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-rw-r--r--sound/soc/Kconfig1
-rw-r--r--sound/soc/Makefile1
-rw-r--r--sound/soc/atmel/Kconfig2
-rw-r--r--sound/soc/atmel/atmel_ssc_dai.c13
-rw-r--r--sound/soc/atmel/sam9g20_wm8731.c20
-rw-r--r--sound/soc/blackfin/Kconfig9
-rw-r--r--sound/soc/cirrus/Kconfig4
-rw-r--r--sound/soc/cirrus/snappercl15.c18
-rw-r--r--sound/soc/codecs/88pm860x-codec.c119
-rw-r--r--sound/soc/codecs/Kconfig195
-rw-r--r--sound/soc/codecs/Makefile39
-rw-r--r--sound/soc/codecs/ad1836.c4
-rw-r--r--sound/soc/codecs/ad193x-i2c.c54
-rw-r--r--sound/soc/codecs/ad193x-spi.c48
-rw-r--r--sound/soc/codecs/ad193x.c154
-rw-r--r--sound/soc/codecs/ad193x.h7
-rw-r--r--sound/soc/codecs/adau1373.c39
-rw-r--r--sound/soc/codecs/adau1977-i2c.c59
-rw-r--r--sound/soc/codecs/adau1977-spi.c76
-rw-r--r--sound/soc/codecs/adau1977.c1018
-rw-r--r--sound/soc/codecs/adau1977.h37
-rw-r--r--sound/soc/codecs/adav801.c53
-rw-r--r--sound/soc/codecs/adav803.c50
-rw-r--r--sound/soc/codecs/adav80x.c159
-rw-r--r--sound/soc/codecs/adav80x.h7
-rw-r--r--sound/soc/codecs/ak4104.c2
-rw-r--r--sound/soc/codecs/ak4535.c9
-rw-r--r--sound/soc/codecs/ak4641.c24
-rw-r--r--sound/soc/codecs/ak4642.c8
-rw-r--r--sound/soc/codecs/ak4671.c250
-rw-r--r--sound/soc/codecs/ak4671.h2
-rw-r--r--sound/soc/codecs/alc5623.c120
-rw-r--r--sound/soc/codecs/alc5632.c48
-rw-r--r--sound/soc/codecs/arizona.c325
-rw-r--r--sound/soc/codecs/cq93vc.c3
-rw-r--r--sound/soc/codecs/cs4270.c9
-rw-r--r--sound/soc/codecs/cs4271.c63
-rw-r--r--sound/soc/codecs/cs42l51.c99
-rw-r--r--sound/soc/codecs/cs42l52.c109
-rw-r--r--sound/soc/codecs/cs42l73.c72
-rw-r--r--sound/soc/codecs/cs42xx8-i2c.c64
-rw-r--r--sound/soc/codecs/cs42xx8.c602
-rw-r--r--sound/soc/codecs/cs42xx8.h238
-rw-r--r--sound/soc/codecs/da7210.c28
-rw-r--r--sound/soc/codecs/da7213.c159
-rw-r--r--sound/soc/codecs/da732x.c196
-rw-r--r--sound/soc/codecs/da732x.h3
-rw-r--r--sound/soc/codecs/da9055.c100
-rw-r--r--sound/soc/codecs/isabelle.c19
-rw-r--r--sound/soc/codecs/lm4857.c3
-rw-r--r--sound/soc/codecs/lm49453.c47
-rw-r--r--sound/soc/codecs/max9768.c5
-rw-r--r--sound/soc/codecs/max98088.c47
-rw-r--r--sound/soc/codecs/max98090.c181
-rw-r--r--sound/soc/codecs/max98090.h1
-rw-r--r--sound/soc/codecs/max98095.c60
-rw-r--r--sound/soc/codecs/max9850.c8
-rw-r--r--sound/soc/codecs/mc13783.c30
-rw-r--r--sound/soc/codecs/ml26124.c22
-rw-r--r--sound/soc/codecs/pcm1681.c15
-rw-r--r--sound/soc/codecs/pcm1792a.c33
-rw-r--r--sound/soc/codecs/pcm512x-i2c.c71
-rw-r--r--sound/soc/codecs/pcm512x-spi.c69
-rw-r--r--sound/soc/codecs/pcm512x.c589
-rw-r--r--sound/soc/codecs/pcm512x.h171
-rw-r--r--sound/soc/codecs/rt5631.c84
-rw-r--r--sound/soc/codecs/rt5640.c86
-rw-r--r--sound/soc/codecs/sgtl5000.c18
-rw-r--r--sound/soc/codecs/si476x.c6
-rw-r--r--sound/soc/codecs/sirf-audio-codec.c524
-rw-r--r--sound/soc/codecs/sirf-audio-codec.h75
-rw-r--r--sound/soc/codecs/sn95031.c46
-rw-r--r--sound/soc/codecs/ssm2518.c24
-rw-r--r--sound/soc/codecs/ssm2602-i2c.c57
-rw-r--r--sound/soc/codecs/ssm2602-spi.c41
-rw-r--r--sound/soc/codecs/ssm2602.c180
-rw-r--r--sound/soc/codecs/ssm2602.h14
-rw-r--r--sound/soc/codecs/sta32x.c14
-rw-r--r--sound/soc/codecs/sta529.c15
-rw-r--r--sound/soc/codecs/stac9766.c38
-rw-r--r--sound/soc/codecs/tlv320aic23-i2c.c59
-rw-r--r--sound/soc/codecs/tlv320aic23-spi.c56
-rw-r--r--sound/soc/codecs/tlv320aic23.c79
-rw-r--r--sound/soc/codecs/tlv320aic23.h6
-rw-r--r--sound/soc/codecs/tlv320aic26.c7
-rw-r--r--sound/soc/codecs/tlv320aic31xx.c1280
-rw-r--r--sound/soc/codecs/tlv320aic31xx.h258
-rw-r--r--sound/soc/codecs/tlv320aic32x4.c234
-rw-r--r--sound/soc/codecs/tlv320aic3x.c6
-rw-r--r--sound/soc/codecs/tlv320dac33.c35
-rw-r--r--sound/soc/codecs/twl4030.c101
-rw-r--r--sound/soc/codecs/twl6040.c17
-rw-r--r--sound/soc/codecs/uda134x.c3
-rw-r--r--sound/soc/codecs/uda1380.c43
-rw-r--r--sound/soc/codecs/wl1273.c9
-rw-r--r--sound/soc/codecs/wm2000.c2
-rw-r--r--sound/soc/codecs/wm2200.c25
-rw-r--r--sound/soc/codecs/wm5100.c43
-rw-r--r--sound/soc/codecs/wm5102.c32
-rw-r--r--sound/soc/codecs/wm5110.c22
-rw-r--r--sound/soc/codecs/wm8350.c4
-rw-r--r--sound/soc/codecs/wm8400.c3
-rw-r--r--sound/soc/codecs/wm8510.c10
-rw-r--r--sound/soc/codecs/wm8523.c11
-rw-r--r--sound/soc/codecs/wm8580.c9
-rw-r--r--sound/soc/codecs/wm8711.c8
-rw-r--r--sound/soc/codecs/wm8728.c11
-rw-r--r--sound/soc/codecs/wm8731.c11
-rw-r--r--sound/soc/codecs/wm8737.c56
-rw-r--r--sound/soc/codecs/wm8741.c40
-rw-r--r--sound/soc/codecs/wm8750.c6
-rw-r--r--sound/soc/codecs/wm8753.c12
-rw-r--r--sound/soc/codecs/wm8770.c6
-rw-r--r--sound/soc/codecs/wm8776.c6
-rw-r--r--sound/soc/codecs/wm8804.c10
-rw-r--r--sound/soc/codecs/wm8900.c8
-rw-r--r--sound/soc/codecs/wm8903.c118
-rw-r--r--sound/soc/codecs/wm8904.c86
-rw-r--r--sound/soc/codecs/wm8940.c26
-rw-r--r--sound/soc/codecs/wm8955.c19
-rw-r--r--sound/soc/codecs/wm8958-dsp2.c8
-rw-r--r--sound/soc/codecs/wm8960.c6
-rw-r--r--sound/soc/codecs/wm8961.c23
-rw-r--r--sound/soc/codecs/wm8962.c87
-rw-r--r--sound/soc/codecs/wm8971.c6
-rw-r--r--sound/soc/codecs/wm8974.c10
-rw-r--r--sound/soc/codecs/wm8978.c38
-rw-r--r--sound/soc/codecs/wm8983.c51
-rw-r--r--sound/soc/codecs/wm8985.c46
-rw-r--r--sound/soc/codecs/wm8988.c70
-rw-r--r--sound/soc/codecs/wm8990.c49
-rw-r--r--sound/soc/codecs/wm8991.c52
-rw-r--r--sound/soc/codecs/wm8993.c73
-rw-r--r--sound/soc/codecs/wm8994.c55
-rw-r--r--sound/soc/codecs/wm8995.c50
-rw-r--r--sound/soc/codecs/wm8996.c87
-rw-r--r--sound/soc/codecs/wm8997.c29
-rw-r--r--sound/soc/codecs/wm9081.c34
-rw-r--r--sound/soc/codecs/wm9090.c10
-rw-r--r--sound/soc/codecs/wm9705.c12
-rw-r--r--sound/soc/codecs/wm_adsp.c50
-rw-r--r--sound/soc/codecs/wm_hubs.c16
-rw-r--r--sound/soc/davinci/davinci-evm.c80
-rw-r--r--sound/soc/davinci/davinci-mcasp.c213
-rw-r--r--sound/soc/davinci/edma-pcm.c57
-rw-r--r--sound/soc/davinci/edma-pcm.h25
-rw-r--r--sound/soc/fsl/Kconfig12
-rw-r--r--sound/soc/fsl/eukrea-tlv320.c108
-rw-r--r--sound/soc/fsl/fsl_esai.c32
-rw-r--r--sound/soc/fsl/fsl_sai.c332
-rw-r--r--sound/soc/fsl/fsl_sai.h48
-rw-r--r--sound/soc/fsl/fsl_spdif.c9
-rw-r--r--sound/soc/fsl/imx-pcm-fiq.c7
-rw-r--r--sound/soc/fsl/wm1133-ev1.c11
-rw-r--r--sound/soc/generic/simple-card.c295
-rw-r--r--sound/soc/intel/Kconfig42
-rw-r--r--sound/soc/intel/Makefile27
-rw-r--r--sound/soc/intel/byt-rt5640.c187
-rw-r--r--sound/soc/intel/haswell.c235
-rw-r--r--sound/soc/intel/mfld_machine.c108
-rw-r--r--sound/soc/intel/sst-acpi.c284
-rw-r--r--sound/soc/intel/sst-baytrail-dsp.c372
-rw-r--r--sound/soc/intel/sst-baytrail-ipc.c867
-rw-r--r--sound/soc/intel/sst-baytrail-ipc.h69
-rw-r--r--sound/soc/intel/sst-baytrail-pcm.c422
-rw-r--r--sound/soc/intel/sst-dsp-priv.h309
-rw-r--r--sound/soc/intel/sst-dsp.c385
-rw-r--r--sound/soc/intel/sst-dsp.h233
-rw-r--r--sound/soc/intel/sst-firmware.c587
-rw-r--r--sound/soc/intel/sst-haswell-dsp.c517
-rw-r--r--sound/soc/intel/sst-haswell-ipc.c1785
-rw-r--r--sound/soc/intel/sst-haswell-ipc.h488
-rw-r--r--sound/soc/intel/sst-haswell-pcm.c872
-rw-r--r--sound/soc/intel/sst-mfld-dsp.h (renamed from sound/soc/intel/sst_dsp.h)8
-rw-r--r--sound/soc/intel/sst-mfld-platform.c (renamed from sound/soc/intel/sst_platform.c)8
-rw-r--r--sound/soc/intel/sst-mfld-platform.h (renamed from sound/soc/intel/sst_platform.h)4
-rw-r--r--sound/soc/kirkwood/Kconfig11
-rw-r--r--sound/soc/kirkwood/Makefile2
-rw-r--r--sound/soc/kirkwood/armada-370-db.c148
-rw-r--r--sound/soc/kirkwood/kirkwood-i2s.c1
-rw-r--r--sound/soc/omap/Kconfig4
-rw-r--r--sound/soc/omap/ams-delta.c55
-rw-r--r--sound/soc/omap/n810.c22
-rw-r--r--sound/soc/omap/omap-abe-twl6040.c3
-rw-r--r--sound/soc/omap/rx51.c22
-rw-r--r--sound/soc/pxa/corgi.c49
-rw-r--r--sound/soc/pxa/e740_wm9705.c10
-rw-r--r--sound/soc/pxa/e750_wm9705.c10
-rw-r--r--sound/soc/pxa/e800_wm9712.c19
-rw-r--r--sound/soc/pxa/magician.c60
-rw-r--r--sound/soc/pxa/mioa701_wm9713.c19
-rw-r--r--sound/soc/pxa/poodle.c7
-rw-r--r--sound/soc/pxa/spitz.c58
-rw-r--r--sound/soc/pxa/tosa.c67
-rw-r--r--sound/soc/pxa/zylonite.c17
-rw-r--r--sound/soc/s6000/s6105-ipcam.c28
-rw-r--r--sound/soc/samsung/Kconfig2
-rw-r--r--sound/soc/samsung/h1940_uda1380.c7
-rw-r--r--sound/soc/samsung/neo1973_wm8753.c168
-rw-r--r--sound/soc/samsung/rx1950_uda1380.c5
-rw-r--r--sound/soc/samsung/smdk_wm8994.c2
-rw-r--r--sound/soc/samsung/tobermory.c2
-rw-r--r--sound/soc/sh/fsi.c2
-rw-r--r--sound/soc/sh/migor.c19
-rw-r--r--sound/soc/sh/rcar/Makefile2
-rw-r--r--sound/soc/sh/rcar/adg.c229
-rw-r--r--sound/soc/sh/rcar/core.c426
-rw-r--r--sound/soc/sh/rcar/gen.c113
-rw-r--r--sound/soc/sh/rcar/rsnd.h206
-rw-r--r--sound/soc/sh/rcar/scu.c384
-rw-r--r--sound/soc/sh/rcar/src.c723
-rw-r--r--sound/soc/sh/rcar/ssi.c388
-rw-r--r--sound/soc/sirf/Kconfig14
-rw-r--r--sound/soc/sirf/Makefile5
-rw-r--r--sound/soc/sirf/sirf-audio-port.c194
-rw-r--r--sound/soc/sirf/sirf-audio-port.h62
-rw-r--r--sound/soc/sirf/sirf-audio.c156
-rw-r--r--sound/soc/soc-cache.c13
-rw-r--r--sound/soc/soc-compress.c65
-rw-r--r--sound/soc/soc-core.c599
-rw-r--r--sound/soc/soc-dapm.c465
-rw-r--r--sound/soc/soc-io.c99
-rw-r--r--sound/soc/soc-jack.c5
-rw-r--r--sound/soc/soc-pcm.c112
-rw-r--r--sound/soc/spear/spdif_out.c10
-rw-r--r--sound/soc/tegra/Kconfig2
-rw-r--r--sound/soc/tegra/tegra20_ac97.c2
-rw-r--r--sound/soc/tegra/tegra20_das.c2
-rw-r--r--sound/soc/tegra/tegra20_i2s.c2
-rw-r--r--sound/soc/tegra/tegra20_spdif.c2
-rw-r--r--sound/soc/tegra/tegra30_ahub.c4
-rw-r--r--sound/soc/tegra/tegra30_i2s.c2
232 files changed, 19701 insertions, 5393 deletions
diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig
index d62ce48..0060b31 100644
--- a/sound/soc/Kconfig
+++ b/sound/soc/Kconfig
@@ -50,6 +50,7 @@ source "sound/soc/pxa/Kconfig"
source "sound/soc/samsung/Kconfig"
source "sound/soc/s6000/Kconfig"
source "sound/soc/sh/Kconfig"
+source "sound/soc/sirf/Kconfig"
source "sound/soc/spear/Kconfig"
source "sound/soc/tegra/Kconfig"
source "sound/soc/txx9/Kconfig"
diff --git a/sound/soc/Makefile b/sound/soc/Makefile
index 62a1822e7..5f1df02 100644
--- a/sound/soc/Makefile
+++ b/sound/soc/Makefile
@@ -27,6 +27,7 @@ obj-$(CONFIG_SND_SOC) += pxa/
obj-$(CONFIG_SND_SOC) += samsung/
obj-$(CONFIG_SND_SOC) += s6000/
obj-$(CONFIG_SND_SOC) += sh/
+obj-$(CONFIG_SND_SOC) += sirf/
obj-$(CONFIG_SND_SOC) += spear/
obj-$(CONFIG_SND_SOC) += tegra/
obj-$(CONFIG_SND_SOC) += txx9/
diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig
index e634eb7..4789619 100644
--- a/sound/soc/atmel/Kconfig
+++ b/sound/soc/atmel/Kconfig
@@ -58,6 +58,6 @@ config SND_AT91_SOC_AFEB9260
depends on ARCH_AT91 && ATMEL_SSC && ARCH_AT91 && MACH_AFEB9260 && SND_ATMEL_SOC
select SND_ATMEL_SOC_PDC
select SND_ATMEL_SOC_SSC
- select SND_SOC_TLV320AIC23
+ select SND_SOC_TLV320AIC23_I2C
help
Say Y here to support sound on AFEB9260 board.
diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c
index 1ead3c9..de433cfd 100644
--- a/sound/soc/atmel/atmel_ssc_dai.c
+++ b/sound/soc/atmel/atmel_ssc_dai.c
@@ -341,6 +341,7 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream,
{
int id = dai->id;
struct atmel_ssc_info *ssc_p = &ssc_info[id];
+ struct ssc_device *ssc = ssc_p->ssc;
struct atmel_pcm_dma_params *dma_params;
int dir, channels, bits;
u32 tfmr, rfmr, tcmr, rcmr;
@@ -466,7 +467,8 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream,
| SSC_BF(RCMR_START, start_event)
| SSC_BF(RCMR_CKI, SSC_CKI_RISING)
| SSC_BF(RCMR_CKO, SSC_CKO_NONE)
- | SSC_BF(RCMR_CKS, SSC_CKS_CLOCK);
+ | SSC_BF(RCMR_CKS, ssc->clk_from_rk_pin ?
+ SSC_CKS_PIN : SSC_CKS_CLOCK);
rfmr = SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE)
| SSC_BF(RFMR_FSOS, SSC_FSOS_NONE)
@@ -481,7 +483,8 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream,
| SSC_BF(TCMR_START, start_event)
| SSC_BF(TCMR_CKI, SSC_CKI_FALLING)
| SSC_BF(TCMR_CKO, SSC_CKO_NONE)
- | SSC_BF(TCMR_CKS, SSC_CKS_PIN);
+ | SSC_BF(TCMR_CKS, ssc->clk_from_rk_pin ?
+ SSC_CKS_CLOCK : SSC_CKS_PIN);
tfmr = SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE)
| SSC_BF(TFMR_FSDEN, 0)
@@ -550,7 +553,8 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream,
| SSC_BF(RCMR_START, SSC_START_RISING_RF)
| SSC_BF(RCMR_CKI, SSC_CKI_RISING)
| SSC_BF(RCMR_CKO, SSC_CKO_NONE)
- | SSC_BF(RCMR_CKS, SSC_CKS_PIN);
+ | SSC_BF(RCMR_CKS, ssc->clk_from_rk_pin ?
+ SSC_CKS_PIN : SSC_CKS_CLOCK);
rfmr = SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE)
| SSC_BF(RFMR_FSOS, SSC_FSOS_NONE)
@@ -565,7 +569,8 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream,
| SSC_BF(TCMR_START, SSC_START_RISING_RF)
| SSC_BF(TCMR_CKI, SSC_CKI_FALLING)
| SSC_BF(TCMR_CKO, SSC_CKO_NONE)
- | SSC_BF(TCMR_CKS, SSC_CKS_PIN);
+ | SSC_BF(RCMR_CKS, ssc->clk_from_rk_pin ?
+ SSC_CKS_CLOCK : SSC_CKS_PIN);
tfmr = SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE)
| SSC_BF(TFMR_FSDEN, 0)
diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c
index f15bff1..174bd54 100644
--- a/sound/soc/atmel/sam9g20_wm8731.c
+++ b/sound/soc/atmel/sam9g20_wm8731.c
@@ -155,25 +155,14 @@ static int at91sam9g20ek_wm8731_init(struct snd_soc_pcm_runtime *rtd)
return ret;
}
- /* Add specific widgets */
- snd_soc_dapm_new_controls(dapm, at91sam9g20ek_dapm_widgets,
- ARRAY_SIZE(at91sam9g20ek_dapm_widgets));
- /* Set up specific audio path interconnects */
- snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon));
-
/* not connected */
snd_soc_dapm_nc_pin(dapm, "RLINEIN");
snd_soc_dapm_nc_pin(dapm, "LLINEIN");
-#ifdef ENABLE_MIC_INPUT
- snd_soc_dapm_enable_pin(dapm, "Int Mic");
-#else
- snd_soc_dapm_nc_pin(dapm, "Int Mic");
+#ifndef ENABLE_MIC_INPUT
+ snd_soc_dapm_nc_pin(&rtd->card->dapm, "Int Mic");
#endif
- /* always connected */
- snd_soc_dapm_enable_pin(dapm, "Ext Spk");
-
return 0;
}
@@ -194,6 +183,11 @@ static struct snd_soc_card snd_soc_at91sam9g20ek = {
.dai_link = &at91sam9g20ek_dai,
.num_links = 1,
.set_bias_level = at91sam9g20ek_set_bias_level,
+
+ .dapm_widgets = at91sam9g20ek_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(at91sam9g20ek_dapm_widgets),
+ .dapm_routes = intercon,
+ .num_dapm_routes = ARRAY_SIZE(intercon),
};
static int at91sam9g20ek_audio_probe(struct platform_device *pdev)
diff --git a/sound/soc/blackfin/Kconfig b/sound/soc/blackfin/Kconfig
index 4544d8e..6347d59 100644
--- a/sound/soc/blackfin/Kconfig
+++ b/sound/soc/blackfin/Kconfig
@@ -14,7 +14,8 @@ config SND_BF5XX_SOC_SSM2602
depends on SND_BF5XX_I2S && SND_SOC_I2C_AND_SPI
select SND_BF5XX_SOC_I2S if !BF60x
select SND_BF6XX_SOC_I2S if BF60x
- select SND_SOC_SSM2602
+ select SND_SOC_SSM2602_SPI if SPI_MASTER
+ select SND_SOC_SSM2602_I2C if I2C
help
Say Y if you want to add support for the Analog Devices
SSM2602 Audio Codec Add-On Card.
@@ -46,7 +47,8 @@ config SND_SOC_BFIN_EVAL_ADAV80X
tristate "Support for the EVAL-ADAV80X boards on Blackfin eval boards"
depends on SND_BF5XX_I2S && SND_SOC_I2C_AND_SPI
select SND_BF5XX_SOC_I2S
- select SND_SOC_ADAV80X
+ select SND_SOC_ADAV801 if SPI_MASTER
+ select SND_SOC_ADAV803 if I2C
help
Say Y if you want to add support for the Analog Devices EVAL-ADAV801 or
EVAL-ADAV803 board connected to one of the Blackfin evaluation boards
@@ -67,7 +69,8 @@ config SND_BF5XX_SOC_AD193X
tristate "SoC AD193X Audio support for Blackfin"
depends on SND_BF5XX_I2S && SND_SOC_I2C_AND_SPI
select SND_BF5XX_SOC_I2S
- select SND_SOC_AD193X
+ select SND_SOC_AD193X_I2C if I2C
+ select SND_SOC_AD193X_SPI if SPI_MASTER
help
Say Y if you want to add support for AD193X codec on Blackfin.
This driver supports AD1936, AD1937, AD1938 and AD1939.
diff --git a/sound/soc/cirrus/Kconfig b/sound/soc/cirrus/Kconfig
index 06f938d..5477c54 100644
--- a/sound/soc/cirrus/Kconfig
+++ b/sound/soc/cirrus/Kconfig
@@ -1,6 +1,6 @@
config SND_EP93XX_SOC
tristate "SoC Audio support for the Cirrus Logic EP93xx series"
- depends on (ARCH_EP93XX || COMPILE_TEST) && SND_SOC
+ depends on ARCH_EP93XX || COMPILE_TEST
select SND_SOC_GENERIC_DMAENGINE_PCM
help
Say Y or M if you want to add support for codecs attached to
@@ -18,7 +18,7 @@ config SND_EP93XX_SOC_SNAPPERCL15
tristate "SoC Audio support for Bluewater Systems Snapper CL15 module"
depends on SND_EP93XX_SOC && MACH_SNAPPER_CL15
select SND_EP93XX_SOC_I2S
- select SND_SOC_TLV320AIC23
+ select SND_SOC_TLV320AIC23_I2C
help
Say Y or M here if you want to add support for I2S audio on the
Bluewater Systems Snapper CL15 module.
diff --git a/sound/soc/cirrus/snappercl15.c b/sound/soc/cirrus/snappercl15.c
index 29238a7..5b68b10 100644
--- a/sound/soc/cirrus/snappercl15.c
+++ b/sound/soc/cirrus/snappercl15.c
@@ -65,18 +65,6 @@ static const struct snd_soc_dapm_route audio_map[] = {
{"MICIN", NULL, "Mic Jack"},
};
-static int snappercl15_tlv320aic23_init(struct snd_soc_pcm_runtime *rtd)
-{
- struct snd_soc_codec *codec = rtd->codec;
- struct snd_soc_dapm_context *dapm = &codec->dapm;
-
- snd_soc_dapm_new_controls(dapm, tlv320aic23_dapm_widgets,
- ARRAY_SIZE(tlv320aic23_dapm_widgets));
-
- snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
- return 0;
-}
-
static struct snd_soc_dai_link snappercl15_dai = {
.name = "tlv320aic23",
.stream_name = "AIC23",
@@ -84,7 +72,6 @@ static struct snd_soc_dai_link snappercl15_dai = {
.codec_dai_name = "tlv320aic23-hifi",
.codec_name = "tlv320aic23-codec.0-001a",
.platform_name = "ep93xx-i2s",
- .init = snappercl15_tlv320aic23_init,
.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_IF |
SND_SOC_DAIFMT_CBS_CFS,
.ops = &snappercl15_ops,
@@ -95,6 +82,11 @@ static struct snd_soc_card snd_soc_snappercl15 = {
.owner = THIS_MODULE,
.dai_link = &snappercl15_dai,
.num_links = 1,
+
+ .dapm_widgets = tlv320aic23_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(tlv320aic23_dapm_widgets),
+ .dapm_routes = audio_map,
+ .num_dapm_routes = ARRAY_SIZE(audio_map),
};
static int snappercl15_probe(struct platform_device *pdev)
diff --git a/sound/soc/codecs/88pm860x-codec.c b/sound/soc/codecs/88pm860x-codec.c
index 647a72c..b07e171 100644
--- a/sound/soc/codecs/88pm860x-codec.c
+++ b/sound/soc/codecs/88pm860x-codec.c
@@ -448,38 +448,38 @@ static const char *pm860x_opamp_texts[] = {"-50%", "-25%", "0%", "75%"};
static const char *pm860x_pa_texts[] = {"-33%", "0%", "33%", "66%"};
-static const struct soc_enum pm860x_hs1_opamp_enum =
- SOC_ENUM_SINGLE(PM860X_HS1_CTRL, 5, 4, pm860x_opamp_texts);
+static SOC_ENUM_SINGLE_DECL(pm860x_hs1_opamp_enum,
+ PM860X_HS1_CTRL, 5, pm860x_opamp_texts);
-static const struct soc_enum pm860x_hs2_opamp_enum =
- SOC_ENUM_SINGLE(PM860X_HS2_CTRL, 5, 4, pm860x_opamp_texts);
+static SOC_ENUM_SINGLE_DECL(pm860x_hs2_opamp_enum,
+ PM860X_HS2_CTRL, 5, pm860x_opamp_texts);
-static const struct soc_enum pm860x_hs1_pa_enum =
- SOC_ENUM_SINGLE(PM860X_HS1_CTRL, 3, 4, pm860x_pa_texts);
+static SOC_ENUM_SINGLE_DECL(pm860x_hs1_pa_enum,
+ PM860X_HS1_CTRL, 3, pm860x_pa_texts);
-static const struct soc_enum pm860x_hs2_pa_enum =
- SOC_ENUM_SINGLE(PM860X_HS2_CTRL, 3, 4, pm860x_pa_texts);
+static SOC_ENUM_SINGLE_DECL(pm860x_hs2_pa_enum,
+ PM860X_HS2_CTRL, 3, pm860x_pa_texts);
-static const struct soc_enum pm860x_lo1_opamp_enum =
- SOC_ENUM_SINGLE(PM860X_LO1_CTRL, 5, 4, pm860x_opamp_texts);
+static SOC_ENUM_SINGLE_DECL(pm860x_lo1_opamp_enum,
+ PM860X_LO1_CTRL, 5, pm860x_opamp_texts);
-static const struct soc_enum pm860x_lo2_opamp_enum =
- SOC_ENUM_SINGLE(PM860X_LO2_CTRL, 5, 4, pm860x_opamp_texts);
+static SOC_ENUM_SINGLE_DECL(pm860x_lo2_opamp_enum,
+ PM860X_LO2_CTRL, 5, pm860x_opamp_texts);
-static const struct soc_enum pm860x_lo1_pa_enum =
- SOC_ENUM_SINGLE(PM860X_LO1_CTRL, 3, 4, pm860x_pa_texts);
+static SOC_ENUM_SINGLE_DECL(pm860x_lo1_pa_enum,
+ PM860X_LO1_CTRL, 3, pm860x_pa_texts);
-static const struct soc_enum pm860x_lo2_pa_enum =
- SOC_ENUM_SINGLE(PM860X_LO2_CTRL, 3, 4, pm860x_pa_texts);
+static SOC_ENUM_SINGLE_DECL(pm860x_lo2_pa_enum,
+ PM860X_LO2_CTRL, 3, pm860x_pa_texts);
-static const struct soc_enum pm860x_spk_pa_enum =
- SOC_ENUM_SINGLE(PM860X_EAR_CTRL_1, 5, 4, pm860x_pa_texts);
+static SOC_ENUM_SINGLE_DECL(pm860x_spk_pa_enum,
+ PM860X_EAR_CTRL_1, 5, pm860x_pa_texts);
-static const struct soc_enum pm860x_ear_pa_enum =
- SOC_ENUM_SINGLE(PM860X_EAR_CTRL_2, 0, 4, pm860x_pa_texts);
+static SOC_ENUM_SINGLE_DECL(pm860x_ear_pa_enum,
+ PM860X_EAR_CTRL_2, 0, pm860x_pa_texts);
-static const struct soc_enum pm860x_spk_ear_opamp_enum =
- SOC_ENUM_SINGLE(PM860X_EAR_CTRL_1, 3, 4, pm860x_opamp_texts);
+static SOC_ENUM_SINGLE_DECL(pm860x_spk_ear_opamp_enum,
+ PM860X_EAR_CTRL_1, 3, pm860x_opamp_texts);
static const struct snd_kcontrol_new pm860x_snd_controls[] = {
SOC_DOUBLE_R_TLV("ADC Capture Volume", PM860X_ADC_ANA_2,
@@ -561,8 +561,8 @@ static const char *aif1_text[] = {
"PCM L", "PCM R",
};
-static const struct soc_enum aif1_enum =
- SOC_ENUM_SINGLE(PM860X_PCM_IFACE_3, 6, 2, aif1_text);
+static SOC_ENUM_SINGLE_DECL(aif1_enum,
+ PM860X_PCM_IFACE_3, 6, aif1_text);
static const struct snd_kcontrol_new aif1_mux =
SOC_DAPM_ENUM("PCM Mux", aif1_enum);
@@ -572,8 +572,8 @@ static const char *i2s_din_text[] = {
"DIN", "DIN1",
};
-static const struct soc_enum i2s_din_enum =
- SOC_ENUM_SINGLE(PM860X_I2S_IFACE_3, 1, 2, i2s_din_text);
+static SOC_ENUM_SINGLE_DECL(i2s_din_enum,
+ PM860X_I2S_IFACE_3, 1, i2s_din_text);
static const struct snd_kcontrol_new i2s_din_mux =
SOC_DAPM_ENUM("I2S DIN Mux", i2s_din_enum);
@@ -583,8 +583,8 @@ static const char *i2s_mic_text[] = {
"Ex PA", "ADC",
};
-static const struct soc_enum i2s_mic_enum =
- SOC_ENUM_SINGLE(PM860X_I2S_IFACE_3, 4, 2, i2s_mic_text);
+static SOC_ENUM_SINGLE_DECL(i2s_mic_enum,
+ PM860X_I2S_IFACE_3, 4, i2s_mic_text);
static const struct snd_kcontrol_new i2s_mic_mux =
SOC_DAPM_ENUM("I2S Mic Mux", i2s_mic_enum);
@@ -594,8 +594,8 @@ static const char *adcl_text[] = {
"ADCR", "ADCL",
};
-static const struct soc_enum adcl_enum =
- SOC_ENUM_SINGLE(PM860X_PCM_IFACE_3, 4, 2, adcl_text);
+static SOC_ENUM_SINGLE_DECL(adcl_enum,
+ PM860X_PCM_IFACE_3, 4, adcl_text);
static const struct snd_kcontrol_new adcl_mux =
SOC_DAPM_ENUM("ADC Left Mux", adcl_enum);
@@ -605,8 +605,8 @@ static const char *adcr_text[] = {
"ADCL", "ADCR",
};
-static const struct soc_enum adcr_enum =
- SOC_ENUM_SINGLE(PM860X_PCM_IFACE_3, 2, 2, adcr_text);
+static SOC_ENUM_SINGLE_DECL(adcr_enum,
+ PM860X_PCM_IFACE_3, 2, adcr_text);
static const struct snd_kcontrol_new adcr_mux =
SOC_DAPM_ENUM("ADC Right Mux", adcr_enum);
@@ -616,8 +616,8 @@ static const char *adcr_ec_text[] = {
"ADCR", "EC",
};
-static const struct soc_enum adcr_ec_enum =
- SOC_ENUM_SINGLE(PM860X_ADC_EN_2, 3, 2, adcr_ec_text);
+static SOC_ENUM_SINGLE_DECL(adcr_ec_enum,
+ PM860X_ADC_EN_2, 3, adcr_ec_text);
static const struct snd_kcontrol_new adcr_ec_mux =
SOC_DAPM_ENUM("ADCR EC Mux", adcr_ec_enum);
@@ -627,8 +627,8 @@ static const char *ec_text[] = {
"Left", "Right", "Left + Right",
};
-static const struct soc_enum ec_enum =
- SOC_ENUM_SINGLE(PM860X_EC_PATH, 1, 3, ec_text);
+static SOC_ENUM_SINGLE_DECL(ec_enum,
+ PM860X_EC_PATH, 1, ec_text);
static const struct snd_kcontrol_new ec_mux =
SOC_DAPM_ENUM("EC Mux", ec_enum);
@@ -638,36 +638,36 @@ static const char *dac_text[] = {
};
/* DAC Headset 1 Mux / Mux10 */
-static const struct soc_enum dac_hs1_enum =
- SOC_ENUM_SINGLE(PM860X_ANA_INPUT_SEL_1, 0, 4, dac_text);
+static SOC_ENUM_SINGLE_DECL(dac_hs1_enum,
+ PM860X_ANA_INPUT_SEL_1, 0, dac_text);
static const struct snd_kcontrol_new dac_hs1_mux =
SOC_DAPM_ENUM("DAC HS1 Mux", dac_hs1_enum);
/* DAC Headset 2 Mux / Mux11 */
-static const struct soc_enum dac_hs2_enum =
- SOC_ENUM_SINGLE(PM860X_ANA_INPUT_SEL_1, 2, 4, dac_text);
+static SOC_ENUM_SINGLE_DECL(dac_hs2_enum,
+ PM860X_ANA_INPUT_SEL_1, 2, dac_text);
static const struct snd_kcontrol_new dac_hs2_mux =
SOC_DAPM_ENUM("DAC HS2 Mux", dac_hs2_enum);
/* DAC Lineout 1 Mux / Mux12 */
-static const struct soc_enum dac_lo1_enum =
- SOC_ENUM_SINGLE(PM860X_ANA_INPUT_SEL_1, 4, 4, dac_text);
+static SOC_ENUM_SINGLE_DECL(dac_lo1_enum,
+ PM860X_ANA_INPUT_SEL_1, 4, dac_text);
static const struct snd_kcontrol_new dac_lo1_mux =
SOC_DAPM_ENUM("DAC LO1 Mux", dac_lo1_enum);
/* DAC Lineout 2 Mux / Mux13 */
-static const struct soc_enum dac_lo2_enum =
- SOC_ENUM_SINGLE(PM860X_ANA_INPUT_SEL_1, 6, 4, dac_text);
+static SOC_ENUM_SINGLE_DECL(dac_lo2_enum,
+ PM860X_ANA_INPUT_SEL_1, 6, dac_text);
static const struct snd_kcontrol_new dac_lo2_mux =
SOC_DAPM_ENUM("DAC LO2 Mux", dac_lo2_enum);
/* DAC Spearker Earphone Mux / Mux14 */
-static const struct soc_enum dac_spk_ear_enum =
- SOC_ENUM_SINGLE(PM860X_ANA_INPUT_SEL_2, 0, 4, dac_text);
+static SOC_ENUM_SINGLE_DECL(dac_spk_ear_enum,
+ PM860X_ANA_INPUT_SEL_2, 0, dac_text);
static const struct snd_kcontrol_new dac_spk_ear_mux =
SOC_DAPM_ENUM("DAC SP Mux", dac_spk_ear_enum);
@@ -677,29 +677,29 @@ static const char *in_text[] = {
"Digital", "Analog",
};
-static const struct soc_enum hs1_enum =
- SOC_ENUM_SINGLE(PM860X_ANA_TO_ANA, 0, 2, in_text);
+static SOC_ENUM_SINGLE_DECL(hs1_enum,
+ PM860X_ANA_TO_ANA, 0, in_text);
static const struct snd_kcontrol_new hs1_mux =
SOC_DAPM_ENUM("Headset1 Mux", hs1_enum);
/* Headset 2 Mux / Mux16 */
-static const struct soc_enum hs2_enum =
- SOC_ENUM_SINGLE(PM860X_ANA_TO_ANA, 1, 2, in_text);
+static SOC_ENUM_SINGLE_DECL(hs2_enum,
+ PM860X_ANA_TO_ANA, 1, in_text);
static const struct snd_kcontrol_new hs2_mux =
SOC_DAPM_ENUM("Headset2 Mux", hs2_enum);
/* Lineout 1 Mux / Mux17 */
-static const struct soc_enum lo1_enum =
- SOC_ENUM_SINGLE(PM860X_ANA_TO_ANA, 2, 2, in_text);
+static SOC_ENUM_SINGLE_DECL(lo1_enum,
+ PM860X_ANA_TO_ANA, 2, in_text);
static const struct snd_kcontrol_new lo1_mux =
SOC_DAPM_ENUM("Lineout1 Mux", lo1_enum);
/* Lineout 2 Mux / Mux18 */
-static const struct soc_enum lo2_enum =
- SOC_ENUM_SINGLE(PM860X_ANA_TO_ANA, 3, 2, in_text);
+static SOC_ENUM_SINGLE_DECL(lo2_enum,
+ PM860X_ANA_TO_ANA, 3, in_text);
static const struct snd_kcontrol_new lo2_mux =
SOC_DAPM_ENUM("Lineout2 Mux", lo2_enum);
@@ -709,8 +709,8 @@ static const char *spk_text[] = {
"Earpiece", "Speaker",
};
-static const struct soc_enum spk_enum =
- SOC_ENUM_SINGLE(PM860X_ANA_TO_ANA, 6, 2, spk_text);
+static SOC_ENUM_SINGLE_DECL(spk_enum,
+ PM860X_ANA_TO_ANA, 6, spk_text);
static const struct snd_kcontrol_new spk_demux =
SOC_DAPM_ENUM("Speaker Earpiece Demux", spk_enum);
@@ -720,8 +720,8 @@ static const char *mic_text[] = {
"Mic 1", "Mic 2",
};
-static const struct soc_enum mic_enum =
- SOC_ENUM_SINGLE(PM860X_ADC_ANA_4, 4, 2, mic_text);
+static SOC_ENUM_SINGLE_DECL(mic_enum,
+ PM860X_ADC_ANA_4, 4, mic_text);
static const struct snd_kcontrol_new mic_mux =
SOC_DAPM_ENUM("MIC Mux", mic_enum);
@@ -1327,8 +1327,7 @@ static int pm860x_probe(struct snd_soc_codec *codec)
pm860x->codec = codec;
- codec->control_data = pm860x->regmap;
- ret = snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP);
+ ret = snd_soc_codec_set_cache_io(codec, pm860x->regmap);
if (ret)
return ret;
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 983d087a..f0e8401 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -8,6 +8,8 @@ config SND_SOC_I2C_AND_SPI
default y if I2C=y
default y if SPI_MASTER=y
+menu "CODEC drivers"
+
config SND_SOC_ALL_CODECS
tristate "Build all ASoC CODEC drivers"
depends on COMPILE_TEST
@@ -16,15 +18,20 @@ config SND_SOC_ALL_CODECS
select SND_SOC_AB8500_CODEC if ABX500_CORE
select SND_SOC_AC97_CODEC if SND_SOC_AC97_BUS
select SND_SOC_AD1836 if SPI_MASTER
- select SND_SOC_AD193X if SND_SOC_I2C_AND_SPI
+ select SND_SOC_AD193X_SPI if SPI_MASTER
+ select SND_SOC_AD193X_I2C if I2C
select SND_SOC_AD1980 if SND_SOC_AC97_BUS
select SND_SOC_AD73311
select SND_SOC_ADAU1373 if I2C
- select SND_SOC_ADAV80X if SND_SOC_I2C_AND_SPI
+ select SND_SOC_ADAV801 if SPI_MASTER
+ select SND_SOC_ADAV803 if I2C
+ select SND_SOC_ADAU1977_SPI if SPI_MASTER
+ select SND_SOC_ADAU1977_I2C if I2C
select SND_SOC_ADAU1701 if I2C
select SND_SOC_ADS117X
select SND_SOC_AK4104 if SPI_MASTER
select SND_SOC_AK4535 if I2C
+ select SND_SOC_AK4554
select SND_SOC_AK4641 if I2C
select SND_SOC_AK4642 if I2C
select SND_SOC_AK4671 if I2C
@@ -37,6 +44,7 @@ config SND_SOC_ALL_CODECS
select SND_SOC_CS42L73 if I2C
select SND_SOC_CS4270 if I2C
select SND_SOC_CS4271 if SND_SOC_I2C_AND_SPI
+ select SND_SOC_CS42XX8_I2C if I2C
select SND_SOC_CX20442 if TTY
select SND_SOC_DA7210 if I2C
select SND_SOC_DA7213 if I2C
@@ -59,20 +67,26 @@ config SND_SOC_ALL_CODECS
select SND_SOC_PCM1681 if I2C
select SND_SOC_PCM1792A if SPI_MASTER
select SND_SOC_PCM3008
+ select SND_SOC_PCM512x_I2C if I2C
+ select SND_SOC_PCM512x_SPI if SPI_MASTER
select SND_SOC_RT5631 if I2C
select SND_SOC_RT5640 if I2C
select SND_SOC_SGTL5000 if I2C
select SND_SOC_SI476X if MFD_SI476X_CORE
+ select SND_SOC_SIRF_AUDIO_CODEC
select SND_SOC_SN95031 if INTEL_SCU_IPC
select SND_SOC_SPDIF
select SND_SOC_SSM2518 if I2C
- select SND_SOC_SSM2602 if SND_SOC_I2C_AND_SPI
+ select SND_SOC_SSM2602_SPI if SPI_MASTER
+ select SND_SOC_SSM2602_I2C if I2C
select SND_SOC_STA32X if I2C
select SND_SOC_STA529 if I2C
select SND_SOC_STAC9766 if SND_SOC_AC97_BUS
select SND_SOC_TAS5086 if I2C
- select SND_SOC_TLV320AIC23 if I2C
+ select SND_SOC_TLV320AIC23_I2C if I2C
+ select SND_SOC_TLV320AIC23_SPI if SPI_MASTER
select SND_SOC_TLV320AIC26 if SPI_MASTER
+ select SND_SOC_TLV320AIC31XX if I2C
select SND_SOC_TLV320AIC32X4 if I2C
select SND_SOC_TLV320AIC3X if I2C
select SND_SOC_TPA6130A2 if I2C
@@ -182,6 +196,14 @@ config SND_SOC_AD1836
config SND_SOC_AD193X
tristate
+config SND_SOC_AD193X_SPI
+ tristate
+ select SND_SOC_AD193X
+
+config SND_SOC_AD193X_I2C
+ tristate
+ select SND_SOC_AD193X
+
config SND_SOC_AD1980
tristate
@@ -189,41 +211,66 @@ config SND_SOC_AD73311
tristate
config SND_SOC_ADAU1701
+ tristate "Analog Devices ADAU1701 CODEC"
+ depends on I2C
select SND_SOC_SIGMADSP
- tristate
config SND_SOC_ADAU1373
tristate
+config SND_SOC_ADAU1977
+ tristate
+
+config SND_SOC_ADAU1977_SPI
+ tristate
+ select SND_SOC_ADAU1977
+ select REGMAP_SPI
+
+config SND_SOC_ADAU1977_I2C
+ tristate
+ select SND_SOC_ADAU1977
+ select REGMAP_I2C
+
config SND_SOC_ADAV80X
tristate
+config SND_SOC_ADAV801
+ tristate
+ select SND_SOC_ADAV80X
+
+config SND_SOC_ADAV803
+ tristate
+ select SND_SOC_ADAV80X
+
config SND_SOC_ADS117X
tristate
config SND_SOC_AK4104
- tristate
+ tristate "AKM AK4104 CODEC"
+ depends on SPI_MASTER
config SND_SOC_AK4535
tristate
config SND_SOC_AK4554
- tristate
+ tristate "AKM AK4554 CODEC"
config SND_SOC_AK4641
tristate
config SND_SOC_AK4642
- tristate
+ tristate "AKM AK4642 CODEC"
+ depends on I2C
config SND_SOC_AK4671
tristate
config SND_SOC_AK5386
- tristate
+ tristate "AKM AK5638 CODEC"
config SND_SOC_ALC5623
tristate
+
config SND_SOC_ALC5632
tristate
@@ -234,14 +281,17 @@ config SND_SOC_CS42L51
tristate
config SND_SOC_CS42L52
- tristate
+ tristate "Cirrus Logic CS42L52 CODEC"
+ depends on I2C
config SND_SOC_CS42L73
- tristate
+ tristate "Cirrus Logic CS42L73 CODEC"
+ depends on I2C
# Cirrus Logic CS4270 Codec
config SND_SOC_CS4270
- tristate
+ tristate "Cirrus Logic CS4270 CODEC"
+ depends on I2C
# Cirrus Logic CS4270 Codec VD = 3.3V Errata
# Select if you are affected by the errata where the part will not function
@@ -252,8 +302,18 @@ config SND_SOC_CS4270_VD33_ERRATA
depends on SND_SOC_CS4270
config SND_SOC_CS4271
+ tristate "Cirrus Logic CS4271 CODEC"
+ depends on SND_SOC_I2C_AND_SPI
+
+config SND_SOC_CS42XX8
tristate
+config SND_SOC_CS42XX8_I2C
+ tristate "Cirrus Logic CS42448/CS42888 CODEC (I2C)"
+ depends on I2C
+ select SND_SOC_CS42XX8
+ select REGMAP_I2C
+
config SND_SOC_CX20442
tristate
depends on TTY
@@ -283,6 +343,9 @@ config SND_SOC_BT_SCO
config SND_SOC_DMIC
tristate
+config SND_SOC_HDMI_CODEC
+ tristate "HDMI stub CODEC"
+
config SND_SOC_ISABELLE
tristate
@@ -301,18 +364,32 @@ config SND_SOC_MAX98095
config SND_SOC_MAX9850
tristate
-config SND_SOC_HDMI_CODEC
- tristate
-
config SND_SOC_PCM1681
- tristate
+ tristate "Texas Instruments PCM1681 CODEC"
+ depends on I2C
config SND_SOC_PCM1792A
- tristate
+ tristate "Texas Instruments PCM1792A CODEC"
+ depends on SPI_MASTER
config SND_SOC_PCM3008
tristate
+config SND_SOC_PCM512x
+ tristate
+
+config SND_SOC_PCM512x_I2C
+ tristate "Texas Instruments PCM512x CODECs - I2C"
+ depends on I2C
+ select SND_SOC_PCM512x
+ select REGMAP_I2C
+
+config SND_SOC_PCM512x_SPI
+ tristate "Texas Instruments PCM512x CODECs - SPI"
+ depends on SPI_MASTER
+ select SND_SOC_PCM512x
+ select REGMAP_SPI
+
config SND_SOC_RT5631
tristate
@@ -321,7 +398,8 @@ config SND_SOC_RT5640
#Freescale sgtl5000 codec
config SND_SOC_SGTL5000
- tristate
+ tristate "Freescale SGTL5000 CODEC"
+ depends on I2C
config SND_SOC_SI476X
tristate
@@ -330,11 +408,15 @@ config SND_SOC_SIGMADSP
tristate
select CRC32
+config SND_SOC_SIRF_AUDIO_CODEC
+ tristate "SiRF SoC internal audio codec"
+ select REGMAP_MMIO
+
config SND_SOC_SN95031
tristate
config SND_SOC_SPDIF
- tristate
+ tristate "S/PDIF CODEC"
config SND_SOC_SSM2518
tristate
@@ -342,6 +424,14 @@ config SND_SOC_SSM2518
config SND_SOC_SSM2602
tristate
+config SND_SOC_SSM2602_SPI
+ select SND_SOC_SSM2602
+ tristate
+
+config SND_SOC_SSM2602_I2C
+ select SND_SOC_SSM2602
+ tristate
+
config SND_SOC_STA32X
tristate
@@ -352,20 +442,33 @@ config SND_SOC_STAC9766
tristate
config SND_SOC_TAS5086
- tristate
+ tristate "Texas Instruments TAS5086 speaker amplifier"
+ depends on I2C
config SND_SOC_TLV320AIC23
tristate
+config SND_SOC_TLV320AIC23_I2C
+ tristate
+ select SND_SOC_TLV320AIC23
+
+config SND_SOC_TLV320AIC23_SPI
+ tristate
+ select SND_SOC_TLV320AIC23
+
config SND_SOC_TLV320AIC26
tristate
depends on SPI
+config SND_SOC_TLV320AIC31XX
+ tristate
+
config SND_SOC_TLV320AIC32X4
tristate
config SND_SOC_TLV320AIC3X
- tristate
+ tristate "Texas Instruments TLV320AIC3x CODECs"
+ depends on I2C
config SND_SOC_TLV320DAC33
tristate
@@ -414,55 +517,69 @@ config SND_SOC_WM8400
tristate
config SND_SOC_WM8510
- tristate
+ tristate "Wolfson Microelectronics WM8510 CODEC"
+ depends on SND_SOC_I2C_AND_SPI
config SND_SOC_WM8523
- tristate
+ tristate "Wolfson Microelectronics WM8523 DAC"
+ depends on I2C
config SND_SOC_WM8580
- tristate
+ tristate "Wolfson Microelectronics WM8523 CODEC"
+ depends on I2C
config SND_SOC_WM8711
- tristate
+ tristate "Wolfson Microelectronics WM8711 CODEC"
+ depends on SND_SOC_I2C_AND_SPI
config SND_SOC_WM8727
tristate
config SND_SOC_WM8728
- tristate
+ tristate "Wolfson Microelectronics WM8728 DAC"
+ depends on SND_SOC_I2C_AND_SPI
config SND_SOC_WM8731
- tristate
+ tristate "Wolfson Microelectronics WM8731 CODEC"
+ depends on SND_SOC_I2C_AND_SPI
config SND_SOC_WM8737
- tristate
+ tristate "Wolfson Microelectronics WM8737 ADC"
+ depends on SND_SOC_I2C_AND_SPI
config SND_SOC_WM8741
- tristate
+ tristate "Wolfson Microelectronics WM8737 DAC"
+ depends on SND_SOC_I2C_AND_SPI
config SND_SOC_WM8750
- tristate
+ tristate "Wolfson Microelectronics WM8750 CODEC"
+ depends on SND_SOC_I2C_AND_SPI
config SND_SOC_WM8753
- tristate
+ tristate "Wolfson Microelectronics WM8753 CODEC"
+ depends on SND_SOC_I2C_AND_SPI
config SND_SOC_WM8770
- tristate
+ tristate "Wolfson Microelectronics WM8770 CODEC"
+ depends on SPI_MASTER
config SND_SOC_WM8776
- tristate
+ tristate "Wolfson Microelectronics WM8776 CODEC"
+ depends on SND_SOC_I2C_AND_SPI
config SND_SOC_WM8782
tristate
config SND_SOC_WM8804
- tristate
+ tristate "Wolfson Microelectronics WM8804 S/PDIF transceiver"
+ depends on SND_SOC_I2C_AND_SPI
config SND_SOC_WM8900
tristate
config SND_SOC_WM8903
- tristate
+ tristate "Wolfson Microelectronics WM8903 CODEC"
+ depends on I2C
config SND_SOC_WM8904
tristate
@@ -480,7 +597,8 @@ config SND_SOC_WM8961
tristate
config SND_SOC_WM8962
- tristate
+ tristate "Wolfson Microelectronics WM8962 CODEC"
+ depends on I2C
config SND_SOC_WM8971
tristate
@@ -553,4 +671,7 @@ config SND_SOC_ML26124
tristate
config SND_SOC_TPA6130A2
- tristate
+ tristate "Texas Instruments TPA6130A2 headphone amplifier"
+ depends on I2C
+
+endmenu
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index bc12676..3c4d275 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -3,11 +3,18 @@ snd-soc-ab8500-codec-objs := ab8500-codec.o
snd-soc-ac97-objs := ac97.o
snd-soc-ad1836-objs := ad1836.o
snd-soc-ad193x-objs := ad193x.o
+snd-soc-ad193x-spi-objs := ad193x-spi.o
+snd-soc-ad193x-i2c-objs := ad193x-i2c.o
snd-soc-ad1980-objs := ad1980.o
snd-soc-ad73311-objs := ad73311.o
snd-soc-adau1701-objs := adau1701.o
snd-soc-adau1373-objs := adau1373.o
+snd-soc-adau1977-objs := adau1977.o
+snd-soc-adau1977-spi-objs := adau1977-spi.o
+snd-soc-adau1977-i2c-objs := adau1977-i2c.o
snd-soc-adav80x-objs := adav80x.o
+snd-soc-adav801-objs := adav801.o
+snd-soc-adav803-objs := adav803.o
snd-soc-ads117x-objs := ads117x.o
snd-soc-ak4104-objs := ak4104.o
snd-soc-ak4535-objs := ak4535.o
@@ -23,6 +30,8 @@ snd-soc-cs42l52-objs := cs42l52.o
snd-soc-cs42l73-objs := cs42l73.o
snd-soc-cs4270-objs := cs4270.o
snd-soc-cs4271-objs := cs4271.o
+snd-soc-cs42xx8-objs := cs42xx8.o
+snd-soc-cs42xx8-i2c-objs := cs42xx8-i2c.o
snd-soc-cx20442-objs := cx20442.o
snd-soc-da7210-objs := da7210.o
snd-soc-da7213-objs := da7213.o
@@ -46,6 +55,9 @@ snd-soc-hdmi-codec-objs := hdmi.o
snd-soc-pcm1681-objs := pcm1681.o
snd-soc-pcm1792a-codec-objs := pcm1792a.o
snd-soc-pcm3008-objs := pcm3008.o
+snd-soc-pcm512x-objs := pcm512x.o
+snd-soc-pcm512x-i2c-objs := pcm512x-i2c.o
+snd-soc-pcm512x-spi-objs := pcm512x-spi.o
snd-soc-rt5631-objs := rt5631.o
snd-soc-rt5640-objs := rt5640.o
snd-soc-sgtl5000-objs := sgtl5000.o
@@ -53,19 +65,25 @@ snd-soc-alc5623-objs := alc5623.o
snd-soc-alc5632-objs := alc5632.o
snd-soc-sigmadsp-objs := sigmadsp.o
snd-soc-si476x-objs := si476x.o
+snd-soc-sirf-audio-codec-objs := sirf-audio-codec.o
snd-soc-sn95031-objs := sn95031.o
snd-soc-spdif-tx-objs := spdif_transmitter.o
snd-soc-spdif-rx-objs := spdif_receiver.o
snd-soc-ssm2518-objs := ssm2518.o
snd-soc-ssm2602-objs := ssm2602.o
+snd-soc-ssm2602-spi-objs := ssm2602-spi.o
+snd-soc-ssm2602-i2c-objs := ssm2602-i2c.o
snd-soc-sta32x-objs := sta32x.o
snd-soc-sta529-objs := sta529.o
snd-soc-stac9766-objs := stac9766.o
snd-soc-tas5086-objs := tas5086.o
snd-soc-tlv320aic23-objs := tlv320aic23.o
+snd-soc-tlv320aic23-i2c-objs := tlv320aic23-i2c.o
+snd-soc-tlv320aic23-spi-objs := tlv320aic23-spi.o
snd-soc-tlv320aic26-objs := tlv320aic26.o
-snd-soc-tlv320aic3x-objs := tlv320aic3x.o
+snd-soc-tlv320aic31xx-objs := tlv320aic31xx.o
snd-soc-tlv320aic32x4-objs := tlv320aic32x4.o
+snd-soc-tlv320aic3x-objs := tlv320aic3x.o
snd-soc-tlv320dac33-objs := tlv320dac33.o
snd-soc-twl4030-objs := twl4030.o
snd-soc-twl6040-objs := twl6040.o
@@ -134,11 +152,18 @@ obj-$(CONFIG_SND_SOC_AB8500_CODEC) += snd-soc-ab8500-codec.o
obj-$(CONFIG_SND_SOC_AC97_CODEC) += snd-soc-ac97.o
obj-$(CONFIG_SND_SOC_AD1836) += snd-soc-ad1836.o
obj-$(CONFIG_SND_SOC_AD193X) += snd-soc-ad193x.o
+obj-$(CONFIG_SND_SOC_AD193X_SPI) += snd-soc-ad193x-spi.o
+obj-$(CONFIG_SND_SOC_AD193X_I2C) += snd-soc-ad193x-i2c.o
obj-$(CONFIG_SND_SOC_AD1980) += snd-soc-ad1980.o
obj-$(CONFIG_SND_SOC_AD73311) += snd-soc-ad73311.o
obj-$(CONFIG_SND_SOC_ADAU1373) += snd-soc-adau1373.o
+obj-$(CONFIG_SND_SOC_ADAU1977) += snd-soc-adau1977.o
+obj-$(CONFIG_SND_SOC_ADAU1977_SPI) += snd-soc-adau1977-spi.o
+obj-$(CONFIG_SND_SOC_ADAU1977_I2C) += snd-soc-adau1977-i2c.o
obj-$(CONFIG_SND_SOC_ADAU1701) += snd-soc-adau1701.o
obj-$(CONFIG_SND_SOC_ADAV80X) += snd-soc-adav80x.o
+obj-$(CONFIG_SND_SOC_ADAV801) += snd-soc-adav801.o
+obj-$(CONFIG_SND_SOC_ADAV803) += snd-soc-adav803.o
obj-$(CONFIG_SND_SOC_ADS117X) += snd-soc-ads117x.o
obj-$(CONFIG_SND_SOC_AK4104) += snd-soc-ak4104.o
obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o
@@ -156,6 +181,8 @@ obj-$(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-$(CONFIG_SND_SOC_CS4271) += snd-soc-cs4271.o
+obj-$(CONFIG_SND_SOC_CS42XX8) += snd-soc-cs42xx8.o
+obj-$(CONFIG_SND_SOC_CS42XX8_I2C) += snd-soc-cs42xx8-i2c.o
obj-$(CONFIG_SND_SOC_CX20442) += snd-soc-cx20442.o
obj-$(CONFIG_SND_SOC_DA7210) += snd-soc-da7210.o
obj-$(CONFIG_SND_SOC_DA7213) += snd-soc-da7213.o
@@ -179,6 +206,9 @@ obj-$(CONFIG_SND_SOC_HDMI_CODEC) += snd-soc-hdmi-codec.o
obj-$(CONFIG_SND_SOC_PCM1681) += snd-soc-pcm1681.o
obj-$(CONFIG_SND_SOC_PCM1792A) += snd-soc-pcm1792a-codec.o
obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o
+obj-$(CONFIG_SND_SOC_PCM512x) += snd-soc-pcm512x.o
+obj-$(CONFIG_SND_SOC_PCM512x_I2C) += snd-soc-pcm512x-i2c.o
+obj-$(CONFIG_SND_SOC_PCM512x_SPI) += snd-soc-pcm512x-spi.o
obj-$(CONFIG_SND_SOC_RT5631) += snd-soc-rt5631.o
obj-$(CONFIG_SND_SOC_RT5640) += snd-soc-rt5640.o
obj-$(CONFIG_SND_SOC_SGTL5000) += snd-soc-sgtl5000.o
@@ -188,14 +218,19 @@ obj-$(CONFIG_SND_SOC_SN95031) +=snd-soc-sn95031.o
obj-$(CONFIG_SND_SOC_SPDIF) += snd-soc-spdif-rx.o snd-soc-spdif-tx.o
obj-$(CONFIG_SND_SOC_SSM2518) += snd-soc-ssm2518.o
obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o
+obj-$(CONFIG_SND_SOC_SSM2602_SPI) += snd-soc-ssm2602-spi.o
+obj-$(CONFIG_SND_SOC_SSM2602_I2C) += snd-soc-ssm2602-i2c.o
obj-$(CONFIG_SND_SOC_STA32X) += snd-soc-sta32x.o
obj-$(CONFIG_SND_SOC_STA529) += snd-soc-sta529.o
obj-$(CONFIG_SND_SOC_STAC9766) += snd-soc-stac9766.o
obj-$(CONFIG_SND_SOC_TAS5086) += snd-soc-tas5086.o
obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o
+obj-$(CONFIG_SND_SOC_TLV320AIC23_I2C) += snd-soc-tlv320aic23-i2c.o
+obj-$(CONFIG_SND_SOC_TLV320AIC23_SPI) += snd-soc-tlv320aic23-spi.o
obj-$(CONFIG_SND_SOC_TLV320AIC26) += snd-soc-tlv320aic26.o
-obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o
+obj-$(CONFIG_SND_SOC_TLV320AIC31XX) += snd-soc-tlv320aic31xx.o
obj-$(CONFIG_SND_SOC_TLV320AIC32X4) += snd-soc-tlv320aic32x4.o
+obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o
obj-$(CONFIG_SND_SOC_TLV320DAC33) += snd-soc-tlv320dac33.o
obj-$(CONFIG_SND_SOC_TWL4030) += snd-soc-twl4030.o
obj-$(CONFIG_SND_SOC_TWL6040) += snd-soc-twl6040.o
diff --git a/sound/soc/codecs/ad1836.c b/sound/soc/codecs/ad1836.c
index 77f4598..685998d 100644
--- a/sound/soc/codecs/ad1836.c
+++ b/sound/soc/codecs/ad1836.c
@@ -40,8 +40,8 @@ struct ad1836_priv {
*/
static const char *ad1836_deemp[] = {"None", "44.1kHz", "32kHz", "48kHz"};
-static const struct soc_enum ad1836_deemp_enum =
- SOC_ENUM_SINGLE(AD1836_DAC_CTRL1, 8, 4, ad1836_deemp);
+static SOC_ENUM_SINGLE_DECL(ad1836_deemp_enum,
+ AD1836_DAC_CTRL1, 8, ad1836_deemp);
#define AD1836_DAC_VOLUME(x) \
SOC_DOUBLE_R("DAC" #x " Playback Volume", AD1836_DAC_L_VOL(x), \
diff --git a/sound/soc/codecs/ad193x-i2c.c b/sound/soc/codecs/ad193x-i2c.c
new file mode 100644
index 0000000..df3a1a4
--- /dev/null
+++ b/sound/soc/codecs/ad193x-i2c.c
@@ -0,0 +1,54 @@
+/*
+ * AD1936/AD1937 audio driver
+ *
+ * Copyright 2014 Analog Devices Inc.
+ *
+ * Licensed under the GPL-2.
+ */
+
+#include <linux/module.h>
+#include <linux/i2c.h>
+#include <linux/regmap.h>
+
+#include <sound/soc.h>
+
+#include "ad193x.h"
+
+static const struct i2c_device_id ad193x_id[] = {
+ { "ad1936", 0 },
+ { "ad1937", 0 },
+ { }
+};
+MODULE_DEVICE_TABLE(i2c, ad193x_id);
+
+static int ad193x_i2c_probe(struct i2c_client *client,
+ const struct i2c_device_id *id)
+{
+ struct regmap_config config;
+
+ config = ad193x_regmap_config;
+ config.val_bits = 8;
+ config.reg_bits = 8;
+
+ return ad193x_probe(&client->dev, devm_regmap_init_i2c(client, &config));
+}
+
+static int ad193x_i2c_remove(struct i2c_client *client)
+{
+ snd_soc_unregister_codec(&client->dev);
+ return 0;
+}
+
+static struct i2c_driver ad193x_i2c_driver = {
+ .driver = {
+ .name = "ad193x",
+ },
+ .probe = ad193x_i2c_probe,
+ .remove = ad193x_i2c_remove,
+ .id_table = ad193x_id,
+};
+module_i2c_driver(ad193x_i2c_driver);
+
+MODULE_DESCRIPTION("ASoC AD1936/AD1937 audio CODEC driver");
+MODULE_AUTHOR("Barry Song <21cnbao@gmail.com>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/ad193x-spi.c b/sound/soc/codecs/ad193x-spi.c
new file mode 100644
index 0000000..390cef9
--- /dev/null
+++ b/sound/soc/codecs/ad193x-spi.c
@@ -0,0 +1,48 @@
+/*
+ * AD1938/AD1939 audio driver
+ *
+ * Copyright 2014 Analog Devices Inc.
+ *
+ * Licensed under the GPL-2.
+ */
+
+#include <linux/module.h>
+#include <linux/spi/spi.h>
+#include <linux/regmap.h>
+
+#include <sound/soc.h>
+
+#include "ad193x.h"
+
+static int ad193x_spi_probe(struct spi_device *spi)
+{
+ struct regmap_config config;
+
+ config = ad193x_regmap_config;
+ config.val_bits = 8;
+ config.reg_bits = 16;
+ config.read_flag_mask = 0x09;
+ config.write_flag_mask = 0x08;
+
+ return ad193x_probe(&spi->dev, devm_regmap_init_spi(spi, &config));
+}
+
+static int ad193x_spi_remove(struct spi_device *spi)
+{
+ snd_soc_unregister_codec(&spi->dev);
+ return 0;
+}
+
+static struct spi_driver ad193x_spi_driver = {
+ .driver = {
+ .name = "ad193x",
+ .owner = THIS_MODULE,
+ },
+ .probe = ad193x_spi_probe,
+ .remove = ad193x_spi_remove,
+};
+module_spi_driver(ad193x_spi_driver);
+
+MODULE_DESCRIPTION("ASoC AD1938/AD1939 audio CODEC driver");
+MODULE_AUTHOR("Barry Song <21cnbao@gmail.com>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c
index 5a42dca..6844d0b 100644
--- a/sound/soc/codecs/ad193x.c
+++ b/sound/soc/codecs/ad193x.c
@@ -6,12 +6,10 @@
* Licensed under the GPL-2 or later.
*/
-#include <linux/init.h>
#include <linux/module.h>
#include <linux/kernel.h>
#include <linux/device.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>
@@ -19,6 +17,7 @@
#include <sound/initval.h>
#include <sound/soc.h>
#include <sound/tlv.h>
+
#include "ad193x.h"
/* codec private data */
@@ -32,8 +31,8 @@ struct ad193x_priv {
*/
static const char * const ad193x_deemp[] = {"None", "48kHz", "44.1kHz", "32kHz"};
-static const struct soc_enum ad193x_deemp_enum =
- SOC_ENUM_SINGLE(AD193X_DAC_CTRL2, 1, 4, ad193x_deemp);
+static SOC_ENUM_SINGLE_DECL(ad193x_deemp_enum, AD193X_DAC_CTRL2, 1,
+ ad193x_deemp);
static const DECLARE_TLV_DB_MINMAX(adau193x_tlv, -9563, 0);
@@ -320,17 +319,9 @@ static struct snd_soc_dai_driver ad193x_dai = {
.ops = &ad193x_dai_ops,
};
-static int ad193x_probe(struct snd_soc_codec *codec)
+static int ad193x_codec_probe(struct snd_soc_codec *codec)
{
struct ad193x_priv *ad193x = snd_soc_codec_get_drvdata(codec);
- int ret;
-
- codec->control_data = ad193x->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;
- }
/* default setting for ad193x */
@@ -348,11 +339,11 @@ static int ad193x_probe(struct snd_soc_codec *codec)
regmap_write(ad193x->regmap, AD193X_PLL_CLK_CTRL0, 0x99); /* mclk=24.576Mhz: 0x9D; mclk=12.288Mhz: 0x99 */
regmap_write(ad193x->regmap, AD193X_PLL_CLK_CTRL1, 0x04);
- return ret;
+ return 0;
}
static struct snd_soc_codec_driver soc_codec_dev_ad193x = {
- .probe = ad193x_probe,
+ .probe = ad193x_codec_probe,
.controls = ad193x_snd_controls,
.num_controls = ARRAY_SIZE(ad193x_snd_controls),
.dapm_widgets = ad193x_dapm_widgets,
@@ -366,140 +357,31 @@ static bool adau193x_reg_volatile(struct device *dev, unsigned int reg)
return false;
}
-#if defined(CONFIG_SPI_MASTER)
-
-static const struct regmap_config ad193x_spi_regmap_config = {
- .val_bits = 8,
- .reg_bits = 16,
- .read_flag_mask = 0x09,
- .write_flag_mask = 0x08,
-
+const struct regmap_config ad193x_regmap_config = {
.max_register = AD193X_NUM_REGS - 1,
.volatile_reg = adau193x_reg_volatile,
};
+EXPORT_SYMBOL_GPL(ad193x_regmap_config);
-static int ad193x_spi_probe(struct spi_device *spi)
+int ad193x_probe(struct device *dev, struct regmap *regmap)
{
struct ad193x_priv *ad193x;
- ad193x = devm_kzalloc(&spi->dev, sizeof(struct ad193x_priv),
- GFP_KERNEL);
- if (ad193x == NULL)
- return -ENOMEM;
-
- ad193x->regmap = devm_regmap_init_spi(spi, &ad193x_spi_regmap_config);
- if (IS_ERR(ad193x->regmap))
- return PTR_ERR(ad193x->regmap);
-
- spi_set_drvdata(spi, ad193x);
-
- return snd_soc_register_codec(&spi->dev, &soc_codec_dev_ad193x,
- &ad193x_dai, 1);
-}
-
-static int ad193x_spi_remove(struct spi_device *spi)
-{
- snd_soc_unregister_codec(&spi->dev);
- return 0;
-}
-
-static struct spi_driver ad193x_spi_driver = {
- .driver = {
- .name = "ad193x",
- .owner = THIS_MODULE,
- },
- .probe = ad193x_spi_probe,
- .remove = ad193x_spi_remove,
-};
-#endif
-
-#if IS_ENABLED(CONFIG_I2C)
-
-static const struct regmap_config ad193x_i2c_regmap_config = {
- .val_bits = 8,
- .reg_bits = 8,
-
- .max_register = AD193X_NUM_REGS - 1,
- .volatile_reg = adau193x_reg_volatile,
-};
-
-static const struct i2c_device_id ad193x_id[] = {
- { "ad1936", 0 },
- { "ad1937", 0 },
- { }
-};
-MODULE_DEVICE_TABLE(i2c, ad193x_id);
-
-static int ad193x_i2c_probe(struct i2c_client *client,
- const struct i2c_device_id *id)
-{
- struct ad193x_priv *ad193x;
+ if (IS_ERR(regmap))
+ return PTR_ERR(regmap);
- ad193x = devm_kzalloc(&client->dev, sizeof(struct ad193x_priv),
- GFP_KERNEL);
+ ad193x = devm_kzalloc(dev, sizeof(*ad193x), GFP_KERNEL);
if (ad193x == NULL)
return -ENOMEM;
- ad193x->regmap = devm_regmap_init_i2c(client, &ad193x_i2c_regmap_config);
- if (IS_ERR(ad193x->regmap))
- return PTR_ERR(ad193x->regmap);
-
- i2c_set_clientdata(client, ad193x);
-
- return snd_soc_register_codec(&client->dev, &soc_codec_dev_ad193x,
- &ad193x_dai, 1);
-}
-
-static int ad193x_i2c_remove(struct i2c_client *client)
-{
- snd_soc_unregister_codec(&client->dev);
- return 0;
-}
-
-static struct i2c_driver ad193x_i2c_driver = {
- .driver = {
- .name = "ad193x",
- },
- .probe = ad193x_i2c_probe,
- .remove = ad193x_i2c_remove,
- .id_table = ad193x_id,
-};
-#endif
-
-static int __init ad193x_modinit(void)
-{
- int ret;
-
-#if IS_ENABLED(CONFIG_I2C)
- ret = i2c_add_driver(&ad193x_i2c_driver);
- if (ret != 0) {
- printk(KERN_ERR "Failed to register AD193X I2C driver: %d\n",
- ret);
- }
-#endif
-
-#if defined(CONFIG_SPI_MASTER)
- ret = spi_register_driver(&ad193x_spi_driver);
- if (ret != 0) {
- printk(KERN_ERR "Failed to register AD193X SPI driver: %d\n",
- ret);
- }
-#endif
- return ret;
-}
-module_init(ad193x_modinit);
+ ad193x->regmap = regmap;
-static void __exit ad193x_modexit(void)
-{
-#if defined(CONFIG_SPI_MASTER)
- spi_unregister_driver(&ad193x_spi_driver);
-#endif
+ dev_set_drvdata(dev, ad193x);
-#if IS_ENABLED(CONFIG_I2C)
- i2c_del_driver(&ad193x_i2c_driver);
-#endif
+ return snd_soc_register_codec(dev, &soc_codec_dev_ad193x,
+ &ad193x_dai, 1);
}
-module_exit(ad193x_modexit);
+EXPORT_SYMBOL_GPL(ad193x_probe);
MODULE_DESCRIPTION("ASoC ad193x driver");
MODULE_AUTHOR("Barry Song <21cnbao@gmail.com>");
diff --git a/sound/soc/codecs/ad193x.h b/sound/soc/codecs/ad193x.h
index 4733880..ab9a998 100644
--- a/sound/soc/codecs/ad193x.h
+++ b/sound/soc/codecs/ad193x.h
@@ -9,6 +9,13 @@
#ifndef __AD193X_H__
#define __AD193X_H__
+#include <linux/regmap.h>
+
+struct device;
+
+extern const struct regmap_config ad193x_regmap_config;
+int ad193x_probe(struct device *dev, struct regmap *regmap);
+
#define AD193X_PLL_CLK_CTRL0 0x00
#define AD193X_PLL_POWERDOWN 0x01
#define AD193X_PLL_INPUT_MASK 0x6
diff --git a/sound/soc/codecs/adau1373.c b/sound/soc/codecs/adau1373.c
index eb836ed..877f573 100644
--- a/sound/soc/codecs/adau1373.c
+++ b/sound/soc/codecs/adau1373.c
@@ -345,15 +345,15 @@ static const char *adau1373_fdsp_sel_text[] = {
"Channel 5",
};
-static const SOC_ENUM_SINGLE_DECL(adau1373_drc1_channel_enum,
+static SOC_ENUM_SINGLE_DECL(adau1373_drc1_channel_enum,
ADAU1373_FDSP_SEL1, 4, adau1373_fdsp_sel_text);
-static const SOC_ENUM_SINGLE_DECL(adau1373_drc2_channel_enum,
+static SOC_ENUM_SINGLE_DECL(adau1373_drc2_channel_enum,
ADAU1373_FDSP_SEL1, 0, adau1373_fdsp_sel_text);
-static const SOC_ENUM_SINGLE_DECL(adau1373_drc3_channel_enum,
+static SOC_ENUM_SINGLE_DECL(adau1373_drc3_channel_enum,
ADAU1373_FDSP_SEL2, 0, adau1373_fdsp_sel_text);
-static const SOC_ENUM_SINGLE_DECL(adau1373_hpf_channel_enum,
+static SOC_ENUM_SINGLE_DECL(adau1373_hpf_channel_enum,
ADAU1373_FDSP_SEL3, 0, adau1373_fdsp_sel_text);
-static const SOC_ENUM_SINGLE_DECL(adau1373_bass_channel_enum,
+static SOC_ENUM_SINGLE_DECL(adau1373_bass_channel_enum,
ADAU1373_FDSP_SEL4, 4, adau1373_fdsp_sel_text);
static const char *adau1373_hpf_cutoff_text[] = {
@@ -362,7 +362,7 @@ static const char *adau1373_hpf_cutoff_text[] = {
"800Hz",
};
-static const SOC_ENUM_SINGLE_DECL(adau1373_hpf_cutoff_enum,
+static SOC_ENUM_SINGLE_DECL(adau1373_hpf_cutoff_enum,
ADAU1373_HPF_CTRL, 3, adau1373_hpf_cutoff_text);
static const char *adau1373_bass_lpf_cutoff_text[] = {
@@ -388,14 +388,14 @@ static const unsigned int adau1373_bass_tlv[] = {
5, 7, TLV_DB_SCALE_ITEM(1400, 150, 0),
};
-static const SOC_ENUM_SINGLE_DECL(adau1373_bass_lpf_cutoff_enum,
+static SOC_ENUM_SINGLE_DECL(adau1373_bass_lpf_cutoff_enum,
ADAU1373_BASS1, 5, adau1373_bass_lpf_cutoff_text);
-static const SOC_VALUE_ENUM_SINGLE_DECL(adau1373_bass_clip_level_enum,
+static SOC_VALUE_ENUM_SINGLE_DECL(adau1373_bass_clip_level_enum,
ADAU1373_BASS1, 2, 7, adau1373_bass_clip_level_text,
adau1373_bass_clip_level_values);
-static const SOC_ENUM_SINGLE_DECL(adau1373_bass_hpf_cutoff_enum,
+static SOC_ENUM_SINGLE_DECL(adau1373_bass_hpf_cutoff_enum,
ADAU1373_BASS1, 0, adau1373_bass_hpf_cutoff_text);
static const char *adau1373_3d_level_text[] = {
@@ -409,9 +409,9 @@ static const char *adau1373_3d_cutoff_text[] = {
"0.16875 fs", "0.27083 fs"
};
-static const SOC_ENUM_SINGLE_DECL(adau1373_3d_level_enum,
+static SOC_ENUM_SINGLE_DECL(adau1373_3d_level_enum,
ADAU1373_3D_CTRL1, 4, adau1373_3d_level_text);
-static const SOC_ENUM_SINGLE_DECL(adau1373_3d_cutoff_enum,
+static SOC_ENUM_SINGLE_DECL(adau1373_3d_cutoff_enum,
ADAU1373_3D_CTRL1, 0, adau1373_3d_cutoff_text);
static const unsigned int adau1373_3d_tlv[] = {
@@ -427,11 +427,11 @@ static const char *adau1373_lr_mux_text[] = {
"Stereo",
};
-static const SOC_ENUM_SINGLE_DECL(adau1373_lineout1_lr_mux_enum,
+static SOC_ENUM_SINGLE_DECL(adau1373_lineout1_lr_mux_enum,
ADAU1373_OUTPUT_CTRL, 4, adau1373_lr_mux_text);
-static const SOC_ENUM_SINGLE_DECL(adau1373_lineout2_lr_mux_enum,
+static SOC_ENUM_SINGLE_DECL(adau1373_lineout2_lr_mux_enum,
ADAU1373_OUTPUT_CTRL, 6, adau1373_lr_mux_text);
-static const SOC_ENUM_SINGLE_DECL(adau1373_speaker_lr_mux_enum,
+static SOC_ENUM_SINGLE_DECL(adau1373_speaker_lr_mux_enum,
ADAU1373_LS_CTRL, 4, adau1373_lr_mux_text);
static const struct snd_kcontrol_new adau1373_controls[] = {
@@ -576,8 +576,8 @@ static const char *adau1373_decimator_text[] = {
"DMIC1",
};
-static const struct soc_enum adau1373_decimator_enum =
- SOC_ENUM_SINGLE(0, 0, 2, adau1373_decimator_text);
+static SOC_ENUM_SINGLE_VIRT_DECL(adau1373_decimator_enum,
+ adau1373_decimator_text);
static const struct snd_kcontrol_new adau1373_decimator_mux =
SOC_DAPM_ENUM_VIRT("Decimator Mux", adau1373_decimator_enum);
@@ -1376,15 +1376,8 @@ static int adau1373_probe(struct snd_soc_codec *codec)
struct adau1373_platform_data *pdata = codec->dev->platform_data;
bool lineout_differential = false;
unsigned int val;
- int ret;
int i;
- ret = snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP);
- if (ret) {
- dev_err(codec->dev, "failed to set cache I/O: %d\n", ret);
- return ret;
- }
-
if (pdata) {
if (pdata->num_drc > ARRAY_SIZE(pdata->drc_setting))
return -EINVAL;
diff --git a/sound/soc/codecs/adau1977-i2c.c b/sound/soc/codecs/adau1977-i2c.c
new file mode 100644
index 0000000..9700e8c
--- /dev/null
+++ b/sound/soc/codecs/adau1977-i2c.c
@@ -0,0 +1,59 @@
+/*
+ * ADAU1977/ADAU1978/ADAU1979 driver
+ *
+ * Copyright 2014 Analog Devices Inc.
+ * Author: Lars-Peter Clausen <lars@metafoo.de>
+ *
+ * Licensed under the GPL-2.
+ */
+
+#include <linux/i2c.h>
+#include <linux/mod_devicetable.h>
+#include <linux/module.h>
+#include <linux/regmap.h>
+#include <sound/soc.h>
+
+#include "adau1977.h"
+
+static int adau1977_i2c_probe(struct i2c_client *client,
+ const struct i2c_device_id *id)
+{
+ struct regmap_config config;
+
+ config = adau1977_regmap_config;
+ config.val_bits = 8;
+ config.reg_bits = 8;
+
+ return adau1977_probe(&client->dev,
+ devm_regmap_init_i2c(client, &config),
+ id->driver_data, NULL);
+}
+
+static int adau1977_i2c_remove(struct i2c_client *client)
+{
+ snd_soc_unregister_codec(&client->dev);
+ return 0;
+}
+
+static const struct i2c_device_id adau1977_i2c_ids[] = {
+ { "adau1977", ADAU1977 },
+ { "adau1978", ADAU1978 },
+ { "adau1979", ADAU1978 },
+ { }
+};
+MODULE_DEVICE_TABLE(i2c, adau1977_i2c_ids);
+
+static struct i2c_driver adau1977_i2c_driver = {
+ .driver = {
+ .name = "adau1977",
+ .owner = THIS_MODULE,
+ },
+ .probe = adau1977_i2c_probe,
+ .remove = adau1977_i2c_remove,
+ .id_table = adau1977_i2c_ids,
+};
+module_i2c_driver(adau1977_i2c_driver);
+
+MODULE_DESCRIPTION("ASoC ADAU1977/ADAU1978/ADAU1979 driver");
+MODULE_AUTHOR("Lars-Peter Clausen <lars@metafoo.de>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/adau1977-spi.c b/sound/soc/codecs/adau1977-spi.c
new file mode 100644
index 0000000..b05cf5d
--- /dev/null
+++ b/sound/soc/codecs/adau1977-spi.c
@@ -0,0 +1,76 @@
+/*
+ * ADAU1977/ADAU1978/ADAU1979 driver
+ *
+ * Copyright 2014 Analog Devices Inc.
+ * Author: Lars-Peter Clausen <lars@metafoo.de>
+ *
+ * Licensed under the GPL-2.
+ */
+
+#include <linux/mod_devicetable.h>
+#include <linux/module.h>
+#include <linux/regmap.h>
+#include <linux/spi/spi.h>
+#include <sound/soc.h>
+
+#include "adau1977.h"
+
+static void adau1977_spi_switch_mode(struct device *dev)
+{
+ struct spi_device *spi = to_spi_device(dev);
+
+ /*
+ * To get the device into SPI mode CLATCH has to be pulled low three
+ * times. Do this by issuing three dummy reads.
+ */
+ spi_w8r8(spi, 0x00);
+ spi_w8r8(spi, 0x00);
+ spi_w8r8(spi, 0x00);
+}
+
+static int adau1977_spi_probe(struct spi_device *spi)
+{
+ const struct spi_device_id *id = spi_get_device_id(spi);
+ struct regmap_config config;
+
+ if (!id)
+ return -EINVAL;
+
+ config = adau1977_regmap_config;
+ config.val_bits = 8;
+ config.reg_bits = 16;
+ config.read_flag_mask = 0x1;
+
+ return adau1977_probe(&spi->dev,
+ devm_regmap_init_spi(spi, &config),
+ id->driver_data, adau1977_spi_switch_mode);
+}
+
+static int adau1977_spi_remove(struct spi_device *spi)
+{
+ snd_soc_unregister_codec(&spi->dev);
+ return 0;
+}
+
+static const struct spi_device_id adau1977_spi_ids[] = {
+ { "adau1977", ADAU1977 },
+ { "adau1978", ADAU1978 },
+ { "adau1979", ADAU1978 },
+ { }
+};
+MODULE_DEVICE_TABLE(spi, adau1977_spi_ids);
+
+static struct spi_driver adau1977_spi_driver = {
+ .driver = {
+ .name = "adau1977",
+ .owner = THIS_MODULE,
+ },
+ .probe = adau1977_spi_probe,
+ .remove = adau1977_spi_remove,
+ .id_table = adau1977_spi_ids,
+};
+module_spi_driver(adau1977_spi_driver);
+
+MODULE_DESCRIPTION("ASoC ADAU1977/ADAU1978/ADAU1979 driver");
+MODULE_AUTHOR("Lars-Peter Clausen <lars@metafoo.de>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/adau1977.c b/sound/soc/codecs/adau1977.c
new file mode 100644
index 0000000..fd55da7
--- /dev/null
+++ b/sound/soc/codecs/adau1977.c
@@ -0,0 +1,1018 @@
+/*
+ * ADAU1977/ADAU1978/ADAU1979 driver
+ *
+ * Copyright 2014 Analog Devices Inc.
+ * Author: Lars-Peter Clausen <lars@metafoo.de>
+ *
+ * Licensed under the GPL-2.
+ */
+
+#include <linux/delay.h>
+#include <linux/device.h>
+#include <linux/gpio/consumer.h>
+#include <linux/i2c.h>
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/platform_data/adau1977.h>
+#include <linux/regmap.h>
+#include <linux/regulator/consumer.h>
+#include <linux/slab.h>
+
+#include <sound/core.h>
+#include <sound/initval.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/tlv.h>
+
+#include "adau1977.h"
+
+#define ADAU1977_REG_POWER 0x00
+#define ADAU1977_REG_PLL 0x01
+#define ADAU1977_REG_BOOST 0x02
+#define ADAU1977_REG_MICBIAS 0x03
+#define ADAU1977_REG_BLOCK_POWER_SAI 0x04
+#define ADAU1977_REG_SAI_CTRL0 0x05
+#define ADAU1977_REG_SAI_CTRL1 0x06
+#define ADAU1977_REG_CMAP12 0x07
+#define ADAU1977_REG_CMAP34 0x08
+#define ADAU1977_REG_SAI_OVERTEMP 0x09
+#define ADAU1977_REG_POST_ADC_GAIN(x) (0x0a + (x))
+#define ADAU1977_REG_MISC_CONTROL 0x0e
+#define ADAU1977_REG_DIAG_CONTROL 0x10
+#define ADAU1977_REG_STATUS(x) (0x11 + (x))
+#define ADAU1977_REG_DIAG_IRQ1 0x15
+#define ADAU1977_REG_DIAG_IRQ2 0x16
+#define ADAU1977_REG_ADJUST1 0x17
+#define ADAU1977_REG_ADJUST2 0x18
+#define ADAU1977_REG_ADC_CLIP 0x19
+#define ADAU1977_REG_DC_HPF_CAL 0x1a
+
+#define ADAU1977_POWER_RESET BIT(7)
+#define ADAU1977_POWER_PWUP BIT(0)
+
+#define ADAU1977_PLL_CLK_S BIT(4)
+#define ADAU1977_PLL_MCS_MASK 0x7
+
+#define ADAU1977_MICBIAS_MB_VOLTS_MASK 0xf0
+#define ADAU1977_MICBIAS_MB_VOLTS_OFFSET 4
+
+#define ADAU1977_BLOCK_POWER_SAI_LR_POL BIT(7)
+#define ADAU1977_BLOCK_POWER_SAI_BCLK_EDGE BIT(6)
+#define ADAU1977_BLOCK_POWER_SAI_LDO_EN BIT(5)
+
+#define ADAU1977_SAI_CTRL0_FMT_MASK (0x3 << 6)
+#define ADAU1977_SAI_CTRL0_FMT_I2S (0x0 << 6)
+#define ADAU1977_SAI_CTRL0_FMT_LJ (0x1 << 6)
+#define ADAU1977_SAI_CTRL0_FMT_RJ_24BIT (0x2 << 6)
+#define ADAU1977_SAI_CTRL0_FMT_RJ_16BIT (0x3 << 6)
+
+#define ADAU1977_SAI_CTRL0_SAI_MASK (0x7 << 3)
+#define ADAU1977_SAI_CTRL0_SAI_I2S (0x0 << 3)
+#define ADAU1977_SAI_CTRL0_SAI_TDM_2 (0x1 << 3)
+#define ADAU1977_SAI_CTRL0_SAI_TDM_4 (0x2 << 3)
+#define ADAU1977_SAI_CTRL0_SAI_TDM_8 (0x3 << 3)
+#define ADAU1977_SAI_CTRL0_SAI_TDM_16 (0x4 << 3)
+
+#define ADAU1977_SAI_CTRL0_FS_MASK (0x7)
+#define ADAU1977_SAI_CTRL0_FS_8000_12000 (0x0)
+#define ADAU1977_SAI_CTRL0_FS_16000_24000 (0x1)
+#define ADAU1977_SAI_CTRL0_FS_32000_48000 (0x2)
+#define ADAU1977_SAI_CTRL0_FS_64000_96000 (0x3)
+#define ADAU1977_SAI_CTRL0_FS_128000_192000 (0x4)
+
+#define ADAU1977_SAI_CTRL1_SLOT_WIDTH_MASK (0x3 << 5)
+#define ADAU1977_SAI_CTRL1_SLOT_WIDTH_32 (0x0 << 5)
+#define ADAU1977_SAI_CTRL1_SLOT_WIDTH_24 (0x1 << 5)
+#define ADAU1977_SAI_CTRL1_SLOT_WIDTH_16 (0x2 << 5)
+#define ADAU1977_SAI_CTRL1_DATA_WIDTH_MASK (0x1 << 4)
+#define ADAU1977_SAI_CTRL1_DATA_WIDTH_16BIT (0x1 << 4)
+#define ADAU1977_SAI_CTRL1_DATA_WIDTH_24BIT (0x0 << 4)
+#define ADAU1977_SAI_CTRL1_LRCLK_PULSE BIT(3)
+#define ADAU1977_SAI_CTRL1_MSB BIT(2)
+#define ADAU1977_SAI_CTRL1_BCLKRATE_16 (0x1 << 1)
+#define ADAU1977_SAI_CTRL1_BCLKRATE_32 (0x0 << 1)
+#define ADAU1977_SAI_CTRL1_BCLKRATE_MASK (0x1 << 1)
+#define ADAU1977_SAI_CTRL1_MASTER BIT(0)
+
+#define ADAU1977_SAI_OVERTEMP_DRV_C(x) BIT(4 + (x))
+#define ADAU1977_SAI_OVERTEMP_DRV_HIZ BIT(3)
+
+#define ADAU1977_MISC_CONTROL_SUM_MODE_MASK (0x3 << 6)
+#define ADAU1977_MISC_CONTROL_SUM_MODE_1CH (0x2 << 6)
+#define ADAU1977_MISC_CONTROL_SUM_MODE_2CH (0x1 << 6)
+#define ADAU1977_MISC_CONTROL_SUM_MODE_4CH (0x0 << 6)
+#define ADAU1977_MISC_CONTROL_MMUTE BIT(4)
+#define ADAU1977_MISC_CONTROL_DC_CAL BIT(0)
+
+#define ADAU1977_CHAN_MAP_SECOND_SLOT_OFFSET 4
+#define ADAU1977_CHAN_MAP_FIRST_SLOT_OFFSET 0
+
+struct adau1977 {
+ struct regmap *regmap;
+ bool right_j;
+ unsigned int sysclk;
+ enum adau1977_sysclk_src sysclk_src;
+ struct gpio_desc *reset_gpio;
+ enum adau1977_type type;
+
+ struct regulator *avdd_reg;
+ struct regulator *dvdd_reg;
+
+ struct snd_pcm_hw_constraint_list constraints;
+
+ struct device *dev;
+ void (*switch_mode)(struct device *dev);
+
+ unsigned int max_master_fs;
+ unsigned int slot_width;
+ bool enabled;
+ bool master;
+};
+
+static const struct reg_default adau1977_reg_defaults[] = {
+ { 0x00, 0x00 },
+ { 0x01, 0x41 },
+ { 0x02, 0x4a },
+ { 0x03, 0x7d },
+ { 0x04, 0x3d },
+ { 0x05, 0x02 },
+ { 0x06, 0x00 },
+ { 0x07, 0x10 },
+ { 0x08, 0x32 },
+ { 0x09, 0xf0 },
+ { 0x0a, 0xa0 },
+ { 0x0b, 0xa0 },
+ { 0x0c, 0xa0 },
+ { 0x0d, 0xa0 },
+ { 0x0e, 0x02 },
+ { 0x10, 0x0f },
+ { 0x15, 0x20 },
+ { 0x16, 0x00 },
+ { 0x17, 0x00 },
+ { 0x18, 0x00 },
+ { 0x1a, 0x00 },
+};
+
+static const DECLARE_TLV_DB_MINMAX_MUTE(adau1977_adc_gain, -3562, 6000);
+
+static const struct snd_soc_dapm_widget adau1977_micbias_dapm_widgets[] = {
+ SND_SOC_DAPM_SUPPLY("MICBIAS", ADAU1977_REG_MICBIAS,
+ 3, 0, NULL, 0)
+};
+
+static const struct snd_soc_dapm_widget adau1977_dapm_widgets[] = {
+ SND_SOC_DAPM_SUPPLY("Vref", ADAU1977_REG_BLOCK_POWER_SAI,
+ 4, 0, NULL, 0),
+
+ SND_SOC_DAPM_ADC("ADC1", "Capture", ADAU1977_REG_BLOCK_POWER_SAI, 0, 0),
+ SND_SOC_DAPM_ADC("ADC2", "Capture", ADAU1977_REG_BLOCK_POWER_SAI, 1, 0),
+ SND_SOC_DAPM_ADC("ADC3", "Capture", ADAU1977_REG_BLOCK_POWER_SAI, 2, 0),
+ SND_SOC_DAPM_ADC("ADC4", "Capture", ADAU1977_REG_BLOCK_POWER_SAI, 3, 0),
+
+ SND_SOC_DAPM_INPUT("AIN1"),
+ SND_SOC_DAPM_INPUT("AIN2"),
+ SND_SOC_DAPM_INPUT("AIN3"),
+ SND_SOC_DAPM_INPUT("AIN4"),
+
+ SND_SOC_DAPM_OUTPUT("VREF"),
+};
+
+static const struct snd_soc_dapm_route adau1977_dapm_routes[] = {
+ { "ADC1", NULL, "AIN1" },
+ { "ADC2", NULL, "AIN2" },
+ { "ADC3", NULL, "AIN3" },
+ { "ADC4", NULL, "AIN4" },
+
+ { "ADC1", NULL, "Vref" },
+ { "ADC2", NULL, "Vref" },
+ { "ADC3", NULL, "Vref" },
+ { "ADC4", NULL, "Vref" },
+
+ { "VREF", NULL, "Vref" },
+};
+
+#define ADAU1977_VOLUME(x) \
+ SOC_SINGLE_TLV("ADC" #x " Capture Volume", \
+ ADAU1977_REG_POST_ADC_GAIN((x) - 1), \
+ 0, 255, 1, adau1977_adc_gain)
+
+#define ADAU1977_HPF_SWITCH(x) \
+ SOC_SINGLE("ADC" #x " Highpass-Filter Capture Switch", \
+ ADAU1977_REG_DC_HPF_CAL, (x) - 1, 1, 0)
+
+#define ADAU1977_DC_SUB_SWITCH(x) \
+ SOC_SINGLE("ADC" #x " DC Substraction Capture Switch", \
+ ADAU1977_REG_DC_HPF_CAL, (x) + 3, 1, 0)
+
+static const struct snd_kcontrol_new adau1977_snd_controls[] = {
+ ADAU1977_VOLUME(1),
+ ADAU1977_VOLUME(2),
+ ADAU1977_VOLUME(3),
+ ADAU1977_VOLUME(4),
+
+ ADAU1977_HPF_SWITCH(1),
+ ADAU1977_HPF_SWITCH(2),
+ ADAU1977_HPF_SWITCH(3),
+ ADAU1977_HPF_SWITCH(4),
+
+ ADAU1977_DC_SUB_SWITCH(1),
+ ADAU1977_DC_SUB_SWITCH(2),
+ ADAU1977_DC_SUB_SWITCH(3),
+ ADAU1977_DC_SUB_SWITCH(4),
+};
+
+static int adau1977_reset(struct adau1977 *adau1977)
+{
+ int ret;
+
+ /*
+ * The reset bit is obviously volatile, but we need to be able to cache
+ * the other bits in the register, so we can't just mark the whole
+ * register as volatile. Since this is the only place where we'll ever
+ * touch the reset bit just bypass the cache for this operation.
+ */
+ regcache_cache_bypass(adau1977->regmap, true);
+ ret = regmap_write(adau1977->regmap, ADAU1977_REG_POWER,
+ ADAU1977_POWER_RESET);
+ regcache_cache_bypass(adau1977->regmap, false);
+ if (ret)
+ return ret;
+
+ return ret;
+}
+
+/*
+ * Returns the appropriate setting for ths FS field in the CTRL0 register
+ * depending on the rate.
+ */
+static int adau1977_lookup_fs(unsigned int rate)
+{
+ if (rate >= 8000 && rate <= 12000)
+ return ADAU1977_SAI_CTRL0_FS_8000_12000;
+ else if (rate >= 16000 && rate <= 24000)
+ return ADAU1977_SAI_CTRL0_FS_16000_24000;
+ else if (rate >= 32000 && rate <= 48000)
+ return ADAU1977_SAI_CTRL0_FS_32000_48000;
+ else if (rate >= 64000 && rate <= 96000)
+ return ADAU1977_SAI_CTRL0_FS_64000_96000;
+ else if (rate >= 128000 && rate <= 192000)
+ return ADAU1977_SAI_CTRL0_FS_128000_192000;
+ else
+ return -EINVAL;
+}
+
+static int adau1977_lookup_mcs(struct adau1977 *adau1977, unsigned int rate,
+ unsigned int fs)
+{
+ unsigned int mcs;
+
+ /*
+ * rate = sysclk / (512 * mcs_lut[mcs]) * 2**fs
+ * => mcs_lut[mcs] = sysclk / (512 * rate) * 2**fs
+ * => mcs_lut[mcs] = sysclk / ((512 / 2**fs) * rate)
+ */
+
+ rate *= 512 >> fs;
+
+ if (adau1977->sysclk % rate != 0)
+ return -EINVAL;
+
+ mcs = adau1977->sysclk / rate;
+
+ /* The factors configured by MCS are 1, 2, 3, 4, 6 */
+ if (mcs < 1 || mcs > 6 || mcs == 5)
+ return -EINVAL;
+
+ mcs = mcs - 1;
+ if (mcs == 5)
+ mcs = 4;
+
+ return mcs;
+}
+
+static int adau1977_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 adau1977 *adau1977 = snd_soc_codec_get_drvdata(codec);
+ unsigned int rate = params_rate(params);
+ unsigned int slot_width;
+ unsigned int ctrl0, ctrl0_mask;
+ unsigned int ctrl1;
+ int mcs, fs;
+ int ret;
+
+ fs = adau1977_lookup_fs(rate);
+ if (fs < 0)
+ return fs;
+
+ if (adau1977->sysclk_src == ADAU1977_SYSCLK_SRC_MCLK) {
+ mcs = adau1977_lookup_mcs(adau1977, rate, fs);
+ if (mcs < 0)
+ return mcs;
+ } else {
+ mcs = 0;
+ }
+
+ ctrl0_mask = ADAU1977_SAI_CTRL0_FS_MASK;
+ ctrl0 = fs;
+
+ if (adau1977->right_j) {
+ switch (params_width(params)) {
+ case 16:
+ ctrl0 |= ADAU1977_SAI_CTRL0_FMT_RJ_16BIT;
+ break;
+ case 24:
+ ctrl0 |= ADAU1977_SAI_CTRL0_FMT_RJ_24BIT;
+ break;
+ default:
+ return -EINVAL;
+ }
+ ctrl0_mask |= ADAU1977_SAI_CTRL0_FMT_MASK;
+ }
+
+ if (adau1977->master) {
+ switch (params_width(params)) {
+ case 16:
+ ctrl1 = ADAU1977_SAI_CTRL1_DATA_WIDTH_16BIT;
+ slot_width = 16;
+ break;
+ case 24:
+ case 32:
+ ctrl1 = ADAU1977_SAI_CTRL1_DATA_WIDTH_24BIT;
+ slot_width = 32;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ /* In TDM mode there is a fixed slot width */
+ if (adau1977->slot_width)
+ slot_width = adau1977->slot_width;
+
+ if (slot_width == 16)
+ ctrl1 |= ADAU1977_SAI_CTRL1_BCLKRATE_16;
+ else
+ ctrl1 |= ADAU1977_SAI_CTRL1_BCLKRATE_32;
+
+ ret = regmap_update_bits(adau1977->regmap,
+ ADAU1977_REG_SAI_CTRL1,
+ ADAU1977_SAI_CTRL1_DATA_WIDTH_MASK |
+ ADAU1977_SAI_CTRL1_BCLKRATE_MASK,
+ ctrl1);
+ if (ret < 0)
+ return ret;
+ }
+
+ ret = regmap_update_bits(adau1977->regmap, ADAU1977_REG_SAI_CTRL0,
+ ctrl0_mask, ctrl0);
+ if (ret < 0)
+ return ret;
+
+ return regmap_update_bits(adau1977->regmap, ADAU1977_REG_PLL,
+ ADAU1977_PLL_MCS_MASK, mcs);
+}
+
+static int adau1977_power_disable(struct adau1977 *adau1977)
+{
+ int ret = 0;
+
+ if (!adau1977->enabled)
+ return 0;
+
+ ret = regmap_update_bits(adau1977->regmap, ADAU1977_REG_POWER,
+ ADAU1977_POWER_PWUP, 0);
+ if (ret)
+ return ret;
+
+ regcache_mark_dirty(adau1977->regmap);
+
+ if (adau1977->reset_gpio)
+ gpiod_set_value_cansleep(adau1977->reset_gpio, 0);
+
+ regcache_cache_only(adau1977->regmap, true);
+
+ regulator_disable(adau1977->avdd_reg);
+ if (adau1977->dvdd_reg)
+ regulator_disable(adau1977->dvdd_reg);
+
+ adau1977->enabled = false;
+
+ return 0;
+}
+
+static int adau1977_power_enable(struct adau1977 *adau1977)
+{
+ unsigned int val;
+ int ret = 0;
+
+ if (adau1977->enabled)
+ return 0;
+
+ ret = regulator_enable(adau1977->avdd_reg);
+ if (ret)
+ return ret;
+
+ if (adau1977->dvdd_reg) {
+ ret = regulator_enable(adau1977->dvdd_reg);
+ if (ret)
+ goto err_disable_avdd;
+ }
+
+ if (adau1977->reset_gpio)
+ gpiod_set_value_cansleep(adau1977->reset_gpio, 1);
+
+ regcache_cache_only(adau1977->regmap, false);
+
+ if (adau1977->switch_mode)
+ adau1977->switch_mode(adau1977->dev);
+
+ ret = adau1977_reset(adau1977);
+ if (ret)
+ goto err_disable_dvdd;
+
+ ret = regmap_update_bits(adau1977->regmap, ADAU1977_REG_POWER,
+ ADAU1977_POWER_PWUP, ADAU1977_POWER_PWUP);
+ if (ret)
+ goto err_disable_dvdd;
+
+ ret = regcache_sync(adau1977->regmap);
+ if (ret)
+ goto err_disable_dvdd;
+
+ /*
+ * The PLL register is not affected by the software reset. It is
+ * possible that the value of the register was changed to the
+ * default value while we were in cache only mode. In this case
+ * regcache_sync will skip over it and we have to manually sync
+ * it.
+ */
+ ret = regmap_read(adau1977->regmap, ADAU1977_REG_PLL, &val);
+ if (ret)
+ goto err_disable_dvdd;
+
+ if (val == 0x41) {
+ regcache_cache_bypass(adau1977->regmap, true);
+ ret = regmap_write(adau1977->regmap, ADAU1977_REG_PLL,
+ 0x41);
+ if (ret)
+ goto err_disable_dvdd;
+ regcache_cache_bypass(adau1977->regmap, false);
+ }
+
+ adau1977->enabled = true;
+
+ return ret;
+
+err_disable_dvdd:
+ if (adau1977->dvdd_reg)
+ regulator_disable(adau1977->dvdd_reg);
+err_disable_avdd:
+ regulator_disable(adau1977->avdd_reg);
+ return ret;
+}
+
+static int adau1977_set_bias_level(struct snd_soc_codec *codec,
+ enum snd_soc_bias_level level)
+{
+ struct adau1977 *adau1977 = snd_soc_codec_get_drvdata(codec);
+ int ret = 0;
+
+ switch (level) {
+ case SND_SOC_BIAS_ON:
+ break;
+ case SND_SOC_BIAS_PREPARE:
+ break;
+ case SND_SOC_BIAS_STANDBY:
+ if (codec->dapm.bias_level == SND_SOC_BIAS_OFF)
+ ret = adau1977_power_enable(adau1977);
+ break;
+ case SND_SOC_BIAS_OFF:
+ ret = adau1977_power_disable(adau1977);
+ break;
+ }
+
+ if (ret)
+ return ret;
+
+ codec->dapm.bias_level = level;
+
+ return 0;
+}
+
+static int adau1977_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask,
+ unsigned int rx_mask, int slots, int width)
+{
+ struct adau1977 *adau1977 = snd_soc_codec_get_drvdata(dai->codec);
+ unsigned int ctrl0, ctrl1, drv;
+ unsigned int slot[4];
+ unsigned int i;
+ int ret;
+
+ if (slots == 0) {
+ /* 0 = No fixed slot width */
+ adau1977->slot_width = 0;
+ adau1977->max_master_fs = 192000;
+ return regmap_update_bits(adau1977->regmap,
+ ADAU1977_REG_SAI_CTRL0, ADAU1977_SAI_CTRL0_SAI_MASK,
+ ADAU1977_SAI_CTRL0_SAI_I2S);
+ }
+
+ if (rx_mask == 0 || tx_mask != 0)
+ return -EINVAL;
+
+ drv = 0;
+ for (i = 0; i < 4; i++) {
+ slot[i] = __ffs(rx_mask);
+ drv |= ADAU1977_SAI_OVERTEMP_DRV_C(i);
+ rx_mask &= ~(1 << slot[i]);
+ if (slot[i] >= slots)
+ return -EINVAL;
+ if (rx_mask == 0)
+ break;
+ }
+
+ if (rx_mask != 0)
+ return -EINVAL;
+
+ switch (width) {
+ case 16:
+ ctrl1 = ADAU1977_SAI_CTRL1_SLOT_WIDTH_16;
+ break;
+ case 24:
+ /* We can only generate 16 bit or 32 bit wide slots */
+ if (adau1977->master)
+ return -EINVAL;
+ ctrl1 = ADAU1977_SAI_CTRL1_SLOT_WIDTH_24;
+ break;
+ case 32:
+ ctrl1 = ADAU1977_SAI_CTRL1_SLOT_WIDTH_32;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ switch (slots) {
+ case 2:
+ ctrl0 = ADAU1977_SAI_CTRL0_SAI_TDM_2;
+ break;
+ case 4:
+ ctrl0 = ADAU1977_SAI_CTRL0_SAI_TDM_4;
+ break;
+ case 8:
+ ctrl0 = ADAU1977_SAI_CTRL0_SAI_TDM_8;
+ break;
+ case 16:
+ ctrl0 = ADAU1977_SAI_CTRL0_SAI_TDM_16;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ ret = regmap_update_bits(adau1977->regmap, ADAU1977_REG_SAI_OVERTEMP,
+ ADAU1977_SAI_OVERTEMP_DRV_C(0) |
+ ADAU1977_SAI_OVERTEMP_DRV_C(1) |
+ ADAU1977_SAI_OVERTEMP_DRV_C(2) |
+ ADAU1977_SAI_OVERTEMP_DRV_C(3), drv);
+ if (ret)
+ return ret;
+
+ ret = regmap_write(adau1977->regmap, ADAU1977_REG_CMAP12,
+ (slot[1] << ADAU1977_CHAN_MAP_SECOND_SLOT_OFFSET) |
+ (slot[0] << ADAU1977_CHAN_MAP_FIRST_SLOT_OFFSET));
+ if (ret)
+ return ret;
+
+ ret = regmap_write(adau1977->regmap, ADAU1977_REG_CMAP34,
+ (slot[3] << ADAU1977_CHAN_MAP_SECOND_SLOT_OFFSET) |
+ (slot[2] << ADAU1977_CHAN_MAP_FIRST_SLOT_OFFSET));
+ if (ret)
+ return ret;
+
+ ret = regmap_update_bits(adau1977->regmap, ADAU1977_REG_SAI_CTRL0,
+ ADAU1977_SAI_CTRL0_SAI_MASK, ctrl0);
+ if (ret)
+ return ret;
+
+ ret = regmap_update_bits(adau1977->regmap, ADAU1977_REG_SAI_CTRL1,
+ ADAU1977_SAI_CTRL1_SLOT_WIDTH_MASK, ctrl1);
+ if (ret)
+ return ret;
+
+ adau1977->slot_width = width;
+
+ /* In master mode the maximum bitclock is 24.576 MHz */
+ adau1977->max_master_fs = min(192000, 24576000 / width / slots);
+
+ return 0;
+}
+
+static int adau1977_mute(struct snd_soc_dai *dai, int mute, int stream)
+{
+ struct adau1977 *adau1977 = snd_soc_codec_get_drvdata(dai->codec);
+ unsigned int val;
+
+ if (mute)
+ val = ADAU1977_MISC_CONTROL_MMUTE;
+ else
+ val = 0;
+
+ return regmap_update_bits(adau1977->regmap, ADAU1977_REG_MISC_CONTROL,
+ ADAU1977_MISC_CONTROL_MMUTE, val);
+}
+
+static int adau1977_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt)
+{
+ struct adau1977 *adau1977 = snd_soc_codec_get_drvdata(dai->codec);
+ unsigned int ctrl0 = 0, ctrl1 = 0, block_power = 0;
+ bool invert_lrclk;
+ int ret;
+
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBS_CFS:
+ adau1977->master = false;
+ break;
+ case SND_SOC_DAIFMT_CBM_CFM:
+ ctrl1 |= ADAU1977_SAI_CTRL1_MASTER;
+ adau1977->master = true;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_NB_NF:
+ invert_lrclk = false;
+ break;
+ case SND_SOC_DAIFMT_IB_NF:
+ block_power |= ADAU1977_BLOCK_POWER_SAI_BCLK_EDGE;
+ invert_lrclk = false;
+ break;
+ case SND_SOC_DAIFMT_NB_IF:
+ invert_lrclk = true;
+ break;
+ case SND_SOC_DAIFMT_IB_IF:
+ block_power |= ADAU1977_BLOCK_POWER_SAI_BCLK_EDGE;
+ invert_lrclk = true;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ adau1977->right_j = false;
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ ctrl0 |= ADAU1977_SAI_CTRL0_FMT_I2S;
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ ctrl0 |= ADAU1977_SAI_CTRL0_FMT_LJ;
+ invert_lrclk = !invert_lrclk;
+ break;
+ case SND_SOC_DAIFMT_RIGHT_J:
+ ctrl0 |= ADAU1977_SAI_CTRL0_FMT_RJ_24BIT;
+ adau1977->right_j = true;
+ invert_lrclk = !invert_lrclk;
+ break;
+ case SND_SOC_DAIFMT_DSP_A:
+ ctrl1 |= ADAU1977_SAI_CTRL1_LRCLK_PULSE;
+ ctrl0 |= ADAU1977_SAI_CTRL0_FMT_I2S;
+ invert_lrclk = false;
+ break;
+ case SND_SOC_DAIFMT_DSP_B:
+ ctrl1 |= ADAU1977_SAI_CTRL1_LRCLK_PULSE;
+ ctrl0 |= ADAU1977_SAI_CTRL0_FMT_LJ;
+ invert_lrclk = false;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ if (invert_lrclk)
+ block_power |= ADAU1977_BLOCK_POWER_SAI_LR_POL;
+
+ ret = regmap_update_bits(adau1977->regmap, ADAU1977_REG_BLOCK_POWER_SAI,
+ ADAU1977_BLOCK_POWER_SAI_LR_POL |
+ ADAU1977_BLOCK_POWER_SAI_BCLK_EDGE, block_power);
+ if (ret)
+ return ret;
+
+ ret = regmap_update_bits(adau1977->regmap, ADAU1977_REG_SAI_CTRL0,
+ ADAU1977_SAI_CTRL0_FMT_MASK,
+ ctrl0);
+ if (ret)
+ return ret;
+
+ return regmap_update_bits(adau1977->regmap, ADAU1977_REG_SAI_CTRL1,
+ ADAU1977_SAI_CTRL1_MASTER | ADAU1977_SAI_CTRL1_LRCLK_PULSE,
+ ctrl1);
+}
+
+static int adau1977_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct adau1977 *adau1977 = snd_soc_codec_get_drvdata(dai->codec);
+ u64 formats = 0;
+
+ if (adau1977->slot_width == 16)
+ formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE;
+ else if (adau1977->right_j || adau1977->slot_width == 24)
+ formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE |
+ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S24_BE;
+
+ snd_pcm_hw_constraint_list(substream->runtime, 0,
+ SNDRV_PCM_HW_PARAM_RATE, &adau1977->constraints);
+
+ if (adau1977->master)
+ snd_pcm_hw_constraint_minmax(substream->runtime,
+ SNDRV_PCM_HW_PARAM_RATE, 8000, adau1977->max_master_fs);
+
+ if (formats != 0)
+ snd_pcm_hw_constraint_mask64(substream->runtime,
+ SNDRV_PCM_HW_PARAM_FORMAT, formats);
+
+ return 0;
+}
+
+static int adau1977_set_tristate(struct snd_soc_dai *dai, int tristate)
+{
+ struct adau1977 *adau1977 = snd_soc_codec_get_drvdata(dai->codec);
+ unsigned int val;
+
+ if (tristate)
+ val = ADAU1977_SAI_OVERTEMP_DRV_HIZ;
+ else
+ val = 0;
+
+ return regmap_update_bits(adau1977->regmap, ADAU1977_REG_SAI_OVERTEMP,
+ ADAU1977_SAI_OVERTEMP_DRV_HIZ, val);
+}
+
+static const struct snd_soc_dai_ops adau1977_dai_ops = {
+ .startup = adau1977_startup,
+ .hw_params = adau1977_hw_params,
+ .mute_stream = adau1977_mute,
+ .set_fmt = adau1977_set_dai_fmt,
+ .set_tdm_slot = adau1977_set_tdm_slot,
+ .set_tristate = adau1977_set_tristate,
+};
+
+static struct snd_soc_dai_driver adau1977_dai = {
+ .name = "adau1977-hifi",
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 1,
+ .channels_max = 4,
+ .rates = SNDRV_PCM_RATE_KNOT,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE |
+ SNDRV_PCM_FMTBIT_S32_LE,
+ .sig_bits = 24,
+ },
+ .ops = &adau1977_dai_ops,
+};
+
+static const unsigned int adau1977_rates[] = {
+ 8000, 16000, 32000, 64000, 128000,
+ 11025, 22050, 44100, 88200, 172400,
+ 12000, 24000, 48000, 96000, 192000,
+};
+
+#define ADAU1977_RATE_CONSTRAINT_MASK_32000 0x001f
+#define ADAU1977_RATE_CONSTRAINT_MASK_44100 0x03e0
+#define ADAU1977_RATE_CONSTRAINT_MASK_48000 0x7c00
+/* All rates >= 32000 */
+#define ADAU1977_RATE_CONSTRAINT_MASK_LRCLK 0x739c
+
+static bool adau1977_check_sysclk(unsigned int mclk, unsigned int base_freq)
+{
+ unsigned int mcs;
+
+ if (mclk % (base_freq * 128) != 0)
+ return false;
+
+ mcs = mclk / (128 * base_freq);
+ if (mcs < 1 || mcs > 6 || mcs == 5)
+ return false;
+
+ return true;
+}
+
+static int adau1977_set_sysclk(struct snd_soc_codec *codec,
+ int clk_id, int source, unsigned int freq, int dir)
+{
+ struct adau1977 *adau1977 = snd_soc_codec_get_drvdata(codec);
+ unsigned int mask = 0;
+ unsigned int clk_src;
+ unsigned int ret;
+
+ if (dir != SND_SOC_CLOCK_IN)
+ return -EINVAL;
+
+ if (clk_id != ADAU1977_SYSCLK)
+ return -EINVAL;
+
+ switch (source) {
+ case ADAU1977_SYSCLK_SRC_MCLK:
+ clk_src = 0;
+ break;
+ case ADAU1977_SYSCLK_SRC_LRCLK:
+ clk_src = ADAU1977_PLL_CLK_S;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ if (freq != 0 && source == ADAU1977_SYSCLK_SRC_MCLK) {
+ if (freq < 4000000 || freq > 36864000)
+ return -EINVAL;
+
+ if (adau1977_check_sysclk(freq, 32000))
+ mask |= ADAU1977_RATE_CONSTRAINT_MASK_32000;
+ if (adau1977_check_sysclk(freq, 44100))
+ mask |= ADAU1977_RATE_CONSTRAINT_MASK_44100;
+ if (adau1977_check_sysclk(freq, 48000))
+ mask |= ADAU1977_RATE_CONSTRAINT_MASK_48000;
+
+ if (mask == 0)
+ return -EINVAL;
+ } else if (source == ADAU1977_SYSCLK_SRC_LRCLK) {
+ mask = ADAU1977_RATE_CONSTRAINT_MASK_LRCLK;
+ }
+
+ ret = regmap_update_bits(adau1977->regmap, ADAU1977_REG_PLL,
+ ADAU1977_PLL_CLK_S, clk_src);
+ if (ret)
+ return ret;
+
+ adau1977->constraints.mask = mask;
+ adau1977->sysclk_src = source;
+ adau1977->sysclk = freq;
+
+ return 0;
+}
+
+static int adau1977_codec_probe(struct snd_soc_codec *codec)
+{
+ struct adau1977 *adau1977 = snd_soc_codec_get_drvdata(codec);
+ int ret;
+
+ switch (adau1977->type) {
+ case ADAU1977:
+ ret = snd_soc_dapm_new_controls(&codec->dapm,
+ adau1977_micbias_dapm_widgets,
+ ARRAY_SIZE(adau1977_micbias_dapm_widgets));
+ if (ret < 0)
+ return ret;
+ break;
+ default:
+ break;
+ }
+
+ return 0;
+}
+
+static struct snd_soc_codec_driver adau1977_codec_driver = {
+ .probe = adau1977_codec_probe,
+ .set_bias_level = adau1977_set_bias_level,
+ .set_sysclk = adau1977_set_sysclk,
+ .idle_bias_off = true,
+
+ .controls = adau1977_snd_controls,
+ .num_controls = ARRAY_SIZE(adau1977_snd_controls),
+ .dapm_widgets = adau1977_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(adau1977_dapm_widgets),
+ .dapm_routes = adau1977_dapm_routes,
+ .num_dapm_routes = ARRAY_SIZE(adau1977_dapm_routes),
+};
+
+static int adau1977_setup_micbias(struct adau1977 *adau1977)
+{
+ struct adau1977_platform_data *pdata = adau1977->dev->platform_data;
+ unsigned int micbias;
+
+ if (pdata) {
+ micbias = pdata->micbias;
+ if (micbias > ADAU1977_MICBIAS_9V0)
+ return -EINVAL;
+
+ } else {
+ micbias = ADAU1977_MICBIAS_8V5;
+ }
+
+ return regmap_update_bits(adau1977->regmap, ADAU1977_REG_MICBIAS,
+ ADAU1977_MICBIAS_MB_VOLTS_MASK,
+ micbias << ADAU1977_MICBIAS_MB_VOLTS_OFFSET);
+}
+
+int adau1977_probe(struct device *dev, struct regmap *regmap,
+ enum adau1977_type type, void (*switch_mode)(struct device *dev))
+{
+ unsigned int power_off_mask;
+ struct adau1977 *adau1977;
+ int ret;
+
+ if (IS_ERR(regmap))
+ return PTR_ERR(regmap);
+
+ adau1977 = devm_kzalloc(dev, sizeof(*adau1977), GFP_KERNEL);
+ if (adau1977 == NULL)
+ return -ENOMEM;
+
+ adau1977->dev = dev;
+ adau1977->type = type;
+ adau1977->regmap = regmap;
+ adau1977->switch_mode = switch_mode;
+ adau1977->max_master_fs = 192000;
+
+ adau1977->constraints.list = adau1977_rates;
+ adau1977->constraints.count = ARRAY_SIZE(adau1977_rates);
+
+ adau1977->avdd_reg = devm_regulator_get(dev, "AVDD");
+ if (IS_ERR(adau1977->avdd_reg))
+ return PTR_ERR(adau1977->avdd_reg);
+
+ adau1977->dvdd_reg = devm_regulator_get_optional(dev, "DVDD");
+ if (IS_ERR(adau1977->dvdd_reg)) {
+ if (PTR_ERR(adau1977->dvdd_reg) != -ENODEV)
+ return PTR_ERR(adau1977->dvdd_reg);
+ adau1977->dvdd_reg = NULL;
+ }
+
+ adau1977->reset_gpio = devm_gpiod_get(dev, "reset");
+ if (IS_ERR(adau1977->reset_gpio)) {
+ ret = PTR_ERR(adau1977->reset_gpio);
+ if (ret != -ENOENT && ret != -ENOSYS)
+ return PTR_ERR(adau1977->reset_gpio);
+ adau1977->reset_gpio = NULL;
+ }
+
+ dev_set_drvdata(dev, adau1977);
+
+ if (adau1977->reset_gpio) {
+ ret = gpiod_direction_output(adau1977->reset_gpio, 0);
+ if (ret)
+ return ret;
+ ndelay(100);
+ }
+
+ ret = adau1977_power_enable(adau1977);
+ if (ret)
+ return ret;
+
+ if (type == ADAU1977) {
+ ret = adau1977_setup_micbias(adau1977);
+ if (ret)
+ goto err_poweroff;
+ }
+
+ if (adau1977->dvdd_reg)
+ power_off_mask = ~0;
+ else
+ power_off_mask = ~ADAU1977_BLOCK_POWER_SAI_LDO_EN;
+
+ ret = regmap_update_bits(adau1977->regmap, ADAU1977_REG_BLOCK_POWER_SAI,
+ power_off_mask, 0x00);
+ if (ret)
+ goto err_poweroff;
+
+ ret = adau1977_power_disable(adau1977);
+ if (ret)
+ return ret;
+
+ return snd_soc_register_codec(dev, &adau1977_codec_driver,
+ &adau1977_dai, 1);
+
+err_poweroff:
+ adau1977_power_disable(adau1977);
+ return ret;
+
+}
+EXPORT_SYMBOL_GPL(adau1977_probe);
+
+static bool adau1977_register_volatile(struct device *dev, unsigned int reg)
+{
+ switch (reg) {
+ case ADAU1977_REG_STATUS(0):
+ case ADAU1977_REG_STATUS(1):
+ case ADAU1977_REG_STATUS(2):
+ case ADAU1977_REG_STATUS(3):
+ case ADAU1977_REG_ADC_CLIP:
+ return true;
+ }
+
+ return false;
+}
+
+const struct regmap_config adau1977_regmap_config = {
+ .max_register = ADAU1977_REG_DC_HPF_CAL,
+ .volatile_reg = adau1977_register_volatile,
+
+ .cache_type = REGCACHE_RBTREE,
+ .reg_defaults = adau1977_reg_defaults,
+ .num_reg_defaults = ARRAY_SIZE(adau1977_reg_defaults),
+};
+EXPORT_SYMBOL_GPL(adau1977_regmap_config);
+
+MODULE_DESCRIPTION("ASoC ADAU1977/ADAU1978/ADAU1979 driver");
+MODULE_AUTHOR("Lars-Peter Clausen <lars@metafoo.de>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/adau1977.h b/sound/soc/codecs/adau1977.h
new file mode 100644
index 0000000..95e7143
--- /dev/null
+++ b/sound/soc/codecs/adau1977.h
@@ -0,0 +1,37 @@
+/*
+ * ADAU1977/ADAU1978/ADAU1979 driver
+ *
+ * Copyright 2014 Analog Devices Inc.
+ * Author: Lars-Peter Clausen <lars@metafoo.de>
+ *
+ * Licensed under the GPL-2.
+ */
+
+#ifndef __SOUND_SOC_CODECS_ADAU1977_H__
+#define __SOUND_SOC_CODECS_ADAU1977_H__
+
+#include <linux/regmap.h>
+
+struct device;
+
+enum adau1977_type {
+ ADAU1977,
+ ADAU1978,
+ ADAU1979,
+};
+
+int adau1977_probe(struct device *dev, struct regmap *regmap,
+ enum adau1977_type type, void (*switch_mode)(struct device *dev));
+
+extern const struct regmap_config adau1977_regmap_config;
+
+enum adau1977_clk_id {
+ ADAU1977_SYSCLK,
+};
+
+enum adau1977_sysclk_src {
+ ADAU1977_SYSCLK_SRC_MCLK,
+ ADAU1977_SYSCLK_SRC_LRCLK,
+};
+
+#endif
diff --git a/sound/soc/codecs/adav801.c b/sound/soc/codecs/adav801.c
new file mode 100644
index 0000000..790fce3
--- /dev/null
+++ b/sound/soc/codecs/adav801.c
@@ -0,0 +1,53 @@
+/*
+ * ADAV801 audio driver
+ *
+ * Copyright 2014 Analog Devices Inc.
+ *
+ * Licensed under the GPL-2.
+ */
+
+#include <linux/module.h>
+#include <linux/spi/spi.h>
+#include <linux/regmap.h>
+
+#include <sound/soc.h>
+
+#include "adav80x.h"
+
+static const struct spi_device_id adav80x_spi_id[] = {
+ { "adav801", 0 },
+ { }
+};
+MODULE_DEVICE_TABLE(spi, adav80x_spi_id);
+
+static int adav80x_spi_probe(struct spi_device *spi)
+{
+ struct regmap_config config;
+
+ config = adav80x_regmap_config;
+ config.read_flag_mask = 0x01;
+
+ return adav80x_bus_probe(&spi->dev, devm_regmap_init_spi(spi, &config));
+}
+
+static int adav80x_spi_remove(struct spi_device *spi)
+{
+ snd_soc_unregister_codec(&spi->dev);
+ return 0;
+}
+
+static struct spi_driver adav80x_spi_driver = {
+ .driver = {
+ .name = "adav801",
+ .owner = THIS_MODULE,
+ },
+ .probe = adav80x_spi_probe,
+ .remove = adav80x_spi_remove,
+ .id_table = adav80x_spi_id,
+};
+module_spi_driver(adav80x_spi_driver);
+
+MODULE_DESCRIPTION("ASoC ADAV801 driver");
+MODULE_AUTHOR("Lars-Peter Clausen <lars@metafoo.de>");
+MODULE_AUTHOR("Yi Li <yi.li@analog.com>>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/adav803.c b/sound/soc/codecs/adav803.c
new file mode 100644
index 0000000..66d9fce
--- /dev/null
+++ b/sound/soc/codecs/adav803.c
@@ -0,0 +1,50 @@
+/*
+ * ADAV803 audio driver
+ *
+ * Copyright 2014 Analog Devices Inc.
+ *
+ * Licensed under the GPL-2.
+ */
+
+#include <linux/module.h>
+#include <linux/i2c.h>
+#include <linux/regmap.h>
+
+#include <sound/soc.h>
+
+#include "adav80x.h"
+
+static const struct i2c_device_id adav803_id[] = {
+ { "adav803", 0 },
+ { }
+};
+MODULE_DEVICE_TABLE(i2c, adav803_id);
+
+static int adav803_probe(struct i2c_client *client,
+ const struct i2c_device_id *id)
+{
+ return adav80x_bus_probe(&client->dev,
+ devm_regmap_init_i2c(client, &adav80x_regmap_config));
+}
+
+static int adav803_remove(struct i2c_client *client)
+{
+ snd_soc_unregister_codec(&client->dev);
+ return 0;
+}
+
+static struct i2c_driver adav803_driver = {
+ .driver = {
+ .name = "adav803",
+ .owner = THIS_MODULE,
+ },
+ .probe = adav803_probe,
+ .remove = adav803_remove,
+ .id_table = adav803_id,
+};
+module_i2c_driver(adav803_driver);
+
+MODULE_DESCRIPTION("ASoC ADAV803 driver");
+MODULE_AUTHOR("Lars-Peter Clausen <lars@metafoo.de>");
+MODULE_AUTHOR("Yi Li <yi.li@analog.com>>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/adav80x.c b/sound/soc/codecs/adav80x.c
index f78b27a..5062e34 100644
--- a/sound/soc/codecs/adav80x.c
+++ b/sound/soc/codecs/adav80x.c
@@ -8,17 +8,15 @@
* Licensed under the GPL-2 or later.
