *pdata = i2c->dev.platform_data;
struct ak4641_priv *ak4641;
int ret;
@@ -612,16 +569,62 @@ static int __devinit ak4641_i2c_probe(struct i2c_client *i2c,
if (!ak4641)
return -ENOMEM;
+ if (pdata) {
+ if (gpio_is_valid(pdata->gpio_power)) {
+ ret = gpio_request_one(pdata->gpio_power,
+ GPIOF_OUT_INIT_LOW, "ak4641 power");
+ if (ret)
+ goto err_out;
+ }
+ if (gpio_is_valid(pdata->gpio_npdn)) {
+ ret = gpio_request_one(pdata->gpio_npdn,
+ GPIOF_OUT_INIT_LOW, "ak4641 npdn");
+ if (ret)
+ goto err_gpio;
+
+ udelay(1); /* > 150 ns */
+ gpio_set_value(pdata->gpio_npdn, 1);
+ }
+ }
+
i2c_set_clientdata(i2c, ak4641);
ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_ak4641,
ak4641_dai, ARRAY_SIZE(ak4641_dai));
+ if (ret != 0)
+ goto err_gpio2;
+
+ return 0;
+
+err_gpio2:
+ if (pdata) {
+ if (gpio_is_valid(pdata->gpio_power))
+ gpio_set_value(pdata->gpio_power, 0);
+ if (gpio_is_valid(pdata->gpio_npdn))
+ gpio_free(pdata->gpio_npdn);
+ }
+err_gpio:
+ if (pdata && gpio_is_valid(pdata->gpio_power))
+ gpio_free(pdata->gpio_power);
+err_out:
return ret;
}
static int __devexit ak4641_i2c_remove(struct i2c_client *i2c)
{
+ struct ak4641_platform_data *pdata = i2c->dev.platform_data;
+
snd_soc_unregister_codec(&i2c->dev);
+
+ if (pdata) {
+ if (gpio_is_valid(pdata->gpio_power)) {
+ gpio_set_value(pdata->gpio_power, 0);
+ gpio_free(pdata->gpio_power);
+ }
+ if (gpio_is_valid(pdata->gpio_npdn))
+ gpio_free(pdata->gpio_npdn);
+ }
+
return 0;
}
@@ -641,23 +644,7 @@ static struct i2c_driver ak4641_i2c_driver = {
.id_table = ak4641_i2c_id,
};
-static int __init ak4641_modinit(void)
-{
- int ret;
-
- ret = i2c_add_driver(&ak4641_i2c_driver);
- if (ret != 0)
- pr_err("Failed to register AK4641 I2C driver: %d\n", ret);
-
- return ret;
-}
-module_init(ak4641_modinit);
-
-static void __exit ak4641_exit(void)
-{
- i2c_del_driver(&ak4641_i2c_driver);
-}
-module_exit(ak4641_exit);
+module_i2c_driver(ak4641_i2c_driver);
MODULE_DESCRIPTION("SoC AK4641 driver");
MODULE_AUTHOR("Harald Welte <laforge@gnufiish.org>");
diff --git a/sound/soc/codecs/alc5623.c b/sound/soc/codecs/alc5623.c
index d47b62ddb21..1960478ce6b 100644
--- a/sound/soc/codecs/alc5623.c
+++ b/sound/soc/codecs/alc5623.c
@@ -705,8 +705,7 @@ static int alc5623_set_dai_fmt(struct snd_soc_dai *codec_dai,
static int alc5623_pcm_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;
+ struct snd_soc_codec *codec = dai->codec;
struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec);
int coeff, rate;
u16 iface;
@@ -1084,25 +1083,7 @@ static struct i2c_driver alc5623_i2c_driver = {
.id_table = alc5623_i2c_table,
};
-static int __init alc5623_modinit(void)
-{
- int ret;
-
- ret = i2c_add_driver(&alc5623_i2c_driver);
- if (ret != 0) {
- printk(KERN_ERR "%s: can't add i2c driver", __func__);
- return ret;
- }
-
- return ret;
-}
-module_init(alc5623_modinit);
-
-static void __exit alc5623_modexit(void)
-{
- i2c_del_driver(&alc5623_i2c_driver);
-}
-module_exit(alc5623_modexit);
+module_i2c_driver(alc5623_i2c_driver);
MODULE_DESCRIPTION("ASoC alc5621/2/3 driver");
MODULE_AUTHOR("Arnaud Patard <arnaud.patard@rtp-net.org>");
diff --git a/sound/soc/codecs/alc5632.c b/sound/soc/codecs/alc5632.c
index e2111e0ccad..7dd02420b36 100644
--- a/sound/soc/codecs/alc5632.c
+++ b/sound/soc/codecs/alc5632.c
@@ -861,8 +861,7 @@ static int alc5632_set_dai_fmt(struct snd_soc_dai *codec_dai,
static int alc5632_pcm_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;
+ struct snd_soc_codec *codec = dai->codec;
int coeff, rate;
u16 iface;
@@ -1131,7 +1130,7 @@ static __devinit int alc5632_i2c_probe(struct i2c_client *client,
i2c_set_clientdata(client, alc5632);
- alc5632->regmap = regmap_init_i2c(client, &alc5632_regmap);
+ alc5632->regmap = devm_regmap_init_i2c(client, &alc5632_regmap);
if (IS_ERR(alc5632->regmap)) {
ret = PTR_ERR(alc5632->regmap);
dev_err(&client->dev, "regmap_init() failed: %d\n", ret);
@@ -1143,7 +1142,6 @@ static __devinit int alc5632_i2c_probe(struct i2c_client *client,
if (ret1 != 0 || ret2 != 0) {
dev_err(&client->dev,
"Failed to read chip ID: ret1=%d, ret2=%d\n", ret1, ret2);
- regmap_exit(alc5632->regmap);
return -EIO;
}
@@ -1152,14 +1150,12 @@ static __devinit int alc5632_i2c_probe(struct i2c_client *client,
if ((vid1 != 0x10EC) || (vid2 != id->driver_data)) {
dev_err(&client->dev,
"Device is not a ALC5632: VID1=0x%x, VID2=0x%x\n", vid1, vid2);
- regmap_exit(alc5632->regmap);
return -EINVAL;
}
ret = alc5632_reset(alc5632->regmap);
if (ret < 0) {
dev_err(&client->dev, "Failed to issue reset\n");
- regmap_exit(alc5632->regmap);
return ret;
}
@@ -1177,7 +1173,6 @@ static __devinit int alc5632_i2c_probe(struct i2c_client *client,
if (ret < 0) {
dev_err(&client->dev, "Failed to register codec: %d\n", ret);
- regmap_exit(alc5632->regmap);
return ret;
}
@@ -1186,9 +1181,7 @@ static __devinit int alc5632_i2c_probe(struct i2c_client *client,
static __devexit int alc5632_i2c_remove(struct i2c_client *client)
{
- struct alc5632_priv *alc5632 = i2c_get_clientdata(client);
snd_soc_unregister_codec(&client->dev);
- regmap_exit(alc5632->regmap);
return 0;
}
@@ -1209,25 +1202,7 @@ static struct i2c_driver alc5632_i2c_driver = {
.id_table = alc5632_i2c_table,
};
-static int __init alc5632_modinit(void)
-{
- int ret;
-
- ret = i2c_add_driver(&alc5632_i2c_driver);
- if (ret != 0) {
- printk(KERN_ERR "%s: can't add i2c driver", __func__);
- return ret;
- }
-
- return ret;
-}
-module_init(alc5632_modinit);
-
-static void __exit alc5632_modexit(void)
-{
- i2c_del_driver(&alc5632_i2c_driver);
-}
-module_exit(alc5632_modexit);
+module_i2c_driver(alc5632_i2c_driver);
MODULE_DESCRIPTION("ASoC ALC5632 driver");
MODULE_AUTHOR("Leon Romanovsky <leon@leon.nu>");
diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c
index 1d672f52866..047917f0b8a 100644
--- a/sound/soc/codecs/cs4270.c
+++ b/sound/soc/codecs/cs4270.c
@@ -307,8 +307,7 @@ static int cs4270_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;
+ struct snd_soc_codec *codec = dai->codec;
struct cs4270_private *cs4270 = snd_soc_codec_get_drvdata(codec);
int ret;
unsigned int i;
@@ -600,10 +599,12 @@ static int cs4270_soc_suspend(struct snd_soc_codec *codec)
static int cs4270_soc_resume(struct snd_soc_codec *codec)
{
struct cs4270_private *cs4270 = snd_soc_codec_get_drvdata(codec);
- int reg;
+ int reg, ret;
- regulator_bulk_enable(ARRAY_SIZE(cs4270->supplies),
- cs4270->supplies);
+ ret = regulator_bulk_enable(ARRAY_SIZE(cs4270->supplies),
+ cs4270->supplies);
+ if (ret != 0)
+ return ret;
/* In case the device was put to hard reset during sleep, we need to
* wait 500ns here before any I2C communication. */
diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c
index bf7141280a7..9eb01d7d58a 100644
--- a/sound/soc/codecs/cs4271.c
+++ b/sound/soc/codecs/cs4271.c
@@ -318,8 +318,7 @@ static int cs4271_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;
+ struct snd_soc_codec *codec = dai->codec;
struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec);
int i, ret;
unsigned int ratio, val;
diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c
index a8bf588e874..091d0193f50 100644
--- a/sound/soc/codecs/cs42l51.c
+++ b/sound/soc/codecs/cs42l51.c
@@ -141,15 +141,15 @@ static const struct soc_enum cs42l51_chan_mix =
static const struct snd_kcontrol_new cs42l51_snd_controls[] = {
SOC_DOUBLE_R_SX_TLV("PCM Playback Volume",
CS42L51_PCMA_VOL, CS42L51_PCMB_VOL,
- 7, 0xffffff99, 0x18, adc_pcm_tlv),
+ 6, 0x19, 0x7F, adc_pcm_tlv),
SOC_DOUBLE_R("PCM Playback Switch",
CS42L51_PCMA_VOL, CS42L51_PCMB_VOL, 7, 1, 1),
SOC_DOUBLE_R_SX_TLV("Analog Playback Volume",
CS42L51_AOUTA_VOL, CS42L51_AOUTB_VOL,
- 8, 0xffffff19, 0x18, aout_tlv),
+ 0, 0x34, 0xE4, aout_tlv),
SOC_DOUBLE_R_SX_TLV("ADC Mixer Volume",
CS42L51_ADCA_VOL, CS42L51_ADCB_VOL,
- 7, 0xffffff99, 0x18, adc_pcm_tlv),
+ 6, 0x19, 0x7F, adc_pcm_tlv),
SOC_DOUBLE_R("ADC Mixer Switch",
CS42L51_ADCA_VOL, CS42L51_ADCB_VOL, 7, 1, 1),
SOC_SINGLE("Playback Deemphasis Switch", CS42L51_DAC_CTL, 3, 1, 0),
@@ -356,8 +356,7 @@ static int cs42l51_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;
+ struct snd_soc_codec *codec = dai->codec;
struct cs42l51_private *cs42l51 = snd_soc_codec_get_drvdata(codec);
int ret;
unsigned int i;
diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c
new file mode 100644
index 00000000000..a7109413aef
--- /dev/null
+++ b/sound/soc/codecs/cs42l52.c
@@ -0,0 +1,1295 @@
+/*
+ * cs42l52.c -- CS42L52 ALSA SoC audio driver
+ *
+ * Copyright 2012 CirrusLogic, Inc.
+ *
+ * Author: Georgi Vlaev <joe@nucleusys.com>
+ * Author: Brian Austin <brian.austin@cirrus.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/version.h>
+#include <linux/kernel.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/i2c.h>
+#include <linux/input.h>
+#include <linux/regmap.h>
+#include <linux/slab.h>
+#include <linux/workqueue.h>
+#include <linux/platform_device.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/initval.h>
+#include <sound/tlv.h>
+#include <sound/cs42l52.h>
+#include "cs42l52.h"
+
+struct sp_config {
+ u8 spc, format, spfs;
+ u32 srate;
+};
+
+struct cs42l52_private {
+ struct regmap *regmap;
+ struct snd_soc_codec *codec;
+ struct device *dev;
+ struct sp_config config;
+ struct cs42l52_platform_data pdata;
+ u32 sysclk;
+ u8 mclksel;
+ u32 mclk;
+ u8 flags;
+#if defined(CONFIG_INPUT) || defined(CONFIG_INPUT_MODULE)
+ struct input_dev *beep;
+ struct work_struct beep_work;
+ int beep_rate;
+#endif
+};
+
+static const struct reg_default cs42l52_reg_defaults[] = {
+ { CS42L52_PWRCTL1, 0x9F }, /* r02 PWRCTL 1 */
+ { CS42L52_PWRCTL2, 0x07 }, /* r03 PWRCTL 2 */
+ { CS42L52_PWRCTL3, 0xFF }, /* r04 PWRCTL 3 */
+ { CS42L52_CLK_CTL, 0xA0 }, /* r05 Clocking Ctl */
+ { CS42L52_IFACE_CTL1, 0x00 }, /* r06 Interface Ctl 1 */
+ { CS42L52_ADC_PGA_A, 0x80 }, /* r08 Input A Select */
+ { CS42L52_ADC_PGA_B, 0x80 }, /* r09 Input B Select */
+ { CS42L52_ANALOG_HPF_CTL, 0xA5 }, /* r0A Analog HPF Ctl */
+ { CS42L52_ADC_HPF_FREQ, 0x00 }, /* r0B ADC HPF Corner Freq */
+ { CS42L52_ADC_MISC_CTL, 0x00 }, /* r0C Misc. ADC Ctl */
+ { CS42L52_PB_CTL1, 0x60 }, /* r0D Playback Ctl 1 */
+ { CS42L52_MISC_CTL, 0x02 }, /* r0E Misc. Ctl */
+ { CS42L52_PB_CTL2, 0x00 }, /* r0F Playback Ctl 2 */
+ { CS42L52_MICA_CTL, 0x00 }, /* r10 MICA Amp Ctl */
+ { CS42L52_MICB_CTL, 0x00 }, /* r11 MICB Amp Ctl */
+ { CS42L52_PGAA_CTL, 0x00 }, /* r12 PGAA Vol, Misc. */
+ { CS42L52_PGAB_CTL, 0x00 }, /* r13 PGAB Vol, Misc. */
+ { CS42L52_PASSTHRUA_VOL, 0x00 }, /* r14 Bypass A Vol */
+ { CS42L52_PASSTHRUB_VOL, 0x00 }, /* r15 Bypass B Vol */
+ { CS42L52_ADCA_VOL, 0x00 }, /* r16 ADCA Volume */
+ { CS42L52_ADCB_VOL, 0x00 }, /* r17 ADCB Volume */
+ { CS42L52_ADCA_MIXER_VOL, 0x80 }, /* r18 ADCA Mixer Volume */
+ { CS42L52_ADCB_MIXER_VOL, 0x80 }, /* r19 ADCB Mixer Volume */
+ { CS42L52_PCMA_MIXER_VOL, 0x00 }, /* r1A PCMA Mixer Volume */
+ { CS42L52_PCMB_MIXER_VOL, 0x00 }, /* r1B PCMB Mixer Volume */
+ { CS42L52_BEEP_FREQ, 0x00 }, /* r1C Beep Freq on Time */
+ { CS42L52_BEEP_VOL, 0x00 }, /* r1D Beep Volume off Time */
+ { CS42L52_BEEP_TONE_CTL, 0x00 }, /* r1E Beep Tone Cfg. */
+ { CS42L52_TONE_CTL, 0x00 }, /* r1F Tone Ctl */
+ { CS42L52_MASTERA_VOL, 0x88 }, /* r20 Master A Volume */
+ { CS42L52_MASTERB_VOL, 0x00 }, /* r21 Master B Volume */
+ { CS42L52_HPA_VOL, 0x00 }, /* r22 Headphone A Volume */
+ { CS42L52_HPB_VOL, 0x00 }, /* r23 Headphone B Volume */
+ { CS42L52_SPKA_VOL, 0x00 }, /* r24 Speaker A Volume */
+ { CS42L52_SPKB_VOL, 0x00 }, /* r25 Speaker B Volume */
+ { CS42L52_ADC_PCM_MIXER, 0x00 }, /* r26 Channel Mixer and Swap */
+ { CS42L52_LIMITER_CTL1, 0x00 }, /* r27 Limit Ctl 1 Thresholds */
+ { CS42L52_LIMITER_CTL2, 0x7F }, /* r28 Limit Ctl 2 Release Rate */
+ { CS42L52_LIMITER_AT_RATE, 0xC0 }, /* r29 Limiter Attack Rate */
+ { CS42L52_ALC_CTL, 0x00 }, /* r2A ALC Ctl 1 Attack Rate */
+ { CS42L52_ALC_RATE, 0x3F }, /* r2B ALC Release Rate */
+ { CS42L52_ALC_THRESHOLD, 0x3f }, /* r2C ALC Thresholds */
+ { CS42L52_NOISE_GATE_CTL, 0x00 }, /* r2D Noise Gate Ctl */
+ { CS42L52_CLK_STATUS, 0x00 }, /* r2E Overflow and Clock Status */
+ { CS42L52_BATT_COMPEN, 0x00 }, /* r2F battery Compensation */
+ { CS42L52_BATT_LEVEL, 0x00 }, /* r30 VP Battery Level */
+ { CS42L52_SPK_STATUS, 0x00 }, /* r31 Speaker Status */
+ { CS42L52_TEM_CTL, 0x3B }, /* r32 Temp Ctl */
+ { CS42L52_THE_FOLDBACK, 0x00 }, /* r33 Foldback */
+};
+
+static bool cs42l52_readable_register(struct device *dev, unsigned int reg)
+{
+ switch (reg) {
+ case CS42L52_CHIP:
+ case CS42L52_PWRCTL1:
+ case CS42L52_PWRCTL2:
+ case CS42L52_PWRCTL3:
+ case CS42L52_CLK_CTL:
+ case CS42L52_IFACE_CTL1:
+ case CS42L52_IFACE_CTL2:
+ case CS42L52_ADC_PGA_A:
+ case CS42L52_ADC_PGA_B:
+ case CS42L52_ANALOG_HPF_CTL:
+ case CS42L52_ADC_HPF_FREQ:
+ case CS42L52_ADC_MISC_CTL:
+ case CS42L52_PB_CTL1:
+ case CS42L52_MISC_CTL:
+ case CS42L52_PB_CTL2:
+ case CS42L52_MICA_CTL:
+ case CS42L52_MICB_CTL:
+ case CS42L52_PGAA_CTL:
+ case CS42L52_PGAB_CTL:
+ case CS42L52_PASSTHRUA_VOL:
+ case CS42L52_PASSTHRUB_VOL:
+ case CS42L52_ADCA_VOL:
+ case CS42L52_ADCB_VOL:
+ case CS42L52_ADCA_MIXER_VOL:
+ case CS42L52_ADCB_MIXER_VOL:
+ case CS42L52_PCMA_MIXER_VOL:
+ case CS42L52_PCMB_MIXER_VOL:
+ case CS42L52_BEEP_FREQ:
+ case CS42L52_BEEP_VOL:
+ case CS42L52_BEEP_TONE_CTL:
+ case CS42L52_TONE_CTL:
+ case CS42L52_MASTERA_VOL:
+ case CS42L52_MASTERB_VOL:
+ case CS42L52_HPA_VOL:
+ case CS42L52_HPB_VOL:
+ case CS42L52_SPKA_VOL:
+ case CS42L52_SPKB_VOL:
+ case CS42L52_ADC_PCM_MIXER:
+ case CS42L52_LIMITER_CTL1:
+ case CS42L52_LIMITER_CTL2:
+ case CS42L52_LIMITER_AT_RATE:
+ case CS42L52_ALC_CTL:
+ case CS42L52_ALC_RATE:
+ case CS42L52_ALC_THRESHOLD:
+ case CS42L52_NOISE_GATE_CTL:
+ case CS42L52_CLK_STATUS:
+ case CS42L52_BATT_COMPEN:
+ case CS42L52_BATT_LEVEL:
+ case CS42L52_SPK_STATUS:
+ case CS42L52_TEM_CTL:
+ case CS42L52_THE_FOLDBACK:
+ case CS42L52_CHARGE_PUMP:
+ return true;
+ default:
+ return false;
+ }
+}
+
+static bool cs42l52_volatile_register(struct device *dev, unsigned int reg)
+{
+ switch (reg) {
+ case CS42L52_IFACE_CTL2:
+ case CS42L52_CLK_STATUS:
+ case CS42L52_BATT_LEVEL:
+ case CS42L52_SPK_STATUS:
+ case CS42L52_CHARGE_PUMP:
+ return 1;
+ default:
+ return 0;
+ }
+}
+
+static DECLARE_TLV_DB_SCALE(hl_tlv, -10200, 50, 0);
+
+static DECLARE_TLV_DB_SCALE(hpd_tlv, -9600, 50, 1);
+
+static DECLARE_TLV_DB_SCALE(ipd_tlv, -9600, 100, 0);
+
+static DECLARE_TLV_DB_SCALE(mic_tlv, 1600, 100, 0);
+
+static DECLARE_TLV_DB_SCALE(pga_tlv, -600, 50, 0);
+
+static const unsigned int limiter_tlv[] = {
+ TLV_DB_RANGE_HEAD(2),
+ 0, 2, TLV_DB_SCALE_ITEM(-3000, 600, 0),
+ 3, 7, TLV_DB_SCALE_ITEM(-1200, 300, 0),
+};
+
+static const char * const cs42l52_adca_text[] = {
+ "Input1A", "Input2A", "Input3A", "Input4A", "PGA Input Left"};
+
+static const char * const cs42l52_adcb_text[] = {
+ "Input1B", "Input2B", "Input3B", "Input4B", "PGA Input Right"};
+
+static const struct soc_enum adca_enum =
+ SOC_ENUM_SINGLE(CS42L52_ADC_PGA_A, 5,
+ ARRAY_SIZE(cs42l52_adca_text), cs42l52_adca_text);
+
+static const struct soc_enum adcb_enum =
+ SOC_ENUM_SINGLE(CS42L52_ADC_PGA_B, 5,
+ ARRAY_SIZE(cs42l52_adcb_text), cs42l52_adcb_text);
+
+static const struct snd_kcontrol_new adca_mux =
+ SOC_DAPM_ENUM("Left ADC Input Capture Mux", adca_enum);
+
+static const struct snd_kcontrol_new adcb_mux =
+ SOC_DAPM_ENUM("Right ADC Input Capture Mux", adcb_enum);
+
+static const char * const mic_bias_level_text[] = {
+ "0.5 +VA", "0.6 +VA", "0.7 +VA",
+ "0.8 +VA", "0.83 +VA", "0.91 +VA"
+};
+
+static const struct soc_enum mic_bias_level_enum =
+ SOC_ENUM_SINGLE(CS42L52_IFACE_CTL1, 0,
+ ARRAY_SIZE(mic_bias_level_text), mic_bias_level_text);
+
+static const char * const cs42l52_mic_text[] = { "Single", "Differential" };
+
+static const struct soc_enum mica_enum =
+ SOC_ENUM_SINGLE(CS42L52_MICA_CTL, 5,
+ ARRAY_SIZE(cs42l52_mic_text), cs42l52_mic_text);
+
+static const struct soc_enum micb_enum =
+ SOC_ENUM_SINGLE(CS42L52_MICB_CTL, 5,
+ ARRAY_SIZE(cs42l52_mic_text), cs42l52_mic_text);
+
+static const struct snd_kcontrol_new mica_mux =
+ SOC_DAPM_ENUM("Left Mic Input Capture Mux", mica_enum);
+
+static const struct snd_kcontrol_new micb_mux =
+ SOC_DAPM_ENUM("Right Mic Input Capture Mux", micb_enum);
+
+static const char * const digital_output_mux_text[] = {"ADC", "DSP"};
+
+static const struct soc_enum digital_output_mux_enum =
+ SOC_ENUM_SINGLE(CS42L52_ADC_MISC_CTL, 6,
+ ARRAY_SIZE(digital_output_mux_text),
+ digital_output_mux_text);
+
+static const struct snd_kcontrol_new digital_output_mux =
+ SOC_DAPM_ENUM("Digital Output Mux", digital_output_mux_enum);
+
+static const char * const hp_gain_num_text[] = {
+ "0.3959", "0.4571", "0.5111", "0.6047",
+ "0.7099", "0.8399", "1.000", "1.1430"
+};
+
+static const struct soc_enum hp_gain_enum =
+ SOC_ENUM_SINGLE(CS42L52_PB_CTL1, 4,
+ ARRAY_SIZE(hp_gain_num_text), hp_gain_num_text);
+
+static const char * const beep_pitch_text[] = {
+ "C4", "C5", "D5", "E5", "F5", "G5", "A5", "B5",
+ "C6", "D6", "E6", "F6", "G6", "A6", "B6", "C7"
+};
+
+static const struct soc_enum beep_pitch_enum =
+ SOC_ENUM_SINGLE(CS42L52_BEEP_FREQ, 4,
+ ARRAY_SIZE(beep_pitch_text), beep_pitch_text);
+
+static const char * const beep_ontime_text[] = {
+ "86 ms", "430 ms", "780 ms", "1.20 s", "1.50 s",
+ "1.80 s", "2.20 s", "2.50 s", "2.80 s", "3.20 s",
+ "3.50 s", "3.80 s", "4.20 s", "4.50 s", "4.80 s", "5.20 s"
+};
+
+static const struct soc_enum beep_ontime_enum =
+ SOC_ENUM_SINGLE(CS42L52_BEEP_FREQ, 0,
+ ARRAY_SIZE(beep_ontime_text), beep_ontime_text);
+
+static const char * const beep_offtime_text[] = {
+ "1.23 s", "2.58 s", "3.90 s", "5.20 s",
+ "6.60 s", "8.05 s", "9.35 s", "10.80 s"
+};
+
+static const struct soc_enum beep_offtime_enum =
+ SOC_ENUM_SINGLE(CS42L52_BEEP_VOL, 5,
+ ARRAY_SIZE(beep_offtime_text), beep_offtime_text);
+
+static const char * const beep_config_text[] = {
+ "Off", "Single", "Multiple", "Continuous"
+};
+
+static const struct soc_enum beep_config_enum =
+ SOC_ENUM_SINGLE(CS42L52_BEEP_TONE_CTL, 6,
+ ARRAY_SIZE(beep_config_text), beep_config_text);
+
+static const char * const beep_bass_text[] = {
+ "50 Hz", "100 Hz", "200 Hz", "250 Hz"
+};
+
+static const struct soc_enum beep_bass_enum =
+ SOC_ENUM_SINGLE(CS42L52_BEEP_TONE_CTL, 1,
+ ARRAY_SIZE(beep_bass_text), beep_bass_text);
+
+static const char * const beep_treble_text[] = {
+ "5 kHz", "7 kHz", "10 kHz", " 15 kHz"
+};
+
+static const struct soc_enum beep_treble_enum =
+ SOC_ENUM_SINGLE(CS42L52_BEEP_TONE_CTL, 3,
+ ARRAY_SIZE(beep_treble_text), beep_treble_text);
+
+static const char * const ng_threshold_text[] = {
+ "-34dB", "-37dB", "-40dB", "-43dB",
+ "-46dB", "-52dB", "-58dB", "-64dB"
+};
+
+static const struct soc_enum ng_threshold_enum =
+ SOC_ENUM_SINGLE(CS42L52_NOISE_GATE_CTL, 2,
+ ARRAY_SIZE(ng_threshold_text), ng_threshold_text);
+
+static const char * const cs42l52_ng_delay_text[] = {
+ "50ms", "100ms", "150ms", "200ms"};
+
+static const struct soc_enum ng_delay_enum =
+ SOC_ENUM_SINGLE(CS42L52_NOISE_GATE_CTL, 0,
+ ARRAY_SIZE(cs42l52_ng_delay_text), cs42l52_ng_delay_text);
+
+static const char * const cs42l52_ng_type_text[] = {
+ "Apply Specific", "Apply All"
+};
+
+static const struct soc_enum ng_type_enum =
+ SOC_ENUM_SINGLE(CS42L52_NOISE_GATE_CTL, 6,
+ ARRAY_SIZE(cs42l52_ng_type_text), cs42l52_ng_type_text);
+
+static const char * const left_swap_text[] = {
+ "Left", "LR 2", "Right"};
+
+static const char * const right_swap_text[] = {
+ "Right", "LR 2", "Left"};
+
+static const unsigned int swap_values[] = { 0, 1, 3 };
+
+static const struct soc_enum adca_swap_enum =
+ SOC_VALUE_ENUM_SINGLE(CS42L52_ADC_PCM_MIXER, 2, 1,
+ ARRAY_SIZE(left_swap_text),
+ left_swap_text,
+ swap_values);
+
+static const struct snd_kcontrol_new adca_mixer =
+ SOC_DAPM_ENUM("Route", adca_swap_enum);
+
+static const struct soc_enum pcma_swap_enum =
+ SOC_VALUE_ENUM_SINGLE(CS42L52_ADC_PCM_MIXER, 6, 1,
+ ARRAY_SIZE(left_swap_text),
+ left_swap_text,
+ swap_values);
+
+static const struct snd_kcontrol_new pcma_mixer =
+ SOC_DAPM_ENUM("Route", pcma_swap_enum);
+
+static const struct soc_enum adcb_swap_enum =
+ SOC_VALUE_ENUM_SINGLE(CS42L52_ADC_PCM_MIXER, 0, 1,
+ ARRAY_SIZE(right_swap_text),
+ right_swap_text,
+ swap_values);
+
+static const struct snd_kcontrol_new adcb_mixer =
+ SOC_DAPM_ENUM("Route", adcb_swap_enum);
+
+static const struct soc_enum pcmb_swap_enum =
+ SOC_VALUE_ENUM_SINGLE(CS42L52_ADC_PCM_MIXER, 4, 1,
+ ARRAY_SIZE(right_swap_text),
+ right_swap_text,
+ swap_values);
+
+static const struct snd_kcontrol_new pcmb_mixer =
+ SOC_DAPM_ENUM("Route", pcmb_swap_enum);
+
+
+static const struct snd_kcontrol_new passthrul_ctl =
+ SOC_DAPM_SINGLE("Switch", CS42L52_MISC_CTL, 6, 1, 0);
+
+static const struct snd_kcontrol_new passthrur_ctl =
+ SOC_DAPM_SINGLE("Switch", CS42L52_MISC_CTL, 7, 1, 0);
+
+static const struct snd_kcontrol_new spkl_ctl =
+ SOC_DAPM_SINGLE("Switch", CS42L52_PWRCTL3, 0, 1, 1);
+
+static const struct snd_kcontrol_new spkr_ctl =
+ SOC_DAPM_SINGLE("Switch", CS42L52_PWRCTL3, 2, 1, 1);
+
+static const struct snd_kcontrol_new hpl_ctl =
+ SOC_DAPM_SINGLE("Switch", CS42L52_PWRCTL3, 4, 1, 1);
+
+static const struct snd_kcontrol_new hpr_ctl =
+ SOC_DAPM_SINGLE("Switch", CS42L52_PWRCTL3, 6, 1, 1);
+
+static const struct snd_kcontrol_new cs42l52_snd_controls[] = {
+
+ SOC_DOUBLE_R_SX_TLV("Master Volume", CS42L52_MASTERA_VOL,
+ CS42L52_MASTERB_VOL, 0, 0x34, 0xE4, hl_tlv),
+
+ SOC_DOUBLE_R_SX_TLV("Headphone Volume", CS42L52_HPA_VOL,
+ CS42L52_HPB_VOL, 0, 0x34, 0xCC, hpd_tlv),
+
+ SOC_ENUM("Headphone Analog Gain", hp_gain_enum),
+
+ SOC_DOUBLE_R_SX_TLV("Speaker Volume", CS42L52_SPKA_VOL,
+ CS42L52_SPKB_VOL, 7, 0x1, 0xff, hl_tlv),
+
+ SOC_DOUBLE_R_SX_TLV("Bypass Volume", CS42L52_PASSTHRUA_VOL,
+ CS42L52_PASSTHRUB_VOL, 6, 0x18, 0x90, pga_tlv),
+
+ SOC_DOUBLE("Bypass Mute", CS42L52_MISC_CTL, 4, 5, 1, 0),
+
+ SOC_DOUBLE_R_TLV("MIC Gain Volume", CS42L52_MICA_CTL,
+ CS42L52_MICB_CTL, 0, 0x10, 0, mic_tlv),
+
+ SOC_ENUM("MIC Bias Level", mic_bias_level_enum),
+
+ SOC_DOUBLE_R_SX_TLV("ADC Volume", CS42L52_ADCA_VOL,
+ CS42L52_ADCB_VOL, 7, 0x80, 0xA0, ipd_tlv),
+ SOC_DOUBLE_R_SX_TLV("ADC Mixer Volume",
+ CS42L52_ADCA_MIXER_VOL, CS42L52_ADCB_MIXER_VOL,
+ 6, 0x7f, 0x19, ipd_tlv),
+
+ SOC_DOUBLE("ADC Switch", CS42L52_ADC_MISC_CTL, 0, 1, 1, 0),
+
+ SOC_DOUBLE_R("ADC Mixer Switch", CS42L52_ADCA_MIXER_VOL,
+ CS42L52_ADCB_MIXER_VOL, 7, 1, 1),
+
+ SOC_DOUBLE_R_SX_TLV("PGA Volume", CS42L52_PGAA_CTL,
+ CS42L52_PGAB_CTL, 0, 0x28, 0x30, pga_tlv),
+
+ SOC_DOUBLE_R_SX_TLV("PCM Mixer Volume",
+ CS42L52_PCMA_MIXER_VOL, CS42L52_PCMB_MIXER_VOL,
+ 6, 0x7f, 0x19, hl_tlv),
+ SOC_DOUBLE_R("PCM Mixer Switch",
+ CS42L52_PCMA_MIXER_VOL, CS42L52_PCMB_MIXER_VOL, 7, 1, 1),
+
+ SOC_ENUM("Beep Config", beep_config_enum),
+ SOC_ENUM("Beep Pitch", beep_pitch_enum),
+ SOC_ENUM("Beep on Time", beep_ontime_enum),
+ SOC_ENUM("Beep off Time", beep_offtime_enum),
+ SOC_SINGLE_TLV("Beep Volume", CS42L52_BEEP_VOL, 0, 0x1f, 0x07, hl_tlv),
+ SOC_SINGLE("Beep Mixer Switch", CS42L52_BEEP_TONE_CTL, 5, 1, 1),
+ SOC_ENUM("Beep Treble Corner Freq", beep_treble_enum),
+ SOC_ENUM("Beep Bass Corner Freq", beep_bass_enum),
+
+ SOC_SINGLE("Tone Control Switch", CS42L52_BEEP_TONE_CTL, 0, 1, 1),
+ SOC_SINGLE_TLV("Treble Gain Volume",
+ CS42L52_TONE_CTL, 4, 15, 1, hl_tlv),
+ SOC_SINGLE_TLV("Bass Gain Volume",
+ CS42L52_TONE_CTL, 0, 15, 1, hl_tlv),
+
+ /* Limiter */
+ SOC_SINGLE_TLV("Limiter Max Threshold Volume",
+ CS42L52_LIMITER_CTL1, 5, 7, 0, limiter_tlv),
+ SOC_SINGLE_TLV("Limiter Cushion Threshold Volume",
+ CS42L52_LIMITER_CTL1, 2, 7, 0, limiter_tlv),
+ SOC_SINGLE_TLV("Limiter Release Rate Volume",
+ CS42L52_LIMITER_CTL2, 0, 63, 0, limiter_tlv),
+ SOC_SINGLE_TLV("Limiter Attack Rate Volume",
+ CS42L52_LIMITER_AT_RATE, 0, 63, 0, limiter_tlv),
+
+ SOC_SINGLE("Limiter SR Switch", CS42L52_LIMITER_CTL1, 1, 1, 0),
+ SOC_SINGLE("Limiter ZC Switch", CS42L52_LIMITER_CTL1, 0, 1, 0),
+ SOC_SINGLE("Limiter Switch", CS42L52_LIMITER_CTL2, 7, 1, 0),
+
+ /* ALC */
+ SOC_SINGLE_TLV("ALC Attack Rate Volume", CS42L52_ALC_CTL,
+ 0, 63, 0, limiter_tlv),
+ SOC_SINGLE_TLV("ALC Release Rate Volume", CS42L52_ALC_RATE,
+ 0, 63, 0, limiter_tlv),
+ SOC_SINGLE_TLV("ALC Max Threshold Volume", CS42L52_ALC_THRESHOLD,
+ 5, 7, 0, limiter_tlv),
+ SOC_SINGLE_TLV("ALC Min Threshold Volume", CS42L52_ALC_THRESHOLD,
+ 2, 7, 0, limiter_tlv),
+
+ SOC_DOUBLE_R("ALC SR Capture Switch", CS42L52_PGAA_CTL,
+ CS42L52_PGAB_CTL, 7, 1, 1),
+ SOC_DOUBLE_R("ALC ZC Capture Switch", CS42L52_PGAA_CTL,
+ CS42L52_PGAB_CTL, 6, 1, 1),
+ SOC_DOUBLE("ALC Capture Switch", CS42L52_ALC_CTL, 6, 7, 1, 0),
+
+ /* Noise gate */
+ SOC_ENUM("NG Type Switch", ng_type_enum),
+ SOC_SINGLE("NG Enable Switch", CS42L52_NOISE_GATE_CTL, 6, 1, 0),
+ SOC_SINGLE("NG Boost Switch", CS42L52_NOISE_GATE_CTL, 5, 1, 1),
+ SOC_ENUM("NG Threshold", ng_threshold_enum),
+ SOC_ENUM("NG Delay", ng_delay_enum),
+
+ SOC_DOUBLE("HPF Switch", CS42L52_ANALOG_HPF_CTL, 5, 7, 1, 0),
+
+ SOC_DOUBLE("Analog SR Switch", CS42L52_ANALOG_HPF_CTL, 1, 3, 1, 1),
+ SOC_DOUBLE("Analog ZC Switch", CS42L52_ANALOG_HPF_CTL, 0, 2, 1, 1),
+ SOC_SINGLE("Digital SR Switch", CS42L52_MISC_CTL, 1, 1, 0),
+ SOC_SINGLE("Digital ZC Switch", CS42L52_MISC_CTL, 0, 1, 0),
+ SOC_SINGLE("Deemphasis Switch", CS42L52_MISC_CTL, 2, 1, 0),
+
+ SOC_SINGLE("Batt Compensation Switch", CS42L52_BATT_COMPEN, 7, 1, 0),
+ SOC_SINGLE("Batt VP Monitor Switch", CS42L52_BATT_COMPEN, 6, 1, 0),
+ SOC_SINGLE("Batt VP ref", CS42L52_BATT_COMPEN, 0, 0x0f, 0),
+
+ SOC_SINGLE("PGA AIN1L Switch", CS42L52_ADC_PGA_A, 0, 1, 0),
+ SOC_SINGLE("PGA AIN1R Switch", CS42L52_ADC_PGA_B, 0, 1, 0),
+ SOC_SINGLE("PGA AIN2L Switch", CS42L52_ADC_PGA_A, 1, 1, 0),
+ SOC_SINGLE("PGA AIN2R Switch", CS42L52_ADC_PGA_B, 1, 1, 0),
+
+ SOC_SINGLE("PGA AIN3L Switch", CS42L52_ADC_PGA_A, 2, 1, 0),
+ SOC_SINGLE("PGA AIN3R Switch", CS42L52_ADC_PGA_B, 2, 1, 0),
+
+ SOC_SINGLE("PGA AIN4L Switch", CS42L52_ADC_PGA_A, 3, 1, 0),
+ SOC_SINGLE("PGA AIN4R Switch", CS42L52_ADC_PGA_B, 3, 1, 0),
+
+ SOC_SINGLE("PGA MICA Switch", CS42L52_ADC_PGA_A, 4, 1, 0),
+ SOC_SINGLE("PGA MICB Switch", CS42L52_ADC_PGA_B, 4, 1, 0),
+
+};
+
+static const struct snd_soc_dapm_widget cs42l52_dapm_widgets[] = {
+
+ SND_SOC_DAPM_INPUT("AIN1L"),
+ SND_SOC_DAPM_INPUT("AIN1R"),
+ SND_SOC_DAPM_INPUT("AIN2L"),
+ SND_SOC_DAPM_INPUT("AIN2R"),
+ SND_SOC_DAPM_INPUT("AIN3L"),
+ SND_SOC_DAPM_INPUT("AIN3R"),
+ SND_SOC_DAPM_INPUT("AIN4L"),
+ SND_SOC_DAPM_INPUT("AIN4R"),
+ SND_SOC_DAPM_INPUT("MICA"),
+ SND_SOC_DAPM_INPUT("MICB"),
+ SND_SOC_DAPM_SIGGEN("Beep"),
+
+ SND_SOC_DAPM_AIF_OUT("AIFOUTL", NULL, 0,
+ SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("AIFOUTR", NULL, 0,
+ SND_SOC_NOPM, 0, 0),
+
+ SND_SOC_DAPM_MUX("MICA Mux", SND_SOC_NOPM, 0, 0, &mica_mux),
+ SND_SOC_DAPM_MUX("MICB Mux", SND_SOC_NOPM, 0, 0, &micb_mux),
+
+ SND_SOC_DAPM_ADC("ADC Left", NULL, CS42L52_PWRCTL1, 1, 1),
+ SND_SOC_DAPM_ADC("ADC Right", NULL, CS42L52_PWRCTL1, 2, 1),
+ SND_SOC_DAPM_PGA("PGA Left", CS42L52_PWRCTL1, 3, 1, NULL, 0),
+ SND_SOC_DAPM_PGA("PGA Right", CS42L52_PWRCTL1, 4, 1, NULL, 0),
+
+ SND_SOC_DAPM_MUX("ADC Left Mux", SND_SOC_NOPM, 0, 0, &adca_mux),
+ SND_SOC_DAPM_MUX("ADC Right Mux", SND_SOC_NOPM, 0, 0, &adcb_mux),
+
+ SND_SOC_DAPM_MUX("ADC Left Swap", SND_SOC_NOPM,
+ 0, 0, &adca_mixer),
+ SND_SOC_DAPM_MUX("ADC Right Swap", SND_SOC_NOPM,
+ 0, 0, &adcb_mixer),
+
+ SND_SOC_DAPM_MUX("Output Mux", SND_SOC_NOPM,
+ 0, 0, &digital_output_mux),
+
+ SND_SOC_DAPM_PGA("PGA MICA", CS42L52_PWRCTL2, 1, 1, NULL, 0),
+ SND_SOC_DAPM_PGA("PGA MICB", CS42L52_PWRCTL2, 2, 1, NULL, 0),
+
+ SND_SOC_DAPM_SUPPLY("Mic Bias", CS42L52_PWRCTL2, 0, 1, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("Charge Pump", CS42L52_PWRCTL1, 7, 1, NULL, 0),
+
+ SND_SOC_DAPM_AIF_IN("AIFINL", NULL, 0,
+ SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_IN("AIFINR", NULL, 0,
+ SND_SOC_NOPM, 0, 0),
+
+ SND_SOC_DAPM_DAC("DAC Left", NULL, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_DAC("DAC Right", NULL, SND_SOC_NOPM, 0, 0),
+
+ SND_SOC_DAPM_SWITCH("Bypass Left", CS42L52_MISC_CTL,
+ 6, 0, &passthrul_ctl),
+ SND_SOC_DAPM_SWITCH("Bypass Right", CS42L52_MISC_CTL,
+ 7, 0, &passthrur_ctl),
+
+ SND_SOC_DAPM_MUX("PCM Left Swap", SND_SOC_NOPM,
+ 0, 0, &pcma_mixer),
+ SND_SOC_DAPM_MUX("PCM Right Swap", SND_SOC_NOPM,
+ 0, 0, &pcmb_mixer),
+
+ SND_SOC_DAPM_SWITCH("HP Left Amp", SND_SOC_NOPM, 0, 0, &hpl_ctl),
+ SND_SOC_DAPM_SWITCH("HP Right Amp", SND_SOC_NOPM, 0, 0, &hpr_ctl),
+
+ SND_SOC_DAPM_SWITCH("SPK Left Amp", SND_SOC_NOPM, 0, 0, &spkl_ctl),
+ SND_SOC_DAPM_SWITCH("SPK Right Amp", SND_SOC_NOPM, 0, 0, &spkr_ctl),
+
+ SND_SOC_DAPM_OUTPUT("HPOUTA"),
+ SND_SOC_DAPM_OUTPUT("HPOUTB"),
+ SND_SOC_DAPM_OUTPUT("SPKOUTA"),
+ SND_SOC_DAPM_OUTPUT("SPKOUTB"),
+
+};
+
+static const struct snd_soc_dapm_route cs42l52_audio_map[] = {
+
+ {"Capture", NULL, "AIFOUTL"},
+ {"Capture", NULL, "AIFOUTL"},
+
+ {"AIFOUTL", NULL, "Output Mux"},
+ {"AIFOUTR", NULL, "Output Mux"},
+
+ {"Output Mux", "ADC", "ADC Left"},
+ {"Output Mux", "ADC", "ADC Right"},
+
+ {"ADC Left", NULL, "Charge Pump"},
+ {"ADC Right", NULL, "Charge Pump"},
+
+ {"Charge Pump", NULL, "ADC Left Mux"},
+ {"Charge Pump", NULL, "ADC Right Mux"},
+
+ {"ADC Left Mux", "Input1A", "AIN1L"},
+ {"ADC Right Mux", "Input1B", "AIN1R"},
+ {"ADC Left Mux", "Input2A", "AIN2L"},
+ {"ADC Right Mux", "Input2B", "AIN2R"},
+ {"ADC Left Mux", "Input3A", "AIN3L"},
+ {"ADC Right Mux", "Input3B", "AIN3R"},
+ {"ADC Left Mux", "Input4A", "AIN4L"},
+ {"ADC Right Mux", "Input4B", "AIN4R"},
+ {"ADC Left Mux", "PGA Input Left", "PGA Left"},
+ {"ADC Right Mux", "PGA Input Right" , "PGA Right"},
+
+ {"PGA Left", "Switch", "AIN1L"},
+ {"PGA Right", "Switch", "AIN1R"},
+ {"PGA Left", "Switch", "AIN2L"},
+ {"PGA Right", "Switch", "AIN2R"},
+ {"PGA Left", "Switch", "AIN3L"},
+ {"PGA Right", "Switch", "AIN3R"},
+ {"PGA Left", "Switch", "AIN4L"},
+ {"PGA Right", "Switch", "AIN4R"},
+
+ {"PGA Left", "Switch", "PGA MICA"},
+ {"PGA MICA", NULL, "MICA"},
+
+ {"PGA Right", "Switch", "PGA MICB"},
+ {"PGA MICB", NULL, "MICB"},
+
+ {"HPOUTA", NULL, "HP Left Amp"},
+ {"HPOUTB", NULL, "HP Right Amp"},
+ {"HP Left Amp", NULL, "Bypass Left"},
+ {"HP Right Amp", NULL, "Bypass Right"},
+ {"Bypass Left", "Switch", "PGA Left"},
+ {"Bypass Right", "Switch", "PGA Right"},
+ {"HP Left Amp", "Switch", "DAC Left"},
+ {"HP Right Amp", "Switch", "DAC Right"},
+
+ {"SPKOUTA", NULL, "SPK Left Amp"},
+ {"SPKOUTB", NULL, "SPK Right Amp"},
+
+ {"SPK Left Amp", NULL, "Beep"},
+ {"SPK Right Amp", NULL, "Beep"},
+ {"SPK Left Amp", "Switch", "Playback"},
+ {"SPK Right Amp", "Switch", "Playback"},
+
+ {"DAC Left", NULL, "Beep"},
+ {"DAC Right", NULL, "Beep"},
+ {"DAC Left", NULL, "Playback"},
+ {"DAC Right", NULL, "Playback"},
+
+ {"Output Mux", "DSP", "Playback"},
+ {"Output Mux", "DSP", "Playback"},
+
+ {"AIFINL", NULL, "Playback"},
+ {"AIFINR", NULL, "Playback"},
+
+};
+
+struct cs42l52_clk_para {
+ u32 mclk;
+ u32 rate;
+ u8 speed;
+ u8 group;
+ u8 videoclk;
+ u8 ratio;
+ u8 mclkdiv2;
+};
+
+static const struct cs42l52_clk_para clk_map_table[] = {
+ /*8k*/
+ {12288000, 8000, CLK_QS_MODE, CLK_32K, CLK_NO_27M, CLK_R_128, 0},
+ {18432000, 8000, CLK_QS_MODE, CLK_32K, CLK_NO_27M, CLK_R_128, 0},
+ {12000000, 8000, CLK_QS_MODE, CLK_32K, CLK_NO_27M, CLK_R_125, 0},
+ {24000000, 8000, CLK_QS_MODE, CLK_32K, CLK_NO_27M, CLK_R_125, 1},
+ {27000000, 8000, CLK_QS_MODE, CLK_32K, CLK_27M_MCLK, CLK_R_125, 0},
+
+ /*11.025k*/
+ {11289600, 11025, CLK_QS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_128, 0},
+ {16934400, 11025, CLK_QS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_128, 0},
+
+ /*16k*/
+ {12288000, 16000, CLK_HS_MODE, CLK_32K, CLK_NO_27M, CLK_R_128, 0},
+ {18432000, 16000, CLK_HS_MODE, CLK_32K, CLK_NO_27M, CLK_R_128, 0},
+ {12000000, 16000, CLK_HS_MODE, CLK_32K, CLK_NO_27M, CLK_R_125, 0},
+ {24000000, 16000, CLK_HS_MODE, CLK_32K, CLK_NO_27M, CLK_R_125, 1},
+ {27000000, 16000, CLK_HS_MODE, CLK_32K, CLK_27M_MCLK, CLK_R_125, 1},
+
+ /*22.05k*/
+ {11289600, 22050, CLK_HS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_128, 0},
+ {16934400, 22050, CLK_HS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_128, 0},
+
+ /* 32k */
+ {12288000, 32000, CLK_SS_MODE, CLK_32K, CLK_NO_27M, CLK_R_128, 0},
+ {18432000, 32000, CLK_SS_MODE, CLK_32K, CLK_NO_27M, CLK_R_128, 0},
+ {12000000, 32000, CLK_SS_MODE, CLK_32K, CLK_NO_27M, CLK_R_125, 0},
+ {24000000, 32000, CLK_SS_MODE, CLK_32K, CLK_NO_27M, CLK_R_125, 1},
+ {27000000, 32000, CLK_SS_MODE, CLK_32K, CLK_27M_MCLK, CLK_R_125, 0},
+
+ /* 44.1k */
+ {11289600, 44100, CLK_SS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_128, 0},
+ {16934400, 44100, CLK_SS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_128, 0},
+
+ /* 48k */
+ {12288000, 48000, CLK_SS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_128, 0},
+ {18432000, 48000, CLK_SS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_128, 0},
+ {12000000, 48000, CLK_SS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_125, 0},
+ {24000000, 48000, CLK_SS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_125, 1},
+ {27000000, 48000, CLK_SS_MODE, CLK_NO_32K, CLK_27M_MCLK, CLK_R_125, 1},
+
+ /* 88.2k */
+ {11289600, 88200, CLK_DS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_128, 0},
+ {16934400, 88200, CLK_DS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_128, 0},
+
+ /* 96k */
+ {12288000, 96000, CLK_DS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_128, 0},
+ {18432000, 96000, CLK_DS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_128, 0},
+ {12000000, 96000, CLK_DS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_125, 0},
+ {24000000, 96000, CLK_DS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_125, 1},
+};
+
+static int cs42l52_get_clk(int mclk, int rate)
+{
+ int i, ret = 0;
+ u_int mclk1, mclk2 = 0;
+
+ for (i = 0; i < ARRAY_SIZE(clk_map_table); i++) {
+ if (clk_map_table[i].rate == rate) {
+ mclk1 = clk_map_table[i].mclk;
+ if (abs(mclk - mclk1) < abs(mclk - mclk2)) {
+ mclk2 = mclk1;
+ ret = i;
+ }
+ }
+ }
+ if (ret > ARRAY_SIZE(clk_map_table))
+ return -EINVAL;
+ return ret;
+}
+
+static int cs42l52_set_sysclk(struct snd_soc_dai *codec_dai,
+ int clk_id, unsigned int freq, int dir)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ struct cs42l52_private *cs42l52 = snd_soc_codec_get_drvdata(codec);
+
+ if ((freq >= CS42L52_MIN_CLK) && (freq <= CS42L52_MAX_CLK)) {
+ cs42l52->sysclk = freq;
+ } else {
+ dev_err(codec->dev, "Invalid freq paramter\n");
+ return -EINVAL;
+ }
+ return 0;
+}
+
+static int cs42l52_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ struct cs42l52_private *cs42l52 = snd_soc_codec_get_drvdata(codec);
+ int ret = 0;
+ u8 iface = 0;
+
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBM_CFM:
+ iface = CS42L52_IFACE_CTL1_MASTER;
+ break;
+ case SND_SOC_DAIFMT_CBS_CFS:
+ iface = CS42L52_IFACE_CTL1_SLAVE;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ /* interface format */
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ iface |= CS42L52_IFACE_CTL1_ADC_FMT_I2S |
+ CS42L52_IFACE_CTL1_DAC_FMT_I2S;
+ break;
+ case SND_SOC_DAIFMT_RIGHT_J:
+ iface |= CS42L52_IFACE_CTL1_DAC_FMT_RIGHT_J;
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ iface |= CS42L52_IFACE_CTL1_ADC_FMT_LEFT_J |
+ CS42L52_IFACE_CTL1_DAC_FMT_LEFT_J;
+ break;
+ case SND_SOC_DAIFMT_DSP_A:
+ iface |= CS42L52_IFACE_CTL1_DSP_MODE_EN;
+ break;
+ case SND_SOC_DAIFMT_DSP_B:
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ /* clock inversion */
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_NB_NF:
+ break;
+ case SND_SOC_DAIFMT_IB_IF:
+ iface |= CS42L52_IFACE_CTL1_INV_SCLK;
+ break;
+ case SND_SOC_DAIFMT_IB_NF:
+ iface |= CS42L52_IFACE_CTL1_INV_SCLK;
+ break;
+ case SND_SOC_DAIFMT_NB_IF:
+ break;
+ default:
+ ret = -EINVAL;
+ }
+ cs42l52->config.format = iface;
+ snd_soc_write(codec, CS42L52_IFACE_CTL1, cs42l52->config.format);
+
+ return 0;
+}
+
+static int cs42l52_digital_mute(struct snd_soc_dai *dai, int mute)
+{
+ struct snd_soc_codec *codec = dai->codec;
+
+ if (mute)
+ snd_soc_update_bits(codec, CS42L52_PB_CTL1,
+ CS42L52_PB_CTL1_MUTE_MASK,
+ CS42L52_PB_CTL1_MUTE);
+ else
+ snd_soc_update_bits(codec, CS42L52_PB_CTL1,
+ CS42L52_PB_CTL1_MUTE_MASK,
+ CS42L52_PB_CTL1_UNMUTE);
+
+ return 0;
+}
+
+static int cs42l52_pcm_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 cs42l52_private *cs42l52 = snd_soc_codec_get_drvdata(codec);
+ u32 clk = 0;
+ int index;
+
+ index = cs42l52_get_clk(cs42l52->sysclk, params_rate(params));
+ if (index >= 0) {
+ cs42l52->sysclk = clk_map_table[index].mclk;
+
+ clk |= (clk_map_table[index].speed << CLK_SPEED_SHIFT) |
+ (clk_map_table[index].group << CLK_32K_SR_SHIFT) |
+ (clk_map_table[index].videoclk << CLK_27M_MCLK_SHIFT) |
+ (clk_map_table[index].ratio << CLK_RATIO_SHIFT) |
+ clk_map_table[index].mclkdiv2;
+
+ snd_soc_write(codec, CS42L52_CLK_CTL, clk);
+ } else {
+ dev_err(codec->dev, "can't get correct mclk\n");
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static int cs42l52_set_bias_level(struct snd_soc_codec *codec,
+ enum snd_soc_bias_level level)
+{
+ struct cs42l52_private *cs42l52 = snd_soc_codec_get_drvdata(codec);
+
+ switch (level) {
+ case SND_SOC_BIAS_ON:
+ break;
+ case SND_SOC_BIAS_PREPARE:
+ snd_soc_update_bits(codec, CS42L52_PWRCTL1,
+ CS42L52_PWRCTL1_PDN_CODEC, 0);
+ break;
+ case SND_SOC_BIAS_STANDBY:
+ if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
+ regcache_cache_only(cs42l52->regmap, false);
+ regcache_sync(cs42l52->regmap);
+ }
+ snd_soc_write(codec, CS42L52_PWRCTL1, CS42L52_PWRCTL1_PDN_ALL);
+ break;
+ case SND_SOC_BIAS_OFF:
+ snd_soc_write(codec, CS42L52_PWRCTL1, CS42L52_PWRCTL1_PDN_ALL);
+ regcache_cache_only(cs42l52->regmap, true);
+ break;
+ }
+ codec->dapm.bias_level = level;
+
+ return 0;
+}
+
+#define CS42L52_RATES (SNDRV_PCM_RATE_8000_96000)
+
+#define CS42L52_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_U16_LE | \
+ SNDRV_PCM_FMTBIT_S18_3LE | SNDRV_PCM_FMTBIT_U18_3LE | \
+ SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_U20_3LE | \
+ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_U24_LE)
+
+static struct snd_soc_dai_ops cs42l52_ops = {
+ .hw_params = cs42l52_pcm_hw_params,
+ .digital_mute = cs42l52_digital_mute,
+ .set_fmt = cs42l52_set_fmt,
+ .set_sysclk = cs42l52_set_sysclk,
+};
+
+static struct snd_soc_dai_driver cs42l52_dai = {
+ .name = "cs42l52",
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = CS42L52_RATES,
+ .formats = CS42L52_FORMATS,
+ },
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = CS42L52_RATES,
+ .formats = CS42L52_FORMATS,
+ },
+ .ops = &cs42l52_ops,
+};
+
+static int cs42l52_suspend(struct snd_soc_codec *codec)
+{
+ cs42l52_set_bias_level(codec, SND_SOC_BIAS_OFF);
+
+ return 0;
+}
+
+static int cs42l52_resume(struct snd_soc_codec *codec)
+{
+ cs42l52_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+ return 0;
+}
+
+#if defined(CONFIG_INPUT) || defined(CONFIG_INPUT_MODULE)
+static int beep_rates[] = {
+ 261, 522, 585, 667, 706, 774, 889, 1000,
+ 1043, 1200, 1333, 1412, 1600, 1714, 2000, 2182
+};
+
+static void cs42l52_beep_work(struct work_struct *work)
+{
+ struct cs42l52_private *cs42l52 =
+ container_of(work, struct cs42l52_private, beep_work);
+ struct snd_soc_codec *codec = cs42l52->codec;
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
+ int i;
+ int val = 0;
+ int best = 0;
+
+ if (cs42l52->beep_rate) {
+ for (i = 0; i < ARRAY_SIZE(beep_rates); i++) {
+ if (abs(cs42l52->beep_rate - beep_rates[i]) <
+ abs(cs42l52->beep_rate - beep_rates[best]))
+ best = i;
+ }
+
+ dev_dbg(codec->dev, "Set beep rate %dHz for requested %dHz\n",
+ beep_rates[best], cs42l52->beep_rate);
+
+ val = (best << CS42L52_BEEP_RATE_SHIFT);
+
+ snd_soc_dapm_enable_pin(dapm, "Beep");
+ } else {
+ dev_dbg(codec->dev, "Disabling beep\n");
+ snd_soc_dapm_disable_pin(dapm, "Beep");
+ }
+
+ snd_soc_update_bits(codec, CS42L52_BEEP_FREQ,
+ CS42L52_BEEP_RATE_MASK, val);
+
+ snd_soc_dapm_sync(dapm);
+}
+
+/* For usability define a way of injecting beep events for the device -
+ * many systems will not have a keyboard.
+ */
+static int cs42l52_beep_event(struct input_dev *dev, unsigned int type,
+ unsigned int code, int hz)
+{
+ struct snd_soc_codec *codec = input_get_drvdata(dev);
+ struct cs42l52_private *cs42l52 = snd_soc_codec_get_drvdata(codec);
+
+ dev_dbg(codec->dev, "Beep event %x %x\n", code, hz);
+
+ switch (code) {
+ case SND_BELL:
+ if (hz)
+ hz = 261;
+ case SND_TONE:
+ break;
+ default:
+ return -1;
+ }
+
+ /* Kick the beep from a workqueue */
+ cs42l52->beep_rate = hz;
+ schedule_work(&cs42l52->beep_work);
+ return 0;
+}
+
+static ssize_t cs42l52_beep_set(struct device *dev,
+ struct device_attribute *attr,
+ const char *buf, size_t count)
+{
+ struct cs42l52_private *cs42l52 = dev_get_drvdata(dev);
+ long int time;
+ int ret;
+
+ ret = kstrtol(buf, 10, &time);
+ if (ret != 0)
+ return ret;
+
+ input_event(cs42l52->beep, EV_SND, SND_TONE, time);
+
+ return count;
+}
+
+static DEVICE_ATTR(beep, 0200, NULL, cs42l52_beep_set);
+
+static void cs42l52_init_beep(struct snd_soc_codec *codec)
+{
+ struct cs42l52_private *cs42l52 = snd_soc_codec_get_drvdata(codec);
+ int ret;
+
+ cs42l52->beep = input_allocate_device();
+ if (!cs42l52->beep) {
+ dev_err(codec->dev, "Failed to allocate beep device\n");
+ return;
+ }
+
+ INIT_WORK(&cs42l52->beep_work, cs42l52_beep_work);
+ cs42l52->beep_rate = 0;
+
+ cs42l52->beep->name = "CS42L52 Beep Generator";
+ cs42l52->beep->phys = dev_name(codec->dev);
+ cs42l52->beep->id.bustype = BUS_I2C;
+
+ cs42l52->beep->evbit[0] = BIT_MASK(EV_SND);
+ cs42l52->beep->sndbit[0] = BIT_MASK(SND_BELL) | BIT_MASK(SND_TONE);
+ cs42l52->beep->event = cs42l52_beep_event;
+ cs42l52->beep->dev.parent = codec->dev;
+ input_set_drvdata(cs42l52->beep, codec);
+
+ ret = input_register_device(cs42l52->beep);
+ if (ret != 0) {
+ input_free_device(cs42l52->beep);
+ cs42l52->beep = NULL;
+ dev_err(codec->dev, "Failed to register beep device\n");
+ }
+
+ ret = device_create_file(codec->dev, &dev_attr_beep);
+ if (ret != 0) {
+ dev_err(codec->dev, "Failed to create keyclick file: %d\n",
+ ret);
+ }
+}
+
+static void cs42l52_free_beep(struct snd_soc_codec *codec)
+{
+ struct cs42l52_private *cs42l52 = snd_soc_codec_get_drvdata(codec);
+
+ device_remove_file(codec->dev, &dev_attr_beep);
+ input_unregister_device(cs42l52->beep);
+ cancel_work_sync(&cs42l52->beep_work);
+ cs42l52->beep = NULL;
+
+ snd_soc_update_bits(codec, CS42L52_BEEP_TONE_CTL,
+ CS42L52_BEEP_EN_MASK, 0);
+}
+#else
+static void cs42l52_init_beep(struct snd_soc_codec *codec)
+{
+}
+
+static void cs42l52_free_beep(struct snd_soc_codec *codec)
+{
+}
+#endif
+
+static int cs42l52_probe(struct snd_soc_codec *codec)
+{
+ struct cs42l52_private *cs42l52 = snd_soc_codec_get_drvdata(codec);
+ int ret;
+
+ codec->control_data = cs42l52->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;
+ }
+ regcache_cache_only(cs42l52->regmap, true);
+
+ cs42l52_init_beep(codec);
+
+ cs42l52_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+ cs42l52->sysclk = CS42L52_DEFAULT_CLK;
+ cs42l52->config.format = CS42L52_DEFAULT_FORMAT;
+
+ /* Set Platform MICx CFG */
+ snd_soc_update_bits(codec, CS42L52_MICA_CTL,
+ CS42L52_MIC_CTL_TYPE_MASK,
+ cs42l52->pdata.mica_cfg <<
+ CS42L52_MIC_CTL_TYPE_SHIFT);
+
+ snd_soc_update_bits(codec, CS42L52_MICB_CTL,
+ CS42L52_MIC_CTL_TYPE_MASK,
+ cs42l52->pdata.micb_cfg <<
+ CS42L52_MIC_CTL_TYPE_SHIFT);
+
+ /* if Single Ended, Get Mic_Select */
+ if (cs42l52->pdata.mica_cfg)
+ snd_soc_update_bits(codec, CS42L52_MICA_CTL,
+ CS42L52_MIC_CTL_MIC_SEL_MASK,
+ cs42l52->pdata.mica_sel <<
+ CS42L52_MIC_CTL_MIC_SEL_SHIFT);
+ if (cs42l52->pdata.micb_cfg)
+ snd_soc_update_bits(codec, CS42L52_MICB_CTL,
+ CS42L52_MIC_CTL_MIC_SEL_MASK,
+ cs42l52->pdata.micb_sel <<
+ CS42L52_MIC_CTL_MIC_SEL_SHIFT);
+
+ /* Set Platform Charge Pump Freq */
+ snd_soc_update_bits(codec, CS42L52_CHARGE_PUMP,
+ CS42L52_CHARGE_PUMP_MASK,
+ cs42l52->pdata.chgfreq <<
+ CS42L52_CHARGE_PUMP_SHIFT);
+
+ /* Set Platform Bias Level */
+ snd_soc_update_bits(codec, CS42L52_IFACE_CTL2,
+ CS42L52_IFACE_CTL2_BIAS_LVL,
+ cs42l52->pdata.micbias_lvl);
+
+ return ret;
+}
+
+static int cs42l52_remove(struct snd_soc_codec *codec)
+{
+ cs42l52_free_beep(codec);
+ cs42l52_set_bias_level(codec, SND_SOC_BIAS_OFF);
+
+ return 0;
+}
+
+static struct snd_soc_codec_driver soc_codec_dev_cs42l52 = {
+ .probe = cs42l52_probe,
+ .remove = cs42l52_remove,
+ .suspend = cs42l52_suspend,
+ .resume = cs42l52_resume,
+ .set_bias_level = cs42l52_set_bias_level,
+
+ .dapm_widgets = cs42l52_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(cs42l52_dapm_widgets),
+ .dapm_routes = cs42l52_audio_map,
+ .num_dapm_routes = ARRAY_SIZE(cs42l52_audio_map),
+
+ .controls = cs42l52_snd_controls,
+ .num_controls = ARRAY_SIZE(cs42l52_snd_controls),
+};
+
+/* Current and threshold powerup sequence Pg37 */
+static const struct reg_default cs42l52_threshold_patch[] = {
+
+ { 0x00, 0x99 },
+ { 0x3E, 0xBA },
+ { 0x47, 0x80 },
+ { 0x32, 0xBB },
+ { 0x32, 0x3B },
+ { 0x00, 0x00 },
+
+};
+
+static struct regmap_config cs42l52_regmap = {
+ .reg_bits = 8,
+ .val_bits = 8,
+
+ .max_register = CS42L52_MAX_REGISTER,
+ .reg_defaults = cs42l52_reg_defaults,
+ .num_reg_defaults = ARRAY_SIZE(cs42l52_reg_defaults),
+ .readable_reg = cs42l52_readable_register,
+ .volatile_reg = cs42l52_volatile_register,
+ .cache_type = REGCACHE_RBTREE,
+};
+
+static int cs42l52_i2c_probe(struct i2c_client *i2c_client,
+ const struct i2c_device_id *id)
+{
+ struct cs42l52_private *cs42l52;
+ int ret;
+ unsigned int devid = 0;
+ unsigned int reg;
+
+ cs42l52 = devm_kzalloc(&i2c_client->dev, sizeof(struct cs42l52_private),
+ GFP_KERNEL);
+ if (cs42l52 == NULL)
+ return -ENOMEM;
+ cs42l52->dev = &i2c_client->dev;
+
+ cs42l52->regmap = 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;
+ }
+
+ i2c_set_clientdata(i2c_client, cs42l52);
+
+ if (dev_get_platdata(&i2c_client->dev))
+ memcpy(&cs42l52->pdata, dev_get_platdata(&i2c_client->dev),
+ sizeof(cs42l52->pdata));
+
+ ret = regmap_register_patch(cs42l52->regmap, cs42l52_threshold_patch,
+ ARRAY_SIZE(cs42l52_threshold_patch));
+ if (ret != 0)
+ dev_warn(cs42l52->dev, "Failed to apply regmap patch: %d\n",
+ ret);
+
+ ret = regmap_read(cs42l52->regmap, CS42L52_CHIP, &reg);
+ devid = reg & CS42L52_CHIP_ID_MASK;
+ if (devid != CS42L52_CHIP_ID) {
+ ret = -ENODEV;
+ dev_err(&i2c_client->dev,
+ "CS42L52 Device ID (%X). Expected %X\n",
+ devid, CS42L52_CHIP_ID);
+ goto err_regmap;
+ }
+
+ regcache_cache_only(cs42l52->regmap, true);
+
+ ret = snd_soc_register_codec(&i2c_client->dev,
+ &soc_codec_dev_cs42l52, &cs42l52_dai, 1);
+ if (ret < 0)
+ goto err_regmap;
+ 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;
+}
+
+static const struct i2c_device_id cs42l52_id[] = {
+ { "cs42l52", 0 },
+ { }
+};
+MODULE_DEVICE_TABLE(i2c, cs42l52_id);
+
+static struct i2c_driver cs42l52_i2c_driver = {
+ .driver = {
+ .name = "cs42l52",
+ .owner = THIS_MODULE,
+ },
+ .id_table = cs42l52_id,
+ .probe = cs42l52_i2c_probe,
+ .remove = __devexit_p(cs42l52_i2c_remove),
+};
+
+module_i2c_driver(cs42l52_i2c_driver);
+
+MODULE_DESCRIPTION("ASoC CS42L52 driver");
+MODULE_AUTHOR("Georgi Vlaev, Nucleus Systems Ltd, <joe@nucleusys.com>");
+MODULE_AUTHOR("Brian Austin, Cirrus Logic Inc, <brian.austin@cirrus.com>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/cs42l52.h b/sound/soc/codecs/cs42l52.h
new file mode 100644
index 00000000000..60985c05907
--- /dev/null
+++ b/sound/soc/codecs/cs42l52.h
@@ -0,0 +1,274 @@
+/*
+ * cs42l52.h -- CS42L52 ALSA SoC audio driver
+ *
+ * Copyright 2012 CirrusLogic, Inc.