*/
-#include <linux/init.h>
#include <linux/module.h>
#include <linux/kernel.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>
#include <sound/pcm_params.h>
-#include <sound/tlv.h>
#include <sound/soc.h>
+#include <sound/tlv.h>
#include "adav80x.h"
@@ -541,6 +539,7 @@ static int adav80x_set_sysclk(struct snd_soc_codec *codec,
unsigned int freq, int dir)
{
struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec);
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
if (dir == SND_SOC_CLOCK_IN) {
switch (clk_id) {
@@ -573,7 +572,7 @@ static int adav80x_set_sysclk(struct snd_soc_codec *codec,
regmap_write(adav80x->regmap, ADAV80X_ICLK_CTRL2,
iclk_ctrl2);
- snd_soc_dapm_sync(&codec->dapm);
+ snd_soc_dapm_sync(dapm);
}
} else {
unsigned int mask;
@@ -600,17 +599,21 @@ static int adav80x_set_sysclk(struct snd_soc_codec *codec,
adav80x->sysclk_pd[clk_id] = false;
}
+ snd_soc_dapm_mutex_lock(dapm);
+
if (adav80x->sysclk_pd[0])
- snd_soc_dapm_disable_pin(&codec->dapm, "PLL1");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "PLL1");
else
- snd_soc_dapm_force_enable_pin(&codec->dapm, "PLL1");
+ snd_soc_dapm_force_enable_pin_unlocked(dapm, "PLL1");
if (adav80x->sysclk_pd[1] || adav80x->sysclk_pd[2])
- snd_soc_dapm_disable_pin(&codec->dapm, "PLL2");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "PLL2");
else
- snd_soc_dapm_force_enable_pin(&codec->dapm, "PLL2");
+ snd_soc_dapm_force_enable_pin_unlocked(dapm, "PLL2");
- snd_soc_dapm_sync(&codec->dapm);
+ snd_soc_dapm_sync_unlocked(dapm);
+
+ snd_soc_dapm_mutex_unlock(dapm);
}
return 0;
@@ -722,7 +725,7 @@ static int adav80x_dai_startup(struct snd_pcm_substream *substream,
struct snd_soc_codec *codec = dai->codec;
struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec);
- if (!codec->active || !adav80x->rate)
+ if (!snd_soc_codec_is_active(codec) || !adav80x->rate)
return 0;
return snd_pcm_hw_constraint_minmax(substream->runtime,
@@ -735,7 +738,7 @@ static void adav80x_dai_shutdown(struct snd_pcm_substream *substream,
struct snd_soc_codec *codec = dai->codec;
struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec);
- if (!codec->active)
+ if (!snd_soc_codec_is_active(codec))
adav80x->rate = 0;
}
@@ -798,15 +801,8 @@ static struct snd_soc_dai_driver adav80x_dais[] = {
static int adav80x_probe(struct snd_soc_codec *codec)
{
- int ret;
struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec);
- ret = snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP);
- if (ret) {
- dev_err(codec->dev, "failed to set cache I/O: %d\n", ret);
- return ret;
- }
-
/* Force PLLs on for SYSCLK output */
snd_soc_dapm_force_enable_pin(&codec->dapm, "PLL1");
snd_soc_dapm_force_enable_pin(&codec->dapm, "PLL2");
@@ -864,39 +860,26 @@ static struct snd_soc_codec_driver adav80x_codec_driver = {
.num_dapm_routes = ARRAY_SIZE(adav80x_dapm_routes),
};
-static int adav80x_bus_probe(struct device *dev, struct regmap *regmap)
+int adav80x_bus_probe(struct device *dev, struct regmap *regmap)
{
struct adav80x *adav80x;
- int ret;
if (IS_ERR(regmap))
return PTR_ERR(regmap);
- adav80x = kzalloc(sizeof(*adav80x), GFP_KERNEL);
+ adav80x = devm_kzalloc(dev, sizeof(*adav80x), GFP_KERNEL);
if (!adav80x)
return -ENOMEM;
-
dev_set_drvdata(dev, adav80x);
adav80x->regmap = regmap;
- ret = snd_soc_register_codec(dev, &adav80x_codec_driver,
+ return snd_soc_register_codec(dev, &adav80x_codec_driver,
adav80x_dais, ARRAY_SIZE(adav80x_dais));
- if (ret)
- kfree(adav80x);
-
- return ret;
-}
-
-static int adav80x_bus_remove(struct device *dev)
-{
- snd_soc_unregister_codec(dev);
- kfree(dev_get_drvdata(dev));
- return 0;
}
+EXPORT_SYMBOL_GPL(adav80x_bus_probe);
-#if defined(CONFIG_SPI_MASTER)
-static const struct regmap_config adav80x_spi_regmap_config = {
+const struct regmap_config adav80x_regmap_config = {
.val_bits = 8,
.pad_bits = 1,
.reg_bits = 7,
@@ -908,105 +891,7 @@ static const struct regmap_config adav80x_spi_regmap_config = {
.reg_defaults = adav80x_reg_defaults,
.num_reg_defaults = ARRAY_SIZE(adav80x_reg_defaults),
};
-
-static const struct spi_device_id adav80x_spi_id[] = {
- { "adav801", 0 },
- { }
-};
-MODULE_DEVICE_TABLE(spi, adav80x_spi_id);
-
-static int adav80x_spi_probe(struct spi_device *spi)
-{
- return adav80x_bus_probe(&spi->dev,
- devm_regmap_init_spi(spi, &adav80x_spi_regmap_config));
-}
-
-static int adav80x_spi_remove(struct spi_device *spi)
-{
- return adav80x_bus_remove(&spi->dev);
-}
-
-static struct spi_driver adav80x_spi_driver = {
- .driver = {
- .name = "adav801",
- .owner = THIS_MODULE,
- },
- .probe = adav80x_spi_probe,
- .remove = adav80x_spi_remove,
- .id_table = adav80x_spi_id,
-};
-#endif
-
-#if IS_ENABLED(CONFIG_I2C)
-static const struct regmap_config adav80x_i2c_regmap_config = {
- .val_bits = 8,
- .pad_bits = 1,
- .reg_bits = 7,
-
- .max_register = ADAV80X_PLL_OUTE,
-
- .cache_type = REGCACHE_RBTREE,
- .reg_defaults = adav80x_reg_defaults,
- .num_reg_defaults = ARRAY_SIZE(adav80x_reg_defaults),
-};
-
-static const struct i2c_device_id adav80x_i2c_id[] = {
- { "adav803", 0 },
- { }
-};
-MODULE_DEVICE_TABLE(i2c, adav80x_i2c_id);
-
-static int adav80x_i2c_probe(struct i2c_client *client,
- const struct i2c_device_id *id)
-{
- return adav80x_bus_probe(&client->dev,
- devm_regmap_init_i2c(client, &adav80x_i2c_regmap_config));
-}
-
-static int adav80x_i2c_remove(struct i2c_client *client)
-{
- return adav80x_bus_remove(&client->dev);
-}
-
-static struct i2c_driver adav80x_i2c_driver = {
- .driver = {
- .name = "adav803",
- .owner = THIS_MODULE,
- },
- .probe = adav80x_i2c_probe,
- .remove = adav80x_i2c_remove,
- .id_table = adav80x_i2c_id,
-};
-#endif
-
-static int __init adav80x_init(void)
-{
- int ret = 0;
-
-#if IS_ENABLED(CONFIG_I2C)
- ret = i2c_add_driver(&adav80x_i2c_driver);
- if (ret)
- return ret;
-#endif
-
-#if defined(CONFIG_SPI_MASTER)
- ret = spi_register_driver(&adav80x_spi_driver);
-#endif
-
- return ret;
-}
-module_init(adav80x_init);
-
-static void __exit adav80x_exit(void)
-{
-#if IS_ENABLED(CONFIG_I2C)
- i2c_del_driver(&adav80x_i2c_driver);
-#endif
-#if defined(CONFIG_SPI_MASTER)
- spi_unregister_driver(&adav80x_spi_driver);
-#endif
-}
-module_exit(adav80x_exit);
+EXPORT_SYMBOL_GPL(adav80x_regmap_config);
MODULE_DESCRIPTION("ASoC ADAV80x driver");
MODULE_AUTHOR("Lars-Peter Clausen <lars@metafoo.de>");
diff --git a/sound/soc/codecs/adav80x.h b/sound/soc/codecs/adav80x.h
index adb0fc7..8a1d7c0 100644
--- a/sound/soc/codecs/adav80x.h
+++ b/sound/soc/codecs/adav80x.h
@@ -9,6 +9,13 @@
#ifndef _ADAV80X_H
#define _ADAV80X_H
+#include <linux/regmap.h>
+
+struct device;
+
+extern const struct regmap_config adav80x_regmap_config;
+int adav80x_bus_probe(struct device *dev, struct regmap *regmap);
+
enum adav80x_pll_src {
ADAV80X_PLL_SRC_XIN,
ADAV80X_PLL_SRC_XTAL,
diff --git a/sound/soc/codecs/ak4104.c b/sound/soc/codecs/ak4104.c
index b4819dc..10adf25 100644
--- a/sound/soc/codecs/ak4104.c
+++ b/sound/soc/codecs/ak4104.c
@@ -174,8 +174,6 @@ static int ak4104_probe(struct snd_soc_codec *codec)
struct ak4104_private *ak4104 = snd_soc_codec_get_drvdata(codec);
int ret;
- codec->control_data = ak4104->regmap;
-
/* set power-up and non-reset bits */
ret = regmap_update_bits(ak4104->regmap, AK4104_REG_CONTROL1,
AK4104_CONTROL1_PW | AK4104_CONTROL1_RSTN,
diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c
index 684fe91..30e2978 100644
--- a/sound/soc/codecs/ak4535.c
+++ b/sound/soc/codecs/ak4535.c
@@ -388,15 +388,6 @@ static int ak4535_resume(struct snd_soc_codec *codec)
static int ak4535_probe(struct snd_soc_codec *codec)
{
- struct ak4535_priv *ak4535 = snd_soc_codec_get_drvdata(codec);
- int ret;
-
- codec->control_data = ak4535->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;
- }
/* power on device */
ak4535_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
diff --git a/sound/soc/codecs/ak4641.c b/sound/soc/codecs/ak4641.c
index 94cbe50..868c0e2 100644
--- a/sound/soc/codecs/ak4641.c
+++ b/sound/soc/codecs/ak4641.c
@@ -113,14 +113,14 @@ static const DECLARE_TLV_DB_SCALE(alc_tlv, -800, 50, 0);
static const DECLARE_TLV_DB_SCALE(aux_in_tlv, -2100, 300, 0);
-static const struct soc_enum ak4641_mono_out_enum =
- SOC_ENUM_SINGLE(AK4641_SIG1, 6, 2, ak4641_mono_out);
-static const struct soc_enum ak4641_hp_out_enum =
- SOC_ENUM_SINGLE(AK4641_MODE2, 2, 2, ak4641_hp_out);
-static const struct soc_enum ak4641_mic_select_enum =
- SOC_ENUM_SINGLE(AK4641_MIC, 1, 2, ak4641_mic_select);
-static const struct soc_enum ak4641_mic_or_dac_enum =
- SOC_ENUM_SINGLE(AK4641_BTIF, 4, 2, ak4641_mic_or_dac);
+static SOC_ENUM_SINGLE_DECL(ak4641_mono_out_enum,
+ AK4641_SIG1, 6, ak4641_mono_out);
+static SOC_ENUM_SINGLE_DECL(ak4641_hp_out_enum,
+ AK4641_MODE2, 2, ak4641_hp_out);
+static SOC_ENUM_SINGLE_DECL(ak4641_mic_select_enum,
+ AK4641_MIC, 1, ak4641_mic_select);
+static SOC_ENUM_SINGLE_DECL(ak4641_mic_or_dac_enum,
+ AK4641_BTIF, 4, ak4641_mic_or_dac);
static const struct snd_kcontrol_new ak4641_snd_controls[] = {
SOC_ENUM("Mono 1 Output", ak4641_mono_out_enum),
@@ -519,14 +519,6 @@ static int ak4641_resume(struct snd_soc_codec *codec)
static int ak4641_probe(struct snd_soc_codec *codec)
{
- int ret;
-
- 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;
- }
-
/* power on device */
ak4641_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c
index 1f646c6..92655cc 100644
--- a/sound/soc/codecs/ak4642.c
+++ b/sound/soc/codecs/ak4642.c
@@ -465,14 +465,6 @@ static int ak4642_resume(struct snd_soc_codec *codec)
static int ak4642_probe(struct snd_soc_codec *codec)
{
- int ret;
-
- 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;
- }
-
ak4642_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
return 0;
diff --git a/sound/soc/codecs/ak4671.c b/sound/soc/codecs/ak4671.c
index 25bdf6a..998fa0c 100644
--- a/sound/soc/codecs/ak4671.c
+++ b/sound/soc/codecs/ak4671.c
@@ -15,6 +15,7 @@
#include <linux/init.h>
#include <linux/i2c.h>
#include <linux/delay.h>
+#include <linux/regmap.h>
#include <linux/slab.h>
#include <sound/soc.h>
#include <sound/initval.h>
@@ -23,104 +24,99 @@
#include "ak4671.h"
-/* codec private data */
-struct ak4671_priv {
- enum snd_soc_control_type control_type;
-};
-
/* ak4671 register cache & default register settings */
-static const u8 ak4671_reg[AK4671_CACHEREGNUM] = {
- 0x00, /* AK4671_AD_DA_POWER_MANAGEMENT (0x00) */
- 0xf6, /* AK4671_PLL_MODE_SELECT0 (0x01) */
- 0x00, /* AK4671_PLL_MODE_SELECT1 (0x02) */
- 0x02, /* AK4671_FORMAT_SELECT (0x03) */
- 0x00, /* AK4671_MIC_SIGNAL_SELECT (0x04) */
- 0x55, /* AK4671_MIC_AMP_GAIN (0x05) */
- 0x00, /* AK4671_MIXING_POWER_MANAGEMENT0 (0x06) */
- 0x00, /* AK4671_MIXING_POWER_MANAGEMENT1 (0x07) */
- 0xb5, /* AK4671_OUTPUT_VOLUME_CONTROL (0x08) */
- 0x00, /* AK4671_LOUT1_SIGNAL_SELECT (0x09) */
- 0x00, /* AK4671_ROUT1_SIGNAL_SELECT (0x0a) */
- 0x00, /* AK4671_LOUT2_SIGNAL_SELECT (0x0b) */
- 0x00, /* AK4671_ROUT2_SIGNAL_SELECT (0x0c) */
- 0x00, /* AK4671_LOUT3_SIGNAL_SELECT (0x0d) */
- 0x00, /* AK4671_ROUT3_SIGNAL_SELECT (0x0e) */
- 0x00, /* AK4671_LOUT1_POWER_MANAGERMENT (0x0f) */
- 0x00, /* AK4671_LOUT2_POWER_MANAGERMENT (0x10) */
- 0x80, /* AK4671_LOUT3_POWER_MANAGERMENT (0x11) */
- 0x91, /* AK4671_LCH_INPUT_VOLUME_CONTROL (0x12) */
- 0x91, /* AK4671_RCH_INPUT_VOLUME_CONTROL (0x13) */
- 0xe1, /* AK4671_ALC_REFERENCE_SELECT (0x14) */
- 0x00, /* AK4671_DIGITAL_MIXING_CONTROL (0x15) */
- 0x00, /* AK4671_ALC_TIMER_SELECT (0x16) */
- 0x00, /* AK4671_ALC_MODE_CONTROL (0x17) */
- 0x02, /* AK4671_MODE_CONTROL1 (0x18) */
- 0x01, /* AK4671_MODE_CONTROL2 (0x19) */
- 0x18, /* AK4671_LCH_OUTPUT_VOLUME_CONTROL (0x1a) */
- 0x18, /* AK4671_RCH_OUTPUT_VOLUME_CONTROL (0x1b) */
- 0x00, /* AK4671_SIDETONE_A_CONTROL (0x1c) */
- 0x02, /* AK4671_DIGITAL_FILTER_SELECT (0x1d) */
- 0x00, /* AK4671_FIL3_COEFFICIENT0 (0x1e) */
- 0x00, /* AK4671_FIL3_COEFFICIENT1 (0x1f) */
- 0x00, /* AK4671_FIL3_COEFFICIENT2 (0x20) */
- 0x00, /* AK4671_FIL3_COEFFICIENT3 (0x21) */
- 0x00, /* AK4671_EQ_COEFFICIENT0 (0x22) */
- 0x00, /* AK4671_EQ_COEFFICIENT1 (0x23) */
- 0x00, /* AK4671_EQ_COEFFICIENT2 (0x24) */
- 0x00, /* AK4671_EQ_COEFFICIENT3 (0x25) */
- 0x00, /* AK4671_EQ_COEFFICIENT4 (0x26) */
- 0x00, /* AK4671_EQ_COEFFICIENT5 (0x27) */
- 0xa9, /* AK4671_FIL1_COEFFICIENT0 (0x28) */
- 0x1f, /* AK4671_FIL1_COEFFICIENT1 (0x29) */
- 0xad, /* AK4671_FIL1_COEFFICIENT2 (0x2a) */
- 0x20, /* AK4671_FIL1_COEFFICIENT3 (0x2b) */
- 0x00, /* AK4671_FIL2_COEFFICIENT0 (0x2c) */
- 0x00, /* AK4671_FIL2_COEFFICIENT1 (0x2d) */
- 0x00, /* AK4671_FIL2_COEFFICIENT2 (0x2e) */
- 0x00, /* AK4671_FIL2_COEFFICIENT3 (0x2f) */
- 0x00, /* AK4671_DIGITAL_FILTER_SELECT2 (0x30) */
- 0x00, /* this register not used */
- 0x00, /* AK4671_E1_COEFFICIENT0 (0x32) */
- 0x00, /* AK4671_E1_COEFFICIENT1 (0x33) */
- 0x00, /* AK4671_E1_COEFFICIENT2 (0x34) */
- 0x00, /* AK4671_E1_COEFFICIENT3 (0x35) */
- 0x00, /* AK4671_E1_COEFFICIENT4 (0x36) */
- 0x00, /* AK4671_E1_COEFFICIENT5 (0x37) */
- 0x00, /* AK4671_E2_COEFFICIENT0 (0x38) */
- 0x00, /* AK4671_E2_COEFFICIENT1 (0x39) */
- 0x00, /* AK4671_E2_COEFFICIENT2 (0x3a) */
- 0x00, /* AK4671_E2_COEFFICIENT3 (0x3b) */
- 0x00, /* AK4671_E2_COEFFICIENT4 (0x3c) */
- 0x00, /* AK4671_E2_COEFFICIENT5 (0x3d) */
- 0x00, /* AK4671_E3_COEFFICIENT0 (0x3e) */
- 0x00, /* AK4671_E3_COEFFICIENT1 (0x3f) */
- 0x00, /* AK4671_E3_COEFFICIENT2 (0x40) */
- 0x00, /* AK4671_E3_COEFFICIENT3 (0x41) */
- 0x00, /* AK4671_E3_COEFFICIENT4 (0x42) */
- 0x00, /* AK4671_E3_COEFFICIENT5 (0x43) */
- 0x00, /* AK4671_E4_COEFFICIENT0 (0x44) */
- 0x00, /* AK4671_E4_COEFFICIENT1 (0x45) */
- 0x00, /* AK4671_E4_COEFFICIENT2 (0x46) */
- 0x00, /* AK4671_E4_COEFFICIENT3 (0x47) */
- 0x00, /* AK4671_E4_COEFFICIENT4 (0x48) */
- 0x00, /* AK4671_E4_COEFFICIENT5 (0x49) */
- 0x00, /* AK4671_E5_COEFFICIENT0 (0x4a) */
- 0x00, /* AK4671_E5_COEFFICIENT1 (0x4b) */
- 0x00, /* AK4671_E5_COEFFICIENT2 (0x4c) */
- 0x00, /* AK4671_E5_COEFFICIENT3 (0x4d) */
- 0x00, /* AK4671_E5_COEFFICIENT4 (0x4e) */
- 0x00, /* AK4671_E5_COEFFICIENT5 (0x4f) */
- 0x88, /* AK4671_EQ_CONTROL_250HZ_100HZ (0x50) */
- 0x88, /* AK4671_EQ_CONTROL_3500HZ_1KHZ (0x51) */
- 0x08, /* AK4671_EQ_CONTRO_10KHZ (0x52) */
- 0x00, /* AK4671_PCM_IF_CONTROL0 (0x53) */
- 0x00, /* AK4671_PCM_IF_CONTROL1 (0x54) */
- 0x00, /* AK4671_PCM_IF_CONTROL2 (0x55) */
- 0x18, /* AK4671_DIGITAL_VOLUME_B_CONTROL (0x56) */
- 0x18, /* AK4671_DIGITAL_VOLUME_C_CONTROL (0x57) */
- 0x00, /* AK4671_SIDETONE_VOLUME_CONTROL (0x58) */
- 0x00, /* AK4671_DIGITAL_MIXING_CONTROL2 (0x59) */
- 0x00, /* AK4671_SAR_ADC_CONTROL (0x5a) */
+static const struct reg_default ak4671_reg_defaults[] = {
+ { 0x00, 0x00 }, /* AK4671_AD_DA_POWER_MANAGEMENT (0x00) */
+ { 0x01, 0xf6 }, /* AK4671_PLL_MODE_SELECT0 (0x01) */
+ { 0x02, 0x00 }, /* AK4671_PLL_MODE_SELECT1 (0x02) */
+ { 0x03, 0x02 }, /* AK4671_FORMAT_SELECT (0x03) */
+ { 0x04, 0x00 }, /* AK4671_MIC_SIGNAL_SELECT (0x04) */
+ { 0x05, 0x55 }, /* AK4671_MIC_AMP_GAIN (0x05) */
+ { 0x06, 0x00 }, /* AK4671_MIXING_POWER_MANAGEMENT0 (0x06) */
+ { 0x07, 0x00 }, /* AK4671_MIXING_POWER_MANAGEMENT1 (0x07) */
+ { 0x08, 0xb5 }, /* AK4671_OUTPUT_VOLUME_CONTROL (0x08) */
+ { 0x09, 0x00 }, /* AK4671_LOUT1_SIGNAL_SELECT (0x09) */
+ { 0x0a, 0x00 }, /* AK4671_ROUT1_SIGNAL_SELECT (0x0a) */
+ { 0x0b, 0x00 }, /* AK4671_LOUT2_SIGNAL_SELECT (0x0b) */
+ { 0x0c, 0x00 }, /* AK4671_ROUT2_SIGNAL_SELECT (0x0c) */
+ { 0x0d, 0x00 }, /* AK4671_LOUT3_SIGNAL_SELECT (0x0d) */
+ { 0x0e, 0x00 }, /* AK4671_ROUT3_SIGNAL_SELECT (0x0e) */
+ { 0x0f, 0x00 }, /* AK4671_LOUT1_POWER_MANAGERMENT (0x0f) */
+ { 0x10, 0x00 }, /* AK4671_LOUT2_POWER_MANAGERMENT (0x10) */
+ { 0x11, 0x80 }, /* AK4671_LOUT3_POWER_MANAGERMENT (0x11) */
+ { 0x12, 0x91 }, /* AK4671_LCH_INPUT_VOLUME_CONTROL (0x12) */
+ { 0x13, 0x91 }, /* AK4671_RCH_INPUT_VOLUME_CONTROL (0x13) */
+ { 0x14, 0xe1 }, /* AK4671_ALC_REFERENCE_SELECT (0x14) */
+ { 0x15, 0x00 }, /* AK4671_DIGITAL_MIXING_CONTROL (0x15) */
+ { 0x16, 0x00 }, /* AK4671_ALC_TIMER_SELECT (0x16) */
+ { 0x17, 0x00 }, /* AK4671_ALC_MODE_CONTROL (0x17) */
+ { 0x18, 0x02 }, /* AK4671_MODE_CONTROL1 (0x18) */
+ { 0x19, 0x01 }, /* AK4671_MODE_CONTROL2 (0x19) */
+ { 0x1a, 0x18 }, /* AK4671_LCH_OUTPUT_VOLUME_CONTROL (0x1a) */
+ { 0x1b, 0x18 }, /* AK4671_RCH_OUTPUT_VOLUME_CONTROL (0x1b) */
+ { 0x1c, 0x00 }, /* AK4671_SIDETONE_A_CONTROL (0x1c) */
+ { 0x1d, 0x02 }, /* AK4671_DIGITAL_FILTER_SELECT (0x1d) */
+ { 0x1e, 0x00 }, /* AK4671_FIL3_COEFFICIENT0 (0x1e) */
+ { 0x1f, 0x00 }, /* AK4671_FIL3_COEFFICIENT1 (0x1f) */
+ { 0x20, 0x00 }, /* AK4671_FIL3_COEFFICIENT2 (0x20) */
+ { 0x21, 0x00 }, /* AK4671_FIL3_COEFFICIENT3 (0x21) */
+ { 0x22, 0x00 }, /* AK4671_EQ_COEFFICIENT0 (0x22) */
+ { 0x23, 0x00 }, /* AK4671_EQ_COEFFICIENT1 (0x23) */
+ { 0x24, 0x00 }, /* AK4671_EQ_COEFFICIENT2 (0x24) */
+ { 0x25, 0x00 }, /* AK4671_EQ_COEFFICIENT3 (0x25) */
+ { 0x26, 0x00 }, /* AK4671_EQ_COEFFICIENT4 (0x26) */
+ { 0x27, 0x00 }, /* AK4671_EQ_COEFFICIENT5 (0x27) */
+ { 0x28, 0xa9 }, /* AK4671_FIL1_COEFFICIENT0 (0x28) */
+ { 0x29, 0x1f }, /* AK4671_FIL1_COEFFICIENT1 (0x29) */
+ { 0x2a, 0xad }, /* AK4671_FIL1_COEFFICIENT2 (0x2a) */
+ { 0x2b, 0x20 }, /* AK4671_FIL1_COEFFICIENT3 (0x2b) */
+ { 0x2c, 0x00 }, /* AK4671_FIL2_COEFFICIENT0 (0x2c) */
+ { 0x2d, 0x00 }, /* AK4671_FIL2_COEFFICIENT1 (0x2d) */
+ { 0x2e, 0x00 }, /* AK4671_FIL2_COEFFICIENT2 (0x2e) */
+ { 0x2f, 0x00 }, /* AK4671_FIL2_COEFFICIENT3 (0x2f) */
+ { 0x30, 0x00 }, /* AK4671_DIGITAL_FILTER_SELECT2 (0x30) */
+
+ { 0x32, 0x00 }, /* AK4671_E1_COEFFICIENT0 (0x32) */
+ { 0x33, 0x00 }, /* AK4671_E1_COEFFICIENT1 (0x33) */
+ { 0x34, 0x00 }, /* AK4671_E1_COEFFICIENT2 (0x34) */
+ { 0x35, 0x00 }, /* AK4671_E1_COEFFICIENT3 (0x35) */
+ { 0x36, 0x00 }, /* AK4671_E1_COEFFICIENT4 (0x36) */
+ { 0x37, 0x00 }, /* AK4671_E1_COEFFICIENT5 (0x37) */
+ { 0x38, 0x00 }, /* AK4671_E2_COEFFICIENT0 (0x38) */
+ { 0x39, 0x00 }, /* AK4671_E2_COEFFICIENT1 (0x39) */
+ { 0x3a, 0x00 }, /* AK4671_E2_COEFFICIENT2 (0x3a) */
+ { 0x3b, 0x00 }, /* AK4671_E2_COEFFICIENT3 (0x3b) */
+ { 0x3c, 0x00 }, /* AK4671_E2_COEFFICIENT4 (0x3c) */
+ { 0x3d, 0x00 }, /* AK4671_E2_COEFFICIENT5 (0x3d) */
+ { 0x3e, 0x00 }, /* AK4671_E3_COEFFICIENT0 (0x3e) */
+ { 0x3f, 0x00 }, /* AK4671_E3_COEFFICIENT1 (0x3f) */
+ { 0x40, 0x00 }, /* AK4671_E3_COEFFICIENT2 (0x40) */
+ { 0x41, 0x00 }, /* AK4671_E3_COEFFICIENT3 (0x41) */
+ { 0x42, 0x00 }, /* AK4671_E3_COEFFICIENT4 (0x42) */
+ { 0x43, 0x00 }, /* AK4671_E3_COEFFICIENT5 (0x43) */
+ { 0x44, 0x00 }, /* AK4671_E4_COEFFICIENT0 (0x44) */
+ { 0x45, 0x00 }, /* AK4671_E4_COEFFICIENT1 (0x45) */
+ { 0x46, 0x00 }, /* AK4671_E4_COEFFICIENT2 (0x46) */
+ { 0x47, 0x00 }, /* AK4671_E4_COEFFICIENT3 (0x47) */
+ { 0x48, 0x00 }, /* AK4671_E4_COEFFICIENT4 (0x48) */
+ { 0x49, 0x00 }, /* AK4671_E4_COEFFICIENT5 (0x49) */
+ { 0x4a, 0x00 }, /* AK4671_E5_COEFFICIENT0 (0x4a) */
+ { 0x4b, 0x00 }, /* AK4671_E5_COEFFICIENT1 (0x4b) */
+ { 0x4c, 0x00 }, /* AK4671_E5_COEFFICIENT2 (0x4c) */
+ { 0x4d, 0x00 }, /* AK4671_E5_COEFFICIENT3 (0x4d) */
+ { 0x4e, 0x00 }, /* AK4671_E5_COEFFICIENT4 (0x4e) */
+ { 0x4f, 0x00 }, /* AK4671_E5_COEFFICIENT5 (0x4f) */
+ { 0x50, 0x88 }, /* AK4671_EQ_CONTROL_250HZ_100HZ (0x50) */
+ { 0x51, 0x88 }, /* AK4671_EQ_CONTROL_3500HZ_1KHZ (0x51) */
+ { 0x52, 0x08 }, /* AK4671_EQ_CONTRO_10KHZ (0x52) */
+ { 0x53, 0x00 }, /* AK4671_PCM_IF_CONTROL0 (0x53) */
+ { 0x54, 0x00 }, /* AK4671_PCM_IF_CONTROL1 (0x54) */
+ { 0x55, 0x00 }, /* AK4671_PCM_IF_CONTROL2 (0x55) */
+ { 0x56, 0x18 }, /* AK4671_DIGITAL_VOLUME_B_CONTROL (0x56) */
+ { 0x57, 0x18 }, /* AK4671_DIGITAL_VOLUME_C_CONTROL (0x57) */
+ { 0x58, 0x00 }, /* AK4671_SIDETONE_VOLUME_CONTROL (0x58) */
+ { 0x59, 0x00 }, /* AK4671_DIGITAL_MIXING_CONTROL2 (0x59) */
+ { 0x5a, 0x00 }, /* AK4671_SAR_ADC_CONTROL (0x5a) */
};
/*
@@ -241,19 +237,17 @@ static const struct snd_kcontrol_new ak4671_rout3_mixer_controls[] = {
/* Input MUXs */
static const char *ak4671_lin_mux_texts[] =
{"LIN1", "LIN2", "LIN3", "LIN4"};
-static const struct soc_enum ak4671_lin_mux_enum =
- SOC_ENUM_SINGLE(AK4671_MIC_SIGNAL_SELECT, 0,
- ARRAY_SIZE(ak4671_lin_mux_texts),
- ak4671_lin_mux_texts);
+static SOC_ENUM_SINGLE_DECL(ak4671_lin_mux_enum,
+ AK4671_MIC_SIGNAL_SELECT, 0,
+ ak4671_lin_mux_texts);
static const struct snd_kcontrol_new ak4671_lin_mux_control =
SOC_DAPM_ENUM("Route", ak4671_lin_mux_enum);
static const char *ak4671_rin_mux_texts[] =
{"RIN1", "RIN2", "RIN3", "RIN4"};
-static const struct soc_enum ak4671_rin_mux_enum =
- SOC_ENUM_SINGLE(AK4671_MIC_SIGNAL_SELECT, 2,
- ARRAY_SIZE(ak4671_rin_mux_texts),
- ak4671_rin_mux_texts);
+static SOC_ENUM_SINGLE_DECL(ak4671_rin_mux_enum,
+ AK4671_MIC_SIGNAL_SELECT, 2,
+ ak4671_rin_mux_texts);
static const struct snd_kcontrol_new ak4671_rin_mux_control =
SOC_DAPM_ENUM("Route", ak4671_rin_mux_enum);
@@ -619,21 +613,7 @@ static struct snd_soc_dai_driver ak4671_dai = {
static int ak4671_probe(struct snd_soc_codec *codec)
{
- struct ak4671_priv *ak4671 = snd_soc_codec_get_drvdata(codec);
- int ret;
-
- ret = snd_soc_codec_set_cache_io(codec, 8, 8, ak4671->control_type);
- if (ret < 0) {
- dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
- return ret;
- }
-
- snd_soc_add_codec_controls(codec, ak4671_snd_controls,
- ARRAY_SIZE(ak4671_snd_controls));
-
- ak4671_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-
- return ret;
+ return ak4671_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
}
static int ak4671_remove(struct snd_soc_codec *codec)
@@ -646,28 +626,36 @@ static struct snd_soc_codec_driver soc_codec_dev_ak4671 = {
.probe = ak4671_probe,
.remove = ak4671_remove,
.set_bias_level = ak4671_set_bias_level,
- .reg_cache_size = AK4671_CACHEREGNUM,
- .reg_word_size = sizeof(u8),
- .reg_cache_default = ak4671_reg,
+ .controls = ak4671_snd_controls,
+ .num_controls = ARRAY_SIZE(ak4671_snd_controls),
.dapm_widgets = ak4671_dapm_widgets,
.num_dapm_widgets = ARRAY_SIZE(ak4671_dapm_widgets),
.dapm_routes = ak4671_intercon,
.num_dapm_routes = ARRAY_SIZE(ak4671_intercon),
};
+static const struct regmap_config ak4671_regmap = {
+ .reg_bits = 8,
+ .val_bits = 8,
+
+ .max_register = AK4671_SAR_ADC_CONTROL,
+ .reg_defaults = ak4671_reg_defaults,
+ .num_reg_defaults = ARRAY_SIZE(ak4671_reg_defaults),
+ .cache_type = REGCACHE_RBTREE,
+};
+
static int ak4671_i2c_probe(struct i2c_client *client,
const struct i2c_device_id *id)
{
- struct ak4671_priv *ak4671;
+ struct regmap *regmap;
int ret;
- ak4671 = devm_kzalloc(&client->dev, sizeof(struct ak4671_priv),
- GFP_KERNEL);
- if (ak4671 == NULL)
- return -ENOMEM;
-
- i2c_set_clientdata(client, ak4671);
- ak4671->control_type = SND_SOC_I2C;
+ regmap = devm_regmap_init_i2c(client, &ak4671_regmap);
+ if (IS_ERR(regmap)) {
+ ret = PTR_ERR(regmap);
+ dev_err(&client->dev, "Failed to create regmap: %d\n", ret);
+ return ret;
+ }
ret = snd_soc_register_codec(&client->dev,
&soc_codec_dev_ak4671, &ak4671_dai, 1);
diff --git a/sound/soc/codecs/ak4671.h b/sound/soc/codecs/ak4671.h
index 61cb7ab..394a34d 100644
--- a/sound/soc/codecs/ak4671.h
+++ b/sound/soc/codecs/ak4671.h
@@ -105,8 +105,6 @@
#define AK4671_DIGITAL_MIXING_CONTROL2 0x59
#define AK4671_SAR_ADC_CONTROL 0x5a
-#define AK4671_CACHEREGNUM (AK4671_SAR_ADC_CONTROL + 1)
-
/* Bitfield Definitions */
/* AK4671_AD_DA_POWER_MANAGEMENT (0x00) Fields */
diff --git a/sound/soc/codecs/alc5623.c b/sound/soc/codecs/alc5623.c
index d303628..09f7e77 100644
--- a/sound/soc/codecs/alc5623.c
+++ b/sound/soc/codecs/alc5623.c
@@ -21,6 +21,7 @@
#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>
@@ -38,26 +39,13 @@ MODULE_PARM_DESC(caps_charge, "ALC5623 cap charge time (msecs)");
/* codec private data */
struct alc5623_priv {
- enum snd_soc_control_type control_type;
+ struct regmap *regmap;
u8 id;
unsigned int sysclk;
- u16 reg_cache[ALC5623_VENDOR_ID2+2];
unsigned int add_ctrl;
unsigned int jack_det_ctrl;
};
-static void alc5623_fill_cache(struct snd_soc_codec *codec)
-{
- int i, step = codec->driver->reg_cache_step;
- u16 *cache = codec->reg_cache;
-
- /* not really efficient ... */
- codec->cache_bypass = 1;
- for (i = 0 ; i < codec->driver->reg_cache_size ; i += step)
- cache[i] = snd_soc_read(codec, i);
- codec->cache_bypass = 0;
-}
-
static inline int alc5623_reset(struct snd_soc_codec *codec)
{
return snd_soc_write(codec, ALC5623_RESET, 0);
@@ -228,32 +216,37 @@ static const char *alc5623_aux_out_input_sel[] = {
"Vmid", "HPOut Mix", "Speaker Mix", "Mono Mix"};
/* auxout output mux */
-static const struct soc_enum alc5623_aux_out_input_enum =
-SOC_ENUM_SINGLE(ALC5623_OUTPUT_MIXER_CTRL, 6, 4, alc5623_aux_out_input_sel);
+static SOC_ENUM_SINGLE_DECL(alc5623_aux_out_input_enum,
+ ALC5623_OUTPUT_MIXER_CTRL, 6,
+ alc5623_aux_out_input_sel);
static const struct snd_kcontrol_new alc5623_auxout_mux_controls =
SOC_DAPM_ENUM("Route", alc5623_aux_out_input_enum);
/* speaker output mux */
-static const struct soc_enum alc5623_spkout_input_enum =
-SOC_ENUM_SINGLE(ALC5623_OUTPUT_MIXER_CTRL, 10, 4, alc5623_spkout_input_sel);
+static SOC_ENUM_SINGLE_DECL(alc5623_spkout_input_enum,
+ ALC5623_OUTPUT_MIXER_CTRL, 10,
+ alc5623_spkout_input_sel);
static const struct snd_kcontrol_new alc5623_spkout_mux_controls =
SOC_DAPM_ENUM("Route", alc5623_spkout_input_enum);
/* headphone left output mux */
-static const struct soc_enum alc5623_hpl_out_input_enum =
-SOC_ENUM_SINGLE(ALC5623_OUTPUT_MIXER_CTRL, 9, 2, alc5623_hpl_out_input_sel);
+static SOC_ENUM_SINGLE_DECL(alc5623_hpl_out_input_enum,
+ ALC5623_OUTPUT_MIXER_CTRL, 9,
+ alc5623_hpl_out_input_sel);
static const struct snd_kcontrol_new alc5623_hpl_out_mux_controls =
SOC_DAPM_ENUM("Route", alc5623_hpl_out_input_enum);
/* headphone right output mux */
-static const struct soc_enum alc5623_hpr_out_input_enum =
-SOC_ENUM_SINGLE(ALC5623_OUTPUT_MIXER_CTRL, 8, 2, alc5623_hpr_out_input_sel);
+static SOC_ENUM_SINGLE_DECL(alc5623_hpr_out_input_enum,
+ ALC5623_OUTPUT_MIXER_CTRL, 8,
+ alc5623_hpr_out_input_sel);
static const struct snd_kcontrol_new alc5623_hpr_out_mux_controls =
SOC_DAPM_ENUM("Route", alc5623_hpr_out_input_enum);
/* speaker output N select */
-static const struct soc_enum alc5623_spk_n_sour_enum =
-SOC_ENUM_SINGLE(ALC5623_OUTPUT_MIXER_CTRL, 14, 4, alc5623_spk_n_sour_sel);
+static SOC_ENUM_SINGLE_DECL(alc5623_spk_n_sour_enum,
+ ALC5623_OUTPUT_MIXER_CTRL, 14,
+ alc5623_spk_n_sour_sel);
static const struct snd_kcontrol_new alc5623_spkoutn_mux_controls =
SOC_DAPM_ENUM("Route", alc5623_spk_n_sour_enum);
@@ -338,8 +331,9 @@ SND_SOC_DAPM_VMID("Vmid"),
};
static const char *alc5623_amp_names[] = {"AB Amp", "D Amp"};
-static const struct soc_enum alc5623_amp_enum =
- SOC_ENUM_SINGLE(ALC5623_OUTPUT_MIXER_CTRL, 13, 2, alc5623_amp_names);
+static SOC_ENUM_SINGLE_DECL(alc5623_amp_enum,
+ ALC5623_OUTPUT_MIXER_CTRL, 13,
+ alc5623_amp_names);
static const struct snd_kcontrol_new alc5623_amp_mux_controls =
SOC_DAPM_ENUM("Route", alc5623_amp_enum);
@@ -869,18 +863,28 @@ static struct snd_soc_dai_driver alc5623_dai = {
static int alc5623_suspend(struct snd_soc_codec *codec)
{
+ struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec);
+
alc5623_set_bias_level(codec, SND_SOC_BIAS_OFF);
+ regcache_cache_only(alc5623->regmap, true);
+
return 0;
}
static int alc5623_resume(struct snd_soc_codec *codec)
{
- int i, step = codec->driver->reg_cache_step;
- u16 *cache = codec->reg_cache;
+ struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec);
+ int ret;
/* Sync reg_cache with the hardware */
- for (i = 2 ; i < codec->driver->reg_cache_size ; i += step)
- snd_soc_write(codec, i, cache[i]);
+ regcache_cache_only(alc5623->regmap, false);
+ ret = regcache_sync(alc5623->regmap);
+ if (ret != 0) {
+ dev_err(codec->dev, "Failed to sync register cache: %d\n",
+ ret);
+ regcache_cache_only(alc5623->regmap, true);
+ return ret;
+ }
alc5623_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
@@ -900,14 +904,7 @@ static int alc5623_probe(struct snd_soc_codec *codec)
struct snd_soc_dapm_context *dapm = &codec->dapm;
int ret;
- ret = snd_soc_codec_set_cache_io(codec, 8, 16, alc5623->control_type);
- if (ret < 0) {
- dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
- return ret;
- }
-
alc5623_reset(codec);
- alc5623_fill_cache(codec);
/* power on device */
alc5623_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
@@ -980,9 +977,15 @@ static struct snd_soc_codec_driver soc_codec_device_alc5623 = {
.suspend = alc5623_suspend,
.resume = alc5623_resume,
.set_bias_level = alc5623_set_bias_level,
- .reg_cache_size = ALC5623_VENDOR_ID2+2,
- .reg_word_size = sizeof(u16),
- .reg_cache_step = 2,
+};
+
+static const struct regmap_config alc5623_regmap = {
+ .reg_bits = 8,
+ .val_bits = 16,
+ .reg_stride = 2,
+
+ .max_register = ALC5623_VENDOR_ID2,
+ .cache_type = REGCACHE_RBTREE,
};
/*
@@ -996,19 +999,32 @@ static int alc5623_i2c_probe(struct i2c_client *client,
{
struct alc5623_platform_data *pdata;
struct alc5623_priv *alc5623;
- int ret, vid1, vid2;
+ unsigned int vid1, vid2;
+ int ret;
- vid1 = i2c_smbus_read_word_data(client, ALC5623_VENDOR_ID1);
- if (vid1 < 0) {
- dev_err(&client->dev, "failed to read I2C\n");
- return -EIO;
+ alc5623 = devm_kzalloc(&client->dev, sizeof(struct alc5623_priv),
+ GFP_KERNEL);
+ if (alc5623 == NULL)
+ return -ENOMEM;
+
+ alc5623->regmap = devm_regmap_init_i2c(client, &alc5623_regmap);
+ if (IS_ERR(alc5623->regmap)) {
+ ret = PTR_ERR(alc5623->regmap);
+ dev_err(&client->dev, "Failed to initialise I/O: %d\n", ret);
+ return ret;
+ }
+
+ ret = regmap_read(alc5623->regmap, ALC5623_VENDOR_ID1, &vid1);
+ if (ret < 0) {
+ dev_err(&client->dev, "failed to read vendor ID1: %d\n", ret);
+ return ret;
}
vid1 = ((vid1 & 0xff) << 8) | (vid1 >> 8);
- vid2 = i2c_smbus_read_byte_data(client, ALC5623_VENDOR_ID2);
- if (vid2 < 0) {
- dev_err(&client->dev, "failed to read I2C\n");
- return -EIO;
+ ret = regmap_read(alc5623->regmap, ALC5623_VENDOR_ID2, &vid2);
+ if (ret < 0) {
+ dev_err(&client->dev, "failed to read vendor ID2: %d\n", ret);
+ return ret;
}
if ((vid1 != 0x10ec) || (vid2 != id->driver_data)) {
@@ -1021,11 +1037,6 @@ static int alc5623_i2c_probe(struct i2c_client *client,
dev_dbg(&client->dev, "Found codec id : alc56%02x\n", vid2);
- alc5623 = devm_kzalloc(&client->dev, sizeof(struct alc5623_priv),
- GFP_KERNEL);
- if (alc5623 == NULL)
- return -ENOMEM;
-
pdata = client->dev.platform_data;
if (pdata) {
alc5623->add_ctrl = pdata->add_ctrl;
@@ -1048,7 +1059,6 @@ static int alc5623_i2c_probe(struct i2c_client *client,
}
i2c_set_clientdata(client, alc5623);
- alc5623->control_type = SND_SOC_I2C;
ret = snd_soc_register_codec(&client->dev,
&soc_codec_device_alc5623, &alc5623_dai, 1);
diff --git a/sound/soc/codecs/alc5632.c b/sound/soc/codecs/alc5632.c
index fb001c5..ec071a6 100644
--- a/sound/soc/codecs/alc5632.c
+++ b/sound/soc/codecs/alc5632.c
@@ -293,51 +293,59 @@ static const char * const alc5632_i2s_out_sel[] = {
"ADC LR", "Voice Stereo Digital"};
/* auxout output mux */
-static const struct soc_enum alc5632_aux_out_input_enum =
-SOC_ENUM_SINGLE(ALC5632_OUTPUT_MIXER_CTRL, 6, 4, alc5632_aux_out_input_sel);
+static SOC_ENUM_SINGLE_DECL(alc5632_aux_out_input_enum,
+ ALC5632_OUTPUT_MIXER_CTRL, 6,
+ alc5632_aux_out_input_sel);
static const struct snd_kcontrol_new alc5632_auxout_mux_controls =
SOC_DAPM_ENUM("AuxOut Mux", alc5632_aux_out_input_enum);
/* speaker output mux */
-static const struct soc_enum alc5632_spkout_input_enum =
-SOC_ENUM_SINGLE(ALC5632_OUTPUT_MIXER_CTRL, 10, 4, alc5632_spkout_input_sel);
+static SOC_ENUM_SINGLE_DECL(alc5632_spkout_input_enum,
+ ALC5632_OUTPUT_MIXER_CTRL, 10,
+ alc5632_spkout_input_sel);
static const struct snd_kcontrol_new alc5632_spkout_mux_controls =
SOC_DAPM_ENUM("SpeakerOut Mux", alc5632_spkout_input_enum);
/* headphone left output mux */
-static const struct soc_enum alc5632_hpl_out_input_enum =
-SOC_ENUM_SINGLE(ALC5632_OUTPUT_MIXER_CTRL, 9, 2, alc5632_hpl_out_input_sel);
+static SOC_ENUM_SINGLE_DECL(alc5632_hpl_out_input_enum,
+ ALC5632_OUTPUT_MIXER_CTRL, 9,
+ alc5632_hpl_out_input_sel);
static const struct snd_kcontrol_new alc5632_hpl_out_mux_controls =
SOC_DAPM_ENUM("Left Headphone Mux", alc5632_hpl_out_input_enum);
/* headphone right output mux */
-static const struct soc_enum alc5632_hpr_out_input_enum =
-SOC_ENUM_SINGLE(ALC5632_OUTPUT_MIXER_CTRL, 8, 2, alc5632_hpr_out_input_sel);
+static SOC_ENUM_SINGLE_DECL(alc5632_hpr_out_input_enum,
+ ALC5632_OUTPUT_MIXER_CTRL, 8,
+ alc5632_hpr_out_input_sel);
static const struct snd_kcontrol_new alc5632_hpr_out_mux_controls =
SOC_DAPM_ENUM("Right Headphone Mux", alc5632_hpr_out_input_enum);
/* speaker output N select */
-static const struct soc_enum alc5632_spk_n_sour_enum =
-SOC_ENUM_SINGLE(ALC5632_OUTPUT_MIXER_CTRL, 14, 4, alc5632_spk_n_sour_sel);
+static SOC_ENUM_SINGLE_DECL(alc5632_spk_n_sour_enum,
+ ALC5632_OUTPUT_MIXER_CTRL, 14,
+ alc5632_spk_n_sour_sel);
static const struct snd_kcontrol_new alc5632_spkoutn_mux_controls =
SOC_DAPM_ENUM("SpeakerOut N Mux", alc5632_spk_n_sour_enum);
/* speaker amplifier */
static const char *alc5632_amp_names[] = {"AB Amp", "D Amp"};
-static const struct soc_enum alc5632_amp_enum =
- SOC_ENUM_SINGLE(ALC5632_OUTPUT_MIXER_CTRL, 13, 2, alc5632_amp_names);
+static SOC_ENUM_SINGLE_DECL(alc5632_amp_enum,
+ ALC5632_OUTPUT_MIXER_CTRL, 13,
+ alc5632_amp_names);
static const struct snd_kcontrol_new alc5632_amp_mux_controls =
SOC_DAPM_ENUM("AB-D Amp Mux", alc5632_amp_enum);
/* ADC output select */
-static const struct soc_enum alc5632_adcr_func_enum =
- SOC_ENUM_SINGLE(ALC5632_DAC_FUNC_SELECT, 5, 2, alc5632_adcr_func_sel);
+static SOC_ENUM_SINGLE_DECL(alc5632_adcr_func_enum,
+ ALC5632_DAC_FUNC_SELECT, 5,
+ alc5632_adcr_func_sel);
static const struct snd_kcontrol_new alc5632_adcr_func_controls =
SOC_DAPM_ENUM("ADCR Mux", alc5632_adcr_func_enum);
/* I2S out select */
-static const struct soc_enum alc5632_i2s_out_enum =
- SOC_ENUM_SINGLE(ALC5632_I2S_OUT_CTL, 5, 2, alc5632_i2s_out_sel);
+static SOC_ENUM_SINGLE_DECL(alc5632_i2s_out_enum,
+ ALC5632_I2S_OUT_CTL, 5,
+ alc5632_i2s_out_sel);
static const struct snd_kcontrol_new alc5632_i2s_out_controls =
SOC_DAPM_ENUM("I2SOut Mux", alc5632_i2s_out_enum);
@@ -1055,14 +1063,6 @@ static int alc5632_probe(struct snd_soc_codec *codec)
struct alc5632_priv *alc5632 = snd_soc_codec_get_drvdata(codec);
int ret;
- codec->control_data = alc5632->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;
- }
-
/* power on device */
alc5632_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c
index e4295fe..29e198f 100644
--- a/sound/soc/codecs/arizona.c
+++ b/sound/soc/codecs/arizona.c
@@ -53,6 +53,14 @@
#define ARIZONA_AIF_RX_ENABLES 0x1A
#define ARIZONA_AIF_FORCE_WRITE 0x1B
+#define ARIZONA_FLL_VCO_CORNER 141900000
+#define ARIZONA_FLL_MAX_FREF 13500000
+#define ARIZONA_FLL_MIN_FVCO 90000000
+#define ARIZONA_FLL_MAX_FRATIO 16
+#define ARIZONA_FLL_MAX_REFDIV 8
+#define ARIZONA_FLL_MIN_OUTDIV 2
+#define ARIZONA_FLL_MAX_OUTDIV 7
+
#define arizona_fll_err(_fll, fmt, ...) \
dev_err(_fll->arizona->dev, "FLL%d: " fmt, _fll->id, ##__VA_ARGS__)
#define arizona_fll_warn(_fll, fmt, ...) \
@@ -542,67 +550,76 @@ static const char *arizona_vol_ramp_text[] = {
"15ms/6dB", "30ms/6dB",
};
-const struct soc_enum arizona_in_vd_ramp =
- SOC_ENUM_SINGLE(ARIZONA_INPUT_VOLUME_RAMP,
- ARIZONA_IN_VD_RAMP_SHIFT, 7, arizona_vol_ramp_text);
+SOC_ENUM_SINGLE_DECL(arizona_in_vd_ramp,
+ ARIZONA_INPUT_VOLUME_RAMP,
+ ARIZONA_IN_VD_RAMP_SHIFT,
+ arizona_vol_ramp_text);
EXPORT_SYMBOL_GPL(arizona_in_vd_ramp);
-const struct soc_enum arizona_in_vi_ramp =
- SOC_ENUM_SINGLE(ARIZONA_INPUT_VOLUME_RAMP,
- ARIZONA_IN_VI_RAMP_SHIFT, 7, arizona_vol_ramp_text);
+SOC_ENUM_SINGLE_DECL(arizona_in_vi_ramp,
+ ARIZONA_INPUT_VOLUME_RAMP,
+ ARIZONA_IN_VI_RAMP_SHIFT,
+ arizona_vol_ramp_text);
EXPORT_SYMBOL_GPL(arizona_in_vi_ramp);
-const struct soc_enum arizona_out_vd_ramp =
- SOC_ENUM_SINGLE(ARIZONA_OUTPUT_VOLUME_RAMP,
- ARIZONA_OUT_VD_RAMP_SHIFT, 7, arizona_vol_ramp_text);
+SOC_ENUM_SINGLE_DECL(arizona_out_vd_ramp,
+ ARIZONA_OUTPUT_VOLUME_RAMP,
+ ARIZONA_OUT_VD_RAMP_SHIFT,
+ arizona_vol_ramp_text);
EXPORT_SYMBOL_GPL(arizona_out_vd_ramp);
-const struct soc_enum arizona_out_vi_ramp =
- SOC_ENUM_SINGLE(ARIZONA_OUTPUT_VOLUME_RAMP,
- ARIZONA_OUT_VI_RAMP_SHIFT, 7, arizona_vol_ramp_text);
+SOC_ENUM_SINGLE_DECL(arizona_out_vi_ramp,
+ ARIZONA_OUTPUT_VOLUME_RAMP,
+ ARIZONA_OUT_VI_RAMP_SHIFT,
+ arizona_vol_ramp_text);
EXPORT_SYMBOL_GPL(arizona_out_vi_ramp);
static const char *arizona_lhpf_mode_text[] = {
"Low-pass", "High-pass"
};
-const struct soc_enum arizona_lhpf1_mode =
- SOC_ENUM_SINGLE(ARIZONA_HPLPF1_1, ARIZONA_LHPF1_MODE_SHIFT, 2,
- arizona_lhpf_mode_text);
+SOC_ENUM_SINGLE_DECL(arizona_lhpf1_mode,
+ ARIZONA_HPLPF1_1,
+ ARIZONA_LHPF1_MODE_SHIFT,
+ arizona_lhpf_mode_text);
EXPORT_SYMBOL_GPL(arizona_lhpf1_mode);
-const struct soc_enum arizona_lhpf2_mode =
- SOC_ENUM_SINGLE(ARIZONA_HPLPF2_1, ARIZONA_LHPF2_MODE_SHIFT, 2,
- arizona_lhpf_mode_text);
+SOC_ENUM_SINGLE_DECL(arizona_lhpf2_mode,
+ ARIZONA_HPLPF2_1,
+ ARIZONA_LHPF2_MODE_SHIFT,
+ arizona_lhpf_mode_text);
EXPORT_SYMBOL_GPL(arizona_lhpf2_mode);
-const struct soc_enum arizona_lhpf3_mode =
- SOC_ENUM_SINGLE(ARIZONA_HPLPF3_1, ARIZONA_LHPF3_MODE_SHIFT, 2,
- arizona_lhpf_mode_text);
+SOC_ENUM_SINGLE_DECL(arizona_lhpf3_mode,
+ ARIZONA_HPLPF3_1,
+ ARIZONA_LHPF3_MODE_SHIFT,
+ arizona_lhpf_mode_text);
EXPORT_SYMBOL_GPL(arizona_lhpf3_mode);
-const struct soc_enum arizona_lhpf4_mode =
- SOC_ENUM_SINGLE(ARIZONA_HPLPF4_1, ARIZONA_LHPF4_MODE_SHIFT, 2,
- arizona_lhpf_mode_text);
+SOC_ENUM_SINGLE_DECL(arizona_lhpf4_mode,
+ ARIZONA_HPLPF4_1,
+ ARIZONA_LHPF4_MODE_SHIFT,
+ arizona_lhpf_mode_text);
EXPORT_SYMBOL_GPL(arizona_lhpf4_mode);
static const char *arizona_ng_hold_text[] = {
"30ms", "120ms", "250ms", "500ms",
};
-const struct soc_enum arizona_ng_hold =
- SOC_ENUM_SINGLE(ARIZONA_NOISE_GATE_CONTROL, ARIZONA_NGATE_HOLD_SHIFT,
- 4, arizona_ng_hold_text);
+SOC_ENUM_SINGLE_DECL(arizona_ng_hold,
+ ARIZONA_NOISE_GATE_CONTROL,
+ ARIZONA_NGATE_HOLD_SHIFT,
+ arizona_ng_hold_text);
EXPORT_SYMBOL_GPL(arizona_ng_hold);
static const char * const arizona_in_hpf_cut_text[] = {
"2.5Hz", "5Hz", "10Hz", "20Hz", "40Hz"
};
-const struct soc_enum arizona_in_hpf_cut_enum =
- SOC_ENUM_SINGLE(ARIZONA_HPF_CONTROL, ARIZONA_IN_HPF_CUT_SHIFT,
- ARRAY_SIZE(arizona_in_hpf_cut_text),
- arizona_in_hpf_cut_text);
+SOC_ENUM_SINGLE_DECL(arizona_in_hpf_cut_enum,
+ ARIZONA_HPF_CONTROL,
+ ARIZONA_IN_HPF_CUT_SHIFT,
+ arizona_in_hpf_cut_text);
EXPORT_SYMBOL_GPL(arizona_in_hpf_cut_enum);
static const char * const arizona_in_dmic_osr_text[] = {
@@ -1377,74 +1394,147 @@ struct arizona_fll_cfg {
int gain;
};
-static int arizona_calc_fll(struct arizona_fll *fll,
- struct arizona_fll_cfg *cfg,
- unsigned int Fref,
- unsigned int Fout)
+static int arizona_validate_fll(struct arizona_fll *fll,
+ unsigned int Fref,
+ unsigned int Fout)
{
- unsigned int target, div, gcd_fll;
- int i, ratio;
+ unsigned int Fvco_min;
+
+ if (Fref / ARIZONA_FLL_MAX_REFDIV > ARIZONA_FLL_MAX_FREF) {
+ arizona_fll_err(fll,
+ "Can't scale %dMHz in to <=13.5MHz\n",
+ Fref);
+ return -EINVAL;
+ }
- arizona_fll_dbg(fll, "Fref=%u Fout=%u\n", Fref, Fout);
+ Fvco_min = ARIZONA_FLL_MIN_FVCO * fll->vco_mult;
+ if (Fout * ARIZONA_FLL_MAX_OUTDIV < Fvco_min) {
+ arizona_fll_err(fll, "No FLL_OUTDIV for Fout=%uHz\n",
+ Fout);
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static int arizona_find_fratio(unsigned int Fref, int *fratio)
+{
+ int i;
+
+ /* Find an appropriate FLL_FRATIO */
+ for (i = 0; i < ARRAY_SIZE(fll_fratios); i++) {
+ if (fll_fratios[i].min <= Fref && Fref <= fll_fratios[i].max) {
+ if (fratio)
+ *fratio = fll_fratios[i].fratio;
+ return fll_fratios[i].ratio;
+ }
+ }
+
+ return -EINVAL;
+}
+
+static int arizona_calc_fratio(struct arizona_fll *fll,
+ struct arizona_fll_cfg *cfg,
+ unsigned int target,
+ unsigned int Fref, bool sync)
+{
+ int init_ratio, ratio;
+ int refdiv, div;
- /* Fref must be <=13.5MHz */
+ /* Fref must be <=13.5MHz, find initial refdiv */
div = 1;
cfg->refdiv = 0;
- while ((Fref / div) > 13500000) {
+ while (Fref > ARIZONA_FLL_MAX_FREF) {
div *= 2;
+ Fref /= 2;
cfg->refdiv++;
- if (div > 8) {
- arizona_fll_err(fll,
- "Can't scale %dMHz in to <=13.5MHz\n",
- Fref);
+ if (div > ARIZONA_FLL_MAX_REFDIV)
return -EINVAL;
+ }
+
+ /* Find an appropriate FLL_FRATIO */
+ init_ratio = arizona_find_fratio(Fref, &cfg->fratio);
+ if (init_ratio < 0) {
+ arizona_fll_err(fll, "Unable to find FRATIO for Fref=%uHz\n",
+ Fref);
+ return init_ratio;
+ }
+
+ switch (fll->arizona->type) {
+ case WM5110:
+ if (fll->arizona->rev < 3 || sync)
+ return init_ratio;
+ break;
+ default:
+ return init_ratio;
+ }
+
+ cfg->fratio = init_ratio - 1;
+
+ /* Adjust FRATIO/refdiv to avoid integer mode if possible */
+ refdiv = cfg->refdiv;
+
+ while (div <= ARIZONA_FLL_MAX_REFDIV) {
+ for (ratio = init_ratio; ratio <= ARIZONA_FLL_MAX_FRATIO;
+ ratio++) {
+ if (target % (ratio * Fref)) {
+ cfg->refdiv = refdiv;
+ cfg->fratio = ratio - 1;
+ return ratio;
+ }
}
+
+ for (ratio = init_ratio - 1; ratio >= 0; ratio--) {
+ if (ARIZONA_FLL_VCO_CORNER / (fll->vco_mult * ratio) <
+ Fref)
+ break;
+
+ if (target % (ratio * Fref)) {
+ cfg->refdiv = refdiv;
+ cfg->fratio = ratio - 1;
+ return ratio;
+ }
+ }
+
+ div *= 2;
+ Fref /= 2;
+ refdiv++;
+ init_ratio = arizona_find_fratio(Fref, NULL);
}
- /* Apply the division for our remaining calculations */
- Fref /= div;
+ arizona_fll_warn(fll, "Falling back to integer mode operation\n");
+ return cfg->fratio + 1;
+}
+
+static int arizona_calc_fll(struct arizona_fll *fll,
+ struct arizona_fll_cfg *cfg,
+ unsigned int Fref, bool sync)
+{
+ unsigned int target, div, gcd_fll;
+ int i, ratio;
+
+ arizona_fll_dbg(fll, "Fref=%u Fout=%u\n", Fref, fll->fout);
/* Fvco should be over the targt; don't check the upper bound */
- div = 1;
- while (Fout * div < 90000000 * fll->vco_mult) {
+ div = ARIZONA_FLL_MIN_OUTDIV;
+ while (fll->fout * div < ARIZONA_FLL_MIN_FVCO * fll->vco_mult) {
div++;
- if (div > 7) {
- arizona_fll_err(fll, "No FLL_OUTDIV for Fout=%uHz\n",
- Fout);
+ if (div > ARIZONA_FLL_MAX_OUTDIV)
return -EINVAL;
- }
}
- target = Fout * div / fll->vco_mult;
+ target = fll->fout * div / fll->vco_mult;
cfg->outdiv = div;
arizona_fll_dbg(fll, "Fvco=%dHz\n", target);
- /* Find an appropraite FLL_FRATIO and factor it out of the target */
- for (i = 0; i < ARRAY_SIZE(fll_fratios); i++) {
- if (fll_fratios[i].min <= Fref && Fref <= fll_fratios[i].max) {
- cfg->fratio = fll_fratios[i].fratio;
- ratio = fll_fratios[i].ratio;
- break;
- }
- }
- if (i == ARRAY_SIZE(fll_fratios)) {
- arizona_fll_err(fll, "Unable to find FRATIO for Fref=%uHz\n",
- Fref);
- return -EINVAL;
- }
+ /* Find an appropriate FLL_FRATIO and refdiv */
+ ratio = arizona_calc_fratio(fll, cfg, target, Fref, sync);
+ if (ratio < 0)
+ return ratio;
- for (i = 0; i < ARRAY_SIZE(fll_gains); i++) {
- if (fll_gains[i].min <= Fref && Fref <= fll_gains[i].max) {
- cfg->gain = fll_gains[i].gain;
- break;
- }
- }
- if (i == ARRAY_SIZE(fll_gains)) {
- arizona_fll_err(fll, "Unable to find gain for Fref=%uHz\n",
- Fref);
- return -EINVAL;
- }
+ /* Apply the division for our remaining calculations */
+ Fref = Fref / (1 << cfg->refdiv);
cfg->n = target / (ratio * Fref);
@@ -1469,6 +1559,18 @@ static int arizona_calc_fll(struct arizona_fll *fll,
cfg->lambda >>= 1;
}
+ for (i = 0; i < ARRAY_SIZE(fll_gains); i++) {
+ if (fll_gains[i].min <= Fref && Fref <= fll_gains[i].max) {
+ cfg->gain = fll_gains[i].gain;
+ break;
+ }
+ }
+ if (i == ARRAY_SIZE(fll_gains)) {
+ arizona_fll_err(fll, "Unable to find gain for Fref=%uHz\n",
+ Fref);
+ return -EINVAL;
+ }
+
arizona_fll_dbg(fll, "N=%x THETA=%x LAMBDA=%x\n",
cfg->n, cfg->theta, cfg->lambda);
arizona_fll_dbg(fll, "FRATIO=%x(%d) OUTDIV=%x REFCLK_DIV=%x\n",
@@ -1496,14 +1598,18 @@ static void arizona_apply_fll(struct arizona *arizona, unsigned int base,
cfg->refdiv << ARIZONA_FLL1_CLK_REF_DIV_SHIFT |
source << ARIZONA_FLL1_CLK_REF_SRC_SHIFT);
- if (sync)
- regmap_update_bits_async(arizona->regmap, base + 0x7,
- ARIZONA_FLL1_GAIN_MASK,
- cfg->gain << ARIZONA_FLL1_GAIN_SHIFT);
- else
- regmap_update_bits_async(arizona->regmap, base + 0x9,
- ARIZONA_FLL1_GAIN_MASK,
- cfg->gain << ARIZONA_FLL1_GAIN_SHIFT);
+ if (sync) {
+ regmap_update_bits(arizona->regmap, base + 0x7,
+ ARIZONA_FLL1_GAIN_MASK,
+ cfg->gain << ARIZONA_FLL1_GAIN_SHIFT);
+ } else {
+ regmap_update_bits(arizona->regmap, base + 0x5,
+ ARIZONA_FLL1_OUTDIV_MASK,
+ cfg->outdiv << ARIZONA_FLL1_OUTDIV_SHIFT);
+ regmap_update_bits(arizona->regmap, base + 0x9,
+ ARIZONA_FLL1_GAIN_MASK,
+ cfg->gain << ARIZONA_FLL1_GAIN_SHIFT);
+ }
regmap_update_bits_async(arizona->regmap, base + 2,
ARIZONA_FLL1_CTRL_UPD | ARIZONA_FLL1_N_MASK,
@@ -1526,13 +1632,12 @@ static bool arizona_is_enabled_fll(struct arizona_fll *fll)
return reg & ARIZONA_FLL1_ENA;
}
-static void arizona_enable_fll(struct arizona_fll *fll,
- struct arizona_fll_cfg *ref,
- struct arizona_fll_cfg *sync)
+static void arizona_enable_fll(struct arizona_fll *fll)
{
struct arizona *arizona = fll->arizona;
int ret;
bool use_sync = false;
+ struct arizona_fll_cfg cfg;
/*
* If we have both REFCLK and SYNCCLK then enable both,
@@ -1540,23 +1645,21 @@ static void arizona_enable_fll(struct arizona_fll *fll,
*/
if (fll->ref_src >= 0 && fll->ref_freq &&
fll->ref_src != fll->sync_src) {
- regmap_update_bits_async(arizona->regmap, fll->base + 5,
- ARIZONA_FLL1_OUTDIV_MASK,
- ref->outdiv << ARIZONA_FLL1_OUTDIV_SHIFT);
+ arizona_calc_fll(fll, &cfg, fll->ref_freq, false);
- arizona_apply_fll(arizona, fll->base, ref, fll->ref_src,
+ arizona_apply_fll(arizona, fll->base, &cfg, fll->ref_src,
false);
if (fll->sync_src >= 0) {
- arizona_apply_fll(arizona, fll->base + 0x10, sync,
+ arizona_calc_fll(fll, &cfg, fll->sync_freq, true);
+
+ arizona_apply_fll(arizona, fll->base + 0x10, &cfg,
fll->sync_src, true);
use_sync = true;
}
} else if (fll->sync_src >= 0) {
- regmap_update_bits_async(arizona->regmap, fll->base + 5,
- ARIZONA_FLL1_OUTDIV_MASK,
- sync->outdiv << ARIZONA_FLL1_OUTDIV_SHIFT);
+ arizona_calc_fll(fll, &cfg, fll->sync_freq, false);
- arizona_apply_fll(arizona, fll->base, sync,
+ arizona_apply_fll(arizona, fll->base, &cfg,
fll->sync_src, false);
regmap_update_bits_async(arizona->regmap, fll->base + 0x11,
@@ -1618,32 +1721,22 @@ static void arizona_disable_fll(struct arizona_fll *fll)
int arizona_set_fll_refclk(struct arizona_fll *fll, int source,
unsigned int Fref, unsigned int Fout)
{
- struct arizona_fll_cfg ref, sync;
int ret;
if (fll->ref_src == source && fll->ref_freq == Fref)
return 0;
- if (fll->fout) {
- if (Fref > 0) {
- ret = arizona_calc_fll(fll, &ref, Fref, fll->fout);
- if (ret != 0)
- return ret;
- }
-
- if (fll->sync_src >= 0) {
- ret = arizona_calc_fll(fll, &sync, fll->sync_freq,
- fll->fout);
- if (ret != 0)
- return ret;
- }
+ if (fll->fout && Fref > 0) {
+ ret = arizona_validate_fll(fll, Fref, fll->fout);
+ if (ret != 0)
+ return ret;
}
fll->ref_src = source;
fll->ref_freq = Fref;
if (fll->fout && Fref > 0) {
- arizona_enable_fll(fll, &ref, &sync);
+ arizona_enable_fll(fll);
}
return 0;
@@ -1653,7 +1746,6 @@ EXPORT_SYMBOL_GPL(arizona_set_fll_refclk);
int arizona_set_fll(struct arizona_fll *fll, int source,
unsigned int Fref, unsigned int Fout)
{
- struct arizona_fll_cfg ref, sync;
int ret;
if (fll->sync_src == source &&
@@ -1662,13 +1754,12 @@ int arizona_set_fll(struct arizona_fll *fll, int source,
if (Fout) {
if (fll->ref_src >= 0) {
- ret = arizona_calc_fll(fll, &ref, fll->ref_freq,
- Fout);
+ ret = arizona_validate_fll(fll, fll->ref_freq, Fout);
if (ret != 0)
return ret;
}
- ret = arizona_calc_fll(fll, &sync, Fref, Fout);
+ ret = arizona_validate_fll(fll, Fref, Fout);
if (ret != 0)
return ret;
}
@@ -1678,7 +1769,7 @@ int arizona_set_fll(struct arizona_fll *fll, int source,
fll->fout = Fout;
if (Fout) {
- arizona_enable_fll(fll, &ref, &sync);
+ arizona_enable_fll(fll);
} else {
arizona_disable_fll(fll);
}
diff --git a/sound/soc/codecs/cq93vc.c b/sound/soc/codecs/cq93vc.c
index 43737a27..1e25c7a 100644
--- a/sound/soc/codecs/cq93vc.c
+++ b/sound/soc/codecs/cq93vc.c
@@ -138,9 +138,8 @@ static int cq93vc_probe(struct snd_soc_codec *codec)
struct davinci_vc *davinci_vc = codec->dev->platform_data;
davinci_vc->cq93vc.codec = codec;
- codec->control_data = davinci_vc->regmap;
- snd_soc_codec_set_cache_io(codec, 32, 32, SND_SOC_REGMAP);
+ snd_soc_codec_set_cache_io(codec, davinci_vc->regmap);
/* Off, with power on */
cq93vc_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c
index 83c835d..3920e62 100644
--- a/sound/soc/codecs/cs4270.c
+++ b/sound/soc/codecs/cs4270.c
@@ -506,15 +506,6 @@ static int cs4270_probe(struct snd_soc_codec *codec)
struct cs4270_private *cs4270 = snd_soc_codec_get_drvdata(codec);
int ret;
- /* Tell ASoC what kind of I/O to use to read the registers. ASoC will
- * then do the I2C transactions itself.
- */
- 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 (ret=%i)\n", ret);
- return ret;
- }
-
/* Disable auto-mute. This feature appears to be buggy. In some
* situations, auto-mute will not deactivate when it should, so we want
* this feature disabled by default. An application (e.g. alsactl) can
diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c
index ce05fd9..aef4965 100644
--- a/sound/soc/codecs/cs4271.c
+++ b/sound/soc/codecs/cs4271.c
@@ -159,7 +159,6 @@ static bool cs4271_volatile_reg(struct device *dev, unsigned int reg)
}
struct cs4271_private {
- /* SND_SOC_I2C or SND_SOC_SPI */
unsigned int mclk;
bool master;
bool deemph;
@@ -540,14 +539,10 @@ static int cs4271_probe(struct snd_soc_codec *codec)
struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec);
struct cs4271_platform_data *cs4271plat = codec->dev->platform_data;
int ret;
- int gpio_nreset = -EINVAL;
bool amutec_eq_bmutec = false;
#ifdef CONFIG_OF
if (of_match_device(cs4271_dt_ids, codec->dev)) {
- gpio_nreset = of_get_named_gpio(codec->dev->of_node,
- "reset-gpio", 0);
-
if (of_get_property(codec->dev->of_node,
"cirrus,amutec-eq-bmutec", NULL))
amutec_eq_bmutec = true;
@@ -559,27 +554,19 @@ static int cs4271_probe(struct snd_soc_codec *codec)
#endif
if (cs4271plat) {
- if (gpio_is_valid(cs4271plat->gpio_nreset))
- gpio_nreset = cs4271plat->gpio_nreset;
-
amutec_eq_bmutec = cs4271plat->amutec_eq_bmutec;
cs4271->enable_soft_reset = cs4271plat->enable_soft_reset;
}
- if (gpio_nreset >= 0)
- if (devm_gpio_request(codec->dev, gpio_nreset, "CS4271 Reset"))
- gpio_nreset = -EINVAL;
- if (gpio_nreset >= 0) {
+ if (gpio_is_valid(cs4271->gpio_nreset)) {
/* Reset codec */
- gpio_direction_output(gpio_nreset, 0);
+ gpio_direction_output(cs4271->gpio_nreset, 0);
udelay(1);
- gpio_set_value(gpio_nreset, 1);
+ gpio_set_value(cs4271->gpio_nreset, 1);
/* Give the codec time to wake up */
udelay(1);
}
- cs4271->gpio_nreset = gpio_nreset;
-
ret = regmap_update_bits(cs4271->regmap, CS4271_MODE2,
CS4271_MODE2_PDN | CS4271_MODE2_CPEN,
CS4271_MODE2_PDN | CS4271_MODE2_CPEN);
@@ -625,6 +612,36 @@ static struct snd_soc_codec_driver soc_codec_dev_cs4271 = {
.num_dapm_routes = ARRAY_SIZE(cs4271_dapm_routes),
};
+static int cs4271_common_probe(struct device *dev,
+ struct cs4271_private **c)
+{
+ struct cs4271_platform_data *cs4271plat = dev->platform_data;
+ struct cs4271_private *cs4271;
+
+ cs4271 = devm_kzalloc(dev, sizeof(*cs4271), GFP_KERNEL);
+ if (!cs4271)
+ return -ENOMEM;
+
+ if (of_match_device(cs4271_dt_ids, dev))
+ cs4271->gpio_nreset =
+ of_get_named_gpio(dev->of_node, "reset-gpio", 0);
+
+ if (cs4271plat)
+ cs4271->gpio_nreset = cs4271plat->gpio_nreset;
+
+ if (gpio_is_valid(cs4271->gpio_nreset)) {
+ int ret;
+
+ ret = devm_gpio_request(dev, cs4271->gpio_nreset,
+ "CS4271 Reset");
+ if (ret < 0)
+ return ret;
+ }
+
+ *c = cs4271;
+ return 0;
+}
+
#if defined(CONFIG_SPI_MASTER)
static const struct regmap_config cs4271_spi_regmap = {
@@ -644,10 +661,11 @@ static const struct regmap_config cs4271_spi_regmap = {
static int cs4271_spi_probe(struct spi_device *spi)
{
struct cs4271_private *cs4271;
+ int ret;
- cs4271 = devm_kzalloc(&spi->dev, sizeof(*cs4271), GFP_KERNEL);
- if (!cs4271)
- return -ENOMEM;
+ ret = cs4271_common_probe(&spi->dev, &cs4271);
+ if (ret < 0)
+ return ret;
spi_set_drvdata(spi, cs4271);
cs4271->regmap = devm_regmap_init_spi(spi, &cs4271_spi_regmap);
@@ -698,10 +716,11 @@ static int cs4271_i2c_probe(struct i2c_client *client,
const struct i2c_device_id *id)
{
struct cs4271_private *cs4271;
+ int ret;
- cs4271 = devm_kzalloc(&client->dev, sizeof(*cs4271), GFP_KERNEL);
- if (!cs4271)
- return -ENOMEM;
+ ret = cs4271_common_probe(&client->dev, &cs4271);
+ if (ret < 0)
+ return ret;
i2c_set_clientdata(client, cs4271);
cs4271->regmap = devm_regmap_init_i2c(client, &cs4271_i2c_regmap);
diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c
index 6e9ea83..6c0da2b 100644
--- a/sound/soc/codecs/cs42l51.c
+++ b/sound/soc/codecs/cs42l51.c
@@ -30,6 +30,7 @@
#include <sound/pcm_params.h>
#include <sound/pcm.h>
#include <linux/i2c.h>
+#include <linux/regmap.h>
#include "cs42l51.h"
@@ -40,7 +41,6 @@ enum master_slave_mode {
};
struct cs42l51_private {
- enum snd_soc_control_type control_type;
unsigned int mclk;
unsigned int audio_mode; /* The mode (I2S or left-justified) */
enum master_slave_mode func;
@@ -52,24 +52,6 @@ struct cs42l51_private {
SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S20_3BE | \
SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S24_BE)
-static int cs42l51_fill_cache(struct snd_soc_codec *codec)
-{
- u8 *cache = codec->reg_cache + 1;
- struct i2c_client *i2c_client = to_i2c_client(codec->dev);
- s32 length;
-
- length = i2c_smbus_read_i2c_block_data(i2c_client,
- CS42L51_FIRSTREG | 0x80, CS42L51_NUMREGS, cache);
- if (length != CS42L51_NUMREGS) {
- dev_err(&i2c_client->dev,
- "I2C read failure, addr=0x%x (ret=%d vs %d)\n",
- i2c_client->addr, length, CS42L51_NUMREGS);
- return -EIO;
- }
-
- return 0;
-}
-
static int cs42l51_get_chan_mix(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
@@ -124,9 +106,8 @@ static int cs42l51_set_chan_mix(struct snd_kcontrol *kcontrol,
static const DECLARE_TLV_DB_SCALE(adc_pcm_tlv, -5150, 50, 0);
static const DECLARE_TLV_DB_SCALE(tone_tlv, -1050, 150, 0);
-/* This is a lie. after -102 db, it stays at -102 */
-/* maybe a range would be better */
-static const DECLARE_TLV_DB_SCALE(aout_tlv, -11550, 50, 0);
+
+static const DECLARE_TLV_DB_SCALE(aout_tlv, -10200, 50, 0);
static const DECLARE_TLV_DB_SCALE(boost_tlv, 1600, 1600, 0);
static const char *chan_mix[] = {
@@ -135,13 +116,12 @@ static const char *chan_mix[] = {
"R L",
};
-static const struct soc_enum cs42l51_chan_mix =
- SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(chan_mix), chan_mix);
+static SOC_ENUM_SINGLE_EXT_DECL(cs42l51_chan_mix, 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,
- 6, 0x19, 0x7F, adc_pcm_tlv),
+ 0, 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",
@@ -149,7 +129,7 @@ static const struct snd_kcontrol_new cs42l51_snd_controls[] = {
0, 0x34, 0xE4, aout_tlv),
SOC_DOUBLE_R_SX_TLV("ADC Mixer Volume",
CS42L51_ADCA_VOL, CS42L51_ADCB_VOL,
- 6, 0x19, 0x7F, adc_pcm_tlv),
+ 0, 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),
@@ -192,22 +172,22 @@ static int cs42l51_pdn_event(struct snd_soc_dapm_widget *w,
static const char *cs42l51_dac_names[] = {"Direct PCM",
"DSP PCM", "ADC"};
-static const struct soc_enum cs42l51_dac_mux_enum =
- SOC_ENUM_SINGLE(CS42L51_DAC_CTL, 6, 3, cs42l51_dac_names);
+static SOC_ENUM_SINGLE_DECL(cs42l51_dac_mux_enum,
+ CS42L51_DAC_CTL, 6, cs42l51_dac_names);
static const struct snd_kcontrol_new cs42l51_dac_mux_controls =
SOC_DAPM_ENUM("Route", cs42l51_dac_mux_enum);
static const char *cs42l51_adcl_names[] = {"AIN1 Left", "AIN2 Left",
"MIC Left", "MIC+preamp Left"};
-static const struct soc_enum cs42l51_adcl_mux_enum =
- SOC_ENUM_SINGLE(CS42L51_ADC_INPUT, 4, 4, cs42l51_adcl_names);
+static SOC_ENUM_SINGLE_DECL(cs42l51_adcl_mux_enum,
+ CS42L51_ADC_INPUT, 4, cs42l51_adcl_names);
static const struct snd_kcontrol_new cs42l51_adcl_mux_controls =
SOC_DAPM_ENUM("Route", cs42l51_adcl_mux_enum);
static const char *cs42l51_adcr_names[] = {"AIN1 Right", "AIN2 Right",
"MIC Right", "MIC+preamp Right"};
-static const struct soc_enum cs42l51_adcr_mux_enum =
- SOC_ENUM_SINGLE(CS42L51_ADC_INPUT, 6, 4, cs42l51_adcr_names);
+static SOC_ENUM_SINGLE_DECL(cs42l51_adcr_mux_enum,
+ CS42L51_ADC_INPUT, 6, cs42l51_adcr_names);
static const struct snd_kcontrol_new cs42l51_adcr_mux_controls =
SOC_DAPM_ENUM("Route", cs42l51_adcr_mux_enum);
@@ -505,21 +485,8 @@ static struct snd_soc_dai_driver cs42l51_dai = {
static int cs42l51_probe(struct snd_soc_codec *codec)
{
- struct cs42l51_private *cs42l51 = snd_soc_codec_get_drvdata(codec);
int ret, reg;
- ret = cs42l51_fill_cache(codec);
- if (ret < 0) {
- dev_err(codec->dev, "failed to fill register cache\n");
- return ret;
- }
-
- ret = snd_soc_codec_set_cache_io(codec, 8, 8, cs42l51->control_type);
- if (ret < 0) {
- dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
- return ret;
- }
-
/*
* DAC configuration
* - Use signal processor
@@ -538,8 +505,6 @@ static int cs42l51_probe(struct snd_soc_codec *codec)
static struct snd_soc_codec_driver soc_codec_device_cs42l51 = {
.probe = cs42l51_probe,
- .reg_cache_size = CS42L51_NUMREGS + 1,
- .reg_word_size = sizeof(u8),
.controls = cs42l51_snd_controls,
.num_controls = ARRAY_SIZE(cs42l51_snd_controls),
@@ -549,38 +514,53 @@ static struct snd_soc_codec_driver soc_codec_device_cs42l51 = {
.num_dapm_routes = ARRAY_SIZE(cs42l51_routes),
};
+static const struct regmap_config cs42l51_regmap = {
+ .reg_bits = 8,
+ .val_bits = 8,
+
+ .max_register = CS42L51_CHARGE_FREQ,
+ .cache_type = REGCACHE_RBTREE,
+};
+
static int cs42l51_i2c_probe(struct i2c_client *i2c_client,
const struct i2c_device_id *id)
{
struct cs42l51_private *cs42l51;
+ struct regmap *regmap;
+ unsigned int val;
int ret;
+ regmap = devm_regmap_init_i2c(i2c_client, &cs42l51_regmap);
+ if (IS_ERR(regmap)) {
+ ret = PTR_ERR(regmap);
+ dev_err(&i2c_client->dev, "Failed to create regmap: %d\n",
+ ret);
+ return ret;
+ }
+
/* Verify that we have a CS42L51 */
- ret = i2c_smbus_read_byte_data(i2c_client, CS42L51_CHIP_REV_ID);
+ ret = regmap_read(regmap, CS42L51_CHIP_REV_ID, &val);
if (ret < 0) {
dev_err(&i2c_client->dev, "failed to read I2C\n");
goto error;
}
- if ((ret != CS42L51_MK_CHIP_REV(CS42L51_CHIP_ID, CS42L51_CHIP_REV_A)) &&
- (ret != CS42L51_MK_CHIP_REV(CS42L51_CHIP_ID, CS42L51_CHIP_REV_B))) {
- dev_err(&i2c_client->dev, "Invalid chip id\n");
+ if ((val != CS42L51_MK_CHIP_REV(CS42L51_CHIP_ID, CS42L51_CHIP_REV_A)) &&
+ (val != CS42L51_MK_CHIP_REV(CS42L51_CHIP_ID, CS42L51_CHIP_REV_B))) {
+ dev_err(&i2c_client->dev, "Invalid chip id: %x\n", val);
ret = -ENODEV;
goto error;
}
dev_info(&i2c_client->dev, "found device cs42l51 rev %d\n",
- ret & 7);
+ val & 7);
cs42l51 = devm_kzalloc(&i2c_client->dev, sizeof(struct cs42l51_private),
GFP_KERNEL);
- if (!cs42l51) {
- dev_err(&i2c_client->dev, "could not allocate codec\n");
+ if (!cs42l51)
return -ENOMEM;
- }
i2c_set_clientdata(i2c_client, cs42l51);
- cs42l51->control_type = SND_SOC_I2C;
ret = snd_soc_register_codec(&i2c_client->dev,
&soc_codec_device_cs42l51, &cs42l51_dai, 1);
@@ -600,10 +580,17 @@ static const struct i2c_device_id cs42l51_id[] = {
};
MODULE_DEVICE_TABLE(i2c, cs42l51_id);
+static const struct of_device_id cs42l51_of_match[] = {
+ { .compatible = "cirrus,cs42l51", },
+ { }
+};
+MODULE_DEVICE_TABLE(of, cs42l51_of_match);
+
static struct i2c_driver cs42l51_i2c_driver = {
.driver = {
.name = "cs42l51-codec",
.owner = THIS_MODULE,
+ .of_match_table = cs42l51_of_match,
},
.id_table = cs42l51_id,
.probe = cs42l51_i2c_probe,
diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c
index 0bac6d5..f0ca6be 100644
--- a/sound/soc/codecs/cs42l52.c
+++ b/sound/soc/codecs/cs42l52.c
@@ -210,13 +210,11 @@ static const char * const cs42l52_adca_text[] = {
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 SOC_ENUM_SINGLE_DECL(adca_enum,
+ CS42L52_ADC_PGA_A, 5, 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 SOC_ENUM_SINGLE_DECL(adcb_enum,
+ CS42L52_ADC_PGA_B, 5, cs42l52_adcb_text);
static const struct snd_kcontrol_new adca_mux =
SOC_DAPM_ENUM("Left ADC Input Capture Mux", adca_enum);
@@ -229,26 +227,22 @@ static const char * const mic_bias_level_text[] = {
"0.8 +VA", "0.83 +VA", "0.91 +VA"
};
-static const struct soc_enum mic_bias_level_enum =
- SOC_ENUM_SINGLE(CS42L52_IFACE_CTL2, 0,
- ARRAY_SIZE(mic_bias_level_text), mic_bias_level_text);
+static SOC_ENUM_SINGLE_DECL(mic_bias_level_enum,
+ CS42L52_IFACE_CTL2, 0, mic_bias_level_text);
static const char * const cs42l52_mic_text[] = { "MIC1", "MIC2" };
-static const struct soc_enum mica_enum =
- SOC_ENUM_SINGLE(CS42L52_MICA_CTL, 5,
- ARRAY_SIZE(cs42l52_mic_text), cs42l52_mic_text);
+static SOC_ENUM_SINGLE_DECL(mica_enum,
+ CS42L52_MICA_CTL, 5, 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 SOC_ENUM_SINGLE_DECL(micb_enum,
+ CS42L52_MICB_CTL, 5, cs42l52_mic_text);
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 SOC_ENUM_SINGLE_DECL(digital_output_mux_enum,
+ CS42L52_ADC_MISC_CTL, 6,
+ digital_output_mux_text);
static const struct snd_kcontrol_new digital_output_mux =
SOC_DAPM_ENUM("Digital Output Mux", digital_output_mux_enum);
@@ -258,18 +252,18 @@ static const char * const hp_gain_num_text[] = {
"0.7099", "0.8399", "1.000", "1.1430"
};
-static const struct soc_enum hp_gain_enum =
- SOC_ENUM_SINGLE(CS42L52_PB_CTL1, 5,
- ARRAY_SIZE(hp_gain_num_text), hp_gain_num_text);
+static SOC_ENUM_SINGLE_DECL(hp_gain_enum,
+ CS42L52_PB_CTL1, 5,
+ 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 SOC_ENUM_SINGLE_DECL(beep_pitch_enum,
+ CS42L52_BEEP_FREQ, 4,
+ beep_pitch_text);
static const char * const beep_ontime_text[] = {
"86 ms", "430 ms", "780 ms", "1.20 s", "1.50 s",
@@ -277,66 +271,66 @@ static const char * const beep_ontime_text[] = {
"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 SOC_ENUM_SINGLE_DECL(beep_ontime_enum,
+ CS42L52_BEEP_FREQ, 0,
+ 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 SOC_ENUM_SINGLE_DECL(beep_offtime_enum,
+ CS42L52_BEEP_VOL, 5,
+ 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 SOC_ENUM_SINGLE_DECL(beep_config_enum,
+ CS42L52_BEEP_TONE_CTL, 6,
+ 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 SOC_ENUM_SINGLE_DECL(beep_bass_enum,
+ CS42L52_BEEP_TONE_CTL, 1,
+ 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 SOC_ENUM_SINGLE_DECL(beep_treble_enum,
+ CS42L52_BEEP_TONE_CTL, 3,
+ 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 SOC_ENUM_SINGLE_DECL(ng_threshold_enum,
+ CS42L52_NOISE_GATE_CTL, 2,
+ 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 SOC_ENUM_SINGLE_DECL(ng_delay_enum,
+ CS42L52_NOISE_GATE_CTL, 0,
+ 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 SOC_ENUM_SINGLE_DECL(ng_type_enum,
+ CS42L52_NOISE_GATE_CTL, 6,
+ cs42l52_ng_type_text);
static const char * const left_swap_text[] = {
"Left", "LR 2", "Right"};
@@ -347,7 +341,7 @@ static const char * const right_swap_text[] = {
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,
+ SOC_VALUE_ENUM_SINGLE(CS42L52_ADC_PCM_MIXER, 2, 3,
ARRAY_SIZE(left_swap_text),
left_swap_text,
swap_values);
@@ -356,7 +350,7 @@ 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,
+ SOC_VALUE_ENUM_SINGLE(CS42L52_ADC_PCM_MIXER, 6, 3,
ARRAY_SIZE(left_swap_text),
left_swap_text,
swap_values);
@@ -365,7 +359,7 @@ 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,
+ SOC_VALUE_ENUM_SINGLE(CS42L52_ADC_PCM_MIXER, 0, 3,
ARRAY_SIZE(right_swap_text),
right_swap_text,
swap_values);
@@ -374,7 +368,7 @@ 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,
+ SOC_VALUE_ENUM_SINGLE(CS42L52_ADC_PCM_MIXER, 4, 3,
ARRAY_SIZE(right_swap_text),
right_swap_text,
swap_values);
@@ -1115,14 +1109,7 @@ static void cs42l52_free_beep(struct snd_soc_codec *codec)
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_add_mic_controls(codec);
@@ -1134,7 +1121,7 @@ static int cs42l52_probe(struct snd_soc_codec *codec)
cs42l52->sysclk = CS42L52_DEFAULT_CLK;
cs42l52->config.format = CS42L52_DEFAULT_FORMAT;
- return ret;
+ return 0;
}
static int cs42l52_remove(struct snd_soc_codec *codec)
diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c
index 549d5d6..0ee60a1 100644
--- a/sound/soc/codecs/cs42l73.c
+++ b/sound/soc/codecs/cs42l73.c
@@ -278,13 +278,13 @@ static const DECLARE_TLV_DB_SCALE(attn_tlv, -6300, 100, 1);
static const char * const cs42l73_pgaa_text[] = { "Line A", "Mic 1" };
static const char * const cs42l73_pgab_text[] = { "Line B", "Mic 2" };
-static const struct soc_enum pgaa_enum =
- SOC_ENUM_SINGLE(CS42L73_ADCIPC, 3,
- ARRAY_SIZE(cs42l73_pgaa_text), cs42l73_pgaa_text);
+static SOC_ENUM_SINGLE_DECL(pgaa_enum,
+ CS42L73_ADCIPC, 3,
+ cs42l73_pgaa_text);
-static const struct soc_enum pgab_enum =
- SOC_ENUM_SINGLE(CS42L73_ADCIPC, 7,
- ARRAY_SIZE(cs42l73_pgab_text), cs42l73_pgab_text);
+static SOC_ENUM_SINGLE_DECL(pgab_enum,
+ CS42L73_ADCIPC, 7,
+ cs42l73_pgab_text);
static const struct snd_kcontrol_new pgaa_mux =
SOC_DAPM_ENUM("Left Analog Input Capture Mux", pgaa_enum);
@@ -309,9 +309,9 @@ static const struct snd_kcontrol_new input_right_mixer[] = {
static const char * const cs42l73_ng_delay_text[] = {
"50ms", "100ms", "150ms", "200ms" };
-static const struct soc_enum ng_delay_enum =
- SOC_ENUM_SINGLE(CS42L73_NGCAB, 0,
- ARRAY_SIZE(cs42l73_ng_delay_text), cs42l73_ng_delay_text);
+static SOC_ENUM_SINGLE_DECL(ng_delay_enum,
+ CS42L73_NGCAB, 0,
+ cs42l73_ng_delay_text);
static const char * const cs42l73_mono_mix_texts[] = {
"Left", "Right", "Mono Mix"};
@@ -319,7 +319,7 @@ static const char * const cs42l73_mono_mix_texts[] = {
static const unsigned int cs42l73_mono_mix_values[] = { 0, 1, 2 };
static const struct soc_enum spk_asp_enum =
- SOC_VALUE_ENUM_SINGLE(CS42L73_MMIXCTL, 6, 1,
+ SOC_VALUE_ENUM_SINGLE(CS42L73_MMIXCTL, 6, 3,
ARRAY_SIZE(cs42l73_mono_mix_texts),
cs42l73_mono_mix_texts,
cs42l73_mono_mix_values);
@@ -337,7 +337,7 @@ static const struct snd_kcontrol_new spk_xsp_mixer =
SOC_DAPM_ENUM("Route", spk_xsp_enum);
static const struct soc_enum esl_asp_enum =
- SOC_VALUE_ENUM_SINGLE(CS42L73_MMIXCTL, 2, 5,
+ SOC_VALUE_ENUM_SINGLE(CS42L73_MMIXCTL, 2, 3,
ARRAY_SIZE(cs42l73_mono_mix_texts),
cs42l73_mono_mix_texts,
cs42l73_mono_mix_values);
@@ -346,7 +346,7 @@ static const struct snd_kcontrol_new esl_asp_mixer =
SOC_DAPM_ENUM("Route", esl_asp_enum);
static const struct soc_enum esl_xsp_enum =
- SOC_VALUE_ENUM_SINGLE(CS42L73_MMIXCTL, 0, 7,
+ SOC_VALUE_ENUM_SINGLE(CS42L73_MMIXCTL, 0, 3,
ARRAY_SIZE(cs42l73_mono_mix_texts),
cs42l73_mono_mix_texts,
cs42l73_mono_mix_values);
@@ -357,19 +357,19 @@ static const struct snd_kcontrol_new esl_xsp_mixer =
static const char * const cs42l73_ip_swap_text[] = {
"Stereo", "Mono A", "Mono B", "Swap A-B"};
-static const struct soc_enum ip_swap_enum =
- SOC_ENUM_SINGLE(CS42L73_MIOPC, 6,
- ARRAY_SIZE(cs42l73_ip_swap_text), cs42l73_ip_swap_text);
+static SOC_ENUM_SINGLE_DECL(ip_swap_enum,
+ CS42L73_MIOPC, 6,
+ cs42l73_ip_swap_text);
static const char * const cs42l73_spo_mixer_text[] = {"Mono", "Stereo"};
-static const struct soc_enum vsp_output_mux_enum =
- SOC_ENUM_SINGLE(CS42L73_MIXERCTL, 5,
- ARRAY_SIZE(cs42l73_spo_mixer_text), cs42l73_spo_mixer_text);
+static SOC_ENUM_SINGLE_DECL(vsp_output_mux_enum,
+ CS42L73_MIXERCTL, 5,
+ cs42l73_spo_mixer_text);
-static const struct soc_enum xsp_output_mux_enum =
- SOC_ENUM_SINGLE(CS42L73_MIXERCTL, 4,
- ARRAY_SIZE(cs42l73_spo_mixer_text), cs42l73_spo_mixer_text);
+static SOC_ENUM_SINGLE_DECL(xsp_output_mux_enum,
+ CS42L73_MIXERCTL, 4,
+ cs42l73_spo_mixer_text);
static const struct snd_kcontrol_new vsp_output_mux =
SOC_DAPM_ENUM("Route", vsp_output_mux_enum);
@@ -1108,7 +1108,7 @@ static int cs42l73_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
return 0;
}
-static u32 cs42l73_asrc_rates[] = {
+static const unsigned int cs42l73_asrc_rates[] = {
8000, 11025, 12000, 16000, 22050,
24000, 32000, 44100, 48000
};
@@ -1241,7 +1241,7 @@ static int cs42l73_set_tristate(struct snd_soc_dai *dai, int tristate)
0x7F, tristate << 7);
}
-static struct snd_pcm_hw_constraint_list constraints_12_24 = {
+static const struct snd_pcm_hw_constraint_list constraints_12_24 = {
.count = ARRAY_SIZE(cs42l73_asrc_rates),
.list = cs42l73_asrc_rates,
};
@@ -1255,9 +1255,6 @@ static int cs42l73_pcm_startup(struct snd_pcm_substream *substream,
return 0;
}
-/* SNDRV_PCM_RATE_KNOT -> 12000, 24000 Hz, limit with constraint list */
-#define CS42L73_RATES (SNDRV_PCM_RATE_8000_48000 | SNDRV_PCM_RATE_KNOT)
-
#define CS42L73_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
SNDRV_PCM_FMTBIT_S24_LE)
@@ -1278,14 +1275,14 @@ static struct snd_soc_dai_driver cs42l73_dai[] = {
.stream_name = "XSP Playback",
.channels_min = 1,
.channels_max = 2,
- .rates = CS42L73_RATES,
+ .rates = SNDRV_PCM_RATE_KNOT,
.formats = CS42L73_FORMATS,
},
.capture = {
.stream_name = "XSP Capture",
.channels_min = 1,
.channels_max = 2,
- .rates = CS42L73_RATES,
+ .rates = SNDRV_PCM_RATE_KNOT,
.formats = CS42L73_FORMATS,
},
.ops = &cs42l73_ops,
@@ -1298,14 +1295,14 @@ static struct snd_soc_dai_driver cs42l73_dai[] = {
.stream_name = "ASP Playback",
.channels_min = 2,
.channels_max = 2,
- .rates = CS42L73_RATES,
+ .rates = SNDRV_PCM_RATE_KNOT,
.formats = CS42L73_FORMATS,
},
.capture = {
.stream_name = "ASP Capture",
.channels_min = 2,
.channels_max = 2,
- .rates = CS42L73_RATES,
+ .rates = SNDRV_PCM_RATE_KNOT,
.formats = CS42L73_FORMATS,
},
.ops = &cs42l73_ops,
@@ -1318,14 +1315,14 @@ static struct snd_soc_dai_driver cs42l73_dai[] = {
.stream_name = "VSP Playback",
.channels_min = 1,
.channels_max = 2,
- .rates = CS42L73_RATES,
+ .rates = SNDRV_PCM_RATE_KNOT,
.formats = CS42L73_FORMATS,
},
.capture = {
.stream_name = "VSP Capture",
.channels_min = 1,
.channels_max = 2,
- .rates = CS42L73_RATES,
+ .rates = SNDRV_PCM_RATE_KNOT,
.formats = CS42L73_FORMATS,
},
.ops = &cs42l73_ops,
@@ -1348,17 +1345,8 @@ static int cs42l73_resume(struct snd_soc_codec *codec)
static int cs42l73_probe(struct snd_soc_codec *codec)
{
- int ret;
struct cs42l73_private *cs42l73 = snd_soc_codec_get_drvdata(codec);
- codec->control_data = cs42l73->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;
- }
-
cs42l73_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
/* Set Charge Pump Frequency */
@@ -1371,7 +1359,7 @@ static int cs42l73_probe(struct snd_soc_codec *codec)
cs42l73->mclksel = CS42L73_CLKID_MCLK1;
cs42l73->mclk = 0;
- return ret;
+ return 0;
}
static int cs42l73_remove(struct snd_soc_codec *codec)
diff --git a/sound/soc/codecs/cs42xx8-i2c.c b/sound/soc/codecs/cs42xx8-i2c.c
new file mode 100644
index 0000000..657dce2
--- /dev/null
+++ b/sound/soc/codecs/cs42xx8-i2c.c
@@ -0,0 +1,64 @@
+/*
+ * Cirrus Logic CS42448/CS42888 Audio CODEC DAI I2C driver
+ *
+ * Copyright (C) 2014 Freescale Semiconductor, Inc.
+ *
+ * Author: Nicolin Chen <Guangyu.Chen@freescale.com>
+ *
+ * This file is licensed under the terms of the GNU General Public License
+ * version 2. This program is licensed "as is" without any warranty of any
+ * kind, whether express or implied.
+ */
+
+#include <linux/i2c.h>
+#include <linux/module.h>
+#include <linux/pm_runtime.h>
+#include <sound/soc.h>
+
+#include "cs42xx8.h"
+
+static int cs42xx8_i2c_probe(struct i2c_client *i2c,
+ const struct i2c_device_id *id)
+{
+ u32 ret = cs42xx8_probe(&i2c->dev,
+ devm_regmap_init_i2c(i2c, &cs42xx8_regmap_config));
+ if (ret)
+ return ret;
+
+ pm_runtime_enable(&i2c->dev);
+ pm_request_idle(&i2c->dev);
+
+ return 0;
+}
+
+static int cs42xx8_i2c_remove(struct i2c_client *i2c)
+{
+ snd_soc_unregister_codec(&i2c->dev);
+ pm_runtime_disable(&i2c->dev);
+
+ return 0;
+}
+
+static struct i2c_device_id cs42xx8_i2c_id[] = {
+ {"cs42448", (kernel_ulong_t)&cs42448_data},
+ {"cs42888", (kernel_ulong_t)&cs42888_data},
+ {}
+};
+MODULE_DEVICE_TABLE(i2c, cs42xx8_i2c_id);
+
+static struct i2c_driver cs42xx8_i2c_driver = {
+ .driver = {
+ .name = "cs42xx8",
+ .owner = THIS_MODULE,
+ .pm = &cs42xx8_pm,
+ },
+ .probe = cs42xx8_i2c_probe,
+ .remove = cs42xx8_i2c_remove,
+ .id_table = cs42xx8_i2c_id,
+};
+
+module_i2c_driver(cs42xx8_i2c_driver);
+
+MODULE_DESCRIPTION("Cirrus Logic CS42448/CS42888 ALSA SoC Codec I2C Driver");
+MODULE_AUTHOR("Freescale Semiconductor, Inc.");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/cs42xx8.c b/sound/soc/codecs/cs42xx8.c
new file mode 100644
index 0000000..082299a
--- /dev/null
+++ b/sound/soc/codecs/cs42xx8.c
@@ -0,0 +1,602 @@
+/*
+ * Cirrus Logic CS42448/CS42888 Audio CODEC Digital Audio Interface (DAI) driver
+ *
+ * Copyright (C) 2014 Freescale Semiconductor, Inc.
+ *
+ * Author: Nicolin Chen <Guangyu.Chen@freescale.com>
+ *
+ * This file is licensed under the terms of the GNU General Public License
+ * version 2. This program is licensed "as is" without any warranty of any
+ * kind, whether express or implied.
+ */
+
+#include <linux/clk.h>
+#include <linux/delay.h>
+#include <linux/module.h>
+#include <linux/of_device.h>
+#include <linux/pm_runtime.h>
+#include <linux/regulator/consumer.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/tlv.h>
+
+#include "cs42xx8.h"
+
+#define CS42XX8_NUM_SUPPLIES 4
+static const char *const cs42xx8_supply_names[CS42XX8_NUM_SUPPLIES] = {
+ "VA",
+ "VD",
+ "VLS",
+ "VLC",
+};
+
+#define CS42XX8_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \
+ SNDRV_PCM_FMTBIT_S20_3LE | \
+ SNDRV_PCM_FMTBIT_S24_LE | \
+ SNDRV_PCM_FMTBIT_S32_LE)
+
+/* codec private data */
+struct cs42xx8_priv {
+ struct regulator_bulk_data supplies[CS42XX8_NUM_SUPPLIES];
+ const struct cs42xx8_driver_data *drvdata;
+ struct regmap *regmap;
+ struct clk *clk;
+
+ bool slave_mode;
+ unsigned long sysclk;
+};
+
+/* -127.5dB to 0dB with step of 0.5dB */
+static const DECLARE_TLV_DB_SCALE(dac_tlv, -12750, 50, 1);
+/* -64dB to 24dB with step of 0.5dB */
+static const DECLARE_TLV_DB_SCALE(adc_tlv, -6400, 50, 0);
+
+static const char *const cs42xx8_adc_single[] = { "Differential", "Single-Ended" };
+static const char *const cs42xx8_szc[] = { "Immediate Change", "Zero Cross",
+ "Soft Ramp", "Soft Ramp on Zero Cross" };
+
+static const struct soc_enum adc1_single_enum =
+ SOC_ENUM_SINGLE(CS42XX8_ADCCTL, 4, 2, cs42xx8_adc_single);
+static const struct soc_enum adc2_single_enum =
+ SOC_ENUM_SINGLE(CS42XX8_ADCCTL, 3, 2, cs42xx8_adc_single);
+static const struct soc_enum adc3_single_enum =
+ SOC_ENUM_SINGLE(CS42XX8_ADCCTL, 2, 2, cs42xx8_adc_single);
+static const struct soc_enum dac_szc_enum =
+ SOC_ENUM_SINGLE(CS42XX8_TXCTL, 5, 4, cs42xx8_szc);
+static const struct soc_enum adc_szc_enum =
+ SOC_ENUM_SINGLE(CS42XX8_TXCTL, 0, 4, cs42xx8_szc);
+
+static const struct snd_kcontrol_new cs42xx8_snd_controls[] = {
+ SOC_DOUBLE_R_TLV("DAC1 Playback Volume", CS42XX8_VOLAOUT1,
+ CS42XX8_VOLAOUT2, 0, 0xff, 1, dac_tlv),
+ SOC_DOUBLE_R_TLV("DAC2 Playback Volume", CS42XX8_VOLAOUT3,
+ CS42XX8_VOLAOUT4, 0, 0xff, 1, dac_tlv),
+ SOC_DOUBLE_R_TLV("DAC3 Playback Volume", CS42XX8_VOLAOUT5,
+ CS42XX8_VOLAOUT6, 0, 0xff, 1, dac_tlv),
+ SOC_DOUBLE_R_TLV("DAC4 Playback Volume", CS42XX8_VOLAOUT7,
+ CS42XX8_VOLAOUT8, 0, 0xff, 1, dac_tlv),
+ SOC_DOUBLE_R_S_TLV("ADC1 Capture Volume", CS42XX8_VOLAIN1,
+ CS42XX8_VOLAIN2, 0, -0x80, 0x30, 7, 0, adc_tlv),
+ SOC_DOUBLE_R_S_TLV("ADC2 Capture Volume", CS42XX8_VOLAIN3,
+ CS42XX8_VOLAIN4, 0, -0x80, 0x30, 7, 0, adc_tlv),
+ SOC_DOUBLE("DAC1 Invert Switch", CS42XX8_DACINV, 0, 1, 1, 0),
+ SOC_DOUBLE("DAC2 Invert Switch", CS42XX8_DACINV, 2, 3, 1, 0),
+ SOC_DOUBLE("DAC3 Invert Switch", CS42XX8_DACINV, 4, 5, 1, 0),
+ SOC_DOUBLE("DAC4 Invert Switch", CS42XX8_DACINV, 6, 7, 1, 0),
+ SOC_DOUBLE("ADC1 Invert Switch", CS42XX8_ADCINV, 0, 1, 1, 0),
+ SOC_DOUBLE("ADC2 Invert Switch", CS42XX8_ADCINV, 2, 3, 1, 0),
+ SOC_SINGLE("ADC High-Pass Filter Switch", CS42XX8_ADCCTL, 7, 1, 1),
+ SOC_SINGLE("DAC De-emphasis Switch", CS42XX8_ADCCTL, 5, 1, 0),
+ SOC_ENUM("ADC1 Single Ended Mode Switch", adc1_single_enum),
+ SOC_ENUM("ADC2 Single Ended Mode Switch", adc2_single_enum),
+ SOC_SINGLE("DAC Single Volume Control Switch", CS42XX8_TXCTL, 7, 1, 0),
+ SOC_ENUM("DAC Soft Ramp & Zero Cross Control Switch", dac_szc_enum),
+ SOC_SINGLE("DAC Auto Mute Switch", CS42XX8_TXCTL, 4, 1, 0),
+ SOC_SINGLE("Mute ADC Serial Port Switch", CS42XX8_TXCTL, 3, 1, 0),
+ SOC_SINGLE("ADC Single Volume Control Switch", CS42XX8_TXCTL, 2, 1, 0),
+ SOC_ENUM("ADC Soft Ramp & Zero Cross Control Switch", adc_szc_enum),
+};
+
+static const struct snd_kcontrol_new cs42xx8_adc3_snd_controls[] = {
+ SOC_DOUBLE_R_S_TLV("ADC3 Capture Volume", CS42XX8_VOLAIN5,
+ CS42XX8_VOLAIN6, 0, -0x80, 0x30, 7, 0, adc_tlv),
+ SOC_DOUBLE("ADC3 Invert Switch", CS42XX8_ADCINV, 4, 5, 1, 0),
+ SOC_ENUM("ADC3 Single Ended Mode Switch", adc3_single_enum),
+};
+
+static const struct snd_soc_dapm_widget cs42xx8_dapm_widgets[] = {
+ SND_SOC_DAPM_DAC("DAC1", "Playback", CS42XX8_PWRCTL, 1, 1),
+ SND_SOC_DAPM_DAC("DAC2", "Playback", CS42XX8_PWRCTL, 2, 1),
+ SND_SOC_DAPM_DAC("DAC3", "Playback", CS42XX8_PWRCTL, 3, 1),
+ SND_SOC_DAPM_DAC("DAC4", "Playback", CS42XX8_PWRCTL, 4, 1),
+
+ SND_SOC_DAPM_OUTPUT("AOUT1L"),
+ SND_SOC_DAPM_OUTPUT("AOUT1R"),
+ SND_SOC_DAPM_OUTPUT("AOUT2L"),
+ SND_SOC_DAPM_OUTPUT("AOUT2R"),
+ SND_SOC_DAPM_OUTPUT("AOUT3L"),
+ SND_SOC_DAPM_OUTPUT("AOUT3R"),
+ SND_SOC_DAPM_OUTPUT("AOUT4L"),
+ SND_SOC_DAPM_OUTPUT("AOUT4R"),
+
+ SND_SOC_DAPM_ADC("ADC1", "Capture", CS42XX8_PWRCTL, 5, 1),
+ SND_SOC_DAPM_ADC("ADC2", "Capture", CS42XX8_PWRCTL, 6, 1),
+
+ SND_SOC_DAPM_INPUT("AIN1L"),
+ SND_SOC_DAPM_INPUT("AIN1R"),
+ SND_SOC_DAPM_INPUT("AIN2L"),
+ SND_SOC_DAPM_INPUT("AIN2R"),
+
+ SND_SOC_DAPM_SUPPLY("PWR", CS42XX8_PWRCTL, 0, 1, NULL, 0),
+};
+
+static const struct snd_soc_dapm_widget cs42xx8_adc3_dapm_widgets[] = {
+ SND_SOC_DAPM_ADC("ADC3", "Capture", CS42XX8_PWRCTL, 7, 1),
+
+ SND_SOC_DAPM_INPUT("AIN3L"),
+ SND_SOC_DAPM_INPUT("AIN3R"),
+};
+
+static const struct snd_soc_dapm_route cs42xx8_dapm_routes[] = {
+ /* Playback */
+ { "AOUT1L", NULL, "DAC1" },
+ { "AOUT1R", NULL, "DAC1" },
+ { "DAC1", NULL, "PWR" },
+
+ { "AOUT2L", NULL, "DAC2" },
+ { "AOUT2R", NULL, "DAC2" },
+ { "DAC2", NULL, "PWR" },
+
+ { "AOUT3L", NULL, "DAC3" },
+ { "AOUT3R", NULL, "DAC3" },
+ { "DAC3", NULL, "PWR" },
+
+ { "AOUT4L", NULL, "DAC4" },
+ { "AOUT4R", NULL, "DAC4" },
+ { "DAC4", NULL, "PWR" },
+
+ /* Capture */
+ { "ADC1", NULL, "AIN1L" },
+ { "ADC1", NULL, "AIN1R" },
+ { "ADC1", NULL, "PWR" },
+
+ { "ADC2", NULL, "AIN2L" },
+ { "ADC2", NULL, "AIN2R" },
+ { "ADC2", NULL, "PWR" },
+};
+
+static const struct snd_soc_dapm_route cs42xx8_adc3_dapm_routes[] = {
+ /* Capture */
+ { "ADC3", NULL, "AIN3L" },
+ { "ADC3", NULL, "AIN3R" },
+ { "ADC3", NULL, "PWR" },
+};
+
+struct cs42xx8_ratios {
+ unsigned int ratio;
+ unsigned char speed;
+ unsigned char mclk;
+};
+
+static const struct cs42xx8_ratios cs42xx8_ratios[] = {
+ { 64, CS42XX8_FM_QUAD, CS42XX8_FUNCMOD_MFREQ_256(4) },
+ { 96, CS42XX8_FM_QUAD, CS42XX8_FUNCMOD_MFREQ_384(4) },
+ { 128, CS42XX8_FM_QUAD, CS42XX8_FUNCMOD_MFREQ_512(4) },
+ { 192, CS42XX8_FM_QUAD, CS42XX8_FUNCMOD_MFREQ_768(4) },
+ { 256, CS42XX8_FM_SINGLE, CS42XX8_FUNCMOD_MFREQ_256(1) },
+ { 384, CS42XX8_FM_SINGLE, CS42XX8_FUNCMOD_MFREQ_384(1) },
+ { 512, CS42XX8_FM_SINGLE, CS42XX8_FUNCMOD_MFREQ_512(1) },
+ { 768, CS42XX8_FM_SINGLE, CS42XX8_FUNCMOD_MFREQ_768(1) },
+ { 1024, CS42XX8_FM_SINGLE, CS42XX8_FUNCMOD_MFREQ_1024(1) }
+};
+
+static int cs42xx8_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 cs42xx8_priv *cs42xx8 = snd_soc_codec_get_drvdata(codec);
+
+ cs42xx8->sysclk = freq;
+
+ return 0;
+}
+
+static int cs42xx8_set_dai_fmt(struct snd_soc_dai *codec_dai,
+ unsigned int format)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ struct cs42xx8_priv *cs42xx8 = snd_soc_codec_get_drvdata(codec);
+ u32 val;
+
+ /* Set DAI format */
+ switch (format & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_LEFT_J:
+ val = CS42XX8_INTF_DAC_DIF_LEFTJ | CS42XX8_INTF_ADC_DIF_LEFTJ;
+ break;
+ case SND_SOC_DAIFMT_I2S:
+ val = CS42XX8_INTF_DAC_DIF_I2S | CS42XX8_INTF_ADC_DIF_I2S;
+ break;
+ case SND_SOC_DAIFMT_RIGHT_J:
+ val = CS42XX8_INTF_DAC_DIF_RIGHTJ | CS42XX8_INTF_ADC_DIF_RIGHTJ;
+ break;
+ default:
+ dev_err(codec->dev, "unsupported dai format\n");
+ return -EINVAL;
+ }
+
+ regmap_update_bits(cs42xx8->regmap, CS42XX8_INTF,
+ CS42XX8_INTF_DAC_DIF_MASK |
+ CS42XX8_INTF_ADC_DIF_MASK, val);
+
+ /* Set master/slave audio interface */
+ switch (format & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBS_CFS:
+ cs42xx8->slave_mode = true;
+ break;
+ case SND_SOC_DAIFMT_CBM_CFM:
+ cs42xx8->slave_mode = false;
+ break;
+ default:
+ dev_err(codec->dev, "unsupported master/slave mode\n");
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static int cs42xx8_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 cs42xx8_priv *cs42xx8 = snd_soc_codec_get_drvdata(codec);
+ bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
+ u32 ratio = cs42xx8->sysclk / params_rate(params);
+ u32 i, fm, val, mask;
+
+ for (i = 0; i < ARRAY_SIZE(cs42xx8_ratios); i++) {
+ if (cs42xx8_ratios[i].ratio == ratio)
+ break;
+ }
+
+ if (i == ARRAY_SIZE(cs42xx8_ratios)) {
+ dev_err(codec->dev, "unsupported sysclk ratio\n");
+ return -EINVAL;
+ }
+
+ mask = CS42XX8_FUNCMOD_MFREQ_MASK;
+ val = cs42xx8_ratios[i].mclk;
+
+ fm = cs42xx8->slave_mode ? CS42XX8_FM_AUTO : cs42xx8_ratios[i].speed;
+
+ regmap_update_bits(cs42xx8->regmap, CS42XX8_FUNCMOD,
+ CS42XX8_FUNCMOD_xC_FM_MASK(tx) | mask,
+ CS42XX8_FUNCMOD_xC_FM(tx, fm) | val);
+
+ return 0;
+}
+
+static int cs42xx8_digital_mute(struct snd_soc_dai *dai, int mute)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ struct cs42xx8_priv *cs42xx8 = snd_soc_codec_get_drvdata(codec);
+
+ regmap_update_bits(cs42xx8->regmap, CS42XX8_DACMUTE,
+ CS42XX8_DACMUTE_ALL, mute ? CS42XX8_DACMUTE_ALL : 0);
+
+ return 0;
+}
+
+static const struct snd_soc_dai_ops cs42xx8_dai_ops = {
+ .set_fmt = cs42xx8_set_dai_fmt,
+ .set_sysclk = cs42xx8_set_dai_sysclk,
+ .hw_params = cs42xx8_hw_params,
+ .digital_mute = cs42xx8_digital_mute,
+};
+
+static struct snd_soc_dai_driver cs42xx8_dai = {
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 1,
+ .channels_max = 8,
+ .rates = SNDRV_PCM_RATE_8000_192000,
+ .formats = CS42XX8_FORMATS,
+ },
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 1,
+ .rates = SNDRV_PCM_RATE_8000_192000,
+ .formats = CS42XX8_FORMATS,
+ },
+ .ops = &cs42xx8_dai_ops,
+};
+
+static const struct reg_default cs42xx8_reg[] = {
+ { 0x01, 0x01 }, /* Chip I.D. and Revision Register */
+ { 0x02, 0x00 }, /* Power Control */
+ { 0x03, 0xF0 }, /* Functional Mode */
+ { 0x04, 0x46 }, /* Interface Formats */
+ { 0x05, 0x00 }, /* ADC Control & DAC De-Emphasis */
+ { 0x06, 0x10 }, /* Transition Control */
+ { 0x07, 0x00 }, /* DAC Channel Mute */
+ { 0x08, 0x00 }, /* Volume Control AOUT1 */
+ { 0x09, 0x00 }, /* Volume Control AOUT2 */
+ { 0x0a, 0x00 }, /* Volume Control AOUT3 */
+ { 0x0b, 0x00 }, /* Volume Control AOUT4 */
+ { 0x0c, 0x00 }, /* Volume Control AOUT5 */
+ { 0x0d, 0x00 }, /* Volume Control AOUT6 */
+ { 0x0e, 0x00 }, /* Volume Control AOUT7 */
+ { 0x0f, 0x00 }, /* Volume Control AOUT8 */
+ { 0x10, 0x00 }, /* DAC Channel Invert */
+ { 0x11, 0x00 }, /* Volume Control AIN1 */
+ { 0x12, 0x00 }, /* Volume Control AIN2 */
+ { 0x13, 0x00 }, /* Volume Control AIN3 */
+ { 0x14, 0x00 }, /* Volume Control AIN4 */
+ { 0x15, 0x00 }, /* Volume Control AIN5 */
+ { 0x16, 0x00 }, /* Volume Control AIN6 */
+ { 0x17, 0x00 }, /* ADC Channel Invert */
+ { 0x18, 0x00 }, /* Status Control */
+ { 0x1a, 0x00 }, /* Status Mask */
+ { 0x1b, 0x00 }, /* MUTEC Pin Control */
+};
+
+static bool cs42xx8_volatile_register(struct device *dev, unsigned int reg)
+{
+ switch (reg) {
+ case CS42XX8_STATUS:
+ return true;
+ default:
+ return false;
+ }
+}
+
+static bool cs42xx8_writeable_register(struct device *dev, unsigned int reg)
+{
+ switch (reg) {
+ case CS42XX8_CHIPID:
+ case CS42XX8_STATUS:
+ return false;
+ default:
+ return true;
+ }
+}
+
+const struct regmap_config cs42xx8_regmap_config = {
+ .reg_bits = 8,
+ .val_bits = 8,
+
+ .max_register = CS42XX8_LASTREG,
+ .reg_defaults = cs42xx8_reg,
+ .num_reg_defaults = ARRAY_SIZE(cs42xx8_reg),
+ .volatile_reg = cs42xx8_volatile_register,
+ .writeable_reg = cs42xx8_writeable_register,
+ .cache_type = REGCACHE_RBTREE,
+};
+EXPORT_SYMBOL_GPL(cs42xx8_regmap_config);
+
+static int cs42xx8_codec_probe(struct snd_soc_codec *codec)
+{
+ struct cs42xx8_priv *cs42xx8 = snd_soc_codec_get_drvdata(codec);
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
+
+ switch (cs42xx8->drvdata->num_adcs) {
+ case 3:
+ snd_soc_add_codec_controls(codec, cs42xx8_adc3_snd_controls,
+ ARRAY_SIZE(cs42xx8_adc3_snd_controls));
+ snd_soc_dapm_new_controls(dapm, cs42xx8_adc3_dapm_widgets,
+ ARRAY_SIZE(cs42xx8_adc3_dapm_widgets));
+ snd_soc_dapm_add_routes(dapm, cs42xx8_adc3_dapm_routes,
+ ARRAY_SIZE(cs42xx8_adc3_dapm_routes));
+ break;
+ default:
+ break;
+ }
+
+ /* Mute all DAC channels */
+ regmap_write(cs42xx8->regmap, CS42XX8_DACMUTE, CS42XX8_DACMUTE_ALL);
+
+ return 0;
+}
+
+static const struct snd_soc_codec_driver cs42xx8_driver = {
+ .probe = cs42xx8_codec_probe,
+ .idle_bias_off = true,
+
+ .controls = cs42xx8_snd_controls,
+ .num_controls = ARRAY_SIZE(cs42xx8_snd_controls),
+ .dapm_widgets = cs42xx8_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(cs42xx8_dapm_widgets),
+ .dapm_routes = cs42xx8_dapm_routes,
+ .num_dapm_routes = ARRAY_SIZE(cs42xx8_dapm_routes),
+};
+
+const struct cs42xx8_driver_data cs42448_data = {
+ .name = "cs42448",
+ .num_adcs = 3,
+};
+EXPORT_SYMBOL_GPL(cs42448_data);
+
+const struct cs42xx8_driver_data cs42888_data = {
+ .name = "cs42888",
+ .num_adcs = 2,
+};
+EXPORT_SYMBOL_GPL(cs42888_data);
+
+const struct of_device_id cs42xx8_of_match[] = {
+ { .compatible = "cirrus,cs42448", .data = &cs42448_data, },
+ { .compatible = "cirrus,cs42888", .data = &cs42888_data, },
+ { /* sentinel */ }
+};
+MODULE_DEVICE_TABLE(of, cs42xx8_of_match);
+EXPORT_SYMBOL_GPL(cs42xx8_of_match);
+
+int cs42xx8_probe(struct device *dev, struct regmap *regmap)
+{
+ const struct of_device_id *of_id = of_match_device(cs42xx8_of_match, dev);
+ struct cs42xx8_priv *cs42xx8;
+ int ret, val, i;
+
+ cs42xx8 = devm_kzalloc(dev, sizeof(*cs42xx8), GFP_KERNEL);
+ if (cs42xx8 == NULL)
+ return -ENOMEM;
+
+ dev_set_drvdata(dev, cs42xx8);
+
+ if (of_id)
+ cs42xx8->drvdata = of_id->data;
+
+ if (!cs42xx8->drvdata) {
+ dev_err(dev, "failed to find driver data\n");
+ return -EINVAL;
+ }
+
+ cs42xx8->clk = devm_clk_get(dev, "mclk");
+ if (IS_ERR(cs42xx8->clk)) {
+ dev_err(dev, "failed to get the clock: %ld\n",
+ PTR_ERR(cs42xx8->clk));
+ return -EINVAL;
+ }
+
+ cs42xx8->sysclk = clk_get_rate(cs42xx8->clk);
+
+ for (i = 0; i < ARRAY_SIZE(cs42xx8->supplies); i++)
+ cs42xx8->supplies[i].supply = cs42xx8_supply_names[i];
+
+ ret = devm_regulator_bulk_get(dev,
+ ARRAY_SIZE(cs42xx8->supplies), cs42xx8->supplies);
+ if (ret) {
+ dev_err(dev, "failed to request supplies: %d\n", ret);
+ return ret;
+ }
+
+ ret = regulator_bulk_enable(ARRAY_SIZE(cs42xx8->supplies),
+ cs42xx8->supplies);
+ if (ret) {
+ dev_err(dev, "failed to enable supplies: %d\n", ret);
+ return ret;
+ }
+
+ /* Make sure hardware reset done */
+ msleep(5);
+
+ cs42xx8->regmap = regmap;
+ if (IS_ERR(cs42xx8->regmap)) {
+ ret = PTR_ERR(cs42xx8->regmap);
+ dev_err(dev, "failed to allocate regmap: %d\n", ret);
+ goto err_enable;
+ }
+
+ /*
+ * We haven't marked the chip revision as volatile due to
+ * sharing a register with the right input volume; explicitly
+ * bypass the cache to read it.
+ */
+ regcache_cache_bypass(cs42xx8->regmap, true);
+
+ /* Validate the chip ID */
+ regmap_read(cs42xx8->regmap, CS42XX8_CHIPID, &val);
+ if (val < 0) {
+ dev_err(dev, "failed to get device ID: %x", val);
+ ret = -EINVAL;
+ goto err_enable;
+ }
+
+ /* The top four bits of the chip ID should be 0000 */
+ if ((val & CS42XX8_CHIPID_CHIP_ID_MASK) != 0x00) {
+ dev_err(dev, "unmatched chip ID: %d\n",
+ val & CS42XX8_CHIPID_CHIP_ID_MASK);
+ ret = -EINVAL;
+ goto err_enable;
+ }
+
+ dev_info(dev, "found device, revision %X\n",
+ val & CS42XX8_CHIPID_REV_ID_MASK);
+
+ regcache_cache_bypass(cs42xx8->regmap, false);
+
+ cs42xx8_dai.name = cs42xx8->drvdata->name;
+
+ /* Each adc supports stereo input */
+ cs42xx8_dai.capture.channels_max = cs42xx8->drvdata->num_adcs * 2;
+
+ ret = snd_soc_register_codec(dev, &cs42xx8_driver, &cs42xx8_dai, 1);
+ if (ret) {
+ dev_err(dev, "failed to register codec:%d\n", ret);
+ goto err_enable;
+ }
+
+ regcache_cache_only(cs42xx8->regmap, true);
+
+err_enable:
+ regulator_bulk_disable(ARRAY_SIZE(cs42xx8->supplies),
+ cs42xx8->supplies);
+
+ return ret;
+}
+EXPORT_SYMBOL_GPL(cs42xx8_probe);
+
+#ifdef CONFIG_PM_RUNTIME
+static int cs42xx8_runtime_resume(struct device *dev)
+{
+ struct cs42xx8_priv *cs42xx8 = dev_get_drvdata(dev);
+ int ret;
+
+ ret = clk_prepare_enable(cs42xx8->clk);
+ if (ret) {
+ dev_err(dev, "failed to enable mclk: %d\n", ret);
+ return ret;
+ }
+
+ ret = regulator_bulk_enable(ARRAY_SIZE(cs42xx8->supplies),
+ cs42xx8->supplies);
+ if (ret) {
+ dev_err(dev, "failed to enable supplies: %d\n", ret);
+ goto err_clk;
+ }
+
+ /* Make sure hardware reset done */
+ msleep(5);
+
+ regcache_cache_only(cs42xx8->regmap, false);
+
+ ret = regcache_sync(cs42xx8->regmap);
+ if (ret) {
+ dev_err(dev, "failed to sync regmap: %d\n", ret);
+ goto err_bulk;
+ }
+
+ return 0;
+
+err_bulk:
+ regulator_bulk_disable(ARRAY_SIZE(cs42xx8->supplies),
+ cs42xx8->supplies);
+err_clk:
+ clk_disable_unprepare(cs42xx8->clk);
+
+ return ret;
+}
+
+static int cs42xx8_runtime_suspend(struct device *dev)
+{
+ struct cs42xx8_priv *cs42xx8 = dev_get_drvdata(dev);
+
+ regcache_cache_only(cs42xx8->regmap, true);
+
+ regulator_bulk_disable(ARRAY_SIZE(cs42xx8->supplies),
+ cs42xx8->supplies);
+
+ clk_disable_unprepare(cs42xx8->clk);
+
+ return 0;
+}
+#endif
+
+const struct dev_pm_ops cs42xx8_pm = {
+ SET_RUNTIME_PM_OPS(cs42xx8_runtime_suspend, cs42xx8_runtime_resume, NULL)
+};
+EXPORT_SYMBOL_GPL(cs42xx8_pm);
+
+MODULE_DESCRIPTION("Cirrus Logic CS42448/CS42888 ALSA SoC Codec Driver");
+MODULE_AUTHOR("Freescale Semiconductor, Inc.");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/cs42xx8.h b/sound/soc/codecs/cs42xx8.h
new file mode 100644
index 0000000..da0b94a
--- /dev/null
+++ b/sound/soc/codecs/cs42xx8.h
@@ -0,0 +1,238 @@
+/*
+ * cs42xx8.h - Cirrus Logic CS42448/CS42888 Audio CODEC driver header file
+ *
+ * Copyright (C) 2014 Freescale Semiconductor, Inc.
+ *
+ * Author: Nicolin Chen <Guangyu.Chen@freescale.com>
+ *
+ * This file is licensed under the terms of the GNU General Public License
+ * version 2. This program is licensed "as is" without any warranty of any
+ * kind, whether express or implied.
+ */
+
+#ifndef _CS42XX8_H
+#define _CS42XX8_H
+
+struct cs42xx8_driver_data {
+ char name[32];
+ int num_adcs;
+};
+
+extern const struct dev_pm_ops cs42xx8_pm;
+extern const struct cs42xx8_driver_data cs42448_data;
+extern const struct cs42xx8_driver_data cs42888_data;
+extern const struct regmap_config cs42xx8_regmap_config;
+int cs42xx8_probe(struct device *dev, struct regmap *regmap);
+
+/* CS42888 register map */
+#define CS42XX8_CHIPID 0x01 /* Chip ID */
+#define CS42XX8_PWRCTL 0x02 /* Power Control */
+#define CS42XX8_FUNCMOD 0x03 /* Functional Mode */
+#define CS42XX8_INTF 0x04 /* Interface Formats */
+#define CS42XX8_ADCCTL 0x05 /* ADC Control */
+#define CS42XX8_TXCTL 0x06 /* Transition Control */
+#define CS42XX8_DACMUTE 0x07 /* DAC Mute Control */
+#define CS42XX8_VOLAOUT1 0x08 /* Volume Control AOUT1 */
+#define CS42XX8_VOLAOUT2 0x09 /* Volume Control AOUT2 */
+#define CS42XX8_VOLAOUT3 0x0A /* Volume Control AOUT3 */
+#define CS42XX8_VOLAOUT4 0x0B /* Volume Control AOUT4 */
+#define CS42XX8_VOLAOUT5 0x0C /* Volume Control AOUT5 */
+#define CS42XX8_VOLAOUT6 0x0D /* Volume Control AOUT6 */
+#define CS42XX8_VOLAOUT7 0x0E /* Volume Control AOUT7 */
+#define CS42XX8_VOLAOUT8 0x0F /* Volume Control AOUT8 */
+#define CS42XX8_DACINV 0x10 /* DAC Channel Invert */
+#define CS42XX8_VOLAIN1 0x11 /* Volume Control AIN1 */
+#define CS42XX8_VOLAIN2 0x12 /* Volume Control AIN2 */
+#define CS42XX8_VOLAIN3 0x13 /* Volume Control AIN3 */
+#define CS42XX8_VOLAIN4 0x14 /* Volume Control AIN4 */
+#define CS42XX8_VOLAIN5 0x15 /* Volume Control AIN5 */
+#define CS42XX8_VOLAIN6 0x16 /* Volume Control AIN6 */
+#define CS42XX8_ADCINV 0x17 /* ADC Channel Invert */
+#define CS42XX8_STATUSCTL 0x18 /* Status Control */
+#define CS42XX8_STATUS 0x19 /* Status */
+#define CS42XX8_STATUSM 0x1A /* Status Mask */
+#define CS42XX8_MUTEC 0x1B /* MUTEC Pin Control */
+
+#define CS42XX8_FIRSTREG CS42XX8_CHIPID
+#define CS42XX8_LASTREG CS42XX8_MUTEC
+#define CS42XX8_NUMREGS (CS42XX8_LASTREG - CS42XX8_FIRSTREG + 1)
+#define CS42XX8_I2C_INCR 0x80
+
+/* Chip I.D. and Revision Register (Address 01h) */
+#define CS42XX8_CHIPID_CHIP_ID_MASK 0xF0
+#define CS42XX8_CHIPID_REV_ID_MASK 0x0F
+
+/* Power Control (Address 02h) */
+#define CS42XX8_PWRCTL_PDN_ADC3_SHIFT 7
+#define CS42XX8_PWRCTL_PDN_ADC3_MASK (1 << CS42XX8_PWRCTL_PDN_ADC3_SHIFT)
+#define CS42XX8_PWRCTL_PDN_ADC3 (1 << CS42XX8_PWRCTL_PDN_ADC3_SHIFT)
+#define CS42XX8_PWRCTL_PDN_ADC2_SHIFT 6
+#define CS42XX8_PWRCTL_PDN_ADC2_MASK (1 << CS42XX8_PWRCTL_PDN_ADC2_SHIFT)
+#define CS42XX8_PWRCTL_PDN_ADC2 (1 << CS42XX8_PWRCTL_PDN_ADC2_SHIFT)
+#define CS42XX8_PWRCTL_PDN_ADC1_SHIFT 5
+#define CS42XX8_PWRCTL_PDN_ADC1_MASK (1 << CS42XX8_PWRCTL_PDN_ADC1_SHIFT)
+#define CS42XX8_PWRCTL_PDN_ADC1 (1 << CS42XX8_PWRCTL_PDN_ADC1_SHIFT)
+#define CS42XX8_PWRCTL_PDN_DAC4_SHIFT 4
+#define CS42XX8_PWRCTL_PDN_DAC4_MASK (1 << CS42XX8_PWRCTL_PDN_DAC4_SHIFT)
+#define CS42XX8_PWRCTL_PDN_DAC4 (1 << CS42XX8_PWRCTL_PDN_DAC4_SHIFT)
+#define CS42XX8_PWRCTL_PDN_DAC3_SHIFT 3
+#define CS42XX8_PWRCTL_PDN_DAC3_MASK (1 << CS42XX8_PWRCTL_PDN_DAC3_SHIFT)
+#define CS42XX8_PWRCTL_PDN_DAC3 (1 << CS42XX8_PWRCTL_PDN_DAC3_SHIFT)
+#define CS42XX8_PWRCTL_PDN_DAC2_SHIFT 2
+#define CS42XX8_PWRCTL_PDN_DAC2_MASK (1 << CS42XX8_PWRCTL_PDN_DAC2_SHIFT)
+#define CS42XX8_PWRCTL_PDN_DAC2 (1 << CS42XX8_PWRCTL_PDN_DAC2_SHIFT)
+#define CS42XX8_PWRCTL_PDN_DAC1_SHIFT 1
+#define CS42XX8_PWRCTL_PDN_DAC1_MASK (1 << CS42XX8_PWRCTL_PDN_DAC1_SHIFT)
+#define CS42XX8_PWRCTL_PDN_DAC1 (1 << CS42XX8_PWRCTL_PDN_DAC1_SHIFT)
+#define CS42XX8_PWRCTL_PDN_SHIFT 0
+#define CS42XX8_PWRCTL_PDN_MASK (1 << CS42XX8_PWRCTL_PDN_SHIFT)
+#define CS42XX8_PWRCTL_PDN (1 << CS42XX8_PWRCTL_PDN_SHIFT)
+
+/* Functional Mode (Address 03h) */
+#define CS42XX8_FUNCMOD_DAC_FM_SHIFT 6
+#define CS42XX8_FUNCMOD_DAC_FM_WIDTH 2
+#define CS42XX8_FUNCMOD_DAC_FM_MASK (((1 << CS42XX8_FUNCMOD_DAC_FM_WIDTH) - 1) << CS42XX8_FUNCMOD_DAC_FM_SHIFT)
+#define CS42XX8_FUNCMOD_DAC_FM(v) ((v) << CS42XX8_FUNCMOD_DAC_FM_SHIFT)
+#define CS42XX8_FUNCMOD_ADC_FM_SHIFT 4
+#define CS42XX8_FUNCMOD_ADC_FM_WIDTH 2
+#define CS42XX8_FUNCMOD_ADC_FM_MASK (((1 << CS42XX8_FUNCMOD_ADC_FM_WIDTH) - 1) << CS42XX8_FUNCMOD_ADC_FM_SHIFT)
+#define CS42XX8_FUNCMOD_ADC_FM(v) ((v) << CS42XX8_FUNCMOD_ADC_FM_SHIFT)
+#define CS42XX8_FUNCMOD_xC_FM_MASK(x) ((x) ? CS42XX8_FUNCMOD_DAC_FM_MASK : CS42XX8_FUNCMOD_ADC_FM_MASK)
+#define CS42XX8_FUNCMOD_xC_FM(x, v) ((x) ? CS42XX8_FUNCMOD_DAC_FM(v) : CS42XX8_FUNCMOD_ADC_FM(v))
+#define CS42XX8_FUNCMOD_MFREQ_SHIFT 1
+#define CS42XX8_FUNCMOD_MFREQ_WIDTH 3
+#define CS42XX8_FUNCMOD_MFREQ_MASK (((1 << CS42XX8_FUNCMOD_MFREQ_WIDTH) - 1) << CS42XX8_FUNCMOD_MFREQ_SHIFT)
+#define CS42XX8_FUNCMOD_MFREQ_256(s) ((0 << CS42XX8_FUNCMOD_MFREQ_SHIFT) >> (s >> 1))
+#define CS42XX8_FUNCMOD_MFREQ_384(s) ((1 << CS42XX8_FUNCMOD_MFREQ_SHIFT) >> (s >> 1))
+#define CS42XX8_FUNCMOD_MFREQ_512(s) ((2 << CS42XX8_FUNCMOD_MFREQ_SHIFT) >> (s >> 1))
+#define CS42XX8_FUNCMOD_MFREQ_768(s) ((3 << CS42XX8_FUNCMOD_MFREQ_SHIFT) >> (s >> 1))
+#define CS42XX8_FUNCMOD_MFREQ_1024(s) ((4 << CS42XX8_FUNCMOD_MFREQ_SHIFT) >> (s >> 1))
+
+#define CS42XX8_FM_SINGLE 0
+#define CS42XX8_FM_DOUBLE 1
+#define CS42XX8_FM_QUAD 2
+#define CS42XX8_FM_AUTO 3
+
+/* Interface Formats (Address 04h) */
+#define CS42XX8_INTF_FREEZE_SHIFT 7
+#define CS42XX8_INTF_FREEZE_MASK (1 << CS42XX8_INTF_FREEZE_SHIFT)
+#define CS42XX8_INTF_FREEZE (1 << CS42XX8_INTF_FREEZE_SHIFT)
+#define CS42XX8_INTF_AUX_DIF_SHIFT 6
+#define CS42XX8_INTF_AUX_DIF_MASK (1 << CS42XX8_INTF_AUX_DIF_SHIFT)
+#define CS42XX8_INTF_AUX_DIF (1 << CS42XX8_INTF_AUX_DIF_SHIFT)
+#define CS42XX8_INTF_DAC_DIF_SHIFT 3
+#define CS42XX8_INTF_DAC_DIF_WIDTH 3
+#define CS42XX8_INTF_DAC_DIF_MASK (((1 << CS42XX8_INTF_DAC_DIF_WIDTH) - 1) << CS42XX8_INTF_DAC_DIF_SHIFT)
+#define CS42XX8_INTF_DAC_DIF_LEFTJ (0 << CS42XX8_INTF_DAC_DIF_SHIFT)
+#define CS42XX8_INTF_DAC_DIF_I2S (1 << CS42XX8_INTF_DAC_DIF_SHIFT)
+#define CS42XX8_INTF_DAC_DIF_RIGHTJ (2 << CS42XX8_INTF_DAC_DIF_SHIFT)
+#define CS42XX8_INTF_DAC_DIF_RIGHTJ_16 (3 << CS42XX8_INTF_DAC_DIF_SHIFT)
+#define CS42XX8_INTF_DAC_DIF_ONELINE_20 (4 << CS42XX8_INTF_DAC_DIF_SHIFT)
+#define CS42XX8_INTF_DAC_DIF_ONELINE_24 (6 << CS42XX8_INTF_DAC_DIF_SHIFT)
+#define CS42XX8_INTF_DAC_DIF_TDM (7 << CS42XX8_INTF_DAC_DIF_SHIFT)
+#define CS42XX8_INTF_ADC_DIF_SHIFT 0
+#define CS42XX8_INTF_ADC_DIF_WIDTH 3
+#define CS42XX8_INTF_ADC_DIF_MASK (((1 << CS42XX8_INTF_ADC_DIF_WIDTH) - 1) << CS42XX8_INTF_ADC_DIF_SHIFT)
+#define CS42XX8_INTF_ADC_DIF_LEFTJ (0 << CS42XX8_INTF_ADC_DIF_SHIFT)
+#define CS42XX8_INTF_ADC_DIF_I2S (1 << CS42XX8_INTF_ADC_DIF_SHIFT)
+#define CS42XX8_INTF_ADC_DIF_RIGHTJ (2 << CS42XX8_INTF_ADC_DIF_SHIFT)
+#define CS42XX8_INTF_ADC_DIF_RIGHTJ_16 (3 << CS42XX8_INTF_ADC_DIF_SHIFT)
+#define CS42XX8_INTF_ADC_DIF_ONELINE_20 (4 << CS42XX8_INTF_ADC_DIF_SHIFT)
+#define CS42XX8_INTF_ADC_DIF_ONELINE_24 (6 << CS42XX8_INTF_ADC_DIF_SHIFT)
+#define CS42XX8_INTF_ADC_DIF_TDM (7 << CS42XX8_INTF_ADC_DIF_SHIFT)
+
+/* ADC Control & DAC De-Emphasis (Address 05h) */
+#define CS42XX8_ADCCTL_ADC_HPF_FREEZE_SHIFT 7
+#define CS42XX8_ADCCTL_ADC_HPF_FREEZE_MASK (1 << CS42XX8_ADCCTL_ADC_HPF_FREEZE_SHIFT)
+#define CS42XX8_ADCCTL_ADC_HPF_FREEZE (1 << CS42XX8_ADCCTL_ADC_HPF_FREEZE_SHIFT)
+#define CS42XX8_ADCCTL_DAC_DEM_SHIFT 5
+#define CS42XX8_ADCCTL_DAC_DEM_MASK (1 << CS42XX8_ADCCTL_DAC_DEM_SHIFT)
+#define CS42XX8_ADCCTL_DAC_DEM (1 << CS42XX8_ADCCTL_DAC_DEM_SHIFT)
+#define CS42XX8_ADCCTL_ADC1_SINGLE_SHIFT 4
+#define CS42XX8_ADCCTL_ADC1_SINGLE_MASK (1 << CS42XX8_ADCCTL_ADC1_SINGLE_SHIFT)
+#define CS42XX8_ADCCTL_ADC1_SINGLE (1 << CS42XX8_ADCCTL_ADC1_SINGLE_SHIFT)
+#define CS42XX8_ADCCTL_ADC2_SINGLE_SHIFT 3
+#define CS42XX8_ADCCTL_ADC2_SINGLE_MASK (1 << CS42XX8_ADCCTL_ADC2_SINGLE_SHIFT)
+#define CS42XX8_ADCCTL_ADC2_SINGLE (1 << CS42XX8_ADCCTL_ADC2_SINGLE_SHIFT)
+#define CS42XX8_ADCCTL_ADC3_SINGLE_SHIFT 2
+#define CS42XX8_ADCCTL_ADC3_SINGLE_MASK (1 << CS42XX8_ADCCTL_ADC3_SINGLE_SHIFT)
+#define CS42XX8_ADCCTL_ADC3_SINGLE (1 << CS42XX8_ADCCTL_ADC3_SINGLE_SHIFT)
+#define CS42XX8_ADCCTL_AIN5_MUX_SHIFT 1
+#define CS42XX8_ADCCTL_AIN5_MUX_MASK (1 << CS42XX8_ADCCTL_AIN5_MUX_SHIFT)
+#define CS42XX8_ADCCTL_AIN5_MUX (1 << CS42XX8_ADCCTL_AIN5_MUX_SHIFT)
+#define CS42XX8_ADCCTL_AIN6_MUX_SHIFT 0
+#define CS42XX8_ADCCTL_AIN6_MUX_MASK (1 << CS42XX8_ADCCTL_AIN6_MUX_SHIFT)
+#define CS42XX8_ADCCTL_AIN6_MUX (1 << CS42XX8_ADCCTL_AIN6_MUX_SHIFT)
+
+/* Transition Control (Address 06h) */
+#define CS42XX8_TXCTL_DAC_SNGVOL_SHIFT 7
+#define CS42XX8_TXCTL_DAC_SNGVOL_MASK (1 << CS42XX8_TXCTL_DAC_SNGVOL_SHIFT)
+#define CS42XX8_TXCTL_DAC_SNGVOL (1 << CS42XX8_TXCTL_DAC_SNGVOL_SHIFT)
+#define CS42XX8_TXCTL_DAC_SZC_SHIFT 5
+#define CS42XX8_TXCTL_DAC_SZC_WIDTH 2
+#define CS42XX8_TXCTL_DAC_SZC_MASK (((1 << CS42XX8_TXCTL_DAC_SZC_WIDTH) - 1) << CS42XX8_TXCTL_DAC_SZC_SHIFT)
+#define CS42XX8_TXCTL_DAC_SZC_IC (0 << CS42XX8_TXCTL_DAC_SZC_SHIFT)
+#define CS42XX8_TXCTL_DAC_SZC_ZC (1 << CS42XX8_TXCTL_DAC_SZC_SHIFT)
+#define CS42XX8_TXCTL_DAC_SZC_SR (2 << CS42XX8_TXCTL_DAC_SZC_SHIFT)
+#define CS42XX8_TXCTL_DAC_SZC_SRZC (3 << CS42XX8_TXCTL_DAC_SZC_SHIFT)
+#define CS42XX8_TXCTL_AMUTE_SHIFT 4
+#define CS42XX8_TXCTL_AMUTE_MASK (1 << CS42XX8_TXCTL_AMUTE_SHIFT)
+#define CS42XX8_TXCTL_AMUTE (1 << CS42XX8_TXCTL_AMUTE_SHIFT)
+#define CS42XX8_TXCTL_MUTE_ADC_SP_SHIFT 3
+#define CS42XX8_TXCTL_MUTE_ADC_SP_MASK (1 << CS42XX8_TXCTL_MUTE_ADC_SP_SHIFT)
+#define CS42XX8_TXCTL_MUTE_ADC_SP (1 << CS42XX8_TXCTL_MUTE_ADC_SP_SHIFT)
+#define CS42XX8_TXCTL_ADC_SNGVOL_SHIFT 2
+#define CS42XX8_TXCTL_ADC_SNGVOL_MASK (1 << CS42XX8_TXCTL_ADC_SNGVOL_SHIFT)
+#define CS42XX8_TXCTL_ADC_SNGVOL (1 << CS42XX8_TXCTL_ADC_SNGVOL_SHIFT)
+#define CS42XX8_TXCTL_ADC_SZC_SHIFT 0
+#define CS42XX8_TXCTL_ADC_SZC_MASK (((1 << CS42XX8_TXCTL_ADC_SZC_WIDTH) - 1) << CS42XX8_TXCTL_ADC_SZC_SHIFT)
+#define CS42XX8_TXCTL_ADC_SZC_IC (0 << CS42XX8_TXCTL_ADC_SZC_SHIFT)
+#define CS42XX8_TXCTL_ADC_SZC_ZC (1 << CS42XX8_TXCTL_ADC_SZC_SHIFT)
+#define CS42XX8_TXCTL_ADC_SZC_SR (2 << CS42XX8_TXCTL_ADC_SZC_SHIFT)
+#define CS42XX8_TXCTL_ADC_SZC_SRZC (3 << CS42XX8_TXCTL_ADC_SZC_SHIFT)
+
+/* DAC Channel Mute (Address 07h) */
+#define CS42XX8_DACMUTE_AOUT(n) (0x1 << n)
+#define CS42XX8_DACMUTE_ALL 0xff
+
+/* Status Control (Address 18h)*/
+#define CS42XX8_STATUSCTL_INI_SHIFT 2
+#define CS42XX8_STATUSCTL_INI_WIDTH 2
+#define CS42XX8_STATUSCTL_INI_MASK (((1 << CS42XX8_STATUSCTL_INI_WIDTH) - 1) << CS42XX8_STATUSCTL_INI_SHIFT)
+#define CS42XX8_STATUSCTL_INT_ACTIVE_HIGH (0 << CS42XX8_STATUSCTL_INI_SHIFT)
+#define CS42XX8_STATUSCTL_INT_ACTIVE_LOW (1 << CS42XX8_STATUSCTL_INI_SHIFT)
+#define CS42XX8_STATUSCTL_INT_OPEN_DRAIN (2 << CS42XX8_STATUSCTL_INI_SHIFT)
+
+/* Status (Address 19h)*/
+#define CS42XX8_STATUS_DAC_CLK_ERR_SHIFT 4
+#define CS42XX8_STATUS_DAC_CLK_ERR_MASK (1 << CS42XX8_STATUS_DAC_CLK_ERR_SHIFT)
+#define CS42XX8_STATUS_ADC_CLK_ERR_SHIFT 3
+#define CS42XX8_STATUS_ADC_CLK_ERR_MASK (1 << CS42XX8_STATUS_ADC_CLK_ERR_SHIFT)
+#define CS42XX8_STATUS_ADC3_OVFL_SHIFT 2
+#define CS42XX8_STATUS_ADC3_OVFL_MASK (1 << CS42XX8_STATUS_ADC3_OVFL_SHIFT)
+#define CS42XX8_STATUS_ADC2_OVFL_SHIFT 1
+#define CS42XX8_STATUS_ADC2_OVFL_MASK (1 << CS42XX8_STATUS_ADC2_OVFL_SHIFT)
+#define CS42XX8_STATUS_ADC1_OVFL_SHIFT 0
+#define CS42XX8_STATUS_ADC1_OVFL_MASK (1 << CS42XX8_STATUS_ADC1_OVFL_SHIFT)
+
+/* Status Mask (Address 1Ah) */
+#define CS42XX8_STATUS_DAC_CLK_ERR_M_SHIFT 4
+#define CS42XX8_STATUS_DAC_CLK_ERR_M_MASK (1 << CS42XX8_STATUS_DAC_CLK_ERR_M_SHIFT)
+#define CS42XX8_STATUS_ADC_CLK_ERR_M_SHIFT 3
+#define CS42XX8_STATUS_ADC_CLK_ERR_M_MASK (1 << CS42XX8_STATUS_ADC_CLK_ERR_M_SHIFT)
+#define CS42XX8_STATUS_ADC3_OVFL_M_SHIFT 2
+#define CS42XX8_STATUS_ADC3_OVFL_M_MASK (1 << CS42XX8_STATUS_ADC3_OVFL_M_SHIFT)
+#define CS42XX8_STATUS_ADC2_OVFL_M_SHIFT 1
+#define CS42XX8_STATUS_ADC2_OVFL_M_MASK (1 << CS42XX8_STATUS_ADC2_OVFL_M_SHIFT)
+#define CS42XX8_STATUS_ADC1_OVFL_M_SHIFT 0
+#define CS42XX8_STATUS_ADC1_OVFL_M_MASK (1 << CS42XX8_STATUS_ADC1_OVFL_M_SHIFT)
+
+/* MUTEC Pin Control (Address 1Bh) */
+#define CS42XX8_MUTEC_MCPOLARITY_SHIFT 1
+#define CS42XX8_MUTEC_MCPOLARITY_MASK (1 << CS42XX8_MUTEC_MCPOLARITY_SHIFT)
+#define CS42XX8_MUTEC_MCPOLARITY_ACTIVE_LOW (0 << CS42XX8_MUTEC_MCPOLARITY_SHIFT)
+#define CS42XX8_MUTEC_MCPOLARITY_ACTIVE_HIGH (1 << CS42XX8_MUTEC_MCPOLARITY_SHIFT)
+#define CS42XX8_MUTEC_MUTEC_ACTIVE_SHIFT 0
+#define CS42XX8_MUTEC_MUTEC_ACTIVE_MASK (1 << CS42XX8_MUTEC_MUTEC_ACTIVE_SHIFT)
+#define CS42XX8_MUTEC_MUTEC_ACTIVE (1 << CS42XX8_MUTEC_MUTEC_ACTIVE_SHIFT)
+#endif /* _CS42XX8_H */
diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c
index e62e294..137e8eb 100644
--- a/sound/soc/codecs/da7210.c
+++ b/sound/soc/codecs/da7210.c
@@ -307,29 +307,29 @@ static const char * const da7210_hpf_cutoff_txt[] = {
"Fs/8192*pi", "Fs/4096*pi", "Fs/2048*pi", "Fs/1024*pi"
};
-static const struct soc_enum da7210_dac_hpf_cutoff =
- SOC_ENUM_SINGLE(DA7210_DAC_HPF, 0, 4, da7210_hpf_cutoff_txt);
+static SOC_ENUM_SINGLE_DECL(da7210_dac_hpf_cutoff,
+ DA7210_DAC_HPF, 0, da7210_hpf_cutoff_txt);
-static const struct soc_enum da7210_adc_hpf_cutoff =
- SOC_ENUM_SINGLE(DA7210_ADC_HPF, 0, 4, da7210_hpf_cutoff_txt);
+static SOC_ENUM_SINGLE_DECL(da7210_adc_hpf_cutoff,
+ DA7210_ADC_HPF, 0, da7210_hpf_cutoff_txt);
/* ADC and DAC voice (8kHz) high pass cutoff value */
static const char * const da7210_vf_cutoff_txt[] = {
"2.5Hz", "25Hz", "50Hz", "100Hz", "150Hz", "200Hz", "300Hz", "400Hz"
};
-static const struct soc_enum da7210_dac_vf_cutoff =
- SOC_ENUM_SINGLE(DA7210_DAC_HPF, 4, 8, da7210_vf_cutoff_txt);
+static SOC_ENUM_SINGLE_DECL(da7210_dac_vf_cutoff,
+ DA7210_DAC_HPF, 4, da7210_vf_cutoff_txt);
-static const struct soc_enum da7210_adc_vf_cutoff =
- SOC_ENUM_SINGLE(DA7210_ADC_HPF, 4, 8, da7210_vf_cutoff_txt);
+static SOC_ENUM_SINGLE_DECL(da7210_adc_vf_cutoff,
+ DA7210_ADC_HPF, 4, da7210_vf_cutoff_txt);
static const char *da7210_hp_mode_txt[] = {
"Class H", "Class G"
};
-static const struct soc_enum da7210_hp_mode_sel =
- SOC_ENUM_SINGLE(DA7210_HP_CFG, 0, 2, da7210_hp_mode_txt);
+static SOC_ENUM_SINGLE_DECL(da7210_hp_mode_sel,
+ DA7210_HP_CFG, 0, da7210_hp_mode_txt);
/* ALC can be enabled only if noise suppression is disabled */
static int da7210_put_alc_sw(struct snd_kcontrol *kcontrol,
@@ -1071,17 +1071,9 @@ static struct snd_soc_dai_driver da7210_dai = {
static int da7210_probe(struct snd_soc_codec *codec)
{
struct da7210_priv *da7210 = snd_soc_codec_get_drvdata(codec);
- int ret;
dev_info(codec->dev, "DA7210 Audio Codec %s\n", DA7210_VERSION);
- codec->control_data = da7210->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;
- }
-
da7210->mclk_rate = 0; /* This will be set from set_sysclk() */
da7210->master = 0; /* This will be set from set_fmt() */
diff --git a/sound/soc/codecs/da7213.c b/sound/soc/codecs/da7213.c
index 0c77e7a..738fa18 100644
--- a/sound/soc/codecs/da7213.c
+++ b/sound/soc/codecs/da7213.c
@@ -63,30 +63,30 @@ static const char * const da7213_voice_hpf_corner_txt[] = {
"2.5Hz", "25Hz", "50Hz", "100Hz", "150Hz", "200Hz", "300Hz", "400Hz"
};
-static const struct soc_enum da7213_dac_voice_hpf_corner =
- SOC_ENUM_SINGLE(DA7213_DAC_FILTERS1, DA7213_VOICE_HPF_CORNER_SHIFT,
- DA7213_VOICE_HPF_CORNER_MAX,
- da7213_voice_hpf_corner_txt);
+static SOC_ENUM_SINGLE_DECL(da7213_dac_voice_hpf_corner,
+ DA7213_DAC_FILTERS1,
+ DA7213_VOICE_HPF_CORNER_SHIFT,
+ da7213_voice_hpf_corner_txt);
-static const struct soc_enum da7213_adc_voice_hpf_corner =
- SOC_ENUM_SINGLE(DA7213_ADC_FILTERS1, DA7213_VOICE_HPF_CORNER_SHIFT,
- DA7213_VOICE_HPF_CORNER_MAX,
- da7213_voice_hpf_corner_txt);
+static SOC_ENUM_SINGLE_DECL(da7213_adc_voice_hpf_corner,
+ DA7213_ADC_FILTERS1,
+ DA7213_VOICE_HPF_CORNER_SHIFT,
+ da7213_voice_hpf_corner_txt);
/* ADC and DAC high pass filter cutoff value */
static const char * const da7213_audio_hpf_corner_txt[] = {
"Fs/24000", "Fs/12000", "Fs/6000", "Fs/3000"
};
-static const struct soc_enum da7213_dac_audio_hpf_corner =
- SOC_ENUM_SINGLE(DA7213_DAC_FILTERS1, DA7213_AUDIO_HPF_CORNER_SHIFT,
- DA7213_AUDIO_HPF_CORNER_MAX,
- da7213_audio_hpf_corner_txt);
+static SOC_ENUM_SINGLE_DECL(da7213_dac_audio_hpf_corner,
+ DA7213_DAC_FILTERS1
+ , DA7213_AUDIO_HPF_CORNER_SHIFT,
+ da7213_audio_hpf_corner_txt);
-static const struct soc_enum da7213_adc_audio_hpf_corner =
- SOC_ENUM_SINGLE(DA7213_ADC_FILTERS1, DA7213_AUDIO_HPF_CORNER_SHIFT,
- DA7213_AUDIO_HPF_CORNER_MAX,
- da7213_audio_hpf_corner_txt);
+static SOC_ENUM_SINGLE_DECL(da7213_adc_audio_hpf_corner,
+ DA7213_ADC_FILTERS1,
+ DA7213_AUDIO_HPF_CORNER_SHIFT,
+ da7213_audio_hpf_corner_txt);
/* Gain ramping rate value */
static const char * const da7213_gain_ramp_rate_txt[] = {
@@ -94,52 +94,50 @@ static const char * const da7213_gain_ramp_rate_txt[] = {
"nominal rate / 32"
};
-static const struct soc_enum da7213_gain_ramp_rate =
- SOC_ENUM_SINGLE(DA7213_GAIN_RAMP_CTRL, DA7213_GAIN_RAMP_RATE_SHIFT,
- DA7213_GAIN_RAMP_RATE_MAX, da7213_gain_ramp_rate_txt);
+static SOC_ENUM_SINGLE_DECL(da7213_gain_ramp_rate,
+ DA7213_GAIN_RAMP_CTRL,
+ DA7213_GAIN_RAMP_RATE_SHIFT,
+ da7213_gain_ramp_rate_txt);
/* DAC noise gate setup time value */
static const char * const da7213_dac_ng_setup_time_txt[] = {
"256 samples", "512 samples", "1024 samples", "2048 samples"
};
-static const struct soc_enum da7213_dac_ng_setup_time =
- SOC_ENUM_SINGLE(DA7213_DAC_NG_SETUP_TIME,
- DA7213_DAC_NG_SETUP_TIME_SHIFT,
- DA7213_DAC_NG_SETUP_TIME_MAX,
- da7213_dac_ng_setup_time_txt);
+static SOC_ENUM_SINGLE_DECL(da7213_dac_ng_setup_time,
+ DA7213_DAC_NG_SETUP_TIME,
+ DA7213_DAC_NG_SETUP_TIME_SHIFT,
+ da7213_dac_ng_setup_time_txt);
/* DAC noise gate rampup rate value */
static const char * const da7213_dac_ng_rampup_txt[] = {
"0.02 ms/dB", "0.16 ms/dB"
};
-static const struct soc_enum da7213_dac_ng_rampup_rate =
- SOC_ENUM_SINGLE(DA7213_DAC_NG_SETUP_TIME,
- DA7213_DAC_NG_RAMPUP_RATE_SHIFT,
- DA7213_DAC_NG_RAMP_RATE_MAX,
- da7213_dac_ng_rampup_txt);
+static SOC_ENUM_SINGLE_DECL(da7213_dac_ng_rampup_rate,
+ DA7213_DAC_NG_SETUP_TIME,
+ DA7213_DAC_NG_RAMPUP_RATE_SHIFT,
+ da7213_dac_ng_rampup_txt);
/* DAC noise gate rampdown rate value */
static const char * const da7213_dac_ng_rampdown_txt[] = {
"0.64 ms/dB", "20.48 ms/dB"
};
-static const struct soc_enum da7213_dac_ng_rampdown_rate =
- SOC_ENUM_SINGLE(DA7213_DAC_NG_SETUP_TIME,
- DA7213_DAC_NG_RAMPDN_RATE_SHIFT,
- DA7213_DAC_NG_RAMP_RATE_MAX,
- da7213_dac_ng_rampdown_txt);
+static SOC_ENUM_SINGLE_DECL(da7213_dac_ng_rampdown_rate,
+ DA7213_DAC_NG_SETUP_TIME,
+ DA7213_DAC_NG_RAMPDN_RATE_SHIFT,
+ da7213_dac_ng_rampdown_txt);
/* DAC soft mute rate value */
static const char * const da7213_dac_soft_mute_rate_txt[] = {
"1", "2", "4", "8", "16", "32", "64"
};
-static const struct soc_enum da7213_dac_soft_mute_rate =
- SOC_ENUM_SINGLE(DA7213_DAC_FILTERS5, DA7213_DAC_SOFTMUTE_RATE_SHIFT,
- DA7213_DAC_SOFTMUTE_RATE_MAX,
- da7213_dac_soft_mute_rate_txt);
+static SOC_ENUM_SINGLE_DECL(da7213_dac_soft_mute_rate,
+ DA7213_DAC_FILTERS5,
+ DA7213_DAC_SOFTMUTE_RATE_SHIFT,
+ da7213_dac_soft_mute_rate_txt);
/* ALC Attack Rate select */
static const char * const da7213_alc_attack_rate_txt[] = {
@@ -147,9 +145,10 @@ static const char * const da7213_alc_attack_rate_txt[] = {
"5632/fs", "11264/fs", "22528/fs", "45056/fs", "90112/fs", "180224/fs"
};
-static const struct soc_enum da7213_alc_attack_rate =
- SOC_ENUM_SINGLE(DA7213_ALC_CTRL2, DA7213_ALC_ATTACK_SHIFT,
- DA7213_ALC_ATTACK_MAX, da7213_alc_attack_rate_txt);
+static SOC_ENUM_SINGLE_DECL(da7213_alc_attack_rate,
+ DA7213_ALC_CTRL2,
+ DA7213_ALC_ATTACK_SHIFT,
+ da7213_alc_attack_rate_txt);
/* ALC Release Rate select */
static const char * const da7213_alc_release_rate_txt[] = {
@@ -157,9 +156,10 @@ static const char * const da7213_alc_release_rate_txt[] = {
"11264/fs", "22528/fs", "45056/fs", "90112/fs", "180224/fs"
};
-static const struct soc_enum da7213_alc_release_rate =
- SOC_ENUM_SINGLE(DA7213_ALC_CTRL2, DA7213_ALC_RELEASE_SHIFT,
- DA7213_ALC_RELEASE_MAX, da7213_alc_release_rate_txt);
+static SOC_ENUM_SINGLE_DECL(da7213_alc_release_rate,
+ DA7213_ALC_CTRL2,
+ DA7213_ALC_RELEASE_SHIFT,
+ da7213_alc_release_rate_txt);
/* ALC Hold Time select */
static const char * const da7213_alc_hold_time_txt[] = {
@@ -168,22 +168,25 @@ static const char * const da7213_alc_hold_time_txt[] = {
"253952/fs", "507904/fs", "1015808/fs", "2031616/fs"
};
-static const struct soc_enum da7213_alc_hold_time =
- SOC_ENUM_SINGLE(DA7213_ALC_CTRL3, DA7213_ALC_HOLD_SHIFT,
- DA7213_ALC_HOLD_MAX, da7213_alc_hold_time_txt);
+static SOC_ENUM_SINGLE_DECL(da7213_alc_hold_time,
+ DA7213_ALC_CTRL3,
+ DA7213_ALC_HOLD_SHIFT,
+ da7213_alc_hold_time_txt);
/* ALC Input Signal Tracking rate select */
static const char * const da7213_alc_integ_rate_txt[] = {
"1/4", "1/16", "1/256", "1/65536"
};
-static const struct soc_enum da7213_alc_integ_attack_rate =
- SOC_ENUM_SINGLE(DA7213_ALC_CTRL3, DA7213_ALC_INTEG_ATTACK_SHIFT,
- DA7213_ALC_INTEG_MAX, da7213_alc_integ_rate_txt);
+static SOC_ENUM_SINGLE_DECL(da7213_alc_integ_attack_rate,
+ DA7213_ALC_CTRL3,
+ DA7213_ALC_INTEG_ATTACK_SHIFT,
+ da7213_alc_integ_rate_txt);
-static const struct soc_enum da7213_alc_integ_release_rate =
- SOC_ENUM_SINGLE(DA7213_ALC_CTRL3, DA7213_ALC_INTEG_RELEASE_SHIFT,
- DA7213_ALC_INTEG_MAX, da7213_alc_integ_rate_txt);
+static SOC_ENUM_SINGLE_DECL(da7213_alc_integ_release_rate,
+ DA7213_ALC_CTRL3,
+ DA7213_ALC_INTEG_RELEASE_SHIFT,
+ da7213_alc_integ_rate_txt);
/*
@@ -584,15 +587,17 @@ static const char * const da7213_mic_amp_in_sel_txt[] = {
"Differential", "MIC_P", "MIC_N"
};
-static const struct soc_enum da7213_mic_1_amp_in_sel =
- SOC_ENUM_SINGLE(DA7213_MIC_1_CTRL, DA7213_MIC_AMP_IN_SEL_SHIFT,
- DA7213_MIC_AMP_IN_SEL_MAX, da7213_mic_amp_in_sel_txt);
+static SOC_ENUM_SINGLE_DECL(da7213_mic_1_amp_in_sel,
+ DA7213_MIC_1_CTRL,
+ DA7213_MIC_AMP_IN_SEL_SHIFT,
+ da7213_mic_amp_in_sel_txt);
static const struct snd_kcontrol_new da7213_mic_1_amp_in_sel_mux =
SOC_DAPM_ENUM("Mic 1 Amp Source MUX", da7213_mic_1_amp_in_sel);
-static const struct soc_enum da7213_mic_2_amp_in_sel =
- SOC_ENUM_SINGLE(DA7213_MIC_2_CTRL, DA7213_MIC_AMP_IN_SEL_SHIFT,
- DA7213_MIC_AMP_IN_SEL_MAX, da7213_mic_amp_in_sel_txt);
+static SOC_ENUM_SINGLE_DECL(da7213_mic_2_amp_in_sel,
+ DA7213_MIC_2_CTRL,
+ DA7213_MIC_AMP_IN_SEL_SHIFT,
+ da7213_mic_amp_in_sel_txt);
static const struct snd_kcontrol_new da7213_mic_2_amp_in_sel_mux =
SOC_DAPM_ENUM("Mic 2 Amp Source MUX", da7213_mic_2_amp_in_sel);
@@ -601,15 +606,17 @@ static const char * const da7213_dai_src_txt[] = {
"ADC Left", "ADC Right", "DAI Input Left", "DAI Input Right"
};
-static const struct soc_enum da7213_dai_l_src =
- SOC_ENUM_SINGLE(DA7213_DIG_ROUTING_DAI, DA7213_DAI_L_SRC_SHIFT,
- DA7213_DAI_SRC_MAX, da7213_dai_src_txt);
+static SOC_ENUM_SINGLE_DECL(da7213_dai_l_src,
+ DA7213_DIG_ROUTING_DAI,
+ DA7213_DAI_L_SRC_SHIFT,
+ da7213_dai_src_txt);
static const struct snd_kcontrol_new da7213_dai_l_src_mux =
SOC_DAPM_ENUM("DAI Left Source MUX", da7213_dai_l_src);
-static const struct soc_enum da7213_dai_r_src =
- SOC_ENUM_SINGLE(DA7213_DIG_ROUTING_DAI, DA7213_DAI_R_SRC_SHIFT,
- DA7213_DAI_SRC_MAX, da7213_dai_src_txt);
+static SOC_ENUM_SINGLE_DECL(da7213_dai_r_src,
+ DA7213_DIG_ROUTING_DAI,
+ DA7213_DAI_R_SRC_SHIFT,
+ da7213_dai_src_txt);
static const struct snd_kcontrol_new da7213_dai_r_src_mux =
SOC_DAPM_ENUM("DAI Right Source MUX", da7213_dai_r_src);
@@ -619,15 +626,17 @@ static const char * const da7213_dac_src_txt[] = {
"DAI Input Right"
};
-static const struct soc_enum da7213_dac_l_src =
- SOC_ENUM_SINGLE(DA7213_DIG_ROUTING_DAC, DA7213_DAC_L_SRC_SHIFT,
- DA7213_DAC_SRC_MAX, da7213_dac_src_txt);
+static SOC_ENUM_SINGLE_DECL(da7213_dac_l_src,
+ DA7213_DIG_ROUTING_DAC,
+ DA7213_DAC_L_SRC_SHIFT,
+ da7213_dac_src_txt);
static const struct snd_kcontrol_new da7213_dac_l_src_mux =
SOC_DAPM_ENUM("DAC Left Source MUX", da7213_dac_l_src);
-static const struct soc_enum da7213_dac_r_src =
- SOC_ENUM_SINGLE(DA7213_DIG_ROUTING_DAC, DA7213_DAC_R_SRC_SHIFT,
- DA7213_DAC_SRC_MAX, da7213_dac_src_txt);
+static SOC_ENUM_SINGLE_DECL(da7213_dac_r_src,
+ DA7213_DIG_ROUTING_DAC,
+ DA7213_DAC_R_SRC_SHIFT,
+ da7213_dac_src_txt);
static const struct snd_kcontrol_new da7213_dac_r_src_mux =
SOC_DAPM_ENUM("DAC Right Source MUX", da7213_dac_r_src);
@@ -1384,17 +1393,9 @@ static int da7213_set_bias_level(struct snd_soc_codec *codec,
static int da7213_probe(struct snd_soc_codec *codec)
{
- int ret;
struct da7213_priv *da7213 = snd_soc_codec_get_drvdata(codec);
struct da7213_platform_data *pdata = da7213->pdata;
- codec->control_data = da7213->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;
- }
-
/* Default to using ALC auto offset calibration mode. */
snd_soc_update_bits(codec, DA7213_ALC_CTRL1,
DA7213_ALC_CALIB_MODE_MAN, 0);
diff --git a/sound/soc/codecs/da732x.c b/sound/soc/codecs/da732x.c
index f4d965ebc..7d168ec 100644
--- a/sound/soc/codecs/da732x.c
+++ b/sound/soc/codecs/da732x.c
@@ -269,81 +269,65 @@ static const char *da732x_hpf_voice[] = {
"150Hz", "200Hz", "300Hz", "400Hz"
};
-static const struct soc_enum da732x_dac1_hpf_mode_enum[] = {
- SOC_ENUM_SINGLE(DA732X_REG_DAC1_HPF, DA732X_HPF_MODE_SHIFT,
- DA732X_HPF_MODE_MAX, da732x_hpf_mode)
-};
-
-static const struct soc_enum da732x_dac2_hpf_mode_enum[] = {
- SOC_ENUM_SINGLE(DA732X_REG_DAC2_HPF, DA732X_HPF_MODE_SHIFT,
- DA732X_HPF_MODE_MAX, da732x_hpf_mode)
-};
+static SOC_ENUM_SINGLE_DECL(da732x_dac1_hpf_mode_enum,
+ DA732X_REG_DAC1_HPF, DA732X_HPF_MODE_SHIFT,
+ da732x_hpf_mode);
-static const struct soc_enum da732x_dac3_hpf_mode_enum[] = {
- SOC_ENUM_SINGLE(DA732X_REG_DAC3_HPF, DA732X_HPF_MODE_SHIFT,
- DA732X_HPF_MODE_MAX, da732x_hpf_mode)
-};
+static SOC_ENUM_SINGLE_DECL(da732x_dac2_hpf_mode_enum,
+ DA732X_REG_DAC2_HPF, DA732X_HPF_MODE_SHIFT,
+ da732x_hpf_mode);
-static const struct soc_enum da732x_adc1_hpf_mode_enum[] = {
- SOC_ENUM_SINGLE(DA732X_REG_ADC1_HPF, DA732X_HPF_MODE_SHIFT,
- DA732X_HPF_MODE_MAX, da732x_hpf_mode)
-};
+static SOC_ENUM_SINGLE_DECL(da732x_dac3_hpf_mode_enum,
+ DA732X_REG_DAC3_HPF, DA732X_HPF_MODE_SHIFT,
+ da732x_hpf_mode);
-static const struct soc_enum da732x_adc2_hpf_mode_enum[] = {
- SOC_ENUM_SINGLE(DA732X_REG_ADC2_HPF, DA732X_HPF_MODE_SHIFT,
- DA732X_HPF_MODE_MAX, da732x_hpf_mode)
-};
+static SOC_ENUM_SINGLE_DECL(da732x_adc1_hpf_mode_enum,
+ DA732X_REG_ADC1_HPF, DA732X_HPF_MODE_SHIFT,
+ da732x_hpf_mode);
-static const struct soc_enum da732x_dac1_hp_filter_enum[] = {
- SOC_ENUM_SINGLE(DA732X_REG_DAC1_HPF, DA732X_HPF_MUSIC_SHIFT,
- DA732X_HPF_MUSIC_MAX, da732x_hpf_music)
-};
+static SOC_ENUM_SINGLE_DECL(da732x_adc2_hpf_mode_enum,
+ DA732X_REG_ADC2_HPF, DA732X_HPF_MODE_SHIFT,
+ da732x_hpf_mode);
-static const struct soc_enum da732x_dac2_hp_filter_enum[] = {
- SOC_ENUM_SINGLE(DA732X_REG_DAC2_HPF, DA732X_HPF_MUSIC_SHIFT,
- DA732X_HPF_MUSIC_MAX, da732x_hpf_music)
-};
+static SOC_ENUM_SINGLE_DECL(da732x_dac1_hp_filter_enum,
+ DA732X_REG_DAC1_HPF, DA732X_HPF_MUSIC_SHIFT,
+ da732x_hpf_music);
-static const struct soc_enum da732x_dac3_hp_filter_enum[] = {
- SOC_ENUM_SINGLE(DA732X_REG_DAC3_HPF, DA732X_HPF_MUSIC_SHIFT,
- DA732X_HPF_MUSIC_MAX, da732x_hpf_music)
-};
+static SOC_ENUM_SINGLE_DECL(da732x_dac2_hp_filter_enum,
+ DA732X_REG_DAC2_HPF, DA732X_HPF_MUSIC_SHIFT,
+ da732x_hpf_music);
-static const struct soc_enum da732x_adc1_hp_filter_enum[] = {
- SOC_ENUM_SINGLE(DA732X_REG_ADC1_HPF, DA732X_HPF_MUSIC_SHIFT,
- DA732X_HPF_MUSIC_MAX, da732x_hpf_music)
-};
+static SOC_ENUM_SINGLE_DECL(da732x_dac3_hp_filter_enum,
+ DA732X_REG_DAC3_HPF, DA732X_HPF_MUSIC_SHIFT,
+ da732x_hpf_music);
-static const struct soc_enum da732x_adc2_hp_filter_enum[] = {
- SOC_ENUM_SINGLE(DA732X_REG_ADC2_HPF, DA732X_HPF_MUSIC_SHIFT,
- DA732X_HPF_MUSIC_MAX, da732x_hpf_music)
-};
+static SOC_ENUM_SINGLE_DECL(da732x_adc1_hp_filter_enum,
+ DA732X_REG_ADC1_HPF, DA732X_HPF_MUSIC_SHIFT,
+ da732x_hpf_music);
-static const struct soc_enum da732x_dac1_voice_filter_enum[] = {
- SOC_ENUM_SINGLE(DA732X_REG_DAC1_HPF, DA732X_HPF_VOICE_SHIFT,
- DA732X_HPF_VOICE_MAX, da732x_hpf_voice)
-};
+static SOC_ENUM_SINGLE_DECL(da732x_adc2_hp_filter_enum,
+ DA732X_REG_ADC2_HPF, DA732X_HPF_MUSIC_SHIFT,
+ da732x_hpf_music);
-static const struct soc_enum da732x_dac2_voice_filter_enum[] = {
- SOC_ENUM_SINGLE(DA732X_REG_DAC2_HPF, DA732X_HPF_VOICE_SHIFT,
- DA732X_HPF_VOICE_MAX, da732x_hpf_voice)
-};
+static SOC_ENUM_SINGLE_DECL(da732x_dac1_voice_filter_enum,
+ DA732X_REG_DAC1_HPF, DA732X_HPF_VOICE_SHIFT,
+ da732x_hpf_voice);
-static const struct soc_enum da732x_dac3_voice_filter_enum[] = {
- SOC_ENUM_SINGLE(DA732X_REG_DAC3_HPF, DA732X_HPF_VOICE_SHIFT,
- DA732X_HPF_VOICE_MAX, da732x_hpf_voice)
-};
+static SOC_ENUM_SINGLE_DECL(da732x_dac2_voice_filter_enum,
+ DA732X_REG_DAC2_HPF, DA732X_HPF_VOICE_SHIFT,
+ da732x_hpf_voice);
-static const struct soc_enum da732x_adc1_voice_filter_enum[] = {
- SOC_ENUM_SINGLE(DA732X_REG_ADC1_HPF, DA732X_HPF_VOICE_SHIFT,
- DA732X_HPF_VOICE_MAX, da732x_hpf_voice)
-};
+static SOC_ENUM_SINGLE_DECL(da732x_dac3_voice_filter_enum,
+ DA732X_REG_DAC3_HPF, DA732X_HPF_VOICE_SHIFT,
+ da732x_hpf_voice);
-static const struct soc_enum da732x_adc2_voice_filter_enum[] = {
- SOC_ENUM_SINGLE(DA732X_REG_ADC2_HPF, DA732X_HPF_VOICE_SHIFT,
- DA732X_HPF_VOICE_MAX, da732x_hpf_voice)
-};
+static SOC_ENUM_SINGLE_DECL(da732x_adc1_voice_filter_enum,
+ DA732X_REG_ADC1_HPF, DA732X_HPF_VOICE_SHIFT,
+ da732x_hpf_voice);
+static SOC_ENUM_SINGLE_DECL(da732x_adc2_voice_filter_enum,
+ DA732X_REG_ADC2_HPF, DA732X_HPF_VOICE_SHIFT,
+ da732x_hpf_voice);
static int da732x_hpf_set(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
@@ -714,65 +698,65 @@ static const char *enable_text[] = {
};
/* ADC1LMUX */
-static const struct soc_enum adc1l_enum =
- SOC_ENUM_SINGLE(DA732X_REG_INP_MUX, DA732X_ADC1L_MUX_SEL_SHIFT,
- DA732X_ADCL_MUX_MAX, adcl_text);
+static SOC_ENUM_SINGLE_DECL(adc1l_enum,
+ DA732X_REG_INP_MUX, DA732X_ADC1L_MUX_SEL_SHIFT,
+ adcl_text);
static const struct snd_kcontrol_new adc1l_mux =
SOC_DAPM_ENUM("ADC Route", adc1l_enum);
/* ADC1RMUX */
-static const struct soc_enum adc1r_enum =
- SOC_ENUM_SINGLE(DA732X_REG_INP_MUX, DA732X_ADC1R_MUX_SEL_SHIFT,
- DA732X_ADCR_MUX_MAX, adcr_text);
+static SOC_ENUM_SINGLE_DECL(adc1r_enum,
+ DA732X_REG_INP_MUX, DA732X_ADC1R_MUX_SEL_SHIFT,
+ adcr_text);
static const struct snd_kcontrol_new adc1r_mux =
SOC_DAPM_ENUM("ADC Route", adc1r_enum);
/* ADC2LMUX */
-static const struct soc_enum adc2l_enum =
- SOC_ENUM_SINGLE(DA732X_REG_INP_MUX, DA732X_ADC2L_MUX_SEL_SHIFT,
- DA732X_ADCL_MUX_MAX, adcl_text);
+static SOC_ENUM_SINGLE_DECL(adc2l_enum,
+ DA732X_REG_INP_MUX, DA732X_ADC2L_MUX_SEL_SHIFT,
+ adcl_text);
static const struct snd_kcontrol_new adc2l_mux =
SOC_DAPM_ENUM("ADC Route", adc2l_enum);
/* ADC2RMUX */
-static const struct soc_enum adc2r_enum =
- SOC_ENUM_SINGLE(DA732X_REG_INP_MUX, DA732X_ADC2R_MUX_SEL_SHIFT,
- DA732X_ADCR_MUX_MAX, adcr_text);
+static SOC_ENUM_SINGLE_DECL(adc2r_enum,
+ DA732X_REG_INP_MUX, DA732X_ADC2R_MUX_SEL_SHIFT,
+ adcr_text);
static const struct snd_kcontrol_new adc2r_mux =
SOC_DAPM_ENUM("ADC Route", adc2r_enum);
-static const struct soc_enum da732x_hp_left_output =
- SOC_ENUM_SINGLE(DA732X_REG_HPL, DA732X_HP_OUT_DAC_EN_SHIFT,
- DA732X_DAC_EN_MAX, enable_text);
+static SOC_ENUM_SINGLE_DECL(da732x_hp_left_output,
+ DA732X_REG_HPL, DA732X_HP_OUT_DAC_EN_SHIFT,
+ enable_text);
static const struct snd_kcontrol_new hpl_mux =
SOC_DAPM_ENUM("HPL Switch", da732x_hp_left_output);
-static const struct soc_enum da732x_hp_right_output =
- SOC_ENUM_SINGLE(DA732X_REG_HPR, DA732X_HP_OUT_DAC_EN_SHIFT,
- DA732X_DAC_EN_MAX, enable_text);
+static SOC_ENUM_SINGLE_DECL(da732x_hp_right_output,
+ DA732X_REG_HPR, DA732X_HP_OUT_DAC_EN_SHIFT,
+ enable_text);
static const struct snd_kcontrol_new hpr_mux =
SOC_DAPM_ENUM("HPR Switch", da732x_hp_right_output);
-static const struct soc_enum da732x_speaker_output =
- SOC_ENUM_SINGLE(DA732X_REG_LIN3, DA732X_LOUT_DAC_EN_SHIFT,
- DA732X_DAC_EN_MAX, enable_text);
+static SOC_ENUM_SINGLE_DECL(da732x_speaker_output,
+ DA732X_REG_LIN3, DA732X_LOUT_DAC_EN_SHIFT,
+ enable_text);
static const struct snd_kcontrol_new spk_mux =
SOC_DAPM_ENUM("SPK Switch", da732x_speaker_output);
-static const struct soc_enum da732x_lout4_output =
- SOC_ENUM_SINGLE(DA732X_REG_LIN4, DA732X_LOUT_DAC_EN_SHIFT,
- DA732X_DAC_EN_MAX, enable_text);
+static SOC_ENUM_SINGLE_DECL(da732x_lout4_output,
+ DA732X_REG_LIN4, DA732X_LOUT_DAC_EN_SHIFT,
+ enable_text);
static const struct snd_kcontrol_new lout4_mux =
SOC_DAPM_ENUM("LOUT4 Switch", da732x_lout4_output);
-static const struct soc_enum da732x_lout2_output =
- SOC_ENUM_SINGLE(DA732X_REG_LIN2, DA732X_LOUT_DAC_EN_SHIFT,
- DA732X_DAC_EN_MAX, enable_text);
+static SOC_ENUM_SINGLE_DECL(da732x_lout2_output,
+ DA732X_REG_LIN2, DA732X_LOUT_DAC_EN_SHIFT,
+ enable_text);
static const struct snd_kcontrol_new lout2_mux =
SOC_DAPM_ENUM("LOUT2 Switch", da732x_lout2_output);
@@ -1313,9 +1297,9 @@ static void da732x_dac_offset_adjust(struct snd_soc_codec *codec)
msleep(DA732X_WAIT_FOR_STABILIZATION);
/* Check DAC offset sign */
- sign[DA732X_HPL_DAC] = (codec->hw_read(codec, DA732X_REG_HPL_DAC_OFF_CNTL) &
+ sign[DA732X_HPL_DAC] = (snd_soc_read(codec, DA732X_REG_HPL_DAC_OFF_CNTL) &
DA732X_HP_DAC_OFF_CNTL_COMPO);
- sign[DA732X_HPR_DAC] = (codec->hw_read(codec, DA732X_REG_HPR_DAC_OFF_CNTL) &
+ sign[DA732X_HPR_DAC] = (snd_soc_read(codec, DA732X_REG_HPR_DAC_OFF_CNTL) &
DA732X_HP_DAC_OFF_CNTL_COMPO);
/* Binary search DAC offset values (both channels at once) */
@@ -1332,10 +1316,10 @@ static void da732x_dac_offset_adjust(struct snd_soc_codec *codec)
msleep(DA732X_WAIT_FOR_STABILIZATION);
- if ((codec->hw_read(codec, DA732X_REG_HPL_DAC_OFF_CNTL) &
+ if ((snd_soc_read(codec, DA732X_REG_HPL_DAC_OFF_CNTL) &
DA732X_HP_DAC_OFF_CNTL_COMPO) ^ sign[DA732X_HPL_DAC])
offset[DA732X_HPL_DAC] &= ~step;
- if ((codec->hw_read(codec, DA732X_REG_HPR_DAC_OFF_CNTL) &
+ if ((snd_soc_read(codec, DA732X_REG_HPR_DAC_OFF_CNTL) &
DA732X_HP_DAC_OFF_CNTL_COMPO) ^ sign[DA732X_HPR_DAC])
offset[DA732X_HPR_DAC] &= ~step;
@@ -1376,9 +1360,9 @@ static void da732x_output_offset_adjust(struct snd_soc_codec *codec)
msleep(DA732X_WAIT_FOR_STABILIZATION);
/* Check output offset sign */
- sign[DA732X_HPL_AMP] = codec->hw_read(codec, DA732X_REG_HPL) &
+ sign[DA732X_HPL_AMP] = snd_soc_read(codec, DA732X_REG_HPL) &
DA732X_HP_OUT_COMPO;
- sign[DA732X_HPR_AMP] = codec->hw_read(codec, DA732X_REG_HPR) &
+ sign[DA732X_HPR_AMP] = snd_soc_read(codec, DA732X_REG_HPR) &
DA732X_HP_OUT_COMPO;
snd_soc_write(codec, DA732X_REG_HPL, DA732X_HP_OUT_COMP |
@@ -1399,10 +1383,10 @@ static void da732x_output_offset_adjust(struct snd_soc_codec *codec)
msleep(DA732X_WAIT_FOR_STABILIZATION);
- if ((codec->hw_read(codec, DA732X_REG_HPL) &
+ if ((snd_soc_read(codec, DA732X_REG_HPL) &
DA732X_HP_OUT_COMPO) ^ sign[DA732X_HPL_AMP])
offset[DA732X_HPL_AMP] &= ~step;
- if ((codec->hw_read(codec, DA732X_REG_HPR) &
+ if ((snd_soc_read(codec, DA732X_REG_HPR) &
DA732X_HP_OUT_COMPO) ^ sign[DA732X_HPR_AMP])
offset[DA732X_HPR_AMP] &= ~step;
@@ -1499,8 +1483,8 @@ static int da732x_set_bias_level(struct snd_soc_codec *codec,
da732x_hp_dc_offset_cancellation(codec);
- regcache_cache_only(codec->control_data, false);
- regcache_sync(codec->control_data);
+ regcache_cache_only(da732x->regmap, false);
+ regcache_sync(da732x->regmap);
} else {
snd_soc_update_bits(codec, DA732X_REG_BIAS_EN,
DA732X_BIAS_BOOST_MASK,
@@ -1511,7 +1495,7 @@ static int da732x_set_bias_level(struct snd_soc_codec *codec,
}
break;
case SND_SOC_BIAS_OFF:
- regcache_cache_only(codec->control_data, true);
+ regcache_cache_only(da732x->regmap, true);
da732x_set_charge_pump(codec, DA732X_DISABLE_CP);
snd_soc_update_bits(codec, DA732X_REG_BIAS_EN, DA732X_BIAS_EN,
DA732X_BIAS_DIS);
@@ -1528,23 +1512,14 @@ static int da732x_probe(struct snd_soc_codec *codec)
{
struct da732x_priv *da732x = snd_soc_codec_get_drvdata(codec);
struct snd_soc_dapm_context *dapm = &codec->dapm;
- int ret = 0;
da732x->codec = codec;
dapm->idle_bias_off = false;
- codec->control_data = da732x->regmap;
-
- ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP);
- if (ret != 0) {
- dev_err(codec->dev, "Failed to register codec.\n");
- goto err;
- }
-
da732x_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-err:
- return ret;
+
+ return 0;
}
static int da732x_remove(struct snd_soc_codec *codec)
@@ -1566,7 +1541,6 @@ static struct snd_soc_codec_driver soc_codec_dev_da732x = {
.dapm_routes = da732x_dapm_routes,
.num_dapm_routes = ARRAY_SIZE(da732x_dapm_routes),
.set_pll = da732x_set_dai_pll,
- .reg_cache_size = ARRAY_SIZE(da732x_reg_cache),
};
static int da732x_i2c_probe(struct i2c_client *i2c,
diff --git a/sound/soc/codecs/da732x.h b/sound/soc/codecs/da732x.h
index c8ce547..1dceafe 100644
--- a/sound/soc/codecs/da732x.h
+++ b/sound/soc/codecs/da732x.h
@@ -113,9 +113,6 @@
#define DA732X_EQ_OVERALL_VOL_DB_MIN -1800
#define DA732X_EQ_OVERALL_VOL_DB_INC 600
-#define DA732X_SOC_ENUM_DOUBLE_R(xreg, xrreg, xmax, xtext) \
- {.reg = xreg, .reg2 = xrreg, .max = xmax, .texts = xtext}
-
enum da732x_sysctl {
DA732X_SR_8KHZ = 0x1,
DA732X_SR_11_025KHZ = 0x2,
diff --git a/sound/soc/codecs/da9055.c b/sound/soc/codecs/da9055.c
index 4228126..4ff06b5 100644
--- a/sound/soc/codecs/da9055.c
+++ b/sound/soc/codecs/da9055.c
@@ -18,6 +18,8 @@
#include <linux/regmap.h>
#include <linux/slab.h>
#include <linux/module.h>
+#include <linux/of.h>
+#include <linux/of_device.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
#include <sound/soc.h>
@@ -321,22 +323,22 @@ static const char * const da9055_hpf_cutoff_txt[] = {
"Fs/24000", "Fs/12000", "Fs/6000", "Fs/3000"
};
-static const struct soc_enum da9055_dac_hpf_cutoff =
- SOC_ENUM_SINGLE(DA9055_DAC_FILTERS1, 4, 4, da9055_hpf_cutoff_txt);
+static SOC_ENUM_SINGLE_DECL(da9055_dac_hpf_cutoff,
+ DA9055_DAC_FILTERS1, 4, da9055_hpf_cutoff_txt);
-static const struct soc_enum da9055_adc_hpf_cutoff =
- SOC_ENUM_SINGLE(DA9055_ADC_FILTERS1, 4, 4, da9055_hpf_cutoff_txt);
+static SOC_ENUM_SINGLE_DECL(da9055_adc_hpf_cutoff,
+ DA9055_ADC_FILTERS1, 4, da9055_hpf_cutoff_txt);
/* ADC and DAC voice mode (8kHz) high pass cutoff value */
static const char * const da9055_vf_cutoff_txt[] = {
"2.5Hz", "25Hz", "50Hz", "100Hz", "150Hz", "200Hz", "300Hz", "400Hz"
};
-static const struct soc_enum da9055_dac_vf_cutoff =
- SOC_ENUM_SINGLE(DA9055_DAC_FILTERS1, 0, 8, da9055_vf_cutoff_txt);
+static SOC_ENUM_SINGLE_DECL(da9055_dac_vf_cutoff,
+ DA9055_DAC_FILTERS1, 0, da9055_vf_cutoff_txt);
-static const struct soc_enum da9055_adc_vf_cutoff =
- SOC_ENUM_SINGLE(DA9055_ADC_FILTERS1, 0, 8, da9055_vf_cutoff_txt);
+static SOC_ENUM_SINGLE_DECL(da9055_adc_vf_cutoff,
+ DA9055_ADC_FILTERS1, 0, da9055_vf_cutoff_txt);
/* Gain ramping rate value */
static const char * const da9055_gain_ramping_txt[] = {
@@ -344,44 +346,44 @@ static const char * const da9055_gain_ramping_txt[] = {
"nominal rate / 8"
};
-static const struct soc_enum da9055_gain_ramping_rate =
- SOC_ENUM_SINGLE(DA9055_GAIN_RAMP_CTRL, 0, 4, da9055_gain_ramping_txt);
+static SOC_ENUM_SINGLE_DECL(da9055_gain_ramping_rate,
+ DA9055_GAIN_RAMP_CTRL, 0, da9055_gain_ramping_txt);
/* DAC noise gate setup time value */
static const char * const da9055_dac_ng_setup_time_txt[] = {
"256 samples", "512 samples", "1024 samples", "2048 samples"
};
-static const struct soc_enum da9055_dac_ng_setup_time =
- SOC_ENUM_SINGLE(DA9055_DAC_NG_SETUP_TIME, 0, 4,
- da9055_dac_ng_setup_time_txt);
+static SOC_ENUM_SINGLE_DECL(da9055_dac_ng_setup_time,
+ DA9055_DAC_NG_SETUP_TIME, 0,
+ da9055_dac_ng_setup_time_txt);
/* DAC noise gate rampup rate value */
static const char * const da9055_dac_ng_rampup_txt[] = {
"0.02 ms/dB", "0.16 ms/dB"
};
-static const struct soc_enum da9055_dac_ng_rampup_rate =
- SOC_ENUM_SINGLE(DA9055_DAC_NG_SETUP_TIME, 2, 2,
- da9055_dac_ng_rampup_txt);
+static SOC_ENUM_SINGLE_DECL(da9055_dac_ng_rampup_rate,
+ DA9055_DAC_NG_SETUP_TIME, 2,
+ da9055_dac_ng_rampup_txt);
/* DAC noise gate rampdown rate value */
static const char * const da9055_dac_ng_rampdown_txt[] = {
"0.64 ms/dB", "20.48 ms/dB"
};
-static const struct soc_enum da9055_dac_ng_rampdown_rate =
- SOC_ENUM_SINGLE(DA9055_DAC_NG_SETUP_TIME, 3, 2,
- da9055_dac_ng_rampdown_txt);
+static SOC_ENUM_SINGLE_DECL(da9055_dac_ng_rampdown_rate,
+ DA9055_DAC_NG_SETUP_TIME, 3,
+ da9055_dac_ng_rampdown_txt);
/* DAC soft mute rate value */
static const char * const da9055_dac_soft_mute_rate_txt[] = {
"1", "2", "4", "8", "16", "32", "64"
};
-static const struct soc_enum da9055_dac_soft_mute_rate =
- SOC_ENUM_SINGLE(DA9055_DAC_FILTERS5, 4, 7,
- da9055_dac_soft_mute_rate_txt);
+static SOC_ENUM_SINGLE_DECL(da9055_dac_soft_mute_rate,
+ DA9055_DAC_FILTERS5, 4,
+ da9055_dac_soft_mute_rate_txt);
/* DAC routing select */
static const char * const da9055_dac_src_txt[] = {
@@ -389,40 +391,40 @@ static const char * const da9055_dac_src_txt[] = {
"AIF input right"
};
-static const struct soc_enum da9055_dac_l_src =
- SOC_ENUM_SINGLE(DA9055_DIG_ROUTING_DAC, 0, 4, da9055_dac_src_txt);
+static SOC_ENUM_SINGLE_DECL(da9055_dac_l_src,
+ DA9055_DIG_ROUTING_DAC, 0, da9055_dac_src_txt);
-static const struct soc_enum da9055_dac_r_src =
- SOC_ENUM_SINGLE(DA9055_DIG_ROUTING_DAC, 4, 4, da9055_dac_src_txt);
+static SOC_ENUM_SINGLE_DECL(da9055_dac_r_src,
+ DA9055_DIG_ROUTING_DAC, 4, da9055_dac_src_txt);
/* MIC PGA Left source select */
static const char * const da9055_mic_l_src_txt[] = {
"MIC1_P_N", "MIC1_P", "MIC1_N", "MIC2_L"
};
-static const struct soc_enum da9055_mic_l_src =
- SOC_ENUM_SINGLE(DA9055_MIXIN_L_SELECT, 4, 4, da9055_mic_l_src_txt);
+static SOC_ENUM_SINGLE_DECL(da9055_mic_l_src,
+ DA9055_MIXIN_L_SELECT, 4, da9055_mic_l_src_txt);
/* MIC PGA Right source select */
static const char * const da9055_mic_r_src_txt[] = {
"MIC2_R_L", "MIC2_R", "MIC2_L"
};
-static const struct soc_enum da9055_mic_r_src =
- SOC_ENUM_SINGLE(DA9055_MIXIN_R_SELECT, 4, 3, da9055_mic_r_src_txt);
+static SOC_ENUM_SINGLE_DECL(da9055_mic_r_src,
+ DA9055_MIXIN_R_SELECT, 4, da9055_mic_r_src_txt);
/* ALC Input Signal Tracking rate select */
static const char * const da9055_signal_tracking_rate_txt[] = {
"1/4", "1/16", "1/256", "1/65536"
};
-static const struct soc_enum da9055_integ_attack_rate =
- SOC_ENUM_SINGLE(DA9055_ALC_CTRL3, 4, 4,
- da9055_signal_tracking_rate_txt);
+static SOC_ENUM_SINGLE_DECL(da9055_integ_attack_rate,
+ DA9055_ALC_CTRL3, 4,
+ da9055_signal_tracking_rate_txt);
-static const struct soc_enum da9055_integ_release_rate =
- SOC_ENUM_SINGLE(DA9055_ALC_CTRL3, 6, 4,
- da9055_signal_tracking_rate_txt);
+static SOC_ENUM_SINGLE_DECL(da9055_integ_release_rate,
+ DA9055_ALC_CTRL3, 6,
+ da9055_signal_tracking_rate_txt);
/* ALC Attack Rate select */
static const char * const da9055_attack_rate_txt[] = {
@@ -430,8 +432,8 @@ static const char * const da9055_attack_rate_txt[] = {
"5632/fs", "11264/fs", "22528/fs", "45056/fs", "90112/fs", "180224/fs"
};
-static const struct soc_enum da9055_attack_rate =
- SOC_ENUM_SINGLE(DA9055_ALC_CTRL2, 0, 13, da9055_attack_rate_txt);
+static SOC_ENUM_SINGLE_DECL(da9055_attack_rate,
+ DA9055_ALC_CTRL2, 0, da9055_attack_rate_txt);
/* ALC Release Rate select */
static const char * const da9055_release_rate_txt[] = {
@@ -439,8 +441,8 @@ static const char * const da9055_release_rate_txt[] = {
"11264/fs", "22528/fs", "45056/fs", "90112/fs", "180224/fs"
};
-static const struct soc_enum da9055_release_rate =
- SOC_ENUM_SINGLE(DA9055_ALC_CTRL2, 4, 11, da9055_release_rate_txt);
+static SOC_ENUM_SINGLE_DECL(da9055_release_rate,
+ DA9055_ALC_CTRL2, 4, da9055_release_rate_txt);
/* ALC Hold Time select */
static const char * const da9055_hold_time_txt[] = {
@@ -449,8 +451,8 @@ static const char * const da9055_hold_time_txt[] = {
"253952/fs", "507904/fs", "1015808/fs", "2031616/fs"
};
-static const struct soc_enum da9055_hold_time =
- SOC_ENUM_SINGLE(DA9055_ALC_CTRL3, 0, 16, da9055_hold_time_txt);
+static SOC_ENUM_SINGLE_DECL(da9055_hold_time,
+ DA9055_ALC_CTRL3, 0, da9055_hold_time_txt);
static int da9055_get_alc_data(struct snd_soc_codec *codec, u8 reg_val)
{
@@ -1381,16 +1383,8 @@ static int da9055_set_bias_level(struct snd_soc_codec *codec,
static int da9055_probe(struct snd_soc_codec *codec)
{
- int ret;
struct da9055_priv *da9055 = snd_soc_codec_get_drvdata(codec);
- codec->control_data = da9055->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;
- }
-
/* Enable all Gain Ramps */
snd_soc_update_bits(codec, DA9055_AUX_L_CTRL,
DA9055_GAIN_RAMPING_EN, DA9055_GAIN_RAMPING_EN);
@@ -1536,11 +1530,17 @@ static const struct i2c_device_id da9055_i2c_id[] = {
};
MODULE_DEVICE_TABLE(i2c, da9055_i2c_id);
+static const struct of_device_id da9055_of_match[] = {
+ { .compatible = "dlg,da9055-codec", },
+ { }
+};
+
/* I2C codec control layer */
static struct i2c_driver da9055_i2c_driver = {
.driver = {
.name = "da9055-codec",
.owner = THIS_MODULE,
+ .of_match_table = of_match_ptr(da9055_of_match),
},
.probe = da9055_i2c_probe,
.remove = da9055_remove,
diff --git a/sound/soc/codecs/isabelle.c b/sound/soc/codecs/isabelle.c
index cb736dd..3a89ce6 100644
--- a/sound/soc/codecs/isabelle.c
+++ b/sound/soc/codecs/isabelle.c
@@ -918,8 +918,7 @@ static int isabelle_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_codec *codec = dai->codec;
u16 aif = 0;
unsigned int fs_val = 0;
@@ -1090,23 +1089,7 @@ static struct snd_soc_dai_driver isabelle_dai[] = {
},
};
-static int isabelle_probe(struct snd_soc_codec *codec)
-{
- int ret = 0;
-
- codec->control_data = dev_get_regmap(codec->dev, NULL);
-
- ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP);
- if (ret < 0) {
- dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
- return ret;
- }
-
- return 0;
-}
-
static struct snd_soc_codec_driver soc_codec_dev_isabelle = {
- .probe = isabelle_probe,
.set_bias_level = isabelle_set_bias_level,
.controls = isabelle_snd_controls,
.num_controls = ARRAY_SIZE(isabelle_snd_controls),
diff --git a/sound/soc/codecs/lm4857.c b/sound/soc/codecs/lm4857.c
index 0e5743e..4f048db 100644
--- a/sound/soc/codecs/lm4857.c
+++ b/sound/soc/codecs/lm4857.c
@@ -101,8 +101,7 @@ static const char *lm4857_mode[] = {
"Headphone",
};
-static const struct soc_enum lm4857_mode_enum =
- SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(lm4857_mode), lm4857_mode);
+static SOC_ENUM_SINGLE_EXT_DECL(lm4857_mode_enum, lm4857_mode);
static const struct snd_soc_dapm_widget lm4857_dapm_widgets[] = {
SND_SOC_DAPM_INPUT("IN"),
diff --git a/sound/soc/codecs/lm49453.c b/sound/soc/codecs/lm49453.c
index e19490c..275b3f7 100644
--- a/sound/soc/codecs/lm49453.c
+++ b/sound/soc/codecs/lm49453.c
@@ -195,33 +195,31 @@ struct lm49453_priv {
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 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 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);
+static 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 SOC_ENUM_SINGLE_DECL(lm49453_adcl_enum,
+ LM49453_P0_ANALOG_MIXER_ADC_REG, 0,
+ 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 SOC_ENUM_SINGLE_DECL(lm49453_adcr_enum,
+ LM49453_P0_ANALOG_MIXER_ADC_REG, 1,
+ lm49453_adcr_mux_text);
static const struct snd_kcontrol_new lm49453_adcl_mux_control =
SOC_DAPM_ENUM("ADC Left Mux", lm49453_adcl_enum);
@@ -1409,22 +1407,6 @@ static int lm49453_resume(struct snd_soc_codec *codec)
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)
{
@@ -1433,7 +1415,6 @@ static int lm49453_remove(struct snd_soc_codec *codec)
}
static struct snd_soc_codec_driver soc_codec_dev_lm49453 = {
- .probe = lm49453_probe,
.remove = lm49453_remove,
.suspend = lm49453_suspend,
.resume = lm49453_resume,
diff --git a/sound/soc/codecs/max9768.c b/sound/soc/codecs/max9768.c
index 31f9156..ec481fc 100644
--- a/sound/soc/codecs/max9768.c
+++ b/sound/soc/codecs/max9768.c
@@ -135,11 +135,6 @@ static int max9768_probe(struct snd_soc_codec *codec)
struct max9768 *max9768 = snd_soc_codec_get_drvdata(codec);
int ret;
- codec->control_data = max9768->regmap;
- ret = snd_soc_codec_set_cache_io(codec, 2, 6, SND_SOC_REGMAP);
- if (ret)
- return ret;
-
if (max9768->flags & MAX9768_FLAG_CLASSIC_PWM) {
ret = snd_soc_write(codec, MAX9768_CTRL, MAX9768_CTRL_PWM);
if (ret)
diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c
index ee660e2..ef7cf89 100644
--- a/sound/soc/codecs/max98088.c
+++ b/sound/soc/codecs/max98088.c
@@ -597,28 +597,27 @@ static const unsigned int max98088_exmode_values[] = {
0x00, 0x43, 0x10, 0x20, 0x30, 0x40, 0x11, 0x22, 0x32
};
-static const struct soc_enum max98088_exmode_enum =
- SOC_VALUE_ENUM_SINGLE(M98088_REG_41_SPKDHP, 0, 127,
- ARRAY_SIZE(max98088_exmode_texts),
- max98088_exmode_texts,
- max98088_exmode_values);
+static SOC_VALUE_ENUM_SINGLE_DECL(max98088_exmode_enum,
+ M98088_REG_41_SPKDHP, 0, 127,
+ max98088_exmode_texts,
+ max98088_exmode_values);
static const char *max98088_ex_thresh[] = { /* volts PP */
"0.6", "1.2", "1.8", "2.4", "3.0", "3.6", "4.2", "4.8"};
-static const struct soc_enum max98088_ex_thresh_enum[] = {
- SOC_ENUM_SINGLE(M98088_REG_42_SPKDHP_THRESH, 0, 8,
- max98088_ex_thresh),
-};
+static SOC_ENUM_SINGLE_DECL(max98088_ex_thresh_enum,
+ M98088_REG_42_SPKDHP_THRESH, 0,
+ max98088_ex_thresh);
static const char *max98088_fltr_mode[] = {"Voice", "Music" };
-static const struct soc_enum max98088_filter_mode_enum[] = {
- SOC_ENUM_SINGLE(M98088_REG_18_DAI1_FILTERS, 7, 2, max98088_fltr_mode),
-};
+static SOC_ENUM_SINGLE_DECL(max98088_filter_mode_enum,
+ M98088_REG_18_DAI1_FILTERS, 7,
+ max98088_fltr_mode);
static const char *max98088_extmic_text[] = { "None", "MIC1", "MIC2" };
-static const struct soc_enum max98088_extmic_enum =
- SOC_ENUM_SINGLE(M98088_REG_48_CFG_MIC, 0, 3, max98088_extmic_text);
+static SOC_ENUM_SINGLE_DECL(max98088_extmic_enum,
+ M98088_REG_48_CFG_MIC, 0,
+ max98088_extmic_text);
static const struct snd_kcontrol_new max98088_extmic_mux =
SOC_DAPM_ENUM("External MIC Mux", max98088_extmic_enum);
@@ -626,12 +625,12 @@ static const struct snd_kcontrol_new max98088_extmic_mux =
static const char *max98088_dai1_fltr[] = {
"Off", "fc=258/fs=16k", "fc=500/fs=16k",
"fc=258/fs=8k", "fc=500/fs=8k", "fc=200"};
-static const struct soc_enum max98088_dai1_dac_filter_enum[] = {
- SOC_ENUM_SINGLE(M98088_REG_18_DAI1_FILTERS, 0, 6, max98088_dai1_fltr),
-};
-static const struct soc_enum max98088_dai1_adc_filter_enum[] = {
- SOC_ENUM_SINGLE(M98088_REG_18_DAI1_FILTERS, 4, 6, max98088_dai1_fltr),
-};
+static SOC_ENUM_SINGLE_DECL(max98088_dai1_dac_filter_enum,
+ M98088_REG_18_DAI1_FILTERS, 0,
+ max98088_dai1_fltr);
+static SOC_ENUM_SINGLE_DECL(max98088_dai1_adc_filter_enum,
+ M98088_REG_18_DAI1_FILTERS, 4,
+ max98088_dai1_fltr);
static int max98088_mic1pre_set(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
@@ -1849,7 +1848,7 @@ static void max98088_handle_eq_pdata(struct snd_soc_codec *codec)
/* Now point the soc_enum to .texts array items */
max98088->eq_enum.texts = max98088->eq_texts;
- max98088->eq_enum.max = max98088->eq_textcnt;
+ max98088->eq_enum.items = max98088->eq_textcnt;
ret = snd_soc_add_codec_controls(codec, controls, ARRAY_SIZE(controls));
if (ret != 0)
@@ -1915,12 +1914,6 @@ static int max98088_probe(struct snd_soc_codec *codec)
regcache_mark_dirty(max98088->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;
- }
-
/* initialize private data */
max98088->sysclk = (unsigned)-1;
diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c
index 9f714ea..98c6e10 100644
--- a/sound/soc/codecs/max98090.c
+++ b/sound/soc/codecs/max98090.c
@@ -513,65 +513,75 @@ static const char *max98090_perf_pwr_text[] =
static const char *max98090_pwr_perf_text[] =
{ "Low Power", "High Performance" };
-static const struct soc_enum max98090_vcmbandgap_enum =
- SOC_ENUM_SINGLE(M98090_REG_BIAS_CONTROL, M98090_VCM_MODE_SHIFT,
- ARRAY_SIZE(max98090_pwr_perf_text), max98090_pwr_perf_text);
+static SOC_ENUM_SINGLE_DECL(max98090_vcmbandgap_enum,
+ M98090_REG_BIAS_CONTROL,
+ M98090_VCM_MODE_SHIFT,
+ max98090_pwr_perf_text);
static const char *max98090_osr128_text[] = { "64*fs", "128*fs" };
-static const struct soc_enum max98090_osr128_enum =
- SOC_ENUM_SINGLE(M98090_REG_ADC_CONTROL, M98090_OSR128_SHIFT,
- ARRAY_SIZE(max98090_osr128_text), max98090_osr128_text);
+static SOC_ENUM_SINGLE_DECL(max98090_osr128_enum,
+ M98090_REG_ADC_CONTROL,
+ M98090_OSR128_SHIFT,
+ max98090_osr128_text);
static const char *max98090_mode_text[] = { "Voice", "Music" };
-static const struct soc_enum max98090_mode_enum =
- SOC_ENUM_SINGLE(M98090_REG_FILTER_CONFIG, M98090_MODE_SHIFT,
- ARRAY_SIZE(max98090_mode_text), max98090_mode_text);
+static SOC_ENUM_SINGLE_DECL(max98090_mode_enum,
+ M98090_REG_FILTER_CONFIG,
+ M98090_MODE_SHIFT,
+ max98090_mode_text);
-static const struct soc_enum max98090_filter_dmic34mode_enum =
- SOC_ENUM_SINGLE(M98090_REG_FILTER_CONFIG,
- M98090_FLT_DMIC34MODE_SHIFT,
- ARRAY_SIZE(max98090_mode_text), max98090_mode_text);
+static SOC_ENUM_SINGLE_DECL(max98090_filter_dmic34mode_enum,
+ M98090_REG_FILTER_CONFIG,
+ M98090_FLT_DMIC34MODE_SHIFT,
+ max98090_mode_text);
static const char *max98090_drcatk_text[] =
{ "0.5ms", "1ms", "5ms", "10ms", "25ms", "50ms", "100ms", "200ms" };
-static const struct soc_enum max98090_drcatk_enum =
- SOC_ENUM_SINGLE(M98090_REG_DRC_TIMING, M98090_DRCATK_SHIFT,
- ARRAY_SIZE(max98090_drcatk_text), max98090_drcatk_text);
+static SOC_ENUM_SINGLE_DECL(max98090_drcatk_enum,
+ M98090_REG_DRC_TIMING,
+ M98090_DRCATK_SHIFT,
+ max98090_drcatk_text);
static const char *max98090_drcrls_text[] =
{ "8s", "4s", "2s", "1s", "0.5s", "0.25s", "0.125s", "0.0625s" };
-static const struct soc_enum max98090_drcrls_enum =
- SOC_ENUM_SINGLE(M98090_REG_DRC_TIMING, M98090_DRCRLS_SHIFT,
- ARRAY_SIZE(max98090_drcrls_text), max98090_drcrls_text);
+static SOC_ENUM_SINGLE_DECL(max98090_drcrls_enum,
+ M98090_REG_DRC_TIMING,
+ M98090_DRCRLS_SHIFT,
+ max98090_drcrls_text);
static const char *max98090_alccmp_text[] =
{ "1:1", "1:1.5", "1:2", "1:4", "1:INF" };
-static const struct soc_enum max98090_alccmp_enum =
- SOC_ENUM_SINGLE(M98090_REG_DRC_COMPRESSOR, M98090_DRCCMP_SHIFT,
- ARRAY_SIZE(max98090_alccmp_text), max98090_alccmp_text);
+static SOC_ENUM_SINGLE_DECL(max98090_alccmp_enum,
+ M98090_REG_DRC_COMPRESSOR,
+ M98090_DRCCMP_SHIFT,
+ max98090_alccmp_text);
static const char *max98090_drcexp_text[] = { "1:1", "2:1", "3:1" };
-static const struct soc_enum max98090_drcexp_enum =
- SOC_ENUM_SINGLE(M98090_REG_DRC_EXPANDER, M98090_DRCEXP_SHIFT,
- ARRAY_SIZE(max98090_drcexp_text), max98090_drcexp_text);
+static SOC_ENUM_SINGLE_DECL(max98090_drcexp_enum,
+ M98090_REG_DRC_EXPANDER,
+ M98090_DRCEXP_SHIFT,
+ max98090_drcexp_text);
-static const struct soc_enum max98090_dac_perfmode_enum =
- SOC_ENUM_SINGLE(M98090_REG_DAC_CONTROL, M98090_PERFMODE_SHIFT,
- ARRAY_SIZE(max98090_perf_pwr_text), max98090_perf_pwr_text);
+static SOC_ENUM_SINGLE_DECL(max98090_dac_perfmode_enum,
+ M98090_REG_DAC_CONTROL,
+ M98090_PERFMODE_SHIFT,
+ max98090_perf_pwr_text);
-static const struct soc_enum max98090_dachp_enum =
- SOC_ENUM_SINGLE(M98090_REG_DAC_CONTROL, M98090_DACHP_SHIFT,
- ARRAY_SIZE(max98090_pwr_perf_text), max98090_pwr_perf_text);
+static SOC_ENUM_SINGLE_DECL(max98090_dachp_enum,
+ M98090_REG_DAC_CONTROL,
+ M98090_DACHP_SHIFT,
+ max98090_pwr_perf_text);
-static const struct soc_enum max98090_adchp_enum =
- SOC_ENUM_SINGLE(M98090_REG_ADC_CONTROL, M98090_ADCHP_SHIFT,
- ARRAY_SIZE(max98090_pwr_perf_text), max98090_pwr_perf_text);
+static SOC_ENUM_SINGLE_DECL(max98090_adchp_enum,
+ M98090_REG_ADC_CONTROL,
+ M98090_ADCHP_SHIFT,
+ max98090_pwr_perf_text);
static const struct snd_kcontrol_new max98090_snd_controls[] = {
SOC_ENUM("MIC Bias VCM Bandgap", max98090_vcmbandgap_enum),
@@ -842,39 +852,42 @@ static int max98090_micinput_event(struct snd_soc_dapm_widget *w,
static const char *mic1_mux_text[] = { "IN12", "IN56" };
-static const struct soc_enum mic1_mux_enum =
- SOC_ENUM_SINGLE(M98090_REG_INPUT_MODE, M98090_EXTMIC1_SHIFT,
- ARRAY_SIZE(mic1_mux_text), mic1_mux_text);
+static SOC_ENUM_SINGLE_DECL(mic1_mux_enum,
+ M98090_REG_INPUT_MODE,
+ M98090_EXTMIC1_SHIFT,
+ mic1_mux_text);
static const struct snd_kcontrol_new max98090_mic1_mux =
SOC_DAPM_ENUM("MIC1 Mux", mic1_mux_enum);
static const char *mic2_mux_text[] = { "IN34", "IN56" };
-static const struct soc_enum mic2_mux_enum =
- SOC_ENUM_SINGLE(M98090_REG_INPUT_MODE, M98090_EXTMIC2_SHIFT,
- ARRAY_SIZE(mic2_mux_text), mic2_mux_text);
+static SOC_ENUM_SINGLE_DECL(mic2_mux_enum,
+ M98090_REG_INPUT_MODE,
+ M98090_EXTMIC2_SHIFT,
+ mic2_mux_text);
static const struct snd_kcontrol_new max98090_mic2_mux =
SOC_DAPM_ENUM("MIC2 Mux", mic2_mux_enum);
static const char *dmic_mux_text[] = { "ADC", "DMIC" };
-static const struct soc_enum dmic_mux_enum =
- SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(dmic_mux_text), dmic_mux_text);
+static SOC_ENUM_SINGLE_VIRT_DECL(dmic_mux_enum, dmic_mux_text);
static const struct snd_kcontrol_new max98090_dmic_mux =
SOC_DAPM_ENUM_VIRT("DMIC Mux", dmic_mux_enum);
static const char *max98090_micpre_text[] = { "Off", "On" };
-static const struct soc_enum max98090_pa1en_enum =
- SOC_ENUM_SINGLE(M98090_REG_MIC1_INPUT_LEVEL, M98090_MIC_PA1EN_SHIFT,
- ARRAY_SIZE(max98090_micpre_text), max98090_micpre_text);
+static SOC_ENUM_SINGLE_DECL(max98090_pa1en_enum,
+ M98090_REG_MIC1_INPUT_LEVEL,
+ M98090_MIC_PA1EN_SHIFT,
+ max98090_micpre_text);
-static const struct soc_enum max98090_pa2en_enum =
- SOC_ENUM_SINGLE(M98090_REG_MIC2_INPUT_LEVEL, M98090_MIC_PA2EN_SHIFT,
- ARRAY_SIZE(max98090_micpre_text), max98090_micpre_text);
+static SOC_ENUM_SINGLE_DECL(max98090_pa2en_enum,
+ M98090_REG_MIC2_INPUT_LEVEL,
+ M98090_MIC_PA2EN_SHIFT,
+ max98090_micpre_text);
/* LINEA mixer switch */
static const struct snd_kcontrol_new max98090_linea_mixer_controls[] = {
@@ -938,13 +951,15 @@ static const struct snd_kcontrol_new max98090_right_adc_mixer_controls[] = {
static const char *lten_mux_text[] = { "Normal", "Loopthrough" };
-static const struct soc_enum ltenl_mux_enum =
- SOC_ENUM_SINGLE(M98090_REG_IO_CONFIGURATION, M98090_LTEN_SHIFT,
- ARRAY_SIZE(lten_mux_text), lten_mux_text);
+static SOC_ENUM_SINGLE_DECL(ltenl_mux_enum,
+ M98090_REG_IO_CONFIGURATION,
+ M98090_LTEN_SHIFT,
+ lten_mux_text);
-static const struct soc_enum ltenr_mux_enum =
- SOC_ENUM_SINGLE(M98090_REG_IO_CONFIGURATION, M98090_LTEN_SHIFT,
- ARRAY_SIZE(lten_mux_text), lten_mux_text);
+static SOC_ENUM_SINGLE_DECL(ltenr_mux_enum,
+ M98090_REG_IO_CONFIGURATION,
+ M98090_LTEN_SHIFT,
+ lten_mux_text);
static const struct snd_kcontrol_new max98090_ltenl_mux =
SOC_DAPM_ENUM("LTENL Mux", ltenl_mux_enum);
@@ -954,13 +969,15 @@ static const struct snd_kcontrol_new max98090_ltenr_mux =
static const char *lben_mux_text[] = { "Normal", "Loopback" };
-static const struct soc_enum lbenl_mux_enum =
- SOC_ENUM_SINGLE(M98090_REG_IO_CONFIGURATION, M98090_LBEN_SHIFT,
- ARRAY_SIZE(lben_mux_text), lben_mux_text);
+static SOC_ENUM_SINGLE_DECL(lbenl_mux_enum,
+ M98090_REG_IO_CONFIGURATION,
+ M98090_LBEN_SHIFT,
+ lben_mux_text);
-static const struct soc_enum lbenr_mux_enum =
- SOC_ENUM_SINGLE(M98090_REG_IO_CONFIGURATION, M98090_LBEN_SHIFT,
- ARRAY_SIZE(lben_mux_text), lben_mux_text);
+static SOC_ENUM_SINGLE_DECL(lbenr_mux_enum,
+ M98090_REG_IO_CONFIGURATION,
+ M98090_LBEN_SHIFT,
+ lben_mux_text);
static const struct snd_kcontrol_new max98090_lbenl_mux =
SOC_DAPM_ENUM("LBENL Mux", lbenl_mux_enum);
@@ -972,13 +989,15 @@ static const char *stenl_mux_text[] = { "Normal", "Sidetone Left" };
static const char *stenr_mux_text[] = { "Normal", "Sidetone Right" };
-static const struct soc_enum stenl_mux_enum =
- SOC_ENUM_SINGLE(M98090_REG_ADC_SIDETONE, M98090_DSTSL_SHIFT,
- ARRAY_SIZE(stenl_mux_text), stenl_mux_text);
+static SOC_ENUM_SINGLE_DECL(stenl_mux_enum,
+ M98090_REG_ADC_SIDETONE,
+ M98090_DSTSL_SHIFT,
+ stenl_mux_text);
-static const struct soc_enum stenr_mux_enum =
- SOC_ENUM_SINGLE(M98090_REG_ADC_SIDETONE, M98090_DSTSR_SHIFT,
- ARRAY_SIZE(stenr_mux_text), stenr_mux_text);
+static SOC_ENUM_SINGLE_DECL(stenr_mux_enum,
+ M98090_REG_ADC_SIDETONE,
+ M98090_DSTSR_SHIFT,
+ stenr_mux_text);
static const struct snd_kcontrol_new max98090_stenl_mux =
SOC_DAPM_ENUM("STENL Mux", stenl_mux_enum);
@@ -1086,9 +1105,10 @@ static const struct snd_kcontrol_new max98090_right_rcv_mixer_controls[] = {
static const char *linmod_mux_text[] = { "Left Only", "Left and Right" };
-static const struct soc_enum linmod_mux_enum =
- SOC_ENUM_SINGLE(M98090_REG_LOUTR_MIXER, M98090_LINMOD_SHIFT,
- ARRAY_SIZE(linmod_mux_text), linmod_mux_text);
+static SOC_ENUM_SINGLE_DECL(linmod_mux_enum,
+ M98090_REG_LOUTR_MIXER,
+ M98090_LINMOD_SHIFT,
+ linmod_mux_text);
static const struct snd_kcontrol_new max98090_linmod_mux =
SOC_DAPM_ENUM("LINMOD Mux", linmod_mux_enum);
@@ -1098,16 +1118,18 @@ static const char *mixhpsel_mux_text[] = { "DAC Only", "HP Mixer" };
/*
* This is a mux as it selects the HP output, but to DAPM it is a Mixer enable
*/
-static const struct soc_enum mixhplsel_mux_enum =
- SOC_ENUM_SINGLE(M98090_REG_HP_CONTROL, M98090_MIXHPLSEL_SHIFT,
- ARRAY_SIZE(mixhpsel_mux_text), mixhpsel_mux_text);
+static SOC_ENUM_SINGLE_DECL(mixhplsel_mux_enum,
+ M98090_REG_HP_CONTROL,
+ M98090_MIXHPLSEL_SHIFT,
+ mixhpsel_mux_text);
static const struct snd_kcontrol_new max98090_mixhplsel_mux =
SOC_DAPM_ENUM("MIXHPLSEL Mux", mixhplsel_mux_enum);
-static const struct soc_enum mixhprsel_mux_enum =
- SOC_ENUM_SINGLE(M98090_REG_HP_CONTROL, M98090_MIXHPRSEL_SHIFT,
- ARRAY_SIZE(mixhpsel_mux_text), mixhpsel_mux_text);
+static SOC_ENUM_SINGLE_DECL(mixhprsel_mux_enum,
+ M98090_REG_HP_CONTROL,
+ M98090_MIXHPRSEL_SHIFT,
+ mixhpsel_mux_text);
static const struct snd_kcontrol_new max98090_mixhprsel_mux =
SOC_DAPM_ENUM("MIXHPRSEL Mux", mixhprsel_mux_enum);
@@ -2196,14 +2218,6 @@ static int max98090_probe(struct snd_soc_codec *codec)
max98090->codec = codec;
- codec->control_data = max98090->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;
- }
-
/* Reset the codec, the DSP core, and disable all interrupts */
max98090_reset(max98090);
@@ -2329,7 +2343,6 @@ static int max98090_i2c_probe(struct i2c_client *i2c,
max98090->devtype = id->driver_data;
i2c_set_clientdata(i2c, max98090);
- max98090->control_data = i2c;
max98090->pdata = i2c->dev.platform_data;
max98090->irq = i2c->irq;
diff --git a/sound/soc/codecs/max98090.h b/sound/soc/codecs/max98090.h
index 7e103f2..1a4e233 100644
--- a/sound/soc/codecs/max98090.h
+++ b/sound/soc/codecs/max98090.h
@@ -1523,7 +1523,6 @@ struct max98090_priv {
struct regmap *regmap;
struct snd_soc_codec *codec;
enum max98090_type devtype;
- void *control_data;
struct max98090_pdata *pdata;
unsigned int sysclk;
unsigned int bclk;
diff --git a/sound/soc/codecs/max98095.c b/sound/soc/codecs/max98095.c
index 3ba1170..03f0536 100644
--- a/sound/soc/codecs/max98095.c
+++ b/sound/soc/codecs/max98095.c
@@ -560,25 +560,27 @@ static void m98095_biquad_band(struct snd_soc_codec *codec, unsigned int dai,
}
static const char * const max98095_fltr_mode[] = { "Voice", "Music" };
-static const struct soc_enum max98095_dai1_filter_mode_enum[] = {
- SOC_ENUM_SINGLE(M98095_02E_DAI1_FILTERS, 7, 2, max98095_fltr_mode),
-};
-static const struct soc_enum max98095_dai2_filter_mode_enum[] = {
- SOC_ENUM_SINGLE(M98095_038_DAI2_FILTERS, 7, 2, max98095_fltr_mode),
-};
+static SOC_ENUM_SINGLE_DECL(max98095_dai1_filter_mode_enum,
+ M98095_02E_DAI1_FILTERS, 7,
+ max98095_fltr_mode);
+static SOC_ENUM_SINGLE_DECL(max98095_dai2_filter_mode_enum,
+ M98095_038_DAI2_FILTERS, 7,
+ max98095_fltr_mode);
static const char * const max98095_extmic_text[] = { "None", "MIC1", "MIC2" };
-static const struct soc_enum max98095_extmic_enum =
- SOC_ENUM_SINGLE(M98095_087_CFG_MIC, 0, 3, max98095_extmic_text);
+static SOC_ENUM_SINGLE_DECL(max98095_extmic_enum,
+ M98095_087_CFG_MIC, 0,
+ max98095_extmic_text);
static const struct snd_kcontrol_new max98095_extmic_mux =
SOC_DAPM_ENUM("External MIC Mux", max98095_extmic_enum);
static const char * const max98095_linein_text[] = { "INA", "INB" };
-static const struct soc_enum max98095_linein_enum =
- SOC_ENUM_SINGLE(M98095_086_CFG_LINE, 6, 2, max98095_linein_text);
+static SOC_ENUM_SINGLE_DECL(max98095_linein_enum,
+ M98095_086_CFG_LINE, 6,
+ max98095_linein_text);
static const struct snd_kcontrol_new max98095_linein_mux =
SOC_DAPM_ENUM("Linein Input Mux", max98095_linein_enum);
@@ -586,24 +588,26 @@ static const struct snd_kcontrol_new max98095_linein_mux =
static const char * const max98095_line_mode_text[] = {
"Stereo", "Differential"};
-static const struct soc_enum max98095_linein_mode_enum =
- SOC_ENUM_SINGLE(M98095_086_CFG_LINE, 7, 2, max98095_line_mode_text);
+static SOC_ENUM_SINGLE_DECL(max98095_linein_mode_enum,
+ M98095_086_CFG_LINE, 7,
+ max98095_line_mode_text);
-static const struct soc_enum max98095_lineout_mode_enum =
- SOC_ENUM_SINGLE(M98095_086_CFG_LINE, 4, 2, max98095_line_mode_text);
+static SOC_ENUM_SINGLE_DECL(max98095_lineout_mode_enum,
+ M98095_086_CFG_LINE, 4,
+ max98095_line_mode_text);
static const char * const max98095_dai_fltr[] = {
"Off", "Elliptical-HPF-16k", "Butterworth-HPF-16k",
"Elliptical-HPF-8k", "Butterworth-HPF-8k", "Butterworth-HPF-Fs/240"};
-static const struct soc_enum max98095_dai1_dac_filter_enum[] = {
- SOC_ENUM_SINGLE(M98095_02E_DAI1_FILTERS, 0, 6, max98095_dai_fltr),
-};
-static const struct soc_enum max98095_dai2_dac_filter_enum[] = {
- SOC_ENUM_SINGLE(M98095_038_DAI2_FILTERS, 0, 6, max98095_dai_fltr),
-};
-static const struct soc_enum max98095_dai3_dac_filter_enum[] = {
- SOC_ENUM_SINGLE(M98095_042_DAI3_FILTERS, 0, 6, max98095_dai_fltr),
-};
+static SOC_ENUM_SINGLE_DECL(max98095_dai1_dac_filter_enum,
+ M98095_02E_DAI1_FILTERS, 0,
+ max98095_dai_fltr);
+static SOC_ENUM_SINGLE_DECL(max98095_dai2_dac_filter_enum,
+ M98095_038_DAI2_FILTERS, 0,
+ max98095_dai_fltr);
+static SOC_ENUM_SINGLE_DECL(max98095_dai3_dac_filter_enum,
+ M98095_042_DAI3_FILTERS, 0,
+ max98095_dai_fltr);
static int max98095_mic1pre_set(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
@@ -1861,7 +1865,7 @@ static void max98095_handle_eq_pdata(struct snd_soc_codec *codec)
/* Now point the soc_enum to .texts array items */
max98095->eq_enum.texts = max98095->eq_texts;
- max98095->eq_enum.max = max98095->eq_textcnt;
+ max98095->eq_enum.items = max98095->eq_textcnt;
ret = snd_soc_add_codec_controls(codec, controls, ARRAY_SIZE(controls));
if (ret != 0)
@@ -2016,7 +2020,7 @@ static void max98095_handle_bq_pdata(struct snd_soc_codec *codec)
/* Now point the soc_enum to .texts array items */
max98095->bq_enum.texts = max98095->bq_texts;
- max98095->bq_enum.max = max98095->bq_textcnt;
+ max98095->bq_enum.items = max98095->bq_textcnt;
ret = snd_soc_add_codec_controls(codec, controls, ARRAY_SIZE(controls));
if (ret != 0)
@@ -2234,12 +2238,6 @@ static int max98095_probe(struct snd_soc_codec *codec)
struct i2c_client *client;
int ret = 0;
- 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;
- }
-
/* reset the codec, the DSP core, and disable all interrupts */
max98095_reset(codec);
diff --git a/sound/soc/codecs/max9850.c b/sound/soc/codecs/max9850.c
index 82757eb..4fdf5aa 100644
--- a/sound/soc/codecs/max9850.c
+++ b/sound/soc/codecs/max9850.c
@@ -312,14 +312,6 @@ static int max9850_resume(struct snd_soc_codec *codec)
static int max9850_probe(struct snd_soc_codec *codec)
{
- int ret;
-
- 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;
- }
-
/* enable zero-detect */
snd_soc_update_bits(codec, MAX9850_GENERAL_PURPOSE, 1, 1);
/* enable slew-rate control */
diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c
index 582c2bb..2c59b1f 100644
--- a/sound/soc/codecs/mc13783.c
+++ b/sound/soc/codecs/mc13783.c
@@ -106,8 +106,7 @@ 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;
+ struct snd_soc_codec *codec = dai->codec;
unsigned int rate = params_rate(params);
int i;
@@ -126,8 +125,7 @@ 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;
+ struct snd_soc_codec *codec = dai->codec;
unsigned int rate = params_rate(params);
unsigned int val;
@@ -408,8 +406,7 @@ 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 SOC_ENUM_SINGLE_VIRT_DECL(adcl_enum, adcl_enum_text);
static const struct snd_kcontrol_new left_input_mux =
SOC_DAPM_ENUM_VIRT("Route", adcl_enum);
@@ -418,8 +415,7 @@ 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 SOC_ENUM_SINGLE_VIRT_DECL(adcr_enum, adcr_enum_text);
static const struct snd_kcontrol_new right_input_mux =
SOC_DAPM_ENUM_VIRT("Route", adcr_enum);
@@ -430,8 +426,8 @@ static const struct snd_kcontrol_new samp_ctl =
static const char * const speaker_amp_source_text[] = {
"CODEC", "Right"
};
-static const SOC_ENUM_SINGLE_DECL(speaker_amp_source, MC13783_AUDIO_RX0, 4,
- speaker_amp_source_text);
+static SOC_ENUM_SINGLE_DECL(speaker_amp_source, MC13783_AUDIO_RX0, 4,
+ speaker_amp_source_text);
static const struct snd_kcontrol_new speaker_amp_source_mux =
SOC_DAPM_ENUM("Speaker Amp Source MUX", speaker_amp_source);
@@ -439,8 +435,8 @@ static const char * const headset_amp_source_text[] = {
"CODEC", "Mixer"
};
-static const SOC_ENUM_SINGLE_DECL(headset_amp_source, MC13783_AUDIO_RX0, 11,
- headset_amp_source_text);
+static SOC_ENUM_SINGLE_DECL(headset_amp_source, MC13783_AUDIO_RX0, 11,
+ headset_amp_source_text);
static const struct snd_kcontrol_new headset_amp_source_mux =
SOC_DAPM_ENUM("Headset Amp Source MUX", headset_amp_source);
@@ -580,9 +576,9 @@ static struct snd_soc_dapm_route mc13783_routes[] = {
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 SOC_ENUM_SINGLE_DECL(mc13783_enum_3d_mixer,
+ MC13783_AUDIO_RX1, 16,
+ mc13783_3d_mixer);
static struct snd_kcontrol_new mc13783_control_list[] = {
SOC_SINGLE("Loudspeaker enable", MC13783_AUDIO_RX0, 5, 1, 0),
@@ -614,8 +610,8 @@ static int mc13783_probe(struct snd_soc_codec *codec)
struct mc13783_priv *priv = snd_soc_codec_get_drvdata(codec);
int ret;
- codec->control_data = dev_get_regmap(codec->dev->parent, NULL);
- ret = snd_soc_codec_set_cache_io(codec, 8, 24, SND_SOC_REGMAP);
+ ret = snd_soc_codec_set_cache_io(codec,
+ dev_get_regmap(codec->dev->parent, NULL));
if (ret != 0) {
dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
return ret;
diff --git a/sound/soc/codecs/ml26124.c b/sound/soc/codecs/ml26124.c
index 185fa3bc..e661e84 100644
--- a/sound/soc/codecs/ml26124.c
+++ b/sound/soc/codecs/ml26124.c
@@ -73,11 +73,11 @@ 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 SOC_ENUM_SINGLE_DECL(ml26124_adc_companding_enum,
+ ML26124_SAI_TRANS_CTL, 6, ml26124_companding);
-static const struct soc_enum ml26124_dac_companding_enum
- = SOC_ENUM_SINGLE(ML26124_SAI_RCV_CTL, 6, 3, ml26124_companding);
+static SOC_ENUM_SINGLE_DECL(ml26124_dac_companding_enum,
+ ML26124_SAI_RCV_CTL, 6, ml26124_companding);
static const struct snd_kcontrol_new ml26124_snd_controls[] = {
SOC_SINGLE_TLV("Capture Digital Volume", ML26124_RECORD_DIG_VOL, 0,
@@ -136,8 +136,8 @@ static const struct snd_kcontrol_new ml26124_output_mixer_controls[] = {
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 SOC_ENUM_SINGLE_DECL(ml26124_insel_enum,
+ ML26124_MIC_IF_CTL, 0, ml26124_input_select);
static const struct snd_kcontrol_new ml26124_input_mux_controls =
SOC_DAPM_ENUM("Input Select", ml26124_insel_enum);
@@ -586,16 +586,6 @@ static int ml26124_resume(struct snd_soc_codec *codec)
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);
diff --git a/sound/soc/codecs/pcm1681.c b/sound/soc/codecs/pcm1681.c
index 73f9c36..e427544 100644
--- a/sound/soc/codecs/pcm1681.c
+++ b/sound/soc/codecs/pcm1681.c
@@ -172,16 +172,21 @@ static int pcm1681_hw_params(struct snd_pcm_substream *substream,
struct snd_soc_codec *codec = dai->codec;
struct pcm1681_private *priv = snd_soc_codec_get_drvdata(codec);
int val = 0, ret;
- int pcm_format = params_format(params);
priv->rate = params_rate(params);
switch (priv->format & SND_SOC_DAIFMT_FORMAT_MASK) {
case SND_SOC_DAIFMT_RIGHT_J:
- if (pcm_format == SNDRV_PCM_FORMAT_S24_LE)
- val = 0x00;
- else if (pcm_format == SNDRV_PCM_FORMAT_S16_LE)
- val = 0x03;
+ switch (params_width(params)) {
+ case 24:
+ val = 0;
+ break;
+ case 16:
+ val = 3;
+ break;
+ default:
+ return -EINVAL;
+ }
break;
case SND_SOC_DAIFMT_I2S:
val = 0x04;
diff --git a/sound/soc/codecs/pcm1792a.c b/sound/soc/codecs/pcm1792a.c
index 7146653a..3a80ba4 100644
--- a/sound/soc/codecs/pcm1792a.c
+++ b/sound/soc/codecs/pcm1792a.c
@@ -107,24 +107,35 @@ static int pcm1792a_hw_params(struct snd_pcm_substream *substream,
struct snd_soc_codec *codec = dai->codec;
struct pcm1792a_private *priv = snd_soc_codec_get_drvdata(codec);
int val = 0, ret;
- int pcm_format = params_format(params);
priv->rate = params_rate(params);
switch (priv->format & SND_SOC_DAIFMT_FORMAT_MASK) {
case SND_SOC_DAIFMT_RIGHT_J:
- if (pcm_format == SNDRV_PCM_FORMAT_S24_LE ||
- pcm_format == SNDRV_PCM_FORMAT_S32_LE)
- val = 0x02;
- else if (pcm_format == SNDRV_PCM_FORMAT_S16_LE)
- val = 0x00;
+ switch (params_width(params)) {
+ case 24:
+ case 32:
+ val = 2;
+ break;
+ case 16:
+ val = 0;
+ break;
+ default:
+ return -EINVAL;
+ }
break;
case SND_SOC_DAIFMT_I2S:
- if (pcm_format == SNDRV_PCM_FORMAT_S24_LE ||
- pcm_format == SNDRV_PCM_FORMAT_S32_LE)
- val = 0x05;
- else if (pcm_format == SNDRV_PCM_FORMAT_S16_LE)
- val = 0x04;
+ switch (params_width(params)) {
+ case 24:
+ case 32:
+ val = 5;
+ break;
+ case 16:
+ val = 4;
+ break;
+ default:
+ return -EINVAL;
+ }
break;
default:
dev_err(codec->dev, "Invalid DAI format\n");
diff --git a/sound/soc/codecs/pcm512x-i2c.c b/sound/soc/codecs/pcm512x-i2c.c
new file mode 100644
index 0000000..4d62230
--- /dev/null
+++ b/sound/soc/codecs/pcm512x-i2c.c
@@ -0,0 +1,71 @@
+/*
+ * Driver for the PCM512x CODECs
+ *
+ * Author: Mark Brown <broonie@linaro.org>
+ * Copyright 2014 Linaro Ltd
+ *
+ * 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.
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/i2c.h>
+
+#include "pcm512x.h"
+
+static int pcm512x_i2c_probe(struct i2c_client *i2c,
+ const struct i2c_device_id *id)
+{
+ struct regmap *regmap;
+
+ regmap = devm_regmap_init_i2c(i2c, &pcm512x_regmap);
+ if (IS_ERR(regmap))
+ return PTR_ERR(regmap);
+
+ return pcm512x_probe(&i2c->dev, regmap);
+}
+
+static int pcm512x_i2c_remove(struct i2c_client *i2c)
+{
+ pcm512x_remove(&i2c->dev);
+ return 0;
+}
+
+static const struct i2c_device_id pcm512x_i2c_id[] = {
+ { "pcm5121", },
+ { "pcm5122", },
+ { }
+};
+MODULE_DEVICE_TABLE(i2c, pcm512x_i2c_id);
+
+static const struct of_device_id pcm512x_of_match[] = {
+ { .compatible = "ti,pcm5121", },
+ { .compatible = "ti,pcm5122", },
+ { }
+};
+MODULE_DEVICE_TABLE(of, pcm512x_of_match);
+
+static struct i2c_driver pcm512x_i2c_driver = {
+ .probe = pcm512x_i2c_probe,
+ .remove = pcm512x_i2c_remove,
+ .id_table = pcm512x_i2c_id,
+ .driver = {
+ .name = "pcm512x",
+ .owner = THIS_MODULE,
+ .of_match_table = pcm512x_of_match,
+ .pm = &pcm512x_pm_ops,
+ },
+};
+
+module_i2c_driver(pcm512x_i2c_driver);
+
+MODULE_DESCRIPTION("ASoC PCM512x codec driver - I2C");
+MODULE_AUTHOR("Mark Brown <broonie@linaro.org>");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/codecs/pcm512x-spi.c b/sound/soc/codecs/pcm512x-spi.c
new file mode 100644
index 0000000..f297058
--- /dev/null
+++ b/sound/soc/codecs/pcm512x-spi.c
@@ -0,0 +1,69 @@
+/*
+ * Driver for the PCM512x CODECs
+ *
+ * Author: Mark Brown <broonie@linaro.org>
+ * Copyright 2014 Linaro Ltd
+ *
+ * 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.
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/spi/spi.h>
+
+#include "pcm512x.h"
+
+static int pcm512x_spi_probe(struct spi_device *spi)
+{
+ struct regmap *regmap;
+ int ret;
+
+ regmap = devm_regmap_init_spi(spi, &pcm512x_regmap);
+ if (IS_ERR(regmap)) {
+ ret = PTR_ERR(regmap);
+ return ret;
+ }
+
+ return pcm512x_probe(&spi->dev, regmap);
+}
+
+static int pcm512x_spi_remove(struct spi_device *spi)
+{
+ pcm512x_remove(&spi->dev);
+ return 0;
+}
+
+static const struct spi_device_id pcm512x_spi_id[] = {
+ { "pcm5121", },
+ { "pcm5122", },
+ { },
+};
+MODULE_DEVICE_TABLE(spi, pcm512x_spi_id);
+
+static const struct of_device_id pcm512x_of_match[] = {
+ { .compatible = "ti,pcm5121", },
+ { .compatible = "ti,pcm5122", },
+ { }
+};
+MODULE_DEVICE_TABLE(of, pcm512x_of_match);
+
+static struct spi_driver pcm512x_spi_driver = {
+ .probe = pcm512x_spi_probe,
+ .remove = pcm512x_spi_remove,
+ .id_table = pcm512x_spi_id,
+ .driver = {
+ .name = "pcm512x",
+ .owner = THIS_MODULE,
+ .of_match_table = pcm512x_of_match,
+ .pm = &pcm512x_pm_ops,
+ },
+};
+
+module_spi_driver(pcm512x_spi_driver);
diff --git a/sound/soc/codecs/pcm512x.c b/sound/soc/codecs/pcm512x.c
new file mode 100644
index 0000000..4b4c0c7
--- /dev/null
+++ b/sound/soc/codecs/pcm512x.c
@@ -0,0 +1,589 @@
+/*
+ * Driver for the PCM512x CODECs
+ *
+ * Author: Mark Brown <broonie@linaro.org>
+ * Copyright 2014 Linaro Ltd
+ *
+ * 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.
+ */
+
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/clk.h>
+#include <linux/pm_runtime.h>
+#include <linux/regmap.h>
+#include <linux/regulator/consumer.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/tlv.h>
+
+#include "pcm512x.h"
+
+#define PCM512x_NUM_SUPPLIES 3
+static const char * const pcm512x_supply_names[PCM512x_NUM_SUPPLIES] = {
+ "AVDD",
+ "DVDD",
+ "CPVDD",
+};
+
+struct pcm512x_priv {
+ struct regmap *regmap;
+ struct clk *sclk;
+ struct regulator_bulk_data supplies[PCM512x_NUM_SUPPLIES];
+ struct notifier_block supply_nb[PCM512x_NUM_SUPPLIES];
+};
+
+/*
+ * We can't use the same notifier block for more than one supply and
+ * there's no way I can see to get from a callback to the caller
+ * except container_of().
+ */
+#define PCM512x_REGULATOR_EVENT(n) \
+static int pcm512x_regulator_event_##n(struct notifier_block *nb, \
+ unsigned long event, void *data) \
+{ \
+ struct pcm512x_priv *pcm512x = container_of(nb, struct pcm512x_priv, \
+ supply_nb[n]); \
+ if (event & REGULATOR_EVENT_DISABLE) { \
+ regcache_mark_dirty(pcm512x->regmap); \
+ regcache_cache_only(pcm512x->regmap, true); \
+ } \
+ return 0; \
+}
+
+PCM512x_REGULATOR_EVENT(0)
+PCM512x_REGULATOR_EVENT(1)
+PCM512x_REGULATOR_EVENT(2)
+
+static const struct reg_default pcm512x_reg_defaults[] = {
+ { PCM512x_RESET, 0x00 },
+ { PCM512x_POWER, 0x00 },
+ { PCM512x_MUTE, 0x00 },
+ { PCM512x_DSP, 0x00 },
+ { PCM512x_PLL_REF, 0x00 },
+ { PCM512x_DAC_ROUTING, 0x11 },
+ { PCM512x_DSP_PROGRAM, 0x01 },
+ { PCM512x_CLKDET, 0x00 },
+ { PCM512x_AUTO_MUTE, 0x00 },
+ { PCM512x_ERROR_DETECT, 0x00 },
+ { PCM512x_DIGITAL_VOLUME_1, 0x00 },
+ { PCM512x_DIGITAL_VOLUME_2, 0x30 },
+ { PCM512x_DIGITAL_VOLUME_3, 0x30 },
+ { PCM512x_DIGITAL_MUTE_1, 0x22 },
+ { PCM512x_DIGITAL_MUTE_2, 0x00 },
+ { PCM512x_DIGITAL_MUTE_3, 0x07 },
+ { PCM512x_OUTPUT_AMPLITUDE, 0x00 },
+ { PCM512x_ANALOG_GAIN_CTRL, 0x00 },
+ { PCM512x_UNDERVOLTAGE_PROT, 0x00 },
+ { PCM512x_ANALOG_MUTE_CTRL, 0x00 },
+ { PCM512x_ANALOG_GAIN_BOOST, 0x00 },
+ { PCM512x_VCOM_CTRL_1, 0x00 },
+ { PCM512x_VCOM_CTRL_2, 0x01 },
+};
+
+static bool pcm512x_readable(struct device *dev, unsigned int reg)
+{
+ switch (reg) {
+ case PCM512x_RESET:
+ case PCM512x_POWER:
+ case PCM512x_MUTE:
+ case PCM512x_PLL_EN:
+ case PCM512x_SPI_MISO_FUNCTION:
+ case PCM512x_DSP:
+ case PCM512x_GPIO_EN:
+ case PCM512x_BCLK_LRCLK_CFG:
+ case PCM512x_DSP_GPIO_INPUT:
+ case PCM512x_MASTER_MODE:
+ case PCM512x_PLL_REF:
+ case PCM512x_PLL_COEFF_0:
+ case PCM512x_PLL_COEFF_1:
+ case PCM512x_PLL_COEFF_2:
+ case PCM512x_PLL_COEFF_3:
+ case PCM512x_PLL_COEFF_4:
+ case PCM512x_DSP_CLKDIV:
+ case PCM512x_DAC_CLKDIV:
+ case PCM512x_NCP_CLKDIV:
+ case PCM512x_OSR_CLKDIV:
+ case PCM512x_MASTER_CLKDIV_1:
+ case PCM512x_MASTER_CLKDIV_2:
+ case PCM512x_FS_SPEED_MODE:
+ case PCM512x_IDAC_1:
+ case PCM512x_IDAC_2:
+ case PCM512x_ERROR_DETECT:
+ case PCM512x_I2S_1:
+ case PCM512x_I2S_2:
+ case PCM512x_DAC_ROUTING:
+ case PCM512x_DSP_PROGRAM:
+ case PCM512x_CLKDET:
+ case PCM512x_AUTO_MUTE:
+ case PCM512x_DIGITAL_VOLUME_1:
+ case PCM512x_DIGITAL_VOLUME_2:
+ case PCM512x_DIGITAL_VOLUME_3:
+ case PCM512x_DIGITAL_MUTE_1:
+ case PCM512x_DIGITAL_MUTE_2:
+ case PCM512x_DIGITAL_MUTE_3:
+ case PCM512x_GPIO_OUTPUT_1:
+ case PCM512x_GPIO_OUTPUT_2:
+ case PCM512x_GPIO_OUTPUT_3:
+ case PCM512x_GPIO_OUTPUT_4:
+ case PCM512x_GPIO_OUTPUT_5:
+ case PCM512x_GPIO_OUTPUT_6:
+ case PCM512x_GPIO_CONTROL_1:
+ case PCM512x_GPIO_CONTROL_2:
+ case PCM512x_OVERFLOW:
+ case PCM512x_RATE_DET_1:
+ case PCM512x_RATE_DET_2:
+ case PCM512x_RATE_DET_3:
+ case PCM512x_RATE_DET_4:
+ case PCM512x_ANALOG_MUTE_DET:
+ case PCM512x_GPIN:
+ case PCM512x_DIGITAL_MUTE_DET:
+ case PCM512x_OUTPUT_AMPLITUDE:
+ case PCM512x_ANALOG_GAIN_CTRL:
+ case PCM512x_UNDERVOLTAGE_PROT:
+ case PCM512x_ANALOG_MUTE_CTRL:
+ case PCM512x_ANALOG_GAIN_BOOST:
+ case PCM512x_VCOM_CTRL_1:
+ case PCM512x_VCOM_CTRL_2:
+ case PCM512x_CRAM_CTRL:
+ return true;
+ default:
+ /* There are 256 raw register addresses */
+ return reg < 0xff;
+ }
+}
+
+static bool pcm512x_volatile(struct device *dev, unsigned int reg)
+{
+ switch (reg) {
+ case PCM512x_PLL_EN:
+ case PCM512x_OVERFLOW:
+ case PCM512x_RATE_DET_1:
+ case PCM512x_RATE_DET_2:
+ case PCM512x_RATE_DET_3:
+ case PCM512x_RATE_DET_4:
+ case PCM512x_ANALOG_MUTE_DET:
+ case PCM512x_GPIN:
+ case PCM512x_DIGITAL_MUTE_DET:
+ case PCM512x_CRAM_CTRL:
+ return true;
+ default:
+ /* There are 256 raw register addresses */
+ return reg < 0xff;
+ }
+}
+
+static const DECLARE_TLV_DB_SCALE(digital_tlv, -10350, 50, 1);
+static const DECLARE_TLV_DB_SCALE(analog_tlv, -600, 600, 0);
+static const DECLARE_TLV_DB_SCALE(boost_tlv, 0, 80, 0);
+
+static const char * const pcm512x_dsp_program_texts[] = {
+ "FIR interpolation with de-emphasis",
+ "Low latency IIR with de-emphasis",
+ "Fixed process flow",
+ "High attenuation with de-emphasis",
+ "Ringing-less low latency FIR",
+};
+
+static const unsigned int pcm512x_dsp_program_values[] = {
+ 1,
+ 2,
+ 3,
+ 5,
+ 7,
+};
+
+static SOC_VALUE_ENUM_SINGLE_DECL(pcm512x_dsp_program,
+ PCM512x_DSP_PROGRAM, 0, 0x1f,
+ pcm512x_dsp_program_texts,
+ pcm512x_dsp_program_values);
+
+static const char * const pcm512x_clk_missing_text[] = {
+ "1s", "2s", "3s", "4s", "5s", "6s", "7s", "8s"
+};
+
+static const struct soc_enum pcm512x_clk_missing =
+ SOC_ENUM_SINGLE(PCM512x_CLKDET, 0, 8, pcm512x_clk_missing_text);
+
+static const char * const pcm512x_autom_text[] = {
+ "21ms", "106ms", "213ms", "533ms", "1.07s", "2.13s", "5.33s", "10.66s"
+};
+
+static const struct soc_enum pcm512x_autom_l =
+ SOC_ENUM_SINGLE(PCM512x_AUTO_MUTE, PCM512x_ATML_SHIFT, 8,
+ pcm512x_autom_text);
+
+static const struct soc_enum pcm512x_autom_r =
+ SOC_ENUM_SINGLE(PCM512x_AUTO_MUTE, PCM512x_ATMR_SHIFT, 8,
+ pcm512x_autom_text);
+
+static const char * const pcm512x_ramp_rate_text[] = {
+ "1 sample/update", "2 samples/update", "4 samples/update",
+ "Immediate"
+};
+
+static const struct soc_enum pcm512x_vndf =
+ SOC_ENUM_SINGLE(PCM512x_DIGITAL_MUTE_1, PCM512x_VNDF_SHIFT, 4,
+ pcm512x_ramp_rate_text);
+
+static const struct soc_enum pcm512x_vnuf =
+ SOC_ENUM_SINGLE(PCM512x_DIGITAL_MUTE_1, PCM512x_VNUF_SHIFT, 4,
+ pcm512x_ramp_rate_text);
+
+static const struct soc_enum pcm512x_vedf =
+ SOC_ENUM_SINGLE(PCM512x_DIGITAL_MUTE_2, PCM512x_VEDF_SHIFT, 4,
+ pcm512x_ramp_rate_text);
+
+static const char * const pcm512x_ramp_step_text[] = {
+ "4dB/step", "2dB/step", "1dB/step", "0.5dB/step"
+};
+
+static const struct soc_enum pcm512x_vnds =
+ SOC_ENUM_SINGLE(PCM512x_DIGITAL_MUTE_1, PCM512x_VNDS_SHIFT, 4,
+ pcm512x_ramp_step_text);
+
+static const struct soc_enum pcm512x_vnus =
+ SOC_ENUM_SINGLE(PCM512x_DIGITAL_MUTE_1, PCM512x_VNUS_SHIFT, 4,
+ pcm512x_ramp_step_text);
+
+static const struct soc_enum pcm512x_veds =
+ SOC_ENUM_SINGLE(PCM512x_DIGITAL_MUTE_2, PCM512x_VEDS_SHIFT, 4,
+ pcm512x_ramp_step_text);
+
+static const struct snd_kcontrol_new pcm512x_controls[] = {
+SOC_DOUBLE_R_TLV("Playback Digital Volume", PCM512x_DIGITAL_VOLUME_2,
+ PCM512x_DIGITAL_VOLUME_3, 0, 255, 1, digital_tlv),
+SOC_DOUBLE_TLV("Playback Volume", PCM512x_ANALOG_GAIN_CTRL,
+ PCM512x_LAGN_SHIFT, PCM512x_RAGN_SHIFT, 1, 1, analog_tlv),
+SOC_DOUBLE_TLV("Playback Boost Volume", PCM512x_ANALOG_GAIN_BOOST,
+ PCM512x_AGBL_SHIFT, PCM512x_AGBR_SHIFT, 1, 0, boost_tlv),
+SOC_DOUBLE("Playback Digital Switch", PCM512x_MUTE, PCM512x_RQML_SHIFT,
+ PCM512x_RQMR_SHIFT, 1, 1),
+
+SOC_SINGLE("Deemphasis Switch", PCM512x_DSP, PCM512x_DEMP_SHIFT, 1, 1),
+SOC_VALUE_ENUM("DSP Program", pcm512x_dsp_program),
+
+SOC_ENUM("Clock Missing Period", pcm512x_clk_missing),
+SOC_ENUM("Auto Mute Time Left", pcm512x_autom_l),
+SOC_ENUM("Auto Mute Time Right", pcm512x_autom_r),
+SOC_SINGLE("Auto Mute Mono Switch", PCM512x_DIGITAL_MUTE_3,
+ PCM512x_ACTL_SHIFT, 1, 0),
+SOC_DOUBLE("Auto Mute Switch", PCM512x_DIGITAL_MUTE_3, PCM512x_AMLE_SHIFT,
+ PCM512x_AMLR_SHIFT, 1, 0),
+
+SOC_ENUM("Volume Ramp Down Rate", pcm512x_vndf),
+SOC_ENUM("Volume Ramp Down Step", pcm512x_vnds),
+SOC_ENUM("Volume Ramp Up Rate", pcm512x_vnuf),
+SOC_ENUM("Volume Ramp Up Step", pcm512x_vnus),
+SOC_ENUM("Volume Ramp Down Emergency Rate", pcm512x_vedf),
+SOC_ENUM("Volume Ramp Down Emergency Step", pcm512x_veds),
+};
+
+static const struct snd_soc_dapm_widget pcm512x_dapm_widgets[] = {
+SND_SOC_DAPM_DAC("DACL", NULL, SND_SOC_NOPM, 0, 0),
+SND_SOC_DAPM_DAC("DACR", NULL, SND_SOC_NOPM, 0, 0),
+
+SND_SOC_DAPM_OUTPUT("OUTL"),
+SND_SOC_DAPM_OUTPUT("OUTR"),
+};
+
+static const struct snd_soc_dapm_route pcm512x_dapm_routes[] = {
+ { "DACL", NULL, "Playback" },
+ { "DACR", NULL, "Playback" },
+
+ { "OUTL", NULL, "DACL" },
+ { "OUTR", NULL, "DACR" },
+};
+
+static int pcm512x_set_bias_level(struct snd_soc_codec *codec,
+ enum snd_soc_bias_level level)
+{
+ struct pcm512x_priv *pcm512x = dev_get_drvdata(codec->dev);
+ int ret;
+
+ switch (level) {
+ case SND_SOC_BIAS_ON:
+ case SND_SOC_BIAS_PREPARE:
+ break;
+
+ case SND_SOC_BIAS_STANDBY:
+ ret = regmap_update_bits(pcm512x->regmap, PCM512x_POWER,
+ PCM512x_RQST, 0);
+ if (ret != 0) {
+ dev_err(codec->dev, "Failed to remove standby: %d\n",
+ ret);
+ return ret;
+ }
+ break;
+
+ case SND_SOC_BIAS_OFF:
+ ret = regmap_update_bits(pcm512x->regmap, PCM512x_POWER,
+ PCM512x_RQST, PCM512x_RQST);
+ if (ret != 0) {
+ dev_err(codec->dev, "Failed to request standby: %d\n",
+ ret);
+ return ret;
+ }
+ break;
+ }
+
+ codec->dapm.bias_level = level;
+
+ return 0;
+}
+
+static struct snd_soc_dai_driver pcm512x_dai = {
+ .name = "pcm512x-hifi",
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_8000_192000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S24_LE |
+ SNDRV_PCM_FMTBIT_S32_LE
+ },
+};
+
+static struct snd_soc_codec_driver pcm512x_codec_driver = {
+ .set_bias_level = pcm512x_set_bias_level,
+ .idle_bias_off = true,
+
+ .controls = pcm512x_controls,
+ .num_controls = ARRAY_SIZE(pcm512x_controls),
+ .dapm_widgets = pcm512x_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(pcm512x_dapm_widgets),
+ .dapm_routes = pcm512x_dapm_routes,
+ .num_dapm_routes = ARRAY_SIZE(pcm512x_dapm_routes),
+};
+
+static const struct regmap_range_cfg pcm512x_range = {
+ .name = "Pages", .range_min = PCM512x_VIRT_BASE,
+ .range_max = PCM512x_MAX_REGISTER,
+ .selector_reg = PCM512x_PAGE,
+ .selector_mask = 0xff,
+ .window_start = 0, .window_len = 0x100,
+};
+
+const struct regmap_config pcm512x_regmap = {
+ .reg_bits = 8,
+ .val_bits = 8,
+
+ .readable_reg = pcm512x_readable,
+ .volatile_reg = pcm512x_volatile,
+
+ .ranges = &pcm512x_range,
+ .num_ranges = 1,
+
+ .max_register = PCM512x_MAX_REGISTER,
+ .reg_defaults = pcm512x_reg_defaults,
+ .num_reg_defaults = ARRAY_SIZE(pcm512x_reg_defaults),
+ .cache_type = REGCACHE_RBTREE,
+};
+EXPORT_SYMBOL_GPL(pcm512x_regmap);
+
+int pcm512x_probe(struct device *dev, struct regmap *regmap)
+{
+ struct pcm512x_priv *pcm512x;
+ int i, ret;
+
+ pcm512x = devm_kzalloc(dev, sizeof(struct pcm512x_priv), GFP_KERNEL);
+ if (!pcm512x)
+ return -ENOMEM;
+
+ dev_set_drvdata(dev, pcm512x);
+ pcm512x->regmap = regmap;
+
+ for (i = 0; i < ARRAY_SIZE(pcm512x->supplies); i++)
+ pcm512x->supplies[i].supply = pcm512x_supply_names[i];
+
+ ret = devm_regulator_bulk_get(dev, ARRAY_SIZE(pcm512x->supplies),
+ pcm512x->supplies);
+ if (ret != 0) {
+ dev_err(dev, "Failed to get supplies: %d\n", ret);
+ return ret;
+ }
+
+ pcm512x->supply_nb[0].notifier_call = pcm512x_regulator_event_0;
+ pcm512x->supply_nb[1].notifier_call = pcm512x_regulator_event_1;
+ pcm512x->supply_nb[2].notifier_call = pcm512x_regulator_event_2;
+
+ for (i = 0; i < ARRAY_SIZE(pcm512x->supplies); i++) {
+ ret = regulator_register_notifier(pcm512x->supplies[i].consumer,
+ &pcm512x->supply_nb[i]);
+ if (ret != 0) {
+ dev_err(dev,
+ "Failed to register regulator notifier: %d\n",
+ ret);
+ }
+ }
+
+ ret = regulator_bulk_enable(ARRAY_SIZE(pcm512x->supplies),
+ pcm512x->supplies);
+ if (ret != 0) {
+ dev_err(dev, "Failed to enable supplies: %d\n", ret);
+ return ret;
+ }
+
+ /* Reset the device, verifying I/O in the process for I2C */
+ ret = regmap_write(regmap, PCM512x_RESET,
+ PCM512x_RSTM | PCM512x_RSTR);
+ if (ret != 0) {
+ dev_err(dev, "Failed to reset device: %d\n", ret);
+ goto err;
+ }
+
+ ret = regmap_write(regmap, PCM512x_RESET, 0);
+ if (ret != 0) {
+ dev_err(dev, "Failed to reset device: %d\n", ret);
+ goto err;
+ }
+
+ pcm512x->sclk = devm_clk_get(dev, NULL);
+ if (IS_ERR(pcm512x->sclk)) {
+ if (PTR_ERR(pcm512x->sclk) == -EPROBE_DEFER)
+ return -EPROBE_DEFER;
+
+ dev_info(dev, "No SCLK, using BCLK: %ld\n",
+ PTR_ERR(pcm512x->sclk));
+
+ /* Disable reporting of missing SCLK as an error */
+ regmap_update_bits(regmap, PCM512x_ERROR_DETECT,
+ PCM512x_IDCH, PCM512x_IDCH);
+
+ /* Switch PLL input to BCLK */
+ regmap_update_bits(regmap, PCM512x_PLL_REF,
+ PCM512x_SREF, PCM512x_SREF);
+ } else {
+ ret = clk_prepare_enable(pcm512x->sclk);
+ if (ret != 0) {
+ dev_err(dev, "Failed to enable SCLK: %d\n", ret);
+ return ret;
+ }
+ }
+
+ /* Default to standby mode */
+ ret = regmap_update_bits(pcm512x->regmap, PCM512x_POWER,
+ PCM512x_RQST, PCM512x_RQST);
+ if (ret != 0) {
+ dev_err(dev, "Failed to request standby: %d\n",
+ ret);
+ goto err_clk;
+ }
+
+ pm_runtime_set_active(dev);
+ pm_runtime_enable(dev);
+ pm_runtime_idle(dev);
+
+ ret = snd_soc_register_codec(dev, &pcm512x_codec_driver,
+ &pcm512x_dai, 1);
+ if (ret != 0) {
+ dev_err(dev, "Failed to register CODEC: %d\n", ret);
+ goto err_pm;
+ }
+
+ return 0;
+
+err_pm:
+ pm_runtime_disable(dev);
+err_clk:
+ if (!IS_ERR(pcm512x->sclk))
+ clk_disable_unprepare(pcm512x->sclk);
+err:
+ regulator_bulk_disable(ARRAY_SIZE(pcm512x->supplies),
+ pcm512x->supplies);
+ return ret;
+}
+EXPORT_SYMBOL_GPL(pcm512x_probe);
+
+void pcm512x_remove(struct device *dev)
+{
+ struct pcm512x_priv *pcm512x = dev_get_drvdata(dev);
+
+ snd_soc_unregister_codec(dev);
+ pm_runtime_disable(dev);
+ if (!IS_ERR(pcm512x->sclk))
+ clk_disable_unprepare(pcm512x->sclk);
+ regulator_bulk_disable(ARRAY_SIZE(pcm512x->supplies),
+ pcm512x->supplies);
+}
+EXPORT_SYMBOL_GPL(pcm512x_remove);
+
+static int pcm512x_suspend(struct device *dev)
+{
+ struct pcm512x_priv *pcm512x = dev_get_drvdata(dev);
+ int ret;
+
+ ret = regmap_update_bits(pcm512x->regmap, PCM512x_POWER,
+ PCM512x_RQPD, PCM512x_RQPD);
+ if (ret != 0) {
+ dev_err(dev, "Failed to request power down: %d\n", ret);
+ return ret;
+ }
+
+ ret = regulator_bulk_disable(ARRAY_SIZE(pcm512x->supplies),
+ pcm512x->supplies);
+ if (ret != 0) {
+ dev_err(dev, "Failed to disable supplies: %d\n", ret);
+ return ret;
+ }
+
+ if (!IS_ERR(pcm512x->sclk))
+ clk_disable_unprepare(pcm512x->sclk);
+
+ return 0;
+}
+
+static int pcm512x_resume(struct device *dev)
+{
+ struct pcm512x_priv *pcm512x = dev_get_drvdata(dev);
+ int ret;
+
+ if (!IS_ERR(pcm512x->sclk)) {
+ ret = clk_prepare_enable(pcm512x->sclk);
+ if (ret != 0) {
+ dev_err(dev, "Failed to enable SCLK: %d\n", ret);
+ return ret;
+ }
+ }
+
+ ret = regulator_bulk_enable(ARRAY_SIZE(pcm512x->supplies),
+ pcm512x->supplies);
+ if (ret != 0) {
+ dev_err(dev, "Failed to enable supplies: %d\n", ret);
+ return ret;
+ }
+
+ regcache_cache_only(pcm512x->regmap, false);
+ ret = regcache_sync(pcm512x->regmap);
+ if (ret != 0) {
+ dev_err(dev, "Failed to sync cache: %d\n", ret);
+ return ret;
+ }
+
+ ret = regmap_update_bits(pcm512x->regmap, PCM512x_POWER,
+ PCM512x_RQPD, 0);
+ if (ret != 0) {
+ dev_err(dev, "Failed to remove power down: %d\n", ret);
+ return ret;
+ }
+
+ return 0;
+}
+
+const struct dev_pm_ops pcm512x_pm_ops = {
+ SET_RUNTIME_PM_OPS(pcm512x_suspend, pcm512x_resume, NULL)
+};
+EXPORT_SYMBOL_GPL(pcm512x_pm_ops);
+
+MODULE_DESCRIPTION("ASoC PCM512x codec driver");
+MODULE_AUTHOR("Mark Brown <broonie@linaro.org>");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/codecs/pcm512x.h b/sound/soc/codecs/pcm512x.h
new file mode 100644
index 0000000..6ee76aa
--- /dev/null
+++ b/sound/soc/codecs/pcm512x.h
@@ -0,0 +1,171 @@
+/*
+ * Driver for the PCM512x CODECs
+ *
+ * Author: Mark Brown <broonie@linaro.org>
+ * Copyright 2014 Linaro Ltd
+ *
+ * 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.
+ */
+
+#ifndef _SND_SOC_PCM512X
+#define _SND_SOC_PCM512X
+
+#include <linux/pm.h>
+#include <linux/regmap.h>
+
+#define PCM512x_VIRT_BASE 0x100
+#define PCM512x_PAGE_LEN 0x100
+#define PCM512x_PAGE_BASE(n) (PCM512x_VIRT_BASE + (PCM512x_PAGE_LEN * n))
+
+#define PCM512x_PAGE 0
+
+#define PCM512x_RESET (PCM512x_PAGE_BASE(0) + 1)
+#define PCM512x_POWER (PCM512x_PAGE_BASE(0) + 2)
+#define PCM512x_MUTE (PCM512x_PAGE_BASE(0) + 3)
+#define PCM512x_PLL_EN (PCM512x_PAGE_BASE(0) + 4)
+#define PCM512x_SPI_MISO_FUNCTION (PCM512x_PAGE_BASE(0) + 6)
+#define PCM512x_DSP (PCM512x_PAGE_BASE(0) + 7)
+#define PCM512x_GPIO_EN (PCM512x_PAGE_BASE(0) + 8)
+#define PCM512x_BCLK_LRCLK_CFG (PCM512x_PAGE_BASE(0) + 9)
+#define PCM512x_DSP_GPIO_INPUT (PCM512x_PAGE_BASE(0) + 10)
+#define PCM512x_MASTER_MODE (PCM512x_PAGE_BASE(0) + 12)
+#define PCM512x_PLL_REF (PCM512x_PAGE_BASE(0) + 13)
+#define PCM512x_PLL_COEFF_0 (PCM512x_PAGE_BASE(0) + 20)
+#define PCM512x_PLL_COEFF_1 (PCM512x_PAGE_BASE(0) + 21)
+#define PCM512x_PLL_COEFF_2 (PCM512x_PAGE_BASE(0) + 22)
+#define PCM512x_PLL_COEFF_3 (PCM512x_PAGE_BASE(0) + 23)
+#define PCM512x_PLL_COEFF_4 (PCM512x_PAGE_BASE(0) + 24)
+#define PCM512x_DSP_CLKDIV (PCM512x_PAGE_BASE(0) + 27)
+#define PCM512x_DAC_CLKDIV (PCM512x_PAGE_BASE(0) + 28)
+#define PCM512x_NCP_CLKDIV (PCM512x_PAGE_BASE(0) + 29)
+#define PCM512x_OSR_CLKDIV (PCM512x_PAGE_BASE(0) + 30)
+#define PCM512x_MASTER_CLKDIV_1 (PCM512x_PAGE_BASE(0) + 32)
+#define PCM512x_MASTER_CLKDIV_2 (PCM512x_PAGE_BASE(0) + 33)
+#define PCM512x_FS_SPEED_MODE (PCM512x_PAGE_BASE(0) + 34)
+#define PCM512x_IDAC_1 (PCM512x_PAGE_BASE(0) + 35)
+#define PCM512x_IDAC_2 (PCM512x_PAGE_BASE(0) + 36)
+#define PCM512x_ERROR_DETECT (PCM512x_PAGE_BASE(0) + 37)
+#define PCM512x_I2S_1 (PCM512x_PAGE_BASE(0) + 40)
+#define PCM512x_I2S_2 (PCM512x_PAGE_BASE(0) + 41)
+#define PCM512x_DAC_ROUTING (PCM512x_PAGE_BASE(0) + 42)
+#define PCM512x_DSP_PROGRAM (PCM512x_PAGE_BASE(0) + 43)
+#define PCM512x_CLKDET (PCM512x_PAGE_BASE(0) + 44)
+#define PCM512x_AUTO_MUTE (PCM512x_PAGE_BASE(0) + 59)
+#define PCM512x_DIGITAL_VOLUME_1 (PCM512x_PAGE_BASE(0) + 60)
+#define PCM512x_DIGITAL_VOLUME_2 (PCM512x_PAGE_BASE(0) + 61)
+#define PCM512x_DIGITAL_VOLUME_3 (PCM512x_PAGE_BASE(0) + 62)
+#define PCM512x_DIGITAL_MUTE_1 (PCM512x_PAGE_BASE(0) + 63)
+#define PCM512x_DIGITAL_MUTE_2 (PCM512x_PAGE_BASE(0) + 64)
+#define PCM512x_DIGITAL_MUTE_3 (PCM512x_PAGE_BASE(0) + 65)
+#define PCM512x_GPIO_OUTPUT_1 (PCM512x_PAGE_BASE(0) + 80)
+#define PCM512x_GPIO_OUTPUT_2 (PCM512x_PAGE_BASE(0) + 81)
+#define PCM512x_GPIO_OUTPUT_3 (PCM512x_PAGE_BASE(0) + 82)
+#define PCM512x_GPIO_OUTPUT_4 (PCM512x_PAGE_BASE(0) + 83)
+#define PCM512x_GPIO_OUTPUT_5 (PCM512x_PAGE_BASE(0) + 84)
+#define PCM512x_GPIO_OUTPUT_6 (PCM512x_PAGE_BASE(0) + 85)
+#define PCM512x_GPIO_CONTROL_1 (PCM512x_PAGE_BASE(0) + 86)
+#define PCM512x_GPIO_CONTROL_2 (PCM512x_PAGE_BASE(0) + 87)
+#define PCM512x_OVERFLOW (PCM512x_PAGE_BASE(0) + 90)
+#define PCM512x_RATE_DET_1 (PCM512x_PAGE_BASE(0) + 91)
+#define PCM512x_RATE_DET_2 (PCM512x_PAGE_BASE(0) + 92)
+#define PCM512x_RATE_DET_3 (PCM512x_PAGE_BASE(0) + 93)
+#define PCM512x_RATE_DET_4 (PCM512x_PAGE_BASE(0) + 94)
+#define PCM512x_ANALOG_MUTE_DET (PCM512x_PAGE_BASE(0) + 108)
+#define PCM512x_GPIN (PCM512x_PAGE_BASE(0) + 119)
+#define PCM512x_DIGITAL_MUTE_DET (PCM512x_PAGE_BASE(0) + 120)
+
+#define PCM512x_OUTPUT_AMPLITUDE (PCM512x_PAGE_BASE(1) + 1)
+#define PCM512x_ANALOG_GAIN_CTRL (PCM512x_PAGE_BASE(1) + 2)
+#define PCM512x_UNDERVOLTAGE_PROT (PCM512x_PAGE_BASE(1) + 5)
+#define PCM512x_ANALOG_MUTE_CTRL (PCM512x_PAGE_BASE(1) + 6)
+#define PCM512x_ANALOG_GAIN_BOOST (PCM512x_PAGE_BASE(1) + 7)
+#define PCM512x_VCOM_CTRL_1 (PCM512x_PAGE_BASE(1) + 8)
+#define PCM512x_VCOM_CTRL_2 (PCM512x_PAGE_BASE(1) + 9)
+
+#define PCM512x_CRAM_CTRL (PCM512x_PAGE_BASE(44) + 1)
+
+#define PCM512x_MAX_REGISTER (PCM512x_PAGE_BASE(44) + 1)
+
+/* Page 0, Register 1 - reset */
+#define PCM512x_RSTR (1 << 0)
+#define PCM512x_RSTM (1 << 4)
+
+/* Page 0, Register 2 - power */
+#define PCM512x_RQPD (1 << 0)
+#define PCM512x_RQPD_SHIFT 0
+#define PCM512x_RQST (1 << 4)
+#define PCM512x_RQST_SHIFT 4
+
+/* Page 0, Register 3 - mute */
+#define PCM512x_RQMR_SHIFT 0
+#define PCM512x_RQML_SHIFT 4
+
+/* Page 0, Register 4 - PLL */
+#define PCM512x_PLCE (1 << 0)
+#define PCM512x_RLCE_SHIFT 0
+#define PCM512x_PLCK (1 << 4)
+#define PCM512x_PLCK_SHIFT 4
+
+/* Page 0, Register 7 - DSP */
+#define PCM512x_SDSL (1 << 0)
+#define PCM512x_SDSL_SHIFT 0
+#define PCM512x_DEMP (1 << 4)
+#define PCM512x_DEMP_SHIFT 4
+
+/* Page 0, Register 13 - PLL reference */
+#define PCM512x_SREF (1 << 4)
+
+/* Page 0, Register 37 - Error detection */
+#define PCM512x_IPLK (1 << 0)
+#define PCM512x_DCAS (1 << 1)
+#define PCM512x_IDCM (1 << 2)
+#define PCM512x_IDCH (1 << 3)
+#define PCM512x_IDSK (1 << 4)
+#define PCM512x_IDBK (1 << 5)
+#define PCM512x_IDFS (1 << 6)
+
+/* Page 0, Register 42 - DAC routing */
+#define PCM512x_AUPR_SHIFT 0
+#define PCM512x_AUPL_SHIFT 4
+
+/* Page 0, Register 59 - auto mute */
+#define PCM512x_ATMR_SHIFT 0
+#define PCM512x_ATML_SHIFT 4
+
+/* Page 0, Register 63 - ramp rates */
+#define PCM512x_VNDF_SHIFT 6
+#define PCM512x_VNDS_SHIFT 4
+#define PCM512x_VNUF_SHIFT 2
+#define PCM512x_VNUS_SHIFT 0
+
+/* Page 0, Register 64 - emergency ramp rates */
+#define PCM512x_VEDF_SHIFT 6
+#define PCM512x_VEDS_SHIFT 4
+
+/* Page 0, Register 65 - Digital mute enables */
+#define PCM512x_ACTL_SHIFT 2
+#define PCM512x_AMLE_SHIFT 1
+#define PCM512x_AMLR_SHIFT 0
+
+/* Page 1, Register 2 - analog volume control */
+#define PCM512x_RAGN_SHIFT 0
+#define PCM512x_LAGN_SHIFT 4
+
+/* Page 1, Register 7 - analog boost control */
+#define PCM512x_AGBR_SHIFT 0
+#define PCM512x_AGBL_SHIFT 4
+
+extern const struct dev_pm_ops pcm512x_pm_ops;
+extern const struct regmap_config pcm512x_regmap;
+
+int pcm512x_probe(struct device *dev, struct regmap *regmap);
+void pcm512x_remove(struct device *dev);
+
+#endif
diff --git a/sound/soc/codecs/rt5631.c b/sound/soc/codecs/rt5631.c
index 912c9cb..d4c229f 100644
--- a/sound/soc/codecs/rt5631.c
+++ b/sound/soc/codecs/rt5631.c
@@ -210,26 +210,22 @@ static int rt5631_dmic_put(struct snd_kcontrol *kcontrol,
static const char *rt5631_input_mode[] = {
"Single ended", "Differential"};
-static const SOC_ENUM_SINGLE_DECL(
- rt5631_mic1_mode_enum, RT5631_MIC_CTRL_1,
- RT5631_MIC1_DIFF_INPUT_SHIFT, rt5631_input_mode);
+static SOC_ENUM_SINGLE_DECL(rt5631_mic1_mode_enum, RT5631_MIC_CTRL_1,
+ RT5631_MIC1_DIFF_INPUT_SHIFT, rt5631_input_mode);
-static const SOC_ENUM_SINGLE_DECL(
- rt5631_mic2_mode_enum, RT5631_MIC_CTRL_1,
- RT5631_MIC2_DIFF_INPUT_SHIFT, rt5631_input_mode);
+static SOC_ENUM_SINGLE_DECL(rt5631_mic2_mode_enum, RT5631_MIC_CTRL_1,
+ RT5631_MIC2_DIFF_INPUT_SHIFT, rt5631_input_mode);
/* MONO Input Type */
-static const SOC_ENUM_SINGLE_DECL(
- rt5631_monoin_mode_enum, RT5631_MONO_INPUT_VOL,
- RT5631_MONO_DIFF_INPUT_SHIFT, rt5631_input_mode);
+static SOC_ENUM_SINGLE_DECL(rt5631_monoin_mode_enum, RT5631_MONO_INPUT_VOL,
+ RT5631_MONO_DIFF_INPUT_SHIFT, rt5631_input_mode);
/* SPK Ratio Gain Control */
static const char *rt5631_spk_ratio[] = {"1.00x", "1.09x", "1.27x", "1.44x",
"1.56x", "1.68x", "1.99x", "2.34x"};
-static const SOC_ENUM_SINGLE_DECL(
- rt5631_spk_ratio_enum, RT5631_GEN_PUR_CTRL_REG,
- RT5631_SPK_AMP_RATIO_CTRL_SHIFT, rt5631_spk_ratio);
+static SOC_ENUM_SINGLE_DECL(rt5631_spk_ratio_enum, RT5631_GEN_PUR_CTRL_REG,
+ RT5631_SPK_AMP_RATIO_CTRL_SHIFT, rt5631_spk_ratio);
static const struct snd_kcontrol_new rt5631_snd_controls[] = {
/* MIC */
@@ -759,9 +755,8 @@ static const struct snd_kcontrol_new rt5631_monomix_mixer_controls[] = {
/* Left SPK Volume Input */
static const char *rt5631_spkvoll_sel[] = {"Vmid", "SPKMIXL"};
-static const SOC_ENUM_SINGLE_DECL(
- rt5631_spkvoll_enum, RT5631_SPK_OUT_VOL,
- RT5631_L_EN_SHIFT, rt5631_spkvoll_sel);
+static SOC_ENUM_SINGLE_DECL(rt5631_spkvoll_enum, RT5631_SPK_OUT_VOL,
+ RT5631_L_EN_SHIFT, rt5631_spkvoll_sel);
static const struct snd_kcontrol_new rt5631_spkvoll_mux_control =
SOC_DAPM_ENUM("Left SPKVOL SRC", rt5631_spkvoll_enum);
@@ -769,9 +764,8 @@ static const struct snd_kcontrol_new rt5631_spkvoll_mux_control =
/* Left HP Volume Input */
static const char *rt5631_hpvoll_sel[] = {"Vmid", "OUTMIXL"};
-static const SOC_ENUM_SINGLE_DECL(
- rt5631_hpvoll_enum, RT5631_HP_OUT_VOL,
- RT5631_L_EN_SHIFT, rt5631_hpvoll_sel);
+static SOC_ENUM_SINGLE_DECL(rt5631_hpvoll_enum, RT5631_HP_OUT_VOL,
+ RT5631_L_EN_SHIFT, rt5631_hpvoll_sel);
static const struct snd_kcontrol_new rt5631_hpvoll_mux_control =
SOC_DAPM_ENUM("Left HPVOL SRC", rt5631_hpvoll_enum);
@@ -779,9 +773,8 @@ static const struct snd_kcontrol_new rt5631_hpvoll_mux_control =
/* Left Out Volume Input */
static const char *rt5631_outvoll_sel[] = {"Vmid", "OUTMIXL"};
-static const SOC_ENUM_SINGLE_DECL(
- rt5631_outvoll_enum, RT5631_MONO_AXO_1_2_VOL,
- RT5631_L_EN_SHIFT, rt5631_outvoll_sel);
+static SOC_ENUM_SINGLE_DECL(rt5631_outvoll_enum, RT5631_MONO_AXO_1_2_VOL,
+ RT5631_L_EN_SHIFT, rt5631_outvoll_sel);
static const struct snd_kcontrol_new rt5631_outvoll_mux_control =
SOC_DAPM_ENUM("Left OUTVOL SRC", rt5631_outvoll_enum);
@@ -789,9 +782,8 @@ static const struct snd_kcontrol_new rt5631_outvoll_mux_control =
/* Right Out Volume Input */
static const char *rt5631_outvolr_sel[] = {"Vmid", "OUTMIXR"};
-static const SOC_ENUM_SINGLE_DECL(
- rt5631_outvolr_enum, RT5631_MONO_AXO_1_2_VOL,
- RT5631_R_EN_SHIFT, rt5631_outvolr_sel);
+static SOC_ENUM_SINGLE_DECL(rt5631_outvolr_enum, RT5631_MONO_AXO_1_2_VOL,
+ RT5631_R_EN_SHIFT, rt5631_outvolr_sel);
static const struct snd_kcontrol_new rt5631_outvolr_mux_control =
SOC_DAPM_ENUM("Right OUTVOL SRC", rt5631_outvolr_enum);
@@ -799,9 +791,8 @@ static const struct snd_kcontrol_new rt5631_outvolr_mux_control =
/* Right HP Volume Input */
static const char *rt5631_hpvolr_sel[] = {"Vmid", "OUTMIXR"};
-static const SOC_ENUM_SINGLE_DECL(
- rt5631_hpvolr_enum, RT5631_HP_OUT_VOL,
- RT5631_R_EN_SHIFT, rt5631_hpvolr_sel);
+static SOC_ENUM_SINGLE_DECL(rt5631_hpvolr_enum, RT5631_HP_OUT_VOL,
+ RT5631_R_EN_SHIFT, rt5631_hpvolr_sel);
static const struct snd_kcontrol_new rt5631_hpvolr_mux_control =
SOC_DAPM_ENUM("Right HPVOL SRC", rt5631_hpvolr_enum);
@@ -809,9 +800,8 @@ static const struct snd_kcontrol_new rt5631_hpvolr_mux_control =
/* Right SPK Volume Input */
static const char *rt5631_spkvolr_sel[] = {"Vmid", "SPKMIXR"};
-static const SOC_ENUM_SINGLE_DECL(
- rt5631_spkvolr_enum, RT5631_SPK_OUT_VOL,
- RT5631_R_EN_SHIFT, rt5631_spkvolr_sel);
+static SOC_ENUM_SINGLE_DECL(rt5631_spkvolr_enum, RT5631_SPK_OUT_VOL,
+ RT5631_R_EN_SHIFT, rt5631_spkvolr_sel);
static const struct snd_kcontrol_new rt5631_spkvolr_mux_control =
SOC_DAPM_ENUM("Right SPKVOL SRC", rt5631_spkvolr_enum);
@@ -820,9 +810,8 @@ static const struct snd_kcontrol_new rt5631_spkvolr_mux_control =
static const char *rt5631_spol_src_sel[] = {
"SPOLMIX", "MONOIN_RX", "VDAC", "DACL"};
-static const SOC_ENUM_SINGLE_DECL(
- rt5631_spol_src_enum, RT5631_SPK_MONO_HP_OUT_CTRL,
- RT5631_SPK_L_MUX_SEL_SHIFT, rt5631_spol_src_sel);
+static SOC_ENUM_SINGLE_DECL(rt5631_spol_src_enum, RT5631_SPK_MONO_HP_OUT_CTRL,
+ RT5631_SPK_L_MUX_SEL_SHIFT, rt5631_spol_src_sel);
static const struct snd_kcontrol_new rt5631_spol_mux_control =
SOC_DAPM_ENUM("SPOL SRC", rt5631_spol_src_enum);
@@ -831,9 +820,8 @@ static const struct snd_kcontrol_new rt5631_spol_mux_control =
static const char *rt5631_spor_src_sel[] = {
"SPORMIX", "MONOIN_RX", "VDAC", "DACR"};
-static const SOC_ENUM_SINGLE_DECL(
- rt5631_spor_src_enum, RT5631_SPK_MONO_HP_OUT_CTRL,
- RT5631_SPK_R_MUX_SEL_SHIFT, rt5631_spor_src_sel);
+static SOC_ENUM_SINGLE_DECL(rt5631_spor_src_enum, RT5631_SPK_MONO_HP_OUT_CTRL,
+ RT5631_SPK_R_MUX_SEL_SHIFT, rt5631_spor_src_sel);
static const struct snd_kcontrol_new rt5631_spor_mux_control =
SOC_DAPM_ENUM("SPOR SRC", rt5631_spor_src_enum);
@@ -841,9 +829,8 @@ static const struct snd_kcontrol_new rt5631_spor_mux_control =
/* MONO Input */
static const char *rt5631_mono_src_sel[] = {"MONOMIX", "MONOIN_RX", "VDAC"};
-static const SOC_ENUM_SINGLE_DECL(
- rt5631_mono_src_enum, RT5631_SPK_MONO_HP_OUT_CTRL,
- RT5631_MONO_MUX_SEL_SHIFT, rt5631_mono_src_sel);
+static SOC_ENUM_SINGLE_DECL(rt5631_mono_src_enum, RT5631_SPK_MONO_HP_OUT_CTRL,
+ RT5631_MONO_MUX_SEL_SHIFT, rt5631_mono_src_sel);
static const struct snd_kcontrol_new rt5631_mono_mux_control =
SOC_DAPM_ENUM("MONO SRC", rt5631_mono_src_enum);
@@ -851,9 +838,8 @@ static const struct snd_kcontrol_new rt5631_mono_mux_control =
/* Left HPO Input */
static const char *rt5631_hpl_src_sel[] = {"Left HPVOL", "Left DAC"};
-static const SOC_ENUM_SINGLE_DECL(
- rt5631_hpl_src_enum, RT5631_SPK_MONO_HP_OUT_CTRL,
- RT5631_HP_L_MUX_SEL_SHIFT, rt5631_hpl_src_sel);
+static SOC_ENUM_SINGLE_DECL(rt5631_hpl_src_enum, RT5631_SPK_MONO_HP_OUT_CTRL,
+ RT5631_HP_L_MUX_SEL_SHIFT, rt5631_hpl_src_sel);
static const struct snd_kcontrol_new rt5631_hpl_mux_control =
SOC_DAPM_ENUM("HPL SRC", rt5631_hpl_src_enum);
@@ -861,9 +847,8 @@ static const struct snd_kcontrol_new rt5631_hpl_mux_control =
/* Right HPO Input */
static const char *rt5631_hpr_src_sel[] = {"Right HPVOL", "Right DAC"};
-static const SOC_ENUM_SINGLE_DECL(
- rt5631_hpr_src_enum, RT5631_SPK_MONO_HP_OUT_CTRL,
- RT5631_HP_R_MUX_SEL_SHIFT, rt5631_hpr_src_sel);
+static SOC_ENUM_SINGLE_DECL(rt5631_hpr_src_enum, RT5631_SPK_MONO_HP_OUT_CTRL,
+ RT5631_HP_R_MUX_SEL_SHIFT, rt5631_hpr_src_sel);
static const struct snd_kcontrol_new rt5631_hpr_mux_control =
SOC_DAPM_ENUM("HPR SRC", rt5631_hpr_src_enum);
@@ -1585,15 +1570,6 @@ static int rt5631_probe(struct snd_soc_codec *codec)
{
struct rt5631_priv *rt5631 = snd_soc_codec_get_drvdata(codec);
unsigned int val;
- int ret;
-
- 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;
- }
val = rt5631_read_index(codec, RT5631_ADDA_MIXER_INTL_REG3);
if (val & 0x0002)
diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c
index 8869249..0061ae6 100644
--- a/sound/soc/codecs/rt5640.c
+++ b/sound/soc/codecs/rt5640.c
@@ -361,25 +361,24 @@ static unsigned int bst_tlv[] = {
static const char * const rt5640_data_select[] = {
"Normal", "left copy to right", "right copy to left", "Swap"};
-static const SOC_ENUM_SINGLE_DECL(rt5640_if1_dac_enum, RT5640_DIG_INF_DATA,
- RT5640_IF1_DAC_SEL_SFT, rt5640_data_select);
+static SOC_ENUM_SINGLE_DECL(rt5640_if1_dac_enum, RT5640_DIG_INF_DATA,
+ RT5640_IF1_DAC_SEL_SFT, rt5640_data_select);
-static const SOC_ENUM_SINGLE_DECL(rt5640_if1_adc_enum, RT5640_DIG_INF_DATA,
- RT5640_IF1_ADC_SEL_SFT, rt5640_data_select);
+static SOC_ENUM_SINGLE_DECL(rt5640_if1_adc_enum, RT5640_DIG_INF_DATA,
+ RT5640_IF1_ADC_SEL_SFT, rt5640_data_select);
-static const SOC_ENUM_SINGLE_DECL(rt5640_if2_dac_enum, RT5640_DIG_INF_DATA,
- RT5640_IF2_DAC_SEL_SFT, rt5640_data_select);
+static SOC_ENUM_SINGLE_DECL(rt5640_if2_dac_enum, RT5640_DIG_INF_DATA,
+ RT5640_IF2_DAC_SEL_SFT, rt5640_data_select);
-static const SOC_ENUM_SINGLE_DECL(rt5640_if2_adc_enum, RT5640_DIG_INF_DATA,
- RT5640_IF2_ADC_SEL_SFT, rt5640_data_select);
+static SOC_ENUM_SINGLE_DECL(rt5640_if2_adc_enum, RT5640_DIG_INF_DATA,
+ RT5640_IF2_ADC_SEL_SFT, rt5640_data_select);
/* Class D speaker gain ratio */
static const char * const rt5640_clsd_spk_ratio[] = {"1.66x", "1.83x", "1.94x",
"2x", "2.11x", "2.22x", "2.33x", "2.44x", "2.55x", "2.66x", "2.77x"};
-static const SOC_ENUM_SINGLE_DECL(
- rt5640_clsd_spk_ratio_enum, RT5640_CLS_D_OUT,
- RT5640_CLSD_RATIO_SFT, rt5640_clsd_spk_ratio);
+static SOC_ENUM_SINGLE_DECL(rt5640_clsd_spk_ratio_enum, RT5640_CLS_D_OUT,
+ RT5640_CLSD_RATIO_SFT, rt5640_clsd_spk_ratio);
static const struct snd_kcontrol_new rt5640_snd_controls[] = {
/* Speaker Output Volume */
@@ -753,9 +752,8 @@ static const char * const rt5640_stereo_adc1_src[] = {
"DIG MIX", "ADC"
};
-static const SOC_ENUM_SINGLE_DECL(
- rt5640_stereo_adc1_enum, RT5640_STO_ADC_MIXER,
- RT5640_ADC_1_SRC_SFT, rt5640_stereo_adc1_src);
+static SOC_ENUM_SINGLE_DECL(rt5640_stereo_adc1_enum, RT5640_STO_ADC_MIXER,
+ RT5640_ADC_1_SRC_SFT, rt5640_stereo_adc1_src);
static const struct snd_kcontrol_new rt5640_sto_adc_1_mux =
SOC_DAPM_ENUM("Stereo ADC1 Mux", rt5640_stereo_adc1_enum);
@@ -764,9 +762,8 @@ static const char * const rt5640_stereo_adc2_src[] = {
"DMIC1", "DMIC2", "DIG MIX"
};
-static const SOC_ENUM_SINGLE_DECL(
- rt5640_stereo_adc2_enum, RT5640_STO_ADC_MIXER,
- RT5640_ADC_2_SRC_SFT, rt5640_stereo_adc2_src);
+static SOC_ENUM_SINGLE_DECL(rt5640_stereo_adc2_enum, RT5640_STO_ADC_MIXER,
+ RT5640_ADC_2_SRC_SFT, rt5640_stereo_adc2_src);
static const struct snd_kcontrol_new rt5640_sto_adc_2_mux =
SOC_DAPM_ENUM("Stereo ADC2 Mux", rt5640_stereo_adc2_enum);
@@ -776,9 +773,8 @@ static const char * const rt5640_mono_adc_l1_src[] = {
"Mono DAC MIXL", "ADCL"
};
-static const SOC_ENUM_SINGLE_DECL(
- rt5640_mono_adc_l1_enum, RT5640_MONO_ADC_MIXER,
- RT5640_MONO_ADC_L1_SRC_SFT, rt5640_mono_adc_l1_src);
+static SOC_ENUM_SINGLE_DECL(rt5640_mono_adc_l1_enum, RT5640_MONO_ADC_MIXER,
+ RT5640_MONO_ADC_L1_SRC_SFT, rt5640_mono_adc_l1_src);
static const struct snd_kcontrol_new rt5640_mono_adc_l1_mux =
SOC_DAPM_ENUM("Mono ADC1 left source", rt5640_mono_adc_l1_enum);
@@ -787,9 +783,8 @@ static const char * const rt5640_mono_adc_l2_src[] = {
"DMIC L1", "DMIC L2", "Mono DAC MIXL"
};
-static const SOC_ENUM_SINGLE_DECL(
- rt5640_mono_adc_l2_enum, RT5640_MONO_ADC_MIXER,
- RT5640_MONO_ADC_L2_SRC_SFT, rt5640_mono_adc_l2_src);
+static SOC_ENUM_SINGLE_DECL(rt5640_mono_adc_l2_enum, RT5640_MONO_ADC_MIXER,
+ RT5640_MONO_ADC_L2_SRC_SFT, rt5640_mono_adc_l2_src);
static const struct snd_kcontrol_new rt5640_mono_adc_l2_mux =
SOC_DAPM_ENUM("Mono ADC2 left source", rt5640_mono_adc_l2_enum);
@@ -798,9 +793,8 @@ static const char * const rt5640_mono_adc_r1_src[] = {
"Mono DAC MIXR", "ADCR"
};
-static const SOC_ENUM_SINGLE_DECL(
- rt5640_mono_adc_r1_enum, RT5640_MONO_ADC_MIXER,
- RT5640_MONO_ADC_R1_SRC_SFT, rt5640_mono_adc_r1_src);
+static SOC_ENUM_SINGLE_DECL(rt5640_mono_adc_r1_enum, RT5640_MONO_ADC_MIXER,
+ RT5640_MONO_ADC_R1_SRC_SFT, rt5640_mono_adc_r1_src);
static const struct snd_kcontrol_new rt5640_mono_adc_r1_mux =
SOC_DAPM_ENUM("Mono ADC1 right source", rt5640_mono_adc_r1_enum);
@@ -809,9 +803,8 @@ static const char * const rt5640_mono_adc_r2_src[] = {
"DMIC R1", "DMIC R2", "Mono DAC MIXR"
};
-static const SOC_ENUM_SINGLE_DECL(
- rt5640_mono_adc_r2_enum, RT5640_MONO_ADC_MIXER,
- RT5640_MONO_ADC_R2_SRC_SFT, rt5640_mono_adc_r2_src);
+static SOC_ENUM_SINGLE_DECL(rt5640_mono_adc_r2_enum, RT5640_MONO_ADC_MIXER,
+ RT5640_MONO_ADC_R2_SRC_SFT, rt5640_mono_adc_r2_src);
static const struct snd_kcontrol_new rt5640_mono_adc_r2_mux =
SOC_DAPM_ENUM("Mono ADC2 right source", rt5640_mono_adc_r2_enum);
@@ -826,9 +819,9 @@ static int rt5640_dac_l2_values[] = {
3,
};
-static const SOC_VALUE_ENUM_SINGLE_DECL(
- rt5640_dac_l2_enum, RT5640_DSP_PATH2, RT5640_DAC_L2_SEL_SFT,
- 0x3, rt5640_dac_l2_src, rt5640_dac_l2_values);
+static SOC_VALUE_ENUM_SINGLE_DECL(rt5640_dac_l2_enum,
+ RT5640_DSP_PATH2, RT5640_DAC_L2_SEL_SFT,
+ 0x3, rt5640_dac_l2_src, rt5640_dac_l2_values);
static const struct snd_kcontrol_new rt5640_dac_l2_mux =
SOC_DAPM_VALUE_ENUM("DAC2 left channel source", rt5640_dac_l2_enum);
@@ -841,9 +834,9 @@ static int rt5640_dac_r2_values[] = {
0,
};
-static const SOC_VALUE_ENUM_SINGLE_DECL(
- rt5640_dac_r2_enum, RT5640_DSP_PATH2, RT5640_DAC_R2_SEL_SFT,
- 0x3, rt5640_dac_r2_src, rt5640_dac_r2_values);
+static SOC_VALUE_ENUM_SINGLE_DECL(rt5640_dac_r2_enum,
+ RT5640_DSP_PATH2, RT5640_DAC_R2_SEL_SFT,
+ 0x3, rt5640_dac_r2_src, rt5640_dac_r2_values);
static const struct snd_kcontrol_new rt5640_dac_r2_mux =
SOC_DAPM_ENUM("DAC2 right channel source", rt5640_dac_r2_enum);
@@ -860,9 +853,10 @@ static int rt5640_dai_iis_map_values[] = {
7,
};
-static const SOC_VALUE_ENUM_SINGLE_DECL(
- rt5640_dai_iis_map_enum, RT5640_I2S1_SDP, RT5640_I2S_IF_SFT,
- 0x7, rt5640_dai_iis_map, rt5640_dai_iis_map_values);
+static SOC_VALUE_ENUM_SINGLE_DECL(rt5640_dai_iis_map_enum,
+ RT5640_I2S1_SDP, RT5640_I2S_IF_SFT,
+ 0x7, rt5640_dai_iis_map,
+ rt5640_dai_iis_map_values);
static const struct snd_kcontrol_new rt5640_dai_mux =
SOC_DAPM_VALUE_ENUM("DAI select", rt5640_dai_iis_map_enum);
@@ -872,9 +866,8 @@ static const char * const rt5640_sdi_sel[] = {
"IF1", "IF2"
};
-static const SOC_ENUM_SINGLE_DECL(
- rt5640_sdi_sel_enum, RT5640_I2S2_SDP,
- RT5640_I2S2_SDI_SFT, rt5640_sdi_sel);
+static SOC_ENUM_SINGLE_DECL(rt5640_sdi_sel_enum, RT5640_I2S2_SDP,
+ RT5640_I2S2_SDI_SFT, rt5640_sdi_sel);
static const struct snd_kcontrol_new rt5640_sdi_mux =
SOC_DAPM_ENUM("SDI select", rt5640_sdi_sel_enum);
@@ -1601,8 +1594,7 @@ static int get_clk_info(int sclk, int rate)
static int rt5640_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 rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec);
unsigned int val_len = 0, val_clk, mask_clk;
int dai_sel, pre_div, bclk_ms, frame_size;
@@ -1943,16 +1935,8 @@ static int rt5640_set_bias_level(struct snd_soc_codec *codec,
static int rt5640_probe(struct snd_soc_codec *codec)
{
struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec);
- int ret;
rt5640->codec = codec;
- codec->control_data = rt5640->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;
- }
codec->dapm.idle_bias_off = 1;
rt5640_set_bias_level(codec, SND_SOC_BIAS_OFF);
diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c
index 0fcbe90..d3ed1be 100644
--- a/sound/soc/codecs/sgtl5000.c
+++ b/sound/soc/codecs/sgtl5000.c
@@ -187,8 +187,9 @@ static const char *adc_mux_text[] = {
"MIC_IN", "LINE_IN"
};
-static const struct soc_enum adc_enum =
-SOC_ENUM_SINGLE(SGTL5000_CHIP_ANA_CTRL, 2, 2, adc_mux_text);
+static SOC_ENUM_SINGLE_DECL(adc_enum,
+ SGTL5000_CHIP_ANA_CTRL, 2,
+ adc_mux_text);
static const struct snd_kcontrol_new adc_mux =
SOC_DAPM_ENUM("Capture Mux", adc_enum);
@@ -198,8 +199,9 @@ static const char *dac_mux_text[] = {
"DAC", "LINE_IN"
};
-static const struct soc_enum dac_enum =
-SOC_ENUM_SINGLE(SGTL5000_CHIP_ANA_CTRL, 6, 2, dac_mux_text);
+static SOC_ENUM_SINGLE_DECL(dac_enum,
+ SGTL5000_CHIP_ANA_CTRL, 6,
+ dac_mux_text);
static const struct snd_kcontrol_new dac_mux =
SOC_DAPM_ENUM("Headphone Mux", dac_enum);
@@ -1350,14 +1352,6 @@ static int sgtl5000_probe(struct snd_soc_codec *codec)
int ret;
struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec);
- /* setup i2c data ops */
- codec->control_data = sgtl5000->regmap;
- ret = snd_soc_codec_set_cache_io(codec, 16, 16, SND_SOC_REGMAP);
- if (ret < 0) {
- dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
- return ret;
- }
-
ret = sgtl5000_enable_regulators(codec);
if (ret)
return ret;
diff --git a/sound/soc/codecs/si476x.c b/sound/soc/codecs/si476x.c
index fa2b8e0..244c097 100644
--- a/sound/soc/codecs/si476x.c
+++ b/sound/soc/codecs/si476x.c
@@ -21,6 +21,7 @@
#include <linux/slab.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
+#include <linux/regmap.h>
#include <sound/soc.h>
#include <sound/initval.h>
@@ -209,8 +210,9 @@ out:
static int si476x_codec_probe(struct snd_soc_codec *codec)
{
- codec->control_data = dev_get_regmap(codec->dev->parent, NULL);
- return snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP);
+ struct regmap *regmap = dev_get_regmap(codec->dev->parent, NULL);
+
+ return snd_soc_codec_set_cache_io(codec, regmap);
}
static struct snd_soc_dai_ops si476x_dai_ops = {
diff --git a/sound/soc/codecs/sirf-audio-codec.c b/sound/soc/codecs/sirf-audio-codec.c
new file mode 100644
index 0000000..58e7c1f
--- /dev/null
+++ b/sound/soc/codecs/sirf-audio-codec.c
@@ -0,0 +1,524 @@
+/*
+ * SiRF audio codec driver
+ *
+ * Copyright (c) 2011 Cambridge Silicon Radio Limited, a CSR plc group company.