+ *
+ * Author: Georgi Vlaev <joe@nucleusys.com>
+ * Author: Brian Austin <brian.austin@cirrus.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 __CS42L52_H__
+#define __CS42L52_H__
+
+#define CS42L52_NAME "CS42L52"
+#define CS42L52_DEFAULT_CLK 12000000
+#define CS42L52_MIN_CLK 11000000
+#define CS42L52_MAX_CLK 27000000
+#define CS42L52_DEFAULT_FORMAT SNDRV_PCM_FMTBIT_S16_LE
+#define CS42L52_DEFAULT_MAX_CHANS 2
+#define CS42L52_SYSCLK 1
+
+#define CS42L52_CHIP_SWICTH (1 << 17)
+#define CS42L52_ALL_IN_ONE (1 << 16)
+#define CS42L52_CHIP_ONE 0x00
+#define CS42L52_CHIP_TWO 0x01
+#define CS42L52_CHIP_THR 0x02
+#define CS42L52_CHIP_MASK 0x0f
+
+#define CS42L52_FIX_BITS_CTL 0x00
+#define CS42L52_CHIP 0x01
+#define CS42L52_CHIP_ID 0xE0
+#define CS42L52_CHIP_ID_MASK 0xF8
+#define CS42L52_CHIP_REV_A0 0x00
+#define CS42L52_CHIP_REV_A1 0x01
+#define CS42L52_CHIP_REV_B0 0x02
+#define CS42L52_CHIP_REV_MASK 0x03
+
+#define CS42L52_PWRCTL1 0x02
+#define CS42L52_PWRCTL1_PDN_ALL 0x9F
+#define CS42L52_PWRCTL1_PDN_CHRG 0x80
+#define CS42L52_PWRCTL1_PDN_PGAB 0x10
+#define CS42L52_PWRCTL1_PDN_PGAA 0x08
+#define CS42L52_PWRCTL1_PDN_ADCB 0x04
+#define CS42L52_PWRCTL1_PDN_ADCA 0x02
+#define CS42L52_PWRCTL1_PDN_CODEC 0x01
+
+#define CS42L52_PWRCTL2 0x03
+#define CS42L52_PWRCTL2_OVRDB (1 << 4)
+#define CS42L52_PWRCTL2_OVRDA (1 << 3)
+#define CS42L52_PWRCTL2_PDN_MICB (1 << 2)
+#define CS42L52_PWRCTL2_PDN_MICB_SHIFT 2
+#define CS42L52_PWRCTL2_PDN_MICA (1 << 1)
+#define CS42L52_PWRCTL2_PDN_MICA_SHIFT 1
+#define CS42L52_PWRCTL2_PDN_MICBIAS (1 << 0)
+#define CS42L52_PWRCTL2_PDN_MICBIAS_SHIFT 0
+
+#define CS42L52_PWRCTL3 0x04
+#define CS42L52_PWRCTL3_HPB_PDN_SHIFT 6
+#define CS42L52_PWRCTL3_HPB_ON_LOW 0x00
+#define CS42L52_PWRCTL3_HPB_ON_HIGH 0x01
+#define CS42L52_PWRCTL3_HPB_ALWAYS_ON 0x02
+#define CS42L52_PWRCTL3_HPB_ALWAYS_OFF 0x03
+#define CS42L52_PWRCTL3_HPA_PDN_SHIFT 4
+#define CS42L52_PWRCTL3_HPA_ON_LOW 0x00
+#define CS42L52_PWRCTL3_HPA_ON_HIGH 0x01
+#define CS42L52_PWRCTL3_HPA_ALWAYS_ON 0x02
+#define CS42L52_PWRCTL3_HPA_ALWAYS_OFF 0x03
+#define CS42L52_PWRCTL3_SPKB_PDN_SHIFT 2
+#define CS42L52_PWRCTL3_SPKB_ON_LOW 0x00
+#define CS42L52_PWRCTL3_SPKB_ON_HIGH 0x01
+#define CS42L52_PWRCTL3_SPKB_ALWAYS_ON 0x02
+#define CS42L52_PWRCTL3_PDN_SPKB (1 << 2)
+#define CS42L52_PWRCTL3_PDN_SPKA (1 << 0)
+#define CS42L52_PWRCTL3_SPKA_PDN_SHIFT 0
+#define CS42L52_PWRCTL3_SPKA_ON_LOW 0x00
+#define CS42L52_PWRCTL3_SPKA_ON_HIGH 0x01
+#define CS42L52_PWRCTL3_SPKA_ALWAYS_ON 0x02
+
+#define CS42L52_DEFAULT_OUTPUT_STATE 0x05
+#define CS42L52_PWRCTL3_CONF_MASK 0x03
+
+#define CS42L52_CLK_CTL 0x05
+#define CLK_AUTODECT_ENABLE (1 << 7)
+#define CLK_SPEED_SHIFT 5
+#define CLK_DS_MODE 0x00
+#define CLK_SS_MODE 0x01
+#define CLK_HS_MODE 0x02
+#define CLK_QS_MODE 0x03
+#define CLK_32K_SR_SHIFT 4
+#define CLK_32K 0x01
+#define CLK_NO_32K 0x00
+#define CLK_27M_MCLK_SHIFT 3
+#define CLK_27M_MCLK 0x01
+#define CLK_NO_27M 0x00
+#define CLK_RATIO_SHIFT 1
+#define CLK_R_128 0x00
+#define CLK_R_125 0x01
+#define CLK_R_132 0x02
+#define CLK_R_136 0x03
+
+#define CS42L52_IFACE_CTL1 0x06
+#define CS42L52_IFACE_CTL1_MASTER (1 << 7)
+#define CS42L52_IFACE_CTL1_SLAVE (0 << 7)
+#define CS42L52_IFACE_CTL1_INV_SCLK (1 << 6)
+#define CS42L52_IFACE_CTL1_ADC_FMT_I2S (1 << 5)
+#define CS42L52_IFACE_CTL1_ADC_FMT_LEFT_J (0 << 5)
+#define CS42L52_IFACE_CTL1_DSP_MODE_EN (1 << 4)
+#define CS42L52_IFACE_CTL1_DAC_FMT_LEFT_J (0 << 2)
+#define CS42L52_IFACE_CTL1_DAC_FMT_I2S (1 << 2)
+#define CS42L52_IFACE_CTL1_DAC_FMT_RIGHT_J (2 << 2)
+#define CS42L52_IFACE_CTL1_WL_32BIT (0x00)
+#define CS42L52_IFACE_CTL1_WL_24BIT (0x01)
+#define CS42L52_IFACE_CTL1_WL_20BIT (0x02)
+#define CS42L52_IFACE_CTL1_WL_16BIT (0x03)
+#define CS42L52_IFACE_CTL1_WL_MASK 0xFFFF
+
+#define CS42L52_IFACE_CTL2 0x07
+#define CS42L52_IFACE_CTL2_SC_MC_EQ (1 << 6)
+#define CS42L52_IFACE_CTL2_LOOPBACK (1 << 5)
+#define CS42L52_IFACE_CTL2_S_MODE_OUTPUT_EN (0 << 4)
+#define CS42L52_IFACE_CTL2_S_MODE_OUTPUT_HIZ (1 << 4)
+#define CS42L52_IFACE_CTL2_HP_SW_INV (1 << 3)
+#define CS42L52_IFACE_CTL2_BIAS_LVL 0x07
+
+#define CS42L52_ADC_PGA_A 0x08
+#define CS42L52_ADC_PGA_B 0x09
+#define CS42L52_ADC_SEL_SHIFT 5
+#define CS42L52_ADC_SEL_AIN1 0x00
+#define CS42L52_ADC_SEL_AIN2 0x01
+#define CS42L52_ADC_SEL_AIN3 0x02
+#define CS42L52_ADC_SEL_AIN4 0x03
+#define CS42L52_ADC_SEL_PGA 0x04
+
+#define CS42L52_ANALOG_HPF_CTL 0x0A
+#define CS42L52_HPF_CTL_ANLGSFTB (1 << 3)
+#define CS42L52_HPF_CTL_ANLGSFTA (1 << 0)
+
+#define CS42L52_ADC_HPF_FREQ 0x0B
+#define CS42L52_ADC_MISC_CTL 0x0C
+#define CS42L52_ADC_MISC_CTL_SOURCE_DSP (1 << 6)
+
+#define CS42L52_PB_CTL1 0x0D
+#define CS42L52_PB_CTL1_HP_GAIN_SHIFT 5
+#define CS42L52_PB_CTL1_HP_GAIN_03959 0x00
+#define CS42L52_PB_CTL1_HP_GAIN_04571 0x01
+#define CS42L52_PB_CTL1_HP_GAIN_05111 0x02
+#define CS42L52_PB_CTL1_HP_GAIN_06047 0x03
+#define CS42L52_PB_CTL1_HP_GAIN_07099 0x04
+#define CS42L52_PB_CTL1_HP_GAIN_08399 0x05
+#define CS42L52_PB_CTL1_HP_GAIN_10000 0x06
+#define CS42L52_PB_CTL1_HP_GAIN_11430 0x07
+#define CS42L52_PB_CTL1_INV_PCMB (1 << 3)
+#define CS42L52_PB_CTL1_INV_PCMA (1 << 2)
+#define CS42L52_PB_CTL1_MSTB_MUTE (1 << 1)
+#define CS42L52_PB_CTL1_MSTA_MUTE (1 << 0)
+#define CS42L52_PB_CTL1_MUTE_MASK 0xFFFD
+#define CS42L52_PB_CTL1_MUTE 3
+#define CS42L52_PB_CTL1_UNMUTE 0
+
+#define CS42L52_MISC_CTL 0x0E
+#define CS42L52_MISC_CTL_DEEMPH (1 << 2)
+#define CS42L52_MISC_CTL_DIGSFT (1 << 1)
+#define CS42L52_MISC_CTL_DIGZC (1 << 0)
+
+#define CS42L52_PB_CTL2 0x0F
+#define CS42L52_PB_CTL2_HPB_MUTE (1 << 7)
+#define CS42L52_PB_CTL2_HPA_MUTE (1 << 6)
+#define CS42L52_PB_CTL2_SPKB_MUTE (1 << 5)
+#define CS42L52_PB_CTL2_SPKA_MUTE (1 << 4)
+#define CS42L52_PB_CTL2_SPK_SWAP (1 << 2)
+#define CS42L52_PB_CTL2_SPK_MONO (1 << 1)
+#define CS42L52_PB_CTL2_SPK_MUTE50 (1 << 0)
+
+#define CS42L52_MICA_CTL 0x10
+#define CS42L52_MICB_CTL 0x11
+#define CS42L52_MIC_CTL_MIC_SEL_MASK 0xBF
+#define CS42L52_MIC_CTL_MIC_SEL_SHIFT 6
+#define CS42L52_MIC_CTL_TYPE_MASK 0xDF
+#define CS42L52_MIC_CTL_TYPE_SHIFT 5
+
+
+#define CS42L52_PGAA_CTL 0x12
+#define CS42L52_PGAB_CTL 0x13
+#define CS42L52_PGAX_CTL_VOL_12DB 24
+#define CS42L52_PGAX_CTL_VOL_6DB 12 /*step size 0.5db*/
+
+#define CS42L52_PASSTHRUA_VOL 0x14
+#define CS42L52_PASSTHRUB_VOL 0x15
+
+#define CS42L52_ADCA_VOL 0x16
+#define CS42L52_ADCB_VOL 0x17
+#define CS42L52_ADCX_VOL_24DB 24 /*step size 1db*/
+#define CS42L52_ADCX_VOL_12DB 12
+#define CS42L52_ADCX_VOL_6DB 6
+
+#define CS42L52_ADCA_MIXER_VOL 0x18
+#define CS42L52_ADCB_MIXER_VOL 0x19
+#define CS42L52_ADC_MIXER_VOL_12DB 0x18
+
+#define CS42L52_PCMA_MIXER_VOL 0x1A
+#define CS42L52_PCMB_MIXER_VOL 0x1B
+
+#define CS42L52_BEEP_FREQ 0x1C
+#define CS42L52_BEEP_VOL 0x1D
+#define CS42L52_BEEP_TONE_CTL 0x1E
+#define CS42L52_BEEP_RATE_SHIFT 4
+#define CS42L52_BEEP_RATE_MASK 0x0F
+
+#define CS42L52_TONE_CTL 0x1F
+#define CS42L52_BEEP_EN_MASK 0x3F
+
+#define CS42L52_MASTERA_VOL 0x20
+#define CS42L52_MASTERB_VOL 0x21
+
+#define CS42L52_HPA_VOL 0x22
+#define CS42L52_HPB_VOL 0x23
+#define CS42L52_DEFAULT_HP_VOL 0xF0
+
+#define CS42L52_SPKA_VOL 0x24
+#define CS42L52_SPKB_VOL 0x25
+#define CS42L52_DEFAULT_SPK_VOL 0xF0
+
+#define CS42L52_ADC_PCM_MIXER 0x26
+
+#define CS42L52_LIMITER_CTL1 0x27
+#define CS42L52_LIMITER_CTL2 0x28
+#define CS42L52_LIMITER_AT_RATE 0x29
+
+#define CS42L52_ALC_CTL 0x2A
+#define CS42L52_ALC_CTL_ALCB_ENABLE_SHIFT 7
+#define CS42L52_ALC_CTL_ALCA_ENABLE_SHIFT 6
+#define CS42L52_ALC_CTL_FASTEST_ATTACK 0
+
+#define CS42L52_ALC_RATE 0x2B
+#define CS42L52_ALC_SLOWEST_RELEASE 0x3F
+
+#define CS42L52_ALC_THRESHOLD 0x2C
+#define CS42L52_ALC_MAX_RATE_SHIFT 5
+#define CS42L52_ALC_MIN_RATE_SHIFT 2
+#define CS42L52_ALC_RATE_0DB 0
+#define CS42L52_ALC_RATE_3DB 1
+#define CS42L52_ALC_RATE_6DB 2
+
+#define CS42L52_NOISE_GATE_CTL 0x2D
+#define CS42L52_NG_ENABLE_SHIFT 6
+#define CS42L52_NG_THRESHOLD_SHIFT 2
+#define CS42L52_NG_MIN_70DB 2
+#define CS42L52_NG_DELAY_SHIFT 0
+#define CS42L52_NG_DELAY_100MS 1
+
+#define CS42L52_CLK_STATUS 0x2E
+#define CS42L52_BATT_COMPEN 0x2F
+
+#define CS42L52_BATT_LEVEL 0x30
+#define CS42L52_SPK_STATUS 0x31
+#define CS42L52_SPK_STATUS_PIN_SHIFT 3
+#define CS42L52_SPK_STATUS_PIN_HIGH 1
+
+#define CS42L52_TEM_CTL 0x32
+#define CS42L52_TEM_CTL_SET 0x80
+#define CS42L52_THE_FOLDBACK 0x33
+#define CS42L52_CHARGE_PUMP 0x34
+#define CS42L52_CHARGE_PUMP_MASK 0xF0
+#define CS42L52_CHARGE_PUMP_SHIFT 4
+#define CS42L52_FIX_BITS1 0x3E
+#define CS42L52_FIX_BITS2 0x47
+
+#define CS42L52_MAX_REGISTER 0x34
+
+#endif
diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c
index 3686417f5ea..e0d45fdaa75 100644
--- a/sound/soc/codecs/cs42l73.c
+++ b/sound/soc/codecs/cs42l73.c
@@ -43,9 +43,6 @@ struct cs42l73_private {
};
static const struct reg_default cs42l73_reg_defaults[] = {
- { 1, 0x42 }, /* r01 - Device ID A&B */
- { 2, 0xA7 }, /* r02 - Device ID C&D */
- { 3, 0x30 }, /* r03 - Device ID E */
{ 6, 0xF1 }, /* r06 - Power Ctl 1 */
{ 7, 0xDF }, /* r07 - Power Ctl 2 */
{ 8, 0x3F }, /* r08 - Power Ctl 3 */
@@ -402,37 +399,37 @@ static const struct snd_kcontrol_new ear_amp_ctl =
static const struct snd_kcontrol_new cs42l73_snd_controls[] = {
SOC_DOUBLE_R_SX_TLV("Headphone Analog Playback Volume",
- CS42L73_HPAAVOL, CS42L73_HPBAVOL, 7,
- 0xffffffC1, 0x0C, hpaloa_tlv),
+ CS42L73_HPAAVOL, CS42L73_HPBAVOL, 0,
+ 0x41, 0x4B, hpaloa_tlv),
SOC_DOUBLE_R_SX_TLV("LineOut Analog Playback Volume", CS42L73_LOAAVOL,
- CS42L73_LOBAVOL, 7, 0xffffffC1, 0x0C, hpaloa_tlv),
+ CS42L73_LOBAVOL, 0, 0x41, 0x4B, hpaloa_tlv),
SOC_DOUBLE_R_SX_TLV("Input PGA Analog Volume", CS42L73_MICAPREPGAAVOL,
- CS42L73_MICBPREPGABVOL, 5, 0xffffff35,
- 0x34, micpga_tlv),
+ CS42L73_MICBPREPGABVOL, 5, 0x34,
+ 0x24, micpga_tlv),
SOC_DOUBLE_R("MIC Preamp Switch", CS42L73_MICAPREPGAAVOL,
CS42L73_MICBPREPGABVOL, 6, 1, 1),
SOC_DOUBLE_R_SX_TLV("Input Path Digital Volume", CS42L73_IPADVOL,
- CS42L73_IPBDVOL, 7, 0xffffffA0, 0xA0, ipd_tlv),
+ CS42L73_IPBDVOL, 0, 0xA0, 0x6C, ipd_tlv),
SOC_DOUBLE_R_SX_TLV("HL Digital Playback Volume",
- CS42L73_HLADVOL, CS42L73_HLBDVOL, 7, 0xffffffE5,
- 0xE4, hl_tlv),
+ CS42L73_HLADVOL, CS42L73_HLBDVOL,
+ 0, 0x34, 0xE4, hl_tlv),
SOC_SINGLE_TLV("ADC A Boost Volume",
CS42L73_ADCIPC, 2, 0x01, 1, adc_boost_tlv),
SOC_SINGLE_TLV("ADC B Boost Volume",
- CS42L73_ADCIPC, 6, 0x01, 1, adc_boost_tlv),
+ CS42L73_ADCIPC, 6, 0x01, 1, adc_boost_tlv),
- SOC_SINGLE_TLV("Speakerphone Digital Playback Volume",
- CS42L73_SPKDVOL, 0, 0xE4, 1, hl_tlv),
+ SOC_SINGLE_SX_TLV("Speakerphone Digital Volume",
+ CS42L73_SPKDVOL, 0, 0x34, 0xE4, hl_tlv),
- SOC_SINGLE_TLV("Ear Speaker Digital Playback Volume",
- CS42L73_ESLDVOL, 0, 0xE4, 1, hl_tlv),
+ SOC_SINGLE_SX_TLV("Ear Speaker Digital Volume",
+ CS42L73_ESLDVOL, 0, 0x34, 0xE4, hl_tlv),
SOC_DOUBLE_R("Headphone Analog Playback Switch", CS42L73_HPAAVOL,
CS42L73_HPBAVOL, 7, 1, 1),
@@ -599,17 +596,17 @@ static const struct snd_soc_dapm_widget cs42l73_dapm_widgets[] = {
SND_SOC_DAPM_INPUT("MIC2"),
SND_SOC_DAPM_SUPPLY("MIC2 Bias", CS42L73_PWRCTL2, 7, 1, NULL, 0),
- SND_SOC_DAPM_AIF_OUT("XSPOUTL", "XSP Capture", 0,
+ SND_SOC_DAPM_AIF_OUT("XSPOUTL", NULL, 0,
CS42L73_PWRCTL2, 1, 1),
- SND_SOC_DAPM_AIF_OUT("XSPOUTR", "XSP Capture", 0,
+ SND_SOC_DAPM_AIF_OUT("XSPOUTR", NULL, 0,
CS42L73_PWRCTL2, 1, 1),
- SND_SOC_DAPM_AIF_OUT("ASPOUTL", "ASP Capture", 0,
+ SND_SOC_DAPM_AIF_OUT("ASPOUTL", NULL, 0,
CS42L73_PWRCTL2, 3, 1),
- SND_SOC_DAPM_AIF_OUT("ASPOUTR", "ASP Capture", 0,
+ SND_SOC_DAPM_AIF_OUT("ASPOUTR", NULL, 0,
CS42L73_PWRCTL2, 3, 1),
- SND_SOC_DAPM_AIF_OUT("VSPOUTL", "VSP Capture", 0,
+ SND_SOC_DAPM_AIF_OUT("VSPOUTL", NULL, 0,
CS42L73_PWRCTL2, 4, 1),
- SND_SOC_DAPM_AIF_OUT("VSPOUTR", "VSP Capture", 0,
+ SND_SOC_DAPM_AIF_OUT("VSPOUTR", NULL, 0,
CS42L73_PWRCTL2, 4, 1),
SND_SOC_DAPM_PGA("PGA Left", SND_SOC_NOPM, 0, 0, NULL, 0),
@@ -638,21 +635,21 @@ static const struct snd_soc_dapm_widget cs42l73_dapm_widgets[] = {
SND_SOC_DAPM_MIXER("VSPL Output Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
SND_SOC_DAPM_MIXER("VSPR Output Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
- SND_SOC_DAPM_AIF_IN("XSPINL", "XSP Playback", 0,
+ SND_SOC_DAPM_AIF_IN("XSPINL", NULL, 0,
CS42L73_PWRCTL2, 0, 1),
- SND_SOC_DAPM_AIF_IN("XSPINR", "XSP Playback", 0,
+ SND_SOC_DAPM_AIF_IN("XSPINR", NULL, 0,
CS42L73_PWRCTL2, 0, 1),
- SND_SOC_DAPM_AIF_IN("XSPINM", "XSP Playback", 0,
+ SND_SOC_DAPM_AIF_IN("XSPINM", NULL, 0,
CS42L73_PWRCTL2, 0, 1),
- SND_SOC_DAPM_AIF_IN("ASPINL", "ASP Playback", 0,
+ SND_SOC_DAPM_AIF_IN("ASPINL", NULL, 0,
CS42L73_PWRCTL2, 2, 1),
- SND_SOC_DAPM_AIF_IN("ASPINR", "ASP Playback", 0,
+ SND_SOC_DAPM_AIF_IN("ASPINR", NULL, 0,
CS42L73_PWRCTL2, 2, 1),
- SND_SOC_DAPM_AIF_IN("ASPINM", "ASP Playback", 0,
+ SND_SOC_DAPM_AIF_IN("ASPINM", NULL, 0,
CS42L73_PWRCTL2, 2, 1),
- SND_SOC_DAPM_AIF_IN("VSPIN", "VSP Playback", 0,
+ SND_SOC_DAPM_AIF_IN("VSPIN", NULL, 0,
CS42L73_PWRCTL2, 4, 1),
SND_SOC_DAPM_MIXER("HL Left Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
@@ -776,6 +773,14 @@ static const struct snd_soc_dapm_route cs42l73_audio_map[] = {
{"HL Left Mixer", NULL, "VSPIN"},
{"HL Right Mixer", NULL, "VSPIN"},
+ {"ASPINL", NULL, "ASP Playback"},
+ {"ASPINM", NULL, "ASP Playback"},
+ {"ASPINR", NULL, "ASP Playback"},
+ {"XSPINL", NULL, "XSP Playback"},
+ {"XSPINM", NULL, "XSP Playback"},
+ {"XSPINR", NULL, "XSP Playback"},
+ {"VSPIN", NULL, "VSP Playback"},
+
/* Capture Paths */
{"MIC1", NULL, "MIC1 Bias"},
{"PGA Left Mux", "Mic 1", "MIC1"},
@@ -822,6 +827,13 @@ static const struct snd_soc_dapm_route cs42l73_audio_map[] = {
{"VSPOUTL", NULL, "VSPL Output Mixer"},
{"VSPOUTR", NULL, "VSPR Output Mixer"},
+
+ {"ASP Capture", NULL, "ASPOUTL"},
+ {"ASP Capture", NULL, "ASPOUTR"},
+ {"XSP Capture", NULL, "XSPOUTL"},
+ {"XSP Capture", NULL, "XSPOUTR"},
+ {"VSP Capture", NULL, "VSPOUTL"},
+ {"VSP Capture", NULL, "VSPOUTR"},
};
struct cs42l73_mclk_div {
@@ -1091,8 +1103,7 @@ static int cs42l73_pcm_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;
+ struct snd_soc_codec *codec = dai->codec;
struct cs42l73_private *priv = snd_soc_codec_get_drvdata(codec);
int id = dai->id;
int mclk_coeff;
@@ -1429,25 +1440,7 @@ static struct i2c_driver cs42l73_i2c_driver = {
};
-static int __init cs42l73_modinit(void)
-{
- int ret;
- ret = i2c_add_driver(&cs42l73_i2c_driver);
- if (ret != 0) {
- pr_err("Failed to register CS42L73 I2C driver: %d\n", ret);
- return ret;
- }
- return 0;
-}
-
-module_init(cs42l73_modinit);
-
-static void __exit cs42l73_exit(void)
-{
- i2c_del_driver(&cs42l73_i2c_driver);
-}
-
-module_exit(cs42l73_exit);
+module_i2c_driver(cs42l73_i2c_driver);
MODULE_DESCRIPTION("ASoC CS42L73 driver");
MODULE_AUTHOR("Georgi Vlaev, Nucleus Systems Ltd, <joe@nucleusys.com>");
diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c
index 7843711729b..af5db708051 100644
--- a/sound/soc/codecs/da7210.c
+++ b/sound/soc/codecs/da7210.c
@@ -17,6 +17,7 @@
#include <linux/delay.h>
#include <linux/i2c.h>
+#include <linux/spi/spi.h>
#include <linux/regmap.h>
#include <linux/slab.h>
#include <linux/module.h>
@@ -27,6 +28,7 @@
#include <sound/tlv.h>
/* DA7210 register space */
+#define DA7210_PAGE_CONTROL 0x00
#define DA7210_CONTROL 0x01
#define DA7210_STATUS 0x02
#define DA7210_STARTUP1 0x03
@@ -146,6 +148,7 @@
#define DA7210_DAI_EN (1 << 7)
/*PLL_DIV3 bit fields */
+#define DA7210_PLL_DIV_L_MASK (0xF << 0)
#define DA7210_MCLK_RANGE_10_20_MHZ (1 << 4)
#define DA7210_PLL_BYP (1 << 6)
@@ -162,12 +165,16 @@
#define DA7210_PLL_FS_48000 (0xB << 0)
#define DA7210_PLL_FS_88200 (0xE << 0)
#define DA7210_PLL_FS_96000 (0xF << 0)
+#define DA7210_MCLK_DET_EN (0x1 << 5)
+#define DA7210_MCLK_SRM_EN (0x1 << 6)
#define DA7210_PLL_EN (0x1 << 7)
/* SOFTMUTE bit fields */
#define DA7210_RAMP_EN (1 << 6)
/* CONTROL bit fields */
+#define DA7210_REG_EN (1 << 0)
+#define DA7210_BIAS_EN (1 << 2)
#define DA7210_NOISE_SUP_EN (1 << 3)
/* IN_GAIN bit fields */
@@ -206,6 +213,47 @@
#define DA7210_OUT2_OUTMIX_L (1 << 6)
#define DA7210_OUT2_EN (1 << 7)
+struct pll_div {
+ int fref;
+ int fout;
+ u8 div1;
+ u8 div2;
+ u8 div3;
+ u8 mode; /* 0 = slave, 1 = master */
+};
+
+/* PLL dividers table */
+static const struct pll_div da7210_pll_div[] = {
+ /* for MASTER mode, fs = 44.1Khz */
+ { 12000000, 2822400, 0xE8, 0x6C, 0x2, 1}, /* MCLK=12Mhz */
+ { 13000000, 2822400, 0xDF, 0x28, 0xC, 1}, /* MCLK=13Mhz */
+ { 13500000, 2822400, 0xDB, 0x0A, 0xD, 1}, /* MCLK=13.5Mhz */
+ { 14400000, 2822400, 0xD4, 0x5A, 0x2, 1}, /* MCLK=14.4Mhz */
+ { 19200000, 2822400, 0xBB, 0x43, 0x9, 1}, /* MCLK=19.2Mhz */
+ { 19680000, 2822400, 0xB9, 0x6D, 0xA, 1}, /* MCLK=19.68Mhz */
+ { 19800000, 2822400, 0xB8, 0xFB, 0xB, 1}, /* MCLK=19.8Mhz */
+ /* for MASTER mode, fs = 48Khz */
+ { 12000000, 3072000, 0xF3, 0x12, 0x7, 1}, /* MCLK=12Mhz */
+ { 13000000, 3072000, 0xE8, 0xFD, 0x5, 1}, /* MCLK=13Mhz */
+ { 13500000, 3072000, 0xE4, 0x82, 0x3, 1}, /* MCLK=13.5Mhz */
+ { 14400000, 3072000, 0xDD, 0x3A, 0x0, 1}, /* MCLK=14.4Mhz */
+ { 19200000, 3072000, 0xC1, 0xEB, 0x8, 1}, /* MCLK=19.2Mhz */
+ { 19680000, 3072000, 0xBF, 0xEC, 0x0, 1}, /* MCLK=19.68Mhz */
+ { 19800000, 3072000, 0xBF, 0x70, 0x0, 1}, /* MCLK=19.8Mhz */
+ /* for SLAVE mode with SRM */
+ { 12000000, 2822400, 0xED, 0xBF, 0x5, 0}, /* MCLK=12Mhz */
+ { 13000000, 2822400, 0xE4, 0x13, 0x0, 0}, /* MCLK=13Mhz */
+ { 13500000, 2822400, 0xDF, 0xC6, 0x8, 0}, /* MCLK=13.5Mhz */
+ { 14400000, 2822400, 0xD8, 0xCA, 0x1, 0}, /* MCLK=14.4Mhz */
+ { 19200000, 2822400, 0xBE, 0x97, 0x9, 0}, /* MCLK=19.2Mhz */
+ { 19680000, 2822400, 0xBC, 0xAC, 0xD, 0}, /* MCLK=19.68Mhz */
+ { 19800000, 2822400, 0xBC, 0x35, 0xE, 0}, /* MCLK=19.8Mhz */
+};
+
+enum clk_src {
+ DA7210_CLKSRC_MCLK
+};
+
#define DA7210_VERSION "0.0.1"
/*
@@ -628,9 +676,12 @@ static const struct snd_soc_dapm_route da7210_audio_map[] = {
/* Codec private data */
struct da7210_priv {
struct regmap *regmap;
+ unsigned int mclk_rate;
+ int master;
};
static struct reg_default da7210_reg_defaults[] = {
+ { 0x00, 0x00 },
{ 0x01, 0x11 },
{ 0x03, 0x00 },
{ 0x04, 0x00 },
@@ -713,10 +764,10 @@ static int da7210_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;
+ struct snd_soc_codec *codec = dai->codec;
+ struct da7210_priv *da7210 = snd_soc_codec_get_drvdata(codec);
u32 dai_cfg1;
- u32 fs, bypass;
+ u32 fs, sysclk;
/* set DAI source to Left and Right ADC */
snd_soc_write(codec, DA7210_DAI_SRC_SEL,
@@ -749,43 +800,43 @@ static int da7210_hw_params(struct snd_pcm_substream *substream,
switch (params_rate(params)) {
case 8000:
fs = DA7210_PLL_FS_8000;
- bypass = DA7210_PLL_BYP;
+ sysclk = 3072000;
break;
case 11025:
fs = DA7210_PLL_FS_11025;
- bypass = 0;
+ sysclk = 2822400;
break;
case 12000:
fs = DA7210_PLL_FS_12000;
- bypass = DA7210_PLL_BYP;
+ sysclk = 3072000;
break;
case 16000:
fs = DA7210_PLL_FS_16000;
- bypass = DA7210_PLL_BYP;
+ sysclk = 3072000;
break;
case 22050:
fs = DA7210_PLL_FS_22050;
- bypass = 0;
+ sysclk = 2822400;
break;
case 32000:
fs = DA7210_PLL_FS_32000;
- bypass = DA7210_PLL_BYP;
+ sysclk = 3072000;
break;
case 44100:
fs = DA7210_PLL_FS_44100;
- bypass = 0;
+ sysclk = 2822400;
break;
case 48000:
fs = DA7210_PLL_FS_48000;
- bypass = DA7210_PLL_BYP;
+ sysclk = 3072000;
break;
case 88200:
fs = DA7210_PLL_FS_88200;
- bypass = 0;
+ sysclk = 2822400;
break;
case 96000:
fs = DA7210_PLL_FS_96000;
- bypass = DA7210_PLL_BYP;
+ sysclk = 3072000;
break;
default:
return -EINVAL;
@@ -795,8 +846,26 @@ static int da7210_hw_params(struct snd_pcm_substream *substream,
snd_soc_update_bits(codec, DA7210_STARTUP1, DA7210_SC_MST_EN, 0);
snd_soc_update_bits(codec, DA7210_PLL, DA7210_PLL_FS_MASK, fs);
- snd_soc_update_bits(codec, DA7210_PLL_DIV3, DA7210_PLL_BYP, bypass);
+ if (da7210->mclk_rate && (da7210->mclk_rate != sysclk)) {
+ /* PLL mode, disable PLL bypass */
+ snd_soc_update_bits(codec, DA7210_PLL_DIV3, DA7210_PLL_BYP, 0);
+
+ if (!da7210->master) {
+ /* PLL slave mode, also enable SRM */
+ snd_soc_update_bits(codec, DA7210_PLL,
+ (DA7210_MCLK_SRM_EN |
+ DA7210_MCLK_DET_EN),
+ (DA7210_MCLK_SRM_EN |
+ DA7210_MCLK_DET_EN));
+ }
+ } else {
+ /* PLL bypass mode, enable PLL bypass and Auto Detection */
+ snd_soc_update_bits(codec, DA7210_PLL, DA7210_MCLK_DET_EN,
+ DA7210_MCLK_DET_EN);
+ snd_soc_update_bits(codec, DA7210_PLL_DIV3, DA7210_PLL_BYP,
+ DA7210_PLL_BYP);
+ }
/* Enable active mode */
snd_soc_update_bits(codec, DA7210_STARTUP1,
DA7210_SC_MST_EN, DA7210_SC_MST_EN);
@@ -810,17 +879,24 @@ static int da7210_hw_params(struct snd_pcm_substream *substream,
static int da7210_set_dai_fmt(struct snd_soc_dai *codec_dai, u32 fmt)
{
struct snd_soc_codec *codec = codec_dai->codec;
+ struct da7210_priv *da7210 = snd_soc_codec_get_drvdata(codec);
u32 dai_cfg1;
u32 dai_cfg3;
dai_cfg1 = 0x7f & snd_soc_read(codec, DA7210_DAI_CFG1);
dai_cfg3 = 0xfc & snd_soc_read(codec, DA7210_DAI_CFG3);
+ if ((snd_soc_read(codec, DA7210_PLL) & DA7210_PLL_EN) &&
+ (!(snd_soc_read(codec, DA7210_PLL_DIV3) & DA7210_PLL_BYP)))
+ return -EINVAL;
+
switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
case SND_SOC_DAIFMT_CBM_CFM:
+ da7210->master = 1;
dai_cfg1 |= DA7210_DAI_MODE_MASTER;
break;
case SND_SOC_DAIFMT_CBS_CFS:
+ da7210->master = 0;
dai_cfg1 |= DA7210_DAI_MODE_SLAVE;
break;
default:
@@ -872,10 +948,101 @@ static int da7210_mute(struct snd_soc_dai *dai, int mute)
#define DA7210_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
+static int da7210_set_dai_sysclk(struct snd_soc_dai *codec_dai,
+ int clk_id, unsigned int freq, int dir)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ struct da7210_priv *da7210 = snd_soc_codec_get_drvdata(codec);
+
+ switch (clk_id) {
+ case DA7210_CLKSRC_MCLK:
+ switch (freq) {
+ case 12000000:
+ case 13000000:
+ case 13500000:
+ case 14400000:
+ case 19200000:
+ case 19680000:
+ case 19800000:
+ da7210->mclk_rate = freq;
+ return 0;
+ default:
+ dev_err(codec_dai->dev, "Unsupported MCLK value %d\n",
+ freq);
+ return -EINVAL;
+ }
+ break;
+ default:
+ dev_err(codec_dai->dev, "Unknown clock source %d\n", clk_id);
+ return -EINVAL;
+ }
+}
+
+/**
+ * da7210_set_dai_pll :Configure the codec PLL
+ * @param codec_dai : pointer to codec DAI
+ * @param pll_id : da7210 has only one pll, so pll_id is always zero
+ * @param fref : MCLK frequency, should be < 20MHz
+ * @param fout : FsDM value, Refer page 44 & 45 of datasheet
+ * @return int : Zero for success, negative error code for error
+ *
+ * Note: Supported PLL input frequencies are 12MHz, 13MHz, 13.5MHz, 14.4MHz,
+ * 19.2MHz, 19.6MHz and 19.8MHz
+ */
+static int da7210_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id,
+ int source, unsigned int fref, unsigned int fout)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ struct da7210_priv *da7210 = snd_soc_codec_get_drvdata(codec);
+
+ u8 pll_div1, pll_div2, pll_div3, cnt;
+
+ /* In slave mode, there is only one set of divisors */
+ if (!da7210->master)
+ fout = 2822400;
+
+ /* Search pll div array for correct divisors */
+ for (cnt = 0; cnt < ARRAY_SIZE(da7210_pll_div); cnt++) {
+ /* check fref, mode and fout */
+ if ((fref == da7210_pll_div[cnt].fref) &&
+ (da7210->master == da7210_pll_div[cnt].mode) &&
+ (fout == da7210_pll_div[cnt].fout)) {
+ /* all match, pick up divisors */
+ pll_div1 = da7210_pll_div[cnt].div1;
+ pll_div2 = da7210_pll_div[cnt].div2;
+ pll_div3 = da7210_pll_div[cnt].div3;
+ break;
+ }
+ }
+ if (cnt >= ARRAY_SIZE(da7210_pll_div))
+ goto err;
+
+ /* Disable active mode */
+ snd_soc_update_bits(codec, DA7210_STARTUP1, DA7210_SC_MST_EN, 0);
+ /* Write PLL dividers */
+ snd_soc_write(codec, DA7210_PLL_DIV1, pll_div1);
+ snd_soc_write(codec, DA7210_PLL_DIV2, pll_div2);
+ snd_soc_update_bits(codec, DA7210_PLL_DIV3,
+ DA7210_PLL_DIV_L_MASK, pll_div3);
+
+ /* Enable PLL */
+ snd_soc_update_bits(codec, DA7210_PLL, DA7210_PLL_EN, DA7210_PLL_EN);
+
+ /* Enable active mode */
+ snd_soc_update_bits(codec, DA7210_STARTUP1, DA7210_SC_MST_EN,
+ DA7210_SC_MST_EN);
+ return 0;
+err:
+ dev_err(codec_dai->dev, "Unsupported PLL input frequency %d\n", fref);
+ return -EINVAL;
+}
+
/* DAI operations */
static const struct snd_soc_dai_ops da7210_dai_ops = {
.hw_params = da7210_hw_params,
.set_fmt = da7210_set_dai_fmt,
+ .set_sysclk = da7210_set_dai_sysclk,
+ .set_pll = da7210_set_dai_pll,
.digital_mute = da7210_mute,
};
@@ -915,24 +1082,11 @@ static int da7210_probe(struct snd_soc_codec *codec)
return ret;
}
- /* FIXME
- *
- * This driver use fixed value here
- * And below settings expects MCLK = 12.288MHz
- *
- * When you select different MCLK, please check...
- * DA7210_PLL_DIV1 val
- * DA7210_PLL_DIV2 val
- * DA7210_PLL_DIV3 val
- * DA7210_PLL_DIV3 :: DA7210_MCLK_RANGExxx
- */
+ da7210->mclk_rate = 0; /* This will be set from set_sysclk() */
+ da7210->master = 0; /* This will be set from set_fmt() */
- /*
- * make sure that DA7210 use bypass mode before start up
- */
- snd_soc_write(codec, DA7210_STARTUP1, 0);
- snd_soc_write(codec, DA7210_PLL_DIV3,
- DA7210_MCLK_RANGE_10_20_MHZ | DA7210_PLL_BYP);
+ /* Enable internal regulator & bias current */
+ snd_soc_write(codec, DA7210_CONTROL, DA7210_REG_EN | DA7210_BIAS_EN);
/*
* ADC settings
@@ -1007,34 +1161,13 @@ static int da7210_probe(struct snd_soc_codec *codec)
/* Enable Aux2 */
snd_soc_write(codec, DA7210_AUX2, DA7210_AUX2_EN);
+ /* Set PLL Master clock range 10-20 MHz, enable PLL bypass */
+ snd_soc_write(codec, DA7210_PLL_DIV3, DA7210_MCLK_RANGE_10_20_MHZ |
+ DA7210_PLL_BYP);
+
/* Diable PLL and bypass it */
snd_soc_write(codec, DA7210_PLL, DA7210_PLL_FS_48000);
- /*
- * If 48kHz sound came, it use bypass mode,
- * and when it is 44.1kHz, it use PLL.
- *
- * This time, this driver sets PLL always ON
- * and controls bypass/PLL mode by switching
- * DA7210_PLL_DIV3 :: DA7210_PLL_BYP bit.
- * see da7210_hw_params
- */
- snd_soc_write(codec, DA7210_PLL_DIV1, 0xE5); /* MCLK = 12.288MHz */
- snd_soc_write(codec, DA7210_PLL_DIV2, 0x99);
- snd_soc_write(codec, DA7210_PLL_DIV3, 0x0A |
- DA7210_MCLK_RANGE_10_20_MHZ | DA7210_PLL_BYP);
- snd_soc_update_bits(codec, DA7210_PLL, DA7210_PLL_EN, DA7210_PLL_EN);
-
- /* As suggested by Dialog */
- /* unlock */
- regmap_write(da7210->regmap, DA7210_A_HID_UNLOCK, 0x8B);
- regmap_write(da7210->regmap, DA7210_A_TEST_UNLOCK, 0xB4);
- regmap_write(da7210->regmap, DA7210_A_PLL1, 0x01);
- regmap_write(da7210->regmap, DA7210_A_CP_MODE, 0x7C);
- /* re-lock */
- regmap_write(da7210->regmap, DA7210_A_HID_UNLOCK, 0x00);
- regmap_write(da7210->regmap, DA7210_A_TEST_UNLOCK, 0x00);
-
/* Activate all enabled subsystem */
snd_soc_write(codec, DA7210_STARTUP1, DA7210_SC_MST_EN);
@@ -1055,7 +1188,26 @@ static struct snd_soc_codec_driver soc_codec_dev_da7210 = {
.num_dapm_routes = ARRAY_SIZE(da7210_audio_map),
};
-static struct regmap_config da7210_regmap = {
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+
+static struct reg_default da7210_regmap_i2c_patch[] = {
+
+ /* System controller master disable */
+ { DA7210_STARTUP1, 0x00 },
+ /* Set PLL Master clock range 10-20 MHz */
+ { DA7210_PLL_DIV3, DA7210_MCLK_RANGE_10_20_MHZ },
+
+ /* to unlock */
+ { DA7210_A_HID_UNLOCK, 0x8B},
+ { DA7210_A_TEST_UNLOCK, 0xB4},
+ { DA7210_A_PLL1, 0x01},
+ { DA7210_A_CP_MODE, 0x7C},
+ /* to re-lock */
+ { DA7210_A_HID_UNLOCK, 0x00},
+ { DA7210_A_TEST_UNLOCK, 0x00},
+};
+
+static const struct regmap_config da7210_regmap_config_i2c = {
.reg_bits = 8,
.val_bits = 8,
@@ -1066,7 +1218,6 @@ static struct regmap_config da7210_regmap = {
.cache_type = REGCACHE_RBTREE,
};
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
static int __devinit da7210_i2c_probe(struct i2c_client *i2c,
const struct i2c_device_id *id)
{
@@ -1080,13 +1231,18 @@ static int __devinit da7210_i2c_probe(struct i2c_client *i2c,
i2c_set_clientdata(i2c, da7210);
- da7210->regmap = regmap_init_i2c(i2c, &da7210_regmap);
+ da7210->regmap = regmap_init_i2c(i2c, &da7210_regmap_config_i2c);
if (IS_ERR(da7210->regmap)) {
ret = PTR_ERR(da7210->regmap);
dev_err(&i2c->dev, "regmap_init() failed: %d\n", ret);
return ret;
}
+ ret = regmap_register_patch(da7210->regmap, da7210_regmap_i2c_patch,
+ ARRAY_SIZE(da7210_regmap_i2c_patch));
+ if (ret != 0)
+ dev_warn(&i2c->dev, "Failed to apply regmap patch: %d\n", ret);
+
ret = snd_soc_register_codec(&i2c->dev,
&soc_codec_dev_da7210, &da7210_dai, 1);
if (ret < 0) {
@@ -1119,7 +1275,7 @@ MODULE_DEVICE_TABLE(i2c, da7210_i2c_id);
/* I2C codec control layer */
static struct i2c_driver da7210_i2c_driver = {
.driver = {
- .name = "da7210-codec",
+ .name = "da7210",
.owner = THIS_MODULE,
},
.probe = da7210_i2c_probe,
@@ -1128,12 +1284,112 @@ static struct i2c_driver da7210_i2c_driver = {
};
#endif
+#if defined(CONFIG_SPI_MASTER)
+
+static struct reg_default da7210_regmap_spi_patch[] = {
+ /* Dummy read to give two pulses over nCS for SPI */
+ { DA7210_AUX2, 0x00 },
+ { DA7210_AUX2, 0x00 },
+
+ /* System controller master disable */
+ { DA7210_STARTUP1, 0x00 },
+ /* Set PLL Master clock range 10-20 MHz */
+ { DA7210_PLL_DIV3, DA7210_MCLK_RANGE_10_20_MHZ },
+
+ /* to set PAGE1 of SPI register space */
+ { DA7210_PAGE_CONTROL, 0x80 },
+ /* to unlock */
+ { DA7210_A_HID_UNLOCK, 0x8B},
+ { DA7210_A_TEST_UNLOCK, 0xB4},
+ { DA7210_A_PLL1, 0x01},
+ { DA7210_A_CP_MODE, 0x7C},
+ /* to re-lock */
+ { DA7210_A_HID_UNLOCK, 0x00},
+ { DA7210_A_TEST_UNLOCK, 0x00},
+ /* to set back PAGE0 of SPI register space */
+ { DA7210_PAGE_CONTROL, 0x00 },
+};
+
+static const struct regmap_config da7210_regmap_config_spi = {
+ .reg_bits = 8,
+ .val_bits = 8,
+ .read_flag_mask = 0x01,
+ .write_flag_mask = 0x00,
+
+ .reg_defaults = da7210_reg_defaults,
+ .num_reg_defaults = ARRAY_SIZE(da7210_reg_defaults),
+ .volatile_reg = da7210_volatile_register,
+ .readable_reg = da7210_readable_register,
+ .cache_type = REGCACHE_RBTREE,
+};
+
+static int __devinit da7210_spi_probe(struct spi_device *spi)
+{
+ struct da7210_priv *da7210;
+ int ret;
+
+ da7210 = devm_kzalloc(&spi->dev, sizeof(struct da7210_priv),
+ GFP_KERNEL);
+ if (!da7210)
+ return -ENOMEM;
+
+ spi_set_drvdata(spi, da7210);
+ da7210->regmap = devm_regmap_init_spi(spi, &da7210_regmap_config_spi);
+ if (IS_ERR(da7210->regmap)) {
+ ret = PTR_ERR(da7210->regmap);
+ dev_err(&spi->dev, "Failed to register regmap: %d\n", ret);
+ return ret;
+ }
+
+ ret = regmap_register_patch(da7210->regmap, da7210_regmap_spi_patch,
+ ARRAY_SIZE(da7210_regmap_spi_patch));
+ if (ret != 0)
+ dev_warn(&spi->dev, "Failed to apply regmap patch: %d\n", ret);
+
+ ret = snd_soc_register_codec(&spi->dev,
+ &soc_codec_dev_da7210, &da7210_dai, 1);
+ if (ret < 0)
+ goto err_regmap;
+
+ return ret;
+
+err_regmap:
+ regmap_exit(da7210->regmap);
+
+ return ret;
+}
+
+static int __devexit da7210_spi_remove(struct spi_device *spi)
+{
+ struct da7210_priv *da7210 = spi_get_drvdata(spi);
+ snd_soc_unregister_codec(&spi->dev);
+ regmap_exit(da7210->regmap);
+ return 0;
+}
+
+static struct spi_driver da7210_spi_driver = {
+ .driver = {
+ .name = "da7210",
+ .owner = THIS_MODULE,
+ },
+ .probe = da7210_spi_probe,
+ .remove = __devexit_p(da7210_spi_remove)
+};
+#endif
+
static int __init da7210_modinit(void)
{
int ret = 0;
#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
ret = i2c_add_driver(&da7210_i2c_driver);
#endif
+#if defined(CONFIG_SPI_MASTER)
+ ret = spi_register_driver(&da7210_spi_driver);
+ if (ret) {
+ printk(KERN_ERR "Failed to register da7210 SPI driver: %d\n",
+ ret);
+ }
+#endif
return ret;
}
module_init(da7210_modinit);
@@ -1143,6 +1399,9 @@ static void __exit da7210_exit(void)
#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
i2c_del_driver(&da7210_i2c_driver);
#endif
+#if defined(CONFIG_SPI_MASTER)
+ spi_unregister_driver(&da7210_spi_driver);
+#endif
}
module_exit(da7210_exit);
diff --git a/sound/soc/codecs/jz4740.c b/sound/soc/codecs/jz4740.c
index 4624e752a18..85d9cabe6d5 100644
--- a/sound/soc/codecs/jz4740.c
+++ b/sound/soc/codecs/jz4740.c
@@ -164,8 +164,7 @@ static int jz4740_codec_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params, struct snd_soc_dai *dai)
{
uint32_t val;
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_codec *codec =rtd->codec;
+ struct snd_soc_codec *codec = dai->codec;
switch (params_rate(params)) {
case 8000:
diff --git a/sound/soc/codecs/lm49453.c b/sound/soc/codecs/lm49453.c
new file mode 100644
index 00000000000..802b9f176b1
--- /dev/null
+++ b/sound/soc/codecs/lm49453.c
@@ -0,0 +1,1550 @@
+/*
+ * lm49453.c - LM49453 ALSA Soc Audio 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/wm8350.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/i2c.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/soc-dapm.h>
+#include <sound/tlv.h>
+#include <sound/jack.h>
+#include <sound/initval.h>
+#include <asm/div64.h>
+#include "lm49453.h"
+
+static struct reg_default lm49453_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, 0x00 },
+ { 22, 0x00 },
+ { 23, 0x00 },
+ { 32, 0x00 },
+ { 33, 0x00 },
+ { 35, 0x00 },
+ { 36, 0x00 },
+ { 37, 0x00 },
+ { 46, 0x00 },
+ { 48, 0x00 },
+ { 49, 0x00 },
+ { 51, 0x00 },
+ { 56, 0x00 },
+ { 58, 0x00 },
+ { 59, 0x00 },
+ { 60, 0x00 },
+ { 61, 0x00 },
+ { 62, 0x00 },
+ { 63, 0x00 },
+ { 64, 0x00 },
+ { 65, 0x00 },
+ { 66, 0x00 },
+ { 67, 0x00 },
+ { 68, 0x00 },
+ { 69, 0x00 },
+ { 70, 0x00 },
+ { 71, 0x00 },
+ { 72, 0x00 },
+ { 73, 0x00 },
+ { 74, 0x00 },
+ { 75, 0x00 },
+ { 76, 0x00 },
+ { 77, 0x00 },
+ { 78, 0x00 },
+ { 79, 0x00 },
+ { 80, 0x00 },
+ { 81, 0x00 },
+ { 82, 0x00 },
+ { 83, 0x00 },
+ { 85, 0x00 },
+ { 85, 0x00 },
+ { 86, 0x00 },
+ { 87, 0x00 },
+ { 88, 0x00 },
+ { 89, 0x00 },
+ { 90, 0x00 },
+ { 91, 0x00 },
+ { 92, 0x00 },
+ { 93, 0x00 },
+ { 94, 0x00 },
+ { 95, 0x00 },
+ { 96, 0x01 },
+ { 97, 0x00 },
+ { 98, 0x00 },
+ { 99, 0x00 },
+ { 100, 0x00 },
+ { 101, 0x00 },
+ { 102, 0x00 },
+ { 103, 0x01 },
+ { 105, 0x01 },
+ { 106, 0x00 },
+ { 107, 0x01 },
+ { 107, 0x00 },
+ { 108, 0x00 },
+ { 109, 0x00 },
+ { 110, 0x00 },
+ { 111, 0x02 },
+ { 112, 0x02 },
+ { 113, 0x00 },
+ { 121, 0x80 },
+ { 122, 0xBB },
+ { 123, 0x80 },
+ { 124, 0xBB },
+ { 128, 0x00 },
+ { 130, 0x00 },
+ { 131, 0x00 },
+ { 132, 0x00 },
+ { 133, 0x0A },
+ { 134, 0x0A },
+ { 135, 0x0A },
+ { 136, 0x0F },
+ { 137, 0x00 },
+ { 138, 0x73 },
+ { 139, 0x33 },
+ { 140, 0x73 },
+ { 141, 0x33 },
+ { 142, 0x73 },
+ { 143, 0x33 },
+ { 144, 0x73 },
+ { 145, 0x33 },
+ { 146, 0x73 },
+ { 147, 0x33 },
+ { 148, 0x73 },
+ { 149, 0x33 },
+ { 150, 0x73 },
+ { 151, 0x33 },
+ { 152, 0x00 },
+ { 153, 0x00 },
+ { 154, 0x00 },
+ { 155, 0x00 },
+ { 176, 0x00 },
+ { 177, 0x00 },
+ { 178, 0x00 },
+ { 179, 0x00 },
+ { 180, 0x00 },
+ { 181, 0x00 },
+ { 182, 0x00 },
+ { 183, 0x00 },
+ { 184, 0x00 },
+ { 185, 0x00 },
+ { 186, 0x00 },
+ { 189, 0x00 },
+ { 188, 0x00 },
+ { 194, 0x00 },
+ { 195, 0x00 },
+ { 196, 0x00 },
+ { 197, 0x00 },
+ { 200, 0x00 },
+ { 201, 0x00 },
+ { 202, 0x00 },
+ { 203, 0x00 },
+ { 204, 0x00 },
+ { 205, 0x00 },
+ { 208, 0x00 },
+ { 209, 0x00 },
+ { 210, 0x00 },
+ { 211, 0x00 },
+ { 213, 0x00 },
+ { 214, 0x00 },
+ { 215, 0x00 },
+ { 216, 0x00 },
+ { 217, 0x00 },
+ { 218, 0x00 },
+ { 219, 0x00 },
+ { 221, 0x00 },
+ { 222, 0x00 },
+ { 224, 0x00 },
+ { 225, 0x00 },
+ { 226, 0x00 },
+ { 227, 0x00 },
+ { 228, 0x00 },
+ { 229, 0x00 },
+ { 230, 0x13 },
+ { 231, 0x00 },
+ { 232, 0x80 },
+ { 233, 0x0C },
+ { 234, 0xDD },
+ { 235, 0x00 },
+ { 236, 0x04 },
+ { 237, 0x00 },
+ { 238, 0x00 },
+ { 239, 0x00 },
+ { 240, 0x00 },
+ { 241, 0x00 },
+ { 242, 0x00 },
+ { 243, 0x00 },
+ { 244, 0x00 },
+ { 245, 0x00 },
+ { 248, 0x00 },
+ { 249, 0x00 },
+ { 254, 0x00 },
+ { 255, 0x00 },
+};
+
+/* codec private data */
+struct lm49453_priv {
+ struct regmap *regmap;
+ int fs_rate;
+};
+
+/* capture path controls */
+
+static const char *lm49453_mic2mode_text[] = {"Single Ended", "Differential"};
+
+static const SOC_ENUM_SINGLE_DECL(lm49453_mic2mode_enum, LM49453_P0_MICR_REG, 5,
+ lm49453_mic2mode_text);
+
+static const char *lm49453_dmic_cfg_text[] = {"DMICDAT1", "DMICDAT2"};
+
+static const SOC_ENUM_SINGLE_DECL(lm49453_dmic12_cfg_enum,
+ LM49453_P0_DIGITAL_MIC1_CONFIG_REG,
+ 7, lm49453_dmic_cfg_text);
+
+static const SOC_ENUM_SINGLE_DECL(lm49453_dmic34_cfg_enum,
+ LM49453_P0_DIGITAL_MIC2_CONFIG_REG,
+ 7, lm49453_dmic_cfg_text);
+
+/* MUX Controls */
+static const char *lm49453_adcl_mux_text[] = { "MIC1", "Aux_L" };
+
+static const char *lm49453_adcr_mux_text[] = { "MIC2", "Aux_R" };
+
+static const struct soc_enum lm49453_adcl_enum =
+ SOC_ENUM_SINGLE(LM49453_P0_ANALOG_MIXER_ADC_REG, 0,
+ ARRAY_SIZE(lm49453_adcl_mux_text),
+ lm49453_adcl_mux_text);
+
+static const struct soc_enum lm49453_adcr_enum =
+ SOC_ENUM_SINGLE(LM49453_P0_ANALOG_MIXER_ADC_REG, 1,
+ ARRAY_SIZE(lm49453_adcr_mux_text),
+ lm49453_adcr_mux_text);
+
+static const struct snd_kcontrol_new lm49453_adcl_mux_control =
+ SOC_DAPM_ENUM("ADC Left Mux", lm49453_adcl_enum);
+
+static const struct snd_kcontrol_new lm49453_adcr_mux_control =
+ SOC_DAPM_ENUM("ADC Right Mux", lm49453_adcr_enum);
+
+static const struct snd_kcontrol_new lm49453_headset_left_mixer[] = {
+SOC_DAPM_SINGLE("Port1_1 Switch", LM49453_P0_DACHPL1_REG, 0, 1, 0),
+SOC_DAPM_SINGLE("Port1_2 Switch", LM49453_P0_DACHPL1_REG, 1, 1, 0),
+SOC_DAPM_SINGLE("Port1_3 Switch", LM49453_P0_DACHPL1_REG, 2, 1, 0),
+SOC_DAPM_SINGLE("Port1_4 Switch", LM49453_P0_DACHPL1_REG, 3, 1, 0),
+SOC_DAPM_SINGLE("Port1_5 Switch", LM49453_P0_DACHPL1_REG, 4, 1, 0),
+SOC_DAPM_SINGLE("Port1_6 Switch", LM49453_P0_DACHPL1_REG, 5, 1, 0),
+SOC_DAPM_SINGLE("Port1_7 Switch", LM49453_P0_DACHPL1_REG, 6, 1, 0),
+SOC_DAPM_SINGLE("Port1_8 Switch", LM49453_P0_DACHPL1_REG, 7, 1, 0),
+SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_DACHPL2_REG, 0, 1, 0),
+SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_DACHPL2_REG, 1, 1, 0),
+SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_DACHPL2_REG, 2, 1, 0),
+SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_DACHPL2_REG, 3, 1, 0),
+SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_DACHPL2_REG, 4, 1, 0),
+SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_DACHPL2_REG, 5, 1, 0),
+SOC_DAPM_SINGLE("Port2_1 Switch", LM49453_P0_DACHPL2_REG, 6, 1, 0),
+SOC_DAPM_SINGLE("Port2_2 Switch", LM49453_P0_DACHPL2_REG, 7, 1, 0),
+SOC_DAPM_SINGLE("Sidetone Switch", LM49453_P0_STN_SEL_REG, 0, 0, 0),
+};
+
+static const struct snd_kcontrol_new lm49453_headset_right_mixer[] = {
+SOC_DAPM_SINGLE("Port1_1 Switch", LM49453_P0_DACHPR1_REG, 0, 1, 0),
+SOC_DAPM_SINGLE("Port1_2 Switch", LM49453_P0_DACHPR1_REG, 1, 1, 0),
+SOC_DAPM_SINGLE("Port1_3 Switch", LM49453_P0_DACHPR1_REG, 2, 1, 0),
+SOC_DAPM_SINGLE("Port1_4 Switch", LM49453_P0_DACHPR1_REG, 3, 1, 0),
+SOC_DAPM_SINGLE("Port1_5 Switch", LM49453_P0_DACHPR1_REG, 4, 1, 0),
+SOC_DAPM_SINGLE("Port1_6 Switch", LM49453_P0_DACHPR1_REG, 5, 1, 0),
+SOC_DAPM_SINGLE("Port1_7 Switch", LM49453_P0_DACHPR1_REG, 6, 1, 0),
+SOC_DAPM_SINGLE("Port1_8 Switch", LM49453_P0_DACHPR1_REG, 7, 1, 0),
+SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_DACHPR2_REG, 0, 1, 0),
+SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_DACHPR2_REG, 1, 1, 0),
+SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_DACHPR2_REG, 2, 1, 0),
+SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_DACHPR2_REG, 3, 1, 0),
+SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_DACHPR2_REG, 4, 1, 0),
+SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_DACHPR2_REG, 5, 1, 0),
+SOC_DAPM_SINGLE("Port2_1 Switch", LM49453_P0_DACHPR2_REG, 6, 1, 0),
+SOC_DAPM_SINGLE("Port2_2 Switch", LM49453_P0_DACHPR2_REG, 7, 1, 0),
+SOC_DAPM_SINGLE("Sidetone Switch", LM49453_P0_STN_SEL_REG, 1, 0, 0),
+};
+
+static const struct snd_kcontrol_new lm49453_speaker_left_mixer[] = {
+SOC_DAPM_SINGLE("Port1_1 Switch", LM49453_P0_DACLSL1_REG, 0, 1, 0),
+SOC_DAPM_SINGLE("Port1_2 Switch", LM49453_P0_DACLSL1_REG, 1, 1, 0),
+SOC_DAPM_SINGLE("Port1_3 Switch", LM49453_P0_DACLSL1_REG, 2, 1, 0),
+SOC_DAPM_SINGLE("Port1_4 Switch", LM49453_P0_DACLSL1_REG, 3, 1, 0),
+SOC_DAPM_SINGLE("Port1_5 Switch", LM49453_P0_DACLSL1_REG, 4, 1, 0),
+SOC_DAPM_SINGLE("Port1_6 Switch", LM49453_P0_DACLSL1_REG, 5, 1, 0),
+SOC_DAPM_SINGLE("Port1_7 Switch", LM49453_P0_DACLSL1_REG, 6, 1, 0),
+SOC_DAPM_SINGLE("Port1_8 Switch", LM49453_P0_DACLSL1_REG, 7, 1, 0),
+SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_DACLSL2_REG, 0, 1, 0),
+SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_DACLSL2_REG, 1, 1, 0),
+SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_DACLSL2_REG, 2, 1, 0),
+SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_DACLSL2_REG, 3, 1, 0),
+SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_DACLSL2_REG, 4, 1, 0),
+SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_DACLSL2_REG, 5, 1, 0),
+SOC_DAPM_SINGLE("Port2_1 Switch", LM49453_P0_DACLSL2_REG, 6, 1, 0),
+SOC_DAPM_SINGLE("Port2_2 Switch", LM49453_P0_DACLSL2_REG, 7, 1, 0),
+SOC_DAPM_SINGLE("Sidetone Switch", LM49453_P0_STN_SEL_REG, 2, 0, 0),
+};
+
+static const struct snd_kcontrol_new lm49453_speaker_right_mixer[] = {
+SOC_DAPM_SINGLE("Port1_1 Switch", LM49453_P0_DACLSR1_REG, 0, 1, 0),
+SOC_DAPM_SINGLE("Port1_2 Switch", LM49453_P0_DACLSR1_REG, 1, 1, 0),
+SOC_DAPM_SINGLE("Port1_3 Switch", LM49453_P0_DACLSR1_REG, 2, 1, 0),
+SOC_DAPM_SINGLE("Port1_4 Switch", LM49453_P0_DACLSR1_REG, 3, 1, 0),
+SOC_DAPM_SINGLE("Port1_5 Switch", LM49453_P0_DACLSR1_REG, 4, 1, 0),
+SOC_DAPM_SINGLE("Port1_6 Switch", LM49453_P0_DACLSR1_REG, 5, 1, 0),
+SOC_DAPM_SINGLE("Port1_7 Switch", LM49453_P0_DACLSR1_REG, 6, 1, 0),
+SOC_DAPM_SINGLE("Port1_8 Switch", LM49453_P0_DACLSR1_REG, 7, 1, 0),
+SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_DACLSR2_REG, 0, 1, 0),
+SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_DACLSR2_REG, 1, 1, 0),
+SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_DACLSR2_REG, 2, 1, 0),
+SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_DACLSR2_REG, 3, 1, 0),
+SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_DACLSR2_REG, 4, 1, 0),
+SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_DACLSR2_REG, 5, 1, 0),
+SOC_DAPM_SINGLE("Port2_1 Switch", LM49453_P0_DACLSR2_REG, 6, 1, 0),
+SOC_DAPM_SINGLE("Port2_2 Switch", LM49453_P0_DACLSR2_REG, 7, 1, 0),
+SOC_DAPM_SINGLE("Sidetone Switch", LM49453_P0_STN_SEL_REG, 3, 0, 0),
+};
+
+static const struct snd_kcontrol_new lm49453_haptic_left_mixer[] = {
+SOC_DAPM_SINGLE("Port1_1 Switch", LM49453_P0_DACHAL1_REG, 0, 1, 0),
+SOC_DAPM_SINGLE("Port1_2 Switch", LM49453_P0_DACHAL1_REG, 1, 1, 0),
+SOC_DAPM_SINGLE("Port1_3 Switch", LM49453_P0_DACHAL1_REG, 2, 1, 0),
+SOC_DAPM_SINGLE("Port1_4 Switch", LM49453_P0_DACHAL1_REG, 3, 1, 0),
+SOC_DAPM_SINGLE("Port1_5 Switch", LM49453_P0_DACHAL1_REG, 4, 1, 0),
+SOC_DAPM_SINGLE("Port1_6 Switch", LM49453_P0_DACHAL1_REG, 5, 1, 0),
+SOC_DAPM_SINGLE("Port1_7 Switch", LM49453_P0_DACHAL1_REG, 6, 1, 0),
+SOC_DAPM_SINGLE("Port1_8 Switch", LM49453_P0_DACHAL1_REG, 7, 1, 0),
+SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_DACHAL2_REG, 0, 1, 0),
+SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_DACHAL2_REG, 1, 1, 0),
+SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_DACHAL2_REG, 2, 1, 0),
+SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_DACHAL2_REG, 3, 1, 0),
+SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_DACHAL2_REG, 4, 1, 0),
+SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_DACHAL2_REG, 5, 1, 0),
+SOC_DAPM_SINGLE("Port2_1 Switch", LM49453_P0_DACHAL2_REG, 6, 1, 0),
+SOC_DAPM_SINGLE("Port2_2 Switch", LM49453_P0_DACHAL2_REG, 7, 1, 0),
+SOC_DAPM_SINGLE("Sidetone Switch", LM49453_P0_STN_SEL_REG, 4, 0, 0),
+};
+
+static const struct snd_kcontrol_new lm49453_haptic_right_mixer[] = {
+SOC_DAPM_SINGLE("Port1_1 Switch", LM49453_P0_DACHAR1_REG, 0, 1, 0),
+SOC_DAPM_SINGLE("Port1_2 Switch", LM49453_P0_DACHAR1_REG, 1, 1, 0),
+SOC_DAPM_SINGLE("Port1_3 Switch", LM49453_P0_DACHAR1_REG, 2, 1, 0),
+SOC_DAPM_SINGLE("Port1_4 Switch", LM49453_P0_DACHAR1_REG, 3, 1, 0),
+SOC_DAPM_SINGLE("Port1_5 Switch", LM49453_P0_DACHAR1_REG, 4, 1, 0),
+SOC_DAPM_SINGLE("Port1_6 Switch", LM49453_P0_DACHAR1_REG, 5, 1, 0),
+SOC_DAPM_SINGLE("Port1_7 Switch", LM49453_P0_DACHAR1_REG, 6, 1, 0),
+SOC_DAPM_SINGLE("Port1_8 Switch", LM49453_P0_DACHAR1_REG, 7, 1, 0),
+SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_DACHAR2_REG, 0, 1, 0),
+SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_DACHAR2_REG, 1, 1, 0),
+SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_DACHAR2_REG, 2, 1, 0),
+SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_DACHAR2_REG, 3, 1, 0),
+SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_DACHAR2_REG, 4, 1, 0),
+SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_DACHAR2_REG, 5, 1, 0),
+SOC_DAPM_SINGLE("Port2_1 Switch", LM49453_P0_DACHAR2_REG, 6, 1, 0),
+SOC_DAPM_SINGLE("Port2_2 Switch", LM49453_P0_DACHAR2_REG, 7, 1, 0),
+SOC_DAPM_SINGLE("Sidetone Switch", LM49453_P0_STN_SEL_REG, 5, 0, 0),
+};
+
+static const struct snd_kcontrol_new lm49453_lineout_left_mixer[] = {
+SOC_DAPM_SINGLE("Port1_1 Switch", LM49453_P0_DACLOL1_REG, 0, 1, 0),
+SOC_DAPM_SINGLE("Port1_2 Switch", LM49453_P0_DACLOL1_REG, 1, 1, 0),
+SOC_DAPM_SINGLE("Port1_3 Switch", LM49453_P0_DACLOL1_REG, 2, 1, 0),
+SOC_DAPM_SINGLE("Port1_4 Switch", LM49453_P0_DACLOL1_REG, 3, 1, 0),
+SOC_DAPM_SINGLE("Port1_5 Switch", LM49453_P0_DACLOL1_REG, 4, 1, 0),
+SOC_DAPM_SINGLE("Port1_6 Switch", LM49453_P0_DACLOL1_REG, 5, 1, 0),
+SOC_DAPM_SINGLE("Port1_7 Switch", LM49453_P0_DACLOL1_REG, 6, 1, 0),
+SOC_DAPM_SINGLE("Port1_8 Switch", LM49453_P0_DACLOL1_REG, 7, 1, 0),
+SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_DACLOL2_REG, 0, 1, 0),
+SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_DACLOL2_REG, 1, 1, 0),
+SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_DACLOL2_REG, 2, 1, 0),
+SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_DACLOL2_REG, 3, 1, 0),
+SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_DACLOL2_REG, 4, 1, 0),
+SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_DACLOL2_REG, 5, 1, 0),
+SOC_DAPM_SINGLE("Port2_1 Switch", LM49453_P0_DACLOL2_REG, 6, 1, 0),
+SOC_DAPM_SINGLE("Port2_2 Switch", LM49453_P0_DACLOL2_REG, 7, 1, 0),
+SOC_DAPM_SINGLE("Sidetone Switch", LM49453_P0_STN_SEL_REG, 6, 0, 0),
+};
+
+static const struct snd_kcontrol_new lm49453_lineout_right_mixer[] = {
+SOC_DAPM_SINGLE("Port1_1 Switch", LM49453_P0_DACLOR1_REG, 0, 1, 0),
+SOC_DAPM_SINGLE("Port1_2 Switch", LM49453_P0_DACLOR1_REG, 1, 1, 0),
+SOC_DAPM_SINGLE("Port1_3 Switch", LM49453_P0_DACLOR1_REG, 2, 1, 0),
+SOC_DAPM_SINGLE("Port1_4 Switch", LM49453_P0_DACLOR1_REG, 3, 1, 0),
+SOC_DAPM_SINGLE("Port1_5 Switch", LM49453_P0_DACLOR1_REG, 4, 1, 0),
+SOC_DAPM_SINGLE("Port1_6 Switch", LM49453_P0_DACLOR1_REG, 5, 1, 0),
+SOC_DAPM_SINGLE("Port1_7 Switch", LM49453_P0_DACLOR1_REG, 6, 1, 0),
+SOC_DAPM_SINGLE("Port1_8 Switch", LM49453_P0_DACLOR1_REG, 7, 1, 0),
+SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_DACLOR2_REG, 0, 1, 0),
+SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_DACLOR2_REG, 1, 1, 0),
+SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_DACLOR2_REG, 2, 1, 0),
+SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_DACLOR2_REG, 3, 1, 0),
+SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_DACLOR2_REG, 4, 1, 0),
+SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_DACLOR2_REG, 5, 1, 0),
+SOC_DAPM_SINGLE("Port2_1 Switch", LM49453_P0_DACLOR2_REG, 6, 1, 0),
+SOC_DAPM_SINGLE("Port2_2 Switch", LM49453_P0_DACLOR2_REG, 7, 1, 0),
+SOC_DAPM_SINGLE("Sidetone Switch", LM49453_P0_STN_SEL_REG, 7, 0, 0),
+};
+
+static const struct snd_kcontrol_new lm49453_port1_tx1_mixer[] = {
+SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_PORT1_TX1_REG, 0, 1, 0),
+SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_PORT1_TX1_REG, 1, 1, 0),
+SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_PORT1_TX1_REG, 2, 1, 0),
+SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_PORT1_TX1_REG, 3, 1, 0),
+SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_PORT1_TX1_REG, 4, 1, 0),
+SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_PORT1_TX1_REG, 5, 1, 0),
+SOC_DAPM_SINGLE("Port1_1 Switch", LM49453_P0_PORT1_TX1_REG, 6, 1, 0),
+SOC_DAPM_SINGLE("Port2_1 Switch", LM49453_P0_PORT1_TX1_REG, 7, 1, 0),
+};
+
+static const struct snd_kcontrol_new lm49453_port1_tx2_mixer[] = {
+SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_PORT1_TX2_REG, 0, 1, 0),
+SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_PORT1_TX2_REG, 1, 1, 0),
+SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_PORT1_TX2_REG, 2, 1, 0),
+SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_PORT1_TX2_REG, 3, 1, 0),
+SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_PORT1_TX2_REG, 4, 1, 0),
+SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_PORT1_TX2_REG, 5, 1, 0),
+SOC_DAPM_SINGLE("Port1_2 Switch", LM49453_P0_PORT1_TX2_REG, 6, 1, 0),
+SOC_DAPM_SINGLE("Port2_2 Switch", LM49453_P0_PORT1_TX2_REG, 7, 1, 0),
+};
+
+static const struct snd_kcontrol_new lm49453_port1_tx3_mixer[] = {
+SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_PORT1_TX3_REG, 0, 1, 0),
+SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_PORT1_TX3_REG, 1, 1, 0),
+SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_PORT1_TX3_REG, 2, 1, 0),
+SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_PORT1_TX3_REG, 3, 1, 0),
+SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_PORT1_TX3_REG, 4, 1, 0),
+SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_PORT1_TX3_REG, 5, 1, 0),
+SOC_DAPM_SINGLE("Port1_3 Switch", LM49453_P0_PORT1_TX3_REG, 6, 1, 0),
+};
+
+static const struct snd_kcontrol_new lm49453_port1_tx4_mixer[] = {
+SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_PORT1_TX4_REG, 0, 1, 0),
+SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_PORT1_TX4_REG, 1, 1, 0),
+SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_PORT1_TX4_REG, 2, 1, 0),
+SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_PORT1_TX4_REG, 3, 1, 0),
+SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_PORT1_TX4_REG, 4, 1, 0),
+SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_PORT1_TX4_REG, 5, 1, 0),
+SOC_DAPM_SINGLE("Port1_4 Switch", LM49453_P0_PORT1_TX4_REG, 6, 1, 0),
+};
+
+static const struct snd_kcontrol_new lm49453_port1_tx5_mixer[] = {
+SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_PORT1_TX5_REG, 0, 1, 0),
+SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_PORT1_TX5_REG, 1, 1, 0),
+SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_PORT1_TX5_REG, 2, 1, 0),
+SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_PORT1_TX5_REG, 3, 1, 0),
+SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_PORT1_TX5_REG, 4, 1, 0),
+SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_PORT1_TX5_REG, 5, 1, 0),
+SOC_DAPM_SINGLE("Port1_5 Switch", LM49453_P0_PORT1_TX5_REG, 6, 1, 0),
+};
+
+static const struct snd_kcontrol_new lm49453_port1_tx6_mixer[] = {
+SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_PORT1_TX6_REG, 0, 1, 0),
+SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_PORT1_TX6_REG, 1, 1, 0),
+SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_PORT1_TX6_REG, 2, 1, 0),
+SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_PORT1_TX6_REG, 3, 1, 0),
+SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_PORT1_TX6_REG, 4, 1, 0),
+SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_PORT1_TX6_REG, 5, 1, 0),
+SOC_DAPM_SINGLE("Port1_6 Switch", LM49453_P0_PORT1_TX6_REG, 6, 1, 0),
+};
+
+static const struct snd_kcontrol_new lm49453_port1_tx7_mixer[] = {
+SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_PORT1_TX7_REG, 0, 1, 0),
+SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_PORT1_TX7_REG, 1, 1, 0),
+SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_PORT1_TX7_REG, 2, 1, 0),
+SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_PORT1_TX7_REG, 3, 1, 0),
+SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_PORT1_TX7_REG, 4, 1, 0),
+SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_PORT1_TX7_REG, 5, 1, 0),
+SOC_DAPM_SINGLE("Port1_7 Switch", LM49453_P0_PORT1_TX7_REG, 6, 1, 0),
+};
+
+static const struct snd_kcontrol_new lm49453_port1_tx8_mixer[] = {
+SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_PORT1_TX8_REG, 0, 1, 0),
+SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_PORT1_TX8_REG, 1, 1, 0),
+SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_PORT1_TX8_REG, 2, 1, 0),
+SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_PORT1_TX8_REG, 3, 1, 0),
+SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_PORT1_TX8_REG, 4, 1, 0),
+SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_PORT1_TX8_REG, 5, 1, 0),
+SOC_DAPM_SINGLE("Port1_8 Switch", LM49453_P0_PORT1_TX8_REG, 6, 1, 0),
+};
+
+static const struct snd_kcontrol_new lm49453_port2_tx1_mixer[] = {
+SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_PORT2_TX1_REG, 0, 1, 0),
+SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_PORT2_TX1_REG, 1, 1, 0),
+SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_PORT2_TX1_REG, 2, 1, 0),
+SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_PORT2_TX1_REG, 3, 1, 0),
+SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_PORT2_TX1_REG, 4, 1, 0),
+SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_PORT2_TX1_REG, 5, 1, 0),
+SOC_DAPM_SINGLE("Port1_1 Switch", LM49453_P0_PORT2_TX1_REG, 6, 1, 0),
+SOC_DAPM_SINGLE("Port2_1 Switch", LM49453_P0_PORT2_TX1_REG, 7, 1, 0),
+};
+
+static const struct snd_kcontrol_new lm49453_port2_tx2_mixer[] = {
+SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_PORT2_TX2_REG, 0, 1, 0),
+SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_PORT2_TX2_REG, 1, 1, 0),
+SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_PORT2_TX2_REG, 2, 1, 0),
+SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_PORT2_TX2_REG, 3, 1, 0),
+SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_PORT2_TX2_REG, 4, 1, 0),
+SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_PORT2_TX2_REG, 5, 1, 0),
+SOC_DAPM_SINGLE("Port1_2 Switch", LM49453_P0_PORT2_TX2_REG, 6, 1, 0),
+SOC_DAPM_SINGLE("Port2_2 Switch", LM49453_P0_PORT2_TX2_REG, 7, 1, 0),
+};
+
+/* TLV Declarations */
+static const DECLARE_TLV_DB_SCALE(digital_tlv, -7650, 150, 1);
+static const DECLARE_TLV_DB_SCALE(port_tlv, 0, 600, 0);
+
+static const struct snd_kcontrol_new lm49453_sidetone_mixer_controls[] = {
+/* Sidetone supports mono only */
+SOC_DAPM_SINGLE_TLV("Sidetone ADCL Volume", LM49453_P0_STN_VOL_ADCL_REG,
+ 0, 0x3F, 0, digital_tlv),
+SOC_DAPM_SINGLE_TLV("Sidetone ADCR Volume", LM49453_P0_STN_VOL_ADCR_REG,
+ 0, 0x3F, 0, digital_tlv),
+SOC_DAPM_SINGLE_TLV("Sidetone DMIC1L Volume", LM49453_P0_STN_VOL_DMIC1L_REG,
+ 0, 0x3F, 0, digital_tlv),
+SOC_DAPM_SINGLE_TLV("Sidetone DMIC1R Volume", LM49453_P0_STN_VOL_DMIC1R_REG,
+ 0, 0x3F, 0, digital_tlv),
+SOC_DAPM_SINGLE_TLV("Sidetone DMIC2L Volume", LM49453_P0_STN_VOL_DMIC2L_REG,
+ 0, 0x3F, 0, digital_tlv),
+SOC_DAPM_SINGLE_TLV("Sidetone DMIC2R Volume", LM49453_P0_STN_VOL_DMIC2R_REG,
+ 0, 0x3F, 0, digital_tlv),
+};
+
+static const struct snd_kcontrol_new lm49453_snd_controls[] = {
+ /* mic1 and mic2 supports mono only */
+ SOC_SINGLE_TLV("Mic1 Volume", LM49453_P0_ADC_LEVELL_REG, 0, 6,
+ 0, digital_tlv),
+ SOC_SINGLE_TLV("Mic2 Volume", LM49453_P0_ADC_LEVELR_REG, 0, 6,
+ 0, digital_tlv),
+
+ SOC_DOUBLE_R_TLV("DMIC1 Volume", LM49453_P0_DMIC1_LEVELL_REG,
+ LM49453_P0_DMIC1_LEVELR_REG, 0, 6, 0, digital_tlv),
+ SOC_DOUBLE_R_TLV("DMIC2 Volume", LM49453_P0_DMIC2_LEVELL_REG,
+ LM49453_P0_DMIC2_LEVELR_REG, 0, 6, 0, digital_tlv),
+
+ SOC_DAPM_ENUM("Mic2Mode", lm49453_mic2mode_enum),
+ SOC_DAPM_ENUM("DMIC12 SRC", lm49453_dmic12_cfg_enum),
+ SOC_DAPM_ENUM("DMIC34 SRC", lm49453_dmic34_cfg_enum),
+
+ /* Capture path filter enable */
+ SOC_SINGLE("DMIC1 HPFilter Switch", LM49453_P0_ADC_FX_ENABLES_REG,
+ 0, 1, 0),
+ SOC_SINGLE("DMIC2 HPFilter Switch", LM49453_P0_ADC_FX_ENABLES_REG,
+ 1, 1, 0),
+ SOC_SINGLE("ADC HPFilter Switch", LM49453_P0_ADC_FX_ENABLES_REG,
+ 2, 1, 0),
+
+ SOC_DOUBLE_R_TLV("DAC HP Volume", LM49453_P0_DAC_HP_LEVELL_REG,
+ LM49453_P0_DAC_HP_LEVELR_REG, 0, 6, 0, digital_tlv),
+ SOC_DOUBLE_R_TLV("DAC LO Volume", LM49453_P0_DAC_LO_LEVELL_REG,
+ LM49453_P0_DAC_LO_LEVELR_REG, 0, 6, 0, digital_tlv),
+ SOC_DOUBLE_R_TLV("DAC LS Volume", LM49453_P0_DAC_LS_LEVELL_REG,
+ LM49453_P0_DAC_LS_LEVELR_REG, 0, 6, 0, digital_tlv),
+ SOC_DOUBLE_R_TLV("DAC HA Volume", LM49453_P0_DAC_HA_LEVELL_REG,
+ LM49453_P0_DAC_HA_LEVELR_REG, 0, 6, 0, digital_tlv),
+
+ SOC_SINGLE_TLV("EP Volume", LM49453_P0_DAC_LS_LEVELL_REG,
+ 0, 6, 0, digital_tlv),
+
+ SOC_SINGLE_TLV("PORT1_1_RX_LVL Volume", LM49453_P0_PORT1_RX_LVL1_REG,
+ 0, 3, 0, port_tlv),
+ SOC_SINGLE_TLV("PORT1_2_RX_LVL Volume", LM49453_P0_PORT1_RX_LVL1_REG,
+ 2, 3, 0, port_tlv),
+ SOC_SINGLE_TLV("PORT1_3_RX_LVL Volume", LM49453_P0_PORT1_RX_LVL1_REG,
+ 4, 3, 0, port_tlv),
+ SOC_SINGLE_TLV("PORT1_4_RX_LVL Volume", LM49453_P0_PORT1_RX_LVL1_REG,
+ 6, 3, 0, port_tlv),
+ SOC_SINGLE_TLV("PORT1_5_RX_LVL Volume", LM49453_P0_PORT1_RX_LVL2_REG,
+ 0, 3, 0, port_tlv),
+ SOC_SINGLE_TLV("PORT1_6_RX_LVL Volume", LM49453_P0_PORT1_RX_LVL2_REG,
+ 2, 3, 0, port_tlv),
+ SOC_SINGLE_TLV("PORT1_7_RX_LVL Volume", LM49453_P0_PORT1_RX_LVL2_REG,
+ 4, 3, 0, port_tlv),
+ SOC_SINGLE_TLV("PORT1_8_RX_LVL Volume", LM49453_P0_PORT1_RX_LVL2_REG,
+ 6, 3, 0, port_tlv),
+
+ SOC_SINGLE_TLV("PORT2_1_RX_LVL Volume", LM49453_P0_PORT2_RX_LVL_REG,
+ 0, 3, 0, port_tlv),
+ SOC_SINGLE_TLV("PORT2_2_RX_LVL Volume", LM49453_P0_PORT2_RX_LVL_REG,
+ 2, 3, 0, port_tlv),
+
+ SOC_SINGLE("Port1 Playback Switch", LM49453_P0_AUDIO_PORT1_BASIC_REG,
+ 1, 1, 0),
+ SOC_SINGLE("Port2 Playback Switch", LM49453_P0_AUDIO_PORT2_BASIC_REG,
+ 1, 1, 0),
+ SOC_SINGLE("Port1 Capture Switch", LM49453_P0_AUDIO_PORT1_BASIC_REG,
+ 2, 1, 0),
+ SOC_SINGLE("Port2 Capture Switch", LM49453_P0_AUDIO_PORT2_BASIC_REG,
+ 2, 1, 0)
+
+};
+
+/* DAPM widgets */
+static const struct snd_soc_dapm_widget lm49453_dapm_widgets[] = {
+
+ /* All end points HP,EP, LS, Lineout and Haptic */
+ SND_SOC_DAPM_OUTPUT("HPOUTL"),
+ SND_SOC_DAPM_OUTPUT("HPOUTR"),
+ SND_SOC_DAPM_OUTPUT("EPOUT"),
+ SND_SOC_DAPM_OUTPUT("LSOUTL"),
+ SND_SOC_DAPM_OUTPUT("LSOUTR"),
+ SND_SOC_DAPM_OUTPUT("LOOUTR"),
+ SND_SOC_DAPM_OUTPUT("LOOUTL"),
+ SND_SOC_DAPM_OUTPUT("HAOUTL"),
+ SND_SOC_DAPM_OUTPUT("HAOUTR"),
+
+ SND_SOC_DAPM_INPUT("AMIC1"),
+ SND_SOC_DAPM_INPUT("AMIC2"),
+ SND_SOC_DAPM_INPUT("DMIC1DAT"),
+ SND_SOC_DAPM_INPUT("DMIC2DAT"),
+ SND_SOC_DAPM_INPUT("AUXL"),
+ SND_SOC_DAPM_INPUT("AUXR"),
+
+ SND_SOC_DAPM_PGA("PORT1_1_RX", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("PORT1_2_RX", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("PORT1_3_RX", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("PORT1_4_RX", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("PORT1_5_RX", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("PORT1_6_RX", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("PORT1_7_RX", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("PORT1_8_RX", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("PORT2_1_RX", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("PORT2_2_RX", SND_SOC_NOPM, 0, 0, NULL, 0),
+
+ SND_SOC_DAPM_SUPPLY("AMIC1Bias", LM49453_P0_MICL_REG, 6, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("AMIC2Bias", LM49453_P0_MICR_REG, 6, 0, NULL, 0),
+
+ /* playback path driver enables */
+ SND_SOC_DAPM_OUT_DRV("Headset Switch",
+ LM49453_P0_PMC_SETUP_REG, 0, 0, NULL, 0),
+ SND_SOC_DAPM_OUT_DRV("Earpiece Switch",
+ LM49453_P0_EP_REG, 0, 0, NULL, 0),
+ SND_SOC_DAPM_OUT_DRV("Speaker Left Switch",
+ LM49453_P0_DIS_PKVL_FB_REG, 0, 1, NULL, 0),
+ SND_SOC_DAPM_OUT_DRV("Speaker Right Switch",
+ LM49453_P0_DIS_PKVL_FB_REG, 1, 1, NULL, 0),
+ SND_SOC_DAPM_OUT_DRV("Haptic Left Switch",
+ LM49453_P0_DIS_PKVL_FB_REG, 2, 1, NULL, 0),
+ SND_SOC_DAPM_OUT_DRV("Haptic Right Switch",
+ LM49453_P0_DIS_PKVL_FB_REG, 3, 1, NULL, 0),
+
+ /* DAC */
+ SND_SOC_DAPM_DAC("HPL DAC", "Headset", SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_DAC("HPR DAC", "Headset", SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_DAC("LSL DAC", "Speaker", SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_DAC("LSR DAC", "Speaker", SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_DAC("HAL DAC", "Haptic", SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_DAC("HAR DAC", "Haptic", SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_DAC("LOL DAC", "Lineout", SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_DAC("LOR DAC", "Lineout", SND_SOC_NOPM, 0, 0),
+
+
+ SND_SOC_DAPM_PGA("AUXL Input",
+ LM49453_P0_ANALOG_MIXER_ADC_REG, 2, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("AUXR Input",
+ LM49453_P0_ANALOG_MIXER_ADC_REG, 3, 0, NULL, 0),
+
+ SND_SOC_DAPM_PGA("Sidetone", SND_SOC_NOPM, 0, 0, NULL, 0),
+
+ /* ADC */
+ SND_SOC_DAPM_ADC("DMIC1 Left", "Capture", SND_SOC_NOPM, 1, 0),
+ SND_SOC_DAPM_ADC("DMIC1 Right", "Capture", SND_SOC_NOPM, 1, 0),
+ SND_SOC_DAPM_ADC("DMIC2 Left", "Capture", SND_SOC_NOPM, 1, 0),
+ SND_SOC_DAPM_ADC("DMIC2 Right", "Capture", SND_SOC_NOPM, 1, 0),
+
+ SND_SOC_DAPM_ADC("ADC Left", "Capture", SND_SOC_NOPM, 1, 0),
+ SND_SOC_DAPM_ADC("ADC Right", "Capture", SND_SOC_NOPM, 0, 0),
+
+ SND_SOC_DAPM_MUX("ADCL Mux", SND_SOC_NOPM, 0, 0,
+ &lm49453_adcl_mux_control),
+ SND_SOC_DAPM_MUX("ADCR Mux", SND_SOC_NOPM, 0, 0,
+ &lm49453_adcr_mux_control),
+
+ SND_SOC_DAPM_MUX("Mic1 Input",
+ SND_SOC_NOPM, 0, 0, &lm49453_adcl_mux_control),
+
+ SND_SOC_DAPM_MUX("Mic2 Input",
+ SND_SOC_NOPM, 0, 0, &lm49453_adcr_mux_control),
+
+ /* AIF */
+ SND_SOC_DAPM_AIF_IN("PORT1_SDI", NULL, 0,
+ LM49453_P0_PULL_CONFIG1_REG, 2, 0),
+ SND_SOC_DAPM_AIF_IN("PORT2_SDI", NULL, 0,
+ LM49453_P0_PULL_CONFIG1_REG, 6, 0),
+
+ SND_SOC_DAPM_AIF_OUT("PORT1_SDO", NULL, 0,
+ LM49453_P0_PULL_CONFIG1_REG, 3, 0),
+ SND_SOC_DAPM_AIF_OUT("PORT2_SDO", NULL, 0,
+ LM49453_P0_PULL_CONFIG1_REG, 7, 0),
+
+ /* Port1 TX controls */
+ SND_SOC_DAPM_OUT_DRV("P1_1_TX", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_OUT_DRV("P1_2_TX", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_OUT_DRV("P1_3_TX", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_OUT_DRV("P1_4_TX", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_OUT_DRV("P1_5_TX", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_OUT_DRV("P1_6_TX", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_OUT_DRV("P1_7_TX", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_OUT_DRV("P1_8_TX", SND_SOC_NOPM, 0, 0, NULL, 0),
+
+ /* Port2 TX controls */
+ SND_SOC_DAPM_OUT_DRV("P2_1_TX", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_OUT_DRV("P2_2_TX", SND_SOC_NOPM, 0, 0, NULL, 0),
+
+ /* Sidetone Mixer */
+ SND_SOC_DAPM_MIXER("Sidetone Mixer", SND_SOC_NOPM, 0, 0,
+ lm49453_sidetone_mixer_controls,
+ ARRAY_SIZE(lm49453_sidetone_mixer_controls)),
+
+ /* DAC MIXERS */
+ SND_SOC_DAPM_MIXER("HPL Mixer", SND_SOC_NOPM, 0, 0,
+ lm49453_headset_left_mixer,
+ ARRAY_SIZE(lm49453_headset_left_mixer)),
+ SND_SOC_DAPM_MIXER("HPR Mixer", SND_SOC_NOPM, 0, 0,
+ lm49453_headset_right_mixer,
+ ARRAY_SIZE(lm49453_headset_right_mixer)),
+ SND_SOC_DAPM_MIXER("LOL Mixer", SND_SOC_NOPM, 0, 0,
+ lm49453_lineout_left_mixer,
+ ARRAY_SIZE(lm49453_lineout_left_mixer)),
+ SND_SOC_DAPM_MIXER("LOR Mixer", SND_SOC_NOPM, 0, 0,
+ lm49453_lineout_right_mixer,
+ ARRAY_SIZE(lm49453_lineout_right_mixer)),
+ SND_SOC_DAPM_MIXER("LSL Mixer", SND_SOC_NOPM, 0, 0,
+ lm49453_speaker_left_mixer,
+ ARRAY_SIZE(lm49453_speaker_left_mixer)),
+ SND_SOC_DAPM_MIXER("LSR Mixer", SND_SOC_NOPM, 0, 0,
+ lm49453_speaker_right_mixer,
+ ARRAY_SIZE(lm49453_speaker_right_mixer)),
+ SND_SOC_DAPM_MIXER("HAL Mixer", SND_SOC_NOPM, 0, 0,
+ lm49453_haptic_left_mixer,
+ ARRAY_SIZE(lm49453_haptic_left_mixer)),
+ SND_SOC_DAPM_MIXER("HAR Mixer", SND_SOC_NOPM, 0, 0,