+ *
+ * Licensed under GPLv2 or later.
+ */
+
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <linux/pm_runtime.h>
+#include <linux/of.h>
+#include <linux/of_device.h>
+#include <linux/clk.h>
+#include <linux/delay.h>
+#include <linux/io.h>
+#include <linux/regmap.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/initval.h>
+#include <sound/tlv.h>
+#include <sound/soc.h>
+#include <sound/dmaengine_pcm.h>
+
+#include "sirf-audio-codec.h"
+
+struct sirf_audio_codec {
+ struct clk *clk;
+ struct regmap *regmap;
+ u32 reg_ctrl0, reg_ctrl1;
+};
+
+static const char * const input_mode_mux[] = {"Single-ended",
+ "Differential"};
+
+static const struct soc_enum input_mode_mux_enum =
+ SOC_ENUM_SINGLE(AUDIO_IC_CODEC_CTRL1, 4, 2, input_mode_mux);
+
+static const struct snd_kcontrol_new sirf_audio_codec_input_mode_control =
+ SOC_DAPM_ENUM("Route", input_mode_mux_enum);
+
+static const DECLARE_TLV_DB_SCALE(playback_vol_tlv, -12400, 100, 0);
+static const DECLARE_TLV_DB_SCALE(capture_vol_tlv_prima2, 500, 100, 0);
+static const DECLARE_TLV_DB_RANGE(capture_vol_tlv_atlas6,
+ 0, 7, TLV_DB_SCALE_ITEM(-100, 100, 0),
+ 0x22, 0x3F, TLV_DB_SCALE_ITEM(700, 100, 0),
+);
+
+static struct snd_kcontrol_new volume_controls_atlas6[] = {
+ SOC_DOUBLE_TLV("Playback Volume", AUDIO_IC_CODEC_CTRL0, 21, 14,
+ 0x7F, 0, playback_vol_tlv),
+ SOC_DOUBLE_TLV("Capture Volume", AUDIO_IC_CODEC_CTRL1, 16, 10,
+ 0x3F, 0, capture_vol_tlv_atlas6),
+};
+
+static struct snd_kcontrol_new volume_controls_prima2[] = {
+ SOC_DOUBLE_TLV("Speaker Volume", AUDIO_IC_CODEC_CTRL0, 21, 14,
+ 0x7F, 0, playback_vol_tlv),
+ SOC_DOUBLE_TLV("Capture Volume", AUDIO_IC_CODEC_CTRL1, 15, 10,
+ 0x1F, 0, capture_vol_tlv_prima2),
+};
+
+static struct snd_kcontrol_new left_input_path_controls[] = {
+ SOC_DAPM_SINGLE("Line Left Switch", AUDIO_IC_CODEC_CTRL1, 6, 1, 0),
+ SOC_DAPM_SINGLE("Mic Left Switch", AUDIO_IC_CODEC_CTRL1, 3, 1, 0),
+};
+
+static struct snd_kcontrol_new right_input_path_controls[] = {
+ SOC_DAPM_SINGLE("Line Right Switch", AUDIO_IC_CODEC_CTRL1, 5, 1, 0),
+ SOC_DAPM_SINGLE("Mic Right Switch", AUDIO_IC_CODEC_CTRL1, 2, 1, 0),
+};
+
+static struct snd_kcontrol_new left_dac_to_hp_left_amp_switch_control =
+ SOC_DAPM_SINGLE("Switch", AUDIO_IC_CODEC_CTRL0, 9, 1, 0);
+
+static struct snd_kcontrol_new left_dac_to_hp_right_amp_switch_control =
+ SOC_DAPM_SINGLE("Switch", AUDIO_IC_CODEC_CTRL0, 8, 1, 0);
+
+static struct snd_kcontrol_new right_dac_to_hp_left_amp_switch_control =
+ SOC_DAPM_SINGLE("Switch", AUDIO_IC_CODEC_CTRL0, 7, 1, 0);
+
+static struct snd_kcontrol_new right_dac_to_hp_right_amp_switch_control =
+ SOC_DAPM_SINGLE("Switch", AUDIO_IC_CODEC_CTRL0, 6, 1, 0);
+
+static struct snd_kcontrol_new left_dac_to_speaker_lineout_switch_control =
+ SOC_DAPM_SINGLE("Switch", AUDIO_IC_CODEC_CTRL0, 11, 1, 0);
+
+static struct snd_kcontrol_new right_dac_to_speaker_lineout_switch_control =
+ SOC_DAPM_SINGLE("Switch", AUDIO_IC_CODEC_CTRL0, 10, 1, 0);
+
+/* After enable adc, Delay 200ms to avoid pop noise */
+static int adc_enable_delay_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ switch (event) {
+ case SND_SOC_DAPM_POST_PMU:
+ msleep(200);
+ break;
+ default:
+ break;
+ }
+
+ return 0;
+}
+
+static void enable_and_reset_codec(struct regmap *regmap,
+ u32 codec_enable_bits, u32 codec_reset_bits)
+{
+ regmap_update_bits(regmap, AUDIO_IC_CODEC_CTRL1,
+ codec_enable_bits | codec_reset_bits,
+ codec_enable_bits | ~codec_reset_bits);
+ msleep(20);
+ regmap_update_bits(regmap, AUDIO_IC_CODEC_CTRL1,
+ codec_reset_bits, codec_reset_bits);
+}
+
+static int atlas6_codec_enable_and_reset_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+#define ATLAS6_CODEC_ENABLE_BITS (1 << 29)
+#define ATLAS6_CODEC_RESET_BITS (1 << 28)
+ struct sirf_audio_codec *sirf_audio_codec = dev_get_drvdata(w->codec->dev);
+ switch (event) {
+ case SND_SOC_DAPM_PRE_PMU:
+ enable_and_reset_codec(sirf_audio_codec->regmap,
+ ATLAS6_CODEC_ENABLE_BITS, ATLAS6_CODEC_RESET_BITS);
+ break;
+ case SND_SOC_DAPM_POST_PMD:
+ regmap_update_bits(sirf_audio_codec->regmap,
+ AUDIO_IC_CODEC_CTRL1, ATLAS6_CODEC_ENABLE_BITS,
+ ~ATLAS6_CODEC_ENABLE_BITS);
+ break;
+ default:
+ break;
+ }
+
+ return 0;
+}
+
+static int prima2_codec_enable_and_reset_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+#define PRIMA2_CODEC_ENABLE_BITS (1 << 27)
+#define PRIMA2_CODEC_RESET_BITS (1 << 26)
+ struct sirf_audio_codec *sirf_audio_codec = dev_get_drvdata(w->codec->dev);
+ switch (event) {
+ case SND_SOC_DAPM_POST_PMU:
+ enable_and_reset_codec(sirf_audio_codec->regmap,
+ PRIMA2_CODEC_ENABLE_BITS, PRIMA2_CODEC_RESET_BITS);
+ break;
+ case SND_SOC_DAPM_POST_PMD:
+ regmap_update_bits(sirf_audio_codec->regmap,
+ AUDIO_IC_CODEC_CTRL1, PRIMA2_CODEC_ENABLE_BITS,
+ ~PRIMA2_CODEC_ENABLE_BITS);
+ break;
+ default:
+ break;
+ }
+
+ return 0;
+}
+
+static const struct snd_soc_dapm_widget atlas6_output_driver_dapm_widgets[] = {
+ SND_SOC_DAPM_OUT_DRV("HP Left Driver", AUDIO_IC_CODEC_CTRL1,
+ 25, 0, NULL, 0),
+ SND_SOC_DAPM_OUT_DRV("HP Right Driver", AUDIO_IC_CODEC_CTRL1,
+ 26, 0, NULL, 0),
+ SND_SOC_DAPM_OUT_DRV("Speaker Driver", AUDIO_IC_CODEC_CTRL1,
+ 27, 0, NULL, 0),
+};
+
+static const struct snd_soc_dapm_widget prima2_output_driver_dapm_widgets[] = {
+ SND_SOC_DAPM_OUT_DRV("HP Left Driver", AUDIO_IC_CODEC_CTRL1,
+ 23, 0, NULL, 0),
+ SND_SOC_DAPM_OUT_DRV("HP Right Driver", AUDIO_IC_CODEC_CTRL1,
+ 24, 0, NULL, 0),
+ SND_SOC_DAPM_OUT_DRV("Speaker Driver", AUDIO_IC_CODEC_CTRL1,
+ 25, 0, NULL, 0),
+};
+
+static const struct snd_soc_dapm_widget atlas6_codec_clock_dapm_widget =
+ SND_SOC_DAPM_SUPPLY("codecclk", SND_SOC_NOPM, 0, 0,
+ atlas6_codec_enable_and_reset_event,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD);
+
+static const struct snd_soc_dapm_widget prima2_codec_clock_dapm_widget =
+ SND_SOC_DAPM_SUPPLY("codecclk", SND_SOC_NOPM, 0, 0,
+ prima2_codec_enable_and_reset_event,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD);
+
+static const struct snd_soc_dapm_widget sirf_audio_codec_dapm_widgets[] = {
+ SND_SOC_DAPM_DAC("DAC left", NULL, AUDIO_IC_CODEC_CTRL0, 1, 0),
+ SND_SOC_DAPM_DAC("DAC right", NULL, AUDIO_IC_CODEC_CTRL0, 0, 0),
+ SND_SOC_DAPM_SWITCH("Left dac to hp left amp", SND_SOC_NOPM, 0, 0,
+ &left_dac_to_hp_left_amp_switch_control),
+ SND_SOC_DAPM_SWITCH("Left dac to hp right amp", SND_SOC_NOPM, 0, 0,
+ &left_dac_to_hp_right_amp_switch_control),
+ SND_SOC_DAPM_SWITCH("Right dac to hp left amp", SND_SOC_NOPM, 0, 0,
+ &right_dac_to_hp_left_amp_switch_control),
+ SND_SOC_DAPM_SWITCH("Right dac to hp right amp", SND_SOC_NOPM, 0, 0,
+ &right_dac_to_hp_right_amp_switch_control),
+ SND_SOC_DAPM_OUT_DRV("HP amp left driver", AUDIO_IC_CODEC_CTRL0, 3, 0,
+ NULL, 0),
+ SND_SOC_DAPM_OUT_DRV("HP amp right driver", AUDIO_IC_CODEC_CTRL0, 3, 0,
+ NULL, 0),
+
+ SND_SOC_DAPM_SWITCH("Left dac to speaker lineout", SND_SOC_NOPM, 0, 0,
+ &left_dac_to_speaker_lineout_switch_control),
+ SND_SOC_DAPM_SWITCH("Right dac to speaker lineout", SND_SOC_NOPM, 0, 0,
+ &right_dac_to_speaker_lineout_switch_control),
+ SND_SOC_DAPM_OUT_DRV("Speaker amp driver", AUDIO_IC_CODEC_CTRL0, 4, 0,
+ NULL, 0),
+
+ SND_SOC_DAPM_OUTPUT("HPOUTL"),
+ SND_SOC_DAPM_OUTPUT("HPOUTR"),
+ SND_SOC_DAPM_OUTPUT("SPKOUT"),
+
+ SND_SOC_DAPM_ADC_E("ADC left", NULL, AUDIO_IC_CODEC_CTRL1, 8, 0,
+ adc_enable_delay_event, SND_SOC_DAPM_POST_PMU),
+ SND_SOC_DAPM_ADC_E("ADC right", NULL, AUDIO_IC_CODEC_CTRL1, 7, 0,
+ adc_enable_delay_event, SND_SOC_DAPM_POST_PMU),
+ SND_SOC_DAPM_MIXER("Left PGA mixer", AUDIO_IC_CODEC_CTRL1, 1, 0,
+ &left_input_path_controls[0],
+ ARRAY_SIZE(left_input_path_controls)),
+ SND_SOC_DAPM_MIXER("Right PGA mixer", AUDIO_IC_CODEC_CTRL1, 0, 0,
+ &right_input_path_controls[0],
+ ARRAY_SIZE(right_input_path_controls)),
+
+ SND_SOC_DAPM_MUX("Mic input mode mux", SND_SOC_NOPM, 0, 0,
+ &sirf_audio_codec_input_mode_control),
+ SND_SOC_DAPM_MICBIAS("Mic Bias", AUDIO_IC_CODEC_PWR, 3, 0),
+ SND_SOC_DAPM_INPUT("MICIN1"),
+ SND_SOC_DAPM_INPUT("MICIN2"),
+ SND_SOC_DAPM_INPUT("LINEIN1"),
+ SND_SOC_DAPM_INPUT("LINEIN2"),
+
+ SND_SOC_DAPM_SUPPLY("HSL Phase Opposite", AUDIO_IC_CODEC_CTRL0,
+ 30, 0, NULL, 0),
+};
+
+static const struct snd_soc_dapm_route sirf_audio_codec_map[] = {
+ {"SPKOUT", NULL, "Speaker Driver"},
+ {"Speaker Driver", NULL, "Speaker amp driver"},
+ {"Speaker amp driver", NULL, "Left dac to speaker lineout"},
+ {"Speaker amp driver", NULL, "Right dac to speaker lineout"},
+ {"Left dac to speaker lineout", "Switch", "DAC left"},
+ {"Right dac to speaker lineout", "Switch", "DAC right"},
+ {"HPOUTL", NULL, "HP Left Driver"},
+ {"HPOUTR", NULL, "HP Right Driver"},
+ {"HP Left Driver", NULL, "HP amp left driver"},
+ {"HP Right Driver", NULL, "HP amp right driver"},
+ {"HP amp left driver", NULL, "Right dac to hp left amp"},
+ {"HP amp right driver", NULL , "Right dac to hp right amp"},
+ {"HP amp left driver", NULL, "Left dac to hp left amp"},
+ {"HP amp right driver", NULL , "Right dac to hp right amp"},
+ {"Right dac to hp left amp", "Switch", "DAC left"},
+ {"Right dac to hp right amp", "Switch", "DAC right"},
+ {"Left dac to hp left amp", "Switch", "DAC left"},
+ {"Left dac to hp right amp", "Switch", "DAC right"},
+ {"DAC left", NULL, "codecclk"},
+ {"DAC right", NULL, "codecclk"},
+ {"DAC left", NULL, "Playback"},
+ {"DAC right", NULL, "Playback"},
+ {"DAC left", NULL, "HSL Phase Opposite"},
+ {"DAC right", NULL, "HSL Phase Opposite"},
+
+ {"Capture", NULL, "ADC left"},
+ {"Capture", NULL, "ADC right"},
+ {"ADC left", NULL, "codecclk"},
+ {"ADC right", NULL, "codecclk"},
+ {"ADC left", NULL, "Left PGA mixer"},
+ {"ADC right", NULL, "Right PGA mixer"},
+ {"Left PGA mixer", "Line Left Switch", "LINEIN2"},
+ {"Right PGA mixer", "Line Right Switch", "LINEIN1"},
+ {"Left PGA mixer", "Mic Left Switch", "MICIN2"},
+ {"Right PGA mixer", "Mic Right Switch", "Mic input mode mux"},
+ {"Mic input mode mux", "Single-ended", "MICIN1"},
+ {"Mic input mode mux", "Differential", "MICIN1"},
+};
+
+static int sirf_audio_codec_trigger(struct snd_pcm_substream *substream,
+ int cmd,
+ struct snd_soc_dai *dai)
+{
+ int playback = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
+ struct snd_soc_codec *codec = dai->codec;
+ u32 val = 0;
+
+ /*
+ * This is a workaround, When stop playback,
+ * need disable HP amp, avoid the current noise.
+ */
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ break;
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ if (playback)
+ val = IC_HSLEN | IC_HSREN;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ if (playback)
+ snd_soc_update_bits(codec, AUDIO_IC_CODEC_CTRL0,
+ IC_HSLEN | IC_HSREN, val);
+ return 0;
+}
+
+struct snd_soc_dai_ops sirf_audio_codec_dai_ops = {
+ .trigger = sirf_audio_codec_trigger,
+};
+
+struct snd_soc_dai_driver sirf_audio_codec_dai = {
+ .name = "sirf-audio-codec",
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_48000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ },
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_48000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ },
+ .ops = &sirf_audio_codec_dai_ops,
+};
+
+static int sirf_audio_codec_probe(struct snd_soc_codec *codec)
+{
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
+
+ pm_runtime_enable(codec->dev);
+
+ if (of_device_is_compatible(codec->dev->of_node, "sirf,prima2-audio-codec")) {
+ snd_soc_dapm_new_controls(dapm,
+ prima2_output_driver_dapm_widgets,
+ ARRAY_SIZE(prima2_output_driver_dapm_widgets));
+ snd_soc_dapm_new_controls(dapm,
+ &prima2_codec_clock_dapm_widget, 1);
+ return snd_soc_add_codec_controls(codec,
+ volume_controls_prima2,
+ ARRAY_SIZE(volume_controls_prima2));
+ }
+ if (of_device_is_compatible(codec->dev->of_node, "sirf,atlas6-audio-codec")) {
+ snd_soc_dapm_new_controls(dapm,
+ atlas6_output_driver_dapm_widgets,
+ ARRAY_SIZE(atlas6_output_driver_dapm_widgets));
+ snd_soc_dapm_new_controls(dapm,
+ &atlas6_codec_clock_dapm_widget, 1);
+ return snd_soc_add_codec_controls(codec,
+ volume_controls_atlas6,
+ ARRAY_SIZE(volume_controls_atlas6));
+ }
+
+ return -EINVAL;
+}
+
+static int sirf_audio_codec_remove(struct snd_soc_codec *codec)
+{
+ pm_runtime_disable(codec->dev);
+ return 0;
+}
+
+static struct snd_soc_codec_driver soc_codec_device_sirf_audio_codec = {
+ .probe = sirf_audio_codec_probe,
+ .remove = sirf_audio_codec_remove,
+ .dapm_widgets = sirf_audio_codec_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(sirf_audio_codec_dapm_widgets),
+ .dapm_routes = sirf_audio_codec_map,
+ .num_dapm_routes = ARRAY_SIZE(sirf_audio_codec_map),
+ .idle_bias_off = true,
+};
+
+static const struct of_device_id sirf_audio_codec_of_match[] = {
+ { .compatible = "sirf,prima2-audio-codec" },
+ { .compatible = "sirf,atlas6-audio-codec" },
+ {}
+};
+MODULE_DEVICE_TABLE(of, sirf_audio_codec_of_match);
+
+static const struct regmap_config sirf_audio_codec_regmap_config = {
+ .reg_bits = 32,
+ .reg_stride = 4,
+ .val_bits = 32,
+ .max_register = AUDIO_IC_CODEC_CTRL3,
+ .cache_type = REGCACHE_NONE,
+};
+
+static int sirf_audio_codec_driver_probe(struct platform_device *pdev)
+{
+ int ret;
+ struct sirf_audio_codec *sirf_audio_codec;
+ void __iomem *base;
+ struct resource *mem_res;
+ const struct of_device_id *match;
+
+ match = of_match_node(sirf_audio_codec_of_match, pdev->dev.of_node);
+
+ sirf_audio_codec = devm_kzalloc(&pdev->dev,
+ sizeof(struct sirf_audio_codec), GFP_KERNEL);
+ if (!sirf_audio_codec)
+ return -ENOMEM;
+
+ platform_set_drvdata(pdev, sirf_audio_codec);
+
+ mem_res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+ base = devm_ioremap_resource(&pdev->dev, mem_res);
+ if (base == NULL)
+ return -ENOMEM;
+
+ sirf_audio_codec->regmap = devm_regmap_init_mmio(&pdev->dev, base,
+ &sirf_audio_codec_regmap_config);
+ if (IS_ERR(sirf_audio_codec->regmap))
+ return PTR_ERR(sirf_audio_codec->regmap);
+
+ sirf_audio_codec->clk = devm_clk_get(&pdev->dev, NULL);
+ if (IS_ERR(sirf_audio_codec->clk)) {
+ dev_err(&pdev->dev, "Get clock failed.\n");
+ return PTR_ERR(sirf_audio_codec->clk);
+ }
+
+ ret = clk_prepare_enable(sirf_audio_codec->clk);
+ if (ret) {
+ dev_err(&pdev->dev, "Enable clock failed.\n");
+ return ret;
+ }
+
+ ret = snd_soc_register_codec(&(pdev->dev),
+ &soc_codec_device_sirf_audio_codec,
+ &sirf_audio_codec_dai, 1);
+ if (ret) {
+ dev_err(&pdev->dev, "Register Audio Codec dai failed.\n");
+ goto err_clk_put;
+ }
+
+ /*
+ * Always open charge pump, if not, when the charge pump closed the
+ * adc will not stable
+ */
+ regmap_update_bits(sirf_audio_codec->regmap, AUDIO_IC_CODEC_CTRL0,
+ IC_CPFREQ, IC_CPFREQ);
+
+ if (of_device_is_compatible(pdev->dev.of_node, "sirf,atlas6-audio-codec"))
+ regmap_update_bits(sirf_audio_codec->regmap,
+ AUDIO_IC_CODEC_CTRL0, IC_CPEN, IC_CPEN);
+ return 0;
+
+err_clk_put:
+ clk_disable_unprepare(sirf_audio_codec->clk);
+ return ret;
+}
+
+static int sirf_audio_codec_driver_remove(struct platform_device *pdev)
+{
+ struct sirf_audio_codec *sirf_audio_codec = platform_get_drvdata(pdev);
+
+ clk_disable_unprepare(sirf_audio_codec->clk);
+ snd_soc_unregister_codec(&(pdev->dev));
+
+ return 0;
+}
+
+#ifdef CONFIG_PM_SLEEP
+static int sirf_audio_codec_suspend(struct device *dev)
+{
+ struct sirf_audio_codec *sirf_audio_codec = dev_get_drvdata(dev);
+
+ regmap_read(sirf_audio_codec->regmap, AUDIO_IC_CODEC_CTRL0,
+ &sirf_audio_codec->reg_ctrl0);
+ regmap_read(sirf_audio_codec->regmap, AUDIO_IC_CODEC_CTRL1,
+ &sirf_audio_codec->reg_ctrl1);
+ clk_disable_unprepare(sirf_audio_codec->clk);
+
+ return 0;
+}
+
+static int sirf_audio_codec_resume(struct device *dev)
+{
+ struct sirf_audio_codec *sirf_audio_codec = dev_get_drvdata(dev);
+ int ret;
+
+ ret = clk_prepare_enable(sirf_audio_codec->clk);
+ if (ret)
+ return ret;
+
+ regmap_write(sirf_audio_codec->regmap, AUDIO_IC_CODEC_CTRL0,
+ sirf_audio_codec->reg_ctrl0);
+ regmap_write(sirf_audio_codec->regmap, AUDIO_IC_CODEC_CTRL1,
+ sirf_audio_codec->reg_ctrl1);
+
+ return 0;
+}
+#endif
+
+static const struct dev_pm_ops sirf_audio_codec_pm_ops = {
+ SET_SYSTEM_SLEEP_PM_OPS(sirf_audio_codec_suspend, sirf_audio_codec_resume)
+};
+
+static struct platform_driver sirf_audio_codec_driver = {
+ .driver = {
+ .name = "sirf-audio-codec",
+ .owner = THIS_MODULE,
+ .of_match_table = sirf_audio_codec_of_match,
+ .pm = &sirf_audio_codec_pm_ops,
+ },
+ .probe = sirf_audio_codec_driver_probe,
+ .remove = sirf_audio_codec_driver_remove,
+};
+
+module_platform_driver(sirf_audio_codec_driver);
+
+MODULE_DESCRIPTION("SiRF audio codec driver");
+MODULE_AUTHOR("RongJun Ying <Rongjun.Ying@csr.com>");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/codecs/sirf-audio-codec.h b/sound/soc/codecs/sirf-audio-codec.h
new file mode 100644
index 0000000..d4c187b
--- /dev/null
+++ b/sound/soc/codecs/sirf-audio-codec.h
@@ -0,0 +1,75 @@
+/*
+ * SiRF inner codec controllers define
+ *
+ * Copyright (c) 2011 Cambridge Silicon Radio Limited, a CSR plc group company.
+ *
+ * Licensed under GPLv2 or later.
+ */
+
+#ifndef _SIRF_AUDIO_CODEC_H
+#define _SIRF_AUDIO_CODEC_H
+
+
+#define AUDIO_IC_CODEC_PWR (0x00E0)
+#define AUDIO_IC_CODEC_CTRL0 (0x00E4)
+#define AUDIO_IC_CODEC_CTRL1 (0x00E8)
+#define AUDIO_IC_CODEC_CTRL2 (0x00EC)
+#define AUDIO_IC_CODEC_CTRL3 (0x00F0)
+
+#define MICBIASEN (1 << 3)
+
+#define IC_RDACEN (1 << 0)
+#define IC_LDACEN (1 << 1)
+#define IC_HSREN (1 << 2)
+#define IC_HSLEN (1 << 3)
+#define IC_SPEN (1 << 4)
+#define IC_CPEN (1 << 5)
+
+#define IC_HPRSELR (1 << 6)
+#define IC_HPLSELR (1 << 7)
+#define IC_HPRSELL (1 << 8)
+#define IC_HPLSELL (1 << 9)
+#define IC_SPSELR (1 << 10)
+#define IC_SPSELL (1 << 11)
+
+#define IC_MONOR (1 << 12)
+#define IC_MONOL (1 << 13)
+
+#define IC_RXOSRSEL (1 << 28)
+#define IC_CPFREQ (1 << 29)
+#define IC_HSINVEN (1 << 30)
+
+#define IC_MICINREN (1 << 0)
+#define IC_MICINLEN (1 << 1)
+#define IC_MICIN1SEL (1 << 2)
+#define IC_MICIN2SEL (1 << 3)
+#define IC_MICDIFSEL (1 << 4)
+#define IC_LINEIN1SEL (1 << 5)
+#define IC_LINEIN2SEL (1 << 6)
+#define IC_RADCEN (1 << 7)
+#define IC_LADCEN (1 << 8)
+#define IC_ALM (1 << 9)
+
+#define IC_DIGMICEN (1 << 22)
+#define IC_DIGMICFREQ (1 << 23)
+#define IC_ADC14B_12 (1 << 24)
+#define IC_FIRDAC_HSL_EN (1 << 25)
+#define IC_FIRDAC_HSR_EN (1 << 26)
+#define IC_FIRDAC_LOUT_EN (1 << 27)
+#define IC_POR (1 << 28)
+#define IC_CODEC_CLK_EN (1 << 29)
+#define IC_HP_3DB_BOOST (1 << 30)
+
+#define IC_ADC_LEFT_GAIN_SHIFT 16
+#define IC_ADC_RIGHT_GAIN_SHIFT 10
+#define IC_ADC_GAIN_MASK 0x3F
+#define IC_MIC_MAX_GAIN 0x39
+
+#define IC_RXPGAR_MASK 0x3F
+#define IC_RXPGAR_SHIFT 14
+#define IC_RXPGAL_MASK 0x3F
+#define IC_RXPGAL_SHIFT 21
+#define IC_RXPGAR 0x7B
+#define IC_RXPGAL 0x7B
+
+#endif /*__SIRF_AUDIO_CODEC_H*/
diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c
index 13045f2..42dff26 100644
--- a/sound/soc/codecs/sn95031.c
+++ b/sound/soc/codecs/sn95031.c
@@ -312,14 +312,14 @@ static int sn95031_dmic56_event(struct snd_soc_dapm_widget *w,
/* mux controls */
static const char *sn95031_mic_texts[] = { "AMIC", "LineIn" };
-static const struct soc_enum sn95031_micl_enum =
- SOC_ENUM_SINGLE(SN95031_ADCCONFIG, 1, 2, sn95031_mic_texts);
+static SOC_ENUM_SINGLE_DECL(sn95031_micl_enum,
+ SN95031_ADCCONFIG, 1, sn95031_mic_texts);
static const struct snd_kcontrol_new sn95031_micl_mux_control =
SOC_DAPM_ENUM("Route", sn95031_micl_enum);
-static const struct soc_enum sn95031_micr_enum =
- SOC_ENUM_SINGLE(SN95031_ADCCONFIG, 3, 2, sn95031_mic_texts);
+static SOC_ENUM_SINGLE_DECL(sn95031_micr_enum,
+ SN95031_ADCCONFIG, 3, sn95031_mic_texts);
static const struct snd_kcontrol_new sn95031_micr_mux_control =
SOC_DAPM_ENUM("Route", sn95031_micr_enum);
@@ -328,26 +328,26 @@ static const char *sn95031_input_texts[] = { "DMIC1", "DMIC2", "DMIC3",
"DMIC4", "DMIC5", "DMIC6",
"ADC Left", "ADC Right" };
-static const struct soc_enum sn95031_input1_enum =
- SOC_ENUM_SINGLE(SN95031_AUDIOMUX12, 0, 8, sn95031_input_texts);
+static SOC_ENUM_SINGLE_DECL(sn95031_input1_enum,
+ SN95031_AUDIOMUX12, 0, sn95031_input_texts);
static const struct snd_kcontrol_new sn95031_input1_mux_control =
SOC_DAPM_ENUM("Route", sn95031_input1_enum);
-static const struct soc_enum sn95031_input2_enum =
- SOC_ENUM_SINGLE(SN95031_AUDIOMUX12, 4, 8, sn95031_input_texts);
+static SOC_ENUM_SINGLE_DECL(sn95031_input2_enum,
+ SN95031_AUDIOMUX12, 4, sn95031_input_texts);
static const struct snd_kcontrol_new sn95031_input2_mux_control =
SOC_DAPM_ENUM("Route", sn95031_input2_enum);
-static const struct soc_enum sn95031_input3_enum =
- SOC_ENUM_SINGLE(SN95031_AUDIOMUX34, 0, 8, sn95031_input_texts);
+static SOC_ENUM_SINGLE_DECL(sn95031_input3_enum,
+ SN95031_AUDIOMUX34, 0, sn95031_input_texts);
static const struct snd_kcontrol_new sn95031_input3_mux_control =
SOC_DAPM_ENUM("Route", sn95031_input3_enum);
-static const struct soc_enum sn95031_input4_enum =
- SOC_ENUM_SINGLE(SN95031_AUDIOMUX34, 4, 8, sn95031_input_texts);
+static SOC_ENUM_SINGLE_DECL(sn95031_input4_enum,
+ SN95031_AUDIOMUX34, 4, sn95031_input_texts);
static const struct snd_kcontrol_new sn95031_input4_mux_control =
SOC_DAPM_ENUM("Route", sn95031_input4_enum);
@@ -359,19 +359,19 @@ static const char *sn95031_micmode_text[] = {"Single Ended", "Differential"};
/* 0dB to 30dB in 10dB steps */
static const DECLARE_TLV_DB_SCALE(mic_tlv, 0, 10, 0);
-static const struct soc_enum sn95031_micmode1_enum =
- SOC_ENUM_SINGLE(SN95031_MICAMP1, 1, 2, sn95031_micmode_text);
-static const struct soc_enum sn95031_micmode2_enum =
- SOC_ENUM_SINGLE(SN95031_MICAMP2, 1, 2, sn95031_micmode_text);
+static SOC_ENUM_SINGLE_DECL(sn95031_micmode1_enum,
+ SN95031_MICAMP1, 1, sn95031_micmode_text);
+static SOC_ENUM_SINGLE_DECL(sn95031_micmode2_enum,
+ SN95031_MICAMP2, 1, sn95031_micmode_text);
static const char *sn95031_dmic_cfg_text[] = {"GPO", "DMIC"};
-static const struct soc_enum sn95031_dmic12_cfg_enum =
- SOC_ENUM_SINGLE(SN95031_DMICMUX, 0, 2, sn95031_dmic_cfg_text);
-static const struct soc_enum sn95031_dmic34_cfg_enum =
- SOC_ENUM_SINGLE(SN95031_DMICMUX, 1, 2, sn95031_dmic_cfg_text);
-static const struct soc_enum sn95031_dmic56_cfg_enum =
- SOC_ENUM_SINGLE(SN95031_DMICMUX, 2, 2, sn95031_dmic_cfg_text);
+static SOC_ENUM_SINGLE_DECL(sn95031_dmic12_cfg_enum,
+ SN95031_DMICMUX, 0, sn95031_dmic_cfg_text);
+static SOC_ENUM_SINGLE_DECL(sn95031_dmic34_cfg_enum,
+ SN95031_DMICMUX, 1, sn95031_dmic_cfg_text);
+static SOC_ENUM_SINGLE_DECL(sn95031_dmic56_cfg_enum,
+ SN95031_DMICMUX, 2, sn95031_dmic_cfg_text);
static const struct snd_kcontrol_new sn95031_snd_controls[] = {
SOC_ENUM("Mic1Mode Capture Route", sn95031_micmode1_enum),
@@ -825,8 +825,6 @@ static int sn95031_codec_probe(struct snd_soc_codec *codec)
{
pr_debug("codec_probe called\n");
- snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP);
-
/* PCM interface config
* This sets the pcm rx slot conguration to max 6 slots
* for max 4 dais (2 stereo and 2 mono)
diff --git a/sound/soc/codecs/ssm2518.c b/sound/soc/codecs/ssm2518.c
index cc8debc..56adb3e 100644
--- a/sound/soc/codecs/ssm2518.c
+++ b/sound/soc/codecs/ssm2518.c
@@ -169,19 +169,19 @@ static const char * const ssm2518_drc_hold_time_text[] = {
"682.24 ms", "1364 ms",
};
-static const SOC_ENUM_SINGLE_DECL(ssm2518_drc_peak_detector_attack_time_enum,
+static SOC_ENUM_SINGLE_DECL(ssm2518_drc_peak_detector_attack_time_enum,
SSM2518_REG_DRC_2, 4, ssm2518_drc_peak_detector_attack_time_text);
-static const SOC_ENUM_SINGLE_DECL(ssm2518_drc_peak_detector_release_time_enum,
+static SOC_ENUM_SINGLE_DECL(ssm2518_drc_peak_detector_release_time_enum,
SSM2518_REG_DRC_2, 0, ssm2518_drc_peak_detector_release_time_text);
-static const SOC_ENUM_SINGLE_DECL(ssm2518_drc_attack_time_enum,
+static SOC_ENUM_SINGLE_DECL(ssm2518_drc_attack_time_enum,
SSM2518_REG_DRC_6, 4, ssm2518_drc_peak_detector_attack_time_text);
-static const SOC_ENUM_SINGLE_DECL(ssm2518_drc_decay_time_enum,
+static SOC_ENUM_SINGLE_DECL(ssm2518_drc_decay_time_enum,
SSM2518_REG_DRC_6, 0, ssm2518_drc_peak_detector_release_time_text);
-static const SOC_ENUM_SINGLE_DECL(ssm2518_drc_hold_time_enum,
+static SOC_ENUM_SINGLE_DECL(ssm2518_drc_hold_time_enum,
SSM2518_REG_DRC_7, 4, ssm2518_drc_hold_time_text);
-static const SOC_ENUM_SINGLE_DECL(ssm2518_drc_noise_gate_hold_time_enum,
+static SOC_ENUM_SINGLE_DECL(ssm2518_drc_noise_gate_hold_time_enum,
SSM2518_REG_DRC_7, 0, ssm2518_drc_hold_time_text);
-static const SOC_ENUM_SINGLE_DECL(ssm2518_drc_rms_averaging_time_enum,
+static SOC_ENUM_SINGLE_DECL(ssm2518_drc_rms_averaging_time_enum,
SSM2518_REG_DRC_9, 0, ssm2518_drc_peak_detector_release_time_text);
static const struct snd_kcontrol_new ssm2518_snd_controls[] = {
@@ -648,16 +648,6 @@ static struct snd_soc_dai_driver ssm2518_dai = {
static int ssm2518_probe(struct snd_soc_codec *codec)
{
- struct ssm2518 *ssm2518 = snd_soc_codec_get_drvdata(codec);
- int ret;
-
- codec->control_data = ssm2518->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;
- }
-
return ssm2518_set_bias_level(codec, SND_SOC_BIAS_OFF);
}
diff --git a/sound/soc/codecs/ssm2602-i2c.c b/sound/soc/codecs/ssm2602-i2c.c
new file mode 100644
index 0000000..abd63d5
--- /dev/null
+++ b/sound/soc/codecs/ssm2602-i2c.c
@@ -0,0 +1,57 @@
+/*
+ * SSM2602/SSM2603/SSM2604 I2C audio driver
+ *
+ * Copyright 2014 Analog Devices Inc.
+ *
+ * Licensed under the GPL-2.
+ */
+
+#include <linux/module.h>
+#include <linux/i2c.h>
+#include <linux/regmap.h>
+
+#include <sound/soc.h>
+
+#include "ssm2602.h"
+
+/*
+ * ssm2602 2 wire address is determined by GPIO5
+ * state during powerup.
+ * low = 0x1a
+ * high = 0x1b
+ */
+static int ssm2602_i2c_probe(struct i2c_client *client,
+ const struct i2c_device_id *id)
+{
+ return ssm2602_probe(&client->dev, id->driver_data,
+ devm_regmap_init_i2c(client, &ssm2602_regmap_config));
+}
+
+static int ssm2602_i2c_remove(struct i2c_client *client)
+{
+ snd_soc_unregister_codec(&client->dev);
+ return 0;
+}
+
+static const struct i2c_device_id ssm2602_i2c_id[] = {
+ { "ssm2602", SSM2602 },
+ { "ssm2603", SSM2602 },
+ { "ssm2604", SSM2604 },
+ { }
+};
+MODULE_DEVICE_TABLE(i2c, ssm2602_i2c_id);
+
+static struct i2c_driver ssm2602_i2c_driver = {
+ .driver = {
+ .name = "ssm2602",
+ .owner = THIS_MODULE,
+ },
+ .probe = ssm2602_i2c_probe,
+ .remove = ssm2602_i2c_remove,
+ .id_table = ssm2602_i2c_id,
+};
+module_i2c_driver(ssm2602_i2c_driver);
+
+MODULE_DESCRIPTION("ASoC SSM2602/SSM2603/SSM2604 I2C driver");
+MODULE_AUTHOR("Cliff Cai");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/ssm2602-spi.c b/sound/soc/codecs/ssm2602-spi.c
new file mode 100644
index 0000000..2bf55e2
--- /dev/null
+++ b/sound/soc/codecs/ssm2602-spi.c
@@ -0,0 +1,41 @@
+/*
+ * SSM2602 SPI audio driver
+ *
+ * Copyright 2014 Analog Devices Inc.
+ *
+ * Licensed under the GPL-2.
+ */
+
+#include <linux/module.h>
+#include <linux/spi/spi.h>
+#include <linux/regmap.h>
+
+#include <sound/soc.h>
+
+#include "ssm2602.h"
+
+static int ssm2602_spi_probe(struct spi_device *spi)
+{
+ return ssm2602_probe(&spi->dev, SSM2602,
+ devm_regmap_init_spi(spi, &ssm2602_regmap_config));
+}
+
+static int ssm2602_spi_remove(struct spi_device *spi)
+{
+ snd_soc_unregister_codec(&spi->dev);
+ return 0;
+}
+
+static struct spi_driver ssm2602_spi_driver = {
+ .driver = {
+ .name = "ssm2602",
+ .owner = THIS_MODULE,
+ },
+ .probe = ssm2602_spi_probe,
+ .remove = ssm2602_spi_remove,
+};
+module_spi_driver(ssm2602_spi_driver);
+
+MODULE_DESCRIPTION("ASoC SSM2602 SPI driver");
+MODULE_AUTHOR("Cliff Cai");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c
index af76bbd..97b0454 100644
--- a/sound/soc/codecs/ssm2602.c
+++ b/sound/soc/codecs/ssm2602.c
@@ -27,32 +27,20 @@
*/
#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/spi/spi.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/initval.h>
#include <sound/tlv.h>
#include "ssm2602.h"
-enum ssm2602_type {
- SSM2602,
- SSM2604,
-};
-
/* codec private data */
struct ssm2602_priv {
unsigned int sysclk;
- struct snd_pcm_hw_constraint_list *sysclk_constraints;
+ const struct snd_pcm_hw_constraint_list *sysclk_constraints;
struct regmap *regmap;
@@ -75,15 +63,16 @@ static const u16 ssm2602_reg[SSM2602_CACHEREGNUM] = {
/*Appending several "None"s just for OSS mixer use*/
static const char *ssm2602_input_select[] = {
- "Line", "Mic", "None", "None", "None",
- "None", "None", "None",
+ "Line", "Mic",
};
static const char *ssm2602_deemph[] = {"None", "32Khz", "44.1Khz", "48Khz"};
static const struct soc_enum ssm2602_enum[] = {
- SOC_ENUM_SINGLE(SSM2602_APANA, 2, 2, ssm2602_input_select),
- SOC_ENUM_SINGLE(SSM2602_APDIGI, 1, 4, ssm2602_deemph),
+ SOC_ENUM_SINGLE(SSM2602_APANA, 2, ARRAY_SIZE(ssm2602_input_select),
+ ssm2602_input_select),
+ SOC_ENUM_SINGLE(SSM2602_APDIGI, 1, ARRAY_SIZE(ssm2602_deemph),
+ ssm2602_deemph),
};
static const unsigned int ssm260x_outmix_tlv[] = {
@@ -197,7 +186,7 @@ static const unsigned int ssm2602_rates_12288000[] = {
8000, 16000, 32000, 48000, 96000,
};
-static struct snd_pcm_hw_constraint_list ssm2602_constraints_12288000 = {
+static const struct snd_pcm_hw_constraint_list ssm2602_constraints_12288000 = {
.list = ssm2602_rates_12288000,
.count = ARRAY_SIZE(ssm2602_rates_12288000),
};
@@ -206,7 +195,7 @@ static const unsigned int ssm2602_rates_11289600[] = {
8000, 44100, 88200,
};
-static struct snd_pcm_hw_constraint_list ssm2602_constraints_11289600 = {
+static const struct snd_pcm_hw_constraint_list ssm2602_constraints_11289600 = {
.list = ssm2602_rates_11289600,
.count = ARRAY_SIZE(ssm2602_rates_11289600),
};
@@ -529,7 +518,7 @@ static int ssm2602_resume(struct snd_soc_codec *codec)
return 0;
}
-static int ssm2602_probe(struct snd_soc_codec *codec)
+static int ssm2602_codec_probe(struct snd_soc_codec *codec)
{
struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec);
struct snd_soc_dapm_context *dapm = &codec->dapm;
@@ -554,7 +543,7 @@ static int ssm2602_probe(struct snd_soc_codec *codec)
ARRAY_SIZE(ssm2602_routes));
}
-static int ssm2604_probe(struct snd_soc_codec *codec)
+static int ssm2604_codec_probe(struct snd_soc_codec *codec)
{
struct snd_soc_dapm_context *dapm = &codec->dapm;
int ret;
@@ -568,18 +557,11 @@ static int ssm2604_probe(struct snd_soc_codec *codec)
ARRAY_SIZE(ssm2604_routes));
}
-static int ssm260x_probe(struct snd_soc_codec *codec)
+static int ssm260x_codec_probe(struct snd_soc_codec *codec)
{
struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec);
int ret;
- 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 = regmap_write(ssm2602->regmap, SSM2602_RESET, 0);
if (ret < 0) {
dev_err(codec->dev, "Failed to issue reset: %d\n", ret);
@@ -597,10 +579,10 @@ static int ssm260x_probe(struct snd_soc_codec *codec)
switch (ssm2602->type) {
case SSM2602:
- ret = ssm2602_probe(codec);
+ ret = ssm2602_codec_probe(codec);
break;
case SSM2604:
- ret = ssm2604_probe(codec);
+ ret = ssm2604_codec_probe(codec);
break;
}
@@ -620,7 +602,7 @@ static int ssm2602_remove(struct snd_soc_codec *codec)
}
static struct snd_soc_codec_driver soc_codec_dev_ssm2602 = {
- .probe = ssm260x_probe,
+ .probe = ssm260x_codec_probe,
.remove = ssm2602_remove,
.suspend = ssm2602_suspend,
.resume = ssm2602_resume,
@@ -639,7 +621,7 @@ static bool ssm2602_register_volatile(struct device *dev, unsigned int reg)
return reg == SSM2602_RESET;
}
-static const struct regmap_config ssm2602_regmap_config = {
+const struct regmap_config ssm2602_regmap_config = {
.val_bits = 9,
.reg_bits = 7,
@@ -650,134 +632,28 @@ static const struct regmap_config ssm2602_regmap_config = {
.reg_defaults_raw = ssm2602_reg,
.num_reg_defaults_raw = ARRAY_SIZE(ssm2602_reg),
};
+EXPORT_SYMBOL_GPL(ssm2602_regmap_config);
-#if defined(CONFIG_SPI_MASTER)
-static int ssm2602_spi_probe(struct spi_device *spi)
+int ssm2602_probe(struct device *dev, enum ssm2602_type type,
+ struct regmap *regmap)
{
struct ssm2602_priv *ssm2602;
- int ret;
-
- ssm2602 = devm_kzalloc(&spi->dev, sizeof(struct ssm2602_priv),
- GFP_KERNEL);
- if (ssm2602 == NULL)
- return -ENOMEM;
-
- spi_set_drvdata(spi, ssm2602);
- ssm2602->type = SSM2602;
- ssm2602->regmap = devm_regmap_init_spi(spi, &ssm2602_regmap_config);
- if (IS_ERR(ssm2602->regmap))
- return PTR_ERR(ssm2602->regmap);
+ if (IS_ERR(regmap))
+ return PTR_ERR(regmap);
- ret = snd_soc_register_codec(&spi->dev,
- &soc_codec_dev_ssm2602, &ssm2602_dai, 1);
- return ret;
-}
-
-static int ssm2602_spi_remove(struct spi_device *spi)
-{
- snd_soc_unregister_codec(&spi->dev);
- return 0;
-}
-
-static struct spi_driver ssm2602_spi_driver = {
- .driver = {
- .name = "ssm2602",
- .owner = THIS_MODULE,
- },
- .probe = ssm2602_spi_probe,
- .remove = ssm2602_spi_remove,
-};
-#endif
-
-#if IS_ENABLED(CONFIG_I2C)
-/*
- * ssm2602 2 wire address is determined by GPIO5
- * state during powerup.
- * low = 0x1a
- * high = 0x1b
- */
-static int ssm2602_i2c_probe(struct i2c_client *i2c,
- const struct i2c_device_id *id)
-{
- struct ssm2602_priv *ssm2602;
- int ret;
-
- ssm2602 = devm_kzalloc(&i2c->dev, sizeof(struct ssm2602_priv),
- GFP_KERNEL);
+ ssm2602 = devm_kzalloc(dev, sizeof(*ssm2602), GFP_KERNEL);
if (ssm2602 == NULL)
return -ENOMEM;
- i2c_set_clientdata(i2c, ssm2602);
- 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;
-}
-
-static int ssm2602_i2c_remove(struct i2c_client *client)
-{
- snd_soc_unregister_codec(&client->dev);
- return 0;
-}
-
-static const struct i2c_device_id ssm2602_i2c_id[] = {
- { "ssm2602", SSM2602 },
- { "ssm2603", SSM2602 },
- { "ssm2604", SSM2604 },
- { }
-};
-MODULE_DEVICE_TABLE(i2c, ssm2602_i2c_id);
-
-/* corgi i2c codec control layer */
-static struct i2c_driver ssm2602_i2c_driver = {
- .driver = {
- .name = "ssm2602",
- .owner = THIS_MODULE,
- },
- .probe = ssm2602_i2c_probe,
- .remove = ssm2602_i2c_remove,
- .id_table = ssm2602_i2c_id,
-};
-#endif
-
-
-static int __init ssm2602_modinit(void)
-{
- int ret = 0;
-
-#if defined(CONFIG_SPI_MASTER)
- ret = spi_register_driver(&ssm2602_spi_driver);
- if (ret)
- return ret;
-#endif
-
-#if IS_ENABLED(CONFIG_I2C)
- ret = i2c_add_driver(&ssm2602_i2c_driver);
- if (ret)
- return ret;
-#endif
-
- return ret;
-}
-module_init(ssm2602_modinit);
-
-static void __exit ssm2602_exit(void)
-{
-#if defined(CONFIG_SPI_MASTER)
- spi_unregister_driver(&ssm2602_spi_driver);
-#endif
+ dev_set_drvdata(dev, ssm2602);
+ ssm2602->type = SSM2602;
+ ssm2602->regmap = regmap;
-#if IS_ENABLED(CONFIG_I2C)
- i2c_del_driver(&ssm2602_i2c_driver);
-#endif
+ return snd_soc_register_codec(dev, &soc_codec_dev_ssm2602,
+ &ssm2602_dai, 1);
}
-module_exit(ssm2602_exit);
+EXPORT_SYMBOL_GPL(ssm2602_probe);
MODULE_DESCRIPTION("ASoC SSM2602/SSM2603/SSM2604 driver");
MODULE_AUTHOR("Cliff Cai");
diff --git a/sound/soc/codecs/ssm2602.h b/sound/soc/codecs/ssm2602.h
index fbd07d7..7475388 100644
--- a/sound/soc/codecs/ssm2602.h
+++ b/sound/soc/codecs/ssm2602.h
@@ -28,6 +28,20 @@
#ifndef _SSM2602_H
#define _SSM2602_H
+#include <linux/regmap.h>
+
+struct device;
+
+enum ssm2602_type {
+ SSM2602,
+ SSM2604,
+};
+
+extern const struct regmap_config ssm2602_regmap_config;
+
+int ssm2602_probe(struct device *dev, enum ssm2602_type type,
+ struct regmap *regmap);
+
/* SSM2602 Codec Register definitions */
#define SSM2602_LINVOL 0x00
diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c
index 2735361..1257774 100644
--- a/sound/soc/codecs/sta32x.c
+++ b/sound/soc/codecs/sta32x.c
@@ -872,16 +872,6 @@ static int sta32x_probe(struct snd_soc_codec *codec)
return ret;
}
- /* Tell ASoC what kind of I/O to use to read the registers. ASoC will
- * then do the I2C transactions itself.
- */
- codec->control_data = sta32x->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 (ret=%i)\n", ret);
- goto err;
- }
-
/* Chip documentation explicitly requires that the reset values
* of reserved register bits are left untouched.
* Write the register default value to cache for reserved registers,
@@ -946,10 +936,6 @@ static int sta32x_probe(struct snd_soc_codec *codec)
regulator_bulk_disable(ARRAY_SIZE(sta32x->supplies), sta32x->supplies);
return 0;
-
-err:
- regulator_bulk_disable(ARRAY_SIZE(sta32x->supplies), sta32x->supplies);
- return ret;
}
static int sta32x_remove(struct snd_soc_codec *codec)
diff --git a/sound/soc/codecs/sta529.c b/sound/soc/codecs/sta529.c
index 40c07be..a40c4b0 100644
--- a/sound/soc/codecs/sta529.c
+++ b/sound/soc/codecs/sta529.c
@@ -141,7 +141,7 @@ static const char *pwm_mode_text[] = { "Binary", "Headphone", "Ternary",
static const DECLARE_TLV_DB_SCALE(out_gain_tlv, -9150, 50, 0);
static const DECLARE_TLV_DB_SCALE(master_vol_tlv, -12750, 50, 0);
-static const SOC_ENUM_SINGLE_DECL(pwm_src, STA529_FFXCFG1, 4, pwm_mode_text);
+static SOC_ENUM_SINGLE_DECL(pwm_src, STA529_FFXCFG1, 4, pwm_mode_text);
static const struct snd_kcontrol_new sta529_snd_controls[] = {
SOC_DOUBLE_R_TLV("Digital Playback Volume", STA529_LVOL, STA529_RVOL, 0,
@@ -193,8 +193,7 @@ static int sta529_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_codec *codec = dai->codec;
int pdata, play_freq_val, record_freq_val;
int bclk_to_fs_ratio;
@@ -322,16 +321,6 @@ static struct snd_soc_dai_driver sta529_dai = {
static int sta529_probe(struct snd_soc_codec *codec)
{
- struct sta529 *sta529 = snd_soc_codec_get_drvdata(codec);
- int ret;
-
- codec->control_data = sta529->regmap;
- ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP);
-
- if (ret < 0) {
- dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
- return ret;
- }
sta529_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
return 0;
diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c
index a5455c1..53b810d 100644
--- a/sound/soc/codecs/stac9766.c
+++ b/sound/soc/codecs/stac9766.c
@@ -62,25 +62,25 @@ static const char *stac9766_boost1[] = {"0dB", "10dB"};
static const char *stac9766_boost2[] = {"0dB", "20dB"};
static const char *stac9766_stereo_mic[] = {"Off", "On"};
-static const struct soc_enum stac9766_record_enum =
- SOC_ENUM_DOUBLE(AC97_REC_SEL, 8, 0, 8, stac9766_record_mux);
-static const struct soc_enum stac9766_mono_enum =
- SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 9, 2, stac9766_mono_mux);
-static const struct soc_enum stac9766_mic_enum =
- SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 8, 2, stac9766_mic_mux);
-static const struct soc_enum stac9766_SPDIF_enum =
- SOC_ENUM_SINGLE(AC97_STAC_DA_CONTROL, 1, 2, stac9766_SPDIF_mux);
-static const struct soc_enum stac9766_popbypass_enum =
- SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 15, 2, stac9766_popbypass_mux);
-static const struct soc_enum stac9766_record_all_enum =
- SOC_ENUM_SINGLE(AC97_STAC_ANALOG_SPECIAL, 12, 2,
- stac9766_record_all_mux);
-static const struct soc_enum stac9766_boost1_enum =
- SOC_ENUM_SINGLE(AC97_MIC, 6, 2, stac9766_boost1); /* 0/10dB */
-static const struct soc_enum stac9766_boost2_enum =
- SOC_ENUM_SINGLE(AC97_STAC_ANALOG_SPECIAL, 2, 2, stac9766_boost2); /* 0/20dB */
-static const struct soc_enum stac9766_stereo_mic_enum =
- SOC_ENUM_SINGLE(AC97_STAC_STEREO_MIC, 2, 1, stac9766_stereo_mic);
+static SOC_ENUM_DOUBLE_DECL(stac9766_record_enum,
+ AC97_REC_SEL, 8, 0, stac9766_record_mux);
+static SOC_ENUM_SINGLE_DECL(stac9766_mono_enum,
+ AC97_GENERAL_PURPOSE, 9, stac9766_mono_mux);
+static SOC_ENUM_SINGLE_DECL(stac9766_mic_enum,
+ AC97_GENERAL_PURPOSE, 8, stac9766_mic_mux);
+static SOC_ENUM_SINGLE_DECL(stac9766_SPDIF_enum,
+ AC97_STAC_DA_CONTROL, 1, stac9766_SPDIF_mux);
+static SOC_ENUM_SINGLE_DECL(stac9766_popbypass_enum,
+ AC97_GENERAL_PURPOSE, 15, stac9766_popbypass_mux);
+static SOC_ENUM_SINGLE_DECL(stac9766_record_all_enum,
+ AC97_STAC_ANALOG_SPECIAL, 12,
+ stac9766_record_all_mux);
+static SOC_ENUM_SINGLE_DECL(stac9766_boost1_enum,
+ AC97_MIC, 6, stac9766_boost1); /* 0/10dB */
+static SOC_ENUM_SINGLE_DECL(stac9766_boost2_enum,
+ AC97_STAC_ANALOG_SPECIAL, 2, stac9766_boost2); /* 0/20dB */
+static SOC_ENUM_SINGLE_DECL(stac9766_stereo_mic_enum,
+ AC97_STAC_STEREO_MIC, 2, stac9766_stereo_mic);
static const DECLARE_TLV_DB_LINEAR(master_tlv, -4600, 0);
static const DECLARE_TLV_DB_LINEAR(record_tlv, 0, 2250);
diff --git a/sound/soc/codecs/tlv320aic23-i2c.c b/sound/soc/codecs/tlv320aic23-i2c.c
new file mode 100644
index 0000000..20fc460
--- /dev/null
+++ b/sound/soc/codecs/tlv320aic23-i2c.c
@@ -0,0 +1,59 @@
+/*
+ * ALSA SoC TLV320AIC23 codec driver I2C interface
+ *
+ * Author: Arun KS, <arunks@mistralsolutions.com>
+ * Copyright: (C) 2008 Mistral Solutions Pvt Ltd.,
+ *
+ * Based on sound/soc/codecs/wm8731.c by Richard Purdie
+ *
+ * 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/i2c.h>
+#include <linux/module.h>
+#include <linux/regmap.h>
+#include <sound/soc.h>
+
+#include "tlv320aic23.h"
+
+static int tlv320aic23_i2c_probe(struct i2c_client *i2c,
+ const struct i2c_device_id *i2c_id)
+{
+ struct regmap *regmap;
+
+ if (!i2c_check_functionality(i2c->adapter, I2C_FUNC_SMBUS_BYTE_DATA))
+ return -EINVAL;
+
+ regmap = devm_regmap_init_i2c(i2c, &tlv320aic23_regmap);
+ return tlv320aic23_probe(&i2c->dev, regmap);
+}
+
+static int __exit tlv320aic23_i2c_remove(struct i2c_client *i2c)
+{
+ snd_soc_unregister_codec(&i2c->dev);
+ return 0;
+}
+
+static const struct i2c_device_id tlv320aic23_id[] = {
+ {"tlv320aic23", 0},
+ {}
+};
+
+MODULE_DEVICE_TABLE(i2c, tlv320aic23_id);
+
+static struct i2c_driver tlv320aic23_i2c_driver = {
+ .driver = {
+ .name = "tlv320aic23-codec",
+ },
+ .probe = tlv320aic23_i2c_probe,
+ .remove = __exit_p(tlv320aic23_i2c_remove),
+ .id_table = tlv320aic23_id,
+};
+
+module_i2c_driver(tlv320aic23_i2c_driver);
+
+MODULE_DESCRIPTION("ASoC TLV320AIC23 codec driver I2C");
+MODULE_AUTHOR("Arun KS <arunks@mistralsolutions.com>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/tlv320aic23-spi.c b/sound/soc/codecs/tlv320aic23-spi.c
new file mode 100644
index 0000000..3b387e4
--- /dev/null
+++ b/sound/soc/codecs/tlv320aic23-spi.c
@@ -0,0 +1,56 @@
+/*
+ * ALSA SoC TLV320AIC23 codec driver SPI interface
+ *
+ * Author: Arun KS, <arunks@mistralsolutions.com>
+ * Copyright: (C) 2008 Mistral Solutions Pvt Ltd.,
+ *
+ * Based on sound/soc/codecs/wm8731.c by Richard Purdie
+ *
+ * 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/regmap.h>
+#include <linux/spi/spi.h>
+#include <sound/soc.h>
+
+#include "tlv320aic23.h"
+
+static int aic23_spi_probe(struct spi_device *spi)
+{
+ int ret;
+ struct regmap *regmap;
+
+ dev_dbg(&spi->dev, "probing tlv320aic23 spi device\n");
+
+ spi->mode = SPI_MODE_0;
+ ret = spi_setup(spi);
+ if (ret < 0)
+ return ret;
+
+ regmap = devm_regmap_init_spi(spi, &tlv320aic23_regmap);
+ return tlv320aic23_probe(&spi->dev, regmap);
+}
+
+static int aic23_spi_remove(struct spi_device *spi)
+{
+ snd_soc_unregister_codec(&spi->dev);
+ return 0;
+}
+
+static struct spi_driver aic23_spi = {
+ .driver = {
+ .name = "tlv320aic23",
+ .owner = THIS_MODULE,
+ },
+ .probe = aic23_spi_probe,
+ .remove = aic23_spi_remove,
+};
+
+module_spi_driver(aic23_spi);
+
+MODULE_DESCRIPTION("ASoC TLV320AIC23 codec driver SPI");
+MODULE_AUTHOR("Arun KS <arunks@mistralsolutions.com>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c
index 5d430cc..20864ee 100644
--- a/sound/soc/codecs/tlv320aic23.c
+++ b/sound/soc/codecs/tlv320aic23.c
@@ -23,7 +23,6 @@
#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>
@@ -51,7 +50,7 @@ static const struct reg_default tlv320aic23_reg[] = {
{ 9, 0x0000 },
};
-static const struct regmap_config tlv320aic23_regmap = {
+const struct regmap_config tlv320aic23_regmap = {
.reg_bits = 7,
.val_bits = 9,
@@ -60,20 +59,21 @@ static const struct regmap_config tlv320aic23_regmap = {
.num_reg_defaults = ARRAY_SIZE(tlv320aic23_reg),
.cache_type = REGCACHE_RBTREE,
};
+EXPORT_SYMBOL(tlv320aic23_regmap);
static const char *rec_src_text[] = { "Line", "Mic" };
static const char *deemph_text[] = {"None", "32Khz", "44.1Khz", "48Khz"};
-static const struct soc_enum rec_src_enum =
- SOC_ENUM_SINGLE(TLV320AIC23_ANLG, 2, 2, rec_src_text);
+static SOC_ENUM_SINGLE_DECL(rec_src_enum,
+ TLV320AIC23_ANLG, 2, rec_src_text);
static const struct snd_kcontrol_new tlv320aic23_rec_src_mux_controls =
SOC_DAPM_ENUM("Input Select", rec_src_enum);
-static const struct soc_enum tlv320aic23_rec_src =
- SOC_ENUM_SINGLE(TLV320AIC23_ANLG, 2, 2, rec_src_text);
-static const struct soc_enum tlv320aic23_deemph =
- SOC_ENUM_SINGLE(TLV320AIC23_DIGT, 1, 4, deemph_text);
+static SOC_ENUM_SINGLE_DECL(tlv320aic23_rec_src,
+ TLV320AIC23_ANLG, 2, rec_src_text);
+static SOC_ENUM_SINGLE_DECL(tlv320aic23_deemph,
+ TLV320AIC23_DIGT, 1, deemph_text);
static const DECLARE_TLV_DB_SCALE(out_gain_tlv, -12100, 100, 0);
static const DECLARE_TLV_DB_SCALE(input_gain_tlv, -1725, 75, 0);
@@ -400,7 +400,7 @@ static void tlv320aic23_shutdown(struct snd_pcm_substream *substream,
struct aic23 *aic23 = snd_soc_codec_get_drvdata(codec);
/* deactivate */
- if (!codec->active) {
+ if (!snd_soc_codec_is_active(codec)) {
udelay(50);
snd_soc_write(codec, TLV320AIC23_ACTIVE, 0x0);
}
@@ -557,16 +557,8 @@ static int tlv320aic23_resume(struct snd_soc_codec *codec)
return 0;
}
-static int tlv320aic23_probe(struct snd_soc_codec *codec)
+static int tlv320aic23_codec_probe(struct snd_soc_codec *codec)
{
- int ret;
-
- 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;
- }
-
/* Reset codec */
snd_soc_write(codec, TLV320AIC23_RESET, 0);
@@ -604,7 +596,7 @@ static int tlv320aic23_remove(struct snd_soc_codec *codec)
}
static struct snd_soc_codec_driver soc_codec_dev_tlv320aic23 = {
- .probe = tlv320aic23_probe,
+ .probe = tlv320aic23_codec_probe,
.remove = tlv320aic23_remove,
.suspend = tlv320aic23_suspend,
.resume = tlv320aic23_resume,
@@ -617,56 +609,25 @@ static struct snd_soc_codec_driver soc_codec_dev_tlv320aic23 = {
.num_dapm_routes = ARRAY_SIZE(tlv320aic23_intercon),
};
-/*
- * If the i2c layer weren't so broken, we could pass this kind of data
- * around
- */
-static int tlv320aic23_codec_probe(struct i2c_client *i2c,
- const struct i2c_device_id *i2c_id)
+int tlv320aic23_probe(struct device *dev, struct regmap *regmap)
{
struct aic23 *aic23;
- int ret;
- if (!i2c_check_functionality(i2c->adapter, I2C_FUNC_SMBUS_BYTE_DATA))
- return -EINVAL;
+ if (IS_ERR(regmap))
+ return PTR_ERR(regmap);
- aic23 = devm_kzalloc(&i2c->dev, sizeof(struct aic23), GFP_KERNEL);
+ aic23 = devm_kzalloc(dev, sizeof(struct aic23), GFP_KERNEL);
if (aic23 == NULL)
return -ENOMEM;
- aic23->regmap = devm_regmap_init_i2c(i2c, &tlv320aic23_regmap);
- if (IS_ERR(aic23->regmap))
- return PTR_ERR(aic23->regmap);
+ aic23->regmap = regmap;
- i2c_set_clientdata(i2c, aic23);
+ dev_set_drvdata(dev, aic23);
- ret = snd_soc_register_codec(&i2c->dev,
- &soc_codec_dev_tlv320aic23, &tlv320aic23_dai, 1);
- return ret;
+ return snd_soc_register_codec(dev, &soc_codec_dev_tlv320aic23,
+ &tlv320aic23_dai, 1);
}
-static int __exit tlv320aic23_i2c_remove(struct i2c_client *i2c)
-{
- snd_soc_unregister_codec(&i2c->dev);
- return 0;
-}
-
-static const struct i2c_device_id tlv320aic23_id[] = {
- {"tlv320aic23", 0},
- {}
-};
-
-MODULE_DEVICE_TABLE(i2c, tlv320aic23_id);
-
-static struct i2c_driver tlv320aic23_i2c_driver = {
- .driver = {
- .name = "tlv320aic23-codec",
- },
- .probe = tlv320aic23_codec_probe,
- .remove = __exit_p(tlv320aic23_i2c_remove),
- .id_table = tlv320aic23_id,
-};
-
-module_i2c_driver(tlv320aic23_i2c_driver);
+EXPORT_SYMBOL(tlv320aic23_probe);
MODULE_DESCRIPTION("ASoC TLV320AIC23 codec driver");
MODULE_AUTHOR("Arun KS <arunks@mistralsolutions.com>");
diff --git a/sound/soc/codecs/tlv320aic23.h b/sound/soc/codecs/tlv320aic23.h
index e804120..3a7235a 100644
--- a/sound/soc/codecs/tlv320aic23.h
+++ b/sound/soc/codecs/tlv320aic23.h
@@ -12,6 +12,12 @@
#ifndef _TLV320AIC23_H
#define _TLV320AIC23_H
+struct device;
+struct regmap_config;
+
+extern const struct regmap_config tlv320aic23_regmap;
+int tlv320aic23_probe(struct device *dev, struct regmap *regmap);
+
/* Codec TLV320AIC23 */
#define TLV320AIC23_LINVOL 0x00
#define TLV320AIC23_RINVOL 0x01
diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c
index 94a658f..43069de 100644
--- a/sound/soc/codecs/tlv320aic26.c
+++ b/sound/soc/codecs/tlv320aic26.c
@@ -238,8 +238,9 @@ static struct snd_soc_dai_driver aic26_dai = {
* ALSA controls
*/
static const char *aic26_capture_src_text[] = {"Mic", "Aux"};
-static const struct soc_enum aic26_capture_src_enum =
- SOC_ENUM_SINGLE(AIC26_REG_AUDIO_CTRL1, 12, 2, aic26_capture_src_text);
+static SOC_ENUM_SINGLE_DECL(aic26_capture_src_enum,
+ AIC26_REG_AUDIO_CTRL1, 12,
+ aic26_capture_src_text);
static const struct snd_kcontrol_new aic26_snd_controls[] = {
/* Output */
@@ -295,8 +296,6 @@ static int aic26_probe(struct snd_soc_codec *codec)
struct aic26 *aic26 = dev_get_drvdata(codec->dev);
int ret, reg;
- snd_soc_codec_set_cache_io(codec, 16, 16, SND_SOC_REGMAP);
-
aic26->codec = codec;
/* Reset the codec to power on defaults */
diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c
new file mode 100644
index 0000000..fa158cf
--- /dev/null
+++ b/sound/soc/codecs/tlv320aic31xx.c
@@ -0,0 +1,1280 @@
+/*
+ * ALSA SoC TLV320AIC31XX codec driver
+ *
+ * Copyright (C) 2014 Texas Instruments, Inc.
+ *
+ * Author: Jyri Sarha <jsarha@ti.com>
+ *
+ * Based on ground work by: Ajit Kulkarni <x0175765@ti.com>
+ *
+ * This package 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 PACKAGE IS PROVIDED AS IS AND WITHOUT ANY EXPRESS OR
+ * IMPLIED WARRANTIES, INCLUDING, WITHOUT LIMITATION, THE IMPLIED
+ * WARRANTIES OF MERCHANTIBILITY AND FITNESS FOR A PARTICULAR PURPOSE.
+ *
+ * The TLV320AIC31xx series of audio codec is a low-power, highly integrated
+ * high performance codec which provides a stereo DAC, a mono ADC,
+ * and mono/stereo Class-D speaker driver.
+ */
+
+#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/gpio.h>
+#include <linux/regulator/consumer.h>
+#include <linux/of_gpio.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/initval.h>
+#include <sound/tlv.h>
+#include <dt-bindings/sound/tlv320aic31xx-micbias.h>
+
+#include "tlv320aic31xx.h"
+
+static const struct reg_default aic31xx_reg_defaults[] = {
+ { AIC31XX_CLKMUX, 0x00 },
+ { AIC31XX_PLLPR, 0x11 },
+ { AIC31XX_PLLJ, 0x04 },
+ { AIC31XX_PLLDMSB, 0x00 },
+ { AIC31XX_PLLDLSB, 0x00 },
+ { AIC31XX_NDAC, 0x01 },
+ { AIC31XX_MDAC, 0x01 },
+ { AIC31XX_DOSRMSB, 0x00 },
+ { AIC31XX_DOSRLSB, 0x80 },
+ { AIC31XX_NADC, 0x01 },
+ { AIC31XX_MADC, 0x01 },
+ { AIC31XX_AOSR, 0x80 },
+ { AIC31XX_IFACE1, 0x00 },
+ { AIC31XX_DATA_OFFSET, 0x00 },
+ { AIC31XX_IFACE2, 0x00 },
+ { AIC31XX_BCLKN, 0x01 },
+ { AIC31XX_DACSETUP, 0x14 },
+ { AIC31XX_DACMUTE, 0x0c },
+ { AIC31XX_LDACVOL, 0x00 },
+ { AIC31XX_RDACVOL, 0x00 },
+ { AIC31XX_ADCSETUP, 0x00 },
+ { AIC31XX_ADCFGA, 0x80 },
+ { AIC31XX_ADCVOL, 0x00 },
+ { AIC31XX_HPDRIVER, 0x04 },
+ { AIC31XX_SPKAMP, 0x06 },
+ { AIC31XX_DACMIXERROUTE, 0x00 },
+ { AIC31XX_LANALOGHPL, 0x7f },
+ { AIC31XX_RANALOGHPR, 0x7f },
+ { AIC31XX_LANALOGSPL, 0x7f },
+ { AIC31XX_RANALOGSPR, 0x7f },
+ { AIC31XX_HPLGAIN, 0x02 },
+ { AIC31XX_HPRGAIN, 0x02 },
+ { AIC31XX_SPLGAIN, 0x00 },
+ { AIC31XX_SPRGAIN, 0x00 },
+ { AIC31XX_MICBIAS, 0x00 },
+ { AIC31XX_MICPGA, 0x80 },
+ { AIC31XX_MICPGAPI, 0x00 },
+ { AIC31XX_MICPGAMI, 0x00 },
+};
+
+static bool aic31xx_volatile(struct device *dev, unsigned int reg)
+{
+ switch (reg) {
+ case AIC31XX_PAGECTL: /* regmap implementation requires this */
+ case AIC31XX_RESET: /* always clears after write */
+ case AIC31XX_OT_FLAG:
+ case AIC31XX_ADCFLAG:
+ case AIC31XX_DACFLAG1:
+ case AIC31XX_DACFLAG2:
+ case AIC31XX_OFFLAG: /* Sticky interrupt flags */
+ case AIC31XX_INTRDACFLAG: /* Sticky interrupt flags */
+ case AIC31XX_INTRADCFLAG: /* Sticky interrupt flags */
+ case AIC31XX_INTRDACFLAG2:
+ case AIC31XX_INTRADCFLAG2:
+ return true;
+ }
+ return false;
+}
+
+static bool aic31xx_writeable(struct device *dev, unsigned int reg)
+{
+ switch (reg) {
+ case AIC31XX_OT_FLAG:
+ case AIC31XX_ADCFLAG:
+ case AIC31XX_DACFLAG1:
+ case AIC31XX_DACFLAG2:
+ case AIC31XX_OFFLAG: /* Sticky interrupt flags */
+ case AIC31XX_INTRDACFLAG: /* Sticky interrupt flags */
+ case AIC31XX_INTRADCFLAG: /* Sticky interrupt flags */
+ case AIC31XX_INTRDACFLAG2:
+ case AIC31XX_INTRADCFLAG2:
+ return false;
+ }
+ return true;
+}
+
+static const struct regmap_range_cfg aic31xx_ranges[] = {
+ {
+ .range_min = 0,
+ .range_max = 12 * 128,
+ .selector_reg = AIC31XX_PAGECTL,
+ .selector_mask = 0xff,
+ .selector_shift = 0,
+ .window_start = 0,
+ .window_len = 128,
+ },
+};
+
+static const struct regmap_config aic31xx_i2c_regmap = {
+ .reg_bits = 8,
+ .val_bits = 8,
+ .writeable_reg = aic31xx_writeable,
+ .volatile_reg = aic31xx_volatile,
+ .reg_defaults = aic31xx_reg_defaults,
+ .num_reg_defaults = ARRAY_SIZE(aic31xx_reg_defaults),
+ .cache_type = REGCACHE_RBTREE,
+ .ranges = aic31xx_ranges,
+ .num_ranges = ARRAY_SIZE(aic31xx_ranges),
+ .max_register = 12 * 128,
+};
+
+#define AIC31XX_NUM_SUPPLIES 6
+static const char * const aic31xx_supply_names[AIC31XX_NUM_SUPPLIES] = {
+ "HPVDD",
+ "SPRVDD",
+ "SPLVDD",
+ "AVDD",
+ "IOVDD",
+ "DVDD",
+};
+
+struct aic31xx_disable_nb {
+ struct notifier_block nb;
+ struct aic31xx_priv *aic31xx;
+};
+
+struct aic31xx_priv {
+ struct snd_soc_codec *codec;
+ u8 i2c_regs_status;
+ struct device *dev;
+ struct regmap *regmap;
+ struct aic31xx_pdata pdata;
+ struct regulator_bulk_data supplies[AIC31XX_NUM_SUPPLIES];
+ struct aic31xx_disable_nb disable_nb[AIC31XX_NUM_SUPPLIES];
+ unsigned int sysclk;
+ int rate_div_line;
+};
+
+struct aic31xx_rate_divs {
+ u32 mclk;
+ u32 rate;
+ u8 p_val;
+ u8 pll_j;
+ u16 pll_d;
+ u16 dosr;
+ u8 ndac;
+ u8 mdac;
+ u8 aosr;
+ u8 nadc;
+ u8 madc;
+};
+
+/* ADC dividers can be disabled by cofiguring them to 0 */
+static const struct aic31xx_rate_divs aic31xx_divs[] = {
+ /* mclk rate pll: p j d dosr ndac mdac aors nadc madc */
+ /* 8k rate */
+ {12000000, 8000, 1, 8, 1920, 128, 48, 2, 128, 48, 2},
+ {24000000, 8000, 2, 8, 1920, 128, 48, 2, 128, 48, 2},
+ {25000000, 8000, 2, 7, 8643, 128, 48, 2, 128, 48, 2},
+ /* 11.025k rate */
+ {12000000, 11025, 1, 7, 5264, 128, 32, 2, 128, 32, 2},
+ {24000000, 11025, 2, 7, 5264, 128, 32, 2, 128, 32, 2},
+ {25000000, 11025, 2, 7, 2253, 128, 32, 2, 128, 32, 2},
+ /* 16k rate */
+ {12000000, 16000, 1, 8, 1920, 128, 24, 2, 128, 24, 2},
+ {24000000, 16000, 2, 8, 1920, 128, 24, 2, 128, 24, 2},
+ {25000000, 16000, 2, 7, 8643, 128, 24, 2, 128, 24, 2},
+ /* 22.05k rate */
+ {12000000, 22050, 1, 7, 5264, 128, 16, 2, 128, 16, 2},
+ {24000000, 22050, 2, 7, 5264, 128, 16, 2, 128, 16, 2},
+ {25000000, 22050, 2, 7, 2253, 128, 16, 2, 128, 16, 2},
+ /* 32k rate */
+ {12000000, 32000, 1, 8, 1920, 128, 12, 2, 128, 12, 2},
+ {24000000, 32000, 2, 8, 1920, 128, 12, 2, 128, 12, 2},
+ {25000000, 32000, 2, 7, 8643, 128, 12, 2, 128, 12, 2},
+ /* 44.1k rate */
+ {12000000, 44100, 1, 7, 5264, 128, 8, 2, 128, 8, 2},
+ {24000000, 44100, 2, 7, 5264, 128, 8, 2, 128, 8, 2},
+ {25000000, 44100, 2, 7, 2253, 128, 8, 2, 128, 8, 2},
+ /* 48k rate */
+ {12000000, 48000, 1, 8, 1920, 128, 8, 2, 128, 8, 2},
+ {24000000, 48000, 2, 8, 1920, 128, 8, 2, 128, 8, 2},
+ {25000000, 48000, 2, 7, 8643, 128, 8, 2, 128, 8, 2},
+ /* 88.2k rate */
+ {12000000, 88200, 1, 7, 5264, 64, 8, 2, 64, 8, 2},
+ {24000000, 88200, 2, 7, 5264, 64, 8, 2, 64, 8, 2},
+ {25000000, 88200, 2, 7, 2253, 64, 8, 2, 64, 8, 2},
+ /* 96k rate */
+ {12000000, 96000, 1, 8, 1920, 64, 8, 2, 64, 8, 2},
+ {24000000, 96000, 2, 8, 1920, 64, 8, 2, 64, 8, 2},
+ {25000000, 96000, 2, 7, 8643, 64, 8, 2, 64, 8, 2},
+ /* 176.4k rate */
+ {12000000, 176400, 1, 7, 5264, 32, 8, 2, 32, 8, 2},
+ {24000000, 176400, 2, 7, 5264, 32, 8, 2, 32, 8, 2},
+ {25000000, 176400, 2, 7, 2253, 32, 8, 2, 32, 8, 2},
+ /* 192k rate */
+ {12000000, 192000, 1, 8, 1920, 32, 8, 2, 32, 8, 2},
+ {24000000, 192000, 2, 8, 1920, 32, 8, 2, 32, 8, 2},
+ {25000000, 192000, 2, 7, 8643, 32, 8, 2, 32, 8, 2},
+};
+
+static const char * const ldac_in_text[] = {
+ "Off", "Left Data", "Right Data", "Mono"
+};
+
+static const char * const rdac_in_text[] = {
+ "Off", "Right Data", "Left Data", "Mono"
+};
+
+static SOC_ENUM_SINGLE_DECL(ldac_in_enum, AIC31XX_DACSETUP, 4, ldac_in_text);
+
+static SOC_ENUM_SINGLE_DECL(rdac_in_enum, AIC31XX_DACSETUP, 2, rdac_in_text);
+
+static const char * const mic_select_text[] = {
+ "Off", "FFR 10 Ohm", "FFR 20 Ohm", "FFR 40 Ohm"
+};
+
+static const
+SOC_ENUM_SINGLE_DECL(mic1lp_p_enum, AIC31XX_MICPGAPI, 6, mic_select_text);
+static const
+SOC_ENUM_SINGLE_DECL(mic1rp_p_enum, AIC31XX_MICPGAPI, 4, mic_select_text);
+static const
+SOC_ENUM_SINGLE_DECL(mic1lm_p_enum, AIC31XX_MICPGAPI, 2, mic_select_text);
+
+static const
+SOC_ENUM_SINGLE_DECL(cm_m_enum, AIC31XX_MICPGAMI, 6, mic_select_text);
+static const
+SOC_ENUM_SINGLE_DECL(mic1lm_m_enum, AIC31XX_MICPGAMI, 4, mic_select_text);
+
+static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -6350, 50, 0);
+static const DECLARE_TLV_DB_SCALE(adc_fgain_tlv, 0, 10, 0);
+static const DECLARE_TLV_DB_SCALE(adc_cgain_tlv, -2000, 50, 0);
+static const DECLARE_TLV_DB_SCALE(mic_pga_tlv, 0, 50, 0);
+static const DECLARE_TLV_DB_SCALE(hp_drv_tlv, 0, 100, 0);
+static const DECLARE_TLV_DB_SCALE(class_D_drv_tlv, 600, 600, 0);
+static const DECLARE_TLV_DB_SCALE(hp_vol_tlv, -6350, 50, 0);
+static const DECLARE_TLV_DB_SCALE(sp_vol_tlv, -6350, 50, 0);
+
+/*
+ * controls to be exported to the user space
+ */
+static const struct snd_kcontrol_new aic31xx_snd_controls[] = {
+ SOC_DOUBLE_R_S_TLV("DAC Playback Volume", AIC31XX_LDACVOL,
+ AIC31XX_RDACVOL, 0, -127, 48, 7, 0, dac_vol_tlv),
+
+ SOC_SINGLE_TLV("ADC Fine Capture Volume", AIC31XX_ADCFGA, 4, 4, 1,
+ adc_fgain_tlv),
+
+ SOC_SINGLE("ADC Capture Switch", AIC31XX_ADCFGA, 7, 1, 1),
+ SOC_DOUBLE_R_S_TLV("ADC Capture Volume", AIC31XX_ADCVOL, AIC31XX_ADCVOL,
+ 0, -24, 40, 6, 0, adc_cgain_tlv),
+
+ SOC_SINGLE_TLV("Mic PGA Capture Volume", AIC31XX_MICPGA, 0,
+ 119, 0, mic_pga_tlv),
+
+ SOC_DOUBLE_R("HP Driver Playback Switch", AIC31XX_HPLGAIN,
+ AIC31XX_HPRGAIN, 2, 1, 0),
+ SOC_DOUBLE_R_TLV("HP Driver Playback Volume", AIC31XX_HPLGAIN,
+ AIC31XX_HPRGAIN, 3, 0x09, 0, hp_drv_tlv),
+
+ SOC_DOUBLE_R_TLV("HP Analog Playback Volume", AIC31XX_LANALOGHPL,
+ AIC31XX_RANALOGHPR, 0, 0x7F, 1, hp_vol_tlv),
+};
+
+static const struct snd_kcontrol_new aic311x_snd_controls[] = {
+ SOC_DOUBLE_R("Speaker Driver Playback Switch", AIC31XX_SPLGAIN,
+ AIC31XX_SPRGAIN, 2, 1, 0),
+ SOC_DOUBLE_R_TLV("Speaker Driver Playback Volume", AIC31XX_SPLGAIN,
+ AIC31XX_SPRGAIN, 3, 3, 0, class_D_drv_tlv),
+
+ SOC_DOUBLE_R_TLV("Speaker Analog Playback Volume", AIC31XX_LANALOGSPL,
+ AIC31XX_RANALOGSPR, 0, 0x7F, 1, sp_vol_tlv),
+};
+
+static const struct snd_kcontrol_new aic310x_snd_controls[] = {
+ SOC_SINGLE("Speaker Driver Playback Switch", AIC31XX_SPLGAIN,
+ 2, 1, 0),
+ SOC_SINGLE_TLV("Speaker Driver Playback Volume", AIC31XX_SPLGAIN,
+ 3, 3, 0, class_D_drv_tlv),
+
+ SOC_SINGLE_TLV("Speaker Analog Playback Volume", AIC31XX_LANALOGSPL,
+ 0, 0x7F, 1, sp_vol_tlv),
+};
+
+static const struct snd_kcontrol_new ldac_in_control =
+ SOC_DAPM_ENUM("DAC Left Input", ldac_in_enum);
+
+static const struct snd_kcontrol_new rdac_in_control =
+ SOC_DAPM_ENUM("DAC Right Input", rdac_in_enum);
+
+static int aic31xx_wait_bits(struct aic31xx_priv *aic31xx, unsigned int reg,
+ unsigned int mask, unsigned int wbits, int sleep,
+ int count)
+{
+ unsigned int bits;
+ int counter = count;
+ int ret = regmap_read(aic31xx->regmap, reg, &bits);
+ while ((bits & mask) != wbits && counter && !ret) {
+ usleep_range(sleep, sleep * 2);
+ ret = regmap_read(aic31xx->regmap, reg, &bits);
+ counter--;
+ }
+ if ((bits & mask) != wbits) {
+ dev_err(aic31xx->dev,
+ "%s: Failed! 0x%x was 0x%x expected 0x%x (%d, 0x%x, %d us)\n",
+ __func__, reg, bits, wbits, ret, mask,
+ (count - counter) * sleep);
+ ret = -1;
+ }
+ return ret;
+}
+
+#define WIDGET_BIT(reg, shift) (((shift) << 8) | (reg))
+
+static int aic31xx_dapm_power_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(w->codec);
+ unsigned int reg = AIC31XX_DACFLAG1;
+ unsigned int mask;
+
+ switch (WIDGET_BIT(w->reg, w->shift)) {
+ case WIDGET_BIT(AIC31XX_DACSETUP, 7):
+ mask = AIC31XX_LDACPWRSTATUS_MASK;
+ break;
+ case WIDGET_BIT(AIC31XX_DACSETUP, 6):
+ mask = AIC31XX_RDACPWRSTATUS_MASK;
+ break;
+ case WIDGET_BIT(AIC31XX_HPDRIVER, 7):
+ mask = AIC31XX_HPLDRVPWRSTATUS_MASK;
+ break;
+ case WIDGET_BIT(AIC31XX_HPDRIVER, 6):
+ mask = AIC31XX_HPRDRVPWRSTATUS_MASK;
+ break;
+ case WIDGET_BIT(AIC31XX_SPKAMP, 7):
+ mask = AIC31XX_SPLDRVPWRSTATUS_MASK;
+ break;
+ case WIDGET_BIT(AIC31XX_SPKAMP, 6):
+ mask = AIC31XX_SPRDRVPWRSTATUS_MASK;
+ break;
+ case WIDGET_BIT(AIC31XX_ADCSETUP, 7):
+ mask = AIC31XX_ADCPWRSTATUS_MASK;
+ reg = AIC31XX_ADCFLAG;
+ break;
+ default:
+ dev_err(w->codec->dev, "Unknown widget '%s' calling %s/n",
+ w->name, __func__);
+ return -EINVAL;
+ }
+
+ switch (event) {
+ case SND_SOC_DAPM_POST_PMU:
+ return aic31xx_wait_bits(aic31xx, reg, mask, mask, 5000, 100);
+ case SND_SOC_DAPM_POST_PMD:
+ return aic31xx_wait_bits(aic31xx, reg, mask, 0, 5000, 100);
+ default:
+ dev_dbg(w->codec->dev,
+ "Unhandled dapm widget event %d from %s\n",
+ event, w->name);
+ }
+ return 0;
+}
+
+static const struct snd_kcontrol_new left_output_switches[] = {
+ SOC_DAPM_SINGLE("From Left DAC", AIC31XX_DACMIXERROUTE, 6, 1, 0),
+ SOC_DAPM_SINGLE("From MIC1LP", AIC31XX_DACMIXERROUTE, 5, 1, 0),
+ SOC_DAPM_SINGLE("From MIC1RP", AIC31XX_DACMIXERROUTE, 4, 1, 0),
+};
+
+static const struct snd_kcontrol_new right_output_switches[] = {
+ SOC_DAPM_SINGLE("From Right DAC", AIC31XX_DACMIXERROUTE, 2, 1, 0),
+ SOC_DAPM_SINGLE("From MIC1RP", AIC31XX_DACMIXERROUTE, 1, 1, 0),
+};
+
+static const struct snd_kcontrol_new p_term_mic1lp =
+ SOC_DAPM_ENUM("MIC1LP P-Terminal", mic1lp_p_enum);
+
+static const struct snd_kcontrol_new p_term_mic1rp =
+ SOC_DAPM_ENUM("MIC1RP P-Terminal", mic1rp_p_enum);
+
+static const struct snd_kcontrol_new p_term_mic1lm =
+ SOC_DAPM_ENUM("MIC1LM P-Terminal", mic1lm_p_enum);
+
+static const struct snd_kcontrol_new m_term_mic1lm =
+ SOC_DAPM_ENUM("MIC1LM M-Terminal", mic1lm_m_enum);
+
+static const struct snd_kcontrol_new aic31xx_dapm_hpl_switch =
+ SOC_DAPM_SINGLE("Switch", AIC31XX_LANALOGHPL, 7, 1, 0);
+
+static const struct snd_kcontrol_new aic31xx_dapm_hpr_switch =
+ SOC_DAPM_SINGLE("Switch", AIC31XX_RANALOGHPR, 7, 1, 0);
+
+static const struct snd_kcontrol_new aic31xx_dapm_spl_switch =
+ SOC_DAPM_SINGLE("Switch", AIC31XX_LANALOGSPL, 7, 1, 0);
+
+static const struct snd_kcontrol_new aic31xx_dapm_spr_switch =
+ SOC_DAPM_SINGLE("Switch", AIC31XX_RANALOGSPR, 7, 1, 0);
+
+static int mic_bias_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_codec *codec = w->codec;
+ struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec);
+ switch (event) {
+ case SND_SOC_DAPM_POST_PMU:
+ /* change mic bias voltage to user defined */
+ snd_soc_update_bits(codec, AIC31XX_MICBIAS,
+ AIC31XX_MICBIAS_MASK,
+ aic31xx->pdata.micbias_vg <<
+ AIC31XX_MICBIAS_SHIFT);
+ dev_dbg(codec->dev, "%s: turned on\n", __func__);
+ break;
+ case SND_SOC_DAPM_PRE_PMD:
+ /* turn mic bias off */
+ snd_soc_update_bits(codec, AIC31XX_MICBIAS,
+ AIC31XX_MICBIAS_MASK, 0);
+ dev_dbg(codec->dev, "%s: turned off\n", __func__);
+ break;
+ }
+ return 0;
+}
+
+static const struct snd_soc_dapm_widget aic31xx_dapm_widgets[] = {
+ SND_SOC_DAPM_AIF_IN("DAC IN", "DAC Playback", 0, SND_SOC_NOPM, 0, 0),
+
+ SND_SOC_DAPM_MUX("DAC Left Input",
+ SND_SOC_NOPM, 0, 0, &ldac_in_control),
+ SND_SOC_DAPM_MUX("DAC Right Input",
+ SND_SOC_NOPM, 0, 0, &rdac_in_control),
+ /* DACs */
+ SND_SOC_DAPM_DAC_E("DAC Left", "Left Playback",
+ AIC31XX_DACSETUP, 7, 0, aic31xx_dapm_power_event,
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+
+ SND_SOC_DAPM_DAC_E("DAC Right", "Right Playback",
+ AIC31XX_DACSETUP, 6, 0, aic31xx_dapm_power_event,
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+
+ /* Output Mixers */
+ SND_SOC_DAPM_MIXER("Output Left", SND_SOC_NOPM, 0, 0,
+ left_output_switches,
+ ARRAY_SIZE(left_output_switches)),
+ SND_SOC_DAPM_MIXER("Output Right", SND_SOC_NOPM, 0, 0,
+ right_output_switches,
+ ARRAY_SIZE(right_output_switches)),
+
+ SND_SOC_DAPM_SWITCH("HP Left", SND_SOC_NOPM, 0, 0,
+ &aic31xx_dapm_hpl_switch),
+ SND_SOC_DAPM_SWITCH("HP Right", SND_SOC_NOPM, 0, 0,
+ &aic31xx_dapm_hpr_switch),
+
+ /* Output drivers */
+ SND_SOC_DAPM_OUT_DRV_E("HPL Driver", AIC31XX_HPDRIVER, 7, 0,
+ NULL, 0, aic31xx_dapm_power_event,
+ SND_SOC_DAPM_POST_PMD | SND_SOC_DAPM_POST_PMU),
+ SND_SOC_DAPM_OUT_DRV_E("HPR Driver", AIC31XX_HPDRIVER, 6, 0,
+ NULL, 0, aic31xx_dapm_power_event,
+ SND_SOC_DAPM_POST_PMD | SND_SOC_DAPM_POST_PMU),
+
+ /* ADC */
+ SND_SOC_DAPM_ADC_E("ADC", "Capture", AIC31XX_ADCSETUP, 7, 0,
+ aic31xx_dapm_power_event, SND_SOC_DAPM_POST_PMU |
+ SND_SOC_DAPM_POST_PMD),
+
+ /* Input Selection to MIC_PGA */
+ SND_SOC_DAPM_MUX("MIC1LP P-Terminal", SND_SOC_NOPM, 0, 0,
+ &p_term_mic1lp),
+ SND_SOC_DAPM_MUX("MIC1RP P-Terminal", SND_SOC_NOPM, 0, 0,
+ &p_term_mic1rp),
+ SND_SOC_DAPM_MUX("MIC1LM P-Terminal", SND_SOC_NOPM, 0, 0,
+ &p_term_mic1lm),
+
+ SND_SOC_DAPM_MUX("MIC1LM M-Terminal", SND_SOC_NOPM, 0, 0,
+ &m_term_mic1lm),
+ /* Enabling & Disabling MIC Gain Ctl */
+ SND_SOC_DAPM_PGA("MIC_GAIN_CTL", AIC31XX_MICPGA,
+ 7, 1, NULL, 0),
+
+ /* Mic Bias */
+ SND_SOC_DAPM_SUPPLY("MICBIAS", SND_SOC_NOPM, 0, 0, mic_bias_event,
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
+
+ /* Outputs */
+ SND_SOC_DAPM_OUTPUT("HPL"),
+ SND_SOC_DAPM_OUTPUT("HPR"),
+
+ /* Inputs */
+ SND_SOC_DAPM_INPUT("MIC1LP"),
+ SND_SOC_DAPM_INPUT("MIC1RP"),
+ SND_SOC_DAPM_INPUT("MIC1LM"),
+};
+
+static const struct snd_soc_dapm_widget aic311x_dapm_widgets[] = {
+ /* AIC3111 and AIC3110 have stereo class-D amplifier */
+ SND_SOC_DAPM_OUT_DRV_E("SPL ClassD", AIC31XX_SPKAMP, 7, 0, NULL, 0,
+ aic31xx_dapm_power_event, SND_SOC_DAPM_POST_PMU |
+ SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_OUT_DRV_E("SPR ClassD", AIC31XX_SPKAMP, 6, 0, NULL, 0,
+ aic31xx_dapm_power_event, SND_SOC_DAPM_POST_PMU |
+ SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_SWITCH("Speaker Left", SND_SOC_NOPM, 0, 0,
+ &aic31xx_dapm_spl_switch),
+ SND_SOC_DAPM_SWITCH("Speaker Right", SND_SOC_NOPM, 0, 0,
+ &aic31xx_dapm_spr_switch),
+ SND_SOC_DAPM_OUTPUT("SPL"),
+ SND_SOC_DAPM_OUTPUT("SPR"),
+};
+
+/* AIC3100 and AIC3120 have only mono class-D amplifier */
+static const struct snd_soc_dapm_widget aic310x_dapm_widgets[] = {
+ SND_SOC_DAPM_OUT_DRV_E("SPK ClassD", AIC31XX_SPKAMP, 7, 0, NULL, 0,
+ aic31xx_dapm_power_event, SND_SOC_DAPM_POST_PMU |
+ SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_SWITCH("Speaker", SND_SOC_NOPM, 0, 0,
+ &aic31xx_dapm_spl_switch),
+ SND_SOC_DAPM_OUTPUT("SPK"),
+};
+
+static const struct snd_soc_dapm_route
+aic31xx_audio_map[] = {
+ /* DAC Input Routing */
+ {"DAC Left Input", "Left Data", "DAC IN"},
+ {"DAC Left Input", "Right Data", "DAC IN"},
+ {"DAC Left Input", "Mono", "DAC IN"},
+ {"DAC Right Input", "Left Data", "DAC IN"},
+ {"DAC Right Input", "Right Data", "DAC IN"},
+ {"DAC Right Input", "Mono", "DAC IN"},
+ {"DAC Left", NULL, "DAC Left Input"},
+ {"DAC Right", NULL, "DAC Right Input"},
+
+ /* Mic input */
+ {"MIC1LP P-Terminal", "FFR 10 Ohm", "MIC1LP"},
+ {"MIC1LP P-Terminal", "FFR 20 Ohm", "MIC1LP"},
+ {"MIC1LP P-Terminal", "FFR 40 Ohm", "MIC1LP"},
+ {"MIC1RP P-Terminal", "FFR 10 Ohm", "MIC1RP"},
+ {"MIC1RP P-Terminal", "FFR 20 Ohm", "MIC1RP"},
+ {"MIC1RP P-Terminal", "FFR 40 Ohm", "MIC1RP"},
+ {"MIC1LM P-Terminal", "FFR 10 Ohm", "MIC1LM"},
+ {"MIC1LM P-Terminal", "FFR 20 Ohm", "MIC1LM"},
+ {"MIC1LM P-Terminal", "FFR 40 Ohm", "MIC1LM"},
+
+ {"MIC1LM M-Terminal", "FFR 10 Ohm", "MIC1LM"},
+ {"MIC1LM M-Terminal", "FFR 20 Ohm", "MIC1LM"},
+ {"MIC1LM M-Terminal", "FFR 40 Ohm", "MIC1LM"},
+
+ {"MIC_GAIN_CTL", NULL, "MIC1LP P-Terminal"},
+ {"MIC_GAIN_CTL", NULL, "MIC1RP P-Terminal"},
+ {"MIC_GAIN_CTL", NULL, "MIC1LM P-Terminal"},
+ {"MIC_GAIN_CTL", NULL, "MIC1LM M-Terminal"},
+
+ {"ADC", NULL, "MIC_GAIN_CTL"},
+
+ /* Left Output */
+ {"Output Left", "From Left DAC", "DAC Left"},
+ {"Output Left", "From MIC1LP", "MIC1LP"},
+ {"Output Left", "From MIC1RP", "MIC1RP"},
+
+ /* Right Output */
+ {"Output Right", "From Right DAC", "DAC Right"},
+ {"Output Right", "From MIC1RP", "MIC1RP"},
+
+ /* HPL path */
+ {"HP Left", "Switch", "Output Left"},
+ {"HPL Driver", NULL, "HP Left"},
+ {"HPL", NULL, "HPL Driver"},
+
+ /* HPR path */
+ {"HP Right", "Switch", "Output Right"},
+ {"HPR Driver", NULL, "HP Right"},
+ {"HPR", NULL, "HPR Driver"},
+};
+
+static const struct snd_soc_dapm_route
+aic311x_audio_map[] = {
+ /* SP L path */
+ {"Speaker Left", "Switch", "Output Left"},
+ {"SPL ClassD", NULL, "Speaker Left"},
+ {"SPL", NULL, "SPL ClassD"},
+
+ /* SP R path */
+ {"Speaker Right", "Switch", "Output Right"},
+ {"SPR ClassD", NULL, "Speaker Right"},
+ {"SPR", NULL, "SPR ClassD"},
+};
+
+static const struct snd_soc_dapm_route
+aic310x_audio_map[] = {
+ /* SP L path */
+ {"Speaker", "Switch", "Output Left"},
+ {"SPK ClassD", NULL, "Speaker"},
+ {"SPK", NULL, "SPK ClassD"},
+};
+
+static int aic31xx_add_controls(struct snd_soc_codec *codec)
+{
+ int ret = 0;
+ struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec);
+
+ if (aic31xx->pdata.codec_type & AIC31XX_STEREO_CLASS_D_BIT)
+ ret = snd_soc_add_codec_controls(
+ codec, aic311x_snd_controls,
+ ARRAY_SIZE(aic311x_snd_controls));
+ else
+ ret = snd_soc_add_codec_controls(
+ codec, aic310x_snd_controls,
+ ARRAY_SIZE(aic310x_snd_controls));
+
+ return ret;
+}
+
+static int aic31xx_add_widgets(struct snd_soc_codec *codec)
+{
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
+ struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec);
+ int ret = 0;
+
+ if (aic31xx->pdata.codec_type & AIC31XX_STEREO_CLASS_D_BIT) {
+ ret = snd_soc_dapm_new_controls(
+ dapm, aic311x_dapm_widgets,
+ ARRAY_SIZE(aic311x_dapm_widgets));
+ if (ret)
+ return ret;
+
+ ret = snd_soc_dapm_add_routes(dapm, aic311x_audio_map,
+ ARRAY_SIZE(aic311x_audio_map));
+ if (ret)
+ return ret;
+ } else {
+ ret = snd_soc_dapm_new_controls(
+ dapm, aic310x_dapm_widgets,
+ ARRAY_SIZE(aic310x_dapm_widgets));
+ if (ret)
+ return ret;
+
+ ret = snd_soc_dapm_add_routes(dapm, aic310x_audio_map,
+ ARRAY_SIZE(aic310x_audio_map));
+ if (ret)
+ return ret;
+ }
+
+ return 0;
+}
+
+static int aic31xx_setup_pll(struct snd_soc_codec *codec,
+ struct snd_pcm_hw_params *params)
+{
+ struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec);
+ int bclk_n = 0;
+ int i;
+
+ /* Use PLL as CODEC_CLKIN and DAC_CLK as BDIV_CLKIN */
+ snd_soc_update_bits(codec, AIC31XX_CLKMUX,
+ AIC31XX_CODEC_CLKIN_MASK, AIC31XX_CODEC_CLKIN_PLL);
+ snd_soc_update_bits(codec, AIC31XX_IFACE2,
+ AIC31XX_BDIVCLK_MASK, AIC31XX_DAC2BCLK);
+
+ for (i = 0; i < ARRAY_SIZE(aic31xx_divs); i++) {
+ if (aic31xx_divs[i].rate == params_rate(params) &&
+ aic31xx_divs[i].mclk == aic31xx->sysclk)
+ break;
+ }
+
+ if (i == ARRAY_SIZE(aic31xx_divs)) {
+ dev_err(codec->dev, "%s: Sampling rate %u not supported\n",
+ __func__, params_rate(params));
+ return -EINVAL;
+ }
+
+ /* PLL configuration */
+ snd_soc_update_bits(codec, AIC31XX_PLLPR, AIC31XX_PLL_MASK,
+ (aic31xx_divs[i].p_val << 4) | 0x01);
+ snd_soc_write(codec, AIC31XX_PLLJ, aic31xx_divs[i].pll_j);
+
+ snd_soc_write(codec, AIC31XX_PLLDMSB,
+ aic31xx_divs[i].pll_d >> 8);
+ snd_soc_write(codec, AIC31XX_PLLDLSB,
+ aic31xx_divs[i].pll_d & 0xff);
+
+ /* DAC dividers configuration */
+ snd_soc_update_bits(codec, AIC31XX_NDAC, AIC31XX_PLL_MASK,
+ aic31xx_divs[i].ndac);
+ snd_soc_update_bits(codec, AIC31XX_MDAC, AIC31XX_PLL_MASK,
+ aic31xx_divs[i].mdac);
+
+ snd_soc_write(codec, AIC31XX_DOSRMSB, aic31xx_divs[i].dosr >> 8);
+ snd_soc_write(codec, AIC31XX_DOSRLSB, aic31xx_divs[i].dosr & 0xff);
+
+ /* ADC dividers configuration. Write reset value 1 if not used. */
+ snd_soc_update_bits(codec, AIC31XX_NADC, AIC31XX_PLL_MASK,
+ aic31xx_divs[i].nadc ? aic31xx_divs[i].nadc : 1);
+ snd_soc_update_bits(codec, AIC31XX_MADC, AIC31XX_PLL_MASK,
+ aic31xx_divs[i].madc ? aic31xx_divs[i].madc : 1);
+
+ snd_soc_write(codec, AIC31XX_AOSR, aic31xx_divs[i].aosr);
+
+ /* Bit clock divider configuration. */
+ bclk_n = (aic31xx_divs[i].dosr * aic31xx_divs[i].mdac)
+ / snd_soc_params_to_frame_size(params);
+ if (bclk_n == 0) {
+ dev_err(codec->dev, "%s: Not enough BLCK bandwidth\n",
+ __func__);
+ return -EINVAL;
+ }
+
+ snd_soc_update_bits(codec, AIC31XX_BCLKN,
+ AIC31XX_PLL_MASK, bclk_n);
+
+ aic31xx->rate_div_line = i;
+
+ dev_dbg(codec->dev,
+ "pll %d.%04d/%d dosr %d n %d m %d aosr %d n %d m %d bclk_n %d\n",
+ aic31xx_divs[i].pll_j, aic31xx_divs[i].pll_d,
+ aic31xx_divs[i].p_val, aic31xx_divs[i].dosr,
+ aic31xx_divs[i].ndac, aic31xx_divs[i].mdac,
+ aic31xx_divs[i].aosr, aic31xx_divs[i].nadc,
+ aic31xx_divs[i].madc, bclk_n);
+
+ return 0;
+}
+
+static int aic31xx_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;
+ u8 data = 0;
+
+ dev_dbg(codec->dev, "## %s: format %d width %d rate %d\n",
+ __func__, params_format(params), params_width(params),
+ params_rate(params));
+
+ switch (params_width(params)) {
+ case 16:
+ break;
+ case 20:
+ data = (AIC31XX_WORD_LEN_20BITS <<
+ AIC31XX_IFACE1_DATALEN_SHIFT);
+ break;
+ case 24:
+ data = (AIC31XX_WORD_LEN_24BITS <<
+ AIC31XX_IFACE1_DATALEN_SHIFT);
+ break;
+ case 32:
+ data = (AIC31XX_WORD_LEN_32BITS <<
+ AIC31XX_IFACE1_DATALEN_SHIFT);
+ break;
+ default:
+ dev_err(codec->dev, "%s: Unsupported format %d\n",
+ __func__, params_format(params));
+ return -EINVAL;
+ }
+
+ snd_soc_update_bits(codec, AIC31XX_IFACE1,
+ AIC31XX_IFACE1_DATALEN_MASK,
+ data);
+
+ return aic31xx_setup_pll(codec, params);
+}
+
+static int aic31xx_dac_mute(struct snd_soc_dai *codec_dai, int mute)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+
+ if (mute) {
+ snd_soc_update_bits(codec, AIC31XX_DACMUTE,
+ AIC31XX_DACMUTE_MASK,
+ AIC31XX_DACMUTE_MASK);
+ } else {
+ snd_soc_update_bits(codec, AIC31XX_DACMUTE,
+ AIC31XX_DACMUTE_MASK, 0x0);
+ }
+
+ return 0;
+}
+
+static int aic31xx_set_dai_fmt(struct snd_soc_dai *codec_dai,
+ unsigned int fmt)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ u8 iface_reg1 = 0;
+ u8 iface_reg3 = 0;
+ u8 dsp_a_val = 0;
+
+ dev_dbg(codec->dev, "## %s: fmt = 0x%x\n", __func__, fmt);
+
+ /* set master/slave audio interface */
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBM_CFM:
+ iface_reg1 |= AIC31XX_BCLK_MASTER | AIC31XX_WCLK_MASTER;
+ break;
+ default:
+ dev_alert(codec->dev, "Invalid DAI master/slave interface\n");
+ return -EINVAL;
+ }
+
+ /* interface format */
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ break;
+ case SND_SOC_DAIFMT_DSP_A:
+ dsp_a_val = 0x1;
+ case SND_SOC_DAIFMT_DSP_B:
+ /* NOTE: BCLKINV bit value 1 equas NB and 0 equals IB */
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_NB_NF:
+ iface_reg3 |= AIC31XX_BCLKINV_MASK;
+ break;
+ case SND_SOC_DAIFMT_IB_NF:
+ break;
+ default:
+ return -EINVAL;
+ }
+ iface_reg1 |= (AIC31XX_DSP_MODE <<
+ AIC31XX_IFACE1_DATATYPE_SHIFT);
+ break;
+ case SND_SOC_DAIFMT_RIGHT_J:
+ iface_reg1 |= (AIC31XX_RIGHT_JUSTIFIED_MODE <<
+ AIC31XX_IFACE1_DATATYPE_SHIFT);
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ iface_reg1 |= (AIC31XX_LEFT_JUSTIFIED_MODE <<
+ AIC31XX_IFACE1_DATATYPE_SHIFT);
+ break;
+ default:
+ dev_err(codec->dev, "Invalid DAI interface format\n");
+ return -EINVAL;
+ }
+
+ snd_soc_update_bits(codec, AIC31XX_IFACE1,
+ AIC31XX_IFACE1_DATATYPE_MASK |
+ AIC31XX_IFACE1_MASTER_MASK,
+ iface_reg1);
+ snd_soc_update_bits(codec, AIC31XX_DATA_OFFSET,
+ AIC31XX_DATA_OFFSET_MASK,
+ dsp_a_val);
+ snd_soc_update_bits(codec, AIC31XX_IFACE2,
+ AIC31XX_BCLKINV_MASK,
+ iface_reg3);
+
+ return 0;
+}
+
+static int aic31xx_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 aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec);
+ int i;
+
+ dev_dbg(codec->dev, "## %s: clk_id = %d, freq = %d, dir = %d\n",
+ __func__, clk_id, freq, dir);
+
+ for (i = 0; aic31xx_divs[i].mclk != freq; i++) {
+ if (i == ARRAY_SIZE(aic31xx_divs)) {
+ dev_err(aic31xx->dev, "%s: Unsupported frequency %d\n",
+ __func__, freq);
+ return -EINVAL;
+ }
+ }
+
+ /* set clock on MCLK, BCLK, or GPIO1 as PLL input */
+ snd_soc_update_bits(codec, AIC31XX_CLKMUX, AIC31XX_PLL_CLKIN_MASK,
+ clk_id << AIC31XX_PLL_CLKIN_SHIFT);
+
+ aic31xx->sysclk = freq;
+ return 0;
+}
+
+static int aic31xx_regulator_event(struct notifier_block *nb,
+ unsigned long event, void *data)
+{
+ struct aic31xx_disable_nb *disable_nb =
+ container_of(nb, struct aic31xx_disable_nb, nb);
+ struct aic31xx_priv *aic31xx = disable_nb->aic31xx;
+
+ if (event & REGULATOR_EVENT_DISABLE) {
+ /*
+ * Put codec to reset and as at least one of the
+ * supplies was disabled.
+ */
+ if (gpio_is_valid(aic31xx->pdata.gpio_reset))
+ gpio_set_value(aic31xx->pdata.gpio_reset, 0);
+
+ regcache_mark_dirty(aic31xx->regmap);
+ dev_dbg(aic31xx->dev, "## %s: DISABLE received\n", __func__);
+ }
+
+ return 0;
+}
+
+static void aic31xx_clk_on(struct snd_soc_codec *codec)
+{
+ struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec);
+ u8 mask = AIC31XX_PM_MASK;
+ u8 on = AIC31XX_PM_MASK;
+
+ dev_dbg(codec->dev, "codec clock -> on (rate %d)\n",
+ aic31xx_divs[aic31xx->rate_div_line].rate);
+ snd_soc_update_bits(codec, AIC31XX_PLLPR, mask, on);
+ mdelay(10);
+ snd_soc_update_bits(codec, AIC31XX_NDAC, mask, on);
+ snd_soc_update_bits(codec, AIC31XX_MDAC, mask, on);
+ if (aic31xx_divs[aic31xx->rate_div_line].nadc)
+ snd_soc_update_bits(codec, AIC31XX_NADC, mask, on);
+ if (aic31xx_divs[aic31xx->rate_div_line].madc)
+ snd_soc_update_bits(codec, AIC31XX_MADC, mask, on);
+ snd_soc_update_bits(codec, AIC31XX_BCLKN, mask, on);
+}
+
+static void aic31xx_clk_off(struct snd_soc_codec *codec)
+{
+ u8 mask = AIC31XX_PM_MASK;
+ u8 off = 0;
+
+ dev_dbg(codec->dev, "codec clock -> off\n");
+ snd_soc_update_bits(codec, AIC31XX_BCLKN, mask, off);
+ snd_soc_update_bits(codec, AIC31XX_MADC, mask, off);
+ snd_soc_update_bits(codec, AIC31XX_NADC, mask, off);
+ snd_soc_update_bits(codec, AIC31XX_MDAC, mask, off);
+ snd_soc_update_bits(codec, AIC31XX_NDAC, mask, off);
+ snd_soc_update_bits(codec, AIC31XX_PLLPR, mask, off);
+}
+
+static int aic31xx_power_on(struct snd_soc_codec *codec)
+{
+ struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec);
+ int ret = 0;
+
+ ret = regulator_bulk_enable(ARRAY_SIZE(aic31xx->supplies),
+ aic31xx->supplies);
+ if (ret)
+ return ret;
+
+ if (gpio_is_valid(aic31xx->pdata.gpio_reset)) {
+ gpio_set_value(aic31xx->pdata.gpio_reset, 1);
+ udelay(100);
+ }
+ regcache_cache_only(aic31xx->regmap, false);
+ ret = regcache_sync(aic31xx->regmap);
+ if (ret != 0) {
+ dev_err(codec->dev,
+ "Failed to restore cache: %d\n", ret);
+ regcache_cache_only(aic31xx->regmap, true);
+ regulator_bulk_disable(ARRAY_SIZE(aic31xx->supplies),
+ aic31xx->supplies);
+ return ret;
+ }
+ return 0;
+}
+
+static int aic31xx_power_off(struct snd_soc_codec *codec)
+{
+ struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec);
+ int ret = 0;
+
+ regcache_cache_only(aic31xx->regmap, true);
+ ret = regulator_bulk_disable(ARRAY_SIZE(aic31xx->supplies),
+ aic31xx->supplies);
+
+ return ret;
+}
+
+static int aic31xx_set_bias_level(struct snd_soc_codec *codec,
+ enum snd_soc_bias_level level)
+{
+ dev_dbg(codec->dev, "## %s: %d -> %d\n", __func__,
+ codec->dapm.bias_level, level);
+
+ switch (level) {
+ case SND_SOC_BIAS_ON:
+ break;
+ case SND_SOC_BIAS_PREPARE:
+ if (codec->dapm.bias_level == SND_SOC_BIAS_STANDBY)
+ aic31xx_clk_on(codec);
+ break;
+ case SND_SOC_BIAS_STANDBY:
+ switch (codec->dapm.bias_level) {
+ case SND_SOC_BIAS_OFF:
+ aic31xx_power_on(codec);
+ break;
+ case SND_SOC_BIAS_PREPARE:
+ aic31xx_clk_off(codec);
+ break;
+ default:
+ BUG();
+ }
+ break;
+ case SND_SOC_BIAS_OFF:
+ if (codec->dapm.bias_level == SND_SOC_BIAS_STANDBY)
+ aic31xx_power_off(codec);
+ break;
+ }
+ codec->dapm.bias_level = level;
+
+ return 0;
+}
+
+static int aic31xx_suspend(struct snd_soc_codec *codec)
+{
+ aic31xx_set_bias_level(codec, SND_SOC_BIAS_OFF);
+ return 0;
+}
+
+static int aic31xx_resume(struct snd_soc_codec *codec)
+{
+ aic31xx_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+ return 0;
+}
+
+static int aic31xx_codec_probe(struct snd_soc_codec *codec)
+{
+ int ret = 0;
+ struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec);
+ int i;
+
+ dev_dbg(aic31xx->dev, "## %s\n", __func__);
+
+ aic31xx = snd_soc_codec_get_drvdata(codec);
+
+ aic31xx->codec = codec;
+
+ for (i = 0; i < ARRAY_SIZE(aic31xx->supplies); i++) {
+ aic31xx->disable_nb[i].nb.notifier_call =
+ aic31xx_regulator_event;
+ aic31xx->disable_nb[i].aic31xx = aic31xx;
+ ret = regulator_register_notifier(aic31xx->supplies[i].consumer,
+ &aic31xx->disable_nb[i].nb);
+ if (ret) {
+ dev_err(codec->dev,
+ "Failed to request regulator notifier: %d\n",
+ ret);
+ return ret;
+ }
+ }
+
+ regcache_cache_only(aic31xx->regmap, true);
+ regcache_mark_dirty(aic31xx->regmap);
+
+ ret = aic31xx_add_controls(codec);
+ if (ret)
+ return ret;
+
+ ret = aic31xx_add_widgets(codec);
+
+ return ret;
+}
+
+static int aic31xx_codec_remove(struct snd_soc_codec *codec)
+{
+ struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec);
+ int i;
+ /* power down chip */
+ aic31xx_set_bias_level(codec, SND_SOC_BIAS_OFF);
+
+ for (i = 0; i < ARRAY_SIZE(aic31xx->supplies); i++)
+ regulator_unregister_notifier(aic31xx->supplies[i].consumer,
+ &aic31xx->disable_nb[i].nb);
+
+ return 0;
+}
+
+static struct snd_soc_codec_driver soc_codec_driver_aic31xx = {
+ .probe = aic31xx_codec_probe,
+ .remove = aic31xx_codec_remove,
+ .suspend = aic31xx_suspend,
+ .resume = aic31xx_resume,
+ .set_bias_level = aic31xx_set_bias_level,
+ .controls = aic31xx_snd_controls,
+ .num_controls = ARRAY_SIZE(aic31xx_snd_controls),
+ .dapm_widgets = aic31xx_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(aic31xx_dapm_widgets),
+ .dapm_routes = aic31xx_audio_map,
+ .num_dapm_routes = ARRAY_SIZE(aic31xx_audio_map),
+};
+
+static struct snd_soc_dai_ops aic31xx_dai_ops = {
+ .hw_params = aic31xx_hw_params,
+ .set_sysclk = aic31xx_set_dai_sysclk,
+ .set_fmt = aic31xx_set_dai_fmt,
+ .digital_mute = aic31xx_dac_mute,
+};
+
+static struct snd_soc_dai_driver aic31xx_dai_driver[] = {
+ {
+ .name = "tlv320aic31xx-hifi",
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = AIC31XX_RATES,
+ .formats = AIC31XX_FORMATS,
+ },
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = AIC31XX_RATES,
+ .formats = AIC31XX_FORMATS,
+ },
+ .ops = &aic31xx_dai_ops,
+ .symmetric_rates = 1,
+ }
+};
+
+#if defined(CONFIG_OF)
+static const struct of_device_id tlv320aic31xx_of_match[] = {
+ { .compatible = "ti,tlv320aic310x" },
+ { .compatible = "ti,tlv320aic311x" },
+ { .compatible = "ti,tlv320aic3100" },
+ { .compatible = "ti,tlv320aic3110" },
+ { .compatible = "ti,tlv320aic3120" },
+ { .compatible = "ti,tlv320aic3111" },
+ {},
+};
+MODULE_DEVICE_TABLE(of, tlv320aic31xx_of_match);
+
+static void aic31xx_pdata_from_of(struct aic31xx_priv *aic31xx)
+{
+ struct device_node *np = aic31xx->dev->of_node;
+ unsigned int value = MICBIAS_2_0V;
+ int ret;
+
+ of_property_read_u32(np, "ai31xx-micbias-vg", &value);
+ switch (value) {
+ case MICBIAS_2_0V:
+ case MICBIAS_2_5V:
+ case MICBIAS_AVDDV:
+ aic31xx->pdata.micbias_vg = value;
+ break;
+ default:
+ dev_err(aic31xx->dev,
+ "Bad ai31xx-micbias-vg value %d DT\n",
+ value);
+ aic31xx->pdata.micbias_vg = MICBIAS_2_0V;
+ }
+
+ ret = of_get_named_gpio(np, "gpio-reset", 0);
+ if (ret > 0)
+ aic31xx->pdata.gpio_reset = ret;
+}
+#else /* CONFIG_OF */
+static void aic31xx_pdata_from_of(struct aic31xx_priv *aic31xx)
+{
+}
+#endif /* CONFIG_OF */
+
+static void aic31xx_device_init(struct aic31xx_priv *aic31xx)
+{
+ int ret, i;
+
+ dev_set_drvdata(aic31xx->dev, aic31xx);
+
+ if (dev_get_platdata(aic31xx->dev))
+ memcpy(&aic31xx->pdata, dev_get_platdata(aic31xx->dev),
+ sizeof(aic31xx->pdata));
+ else if (aic31xx->dev->of_node)
+ aic31xx_pdata_from_of(aic31xx);
+
+ if (aic31xx->pdata.gpio_reset) {
+ ret = devm_gpio_request_one(aic31xx->dev,
+ aic31xx->pdata.gpio_reset,
+ GPIOF_OUT_INIT_HIGH,
+ "aic31xx-reset-pin");
+ if (ret < 0) {
+ dev_err(aic31xx->dev, "not able to acquire gpio\n");
+ return;
+ }
+ }
+
+ for (i = 0; i < ARRAY_SIZE(aic31xx->supplies); i++)
+ aic31xx->supplies[i].supply = aic31xx_supply_names[i];
+
+ ret = devm_regulator_bulk_get(aic31xx->dev,
+ ARRAY_SIZE(aic31xx->supplies),
+ aic31xx->supplies);
+ if (ret != 0)
+ dev_err(aic31xx->dev, "Failed to request supplies: %d\n", ret);
+
+}
+
+static int aic31xx_i2c_probe(struct i2c_client *i2c,
+ const struct i2c_device_id *id)
+{
+ struct aic31xx_priv *aic31xx;
+ int ret;
+ const struct regmap_config *regmap_config;
+
+ dev_dbg(&i2c->dev, "## %s: %s codec_type = %d\n", __func__,
+ id->name, (int) id->driver_data);
+
+ regmap_config = &aic31xx_i2c_regmap;
+
+ aic31xx = devm_kzalloc(&i2c->dev, sizeof(*aic31xx), GFP_KERNEL);
+ if (aic31xx == NULL)
+ return -ENOMEM;
+
+ aic31xx->regmap = devm_regmap_init_i2c(i2c, regmap_config);
+ if (IS_ERR(aic31xx->regmap)) {
+ ret = PTR_ERR(aic31xx->regmap);
+ dev_err(&i2c->dev, "Failed to allocate register map: %d\n",
+ ret);
+ return ret;
+ }
+ aic31xx->dev = &i2c->dev;
+
+ aic31xx->pdata.codec_type = id->driver_data;
+
+ aic31xx_device_init(aic31xx);
+
+ return snd_soc_register_codec(&i2c->dev, &soc_codec_driver_aic31xx,
+ aic31xx_dai_driver,
+ ARRAY_SIZE(aic31xx_dai_driver));
+}
+
+static int aic31xx_i2c_remove(struct i2c_client *i2c)
+{
+ snd_soc_unregister_codec(&i2c->dev);
+ return 0;
+}
+
+static const struct i2c_device_id aic31xx_i2c_id[] = {
+ { "tlv320aic310x", AIC3100 },
+ { "tlv320aic311x", AIC3110 },
+ { "tlv320aic3100", AIC3100 },
+ { "tlv320aic3110", AIC3110 },
+ { "tlv320aic3120", AIC3120 },
+ { "tlv320aic3111", AIC3111 },
+ { }
+};
+MODULE_DEVICE_TABLE(i2c, aic31xx_i2c_id);
+
+static struct i2c_driver aic31xx_i2c_driver = {
+ .driver = {
+ .name = "tlv320aic31xx-codec",
+ .owner = THIS_MODULE,
+ .of_match_table = of_match_ptr(tlv320aic31xx_of_match),
+ },
+ .probe = aic31xx_i2c_probe,
+ .remove = aic31xx_i2c_remove,
+ .id_table = aic31xx_i2c_id,
+};
+
+module_i2c_driver(aic31xx_i2c_driver);
+
+MODULE_DESCRIPTION("ASoC TLV320AIC3111 codec driver");
+MODULE_AUTHOR("Jyri Sarha");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/tlv320aic31xx.h b/sound/soc/codecs/tlv320aic31xx.h
new file mode 100644
index 0000000..52ed57c
--- /dev/null
+++ b/sound/soc/codecs/tlv320aic31xx.h
@@ -0,0 +1,258 @@
+/*
+ * ALSA SoC TLV320AIC31XX codec driver
+ *
+ * Copyright (C) 2013 Texas Instruments, Inc.
+ *
+ * This package 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 PACKAGE IS PROVIDED ``AS IS'' AND WITHOUT ANY EXPRESS OR
+ * IMPLIED WARRANTIES, INCLUDING, WITHOUT LIMITATION, THE IMPLIED
+ * WARRANTIES OF MERCHANTIBILITY AND FITNESS FOR A PARTICULAR PURPOSE.
+ *
+ */
+#ifndef _TLV320AIC31XX_H
+#define _TLV320AIC31XX_H
+
+#define AIC31XX_RATES SNDRV_PCM_RATE_8000_192000
+
+#define AIC31XX_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE \
+ | SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE)
+
+
+#define AIC31XX_STEREO_CLASS_D_BIT 0x1
+#define AIC31XX_MINIDSP_BIT 0x2
+
+enum aic31xx_type {
+ AIC3100 = 0,
+ AIC3110 = AIC31XX_STEREO_CLASS_D_BIT,
+ AIC3120 = AIC31XX_MINIDSP_BIT,
+ AIC3111 = (AIC31XX_STEREO_CLASS_D_BIT | AIC31XX_MINIDSP_BIT),
+};
+
+struct aic31xx_pdata {
+ enum aic31xx_type codec_type;
+ unsigned int gpio_reset;
+ int micbias_vg;
+};
+
+/* Page Control Register */
+#define AIC31XX_PAGECTL 0x00
+
+/* Page 0 Registers */
+/* Software reset register */
+#define AIC31XX_RESET 0x01
+/* OT FLAG register */
+#define AIC31XX_OT_FLAG 0x03
+/* Clock clock Gen muxing, Multiplexers*/
+#define AIC31XX_CLKMUX 0x04
+/* PLL P and R-VAL register */
+#define AIC31XX_PLLPR 0x05
+/* PLL J-VAL register */
+#define AIC31XX_PLLJ 0x06
+/* PLL D-VAL MSB register */
+#define AIC31XX_PLLDMSB 0x07
+/* PLL D-VAL LSB register */
+#define AIC31XX_PLLDLSB 0x08
+/* DAC NDAC_VAL register*/
+#define AIC31XX_NDAC 0x0B
+/* DAC MDAC_VAL register */
+#define AIC31XX_MDAC 0x0C
+/* DAC OSR setting register 1, MSB value */
+#define AIC31XX_DOSRMSB 0x0D
+/* DAC OSR setting register 2, LSB value */
+#define AIC31XX_DOSRLSB 0x0E
+#define AIC31XX_MINI_DSP_INPOL 0x10
+/* Clock setting register 8, PLL */
+#define AIC31XX_NADC 0x12
+/* Clock setting register 9, PLL */
+#define AIC31XX_MADC 0x13
+/* ADC Oversampling (AOSR) Register */
+#define AIC31XX_AOSR 0x14
+/* Clock setting register 9, Multiplexers */
+#define AIC31XX_CLKOUTMUX 0x19
+/* Clock setting register 10, CLOCKOUT M divider value */
+#define AIC31XX_CLKOUTMVAL 0x1A
+/* Audio Interface Setting Register 1 */
+#define AIC31XX_IFACE1 0x1B
+/* Audio Data Slot Offset Programming */
+#define AIC31XX_DATA_OFFSET 0x1C
+/* Audio Interface Setting Register 2 */
+#define AIC31XX_IFACE2 0x1D
+/* Clock setting register 11, BCLK N Divider */
+#define AIC31XX_BCLKN 0x1E
+/* Audio Interface Setting Register 3, Secondary Audio Interface */
+#define AIC31XX_IFACESEC1 0x1F
+/* Audio Interface Setting Register 4 */
+#define AIC31XX_IFACESEC2 0x20
+/* Audio Interface Setting Register 5 */
+#define AIC31XX_IFACESEC3 0x21
+/* I2C Bus Condition */
+#define AIC31XX_I2C 0x22
+/* ADC FLAG */
+#define AIC31XX_ADCFLAG 0x24
+/* DAC Flag Registers */
+#define AIC31XX_DACFLAG1 0x25
+#define AIC31XX_DACFLAG2 0x26
+/* Sticky Interrupt flag (overflow) */
+#define AIC31XX_OFFLAG 0x27
+/* Sticy DAC Interrupt flags */
+#define AIC31XX_INTRDACFLAG 0x2C
+/* Sticy ADC Interrupt flags */
+#define AIC31XX_INTRADCFLAG 0x2D
+/* DAC Interrupt flags 2 */
+#define AIC31XX_INTRDACFLAG2 0x2E
+/* ADC Interrupt flags 2 */
+#define AIC31XX_INTRADCFLAG2 0x2F
+/* INT1 interrupt control */
+#define AIC31XX_INT1CTRL 0x30
+/* INT2 interrupt control */
+#define AIC31XX_INT2CTRL 0x31
+/* GPIO1 control */
+#define AIC31XX_GPIO1 0x33
+
+#define AIC31XX_DACPRB 0x3C
+/* ADC Instruction Set Register */
+#define AIC31XX_ADCPRB 0x3D
+/* DAC channel setup register */
+#define AIC31XX_DACSETUP 0x3F
+/* DAC Mute and volume control register */
+#define AIC31XX_DACMUTE 0x40
+/* Left DAC channel digital volume control */
+#define AIC31XX_LDACVOL 0x41
+/* Right DAC channel digital volume control */
+#define AIC31XX_RDACVOL 0x42
+/* Headset detection */
+#define AIC31XX_HSDETECT 0x43
+/* ADC Digital Mic */
+#define AIC31XX_ADCSETUP 0x51
+/* ADC Digital Volume Control Fine Adjust */
+#define AIC31XX_ADCFGA 0x52
+/* ADC Digital Volume Control Coarse Adjust */
+#define AIC31XX_ADCVOL 0x53
+
+
+/* Page 1 Registers */
+/* Headphone drivers */
+#define AIC31XX_HPDRIVER 0x9F
+/* Class-D Speakear Amplifier */
+#define AIC31XX_SPKAMP 0xA0
+/* HP Output Drivers POP Removal Settings */
+#define AIC31XX_HPPOP 0xA1
+/* Output Driver PGA Ramp-Down Period Control */
+#define AIC31XX_SPPGARAMP 0xA2
+/* DAC_L and DAC_R Output Mixer Routing */
+#define AIC31XX_DACMIXERROUTE 0xA3
+/* Left Analog Vol to HPL */
+#define AIC31XX_LANALOGHPL 0xA4
+/* Right Analog Vol to HPR */
+#define AIC31XX_RANALOGHPR 0xA5
+/* Left Analog Vol to SPL */
+#define AIC31XX_LANALOGSPL 0xA6
+/* Right Analog Vol to SPR */
+#define AIC31XX_RANALOGSPR 0xA7
+/* HPL Driver */
+#define AIC31XX_HPLGAIN 0xA8
+/* HPR Driver */
+#define AIC31XX_HPRGAIN 0xA9
+/* SPL Driver */
+#define AIC31XX_SPLGAIN 0xAA
+/* SPR Driver */
+#define AIC31XX_SPRGAIN 0xAB
+/* HP Driver Control */
+#define AIC31XX_HPCONTROL 0xAC
+/* MIC Bias Control */
+#define AIC31XX_MICBIAS 0xAE
+/* MIC PGA*/
+#define AIC31XX_MICPGA 0xAF
+/* Delta-Sigma Mono ADC Channel Fine-Gain Input Selection for P-Terminal */
+#define AIC31XX_MICPGAPI 0xB0
+/* ADC Input Selection for M-Terminal */
+#define AIC31XX_MICPGAMI 0xB1
+/* Input CM Settings */
+#define AIC31XX_MICPGACM 0xB2
+
+/* Bits, masks and shifts */
+
+/* AIC31XX_CLKMUX */
+#define AIC31XX_PLL_CLKIN_MASK 0x0c
+#define AIC31XX_PLL_CLKIN_SHIFT 2
+#define AIC31XX_PLL_CLKIN_MCLK 0
+#define AIC31XX_CODEC_CLKIN_MASK 0x03
+#define AIC31XX_CODEC_CLKIN_SHIFT 0
+#define AIC31XX_CODEC_CLKIN_PLL 3
+#define AIC31XX_CODEC_CLKIN_BCLK 1
+
+/* AIC31XX_PLLPR, AIC31XX_NDAC, AIC31XX_MDAC, AIC31XX_NADC, AIC31XX_MADC,
+ AIC31XX_BCLKN */
+#define AIC31XX_PLL_MASK 0x7f
+#define AIC31XX_PM_MASK 0x80
+
+/* AIC31XX_IFACE1 */
+#define AIC31XX_WORD_LEN_16BITS 0x00
+#define AIC31XX_WORD_LEN_20BITS 0x01
+#define AIC31XX_WORD_LEN_24BITS 0x02
+#define AIC31XX_WORD_LEN_32BITS 0x03
+#define AIC31XX_IFACE1_DATALEN_MASK 0x30
+#define AIC31XX_IFACE1_DATALEN_SHIFT (4)
+#define AIC31XX_IFACE1_DATATYPE_MASK 0xC0
+#define AIC31XX_IFACE1_DATATYPE_SHIFT (6)
+#define AIC31XX_I2S_MODE 0x00
+#define AIC31XX_DSP_MODE 0x01
+#define AIC31XX_RIGHT_JUSTIFIED_MODE 0x02
+#define AIC31XX_LEFT_JUSTIFIED_MODE 0x03
+#define AIC31XX_IFACE1_MASTER_MASK 0x0C
+#define AIC31XX_BCLK_MASTER 0x08
+#define AIC31XX_WCLK_MASTER 0x04
+
+/* AIC31XX_DATA_OFFSET */
+#define AIC31XX_DATA_OFFSET_MASK 0xFF
+
+/* AIC31XX_IFACE2 */
+#define AIC31XX_BCLKINV_MASK 0x08
+#define AIC31XX_BDIVCLK_MASK 0x03
+#define AIC31XX_DAC2BCLK 0x00
+#define AIC31XX_DACMOD2BCLK 0x01
+#define AIC31XX_ADC2BCLK 0x02
+#define AIC31XX_ADCMOD2BCLK 0x03
+
+/* AIC31XX_ADCFLAG */
+#define AIC31XX_ADCPWRSTATUS_MASK 0x40
+
+/* AIC31XX_DACFLAG1 */
+#define AIC31XX_LDACPWRSTATUS_MASK 0x80
+#define AIC31XX_RDACPWRSTATUS_MASK 0x08
+#define AIC31XX_HPLDRVPWRSTATUS_MASK 0x20
+#define AIC31XX_HPRDRVPWRSTATUS_MASK 0x02
+#define AIC31XX_SPLDRVPWRSTATUS_MASK 0x10
+#define AIC31XX_SPRDRVPWRSTATUS_MASK 0x01
+
+/* AIC31XX_INTRDACFLAG */
+#define AIC31XX_HPSCDETECT_MASK 0x80
+#define AIC31XX_BUTTONPRESS_MASK 0x20
+#define AIC31XX_HSPLUG_MASK 0x10
+#define AIC31XX_LDRCTHRES_MASK 0x08
+#define AIC31XX_RDRCTHRES_MASK 0x04
+#define AIC31XX_DACSINT_MASK 0x02
+#define AIC31XX_DACAINT_MASK 0x01
+
+/* AIC31XX_INT1CTRL */
+#define AIC31XX_HSPLUGDET_MASK 0x80
+#define AIC31XX_BUTTONPRESSDET_MASK 0x40
+#define AIC31XX_DRCTHRES_MASK 0x20
+#define AIC31XX_AGCNOISE_MASK 0x10
+#define AIC31XX_OC_MASK 0x08
+#define AIC31XX_ENGINE_MASK 0x04
+
+/* AIC31XX_DACSETUP */
+#define AIC31XX_SOFTSTEP_MASK 0x03
+
+/* AIC31XX_DACMUTE */
+#define AIC31XX_DACMUTE_MASK 0x0C
+
+/* AIC31XX_MICBIAS */
+#define AIC31XX_MICBIAS_MASK 0x03
+#define AIC31XX_MICBIAS_SHIFT 0
+
+#endif /* _TLV320AIC31XX_H */
diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c
index 688151b..1d9b117 100644
--- a/sound/soc/codecs/tlv320aic32x4.c
+++ b/sound/soc/codecs/tlv320aic32x4.c
@@ -29,9 +29,12 @@
#include <linux/delay.h>
#include <linux/pm.h>
#include <linux/gpio.h>
+#include <linux/of_gpio.h>
#include <linux/i2c.h>
#include <linux/cdev.h>
#include <linux/slab.h>
+#include <linux/clk.h>
+#include <linux/regulator/consumer.h>
#include <sound/tlv320aic32x4.h>
#include <sound/core.h>
@@ -66,20 +69,32 @@ struct aic32x4_priv {
u32 micpga_routing;
bool swapdacs;
int rstn_gpio;
+ struct clk *mclk;
+
+ struct regulator *supply_ldo;
+ struct regulator *supply_iov;
+ struct regulator *supply_dv;
+ struct regulator *supply_av;
};
-/* 0dB min, 1dB steps */
-static DECLARE_TLV_DB_SCALE(tlv_step_1, 0, 100, 0);
/* 0dB min, 0.5dB steps */
static DECLARE_TLV_DB_SCALE(tlv_step_0_5, 0, 50, 0);
+/* -63.5dB min, 0.5dB steps */
+static DECLARE_TLV_DB_SCALE(tlv_pcm, -6350, 50, 0);
+/* -6dB min, 1dB steps */
+static DECLARE_TLV_DB_SCALE(tlv_driver_gain, -600, 100, 0);
+/* -12dB min, 0.5dB steps */
+static DECLARE_TLV_DB_SCALE(tlv_adc_vol, -1200, 50, 0);
static const struct snd_kcontrol_new aic32x4_snd_controls[] = {
- SOC_DOUBLE_R_TLV("PCM Playback Volume", AIC32X4_LDACVOL,
- AIC32X4_RDACVOL, 0, 0x30, 0, tlv_step_0_5),
- SOC_DOUBLE_R_TLV("HP Driver Gain Volume", AIC32X4_HPLGAIN,
- AIC32X4_HPRGAIN, 0, 0x1D, 0, tlv_step_1),
- SOC_DOUBLE_R_TLV("LO Driver Gain Volume", AIC32X4_LOLGAIN,
- AIC32X4_LORGAIN, 0, 0x1D, 0, tlv_step_1),
+ SOC_DOUBLE_R_S_TLV("PCM Playback Volume", AIC32X4_LDACVOL,
+ AIC32X4_RDACVOL, 0, -0x7f, 0x30, 7, 0, tlv_pcm),
+ SOC_DOUBLE_R_S_TLV("HP Driver Gain Volume", AIC32X4_HPLGAIN,
+ AIC32X4_HPRGAIN, 0, -0x6, 0x1d, 5, 0,
+ tlv_driver_gain),
+ SOC_DOUBLE_R_S_TLV("LO Driver Gain Volume", AIC32X4_LOLGAIN,
+ AIC32X4_LORGAIN, 0, -0x6, 0x1d, 5, 0,
+ tlv_driver_gain),
SOC_DOUBLE_R("HP DAC Playback Switch", AIC32X4_HPLGAIN,
AIC32X4_HPRGAIN, 6, 0x01, 1),
SOC_DOUBLE_R("LO DAC Playback Switch", AIC32X4_LOLGAIN,
@@ -90,8 +105,8 @@ static const struct snd_kcontrol_new aic32x4_snd_controls[] = {
SOC_SINGLE("ADCFGA Left Mute Switch", AIC32X4_ADCFGA, 7, 1, 0),
SOC_SINGLE("ADCFGA Right Mute Switch", AIC32X4_ADCFGA, 3, 1, 0),
- SOC_DOUBLE_R_TLV("ADC Level Volume", AIC32X4_LADCVOL,
- AIC32X4_RADCVOL, 0, 0x28, 0, tlv_step_0_5),
+ SOC_DOUBLE_R_S_TLV("ADC Level Volume", AIC32X4_LADCVOL,
+ AIC32X4_RADCVOL, 0, -0x18, 0x28, 6, 0, tlv_adc_vol),
SOC_DOUBLE_R_TLV("PGA Level Volume", AIC32X4_LMICPGAVOL,
AIC32X4_RMICPGAVOL, 0, 0x5f, 0, tlv_step_0_5),
@@ -480,8 +495,18 @@ static int aic32x4_mute(struct snd_soc_dai *dai, int mute)
static int aic32x4_set_bias_level(struct snd_soc_codec *codec,
enum snd_soc_bias_level level)
{
+ struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec);
+ int ret;
+
switch (level) {
case SND_SOC_BIAS_ON:
+ /* Switch on master clock */
+ ret = clk_prepare_enable(aic32x4->mclk);
+ if (ret) {
+ dev_err(codec->dev, "Failed to enable master clock\n");
+ return ret;
+ }
+
/* Switch on PLL */
snd_soc_update_bits(codec, AIC32X4_PLLPR,
AIC32X4_PLLEN, AIC32X4_PLLEN);
@@ -509,29 +534,32 @@ static int aic32x4_set_bias_level(struct snd_soc_codec *codec,
case SND_SOC_BIAS_PREPARE:
break;
case SND_SOC_BIAS_STANDBY:
- /* Switch off PLL */
- snd_soc_update_bits(codec, AIC32X4_PLLPR,
- AIC32X4_PLLEN, 0);
+ /* Switch off BCLK_N Divider */
+ snd_soc_update_bits(codec, AIC32X4_BCLKN,
+ AIC32X4_BCLKEN, 0);
- /* Switch off NDAC Divider */
- snd_soc_update_bits(codec, AIC32X4_NDAC,
- AIC32X4_NDACEN, 0);
+ /* Switch off MADC Divider */
+ snd_soc_update_bits(codec, AIC32X4_MADC,
+ AIC32X4_MADCEN, 0);
+
+ /* Switch off NADC Divider */
+ snd_soc_update_bits(codec, AIC32X4_NADC,
+ AIC32X4_NADCEN, 0);
/* Switch off MDAC Divider */
snd_soc_update_bits(codec, AIC32X4_MDAC,
AIC32X4_MDACEN, 0);
- /* Switch off NADC Divider */
- snd_soc_update_bits(codec, AIC32X4_NADC,
- AIC32X4_NADCEN, 0);
+ /* Switch off NDAC Divider */
+ snd_soc_update_bits(codec, AIC32X4_NDAC,
+ AIC32X4_NDACEN, 0);
- /* Switch off MADC Divider */
- snd_soc_update_bits(codec, AIC32X4_MADC,
- AIC32X4_MADCEN, 0);
+ /* Switch off PLL */
+ snd_soc_update_bits(codec, AIC32X4_PLLPR,
+ AIC32X4_PLLEN, 0);
- /* Switch off BCLK_N Divider */
- snd_soc_update_bits(codec, AIC32X4_BCLKN,
- AIC32X4_BCLKEN, 0);
+ /* Switch off master clock */
+ clk_disable_unprepare(aic32x4->mclk);
break;
case SND_SOC_BIAS_OFF:
break;
@@ -586,9 +614,7 @@ static int aic32x4_probe(struct snd_soc_codec *codec)
struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec);
u32 tmp_reg;
- snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP);
-
- if (aic32x4->rstn_gpio >= 0) {
+ if (gpio_is_valid(aic32x4->rstn_gpio)) {
ndelay(10);
gpio_set_value(aic32x4->rstn_gpio, 1);
}
@@ -663,11 +689,122 @@ static struct snd_soc_codec_driver soc_codec_dev_aic32x4 = {
.num_dapm_routes = ARRAY_SIZE(aic32x4_dapm_routes),
};
+static int aic32x4_parse_dt(struct aic32x4_priv *aic32x4,
+ struct device_node *np)
+{
+ aic32x4->swapdacs = false;
+ aic32x4->micpga_routing = 0;
+ aic32x4->rstn_gpio = of_get_named_gpio(np, "reset-gpios", 0);
+
+ return 0;
+}
+
+static void aic32x4_disable_regulators(struct aic32x4_priv *aic32x4)
+{
+ regulator_disable(aic32x4->supply_iov);
+
+ if (!IS_ERR(aic32x4->supply_ldo))
+ regulator_disable(aic32x4->supply_ldo);
+
+ if (!IS_ERR(aic32x4->supply_dv))
+ regulator_disable(aic32x4->supply_dv);
+
+ if (!IS_ERR(aic32x4->supply_av))
+ regulator_disable(aic32x4->supply_av);
+}
+
+static int aic32x4_setup_regulators(struct device *dev,
+ struct aic32x4_priv *aic32x4)
+{
+ int ret = 0;
+
+ aic32x4->supply_ldo = devm_regulator_get_optional(dev, "ldoin");
+ aic32x4->supply_iov = devm_regulator_get(dev, "iov");
+ aic32x4->supply_dv = devm_regulator_get_optional(dev, "dv");
+ aic32x4->supply_av = devm_regulator_get_optional(dev, "av");
+
+ /* Check if the regulator requirements are fulfilled */
+
+ if (IS_ERR(aic32x4->supply_iov)) {
+ dev_err(dev, "Missing supply 'iov'\n");
+ return PTR_ERR(aic32x4->supply_iov);
+ }
+
+ if (IS_ERR(aic32x4->supply_ldo)) {
+ if (PTR_ERR(aic32x4->supply_ldo) == -EPROBE_DEFER)
+ return -EPROBE_DEFER;
+
+ if (IS_ERR(aic32x4->supply_dv)) {
+ dev_err(dev, "Missing supply 'dv' or 'ldoin'\n");
+ return PTR_ERR(aic32x4->supply_dv);
+ }
+ if (IS_ERR(aic32x4->supply_av)) {
+ dev_err(dev, "Missing supply 'av' or 'ldoin'\n");
+ return PTR_ERR(aic32x4->supply_av);
+ }
+ } else {
+ if (IS_ERR(aic32x4->supply_dv) &&
+ PTR_ERR(aic32x4->supply_dv) == -EPROBE_DEFER)
+ return -EPROBE_DEFER;
+ if (IS_ERR(aic32x4->supply_av) &&
+ PTR_ERR(aic32x4->supply_av) == -EPROBE_DEFER)
+ return -EPROBE_DEFER;
+ }
+
+ ret = regulator_enable(aic32x4->supply_iov);
+ if (ret) {
+ dev_err(dev, "Failed to enable regulator iov\n");
+ return ret;
+ }
+
+ if (!IS_ERR(aic32x4->supply_ldo)) {
+ ret = regulator_enable(aic32x4->supply_ldo);
+ if (ret) {
+ dev_err(dev, "Failed to enable regulator ldo\n");
+ goto error_ldo;
+ }
+ }
+
+ if (!IS_ERR(aic32x4->supply_dv)) {
+ ret = regulator_enable(aic32x4->supply_dv);
+ if (ret) {
+ dev_err(dev, "Failed to enable regulator dv\n");
+ goto error_dv;
+ }
+ }
+
+ if (!IS_ERR(aic32x4->supply_av)) {
+ ret = regulator_enable(aic32x4->supply_av);
+ if (ret) {
+ dev_err(dev, "Failed to enable regulator av\n");
+ goto error_av;
+ }
+ }
+
+ if (!IS_ERR(aic32x4->supply_ldo) && IS_ERR(aic32x4->supply_av))
+ aic32x4->power_cfg |= AIC32X4_PWR_AIC32X4_LDO_ENABLE;
+
+ return 0;
+
+error_av:
+ if (!IS_ERR(aic32x4->supply_dv))
+ regulator_disable(aic32x4->supply_dv);
+
+error_dv:
+ if (!IS_ERR(aic32x4->supply_ldo))
+ regulator_disable(aic32x4->supply_ldo);
+
+error_ldo:
+ regulator_disable(aic32x4->supply_iov);
+ return ret;
+}
+
static int aic32x4_i2c_probe(struct i2c_client *i2c,
const struct i2c_device_id *id)
{
struct aic32x4_pdata *pdata = i2c->dev.platform_data;
struct aic32x4_priv *aic32x4;
+ struct device_node *np = i2c->dev.of_node;
int ret;
aic32x4 = devm_kzalloc(&i2c->dev, sizeof(struct aic32x4_priv),
@@ -686,6 +823,12 @@ static int aic32x4_i2c_probe(struct i2c_client *i2c,
aic32x4->swapdacs = pdata->swapdacs;
aic32x4->micpga_routing = pdata->micpga_routing;
aic32x4->rstn_gpio = pdata->rstn_gpio;
+ } else if (np) {
+ ret = aic32x4_parse_dt(aic32x4, np);
+ if (ret) {
+ dev_err(&i2c->dev, "Failed to parse DT node\n");
+ return ret;
+ }
} else {
aic32x4->power_cfg = 0;
aic32x4->swapdacs = false;
@@ -693,20 +836,44 @@ static int aic32x4_i2c_probe(struct i2c_client *i2c,
aic32x4->rstn_gpio = -1;
}
- if (aic32x4->rstn_gpio >= 0) {
+ aic32x4->mclk = devm_clk_get(&i2c->dev, "mclk");
+ if (IS_ERR(aic32x4->mclk)) {
+ dev_err(&i2c->dev, "Failed getting the mclk. The current implementation does not support the usage of this codec without mclk\n");
+ return PTR_ERR(aic32x4->mclk);
+ }
+
+ if (gpio_is_valid(aic32x4->rstn_gpio)) {
ret = devm_gpio_request_one(&i2c->dev, aic32x4->rstn_gpio,
GPIOF_OUT_INIT_LOW, "tlv320aic32x4 rstn");
if (ret != 0)
return ret;
}
+ ret = aic32x4_setup_regulators(&i2c->dev, aic32x4);
+ if (ret) {
+ dev_err(&i2c->dev, "Failed to setup regulators\n");
+ return ret;
+ }
+
ret = snd_soc_register_codec(&i2c->dev,
&soc_codec_dev_aic32x4, &aic32x4_dai, 1);
- return ret;
+ if (ret) {
+ dev_err(&i2c->dev, "Failed to register codec\n");
+ aic32x4_disable_regulators(aic32x4);
+ return ret;
+ }
+
+ i2c_set_clientdata(i2c, aic32x4);
+
+ return 0;
}
static int aic32x4_i2c_remove(struct i2c_client *client)
{
+ struct aic32x4_priv *aic32x4 = i2c_get_clientdata(client);
+
+ aic32x4_disable_regulators(aic32x4);
+
snd_soc_unregister_codec(&client->dev);
return 0;
}
@@ -717,10 +884,17 @@ static const struct i2c_device_id aic32x4_i2c_id[] = {
};
MODULE_DEVICE_TABLE(i2c, aic32x4_i2c_id);
+static const struct of_device_id aic32x4_of_id[] = {
+ { .compatible = "ti,tlv320aic32x4", },
+ { /* senitel */ }
+};
+MODULE_DEVICE_TABLE(of, aic32x4_of_id);
+
static struct i2c_driver aic32x4_i2c_driver = {
.driver = {
.name = "tlv320aic32x4",
.owner = THIS_MODULE,
+ .of_match_table = aic32x4_of_id,
},
.probe = aic32x4_i2c_probe,
.remove = aic32x4_i2c_remove,
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index 470fbfb..b183510 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -1344,12 +1344,6 @@ static int aic3x_probe(struct snd_soc_codec *codec)
INIT_LIST_HEAD(&aic3x->list);
aic3x->codec = codec;
- 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;
- }
-
for (i = 0; i < ARRAY_SIZE(aic3x->supplies); i++) {
aic3x->disable_nb[i].nb.notifier_call = aic3x_regulator_event;
aic3x->disable_nb[i].aic3x = aic3x;
diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c
index 4f35839..6bfc8a1 100644
--- a/sound/soc/codecs/tlv320dac33.c
+++ b/sound/soc/codecs/tlv320dac33.c
@@ -122,7 +122,6 @@ struct tlv320dac33_priv {
unsigned int uthr;
enum dac33_state state;
- enum snd_soc_control_type control_type;
void *control_data;
};
@@ -461,7 +460,7 @@ static int dac33_set_fifo_mode(struct snd_kcontrol *kcontrol,
if (dac33->fifo_mode == ucontrol->value.integer.value[0])
return 0;
/* Do not allow changes while stream is running*/
- if (codec->active)
+ if (snd_soc_codec_is_active(codec))
return -EPERM;
if (ucontrol->value.integer.value[0] < 0 ||
@@ -478,9 +477,7 @@ static const char *dac33_fifo_mode_texts[] = {
"Bypass", "Mode 1", "Mode 7"
};
-static const struct soc_enum dac33_fifo_mode_enum =
- SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(dac33_fifo_mode_texts),
- dac33_fifo_mode_texts);
+static SOC_ENUM_SINGLE_EXT_DECL(dac33_fifo_mode_enum, dac33_fifo_mode_texts);
/* L/R Line Output Gain */
static const char *lr_lineout_gain_texts[] = {
@@ -488,15 +485,13 @@ static const char *lr_lineout_gain_texts[] = {
"Line 0dB DAC 12dB", "Line 6dB DAC 18dB",
};
-static const struct soc_enum l_lineout_gain_enum =
- SOC_ENUM_SINGLE(DAC33_LDAC_PWR_CTRL, 0,
- ARRAY_SIZE(lr_lineout_gain_texts),
- lr_lineout_gain_texts);
+static SOC_ENUM_SINGLE_DECL(l_lineout_gain_enum,
+ DAC33_LDAC_PWR_CTRL, 0,
+ lr_lineout_gain_texts);
-static const struct soc_enum r_lineout_gain_enum =
- SOC_ENUM_SINGLE(DAC33_RDAC_PWR_CTRL, 0,
- ARRAY_SIZE(lr_lineout_gain_texts),
- lr_lineout_gain_texts);
+static SOC_ENUM_SINGLE_DECL(r_lineout_gain_enum,
+ DAC33_RDAC_PWR_CTRL, 0,
+ lr_lineout_gain_texts);
/*
* DACL/R digital volume control:
@@ -534,18 +529,16 @@ static const struct snd_kcontrol_new dac33_dapm_abypassr_control =
/* LOP L/R invert selection */
static const char *dac33_lr_lom_texts[] = {"DAC", "LOP"};
-static const struct soc_enum dac33_left_lom_enum =
- SOC_ENUM_SINGLE(DAC33_OUT_AMP_CTRL, 3,
- ARRAY_SIZE(dac33_lr_lom_texts),
- dac33_lr_lom_texts);
+static SOC_ENUM_SINGLE_DECL(dac33_left_lom_enum,
+ DAC33_OUT_AMP_CTRL, 3,
+ dac33_lr_lom_texts);
static const struct snd_kcontrol_new dac33_dapm_left_lom_control =
SOC_DAPM_ENUM("Route", dac33_left_lom_enum);
-static const struct soc_enum dac33_right_lom_enum =
- SOC_ENUM_SINGLE(DAC33_OUT_AMP_CTRL, 2,
- ARRAY_SIZE(dac33_lr_lom_texts),
- dac33_lr_lom_texts);
+static SOC_ENUM_SINGLE_DECL(dac33_right_lom_enum,
+ DAC33_OUT_AMP_CTRL, 2,
+ dac33_lr_lom_texts);
static const struct snd_kcontrol_new dac33_dapm_right_lom_control =
SOC_DAPM_ENUM("Route", dac33_right_lom_enum);
diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c
index 00665ad..975e0f7 100644
--- a/sound/soc/codecs/twl4030.c
+++ b/sound/soc/codecs/twl4030.c
@@ -415,10 +415,9 @@ static const struct snd_kcontrol_new twl4030_dapm_carkitr_controls[] = {
static const char *twl4030_handsfreel_texts[] =
{"Voice", "AudioL1", "AudioL2", "AudioR2"};
-static const struct soc_enum twl4030_handsfreel_enum =
- SOC_ENUM_SINGLE(TWL4030_REG_HFL_CTL, 0,
- ARRAY_SIZE(twl4030_handsfreel_texts),
- twl4030_handsfreel_texts);
+static SOC_ENUM_SINGLE_DECL(twl4030_handsfreel_enum,
+ TWL4030_REG_HFL_CTL, 0,
+ twl4030_handsfreel_texts);
static const struct snd_kcontrol_new twl4030_dapm_handsfreel_control =
SOC_DAPM_ENUM("Route", twl4030_handsfreel_enum);
@@ -431,10 +430,9 @@ static const struct snd_kcontrol_new twl4030_dapm_handsfreelmute_control =
static const char *twl4030_handsfreer_texts[] =
{"Voice", "AudioR1", "AudioR2", "AudioL2"};
-static const struct soc_enum twl4030_handsfreer_enum =
- SOC_ENUM_SINGLE(TWL4030_REG_HFR_CTL, 0,
- ARRAY_SIZE(twl4030_handsfreer_texts),
- twl4030_handsfreer_texts);
+static SOC_ENUM_SINGLE_DECL(twl4030_handsfreer_enum,
+ TWL4030_REG_HFR_CTL, 0,
+ twl4030_handsfreer_texts);
static const struct snd_kcontrol_new twl4030_dapm_handsfreer_control =
SOC_DAPM_ENUM("Route", twl4030_handsfreer_enum);
@@ -448,10 +446,9 @@ static const struct snd_kcontrol_new twl4030_dapm_handsfreermute_control =
static const char *twl4030_vibra_texts[] =
{"AudioL1", "AudioR1", "AudioL2", "AudioR2"};
-static const struct soc_enum twl4030_vibra_enum =
- SOC_ENUM_SINGLE(TWL4030_REG_VIBRA_CTL, 2,
- ARRAY_SIZE(twl4030_vibra_texts),
- twl4030_vibra_texts);
+static SOC_ENUM_SINGLE_DECL(twl4030_vibra_enum,
+ TWL4030_REG_VIBRA_CTL, 2,
+ twl4030_vibra_texts);
static const struct snd_kcontrol_new twl4030_dapm_vibra_control =
SOC_DAPM_ENUM("Route", twl4030_vibra_enum);
@@ -460,10 +457,9 @@ SOC_DAPM_ENUM("Route", twl4030_vibra_enum);
static const char *twl4030_vibrapath_texts[] =
{"Local vibrator", "Audio"};
-static const struct soc_enum twl4030_vibrapath_enum =
- SOC_ENUM_SINGLE(TWL4030_REG_VIBRA_CTL, 4,
- ARRAY_SIZE(twl4030_vibrapath_texts),
- twl4030_vibrapath_texts);
+static SOC_ENUM_SINGLE_DECL(twl4030_vibrapath_enum,
+ TWL4030_REG_VIBRA_CTL, 4,
+ twl4030_vibrapath_texts);
static const struct snd_kcontrol_new twl4030_dapm_vibrapath_control =
SOC_DAPM_ENUM("Route", twl4030_vibrapath_enum);
@@ -490,10 +486,9 @@ static const struct snd_kcontrol_new twl4030_dapm_analogrmic_controls[] = {
static const char *twl4030_micpathtx1_texts[] =
{"Analog", "Digimic0"};
-static const struct soc_enum twl4030_micpathtx1_enum =
- SOC_ENUM_SINGLE(TWL4030_REG_ADCMICSEL, 0,
- ARRAY_SIZE(twl4030_micpathtx1_texts),
- twl4030_micpathtx1_texts);
+static SOC_ENUM_SINGLE_DECL(twl4030_micpathtx1_enum,
+ TWL4030_REG_ADCMICSEL, 0,
+ twl4030_micpathtx1_texts);
static const struct snd_kcontrol_new twl4030_dapm_micpathtx1_control =
SOC_DAPM_ENUM("Route", twl4030_micpathtx1_enum);
@@ -502,10 +497,9 @@ SOC_DAPM_ENUM("Route", twl4030_micpathtx1_enum);
static const char *twl4030_micpathtx2_texts[] =
{"Analog", "Digimic1"};
-static const struct soc_enum twl4030_micpathtx2_enum =
- SOC_ENUM_SINGLE(TWL4030_REG_ADCMICSEL, 2,
- ARRAY_SIZE(twl4030_micpathtx2_texts),
- twl4030_micpathtx2_texts);
+static SOC_ENUM_SINGLE_DECL(twl4030_micpathtx2_enum,
+ TWL4030_REG_ADCMICSEL, 2,
+ twl4030_micpathtx2_texts);
static const struct snd_kcontrol_new twl4030_dapm_micpathtx2_control =
SOC_DAPM_ENUM("Route", twl4030_micpathtx2_enum);
@@ -955,19 +949,15 @@ static const char *twl4030_op_modes_texts[] = {
"Option 2 (voice/audio)", "Option 1 (audio)"
};
-static const struct soc_enum twl4030_op_modes_enum =
- SOC_ENUM_SINGLE(TWL4030_REG_CODEC_MODE, 0,
- ARRAY_SIZE(twl4030_op_modes_texts),
- twl4030_op_modes_texts);
+static SOC_ENUM_SINGLE_DECL(twl4030_op_modes_enum,
+ TWL4030_REG_CODEC_MODE, 0,
+ twl4030_op_modes_texts);
static int snd_soc_put_twl4030_opmode_enum_double(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec);
- struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
- unsigned short val;
- unsigned short mask;
if (twl4030->configured) {
dev_err(codec->dev,
@@ -975,19 +965,7 @@ static int snd_soc_put_twl4030_opmode_enum_double(struct snd_kcontrol *kcontrol,
return -EBUSY;
}
- if (ucontrol->value.enumerated.item[0] > e->max - 1)
- return -EINVAL;
-
- val = ucontrol->value.enumerated.item[0] << e->shift_l;
- mask = e->mask << e->shift_l;
- if (e->shift_l != e->shift_r) {
- if (ucontrol->value.enumerated.item[1] > e->max - 1)
- return -EINVAL;
- val |= ucontrol->value.enumerated.item[1] << e->shift_r;
- mask |= e->mask << e->shift_r;
- }
-
- return snd_soc_update_bits(codec, e->reg, mask, val);
+ return snd_soc_put_enum_double(kcontrol, ucontrol);
}
/*
@@ -1044,10 +1022,9 @@ static const char *twl4030_avadc_clk_priority_texts[] = {
"Voice high priority", "HiFi high priority"
};
-static const struct soc_enum twl4030_avadc_clk_priority_enum =
- SOC_ENUM_SINGLE(TWL4030_REG_AVADC_CTL, 2,
- ARRAY_SIZE(twl4030_avadc_clk_priority_texts),
- twl4030_avadc_clk_priority_texts);
+static SOC_ENUM_SINGLE_DECL(twl4030_avadc_clk_priority_enum,
+ TWL4030_REG_AVADC_CTL, 2,
+ twl4030_avadc_clk_priority_texts);
static const char *twl4030_rampdelay_texts[] = {
"27/20/14 ms", "55/40/27 ms", "109/81/55 ms", "218/161/109 ms",
@@ -1055,40 +1032,36 @@ static const char *twl4030_rampdelay_texts[] = {
"3495/2581/1748 ms"
};
-static const struct soc_enum twl4030_rampdelay_enum =
- SOC_ENUM_SINGLE(TWL4030_REG_HS_POPN_SET, 2,
- ARRAY_SIZE(twl4030_rampdelay_texts),
- twl4030_rampdelay_texts);
+static SOC_ENUM_SINGLE_DECL(twl4030_rampdelay_enum,
+ TWL4030_REG_HS_POPN_SET, 2,
+ twl4030_rampdelay_texts);
/* Vibra H-bridge direction mode */
static const char *twl4030_vibradirmode_texts[] = {
"Vibra H-bridge direction", "Audio data MSB",
};
-static const struct soc_enum twl4030_vibradirmode_enum =
- SOC_ENUM_SINGLE(TWL4030_REG_VIBRA_CTL, 5,
- ARRAY_SIZE(twl4030_vibradirmode_texts),
- twl4030_vibradirmode_texts);
+static SOC_ENUM_SINGLE_DECL(twl4030_vibradirmode_enum,
+ TWL4030_REG_VIBRA_CTL, 5,
+ twl4030_vibradirmode_texts);
/* Vibra H-bridge direction */
static const char *twl4030_vibradir_texts[] = {
"Positive polarity", "Negative polarity",
};
-static const struct soc_enum twl4030_vibradir_enum =
- SOC_ENUM_SINGLE(TWL4030_REG_VIBRA_CTL, 1,
- ARRAY_SIZE(twl4030_vibradir_texts),
- twl4030_vibradir_texts);
+static SOC_ENUM_SINGLE_DECL(twl4030_vibradir_enum,
+ TWL4030_REG_VIBRA_CTL, 1,
+ twl4030_vibradir_texts);
/* Digimic Left and right swapping */
static const char *twl4030_digimicswap_texts[] = {
"Not swapped", "Swapped",
};
-static const struct soc_enum twl4030_digimicswap_enum =
- SOC_ENUM_SINGLE(TWL4030_REG_MISC_SET_1, 0,
- ARRAY_SIZE(twl4030_digimicswap_texts),
- twl4030_digimicswap_texts);
+static SOC_ENUM_SINGLE_DECL(twl4030_digimicswap_enum,
+ TWL4030_REG_MISC_SET_1, 0,
+ twl4030_digimicswap_texts);
static const struct snd_kcontrol_new twl4030_snd_controls[] = {
/* Codec operation mode control */
diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c
index 0afe8be..bd3a206 100644
--- a/sound/soc/codecs/twl6040.c
+++ b/sound/soc/codecs/twl6040.c
@@ -81,7 +81,7 @@ struct twl6040_data {
};
/* set of rates for each pll: low-power and high-performance */
-static unsigned int lp_rates[] = {
+static const unsigned int lp_rates[] = {
8000,
11250,
16000,
@@ -93,7 +93,7 @@ static unsigned int lp_rates[] = {
96000,
};
-static unsigned int hp_rates[] = {
+static const unsigned int hp_rates[] = {
8000,
16000,
32000,
@@ -101,7 +101,7 @@ static unsigned int hp_rates[] = {
96000,
};
-static struct snd_pcm_hw_constraint_list sysclk_constraints[] = {
+static const struct snd_pcm_hw_constraint_list sysclk_constraints[] = {
{ .count = ARRAY_SIZE(lp_rates), .list = lp_rates, },
{ .count = ARRAY_SIZE(hp_rates), .list = hp_rates, },
};
@@ -392,8 +392,10 @@ static const char *twl6040_amicr_texts[] =
{"Headset Mic", "Sub Mic", "Aux/FM Right", "Off"};
static const struct soc_enum twl6040_enum[] = {
- SOC_ENUM_SINGLE(TWL6040_REG_MICLCTL, 3, 4, twl6040_amicl_texts),
- SOC_ENUM_SINGLE(TWL6040_REG_MICRCTL, 3, 4, twl6040_amicr_texts),
+ SOC_ENUM_SINGLE(TWL6040_REG_MICLCTL, 3,
+ ARRAY_SIZE(twl6040_amicl_texts), twl6040_amicl_texts),
+ SOC_ENUM_SINGLE(TWL6040_REG_MICRCTL, 3,
+ ARRAY_SIZE(twl6040_amicr_texts), twl6040_amicr_texts),
};
static const char *twl6040_hs_texts[] = {
@@ -476,9 +478,8 @@ static const char *twl6040_power_mode_texts[] = {
"Low-Power", "High-Performance",
};
-static const struct soc_enum twl6040_power_mode_enum =
- SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(twl6040_power_mode_texts),
- twl6040_power_mode_texts);
+static SOC_ENUM_SINGLE_EXT_DECL(twl6040_power_mode_enum,
+ twl6040_power_mode_texts);
static int twl6040_headset_power_get_enum(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c
index c94d4c1..edf27ac 100644
--- a/sound/soc/codecs/uda134x.c
+++ b/sound/soc/codecs/uda134x.c
@@ -203,8 +203,7 @@ static int uda134x_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 uda134x_priv *uda134x = snd_soc_codec_get_drvdata(codec);
u8 hw_params;
diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c
index 726df6d..e62e707 100644
--- a/sound/soc/codecs/uda1380.c
+++ b/sound/soc/codecs/uda1380.c
@@ -108,7 +108,7 @@ static int uda1380_write(struct snd_soc_codec *codec, unsigned int reg,
/* the interpolator & decimator regs must only be written when the
* codec DAI is active.