+ lm49453_haptic_right_mixer,
+ ARRAY_SIZE(lm49453_haptic_right_mixer)),
+
+ /* Capture Mixer */
+ SND_SOC_DAPM_MIXER("Port1_1 Mixer", SND_SOC_NOPM, 0, 0,
+ lm49453_port1_tx1_mixer,
+ ARRAY_SIZE(lm49453_port1_tx1_mixer)),
+ SND_SOC_DAPM_MIXER("Port1_2 Mixer", SND_SOC_NOPM, 0, 0,
+ lm49453_port1_tx2_mixer,
+ ARRAY_SIZE(lm49453_port1_tx2_mixer)),
+ SND_SOC_DAPM_MIXER("Port1_3 Mixer", SND_SOC_NOPM, 0, 0,
+ lm49453_port1_tx3_mixer,
+ ARRAY_SIZE(lm49453_port1_tx3_mixer)),
+ SND_SOC_DAPM_MIXER("Port1_4 Mixer", SND_SOC_NOPM, 0, 0,
+ lm49453_port1_tx4_mixer,
+ ARRAY_SIZE(lm49453_port1_tx4_mixer)),
+ SND_SOC_DAPM_MIXER("Port1_5 Mixer", SND_SOC_NOPM, 0, 0,
+ lm49453_port1_tx5_mixer,
+ ARRAY_SIZE(lm49453_port1_tx5_mixer)),
+ SND_SOC_DAPM_MIXER("Port1_6 Mixer", SND_SOC_NOPM, 0, 0,
+ lm49453_port1_tx6_mixer,
+ ARRAY_SIZE(lm49453_port1_tx6_mixer)),
+ SND_SOC_DAPM_MIXER("Port1_7 Mixer", SND_SOC_NOPM, 0, 0,
+ lm49453_port1_tx7_mixer,
+ ARRAY_SIZE(lm49453_port1_tx7_mixer)),
+ SND_SOC_DAPM_MIXER("Port1_8 Mixer", SND_SOC_NOPM, 0, 0,
+ lm49453_port1_tx8_mixer,
+ ARRAY_SIZE(lm49453_port1_tx8_mixer)),
+
+ SND_SOC_DAPM_MIXER("Port2_1 Mixer", SND_SOC_NOPM, 0, 0,
+ lm49453_port2_tx1_mixer,
+ ARRAY_SIZE(lm49453_port2_tx1_mixer)),
+ SND_SOC_DAPM_MIXER("Port2_2 Mixer", SND_SOC_NOPM, 0, 0,
+ lm49453_port2_tx2_mixer,
+ ARRAY_SIZE(lm49453_port2_tx2_mixer)),
+};
+
+static const struct snd_soc_dapm_route lm49453_audio_map[] = {
+ /* Port SDI mapping */
+ { "PORT1_1_RX", "Port1 Playback Switch", "PORT1_SDI" },
+ { "PORT1_2_RX", "Port1 Playback Switch", "PORT1_SDI" },
+ { "PORT1_3_RX", "Port1 Playback Switch", "PORT1_SDI" },
+ { "PORT1_4_RX", "Port1 Playback Switch", "PORT1_SDI" },
+ { "PORT1_5_RX", "Port1 Playback Switch", "PORT1_SDI" },
+ { "PORT1_6_RX", "Port1 Playback Switch", "PORT1_SDI" },
+ { "PORT1_7_RX", "Port1 Playback Switch", "PORT1_SDI" },
+ { "PORT1_8_RX", "Port1 Playback Switch", "PORT1_SDI" },
+
+ { "PORT2_1_RX", "Port2 Playback Switch", "PORT2_SDI" },
+ { "PORT2_2_RX", "Port2 Playback Switch", "PORT2_SDI" },
+
+ /* HP mapping */
+ { "HPL Mixer", "Port1_1 Switch", "PORT1_1_RX" },
+ { "HPL Mixer", "Port1_2 Switch", "PORT1_2_RX" },
+ { "HPL Mixer", "Port1_3 Switch", "PORT1_3_RX" },
+ { "HPL Mixer", "Port1_4 Switch", "PORT1_4_RX" },
+ { "HPL Mixer", "Port1_5 Switch", "PORT1_5_RX" },
+ { "HPL Mixer", "Port1_6 Switch", "PORT1_6_RX" },
+ { "HPL Mixer", "Port1_7 Switch", "PORT1_7_RX" },
+ { "HPL Mixer", "Port1_8 Switch", "PORT1_8_RX" },
+
+ { "HPL Mixer", "Port2_1 Switch", "PORT2_1_RX" },
+ { "HPL Mixer", "Port2_2 Switch", "PORT2_2_RX" },
+
+ { "HPL Mixer", "ADCL Switch", "ADC Left" },
+ { "HPL Mixer", "ADCR Switch", "ADC Right" },
+ { "HPL Mixer", "DMIC1L Switch", "DMIC1 Left" },
+ { "HPL Mixer", "DMIC1R Switch", "DMIC1 Right" },
+ { "HPL Mixer", "DMIC2L Switch", "DMIC2 Left" },
+ { "HPL Mixer", "DMIC2R Switch", "DMIC2 Right" },
+ { "HPL Mixer", "Sidetone Switch", "Sidetone" },
+
+ { "HPL DAC", NULL, "HPL Mixer" },
+
+ { "HPR Mixer", "Port1_1 Switch", "PORT1_1_RX" },
+ { "HPR Mixer", "Port1_2 Switch", "PORT1_2_RX" },
+ { "HPR Mixer", "Port1_3 Switch", "PORT1_3_RX" },
+ { "HPR Mixer", "Port1_4 Switch", "PORT1_4_RX" },
+ { "HPR Mixer", "Port1_5 Switch", "PORT1_5_RX" },
+ { "HPR Mixer", "Port1_6 Switch", "PORT1_6_RX" },
+ { "HPR Mixer", "Port1_7 Switch", "PORT1_7_RX" },
+ { "HPR Mixer", "Port1_8 Switch", "PORT1_8_RX" },
+
+ /* Port 2 */
+ { "HPR Mixer", "Port2_1 Switch", "PORT2_1_RX" },
+ { "HPR Mixer", "Port2_2 Switch", "PORT2_2_RX" },
+
+ { "HPR Mixer", "ADCL Switch", "ADC Left" },
+ { "HPR Mixer", "ADCR Switch", "ADC Right" },
+ { "HPR Mixer", "DMIC1L Switch", "DMIC1 Left" },
+ { "HPR Mixer", "DMIC1R Switch", "DMIC1 Right" },
+ { "HPR Mixer", "DMIC2L Switch", "DMIC2 Left" },
+ { "HPR Mixer", "DMIC2L Switch", "DMIC2 Right" },
+ { "HPR Mixer", "Sidetone Switch", "Sidetone" },
+
+ { "HPR DAC", NULL, "HPR Mixer" },
+
+ { "HPOUTL", "Headset Switch", "HPL DAC"},
+ { "HPOUTR", "Headset Switch", "HPR DAC"},
+
+ /* EP map */
+ { "EPOUT", "Earpiece Switch", "HPL DAC" },
+
+ /* Speaker map */
+ { "LSL Mixer", "Port1_1 Switch", "PORT1_1_RX" },
+ { "LSL Mixer", "Port1_2 Switch", "PORT1_2_RX" },
+ { "LSL Mixer", "Port1_3 Switch", "PORT1_3_RX" },
+ { "LSL Mixer", "Port1_4 Switch", "PORT1_4_RX" },
+ { "LSL Mixer", "Port1_5 Switch", "PORT1_5_RX" },
+ { "LSL Mixer", "Port1_6 Switch", "PORT1_6_RX" },
+ { "LSL Mixer", "Port1_7 Switch", "PORT1_7_RX" },
+ { "LSL Mixer", "Port1_8 Switch", "PORT1_8_RX" },
+
+ /* Port 2 */
+ { "LSL Mixer", "Port2_1 Switch", "PORT2_1_RX" },
+ { "LSL Mixer", "Port2_2 Switch", "PORT2_2_RX" },
+
+ { "LSL Mixer", "ADCL Switch", "ADC Left" },
+ { "LSL Mixer", "ADCR Switch", "ADC Right" },
+ { "LSL Mixer", "DMIC1L Switch", "DMIC1 Left" },
+ { "LSL Mixer", "DMIC1R Switch", "DMIC1 Right" },
+ { "LSL Mixer", "DMIC2L Switch", "DMIC2 Left" },
+ { "LSL Mixer", "DMIC2R Switch", "DMIC2 Right" },
+ { "LSL Mixer", "Sidetone Switch", "Sidetone" },
+
+ { "LSL DAC", NULL, "LSL Mixer" },
+
+ { "LSR Mixer", "Port1_1 Switch", "PORT1_1_RX" },
+ { "LSR Mixer", "Port1_2 Switch", "PORT1_2_RX" },
+ { "LSR Mixer", "Port1_3 Switch", "PORT1_3_RX" },
+ { "LSR Mixer", "Port1_4 Switch", "PORT1_4_RX" },
+ { "LSR Mixer", "Port1_5 Switch", "PORT1_5_RX" },
+ { "LSR Mixer", "Port1_6 Switch", "PORT1_6_RX" },
+ { "LSR Mixer", "Port1_7 Switch", "PORT1_7_RX" },
+ { "LSR Mixer", "Port1_8 Switch", "PORT1_8_RX" },
+
+ /* Port 2 */
+ { "LSR Mixer", "Port2_1 Switch", "PORT2_1_RX" },
+ { "LSR Mixer", "Port2_2 Switch", "PORT2_2_RX" },
+
+ { "LSR Mixer", "ADCL Switch", "ADC Left" },
+ { "LSR Mixer", "ADCR Switch", "ADC Right" },
+ { "LSR Mixer", "DMIC1L Switch", "DMIC1 Left" },
+ { "LSR Mixer", "DMIC1R Switch", "DMIC1 Right" },
+ { "LSR Mixer", "DMIC2L Switch", "DMIC2 Left" },
+ { "LSR Mixer", "DMIC2R Switch", "DMIC2 Right" },
+ { "LSR Mixer", "Sidetone Switch", "Sidetone" },
+
+ { "LSR DAC", NULL, "LSR Mixer" },
+
+ { "LSOUTL", "Speaker Left Switch", "LSL DAC"},
+ { "LSOUTR", "Speaker Left Switch", "LSR DAC"},
+
+ /* Haptic map */
+ { "HAL Mixer", "Port1_1 Switch", "PORT1_1_RX" },
+ { "HAL Mixer", "Port1_2 Switch", "PORT1_2_RX" },
+ { "HAL Mixer", "Port1_3 Switch", "PORT1_3_RX" },
+ { "HAL Mixer", "Port1_4 Switch", "PORT1_4_RX" },
+ { "HAL Mixer", "Port1_5 Switch", "PORT1_5_RX" },
+ { "HAL Mixer", "Port1_6 Switch", "PORT1_6_RX" },
+ { "HAL Mixer", "Port1_7 Switch", "PORT1_7_RX" },
+ { "HAL Mixer", "Port1_8 Switch", "PORT1_8_RX" },
+
+ /* Port 2 */
+ { "HAL Mixer", "Port2_1 Switch", "PORT2_1_RX" },
+ { "HAL Mixer", "Port2_2 Switch", "PORT2_2_RX" },
+
+ { "HAL Mixer", "ADCL Switch", "ADC Left" },
+ { "HAL Mixer", "ADCR Switch", "ADC Right" },
+ { "HAL Mixer", "DMIC1L Switch", "DMIC1 Left" },
+ { "HAL Mixer", "DMIC1R Switch", "DMIC1 Right" },
+ { "HAL Mixer", "DMIC2L Switch", "DMIC2 Left" },
+ { "HAL Mixer", "DMIC2R Switch", "DMIC2 Right" },
+ { "HAL Mixer", "Sidetone Switch", "Sidetone" },
+
+ { "HAL DAC", NULL, "HAL Mixer" },
+
+ { "HAR Mixer", "Port1_1 Switch", "PORT1_1_RX" },
+ { "HAR Mixer", "Port1_2 Switch", "PORT1_2_RX" },
+ { "HAR Mixer", "Port1_3 Switch", "PORT1_3_RX" },
+ { "HAR Mixer", "Port1_4 Switch", "PORT1_4_RX" },
+ { "HAR Mixer", "Port1_5 Switch", "PORT1_5_RX" },
+ { "HAR Mixer", "Port1_6 Switch", "PORT1_6_RX" },
+ { "HAR Mixer", "Port1_7 Switch", "PORT1_7_RX" },
+ { "HAR Mixer", "Port1_8 Switch", "PORT1_8_RX" },
+
+ /* Port 2 */
+ { "HAR Mixer", "Port2_1 Switch", "PORT2_1_RX" },
+ { "HAR Mixer", "Port2_2 Switch", "PORT2_2_RX" },
+
+ { "HAR Mixer", "ADCL Switch", "ADC Left" },
+ { "HAR Mixer", "ADCR Switch", "ADC Right" },
+ { "HAR Mixer", "DMIC1L Switch", "DMIC1 Left" },
+ { "HAR Mixer", "DMIC1R Switch", "DMIC1 Right" },
+ { "HAR Mixer", "DMIC2L Switch", "DMIC2 Left" },
+ { "HAR Mixer", "DMIC2R Switch", "DMIC2 Right" },
+ { "HAR Mixer", "Sideton Switch", "Sidetone" },
+
+ { "HAR DAC", NULL, "HAR Mixer" },
+
+ { "HAOUTL", "Haptic Left Switch", "HAL DAC" },
+ { "HAOUTR", "Haptic Right Switch", "HAR DAC" },
+
+ /* Lineout map */
+ { "LOL Mixer", "Port1_1 Switch", "PORT1_1_RX" },
+ { "LOL Mixer", "Port1_2 Switch", "PORT1_2_RX" },
+ { "LOL Mixer", "Port1_3 Switch", "PORT1_3_RX" },
+ { "LOL Mixer", "Port1_4 Switch", "PORT1_4_RX" },
+ { "LOL Mixer", "Port1_5 Switch", "PORT1_5_RX" },
+ { "LOL Mixer", "Port1_6 Switch", "PORT1_6_RX" },
+ { "LOL Mixer", "Port1_7 Switch", "PORT1_7_RX" },
+ { "LOL Mixer", "Port1_8 Switch", "PORT1_8_RX" },
+
+ /* Port 2 */
+ { "LOL Mixer", "Port2_1 Switch", "PORT2_1_RX" },
+ { "LOL Mixer", "Port2_2 Switch", "PORT2_2_RX" },
+
+ { "LOL Mixer", "ADCL Switch", "ADC Left" },
+ { "LOL Mixer", "ADCR Switch", "ADC Right" },
+ { "LOL Mixer", "DMIC1L Switch", "DMIC1 Left" },
+ { "LOL Mixer", "DMIC1R Switch", "DMIC1 Right" },
+ { "LOL Mixer", "DMIC2L Switch", "DMIC2 Left" },
+ { "LOL Mixer", "DMIC2R Switch", "DMIC2 Right" },
+ { "LOL Mixer", "Sidetone Switch", "Sidetone" },
+
+ { "LOL DAC", NULL, "LOL Mixer" },
+
+ { "LOR Mixer", "Port1_1 Switch", "PORT1_1_RX" },
+ { "LOR Mixer", "Port1_2 Switch", "PORT1_2_RX" },
+ { "LOR Mixer", "Port1_3 Switch", "PORT1_3_RX" },
+ { "LOR Mixer", "Port1_4 Switch", "PORT1_4_RX" },
+ { "LOR Mixer", "Port1_5 Switch", "PORT1_5_RX" },
+ { "LOR Mixer", "Port1_6 Switch", "PORT1_6_RX" },
+ { "LOR Mixer", "Port1_7 Switch", "PORT1_7_RX" },
+ { "LOR Mixer", "Port1_8 Switch", "PORT1_8_RX" },
+
+ /* Port 2 */
+ { "LOR Mixer", "Port2_1 Switch", "PORT2_1_RX" },
+ { "LOR Mixer", "Port2_2 Switch", "PORT2_2_RX" },
+
+ { "LOR Mixer", "ADCL Switch", "ADC Left" },
+ { "LOR Mixer", "ADCR Switch", "ADC Right" },
+ { "LOR Mixer", "DMIC1L Switch", "DMIC1 Left" },
+ { "LOR Mixer", "DMIC1R Switch", "DMIC1 Right" },
+ { "LOR Mixer", "DMIC2L Switch", "DMIC2 Left" },
+ { "LOR Mixer", "DMIC2R Switch", "DMIC2 Right" },
+ { "LOR Mixer", "Sidetone Switch", "Sidetone" },
+
+ { "LOR DAC", NULL, "LOR Mixer" },
+
+ { "LOOUTL", NULL, "LOL DAC" },
+ { "LOOUTR", NULL, "LOR DAC" },
+
+ /* TX map */
+ /* Port1 mappings */
+ { "Port1_1 Mixer", "ADCL Switch", "ADC Left" },
+ { "Port1_1 Mixer", "ADCR Switch", "ADC Right" },
+ { "Port1_1 Mixer", "DMIC1L Switch", "DMIC1 Left" },
+ { "Port1_1 Mixer", "DMIC1R Switch", "DMIC1 Right" },
+ { "Port1_1 Mixer", "DMIC2L Switch", "DMIC2 Left" },
+ { "Port1_1 Mixer", "DMIC2R Switch", "DMIC2 Right" },
+
+ { "Port1_2 Mixer", "ADCL Switch", "ADC Left" },
+ { "Port1_2 Mixer", "ADCR Switch", "ADC Right" },
+ { "Port1_2 Mixer", "DMIC1L Switch", "DMIC1 Left" },
+ { "Port1_2 Mixer", "DMIC1R Switch", "DMIC1 Right" },
+ { "Port1_2 Mixer", "DMIC2L Switch", "DMIC2 Left" },
+ { "Port1_2 Mixer", "DMIC2R Switch", "DMIC2 Right" },
+
+ { "Port1_3 Mixer", "ADCL Switch", "ADC Left" },
+ { "Port1_3 Mixer", "ADCR Switch", "ADC Right" },
+ { "Port1_3 Mixer", "DMIC1L Switch", "DMIC1 Left" },
+ { "Port1_3 Mixer", "DMIC1R Switch", "DMIC1 Right" },
+ { "Port1_3 Mixer", "DMIC2L Switch", "DMIC2 Left" },
+ { "Port1_3 Mixer", "DMIC2R Switch", "DMIC2 Right" },
+
+ { "Port1_4 Mixer", "ADCL Switch", "ADC Left" },
+ { "Port1_4 Mixer", "ADCR Switch", "ADC Right" },
+ { "Port1_4 Mixer", "DMIC1L Switch", "DMIC1 Left" },
+ { "Port1_4 Mixer", "DMIC1R Switch", "DMIC1 Right" },
+ { "Port1_4 Mixer", "DMIC2L Switch", "DMIC2 Left" },
+ { "Port1_4 Mixer", "DMIC2R Switch", "DMIC2 Right" },
+
+ { "Port1_5 Mixer", "ADCL Switch", "ADC Left" },
+ { "Port1_5 Mixer", "ADCR Switch", "ADC Right" },
+ { "Port1_5 Mixer", "DMIC1L Switch", "DMIC1 Left" },
+ { "Port1_5 Mixer", "DMIC1R Switch", "DMIC1 Right" },
+ { "Port1_5 Mixer", "DMIC2L Switch", "DMIC2 Left" },
+ { "Port1_5 Mixer", "DMIC2R Switch", "DMIC2 Right" },
+
+ { "Port1_6 Mixer", "ADCL Switch", "ADC Left" },
+ { "Port1_6 Mixer", "ADCR Switch", "ADC Right" },
+ { "Port1_6 Mixer", "DMIC1L Switch", "DMIC1 Left" },
+ { "Port1_6 Mixer", "DMIC1R Switch", "DMIC1 Right" },
+ { "Port1_6 Mixer", "DMIC2L Switch", "DMIC2 Left" },
+ { "Port1_6 Mixer", "DMIC2R Switch", "DMIC2 Right" },
+
+ { "Port1_7 Mixer", "ADCL Switch", "ADC Left" },
+ { "Port1_7 Mixer", "ADCR Switch", "ADC Right" },
+ { "Port1_7 Mixer", "DMIC1L Switch", "DMIC1 Left" },
+ { "Port1_7 Mixer", "DMIC1R Switch", "DMIC1 Right" },
+ { "Port1_7 Mixer", "DMIC2L Switch", "DMIC2 Left" },
+ { "Port1_7 Mixer", "DMIC2R Switch", "DMIC2 Right" },
+
+ { "Port1_8 Mixer", "ADCL Switch", "ADC Left" },
+ { "Port1_8 Mixer", "ADCR Switch", "ADC Right" },
+ { "Port1_8 Mixer", "DMIC1L Switch", "DMIC1 Left" },
+ { "Port1_8 Mixer", "DMIC1R Switch", "DMIC1 Right" },
+ { "Port1_8 Mixer", "DMIC2L Switch", "DMIC2 Left" },
+ { "Port1_8 Mixer", "DMIC2R Switch", "DMIC2 Right" },
+
+ { "Port2_1 Mixer", "ADCL Switch", "ADC Left" },
+ { "Port2_1 Mixer", "ADCR Switch", "ADC Right" },
+ { "Port2_1 Mixer", "DMIC1L Switch", "DMIC1 Left" },
+ { "Port2_1 Mixer", "DMIC1R Switch", "DMIC1 Right" },
+ { "Port2_1 Mixer", "DMIC2L Switch", "DMIC2 Left" },
+ { "Port2_1 Mixer", "DMIC2R Switch", "DMIC2 Right" },
+
+ { "Port2_2 Mixer", "ADCL Switch", "ADC Left" },
+ { "Port2_2 Mixer", "ADCR Switch", "ADC Right" },
+ { "Port2_2 Mixer", "DMIC1L Switch", "DMIC1 Left" },
+ { "Port2_2 Mixer", "DMIC1R Switch", "DMIC1 Right" },
+ { "Port2_2 Mixer", "DMIC2L Switch", "DMIC2 Left" },
+ { "Port2_2 Mixer", "DMIC2R Switch", "DMIC2 Right" },
+
+ { "P1_1_TX", NULL, "Port1_1 Mixer" },
+ { "P1_2_TX", NULL, "Port1_2 Mixer" },
+ { "P1_3_TX", NULL, "Port1_3 Mixer" },
+ { "P1_4_TX", NULL, "Port1_4 Mixer" },
+ { "P1_5_TX", NULL, "Port1_5 Mixer" },
+ { "P1_6_TX", NULL, "Port1_6 Mixer" },
+ { "P1_7_TX", NULL, "Port1_7 Mixer" },
+ { "P1_8_TX", NULL, "Port1_8 Mixer" },
+
+ { "P2_1_TX", NULL, "Port2_1 Mixer" },
+ { "P2_2_TX", NULL, "Port2_2 Mixer" },
+
+ { "PORT1_SDO", "Port1 Capture Switch", "P1_1_TX"},
+ { "PORT1_SDO", "Port1 Capture Switch", "P1_2_TX"},
+ { "PORT1_SDO", "Port1 Capture Switch", "P1_3_TX"},
+ { "PORT1_SDO", "Port1 Capture Switch", "P1_4_TX"},
+ { "PORT1_SDO", "Port1 Capture Switch", "P1_5_TX"},
+ { "PORT1_SDO", "Port1 Capture Switch", "P1_6_TX"},
+ { "PORT1_SDO", "Port1 Capture Switch", "P1_7_TX"},
+ { "PORT1_SDO", "Port1 Capture Switch", "P1_8_TX"},
+
+ { "PORT2_SDO", "Port2 Capture Switch", "P2_1_TX"},
+ { "PORT2_SDO", "Port2 Capture Switch", "P2_2_TX"},
+
+ { "Mic1 Input", NULL, "AMIC1" },
+ { "Mic2 Input", NULL, "AMIC2" },
+
+ { "AUXL Input", NULL, "AUXL" },
+ { "AUXR Input", NULL, "AUXR" },
+
+ /* AUX connections */
+ { "ADCL Mux", "Aux_L", "AUXL Input" },
+ { "ADCL Mux", "MIC1", "Mic1 Input" },
+
+ { "ADCR Mux", "Aux_R", "AUXR Input" },
+ { "ADCR Mux", "MIC2", "Mic2 Input" },
+
+ /* ADC connection */
+ { "ADC Left", NULL, "ADCL Mux"},
+ { "ADC Right", NULL, "ADCR Mux"},
+
+ { "DMIC1 Left", NULL, "DMIC1DAT"},
+ { "DMIC1 Right", NULL, "DMIC1DAT"},
+ { "DMIC2 Left", NULL, "DMIC2DAT"},
+ { "DMIC2 Right", NULL, "DMIC2DAT"},
+
+ /* Sidetone map */
+ { "Sidetone Mixer", NULL, "ADC Left" },
+ { "Sidetone Mixer", NULL, "ADC Right" },
+ { "Sidetone Mixer", NULL, "DMIC1 Left" },
+ { "Sidetone Mixer", NULL, "DMIC1 Right" },
+ { "Sidetone Mixer", NULL, "DMIC2 Left" },
+ { "Sidetone Mixer", NULL, "DMIC2 Right" },
+
+ { "Sidetone", "Sidetone Switch", "Sidetone Mixer" },
+};
+
+static int lm49453_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 lm49453_priv *lm49453 = snd_soc_codec_get_drvdata(codec);
+ u16 clk_div = 0;
+
+ lm49453->fs_rate = params_rate(params);
+
+ /* Setting DAC clock dividers based on substream sample rate. */
+ switch (lm49453->fs_rate) {
+ case 8000:
+ case 16000:
+ case 32000:
+ case 24000:
+ case 48000:
+ clk_div = 256;
+ break;
+ case 11025:
+ case 22050:
+ case 44100:
+ clk_div = 216;
+ break;
+ case 96000:
+ clk_div = 127;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ snd_soc_write(codec, LM49453_P0_ADC_CLK_DIV_REG, clk_div);
+ snd_soc_write(codec, LM49453_P0_DAC_HP_CLK_DIV_REG, clk_div);
+
+ return 0;
+}
+
+static int lm49453_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+
+ u16 aif_val;
+ int mode = 0;
+ int clk_phase = 0;
+ int clk_shift = 0;
+
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBS_CFS:
+ aif_val = 0;
+ break;
+ case SND_SOC_DAIFMT_CBS_CFM:
+ aif_val = LM49453_AUDIO_PORT1_BASIC_SYNC_MS;
+ break;
+ case SND_SOC_DAIFMT_CBM_CFS:
+ aif_val = LM49453_AUDIO_PORT1_BASIC_CLK_MS;
+ break;
+ case SND_SOC_DAIFMT_CBM_CFM:
+ aif_val = LM49453_AUDIO_PORT1_BASIC_CLK_MS |
+ LM49453_AUDIO_PORT1_BASIC_SYNC_MS;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ break;
+ case SND_SOC_DAIFMT_DSP_A:
+ mode = 1;
+ clk_phase = (1 << 5);
+ clk_shift = 1;
+ break;
+ case SND_SOC_DAIFMT_DSP_B:
+ mode = 1;
+ clk_phase = (1 << 5);
+ clk_shift = 0;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ snd_soc_update_bits(codec, LM49453_P0_AUDIO_PORT1_BASIC_REG,
+ LM49453_AUDIO_PORT1_BASIC_FMT_MASK|BIT(1)|BIT(5),
+ (aif_val | mode | clk_phase));
+
+ snd_soc_write(codec, LM49453_P0_AUDIO_PORT1_RX_MSB_REG, clk_shift);
+
+ return 0;
+}
+
+static int lm49453_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id,
+ unsigned int freq, int dir)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ u16 pll_clk = 0;
+
+ switch (freq) {
+ case 12288000:
+ case 26000000:
+ case 19200000:
+ /* pll clk slection */
+ pll_clk = 0;
+ break;
+ case 48000:
+ case 32576:
+ /* fll clk slection */
+ pll_clk = BIT(4);
+ return 0;
+ default:
+ return -EINVAL;
+ }
+
+ snd_soc_update_bits(codec, LM49453_P0_PMC_SETUP_REG, BIT(4), pll_clk);
+
+ return 0;
+}
+
+static int lm49453_hp_mute(struct snd_soc_dai *dai, int mute)
+{
+ snd_soc_update_bits(dai->codec, LM49453_P0_DAC_DSP_REG, BIT(1)|BIT(0),
+ (mute ? (BIT(1)|BIT(0)) : 0));
+ return 0;
+}
+
+static int lm49453_lo_mute(struct snd_soc_dai *dai, int mute)
+{
+ snd_soc_update_bits(dai->codec, LM49453_P0_DAC_DSP_REG, BIT(3)|BIT(2),
+ (mute ? (BIT(3)|BIT(2)) : 0));
+ return 0;
+}
+
+static int lm49453_ls_mute(struct snd_soc_dai *dai, int mute)
+{
+ snd_soc_update_bits(dai->codec, LM49453_P0_DAC_DSP_REG, BIT(5)|BIT(4),
+ (mute ? (BIT(5)|BIT(4)) : 0));
+ return 0;
+}
+
+static int lm49453_ep_mute(struct snd_soc_dai *dai, int mute)
+{
+ snd_soc_update_bits(dai->codec, LM49453_P0_DAC_DSP_REG, BIT(4),
+ (mute ? BIT(4) : 0));
+ return 0;
+}
+
+static int lm49453_ha_mute(struct snd_soc_dai *dai, int mute)
+{
+ snd_soc_update_bits(dai->codec, LM49453_P0_DAC_DSP_REG, BIT(7)|BIT(6),
+ (mute ? (BIT(7)|BIT(6)) : 0));
+ return 0;
+}
+
+static int lm49453_set_bias_level(struct snd_soc_codec *codec,
+ enum snd_soc_bias_level level)
+{
+ struct lm49453_priv *lm49453 = snd_soc_codec_get_drvdata(codec);
+
+ switch (level) {
+ case SND_SOC_BIAS_ON:
+ case SND_SOC_BIAS_PREPARE:
+ break;
+
+ case SND_SOC_BIAS_STANDBY:
+ if (codec->dapm.bias_level == SND_SOC_BIAS_OFF)
+ regcache_sync(lm49453->regmap);
+
+ snd_soc_update_bits(codec, LM49453_P0_PMC_SETUP_REG,
+ LM49453_PMC_SETUP_CHIP_EN, LM49453_CHIP_EN);
+ break;
+
+ case SND_SOC_BIAS_OFF:
+ snd_soc_update_bits(codec, LM49453_P0_PMC_SETUP_REG,
+ LM49453_PMC_SETUP_CHIP_EN, 0);
+ break;
+ }
+
+ codec->dapm.bias_level = level;
+
+ return 0;
+}
+
+/* Formates supported by LM49453 driver. */
+#define LM49453_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 lm49453_headset_dai_ops = {
+ .hw_params = lm49453_hw_params,
+ .set_sysclk = lm49453_set_dai_sysclk,
+ .set_fmt = lm49453_set_dai_fmt,
+ .digital_mute = lm49453_hp_mute,
+};
+
+static struct snd_soc_dai_ops lm49453_speaker_dai_ops = {
+ .hw_params = lm49453_hw_params,
+ .set_sysclk = lm49453_set_dai_sysclk,
+ .set_fmt = lm49453_set_dai_fmt,
+ .digital_mute = lm49453_ls_mute,
+};
+
+static struct snd_soc_dai_ops lm49453_haptic_dai_ops = {
+ .hw_params = lm49453_hw_params,
+ .set_sysclk = lm49453_set_dai_sysclk,
+ .set_fmt = lm49453_set_dai_fmt,
+ .digital_mute = lm49453_ha_mute,
+};
+
+static struct snd_soc_dai_ops lm49453_ep_dai_ops = {
+ .hw_params = lm49453_hw_params,
+ .set_sysclk = lm49453_set_dai_sysclk,
+ .set_fmt = lm49453_set_dai_fmt,
+ .digital_mute = lm49453_ep_mute,
+};
+
+static struct snd_soc_dai_ops lm49453_lineout_dai_ops = {
+ .hw_params = lm49453_hw_params,
+ .set_sysclk = lm49453_set_dai_sysclk,
+ .set_fmt = lm49453_set_dai_fmt,
+ .digital_mute = lm49453_lo_mute,
+};
+
+/* LM49453 dai structure. */
+static const struct snd_soc_dai_driver lm49453_dai[] = {
+ {
+ .name = "LM49453 Headset",
+ .playback = {
+ .stream_name = "Headset",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_8000_192000,
+ .formats = LM49453_FORMATS,
+ },
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 1,
+ .channels_max = 5,
+ .rates = SNDRV_PCM_RATE_8000_192000,
+ .formats = LM49453_FORMATS,
+ },
+ .ops = &lm49453_headset_dai_ops,
+ .symmetric_rates = 1,
+ },
+ {
+ .name = "LM49453 Speaker",
+ .playback = {
+ .stream_name = "Speaker",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_8000_192000,
+ .formats = LM49453_FORMATS,
+ },
+ .ops = &lm49453_speaker_dai_ops,
+ },
+ {
+ .name = "LM49453 Haptic",
+ .playback = {
+ .stream_name = "Haptic",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_8000_192000,
+ .formats = LM49453_FORMATS,
+ },
+ .ops = &lm49453_haptic_dai_ops,
+ },
+ {
+ .name = "LM49453 Earpiece",
+ .playback = {
+ .stream_name = "Earpiece",
+ .channels_min = 1,
+ .channels_max = 1,
+ .rates = SNDRV_PCM_RATE_8000_192000,
+ .formats = LM49453_FORMATS,
+ },
+ .ops = &lm49453_ep_dai_ops,
+ },
+ {
+ .name = "LM49453 line out",
+ .playback = {
+ .stream_name = "Lineout",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_8000_192000,
+ .formats = LM49453_FORMATS,
+ },
+ .ops = &lm49453_lineout_dai_ops,
+ },
+};
+
+static int lm49453_suspend(struct snd_soc_codec *codec)
+{
+ lm49453_set_bias_level(codec, SND_SOC_BIAS_OFF);
+ return 0;
+}
+
+static int lm49453_resume(struct snd_soc_codec *codec)
+{
+ lm49453_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+ return 0;
+}
+
+static int lm49453_probe(struct snd_soc_codec *codec)
+{
+ struct lm49453_priv *lm49453 = snd_soc_codec_get_drvdata(codec);
+ int ret = 0;
+
+ codec->control_data = lm49453->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;
+ }
+
+ return 0;
+}
+
+/* power down chip */
+static int lm49453_remove(struct snd_soc_codec *codec)
+{
+ lm49453_set_bias_level(codec, SND_SOC_BIAS_OFF);
+ return 0;
+}
+
+static struct snd_soc_codec_driver soc_codec_dev_lm49453 = {
+ .probe = lm49453_probe,
+ .remove = lm49453_remove,
+ .suspend = lm49453_suspend,
+ .resume = lm49453_resume,
+ .set_bias_level = lm49453_set_bias_level,
+ .controls = lm49453_snd_controls,
+ .num_controls = ARRAY_SIZE(lm49453_snd_controls),
+ .dapm_widgets = lm49453_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(lm49453_dapm_widgets),
+ .dapm_routes = lm49453_audio_map,
+ .num_dapm_routes = ARRAY_SIZE(lm49453_audio_map),
+ .idle_bias_off = true,
+};
+
+static const struct regmap_config lm49453_regmap_config = {
+ .reg_bits = 8,
+ .val_bits = 8,
+
+ .max_register = LM49453_MAX_REGISTER,
+ .reg_defaults = lm49453_reg_defs,
+ .num_reg_defaults = ARRAY_SIZE(lm49453_reg_defs),
+ .cache_type = REGCACHE_RBTREE,
+};
+
+static __devinit int lm49453_i2c_probe(struct i2c_client *i2c,
+ const struct i2c_device_id *id)
+{
+ struct lm49453_priv *lm49453;
+ int ret = 0;
+
+ lm49453 = devm_kzalloc(&i2c->dev, sizeof(struct lm49453_priv),
+ GFP_KERNEL);
+
+ if (lm49453 == NULL)
+ return -ENOMEM;
+
+ i2c_set_clientdata(i2c, lm49453);
+
+ lm49453->regmap = regmap_init_i2c(i2c, &lm49453_regmap_config);
+ if (IS_ERR(lm49453->regmap)) {
+ ret = PTR_ERR(lm49453->regmap);
+ dev_err(&i2c->dev, "Failed to allocate register map: %d\n",
+ ret);
+ return ret;
+ }
+
+ ret = snd_soc_register_codec(&i2c->dev,
+ &soc_codec_dev_lm49453,
+ lm49453_dai, ARRAY_SIZE(lm49453_dai));
+ if (ret < 0) {
+ dev_err(&i2c->dev, "Failed to register codec: %d\n", ret);
+ regmap_exit(lm49453->regmap);
+ return ret;
+ }
+
+ return ret;
+}
+
+static int __devexit lm49453_i2c_remove(struct i2c_client *client)
+{
+ struct lm49453_priv *lm49453 = i2c_get_clientdata(client);
+
+ snd_soc_unregister_codec(&client->dev);
+ regmap_exit(lm49453->regmap);
+ return 0;
+}
+
+static const struct i2c_device_id lm49453_i2c_id[] = {
+ { "lm49453", 0 },
+ { }
+};
+MODULE_DEVICE_TABLE(i2c, lm49453_i2c_id);
+
+static struct i2c_driver lm49453_i2c_driver = {
+ .driver = {
+ .name = "lm49453",
+ .owner = THIS_MODULE,
+ },
+ .probe = lm49453_i2c_probe,
+ .remove = __devexit_p(lm49453_i2c_remove),
+ .id_table = lm49453_i2c_id,
+};
+
+module_i2c_driver(lm49453_i2c_driver);
+
+MODULE_DESCRIPTION("ASoC LM49453 driver");
+MODULE_AUTHOR("M R Swami Reddy <MR.Swami.Reddy@ti.com>");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/codecs/lm49453.h b/sound/soc/codecs/lm49453.h
new file mode 100644
index 00000000000..a63cfa5c088
--- /dev/null
+++ b/sound/soc/codecs/lm49453.h
@@ -0,0 +1,380 @@
+/*
+ * lm49453.h - LM49453 ALSA Soc Audio drive
+ *
+ * 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 _LM49453_H
+#define _LM49453_H
+
+#include <linux/bitops.h>
+
+/* LM49453_P0 register space for page0 */
+#define LM49453_P0_PMC_SETUP_REG 0x00
+#define LM49453_P0_PLL_CLK_SEL1_REG 0x01
+#define LM49453_P0_PLL_CLK_SEL2_REG 0x02
+#define LM49453_P0_PMC_CLK_DIV_REG 0x03
+#define LM49453_P0_HSDET_CLK_DIV_REG 0x04
+#define LM49453_P0_DMIC_CLK_DIV_REG 0x05
+#define LM49453_P0_ADC_CLK_DIV_REG 0x06
+#define LM49453_P0_DAC_OT_CLK_DIV_REG 0x07
+#define LM49453_P0_PLL_HF_M_REG 0x08
+#define LM49453_P0_PLL_LF_M_REG 0x09
+#define LM49453_P0_PLL_NL_REG 0x0A
+#define LM49453_P0_PLL_N_MODL_REG 0x0B
+#define LM49453_P0_PLL_N_MODH_REG 0x0C
+#define LM49453_P0_PLL_P1_REG 0x0D
+#define LM49453_P0_PLL_P2_REG 0x0E
+#define LM49453_P0_FLL_REF_FREQL_REG 0x0F
+#define LM49453_P0_FLL_REF_FREQH_REG 0x10
+#define LM49453_P0_VCO_TARGETLL_REG 0x11
+#define LM49453_P0_VCO_TARGETLH_REG 0x12
+#define LM49453_P0_VCO_TARGETHL_REG 0x13
+#define LM49453_P0_VCO_TARGETHH_REG 0x14
+#define LM49453_P0_PLL_CONFIG_REG 0x15
+#define LM49453_P0_DAC_CLK_SEL_REG 0x16
+#define LM49453_P0_DAC_HP_CLK_DIV_REG 0x17
+
+/* Analog Mixer Input Stages */
+#define LM49453_P0_MICL_REG 0x20
+#define LM49453_P0_MICR_REG 0x21
+#define LM49453_P0_EP_REG 0x24
+#define LM49453_P0_DIS_PKVL_FB_REG 0x25
+
+/* Analog Mixer Output Stages */
+#define LM49453_P0_ANALOG_MIXER_ADC_REG 0x2E
+
+/*ADC or DAC */
+#define LM49453_P0_ADC_DSP_REG 0x30
+#define LM49453_P0_DAC_DSP_REG 0x31
+
+/* EFFECTS ENABLES */
+#define LM49453_P0_ADC_FX_ENABLES_REG 0x33
+
+/* GPIO */
+#define LM49453_P0_GPIO1_REG 0x38
+#define LM49453_P0_GPIO2_REG 0x39
+#define LM49453_P0_GPIO3_REG 0x3A
+#define LM49453_P0_HAP_CTL_REG 0x3B
+#define LM49453_P0_HAP_FREQ_PROG_LEFTL_REG 0x3C
+#define LM49453_P0_HAP_FREQ_PROG_LEFTH_REG 0x3D
+#define LM49453_P0_HAP_FREQ_PROG_RIGHTL_REG 0x3E
+#define LM49453_P0_HAP_FREQ_PROG_RIGHTH_REG 0x3F
+
+/* DIGITAL MIXER */
+#define LM49453_P0_DMIX_CLK_SEL_REG 0x40
+#define LM49453_P0_PORT1_RX_LVL1_REG 0x41
+#define LM49453_P0_PORT1_RX_LVL2_REG 0x42
+#define LM49453_P0_PORT2_RX_LVL_REG 0x43
+#define LM49453_P0_PORT1_TX1_REG 0x44
+#define LM49453_P0_PORT1_TX2_REG 0x45
+#define LM49453_P0_PORT1_TX3_REG 0x46
+#define LM49453_P0_PORT1_TX4_REG 0x47
+#define LM49453_P0_PORT1_TX5_REG 0x48
+#define LM49453_P0_PORT1_TX6_REG 0x49
+#define LM49453_P0_PORT1_TX7_REG 0x4A
+#define LM49453_P0_PORT1_TX8_REG 0x4B
+#define LM49453_P0_PORT2_TX1_REG 0x4C
+#define LM49453_P0_PORT2_TX2_REG 0x4D
+#define LM49453_P0_STN_SEL_REG 0x4F
+#define LM49453_P0_DACHPL1_REG 0x50
+#define LM49453_P0_DACHPL2_REG 0x51
+#define LM49453_P0_DACHPR1_REG 0x52
+#define LM49453_P0_DACHPR2_REG 0x53
+#define LM49453_P0_DACLOL1_REG 0x54
+#define LM49453_P0_DACLOL2_REG 0x55
+#define LM49453_P0_DACLOR1_REG 0x56
+#define LM49453_P0_DACLOR2_REG 0x57
+#define LM49453_P0_DACLSL1_REG 0x58
+#define LM49453_P0_DACLSL2_REG 0x59
+#define LM49453_P0_DACLSR1_REG 0x5A
+#define LM49453_P0_DACLSR2_REG 0x5B
+#define LM49453_P0_DACHAL1_REG 0x5C
+#define LM49453_P0_DACHAL2_REG 0x5D
+#define LM49453_P0_DACHAR1_REG 0x5E
+#define LM49453_P0_DACHAR2_REG 0x5F
+
+/* AUDIO PORT 1 (TDM) */
+#define LM49453_P0_AUDIO_PORT1_BASIC_REG 0x60
+#define LM49453_P0_AUDIO_PORT1_CLK_GEN1_REG 0x61
+#define LM49453_P0_AUDIO_PORT1_CLK_GEN2_REG 0x62
+#define LM49453_P0_AUDIO_PORT1_CLK_GEN3_REG 0x63
+#define LM49453_P0_AUDIO_PORT1_SYNC_RATE_REG 0x64
+#define LM49453_P0_AUDIO_PORT1_SYNC_SDO_SETUP_REG 0x65
+#define LM49453_P0_AUDIO_PORT1_DATA_WIDTH_REG 0x66
+#define LM49453_P0_AUDIO_PORT1_RX_MSB_REG 0x67
+#define LM49453_P0_AUDIO_PORT1_TX_MSB_REG 0x68
+#define LM49453_P0_AUDIO_PORT1_TDM_CHANNELS_REG 0x69
+
+/* AUDIO PORT 2 */
+#define LM49453_P0_AUDIO_PORT2_BASIC_REG 0x6A
+#define LM49453_P0_AUDIO_PORT2_CLK_GEN1_REG 0x6B
+#define LM49453_P0_AUDIO_PORT2_CLK_GEN2_REG 0x6C
+#define LM49453_P0_AUDIO_PORT2_SYNC_GEN_REG 0x6D
+#define LM49453_P0_AUDIO_PORT2_DATA_WIDTH_REG 0x6E
+#define LM49453_P0_AUDIO_PORT2_RX_MODE_REG 0x6F
+#define LM49453_P0_AUDIO_PORT2_TX_MODE_REG 0x70
+
+/* SAMPLE RATE */
+#define LM49453_P0_PORT1_SR_LSB_REG 0x79
+#define LM49453_P0_PORT1_SR_MSB_REG 0x7A
+#define LM49453_P0_PORT2_SR_LSB_REG 0x7B
+#define LM49453_P0_PORT2_SR_MSB_REG 0x7C
+
+/* EFFECTS - HPFs */
+#define LM49453_P0_HPF_REG 0x80
+
+/* EFFECTS ADC ALC */
+#define LM49453_P0_ADC_ALC1_REG 0x82
+#define LM49453_P0_ADC_ALC2_REG 0x83
+#define LM49453_P0_ADC_ALC3_REG 0x84
+#define LM49453_P0_ADC_ALC4_REG 0x85
+#define LM49453_P0_ADC_ALC5_REG 0x86
+#define LM49453_P0_ADC_ALC6_REG 0x87
+#define LM49453_P0_ADC_ALC7_REG 0x88
+#define LM49453_P0_ADC_ALC8_REG 0x89
+#define LM49453_P0_DMIC1_LEVELL_REG 0x8A
+#define LM49453_P0_DMIC1_LEVELR_REG 0x8B
+#define LM49453_P0_DMIC2_LEVELL_REG 0x8C
+#define LM49453_P0_DMIC2_LEVELR_REG 0x8D
+#define LM49453_P0_ADC_LEVELL_REG 0x8E
+#define LM49453_P0_ADC_LEVELR_REG 0x8F
+#define LM49453_P0_DAC_HP_LEVELL_REG 0x90
+#define LM49453_P0_DAC_HP_LEVELR_REG 0x91
+#define LM49453_P0_DAC_LO_LEVELL_REG 0x92
+#define LM49453_P0_DAC_LO_LEVELR_REG 0x93
+#define LM49453_P0_DAC_LS_LEVELL_REG 0x94
+#define LM49453_P0_DAC_LS_LEVELR_REG 0x95
+#define LM49453_P0_DAC_HA_LEVELL_REG 0x96
+#define LM49453_P0_DAC_HA_LEVELR_REG 0x97
+#define LM49453_P0_SOFT_MUTE_REG 0x98
+#define LM49453_P0_DMIC_MUTE_CFG_REG 0x99
+#define LM49453_P0_ADC_MUTE_CFG_REG 0x9A
+#define LM49453_P0_DAC_MUTE_CFG_REG 0x9B
+
+/*DIGITAL MIC1 */
+#define LM49453_P0_DIGITAL_MIC1_CONFIG_REG 0xB0
+#define LM49453_P0_DIGITAL_MIC1_DATA_DELAYL_REG 0xB1
+#define LM49453_P0_DIGITAL_MIC1_DATA_DELAYR_REG 0xB2
+
+/*DIGITAL MIC2 */
+#define LM49453_P0_DIGITAL_MIC2_CONFIG_REG 0xB3
+#define LM49453_P0_DIGITAL_MIC2_DATA_DELAYL_REG 0xB4
+#define LM49453_P0_DIGITAL_MIC2_DATA_DELAYR_REG 0xB5
+
+/* ADC DECIMATOR */
+#define LM49453_P0_ADC_DECIMATOR_REG 0xB6
+
+/* DAC CONFIGURE */
+#define LM49453_P0_DAC_CONFIG_REG 0xB7
+
+/* SIDETONE */
+#define LM49453_P0_STN_VOL_ADCL_REG 0xB8
+#define LM49453_P0_STN_VOL_ADCR_REG 0xB9
+#define LM49453_P0_STN_VOL_DMIC1L_REG 0xBA
+#define LM49453_P0_STN_VOL_DMIC1R_REG 0xBB
+#define LM49453_P0_STN_VOL_DMIC2L_REG 0xBC
+#define LM49453_P0_STN_VOL_DMIC2R_REG 0xBD
+
+/* ADC/DAC CLIPPING MONITORS (Read Only/Write to Clear) */
+#define LM49453_P0_ADC_DEC_CLIP_REG 0xC2
+#define LM49453_P0_ADC_HPF_CLIP_REG 0xC3
+#define LM49453_P0_ADC_LVL_CLIP_REG 0xC4
+#define LM49453_P0_DAC_LVL_CLIP_REG 0xC5
+
+/* ADC ALC EFFECT MONITORS (Read Only) */
+#define LM49453_P0_ADC_LVLMONL_REG 0xC8
+#define LM49453_P0_ADC_LVLMONR_REG 0xC9
+#define LM49453_P0_ADC_ALCMONL_REG 0xCA
+#define LM49453_P0_ADC_ALCMONR_REG 0xCB
+#define LM49453_P0_ADC_MUTED_REG 0xCC
+#define LM49453_P0_DAC_MUTED_REG 0xCD
+
+/* HEADSET DETECT */
+#define LM49453_P0_HSD_PPB_LONG_CNT_LIMITL_REG 0xD0
+#define LM49453_P0_HSD_PPB_LONG_CNT_LIMITR_REG 0xD1
+#define LM49453_P0_HSD_PIN3_4_EX_LOOP_CNT_LIMITL_REG 0xD2
+#define LM49453_P0_HSD_PIN3_4_EX_LOOP_CNT_LIMITH_REG 0xD3
+#define LM49453_P0_HSD_TIMEOUT1_REG 0xD4
+#define LM49453_P0_HSD_TIMEOUT2_REG 0xD5
+#define LM49453_P0_HSD_TIMEOUT3_REG 0xD6
+#define LM49453_P0_HSD_PIN3_4_CFG_REG 0xD7
+#define LM49453_P0_HSD_IRQ1_REG 0xD8
+#define LM49453_P0_HSD_IRQ2_REG 0xD9
+#define LM49453_P0_HSD_IRQ3_REG 0xDA
+#define LM49453_P0_HSD_IRQ4_REG 0xDB
+#define LM49453_P0_HSD_IRQ_MASK1_REG 0xDC
+#define LM49453_P0_HSD_IRQ_MASK2_REG 0xDD
+#define LM49453_P0_HSD_IRQ_MASK3_REG 0xDE
+#define LM49453_P0_HSD_R_HPLL_REG 0xE0
+#define LM49453_P0_HSD_R_HPLH_REG 0xE1
+#define LM49453_P0_HSD_R_HPLU_REG 0xE2
+#define LM49453_P0_HSD_R_HPRL_REG 0xE3
+#define LM49453_P0_HSD_R_HPRH_REG 0xE4
+#define LM49453_P0_HSD_R_HPRU_REG 0xE5
+#define LM49453_P0_HSD_VEL_L_FINALL_REG 0xE6
+#define LM49453_P0_HSD_VEL_L_FINALH_REG 0xE7
+#define LM49453_P0_HSD_VEL_L_FINALU_REG 0xE8
+#define LM49453_P0_HSD_RO_FINALL_REG 0xE9
+#define LM49453_P0_HSD_RO_FINALH_REG 0xEA
+#define LM49453_P0_HSD_RO_FINALU_REG 0xEB
+#define LM49453_P0_HSD_VMIC_BIAS_FINALL_REG 0xEC
+#define LM49453_P0_HSD_VMIC_BIAS_FINALH_REG 0xED
+#define LM49453_P0_HSD_VMIC_BIAS_FINALU_REG 0xEE
+#define LM49453_P0_HSD_PIN_CONFIG_REG 0xEF
+#define LM49453_P0_HSD_PLUG_DETECT_BB_IRQ_STATUS1_REG 0xF1
+#define LM49453_P0_HSD_PLUG_DETECT_BB_IRQ_STATUS2_REG 0xF2
+#define LM49453_P0_HSD_PLUG_DETECT_BB_IRQ_STATUS3_REG 0xF3
+#define LM49453_P0_HSD_PLUG_DETECT_BB_IRQ_STATEL_REG 0xF4
+#define LM49453_P0_HSD_PLUG_DETECT_BB_IRQ_STATEH_REG 0xF5
+
+/* I/O PULLDOWN CONFIG */
+#define LM49453_P0_PULL_CONFIG1_REG 0xF8
+#define LM49453_P0_PULL_CONFIG2_REG 0xF9
+#define LM49453_P0_PULL_CONFIG3_REG 0xFA
+
+/* RESET */
+#define LM49453_P0_RESET_REG 0xFE
+
+/* PAGE */
+#define LM49453_PAGE_REG 0xFF
+
+#define LM49453_MAX_REGISTER (0xFF+1)
+
+/* LM49453_P0_PMC_SETUP_REG (0x00h) */
+#define LM49453_PMC_SETUP_CHIP_EN (BIT(1)|BIT(0))
+#define LM49453_PMC_SETUP_PLL_EN BIT(2)
+#define LM49453_PMC_SETUP_PLL_P2_EN BIT(3)
+#define LM49453_PMC_SETUP_PLL_FLL BIT(4)
+#define LM49453_PMC_SETUP_MCLK_OVER BIT(5)
+#define LM49453_PMC_SETUP_RTC_CLK_OVER BIT(6)
+#define LM49453_PMC_SETUP_CHIP_ACTIVE BIT(7)
+
+/* Chip Enable bits */
+#define LM49453_CHIP_EN_SHUTDOWN 0x00
+#define LM49453_CHIP_EN 0x01
+#define LM49453_CHIP_EN_HSD_DETECT 0x02
+#define LM49453_CHIP_EN_INVALID_HSD 0x03
+
+/* LM49453_P0_PLL_CLK_SEL1_REG (0x01h) */
+#define LM49453_CLK_SEL1_MCLK_SEL 0x11
+#define LM49453_CLK_SEL1_RTC_SEL 0x11
+#define LM49453_CLK_SEL1_PORT1_SEL 0x10
+#define LM49453_CLK_SEL1_PORT2_SEL 0x11
+
+/* LM49453_P0_PLL_CLK_SEL2_REG (0x02h) */
+#define LM49453_CLK_SEL2_ADC_CLK_SEL 0x38
+
+/* LM49453_P0_FLL_REF_FREQL_REG (0x0F) */
+#define LM49453_FLL_REF_FREQ_VAL 0x8ca0001
+
+/* LM49453_P0_VCO_TARGETLL_REG (0x11) */
+#define LM49453_VCO_TARGET_VAL 0x8ca0001
+
+/* LM49453_P0_ADC_DSP_REG (0x30h) */
+#define LM49453_ADC_DSP_ADC_MUTEL BIT(0)
+#define LM49453_ADC_DSP_ADC_MUTER BIT(1)
+#define LM49453_ADC_DSP_DMIC1_MUTEL BIT(2)
+#define LM49453_ADC_DSP_DMIC1_MUTER BIT(3)
+#define LM49453_ADC_DSP_DMIC2_MUTEL BIT(4)
+#define LM49453_ADC_DSP_DMIC2_MUTER BIT(5)
+#define LM49453_ADC_DSP_MUTE_ALL 0x3F
+
+/* LM49453_P0_DAC_DSP_REG (0x31h) */
+#define LM49453_DAC_DSP_MUTE_ALL 0xFF
+
+/* LM49453_P0_AUDIO_PORT1_BASIC_REG (0x60h) */
+#define LM49453_AUDIO_PORT1_BASIC_FMT_MASK (BIT(4)|BIT(3))
+#define LM49453_AUDIO_PORT1_BASIC_CLK_MS BIT(3)
+#define LM49453_AUDIO_PORT1_BASIC_SYNC_MS BIT(4)
+
+/* LM49453_P0_RESET_REG (0xFEh) */
+#define LM49453_RESET_REG_RST BIT(0)
+
+/* Page select register bits (0xFF) */
+#define LM49453_PAGE0_SELECT 0x0
+#define LM49453_PAGE1_SELECT 0x1
+
+/* LM49453_P0_HSD_PIN3_4_CFG_REG (Jack Pin config - 0xD7) */
+#define LM49453_JACK_DISABLE 0x00
+#define LM49453_JACK_CONFIG1 0x01
+#define LM49453_JACK_CONFIG2 0x02
+#define LM49453_JACK_CONFIG3 0x03
+#define LM49453_JACK_CONFIG4 0x04
+#define LM49453_JACK_CONFIG5 0x05
+
+/* Page 1 REGISTERS */
+
+/* SIDETONE */
+#define LM49453_P1_SIDETONE_SA0L_REG 0x80
+#define LM49453_P1_SIDETONE_SA0H_REG 0x81
+#define LM49453_P1_SIDETONE_SAB0U_REG 0x82
+#define LM49453_P1_SIDETONE_SB0L_REG 0x83
+#define LM49453_P1_SIDETONE_SB0H_REG 0x84
+#define LM49453_P1_SIDETONE_SH0L_REG 0x85
+#define LM49453_P1_SIDETONE_SH0H_REG 0x86
+#define LM49453_P1_SIDETONE_SH0U_REG 0x87
+#define LM49453_P1_SIDETONE_SA1L_REG 0x88
+#define LM49453_P1_SIDETONE_SA1H_REG 0x89
+#define LM49453_P1_SIDETONE_SAB1U_REG 0x8A
+#define LM49453_P1_SIDETONE_SB1L_REG 0x8B
+#define LM49453_P1_SIDETONE_SB1H_REG 0x8C
+#define LM49453_P1_SIDETONE_SH1L_REG 0x8D
+#define LM49453_P1_SIDETONE_SH1H_REG 0x8E
+#define LM49453_P1_SIDETONE_SH1U_REG 0x8F
+#define LM49453_P1_SIDETONE_SA2L_REG 0x90
+#define LM49453_P1_SIDETONE_SA2H_REG 0x91
+#define LM49453_P1_SIDETONE_SAB2U_REG 0x92
+#define LM49453_P1_SIDETONE_SB2L_REG 0x93
+#define LM49453_P1_SIDETONE_SB2H_REG 0x94
+#define LM49453_P1_SIDETONE_SH2L_REG 0x95
+#define LM49453_P1_SIDETONE_SH2H_REG 0x96
+#define LM49453_P1_SIDETONE_SH2U_REG 0x97
+#define LM49453_P1_SIDETONE_SA3L_REG 0x98
+#define LM49453_P1_SIDETONE_SA3H_REG 0x99
+#define LM49453_P1_SIDETONE_SAB3U_REG 0x9A
+#define LM49453_P1_SIDETONE_SB3L_REG 0x9B
+#define LM49453_P1_SIDETONE_SB3H_REG 0x9C
+#define LM49453_P1_SIDETONE_SH3L_REG 0x9D
+#define LM49453_P1_SIDETONE_SH3H_REG 0x9E
+#define LM49453_P1_SIDETONE_SH3U_REG 0x9F
+#define LM49453_P1_SIDETONE_SA4L_REG 0xA0
+#define LM49453_P1_SIDETONE_SA4H_REG 0xA1
+#define LM49453_P1_SIDETONE_SAB4U_REG 0xA2
+#define LM49453_P1_SIDETONE_SB4L_REG 0xA3
+#define LM49453_P1_SIDETONE_SB4H_REG 0xA4
+#define LM49453_P1_SIDETONE_SH4L_REG 0xA5
+#define LM49453_P1_SIDETONE_SH4H_REG 0xA6
+#define LM49453_P1_SIDETONE_SH4U_REG 0xA7
+#define LM49453_P1_SIDETONE_SA5L_REG 0xA8
+#define LM49453_P1_SIDETONE_SA5H_REG 0xA9
+#define LM49453_P1_SIDETONE_SAB5U_REG 0xAA
+#define LM49453_P1_SIDETONE_SB5L_REG 0xAB
+#define LM49453_P1_SIDETONE_SB5H_REG 0xAC
+#define LM49453_P1_SIDETONE_SH5L_REG 0xAD
+#define LM49453_P1_SIDETONE_SH5H_REG 0xAE
+#define LM49453_P1_SIDETONE_SH5U_REG 0xAF
+
+/* CHARGE PUMP CONFIG */
+#define LM49453_P1_CP_CONFIG1_REG 0xB0
+#define LM49453_P1_CP_CONFIG2_REG 0xB1
+#define LM49453_P1_CP_CONFIG3_REG 0xB2
+#define LM49453_P1_CP_CONFIG4_REG 0xB3
+#define LM49453_P1_CP_LA_VTH1L_REG 0xB4
+#define LM49453_P1_CP_LA_VTH1M_REG 0xB5
+#define LM49453_P1_CP_LA_VTH2L_REG 0xB6
+#define LM49453_P1_CP_LA_VTH2M_REG 0xB7
+#define LM49453_P1_CP_LA_VTH3L_REG 0xB8
+#define LM49453_P1_CP_LA_VTH3H_REG 0xB9
+#define LM49453_P1_CP_CLK_DIV_REG 0xBA
+
+/* DAC */
+#define LM49453_P1_DAC_CHOP_REG 0xC0
+
+#define LM49453_CLK_SRC_MCLK 1
+#endif
diff --git a/sound/soc/codecs/max98095.c b/sound/soc/codecs/max98095.c
index 0bb511a0388..35179e2c23c 100644
--- a/sound/soc/codecs/max98095.c
+++ b/sound/soc/codecs/max98095.c
@@ -24,6 +24,7 @@
#include <linux/slab.h>
#include <asm/div64.h>
#include <sound/max98095.h>
+#include <sound/jack.h>
#include "max98095.h"
enum max98095_type {
@@ -51,6 +52,8 @@ struct max98095_priv {
u8 lin_state;
unsigned int mic1pre;
unsigned int mic2pre;
+ struct snd_soc_jack *headphone_jack;
+ struct snd_soc_jack *mic_jack;
};
static const u8 max98095_reg_def[M98095_REG_CNT] = {
@@ -2173,9 +2176,125 @@ static void max98095_handle_pdata(struct snd_soc_codec *codec)
max98095_handle_bq_pdata(codec);
}
+static irqreturn_t max98095_report_jack(int irq, void *data)
+{
+ struct snd_soc_codec *codec = data;
+ struct max98095_priv *max98095 = snd_soc_codec_get_drvdata(codec);
+ unsigned int value;
+ int hp_report = 0;
+ int mic_report = 0;
+
+ /* Read the Jack Status Register */
+ value = snd_soc_read(codec, M98095_007_JACK_AUTO_STS);
+
+ /* If ddone is not set, then detection isn't finished yet */
+ if ((value & M98095_DDONE) == 0)
+ return IRQ_NONE;
+
+ /* if hp, check its bit, and if set, clear it */
+ if ((value & M98095_HP_IN || value & M98095_LO_IN) &&
+ max98095->headphone_jack)
+ hp_report |= SND_JACK_HEADPHONE;
+
+ /* if mic, check its bit, and if set, clear it */
+ if ((value & M98095_MIC_IN) && max98095->mic_jack)
+ mic_report |= SND_JACK_MICROPHONE;
+
+ if (max98095->headphone_jack == max98095->mic_jack) {
+ snd_soc_jack_report(max98095->headphone_jack,
+ hp_report | mic_report,
+ SND_JACK_HEADSET);
+ } else {
+ if (max98095->headphone_jack)
+ snd_soc_jack_report(max98095->headphone_jack,
+ hp_report, SND_JACK_HEADPHONE);
+ if (max98095->mic_jack)
+ snd_soc_jack_report(max98095->mic_jack,
+ mic_report, SND_JACK_MICROPHONE);
+ }
+
+ return IRQ_HANDLED;
+}
+
+int max98095_jack_detect_enable(struct snd_soc_codec *codec)
+{
+ struct max98095_priv *max98095 = snd_soc_codec_get_drvdata(codec);
+ int ret = 0;
+ int detect_enable = M98095_JDEN;
+ unsigned int slew = M98095_DEFAULT_SLEW_DELAY;
+
+ if (max98095->pdata->jack_detect_pin5en)
+ detect_enable |= M98095_PIN5EN;
+
+ if (max98095->pdata->jack_detect_delay)
+ slew = max98095->pdata->jack_detect_delay;
+
+ ret = snd_soc_write(codec, M98095_08E_JACK_DC_SLEW, slew);
+ if (ret < 0) {
+ dev_err(codec->dev, "Failed to cfg auto detect %d\n", ret);
+ return ret;
+ }
+
+ /* configure auto detection to be enabled */
+ ret = snd_soc_write(codec, M98095_089_JACK_DET_AUTO, detect_enable);
+ if (ret < 0) {
+ dev_err(codec->dev, "Failed to cfg auto detect %d\n", ret);
+ return ret;
+ }
+
+ return ret;
+}
+
+int max98095_jack_detect_disable(struct snd_soc_codec *codec)
+{
+ int ret = 0;
+
+ /* configure auto detection to be disabled */
+ ret = snd_soc_write(codec, M98095_089_JACK_DET_AUTO, 0x0);
+ if (ret < 0) {
+ dev_err(codec->dev, "Failed to cfg auto detect %d\n", ret);
+ return ret;
+ }
+
+ return ret;
+}
+
+int max98095_jack_detect(struct snd_soc_codec *codec,
+ struct snd_soc_jack *hp_jack, struct snd_soc_jack *mic_jack)
+{
+ struct max98095_priv *max98095 = snd_soc_codec_get_drvdata(codec);
+ struct i2c_client *client = to_i2c_client(codec->dev);
+ int ret = 0;
+
+ max98095->headphone_jack = hp_jack;
+ max98095->mic_jack = mic_jack;
+
+ /* only progress if we have at least 1 jack pointer */
+ if (!hp_jack && !mic_jack)
+ return -EINVAL;
+
+ max98095_jack_detect_enable(codec);
+
+ /* enable interrupts for headphone jack detection */
+ ret = snd_soc_update_bits(codec, M98095_013_JACK_INT_EN,
+ M98095_IDDONE, M98095_IDDONE);
+ if (ret < 0) {
+ dev_err(codec->dev, "Failed to cfg jack irqs %d\n", ret);
+ return ret;
+ }
+
+ max98095_report_jack(client->irq, codec);
+ return 0;
+}
+
#ifdef CONFIG_PM
static int max98095_suspend(struct snd_soc_codec *codec)
{
+ struct max98095_priv *max98095 = snd_soc_codec_get_drvdata(codec);
+
+ if (max98095->headphone_jack || max98095->mic_jack)
+ max98095_jack_detect_disable(codec);
+
max98095_set_bias_level(codec, SND_SOC_BIAS_OFF);
return 0;
@@ -2183,8 +2302,16 @@ static int max98095_suspend(struct snd_soc_codec *codec)
static int max98095_resume(struct snd_soc_codec *codec)
{
+ struct max98095_priv *max98095 = snd_soc_codec_get_drvdata(codec);
+ struct i2c_client *client = to_i2c_client(codec->dev);
+
max98095_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+ if (max98095->headphone_jack || max98095->mic_jack) {
+ max98095_jack_detect_enable(codec);
+ max98095_report_jack(client->irq, codec);
+ }
+
return 0;
}
#else
@@ -2227,6 +2354,7 @@ static int max98095_probe(struct snd_soc_codec *codec)
{
struct max98095_priv *max98095 = snd_soc_codec_get_drvdata(codec);
struct max98095_cdata *cdata;
+ struct i2c_client *client;
int ret = 0;
ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_I2C);
@@ -2238,6 +2366,8 @@ static int max98095_probe(struct snd_soc_codec *codec)
/* reset the codec, the DSP core, and disable all interrupts */
max98095_reset(codec);
+ client = to_i2c_client(codec->dev);
+
/* initialize private data */
max98095->sysclk = (unsigned)-1;
@@ -2266,11 +2396,23 @@ static int max98095_probe(struct snd_soc_codec *codec)
max98095->mic1pre = 0;
max98095->mic2pre = 0;
+ if (client->irq) {
+ /* register an audio interrupt */
+ ret = request_threaded_irq(client->irq, NULL,
+ max98095_report_jack,
+ IRQF_TRIGGER_FALLING | IRQF_TRIGGER_RISING,
+ "max98095", codec);
+ if (ret) {
+ dev_err(codec->dev, "Failed to request IRQ: %d\n", ret);
+ goto err_access;
+ }
+ }
+
ret = snd_soc_read(codec, M98095_0FF_REV_ID);
if (ret < 0) {
dev_err(codec->dev, "Failure reading hardware revision: %d\n",
ret);
- goto err_access;
+ goto err_irq;
}
dev_info(codec->dev, "Hardware revision: %c\n", ret - 0x40 + 'A');
@@ -2306,14 +2448,28 @@ static int max98095_probe(struct snd_soc_codec *codec)
max98095_add_widgets(codec);
+ return 0;
+
+err_irq:
+ if (client->irq)
+ free_irq(client->irq, codec);
err_access:
return ret;
}
static int max98095_remove(struct snd_soc_codec *codec)
{
+ struct max98095_priv *max98095 = snd_soc_codec_get_drvdata(codec);
+ struct i2c_client *client = to_i2c_client(codec->dev);
+
max98095_set_bias_level(codec, SND_SOC_BIAS_OFF);
+ if (max98095->headphone_jack || max98095->mic_jack)
+ max98095_jack_detect_disable(codec);
+
+ if (client->irq)
+ free_irq(client->irq, codec);
+
return 0;
}
diff --git a/sound/soc/codecs/max98095.h b/sound/soc/codecs/max98095.h
index 891584a0eb0..2ebbe4e894b 100644
--- a/sound/soc/codecs/max98095.h
+++ b/sound/soc/codecs/max98095.h
@@ -175,11 +175,23 @@
/* MAX98095 Registers Bit Fields */
+/* M98095_007_JACK_AUTO_STS */
+ #define M98095_MIC_IN (1<<3)
+ #define M98095_LO_IN (1<<5)
+ #define M98095_HP_IN (1<<6)
+ #define M98095_DDONE (1<<7)
+
/* M98095_00F_HOST_CFG */
#define M98095_SEG (1<<0)
#define M98095_XTEN (1<<1)
#define M98095_MDLLEN (1<<2)
+/* M98095_013_JACK_INT_EN */
+ #define M98095_IMIC_IN (1<<3)
+ #define M98095_ILO_IN (1<<5)
+ #define M98095_IHP_IN (1<<6)
+ #define M98095_IDDONE (1<<7)
+
/* M98095_027_DAI1_CLKMODE, M98095_031_DAI2_CLKMODE, M98095_03B_DAI3_CLKMODE */
#define M98095_CLKMODE_MASK 0xFF
@@ -255,6 +267,10 @@
#define M98095_EQ2EN (1<<1)
#define M98095_EQ1EN (1<<0)
+/* M98095_089_JACK_DET_AUTO */
+ #define M98095_PIN5EN (1<<2)
+ #define M98095_JDEN (1<<7)
+
/* M98095_090_PWR_EN_IN */
#define M98095_INEN (1<<7)
#define M98095_MB2EN (1<<3)
@@ -296,4 +312,10 @@
#define M98095_174_DAI1_BQ_BASE 0x74
#define M98095_17E_DAI2_BQ_BASE 0x7E
+/* Default Delay used in Slew Rate Calculation for Jack detection */
+#define M98095_DEFAULT_SLEW_DELAY 0x18
+
+extern int max98095_jack_detect(struct snd_soc_codec *codec,
+ struct snd_soc_jack *hp_jack, struct snd_soc_jack *mic_jack);
+
#endif
diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c
new file mode 100644
index 00000000000..6276e352125
--- /dev/null
+++ b/sound/soc/codecs/mc13783.c
@@ -0,0 +1,786 @@
+/*
+ * Copyright 2008 Juergen Beisert, kernel@pengutronix.de
+ * Copyright 2009 Sascha Hauer, s.hauer@pengutronix.de
+ * Copyright 2012 Philippe Retornaz, philippe.retornaz@epfl.ch
+ *
+ * Initial development of this code was funded by
+ * Phytec Messtechnik GmbH, http://www.phytec.de
+ *
+ * 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., 51 Franklin Street, Fifth Floor, Boston,
+ * MA 02110-1301, USA.
+ */
+#include <linux/module.h>
+#include <linux/device.h>
+#include <linux/mfd/mc13xxx.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/control.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/initval.h>
+#include <sound/soc-dapm.h>
+
+#include "mc13783.h"
+
+#define MC13783_AUDIO_RX0 36
+#define MC13783_AUDIO_RX1 37
+#define MC13783_AUDIO_TX 38
+#define MC13783_SSI_NETWORK 39
+#define MC13783_AUDIO_CODEC 40
+#define MC13783_AUDIO_DAC 41
+
+#define AUDIO_RX0_ALSPEN (1 << 5)
+#define AUDIO_RX0_ALSPSEL (1 << 7)
+#define AUDIO_RX0_ADDCDC (1 << 21)
+#define AUDIO_RX0_ADDSTDC (1 << 22)
+#define AUDIO_RX0_ADDRXIN (1 << 23)
+
+#define AUDIO_RX1_PGARXEN (1 << 0);
+#define AUDIO_RX1_PGASTEN (1 << 5)
+#define AUDIO_RX1_ARXINEN (1 << 10)
+
+#define AUDIO_TX_AMC1REN (1 << 5)
+#define AUDIO_TX_AMC1LEN (1 << 7)
+#define AUDIO_TX_AMC2EN (1 << 9)
+#define AUDIO_TX_ATXINEN (1 << 11)
+#define AUDIO_TX_RXINREC (1 << 13)
+
+#define SSI_NETWORK_CDCTXRXSLOT(x) (((x) & 0x3) << 2)
+#define SSI_NETWORK_CDCTXSECSLOT(x) (((x) & 0x3) << 4)
+#define SSI_NETWORK_CDCRXSECSLOT(x) (((x) & 0x3) << 6)
+#define SSI_NETWORK_CDCRXSECGAIN(x) (((x) & 0x3) << 8)
+#define SSI_NETWORK_CDCSUMGAIN(x) (1 << 10)
+#define SSI_NETWORK_CDCFSDLY(x) (1 << 11)
+#define SSI_NETWORK_DAC_SLOTS_8 (1 << 12)
+#define SSI_NETWORK_DAC_SLOTS_4 (2 << 12)
+#define SSI_NETWORK_DAC_SLOTS_2 (3 << 12)
+#define SSI_NETWORK_DAC_SLOT_MASK (3 << 12)
+#define SSI_NETWORK_DAC_RXSLOT_0_1 (0 << 14)
+#define SSI_NETWORK_DAC_RXSLOT_2_3 (1 << 14)
+#define SSI_NETWORK_DAC_RXSLOT_4_5 (2 << 14)
+#define SSI_NETWORK_DAC_RXSLOT_6_7 (3 << 14)
+#define SSI_NETWORK_DAC_RXSLOT_MASK (3 << 14)
+#define SSI_NETWORK_STDCRXSECSLOT(x) (((x) & 0x3) << 16)
+#define SSI_NETWORK_STDCRXSECGAIN(x) (((x) & 0x3) << 18)
+#define SSI_NETWORK_STDCSUMGAIN (1 << 20)
+
+/*
+ * MC13783_AUDIO_CODEC and MC13783_AUDIO_DAC mostly share the same
+ * register layout
+ */
+#define AUDIO_SSI_SEL (1 << 0)
+#define AUDIO_CLK_SEL (1 << 1)
+#define AUDIO_CSM (1 << 2)
+#define AUDIO_BCL_INV (1 << 3)
+#define AUDIO_CFS_INV (1 << 4)
+#define AUDIO_CFS(x) (((x) & 0x3) << 5)
+#define AUDIO_CLK(x) (((x) & 0x7) << 7)
+#define AUDIO_C_EN (1 << 11)
+#define AUDIO_C_CLK_EN (1 << 12)
+#define AUDIO_C_RESET (1 << 15)
+
+#define AUDIO_CODEC_CDCFS8K16K (1 << 10)
+#define AUDIO_DAC_CFS_DLY_B (1 << 10)
+
+struct mc13783_priv {
+ struct snd_soc_codec codec;
+ struct mc13xxx *mc13xxx;
+
+ enum mc13783_ssi_port adc_ssi_port;
+ enum mc13783_ssi_port dac_ssi_port;
+};
+
+static unsigned int mc13783_read(struct snd_soc_codec *codec,
+ unsigned int reg)
+{
+ struct mc13783_priv *priv = snd_soc_codec_get_drvdata(codec);
+ unsigned int value = 0;
+
+ mc13xxx_lock(priv->mc13xxx);
+
+ mc13xxx_reg_read(priv->mc13xxx, reg, &value);
+
+ mc13xxx_unlock(priv->mc13xxx);
+
+ return value;
+}
+
+static int mc13783_write(struct snd_soc_codec *codec,
+ unsigned int reg, unsigned int value)
+{
+ struct mc13783_priv *priv = snd_soc_codec_get_drvdata(codec);
+ int ret;
+
+ mc13xxx_lock(priv->mc13xxx);
+
+ ret = mc13xxx_reg_write(priv->mc13xxx, reg, value);
+
+ mc13xxx_unlock(priv->mc13xxx);
+
+ return ret;
+}
+
+/* Mapping between sample rates and register value */
+static unsigned int mc13783_rates[] = {
+ 8000, 11025, 12000, 16000,
+ 22050, 24000, 32000, 44100,
+ 48000, 64000, 96000
+};
+
+static int mc13783_pcm_hw_params_dac(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;
+ unsigned int rate = params_rate(params);
+ int i;
+
+ for (i = 0; i < ARRAY_SIZE(mc13783_rates); i++) {
+ if (rate == mc13783_rates[i]) {
+ snd_soc_update_bits(codec, MC13783_AUDIO_DAC,
+ 0xf << 17, i << 17);
+ return 0;
+ }
+ }
+
+ return -EINVAL;
+}
+
+static int mc13783_pcm_hw_params_codec(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;
+ unsigned int rate = params_rate(params);
+ unsigned int val;
+
+ switch (rate) {
+ case 8000:
+ val = 0;
+ break;
+ case 16000:
+ val = AUDIO_CODEC_CDCFS8K16K;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ snd_soc_update_bits(codec, MC13783_AUDIO_CODEC, AUDIO_CODEC_CDCFS8K16K,
+ val);
+
+ return 0;
+}
+
+static int mc13783_pcm_hw_params_sync(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ return mc13783_pcm_hw_params_dac(substream, params, dai);
+ else
+ return mc13783_pcm_hw_params_codec(substream, params, dai);
+}
+
+static int mc13783_set_fmt(struct snd_soc_dai *dai, unsigned int fmt,
+ unsigned int reg)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ unsigned int val = 0;
+ unsigned int mask = AUDIO_CFS(3) | AUDIO_BCL_INV | AUDIO_CFS_INV |
+ AUDIO_CSM | AUDIO_C_CLK_EN | AUDIO_C_RESET;
+
+
+ /* DAI mode */
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ val |= AUDIO_CFS(2);
+ break;
+ case SND_SOC_DAIFMT_DSP_A:
+ val |= AUDIO_CFS(1);
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ /* DAI clock inversion */
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_NB_NF:
+ val |= AUDIO_BCL_INV;
+ break;
+ case SND_SOC_DAIFMT_NB_IF:
+ val |= AUDIO_BCL_INV | AUDIO_CFS_INV;
+ break;
+ case SND_SOC_DAIFMT_IB_NF:
+ break;
+ case SND_SOC_DAIFMT_IB_IF:
+ val |= AUDIO_CFS_INV;
+ break;
+ }
+
+ /* DAI clock master masks */
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBM_CFM:
+ val |= AUDIO_C_CLK_EN;
+ break;
+ case SND_SOC_DAIFMT_CBS_CFS:
+ val |= AUDIO_CSM;
+ break;
+ case SND_SOC_DAIFMT_CBM_CFS:
+ case SND_SOC_DAIFMT_CBS_CFM:
+ return -EINVAL;
+ }
+
+ val |= AUDIO_C_RESET;
+
+ snd_soc_update_bits(codec, reg, mask, val);
+
+ return 0;
+}
+
+static int mc13783_set_fmt_async(struct snd_soc_dai *dai, unsigned int fmt)
+{
+ if (dai->id == MC13783_ID_STEREO_DAC)
+ return mc13783_set_fmt(dai, fmt, MC13783_AUDIO_DAC);
+ else
+ return mc13783_set_fmt(dai, fmt, MC13783_AUDIO_CODEC);
+}
+
+static int mc13783_set_fmt_sync(struct snd_soc_dai *dai, unsigned int fmt)
+{
+ int ret;
+
+ ret = mc13783_set_fmt(dai, fmt, MC13783_AUDIO_DAC);
+ if (ret)
+ return ret;
+
+ /*
+ * In synchronous mode force the voice codec into slave mode
+ * so that the clock / framesync from the stereo DAC is used
+ */
+ fmt &= ~SND_SOC_DAIFMT_MASTER_MASK;
+ fmt |= SND_SOC_DAIFMT_CBS_CFS;
+ ret = mc13783_set_fmt(dai, fmt, MC13783_AUDIO_CODEC);
+
+ return ret;
+}
+
+static int mc13783_sysclk[] = {
+ 13000000,
+ 15360000,
+ 16800000,
+ -1,
+ 26000000,
+ -1, /* 12000000, invalid for voice codec */
+ -1, /* 3686400, invalid for voice codec */
+ 33600000,
+};
+
+static int mc13783_set_sysclk(struct snd_soc_dai *dai,
+ int clk_id, unsigned int freq, int dir,
+ unsigned int reg)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ int clk;
+ unsigned int val = 0;
+ unsigned int mask = AUDIO_CLK(0x7) | AUDIO_CLK_SEL;
+
+ for (clk = 0; clk < ARRAY_SIZE(mc13783_sysclk); clk++) {
+ if (mc13783_sysclk[clk] < 0)
+ continue;
+ if (mc13783_sysclk[clk] == freq)
+ break;
+ }
+
+ if (clk == ARRAY_SIZE(mc13783_sysclk))
+ return -EINVAL;
+
+ if (clk_id == MC13783_CLK_CLIB)
+ val |= AUDIO_CLK_SEL;
+
+ val |= AUDIO_CLK(clk);
+
+ snd_soc_update_bits(codec, reg, mask, val);
+
+ return 0;
+}
+
+static int mc13783_set_sysclk_dac(struct snd_soc_dai *dai,
+ int clk_id, unsigned int freq, int dir)
+{
+ return mc13783_set_sysclk(dai, clk_id, freq, dir, MC13783_AUDIO_DAC);
+}
+
+static int mc13783_set_sysclk_codec(struct snd_soc_dai *dai,
+ int clk_id, unsigned int freq, int dir)
+{
+ return mc13783_set_sysclk(dai, clk_id, freq, dir, MC13783_AUDIO_CODEC);
+}
+
+static int mc13783_set_sysclk_sync(struct snd_soc_dai *dai,
+ int clk_id, unsigned int freq, int dir)
+{
+ int ret;
+
+ ret = mc13783_set_sysclk(dai, clk_id, freq, dir, MC13783_AUDIO_DAC);
+ if (ret)
+ return ret;
+
+ return mc13783_set_sysclk(dai, clk_id, freq, dir, MC13783_AUDIO_CODEC);
+}
+
+static int mc13783_set_tdm_slot_dac(struct snd_soc_dai *dai,
+ unsigned int tx_mask, unsigned int rx_mask, int slots,
+ int slot_width)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ unsigned int val = 0;
+ unsigned int mask = SSI_NETWORK_DAC_SLOT_MASK |
+ SSI_NETWORK_DAC_RXSLOT_MASK;
+
+ switch (slots) {
+ case 2:
+ val |= SSI_NETWORK_DAC_SLOTS_2;
+ break;
+ case 4:
+ val |= SSI_NETWORK_DAC_SLOTS_4;
+ break;
+ case 8:
+ val |= SSI_NETWORK_DAC_SLOTS_8;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ switch (rx_mask) {
+ case 0xfffffffc:
+ val |= SSI_NETWORK_DAC_RXSLOT_0_1;
+ break;
+ case 0xfffffff3:
+ val |= SSI_NETWORK_DAC_RXSLOT_2_3;
+ break;
+ case 0xffffffcf:
+ val |= SSI_NETWORK_DAC_RXSLOT_4_5;
+ break;
+ case 0xffffff3f:
+ val |= SSI_NETWORK_DAC_RXSLOT_6_7;
+ break;
+ default:
+ return -EINVAL;
+ };
+
+ snd_soc_update_bits(codec, MC13783_SSI_NETWORK, mask, val);
+
+ return 0;
+}
+
+static int mc13783_set_tdm_slot_codec(struct snd_soc_dai *dai,
+ unsigned int tx_mask, unsigned int rx_mask, int slots,
+ int slot_width)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ unsigned int val = 0;
+ unsigned int mask = 0x3f;
+
+ if (slots != 4)
+ return -EINVAL;
+
+ if (tx_mask != 0xfffffffc)
+ return -EINVAL;
+
+ val |= (0x00 << 2); /* primary timeslot RX/TX(?) is 0 */
+ val |= (0x01 << 4); /* secondary timeslot TX is 1 */
+
+ snd_soc_update_bits(codec, MC13783_SSI_NETWORK, mask, val);
+
+ return 0;
+}
+
+static int mc13783_set_tdm_slot_sync(struct snd_soc_dai *dai,
+ unsigned int tx_mask, unsigned int rx_mask, int slots,
+ int slot_width)
+{
+ int ret;
+
+ ret = mc13783_set_tdm_slot_dac(dai, tx_mask, rx_mask, slots,
+ slot_width);
+ if (ret)
+ return ret;
+
+ ret = mc13783_set_tdm_slot_codec(dai, tx_mask, rx_mask, slots,
+ slot_width);
+
+ return ret;
+}
+
+static const struct snd_kcontrol_new mc1l_amp_ctl =
+ SOC_DAPM_SINGLE("Switch", 38, 7, 1, 0);
+
+static const struct snd_kcontrol_new mc1r_amp_ctl =
+ SOC_DAPM_SINGLE("Switch", 38, 5, 1, 0);
+
+static const struct snd_kcontrol_new mc2_amp_ctl =
+ SOC_DAPM_SINGLE("Switch", 38, 9, 1, 0);
+
+static const struct snd_kcontrol_new atx_amp_ctl =
+ SOC_DAPM_SINGLE("Switch", 38, 11, 1, 0);
+
+
+/* Virtual mux. The chip does the input selection automatically
+ * as soon as we enable one input. */
+static const char * const adcl_enum_text[] = {
+ "MC1L", "RXINL",
+};
+
+static const struct soc_enum adcl_enum =
+ SOC_ENUM_SINGLE(0, 0, ARRAY_SIZE(adcl_enum_text), adcl_enum_text);
+
+static const struct snd_kcontrol_new left_input_mux =
+ SOC_DAPM_ENUM_VIRT("Route", adcl_enum);
+
+static const char * const adcr_enum_text[] = {
+ "MC1R", "MC2", "RXINR", "TXIN",
+};
+
+static const struct soc_enum adcr_enum =
+ SOC_ENUM_SINGLE(0, 0, ARRAY_SIZE(adcr_enum_text), adcr_enum_text);
+
+static const struct snd_kcontrol_new right_input_mux =
+ SOC_DAPM_ENUM_VIRT("Route", adcr_enum);
+
+static const struct snd_kcontrol_new samp_ctl =
+ SOC_DAPM_SINGLE("Switch", 36, 3, 1, 0);
+
+static const struct snd_kcontrol_new lamp_ctl =
+ SOC_DAPM_SINGLE("Switch", 36, 5, 1, 0);
+
+static const struct snd_kcontrol_new hlamp_ctl =
+ SOC_DAPM_SINGLE("Switch", 36, 10, 1, 0);
+
+static const struct snd_kcontrol_new hramp_ctl =
+ SOC_DAPM_SINGLE("Switch", 36, 9, 1, 0);
+
+static const struct snd_kcontrol_new llamp_ctl =
+ SOC_DAPM_SINGLE("Switch", 36, 16, 1, 0);
+
+static const struct snd_kcontrol_new lramp_ctl =
+ SOC_DAPM_SINGLE("Switch", 36, 15, 1, 0);
+
+static const struct snd_soc_dapm_widget mc13783_dapm_widgets[] = {
+/* Input */
+ SND_SOC_DAPM_INPUT("MC1LIN"),
+ SND_SOC_DAPM_INPUT("MC1RIN"),
+ SND_SOC_DAPM_INPUT("MC2IN"),
+ SND_SOC_DAPM_INPUT("RXINR"),
+ SND_SOC_DAPM_INPUT("RXINL"),
+ SND_SOC_DAPM_INPUT("TXIN"),
+
+ SND_SOC_DAPM_SUPPLY("MC1 Bias", 38, 0, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("MC2 Bias", 38, 1, 0, NULL, 0),
+
+ SND_SOC_DAPM_SWITCH("MC1L Amp", 38, 7, 0, &mc1l_amp_ctl),
+ SND_SOC_DAPM_SWITCH("MC1R Amp", 38, 5, 0, &mc1r_amp_ctl),
+ SND_SOC_DAPM_SWITCH("MC2 Amp", 38, 9, 0, &mc2_amp_ctl),
+ SND_SOC_DAPM_SWITCH("TXIN Amp", 38, 11, 0, &atx_amp_ctl),
+
+ SND_SOC_DAPM_VIRT_MUX("PGA Left Input Mux", SND_SOC_NOPM, 0, 0,
+ &left_input_mux),
+ SND_SOC_DAPM_VIRT_MUX("PGA Right Input Mux", SND_SOC_NOPM, 0, 0,
+ &right_input_mux),
+
+ SND_SOC_DAPM_PGA("PGA Left Input", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("PGA Right Input", SND_SOC_NOPM, 0, 0, NULL, 0),
+
+ SND_SOC_DAPM_ADC("ADC", "Capture", 40, 11, 0),
+ SND_SOC_DAPM_SUPPLY("ADC_Reset", 40, 15, 0, NULL, 0),
+
+/* Output */
+ SND_SOC_DAPM_SUPPLY("DAC_E", 41, 11, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("DAC_Reset", 41, 15, 0, NULL, 0),
+ SND_SOC_DAPM_OUTPUT("RXOUTL"),
+ SND_SOC_DAPM_OUTPUT("RXOUTR"),
+ SND_SOC_DAPM_OUTPUT("HSL"),
+ SND_SOC_DAPM_OUTPUT("HSR"),
+ SND_SOC_DAPM_OUTPUT("LSP"),
+ SND_SOC_DAPM_OUTPUT("SP"),
+
+ SND_SOC_DAPM_SWITCH("Speaker Amp", 36, 3, 0, &samp_ctl),
+ SND_SOC_DAPM_SWITCH("Loudspeaker Amp", SND_SOC_NOPM, 0, 0, &lamp_ctl),
+ SND_SOC_DAPM_SWITCH("Headset Amp Left", 36, 10, 0, &hlamp_ctl),
+ SND_SOC_DAPM_SWITCH("Headset Amp Right", 36, 9, 0, &hramp_ctl),
+ SND_SOC_DAPM_SWITCH("Line out Amp Left", 36, 16, 0, &llamp_ctl),
+ SND_SOC_DAPM_SWITCH("Line out Amp Right", 36, 15, 0, &lramp_ctl),
+ SND_SOC_DAPM_DAC("DAC", "Playback", 36, 22, 0),
+ SND_SOC_DAPM_PGA("DAC PGA", 37, 5, 0, NULL, 0),
+};
+
+static struct snd_soc_dapm_route mc13783_routes[] = {
+/* Input */
+ { "MC1L Amp", NULL, "MC1LIN"},
+ { "MC1R Amp", NULL, "MC1RIN" },
+ { "MC2 Amp", NULL, "MC2IN" },
+ { "TXIN Amp", NULL, "TXIN"},
+
+ { "PGA Left Input Mux", "MC1L", "MC1L Amp" },
+ { "PGA Left Input Mux", "RXINL", "RXINL"},
+ { "PGA Right Input Mux", "MC1R", "MC1R Amp" },
+ { "PGA Right Input Mux", "MC2", "MC2 Amp"},
+ { "PGA Right Input Mux", "TXIN", "TXIN Amp"},
+ { "PGA Right Input Mux", "RXINR", "RXINR"},
+
+ { "PGA Left Input", NULL, "PGA Left Input Mux"},
+ { "PGA Right Input", NULL, "PGA Right Input Mux"},
+
+ { "ADC", NULL, "PGA Left Input"},
+ { "ADC", NULL, "PGA Right Input"},
+ { "ADC", NULL, "ADC_Reset"},
+
+/* Output */
+ { "HSL", NULL, "Headset Amp Left" },
+ { "HSR", NULL, "Headset Amp Right"},
+ { "RXOUTL", NULL, "Line out Amp Left"},
+ { "RXOUTR", NULL, "Line out Amp Right"},
+ { "SP", NULL, "Speaker Amp"},
+ { "Speaker Amp", NULL, "DAC PGA"},
+ { "LSP", NULL, "DAC PGA"},
+ { "Headset Amp Left", NULL, "DAC PGA"},
+ { "Headset Amp Right", NULL, "DAC PGA"},
+ { "Line out Amp Left", NULL, "DAC PGA"},
+ { "Line out Amp Right", NULL, "DAC PGA"},
+ { "DAC PGA", NULL, "DAC"},
+ { "DAC", NULL, "DAC_E"},
+};
+
+static const char * const mc13783_3d_mixer[] = {"Stereo", "Phase Mix",
+ "Mono", "Mono Mix"};
+
+static const struct soc_enum mc13783_enum_3d_mixer =
+ SOC_ENUM_SINGLE(MC13783_AUDIO_RX1, 16, ARRAY_SIZE(mc13783_3d_mixer),
+ mc13783_3d_mixer);
+
+static struct snd_kcontrol_new mc13783_control_list[] = {
+ SOC_SINGLE("Loudspeaker enable", MC13783_AUDIO_RX0, 5, 1, 0),
+ SOC_SINGLE("PCM Playback Volume", MC13783_AUDIO_RX1, 6, 15, 0),
+ SOC_DOUBLE("PCM Capture Volume", MC13783_AUDIO_TX, 19, 14, 31, 0),
+ SOC_ENUM("3D Control", mc13783_enum_3d_mixer),
+};
+
+static int mc13783_probe(struct snd_soc_codec *codec)
+{
+ struct mc13783_priv *priv = snd_soc_codec_get_drvdata(codec);
+
+ mc13xxx_lock(priv->mc13xxx);
+
+ /* these are the reset values */
+ mc13xxx_reg_write(priv->mc13xxx, MC13783_AUDIO_RX0, 0x25893);
+ mc13xxx_reg_write(priv->mc13xxx, MC13783_AUDIO_RX1, 0x00d35A);
+ mc13xxx_reg_write(priv->mc13xxx, MC13783_AUDIO_TX, 0x420000);
+ mc13xxx_reg_write(priv->mc13xxx, MC13783_SSI_NETWORK, 0x013060);
+ mc13xxx_reg_write(priv->mc13xxx, MC13783_AUDIO_CODEC, 0x180027);
+ mc13xxx_reg_write(priv->mc13xxx, MC13783_AUDIO_DAC, 0x0e0004);
+
+ if (priv->adc_ssi_port == MC13783_SSI1_PORT)
+ mc13xxx_reg_rmw(priv->mc13xxx, MC13783_AUDIO_CODEC,
+ AUDIO_SSI_SEL, 0);
+ else
+ mc13xxx_reg_rmw(priv->mc13xxx, MC13783_AUDIO_CODEC,
+ 0, AUDIO_SSI_SEL);
+
+ if (priv->dac_ssi_port == MC13783_SSI1_PORT)
+ mc13xxx_reg_rmw(priv->mc13xxx, MC13783_AUDIO_DAC,
+ AUDIO_SSI_SEL, 0);
+ else
+ mc13xxx_reg_rmw(priv->mc13xxx, MC13783_AUDIO_DAC,
+ 0, AUDIO_SSI_SEL);
+
+ mc13xxx_unlock(priv->mc13xxx);
+
+ return 0;
+}
+
+static int mc13783_remove(struct snd_soc_codec *codec)
+{
+ struct mc13783_priv *priv = snd_soc_codec_get_drvdata(codec);
+
+ mc13xxx_lock(priv->mc13xxx);
+
+ /* Make sure VAUDIOON is off */
+ mc13xxx_reg_rmw(priv->mc13xxx, MC13783_AUDIO_RX0, 0x3, 0);
+
+ mc13xxx_unlock(priv->mc13xxx);
+
+ return 0;
+}
+
+#define MC13783_RATES_RECORD (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000)
+
+#define MC13783_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
+ SNDRV_PCM_FMTBIT_S24_LE)
+
+static struct snd_soc_dai_ops mc13783_ops_dac = {
+ .hw_params = mc13783_pcm_hw_params_dac,
+ .set_fmt = mc13783_set_fmt_async,
+ .set_sysclk = mc13783_set_sysclk_dac,
+ .set_tdm_slot = mc13783_set_tdm_slot_dac,
+};
+
+static struct snd_soc_dai_ops mc13783_ops_codec = {
+ .hw_params = mc13783_pcm_hw_params_codec,
+ .set_fmt = mc13783_set_fmt_async,
+ .set_sysclk = mc13783_set_sysclk_codec,
+ .set_tdm_slot = mc13783_set_tdm_slot_codec,
+};
+
+/*
+ * The mc13783 has two SSI ports, both of them can be routed either
+ * to the voice codec or the stereo DAC. When two different SSI ports
+ * are used for the voice codec and the stereo DAC we can do different
+ * formats and sysclock settings for playback and capture
+ * (mc13783-hifi-playback and mc13783-hifi-capture). Using the same port
+ * forces us to use symmetric rates (mc13783-hifi).
+ */
+static struct snd_soc_dai_driver mc13783_dai_async[] = {
+ {
+ .name = "mc13783-hifi-playback",
+ .id = MC13783_ID_STEREO_DAC,
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_8000_96000,
+ .formats = MC13783_FORMATS,
+ },
+ .ops = &mc13783_ops_dac,
+ }, {
+ .name = "mc13783-hifi-capture",
+ .id = MC13783_ID_STEREO_CODEC,
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = MC13783_RATES_RECORD,
+ .formats = MC13783_FORMATS,
+ },
+ .ops = &mc13783_ops_codec,
+ },
+};
+
+static struct snd_soc_dai_ops mc13783_ops_sync = {
+ .hw_params = mc13783_pcm_hw_params_sync,
+ .set_fmt = mc13783_set_fmt_sync,
+ .set_sysclk = mc13783_set_sysclk_sync,
+ .set_tdm_slot = mc13783_set_tdm_slot_sync,
+};
+
+static struct snd_soc_dai_driver mc13783_dai_sync[] = {
+ {
+ .name = "mc13783-hifi",
+ .id = MC13783_ID_SYNC,
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_8000_96000,
+ .formats = MC13783_FORMATS,
+ },
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = MC13783_RATES_RECORD,
+ .formats = MC13783_FORMATS,
+ },
+ .ops = &mc13783_ops_sync,
+ .symmetric_rates = 1,
+ }
+};
+
+static struct snd_soc_codec_driver soc_codec_dev_mc13783 = {
+ .probe = mc13783_probe,
+ .remove = mc13783_remove,
+ .read = mc13783_read,
+ .write = mc13783_write,
+ .controls = mc13783_control_list,
+ .num_controls = ARRAY_SIZE(mc13783_control_list),
+ .dapm_widgets = mc13783_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(mc13783_dapm_widgets),
+ .dapm_routes = mc13783_routes,
+ .num_dapm_routes = ARRAY_SIZE(mc13783_routes),
+};
+
+static int mc13783_codec_probe(struct platform_device *pdev)
+{
+ struct mc13xxx *mc13xxx;
+ struct mc13783_priv *priv;
+ struct mc13xxx_codec_platform_data *pdata = pdev->dev.platform_data;
+ int ret;
+
+ mc13xxx = dev_get_drvdata(pdev->dev.parent);
+
+
+ priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL);
+ if (priv == NULL)
+ return -ENOMEM;
+
+ dev_set_drvdata(&pdev->dev, priv);
+ priv->mc13xxx = mc13xxx;
+ if (pdata) {
+ priv->adc_ssi_port = pdata->adc_ssi_port;
+ priv->dac_ssi_port = pdata->dac_ssi_port;
+ } else {
+ priv->adc_ssi_port = MC13783_SSI1_PORT;
+ priv->dac_ssi_port = MC13783_SSI2_PORT;
+ }
+
+ if (priv->adc_ssi_port == priv->dac_ssi_port)
+ ret = snd_soc_register_codec(&pdev->dev, &soc_codec_dev_mc13783,
+ mc13783_dai_sync, ARRAY_SIZE(mc13783_dai_sync));
+ else
+ ret = snd_soc_register_codec(&pdev->dev, &soc_codec_dev_mc13783,
+ mc13783_dai_async, ARRAY_SIZE(mc13783_dai_async));
+
+ if (ret)
+ goto err_register_codec;
+
+ return 0;
+
+err_register_codec:
+ dev_err(&pdev->dev, "register codec failed with %d\n", ret);
+
+ return ret;
+}
+
+static int mc13783_codec_remove(struct platform_device *pdev)
+{
+ snd_soc_unregister_codec(&pdev->dev);
+
+ return 0;
+}
+
+static struct platform_driver mc13783_codec_driver = {
+ .driver = {
+ .name = "mc13783-codec",
+ .owner = THIS_MODULE,
+ },
+ .probe = mc13783_codec_probe,
+ .remove = __devexit_p(mc13783_codec_remove),
+};
+
+module_platform_driver(mc13783_codec_driver);
+
+MODULE_DESCRIPTION("ASoC MC13783 driver");
+MODULE_AUTHOR("Sascha Hauer, Pengutronix <s.hauer@pengutronix.de>");
+MODULE_AUTHOR("Philippe Retornaz <philippe.retornaz@epfl.ch>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/mc13783.h b/sound/soc/codecs/mc13783.h
new file mode 100644
index 00000000000..3a6d1993a21
--- /dev/null
+++ b/sound/soc/codecs/mc13783.h
@@ -0,0 +1,28 @@
+/*
+ * Copyright 2008 Juergen Beisert, kernel@pengutronix.de
+ *
+ * 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.
+ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301, USA.
+ */
+
+#ifndef MC13783_MIXER_H
+#define MC13783_MIXER_H
+
+#define MC13783_CLK_CLIA 1
+#define MC13783_CLK_CLIB 2
+
+#define MC13783_ID_STEREO_DAC 1
+#define MC13783_ID_STEREO_CODEC 2
+#define MC13783_ID_SYNC 3
+
+#endif /* MC13783_MIXER_H */
diff --git a/sound/soc/codecs/ml26124.c b/sound/soc/codecs/ml26124.c
new file mode 100644
index 00000000000..22cb5bf5927
--- /dev/null
+++ b/sound/soc/codecs/ml26124.c
@@ -0,0 +1,681 @@
+/*
+ * Copyright (C) 2011 LAPIS Semiconductor Co., 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; version 2 of the License.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * 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 <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/i2c.h>
+#include <linux/slab.h>
+#include <linux/platform_device.h>
+#include <linux/regmap.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/tlv.h>
+#include "ml26124.h"
+
+#define DVOL_CTL_DVMUTE_ON BIT(4) /* Digital volume MUTE On */
+#define DVOL_CTL_DVMUTE_OFF 0 /* Digital volume MUTE Off */
+#define ML26124_SAI_NO_DELAY BIT(1)
+#define ML26124_SAI_FRAME_SYNC (BIT(5) | BIT(0)) /* For mono (Telecodec) */
+#define ML26134_CACHESIZE 212
+#define ML26124_VMID BIT(1)
+#define ML26124_RATES (SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_32000 |\
+ SNDRV_PCM_RATE_48000)
+#define ML26124_FORMATS (SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE |\
+ SNDRV_PCM_FMTBIT_S32_LE)
+#define ML26124_NUM_REGISTER ML26134_CACHESIZE
+
+struct ml26124_priv {
+ u32 mclk;
+ u32 rate;
+ struct regmap *regmap;
+ int clk_in;
+ struct snd_pcm_substream *substream;
+};
+
+struct clk_coeff {
+ u32 mclk;
+ u32 rate;
+ u8 pllnl;
+ u8 pllnh;
+ u8 pllml;
+ u8 pllmh;
+ u8 plldiv;
+};
+
+/* ML26124 configuration */
+static const DECLARE_TLV_DB_SCALE(digital_tlv, -7150, 50, 0);
+
+static const DECLARE_TLV_DB_SCALE(alclvl, -2250, 150, 0);
+static const DECLARE_TLV_DB_SCALE(mingain, -1200, 600, 0);
+static const DECLARE_TLV_DB_SCALE(maxgain, -675, 600, 0);
+static const DECLARE_TLV_DB_SCALE(boost_vol, -1200, 75, 0);
+static const DECLARE_TLV_DB_SCALE(ngth, -7650, 150, 0);
+
+static const char * const ml26124_companding[] = {"16bit PCM", "u-law",
+ "A-law"};
+
+static const struct soc_enum ml26124_adc_companding_enum
+ = SOC_ENUM_SINGLE(ML26124_SAI_TRANS_CTL, 6, 3, ml26124_companding);
+
+static const struct soc_enum ml26124_dac_companding_enum
+ = SOC_ENUM_SINGLE(ML26124_SAI_RCV_CTL, 6, 3, ml26124_companding);
+
+static const struct snd_kcontrol_new ml26124_snd_controls[] = {
+ SOC_SINGLE_TLV("Capture Digital Volume", ML26124_RECORD_DIG_VOL, 0,
+ 0xff, 1, digital_tlv),
+ SOC_SINGLE_TLV("Playback Digital Volume", ML26124_PLBAK_DIG_VOL, 0,
+ 0xff, 1, digital_tlv),
+ SOC_SINGLE_TLV("Digital Boost Volume", ML26124_DIGI_BOOST_VOL, 0,
+ 0x3f, 0, boost_vol),
+ SOC_SINGLE_TLV("EQ Band0 Volume", ML26124_EQ_GAIN_BRAND0, 0,
+ 0xff, 1, digital_tlv),
+ SOC_SINGLE_TLV("EQ Band1 Volume", ML26124_EQ_GAIN_BRAND1, 0,
+ 0xff, 1, digital_tlv),
+ SOC_SINGLE_TLV("EQ Band2 Volume", ML26124_EQ_GAIN_BRAND2, 0,
+ 0xff, 1, digital_tlv),
+ SOC_SINGLE_TLV("EQ Band3 Volume", ML26124_EQ_GAIN_BRAND3, 0,
+ 0xff, 1, digital_tlv),
+ SOC_SINGLE_TLV("EQ Band4 Volume", ML26124_EQ_GAIN_BRAND4, 0,
+ 0xff, 1, digital_tlv),
+ SOC_SINGLE_TLV("ALC Target Level", ML26124_ALC_TARGET_LEV, 0,
+ 0xf, 1, alclvl),
+ SOC_SINGLE_TLV("ALC Min Input Volume", ML26124_ALC_MAXMIN_GAIN, 0,
+ 7, 0, mingain),
+ SOC_SINGLE_TLV("ALC Max Input Volume", ML26124_ALC_MAXMIN_GAIN, 4,
+ 7, 1, maxgain),
+ SOC_SINGLE_TLV("Playback Limiter Min Input Volume",
+ ML26124_PL_MAXMIN_GAIN, 0, 7, 0, mingain),
+ SOC_SINGLE_TLV("Playback Limiter Max Input Volume",
+ ML26124_PL_MAXMIN_GAIN, 4, 7, 1, maxgain),
+ SOC_SINGLE_TLV("Playback Boost Volume", ML26124_PLYBAK_BOST_VOL, 0,
+ 0x3f, 0, boost_vol),
+ SOC_SINGLE("DC High Pass Filter Switch", ML26124_FILTER_EN, 0, 1, 0),
+ SOC_SINGLE("Noise High Pass Filter Switch", ML26124_FILTER_EN, 1, 1, 0),
+ SOC_SINGLE("ZC Switch", ML26124_PW_ZCCMP_PW_MNG, 1,
+ 1, 0),
+ SOC_SINGLE("EQ Band0 Switch", ML26124_FILTER_EN, 2, 1, 0),
+ SOC_SINGLE("EQ Band1 Switch", ML26124_FILTER_EN, 3, 1, 0),
+ SOC_SINGLE("EQ Band2 Switch", ML26124_FILTER_EN, 4, 1, 0),
+ SOC_SINGLE("EQ Band3 Switch", ML26124_FILTER_EN, 5, 1, 0),
+ SOC_SINGLE("EQ Band4 Switch", ML26124_FILTER_EN, 6, 1, 0),
+ SOC_SINGLE("Play Limiter", ML26124_DVOL_CTL, 0, 1, 0),
+ SOC_SINGLE("Capture Limiter", ML26124_DVOL_CTL, 1, 1, 0),
+ SOC_SINGLE("Digital Volume Fade Switch", ML26124_DVOL_CTL, 3, 1, 0),
+ SOC_SINGLE("Digital Switch", ML26124_DVOL_CTL, 4, 1, 0),
+ SOC_ENUM("DAC Companding", ml26124_dac_companding_enum),
+ SOC_ENUM("ADC Companding", ml26124_adc_companding_enum),
+};
+
+static const struct snd_kcontrol_new ml26124_output_mixer_controls[] = {
+ SOC_DAPM_SINGLE("DAC Switch", ML26124_SPK_AMP_OUT, 1, 1, 0),
+ SOC_DAPM_SINGLE("Line in loopback Switch", ML26124_SPK_AMP_OUT, 3, 1,
+ 0),
+ SOC_DAPM_SINGLE("PGA Switch", ML26124_SPK_AMP_OUT, 5, 1, 0),
+};
+
+/* Input mux */
+static const char * const ml26124_input_select[] = {"Analog MIC SingleEnded in",
+ "Digital MIC in", "Analog MIC Differential in"};
+
+static const struct soc_enum ml26124_insel_enum =
+ SOC_ENUM_SINGLE(ML26124_MIC_IF_CTL, 0, 3, ml26124_input_select);
+
+static const struct snd_kcontrol_new ml26124_input_mux_controls =
+ SOC_DAPM_ENUM("Input Select", ml26124_insel_enum);
+
+static const struct snd_kcontrol_new ml26124_line_control =
+ SOC_DAPM_SINGLE("Switch", ML26124_PW_LOUT_PW_MNG, 1, 1, 0);
+
+static const struct snd_soc_dapm_widget ml26124_dapm_widgets[] = {
+ SND_SOC_DAPM_SUPPLY("MCLKEN", ML26124_CLK_EN, 0, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("PLLEN", ML26124_CLK_EN, 1, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("PLLOE", ML26124_CLK_EN, 2, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("MICBIAS", ML26124_PW_REF_PW_MNG, 2, 0, NULL, 0),
+ SND_SOC_DAPM_MIXER("Output Mixer", SND_SOC_NOPM, 0, 0,
+ &ml26124_output_mixer_controls[0],
+ ARRAY_SIZE(ml26124_output_mixer_controls)),
+ SND_SOC_DAPM_DAC("DAC", "Playback", ML26124_PW_DAC_PW_MNG, 1, 0),
+ SND_SOC_DAPM_ADC("ADC", "Capture", ML26124_PW_IN_PW_MNG, 1, 0),
+ SND_SOC_DAPM_PGA("PGA", ML26124_PW_IN_PW_MNG, 3, 0, NULL, 0),
+ SND_SOC_DAPM_MUX("Input Mux", SND_SOC_NOPM, 0, 0,
+ &ml26124_input_mux_controls),
+ SND_SOC_DAPM_SWITCH("Line Out Enable", SND_SOC_NOPM, 0, 0,
+ &ml26124_line_control),
+ SND_SOC_DAPM_INPUT("MDIN"),
+ SND_SOC_DAPM_INPUT("MIN"),
+ SND_SOC_DAPM_INPUT("LIN"),
+ SND_SOC_DAPM_OUTPUT("SPOUT"),
+ SND_SOC_DAPM_OUTPUT("LOUT"),
+};
+
+static const struct snd_soc_dapm_route ml26124_intercon[] = {
+ /* Supply */
+ {"DAC", NULL, "MCLKEN"},
+ {"ADC", NULL, "MCLKEN"},
+ {"DAC", NULL, "PLLEN"},
+ {"ADC", NULL, "PLLEN"},
+ {"DAC", NULL, "PLLOE"},
+ {"ADC", NULL, "PLLOE"},
+
+ /* output mixer */
+ {"Output Mixer", "DAC Switch", "DAC"},
+ {"Output Mixer", "Line in loopback Switch", "LIN"},
+
+ /* outputs */
+ {"LOUT", NULL, "Output Mixer"},
+ {"SPOUT", NULL, "Output Mixer"},
+ {"Line Out Enable", NULL, "LOUT"},
+
+ /* input */
+ {"ADC", NULL, "Input Mux"},
+ {"Input Mux", "Analog MIC SingleEnded in", "PGA"},
+ {"Input Mux", "Analog MIC Differential in", "PGA"},
+ {"PGA", NULL, "MIN"},
+};
+
+/* PLLOutputFreq(Hz) = InputMclkFreq(Hz) * PLLM / (PLLN * PLLDIV) */
+static const struct clk_coeff coeff_div[] = {
+ {12288000, 16000, 0xc, 0x0, 0x20, 0x0, 0x4},
+ {12288000, 32000, 0xc, 0x0, 0x20, 0x0, 0x4},
+ {12288000, 48000, 0xc, 0x0, 0x30, 0x0, 0x4},
+};
+
+static struct reg_default ml26124_reg[] = {
+ /* CLOCK control Register */
+ {0x00, 0x00 }, /* Sampling Rate */
+ {0x02, 0x00}, /* PLL NL */
+ {0x04, 0x00}, /* PLLNH */
+ {0x06, 0x00}, /* PLLML */
+ {0x08, 0x00}, /* MLLMH */
+ {0x0a, 0x00}, /* PLLDIV */
+ {0x0c, 0x00}, /* Clock Enable */
+ {0x0e, 0x00}, /* CLK Input/Output Control */
+
+ /* System Control Register */
+ {0x10, 0x00}, /* Software RESET */
+ {0x12, 0x00}, /* Record/Playback Run */
+ {0x14, 0x00}, /* Mic Input/Output control */
+
+ /* Power Management Register */
+ {0x20, 0x00}, /* Reference Power Management */
+ {0x22, 0x00}, /* Input Power Management */
+ {0x24, 0x00}, /* DAC Power Management */
+ {0x26, 0x00}, /* SP-AMP Power Management */
+ {0x28, 0x00}, /* LINEOUT Power Management */
+ {0x2a, 0x00}, /* VIDEO Power Management */
+ {0x2e, 0x00}, /* AC-CMP Power Management */
+
+ /* Analog reference Control Register */
+ {0x30, 0x04}, /* MICBIAS Voltage Control */
+
+ /* Input/Output Amplifier Control Register */
+ {0x32, 0x10}, /* MIC Input Volume */
+ {0x38, 0x00}, /* Mic Boost Volume */
+ {0x3a, 0x33}, /* Speaker AMP Volume */
+ {0x48, 0x00}, /* AMP Volume Control Function Enable */
+ {0x4a, 0x00}, /* Amplifier Volume Fader Control */
+
+ /* Analog Path Control Register */
+ {0x54, 0x00}, /* Speaker AMP Output Control */
+ {0x5a, 0x00}, /* Mic IF Control */
+ {0xe8, 0x01}, /* Mic Select Control */
+
+ /* Audio Interface Control Register */
+ {0x60, 0x00}, /* SAI-Trans Control */
+ {0x62, 0x00}, /* SAI-Receive Control */
+ {0x64, 0x00}, /* SAI Mode select */
+
+ /* DSP Control Register */
+ {0x66, 0x01}, /* Filter Func Enable */
+ {0x68, 0x00}, /* Volume Control Func Enable */
+ {0x6A, 0x00}, /* Mixer & Volume Control*/
+ {0x6C, 0xff}, /* Record Digital Volume */
+ {0x70, 0xff}, /* Playback Digital Volume */
+ {0x72, 0x10}, /* Digital Boost Volume */
+ {0x74, 0xe7}, /* EQ gain Band0 */
+ {0x76, 0xe7}, /* EQ gain Band1 */
+ {0x78, 0xe7}, /* EQ gain Band2 */
+ {0x7A, 0xe7}, /* EQ gain Band3 */
+ {0x7C, 0xe7}, /* EQ gain Band4 */
+ {0x7E, 0x00}, /* HPF2 CutOff*/
+ {0x80, 0x00}, /* EQ Band0 Coef0L */
+ {0x82, 0x00}, /* EQ Band0 Coef0H */
+ {0x84, 0x00}, /* EQ Band0 Coef0L */
+ {0x86, 0x00}, /* EQ Band0 Coef0H */
+ {0x88, 0x00}, /* EQ Band1 Coef0L */
+ {0x8A, 0x00}, /* EQ Band1 Coef0H */
+ {0x8C, 0x00}, /* EQ Band1 Coef0L */
+ {0x8E, 0x00}, /* EQ Band1 Coef0H */
+ {0x90, 0x00}, /* EQ Band2 Coef0L */
+ {0x92, 0x00}, /* EQ Band2 Coef0H */
+ {0x94, 0x00}, /* EQ Band2 Coef0L */
+ {0x96, 0x00}, /* EQ Band2 Coef0H */
+ {0x98, 0x00}, /* EQ Band3 Coef0L */
+ {0x9A, 0x00}, /* EQ Band3 Coef0H */
+ {0x9C, 0x00}, /* EQ Band3 Coef0L */
+ {0x9E, 0x00}, /* EQ Band3 Coef0H */
+ {0xA0, 0x00}, /* EQ Band4 Coef0L */
+ {0xA2, 0x00}, /* EQ Band4 Coef0H */
+ {0xA4, 0x00}, /* EQ Band4 Coef0L */
+ {0xA6, 0x00}, /* EQ Band4 Coef0H */
+
+ /* ALC Control Register */
+ {0xb0, 0x00}, /* ALC Mode */
+ {0xb2, 0x02}, /* ALC Attack Time */
+ {0xb4, 0x03}, /* ALC Decay Time */
+ {0xb6, 0x00}, /* ALC Hold Time */
+ {0xb8, 0x0b}, /* ALC Target Level */
+ {0xba, 0x70}, /* ALC Max/Min Gain */
+ {0xbc, 0x00}, /* Noise Gate Threshold */
+ {0xbe, 0x00}, /* ALC ZeroCross TimeOut */
+
+ /* Playback Limiter Control Register */
+ {0xc0, 0x04}, /* PL Attack Time */
+ {0xc2, 0x05}, /* PL Decay Time */
+ {0xc4, 0x0d}, /* PL Target Level */
+ {0xc6, 0x70}, /* PL Max/Min Gain */
+ {0xc8, 0x10}, /* Playback Boost Volume */
+ {0xca, 0x00}, /* PL ZeroCross TimeOut */
+
+ /* Video Amplifier Control Register */
+ {0xd0, 0x01}, /* VIDEO AMP Gain Control */
+ {0xd2, 0x01}, /* VIDEO AMP Setup 1 */
+ {0xd4, 0x01}, /* VIDEO AMP Control2 */
+};
+
+/* Get sampling rate value of sampling rate setting register (0x0) */
+static inline int get_srate(int rate)
+{
+ int srate;
+
+ switch (rate) {
+ case 16000:
+ srate = 3;
+ break;
+ case 32000:
+ srate = 6;
+ break;
+ case 48000:
+ srate = 8;
+ break;
+ default:
+ return -EINVAL;
+ }
+ return srate;
+}
+
+static inline int get_coeff(int mclk, int rate)
+{
+ int i;
+
+ for (i = 0; i < ARRAY_SIZE(coeff_div); i++) {
+ if (coeff_div[i].rate == rate && coeff_div[i].mclk == mclk)
+ return i;
+ }
+ return -EINVAL;
+}
+
+static int ml26124_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ struct ml26124_priv *priv = snd_soc_codec_get_drvdata(codec);
+ int i = get_coeff(priv->mclk, params_rate(hw_params));
+
+ priv->substream = substream;
+ priv->rate = params_rate(hw_params);
+
+ if (priv->clk_in) {
+ switch (priv->mclk / params_rate(hw_params)) {
+ case 256:
+ snd_soc_update_bits(codec, ML26124_CLK_CTL,
+ BIT(0) | BIT(1), 1);
+ break;
+ case 512:
+ snd_soc_update_bits(codec, ML26124_CLK_CTL,
+ BIT(0) | BIT(1), 2);
+ break;
+ case 1024:
+ snd_soc_update_bits(codec, ML26124_CLK_CTL,
+ BIT(0) | BIT(1), 3);
+ break;
+ default:
+ dev_err(codec->dev, "Unsupported MCLKI\n");
+ break;
+ }
+ } else {
+ snd_soc_update_bits(codec, ML26124_CLK_CTL,
+ BIT(0) | BIT(1), 0);
+ }
+
+ switch (params_rate(hw_params)) {
+ case 16000:
+ snd_soc_update_bits(codec, ML26124_SMPLING_RATE, 0xf,
+ get_srate(params_rate(hw_params)));
+ snd_soc_update_bits(codec, ML26124_PLLNL, 0xff,
+ coeff_div[i].pllnl);
+ snd_soc_update_bits(codec, ML26124_PLLNH, 0x1,
+ coeff_div[i].pllnh);
+ snd_soc_update_bits(codec, ML26124_PLLML, 0xff,
+ coeff_div[i].pllml);
+ snd_soc_update_bits(codec, ML26124_PLLMH, 0x3f,
+ coeff_div[i].pllmh);
+ snd_soc_update_bits(codec, ML26124_PLLDIV, 0x1f,
+ coeff_div[i].plldiv);
+ break;
+ case 32000:
+ snd_soc_update_bits(codec, ML26124_SMPLING_RATE, 0xf,
+ get_srate(params_rate(hw_params)));
+ snd_soc_update_bits(codec, ML26124_PLLNL, 0xff,
+ coeff_div[i].pllnl);
+ snd_soc_update_bits(codec, ML26124_PLLNH, 0x1,
+ coeff_div[i].pllnh);
+ snd_soc_update_bits(codec, ML26124_PLLML, 0xff,
+ coeff_div[i].pllml);
+ snd_soc_update_bits(codec, ML26124_PLLMH, 0x3f,
+ coeff_div[i].pllmh);
+ snd_soc_update_bits(codec, ML26124_PLLDIV, 0x1f,
+ coeff_div[i].plldiv);
+ break;
+ case 48000:
+ snd_soc_update_bits(codec, ML26124_SMPLING_RATE, 0xf,
+ get_srate(params_rate(hw_params)));
+ snd_soc_update_bits(codec, ML26124_PLLNL, 0xff,
+ coeff_div[i].pllnl);
+ snd_soc_update_bits(codec, ML26124_PLLNH, 0x1,
+ coeff_div[i].pllnh);
+ snd_soc_update_bits(codec, ML26124_PLLML, 0xff,
+ coeff_div[i].pllml);
+ snd_soc_update_bits(codec, ML26124_PLLMH, 0x3f,
+ coeff_div[i].pllmh);
+ snd_soc_update_bits(codec, ML26124_PLLDIV, 0x1f,
+ coeff_div[i].plldiv);
+ break;
+ default:
+ pr_err("%s:this rate is no support for ml26124\n", __func__);
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static int ml26124_mute(struct snd_soc_dai *dai, int mute)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ struct ml26124_priv *priv = snd_soc_codec_get_drvdata(codec);
+
+ switch (priv->substream->stream) {
+ case SNDRV_PCM_STREAM_CAPTURE:
+ snd_soc_update_bits(codec, ML26124_REC_PLYBAK_RUN, BIT(0), 1);
+ break;
+ case SNDRV_PCM_STREAM_PLAYBACK:
+ snd_soc_update_bits(codec, ML26124_REC_PLYBAK_RUN, BIT(1), 2);
+ break;
+ }
+
+ if (mute)
+ snd_soc_update_bits(codec, ML26124_DVOL_CTL, BIT(4),
+ DVOL_CTL_DVMUTE_ON);
+ else
+ snd_soc_update_bits(codec, ML26124_DVOL_CTL, BIT(4),
+ DVOL_CTL_DVMUTE_OFF);
+
+ return 0;
+}
+
+static int ml26124_set_dai_fmt(struct snd_soc_dai *codec_dai,
+ unsigned int fmt)
+{
+ unsigned char mode;
+ struct snd_soc_codec *codec = codec_dai->codec;
+
+ /* set master/slave audio interface */
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBM_CFM:
+ mode = 1;
+ break;
+ case SND_SOC_DAIFMT_CBS_CFS:
+ mode = 0;
+ break;
+ default:
+ return -EINVAL;
+ }
+ snd_soc_update_bits(codec, ML26124_SAI_MODE_SEL, BIT(0), mode);
+
+ /* interface format */
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ /* clock inversion */
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_NB_NF:
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static int ml26124_set_dai_sysclk(struct snd_soc_dai *codec_dai,
+ int clk_id, unsigned int freq, int dir)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ struct ml26124_priv *priv = snd_soc_codec_get_drvdata(codec);
+
+ switch (clk_id) {
+ case ML26124_USE_PLLOUT:
+ priv->clk_in = ML26124_USE_PLLOUT;
+ break;
+ case ML26124_USE_MCLKI:
+ priv->clk_in = ML26124_USE_MCLKI;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ priv->mclk = freq;
+
+ return 0;
+}
+
+static int ml26124_set_bias_level(struct snd_soc_codec *codec,
+ enum snd_soc_bias_level level)
+{
+ struct ml26124_priv *priv = snd_soc_codec_get_drvdata(codec);
+
+ switch (level) {
+ case SND_SOC_BIAS_ON:
+ snd_soc_update_bits(codec, ML26124_PW_SPAMP_PW_MNG,
+ ML26124_R26_MASK, ML26124_BLT_PREAMP_ON);
+ msleep(100);
+ snd_soc_update_bits(codec, ML26124_PW_SPAMP_PW_MNG,
+ ML26124_R26_MASK,
+ ML26124_MICBEN_ON | ML26124_BLT_ALL_ON);
+ break;
+ case SND_SOC_BIAS_PREPARE:
+ break;
+ case SND_SOC_BIAS_STANDBY:
+ /* VMID ON */
+ if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
+ snd_soc_update_bits(codec, ML26124_PW_REF_PW_MNG,
+ ML26124_VMID, ML26124_VMID);
+ msleep(500);
+ regcache_sync(priv->regmap);
+ }
+ break;
+ case SND_SOC_BIAS_OFF:
+ /* VMID OFF */
+ snd_soc_update_bits(codec, ML26124_PW_REF_PW_MNG,
+ ML26124_VMID, 0);
+ break;
+ }
+ codec->dapm.bias_level = level;
+ return 0;
+}
+
+static const struct snd_soc_dai_ops ml26124_dai_ops = {
+ .hw_params = ml26124_hw_params,
+ .digital_mute = ml26124_mute,
+ .set_fmt = ml26124_set_dai_fmt,
+ .set_sysclk = ml26124_set_dai_sysclk,
+};
+
+static struct snd_soc_dai_driver ml26124_dai = {
+ .name = "ml26124-hifi",
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = ML26124_RATES,
+ .formats = ML26124_FORMATS,},
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = ML26124_RATES,
+ .formats = ML26124_FORMATS,},
+ .ops = &ml26124_dai_ops,
+ .symmetric_rates = 1,
+};
+
+#ifdef CONFIG_PM
+static int ml26124_suspend(struct snd_soc_codec *codec)
+{
+ ml26124_set_bias_level(codec, SND_SOC_BIAS_OFF);
+
+ return 0;
+}
+
+static int ml26124_resume(struct snd_soc_codec *codec)
+{
+ ml26124_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+ return 0;
+}
+#else
+#define ml26124_suspend NULL
+#define ml26124_resume NULL
+#endif
+
+static int ml26124_probe(struct snd_soc_codec *codec)
+{
+ int ret;
+ struct ml26124_priv *priv = snd_soc_codec_get_drvdata(codec);
+ codec->control_data = priv->regmap;
+
+ ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP);
+ if (ret < 0) {
+ dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
+ return ret;
+ }
+
+ /* Software Reset */
+ snd_soc_update_bits(codec, ML26124_SW_RST, 0x01, 1);
+ snd_soc_update_bits(codec, ML26124_SW_RST, 0x01, 0);
+
+ ml26124_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+ return 0;
+}
+
+static struct snd_soc_codec_driver soc_codec_dev_ml26124 = {
+ .probe = ml26124_probe,
+ .suspend = ml26124_suspend,
+ .resume = ml26124_resume,
+ .set_bias_level = ml26124_set_bias_level,
+ .dapm_widgets = ml26124_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(ml26124_dapm_widgets),
+ .dapm_routes = ml26124_intercon,
+ .num_dapm_routes = ARRAY_SIZE(ml26124_intercon),
+ .controls = ml26124_snd_controls,
+ .num_controls = ARRAY_SIZE(ml26124_snd_controls),
+};
+
+static const struct regmap_config ml26124_i2c_regmap = {
+ .val_bits = 8,
+ .reg_bits = 8,
+ .max_register = ML26124_NUM_REGISTER,
+ .reg_defaults = ml26124_reg,
+ .num_reg_defaults = ARRAY_SIZE(ml26124_reg),
+ .cache_type = REGCACHE_RBTREE,
+ .write_flag_mask = 0x01,
+};
+
+static __devinit int ml26124_i2c_probe(struct i2c_client *i2c,
+ const struct i2c_device_id *id)
+{
+ struct ml26124_priv *priv;
+ int ret;
+
+ priv = devm_kzalloc(&i2c->dev, sizeof(*priv), GFP_KERNEL);
+ if (!priv)
+ return -ENOMEM;
+
+ i2c_set_clientdata(i2c, priv);
+
+ priv->regmap = 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);
+ return ret;
+ }
+
+ return snd_soc_register_codec(&i2c->dev,
+ &soc_codec_dev_ml26124, &ml26124_dai, 1);
+}
+
+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;
+}
+
+static const struct i2c_device_id ml26124_i2c_id[] = {
+ { "ml26124", 0 },
+ { }
+};
+MODULE_DEVICE_TABLE(i2c, ml26124_i2c_id);
+
+static struct i2c_driver ml26124_i2c_driver = {
+ .driver = {
+ .name = "ml26124",
+ .owner = THIS_MODULE,
+ },
+ .probe = ml26124_i2c_probe,
+ .remove = __devexit_p(ml26124_i2c_remove),
+ .id_table = ml26124_i2c_id,
+};
+
+module_i2c_driver(ml26124_i2c_driver);
+
+MODULE_AUTHOR("Tomoya MORINAGA <tomoya.rohm@gmail.com>");
+MODULE_DESCRIPTION("LAPIS Semiconductor ML26124 ALSA SoC codec driver");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/ml26124.h b/sound/soc/codecs/ml26124.h
new file mode 100644
index 00000000000..5ea0cbb8c46
--- /dev/null
+++ b/sound/soc/codecs/ml26124.h
@@ -0,0 +1,184 @@
+/*
+ * Copyright (C) 2011 LAPIS Semiconductor Co., 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; version 2 of the License.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * 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 ML26124_H
+#define ML26124_H
+
+/* Clock Control Register */
+#define ML26124_SMPLING_RATE 0x00
+#define ML26124_PLLNL 0x02
+#define ML26124_PLLNH 0x04
+#define ML26124_PLLML 0x06
+#define ML26124_PLLMH 0x08
+#define ML26124_PLLDIV 0x0a
+#define ML26124_CLK_EN 0x0c
+#define ML26124_CLK_CTL 0x0e
+
+/* System Control Register */
+#define ML26124_SW_RST 0x10
+#define ML26124_REC_PLYBAK_RUN 0x12
+#define ML26124_MIC_TIM 0x14
+
+/* Power Mnagement Register */
+#define ML26124_PW_REF_PW_MNG 0x20
+#define ML26124_PW_IN_PW_MNG 0x22
+#define ML26124_PW_DAC_PW_MNG 0x24
+#define ML26124_PW_SPAMP_PW_MNG 0x26
+#define ML26124_PW_LOUT_PW_MNG 0x28
+#define ML26124_PW_VOUT_PW_MNG 0x2a
+#define ML26124_PW_ZCCMP_PW_MNG 0x2e
+
+/* Analog Reference Control Register */
+#define ML26124_PW_MICBIAS_VOL 0x30
+
+/* Input/Output Amplifier Control Register */
+#define ML26124_PW_MIC_IN_VOL 0x32
+#define ML26124_PW_MIC_BOST_VOL 0x38
+#define ML26124_PW_SPK_AMP_VOL 0x3a
+#define ML26124_PW_AMP_VOL_FUNC 0x48
+#define ML26124_PW_AMP_VOL_FADE 0x4a
+
+/* Analog Path Control Register */
+#define ML26124_SPK_AMP_OUT 0x54
+#define ML26124_MIC_IF_CTL 0x5a
+#define ML26124_MIC_SELECT 0xe8
+
+/* Audio Interface Control Register */
+#define ML26124_SAI_TRANS_CTL 0x60
+#define ML26124_SAI_RCV_CTL 0x62
+#define ML26124_SAI_MODE_SEL 0x64
+
+/* DSP Control Register */
+#define ML26124_FILTER_EN 0x66
+#define ML26124_DVOL_CTL 0x68
+#define ML26124_MIXER_VOL_CTL 0x6a
+#define ML26124_RECORD_DIG_VOL 0x6c
+#define ML26124_PLBAK_DIG_VOL 0x70
+#define ML26124_DIGI_BOOST_VOL 0x72
+#define ML26124_EQ_GAIN_BRAND0 0x74
+#define ML26124_EQ_GAIN_BRAND1 0x76
+#define ML26124_EQ_GAIN_BRAND2 0x78
+#define ML26124_EQ_GAIN_BRAND3 0x7a
+#define ML26124_EQ_GAIN_BRAND4 0x7c
+#define ML26124_HPF2_CUTOFF 0x7e
+#define ML26124_EQBRAND0_F0L 0x80
+#define ML26124_EQBRAND0_F0H 0x82
+#define ML26124_EQBRAND0_F1L 0x84
+#define ML26124_EQBRAND0_F1H 0x86
+#define ML26124_EQBRAND1_F0L 0x88
+#define ML26124_EQBRAND1_F0H 0x8a
+#define ML26124_EQBRAND1_F1L 0x8c
+#define ML26124_EQBRAND1_F1H 0x8e
+#define ML26124_EQBRAND2_F0L 0x90
+#define ML26124_EQBRAND2_F0H 0x92
+#define ML26124_EQBRAND2_F1L 0x94
+#define ML26124_EQBRAND2_F1H 0x96
+#define ML26124_EQBRAND3_F0L 0x98
+#define ML26124_EQBRAND3_F0H 0x9a
+#define ML26124_EQBRAND3_F1L 0x9c
+#define ML26124_EQBRAND3_F1H 0x9e
+#define ML26124_EQBRAND4_F0L 0xa0
+#define ML26124_EQBRAND4_F0H 0xa2
+#define ML26124_EQBRAND4_F1L 0xa4
+#define ML26124_EQBRAND4_F1H 0xa6
+
+/* ALC Control Register */
+#define ML26124_ALC_MODE 0xb0
+#define ML26124_ALC_ATTACK_TIM 0xb2
+#define ML26124_ALC_DECAY_TIM 0xb4
+#define ML26124_ALC_HOLD_TIM 0xb6
+#define ML26124_ALC_TARGET_LEV 0xb8
+#define ML26124_ALC_MAXMIN_GAIN 0xba
+#define ML26124_NOIS_GATE_THRSH 0xbc
+#define ML26124_ALC_ZERO_TIMOUT 0xbe
+
+/* Playback Limiter Control Register */
+#define ML26124_PL_ATTACKTIME 0xc0
+#define ML26124_PL_DECAYTIME 0xc2
+#define ML26124_PL_TARGETTIME 0xc4
+#define ML26124_PL_MAXMIN_GAIN 0xc6
+#define ML26124_PLYBAK_BOST_VOL 0xc8
+#define ML26124_PL_0CROSS_TIMOUT 0xca
+
+/* Video Amplifer Control Register */
+#define ML26124_VIDEO_AMP_GAIN_CTL 0xd0
+#define ML26124_VIDEO_AMP_SETUP1 0xd2
+#define ML26124_VIDEO_AMP_CTL2 0xd4
+
+/* Clock select for machine driver */
+#define ML26124_USE_PLL 0
+#define ML26124_USE_MCLKI_256FS 1
+#define ML26124_USE_MCLKI_512FS 2
+#define ML26124_USE_MCLKI_1024FS 3
+
+/* Register Mask */
+#define ML26124_R0_MASK 0xf
+#define ML26124_R2_MASK 0xff
+#define ML26124_R4_MASK 0x1
+#define ML26124_R6_MASK 0xf
+#define ML26124_R8_MASK 0x3f
+#define ML26124_Ra_MASK 0x1f
+#define ML26124_Rc_MASK 0x1f
+#define ML26124_Re_MASK 0x7
+#define ML26124_R10_MASK 0x1
+#define ML26124_R12_MASK 0x17
+#define ML26124_R14_MASK 0x3f
+#define ML26124_R20_MASK 0x47
+#define ML26124_R22_MASK 0xa
+#define ML26124_R24_MASK 0x2
+#define ML26124_R26_MASK 0x1f
+#define ML26124_R28_MASK 0x2
+#define ML26124_R2a_MASK 0x2
+#define ML26124_R2e_MASK 0x2
+#define ML26124_R30_MASK 0x7
+#define ML26124_R32_MASK 0x3f
+#define ML26124_R38_MASK 0x38
+#define ML26124_R3a_MASK 0x3f
+#define ML26124_R48_MASK 0x3
+#define ML26124_R4a_MASK 0x7
+#define ML26124_R54_MASK 0x2a
+#define ML26124_R5a_MASK 0x3
+#define ML26124_Re8_MASK 0x3
+#define ML26124_R60_MASK 0xff
+#define ML26124_R62_MASK 0xff
+#define ML26124_R64_MASK 0x1
+#define ML26124_R66_MASK 0xff
+#define ML26124_R68_MASK 0x3b
+#define ML26124_R6a_MASK 0xf3
+#define ML26124_R6c_MASK 0xff
+#define ML26124_R70_MASK 0xff
+
+#define ML26124_MCLKEN BIT(0)
+#define ML26124_PLLEN BIT(1)
+#define ML26124_PLLOE BIT(2)
+#define ML26124_MCLKOE BIT(3)
+
+#define ML26124_BLT_ALL_ON 0x1f
+#define ML26124_BLT_PREAMP_ON 0x13
+
+#define ML26124_MICBEN_ON BIT(2)
+
+enum ml26124_regs {
+ ML26124_MCLK = 0,
+};
+
+enum ml26124_clk_in {
+ ML26124_USE_PLLOUT = 0,
+ ML26124_USE_MCLKI,
+};
+
+#endif
diff --git a/sound/soc/codecs/omap-hdmi.c b/sound/soc/codecs/omap-hdmi.c
new file mode 100644
index 00000000000..1bf5c74f5f9
--- /dev/null
+++ b/sound/soc/codecs/omap-hdmi.c
@@ -0,0 +1,69 @@
+/*
+ * ALSA SoC codec driver for HDMI audio on OMAP processors.
+ * Copyright (C) 2012 Texas Instruments Incorporated - http://www.ti.com/
+ * Author: Ricardo Neri <ricardo.neri@ti.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.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ *
+ */
+#include <linux/module.h>
+#include <sound/soc.h>
+
+#define DRV_NAME "hdmi-audio-codec"
+
+static struct snd_soc_codec_driver omap_hdmi_codec;
+
+static struct snd_soc_dai_driver omap_hdmi_codec_dai = {
+ .name = "omap-hdmi-hifi",
+ .playback = {
+ .channels_min = 2,
+ .channels_max = 8,
+ .rates = SNDRV_PCM_RATE_32000 |
+ SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |
+ SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 |
+ SNDRV_PCM_RATE_176400 | SNDRV_PCM_RATE_192000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S24_LE,
+ },
+};
+
+static __devinit int omap_hdmi_codec_probe(struct platform_device *pdev)
+{
+ return snd_soc_register_codec(&pdev->dev, &omap_hdmi_codec,
+ &omap_hdmi_codec_dai, 1);
+}
+
+static __devexit int omap_hdmi_codec_remove(struct platform_device *pdev)
+{
+ snd_soc_unregister_codec(&pdev->dev);
+ return 0;
+}
+
+static struct platform_driver omap_hdmi_codec_driver = {
+ .driver = {
+ .name = DRV_NAME,
+ .owner = THIS_MODULE,
+ },
+
+ .probe = omap_hdmi_codec_probe,
+ .remove = __devexit_p(omap_hdmi_codec_remove),
+};
+
+module_platform_driver(omap_hdmi_codec_driver);
+
+MODULE_AUTHOR("Ricardo Neri <ricardo.neri@ti.com>");
+MODULE_DESCRIPTION("ASoC OMAP HDMI codec driver");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:" DRV_NAME);
diff --git a/sound/soc/codecs/rt5631.c b/sound/soc/codecs/rt5631.c
index 20c324c7c34..960d0e93cce 100644
--- a/sound/soc/codecs/rt5631.c
+++ b/sound/soc/codecs/rt5631.c
@@ -18,7 +18,7 @@
#include <linux/delay.h>
#include <linux/pm.h>
#include <linux/i2c.h>
-#include <linux/spi/spi.h>
+#include <linux/regmap.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
@@ -30,6 +30,7 @@
#include "rt5631.h"
struct rt5631_priv {
+ struct regmap *regmap;
int codec_version;
int master;
int sysclk;
@@ -38,33 +39,33 @@ struct rt5631_priv {
int dmic_used_flag;
};
-static const u16 rt5631_reg[RT5631_VENDOR_ID2 + 1] = {
- [RT5631_SPK_OUT_VOL] = 0x8888,
- [RT5631_HP_OUT_VOL] = 0x8080,
- [RT5631_MONO_AXO_1_2_VOL] = 0xa080,
- [RT5631_AUX_IN_VOL] = 0x0808,
- [RT5631_ADC_REC_MIXER] = 0xf0f0,
- [RT5631_VDAC_DIG_VOL] = 0x0010,
- [RT5631_OUTMIXER_L_CTRL] = 0xffc0,
- [RT5631_OUTMIXER_R_CTRL] = 0xffc0,
- [RT5631_AXO1MIXER_CTRL] = 0x88c0,
- [RT5631_AXO2MIXER_CTRL] = 0x88c0,
- [RT5631_DIG_MIC_CTRL] = 0x3000,
- [RT5631_MONO_INPUT_VOL] = 0x8808,
- [RT5631_SPK_MIXER_CTRL] = 0xf8f8,
- [RT5631_SPK_MONO_OUT_CTRL] = 0xfc00,
- [RT5631_SPK_MONO_HP_OUT_CTRL] = 0x4440,
- [RT5631_SDP_CTRL] = 0x8000,
- [RT5631_MONO_SDP_CTRL] = 0x8000,
- [RT5631_STEREO_AD_DA_CLK_CTRL] = 0x2010,
- [RT5631_GEN_PUR_CTRL_REG] = 0x0e00,
- [RT5631_INT_ST_IRQ_CTRL_2] = 0x071a,
- [RT5631_MISC_CTRL] = 0x2040,
- [RT5631_DEPOP_FUN_CTRL_2] = 0x8000,
- [RT5631_SOFT_VOL_CTRL] = 0x07e0,
- [RT5631_ALC_CTRL_1] = 0x0206,
- [RT5631_ALC_CTRL_3] = 0x2000,
- [RT5631_PSEUDO_SPATL_CTRL] = 0x0553,
+static const struct reg_default rt5631_reg[] = {
+ { RT5631_SPK_OUT_VOL, 0x8888 },
+ { RT5631_HP_OUT_VOL, 0x8080 },
+ { RT5631_MONO_AXO_1_2_VOL, 0xa080 },
+ { RT5631_AUX_IN_VOL, 0x0808 },
+ { RT5631_ADC_REC_MIXER, 0xf0f0 },
+ { RT5631_VDAC_DIG_VOL, 0x0010 },
+ { RT5631_OUTMIXER_L_CTRL, 0xffc0 },
+ { RT5631_OUTMIXER_R_CTRL, 0xffc0 },
+ { RT5631_AXO1MIXER_CTRL, 0x88c0 },
+ { RT5631_AXO2MIXER_CTRL, 0x88c0 },
+ { RT5631_DIG_MIC_CTRL, 0x3000 },
+ { RT5631_MONO_INPUT_VOL, 0x8808 },
+ { RT5631_SPK_MIXER_CTRL, 0xf8f8 },
+ { RT5631_SPK_MONO_OUT_CTRL, 0xfc00 },
+ { RT5631_SPK_MONO_HP_OUT_CTRL, 0x4440 },
+ { RT5631_SDP_CTRL, 0x8000 },
+ { RT5631_MONO_SDP_CTRL, 0x8000 },
+ { RT5631_STEREO_AD_DA_CLK_CTRL, 0x2010 },
+ { RT5631_GEN_PUR_CTRL_REG, 0x0e00 },
+ { RT5631_INT_ST_IRQ_CTRL_2, 0x071a },
+ { RT5631_MISC_CTRL, 0x2040 },
+ { RT5631_DEPOP_FUN_CTRL_2, 0x8000 },
+ { RT5631_SOFT_VOL_CTRL, 0x07e0 },
+ { RT5631_ALC_CTRL_1, 0x0206 },
+ { RT5631_ALC_CTRL_3, 0x2000 },
+ { RT5631_PSEUDO_SPATL_CTRL, 0x0553 },
};
/**
@@ -96,8 +97,7 @@ static int rt5631_reset(struct snd_soc_codec *codec)
return snd_soc_write(codec, RT5631_RESET, 0);
}
-static int rt5631_volatile_register(struct snd_soc_codec *codec,
- unsigned int reg)
+static bool rt5631_volatile_register(struct device *dev, unsigned int reg)
{
switch (reg) {
case RT5631_RESET:
@@ -111,8 +111,7 @@ static int rt5631_volatile_register(struct snd_soc_codec *codec,
}
}
-static int rt5631_readable_register(struct snd_soc_codec *codec,
- unsigned int reg)
+static bool rt5631_readable_register(struct device *dev, unsigned int reg)
{
switch (reg) {
case RT5631_RESET:
@@ -1361,8 +1360,7 @@ static int get_coeff(int mclk, int rate, int timesofbclk)
static int rt5631_hifi_pcm_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params, struct snd_soc_dai *dai)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_codec *codec = dai->codec;
struct rt5631_priv *rt5631 = snd_soc_codec_get_drvdata(codec);
int timesofbclk = 32, coeff;
unsigned int iface = 0;
@@ -1544,6 +1542,8 @@ static int rt5631_codec_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id,
static int rt5631_set_bias_level(struct snd_soc_codec *codec,
enum snd_soc_bias_level level)
{
+ struct rt5631_priv *rt5631 = snd_soc_codec_get_drvdata(codec);
+
switch (level) {
case SND_SOC_BIAS_ON:
case SND_SOC_BIAS_PREPARE:
@@ -1561,8 +1561,8 @@ static int rt5631_set_bias_level(struct snd_soc_codec *codec,
snd_soc_update_bits(codec, RT5631_PWR_MANAG_ADD3,
RT5631_PWR_FAST_VREF_CTRL,
RT5631_PWR_FAST_VREF_CTRL);
- codec->cache_only = false;
- snd_soc_cache_sync(codec);
+ regcache_cache_only(rt5631->regmap, false);
+ regcache_sync(rt5631->regmap);
}
break;
@@ -1587,7 +1587,9 @@ static int rt5631_probe(struct snd_soc_codec *codec)
unsigned int val;
int ret;
- ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_I2C);
+ codec->control_data = rt5631->regmap;
+
+ ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP);
if (ret != 0) {
dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
return ret;
@@ -1698,12 +1700,6 @@ static struct snd_soc_codec_driver soc_codec_dev_rt5631 = {
.suspend = rt5631_suspend,
.resume = rt5631_resume,
.set_bias_level = rt5631_set_bias_level,
- .reg_cache_size = RT5631_VENDOR_ID2 + 1,
- .reg_word_size = sizeof(u16),
- .reg_cache_default = rt5631_reg,
- .volatile_register = rt5631_volatile_register,
- .readable_register = rt5631_readable_register,
- .reg_cache_step = 1,
.controls = rt5631_snd_controls,
.num_controls = ARRAY_SIZE(rt5631_snd_controls),
.dapm_widgets = rt5631_dapm_widgets,
@@ -1718,6 +1714,18 @@ static const struct i2c_device_id rt5631_i2c_id[] = {
};
MODULE_DEVICE_TABLE(i2c, rt5631_i2c_id);
+static const struct regmap_config rt5631_regmap_config = {
+ .reg_bits = 8,
+ .val_bits = 16,
+
+ .readable_reg = rt5631_readable_register,
+ .volatile_reg = rt5631_volatile_register,
+ .max_register = RT5631_VENDOR_ID2,
+ .reg_defaults = rt5631_reg,
+ .num_reg_defaults = ARRAY_SIZE(rt5631_reg),
+ .cache_type = REGCACHE_RBTREE,
+};
+
static int rt5631_i2c_probe(struct i2c_client *i2c,
const struct i2c_device_id *id)
{
@@ -1731,6 +1739,10 @@ static int rt5631_i2c_probe(struct i2c_client *i2c,
i2c_set_clientdata(i2c, rt5631);
+ rt5631->regmap = devm_regmap_init_i2c(i2c, &rt5631_regmap_config);
+ if (IS_ERR(rt5631->regmap))
+ return PTR_ERR(rt5631->regmap);
+
ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5631,
rt5631_dai, ARRAY_SIZE(rt5631_dai));
return ret;
@@ -1752,17 +1764,7 @@ static struct i2c_driver rt5631_i2c_driver = {
.id_table = rt5631_i2c_id,
};
-static int __init rt5631_modinit(void)
-{
- return i2c_add_driver(&rt5631_i2c_driver);
-}
-module_init(rt5631_modinit);
-
-static void __exit rt5631_modexit(void)
-{
- i2c_del_driver(&rt5631_i2c_driver);
-}
-module_exit(rt5631_modexit);
+module_i2c_driver(rt5631_i2c_driver);
MODULE_DESCRIPTION("ASoC RT5631 driver");
MODULE_AUTHOR("flove <flove@realtek.com>");
diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c
index 8e92fb88ed0..8af6a5245b1 100644
--- a/sound/soc/codecs/sgtl5000.c
+++ b/sound/soc/codecs/sgtl5000.c
@@ -84,8 +84,8 @@ static struct regulator_consumer_supply ldo_consumer[] = {
static struct regulator_init_data ldo_init_data = {
.constraints = {
- .min_uV = 850000,
- .max_uV = 1600000,
+ .min_uV = 1200000,
+ .max_uV = 1200000,
.valid_modes_mask = REGULATOR_MODE_NORMAL,
.valid_ops_mask = REGULATOR_CHANGE_STATUS,
},
@@ -197,9 +197,9 @@ static const struct snd_soc_dapm_widget sgtl5000_dapm_widgets[] = {
SND_SOC_DAPM_OUTPUT("HP_OUT"),
SND_SOC_DAPM_OUTPUT("LINE_OUT"),
- SND_SOC_DAPM_MICBIAS_E("Mic Bias", SGTL5000_CHIP_MIC_CTRL, 8, 0,
- mic_bias_event,
- SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
+ SND_SOC_DAPM_SUPPLY("Mic Bias", SGTL5000_CHIP_MIC_CTRL, 8, 0,
+ mic_bias_event,
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
SND_SOC_DAPM_PGA("HP", SGTL5000_CHIP_ANA_POWER, 4, 0, NULL, 0),
SND_SOC_DAPM_PGA("LO", SGTL5000_CHIP_ANA_POWER, 0, 0, NULL, 0),
@@ -665,8 +665,7 @@ static int sgtl5000_pcm_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;
+ struct snd_soc_codec *codec = dai->codec;
struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec);
int channels = params_channels(params);
int i2s_ctl = 0;
@@ -809,6 +808,7 @@ static int ldo_regulator_register(struct snd_soc_codec *codec,
{
struct ldo_regulator *ldo;
struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec);
+ struct regulator_config config = { };
ldo = kzalloc(sizeof(struct ldo_regulator), GFP_KERNEL);
@@ -832,8 +832,11 @@ static int ldo_regulator_register(struct snd_soc_codec *codec,
ldo->codec_data = codec;
ldo->voltage = voltage;
- ldo->dev = regulator_register(&ldo->desc, codec->dev,
- init_data, ldo, NULL);
+ config.dev = codec->dev;
+ config.driver_data = ldo;
+ config.init_data = init_data;
+
+ ldo->dev = regulator_register(&ldo->desc, &config);
if (IS_ERR(ldo->dev)) {
int ret = PTR_ERR(ldo->dev);
@@ -1451,17 +1454,7 @@ static struct i2c_driver sgtl5000_i2c_driver = {
.id_table = sgtl5000_id,
};
-static int __init sgtl5000_modinit(void)
-{
- return i2c_add_driver(&sgtl5000_i2c_driver);
-}
-module_init(sgtl5000_modinit);
-
-static void __exit sgtl5000_exit(void)
-{
- i2c_del_driver(&sgtl5000_i2c_driver);
-}
-module_exit(sgtl5000_exit);
+module_i2c_driver(sgtl5000_i2c_driver);
MODULE_DESCRIPTION("Freescale SGTL5000 ALSA SoC Codec Driver");
MODULE_AUTHOR("Zeng Zhaoming <zengzm.kernel@gmail.com>");
diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c
index de2b20544ce..079066fef42 100644
--- a/sound/soc/codecs/ssm2602.c
+++ b/sound/soc/codecs/ssm2602.c
@@ -33,6 +33,7 @@
#include <linux/pm.h>
#include <linux/i2c.h>
#include <linux/spi/spi.h>
+#include <linux/regmap.h>
#include <linux/slab.h>
#include <sound/core.h>
#include <sound/pcm.h>
@@ -43,8 +44,6 @@
#include "ssm2602.h"
-#define SSM2602_VERSION "0.1"
-
enum ssm2602_type {
SSM2602,
SSM2604,
@@ -53,10 +52,12 @@ enum ssm2602_type {
/* codec private data */
struct ssm2602_priv {
unsigned int sysclk;
- enum snd_soc_control_type control_type;
+ struct snd_pcm_hw_constraint_list *sysclk_constraints;
struct snd_pcm_substream *master_substream;
struct snd_pcm_substream *slave_substream;
+ struct regmap *regmap;
+
enum ssm2602_type type;
unsigned int clk_out_pwr;
};
@@ -73,7 +74,6 @@ static const u16 ssm2602_reg[SSM2602_CACHEREGNUM] = {
0x0000, 0x0000
};
-#define ssm2602_reset(c) snd_soc_write(c, SSM2602_RESET, 0)
/*Appending several "None"s just for OSS mixer use*/
static const char *ssm2602_input_select[] = {
@@ -195,6 +195,24 @@ static const struct snd_soc_dapm_route ssm2604_routes[] = {
{"ADC", NULL, "Line Input"},
};
+static const unsigned int ssm2602_rates_12288000[] = {
+ 8000, 32000, 48000, 96000,
+};
+
+static struct snd_pcm_hw_constraint_list ssm2602_constraints_12288000 = {
+ .list = ssm2602_rates_12288000,
+ .count = ARRAY_SIZE(ssm2602_rates_12288000),
+};
+
+static const unsigned int ssm2602_rates_11289600[] = {
+ 8000, 44100, 88200,
+};
+
+static struct snd_pcm_hw_constraint_list ssm2602_constraints_11289600 = {
+ .list = ssm2602_rates_11289600,
+ .count = ARRAY_SIZE(ssm2602_rates_11289600),
+};
+
struct ssm2602_coeff {
u32 mclk;
u32 rate;
@@ -254,11 +272,10 @@ static int ssm2602_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;
+ struct snd_soc_codec *codec = dai->codec;
struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec);
- u16 iface = snd_soc_read(codec, SSM2602_IFACE) & 0xfff3;
int srate = ssm2602_get_coeff(ssm2602->sysclk, params_rate(params));
+ unsigned int iface;
if (substream == ssm2602->slave_substream) {
dev_dbg(codec->dev, "Ignoring hw_params for slave substream\n");
@@ -268,31 +285,34 @@ static int ssm2602_hw_params(struct snd_pcm_substream *substream,
if (srate < 0)
return srate;
- snd_soc_write(codec, SSM2602_SRATE, srate);
+ regmap_write(ssm2602->regmap, SSM2602_SRATE, srate);
/* bit size */
switch (params_format(params)) {
case SNDRV_PCM_FORMAT_S16_LE:
+ iface = 0x0;
break;
case SNDRV_PCM_FORMAT_S20_3LE:
- iface |= 0x0004;
+ iface = 0x4;
break;
case SNDRV_PCM_FORMAT_S24_LE:
- iface |= 0x0008;
+ iface = 0x8;
break;
case SNDRV_PCM_FORMAT_S32_LE:
- iface |= 0x000c;
+ iface = 0xc;
break;
+ default:
+ return -EINVAL;
}
- snd_soc_write(codec, SSM2602_IFACE, iface);
+ regmap_update_bits(ssm2602->regmap, SSM2602_IFACE,
+ IFACE_AUDIO_DATA_LEN, iface);
return 0;
}
static int ssm2602_startup(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_codec *codec = dai->codec;
struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec);
struct snd_pcm_runtime *master_runtime;
@@ -322,14 +342,19 @@ static int ssm2602_startup(struct snd_pcm_substream *substream,
} else
ssm2602->master_substream = substream;
+ if (ssm2602->sysclk_constraints) {
+ snd_pcm_hw_constraint_list(substream->runtime, 0,
+ SNDRV_PCM_HW_PARAM_RATE,
+ ssm2602->sysclk_constraints);
+ }
+
return 0;
}
static void ssm2602_shutdown(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_codec *codec = dai->codec;
struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec);
if (ssm2602->master_substream == substream)
@@ -341,14 +366,14 @@ static void ssm2602_shutdown(struct snd_pcm_substream *substream,
static int ssm2602_mute(struct snd_soc_dai *dai, int mute)
{
- struct snd_soc_codec *codec = dai->codec;
+ struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(dai->codec);
if (mute)
- snd_soc_update_bits(codec, SSM2602_APDIGI,
+ regmap_update_bits(ssm2602->regmap, SSM2602_APDIGI,
APDIGI_ENABLE_DAC_MUTE,
APDIGI_ENABLE_DAC_MUTE);
else
- snd_soc_update_bits(codec, SSM2602_APDIGI,
+ regmap_update_bits(ssm2602->regmap, SSM2602_APDIGI,
APDIGI_ENABLE_DAC_MUTE, 0);
return 0;
}
@@ -364,16 +389,21 @@ static int ssm2602_set_dai_sysclk(struct snd_soc_dai *codec_dai,
return -EINVAL;
switch (freq) {
- case 11289600:
- case 12000000:
case 12288000:
- case 16934400:
case 18432000:
- ssm2602->sysclk = freq;
+ ssm2602->sysclk_constraints = &ssm2602_constraints_12288000;
+ break;
+ case 11289600:
+ case 16934400:
+ ssm2602->sysclk_constraints = &ssm2602_constraints_11289600;
+ break;
+ case 12000000:
+ ssm2602->sysclk_constraints = NULL;
break;
default:
return -EINVAL;
}
+ ssm2602->sysclk = freq;
} else {
unsigned int mask;
@@ -393,7 +423,7 @@ static int ssm2602_set_dai_sysclk(struct snd_soc_dai *codec_dai,
else
ssm2602->clk_out_pwr &= ~mask;
- snd_soc_update_bits(codec, SSM2602_PWR,
+ regmap_update_bits(ssm2602->regmap, SSM2602_PWR,
PWR_CLK_OUT_PDN | PWR_OSC_PDN, ssm2602->clk_out_pwr);
}
@@ -403,8 +433,8 @@ static int ssm2602_set_dai_sysclk(struct snd_soc_dai *codec_dai,
static int ssm2602_set_dai_fmt(struct snd_soc_dai *codec_dai,
unsigned int fmt)
{
- struct snd_soc_codec *codec = codec_dai->codec;
- u16 iface = 0;
+ struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec_dai->codec);
+ unsigned int iface = 0;
/* set master/slave audio interface */
switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
@@ -455,7 +485,7 @@ static int ssm2602_set_dai_fmt(struct snd_soc_dai *codec_dai,
}
/* set iface */
- snd_soc_write(codec, SSM2602_IFACE, iface);
+ regmap_write(ssm2602->regmap, SSM2602_IFACE, iface);
return 0;
}
@@ -467,7 +497,7 @@ static int ssm2602_set_bias_level(struct snd_soc_codec *codec,
switch (level) {
case SND_SOC_BIAS_ON:
/* vref/mid on, osc and clkout on if enabled */
- snd_soc_update_bits(codec, SSM2602_PWR,
+ regmap_update_bits(ssm2602->regmap, SSM2602_PWR,
PWR_POWER_OFF | PWR_CLK_OUT_PDN | PWR_OSC_PDN,
ssm2602->clk_out_pwr);
break;
@@ -475,13 +505,13 @@ static int ssm2602_set_bias_level(struct snd_soc_codec *codec,
break;
case SND_SOC_BIAS_STANDBY:
/* everything off except vref/vmid, */
- snd_soc_update_bits(codec, SSM2602_PWR,
+ regmap_update_bits(ssm2602->regmap, SSM2602_PWR,
PWR_POWER_OFF | PWR_CLK_OUT_PDN | PWR_OSC_PDN,
PWR_CLK_OUT_PDN | PWR_OSC_PDN);
break;
case SND_SOC_BIAS_OFF:
/* everything off */
- snd_soc_update_bits(codec, SSM2602_PWR,
+ regmap_update_bits(ssm2602->regmap, SSM2602_PWR,
PWR_POWER_OFF, PWR_POWER_OFF);
break;
@@ -540,12 +570,13 @@ static int ssm2602_resume(struct snd_soc_codec *codec)
static int ssm2602_probe(struct snd_soc_codec *codec)
{
+ struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec);
struct snd_soc_dapm_context *dapm = &codec->dapm;
int ret;
- snd_soc_update_bits(codec, SSM2602_LOUT1V,
+ regmap_update_bits(ssm2602->regmap, SSM2602_LOUT1V,
LOUT1V_LRHP_BOTH, LOUT1V_LRHP_BOTH);
- snd_soc_update_bits(codec, SSM2602_ROUT1V,
+ regmap_update_bits(ssm2602->regmap, SSM2602_ROUT1V,
ROUT1V_RLHP_BOTH, ROUT1V_RLHP_BOTH);
ret = snd_soc_add_codec_controls(codec, ssm2602_snd_controls,
@@ -581,27 +612,26 @@ static int ssm260x_probe(struct snd_soc_codec *codec)
struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec);
int ret;
- pr_info("ssm2602 Audio Codec %s", SSM2602_VERSION);
-
- ret = snd_soc_codec_set_cache_io(codec, 7, 9, ssm2602->control_type);
+ codec->control_data = ssm2602->regmap;
+ ret = snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP);
if (ret < 0) {
dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
return ret;
}
- ret = ssm2602_reset(codec);
+ ret = regmap_write(ssm2602->regmap, SSM2602_RESET, 0);
if (ret < 0) {
dev_err(codec->dev, "Failed to issue reset: %d\n", ret);
return ret;
}
/* set the update bits */
- snd_soc_update_bits(codec, SSM2602_LINVOL,
+ regmap_update_bits(ssm2602->regmap, SSM2602_LINVOL,
LINVOL_LRIN_BOTH, LINVOL_LRIN_BOTH);
- snd_soc_update_bits(codec, SSM2602_RINVOL,
+ regmap_update_bits(ssm2602->regmap, SSM2602_RINVOL,
RINVOL_RLIN_BOTH, RINVOL_RLIN_BOTH);
/*select Line in as default input*/
- snd_soc_write(codec, SSM2602_APANA, APANA_SELECT_DAC |
+ regmap_write(ssm2602->regmap, SSM2602_APANA, APANA_SELECT_DAC |
APANA_ENABLE_MIC_BOOST);
switch (ssm2602->type) {
@@ -634,9 +664,6 @@ static struct snd_soc_codec_driver soc_codec_dev_ssm2602 = {
.suspend = ssm2602_suspend,
.resume = ssm2602_resume,
.set_bias_level = ssm2602_set_bias_level,
- .reg_cache_size = ARRAY_SIZE(ssm2602_reg),
- .reg_word_size = sizeof(u16),
- .reg_cache_default = ssm2602_reg,
.controls = ssm260x_snd_controls,
.num_controls = ARRAY_SIZE(ssm260x_snd_controls),
@@ -646,6 +673,23 @@ static struct snd_soc_codec_driver soc_codec_dev_ssm2602 = {
.num_dapm_routes = ARRAY_SIZE(ssm260x_routes),
};
+static bool ssm2602_register_volatile(struct device *dev, unsigned int reg)
+{
+ return reg == SSM2602_RESET;
+}
+
+static const struct regmap_config ssm2602_regmap_config = {
+ .val_bits = 9,
+ .reg_bits = 7,
+
+ .max_register = SSM2602_RESET,
+ .volatile_reg = ssm2602_register_volatile,
+
+ .cache_type = REGCACHE_RBTREE,
+ .reg_defaults_raw = ssm2602_reg,
+ .num_reg_defaults_raw = ARRAY_SIZE(ssm2602_reg),
+};
+
#if defined(CONFIG_SPI_MASTER)
static int __devinit ssm2602_spi_probe(struct spi_device *spi)
{
@@ -658,9 +702,12 @@ static int __devinit ssm2602_spi_probe(struct spi_device *spi)
return -ENOMEM;
spi_set_drvdata(spi, ssm2602);
- ssm2602->control_type = SND_SOC_SPI;
ssm2602->type = SSM2602;
+ ssm2602->regmap = devm_regmap_init_spi(spi, &ssm2602_regmap_config);
+ if (IS_ERR(ssm2602->regmap))
+ return PTR_ERR(ssm2602->regmap);
+
ret = snd_soc_register_codec(&spi->dev,
&soc_codec_dev_ssm2602, &ssm2602_dai, 1);
return ret;
@@ -701,9 +748,12 @@ static int __devinit ssm2602_i2c_probe(struct i2c_client *i2c,
return -ENOMEM;
i2c_set_clientdata(i2c, ssm2602);
- ssm2602->control_type = SND_SOC_I2C;
ssm2602->type = id->driver_data;
+ ssm2602->regmap = devm_regmap_init_i2c(i2c, &ssm2602_regmap_config);
+ if (IS_ERR(ssm2602->regmap))
+ return PTR_ERR(ssm2602->regmap);
+
ret = snd_soc_register_codec(&i2c->dev,
&soc_codec_dev_ssm2602, &ssm2602_dai, 1);
return ret;
diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c
index 7db6fa51502..8d717f4b5a8 100644
--- a/sound/soc/codecs/sta32x.c
+++ b/sound/soc/codecs/sta32x.c
@@ -609,8 +609,7 @@ static int sta32x_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;
+ struct snd_soc_codec *codec = dai->codec;
struct sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec);
unsigned int rate;
int i, mcs = -1, ir = -1;
diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c
index df1e07ffac3..31762ebdd77 100644
--- a/sound/soc/codecs/tlv320aic23.c
+++ b/sound/soc/codecs/tlv320aic23.c
@@ -34,8 +34,6 @@
#include "tlv320aic23.h"
-#define AIC23_VERSION "0.1"
-
/*
* AIC23 register cache
*/
@@ -325,8 +323,7 @@ static int tlv320aic23_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;
+ struct snd_soc_codec *codec = dai->codec;
u16 iface_reg;
int ret;
struct aic23 *aic23 = snd_soc_codec_get_drvdata(codec);
@@ -371,8 +368,7 @@ static int tlv320aic23_hw_params(struct snd_pcm_substream *substream,
static int tlv320aic23_pcm_prepare(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_codec *codec = dai->codec;
/* set active */
snd_soc_write(codec, TLV320AIC23_ACTIVE, 0x0001);
@@ -383,8 +379,7 @@ static int tlv320aic23_pcm_prepare(struct snd_pcm_substream *substream,
static void tlv320aic23_shutdown(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_codec *codec = dai->codec;
struct aic23 *aic23 = snd_soc_codec_get_drvdata(codec);
/* deactivate */
@@ -548,8 +543,6 @@ static int tlv320aic23_probe(struct snd_soc_codec *codec)
struct aic23 *aic23 = snd_soc_codec_get_drvdata(codec);
int ret;
- printk(KERN_INFO "AIC23 Audio Codec %s\n", AIC23_VERSION);
-
ret = snd_soc_codec_set_cache_io(codec, 7, 9, aic23->control_type);
if (ret < 0) {
dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c
index 802064b5030..85944e95357 100644
--- a/sound/soc/codecs/tlv320aic26.c
+++ b/sound/soc/codecs/tlv320aic26.c
@@ -126,8 +126,7 @@ static int aic26_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;
+ struct snd_soc_codec *codec = dai->codec;
struct aic26 *aic26 = snd_soc_codec_get_drvdata(codec);
int fsref, divisor, wlen, pval, jval, dval, qval;
u16 reg;
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index 8d20f6ec20f..e9b62b5ea63 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -802,8 +802,7 @@ static int aic3x_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;
+ struct snd_soc_codec *codec = dai->codec;
struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec);
int codec_clk = 0, bypass_pll = 0, fsref, last_clk = 0;
u8 data, j, r, p, pll_q, pll_p = 1, pll_r = 1, pll_j = 1;
@@ -936,9 +935,7 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream,
}
found:
- data = snd_soc_read(codec, AIC3X_PLL_PROGA_REG);
- snd_soc_write(codec, AIC3X_PLL_PROGA_REG,
- data | (pll_p << PLLP_SHIFT));
+ snd_soc_update_bits(codec, AIC3X_PLL_PROGA_REG, PLLP_MASK, pll_p);
snd_soc_write(codec, AIC3X_OVRF_STATUS_AND_PLLR_REG,
pll_r << PLLR_SHIFT);
snd_soc_write(codec, AIC3X_PLL_PROGB_REG, pll_j << PLLJ_SHIFT);
@@ -1161,24 +1158,6 @@ static int aic3x_set_bias_level(struct snd_soc_codec *codec,
return 0;
}
-void aic3x_set_headset_detection(struct snd_soc_codec *codec, int detect,
- int headset_debounce, int button_debounce)
-{
- u8 val;
-
- val = ((detect & AIC3X_HEADSET_DETECT_MASK)
- << AIC3X_HEADSET_DETECT_SHIFT) |
- ((headset_debounce & AIC3X_HEADSET_DEBOUNCE_MASK)
- << AIC3X_HEADSET_DEBOUNCE_SHIFT) |
- ((button_debounce & AIC3X_BUTTON_DEBOUNCE_MASK)
- << AIC3X_BUTTON_DEBOUNCE_SHIFT);
-
- if (detect & AIC3X_HEADSET_DETECT_MASK)
- val |= AIC3X_HEADSET_DETECT_ENABLED;
-
- snd_soc_write(codec, AIC3X_HEADSET_DETECT_CTRL_A, val);
-}
-
#define AIC3X_RATES SNDRV_PCM_RATE_8000_96000
#define AIC3X_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE)
diff --git a/sound/soc/codecs/tlv320aic3x.h b/sound/soc/codecs/tlv320aic3x.h
index 6f097fb6068..08c7f6685ff 100644
--- a/sound/soc/codecs/tlv320aic3x.h
+++ b/sound/soc/codecs/tlv320aic3x.h
@@ -166,6 +166,7 @@
/* PLL registers bitfields */
#define PLLP_SHIFT 0
+#define PLLP_MASK 7
#define PLLQ_SHIFT 3
#define PLLR_SHIFT 0
#define PLLJ_SHIFT 2
diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c
index 4587ddd0fbf..0dd41077ab7 100644
--- a/sound/soc/codecs/tlv320dac33.c
+++ b/sound/soc/codecs/tlv320dac33.c
@@ -62,8 +62,10 @@
#define UTHR_FROM_PERIOD_SIZE(samples, playrate, burstrate) \
(((samples)*5000) / (((burstrate)*5000) / ((burstrate) - (playrate))))
-static void dac33_calculate_times(struct snd_pcm_substream *substream);
-static int dac33_prepare_chip(struct snd_pcm_substream *substream);
+static void dac33_calculate_times(struct snd_pcm_substream *substream,
+ struct snd_soc_codec *codec);
+static int dac33_prepare_chip(struct snd_pcm_substream *substream,
+ struct snd_soc_codec *codec);
enum dac33_state {
DAC33_IDLE = 0,
@@ -427,8 +429,8 @@ static int dac33_playback_event(struct snd_soc_dapm_widget *w,
switch (event) {
case SND_SOC_DAPM_PRE_PMU:
if (likely(dac33->substream)) {
- dac33_calculate_times(dac33->substream);
- dac33_prepare_chip(dac33->substream);
+ dac33_calculate_times(dac33->substream, w->codec);
+ dac33_prepare_chip(dac33->substream, w->codec);
}
break;
case SND_SOC_DAPM_POST_PMD:
@@ -799,8 +801,7 @@ static void dac33_oscwait(struct snd_soc_codec *codec)
static int dac33_startup(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_codec *codec = dai->codec;
struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec);
/* Stream started, save the substream pointer */
@@ -812,8 +813,7 @@ static int dac33_startup(struct snd_pcm_substream *substream,
static void dac33_shutdown(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_codec *codec = dai->codec;
struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec);
dac33->substream = NULL;
@@ -825,8 +825,7 @@ static int dac33_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;
+ struct snd_soc_codec *codec = dai->codec;
struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec);
/* Check parameters for validity */
@@ -868,10 +867,9 @@ static int dac33_hw_params(struct snd_pcm_substream *substream,
* writes happens in different order, than dac33 might end up in unknown state.
* Use the known, working sequence of register writes to initialize the dac33.
*/
-static int dac33_prepare_chip(struct snd_pcm_substream *substream)
+static int dac33_prepare_chip(struct snd_pcm_substream *substream,
+ struct snd_soc_codec *codec)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_codec *codec = rtd->codec;
struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec);
unsigned int oscset, ratioset, pwr_ctrl, reg_tmp;
u8 aictrl_a, aictrl_b, fifoctrl_a;
@@ -1067,10 +1065,9 @@ static int dac33_prepare_chip(struct snd_pcm_substream *substream)
return 0;
}
-static void dac33_calculate_times(struct snd_pcm_substream *substream)
+static void dac33_calculate_times(struct snd_pcm_substream *substream,
+ struct snd_soc_codec *codec)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_codec *codec = rtd->codec;
struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec);
unsigned int period_size = substream->runtime->period_size;
unsigned int rate = substream->runtime->rate;
@@ -1128,8 +1125,7 @@ static void dac33_calculate_times(struct snd_pcm_substream *substream)
static int dac33_pcm_trigger(struct snd_pcm_substream *substream, int cmd,
struct snd_soc_dai *dai)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_codec *codec = dai->codec;
struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec);
int ret = 0;
@@ -1161,8 +1157,7 @@ static snd_pcm_sframes_t dac33_dai_delay(
struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_codec *codec = dai->codec;
struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec);
unsigned long long t0, t1, t_now;
unsigned int time_delta, uthr;
diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c
index 170cf9a8fc7..391fcfc7b63 100644
--- a/sound/soc/codecs/twl4030.c
+++ b/sound/soc/codecs/twl4030.c
@@ -1685,8 +1685,7 @@ static void twl4030_tdm_enable(struct snd_soc_codec *codec, int direction,
static int twl4030_startup(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_codec *codec = dai->codec;
struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec);
if (twl4030->master_substream) {
@@ -1715,8 +1714,7 @@ static int twl4030_startup(struct snd_pcm_substream *substream,
static void twl4030_shutdown(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_codec *codec = dai->codec;
struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec);
if (twl4030->master_substream == substream)
@@ -1740,8 +1738,7 @@ static int twl4030_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;
+ struct snd_soc_codec *codec = dai->codec;
struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec);
u8 mode, old_mode, format, old_format;
@@ -1974,8 +1971,7 @@ static void twl4030_voice_enable(struct snd_soc_codec *codec, int direction,
static int twl4030_voice_startup(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_codec *codec = dai->codec;
struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec);
u8 mode;
@@ -2007,8 +2003,7 @@ static int twl4030_voice_startup(struct snd_pcm_substream *substream,
static void twl4030_voice_shutdown(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_codec *codec = dai->codec;
/* Enable voice digital filters */
twl4030_voice_enable(codec, substream->stream, 0);
@@ -2017,8 +2012,7 @@ static void twl4030_voice_shutdown(struct snd_pcm_substream *substream,
static int twl4030_voice_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;
+ struct snd_soc_codec *codec = dai->codec;
struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec);
u8 old_mode, mode;
diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c
index dc7509b9d53..a36e9fcdf18 100644
--- a/sound/soc/codecs/twl6040.c
+++ b/sound/soc/codecs/twl6040.c
@@ -46,17 +46,6 @@
#define TWL6040_OUTHF_0dB 0x03
#define TWL6040_OUTHF_M52dB 0x1D
-#define TWL6040_RAMP_NONE 0
-#define TWL6040_RAMP_UP 1
-#define TWL6040_RAMP_DOWN 2
-
-#define TWL6040_HSL_VOL_MASK 0x0F
-#define TWL6040_HSL_VOL_SHIFT 0
-#define TWL6040_HSR_VOL_MASK 0xF0
-#define TWL6040_HSR_VOL_SHIFT 4
-#define TWL6040_HF_VOL_MASK 0x1F
-#define TWL6040_HF_VOL_SHIFT 0
-
/* Shadow register used by the driver */
#define TWL6040_REG_SW_SHADOW 0x2F
#define TWL6040_CACHEREGNUM (TWL6040_REG_SW_SHADOW + 1)
@@ -64,18 +53,6 @@
/* TWL6040_REG_SW_SHADOW (0x2F) fields */
#define TWL6040_EAR_PATH_ENABLE 0x01
-struct twl6040_output {
- u16 active;
- u16 left_vol;
- u16 right_vol;
- u16 left_step;
- u16 right_step;
- unsigned int step_delay;
- u16 ramp;
- struct delayed_work work;
- struct completion ramp_done;
-};
-
struct twl6040_jack_data {
struct snd_soc_jack *jack;
struct delayed_work work;
@@ -100,8 +77,6 @@ struct twl6040_data {
struct snd_soc_codec *codec;
struct workqueue_struct *workqueue;
struct mutex mutex;
- struct twl6040_output headset;
- struct twl6040_output handsfree;
};
/*
@@ -311,318 +286,6 @@ static void twl6040_restore_regs(struct snd_soc_codec *codec)
}
}
-/*
- * Ramp HS PGA volume to minimise pops at stream startup and shutdown.
- */
-static inline int twl6040_hs_ramp_step(struct snd_soc_codec *codec,
- unsigned int left_step, unsigned int right_step)
-{
-
- struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec);
- struct twl6040_output *headset = &priv->headset;
- int left_complete = 0, right_complete = 0;
- u8 reg, val;
-
- /* left channel */
- left_step = (left_step > 0xF) ? 0xF : left_step;
- reg = twl6040_read_reg_cache(codec, TWL6040_REG_HSGAIN);
- val = (~reg & TWL6040_HSL_VOL_MASK);
-
- if (headset->ramp == TWL6040_RAMP_UP) {
- /* ramp step up */
- if (val < headset->left_vol) {
- if (val + left_step > headset->left_vol)
- val = headset->left_vol;
- else
- val += left_step;
-
- reg &= ~TWL6040_HSL_VOL_MASK;
- twl6040_write(codec, TWL6040_REG_HSGAIN,
- (reg | (~val & TWL6040_HSL_VOL_MASK)));
- } else {
- left_complete = 1;
- }
- } else if (headset->ramp == TWL6040_RAMP_DOWN) {
- /* ramp step down */
- if (val > 0x0) {
- if ((int)val - (int)left_step < 0)
- val = 0;
- else
- val -= left_step;
-
- reg &= ~TWL6040_HSL_VOL_MASK;
- twl6040_write(codec, TWL6040_REG_HSGAIN, reg |
- (~val & TWL6040_HSL_VOL_MASK));
- } else {
- left_complete = 1;
- }
- }
-
- /* right channel */
- right_step = (right_step > 0xF) ? 0xF : right_step;
- reg = twl6040_read_reg_cache(codec, TWL6040_REG_HSGAIN);
- val = (~reg & TWL6040_HSR_VOL_MASK) >> TWL6040_HSR_VOL_SHIFT;
-
- if (headset->ramp == TWL6040_RAMP_UP) {
- /* ramp step up */
- if (val < headset->right_vol) {
- if (val + right_step > headset->right_vol)
- val = headset->right_vol;
- else
- val += right_step;
-
- reg &= ~TWL6040_HSR_VOL_MASK;
- twl6040_write(codec, TWL6040_REG_HSGAIN,
- (reg | (~val << TWL6040_HSR_VOL_SHIFT)));
- } else {
- right_complete = 1;
- }
- } else if (headset->ramp == TWL6040_RAMP_DOWN) {
- /* ramp step down */
- if (val > 0x0) {
- if ((int)val - (int)right_step < 0)
- val = 0;
- else
- val -= right_step;
-
- reg &= ~TWL6040_HSR_VOL_MASK;
- twl6040_write(codec, TWL6040_REG_HSGAIN,
- reg | (~val << TWL6040_HSR_VOL_SHIFT));
- } else {
- right_complete = 1;
- }
- }
-
- return left_complete & right_complete;
-}
-
-/*
- * Ramp HF PGA volume to minimise pops at stream startup and shutdown.