*/
- if (!codec->active && (reg >= UDA1380_MVOL))
+ if (!snd_soc_codec_is_active(codec) && (reg >= UDA1380_MVOL))
return 0;
pr_debug("uda1380: hw write %x val %x\n", reg, value);
if (codec->hw_write(codec->control_data, data, 3) == 3) {
@@ -237,25 +237,27 @@ static const char *uda1380_os_setting[] = {
};
static const struct soc_enum uda1380_deemp_enum[] = {
- SOC_ENUM_SINGLE(UDA1380_DEEMP, 8, 5, uda1380_deemp),
- SOC_ENUM_SINGLE(UDA1380_DEEMP, 0, 5, uda1380_deemp),
+ SOC_ENUM_SINGLE(UDA1380_DEEMP, 8, ARRAY_SIZE(uda1380_deemp),
+ uda1380_deemp),
+ SOC_ENUM_SINGLE(UDA1380_DEEMP, 0, ARRAY_SIZE(uda1380_deemp),
+ uda1380_deemp),
};
-static const struct soc_enum uda1380_input_sel_enum =
- SOC_ENUM_SINGLE(UDA1380_ADC, 2, 4, uda1380_input_sel); /* SEL_MIC, SEL_LNA */
-static const struct soc_enum uda1380_output_sel_enum =
- SOC_ENUM_SINGLE(UDA1380_PM, 7, 2, uda1380_output_sel); /* R02_EN_AVC */
-static const struct soc_enum uda1380_spf_enum =
- SOC_ENUM_SINGLE(UDA1380_MODE, 14, 4, uda1380_spf_mode); /* M */
-static const struct soc_enum uda1380_capture_sel_enum =
- SOC_ENUM_SINGLE(UDA1380_IFACE, 6, 2, uda1380_capture_sel); /* SEL_SOURCE */
-static const struct soc_enum uda1380_sel_ns_enum =
- SOC_ENUM_SINGLE(UDA1380_MIXER, 14, 2, uda1380_sel_ns); /* SEL_NS */
-static const struct soc_enum uda1380_mix_enum =
- SOC_ENUM_SINGLE(UDA1380_MIXER, 12, 4, uda1380_mix_control); /* MIX, MIX_POS */
-static const struct soc_enum uda1380_sdet_enum =
- SOC_ENUM_SINGLE(UDA1380_MIXER, 4, 4, uda1380_sdet_setting); /* SD_VALUE */
-static const struct soc_enum uda1380_os_enum =
- SOC_ENUM_SINGLE(UDA1380_MIXER, 0, 3, uda1380_os_setting); /* OS */
+static SOC_ENUM_SINGLE_DECL(uda1380_input_sel_enum,
+ UDA1380_ADC, 2, uda1380_input_sel); /* SEL_MIC, SEL_LNA */
+static SOC_ENUM_SINGLE_DECL(uda1380_output_sel_enum,
+ UDA1380_PM, 7, uda1380_output_sel); /* R02_EN_AVC */
+static SOC_ENUM_SINGLE_DECL(uda1380_spf_enum,
+ UDA1380_MODE, 14, uda1380_spf_mode); /* M */
+static SOC_ENUM_SINGLE_DECL(uda1380_capture_sel_enum,
+ UDA1380_IFACE, 6, uda1380_capture_sel); /* SEL_SOURCE */
+static SOC_ENUM_SINGLE_DECL(uda1380_sel_ns_enum,
+ UDA1380_MIXER, 14, uda1380_sel_ns); /* SEL_NS */
+static SOC_ENUM_SINGLE_DECL(uda1380_mix_enum,
+ UDA1380_MIXER, 12, uda1380_mix_control); /* MIX, MIX_POS */
+static SOC_ENUM_SINGLE_DECL(uda1380_sdet_enum,
+ UDA1380_MIXER, 4, uda1380_sdet_setting); /* SD_VALUE */
+static SOC_ENUM_SINGLE_DECL(uda1380_os_enum,
+ UDA1380_MIXER, 0, uda1380_os_setting); /* OS */
/*
* from -48 dB in 1.5 dB steps (mute instead of -49.5 dB)
@@ -564,8 +566,7 @@ static int uda1380_pcm_hw_params(struct snd_pcm_substream *substream,
static void uda1380_pcm_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;
u16 clk = uda1380_read_reg_cache(codec, UDA1380_CLK);
/* shut down WSPLL power if running from this clock */
diff --git a/sound/soc/codecs/wl1273.c b/sound/soc/codecs/wl1273.c
index b7ab2ef..6be5f80 100644
--- a/sound/soc/codecs/wl1273.c
+++ b/sound/soc/codecs/wl1273.c
@@ -197,7 +197,7 @@ static int snd_wl1273_set_audio_route(struct snd_kcontrol *kcontrol,
return 0;
/* Do not allow changes while stream is running */
- if (codec->active)
+ if (snd_soc_codec_is_active(codec))
return -EPERM;
if (ucontrol->value.integer.value[0] < 0 ||
@@ -209,8 +209,7 @@ static int snd_wl1273_set_audio_route(struct snd_kcontrol *kcontrol,
return 1;
}
-static const struct soc_enum wl1273_enum =
- SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(wl1273_audio_route), wl1273_audio_route);
+static SOC_ENUM_SINGLE_EXT_DECL(wl1273_enum, wl1273_audio_route);
static int snd_wl1273_fm_audio_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
@@ -247,9 +246,7 @@ static int snd_wl1273_fm_audio_put(struct snd_kcontrol *kcontrol,
static const char * const wl1273_audio_strings[] = { "Digital", "Analog" };
-static const struct soc_enum wl1273_audio_enum =
- SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(wl1273_audio_strings),
- wl1273_audio_strings);
+static SOC_ENUM_SINGLE_EXT_DECL(wl1273_audio_enum, wl1273_audio_strings);
static int snd_wl1273_fm_volume_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c
index 8ae5027..83a2c87 100644
--- a/sound/soc/codecs/wm2000.c
+++ b/sound/soc/codecs/wm2000.c
@@ -786,8 +786,6 @@ static int wm2000_probe(struct snd_soc_codec *codec)
{
struct wm2000_priv *wm2000 = dev_get_drvdata(codec->dev);
- snd_soc_codec_set_cache_io(codec, 16, 8, SND_SOC_REGMAP);
-
/* This will trigger a transition to standby mode by default */
wm2000_anc_set_mode(wm2000);
diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c
index 57ba315..2e721e0 100644
--- a/sound/soc/codecs/wm2200.c
+++ b/sound/soc/codecs/wm2200.c
@@ -1113,11 +1113,10 @@ static const char *wm2200_rxanc_input_sel_texts[] = {
"None", "IN1", "IN2", "IN3",
};
-static const struct soc_enum wm2200_rxanc_input_sel =
- SOC_ENUM_SINGLE(WM2200_RXANC_SRC,
- WM2200_IN_RXANC_SEL_SHIFT,
- ARRAY_SIZE(wm2200_rxanc_input_sel_texts),
- wm2200_rxanc_input_sel_texts);
+static SOC_ENUM_SINGLE_DECL(wm2200_rxanc_input_sel,
+ WM2200_RXANC_SRC,
+ WM2200_IN_RXANC_SEL_SHIFT,
+ wm2200_rxanc_input_sel_texts);
static const struct snd_kcontrol_new wm2200_snd_controls[] = {
SOC_SINGLE("IN1 High Performance Switch", WM2200_IN1L_CONTROL,
@@ -1288,11 +1287,10 @@ static const char *wm2200_aec_loopback_texts[] = {
"OUT1L", "OUT1R", "OUT2L", "OUT2R",
};
-static const struct soc_enum wm2200_aec_loopback =
- SOC_ENUM_SINGLE(WM2200_DAC_AEC_CONTROL_1,
- WM2200_AEC_LOOPBACK_SRC_SHIFT,
- ARRAY_SIZE(wm2200_aec_loopback_texts),
- wm2200_aec_loopback_texts);
+static SOC_ENUM_SINGLE_DECL(wm2200_aec_loopback,
+ WM2200_DAC_AEC_CONTROL_1,
+ WM2200_AEC_LOOPBACK_SRC_SHIFT,
+ wm2200_aec_loopback_texts);
static const struct snd_kcontrol_new wm2200_aec_loopback_mux =
SOC_DAPM_ENUM("AEC Loopback", wm2200_aec_loopback);
@@ -1556,15 +1554,8 @@ static int wm2200_probe(struct snd_soc_codec *codec)
int ret;
wm2200->codec = codec;
- codec->control_data = wm2200->regmap;
codec->dapm.bias_level = SND_SOC_BIAS_OFF;
- ret = snd_soc_codec_set_cache_io(codec, 16, 16, SND_SOC_REGMAP);
- if (ret != 0) {
- dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
- return ret;
- }
-
ret = snd_soc_add_codec_controls(codec, wm_adsp1_fw_controls, 2);
if (ret != 0)
return ret;
diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c
index 4e3e31a..eca983f 100644
--- a/sound/soc/codecs/wm5100.c
+++ b/sound/soc/codecs/wm5100.c
@@ -506,21 +506,21 @@ static const char *wm5100_lhpf_mode_text[] = {
"Low-pass", "High-pass"
};
-static const struct soc_enum wm5100_lhpf1_mode =
- SOC_ENUM_SINGLE(WM5100_HPLPF1_1, WM5100_LHPF1_MODE_SHIFT, 2,
- wm5100_lhpf_mode_text);
+static SOC_ENUM_SINGLE_DECL(wm5100_lhpf1_mode,
+ WM5100_HPLPF1_1, WM5100_LHPF1_MODE_SHIFT,
+ wm5100_lhpf_mode_text);
-static const struct soc_enum wm5100_lhpf2_mode =
- SOC_ENUM_SINGLE(WM5100_HPLPF2_1, WM5100_LHPF2_MODE_SHIFT, 2,
- wm5100_lhpf_mode_text);
+static SOC_ENUM_SINGLE_DECL(wm5100_lhpf2_mode,
+ WM5100_HPLPF2_1, WM5100_LHPF2_MODE_SHIFT,
+ wm5100_lhpf_mode_text);
-static const struct soc_enum wm5100_lhpf3_mode =
- SOC_ENUM_SINGLE(WM5100_HPLPF3_1, WM5100_LHPF3_MODE_SHIFT, 2,
- wm5100_lhpf_mode_text);
+static SOC_ENUM_SINGLE_DECL(wm5100_lhpf3_mode,
+ WM5100_HPLPF3_1, WM5100_LHPF3_MODE_SHIFT,
+ wm5100_lhpf_mode_text);
-static const struct soc_enum wm5100_lhpf4_mode =
- SOC_ENUM_SINGLE(WM5100_HPLPF4_1, WM5100_LHPF4_MODE_SHIFT, 2,
- wm5100_lhpf_mode_text);
+static SOC_ENUM_SINGLE_DECL(wm5100_lhpf4_mode,
+ WM5100_HPLPF4_1, WM5100_LHPF4_MODE_SHIFT,
+ wm5100_lhpf_mode_text);
static const struct snd_kcontrol_new wm5100_snd_controls[] = {
SOC_SINGLE("IN1 High Performance Switch", WM5100_IN1L_CONTROL,
@@ -2100,6 +2100,7 @@ static void wm5100_micd_irq(struct wm5100_priv *wm5100)
int wm5100_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack)
{
struct wm5100_priv *wm5100 = snd_soc_codec_get_drvdata(codec);
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
if (jack) {
wm5100->jack = jack;
@@ -2117,9 +2118,14 @@ int wm5100_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack)
WM5100_ACCDET_RATE_MASK);
/* We need the charge pump to power MICBIAS */
- snd_soc_dapm_force_enable_pin(&codec->dapm, "CP2");
- snd_soc_dapm_force_enable_pin(&codec->dapm, "SYSCLK");
- snd_soc_dapm_sync(&codec->dapm);
+ snd_soc_dapm_mutex_lock(dapm);
+
+ snd_soc_dapm_force_enable_pin_unlocked(dapm, "CP2");
+ snd_soc_dapm_force_enable_pin_unlocked(dapm, "SYSCLK");
+
+ snd_soc_dapm_sync_unlocked(dapm);
+
+ snd_soc_dapm_mutex_unlock(dapm);
/* We start off just enabling microphone detection - even a
* plain headphone will trigger detection.
@@ -2337,13 +2343,6 @@ static int wm5100_probe(struct snd_soc_codec *codec)
int ret, i;
wm5100->codec = codec;
- codec->control_data = wm5100->regmap;
-
- ret = snd_soc_codec_set_cache_io(codec, 16, 16, SND_SOC_REGMAP);
- if (ret != 0) {
- dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
- return ret;
- }
for (i = 0; i < ARRAY_SIZE(wm5100_dig_vu); i++)
snd_soc_update_bits(codec, wm5100_dig_vu[i], WM5100_OUT_VU,
diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c
index ce9c8e1..dcf1d12 100644
--- a/sound/soc/codecs/wm5102.c
+++ b/sound/soc/codecs/wm5102.c
@@ -582,7 +582,7 @@ static int wm5102_sysclk_ev(struct snd_soc_dapm_widget *w,
{
struct snd_soc_codec *codec = w->codec;
struct arizona *arizona = dev_get_drvdata(codec->dev->parent);
- struct regmap *regmap = codec->control_data;
+ struct regmap *regmap = arizona->regmap;
const struct reg_default *patch = NULL;
int i, patch_size;
@@ -622,13 +622,16 @@ static const unsigned int wm5102_osr_val[] = {
static const struct soc_enum wm5102_hpout_osr[] = {
SOC_VALUE_ENUM_SINGLE(ARIZONA_OUTPUT_PATH_CONFIG_1L,
- ARIZONA_OUT1_OSR_SHIFT, 0x7, 3,
+ ARIZONA_OUT1_OSR_SHIFT, 0x7,
+ ARRAY_SIZE(wm5102_osr_text),
wm5102_osr_text, wm5102_osr_val),
SOC_VALUE_ENUM_SINGLE(ARIZONA_OUTPUT_PATH_CONFIG_2L,
- ARIZONA_OUT2_OSR_SHIFT, 0x7, 3,
+ ARIZONA_OUT2_OSR_SHIFT, 0x7,
+ ARRAY_SIZE(wm5102_osr_text),
wm5102_osr_text, wm5102_osr_val),
SOC_VALUE_ENUM_SINGLE(ARIZONA_OUTPUT_PATH_CONFIG_3L,
- ARIZONA_OUT3_OSR_SHIFT, 0x7, 3,
+ ARIZONA_OUT3_OSR_SHIFT, 0x7,
+ ARRAY_SIZE(wm5102_osr_text),
wm5102_osr_text, wm5102_osr_val),
};
@@ -685,15 +688,8 @@ ARIZONA_MIXER_CONTROLS("EQ2", ARIZONA_EQ2MIX_INPUT_1_SOURCE),
ARIZONA_MIXER_CONTROLS("EQ3", ARIZONA_EQ3MIX_INPUT_1_SOURCE),
ARIZONA_MIXER_CONTROLS("EQ4", ARIZONA_EQ4MIX_INPUT_1_SOURCE),
-SND_SOC_BYTES_MASK("EQ1 Coefficients", ARIZONA_EQ1_1, 21,
- ARIZONA_EQ1_ENA_MASK),
-SND_SOC_BYTES_MASK("EQ2 Coefficients", ARIZONA_EQ2_1, 21,
- ARIZONA_EQ2_ENA_MASK),
-SND_SOC_BYTES_MASK("EQ3 Coefficients", ARIZONA_EQ3_1, 21,
- ARIZONA_EQ3_ENA_MASK),
-SND_SOC_BYTES_MASK("EQ4 Coefficients", ARIZONA_EQ4_1, 21,
- ARIZONA_EQ4_ENA_MASK),
-
+SND_SOC_BYTES("EQ1 Coefficients", ARIZONA_EQ1_3, 19),
+SOC_SINGLE("EQ1 Mode Switch", ARIZONA_EQ1_2, ARIZONA_EQ1_B1_MODE, 1, 0),
SOC_SINGLE_TLV("EQ1 B1 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B1_GAIN_SHIFT,
24, 0, eq_tlv),
SOC_SINGLE_TLV("EQ1 B2 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B2_GAIN_SHIFT,
@@ -705,6 +701,8 @@ SOC_SINGLE_TLV("EQ1 B4 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B4_GAIN_SHIFT,
SOC_SINGLE_TLV("EQ1 B5 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B5_GAIN_SHIFT,
24, 0, eq_tlv),
+SND_SOC_BYTES("EQ2 Coefficients", ARIZONA_EQ2_3, 19),
+SOC_SINGLE("EQ2 Mode Switch", ARIZONA_EQ2_2, ARIZONA_EQ2_B1_MODE, 1, 0),
SOC_SINGLE_TLV("EQ2 B1 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B1_GAIN_SHIFT,
24, 0, eq_tlv),
SOC_SINGLE_TLV("EQ2 B2 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B2_GAIN_SHIFT,
@@ -716,6 +714,8 @@ SOC_SINGLE_TLV("EQ2 B4 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B4_GAIN_SHIFT,
SOC_SINGLE_TLV("EQ2 B5 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B5_GAIN_SHIFT,
24, 0, eq_tlv),
+SND_SOC_BYTES("EQ3 Coefficients", ARIZONA_EQ3_3, 19),
+SOC_SINGLE("EQ3 Mode Switch", ARIZONA_EQ3_2, ARIZONA_EQ3_B1_MODE, 1, 0),
SOC_SINGLE_TLV("EQ3 B1 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B1_GAIN_SHIFT,
24, 0, eq_tlv),
SOC_SINGLE_TLV("EQ3 B2 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B2_GAIN_SHIFT,
@@ -727,6 +727,8 @@ SOC_SINGLE_TLV("EQ3 B4 Volume", ARIZONA_EQ3_2, ARIZONA_EQ3_B4_GAIN_SHIFT,
SOC_SINGLE_TLV("EQ3 B5 Volume", ARIZONA_EQ3_2, ARIZONA_EQ3_B5_GAIN_SHIFT,
24, 0, eq_tlv),
+SND_SOC_BYTES("EQ4 Coefficients", ARIZONA_EQ4_3, 19),
+SOC_SINGLE("EQ4 Mode Switch", ARIZONA_EQ4_2, ARIZONA_EQ4_B1_MODE, 1, 0),
SOC_SINGLE_TLV("EQ4 B1 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B1_GAIN_SHIFT,
24, 0, eq_tlv),
SOC_SINGLE_TLV("EQ4 B2 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B2_GAIN_SHIFT,
@@ -1758,9 +1760,7 @@ static int wm5102_codec_probe(struct snd_soc_codec *codec)
struct wm5102_priv *priv = snd_soc_codec_get_drvdata(codec);
int ret;
- codec->control_data = priv->core.arizona->regmap;
-
- ret = snd_soc_codec_set_cache_io(codec, 32, 16, SND_SOC_REGMAP);
+ ret = snd_soc_codec_set_cache_io(codec, priv->core.arizona->regmap);
if (ret != 0)
return ret;
diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c
index 2c3c962..df5a38d 100644
--- a/sound/soc/codecs/wm5110.c
+++ b/sound/soc/codecs/wm5110.c
@@ -136,7 +136,7 @@ static int wm5110_sysclk_ev(struct snd_soc_dapm_widget *w,
{
struct snd_soc_codec *codec = w->codec;
struct arizona *arizona = dev_get_drvdata(codec->dev->parent);
- struct regmap *regmap = codec->control_data;
+ struct regmap *regmap = arizona->regmap;
const struct reg_default *patch = NULL;
int i, patch_size;
@@ -247,15 +247,8 @@ ARIZONA_MIXER_CONTROLS("EQ2", ARIZONA_EQ2MIX_INPUT_1_SOURCE),
ARIZONA_MIXER_CONTROLS("EQ3", ARIZONA_EQ3MIX_INPUT_1_SOURCE),
ARIZONA_MIXER_CONTROLS("EQ4", ARIZONA_EQ4MIX_INPUT_1_SOURCE),
-SND_SOC_BYTES_MASK("EQ1 Coefficients", ARIZONA_EQ1_1, 21,
- ARIZONA_EQ1_ENA_MASK),
-SND_SOC_BYTES_MASK("EQ2 Coefficients", ARIZONA_EQ2_1, 21,
- ARIZONA_EQ2_ENA_MASK),
-SND_SOC_BYTES_MASK("EQ3 Coefficients", ARIZONA_EQ3_1, 21,
- ARIZONA_EQ3_ENA_MASK),
-SND_SOC_BYTES_MASK("EQ4 Coefficients", ARIZONA_EQ4_1, 21,
- ARIZONA_EQ4_ENA_MASK),
-
+SND_SOC_BYTES("EQ1 Coefficients", ARIZONA_EQ1_3, 19),
+SOC_SINGLE("EQ1 Mode Switch", ARIZONA_EQ1_2, ARIZONA_EQ1_B1_MODE, 1, 0),
SOC_SINGLE_TLV("EQ1 B1 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B1_GAIN_SHIFT,
24, 0, eq_tlv),
SOC_SINGLE_TLV("EQ1 B2 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B2_GAIN_SHIFT,
@@ -267,6 +260,8 @@ SOC_SINGLE_TLV("EQ1 B4 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B4_GAIN_SHIFT,
SOC_SINGLE_TLV("EQ1 B5 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B5_GAIN_SHIFT,
24, 0, eq_tlv),
+SND_SOC_BYTES("EQ2 Coefficients", ARIZONA_EQ2_3, 19),
+SOC_SINGLE("EQ2 Mode Switch", ARIZONA_EQ2_2, ARIZONA_EQ2_B1_MODE, 1, 0),
SOC_SINGLE_TLV("EQ2 B1 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B1_GAIN_SHIFT,
24, 0, eq_tlv),
SOC_SINGLE_TLV("EQ2 B2 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B2_GAIN_SHIFT,
@@ -278,6 +273,8 @@ SOC_SINGLE_TLV("EQ2 B4 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B4_GAIN_SHIFT,
SOC_SINGLE_TLV("EQ2 B5 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B5_GAIN_SHIFT,
24, 0, eq_tlv),
+SND_SOC_BYTES("EQ3 Coefficients", ARIZONA_EQ3_3, 19),
+SOC_SINGLE("EQ3 Mode Switch", ARIZONA_EQ3_2, ARIZONA_EQ3_B1_MODE, 1, 0),
SOC_SINGLE_TLV("EQ3 B1 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B1_GAIN_SHIFT,
24, 0, eq_tlv),
SOC_SINGLE_TLV("EQ3 B2 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B2_GAIN_SHIFT,
@@ -289,6 +286,8 @@ SOC_SINGLE_TLV("EQ3 B4 Volume", ARIZONA_EQ3_2, ARIZONA_EQ3_B4_GAIN_SHIFT,
SOC_SINGLE_TLV("EQ3 B5 Volume", ARIZONA_EQ3_2, ARIZONA_EQ3_B5_GAIN_SHIFT,
24, 0, eq_tlv),
+SND_SOC_BYTES("EQ4 Coefficients", ARIZONA_EQ4_3, 19),
+SOC_SINGLE("EQ4 Mode Switch", ARIZONA_EQ4_2, ARIZONA_EQ4_B1_MODE, 1, 0),
SOC_SINGLE_TLV("EQ4 B1 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B1_GAIN_SHIFT,
24, 0, eq_tlv),
SOC_SINGLE_TLV("EQ4 B2 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B2_GAIN_SHIFT,
@@ -1588,10 +1587,9 @@ static int wm5110_codec_probe(struct snd_soc_codec *codec)
struct wm5110_priv *priv = snd_soc_codec_get_drvdata(codec);
int ret;
- codec->control_data = priv->core.arizona->regmap;
priv->core.arizona->dapm = &codec->dapm;
- ret = snd_soc_codec_set_cache_io(codec, 32, 16, SND_SOC_REGMAP);
+ ret = snd_soc_codec_set_cache_io(codec, priv->core.arizona->regmap);
if (ret != 0)
return ret;
diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c
index a183dcf..757256b 100644
--- a/sound/soc/codecs/wm8350.c
+++ b/sound/soc/codecs/wm8350.c
@@ -1505,9 +1505,7 @@ static int wm8350_codec_probe(struct snd_soc_codec *codec)
if (ret != 0)
return ret;
- codec->control_data = wm8350->regmap;
-
- snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP);
+ snd_soc_codec_set_cache_io(codec, wm8350->regmap);
/* Put the codec into reset if it wasn't already */
wm8350_clear_bits(wm8350, WM8350_POWER_MGMT_5, WM8350_CODEC_ENA);
diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c
index 6d684d9..146564f 100644
--- a/sound/soc/codecs/wm8400.c
+++ b/sound/soc/codecs/wm8400.c
@@ -1316,10 +1316,9 @@ static int wm8400_codec_probe(struct snd_soc_codec *codec)
snd_soc_codec_set_drvdata(codec, priv);
priv->wm8400 = wm8400;
- codec->control_data = wm8400->regmap;
priv->codec = codec;
- snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP);
+ snd_soc_codec_set_cache_io(codec, wm8400->regmap);
ret = devm_regulator_bulk_get(wm8400->dev,
ARRAY_SIZE(power), &power[0]);
diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c
index 7df7d45..1c1e328 100644
--- a/sound/soc/codecs/wm8510.c
+++ b/sound/soc/codecs/wm8510.c
@@ -589,20 +589,12 @@ static int wm8510_resume(struct snd_soc_codec *codec)
static int wm8510_probe(struct snd_soc_codec *codec)
{
- int ret;
-
- ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP);
- if (ret < 0) {
- printk(KERN_ERR "wm8510: failed to set cache I/O: %d\n", ret);
- return ret;
- }
-
wm8510_reset(codec);
/* power on device */
wm8510_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
- return ret;
+ return 0;
}
/* power down chip */
diff --git a/sound/soc/codecs/wm8523.c b/sound/soc/codecs/wm8523.c
index 74d106d..601ee81 100644
--- a/sound/soc/codecs/wm8523.c
+++ b/sound/soc/codecs/wm8523.c
@@ -75,8 +75,8 @@ static const char *wm8523_zd_count_text[] = {
"2048",
};
-static const struct soc_enum wm8523_zc_count =
- SOC_ENUM_SINGLE(WM8523_ZERO_DETECT, 0, 2, wm8523_zd_count_text);
+static SOC_ENUM_SINGLE_DECL(wm8523_zc_count, WM8523_ZERO_DETECT, 0,
+ wm8523_zd_count_text);
static const struct snd_kcontrol_new wm8523_controls[] = {
SOC_DOUBLE_R_TLV("Playback Volume", WM8523_DAC_GAINL, WM8523_DAC_GAINR,
@@ -392,18 +392,11 @@ static int wm8523_resume(struct snd_soc_codec *codec)
static int wm8523_probe(struct snd_soc_codec *codec)
{
struct wm8523_priv *wm8523 = snd_soc_codec_get_drvdata(codec);
- int ret;
wm8523->rate_constraint.list = &wm8523->rate_constraint_list[0];
wm8523->rate_constraint.count =
ARRAY_SIZE(wm8523->rate_constraint_list);
- 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;
- }
-
/* Change some default settings - latch VU and enable ZC */
snd_soc_update_bits(codec, WM8523_DAC_GAINR,
WM8523_DACR_VU, WM8523_DACR_VU);
diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c
index 318989a..af7ed8b 100644
--- a/sound/soc/codecs/wm8580.c
+++ b/sound/soc/codecs/wm8580.c
@@ -504,8 +504,7 @@ static int wm8580_paif_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 wm8580_priv *wm8580 = snd_soc_codec_get_drvdata(codec);
u16 paifa = 0;
u16 paifb = 0;
@@ -869,12 +868,6 @@ static int wm8580_probe(struct snd_soc_codec *codec)
struct wm8580_priv *wm8580 = snd_soc_codec_get_drvdata(codec);
int ret = 0;
- 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;
- }
-
ret = regulator_bulk_enable(ARRAY_SIZE(wm8580->supplies),
wm8580->supplies);
if (ret != 0) {
diff --git a/sound/soc/codecs/wm8711.c b/sound/soc/codecs/wm8711.c
index d99f948..b0fbcb3 100644
--- a/sound/soc/codecs/wm8711.c
+++ b/sound/soc/codecs/wm8711.c
@@ -201,7 +201,7 @@ static void wm8711_shutdown(struct snd_pcm_substream *substream,
struct snd_soc_codec *codec = dai->codec;
/* deactivate */
- if (!codec->active) {
+ if (!snd_soc_codec_is_active(codec)) {
udelay(50);
snd_soc_write(codec, WM8711_ACTIVE, 0x0);
}
@@ -367,12 +367,6 @@ static int wm8711_probe(struct snd_soc_codec *codec)
{
int ret;
- 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;
- }
-
ret = wm8711_reset(codec);
if (ret < 0) {
dev_err(codec->dev, "Failed to issue reset\n");
diff --git a/sound/soc/codecs/wm8728.c b/sound/soc/codecs/wm8728.c
index cd89033..bac7fc2 100644
--- a/sound/soc/codecs/wm8728.c
+++ b/sound/soc/codecs/wm8728.c
@@ -228,19 +228,10 @@ static int wm8728_resume(struct snd_soc_codec *codec)
static int wm8728_probe(struct snd_soc_codec *codec)
{
- int ret;
-
- ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP);
- if (ret < 0) {
- printk(KERN_ERR "wm8728: failed to configure cache I/O: %d\n",
- ret);
- return ret;
- }
-
/* power on device */
wm8728_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
- return ret;
+ return 0;
}
static int wm8728_remove(struct snd_soc_codec *codec)
diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c
index 0297203..d74f439 100644
--- a/sound/soc/codecs/wm8731.c
+++ b/sound/soc/codecs/wm8731.c
@@ -83,8 +83,8 @@ static bool wm8731_writeable(struct device *dev, unsigned int reg)
static const char *wm8731_input_select[] = {"Line In", "Mic"};
-static const struct soc_enum wm8731_insel_enum =
- SOC_ENUM_SINGLE(WM8731_APANA, 2, 2, wm8731_input_select);
+static SOC_ENUM_SINGLE_DECL(wm8731_insel_enum,
+ WM8731_APANA, 2, wm8731_input_select);
static int wm8731_deemph[] = { 0, 32000, 44100, 48000 };
@@ -583,13 +583,6 @@ static int wm8731_probe(struct snd_soc_codec *codec)
struct wm8731_priv *wm8731 = snd_soc_codec_get_drvdata(codec);
int ret = 0, i;
- codec->control_data = wm8731->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;
- }
-
for (i = 0; i < ARRAY_SIZE(wm8731->supplies); i++)
wm8731->supplies[i].supply = wm8731_supply_names[i];
diff --git a/sound/soc/codecs/wm8737.c b/sound/soc/codecs/wm8737.c
index 2f167a8..b27f26cd 100644
--- a/sound/soc/codecs/wm8737.c
+++ b/sound/soc/codecs/wm8737.c
@@ -99,29 +99,29 @@ static const char *micbias_enum_text[] = {
"100%",
};
-static const struct soc_enum micbias_enum =
- SOC_ENUM_SINGLE(WM8737_MIC_PREAMP_CONTROL, 0, 4, micbias_enum_text);
+static SOC_ENUM_SINGLE_DECL(micbias_enum,
+ WM8737_MIC_PREAMP_CONTROL, 0, micbias_enum_text);
static const char *low_cutoff_text[] = {
"Low", "High"
};
-static const struct soc_enum low_3d =
- SOC_ENUM_SINGLE(WM8737_3D_ENHANCE, 6, 2, low_cutoff_text);
+static SOC_ENUM_SINGLE_DECL(low_3d,
+ WM8737_3D_ENHANCE, 6, low_cutoff_text);
static const char *high_cutoff_text[] = {
"High", "Low"
};
-static const struct soc_enum high_3d =
- SOC_ENUM_SINGLE(WM8737_3D_ENHANCE, 5, 2, high_cutoff_text);
+static SOC_ENUM_SINGLE_DECL(high_3d,
+ WM8737_3D_ENHANCE, 5, high_cutoff_text);
static const char *alc_fn_text[] = {
"Disabled", "Right", "Left", "Stereo"
};
-static const struct soc_enum alc_fn =
- SOC_ENUM_SINGLE(WM8737_ALC1, 7, 4, alc_fn_text);
+static SOC_ENUM_SINGLE_DECL(alc_fn,
+ WM8737_ALC1, 7, alc_fn_text);
static const char *alc_hold_text[] = {
"0", "2.67ms", "5.33ms", "10.66ms", "21.32ms", "42.64ms", "85.28ms",
@@ -129,24 +129,24 @@ static const char *alc_hold_text[] = {
"10.916s", "21.832s", "43.691s"
};
-static const struct soc_enum alc_hold =
- SOC_ENUM_SINGLE(WM8737_ALC2, 0, 16, alc_hold_text);
+static SOC_ENUM_SINGLE_DECL(alc_hold,
+ WM8737_ALC2, 0, alc_hold_text);
static const char *alc_atk_text[] = {
"8.4ms", "16.8ms", "33.6ms", "67.2ms", "134.4ms", "268.8ms", "537.6ms",
"1.075s", "2.15s", "4.3s", "8.6s"
};
-static const struct soc_enum alc_atk =
- SOC_ENUM_SINGLE(WM8737_ALC3, 0, 11, alc_atk_text);
+static SOC_ENUM_SINGLE_DECL(alc_atk,
+ WM8737_ALC3, 0, alc_atk_text);
static const char *alc_dcy_text[] = {
"33.6ms", "67.2ms", "134.4ms", "268.8ms", "537.6ms", "1.075s", "2.15s",
"4.3s", "8.6s", "17.2s", "34.41s"
};
-static const struct soc_enum alc_dcy =
- SOC_ENUM_SINGLE(WM8737_ALC3, 4, 11, alc_dcy_text);
+static SOC_ENUM_SINGLE_DECL(alc_dcy,
+ WM8737_ALC3, 4, alc_dcy_text);
static const struct snd_kcontrol_new wm8737_snd_controls[] = {
SOC_DOUBLE_R_TLV("Mic Boost Volume", WM8737_AUDIO_PATH_L, WM8737_AUDIO_PATH_R,
@@ -191,8 +191,8 @@ static const char *linsel_text[] = {
"LINPUT1", "LINPUT2", "LINPUT3", "LINPUT1 DC",
};
-static const struct soc_enum linsel_enum =
- SOC_ENUM_SINGLE(WM8737_AUDIO_PATH_L, 7, 4, linsel_text);
+static SOC_ENUM_SINGLE_DECL(linsel_enum,
+ WM8737_AUDIO_PATH_L, 7, linsel_text);
static const struct snd_kcontrol_new linsel_mux =
SOC_DAPM_ENUM("LINSEL", linsel_enum);
@@ -202,8 +202,8 @@ static const char *rinsel_text[] = {
"RINPUT1", "RINPUT2", "RINPUT3", "RINPUT1 DC",
};
-static const struct soc_enum rinsel_enum =
- SOC_ENUM_SINGLE(WM8737_AUDIO_PATH_R, 7, 4, rinsel_text);
+static SOC_ENUM_SINGLE_DECL(rinsel_enum,
+ WM8737_AUDIO_PATH_R, 7, rinsel_text);
static const struct snd_kcontrol_new rinsel_mux =
SOC_DAPM_ENUM("RINSEL", rinsel_enum);
@@ -212,15 +212,15 @@ static const char *bypass_text[] = {
"Direct", "Preamp"
};
-static const struct soc_enum lbypass_enum =
- SOC_ENUM_SINGLE(WM8737_MIC_PREAMP_CONTROL, 2, 2, bypass_text);
+static SOC_ENUM_SINGLE_DECL(lbypass_enum,
+ WM8737_MIC_PREAMP_CONTROL, 2, bypass_text);
static const struct snd_kcontrol_new lbypass_mux =
SOC_DAPM_ENUM("Left Bypass", lbypass_enum);
-static const struct soc_enum rbypass_enum =
- SOC_ENUM_SINGLE(WM8737_MIC_PREAMP_CONTROL, 3, 2, bypass_text);
+static SOC_ENUM_SINGLE_DECL(rbypass_enum,
+ WM8737_MIC_PREAMP_CONTROL, 3, bypass_text);
static const struct snd_kcontrol_new rbypass_mux =
SOC_DAPM_ENUM("Left Bypass", rbypass_enum);
@@ -570,12 +570,6 @@ static int wm8737_probe(struct snd_soc_codec *codec)
struct wm8737_priv *wm8737 = snd_soc_codec_get_drvdata(codec);
int ret;
- 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;
- }
-
ret = regulator_bulk_enable(ARRAY_SIZE(wm8737->supplies),
wm8737->supplies);
if (ret != 0) {
@@ -644,7 +638,7 @@ static const struct regmap_config wm8737_regmap = {
.volatile_reg = wm8737_volatile,
};
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+#if IS_ENABLED(CONFIG_I2C)
static int wm8737_i2c_probe(struct i2c_client *i2c,
const struct i2c_device_id *id)
{
@@ -758,7 +752,7 @@ static struct spi_driver wm8737_spi_driver = {
static int __init wm8737_modinit(void)
{
int ret;
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+#if IS_ENABLED(CONFIG_I2C)
ret = i2c_add_driver(&wm8737_i2c_driver);
if (ret != 0) {
printk(KERN_ERR "Failed to register WM8737 I2C driver: %d\n",
@@ -781,7 +775,7 @@ static void __exit wm8737_exit(void)
#if defined(CONFIG_SPI_MASTER)
spi_unregister_driver(&wm8737_spi_driver);
#endif
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+#if IS_ENABLED(CONFIG_I2C)
i2c_del_driver(&wm8737_i2c_driver);
#endif
}
diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c
index 2895c8d..b33542a 100644
--- a/sound/soc/codecs/wm8741.c
+++ b/sound/soc/codecs/wm8741.c
@@ -44,7 +44,7 @@ struct wm8741_priv {
struct regmap *regmap;
struct regulator_bulk_data supplies[WM8741_NUM_SUPPLIES];
unsigned int sysclk;
- struct snd_pcm_hw_constraint_list *sysclk_constraints;
+ const struct snd_pcm_hw_constraint_list *sysclk_constraints;
};
static const struct reg_default wm8741_reg_defaults[] = {
@@ -122,74 +122,74 @@ static struct {
{ 6, 768 },
};
-static unsigned int rates_11289[] = {
+static const unsigned int rates_11289[] = {
44100, 88235,
};
-static struct snd_pcm_hw_constraint_list constraints_11289 = {
+static const struct snd_pcm_hw_constraint_list constraints_11289 = {
.count = ARRAY_SIZE(rates_11289),
.list = rates_11289,
};
-static unsigned int rates_12288[] = {
+static const unsigned int rates_12288[] = {
32000, 48000, 96000,
};
-static struct snd_pcm_hw_constraint_list constraints_12288 = {
+static const struct snd_pcm_hw_constraint_list constraints_12288 = {
.count = ARRAY_SIZE(rates_12288),
.list = rates_12288,
};
-static unsigned int rates_16384[] = {
+static const unsigned int rates_16384[] = {
32000,
};
-static struct snd_pcm_hw_constraint_list constraints_16384 = {
+static const struct snd_pcm_hw_constraint_list constraints_16384 = {
.count = ARRAY_SIZE(rates_16384),
.list = rates_16384,
};
-static unsigned int rates_16934[] = {
+static const unsigned int rates_16934[] = {
44100, 88235,
};
-static struct snd_pcm_hw_constraint_list constraints_16934 = {
+static const struct snd_pcm_hw_constraint_list constraints_16934 = {
.count = ARRAY_SIZE(rates_16934),
.list = rates_16934,
};
-static unsigned int rates_18432[] = {
+static const unsigned int rates_18432[] = {
48000, 96000,
};
-static struct snd_pcm_hw_constraint_list constraints_18432 = {
+static const struct snd_pcm_hw_constraint_list constraints_18432 = {
.count = ARRAY_SIZE(rates_18432),
.list = rates_18432,
};
-static unsigned int rates_22579[] = {
+static const unsigned int rates_22579[] = {
44100, 88235, 1764000
};
-static struct snd_pcm_hw_constraint_list constraints_22579 = {
+static const struct snd_pcm_hw_constraint_list constraints_22579 = {
.count = ARRAY_SIZE(rates_22579),
.list = rates_22579,
};
-static unsigned int rates_24576[] = {
+static const unsigned int rates_24576[] = {
32000, 48000, 96000, 192000
};
-static struct snd_pcm_hw_constraint_list constraints_24576 = {
+static const struct snd_pcm_hw_constraint_list constraints_24576 = {
.count = ARRAY_SIZE(rates_24576),
.list = rates_24576,
};
-static unsigned int rates_36864[] = {
+static const unsigned int rates_36864[] = {
48000, 96000, 19200
};
-static struct snd_pcm_hw_constraint_list constraints_36864 = {
+static const struct snd_pcm_hw_constraint_list constraints_36864 = {
.count = ARRAY_SIZE(rates_36864),
.list = rates_36864,
};
@@ -429,12 +429,6 @@ static int wm8741_probe(struct snd_soc_codec *codec)
goto err_get;
}
- 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);
- goto err_enable;
- }
-
ret = wm8741_reset(codec);
if (ret < 0) {
dev_err(codec->dev, "Failed to issue reset\n");
diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c
index 78616a6..33990b6 100644
--- a/sound/soc/codecs/wm8750.c
+++ b/sound/soc/codecs/wm8750.c
@@ -702,12 +702,6 @@ static int wm8750_probe(struct snd_soc_codec *codec)
{
int ret;
- ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP);
- if (ret < 0) {
- printk(KERN_ERR "wm8750: failed to set cache I/O: %d\n", ret);
- return ret;
- }
-
ret = wm8750_reset(codec);
if (ret < 0) {
printk(KERN_ERR "wm8750: failed to reset: %d\n", ret);
diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c
index be85da9..cbb8d55 100644
--- a/sound/soc/codecs/wm8753.c
+++ b/sound/soc/codecs/wm8753.c
@@ -251,7 +251,7 @@ static int wm8753_set_dai(struct snd_kcontrol *kcontrol,
if (wm8753->dai_func == ucontrol->value.integer.value[0])
return 0;
- if (codec->active)
+ if (snd_soc_codec_is_active(codec))
return -EBUSY;
ioctl = snd_soc_read(codec, WM8753_IOCTL);
@@ -1314,7 +1314,7 @@ static int wm8753_mute(struct snd_soc_dai *dai, int mute)
/* the digital mute covers the HiFi and Voice DAC's on the WM8753.
* make sure we check if they are not both active when we mute */
if (mute && wm8753->dai_func == 1) {
- if (!codec->active)
+ if (!snd_soc_codec_is_active(codec))
snd_soc_write(codec, WM8753_DAC, mute_reg | 0x8);
} else {
if (mute)
@@ -1440,7 +1440,6 @@ static void wm8753_work(struct work_struct *work)
static int wm8753_suspend(struct snd_soc_codec *codec)
{
wm8753_set_bias_level(codec, SND_SOC_BIAS_OFF);
- codec->cache_sync = 1;
return 0;
}
@@ -1471,13 +1470,6 @@ static int wm8753_probe(struct snd_soc_codec *codec)
INIT_DELAYED_WORK(&codec->dapm.delayed_work, wm8753_work);
- codec->control_data = wm8753->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;
- }
-
ret = wm8753_reset(codec);
if (ret < 0) {
dev_err(codec->dev, "Failed to issue reset: %d\n", ret);
diff --git a/sound/soc/codecs/wm8770.c b/sound/soc/codecs/wm8770.c
index 5bce210..c61aeb3 100644
--- a/sound/soc/codecs/wm8770.c
+++ b/sound/soc/codecs/wm8770.c
@@ -580,12 +580,6 @@ static int wm8770_probe(struct snd_soc_codec *codec)
wm8770 = snd_soc_codec_get_drvdata(codec);
wm8770->codec = codec;
- 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;
- }
-
ret = regulator_bulk_enable(ARRAY_SIZE(wm8770->supplies),
wm8770->supplies);
if (ret) {
diff --git a/sound/soc/codecs/wm8776.c b/sound/soc/codecs/wm8776.c
index ef82467..70952ce 100644
--- a/sound/soc/codecs/wm8776.c
+++ b/sound/soc/codecs/wm8776.c
@@ -430,12 +430,6 @@ static int wm8776_probe(struct snd_soc_codec *codec)
{
int ret = 0;
- 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;
- }
-
ret = wm8776_reset(codec);
if (ret < 0) {
dev_err(codec->dev, "Failed to issue reset: %d\n", ret);
diff --git a/sound/soc/codecs/wm8804.c b/sound/soc/codecs/wm8804.c
index 9bc8206..ee76f0f 100644
--- a/sound/soc/codecs/wm8804.c
+++ b/sound/soc/codecs/wm8804.c
@@ -92,7 +92,7 @@ WM8804_REGULATOR_EVENT(0)
WM8804_REGULATOR_EVENT(1)
static const char *txsrc_text[] = { "S/PDIF RX", "AIF" };
-static const SOC_ENUM_SINGLE_EXT_DECL(txsrc, txsrc_text);
+static SOC_ENUM_SINGLE_EXT_DECL(txsrc, txsrc_text);
static const struct snd_kcontrol_new wm8804_snd_controls[] = {
SOC_ENUM_EXT("Input Source", txsrc, txsrc_get, txsrc_put),
@@ -546,14 +546,6 @@ static int wm8804_probe(struct snd_soc_codec *codec)
wm8804 = snd_soc_codec_get_drvdata(codec);
- codec->control_data = wm8804->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;
- }
-
for (i = 0; i < ARRAY_SIZE(wm8804->supplies); i++)
wm8804->supplies[i].supply = wm8804_supply_names[i];
diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c
index 43c2201..d09fdce 100644
--- a/sound/soc/codecs/wm8900.c
+++ b/sound/soc/codecs/wm8900.c
@@ -1178,13 +1178,7 @@ static int wm8900_resume(struct snd_soc_codec *codec)
static int wm8900_probe(struct snd_soc_codec *codec)
{
- int ret = 0, reg;
-
- 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;
- }
+ int reg;
reg = snd_soc_read(codec, WM8900_REG_ID);
if (reg != 0x8900) {
diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c
index eebcb1d..b0084a1 100644
--- a/sound/soc/codecs/wm8903.c
+++ b/sound/soc/codecs/wm8903.c
@@ -489,28 +489,28 @@ static const char *hpf_mode_text[] = {
"Hi-fi", "Voice 1", "Voice 2", "Voice 3"
};
-static const struct soc_enum hpf_mode =
- SOC_ENUM_SINGLE(WM8903_ADC_DIGITAL_0, 5, 4, hpf_mode_text);
+static SOC_ENUM_SINGLE_DECL(hpf_mode,
+ WM8903_ADC_DIGITAL_0, 5, hpf_mode_text);
static const char *osr_text[] = {
"Low power", "High performance"
};
-static const struct soc_enum adc_osr =
- SOC_ENUM_SINGLE(WM8903_ANALOGUE_ADC_0, 0, 2, osr_text);
+static SOC_ENUM_SINGLE_DECL(adc_osr,
+ WM8903_ANALOGUE_ADC_0, 0, osr_text);
-static const struct soc_enum dac_osr =
- SOC_ENUM_SINGLE(WM8903_DAC_DIGITAL_1, 0, 2, osr_text);
+static SOC_ENUM_SINGLE_DECL(dac_osr,
+ WM8903_DAC_DIGITAL_1, 0, osr_text);
static const char *drc_slope_text[] = {
"1", "1/2", "1/4", "1/8", "1/16", "0"
};
-static const struct soc_enum drc_slope_r0 =
- SOC_ENUM_SINGLE(WM8903_DRC_2, 3, 6, drc_slope_text);
+static SOC_ENUM_SINGLE_DECL(drc_slope_r0,
+ WM8903_DRC_2, 3, drc_slope_text);
-static const struct soc_enum drc_slope_r1 =
- SOC_ENUM_SINGLE(WM8903_DRC_2, 0, 6, drc_slope_text);
+static SOC_ENUM_SINGLE_DECL(drc_slope_r1,
+ WM8903_DRC_2, 0, drc_slope_text);
static const char *drc_attack_text[] = {
"instantaneous",
@@ -518,125 +518,125 @@ static const char *drc_attack_text[] = {
"46.4ms", "92.8ms", "185.6ms"
};
-static const struct soc_enum drc_attack =
- SOC_ENUM_SINGLE(WM8903_DRC_1, 12, 11, drc_attack_text);
+static SOC_ENUM_SINGLE_DECL(drc_attack,
+ WM8903_DRC_1, 12, drc_attack_text);
static const char *drc_decay_text[] = {
"186ms", "372ms", "743ms", "1.49s", "2.97s", "5.94s", "11.89s",
"23.87s", "47.56s"
};
-static const struct soc_enum drc_decay =
- SOC_ENUM_SINGLE(WM8903_DRC_1, 8, 9, drc_decay_text);
+static SOC_ENUM_SINGLE_DECL(drc_decay,
+ WM8903_DRC_1, 8, drc_decay_text);
static const char *drc_ff_delay_text[] = {
"5 samples", "9 samples"
};
-static const struct soc_enum drc_ff_delay =
- SOC_ENUM_SINGLE(WM8903_DRC_0, 5, 2, drc_ff_delay_text);
+static SOC_ENUM_SINGLE_DECL(drc_ff_delay,
+ WM8903_DRC_0, 5, drc_ff_delay_text);
static const char *drc_qr_decay_text[] = {
"0.725ms", "1.45ms", "5.8ms"
};
-static const struct soc_enum drc_qr_decay =
- SOC_ENUM_SINGLE(WM8903_DRC_1, 4, 3, drc_qr_decay_text);
+static SOC_ENUM_SINGLE_DECL(drc_qr_decay,
+ WM8903_DRC_1, 4, drc_qr_decay_text);
static const char *drc_smoothing_text[] = {
"Low", "Medium", "High"
};
-static const struct soc_enum drc_smoothing =
- SOC_ENUM_SINGLE(WM8903_DRC_0, 11, 3, drc_smoothing_text);
+static SOC_ENUM_SINGLE_DECL(drc_smoothing,
+ WM8903_DRC_0, 11, drc_smoothing_text);
static const char *soft_mute_text[] = {
"Fast (fs/2)", "Slow (fs/32)"
};
-static const struct soc_enum soft_mute =
- SOC_ENUM_SINGLE(WM8903_DAC_DIGITAL_1, 10, 2, soft_mute_text);
+static SOC_ENUM_SINGLE_DECL(soft_mute,
+ WM8903_DAC_DIGITAL_1, 10, soft_mute_text);
static const char *mute_mode_text[] = {
"Hard", "Soft"
};
-static const struct soc_enum mute_mode =
- SOC_ENUM_SINGLE(WM8903_DAC_DIGITAL_1, 9, 2, mute_mode_text);
+static SOC_ENUM_SINGLE_DECL(mute_mode,
+ WM8903_DAC_DIGITAL_1, 9, mute_mode_text);
static const char *companding_text[] = {
"ulaw", "alaw"
};
-static const struct soc_enum dac_companding =
- SOC_ENUM_SINGLE(WM8903_AUDIO_INTERFACE_0, 0, 2, companding_text);
+static SOC_ENUM_SINGLE_DECL(dac_companding,
+ WM8903_AUDIO_INTERFACE_0, 0, companding_text);
-static const struct soc_enum adc_companding =
- SOC_ENUM_SINGLE(WM8903_AUDIO_INTERFACE_0, 2, 2, companding_text);
+static SOC_ENUM_SINGLE_DECL(adc_companding,
+ WM8903_AUDIO_INTERFACE_0, 2, companding_text);
static const char *input_mode_text[] = {
"Single-Ended", "Differential Line", "Differential Mic"
};
-static const struct soc_enum linput_mode_enum =
- SOC_ENUM_SINGLE(WM8903_ANALOGUE_LEFT_INPUT_1, 0, 3, input_mode_text);
+static SOC_ENUM_SINGLE_DECL(linput_mode_enum,
+ WM8903_ANALOGUE_LEFT_INPUT_1, 0, input_mode_text);
-static const struct soc_enum rinput_mode_enum =
- SOC_ENUM_SINGLE(WM8903_ANALOGUE_RIGHT_INPUT_1, 0, 3, input_mode_text);
+static SOC_ENUM_SINGLE_DECL(rinput_mode_enum,
+ WM8903_ANALOGUE_RIGHT_INPUT_1, 0, input_mode_text);
static const char *linput_mux_text[] = {
"IN1L", "IN2L", "IN3L"
};
-static const struct soc_enum linput_enum =
- SOC_ENUM_SINGLE(WM8903_ANALOGUE_LEFT_INPUT_1, 2, 3, linput_mux_text);
+static SOC_ENUM_SINGLE_DECL(linput_enum,
+ WM8903_ANALOGUE_LEFT_INPUT_1, 2, linput_mux_text);
-static const struct soc_enum linput_inv_enum =
- SOC_ENUM_SINGLE(WM8903_ANALOGUE_LEFT_INPUT_1, 4, 3, linput_mux_text);
+static SOC_ENUM_SINGLE_DECL(linput_inv_enum,
+ WM8903_ANALOGUE_LEFT_INPUT_1, 4, linput_mux_text);
static const char *rinput_mux_text[] = {
"IN1R", "IN2R", "IN3R"
};
-static const struct soc_enum rinput_enum =
- SOC_ENUM_SINGLE(WM8903_ANALOGUE_RIGHT_INPUT_1, 2, 3, rinput_mux_text);
+static SOC_ENUM_SINGLE_DECL(rinput_enum,
+ WM8903_ANALOGUE_RIGHT_INPUT_1, 2, rinput_mux_text);
-static const struct soc_enum rinput_inv_enum =
- SOC_ENUM_SINGLE(WM8903_ANALOGUE_RIGHT_INPUT_1, 4, 3, rinput_mux_text);
+static SOC_ENUM_SINGLE_DECL(rinput_inv_enum,
+ WM8903_ANALOGUE_RIGHT_INPUT_1, 4, rinput_mux_text);
static const char *sidetone_text[] = {
"None", "Left", "Right"
};
-static const struct soc_enum lsidetone_enum =
- SOC_ENUM_SINGLE(WM8903_DAC_DIGITAL_0, 2, 3, sidetone_text);
+static SOC_ENUM_SINGLE_DECL(lsidetone_enum,
+ WM8903_DAC_DIGITAL_0, 2, sidetone_text);
-static const struct soc_enum rsidetone_enum =
- SOC_ENUM_SINGLE(WM8903_DAC_DIGITAL_0, 0, 3, sidetone_text);
+static SOC_ENUM_SINGLE_DECL(rsidetone_enum,
+ WM8903_DAC_DIGITAL_0, 0, sidetone_text);
static const char *adcinput_text[] = {
"ADC", "DMIC"
};
-static const struct soc_enum adcinput_enum =
- SOC_ENUM_SINGLE(WM8903_CLOCK_RATE_TEST_4, 9, 2, adcinput_text);
+static SOC_ENUM_SINGLE_DECL(adcinput_enum,
+ WM8903_CLOCK_RATE_TEST_4, 9, adcinput_text);
static const char *aif_text[] = {
"Left", "Right"
};
-static const struct soc_enum lcapture_enum =
- SOC_ENUM_SINGLE(WM8903_AUDIO_INTERFACE_0, 7, 2, aif_text);
+static SOC_ENUM_SINGLE_DECL(lcapture_enum,
+ WM8903_AUDIO_INTERFACE_0, 7, aif_text);
-static const struct soc_enum rcapture_enum =
- SOC_ENUM_SINGLE(WM8903_AUDIO_INTERFACE_0, 6, 2, aif_text);
+static SOC_ENUM_SINGLE_DECL(rcapture_enum,
+ WM8903_AUDIO_INTERFACE_0, 6, aif_text);
-static const struct soc_enum lplay_enum =
- SOC_ENUM_SINGLE(WM8903_AUDIO_INTERFACE_0, 5, 2, aif_text);
+static SOC_ENUM_SINGLE_DECL(lplay_enum,
+ WM8903_AUDIO_INTERFACE_0, 5, aif_text);
-static const struct soc_enum rplay_enum =
- SOC_ENUM_SINGLE(WM8903_AUDIO_INTERFACE_0, 4, 2, aif_text);
+static SOC_ENUM_SINGLE_DECL(rplay_enum,
+ WM8903_AUDIO_INTERFACE_0, 4, aif_text);
static const struct snd_kcontrol_new wm8903_snd_controls[] = {
@@ -1897,21 +1897,13 @@ static void wm8903_free_gpio(struct wm8903_priv *wm8903)
static int wm8903_probe(struct snd_soc_codec *codec)
{
struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec);
- int ret;
wm8903->codec = codec;
- codec->control_data = wm8903->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;
- }
/* power on device */
wm8903_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
- return ret;
+ return 0;
}
/* power down chip */
diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c
index 53bbfac..49c35c3 100644
--- a/sound/soc/codecs/wm8904.c
+++ b/sound/soc/codecs/wm8904.c
@@ -552,18 +552,20 @@ static const char *input_mode_text[] = {
"Single-Ended", "Differential Line", "Differential Mic"
};
-static const struct soc_enum lin_mode =
- SOC_ENUM_SINGLE(WM8904_ANALOGUE_LEFT_INPUT_1, 0, 3, input_mode_text);
+static SOC_ENUM_SINGLE_DECL(lin_mode,
+ WM8904_ANALOGUE_LEFT_INPUT_1, 0,
+ input_mode_text);
-static const struct soc_enum rin_mode =
- SOC_ENUM_SINGLE(WM8904_ANALOGUE_RIGHT_INPUT_1, 0, 3, input_mode_text);
+static SOC_ENUM_SINGLE_DECL(rin_mode,
+ WM8904_ANALOGUE_RIGHT_INPUT_1, 0,
+ input_mode_text);
static const char *hpf_mode_text[] = {
"Hi-fi", "Voice 1", "Voice 2", "Voice 3"
};
-static const struct soc_enum hpf_mode =
- SOC_ENUM_SINGLE(WM8904_ADC_DIGITAL_0, 5, 4, hpf_mode_text);
+static SOC_ENUM_SINGLE_DECL(hpf_mode, WM8904_ADC_DIGITAL_0, 5,
+ hpf_mode_text);
static int wm8904_adc_osr_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
@@ -611,8 +613,7 @@ static const char *drc_path_text[] = {
"ADC", "DAC"
};
-static const struct soc_enum drc_path =
- SOC_ENUM_SINGLE(WM8904_DRC_0, 14, 2, drc_path_text);
+static SOC_ENUM_SINGLE_DECL(drc_path, WM8904_DRC_0, 14, drc_path_text);
static const struct snd_kcontrol_new wm8904_dac_snd_controls[] = {
SOC_SINGLE_TLV("Digital Playback Boost Volume",
@@ -858,14 +859,14 @@ static const char *lin_text[] = {
"IN1L", "IN2L", "IN3L"
};
-static const struct soc_enum lin_enum =
- SOC_ENUM_SINGLE(WM8904_ANALOGUE_LEFT_INPUT_1, 2, 3, lin_text);
+static SOC_ENUM_SINGLE_DECL(lin_enum, WM8904_ANALOGUE_LEFT_INPUT_1, 2,
+ lin_text);
static const struct snd_kcontrol_new lin_mux =
SOC_DAPM_ENUM("Left Capture Mux", lin_enum);
-static const struct soc_enum lin_inv_enum =
- SOC_ENUM_SINGLE(WM8904_ANALOGUE_LEFT_INPUT_1, 4, 3, lin_text);
+static SOC_ENUM_SINGLE_DECL(lin_inv_enum, WM8904_ANALOGUE_LEFT_INPUT_1, 4,
+ lin_text);
static const struct snd_kcontrol_new lin_inv_mux =
SOC_DAPM_ENUM("Left Capture Inveting Mux", lin_inv_enum);
@@ -874,14 +875,14 @@ static const char *rin_text[] = {
"IN1R", "IN2R", "IN3R"
};
-static const struct soc_enum rin_enum =
- SOC_ENUM_SINGLE(WM8904_ANALOGUE_RIGHT_INPUT_1, 2, 3, rin_text);
+static SOC_ENUM_SINGLE_DECL(rin_enum, WM8904_ANALOGUE_RIGHT_INPUT_1, 2,
+ rin_text);
static const struct snd_kcontrol_new rin_mux =
SOC_DAPM_ENUM("Right Capture Mux", rin_enum);
-static const struct soc_enum rin_inv_enum =
- SOC_ENUM_SINGLE(WM8904_ANALOGUE_RIGHT_INPUT_1, 4, 3, rin_text);
+static SOC_ENUM_SINGLE_DECL(rin_inv_enum, WM8904_ANALOGUE_RIGHT_INPUT_1, 4,
+ rin_text);
static const struct snd_kcontrol_new rin_inv_mux =
SOC_DAPM_ENUM("Right Capture Inveting Mux", rin_inv_enum);
@@ -890,26 +891,26 @@ static const char *aif_text[] = {
"Left", "Right"
};
-static const struct soc_enum aifoutl_enum =
- SOC_ENUM_SINGLE(WM8904_AUDIO_INTERFACE_0, 7, 2, aif_text);
+static SOC_ENUM_SINGLE_DECL(aifoutl_enum, WM8904_AUDIO_INTERFACE_0, 7,
+ aif_text);
static const struct snd_kcontrol_new aifoutl_mux =
SOC_DAPM_ENUM("AIFOUTL Mux", aifoutl_enum);
-static const struct soc_enum aifoutr_enum =
- SOC_ENUM_SINGLE(WM8904_AUDIO_INTERFACE_0, 6, 2, aif_text);
+static SOC_ENUM_SINGLE_DECL(aifoutr_enum, WM8904_AUDIO_INTERFACE_0, 6,
+ aif_text);
static const struct snd_kcontrol_new aifoutr_mux =
SOC_DAPM_ENUM("AIFOUTR Mux", aifoutr_enum);
-static const struct soc_enum aifinl_enum =
- SOC_ENUM_SINGLE(WM8904_AUDIO_INTERFACE_0, 5, 2, aif_text);
+static SOC_ENUM_SINGLE_DECL(aifinl_enum, WM8904_AUDIO_INTERFACE_0, 5,
+ aif_text);
static const struct snd_kcontrol_new aifinl_mux =
SOC_DAPM_ENUM("AIFINL Mux", aifinl_enum);
-static const struct soc_enum aifinr_enum =
- SOC_ENUM_SINGLE(WM8904_AUDIO_INTERFACE_0, 4, 2, aif_text);
+static SOC_ENUM_SINGLE_DECL(aifinr_enum, WM8904_AUDIO_INTERFACE_0, 4,
+ aif_text);
static const struct snd_kcontrol_new aifinr_mux =
SOC_DAPM_ENUM("AIFINR Mux", aifinr_enum);
@@ -991,26 +992,26 @@ static const char *out_mux_text[] = {
"DAC", "Bypass"
};
-static const struct soc_enum hpl_enum =
- SOC_ENUM_SINGLE(WM8904_ANALOGUE_OUT12_ZC, 3, 2, out_mux_text);
+static SOC_ENUM_SINGLE_DECL(hpl_enum, WM8904_ANALOGUE_OUT12_ZC, 3,
+ out_mux_text);
static const struct snd_kcontrol_new hpl_mux =
SOC_DAPM_ENUM("HPL Mux", hpl_enum);
-static const struct soc_enum hpr_enum =
- SOC_ENUM_SINGLE(WM8904_ANALOGUE_OUT12_ZC, 2, 2, out_mux_text);
+static SOC_ENUM_SINGLE_DECL(hpr_enum, WM8904_ANALOGUE_OUT12_ZC, 2,
+ out_mux_text);
static const struct snd_kcontrol_new hpr_mux =
SOC_DAPM_ENUM("HPR Mux", hpr_enum);
-static const struct soc_enum linel_enum =
- SOC_ENUM_SINGLE(WM8904_ANALOGUE_OUT12_ZC, 1, 2, out_mux_text);
+static SOC_ENUM_SINGLE_DECL(linel_enum, WM8904_ANALOGUE_OUT12_ZC, 1,
+ out_mux_text);
static const struct snd_kcontrol_new linel_mux =
SOC_DAPM_ENUM("LINEL Mux", linel_enum);
-static const struct soc_enum liner_enum =
- SOC_ENUM_SINGLE(WM8904_ANALOGUE_OUT12_ZC, 0, 2, out_mux_text);
+static SOC_ENUM_SINGLE_DECL(liner_enum, WM8904_ANALOGUE_OUT12_ZC, 0,
+ out_mux_text);
static const struct snd_kcontrol_new liner_mux =
SOC_DAPM_ENUM("LINER Mux", liner_enum);
@@ -1019,14 +1020,14 @@ static const char *sidetone_text[] = {
"None", "Left", "Right"
};
-static const struct soc_enum dacl_sidetone_enum =
- SOC_ENUM_SINGLE(WM8904_DAC_DIGITAL_0, 2, 3, sidetone_text);
+static SOC_ENUM_SINGLE_DECL(dacl_sidetone_enum, WM8904_DAC_DIGITAL_0, 2,
+ sidetone_text);
static const struct snd_kcontrol_new dacl_sidetone_mux =
SOC_DAPM_ENUM("Left Sidetone Mux", dacl_sidetone_enum);
-static const struct soc_enum dacr_sidetone_enum =
- SOC_ENUM_SINGLE(WM8904_DAC_DIGITAL_0, 0, 3, sidetone_text);
+static SOC_ENUM_SINGLE_DECL(dacr_sidetone_enum, WM8904_DAC_DIGITAL_0, 0,
+ sidetone_text);
static const struct snd_kcontrol_new dacr_sidetone_mux =
SOC_DAPM_ENUM("Right Sidetone Mux", dacr_sidetone_enum);
@@ -1981,7 +1982,7 @@ static void wm8904_handle_retune_mobile_pdata(struct snd_soc_codec *codec)
dev_dbg(codec->dev, "Allocated %d unique ReTune Mobile names\n",
wm8904->num_retune_mobile_texts);
- wm8904->retune_mobile_enum.max = wm8904->num_retune_mobile_texts;
+ wm8904->retune_mobile_enum.items = wm8904->num_retune_mobile_texts;
wm8904->retune_mobile_enum.texts = wm8904->retune_mobile_texts;
ret = snd_soc_add_codec_controls(codec, &control, 1);
@@ -2022,7 +2023,7 @@ static void wm8904_handle_pdata(struct snd_soc_codec *codec)
for (i = 0; i < pdata->num_drc_cfgs; i++)
wm8904->drc_texts[i] = pdata->drc_cfgs[i].name;
- wm8904->drc_enum.max = pdata->num_drc_cfgs;
+ wm8904->drc_enum.items = pdata->num_drc_cfgs;
wm8904->drc_enum.texts = wm8904->drc_texts;
ret = snd_soc_add_codec_controls(codec, &control, 1);
@@ -2047,9 +2048,6 @@ static void wm8904_handle_pdata(struct snd_soc_codec *codec)
static int wm8904_probe(struct snd_soc_codec *codec)
{
struct wm8904_priv *wm8904 = snd_soc_codec_get_drvdata(codec);
- int ret;
-
- codec->control_data = wm8904->regmap;
switch (wm8904->devtype) {
case WM8904:
@@ -2063,12 +2061,6 @@ static int wm8904_probe(struct snd_soc_codec *codec)
return -EINVAL;
}
- 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;
- }
-
wm8904_handle_pdata(codec);
wm8904_add_widgets(codec);
diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c
index b404c26..fc6eec9 100644
--- a/sound/soc/codecs/wm8940.c
+++ b/sound/soc/codecs/wm8940.c
@@ -154,22 +154,22 @@ static const struct reg_default wm8940_reg_defaults[] = {
};
static const char *wm8940_companding[] = { "Off", "NC", "u-law", "A-law" };
-static const struct soc_enum wm8940_adc_companding_enum
-= SOC_ENUM_SINGLE(WM8940_COMPANDINGCTL, 1, 4, wm8940_companding);
-static const struct soc_enum wm8940_dac_companding_enum
-= SOC_ENUM_SINGLE(WM8940_COMPANDINGCTL, 3, 4, wm8940_companding);
+static SOC_ENUM_SINGLE_DECL(wm8940_adc_companding_enum,
+ WM8940_COMPANDINGCTL, 1, wm8940_companding);
+static SOC_ENUM_SINGLE_DECL(wm8940_dac_companding_enum,
+ WM8940_COMPANDINGCTL, 3, wm8940_companding);
static const char *wm8940_alc_mode_text[] = {"ALC", "Limiter"};
-static const struct soc_enum wm8940_alc_mode_enum
-= SOC_ENUM_SINGLE(WM8940_ALC3, 8, 2, wm8940_alc_mode_text);
+static SOC_ENUM_SINGLE_DECL(wm8940_alc_mode_enum,
+ WM8940_ALC3, 8, wm8940_alc_mode_text);
static const char *wm8940_mic_bias_level_text[] = {"0.9", "0.65"};
-static const struct soc_enum wm8940_mic_bias_level_enum
-= SOC_ENUM_SINGLE(WM8940_INPUTCTL, 8, 2, wm8940_mic_bias_level_text);
+static SOC_ENUM_SINGLE_DECL(wm8940_mic_bias_level_enum,
+ WM8940_INPUTCTL, 8, wm8940_mic_bias_level_text);
static const char *wm8940_filter_mode_text[] = {"Audio", "Application"};
-static const struct soc_enum wm8940_filter_mode_enum
-= SOC_ENUM_SINGLE(WM8940_ADC, 7, 2, wm8940_filter_mode_text);
+static SOC_ENUM_SINGLE_DECL(wm8940_filter_mode_enum,
+ WM8940_ADC, 7, wm8940_filter_mode_text);
static DECLARE_TLV_DB_SCALE(wm8940_spk_vol_tlv, -5700, 100, 1);
static DECLARE_TLV_DB_SCALE(wm8940_att_tlv, -1000, 1000, 0);
@@ -712,12 +712,6 @@ static int wm8940_probe(struct snd_soc_codec *codec)
int ret;
u16 reg;
- 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;
- }
-
ret = wm8940_reset(codec);
if (ret < 0) {
dev_err(codec->dev, "Failed to issue reset\n");
diff --git a/sound/soc/codecs/wm8955.c b/sound/soc/codecs/wm8955.c
index 82c8ba9..fecd4e4 100644
--- a/sound/soc/codecs/wm8955.c
+++ b/sound/soc/codecs/wm8955.c
@@ -416,22 +416,21 @@ static const char *bass_mode_text[] = {
"Linear", "Adaptive",
};
-static const struct soc_enum bass_mode =
- SOC_ENUM_SINGLE(WM8955_BASS_CONTROL, 7, 2, bass_mode_text);
+static SOC_ENUM_SINGLE_DECL(bass_mode, WM8955_BASS_CONTROL, 7, bass_mode_text);
static const char *bass_cutoff_text[] = {
"Low", "High"
};
-static const struct soc_enum bass_cutoff =
- SOC_ENUM_SINGLE(WM8955_BASS_CONTROL, 6, 2, bass_cutoff_text);
+static SOC_ENUM_SINGLE_DECL(bass_cutoff, WM8955_BASS_CONTROL, 6,
+ bass_cutoff_text);
static const char *treble_cutoff_text[] = {
"High", "Low"
};
-static const struct soc_enum treble_cutoff =
- SOC_ENUM_SINGLE(WM8955_TREBLE_CONTROL, 6, 2, treble_cutoff_text);
+static SOC_ENUM_SINGLE_DECL(treble_cutoff, WM8955_TREBLE_CONTROL, 2,
+ treble_cutoff_text);
static const DECLARE_TLV_DB_SCALE(digital_tlv, -12750, 50, 1);
static const DECLARE_TLV_DB_SCALE(atten_tlv, -600, 600, 0);
@@ -896,14 +895,6 @@ static int wm8955_probe(struct snd_soc_codec *codec)
struct wm8955_pdata *pdata = dev_get_platdata(codec->dev);
int ret, i;
- codec->control_data = wm8955->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;
- }
-
for (i = 0; i < ARRAY_SIZE(wm8955->supplies); i++)
wm8955->supplies[i].supply = wm8955_supply_names[i];
diff --git a/sound/soc/codecs/wm8958-dsp2.c b/sound/soc/codecs/wm8958-dsp2.c
index d4248e0..7ac2e51 100644
--- a/sound/soc/codecs/wm8958-dsp2.c
+++ b/sound/soc/codecs/wm8958-dsp2.c
@@ -944,7 +944,7 @@ void wm8958_dsp2_init(struct snd_soc_codec *codec)
for (i = 0; i < pdata->num_mbc_cfgs; i++)
wm8994->mbc_texts[i] = pdata->mbc_cfgs[i].name;
- wm8994->mbc_enum.max = pdata->num_mbc_cfgs;
+ wm8994->mbc_enum.items = pdata->num_mbc_cfgs;
wm8994->mbc_enum.texts = wm8994->mbc_texts;
ret = snd_soc_add_codec_controls(wm8994->hubs.codec,
@@ -973,7 +973,7 @@ void wm8958_dsp2_init(struct snd_soc_codec *codec)
for (i = 0; i < pdata->num_vss_cfgs; i++)
wm8994->vss_texts[i] = pdata->vss_cfgs[i].name;
- wm8994->vss_enum.max = pdata->num_vss_cfgs;
+ wm8994->vss_enum.items = pdata->num_vss_cfgs;
wm8994->vss_enum.texts = wm8994->vss_texts;
ret = snd_soc_add_codec_controls(wm8994->hubs.codec,
@@ -1003,7 +1003,7 @@ void wm8958_dsp2_init(struct snd_soc_codec *codec)
for (i = 0; i < pdata->num_vss_hpf_cfgs; i++)
wm8994->vss_hpf_texts[i] = pdata->vss_hpf_cfgs[i].name;
- wm8994->vss_hpf_enum.max = pdata->num_vss_hpf_cfgs;
+ wm8994->vss_hpf_enum.items = pdata->num_vss_hpf_cfgs;
wm8994->vss_hpf_enum.texts = wm8994->vss_hpf_texts;
ret = snd_soc_add_codec_controls(wm8994->hubs.codec,
@@ -1034,7 +1034,7 @@ void wm8958_dsp2_init(struct snd_soc_codec *codec)
for (i = 0; i < pdata->num_enh_eq_cfgs; i++)
wm8994->enh_eq_texts[i] = pdata->enh_eq_cfgs[i].name;
- wm8994->enh_eq_enum.max = pdata->num_enh_eq_cfgs;
+ wm8994->enh_eq_enum.items = pdata->num_enh_eq_cfgs;
wm8994->enh_eq_enum.texts = wm8994->enh_eq_texts;
ret = snd_soc_add_codec_controls(wm8994->hubs.codec,
diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c
index f156010..d04e9ca 100644
--- a/sound/soc/codecs/wm8960.c
+++ b/sound/soc/codecs/wm8960.c
@@ -976,12 +976,6 @@ static int wm8960_probe(struct snd_soc_codec *codec)
wm8960->set_bias_level = wm8960_set_bias_level_capless;
}
- 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;
- }
-
ret = wm8960_reset(codec);
if (ret < 0) {
dev_err(codec->dev, "Failed to issue reset\n");
diff --git a/sound/soc/codecs/wm8961.c b/sound/soc/codecs/wm8961.c
index 900328e..9c88f04 100644
--- a/sound/soc/codecs/wm8961.c
+++ b/sound/soc/codecs/wm8961.c
@@ -317,15 +317,15 @@ static const char *adc_hpf_text[] = {
"Hi-fi", "Voice 1", "Voice 2", "Voice 3",
};
-static const struct soc_enum adc_hpf =
- SOC_ENUM_SINGLE(WM8961_ADC_DAC_CONTROL_2, 7, 4, adc_hpf_text);
+static SOC_ENUM_SINGLE_DECL(adc_hpf,
+ WM8961_ADC_DAC_CONTROL_2, 7, adc_hpf_text);
static const char *dac_deemph_text[] = {
"None", "32kHz", "44.1kHz", "48kHz",
};
-static const struct soc_enum dac_deemph =
- SOC_ENUM_SINGLE(WM8961_ADC_DAC_CONTROL_1, 1, 4, dac_deemph_text);
+static SOC_ENUM_SINGLE_DECL(dac_deemph,
+ WM8961_ADC_DAC_CONTROL_1, 1, dac_deemph_text);
static const DECLARE_TLV_DB_SCALE(out_tlv, -12100, 100, 1);
static const DECLARE_TLV_DB_SCALE(hp_sec_tlv, -700, 100, 0);
@@ -385,11 +385,11 @@ static const char *sidetone_text[] = {
"None", "Left", "Right"
};
-static const struct soc_enum dacl_sidetone =
- SOC_ENUM_SINGLE(WM8961_DSP_SIDETONE_0, 2, 3, sidetone_text);
+static SOC_ENUM_SINGLE_DECL(dacl_sidetone,
+ WM8961_DSP_SIDETONE_0, 2, sidetone_text);
-static const struct soc_enum dacr_sidetone =
- SOC_ENUM_SINGLE(WM8961_DSP_SIDETONE_1, 2, 3, sidetone_text);
+static SOC_ENUM_SINGLE_DECL(dacr_sidetone,
+ WM8961_DSP_SIDETONE_1, 2, sidetone_text);
static const struct snd_kcontrol_new dacl_mux =
SOC_DAPM_ENUM("DACL Sidetone", dacl_sidetone);
@@ -836,15 +836,8 @@ static struct snd_soc_dai_driver wm8961_dai = {
static int wm8961_probe(struct snd_soc_codec *codec)
{
struct snd_soc_dapm_context *dapm = &codec->dapm;
- int ret = 0;
u16 reg;
- 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;
- }
-
/* Enable class W */
reg = snd_soc_read(codec, WM8961_CHARGE_PUMP_B);
reg |= WM8961_CP_DYN_PWR_MASK;
diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index 97db3b4..5522d25 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -1479,7 +1479,9 @@ static const DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0);
static int wm8962_dsp2_write_config(struct snd_soc_codec *codec)
{
- return regcache_sync_region(codec->control_data,
+ struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec);
+
+ return regcache_sync_region(wm8962->regmap,
WM8962_HDBASS_AI_1, WM8962_MAX_REGISTER);
}
@@ -1658,16 +1660,16 @@ static const char *cap_hpf_mode_text[] = {
"Hi-fi", "Application"
};
-static const struct soc_enum cap_hpf_mode =
- SOC_ENUM_SINGLE(WM8962_ADC_DAC_CONTROL_2, 10, 2, cap_hpf_mode_text);
+static SOC_ENUM_SINGLE_DECL(cap_hpf_mode,
+ WM8962_ADC_DAC_CONTROL_2, 10, cap_hpf_mode_text);
static const char *cap_lhpf_mode_text[] = {
"LPF", "HPF"
};
-static const struct soc_enum cap_lhpf_mode =
- SOC_ENUM_SINGLE(WM8962_LHPF1, 1, 2, cap_lhpf_mode_text);
+static SOC_ENUM_SINGLE_DECL(cap_lhpf_mode,
+ WM8962_LHPF1, 1, cap_lhpf_mode_text);
static const struct snd_kcontrol_new wm8962_snd_controls[] = {
SOC_DOUBLE("Input Mixer Switch", WM8962_INPUT_MIXER_CONTROL_1, 3, 2, 1, 1),
@@ -2014,40 +2016,40 @@ static int dsp2_event(struct snd_soc_dapm_widget *w,
static const char *st_text[] = { "None", "Left", "Right" };
-static const struct soc_enum str_enum =
- SOC_ENUM_SINGLE(WM8962_DAC_DSP_MIXING_1, 2, 3, st_text);
+static SOC_ENUM_SINGLE_DECL(str_enum,
+ WM8962_DAC_DSP_MIXING_1, 2, st_text);
static const struct snd_kcontrol_new str_mux =
SOC_DAPM_ENUM("Right Sidetone", str_enum);
-static const struct soc_enum stl_enum =
- SOC_ENUM_SINGLE(WM8962_DAC_DSP_MIXING_2, 2, 3, st_text);
+static SOC_ENUM_SINGLE_DECL(stl_enum,
+ WM8962_DAC_DSP_MIXING_2, 2, st_text);
static const struct snd_kcontrol_new stl_mux =
SOC_DAPM_ENUM("Left Sidetone", stl_enum);
static const char *outmux_text[] = { "DAC", "Mixer" };
-static const struct soc_enum spkoutr_enum =
- SOC_ENUM_SINGLE(WM8962_SPEAKER_MIXER_2, 7, 2, outmux_text);
+static SOC_ENUM_SINGLE_DECL(spkoutr_enum,
+ WM8962_SPEAKER_MIXER_2, 7, outmux_text);
static const struct snd_kcontrol_new spkoutr_mux =
SOC_DAPM_ENUM("SPKOUTR Mux", spkoutr_enum);
-static const struct soc_enum spkoutl_enum =
- SOC_ENUM_SINGLE(WM8962_SPEAKER_MIXER_1, 7, 2, outmux_text);
+static SOC_ENUM_SINGLE_DECL(spkoutl_enum,
+ WM8962_SPEAKER_MIXER_1, 7, outmux_text);
static const struct snd_kcontrol_new spkoutl_mux =
SOC_DAPM_ENUM("SPKOUTL Mux", spkoutl_enum);
-static const struct soc_enum hpoutr_enum =
- SOC_ENUM_SINGLE(WM8962_HEADPHONE_MIXER_2, 7, 2, outmux_text);
+static SOC_ENUM_SINGLE_DECL(hpoutr_enum,
+ WM8962_HEADPHONE_MIXER_2, 7, outmux_text);
static const struct snd_kcontrol_new hpoutr_mux =
SOC_DAPM_ENUM("HPOUTR Mux", hpoutr_enum);
-static const struct soc_enum hpoutl_enum =
- SOC_ENUM_SINGLE(WM8962_HEADPHONE_MIXER_1, 7, 2, outmux_text);
+static SOC_ENUM_SINGLE_DECL(hpoutl_enum,
+ WM8962_HEADPHONE_MIXER_1, 7, outmux_text);
static const struct snd_kcontrol_new hpoutl_mux =
SOC_DAPM_ENUM("HPOUTL Mux", hpoutl_enum);
@@ -2884,9 +2886,13 @@ static int wm8962_set_fll(struct snd_soc_codec *codec, int fll_id, int source,
snd_soc_write(codec, WM8962_FLL_CONTROL_7, fll_div.lambda);
snd_soc_write(codec, WM8962_FLL_CONTROL_8, fll_div.n);
- try_wait_for_completion(&wm8962->fll_lock);
+ reinit_completion(&wm8962->fll_lock);
- pm_runtime_get_sync(codec->dev);
+ ret = pm_runtime_get_sync(codec->dev);
+ if (ret < 0) {
+ dev_err(codec->dev, "Failed to resume device: %d\n", ret);
+ return ret;
+ }
snd_soc_update_bits(codec, WM8962_FLL_CONTROL_1,
WM8962_FLL_FRAC | WM8962_FLL_REFCLK_SRC_MASK |
@@ -2894,8 +2900,6 @@ static int wm8962_set_fll(struct snd_soc_codec *codec, int fll_id, int source,
dev_dbg(codec->dev, "FLL configured for %dHz->%dHz\n", Fref, Fout);
- ret = 0;
-
/* This should be a massive overestimate but go even
* higher if we'll error out
*/
@@ -2909,14 +2913,17 @@ static int wm8962_set_fll(struct snd_soc_codec *codec, int fll_id, int source,
if (timeout == 0 && wm8962->irq) {
dev_err(codec->dev, "FLL lock timed out");
- ret = -ETIMEDOUT;
+ snd_soc_update_bits(codec, WM8962_FLL_CONTROL_1,
+ WM8962_FLL_ENA, 0);
+ pm_runtime_put(codec->dev);
+ return -ETIMEDOUT;
}
wm8962->fll_fref = Fref;
wm8962->fll_fout = Fout;
wm8962->fll_src = source;
- return ret;
+ return 0;
}
static int wm8962_mute(struct snd_soc_dai *dai, int mute)
@@ -3003,9 +3010,16 @@ static irqreturn_t wm8962_irq(int irq, void *data)
unsigned int active;
int reg, ret;
+ ret = pm_runtime_get_sync(dev);
+ if (ret < 0) {
+ dev_err(dev, "Failed to resume: %d\n", ret);
+ return IRQ_NONE;
+ }
+
ret = regmap_read(wm8962->regmap, WM8962_INTERRUPT_STATUS_2_MASK,
&mask);
if (ret != 0) {
+ pm_runtime_put(dev);
dev_err(dev, "Failed to read interrupt mask: %d\n",
ret);
return IRQ_NONE;
@@ -3013,14 +3027,17 @@ static irqreturn_t wm8962_irq(int irq, void *data)
ret = regmap_read(wm8962->regmap, WM8962_INTERRUPT_STATUS_2, &active);
if (ret != 0) {
+ pm_runtime_put(dev);
dev_err(dev, "Failed to read interrupt: %d\n", ret);
return IRQ_NONE;
}
active &= ~mask;
- if (!active)
+ if (!active) {
+ pm_runtime_put(dev);
return IRQ_NONE;
+ }
/* Acknowledge the interrupts */
ret = regmap_write(wm8962->regmap, WM8962_INTERRUPT_STATUS_2, active);
@@ -3070,6 +3087,8 @@ static irqreturn_t wm8962_irq(int irq, void *data)
msecs_to_jiffies(250));
}
+ pm_runtime_put(dev);
+
return IRQ_HANDLED;
}
@@ -3089,6 +3108,7 @@ static irqreturn_t wm8962_irq(int irq, void *data)
int wm8962_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack)
{
struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec);
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
int irq_mask, enable;
wm8962->jack = jack;
@@ -3109,14 +3129,18 @@ int wm8962_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack)
snd_soc_jack_report(wm8962->jack, 0,
SND_JACK_MICROPHONE | SND_JACK_BTN_0);
+ snd_soc_dapm_mutex_lock(dapm);
+
if (jack) {
- snd_soc_dapm_force_enable_pin(&codec->dapm, "SYSCLK");
- snd_soc_dapm_force_enable_pin(&codec->dapm, "MICBIAS");
+ snd_soc_dapm_force_enable_pin_unlocked(dapm, "SYSCLK");
+ snd_soc_dapm_force_enable_pin_unlocked(dapm, "MICBIAS");
} else {
- snd_soc_dapm_disable_pin(&codec->dapm, "SYSCLK");
- snd_soc_dapm_disable_pin(&codec->dapm, "MICBIAS");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "SYSCLK");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "MICBIAS");
}
+ snd_soc_dapm_mutex_unlock(dapm);
+
return 0;
}
EXPORT_SYMBOL_GPL(wm8962_mic_detect);
@@ -3400,13 +3424,6 @@ static int wm8962_probe(struct snd_soc_codec *codec)
bool dmicclk, dmicdat;
wm8962->codec = codec;
- codec->control_data = wm8962->regmap;
-
- ret = snd_soc_codec_set_cache_io(codec, 16, 16, SND_SOC_REGMAP);
- if (ret != 0) {
- dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
- return ret;
- }
wm8962->disable_nb[0].notifier_call = wm8962_regulator_event_0;
wm8962->disable_nb[1].notifier_call = wm8962_regulator_event_1;
diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c
index 67aba78..09b7b42 100644
--- a/sound/soc/codecs/wm8971.c
+++ b/sound/soc/codecs/wm8971.c
@@ -648,12 +648,6 @@ static int wm8971_probe(struct snd_soc_codec *codec)
int ret = 0;
u16 reg;
- ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP);
- if (ret < 0) {
- printk(KERN_ERR "wm8971: failed to set cache I/O: %d\n", ret);
- return ret;
- }
-
INIT_DELAYED_WORK(&codec->dapm.delayed_work, wm8971_work);
wm8971_workq = create_workqueue("wm8971");
if (wm8971_workq == NULL)
diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c
index 15f45c7..0627c56 100644
--- a/sound/soc/codecs/wm8974.c
+++ b/sound/soc/codecs/wm8974.c
@@ -84,8 +84,8 @@ static const struct soc_enum wm8974_enum[] = {
static const char *wm8974_auxmode_text[] = { "Buffer", "Mixer" };
-static const struct soc_enum wm8974_auxmode =
- SOC_ENUM_SINGLE(WM8974_INPUT, 3, 2, wm8974_auxmode_text);
+static SOC_ENUM_SINGLE_DECL(wm8974_auxmode,
+ WM8974_INPUT, 3, wm8974_auxmode_text);
static const DECLARE_TLV_DB_SCALE(digital_tlv, -12750, 50, 1);
static const DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0);
@@ -593,12 +593,6 @@ static int wm8974_probe(struct snd_soc_codec *codec)
{
int ret = 0;
- 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;
- }
-
ret = wm8974_reset(codec);
if (ret < 0) {
dev_err(codec->dev, "Failed to issue reset\n");
diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c
index d8fc531..28ef46c 100644
--- a/sound/soc/codecs/wm8978.c
+++ b/sound/soc/codecs/wm8978.c
@@ -117,21 +117,21 @@ static const char *wm8978_eq5[] = {"5.3kHz", "6.9kHz", "9kHz", "11.7kHz"};
static const char *wm8978_alc3[] = {"ALC", "Limiter"};
static const char *wm8978_alc1[] = {"Off", "Right", "Left", "Both"};
-static const SOC_ENUM_SINGLE_DECL(adc_compand, WM8978_COMPANDING_CONTROL, 1,
- wm8978_companding);
-static const SOC_ENUM_SINGLE_DECL(dac_compand, WM8978_COMPANDING_CONTROL, 3,
- wm8978_companding);
-static const SOC_ENUM_SINGLE_DECL(eqmode, WM8978_EQ1, 8, wm8978_eqmode);
-static const SOC_ENUM_SINGLE_DECL(eq1, WM8978_EQ1, 5, wm8978_eq1);
-static const SOC_ENUM_SINGLE_DECL(eq2bw, WM8978_EQ2, 8, wm8978_bw);
-static const SOC_ENUM_SINGLE_DECL(eq2, WM8978_EQ2, 5, wm8978_eq2);
-static const SOC_ENUM_SINGLE_DECL(eq3bw, WM8978_EQ3, 8, wm8978_bw);
-static const SOC_ENUM_SINGLE_DECL(eq3, WM8978_EQ3, 5, wm8978_eq3);
-static const SOC_ENUM_SINGLE_DECL(eq4bw, WM8978_EQ4, 8, wm8978_bw);
-static const SOC_ENUM_SINGLE_DECL(eq4, WM8978_EQ4, 5, wm8978_eq4);
-static const SOC_ENUM_SINGLE_DECL(eq5, WM8978_EQ5, 5, wm8978_eq5);
-static const SOC_ENUM_SINGLE_DECL(alc3, WM8978_ALC_CONTROL_3, 8, wm8978_alc3);
-static const SOC_ENUM_SINGLE_DECL(alc1, WM8978_ALC_CONTROL_1, 7, wm8978_alc1);
+static SOC_ENUM_SINGLE_DECL(adc_compand, WM8978_COMPANDING_CONTROL, 1,
+ wm8978_companding);
+static SOC_ENUM_SINGLE_DECL(dac_compand, WM8978_COMPANDING_CONTROL, 3,
+ wm8978_companding);
+static SOC_ENUM_SINGLE_DECL(eqmode, WM8978_EQ1, 8, wm8978_eqmode);
+static SOC_ENUM_SINGLE_DECL(eq1, WM8978_EQ1, 5, wm8978_eq1);
+static SOC_ENUM_SINGLE_DECL(eq2bw, WM8978_EQ2, 8, wm8978_bw);
+static SOC_ENUM_SINGLE_DECL(eq2, WM8978_EQ2, 5, wm8978_eq2);
+static SOC_ENUM_SINGLE_DECL(eq3bw, WM8978_EQ3, 8, wm8978_bw);
+static SOC_ENUM_SINGLE_DECL(eq3, WM8978_EQ3, 5, wm8978_eq3);
+static SOC_ENUM_SINGLE_DECL(eq4bw, WM8978_EQ4, 8, wm8978_bw);
+static SOC_ENUM_SINGLE_DECL(eq4, WM8978_EQ4, 5, wm8978_eq4);
+static SOC_ENUM_SINGLE_DECL(eq5, WM8978_EQ5, 5, wm8978_eq5);
+static SOC_ENUM_SINGLE_DECL(alc3, WM8978_ALC_CONTROL_3, 8, wm8978_alc3);
+static SOC_ENUM_SINGLE_DECL(alc1, WM8978_ALC_CONTROL_1, 7, wm8978_alc1);
static const DECLARE_TLV_DB_SCALE(digital_tlv, -12750, 50, 1);
static const DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0);
@@ -975,19 +975,13 @@ static const int update_reg[] = {
static int wm8978_probe(struct snd_soc_codec *codec)
{
struct wm8978_priv *wm8978 = snd_soc_codec_get_drvdata(codec);
- int ret = 0, i;
+ int i;
/*
* Set default system clock to PLL, it is more precise, this is also the
* default hardware setting
*/
wm8978->sysclk = WM8978_PLL;
- codec->control_data = wm8978->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;
- }
/*
* Set the update bit in all registers, that have one. This way all
diff --git a/sound/soc/codecs/wm8983.c b/sound/soc/codecs/wm8983.c
index aa41ba0..2b9bfa5 100644
--- a/sound/soc/codecs/wm8983.c
+++ b/sound/soc/codecs/wm8983.c
@@ -205,49 +205,44 @@ static const DECLARE_TLV_DB_SCALE(bypass_tlv, -1500, 300, 0);
static const DECLARE_TLV_DB_SCALE(pga_boost_tlv, 0, 2000, 0);
static const char *alc_sel_text[] = { "Off", "Right", "Left", "Stereo" };
-static const SOC_ENUM_SINGLE_DECL(alc_sel, WM8983_ALC_CONTROL_1, 7,
- alc_sel_text);
+static SOC_ENUM_SINGLE_DECL(alc_sel, WM8983_ALC_CONTROL_1, 7, alc_sel_text);
static const char *alc_mode_text[] = { "ALC", "Limiter" };
-static const SOC_ENUM_SINGLE_DECL(alc_mode, WM8983_ALC_CONTROL_3, 8,
- alc_mode_text);
+static SOC_ENUM_SINGLE_DECL(alc_mode, WM8983_ALC_CONTROL_3, 8, alc_mode_text);
static const char *filter_mode_text[] = { "Audio", "Application" };
-static const SOC_ENUM_SINGLE_DECL(filter_mode, WM8983_ADC_CONTROL, 7,
- filter_mode_text);
+static SOC_ENUM_SINGLE_DECL(filter_mode, WM8983_ADC_CONTROL, 7,
+ filter_mode_text);
static const char *eq_bw_text[] = { "Narrow", "Wide" };
static const char *eqmode_text[] = { "Capture", "Playback" };
-static const SOC_ENUM_SINGLE_EXT_DECL(eqmode, eqmode_text);
+static SOC_ENUM_SINGLE_EXT_DECL(eqmode, eqmode_text);
static const char *eq1_cutoff_text[] = {
"80Hz", "105Hz", "135Hz", "175Hz"
};
-static const SOC_ENUM_SINGLE_DECL(eq1_cutoff, WM8983_EQ1_LOW_SHELF, 5,
- eq1_cutoff_text);
+static SOC_ENUM_SINGLE_DECL(eq1_cutoff, WM8983_EQ1_LOW_SHELF, 5,
+ eq1_cutoff_text);
static const char *eq2_cutoff_text[] = {
"230Hz", "300Hz", "385Hz", "500Hz"
};
-static const SOC_ENUM_SINGLE_DECL(eq2_bw, WM8983_EQ2_PEAK_1, 8, eq_bw_text);
-static const SOC_ENUM_SINGLE_DECL(eq2_cutoff, WM8983_EQ2_PEAK_1, 5,
- eq2_cutoff_text);
+static SOC_ENUM_SINGLE_DECL(eq2_bw, WM8983_EQ2_PEAK_1, 8, eq_bw_text);
+static SOC_ENUM_SINGLE_DECL(eq2_cutoff, WM8983_EQ2_PEAK_1, 5, eq2_cutoff_text);
static const char *eq3_cutoff_text[] = {
"650Hz", "850Hz", "1.1kHz", "1.4kHz"
};
-static const SOC_ENUM_SINGLE_DECL(eq3_bw, WM8983_EQ3_PEAK_2, 8, eq_bw_text);
-static const SOC_ENUM_SINGLE_DECL(eq3_cutoff, WM8983_EQ3_PEAK_2, 5,
- eq3_cutoff_text);
+static SOC_ENUM_SINGLE_DECL(eq3_bw, WM8983_EQ3_PEAK_2, 8, eq_bw_text);
+static SOC_ENUM_SINGLE_DECL(eq3_cutoff, WM8983_EQ3_PEAK_2, 5, eq3_cutoff_text);
static const char *eq4_cutoff_text[] = {
"1.8kHz", "2.4kHz", "3.2kHz", "4.1kHz"
};
-static const SOC_ENUM_SINGLE_DECL(eq4_bw, WM8983_EQ4_PEAK_3, 8, eq_bw_text);
-static const SOC_ENUM_SINGLE_DECL(eq4_cutoff, WM8983_EQ4_PEAK_3, 5,
- eq4_cutoff_text);
+static SOC_ENUM_SINGLE_DECL(eq4_bw, WM8983_EQ4_PEAK_3, 8, eq_bw_text);
+static SOC_ENUM_SINGLE_DECL(eq4_cutoff, WM8983_EQ4_PEAK_3, 5, eq4_cutoff_text);
static const char *eq5_cutoff_text[] = {
"5.3kHz", "6.9kHz", "9kHz", "11.7kHz"
};
-static const SOC_ENUM_SINGLE_DECL(eq5_cutoff, WM8983_EQ5_HIGH_SHELF, 5,
- eq5_cutoff_text);
+static SOC_ENUM_SINGLE_DECL(eq5_cutoff, WM8983_EQ5_HIGH_SHELF, 5,
+ eq5_cutoff_text);
static const char *depth_3d_text[] = {
"Off",
@@ -267,8 +262,8 @@ static const char *depth_3d_text[] = {
"93.3%",
"100%"
};
-static const SOC_ENUM_SINGLE_DECL(depth_3d, WM8983_3D_CONTROL, 0,
- depth_3d_text);
+static SOC_ENUM_SINGLE_DECL(depth_3d, WM8983_3D_CONTROL, 0,
+ depth_3d_text);
static const struct snd_kcontrol_new wm8983_snd_controls[] = {
SOC_SINGLE("Digital Loopback Switch", WM8983_COMPANDING_CONTROL,
@@ -1000,12 +995,6 @@ static int wm8983_probe(struct snd_soc_codec *codec)
int ret;
int i;
- 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;
- }
-
ret = snd_soc_write(codec, WM8983_SOFTWARE_RESET, 0);
if (ret < 0) {
dev_err(codec->dev, "Failed to issue reset: %d\n", ret);
@@ -1129,7 +1118,7 @@ static struct spi_driver wm8983_spi_driver = {
};
#endif
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+#if IS_ENABLED(CONFIG_I2C)
static int wm8983_i2c_probe(struct i2c_client *i2c,
const struct i2c_device_id *id)
{
@@ -1182,7 +1171,7 @@ static int __init wm8983_modinit(void)
{
int ret = 0;
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+#if IS_ENABLED(CONFIG_I2C)
ret = i2c_add_driver(&wm8983_i2c_driver);
if (ret) {
printk(KERN_ERR "Failed to register wm8983 I2C driver: %d\n",
@@ -1202,7 +1191,7 @@ module_init(wm8983_modinit);
static void __exit wm8983_exit(void)
{
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+#if IS_ENABLED(CONFIG_I2C)
i2c_del_driver(&wm8983_i2c_driver);
#endif
#if defined(CONFIG_SPI_MASTER)
diff --git a/sound/soc/codecs/wm8985.c b/sound/soc/codecs/wm8985.c
index 271b517..5473dc9 100644
--- a/sound/soc/codecs/wm8985.c
+++ b/sound/soc/codecs/wm8985.c
@@ -226,52 +226,48 @@ static const DECLARE_TLV_DB_SCALE(bypass_tlv, -1500, 300, 0);
static const DECLARE_TLV_DB_SCALE(pga_boost_tlv, 0, 2000, 0);
static const char *alc_sel_text[] = { "Off", "Right", "Left", "Stereo" };
-static const SOC_ENUM_SINGLE_DECL(alc_sel, WM8985_ALC_CONTROL_1, 7,
- alc_sel_text);
+static SOC_ENUM_SINGLE_DECL(alc_sel, WM8985_ALC_CONTROL_1, 7, alc_sel_text);
static const char *alc_mode_text[] = { "ALC", "Limiter" };
-static const SOC_ENUM_SINGLE_DECL(alc_mode, WM8985_ALC_CONTROL_3, 8,
- alc_mode_text);
+static SOC_ENUM_SINGLE_DECL(alc_mode, WM8985_ALC_CONTROL_3, 8, alc_mode_text);
static const char *filter_mode_text[] = { "Audio", "Application" };
-static const SOC_ENUM_SINGLE_DECL(filter_mode, WM8985_ADC_CONTROL, 7,
- filter_mode_text);
+static SOC_ENUM_SINGLE_DECL(filter_mode, WM8985_ADC_CONTROL, 7,
+ filter_mode_text);
static const char *eq_bw_text[] = { "Narrow", "Wide" };
static const char *eqmode_text[] = { "Capture", "Playback" };
-static const SOC_ENUM_SINGLE_EXT_DECL(eqmode, eqmode_text);
+static SOC_ENUM_SINGLE_EXT_DECL(eqmode, eqmode_text);
static const char *eq1_cutoff_text[] = {
"80Hz", "105Hz", "135Hz", "175Hz"
};
-static const SOC_ENUM_SINGLE_DECL(eq1_cutoff, WM8985_EQ1_LOW_SHELF, 5,
- eq1_cutoff_text);
+static SOC_ENUM_SINGLE_DECL(eq1_cutoff, WM8985_EQ1_LOW_SHELF, 5,
+ eq1_cutoff_text);
static const char *eq2_cutoff_text[] = {
"230Hz", "300Hz", "385Hz", "500Hz"
};
-static const SOC_ENUM_SINGLE_DECL(eq2_bw, WM8985_EQ2_PEAK_1, 8, eq_bw_text);
-static const SOC_ENUM_SINGLE_DECL(eq2_cutoff, WM8985_EQ2_PEAK_1, 5,
- eq2_cutoff_text);
+static SOC_ENUM_SINGLE_DECL(eq2_bw, WM8985_EQ2_PEAK_1, 8, eq_bw_text);
+static SOC_ENUM_SINGLE_DECL(eq2_cutoff, WM8985_EQ2_PEAK_1, 5, eq2_cutoff_text);
static const char *eq3_cutoff_text[] = {
"650Hz", "850Hz", "1.1kHz", "1.4kHz"
};
-static const SOC_ENUM_SINGLE_DECL(eq3_bw, WM8985_EQ3_PEAK_2, 8, eq_bw_text);
-static const SOC_ENUM_SINGLE_DECL(eq3_cutoff, WM8985_EQ3_PEAK_2, 5,
- eq3_cutoff_text);
+static SOC_ENUM_SINGLE_DECL(eq3_bw, WM8985_EQ3_PEAK_2, 8, eq_bw_text);
+static SOC_ENUM_SINGLE_DECL(eq3_cutoff, WM8985_EQ3_PEAK_2, 5,
+ eq3_cutoff_text);
static const char *eq4_cutoff_text[] = {
"1.8kHz", "2.4kHz", "3.2kHz", "4.1kHz"
};
-static const SOC_ENUM_SINGLE_DECL(eq4_bw, WM8985_EQ4_PEAK_3, 8, eq_bw_text);
-static const SOC_ENUM_SINGLE_DECL(eq4_cutoff, WM8985_EQ4_PEAK_3, 5,
- eq4_cutoff_text);
+static SOC_ENUM_SINGLE_DECL(eq4_bw, WM8985_EQ4_PEAK_3, 8, eq_bw_text);
+static SOC_ENUM_SINGLE_DECL(eq4_cutoff, WM8985_EQ4_PEAK_3, 5, eq4_cutoff_text);
static const char *eq5_cutoff_text[] = {
"5.3kHz", "6.9kHz", "9kHz", "11.7kHz"
};
-static const SOC_ENUM_SINGLE_DECL(eq5_cutoff, WM8985_EQ5_HIGH_SHELF, 5,
+static SOC_ENUM_SINGLE_DECL(eq5_cutoff, WM8985_EQ5_HIGH_SHELF, 5,
eq5_cutoff_text);
static const char *speaker_mode_text[] = { "Class A/B", "Class D" };
-static const SOC_ENUM_SINGLE_DECL(speaker_mode, 0x17, 8, speaker_mode_text);
+static SOC_ENUM_SINGLE_DECL(speaker_mode, 0x17, 8, speaker_mode_text);
static const char *depth_3d_text[] = {
"Off",
@@ -291,8 +287,7 @@ static const char *depth_3d_text[] = {
"93.3%",
"100%"
};
-static const SOC_ENUM_SINGLE_DECL(depth_3d, WM8985_3D_CONTROL, 0,
- depth_3d_text);
+static SOC_ENUM_SINGLE_DECL(depth_3d, WM8985_3D_CONTROL, 0, depth_3d_text);
static const struct snd_kcontrol_new wm8985_snd_controls[] = {
SOC_SINGLE("Digital Loopback Switch", WM8985_COMPANDING_CONTROL,
@@ -1000,13 +995,6 @@ static int wm8985_probe(struct snd_soc_codec *codec)
int ret;
wm8985 = snd_soc_codec_get_drvdata(codec);
- codec->control_data = wm8985->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;
- }
for (i = 0; i < ARRAY_SIZE(wm8985->supplies); i++)
wm8985->supplies[i].supply = wm8985_supply_names[i];
diff --git a/sound/soc/codecs/wm8988.c b/sound/soc/codecs/wm8988.c
index a55e1c2..3a1ae4f5 100644
--- a/sound/soc/codecs/wm8988.c
+++ b/sound/soc/codecs/wm8988.c
@@ -116,7 +116,7 @@ static bool wm8988_writeable(struct device *dev, unsigned int reg)
struct wm8988_priv {
struct regmap *regmap;
unsigned int sysclk;
- struct snd_pcm_hw_constraint_list *sysclk_constraints;
+ const struct snd_pcm_hw_constraint_list *sysclk_constraints;
};
#define wm8988_reset(c) snd_soc_write(c, WM8988_RESET, 0)
@@ -126,46 +126,46 @@ struct wm8988_priv {
*/
static const char *bass_boost_txt[] = {"Linear Control", "Adaptive Boost"};
-static const struct soc_enum bass_boost =
- SOC_ENUM_SINGLE(WM8988_BASS, 7, 2, bass_boost_txt);
+static SOC_ENUM_SINGLE_DECL(bass_boost,
+ WM8988_BASS, 7, bass_boost_txt);
static const char *bass_filter_txt[] = { "130Hz @ 48kHz", "200Hz @ 48kHz" };
-static const struct soc_enum bass_filter =
- SOC_ENUM_SINGLE(WM8988_BASS, 6, 2, bass_filter_txt);
+static SOC_ENUM_SINGLE_DECL(bass_filter,
+ WM8988_BASS, 6, bass_filter_txt);
static const char *treble_txt[] = {"8kHz", "4kHz"};
-static const struct soc_enum treble =
- SOC_ENUM_SINGLE(WM8988_TREBLE, 6, 2, treble_txt);
+static SOC_ENUM_SINGLE_DECL(treble,
+ WM8988_TREBLE, 6, treble_txt);
static const char *stereo_3d_lc_txt[] = {"200Hz", "500Hz"};
-static const struct soc_enum stereo_3d_lc =
- SOC_ENUM_SINGLE(WM8988_3D, 5, 2, stereo_3d_lc_txt);
+static SOC_ENUM_SINGLE_DECL(stereo_3d_lc,
+ WM8988_3D, 5, stereo_3d_lc_txt);
static const char *stereo_3d_uc_txt[] = {"2.2kHz", "1.5kHz"};
-static const struct soc_enum stereo_3d_uc =
- SOC_ENUM_SINGLE(WM8988_3D, 6, 2, stereo_3d_uc_txt);
+static SOC_ENUM_SINGLE_DECL(stereo_3d_uc,
+ WM8988_3D, 6, stereo_3d_uc_txt);
static const char *stereo_3d_func_txt[] = {"Capture", "Playback"};
-static const struct soc_enum stereo_3d_func =
- SOC_ENUM_SINGLE(WM8988_3D, 7, 2, stereo_3d_func_txt);
+static SOC_ENUM_SINGLE_DECL(stereo_3d_func,
+ WM8988_3D, 7, stereo_3d_func_txt);
static const char *alc_func_txt[] = {"Off", "Right", "Left", "Stereo"};
-static const struct soc_enum alc_func =
- SOC_ENUM_SINGLE(WM8988_ALC1, 7, 4, alc_func_txt);
+static SOC_ENUM_SINGLE_DECL(alc_func,
+ WM8988_ALC1, 7, alc_func_txt);
static const char *ng_type_txt[] = {"Constant PGA Gain",
"Mute ADC Output"};
-static const struct soc_enum ng_type =
- SOC_ENUM_SINGLE(WM8988_NGATE, 1, 2, ng_type_txt);
+static SOC_ENUM_SINGLE_DECL(ng_type,
+ WM8988_NGATE, 1, ng_type_txt);
static const char *deemph_txt[] = {"None", "32Khz", "44.1Khz", "48Khz"};
-static const struct soc_enum deemph =
- SOC_ENUM_SINGLE(WM8988_ADCDAC, 1, 4, deemph_txt);
+static SOC_ENUM_SINGLE_DECL(deemph,
+ WM8988_ADCDAC, 1, deemph_txt);
static const char *adcpol_txt[] = {"Normal", "L Invert", "R Invert",
"L + R Invert"};
-static const struct soc_enum adcpol =
- SOC_ENUM_SINGLE(WM8988_ADCDAC, 5, 4, adcpol_txt);
+static SOC_ENUM_SINGLE_DECL(adcpol,
+ WM8988_ADCDAC, 5, adcpol_txt);
static const DECLARE_TLV_DB_SCALE(pga_tlv, -1725, 75, 0);
static const DECLARE_TLV_DB_SCALE(adc_tlv, -9750, 50, 1);
@@ -317,16 +317,16 @@ static const struct snd_kcontrol_new wm8988_right_pga_controls =
/* Differential Mux */
static const char *wm8988_diff_sel[] = {"Line 1", "Line 2"};
-static const struct soc_enum diffmux =
- SOC_ENUM_SINGLE(WM8988_ADCIN, 8, 2, wm8988_diff_sel);
+static SOC_ENUM_SINGLE_DECL(diffmux,
+ WM8988_ADCIN, 8, wm8988_diff_sel);
static const struct snd_kcontrol_new wm8988_diffmux_controls =
SOC_DAPM_ENUM("Route", diffmux);
/* Mono ADC Mux */
static const char *wm8988_mono_mux[] = {"Stereo", "Mono (Left)",
"Mono (Right)", "Digital Mono"};
-static const struct soc_enum monomux =
- SOC_ENUM_SINGLE(WM8988_ADCIN, 6, 4, wm8988_mono_mux);
+static SOC_ENUM_SINGLE_DECL(monomux,
+ WM8988_ADCIN, 6, wm8988_mono_mux);
static const struct snd_kcontrol_new wm8988_monomux_controls =
SOC_DAPM_ENUM("Route", monomux);
@@ -521,30 +521,30 @@ static inline int get_coeff(int mclk, int rate)
/* The set of rates we can generate from the above for each SYSCLK */
-static unsigned int rates_12288[] = {
+static const unsigned int rates_12288[] = {
8000, 12000, 16000, 24000, 24000, 32000, 48000, 96000,
};
-static struct snd_pcm_hw_constraint_list constraints_12288 = {
+static const struct snd_pcm_hw_constraint_list constraints_12288 = {
.count = ARRAY_SIZE(rates_12288),
.list = rates_12288,
};
-static unsigned int rates_112896[] = {
+static const unsigned int rates_112896[] = {
8000, 11025, 22050, 44100,
};
-static struct snd_pcm_hw_constraint_list constraints_112896 = {
+static const struct snd_pcm_hw_constraint_list constraints_112896 = {
.count = ARRAY_SIZE(rates_112896),
.list = rates_112896,
};
-static unsigned int rates_12[] = {
+static const unsigned int rates_12[] = {
8000, 11025, 12000, 16000, 22050, 2400, 32000, 41100, 48000,
48000, 88235, 96000,
};
-static struct snd_pcm_hw_constraint_list constraints_12 = {
+static const struct snd_pcm_hw_constraint_list constraints_12 = {
.count = ARRAY_SIZE(rates_12),
.list = rates_12,
};
@@ -810,16 +810,8 @@ static int wm8988_resume(struct snd_soc_codec *codec)
static int wm8988_probe(struct snd_soc_codec *codec)
{
- struct wm8988_priv *wm8988 = snd_soc_codec_get_drvdata(codec);
int ret = 0;
- codec->control_data = wm8988->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;
- }
-
ret = wm8988_reset(codec);
if (ret < 0) {
dev_err(codec->dev, "Failed to issue reset\n");
diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c
index 0ccd4d8..c413c19 100644
--- a/sound/soc/codecs/wm8990.c
+++ b/sound/soc/codecs/wm8990.c
@@ -157,26 +157,23 @@ static int wm899x_outpga_put_volsw_vu(struct snd_kcontrol *kcontrol,
static const char *wm8990_digital_sidetone[] =
{"None", "Left ADC", "Right ADC", "Reserved"};
-static const struct soc_enum wm8990_left_digital_sidetone_enum =
-SOC_ENUM_SINGLE(WM8990_DIGITAL_SIDE_TONE,
- WM8990_ADC_TO_DACL_SHIFT,
- WM8990_ADC_TO_DACL_MASK,
- wm8990_digital_sidetone);
-
-static const struct soc_enum wm8990_right_digital_sidetone_enum =
-SOC_ENUM_SINGLE(WM8990_DIGITAL_SIDE_TONE,
- WM8990_ADC_TO_DACR_SHIFT,
- WM8990_ADC_TO_DACR_MASK,
- wm8990_digital_sidetone);
+static SOC_ENUM_SINGLE_DECL(wm8990_left_digital_sidetone_enum,
+ WM8990_DIGITAL_SIDE_TONE,
+ WM8990_ADC_TO_DACL_SHIFT,
+ wm8990_digital_sidetone);
+
+static SOC_ENUM_SINGLE_DECL(wm8990_right_digital_sidetone_enum,
+ WM8990_DIGITAL_SIDE_TONE,
+ WM8990_ADC_TO_DACR_SHIFT,
+ wm8990_digital_sidetone);
static const char *wm8990_adcmode[] =
{"Hi-fi mode", "Voice mode 1", "Voice mode 2", "Voice mode 3"};
-static const struct soc_enum wm8990_right_adcmode_enum =
-SOC_ENUM_SINGLE(WM8990_ADC_CTRL,
- WM8990_ADC_HPF_CUT_SHIFT,
- WM8990_ADC_HPF_CUT_MASK,
- wm8990_adcmode);
+static SOC_ENUM_SINGLE_DECL(wm8990_right_adcmode_enum,
+ WM8990_ADC_CTRL,
+ WM8990_ADC_HPF_CUT_SHIFT,
+ wm8990_adcmode);
static const struct snd_kcontrol_new wm8990_snd_controls[] = {
/* INMIXL */
@@ -475,9 +472,9 @@ SOC_DAPM_SINGLE("RINPGA34 Switch", WM8990_INPUT_MIXER3, WM8990_L34MNB_BIT,
static const char *wm8990_ainlmux[] =
{"INMIXL Mix", "RXVOICE Mix", "DIFFINL Mix"};
-static const struct soc_enum wm8990_ainlmux_enum =
-SOC_ENUM_SINGLE(WM8990_INPUT_MIXER1, WM8990_AINLMODE_SHIFT,
- ARRAY_SIZE(wm8990_ainlmux), wm8990_ainlmux);
+static SOC_ENUM_SINGLE_DECL(wm8990_ainlmux_enum,
+ WM8990_INPUT_MIXER1, WM8990_AINLMODE_SHIFT,
+ wm8990_ainlmux);
static const struct snd_kcontrol_new wm8990_dapm_ainlmux_controls =
SOC_DAPM_ENUM("Route", wm8990_ainlmux_enum);
@@ -488,9 +485,9 @@ SOC_DAPM_ENUM("Route", wm8990_ainlmux_enum);
static const char *wm8990_ainrmux[] =
{"INMIXR Mix", "RXVOICE Mix", "DIFFINR Mix"};
-static const struct soc_enum wm8990_ainrmux_enum =
-SOC_ENUM_SINGLE(WM8990_INPUT_MIXER1, WM8990_AINRMODE_SHIFT,
- ARRAY_SIZE(wm8990_ainrmux), wm8990_ainrmux);
+static SOC_ENUM_SINGLE_DECL(wm8990_ainrmux_enum,
+ WM8990_INPUT_MIXER1, WM8990_AINRMODE_SHIFT,
+ wm8990_ainrmux);
static const struct snd_kcontrol_new wm8990_dapm_ainrmux_controls =
SOC_DAPM_ENUM("Route", wm8990_ainrmux_enum);
@@ -1292,14 +1289,6 @@ static int wm8990_resume(struct snd_soc_codec *codec)
*/
static int wm8990_probe(struct snd_soc_codec *codec)
{
- int ret;
-
- ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP);
- if (ret < 0) {
- printk(KERN_ERR "wm8990: failed to set cache I/O: %d\n", ret);
- return ret;
- }
-
wm8990_reset(codec);
/* charge output caps */
diff --git a/sound/soc/codecs/wm8991.c b/sound/soc/codecs/wm8991.c
index dba0306..844cc4a 100644
--- a/sound/soc/codecs/wm8991.c
+++ b/sound/soc/codecs/wm8991.c
@@ -171,26 +171,23 @@ static int wm899x_outpga_put_volsw_vu(struct snd_kcontrol *kcontrol,
static const char *wm8991_digital_sidetone[] =
{"None", "Left ADC", "Right ADC", "Reserved"};
-static const struct soc_enum wm8991_left_digital_sidetone_enum =
- SOC_ENUM_SINGLE(WM8991_DIGITAL_SIDE_TONE,
- WM8991_ADC_TO_DACL_SHIFT,
- WM8991_ADC_TO_DACL_MASK,
- wm8991_digital_sidetone);
-
-static const struct soc_enum wm8991_right_digital_sidetone_enum =
- SOC_ENUM_SINGLE(WM8991_DIGITAL_SIDE_TONE,
- WM8991_ADC_TO_DACR_SHIFT,
- WM8991_ADC_TO_DACR_MASK,
- wm8991_digital_sidetone);
+static SOC_ENUM_SINGLE_DECL(wm8991_left_digital_sidetone_enum,
+ WM8991_DIGITAL_SIDE_TONE,
+ WM8991_ADC_TO_DACL_SHIFT,
+ wm8991_digital_sidetone);
+
+static SOC_ENUM_SINGLE_DECL(wm8991_right_digital_sidetone_enum,
+ WM8991_DIGITAL_SIDE_TONE,
+ WM8991_ADC_TO_DACR_SHIFT,
+ wm8991_digital_sidetone);
static const char *wm8991_adcmode[] =
{"Hi-fi mode", "Voice mode 1", "Voice mode 2", "Voice mode 3"};
-static const struct soc_enum wm8991_right_adcmode_enum =
- SOC_ENUM_SINGLE(WM8991_ADC_CTRL,
- WM8991_ADC_HPF_CUT_SHIFT,
- WM8991_ADC_HPF_CUT_MASK,
- wm8991_adcmode);
+static SOC_ENUM_SINGLE_DECL(wm8991_right_adcmode_enum,
+ WM8991_ADC_CTRL,
+ WM8991_ADC_HPF_CUT_SHIFT,
+ wm8991_adcmode);
static const struct snd_kcontrol_new wm8991_snd_controls[] = {
/* INMIXL */
@@ -486,9 +483,9 @@ static const struct snd_kcontrol_new wm8991_dapm_inmixr_controls[] = {
static const char *wm8991_ainlmux[] =
{"INMIXL Mix", "RXVOICE Mix", "DIFFINL Mix"};
-static const struct soc_enum wm8991_ainlmux_enum =
- SOC_ENUM_SINGLE(WM8991_INPUT_MIXER1, WM8991_AINLMODE_SHIFT,
- ARRAY_SIZE(wm8991_ainlmux), wm8991_ainlmux);
+static SOC_ENUM_SINGLE_DECL(wm8991_ainlmux_enum,
+ WM8991_INPUT_MIXER1, WM8991_AINLMODE_SHIFT,
+ wm8991_ainlmux);
static const struct snd_kcontrol_new wm8991_dapm_ainlmux_controls =
SOC_DAPM_ENUM("Route", wm8991_ainlmux_enum);
@@ -499,9 +496,9 @@ static const struct snd_kcontrol_new wm8991_dapm_ainlmux_controls =
static const char *wm8991_ainrmux[] =
{"INMIXR Mix", "RXVOICE Mix", "DIFFINR Mix"};
-static const struct soc_enum wm8991_ainrmux_enum =
- SOC_ENUM_SINGLE(WM8991_INPUT_MIXER1, WM8991_AINRMODE_SHIFT,
- ARRAY_SIZE(wm8991_ainrmux), wm8991_ainrmux);
+static SOC_ENUM_SINGLE_DECL(wm8991_ainrmux_enum,
+ WM8991_INPUT_MIXER1, WM8991_AINRMODE_SHIFT,
+ wm8991_ainrmux);
static const struct snd_kcontrol_new wm8991_dapm_ainrmux_controls =
SOC_DAPM_ENUM("Route", wm8991_ainrmux_enum);
@@ -1251,17 +1248,6 @@ static int wm8991_remove(struct snd_soc_codec *codec)
static int wm8991_probe(struct snd_soc_codec *codec)
{
- struct wm8991_priv *wm8991;
- int ret;
-
- wm8991 = snd_soc_codec_get_drvdata(codec);
-
- 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;
- }
-
wm8991_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
return 0;
diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c
index 2ee23a3..f825dc0 100644
--- a/sound/soc/codecs/wm8993.c
+++ b/sound/soc/codecs/wm8993.c
@@ -646,8 +646,8 @@ static const char *dac_deemph_text[] = {
"48kHz",
};
-static const struct soc_enum dac_deemph =
- SOC_ENUM_SINGLE(WM8993_DAC_CTRL, 4, 4, dac_deemph_text);
+static SOC_ENUM_SINGLE_DECL(dac_deemph,
+ WM8993_DAC_CTRL, 4, dac_deemph_text);
static const char *adc_hpf_text[] = {
"Hi-Fi",
@@ -656,16 +656,16 @@ static const char *adc_hpf_text[] = {
"Voice 3",
};
-static const struct soc_enum adc_hpf =
- SOC_ENUM_SINGLE(WM8993_ADC_CTRL, 5, 4, adc_hpf_text);
+static SOC_ENUM_SINGLE_DECL(adc_hpf,
+ WM8993_ADC_CTRL, 5, adc_hpf_text);
static const char *drc_path_text[] = {
"ADC",
"DAC"
};
-static const struct soc_enum drc_path =
- SOC_ENUM_SINGLE(WM8993_DRC_CONTROL_1, 14, 2, drc_path_text);
+static SOC_ENUM_SINGLE_DECL(drc_path,
+ WM8993_DRC_CONTROL_1, 14, drc_path_text);
static const char *drc_r0_text[] = {
"1",
@@ -676,8 +676,8 @@ static const char *drc_r0_text[] = {
"0",
};
-static const struct soc_enum drc_r0 =
- SOC_ENUM_SINGLE(WM8993_DRC_CONTROL_3, 8, 6, drc_r0_text);
+static SOC_ENUM_SINGLE_DECL(drc_r0,
+ WM8993_DRC_CONTROL_3, 8, drc_r0_text);
static const char *drc_r1_text[] = {
"1",
@@ -687,8 +687,8 @@ static const char *drc_r1_text[] = {
"0",
};
-static const struct soc_enum drc_r1 =
- SOC_ENUM_SINGLE(WM8993_DRC_CONTROL_4, 13, 5, drc_r1_text);
+static SOC_ENUM_SINGLE_DECL(drc_r1,
+ WM8993_DRC_CONTROL_4, 13, drc_r1_text);
static const char *drc_attack_text[] = {
"Reserved",
@@ -705,8 +705,8 @@ static const char *drc_attack_text[] = {
"185.6ms",
};
-static const struct soc_enum drc_attack =
- SOC_ENUM_SINGLE(WM8993_DRC_CONTROL_2, 12, 12, drc_attack_text);
+static SOC_ENUM_SINGLE_DECL(drc_attack,
+ WM8993_DRC_CONTROL_2, 12, drc_attack_text);
static const char *drc_decay_text[] = {
"186ms",
@@ -720,16 +720,16 @@ static const char *drc_decay_text[] = {
"47.56ms",
};
-static const struct soc_enum drc_decay =
- SOC_ENUM_SINGLE(WM8993_DRC_CONTROL_2, 8, 9, drc_decay_text);
+static SOC_ENUM_SINGLE_DECL(drc_decay,
+ WM8993_DRC_CONTROL_2, 8, drc_decay_text);
static const char *drc_ff_text[] = {
"5 samples",
"9 samples",
};
-static const struct soc_enum drc_ff =
- SOC_ENUM_SINGLE(WM8993_DRC_CONTROL_3, 7, 2, drc_ff_text);
+static SOC_ENUM_SINGLE_DECL(drc_ff,
+ WM8993_DRC_CONTROL_3, 7, drc_ff_text);
static const char *drc_qr_rate_text[] = {
"0.725ms",
@@ -737,8 +737,8 @@ static const char *drc_qr_rate_text[] = {
"5.8ms",
};
-static const struct soc_enum drc_qr_rate =
- SOC_ENUM_SINGLE(WM8993_DRC_CONTROL_3, 0, 3, drc_qr_rate_text);
+static SOC_ENUM_SINGLE_DECL(drc_qr_rate,
+ WM8993_DRC_CONTROL_3, 0, drc_qr_rate_text);
static const char *drc_smooth_text[] = {
"Low",
@@ -746,8 +746,8 @@ static const char *drc_smooth_text[] = {
"High",
};
-static const struct soc_enum drc_smooth =
- SOC_ENUM_SINGLE(WM8993_DRC_CONTROL_1, 4, 3, drc_smooth_text);
+static SOC_ENUM_SINGLE_DECL(drc_smooth,
+ WM8993_DRC_CONTROL_1, 4, drc_smooth_text);
static const struct snd_kcontrol_new wm8993_snd_controls[] = {
SOC_DOUBLE_TLV("Digital Sidetone Volume", WM8993_DIGITAL_SIDE_TONE,
@@ -841,26 +841,26 @@ static const char *aif_text[] = {
"Left", "Right"
};
-static const struct soc_enum aifoutl_enum =
- SOC_ENUM_SINGLE(WM8993_AUDIO_INTERFACE_1, 15, 2, aif_text);
+static SOC_ENUM_SINGLE_DECL(aifoutl_enum,
+ WM8993_AUDIO_INTERFACE_1, 15, aif_text);
static const struct snd_kcontrol_new aifoutl_mux =
SOC_DAPM_ENUM("AIFOUTL Mux", aifoutl_enum);
-static const struct soc_enum aifoutr_enum =
- SOC_ENUM_SINGLE(WM8993_AUDIO_INTERFACE_1, 14, 2, aif_text);
+static SOC_ENUM_SINGLE_DECL(aifoutr_enum,
+ WM8993_AUDIO_INTERFACE_1, 14, aif_text);
static const struct snd_kcontrol_new aifoutr_mux =
SOC_DAPM_ENUM("AIFOUTR Mux", aifoutr_enum);
-static const struct soc_enum aifinl_enum =
- SOC_ENUM_SINGLE(WM8993_AUDIO_INTERFACE_2, 15, 2, aif_text);
+static SOC_ENUM_SINGLE_DECL(aifinl_enum,
+ WM8993_AUDIO_INTERFACE_2, 15, aif_text);
static const struct snd_kcontrol_new aifinl_mux =
SOC_DAPM_ENUM("AIFINL Mux", aifinl_enum);
-static const struct soc_enum aifinr_enum =
- SOC_ENUM_SINGLE(WM8993_AUDIO_INTERFACE_2, 14, 2, aif_text);
+static SOC_ENUM_SINGLE_DECL(aifinr_enum,
+ WM8993_AUDIO_INTERFACE_2, 14, aif_text);
static const struct snd_kcontrol_new aifinr_mux =
SOC_DAPM_ENUM("AIFINR Mux", aifinr_enum);
@@ -869,14 +869,14 @@ static const char *sidetone_text[] = {
"None", "Left", "Right"
};
-static const struct soc_enum sidetonel_enum =
- SOC_ENUM_SINGLE(WM8993_DIGITAL_SIDE_TONE, 2, 3, sidetone_text);
+static SOC_ENUM_SINGLE_DECL(sidetonel_enum,
+ WM8993_DIGITAL_SIDE_TONE, 2, sidetone_text);
static const struct snd_kcontrol_new sidetonel_mux =
SOC_DAPM_ENUM("Left Sidetone", sidetonel_enum);
-static const struct soc_enum sidetoner_enum =
- SOC_ENUM_SINGLE(WM8993_DIGITAL_SIDE_TONE, 0, 3, sidetone_text);
+static SOC_ENUM_SINGLE_DECL(sidetoner_enum,
+ WM8993_DIGITAL_SIDE_TONE, 0, sidetone_text);
static const struct snd_kcontrol_new sidetoner_mux =
SOC_DAPM_ENUM("Right Sidetone", sidetoner_enum);
@@ -1493,13 +1493,6 @@ static int wm8993_probe(struct snd_soc_codec *codec)
wm8993->hubs_data.dcs_codes_r = -2;
wm8993->hubs_data.series_startup = 1;
- codec->control_data = wm8993->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;
- }
-
/* Latch volume update bits and default ZC on */
snd_soc_update_bits(codec, WM8993_RIGHT_DAC_DIGITAL_VOLUME,
WM8993_DAC_VU, WM8993_DAC_VU);
@@ -1559,8 +1552,6 @@ static int wm8993_probe(struct snd_soc_codec *codec)
static int wm8993_remove(struct snd_soc_codec *codec)
{
- struct wm8993_priv *wm8993 = snd_soc_codec_get_drvdata(codec);
-
wm8993_set_bias_level(codec, SND_SOC_BIAS_OFF);
return 0;
}
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index adb7206..6303537 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -1344,8 +1344,7 @@ static const char *adc_mux_text[] = {
"DMIC",
};
-static SOC_ENUM_SINGLE_DECL(adc_enum,
- 0, 0, adc_mux_text);
+static SOC_ENUM_SINGLE_VIRT_DECL(adc_enum, adc_mux_text);
static const struct snd_kcontrol_new adcl_mux =
SOC_DAPM_ENUM_VIRT("ADCL Mux", adc_enum);
@@ -2554,43 +2553,52 @@ static int wm8994_set_bias_level(struct snd_soc_codec *codec,
int wm8994_vmid_mode(struct snd_soc_codec *codec, enum wm8994_vmid_mode mode)
{
struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
switch (mode) {
case WM8994_VMID_NORMAL:
+ snd_soc_dapm_mutex_lock(dapm);
+
if (wm8994->hubs.lineout1_se) {
- snd_soc_dapm_disable_pin(&codec->dapm,
- "LINEOUT1N Driver");
- snd_soc_dapm_disable_pin(&codec->dapm,
- "LINEOUT1P Driver");
+ snd_soc_dapm_disable_pin_unlocked(dapm,
+ "LINEOUT1N Driver");
+ snd_soc_dapm_disable_pin_unlocked(dapm,
+ "LINEOUT1P Driver");
}
if (wm8994->hubs.lineout2_se) {
- snd_soc_dapm_disable_pin(&codec->dapm,
- "LINEOUT2N Driver");
- snd_soc_dapm_disable_pin(&codec->dapm,
- "LINEOUT2P Driver");
+ snd_soc_dapm_disable_pin_unlocked(dapm,
+ "LINEOUT2N Driver");
+ snd_soc_dapm_disable_pin_unlocked(dapm,
+ "LINEOUT2P Driver");
}
/* Do the sync with the old mode to allow it to clean up */
- snd_soc_dapm_sync(&codec->dapm);
+ snd_soc_dapm_sync_unlocked(dapm);
wm8994->vmid_mode = mode;
+
+ snd_soc_dapm_mutex_unlock(dapm);
break;
case WM8994_VMID_FORCE:
+ snd_soc_dapm_mutex_lock(dapm);
+
if (wm8994->hubs.lineout1_se) {
- snd_soc_dapm_force_enable_pin(&codec->dapm,
- "LINEOUT1N Driver");
- snd_soc_dapm_force_enable_pin(&codec->dapm,
- "LINEOUT1P Driver");
+ snd_soc_dapm_force_enable_pin_unlocked(dapm,
+ "LINEOUT1N Driver");
+ snd_soc_dapm_force_enable_pin_unlocked(dapm,
+ "LINEOUT1P Driver");
}
if (wm8994->hubs.lineout2_se) {
- snd_soc_dapm_force_enable_pin(&codec->dapm,
- "LINEOUT2N Driver");
- snd_soc_dapm_force_enable_pin(&codec->dapm,
- "LINEOUT2P Driver");
+ snd_soc_dapm_force_enable_pin_unlocked(dapm,
+ "LINEOUT2N Driver");
+ snd_soc_dapm_force_enable_pin_unlocked(dapm,
+ "LINEOUT2P Driver");
}
wm8994->vmid_mode = mode;
- snd_soc_dapm_sync(&codec->dapm);
+ snd_soc_dapm_sync_unlocked(dapm);
+
+ snd_soc_dapm_mutex_unlock(dapm);
break;
default:
@@ -3242,7 +3250,7 @@ static void wm8994_handle_retune_mobile_pdata(struct wm8994_priv *wm8994)
dev_dbg(codec->dev, "Allocated %d unique ReTune Mobile names\n",
wm8994->num_retune_mobile_texts);
- wm8994->retune_mobile_enum.max = wm8994->num_retune_mobile_texts;
+ wm8994->retune_mobile_enum.items = wm8994->num_retune_mobile_texts;
wm8994->retune_mobile_enum.texts = wm8994->retune_mobile_texts;
ret = snd_soc_add_codec_controls(wm8994->hubs.codec, controls,
@@ -3298,7 +3306,7 @@ static void wm8994_handle_pdata(struct wm8994_priv *wm8994)
for (i = 0; i < pdata->num_drc_cfgs; i++)
wm8994->drc_texts[i] = pdata->drc_cfgs[i].name;
- wm8994->drc_enum.max = pdata->num_drc_cfgs;
+ wm8994->drc_enum.items = pdata->num_drc_cfgs;
wm8994->drc_enum.texts = wm8994->drc_texts;
ret = snd_soc_add_codec_controls(wm8994->hubs.codec, controls,
@@ -3990,9 +3998,8 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec)
int ret, i;
wm8994->hubs.codec = codec;
- codec->control_data = control->regmap;
- snd_soc_codec_set_cache_io(codec, 16, 16, SND_SOC_REGMAP);
+ snd_soc_codec_set_cache_io(codec, control->regmap);
mutex_init(&wm8994->accdet_lock);
INIT_DELAYED_WORK(&wm8994->jackdet_bootstrap,
diff --git a/sound/soc/codecs/wm8995.c b/sound/soc/codecs/wm8995.c
index 4300caf..d3152cf5 100644
--- a/sound/soc/codecs/wm8995.c
+++ b/sound/soc/codecs/wm8995.c
@@ -423,24 +423,24 @@ static const char *in1l_text[] = {
"Differential", "Single-ended IN1LN", "Single-ended IN1LP"
};
-static const SOC_ENUM_SINGLE_DECL(in1l_enum, WM8995_LEFT_LINE_INPUT_CONTROL,
- 2, in1l_text);
+static SOC_ENUM_SINGLE_DECL(in1l_enum, WM8995_LEFT_LINE_INPUT_CONTROL,
+ 2, in1l_text);
static const char *in1r_text[] = {
"Differential", "Single-ended IN1RN", "Single-ended IN1RP"
};
-static const SOC_ENUM_SINGLE_DECL(in1r_enum, WM8995_LEFT_LINE_INPUT_CONTROL,
- 0, in1r_text);
+static SOC_ENUM_SINGLE_DECL(in1r_enum, WM8995_LEFT_LINE_INPUT_CONTROL,
+ 0, in1r_text);
static const char *dmic_src_text[] = {
"DMICDAT1", "DMICDAT2", "DMICDAT3"
};
-static const SOC_ENUM_SINGLE_DECL(dmic_src1_enum, WM8995_POWER_MANAGEMENT_5,
- 8, dmic_src_text);
-static const SOC_ENUM_SINGLE_DECL(dmic_src2_enum, WM8995_POWER_MANAGEMENT_5,
- 6, dmic_src_text);
+static SOC_ENUM_SINGLE_DECL(dmic_src1_enum, WM8995_POWER_MANAGEMENT_5,
+ 8, dmic_src_text);
+static SOC_ENUM_SINGLE_DECL(dmic_src2_enum, WM8995_POWER_MANAGEMENT_5,
+ 6, dmic_src_text);
static const struct snd_kcontrol_new wm8995_snd_controls[] = {
SOC_DOUBLE_R_TLV("DAC1 Volume", WM8995_DAC1_LEFT_VOLUME,
@@ -561,10 +561,8 @@ static int hp_supply_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
struct snd_soc_codec *codec;
- struct wm8995_priv *wm8995;
codec = w->codec;
- wm8995 = snd_soc_codec_get_drvdata(codec);
switch (event) {
case SND_SOC_DAPM_PRE_PMU:
@@ -783,14 +781,12 @@ static const char *sidetone_text[] = {
"ADC/DMIC1", "DMIC2",
};
-static const struct soc_enum sidetone1_enum =
- SOC_ENUM_SINGLE(WM8995_SIDETONE, 0, 2, sidetone_text);
+static SOC_ENUM_SINGLE_DECL(sidetone1_enum, WM8995_SIDETONE, 0, sidetone_text);
static const struct snd_kcontrol_new sidetone1_mux =
SOC_DAPM_ENUM("Left Sidetone Mux", sidetone1_enum);
-static const struct soc_enum sidetone2_enum =
- SOC_ENUM_SINGLE(WM8995_SIDETONE, 1, 2, sidetone_text);
+static SOC_ENUM_SINGLE_DECL(sidetone2_enum, WM8995_SIDETONE, 1, sidetone_text);
static const struct snd_kcontrol_new sidetone2_mux =
SOC_DAPM_ENUM("Right Sidetone Mux", sidetone2_enum);
@@ -886,8 +882,7 @@ static const char *adc_mux_text[] = {
"DMIC",
};
-static const struct soc_enum adc_enum =
- SOC_ENUM_SINGLE(0, 0, 2, adc_mux_text);
+static SOC_ENUM_SINGLE_VIRT_DECL(adc_enum, adc_mux_text);
static const struct snd_kcontrol_new adcl_mux =
SOC_DAPM_ENUM_VIRT("ADCL Mux", adc_enum);
@@ -899,14 +894,14 @@ static const char *spk_src_text[] = {
"DAC1L", "DAC1R", "DAC2L", "DAC2R"
};
-static const SOC_ENUM_SINGLE_DECL(spk1l_src_enum, WM8995_LEFT_PDM_SPEAKER_1,
- 0, spk_src_text);
-static const SOC_ENUM_SINGLE_DECL(spk1r_src_enum, WM8995_RIGHT_PDM_SPEAKER_1,
- 0, spk_src_text);
-static const SOC_ENUM_SINGLE_DECL(spk2l_src_enum, WM8995_LEFT_PDM_SPEAKER_2,
- 0, spk_src_text);
-static const SOC_ENUM_SINGLE_DECL(spk2r_src_enum, WM8995_RIGHT_PDM_SPEAKER_2,
- 0, spk_src_text);
+static SOC_ENUM_SINGLE_DECL(spk1l_src_enum, WM8995_LEFT_PDM_SPEAKER_1,
+ 0, spk_src_text);
+static SOC_ENUM_SINGLE_DECL(spk1r_src_enum, WM8995_RIGHT_PDM_SPEAKER_1,
+ 0, spk_src_text);
+static SOC_ENUM_SINGLE_DECL(spk2l_src_enum, WM8995_LEFT_PDM_SPEAKER_2,
+ 0, spk_src_text);
+static SOC_ENUM_SINGLE_DECL(spk2r_src_enum, WM8995_RIGHT_PDM_SPEAKER_2,
+ 0, spk_src_text);
static const struct snd_kcontrol_new spk1l_mux =
SOC_DAPM_ENUM("SPK1L SRC", spk1l_src_enum);
@@ -2047,13 +2042,6 @@ static int wm8995_probe(struct snd_soc_codec *codec)
wm8995 = snd_soc_codec_get_drvdata(codec);
wm8995->codec = codec;
- codec->control_data = wm8995->regmap;
- ret = snd_soc_codec_set_cache_io(codec, 16, 16, SND_SOC_REGMAP);
- if (ret < 0) {
- dev_err(codec->dev, "Failed to set cache i/o: %d\n", ret);
- return ret;
- }
-
for (i = 0; i < ARRAY_SIZE(wm8995->supplies); i++)
wm8995->supplies[i].supply = wm8995_supply_names[i];
diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c
index 1a7655b..c6cbb3b 100644
--- a/sound/soc/codecs/wm8996.c
+++ b/sound/soc/codecs/wm8996.c
@@ -311,28 +311,28 @@ static const char *sidetone_hpf_text[] = {
"2.9kHz", "1.5kHz", "735Hz", "403Hz", "196Hz", "98Hz", "49Hz"
};
-static const struct soc_enum sidetone_hpf =
- SOC_ENUM_SINGLE(WM8996_SIDETONE, 7, 7, sidetone_hpf_text);
+static SOC_ENUM_SINGLE_DECL(sidetone_hpf,
+ WM8996_SIDETONE, 7, sidetone_hpf_text);
static const char *hpf_mode_text[] = {
"HiFi", "Custom", "Voice"
};
-static const struct soc_enum dsp1tx_hpf_mode =
- SOC_ENUM_SINGLE(WM8996_DSP1_TX_FILTERS, 3, 3, hpf_mode_text);
+static SOC_ENUM_SINGLE_DECL(dsp1tx_hpf_mode,
+ WM8996_DSP1_TX_FILTERS, 3, hpf_mode_text);
-static const struct soc_enum dsp2tx_hpf_mode =
- SOC_ENUM_SINGLE(WM8996_DSP2_TX_FILTERS, 3, 3, hpf_mode_text);
+static SOC_ENUM_SINGLE_DECL(dsp2tx_hpf_mode,
+ WM8996_DSP2_TX_FILTERS, 3, hpf_mode_text);
static const char *hpf_cutoff_text[] = {
"50Hz", "75Hz", "100Hz", "150Hz", "200Hz", "300Hz", "400Hz"
};
-static const struct soc_enum dsp1tx_hpf_cutoff =
- SOC_ENUM_SINGLE(WM8996_DSP1_TX_FILTERS, 0, 7, hpf_cutoff_text);
+static SOC_ENUM_SINGLE_DECL(dsp1tx_hpf_cutoff,
+ WM8996_DSP1_TX_FILTERS, 0, hpf_cutoff_text);
-static const struct soc_enum dsp2tx_hpf_cutoff =
- SOC_ENUM_SINGLE(WM8996_DSP2_TX_FILTERS, 0, 7, hpf_cutoff_text);
+static SOC_ENUM_SINGLE_DECL(dsp2tx_hpf_cutoff,
+ WM8996_DSP2_TX_FILTERS, 0, hpf_cutoff_text);
static void wm8996_set_retune_mobile(struct snd_soc_codec *codec, int block)
{
@@ -780,14 +780,14 @@ static const char *sidetone_text[] = {
"IN1", "IN2",
};
-static const struct soc_enum left_sidetone_enum =
- SOC_ENUM_SINGLE(WM8996_SIDETONE, 0, 2, sidetone_text);
+static SOC_ENUM_SINGLE_DECL(left_sidetone_enum,
+ WM8996_SIDETONE, 0, sidetone_text);
static const struct snd_kcontrol_new left_sidetone =
SOC_DAPM_ENUM("Left Sidetone", left_sidetone_enum);
-static const struct soc_enum right_sidetone_enum =
- SOC_ENUM_SINGLE(WM8996_SIDETONE, 1, 2, sidetone_text);
+static SOC_ENUM_SINGLE_DECL(right_sidetone_enum,
+ WM8996_SIDETONE, 1, sidetone_text);
static const struct snd_kcontrol_new right_sidetone =
SOC_DAPM_ENUM("Right Sidetone", right_sidetone_enum);
@@ -796,14 +796,14 @@ static const char *spk_text[] = {
"DAC1L", "DAC1R", "DAC2L", "DAC2R"
};
-static const struct soc_enum spkl_enum =
- SOC_ENUM_SINGLE(WM8996_LEFT_PDM_SPEAKER, 0, 4, spk_text);
+static SOC_ENUM_SINGLE_DECL(spkl_enum,
+ WM8996_LEFT_PDM_SPEAKER, 0, spk_text);
static const struct snd_kcontrol_new spkl_mux =
SOC_DAPM_ENUM("SPKL", spkl_enum);
-static const struct soc_enum spkr_enum =
- SOC_ENUM_SINGLE(WM8996_RIGHT_PDM_SPEAKER, 0, 4, spk_text);
+static SOC_ENUM_SINGLE_DECL(spkr_enum,
+ WM8996_RIGHT_PDM_SPEAKER, 0, spk_text);
static const struct snd_kcontrol_new spkr_mux =
SOC_DAPM_ENUM("SPKR", spkr_enum);
@@ -812,8 +812,8 @@ static const char *dsp1rx_text[] = {
"AIF1", "AIF2"
};
-static const struct soc_enum dsp1rx_enum =
- SOC_ENUM_SINGLE(WM8996_POWER_MANAGEMENT_8, 0, 2, dsp1rx_text);
+static SOC_ENUM_SINGLE_DECL(dsp1rx_enum,
+ WM8996_POWER_MANAGEMENT_8, 0, dsp1rx_text);
static const struct snd_kcontrol_new dsp1rx =
SOC_DAPM_ENUM("DSP1RX", dsp1rx_enum);
@@ -822,8 +822,8 @@ static const char *dsp2rx_text[] = {
"AIF2", "AIF1"
};
-static const struct soc_enum dsp2rx_enum =
- SOC_ENUM_SINGLE(WM8996_POWER_MANAGEMENT_8, 4, 2, dsp2rx_text);
+static SOC_ENUM_SINGLE_DECL(dsp2rx_enum,
+ WM8996_POWER_MANAGEMENT_8, 4, dsp2rx_text);
static const struct snd_kcontrol_new dsp2rx =
SOC_DAPM_ENUM("DSP2RX", dsp2rx_enum);
@@ -832,8 +832,8 @@ static const char *aif2tx_text[] = {
"DSP2", "DSP1", "AIF1"
};
-static const struct soc_enum aif2tx_enum =
- SOC_ENUM_SINGLE(WM8996_POWER_MANAGEMENT_8, 6, 3, aif2tx_text);
+static SOC_ENUM_SINGLE_DECL(aif2tx_enum,
+ WM8996_POWER_MANAGEMENT_8, 6, aif2tx_text);
static const struct snd_kcontrol_new aif2tx =
SOC_DAPM_ENUM("AIF2TX", aif2tx_enum);
@@ -842,14 +842,14 @@ static const char *inmux_text[] = {
"ADC", "DMIC1", "DMIC2"
};
-static const struct soc_enum in1_enum =
- SOC_ENUM_SINGLE(WM8996_POWER_MANAGEMENT_7, 0, 3, inmux_text);
+static SOC_ENUM_SINGLE_DECL(in1_enum,
+ WM8996_POWER_MANAGEMENT_7, 0, inmux_text);
static const struct snd_kcontrol_new in1_mux =
SOC_DAPM_ENUM("IN1 Mux", in1_enum);
-static const struct soc_enum in2_enum =
- SOC_ENUM_SINGLE(WM8996_POWER_MANAGEMENT_7, 4, 3, inmux_text);
+static SOC_ENUM_SINGLE_DECL(in2_enum,
+ WM8996_POWER_MANAGEMENT_7, 4, inmux_text);
static const struct snd_kcontrol_new in2_mux =
SOC_DAPM_ENUM("IN2 Mux", in2_enum);
@@ -1608,8 +1608,8 @@ static int wm8996_set_bias_level(struct snd_soc_codec *codec,
msleep(5);
}
- regcache_cache_only(codec->control_data, false);
- regcache_sync(codec->control_data);
+ regcache_cache_only(wm8996->regmap, false);
+ regcache_sync(wm8996->regmap);
}
/* Bypass the MICBIASes for lowest power */
@@ -1620,10 +1620,10 @@ static int wm8996_set_bias_level(struct snd_soc_codec *codec,
break;
case SND_SOC_BIAS_OFF:
- regcache_cache_only(codec->control_data, true);
+ regcache_cache_only(wm8996->regmap, true);
if (wm8996->pdata.ldo_ena >= 0) {
gpio_set_value_cansleep(wm8996->pdata.ldo_ena, 0);
- regcache_cache_only(codec->control_data, true);
+ regcache_cache_only(wm8996->regmap, true);
}
regulator_bulk_disable(ARRAY_SIZE(wm8996->supplies),
wm8996->supplies);
@@ -2251,6 +2251,7 @@ int wm8996_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack,
wm8996_polarity_fn polarity_cb)
{
struct wm8996_priv *wm8996 = snd_soc_codec_get_drvdata(codec);
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
wm8996->jack = jack;
wm8996->detecting = true;
@@ -2267,8 +2268,12 @@ int wm8996_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack,
WM8996_MICB2_DISCH, 0);
/* LDO2 powers the microphones, SYSCLK clocks detection */
- snd_soc_dapm_force_enable_pin(&codec->dapm, "LDO2");
- snd_soc_dapm_force_enable_pin(&codec->dapm, "SYSCLK");
+ snd_soc_dapm_mutex_lock(dapm);
+
+ snd_soc_dapm_force_enable_pin_unlocked(dapm, "LDO2");
+ snd_soc_dapm_force_enable_pin_unlocked(dapm, "SYSCLK");
+
+ snd_soc_dapm_mutex_unlock(dapm);
/* We start off just enabling microphone detection - even a
* plain headphone will trigger detection.
@@ -2595,7 +2600,7 @@ static void wm8996_retune_mobile_pdata(struct snd_soc_codec *codec)
dev_dbg(codec->dev, "Allocated %d unique ReTune Mobile names\n",
wm8996->num_retune_mobile_texts);
- wm8996->retune_mobile_enum.max = wm8996->num_retune_mobile_texts;
+ wm8996->retune_mobile_enum.items = wm8996->num_retune_mobile_texts;
wm8996->retune_mobile_enum.texts = wm8996->retune_mobile_texts;
ret = snd_soc_add_codec_controls(codec, controls, ARRAY_SIZE(controls));
@@ -2628,14 +2633,6 @@ static int wm8996_probe(struct snd_soc_codec *codec)
init_completion(&wm8996->dcs_done);
init_completion(&wm8996->fll_lock);
- codec->control_data = wm8996->regmap;
-
- ret = snd_soc_codec_set_cache_io(codec, 16, 16, SND_SOC_REGMAP);
- if (ret != 0) {
- dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
- goto err;
- }
-
if (wm8996->pdata.num_retune_mobile_cfgs)
wm8996_retune_mobile_pdata(codec);
else
@@ -2674,13 +2671,11 @@ static int wm8996_probe(struct snd_soc_codec *codec)
} else {
dev_err(codec->dev, "Failed to request IRQ: %d\n",
ret);
+ return ret;
}
}
return 0;
-
-err:
- return ret;
}
static int wm8996_remove(struct snd_soc_codec *codec)
diff --git a/sound/soc/codecs/wm8997.c b/sound/soc/codecs/wm8997.c
index 555115e..004186b 100644
--- a/sound/soc/codecs/wm8997.c
+++ b/sound/soc/codecs/wm8997.c
@@ -86,7 +86,7 @@ static int wm8997_sysclk_ev(struct snd_soc_dapm_widget *w,
{
struct snd_soc_codec *codec = w->codec;
struct arizona *arizona = dev_get_drvdata(codec->dev->parent);
- struct regmap *regmap = codec->control_data;
+ struct regmap *regmap = arizona->regmap;
const struct reg_default *patch = NULL;
int i, patch_size;
@@ -123,10 +123,12 @@ static const unsigned int wm8997_osr_val[] = {
static const struct soc_enum wm8997_hpout_osr[] = {
SOC_VALUE_ENUM_SINGLE(ARIZONA_OUTPUT_PATH_CONFIG_1L,
- ARIZONA_OUT1_OSR_SHIFT, 0x7, 3,
+ ARIZONA_OUT1_OSR_SHIFT, 0x7,
+ ARRAY_SIZE(wm8997_osr_text),
wm8997_osr_text, wm8997_osr_val),
SOC_VALUE_ENUM_SINGLE(ARIZONA_OUTPUT_PATH_CONFIG_3L,
- ARIZONA_OUT3_OSR_SHIFT, 0x7, 3,
+ ARIZONA_OUT3_OSR_SHIFT, 0x7,
+ ARRAY_SIZE(wm8997_osr_text),
wm8997_osr_text, wm8997_osr_val),
};
@@ -170,15 +172,8 @@ ARIZONA_MIXER_CONTROLS("EQ2", ARIZONA_EQ2MIX_INPUT_1_SOURCE),
ARIZONA_MIXER_CONTROLS("EQ3", ARIZONA_EQ3MIX_INPUT_1_SOURCE),
ARIZONA_MIXER_CONTROLS("EQ4", ARIZONA_EQ4MIX_INPUT_1_SOURCE),
-SND_SOC_BYTES_MASK("EQ1 Coefficients", ARIZONA_EQ1_1, 21,
- ARIZONA_EQ1_ENA_MASK),
-SND_SOC_BYTES_MASK("EQ2 Coefficients", ARIZONA_EQ2_1, 21,
- ARIZONA_EQ2_ENA_MASK),
-SND_SOC_BYTES_MASK("EQ3 Coefficients", ARIZONA_EQ3_1, 21,
- ARIZONA_EQ3_ENA_MASK),
-SND_SOC_BYTES_MASK("EQ4 Coefficients", ARIZONA_EQ4_1, 21,
- ARIZONA_EQ4_ENA_MASK),
-
+SND_SOC_BYTES("EQ1 Coefficients", ARIZONA_EQ1_3, 19),
+SOC_SINGLE("EQ1 Mode Switch", ARIZONA_EQ1_2, ARIZONA_EQ1_B1_MODE, 1, 0),
SOC_SINGLE_TLV("EQ1 B1 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B1_GAIN_SHIFT,
24, 0, eq_tlv),
SOC_SINGLE_TLV("EQ1 B2 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B2_GAIN_SHIFT,
@@ -190,6 +185,8 @@ SOC_SINGLE_TLV("EQ1 B4 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B4_GAIN_SHIFT,
SOC_SINGLE_TLV("EQ1 B5 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B5_GAIN_SHIFT,
24, 0, eq_tlv),
+SND_SOC_BYTES("EQ2 Coefficients", ARIZONA_EQ2_3, 19),
+SOC_SINGLE("EQ2 Mode Switch", ARIZONA_EQ2_2, ARIZONA_EQ2_B1_MODE, 1, 0),
SOC_SINGLE_TLV("EQ2 B1 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B1_GAIN_SHIFT,
24, 0, eq_tlv),
SOC_SINGLE_TLV("EQ2 B2 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B2_GAIN_SHIFT,
@@ -201,6 +198,8 @@ SOC_SINGLE_TLV("EQ2 B4 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B4_GAIN_SHIFT,
SOC_SINGLE_TLV("EQ2 B5 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B5_GAIN_SHIFT,
24, 0, eq_tlv),
+SND_SOC_BYTES("EQ3 Coefficients", ARIZONA_EQ3_3, 19),
+SOC_SINGLE("EQ3 Mode Switch", ARIZONA_EQ3_2, ARIZONA_EQ3_B1_MODE, 1, 0),
SOC_SINGLE_TLV("EQ3 B1 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B1_GAIN_SHIFT,
24, 0, eq_tlv),
SOC_SINGLE_TLV("EQ3 B2 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B2_GAIN_SHIFT,
@@ -212,6 +211,8 @@ SOC_SINGLE_TLV("EQ3 B4 Volume", ARIZONA_EQ3_2, ARIZONA_EQ3_B4_GAIN_SHIFT,
SOC_SINGLE_TLV("EQ3 B5 Volume", ARIZONA_EQ3_2, ARIZONA_EQ3_B5_GAIN_SHIFT,
24, 0, eq_tlv),
+SND_SOC_BYTES("EQ4 Coefficients", ARIZONA_EQ4_3, 19),
+SOC_SINGLE("EQ4 Mode Switch", ARIZONA_EQ4_2, ARIZONA_EQ4_B1_MODE, 1, 0),
SOC_SINGLE_TLV("EQ4 B1 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B1_GAIN_SHIFT,
24, 0, eq_tlv),
SOC_SINGLE_TLV("EQ4 B2 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B2_GAIN_SHIFT,
@@ -1052,9 +1053,7 @@ static int wm8997_codec_probe(struct snd_soc_codec *codec)
struct wm8997_priv *priv = snd_soc_codec_get_drvdata(codec);
int ret;
- codec->control_data = priv->core.arizona->regmap;
-
- ret = snd_soc_codec_set_cache_io(codec, 32, 16, SND_SOC_REGMAP);
+ ret = snd_soc_codec_set_cache_io(codec, priv->core.arizona->regmap);
if (ret != 0)
return ret;
diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c
index 0982c1d..d18eff3 100644
--- a/sound/soc/codecs/wm9081.c
+++ b/sound/soc/codecs/wm9081.c
@@ -268,8 +268,7 @@ static const char *drc_high_text[] = {
"0",
};
-static const struct soc_enum drc_high =
- SOC_ENUM_SINGLE(WM9081_DRC_3, 3, 6, drc_high_text);
+static SOC_ENUM_SINGLE_DECL(drc_high, WM9081_DRC_3, 3, drc_high_text);
static const char *drc_low_text[] = {
"1",
@@ -279,8 +278,7 @@ static const char *drc_low_text[] = {
"0",
};
-static const struct soc_enum drc_low =
- SOC_ENUM_SINGLE(WM9081_DRC_3, 0, 5, drc_low_text);
+static SOC_ENUM_SINGLE_DECL(drc_low, WM9081_DRC_3, 0, drc_low_text);
static const char *drc_atk_text[] = {
"181us",
@@ -297,8 +295,7 @@ static const char *drc_atk_text[] = {
"185.6ms",
};
-static const struct soc_enum drc_atk =
- SOC_ENUM_SINGLE(WM9081_DRC_2, 12, 12, drc_atk_text);
+static SOC_ENUM_SINGLE_DECL(drc_atk, WM9081_DRC_2, 12, drc_atk_text);
static const char *drc_dcy_text[] = {
"186ms",
@@ -312,8 +309,7 @@ static const char *drc_dcy_text[] = {
"47.56s",
};
-static const struct soc_enum drc_dcy =
- SOC_ENUM_SINGLE(WM9081_DRC_2, 8, 9, drc_dcy_text);
+static SOC_ENUM_SINGLE_DECL(drc_dcy, WM9081_DRC_2, 8, drc_dcy_text);
static const char *drc_qr_dcy_text[] = {
"0.725ms",
@@ -321,8 +317,7 @@ static const char *drc_qr_dcy_text[] = {
"5.8ms",
};
-static const struct soc_enum drc_qr_dcy =
- SOC_ENUM_SINGLE(WM9081_DRC_2, 4, 3, drc_qr_dcy_text);
+static SOC_ENUM_SINGLE_DECL(drc_qr_dcy, WM9081_DRC_2, 4, drc_qr_dcy_text);
static const char *dac_deemph_text[] = {
"None",
@@ -331,16 +326,16 @@ static const char *dac_deemph_text[] = {
"48kHz",
};
-static const struct soc_enum dac_deemph =
- SOC_ENUM_SINGLE(WM9081_DAC_DIGITAL_2, 1, 4, dac_deemph_text);
+static SOC_ENUM_SINGLE_DECL(dac_deemph, WM9081_DAC_DIGITAL_2, 1,
+ dac_deemph_text);
static const char *speaker_mode_text[] = {
"Class D",
"Class AB",
};
-static const struct soc_enum speaker_mode =
- SOC_ENUM_SINGLE(WM9081_ANALOGUE_SPEAKER_2, 6, 2, speaker_mode_text);
+static SOC_ENUM_SINGLE_DECL(speaker_mode, WM9081_ANALOGUE_SPEAKER_2, 6,
+ speaker_mode_text);
static int speaker_mode_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
@@ -1265,15 +1260,6 @@ static struct snd_soc_dai_driver wm9081_dai = {
static int wm9081_probe(struct snd_soc_codec *codec)
{
struct wm9081_priv *wm9081 = snd_soc_codec_get_drvdata(codec);
- int ret;
-
- codec->control_data = wm9081->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;
- }
/* Enable zero cross by default */
snd_soc_update_bits(codec, WM9081_ANALOGUE_LINEOUT,
@@ -1288,7 +1274,7 @@ static int wm9081_probe(struct snd_soc_codec *codec)
ARRAY_SIZE(wm9081_eq_controls));
}
- return ret;
+ return 0;
}
static int wm9081_remove(struct snd_soc_codec *codec)
diff --git a/sound/soc/codecs/wm9090.c b/sound/soc/codecs/wm9090.c
index a07fe16..8793417 100644
--- a/sound/soc/codecs/wm9090.c
+++ b/sound/soc/codecs/wm9090.c
@@ -522,16 +522,6 @@ static int wm9090_set_bias_level(struct snd_soc_codec *codec,
static int wm9090_probe(struct snd_soc_codec *codec)
{
- struct wm9090_priv *wm9090 = dev_get_drvdata(codec->dev);
- int ret;
-
- codec->control_data = wm9090->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;
- }
-
/* Configure some defaults; they will be written out when we
* bring the bias up.
*/
diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c
index 70ce6793..c0b7f45 100644
--- a/sound/soc/codecs/wm9705.c
+++ b/sound/soc/codecs/wm9705.c
@@ -67,12 +67,12 @@ static const char *wm9705_mic[] = {"Mic 1", "Mic 2"};
static const char *wm9705_rec_sel[] = {"Mic", "CD", "NC", "NC",
"Line", "Stereo Mix", "Mono Mix", "Phone"};
-static const struct soc_enum wm9705_enum_mic =
- SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 8, 2, wm9705_mic);
-static const struct soc_enum wm9705_enum_rec_l =
- SOC_ENUM_SINGLE(AC97_REC_SEL, 8, 8, wm9705_rec_sel);
-static const struct soc_enum wm9705_enum_rec_r =
- SOC_ENUM_SINGLE(AC97_REC_SEL, 0, 8, wm9705_rec_sel);
+static SOC_ENUM_SINGLE_DECL(wm9705_enum_mic,
+ AC97_GENERAL_PURPOSE, 8, wm9705_mic);
+static SOC_ENUM_SINGLE_DECL(wm9705_enum_rec_l,
+ AC97_REC_SEL, 8, wm9705_rec_sel);
+static SOC_ENUM_SINGLE_DECL(wm9705_enum_rec_r,
+ AC97_REC_SEL, 0, wm9705_rec_sel);
/* Headphone Mixer */
static const struct snd_kcontrol_new wm9705_hp_mixer_controls[] = {
diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c
index 444626f..bb5f7b4 100644
--- a/sound/soc/codecs/wm_adsp.c
+++ b/sound/soc/codecs/wm_adsp.c
@@ -684,24 +684,38 @@ static int wm_adsp_load(struct wm_adsp *dsp)
}
if (reg) {
- buf = wm_adsp_buf_alloc(region->data,
- le32_to_cpu(region->len),
- &buf_list);
- if (!buf) {
- adsp_err(dsp, "Out of memory\n");
- ret = -ENOMEM;
- goto out_fw;
- }
+ size_t to_write = PAGE_SIZE;
+ size_t remain = le32_to_cpu(region->len);
+ const u8 *data = region->data;
+
+ while (remain > 0) {
+ if (remain < PAGE_SIZE)
+ to_write = remain;
+
+ buf = wm_adsp_buf_alloc(data,
+ to_write,
+ &buf_list);
+ if (!buf) {
+ adsp_err(dsp, "Out of memory\n");
+ ret = -ENOMEM;
+ goto out_fw;
+ }
- ret = regmap_raw_write_async(regmap, reg, buf->buf,
- le32_to_cpu(region->len));
- if (ret != 0) {
- adsp_err(dsp,
- "%s.%d: Failed to write %d bytes at %d in %s: %d\n",
- file, regions,
- le32_to_cpu(region->len), offset,
- region_name, ret);
- goto out_fw;
+ ret = regmap_raw_write_async(regmap, reg,
+ buf->buf,
+ to_write);
+ if (ret != 0) {
+ adsp_err(dsp,
+ "%s.%d: Failed to write %zd bytes at %d in %s: %d\n",
+ file, regions,
+ to_write, offset,
+ region_name, ret);
+ goto out_fw;
+ }
+
+ data += to_write;
+ reg += to_write / 2;
+ remain -= to_write;
}
}
@@ -1679,6 +1693,8 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w,
list_del(&alg_region->list);
kfree(alg_region);
}
+
+ adsp_dbg(dsp, "Shutdown complete\n");
break;
default:
diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c
index b371066..b620966 100644
--- a/sound/soc/codecs/wm_hubs.c
+++ b/sound/soc/codecs/wm_hubs.c
@@ -50,16 +50,16 @@ static const char *speaker_ref_text[] = {
"VMID",
};
-static const struct soc_enum speaker_ref =
- SOC_ENUM_SINGLE(WM8993_SPEAKER_MIXER, 8, 2, speaker_ref_text);
+static SOC_ENUM_SINGLE_DECL(speaker_ref,
+ WM8993_SPEAKER_MIXER, 8, speaker_ref_text);
static const char *speaker_mode_text[] = {
"Class D",
"Class AB",
};
-static const struct soc_enum speaker_mode =
- SOC_ENUM_SINGLE(WM8993_SPKMIXR_ATTENUATION, 8, 2, speaker_mode_text);
+static SOC_ENUM_SINGLE_DECL(speaker_mode,
+ WM8993_SPKMIXR_ATTENUATION, 8, speaker_mode_text);
static void wait_for_dc_servo(struct snd_soc_codec *codec, unsigned int op)
{
@@ -735,15 +735,15 @@ static const char *hp_mux_text[] = {
"DAC",
};
-static const struct soc_enum hpl_enum =
- SOC_ENUM_SINGLE(WM8993_OUTPUT_MIXER1, 8, 2, hp_mux_text);
+static SOC_ENUM_SINGLE_DECL(hpl_enum,
+ WM8993_OUTPUT_MIXER1, 8, 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);
+static SOC_ENUM_SINGLE_DECL(hpr_enum,
+ WM8993_OUTPUT_MIXER2, 8, hp_mux_text);
const struct snd_kcontrol_new wm_hubs_hpr_mux =
WM_HUBS_ENUM_W("Right Headphone Mux", hpr_enum);
diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c
index 5e3bc3c..cab98a5 100644
--- a/sound/soc/davinci/davinci-evm.c
+++ b/sound/soc/davinci/davinci-evm.c
@@ -17,6 +17,7 @@
#include <linux/platform_data/edma.h>
#include <linux/i2c.h>
#include <linux/of_platform.h>
+#include <linux/clk.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/soc.h>
@@ -30,9 +31,34 @@
#include "davinci-i2s.h"
struct snd_soc_card_drvdata_davinci {
+ struct clk *mclk;
unsigned sysclk;
};
+static int evm_startup(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_card *soc_card = rtd->codec->card;
+ struct snd_soc_card_drvdata_davinci *drvdata =
+ snd_soc_card_get_drvdata(soc_card);
+
+ if (drvdata->mclk)
+ return clk_prepare_enable(drvdata->mclk);
+
+ return 0;
+}
+
+static void evm_shutdown(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_card *soc_card = rtd->codec->card;
+ struct snd_soc_card_drvdata_davinci *drvdata =
+ snd_soc_card_get_drvdata(soc_card);
+
+ if (drvdata->mclk)
+ clk_disable_unprepare(drvdata->mclk);
+}
+
static int evm_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
@@ -59,6 +85,8 @@ static int evm_hw_params(struct snd_pcm_substream *substream,
}
static struct snd_soc_ops evm_ops = {
+ .startup = evm_startup,
+ .shutdown = evm_shutdown,
.hw_params = evm_hw_params,
};
@@ -95,35 +123,29 @@ static const struct snd_soc_dapm_route audio_map[] = {
/* Logic for a aic3x as connected on a davinci-evm */
static int evm_aic3x_init(struct snd_soc_pcm_runtime *rtd)
{
+ struct snd_soc_card *card = rtd->card;
struct snd_soc_codec *codec = rtd->codec;
- struct snd_soc_dapm_context *dapm = &codec->dapm;
struct device_node *np = codec->card->dev->of_node;
int ret;
/* Add davinci-evm specific widgets */
- snd_soc_dapm_new_controls(dapm, aic3x_dapm_widgets,
+ snd_soc_dapm_new_controls(&card->dapm, aic3x_dapm_widgets,
ARRAY_SIZE(aic3x_dapm_widgets));
if (np) {
- ret = snd_soc_of_parse_audio_routing(codec->card,
- "ti,audio-routing");
+ ret = snd_soc_of_parse_audio_routing(card, "ti,audio-routing");
if (ret)
return ret;
} else {
/* Set up davinci-evm specific audio path audio_map */
- snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
+ snd_soc_dapm_add_routes(&card->dapm, audio_map,
+ ARRAY_SIZE(audio_map));
}
/* not connected */
- snd_soc_dapm_disable_pin(dapm, "MONO_LOUT");
- snd_soc_dapm_disable_pin(dapm, "HPLCOM");
- snd_soc_dapm_disable_pin(dapm, "HPRCOM");
-
- /* always connected */
- snd_soc_dapm_enable_pin(dapm, "Headphone Jack");
- snd_soc_dapm_enable_pin(dapm, "Line Out");
- snd_soc_dapm_enable_pin(dapm, "Mic Jack");
- snd_soc_dapm_enable_pin(dapm, "Line In");
+ snd_soc_dapm_nc_pin(&codec->dapm, "MONO_LOUT");
+ snd_soc_dapm_nc_pin(&codec->dapm, "HPLCOM");
+ snd_soc_dapm_nc_pin(&codec->dapm, "HPRCOM");
return 0;
}
@@ -348,6 +370,7 @@ static int davinci_evm_probe(struct platform_device *pdev)
of_match_device(of_match_ptr(davinci_evm_dt_ids), &pdev->dev);
struct snd_soc_dai_link *dai = (struct snd_soc_dai_link *) match->data;
struct snd_soc_card_drvdata_davinci *drvdata = NULL;
+ struct clk *mclk;
int ret = 0;
evm_soc_card.dai_link = dai;
@@ -367,13 +390,38 @@ static int davinci_evm_probe(struct platform_device *pdev)
if (ret)
return ret;
+ mclk = devm_clk_get(&pdev->dev, "mclk");
+ if (PTR_ERR(mclk) == -EPROBE_DEFER) {
+ return -EPROBE_DEFER;
+ } else if (IS_ERR(mclk)) {
+ dev_dbg(&pdev->dev, "mclk not found.\n");
+ mclk = NULL;
+ }
+
drvdata = devm_kzalloc(&pdev->dev, sizeof(*drvdata), GFP_KERNEL);
if (!drvdata)
return -ENOMEM;
+ drvdata->mclk = mclk;
+
ret = of_property_read_u32(np, "ti,codec-clock-rate", &drvdata->sysclk);
- if (ret < 0)
- return -EINVAL;
+
+ if (ret < 0) {
+ if (!drvdata->mclk) {
+ dev_err(&pdev->dev,
+ "No clock or clock rate defined.\n");
+ return -EINVAL;
+ }
+ drvdata->sysclk = clk_get_rate(drvdata->mclk);
+ } else if (drvdata->mclk) {
+ unsigned int requestd_rate = drvdata->sysclk;
+ clk_set_rate(drvdata->mclk, drvdata->sysclk);
+ drvdata->sysclk = clk_get_rate(drvdata->mclk);
+ if (drvdata->sysclk != requestd_rate)
+ dev_warn(&pdev->dev,
+ "Could not get requested rate %u using %u.\n",
+ requestd_rate, drvdata->sysclk);
+ }
snd_soc_card_set_drvdata(&evm_soc_card, drvdata);
ret = devm_snd_soc_register_card(&pdev->dev, &evm_soc_card);
diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index 670afa2..a01ae97 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -37,6 +37,16 @@
#include "davinci-pcm.h"
#include "davinci-mcasp.h"
+struct davinci_mcasp_context {
+ u32 txfmtctl;
+ u32 rxfmtctl;
+ u32 txfmt;
+ u32 rxfmt;
+ u32 aclkxctl;
+ u32 aclkrctl;
+ u32 pdir;
+};
+
struct davinci_mcasp {
struct davinci_pcm_dma_params dma_params[2];
struct snd_dmaengine_dai_dma_data dma_data[2];
@@ -53,6 +63,9 @@ struct davinci_mcasp {
u16 bclk_lrclk_ratio;
int streams;
+ int sysclk_freq;
+ bool bclk_master;
+
/* McASP FIFO related */
u8 txnumevt;
u8 rxnumevt;
@@ -60,15 +73,7 @@ struct davinci_mcasp {
bool dat_port;
#ifdef CONFIG_PM_SLEEP
- struct {
- u32 txfmtctl;
- u32 rxfmtctl;
- u32 txfmt;
- u32 rxfmt;
- u32 aclkxctl;
- u32 aclkrctl;
- u32 pdir;
- } context;
+ struct davinci_mcasp_context context;
#endif
};
@@ -294,6 +299,7 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai,
mcasp_set_bits(mcasp, DAVINCI_MCASP_PDIR_REG, ACLKX | ACLKR);
mcasp_set_bits(mcasp, DAVINCI_MCASP_PDIR_REG, AFSX | AFSR);
+ mcasp->bclk_master = 1;
break;
case SND_SOC_DAIFMT_CBM_CFS:
/* codec is clock master and frame slave */
@@ -305,6 +311,7 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai,
mcasp_clr_bits(mcasp, DAVINCI_MCASP_PDIR_REG, ACLKX | ACLKR);
mcasp_set_bits(mcasp, DAVINCI_MCASP_PDIR_REG, AFSX | AFSR);
+ mcasp->bclk_master = 0;
break;
case SND_SOC_DAIFMT_CBM_CFM:
/* codec is clock and frame master */
@@ -316,6 +323,7 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai,
mcasp_clr_bits(mcasp, DAVINCI_MCASP_PDIR_REG,
ACLKX | AHCLKX | AFSX | ACLKR | AHCLKR | AFSR);
+ mcasp->bclk_master = 0;
break;
default:
@@ -410,6 +418,8 @@ static int davinci_mcasp_set_sysclk(struct snd_soc_dai *dai, int clk_id,
mcasp_clr_bits(mcasp, DAVINCI_MCASP_PDIR_REG, AHCLKX);
}
+ mcasp->sysclk_freq = freq;
+
return 0;
}
@@ -603,20 +613,23 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream,
u8 fifo_level;
u8 slots = mcasp->tdm_slots;
u8 active_serializers;
- int channels;
+ int channels = params_channels(params);
int ret;
- struct snd_interval *pcm_channels = hw_param_interval(params,
- SNDRV_PCM_HW_PARAM_CHANNELS);
- channels = pcm_channels->min;
- active_serializers = (channels + slots - 1) / slots;
+ /* If mcasp is BCLK master we need to set BCLK divider */
+ if (mcasp->bclk_master) {
+ unsigned int bclk_freq = snd_soc_params_to_bclk(params);
+ if (mcasp->sysclk_freq % bclk_freq != 0) {
+ dev_err(mcasp->dev, "Can't produce requred BCLK\n");
+ return -EINVAL;
+ }
+ davinci_mcasp_set_clkdiv(
+ cpu_dai, 1, mcasp->sysclk_freq / bclk_freq);
+ }
- if (mcasp_common_hw_param(mcasp, substream->stream, channels) == -EINVAL)
- return -EINVAL;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
- fifo_level = mcasp->txnumevt * active_serializers;
- else
- fifo_level = mcasp->rxnumevt * active_serializers;
+ ret = mcasp_common_hw_param(mcasp, substream->stream, channels);
+ if (ret)
+ return ret;
if (mcasp->op_mode == DAVINCI_MCASP_DIT_MODE)
ret = mcasp_dit_hw_param(mcasp);
@@ -658,6 +671,13 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream,
return -EINVAL;
}
+ /* Calculate FIFO level */
+ active_serializers = (channels + slots - 1) / slots;
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ fifo_level = mcasp->txnumevt * active_serializers;
+ else
+ fifo_level = mcasp->rxnumevt * active_serializers;
+
if (mcasp->version == MCASP_VERSION_2 && !fifo_level)
dma_params->acnt = 4;
else
@@ -719,6 +739,43 @@ static const struct snd_soc_dai_ops davinci_mcasp_dai_ops = {
.set_sysclk = davinci_mcasp_set_sysclk,
};
+#ifdef CONFIG_PM_SLEEP
+static int davinci_mcasp_suspend(struct snd_soc_dai *dai)
+{
+ struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(dai);
+ struct davinci_mcasp_context *context = &mcasp->context;
+
+ context->txfmtctl = mcasp_get_reg(mcasp, DAVINCI_MCASP_TXFMCTL_REG);
+ context->rxfmtctl = mcasp_get_reg(mcasp, DAVINCI_MCASP_RXFMCTL_REG);
+ context->txfmt = mcasp_get_reg(mcasp, DAVINCI_MCASP_TXFMT_REG);
+ context->rxfmt = mcasp_get_reg(mcasp, DAVINCI_MCASP_RXFMT_REG);
+ context->aclkxctl = mcasp_get_reg(mcasp, DAVINCI_MCASP_ACLKXCTL_REG);
+ context->aclkrctl = mcasp_get_reg(mcasp, DAVINCI_MCASP_ACLKRCTL_REG);
+ context->pdir = mcasp_get_reg(mcasp, DAVINCI_MCASP_PDIR_REG);
+
+ return 0;
+}
+
+static int davinci_mcasp_resume(struct snd_soc_dai *dai)
+{
+ struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(dai);
+ struct davinci_mcasp_context *context = &mcasp->context;
+
+ mcasp_set_reg(mcasp, DAVINCI_MCASP_TXFMCTL_REG, context->txfmtctl);
+ mcasp_set_reg(mcasp, DAVINCI_MCASP_RXFMCTL_REG, context->rxfmtctl);
+ mcasp_set_reg(mcasp, DAVINCI_MCASP_TXFMT_REG, context->txfmt);
+ mcasp_set_reg(mcasp, DAVINCI_MCASP_RXFMT_REG, context->rxfmt);
+ mcasp_set_reg(mcasp, DAVINCI_MCASP_ACLKXCTL_REG, context->aclkxctl);
+ mcasp_set_reg(mcasp, DAVINCI_MCASP_ACLKRCTL_REG, context->aclkrctl);
+ mcasp_set_reg(mcasp, DAVINCI_MCASP_PDIR_REG, context->pdir);
+
+ return 0;
+}
+#else
+#define davinci_mcasp_suspend NULL
+#define davinci_mcasp_resume NULL
+#endif
+
#define DAVINCI_MCASP_RATES SNDRV_PCM_RATE_8000_192000
#define DAVINCI_MCASP_PCM_FMTS (SNDRV_PCM_FMTBIT_S8 | \
@@ -735,6 +792,8 @@ static const struct snd_soc_dai_ops davinci_mcasp_dai_ops = {
static struct snd_soc_dai_driver davinci_mcasp_dai[] = {
{
.name = "davinci-mcasp.0",
+ .suspend = davinci_mcasp_suspend,
+ .resume = davinci_mcasp_resume,
.playback = {
.channels_min = 2,
.channels_max = 32 * 16,
@@ -768,28 +827,28 @@ static const struct snd_soc_component_driver davinci_mcasp_component = {
};
/* Some HW specific values and defaults. The rest is filled in from DT. */
-static struct snd_platform_data dm646x_mcasp_pdata = {
+static struct davinci_mcasp_pdata dm646x_mcasp_pdata = {
.tx_dma_offset = 0x400,
.rx_dma_offset = 0x400,
.asp_chan_q = EVENTQ_0,
.version = MCASP_VERSION_1,
};
-static struct snd_platform_data da830_mcasp_pdata = {
+static struct davinci_mcasp_pdata da830_mcasp_pdata = {
.tx_dma_offset = 0x2000,
.rx_dma_offset = 0x2000,
.asp_chan_q = EVENTQ_0,
.version = MCASP_VERSION_2,
};
-static struct snd_platform_data am33xx_mcasp_pdata = {
+static struct davinci_mcasp_pdata am33xx_mcasp_pdata = {
.tx_dma_offset = 0,
.rx_dma_offset = 0,
.asp_chan_q = EVENTQ_0,
.version = MCASP_VERSION_3,
};
-static struct snd_platform_data dra7_mcasp_pdata = {
+static struct davinci_mcasp_pdata dra7_mcasp_pdata = {
.tx_dma_offset = 0x200,
.rx_dma_offset = 0x284,
.asp_chan_q = EVENTQ_0,
@@ -857,11 +916,11 @@ err1:
return ret;
}
-static struct snd_platform_data *davinci_mcasp_set_pdata_from_of(
+static struct davinci_mcasp_pdata *davinci_mcasp_set_pdata_from_of(
struct platform_device *pdev)
{
struct device_node *np = pdev->dev.of_node;
- struct snd_platform_data *pdata = NULL;
+ struct davinci_mcasp_pdata *pdata = NULL;
const struct of_device_id *match =
of_match_device(mcasp_dt_ids, &pdev->dev);
struct of_phandle_args dma_spec;
@@ -874,7 +933,7 @@ static struct snd_platform_data *davinci_mcasp_set_pdata_from_of(
pdata = pdev->dev.platform_data;
return pdata;
} else if (match) {
- pdata = (struct snd_platform_data *) match->data;
+ pdata = (struct davinci_mcasp_pdata*) match->data;
} else {
/* control shouldn't reach here. something is wrong */
ret = -EINVAL;
@@ -966,9 +1025,10 @@ nodata:
static int davinci_mcasp_probe(struct platform_device *pdev)
{
- struct davinci_pcm_dma_params *dma_data;
+ struct davinci_pcm_dma_params *dma_params;
+ struct snd_dmaengine_dai_dma_data *dma_data;
struct resource *mem, *ioarea, *res, *dat;
- struct snd_platform_data *pdata;
+ struct davinci_mcasp_pdata *pdata;
struct davinci_mcasp *mcasp;
int ret;
@@ -1035,41 +1095,49 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
if (dat)
mcasp->dat_port = true;
- dma_data = &mcasp->dma_params[SNDRV_PCM_STREAM_PLAYBACK];
- dma_data->asp_chan_q = pdata->asp_chan_q;
- dma_data->ram_chan_q = pdata->ram_chan_q;
- dma_data->sram_pool = pdata->sram_pool;
- dma_data->sram_size = pdata->sram_size_playback;
+ dma_params = &mcasp->dma_params[SNDRV_PCM_STREAM_PLAYBACK];
+ dma_data = &mcasp->dma_data[SNDRV_PCM_STREAM_PLAYBACK];
+ dma_params->asp_chan_q = pdata->asp_chan_q;
+ dma_params->ram_chan_q = pdata->ram_chan_q;
+ dma_params->sram_pool = pdata->sram_pool;
+ dma_params->sram_size = pdata->sram_size_playback;
if (dat)
- dma_data->dma_addr = dat->start;
+ dma_params->dma_addr = dat->start;
else
- dma_data->dma_addr = mem->start + pdata->tx_dma_offset;
+ dma_params->dma_addr = mem->start + pdata->tx_dma_offset;
/* Unconditional dmaengine stuff */
- mcasp->dma_data[SNDRV_PCM_STREAM_PLAYBACK].addr = dma_data->dma_addr;
+ dma_data->addr = dma_params->dma_addr;
res = platform_get_resource(pdev, IORESOURCE_DMA, 0);
if (res)
- dma_data->channel = res->start;
+ dma_params->channel = res->start;
else
- dma_data->channel = pdata->tx_dma_channel;
+ dma_params->channel = pdata->tx_dma_channel;
- dma_data = &mcasp->dma_params[SNDRV_PCM_STREAM_CAPTURE];
- dma_data->asp_chan_q = pdata->asp_chan_q;
- dma_data->ram_chan_q = pdata->ram_chan_q;
- dma_data->sram_pool = pdata->sram_pool;
- dma_data->sram_size = pdata->sram_size_capture;
+ /* dmaengine filter data for DT and non-DT boot */
+ if (pdev->dev.of_node)
+ dma_data->filter_data = "tx";
+ else
+ dma_data->filter_data = &dma_params->channel;
+
+ dma_params = &mcasp->dma_params[SNDRV_PCM_STREAM_CAPTURE];
+ dma_data = &mcasp->dma_data[SNDRV_PCM_STREAM_CAPTURE];
+ dma_params->asp_chan_q = pdata->asp_chan_q;
+ dma_params->ram_chan_q = pdata->ram_chan_q;
+ dma_params->sram_pool = pdata->sram_pool;
+ dma_params->sram_size = pdata->sram_size_capture;
if (dat)
- dma_data->dma_addr = dat->start;
+ dma_params->dma_addr = dat->start;
else
- dma_data->dma_addr = mem->start + pdata->rx_dma_offset;
+ dma_params->dma_addr = mem->start + pdata->rx_dma_offset;
/* Unconditional dmaengine stuff */
- mcasp->dma_data[SNDRV_PCM_STREAM_CAPTURE].addr = dma_data->dma_addr;
+ dma_data->addr = dma_params->dma_addr;
if (mcasp->version < MCASP_VERSION_3) {
mcasp->fifo_base = DAVINCI_MCASP_V2_AFIFO_BASE;
- /* dma_data->dma_addr is pointing to the data port address */
+ /* dma_params->dma_addr is pointing to the data port address */
mcasp->dat_port = true;
} else {
mcasp->fifo_base = DAVINCI_MCASP_V3_AFIFO_BASE;
@@ -1077,13 +1145,15 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
res = platform_get_resource(pdev, IORESOURCE_DMA, 1);
if (res)
- dma_data->channel = res->start;
+ dma_params->channel = res->start;
else
- dma_data->channel = pdata->rx_dma_channel;
+ dma_params->channel = pdata->rx_dma_channel;
- /* Unconditional dmaengine stuff */
- mcasp->dma_data[SNDRV_PCM_STREAM_PLAYBACK].filter_data = "tx";
- mcasp->dma_data[SNDRV_PCM_STREAM_CAPTURE].filter_data = "rx";
+ /* dmaengine filter data for DT and non-DT boot */
+ if (pdev->dev.of_node)
+ dma_data->filter_data = "rx";
+ else
+ dma_data->filter_data = &dma_params->channel;
dev_set_drvdata(&pdev->dev, mcasp);
@@ -1127,49 +1197,12 @@ static int davinci_mcasp_remove(struct platform_device *pdev)
return 0;
}
-#ifdef CONFIG_PM_SLEEP
-static int davinci_mcasp_suspend(struct device *dev)
-{
- struct davinci_mcasp *mcasp = dev_get_drvdata(dev);
-
- mcasp->context.txfmtctl = mcasp_get_reg(mcasp, DAVINCI_MCASP_TXFMCTL_REG);
- mcasp->context.rxfmtctl = mcasp_get_reg(mcasp, DAVINCI_MCASP_RXFMCTL_REG);
- mcasp->context.txfmt = mcasp_get_reg(mcasp, DAVINCI_MCASP_TXFMT_REG);
- mcasp->context.rxfmt = mcasp_get_reg(mcasp, DAVINCI_MCASP_RXFMT_REG);
- mcasp->context.aclkxctl = mcasp_get_reg(mcasp, DAVINCI_MCASP_ACLKXCTL_REG);
- mcasp->context.aclkrctl = mcasp_get_reg(mcasp, DAVINCI_MCASP_ACLKRCTL_REG);
- mcasp->context.pdir = mcasp_get_reg(mcasp, DAVINCI_MCASP_PDIR_REG);
-
- return 0;
-}
-
-static int davinci_mcasp_resume(struct device *dev)
-{
- struct davinci_mcasp *mcasp = dev_get_drvdata(dev);
-
- mcasp_set_reg(mcasp, DAVINCI_MCASP_TXFMCTL_REG, mcasp->context.txfmtctl);
- mcasp_set_reg(mcasp, DAVINCI_MCASP_RXFMCTL_REG, mcasp->context.rxfmtctl);
- mcasp_set_reg(mcasp, DAVINCI_MCASP_TXFMT_REG, mcasp->context.txfmt);
- mcasp_set_reg(mcasp, DAVINCI_MCASP_RXFMT_REG, mcasp->context.rxfmt);
- mcasp_set_reg(mcasp, DAVINCI_MCASP_ACLKXCTL_REG, mcasp->context.aclkxctl);
- mcasp_set_reg(mcasp, DAVINCI_MCASP_ACLKRCTL_REG, mcasp->context.aclkrctl);
- mcasp_set_reg(mcasp, DAVINCI_MCASP_PDIR_REG, mcasp->context.pdir);
-
- return 0;
-}
-#endif
-
-SIMPLE_DEV_PM_OPS(davinci_mcasp_pm_ops,
- davinci_mcasp_suspend,
- davinci_mcasp_resume);
-
static struct platform_driver davinci_mcasp_driver = {
.probe = davinci_mcasp_probe,
.remove = davinci_mcasp_remove,
.driver = {
.name = "davinci-mcasp",
.owner = THIS_MODULE,
- .pm = &davinci_mcasp_pm_ops,
.of_match_table = mcasp_dt_ids,
},
};
diff --git a/sound/soc/davinci/edma-pcm.c b/sound/soc/davinci/edma-pcm.c
new file mode 100644
index 0000000..d38afb1
--- /dev/null
+++ b/sound/soc/davinci/edma-pcm.c
@@ -0,0 +1,57 @@
+/*
+ * edma-pcm.c - eDMA PCM driver using dmaengine for AM3xxx, AM4xxx
+ *
+ * Copyright (C) 2014 Texas Instruments, Inc.
+ *
+ * Author: Peter Ujfalusi <peter.ujfalusi@ti.com>
+ *
+ * Based on: sound/soc/tegra/tegra_pcm.c
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * version 2 as published by the Free Software Foundation.
+ *
+ * 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.
+ */
+
+#include <linux/module.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/dmaengine_pcm.h>
+#include <linux/edma.h>
+
+static const struct snd_pcm_hardware edma_pcm_hardware = {
+ .info = SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_BATCH |
+ SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME |
+ SNDRV_PCM_INFO_INTERLEAVED,
+ .buffer_bytes_max = 128 * 1024,
+ .period_bytes_min = 32,
+ .period_bytes_max = 64 * 1024,
+ .periods_min = 2,
+ .periods_max = 19, /* Limit by edma dmaengine driver */
+};
+
+static const struct snd_dmaengine_pcm_config edma_dmaengine_pcm_config = {
+ .pcm_hardware = &edma_pcm_hardware,
+ .prepare_slave_config = snd_dmaengine_pcm_prepare_slave_config,
+ .compat_filter_fn = edma_filter_fn,
+ .prealloc_buffer_size = 128 * 1024,
+};
+
+int edma_pcm_platform_register(struct device *dev)
+{
+ return devm_snd_dmaengine_pcm_register(dev, &edma_dmaengine_pcm_config,
+ SND_DMAENGINE_PCM_FLAG_COMPAT);
+}
+EXPORT_SYMBOL_GPL(edma_pcm_platform_register);
+
+MODULE_AUTHOR("Peter Ujfalusi <peter.ujfalusi@ti.com>");
+MODULE_DESCRIPTION("eDMA PCM ASoC platform driver");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/davinci/edma-pcm.h b/sound/soc/davinci/edma-pcm.h
new file mode 100644
index 0000000..894c378
--- /dev/null
+++ b/sound/soc/davinci/edma-pcm.h
@@ -0,0 +1,25 @@
+/*
+ * edma-pcm.h - eDMA PCM driver using dmaengine for AM3xxx, AM4xxx
+ *
+ * Copyright (C) 2014 Texas Instruments, Inc.
+ *
+ * Author: Peter Ujfalusi <peter.ujfalusi@ti.com>
+ *
+ * Based on: sound/soc/tegra/tegra_pcm.h
+ *
+ * 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.
+ */
+
+#ifndef __EDMA_PCM_H__
+#define __EDMA_PCM_H__
+
+int edma_pcm_platform_register(struct device *dev);
+
+#endif /* __EDMA_PCM_H__ */
diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig
index 07f8f14..597962e 100644
--- a/sound/soc/fsl/Kconfig
+++ b/sound/soc/fsl/Kconfig
@@ -1,5 +1,6 @@
config SND_SOC_FSL_SAI
tristate
+ select REGMAP_MMIO
select SND_SOC_GENERIC_DMAENGINE_PCM
config SND_SOC_FSL_SSI
@@ -7,9 +8,11 @@ config SND_SOC_FSL_SSI
config SND_SOC_FSL_SPDIF
tristate
+ select REGMAP_MMIO
config SND_SOC_FSL_ESAI
tristate
+ select REGMAP_MMIO
config SND_SOC_FSL_UTILS
tristate
@@ -168,12 +171,14 @@ config SND_SOC_EUKREA_TLV320
depends on MACH_EUKREA_MBIMX27_BASEBOARD \
|| MACH_EUKREA_MBIMXSD25_BASEBOARD \
|| MACH_EUKREA_MBIMXSD35_BASEBOARD \
- || MACH_EUKREA_MBIMXSD51_BASEBOARD
+ || MACH_EUKREA_MBIMXSD51_BASEBOARD \
+ || (OF && ARM)
depends on I2C
- select SND_SOC_TLV320AIC23
- select SND_SOC_IMX_PCM_FIQ
+ select SND_SOC_TLV320AIC23_I2C
select SND_SOC_IMX_AUDMUX
select SND_SOC_IMX_SSI
+ select SND_SOC_FSL_SSI
+ select SND_SOC_IMX_PCM_DMA
help
Enable I2S based access to the TLV320AIC23B codec attached
to the SSI interface
@@ -204,7 +209,6 @@ config SND_SOC_IMX_SPDIF
tristate "SoC Audio support for i.MX boards with S/PDIF"
select SND_SOC_IMX_PCM_DMA
select SND_SOC_FSL_SPDIF
- select REGMAP_MMIO
help
SoC Audio support for i.MX boards with S/PDIF
Say Y if you want to add support for SoC audio on an i.MX board with
diff --git a/sound/soc/fsl/eukrea-tlv320.c b/sound/soc/fsl/eukrea-tlv320.c
index 5983740..eb093d5 100644
--- a/sound/soc/fsl/eukrea-tlv320.c
+++ b/sound/soc/fsl/eukrea-tlv320.c
@@ -15,8 +15,11 @@
*
*/
+#include <linux/errno.h>
#include <linux/module.h>
#include <linux/moduleparam.h>
+#include <linux/of.h>
+#include <linux/of_platform.h>
#include <linux/device.h>
#include <linux/i2c.h>
#include <sound/core.h>
@@ -26,6 +29,7 @@
#include "../codecs/tlv320aic23.h"
#include "imx-ssi.h"
+#include "fsl_ssi.h"
#include "imx-audmux.h"
#define CODEC_CLOCK 12000000
@@ -41,7 +45,8 @@ static int eukrea_tlv320_hw_params(struct snd_pcm_substream *substream,
ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
SND_SOC_DAIFMT_NB_NF |
SND_SOC_DAIFMT_CBM_CFM);
- if (ret) {
+ /* fsl_ssi lacks the set_fmt ops. */
+ if (ret && ret != -ENOTSUPP) {
dev_err(cpu_dai->dev,
"Failed to set the cpu dai format.\n");
return ret;
@@ -63,11 +68,13 @@ static int eukrea_tlv320_hw_params(struct snd_pcm_substream *substream,
"Failed to set the codec sysclk.\n");
return ret;
}
+
snd_soc_dai_set_tdm_slot(cpu_dai, 0xffffffc, 0xffffffc, 2, 0);
ret = snd_soc_dai_set_sysclk(cpu_dai, IMX_SSP_SYS_CLK, 0,
SND_SOC_CLOCK_IN);
- if (ret) {
+ /* fsl_ssi lacks the set_sysclk ops */
+ if (ret && ret != -EINVAL) {
dev_err(cpu_dai->dev,
"Can't set the IMX_SSP_SYS_CLK CPU system clock.\n");
return ret;
@@ -84,14 +91,10 @@ static struct snd_soc_dai_link eukrea_tlv320_dai = {
.name = "tlv320aic23",
.stream_name = "TLV320AIC23",
.codec_dai_name = "tlv320aic23-hifi",
- .platform_name = "imx-ssi.0",
- .codec_name = "tlv320aic23-codec.0-001a",
- .cpu_dai_name = "imx-ssi.0",
.ops = &eukrea_tlv320_snd_ops,
};
static struct snd_soc_card eukrea_tlv320 = {
- .name = "cpuimx-audio",
.owner = THIS_MODULE,
.dai_link = &eukrea_tlv320_dai,
.num_links = 1,
@@ -101,8 +104,65 @@ static int eukrea_tlv320_probe(struct platform_device *pdev)
{
int ret;
int int_port = 0, ext_port;
+ struct device_node *np = pdev->dev.of_node;
+ struct device_node *ssi_np, *codec_np;
- if (machine_is_eukrea_cpuimx27()) {
+ eukrea_tlv320.dev = &pdev->dev;
+ if (np) {
+ ret = snd_soc_of_parse_card_name(&eukrea_tlv320,
+ "eukrea,model");
+ if (ret) {
+ dev_err(&pdev->dev,
+ "eukrea,model node missing or invalid.\n");
+ goto err;
+ }
+
+ ssi_np = of_parse_phandle(pdev->dev.of_node,
+ "ssi-controller", 0);
+ if (!ssi_np) {
+ dev_err(&pdev->dev,
+ "ssi-controller missing or invalid.\n");
+ ret = -ENODEV;
+ goto err;
+ }
+
+ codec_np = of_parse_phandle(ssi_np, "codec-handle", 0);
+ if (codec_np)
+ eukrea_tlv320_dai.codec_of_node = codec_np;
+ else
+ dev_err(&pdev->dev, "codec-handle node missing or invalid.\n");
+
+ ret = of_property_read_u32(np, "fsl,mux-int-port", &int_port);
+ if (ret) {
+ dev_err(&pdev->dev,
+ "fsl,mux-int-port node missing or invalid.\n");
+ return ret;
+ }
+ ret = of_property_read_u32(np, "fsl,mux-ext-port", &ext_port);
+ if (ret) {
+ dev_err(&pdev->dev,
+ "fsl,mux-ext-port node 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--;
+
+ eukrea_tlv320_dai.cpu_of_node = ssi_np;
+ eukrea_tlv320_dai.platform_of_node = ssi_np;
+ } else {
+ eukrea_tlv320_dai.cpu_dai_name = "imx-ssi.0";
+ eukrea_tlv320_dai.platform_name = "imx-ssi.0";
+ eukrea_tlv320_dai.codec_name = "tlv320aic23-codec.0-001a";
+ eukrea_tlv320.name = "cpuimx-audio";
+ }
+
+ if (machine_is_eukrea_cpuimx27() ||
+ of_find_compatible_node(NULL, NULL, "fsl,imx21-audmux")) {
imx_audmux_v1_configure_port(MX27_AUDMUX_HPCR1_SSI0,
IMX_AUDMUX_V1_PCR_SYN |
IMX_AUDMUX_V1_PCR_TFSDIR |
@@ -119,8 +179,12 @@ static int eukrea_tlv320_probe(struct platform_device *pdev)
);
} else if (machine_is_eukrea_cpuimx25sd() ||
machine_is_eukrea_cpuimx35sd() ||
- machine_is_eukrea_cpuimx51sd()) {
- ext_port = machine_is_eukrea_cpuimx25sd() ? 4 : 3;
+ machine_is_eukrea_cpuimx51sd() ||
+ of_find_compatible_node(NULL, NULL, "fsl,imx31-audmux")) {
+ if (!np)
+ ext_port = machine_is_eukrea_cpuimx25sd() ?
+ 4 : 3;
+
imx_audmux_v2_configure_port(int_port,
IMX_AUDMUX_V2_PTCR_SYN |
IMX_AUDMUX_V2_PTCR_TFSDIR |
@@ -134,14 +198,27 @@ static int eukrea_tlv320_probe(struct platform_device *pdev)
IMX_AUDMUX_V2_PDCR_RXDSEL(int_port)
);
} else {
- /* return happy. We might run on a totally different machine */
- return 0;
+ if (np) {
+ /* The eukrea,asoc-tlv320 driver was explicitely
+ * requested (through the device tree).
+ */
+ dev_err(&pdev->dev,
+ "Missing or invalid audmux DT node.\n");
+ return -ENODEV;
+ } else {
+ /* Return happy.
+ * We might run on a totally different machine.
+ */
+ return 0;
+ }
}
- eukrea_tlv320.dev = &pdev->dev;
ret = snd_soc_register_card(&eukrea_tlv320);
+err:
if (ret)
dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret);
+ if (np)
+ of_node_put(ssi_np);
return ret;
}
@@ -153,10 +230,17 @@ static int eukrea_tlv320_remove(struct platform_device *pdev)
return 0;
}
+static const struct of_device_id imx_tlv320_dt_ids[] = {
+ { .compatible = "eukrea,asoc-tlv320"},
+ { /* sentinel */ }
+};
+MODULE_DEVICE_TABLE(of, imx_tlv320_dt_ids);
+
static struct platform_driver eukrea_tlv320_driver = {
.driver = {
.name = "eukrea_tlv320",
.owner = THIS_MODULE,
+ .of_match_table = imx_tlv320_dt_ids,
},
.probe = eukrea_tlv320_probe,
.remove = eukrea_tlv320_remove,
diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c
index c84026c..0ba3700 100644
--- a/sound/soc/fsl/fsl_esai.c
+++ b/sound/soc/fsl/fsl_esai.c
@@ -431,17 +431,26 @@ static int fsl_esai_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt)
static int fsl_esai_startup(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
+ int ret;
struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai);
/*
* Some platforms might use the same bit to gate all three or two of
* clocks, so keep all clocks open/close at the same time for safety
*/
- clk_prepare_enable(esai_priv->coreclk);
- if (!IS_ERR(esai_priv->extalclk))
- clk_prepare_enable(esai_priv->extalclk);
- if (!IS_ERR(esai_priv->fsysclk))
- clk_prepare_enable(esai_priv->fsysclk);
+ ret = clk_prepare_enable(esai_priv->coreclk);
+ if (ret)
+ return ret;
+ if (!IS_ERR(esai_priv->extalclk)) {
+ ret = clk_prepare_enable(esai_priv->extalclk);
+ if (ret)
+ goto err_extalck;
+ }
+ if (!IS_ERR(esai_priv->fsysclk)) {
+ ret = clk_prepare_enable(esai_priv->fsysclk);
+ if (ret)
+ goto err_fsysclk;
+ }
if (!dai->active) {
/* Reset Port C */
@@ -463,6 +472,14 @@ static int fsl_esai_startup(struct snd_pcm_substream *substream,
}
return 0;
+
+err_fsysclk:
+ if (!IS_ERR(esai_priv->extalclk))
+ clk_disable_unprepare(esai_priv->extalclk);
+err_extalck:
+ clk_disable_unprepare(esai_priv->coreclk);
+
+ return ret;
}
static int fsl_esai_hw_params(struct snd_pcm_substream *substream,
@@ -661,7 +678,7 @@ static bool fsl_esai_writeable_reg(struct device *dev, unsigned int reg)
}
}
-static const struct regmap_config fsl_esai_regmap_config = {
+static struct regmap_config fsl_esai_regmap_config = {
.reg_bits = 32,
.reg_stride = 4,
.val_bits = 32,
@@ -687,6 +704,9 @@ static int fsl_esai_probe(struct platform_device *pdev)
esai_priv->pdev = pdev;
strcpy(esai_priv->name, np->name);
+ if (of_property_read_bool(np, "big-endian"))
+ fsl_esai_regmap_config.val_format_endian = REGMAP_ENDIAN_BIG;
+
/* Get the addresses and IRQ */
res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
regs = devm_ioremap_resource(&pdev->dev, res);
diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c
index cdd3fa8..c4a4231 100644
--- a/sound/soc/fsl/fsl_sai.c
+++ b/sound/soc/fsl/fsl_sai.c
@@ -15,6 +15,7 @@
#include <linux/dmaengine.h>
#include <linux/module.h>
#include <linux/of_address.h>
+#include <linux/regmap.h>
#include <linux/slab.h>
#include <sound/core.h>
#include <sound/dmaengine_pcm.h>
@@ -22,34 +23,6 @@
#include "fsl_sai.h"
-static inline u32 sai_readl(struct fsl_sai *sai,
- const void __iomem *addr)
-{
- u32 val;
-
- val = __raw_readl(addr);
-
- if (likely(sai->big_endian_regs))
- val = be32_to_cpu(val);
- else
- val = le32_to_cpu(val);
- rmb();
-
- return val;
-}
-
-static inline void sai_writel(struct fsl_sai *sai,
- u32 val, void __iomem *addr)
-{
- wmb();
- if (likely(sai->big_endian_regs))
- val = cpu_to_be32(val);
- else
- val = cpu_to_le32(val);
-
- __raw_writel(val, addr);
-}
-
static int fsl_sai_set_dai_sysclk_tr(struct snd_soc_dai *cpu_dai,
int clk_id, unsigned int freq, int fsl_dir)
{
@@ -61,7 +34,8 @@ static int fsl_sai_set_dai_sysclk_tr(struct snd_soc_dai *cpu_dai,
else
reg_cr2 = FSL_SAI_RCR2;
- val_cr2 = sai_readl(sai, sai->base + reg_cr2);
+ regmap_read(sai->regmap, reg_cr2, &val_cr2);
+
val_cr2 &= ~FSL_SAI_CR2_MSEL_MASK;
switch (clk_id) {
@@ -81,7 +55,7 @@ static int fsl_sai_set_dai_sysclk_tr(struct snd_soc_dai *cpu_dai,
return -EINVAL;
}
- sai_writel(sai, val_cr2, sai->base + reg_cr2);
+ regmap_write(sai->regmap, reg_cr2, val_cr2);
return 0;
}
@@ -89,32 +63,22 @@ static int fsl_sai_set_dai_sysclk_tr(struct snd_soc_dai *cpu_dai,
static int fsl_sai_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
int clk_id, unsigned int freq, int dir)
{
- struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai);
int ret;
if (dir == SND_SOC_CLOCK_IN)
return 0;
- ret = clk_prepare_enable(sai->clk);
- if (ret)
- return ret;
-
ret = fsl_sai_set_dai_sysclk_tr(cpu_dai, clk_id, freq,
FSL_FMT_TRANSMITTER);
if (ret) {
dev_err(cpu_dai->dev, "Cannot set tx sysclk: %d\n", ret);
- goto err_clk;
+ return ret;
}
ret = fsl_sai_set_dai_sysclk_tr(cpu_dai, clk_id, freq,
FSL_FMT_RECEIVER);
- if (ret) {
+ if (ret)
dev_err(cpu_dai->dev, "Cannot set rx sysclk: %d\n", ret);
- goto err_clk;
- }
-
-err_clk:
- clk_disable_unprepare(sai->clk);
return ret;
}
@@ -133,43 +97,84 @@ static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai,
reg_cr4 = FSL_SAI_RCR4;
}
- val_cr2 = sai_readl(sai, sai->base + reg_cr2);
- val_cr4 = sai_readl(sai, sai->base + reg_cr4);
+ regmap_read(sai->regmap, reg_cr2, &val_cr2);
+ regmap_read(sai->regmap, reg_cr4, &val_cr4);
if (sai->big_endian_data)
val_cr4 &= ~FSL_SAI_CR4_MF;
else
val_cr4 |= FSL_SAI_CR4_MF;
+ /* DAI mode */
switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
case SND_SOC_DAIFMT_I2S:
+ /*
+ * Frame low, 1clk before data, one word length for frame sync,
+ * frame sync starts one serial clock cycle earlier,
+ * that is, together with the last bit of the previous
+ * data word.
+ */
+ val_cr2 &= ~FSL_SAI_CR2_BCP;
+ val_cr4 |= FSL_SAI_CR4_FSE | FSL_SAI_CR4_FSP;
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ /*
+ * Frame high, one word length for frame sync,
+ * frame sync asserts with the first bit of the frame.
+ */
+ val_cr2 &= ~FSL_SAI_CR2_BCP;
+ val_cr4 &= ~(FSL_SAI_CR4_FSE | FSL_SAI_CR4_FSP);
+ break;
+ case SND_SOC_DAIFMT_DSP_A:
+ /*
+ * Frame high, 1clk before data, one bit for frame sync,
+ * frame sync starts one serial clock cycle earlier,
+ * that is, together with the last bit of the previous
+ * data word.
+ */
+ val_cr2 &= ~FSL_SAI_CR2_BCP;
+ val_cr4 &= ~FSL_SAI_CR4_FSP;
val_cr4 |= FSL_SAI_CR4_FSE;
+ sai->is_dsp_mode = true;
+ break;
+ case SND_SOC_DAIFMT_DSP_B:
+ /*
+ * Frame high, one bit for frame sync,
+ * frame sync asserts with the first bit of the frame.
+ */
+ val_cr2 &= ~FSL_SAI_CR2_BCP;
+ val_cr4 &= ~(FSL_SAI_CR4_FSE | FSL_SAI_CR4_FSP);
+ sai->is_dsp_mode = true;
break;
+ case SND_SOC_DAIFMT_RIGHT_J:
+ /* To be done */
default:
return -EINVAL;
}
+ /* DAI clock inversion */
switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
case SND_SOC_DAIFMT_IB_IF:
- val_cr4 |= FSL_SAI_CR4_FSP;
- val_cr2 &= ~FSL_SAI_CR2_BCP;
+ /* Invert both clocks */
+ val_cr2 ^= FSL_SAI_CR2_BCP;
+ val_cr4 ^= FSL_SAI_CR4_FSP;
break;
case SND_SOC_DAIFMT_IB_NF:
- val_cr4 &= ~FSL_SAI_CR4_FSP;
- val_cr2 &= ~FSL_SAI_CR2_BCP;
+ /* Invert bit clock */
+ val_cr2 ^= FSL_SAI_CR2_BCP;
break;
case SND_SOC_DAIFMT_NB_IF:
- val_cr4 |= FSL_SAI_CR4_FSP;
- val_cr2 |= FSL_SAI_CR2_BCP;
+ /* Invert frame clock */
+ val_cr4 ^= FSL_SAI_CR4_FSP;
break;
case SND_SOC_DAIFMT_NB_NF:
- val_cr4 &= ~FSL_SAI_CR4_FSP;
- val_cr2 |= FSL_SAI_CR2_BCP;
+ /* Nothing to do for both normal cases */
break;
default:
return -EINVAL;
}
+ /* DAI clock master masks */
switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
case SND_SOC_DAIFMT_CBS_CFS:
val_cr2 |= FSL_SAI_CR2_BCD_MSTR;
@@ -179,39 +184,37 @@ static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai,
val_cr2 &= ~FSL_SAI_CR2_BCD_MSTR;
val_cr4 &= ~FSL_SAI_CR4_FSD_MSTR;
break;
+ case SND_SOC_DAIFMT_CBS_CFM:
+ val_cr2 |= FSL_SAI_CR2_BCD_MSTR;
+ val_cr4 &= ~FSL_SAI_CR4_FSD_MSTR;
+ break;
+ case SND_SOC_DAIFMT_CBM_CFS:
+ val_cr2 &= ~FSL_SAI_CR2_BCD_MSTR;
+ val_cr4 |= FSL_SAI_CR4_FSD_MSTR;
+ break;
default:
return -EINVAL;
}
- sai_writel(sai, val_cr2, sai->base + reg_cr2);
- sai_writel(sai, val_cr4, sai->base + reg_cr4);
+ regmap_write(sai->regmap, reg_cr2, val_cr2);
+ regmap_write(sai->regmap, reg_cr4, val_cr4);
return 0;
}
static int fsl_sai_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt)
{
- struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai);
int ret;
- ret = clk_prepare_enable(sai->clk);
- if (ret)
- return ret;
-
ret = fsl_sai_set_dai_fmt_tr(cpu_dai, fmt, FSL_FMT_TRANSMITTER);
if (ret) {
dev_err(cpu_dai->dev, "Cannot set tx format: %d\n", ret);
- goto err_clk;
+ return ret;
}
ret = fsl_sai_set_dai_fmt_tr(cpu_dai, fmt, FSL_FMT_RECEIVER);
- if (ret) {
+ if (ret)
dev_err(cpu_dai->dev, "Cannot set rx format: %d\n", ret);
- goto err_clk;
- }
-
-err_clk:
- clk_disable_unprepare(sai->clk);
return ret;
}
@@ -235,16 +238,19 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream,
reg_mr = FSL_SAI_RMR;
}
- val_cr4 = sai_readl(sai, sai->base + reg_cr4);
+ regmap_read(sai->regmap, reg_cr4, &val_cr4);
+ regmap_read(sai->regmap, reg_cr4, &val_cr5);
+
val_cr4 &= ~FSL_SAI_CR4_SYWD_MASK;
val_cr4 &= ~FSL_SAI_CR4_FRSZ_MASK;
- val_cr5 = sai_readl(sai, sai->base + reg_cr5);
val_cr5 &= ~FSL_SAI_CR5_WNW_MASK;
val_cr5 &= ~FSL_SAI_CR5_W0W_MASK;
val_cr5 &= ~FSL_SAI_CR5_FBT_MASK;
- val_cr4 |= FSL_SAI_CR4_SYWD(word_width);
+ if (!sai->is_dsp_mode)
+ val_cr4 |= FSL_SAI_CR4_SYWD(word_width);
+
val_cr5 |= FSL_SAI_CR5_WNW(word_width);
val_cr5 |= FSL_SAI_CR5_W0W(word_width);
@@ -257,9 +263,9 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream,
val_cr4 |= FSL_SAI_CR4_FRSZ(channels);
val_mr = ~0UL - ((1 << channels) - 1);
- sai_writel(sai, val_cr4, sai->base + reg_cr4);
- sai_writel(sai, val_cr5, sai->base + reg_cr5);
- sai_writel(sai, val_mr, sai->base + reg_mr);
+ regmap_write(sai->regmap, reg_cr4, val_cr4);
+ regmap_write(sai->regmap, reg_cr5, val_cr5);
+ regmap_write(sai->regmap, reg_mr, val_mr);
return 0;
}
@@ -268,44 +274,42 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd,
struct snd_soc_dai *cpu_dai)
{
struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai);
- u32 tcsr, rcsr, val_cr2, val_cr3, reg_cr3;
-
- val_cr2 = sai_readl(sai, sai->base + FSL_SAI_TCR2);
- val_cr2 &= ~FSL_SAI_CR2_SYNC;
- sai_writel(sai, val_cr2, sai->base + FSL_SAI_TCR2);
+ u32 tcsr, rcsr;
- val_cr2 = sai_readl(sai, sai->base + FSL_SAI_RCR2);
- val_cr2 |= FSL_SAI_CR2_SYNC;
- sai_writel(sai, val_cr2, sai->base + FSL_SAI_RCR2);
+ /*
+ * The transmitter bit clock and frame sync are to be
+ * used by both the transmitter and receiver.
+ */
+ regmap_update_bits(sai->regmap, FSL_SAI_TCR2, FSL_SAI_CR2_SYNC,
+ ~FSL_SAI_CR2_SYNC);
+ regmap_update_bits(sai->regmap, FSL_SAI_RCR2, FSL_SAI_CR2_SYNC,
+ FSL_SAI_CR2_SYNC);
- tcsr = sai_readl(sai, sai->base + FSL_SAI_TCSR);
- rcsr = sai_readl(sai, sai->base + FSL_SAI_RCSR);
+ regmap_read(sai->regmap, FSL_SAI_TCSR, &tcsr);
+ regmap_read(sai->regmap, FSL_SAI_RCSR, &rcsr);
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
tcsr |= FSL_SAI_CSR_FRDE;
rcsr &= ~FSL_SAI_CSR_FRDE;
- reg_cr3 = FSL_SAI_TCR3;
} else {
rcsr |= FSL_SAI_CSR_FRDE;
tcsr &= ~FSL_SAI_CSR_FRDE;
- reg_cr3 = FSL_SAI_RCR3;
}
- val_cr3 = sai_readl(sai, sai->base + reg_cr3);
-
+ /*
+ * It is recommended that the transmitter is the last enabled
+ * and the first disabled.
+ */
switch (cmd) {
case SNDRV_PCM_TRIGGER_START:
case SNDRV_PCM_TRIGGER_RESUME:
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
tcsr |= FSL_SAI_CSR_TERE;
rcsr |= FSL_SAI_CSR_TERE;
- val_cr3 |= FSL_SAI_CR3_TRCE;
- sai_writel(sai, val_cr3, sai->base + reg_cr3);
- sai_writel(sai, rcsr, sai->base + FSL_SAI_RCSR);
- sai_writel(sai, tcsr, sai->base + FSL_SAI_TCSR);
+ regmap_write(sai->regmap, FSL_SAI_RCSR, rcsr);
+ regmap_write(sai->regmap, FSL_SAI_TCSR, tcsr);
break;
-
case SNDRV_PCM_TRIGGER_STOP:
case SNDRV_PCM_TRIGGER_SUSPEND:
case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
@@ -314,11 +318,8 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd,
rcsr &= ~FSL_SAI_CSR_TERE;
}
- val_cr3 &= ~FSL_SAI_CR3_TRCE;
-
- sai_writel(sai, tcsr, sai->base + FSL_SAI_TCSR);
- sai_writel(sai, rcsr, sai->base + FSL_SAI_RCSR);
- sai_writel(sai, val_cr3, sai->base + reg_cr3);
+ regmap_write(sai->regmap, FSL_SAI_TCSR, tcsr);
+ regmap_write(sai->regmap, FSL_SAI_RCSR, rcsr);
break;
default:
return -EINVAL;
@@ -331,16 +332,32 @@ static int fsl_sai_startup(struct snd_pcm_substream *substream,
struct snd_soc_dai *cpu_dai)
{
struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai);
+ u32 reg;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ reg = FSL_SAI_TCR3;
+ else
+ reg = FSL_SAI_RCR3;
+
+ regmap_update_bits(sai->regmap, reg, FSL_SAI_CR3_TRCE,
+ FSL_SAI_CR3_TRCE);
- return clk_prepare_enable(sai->clk);
+ return 0;
}
static void fsl_sai_shutdown(struct snd_pcm_substream *substream,
struct snd_soc_dai *cpu_dai)
{
struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai);
+ u32 reg;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ reg = FSL_SAI_TCR3;
+ else
+ reg = FSL_SAI_RCR3;
- clk_disable_unprepare(sai->clk);
+ regmap_update_bits(sai->regmap, reg, FSL_SAI_CR3_TRCE,
+ ~FSL_SAI_CR3_TRCE);
}
static const struct snd_soc_dai_ops fsl_sai_pcm_dai_ops = {
@@ -355,18 +372,13 @@ static const struct snd_soc_dai_ops fsl_sai_pcm_dai_ops = {
static int fsl_sai_dai_probe(struct snd_soc_dai *cpu_dai)
{
struct fsl_sai *sai = dev_get_drvdata(cpu_dai->dev);
- int ret;
- ret = clk_prepare_enable(sai->clk);
- if (ret)
- return ret;
-
- sai_writel(sai, 0x0, sai->base + FSL_SAI_RCSR);
- sai_writel(sai, 0x0, sai->base + FSL_SAI_TCSR);
- sai_writel(sai, FSL_SAI_MAXBURST_TX * 2, sai->base + FSL_SAI_TCR1);
- sai_writel(sai, FSL_SAI_MAXBURST_RX - 1, sai->base + FSL_SAI_RCR1);
-
- clk_disable_unprepare(sai->clk);
+ regmap_update_bits(sai->regmap, FSL_SAI_TCSR, 0xffffffff, 0x0);
+ regmap_update_bits(sai->regmap, FSL_SAI_RCSR, 0xffffffff, 0x0);
+ regmap_update_bits(sai->regmap, FSL_SAI_TCR1, FSL_SAI_CR1_RFW_MASK,
+ FSL_SAI_MAXBURST_TX * 2);
+ regmap_update_bits(sai->regmap, FSL_SAI_RCR1, FSL_SAI_CR1_RFW_MASK,
+ FSL_SAI_MAXBURST_RX - 1);
snd_soc_dai_init_dma_data(cpu_dai, &sai->dma_params_tx,
&sai->dma_params_rx);
@@ -397,26 +409,109 @@ static const struct snd_soc_component_driver fsl_component = {
.name = "fsl-sai",
};
+static bool fsl_sai_readable_reg(struct device *dev, unsigned int reg)
+{
+ switch (reg) {
+ case FSL_SAI_TCSR:
+ case FSL_SAI_TCR1:
+ case FSL_SAI_TCR2:
+ case FSL_SAI_TCR3:
+ case FSL_SAI_TCR4:
+ case FSL_SAI_TCR5:
+ case FSL_SAI_TFR:
+ case FSL_SAI_TMR:
+ case FSL_SAI_RCSR:
+ case FSL_SAI_RCR1:
+ case FSL_SAI_RCR2:
+ case FSL_SAI_RCR3:
+ case FSL_SAI_RCR4:
+ case FSL_SAI_RCR5:
+ case FSL_SAI_RDR:
+ case FSL_SAI_RFR:
+ case FSL_SAI_RMR:
+ return true;
+ default:
+ return false;
+ }
+}
+
+static bool fsl_sai_volatile_reg(struct device *dev, unsigned int reg)
+{
+ switch (reg) {
+ case FSL_SAI_TFR:
+ case FSL_SAI_RFR:
+ case FSL_SAI_TDR:
+ case FSL_SAI_RDR:
+ return true;
+ default:
+ return false;
+ }
+
+}
+
+static bool fsl_sai_writeable_reg(struct device *dev, unsigned int reg)
+{
+ switch (reg) {
+ case FSL_SAI_TCSR:
+ case FSL_SAI_TCR1:
+ case FSL_SAI_TCR2:
+ case FSL_SAI_TCR3:
+ case FSL_SAI_TCR4:
+ case FSL_SAI_TCR5:
+ case FSL_SAI_TDR:
+ case FSL_SAI_TMR:
+ case FSL_SAI_RCSR:
+ case FSL_SAI_RCR1:
+ case FSL_SAI_RCR2:
+ case FSL_SAI_RCR3:
+ case FSL_SAI_RCR4:
+ case FSL_SAI_RCR5:
+ case FSL_SAI_RMR:
+ return true;
+ default:
+ return false;
+ }
+}
+
+static struct regmap_config fsl_sai_regmap_config = {
+ .reg_bits = 32,
+ .reg_stride = 4,
+ .val_bits = 32,
+
+ .max_register = FSL_SAI_RMR,
+ .readable_reg = fsl_sai_readable_reg,
+ .volatile_reg = fsl_sai_volatile_reg,
+ .writeable_reg = fsl_sai_writeable_reg,
+};
+
static int fsl_sai_probe(struct platform_device *pdev)
{
struct device_node *np = pdev->dev.of_node;
struct fsl_sai *sai;
struct resource *res;
+ void __iomem *base;
int ret;
sai = devm_kzalloc(&pdev->dev, sizeof(*sai), GFP_KERNEL);
if (!sai)
return -ENOMEM;
+ sai->big_endian_regs = of_property_read_bool(np, "big-endian-regs");
+ if (sai->big_endian_regs)
+ fsl_sai_regmap_config.val_format_endian = REGMAP_ENDIAN_BIG;
+
+ sai->big_endian_data = of_property_read_bool(np, "big-endian-data");
+
res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
- sai->base = devm_ioremap_resource(&pdev->dev, res);
- if (IS_ERR(sai->base))
- return PTR_ERR(sai->base);
-
- sai->clk = devm_clk_get(&pdev->dev, "sai");
- if (IS_ERR(sai->clk)) {
- dev_err(&pdev->dev, "Cannot get SAI's clock\n");
- return PTR_ERR(sai->clk);
+ base = devm_ioremap_resource(&pdev->dev, res);
+ if (IS_ERR(base))
+ return PTR_ERR(base);
+
+ sai->regmap = devm_regmap_init_mmio_clk(&pdev->dev,
+ "sai", base, &fsl_sai_regmap_config);
+ if (IS_ERR(sai->regmap)) {
+ dev_err(&pdev->dev, "regmap init failed\n");
+ return PTR_ERR(sai->regmap);
}
sai->dma_params_rx.addr = res->start + FSL_SAI_RDR;
@@ -424,9 +519,6 @@ static int fsl_sai_probe(struct platform_device *pdev)
sai->dma_params_rx.maxburst = FSL_SAI_MAXBURST_RX;
sai->dma_params_tx.maxburst = FSL_SAI_MAXBURST_TX;
- sai->big_endian_regs = of_property_read_bool(np, "big-endian-regs");
- sai->big_endian_data = of_property_read_bool(np, "big-endian-data");
-
platform_set_drvdata(pdev, sai);
ret = devm_snd_soc_register_component(&pdev->dev, &fsl_component,
diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h
index 41bb62e6..e432260 100644
--- a/sound/soc/fsl/fsl_sai.h
+++ b/sound/soc/fsl/fsl_sai.h
@@ -15,31 +15,36 @@
SNDRV_PCM_FMTBIT_S20_3LE |\
SNDRV_PCM_FMTBIT_S24_LE)
+/* SAI Register Map Register */
+#define FSL_SAI_TCSR 0x00 /* SAI Transmit Control */
+#define FSL_SAI_TCR1 0x04 /* SAI Transmit Configuration 1 */
+#define FSL_SAI_TCR2 0x08 /* SAI Transmit Configuration 2 */
+#define FSL_SAI_TCR3 0x0c /* SAI Transmit Configuration 3 */
+#define FSL_SAI_TCR4 0x10 /* SAI Transmit Configuration 4 */
+#define FSL_SAI_TCR5 0x14 /* SAI Transmit Configuration 5 */
+#define FSL_SAI_TDR 0x20 /* SAI Transmit Data */
+#define FSL_SAI_TFR 0x40 /* SAI Transmit FIFO */
+#define FSL_SAI_TMR 0x60 /* SAI Transmit Mask */
+#define FSL_SAI_RCSR 0x80 /* SAI Receive Control */
+#define FSL_SAI_RCR1 0x84 /* SAI Receive Configuration 1 */
+#define FSL_SAI_RCR2 0x88 /* SAI Receive Configuration 2 */
+#define FSL_SAI_RCR3 0x8c /* SAI Receive Configuration 3 */
+#define FSL_SAI_RCR4 0x90 /* SAI Receive Configuration 4 */
+#define FSL_SAI_RCR5 0x94 /* SAI Receive Configuration 5 */
+#define FSL_SAI_RDR 0xa0 /* SAI Receive Data */
+#define FSL_SAI_RFR 0xc0 /* SAI Receive FIFO */
+#define FSL_SAI_RMR 0xe0 /* SAI Receive Mask */
+
/* SAI Transmit/Recieve Control Register */
-#define FSL_SAI_TCSR 0x00
-#define FSL_SAI_RCSR 0x80
#define FSL_SAI_CSR_TERE BIT(31)
#define FSL_SAI_CSR_FWF BIT(17)
#define FSL_SAI_CSR_FRIE BIT(8)
#define FSL_SAI_CSR_FRDE BIT(0)
-/* SAI Transmit Data/FIFO/MASK Register */
-#define FSL_SAI_TDR 0x20
-#define FSL_SAI_TFR 0x40
-#define FSL_SAI_TMR 0x60
-
-/* SAI Recieve Data/FIFO/MASK Register */
-#define FSL_SAI_RDR 0xa0
-#define FSL_SAI_RFR 0xc0
-#define FSL_SAI_RMR 0xe0
-
/* SAI Transmit and Recieve Configuration 1 Register */
-#define FSL_SAI_TCR1 0x04
-#define FSL_SAI_RCR1 0x84
+#define FSL_SAI_CR1_RFW_MASK 0x1f
/* SAI Transmit and Recieve Configuration 2 Register */
-#define FSL_SAI_TCR2 0x08
-#define FSL_SAI_RCR2 0x88
#define FSL_SAI_CR2_SYNC BIT(30)
#define FSL_SAI_CR2_MSEL_MASK (0xff << 26)
#define FSL_SAI_CR2_MSEL_BUS 0
@@ -50,15 +55,11 @@
#define FSL_SAI_CR2_BCD_MSTR BIT(24)
/* SAI Transmit and Recieve Configuration 3 Register */
-#define FSL_SAI_TCR3 0x0c
-#define FSL_SAI_RCR3 0x8c
#define FSL_SAI_CR3_TRCE BIT(16)
#define FSL_SAI_CR3_WDFL(x) (x)
#define FSL_SAI_CR3_WDFL_MASK 0x1f
/* SAI Transmit and Recieve Configuration 4 Register */
-#define FSL_SAI_TCR4 0x10
-#define FSL_SAI_RCR4 0x90
#define FSL_SAI_CR4_FRSZ(x) (((x) - 1) << 16)
#define FSL_SAI_CR4_FRSZ_MASK (0x1f << 16)
#define FSL_SAI_CR4_SYWD(x) (((x) - 1) << 8)
@@ -69,8 +70,6 @@
#define FSL_SAI_CR4_FSD_MSTR BIT(0)
/* SAI Transmit and Recieve Configuration 5 Register */
-#define FSL_SAI_TCR5 0x14
-#define FSL_SAI_RCR5 0x94
#define FSL_SAI_CR5_WNW(x) (((x) - 1) << 24)
#define FSL_SAI_CR5_WNW_MASK (0x1f << 24)
#define FSL_SAI_CR5_W0W(x) (((x) - 1) << 16)
@@ -100,12 +99,11 @@
#define FSL_SAI_MAXBURST_RX 6
struct fsl_sai {
- struct clk *clk;
-
- void __iomem *base;
+ struct regmap *regmap;
bool big_endian_regs;
bool big_endian_data;
+ bool is_dsp_mode;
struct snd_dmaengine_dai_dma_data dma_params_rx;
struct snd_dmaengine_dai_dma_data dma_params_tx;
diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c
index 4d075f1..6452ca8 100644
--- a/sound/soc/fsl/fsl_spdif.c
+++ b/sound/soc/fsl/fsl_spdif.c
@@ -911,8 +911,8 @@ static int fsl_spdif_dai_probe(struct snd_soc_dai *dai)
{
struct fsl_spdif_priv *spdif_private = snd_soc_dai_get_drvdata(dai);
- dai->playback_dma_data = &spdif_private->dma_params_tx;
- dai->capture_dma_data = &spdif_private->dma_params_rx;
+ snd_soc_dai_init_dma_data(dai, &spdif_private->dma_params_tx,
+ &spdif_private->dma_params_rx);
snd_soc_add_dai_controls(dai, fsl_spdif_ctrls, ARRAY_SIZE(fsl_spdif_ctrls));
@@ -985,7 +985,7 @@ static bool fsl_spdif_writeable_reg(struct device *dev, unsigned int reg)
}
}
-static const struct regmap_config fsl_spdif_regmap_config = {
+static struct regmap_config fsl_spdif_regmap_config = {
.reg_bits = 32,
.reg_stride = 4,
.val_bits = 32,
@@ -1105,6 +1105,9 @@ static int fsl_spdif_probe(struct platform_device *pdev)
memcpy(&spdif_priv->cpu_dai_drv, &fsl_spdif_dai, sizeof(fsl_spdif_dai));
spdif_priv->cpu_dai_drv.name = spdif_priv->name;
+ if (of_property_read_bool(np, "big-endian"))
+ fsl_spdif_regmap_config.val_format_endian = REGMAP_ENDIAN_BIG;
+
/* Get the addresses and IRQ */
res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
regs = devm_ioremap_resource(&pdev->dev, res);
diff --git a/sound/soc/fsl/imx-pcm-fiq.c b/sound/soc/fsl/imx-pcm-fiq.c
index 6553202..7abf6a0 100644
--- a/sound/soc/fsl/imx-pcm-fiq.c
+++ b/sound/soc/fsl/imx-pcm-fiq.c
@@ -270,18 +270,17 @@ static int imx_pcm_new(struct snd_soc_pcm_runtime *rtd)
ret = imx_pcm_preallocate_dma_buffer(pcm,
SNDRV_PCM_STREAM_PLAYBACK);
if (ret)
- goto out;
+ return ret;
}
if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) {
ret = imx_pcm_preallocate_dma_buffer(pcm,
SNDRV_PCM_STREAM_CAPTURE);
if (ret)
- goto out;
+ return ret;
}
-out:
- return ret;
+ return 0;
}
static int ssi_irq = 0;
diff --git a/sound/soc/fsl/wm1133-ev1.c b/sound/soc/fsl/wm1133-ev1.c
index fce6325..804749a 100644
--- a/sound/soc/fsl/wm1133-ev1.c
+++ b/sound/soc/fsl/wm1133-ev1.c
@@ -214,12 +214,6 @@ static int wm1133_ev1_init(struct snd_soc_pcm_runtime *rtd)
struct snd_soc_codec *codec = rtd->codec;
struct snd_soc_dapm_context *dapm = &codec->dapm;
- snd_soc_dapm_new_controls(dapm, wm1133_ev1_widgets,
- ARRAY_SIZE(wm1133_ev1_widgets));
-
- snd_soc_dapm_add_routes(dapm, wm1133_ev1_map,
- ARRAY_SIZE(wm1133_ev1_map));
-
/* Headphone jack detection */
snd_soc_jack_new(codec, "Headphone", SND_JACK_HEADPHONE, &hp_jack);
snd_soc_jack_add_pins(&hp_jack, ARRAY_SIZE(hp_jack_pins),
@@ -257,6 +251,11 @@ static struct snd_soc_card wm1133_ev1 = {
.owner = THIS_MODULE,
.dai_link = &wm1133_ev1_dai,
.num_links = 1,
+
+ .dapm_widgets = wm1133_ev1_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(wm1133_ev1_widgets),
+ .dapm_routes = wm1133_ev1_map,
+ .num_dapm_routes = ARRAY_SIZE(wm1133_ev1_map),
};
static struct platform_device *wm1133_ev1_snd_device;
diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c
index 2a1b1b5..2ee8ed5 100644
--- a/sound/soc/generic/simple-card.c
+++ b/sound/soc/generic/simple-card.c
@@ -9,48 +9,72 @@
* published by the Free Software Foundation.
*/
#include <linux/clk.h>
+#include <linux/device.h>
#include <linux/module.h>
#include <linux/of.h>
#include <linux/platform_device.h>
#include <linux/string.h>
#include <sound/simple_card.h>
+#include <sound/soc-dai.h>
+#include <sound/soc.h>
+
+struct simple_card_data {
+ struct snd_soc_card snd_card;
+ struct asoc_simple_dai cpu_dai;
+ struct asoc_simple_dai codec_dai;
+ struct snd_soc_dai_link snd_link;
+};
static int __asoc_simple_card_dai_init(struct snd_soc_dai *dai,
- struct asoc_simple_dai *set,
- unsigned int daifmt)
+ struct asoc_simple_dai *set)
{
- int ret = 0;
+ int ret;
- daifmt |= set->fmt;
+ if (set->fmt) {
+ ret = snd_soc_dai_set_fmt(dai, set->fmt);
+ if (ret && ret != -ENOTSUPP) {
+ dev_err(dai->dev, "simple-card: set_fmt error\n");
+ goto err;
+ }
+ }
- if (daifmt)
- ret = snd_soc_dai_set_fmt(dai, daifmt);
+ if (set->sysclk) {
+ ret = snd_soc_dai_set_sysclk(dai, 0, set->sysclk, 0);
+ if (ret && ret != -ENOTSUPP) {
+ dev_err(dai->dev, "simple-card: set_sysclk error\n");
+ goto err;
+ }
+ }
- if (ret == -ENOTSUPP) {
- dev_dbg(dai->dev, "ASoC: set_fmt is not supported\n");
- ret = 0;
+ if (set->slots) {
+ ret = snd_soc_dai_set_tdm_slot(dai, 0, 0,
+ set->slots,
+ set->slot_width);
+ if (ret && ret != -ENOTSUPP) {
+ dev_err(dai->dev, "simple-card: set_tdm_slot error\n");
+ goto err;
+ }
}
- if (!ret && set->sysclk)
- ret = snd_soc_dai_set_sysclk(dai, 0, set->sysclk, 0);
+ ret = 0;
+err:
return ret;
}
static int asoc_simple_card_dai_init(struct snd_soc_pcm_runtime *rtd)
{
- struct asoc_simple_card_info *info =
+ struct simple_card_data *priv =
snd_soc_card_get_drvdata(rtd->card);
struct snd_soc_dai *codec = rtd->codec_dai;
struct snd_soc_dai *cpu = rtd->cpu_dai;
- unsigned int daifmt = info->daifmt;
int ret;
- ret = __asoc_simple_card_dai_init(codec, &info->codec_dai, daifmt);
+ ret = __asoc_simple_card_dai_init(codec, &priv->codec_dai);
if (ret < 0)
return ret;
- ret = __asoc_simple_card_dai_init(cpu, &info->cpu_dai, daifmt);
+ ret = __asoc_simple_card_dai_init(cpu, &priv->cpu_dai);
if (ret < 0)
return ret;
@@ -59,9 +83,12 @@ static int asoc_simple_card_dai_init(struct snd_soc_pcm_runtime *rtd)
static int
asoc_simple_card_sub_parse_of(struct device_node *np,
+ unsigned int daifmt,
struct asoc_simple_dai *dai,
- struct device_node **node)
+ const struct device_node **p_node,
+ const char **name)
{
+ struct device_node *node;
struct clk *clk;
int ret;
@@ -69,14 +96,20 @@ asoc_simple_card_sub_parse_of(struct device_node *np,
* get node via "sound-dai = <&phandle port>"
* it will be used as xxx_of_node on soc_bind_dai_link()
*/
- *node = of_parse_phandle(np, "sound-dai", 0);
- if (!*node)
+ node = of_parse_phandle(np, "sound-dai", 0);
+ if (!node)
return -ENODEV;
+ *p_node = node;
/* get dai->name */
- ret = snd_soc_of_get_dai_name(np, &dai->name);
+ ret = snd_soc_of_get_dai_name(np, name);
if (ret < 0)
- goto parse_error;
+ return ret;
+
+ /* parse TDM slot */
+ ret = snd_soc_of_parse_tdm_slot(np, &dai->slots, &dai->slot_width);
+ if (ret)
+ return ret;
/*
* bitclock-inversion, frame-inversion
@@ -84,6 +117,7 @@ asoc_simple_card_sub_parse_of(struct device_node *np,
* and specific "format" if it has
*/
dai->fmt = snd_soc_of_parse_daifmt(np, NULL);
+ dai->fmt |= daifmt;
/*
* dai->sysclk come from
@@ -95,7 +129,7 @@ asoc_simple_card_sub_parse_of(struct device_node *np,
clk = of_clk_get(np, 0);
if (IS_ERR(clk)) {
ret = PTR_ERR(clk);
- goto parse_error;
+ return ret;
}
dai->sysclk = clk_get_rate(clk);
@@ -104,37 +138,44 @@ asoc_simple_card_sub_parse_of(struct device_node *np,
"system-clock-frequency",
&dai->sysclk);
} else {
- clk = of_clk_get(*node, 0);
+ clk = of_clk_get(node, 0);
if (!IS_ERR(clk))
dai->sysclk = clk_get_rate(clk);
}
- ret = 0;
-
-parse_error:
- of_node_put(*node);
-
- return ret;
+ return 0;
}
static int asoc_simple_card_parse_of(struct device_node *node,
- struct asoc_simple_card_info *info,
- struct device *dev,
- struct device_node **of_cpu,
- struct device_node **of_codec,
- struct device_node **of_platform)
+ struct simple_card_data *priv,
+ struct device *dev)
{
+ struct snd_soc_dai_link *dai_link = priv->snd_card.dai_link;
+ struct asoc_simple_dai *codec_dai = &priv->codec_dai;
+ struct asoc_simple_dai *cpu_dai = &priv->cpu_dai;
struct device_node *np;
char *name;
+ unsigned int daifmt;
int ret;
+ /* parsing the card name from DT */
+ snd_soc_of_parse_card_name(&priv->snd_card, "simple-audio-card,name");
+
/* get CPU/CODEC common format via simple-audio-card,format */
- info->daifmt = snd_soc_of_parse_daifmt(node, "simple-audio-card,") &
+ daifmt = snd_soc_of_parse_daifmt(node, "simple-audio-card,") &
(SND_SOC_DAIFMT_FORMAT_MASK | SND_SOC_DAIFMT_INV_MASK);
+ /* off-codec widgets */
+ if (of_property_read_bool(node, "simple-audio-card,widgets")) {
+ ret = snd_soc_of_parse_audio_simple_widgets(&priv->snd_card,
+ "simple-audio-card,widgets");
+ if (ret)
+ return ret;
+ }
+
/* DAPM routes */
if (of_property_read_bool(node, "simple-audio-card,routing")) {
- ret = snd_soc_of_parse_audio_routing(&info->snd_card,
+ ret = snd_soc_of_parse_audio_routing(&priv->snd_card,
"simple-audio-card,routing");
if (ret)
return ret;
@@ -143,125 +184,173 @@ static int asoc_simple_card_parse_of(struct device_node *node,
/* CPU sub-node */
ret = -EINVAL;
np = of_get_child_by_name(node, "simple-audio-card,cpu");
- if (np)
- ret = asoc_simple_card_sub_parse_of(np,
- &info->cpu_dai,
- of_cpu);
+ if (np) {
+ ret = asoc_simple_card_sub_parse_of(np, daifmt,
+ cpu_dai,
+ &dai_link->cpu_of_node,
+ &dai_link->cpu_dai_name);
+ of_node_put(np);
+ }
if (ret < 0)
return ret;
/* CODEC sub-node */
ret = -EINVAL;
np = of_get_child_by_name(node, "simple-audio-card,codec");
- if (np)
- ret = asoc_simple_card_sub_parse_of(np,
- &info->codec_dai,
- of_codec);
+ if (np) {
+ ret = asoc_simple_card_sub_parse_of(np, daifmt,
+ codec_dai,
+ &dai_link->codec_of_node,
+ &dai_link->codec_dai_name);
+ of_node_put(np);
+ }
if (ret < 0)
return ret;
- if (!info->cpu_dai.name || !info->codec_dai.name)
+ /*
+ * overwrite cpu_dai->fmt as its DAIFMT_MASTER bit is based on CODEC
+ * while the other bits should be identical unless buggy SW/HW design.
+ */
+ cpu_dai->fmt = codec_dai->fmt;
+
+ if (!dai_link->cpu_dai_name || !dai_link->codec_dai_name)
return -EINVAL;
/* card name is created from CPU/CODEC dai name */
name = devm_kzalloc(dev,
- strlen(info->cpu_dai.name) +
- strlen(info->codec_dai.name) + 2,
+ strlen(dai_link->cpu_dai_name) +
+ strlen(dai_link->codec_dai_name) + 2,
GFP_KERNEL);
- sprintf(name, "%s-%s", info->cpu_dai.name, info->codec_dai.name);
- info->name = info->card = name;
+ sprintf(name, "%s-%s", dai_link->cpu_dai_name,
+ dai_link->codec_dai_name);
+ if (!priv->snd_card.name)
+ priv->snd_card.name = name;
+ dai_link->name = dai_link->stream_name = name;
/* simple-card assumes platform == cpu */
- *of_platform = *of_cpu;
+ dai_link->platform_of_node = dai_link->cpu_of_node;
- dev_dbg(dev, "card-name : %s\n", info->card);
- dev_dbg(dev, "platform : %04x\n", info->daifmt);
+ dev_dbg(dev, "card-name : %s\n", name);
+ dev_dbg(dev, "platform : %04x\n", daifmt);
dev_dbg(dev, "cpu : %s / %04x / %d\n",
- info->cpu_dai.name,
- info->cpu_dai.fmt,
- info->cpu_dai.sysclk);
+ dai_link->cpu_dai_name,
+ cpu_dai->fmt,
+ cpu_dai->sysclk);
dev_dbg(dev, "codec : %s / %04x / %d\n",
- info->codec_dai.name,
- info->codec_dai.fmt,
- info->codec_dai.sysclk);
+ dai_link->codec_dai_name,
+ codec_dai->fmt,
+ codec_dai->sysclk);
+
+ /*
+ * soc_bind_dai_link() will check cpu name
+ * after of_node matching if dai_link has cpu_dai_name.
+ * but, it will never match if name was created by fmt_single_name()
+ * remove cpu_dai_name to escape name matching.
+ * see
+ * fmt_single_name()
+ * fmt_multiple_name()
+ */
+ dai_link->cpu_dai_name = NULL;
return 0;
}
+/* update the reference count of the devices nodes at end of probe */
+static int asoc_simple_card_unref(struct platform_device *pdev)
+{
+ struct snd_soc_card *card = platform_get_drvdata(pdev);
+ struct snd_soc_dai_link *dai_link;
+ struct device_node *np;
+ int num_links;
+
+ for (num_links = 0, dai_link = card->dai_link;
+ num_links < card->num_links;
+ num_links++, dai_link++) {
+ np = (struct device_node *) dai_link->cpu_of_node;
+ if (np)
+ of_node_put(np);
+ np = (struct device_node *) dai_link->codec_of_node;
+ if (np)
+ of_node_put(np);
+ }
+ return 0;
+}
+
static int asoc_simple_card_probe(struct platform_device *pdev)
{
- struct asoc_simple_card_info *cinfo;
+ struct simple_card_data *priv;
+ struct snd_soc_dai_link *dai_link;
struct device_node *np = pdev->dev.of_node;
- struct device_node *of_cpu, *of_codec, *of_platform;
struct device *dev = &pdev->dev;
int ret;
- cinfo = NULL;
- of_cpu = NULL;
- of_codec = NULL;
- of_platform = NULL;
-
- cinfo = devm_kzalloc(dev, sizeof(*cinfo), GFP_KERNEL);
- if (!cinfo)
+ priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL);
+ if (!priv)
return -ENOMEM;
+ /*
+ * init snd_soc_card
+ */
+ priv->snd_card.owner = THIS_MODULE;
+ priv->snd_card.dev = dev;
+ dai_link = &priv->snd_link;
+ priv->snd_card.dai_link = dai_link;
+ priv->snd_card.num_links = 1;
+
if (np && of_device_is_available(np)) {
- cinfo->snd_card.dev = dev;
- ret = asoc_simple_card_parse_of(np, cinfo, dev,
- &of_cpu,
- &of_codec,
- &of_platform);
+ ret = asoc_simple_card_parse_of(np, priv, dev);
if (ret < 0) {
if (ret != -EPROBE_DEFER)
dev_err(dev, "parse error %d\n", ret);
- return ret;
+ goto err;
}
} else {
- if (!dev->platform_data) {
+ struct asoc_simple_card_info *cinfo;
+
+ cinfo = dev->platform_data;
+ if (!cinfo) {
dev_err(dev, "no info for asoc-simple-card\n");
return -EINVAL;
}
- memcpy(cinfo, dev->platform_data, sizeof(*cinfo));
- cinfo->snd_card.dev = dev;
- }
+ if (!cinfo->name ||
+ !cinfo->codec_dai.name ||
+ !cinfo->codec ||
+ !cinfo->platform ||
+ !cinfo->cpu_dai.name) {
+ dev_err(dev, "insufficient asoc_simple_card_info settings\n");
+ return -EINVAL;
+ }
- if (!cinfo->name ||
- !cinfo->card ||
- !cinfo->codec_dai.name ||
- !(cinfo->codec || of_codec) ||
- !(cinfo->platform || of_platform) ||
- !(cinfo->cpu_dai.name || of_cpu)) {
- dev_err(dev, "insufficient asoc_simple_card_info settings\n");
- return -EINVAL;
+ priv->snd_card.name = (cinfo->card) ? cinfo->card : cinfo->name;
+ dai_link->name = cinfo->name;
+ dai_link->stream_name = cinfo->name;
+ dai_link->platform_name = cinfo->platform;
+ dai_link->codec_name = cinfo->codec;
+ dai_link->cpu_dai_name = cinfo->cpu_dai.name;
+ dai_link->codec_dai_name = cinfo->codec_dai.name;
+ memcpy(&priv->cpu_dai, &cinfo->cpu_dai,
+ sizeof(priv->cpu_dai));
+ memcpy(&priv->codec_dai, &cinfo->codec_dai,
+ sizeof(priv->codec_dai));
+
+ priv->cpu_dai.fmt |= cinfo->daifmt;
+ priv->codec_dai.fmt |= cinfo->daifmt;
}
/*
* 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.name;
- cinfo->snd_link.platform_name = cinfo->platform;
- cinfo->snd_link.codec_name = cinfo->codec;
- cinfo->snd_link.codec_dai_name = cinfo->codec_dai.name;
- cinfo->snd_link.cpu_of_node = of_cpu;
- cinfo->snd_link.codec_of_node = of_codec;
- cinfo->snd_link.platform_of_node = of_platform;
- cinfo->snd_link.init = asoc_simple_card_dai_init;
+ dai_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;
+ snd_soc_card_set_drvdata(&priv->snd_card, priv);
- snd_soc_card_set_drvdata(&cinfo->snd_card, cinfo);
+ ret = devm_snd_soc_register_card(&pdev->dev, &priv->snd_card);
- return devm_snd_soc_register_card(&pdev->dev, &cinfo->snd_card);
+err:
+ asoc_simple_card_unref(pdev);
+ return ret;
}
static const struct of_device_id asoc_simple_of_match[] = {
diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig
index 61c10bf..3c81b38 100644
--- a/sound/soc/intel/Kconfig
+++ b/sound/soc/intel/Kconfig
@@ -2,12 +2,50 @@ config SND_MFLD_MACHINE
tristate "SOC Machine Audio driver for Intel Medfield MID platform"
depends on INTEL_SCU_IPC
select SND_SOC_SN95031
- select SND_SST_PLATFORM
+ select SND_SST_MFLD_PLATFORM
help
This adds support for ASoC machine driver for Intel(R) MID Medfield platform
used as alsa device in audio substem in Intel(R) MID devices
Say Y if you have such a device
If unsure select "N".
-config SND_SST_PLATFORM
+config SND_SST_MFLD_PLATFORM
tristate
+
+config SND_SOC_INTEL_SST
+ tristate "ASoC support for Intel(R) Smart Sound Technology"
+ select SND_SOC_INTEL_SST_ACPI if ACPI
+ depends on (X86 || COMPILE_TEST)
+ help
+ This adds support for Intel(R) Smart Sound Technology (SST).
+ Say Y if you have such a device
+ If unsure select "N".
+
+config SND_SOC_INTEL_SST_ACPI
+ tristate
+
+config SND_SOC_INTEL_HASWELL
+ tristate
+
+config SND_SOC_INTEL_BAYTRAIL
+ tristate
+
+config SND_SOC_INTEL_HASWELL_MACH
+ tristate "ASoC Audio DSP support for Intel Haswell Lynxpoint"
+ depends on SND_SOC_INTEL_SST && X86_INTEL_LPSS && I2C
+ select SND_SOC_INTEL_HASWELL
+ select SND_SOC_RT5640
+ help
+ This adds support for the Lynxpoint Audio DSP on Intel(R) Haswell
+ Ultrabook platforms.
+ Say Y if you have such a device
+ If unsure select "N".
+
+config SND_SOC_INTEL_BYT_RT5640_MACH
+ tristate "ASoC Audio driver for Intel Baytrail with RT5640 codec"
+ depends on SND_SOC_INTEL_SST && X86_INTEL_LPSS && I2C
+ select SND_SOC_INTEL_BAYTRAIL
+ select SND_SOC_RT5640
+ help
+ This adds audio driver for Intel Baytrail platform based boards
+ with the RT5640 audio codec.
diff --git a/sound/soc/intel/Makefile b/sound/soc/intel/Makefile
index 6398833..edeb79ae 100644
--- a/sound/soc/intel/Makefile
+++ b/sound/soc/intel/Makefile
@@ -1,5 +1,28 @@
-snd-soc-sst-platform-objs := sst_platform.o
+# Core support
+snd-soc-sst-dsp-objs := sst-dsp.o sst-firmware.o
+snd-soc-sst-acpi-objs := sst-acpi.o
+
+snd-soc-sst-mfld-platform-objs := sst-mfld-platform.o
snd-soc-mfld-machine-objs := mfld_machine.o
-obj-$(CONFIG_SND_SST_PLATFORM) += snd-soc-sst-platform.o
+obj-$(CONFIG_SND_SST_MFLD_PLATFORM) += snd-soc-sst-mfld-platform.o
obj-$(CONFIG_SND_MFLD_MACHINE) += snd-soc-mfld-machine.o
+
+obj-$(CONFIG_SND_SOC_INTEL_SST) += snd-soc-sst-dsp.o
+obj-$(CONFIG_SND_SOC_INTEL_SST_ACPI) += snd-soc-sst-acpi.o
+
+# Platform Support
+snd-soc-sst-haswell-pcm-objs := \
+ sst-haswell-ipc.o sst-haswell-pcm.o sst-haswell-dsp.o
+snd-soc-sst-baytrail-pcm-objs := \
+ sst-baytrail-ipc.o sst-baytrail-pcm.o sst-baytrail-dsp.o
+
+obj-$(CONFIG_SND_SOC_INTEL_HASWELL) += snd-soc-sst-haswell-pcm.o
+obj-$(CONFIG_SND_SOC_INTEL_BAYTRAIL) += snd-soc-sst-baytrail-pcm.o
+
+# Machine support
+snd-soc-sst-haswell-objs := haswell.o
+snd-soc-sst-byt-rt5640-mach-objs := byt-rt5640.o
+
+obj-$(CONFIG_SND_SOC_INTEL_HASWELL_MACH) += snd-soc-sst-haswell.o
+obj-$(CONFIG_SND_SOC_INTEL_BYT_RT5640_MACH) += snd-soc-sst-byt-rt5640-mach.o
diff --git a/sound/soc/intel/byt-rt5640.c b/sound/soc/intel/byt-rt5640.c
new file mode 100644
index 0000000..eff97c8
--- /dev/null
+++ b/sound/soc/intel/byt-rt5640.c
@@ -0,0 +1,187 @@
+/*
+ * Intel Baytrail SST RT5640 machine driver
+ * Copyright (c) 2014, Intel Corporation.
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms and conditions of the GNU General Public License,
+ * version 2, as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope 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.
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <linux/acpi.h>
+#include <linux/device.h>
+#include <linux/slab.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/jack.h>
+#include "../codecs/rt5640.h"
+
+#include "sst-dsp.h"
+
+static const struct snd_soc_dapm_widget byt_rt5640_widgets[] = {
+ SND_SOC_DAPM_HP("Headphone", NULL),
+ SND_SOC_DAPM_MIC("Headset Mic", NULL),
+ SND_SOC_DAPM_MIC("Internal Mic", NULL),
+ SND_SOC_DAPM_SPK("Speaker", NULL),
+};
+
+static const struct snd_soc_dapm_route byt_rt5640_audio_map[] = {
+ {"IN2P", NULL, "Headset Mic"},
+ {"IN2N", NULL, "Headset Mic"},
+ {"DMIC1", NULL, "Internal Mic"},
+ {"Headphone", NULL, "HPOL"},
+ {"Headphone", NULL, "HPOR"},
+ {"Speaker", NULL, "SPOLP"},
+ {"Speaker", NULL, "SPOLN"},
+ {"Speaker", NULL, "SPORP"},
+ {"Speaker", NULL, "SPORN"},
+};
+
+static const struct snd_kcontrol_new byt_rt5640_controls[] = {
+ SOC_DAPM_PIN_SWITCH("Headphone"),
+ SOC_DAPM_PIN_SWITCH("Headset Mic"),
+ SOC_DAPM_PIN_SWITCH("Internal Mic"),
+ SOC_DAPM_PIN_SWITCH("Speaker"),
+};
+
+static int byt_rt5640_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ int ret;
+
+ ret = snd_soc_dai_set_sysclk(codec_dai, RT5640_SCLK_S_PLL1,
+ params_rate(params) * 256,
+ SND_SOC_CLOCK_IN);
+ if (ret < 0) {
+ dev_err(codec_dai->dev, "can't set codec clock %d\n", ret);
+ return ret;
+ }
+ ret = snd_soc_dai_set_pll(codec_dai, 0, RT5640_PLL1_S_BCLK1,
+ params_rate(params) * 64,
+ params_rate(params) * 256);
+ if (ret < 0) {
+ dev_err(codec_dai->dev, "can't set codec pll: %d\n", ret);
+ return ret;
+ }
+ return 0;
+}
+
+static int byt_rt5640_init(struct snd_soc_pcm_runtime *runtime)
+{
+ int ret;
+ struct snd_soc_codec *codec = runtime->codec;
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
+ struct snd_soc_card *card = runtime->card;
+
+ card->dapm.idle_bias_off = true;
+
+ ret = snd_soc_add_card_controls(card, byt_rt5640_controls,
+ ARRAY_SIZE(byt_rt5640_controls));
+ if (ret) {
+ dev_err(card->dev, "unable to add card controls\n");
+ return ret;
+ }
+
+ snd_soc_dapm_ignore_suspend(dapm, "HPOL");
+ snd_soc_dapm_ignore_suspend(dapm, "HPOR");
+
+ snd_soc_dapm_ignore_suspend(dapm, "SPOLP");
+ snd_soc_dapm_ignore_suspend(dapm, "SPOLN");
+ snd_soc_dapm_ignore_suspend(dapm, "SPORP");
+ snd_soc_dapm_ignore_suspend(dapm, "SPORN");
+
+ snd_soc_dapm_enable_pin(dapm, "Headset Mic");
+ snd_soc_dapm_enable_pin(dapm, "Headphone");
+ snd_soc_dapm_enable_pin(dapm, "Speaker");
+ snd_soc_dapm_enable_pin(dapm, "Internal Mic");
+
+ snd_soc_dapm_sync(dapm);
+ return ret;
+}
+
+static struct snd_soc_ops byt_rt5640_ops = {
+ .hw_params = byt_rt5640_hw_params,
+};
+
+static struct snd_soc_dai_link byt_rt5640_dais[] = {
+ {
+ .name = "Baytrail Audio",
+ .stream_name = "Audio",
+ .cpu_dai_name = "Front-cpu-dai",
+ .codec_dai_name = "rt5640-aif1",
+ .codec_name = "i2c-10EC5640:00",
+ .platform_name = "baytrail-pcm-audio",
+ .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBS_CFS,
+ .init = byt_rt5640_init,
+ .ignore_suspend = 1,
+ .ops = &byt_rt5640_ops,
+ },
+ {
+ .name = "Baytrail Voice",
+ .stream_name = "Voice",
+ .cpu_dai_name = "Mic1-cpu-dai",
+ .codec_dai_name = "rt5640-aif1",
+ .codec_name = "i2c-10EC5640:00",
+ .platform_name = "baytrail-pcm-audio",
+ .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBS_CFS,
+ .init = NULL,
+ .ignore_suspend = 1,
+ .ops = &byt_rt5640_ops,
+ },
+};
+
+static struct snd_soc_card byt_rt5640_card = {
+ .name = "byt-rt5640",
+ .dai_link = byt_rt5640_dais,
+ .num_links = ARRAY_SIZE(byt_rt5640_dais),
+ .dapm_widgets = byt_rt5640_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(byt_rt5640_widgets),
+ .dapm_routes = byt_rt5640_audio_map,
+ .num_dapm_routes = ARRAY_SIZE(byt_rt5640_audio_map),
+};
+
+static int byt_rt5640_probe(struct platform_device *pdev)
+{
+ struct snd_soc_card *card = &byt_rt5640_card;
+ struct device *dev = &pdev->dev;
+
+ card->dev = &pdev->dev;
+ dev_set_drvdata(dev, card);
+ return snd_soc_register_card(card);
+}
+
+static int byt_rt5640_remove(struct platform_device *pdev)
+{
+ struct snd_soc_card *card = platform_get_drvdata(pdev);
+
+ snd_soc_unregister_card(card);
+
+ return 0;
+}
+
+static struct platform_driver byt_rt5640_audio = {
+ .probe = byt_rt5640_probe,
+ .remove = byt_rt5640_remove,
+ .driver = {
+ .name = "byt-rt5640",
+ .owner = THIS_MODULE,
+ },
+};
+module_platform_driver(byt_rt5640_audio)
+
+MODULE_DESCRIPTION("ASoC Intel(R) Baytrail Machine driver");
+MODULE_AUTHOR("Omair Md Abdullah, Jarkko Nikula");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:byt-rt5640");
diff --git a/sound/soc/intel/haswell.c b/sound/soc/intel/haswell.c
new file mode 100644
index 0000000..54345a2
--- /dev/null
+++ b/sound/soc/intel/haswell.c
@@ -0,0 +1,235 @@
+/*
+ * Intel Haswell Lynxpoint SST Audio
+ *
+ * Copyright (C) 2013, Intel Corporation. All rights reserved.
+ *
+ * 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.
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/pcm_params.h>
+
+#include "sst-dsp.h"
+#include "sst-haswell-ipc.h"
+
+#include "../codecs/rt5640.h"
+
+/* Haswell ULT platforms have a Headphone and Mic jack */
+static const struct snd_soc_dapm_widget haswell_widgets[] = {
+ SND_SOC_DAPM_HP("Headphones", NULL),
+ SND_SOC_DAPM_MIC("Mic", NULL),
+};
+
+static const struct snd_soc_dapm_route haswell_rt5640_map[] = {
+
+ {"Headphones", NULL, "HPOR"},
+ {"Headphones", NULL, "HPOL"},
+ {"IN2P", NULL, "Mic"},
+
+ /* CODEC BE connections */
+ {"SSP0 CODEC IN", NULL, "AIF1 Capture"},
+ {"AIF1 Playback", NULL, "SSP0 CODEC OUT"},
+};
+
+static int haswell_ssp0_fixup(struct snd_soc_pcm_runtime *rtd,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_interval *rate = hw_param_interval(params,
+ SNDRV_PCM_HW_PARAM_RATE);
+ struct snd_interval *channels = hw_param_interval(params,
+ SNDRV_PCM_HW_PARAM_CHANNELS);
+
+ /* The ADSP will covert the FE rate to 48k, stereo */
+ rate->min = rate->max = 48000;
+ channels->min = channels->max = 2;
+
+ /* set SSP0 to 16 bit */
+ snd_mask_set(&params->masks[SNDRV_PCM_HW_PARAM_FORMAT -
+ SNDRV_PCM_HW_PARAM_FIRST_MASK],
+ SNDRV_PCM_FORMAT_S16_LE);
+ return 0;
+}
+
+static int haswell_rt5640_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ int ret;
+
+ ret = snd_soc_dai_set_sysclk(codec_dai, RT5640_SCLK_S_MCLK, 12288000,
+ SND_SOC_CLOCK_IN);
+
+ if (ret < 0) {
+ dev_err(rtd->dev, "can't set codec sysclk configuration\n");
+ return ret;
+ }
+
+ /* set correct codec filter for DAI format and clock config */
+ snd_soc_update_bits(rtd->codec, 0x83, 0xffff, 0x8000);
+
+ return ret;
+}
+
+static struct snd_soc_ops haswell_rt5640_ops = {
+ .hw_params = haswell_rt5640_hw_params,
+};
+
+static int haswell_rtd_init(struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
+ struct sst_pdata *pdata = dev_get_platdata(rtd->platform->dev);
+ struct sst_hsw *haswell = pdata->dsp;
+ int ret;
+
+ /* Set ADSP SSP port settings */
+ ret = sst_hsw_device_set_config(haswell, SST_HSW_DEVICE_SSP_0,
+ SST_HSW_DEVICE_MCLK_FREQ_24_MHZ,
+ SST_HSW_DEVICE_CLOCK_MASTER, 9);
+ if (ret < 0) {
+ dev_err(rtd->dev, "failed to set device config\n");
+ return ret;
+ }
+
+ /* always connected */
+ snd_soc_dapm_enable_pin(dapm, "Headphones");
+ snd_soc_dapm_enable_pin(dapm, "Mic");
+
+ return 0;
+}
+
+static struct snd_soc_dai_link haswell_rt5640_dais[] = {
+ /* Front End DAI links */
+ {
+ .name = "System",
+ .stream_name = "System Playback",
+ .cpu_dai_name = "System Pin",
+ .platform_name = "haswell-pcm-audio",
+ .dynamic = 1,
+ .codec_name = "snd-soc-dummy",
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .init = haswell_rtd_init,
+ .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+ .dpcm_playback = 1,
+ },
+ {
+ .name = "Offload0",
+ .stream_name = "Offload0 Playback",
+ .cpu_dai_name = "Offload0 Pin",
+ .platform_name = "haswell-pcm-audio",
+ .dynamic = 1,
+ .codec_name = "snd-soc-dummy",
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+ .dpcm_playback = 1,
+ },
+ {
+ .name = "Offload1",
+ .stream_name = "Offload1 Playback",
+ .cpu_dai_name = "Offload1 Pin",
+ .platform_name = "haswell-pcm-audio",
+ .dynamic = 1,
+ .codec_name = "snd-soc-dummy",
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+ .dpcm_playback = 1,
+ },
+ {
+ .name = "Loopback",
+ .stream_name = "Loopback",
+ .cpu_dai_name = "Loopback Pin",
+ .platform_name = "haswell-pcm-audio",
+ .dynamic = 0,
+ .codec_name = "snd-soc-dummy",
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+ .dpcm_capture = 1,
+ },
+ {
+ .name = "Capture",
+ .stream_name = "Capture",
+ .cpu_dai_name = "Capture Pin",
+ .platform_name = "haswell-pcm-audio",
+ .dynamic = 1,
+ .codec_name = "snd-soc-dummy",
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+ .dpcm_capture = 1,
+ },
+
+ /* Back End DAI links */
+ {
+ /* SSP0 - Codec */
+ .name = "Codec",
+ .be_id = 0,
+ .cpu_dai_name = "snd-soc-dummy-dai",
+ .platform_name = "snd-soc-dummy",
+ .no_pcm = 1,
+ .codec_name = "i2c-INT33CA:00",
+ .codec_dai_name = "rt5640-aif1",
+ .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBS_CFS,
+ .ignore_suspend = 1,
+ .ignore_pmdown_time = 1,
+ .be_hw_params_fixup = haswell_ssp0_fixup,
+ .ops = &haswell_rt5640_ops,
+ .dpcm_playback = 1,
+ .dpcm_capture = 1,
+ },
+};
+
+/* audio machine driver for Haswell Lynxpoint DSP + RT5640 */
+static struct snd_soc_card haswell_rt5640 = {
+ .name = "haswell-rt5640",
+ .owner = THIS_MODULE,
+ .dai_link = haswell_rt5640_dais,
+ .num_links = ARRAY_SIZE(haswell_rt5640_dais),
+ .dapm_widgets = haswell_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(haswell_widgets),
+ .dapm_routes = haswell_rt5640_map,
+ .num_dapm_routes = ARRAY_SIZE(haswell_rt5640_map),
+ .fully_routed = true,
+};
+
+static int haswell_audio_probe(struct platform_device *pdev)
+{
+ haswell_rt5640.dev = &pdev->dev;
+
+ return snd_soc_register_card(&haswell_rt5640);
+}
+
+static int haswell_audio_remove(struct platform_device *pdev)
+{
+ snd_soc_unregister_card(&haswell_rt5640);
+ return 0;
+}
+
+static struct platform_driver haswell_audio = {
+ .probe = haswell_audio_probe,
+ .remove = haswell_audio_remove,
+ .driver = {
+ .name = "haswell-audio",
+ .owner = THIS_MODULE,
+ },
+};
+
+module_platform_driver(haswell_audio)
+
+/* Module information */
+MODULE_AUTHOR("Liam Girdwood, Xingchao Wang");
+MODULE_DESCRIPTION("Intel SST Audio for Haswell Lynxpoint");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:haswell-audio");
diff --git a/sound/soc/intel/mfld_machine.c b/sound/soc/intel/mfld_machine.c
index d3d4c32..031d787 100644
--- a/sound/soc/intel/mfld_machine.c
+++ b/sound/soc/intel/mfld_machine.c
@@ -53,6 +53,7 @@ enum soc_mic_bias_zones {
static unsigned int hs_switch;
static unsigned int lo_dac;
+static struct snd_soc_codec *mfld_codec;
struct mfld_mc_private {
void __iomem *int_base;
@@ -100,40 +101,47 @@ static int headset_get_switch(struct snd_kcontrol *kcontrol,
static int headset_set_switch(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
- struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct snd_soc_card *card = snd_kcontrol_chip(kcontrol);
+ struct snd_soc_dapm_context *dapm = &card->dapm;
if (ucontrol->value.integer.value[0] == hs_switch)
return 0;
+ snd_soc_dapm_mutex_lock(dapm);
+
if (ucontrol->value.integer.value[0]) {
pr_debug("hs_set HS path\n");
- snd_soc_dapm_enable_pin(&codec->dapm, "Headphones");
- snd_soc_dapm_disable_pin(&codec->dapm, "EPOUT");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Headphones");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "EPOUT");
} else {
pr_debug("hs_set EP path\n");
- snd_soc_dapm_disable_pin(&codec->dapm, "Headphones");
- snd_soc_dapm_enable_pin(&codec->dapm, "EPOUT");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Headphones");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "EPOUT");
}
- snd_soc_dapm_sync(&codec->dapm);
+
+ snd_soc_dapm_sync_unlocked(dapm);
+
+ snd_soc_dapm_mutex_unlock(dapm);
+
hs_switch = ucontrol->value.integer.value[0];
return 0;
}
-static void lo_enable_out_pins(struct snd_soc_codec *codec)
+static void lo_enable_out_pins(struct snd_soc_dapm_context *dapm)
{
- snd_soc_dapm_enable_pin(&codec->dapm, "IHFOUTL");
- snd_soc_dapm_enable_pin(&codec->dapm, "IHFOUTR");
- snd_soc_dapm_enable_pin(&codec->dapm, "LINEOUTL");
- snd_soc_dapm_enable_pin(&codec->dapm, "LINEOUTR");
- snd_soc_dapm_enable_pin(&codec->dapm, "VIB1OUT");
- snd_soc_dapm_enable_pin(&codec->dapm, "VIB2OUT");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "IHFOUTL");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "IHFOUTR");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "LINEOUTL");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "LINEOUTR");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "VIB1OUT");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "VIB2OUT");
if (hs_switch) {
- snd_soc_dapm_enable_pin(&codec->dapm, "Headphones");
- snd_soc_dapm_disable_pin(&codec->dapm, "EPOUT");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Headphones");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "EPOUT");
} else {
- snd_soc_dapm_disable_pin(&codec->dapm, "Headphones");
- snd_soc_dapm_enable_pin(&codec->dapm, "EPOUT");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Headphones");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "EPOUT");
}
}
@@ -147,45 +155,53 @@ static int lo_get_switch(struct snd_kcontrol *kcontrol,
static int lo_set_switch(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
- struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct snd_soc_card *card = snd_kcontrol_chip(kcontrol);
+ struct snd_soc_dapm_context *dapm = &card->dapm;
if (ucontrol->value.integer.value[0] == lo_dac)
return 0;
+ snd_soc_dapm_mutex_lock(dapm);
+
/* we dont want to work with last state of lineout so just enable all
* pins and then disable pins not required
*/
- lo_enable_out_pins(codec);
+ lo_enable_out_pins(dapm);
+
switch (ucontrol->value.integer.value[0]) {
case 0:
pr_debug("set vibra path\n");
- snd_soc_dapm_disable_pin(&codec->dapm, "VIB1OUT");
- snd_soc_dapm_disable_pin(&codec->dapm, "VIB2OUT");
- snd_soc_update_bits(codec, SN95031_LOCTL, 0x66, 0);
+ snd_soc_dapm_disable_pin_unlocked(dapm, "VIB1OUT");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "VIB2OUT");
+ snd_soc_update_bits(mfld_codec, SN95031_LOCTL, 0x66, 0);
break;
case 1:
pr_debug("set hs path\n");
- snd_soc_dapm_disable_pin(&codec->dapm, "Headphones");
- snd_soc_dapm_disable_pin(&codec->dapm, "EPOUT");
- snd_soc_update_bits(codec, SN95031_LOCTL, 0x66, 0x22);
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Headphones");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "EPOUT");
+ snd_soc_update_bits(mfld_codec, SN95031_LOCTL, 0x66, 0x22);
break;
case 2:
pr_debug("set spkr path\n");
- snd_soc_dapm_disable_pin(&codec->dapm, "IHFOUTL");
- snd_soc_dapm_disable_pin(&codec->dapm, "IHFOUTR");
- snd_soc_update_bits(codec, SN95031_LOCTL, 0x66, 0x44);
+ snd_soc_dapm_disable_pin_unlocked(dapm, "IHFOUTL");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "IHFOUTR");
+ snd_soc_update_bits(mfld_codec, SN95031_LOCTL, 0x66, 0x44);
break;
case 3:
pr_debug("set null path\n");
- snd_soc_dapm_disable_pin(&codec->dapm, "LINEOUTL");
- snd_soc_dapm_disable_pin(&codec->dapm, "LINEOUTR");
- snd_soc_update_bits(codec, SN95031_LOCTL, 0x66, 0x66);
+ snd_soc_dapm_disable_pin_unlocked(dapm, "LINEOUTL");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "LINEOUTR");
+ snd_soc_update_bits(mfld_codec, SN95031_LOCTL, 0x66, 0x66);
break;
}
- snd_soc_dapm_sync(&codec->dapm);
+
+ snd_soc_dapm_sync_unlocked(dapm);
+
+ snd_soc_dapm_mutex_unlock(dapm);
+
lo_dac = ucontrol->value.integer.value[0];
return 0;
}
@@ -221,26 +237,11 @@ static void mfld_jack_check(unsigned int intr_status)
static int mfld_init(struct snd_soc_pcm_runtime *runtime)
{
- struct snd_soc_codec *codec = runtime->codec;
- struct snd_soc_dapm_context *dapm = &codec->dapm;
+ struct snd_soc_dapm_context *dapm = &runtime->card->dapm;
int ret_val;
- /* Add jack sense widgets */
- snd_soc_dapm_new_controls(dapm, mfld_widgets, ARRAY_SIZE(mfld_widgets));
-
- /* Set up the map */
- snd_soc_dapm_add_routes(dapm, mfld_map, ARRAY_SIZE(mfld_map));
+ mfld_codec = runtime->codec;
- /* always connected */
- snd_soc_dapm_enable_pin(dapm, "Headphones");
- snd_soc_dapm_enable_pin(dapm, "Mic");
-
- ret_val = snd_soc_add_codec_controls(codec, mfld_snd_controls,
- ARRAY_SIZE(mfld_snd_controls));
- if (ret_val) {
- pr_err("soc_add_controls failed %d", ret_val);
- return ret_val;
- }
/* default is earpiece pin, userspace sets it explcitly */
snd_soc_dapm_disable_pin(dapm, "Headphones");
/* default is lineout NC, userspace sets it explcitly */
@@ -253,7 +254,7 @@ static int mfld_init(struct snd_soc_pcm_runtime *runtime)
snd_soc_dapm_disable_pin(dapm, "LINEINR");
/* Headset and button jack detection */
- ret_val = snd_soc_jack_new(codec, "Intel(R) MID Audio Jack",
+ ret_val = snd_soc_jack_new(mfld_codec, "Intel(R) MID Audio Jack",
SND_JACK_HEADSET | SND_JACK_BTN_0 |
SND_JACK_BTN_1, &mfld_jack);
if (ret_val) {
@@ -335,6 +336,13 @@ static struct snd_soc_card snd_soc_card_mfld = {
.owner = THIS_MODULE,
.dai_link = mfld_msic_dailink,
.num_links = ARRAY_SIZE(mfld_msic_dailink),
+
+ .controls = mfld_snd_controls,
+ .num_controls = ARRAY_SIZE(mfld_snd_controls),
+ .dapm_widgets = mfld_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(mfld_widgets),
+ .dapm_routes = mfld_map,
+ .num_dapm_routes = ARRAY_SIZE(mfld_map),
};
static irqreturn_t snd_mfld_jack_intr_handler(int irq, void *dev)
diff --git a/sound/soc/intel/sst-acpi.c b/sound/soc/intel/sst-acpi.c
new file mode 100644
index 0000000..5d06eec
--- /dev/null
+++ b/sound/soc/intel/sst-acpi.c
@@ -0,0 +1,284 @@
+/*
+ * Intel SST loader on ACPI systems
+ *
+ * Copyright (C) 2013, Intel Corporation. All rights reserved.