- */
-static inline int twl6040_hf_ramp_step(struct snd_soc_codec *codec,
- unsigned int left_step, unsigned int right_step)
-{
- struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec);
- struct twl6040_output *handsfree = &priv->handsfree;
- int left_complete = 0, right_complete = 0;
- u16 reg, val;
-
- /* left channel */
- left_step = (left_step > 0x1D) ? 0x1D : left_step;
- reg = twl6040_read_reg_cache(codec, TWL6040_REG_HFLGAIN);
- reg = 0x1D - reg;
- val = (reg & TWL6040_HF_VOL_MASK);
- if (handsfree->ramp == TWL6040_RAMP_UP) {
- /* ramp step up */
- if (val < handsfree->left_vol) {
- if (val + left_step > handsfree->left_vol)
- val = handsfree->left_vol;
- else
- val += left_step;
-
- reg &= ~TWL6040_HF_VOL_MASK;
- twl6040_write(codec, TWL6040_REG_HFLGAIN,
- reg | (0x1D - val));
- } else {
- left_complete = 1;
- }
- } else if (handsfree->ramp == TWL6040_RAMP_DOWN) {
- /* ramp step down */
- if (val > 0) {
- if ((int)val - (int)left_step < 0)
- val = 0;
- else
- val -= left_step;
-
- reg &= ~TWL6040_HF_VOL_MASK;
- twl6040_write(codec, TWL6040_REG_HFLGAIN,
- reg | (0x1D - val));
- } else {
- left_complete = 1;
- }
- }
-
- /* right channel */
- right_step = (right_step > 0x1D) ? 0x1D : right_step;
- reg = twl6040_read_reg_cache(codec, TWL6040_REG_HFRGAIN);
- reg = 0x1D - reg;
- val = (reg & TWL6040_HF_VOL_MASK);
- if (handsfree->ramp == TWL6040_RAMP_UP) {
- /* ramp step up */
- if (val < handsfree->right_vol) {
- if (val + right_step > handsfree->right_vol)
- val = handsfree->right_vol;
- else
- val += right_step;
-
- reg &= ~TWL6040_HF_VOL_MASK;
- twl6040_write(codec, TWL6040_REG_HFRGAIN,
- reg | (0x1D - val));
- } else {
- right_complete = 1;
- }
- } else if (handsfree->ramp == TWL6040_RAMP_DOWN) {
- /* ramp step down */
- if (val > 0) {
- if ((int)val - (int)right_step < 0)
- val = 0;
- else
- val -= right_step;
-
- reg &= ~TWL6040_HF_VOL_MASK;
- twl6040_write(codec, TWL6040_REG_HFRGAIN,
- reg | (0x1D - val));
- }
- }
-
- return left_complete & right_complete;
-}
-
-/*
- * This work ramps both output PGAs at stream start/stop time to
- * minimise pop associated with DAPM power switching.
- */
-static void twl6040_pga_hs_work(struct work_struct *work)
-{
- struct twl6040_data *priv =
- container_of(work, struct twl6040_data, headset.work.work);
- struct snd_soc_codec *codec = priv->codec;
- struct twl6040_output *headset = &priv->headset;
- int i, headset_complete;
-
- /* do we need to ramp at all ? */
- if (headset->ramp == TWL6040_RAMP_NONE)
- return;
-
- /* HS PGA gain range: 0x0 - 0xf (0 - 15) */
- for (i = 0; i < 16; i++) {
- headset_complete = twl6040_hs_ramp_step(codec,
- headset->left_step,
- headset->right_step);
-
- /* ramp finished ? */
- if (headset_complete)
- break;
-
- schedule_timeout_interruptible(
- msecs_to_jiffies(headset->step_delay));
- }
-
- if (headset->ramp == TWL6040_RAMP_DOWN) {
- headset->active = 0;
- complete(&headset->ramp_done);
- } else {
- headset->active = 1;
- }
- headset->ramp = TWL6040_RAMP_NONE;
-}
-
-static void twl6040_pga_hf_work(struct work_struct *work)
-{
- struct twl6040_data *priv =
- container_of(work, struct twl6040_data, handsfree.work.work);
- struct snd_soc_codec *codec = priv->codec;
- struct twl6040_output *handsfree = &priv->handsfree;
- int i, handsfree_complete;
-
- /* do we need to ramp at all ? */
- if (handsfree->ramp == TWL6040_RAMP_NONE)
- return;
-
- /*
- * HF PGA gain range: 0x00 - 0x1d (0 - 29) */
- for (i = 0; i < 30; i++) {
- handsfree_complete = twl6040_hf_ramp_step(codec,
- handsfree->left_step,
- handsfree->right_step);
-
- /* ramp finished ? */
- if (handsfree_complete)
- break;
-
- schedule_timeout_interruptible(
- msecs_to_jiffies(handsfree->step_delay));
- }
-
-
- if (handsfree->ramp == TWL6040_RAMP_DOWN) {
- handsfree->active = 0;
- complete(&handsfree->ramp_done);
- } else
- handsfree->active = 1;
- handsfree->ramp = TWL6040_RAMP_NONE;
-}
-
-static int out_drv_event(struct snd_soc_dapm_widget *w,
- struct snd_kcontrol *kcontrol, int event)
-{
- struct snd_soc_codec *codec = w->codec;
- struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec);
- struct twl6040_output *out;
- struct delayed_work *work;
-
- switch (w->shift) {
- case 2: /* Headset output driver */
- out = &priv->headset;
- work = &out->work;
- /*
- * Make sure, that we do not mess up variables for already
- * executing work.
- */
- cancel_delayed_work_sync(work);
-
- out->left_step = priv->hs_left_step;
- out->right_step = priv->hs_right_step;
- out->step_delay = 5; /* 5 ms between volume ramp steps */
- break;
- case 4: /* Handsfree output driver */
- out = &priv->handsfree;
- work = &out->work;
- /*
- * Make sure, that we do not mess up variables for already
- * executing work.
- */
- cancel_delayed_work_sync(work);
-
- out->left_step = priv->hf_left_step;
- out->right_step = priv->hf_right_step;
- out->step_delay = 5; /* 5 ms between volume ramp steps */
- break;
- default:
- return -1;
- }
-
- switch (event) {
- case SND_SOC_DAPM_POST_PMU:
- if (out->active)
- break;
-
- /* don't use volume ramp for power-up */
- out->ramp = TWL6040_RAMP_UP;
- out->left_step = out->left_vol;
- out->right_step = out->right_vol;
-
- queue_delayed_work(priv->workqueue, work, msecs_to_jiffies(1));
- break;
-
- case SND_SOC_DAPM_PRE_PMD:
- if (!out->active)
- break;
-
- /* use volume ramp for power-down */
- out->ramp = TWL6040_RAMP_DOWN;
- INIT_COMPLETION(out->ramp_done);
-
- queue_delayed_work(priv->workqueue, work, msecs_to_jiffies(1));
-
- wait_for_completion_timeout(&out->ramp_done,
- msecs_to_jiffies(2000));
- break;
- }
-
- return 0;
-}
-
/* set headset dac and driver power mode */
static int headset_power_mode(struct snd_soc_codec *codec, int high_perf)
{
@@ -747,71 +410,6 @@ static irqreturn_t twl6040_audio_handler(int irq, void *data)
return IRQ_HANDLED;
}
-static int twl6040_put_volsw(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
- struct twl6040_data *twl6040_priv = snd_soc_codec_get_drvdata(codec);
- struct twl6040_output *out = NULL;
- struct soc_mixer_control *mc =
- (struct soc_mixer_control *)kcontrol->private_value;
- int ret;
-
- /* For HS and HF we shadow the values and only actually write
- * them out when active in order to ensure the amplifier comes on
- * as quietly as possible. */
- switch (mc->reg) {
- case TWL6040_REG_HSGAIN:
- out = &twl6040_priv->headset;
- break;
- case TWL6040_REG_HFLGAIN:
- out = &twl6040_priv->handsfree;
- break;
- default:
- dev_warn(codec->dev, "%s: Unexpected register: 0x%02x\n",
- __func__, mc->reg);
- return -EINVAL;
- }
-
- out->left_vol = ucontrol->value.integer.value[0];
- out->right_vol = ucontrol->value.integer.value[1];
- if (!out->active)
- return 1;
-
- ret = snd_soc_put_volsw(kcontrol, ucontrol);
- if (ret < 0)
- return ret;
-
- return 1;
-}
-
-static int twl6040_get_volsw(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
- struct twl6040_data *twl6040_priv = snd_soc_codec_get_drvdata(codec);
- struct twl6040_output *out = &twl6040_priv->headset;
- struct soc_mixer_control *mc =
- (struct soc_mixer_control *)kcontrol->private_value;
-
- switch (mc->reg) {
- case TWL6040_REG_HSGAIN:
- out = &twl6040_priv->headset;
- break;
- case TWL6040_REG_HFLGAIN:
- out = &twl6040_priv->handsfree;
- break;
- default:
- dev_warn(codec->dev, "%s: Unexpected register: 0x%02x\n",
- __func__, mc->reg);
- return -EINVAL;
- }
-
- ucontrol->value.integer.value[0] = out->left_vol;
- ucontrol->value.integer.value[1] = out->right_vol;
- return 0;
-}
-
static int twl6040_soc_dapm_put_vibra_enum(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
@@ -1076,12 +674,10 @@ static const struct snd_kcontrol_new twl6040_snd_controls[] = {
TWL6040_REG_LINEGAIN, 0, 3, 7, 0, afm_amp_tlv),
/* Playback gains */
- SOC_DOUBLE_EXT_TLV("Headset Playback Volume",
- TWL6040_REG_HSGAIN, 0, 4, 0xF, 1, twl6040_get_volsw,
- twl6040_put_volsw, hs_tlv),
- SOC_DOUBLE_R_EXT_TLV("Handsfree Playback Volume",
- TWL6040_REG_HFLGAIN, TWL6040_REG_HFRGAIN, 0, 0x1D, 1,
- twl6040_get_volsw, twl6040_put_volsw, hf_tlv),
+ SOC_DOUBLE_TLV("Headset Playback Volume",
+ TWL6040_REG_HSGAIN, 0, 4, 0xF, 1, hs_tlv),
+ SOC_DOUBLE_R_TLV("Handsfree Playback Volume",
+ TWL6040_REG_HFLGAIN, TWL6040_REG_HFRGAIN, 0, 0x1D, 1, hf_tlv),
SOC_SINGLE_TLV("Earphone Playback Volume",
TWL6040_REG_EARCTL, 1, 0xF, 1, ep_tlv),
@@ -1180,22 +776,14 @@ static const struct snd_soc_dapm_widget twl6040_dapm_widgets[] = {
&auxr_switch_control),
/* Analog playback drivers */
- SND_SOC_DAPM_OUT_DRV_E("HF Left Driver",
- TWL6040_REG_HFLCTL, 4, 0, NULL, 0,
- out_drv_event,
- SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
- SND_SOC_DAPM_OUT_DRV_E("HF Right Driver",
- TWL6040_REG_HFRCTL, 4, 0, NULL, 0,
- out_drv_event,
- SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
- SND_SOC_DAPM_OUT_DRV_E("HS Left Driver",
- TWL6040_REG_HSLCTL, 2, 0, NULL, 0,
- out_drv_event,
- SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
- SND_SOC_DAPM_OUT_DRV_E("HS Right Driver",
- TWL6040_REG_HSRCTL, 2, 0, NULL, 0,
- out_drv_event,
- SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
+ SND_SOC_DAPM_OUT_DRV("HF Left Driver",
+ TWL6040_REG_HFLCTL, 4, 0, NULL, 0),
+ SND_SOC_DAPM_OUT_DRV("HF Right Driver",
+ TWL6040_REG_HFRCTL, 4, 0, NULL, 0),
+ SND_SOC_DAPM_OUT_DRV("HS Left Driver",
+ TWL6040_REG_HSLCTL, 2, 0, NULL, 0),
+ SND_SOC_DAPM_OUT_DRV("HS Right Driver",
+ TWL6040_REG_HSRCTL, 2, 0, NULL, 0),
SND_SOC_DAPM_OUT_DRV_E("Earphone Driver",
TWL6040_REG_EARCTL, 0, 0, NULL, 0,
twl6040_ep_drv_event,
@@ -1339,8 +927,7 @@ static int twl6040_set_bias_level(struct snd_soc_codec *codec,
static int twl6040_startup(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_codec *codec = dai->codec;
struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec);
snd_pcm_hw_constraint_list(substream->runtime, 0,
@@ -1354,8 +941,7 @@ static int twl6040_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;
+ struct snd_soc_codec *codec = dai->codec;
struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec);
int rate;
@@ -1391,8 +977,7 @@ static int twl6040_hw_params(struct snd_pcm_substream *substream,
static int twl6040_prepare(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_codec *codec = dai->codec;
struct twl6040 *twl6040 = codec->control_data;
struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec);
int ret;
@@ -1570,14 +1155,9 @@ static int twl6040_probe(struct snd_soc_codec *codec)
}
INIT_DELAYED_WORK(&priv->hs_jack.work, twl6040_accessory_work);
- INIT_DELAYED_WORK(&priv->headset.work, twl6040_pga_hs_work);
- INIT_DELAYED_WORK(&priv->handsfree.work, twl6040_pga_hf_work);
mutex_init(&priv->mutex);
- init_completion(&priv->headset.ramp_done);
- init_completion(&priv->handsfree.ramp_done);
-
ret = request_threaded_irq(priv->plug_irq, NULL, twl6040_audio_handler,
0, "twl6040_irq_plug", codec);
if (ret) {
diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c
index 797b0dde2c6..6c3d43b8ee8 100644
--- a/sound/soc/codecs/uda134x.c
+++ b/sound/soc/codecs/uda134x.c
@@ -159,8 +159,7 @@ static int uda134x_mute(struct snd_soc_dai *dai, int mute)
static int uda134x_startup(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_codec *codec =rtd->codec;
+ struct snd_soc_codec *codec = dai->codec;
struct uda134x_priv *uda134x = snd_soc_codec_get_drvdata(codec);
struct snd_pcm_runtime *master_runtime;
@@ -191,8 +190,7 @@ static int uda134x_startup(struct snd_pcm_substream *substream,
static void uda134x_shutdown(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_codec *codec = dai->codec;
struct uda134x_priv *uda134x = snd_soc_codec_get_drvdata(codec);
if (uda134x->master_substream == substream)
diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c
index 4f1b23d7e40..2502214b84a 100644
--- a/sound/soc/codecs/uda1380.c
+++ b/sound/soc/codecs/uda1380.c
@@ -502,8 +502,7 @@ static int uda1380_set_dai_fmt_capture(struct snd_soc_dai *codec_dai,
static int uda1380_trigger(struct snd_pcm_substream *substream, int cmd,
struct snd_soc_dai *dai)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_codec *codec = dai->codec;
struct uda1380_priv *uda1380 = snd_soc_codec_get_drvdata(codec);
int mixer = uda1380_read_reg_cache(codec, UDA1380_MIXER);
@@ -528,8 +527,7 @@ static int uda1380_pcm_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;
+ struct snd_soc_codec *codec = dai->codec;
u16 clk = uda1380_read_reg_cache(codec, UDA1380_CLK);
/* set WSPLL power and divider if running from this clock */
diff --git a/sound/soc/codecs/wl1273.c b/sound/soc/codecs/wl1273.c
index 3d868dc4009..7b24d6d192e 100644
--- a/sound/soc/codecs/wl1273.c
+++ b/sound/soc/codecs/wl1273.c
@@ -293,8 +293,7 @@ static const struct snd_kcontrol_new wl1273_controls[] = {
static int wl1273_startup(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_codec *codec = dai->codec;
struct wl1273_priv *wl1273 = snd_soc_codec_get_drvdata(codec);
switch (wl1273->mode) {
@@ -329,8 +328,7 @@ static int wl1273_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 wl1273_priv *wl1273 = snd_soc_codec_get_drvdata(rtd->codec);
+ struct wl1273_priv *wl1273 = snd_soc_codec_get_drvdata(dai->codec);
struct wl1273_core *core = wl1273->core;
unsigned int rate, width, r;
diff --git a/sound/soc/codecs/wm1250-ev1.c b/sound/soc/codecs/wm1250-ev1.c
index aefb4f89be0..e0b51e9f8b1 100644
--- a/sound/soc/codecs/wm1250-ev1.c
+++ b/sound/soc/codecs/wm1250-ev1.c
@@ -79,22 +79,65 @@ static const struct snd_soc_dapm_route wm1250_ev1_dapm_routes[] = {
{ "WM1250 Output", NULL, "DAC" },
};
+static int wm1250_ev1_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct wm1250_priv *wm1250 = snd_soc_codec_get_drvdata(dai->codec);
+
+ switch (params_rate(params)) {
+ case 8000:
+ gpio_set_value(wm1250->gpios[WM1250_EV1_GPIO_CLK_SEL0].gpio,
+ 1);
+ gpio_set_value(wm1250->gpios[WM1250_EV1_GPIO_CLK_SEL1].gpio,
+ 1);
+ break;
+ case 16000:
+ gpio_set_value(wm1250->gpios[WM1250_EV1_GPIO_CLK_SEL0].gpio,
+ 0);
+ gpio_set_value(wm1250->gpios[WM1250_EV1_GPIO_CLK_SEL1].gpio,
+ 1);
+ break;
+ case 32000:
+ gpio_set_value(wm1250->gpios[WM1250_EV1_GPIO_CLK_SEL0].gpio,
+ 1);
+ gpio_set_value(wm1250->gpios[WM1250_EV1_GPIO_CLK_SEL1].gpio,
+ 0);
+ break;
+ case 64000:
+ gpio_set_value(wm1250->gpios[WM1250_EV1_GPIO_CLK_SEL0].gpio,
+ 0);
+ gpio_set_value(wm1250->gpios[WM1250_EV1_GPIO_CLK_SEL1].gpio,
+ 0);
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static const struct snd_soc_dai_ops wm1250_ev1_ops = {
+ .hw_params = wm1250_ev1_hw_params,
+};
+
static struct snd_soc_dai_driver wm1250_ev1_dai = {
.name = "wm1250-ev1",
.playback = {
.stream_name = "Playback",
.channels_min = 1,
- .channels_max = 1,
+ .channels_max = 2,
.rates = SNDRV_PCM_RATE_8000,
.formats = SNDRV_PCM_FMTBIT_S16_LE,
},
.capture = {
.stream_name = "Capture",
.channels_min = 1,
- .channels_max = 1,
+ .channels_max = 2,
.rates = SNDRV_PCM_RATE_8000,
.formats = SNDRV_PCM_FMTBIT_S16_LE,
},
+ .ops = &wm1250_ev1_ops,
};
static struct snd_soc_codec_driver soc_codec_dev_wm1250_ev1 = {
@@ -215,23 +258,7 @@ static struct i2c_driver wm1250_ev1_i2c_driver = {
.id_table = wm1250_ev1_i2c_id,
};
-static int __init wm1250_ev1_modinit(void)
-{
- int ret = 0;
-
- ret = i2c_add_driver(&wm1250_ev1_i2c_driver);
- if (ret != 0)
- pr_err("Failed to register WM1250-EV1 I2C driver: %d\n", ret);
-
- return ret;
-}
-module_init(wm1250_ev1_modinit);
-
-static void __exit wm1250_ev1_exit(void)
-{
- i2c_del_driver(&wm1250_ev1_i2c_driver);
-}
-module_exit(wm1250_ev1_exit);
+module_i2c_driver(wm1250_ev1_i2c_driver);
MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>");
MODULE_DESCRIPTION("WM1250-EV1 audio I/O module driver");
diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c
index a75c3766aed..0418fa11e6b 100644
--- a/sound/soc/codecs/wm2000.c
+++ b/sound/soc/codecs/wm2000.c
@@ -99,8 +99,9 @@ static void wm2000_reset(struct wm2000_priv *wm2000)
}
static int wm2000_poll_bit(struct i2c_client *i2c,
- unsigned int reg, u8 mask, int timeout)
+ unsigned int reg, u8 mask)
{
+ int timeout = 4000;
int val;
val = wm2000_read(i2c, reg);
@@ -119,7 +120,7 @@ static int wm2000_poll_bit(struct i2c_client *i2c,
static int wm2000_power_up(struct i2c_client *i2c, int analogue)
{
struct wm2000_priv *wm2000 = dev_get_drvdata(&i2c->dev);
- int ret, timeout;
+ int ret;
BUG_ON(wm2000->anc_mode != ANC_OFF);
@@ -140,13 +141,13 @@ static int wm2000_power_up(struct i2c_client *i2c, int analogue)
/* Wait for ANC engine to become ready */
if (!wm2000_poll_bit(i2c, WM2000_REG_ANC_STAT,
- WM2000_ANC_ENG_IDLE, 1)) {
+ WM2000_ANC_ENG_IDLE)) {
dev_err(&i2c->dev, "ANC engine failed to reset\n");
return -ETIMEDOUT;
}
if (!wm2000_poll_bit(i2c, WM2000_REG_SYS_STATUS,
- WM2000_STATUS_BOOT_COMPLETE, 1)) {
+ WM2000_STATUS_BOOT_COMPLETE)) {
dev_err(&i2c->dev, "ANC engine failed to initialise\n");
return -ETIMEDOUT;
}
@@ -173,16 +174,13 @@ static int wm2000_power_up(struct i2c_client *i2c, int analogue)
dev_dbg(&i2c->dev, "Download complete\n");
if (analogue) {
- timeout = 248;
- wm2000_write(i2c, WM2000_REG_ANA_VMID_PU_TIME, timeout / 4);
+ wm2000_write(i2c, WM2000_REG_ANA_VMID_PU_TIME, 248 / 4);
wm2000_write(i2c, WM2000_REG_SYS_MODE_CNTRL,
WM2000_MODE_ANA_SEQ_INCLUDE |
WM2000_MODE_MOUSE_ENABLE |
WM2000_MODE_THERMAL_ENABLE);
} else {
- timeout = 10;
-
wm2000_write(i2c, WM2000_REG_SYS_MODE_CNTRL,
WM2000_MODE_MOUSE_ENABLE |
WM2000_MODE_THERMAL_ENABLE);
@@ -201,9 +199,8 @@ static int wm2000_power_up(struct i2c_client *i2c, int analogue)
wm2000_write(i2c, WM2000_REG_SYS_CTL2, WM2000_ANC_INT_N_CLR);
if (!wm2000_poll_bit(i2c, WM2000_REG_SYS_STATUS,
- WM2000_STATUS_MOUSE_ACTIVE, timeout)) {
- dev_err(&i2c->dev, "Timed out waiting for device after %dms\n",
- timeout * 10);
+ WM2000_STATUS_MOUSE_ACTIVE)) {
+ dev_err(&i2c->dev, "Timed out waiting for device\n");
return -ETIMEDOUT;
}
@@ -218,28 +215,25 @@ static int wm2000_power_up(struct i2c_client *i2c, int analogue)
static int wm2000_power_down(struct i2c_client *i2c, int analogue)
{
struct wm2000_priv *wm2000 = dev_get_drvdata(&i2c->dev);
- int timeout;
if (analogue) {
- timeout = 248;
- wm2000_write(i2c, WM2000_REG_ANA_VMID_PD_TIME, timeout / 4);
+ wm2000_write(i2c, WM2000_REG_ANA_VMID_PD_TIME, 248 / 4);
wm2000_write(i2c, WM2000_REG_SYS_MODE_CNTRL,
WM2000_MODE_ANA_SEQ_INCLUDE |
WM2000_MODE_POWER_DOWN);
} else {
- timeout = 10;
wm2000_write(i2c, WM2000_REG_SYS_MODE_CNTRL,
WM2000_MODE_POWER_DOWN);
}
if (!wm2000_poll_bit(i2c, WM2000_REG_SYS_STATUS,
- WM2000_STATUS_POWER_DOWN_COMPLETE, timeout)) {
+ WM2000_STATUS_POWER_DOWN_COMPLETE)) {
dev_err(&i2c->dev, "Timeout waiting for ANC power down\n");
return -ETIMEDOUT;
}
if (!wm2000_poll_bit(i2c, WM2000_REG_ANC_STAT,
- WM2000_ANC_ENG_IDLE, 1)) {
+ WM2000_ANC_ENG_IDLE)) {
dev_err(&i2c->dev, "Timeout waiting for ANC engine idle\n");
return -ETIMEDOUT;
}
@@ -268,13 +262,13 @@ static int wm2000_enter_bypass(struct i2c_client *i2c, int analogue)
}
if (!wm2000_poll_bit(i2c, WM2000_REG_SYS_STATUS,
- WM2000_STATUS_ANC_DISABLED, 10)) {
+ WM2000_STATUS_ANC_DISABLED)) {
dev_err(&i2c->dev, "Timeout waiting for ANC disable\n");
return -ETIMEDOUT;
}
if (!wm2000_poll_bit(i2c, WM2000_REG_ANC_STAT,
- WM2000_ANC_ENG_IDLE, 1)) {
+ WM2000_ANC_ENG_IDLE)) {
dev_err(&i2c->dev, "Timeout waiting for ANC engine idle\n");
return -ETIMEDOUT;
}
@@ -311,7 +305,7 @@ static int wm2000_exit_bypass(struct i2c_client *i2c, int analogue)
wm2000_write(i2c, WM2000_REG_SYS_CTL2, WM2000_ANC_INT_N_CLR);
if (!wm2000_poll_bit(i2c, WM2000_REG_SYS_STATUS,
- WM2000_STATUS_MOUSE_ACTIVE, 10)) {
+ WM2000_STATUS_MOUSE_ACTIVE)) {
dev_err(&i2c->dev, "Timed out waiting for MOUSE\n");
return -ETIMEDOUT;
}
@@ -325,38 +319,32 @@ static int wm2000_exit_bypass(struct i2c_client *i2c, int analogue)
static int wm2000_enter_standby(struct i2c_client *i2c, int analogue)
{
struct wm2000_priv *wm2000 = dev_get_drvdata(&i2c->dev);
- int timeout;
BUG_ON(wm2000->anc_mode != ANC_ACTIVE);
if (analogue) {
- timeout = 248;
- wm2000_write(i2c, WM2000_REG_ANA_VMID_PD_TIME, timeout / 4);
+ wm2000_write(i2c, WM2000_REG_ANA_VMID_PD_TIME, 248 / 4);
wm2000_write(i2c, WM2000_REG_SYS_MODE_CNTRL,
WM2000_MODE_ANA_SEQ_INCLUDE |
WM2000_MODE_THERMAL_ENABLE |
WM2000_MODE_STANDBY_ENTRY);
} else {
- timeout = 10;
-
wm2000_write(i2c, WM2000_REG_SYS_MODE_CNTRL,
WM2000_MODE_THERMAL_ENABLE |
WM2000_MODE_STANDBY_ENTRY);
}
if (!wm2000_poll_bit(i2c, WM2000_REG_SYS_STATUS,
- WM2000_STATUS_ANC_DISABLED, timeout)) {
+ WM2000_STATUS_ANC_DISABLED)) {
dev_err(&i2c->dev,
"Timed out waiting for ANC disable after 1ms\n");
return -ETIMEDOUT;
}
- if (!wm2000_poll_bit(i2c, WM2000_REG_ANC_STAT, WM2000_ANC_ENG_IDLE,
- 1)) {
+ if (!wm2000_poll_bit(i2c, WM2000_REG_ANC_STAT, WM2000_ANC_ENG_IDLE)) {
dev_err(&i2c->dev,
- "Timed out waiting for standby after %dms\n",
- timeout * 10);
+ "Timed out waiting for standby\n");
return -ETIMEDOUT;
}
@@ -374,23 +362,19 @@ static int wm2000_enter_standby(struct i2c_client *i2c, int analogue)
static int wm2000_exit_standby(struct i2c_client *i2c, int analogue)
{
struct wm2000_priv *wm2000 = dev_get_drvdata(&i2c->dev);
- int timeout;
BUG_ON(wm2000->anc_mode != ANC_STANDBY);
wm2000_write(i2c, WM2000_REG_SYS_CTL1, 0);
if (analogue) {
- timeout = 248;
- wm2000_write(i2c, WM2000_REG_ANA_VMID_PU_TIME, timeout / 4);
+ wm2000_write(i2c, WM2000_REG_ANA_VMID_PU_TIME, 248 / 4);
wm2000_write(i2c, WM2000_REG_SYS_MODE_CNTRL,
WM2000_MODE_ANA_SEQ_INCLUDE |
WM2000_MODE_THERMAL_ENABLE |
WM2000_MODE_MOUSE_ENABLE);
} else {
- timeout = 10;
-
wm2000_write(i2c, WM2000_REG_SYS_MODE_CNTRL,
WM2000_MODE_THERMAL_ENABLE |
WM2000_MODE_MOUSE_ENABLE);
@@ -400,9 +384,8 @@ static int wm2000_exit_standby(struct i2c_client *i2c, int analogue)
wm2000_write(i2c, WM2000_REG_SYS_CTL2, WM2000_ANC_INT_N_CLR);
if (!wm2000_poll_bit(i2c, WM2000_REG_SYS_STATUS,
- WM2000_STATUS_MOUSE_ACTIVE, timeout)) {
- dev_err(&i2c->dev, "Timed out waiting for MOUSE after %dms\n",
- timeout * 10);
+ WM2000_STATUS_MOUSE_ACTIVE)) {
+ dev_err(&i2c->dev, "Timed out waiting for MOUSE\n");
return -ETIMEDOUT;
}
diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c
index acbdc5fde92..32682c1b7cd 100644
--- a/sound/soc/codecs/wm2200.c
+++ b/sound/soc/codecs/wm2200.c
@@ -1491,6 +1491,7 @@ static int wm2200_bclk_rates_dat[WM2200_NUM_BCLK_RATES] = {
static int wm2200_bclk_rates_cd[WM2200_NUM_BCLK_RATES] = {
5644800,
+ 3763200,
2882400,
1881600,
1411200,
diff --git a/sound/soc/codecs/wm5100-tables.c b/sound/soc/codecs/wm5100-tables.c
index 9a18fae6820..e167207a19c 100644
--- a/sound/soc/codecs/wm5100-tables.c
+++ b/sound/soc/codecs/wm5100-tables.c
@@ -32,7 +32,18 @@ bool wm5100_volatile_register(struct device *dev, unsigned int reg)
case WM5100_MIC_DETECT_3:
return 1;
default:
- return 0;
+ if ((reg >= WM5100_DSP1_PM_0 && reg <= WM5100_DSP1_PM_1535) ||
+ (reg >= WM5100_DSP1_ZM_0 && reg <= WM5100_DSP1_ZM_2047) ||
+ (reg >= WM5100_DSP1_DM_0 && reg <= WM5100_DSP1_DM_511) ||
+ (reg >= WM5100_DSP2_PM_0 && reg <= WM5100_DSP2_PM_1535) ||
+ (reg >= WM5100_DSP2_ZM_0 && reg <= WM5100_DSP2_ZM_2047) ||
+ (reg >= WM5100_DSP2_DM_0 && reg <= WM5100_DSP2_DM_511) ||
+ (reg >= WM5100_DSP3_PM_0 && reg <= WM5100_DSP3_PM_1535) ||
+ (reg >= WM5100_DSP3_ZM_0 && reg <= WM5100_DSP3_ZM_2047) ||
+ (reg >= WM5100_DSP3_DM_0 && reg <= WM5100_DSP3_DM_511))
+ return 1;
+ else
+ return 0;
}
}
@@ -697,9 +708,110 @@ bool wm5100_readable_register(struct device *dev, unsigned int reg)
case WM5100_HPLPF3_2:
case WM5100_HPLPF4_1:
case WM5100_HPLPF4_2:
+ case WM5100_DSP1_CONTROL_1:
+ case WM5100_DSP1_CONTROL_2:
+ case WM5100_DSP1_CONTROL_3:
+ case WM5100_DSP1_CONTROL_4:
+ case WM5100_DSP1_CONTROL_5:
+ case WM5100_DSP1_CONTROL_6:
+ case WM5100_DSP1_CONTROL_7:
+ case WM5100_DSP1_CONTROL_8:
+ case WM5100_DSP1_CONTROL_9:
+ case WM5100_DSP1_CONTROL_10:
+ case WM5100_DSP1_CONTROL_11:
+ case WM5100_DSP1_CONTROL_12:
+ case WM5100_DSP1_CONTROL_13:
+ case WM5100_DSP1_CONTROL_14:
+ case WM5100_DSP1_CONTROL_15:
+ case WM5100_DSP1_CONTROL_16:
+ case WM5100_DSP1_CONTROL_17:
+ case WM5100_DSP1_CONTROL_18:
+ case WM5100_DSP1_CONTROL_19:
+ case WM5100_DSP1_CONTROL_20:
+ case WM5100_DSP1_CONTROL_21:
+ case WM5100_DSP1_CONTROL_22:
+ case WM5100_DSP1_CONTROL_23:
+ case WM5100_DSP1_CONTROL_24:
+ case WM5100_DSP1_CONTROL_25:
+ case WM5100_DSP1_CONTROL_26:
+ case WM5100_DSP1_CONTROL_27:
+ case WM5100_DSP1_CONTROL_28:
+ case WM5100_DSP1_CONTROL_29:
+ case WM5100_DSP1_CONTROL_30:
+ case WM5100_DSP2_CONTROL_1:
+ case WM5100_DSP2_CONTROL_2:
+ case WM5100_DSP2_CONTROL_3:
+ case WM5100_DSP2_CONTROL_4:
+ case WM5100_DSP2_CONTROL_5:
+ case WM5100_DSP2_CONTROL_6:
+ case WM5100_DSP2_CONTROL_7:
+ case WM5100_DSP2_CONTROL_8:
+ case WM5100_DSP2_CONTROL_9:
+ case WM5100_DSP2_CONTROL_10:
+ case WM5100_DSP2_CONTROL_11:
+ case WM5100_DSP2_CONTROL_12:
+ case WM5100_DSP2_CONTROL_13:
+ case WM5100_DSP2_CONTROL_14:
+ case WM5100_DSP2_CONTROL_15:
+ case WM5100_DSP2_CONTROL_16:
+ case WM5100_DSP2_CONTROL_17:
+ case WM5100_DSP2_CONTROL_18:
+ case WM5100_DSP2_CONTROL_19:
+ case WM5100_DSP2_CONTROL_20:
+ case WM5100_DSP2_CONTROL_21:
+ case WM5100_DSP2_CONTROL_22:
+ case WM5100_DSP2_CONTROL_23:
+ case WM5100_DSP2_CONTROL_24:
+ case WM5100_DSP2_CONTROL_25:
+ case WM5100_DSP2_CONTROL_26:
+ case WM5100_DSP2_CONTROL_27:
+ case WM5100_DSP2_CONTROL_28:
+ case WM5100_DSP2_CONTROL_29:
+ case WM5100_DSP2_CONTROL_30:
+ case WM5100_DSP3_CONTROL_1:
+ case WM5100_DSP3_CONTROL_2:
+ case WM5100_DSP3_CONTROL_3:
+ case WM5100_DSP3_CONTROL_4:
+ case WM5100_DSP3_CONTROL_5:
+ case WM5100_DSP3_CONTROL_6:
+ case WM5100_DSP3_CONTROL_7:
+ case WM5100_DSP3_CONTROL_8:
+ case WM5100_DSP3_CONTROL_9:
+ case WM5100_DSP3_CONTROL_10:
+ case WM5100_DSP3_CONTROL_11:
+ case WM5100_DSP3_CONTROL_12:
+ case WM5100_DSP3_CONTROL_13:
+ case WM5100_DSP3_CONTROL_14:
+ case WM5100_DSP3_CONTROL_15:
+ case WM5100_DSP3_CONTROL_16:
+ case WM5100_DSP3_CONTROL_17:
+ case WM5100_DSP3_CONTROL_18:
+ case WM5100_DSP3_CONTROL_19:
+ case WM5100_DSP3_CONTROL_20:
+ case WM5100_DSP3_CONTROL_21:
+ case WM5100_DSP3_CONTROL_22:
+ case WM5100_DSP3_CONTROL_23:
+ case WM5100_DSP3_CONTROL_24:
+ case WM5100_DSP3_CONTROL_25:
+ case WM5100_DSP3_CONTROL_26:
+ case WM5100_DSP3_CONTROL_27:
+ case WM5100_DSP3_CONTROL_28:
+ case WM5100_DSP3_CONTROL_29:
+ case WM5100_DSP3_CONTROL_30:
return 1;
default:
- return 0;
+ if ((reg >= WM5100_DSP1_PM_0 && reg <= WM5100_DSP1_PM_1535) ||
+ (reg >= WM5100_DSP1_ZM_0 && reg <= WM5100_DSP1_ZM_2047) ||
+ (reg >= WM5100_DSP1_DM_0 && reg <= WM5100_DSP1_DM_511) ||
+ (reg >= WM5100_DSP2_PM_0 && reg <= WM5100_DSP2_PM_1535) ||
+ (reg >= WM5100_DSP2_ZM_0 && reg <= WM5100_DSP2_ZM_2047) ||
+ (reg >= WM5100_DSP2_DM_0 && reg <= WM5100_DSP2_DM_511) ||
+ (reg >= WM5100_DSP3_PM_0 && reg <= WM5100_DSP3_PM_1535) ||
+ (reg >= WM5100_DSP3_ZM_0 && reg <= WM5100_DSP3_ZM_2047) ||
+ (reg >= WM5100_DSP3_DM_0 && reg <= WM5100_DSP3_DM_511))
+ return 1;
+ else
+ return 0;
}
}
@@ -1361,4 +1473,13 @@ struct reg_default wm5100_reg_defaults[WM5100_REGISTER_COUNT] = {
{ 0x0EC9, 0x0000 }, /* R3785 - HPLPF3_2 */
{ 0x0ECC, 0x0000 }, /* R3788 - HPLPF4_1 */
{ 0x0ECD, 0x0000 }, /* R3789 - HPLPF4_2 */
+ { 0x0F02, 0x0000 }, /* R3842 - DSP1 Control 2 */
+ { 0x0F03, 0x0000 }, /* R3843 - DSP1 Control 3 */
+ { 0x0F04, 0x0000 }, /* R3844 - DSP1 Control 4 */
+ { 0x1002, 0x0000 }, /* R4098 - DSP2 Control 2 */
+ { 0x1003, 0x0000 }, /* R4099 - DSP2 Control 3 */
+ { 0x1004, 0x0000 }, /* R4100 - DSP2 Control 4 */
+ { 0x1102, 0x0000 }, /* R4354 - DSP3 Control 2 */
+ { 0x1103, 0x0000 }, /* R4355 - DSP3 Control 3 */
+ { 0x1104, 0x0000 }, /* R4356 - DSP3 Control 4 */
};
diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c
index b9c185ce64e..cb6d5372103 100644
--- a/sound/soc/codecs/wm5100.c
+++ b/sound/soc/codecs/wm5100.c
@@ -1265,29 +1265,12 @@ static const __devinitdata struct reg_default wm5100_reva_patches[] = {
{ WM5100_AUDIO_IF_3_19, 1 },
};
-static int wm5100_dai_to_base(struct snd_soc_dai *dai)
-{
- switch (dai->id) {
- case 0:
- return WM5100_AUDIO_IF_1_1 - 1;
- case 1:
- return WM5100_AUDIO_IF_2_1 - 1;
- case 2:
- return WM5100_AUDIO_IF_3_1 - 1;
- default:
- BUG();
- return -EINVAL;
- }
-}
-
static int wm5100_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
{
struct snd_soc_codec *codec = dai->codec;
int lrclk, bclk, mask, base;
- base = wm5100_dai_to_base(dai);
- if (base < 0)
- return base;
+ base = dai->driver->base;
lrclk = 0;
bclk = 0;
@@ -1414,9 +1397,7 @@ static int wm5100_hw_params(struct snd_pcm_substream *substream,
int i, base, bclk, aif_rate, lrclk, wl, fl, sr;
int *bclk_rates;
- base = wm5100_dai_to_base(dai);
- if (base < 0)
- return base;
+ base = dai->driver->base;
/* Data sizes if not using TDM */
wl = snd_pcm_format_width(params_format(params));
@@ -1897,6 +1878,7 @@ static int wm5100_set_fll(struct snd_soc_codec *codec, int fll_id, int source,
static struct snd_soc_dai_driver wm5100_dai[] = {
{
.name = "wm5100-aif1",
+ .base = WM5100_AUDIO_IF_1_1 - 1,
.playback = {
.stream_name = "AIF1 Playback",
.channels_min = 2,
@@ -1916,6 +1898,7 @@ static struct snd_soc_dai_driver wm5100_dai[] = {
{
.name = "wm5100-aif2",
.id = 1,
+ .base = WM5100_AUDIO_IF_2_1 - 1,
.playback = {
.stream_name = "AIF2 Playback",
.channels_min = 2,
@@ -1935,6 +1918,7 @@ static struct snd_soc_dai_driver wm5100_dai[] = {
{
.name = "wm5100-aif3",
.id = 2,
+ .base = WM5100_AUDIO_IF_3_1 - 1,
.playback = {
.stream_name = "AIF3 Playback",
.channels_min = 2,
@@ -2454,7 +2438,7 @@ static __devinit int wm5100_i2c_probe(struct i2c_client *i2c,
wm5100->dev = &i2c->dev;
- wm5100->regmap = regmap_init_i2c(i2c, &wm5100_regmap);
+ wm5100->regmap = devm_regmap_init_i2c(i2c, &wm5100_regmap);
if (IS_ERR(wm5100->regmap)) {
ret = PTR_ERR(wm5100->regmap);
dev_err(&i2c->dev, "Failed to allocate register map: %d\n",
@@ -2479,7 +2463,7 @@ static __devinit int wm5100_i2c_probe(struct i2c_client *i2c,
if (ret != 0) {
dev_err(&i2c->dev, "Failed to request core supplies: %d\n",
ret);
- goto err_regmap;
+ goto err;
}
ret = regulator_bulk_enable(ARRAY_SIZE(wm5100->core_supplies),
@@ -2487,7 +2471,7 @@ static __devinit int wm5100_i2c_probe(struct i2c_client *i2c,
if (ret != 0) {
dev_err(&i2c->dev, "Failed to enable core supplies: %d\n",
ret);
- goto err_regmap;
+ goto err;
}
if (wm5100->pdata.ldo_ena) {
@@ -2660,8 +2644,6 @@ err_ldo:
err_enable:
regulator_bulk_disable(ARRAY_SIZE(wm5100->core_supplies),
wm5100->core_supplies);
-err_regmap:
- regmap_exit(wm5100->regmap);
err:
return ret;
}
@@ -2682,7 +2664,6 @@ static __devexit int wm5100_i2c_remove(struct i2c_client *i2c)
gpio_set_value_cansleep(wm5100->pdata.ldo_ena, 0);
gpio_free(wm5100->pdata.ldo_ena);
}
- regmap_exit(wm5100->regmap);
return 0;
}
@@ -2749,17 +2730,7 @@ static struct i2c_driver wm5100_i2c_driver = {
.id_table = wm5100_i2c_id,
};
-static int __init wm5100_modinit(void)
-{
- return i2c_add_driver(&wm5100_i2c_driver);
-}
-module_init(wm5100_modinit);
-
-static void __exit wm5100_exit(void)
-{
- i2c_del_driver(&wm5100_i2c_driver);
-}
-module_exit(wm5100_exit);
+module_i2c_driver(wm5100_i2c_driver);
MODULE_DESCRIPTION("ASoC WM5100 driver");
MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>");
diff --git a/sound/soc/codecs/wm5100.h b/sound/soc/codecs/wm5100.h
index 25cb6016f9d..935a9b7fb27 100644
--- a/sound/soc/codecs/wm5100.h
+++ b/sound/soc/codecs/wm5100.h
@@ -709,6 +709,96 @@ int wm5100_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack);
#define WM5100_HPLPF3_2 0xEC9
#define WM5100_HPLPF4_1 0xECC
#define WM5100_HPLPF4_2 0xECD
+#define WM5100_DSP1_CONTROL_1 0xF00
+#define WM5100_DSP1_CONTROL_2 0xF02
+#define WM5100_DSP1_CONTROL_3 0xF03
+#define WM5100_DSP1_CONTROL_4 0xF04
+#define WM5100_DSP1_CONTROL_5 0xF06
+#define WM5100_DSP1_CONTROL_6 0xF07
+#define WM5100_DSP1_CONTROL_7 0xF08
+#define WM5100_DSP1_CONTROL_8 0xF09
+#define WM5100_DSP1_CONTROL_9 0xF0A
+#define WM5100_DSP1_CONTROL_10 0xF0B
+#define WM5100_DSP1_CONTROL_11 0xF0C
+#define WM5100_DSP1_CONTROL_12 0xF0D
+#define WM5100_DSP1_CONTROL_13 0xF0F
+#define WM5100_DSP1_CONTROL_14 0xF10
+#define WM5100_DSP1_CONTROL_15 0xF11
+#define WM5100_DSP1_CONTROL_16 0xF12
+#define WM5100_DSP1_CONTROL_17 0xF13
+#define WM5100_DSP1_CONTROL_18 0xF14
+#define WM5100_DSP1_CONTROL_19 0xF16
+#define WM5100_DSP1_CONTROL_20 0xF17
+#define WM5100_DSP1_CONTROL_21 0xF18
+#define WM5100_DSP1_CONTROL_22 0xF1A
+#define WM5100_DSP1_CONTROL_23 0xF1B
+#define WM5100_DSP1_CONTROL_24 0xF1C
+#define WM5100_DSP1_CONTROL_25 0xF1E
+#define WM5100_DSP1_CONTROL_26 0xF20
+#define WM5100_DSP1_CONTROL_27 0xF21
+#define WM5100_DSP1_CONTROL_28 0xF22
+#define WM5100_DSP1_CONTROL_29 0xF23
+#define WM5100_DSP1_CONTROL_30 0xF24
+#define WM5100_DSP2_CONTROL_1 0x1000
+#define WM5100_DSP2_CONTROL_2 0x1002
+#define WM5100_DSP2_CONTROL_3 0x1003
+#define WM5100_DSP2_CONTROL_4 0x1004
+#define WM5100_DSP2_CONTROL_5 0x1006
+#define WM5100_DSP2_CONTROL_6 0x1007
+#define WM5100_DSP2_CONTROL_7 0x1008
+#define WM5100_DSP2_CONTROL_8 0x1009
+#define WM5100_DSP2_CONTROL_9 0x100A
+#define WM5100_DSP2_CONTROL_10 0x100B
+#define WM5100_DSP2_CONTROL_11 0x100C
+#define WM5100_DSP2_CONTROL_12 0x100D
+#define WM5100_DSP2_CONTROL_13 0x100F
+#define WM5100_DSP2_CONTROL_14 0x1010
+#define WM5100_DSP2_CONTROL_15 0x1011
+#define WM5100_DSP2_CONTROL_16 0x1012
+#define WM5100_DSP2_CONTROL_17 0x1013
+#define WM5100_DSP2_CONTROL_18 0x1014
+#define WM5100_DSP2_CONTROL_19 0x1016
+#define WM5100_DSP2_CONTROL_20 0x1017
+#define WM5100_DSP2_CONTROL_21 0x1018
+#define WM5100_DSP2_CONTROL_22 0x101A
+#define WM5100_DSP2_CONTROL_23 0x101B
+#define WM5100_DSP2_CONTROL_24 0x101C
+#define WM5100_DSP2_CONTROL_25 0x101E
+#define WM5100_DSP2_CONTROL_26 0x1020
+#define WM5100_DSP2_CONTROL_27 0x1021
+#define WM5100_DSP2_CONTROL_28 0x1022
+#define WM5100_DSP2_CONTROL_29 0x1023
+#define WM5100_DSP2_CONTROL_30 0x1024
+#define WM5100_DSP3_CONTROL_1 0x1100
+#define WM5100_DSP3_CONTROL_2 0x1102
+#define WM5100_DSP3_CONTROL_3 0x1103
+#define WM5100_DSP3_CONTROL_4 0x1104
+#define WM5100_DSP3_CONTROL_5 0x1106
+#define WM5100_DSP3_CONTROL_6 0x1107
+#define WM5100_DSP3_CONTROL_7 0x1108
+#define WM5100_DSP3_CONTROL_8 0x1109
+#define WM5100_DSP3_CONTROL_9 0x110A
+#define WM5100_DSP3_CONTROL_10 0x110B
+#define WM5100_DSP3_CONTROL_11 0x110C
+#define WM5100_DSP3_CONTROL_12 0x110D
+#define WM5100_DSP3_CONTROL_13 0x110F
+#define WM5100_DSP3_CONTROL_14 0x1110
+#define WM5100_DSP3_CONTROL_15 0x1111
+#define WM5100_DSP3_CONTROL_16 0x1112
+#define WM5100_DSP3_CONTROL_17 0x1113
+#define WM5100_DSP3_CONTROL_18 0x1114
+#define WM5100_DSP3_CONTROL_19 0x1116
+#define WM5100_DSP3_CONTROL_20 0x1117
+#define WM5100_DSP3_CONTROL_21 0x1118
+#define WM5100_DSP3_CONTROL_22 0x111A
+#define WM5100_DSP3_CONTROL_23 0x111B
+#define WM5100_DSP3_CONTROL_24 0x111C
+#define WM5100_DSP3_CONTROL_25 0x111E
+#define WM5100_DSP3_CONTROL_26 0x1120
+#define WM5100_DSP3_CONTROL_27 0x1121
+#define WM5100_DSP3_CONTROL_28 0x1122
+#define WM5100_DSP3_CONTROL_29 0x1123
+#define WM5100_DSP3_CONTROL_30 0x1124
#define WM5100_DSP1_DM_0 0x4000
#define WM5100_DSP1_DM_1 0x4001
#define WM5100_DSP1_DM_2 0x4002
@@ -4561,6 +4651,75 @@ int wm5100_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack);
#define WM5100_LHPF4_COEFF_WIDTH 16 /* LHPF4_COEFF - [15:0] */
/*
+ * R4132 (0x1024) - DSP2 Control 30
+ */
+#define WM5100_DSP2_RATE_MASK 0xC000 /* DSP2_RATE - [15:14] */
+#define WM5100_DSP2_RATE_SHIFT 14 /* DSP2_RATE - [15:14] */
+#define WM5100_DSP2_RATE_WIDTH 2 /* DSP2_RATE - [15:14] */
+#define WM5100_DSP2_DBG_CLK_ENA 0x0008 /* DSP2_DBG_CLK_ENA */
+#define WM5100_DSP2_DBG_CLK_ENA_MASK 0x0008 /* DSP2_DBG_CLK_ENA */
+#define WM5100_DSP2_DBG_CLK_ENA_SHIFT 3 /* DSP2_DBG_CLK_ENA */
+#define WM5100_DSP2_DBG_CLK_ENA_WIDTH 1 /* DSP2_DBG_CLK_ENA */
+#define WM5100_DSP2_SYS_ENA 0x0004 /* DSP2_SYS_ENA */
+#define WM5100_DSP2_SYS_ENA_MASK 0x0004 /* DSP2_SYS_ENA */
+#define WM5100_DSP2_SYS_ENA_SHIFT 2 /* DSP2_SYS_ENA */
+#define WM5100_DSP2_SYS_ENA_WIDTH 1 /* DSP2_SYS_ENA */
+#define WM5100_DSP2_CORE_ENA 0x0002 /* DSP2_CORE_ENA */
+#define WM5100_DSP2_CORE_ENA_MASK 0x0002 /* DSP2_CORE_ENA */
+#define WM5100_DSP2_CORE_ENA_SHIFT 1 /* DSP2_CORE_ENA */
+#define WM5100_DSP2_CORE_ENA_WIDTH 1 /* DSP2_CORE_ENA */
+#define WM5100_DSP2_START 0x0001 /* DSP2_START */
+#define WM5100_DSP2_START_MASK 0x0001 /* DSP2_START */
+#define WM5100_DSP2_START_SHIFT 0 /* DSP2_START */
+#define WM5100_DSP2_START_WIDTH 1 /* DSP2_START */
+
+/*
+ * R3876 (0xF24) - DSP1 Control 30
+ */
+#define WM5100_DSP1_RATE_MASK 0xC000 /* DSP1_RATE - [15:14] */
+#define WM5100_DSP1_RATE_SHIFT 14 /* DSP1_RATE - [15:14] */
+#define WM5100_DSP1_RATE_WIDTH 2 /* DSP1_RATE - [15:14] */
+#define WM5100_DSP1_DBG_CLK_ENA 0x0008 /* DSP1_DBG_CLK_ENA */
+#define WM5100_DSP1_DBG_CLK_ENA_MASK 0x0008 /* DSP1_DBG_CLK_ENA */
+#define WM5100_DSP1_DBG_CLK_ENA_SHIFT 3 /* DSP1_DBG_CLK_ENA */
+#define WM5100_DSP1_DBG_CLK_ENA_WIDTH 1 /* DSP1_DBG_CLK_ENA */
+#define WM5100_DSP1_SYS_ENA 0x0004 /* DSP1_SYS_ENA */
+#define WM5100_DSP1_SYS_ENA_MASK 0x0004 /* DSP1_SYS_ENA */
+#define WM5100_DSP1_SYS_ENA_SHIFT 2 /* DSP1_SYS_ENA */
+#define WM5100_DSP1_SYS_ENA_WIDTH 1 /* DSP1_SYS_ENA */
+#define WM5100_DSP1_CORE_ENA 0x0002 /* DSP1_CORE_ENA */
+#define WM5100_DSP1_CORE_ENA_MASK 0x0002 /* DSP1_CORE_ENA */
+#define WM5100_DSP1_CORE_ENA_SHIFT 1 /* DSP1_CORE_ENA */
+#define WM5100_DSP1_CORE_ENA_WIDTH 1 /* DSP1_CORE_ENA */
+#define WM5100_DSP1_START 0x0001 /* DSP1_START */
+#define WM5100_DSP1_START_MASK 0x0001 /* DSP1_START */
+#define WM5100_DSP1_START_SHIFT 0 /* DSP1_START */
+#define WM5100_DSP1_START_WIDTH 1 /* DSP1_START */
+
+/*
+ * R4388 (0x1124) - DSP3 Control 30
+ */
+#define WM5100_DSP3_RATE_MASK 0xC000 /* DSP3_RATE - [15:14] */
+#define WM5100_DSP3_RATE_SHIFT 14 /* DSP3_RATE - [15:14] */
+#define WM5100_DSP3_RATE_WIDTH 2 /* DSP3_RATE - [15:14] */
+#define WM5100_DSP3_DBG_CLK_ENA 0x0008 /* DSP3_DBG_CLK_ENA */
+#define WM5100_DSP3_DBG_CLK_ENA_MASK 0x0008 /* DSP3_DBG_CLK_ENA */
+#define WM5100_DSP3_DBG_CLK_ENA_SHIFT 3 /* DSP3_DBG_CLK_ENA */
+#define WM5100_DSP3_DBG_CLK_ENA_WIDTH 1 /* DSP3_DBG_CLK_ENA */
+#define WM5100_DSP3_SYS_ENA 0x0004 /* DSP3_SYS_ENA */
+#define WM5100_DSP3_SYS_ENA_MASK 0x0004 /* DSP3_SYS_ENA */
+#define WM5100_DSP3_SYS_ENA_SHIFT 2 /* DSP3_SYS_ENA */
+#define WM5100_DSP3_SYS_ENA_WIDTH 1 /* DSP3_SYS_ENA */
+#define WM5100_DSP3_CORE_ENA 0x0002 /* DSP3_CORE_ENA */
+#define WM5100_DSP3_CORE_ENA_MASK 0x0002 /* DSP3_CORE_ENA */
+#define WM5100_DSP3_CORE_ENA_SHIFT 1 /* DSP3_CORE_ENA */
+#define WM5100_DSP3_CORE_ENA_WIDTH 1 /* DSP3_CORE_ENA */
+#define WM5100_DSP3_START 0x0001 /* DSP3_START */
+#define WM5100_DSP3_START_MASK 0x0001 /* DSP3_START */
+#define WM5100_DSP3_START_SHIFT 0 /* DSP3_START */
+#define WM5100_DSP3_START_WIDTH 1 /* DSP3_START */
+
+/*
* R16384 (0x4000) - DSP1 DM 0
*/
#define WM5100_DSP1_DM_START_1_MASK 0x00FF /* DSP1_DM_START - [7:0] */
diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c
index aa12c6b6bee..555ee146ae0 100644
--- a/sound/soc/codecs/wm8350.c
+++ b/sound/soc/codecs/wm8350.c
@@ -71,13 +71,6 @@ struct wm8350_data {
int fll_freq_in;
};
-static unsigned int wm8350_codec_cache_read(struct snd_soc_codec *codec,
- unsigned int reg)
-{
- struct wm8350 *wm8350 = codec->control_data;
- return wm8350->reg_cache[reg];
-}
-
static unsigned int wm8350_codec_read(struct snd_soc_codec *codec,
unsigned int reg)
{
@@ -99,7 +92,7 @@ static inline int wm8350_out1_ramp_step(struct snd_soc_codec *codec)
{
struct wm8350_data *wm8350_data = snd_soc_codec_get_drvdata(codec);
struct wm8350_output *out1 = &wm8350_data->out1;
- struct wm8350 *wm8350 = codec->control_data;
+ struct wm8350 *wm8350 = wm8350_data->wm8350;
int left_complete = 0, right_complete = 0;
u16 reg, val;
@@ -165,7 +158,7 @@ static inline int wm8350_out2_ramp_step(struct snd_soc_codec *codec)
{
struct wm8350_data *wm8350_data = snd_soc_codec_get_drvdata(codec);
struct wm8350_output *out2 = &wm8350_data->out2;
- struct wm8350 *wm8350 = codec->control_data;
+ struct wm8350 *wm8350 = wm8350_data->wm8350;
int left_complete = 0, right_complete = 0;
u16 reg, val;
@@ -360,8 +353,8 @@ static int wm8350_put_volsw_2r_vu(struct snd_kcontrol *kcontrol,
return ret;
/* now hit the volume update bits (always bit 8) */
- val = wm8350_codec_read(codec, reg);
- wm8350_codec_write(codec, reg, val | WM8350_OUT1_VU);
+ val = snd_soc_read(codec, reg);
+ snd_soc_write(codec, reg, val | WM8350_OUT1_VU);
return 1;
}
@@ -781,7 +774,8 @@ static int wm8350_set_dai_sysclk(struct snd_soc_dai *codec_dai,
int clk_id, unsigned int freq, int dir)
{
struct snd_soc_codec *codec = codec_dai->codec;
- struct wm8350 *wm8350 = codec->control_data;
+ struct wm8350_data *wm8350_data = snd_soc_codec_get_drvdata(codec);
+ struct wm8350 *wm8350 = wm8350_data->wm8350;
u16 fll_4;
switch (clk_id) {
@@ -795,9 +789,9 @@ static int wm8350_set_dai_sysclk(struct snd_soc_dai *codec_dai,
case WM8350_MCLK_SEL_PLL_32K:
wm8350_set_bits(wm8350, WM8350_CLOCK_CONTROL_1,
WM8350_MCLK_SEL);
- fll_4 = wm8350_codec_read(codec, WM8350_FLL_CONTROL_4) &
+ fll_4 = snd_soc_read(codec, WM8350_FLL_CONTROL_4) &
~WM8350_FLL_CLK_SRC_MASK;
- wm8350_codec_write(codec, WM8350_FLL_CONTROL_4, fll_4 | clk_id);
+ snd_soc_write(codec, WM8350_FLL_CONTROL_4, fll_4 | clk_id);
break;
}
@@ -819,39 +813,39 @@ static int wm8350_set_clkdiv(struct snd_soc_dai *codec_dai, int div_id, int div)
switch (div_id) {
case WM8350_ADC_CLKDIV:
- val = wm8350_codec_read(codec, WM8350_ADC_DIVIDER) &
+ val = snd_soc_read(codec, WM8350_ADC_DIVIDER) &
~WM8350_ADC_CLKDIV_MASK;
- wm8350_codec_write(codec, WM8350_ADC_DIVIDER, val | div);
+ snd_soc_write(codec, WM8350_ADC_DIVIDER, val | div);
break;
case WM8350_DAC_CLKDIV:
- val = wm8350_codec_read(codec, WM8350_DAC_CLOCK_CONTROL) &
+ val = snd_soc_read(codec, WM8350_DAC_CLOCK_CONTROL) &
~WM8350_DAC_CLKDIV_MASK;
- wm8350_codec_write(codec, WM8350_DAC_CLOCK_CONTROL, val | div);
+ snd_soc_write(codec, WM8350_DAC_CLOCK_CONTROL, val | div);
break;
case WM8350_BCLK_CLKDIV:
- val = wm8350_codec_read(codec, WM8350_CLOCK_CONTROL_1) &
+ val = snd_soc_read(codec, WM8350_CLOCK_CONTROL_1) &
~WM8350_BCLK_DIV_MASK;
- wm8350_codec_write(codec, WM8350_CLOCK_CONTROL_1, val | div);
+ snd_soc_write(codec, WM8350_CLOCK_CONTROL_1, val | div);
break;
case WM8350_OPCLK_CLKDIV:
- val = wm8350_codec_read(codec, WM8350_CLOCK_CONTROL_1) &
+ val = snd_soc_read(codec, WM8350_CLOCK_CONTROL_1) &
~WM8350_OPCLK_DIV_MASK;
- wm8350_codec_write(codec, WM8350_CLOCK_CONTROL_1, val | div);
+ snd_soc_write(codec, WM8350_CLOCK_CONTROL_1, val | div);
break;
case WM8350_SYS_CLKDIV:
- val = wm8350_codec_read(codec, WM8350_CLOCK_CONTROL_1) &
+ val = snd_soc_read(codec, WM8350_CLOCK_CONTROL_1) &
~WM8350_MCLK_DIV_MASK;
- wm8350_codec_write(codec, WM8350_CLOCK_CONTROL_1, val | div);
+ snd_soc_write(codec, WM8350_CLOCK_CONTROL_1, val | div);
break;
case WM8350_DACLR_CLKDIV:
- val = wm8350_codec_read(codec, WM8350_DAC_LR_RATE) &
+ val = snd_soc_read(codec, WM8350_DAC_LR_RATE) &
~WM8350_DACLRC_RATE_MASK;
- wm8350_codec_write(codec, WM8350_DAC_LR_RATE, val | div);
+ snd_soc_write(codec, WM8350_DAC_LR_RATE, val | div);
break;
case WM8350_ADCLR_CLKDIV:
- val = wm8350_codec_read(codec, WM8350_ADC_LR_RATE) &
+ val = snd_soc_read(codec, WM8350_ADC_LR_RATE) &
~WM8350_ADCLRC_RATE_MASK;
- wm8350_codec_write(codec, WM8350_ADC_LR_RATE, val | div);
+ snd_soc_write(codec, WM8350_ADC_LR_RATE, val | div);
break;
default:
return -EINVAL;
@@ -863,13 +857,13 @@ static int wm8350_set_clkdiv(struct snd_soc_dai *codec_dai, int div_id, int div)
static int wm8350_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
{
struct snd_soc_codec *codec = codec_dai->codec;
- u16 iface = wm8350_codec_read(codec, WM8350_AI_FORMATING) &
+ u16 iface = snd_soc_read(codec, WM8350_AI_FORMATING) &
~(WM8350_AIF_BCLK_INV | WM8350_AIF_LRCLK_INV | WM8350_AIF_FMT_MASK);
- u16 master = wm8350_codec_read(codec, WM8350_AI_DAC_CONTROL) &
+ u16 master = snd_soc_read(codec, WM8350_AI_DAC_CONTROL) &
~WM8350_BCLK_MSTR;
- u16 dac_lrc = wm8350_codec_read(codec, WM8350_DAC_LR_RATE) &
+ u16 dac_lrc = snd_soc_read(codec, WM8350_DAC_LR_RATE) &
~WM8350_DACLRC_ENA;
- u16 adc_lrc = wm8350_codec_read(codec, WM8350_ADC_LR_RATE) &
+ u16 adc_lrc = snd_soc_read(codec, WM8350_ADC_LR_RATE) &
~WM8350_ADCLRC_ENA;
/* set master/slave audio interface */
@@ -922,42 +916,10 @@ static int wm8350_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
return -EINVAL;
}
- wm8350_codec_write(codec, WM8350_AI_FORMATING, iface);
- wm8350_codec_write(codec, WM8350_AI_DAC_CONTROL, master);
- wm8350_codec_write(codec, WM8350_DAC_LR_RATE, dac_lrc);
- wm8350_codec_write(codec, WM8350_ADC_LR_RATE, adc_lrc);
- return 0;
-}
-
-static int wm8350_pcm_trigger(struct snd_pcm_substream *substream,
- int cmd, struct snd_soc_dai *codec_dai)
-{
- struct snd_soc_codec *codec = codec_dai->codec;
- int master = wm8350_codec_cache_read(codec, WM8350_AI_DAC_CONTROL) &
- WM8350_BCLK_MSTR;
- int enabled = 0;
-
- /* Check that the DACs or ADCs are enabled since they are
- * required for LRC in master mode. The DACs or ADCs need a
- * valid audio path i.e. pin -> ADC or DAC -> pin before
- * the LRC will be enabled in master mode. */
- if (!master || cmd != SNDRV_PCM_TRIGGER_START)
- return 0;
-
- if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
- enabled = wm8350_codec_cache_read(codec, WM8350_POWER_MGMT_4) &
- (WM8350_ADCR_ENA | WM8350_ADCL_ENA);
- } else {
- enabled = wm8350_codec_cache_read(codec, WM8350_POWER_MGMT_4) &
- (WM8350_DACR_ENA | WM8350_DACL_ENA);
- }
-
- if (!enabled) {
- dev_err(codec->dev,
- "%s: invalid audio path - no clocks available\n",
- __func__);
- return -EINVAL;
- }
+ snd_soc_write(codec, WM8350_AI_FORMATING, iface);
+ snd_soc_write(codec, WM8350_AI_DAC_CONTROL, master);
+ snd_soc_write(codec, WM8350_DAC_LR_RATE, dac_lrc);
+ snd_soc_write(codec, WM8350_ADC_LR_RATE, adc_lrc);
return 0;
}
@@ -966,8 +928,9 @@ static int wm8350_pcm_hw_params(struct snd_pcm_substream *substream,
struct snd_soc_dai *codec_dai)
{
struct snd_soc_codec *codec = codec_dai->codec;
- struct wm8350 *wm8350 = codec->control_data;
- u16 iface = wm8350_codec_read(codec, WM8350_AI_FORMATING) &
+ struct wm8350_data *wm8350_data = snd_soc_codec_get_drvdata(codec);
+ struct wm8350 *wm8350 = wm8350_data->wm8350;
+ u16 iface = snd_soc_read(codec, WM8350_AI_FORMATING) &
~WM8350_AIF_WL_MASK;
/* bit size */
@@ -985,7 +948,7 @@ static int wm8350_pcm_hw_params(struct snd_pcm_substream *substream,
break;
}
- wm8350_codec_write(codec, WM8350_AI_FORMATING, iface);
+ snd_soc_write(codec, WM8350_AI_FORMATING, iface);
/* The sloping stopband filter is recommended for use with
* lower sample rates to improve performance.
@@ -1005,12 +968,15 @@ static int wm8350_pcm_hw_params(struct snd_pcm_substream *substream,
static int wm8350_mute(struct snd_soc_dai *dai, int mute)
{
struct snd_soc_codec *codec = dai->codec;
- struct wm8350 *wm8350 = codec->control_data;
+ unsigned int val;
if (mute)
- wm8350_set_bits(wm8350, WM8350_DAC_MUTE, WM8350_DAC_MUTE_ENA);
+ val = WM8350_DAC_MUTE_ENA;
else
- wm8350_clear_bits(wm8350, WM8350_DAC_MUTE, WM8350_DAC_MUTE_ENA);
+ val = 0;
+
+ snd_soc_update_bits(codec, WM8350_DAC_MUTE, WM8350_DAC_MUTE_ENA, val);
+
return 0;
}
@@ -1079,8 +1045,8 @@ static int wm8350_set_fll(struct snd_soc_dai *codec_dai,
unsigned int freq_out)
{
struct snd_soc_codec *codec = codec_dai->codec;
- struct wm8350 *wm8350 = codec->control_data;
struct wm8350_data *priv = snd_soc_codec_get_drvdata(codec);
+ struct wm8350 *wm8350 = priv->wm8350;
struct _fll_div fll_div;
int ret = 0;
u16 fll_1, fll_4;
@@ -1104,17 +1070,17 @@ static int wm8350_set_fll(struct snd_soc_dai *codec_dai,
fll_div.ratio);
/* set up N.K & dividers */
- fll_1 = wm8350_codec_read(codec, WM8350_FLL_CONTROL_1) &
+ fll_1 = snd_soc_read(codec, WM8350_FLL_CONTROL_1) &
~(WM8350_FLL_OUTDIV_MASK | WM8350_FLL_RSP_RATE_MASK | 0xc000);
- wm8350_codec_write(codec, WM8350_FLL_CONTROL_1,
+ snd_soc_write(codec, WM8350_FLL_CONTROL_1,
fll_1 | (fll_div.div << 8) | 0x50);
- wm8350_codec_write(codec, WM8350_FLL_CONTROL_2,
+ snd_soc_write(codec, WM8350_FLL_CONTROL_2,
(fll_div.ratio << 11) | (fll_div.
n & WM8350_FLL_N_MASK));
- wm8350_codec_write(codec, WM8350_FLL_CONTROL_3, fll_div.k);
- fll_4 = wm8350_codec_read(codec, WM8350_FLL_CONTROL_4) &
+ snd_soc_write(codec, WM8350_FLL_CONTROL_3, fll_div.k);
+ fll_4 = snd_soc_read(codec, WM8350_FLL_CONTROL_4) &
~(WM8350_FLL_FRAC | WM8350_FLL_SLOW_LOCK_REF);
- wm8350_codec_write(codec, WM8350_FLL_CONTROL_4,
+ snd_soc_write(codec, WM8350_FLL_CONTROL_4,
fll_4 | (fll_div.k ? WM8350_FLL_FRAC : 0) |
(fll_div.ratio == 8 ? WM8350_FLL_SLOW_LOCK_REF : 0));
@@ -1131,8 +1097,8 @@ static int wm8350_set_fll(struct snd_soc_dai *codec_dai,
static int wm8350_set_bias_level(struct snd_soc_codec *codec,
enum snd_soc_bias_level level)
{
- struct wm8350 *wm8350 = codec->control_data;
struct wm8350_data *priv = snd_soc_codec_get_drvdata(codec);
+ struct wm8350 *wm8350 = priv->wm8350;
struct wm8350_audio_platform_data *platform =
wm8350->codec.platform_data;
u16 pm1;
@@ -1339,35 +1305,36 @@ static void wm8350_hpr_work(struct work_struct *work)
wm8350_hp_work(priv, &priv->hpr, WM8350_JACK_R_LVL);
}
-static irqreturn_t wm8350_hp_jack_handler(int irq, void *data)
+static irqreturn_t wm8350_hpl_jack_handler(int irq, void *data)
{
struct wm8350_data *priv = data;
struct wm8350 *wm8350 = priv->wm8350;
- struct wm8350_jack_data *jack = NULL;
- switch (irq - wm8350->irq_base) {
- case WM8350_IRQ_CODEC_JCK_DET_L:
#ifndef CONFIG_SND_SOC_WM8350_MODULE
- trace_snd_soc_jack_irq("WM8350 HPL");
+ trace_snd_soc_jack_irq("WM8350 HPL");
#endif
- jack = &priv->hpl;
- break;
- case WM8350_IRQ_CODEC_JCK_DET_R:
+ if (device_may_wakeup(wm8350->dev))
+ pm_wakeup_event(wm8350->dev, 250);
+
+ schedule_delayed_work(&priv->hpl.work, 200);
+
+ return IRQ_HANDLED;
+}
+
+static irqreturn_t wm8350_hpr_jack_handler(int irq, void *data)
+{
+ struct wm8350_data *priv = data;
+ struct wm8350 *wm8350 = priv->wm8350;
+
#ifndef CONFIG_SND_SOC_WM8350_MODULE
- trace_snd_soc_jack_irq("WM8350 HPR");
+ trace_snd_soc_jack_irq("WM8350 HPR");
#endif
- jack = &priv->hpr;
- break;
-
- default:
- BUG();
- }
if (device_may_wakeup(wm8350->dev))
pm_wakeup_event(wm8350->dev, 250);
- schedule_delayed_work(&jack->work, 200);
+ schedule_delayed_work(&priv->hpr.work, 200);
return IRQ_HANDLED;
}
@@ -1387,7 +1354,7 @@ int wm8350_hp_jack_detect(struct snd_soc_codec *codec, enum wm8350_jack which,
struct snd_soc_jack *jack, int report)
{
struct wm8350_data *priv = snd_soc_codec_get_drvdata(codec);
- struct wm8350 *wm8350 = codec->control_data;
+ struct wm8350 *wm8350 = priv->wm8350;
int irq;
int ena;
@@ -1418,7 +1385,14 @@ int wm8350_hp_jack_detect(struct snd_soc_codec *codec, enum wm8350_jack which,
}
/* Sync status */
- wm8350_hp_jack_handler(irq + wm8350->irq_base, priv);
+ switch (which) {
+ case WM8350_JDL:
+ wm8350_hpl_jack_handler(0, priv);
+ break;
+ case WM8350_JDR:
+ wm8350_hpr_jack_handler(0, priv);
+ break;
+ }
return 0;
}
@@ -1463,7 +1437,7 @@ int wm8350_mic_jack_detect(struct snd_soc_codec *codec,
int detect_report, int short_report)
{
struct wm8350_data *priv = snd_soc_codec_get_drvdata(codec);
- struct wm8350 *wm8350 = codec->control_data;
+ struct wm8350 *wm8350 = priv->wm8350;
priv->mic.jack = jack;
priv->mic.report = detect_report;
@@ -1491,7 +1465,6 @@ EXPORT_SYMBOL_GPL(wm8350_mic_jack_detect);
static const struct snd_soc_dai_ops wm8350_dai_ops = {
.hw_params = wm8350_pcm_hw_params,
.digital_mute = wm8350_mute,
- .trigger = wm8350_pcm_trigger,
.set_fmt = wm8350_set_dai_fmt,
.set_sysclk = wm8350_set_dai_sysclk,
.set_pll = wm8350_set_fll,
@@ -1559,9 +1532,9 @@ static int wm8350_codec_probe(struct snd_soc_codec *codec)
wm8350_set_bits(wm8350, WM8350_POWER_MGMT_5, WM8350_CODEC_ENA);
/* Enable robust clocking mode in ADC */
- wm8350_codec_write(codec, WM8350_SECURITY, 0xa7);
- wm8350_codec_write(codec, 0xde, 0x13);
- wm8350_codec_write(codec, WM8350_SECURITY, 0);
+ snd_soc_write(codec, WM8350_SECURITY, 0xa7);
+ snd_soc_write(codec, 0xde, 0x13);
+ snd_soc_write(codec, WM8350_SECURITY, 0);
/* read OUT1 & OUT2 volumes */
out1 = &priv->out1;
@@ -1601,10 +1574,10 @@ static int wm8350_codec_probe(struct snd_soc_codec *codec)
WM8350_JDL_ENA | WM8350_JDR_ENA);
wm8350_register_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_L,
- wm8350_hp_jack_handler, 0, "Left jack detect",
+ wm8350_hpl_jack_handler, 0, "Left jack detect",
priv);
wm8350_register_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_R,
- wm8350_hp_jack_handler, 0, "Right jack detect",
+ wm8350_hpr_jack_handler, 0, "Right jack detect",
priv);
wm8350_register_irq(wm8350, WM8350_IRQ_CODEC_MICSCD,
wm8350_mic_handler, 0, "Microphone short", priv);
diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c
index 898979d2301..5dc31ebcd0e 100644
--- a/sound/soc/codecs/wm8400.c
+++ b/sound/soc/codecs/wm8400.c
@@ -138,8 +138,8 @@ static int wm8400_outpga_put_volsw_vu(struct snd_kcontrol *kcontrol,
return ret;
/* now hit the volume update bits (always bit 8) */
- val = wm8400_read(codec, reg);
- return wm8400_write(codec, reg, val | 0x0100);
+ val = snd_soc_read(codec, reg);
+ return snd_soc_write(codec, reg, val | 0x0100);
}
#define WM8400_OUTPGA_SINGLE_R_TLV(xname, reg, shift, max, invert, tlv_array) \
@@ -362,8 +362,8 @@ static int inmixer_event (struct snd_soc_dapm_widget *w,
{
u16 reg, fakepower;
- reg = wm8400_read(w->codec, WM8400_POWER_MANAGEMENT_2);
- fakepower = wm8400_read(w->codec, WM8400_INTDRIVBITS);
+ reg = snd_soc_read(w->codec, WM8400_POWER_MANAGEMENT_2);
+ fakepower = snd_soc_read(w->codec, WM8400_INTDRIVBITS);
if (fakepower & ((1 << WM8400_INMIXL_PWR) |
(1 << WM8400_AINLMUX_PWR))) {
@@ -378,7 +378,7 @@ static int inmixer_event (struct snd_soc_dapm_widget *w,
} else {
reg &= ~WM8400_AINR_ENA;
}
- wm8400_write(w->codec, WM8400_POWER_MANAGEMENT_2, reg);
+ snd_soc_write(w->codec, WM8400_POWER_MANAGEMENT_2, reg);
return 0;
}
@@ -394,7 +394,7 @@ static int outmixer_event (struct snd_soc_dapm_widget *w,
switch (reg_shift) {
case WM8400_SPEAKER_MIXER | (WM8400_LDSPK << 8) :
- reg = wm8400_read(w->codec, WM8400_OUTPUT_MIXER1);
+ reg = snd_soc_read(w->codec, WM8400_OUTPUT_MIXER1);
if (reg & WM8400_LDLO) {
printk(KERN_WARNING
"Cannot set as Output Mixer 1 LDLO Set\n");
@@ -402,7 +402,7 @@ static int outmixer_event (struct snd_soc_dapm_widget *w,
}
break;
case WM8400_SPEAKER_MIXER | (WM8400_RDSPK << 8):
- reg = wm8400_read(w->codec, WM8400_OUTPUT_MIXER2);
+ reg = snd_soc_read(w->codec, WM8400_OUTPUT_MIXER2);
if (reg & WM8400_RDRO) {
printk(KERN_WARNING
"Cannot set as Output Mixer 2 RDRO Set\n");
@@ -410,7 +410,7 @@ static int outmixer_event (struct snd_soc_dapm_widget *w,
}
break;
case WM8400_OUTPUT_MIXER1 | (WM8400_LDLO << 8):
- reg = wm8400_read(w->codec, WM8400_SPEAKER_MIXER);
+ reg = snd_soc_read(w->codec, WM8400_SPEAKER_MIXER);
if (reg & WM8400_LDSPK) {
printk(KERN_WARNING
"Cannot set as Speaker Mixer LDSPK Set\n");
@@ -418,7 +418,7 @@ static int outmixer_event (struct snd_soc_dapm_widget *w,
}
break;
case WM8400_OUTPUT_MIXER2 | (WM8400_RDRO << 8):
- reg = wm8400_read(w->codec, WM8400_SPEAKER_MIXER);
+ reg = snd_soc_read(w->codec, WM8400_SPEAKER_MIXER);
if (reg & WM8400_RDSPK) {
printk(KERN_WARNING
"Cannot set as Speaker Mixer RDSPK Set\n");
@@ -1021,13 +1021,13 @@ static int wm8400_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id,
wm8400->fll_in = freq_in;
/* We *must* disable the FLL before any changes */
- reg = wm8400_read(codec, WM8400_POWER_MANAGEMENT_2);
+ reg = snd_soc_read(codec, WM8400_POWER_MANAGEMENT_2);
reg &= ~WM8400_FLL_ENA;
- wm8400_write(codec, WM8400_POWER_MANAGEMENT_2, reg);
+ snd_soc_write(codec, WM8400_POWER_MANAGEMENT_2, reg);
- reg = wm8400_read(codec, WM8400_FLL_CONTROL_1);
+ reg = snd_soc_read(codec, WM8400_FLL_CONTROL_1);
reg &= ~WM8400_FLL_OSC_ENA;
- wm8400_write(codec, WM8400_FLL_CONTROL_1, reg);
+ snd_soc_write(codec, WM8400_FLL_CONTROL_1, reg);
if (!freq_out)
return 0;
@@ -1035,15 +1035,15 @@ static int wm8400_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id,
reg &= ~(WM8400_FLL_REF_FREQ | WM8400_FLL_FRATIO_MASK);
reg |= WM8400_FLL_FRAC | factors.fratio;
reg |= factors.freq_ref << WM8400_FLL_REF_FREQ_SHIFT;
- wm8400_write(codec, WM8400_FLL_CONTROL_1, reg);
+ snd_soc_write(codec, WM8400_FLL_CONTROL_1, reg);
- wm8400_write(codec, WM8400_FLL_CONTROL_2, factors.k);
- wm8400_write(codec, WM8400_FLL_CONTROL_3, factors.n);
+ snd_soc_write(codec, WM8400_FLL_CONTROL_2, factors.k);
+ snd_soc_write(codec, WM8400_FLL_CONTROL_3, factors.n);
- reg = wm8400_read(codec, WM8400_FLL_CONTROL_4);
+ reg = snd_soc_read(codec, WM8400_FLL_CONTROL_4);
reg &= ~WM8400_FLL_OUTDIV_MASK;
reg |= factors.outdiv;
- wm8400_write(codec, WM8400_FLL_CONTROL_4, reg);
+ snd_soc_write(codec, WM8400_FLL_CONTROL_4, reg);
return 0;
}
@@ -1057,8 +1057,8 @@ static int wm8400_set_dai_fmt(struct snd_soc_dai *codec_dai,
struct snd_soc_codec *codec = codec_dai->codec;
u16 audio1, audio3;
- audio1 = wm8400_read(codec, WM8400_AUDIO_INTERFACE_1);
- audio3 = wm8400_read(codec, WM8400_AUDIO_INTERFACE_3);
+ audio1 = snd_soc_read(codec, WM8400_AUDIO_INTERFACE_1);
+ audio3 = snd_soc_read(codec, WM8400_AUDIO_INTERFACE_3);
/* set master/slave audio interface */
switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
@@ -1099,8 +1099,8 @@ static int wm8400_set_dai_fmt(struct snd_soc_dai *codec_dai,
return -EINVAL;
}
- wm8400_write(codec, WM8400_AUDIO_INTERFACE_1, audio1);
- wm8400_write(codec, WM8400_AUDIO_INTERFACE_3, audio3);
+ snd_soc_write(codec, WM8400_AUDIO_INTERFACE_1, audio1);
+ snd_soc_write(codec, WM8400_AUDIO_INTERFACE_3, audio3);
return 0;
}
@@ -1112,24 +1112,24 @@ static int wm8400_set_dai_clkdiv(struct snd_soc_dai *codec_dai,
switch (div_id) {
case WM8400_MCLK_DIV:
- reg = wm8400_read(codec, WM8400_CLOCKING_2) &
+ reg = snd_soc_read(codec, WM8400_CLOCKING_2) &
~WM8400_MCLK_DIV_MASK;
- wm8400_write(codec, WM8400_CLOCKING_2, reg | div);
+ snd_soc_write(codec, WM8400_CLOCKING_2, reg | div);
break;
case WM8400_DACCLK_DIV:
- reg = wm8400_read(codec, WM8400_CLOCKING_2) &
+ reg = snd_soc_read(codec, WM8400_CLOCKING_2) &
~WM8400_DAC_CLKDIV_MASK;
- wm8400_write(codec, WM8400_CLOCKING_2, reg | div);
+ snd_soc_write(codec, WM8400_CLOCKING_2, reg | div);
break;
case WM8400_ADCCLK_DIV:
- reg = wm8400_read(codec, WM8400_CLOCKING_2) &
+ reg = snd_soc_read(codec, WM8400_CLOCKING_2) &
~WM8400_ADC_CLKDIV_MASK;
- wm8400_write(codec, WM8400_CLOCKING_2, reg | div);
+ snd_soc_write(codec, WM8400_CLOCKING_2, reg | div);
break;
case WM8400_BCLK_DIV:
- reg = wm8400_read(codec, WM8400_CLOCKING_1) &
+ reg = snd_soc_read(codec, WM8400_CLOCKING_1) &
~WM8400_BCLK_DIV_MASK;
- wm8400_write(codec, WM8400_CLOCKING_1, reg | div);
+ snd_soc_write(codec, WM8400_CLOCKING_1, reg | div);
break;
default:
return -EINVAL;
@@ -1145,9 +1145,8 @@ static int wm8400_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 audio1 = wm8400_read(codec, WM8400_AUDIO_INTERFACE_1);
+ struct snd_soc_codec *codec = dai->codec;
+ u16 audio1 = snd_soc_read(codec, WM8400_AUDIO_INTERFACE_1);
audio1 &= ~WM8400_AIF_WL_MASK;
/* bit size */
@@ -1165,19 +1164,19 @@ static int wm8400_hw_params(struct snd_pcm_substream *substream,
break;
}
- wm8400_write(codec, WM8400_AUDIO_INTERFACE_1, audio1);
+ snd_soc_write(codec, WM8400_AUDIO_INTERFACE_1, audio1);
return 0;
}
static int wm8400_mute(struct snd_soc_dai *dai, int mute)
{
struct snd_soc_codec *codec = dai->codec;
- u16 val = wm8400_read(codec, WM8400_DAC_CTRL) & ~WM8400_DAC_MUTE;
+ u16 val = snd_soc_read(codec, WM8400_DAC_CTRL) & ~WM8400_DAC_MUTE;
if (mute)
- wm8400_write(codec, WM8400_DAC_CTRL, val | WM8400_DAC_MUTE);
+ snd_soc_write(codec, WM8400_DAC_CTRL, val | WM8400_DAC_MUTE);
else
- wm8400_write(codec, WM8400_DAC_CTRL, val);
+ snd_soc_write(codec, WM8400_DAC_CTRL, val);
return 0;
}
@@ -1196,9 +1195,9 @@ static int wm8400_set_bias_level(struct snd_soc_codec *codec,
case SND_SOC_BIAS_PREPARE:
/* VMID=2*50k */
- val = wm8400_read(codec, WM8400_POWER_MANAGEMENT_1) &
+ val = snd_soc_read(codec, WM8400_POWER_MANAGEMENT_1) &
~WM8400_VMID_MODE_MASK;
- wm8400_write(codec, WM8400_POWER_MANAGEMENT_1, val | 0x2);
+ snd_soc_write(codec, WM8400_POWER_MANAGEMENT_1, val | 0x2);
break;
case SND_SOC_BIAS_STANDBY:
@@ -1212,74 +1211,74 @@ static int wm8400_set_bias_level(struct snd_soc_codec *codec,
return ret;
}
- wm8400_write(codec, WM8400_POWER_MANAGEMENT_1,
+ snd_soc_write(codec, WM8400_POWER_MANAGEMENT_1,
WM8400_CODEC_ENA | WM8400_SYSCLK_ENA);
/* Enable POBCTRL, SOFT_ST, VMIDTOG and BUFDCOPEN */
- wm8400_write(codec, WM8400_ANTIPOP2, WM8400_SOFTST |
+ snd_soc_write(codec, WM8400_ANTIPOP2, WM8400_SOFTST |
WM8400_BUFDCOPEN | WM8400_POBCTRL);
msleep(50);
/* Enable VREF & VMID at 2x50k */
- val = wm8400_read(codec, WM8400_POWER_MANAGEMENT_1);
+ val = snd_soc_read(codec, WM8400_POWER_MANAGEMENT_1);
val |= 0x2 | WM8400_VREF_ENA;
- wm8400_write(codec, WM8400_POWER_MANAGEMENT_1, val);
+ snd_soc_write(codec, WM8400_POWER_MANAGEMENT_1, val);
/* Enable BUFIOEN */
- wm8400_write(codec, WM8400_ANTIPOP2, WM8400_SOFTST |
+ snd_soc_write(codec, WM8400_ANTIPOP2, WM8400_SOFTST |
WM8400_BUFDCOPEN | WM8400_POBCTRL |
WM8400_BUFIOEN);
/* disable POBCTRL, SOFT_ST and BUFDCOPEN */
- wm8400_write(codec, WM8400_ANTIPOP2, WM8400_BUFIOEN);
+ snd_soc_write(codec, WM8400_ANTIPOP2, WM8400_BUFIOEN);
}
/* VMID=2*300k */
- val = wm8400_read(codec, WM8400_POWER_MANAGEMENT_1) &
+ val = snd_soc_read(codec, WM8400_POWER_MANAGEMENT_1) &
~WM8400_VMID_MODE_MASK;
- wm8400_write(codec, WM8400_POWER_MANAGEMENT_1, val | 0x4);
+ snd_soc_write(codec, WM8400_POWER_MANAGEMENT_1, val | 0x4);
break;
case SND_SOC_BIAS_OFF:
/* Enable POBCTRL and SOFT_ST */
- wm8400_write(codec, WM8400_ANTIPOP2, WM8400_SOFTST |
+ snd_soc_write(codec, WM8400_ANTIPOP2, WM8400_SOFTST |
WM8400_POBCTRL | WM8400_BUFIOEN);
/* Enable POBCTRL, SOFT_ST and BUFDCOPEN */
- wm8400_write(codec, WM8400_ANTIPOP2, WM8400_SOFTST |
+ snd_soc_write(codec, WM8400_ANTIPOP2, WM8400_SOFTST |
WM8400_BUFDCOPEN | WM8400_POBCTRL |
WM8400_BUFIOEN);
/* mute DAC */
- val = wm8400_read(codec, WM8400_DAC_CTRL);
- wm8400_write(codec, WM8400_DAC_CTRL, val | WM8400_DAC_MUTE);
+ val = snd_soc_read(codec, WM8400_DAC_CTRL);
+ snd_soc_write(codec, WM8400_DAC_CTRL, val | WM8400_DAC_MUTE);
/* Enable any disabled outputs */
- val = wm8400_read(codec, WM8400_POWER_MANAGEMENT_1);
+ val = snd_soc_read(codec, WM8400_POWER_MANAGEMENT_1);
val |= WM8400_SPK_ENA | WM8400_OUT3_ENA |
WM8400_OUT4_ENA | WM8400_LOUT_ENA |
WM8400_ROUT_ENA;
- wm8400_write(codec, WM8400_POWER_MANAGEMENT_1, val);
+ snd_soc_write(codec, WM8400_POWER_MANAGEMENT_1, val);
/* Disable VMID */
val &= ~WM8400_VMID_MODE_MASK;
- wm8400_write(codec, WM8400_POWER_MANAGEMENT_1, val);
+ snd_soc_write(codec, WM8400_POWER_MANAGEMENT_1, val);
msleep(300);
/* Enable all output discharge bits */
- wm8400_write(codec, WM8400_ANTIPOP1, WM8400_DIS_LLINE |
+ snd_soc_write(codec, WM8400_ANTIPOP1, WM8400_DIS_LLINE |
WM8400_DIS_RLINE | WM8400_DIS_OUT3 |
WM8400_DIS_OUT4 | WM8400_DIS_LOUT |
WM8400_DIS_ROUT);
/* Disable VREF */
val &= ~WM8400_VREF_ENA;
- wm8400_write(codec, WM8400_POWER_MANAGEMENT_1, val);
+ snd_soc_write(codec, WM8400_POWER_MANAGEMENT_1, val);
/* disable POBCTRL, SOFT_ST and BUFDCOPEN */
- wm8400_write(codec, WM8400_ANTIPOP2, 0x0);
+ snd_soc_write(codec, WM8400_ANTIPOP2, 0x0);
ret = regulator_bulk_disable(ARRAY_SIZE(power),
&power[0]);
@@ -1385,19 +1384,19 @@ static int wm8400_codec_probe(struct snd_soc_codec *codec)
wm8400_codec_reset(codec);
- reg = wm8400_read(codec, WM8400_POWER_MANAGEMENT_1);
- wm8400_write(codec, WM8400_POWER_MANAGEMENT_1, reg | WM8400_CODEC_ENA);
+ reg = snd_soc_read(codec, WM8400_POWER_MANAGEMENT_1);
+ snd_soc_write(codec, WM8400_POWER_MANAGEMENT_1, reg | WM8400_CODEC_ENA);
/* Latch volume update bits */
- reg = wm8400_read(codec, WM8400_LEFT_LINE_INPUT_1_2_VOLUME);
- wm8400_write(codec, WM8400_LEFT_LINE_INPUT_1_2_VOLUME,
+ reg = snd_soc_read(codec, WM8400_LEFT_LINE_INPUT_1_2_VOLUME);
+ snd_soc_write(codec, WM8400_LEFT_LINE_INPUT_1_2_VOLUME,
reg & WM8400_IPVU);
- reg = wm8400_read(codec, WM8400_RIGHT_LINE_INPUT_1_2_VOLUME);
- wm8400_write(codec, WM8400_RIGHT_LINE_INPUT_1_2_VOLUME,
+ reg = snd_soc_read(codec, WM8400_RIGHT_LINE_INPUT_1_2_VOLUME);
+ snd_soc_write(codec, WM8400_RIGHT_LINE_INPUT_1_2_VOLUME,
reg & WM8400_IPVU);
- wm8400_write(codec, WM8400_LEFT_OUTPUT_VOLUME, 0x50 | (1<<8));
- wm8400_write(codec, WM8400_RIGHT_OUTPUT_VOLUME, 0x50 | (1<<8));
+ snd_soc_write(codec, WM8400_LEFT_OUTPUT_VOLUME, 0x50 | (1<<8));
+ snd_soc_write(codec, WM8400_RIGHT_OUTPUT_VOLUME, 0x50 | (1<<8));
if (!schedule_work(&priv->work)) {
ret = -EINVAL;
@@ -1414,8 +1413,8 @@ static int wm8400_codec_remove(struct snd_soc_codec *codec)
{
u16 reg;
- reg = wm8400_read(codec, WM8400_POWER_MANAGEMENT_1);
- wm8400_write(codec, WM8400_POWER_MANAGEMENT_1,
+ reg = snd_soc_read(codec, WM8400_POWER_MANAGEMENT_1);
+ snd_soc_write(codec, WM8400_POWER_MANAGEMENT_1,
reg & (~WM8400_CODEC_ENA));
regulator_bulk_free(ARRAY_SIZE(power), power);
@@ -1428,7 +1427,7 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8400 = {
.remove = wm8400_codec_remove,
.suspend = wm8400_suspend,
.resume = wm8400_resume,
- .read = wm8400_read,
+ .read = snd_soc_read,
.write = wm8400_write,
.set_bias_level = wm8400_set_bias_level,
diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c
index 9166126bd31..56a049555e2 100644
--- a/sound/soc/codecs/wm8510.c
+++ b/sound/soc/codecs/wm8510.c
@@ -392,8 +392,7 @@ static int wm8510_pcm_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;
+ struct snd_soc_codec *codec = dai->codec;
u16 iface = snd_soc_read(codec, WM8510_IFACE) & 0x19f;
u16 adn = snd_soc_read(codec, WM8510_ADD) & 0x1f1;
diff --git a/sound/soc/codecs/wm8523.c b/sound/soc/codecs/wm8523.c
index 7fea2c3bf7e..1c3ffb290cd 100644
--- a/sound/soc/codecs/wm8523.c
+++ b/sound/soc/codecs/wm8523.c
@@ -145,8 +145,7 @@ static int wm8523_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;
+ struct snd_soc_codec *codec = dai->codec;
struct wm8523_priv *wm8523 = snd_soc_codec_get_drvdata(codec);
int i;
u16 aifctrl1 = snd_soc_read(codec, WM8523_AIF_CTRL1);
diff --git a/sound/soc/codecs/wm8728.c b/sound/soc/codecs/wm8728.c
index fc3d59e4908..1467f97dce2 100644
--- a/sound/soc/codecs/wm8728.c
+++ b/sound/soc/codecs/wm8728.c
@@ -88,8 +88,7 @@ static int wm8728_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;
+ struct snd_soc_codec *codec = dai->codec;
u16 dac = snd_soc_read(codec, WM8728_DACCTL);
dac &= ~0x18;
diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c
index a32caa72bd7..9d1b9b0271f 100644
--- a/sound/soc/codecs/wm8731.c
+++ b/sound/soc/codecs/wm8731.c
@@ -635,16 +635,17 @@ static int __devinit wm8731_spi_probe(struct spi_device *spi)
struct wm8731_priv *wm8731;
int ret;
- wm8731 = kzalloc(sizeof(struct wm8731_priv), GFP_KERNEL);
+ wm8731 = devm_kzalloc(&spi->dev, sizeof(struct wm8731_priv),
+ GFP_KERNEL);
if (wm8731 == NULL)
return -ENOMEM;
- wm8731->regmap = regmap_init_spi(spi, &wm8731_regmap);
+ wm8731->regmap = devm_regmap_init_spi(spi, &wm8731_regmap);
if (IS_ERR(wm8731->regmap)) {
ret = PTR_ERR(wm8731->regmap);
dev_err(&spi->dev, "Failed to allocate register map: %d\n",
ret);
- goto err;
+ return ret;
}
spi_set_drvdata(spi, wm8731);
@@ -653,25 +654,15 @@ static int __devinit wm8731_spi_probe(struct spi_device *spi)
&soc_codec_dev_wm8731, &wm8731_dai, 1);
if (ret != 0) {
dev_err(&spi->dev, "Failed to register CODEC: %d\n", ret);
- goto err_regmap;
+ return ret;
}
return 0;
-
-err_regmap:
- regmap_exit(wm8731->regmap);
-err:
- kfree(wm8731);
- return ret;
}
static int __devexit wm8731_spi_remove(struct spi_device *spi)
{
- struct wm8731_priv *wm8731 = spi_get_drvdata(spi);
-
snd_soc_unregister_codec(&spi->dev);
- regmap_exit(wm8731->regmap);
- kfree(wm8731);
return 0;
}
@@ -693,16 +684,17 @@ static __devinit int wm8731_i2c_probe(struct i2c_client *i2c,
struct wm8731_priv *wm8731;
int ret;
- wm8731 = kzalloc(sizeof(struct wm8731_priv), GFP_KERNEL);
+ wm8731 = devm_kzalloc(&i2c->dev, sizeof(struct wm8731_priv),
+ GFP_KERNEL);
if (wm8731 == NULL)
return -ENOMEM;
- wm8731->regmap = regmap_init_i2c(i2c, &wm8731_regmap);
+ wm8731->regmap = devm_regmap_init_i2c(i2c, &wm8731_regmap);
if (IS_ERR(wm8731->regmap)) {
ret = PTR_ERR(wm8731->regmap);
dev_err(&i2c->dev, "Failed to allocate register map: %d\n",
ret);
- goto err;
+ return ret;
}
i2c_set_clientdata(i2c, wm8731);
@@ -711,24 +703,15 @@ static __devinit int wm8731_i2c_probe(struct i2c_client *i2c,
&soc_codec_dev_wm8731, &wm8731_dai, 1);
if (ret != 0) {
dev_err(&i2c->dev, "Failed to register CODEC: %d\n", ret);
- goto err_regmap;
+ return ret;
}
return 0;
-
-err_regmap:
- regmap_exit(wm8731->regmap);
-err:
- kfree(wm8731);
- return ret;
}
static __devexit int wm8731_i2c_remove(struct i2c_client *client)
{
- struct wm8731_priv *wm8731 = i2c_get_clientdata(client);
snd_soc_unregister_codec(&client->dev);
- regmap_exit(wm8731->regmap);
- kfree(wm8731);
return 0;
}
diff --git a/sound/soc/codecs/wm8737.c b/sound/soc/codecs/wm8737.c
index 4fe9d191e27..d0520124616 100644
--- a/sound/soc/codecs/wm8737.c
+++ b/sound/soc/codecs/wm8737.c
@@ -329,8 +329,7 @@ static int wm8737_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;
+ struct snd_soc_codec *codec = dai->codec;
struct wm8737_priv *wm8737 = snd_soc_codec_get_drvdata(codec);
int i;
u16 clocking = 0;
diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c
index 3941f50bf18..6e849cb0424 100644
--- a/sound/soc/codecs/wm8741.c
+++ b/sound/soc/codecs/wm8741.c
@@ -203,8 +203,7 @@ static int wm8741_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;
+ struct snd_soc_codec *codec = dai->codec;
struct wm8741_priv *wm8741 = snd_soc_codec_get_drvdata(codec);
u16 iface = snd_soc_read(codec, WM8741_FORMAT_CONTROL) & 0x1FC;
int i;
diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c
index e4c50ce7d9c..89151ca5e77 100644
--- a/sound/soc/codecs/wm8750.c
+++ b/sound/soc/codecs/wm8750.c
@@ -547,8 +547,7 @@ static int wm8750_pcm_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;
+ struct snd_soc_codec *codec = dai->codec;
struct wm8750_priv *wm8750 = snd_soc_codec_get_drvdata(codec);
u16 iface = snd_soc_read(codec, WM8750_IFACE) & 0x1f3;
u16 srate = snd_soc_read(codec, WM8750_SRATE) & 0x1c0;
diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c
index e27e7b62b36..a26482cd765 100644
--- a/sound/soc/codecs/wm8753.c
+++ b/sound/soc/codecs/wm8753.c
@@ -931,8 +931,7 @@ static int wm8753_pcm_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;
+ struct snd_soc_codec *codec = dai->codec;
struct wm8753_priv *wm8753 = snd_soc_codec_get_drvdata(codec);
u16 voice = snd_soc_read(codec, WM8753_PCM) & 0x01f3;
u16 srate = snd_soc_read(codec, WM8753_SRATE1) & 0x017f;
@@ -1161,8 +1160,7 @@ static int wm8753_i2s_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;
+ struct snd_soc_codec *codec = dai->codec;
struct wm8753_priv *wm8753 = snd_soc_codec_get_drvdata(codec);
u16 srate = snd_soc_read(codec, WM8753_SRATE1) & 0x01c0;
u16 hifi = snd_soc_read(codec, WM8753_HIFI) & 0x01f3;
diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c
index f18c554efc9..077c9628c70 100644
--- a/sound/soc/codecs/wm8900.c
+++ b/sound/soc/codecs/wm8900.c
@@ -610,8 +610,7 @@ static int wm8900_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;
+ struct snd_soc_codec *codec = dai->codec;
u16 reg;
reg = snd_soc_read(codec, WM8900_REG_AUDIO1) & ~0x60;
diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c
index c91fb2f99c1..86b8a292659 100644
--- a/sound/soc/codecs/wm8903.c
+++ b/sound/soc/codecs/wm8903.c
@@ -1432,8 +1432,7 @@ static int wm8903_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;
+ struct snd_soc_codec *codec = dai->codec;
struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec);
int fs = params_rate(params);
int bclk;
diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c
index 4e190b5950b..812acd83fb4 100644
--- a/sound/soc/codecs/wm8904.c
+++ b/sound/soc/codecs/wm8904.c
@@ -1863,6 +1863,7 @@ static int wm8904_set_bias_level(struct snd_soc_codec *codec,
return ret;
}
+ regcache_cache_only(wm8904->regmap, false);
regcache_sync(wm8904->regmap);
/* Enable bias */
@@ -1899,14 +1900,8 @@ static int wm8904_set_bias_level(struct snd_soc_codec *codec,
snd_soc_update_bits(codec, WM8904_BIAS_CONTROL_0,
WM8904_BIAS_ENA, 0);
-#ifdef CONFIG_REGULATOR
- /* Post 2.6.34 we will be able to get a callback when
- * the regulators are disabled which we can use but
- * for now just assume that the power will be cut if
- * the regulator API is in use.