+ *
+ * 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.
+ *
+ */
+
+#include <linux/acpi.h>
+#include <linux/device.h>
+#include <linux/firmware.h>
+#include <linux/module.h>
+#include <linux/platform_device.h>
+
+#include "sst-dsp.h"
+
+#define SST_LPT_DSP_DMA_ADDR_OFFSET 0x0F0000
+#define SST_WPT_DSP_DMA_ADDR_OFFSET 0x0FE000
+#define SST_LPT_DSP_DMA_SIZE (1024 - 1)
+
+/* Descriptor for SST ASoC machine driver */
+struct sst_acpi_mach {
+ /* ACPI ID for the matching machine driver. Audio codec for instance */
+ const u8 id[ACPI_ID_LEN];
+ /* machine driver name */
+ const char *drv_name;
+ /* firmware file name */
+ const char *fw_filename;
+};
+
+/* Descriptor for setting up SST platform data */
+struct sst_acpi_desc {
+ const char *drv_name;
+ struct sst_acpi_mach *machines;
+ /* Platform resource indexes. Must set to -1 if not used */
+ int resindex_lpe_base;
+ int resindex_pcicfg_base;
+ int resindex_fw_base;
+ int irqindex_host_ipc;
+ int resindex_dma_base;
+ /* Unique number identifying the SST core on platform */
+ int sst_id;
+ /* DMA only valid when resindex_dma_base != -1*/
+ int dma_engine;
+ int dma_size;
+};
+
+struct sst_acpi_priv {
+ struct platform_device *pdev_mach;
+ struct platform_device *pdev_pcm;
+ struct sst_pdata sst_pdata;
+ struct sst_acpi_desc *desc;
+ struct sst_acpi_mach *mach;
+};
+
+static void sst_acpi_fw_cb(const struct firmware *fw, void *context)
+{
+ struct platform_device *pdev = context;
+ struct device *dev = &pdev->dev;
+ struct sst_acpi_priv *sst_acpi = platform_get_drvdata(pdev);
+ struct sst_pdata *sst_pdata = &sst_acpi->sst_pdata;
+ struct sst_acpi_desc *desc = sst_acpi->desc;
+ struct sst_acpi_mach *mach = sst_acpi->mach;
+
+ sst_pdata->fw = fw;
+ if (!fw) {
+ dev_err(dev, "Cannot load firmware %s\n", mach->fw_filename);
+ return;
+ }
+
+ /* register PCM and DAI driver */
+ sst_acpi->pdev_pcm =
+ platform_device_register_data(dev, desc->drv_name, -1,
+ sst_pdata, sizeof(*sst_pdata));
+ if (IS_ERR(sst_acpi->pdev_pcm)) {
+ dev_err(dev, "Cannot register device %s. Error %d\n",
+ desc->drv_name, (int)PTR_ERR(sst_acpi->pdev_pcm));
+ }
+
+ return;
+}
+
+static acpi_status sst_acpi_mach_match(acpi_handle handle, u32 level,
+ void *context, void **ret)
+{
+ *(bool *)context = true;
+ return AE_OK;
+}
+
+static struct sst_acpi_mach *sst_acpi_find_machine(
+ struct sst_acpi_mach *machines)
+{
+ struct sst_acpi_mach *mach;
+ bool found = false;
+
+ for (mach = machines; mach->id[0]; mach++)
+ if (ACPI_SUCCESS(acpi_get_devices(mach->id,
+ sst_acpi_mach_match,
+ &found, NULL)) && found)
+ return mach;
+
+ return NULL;
+}
+
+static int sst_acpi_probe(struct platform_device *pdev)
+{
+ const struct acpi_device_id *id;
+ struct device *dev = &pdev->dev;
+ struct sst_acpi_priv *sst_acpi;
+ struct sst_pdata *sst_pdata;
+ struct sst_acpi_mach *mach;
+ struct sst_acpi_desc *desc;
+ struct resource *mmio;
+ int ret = 0;
+
+ sst_acpi = devm_kzalloc(dev, sizeof(*sst_acpi), GFP_KERNEL);
+ if (sst_acpi == NULL)
+ return -ENOMEM;
+
+ id = acpi_match_device(dev->driver->acpi_match_table, dev);
+ if (!id)
+ return -ENODEV;
+
+ desc = (struct sst_acpi_desc *)id->driver_data;
+ mach = sst_acpi_find_machine(desc->machines);
+ if (mach == NULL) {
+ dev_err(dev, "No matching ASoC machine driver found\n");
+ return -ENODEV;
+ }
+
+ sst_pdata = &sst_acpi->sst_pdata;
+ sst_pdata->id = desc->sst_id;
+ sst_acpi->desc = desc;
+ sst_acpi->mach = mach;
+
+ if (desc->resindex_dma_base >= 0) {
+ sst_pdata->dma_engine = desc->dma_engine;
+ sst_pdata->dma_base = desc->resindex_dma_base;
+ sst_pdata->dma_size = desc->dma_size;
+ }
+
+ if (desc->irqindex_host_ipc >= 0)
+ sst_pdata->irq = platform_get_irq(pdev, desc->irqindex_host_ipc);
+
+ if (desc->resindex_lpe_base >= 0) {
+ mmio = platform_get_resource(pdev, IORESOURCE_MEM,
+ desc->resindex_lpe_base);
+ if (mmio) {
+ sst_pdata->lpe_base = mmio->start;
+ sst_pdata->lpe_size = resource_size(mmio);
+ }
+ }
+
+ if (desc->resindex_pcicfg_base >= 0) {
+ mmio = platform_get_resource(pdev, IORESOURCE_MEM,
+ desc->resindex_pcicfg_base);
+ if (mmio) {
+ sst_pdata->pcicfg_base = mmio->start;
+ sst_pdata->pcicfg_size = resource_size(mmio);
+ }
+ }
+
+ if (desc->resindex_fw_base >= 0) {
+ mmio = platform_get_resource(pdev, IORESOURCE_MEM,
+ desc->resindex_fw_base);
+ if (mmio) {
+ sst_pdata->fw_base = mmio->start;
+ sst_pdata->fw_size = resource_size(mmio);
+ }
+ }
+
+ platform_set_drvdata(pdev, sst_acpi);
+
+ /* register machine driver */
+ sst_acpi->pdev_mach =
+ platform_device_register_data(dev, mach->drv_name, -1,
+ sst_pdata, sizeof(*sst_pdata));
+ if (IS_ERR(sst_acpi->pdev_mach))
+ return PTR_ERR(sst_acpi->pdev_mach);
+
+ /* continue SST probing after firmware is loaded */
+ ret = request_firmware_nowait(THIS_MODULE, true, mach->fw_filename,
+ dev, GFP_KERNEL, pdev, sst_acpi_fw_cb);
+ if (ret)
+ platform_device_unregister(sst_acpi->pdev_mach);
+
+ return ret;
+}
+
+static int sst_acpi_remove(struct platform_device *pdev)
+{
+ struct sst_acpi_priv *sst_acpi = platform_get_drvdata(pdev);
+ struct sst_pdata *sst_pdata = &sst_acpi->sst_pdata;
+
+ platform_device_unregister(sst_acpi->pdev_mach);
+ if (!IS_ERR_OR_NULL(sst_acpi->pdev_pcm))
+ platform_device_unregister(sst_acpi->pdev_pcm);
+ release_firmware(sst_pdata->fw);
+
+ return 0;
+}
+
+static struct sst_acpi_mach haswell_machines[] = {
+ { "INT33CA", "haswell-audio", "intel/IntcSST1.bin" },
+ {}
+};
+
+static struct sst_acpi_desc sst_acpi_haswell_desc = {
+ .drv_name = "haswell-pcm-audio",
+ .machines = haswell_machines,
+ .resindex_lpe_base = 0,
+ .resindex_pcicfg_base = 1,
+ .resindex_fw_base = -1,
+ .irqindex_host_ipc = 0,
+ .sst_id = SST_DEV_ID_LYNX_POINT,
+ .dma_engine = SST_DMA_TYPE_DW,
+ .resindex_dma_base = SST_LPT_DSP_DMA_ADDR_OFFSET,
+ .dma_size = SST_LPT_DSP_DMA_SIZE,
+};
+
+static struct sst_acpi_mach broadwell_machines[] = {
+ { "INT343A", "broadwell-audio", "intel/IntcSST2.bin" },
+ {}
+};
+
+static struct sst_acpi_desc sst_acpi_broadwell_desc = {
+ .drv_name = "haswell-pcm-audio",
+ .machines = broadwell_machines,
+ .resindex_lpe_base = 0,
+ .resindex_pcicfg_base = 1,
+ .resindex_fw_base = -1,
+ .irqindex_host_ipc = 0,
+ .sst_id = SST_DEV_ID_WILDCAT_POINT,
+ .dma_engine = SST_DMA_TYPE_DW,
+ .resindex_dma_base = SST_WPT_DSP_DMA_ADDR_OFFSET,
+ .dma_size = SST_LPT_DSP_DMA_SIZE,
+};
+
+static struct sst_acpi_mach baytrail_machines[] = {
+ { "10EC5640", "byt-rt5640", "intel/fw_sst_0f28.bin-i2s_master" },
+ {}
+};
+
+static struct sst_acpi_desc sst_acpi_baytrail_desc = {
+ .drv_name = "baytrail-pcm-audio",
+ .machines = baytrail_machines,
+ .resindex_lpe_base = 0,
+ .resindex_pcicfg_base = 1,
+ .resindex_fw_base = 2,
+ .irqindex_host_ipc = 5,
+ .sst_id = SST_DEV_ID_BYT,
+ .resindex_dma_base = -1,
+};
+
+static struct acpi_device_id sst_acpi_match[] = {
+ { "INT33C8", (unsigned long)&sst_acpi_haswell_desc },
+ { "INT3438", (unsigned long)&sst_acpi_broadwell_desc },
+ { "80860F28", (unsigned long)&sst_acpi_baytrail_desc },
+ { }
+};
+MODULE_DEVICE_TABLE(acpi, sst_acpi_match);
+
+static struct platform_driver sst_acpi_driver = {
+ .probe = sst_acpi_probe,
+ .remove = sst_acpi_remove,
+ .driver = {
+ .name = "sst-acpi",
+ .owner = THIS_MODULE,
+ .acpi_match_table = ACPI_PTR(sst_acpi_match),
+ },
+};
+module_platform_driver(sst_acpi_driver);
+
+MODULE_AUTHOR("Jarkko Nikula <jarkko.nikula@linux.intel.com>");
+MODULE_DESCRIPTION("Intel SST loader on ACPI systems");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/intel/sst-baytrail-dsp.c b/sound/soc/intel/sst-baytrail-dsp.c
new file mode 100644
index 0000000..a50bf7f
--- /dev/null
+++ b/sound/soc/intel/sst-baytrail-dsp.c
@@ -0,0 +1,372 @@
+/*
+ * Intel Baytrail SST DSP driver
+ * Copyright (c) 2014, Intel Corporation.
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms and conditions of the GNU General Public License,
+ * version 2, as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope 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.
+ */
+
+#include <linux/delay.h>
+#include <linux/fs.h>
+#include <linux/slab.h>
+#include <linux/device.h>
+#include <linux/interrupt.h>
+#include <linux/module.h>
+#include <linux/dma-mapping.h>
+#include <linux/platform_device.h>
+#include <linux/firmware.h>
+
+#include "sst-dsp.h"
+#include "sst-dsp-priv.h"
+#include "sst-baytrail-ipc.h"
+
+#define SST_BYT_FW_SIGNATURE_SIZE 4
+#define SST_BYT_FW_SIGN "$SST"
+
+#define SST_BYT_IRAM_OFFSET 0xC0000
+#define SST_BYT_DRAM_OFFSET 0x100000
+#define SST_BYT_SHIM_OFFSET 0x140000
+
+enum sst_ram_type {
+ SST_BYT_IRAM = 1,
+ SST_BYT_DRAM = 2,
+ SST_BYT_CACHE = 3,
+};
+
+struct dma_block_info {
+ enum sst_ram_type type; /* IRAM/DRAM */
+ u32 size; /* Bytes */
+ u32 ram_offset; /* Offset in I/DRAM */
+ u32 rsvd; /* Reserved field */
+};
+
+struct fw_header {
+ unsigned char signature[SST_BYT_FW_SIGNATURE_SIZE];
+ u32 file_size; /* size of fw minus this header */
+ u32 modules; /* # of modules */
+ u32 file_format; /* version of header format */
+ u32 reserved[4];
+};
+
+struct sst_byt_fw_module_header {
+ unsigned char signature[SST_BYT_FW_SIGNATURE_SIZE];
+ u32 mod_size; /* size of module */
+ u32 blocks; /* # of blocks */
+ u32 type; /* codec type, pp lib */
+ u32 entry_point;
+};
+
+static int sst_byt_parse_module(struct sst_dsp *dsp, struct sst_fw *fw,
+ struct sst_byt_fw_module_header *module)
+{
+ struct dma_block_info *block;
+ struct sst_module *mod;
+ struct sst_module_data block_data;
+ struct sst_module_template template;
+ int count;
+
+ memset(&template, 0, sizeof(template));
+ template.id = module->type;
+ template.entry = module->entry_point;
+ template.p.type = SST_MEM_DRAM;
+ template.p.data_type = SST_DATA_P;
+ template.s.type = SST_MEM_DRAM;
+ template.s.data_type = SST_DATA_S;
+
+ mod = sst_module_new(fw, &template, NULL);
+ if (mod == NULL)
+ return -ENOMEM;
+
+ block = (void *)module + sizeof(*module);
+
+ for (count = 0; count < module->blocks; count++) {
+
+ if (block->size <= 0) {
+ dev_err(dsp->dev, "block %d size invalid\n", count);
+ return -EINVAL;
+ }
+
+ switch (block->type) {
+ case SST_BYT_IRAM:
+ block_data.offset = block->ram_offset +
+ dsp->addr.iram_offset;
+ block_data.type = SST_MEM_IRAM;
+ break;
+ case SST_BYT_DRAM:
+ block_data.offset = block->ram_offset +
+ dsp->addr.dram_offset;
+ block_data.type = SST_MEM_DRAM;
+ break;
+ case SST_BYT_CACHE:
+ block_data.offset = block->ram_offset +
+ (dsp->addr.fw_ext - dsp->addr.lpe);
+ block_data.type = SST_MEM_CACHE;
+ break;
+ default:
+ dev_err(dsp->dev, "wrong ram type 0x%x in block0x%x\n",
+ block->type, count);
+ return -EINVAL;
+ }
+
+ block_data.size = block->size;
+ block_data.data_type = SST_DATA_M;
+ block_data.data = (void *)block + sizeof(*block);
+
+ sst_module_insert_fixed_block(mod, &block_data);
+
+ block = (void *)block + sizeof(*block) + block->size;
+ }
+ return 0;
+}
+
+static int sst_byt_parse_fw_image(struct sst_fw *sst_fw)
+{
+ struct fw_header *header;
+ struct sst_byt_fw_module_header *module;
+ struct sst_dsp *dsp = sst_fw->dsp;
+ int ret, count;
+
+ /* Read the header information from the data pointer */
+ header = (struct fw_header *)sst_fw->dma_buf;
+
+ /* verify FW */
+ if ((strncmp(header->signature, SST_BYT_FW_SIGN, 4) != 0) ||
+ (sst_fw->size != header->file_size + sizeof(*header))) {
+ /* Invalid FW signature */
+ dev_err(dsp->dev, "Invalid FW sign/filesize mismatch\n");
+ return -EINVAL;
+ }
+
+ dev_dbg(dsp->dev,
+ "header sign=%4s size=0x%x modules=0x%x fmt=0x%x size=%zu\n",
+ header->signature, header->file_size, header->modules,
+ header->file_format, sizeof(*header));
+
+ module = (void *)sst_fw->dma_buf + sizeof(*header);
+ for (count = 0; count < header->modules; count++) {
+ /* module */
+ ret = sst_byt_parse_module(dsp, sst_fw, module);
+ if (ret < 0) {
+ dev_err(dsp->dev, "invalid module %d\n", count);
+ return ret;
+ }
+ module = (void *)module + sizeof(*module) + module->mod_size;
+ }
+
+ return 0;
+}
+
+static void sst_byt_dump_shim(struct sst_dsp *sst)
+{
+ int i;
+ u64 reg;
+
+ for (i = 0; i <= 0xF0; i += 8) {
+ reg = sst_dsp_shim_read64_unlocked(sst, i);
+ if (reg)
+ dev_dbg(sst->dev, "shim 0x%2.2x value 0x%16.16llx\n",
+ i, reg);
+ }
+
+ for (i = 0x00; i <= 0xff; i += 4) {
+ reg = readl(sst->addr.pci_cfg + i);
+ if (reg)
+ dev_dbg(sst->dev, "pci 0x%2.2x value 0x%8.8x\n",
+ i, (u32)reg);
+ }
+}
+
+static irqreturn_t sst_byt_irq(int irq, void *context)
+{
+ struct sst_dsp *sst = (struct sst_dsp *) context;
+ u64 isrx;
+ irqreturn_t ret = IRQ_NONE;
+
+ spin_lock(&sst->spinlock);
+
+ isrx = sst_dsp_shim_read64_unlocked(sst, SST_ISRX);
+ if (isrx & SST_ISRX_DONE) {
+ /* ADSP has processed the message request from IA */
+ sst_dsp_shim_update_bits64_unlocked(sst, SST_IPCX,
+ SST_BYT_IPCX_DONE, 0);
+ ret = IRQ_WAKE_THREAD;
+ }
+ if (isrx & SST_BYT_ISRX_REQUEST) {
+ /* mask message request from ADSP and do processing later */
+ sst_dsp_shim_update_bits64_unlocked(sst, SST_IMRX,
+ SST_BYT_IMRX_REQUEST,
+ SST_BYT_IMRX_REQUEST);
+ ret = IRQ_WAKE_THREAD;
+ }
+
+ spin_unlock(&sst->spinlock);
+
+ return ret;
+}
+
+static void sst_byt_boot(struct sst_dsp *sst)
+{
+ int tries = 10;
+
+ /* release stall and wait to unstall */
+ sst_dsp_shim_update_bits64(sst, SST_CSR, SST_BYT_CSR_STALL, 0x0);
+ while (tries--) {
+ if (!(sst_dsp_shim_read64(sst, SST_CSR) &
+ SST_BYT_CSR_PWAITMODE))
+ break;
+ msleep(100);
+ }
+ if (tries < 0) {
+ dev_err(sst->dev, "unable to start DSP\n");
+ sst_byt_dump_shim(sst);
+ }
+}
+
+static void sst_byt_reset(struct sst_dsp *sst)
+{
+ /* put DSP into reset, set reset vector and stall */
+ sst_dsp_shim_update_bits64(sst, SST_CSR,
+ SST_BYT_CSR_RST | SST_BYT_CSR_VECTOR_SEL | SST_BYT_CSR_STALL,
+ SST_BYT_CSR_RST | SST_BYT_CSR_VECTOR_SEL | SST_BYT_CSR_STALL);
+
+ udelay(10);
+
+ /* take DSP out of reset and keep stalled for FW loading */
+ sst_dsp_shim_update_bits64(sst, SST_CSR, SST_BYT_CSR_RST, 0);
+}
+
+struct sst_adsp_memregion {
+ u32 start;
+ u32 end;
+ int blocks;
+ enum sst_mem_type type;
+};
+
+/* BYT test stuff */
+static const struct sst_adsp_memregion byt_region[] = {
+ {0xC0000, 0x100000, 8, SST_MEM_IRAM}, /* I-SRAM - 8 * 32kB */
+ {0x100000, 0x140000, 8, SST_MEM_DRAM}, /* D-SRAM0 - 8 * 32kB */
+};
+
+static int sst_byt_resource_map(struct sst_dsp *sst, struct sst_pdata *pdata)
+{
+ sst->addr.lpe_base = pdata->lpe_base;
+ sst->addr.lpe = ioremap(pdata->lpe_base, pdata->lpe_size);
+ if (!sst->addr.lpe)
+ return -ENODEV;
+
+ /* ADSP PCI MMIO config space */
+ sst->addr.pci_cfg = ioremap(pdata->pcicfg_base, pdata->pcicfg_size);
+ if (!sst->addr.pci_cfg) {
+ iounmap(sst->addr.lpe);
+ return -ENODEV;
+ }
+
+ /* SST Extended FW allocation */
+ sst->addr.fw_ext = ioremap(pdata->fw_base, pdata->fw_size);
+ if (!sst->addr.fw_ext) {
+ iounmap(sst->addr.pci_cfg);
+ iounmap(sst->addr.lpe);
+ return -ENODEV;
+ }
+
+ /* SST Shim */
+ sst->addr.shim = sst->addr.lpe + sst->addr.shim_offset;
+
+ sst_dsp_mailbox_init(sst, SST_BYT_MAILBOX_OFFSET + 0x204,
+ SST_BYT_IPC_MAX_PAYLOAD_SIZE,
+ SST_BYT_MAILBOX_OFFSET,
+ SST_BYT_IPC_MAX_PAYLOAD_SIZE);
+
+ sst->irq = pdata->irq;
+
+ return 0;
+}
+
+static int sst_byt_init(struct sst_dsp *sst, struct sst_pdata *pdata)
+{
+ const struct sst_adsp_memregion *region;
+ struct device *dev;
+ int ret = -ENODEV, i, j, region_count;
+ u32 offset, size;
+
+ dev = sst->dev;
+
+ switch (sst->id) {
+ case SST_DEV_ID_BYT:
+ region = byt_region;
+ region_count = ARRAY_SIZE(byt_region);
+ sst->addr.iram_offset = SST_BYT_IRAM_OFFSET;
+ sst->addr.dram_offset = SST_BYT_DRAM_OFFSET;
+ sst->addr.shim_offset = SST_BYT_SHIM_OFFSET;
+ break;
+ default:
+ dev_err(dev, "failed to get mem resources\n");
+ return ret;
+ }
+
+ ret = sst_byt_resource_map(sst, pdata);
+ if (ret < 0) {
+ dev_err(dev, "failed to map resources\n");
+ return ret;
+ }
+
+ /*
+ * save the physical address of extended firmware block in the first
+ * 4 bytes of the mailbox
+ */
+ memcpy_toio(sst->addr.lpe + SST_BYT_MAILBOX_OFFSET,
+ &pdata->fw_base, sizeof(u32));
+
+ ret = dma_coerce_mask_and_coherent(dev, DMA_BIT_MASK(32));
+ if (ret)
+ return ret;
+
+ /* enable Interrupt from both sides */
+ sst_dsp_shim_update_bits64(sst, SST_IMRX, 0x3, 0x0);
+ sst_dsp_shim_update_bits64(sst, SST_IMRD, 0x3, 0x0);
+
+ /* register DSP memory blocks - ideally we should get this from ACPI */
+ for (i = 0; i < region_count; i++) {
+ offset = region[i].start;
+ size = (region[i].end - region[i].start) / region[i].blocks;
+
+ /* register individual memory blocks */
+ for (j = 0; j < region[i].blocks; j++) {
+ sst_mem_block_register(sst, offset, size,
+ region[i].type, NULL, j, sst);
+ offset += size;
+ }
+ }
+
+ return 0;
+}
+
+static void sst_byt_free(struct sst_dsp *sst)
+{
+ sst_mem_block_unregister_all(sst);
+ iounmap(sst->addr.lpe);
+ iounmap(sst->addr.pci_cfg);
+ iounmap(sst->addr.fw_ext);
+}
+
+struct sst_ops sst_byt_ops = {
+ .reset = sst_byt_reset,
+ .boot = sst_byt_boot,
+ .write = sst_shim32_write,
+ .read = sst_shim32_read,
+ .write64 = sst_shim32_write64,
+ .read64 = sst_shim32_read64,
+ .ram_read = sst_memcpy_fromio_32,
+ .ram_write = sst_memcpy_toio_32,
+ .irq_handler = sst_byt_irq,
+ .init = sst_byt_init,
+ .free = sst_byt_free,
+ .parse_fw = sst_byt_parse_fw_image,
+};
diff --git a/sound/soc/intel/sst-baytrail-ipc.c b/sound/soc/intel/sst-baytrail-ipc.c
new file mode 100644
index 0000000..d0eaeee
--- /dev/null
+++ b/sound/soc/intel/sst-baytrail-ipc.c
@@ -0,0 +1,867 @@
+/*
+ * Intel Baytrail SST IPC Support
+ * Copyright (c) 2014, Intel Corporation.
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms and conditions of the GNU General Public License,
+ * version 2, as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope 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.
+ */
+
+#include <linux/types.h>
+#include <linux/kernel.h>
+#include <linux/list.h>
+#include <linux/device.h>
+#include <linux/wait.h>
+#include <linux/spinlock.h>
+#include <linux/workqueue.h>
+#include <linux/export.h>
+#include <linux/slab.h>
+#include <linux/delay.h>
+#include <linux/list.h>
+#include <linux/platform_device.h>
+#include <linux/kthread.h>
+#include <linux/firmware.h>
+#include <linux/io.h>
+#include <asm/div64.h>
+
+#include "sst-baytrail-ipc.h"
+#include "sst-dsp.h"
+#include "sst-dsp-priv.h"
+
+/* IPC message timeout */
+#define IPC_TIMEOUT_MSECS 300
+#define IPC_BOOT_MSECS 200
+
+#define IPC_EMPTY_LIST_SIZE 8
+
+/* IPC header bits */
+#define IPC_HEADER_MSG_ID_MASK 0xff
+#define IPC_HEADER_MSG_ID(x) ((x) & IPC_HEADER_MSG_ID_MASK)
+#define IPC_HEADER_STR_ID_SHIFT 8
+#define IPC_HEADER_STR_ID_MASK 0x1f
+#define IPC_HEADER_STR_ID(x) (((x) & 0x1f) << IPC_HEADER_STR_ID_SHIFT)
+#define IPC_HEADER_LARGE_SHIFT 13
+#define IPC_HEADER_LARGE(x) (((x) & 0x1) << IPC_HEADER_LARGE_SHIFT)
+#define IPC_HEADER_DATA_SHIFT 16
+#define IPC_HEADER_DATA_MASK 0x3fff
+#define IPC_HEADER_DATA(x) (((x) & 0x3fff) << IPC_HEADER_DATA_SHIFT)
+
+/* mask for differentiating between notification and reply message */
+#define IPC_NOTIFICATION (0x1 << 7)
+
+/* I2L Stream config/control msgs */
+#define IPC_IA_ALLOC_STREAM 0x20
+#define IPC_IA_FREE_STREAM 0x21
+#define IPC_IA_PAUSE_STREAM 0x24
+#define IPC_IA_RESUME_STREAM 0x25
+#define IPC_IA_DROP_STREAM 0x26
+#define IPC_IA_START_STREAM 0x30
+
+/* notification messages */
+#define IPC_IA_FW_INIT_CMPLT 0x81
+#define IPC_SST_PERIOD_ELAPSED 0x97
+
+/* IPC messages between host and ADSP */
+struct sst_byt_address_info {
+ u32 addr;
+ u32 size;
+} __packed;
+
+struct sst_byt_str_type {
+ u8 codec_type;
+ u8 str_type;
+ u8 operation;
+ u8 protected_str;
+ u8 time_slots;
+ u8 reserved;
+ u16 result;
+} __packed;
+
+struct sst_byt_pcm_params {
+ u8 num_chan;
+ u8 pcm_wd_sz;
+ u8 use_offload_path;
+ u8 reserved;
+ u32 sfreq;
+ u8 channel_map[8];
+} __packed;
+
+struct sst_byt_frames_info {
+ u16 num_entries;
+ u16 rsrvd;
+ u32 frag_size;
+ struct sst_byt_address_info ring_buf_info[8];
+} __packed;
+
+struct sst_byt_alloc_params {
+ struct sst_byt_str_type str_type;
+ struct sst_byt_pcm_params pcm_params;
+ struct sst_byt_frames_info frame_info;
+} __packed;
+
+struct sst_byt_alloc_response {
+ struct sst_byt_str_type str_type;
+ u8 reserved[88];
+} __packed;
+
+struct sst_byt_start_stream_params {
+ u32 byte_offset;
+} __packed;
+
+struct sst_byt_tstamp {
+ u64 ring_buffer_counter;
+ u64 hardware_counter;
+ u64 frames_decoded;
+ u64 bytes_decoded;
+ u64 bytes_copied;
+ u32 sampling_frequency;
+ u32 channel_peak[8];
+} __packed;
+
+/* driver internal IPC message structure */
+struct ipc_message {
+ struct list_head list;
+ u64 header;
+
+ /* direction wrt host CPU */
+ char tx_data[SST_BYT_IPC_MAX_PAYLOAD_SIZE];
+ size_t tx_size;
+ char rx_data[SST_BYT_IPC_MAX_PAYLOAD_SIZE];
+ size_t rx_size;
+
+ wait_queue_head_t waitq;
+ bool complete;
+ bool wait;
+ int errno;
+};
+
+struct sst_byt_stream;
+struct sst_byt;
+
+/* stream infomation */
+struct sst_byt_stream {
+ struct list_head node;
+
+ /* configuration */
+ struct sst_byt_alloc_params request;
+ struct sst_byt_alloc_response reply;
+
+ /* runtime info */
+ struct sst_byt *byt;
+ int str_id;
+ bool commited;
+ bool running;
+
+ /* driver callback */
+ u32 (*notify_position)(struct sst_byt_stream *stream, void *data);
+ void *pdata;
+};
+
+/* SST Baytrail IPC data */
+struct sst_byt {
+ struct device *dev;
+ struct sst_dsp *dsp;
+
+ /* stream */
+ struct list_head stream_list;
+
+ /* boot */
+ wait_queue_head_t boot_wait;
+ bool boot_complete;
+
+ /* IPC messaging */
+ struct list_head tx_list;
+ struct list_head rx_list;
+ struct list_head empty_list;
+ wait_queue_head_t wait_txq;
+ struct task_struct *tx_thread;
+ struct kthread_worker kworker;
+ struct kthread_work kwork;
+ struct ipc_message *msg;
+};
+
+static inline u64 sst_byt_header(int msg_id, int data, bool large, int str_id)
+{
+ u64 header;
+
+ header = IPC_HEADER_MSG_ID(msg_id) |
+ IPC_HEADER_STR_ID(str_id) |
+ IPC_HEADER_LARGE(large) |
+ IPC_HEADER_DATA(data) |
+ SST_BYT_IPCX_BUSY;
+
+ return header;
+}
+
+static inline u16 sst_byt_header_msg_id(u64 header)
+{
+ return header & IPC_HEADER_MSG_ID_MASK;
+}
+
+static inline u8 sst_byt_header_str_id(u64 header)
+{
+ return (header >> IPC_HEADER_STR_ID_SHIFT) & IPC_HEADER_STR_ID_MASK;
+}
+
+static inline u16 sst_byt_header_data(u64 header)
+{
+ return (header >> IPC_HEADER_DATA_SHIFT) & IPC_HEADER_DATA_MASK;
+}
+
+static struct sst_byt_stream *sst_byt_get_stream(struct sst_byt *byt,
+ int stream_id)
+{
+ struct sst_byt_stream *stream;
+
+ list_for_each_entry(stream, &byt->stream_list, node) {
+ if (stream->str_id == stream_id)
+ return stream;
+ }
+
+ return NULL;
+}
+
+static void sst_byt_ipc_shim_dbg(struct sst_byt *byt, const char *text)
+{
+ struct sst_dsp *sst = byt->dsp;
+ u64 isr, ipcd, imrx, ipcx;
+
+ ipcx = sst_dsp_shim_read64_unlocked(sst, SST_IPCX);
+ isr = sst_dsp_shim_read64_unlocked(sst, SST_ISRX);
+ ipcd = sst_dsp_shim_read64_unlocked(sst, SST_IPCD);
+ imrx = sst_dsp_shim_read64_unlocked(sst, SST_IMRX);
+
+ dev_err(byt->dev,
+ "ipc: --%s-- ipcx 0x%llx isr 0x%llx ipcd 0x%llx imrx 0x%llx\n",
+ text, ipcx, isr, ipcd, imrx);
+}
+
+/* locks held by caller */
+static struct ipc_message *sst_byt_msg_get_empty(struct sst_byt *byt)
+{
+ struct ipc_message *msg = NULL;
+
+ if (!list_empty(&byt->empty_list)) {
+ msg = list_first_entry(&byt->empty_list,
+ struct ipc_message, list);
+ list_del(&msg->list);
+ }
+
+ return msg;
+}
+
+static void sst_byt_ipc_tx_msgs(struct kthread_work *work)
+{
+ struct sst_byt *byt =
+ container_of(work, struct sst_byt, kwork);
+ struct ipc_message *msg;
+ u64 ipcx;
+ unsigned long flags;
+
+ spin_lock_irqsave(&byt->dsp->spinlock, flags);
+ if (list_empty(&byt->tx_list)) {
+ spin_unlock_irqrestore(&byt->dsp->spinlock, flags);
+ return;
+ }
+
+ /* if the DSP is busy we will TX messages after IRQ */
+ ipcx = sst_dsp_shim_read64_unlocked(byt->dsp, SST_IPCX);
+ if (ipcx & SST_BYT_IPCX_BUSY) {
+ spin_unlock_irqrestore(&byt->dsp->spinlock, flags);
+ return;
+ }
+
+ msg = list_first_entry(&byt->tx_list, struct ipc_message, list);
+
+ list_move(&msg->list, &byt->rx_list);
+
+ /* send the message */
+ if (msg->header & IPC_HEADER_LARGE(true))
+ sst_dsp_outbox_write(byt->dsp, msg->tx_data, msg->tx_size);
+ sst_dsp_shim_write64_unlocked(byt->dsp, SST_IPCX, msg->header);
+
+ spin_unlock_irqrestore(&byt->dsp->spinlock, flags);
+}
+
+static inline void sst_byt_tx_msg_reply_complete(struct sst_byt *byt,
+ struct ipc_message *msg)
+{
+ msg->complete = true;
+
+ if (!msg->wait)
+ list_add_tail(&msg->list, &byt->empty_list);
+ else
+ wake_up(&msg->waitq);
+}
+
+static int sst_byt_tx_wait_done(struct sst_byt *byt, struct ipc_message *msg,
+ void *rx_data)
+{
+ unsigned long flags;
+ int ret;
+
+ /* wait for DSP completion */
+ ret = wait_event_timeout(msg->waitq, msg->complete,
+ msecs_to_jiffies(IPC_TIMEOUT_MSECS));
+
+ spin_lock_irqsave(&byt->dsp->spinlock, flags);
+ if (ret == 0) {
+ list_del(&msg->list);
+ sst_byt_ipc_shim_dbg(byt, "message timeout");
+
+ ret = -ETIMEDOUT;
+ } else {
+
+ /* copy the data returned from DSP */
+ if (msg->rx_size)
+ memcpy(rx_data, msg->rx_data, msg->rx_size);
+ ret = msg->errno;
+ }
+
+ list_add_tail(&msg->list, &byt->empty_list);
+ spin_unlock_irqrestore(&byt->dsp->spinlock, flags);
+ return ret;
+}
+
+static int sst_byt_ipc_tx_message(struct sst_byt *byt, u64 header,
+ void *tx_data, size_t tx_bytes,
+ void *rx_data, size_t rx_bytes, int wait)
+{
+ unsigned long flags;
+ struct ipc_message *msg;
+
+ spin_lock_irqsave(&byt->dsp->spinlock, flags);
+
+ msg = sst_byt_msg_get_empty(byt);
+ if (msg == NULL) {
+ spin_unlock_irqrestore(&byt->dsp->spinlock, flags);
+ return -EBUSY;
+ }
+
+ msg->header = header;
+ msg->tx_size = tx_bytes;
+ msg->rx_size = rx_bytes;
+ msg->wait = wait;
+ msg->errno = 0;
+ msg->complete = false;
+
+ if (tx_bytes) {
+ /* msg content = lower 32-bit of the header + data */
+ *(u32 *)msg->tx_data = (u32)(header & (u32)-1);
+ memcpy(msg->tx_data + sizeof(u32), tx_data, tx_bytes);
+ msg->tx_size += sizeof(u32);
+ }
+
+ list_add_tail(&msg->list, &byt->tx_list);
+ spin_unlock_irqrestore(&byt->dsp->spinlock, flags);
+
+ queue_kthread_work(&byt->kworker, &byt->kwork);
+
+ if (wait)
+ return sst_byt_tx_wait_done(byt, msg, rx_data);
+ else
+ return 0;
+}
+
+static inline int sst_byt_ipc_tx_msg_wait(struct sst_byt *byt, u64 header,
+ void *tx_data, size_t tx_bytes,
+ void *rx_data, size_t rx_bytes)
+{
+ return sst_byt_ipc_tx_message(byt, header, tx_data, tx_bytes,
+ rx_data, rx_bytes, 1);
+}
+
+static inline int sst_byt_ipc_tx_msg_nowait(struct sst_byt *byt, u64 header,
+ void *tx_data, size_t tx_bytes)
+{
+ return sst_byt_ipc_tx_message(byt, header, tx_data, tx_bytes,
+ NULL, 0, 0);
+}
+
+static struct ipc_message *sst_byt_reply_find_msg(struct sst_byt *byt,
+ u64 header)
+{
+ struct ipc_message *msg = NULL, *_msg;
+ u64 mask;
+
+ /* match reply to message sent based on msg and stream IDs */
+ mask = IPC_HEADER_MSG_ID_MASK |
+ IPC_HEADER_STR_ID_MASK << IPC_HEADER_STR_ID_SHIFT;
+ header &= mask;
+
+ if (list_empty(&byt->rx_list)) {
+ dev_err(byt->dev,
+ "ipc: rx list is empty but received 0x%llx\n", header);
+ goto out;
+ }
+
+ list_for_each_entry(_msg, &byt->rx_list, list) {
+ if ((_msg->header & mask) == header) {
+ msg = _msg;
+ break;
+ }
+ }
+
+out:
+ return msg;
+}
+
+static void sst_byt_stream_update(struct sst_byt *byt, struct ipc_message *msg)
+{
+ struct sst_byt_stream *stream;
+ u64 header = msg->header;
+ u8 stream_id = sst_byt_header_str_id(header);
+ u8 stream_msg = sst_byt_header_msg_id(header);
+
+ stream = sst_byt_get_stream(byt, stream_id);
+ if (stream == NULL)
+ return;
+
+ switch (stream_msg) {
+ case IPC_IA_DROP_STREAM:
+ case IPC_IA_PAUSE_STREAM:
+ case IPC_IA_FREE_STREAM:
+ stream->running = false;
+ break;
+ case IPC_IA_START_STREAM:
+ case IPC_IA_RESUME_STREAM:
+ stream->running = true;
+ break;
+ }
+}
+
+static int sst_byt_process_reply(struct sst_byt *byt, u64 header)
+{
+ struct ipc_message *msg;
+
+ msg = sst_byt_reply_find_msg(byt, header);
+ if (msg == NULL)
+ return 1;
+
+ if (header & IPC_HEADER_LARGE(true)) {
+ msg->rx_size = sst_byt_header_data(header);
+ sst_dsp_inbox_read(byt->dsp, msg->rx_data, msg->rx_size);
+ }
+
+ /* update any stream states */
+ sst_byt_stream_update(byt, msg);
+
+ list_del(&msg->list);
+ /* wake up */
+ sst_byt_tx_msg_reply_complete(byt, msg);
+
+ return 1;
+}
+
+static void sst_byt_fw_ready(struct sst_byt *byt, u64 header)
+{
+ dev_dbg(byt->dev, "ipc: DSP is ready 0x%llX\n", header);
+
+ byt->boot_complete = true;
+ wake_up(&byt->boot_wait);
+}
+
+static int sst_byt_process_notification(struct sst_byt *byt,
+ unsigned long *flags)
+{
+ struct sst_dsp *sst = byt->dsp;
+ struct sst_byt_stream *stream;
+ u64 header;
+ u8 msg_id, stream_id;
+ int handled = 1;
+
+ header = sst_dsp_shim_read64_unlocked(sst, SST_IPCD);
+ msg_id = sst_byt_header_msg_id(header);
+
+ switch (msg_id) {
+ case IPC_SST_PERIOD_ELAPSED:
+ stream_id = sst_byt_header_str_id(header);
+ stream = sst_byt_get_stream(byt, stream_id);
+ if (stream && stream->running && stream->notify_position) {
+ spin_unlock_irqrestore(&sst->spinlock, *flags);
+ stream->notify_position(stream, stream->pdata);
+ spin_lock_irqsave(&sst->spinlock, *flags);
+ }
+ break;
+ case IPC_IA_FW_INIT_CMPLT:
+ sst_byt_fw_ready(byt, header);
+ break;
+ }
+
+ return handled;
+}
+
+static irqreturn_t sst_byt_irq_thread(int irq, void *context)
+{
+ struct sst_dsp *sst = (struct sst_dsp *) context;
+ struct sst_byt *byt = sst_dsp_get_thread_context(sst);
+ u64 header;
+ unsigned long flags;
+
+ spin_lock_irqsave(&sst->spinlock, flags);
+
+ header = sst_dsp_shim_read64_unlocked(sst, SST_IPCD);
+ if (header & SST_BYT_IPCD_BUSY) {
+ if (header & IPC_NOTIFICATION) {
+ /* message from ADSP */
+ sst_byt_process_notification(byt, &flags);
+ } else {
+ /* reply from ADSP */
+ sst_byt_process_reply(byt, header);
+ }
+ /*
+ * clear IPCD BUSY bit and set DONE bit. Tell DSP we have
+ * processed the message and can accept new. Clear data part
+ * of the header
+ */
+ sst_dsp_shim_update_bits64_unlocked(sst, SST_IPCD,
+ SST_BYT_IPCD_DONE | SST_BYT_IPCD_BUSY |
+ IPC_HEADER_DATA(IPC_HEADER_DATA_MASK),
+ SST_BYT_IPCD_DONE);
+ /* unmask message request interrupts */
+ sst_dsp_shim_update_bits64_unlocked(sst, SST_IMRX,
+ SST_BYT_IMRX_REQUEST, 0);
+ }
+
+ spin_unlock_irqrestore(&sst->spinlock, flags);
+
+ /* continue to send any remaining messages... */
+ queue_kthread_work(&byt->kworker, &byt->kwork);
+
+ return IRQ_HANDLED;
+}
+
+/* stream API */
+struct sst_byt_stream *sst_byt_stream_new(struct sst_byt *byt, int id,
+ u32 (*notify_position)(struct sst_byt_stream *stream, void *data),
+ void *data)
+{
+ struct sst_byt_stream *stream;
+
+ stream = kzalloc(sizeof(*stream), GFP_KERNEL);
+ if (stream == NULL)
+ return NULL;
+
+ list_add(&stream->node, &byt->stream_list);
+ stream->notify_position = notify_position;
+ stream->pdata = data;
+ stream->byt = byt;
+ stream->str_id = id;
+
+ return stream;
+}
+
+int sst_byt_stream_set_bits(struct sst_byt *byt, struct sst_byt_stream *stream,
+ int bits)
+{
+ stream->request.pcm_params.pcm_wd_sz = bits;
+ return 0;
+}
+
+int sst_byt_stream_set_channels(struct sst_byt *byt,
+ struct sst_byt_stream *stream, u8 channels)
+{
+ stream->request.pcm_params.num_chan = channels;
+ return 0;
+}
+
+int sst_byt_stream_set_rate(struct sst_byt *byt, struct sst_byt_stream *stream,
+ unsigned int rate)
+{
+ stream->request.pcm_params.sfreq = rate;
+ return 0;
+}
+
+/* stream sonfiguration */
+int sst_byt_stream_type(struct sst_byt *byt, struct sst_byt_stream *stream,
+ int codec_type, int stream_type, int operation)
+{
+ stream->request.str_type.codec_type = codec_type;
+ stream->request.str_type.str_type = stream_type;
+ stream->request.str_type.operation = operation;
+ stream->request.str_type.time_slots = 0xc;
+
+ return 0;
+}
+
+int sst_byt_stream_buffer(struct sst_byt *byt, struct sst_byt_stream *stream,
+ uint32_t buffer_addr, uint32_t buffer_size)
+{
+ stream->request.frame_info.num_entries = 1;
+ stream->request.frame_info.ring_buf_info[0].addr = buffer_addr;
+ stream->request.frame_info.ring_buf_info[0].size = buffer_size;
+ /* calculate bytes per 4 ms fragment */
+ stream->request.frame_info.frag_size =
+ stream->request.pcm_params.sfreq *
+ stream->request.pcm_params.num_chan *
+ stream->request.pcm_params.pcm_wd_sz / 8 *
+ 4 / 1000;
+ return 0;
+}
+
+int sst_byt_stream_commit(struct sst_byt *byt, struct sst_byt_stream *stream)
+{
+ struct sst_byt_alloc_params *str_req = &stream->request;
+ struct sst_byt_alloc_response *reply = &stream->reply;
+ u64 header;
+ int ret;
+
+ header = sst_byt_header(IPC_IA_ALLOC_STREAM,
+ sizeof(*str_req) + sizeof(u32),
+ true, stream->str_id);
+ ret = sst_byt_ipc_tx_msg_wait(byt, header, str_req, sizeof(*str_req),
+ reply, sizeof(*reply));
+ if (ret < 0) {
+ dev_err(byt->dev, "ipc: error stream commit failed\n");
+ return ret;
+ }
+
+ stream->commited = true;
+
+ return 0;
+}
+
+int sst_byt_stream_free(struct sst_byt *byt, struct sst_byt_stream *stream)
+{
+ u64 header;
+ int ret = 0;
+
+ if (!stream->commited)
+ goto out;
+
+ header = sst_byt_header(IPC_IA_FREE_STREAM, 0, false, stream->str_id);
+ ret = sst_byt_ipc_tx_msg_wait(byt, header, NULL, 0, NULL, 0);
+ if (ret < 0) {
+ dev_err(byt->dev, "ipc: free stream %d failed\n",
+ stream->str_id);
+ return -EAGAIN;
+ }
+
+ stream->commited = false;
+out:
+ list_del(&stream->node);
+ kfree(stream);
+
+ return ret;
+}
+
+static int sst_byt_stream_operations(struct sst_byt *byt, int type,
+ int stream_id, int wait)
+{
+ struct sst_byt_start_stream_params start_stream;
+ u64 header;
+ void *tx_msg = NULL;
+ size_t size = 0;
+
+ if (type != IPC_IA_START_STREAM) {
+ header = sst_byt_header(type, 0, false, stream_id);
+ } else {
+ start_stream.byte_offset = 0;
+ header = sst_byt_header(IPC_IA_START_STREAM,
+ sizeof(start_stream) + sizeof(u32),
+ true, stream_id);
+ tx_msg = &start_stream;
+ size = sizeof(start_stream);
+ }
+
+ if (wait)
+ return sst_byt_ipc_tx_msg_wait(byt, header,
+ tx_msg, size, NULL, 0);
+ else
+ return sst_byt_ipc_tx_msg_nowait(byt, header, tx_msg, size);
+}
+
+/* stream ALSA trigger operations */
+int sst_byt_stream_start(struct sst_byt *byt, struct sst_byt_stream *stream)
+{
+ int ret;
+
+ ret = sst_byt_stream_operations(byt, IPC_IA_START_STREAM,
+ stream->str_id, 0);
+ if (ret < 0)
+ dev_err(byt->dev, "ipc: error failed to start stream %d\n",
+ stream->str_id);
+
+ return ret;
+}
+
+int sst_byt_stream_stop(struct sst_byt *byt, struct sst_byt_stream *stream)
+{
+ int ret;
+
+ /* don't stop streams that are not commited */
+ if (!stream->commited)
+ return 0;
+
+ ret = sst_byt_stream_operations(byt, IPC_IA_DROP_STREAM,
+ stream->str_id, 0);
+ if (ret < 0)
+ dev_err(byt->dev, "ipc: error failed to stop stream %d\n",
+ stream->str_id);
+ return ret;
+}
+
+int sst_byt_stream_pause(struct sst_byt *byt, struct sst_byt_stream *stream)
+{
+ int ret;
+
+ ret = sst_byt_stream_operations(byt, IPC_IA_PAUSE_STREAM,
+ stream->str_id, 0);
+ if (ret < 0)
+ dev_err(byt->dev, "ipc: error failed to pause stream %d\n",
+ stream->str_id);
+
+ return ret;
+}
+
+int sst_byt_stream_resume(struct sst_byt *byt, struct sst_byt_stream *stream)
+{
+ int ret;
+
+ ret = sst_byt_stream_operations(byt, IPC_IA_RESUME_STREAM,
+ stream->str_id, 0);
+ if (ret < 0)
+ dev_err(byt->dev, "ipc: error failed to resume stream %d\n",
+ stream->str_id);
+
+ return ret;
+}
+
+int sst_byt_get_dsp_position(struct sst_byt *byt,
+ struct sst_byt_stream *stream, int buffer_size)
+{
+ struct sst_dsp *sst = byt->dsp;
+ struct sst_byt_tstamp fw_tstamp;
+ u8 str_id = stream->str_id;
+ u32 tstamp_offset;
+
+ tstamp_offset = SST_BYT_TIMESTAMP_OFFSET + str_id * sizeof(fw_tstamp);
+ memcpy_fromio(&fw_tstamp,
+ sst->addr.lpe + tstamp_offset, sizeof(fw_tstamp));
+
+ return do_div(fw_tstamp.ring_buffer_counter, buffer_size);
+}
+
+static int msg_empty_list_init(struct sst_byt *byt)
+{
+ struct ipc_message *msg;
+ int i;
+
+ byt->msg = kzalloc(sizeof(*msg) * IPC_EMPTY_LIST_SIZE, GFP_KERNEL);
+ if (byt->msg == NULL)
+ return -ENOMEM;
+
+ for (i = 0; i < IPC_EMPTY_LIST_SIZE; i++) {
+ init_waitqueue_head(&byt->msg[i].waitq);
+ list_add(&byt->msg[i].list, &byt->empty_list);
+ }
+
+ return 0;
+}
+
+struct sst_dsp *sst_byt_get_dsp(struct sst_byt *byt)
+{
+ return byt->dsp;
+}
+
+static struct sst_dsp_device byt_dev = {
+ .thread = sst_byt_irq_thread,
+ .ops = &sst_byt_ops,
+};
+
+int sst_byt_dsp_init(struct device *dev, struct sst_pdata *pdata)
+{
+ struct sst_byt *byt;
+ struct sst_fw *byt_sst_fw;
+ int err;
+
+ dev_dbg(dev, "initialising Byt DSP IPC\n");
+
+ byt = devm_kzalloc(dev, sizeof(*byt), GFP_KERNEL);
+ if (byt == NULL)
+ return -ENOMEM;
+
+ byt->dev = dev;
+ INIT_LIST_HEAD(&byt->stream_list);
+ INIT_LIST_HEAD(&byt->tx_list);
+ INIT_LIST_HEAD(&byt->rx_list);
+ INIT_LIST_HEAD(&byt->empty_list);
+ init_waitqueue_head(&byt->boot_wait);
+ init_waitqueue_head(&byt->wait_txq);
+
+ err = msg_empty_list_init(byt);
+ if (err < 0)
+ return -ENOMEM;
+
+ /* start the IPC message thread */
+ init_kthread_worker(&byt->kworker);
+ byt->tx_thread = kthread_run(kthread_worker_fn,
+ &byt->kworker,
+ dev_name(byt->dev));
+ if (IS_ERR(byt->tx_thread)) {
+ err = PTR_ERR(byt->tx_thread);
+ dev_err(byt->dev, "error failed to create message TX task\n");
+ goto err_free_msg;
+ }
+ init_kthread_work(&byt->kwork, sst_byt_ipc_tx_msgs);
+
+ byt_dev.thread_context = byt;
+
+ /* init SST shim */
+ byt->dsp = sst_dsp_new(dev, &byt_dev, pdata);
+ if (byt->dsp == NULL) {
+ err = -ENODEV;
+ goto err_free_msg;
+ }
+
+ /* keep the DSP in reset state for base FW loading */
+ sst_dsp_reset(byt->dsp);
+
+ byt_sst_fw = sst_fw_new(byt->dsp, pdata->fw, byt);
+ if (byt_sst_fw == NULL) {
+ err = -ENODEV;
+ dev_err(dev, "error: failed to load firmware\n");
+ goto fw_err;
+ }
+
+ /* wait for DSP boot completion */
+ sst_dsp_boot(byt->dsp);
+ err = wait_event_timeout(byt->boot_wait, byt->boot_complete,
+ msecs_to_jiffies(IPC_BOOT_MSECS));
+ if (err == 0) {
+ err = -EIO;
+ dev_err(byt->dev, "ipc: error DSP boot timeout\n");
+ goto boot_err;
+ }
+
+ pdata->dsp = byt;
+
+ return 0;
+
+boot_err:
+ sst_dsp_reset(byt->dsp);
+ sst_fw_free(byt_sst_fw);
+fw_err:
+ sst_dsp_free(byt->dsp);
+err_free_msg:
+ kfree(byt->msg);
+
+ return err;
+}
+EXPORT_SYMBOL_GPL(sst_byt_dsp_init);
+
+void sst_byt_dsp_free(struct device *dev, struct sst_pdata *pdata)
+{
+ struct sst_byt *byt = pdata->dsp;
+
+ sst_dsp_reset(byt->dsp);
+ sst_fw_free_all(byt->dsp);
+ sst_dsp_free(byt->dsp);
+ kfree(byt->msg);
+}
+EXPORT_SYMBOL_GPL(sst_byt_dsp_free);
diff --git a/sound/soc/intel/sst-baytrail-ipc.h b/sound/soc/intel/sst-baytrail-ipc.h
new file mode 100644
index 0000000..f172b64
--- /dev/null
+++ b/sound/soc/intel/sst-baytrail-ipc.h
@@ -0,0 +1,69 @@
+/*
+ * Intel Baytrail SST IPC Support
+ * Copyright (c) 2014, Intel Corporation.
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms and conditions of the GNU General Public License,
+ * version 2, as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope 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.
+ */
+
+#ifndef __SST_BYT_IPC_H
+#define __SST_BYT_IPC_H
+
+#include <linux/types.h>
+
+struct sst_byt;
+struct sst_byt_stream;
+struct sst_pdata;
+extern struct sst_ops sst_byt_ops;
+
+
+#define SST_BYT_MAILBOX_OFFSET 0x144000
+#define SST_BYT_TIMESTAMP_OFFSET (SST_BYT_MAILBOX_OFFSET + 0x800)
+
+/**
+ * Upfront defined maximum message size that is
+ * expected by the in/out communication pipes in FW.
+ */
+#define SST_BYT_IPC_MAX_PAYLOAD_SIZE 200
+
+/* stream API */
+struct sst_byt_stream *sst_byt_stream_new(struct sst_byt *byt, int id,
+ uint32_t (*get_write_position)(struct sst_byt_stream *stream,
+ void *data),
+ void *data);
+
+/* stream configuration */
+int sst_byt_stream_set_bits(struct sst_byt *byt, struct sst_byt_stream *stream,
+ int bits);
+int sst_byt_stream_set_channels(struct sst_byt *byt,
+ struct sst_byt_stream *stream, u8 channels);
+int sst_byt_stream_set_rate(struct sst_byt *byt, struct sst_byt_stream *stream,
+ unsigned int rate);
+int sst_byt_stream_type(struct sst_byt *byt, struct sst_byt_stream *stream,
+ int codec_type, int stream_type, int operation);
+int sst_byt_stream_buffer(struct sst_byt *byt, struct sst_byt_stream *stream,
+ uint32_t buffer_addr, uint32_t buffer_size);
+int sst_byt_stream_commit(struct sst_byt *byt, struct sst_byt_stream *stream);
+int sst_byt_stream_free(struct sst_byt *byt, struct sst_byt_stream *stream);
+
+/* stream ALSA trigger operations */
+int sst_byt_stream_start(struct sst_byt *byt, struct sst_byt_stream *stream);
+int sst_byt_stream_stop(struct sst_byt *byt, struct sst_byt_stream *stream);
+int sst_byt_stream_pause(struct sst_byt *byt, struct sst_byt_stream *stream);
+int sst_byt_stream_resume(struct sst_byt *byt, struct sst_byt_stream *stream);
+
+int sst_byt_get_dsp_position(struct sst_byt *byt,
+ struct sst_byt_stream *stream, int buffer_size);
+
+/* init */
+int sst_byt_dsp_init(struct device *dev, struct sst_pdata *pdata);
+void sst_byt_dsp_free(struct device *dev, struct sst_pdata *pdata);
+struct sst_dsp *sst_byt_get_dsp(struct sst_byt *byt);
+
+#endif
diff --git a/sound/soc/intel/sst-baytrail-pcm.c b/sound/soc/intel/sst-baytrail-pcm.c
new file mode 100644
index 0000000..6d101f3
--- /dev/null
+++ b/sound/soc/intel/sst-baytrail-pcm.c
@@ -0,0 +1,422 @@
+/*
+ * Intel Baytrail SST PCM Support
+ * Copyright (c) 2014, Intel Corporation.
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms and conditions of the GNU General Public License,
+ * version 2, as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope 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.
+ */
+
+#include <linux/module.h>
+#include <linux/dma-mapping.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include "sst-baytrail-ipc.h"
+#include "sst-dsp-priv.h"
+#include "sst-dsp.h"
+
+#define BYT_PCM_COUNT 2
+
+static const struct snd_pcm_hardware sst_byt_pcm_hardware = {
+ .info = SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_PAUSE |
+ SNDRV_PCM_INFO_RESUME,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FORMAT_S24_LE,
+ .period_bytes_min = 384,
+ .period_bytes_max = 48000,
+ .periods_min = 2,
+ .periods_max = 250,
+ .buffer_bytes_max = 96000,
+};
+
+/* private data for each PCM DSP stream */
+struct sst_byt_pcm_data {
+ struct sst_byt_stream *stream;
+ struct snd_pcm_substream *substream;
+ struct mutex mutex;
+};
+
+/* private data for the driver */
+struct sst_byt_priv_data {
+ /* runtime DSP */
+ struct sst_byt *byt;
+
+ /* DAI data */
+ struct sst_byt_pcm_data pcm[BYT_PCM_COUNT];
+};
+
+/* this may get called several times by oss emulation */
+static int sst_byt_pcm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct sst_byt_priv_data *pdata =
+ snd_soc_platform_get_drvdata(rtd->platform);
+ struct sst_byt_pcm_data *pcm_data = snd_soc_pcm_get_drvdata(rtd);
+ struct sst_byt *byt = pdata->byt;
+ u32 rate, bits;
+ u8 channels;
+ int ret, playback = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK);
+
+ dev_dbg(rtd->dev, "PCM: hw_params, pcm_data %p\n", pcm_data);
+
+ ret = sst_byt_stream_type(byt, pcm_data->stream,
+ 1, 1, !playback);
+ if (ret < 0) {
+ dev_err(rtd->dev, "failed to set stream format %d\n", ret);
+ return ret;
+ }
+
+ rate = params_rate(params);
+ ret = sst_byt_stream_set_rate(byt, pcm_data->stream, rate);
+ if (ret < 0) {
+ dev_err(rtd->dev, "could not set rate %d\n", rate);
+ return ret;
+ }
+
+ bits = snd_pcm_format_width(params_format(params));
+ ret = sst_byt_stream_set_bits(byt, pcm_data->stream, bits);
+ if (ret < 0) {
+ dev_err(rtd->dev, "could not set formats %d\n",
+ params_rate(params));
+ return ret;
+ }
+
+ channels = (u8)(params_channels(params) & 0xF);
+ ret = sst_byt_stream_set_channels(byt, pcm_data->stream, channels);
+ if (ret < 0) {
+ dev_err(rtd->dev, "could not set channels %d\n",
+ params_rate(params));
+ return ret;
+ }
+
+ snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params));
+
+ ret = sst_byt_stream_buffer(byt, pcm_data->stream,
+ substream->dma_buffer.addr,
+ params_buffer_bytes(params));
+ if (ret < 0) {
+ dev_err(rtd->dev, "PCM: failed to set DMA buffer %d\n", ret);
+ return ret;
+ }
+
+ ret = sst_byt_stream_commit(byt, pcm_data->stream);
+ if (ret < 0) {
+ dev_err(rtd->dev, "PCM: failed stream commit %d\n", ret);
+ return ret;
+ }
+
+ return 0;
+}
+
+static int sst_byt_pcm_hw_free(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+
+ dev_dbg(rtd->dev, "PCM: hw_free\n");
+ snd_pcm_lib_free_pages(substream);
+
+ return 0;
+}
+
+static int sst_byt_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct sst_byt_priv_data *pdata =
+ snd_soc_platform_get_drvdata(rtd->platform);
+ struct sst_byt_pcm_data *pcm_data = snd_soc_pcm_get_drvdata(rtd);
+ struct sst_byt *byt = pdata->byt;
+
+ dev_dbg(rtd->dev, "PCM: trigger %d\n", cmd);
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ sst_byt_stream_start(byt, pcm_data->stream);
+ break;
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ sst_byt_stream_resume(byt, pcm_data->stream);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ sst_byt_stream_stop(byt, pcm_data->stream);
+ break;
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ sst_byt_stream_pause(byt, pcm_data->stream);
+ break;
+ default:
+ break;
+ }
+
+ return 0;
+}
+
+static u32 byt_notify_pointer(struct sst_byt_stream *stream, void *data)
+{
+ struct sst_byt_pcm_data *pcm_data = data;
+ struct snd_pcm_substream *substream = pcm_data->substream;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ u32 pos;
+
+ pos = frames_to_bytes(runtime,
+ (runtime->control->appl_ptr %
+ runtime->buffer_size));
+
+ dev_dbg(rtd->dev, "PCM: App pointer %d bytes\n", pos);
+
+ snd_pcm_period_elapsed(substream);
+ return pos;
+}
+
+static snd_pcm_uframes_t sst_byt_pcm_pointer(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct sst_byt_priv_data *pdata =
+ snd_soc_platform_get_drvdata(rtd->platform);
+ struct sst_byt_pcm_data *pcm_data = snd_soc_pcm_get_drvdata(rtd);
+ struct sst_byt *byt = pdata->byt;
+ snd_pcm_uframes_t offset;
+ int pos;
+
+ pos = sst_byt_get_dsp_position(byt, pcm_data->stream,
+ snd_pcm_lib_buffer_bytes(substream));
+ offset = bytes_to_frames(runtime, pos);
+
+ dev_dbg(rtd->dev, "PCM: DMA pointer %zu bytes\n",
+ frames_to_bytes(runtime, (u32)offset));
+ return offset;
+}
+
+static int sst_byt_pcm_open(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct sst_byt_priv_data *pdata =
+ snd_soc_platform_get_drvdata(rtd->platform);
+ struct sst_byt_pcm_data *pcm_data = snd_soc_pcm_get_drvdata(rtd);
+ struct sst_byt *byt = pdata->byt;
+
+ dev_dbg(rtd->dev, "PCM: open\n");
+
+ pcm_data = &pdata->pcm[rtd->cpu_dai->id];
+ mutex_lock(&pcm_data->mutex);
+
+ snd_soc_pcm_set_drvdata(rtd, pcm_data);
+ pcm_data->substream = substream;
+
+ snd_soc_set_runtime_hwparams(substream, &sst_byt_pcm_hardware);
+
+ pcm_data->stream = sst_byt_stream_new(byt, rtd->cpu_dai->id + 1,
+ byt_notify_pointer, pcm_data);
+ if (pcm_data->stream == NULL) {
+ dev_err(rtd->dev, "failed to create stream\n");
+ mutex_unlock(&pcm_data->mutex);
+ return -EINVAL;
+ }
+
+ mutex_unlock(&pcm_data->mutex);
+ return 0;
+}
+
+static int sst_byt_pcm_close(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct sst_byt_priv_data *pdata =
+ snd_soc_platform_get_drvdata(rtd->platform);
+ struct sst_byt_pcm_data *pcm_data = snd_soc_pcm_get_drvdata(rtd);
+ struct sst_byt *byt = pdata->byt;
+ int ret;
+
+ dev_dbg(rtd->dev, "PCM: close\n");
+
+ mutex_lock(&pcm_data->mutex);
+ ret = sst_byt_stream_free(byt, pcm_data->stream);
+ if (ret < 0) {
+ dev_dbg(rtd->dev, "Free stream fail\n");
+ goto out;
+ }
+ pcm_data->stream = NULL;
+
+out:
+ mutex_unlock(&pcm_data->mutex);
+ return ret;
+}
+
+static int sst_byt_pcm_mmap(struct snd_pcm_substream *substream,
+ struct vm_area_struct *vma)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+
+ dev_dbg(rtd->dev, "PCM: mmap\n");
+ return snd_pcm_lib_default_mmap(substream, vma);
+}
+
+static struct snd_pcm_ops sst_byt_pcm_ops = {
+ .open = sst_byt_pcm_open,
+ .close = sst_byt_pcm_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = sst_byt_pcm_hw_params,
+ .hw_free = sst_byt_pcm_hw_free,
+ .trigger = sst_byt_pcm_trigger,
+ .pointer = sst_byt_pcm_pointer,
+ .mmap = sst_byt_pcm_mmap,
+};
+
+static void sst_byt_pcm_free(struct snd_pcm *pcm)
+{
+ snd_pcm_lib_preallocate_free_for_all(pcm);
+}
+
+static int sst_byt_pcm_new(struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_pcm *pcm = rtd->pcm;
+ size_t size;
+ int ret = 0;
+
+ ret = dma_coerce_mask_and_coherent(rtd->card->dev, DMA_BIT_MASK(32));
+ if (ret)
+ return ret;
+
+ if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream ||
+ pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) {
+ size = sst_byt_pcm_hardware.buffer_bytes_max;
+ ret = snd_pcm_lib_preallocate_pages_for_all(pcm,
+ SNDRV_DMA_TYPE_DEV,
+ rtd->card->dev,
+ size, size);
+ if (ret) {
+ dev_err(rtd->dev, "dma buffer allocation failed %d\n",
+ ret);
+ return ret;
+ }
+ }
+
+ return ret;
+}
+
+static struct snd_soc_dai_driver byt_dais[] = {
+ {
+ .name = "Front-cpu-dai",
+ .playback = {
+ .stream_name = "System Playback",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_48000,
+ .formats = SNDRV_PCM_FMTBIT_S24_3LE |
+ SNDRV_PCM_FMTBIT_S16_LE,
+ },
+ },
+ {
+ .name = "Mic1-cpu-dai",
+ .capture = {
+ .stream_name = "Analog Capture",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_48000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ },
+ },
+};
+
+static int sst_byt_pcm_probe(struct snd_soc_platform *platform)
+{
+ struct sst_pdata *plat_data = dev_get_platdata(platform->dev);
+ struct sst_byt_priv_data *priv_data;
+ int i;
+
+ if (!plat_data)
+ return -ENODEV;
+
+ priv_data = devm_kzalloc(platform->dev, sizeof(*priv_data),
+ GFP_KERNEL);
+ priv_data->byt = plat_data->dsp;
+ snd_soc_platform_set_drvdata(platform, priv_data);
+
+ for (i = 0; i < ARRAY_SIZE(byt_dais); i++)
+ mutex_init(&priv_data->pcm[i].mutex);
+
+ return 0;
+}
+
+static int sst_byt_pcm_remove(struct snd_soc_platform *platform)
+{
+ return 0;
+}
+
+static struct snd_soc_platform_driver byt_soc_platform = {
+ .probe = sst_byt_pcm_probe,
+ .remove = sst_byt_pcm_remove,
+ .ops = &sst_byt_pcm_ops,
+ .pcm_new = sst_byt_pcm_new,
+ .pcm_free = sst_byt_pcm_free,
+};
+
+static const struct snd_soc_component_driver byt_dai_component = {
+ .name = "byt-dai",
+};
+
+static int sst_byt_pcm_dev_probe(struct platform_device *pdev)
+{
+ struct sst_pdata *sst_pdata = dev_get_platdata(&pdev->dev);
+ int ret;
+
+ ret = sst_byt_dsp_init(&pdev->dev, sst_pdata);
+ if (ret < 0)
+ return -ENODEV;
+
+ ret = snd_soc_register_platform(&pdev->dev, &byt_soc_platform);
+ if (ret < 0)
+ goto err_plat;
+
+ ret = snd_soc_register_component(&pdev->dev, &byt_dai_component,
+ byt_dais, ARRAY_SIZE(byt_dais));
+ if (ret < 0)
+ goto err_comp;
+
+ return 0;
+
+err_comp:
+ snd_soc_unregister_platform(&pdev->dev);
+err_plat:
+ sst_byt_dsp_free(&pdev->dev, sst_pdata);
+ return ret;
+}
+
+static int sst_byt_pcm_dev_remove(struct platform_device *pdev)
+{
+ struct sst_pdata *sst_pdata = dev_get_platdata(&pdev->dev);
+
+ snd_soc_unregister_platform(&pdev->dev);
+ snd_soc_unregister_component(&pdev->dev);
+ sst_byt_dsp_free(&pdev->dev, sst_pdata);
+
+ return 0;
+}
+
+static struct platform_driver sst_byt_pcm_driver = {
+ .driver = {
+ .name = "baytrail-pcm-audio",
+ .owner = THIS_MODULE,
+ },
+
+ .probe = sst_byt_pcm_dev_probe,
+ .remove = sst_byt_pcm_dev_remove,
+};
+module_platform_driver(sst_byt_pcm_driver);
+
+MODULE_AUTHOR("Jarkko Nikula");
+MODULE_DESCRIPTION("Baytrail PCM");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:baytrail-pcm-audio");
diff --git a/sound/soc/intel/sst-dsp-priv.h b/sound/soc/intel/sst-dsp-priv.h
new file mode 100644
index 0000000..fe8e81aa
--- /dev/null
+++ b/sound/soc/intel/sst-dsp-priv.h
@@ -0,0 +1,309 @@
+/*
+ * Intel Smart Sound Technology
+ *
+ * Copyright (C) 2013, Intel Corporation. All rights reserved.
+ *
+ * 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.
+ *
+ */
+
+#ifndef __SOUND_SOC_SST_DSP_PRIV_H
+#define __SOUND_SOC_SST_DSP_PRIV_H
+
+#include <linux/kernel.h>
+#include <linux/types.h>
+#include <linux/interrupt.h>
+#include <linux/firmware.h>
+
+struct sst_mem_block;
+struct sst_module;
+struct sst_fw;
+
+/*
+ * DSP Operations exported by platform Audio DSP driver.
+ */
+struct sst_ops {
+ /* DSP core boot / reset */
+ void (*boot)(struct sst_dsp *);
+ void (*reset)(struct sst_dsp *);
+
+ /* Shim IO */
+ void (*write)(void __iomem *addr, u32 offset, u32 value);
+ u32 (*read)(void __iomem *addr, u32 offset);
+ void (*write64)(void __iomem *addr, u32 offset, u64 value);
+ u64 (*read64)(void __iomem *addr, u32 offset);
+
+ /* DSP I/DRAM IO */
+ void (*ram_read)(struct sst_dsp *sst, void *dest, void __iomem *src,
+ size_t bytes);
+ void (*ram_write)(struct sst_dsp *sst, void __iomem *dest, void *src,
+ size_t bytes);
+
+ void (*dump)(struct sst_dsp *);
+
+ /* IRQ handlers */
+ irqreturn_t (*irq_handler)(int irq, void *context);
+
+ /* SST init and free */
+ int (*init)(struct sst_dsp *sst, struct sst_pdata *pdata);
+ void (*free)(struct sst_dsp *sst);
+
+ /* FW module parser/loader */
+ int (*parse_fw)(struct sst_fw *sst_fw);
+};
+
+/*
+ * Audio DSP memory offsets and addresses.
+ */
+struct sst_addr {
+ u32 lpe_base;
+ u32 shim_offset;
+ u32 iram_offset;
+ u32 dram_offset;
+ void __iomem *lpe;
+ void __iomem *shim;
+ void __iomem *pci_cfg;
+ void __iomem *fw_ext;
+};
+
+/*
+ * Audio DSP Mailbox configuration.
+ */
+struct sst_mailbox {
+ void __iomem *in_base;
+ void __iomem *out_base;
+ size_t in_size;
+ size_t out_size;
+};
+
+/*
+ * Audio DSP Firmware data types.
+ */
+enum sst_data_type {
+ SST_DATA_M = 0, /* module block data */
+ SST_DATA_P = 1, /* peristant data (text, data) */
+ SST_DATA_S = 2, /* scratch data (usually buffers) */
+};
+
+/*
+ * Audio DSP memory block types.
+ */
+enum sst_mem_type {
+ SST_MEM_IRAM = 0,
+ SST_MEM_DRAM = 1,
+ SST_MEM_ANY = 2,
+ SST_MEM_CACHE= 3,
+};
+
+/*
+ * Audio DSP Generic Firmware File.
+ *
+ * SST Firmware files can consist of 1..N modules. This generic structure is
+ * used to manage each firmware file and it's modules regardless of SST firmware
+ * type. A SST driver may load multiple FW files.
+ */
+struct sst_fw {
+ struct sst_dsp *dsp;
+
+ /* base addresses of FW file data */
+ dma_addr_t dmable_fw_paddr; /* physical address of fw data */
+ void *dma_buf; /* virtual address of fw data */
+ u32 size; /* size of fw data */
+
+ /* lists */
+ struct list_head list; /* DSP list of FW */
+ struct list_head module_list; /* FW list of modules */
+
+ void *private; /* core doesn't touch this */
+};
+
+/*
+ * Audio DSP Generic Module data.
+ *
+ * This is used to dsecribe any sections of persistent (text and data) and
+ * scratch (buffers) of module data in ADSP memory space.
+ */
+struct sst_module_data {
+
+ enum sst_mem_type type; /* destination memory type */
+ enum sst_data_type data_type; /* type of module data */
+
+ u32 size; /* size in bytes */
+ u32 offset; /* offset in FW file */
+ u32 data_offset; /* offset in ADSP memory space */
+ void *data; /* module data */
+};
+
+/*
+ * Audio DSP Generic Module Template.
+ *
+ * Used to define and register a new FW module. This data is extracted from
+ * FW module header information.
+ */
+struct sst_module_template {
+ u32 id;
+ u32 entry; /* entry point */
+ struct sst_module_data s; /* scratch data */
+ struct sst_module_data p; /* peristant data */
+};
+
+/*
+ * Audio DSP Generic Module.
+ *
+ * Each Firmware file can consist of 1..N modules. A module can span multiple
+ * ADSP memory blocks. The simplest FW will be a file with 1 module.
+ */
+struct sst_module {
+ struct sst_dsp *dsp;
+ struct sst_fw *sst_fw; /* parent FW we belong too */
+
+ /* module configuration */
+ u32 id;
+ u32 entry; /* module entry point */
+ u32 offset; /* module offset in firmware file */
+ u32 size; /* module size */
+ struct sst_module_data s; /* scratch data */
+ struct sst_module_data p; /* peristant data */
+
+ /* runtime */
+ u32 usage_count; /* can be unloaded if count == 0 */
+ void *private; /* core doesn't touch this */
+
+ /* lists */
+ struct list_head block_list; /* Module list of blocks in use */
+ struct list_head list; /* DSP list of modules */
+ struct list_head list_fw; /* FW list of modules */
+};
+
+/*
+ * SST Memory Block operations.
+ */
+struct sst_block_ops {
+ int (*enable)(struct sst_mem_block *block);
+ int (*disable)(struct sst_mem_block *block);
+};
+
+/*
+ * SST Generic Memory Block.
+ *
+ * SST ADP memory has multiple IRAM and DRAM blocks. Some ADSP blocks can be
+ * power gated.
+ */
+struct sst_mem_block {
+ struct sst_dsp *dsp;
+ struct sst_module *module; /* module that uses this block */
+
+ /* block config */
+ u32 offset; /* offset from base */
+ u32 size; /* block size */
+ u32 index; /* block index 0..N */
+ enum sst_mem_type type; /* block memory type IRAM/DRAM */
+ struct sst_block_ops *ops; /* block operations, if any */
+
+ /* block status */
+ enum sst_data_type data_type; /* data type held in this block */
+ u32 bytes_used; /* bytes in use by modules */
+ void *private; /* generic core does not touch this */
+ int users; /* number of modules using this block */
+
+ /* block lists */
+ struct list_head module_list; /* Module list of blocks */
+ struct list_head list; /* Map list of free/used blocks */
+};
+
+/*
+ * Generic SST Shim Interface.
+ */
+struct sst_dsp {
+
+ /* runtime */
+ struct sst_dsp_device *sst_dev;
+ spinlock_t spinlock; /* IPC locking */
+ struct mutex mutex; /* DSP FW lock */
+ struct device *dev;
+ void *thread_context;
+ int irq;
+ u32 id;
+
+ /* list of free and used ADSP memory blocks */
+ struct list_head used_block_list;
+ struct list_head free_block_list;
+
+ /* operations */
+ struct sst_ops *ops;
+
+ /* debug FS */
+ struct dentry *debugfs_root;
+
+ /* base addresses */
+ struct sst_addr addr;
+
+ /* mailbox */
+ struct sst_mailbox mailbox;
+
+ /* SST FW files loaded and their modules */
+ struct list_head module_list;
+ struct list_head fw_list;
+
+ /* platform data */
+ struct sst_pdata *pdata;
+
+ /* DMA FW loading */
+ struct sst_dma *dma;
+ bool fw_use_dma;
+};
+
+/* Size optimised DRAM/IRAM memcpy */
+static inline void sst_dsp_write(struct sst_dsp *sst, void *src,
+ u32 dest_offset, size_t bytes)
+{
+ sst->ops->ram_write(sst, sst->addr.lpe + dest_offset, src, bytes);
+}
+
+static inline void sst_dsp_read(struct sst_dsp *sst, void *dest,
+ u32 src_offset, size_t bytes)
+{
+ sst->ops->ram_read(sst, dest, sst->addr.lpe + src_offset, bytes);
+}
+
+static inline void *sst_dsp_get_thread_context(struct sst_dsp *sst)
+{
+ return sst->thread_context;
+}
+
+/* Create/Free FW files - can contain multiple modules */
+struct sst_fw *sst_fw_new(struct sst_dsp *dsp,
+ const struct firmware *fw, void *private);
+void sst_fw_free(struct sst_fw *sst_fw);
+void sst_fw_free_all(struct sst_dsp *dsp);
+
+/* Create/Free firmware modules */
+struct sst_module *sst_module_new(struct sst_fw *sst_fw,
+ struct sst_module_template *template, void *private);
+void sst_module_free(struct sst_module *sst_module);
+int sst_module_insert(struct sst_module *sst_module);
+int sst_module_remove(struct sst_module *sst_module);
+int sst_module_insert_fixed_block(struct sst_module *module,
+ struct sst_module_data *data);
+struct sst_module *sst_module_get_from_id(struct sst_dsp *dsp, u32 id);
+
+/* allocate/free pesistent/scratch memory regions managed by drv */
+struct sst_module *sst_mem_block_alloc_scratch(struct sst_dsp *dsp);
+void sst_mem_block_free_scratch(struct sst_dsp *dsp,
+ struct sst_module *scratch);
+int sst_block_module_remove(struct sst_module *module);
+
+/* Register the DSPs memory blocks - would be nice to read from ACPI */
+struct sst_mem_block *sst_mem_block_register(struct sst_dsp *dsp, u32 offset,
+ u32 size, enum sst_mem_type type, struct sst_block_ops *ops, u32 index,
+ void *private);
+void sst_mem_block_unregister_all(struct sst_dsp *dsp);
+
+#endif
diff --git a/sound/soc/intel/sst-dsp.c b/sound/soc/intel/sst-dsp.c
new file mode 100644
index 0000000..0c129fd
--- /dev/null
+++ b/sound/soc/intel/sst-dsp.c
@@ -0,0 +1,385 @@
+/*
+ * Intel Smart Sound Technology (SST) DSP Core Driver
+ *
+ * Copyright (C) 2013, Intel Corporation. All rights reserved.
+ *
+ * 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.
+ *
+ */
+
+#include <linux/slab.h>
+#include <linux/export.h>
+#include <linux/interrupt.h>
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <linux/io.h>
+
+#include "sst-dsp.h"
+#include "sst-dsp-priv.h"
+
+#define CREATE_TRACE_POINTS
+#include <trace/events/intel-sst.h>
+
+/* Internal generic low-level SST IO functions - can be overidden */
+void sst_shim32_write(void __iomem *addr, u32 offset, u32 value)
+{
+ writel(value, addr + offset);
+}
+EXPORT_SYMBOL_GPL(sst_shim32_write);
+
+u32 sst_shim32_read(void __iomem *addr, u32 offset)
+{
+ return readl(addr + offset);
+}
+EXPORT_SYMBOL_GPL(sst_shim32_read);
+
+void sst_shim32_write64(void __iomem *addr, u32 offset, u64 value)
+{
+ memcpy_toio(addr + offset, &value, sizeof(value));
+}
+EXPORT_SYMBOL_GPL(sst_shim32_write64);
+
+u64 sst_shim32_read64(void __iomem *addr, u32 offset)
+{
+ u64 val;
+
+ memcpy_fromio(&val, addr + offset, sizeof(val));
+ return val;
+}
+EXPORT_SYMBOL_GPL(sst_shim32_read64);
+
+static inline void _sst_memcpy_toio_32(volatile u32 __iomem *dest,
+ u32 *src, size_t bytes)
+{
+ int i, words = bytes >> 2;
+
+ for (i = 0; i < words; i++)
+ writel(src[i], dest + i);
+}
+
+static inline void _sst_memcpy_fromio_32(u32 *dest,
+ const volatile __iomem u32 *src, size_t bytes)
+{
+ int i, words = bytes >> 2;
+
+ for (i = 0; i < words; i++)
+ dest[i] = readl(src + i);
+}
+
+void sst_memcpy_toio_32(struct sst_dsp *sst,
+ void __iomem *dest, void *src, size_t bytes)
+{
+ _sst_memcpy_toio_32(dest, src, bytes);
+}
+EXPORT_SYMBOL_GPL(sst_memcpy_toio_32);
+
+void sst_memcpy_fromio_32(struct sst_dsp *sst, void *dest,
+ void __iomem *src, size_t bytes)
+{
+ _sst_memcpy_fromio_32(dest, src, bytes);
+}
+EXPORT_SYMBOL_GPL(sst_memcpy_fromio_32);
+
+/* Public API */
+void sst_dsp_shim_write(struct sst_dsp *sst, u32 offset, u32 value)
+{
+ unsigned long flags;
+
+ spin_lock_irqsave(&sst->spinlock, flags);
+ sst->ops->write(sst->addr.shim, offset, value);
+ spin_unlock_irqrestore(&sst->spinlock, flags);
+}
+EXPORT_SYMBOL_GPL(sst_dsp_shim_write);
+
+u32 sst_dsp_shim_read(struct sst_dsp *sst, u32 offset)
+{
+ unsigned long flags;
+ u32 val;
+
+ spin_lock_irqsave(&sst->spinlock, flags);
+ val = sst->ops->read(sst->addr.shim, offset);
+ spin_unlock_irqrestore(&sst->spinlock, flags);
+
+ return val;
+}
+EXPORT_SYMBOL_GPL(sst_dsp_shim_read);
+
+void sst_dsp_shim_write64(struct sst_dsp *sst, u32 offset, u64 value)
+{
+ unsigned long flags;
+
+ spin_lock_irqsave(&sst->spinlock, flags);
+ sst->ops->write64(sst->addr.shim, offset, value);
+ spin_unlock_irqrestore(&sst->spinlock, flags);
+}
+EXPORT_SYMBOL_GPL(sst_dsp_shim_write64);
+
+u64 sst_dsp_shim_read64(struct sst_dsp *sst, u32 offset)
+{
+ unsigned long flags;
+ u64 val;
+
+ spin_lock_irqsave(&sst->spinlock, flags);
+ val = sst->ops->read64(sst->addr.shim, offset);
+ spin_unlock_irqrestore(&sst->spinlock, flags);
+
+ return val;
+}
+EXPORT_SYMBOL_GPL(sst_dsp_shim_read64);
+
+void sst_dsp_shim_write_unlocked(struct sst_dsp *sst, u32 offset, u32 value)
+{
+ sst->ops->write(sst->addr.shim, offset, value);
+}
+EXPORT_SYMBOL_GPL(sst_dsp_shim_write_unlocked);
+
+u32 sst_dsp_shim_read_unlocked(struct sst_dsp *sst, u32 offset)
+{
+ return sst->ops->read(sst->addr.shim, offset);
+}
+EXPORT_SYMBOL_GPL(sst_dsp_shim_read_unlocked);
+
+void sst_dsp_shim_write64_unlocked(struct sst_dsp *sst, u32 offset, u64 value)
+{
+ sst->ops->write64(sst->addr.shim, offset, value);
+}
+EXPORT_SYMBOL_GPL(sst_dsp_shim_write64_unlocked);
+
+u64 sst_dsp_shim_read64_unlocked(struct sst_dsp *sst, u32 offset)
+{
+ return sst->ops->read64(sst->addr.shim, offset);
+}
+EXPORT_SYMBOL_GPL(sst_dsp_shim_read64_unlocked);
+
+int sst_dsp_shim_update_bits_unlocked(struct sst_dsp *sst, u32 offset,
+ u32 mask, u32 value)
+{
+ bool change;
+ unsigned int old, new;
+ u32 ret;
+
+ ret = sst_dsp_shim_read_unlocked(sst, offset);
+
+ old = ret;
+ new = (old & (~mask)) | (value & mask);
+
+ change = (old != new);
+ if (change)
+ sst_dsp_shim_write_unlocked(sst, offset, new);
+
+ return change;
+}
+EXPORT_SYMBOL_GPL(sst_dsp_shim_update_bits_unlocked);
+
+int sst_dsp_shim_update_bits64_unlocked(struct sst_dsp *sst, u32 offset,
+ u64 mask, u64 value)
+{
+ bool change;
+ u64 old, new;
+
+ old = sst_dsp_shim_read64_unlocked(sst, offset);
+
+ new = (old & (~mask)) | (value & mask);
+
+ change = (old != new);
+ if (change)
+ sst_dsp_shim_write64_unlocked(sst, offset, new);
+
+ return change;
+}
+EXPORT_SYMBOL_GPL(sst_dsp_shim_update_bits64_unlocked);
+
+int sst_dsp_shim_update_bits(struct sst_dsp *sst, u32 offset,
+ u32 mask, u32 value)
+{
+ unsigned long flags;
+ bool change;
+
+ spin_lock_irqsave(&sst->spinlock, flags);
+ change = sst_dsp_shim_update_bits_unlocked(sst, offset, mask, value);
+ spin_unlock_irqrestore(&sst->spinlock, flags);
+ return change;
+}
+EXPORT_SYMBOL_GPL(sst_dsp_shim_update_bits);
+
+int sst_dsp_shim_update_bits64(struct sst_dsp *sst, u32 offset,
+ u64 mask, u64 value)
+{
+ unsigned long flags;
+ bool change;
+
+ spin_lock_irqsave(&sst->spinlock, flags);
+ change = sst_dsp_shim_update_bits64_unlocked(sst, offset, mask, value);
+ spin_unlock_irqrestore(&sst->spinlock, flags);
+ return change;
+}
+EXPORT_SYMBOL_GPL(sst_dsp_shim_update_bits64);
+
+void sst_dsp_dump(struct sst_dsp *sst)
+{
+ sst->ops->dump(sst);
+}
+EXPORT_SYMBOL_GPL(sst_dsp_dump);
+
+void sst_dsp_reset(struct sst_dsp *sst)
+{
+ sst->ops->reset(sst);
+}
+EXPORT_SYMBOL_GPL(sst_dsp_reset);
+
+int sst_dsp_boot(struct sst_dsp *sst)
+{
+ sst->ops->boot(sst);
+ return 0;
+}
+EXPORT_SYMBOL_GPL(sst_dsp_boot);
+
+void sst_dsp_ipc_msg_tx(struct sst_dsp *dsp, u32 msg)
+{
+ sst_dsp_shim_write_unlocked(dsp, SST_IPCX, msg | SST_IPCX_BUSY);
+ trace_sst_ipc_msg_tx(msg);
+}
+EXPORT_SYMBOL_GPL(sst_dsp_ipc_msg_tx);
+
+u32 sst_dsp_ipc_msg_rx(struct sst_dsp *dsp)
+{
+ u32 msg;
+
+ msg = sst_dsp_shim_read_unlocked(dsp, SST_IPCX);
+ trace_sst_ipc_msg_rx(msg);
+
+ return msg;
+}
+EXPORT_SYMBOL_GPL(sst_dsp_ipc_msg_rx);
+
+int sst_dsp_mailbox_init(struct sst_dsp *sst, u32 inbox_offset, size_t inbox_size,
+ u32 outbox_offset, size_t outbox_size)
+{
+ sst->mailbox.in_base = sst->addr.lpe + inbox_offset;
+ sst->mailbox.out_base = sst->addr.lpe + outbox_offset;
+ sst->mailbox.in_size = inbox_size;
+ sst->mailbox.out_size = outbox_size;
+ return 0;
+}
+EXPORT_SYMBOL_GPL(sst_dsp_mailbox_init);
+
+void sst_dsp_outbox_write(struct sst_dsp *sst, void *message, size_t bytes)
+{
+ u32 i;
+
+ trace_sst_ipc_outbox_write(bytes);
+
+ memcpy_toio(sst->mailbox.out_base, message, bytes);
+
+ for (i = 0; i < bytes; i += 4)
+ trace_sst_ipc_outbox_wdata(i, *(u32 *)(message + i));
+}
+EXPORT_SYMBOL_GPL(sst_dsp_outbox_write);
+
+void sst_dsp_outbox_read(struct sst_dsp *sst, void *message, size_t bytes)
+{
+ u32 i;
+
+ trace_sst_ipc_outbox_read(bytes);
+
+ memcpy_fromio(message, sst->mailbox.out_base, bytes);
+
+ for (i = 0; i < bytes; i += 4)
+ trace_sst_ipc_outbox_rdata(i, *(u32 *)(message + i));
+}
+EXPORT_SYMBOL_GPL(sst_dsp_outbox_read);
+
+void sst_dsp_inbox_write(struct sst_dsp *sst, void *message, size_t bytes)
+{
+ u32 i;
+
+ trace_sst_ipc_inbox_write(bytes);
+
+ memcpy_toio(sst->mailbox.in_base, message, bytes);
+
+ for (i = 0; i < bytes; i += 4)
+ trace_sst_ipc_inbox_wdata(i, *(u32 *)(message + i));
+}
+EXPORT_SYMBOL_GPL(sst_dsp_inbox_write);
+
+void sst_dsp_inbox_read(struct sst_dsp *sst, void *message, size_t bytes)
+{
+ u32 i;
+
+ trace_sst_ipc_inbox_read(bytes);
+
+ memcpy_fromio(message, sst->mailbox.in_base, bytes);
+
+ for (i = 0; i < bytes; i += 4)
+ trace_sst_ipc_inbox_rdata(i, *(u32 *)(message + i));
+}
+EXPORT_SYMBOL_GPL(sst_dsp_inbox_read);
+
+struct sst_dsp *sst_dsp_new(struct device *dev,
+ struct sst_dsp_device *sst_dev, struct sst_pdata *pdata)
+{
+ struct sst_dsp *sst;
+ int err;
+
+ dev_dbg(dev, "initialising audio DSP id 0x%x\n", pdata->id);
+
+ sst = devm_kzalloc(dev, sizeof(*sst), GFP_KERNEL);
+ if (sst == NULL)
+ return NULL;
+
+ spin_lock_init(&sst->spinlock);
+ mutex_init(&sst->mutex);
+ sst->dev = dev;
+ sst->thread_context = sst_dev->thread_context;
+ sst->sst_dev = sst_dev;
+ sst->id = pdata->id;
+ sst->irq = pdata->irq;
+ sst->ops = sst_dev->ops;
+ sst->pdata = pdata;
+ INIT_LIST_HEAD(&sst->used_block_list);
+ INIT_LIST_HEAD(&sst->free_block_list);
+ INIT_LIST_HEAD(&sst->module_list);
+ INIT_LIST_HEAD(&sst->fw_list);
+
+ /* Initialise SST Audio DSP */
+ if (sst->ops->init) {
+ err = sst->ops->init(sst, pdata);
+ if (err < 0)
+ return NULL;
+ }
+
+ /* Register the ISR */
+ err = request_threaded_irq(sst->irq, sst->ops->irq_handler,
+ sst_dev->thread, IRQF_SHARED, "AudioDSP", sst);
+ if (err)
+ goto irq_err;
+
+ return sst;
+
+irq_err:
+ if (sst->ops->free)
+ sst->ops->free(sst);
+
+ return NULL;
+}
+EXPORT_SYMBOL_GPL(sst_dsp_new);
+
+void sst_dsp_free(struct sst_dsp *sst)
+{
+ free_irq(sst->irq, sst);
+ if (sst->ops->free)
+ sst->ops->free(sst);
+}
+EXPORT_SYMBOL_GPL(sst_dsp_free);
+
+/* Module information */
+MODULE_AUTHOR("Liam Girdwood");
+MODULE_DESCRIPTION("Intel SST Core");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/intel/sst-dsp.h b/sound/soc/intel/sst-dsp.h
new file mode 100644
index 0000000..74052b5
--- /dev/null
+++ b/sound/soc/intel/sst-dsp.h
@@ -0,0 +1,233 @@
+/*
+ * Intel Smart Sound Technology (SST) Core
+ *
+ * Copyright (C) 2013, Intel Corporation. All rights reserved.
+ *
+ * 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.
+ *
+ */
+
+#ifndef __SOUND_SOC_SST_DSP_H
+#define __SOUND_SOC_SST_DSP_H
+
+#include <linux/kernel.h>
+#include <linux/types.h>
+#include <linux/interrupt.h>
+
+/* SST Device IDs */
+#define SST_DEV_ID_LYNX_POINT 0x33C8
+#define SST_DEV_ID_WILDCAT_POINT 0x3438
+#define SST_DEV_ID_BYT 0x0F28
+
+/* Supported SST DMA Devices */
+#define SST_DMA_TYPE_DW 1
+#define SST_DMA_TYPE_MID 2
+
+/* SST Shim register map
+ * The register naming can differ between products. Some products also
+ * contain extra functionality.
+ */
+#define SST_CSR 0x00
+#define SST_PISR 0x08
+#define SST_PIMR 0x10
+#define SST_ISRX 0x18
+#define SST_ISRD 0x20
+#define SST_IMRX 0x28
+#define SST_IMRD 0x30
+#define SST_IPCX 0x38 /* IPC IA -> SST */
+#define SST_IPCD 0x40 /* IPC SST -> IA */
+#define SST_ISRSC 0x48
+#define SST_ISRLPESC 0x50
+#define SST_IMRSC 0x58
+#define SST_IMRLPESC 0x60
+#define SST_IPCSC 0x68
+#define SST_IPCLPESC 0x70
+#define SST_CLKCTL 0x78
+#define SST_CSR2 0x80
+#define SST_LTRC 0xE0
+#define SST_HDMC 0xE8
+#define SST_DBGO 0xF0
+
+#define SST_SHIM_SIZE 0x100
+#define SST_PWMCTRL 0x1000
+
+/* SST Shim Register bits
+ * The register bit naming can differ between products. Some products also
+ * contain extra functionality.
+ */
+
+/* CSR / CS */
+#define SST_CSR_RST (0x1 << 1)
+#define SST_CSR_SBCS0 (0x1 << 2)
+#define SST_CSR_SBCS1 (0x1 << 3)
+#define SST_CSR_DCS(x) (x << 4)
+#define SST_CSR_DCS_MASK (0x7 << 4)
+#define SST_CSR_STALL (0x1 << 10)
+#define SST_CSR_S0IOCS (0x1 << 21)
+#define SST_CSR_S1IOCS (0x1 << 23)
+#define SST_CSR_LPCS (0x1 << 31)
+#define SST_BYT_CSR_RST (0x1 << 0)
+#define SST_BYT_CSR_VECTOR_SEL (0x1 << 1)
+#define SST_BYT_CSR_STALL (0x1 << 2)
+#define SST_BYT_CSR_PWAITMODE (0x1 << 3)
+
+/* ISRX / ISC */
+#define SST_ISRX_BUSY (0x1 << 1)
+#define SST_ISRX_DONE (0x1 << 0)
+#define SST_BYT_ISRX_REQUEST (0x1 << 1)
+
+/* ISRD / ISD */
+#define SST_ISRD_BUSY (0x1 << 1)
+#define SST_ISRD_DONE (0x1 << 0)
+
+/* IMRX / IMC */
+#define SST_IMRX_BUSY (0x1 << 1)
+#define SST_IMRX_DONE (0x1 << 0)
+#define SST_BYT_IMRX_REQUEST (0x1 << 1)
+
+/* IPCX / IPCC */
+#define SST_IPCX_DONE (0x1 << 30)
+#define SST_IPCX_BUSY (0x1 << 31)
+#define SST_BYT_IPCX_DONE ((u64)0x1 << 62)
+#define SST_BYT_IPCX_BUSY ((u64)0x1 << 63)
+
+/* IPCD */
+#define SST_IPCD_DONE (0x1 << 30)
+#define SST_IPCD_BUSY (0x1 << 31)
+#define SST_BYT_IPCD_DONE ((u64)0x1 << 62)
+#define SST_BYT_IPCD_BUSY ((u64)0x1 << 63)
+
+/* CLKCTL */
+#define SST_CLKCTL_SMOS(x) (x << 24)
+#define SST_CLKCTL_MASK (3 << 24)
+#define SST_CLKCTL_DCPLCG (1 << 18)
+#define SST_CLKCTL_SCOE1 (1 << 17)
+#define SST_CLKCTL_SCOE0 (1 << 16)
+
+/* CSR2 / CS2 */
+#define SST_CSR2_SDFD_SSP0 (1 << 1)
+#define SST_CSR2_SDFD_SSP1 (1 << 2)
+
+/* LTRC */
+#define SST_LTRC_VAL(x) (x << 0)
+
+/* HDMC */
+#define SST_HDMC_HDDA0(x) (x << 0)
+#define SST_HDMC_HDDA1(x) (x << 7)
+
+
+/* SST Vendor Defined Registers and bits */
+#define SST_VDRTCTL0 0xa0
+#define SST_VDRTCTL1 0xa4
+#define SST_VDRTCTL2 0xa8
+#define SST_VDRTCTL3 0xaC
+
+/* VDRTCTL0 */
+#define SST_VDRTCL0_DSRAMPGE_SHIFT 16
+#define SST_VDRTCL0_DSRAMPGE_MASK (0xffff << SST_VDRTCL0_DSRAMPGE_SHIFT)
+#define SST_VDRTCL0_ISRAMPGE_SHIFT 6
+#define SST_VDRTCL0_ISRAMPGE_MASK (0x3ff << SST_VDRTCL0_ISRAMPGE_SHIFT)
+
+struct sst_dsp;
+
+/*
+ * SST Device.
+ *
+ * This structure is populated by the SST core driver.
+ */
+struct sst_dsp_device {
+ /* Mandatory fields */
+ struct sst_ops *ops;
+ irqreturn_t (*thread)(int irq, void *context);
+ void *thread_context;
+};
+
+/*
+ * SST Platform Data.
+ */
+struct sst_pdata {
+ /* ACPI data */
+ u32 lpe_base;
+ u32 lpe_size;
+ u32 pcicfg_base;
+ u32 pcicfg_size;
+ u32 fw_base;
+ u32 fw_size;
+ int irq;
+
+ /* Firmware */
+ const struct firmware *fw;
+
+ /* DMA */
+ u32 dma_base;
+ u32 dma_size;
+ int dma_engine;
+
+ /* DSP */
+ u32 id;
+ void *dsp;
+};
+
+/* Initialization */
+struct sst_dsp *sst_dsp_new(struct device *dev,
+ struct sst_dsp_device *sst_dev, struct sst_pdata *pdata);
+void sst_dsp_free(struct sst_dsp *sst);
+
+/* SHIM Read / Write */
+void sst_dsp_shim_write(struct sst_dsp *sst, u32 offset, u32 value);
+u32 sst_dsp_shim_read(struct sst_dsp *sst, u32 offset);
+int sst_dsp_shim_update_bits(struct sst_dsp *sst, u32 offset,
+ u32 mask, u32 value);
+void sst_dsp_shim_write64(struct sst_dsp *sst, u32 offset, u64 value);
+u64 sst_dsp_shim_read64(struct sst_dsp *sst, u32 offset);
+int sst_dsp_shim_update_bits64(struct sst_dsp *sst, u32 offset,
+ u64 mask, u64 value);
+
+/* SHIM Read / Write Unlocked for callers already holding sst lock */
+void sst_dsp_shim_write_unlocked(struct sst_dsp *sst, u32 offset, u32 value);
+u32 sst_dsp_shim_read_unlocked(struct sst_dsp *sst, u32 offset);
+int sst_dsp_shim_update_bits_unlocked(struct sst_dsp *sst, u32 offset,
+ u32 mask, u32 value);
+void sst_dsp_shim_write64_unlocked(struct sst_dsp *sst, u32 offset, u64 value);
+u64 sst_dsp_shim_read64_unlocked(struct sst_dsp *sst, u32 offset);
+int sst_dsp_shim_update_bits64_unlocked(struct sst_dsp *sst, u32 offset,
+ u64 mask, u64 value);
+
+/* Internal generic low-level SST IO functions - can be overidden */
+void sst_shim32_write(void __iomem *addr, u32 offset, u32 value);
+u32 sst_shim32_read(void __iomem *addr, u32 offset);
+void sst_shim32_write64(void __iomem *addr, u32 offset, u64 value);
+u64 sst_shim32_read64(void __iomem *addr, u32 offset);
+void sst_memcpy_toio_32(struct sst_dsp *sst,
+ void __iomem *dest, void *src, size_t bytes);
+void sst_memcpy_fromio_32(struct sst_dsp *sst,
+ void *dest, void __iomem *src, size_t bytes);
+
+/* DSP reset & boot */
+void sst_dsp_reset(struct sst_dsp *sst);
+int sst_dsp_boot(struct sst_dsp *sst);
+
+/* Msg IO */
+void sst_dsp_ipc_msg_tx(struct sst_dsp *dsp, u32 msg);
+u32 sst_dsp_ipc_msg_rx(struct sst_dsp *dsp);
+
+/* Mailbox management */
+int sst_dsp_mailbox_init(struct sst_dsp *dsp, u32 inbox_offset,
+ size_t inbox_size, u32 outbox_offset, size_t outbox_size);
+void sst_dsp_inbox_write(struct sst_dsp *dsp, void *message, size_t bytes);
+void sst_dsp_inbox_read(struct sst_dsp *dsp, void *message, size_t bytes);
+void sst_dsp_outbox_write(struct sst_dsp *dsp, void *message, size_t bytes);
+void sst_dsp_outbox_read(struct sst_dsp *dsp, void *message, size_t bytes);
+void sst_dsp_mailbox_dump(struct sst_dsp *dsp, size_t bytes);
+
+/* Debug */
+void sst_dsp_dump(struct sst_dsp *sst);
+
+#endif
diff --git a/sound/soc/intel/sst-firmware.c b/sound/soc/intel/sst-firmware.c
new file mode 100644
index 0000000..f768710
--- /dev/null
+++ b/sound/soc/intel/sst-firmware.c
@@ -0,0 +1,587 @@
+/*
+ * Intel SST Firmware Loader
+ *
+ * Copyright (C) 2013, Intel Corporation. All rights reserved.
+ *
+ * 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.
+ *
+ */
+
+#include <linux/kernel.h>
+#include <linux/slab.h>
+#include <linux/sched.h>
+#include <linux/firmware.h>
+#include <linux/export.h>
+#include <linux/platform_device.h>
+#include <linux/dma-mapping.h>
+#include <linux/dmaengine.h>
+#include <linux/pci.h>
+
+#include <asm/page.h>
+#include <asm/pgtable.h>
+
+#include "sst-dsp.h"
+#include "sst-dsp-priv.h"
+
+static void sst_memcpy32(volatile void __iomem *dest, void *src, u32 bytes)
+{
+ u32 i;
+
+ /* copy one 32 bit word at a time as 64 bit access is not supported */
+ for (i = 0; i < bytes; i += 4)
+ memcpy_toio(dest + i, src + i, 4);
+}
+
+/* create new generic firmware object */
+struct sst_fw *sst_fw_new(struct sst_dsp *dsp,
+ const struct firmware *fw, void *private)
+{
+ struct sst_fw *sst_fw;
+ int err;
+
+ if (!dsp->ops->parse_fw)
+ return NULL;
+
+ sst_fw = kzalloc(sizeof(*sst_fw), GFP_KERNEL);
+ if (sst_fw == NULL)
+ return NULL;
+
+ sst_fw->dsp = dsp;
+ sst_fw->private = private;
+ sst_fw->size = fw->size;
+
+ err = dma_coerce_mask_and_coherent(dsp->dev, DMA_BIT_MASK(32));
+ if (err < 0) {
+ kfree(sst_fw);
+ return NULL;
+ }
+
+ /* allocate DMA buffer to store FW data */
+ sst_fw->dma_buf = dma_alloc_coherent(dsp->dev, sst_fw->size,
+ &sst_fw->dmable_fw_paddr, GFP_DMA | GFP_KERNEL);
+ if (!sst_fw->dma_buf) {
+ dev_err(dsp->dev, "error: DMA alloc failed\n");
+ kfree(sst_fw);
+ return NULL;
+ }
+
+ /* copy FW data to DMA-able memory */
+ memcpy((void *)sst_fw->dma_buf, (void *)fw->data, fw->size);
+
+ /* call core specific FW paser to load FW data into DSP */
+ err = dsp->ops->parse_fw(sst_fw);
+ if (err < 0) {
+ dev_err(dsp->dev, "error: parse fw failed %d\n", err);
+ goto parse_err;
+ }
+
+ mutex_lock(&dsp->mutex);
+ list_add(&sst_fw->list, &dsp->fw_list);
+ mutex_unlock(&dsp->mutex);
+
+ return sst_fw;
+
+parse_err:
+ dma_free_coherent(dsp->dev, sst_fw->size,
+ sst_fw->dma_buf,
+ sst_fw->dmable_fw_paddr);
+ kfree(sst_fw);
+ return NULL;
+}
+EXPORT_SYMBOL_GPL(sst_fw_new);
+
+/* free single firmware object */
+void sst_fw_free(struct sst_fw *sst_fw)
+{
+ struct sst_dsp *dsp = sst_fw->dsp;
+
+ mutex_lock(&dsp->mutex);
+ list_del(&sst_fw->list);
+ mutex_unlock(&dsp->mutex);
+
+ dma_free_coherent(dsp->dev, sst_fw->size, sst_fw->dma_buf,
+ sst_fw->dmable_fw_paddr);
+ kfree(sst_fw);
+}
+EXPORT_SYMBOL_GPL(sst_fw_free);
+
+/* free all firmware objects */
+void sst_fw_free_all(struct sst_dsp *dsp)
+{
+ struct sst_fw *sst_fw, *t;
+
+ mutex_lock(&dsp->mutex);
+ list_for_each_entry_safe(sst_fw, t, &dsp->fw_list, list) {
+
+ list_del(&sst_fw->list);
+ dma_free_coherent(dsp->dev, sst_fw->size, sst_fw->dma_buf,
+ sst_fw->dmable_fw_paddr);
+ kfree(sst_fw);
+ }
+ mutex_unlock(&dsp->mutex);
+}
+EXPORT_SYMBOL_GPL(sst_fw_free_all);
+
+/* create a new SST generic module from FW template */
+struct sst_module *sst_module_new(struct sst_fw *sst_fw,
+ struct sst_module_template *template, void *private)
+{
+ struct sst_dsp *dsp = sst_fw->dsp;
+ struct sst_module *sst_module;
+
+ sst_module = kzalloc(sizeof(*sst_module), GFP_KERNEL);
+ if (sst_module == NULL)
+ return NULL;
+
+ sst_module->id = template->id;
+ sst_module->dsp = dsp;
+ sst_module->sst_fw = sst_fw;
+
+ memcpy(&sst_module->s, &template->s, sizeof(struct sst_module_data));
+ memcpy(&sst_module->p, &template->p, sizeof(struct sst_module_data));
+
+ INIT_LIST_HEAD(&sst_module->block_list);
+
+ mutex_lock(&dsp->mutex);
+ list_add(&sst_module->list, &dsp->module_list);
+ mutex_unlock(&dsp->mutex);
+
+ return sst_module;
+}
+EXPORT_SYMBOL_GPL(sst_module_new);
+
+/* free firmware module and remove from available list */
+void sst_module_free(struct sst_module *sst_module)
+{
+ struct sst_dsp *dsp = sst_module->dsp;
+
+ mutex_lock(&dsp->mutex);
+ list_del(&sst_module->list);
+ mutex_unlock(&dsp->mutex);
+
+ kfree(sst_module);
+}
+EXPORT_SYMBOL_GPL(sst_module_free);
+
+static struct sst_mem_block *find_block(struct sst_dsp *dsp, int type,
+ u32 offset)
+{
+ struct sst_mem_block *block;
+
+ list_for_each_entry(block, &dsp->free_block_list, list) {
+ if (block->type == type && block->offset == offset)
+ return block;
+ }
+
+ return NULL;
+}
+
+static int block_alloc_contiguous(struct sst_module *module,
+ struct sst_module_data *data, u32 offset, int size)
+{
+ struct list_head tmp = LIST_HEAD_INIT(tmp);
+ struct sst_dsp *dsp = module->dsp;
+ struct sst_mem_block *block;
+
+ while (size > 0) {
+ block = find_block(dsp, data->type, offset);
+ if (!block) {
+ list_splice(&tmp, &dsp->free_block_list);
+ return -ENOMEM;
+ }
+
+ list_move_tail(&block->list, &tmp);
+ offset += block->size;
+ size -= block->size;
+ }
+
+ list_splice(&tmp, &dsp->used_block_list);
+ return 0;
+}
+
+/* allocate free DSP blocks for module data - callers hold locks */
+static int block_alloc(struct sst_module *module,
+ struct sst_module_data *data)
+{
+ struct sst_dsp *dsp = module->dsp;
+ struct sst_mem_block *block, *tmp;
+ int ret = 0;
+
+ if (data->size == 0)
+ return 0;
+
+ /* find first free whole blocks that can hold module */
+ list_for_each_entry_safe(block, tmp, &dsp->free_block_list, list) {
+
+ /* ignore blocks with wrong type */
+ if (block->type != data->type)
+ continue;
+
+ if (data->size > block->size)
+ continue;
+
+ data->offset = block->offset;
+ block->data_type = data->data_type;
+ block->bytes_used = data->size % block->size;
+ list_add(&block->module_list, &module->block_list);
+ list_move(&block->list, &dsp->used_block_list);
+ dev_dbg(dsp->dev, " *module %d added block %d:%d\n",
+ module->id, block->type, block->index);
+ return 0;
+ }
+
+ /* then find free multiple blocks that can hold module */
+ list_for_each_entry_safe(block, tmp, &dsp->free_block_list, list) {
+
+ /* ignore blocks with wrong type */
+ if (block->type != data->type)
+ continue;
+
+ /* do we span > 1 blocks */
+ if (data->size > block->size) {
+ ret = block_alloc_contiguous(module, data,
+ block->offset + block->size,
+ data->size - block->size);
+ if (ret == 0)
+ return ret;
+ }
+ }
+
+ /* not enough free block space */
+ return -ENOMEM;
+}
+
+/* remove module from memory - callers hold locks */
+static void block_module_remove(struct sst_module *module)
+{
+ struct sst_mem_block *block, *tmp;
+ struct sst_dsp *dsp = module->dsp;
+ int err;
+
+ /* disable each block */
+ list_for_each_entry(block, &module->block_list, module_list) {
+
+ if (block->ops && block->ops->disable) {
+ err = block->ops->disable(block);
+ if (err < 0)
+ dev_err(dsp->dev,
+ "error: cant disable block %d:%d\n",
+ block->type, block->index);
+ }
+ }
+
+ /* mark each block as free */
+ list_for_each_entry_safe(block, tmp, &module->block_list, module_list) {
+ list_del(&block->module_list);
+ list_move(&block->list, &dsp->free_block_list);
+ }
+}
+
+/* prepare the memory block to receive data from host - callers hold locks */
+static int block_module_prepare(struct sst_module *module)
+{
+ struct sst_mem_block *block;
+ int ret = 0;
+
+ /* enable each block so that's it'e ready for module P/S data */
+ list_for_each_entry(block, &module->block_list, module_list) {
+
+ if (block->ops && block->ops->enable) {
+ ret = block->ops->enable(block);
+ if (ret < 0) {
+ dev_err(module->dsp->dev,
+ "error: cant disable block %d:%d\n",
+ block->type, block->index);
+ goto err;
+ }
+ }
+ }
+ return ret;
+
+err:
+ list_for_each_entry(block, &module->block_list, module_list) {
+ if (block->ops && block->ops->disable)
+ block->ops->disable(block);
+ }
+ return ret;
+}
+
+/* allocate memory blocks for static module addresses - callers hold locks */
+static int block_alloc_fixed(struct sst_module *module,
+ struct sst_module_data *data)
+{
+ struct sst_dsp *dsp = module->dsp;
+ struct sst_mem_block *block, *tmp;
+ u32 end = data->offset + data->size, block_end;
+ int err;
+
+ /* only IRAM/DRAM blocks are managed */
+ if (data->type != SST_MEM_IRAM && data->type != SST_MEM_DRAM)
+ return 0;
+
+ /* are blocks already attached to this module */
+ list_for_each_entry_safe(block, tmp, &module->block_list, module_list) {
+
+ /* force compacting mem blocks of the same data_type */
+ if (block->data_type != data->data_type)
+ continue;
+
+ block_end = block->offset + block->size;
+
+ /* find block that holds section */
+ if (data->offset >= block->offset && end < block_end)
+ return 0;
+
+ /* does block span more than 1 section */
+ if (data->offset >= block->offset && data->offset < block_end) {
+
+ err = block_alloc_contiguous(module, data,
+ block->offset + block->size,
+ data->size - block->size + data->offset - block->offset);
+ if (err < 0)
+ return -ENOMEM;
+
+ /* module already owns blocks */
+ return 0;
+ }
+ }
+
+ /* find first free blocks that can hold section in free list */
+ list_for_each_entry_safe(block, tmp, &dsp->free_block_list, list) {
+ block_end = block->offset + block->size;
+
+ /* find block that holds section */
+ if (data->offset >= block->offset && end < block_end) {
+
+ /* add block */
+ block->data_type = data->data_type;
+ list_move(&block->list, &dsp->used_block_list);
+ list_add(&block->module_list, &module->block_list);
+ return 0;
+ }
+
+ /* does block span more than 1 section */
+ if (data->offset >= block->offset && data->offset < block_end) {
+
+ err = block_alloc_contiguous(module, data,
+ block->offset + block->size,
+ data->size - block->size);
+ if (err < 0)
+ return -ENOMEM;
+
+ /* add block */
+ block->data_type = data->data_type;
+ list_move(&block->list, &dsp->used_block_list);
+ list_add(&block->module_list, &module->block_list);
+ return 0;
+ }
+
+ }
+
+ return -ENOMEM;
+}
+
+/* Load fixed module data into DSP memory blocks */
+int sst_module_insert_fixed_block(struct sst_module *module,
+ struct sst_module_data *data)
+{
+ struct sst_dsp *dsp = module->dsp;
+ int ret;
+
+ mutex_lock(&dsp->mutex);
+
+ /* alloc blocks that includes this section */
+ ret = block_alloc_fixed(module, data);
+ if (ret < 0) {
+ dev_err(dsp->dev,
+ "error: no free blocks for section at offset 0x%x size 0x%x\n",
+ data->offset, data->size);
+ mutex_unlock(&dsp->mutex);
+ return -ENOMEM;
+ }
+
+ /* prepare DSP blocks for module copy */
+ ret = block_module_prepare(module);
+ if (ret < 0) {
+ dev_err(dsp->dev, "error: fw module prepare failed\n");
+ goto err;
+ }
+
+ /* copy partial module data to blocks */
+ sst_memcpy32(dsp->addr.lpe + data->offset, data->data, data->size);
+
+ mutex_unlock(&dsp->mutex);
+ return ret;
+
+err:
+ block_module_remove(module);
+ mutex_unlock(&dsp->mutex);
+ return ret;
+}
+EXPORT_SYMBOL_GPL(sst_module_insert_fixed_block);
+
+/* Unload entire module from DSP memory */
+int sst_block_module_remove(struct sst_module *module)
+{
+ struct sst_dsp *dsp = module->dsp;
+
+ mutex_lock(&dsp->mutex);
+ block_module_remove(module);
+ mutex_unlock(&dsp->mutex);
+ return 0;
+}
+EXPORT_SYMBOL_GPL(sst_block_module_remove);
+
+/* register a DSP memory block for use with FW based modules */
+struct sst_mem_block *sst_mem_block_register(struct sst_dsp *dsp, u32 offset,
+ u32 size, enum sst_mem_type type, struct sst_block_ops *ops, u32 index,
+ void *private)
+{
+ struct sst_mem_block *block;
+
+ block = kzalloc(sizeof(*block), GFP_KERNEL);
+ if (block == NULL)
+ return NULL;
+
+ block->offset = offset;
+ block->size = size;
+ block->index = index;
+ block->type = type;
+ block->dsp = dsp;
+ block->private = private;
+ block->ops = ops;
+
+ mutex_lock(&dsp->mutex);
+ list_add(&block->list, &dsp->free_block_list);
+ mutex_unlock(&dsp->mutex);
+
+ return block;
+}
+EXPORT_SYMBOL_GPL(sst_mem_block_register);
+
+/* unregister all DSP memory blocks */
+void sst_mem_block_unregister_all(struct sst_dsp *dsp)
+{
+ struct sst_mem_block *block, *tmp;
+
+ mutex_lock(&dsp->mutex);
+
+ /* unregister used blocks */
+ list_for_each_entry_safe(block, tmp, &dsp->used_block_list, list) {
+ list_del(&block->list);
+ kfree(block);
+ }
+
+ /* unregister free blocks */
+ list_for_each_entry_safe(block, tmp, &dsp->free_block_list, list) {
+ list_del(&block->list);
+ kfree(block);
+ }
+
+ mutex_unlock(&dsp->mutex);
+}
+EXPORT_SYMBOL_GPL(sst_mem_block_unregister_all);
+
+/* allocate scratch buffer blocks */
+struct sst_module *sst_mem_block_alloc_scratch(struct sst_dsp *dsp)
+{
+ struct sst_module *sst_module, *scratch;
+ struct sst_mem_block *block, *tmp;
+ u32 block_size;
+ int ret = 0;
+
+ scratch = kzalloc(sizeof(struct sst_module), GFP_KERNEL);
+ if (scratch == NULL)
+ return NULL;
+
+ mutex_lock(&dsp->mutex);
+
+ /* calculate required scratch size */
+ list_for_each_entry(sst_module, &dsp->module_list, list) {
+ if (scratch->s.size > sst_module->s.size)
+ scratch->s.size = scratch->s.size;
+ else
+ scratch->s.size = sst_module->s.size;
+ }
+
+ dev_dbg(dsp->dev, "scratch buffer required is %d bytes\n",
+ scratch->s.size);
+
+ /* init scratch module */
+ scratch->dsp = dsp;
+ scratch->s.type = SST_MEM_DRAM;
+ scratch->s.data_type = SST_DATA_S;
+ INIT_LIST_HEAD(&scratch->block_list);
+
+ /* check free blocks before looking at used blocks for space */
+ if (!list_empty(&dsp->free_block_list))
+ block = list_first_entry(&dsp->free_block_list,
+ struct sst_mem_block, list);
+ else
+ block = list_first_entry(&dsp->used_block_list,
+ struct sst_mem_block, list);
+ block_size = block->size;
+
+ /* allocate blocks for module scratch buffers */
+ dev_dbg(dsp->dev, "allocating scratch blocks\n");
+ ret = block_alloc(scratch, &scratch->s);
+ if (ret < 0) {
+ dev_err(dsp->dev, "error: can't alloc scratch blocks\n");
+ goto err;
+ }
+
+ /* assign the same offset of scratch to each module */
+ list_for_each_entry(sst_module, &dsp->module_list, list)
+ sst_module->s.offset = scratch->s.offset;
+
+ mutex_unlock(&dsp->mutex);
+ return scratch;
+
+err:
+ list_for_each_entry_safe(block, tmp, &scratch->block_list, module_list)
+ list_del(&block->module_list);
+ mutex_unlock(&dsp->mutex);
+ return NULL;
+}
+EXPORT_SYMBOL_GPL(sst_mem_block_alloc_scratch);
+
+/* free all scratch blocks */
+void sst_mem_block_free_scratch(struct sst_dsp *dsp,
+ struct sst_module *scratch)
+{
+ struct sst_mem_block *block, *tmp;
+
+ mutex_lock(&dsp->mutex);
+
+ list_for_each_entry_safe(block, tmp, &scratch->block_list, module_list)
+ list_del(&block->module_list);
+
+ mutex_unlock(&dsp->mutex);
+}
+EXPORT_SYMBOL_GPL(sst_mem_block_free_scratch);
+
+/* get a module from it's unique ID */
+struct sst_module *sst_module_get_from_id(struct sst_dsp *dsp, u32 id)
+{
+ struct sst_module *module;
+
+ mutex_lock(&dsp->mutex);
+
+ list_for_each_entry(module, &dsp->module_list, list) {
+ if (module->id == id) {
+ mutex_unlock(&dsp->mutex);
+ return module;
+ }
+ }
+
+ mutex_unlock(&dsp->mutex);
+ return NULL;
+}
+EXPORT_SYMBOL_GPL(sst_module_get_from_id);
diff --git a/sound/soc/intel/sst-haswell-dsp.c b/sound/soc/intel/sst-haswell-dsp.c
new file mode 100644
index 0000000..f5ebf36
--- /dev/null
+++ b/sound/soc/intel/sst-haswell-dsp.c
@@ -0,0 +1,517 @@
+/*
+ * Intel Haswell SST DSP driver
+ *
+ * Copyright (C) 2013, Intel Corporation. All rights reserved.
+ *
+ * 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.
+ *
+ */
+
+#include <linux/delay.h>
+#include <linux/fs.h>
+#include <linux/slab.h>
+#include <linux/device.h>
+#include <linux/sched.h>
+#include <linux/export.h>
+#include <linux/interrupt.h>
+#include <linux/module.h>
+#include <linux/dma-mapping.h>
+#include <linux/platform_device.h>
+#include <linux/pci.h>
+#include <linux/firmware.h>
+#include <linux/pm_runtime.h>
+
+#include <linux/acpi.h>
+#include <acpi/acpi_bus.h>
+
+#include "sst-dsp.h"
+#include "sst-dsp-priv.h"
+#include "sst-haswell-ipc.h"
+
+#include <trace/events/hswadsp.h>
+
+#define SST_HSW_FW_SIGNATURE_SIZE 4
+#define SST_HSW_FW_SIGN "$SST"
+#define SST_HSW_FW_LIB_SIGN "$LIB"
+
+#define SST_WPT_SHIM_OFFSET 0xFB000
+#define SST_LP_SHIM_OFFSET 0xE7000
+#define SST_WPT_IRAM_OFFSET 0xA0000
+#define SST_LP_IRAM_OFFSET 0x80000
+
+#define SST_SHIM_PM_REG 0x84
+
+#define SST_HSW_IRAM 1
+#define SST_HSW_DRAM 2
+#define SST_HSW_REGS 3
+
+struct dma_block_info {
+ __le32 type; /* IRAM/DRAM */
+ __le32 size; /* Bytes */
+ __le32 ram_offset; /* Offset in I/DRAM */
+ __le32 rsvd; /* Reserved field */
+} __attribute__((packed));
+
+struct fw_module_info {
+ __le32 persistent_size;
+ __le32 scratch_size;
+} __attribute__((packed));
+
+struct fw_header {
+ unsigned char signature[SST_HSW_FW_SIGNATURE_SIZE]; /* FW signature */
+ __le32 file_size; /* size of fw minus this header */
+ __le32 modules; /* # of modules */
+ __le32 file_format; /* version of header format */
+ __le32 reserved[4];
+} __attribute__((packed));
+
+struct fw_module_header {
+ unsigned char signature[SST_HSW_FW_SIGNATURE_SIZE]; /* module signature */
+ __le32 mod_size; /* size of module */
+ __le32 blocks; /* # of blocks */
+ __le16 padding;
+ __le16 type; /* codec type, pp lib */
+ __le32 entry_point;
+ struct fw_module_info info;
+} __attribute__((packed));
+
+static void hsw_free(struct sst_dsp *sst);
+
+static int hsw_parse_module(struct sst_dsp *dsp, struct sst_fw *fw,
+ struct fw_module_header *module)
+{
+ struct dma_block_info *block;
+ struct sst_module *mod;
+ struct sst_module_data block_data;
+ struct sst_module_template template;
+ int count;
+ void __iomem *ram;
+
+ /* TODO: allowed module types need to be configurable */
+ if (module->type != SST_HSW_MODULE_BASE_FW
+ && module->type != SST_HSW_MODULE_PCM_SYSTEM
+ && module->type != SST_HSW_MODULE_PCM
+ && module->type != SST_HSW_MODULE_PCM_REFERENCE
+ && module->type != SST_HSW_MODULE_PCM_CAPTURE
+ && module->type != SST_HSW_MODULE_LPAL)
+ return 0;
+
+ dev_dbg(dsp->dev, "new module sign 0x%s size 0x%x blocks 0x%x type 0x%x\n",
+ module->signature, module->mod_size,
+ module->blocks, module->type);
+ dev_dbg(dsp->dev, " entrypoint 0x%x\n", module->entry_point);
+ dev_dbg(dsp->dev, " persistent 0x%x scratch 0x%x\n",
+ module->info.persistent_size, module->info.scratch_size);
+
+ memset(&template, 0, sizeof(template));
+ template.id = module->type;
+ template.entry = module->entry_point;
+ template.p.size = module->info.persistent_size;
+ template.p.type = SST_MEM_DRAM;
+ template.p.data_type = SST_DATA_P;
+ template.s.size = module->info.scratch_size;
+ template.s.type = SST_MEM_DRAM;
+ template.s.data_type = SST_DATA_S;
+
+ mod = sst_module_new(fw, &template, NULL);
+ if (mod == NULL)
+ return -ENOMEM;
+
+ block = (void *)module + sizeof(*module);
+
+ for (count = 0; count < module->blocks; count++) {
+
+ if (block->size <= 0) {
+ dev_err(dsp->dev,
+ "error: block %d size invalid\n", count);
+ sst_module_free(mod);
+ return -EINVAL;
+ }
+
+ switch (block->type) {
+ case SST_HSW_IRAM:
+ ram = dsp->addr.lpe;
+ block_data.offset =
+ block->ram_offset + dsp->addr.iram_offset;
+ block_data.type = SST_MEM_IRAM;
+ break;
+ case SST_HSW_DRAM:
+ ram = dsp->addr.lpe;
+ block_data.offset = block->ram_offset;
+ block_data.type = SST_MEM_DRAM;
+ break;
+ default:
+ dev_err(dsp->dev, "error: bad type 0x%x for block 0x%x\n",
+ block->type, count);
+ sst_module_free(mod);
+ return -EINVAL;
+ }
+
+ block_data.size = block->size;
+ block_data.data_type = SST_DATA_M;
+ block_data.data = (void *)block + sizeof(*block);
+ block_data.data_offset = block_data.data - fw->dma_buf;
+
+ dev_dbg(dsp->dev, "copy firmware block %d type 0x%x "
+ "size 0x%x ==> ram %p offset 0x%x\n",
+ count, block->type, block->size, ram,
+ block->ram_offset);
+
+ sst_module_insert_fixed_block(mod, &block_data);
+
+ block = (void *)block + sizeof(*block) + block->size;
+ }
+ return 0;
+}
+
+static int hsw_parse_fw_image(struct sst_fw *sst_fw)
+{
+ struct fw_header *header;
+ struct sst_module *scratch;
+ struct fw_module_header *module;
+ struct sst_dsp *dsp = sst_fw->dsp;
+ struct sst_hsw *hsw = sst_fw->private;
+ int ret, count;
+
+ /* Read the header information from the data pointer */
+ header = (struct fw_header *)sst_fw->dma_buf;
+
+ /* verify FW */
+ if ((strncmp(header->signature, SST_HSW_FW_SIGN, 4) != 0) ||
+ (sst_fw->size != header->file_size + sizeof(*header))) {
+ dev_err(dsp->dev, "error: invalid fw sign/filesize mismatch\n");
+ return -EINVAL;
+ }
+
+ dev_dbg(dsp->dev, "header size=0x%x modules=0x%x fmt=0x%x size=%zu\n",
+ header->file_size, header->modules,
+ header->file_format, sizeof(*header));
+
+ /* parse each module */
+ module = (void *)sst_fw->dma_buf + sizeof(*header);
+ for (count = 0; count < header->modules; count++) {
+
+ /* module */
+ ret = hsw_parse_module(dsp, sst_fw, module);
+ if (ret < 0) {
+ dev_err(dsp->dev, "error: invalid module %d\n", count);
+ return ret;
+ }
+ module = (void *)module + sizeof(*module) + module->mod_size;
+ }
+
+ /* allocate persistent/scratch mem regions */
+ scratch = sst_mem_block_alloc_scratch(dsp);
+ if (scratch == NULL)
+ return -ENOMEM;
+
+ sst_hsw_set_scratch_module(hsw, scratch);
+
+ return 0;
+}
+
+static irqreturn_t hsw_irq(int irq, void *context)
+{
+ struct sst_dsp *sst = (struct sst_dsp *) context;
+ u32 isr;
+ int ret = IRQ_NONE;
+
+ spin_lock(&sst->spinlock);
+
+ /* Interrupt arrived, check src */
+ isr = sst_dsp_shim_read_unlocked(sst, SST_ISRX);
+ if (isr & SST_ISRX_DONE) {
+ trace_sst_irq_done(isr,
+ sst_dsp_shim_read_unlocked(sst, SST_IMRX));
+
+ /* Mask Done interrupt before return */
+ sst_dsp_shim_update_bits_unlocked(sst, SST_IMRX,
+ SST_IMRX_DONE, SST_IMRX_DONE);
+ ret = IRQ_WAKE_THREAD;
+ }
+
+ if (isr & SST_ISRX_BUSY) {
+ trace_sst_irq_busy(isr,
+ sst_dsp_shim_read_unlocked(sst, SST_IMRX));
+
+ /* Mask Busy interrupt before return */
+ sst_dsp_shim_update_bits_unlocked(sst, SST_IMRX,
+ SST_IMRX_BUSY, SST_IMRX_BUSY);
+ ret = IRQ_WAKE_THREAD;
+ }
+
+ spin_unlock(&sst->spinlock);
+ return ret;
+}
+
+static void hsw_boot(struct sst_dsp *sst)
+{
+ /* select SSP1 19.2MHz base clock, SSP clock 0, turn off Low Power Clock */
+ sst_dsp_shim_update_bits_unlocked(sst, SST_CSR,
+ SST_CSR_S1IOCS | SST_CSR_SBCS1 | SST_CSR_LPCS, 0x0);
+
+ /* stall DSP core, set clk to 192/96Mhz */
+ sst_dsp_shim_update_bits_unlocked(sst,
+ SST_CSR, SST_CSR_STALL | SST_CSR_DCS_MASK,
+ SST_CSR_STALL | SST_CSR_DCS(4));
+
+ /* Set 24MHz MCLK, prevent local clock gating, enable SSP0 clock */
+ sst_dsp_shim_update_bits_unlocked(sst, SST_CLKCTL,
+ SST_CLKCTL_MASK | SST_CLKCTL_DCPLCG | SST_CLKCTL_SCOE0,
+ SST_CLKCTL_MASK | SST_CLKCTL_DCPLCG | SST_CLKCTL_SCOE0);
+
+ /* disable DMA finish function for SSP0 & SSP1 */
+ sst_dsp_shim_update_bits_unlocked(sst, SST_CSR2, SST_CSR2_SDFD_SSP1,
+ SST_CSR2_SDFD_SSP1);
+
+ /* enable DMA engine 0,1 all channels to access host memory */
+ sst_dsp_shim_update_bits_unlocked(sst, SST_HDMC,
+ SST_HDMC_HDDA1(0xff) | SST_HDMC_HDDA0(0xff),
+ SST_HDMC_HDDA1(0xff) | SST_HDMC_HDDA0(0xff));
+
+ /* disable all clock gating */
+ writel(0x0, sst->addr.pci_cfg + SST_VDRTCTL2);
+
+ /* set DSP to RUN */
+ sst_dsp_shim_update_bits_unlocked(sst, SST_CSR, SST_CSR_STALL, 0x0);
+}
+
+static void hsw_reset(struct sst_dsp *sst)
+{
+ /* put DSP into reset and stall */
+ sst_dsp_shim_update_bits_unlocked(sst, SST_CSR,
+ SST_CSR_RST | SST_CSR_STALL, SST_CSR_RST | SST_CSR_STALL);
+
+ /* keep in reset for 10ms */
+ mdelay(10);
+
+ /* take DSP out of reset and keep stalled for FW loading */
+ sst_dsp_shim_update_bits_unlocked(sst, SST_CSR,
+ SST_CSR_RST | SST_CSR_STALL, SST_CSR_STALL);
+}
+
+struct sst_adsp_memregion {
+ u32 start;
+ u32 end;
+ int blocks;
+ enum sst_mem_type type;
+};
+
+/* lynx point ADSP mem regions */
+static const struct sst_adsp_memregion lp_region[] = {
+ {0x00000, 0x40000, 8, SST_MEM_DRAM}, /* D-SRAM0 - 8 * 32kB */
+ {0x40000, 0x80000, 8, SST_MEM_DRAM}, /* D-SRAM1 - 8 * 32kB */
+ {0x80000, 0xE0000, 12, SST_MEM_IRAM}, /* I-SRAM - 12 * 32kB */
+};
+
+/* wild cat point ADSP mem regions */
+static const struct sst_adsp_memregion wpt_region[] = {
+ {0x00000, 0x40000, 8, SST_MEM_DRAM}, /* D-SRAM0 - 8 * 32kB */
+ {0x40000, 0x80000, 8, SST_MEM_DRAM}, /* D-SRAM1 - 8 * 32kB */
+ {0x80000, 0xA0000, 4, SST_MEM_DRAM}, /* D-SRAM2 - 4 * 32kB */
+ {0xA0000, 0xF0000, 10, SST_MEM_IRAM}, /* I-SRAM - 10 * 32kB */
+};
+
+static int hsw_acpi_resource_map(struct sst_dsp *sst, struct sst_pdata *pdata)
+{
+ /* ADSP DRAM & IRAM */
+ sst->addr.lpe_base = pdata->lpe_base;
+ sst->addr.lpe = ioremap(pdata->lpe_base, pdata->lpe_size);
+ if (!sst->addr.lpe)
+ return -ENODEV;
+
+ /* ADSP PCI MMIO config space */
+ sst->addr.pci_cfg = ioremap(pdata->pcicfg_base, pdata->pcicfg_size);
+ if (!sst->addr.pci_cfg) {
+ iounmap(sst->addr.lpe);
+ return -ENODEV;
+ }
+
+ /* SST Shim */
+ sst->addr.shim = sst->addr.lpe + sst->addr.shim_offset;
+ return 0;
+}
+
+static u32 hsw_block_get_bit(struct sst_mem_block *block)
+{
+ u32 bit = 0, shift = 0;
+
+ switch (block->type) {
+ case SST_MEM_DRAM:
+ shift = 16;
+ break;
+ case SST_MEM_IRAM:
+ shift = 6;
+ break;
+ default:
+ return 0;
+ }
+
+ bit = 1 << (block->index + shift);
+
+ return bit;
+}
+
+/* enable 32kB memory block - locks held by caller */
+static int hsw_block_enable(struct sst_mem_block *block)
+{
+ struct sst_dsp *sst = block->dsp;
+ u32 bit, val;
+
+ if (block->users++ > 0)
+ return 0;
+
+ dev_dbg(block->dsp->dev, " enabled block %d:%d at offset 0x%x\n",
+ block->type, block->index, block->offset);
+
+ val = readl(sst->addr.pci_cfg + SST_VDRTCTL0);
+ bit = hsw_block_get_bit(block);
+ writel(val & ~bit, sst->addr.pci_cfg + SST_VDRTCTL0);
+
+ /* wait 18 DSP clock ticks */
+ udelay(10);
+
+ return 0;
+}
+
+/* disable 32kB memory block - locks held by caller */
+static int hsw_block_disable(struct sst_mem_block *block)
+{
+ struct sst_dsp *sst = block->dsp;
+ u32 bit, val;
+
+ if (--block->users > 0)
+ return 0;
+
+ dev_dbg(block->dsp->dev, " disabled block %d:%d at offset 0x%x\n",
+ block->type, block->index, block->offset);
+
+ val = readl(sst->addr.pci_cfg + SST_VDRTCTL0);
+ bit = hsw_block_get_bit(block);
+ writel(val | bit, sst->addr.pci_cfg + SST_VDRTCTL0);
+
+ return 0;
+}
+
+static struct sst_block_ops sst_hsw_ops = {
+ .enable = hsw_block_enable,
+ .disable = hsw_block_disable,
+};
+
+static int hsw_enable_shim(struct sst_dsp *sst)
+{
+ int tries = 10;
+ u32 reg;
+
+ /* enable shim */
+ reg = readl(sst->addr.pci_cfg + SST_SHIM_PM_REG);
+ writel(reg & ~0x3, sst->addr.pci_cfg + SST_SHIM_PM_REG);
+
+ /* check that ADSP shim is enabled */
+ while (tries--) {
+ reg = sst_dsp_shim_read_unlocked(sst, SST_CSR);
+ if (reg != 0xffffffff)
+ return 0;
+
+ msleep(1);
+ }
+
+ return -ENODEV;
+}
+
+static int hsw_init(struct sst_dsp *sst, struct sst_pdata *pdata)
+{
+ const struct sst_adsp_memregion *region;
+ struct device *dev;
+ int ret = -ENODEV, i, j, region_count;
+ u32 offset, size;
+
+ dev = sst->dev;
+
+ switch (sst->id) {
+ case SST_DEV_ID_LYNX_POINT:
+ region = lp_region;
+ region_count = ARRAY_SIZE(lp_region);
+ sst->addr.iram_offset = SST_LP_IRAM_OFFSET;
+ sst->addr.shim_offset = SST_LP_SHIM_OFFSET;
+ break;
+ case SST_DEV_ID_WILDCAT_POINT:
+ region = wpt_region;
+ region_count = ARRAY_SIZE(wpt_region);
+ sst->addr.iram_offset = SST_WPT_IRAM_OFFSET;
+ sst->addr.shim_offset = SST_WPT_SHIM_OFFSET;
+ break;
+ default:
+ dev_err(dev, "error: failed to get mem resources\n");
+ return ret;
+ }
+
+ ret = hsw_acpi_resource_map(sst, pdata);
+ if (ret < 0) {
+ dev_err(dev, "error: failed to map resources\n");
+ return ret;
+ }
+
+ /* enable the DSP SHIM */
+ ret = hsw_enable_shim(sst);
+ if (ret < 0) {
+ dev_err(dev, "error: failed to set DSP D0 and reset SHIM\n");
+ return ret;
+ }
+
+ ret = dma_coerce_mask_and_coherent(dev, DMA_BIT_MASK(32));
+ if (ret)
+ return ret;
+
+ /* Enable Interrupt from both sides */
+ sst_dsp_shim_update_bits_unlocked(sst, SST_IMRX, 0x3, 0x0);
+ sst_dsp_shim_update_bits_unlocked(sst, SST_IMRD,
+ (0x3 | 0x1 << 16 | 0x3 << 21), 0x0);
+
+ /* register DSP memory blocks - ideally we should get this from ACPI */
+ for (i = 0; i < region_count; i++) {
+ offset = region[i].start;
+ size = (region[i].end - region[i].start) / region[i].blocks;
+
+ /* register individual memory blocks */
+ for (j = 0; j < region[i].blocks; j++) {
+ sst_mem_block_register(sst, offset, size,
+ region[i].type, &sst_hsw_ops, j, sst);
+ offset += size;
+ }
+ }
+
+ /* set default power gating mask */
+ writel(0x0, sst->addr.pci_cfg + SST_VDRTCTL0);
+
+ return 0;
+}
+
+static void hsw_free(struct sst_dsp *sst)
+{
+ sst_mem_block_unregister_all(sst);
+ iounmap(sst->addr.lpe);
+ iounmap(sst->addr.pci_cfg);
+}
+
+struct sst_ops haswell_ops = {
+ .reset = hsw_reset,
+ .boot = hsw_boot,
+ .write = sst_shim32_write,
+ .read = sst_shim32_read,
+ .write64 = sst_shim32_write64,
+ .read64 = sst_shim32_read64,
+ .ram_read = sst_memcpy_fromio_32,
+ .ram_write = sst_memcpy_toio_32,
+ .irq_handler = hsw_irq,
+ .init = hsw_init,
+ .free = hsw_free,
+ .parse_fw = hsw_parse_fw_image,
+};
diff --git a/sound/soc/intel/sst-haswell-ipc.c b/sound/soc/intel/sst-haswell-ipc.c
new file mode 100644
index 0000000..f46bb4d
--- /dev/null
+++ b/sound/soc/intel/sst-haswell-ipc.c
@@ -0,0 +1,1785 @@
+/*
+ * Intel SST Haswell/Broadwell IPC Support
+ *
+ * Copyright (C) 2013, Intel Corporation. All rights reserved.