- */
- codec->cache_sync = 1;
-#endif
+ regcache_cache_only(wm8904->regmap, true);
+ regcache_mark_dirty(wm8904->regmap);
regulator_bulk_disable(ARRAY_SIZE(wm8904->supplies),
wm8904->supplies);
@@ -2086,7 +2081,6 @@ static int wm8904_probe(struct snd_soc_codec *codec)
struct wm8904_pdata *pdata = wm8904->pdata;
int ret, i;
- codec->cache_sync = 1;
codec->control_data = wm8904->regmap;
switch (wm8904->devtype) {
@@ -2149,6 +2143,7 @@ static int wm8904_probe(struct snd_soc_codec *codec)
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);
diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c
index d2883affea3..481a3d9cfe4 100644
--- a/sound/soc/codecs/wm8940.c
+++ b/sound/soc/codecs/wm8940.c
@@ -371,8 +371,7 @@ static int wm8940_i2s_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;
+ struct snd_soc_codec *codec = dai->codec;
u16 iface = snd_soc_read(codec, WM8940_IFACE) & 0xFD9F;
u16 addcntrl = snd_soc_read(codec, WM8940_ADDCNTRL) & 0xFFF1;
u16 companding = snd_soc_read(codec,
diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c
index 840d72086d0..8bc659d8dd2 100644
--- a/sound/soc/codecs/wm8960.c
+++ b/sound/soc/codecs/wm8960.c
@@ -505,8 +505,7 @@ static int wm8960_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;
+ struct snd_soc_codec *codec = dai->codec;
struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec);
u16 iface = snd_soc_read(codec, WM8960_IFACE1) & 0xfff3;
int i;
diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index 15d467ff91b..21a6727bfc2 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -1478,7 +1478,8 @@ static const DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0);
static int wm8962_dsp2_write_config(struct snd_soc_codec *codec)
{
- return 0;
+ return regcache_sync_region(codec->control_data,
+ WM8962_HDBASS_AI_1, WM8962_MAX_REGISTER);
}
static int wm8962_dsp2_set_enable(struct snd_soc_codec *codec, u16 val)
@@ -1755,10 +1756,22 @@ SOC_DOUBLE_R_TLV("EQ4 Volume", WM8962_EQ3, WM8962_EQ23,
SOC_DOUBLE_R_TLV("EQ5 Volume", WM8962_EQ3, WM8962_EQ23,
WM8962_EQL_B5_GAIN_SHIFT, 31, 0, eq_tlv),
+SOC_SINGLE("3D Switch", WM8962_THREED1, 0, 1, 0),
+SND_SOC_BYTES_MASK("3D Coefficients", WM8962_THREED1, 4, WM8962_THREED_ENA),
+
+SOC_SINGLE("DF1 Switch", WM8962_DF1, 0, 1, 0),
+SND_SOC_BYTES_MASK("DF1 Coefficients", WM8962_DF1, 7, WM8962_DF1_ENA),
+
+SOC_SINGLE("DRC Switch", WM8962_DRC_1, 0, 1, 0),
+SND_SOC_BYTES_MASK("DRC Coefficients", WM8962_DRC_1, 5, WM8962_DRC_ENA),
+
WM8962_DSP2_ENABLE("VSS Switch", WM8962_VSS_ENA_SHIFT),
+SND_SOC_BYTES("VSS Coefficients", WM8962_VSS_XHD2_1, 148),
WM8962_DSP2_ENABLE("HPF1 Switch", WM8962_HPF1_ENA_SHIFT),
WM8962_DSP2_ENABLE("HPF2 Switch", WM8962_HPF2_ENA_SHIFT),
+SND_SOC_BYTES("HPF Coefficients", WM8962_LHPF2, 1),
WM8962_DSP2_ENABLE("HD Bass Switch", WM8962_HDBASS_ENA_SHIFT),
+SND_SOC_BYTES("HD Bass Coefficients", WM8962_HDBASS_AI_1, 30),
};
static const struct snd_kcontrol_new wm8962_spk_mono_controls[] = {
@@ -2519,8 +2532,7 @@ static int wm8962_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;
+ struct snd_soc_codec *codec = dai->codec;
struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec);
int i;
int aif0 = 0;
@@ -3710,6 +3722,9 @@ 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/wm8971.c b/sound/soc/codecs/wm8971.c
index 28fe59e3ce0..eef783f6b6d 100644
--- a/sound/soc/codecs/wm8971.c
+++ b/sound/soc/codecs/wm8971.c
@@ -478,8 +478,7 @@ static int wm8971_pcm_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;
+ struct snd_soc_codec *codec = dai->codec;
struct wm8971_priv *wm8971 = snd_soc_codec_get_drvdata(codec);
u16 iface = snd_soc_read(codec, WM8971_IFACE) & 0x1f3;
u16 srate = snd_soc_read(codec, WM8971_SRATE) & 0x1c0;
diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c
index 72d5fdcd3cc..a5be3adecf7 100644
--- a/sound/soc/codecs/wm8978.c
+++ b/sound/soc/codecs/wm8978.c
@@ -723,8 +723,7 @@ static int wm8978_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;
+ struct snd_soc_codec *codec = dai->codec;
struct wm8978_priv *wm8978 = snd_soc_codec_get_drvdata(codec);
/* Word length mask = 0x60 */
u16 iface_ctl = snd_soc_read(codec, WM8978_AUDIO_INTERFACE) & ~0x60;
diff --git a/sound/soc/codecs/wm8988.c b/sound/soc/codecs/wm8988.c
index 6cdf6a2bc28..1d4c5cf47b0 100644
--- a/sound/soc/codecs/wm8988.c
+++ b/sound/soc/codecs/wm8988.c
@@ -668,8 +668,7 @@ static int wm8988_pcm_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;
+ struct snd_soc_codec *codec = dai->codec;
struct wm8988_priv *wm8988 = snd_soc_codec_get_drvdata(codec);
u16 iface = snd_soc_read(codec, WM8988_IFACE) & 0x1f3;
u16 srate = snd_soc_read(codec, WM8988_SRATE) & 0x180;
diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c
index 9d242351e6e..db63c97ddf5 100644
--- a/sound/soc/codecs/wm8990.c
+++ b/sound/soc/codecs/wm8990.c
@@ -1112,8 +1112,7 @@ static int wm8990_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;
+ struct snd_soc_codec *codec = dai->codec;
u16 audio1 = snd_soc_read(codec, WM8990_AUDIO_INTERFACE_1);
audio1 &= ~WM8990_AIF_WL_MASK;
diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c
index d256a934064..36acfccab99 100644
--- a/sound/soc/codecs/wm8993.c
+++ b/sound/soc/codecs/wm8993.c
@@ -218,7 +218,6 @@ struct wm8993_priv {
unsigned int sysclk_rate;
unsigned int fs;
unsigned int bclk;
- int class_w_users;
unsigned int fll_fref;
unsigned int fll_fout;
int fll_src;
@@ -824,84 +823,6 @@ static int clk_sys_event(struct snd_soc_dapm_widget *w,
return 0;
}
-/*
- * When used with DAC outputs only the WM8993 charge pump supports
- * operation in class W mode, providing very low power consumption
- * when used with digital sources. Enable and disable this mode
- * automatically depending on the mixer configuration.
- *
- * Currently the only supported paths are the direct DAC->headphone
- * paths (which provide minimum power consumption anyway).
- */
-static int class_w_put(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol);
- struct snd_soc_dapm_widget *widget = wlist->widgets[0];
- struct snd_soc_codec *codec = widget->codec;
- struct wm8993_priv *wm8993 = snd_soc_codec_get_drvdata(codec);
- int ret;
-
- /* Turn it off if we're using the main output mixer */
- if (ucontrol->value.integer.value[0] == 0) {
- if (wm8993->class_w_users == 0) {
- dev_dbg(codec->dev, "Disabling Class W\n");
- snd_soc_update_bits(codec, WM8993_CLASS_W_0,
- WM8993_CP_DYN_FREQ |
- WM8993_CP_DYN_V,
- 0);
- }
- wm8993->class_w_users++;
- wm8993->hubs_data.class_w = true;
- }
-
- /* Implement the change */
- ret = snd_soc_dapm_put_enum_double(kcontrol, ucontrol);
-
- /* Enable it if we're using the direct DAC path */
- if (ucontrol->value.integer.value[0] == 1) {
- if (wm8993->class_w_users == 1) {
- dev_dbg(codec->dev, "Enabling Class W\n");
- snd_soc_update_bits(codec, WM8993_CLASS_W_0,
- WM8993_CP_DYN_FREQ |
- WM8993_CP_DYN_V,
- WM8993_CP_DYN_FREQ |
- WM8993_CP_DYN_V);
- }
- wm8993->class_w_users--;
- wm8993->hubs_data.class_w = false;
- }
-
- dev_dbg(codec->dev, "Indirect DAC use count now %d\n",
- wm8993->class_w_users);
-
- return ret;
-}
-
-#define SOC_DAPM_ENUM_W(xname, xenum) \
-{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
- .info = snd_soc_info_enum_double, \
- .get = snd_soc_dapm_get_enum_double, \
- .put = class_w_put, \
- .private_value = (unsigned long)&xenum }
-
-static const char *hp_mux_text[] = {
- "Mixer",
- "DAC",
-};
-
-static const struct soc_enum hpl_enum =
- SOC_ENUM_SINGLE(WM8993_OUTPUT_MIXER1, 8, 2, hp_mux_text);
-
-static const struct snd_kcontrol_new hpl_mux =
- SOC_DAPM_ENUM_W("Left Headphone Mux", hpl_enum);
-
-static const struct soc_enum hpr_enum =
- SOC_ENUM_SINGLE(WM8993_OUTPUT_MIXER2, 8, 2, hp_mux_text);
-
-static const struct snd_kcontrol_new hpr_mux =
- SOC_DAPM_ENUM_W("Right Headphone Mux", hpr_enum);
-
static const struct snd_kcontrol_new left_speaker_mixer[] = {
SOC_DAPM_SINGLE("Input Switch", WM8993_SPEAKER_MIXER, 7, 1, 0),
SOC_DAPM_SINGLE("IN1LP Switch", WM8993_SPEAKER_MIXER, 5, 1, 0),
@@ -988,8 +909,8 @@ SND_SOC_DAPM_MUX("DACR Sidetone", SND_SOC_NOPM, 0, 0, &sidetoner_mux),
SND_SOC_DAPM_DAC("DACL", NULL, WM8993_POWER_MANAGEMENT_3, 1, 0),
SND_SOC_DAPM_DAC("DACR", NULL, WM8993_POWER_MANAGEMENT_3, 0, 0),
-SND_SOC_DAPM_MUX("Left Headphone Mux", SND_SOC_NOPM, 0, 0, &hpl_mux),
-SND_SOC_DAPM_MUX("Right Headphone Mux", SND_SOC_NOPM, 0, 0, &hpr_mux),
+SND_SOC_DAPM_MUX("Left Headphone Mux", SND_SOC_NOPM, 0, 0, &wm_hubs_hpl_mux),
+SND_SOC_DAPM_MUX("Right Headphone Mux", SND_SOC_NOPM, 0, 0, &wm_hubs_hpr_mux),
SND_SOC_DAPM_MIXER("SPKL", WM8993_POWER_MANAGEMENT_3, 8, 0,
left_speaker_mixer, ARRAY_SIZE(left_speaker_mixer)),
@@ -1579,9 +1500,6 @@ static int wm8993_probe(struct snd_soc_codec *codec)
return ret;
}
- /* By default we're using the output mixers */
- wm8993->class_w_users = 2;
-
/* Latch volume update bits and default ZC on */
snd_soc_update_bits(codec, WM8993_RIGHT_DAC_DIGITAL_VOLUME,
WM8993_DAC_VU, WM8993_DAC_VU);
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index f351b933f5c..1436b6ce74d 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -103,8 +103,8 @@ static const struct wm8958_micd_rate micdet_rates[] = {
static const struct wm8958_micd_rate jackdet_rates[] = {
{ 32768, true, 0, 1 },
{ 32768, false, 0, 1 },
- { 44100 * 256, true, 7, 10 },
- { 44100 * 256, false, 7, 10 },
+ { 44100 * 256, true, 10, 10 },
+ { 44100 * 256, false, 7, 8 },
};
static void wm8958_micd_set_rate(struct snd_soc_codec *codec)
@@ -115,7 +115,8 @@ static void wm8958_micd_set_rate(struct snd_soc_codec *codec)
const struct wm8958_micd_rate *rates;
int num_rates;
- if (wm8994->jack_cb != wm8958_default_micdet)
+ if (!(wm8994->pdata && wm8994->pdata->micd_rates) &&
+ wm8994->jack_cb != wm8958_default_micdet)
return;
idle = !wm8994->jack_mic;
@@ -151,6 +152,10 @@ static void wm8958_micd_set_rate(struct snd_soc_codec *codec)
val = rates[best].start << WM8958_MICD_BIAS_STARTTIME_SHIFT
| rates[best].rate << WM8958_MICD_RATE_SHIFT;
+ dev_dbg(codec->dev, "MICD rate %d,%d for %dHz %s\n",
+ rates[best].start, rates[best].rate, sysclk,
+ idle ? "idle" : "active");
+
snd_soc_update_bits(codec, WM8958_MIC_DETECT_1,
WM8958_MICD_BIAS_STARTTIME_MASK |
WM8958_MICD_RATE_MASK, val);
@@ -431,7 +436,7 @@ static void wm8994_set_retune_mobile(struct snd_soc_codec *codec, int block)
wm8994->dac_rates[iface]);
/* The EQ will be disabled while reconfiguring it, remember the
- * current configuration.
+ * current configuration.
*/
save = snd_soc_read(codec, base);
save &= WM8994_AIF1DAC1_EQ_ENA;
@@ -817,7 +822,7 @@ static void vmid_reference(struct snd_soc_codec *codec)
switch (wm8994->vmid_mode) {
default:
- WARN_ON(0 == "Invalid VMID mode");
+ WARN_ON(NULL == "Invalid VMID mode");
case WM8994_VMID_NORMAL:
/* Startup bias, VMID ramp & buffer */
snd_soc_update_bits(codec, WM8994_ANTIPOP_2,
@@ -970,27 +975,12 @@ static int vmid_event(struct snd_soc_dapm_widget *w,
return 0;
}
-static void wm8994_update_class_w(struct snd_soc_codec *codec)
+static bool wm8994_check_class_w_digital(struct snd_soc_codec *codec)
{
- struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
- int enable = 1;
int source = 0; /* GCC flow analysis can't track enable */
int reg, reg_r;
- /* Only support direct DAC->headphone paths */
- reg = snd_soc_read(codec, WM8994_OUTPUT_MIXER_1);
- if (!(reg & WM8994_DAC1L_TO_HPOUT1L)) {
- dev_vdbg(codec->dev, "HPL connected to output mixer\n");
- enable = 0;
- }
-
- reg = snd_soc_read(codec, WM8994_OUTPUT_MIXER_2);
- if (!(reg & WM8994_DAC1R_TO_HPOUT1R)) {
- dev_vdbg(codec->dev, "HPR connected to output mixer\n");
- enable = 0;
- }
-
- /* We also need the same setting for L/R and only one path */
+ /* We also need the same AIF source for L/R and only one path */
reg = snd_soc_read(codec, WM8994_DAC1_LEFT_MIXER_ROUTING);
switch (reg) {
case WM8994_AIF2DACL_TO_DAC1L:
@@ -1007,30 +997,20 @@ static void wm8994_update_class_w(struct snd_soc_codec *codec)
break;
default:
dev_vdbg(codec->dev, "DAC mixer setting: %x\n", reg);
- enable = 0;
- break;
+ return false;
}
reg_r = snd_soc_read(codec, WM8994_DAC1_RIGHT_MIXER_ROUTING);
if (reg_r != reg) {
dev_vdbg(codec->dev, "Left and right DAC mixers different\n");
- enable = 0;
+ return false;
}
- if (enable) {
- dev_dbg(codec->dev, "Class W enabled\n");
- snd_soc_update_bits(codec, WM8994_CLASS_W_1,
- WM8994_CP_DYN_PWR |
- WM8994_CP_DYN_SRC_SEL_MASK,
- source | WM8994_CP_DYN_PWR);
- wm8994->hubs.class_w = true;
-
- } else {
- dev_dbg(codec->dev, "Class W disabled\n");
- snd_soc_update_bits(codec, WM8994_CLASS_W_1,
- WM8994_CP_DYN_PWR, 0);
- wm8994->hubs.class_w = false;
- }
+ /* Set the source up */
+ snd_soc_update_bits(codec, WM8994_CLASS_W_1,
+ WM8994_CP_DYN_SRC_SEL_MASK, source);
+
+ return true;
}
static int aif1clk_ev(struct snd_soc_dapm_widget *w,
@@ -1333,45 +1313,6 @@ static int dac_ev(struct snd_soc_dapm_widget *w,
return 0;
}
-static const char *hp_mux_text[] = {
- "Mixer",
- "DAC",
-};
-
-#define WM8994_HP_ENUM(xname, xenum) \
-{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
- .info = snd_soc_info_enum_double, \
- .get = snd_soc_dapm_get_enum_double, \
- .put = wm8994_put_hp_enum, \
- .private_value = (unsigned long)&xenum }
-
-static int wm8994_put_hp_enum(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol);
- struct snd_soc_dapm_widget *w = wlist->widgets[0];
- struct snd_soc_codec *codec = w->codec;
- int ret;
-
- ret = snd_soc_dapm_put_enum_double(kcontrol, ucontrol);
-
- wm8994_update_class_w(codec);
-
- return ret;
-}
-
-static const struct soc_enum hpl_enum =
- SOC_ENUM_SINGLE(WM8994_OUTPUT_MIXER_1, 8, 2, hp_mux_text);
-
-static const struct snd_kcontrol_new hpl_mux =
- WM8994_HP_ENUM("Left Headphone Mux", hpl_enum);
-
-static const struct soc_enum hpr_enum =
- SOC_ENUM_SINGLE(WM8994_OUTPUT_MIXER_2, 8, 2, hp_mux_text);
-
-static const struct snd_kcontrol_new hpr_mux =
- WM8994_HP_ENUM("Right Headphone Mux", hpr_enum);
-
static const char *adc_mux_text[] = {
"ADC",
"DMIC",
@@ -1483,7 +1424,7 @@ static int wm8994_put_class_w(struct snd_kcontrol *kcontrol,
ret = snd_soc_dapm_put_volsw(kcontrol, ucontrol);
- wm8994_update_class_w(codec);
+ wm_hubs_update_class_w(codec);
return ret;
}
@@ -1577,7 +1518,7 @@ static const struct snd_kcontrol_new wm8958_aif3adc_mux =
SOC_DAPM_ENUM("AIF3ADC Mux", wm8958_aif3adc_enum);
static const char *mono_pcm_out_text[] = {
- "None", "AIF2ADCL", "AIF2ADCR",
+ "None", "AIF2ADCL", "AIF2ADCR",
};
static const struct soc_enum mono_pcm_out_enum =
@@ -1626,9 +1567,9 @@ SND_SOC_DAPM_MIXER_E("SPKL", WM8994_POWER_MANAGEMENT_3, 8, 0,
SND_SOC_DAPM_MIXER_E("SPKR", WM8994_POWER_MANAGEMENT_3, 9, 0,
right_speaker_mixer, ARRAY_SIZE(right_speaker_mixer),
late_enable_ev, SND_SOC_DAPM_PRE_PMU),
-SND_SOC_DAPM_MUX_E("Left Headphone Mux", SND_SOC_NOPM, 0, 0, &hpl_mux,
+SND_SOC_DAPM_MUX_E("Left Headphone Mux", SND_SOC_NOPM, 0, 0, &wm_hubs_hpl_mux,
late_enable_ev, SND_SOC_DAPM_PRE_PMU),
-SND_SOC_DAPM_MUX_E("Right Headphone Mux", SND_SOC_NOPM, 0, 0, &hpr_mux,
+SND_SOC_DAPM_MUX_E("Right Headphone Mux", SND_SOC_NOPM, 0, 0, &wm_hubs_hpr_mux,
late_enable_ev, SND_SOC_DAPM_PRE_PMU),
SND_SOC_DAPM_POST("Late Disable PGA", late_disable_ev)
@@ -1646,8 +1587,8 @@ SND_SOC_DAPM_MIXER("SPKL", WM8994_POWER_MANAGEMENT_3, 8, 0,
left_speaker_mixer, ARRAY_SIZE(left_speaker_mixer)),
SND_SOC_DAPM_MIXER("SPKR", WM8994_POWER_MANAGEMENT_3, 9, 0,
right_speaker_mixer, ARRAY_SIZE(right_speaker_mixer)),
-SND_SOC_DAPM_MUX("Left Headphone Mux", SND_SOC_NOPM, 0, 0, &hpl_mux),
-SND_SOC_DAPM_MUX("Right Headphone Mux", SND_SOC_NOPM, 0, 0, &hpr_mux),
+SND_SOC_DAPM_MUX("Left Headphone Mux", SND_SOC_NOPM, 0, 0, &wm_hubs_hpl_mux),
+SND_SOC_DAPM_MUX("Right Headphone Mux", SND_SOC_NOPM, 0, 0, &wm_hubs_hpr_mux),
};
static const struct snd_soc_dapm_widget wm8994_dac_revd_widgets[] = {
@@ -1787,6 +1728,7 @@ SND_SOC_DAPM_MUX("AIF3ADC Mux", SND_SOC_NOPM, 0, 0, &wm8994_aif3adc_mux),
};
static const struct snd_soc_dapm_widget wm8958_dapm_widgets[] = {
+SND_SOC_DAPM_SUPPLY("AIF3", WM8994_POWER_MANAGEMENT_6, 5, 1, NULL, 0),
SND_SOC_DAPM_MUX("Mono PCM Out Mux", SND_SOC_NOPM, 0, 0, &mono_pcm_out_mux),
SND_SOC_DAPM_MUX("AIF2DACL Mux", SND_SOC_NOPM, 0, 0, &aif2dacl_src_mux),
SND_SOC_DAPM_MUX("AIF2DACR Mux", SND_SOC_NOPM, 0, 0, &aif2dacr_src_mux),
@@ -2027,6 +1969,9 @@ static const struct snd_soc_dapm_route wm8958_intercon[] = {
{ "AIF2DACR Mux", "AIF2", "AIF2DAC Mux" },
{ "AIF2DACR Mux", "AIF3", "AIF3DACDAT" },
+ { "AIF3DACDAT", NULL, "AIF3" },
+ { "AIF3ADCDAT", NULL, "AIF3" },
+
{ "Mono PCM Out Mux", "AIF2ADCL", "AIF2ADCL" },
{ "Mono PCM Out Mux", "AIF2ADCR", "AIF2ADCR" },
@@ -2123,24 +2068,20 @@ static int _wm8994_set_fll(struct snd_soc_codec *codec, int id, int src,
struct wm8994 *control = wm8994->wm8994;
int reg_offset, ret;
struct fll_div fll;
- u16 reg, aif1, aif2;
+ u16 reg, clk1, aif_reg, aif_src;
unsigned long timeout;
bool was_enabled;
- aif1 = snd_soc_read(codec, WM8994_AIF1_CLOCKING_1)
- & WM8994_AIF1CLK_ENA;
-
- aif2 = snd_soc_read(codec, WM8994_AIF2_CLOCKING_1)
- & WM8994_AIF2CLK_ENA;
-
switch (id) {
case WM8994_FLL1:
reg_offset = 0;
id = 0;
+ aif_src = 0x10;
break;
case WM8994_FLL2:
reg_offset = 0x20;
id = 1;
+ aif_src = 0x18;
break;
default:
return -EINVAL;
@@ -2182,16 +2123,33 @@ static int _wm8994_set_fll(struct snd_soc_codec *codec, int id, int src,
if (ret < 0)
return ret;
- /* Gate the AIF clocks while we reclock */
- snd_soc_update_bits(codec, WM8994_AIF1_CLOCKING_1,
- WM8994_AIF1CLK_ENA, 0);
- snd_soc_update_bits(codec, WM8994_AIF2_CLOCKING_1,
- WM8994_AIF2CLK_ENA, 0);
+ /* Make sure that we're not providing SYSCLK right now */
+ clk1 = snd_soc_read(codec, WM8994_CLOCKING_1);
+ if (clk1 & WM8994_SYSCLK_SRC)
+ aif_reg = WM8994_AIF2_CLOCKING_1;
+ else
+ aif_reg = WM8994_AIF1_CLOCKING_1;
+ reg = snd_soc_read(codec, aif_reg);
+
+ if ((reg & WM8994_AIF1CLK_ENA) &&
+ (reg & WM8994_AIF1CLK_SRC_MASK) == aif_src) {
+ dev_err(codec->dev, "FLL%d is currently providing SYSCLK\n",
+ id + 1);
+ return -EBUSY;
+ }
/* We always need to disable the FLL while reconfiguring */
snd_soc_update_bits(codec, WM8994_FLL1_CONTROL_1 + reg_offset,
WM8994_FLL1_ENA, 0);
+ if (wm8994->fll_byp && src == WM8994_FLL_SRC_BCLK &&
+ freq_in == freq_out && freq_out) {
+ dev_dbg(codec->dev, "Bypassing FLL%d\n", id + 1);
+ snd_soc_update_bits(codec, WM8994_FLL1_CONTROL_5 + reg_offset,
+ WM8958_FLL1_BYP, WM8958_FLL1_BYP);
+ goto out;
+ }
+
reg = (fll.outdiv << WM8994_FLL1_OUTDIV_SHIFT) |
(fll.fll_fratio << WM8994_FLL1_FRATIO_SHIFT);
snd_soc_update_bits(codec, WM8994_FLL1_CONTROL_2 + reg_offset,
@@ -2206,6 +2164,7 @@ static int _wm8994_set_fll(struct snd_soc_codec *codec, int id, int src,
fll.n << WM8994_FLL1_N_SHIFT);
snd_soc_update_bits(codec, WM8994_FLL1_CONTROL_5 + reg_offset,
+ WM8958_FLL1_BYP |
WM8994_FLL1_REFCLK_DIV_MASK |
WM8994_FLL1_REFCLK_SRC_MASK,
(fll.clk_ref_div << WM8994_FLL1_REFCLK_DIV_SHIFT) |
@@ -2268,16 +2227,11 @@ static int _wm8994_set_fll(struct snd_soc_codec *codec, int id, int src,
}
}
+out:
wm8994->fll[id].in = freq_in;
wm8994->fll[id].out = freq_out;
wm8994->fll[id].src = src;
- /* Enable any gated AIF clocks */
- snd_soc_update_bits(codec, WM8994_AIF1_CLOCKING_1,
- WM8994_AIF1CLK_ENA, aif1);
- snd_soc_update_bits(codec, WM8994_AIF2_CLOCKING_1,
- WM8994_AIF2CLK_ENA, aif2);
-
configure_clock(codec);
return 0;
@@ -2345,7 +2299,7 @@ static int wm8994_set_dai_sysclk(struct snd_soc_dai *dai,
case WM8994_SYSCLK_OPCLK:
/* Special case - a division (times 10) is given and
- * no effect on main clocking.
+ * no effect on main clocking.
*/
if (freq) {
for (i = 0; i < ARRAY_SIZE(opclk_divs); i++)
@@ -2847,33 +2801,6 @@ static int wm8994_aif3_hw_params(struct snd_pcm_substream *substream,
return snd_soc_update_bits(codec, aif1_reg, WM8994_AIF1_WL_MASK, aif1);
}
-static void wm8994_aif_shutdown(struct snd_pcm_substream *substream,
- struct snd_soc_dai *dai)
-{
- struct snd_soc_codec *codec = dai->codec;
- int rate_reg = 0;
-
- switch (dai->id) {
- case 1:
- rate_reg = WM8994_AIF1_RATE;
- break;
- case 2:
- rate_reg = WM8994_AIF2_RATE;
- break;
- default:
- break;
- }
-
- /* If the DAI is idle then configure the divider tree for the
- * lowest output rate to save a little power if the clock is
- * still active (eg, because it is system clock).
- */
- if (rate_reg && !dai->playback_active && !dai->capture_active)
- snd_soc_update_bits(codec, rate_reg,
- WM8994_AIF1_SR_MASK |
- WM8994_AIF1CLK_RATE_MASK, 0x9);
-}
-
static int wm8994_aif_mute(struct snd_soc_dai *codec_dai, int mute)
{
struct snd_soc_codec *codec = codec_dai->codec;
@@ -2915,10 +2842,6 @@ static int wm8994_set_tristate(struct snd_soc_dai *codec_dai, int tristate)
reg = WM8994_AIF2_MASTER_SLAVE;
mask = WM8994_AIF2_TRI;
break;
- case 3:
- reg = WM8994_POWER_MANAGEMENT_6;
- mask = WM8994_AIF3_TRI;
- break;
default:
return -EINVAL;
}
@@ -2955,7 +2878,6 @@ static const struct snd_soc_dai_ops wm8994_aif1_dai_ops = {
.set_sysclk = wm8994_set_dai_sysclk,
.set_fmt = wm8994_set_dai_fmt,
.hw_params = wm8994_hw_params,
- .shutdown = wm8994_aif_shutdown,
.digital_mute = wm8994_aif_mute,
.set_pll = wm8994_set_fll,
.set_tristate = wm8994_set_tristate,
@@ -2965,7 +2887,6 @@ static const struct snd_soc_dai_ops wm8994_aif2_dai_ops = {
.set_sysclk = wm8994_set_dai_sysclk,
.set_fmt = wm8994_set_dai_fmt,
.hw_params = wm8994_hw_params,
- .shutdown = wm8994_aif_shutdown,
.digital_mute = wm8994_aif_mute,
.set_pll = wm8994_set_fll,
.set_tristate = wm8994_set_tristate,
@@ -2973,7 +2894,6 @@ static const struct snd_soc_dai_ops wm8994_aif2_dai_ops = {
static const struct snd_soc_dai_ops wm8994_aif3_dai_ops = {
.hw_params = wm8994_aif3_hw_params,
- .set_tristate = wm8994_set_tristate,
};
static struct snd_soc_dai_driver wm8994_dai[] = {
@@ -3181,14 +3101,14 @@ static void wm8994_handle_retune_mobile_pdata(struct wm8994_priv *wm8994)
/* Expand the array... */
t = krealloc(wm8994->retune_mobile_texts,
- sizeof(char *) *
+ sizeof(char *) *
(wm8994->num_retune_mobile_texts + 1),
GFP_KERNEL);
if (t == NULL)
continue;
/* ...store the new entry... */
- t[wm8994->num_retune_mobile_texts] =
+ t[wm8994->num_retune_mobile_texts] =
pdata->retune_mobile_cfgs[i].name;
/* ...and remember the new version. */
@@ -3359,25 +3279,25 @@ int wm8994_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack,
}
EXPORT_SYMBOL_GPL(wm8994_mic_detect);
-static irqreturn_t wm8994_mic_irq(int irq, void *data)
+static void wm8994_mic_work(struct work_struct *work)
{
- struct wm8994_priv *priv = data;
- struct snd_soc_codec *codec = priv->codec;
- int reg;
+ struct wm8994_priv *priv = container_of(work,
+ struct wm8994_priv,
+ mic_work.work);
+ struct regmap *regmap = priv->wm8994->regmap;
+ struct device *dev = priv->wm8994->dev;
+ unsigned int reg;
+ int ret;
int report;
-#ifndef CONFIG_SND_SOC_WM8994_MODULE
- trace_snd_soc_jack_irq(dev_name(codec->dev));
-#endif
-
- reg = snd_soc_read(codec, WM8994_INTERRUPT_RAW_STATUS_2);
- if (reg < 0) {
- dev_err(codec->dev, "Failed to read microphone status: %d\n",
- reg);
- return IRQ_HANDLED;
+ ret = regmap_read(regmap, WM8994_INTERRUPT_RAW_STATUS_2, &reg);
+ if (ret < 0) {
+ dev_err(dev, "Failed to read microphone status: %d\n",
+ ret);
+ return;
}
- dev_dbg(codec->dev, "Microphone status: %x\n", reg);
+ dev_dbg(dev, "Microphone status: %x\n", reg);
report = 0;
if (reg & WM8994_MIC1_DET_STS) {
@@ -3416,6 +3336,20 @@ static irqreturn_t wm8994_mic_irq(int irq, void *data)
snd_soc_jack_report(priv->micdet[1].jack, report,
SND_JACK_HEADSET | SND_JACK_BTN_0);
+}
+
+static irqreturn_t wm8994_mic_irq(int irq, void *data)
+{
+ struct wm8994_priv *priv = data;
+ struct snd_soc_codec *codec = priv->codec;
+
+#ifndef CONFIG_SND_SOC_WM8994_MODULE
+ trace_snd_soc_jack_irq(dev_name(codec->dev));
+#endif
+
+ pm_wakeup_event(codec->dev, 300);
+
+ schedule_delayed_work(&priv->mic_work, msecs_to_jiffies(250));
return IRQ_HANDLED;
}
@@ -3470,9 +3404,6 @@ static void wm8958_default_micdet(u16 status, void *data)
wm8958_micd_set_rate(codec);
- snd_soc_jack_report(wm8994->micdet[0].jack, SND_JACK_HEADPHONE,
- SND_JACK_HEADSET);
-
/* If we have jackdet that will detect removal */
if (wm8994->jackdet) {
mutex_lock(&wm8994->accdet_lock);
@@ -3485,14 +3416,13 @@ static void wm8958_default_micdet(u16 status, void *data)
mutex_unlock(&wm8994->accdet_lock);
- if (wm8994->pdata->jd_ext_cap) {
- mutex_lock(&codec->mutex);
+ if (wm8994->pdata->jd_ext_cap)
snd_soc_dapm_disable_pin(&codec->dapm,
"MICBIAS2");
- snd_soc_dapm_sync(&codec->dapm);
- mutex_unlock(&codec->mutex);
- }
}
+
+ snd_soc_jack_report(wm8994->micdet[0].jack, SND_JACK_HEADPHONE,
+ SND_JACK_HEADSET);
}
/* Report short circuit as a button */
@@ -3544,6 +3474,8 @@ static irqreturn_t wm1811_jackdet_irq(int irq, void *data)
if (present) {
dev_dbg(codec->dev, "Jack detected\n");
+ wm8958_micd_set_rate(codec);
+
snd_soc_update_bits(codec, WM8958_MICBIAS2,
WM8958_MICB2_DISCH, 0);
@@ -3581,16 +3513,11 @@ static irqreturn_t wm1811_jackdet_irq(int irq, void *data)
/* If required for an external cap force MICBIAS on */
if (wm8994->pdata->jd_ext_cap) {
- mutex_lock(&codec->mutex);
-
if (present)
snd_soc_dapm_force_enable_pin(&codec->dapm,
"MICBIAS2");
else
snd_soc_dapm_disable_pin(&codec->dapm, "MICBIAS2");
-
- snd_soc_dapm_sync(&codec->dapm);
- mutex_unlock(&codec->mutex);
}
if (present)
@@ -3795,6 +3722,7 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec)
wm8994->codec = codec;
mutex_init(&wm8994->accdet_lock);
+ INIT_DELAYED_WORK(&wm8994->mic_work, wm8994_mic_work);
for (i = 0; i < ARRAY_SIZE(wm8994->fll_locked); i++)
init_completion(&wm8994->fll_locked[i]);
@@ -3838,13 +3766,22 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec)
case WM8958:
wm8994->hubs.dcs_readback_mode = 1;
wm8994->hubs.hp_startup_mode = 1;
+
+ switch (wm8994->revision) {
+ case 0:
+ break;
+ default:
+ wm8994->fll_byp = true;
+ break;
+ }
break;
case WM1811:
wm8994->hubs.dcs_readback_mode = 2;
wm8994->hubs.no_series_update = 1;
wm8994->hubs.hp_startup_mode = 1;
- wm8994->hubs.no_cache_class_w = true;
+ wm8994->hubs.no_cache_dac_hp_direct = true;
+ wm8994->fll_byp = true;
switch (wm8994->revision) {
case 0:
@@ -4037,7 +3974,8 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec)
break;
}
- wm8994_update_class_w(codec);
+ wm8994->hubs.check_class_w_digital = wm8994_check_class_w_digital;
+ wm_hubs_update_class_w(codec);
wm8994_handle_pdata(wm8994);
@@ -4102,7 +4040,6 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec)
ARRAY_SIZE(wm8994_dac_widgets));
break;
}
-
wm_hubs_add_analogue_routes(codec, 0, 0);
snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon));
@@ -4167,7 +4104,7 @@ err_irq:
return ret;
}
-static int wm8994_codec_remove(struct snd_soc_codec *codec)
+static int wm8994_codec_remove(struct snd_soc_codec *codec)
{
struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
struct wm8994 *control = wm8994->wm8994;
@@ -4208,14 +4145,10 @@ static int wm8994_codec_remove(struct snd_soc_codec *codec)
free_irq(wm8994->micdet_irq, wm8994);
break;
}
- if (wm8994->mbc)
- release_firmware(wm8994->mbc);
- if (wm8994->mbc_vss)
- release_firmware(wm8994->mbc_vss);
- if (wm8994->enh_eq)
- release_firmware(wm8994->enh_eq);
+ release_firmware(wm8994->mbc);
+ release_firmware(wm8994->mbc_vss);
+ release_firmware(wm8994->enh_eq);
kfree(wm8994->retune_mobile_texts);
-
return 0;
}
diff --git a/sound/soc/codecs/wm8994.h b/sound/soc/codecs/wm8994.h
index c724112998d..d77e06f0a67 100644
--- a/sound/soc/codecs/wm8994.h
+++ b/sound/soc/codecs/wm8994.h
@@ -12,6 +12,7 @@
#include <sound/soc.h>
#include <linux/firmware.h>
#include <linux/completion.h>
+#include <linux/workqueue.h>
#include "wm_hubs.h"
@@ -79,6 +80,7 @@ struct wm8994_priv {
struct wm8994_fll_config fll[2], fll_suspend[2];
struct completion fll_locked[2];
bool fll_locked_irq;
+ bool fll_byp;
int vmid_refcount;
int active_refcount;
@@ -126,6 +128,7 @@ struct wm8994_priv {
struct mutex accdet_lock;
struct wm8994_micdet micdet[2];
+ struct delayed_work mic_work;
bool mic_detecting;
bool jack_mic;
int btn_mask;
diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c
index 1fd63549404..dc9b42b7fc4 100644
--- a/sound/soc/codecs/wm8996.c
+++ b/sound/soc/codecs/wm8996.c
@@ -1770,7 +1770,13 @@ static int wm8996_set_bias_level(struct snd_soc_codec *codec,
switch (level) {
case SND_SOC_BIAS_ON:
+ break;
case SND_SOC_BIAS_PREPARE:
+ /* Put the MICBIASes into regulating mode */
+ snd_soc_update_bits(codec, WM8996_MICBIAS_1,
+ WM8996_MICB1_MODE, 0);
+ snd_soc_update_bits(codec, WM8996_MICBIAS_2,
+ WM8996_MICB2_MODE, 0);
break;
case SND_SOC_BIAS_STANDBY:
@@ -1793,6 +1799,12 @@ static int wm8996_set_bias_level(struct snd_soc_codec *codec,
regcache_cache_only(codec->control_data, false);
regcache_sync(codec->control_data);
}
+
+ /* Bypass the MICBIASes for lowest power */
+ snd_soc_update_bits(codec, WM8996_MICBIAS_1,
+ WM8996_MICB1_MODE, WM8996_MICB1_MODE);
+ snd_soc_update_bits(codec, WM8996_MICBIAS_2,
+ WM8996_MICB2_MODE, WM8996_MICB2_MODE);
break;
case SND_SOC_BIAS_OFF:
@@ -2825,8 +2837,6 @@ static int wm8996_probe(struct snd_soc_codec *codec)
}
}
- regcache_cache_only(codec->control_data, true);
-
/* Apply platform data settings */
snd_soc_update_bits(codec, WM8996_LINE_INPUT_CONTROL,
WM8996_INL_MODE_MASK | WM8996_INR_MODE_MASK,
@@ -3039,7 +3049,6 @@ static int wm8996_remove(struct snd_soc_codec *codec)
for (i = 0; i < ARRAY_SIZE(wm8996->supplies); i++)
regulator_unregister_notifier(wm8996->supplies[i].consumer,
&wm8996->disable_nb[i]);
- regulator_bulk_free(ARRAY_SIZE(wm8996->supplies), wm8996->supplies);
return 0;
}
@@ -3194,14 +3203,15 @@ static __devinit int wm8996_i2c_probe(struct i2c_client *i2c,
dev_info(&i2c->dev, "revision %c\n",
(reg & WM8996_CHIP_REV_MASK) + 'A');
- regulator_bulk_disable(ARRAY_SIZE(wm8996->supplies), wm8996->supplies);
-
ret = wm8996_reset(wm8996);
if (ret < 0) {
dev_err(&i2c->dev, "Failed to issue reset\n");
goto err_regmap;
}
+ regcache_cache_only(wm8996->regmap, true);
+ regulator_bulk_disable(ARRAY_SIZE(wm8996->supplies), wm8996->supplies);
+
wm8996_init_gpio(wm8996);
ret = snd_soc_register_codec(&i2c->dev,
diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c
index 076c126ed9b..9328270df16 100644
--- a/sound/soc/codecs/wm9081.c
+++ b/sound/soc/codecs/wm9081.c
@@ -774,7 +774,7 @@ static const struct snd_soc_dapm_widget wm9081_dapm_widgets[] = {
SND_SOC_DAPM_INPUT("IN1"),
SND_SOC_DAPM_INPUT("IN2"),
-SND_SOC_DAPM_DAC("DAC", "HiFi Playback", WM9081_POWER_MANAGEMENT, 0, 0),
+SND_SOC_DAPM_DAC("DAC", NULL, WM9081_POWER_MANAGEMENT, 0, 0),
SND_SOC_DAPM_MIXER_NAMED_CTL("Mixer", SND_SOC_NOPM, 0, 0,
mixer, ARRAY_SIZE(mixer)),
@@ -799,6 +799,7 @@ SND_SOC_DAPM_SUPPLY("TSENSE", WM9081_POWER_MANAGEMENT, 7, 0, NULL, 0),
static const struct snd_soc_dapm_route wm9081_audio_paths[] = {
{ "DAC", NULL, "CLK_SYS" },
{ "DAC", NULL, "CLK_DSP" },
+ { "DAC", NULL, "AIF" },
{ "Mixer", "IN1 Switch", "IN1" },
{ "Mixer", "IN2 Switch", "IN2" },
@@ -1252,7 +1253,7 @@ static const struct snd_soc_dai_ops wm9081_dai_ops = {
static struct snd_soc_dai_driver wm9081_dai = {
.name = "wm9081-hifi",
.playback = {
- .stream_name = "HiFi Playback",
+ .stream_name = "AIF",
.channels_min = 1,
.channels_max = 2,
.rates = WM9081_RATES,
diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c
index cacc6a86b46..e8e782a0c78 100644
--- a/sound/soc/codecs/wm9705.c
+++ b/sound/soc/codecs/wm9705.c
@@ -236,9 +236,7 @@ static int ac97_write(struct snd_soc_codec *codec, unsigned int reg,
static int ac97_prepare(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
- struct snd_pcm_runtime *runtime = substream->runtime;
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_codec *codec = dai->codec;
int reg;
u16 vra;
@@ -250,7 +248,7 @@ static int ac97_prepare(struct snd_pcm_substream *substream,
else
reg = AC97_PCM_LR_ADC_RATE;
- return ac97_write(codec, reg, runtime->rate);
+ return ac97_write(codec, reg, substream->runtime->rate);
}
#define WM9705_AC97_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | \
diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c
index b342ae50bcd..a1541414d90 100644
--- a/sound/soc/codecs/wm9712.c
+++ b/sound/soc/codecs/wm9712.c
@@ -467,11 +467,10 @@ static int ac97_write(struct snd_soc_codec *codec, unsigned int reg,
static int ac97_prepare(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
- struct snd_pcm_runtime *runtime = substream->runtime;
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_codec *codec =rtd->codec;
+ struct snd_soc_codec *codec = dai->codec;
int reg;
u16 vra;
+ struct snd_pcm_runtime *runtime = substream->runtime;
vra = ac97_read(codec, AC97_EXTENDED_STATUS);
ac97_write(codec, AC97_EXTENDED_STATUS, vra | 0x1);
@@ -487,10 +486,9 @@ static int ac97_prepare(struct snd_pcm_substream *substream,
static int ac97_aux_prepare(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
- struct snd_pcm_runtime *runtime = substream->runtime;
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_codec *codec = dai->codec;
u16 vra, xsle;
+ struct snd_pcm_runtime *runtime = substream->runtime;
vra = ac97_read(codec, AC97_EXTENDED_STATUS);
ac97_write(codec, AC97_EXTENDED_STATUS, vra | 0x1);
diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c
index 6c028c47060..dfe957a47f2 100644
--- a/sound/soc/codecs/wm_hubs.c
+++ b/sound/soc/codecs/wm_hubs.c
@@ -109,12 +109,103 @@ irqreturn_t wm_hubs_dcs_done(int irq, void *data)
}
EXPORT_SYMBOL_GPL(wm_hubs_dcs_done);
+static bool wm_hubs_dac_hp_direct(struct snd_soc_codec *codec)
+{
+ int reg;
+
+ /* If we're going via the mixer we'll need to do additional checks */
+ reg = snd_soc_read(codec, WM8993_OUTPUT_MIXER1);
+ if (!(reg & WM8993_DACL_TO_HPOUT1L)) {
+ if (reg & ~WM8993_DACL_TO_MIXOUTL) {
+ dev_vdbg(codec->dev, "Analogue paths connected: %x\n",
+ reg & ~WM8993_DACL_TO_HPOUT1L);
+ return false;
+ } else {
+ dev_vdbg(codec->dev, "HPL connected to mixer\n");
+ }
+ } else {
+ dev_vdbg(codec->dev, "HPL connected to DAC\n");
+ }
+
+ reg = snd_soc_read(codec, WM8993_OUTPUT_MIXER2);
+ if (!(reg & WM8993_DACR_TO_HPOUT1R)) {
+ if (reg & ~WM8993_DACR_TO_MIXOUTR) {
+ dev_vdbg(codec->dev, "Analogue paths connected: %x\n",
+ reg & ~WM8993_DACR_TO_HPOUT1R);
+ return false;
+ } else {
+ dev_vdbg(codec->dev, "HPR connected to mixer\n");
+ }
+ } else {
+ dev_vdbg(codec->dev, "HPR connected to DAC\n");
+ }
+
+ return true;
+}
+
+struct wm_hubs_dcs_cache {
+ struct list_head list;
+ unsigned int left;
+ unsigned int right;
+ u16 dcs_cfg;
+};
+
+static bool wm_hubs_dcs_cache_get(struct snd_soc_codec *codec,
+ struct wm_hubs_dcs_cache **entry)
+{
+ struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec);
+ struct wm_hubs_dcs_cache *cache;
+ unsigned int left, right;
+
+ left = snd_soc_read(codec, WM8993_LEFT_OUTPUT_VOLUME);
+ left &= WM8993_HPOUT1L_VOL_MASK;
+
+ right = snd_soc_read(codec, WM8993_RIGHT_OUTPUT_VOLUME);
+ right &= WM8993_HPOUT1R_VOL_MASK;
+
+ list_for_each_entry(cache, &hubs->dcs_cache, list) {
+ if (cache->left != left || cache->right != right)
+ continue;
+
+ *entry = cache;
+ return true;
+ }
+
+ return false;
+}
+
+static void wm_hubs_dcs_cache_set(struct snd_soc_codec *codec, u16 dcs_cfg)
+{
+ struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec);
+ struct wm_hubs_dcs_cache *cache;
+
+ if (hubs->no_cache_dac_hp_direct)
+ return;
+
+ cache = devm_kzalloc(codec->dev, sizeof(*cache), GFP_KERNEL);
+ if (!cache) {
+ dev_err(codec->dev, "Failed to allocate DCS cache entry\n");
+ return;
+ }
+
+ cache->left = snd_soc_read(codec, WM8993_LEFT_OUTPUT_VOLUME);
+ cache->left &= WM8993_HPOUT1L_VOL_MASK;
+
+ cache->right = snd_soc_read(codec, WM8993_RIGHT_OUTPUT_VOLUME);
+ cache->right &= WM8993_HPOUT1R_VOL_MASK;
+
+ cache->dcs_cfg = dcs_cfg;
+
+ list_add_tail(&cache->list, &hubs->dcs_cache);
+}
+
/*
* Startup calibration of the DC servo
*/
static void calibrate_dc_servo(struct snd_soc_codec *codec)
{
struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec);
+ struct wm_hubs_dcs_cache *cache;
s8 offset;
u16 reg, reg_l, reg_r, dcs_cfg, dcs_reg;
@@ -129,10 +220,11 @@ static void calibrate_dc_servo(struct snd_soc_codec *codec)
/* If we're using a digital only path and have a previously
* callibrated DC servo offset stored then use that. */
- if (hubs->class_w && hubs->class_w_dcs) {
- dev_dbg(codec->dev, "Using cached DC servo offset %x\n",
- hubs->class_w_dcs);
- snd_soc_write(codec, dcs_reg, hubs->class_w_dcs);
+ if (wm_hubs_dac_hp_direct(codec) &&
+ wm_hubs_dcs_cache_get(codec, &cache)) {
+ dev_dbg(codec->dev, "Using cached DCS offset %x for %d,%d\n",
+ cache->dcs_cfg, cache->left, cache->right);
+ snd_soc_write(codec, dcs_reg, cache->dcs_cfg);
wait_for_dc_servo(codec,
WM8993_DCS_TRIG_DAC_WR_0 |
WM8993_DCS_TRIG_DAC_WR_1);
@@ -207,8 +299,8 @@ static void calibrate_dc_servo(struct snd_soc_codec *codec)
/* Save the callibrated offset if we're in class W mode and
* therefore don't have any analogue signal mixed in. */
- if (hubs->class_w && !hubs->no_cache_class_w)
- hubs->class_w_dcs = dcs_cfg;
+ if (wm_hubs_dac_hp_direct(codec))
+ wm_hubs_dcs_cache_set(codec, dcs_cfg);
}
/*
@@ -223,9 +315,6 @@ static int wm8993_put_dc_servo(struct snd_kcontrol *kcontrol,
ret = snd_soc_put_volsw(kcontrol, ucontrol);
- /* Updating the analogue gains invalidates the DC servo cache */
- hubs->class_w_dcs = 0;
-
/* If we're applying an offset correction then updating the
* callibration would be likely to introduce further offsets. */
if (hubs->dcs_codes_l || hubs->dcs_codes_r || hubs->no_series_update)
@@ -530,6 +619,86 @@ static int lineout_event(struct snd_soc_dapm_widget *w,
return 0;
}
+void wm_hubs_update_class_w(struct snd_soc_codec *codec)
+{
+ struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec);
+ int enable = WM8993_CP_DYN_V | WM8993_CP_DYN_FREQ;
+
+ if (!wm_hubs_dac_hp_direct(codec))
+ enable = false;
+
+ if (hubs->check_class_w_digital && !hubs->check_class_w_digital(codec))
+ enable = false;
+
+ dev_vdbg(codec->dev, "Class W %s\n", enable ? "enabled" : "disabled");
+
+ snd_soc_update_bits(codec, WM8993_CLASS_W_0,
+ WM8993_CP_DYN_V | WM8993_CP_DYN_FREQ, enable);
+}
+EXPORT_SYMBOL_GPL(wm_hubs_update_class_w);
+
+#define WM_HUBS_SINGLE_W(xname, reg, shift, max, invert) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
+ .info = snd_soc_info_volsw, \
+ .get = snd_soc_dapm_get_volsw, .put = class_w_put_volsw, \
+ .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert) }
+
+static int class_w_put_volsw(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol);
+ struct snd_soc_dapm_widget *widget = wlist->widgets[0];
+ struct snd_soc_codec *codec = widget->codec;
+ int ret;
+
+ ret = snd_soc_dapm_put_volsw(kcontrol, ucontrol);
+
+ wm_hubs_update_class_w(codec);
+
+ return ret;
+}
+
+#define WM_HUBS_ENUM_W(xname, xenum) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
+ .info = snd_soc_info_enum_double, \
+ .get = snd_soc_dapm_get_enum_double, \
+ .put = class_w_put_double, \
+ .private_value = (unsigned long)&xenum }
+
+static int class_w_put_double(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol);
+ struct snd_soc_dapm_widget *widget = wlist->widgets[0];
+ struct snd_soc_codec *codec = widget->codec;
+ int ret;
+
+ ret = snd_soc_dapm_put_enum_double(kcontrol, ucontrol);
+
+ wm_hubs_update_class_w(codec);
+
+ return ret;
+}
+
+static const char *hp_mux_text[] = {
+ "Mixer",
+ "DAC",
+};
+
+static const struct soc_enum hpl_enum =
+ SOC_ENUM_SINGLE(WM8993_OUTPUT_MIXER1, 8, 2, hp_mux_text);
+
+const struct snd_kcontrol_new wm_hubs_hpl_mux =
+ WM_HUBS_ENUM_W("Left Headphone Mux", hpl_enum);
+EXPORT_SYMBOL_GPL(wm_hubs_hpl_mux);
+
+static const struct soc_enum hpr_enum =
+ SOC_ENUM_SINGLE(WM8993_OUTPUT_MIXER2, 8, 2, hp_mux_text);
+
+const struct snd_kcontrol_new wm_hubs_hpr_mux =
+ WM_HUBS_ENUM_W("Right Headphone Mux", hpr_enum);
+EXPORT_SYMBOL_GPL(wm_hubs_hpr_mux);
+
static const struct snd_kcontrol_new in1l_pga[] = {
SOC_DAPM_SINGLE("IN1LP Switch", WM8993_INPUT_MIXER2, 5, 1, 0),
SOC_DAPM_SINGLE("IN1LN Switch", WM8993_INPUT_MIXER2, 4, 1, 0),
@@ -561,25 +730,25 @@ SOC_DAPM_SINGLE("IN1R Switch", WM8993_INPUT_MIXER4, 5, 1, 0),
};
static const struct snd_kcontrol_new left_output_mixer[] = {
-SOC_DAPM_SINGLE("Right Input Switch", WM8993_OUTPUT_MIXER1, 7, 1, 0),
-SOC_DAPM_SINGLE("Left Input Switch", WM8993_OUTPUT_MIXER1, 6, 1, 0),
-SOC_DAPM_SINGLE("IN2RN Switch", WM8993_OUTPUT_MIXER1, 5, 1, 0),
-SOC_DAPM_SINGLE("IN2LN Switch", WM8993_OUTPUT_MIXER1, 4, 1, 0),
-SOC_DAPM_SINGLE("IN2LP Switch", WM8993_OUTPUT_MIXER1, 1, 1, 0),
-SOC_DAPM_SINGLE("IN1R Switch", WM8993_OUTPUT_MIXER1, 3, 1, 0),
-SOC_DAPM_SINGLE("IN1L Switch", WM8993_OUTPUT_MIXER1, 2, 1, 0),
-SOC_DAPM_SINGLE("DAC Switch", WM8993_OUTPUT_MIXER1, 0, 1, 0),
+WM_HUBS_SINGLE_W("Right Input Switch", WM8993_OUTPUT_MIXER1, 7, 1, 0),
+WM_HUBS_SINGLE_W("Left Input Switch", WM8993_OUTPUT_MIXER1, 6, 1, 0),
+WM_HUBS_SINGLE_W("IN2RN Switch", WM8993_OUTPUT_MIXER1, 5, 1, 0),
+WM_HUBS_SINGLE_W("IN2LN Switch", WM8993_OUTPUT_MIXER1, 4, 1, 0),
+WM_HUBS_SINGLE_W("IN2LP Switch", WM8993_OUTPUT_MIXER1, 1, 1, 0),
+WM_HUBS_SINGLE_W("IN1R Switch", WM8993_OUTPUT_MIXER1, 3, 1, 0),
+WM_HUBS_SINGLE_W("IN1L Switch", WM8993_OUTPUT_MIXER1, 2, 1, 0),
+WM_HUBS_SINGLE_W("DAC Switch", WM8993_OUTPUT_MIXER1, 0, 1, 0),
};
static const struct snd_kcontrol_new right_output_mixer[] = {
-SOC_DAPM_SINGLE("Left Input Switch", WM8993_OUTPUT_MIXER2, 7, 1, 0),
-SOC_DAPM_SINGLE("Right Input Switch", WM8993_OUTPUT_MIXER2, 6, 1, 0),
-SOC_DAPM_SINGLE("IN2LN Switch", WM8993_OUTPUT_MIXER2, 5, 1, 0),
-SOC_DAPM_SINGLE("IN2RN Switch", WM8993_OUTPUT_MIXER2, 4, 1, 0),
-SOC_DAPM_SINGLE("IN1L Switch", WM8993_OUTPUT_MIXER2, 3, 1, 0),
-SOC_DAPM_SINGLE("IN1R Switch", WM8993_OUTPUT_MIXER2, 2, 1, 0),
-SOC_DAPM_SINGLE("IN2RP Switch", WM8993_OUTPUT_MIXER2, 1, 1, 0),
-SOC_DAPM_SINGLE("DAC Switch", WM8993_OUTPUT_MIXER2, 0, 1, 0),
+WM_HUBS_SINGLE_W("Left Input Switch", WM8993_OUTPUT_MIXER2, 7, 1, 0),
+WM_HUBS_SINGLE_W("Right Input Switch", WM8993_OUTPUT_MIXER2, 6, 1, 0),
+WM_HUBS_SINGLE_W("IN2LN Switch", WM8993_OUTPUT_MIXER2, 5, 1, 0),
+WM_HUBS_SINGLE_W("IN2RN Switch", WM8993_OUTPUT_MIXER2, 4, 1, 0),
+WM_HUBS_SINGLE_W("IN1L Switch", WM8993_OUTPUT_MIXER2, 3, 1, 0),
+WM_HUBS_SINGLE_W("IN1R Switch", WM8993_OUTPUT_MIXER2, 2, 1, 0),
+WM_HUBS_SINGLE_W("IN2RP Switch", WM8993_OUTPUT_MIXER2, 1, 1, 0),
+WM_HUBS_SINGLE_W("DAC Switch", WM8993_OUTPUT_MIXER2, 0, 1, 0),
};
static const struct snd_kcontrol_new earpiece_mixer[] = {
@@ -943,6 +1112,7 @@ int wm_hubs_add_analogue_routes(struct snd_soc_codec *codec,
struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec);
struct snd_soc_dapm_context *dapm = &codec->dapm;
+ INIT_LIST_HEAD(&hubs->dcs_cache);
init_completion(&hubs->dcs_done);
snd_soc_dapm_add_routes(dapm, analogue_routes,
diff --git a/sound/soc/codecs/wm_hubs.h b/sound/soc/codecs/wm_hubs.h
index 5705276f494..da2dc899ce6 100644
--- a/sound/soc/codecs/wm_hubs.h
+++ b/sound/soc/codecs/wm_hubs.h
@@ -16,6 +16,8 @@
#include <linux/completion.h>
#include <linux/interrupt.h>
+#include <linux/list.h>
+#include <sound/control.h>
struct snd_soc_codec;
@@ -30,9 +32,9 @@ struct wm_hubs_data {
int series_startup;
int no_series_update;
- bool no_cache_class_w;
- bool class_w;
- u16 class_w_dcs;
+ bool no_cache_dac_hp_direct;
+ struct list_head dcs_cache;
+ bool (*check_class_w_digital)(struct snd_soc_codec *);
bool lineout1_se;
bool lineout1n_ena;
@@ -58,5 +60,9 @@ extern irqreturn_t wm_hubs_dcs_done(int irq, void *data);
extern void wm_hubs_vmid_ena(struct snd_soc_codec *codec);
extern void wm_hubs_set_bias_level(struct snd_soc_codec *codec,
enum snd_soc_bias_level level);
+extern void wm_hubs_update_class_w(struct snd_soc_codec *codec);
+
+extern const struct snd_kcontrol_new wm_hubs_hpl_mux;
+extern const struct snd_kcontrol_new wm_hubs_hpr_mux;
#endif
diff --git a/sound/soc/ep93xx/ep93xx-ac97.c b/sound/soc/ep93xx/ep93xx-ac97.c
index 0678637abd6..bdffab33e16 100644
--- a/sound/soc/ep93xx/ep93xx-ac97.c
+++ b/sound/soc/ep93xx/ep93xx-ac97.c
@@ -87,17 +87,13 @@
* struct ep93xx_ac97_info - EP93xx AC97 controller info structure
* @lock: mutex serializing access to the bus (slot 1 & 2 ops)
* @dev: pointer to the platform device dev structure
- * @mem: physical memory resource for the registers
* @regs: mapped AC97 controller registers
- * @irq: AC97 interrupt number
* @done: bus ops wait here for an interrupt
*/
struct ep93xx_ac97_info {
struct mutex lock;
struct device *dev;
- struct resource *mem;
void __iomem *regs;
- int irq;
struct completion done;
};
@@ -359,66 +355,50 @@ static struct snd_soc_dai_driver ep93xx_ac97_dai = {
static int __devinit ep93xx_ac97_probe(struct platform_device *pdev)
{
struct ep93xx_ac97_info *info;
+ struct resource *res;
+ unsigned int irq;
int ret;
- info = kzalloc(sizeof(struct ep93xx_ac97_info), GFP_KERNEL);
+ info = devm_kzalloc(&pdev->dev, sizeof(*info), GFP_KERNEL);
if (!info)
return -ENOMEM;
- dev_set_drvdata(&pdev->dev, info);
-
- mutex_init(&info->lock);
- init_completion(&info->done);
- info->dev = &pdev->dev;
+ res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+ if (!res)
+ return -ENODEV;
- info->mem = platform_get_resource(pdev, IORESOURCE_MEM, 0);
- if (!info->mem) {
- ret = -ENXIO;
- goto fail_free_info;
- }
+ info->regs = devm_request_and_ioremap(&pdev->dev, res);
+ if (!info->regs)
+ return -ENXIO;
- info->irq = platform_get_irq(pdev, 0);
- if (!info->irq) {
- ret = -ENXIO;
- goto fail_free_info;
- }
+ irq = platform_get_irq(pdev, 0);
+ if (!irq)
+ return -ENODEV;
- if (!request_mem_region(info->mem->start, resource_size(info->mem),
- pdev->name)) {
- ret = -EBUSY;
- goto fail_free_info;
- }
+ ret = devm_request_irq(&pdev->dev, irq, ep93xx_ac97_interrupt,
+ IRQF_TRIGGER_HIGH, pdev->name, info);
+ if (ret)
+ goto fail;
- info->regs = ioremap(info->mem->start, resource_size(info->mem));
- if (!info->regs) {
- ret = -ENOMEM;
- goto fail_release_mem;
- }
+ dev_set_drvdata(&pdev->dev, info);
- ret = request_irq(info->irq, ep93xx_ac97_interrupt, IRQF_TRIGGER_HIGH,
- pdev->name, info);
- if (ret)
- goto fail_unmap_mem;
+ mutex_init(&info->lock);
+ init_completion(&info->done);
+ info->dev = &pdev->dev;
ep93xx_ac97_info = info;
platform_set_drvdata(pdev, info);
ret = snd_soc_register_dai(&pdev->dev, &ep93xx_ac97_dai);
if (ret)
- goto fail_free_irq;
+ goto fail;
return 0;
-fail_free_irq:
+fail:
platform_set_drvdata(pdev, NULL);
- free_irq(info->irq, info);
-fail_unmap_mem:
- iounmap(info->regs);
-fail_release_mem:
- release_mem_region(info->mem->start, resource_size(info->mem));
-fail_free_info:
- kfree(info);
-
+ ep93xx_ac97_info = NULL;
+ dev_set_drvdata(&pdev->dev, NULL);
return ret;
}
@@ -431,11 +411,9 @@ static int __devexit ep93xx_ac97_remove(struct platform_device *pdev)
/* disable the AC97 controller */
ep93xx_ac97_write_reg(info, AC97GCR, 0);
- free_irq(info->irq, info);
- iounmap(info->regs);
- release_mem_region(info->mem->start, resource_size(info->mem));
platform_set_drvdata(pdev, NULL);
- kfree(info);
+ ep93xx_ac97_info = NULL;
+ dev_set_drvdata(&pdev->dev, NULL);
return 0;
}
diff --git a/sound/soc/ep93xx/ep93xx-i2s.c b/sound/soc/ep93xx/ep93xx-i2s.c
index f7a62348e3f..8df8f6dc474 100644
--- a/sound/soc/ep93xx/ep93xx-i2s.c
+++ b/sound/soc/ep93xx/ep93xx-i2s.c
@@ -63,7 +63,6 @@ struct ep93xx_i2s_info {
struct clk *sclk;
struct clk *lrclk;
struct ep93xx_pcm_dma_params *dma_params;
- struct resource *mem;
void __iomem *regs;
};
@@ -373,38 +372,22 @@ static int ep93xx_i2s_probe(struct platform_device *pdev)
struct resource *res;
int err;
- info = kzalloc(sizeof(struct ep93xx_i2s_info), GFP_KERNEL);
- if (!info) {
- err = -ENOMEM;
- goto fail;
- }
-
- dev_set_drvdata(&pdev->dev, info);
- info->dma_params = ep93xx_i2s_dma_params;
+ info = devm_kzalloc(&pdev->dev, sizeof(*info), GFP_KERNEL);
+ if (!info)
+ return -ENOMEM;
res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
- if (!res) {
- err = -ENODEV;
- goto fail_free_info;
- }
+ if (!res)
+ return -ENODEV;
- info->mem = request_mem_region(res->start, resource_size(res),
- pdev->name);
- if (!info->mem) {
- err = -EBUSY;
- goto fail_free_info;
- }
-
- info->regs = ioremap(info->mem->start, resource_size(info->mem));
- if (!info->regs) {
- err = -ENXIO;
- goto fail_release_mem;
- }
+ info->regs = devm_request_and_ioremap(&pdev->dev, res);
+ if (!info->regs)
+ return -ENXIO;
info->mclk = clk_get(&pdev->dev, "mclk");
if (IS_ERR(info->mclk)) {
err = PTR_ERR(info->mclk);
- goto fail_unmap_mem;
+ goto fail;
}
info->sclk = clk_get(&pdev->dev, "sclk");
@@ -419,6 +402,9 @@ static int ep93xx_i2s_probe(struct platform_device *pdev)
goto fail_put_sclk;
}
+ dev_set_drvdata(&pdev->dev, info);
+ info->dma_params = ep93xx_i2s_dma_params;
+
err = snd_soc_register_dai(&pdev->dev, &ep93xx_i2s_dai);
if (err)
goto fail_put_lrclk;
@@ -426,17 +412,12 @@ static int ep93xx_i2s_probe(struct platform_device *pdev)
return 0;
fail_put_lrclk:
+ dev_set_drvdata(&pdev->dev, NULL);
clk_put(info->lrclk);
fail_put_sclk:
clk_put(info->sclk);
fail_put_mclk:
clk_put(info->mclk);
-fail_unmap_mem:
- iounmap(info->regs);
-fail_release_mem:
- release_mem_region(info->mem->start, resource_size(info->mem));
-fail_free_info:
- kfree(info);
fail:
return err;
}
@@ -446,12 +427,10 @@ static int __devexit ep93xx_i2s_remove(struct platform_device *pdev)
struct ep93xx_i2s_info *info = dev_get_drvdata(&pdev->dev);
snd_soc_unregister_dai(&pdev->dev);
+ dev_set_drvdata(&pdev->dev, NULL);
clk_put(info->lrclk);
clk_put(info->sclk);
clk_put(info->mclk);
- iounmap(info->regs);
- release_mem_region(info->mem->start, resource_size(info->mem));
- kfree(info);
return 0;
}
diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig
index d754d34d68a..d70133086ac 100644
--- a/sound/soc/fsl/Kconfig
+++ b/sound/soc/fsl/Kconfig
@@ -1,18 +1,31 @@
-config SND_MPC52xx_DMA
+config SND_SOC_FSL_SSI
tristate
-# ASoC platform support for the Freescale PowerPC SOCs that have an SSI and
-# an Elo DMA controller, such as the MPC8610 and P1022. You will still need to
-# select a platform driver and a codec driver.
-config SND_SOC_POWERPC_SSI
+config SND_SOC_FSL_UTILS
tristate
+
+menuconfig SND_POWERPC_SOC
+ tristate "SoC Audio for Freescale PowerPC CPUs"
depends on FSL_SOC
+ help
+ Say Y or M if you want to add support for codecs attached to
+ the PowerPC CPUs.
+
+if SND_POWERPC_SOC
+
+config SND_MPC52xx_DMA
+ tristate
+
+config SND_SOC_POWERPC_DMA
+ tristate
config SND_SOC_MPC8610_HPCD
tristate "ALSA SoC support for the Freescale MPC8610 HPCD board"
# I2C is necessary for the CS4270 driver
depends on MPC8610_HPCD && I2C
- select SND_SOC_POWERPC_SSI
+ select SND_SOC_FSL_SSI
+ select SND_SOC_FSL_UTILS
+ select SND_SOC_POWERPC_DMA
select SND_SOC_CS4270
select SND_SOC_CS4270_VD33_ERRATA
default y if MPC8610_HPCD
@@ -23,7 +36,9 @@ config SND_SOC_P1022_DS
tristate "ALSA SoC support for the Freescale P1022 DS board"
# I2C is necessary for the WM8776 driver
depends on P1022_DS && I2C
- select SND_SOC_POWERPC_SSI
+ select SND_SOC_FSL_SSI
+ select SND_SOC_FSL_UTILS
+ select SND_SOC_POWERPC_DMA
select SND_SOC_WM8776
default y if P1022_DS
help
@@ -65,3 +80,103 @@ config SND_MPC52xx_SOC_EFIKA
help
Say Y if you want to add support for sound on the Efika.
+endif # SND_POWERPC_SOC
+
+menuconfig SND_IMX_SOC
+ tristate "SoC Audio for Freescale i.MX CPUs"
+ depends on ARCH_MXC
+ help
+ Say Y or M if you want to add support for codecs attached to
+ the i.MX CPUs.
+
+if SND_IMX_SOC
+
+config SND_SOC_IMX_SSI
+ tristate
+
+config SND_SOC_IMX_PCM
+ tristate
+
+config SND_SOC_IMX_PCM_FIQ
+ tristate
+ select FIQ
+ select SND_SOC_IMX_PCM
+
+config SND_SOC_IMX_PCM_DMA
+ tristate
+ select SND_SOC_DMAENGINE_PCM
+ select SND_SOC_IMX_PCM
+
+config SND_SOC_IMX_AUDMUX
+ tristate
+
+config SND_MXC_SOC_WM1133_EV1
+ tristate "Audio on the i.MX31ADS with WM1133-EV1 fitted"
+ depends on MACH_MX31ADS_WM1133_EV1 && EXPERIMENTAL
+ select SND_SOC_WM8350
+ select SND_SOC_IMX_PCM_FIQ
+ select SND_SOC_IMX_AUDMUX
+ select SND_SOC_IMX_SSI
+ help
+ Enable support for audio on the i.MX31ADS with the WM1133-EV1
+ PMIC board with WM8835x fitted.
+
+config SND_SOC_MX27VIS_AIC32X4
+ tristate "SoC audio support for Visstrim M10 boards"
+ depends on MACH_IMX27_VISSTRIM_M10 && I2C
+ select SND_SOC_TLV320AIC32X4
+ select SND_SOC_IMX_PCM_DMA
+ select SND_SOC_IMX_AUDMUX
+ select SND_SOC_IMX_SSI
+ help
+ Say Y if you want to add support for SoC audio on Visstrim SM10
+ board with TLV320AIC32X4 codec.
+
+config SND_SOC_PHYCORE_AC97
+ tristate "SoC Audio support for Phytec phyCORE (and phyCARD) boards"
+ depends on MACH_PCM043 || MACH_PCA100
+ select SND_SOC_AC97_BUS
+ select SND_SOC_WM9712
+ select SND_SOC_IMX_PCM_FIQ
+ select SND_SOC_IMX_AUDMUX
+ select SND_SOC_IMX_SSI
+ help
+ Say Y if you want to add support for SoC audio on Phytec phyCORE
+ and phyCARD boards in AC97 mode
+
+config SND_SOC_EUKREA_TLV320
+ tristate "Eukrea TLV320"
+ depends on MACH_EUKREA_MBIMX27_BASEBOARD \
+ || MACH_EUKREA_MBIMXSD25_BASEBOARD \
+ || MACH_EUKREA_MBIMXSD35_BASEBOARD \
+ || MACH_EUKREA_MBIMXSD51_BASEBOARD
+ depends on I2C
+ select SND_SOC_TLV320AIC23
+ select SND_SOC_IMX_PCM_FIQ
+ select SND_SOC_IMX_AUDMUX
+ select SND_SOC_IMX_SSI
+ help
+ Enable I2S based access to the TLV320AIC23B codec attached
+ to the SSI interface
+
+config SND_SOC_IMX_SGTL5000
+ tristate "SoC Audio support for i.MX boards with sgtl5000"
+ depends on OF && I2C
+ select SND_SOC_SGTL5000
+ select SND_SOC_IMX_PCM_DMA
+ select SND_SOC_IMX_AUDMUX
+ select SND_SOC_FSL_SSI
+ select SND_SOC_FSL_UTILS
+ help
+ Say Y if you want to add support for SoC audio on an i.MX board with
+ a sgtl5000 codec.
+
+config SND_SOC_IMX_MC13783
+ tristate "SoC Audio support for I.MX boards with mc13783"
+ depends on MFD_MC13783
+ select SND_SOC_IMX_SSI
+ select SND_SOC_IMX_AUDMUX
+ select SND_SOC_MC13783
+ select SND_SOC_IMX_PCM_DMA
+
+endif # SND_IMX_SOC
diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile
index b4a38c0ac58..5f3cf3f52ea 100644
--- a/sound/soc/fsl/Makefile
+++ b/sound/soc/fsl/Makefile
@@ -8,8 +8,11 @@ obj-$(CONFIG_SND_SOC_P1022_DS) += snd-soc-p1022-ds.o
# Freescale PowerPC SSI/DMA Platform Support
snd-soc-fsl-ssi-objs := fsl_ssi.o
+snd-soc-fsl-utils-objs := fsl_utils.o
snd-soc-fsl-dma-objs := fsl_dma.o
-obj-$(CONFIG_SND_SOC_POWERPC_SSI) += snd-soc-fsl-ssi.o snd-soc-fsl-dma.o
+obj-$(CONFIG_SND_SOC_FSL_SSI) += snd-soc-fsl-ssi.o
+obj-$(CONFIG_SND_SOC_FSL_UTILS) += snd-soc-fsl-utils.o
+obj-$(CONFIG_SND_SOC_POWERPC_DMA) += snd-soc-fsl-dma.o
# MPC5200 Platform Support
obj-$(CONFIG_SND_MPC52xx_DMA) += mpc5200_dma.o
@@ -20,3 +23,29 @@ obj-$(CONFIG_SND_SOC_MPC5200_AC97) += mpc5200_psc_ac97.o
obj-$(CONFIG_SND_MPC52xx_SOC_PCM030) += pcm030-audio-fabric.o
obj-$(CONFIG_SND_MPC52xx_SOC_EFIKA) += efika-audio-fabric.o
+# i.MX Platform Support
+snd-soc-imx-ssi-objs := imx-ssi.o
+snd-soc-imx-audmux-objs := imx-audmux.o
+
+obj-$(CONFIG_SND_SOC_IMX_SSI) += snd-soc-imx-ssi.o
+obj-$(CONFIG_SND_SOC_IMX_AUDMUX) += snd-soc-imx-audmux.o
+
+obj-$(CONFIG_SND_SOC_IMX_PCM) += snd-soc-imx-pcm.o
+snd-soc-imx-pcm-y := imx-pcm.o
+snd-soc-imx-pcm-$(CONFIG_SND_SOC_IMX_PCM_FIQ) += imx-pcm-fiq.o
+snd-soc-imx-pcm-$(CONFIG_SND_SOC_IMX_PCM_DMA) += imx-pcm-dma.o
+
+# i.MX Machine Support
+snd-soc-eukrea-tlv320-objs := eukrea-tlv320.o
+snd-soc-phycore-ac97-objs := phycore-ac97.o
+snd-soc-mx27vis-aic32x4-objs := mx27vis-aic32x4.o
+snd-soc-wm1133-ev1-objs := wm1133-ev1.o
+snd-soc-imx-sgtl5000-objs := imx-sgtl5000.o
+snd-soc-imx-mc13783-objs := imx-mc13783.o
+
+obj-$(CONFIG_SND_SOC_EUKREA_TLV320) += snd-soc-eukrea-tlv320.o
+obj-$(CONFIG_SND_SOC_PHYCORE_AC97) += snd-soc-phycore-ac97.o
+obj-$(CONFIG_SND_SOC_MX27VIS_AIC32X4) += snd-soc-mx27vis-aic32x4.o
+obj-$(CONFIG_SND_MXC_SOC_WM1133_EV1) += snd-soc-wm1133-ev1.o
+obj-$(CONFIG_SND_SOC_IMX_SGTL5000) += snd-soc-imx-sgtl5000.o
+obj-$(CONFIG_SND_SOC_IMX_MC13783) += snd-soc-imx-mc13783.o
diff --git a/sound/soc/imx/eukrea-tlv320.c b/sound/soc/fsl/eukrea-tlv320.c
index 7d4475cfdb2..efb9ede0120 100644
--- a/sound/soc/imx/eukrea-tlv320.c
+++ b/sound/soc/fsl/eukrea-tlv320.c
@@ -7,7 +7,7 @@
* which is Copyright 2009 Simtec Electronics
* and on sound/soc/imx/phycore-ac97.c which is
* Copyright 2009 Sascha Hauer, Pengutronix <s.hauer@pengutronix.de>
- *
+ *
* 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
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index 2eb407fa3b4..4ed2afd4778 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -11,11 +11,15 @@
*/
#include <linux/init.h>
+#include <linux/io.h>
#include <linux/module.h>
#include <linux/interrupt.h>
+#include <linux/clk.h>
#include <linux/device.h>
#include <linux/delay.h>
#include <linux/slab.h>
+#include <linux/of_address.h>
+#include <linux/of_irq.h>
#include <linux/of_platform.h>
#include <sound/core.h>
@@ -25,6 +29,26 @@
#include <sound/soc.h>
#include "fsl_ssi.h"
+#include "imx-pcm.h"
+
+#ifdef PPC
+#define read_ssi(addr) in_be32(addr)
+#define write_ssi(val, addr) out_be32(addr, val)
+#define write_ssi_mask(addr, clear, set) clrsetbits_be32(addr, clear, set)
+#elif defined ARM
+#define read_ssi(addr) readl(addr)
+#define write_ssi(val, addr) writel(val, addr)
+/*
+ * FIXME: Proper locking should be added at write_ssi_mask caller level
+ * to ensure this register read/modify/write sequence is race free.
+ */
+static inline void write_ssi_mask(u32 __iomem *addr, u32 clear, u32 set)
+{
+ u32 val = readl(addr);
+ val = (val & ~clear) | set;
+ writel(val, addr);
+}
+#endif
/**
* FSLSSI_I2S_RATES: sample rates supported by the I2S
@@ -94,6 +118,13 @@ struct fsl_ssi_private {
struct device_attribute dev_attr;
struct platform_device *pdev;
+ bool new_binding;
+ bool ssi_on_imx;
+ struct clk *clk;
+ struct platform_device *imx_pcm_pdev;
+ struct imx_pcm_dma_params dma_params_tx;
+ struct imx_pcm_dma_params dma_params_rx;
+
struct {
unsigned int rfrc;
unsigned int tfrc;
@@ -145,7 +176,7 @@ static irqreturn_t fsl_ssi_isr(int irq, void *dev_id)
were interrupted for. We mask it with the Interrupt Enable register
so that we only check for events that we're interested in.
*/
- sisr = in_be32(&ssi->sisr) & SIER_FLAGS;
+ sisr = read_ssi(&ssi->sisr) & SIER_FLAGS;
if (sisr & CCSR_SSI_SISR_RFRC) {
ssi_private->stats.rfrc++;
@@ -260,7 +291,7 @@ static irqreturn_t fsl_ssi_isr(int irq, void *dev_id)
/* Clear the bits that we set */
if (sisr2)
- out_be32(&ssi->sisr, sisr2);
+ write_ssi(sisr2, &ssi->sisr);
return ret;
}
@@ -295,7 +326,7 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream,
* SSI needs to be disabled before updating the registers we set
* here.
*/
- clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN);
+ write_ssi_mask(&ssi->scr, CCSR_SSI_SCR_SSIEN, 0);
/*
* Program the SSI into I2S Slave Non-Network Synchronous mode.
@@ -303,20 +334,18 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream,
*
* FIXME: Little-endian samples require a different shift dir
*/
- clrsetbits_be32(&ssi->scr,
+ write_ssi_mask(&ssi->scr,
CCSR_SSI_SCR_I2S_MODE_MASK | CCSR_SSI_SCR_SYN,
CCSR_SSI_SCR_TFR_CLK_DIS | CCSR_SSI_SCR_I2S_MODE_SLAVE
| (synchronous ? CCSR_SSI_SCR_SYN : 0));
- out_be32(&ssi->stcr,
- CCSR_SSI_STCR_TXBIT0 | CCSR_SSI_STCR_TFEN0 |
+ write_ssi(CCSR_SSI_STCR_TXBIT0 | CCSR_SSI_STCR_TFEN0 |
CCSR_SSI_STCR_TFSI | CCSR_SSI_STCR_TEFS |
- CCSR_SSI_STCR_TSCKP);
+ CCSR_SSI_STCR_TSCKP, &ssi->stcr);
- out_be32(&ssi->srcr,
- CCSR_SSI_SRCR_RXBIT0 | CCSR_SSI_SRCR_RFEN0 |
+ write_ssi(CCSR_SSI_SRCR_RXBIT0 | CCSR_SSI_SRCR_RFEN0 |
CCSR_SSI_SRCR_RFSI | CCSR_SSI_SRCR_REFS |
- CCSR_SSI_SRCR_RSCKP);
+ CCSR_SSI_SRCR_RSCKP, &ssi->srcr);
/*
* The DC and PM bits are only used if the SSI is the clock
@@ -324,7 +353,7 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream,
*/
/* Enable the interrupts and DMA requests */
- out_be32(&ssi->sier, SIER_FLAGS);
+ write_ssi(SIER_FLAGS, &ssi->sier);
/*
* Set the watermark for transmit FIFI 0 and receive FIFO 0. We
@@ -339,9 +368,9 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream,
* make this value larger (and maybe we should), but this way
* data will be written to memory as soon as it's available.
*/
- out_be32(&ssi->sfcsr,
- CCSR_SSI_SFCSR_TFWM0(ssi_private->fifo_depth - 2) |
- CCSR_SSI_SFCSR_RFWM0(ssi_private->fifo_depth - 2));
+ write_ssi(CCSR_SSI_SFCSR_TFWM0(ssi_private->fifo_depth - 2) |
+ CCSR_SSI_SFCSR_RFWM0(ssi_private->fifo_depth - 2),
+ &ssi->sfcsr);
/*
* We keep the SSI disabled because if we enable it, then the
@@ -393,6 +422,12 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream,
ssi_private->second_stream = substream;
}
+ if (ssi_private->ssi_on_imx)
+ snd_soc_dai_set_dma_data(dai, substream,
+ (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ?
+ &ssi_private->dma_params_tx :
+ &ssi_private->dma_params_rx);
+
return 0;
}
@@ -417,7 +452,7 @@ static int fsl_ssi_hw_params(struct snd_pcm_substream *substream,
unsigned int sample_size =
snd_pcm_format_width(params_format(hw_params));
u32 wl = CCSR_SSI_SxCCR_WL(sample_size);
- int enabled = in_be32(&ssi->scr) & CCSR_SSI_SCR_SSIEN;
+ int enabled = read_ssi(&ssi->scr) & CCSR_SSI_SCR_SSIEN;
/*
* If we're in synchronous mode, and the SSI is already enabled,
@@ -439,9 +474,9 @@ static int fsl_ssi_hw_params(struct snd_pcm_substream *substream,
/* In synchronous mode, the SSI uses STCCR for capture */
if ((substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ||
ssi_private->cpu_dai_drv.symmetric_rates)
- clrsetbits_be32(&ssi->stccr, CCSR_SSI_SxCCR_WL_MASK, wl);
+ write_ssi_mask(&ssi->stccr, CCSR_SSI_SxCCR_WL_MASK, wl);
else
- clrsetbits_be32(&ssi->srccr, CCSR_SSI_SxCCR_WL_MASK, wl);
+ write_ssi_mask(&ssi->srccr, CCSR_SSI_SxCCR_WL_MASK, wl);
return 0;
}
@@ -466,19 +501,19 @@ static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd,
case SNDRV_PCM_TRIGGER_START:
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
- setbits32(&ssi->scr,
+ write_ssi_mask(&ssi->scr, 0,
CCSR_SSI_SCR_SSIEN | CCSR_SSI_SCR_TE);
else
- setbits32(&ssi->scr,
+ write_ssi_mask(&ssi->scr, 0,
CCSR_SSI_SCR_SSIEN | CCSR_SSI_SCR_RE);
break;
case SNDRV_PCM_TRIGGER_STOP:
case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
- clrbits32(&ssi->scr, CCSR_SSI_SCR_TE);
+ write_ssi_mask(&ssi->scr, CCSR_SSI_SCR_TE, 0);
else
- clrbits32(&ssi->scr, CCSR_SSI_SCR_RE);
+ write_ssi_mask(&ssi->scr, CCSR_SSI_SCR_RE, 0);
break;
default:
@@ -510,7 +545,7 @@ static void fsl_ssi_shutdown(struct snd_pcm_substream *substream,
if (!ssi_private->first_stream) {
struct ccsr_ssi __iomem *ssi = ssi_private->ssi;
- clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN);
+ write_ssi_mask(&ssi->scr, CCSR_SSI_SCR_SSIEN, 0);
}
}
@@ -622,12 +657,6 @@ static int __devinit fsl_ssi_probe(struct platform_device *pdev)
if (!of_device_is_available(np))
return -ENODEV;
- /* Check for a codec-handle property. */
- if (!of_get_property(np, "codec-handle", NULL)) {
- dev_err(&pdev->dev, "missing codec-handle property\n");
- return -ENODEV;
- }
-
/* We only support the SSI in "I2S Slave" mode */
sprop = of_get_property(np, "fsl,mode", NULL);
if (!sprop || strcmp(sprop, "i2s-slave")) {
@@ -692,6 +721,50 @@ static int __devinit fsl_ssi_probe(struct platform_device *pdev)
/* Older 8610 DTs didn't have the fifo-depth property */
ssi_private->fifo_depth = 8;
+ if (of_device_is_compatible(pdev->dev.of_node, "fsl,imx21-ssi")) {
+ u32 dma_events[2];
+ ssi_private->ssi_on_imx = true;
+
+ ssi_private->clk = clk_get(&pdev->dev, NULL);
+ if (IS_ERR(ssi_private->clk)) {
+ ret = PTR_ERR(ssi_private->clk);
+ dev_err(&pdev->dev, "could not get clock: %d\n", ret);
+ goto error_irq;
+ }
+ clk_prepare_enable(ssi_private->clk);
+
+ /*
+ * We have burstsize be "fifo_depth - 2" to match the SSI
+ * watermark setting in fsl_ssi_startup().
+ */
+ ssi_private->dma_params_tx.burstsize =
+ ssi_private->fifo_depth - 2;
+ ssi_private->dma_params_rx.burstsize =
+ ssi_private->fifo_depth - 2;
+ ssi_private->dma_params_tx.dma_addr =
+ ssi_private->ssi_phys + offsetof(struct ccsr_ssi, stx0);
+ ssi_private->dma_params_rx.dma_addr =
+ ssi_private->ssi_phys + offsetof(struct ccsr_ssi, srx0);
+ /*
+ * TODO: This is a temporary solution and should be changed
+ * to use generic DMA binding later when the helplers get in.
+ */
+ ret = of_property_read_u32_array(pdev->dev.of_node,
+ "fsl,ssi-dma-events", dma_events, 2);
+ if (ret) {
+ dev_err(&pdev->dev, "could not get dma events\n");
+ goto error_clk;
+ }
+ ssi_private->dma_params_tx.dma = dma_events[0];
+ ssi_private->dma_params_rx.dma = dma_events[1];
+
+ ssi_private->dma_params_tx.shared_peripheral =
+ of_device_is_compatible(of_get_parent(np),
+ "fsl,spba-bus");
+ ssi_private->dma_params_rx.shared_peripheral =
+ ssi_private->dma_params_tx.shared_peripheral;
+ }
+
/* Initialize the the device_attribute structure */
dev_attr = &ssi_private->dev_attr;
sysfs_attr_init(&dev_attr->attr);
@@ -715,6 +788,26 @@ static int __devinit fsl_ssi_probe(struct platform_device *pdev)
goto error_dev;
}
+ if (ssi_private->ssi_on_imx) {
+ ssi_private->imx_pcm_pdev =
+ platform_device_register_simple("imx-pcm-audio",
+ -1, NULL, 0);
+ if (IS_ERR(ssi_private->imx_pcm_pdev)) {
+ ret = PTR_ERR(ssi_private->imx_pcm_pdev);
+ goto error_dev;
+ }
+ }
+
+ /*
+ * If codec-handle property is missing from SSI node, we assume
+ * that the machine driver uses new binding which does not require
+ * SSI driver to trigger machine driver's probe.
+ */
+ if (!of_get_property(np, "codec-handle", NULL)) {
+ ssi_private->new_binding = true;
+ goto done;
+ }
+
/* Trigger the machine driver's probe function. The platform driver
* name of the machine driver is taken from /compatible property of the
* device tree. We also pass the address of the CPU DAI driver
@@ -736,15 +829,24 @@ static int __devinit fsl_ssi_probe(struct platform_device *pdev)
goto error_dai;
}
+done:
return 0;
error_dai:
+ if (ssi_private->ssi_on_imx)
+ platform_device_unregister(ssi_private->imx_pcm_pdev);
snd_soc_unregister_dai(&pdev->dev);
error_dev:
dev_set_drvdata(&pdev->dev, NULL);
device_remove_file(&pdev->dev, dev_attr);
+error_clk:
+ if (ssi_private->ssi_on_imx) {
+ clk_disable_unprepare(ssi_private->clk);
+ clk_put(ssi_private->clk);
+ }
+
error_irq:
free_irq(ssi_private->irq, ssi_private);
@@ -764,7 +866,13 @@ static int fsl_ssi_remove(struct platform_device *pdev)
{
struct fsl_ssi_private *ssi_private = dev_get_drvdata(&pdev->dev);
- platform_device_unregister(ssi_private->pdev);
+ if (!ssi_private->new_binding)
+ platform_device_unregister(ssi_private->pdev);
+ if (ssi_private->ssi_on_imx) {
+ platform_device_unregister(ssi_private->imx_pcm_pdev);
+ clk_disable_unprepare(ssi_private->clk);
+ clk_put(ssi_private->clk);
+ }
snd_soc_unregister_dai(&pdev->dev);
device_remove_file(&pdev->dev, &ssi_private->dev_attr);
@@ -779,6 +887,7 @@ static int fsl_ssi_remove(struct platform_device *pdev)
static const struct of_device_id fsl_ssi_ids[] = {
{ .compatible = "fsl,mpc8610-ssi", },
+ { .compatible = "fsl,imx21-ssi", },
{}
};
MODULE_DEVICE_TABLE(of, fsl_ssi_ids);
diff --git a/sound/soc/fsl/fsl_utils.c b/sound/soc/fsl/fsl_utils.c
new file mode 100644
index 00000000000..b9e42b503a3
--- /dev/null
+++ b/sound/soc/fsl/fsl_utils.c
@@ -0,0 +1,91 @@
+/**
+ * Freescale ALSA SoC Machine driver utility
+ *
+ * Author: Timur Tabi <timur@freescale.com>
+ *
+ * Copyright 2010 Freescale Semiconductor, Inc.
+ *
+ * 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/of_address.h>
+#include <sound/soc.h>
+
+#include "fsl_utils.h"
+
+/**
+ * fsl_asoc_get_dma_channel - determine the dma channel for a SSI node
+ *
+ * @ssi_np: pointer to the SSI device tree node
+ * @name: name of the phandle pointing to the dma channel
+ * @dai: ASoC DAI link pointer to be filled with platform_name
+ * @dma_channel_id: dma channel id to be returned
+ * @dma_id: dma id to be returned
+ *
+ * This function determines the dma and channel id for given SSI node. It
+ * also discovers the platform_name for the ASoC DAI link.
+ */
+int fsl_asoc_get_dma_channel(struct device_node *ssi_np,
+ const char *name,
+ struct snd_soc_dai_link *dai,
+ unsigned int *dma_channel_id,
+ unsigned int *dma_id)
+{
+ struct resource res;
+ struct device_node *dma_channel_np, *dma_np;
+ const u32 *iprop;
+ int ret;
+
+ dma_channel_np = of_parse_phandle(ssi_np, name, 0);
+ if (!dma_channel_np)
+ return -EINVAL;
+
+ if (!of_device_is_compatible(dma_channel_np, "fsl,ssi-dma-channel")) {
+ of_node_put(dma_channel_np);
+ return -EINVAL;
+ }
+
+ /* Determine the dev_name for the device_node. This code mimics the
+ * behavior of of_device_make_bus_id(). We need this because ASoC uses
+ * the dev_name() of the device to match the platform (DMA) device with
+ * the CPU (SSI) device. It's all ugly and hackish, but it works (for
+ * now).
+ *
+ * dai->platform name should already point to an allocated buffer.
+ */
+ ret = of_address_to_resource(dma_channel_np, 0, &res);
+ if (ret) {
+ of_node_put(dma_channel_np);
+ return ret;
+ }
+ snprintf((char *)dai->platform_name, DAI_NAME_SIZE, "%llx.%s",
+ (unsigned long long) res.start, dma_channel_np->name);
+
+ iprop = of_get_property(dma_channel_np, "cell-index", NULL);
+ if (!iprop) {
+ of_node_put(dma_channel_np);
+ return -EINVAL;
+ }
+ *dma_channel_id = be32_to_cpup(iprop);
+
+ dma_np = of_get_parent(dma_channel_np);
+ iprop = of_get_property(dma_np, "cell-index", NULL);
+ if (!iprop) {
+ of_node_put(dma_np);
+ return -EINVAL;
+ }
+ *dma_id = be32_to_cpup(iprop);
+
+ of_node_put(dma_np);
+ of_node_put(dma_channel_np);
+
+ return 0;
+}
+EXPORT_SYMBOL(fsl_asoc_get_dma_channel);
+
+MODULE_AUTHOR("Timur Tabi <timur@freescale.com>");
+MODULE_DESCRIPTION("Freescale ASoC utility code");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/fsl/fsl_utils.h b/sound/soc/fsl/fsl_utils.h
new file mode 100644
index 00000000000..b2951126527
--- /dev/null
+++ b/sound/soc/fsl/fsl_utils.h
@@ -0,0 +1,26 @@
+/**
+ * Freescale ALSA SoC Machine driver utility
+ *
+ * Author: Timur Tabi <timur@freescale.com>
+ *
+ * Copyright 2010 Freescale Semiconductor, Inc.
+ *
+ * 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.
+ */
+
+#ifndef _FSL_UTILS_H
+#define _FSL_UTILS_H
+
+#define DAI_NAME_SIZE 32
+
+struct snd_soc_dai_link;
+struct device_node;
+
+int fsl_asoc_get_dma_channel(struct device_node *ssi_np, const char *name,
+ struct snd_soc_dai_link *dai,
+ unsigned int *dma_channel_id,
+ unsigned int *dma_id);
+
+#endif /* _FSL_UTILS_H */
diff --git a/sound/soc/imx/imx-audmux.c b/sound/soc/fsl/imx-audmux.c
index f23700359c6..080327414c6 100644
--- a/sound/soc/imx/imx-audmux.c
+++ b/sound/soc/fsl/imx-audmux.c
@@ -26,6 +26,7 @@
#include <linux/of_device.h>
#include <linux/platform_device.h>
#include <linux/slab.h>
+#include <linux/pinctrl/consumer.h>
#include "imx-audmux.h"
@@ -249,6 +250,7 @@ EXPORT_SYMBOL_GPL(imx_audmux_v2_configure_port);
static int __devinit imx_audmux_probe(struct platform_device *pdev)
{
struct resource *res;
+ struct pinctrl *pinctrl;
const struct of_device_id *of_id =
of_match_device(imx_audmux_dt_ids, &pdev->dev);
@@ -257,6 +259,12 @@ static int __devinit imx_audmux_probe(struct platform_device *pdev)
if (!audmux_base)
return -EADDRNOTAVAIL;
+ pinctrl = devm_pinctrl_get_select_default(&pdev->dev);
+ if (IS_ERR(pinctrl)) {
+ dev_err(&pdev->dev, "setup pinctrl failed!");
+ return PTR_ERR(pinctrl);
+ }
+
audmux_clk = clk_get(&pdev->dev, "audmux");
if (IS_ERR(audmux_clk)) {
dev_dbg(&pdev->dev, "cannot get clock: %ld\n",
diff --git a/sound/soc/imx/imx-audmux.h b/sound/soc/fsl/imx-audmux.h
index 04ebbab8d7b..04ebbab8d7b 100644
--- a/sound/soc/imx/imx-audmux.h
+++ b/sound/soc/fsl/imx-audmux.h
diff --git a/sound/soc/fsl/imx-mc13783.c b/sound/soc/fsl/imx-mc13783.c
new file mode 100644
index 00000000000..f59c3494366
--- /dev/null
+++ b/sound/soc/fsl/imx-mc13783.c
@@ -0,0 +1,156 @@
+/*
+ * imx-mc13783.c -- SoC audio for imx based boards with mc13783 codec
+ *
+ * Copyright 2012 Philippe Retornaz, <philippe.retornaz@epfl.ch>
+ *
+ * Heavly based on phycore-mc13783:
+ * Copyright 2009 Sascha Hauer, Pengutronix <s.hauer@pengutronix.de>
+ *
+ * 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/moduleparam.h>
+#include <linux/device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <asm/mach-types.h>
+
+#include "../codecs/mc13783.h"
+#include "imx-ssi.h"
+#include "imx-audmux.h"
+
+#define FMT_SSI (SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_NB_NF | \
+ SND_SOC_DAIFMT_CBM_CFM)
+
+static int imx_mc13783_hifi_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ int ret;
+
+ ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xfffffffc, 0xfffffffc,
+ 4, 16);
+ if (ret)
+ return ret;
+
+ ret = snd_soc_dai_set_sysclk(codec_dai, MC13783_CLK_CLIA, 26000000, 0);
+ if (ret)
+ return ret;
+
+ ret = snd_soc_dai_set_tdm_slot(cpu_dai, 0x0, 0xfffffffc, 2, 16);
+ if (ret)
+ return ret;
+
+ return 0;
+}
+
+static struct snd_soc_ops imx_mc13783_hifi_ops = {
+ .hw_params = imx_mc13783_hifi_hw_params,
+};
+
+static struct snd_soc_dai_link imx_mc13783_dai_mc13783[] = {
+ {
+ .name = "MC13783",
+ .stream_name = "Sound",
+ .codec_dai_name = "mc13783-hifi",
+ .codec_name = "mc13783-codec",
+ .cpu_dai_name = "imx-ssi.0",
+ .platform_name = "imx-pcm-audio.0",
+ .ops = &imx_mc13783_hifi_ops,
+ .symmetric_rates = 1,
+ .dai_fmt = FMT_SSI,
+ },
+};
+
+static const struct snd_soc_dapm_widget imx_mc13783_widget[] = {
+ SND_SOC_DAPM_MIC("Mic", NULL),
+ SND_SOC_DAPM_HP("Headphone", NULL),
+ SND_SOC_DAPM_SPK("Speaker", NULL),
+};
+
+static const struct snd_soc_dapm_route imx_mc13783_routes[] = {
+ {"Speaker", NULL, "LSP"},
+ {"Headphone", NULL, "HSL"},
+ {"Headphone", NULL, "HSR"},
+
+ {"MC1LIN", NULL, "MC1 Bias"},
+ {"MC2IN", NULL, "MC2 Bias"},
+ {"MC1 Bias", NULL, "Mic"},
+ {"MC2 Bias", NULL, "Mic"},
+};
+
+static struct snd_soc_card imx_mc13783 = {
+ .name = "imx_mc13783",
+ .dai_link = imx_mc13783_dai_mc13783,
+ .num_links = ARRAY_SIZE(imx_mc13783_dai_mc13783),
+ .dapm_widgets = imx_mc13783_widget,
+ .num_dapm_widgets = ARRAY_SIZE(imx_mc13783_widget),
+ .dapm_routes = imx_mc13783_routes,
+ .num_dapm_routes = ARRAY_SIZE(imx_mc13783_routes),
+};
+
+static int __devinit imx_mc13783_probe(struct platform_device *pdev)
+{
+ int ret;
+
+ imx_mc13783.dev = &pdev->dev;
+
+ ret = snd_soc_register_card(&imx_mc13783);
+ if (ret) {
+ dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n",
+ ret);
+ 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));
+
+ return ret;
+}
+
+static int __devexit imx_mc13783_remove(struct platform_device *pdev)
+{
+ snd_soc_unregister_card(&imx_mc13783);
+
+ return 0;
+}
+
+static struct platform_driver imx_mc13783_audio_driver = {
+ .driver = {
+ .name = "imx_mc13783",
+ .owner = THIS_MODULE,
+ },
+ .probe = imx_mc13783_probe,
+ .remove = __devexit_p(imx_mc13783_remove)
+};
+
+module_platform_driver(imx_mc13783_audio_driver);
+
+MODULE_AUTHOR("Sascha Hauer <s.hauer@pengutronix.de>");
+MODULE_AUTHOR("Philippe Retornaz <philippe.retornaz@epfl.ch");
+MODULE_DESCRIPTION("imx with mc13783 codec ALSA SoC driver");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:imx_mc13783");
diff --git a/sound/soc/imx/imx-pcm-dma-mx2.c b/sound/soc/fsl/imx-pcm-dma.c
index 6b818de2fc0..f3c0a5ef35c 100644
--- a/sound/soc/imx/imx-pcm-dma-mx2.c
+++ b/sound/soc/fsl/imx-pcm-dma.c
@@ -109,7 +109,8 @@ static int snd_imx_open(struct snd_pcm_substream *substream)
dma_params = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
dma_data = kzalloc(sizeof(*dma_data), GFP_KERNEL);
- dma_data->peripheral_type = IMX_DMATYPE_SSI;
+ dma_data->peripheral_type = dma_params->shared_peripheral ?
+ IMX_DMATYPE_SSI_SP : IMX_DMATYPE_SSI;
dma_data->priority = DMA_PRIO_HIGH;
dma_data->dma_request = dma_params->dma;
diff --git a/sound/soc/imx/imx-pcm-fiq.c b/sound/soc/fsl/imx-pcm-fiq.c
index 456b7d723d6..456b7d723d6 100644
--- a/sound/soc/imx/imx-pcm-fiq.c
+++ b/sound/soc/fsl/imx-pcm-fiq.c
diff --git a/sound/soc/imx/imx-pcm.c b/sound/soc/fsl/imx-pcm.c
index 93dc360b177..93dc360b177 100644
--- a/sound/soc/imx/imx-pcm.c
+++ b/sound/soc/fsl/imx-pcm.c
diff --git a/sound/soc/imx/imx-pcm.h b/sound/soc/fsl/imx-pcm.h
index b5f5c3acf34..83c0ed7d55c 100644
--- a/sound/soc/imx/imx-pcm.h
+++ b/sound/soc/fsl/imx-pcm.h
@@ -22,6 +22,7 @@ struct imx_pcm_dma_params {
int dma;
unsigned long dma_addr;
int burstsize;
+ bool shared_peripheral; /* The peripheral is on SPBA bus */
};
int snd_imx_pcm_mmap(struct snd_pcm_substream *substream,
diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c
new file mode 100644
index 00000000000..3a729caeb8c
--- /dev/null
+++ b/sound/soc/fsl/imx-sgtl5000.c
@@ -0,0 +1,221 @@
+/*
+ * Copyright 2012 Freescale Semiconductor, Inc.
+ * Copyright 2012 Linaro Ltd.
+ *
+ * The code contained herein is licensed under the GNU General Public
+ * License. You may obtain a copy of the GNU General Public License
+ * Version 2 or later at the following locations:
+ *
+ * http://www.opensource.org/licenses/gpl-license.html
+ * http://www.gnu.org/copyleft/gpl.html
+ */
+
+#include <linux/module.h>
+#include <linux/of.h>
+#include <linux/of_platform.h>
+#include <linux/of_i2c.h>
+#include <linux/clk.h>
+#include <sound/soc.h>
+
+#include "../codecs/sgtl5000.h"
+#include "imx-audmux.h"
+
+#define DAI_NAME_SIZE 32
+
+struct imx_sgtl5000_data {
+ struct snd_soc_dai_link dai;
+ struct snd_soc_card card;
+ char codec_dai_name[DAI_NAME_SIZE];
+ char platform_name[DAI_NAME_SIZE];
+ struct clk *codec_clk;
+ unsigned int clk_frequency;
+};
+
+static int imx_sgtl5000_dai_init(struct snd_soc_pcm_runtime *rtd)
+{
+ struct imx_sgtl5000_data *data = container_of(rtd->card,
+ struct imx_sgtl5000_data, card);
+ struct device *dev = rtd->card->dev;
+ int ret;
+
+ ret = snd_soc_dai_set_sysclk(rtd->codec_dai, SGTL5000_SYSCLK,
+ data->clk_frequency, SND_SOC_CLOCK_IN);
+ if (ret) {
+ dev_err(dev, "could not set codec driver clock params\n");
+ return ret;
+ }
+
+ return 0;
+}
+
+static const struct snd_soc_dapm_widget imx_sgtl5000_dapm_widgets[] = {
+ SND_SOC_DAPM_MIC("Mic Jack", NULL),
+ SND_SOC_DAPM_LINE("Line In Jack", NULL),
+ SND_SOC_DAPM_HP("Headphone Jack", NULL),
+ SND_SOC_DAPM_SPK("Line Out Jack", NULL),
+ SND_SOC_DAPM_SPK("Ext Spk", NULL),
+};
+
+static int __devinit imx_sgtl5000_probe(struct platform_device *pdev)
+{
+ struct device_node *np = pdev->dev.of_node;
+ struct device_node *ssi_np, *codec_np;
+ struct platform_device *ssi_pdev;
+ struct i2c_client *codec_dev;
+ struct imx_sgtl5000_data *data;
+ int int_port, ext_port;
+ int ret;
+
+ ret = of_property_read_u32(np, "mux-int-port", &int_port);
+ if (ret) {
+ dev_err(&pdev->dev, "mux-int-port missing or invalid\n");
+ return ret;
+ }
+ ret = of_property_read_u32(np, "mux-ext-port", &ext_port);
+ if (ret) {
+ dev_err(&pdev->dev, "mux-ext-port missing or invalid\n");
+ return ret;
+ }
+
+ /*
+ * The port numbering in the hardware manual starts at 1, while
+ * the audmux API expects it starts at 0.
+ */
+ int_port--;
+ ext_port--;
+ ret = imx_audmux_v2_configure_port(int_port,
+ IMX_AUDMUX_V2_PTCR_SYN |
+ IMX_AUDMUX_V2_PTCR_TFSEL(ext_port) |
+ IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) |
+ IMX_AUDMUX_V2_PTCR_TFSDIR |
+ IMX_AUDMUX_V2_PTCR_TCLKDIR,
+ IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port));
+ if (ret) {
+ dev_err(&pdev->dev, "audmux internal port setup failed\n");
+ return ret;
+ }
+ imx_audmux_v2_configure_port(ext_port,
+ IMX_AUDMUX_V2_PTCR_SYN |
+ IMX_AUDMUX_V2_PTCR_TCSEL(int_port),
+ IMX_AUDMUX_V2_PDCR_RXDSEL(int_port));
+ if (ret) {
+ dev_err(&pdev->dev, "audmux external port setup failed\n");
+ return ret;
+ }
+
+ ssi_np = of_parse_phandle(pdev->dev.of_node, "ssi-controller", 0);
+ codec_np = of_parse_phandle(pdev->dev.of_node, "audio-codec", 0);
+ if (!ssi_np || !codec_np) {
+ dev_err(&pdev->dev, "phandle missing or invalid\n");
+ ret = -EINVAL;
+ goto fail;
+ }
+
+ ssi_pdev = of_find_device_by_node(ssi_np);
+ if (!ssi_pdev) {
+ dev_err(&pdev->dev, "failed to find SSI platform device\n");
+ ret = -EINVAL;
+ goto fail;
+ }
+ codec_dev = of_find_i2c_device_by_node(codec_np);
+ if (!codec_dev) {
+ dev_err(&pdev->dev, "failed to find codec platform device\n");
+ return -EINVAL;
+ }
+
+ data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL);
+ if (!data) {
+ ret = -ENOMEM;
+ goto fail;
+ }
+
+ data->codec_clk = clk_get(&codec_dev->dev, NULL);
+ if (IS_ERR(data->codec_clk)) {
+ /* assuming clock enabled by default */
+ data->codec_clk = NULL;
+ ret = of_property_read_u32(codec_np, "clock-frequency",
+ &data->clk_frequency);
+ if (ret) {
+ dev_err(&codec_dev->dev,
+ "clock-frequency missing or invalid\n");
+ goto fail;
+ }
+ } else {
+ data->clk_frequency = clk_get_rate(data->codec_clk);
+ clk_prepare_enable(data->codec_clk);
+ }
+
+ data->dai.name = "HiFi";
+ data->dai.stream_name = "HiFi";
+ data->dai.codec_dai_name = "sgtl5000";
+ data->dai.codec_of_node = codec_np;
+ data->dai.cpu_dai_name = dev_name(&ssi_pdev->dev);
+ data->dai.platform_name = "imx-pcm-audio";
+ data->dai.init = &imx_sgtl5000_dai_init;
+ data->dai.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBM_CFM;
+
+ data->card.dev = &pdev->dev;
+ ret = snd_soc_of_parse_card_name(&data->card, "model");
+ if (ret)
+ goto clk_fail;
+ ret = snd_soc_of_parse_audio_routing(&data->card, "audio-routing");
+ if (ret)
+ goto clk_fail;
+ data->card.num_links = 1;
+ data->card.dai_link = &data->dai;
+ data->card.dapm_widgets = imx_sgtl5000_dapm_widgets;
+ data->card.num_dapm_widgets = ARRAY_SIZE(imx_sgtl5000_dapm_widgets);
+
+ ret = snd_soc_register_card(&data->card);
+ if (ret) {
+ dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret);
+ goto clk_fail;
+ }
+
+ platform_set_drvdata(pdev, data);
+clk_fail:
+ clk_put(data->codec_clk);
+fail:
+ if (ssi_np)
+ of_node_put(ssi_np);
+ if (codec_np)
+ of_node_put(codec_np);
+
+ return ret;
+}
+
+static int __devexit imx_sgtl5000_remove(struct platform_device *pdev)
+{
+ struct imx_sgtl5000_data *data = platform_get_drvdata(pdev);
+
+ if (data->codec_clk) {
+ clk_disable_unprepare(data->codec_clk);
+ clk_put(data->codec_clk);
+ }
+ snd_soc_unregister_card(&data->card);
+
+ return 0;
+}
+
+static const struct of_device_id imx_sgtl5000_dt_ids[] = {
+ { .compatible = "fsl,imx-audio-sgtl5000", },
+ { /* sentinel */ }
+};
+MODULE_DEVICE_TABLE(of, imx_sgtl5000_dt_ids);
+
+static struct platform_driver imx_sgtl5000_driver = {
+ .driver = {
+ .name = "imx-sgtl5000",
+ .owner = THIS_MODULE,
+ .of_match_table = imx_sgtl5000_dt_ids,
+ },
+ .probe = imx_sgtl5000_probe,
+ .remove = __devexit_p(imx_sgtl5000_remove),
+};
+module_platform_driver(imx_sgtl5000_driver);
+
+MODULE_AUTHOR("Shawn Guo <shawn.guo@linaro.org>");
+MODULE_DESCRIPTION("Freescale i.MX SGTL5000 ASoC machine driver");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:imx-sgtl5000");
diff --git a/sound/soc/imx/imx-ssi.c b/sound/soc/fsl/imx-ssi.c
index 4f81ed45632..28dd76c7cb1 100644
--- a/sound/soc/imx/imx-ssi.c
+++ b/sound/soc/fsl/imx-ssi.c
@@ -28,7 +28,7 @@
* value. When we read the same register two times (and the register still
* contains the same value) these status bits are not set. We work
* around this by not polling these bits but only wait a fixed delay.
- *
+ *
*/
#include <linux/clk.h>
@@ -543,7 +543,7 @@ static int imx_ssi_probe(struct platform_device *pdev)
ret);
goto failed_clk;
}
- clk_enable(ssi->clk);
+ clk_prepare_enable(ssi->clk);
res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
if (!res) {
@@ -641,7 +641,7 @@ failed_ac97:
failed_ioremap:
release_mem_region(res->start, resource_size(res));
failed_get_resource:
- clk_disable(ssi->clk);
+ clk_disable_unprepare(ssi->clk);
clk_put(ssi->clk);
failed_clk:
kfree(ssi);
@@ -664,7 +664,7 @@ static int __devexit imx_ssi_remove(struct platform_device *pdev)
iounmap(ssi->base);
release_mem_region(res->start, resource_size(res));
- clk_disable(ssi->clk);
+ clk_disable_unprepare(ssi->clk);
clk_put(ssi->clk);
kfree(ssi);
diff --git a/sound/soc/imx/imx-ssi.h b/sound/soc/fsl/imx-ssi.h
index 5744e86ca87..5744e86ca87 100644
--- a/sound/soc/imx/imx-ssi.h
+++ b/sound/soc/fsl/imx-ssi.h
diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c
index 3fea5a15ffe..60bcba1bc30 100644
--- a/sound/soc/fsl/mpc8610_hpcd.c
+++ b/sound/soc/fsl/mpc8610_hpcd.c
@@ -14,18 +14,16 @@
#include <linux/interrupt.h>
#include <linux/of_device.h>
#include <linux/slab.h>
-#include <linux/of_i2c.h>
#include <sound/soc.h>
#include <asm/fsl_guts.h>
#include "fsl_dma.h"
#include "fsl_ssi.h"
+#include "fsl_utils.h"
/* There's only one global utilities register */
static phys_addr_t guts_phys;
-#define DAI_NAME_SIZE 32
-
/**
* mpc8610_hpcd_data: machine-specific ASoC device data
*
@@ -43,7 +41,6 @@ struct mpc8610_hpcd_data {
unsigned int dma_id[2]; /* 0 = DMA1, 1 = DMA2, etc */
unsigned int dma_channel_id[2]; /* 0 = ch 0, 1 = ch 1, etc*/
char codec_dai_name[DAI_NAME_SIZE];
- char codec_name[DAI_NAME_SIZE];
char platform_name[2][DAI_NAME_SIZE]; /* One for each DMA channel */
};
@@ -181,141 +178,6 @@ static struct snd_soc_ops mpc8610_hpcd_ops = {
};
/**
- * get_node_by_phandle_name - get a node by its phandle name
- *
- * This function takes a node, the name of a property in that node, and a
- * compatible string. Assuming the property is a phandle to another node,
- * it returns that node, (optionally) if that node is compatible.
- *
- * If the property is not a phandle, or the node it points to is not compatible
- * with the specific string, then NULL is returned.
- */
-static struct device_node *get_node_by_phandle_name(struct device_node *np,
- const char *name,
- const char *compatible)
-{
- const phandle *ph;
- int len;
-
- ph = of_get_property(np, name, &len);
- if (!ph || (len != sizeof(phandle)))
- return NULL;
-
- np = of_find_node_by_phandle(*ph);
- if (!np)
- return NULL;
-
- if (compatible && !of_device_is_compatible(np, compatible)) {
- of_node_put(np);
- return NULL;
- }
-
- return np;
-}
-
-/**
- * get_parent_cell_index -- return the cell-index of the parent of a node
- *
- * Return the value of the cell-index property of the parent of the given
- * node. This is used for DMA channel nodes that need to know the DMA ID
- * of the controller they are on.
- */
-static int get_parent_cell_index(struct device_node *np)
-{
- struct device_node *parent = of_get_parent(np);
- const u32 *iprop;
-
- if (!parent)
- return -1;
-
- iprop = of_get_property(parent, "cell-index", NULL);
- of_node_put(parent);
-
- if (!iprop)
- return -1;
-
- return be32_to_cpup(iprop);
-}
-
-/**
- * codec_node_dev_name - determine the dev_name for a codec node
- *
- * This function determines the dev_name for an I2C node. This is the name
- * that would be returned by dev_name() if this device_node were part of a
- * 'struct device' It's ugly and hackish, but it works.
- *
- * The dev_name for such devices include the bus number and I2C address. For
- * example, "cs4270.0-004f".
- */
-static int codec_node_dev_name(struct device_node *np, char *buf, size_t len)
-{
- const u32 *iprop;
- int addr;
- char temp[DAI_NAME_SIZE];
- struct i2c_client *i2c;
-
- of_modalias_node(np, temp, DAI_NAME_SIZE);
-
- iprop = of_get_property(np, "reg", NULL);
- if (!iprop)
- return -EINVAL;
-
- addr = be32_to_cpup(iprop);
-
- /* We need the adapter number */
- i2c = of_find_i2c_device_by_node(np);
- if (!i2c)
- return -ENODEV;
-
- snprintf(buf, len, "%s.%u-%04x", temp, i2c->adapter->nr, addr);
-
- return 0;
-}
-
-static int get_dma_channel(struct device_node *ssi_np,
- const char *name,
- struct snd_soc_dai_link *dai,
- unsigned int *dma_channel_id,
- unsigned int *dma_id)
-{
- struct resource res;
- struct device_node *dma_channel_np;
- const u32 *iprop;
- int ret;
-
- dma_channel_np = get_node_by_phandle_name(ssi_np, name,
- "fsl,ssi-dma-channel");
- if (!dma_channel_np)
- return -EINVAL;
-
- /* Determine the dev_name for the device_node. This code mimics the
- * behavior of of_device_make_bus_id(). We need this because ASoC uses
- * the dev_name() of the device to match the platform (DMA) device with
- * the CPU (SSI) device. It's all ugly and hackish, but it works (for
- * now).
- *
- * dai->platform name should already point to an allocated buffer.
- */
- ret = of_address_to_resource(dma_channel_np, 0, &res);
- if (ret)
- return ret;
- snprintf((char *)dai->platform_name, DAI_NAME_SIZE, "%llx.%s",
- (unsigned long long) res.start, dma_channel_np->name);
-
- iprop = of_get_property(dma_channel_np, "cell-index", NULL);
- if (!iprop) {
- of_node_put(dma_channel_np);
- return -EINVAL;
- }
-
- *dma_channel_id = be32_to_cpup(iprop);
- *dma_id = get_parent_cell_index(dma_channel_np);
- of_node_put(dma_channel_np);
-
- return 0;
-}
-
-/**
* mpc8610_hpcd_probe: platform probe function for the machine driver
*
* Although this is a machine driver, the SSI node is the "master" node with
@@ -352,16 +214,8 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev)
machine_data->dai[0].cpu_dai_name = dev_name(&ssi_pdev->dev);
machine_data->dai[0].ops = &mpc8610_hpcd_ops;
- /* Determine the codec name, it will be used as the codec DAI name */
- ret = codec_node_dev_name(codec_np, machine_data->codec_name,
- DAI_NAME_SIZE);
- if (ret) {
- dev_err(&pdev->dev, "invalid codec node %s\n",
- codec_np->full_name);
- ret = -EINVAL;
- goto error;
- }
- machine_data->dai[0].codec_name = machine_data->codec_name;
+ /* ASoC core can match codec with device node */
+ machine_data->dai[0].codec_of_node = codec_np;
/* The DAI name from the codec (snd_soc_dai_driver.name) */
machine_data->dai[0].codec_dai_name = "cs4270-hifi";
@@ -458,9 +312,10 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev)
/* Find the playback DMA channel to use. */
machine_data->dai[0].platform_name = machine_data->platform_name[0];
- ret = get_dma_channel(np, "fsl,playback-dma", &machine_data->dai[0],
- &machine_data->dma_channel_id[0],
- &machine_data->dma_id[0]);
+ ret = fsl_asoc_get_dma_channel(np, "fsl,playback-dma",
+ &machine_data->dai[0],
+ &machine_data->dma_channel_id[0],
+ &machine_data->dma_id[0]);
if (ret) {
dev_err(&pdev->dev, "missing/invalid playback DMA phandle\n");
goto error;
@@ -468,9 +323,10 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev)
/* Find the capture DMA channel to use. */
machine_data->dai[1].platform_name = machine_data->platform_name[1];
- ret = get_dma_channel(np, "fsl,capture-dma", &machine_data->dai[1],
- &machine_data->dma_channel_id[1],
- &machine_data->dma_id[1]);
+ ret = fsl_asoc_get_dma_channel(np, "fsl,capture-dma",
+ &machine_data->dai[1],
+ &machine_data->dma_channel_id[1],
+ &machine_data->dma_id[1]);
if (ret) {
dev_err(&pdev->dev, "missing/invalid capture DMA phandle\n");
goto error;
diff --git a/sound/soc/imx/mx27vis-aic32x4.c b/sound/soc/fsl/mx27vis-aic32x4.c
index f6d04ad4bb3..f6d04ad4bb3 100644
--- a/sound/soc/imx/mx27vis-aic32x4.c
+++ b/sound/soc/fsl/mx27vis-aic32x4.c
diff --git a/sound/soc/fsl/p1022_ds.c b/sound/soc/fsl/p1022_ds.c
index 982a1c94498..50adf4032bc 100644
--- a/sound/soc/fsl/p1022_ds.c
+++ b/sound/soc/fsl/p1022_ds.c
@@ -14,12 +14,12 @@
#include <linux/interrupt.h>
#include <linux/of_device.h>
#include <linux/slab.h>
-#include <linux/of_i2c.h>
#include <sound/soc.h>
#include <asm/fsl_guts.h>
#include "fsl_dma.h"
#include "fsl_ssi.h"
+#include "fsl_utils.h"
/* P1022-specific PMUXCR and DMUXCR bit definitions */
@@ -57,8 +57,6 @@ static inline void guts_set_dmuxcr(struct ccsr_guts __iomem *guts,
/* There's only one global utilities register */
static phys_addr_t guts_phys;
-#define DAI_NAME_SIZE 32
-
/**
* machine_data: machine-specific ASoC device data
*
@@ -75,7 +73,6 @@ struct machine_data {
unsigned int ssi_id; /* 0 = SSI1, 1 = SSI2, etc */
unsigned int dma_id[2]; /* 0 = DMA1, 1 = DMA2, etc */
unsigned int dma_channel_id[2]; /* 0 = ch 0, 1 = ch 1, etc*/
- char codec_name[DAI_NAME_SIZE];
char platform_name[2][DAI_NAME_SIZE]; /* One for each DMA channel */
};
@@ -191,136 +188,6 @@ static struct snd_soc_ops p1022_ds_ops = {
};
/**
- * get_node_by_phandle_name - get a node by its phandle name
- *
- * This function takes a node, the name of a property in that node, and a
- * compatible string. Assuming the property is a phandle to another node,
- * it returns that node, (optionally) if that node is compatible.
- *
- * If the property is not a phandle, or the node it points to is not compatible
- * with the specific string, then NULL is returned.
- */
-static struct device_node *get_node_by_phandle_name(struct device_node *np,
- const char *name, const char *compatible)
-{
- np = of_parse_phandle(np, name, 0);
- if (!np)
- return NULL;
-
- if (!of_device_is_compatible(np, compatible)) {
- of_node_put(np);
- return NULL;
- }
-
- return np;
-}
-
-/**
- * get_parent_cell_index -- return the cell-index of the parent of a node
- *
- * Return the value of the cell-index property of the parent of the given
- * node. This is used for DMA channel nodes that need to know the DMA ID
- * of the controller they are on.
- */
-static int get_parent_cell_index(struct device_node *np)
-{
- struct device_node *parent = of_get_parent(np);
- const u32 *iprop;
- int ret = -1;
-
- if (!parent)
- return -1;
-
- iprop = of_get_property(parent, "cell-index", NULL);
- if (iprop)
- ret = be32_to_cpup(iprop);
-
- of_node_put(parent);
-
- return ret;
-}
-
-/**
- * codec_node_dev_name - determine the dev_name for a codec node
- *
- * This function determines the dev_name for an I2C node. This is the name
- * that would be returned by dev_name() if this device_node were part of a
- * 'struct device' It's ugly and hackish, but it works.
- *
- * The dev_name for such devices include the bus number and I2C address. For
- * example, "cs4270-codec.0-004f".
- */
-static int codec_node_dev_name(struct device_node *np, char *buf, size_t len)
-{
- const u32 *iprop;
- int addr;
- char temp[DAI_NAME_SIZE];
- struct i2c_client *i2c;
-
- of_modalias_node(np, temp, DAI_NAME_SIZE);
-
- iprop = of_get_property(np, "reg", NULL);
- if (!iprop)
- return -EINVAL;
-
- addr = be32_to_cpup(iprop);
-
- /* We need the adapter number */
- i2c = of_find_i2c_device_by_node(np);
- if (!i2c)
- return -ENODEV;
-
- snprintf(buf, len, "%s.%u-%04x", temp, i2c->adapter->nr, addr);
-
- return 0;
-}
-
-static int get_dma_channel(struct device_node *ssi_np,
- const char *name,
- struct snd_soc_dai_link *dai,
- unsigned int *dma_channel_id,
- unsigned int *dma_id)
-{
- struct resource res;
- struct device_node *dma_channel_np;
- const u32 *iprop;
- int ret;
-
- dma_channel_np = get_node_by_phandle_name(ssi_np, name,
- "fsl,ssi-dma-channel");
- if (!dma_channel_np)
- return -EINVAL;
-
- /* Determine the dev_name for the device_node. This code mimics the
- * behavior of of_device_make_bus_id(). We need this because ASoC uses
- * the dev_name() of the device to match the platform (DMA) device with
- * the CPU (SSI) device. It's all ugly and hackish, but it works (for
- * now).
- *
- * dai->platform name should already point to an allocated buffer.
- */
- ret = of_address_to_resource(dma_channel_np, 0, &res);
- if (ret) {
- of_node_put(dma_channel_np);
- return ret;
- }
- snprintf((char *)dai->platform_name, DAI_NAME_SIZE, "%llx.%s",
- (unsigned long long) res.start, dma_channel_np->name);
-
- iprop = of_get_property(dma_channel_np, "cell-index", NULL);
- if (!iprop) {
- of_node_put(dma_channel_np);
- return -EINVAL;
- }
-
- *dma_channel_id = be32_to_cpup(iprop);
- *dma_id = get_parent_cell_index(dma_channel_np);
- of_node_put(dma_channel_np);
-
- return 0;
-}
-
-/**
* p1022_ds_probe: platform probe function for the machine driver
*
* Although this is a machine driver, the SSI node is the "master" node with
@@ -357,15 +224,8 @@ static int p1022_ds_probe(struct platform_device *pdev)
mdata->dai[0].cpu_dai_name = dev_name(&ssi_pdev->dev);
mdata->dai[0].ops = &p1022_ds_ops;
- /* Determine the codec name, it will be used as the codec DAI name */
- ret = codec_node_dev_name(codec_np, mdata->codec_name, DAI_NAME_SIZE);
- if (ret) {
- dev_err(&pdev->dev, "invalid codec node %s\n",
- codec_np->full_name);
- ret = -EINVAL;
- goto error;
- }
- mdata->dai[0].codec_name = mdata->codec_name;
+ /* ASoC core can match codec with device node */
+ mdata->dai[0].codec_of_node = codec_np;
/* We register two DAIs per SSI, one for playback and the other for
* capture. We support codecs that have separate DAIs for both playback
@@ -462,9 +322,9 @@ static int p1022_ds_probe(struct platform_device *pdev)
/* Find the playback DMA channel to use. */
mdata->dai[0].platform_name = mdata->platform_name[0];
- ret = get_dma_channel(np, "fsl,playback-dma", &mdata->dai[0],
- &mdata->dma_channel_id[0],
- &mdata->dma_id[0]);
+ ret = fsl_asoc_get_dma_channel(np, "fsl,playback-dma", &mdata->dai[0],
+ &mdata->dma_channel_id[0],
+ &mdata->dma_id[0]);
if (ret) {
dev_err(&pdev->dev, "missing/invalid playback DMA phandle\n");
goto error;
@@ -472,9 +332,9 @@ static int p1022_ds_probe(struct platform_device *pdev)
/* Find the capture DMA channel to use. */
mdata->dai[1].platform_name = mdata->platform_name[1];
- ret = get_dma_channel(np, "fsl,capture-dma", &mdata->dai[1],
- &mdata->dma_channel_id[1],
- &mdata->dma_id[1]);
+ ret = fsl_asoc_get_dma_channel(np, "fsl,capture-dma", &mdata->dai[1],
+ &mdata->dma_channel_id[1],
+ &mdata->dma_id[1]);
if (ret) {
dev_err(&pdev->dev, "missing/invalid capture DMA phandle\n");
goto error;
diff --git a/sound/soc/imx/phycore-ac97.c b/sound/soc/fsl/phycore-ac97.c
index f8da6dd115e..f8da6dd115e 100644
--- a/sound/soc/imx/phycore-ac97.c
+++ b/sound/soc/fsl/phycore-ac97.c
diff --git a/sound/soc/imx/wm1133-ev1.c b/sound/soc/fsl/wm1133-ev1.c
index fe54a69073e..fe54a69073e 100644
--- a/sound/soc/imx/wm1133-ev1.c
+++ b/sound/soc/fsl/wm1133-ev1.c
diff --git a/sound/soc/generic/Kconfig b/sound/soc/generic/Kconfig
new file mode 100644
index 00000000000..610f6125164
--- /dev/null
+++ b/sound/soc/generic/Kconfig
@@ -0,0 +1,4 @@
+config SND_SIMPLE_CARD
+ tristate "ASoC Simple sound card support"
+ help
+ This option enables generic simple sound card support
diff --git a/sound/soc/generic/Makefile b/sound/soc/generic/Makefile
new file mode 100644
index 00000000000..9c3b246792b
--- /dev/null
+++ b/sound/soc/generic/Makefile
@@ -0,0 +1,3 @@
+snd-soc-simple-card-objs := simple-card.o
+
+obj-$(CONFIG_SND_SIMPLE_CARD) += snd-soc-simple-card.o
diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c
new file mode 100644
index 00000000000..b4b4cab3023
--- /dev/null
+++ b/sound/soc/generic/simple-card.c
@@ -0,0 +1,114 @@
+/*
+ * ASoC simple sound card support
+ *
+ * Copyright (C) 2012 Renesas Solutions Corp.
+ * Kuninori Morimoto <kuninori.morimoto.gx@renesas.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/platform_device.h>
+#include <linux/module.h>
+#include <sound/simple_card.h>
+
+#define asoc_simple_get_card_info(p) \
+ container_of(p->dai_link, struct asoc_simple_card_info, snd_link)
+
+static int asoc_simple_card_dai_init(struct snd_soc_pcm_runtime *rtd)
+{
+ struct asoc_simple_card_info *cinfo = asoc_simple_get_card_info(rtd);
+ struct asoc_simple_dai_init_info *iinfo = cinfo->init;
+ struct snd_soc_dai *codec = rtd->codec_dai;
+ struct snd_soc_dai *cpu = rtd->cpu_dai;
+ unsigned int cpu_daifmt = iinfo->fmt | iinfo->cpu_daifmt;
+ unsigned int codec_daifmt = iinfo->fmt | iinfo->codec_daifmt;
+ int ret;
+
+ if (codec_daifmt) {
+ ret = snd_soc_dai_set_fmt(codec, codec_daifmt);
+ if (ret < 0)
+ return ret;
+ }
+
+ if (iinfo->sysclk) {
+ ret = snd_soc_dai_set_sysclk(codec, 0, iinfo->sysclk, 0);
+ if (ret < 0)
+ return ret;
+ }
+
+ if (cpu_daifmt) {
+ ret = snd_soc_dai_set_fmt(cpu, cpu_daifmt);
+ if (ret < 0)
+ return ret;
+ }
+
+ return 0;
+}
+
+static int asoc_simple_card_probe(struct platform_device *pdev)
+{
+ struct asoc_simple_card_info *cinfo = pdev->dev.platform_data;
+
+ if (!cinfo) {
+ dev_err(&pdev->dev, "no info for asoc-simple-card\n");
+ return -EINVAL;
+ }
+
+ if (!cinfo->name ||
+ !cinfo->card ||
+ !cinfo->cpu_dai ||
+ !cinfo->codec ||
+ !cinfo->platform ||
+ !cinfo->codec_dai) {
+ dev_err(&pdev->dev, "insufficient asoc_simple_card_info settings\n");
+ return -EINVAL;
+ }
+
+ /*
+ * init snd_soc_dai_link
+ */
+ cinfo->snd_link.name = cinfo->name;
+ cinfo->snd_link.stream_name = cinfo->name;
+ cinfo->snd_link.cpu_dai_name = cinfo->cpu_dai;
+ cinfo->snd_link.platform_name = cinfo->platform;
+ cinfo->snd_link.codec_name = cinfo->codec;
+ cinfo->snd_link.codec_dai_name = cinfo->codec_dai;
+
+ /* enable snd_link.init if cinfo has settings */
+ if (cinfo->init)
+ cinfo->snd_link.init = asoc_simple_card_dai_init;
+
+ /*
+ * init snd_soc_card
+ */
+ cinfo->snd_card.name = cinfo->card;
+ cinfo->snd_card.owner = THIS_MODULE;
+ cinfo->snd_card.dai_link = &cinfo->snd_link;
+ cinfo->snd_card.num_links = 1;
+ cinfo->snd_card.dev = &pdev->dev;
+
+ return snd_soc_register_card(&cinfo->snd_card);
+}
+
+static int asoc_simple_card_remove(struct platform_device *pdev)
+{
+ struct asoc_simple_card_info *cinfo = pdev->dev.platform_data;
+
+ return snd_soc_unregister_card(&cinfo->snd_card);
+}
+
+static struct platform_driver asoc_simple_card = {
+ .driver = {
+ .name = "asoc-simple-card",
+ },
+ .probe = asoc_simple_card_probe,
+ .remove = asoc_simple_card_remove,
+};
+
+module_platform_driver(asoc_simple_card);
+
+MODULE_LICENSE("GPL");
+MODULE_DESCRIPTION("ASoC Simple Sound Card");
+MODULE_AUTHOR("Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>");
diff --git a/sound/soc/imx/Kconfig b/sound/soc/imx/Kconfig
deleted file mode 100644
index 810acaa0900..00000000000
--- a/sound/soc/imx/Kconfig
+++ /dev/null
@@ -1,79 +0,0 @@
-menuconfig SND_IMX_SOC
- tristate "SoC Audio for Freescale i.MX CPUs"
- depends on ARCH_MXC
- help
- Say Y or M if you want to add support for codecs attached to
- the i.MX SSI interface.
-
-
-if SND_IMX_SOC
-
-config SND_SOC_IMX_SSI
- tristate
-
-config SND_SOC_IMX_PCM
- tristate
-
-config SND_MXC_SOC_FIQ
- tristate
- select FIQ
- select SND_SOC_IMX_PCM
-
-config SND_MXC_SOC_MX2
- select SND_SOC_DMAENGINE_PCM
- tristate
- select SND_SOC_IMX_PCM
-
-config SND_SOC_IMX_AUDMUX
- tristate
-
-config SND_MXC_SOC_WM1133_EV1
- tristate "Audio on the the i.MX31ADS with WM1133-EV1 fitted"
- depends on MACH_MX31ADS_WM1133_EV1 && EXPERIMENTAL
- select SND_SOC_WM8350
- select SND_MXC_SOC_FIQ
- select SND_SOC_IMX_AUDMUX
- select SND_SOC_IMX_SSI
- help
- Enable support for audio on the i.MX31ADS with the WM1133-EV1
- PMIC board with WM8835x fitted.
-
-config SND_SOC_MX27VIS_AIC32X4
- tristate "SoC audio support for Visstrim M10 boards"
- depends on MACH_IMX27_VISSTRIM_M10 && I2C
- select SND_SOC_TLV320AIC32X4
- select SND_MXC_SOC_MX2
- select SND_SOC_IMX_AUDMUX
- select SND_SOC_IMX_SSI
- help
- Say Y if you want to add support for SoC audio on Visstrim SM10
- board with TLV320AIC32X4 codec.
-
-config SND_SOC_PHYCORE_AC97
- tristate "SoC Audio support for Phytec phyCORE (and phyCARD) boards"
- depends on MACH_PCM043 || MACH_PCA100
- select SND_SOC_AC97_BUS
- select SND_SOC_WM9712
- select SND_MXC_SOC_FIQ
- select SND_SOC_IMX_AUDMUX
- select SND_SOC_IMX_SSI
- help
- Say Y if you want to add support for SoC audio on Phytec phyCORE
- and phyCARD boards in AC97 mode
-
-config SND_SOC_EUKREA_TLV320
- tristate "Eukrea TLV320"
- depends on MACH_EUKREA_MBIMX27_BASEBOARD \
- || MACH_EUKREA_MBIMXSD25_BASEBOARD \
- || MACH_EUKREA_MBIMXSD35_BASEBOARD \
- || MACH_EUKREA_MBIMXSD51_BASEBOARD
- depends on I2C
- select SND_SOC_TLV320AIC23
- select SND_MXC_SOC_FIQ
- select SND_SOC_IMX_AUDMUX
- select SND_SOC_IMX_SSI
- help
- Enable I2S based access to the TLV320AIC23B codec attached
- to the SSI interface
-
-endif # SND_IMX_SOC
diff --git a/sound/soc/imx/Makefile b/sound/soc/imx/Makefile
deleted file mode 100644
index f5db3e92d0d..00000000000
--- a/sound/soc/imx/Makefile
+++ /dev/null
@@ -1,22 +0,0 @@
-# i.MX Platform Support
-snd-soc-imx-ssi-objs := imx-ssi.o
-snd-soc-imx-audmux-objs := imx-audmux.o
-
-obj-$(CONFIG_SND_SOC_IMX_SSI) += snd-soc-imx-ssi.o
-obj-$(CONFIG_SND_SOC_IMX_AUDMUX) += snd-soc-imx-audmux.o
-
-obj-$(CONFIG_SND_SOC_IMX_PCM) += snd-soc-imx-pcm.o
-snd-soc-imx-pcm-y := imx-pcm.o
-snd-soc-imx-pcm-$(CONFIG_SND_MXC_SOC_FIQ) += imx-pcm-fiq.o
-snd-soc-imx-pcm-$(CONFIG_SND_MXC_SOC_MX2) += imx-pcm-dma-mx2.o
-
-# i.MX Machine Support
-snd-soc-eukrea-tlv320-objs := eukrea-tlv320.o
-snd-soc-phycore-ac97-objs := phycore-ac97.o
-snd-soc-mx27vis-aic32x4-objs := mx27vis-aic32x4.o
-snd-soc-wm1133-ev1-objs := wm1133-ev1.o
-
-obj-$(CONFIG_SND_SOC_EUKREA_TLV320) += snd-soc-eukrea-tlv320.o
-obj-$(CONFIG_SND_SOC_PHYCORE_AC97) += snd-soc-phycore-ac97.o
-obj-$(CONFIG_SND_SOC_MX27VIS_AIC32X4) += snd-soc-mx27vis-aic32x4.o
-obj-$(CONFIG_SND_MXC_SOC_WM1133_EV1) += snd-soc-wm1133-ev1.o
diff --git a/sound/soc/jz4740/jz4740-i2s.c b/sound/soc/jz4740/jz4740-i2s.c
index a5af7c42e62..41349670ada 100644
--- a/sound/soc/jz4740/jz4740-i2s.c
+++ b/sound/soc/jz4740/jz4740-i2s.c
@@ -346,7 +346,7 @@ static void jz4740_i2c_init_pcm_config(struct jz4740_i2s *i2s)
/* Playback */
dma_config = &i2s->pcm_config_playback.dma_config;
- dma_config->src_width = JZ4740_DMA_WIDTH_32BIT,
+ dma_config->src_width = JZ4740_DMA_WIDTH_32BIT;
dma_config->transfer_size = JZ4740_DMA_TRANSFER_SIZE_16BYTE;
dma_config->request_type = JZ4740_DMA_TYPE_AIC_TRANSMIT;
dma_config->flags = JZ4740_DMA_SRC_AUTOINC;
@@ -355,7 +355,7 @@ static void jz4740_i2c_init_pcm_config(struct jz4740_i2s *i2s)
/* Capture */
dma_config = &i2s->pcm_config_capture.dma_config;
- dma_config->dst_width = JZ4740_DMA_WIDTH_32BIT,
+ dma_config->dst_width = JZ4740_DMA_WIDTH_32BIT;
dma_config->transfer_size = JZ4740_DMA_TRANSFER_SIZE_16BYTE;
dma_config->request_type = JZ4740_DMA_TYPE_AIC_RECEIVE;
dma_config->flags = JZ4740_DMA_DST_AUTOINC;
diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c
index 3cb9aa4299d..fa455675045 100644
--- a/sound/soc/kirkwood/kirkwood-i2s.c
+++ b/sound/soc/kirkwood/kirkwood-i2s.c
@@ -17,6 +17,7 @@
#include <linux/slab.h>
#include <linux/mbus.h>
#include <linux/delay.h>
+#include <linux/clk.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
#include <sound/soc.h>
@@ -449,6 +450,14 @@ static __devinit int kirkwood_i2s_dev_probe(struct platform_device *pdev)
priv->burst = data->burst;
+ priv->clk = clk_get(&pdev->dev, NULL);
+ if (IS_ERR(priv->clk)) {
+ dev_err(&pdev->dev, "no clock\n");
+ err = PTR_ERR(priv->clk);
+ goto err_ioremap;
+ }
+ clk_prepare_enable(priv->clk);
+
return snd_soc_register_dai(&pdev->dev, &kirkwood_i2s_dai);
err_ioremap:
@@ -466,6 +475,10 @@ static __devexit int kirkwood_i2s_dev_remove(struct platform_device *pdev)
struct kirkwood_dma_data *priv = dev_get_drvdata(&pdev->dev);
snd_soc_unregister_dai(&pdev->dev);
+
+ clk_disable_unprepare(priv->clk);
+ clk_put(priv->clk);
+
iounmap(priv->io);
release_mem_region(priv->mem->start, SZ_16K);
kfree(priv);
diff --git a/sound/soc/kirkwood/kirkwood.h b/sound/soc/kirkwood/kirkwood.h
index 9047436b393..f9084d83e6b 100644
--- a/sound/soc/kirkwood/kirkwood.h
+++ b/sound/soc/kirkwood/kirkwood.h
@@ -123,6 +123,7 @@ struct kirkwood_dma_data {
void __iomem *io;
int irq;
int burst;
+ struct clk *clk;
};
#endif
diff --git a/sound/soc/mxs/mxs-pcm.c b/sound/soc/mxs/mxs-pcm.c
index e373fbbc97a..373dec90579 100644
--- a/sound/soc/mxs/mxs-pcm.c
+++ b/sound/soc/mxs/mxs-pcm.c
@@ -220,28 +220,16 @@ static struct snd_soc_platform_driver mxs_soc_platform = {
.pcm_free = mxs_pcm_free,
};
-static int __devinit mxs_soc_platform_probe(struct platform_device *pdev)
+int __devinit mxs_pcm_platform_register(struct device *dev)
{
- return snd_soc_register_platform(&pdev->dev, &mxs_soc_platform);
+ return snd_soc_register_platform(dev, &mxs_soc_platform);
}
+EXPORT_SYMBOL_GPL(mxs_pcm_platform_register);
-static int __devexit mxs_soc_platform_remove(struct platform_device *pdev)
+void __devexit mxs_pcm_platform_unregister(struct device *dev)
{
- snd_soc_unregister_platform(&pdev->dev);
-
- return 0;
+ snd_soc_unregister_platform(dev);
}
-
-static struct platform_driver mxs_pcm_driver = {
- .driver = {
- .name = "mxs-pcm-audio",
- .owner = THIS_MODULE,
- },
- .probe = mxs_soc_platform_probe,
- .remove = __devexit_p(mxs_soc_platform_remove),
-};
-
-module_platform_driver(mxs_pcm_driver);
+EXPORT_SYMBOL_GPL(mxs_pcm_platform_unregister);
MODULE_LICENSE("GPL");
-MODULE_ALIAS("platform:mxs-pcm-audio");
diff --git a/sound/soc/mxs/mxs-pcm.h b/sound/soc/mxs/mxs-pcm.h
index 5f01a9124b3..35ba2ca4238 100644
--- a/sound/soc/mxs/mxs-pcm.h
+++ b/sound/soc/mxs/mxs-pcm.h
@@ -24,4 +24,7 @@ struct mxs_pcm_dma_params {
int chan_num;
};
+int mxs_pcm_platform_register(struct device *dev);
+void mxs_pcm_platform_unregister(struct device *dev);
+
#endif
diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c
index 53f4fd8fece..aba71bfa33b 100644
--- a/sound/soc/mxs/mxs-saif.c
+++ b/sound/soc/mxs/mxs-saif.c
@@ -18,6 +18,8 @@
#include <linux/module.h>
#include <linux/init.h>
+#include <linux/of.h>
+#include <linux/of_device.h>
#include <linux/platform_device.h>
#include <linux/slab.h>
#include <linux/dma-mapping.h>
@@ -25,6 +27,7 @@
#include <linux/delay.h>
#include <linux/time.h>
#include <linux/fsl/mxs-dma.h>
+#include <linux/pinctrl/consumer.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
@@ -620,34 +623,61 @@ static irqreturn_t mxs_saif_irq(int irq, void *dev_id)
return IRQ_HANDLED;
}
-static int mxs_saif_probe(struct platform_device *pdev)
+static int __devinit mxs_saif_probe(struct platform_device *pdev)
{
+ struct device_node *np = pdev->dev.of_node;
struct resource *iores, *dmares;
struct mxs_saif *saif;
struct mxs_saif_platform_data *pdata;
+ struct pinctrl *pinctrl;
int ret = 0;
- if (pdev->id >= ARRAY_SIZE(mxs_saif))
+
+ if (!np && pdev->id >= ARRAY_SIZE(mxs_saif))
return -EINVAL;
saif = devm_kzalloc(&pdev->dev, sizeof(*saif), GFP_KERNEL);
if (!saif)
return -ENOMEM;
- mxs_saif[pdev->id] = saif;
- saif->id = pdev->id;
-
- pdata = pdev->dev.platform_data;
- if (pdata && !pdata->master_mode) {
- saif->master_id = pdata->master_id;
- if (saif->master_id < 0 ||
- saif->master_id >= ARRAY_SIZE(mxs_saif) ||
- saif->master_id == saif->id) {
- dev_err(&pdev->dev, "get wrong master id\n");
- return -EINVAL;
+ if (np) {
+ struct device_node *master;
+ saif->id = of_alias_get_id(np, "saif");
+ if (saif->id < 0)
+ return saif->id;
+ /*
+ * If there is no "fsl,saif-master" phandle, it's a saif
+ * master. Otherwise, it's a slave and its phandle points
+ * to the master.
+ */
+ master = of_parse_phandle(np, "fsl,saif-master", 0);
+ if (!master) {
+ saif->master_id = saif->id;
+ } else {
+ saif->master_id = of_alias_get_id(master, "saif");
+ if (saif->master_id < 0)
+ return saif->master_id;
}
} else {
- saif->master_id = saif->id;
+ saif->id = pdev->id;
+ pdata = pdev->dev.platform_data;
+ if (pdata && !pdata->master_mode)
+ saif->master_id = pdata->master_id;
+ else
+ saif->master_id = saif->id;
+ }
+
+ if (saif->master_id < 0 || saif->master_id >= ARRAY_SIZE(mxs_saif)) {
+ dev_err(&pdev->dev, "get wrong master id\n");
+ return -EINVAL;
+ }
+
+ mxs_saif[saif->id] = saif;
+
+ pinctrl = devm_pinctrl_get_select_default(&pdev->dev);
+ if (IS_ERR(pinctrl)) {
+ ret = PTR_ERR(pinctrl);
+ return ret;
}
saif->clk = clk_get(&pdev->dev, NULL);
@@ -669,12 +699,19 @@ static int mxs_saif_probe(struct platform_device *pdev)
dmares = platform_get_resource(pdev, IORESOURCE_DMA, 0);
if (!dmares) {
- ret = -ENODEV;
- dev_err(&pdev->dev, "failed to get dma resource: %d\n",
- ret);
- goto failed_get_resource;
+ /*
+ * TODO: This is a temporary solution and should be changed
+ * to use generic DMA binding later when the helplers get in.
+ */
+ ret = of_property_read_u32(np, "fsl,saif-dma-channel",
+ &saif->dma_param.chan_num);
+ if (ret) {
+ dev_err(&pdev->dev, "failed to get dma channel\n");
+ goto failed_get_resource;
+ }
+ } else {
+ saif->dma_param.chan_num = dmares->start;
}
- saif->dma_param.chan_num = dmares->start;
saif->irq = platform_get_irq(pdev, 0);
if (saif->irq < 0) {
@@ -708,24 +745,14 @@ static int mxs_saif_probe(struct platform_device *pdev)
goto failed_get_resource;
}
- saif->soc_platform_pdev = platform_device_alloc(
- "mxs-pcm-audio", pdev->id);
- if (!saif->soc_platform_pdev) {
- ret = -ENOMEM;
- goto failed_pdev_alloc;
- }
-
- platform_set_drvdata(saif->soc_platform_pdev, saif);
- ret = platform_device_add(saif->soc_platform_pdev);
+ ret = mxs_pcm_platform_register(&pdev->dev);
if (ret) {
- dev_err(&pdev->dev, "failed to add soc platform device\n");
- goto failed_pdev_add;
+ dev_err(&pdev->dev, "register PCM failed: %d\n", ret);
+ goto failed_pdev_alloc;
}
return 0;
-failed_pdev_add:
- platform_device_put(saif->soc_platform_pdev);
failed_pdev_alloc:
snd_soc_unregister_dai(&pdev->dev);
failed_get_resource:
@@ -738,13 +765,19 @@ static int __devexit mxs_saif_remove(struct platform_device *pdev)
{
struct mxs_saif *saif = platform_get_drvdata(pdev);
- platform_device_unregister(saif->soc_platform_pdev);
+ mxs_pcm_platform_unregister(&pdev->dev);
snd_soc_unregister_dai(&pdev->dev);
clk_put(saif->clk);
return 0;
}
+static const struct of_device_id mxs_saif_dt_ids[] = {
+ { .compatible = "fsl,imx28-saif", },
+ { /* sentinel */ }
+};
+MODULE_DEVICE_TABLE(of, mxs_saif_dt_ids);
+
static struct platform_driver mxs_saif_driver = {
.probe = mxs_saif_probe,
.remove = __devexit_p(mxs_saif_remove),
@@ -752,6 +785,7 @@ static struct platform_driver mxs_saif_driver = {
.driver = {
.name = "mxs-saif",
.owner = THIS_MODULE,
+ .of_match_table = mxs_saif_dt_ids,
},
};
diff --git a/sound/soc/mxs/mxs-saif.h b/sound/soc/mxs/mxs-saif.h
index 12c91e4eb94..3cb342e5bc9 100644
--- a/sound/soc/mxs/mxs-saif.h
+++ b/sound/soc/mxs/mxs-saif.h
@@ -123,7 +123,6 @@ struct mxs_saif {
unsigned int cur_rate;
unsigned int ongoing;
- struct platform_device *soc_platform_pdev;
u32 fifo_underrun;
u32 fifo_overrun;
};
diff --git a/sound/soc/mxs/mxs-sgtl5000.c b/sound/soc/mxs/mxs-sgtl5000.c
index 60f052b7cf2..3e6e8764b2e 100644
--- a/sound/soc/mxs/mxs-sgtl5000.c
+++ b/sound/soc/mxs/mxs-sgtl5000.c
@@ -18,6 +18,8 @@
#include <linux/module.h>
#include <linux/device.h>
+#include <linux/of.h>
+#include <linux/of_device.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/soc.h>
@@ -90,7 +92,7 @@ static struct snd_soc_dai_link mxs_sgtl5000_dai[] = {
.codec_dai_name = "sgtl5000",
.codec_name = "sgtl5000.0-000a",
.cpu_dai_name = "mxs-saif.0",
- .platform_name = "mxs-pcm-audio.0",
+ .platform_name = "mxs-saif.0",
.ops = &mxs_sgtl5000_hifi_ops,
}, {
.name = "HiFi Rx",
@@ -98,7 +100,7 @@ static struct snd_soc_dai_link mxs_sgtl5000_dai[] = {
.codec_dai_name = "sgtl5000",
.codec_name = "sgtl5000.0-000a",
.cpu_dai_name = "mxs-saif.1",
- .platform_name = "mxs-pcm-audio.1",
+ .platform_name = "mxs-saif.1",
.ops = &mxs_sgtl5000_hifi_ops,
},
};
@@ -110,11 +112,48 @@ static struct snd_soc_card mxs_sgtl5000 = {
.num_links = ARRAY_SIZE(mxs_sgtl5000_dai),
};
+static int __devinit mxs_sgtl5000_probe_dt(struct platform_device *pdev)
+{
+ struct device_node *np = pdev->dev.of_node;
+ struct device_node *saif_np[2], *codec_np;
+ int i, ret = 0;
+
+ if (!np)
+ return 1; /* no device tree */
+
+ saif_np[0] = of_parse_phandle(np, "saif-controllers", 0);
+ saif_np[1] = of_parse_phandle(np, "saif-controllers", 1);
+ codec_np = of_parse_phandle(np, "audio-codec", 0);
+ if (!saif_np[0] || !saif_np[1] || !codec_np) {
+ dev_err(&pdev->dev, "phandle missing or invalid\n");
+ return -EINVAL;
+ }
+
+ for (i = 0; i < 2; i++) {
+ 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].platform_name = NULL;
+ mxs_sgtl5000_dai[i].platform_of_node = saif_np[i];
+ }
+
+ of_node_put(codec_np);
+ of_node_put(saif_np[0]);
+ of_node_put(saif_np[1]);
+
+ return ret;
+}
+
static int __devinit mxs_sgtl5000_probe(struct platform_device *pdev)
{
struct snd_soc_card *card = &mxs_sgtl5000;
int ret;
+ ret = mxs_sgtl5000_probe_dt(pdev);
+ if (ret < 0)
+ return ret;
+
/*
* Set an init clock(11.28Mhz) for sgtl5000 initialization(i2c r/w).
* The Sgtl5000 sysclk is derived from saif0 mclk and it's range
@@ -148,10 +187,17 @@ static int __devexit mxs_sgtl5000_remove(struct platform_device *pdev)
return 0;
}
+static const struct of_device_id mxs_sgtl5000_dt_ids[] = {
+ { .compatible = "fsl,mxs-audio-sgtl5000", },
+ { /* sentinel */ }
+};
+MODULE_DEVICE_TABLE(of, mxs_sgtl5000_dt_ids);
+
static struct platform_driver mxs_sgtl5000_audio_driver = {
.driver = {
.name = "mxs-sgtl5000",
.owner = THIS_MODULE,
+ .of_match_table = mxs_sgtl5000_dt_ids,
},
.probe = mxs_sgtl5000_probe,
.remove = __devexit_p(mxs_sgtl5000_remove),
diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig
index deafbfaacdb..57a2fa75108 100644
--- a/sound/soc/omap/Kconfig
+++ b/sound/soc/omap/Kconfig
@@ -109,10 +109,12 @@ config SND_OMAP_SOC_OMAP_ABE_TWL6040
- PandaBoard (4430)
- PandaBoardES (4460)
-config SND_OMAP_SOC_OMAP4_HDMI
- tristate "SoC Audio support for Texas Instruments OMAP4 HDMI"
- depends on SND_OMAP_SOC && OMAP4_DSS_HDMI && OMAP2_DSS && ARCH_OMAP4
+config SND_OMAP_SOC_OMAP_HDMI
+ tristate "SoC Audio support for Texas Instruments OMAP HDMI"
+ depends on SND_OMAP_SOC && OMAP4_DSS_HDMI && OMAP2_DSS
select SND_OMAP_SOC_HDMI
+ select SND_SOC_OMAP_HDMI_CODEC
+ select OMAP4_DSS_HDMI_AUDIO
help
Say Y if you want to add support for SoC HDMI audio on Texas Instruments
OMAP4 chips
diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile
index 1d656bce01d..0e14dd32256 100644
--- a/sound/soc/omap/Makefile
+++ b/sound/soc/omap/Makefile
@@ -25,7 +25,7 @@ snd-soc-omap3pandora-objs := omap3pandora.o
snd-soc-omap3beagle-objs := omap3beagle.o
snd-soc-zoom2-objs := zoom2.o
snd-soc-igep0020-objs := igep0020.o
-snd-soc-omap4-hdmi-objs := omap4-hdmi-card.o
+snd-soc-omap-hdmi-card-objs := omap-hdmi-card.o
obj-$(CONFIG_SND_OMAP_SOC_N810) += snd-soc-n810.o
obj-$(CONFIG_SND_OMAP_SOC_RX51) += snd-soc-rx51.o
@@ -41,4 +41,4 @@ obj-$(CONFIG_SND_OMAP_SOC_OMAP3_PANDORA) += snd-soc-omap3pandora.o
obj-$(CONFIG_SND_OMAP_SOC_OMAP3_BEAGLE) += snd-soc-omap3beagle.o
obj-$(CONFIG_SND_OMAP_SOC_ZOOM2) += snd-soc-zoom2.o
obj-$(CONFIG_SND_OMAP_SOC_IGEP0020) += snd-soc-igep0020.o
-obj-$(CONFIG_SND_OMAP_SOC_OMAP4_HDMI) += snd-soc-omap4-hdmi.o
+obj-$(CONFIG_SND_OMAP_SOC_OMAP_HDMI) += snd-soc-omap-hdmi-card.o
diff --git a/sound/soc/omap/mcbsp.c b/sound/soc/omap/mcbsp.c
index e5f44440d1b..34835e8a916 100644
--- a/sound/soc/omap/mcbsp.c
+++ b/sound/soc/omap/mcbsp.c
@@ -109,6 +109,47 @@ static void omap_mcbsp_dump_reg(struct omap_mcbsp *mcbsp)
dev_dbg(mcbsp->dev, "***********************\n");
}
+static irqreturn_t omap_mcbsp_irq_handler(int irq, void *dev_id)
+{
+ struct omap_mcbsp *mcbsp = dev_id;
+ u16 irqst;
+
+ irqst = MCBSP_READ(mcbsp, IRQST);
+ dev_dbg(mcbsp->dev, "IRQ callback : 0x%x\n", irqst);
+
+ if (irqst & RSYNCERREN)
+ dev_err(mcbsp->dev, "RX Frame Sync Error!\n");
+ if (irqst & RFSREN)
+ dev_dbg(mcbsp->dev, "RX Frame Sync\n");
+ if (irqst & REOFEN)
+ dev_dbg(mcbsp->dev, "RX End Of Frame\n");
+ if (irqst & RRDYEN)
+ dev_dbg(mcbsp->dev, "RX Buffer Threshold Reached\n");
+ if (irqst & RUNDFLEN)
+ dev_err(mcbsp->dev, "RX Buffer Underflow!\n");
+ if (irqst & ROVFLEN)
+ dev_err(mcbsp->dev, "RX Buffer Overflow!\n");
+
+ if (irqst & XSYNCERREN)
+ dev_err(mcbsp->dev, "TX Frame Sync Error!\n");
+ if (irqst & XFSXEN)
+ dev_dbg(mcbsp->dev, "TX Frame Sync\n");
+ if (irqst & XEOFEN)
+ dev_dbg(mcbsp->dev, "TX End Of Frame\n");
+ if (irqst & XRDYEN)
+ dev_dbg(mcbsp->dev, "TX Buffer threshold Reached\n");
+ if (irqst & XUNDFLEN)
+ dev_err(mcbsp->dev, "TX Buffer Underflow!\n");
+ if (irqst & XOVFLEN)
+ dev_err(mcbsp->dev, "TX Buffer Overflow!\n");
+ if (irqst & XEMPTYEOFEN)
+ dev_dbg(mcbsp->dev, "TX Buffer empty at end of frame\n");
+
+ MCBSP_WRITE(mcbsp, IRQST, irqst);
+
+ return IRQ_HANDLED;
+}
+
static irqreturn_t omap_mcbsp_tx_irq_handler(int irq, void *dev_id)
{
struct omap_mcbsp *mcbsp_tx = dev_id;
@@ -176,6 +217,10 @@ void omap_mcbsp_config(struct omap_mcbsp *mcbsp,
/* Enable wakeup behavior */
if (mcbsp->pdata->has_wakeup)
MCBSP_WRITE(mcbsp, WAKEUPEN, XRDYEN | RRDYEN);
+
+ /* Enable TX/RX sync error interrupts by default */
+ if (mcbsp->irq)
+ MCBSP_WRITE(mcbsp, IRQEN, RSYNCERREN | XSYNCERREN);
}
/**
@@ -489,23 +534,25 @@ int omap_mcbsp_request(struct omap_mcbsp *mcbsp)
MCBSP_WRITE(mcbsp, SPCR1, 0);
MCBSP_WRITE(mcbsp, SPCR2, 0);
- err = request_irq(mcbsp->tx_irq, omap_mcbsp_tx_irq_handler,
- 0, "McBSP", (void *)mcbsp);
- if (err != 0) {
- dev_err(mcbsp->dev, "Unable to request TX IRQ %d "
- "for McBSP%d\n", mcbsp->tx_irq,
- mcbsp->id);
- goto err_clk_disable;
- }
+ if (mcbsp->irq) {
+ err = request_irq(mcbsp->irq, omap_mcbsp_irq_handler, 0,
+ "McBSP", (void *)mcbsp);
+ if (err != 0) {
+ dev_err(mcbsp->dev, "Unable to request IRQ\n");
+ goto err_clk_disable;
+ }
+ } else {
+ err = request_irq(mcbsp->tx_irq, omap_mcbsp_tx_irq_handler, 0,
+ "McBSP TX", (void *)mcbsp);
+ if (err != 0) {
+ dev_err(mcbsp->dev, "Unable to request TX IRQ\n");
+ goto err_clk_disable;
+ }
- if (mcbsp->rx_irq) {
- err = request_irq(mcbsp->rx_irq,
- omap_mcbsp_rx_irq_handler,
- 0, "McBSP", (void *)mcbsp);
+ err = request_irq(mcbsp->rx_irq, omap_mcbsp_rx_irq_handler, 0,
+ "McBSP RX", (void *)mcbsp);
if (err != 0) {
- dev_err(mcbsp->dev, "Unable to request RX IRQ %d "
- "for McBSP%d\n", mcbsp->rx_irq,
- mcbsp->id);
+ dev_err(mcbsp->dev, "Unable to request RX IRQ\n");
goto err_free_irq;
}
}
@@ -542,9 +589,16 @@ void omap_mcbsp_free(struct omap_mcbsp *mcbsp)
if (mcbsp->pdata->has_wakeup)
MCBSP_WRITE(mcbsp, WAKEUPEN, 0);
- if (mcbsp->rx_irq)
+ /* Disable interrupt requests */
+ if (mcbsp->irq)
+ MCBSP_WRITE(mcbsp, IRQEN, 0);
+
+ if (mcbsp->irq) {
+ free_irq(mcbsp->irq, (void *)mcbsp);
+ } else {
free_irq(mcbsp->rx_irq, (void *)mcbsp);
- free_irq(mcbsp->tx_irq, (void *)mcbsp);
+ free_irq(mcbsp->tx_irq, (void *)mcbsp);
+ }
reg_cache = mcbsp->reg_cache;
@@ -754,7 +808,7 @@ THRESHOLD_PROP_BUILDER(max_tx_thres);
THRESHOLD_PROP_BUILDER(max_rx_thres);
static const char *dma_op_modes[] = {
- "element", "threshold", "frame",
+ "element", "threshold",
};
static ssize_t dma_op_mode_show(struct device *dev,
@@ -949,13 +1003,24 @@ int __devinit omap_mcbsp_init(struct platform_device *pdev)
else
mcbsp->phys_dma_base = res->start;
- mcbsp->tx_irq = platform_get_irq_byname(pdev, "tx");
- mcbsp->rx_irq = platform_get_irq_byname(pdev, "rx");
-
- /* From OMAP4 there will be a single irq line */
- if (mcbsp->tx_irq == -ENXIO) {
- mcbsp->tx_irq = platform_get_irq(pdev, 0);
- mcbsp->rx_irq = 0;
+ /*
+ * OMAP1, 2 uses two interrupt lines: TX, RX
+ * OMAP2430, OMAP3 SoC have combined IRQ line as well.
+ * OMAP4 and newer SoC only have the combined IRQ line.
+ * Use the combined IRQ if available since it gives better debugging
+ * possibilities.
+ */
+ mcbsp->irq = platform_get_irq_byname(pdev, "common");
+ if (mcbsp->irq == -ENXIO) {
+ mcbsp->tx_irq = platform_get_irq_byname(pdev, "tx");
+
+ if (mcbsp->tx_irq == -ENXIO) {
+ mcbsp->irq = platform_get_irq(pdev, 0);
+ mcbsp->tx_irq = 0;
+ } else {
+ mcbsp->rx_irq = platform_get_irq_byname(pdev, "rx");
+ mcbsp->irq = 0;
+ }
}
res = platform_get_resource_byname(pdev, IORESOURCE_DMA, "rx");
diff --git a/sound/soc/omap/mcbsp.h b/sound/soc/omap/mcbsp.h
index a944fcc9073..262a6152111 100644
--- a/sound/soc/omap/mcbsp.h
+++ b/sound/soc/omap/mcbsp.h
@@ -217,17 +217,20 @@ enum {
/********************** McBSP DMA operating modes **************************/
#define MCBSP_DMA_MODE_ELEMENT 0
#define MCBSP_DMA_MODE_THRESHOLD 1
-#define MCBSP_DMA_MODE_FRAME 2
-/********************** McBSP WAKEUPEN bit definitions *********************/
+/********************** McBSP WAKEUPEN/IRQST/IRQEN bit definitions *********/
#define RSYNCERREN BIT(0)
#define RFSREN BIT(1)
#define REOFEN BIT(2)
#define RRDYEN BIT(3)
+#define RUNDFLEN BIT(4)
+#define ROVFLEN BIT(5)
#define XSYNCERREN BIT(7)
#define XFSXEN BIT(8)
#define XEOFEN BIT(9)
#define XRDYEN BIT(10)
+#define XUNDFLEN BIT(11)
+#define XOVFLEN BIT(12)
#define XEMPTYEOFEN BIT(14)
/* Clock signal muxing options */
@@ -295,6 +298,7 @@ struct omap_mcbsp {
int configured;
u8 free;
+ int irq;
int rx_irq;
int tx_irq;
diff --git a/sound/soc/omap/omap-abe-twl6040.c b/sound/soc/omap/omap-abe-twl6040.c
index 93bb8eee22b..9d93793d307 100644
--- a/sound/soc/omap/omap-abe-twl6040.c
+++ b/sound/soc/omap/omap-abe-twl6040.c
@@ -40,6 +40,11 @@
#include "omap-pcm.h"
#include "../codecs/twl6040.h"
+struct abe_twl6040 {
+ int jack_detection; /* board can detect jack events */
+ int mclk_freq; /* MCLK frequency speed for twl6040 */
+};
+
static int omap_abe_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
@@ -47,13 +52,13 @@ static int omap_abe_hw_params(struct snd_pcm_substream *substream,
struct snd_soc_dai *codec_dai = rtd->codec_dai;
struct snd_soc_codec *codec = rtd->codec;
struct snd_soc_card *card = codec->card;
- struct omap_abe_twl6040_data *pdata = dev_get_platdata(card->dev);
+ struct abe_twl6040 *priv = snd_soc_card_get_drvdata(card);
int clk_id, freq;
int ret;
clk_id = twl6040_get_clk_id(rtd->codec);
if (clk_id == TWL6040_SYSCLK_SEL_HPPLL)
- freq = pdata->mclk_freq;
+ freq = priv->mclk_freq;
else if (clk_id == TWL6040_SYSCLK_SEL_LPPLL)
freq = 32768;
else
@@ -128,6 +133,9 @@ static const struct snd_soc_dapm_widget twl6040_dapm_widgets[] = {
SND_SOC_DAPM_MIC("Main Handset Mic", NULL),
SND_SOC_DAPM_MIC("Sub Handset Mic", NULL),
SND_SOC_DAPM_LINE("Line In", NULL),
+
+ /* Digital microphones */
+ SND_SOC_DAPM_MIC("Digital Mic", NULL),
};
static const struct snd_soc_dapm_route audio_map[] = {
@@ -173,6 +181,7 @@ static int omap_abe_twl6040_init(struct snd_soc_pcm_runtime *rtd)
struct snd_soc_card *card = codec->card;
struct snd_soc_dapm_context *dapm = &codec->dapm;
struct omap_abe_twl6040_data *pdata = dev_get_platdata(card->dev);
+ struct abe_twl6040 *priv = snd_soc_card_get_drvdata(card);
int hs_trim;
int ret = 0;
@@ -196,7 +205,7 @@ static int omap_abe_twl6040_init(struct snd_soc_pcm_runtime *rtd)
TWL6040_HSF_TRIM_RIGHT(hs_trim));
/* Headset jack detection only if it is supported */
- if (pdata->jack_detection) {
+ if (priv->jack_detection) {
ret = snd_soc_jack_new(codec, "Headset Jack",
SND_JACK_HEADSET, &hs_jack);
if (ret)
@@ -210,10 +219,6 @@ static int omap_abe_twl6040_init(struct snd_soc_pcm_runtime *rtd)
return ret;
}
-static const struct snd_soc_dapm_widget dmic_dapm_widgets[] = {
- SND_SOC_DAPM_MIC("Digital Mic", NULL),
-};
-
static const struct snd_soc_dapm_route dmic_audio_map[] = {
{"DMic", NULL, "Digital Mic"},
{"Digital Mic", NULL, "Digital Mic1 Bias"},
@@ -223,19 +228,13 @@ static int omap_abe_dmic_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_codec *codec = rtd->codec;
struct snd_soc_dapm_context *dapm = &codec->dapm;
- int ret;
-
- ret = snd_soc_dapm_new_controls(dapm, dmic_dapm_widgets,
- ARRAY_SIZE(dmic_dapm_widgets));
- if (ret)
- return ret;
return snd_soc_dapm_add_routes(dapm, dmic_audio_map,
ARRAY_SIZE(dmic_audio_map));
}
/* Digital audio interface glue - connects codec <--> CPU */
-static struct snd_soc_dai_link twl6040_dmic_dai[] = {
+static struct snd_soc_dai_link abe_twl6040_dai_links[] = {
{
.name = "TWL6040",
.stream_name = "TWL6040",
@@ -258,19 +257,6 @@ static struct snd_soc_dai_link twl6040_dmic_dai[] = {
},
};
-static struct snd_soc_dai_link twl6040_only_dai[] = {
- {
- .name = "TWL6040",
- .stream_name = "TWL6040",
- .cpu_dai_name = "omap-mcpdm",
- .codec_dai_name = "twl6040-legacy",
- .platform_name = "omap-pcm-audio",
- .codec_name = "twl6040-codec",
- .init = omap_abe_twl6040_init,
- .ops = &omap_abe_ops,
- },
-};
-
/* Audio machine driver */
static struct snd_soc_card omap_abe_card = {
.owner = THIS_MODULE,
@@ -285,6 +271,8 @@ static __devinit int omap_abe_probe(struct platform_device *pdev)
{
struct omap_abe_twl6040_data *pdata = dev_get_platdata(&pdev->dev);
struct snd_soc_card *card = &omap_abe_card;
+ struct abe_twl6040 *priv;
+ int num_links = 0;
int ret;
card->dev = &pdev->dev;
@@ -294,6 +282,10 @@ static __devinit int omap_abe_probe(struct platform_device *pdev)
return -ENODEV;
}
+ priv = devm_kzalloc(&pdev->dev, sizeof(struct abe_twl6040), GFP_KERNEL);
+ if (priv == NULL)
+ return -ENOMEM;
+
if (pdata->card_name) {
card->name = pdata->card_name;
} else {
@@ -301,18 +293,24 @@ static __devinit int omap_abe_probe(struct platform_device *pdev)
return -ENODEV;
}
- if (!pdata->mclk_freq) {
+ priv->jack_detection = pdata->jack_detection;
+ priv->mclk_freq = pdata->mclk_freq;
+
+
+ if (!priv->mclk_freq) {
dev_err(&pdev->dev, "MCLK frequency missing\n");
return -ENODEV;
}
- if (pdata->has_dmic) {
- card->dai_link = twl6040_dmic_dai;
- card->num_links = ARRAY_SIZE(twl6040_dmic_dai);
- } else {
- card->dai_link = twl6040_only_dai;
- card->num_links = ARRAY_SIZE(twl6040_only_dai);
- }
+ if (pdata->has_dmic)
+ num_links = 2;
+ else
+ num_links = 1;
+
+ card->dai_link = abe_twl6040_dai_links;
+ card->num_links = num_links;
+
+ snd_soc_card_set_drvdata(card, priv);
ret = snd_soc_register_card(card);
if (ret)
diff --git a/sound/soc/omap/omap-dmic.c b/sound/soc/omap/omap-dmic.c
index 4dcb5a7e40e..75f5dca0e8d 100644
--- a/sound/soc/omap/omap-dmic.c
+++ b/sound/soc/omap/omap-dmic.c
@@ -32,6 +32,7 @@
#include <linux/io.h>
#include <linux/slab.h>
#include <linux/pm_runtime.h>
+#include <linux/of_device.h>
#include <plat/dma.h>
#include <sound/core.h>
@@ -528,10 +529,17 @@ static int __devexit asoc_dmic_remove(struct platform_device *pdev)
return 0;
}
+static const struct of_device_id omap_dmic_of_match[] = {
+ { .compatible = "ti,omap4-dmic", },
+ { }
+};
+MODULE_DEVICE_TABLE(of, omap_dmic_of_match);
+
static struct platform_driver asoc_dmic_driver = {
.driver = {
.name = "omap-dmic",
.owner = THIS_MODULE,
+ .of_match_table = omap_dmic_of_match,
},
.probe = asoc_dmic_probe,
.remove = __devexit_p(asoc_dmic_remove),
diff --git a/sound/soc/omap/omap-hdmi-card.c b/sound/soc/omap/omap-hdmi-card.c
new file mode 100644
index 00000000000..eaa2ea0e3f8
--- /dev/null
+++ b/sound/soc/omap/omap-hdmi-card.c
@@ -0,0 +1,87 @@
+/*
+ * omap-hdmi-card.c
+ *
+ * OMAP ALSA SoC machine driver for TI OMAP HDMI
+ * Copyright (C) 2011 Texas Instruments Incorporated - http://www.ti.com/
+ * Author: Ricardo Neri <ricardo.neri@ti.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.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ *
+ */
+
+#include <linux/module.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <asm/mach-types.h>
+#include <video/omapdss.h>
+
+#define DRV_NAME "omap-hdmi-audio"
+
+static struct snd_soc_dai_link omap_hdmi_dai = {
+ .name = "HDMI",
+ .stream_name = "HDMI",
+ .cpu_dai_name = "omap-hdmi-audio-dai",
+ .platform_name = "omap-pcm-audio",
+ .codec_name = "hdmi-audio-codec",
+ .codec_dai_name = "omap-hdmi-hifi",
+};
+
+static struct snd_soc_card snd_soc_omap_hdmi = {
+ .name = "OMAPHDMI",
+ .owner = THIS_MODULE,
+ .dai_link = &omap_hdmi_dai,
+ .num_links = 1,
+};
+
+static __devinit int omap_hdmi_probe(struct platform_device *pdev)
+{
+ struct snd_soc_card *card = &snd_soc_omap_hdmi;
+ 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);
+ card->dev = NULL;
+ return ret;
+ }
+ return 0;
+}
+
+static int __devexit omap_hdmi_remove(struct platform_device *pdev)
+{
+ struct snd_soc_card *card = platform_get_drvdata(pdev);
+
+ snd_soc_unregister_card(card);
+ card->dev = NULL;
+ return 0;
+}
+
+static struct platform_driver omap_hdmi_driver = {
+ .driver = {
+ .name = DRV_NAME,
+ .owner = THIS_MODULE,
+ },
+ .probe = omap_hdmi_probe,
+ .remove = __devexit_p(omap_hdmi_remove),
+};
+
+module_platform_driver(omap_hdmi_driver);
+
+MODULE_AUTHOR("Ricardo Neri <ricardo.neri@ti.com>");
+MODULE_DESCRIPTION("OMAP HDMI machine ASoC driver");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:" DRV_NAME);
diff --git a/sound/soc/omap/omap-hdmi.c b/sound/soc/omap/omap-hdmi.c
index 38e0defa707..a08245d9203 100644
--- a/sound/soc/omap/omap-hdmi.c
+++ b/sound/soc/omap/omap-hdmi.c
@@ -30,21 +30,28 @@
#include <sound/pcm_params.h>
#include <sound/initval.h>
#include <sound/soc.h>
+#include <sound/asound.h>
+#include <sound/asoundef.h>
+#include <video/omapdss.h>
#include <plat/dma.h>
#include "omap-pcm.h"
#include "omap-hdmi.h"
-#define DRV_NAME "hdmi-audio-dai"
+#define DRV_NAME "omap-hdmi-audio-dai"
-static struct omap_pcm_dma_data omap_hdmi_dai_dma_params = {
- .name = "HDMI playback",
- .sync_mode = OMAP_DMA_SYNC_PACKET,
+struct hdmi_priv {
+ struct omap_pcm_dma_data dma_params;
+ struct omap_dss_audio dss_audio;
+ struct snd_aes_iec958 iec;
+ struct snd_cea_861_aud_if cea;
+ struct omap_dss_device *dssdev;
};
static int omap_hdmi_dai_startup(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
+ struct hdmi_priv *priv = snd_soc_dai_get_drvdata(dai);
int err;
/*
* Make sure that the period bytes are multiple of the DMA packet size.
@@ -52,46 +59,201 @@ static int omap_hdmi_dai_startup(struct snd_pcm_substream *substream,
*/
err = snd_pcm_hw_constraint_step(substream->runtime, 0,
SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 128);
- if (err < 0)
+ if (err < 0) {
+ dev_err(dai->dev, "could not apply constraint\n");
return err;
+ }
+ if (!priv->dssdev->driver->audio_supported(priv->dssdev)) {
+ dev_err(dai->dev, "audio not supported\n");
+ return -ENODEV;
+ }
return 0;
}
+static int omap_hdmi_dai_prepare(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct hdmi_priv *priv = snd_soc_dai_get_drvdata(dai);
+
+ return priv->dssdev->driver->audio_enable(priv->dssdev);
+}
+
static int omap_hdmi_dai_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
{
+ struct hdmi_priv *priv = snd_soc_dai_get_drvdata(dai);
+ struct snd_aes_iec958 *iec = &priv->iec;
+ struct snd_cea_861_aud_if *cea = &priv->cea;
int err = 0;
switch (params_format(params)) {
case SNDRV_PCM_FORMAT_S16_LE:
- omap_hdmi_dai_dma_params.packet_size = 16;
+ priv->dma_params.packet_size = 16;
break;
case SNDRV_PCM_FORMAT_S24_LE:
- omap_hdmi_dai_dma_params.packet_size = 32;
+ priv->dma_params.packet_size = 32;
break;
default:
- err = -EINVAL;
+ dev_err(dai->dev, "format not supported!\n");
+ return -EINVAL;
}
- omap_hdmi_dai_dma_params.data_type = OMAP_DMA_DATA_TYPE_S32;
+ priv->dma_params.data_type = OMAP_DMA_DATA_TYPE_S32;
snd_soc_dai_set_dma_data(dai, substream,
- &omap_hdmi_dai_dma_params);
+ &priv->dma_params);
+
+ /*
+ * fill the IEC-60958 channel status word
+ */
+
+ /* specify IEC-60958-3 (commercial use) */
+ iec->status[0] &= ~IEC958_AES0_PROFESSIONAL;
+
+ /* specify that the audio is LPCM*/
+ iec->status[0] &= ~IEC958_AES0_NONAUDIO;
+
+ iec->status[0] |= IEC958_AES0_CON_NOT_COPYRIGHT;
+
+ iec->status[0] |= IEC958_AES0_CON_EMPHASIS_NONE;
+
+ iec->status[0] |= IEC958_AES1_PRO_MODE_NOTID;
+
+ iec->status[1] = IEC958_AES1_CON_GENERAL;
+
+ iec->status[2] |= IEC958_AES2_CON_SOURCE_UNSPEC;
+
+ iec->status[2] |= IEC958_AES2_CON_CHANNEL_UNSPEC;
+
+ switch (params_rate(params)) {
+ case 32000:
+ iec->status[3] |= IEC958_AES3_CON_FS_32000;
+ break;
+ case 44100:
+ iec->status[3] |= IEC958_AES3_CON_FS_44100;
+ break;
+ case 48000:
+ iec->status[3] |= IEC958_AES3_CON_FS_48000;
+ break;
+ case 88200:
+ iec->status[3] |= IEC958_AES3_CON_FS_88200;
+ break;
+ case 96000:
+ iec->status[3] |= IEC958_AES3_CON_FS_96000;
+ break;
+ case 176400:
+ iec->status[3] |= IEC958_AES3_CON_FS_176400;
+ break;
+ case 192000:
+ iec->status[3] |= IEC958_AES3_CON_FS_192000;
+ break;
+ default:
+ dev_err(dai->dev, "rate not supported!\n");
+ return -EINVAL;
+ }
+
+ /* specify the clock accuracy */
+ iec->status[3] |= IEC958_AES3_CON_CLOCK_1000PPM;
+
+ /*
+ * specify the word length. The same word length value can mean
+ * two different lengths. Hence, we need to specify the maximum
+ * word length as well.
+ */
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ iec->status[4] |= IEC958_AES4_CON_WORDLEN_20_16;
+ iec->status[4] &= ~IEC958_AES4_CON_MAX_WORDLEN_24;
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ iec->status[4] |= IEC958_AES4_CON_WORDLEN_24_20;
+ iec->status[4] |= IEC958_AES4_CON_MAX_WORDLEN_24;
+ break;
+ default:
+ dev_err(dai->dev, "format not supported!\n");
+ return -EINVAL;
+ }
+
+ /*
+ * Fill the CEA-861 audio infoframe (see spec for details)
+ */
+
+ cea->db1_ct_cc = (params_channels(params) - 1)
+ & CEA861_AUDIO_INFOFRAME_DB1CC;
+ cea->db1_ct_cc |= CEA861_AUDIO_INFOFRAME_DB1CT_FROM_STREAM;
+
+ cea->db2_sf_ss = CEA861_AUDIO_INFOFRAME_DB2SF_FROM_STREAM;
+ cea->db2_sf_ss |= CEA861_AUDIO_INFOFRAME_DB2SS_FROM_STREAM;
+
+ cea->db3 = 0; /* not used, all zeros */
+
+ /*
+ * The OMAP HDMI IP requires to use the 8-channel channel code when
+ * transmitting more than two channels.
+ */
+ if (params_channels(params) == 2)
+ cea->db4_ca = 0x0;
+ else
+ cea->db4_ca = 0x13;
+
+ cea->db5_dminh_lsv = CEA861_AUDIO_INFOFRAME_DB5_DM_INH_PROHIBITED;
+ /* the expression is trivial but makes clear what we are doing */
+ cea->db5_dminh_lsv |= (0 & CEA861_AUDIO_INFOFRAME_DB5_LSV);
+
+ priv->dss_audio.iec = iec;
+ priv->dss_audio.cea = cea;
+
+ err = priv->dssdev->driver->audio_config(priv->dssdev,
+ &priv->dss_audio);
return err;
}
+static int omap_hdmi_dai_trigger(struct snd_pcm_substream *substream, int cmd,
+ struct snd_soc_dai *dai)
+{
+ struct hdmi_priv *priv = snd_soc_dai_get_drvdata(dai);
+ int err = 0;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ err = priv->dssdev->driver->audio_start(priv->dssdev);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ priv->dssdev->driver->audio_stop(priv->dssdev);
+ break;
+ default:
+ err = -EINVAL;
+ }
+ return err;
+}
+
+static void omap_hdmi_dai_shutdown(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct hdmi_priv *priv = snd_soc_dai_get_drvdata(dai);
+
+ priv->dssdev->driver->audio_disable(priv->dssdev);
+}
+
static const struct snd_soc_dai_ops omap_hdmi_dai_ops = {
.startup = omap_hdmi_dai_startup,
.hw_params = omap_hdmi_dai_hw_params,
+ .prepare = omap_hdmi_dai_prepare,
+ .trigger = omap_hdmi_dai_trigger,
+ .shutdown = omap_hdmi_dai_shutdown,
};
static struct snd_soc_dai_driver omap_hdmi_dai = {
.playback = {
.channels_min = 2,
- .channels_max = 2,
+ .channels_max = 8,
.rates = OMAP_HDMI_RATES,
.formats = OMAP_HDMI_FORMATS,
},
@@ -102,31 +264,77 @@ static __devinit int omap_hdmi_probe(struct platform_device *pdev)
{
int ret;
struct resource *hdmi_rsrc;
+ struct hdmi_priv *hdmi_data;
+ bool hdmi_dev_found = false;
+
+ hdmi_data = devm_kzalloc(&pdev->dev, sizeof(*hdmi_data), GFP_KERNEL);
+ if (hdmi_data == NULL) {
+ dev_err(&pdev->dev, "Cannot allocate memory for HDMI data\n");
+ return -ENOMEM;
+ }
hdmi_rsrc = platform_get_resource(pdev, IORESOURCE_MEM, 0);
if (!hdmi_rsrc) {
dev_err(&pdev->dev, "Cannot obtain IORESOURCE_MEM HDMI\n");
- return -EINVAL;
+ return -ENODEV;
}
- omap_hdmi_dai_dma_params.port_addr = hdmi_rsrc->start
+ hdmi_data->dma_params.port_addr = hdmi_rsrc->start
+ OMAP_HDMI_AUDIO_DMA_PORT;
hdmi_rsrc = platform_get_resource(pdev, IORESOURCE_DMA, 0);
if (!hdmi_rsrc) {
dev_err(&pdev->dev, "Cannot obtain IORESOURCE_DMA HDMI\n");
- return -EINVAL;
+ return -ENODEV;
}
- omap_hdmi_dai_dma_params.dma_req = hdmi_rsrc->start;
+ hdmi_data->dma_params.dma_req = hdmi_rsrc->start;
+ hdmi_data->dma_params.name = "HDMI playback";
+ hdmi_data->dma_params.sync_mode = OMAP_DMA_SYNC_PACKET;
+
+ /*
+ * TODO: We assume that there is only one DSS HDMI device. Future
+ * OMAP implementations may support more than one HDMI devices and
+ * we should provided separate audio support for all of them.
+ */
+ /* Find an HDMI device. */
+ for_each_dss_dev(hdmi_data->dssdev) {
+ omap_dss_get_device(hdmi_data->dssdev);
+ if (!hdmi_data->dssdev->driver) {
+ omap_dss_put_device(hdmi_data->dssdev);
+ continue;
+ }
+
+ if (hdmi_data->dssdev->type == OMAP_DISPLAY_TYPE_HDMI) {
+ hdmi_dev_found = true;
+ break;
+ }
+ }
+
+ if (!hdmi_dev_found) {
+ dev_err(&pdev->dev, "no driver for HDMI display found\n");
+ return -ENODEV;
+ }
+
+ dev_set_drvdata(&pdev->dev, hdmi_data);
ret = snd_soc_register_dai(&pdev->dev, &omap_hdmi_dai);
+
return ret;
}
static int __devexit omap_hdmi_remove(struct platform_device *pdev)
{
+ struct hdmi_priv *hdmi_data = dev_get_drvdata(&pdev->dev);
+
snd_soc_unregister_dai(&pdev->dev);
+
+ if (hdmi_data == NULL) {
+ dev_err(&pdev->dev, "cannot obtain HDMi data\n");
+ return -ENODEV;
+ }
+
+ omap_dss_put_device(hdmi_data->dssdev);
return 0;
}
diff --git a/sound/soc/omap/omap-hdmi.h b/sound/soc/omap/omap-hdmi.h
index 34c298d5057..6ad2bf4f269 100644
--- a/sound/soc/omap/omap-hdmi.h
+++ b/sound/soc/omap/omap-hdmi.h
@@ -28,7 +28,9 @@
#define OMAP_HDMI_AUDIO_DMA_PORT 0x8c
#define OMAP_HDMI_RATES (SNDRV_PCM_RATE_32000 | \
- SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000)
+ SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | \
+ SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 | \
+ SNDRV_PCM_RATE_176400 | SNDRV_PCM_RATE_192000)
#define OMAP_HDMI_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \
SNDRV_PCM_FMTBIT_S24_LE)
diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c
index 6912ac7cb62..1046083e90a 100644
--- a/sound/soc/omap/omap-mcbsp.c
+++ b/sound/soc/omap/omap-mcbsp.c
@@ -71,18 +71,17 @@ static void omap_mcbsp_set_threshold(struct snd_pcm_substream *substream)
dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
- /* TODO: Currently, MODE_ELEMENT == MODE_FRAME */
- if (mcbsp->dma_op_mode == MCBSP_DMA_MODE_THRESHOLD)
- /*
- * Configure McBSP threshold based on either:
- * packet_size, when the sDMA is in packet mode, or
- * based on the period size.
- */
- if (dma_data->packet_size)
- words = dma_data->packet_size;
- else
- words = snd_pcm_lib_period_bytes(substream) /
- (mcbsp->wlen / 8);
+ /*
+ * Configure McBSP threshold based on either:
+ * packet_size, when the sDMA is in packet mode, or based on the
+ * period size in THRESHOLD mode, otherwise use McBSP threshold = 1
+ * for mono streams.
+ */
+ if (dma_data->packet_size)
+ words = dma_data->packet_size;
+ else if (mcbsp->dma_op_mode == MCBSP_DMA_MODE_THRESHOLD)
+ words = snd_pcm_lib_period_bytes(substream) /
+ (mcbsp->wlen / 8);
else
words = 1;
@@ -139,13 +138,15 @@ static int omap_mcbsp_dai_startup(struct snd_pcm_substream *substream,
if (mcbsp->pdata->buffer_size) {
/*
* Rule for the buffer size. We should not allow
- * smaller buffer than the FIFO size to avoid underruns
+ * smaller buffer than the FIFO size to avoid underruns.
+ * This applies only for the playback stream.
*/
- snd_pcm_hw_rule_add(substream->runtime, 0,
- SNDRV_PCM_HW_PARAM_BUFFER_SIZE,
- omap_mcbsp_hwrule_min_buffersize,
- mcbsp,
- SNDRV_PCM_HW_PARAM_CHANNELS, -1);
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ snd_pcm_hw_rule_add(substream->runtime, 0,
+ SNDRV_PCM_HW_PARAM_BUFFER_SIZE,
+ omap_mcbsp_hwrule_min_buffersize,
+ mcbsp,
+ SNDRV_PCM_HW_PARAM_CHANNELS, -1);
/* Make sure, that the period size is always even */
snd_pcm_hw_constraint_step(substream->runtime, 0,
@@ -230,6 +231,7 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream,
unsigned int format, div, framesize, master;
dma_data = &mcbsp->dma_data[substream->stream];
+ channels = params_channels(params);
switch (params_format(params)) {
case SNDRV_PCM_FORMAT_S16_LE:
@@ -245,7 +247,6 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream,
}
if (mcbsp->pdata->buffer_size) {
dma_data->set_threshold = omap_mcbsp_set_threshold;
- /* TODO: Currently, MODE_ELEMENT == MODE_FRAME */
if (mcbsp->dma_op_mode == MCBSP_DMA_MODE_THRESHOLD) {
int period_words, max_thrsh;
@@ -283,6 +284,10 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream,
} else {
sync_mode = OMAP_DMA_SYNC_FRAME;
}
+ } else if (channels > 1) {
+ /* Use packet mode for non mono streams */
+ pkt_size = channels;
+ sync_mode = OMAP_DMA_SYNC_PACKET;
}
}
@@ -301,7 +306,7 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream,
regs->rcr1 &= ~(RFRLEN1(0x7f) | RWDLEN1(7));
regs->xcr1 &= ~(XFRLEN1(0x7f) | XWDLEN1(7));
format = mcbsp->fmt & SND_SOC_DAIFMT_FORMAT_MASK;
- wpf = channels = params_channels(params);
+ wpf = channels;
if (channels == 2 && (format == SND_SOC_DAIFMT_I2S ||
format == SND_SOC_DAIFMT_LEFT_J)) {
/* Use dual-phase frames */
diff --git a/sound/soc/omap/omap-mcpdm.c b/sound/soc/omap/omap-mcpdm.c
index 39705561131..59d47ab5b15 100644
--- a/sound/soc/omap/omap-mcpdm.c
+++ b/sound/soc/omap/omap-mcpdm.c
@@ -33,6 +33,7 @@
#include <linux/irq.h>
#include <linux/slab.h>
#include <linux/pm_runtime.h>
+#include <linux/of_device.h>
#include <sound/core.h>
#include <sound/pcm.h>
@@ -507,10 +508,17 @@ static int __devexit asoc_mcpdm_remove(struct platform_device *pdev)
return 0;
}
+static const struct of_device_id omap_mcpdm_of_match[] = {
+ { .compatible = "ti,omap4-mcpdm", },
+ { }
+};
+MODULE_DEVICE_TABLE(of, omap_mcpdm_of_match);
+
static struct platform_driver asoc_mcpdm_driver = {
.driver = {
.name = "omap-mcpdm",
.owner = THIS_MODULE,
+ .of_match_table = omap_mcpdm_of_match,
},
.probe = asoc_mcpdm_probe,
diff --git a/sound/soc/omap/omap4-hdmi-card.c b/sound/soc/omap/omap4-hdmi-card.c
deleted file mode 100644
index 28d689b2714..00000000000
--- a/sound/soc/omap/omap4-hdmi-card.c
+++ /dev/null
@@ -1,121 +0,0 @@
-/*
- * omap4-hdmi-card.c
- *
- * OMAP ALSA SoC machine driver for TI OMAP4 HDMI
- * Copyright (C) 2011 Texas Instruments Incorporated - http://www.ti.com/
- * Author: Ricardo Neri <ricardo.neri@ti.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.
- *
- * This program is distributed in the hope that it will be useful, but
- * WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
- * General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
- * 02110-1301 USA
- *
- */
-
-#include <linux/module.h>
-#include <sound/pcm.h>
-#include <sound/soc.h>
-#include <asm/mach-types.h>
-#include <video/omapdss.h>
-
-#define DRV_NAME "omap4-hdmi-audio"
-
-static int omap4_hdmi_dai_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
-{
- int i;
- struct omap_overlay_manager *mgr = NULL;
- struct device *dev = substream->pcm->card->dev;
-
- /* Find DSS HDMI device */
- for (i = 0; i < omap_dss_get_num_overlay_managers(); i++) {
- mgr = omap_dss_get_overlay_manager(i);
- if (mgr && mgr->device
- && mgr->device->type == OMAP_DISPLAY_TYPE_HDMI)
- break;
- }
-
- if (i == omap_dss_get_num_overlay_managers()) {
- dev_err(dev, "HDMI display device not found!\n");
- return -ENODEV;
- }
-
- /* Make sure HDMI is power-on to avoid L3 interconnect errors */
- if (mgr->device->state != OMAP_DSS_DISPLAY_ACTIVE) {
- dev_err(dev, "HDMI display is not active!\n");
- return -EIO;
- }
-
- return 0;
-}
-
-static struct snd_soc_ops omap4_hdmi_dai_ops = {
- .hw_params = omap4_hdmi_dai_hw_params,
-};
-
-static struct snd_soc_dai_link omap4_hdmi_dai = {
- .name = "HDMI",
- .stream_name = "HDMI",
- .cpu_dai_name = "hdmi-audio-dai",
- .platform_name = "omap-pcm-audio",
- .codec_name = "omapdss_hdmi",
- .codec_dai_name = "hdmi-audio-codec",
- .ops = &omap4_hdmi_dai_ops,
-};
-
-static struct snd_soc_card snd_soc_omap4_hdmi = {
- .name = "OMAP4HDMI",
- .owner = THIS_MODULE,
- .dai_link = &omap4_hdmi_dai,
- .num_links = 1,
-};
-
-static __devinit int omap4_hdmi_probe(struct platform_device *pdev)
-{
- struct snd_soc_card *card = &snd_soc_omap4_hdmi;
- 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);
- card->dev = NULL;
- return ret;
- }
- return 0;
-}
-
-static int __devexit omap4_hdmi_remove(struct platform_device *pdev)
-{
- struct snd_soc_card *card = platform_get_drvdata(pdev);
-
- snd_soc_unregister_card(card);
- card->dev = NULL;
- return 0;
-}
-
-static struct platform_driver omap4_hdmi_driver = {
- .driver = {
- .name = "omap4-hdmi-audio",
- .owner = THIS_MODULE,
- },
- .probe = omap4_hdmi_probe,
- .remove = __devexit_p(omap4_hdmi_remove),
-};
-
-module_platform_driver(omap4_hdmi_driver);
-
-MODULE_AUTHOR("Ricardo Neri <ricardo.neri@ti.com>");
-MODULE_DESCRIPTION("OMAP4 HDMI machine ASoC driver");
-MODULE_LICENSE("GPL");
-MODULE_ALIAS("platform:" DRV_NAME);
diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c
index fd04ce13903..4da5fc55c7e 100644
--- a/sound/soc/pxa/pxa-ssp.c
+++ b/sound/soc/pxa/pxa-ssp.c
@@ -33,7 +33,6 @@
#include <mach/hardware.h>
#include <mach/dma.h>
-#include <mach/audio.h>
#include "../../arm/pxa2xx-pcm.h"
#include "pxa-ssp.h"
@@ -85,14 +84,12 @@ struct pxa2xx_pcm_dma_data {
char name[20];
};
-static struct pxa2xx_pcm_dma_params *
-pxa_ssp_get_dma_params(struct ssp_device *ssp, int width4, int out)
+static void pxa_ssp_set_dma_params(struct ssp_device *ssp, int width4,
+ int out, struct pxa2xx_pcm_dma_params *dma_data)
{
struct pxa2xx_pcm_dma_data *dma;
- dma = kzalloc(sizeof(struct pxa2xx_pcm_dma_data), GFP_KERNEL);
- if (dma == NULL)
- return NULL;
+ dma = container_of(dma_data, struct pxa2xx_pcm_dma_data, params);
snprintf(dma->name, 20, "SSP%d PCM %s %s", ssp->port_id,
width4 ? "32-bit" : "16-bit", out ? "out" : "in");
@@ -103,8 +100,6 @@ pxa_ssp_get_dma_params(struct ssp_device *ssp, int width4, int out)
(DCMD_INCTRGADDR | DCMD_FLOWSRC)) |
(width4 ? DCMD_WIDTH4 : DCMD_WIDTH2) | DCMD_BURST16;
dma->params.dev_addr = ssp->phys_base + SSDR;
-
- return &dma->params;
}
static int pxa_ssp_startup(struct snd_pcm_substream *substream,
@@ -112,6 +107,7 @@ static int pxa_ssp_startup(struct snd_pcm_substream *substream,
{
struct ssp_priv *priv = snd_soc_dai_get_drvdata(cpu_dai);
struct ssp_device *ssp = priv->ssp;
+ struct pxa2xx_pcm_dma_data *dma;
int ret = 0;
if (!cpu_dai->active) {
@@ -119,8 +115,10 @@ static int pxa_ssp_startup(struct snd_pcm_substream *substream,
pxa_ssp_disable(ssp);
}
- kfree(snd_soc_dai_get_dma_data(cpu_dai, substream));
- snd_soc_dai_set_dma_data(cpu_dai, substream, NULL);
+ dma = kzalloc(sizeof(struct pxa2xx_pcm_dma_data), GFP_KERNEL);
+ if (!dma)
+ return -ENOMEM;
+ snd_soc_dai_set_dma_data(cpu_dai, substream, &dma->params);
return ret;
}
@@ -195,7 +193,7 @@ static void pxa_ssp_set_scr(struct ssp_device *ssp, u32 div)
{
u32 sscr0 = pxa_ssp_read_reg(ssp, SSCR0);
- if (cpu_is_pxa25x() && ssp->type == PXA25x_SSP) {
+ if (ssp->type == PXA25x_SSP) {
sscr0 &= ~0x0000ff00;
sscr0 |= ((div - 2)/2) << 8; /* 2..512 */
} else {
@@ -213,7 +211,7 @@ static u32 pxa_ssp_get_scr(struct ssp_device *ssp)
u32 sscr0 = pxa_ssp_read_reg(ssp, SSCR0);
u32 div;
- if (cpu_is_pxa25x() && ssp->type == PXA25x_SSP)
+ if (ssp->type == PXA25x_SSP)
div = ((sscr0 >> 8) & 0xff) * 2 + 2;
else
div = ((sscr0 >> 8) & 0xfff) + 1;
@@ -243,7 +241,7 @@ static int pxa_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
break;
case PXA_SSP_CLK_PLL:
/* Internal PLL is fixed */
- if (cpu_is_pxa25x())
+ if (ssp->type == PXA25x_SSP)
priv->sysclk = 1843200;
else
priv->sysclk = 13000000;
@@ -267,11 +265,11 @@ static int pxa_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
/* The SSP clock must be disabled when changing SSP clock mode
* on PXA2xx. On PXA3xx it must be enabled when doing so. */
- if (!cpu_is_pxa3xx())
+ if (ssp->type != PXA3xx_SSP)
clk_disable(ssp->clk);
val = pxa_ssp_read_reg(ssp, SSCR0) | sscr0;
pxa_ssp_write_reg(ssp, SSCR0, val);
- if (!cpu_is_pxa3xx())
+ if (ssp->type != PXA3xx_SSP)
clk_enable(ssp->clk);
return 0;
@@ -295,24 +293,20 @@ static int pxa_ssp_set_dai_clkdiv(struct snd_soc_dai *cpu_dai,
case PXA_SSP_AUDIO_DIV_SCDB:
val = pxa_ssp_read_reg(ssp, SSACD);
val &= ~SSACD_SCDB;
-#if defined(CONFIG_PXA3xx)
- if (cpu_is_pxa3xx())
+ if (ssp->type == PXA3xx_SSP)
val &= ~SSACD_SCDX8;
-#endif
switch (div) {
case PXA_SSP_CLK_SCDB_1:
val |= SSACD_SCDB;
break;
case PXA_SSP_CLK_SCDB_4:
break;
-#if defined(CONFIG_PXA3xx)
case PXA_SSP_CLK_SCDB_8:
- if (cpu_is_pxa3xx())
+ if (ssp->type == PXA3xx_SSP)
val |= SSACD_SCDX8;
else
return -EINVAL;
break;
-#endif
default:
return -EINVAL;
}
@@ -338,10 +332,8 @@ static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, int pll_id,
struct ssp_device *ssp = priv->ssp;
u32 ssacd = pxa_ssp_read_reg(ssp, SSACD) & ~0x70;
-#if defined(CONFIG_PXA3xx)
- if (cpu_is_pxa3xx())
+ if (ssp->type == PXA3xx_SSP)
pxa_ssp_write_reg(ssp, SSACDD, 0);
-#endif
switch (freq_out) {
case 5622000:
@@ -366,11 +358,10 @@ static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, int pll_id,
break;
default:
-#ifdef CONFIG_PXA3xx
/* PXA3xx has a clock ditherer which can be used to generate
* a wider range of frequencies - calculate a value for it.
*/
- if (cpu_is_pxa3xx()) {
+ if (ssp->type == PXA3xx_SSP) {
u32 val;
u64 tmp = 19968;
tmp *= 1000000;
@@ -387,7 +378,6 @@ static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, int pll_id,
val, freq_out);
break;
}
-#endif
return -EINVAL;
}
@@ -573,18 +563,13 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream,
dma_data = snd_soc_dai_get_dma_data(cpu_dai, substream);
- /* generate correct DMA params */
- kfree(dma_data);
-
/* Network mode with one active slot (ttsa == 1) can be used
* to force 16-bit frame width on the wire (for S16_LE), even
* with two channels. Use 16-bit DMA transfers for this case.
*/
- dma_data = pxa_ssp_get_dma_params(ssp,
- ((chn == 2) && (ttsa != 1)) || (width == 32),
- substream->stream == SNDRV_PCM_STREAM_PLAYBACK);
-
- snd_soc_dai_set_dma_data(cpu_dai, substream, dma_data);
+ pxa_ssp_set_dma_params(ssp,
+ ((chn == 2) && (ttsa != 1)) || (width == 32),
+ substream->stream == SNDRV_PCM_STREAM_PLAYBACK, dma_data);
/* we can only change the settings if the port is not in use */
if (pxa_ssp_read_reg(ssp, SSCR0) & SSCR0_SSE)
@@ -596,10 +581,8 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream,
/* bit size */
switch (params_format(params)) {
case SNDRV_PCM_FORMAT_S16_LE:
-#ifdef CONFIG_PXA3xx
- if (cpu_is_pxa3xx())
+ if (ssp->type == PXA3xx_SSP)
sscr0 |= SSCR0_FPCKE;
-#endif
sscr0 |= SSCR0_DataSize(16);
break;
case SNDRV_PCM_FORMAT_S24_LE:
@@ -624,9 +607,7 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream,
* trying and failing a lot; some of the registers
* needed for that mode are only available on PXA3xx.
*/
-
-#ifdef CONFIG_PXA3xx
- if (!cpu_is_pxa3xx())
+ if (ssp->type != PXA3xx_SSP)
return -EINVAL;
sspsp |= SSPSP_SFRMWDTH(width * 2);
@@ -634,9 +615,6 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream,
sspsp |= SSPSP_EDMYSTOP(3);
sspsp |= SSPSP_DMYSTOP(3);
sspsp |= SSPSP_DMYSTRT(1);
-#else
- return -EINVAL;
-#endif
} else {
/* The frame width is the width the LRCLK is
* asserted for; the delay is expressed in
diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c
index d08583790d2..3075a426124 100644
--- a/sound/soc/pxa/pxa2xx-i2s.c
+++ b/sound/soc/pxa/pxa2xx-i2s.c
@@ -166,7 +166,7 @@ static int pxa2xx_i2s_hw_params(struct snd_pcm_substream *substream,
struct pxa2xx_pcm_dma_params *dma_data;
BUG_ON(IS_ERR(clk_i2s));
- clk_enable(clk_i2s);
+ clk_prepare_enable(clk_i2s);
clk_ena = 1;
pxa_i2s_wait();
@@ -259,7 +259,7 @@ static void pxa2xx_i2s_shutdown(struct snd_pcm_substream *substream,
SACR0 &= ~SACR0_ENB;
pxa_i2s_wait();
if (clk_ena) {
- clk_disable(clk_i2s);
+ clk_disable_unprepare(clk_i2s);
clk_ena = 0;
}
}
diff --git a/sound/soc/samsung/littlemill.c b/sound/soc/samsung/littlemill.c
index e7416851bf7..c82c646b8a0 100644
--- a/sound/soc/samsung/littlemill.c
+++ b/sound/soc/samsung/littlemill.c
@@ -23,10 +23,10 @@ static int littlemill_set_bias_level(struct snd_soc_card *card,
struct snd_soc_dapm_context *dapm,
enum snd_soc_bias_level level)
{
- struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai;
+ struct snd_soc_dai *aif1_dai = card->rtd[0].codec_dai;
int ret;
- if (dapm->dev != codec_dai->dev)
+ if (dapm->dev != aif1_dai->dev)
return 0;
switch (level) {
@@ -36,7 +36,7 @@ static int littlemill_set_bias_level(struct snd_soc_card *card,
* then do so now, otherwise these are noops.
*/
if (dapm->bias_level == SND_SOC_BIAS_STANDBY) {
- ret = snd_soc_dai_set_pll(codec_dai, WM8994_FLL1,
+ ret = snd_soc_dai_set_pll(aif1_dai, WM8994_FLL1,
WM8994_FLL_SRC_MCLK2, 32768,
sample_rate * 512);
if (ret < 0) {
@@ -44,7 +44,7 @@ static int littlemill_set_bias_level(struct snd_soc_card *card,
return ret;
}
- ret = snd_soc_dai_set_sysclk(codec_dai,
+ ret = snd_soc_dai_set_sysclk(aif1_dai,
WM8994_SYSCLK_FLL1,
sample_rate * 512,
SND_SOC_CLOCK_IN);
@@ -66,25 +66,25 @@ static int littlemill_set_bias_level_post(struct snd_soc_card *card,
struct snd_soc_dapm_context *dapm,
enum snd_soc_bias_level level)
{
- struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai;
+ struct snd_soc_dai *aif1_dai = card->rtd[0].codec_dai;
int ret;
- if (dapm->dev != codec_dai->dev)
+ if (dapm->dev != aif1_dai->dev)
return 0;
switch (level) {
case SND_SOC_BIAS_STANDBY:
- ret = snd_soc_dai_set_sysclk(codec_dai, WM8994_SYSCLK_MCLK2,
+ ret = snd_soc_dai_set_sysclk(aif1_dai, WM8994_SYSCLK_MCLK2,
32768, SND_SOC_CLOCK_IN);
if (ret < 0) {
- pr_err("Failed to switch away from FLL: %d\n", ret);
+ pr_err("Failed to switch away from FLL1: %d\n", ret);
return ret;
}
- ret = snd_soc_dai_set_pll(codec_dai, WM8994_FLL1,
+ ret = snd_soc_dai_set_pll(aif1_dai, WM8994_FLL1,
0, 0, 0);
if (ret < 0) {
- pr_err("Failed to stop FLL: %d\n", ret);
+ pr_err("Failed to stop FLL1: %d\n", ret);
return ret;
}
break;
@@ -131,6 +131,14 @@ static struct snd_soc_ops littlemill_ops = {
.hw_params = littlemill_hw_params,
};
+static const struct snd_soc_pcm_stream baseband_params = {
+ .formats = SNDRV_PCM_FMTBIT_S32_LE,
+ .rate_min = 8000,
+ .rate_max = 8000,
+ .channels_min = 2,
+ .channels_max = 2,
+};
+
static struct snd_soc_dai_link littlemill_dai[] = {
{
.name = "CPU",
@@ -143,13 +151,75 @@ static struct snd_soc_dai_link littlemill_dai[] = {
| SND_SOC_DAIFMT_CBM_CFM,
.ops = &littlemill_ops,
},
+ {
+ .name = "Baseband",
+ .stream_name = "Baseband",
+ .cpu_dai_name = "wm8994-aif2",
+ .codec_dai_name = "wm1250-ev1",
+ .codec_name = "wm1250-ev1.1-0027",
+ .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
+ | SND_SOC_DAIFMT_CBM_CFM,
+ .ignore_suspend = 1,
+ .params = &baseband_params,
+ },
};
+static int bbclk_ev(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_card *card = w->dapm->card;
+ struct snd_soc_dai *aif2_dai = card->rtd[1].cpu_dai;
+ int ret;
+
+ switch (event) {
+ case SND_SOC_DAPM_PRE_PMU:
+ ret = snd_soc_dai_set_pll(aif2_dai, WM8994_FLL2,
+ WM8994_FLL_SRC_BCLK, 64 * 8000,
+ 8000 * 256);
+ if (ret < 0) {
+ pr_err("Failed to start FLL: %d\n", ret);
+ return ret;
+ }
+
+ ret = snd_soc_dai_set_sysclk(aif2_dai, WM8994_SYSCLK_FLL2,
+ 8000 * 256,
+ SND_SOC_CLOCK_IN);
+ if (ret < 0) {
+ pr_err("Failed to set SYSCLK: %d\n", ret);
+ return ret;
+ }
+ break;
+ case SND_SOC_DAPM_POST_PMD:
+ ret = snd_soc_dai_set_sysclk(aif2_dai, WM8994_SYSCLK_MCLK2,
+ 32768, SND_SOC_CLOCK_IN);
+ if (ret < 0) {
+ pr_err("Failed to switch away from FLL2: %d\n", ret);
+ return ret;
+ }
+
+ ret = snd_soc_dai_set_pll(aif2_dai, WM8994_FLL2,
+ 0, 0, 0);
+ if (ret < 0) {
+ pr_err("Failed to stop FLL2: %d\n", ret);
+ return ret;
+ }
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
static struct snd_soc_dapm_widget widgets[] = {
SND_SOC_DAPM_HP("Headphone", NULL),
SND_SOC_DAPM_MIC("AMIC", NULL),
SND_SOC_DAPM_MIC("DMIC", NULL),
+
+ SND_SOC_DAPM_SUPPLY_S("Baseband Clock", -1, SND_SOC_NOPM, 0, 0,
+ bbclk_ev,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
};
static struct snd_soc_dapm_route audio_paths[] = {
@@ -162,6 +232,8 @@ static struct snd_soc_dapm_route audio_paths[] = {
{ "DMIC", NULL, "MICBIAS2" }, /* Default for DMICBIAS jumper */
{ "DMIC1DAT", NULL, "DMIC" },
{ "DMIC2DAT", NULL, "DMIC" },
+
+ { "AIF2CLK", NULL, "Baseband Clock" },
};
static struct snd_soc_jack littlemill_headset;
@@ -169,10 +241,16 @@ static struct snd_soc_jack littlemill_headset;
static int littlemill_late_probe(struct snd_soc_card *card)
{
struct snd_soc_codec *codec = card->rtd[0].codec;
- struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai;
+ struct snd_soc_dai *aif1_dai = card->rtd[0].codec_dai;
+ struct snd_soc_dai *aif2_dai = card->rtd[1].cpu_dai;
int ret;
- ret = snd_soc_dai_set_sysclk(codec_dai, WM8994_SYSCLK_MCLK2,
+ ret = snd_soc_dai_set_sysclk(aif1_dai, WM8994_SYSCLK_MCLK2,
+ 32768, SND_SOC_CLOCK_IN);
+ if (ret < 0)
+ return ret;
+
+ ret = snd_soc_dai_set_sysclk(aif2_dai, WM8994_SYSCLK_MCLK2,
32768, SND_SOC_CLOCK_IN);
if (ret < 0)
return ret;
diff --git a/sound/soc/samsung/lowland.c b/sound/soc/samsung/lowland.c
index 4adff934f77..6abf341c4a2 100644
--- a/sound/soc/samsung/lowland.c
+++ b/sound/soc/samsung/lowland.c
@@ -21,33 +21,6 @@
#define MCLK1_RATE (44100 * 512)
#define CLKOUT_RATE (44100 * 256)
-static int lowland_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
-{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
- int ret;
-
- ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S
- | SND_SOC_DAIFMT_NB_NF
- | SND_SOC_DAIFMT_CBM_CFM);
- if (ret < 0)
- return ret;
-
- ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S
- | SND_SOC_DAIFMT_NB_NF
- | SND_SOC_DAIFMT_CBM_CFM);
- if (ret < 0)
- return ret;
-
- return 0;
-}
-
-static struct snd_soc_ops lowland_ops = {
- .hw_params = lowland_hw_params,
-};
-
static struct snd_soc_jack lowland_headset;
/* Headset jack detection DAPM pins */
@@ -101,6 +74,25 @@ static int lowland_wm5100_init(struct snd_soc_pcm_runtime *rtd)
return 0;
}
+static int lowland_wm9081_init(struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_soc_codec *codec = rtd->codec;
+
+ snd_soc_dapm_nc_pin(&codec->dapm, "LINEOUT");
+
+ /* At any time the WM9081 is active it will have this clock */
+ return snd_soc_codec_set_sysclk(codec, WM9081_SYSCLK_MCLK, 0,
+ CLKOUT_RATE, 0);
+}
+
+static const struct snd_soc_pcm_stream sub_params = {
+ .formats = SNDRV_PCM_FMTBIT_S32_LE,
+ .rate_min = 44100,
+ .rate_max = 44100,
+ .channels_min = 2,
+ .channels_max = 2,
+};
+
static struct snd_soc_dai_link lowland_dai[] = {
{
.name = "CPU",
@@ -109,7 +101,8 @@ static struct snd_soc_dai_link lowland_dai[] = {
.codec_dai_name = "wm5100-aif1",
.platform_name = "samsung-audio",
.codec_name = "wm5100.1-001a",
- .ops = &lowland_ops,
+ .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBM_CFM,
.init = lowland_wm5100_init,
},
{
@@ -118,24 +111,20 @@ static struct snd_soc_dai_link lowland_dai[] = {
.cpu_dai_name = "wm5100-aif2",
.codec_dai_name = "wm1250-ev1",
.codec_name = "wm1250-ev1.1-0027",
- .ops = &lowland_ops,
+ .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBM_CFM,
.ignore_suspend = 1,
},
-};
-
-static int lowland_wm9081_init(struct snd_soc_dapm_context *dapm)
-{
- snd_soc_dapm_nc_pin(dapm, "LINEOUT");
-
- /* At any time the WM9081 is active it will have this clock */
- return snd_soc_codec_set_sysclk(dapm->codec, WM9081_SYSCLK_MCLK, 0,
- CLKOUT_RATE, 0);
-}
-
-static struct snd_soc_aux_dev lowland_aux_dev[] = {
{
- .name = "wm9081",
+ .name = "Sub Speaker",
+ .stream_name = "Sub Speaker",
+ .cpu_dai_name = "wm5100-aif3",
+ .codec_dai_name = "wm9081-hifi",
.codec_name = "wm9081.1-006c",
+ .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBM_CFM,
+ .ignore_suspend = 1,
+ .params = &sub_params,
.init = lowland_wm9081_init,
},
};
@@ -180,8 +169,6 @@ static struct snd_soc_card lowland = {
.owner = THIS_MODULE,
.dai_link = lowland_dai,
.num_links = ARRAY_SIZE(lowland_dai),
- .aux_dev = lowland_aux_dev,
- .num_aux_devs = ARRAY_SIZE(lowland_aux_dev),
.codec_conf = lowland_codec_conf,
.num_configs = ARRAY_SIZE(lowland_codec_conf),
diff --git a/sound/soc/samsung/speyside.c b/sound/soc/samsung/speyside.c
index f9ab7707a3e..a4a9fc7e8c7 100644
--- a/sound/soc/samsung/speyside.c
+++ b/sound/soc/samsung/speyside.c
@@ -92,33 +92,6 @@ static int speyside_set_bias_level_post(struct snd_soc_card *card,
return 0;
}
-static int speyside_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
-{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
- int ret;
-
- ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S
- | SND_SOC_DAIFMT_NB_NF
- | SND_SOC_DAIFMT_CBM_CFM);
- if (ret < 0)
- return ret;
-
- ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S
- | SND_SOC_DAIFMT_NB_NF
- | SND_SOC_DAIFMT_CBM_CFM);
- if (ret < 0)
- return ret;
-
- return 0;
-}
-
-static struct snd_soc_ops speyside_ops = {
- .hw_params = speyside_hw_params,
-};
-
static struct snd_soc_jack speyside_headset;
/* Headset jack detection DAPM pins */
@@ -208,7 +181,8 @@ static struct snd_soc_dai_link speyside_dai[] = {
.platform_name = "samsung-audio",
.codec_name = "wm8996.1-001a",
.init = speyside_wm8996_init,
- .ops = &speyside_ops,
+ .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
+ | SND_SOC_DAIFMT_CBM_CFM,
},
{
.name = "Baseband",
@@ -216,7 +190,8 @@ static struct snd_soc_dai_link speyside_dai[] = {
.cpu_dai_name = "wm8996-aif2",
.codec_dai_name = "wm1250-ev1",
.codec_name = "wm1250-ev1.1-0027",
- .ops = &speyside_ops,
+ .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
+ | SND_SOC_DAIFMT_CBM_CFM,
.ignore_suspend = 1,
},
};
diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig
index d8e06a607a2..6bcb1164d59 100644
--- a/sound/soc/sh/Kconfig
+++ b/sound/soc/sh/Kconfig
@@ -22,6 +22,7 @@ config SND_SOC_SH4_SSI
config SND_SOC_SH4_FSI
tristate "SH4 FSI support"
+ select SND_SIMPLE_CARD
help
This option enables FSI sound support
@@ -46,29 +47,6 @@ config SND_SH7760_AC97
This option enables generic sound support for the first
AC97 unit of the SH7760.
-config SND_FSI_AK4642
- tristate "FSI-AK4642 sound support"
- depends on SND_SOC_SH4_FSI && I2C
- select SND_SOC_AK4642
- help
- This option enables generic sound support for the
- FSI - AK4642 unit
-
-config SND_FSI_DA7210
- tristate "FSI-DA7210 sound support"
- depends on SND_SOC_SH4_FSI && I2C
- select SND_SOC_DA7210
- help
- This option enables generic sound support for the
- FSI - DA7210 unit
-
-config SND_FSI_HDMI
- tristate "FSI-HDMI sound support"
- depends on SND_SOC_SH4_FSI && FB_SH_MOBILE_HDMI
- help
- This option enables generic sound support for the
- FSI - HDMI unit
-
config SND_SIU_MIGOR
tristate "SIU sound support on Migo-R"
depends on SH_MIGOR
diff --git a/sound/soc/sh/Makefile b/sound/soc/sh/Makefile
index 94476d4c0fd..849b387d17d 100644
--- a/sound/soc/sh/Makefile
+++ b/sound/soc/sh/Makefile
@@ -14,13 +14,7 @@ obj-$(CONFIG_SND_SOC_SH4_SIU) += snd-soc-siu.o
## boards
snd-soc-sh7760-ac97-objs := sh7760-ac97.o
-snd-soc-fsi-ak4642-objs := fsi-ak4642.o
-snd-soc-fsi-da7210-objs := fsi-da7210.o
-snd-soc-fsi-hdmi-objs := fsi-hdmi.o
snd-soc-migor-objs := migor.o
obj-$(CONFIG_SND_SH7760_AC97) += snd-soc-sh7760-ac97.o
-obj-$(CONFIG_SND_FSI_AK4642) += snd-soc-fsi-ak4642.o
-obj-$(CONFIG_SND_FSI_DA7210) += snd-soc-fsi-da7210.o
-obj-$(CONFIG_SND_FSI_HDMI) += snd-soc-fsi-hdmi.o
obj-$(CONFIG_SND_SIU_MIGOR) += snd-soc-migor.o
diff --git a/sound/soc/sh/fsi-ak4642.c b/sound/soc/sh/fsi-ak4642.c
deleted file mode 100644
index 97f540aabbd..00000000000
--- a/sound/soc/sh/fsi-ak4642.c
+++ /dev/null
@@ -1,108 +0,0 @@
-/*
- * FSI-AK464x sound support for ms7724se
- *
- * Copyright (C) 2009 Renesas Solutions Corp.
- * Kuninori Morimoto <morimoto.kuninori@renesas.com>
- *
- * This file is subject to the terms and conditions of the GNU General Public
- * License. See the file "COPYING" in the main directory of this archive
- * for more details.
- */
-
-#include <linux/platform_device.h>
-#include <linux/module.h>
-#include <sound/sh_fsi.h>
-
-struct fsi_ak4642_data {
- const char *name;
- const char *card;
- const char *cpu_dai;
- const char *codec;
- const char *platform;
- int id;
-};
-
-static int fsi_ak4642_dai_init(struct snd_soc_pcm_runtime *rtd)
-{
- struct snd_soc_dai *codec = rtd->codec_dai;
- struct snd_soc_dai *cpu = rtd->cpu_dai;
- int ret;
-
- ret = snd_soc_dai_set_fmt(codec, SND_SOC_DAIFMT_LEFT_J |
- SND_SOC_DAIFMT_CBM_CFM);
- if (ret < 0)
- return ret;
-
- ret = snd_soc_dai_set_sysclk(codec, 0, 11289600, 0);
- if (ret < 0)
- return ret;
-
- ret = snd_soc_dai_set_fmt(cpu, SND_SOC_DAIFMT_LEFT_J |
- SND_SOC_DAIFMT_CBS_CFS);
-
- return ret;
-}
-
-static struct snd_soc_dai_link fsi_dai_link = {
- .codec_dai_name = "ak4642-hifi",
- .init = fsi_ak4642_dai_init,
-};
-
-static struct snd_soc_card fsi_soc_card = {
- .owner = THIS_MODULE,
- .dai_link = &fsi_dai_link,
- .num_links = 1,
-};
-
-static struct platform_device *fsi_snd_device;
-
-static int fsi_ak4642_probe(struct platform_device *pdev)
-{
- int ret = -ENOMEM;
- struct fsi_ak4642_info *pinfo = pdev->dev.platform_data;
-
- if (!pinfo) {
- dev_err(&pdev->dev, "no info for fsi ak4642\n");
- goto out;
- }
-
- fsi_snd_device = platform_device_alloc("soc-audio", pinfo->id);
- if (!fsi_snd_device)
- goto out;
-
- fsi_dai_link.name = pinfo->name;
- fsi_dai_link.stream_name = pinfo->name;
- fsi_dai_link.cpu_dai_name = pinfo->cpu_dai;
- fsi_dai_link.platform_name = pinfo->platform;
- fsi_dai_link.codec_name = pinfo->codec;
- fsi_soc_card.name = pinfo->card;
-
- platform_set_drvdata(fsi_snd_device, &fsi_soc_card);
- ret = platform_device_add(fsi_snd_device);
-
- if (ret)
- platform_device_put(fsi_snd_device);
-
-out:
- return ret;
-}
-
-static int fsi_ak4642_remove(struct platform_device *pdev)
-{
- platform_device_unregister(fsi_snd_device);
- return 0;
-}
-
-static struct platform_driver fsi_ak4642 = {
- .driver = {
- .name = "fsi-ak4642-audio",
- },
- .probe = fsi_ak4642_probe,
- .remove = fsi_ak4642_remove,
-};
-
-module_platform_driver(fsi_ak4642);
-
-MODULE_LICENSE("GPL");
-MODULE_DESCRIPTION("Generic SH4 FSI-AK4642 sound card");
-MODULE_AUTHOR("Kuninori Morimoto <morimoto.kuninori@renesas.com>");
diff --git a/sound/soc/sh/fsi-da7210.c b/sound/soc/sh/fsi-da7210.c
deleted file mode 100644
index 1dd3354c741..00000000000
--- a/sound/soc/sh/fsi-da7210.c
+++ /dev/null
@@ -1,81 +0,0 @@
-/*
- * fsi-da7210.c
- *
- * Copyright (C) 2009 Renesas Solutions Corp.
- * Kuninori Morimoto <morimoto.kuninori@renesas.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.
- */
-
-#include <linux/platform_device.h>
-#include <linux/module.h>
-#include <sound/sh_fsi.h>
-
-static int fsi_da7210_init(struct snd_soc_pcm_runtime *rtd)
-{
- struct snd_soc_dai *codec = rtd->codec_dai;
- struct snd_soc_dai *cpu = rtd->cpu_dai;
- int ret;
-
- ret = snd_soc_dai_set_fmt(codec,
- SND_SOC_DAIFMT_I2S |
- SND_SOC_DAIFMT_CBM_CFM);
- if (ret < 0)
- return ret;
-
- ret = snd_soc_dai_set_fmt(cpu, SND_SOC_DAIFMT_I2S |
- SND_SOC_DAIFMT_CBS_CFS);
-
- return ret;
-}
-
-static struct snd_soc_dai_link fsi_da7210_dai = {
- .name = "DA7210",
- .stream_name = "DA7210",
- .cpu_dai_name = "fsib-dai", /* FSI B */
- .codec_dai_name = "da7210-hifi",
- .platform_name = "sh_fsi.0",
- .codec_name = "da7210-codec.0-001a",
- .init = fsi_da7210_init,
-};
-
-static struct snd_soc_card fsi_soc_card = {
- .name = "FSI-DA7210",
- .owner = THIS_MODULE,
- .dai_link = &fsi_da7210_dai,
- .num_links = 1,
-};
-
-static struct platform_device *fsi_da7210_snd_device;
-
-static int __init fsi_da7210_sound_init(void)
-{
- int ret;
-
- fsi_da7210_snd_device = platform_device_alloc("soc-audio", FSI_PORT_B);
- if (!fsi_da7210_snd_device)
- return -ENOMEM;
-
- platform_set_drvdata(fsi_da7210_snd_device, &fsi_soc_card);
- ret = platform_device_add(fsi_da7210_snd_device);
- if (ret)
- platform_device_put(fsi_da7210_snd_device);
-
- return ret;
-}
-
-static void __exit fsi_da7210_sound_exit(void)
-{
- platform_device_unregister(fsi_da7210_snd_device);
-}
-
-module_init(fsi_da7210_sound_init);
-module_exit(fsi_da7210_sound_exit);
-
-/* Module information */
-MODULE_DESCRIPTION("ALSA SoC FSI DA2710");
-MODULE_AUTHOR("Kuninori Morimoto <morimoto.kuninori@renesas.com>");
-MODULE_LICENSE("GPL");
diff --git a/sound/soc/sh/fsi-hdmi.c b/sound/soc/sh/fsi-hdmi.c
deleted file mode 100644
index 6e41908323e..00000000000
--- a/sound/soc/sh/fsi-hdmi.c
+++ /dev/null
@@ -1,118 +0,0 @@
-/*
- * FSI - HDMI sound support
- *
- * Copyright (C) 2010 Renesas Solutions Corp.
- * Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
- *
- * This file is subject to the terms and conditions of the GNU General Public
- * License. See the file "COPYING" in the main directory of this archive
- * for more details.
- */
-
-#include <linux/platform_device.h>
-#include <linux/module.h>
-#include <sound/sh_fsi.h>
-
-struct fsi_hdmi_data {
- const char *cpu_dai;
- const char *card;
- int id;
-};
-
-static int fsi_hdmi_dai_init(struct snd_soc_pcm_runtime *rtd)
-{
- struct snd_soc_dai *cpu = rtd->cpu_dai;
- int ret;
-
- ret = snd_soc_dai_set_fmt(cpu, SND_SOC_DAIFMT_CBM_CFM);
-
- return ret;
-}
-
-static struct snd_soc_dai_link fsi_dai_link = {
- .name = "HDMI",
- .stream_name = "HDMI",
- .codec_dai_name = "sh_mobile_hdmi-hifi",
- .platform_name = "sh_fsi2",
- .codec_name = "sh-mobile-hdmi",
- .init = fsi_hdmi_dai_init,
-};
-
-static struct snd_soc_card fsi_soc_card = {
- .owner = THIS_MODULE,
- .dai_link = &fsi_dai_link,
- .num_links = 1,
-};
-
-static struct platform_device *fsi_snd_device;
-
-static int fsi_hdmi_probe(struct platform_device *pdev)
-{
- int ret = -ENOMEM;
- const struct platform_device_id *id_entry;
- struct fsi_hdmi_data *pdata;
-
- id_entry = pdev->id_entry;
- if (!id_entry) {
- dev_err(&pdev->dev, "unknown fsi hdmi\n");
- return -ENODEV;
- }
-
- pdata = (struct fsi_hdmi_data *)id_entry->driver_data;
-
- fsi_snd_device = platform_device_alloc("soc-audio", pdata->id);
- if (!fsi_snd_device)
- goto out;
-
- fsi_dai_link.cpu_dai_name = pdata->cpu_dai;
- fsi_soc_card.name = pdata->card;
-
- platform_set_drvdata(fsi_snd_device, &fsi_soc_card);
- ret = platform_device_add(fsi_snd_device);
-
- if (ret)
- platform_device_put(fsi_snd_device);
-
-out:
- return ret;
-}
-
-static int fsi_hdmi_remove(struct platform_device *pdev)
-{
- platform_device_unregister(fsi_snd_device);
- return 0;
-}
-
-static struct fsi_hdmi_data fsi2_a_hdmi = {
- .cpu_dai = "fsia-dai",
- .card = "FSI2A-HDMI",
- .id = FSI_PORT_A,
-};
-
-static struct fsi_hdmi_data fsi2_b_hdmi = {
- .cpu_dai = "fsib-dai",
- .card = "FSI2B-HDMI",
- .id = FSI_PORT_B,
-};
-
-static struct platform_device_id fsi_id_table[] = {
- /* FSI 2 */
- { "sh_fsi2_a_hdmi", (kernel_ulong_t)&fsi2_a_hdmi },
- { "sh_fsi2_b_hdmi", (kernel_ulong_t)&fsi2_b_hdmi },
- {},
-};
-
-static struct platform_driver fsi_hdmi = {
- .driver = {
- .name = "fsi-hdmi-audio",
- },
- .probe = fsi_hdmi_probe,
- .remove = fsi_hdmi_remove,
- .id_table = fsi_id_table,
-};
-
-module_platform_driver(fsi_hdmi);
-
-MODULE_LICENSE("GPL");
-MODULE_DESCRIPTION("Generic SH4 FSI-HDMI sound card");
-MODULE_AUTHOR("Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>");
diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c
index 74ed2dffbff..2ef98536f1d 100644
--- a/sound/soc/sh/fsi.c
+++ b/sound/soc/sh/fsi.c
@@ -132,6 +132,25 @@
typedef int (*set_rate_func)(struct device *dev, int rate, int enable);
/*
+ * bus options
+ *
+ * 0x000000BA
+ *
+ * A : sample widtht 16bit setting
+ * B : sample widtht 24bit setting
+ */
+
+#define SHIFT_16DATA 0
+#define SHIFT_24DATA 4
+
+#define PACKAGE_24BITBUS_BACK 0
+#define PACKAGE_24BITBUS_FRONT 1
+#define PACKAGE_16BITBUS_STREAM 2
+
+#define BUSOP_SET(s, a) ((a) << SHIFT_ ## s ## DATA)
+#define BUSOP_GET(s, a) (((a) >> SHIFT_ ## s ## DATA) & 0xF)
+
+/*
* FSI driver use below type name for variable
*
* xxx_num : number of data
@@ -189,6 +208,11 @@ struct fsi_stream {
int oerr_num;
/*
+ * bus options
+ */
+ u32 bus_option;
+
+ /*
* thse are initialized by fsi_handler_init()
*/
struct fsi_stream_handler *handler;
@@ -211,8 +235,7 @@ struct fsi_priv {
struct fsi_stream playback;
struct fsi_stream capture;
- u32 do_fmt;
- u32 di_fmt;
+ u32 fmt;
int chan_num:16;
int clk_master:1;
@@ -321,6 +344,10 @@ static void _fsi_master_mask_set(struct fsi_master *master,
/*
* basic function
*/
+static int fsi_version(struct fsi_master *master)
+{
+ return master->core->ver;
+}
static struct fsi_master *fsi_get_master(struct fsi_priv *fsi)
{
@@ -495,6 +522,7 @@ static void fsi_stream_init(struct fsi_priv *fsi,
io->period_samples = fsi_frame2sample(fsi, runtime->period_size);
io->period_pos = 0;
io->sample_width = samples_to_bytes(runtime, 1);
+ io->bus_option = 0;
io->oerr_num = -1; /* ignore 1st err */
io->uerr_num = -1; /* ignore 1st err */
fsi_stream_handler_call(io, init, fsi, io);
@@ -522,6 +550,7 @@ static void fsi_stream_quit(struct fsi_priv *fsi, struct fsi_stream *io)
io->period_samples = 0;
io->period_pos = 0;
io->sample_width = 0;
+ io->bus_option = 0;
io->oerr_num = 0;
io->uerr_num = 0;
spin_unlock_irqrestore(&master->lock, flags);
@@ -581,6 +610,53 @@ static int fsi_stream_remove(struct fsi_priv *fsi)
}
/*
+ * format/bus/dma setting
+ */
+static void fsi_format_bus_setup(struct fsi_priv *fsi, struct fsi_stream *io,
+ u32 bus, struct device *dev)
+{
+ struct fsi_master *master = fsi_get_master(fsi);
+ int is_play = fsi_stream_is_play(fsi, io);
+ u32 fmt = fsi->fmt;
+
+ if (fsi_version(master) >= 2) {
+ u32 dma = 0;
+
+ /*
+ * FSI2 needs DMA/Bus setting
+ */
+ switch (bus) {
+ case PACKAGE_24BITBUS_FRONT:
+ fmt |= CR_BWS_24;
+ dma |= VDMD_FRONT;
+ dev_dbg(dev, "24bit bus / package in front\n");
+ break;
+ case PACKAGE_16BITBUS_STREAM:
+ fmt |= CR_BWS_16;
+ dma |= VDMD_STREAM;
+ dev_dbg(dev, "16bit bus / stream mode\n");
+ break;
+ case PACKAGE_24BITBUS_BACK:
+ default:
+ fmt |= CR_BWS_24;
+ dma |= VDMD_BACK;
+ dev_dbg(dev, "24bit bus / package in back\n");
+ break;
+ }
+
+ if (is_play)
+ fsi_reg_write(fsi, OUT_DMAC, dma);
+ else
+ fsi_reg_write(fsi, IN_DMAC, dma);
+ }
+
+ if (is_play)
+ fsi_reg_write(fsi, DO_FMT, fmt);
+ else
+ fsi_reg_write(fsi, DI_FMT, fmt);
+}
+
+/*
* irq function
*/
@@ -629,11 +705,6 @@ static void fsi_spdif_clk_ctrl(struct fsi_priv *fsi, int enable)
struct fsi_master *master = fsi_get_master(fsi);
u32 mask, val;
- if (master->core->ver < 2) {
- pr_err("fsi: register access err (%s)\n", __func__);
- return;
- }
-
mask = BP | SE;
val = enable ? mask : 0;
@@ -648,9 +719,7 @@ static void fsi_spdif_clk_ctrl(struct fsi_priv *fsi, int enable)
static int fsi_set_master_clk(struct device *dev, struct fsi_priv *fsi,
long rate, int enable)
{
- struct fsi_master *master = fsi_get_master(fsi);
set_rate_func set_rate = fsi_get_info_set_rate(fsi);
- int fsi_ver = master->core->ver;
int ret;
if (!set_rate)
@@ -682,10 +751,7 @@ static int fsi_set_master_clk(struct device *dev, struct fsi_priv *fsi,
data |= (0x3 << 12);
break;
case SH_FSI_ACKMD_32:
- if (fsi_ver < 2)
- dev_err(dev, "unsupported ACKMD\n");
- else
- data |= (0x4 << 12);
+ data |= (0x4 << 12);
break;
}
@@ -708,10 +774,7 @@ static int fsi_set_master_clk(struct device *dev, struct fsi_priv *fsi,
data |= (0x4 << 8);
break;
case SH_FSI_BPFMD_16:
- if (fsi_ver < 2)
- dev_err(dev, "unsupported ACKMD\n");
- else
- data |= (0x7 << 8);
+ data |= (0x7 << 8);
break;
}
@@ -728,11 +791,26 @@ static int fsi_set_master_clk(struct device *dev, struct fsi_priv *fsi,
*/
static void fsi_pio_push16(struct fsi_priv *fsi, u8 *_buf, int samples)
{
- u16 *buf = (u16 *)_buf;
+ u32 enable_stream = fsi_get_info_flags(fsi) & SH_FSI_ENABLE_STREAM_MODE;
int i;
- for (i = 0; i < samples; i++)
- fsi_reg_write(fsi, DODT, ((u32)*(buf + i) << 8));
+ if (enable_stream) {
+ /*
+ * stream mode
+ * see
+ * fsi_pio_push_init()
+ */
+ u32 *buf = (u32 *)_buf;
+
+ for (i = 0; i < samples / 2; i++)
+ fsi_reg_write(fsi, DODT, buf[i]);
+ } else {
+ /* normal mode */
+ u16 *buf = (u16 *)_buf;
+
+ for (i = 0; i < samples; i++)
+ fsi_reg_write(fsi, DODT, ((u32)*(buf + i) << 8));
+ }
}
static void fsi_pio_pop16(struct fsi_priv *fsi, u8 *_buf, int samples)
@@ -872,12 +950,44 @@ static void fsi_pio_start_stop(struct fsi_priv *fsi, struct fsi_stream *io,
fsi_master_mask_set(master, CLK_RST, clk, (enable) ? clk : 0);
}
+static int fsi_pio_push_init(struct fsi_priv *fsi, struct fsi_stream *io)
+{
+ u32 enable_stream = fsi_get_info_flags(fsi) & SH_FSI_ENABLE_STREAM_MODE;
+
+ /*
+ * we can use 16bit stream mode
+ * when "playback" and "16bit data"
+ * and platform allows "stream mode"
+ * see
+ * fsi_pio_push16()
+ */
+ if (enable_stream)
+ io->bus_option = BUSOP_SET(24, PACKAGE_24BITBUS_BACK) |
+ BUSOP_SET(16, PACKAGE_16BITBUS_STREAM);
+ else
+ io->bus_option = BUSOP_SET(24, PACKAGE_24BITBUS_BACK) |
+ BUSOP_SET(16, PACKAGE_24BITBUS_BACK);
+ return 0;
+}
+
+static int fsi_pio_pop_init(struct fsi_priv *fsi, struct fsi_stream *io)
+{
+ /*
+ * always 24bit bus, package back when "capture"
+ */
+ io->bus_option = BUSOP_SET(24, PACKAGE_24BITBUS_BACK) |
+ BUSOP_SET(16, PACKAGE_24BITBUS_BACK);
+ return 0;
+}
+
static struct fsi_stream_handler fsi_pio_push_handler = {
+ .init = fsi_pio_push_init,
.transfer = fsi_pio_push,
.start_stop = fsi_pio_start_stop,
};
static struct fsi_stream_handler fsi_pio_pop_handler = {
+ .init = fsi_pio_pop_init,
.transfer = fsi_pio_pop,
.start_stop = fsi_pio_start_stop,
};
@@ -919,6 +1029,13 @@ static int fsi_dma_init(struct fsi_priv *fsi, struct fsi_stream *io)
enum dma_data_direction dir = fsi_stream_is_play(fsi, io) ?
DMA_TO_DEVICE : DMA_FROM_DEVICE;
+ /*
+ * 24bit data : 24bit bus / package in back
+ * 16bit data : 16bit bus / stream mode
+ */
+ io->bus_option = BUSOP_SET(24, PACKAGE_24BITBUS_BACK) |
+ BUSOP_SET(16, PACKAGE_16BITBUS_STREAM);
+
io->dma = dma_map_single(dai->dev, runtime->dma_area,
snd_pcm_lib_buffer_bytes(io->substream), dir);
return 0;
@@ -935,6 +1052,13 @@ static int fsi_dma_quit(struct fsi_priv *fsi, struct fsi_stream *io)
return 0;
}
+static dma_addr_t fsi_dma_get_area(struct fsi_stream *io)
+{
+ struct snd_pcm_runtime *runtime = io->substream->runtime;
+
+ return io->dma + samples_to_bytes(runtime, io->buff_sample_pos);
+}
+
static void fsi_dma_complete(void *data)
{
struct fsi_stream *io = (struct fsi_stream *)data;
@@ -944,7 +1068,7 @@ static void fsi_dma_complete(void *data)
enum dma_data_direction dir = fsi_stream_is_play(fsi, io) ?
DMA_TO_DEVICE : DMA_FROM_DEVICE;
- dma_sync_single_for_cpu(dai->dev, io->dma,
+ dma_sync_single_for_cpu(dai->dev, fsi_dma_get_area(io),
samples_to_bytes(runtime, io->period_samples), dir);
io->buff_sample_pos += io->period_samples;
@@ -961,13 +1085,6 @@ static void fsi_dma_complete(void *data)
snd_pcm_period_elapsed(io->substream);
}
-static dma_addr_t fsi_dma_get_area(struct fsi_stream *io)
-{
- struct snd_pcm_runtime *runtime = io->substream->runtime;
-
- return io->dma + samples_to_bytes(runtime, io->buff_sample_pos);
-}
-
static void fsi_dma_do_tasklet(unsigned long data)
{
struct fsi_stream *io = (struct fsi_stream *)data;
@@ -993,7 +1110,7 @@ static void fsi_dma_do_tasklet(unsigned long data)
len = samples_to_bytes(runtime, io->period_samples);
buf = fsi_dma_get_area(io);
- dma_sync_single_for_device(dai->dev, io->dma, len, dir);
+ dma_sync_single_for_device(dai->dev, buf, len, dir);
s