+ *
+ * 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.
+ *
+ */
+
+#include <linux/types.h>
+#include <linux/kernel.h>
+#include <linux/list.h>
+#include <linux/device.h>
+#include <linux/wait.h>
+#include <linux/spinlock.h>
+#include <linux/workqueue.h>
+#include <linux/export.h>
+#include <linux/slab.h>
+#include <linux/delay.h>
+#include <linux/sched.h>
+#include <linux/list.h>
+#include <linux/platform_device.h>
+#include <linux/kthread.h>
+#include <linux/firmware.h>
+#include <linux/dma-mapping.h>
+#include <linux/debugfs.h>
+
+#include "sst-haswell-ipc.h"
+#include "sst-dsp.h"
+#include "sst-dsp-priv.h"
+
+/* Global Message - Generic */
+#define IPC_GLB_TYPE_SHIFT 24
+#define IPC_GLB_TYPE_MASK (0x1f << IPC_GLB_TYPE_SHIFT)
+#define IPC_GLB_TYPE(x) (x << IPC_GLB_TYPE_SHIFT)
+
+/* Global Message - Reply */
+#define IPC_GLB_REPLY_SHIFT 0
+#define IPC_GLB_REPLY_MASK (0x1f << IPC_GLB_REPLY_SHIFT)
+#define IPC_GLB_REPLY_TYPE(x) (x << IPC_GLB_REPLY_TYPE_SHIFT)
+
+/* Stream Message - Generic */
+#define IPC_STR_TYPE_SHIFT 20
+#define IPC_STR_TYPE_MASK (0xf << IPC_STR_TYPE_SHIFT)
+#define IPC_STR_TYPE(x) (x << IPC_STR_TYPE_SHIFT)
+#define IPC_STR_ID_SHIFT 16
+#define IPC_STR_ID_MASK (0xf << IPC_STR_ID_SHIFT)
+#define IPC_STR_ID(x) (x << IPC_STR_ID_SHIFT)
+
+/* Stream Message - Reply */
+#define IPC_STR_REPLY_SHIFT 0
+#define IPC_STR_REPLY_MASK (0x1f << IPC_STR_REPLY_SHIFT)
+
+/* Stream Stage Message - Generic */
+#define IPC_STG_TYPE_SHIFT 12
+#define IPC_STG_TYPE_MASK (0xf << IPC_STG_TYPE_SHIFT)
+#define IPC_STG_TYPE(x) (x << IPC_STG_TYPE_SHIFT)
+#define IPC_STG_ID_SHIFT 10
+#define IPC_STG_ID_MASK (0x3 << IPC_STG_ID_SHIFT)
+#define IPC_STG_ID(x) (x << IPC_STG_ID_SHIFT)
+
+/* Stream Stage Message - Reply */
+#define IPC_STG_REPLY_SHIFT 0
+#define IPC_STG_REPLY_MASK (0x1f << IPC_STG_REPLY_SHIFT)
+
+/* Debug Log Message - Generic */
+#define IPC_LOG_OP_SHIFT 20
+#define IPC_LOG_OP_MASK (0xf << IPC_LOG_OP_SHIFT)
+#define IPC_LOG_OP_TYPE(x) (x << IPC_LOG_OP_SHIFT)
+#define IPC_LOG_ID_SHIFT 16
+#define IPC_LOG_ID_MASK (0xf << IPC_LOG_ID_SHIFT)
+#define IPC_LOG_ID(x) (x << IPC_LOG_ID_SHIFT)
+
+/* IPC message timeout (msecs) */
+#define IPC_TIMEOUT_MSECS 300
+#define IPC_BOOT_MSECS 200
+#define IPC_MSG_WAIT 0
+#define IPC_MSG_NOWAIT 1
+
+/* Firmware Ready Message */
+#define IPC_FW_READY (0x1 << 29)
+#define IPC_STATUS_MASK (0x3 << 30)
+
+#define IPC_EMPTY_LIST_SIZE 8
+#define IPC_MAX_STREAMS 4
+
+/* Mailbox */
+#define IPC_MAX_MAILBOX_BYTES 256
+
+/* Global Message - Types and Replies */
+enum ipc_glb_type {
+ IPC_GLB_GET_FW_VERSION = 0, /* Retrieves firmware version */
+ IPC_GLB_PERFORMANCE_MONITOR = 1, /* Performance monitoring actions */
+ IPC_GLB_ALLOCATE_STREAM = 3, /* Request to allocate new stream */
+ IPC_GLB_FREE_STREAM = 4, /* Request to free stream */
+ IPC_GLB_GET_FW_CAPABILITIES = 5, /* Retrieves firmware capabilities */
+ IPC_GLB_STREAM_MESSAGE = 6, /* Message directed to stream or its stages */
+ /* Request to store firmware context during D0->D3 transition */
+ IPC_GLB_REQUEST_DUMP = 7,
+ /* Request to restore firmware context during D3->D0 transition */
+ IPC_GLB_RESTORE_CONTEXT = 8,
+ IPC_GLB_GET_DEVICE_FORMATS = 9, /* Set device format */
+ IPC_GLB_SET_DEVICE_FORMATS = 10, /* Get device format */
+ IPC_GLB_SHORT_REPLY = 11,
+ IPC_GLB_ENTER_DX_STATE = 12,
+ IPC_GLB_GET_MIXER_STREAM_INFO = 13, /* Request mixer stream params */
+ IPC_GLB_DEBUG_LOG_MESSAGE = 14, /* Message to or from the debug logger. */
+ IPC_GLB_REQUEST_TRANSFER = 16, /* < Request Transfer for host */
+ IPC_GLB_MAX_IPC_MESSAGE_TYPE = 17, /* Maximum message number */
+};
+
+enum ipc_glb_reply {
+ IPC_GLB_REPLY_SUCCESS = 0, /* The operation was successful. */
+ IPC_GLB_REPLY_ERROR_INVALID_PARAM = 1, /* Invalid parameter was passed. */
+ IPC_GLB_REPLY_UNKNOWN_MESSAGE_TYPE = 2, /* Uknown message type was resceived. */
+ IPC_GLB_REPLY_OUT_OF_RESOURCES = 3, /* No resources to satisfy the request. */
+ IPC_GLB_REPLY_BUSY = 4, /* The system or resource is busy. */
+ IPC_GLB_REPLY_PENDING = 5, /* The action was scheduled for processing. */
+ IPC_GLB_REPLY_FAILURE = 6, /* Critical error happened. */
+ IPC_GLB_REPLY_INVALID_REQUEST = 7, /* Request can not be completed. */
+ IPC_GLB_REPLY_STAGE_UNINITIALIZED = 8, /* Processing stage was uninitialized. */
+ IPC_GLB_REPLY_NOT_FOUND = 9, /* Required resource can not be found. */
+ IPC_GLB_REPLY_SOURCE_NOT_STARTED = 10, /* Source was not started. */
+};
+
+/* Stream Message - Types */
+enum ipc_str_operation {
+ IPC_STR_RESET = 0,
+ IPC_STR_PAUSE = 1,
+ IPC_STR_RESUME = 2,
+ IPC_STR_STAGE_MESSAGE = 3,
+ IPC_STR_NOTIFICATION = 4,
+ IPC_STR_MAX_MESSAGE
+};
+
+/* Stream Stage Message Types */
+enum ipc_stg_operation {
+ IPC_STG_GET_VOLUME = 0,
+ IPC_STG_SET_VOLUME,
+ IPC_STG_SET_WRITE_POSITION,
+ IPC_STG_SET_FX_ENABLE,
+ IPC_STG_SET_FX_DISABLE,
+ IPC_STG_SET_FX_GET_PARAM,
+ IPC_STG_SET_FX_SET_PARAM,
+ IPC_STG_SET_FX_GET_INFO,
+ IPC_STG_MUTE_LOOPBACK,
+ IPC_STG_MAX_MESSAGE
+};
+
+/* Stream Stage Message Types For Notification*/
+enum ipc_stg_operation_notify {
+ IPC_POSITION_CHANGED = 0,
+ IPC_STG_GLITCH,
+ IPC_STG_MAX_NOTIFY
+};
+
+enum ipc_glitch_type {
+ IPC_GLITCH_UNDERRUN = 1,
+ IPC_GLITCH_DECODER_ERROR,
+ IPC_GLITCH_DOUBLED_WRITE_POS,
+ IPC_GLITCH_MAX
+};
+
+/* Debug Control */
+enum ipc_debug_operation {
+ IPC_DEBUG_ENABLE_LOG = 0,
+ IPC_DEBUG_DISABLE_LOG = 1,
+ IPC_DEBUG_REQUEST_LOG_DUMP = 2,
+ IPC_DEBUG_NOTIFY_LOG_DUMP = 3,
+ IPC_DEBUG_MAX_DEBUG_LOG
+};
+
+/* Firmware Ready */
+struct sst_hsw_ipc_fw_ready {
+ u32 inbox_offset;
+ u32 outbox_offset;
+ u32 inbox_size;
+ u32 outbox_size;
+ u32 fw_info_size;
+ u8 fw_info[1];
+} __attribute__((packed));
+
+struct ipc_message {
+ struct list_head list;
+ u32 header;
+
+ /* direction wrt host CPU */
+ char tx_data[IPC_MAX_MAILBOX_BYTES];
+ size_t tx_size;
+ char rx_data[IPC_MAX_MAILBOX_BYTES];
+ size_t rx_size;
+
+ wait_queue_head_t waitq;
+ bool pending;
+ bool complete;
+ bool wait;
+ int errno;
+};
+
+struct sst_hsw_stream;
+struct sst_hsw;
+
+/* Stream infomation */
+struct sst_hsw_stream {
+ /* configuration */
+ struct sst_hsw_ipc_stream_alloc_req request;
+ struct sst_hsw_ipc_stream_alloc_reply reply;
+ struct sst_hsw_ipc_stream_free_req free_req;
+
+ /* Mixer info */
+ u32 mute_volume[SST_HSW_NO_CHANNELS];
+ u32 mute[SST_HSW_NO_CHANNELS];
+
+ /* runtime info */
+ struct sst_hsw *hsw;
+ int host_id;
+ bool commited;
+ bool running;
+
+ /* Notification work */
+ struct work_struct notify_work;
+ u32 header;
+
+ /* Position info from DSP */
+ struct sst_hsw_ipc_stream_set_position wpos;
+ struct sst_hsw_ipc_stream_get_position rpos;
+ struct sst_hsw_ipc_stream_glitch_position glitch;
+
+ /* Volume info */
+ struct sst_hsw_ipc_volume_req vol_req;
+
+ /* driver callback */
+ u32 (*notify_position)(struct sst_hsw_stream *stream, void *data);
+ void *pdata;
+
+ struct list_head node;
+};
+
+/* FW log ring information */
+struct sst_hsw_log_stream {
+ dma_addr_t dma_addr;
+ unsigned char *dma_area;
+ unsigned char *ring_descr;
+ int pages;
+ int size;
+
+ /* Notification work */
+ struct work_struct notify_work;
+ wait_queue_head_t readers_wait_q;
+ struct mutex rw_mutex;
+
+ u32 last_pos;
+ u32 curr_pos;
+ u32 reader_pos;
+
+ /* fw log config */
+ u32 config[SST_HSW_FW_LOG_CONFIG_DWORDS];
+
+ struct sst_hsw *hsw;
+};
+
+/* SST Haswell IPC data */
+struct sst_hsw {
+ struct device *dev;
+ struct sst_dsp *dsp;
+ struct platform_device *pdev_pcm;
+
+ /* FW config */
+ struct sst_hsw_ipc_fw_ready fw_ready;
+ struct sst_hsw_ipc_fw_version version;
+ struct sst_module *scratch;
+ bool fw_done;
+
+ /* stream */
+ struct list_head stream_list;
+
+ /* global mixer */
+ struct sst_hsw_ipc_stream_info_reply mixer_info;
+ enum sst_hsw_volume_curve curve_type;
+ u32 curve_duration;
+ u32 mute[SST_HSW_NO_CHANNELS];
+ u32 mute_volume[SST_HSW_NO_CHANNELS];
+
+ /* DX */
+ struct sst_hsw_ipc_dx_reply dx;
+
+ /* boot */
+ wait_queue_head_t boot_wait;
+ bool boot_complete;
+ bool shutdown;
+
+ /* IPC messaging */
+ struct list_head tx_list;
+ struct list_head rx_list;
+ struct list_head empty_list;
+ wait_queue_head_t wait_txq;
+ struct task_struct *tx_thread;
+ struct kthread_worker kworker;
+ struct kthread_work kwork;
+ bool pending;
+ struct ipc_message *msg;
+
+ /* FW log stream */
+ struct sst_hsw_log_stream log_stream;
+};
+
+#define CREATE_TRACE_POINTS
+#include <trace/events/hswadsp.h>
+
+static inline u32 msg_get_global_type(u32 msg)
+{
+ return (msg & IPC_GLB_TYPE_MASK) >> IPC_GLB_TYPE_SHIFT;
+}
+
+static inline u32 msg_get_global_reply(u32 msg)
+{
+ return (msg & IPC_GLB_REPLY_MASK) >> IPC_GLB_REPLY_SHIFT;
+}
+
+static inline u32 msg_get_stream_type(u32 msg)
+{
+ return (msg & IPC_STR_TYPE_MASK) >> IPC_STR_TYPE_SHIFT;
+}
+
+static inline u32 msg_get_stage_type(u32 msg)
+{
+ return (msg & IPC_STG_TYPE_MASK) >> IPC_STG_TYPE_SHIFT;
+}
+
+static inline u32 msg_set_stage_type(u32 msg, u32 type)
+{
+ return (msg & ~IPC_STG_TYPE_MASK) +
+ (type << IPC_STG_TYPE_SHIFT);
+}
+
+static inline u32 msg_get_stream_id(u32 msg)
+{
+ return (msg & IPC_STR_ID_MASK) >> IPC_STR_ID_SHIFT;
+}
+
+static inline u32 msg_get_notify_reason(u32 msg)
+{
+ return (msg & IPC_STG_TYPE_MASK) >> IPC_STG_TYPE_SHIFT;
+}
+
+u32 create_channel_map(enum sst_hsw_channel_config config)
+{
+ switch (config) {
+ case SST_HSW_CHANNEL_CONFIG_MONO:
+ return (0xFFFFFFF0 | SST_HSW_CHANNEL_CENTER);
+ case SST_HSW_CHANNEL_CONFIG_STEREO:
+ return (0xFFFFFF00 | SST_HSW_CHANNEL_LEFT
+ | (SST_HSW_CHANNEL_RIGHT << 4));
+ case SST_HSW_CHANNEL_CONFIG_2_POINT_1:
+ return (0xFFFFF000 | SST_HSW_CHANNEL_LEFT
+ | (SST_HSW_CHANNEL_RIGHT << 4)
+ | (SST_HSW_CHANNEL_LFE << 8 ));
+ case SST_HSW_CHANNEL_CONFIG_3_POINT_0:
+ return (0xFFFFF000 | SST_HSW_CHANNEL_LEFT
+ | (SST_HSW_CHANNEL_CENTER << 4)
+ | (SST_HSW_CHANNEL_RIGHT << 8));
+ case SST_HSW_CHANNEL_CONFIG_3_POINT_1:
+ return (0xFFFF0000 | SST_HSW_CHANNEL_LEFT
+ | (SST_HSW_CHANNEL_CENTER << 4)
+ | (SST_HSW_CHANNEL_RIGHT << 8)
+ | (SST_HSW_CHANNEL_LFE << 12));
+ case SST_HSW_CHANNEL_CONFIG_QUATRO:
+ return (0xFFFF0000 | SST_HSW_CHANNEL_LEFT
+ | (SST_HSW_CHANNEL_RIGHT << 4)
+ | (SST_HSW_CHANNEL_LEFT_SURROUND << 8)
+ | (SST_HSW_CHANNEL_RIGHT_SURROUND << 12));
+ case SST_HSW_CHANNEL_CONFIG_4_POINT_0:
+ return (0xFFFF0000 | SST_HSW_CHANNEL_LEFT
+ | (SST_HSW_CHANNEL_CENTER << 4)
+ | (SST_HSW_CHANNEL_RIGHT << 8)
+ | (SST_HSW_CHANNEL_CENTER_SURROUND << 12));
+ case SST_HSW_CHANNEL_CONFIG_5_POINT_0:
+ return (0xFFF00000 | SST_HSW_CHANNEL_LEFT
+ | (SST_HSW_CHANNEL_CENTER << 4)
+ | (SST_HSW_CHANNEL_RIGHT << 8)
+ | (SST_HSW_CHANNEL_LEFT_SURROUND << 12)
+ | (SST_HSW_CHANNEL_RIGHT_SURROUND << 16));
+ case SST_HSW_CHANNEL_CONFIG_5_POINT_1:
+ return (0xFF000000 | SST_HSW_CHANNEL_CENTER
+ | (SST_HSW_CHANNEL_LEFT << 4)
+ | (SST_HSW_CHANNEL_RIGHT << 8)
+ | (SST_HSW_CHANNEL_LEFT_SURROUND << 12)
+ | (SST_HSW_CHANNEL_RIGHT_SURROUND << 16)
+ | (SST_HSW_CHANNEL_LFE << 20));
+ case SST_HSW_CHANNEL_CONFIG_DUAL_MONO:
+ return (0xFFFFFF00 | SST_HSW_CHANNEL_LEFT
+ | (SST_HSW_CHANNEL_LEFT << 4));
+ default:
+ return 0xFFFFFFFF;
+ }
+}
+
+static struct sst_hsw_stream *get_stream_by_id(struct sst_hsw *hsw,
+ int stream_id)
+{
+ struct sst_hsw_stream *stream;
+
+ list_for_each_entry(stream, &hsw->stream_list, node) {
+ if (stream->reply.stream_hw_id == stream_id)
+ return stream;
+ }
+
+ return NULL;
+}
+
+static void ipc_shim_dbg(struct sst_hsw *hsw, const char *text)
+{
+ struct sst_dsp *sst = hsw->dsp;
+ u32 isr, ipcd, imrx, ipcx;
+
+ ipcx = sst_dsp_shim_read_unlocked(sst, SST_IPCX);
+ isr = sst_dsp_shim_read_unlocked(sst, SST_ISRX);
+ ipcd = sst_dsp_shim_read_unlocked(sst, SST_IPCD);
+ imrx = sst_dsp_shim_read_unlocked(sst, SST_IMRX);
+
+ dev_err(hsw->dev, "ipc: --%s-- ipcx 0x%8.8x isr 0x%8.8x ipcd 0x%8.8x imrx 0x%8.8x\n",
+ text, ipcx, isr, ipcd, imrx);
+}
+
+/* locks held by caller */
+static struct ipc_message *msg_get_empty(struct sst_hsw *hsw)
+{
+ struct ipc_message *msg = NULL;
+
+ if (!list_empty(&hsw->empty_list)) {
+ msg = list_first_entry(&hsw->empty_list, struct ipc_message,
+ list);
+ list_del(&msg->list);
+ }
+
+ return msg;
+}
+
+static void ipc_tx_msgs(struct kthread_work *work)
+{
+ struct sst_hsw *hsw =
+ container_of(work, struct sst_hsw, kwork);
+ struct ipc_message *msg;
+ unsigned long flags;
+ u32 ipcx;
+
+ spin_lock_irqsave(&hsw->dsp->spinlock, flags);
+
+ if (list_empty(&hsw->tx_list) || hsw->pending) {
+ spin_unlock_irqrestore(&hsw->dsp->spinlock, flags);
+ return;
+ }
+
+ /* if the DSP is busy we will TX messages after IRQ */
+ ipcx = sst_dsp_shim_read_unlocked(hsw->dsp, SST_IPCX);
+ if (ipcx & SST_IPCX_BUSY) {
+ spin_unlock_irqrestore(&hsw->dsp->spinlock, flags);
+ return;
+ }
+
+ msg = list_first_entry(&hsw->tx_list, struct ipc_message, list);
+
+ list_move(&msg->list, &hsw->rx_list);
+
+ /* send the message */
+ sst_dsp_outbox_write(hsw->dsp, msg->tx_data, msg->tx_size);
+ sst_dsp_ipc_msg_tx(hsw->dsp, msg->header | SST_IPCX_BUSY);
+
+ spin_unlock_irqrestore(&hsw->dsp->spinlock, flags);
+}
+
+/* locks held by caller */
+static void tx_msg_reply_complete(struct sst_hsw *hsw, struct ipc_message *msg)
+{
+ msg->complete = true;
+ trace_ipc_reply("completed", msg->header);
+
+ if (!msg->wait)
+ list_add_tail(&msg->list, &hsw->empty_list);
+ else
+ wake_up(&msg->waitq);
+}
+
+static int tx_wait_done(struct sst_hsw *hsw, struct ipc_message *msg,
+ void *rx_data)
+{
+ unsigned long flags;
+ int ret;
+
+ /* wait for DSP completion (in all cases atm inc pending) */
+ ret = wait_event_timeout(msg->waitq, msg->complete,
+ msecs_to_jiffies(IPC_TIMEOUT_MSECS));
+
+ spin_lock_irqsave(&hsw->dsp->spinlock, flags);
+ if (ret == 0) {
+ ipc_shim_dbg(hsw, "message timeout");
+
+ trace_ipc_error("error message timeout for", msg->header);
+ ret = -ETIMEDOUT;
+ } else {
+
+ /* copy the data returned from DSP */
+ if (msg->rx_size)
+ memcpy(rx_data, msg->rx_data, msg->rx_size);
+ ret = msg->errno;
+ }
+
+ list_add_tail(&msg->list, &hsw->empty_list);
+ spin_unlock_irqrestore(&hsw->dsp->spinlock, flags);
+ return ret;
+}
+
+static int ipc_tx_message(struct sst_hsw *hsw, u32 header, void *tx_data,
+ size_t tx_bytes, void *rx_data, size_t rx_bytes, int wait)
+{
+ struct ipc_message *msg;
+ unsigned long flags;
+
+ spin_lock_irqsave(&hsw->dsp->spinlock, flags);
+
+ msg = msg_get_empty(hsw);
+ if (msg == NULL) {
+ spin_unlock_irqrestore(&hsw->dsp->spinlock, flags);
+ return -EBUSY;
+ }
+
+ if (tx_bytes)
+ memcpy(msg->tx_data, tx_data, tx_bytes);
+
+ msg->header = header;
+ msg->tx_size = tx_bytes;
+ msg->rx_size = rx_bytes;
+ msg->wait = wait;
+ msg->errno = 0;
+ msg->pending = false;
+ msg->complete = false;
+
+ list_add_tail(&msg->list, &hsw->tx_list);
+ spin_unlock_irqrestore(&hsw->dsp->spinlock, flags);
+
+ queue_kthread_work(&hsw->kworker, &hsw->kwork);
+
+ if (wait)
+ return tx_wait_done(hsw, msg, rx_data);
+ else
+ return 0;
+}
+
+static inline int ipc_tx_message_wait(struct sst_hsw *hsw, u32 header,
+ void *tx_data, size_t tx_bytes, void *rx_data, size_t rx_bytes)
+{
+ return ipc_tx_message(hsw, header, tx_data, tx_bytes, rx_data,
+ rx_bytes, 1);
+}
+
+static inline int ipc_tx_message_nowait(struct sst_hsw *hsw, u32 header,
+ void *tx_data, size_t tx_bytes)
+{
+ return ipc_tx_message(hsw, header, tx_data, tx_bytes, NULL, 0, 0);
+}
+
+static void hsw_fw_ready(struct sst_hsw *hsw, u32 header)
+{
+ struct sst_hsw_ipc_fw_ready fw_ready;
+ u32 offset;
+
+ offset = (header & 0x1FFFFFFF) << 3;
+
+ dev_dbg(hsw->dev, "ipc: DSP is ready 0x%8.8x offset %d\n",
+ header, offset);
+
+ /* copy data from the DSP FW ready offset */
+ sst_dsp_read(hsw->dsp, &fw_ready, offset, sizeof(fw_ready));
+
+ sst_dsp_mailbox_init(hsw->dsp, fw_ready.inbox_offset,
+ fw_ready.inbox_size, fw_ready.outbox_offset,
+ fw_ready.outbox_size);
+
+ hsw->boot_complete = true;
+ wake_up(&hsw->boot_wait);
+
+ dev_dbg(hsw->dev, " mailbox upstream 0x%x - size 0x%x\n",
+ fw_ready.inbox_offset, fw_ready.inbox_size);
+ dev_dbg(hsw->dev, " mailbox downstream 0x%x - size 0x%x\n",
+ fw_ready.outbox_offset, fw_ready.outbox_size);
+}
+
+static void hsw_notification_work(struct work_struct *work)
+{
+ struct sst_hsw_stream *stream = container_of(work,
+ struct sst_hsw_stream, notify_work);
+ struct sst_hsw_ipc_stream_glitch_position *glitch = &stream->glitch;
+ struct sst_hsw_ipc_stream_get_position *pos = &stream->rpos;
+ struct sst_hsw *hsw = stream->hsw;
+ u32 reason;
+
+ reason = msg_get_notify_reason(stream->header);
+
+ switch (reason) {
+ case IPC_STG_GLITCH:
+ trace_ipc_notification("DSP stream under/overrun",
+ stream->reply.stream_hw_id);
+ sst_dsp_inbox_read(hsw->dsp, glitch, sizeof(*glitch));
+
+ dev_err(hsw->dev, "glitch %d pos 0x%x write pos 0x%x\n",
+ glitch->glitch_type, glitch->present_pos,
+ glitch->write_pos);
+ break;
+
+ case IPC_POSITION_CHANGED:
+ trace_ipc_notification("DSP stream position changed for",
+ stream->reply.stream_hw_id);
+ sst_dsp_inbox_read(hsw->dsp, pos, sizeof(pos));
+
+ if (stream->notify_position)
+ stream->notify_position(stream, stream->pdata);
+
+ break;
+ default:
+ dev_err(hsw->dev, "error: unknown notification 0x%x\n",
+ stream->header);
+ break;
+ }
+
+ /* tell DSP that notification has been handled */
+ sst_dsp_shim_update_bits_unlocked(hsw->dsp, SST_IPCD,
+ SST_IPCD_BUSY | SST_IPCD_DONE, SST_IPCD_DONE);
+
+ /* unmask busy interrupt */
+ sst_dsp_shim_update_bits_unlocked(hsw->dsp, SST_IMRX, SST_IMRX_BUSY, 0);
+}
+
+static struct ipc_message *reply_find_msg(struct sst_hsw *hsw, u32 header)
+{
+ struct ipc_message *msg;
+
+ /* clear reply bits & status bits */
+ header &= ~(IPC_STATUS_MASK | IPC_GLB_REPLY_MASK);
+
+ if (list_empty(&hsw->rx_list)) {
+ dev_err(hsw->dev, "error: rx list empty but received 0x%x\n",
+ header);
+ return NULL;
+ }
+
+ list_for_each_entry(msg, &hsw->rx_list, list) {
+ if (msg->header == header)
+ return msg;
+ }
+
+ return NULL;
+}
+
+static void hsw_stream_update(struct sst_hsw *hsw, struct ipc_message *msg)
+{
+ struct sst_hsw_stream *stream;
+ u32 header = msg->header & ~(IPC_STATUS_MASK | IPC_GLB_REPLY_MASK);
+ u32 stream_id = msg_get_stream_id(header);
+ u32 stream_msg = msg_get_stream_type(header);
+
+ stream = get_stream_by_id(hsw, stream_id);
+ if (stream == NULL)
+ return;
+
+ switch (stream_msg) {
+ case IPC_STR_STAGE_MESSAGE:
+ case IPC_STR_NOTIFICATION:
+ case IPC_STR_RESET:
+ break;
+ case IPC_STR_PAUSE:
+ stream->running = false;
+ trace_ipc_notification("stream paused",
+ stream->reply.stream_hw_id);
+ break;
+ case IPC_STR_RESUME:
+ stream->running = true;
+ trace_ipc_notification("stream running",
+ stream->reply.stream_hw_id);
+ break;
+ }
+}
+
+static int hsw_process_reply(struct sst_hsw *hsw, u32 header)
+{
+ struct ipc_message *msg;
+ u32 reply = msg_get_global_reply(header);
+
+ trace_ipc_reply("processing -->", header);
+
+ msg = reply_find_msg(hsw, header);
+ if (msg == NULL) {
+ trace_ipc_error("error: can't find message header", header);
+ return -EIO;
+ }
+
+ /* first process the header */
+ switch (reply) {
+ case IPC_GLB_REPLY_PENDING:
+ trace_ipc_pending_reply("received", header);
+ msg->pending = true;
+ hsw->pending = true;
+ return 1;
+ case IPC_GLB_REPLY_SUCCESS:
+ if (msg->pending) {
+ trace_ipc_pending_reply("completed", header);
+ sst_dsp_inbox_read(hsw->dsp, msg->rx_data,
+ msg->rx_size);
+ hsw->pending = false;
+ } else {
+ /* copy data from the DSP */
+ sst_dsp_outbox_read(hsw->dsp, msg->rx_data,
+ msg->rx_size);
+ }
+ break;
+ /* these will be rare - but useful for debug */
+ case IPC_GLB_REPLY_UNKNOWN_MESSAGE_TYPE:
+ trace_ipc_error("error: unknown message type", header);
+ msg->errno = -EBADMSG;
+ break;
+ case IPC_GLB_REPLY_OUT_OF_RESOURCES:
+ trace_ipc_error("error: out of resources", header);
+ msg->errno = -ENOMEM;
+ break;
+ case IPC_GLB_REPLY_BUSY:
+ trace_ipc_error("error: reply busy", header);
+ msg->errno = -EBUSY;
+ break;
+ case IPC_GLB_REPLY_FAILURE:
+ trace_ipc_error("error: reply failure", header);
+ msg->errno = -EINVAL;
+ break;
+ case IPC_GLB_REPLY_STAGE_UNINITIALIZED:
+ trace_ipc_error("error: stage uninitialized", header);
+ msg->errno = -EINVAL;
+ break;
+ case IPC_GLB_REPLY_NOT_FOUND:
+ trace_ipc_error("error: reply not found", header);
+ msg->errno = -EINVAL;
+ break;
+ case IPC_GLB_REPLY_SOURCE_NOT_STARTED:
+ trace_ipc_error("error: source not started", header);
+ msg->errno = -EINVAL;
+ break;
+ case IPC_GLB_REPLY_INVALID_REQUEST:
+ trace_ipc_error("error: invalid request", header);
+ msg->errno = -EINVAL;
+ break;
+ case IPC_GLB_REPLY_ERROR_INVALID_PARAM:
+ trace_ipc_error("error: invalid parameter", header);
+ msg->errno = -EINVAL;
+ break;
+ default:
+ trace_ipc_error("error: unknown reply", header);
+ msg->errno = -EINVAL;
+ break;
+ }
+
+ /* update any stream states */
+ hsw_stream_update(hsw, msg);
+
+ /* wake up and return the error if we have waiters on this message ? */
+ list_del(&msg->list);
+ tx_msg_reply_complete(hsw, msg);
+
+ return 1;
+}
+
+static int hsw_stream_message(struct sst_hsw *hsw, u32 header)
+{
+ u32 stream_msg, stream_id, stage_type;
+ struct sst_hsw_stream *stream;
+ int handled = 0;
+
+ stream_msg = msg_get_stream_type(header);
+ stream_id = msg_get_stream_id(header);
+ stage_type = msg_get_stage_type(header);
+
+ stream = get_stream_by_id(hsw, stream_id);
+ if (stream == NULL)
+ return handled;
+
+ stream->header = header;
+
+ switch (stream_msg) {
+ case IPC_STR_STAGE_MESSAGE:
+ dev_err(hsw->dev, "error: stage msg not implemented 0x%8.8x\n",
+ header);
+ break;
+ case IPC_STR_NOTIFICATION:
+ schedule_work(&stream->notify_work);
+ break;
+ default:
+ /* handle pending message complete request */
+ handled = hsw_process_reply(hsw, header);
+ break;
+ }
+
+ return handled;
+}
+
+static int hsw_log_message(struct sst_hsw *hsw, u32 header)
+{
+ u32 operation = (header & IPC_LOG_OP_MASK) >> IPC_LOG_OP_SHIFT;
+ struct sst_hsw_log_stream *stream = &hsw->log_stream;
+ int ret = 1;
+
+ if (operation != IPC_DEBUG_REQUEST_LOG_DUMP) {
+ dev_err(hsw->dev,
+ "error: log msg not implemented 0x%8.8x\n", header);
+ return 0;
+ }
+
+ mutex_lock(&stream->rw_mutex);
+ stream->last_pos = stream->curr_pos;
+ sst_dsp_inbox_read(
+ hsw->dsp, &stream->curr_pos, sizeof(stream->curr_pos));
+ mutex_unlock(&stream->rw_mutex);
+
+ schedule_work(&stream->notify_work);
+
+ return ret;
+}
+
+static int hsw_process_notification(struct sst_hsw *hsw)
+{
+ struct sst_dsp *sst = hsw->dsp;
+ u32 type, header;
+ int handled = 1;
+
+ header = sst_dsp_shim_read_unlocked(sst, SST_IPCD);
+ type = msg_get_global_type(header);
+
+ trace_ipc_request("processing -->", header);
+
+ /* FW Ready is a special case */
+ if (!hsw->boot_complete && header & IPC_FW_READY) {
+ hsw_fw_ready(hsw, header);
+ return handled;
+ }
+
+ switch (type) {
+ case IPC_GLB_GET_FW_VERSION:
+ case IPC_GLB_ALLOCATE_STREAM:
+ case IPC_GLB_FREE_STREAM:
+ case IPC_GLB_GET_FW_CAPABILITIES:
+ case IPC_GLB_REQUEST_DUMP:
+ case IPC_GLB_GET_DEVICE_FORMATS:
+ case IPC_GLB_SET_DEVICE_FORMATS:
+ case IPC_GLB_ENTER_DX_STATE:
+ case IPC_GLB_GET_MIXER_STREAM_INFO:
+ case IPC_GLB_MAX_IPC_MESSAGE_TYPE:
+ case IPC_GLB_RESTORE_CONTEXT:
+ case IPC_GLB_SHORT_REPLY:
+ dev_err(hsw->dev, "error: message type %d header 0x%x\n",
+ type, header);
+ break;
+ case IPC_GLB_STREAM_MESSAGE:
+ handled = hsw_stream_message(hsw, header);
+ break;
+ case IPC_GLB_DEBUG_LOG_MESSAGE:
+ handled = hsw_log_message(hsw, header);
+ break;
+ default:
+ dev_err(hsw->dev, "error: unexpected type %d hdr 0x%8.8x\n",
+ type, header);
+ break;
+ }
+
+ return handled;
+}
+
+static irqreturn_t hsw_irq_thread(int irq, void *context)
+{
+ struct sst_dsp *sst = (struct sst_dsp *) context;
+ struct sst_hsw *hsw = sst_dsp_get_thread_context(sst);
+ u32 ipcx, ipcd;
+ int handled;
+ unsigned long flags;
+
+ spin_lock_irqsave(&sst->spinlock, flags);
+
+ ipcx = sst_dsp_ipc_msg_rx(hsw->dsp);
+ ipcd = sst_dsp_shim_read_unlocked(sst, SST_IPCD);
+
+ /* reply message from DSP */
+ if (ipcx & SST_IPCX_DONE) {
+
+ /* Handle Immediate reply from DSP Core */
+ handled = hsw_process_reply(hsw, ipcx);
+
+ if (handled > 0) {
+ /* clear DONE bit - tell DSP we have completed */
+ sst_dsp_shim_update_bits_unlocked(sst, SST_IPCX,
+ SST_IPCX_DONE, 0);
+
+ /* unmask Done interrupt */
+ sst_dsp_shim_update_bits_unlocked(sst, SST_IMRX,
+ SST_IMRX_DONE, 0);
+ }
+ }
+
+ /* new message from DSP */
+ if (ipcd & SST_IPCD_BUSY) {
+
+ /* Handle Notification and Delayed reply from DSP Core */
+ handled = hsw_process_notification(hsw);
+
+ /* clear BUSY bit and set DONE bit - accept new messages */
+ if (handled > 0) {
+ sst_dsp_shim_update_bits_unlocked(sst, SST_IPCD,
+ SST_IPCD_BUSY | SST_IPCD_DONE, SST_IPCD_DONE);
+
+ /* unmask busy interrupt */
+ sst_dsp_shim_update_bits_unlocked(sst, SST_IMRX,
+ SST_IMRX_BUSY, 0);
+ }
+ }
+
+ spin_unlock_irqrestore(&sst->spinlock, flags);
+
+ /* continue to send any remaining messages... */
+ queue_kthread_work(&hsw->kworker, &hsw->kwork);
+
+ return IRQ_HANDLED;
+}
+
+int sst_hsw_fw_get_version(struct sst_hsw *hsw,
+ struct sst_hsw_ipc_fw_version *version)
+{
+ int ret;
+
+ ret = ipc_tx_message_wait(hsw, IPC_GLB_TYPE(IPC_GLB_GET_FW_VERSION),
+ NULL, 0, version, sizeof(*version));
+ if (ret < 0)
+ dev_err(hsw->dev, "error: get version failed\n");
+
+ return ret;
+}
+
+/* Mixer Controls */
+int sst_hsw_stream_mute(struct sst_hsw *hsw, struct sst_hsw_stream *stream,
+ u32 stage_id, u32 channel)
+{
+ int ret;
+
+ ret = sst_hsw_stream_get_volume(hsw, stream, stage_id, channel,
+ &stream->mute_volume[channel]);
+ if (ret < 0)
+ return ret;
+
+ ret = sst_hsw_stream_set_volume(hsw, stream, stage_id, channel, 0);
+ if (ret < 0) {
+ dev_err(hsw->dev, "error: can't unmute stream %d channel %d\n",
+ stream->reply.stream_hw_id, channel);
+ return ret;
+ }
+
+ stream->mute[channel] = 1;
+ return 0;
+}
+
+int sst_hsw_stream_unmute(struct sst_hsw *hsw, struct sst_hsw_stream *stream,
+ u32 stage_id, u32 channel)
+
+{
+ int ret;
+
+ stream->mute[channel] = 0;
+ ret = sst_hsw_stream_set_volume(hsw, stream, stage_id, channel,
+ stream->mute_volume[channel]);
+ if (ret < 0) {
+ dev_err(hsw->dev, "error: can't unmute stream %d channel %d\n",
+ stream->reply.stream_hw_id, channel);
+ return ret;
+ }
+
+ return 0;
+}
+
+int sst_hsw_stream_get_volume(struct sst_hsw *hsw, struct sst_hsw_stream *stream,
+ u32 stage_id, u32 channel, u32 *volume)
+{
+ if (channel > 1)
+ return -EINVAL;
+
+ sst_dsp_read(hsw->dsp, volume,
+ stream->reply.volume_register_address[channel], sizeof(volume));
+
+ return 0;
+}
+
+int sst_hsw_stream_set_volume_curve(struct sst_hsw *hsw,
+ struct sst_hsw_stream *stream, u64 curve_duration,
+ enum sst_hsw_volume_curve curve)
+{
+ /* curve duration in steps of 100ns */
+ stream->vol_req.curve_duration = curve_duration;
+ stream->vol_req.curve_type = curve;
+
+ return 0;
+}
+
+/* stream volume */
+int sst_hsw_stream_set_volume(struct sst_hsw *hsw,
+ struct sst_hsw_stream *stream, u32 stage_id, u32 channel, u32 volume)
+{
+ struct sst_hsw_ipc_volume_req *req;
+ u32 header;
+ int ret;
+
+ trace_ipc_request("set stream volume", stream->reply.stream_hw_id);
+
+ if (channel > 1)
+ return -EINVAL;
+
+ if (stream->mute[channel]) {
+ stream->mute_volume[channel] = volume;
+ return 0;
+ }
+
+ header = IPC_GLB_TYPE(IPC_GLB_STREAM_MESSAGE) |
+ IPC_STR_TYPE(IPC_STR_STAGE_MESSAGE);
+ header |= (stream->reply.stream_hw_id << IPC_STR_ID_SHIFT);
+ header |= (IPC_STG_SET_VOLUME << IPC_STG_TYPE_SHIFT);
+ header |= (stage_id << IPC_STG_ID_SHIFT);
+
+ req = &stream->vol_req;
+ req->channel = channel;
+ req->target_volume = volume;
+
+ ret = ipc_tx_message_wait(hsw, header, req, sizeof(*req), NULL, 0);
+ if (ret < 0) {
+ dev_err(hsw->dev, "error: set stream volume failed\n");
+ return ret;
+ }
+
+ return 0;
+}
+
+int sst_hsw_mixer_mute(struct sst_hsw *hsw, u32 stage_id, u32 channel)
+{
+ int ret;
+
+ ret = sst_hsw_mixer_get_volume(hsw, stage_id, channel,
+ &hsw->mute_volume[channel]);
+ if (ret < 0)
+ return ret;
+
+ ret = sst_hsw_mixer_set_volume(hsw, stage_id, channel, 0);
+ if (ret < 0) {
+ dev_err(hsw->dev, "error: failed to unmute mixer channel %d\n",
+ channel);
+ return ret;
+ }
+
+ hsw->mute[channel] = 1;
+ return 0;
+}
+
+int sst_hsw_mixer_unmute(struct sst_hsw *hsw, u32 stage_id, u32 channel)
+{
+ int ret;
+
+ ret = sst_hsw_mixer_set_volume(hsw, stage_id, channel,
+ hsw->mixer_info.volume_register_address[channel]);
+ if (ret < 0) {
+ dev_err(hsw->dev, "error: failed to unmute mixer channel %d\n",
+ channel);
+ return ret;
+ }
+
+ hsw->mute[channel] = 0;
+ return 0;
+}
+
+int sst_hsw_mixer_get_volume(struct sst_hsw *hsw, u32 stage_id, u32 channel,
+ u32 *volume)
+{
+ if (channel > 1)
+ return -EINVAL;
+
+ sst_dsp_read(hsw->dsp, volume,
+ hsw->mixer_info.volume_register_address[channel],
+ sizeof(*volume));
+
+ return 0;
+}
+
+int sst_hsw_mixer_set_volume_curve(struct sst_hsw *hsw,
+ u64 curve_duration, enum sst_hsw_volume_curve curve)
+{
+ /* curve duration in steps of 100ns */
+ hsw->curve_duration = curve_duration;
+ hsw->curve_type = curve;
+
+ return 0;
+}
+
+/* global mixer volume */
+int sst_hsw_mixer_set_volume(struct sst_hsw *hsw, u32 stage_id, u32 channel,
+ u32 volume)
+{
+ struct sst_hsw_ipc_volume_req req;
+ u32 header;
+ int ret;
+
+ trace_ipc_request("set mixer volume", volume);
+
+ /* set both at same time ? */
+ if (channel == 2) {
+ if (hsw->mute[0] && hsw->mute[1]) {
+ hsw->mute_volume[0] = hsw->mute_volume[1] = volume;
+ return 0;
+ } else if (hsw->mute[0])
+ req.channel = 1;
+ else if (hsw->mute[1])
+ req.channel = 0;
+ else
+ req.channel = 0xffffffff;
+ } else {
+ /* set only 1 channel */
+ if (hsw->mute[channel]) {
+ hsw->mute_volume[channel] = volume;
+ return 0;
+ }
+ req.channel = channel;
+ }
+
+ header = IPC_GLB_TYPE(IPC_GLB_STREAM_MESSAGE) |
+ IPC_STR_TYPE(IPC_STR_STAGE_MESSAGE);
+ header |= (hsw->mixer_info.mixer_hw_id << IPC_STR_ID_SHIFT);
+ header |= (IPC_STG_SET_VOLUME << IPC_STG_TYPE_SHIFT);
+ header |= (stage_id << IPC_STG_ID_SHIFT);
+
+ req.curve_duration = hsw->curve_duration;
+ req.curve_type = hsw->curve_type;
+ req.target_volume = volume;
+
+ ret = ipc_tx_message_wait(hsw, header, &req, sizeof(req), NULL, 0);
+ if (ret < 0) {
+ dev_err(hsw->dev, "error: set mixer volume failed\n");
+ return ret;
+ }
+
+ return 0;
+}
+
+/* Stream API */
+struct sst_hsw_stream *sst_hsw_stream_new(struct sst_hsw *hsw, int id,
+ u32 (*notify_position)(struct sst_hsw_stream *stream, void *data),
+ void *data)
+{
+ struct sst_hsw_stream *stream;
+
+ stream = kzalloc(sizeof(*stream), GFP_KERNEL);
+ if (stream == NULL)
+ return NULL;
+
+ list_add(&stream->node, &hsw->stream_list);
+ stream->notify_position = notify_position;
+ stream->pdata = data;
+ stream->hsw = hsw;
+ stream->host_id = id;
+
+ /* work to process notification messages */
+ INIT_WORK(&stream->notify_work, hsw_notification_work);
+
+ return stream;
+}
+
+int sst_hsw_stream_free(struct sst_hsw *hsw, struct sst_hsw_stream *stream)
+{
+ u32 header;
+ int ret = 0;
+
+ /* dont free DSP streams that are not commited */
+ if (!stream->commited)
+ goto out;
+
+ trace_ipc_request("stream free", stream->host_id);
+
+ stream->free_req.stream_id = stream->reply.stream_hw_id;
+ header = IPC_GLB_TYPE(IPC_GLB_FREE_STREAM);
+
+ ret = ipc_tx_message_wait(hsw, header, &stream->free_req,
+ sizeof(stream->free_req), NULL, 0);
+ if (ret < 0) {
+ dev_err(hsw->dev, "error: free stream %d failed\n",
+ stream->free_req.stream_id);
+ return -EAGAIN;
+ }
+
+ trace_hsw_stream_free_req(stream, &stream->free_req);
+
+out:
+ list_del(&stream->node);
+ kfree(stream);
+
+ return ret;
+}
+
+int sst_hsw_stream_set_bits(struct sst_hsw *hsw,
+ struct sst_hsw_stream *stream, enum sst_hsw_bitdepth bits)
+{
+ if (stream->commited) {
+ dev_err(hsw->dev, "error: stream committed for set bits\n");
+ return -EINVAL;
+ }
+
+ stream->request.format.bitdepth = bits;
+ return 0;
+}
+
+int sst_hsw_stream_set_channels(struct sst_hsw *hsw,
+ struct sst_hsw_stream *stream, int channels)
+{
+ if (stream->commited) {
+ dev_err(hsw->dev, "error: stream committed for set channels\n");
+ return -EINVAL;
+ }
+
+ /* stereo is only supported atm */
+ if (channels != 2)
+ return -EINVAL;
+
+ stream->request.format.ch_num = channels;
+ return 0;
+}
+
+int sst_hsw_stream_set_rate(struct sst_hsw *hsw,
+ struct sst_hsw_stream *stream, int rate)
+{
+ if (stream->commited) {
+ dev_err(hsw->dev, "error: stream committed for set rate\n");
+ return -EINVAL;
+ }
+
+ stream->request.format.frequency = rate;
+ return 0;
+}
+
+int sst_hsw_stream_set_map_config(struct sst_hsw *hsw,
+ struct sst_hsw_stream *stream, u32 map,
+ enum sst_hsw_channel_config config)
+{
+ if (stream->commited) {
+ dev_err(hsw->dev, "error: stream committed for set map\n");
+ return -EINVAL;
+ }
+
+ stream->request.format.map = map;
+ stream->request.format.config = config;
+ return 0;
+}
+
+int sst_hsw_stream_set_style(struct sst_hsw *hsw,
+ struct sst_hsw_stream *stream, enum sst_hsw_interleaving style)
+{
+ if (stream->commited) {
+ dev_err(hsw->dev, "error: stream committed for set style\n");
+ return -EINVAL;
+ }
+
+ stream->request.format.style = style;
+ return 0;
+}
+
+int sst_hsw_stream_set_valid(struct sst_hsw *hsw,
+ struct sst_hsw_stream *stream, u32 bits)
+{
+ if (stream->commited) {
+ dev_err(hsw->dev, "error: stream committed for set valid bits\n");
+ return -EINVAL;
+ }
+
+ stream->request.format.valid_bit = bits;
+ return 0;
+}
+
+/* Stream Configuration */
+int sst_hsw_stream_format(struct sst_hsw *hsw, struct sst_hsw_stream *stream,
+ enum sst_hsw_stream_path_id path_id,
+ enum sst_hsw_stream_type stream_type,
+ enum sst_hsw_stream_format format_id)
+{
+ if (stream->commited) {
+ dev_err(hsw->dev, "error: stream committed for set format\n");
+ return -EINVAL;
+ }
+
+ stream->request.path_id = path_id;
+ stream->request.stream_type = stream_type;
+ stream->request.format_id = format_id;
+
+ trace_hsw_stream_alloc_request(stream, &stream->request);
+
+ return 0;
+}
+
+int sst_hsw_stream_buffer(struct sst_hsw *hsw, struct sst_hsw_stream *stream,
+ u32 ring_pt_address, u32 num_pages,
+ u32 ring_size, u32 ring_offset, u32 ring_first_pfn)
+{
+ if (stream->commited) {
+ dev_err(hsw->dev, "error: stream committed for buffer\n");
+ return -EINVAL;
+ }
+
+ stream->request.ringinfo.ring_pt_address = ring_pt_address;
+ stream->request.ringinfo.num_pages = num_pages;
+ stream->request.ringinfo.ring_size = ring_size;
+ stream->request.ringinfo.ring_offset = ring_offset;
+ stream->request.ringinfo.ring_first_pfn = ring_first_pfn;
+
+ trace_hsw_stream_buffer(stream);
+
+ return 0;
+}
+
+int sst_hsw_stream_set_module_info(struct sst_hsw *hsw,
+ struct sst_hsw_stream *stream, enum sst_hsw_module_id module_id,
+ u32 entry_point)
+{
+ struct sst_hsw_module_map *map = &stream->request.map;
+
+ if (stream->commited) {
+ dev_err(hsw->dev, "error: stream committed for set module\n");
+ return -EINVAL;
+ }
+
+ /* only support initial module atm */
+ map->module_entries_count = 1;
+ map->module_entries[0].module_id = module_id;
+ map->module_entries[0].entry_point = entry_point;
+
+ return 0;
+}
+
+int sst_hsw_stream_set_pmemory_info(struct sst_hsw *hsw,
+ struct sst_hsw_stream *stream, u32 offset, u32 size)
+{
+ if (stream->commited) {
+ dev_err(hsw->dev, "error: stream committed for set pmem\n");
+ return -EINVAL;
+ }
+
+ stream->request.persistent_mem.offset = offset;
+ stream->request.persistent_mem.size = size;
+
+ return 0;
+}
+
+int sst_hsw_stream_set_smemory_info(struct sst_hsw *hsw,
+ struct sst_hsw_stream *stream, u32 offset, u32 size)
+{
+ if (stream->commited) {
+ dev_err(hsw->dev, "error: stream committed for set smem\n");
+ return -EINVAL;
+ }
+
+ stream->request.scratch_mem.offset = offset;
+ stream->request.scratch_mem.size = size;
+
+ return 0;
+}
+
+int sst_hsw_stream_commit(struct sst_hsw *hsw, struct sst_hsw_stream *stream)
+{
+ struct sst_hsw_ipc_stream_alloc_req *str_req = &stream->request;
+ struct sst_hsw_ipc_stream_alloc_reply *reply = &stream->reply;
+ u32 header;
+ int ret;
+
+ trace_ipc_request("stream alloc", stream->host_id);
+
+ header = IPC_GLB_TYPE(IPC_GLB_ALLOCATE_STREAM);
+
+ ret = ipc_tx_message_wait(hsw, header, str_req, sizeof(*str_req),
+ reply, sizeof(*reply));
+ if (ret < 0) {
+ dev_err(hsw->dev, "error: stream commit failed\n");
+ return ret;
+ }
+
+ stream->commited = 1;
+ trace_hsw_stream_alloc_reply(stream);
+
+ return 0;
+}
+
+/* Stream Information - these calls could be inline but we want the IPC
+ ABI to be opaque to client PCM drivers to cope with any future ABI changes */
+int sst_hsw_stream_get_hw_id(struct sst_hsw *hsw,
+ struct sst_hsw_stream *stream)
+{
+ return stream->reply.stream_hw_id;
+}
+
+int sst_hsw_stream_get_mixer_id(struct sst_hsw *hsw,
+ struct sst_hsw_stream *stream)
+{
+ return stream->reply.mixer_hw_id;
+}
+
+u32 sst_hsw_stream_get_read_reg(struct sst_hsw *hsw,
+ struct sst_hsw_stream *stream)
+{
+ return stream->reply.read_position_register_address;
+}
+
+u32 sst_hsw_stream_get_pointer_reg(struct sst_hsw *hsw,
+ struct sst_hsw_stream *stream)
+{
+ return stream->reply.presentation_position_register_address;
+}
+
+u32 sst_hsw_stream_get_peak_reg(struct sst_hsw *hsw,
+ struct sst_hsw_stream *stream, u32 channel)
+{
+ if (channel >= 2)
+ return 0;
+
+ return stream->reply.peak_meter_register_address[channel];
+}
+
+u32 sst_hsw_stream_get_vol_reg(struct sst_hsw *hsw,
+ struct sst_hsw_stream *stream, u32 channel)
+{
+ if (channel >= 2)
+ return 0;
+
+ return stream->reply.volume_register_address[channel];
+}
+
+int sst_hsw_mixer_get_info(struct sst_hsw *hsw)
+{
+ struct sst_hsw_ipc_stream_info_reply *reply;
+ u32 header;
+ int ret;
+
+ reply = &hsw->mixer_info;
+ header = IPC_GLB_TYPE(IPC_GLB_GET_MIXER_STREAM_INFO);
+
+ trace_ipc_request("get global mixer info", 0);
+
+ ret = ipc_tx_message_wait(hsw, header, NULL, 0, reply, sizeof(*reply));
+ if (ret < 0) {
+ dev_err(hsw->dev, "error: get stream info failed\n");
+ return ret;
+ }
+
+ trace_hsw_mixer_info_reply(reply);
+
+ return 0;
+}
+
+/* Send stream command */
+static int sst_hsw_stream_operations(struct sst_hsw *hsw, int type,
+ int stream_id, int wait)
+{
+ u32 header;
+
+ header = IPC_GLB_TYPE(IPC_GLB_STREAM_MESSAGE) | IPC_STR_TYPE(type);
+ header |= (stream_id << IPC_STR_ID_SHIFT);
+
+ if (wait)
+ return ipc_tx_message_wait(hsw, header, NULL, 0, NULL, 0);
+ else
+ return ipc_tx_message_nowait(hsw, header, NULL, 0);
+}
+
+/* Stream ALSA trigger operations */
+int sst_hsw_stream_pause(struct sst_hsw *hsw, struct sst_hsw_stream *stream,
+ int wait)
+{
+ int ret;
+
+ trace_ipc_request("stream pause", stream->reply.stream_hw_id);
+
+ ret = sst_hsw_stream_operations(hsw, IPC_STR_PAUSE,
+ stream->reply.stream_hw_id, wait);
+ if (ret < 0)
+ dev_err(hsw->dev, "error: failed to pause stream %d\n",
+ stream->reply.stream_hw_id);
+
+ return ret;
+}
+
+int sst_hsw_stream_resume(struct sst_hsw *hsw, struct sst_hsw_stream *stream,
+ int wait)
+{
+ int ret;
+
+ trace_ipc_request("stream resume", stream->reply.stream_hw_id);
+
+ ret = sst_hsw_stream_operations(hsw, IPC_STR_RESUME,
+ stream->reply.stream_hw_id, wait);
+ if (ret < 0)
+ dev_err(hsw->dev, "error: failed to resume stream %d\n",
+ stream->reply.stream_hw_id);
+
+ return ret;
+}
+
+int sst_hsw_stream_reset(struct sst_hsw *hsw, struct sst_hsw_stream *stream)
+{
+ int ret, tries = 10;
+
+ /* dont reset streams that are not commited */
+ if (!stream->commited)
+ return 0;
+
+ /* wait for pause to complete before we reset the stream */
+ while (stream->running && tries--)
+ msleep(1);
+ if (!tries) {
+ dev_err(hsw->dev, "error: reset stream %d still running\n",
+ stream->reply.stream_hw_id);
+ return -EINVAL;
+ }
+
+ trace_ipc_request("stream reset", stream->reply.stream_hw_id);
+
+ ret = sst_hsw_stream_operations(hsw, IPC_STR_RESET,
+ stream->reply.stream_hw_id, 1);
+ if (ret < 0)
+ dev_err(hsw->dev, "error: failed to reset stream %d\n",
+ stream->reply.stream_hw_id);
+ return ret;
+}
+
+/* Stream pointer positions */
+int sst_hsw_get_dsp_position(struct sst_hsw *hsw,
+ struct sst_hsw_stream *stream)
+{
+ return stream->rpos.position;
+}
+
+int sst_hsw_stream_set_write_position(struct sst_hsw *hsw,
+ struct sst_hsw_stream *stream, u32 stage_id, u32 position)
+{
+ u32 header;
+ int ret;
+
+ trace_stream_write_position(stream->reply.stream_hw_id, position);
+
+ header = IPC_GLB_TYPE(IPC_GLB_STREAM_MESSAGE) |
+ IPC_STR_TYPE(IPC_STR_STAGE_MESSAGE);
+ header |= (stream->reply.stream_hw_id << IPC_STR_ID_SHIFT);
+ header |= (IPC_STG_SET_WRITE_POSITION << IPC_STG_TYPE_SHIFT);
+ header |= (stage_id << IPC_STG_ID_SHIFT);
+ stream->wpos.position = position;
+
+ ret = ipc_tx_message_nowait(hsw, header, &stream->wpos,
+ sizeof(stream->wpos));
+ if (ret < 0)
+ dev_err(hsw->dev, "error: stream %d set position %d failed\n",
+ stream->reply.stream_hw_id, position);
+
+ return ret;
+}
+
+/* physical BE config */
+int sst_hsw_device_set_config(struct sst_hsw *hsw,
+ enum sst_hsw_device_id dev, enum sst_hsw_device_mclk mclk,
+ enum sst_hsw_device_mode mode, u32 clock_divider)
+{
+ struct sst_hsw_ipc_device_config_req config;
+ u32 header;
+ int ret;
+
+ trace_ipc_request("set device config", dev);
+
+ config.ssp_interface = dev;
+ config.clock_frequency = mclk;
+ config.mode = mode;
+ config.clock_divider = clock_divider;
+
+ trace_hsw_device_config_req(&config);
+
+ header = IPC_GLB_TYPE(IPC_GLB_SET_DEVICE_FORMATS);
+
+ ret = ipc_tx_message_wait(hsw, header, &config, sizeof(config),
+ NULL, 0);
+ if (ret < 0)
+ dev_err(hsw->dev, "error: set device formats failed\n");
+
+ return ret;
+}
+EXPORT_SYMBOL_GPL(sst_hsw_device_set_config);
+
+/* DX Config */
+int sst_hsw_dx_set_state(struct sst_hsw *hsw,
+ enum sst_hsw_dx_state state, struct sst_hsw_ipc_dx_reply *dx)
+{
+ u32 header, state_;
+ int ret;
+
+ header = IPC_GLB_TYPE(IPC_GLB_ENTER_DX_STATE);
+ state_ = state;
+
+ trace_ipc_request("PM enter Dx state", state);
+
+ ret = ipc_tx_message_wait(hsw, header, &state_, sizeof(state_),
+ dx, sizeof(dx));
+ if (ret < 0) {
+ dev_err(hsw->dev, "ipc: error set dx state %d failed\n", state);
+ return ret;
+ }
+
+ dev_dbg(hsw->dev, "ipc: got %d entry numbers for state %d\n",
+ dx->entries_no, state);
+
+ memcpy(&hsw->dx, dx, sizeof(*dx));
+ return 0;
+}
+
+/* Used to save state into hsw->dx_reply */
+int sst_hsw_dx_get_state(struct sst_hsw *hsw, u32 item,
+ u32 *offset, u32 *size, u32 *source)
+{
+ struct sst_hsw_ipc_dx_memory_item *dx_mem;
+ struct sst_hsw_ipc_dx_reply *dx_reply;
+ int entry_no;
+
+ dx_reply = &hsw->dx;
+ entry_no = dx_reply->entries_no;
+
+ trace_ipc_request("PM get Dx state", entry_no);
+
+ if (item >= entry_no)
+ return -EINVAL;
+
+ dx_mem = &dx_reply->mem_info[item];
+ *offset = dx_mem->offset;
+ *size = dx_mem->size;
+ *source = dx_mem->source;
+
+ return 0;
+}
+
+static int msg_empty_list_init(struct sst_hsw *hsw)
+{
+ int i;
+
+ hsw->msg = kzalloc(sizeof(struct ipc_message) *
+ IPC_EMPTY_LIST_SIZE, GFP_KERNEL);
+ if (hsw->msg == NULL)
+ return -ENOMEM;
+
+ for (i = 0; i < IPC_EMPTY_LIST_SIZE; i++) {
+ init_waitqueue_head(&hsw->msg[i].waitq);
+ list_add(&hsw->msg[i].list, &hsw->empty_list);
+ }
+
+ return 0;
+}
+
+void sst_hsw_set_scratch_module(struct sst_hsw *hsw,
+ struct sst_module *scratch)
+{
+ hsw->scratch = scratch;
+}
+
+struct sst_dsp *sst_hsw_get_dsp(struct sst_hsw *hsw)
+{
+ return hsw->dsp;
+}
+
+static struct sst_dsp_device hsw_dev = {
+ .thread = hsw_irq_thread,
+ .ops = &haswell_ops,
+};
+
+int sst_hsw_dsp_init(struct device *dev, struct sst_pdata *pdata)
+{
+ struct sst_hsw_ipc_fw_version version;
+ struct sst_hsw *hsw;
+ struct sst_fw *hsw_sst_fw;
+ int ret;
+
+ dev_dbg(dev, "initialising Audio DSP IPC\n");
+
+ hsw = devm_kzalloc(dev, sizeof(*hsw), GFP_KERNEL);
+ if (hsw == NULL)
+ return -ENOMEM;
+
+ hsw->dev = dev;
+ INIT_LIST_HEAD(&hsw->stream_list);
+ INIT_LIST_HEAD(&hsw->tx_list);
+ INIT_LIST_HEAD(&hsw->rx_list);
+ INIT_LIST_HEAD(&hsw->empty_list);
+ init_waitqueue_head(&hsw->boot_wait);
+ init_waitqueue_head(&hsw->wait_txq);
+
+ ret = msg_empty_list_init(hsw);
+ if (ret < 0)
+ goto list_err;
+
+ /* start the IPC message thread */
+ init_kthread_worker(&hsw->kworker);
+ hsw->tx_thread = kthread_run(kthread_worker_fn,
+ &hsw->kworker,
+ dev_name(hsw->dev));
+ if (IS_ERR(hsw->tx_thread)) {
+ ret = PTR_ERR(hsw->tx_thread);
+ dev_err(hsw->dev, "error: failed to create message TX task\n");
+ goto list_err;
+ }
+ init_kthread_work(&hsw->kwork, ipc_tx_msgs);
+
+ hsw_dev.thread_context = hsw;
+
+ /* init SST shim */
+ hsw->dsp = sst_dsp_new(dev, &hsw_dev, pdata);
+ if (hsw->dsp == NULL) {
+ ret = -ENODEV;
+ goto list_err;
+ }
+
+ /* keep the DSP in reset state for base FW loading */
+ sst_dsp_reset(hsw->dsp);
+
+ hsw_sst_fw = sst_fw_new(hsw->dsp, pdata->fw, hsw);
+
+ if (hsw_sst_fw == NULL) {
+ ret = -ENODEV;
+ dev_err(dev, "error: failed to load firmware\n");
+ goto fw_err;
+ }
+
+ /* wait for DSP boot completion */
+ sst_dsp_boot(hsw->dsp);
+ ret = wait_event_timeout(hsw->boot_wait, hsw->boot_complete,
+ msecs_to_jiffies(IPC_BOOT_MSECS));
+ if (ret == 0) {
+ ret = -EIO;
+ dev_err(hsw->dev, "error: ADSP boot timeout\n");
+ goto boot_err;
+ }
+
+ /* get the FW version */
+ sst_hsw_fw_get_version(hsw, &version);
+ dev_info(hsw->dev, "FW loaded: type %d - version: %d.%d build %d\n",
+ version.type, version.major, version.minor, version.build);
+
+ /* get the globalmixer */
+ ret = sst_hsw_mixer_get_info(hsw);
+ if (ret < 0) {
+ dev_err(hsw->dev, "error: failed to get stream info\n");
+ goto boot_err;
+ }
+
+ pdata->dsp = hsw;
+ return 0;
+
+boot_err:
+ sst_dsp_reset(hsw->dsp);
+ sst_fw_free(hsw_sst_fw);
+fw_err:
+ sst_dsp_free(hsw->dsp);
+ kfree(hsw->msg);
+list_err:
+ return ret;
+}
+EXPORT_SYMBOL_GPL(sst_hsw_dsp_init);
+
+void sst_hsw_dsp_free(struct device *dev, struct sst_pdata *pdata)
+{
+ struct sst_hsw *hsw = pdata->dsp;
+
+ sst_dsp_reset(hsw->dsp);
+ sst_fw_free_all(hsw->dsp);
+ sst_dsp_free(hsw->dsp);
+ kfree(hsw->scratch);
+ kfree(hsw->msg);
+}
+EXPORT_SYMBOL_GPL(sst_hsw_dsp_free);
diff --git a/sound/soc/intel/sst-haswell-ipc.h b/sound/soc/intel/sst-haswell-ipc.h
new file mode 100644
index 0000000..d517929
--- /dev/null
+++ b/sound/soc/intel/sst-haswell-ipc.h
@@ -0,0 +1,488 @@
+/*
+ * Intel SST Haswell/Broadwell IPC Support
+ *
+ * Copyright (C) 2013, Intel Corporation. All rights reserved.
+ *
+ * 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.
+ *
+ */
+
+#ifndef __SST_HASWELL_IPC_H
+#define __SST_HASWELL_IPC_H
+
+#include <linux/types.h>
+#include <linux/kernel.h>
+#include <linux/platform_device.h>
+
+#define SST_HSW_NO_CHANNELS 2
+#define SST_HSW_MAX_DX_REGIONS 14
+
+#define SST_HSW_FW_LOG_CONFIG_DWORDS 12
+#define SST_HSW_GLOBAL_LOG 15
+
+/**
+ * Upfront defined maximum message size that is
+ * expected by the in/out communication pipes in FW.
+ */
+#define SST_HSW_IPC_MAX_PAYLOAD_SIZE 400
+#define SST_HSW_MAX_INFO_SIZE 64
+#define SST_HSW_BUILD_HASH_LENGTH 40
+
+struct sst_hsw;
+struct sst_hsw_stream;
+struct sst_hsw_log_stream;
+struct sst_pdata;
+struct sst_module;
+extern struct sst_ops haswell_ops;
+
+/* Stream Allocate Path ID */
+enum sst_hsw_stream_path_id {
+ SST_HSW_STREAM_PATH_SSP0_OUT = 0,
+ SST_HSW_STREAM_PATH_SSP0_IN = 1,
+ SST_HSW_STREAM_PATH_MAX_PATH_ID = 2,
+};
+
+/* Stream Allocate Stream Type */
+enum sst_hsw_stream_type {
+ SST_HSW_STREAM_TYPE_RENDER = 0,
+ SST_HSW_STREAM_TYPE_SYSTEM = 1,
+ SST_HSW_STREAM_TYPE_CAPTURE = 2,
+ SST_HSW_STREAM_TYPE_LOOPBACK = 3,
+ SST_HSW_STREAM_TYPE_MAX_STREAM_TYPE = 4,
+};
+
+/* Stream Allocate Stream Format */
+enum sst_hsw_stream_format {
+ SST_HSW_STREAM_FORMAT_PCM_FORMAT = 0,
+ SST_HSW_STREAM_FORMAT_MP3_FORMAT = 1,
+ SST_HSW_STREAM_FORMAT_AAC_FORMAT = 2,
+ SST_HSW_STREAM_FORMAT_MAX_FORMAT_ID = 3,
+};
+
+/* Device ID */
+enum sst_hsw_device_id {
+ SST_HSW_DEVICE_SSP_0 = 0,
+ SST_HSW_DEVICE_SSP_1 = 1,
+};
+
+/* Device Master Clock Frequency */
+enum sst_hsw_device_mclk {
+ SST_HSW_DEVICE_MCLK_OFF = 0,
+ SST_HSW_DEVICE_MCLK_FREQ_6_MHZ = 1,
+ SST_HSW_DEVICE_MCLK_FREQ_12_MHZ = 2,
+ SST_HSW_DEVICE_MCLK_FREQ_24_MHZ = 3,
+};
+
+/* Device Clock Master */
+enum sst_hsw_device_mode {
+ SST_HSW_DEVICE_CLOCK_SLAVE = 0,
+ SST_HSW_DEVICE_CLOCK_MASTER = 1,
+};
+
+/* DX Power State */
+enum sst_hsw_dx_state {
+ SST_HSW_DX_STATE_D0 = 0,
+ SST_HSW_DX_STATE_D1 = 1,
+ SST_HSW_DX_STATE_D3 = 3,
+ SST_HSW_DX_STATE_MAX = 3,
+};
+
+/* Audio stream stage IDs */
+enum sst_hsw_fx_stage_id {
+ SST_HSW_STAGE_ID_WAVES = 0,
+ SST_HSW_STAGE_ID_DTS = 1,
+ SST_HSW_STAGE_ID_DOLBY = 2,
+ SST_HSW_STAGE_ID_BOOST = 3,
+ SST_HSW_STAGE_ID_MAX_FX_ID
+};
+
+/* DX State Type */
+enum sst_hsw_dx_type {
+ SST_HSW_DX_TYPE_FW_IMAGE = 0,
+ SST_HSW_DX_TYPE_MEMORY_DUMP = 1
+};
+
+/* Volume Curve Type*/
+enum sst_hsw_volume_curve {
+ SST_HSW_VOLUME_CURVE_NONE = 0,
+ SST_HSW_VOLUME_CURVE_FADE = 1
+};
+
+/* Sample ordering */
+enum sst_hsw_interleaving {
+ SST_HSW_INTERLEAVING_PER_CHANNEL = 0,
+ SST_HSW_INTERLEAVING_PER_SAMPLE = 1,
+};
+
+/* Channel indices */
+enum sst_hsw_channel_index {
+ SST_HSW_CHANNEL_LEFT = 0,
+ SST_HSW_CHANNEL_CENTER = 1,
+ SST_HSW_CHANNEL_RIGHT = 2,
+ SST_HSW_CHANNEL_LEFT_SURROUND = 3,
+ SST_HSW_CHANNEL_CENTER_SURROUND = 3,
+ SST_HSW_CHANNEL_RIGHT_SURROUND = 4,
+ SST_HSW_CHANNEL_LFE = 7,
+ SST_HSW_CHANNEL_INVALID = 0xF,
+};
+
+/* List of supported channel maps. */
+enum sst_hsw_channel_config {
+ SST_HSW_CHANNEL_CONFIG_MONO = 0, /* mono only. */
+ SST_HSW_CHANNEL_CONFIG_STEREO = 1, /* L & R. */
+ SST_HSW_CHANNEL_CONFIG_2_POINT_1 = 2, /* L, R & LFE; PCM only. */
+ SST_HSW_CHANNEL_CONFIG_3_POINT_0 = 3, /* L, C & R; MP3 & AAC only. */
+ SST_HSW_CHANNEL_CONFIG_3_POINT_1 = 4, /* L, C, R & LFE; PCM only. */
+ SST_HSW_CHANNEL_CONFIG_QUATRO = 5, /* L, R, Ls & Rs; PCM only. */
+ SST_HSW_CHANNEL_CONFIG_4_POINT_0 = 6, /* L, C, R & Cs; MP3 & AAC only. */
+ SST_HSW_CHANNEL_CONFIG_5_POINT_0 = 7, /* L, C, R, Ls & Rs. */
+ SST_HSW_CHANNEL_CONFIG_5_POINT_1 = 8, /* L, C, R, Ls, Rs & LFE. */
+ SST_HSW_CHANNEL_CONFIG_DUAL_MONO = 9, /* One channel replicated in two. */
+ SST_HSW_CHANNEL_CONFIG_INVALID,
+};
+
+/* List of supported bit depths. */
+enum sst_hsw_bitdepth {
+ SST_HSW_DEPTH_8BIT = 8,
+ SST_HSW_DEPTH_16BIT = 16,
+ SST_HSW_DEPTH_24BIT = 24, /* Default. */
+ SST_HSW_DEPTH_32BIT = 32,
+ SST_HSW_DEPTH_INVALID = 33,
+};
+
+enum sst_hsw_module_id {
+ SST_HSW_MODULE_BASE_FW = 0x0,
+ SST_HSW_MODULE_MP3 = 0x1,
+ SST_HSW_MODULE_AAC_5_1 = 0x2,
+ SST_HSW_MODULE_AAC_2_0 = 0x3,
+ SST_HSW_MODULE_SRC = 0x4,
+ SST_HSW_MODULE_WAVES = 0x5,
+ SST_HSW_MODULE_DOLBY = 0x6,
+ SST_HSW_MODULE_BOOST = 0x7,
+ SST_HSW_MODULE_LPAL = 0x8,
+ SST_HSW_MODULE_DTS = 0x9,
+ SST_HSW_MODULE_PCM_CAPTURE = 0xA,
+ SST_HSW_MODULE_PCM_SYSTEM = 0xB,
+ SST_HSW_MODULE_PCM_REFERENCE = 0xC,
+ SST_HSW_MODULE_PCM = 0xD,
+ SST_HSW_MODULE_BLUETOOTH_RENDER_MODULE = 0xE,
+ SST_HSW_MODULE_BLUETOOTH_CAPTURE_MODULE = 0xF,
+ SST_HSW_MAX_MODULE_ID,
+};
+
+enum sst_hsw_performance_action {
+ SST_HSW_PERF_START = 0,
+ SST_HSW_PERF_STOP = 1,
+};
+
+/* SST firmware module info */
+struct sst_hsw_module_info {
+ u8 name[SST_HSW_MAX_INFO_SIZE];
+ u8 version[SST_HSW_MAX_INFO_SIZE];
+} __attribute__((packed));
+
+/* Module entry point */
+struct sst_hsw_module_entry {
+ enum sst_hsw_module_id module_id;
+ u32 entry_point;
+} __attribute__((packed));
+
+/* Module map - alignement matches DSP */
+struct sst_hsw_module_map {
+ u8 module_entries_count;
+ struct sst_hsw_module_entry module_entries[1];
+} __attribute__((packed));
+
+struct sst_hsw_memory_info {
+ u32 offset;
+ u32 size;
+} __attribute__((packed));
+
+struct sst_hsw_fx_enable {
+ struct sst_hsw_module_map module_map;
+ struct sst_hsw_memory_info persistent_mem;
+} __attribute__((packed));
+
+struct sst_hsw_get_fx_param {
+ u32 parameter_id;
+ u32 param_size;
+} __attribute__((packed));
+
+struct sst_hsw_perf_action {
+ u32 action;
+} __attribute__((packed));
+
+struct sst_hsw_perf_data {
+ u64 timestamp;
+ u64 cycles;
+ u64 datatime;
+} __attribute__((packed));
+
+/* FW version */
+struct sst_hsw_ipc_fw_version {
+ u8 build;
+ u8 minor;
+ u8 major;
+ u8 type;
+ u8 fw_build_hash[SST_HSW_BUILD_HASH_LENGTH];
+ u32 fw_log_providers_hash;
+} __attribute__((packed));
+
+/* Stream ring info */
+struct sst_hsw_ipc_stream_ring {
+ u32 ring_pt_address;
+ u32 num_pages;
+ u32 ring_size;
+ u32 ring_offset;
+ u32 ring_first_pfn;
+} __attribute__((packed));
+
+/* Debug Dump Log Enable Request */
+struct sst_hsw_ipc_debug_log_enable_req {
+ struct sst_hsw_ipc_stream_ring ringinfo;
+ u32 config[SST_HSW_FW_LOG_CONFIG_DWORDS];
+} __attribute__((packed));
+
+/* Debug Dump Log Reply */
+struct sst_hsw_ipc_debug_log_reply {
+ u32 log_buffer_begining;
+ u32 log_buffer_size;
+} __attribute__((packed));
+
+/* Stream glitch position */
+struct sst_hsw_ipc_stream_glitch_position {
+ u32 glitch_type;
+ u32 present_pos;
+ u32 write_pos;
+} __attribute__((packed));
+
+/* Stream get position */
+struct sst_hsw_ipc_stream_get_position {
+ u32 position;
+ u32 fw_cycle_count;
+} __attribute__((packed));
+
+/* Stream set position */
+struct sst_hsw_ipc_stream_set_position {
+ u32 position;
+ u32 end_of_buffer;
+} __attribute__((packed));
+
+/* Stream Free Request */
+struct sst_hsw_ipc_stream_free_req {
+ u8 stream_id;
+ u8 reserved[3];
+} __attribute__((packed));
+
+/* Set Volume Request */
+struct sst_hsw_ipc_volume_req {
+ u32 channel;
+ u32 target_volume;
+ u64 curve_duration;
+ u32 curve_type;
+} __attribute__((packed));
+
+/* Device Configuration Request */
+struct sst_hsw_ipc_device_config_req {
+ u32 ssp_interface;
+ u32 clock_frequency;
+ u32 mode;
+ u16 clock_divider;
+ u16 reserved;
+} __attribute__((packed));
+
+/* Audio Data formats */
+struct sst_hsw_audio_data_format_ipc {
+ u32 frequency;
+ u32 bitdepth;
+ u32 map;
+ u32 config;
+ u32 style;
+ u8 ch_num;
+ u8 valid_bit;
+ u8 reserved[2];
+} __attribute__((packed));
+
+/* Stream Allocate Request */
+struct sst_hsw_ipc_stream_alloc_req {
+ u8 path_id;
+ u8 stream_type;
+ u8 format_id;
+ u8 reserved;
+ struct sst_hsw_audio_data_format_ipc format;
+ struct sst_hsw_ipc_stream_ring ringinfo;
+ struct sst_hsw_module_map map;
+ struct sst_hsw_memory_info persistent_mem;
+ struct sst_hsw_memory_info scratch_mem;
+ u32 number_of_notifications;
+} __attribute__((packed));
+
+/* Stream Allocate Reply */
+struct sst_hsw_ipc_stream_alloc_reply {
+ u32 stream_hw_id;
+ u32 mixer_hw_id; // returns rate ????
+ u32 read_position_register_address;
+ u32 presentation_position_register_address;
+ u32 peak_meter_register_address[SST_HSW_NO_CHANNELS];
+ u32 volume_register_address[SST_HSW_NO_CHANNELS];
+} __attribute__((packed));
+
+/* Get Mixer Stream Info */
+struct sst_hsw_ipc_stream_info_reply {
+ u32 mixer_hw_id;
+ u32 peak_meter_register_address[SST_HSW_NO_CHANNELS];
+ u32 volume_register_address[SST_HSW_NO_CHANNELS];
+} __attribute__((packed));
+
+/* DX State Request */
+struct sst_hsw_ipc_dx_req {
+ u8 state;
+ u8 reserved[3];
+} __attribute__((packed));
+
+/* DX State Reply Memory Info Item */
+struct sst_hsw_ipc_dx_memory_item {
+ u32 offset;
+ u32 size;
+ u32 source;
+} __attribute__((packed));
+
+/* DX State Reply */
+struct sst_hsw_ipc_dx_reply {
+ u32 entries_no;
+ struct sst_hsw_ipc_dx_memory_item mem_info[SST_HSW_MAX_DX_REGIONS];
+} __attribute__((packed));
+
+struct sst_hsw_ipc_fw_version;
+
+/* SST Init & Free */
+struct sst_hsw *sst_hsw_new(struct device *dev, const u8 *fw, size_t fw_length,
+ u32 fw_offset);
+void sst_hsw_free(struct sst_hsw *hsw);
+int sst_hsw_fw_get_version(struct sst_hsw *hsw,
+ struct sst_hsw_ipc_fw_version *version);
+u32 create_channel_map(enum sst_hsw_channel_config config);
+
+/* Stream Mixer Controls - */
+int sst_hsw_stream_mute(struct sst_hsw *hsw, struct sst_hsw_stream *stream,
+ u32 stage_id, u32 channel);
+int sst_hsw_stream_unmute(struct sst_hsw *hsw, struct sst_hsw_stream *stream,
+ u32 stage_id, u32 channel);
+
+int sst_hsw_stream_set_volume(struct sst_hsw *hsw,
+ struct sst_hsw_stream *stream, u32 stage_id, u32 channel, u32 volume);
+int sst_hsw_stream_get_volume(struct sst_hsw *hsw,
+ struct sst_hsw_stream *stream, u32 stage_id, u32 channel, u32 *volume);
+
+int sst_hsw_stream_set_volume_curve(struct sst_hsw *hsw,
+ struct sst_hsw_stream *stream, u64 curve_duration,
+ enum sst_hsw_volume_curve curve);
+
+/* Global Mixer Controls - */
+int sst_hsw_mixer_mute(struct sst_hsw *hsw, u32 stage_id, u32 channel);
+int sst_hsw_mixer_unmute(struct sst_hsw *hsw, u32 stage_id, u32 channel);
+
+int sst_hsw_mixer_set_volume(struct sst_hsw *hsw, u32 stage_id, u32 channel,
+ u32 volume);
+int sst_hsw_mixer_get_volume(struct sst_hsw *hsw, u32 stage_id, u32 channel,
+ u32 *volume);
+
+int sst_hsw_mixer_set_volume_curve(struct sst_hsw *hsw,
+ u64 curve_duration, enum sst_hsw_volume_curve curve);
+
+/* Stream API */
+struct sst_hsw_stream *sst_hsw_stream_new(struct sst_hsw *hsw, int id,
+ u32 (*get_write_position)(struct sst_hsw_stream *stream, void *data),
+ void *data);
+
+int sst_hsw_stream_free(struct sst_hsw *hsw, struct sst_hsw_stream *stream);
+
+/* Stream Configuration */
+int sst_hsw_stream_format(struct sst_hsw *hsw, struct sst_hsw_stream *stream,
+ enum sst_hsw_stream_path_id path_id,
+ enum sst_hsw_stream_type stream_type,
+ enum sst_hsw_stream_format format_id);
+
+int sst_hsw_stream_buffer(struct sst_hsw *hsw, struct sst_hsw_stream *stream,
+ u32 ring_pt_address, u32 num_pages,
+ u32 ring_size, u32 ring_offset, u32 ring_first_pfn);
+
+int sst_hsw_stream_commit(struct sst_hsw *hsw, struct sst_hsw_stream *stream);
+
+int sst_hsw_stream_set_valid(struct sst_hsw *hsw, struct sst_hsw_stream *stream,
+ u32 bits);
+int sst_hsw_stream_set_rate(struct sst_hsw *hsw, struct sst_hsw_stream *stream,
+ int rate);
+int sst_hsw_stream_set_bits(struct sst_hsw *hsw, struct sst_hsw_stream *stream,
+ enum sst_hsw_bitdepth bits);
+int sst_hsw_stream_set_channels(struct sst_hsw *hsw,
+ struct sst_hsw_stream *stream, int channels);
+int sst_hsw_stream_set_map_config(struct sst_hsw *hsw,
+ struct sst_hsw_stream *stream, u32 map,
+ enum sst_hsw_channel_config config);
+int sst_hsw_stream_set_style(struct sst_hsw *hsw, struct sst_hsw_stream *stream,
+ enum sst_hsw_interleaving style);
+int sst_hsw_stream_set_module_info(struct sst_hsw *hsw,
+ struct sst_hsw_stream *stream, enum sst_hsw_module_id module_id,
+ u32 entry_point);
+int sst_hsw_stream_set_pmemory_info(struct sst_hsw *hsw,
+ struct sst_hsw_stream *stream, u32 offset, u32 size);
+int sst_hsw_stream_set_smemory_info(struct sst_hsw *hsw,
+ struct sst_hsw_stream *stream, u32 offset, u32 size);
+int sst_hsw_stream_get_hw_id(struct sst_hsw *hsw,
+ struct sst_hsw_stream *stream);
+int sst_hsw_stream_get_mixer_id(struct sst_hsw *hsw,
+ struct sst_hsw_stream *stream);
+u32 sst_hsw_stream_get_read_reg(struct sst_hsw *hsw,
+ struct sst_hsw_stream *stream);
+u32 sst_hsw_stream_get_pointer_reg(struct sst_hsw *hsw,
+ struct sst_hsw_stream *stream);
+u32 sst_hsw_stream_get_peak_reg(struct sst_hsw *hsw,
+ struct sst_hsw_stream *stream, u32 channel);
+u32 sst_hsw_stream_get_vol_reg(struct sst_hsw *hsw,
+ struct sst_hsw_stream *stream, u32 channel);
+int sst_hsw_mixer_get_info(struct sst_hsw *hsw);
+
+/* Stream ALSA trigger operations */
+int sst_hsw_stream_pause(struct sst_hsw *hsw, struct sst_hsw_stream *stream,
+ int wait);
+int sst_hsw_stream_resume(struct sst_hsw *hsw, struct sst_hsw_stream *stream,
+ int wait);
+int sst_hsw_stream_reset(struct sst_hsw *hsw, struct sst_hsw_stream *stream);
+
+/* Stream pointer positions */
+int sst_hsw_stream_get_read_pos(struct sst_hsw *hsw,
+ struct sst_hsw_stream *stream, u32 *position);
+int sst_hsw_stream_get_write_pos(struct sst_hsw *hsw,
+ struct sst_hsw_stream *stream, u32 *position);
+int sst_hsw_stream_set_write_position(struct sst_hsw *hsw,
+ struct sst_hsw_stream *stream, u32 stage_id, u32 position);
+int sst_hsw_get_dsp_position(struct sst_hsw *hsw,
+ struct sst_hsw_stream *stream);
+
+/* HW port config */
+int sst_hsw_device_set_config(struct sst_hsw *hsw,
+ enum sst_hsw_device_id dev, enum sst_hsw_device_mclk mclk,
+ enum sst_hsw_device_mode mode, u32 clock_divider);
+
+/* DX Config */
+int sst_hsw_dx_set_state(struct sst_hsw *hsw,
+ enum sst_hsw_dx_state state, struct sst_hsw_ipc_dx_reply *dx);
+int sst_hsw_dx_get_state(struct sst_hsw *hsw, u32 item,
+ u32 *offset, u32 *size, u32 *source);
+
+/* init */
+int sst_hsw_dsp_init(struct device *dev, struct sst_pdata *pdata);
+void sst_hsw_dsp_free(struct device *dev, struct sst_pdata *pdata);
+struct sst_dsp *sst_hsw_get_dsp(struct sst_hsw *hsw);
+void sst_hsw_set_scratch_module(struct sst_hsw *hsw,
+ struct sst_module *scratch);
+
+#endif
diff --git a/sound/soc/intel/sst-haswell-pcm.c b/sound/soc/intel/sst-haswell-pcm.c
new file mode 100644
index 0000000..0a32dd1
--- /dev/null
+++ b/sound/soc/intel/sst-haswell-pcm.c
@@ -0,0 +1,872 @@
+/*
+ * Intel SST Haswell/Broadwell PCM Support
+ *
+ * Copyright (C) 2013, Intel Corporation. All rights reserved.
+ *
+ * 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.
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/dma-mapping.h>
+#include <linux/slab.h>
+#include <linux/module.h>
+#include <linux/delay.h>
+#include <asm/page.h>
+#include <asm/pgtable.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/dmaengine_pcm.h>
+#include <sound/soc.h>
+#include <sound/tlv.h>
+#include <sound/compress_driver.h>
+
+#include "sst-haswell-ipc.h"
+#include "sst-dsp-priv.h"
+#include "sst-dsp.h"
+
+#define HSW_PCM_COUNT 6
+#define HSW_VOLUME_MAX 0x7FFFFFFF /* 0dB */
+
+/* simple volume table */
+static const u32 volume_map[] = {
+ HSW_VOLUME_MAX >> 30,
+ HSW_VOLUME_MAX >> 29,
+ HSW_VOLUME_MAX >> 28,
+ HSW_VOLUME_MAX >> 27,
+ HSW_VOLUME_MAX >> 26,
+ HSW_VOLUME_MAX >> 25,
+ HSW_VOLUME_MAX >> 24,
+ HSW_VOLUME_MAX >> 23,
+ HSW_VOLUME_MAX >> 22,
+ HSW_VOLUME_MAX >> 21,
+ HSW_VOLUME_MAX >> 20,
+ HSW_VOLUME_MAX >> 19,
+ HSW_VOLUME_MAX >> 18,
+ HSW_VOLUME_MAX >> 17,
+ HSW_VOLUME_MAX >> 16,
+ HSW_VOLUME_MAX >> 15,
+ HSW_VOLUME_MAX >> 14,
+ HSW_VOLUME_MAX >> 13,
+ HSW_VOLUME_MAX >> 12,
+ HSW_VOLUME_MAX >> 11,
+ HSW_VOLUME_MAX >> 10,
+ HSW_VOLUME_MAX >> 9,
+ HSW_VOLUME_MAX >> 8,
+ HSW_VOLUME_MAX >> 7,
+ HSW_VOLUME_MAX >> 6,
+ HSW_VOLUME_MAX >> 5,
+ HSW_VOLUME_MAX >> 4,
+ HSW_VOLUME_MAX >> 3,
+ HSW_VOLUME_MAX >> 2,
+ HSW_VOLUME_MAX >> 1,
+ HSW_VOLUME_MAX >> 0,
+};
+
+#define HSW_PCM_PERIODS_MAX 64
+#define HSW_PCM_PERIODS_MIN 2
+
+static const struct snd_pcm_hardware hsw_pcm_hardware = {
+ .info = SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_PAUSE |
+ SNDRV_PCM_INFO_RESUME |
+ SNDRV_PCM_INFO_NO_PERIOD_WAKEUP,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FORMAT_S24_LE |
+ SNDRV_PCM_FMTBIT_S32_LE,
+ .period_bytes_min = PAGE_SIZE,
+ .period_bytes_max = (HSW_PCM_PERIODS_MAX / HSW_PCM_PERIODS_MIN) * PAGE_SIZE,
+ .periods_min = HSW_PCM_PERIODS_MIN,
+ .periods_max = HSW_PCM_PERIODS_MAX,
+ .buffer_bytes_max = HSW_PCM_PERIODS_MAX * PAGE_SIZE,
+};
+
+/* private data for each PCM DSP stream */
+struct hsw_pcm_data {
+ int dai_id;
+ struct sst_hsw_stream *stream;
+ u32 volume[2];
+ struct snd_pcm_substream *substream;
+ struct snd_compr_stream *cstream;
+ unsigned int wpos;
+ struct mutex mutex;
+};
+
+/* private data for the driver */
+struct hsw_priv_data {
+ /* runtime DSP */
+ struct sst_hsw *hsw;
+
+ /* page tables */
+ unsigned char *pcm_pg[HSW_PCM_COUNT][2];
+
+ /* DAI data */
+ struct hsw_pcm_data pcm[HSW_PCM_COUNT];
+};
+
+static inline u32 hsw_mixer_to_ipc(unsigned int value)
+{
+ if (value >= ARRAY_SIZE(volume_map))
+ return volume_map[0];
+ else
+ return volume_map[value];
+}
+
+static inline unsigned int hsw_ipc_to_mixer(u32 value)
+{
+ int i;
+
+ for (i = 0; i < ARRAY_SIZE(volume_map); i++) {
+ if (volume_map[i] >= value)
+ return i;
+ }
+
+ return i - 1;
+}
+
+static int hsw_stream_volume_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_platform *platform = snd_kcontrol_chip(kcontrol);
+ struct soc_mixer_control *mc =
+ (struct soc_mixer_control *)kcontrol->private_value;
+ struct hsw_priv_data *pdata =
+ snd_soc_platform_get_drvdata(platform);
+ struct hsw_pcm_data *pcm_data = &pdata->pcm[mc->reg];
+ struct sst_hsw *hsw = pdata->hsw;
+ u32 volume;
+
+ mutex_lock(&pcm_data->mutex);
+
+ if (!pcm_data->stream) {
+ pcm_data->volume[0] =
+ hsw_mixer_to_ipc(ucontrol->value.integer.value[0]);
+ pcm_data->volume[1] =
+ hsw_mixer_to_ipc(ucontrol->value.integer.value[1]);
+ mutex_unlock(&pcm_data->mutex);
+ return 0;
+ }
+
+ if (ucontrol->value.integer.value[0] ==
+ ucontrol->value.integer.value[1]) {
+ volume = hsw_mixer_to_ipc(ucontrol->value.integer.value[0]);
+ sst_hsw_stream_set_volume(hsw, pcm_data->stream, 0, 2, volume);
+ } else {
+ volume = hsw_mixer_to_ipc(ucontrol->value.integer.value[0]);
+ sst_hsw_stream_set_volume(hsw, pcm_data->stream, 0, 0, volume);
+ volume = hsw_mixer_to_ipc(ucontrol->value.integer.value[1]);
+ sst_hsw_stream_set_volume(hsw, pcm_data->stream, 0, 1, volume);
+ }
+
+ mutex_unlock(&pcm_data->mutex);
+ return 0;
+}
+
+static int hsw_stream_volume_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_platform *platform = snd_kcontrol_chip(kcontrol);
+ struct soc_mixer_control *mc =
+ (struct soc_mixer_control *)kcontrol->private_value;
+ struct hsw_priv_data *pdata =
+ snd_soc_platform_get_drvdata(platform);
+ struct hsw_pcm_data *pcm_data = &pdata->pcm[mc->reg];
+ struct sst_hsw *hsw = pdata->hsw;
+ u32 volume;
+
+ mutex_lock(&pcm_data->mutex);
+
+ if (!pcm_data->stream) {
+ ucontrol->value.integer.value[0] =
+ hsw_ipc_to_mixer(pcm_data->volume[0]);
+ ucontrol->value.integer.value[1] =
+ hsw_ipc_to_mixer(pcm_data->volume[1]);
+ mutex_unlock(&pcm_data->mutex);
+ return 0;
+ }
+
+ sst_hsw_stream_get_volume(hsw, pcm_data->stream, 0, 0, &volume);
+ ucontrol->value.integer.value[0] = hsw_ipc_to_mixer(volume);
+ sst_hsw_stream_get_volume(hsw, pcm_data->stream, 0, 1, &volume);
+ ucontrol->value.integer.value[1] = hsw_ipc_to_mixer(volume);
+ mutex_unlock(&pcm_data->mutex);
+
+ return 0;
+}
+
+static int hsw_volume_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_platform *platform = snd_kcontrol_chip(kcontrol);
+ struct hsw_priv_data *pdata = snd_soc_platform_get_drvdata(platform);
+ struct sst_hsw *hsw = pdata->hsw;
+ u32 volume;
+
+ if (ucontrol->value.integer.value[0] ==
+ ucontrol->value.integer.value[1]) {
+
+ volume = hsw_mixer_to_ipc(ucontrol->value.integer.value[0]);
+ sst_hsw_mixer_set_volume(hsw, 0, 2, volume);
+
+ } else {
+ volume = hsw_mixer_to_ipc(ucontrol->value.integer.value[0]);
+ sst_hsw_mixer_set_volume(hsw, 0, 0, volume);
+
+ volume = hsw_mixer_to_ipc(ucontrol->value.integer.value[1]);
+ sst_hsw_mixer_set_volume(hsw, 0, 1, volume);
+ }
+
+ return 0;
+}
+
+static int hsw_volume_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_platform *platform = snd_kcontrol_chip(kcontrol);
+ struct hsw_priv_data *pdata = snd_soc_platform_get_drvdata(platform);
+ struct sst_hsw *hsw = pdata->hsw;
+ unsigned int volume = 0;
+
+ sst_hsw_mixer_get_volume(hsw, 0, 0, &volume);
+ ucontrol->value.integer.value[0] = hsw_ipc_to_mixer(volume);
+
+ sst_hsw_mixer_get_volume(hsw, 0, 1, &volume);
+ ucontrol->value.integer.value[1] = hsw_ipc_to_mixer(volume);
+
+ return 0;
+}
+
+/* TLV used by both global and stream volumes */
+static const DECLARE_TLV_DB_SCALE(hsw_vol_tlv, -9000, 300, 1);
+
+/* System Pin has no volume control */
+static const struct snd_kcontrol_new hsw_volume_controls[] = {
+ /* Global DSP volume */
+ SOC_DOUBLE_EXT_TLV("Master Playback Volume", 0, 0, 8,
+ ARRAY_SIZE(volume_map) -1, 0,
+ hsw_volume_get, hsw_volume_put, hsw_vol_tlv),
+ /* Offload 0 volume */
+ SOC_DOUBLE_EXT_TLV("Media0 Playback Volume", 1, 0, 8,
+ ARRAY_SIZE(volume_map), 0,
+ hsw_stream_volume_get, hsw_stream_volume_put, hsw_vol_tlv),
+ /* Offload 1 volume */
+ SOC_DOUBLE_EXT_TLV("Media1 Playback Volume", 2, 0, 8,
+ ARRAY_SIZE(volume_map), 0,
+ hsw_stream_volume_get, hsw_stream_volume_put, hsw_vol_tlv),
+ /* Loopback volume */
+ SOC_DOUBLE_EXT_TLV("Loopback Capture Volume", 3, 0, 8,
+ ARRAY_SIZE(volume_map), 0,
+ hsw_stream_volume_get, hsw_stream_volume_put, hsw_vol_tlv),
+ /* Mic Capture volume */
+ SOC_DOUBLE_EXT_TLV("Mic Capture Volume", 4, 0, 8,
+ ARRAY_SIZE(volume_map), 0,
+ hsw_stream_volume_get, hsw_stream_volume_put, hsw_vol_tlv),
+};
+
+/* Create DMA buffer page table for DSP */
+static int create_adsp_page_table(struct hsw_priv_data *pdata,
+ struct snd_soc_pcm_runtime *rtd,
+ unsigned char *dma_area, size_t size, int pcm, int stream)
+{
+ int i, pages;
+
+ if (size % PAGE_SIZE)
+ pages = (size / PAGE_SIZE) + 1;
+ else
+ pages = size / PAGE_SIZE;
+
+ dev_dbg(rtd->dev, "generating page table for %p size 0x%zu pages %d\n",
+ dma_area, size, pages);
+
+ for (i = 0; i < pages; i++) {
+ u32 idx = (((i << 2) + i)) >> 1;
+ u32 pfn = (virt_to_phys(dma_area + i * PAGE_SIZE)) >> PAGE_SHIFT;
+ u32 *pg_table;
+
+ dev_dbg(rtd->dev, "pfn i %i idx %d pfn %x\n", i, idx, pfn);
+
+ pg_table = (u32*)(pdata->pcm_pg[pcm][stream] + idx);
+
+ if (i & 1)
+ *pg_table |= (pfn << 4);
+ else
+ *pg_table |= pfn;
+ }
+
+ return 0;
+}
+
+/* this may get called several times by oss emulation */
+static int hsw_pcm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct hsw_priv_data *pdata =
+ snd_soc_platform_get_drvdata(rtd->platform);
+ struct hsw_pcm_data *pcm_data = snd_soc_pcm_get_drvdata(rtd);
+ struct sst_hsw *hsw = pdata->hsw;
+ struct sst_module *module_data;
+ struct sst_dsp *dsp;
+ enum sst_hsw_stream_type stream_type;
+ enum sst_hsw_stream_path_id path_id;
+ u32 rate, bits, map, pages, module_id;
+ u8 channels;
+ int ret;
+
+ /* stream direction */
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ path_id = SST_HSW_STREAM_PATH_SSP0_OUT;
+ else
+ path_id = SST_HSW_STREAM_PATH_SSP0_IN;
+
+ /* DSP stream type depends on DAI ID */
+ switch (rtd->cpu_dai->id) {
+ case 0:
+ stream_type = SST_HSW_STREAM_TYPE_SYSTEM;
+ module_id = SST_HSW_MODULE_PCM_SYSTEM;
+ break;
+ case 1:
+ case 2:
+ stream_type = SST_HSW_STREAM_TYPE_RENDER;
+ module_id = SST_HSW_MODULE_PCM;
+ break;
+ case 3:
+ /* path ID needs to be OUT for loopback */
+ stream_type = SST_HSW_STREAM_TYPE_LOOPBACK;
+ path_id = SST_HSW_STREAM_PATH_SSP0_OUT;
+ module_id = SST_HSW_MODULE_PCM_REFERENCE;
+ break;
+ case 4:
+ stream_type = SST_HSW_STREAM_TYPE_CAPTURE;
+ module_id = SST_HSW_MODULE_PCM_CAPTURE;
+ break;
+ default:
+ dev_err(rtd->dev, "error: invalid DAI ID %d\n",
+ rtd->cpu_dai->id);
+ return -EINVAL;
+ };
+
+ ret = sst_hsw_stream_format(hsw, pcm_data->stream,
+ path_id, stream_type, SST_HSW_STREAM_FORMAT_PCM_FORMAT);
+ if (ret < 0) {
+ dev_err(rtd->dev, "error: failed to set format %d\n", ret);
+ return ret;
+ }
+
+ rate = params_rate(params);
+ ret = sst_hsw_stream_set_rate(hsw, pcm_data->stream, rate);
+ if (ret < 0) {
+ dev_err(rtd->dev, "error: could not set rate %d\n", rate);
+ return ret;
+ }
+
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ bits = SST_HSW_DEPTH_16BIT;
+ sst_hsw_stream_set_valid(hsw, pcm_data->stream, 16);
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ bits = SST_HSW_DEPTH_24BIT;
+ sst_hsw_stream_set_valid(hsw, pcm_data->stream, 32);
+ break;
+ default:
+ dev_err(rtd->dev, "error: invalid format %d\n",
+ params_format(params));
+ return -EINVAL;
+ }
+
+ ret = sst_hsw_stream_set_bits(hsw, pcm_data->stream, bits);
+ if (ret < 0) {
+ dev_err(rtd->dev, "error: could not set bits %d\n", bits);
+ return ret;
+ }
+
+ /* we only support stereo atm */
+ channels = params_channels(params);
+ if (channels != 2) {
+ dev_err(rtd->dev, "error: invalid channels %d\n", channels);
+ return -EINVAL;
+ }
+
+ map = create_channel_map(SST_HSW_CHANNEL_CONFIG_STEREO);
+ sst_hsw_stream_set_map_config(hsw, pcm_data->stream,
+ map, SST_HSW_CHANNEL_CONFIG_STEREO);
+
+ ret = sst_hsw_stream_set_channels(hsw, pcm_data->stream, channels);
+ if (ret < 0) {
+ dev_err(rtd->dev, "error: could not set channels %d\n",
+ channels);
+ return ret;
+ }
+
+ ret = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params));
+ if (ret < 0) {
+ dev_err(rtd->dev, "error: could not allocate %d bytes for PCM %d\n",
+ params_buffer_bytes(params), ret);
+ return ret;
+ }
+
+ ret = create_adsp_page_table(pdata, rtd, runtime->dma_area,
+ runtime->dma_bytes, rtd->cpu_dai->id, substream->stream);
+ if (ret < 0)
+ return ret;
+
+ sst_hsw_stream_set_style(hsw, pcm_data->stream,
+ SST_HSW_INTERLEAVING_PER_CHANNEL);
+
+ if (runtime->dma_bytes % PAGE_SIZE)
+ pages = (runtime->dma_bytes / PAGE_SIZE) + 1;
+ else
+ pages = runtime->dma_bytes / PAGE_SIZE;
+
+ ret = sst_hsw_stream_buffer(hsw, pcm_data->stream,
+ virt_to_phys(pdata->pcm_pg[rtd->cpu_dai->id][substream->stream]),
+ pages, runtime->dma_bytes, 0,
+ (u32)(virt_to_phys(runtime->dma_area) >> PAGE_SHIFT));
+ if (ret < 0) {
+ dev_err(rtd->dev, "error: failed to set DMA buffer %d\n", ret);
+ return ret;
+ }
+
+ dsp = sst_hsw_get_dsp(hsw);
+
+ module_data = sst_module_get_from_id(dsp, module_id);
+ if (module_data == NULL) {
+ dev_err(rtd->dev, "error: failed to get module config\n");
+ return -EINVAL;
+ }
+
+ /* we use hardcoded memory offsets atm, will be updated for new FW */
+ if (stream_type == SST_HSW_STREAM_TYPE_CAPTURE) {
+ sst_hsw_stream_set_module_info(hsw, pcm_data->stream,
+ SST_HSW_MODULE_PCM_CAPTURE, module_data->entry);
+ sst_hsw_stream_set_pmemory_info(hsw, pcm_data->stream,
+ 0x449400, 0x4000);
+ sst_hsw_stream_set_smemory_info(hsw, pcm_data->stream,
+ 0x400000, 0);
+ } else { /* stream_type == SST_HSW_STREAM_TYPE_SYSTEM */
+ sst_hsw_stream_set_module_info(hsw, pcm_data->stream,
+ SST_HSW_MODULE_PCM_SYSTEM, module_data->entry);
+
+ sst_hsw_stream_set_pmemory_info(hsw, pcm_data->stream,
+ module_data->offset, module_data->size);
+ sst_hsw_stream_set_pmemory_info(hsw, pcm_data->stream,
+ 0x44d400, 0x3800);
+
+ sst_hsw_stream_set_smemory_info(hsw, pcm_data->stream,
+ module_data->offset, module_data->size);
+ sst_hsw_stream_set_smemory_info(hsw, pcm_data->stream,
+ 0x400000, 0);
+ }
+
+ ret = sst_hsw_stream_commit(hsw, pcm_data->stream);
+ if (ret < 0) {
+ dev_err(rtd->dev, "error: failed to commit stream %d\n", ret);
+ return ret;
+ }
+
+ ret = sst_hsw_stream_pause(hsw, pcm_data->stream, 1);
+ if (ret < 0)
+ dev_err(rtd->dev, "error: failed to pause %d\n", ret);
+
+ return 0;
+}
+
+static int hsw_pcm_hw_free(struct snd_pcm_substream *substream)
+{
+ snd_pcm_lib_free_pages(substream);
+ return 0;
+}
+
+static int hsw_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct hsw_priv_data *pdata =
+ snd_soc_platform_get_drvdata(rtd->platform);
+ struct hsw_pcm_data *pcm_data = snd_soc_pcm_get_drvdata(rtd);
+ struct sst_hsw *hsw = pdata->hsw;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ sst_hsw_stream_resume(hsw, pcm_data->stream, 0);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ sst_hsw_stream_pause(hsw, pcm_data->stream, 0);
+ break;
+ default:
+ break;
+ }
+
+ return 0;
+}
+
+static u32 hsw_notify_pointer(struct sst_hsw_stream *stream, void *data)
+{
+ struct hsw_pcm_data *pcm_data = data;
+ struct snd_pcm_substream *substream = pcm_data->substream;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ u32 pos;
+
+ pos = frames_to_bytes(runtime,
+ (runtime->control->appl_ptr % runtime->buffer_size));
+
+ dev_dbg(rtd->dev, "PCM: App pointer %d bytes\n", pos);
+
+ /* let alsa know we have play a period */
+ snd_pcm_period_elapsed(substream);
+ return pos;
+}
+
+static snd_pcm_uframes_t hsw_pcm_pointer(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct hsw_priv_data *pdata =
+ snd_soc_platform_get_drvdata(rtd->platform);
+ struct hsw_pcm_data *pcm_data = snd_soc_pcm_get_drvdata(rtd);
+ struct sst_hsw *hsw = pdata->hsw;
+ snd_pcm_uframes_t offset;
+
+ offset = bytes_to_frames(runtime,
+ sst_hsw_get_dsp_position(hsw, pcm_data->stream));
+
+ dev_dbg(rtd->dev, "PCM: DMA pointer %zu bytes\n",
+ frames_to_bytes(runtime, (u32)offset));
+ return offset;
+}
+
+static int hsw_pcm_open(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct hsw_priv_data *pdata =
+ snd_soc_platform_get_drvdata(rtd->platform);
+ struct hsw_pcm_data *pcm_data;
+ struct sst_hsw *hsw = pdata->hsw;
+
+ pcm_data = &pdata->pcm[rtd->cpu_dai->id];
+
+ mutex_lock(&pcm_data->mutex);
+
+ snd_soc_pcm_set_drvdata(rtd, pcm_data);
+ pcm_data->substream = substream;
+
+ snd_soc_set_runtime_hwparams(substream, &hsw_pcm_hardware);
+
+ pcm_data->stream = sst_hsw_stream_new(hsw, rtd->cpu_dai->id,
+ hsw_notify_pointer, pcm_data);
+ if (pcm_data->stream == NULL) {
+ dev_err(rtd->dev, "error: failed to create stream\n");
+ mutex_unlock(&pcm_data->mutex);
+ return -EINVAL;
+ }
+
+ /* Set previous saved volume */
+ sst_hsw_stream_set_volume(hsw, pcm_data->stream, 0,
+ 0, pcm_data->volume[0]);
+ sst_hsw_stream_set_volume(hsw, pcm_data->stream, 0,
+ 1, pcm_data->volume[1]);
+
+ mutex_unlock(&pcm_data->mutex);
+ return 0;
+}
+
+static int hsw_pcm_close(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct hsw_priv_data *pdata =
+ snd_soc_platform_get_drvdata(rtd->platform);
+ struct hsw_pcm_data *pcm_data = snd_soc_pcm_get_drvdata(rtd);
+ struct sst_hsw *hsw = pdata->hsw;
+ int ret;
+
+ mutex_lock(&pcm_data->mutex);
+ ret = sst_hsw_stream_reset(hsw, pcm_data->stream);
+ if (ret < 0) {
+ dev_dbg(rtd->dev, "error: reset stream failed %d\n", ret);
+ goto out;
+ }
+
+ ret = sst_hsw_stream_free(hsw, pcm_data->stream);
+ if (ret < 0) {
+ dev_dbg(rtd->dev, "error: free stream failed %d\n", ret);
+ goto out;
+ }
+ pcm_data->stream = NULL;
+
+out:
+ mutex_unlock(&pcm_data->mutex);
+ return ret;
+}
+
+static struct snd_pcm_ops hsw_pcm_ops = {
+ .open = hsw_pcm_open,
+ .close = hsw_pcm_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = hsw_pcm_hw_params,
+ .hw_free = hsw_pcm_hw_free,
+ .trigger = hsw_pcm_trigger,
+ .pointer = hsw_pcm_pointer,
+ .mmap = snd_pcm_lib_default_mmap,
+};
+
+static void hsw_pcm_free(struct snd_pcm *pcm)
+{
+ snd_pcm_lib_preallocate_free_for_all(pcm);
+}
+
+static int hsw_pcm_new(struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_pcm *pcm = rtd->pcm;
+ int ret = 0;
+
+ ret = dma_coerce_mask_and_coherent(rtd->card->dev, DMA_BIT_MASK(32));
+ if (ret)
+ return ret;
+
+ if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream ||
+ pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) {
+ ret = snd_pcm_lib_preallocate_pages_for_all(pcm,
+ SNDRV_DMA_TYPE_DEV,
+ rtd->card->dev,
+ hsw_pcm_hardware.buffer_bytes_max,
+ hsw_pcm_hardware.buffer_bytes_max);
+ if (ret) {
+ dev_err(rtd->dev, "dma buffer allocation failed %d\n",
+ ret);
+ return ret;
+ }
+ }
+
+ return ret;
+}
+
+#define HSW_FORMATS \
+ (SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S16_LE |\
+ SNDRV_PCM_FMTBIT_S32_LE)
+
+static struct snd_soc_dai_driver hsw_dais[] = {
+ {
+ .name = "System Pin",
+ .playback = {
+ .stream_name = "System Playback",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_48000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ },
+ },
+ {
+ /* PCM */
+ .name = "Offload0 Pin",
+ .playback = {
+ .stream_name = "Offload0 Playback",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_8000_192000,
+ .formats = HSW_FORMATS,
+ },
+ },
+ {
+ /* PCM */
+ .name = "Offload1 Pin",
+ .playback = {
+ .stream_name = "Offload1 Playback",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_8000_192000,
+ .formats = HSW_FORMATS,
+ },
+ },
+ {
+ .name = "Loopback Pin",
+ .capture = {
+ .stream_name = "Loopback Capture",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_8000_192000,
+ .formats = HSW_FORMATS,
+ },
+ },
+ {
+ .name = "Capture Pin",
+ .capture = {
+ .stream_name = "Analog Capture",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_8000_192000,
+ .formats = HSW_FORMATS,
+ },
+ },
+};
+
+static const struct snd_soc_dapm_widget widgets[] = {
+
+ /* Backend DAIs */
+ SND_SOC_DAPM_AIF_IN("SSP0 CODEC IN", NULL, 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("SSP0 CODEC OUT", NULL, 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_IN("SSP1 BT IN", NULL, 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("SSP1 BT OUT", NULL, 0, SND_SOC_NOPM, 0, 0),
+
+ /* Global Playback Mixer */
+ SND_SOC_DAPM_MIXER("Playback VMixer", SND_SOC_NOPM, 0, 0, NULL, 0),
+};
+
+static const struct snd_soc_dapm_route graph[] = {
+
+ /* Playback Mixer */
+ {"Playback VMixer", NULL, "System Playback"},
+ {"Playback VMixer", NULL, "Offload0 Playback"},
+ {"Playback VMixer", NULL, "Offload1 Playback"},
+
+ {"SSP0 CODEC OUT", NULL, "Playback VMixer"},
+
+ {"Analog Capture", NULL, "SSP0 CODEC IN"},
+};
+
+static int hsw_pcm_probe(struct snd_soc_platform *platform)
+{
+ struct sst_pdata *pdata = dev_get_platdata(platform->dev);
+ struct hsw_priv_data *priv_data;
+ int i;
+
+ if (!pdata)
+ return -ENODEV;
+
+ priv_data = devm_kzalloc(platform->dev, sizeof(*priv_data), GFP_KERNEL);
+ priv_data->hsw = pdata->dsp;
+ snd_soc_platform_set_drvdata(platform, priv_data);
+
+ /* allocate DSP buffer page tables */
+ for (i = 0; i < ARRAY_SIZE(hsw_dais); i++) {
+
+ mutex_init(&priv_data->pcm[i].mutex);
+
+ /* playback */
+ if (hsw_dais[i].playback.channels_min) {
+ priv_data->pcm_pg[i][0] = kzalloc(PAGE_SIZE, GFP_DMA);
+ if (priv_data->pcm_pg[i][0] == NULL)
+ goto err;
+ }
+
+ /* capture */
+ if (hsw_dais[i].capture.channels_min) {
+ priv_data->pcm_pg[i][1] = kzalloc(PAGE_SIZE, GFP_DMA);
+ if (priv_data->pcm_pg[i][1] == NULL)
+ goto err;
+ }
+ }
+
+ return 0;
+
+err:
+ for (;i >= 0; i--) {
+ if (hsw_dais[i].playback.channels_min)
+ kfree(priv_data->pcm_pg[i][0]);
+ if (hsw_dais[i].capture.channels_min)
+ kfree(priv_data->pcm_pg[i][1]);
+ }
+ return -ENOMEM;
+}
+
+static int hsw_pcm_remove(struct snd_soc_platform *platform)
+{
+ struct hsw_priv_data *priv_data =
+ snd_soc_platform_get_drvdata(platform);
+ int i;
+
+ for (i = 0; i < ARRAY_SIZE(hsw_dais); i++) {
+ if (hsw_dais[i].playback.channels_min)
+ kfree(priv_data->pcm_pg[i][0]);
+ if (hsw_dais[i].capture.channels_min)
+ kfree(priv_data->pcm_pg[i][1]);
+ }
+
+ return 0;
+}
+
+static struct snd_soc_platform_driver hsw_soc_platform = {
+ .probe = hsw_pcm_probe,
+ .remove = hsw_pcm_remove,
+ .ops = &hsw_pcm_ops,
+ .pcm_new = hsw_pcm_new,
+ .pcm_free = hsw_pcm_free,
+ .controls = hsw_volume_controls,
+ .num_controls = ARRAY_SIZE(hsw_volume_controls),
+ .dapm_widgets = widgets,
+ .num_dapm_widgets = ARRAY_SIZE(widgets),
+ .dapm_routes = graph,
+ .num_dapm_routes = ARRAY_SIZE(graph),
+};
+
+static const struct snd_soc_component_driver hsw_dai_component = {
+ .name = "haswell-dai",
+};
+
+static int hsw_pcm_dev_probe(struct platform_device *pdev)
+{
+ struct sst_pdata *sst_pdata = dev_get_platdata(&pdev->dev);
+ int ret;
+
+ ret = sst_hsw_dsp_init(&pdev->dev, sst_pdata);
+ if (ret < 0)
+ return -ENODEV;
+
+ ret = snd_soc_register_platform(&pdev->dev, &hsw_soc_platform);
+ if (ret < 0)
+ goto err_plat;
+
+ ret = snd_soc_register_component(&pdev->dev, &hsw_dai_component,
+ hsw_dais, ARRAY_SIZE(hsw_dais));
+ if (ret < 0)
+ goto err_comp;
+
+ return 0;
+
+err_comp:
+ snd_soc_unregister_platform(&pdev->dev);
+err_plat:
+ sst_hsw_dsp_free(&pdev->dev, sst_pdata);
+ return 0;
+}
+
+static int hsw_pcm_dev_remove(struct platform_device *pdev)
+{
+ struct sst_pdata *sst_pdata = dev_get_platdata(&pdev->dev);
+
+ snd_soc_unregister_platform(&pdev->dev);
+ snd_soc_unregister_component(&pdev->dev);
+ sst_hsw_dsp_free(&pdev->dev, sst_pdata);
+
+ return 0;
+}
+
+static struct platform_driver hsw_pcm_driver = {
+ .driver = {
+ .name = "haswell-pcm-audio",
+ .owner = THIS_MODULE,
+ },
+
+ .probe = hsw_pcm_dev_probe,
+ .remove = hsw_pcm_dev_remove,
+};
+module_platform_driver(hsw_pcm_driver);
+
+MODULE_AUTHOR("Liam Girdwood, Xingchao Wang");
+MODULE_DESCRIPTION("Haswell/Lynxpoint + Broadwell/Wildcatpoint PCM");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:haswell-pcm-audio");
diff --git a/sound/soc/intel/sst_dsp.h b/sound/soc/intel/sst-mfld-dsp.h
index 0fce1de..3b63edc 100644
--- a/sound/soc/intel/sst_dsp.h
+++ b/sound/soc/intel/sst-mfld-dsp.h
@@ -1,7 +1,7 @@
-#ifndef __SST_DSP_H__
-#define __SST_DSP_H__
+#ifndef __SST_MFLD_DSP_H__
+#define __SST_MFLD_DSP_H__
/*
- * sst_dsp.h - Intel SST Driver for audio engine
+ * sst_mfld_dsp.h - Intel SST Driver for audio engine
*
* Copyright (C) 2008-12 Intel Corporation
* Authors: Vinod Koul <vinod.koul@linux.intel.com>
@@ -131,4 +131,4 @@ struct snd_sst_params {
struct snd_sst_alloc_params_ext aparams;
};
-#endif /* __SST_DSP_H__ */
+#endif /* __SST_MFLD_DSP_H__ */
diff --git a/sound/soc/intel/sst_platform.c b/sound/soc/intel/sst-mfld-platform.c
index f465a81..840306c 100644
--- a/sound/soc/intel/sst_platform.c
+++ b/sound/soc/intel/sst-mfld-platform.c
@@ -1,5 +1,5 @@
/*
- * sst_platform.c - Intel MID Platform driver
+ * sst_mfld_platform.c - Intel MID Platform driver
*
* Copyright (C) 2010-2013 Intel Corp
* Author: Vinod Koul <vinod.koul@intel.com>
@@ -33,7 +33,7 @@
#include <sound/pcm_params.h>
#include <sound/soc.h>
#include <sound/compress_driver.h>
-#include "sst_platform.h"
+#include "sst-mfld-platform.h"
static struct sst_device *sst;
static DEFINE_MUTEX(sst_lock);
@@ -709,7 +709,7 @@ static int sst_platform_remove(struct platform_device *pdev)
static struct platform_driver sst_platform_driver = {
.driver = {
- .name = "sst-platform",
+ .name = "sst-mfld-platform",
.owner = THIS_MODULE,
},
.probe = sst_platform_probe,
@@ -722,4 +722,4 @@ MODULE_DESCRIPTION("ASoC Intel(R) MID Platform driver");
MODULE_AUTHOR("Vinod Koul <vinod.koul@intel.com>");
MODULE_AUTHOR("Harsha Priya <priya.harsha@intel.com>");
MODULE_LICENSE("GPL v2");
-MODULE_ALIAS("platform:sst-platform");
+MODULE_ALIAS("platform:sst-mfld-platform");
diff --git a/sound/soc/intel/sst_platform.h b/sound/soc/intel/sst-mfld-platform.h
index bee64fb..0c4e2dd 100644
--- a/sound/soc/intel/sst_platform.h
+++ b/sound/soc/intel/sst-mfld-platform.h
@@ -1,5 +1,5 @@
/*
- * sst_platform.h - Intel MID Platform driver header file
+ * sst_mfld_platform.h - Intel MID Platform driver header file
*
* Copyright (C) 2010 Intel Corp
* Author: Vinod Koul <vinod.koul@intel.com>
@@ -27,7 +27,7 @@
#ifndef __SST_PLATFORMDRV_H__
#define __SST_PLATFORMDRV_H__
-#include "sst_dsp.h"
+#include "sst-mfld-dsp.h"
#define SST_MONO 1
#define SST_STEREO 2
diff --git a/sound/soc/kirkwood/Kconfig b/sound/soc/kirkwood/Kconfig
index 78ed4a4..49f8437 100644
--- a/sound/soc/kirkwood/Kconfig
+++ b/sound/soc/kirkwood/Kconfig
@@ -1,11 +1,20 @@
config SND_KIRKWOOD_SOC
tristate "SoC Audio for the Marvell Kirkwood and Dove chips"
- depends on ARCH_KIRKWOOD || ARCH_DOVE || COMPILE_TEST
+ depends on ARCH_KIRKWOOD || ARCH_DOVE || ARCH_MVEBU || COMPILE_TEST
help
Say Y or M if you want to add support for codecs attached to
the Kirkwood I2S interface. You will also need to select the
audio interfaces to support below.
+config SND_KIRKWOOD_SOC_ARMADA370_DB
+ tristate "SoC Audio support for Armada 370 DB"
+ depends on SND_KIRKWOOD_SOC && (ARCH_MVEBU || COMPILE_TEST) && I2C
+ select SND_SOC_CS42L51
+ select SND_SOC_SPDIF
+ help
+ Say Y if you want to add support for SoC audio on
+ the Armada 370 Development Board.
+
config SND_KIRKWOOD_SOC_OPENRD
tristate "SoC Audio support for Kirkwood Openrd Client"
depends on SND_KIRKWOOD_SOC && (MACH_OPENRD_CLIENT || MACH_OPENRD_ULTIMATE || COMPILE_TEST)
diff --git a/sound/soc/kirkwood/Makefile b/sound/soc/kirkwood/Makefile
index 9e78138..7c1d8fe 100644
--- a/sound/soc/kirkwood/Makefile
+++ b/sound/soc/kirkwood/Makefile
@@ -4,6 +4,8 @@ obj-$(CONFIG_SND_KIRKWOOD_SOC) += snd-soc-kirkwood.o
snd-soc-openrd-objs := kirkwood-openrd.o
snd-soc-t5325-objs := kirkwood-t5325.o
+snd-soc-armada-370-db-objs := armada-370-db.o
+obj-$(CONFIG_SND_KIRKWOOD_SOC_ARMADA370_DB) += snd-soc-armada-370-db.o
obj-$(CONFIG_SND_KIRKWOOD_SOC_OPENRD) += snd-soc-openrd.o
obj-$(CONFIG_SND_KIRKWOOD_SOC_T5325) += snd-soc-t5325.o
diff --git a/sound/soc/kirkwood/armada-370-db.c b/sound/soc/kirkwood/armada-370-db.c
new file mode 100644
index 0000000..c443338
--- /dev/null
+++ b/sound/soc/kirkwood/armada-370-db.c
@@ -0,0 +1,148 @@
+/*
+ * Copyright (C) 2014 Marvell
+ *
+ * Thomas Petazzoni <thomas.petazzoni@free-electrons.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/module.h>
+#include <linux/moduleparam.h>
+#include <linux/interrupt.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+#include <sound/soc.h>
+#include <linux/of.h>
+#include <linux/platform_data/asoc-kirkwood.h>
+#include "../codecs/cs42l51.h"
+
+static int a370db_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ unsigned int freq;
+
+ switch (params_rate(params)) {
+ default:
+ case 44100:
+ freq = 11289600;
+ break;
+ case 48000:
+ freq = 12288000;
+ break;
+ case 96000:
+ freq = 24576000;
+ break;
+ }
+
+ return snd_soc_dai_set_sysclk(codec_dai, 0, freq, SND_SOC_CLOCK_IN);
+}
+
+static struct snd_soc_ops a370db_ops = {
+ .hw_params = a370db_hw_params,
+};
+
+static const struct snd_soc_dapm_widget a370db_dapm_widgets[] = {
+ SND_SOC_DAPM_HP("Out Jack", NULL),
+ SND_SOC_DAPM_LINE("In Jack", NULL),
+};
+
+static const struct snd_soc_dapm_route a370db_route[] = {
+ { "Out Jack", NULL, "HPL" },
+ { "Out Jack", NULL, "HPR" },
+ { "AIN1L", NULL, "In Jack" },
+ { "AIN1L", NULL, "In Jack" },
+};
+
+static struct snd_soc_dai_link a370db_dai[] = {
+{
+ .name = "CS42L51",
+ .stream_name = "analog",
+ .cpu_dai_name = "i2s",
+ .codec_dai_name = "cs42l51-hifi",
+ .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS,
+ .ops = &a370db_ops,
+},
+{
+ .name = "S/PDIF out",
+ .stream_name = "spdif-out",
+ .cpu_dai_name = "spdif",
+ .codec_dai_name = "dit-hifi",
+ .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS,
+},
+{
+ .name = "S/PDIF in",
+ .stream_name = "spdif-in",
+ .cpu_dai_name = "spdif",
+ .codec_dai_name = "dir-hifi",
+ .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS,
+},
+};
+
+static struct snd_soc_card a370db = {
+ .name = "a370db",
+ .owner = THIS_MODULE,
+ .dai_link = a370db_dai,
+ .num_links = ARRAY_SIZE(a370db_dai),
+ .dapm_widgets = a370db_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(a370db_dapm_widgets),
+ .dapm_routes = a370db_route,
+ .num_dapm_routes = ARRAY_SIZE(a370db_route),
+};
+
+static int a370db_probe(struct platform_device *pdev)
+{
+ struct snd_soc_card *card = &a370db;
+
+ card->dev = &pdev->dev;
+
+ a370db_dai[0].cpu_of_node =
+ of_parse_phandle(pdev->dev.of_node,
+ "marvell,audio-controller", 0);
+ a370db_dai[0].platform_of_node = a370db_dai[0].cpu_of_node;
+
+ a370db_dai[0].codec_of_node =
+ of_parse_phandle(pdev->dev.of_node,
+ "marvell,audio-codec", 0);
+
+ a370db_dai[1].cpu_of_node = a370db_dai[0].cpu_of_node;
+ a370db_dai[1].platform_of_node = a370db_dai[0].cpu_of_node;
+
+ a370db_dai[1].codec_of_node =
+ of_parse_phandle(pdev->dev.of_node,
+ "marvell,audio-codec", 1);
+
+ a370db_dai[2].cpu_of_node = a370db_dai[0].cpu_of_node;
+ a370db_dai[2].platform_of_node = a370db_dai[0].cpu_of_node;
+
+ a370db_dai[2].codec_of_node =
+ of_parse_phandle(pdev->dev.of_node,
+ "marvell,audio-codec", 2);
+
+ return devm_snd_soc_register_card(card->dev, card);
+}
+
+static const struct of_device_id a370db_dt_ids[] = {
+ { .compatible = "marvell,a370db-audio" },
+ { },
+};
+
+static struct platform_driver a370db_driver = {
+ .driver = {
+ .name = "a370db-audio",
+ .owner = THIS_MODULE,
+ .of_match_table = of_match_ptr(a370db_dt_ids),
+ },
+ .probe = a370db_probe,
+};
+
+module_platform_driver(a370db_driver);
+
+MODULE_AUTHOR("Thomas Petazzoni <thomas.petazzoni@free-electrons.com>");
+MODULE_DESCRIPTION("ALSA SoC a370db audio client");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:a370db-audio");
diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c
index 3920a5e..9f84222 100644
--- a/sound/soc/kirkwood/kirkwood-i2s.c
+++ b/sound/soc/kirkwood/kirkwood-i2s.c
@@ -633,6 +633,7 @@ static int kirkwood_i2s_dev_remove(struct platform_device *pdev)
static struct of_device_id mvebu_audio_of_match[] = {
{ .compatible = "marvell,kirkwood-audio" },
{ .compatible = "marvell,dove-audio" },
+ { .compatible = "marvell,armada370-audio" },
{ }
};
MODULE_DEVICE_TABLE(of, mvebu_audio_of_match);
diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig
index 22ad9c5..e006593 100644
--- a/sound/soc/omap/Kconfig
+++ b/sound/soc/omap/Kconfig
@@ -58,7 +58,7 @@ config SND_OMAP_SOC_OSK5912
tristate "SoC Audio support for omap osk5912"
depends on SND_OMAP_SOC && MACH_OMAP_OSK && I2C
select SND_OMAP_SOC_MCBSP
- select SND_SOC_TLV320AIC23
+ select SND_SOC_TLV320AIC23_I2C
help
Say Y if you want to add support for SoC audio on osk5912.
@@ -66,7 +66,7 @@ config SND_OMAP_SOC_AM3517EVM
tristate "SoC Audio support for OMAP3517 / AM3517 EVM"
depends on SND_OMAP_SOC && MACH_OMAP3517EVM && I2C
select SND_OMAP_SOC_MCBSP
- select SND_SOC_TLV320AIC23
+ select SND_SOC_TLV320AIC23_I2C
help
Say Y if you want to add support for SoC audio on the OMAP3517 / AM3517
EVM.
diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c
index 6294464..56a5219 100644
--- a/sound/soc/omap/ams-delta.c
+++ b/sound/soc/omap/ams-delta.c
@@ -103,60 +103,62 @@ static int ams_delta_set_audio_mode(struct snd_kcontrol *kcontrol,
if (!codec->hw_write)
return -EUNATCH;
- if (ucontrol->value.enumerated.item[0] >= control->max)
+ if (ucontrol->value.enumerated.item[0] >= control->items)
return -EINVAL;
- mutex_lock(&codec->mutex);
+ snd_soc_dapm_mutex_lock(dapm);
/* Translate selection to bitmap */
pins = ams_delta_audio_mode_pins[ucontrol->value.enumerated.item[0]];
/* Setup pins after corresponding bits if changed */
pin = !!(pins & (1 << AMS_DELTA_MOUTHPIECE));
+
if (pin != snd_soc_dapm_get_pin_status(dapm, "Mouthpiece")) {
changed = 1;
if (pin)
- snd_soc_dapm_enable_pin(dapm, "Mouthpiece");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Mouthpiece");
else
- snd_soc_dapm_disable_pin(dapm, "Mouthpiece");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Mouthpiece");
}
pin = !!(pins & (1 << AMS_DELTA_EARPIECE));
if (pin != snd_soc_dapm_get_pin_status(dapm, "Earpiece")) {
changed = 1;
if (pin)
- snd_soc_dapm_enable_pin(dapm, "Earpiece");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Earpiece");
else
- snd_soc_dapm_disable_pin(dapm, "Earpiece");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Earpiece");
}
pin = !!(pins & (1 << AMS_DELTA_MICROPHONE));
if (pin != snd_soc_dapm_get_pin_status(dapm, "Microphone")) {
changed = 1;
if (pin)
- snd_soc_dapm_enable_pin(dapm, "Microphone");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Microphone");
else
- snd_soc_dapm_disable_pin(dapm, "Microphone");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Microphone");
}
pin = !!(pins & (1 << AMS_DELTA_SPEAKER));
if (pin != snd_soc_dapm_get_pin_status(dapm, "Speaker")) {
changed = 1;
if (pin)
- snd_soc_dapm_enable_pin(dapm, "Speaker");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Speaker");
else
- snd_soc_dapm_disable_pin(dapm, "Speaker");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Speaker");
}
pin = !!(pins & (1 << AMS_DELTA_AGC));
if (pin != ams_delta_audio_agc) {
ams_delta_audio_agc = pin;
changed = 1;
if (pin)
- snd_soc_dapm_enable_pin(dapm, "AGCIN");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "AGCIN");
else
- snd_soc_dapm_disable_pin(dapm, "AGCIN");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "AGCIN");
}
+
if (changed)
- snd_soc_dapm_sync(dapm);
+ snd_soc_dapm_sync_unlocked(dapm);
- mutex_unlock(&codec->mutex);
+ snd_soc_dapm_mutex_unlock(dapm);
return changed;
}
@@ -194,13 +196,11 @@ static int ams_delta_get_audio_mode(struct snd_kcontrol *kcontrol,
return 0;
}
-static const struct soc_enum ams_delta_audio_enum[] = {
- SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(ams_delta_audio_mode),
- ams_delta_audio_mode),
-};
+static const SOC_ENUM_SINGLE_EXT_DECL(ams_delta_audio_enum,
+ ams_delta_audio_mode);
static const struct snd_kcontrol_new ams_delta_audio_controls[] = {
- SOC_ENUM_EXT("Audio Mode", ams_delta_audio_enum[0],
+ SOC_ENUM_EXT("Audio Mode", ams_delta_audio_enum,
ams_delta_get_audio_mode, ams_delta_set_audio_mode),
};
@@ -315,12 +315,17 @@ static void cx81801_close(struct tty_struct *tty)
v253_ops.close(tty);
/* Revert back to default audio input/output constellation */
- snd_soc_dapm_disable_pin(dapm, "Mouthpiece");
- snd_soc_dapm_enable_pin(dapm, "Earpiece");
- snd_soc_dapm_enable_pin(dapm, "Microphone");
- snd_soc_dapm_disable_pin(dapm, "Speaker");
- snd_soc_dapm_disable_pin(dapm, "AGCIN");
- snd_soc_dapm_sync(dapm);
+ snd_soc_dapm_mutex_lock(dapm);
+
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Mouthpiece");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Earpiece");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Microphone");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Speaker");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "AGCIN");
+
+ snd_soc_dapm_sync_unlocked(dapm);
+
+ snd_soc_dapm_mutex_unlock(dapm);
}
/* Line discipline .hangup() */
diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c
index d163e18..fd4d9c8 100644
--- a/sound/soc/omap/n810.c
+++ b/sound/soc/omap/n810.c
@@ -68,26 +68,30 @@ static void n810_ext_control(struct snd_soc_dapm_context *dapm)
break;
}
+ snd_soc_dapm_mutex_lock(dapm);
+
if (n810_spk_func)
- snd_soc_dapm_enable_pin(dapm, "Ext Spk");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Ext Spk");
else
- snd_soc_dapm_disable_pin(dapm, "Ext Spk");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Ext Spk");
if (hp)
- snd_soc_dapm_enable_pin(dapm, "Headphone Jack");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Headphone Jack");
else
- snd_soc_dapm_disable_pin(dapm, "Headphone Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Headphone Jack");
if (line1l)
- snd_soc_dapm_enable_pin(dapm, "LINE1L");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "LINE1L");
else
- snd_soc_dapm_disable_pin(dapm, "LINE1L");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "LINE1L");
if (n810_dmic_func)
- snd_soc_dapm_enable_pin(dapm, "DMic");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "DMic");
else
- snd_soc_dapm_disable_pin(dapm, "DMic");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "DMic");
+
+ snd_soc_dapm_sync_unlocked(dapm);
- snd_soc_dapm_sync(dapm);
+ snd_soc_dapm_mutex_unlock(dapm);
}
static int n810_startup(struct snd_pcm_substream *substream)
diff --git a/sound/soc/omap/omap-abe-twl6040.c b/sound/soc/omap/omap-abe-twl6040.c
index ebb1390..024dafc 100644
--- a/sound/soc/omap/omap-abe-twl6040.c
+++ b/sound/soc/omap/omap-abe-twl6040.c
@@ -203,8 +203,7 @@ static const struct snd_soc_dapm_route dmic_audio_map[] = {
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;
+ struct snd_soc_dapm_context *dapm = &rtd->card->dapm;
return snd_soc_dapm_add_routes(dapm, dmic_audio_map,
ARRAY_SIZE(dmic_audio_map));
diff --git a/sound/soc/omap/rx51.c b/sound/soc/omap/rx51.c
index 611179c..7fb3d4b 100644
--- a/sound/soc/omap/rx51.c
+++ b/sound/soc/omap/rx51.c
@@ -74,26 +74,30 @@ static void rx51_ext_control(struct snd_soc_dapm_context *dapm)
break;
}
+ snd_soc_dapm_mutex_lock(dapm);
+
if (rx51_spk_func)
- snd_soc_dapm_enable_pin(dapm, "Ext Spk");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Ext Spk");
else
- snd_soc_dapm_disable_pin(dapm, "Ext Spk");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Ext Spk");
if (rx51_dmic_func)
- snd_soc_dapm_enable_pin(dapm, "DMic");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "DMic");
else
- snd_soc_dapm_disable_pin(dapm, "DMic");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "DMic");
if (hp)
- snd_soc_dapm_enable_pin(dapm, "Headphone Jack");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Headphone Jack");
else
- snd_soc_dapm_disable_pin(dapm, "Headphone Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Headphone Jack");
if (hs)
- snd_soc_dapm_enable_pin(dapm, "HS Mic");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "HS Mic");
else
- snd_soc_dapm_disable_pin(dapm, "HS Mic");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "HS Mic");
gpio_set_value(RX51_TVOUT_SEL_GPIO, tvout);
- snd_soc_dapm_sync(dapm);
+ snd_soc_dapm_sync_unlocked(dapm);
+
+ snd_soc_dapm_mutex_unlock(dapm);
}
static int rx51_startup(struct snd_pcm_substream *substream)
diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c
index 1853d41..5a88136 100644
--- a/sound/soc/pxa/corgi.c
+++ b/sound/soc/pxa/corgi.c
@@ -47,64 +47,63 @@ static int corgi_spk_func;
static void corgi_ext_control(struct snd_soc_dapm_context *dapm)
{
+ snd_soc_dapm_mutex_lock(dapm);
+
/* set up jack connection */
switch (corgi_jack_func) {
case CORGI_HP:
/* set = unmute headphone */
gpio_set_value(CORGI_GPIO_MUTE_L, 1);
gpio_set_value(CORGI_GPIO_MUTE_R, 1);
- snd_soc_dapm_disable_pin(dapm, "Mic Jack");
- snd_soc_dapm_disable_pin(dapm, "Line Jack");
- snd_soc_dapm_enable_pin(dapm, "Headphone Jack");
- snd_soc_dapm_disable_pin(dapm, "Headset Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Mic Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Line Jack");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Headphone Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Headset Jack");
break;
case CORGI_MIC:
/* reset = mute headphone */
gpio_set_value(CORGI_GPIO_MUTE_L, 0);
gpio_set_value(CORGI_GPIO_MUTE_R, 0);
- snd_soc_dapm_enable_pin(dapm, "Mic Jack");
- snd_soc_dapm_disable_pin(dapm, "Line Jack");
- snd_soc_dapm_disable_pin(dapm, "Headphone Jack");
- snd_soc_dapm_disable_pin(dapm, "Headset Jack");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Mic Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Line Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Headphone Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Headset Jack");
break;
case CORGI_LINE:
gpio_set_value(CORGI_GPIO_MUTE_L, 0);
gpio_set_value(CORGI_GPIO_MUTE_R, 0);
- snd_soc_dapm_disable_pin(dapm, "Mic Jack");
- snd_soc_dapm_enable_pin(dapm, "Line Jack");
- snd_soc_dapm_disable_pin(dapm, "Headphone Jack");
- snd_soc_dapm_disable_pin(dapm, "Headset Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Mic Jack");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Line Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Headphone Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Headset Jack");
break;
case CORGI_HEADSET:
gpio_set_value(CORGI_GPIO_MUTE_L, 0);
gpio_set_value(CORGI_GPIO_MUTE_R, 1);
- snd_soc_dapm_enable_pin(dapm, "Mic Jack");
- snd_soc_dapm_disable_pin(dapm, "Line Jack");
- snd_soc_dapm_disable_pin(dapm, "Headphone Jack");
- snd_soc_dapm_enable_pin(dapm, "Headset Jack");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Mic Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Line Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Headphone Jack");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Headset Jack");
break;
}
if (corgi_spk_func == CORGI_SPK_ON)
- snd_soc_dapm_enable_pin(dapm, "Ext Spk");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Ext Spk");
else
- snd_soc_dapm_disable_pin(dapm, "Ext Spk");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Ext Spk");
/* signal a DAPM event */
- snd_soc_dapm_sync(dapm);
+ snd_soc_dapm_sync_unlocked(dapm);
+
+ snd_soc_dapm_mutex_unlock(dapm);
}
static int corgi_startup(struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_codec *codec = rtd->codec;
-
- mutex_lock(&codec->mutex);
/* check the jack status at stream startup */
- corgi_ext_control(&codec->dapm);
-
- mutex_unlock(&codec->mutex);
+ corgi_ext_control(&rtd->card->dapm);
return 0;
}
diff --git a/sound/soc/pxa/e740_wm9705.c b/sound/soc/pxa/e740_wm9705.c
index 44b5c09..c29feda 100644
--- a/sound/soc/pxa/e740_wm9705.c
+++ b/sound/soc/pxa/e740_wm9705.c
@@ -103,11 +103,6 @@ static int e740_ac97_init(struct snd_soc_pcm_runtime *rtd)
snd_soc_dapm_nc_pin(dapm, "PCBEEP");
snd_soc_dapm_nc_pin(dapm, "MIC2");
- snd_soc_dapm_new_controls(dapm, e740_dapm_widgets,
- ARRAY_SIZE(e740_dapm_widgets));
-
- snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
-
return 0;
}
@@ -136,6 +131,11 @@ static struct snd_soc_card e740 = {
.owner = THIS_MODULE,
.dai_link = e740_dai,
.num_links = ARRAY_SIZE(e740_dai),
+
+ .dapm_widgets = e740_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(e740_dapm_widgets),
+ .dapm_routes = audio_map,
+ .num_dapm_routes = ARRAY_SIZE(audio_map),
};
static struct gpio e740_audio_gpios[] = {
diff --git a/sound/soc/pxa/e750_wm9705.c b/sound/soc/pxa/e750_wm9705.c
index c34e447..ee36aba 100644
--- a/sound/soc/pxa/e750_wm9705.c
+++ b/sound/soc/pxa/e750_wm9705.c
@@ -85,11 +85,6 @@ static int e750_ac97_init(struct snd_soc_pcm_runtime *rtd)
snd_soc_dapm_nc_pin(dapm, "PCBEEP");
snd_soc_dapm_nc_pin(dapm, "MIC2");
- snd_soc_dapm_new_controls(dapm, e750_dapm_widgets,
- ARRAY_SIZE(e750_dapm_widgets));
-
- snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
-
return 0;
}
@@ -119,6 +114,11 @@ static struct snd_soc_card e750 = {
.owner = THIS_MODULE,
.dai_link = e750_dai,
.num_links = ARRAY_SIZE(e750_dai),
+
+ .dapm_widgets = e750_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(e750_dapm_widgets),
+ .dapm_routes = audio_map,
+ .num_dapm_routes = ARRAY_SIZE(audio_map),
};
static struct gpio e750_audio_gpios[] = {
diff --git a/sound/soc/pxa/e800_wm9712.c b/sound/soc/pxa/e800_wm9712.c
index 3137f80..24c2078 100644
--- a/sound/soc/pxa/e800_wm9712.c
+++ b/sound/soc/pxa/e800_wm9712.c
@@ -71,19 +71,6 @@ static const struct snd_soc_dapm_route audio_map[] = {
{"MIC2", NULL, "Mic (Internal2)"},
};
-static int e800_ac97_init(struct snd_soc_pcm_runtime *rtd)
-{
- struct snd_soc_codec *codec = rtd->codec;
- struct snd_soc_dapm_context *dapm = &codec->dapm;
-
- snd_soc_dapm_new_controls(dapm, e800_dapm_widgets,
- ARRAY_SIZE(e800_dapm_widgets));
-
- snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
-
- return 0;
-}
-
static struct snd_soc_dai_link e800_dai[] = {
{
.name = "AC97",
@@ -92,7 +79,6 @@ static struct snd_soc_dai_link e800_dai[] = {
.codec_dai_name = "wm9712-hifi",
.platform_name = "pxa-pcm-audio",
.codec_name = "wm9712-codec",
- .init = e800_ac97_init,
},
{
.name = "AC97 Aux",
@@ -109,6 +95,11 @@ static struct snd_soc_card e800 = {
.owner = THIS_MODULE,
.dai_link = e800_dai,
.num_links = ARRAY_SIZE(e800_dai),
+
+ .dapm_widgets = e800_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(e800_dapm_widgets),
+ .dapm_routes = audio_map,
+ .num_dapm_routes = ARRAY_SIZE(audio_map),
};
static struct gpio e800_audio_gpios[] = {
diff --git a/sound/soc/pxa/magician.c b/sound/soc/pxa/magician.c
index aace19e..259e048 100644
--- a/sound/soc/pxa/magician.c
+++ b/sound/soc/pxa/magician.c
@@ -41,44 +41,42 @@ static int magician_hp_switch;
static int magician_spk_switch = 1;
static int magician_in_sel = MAGICIAN_MIC;
-static void magician_ext_control(struct snd_soc_codec *codec)
+static void magician_ext_control(struct snd_soc_dapm_context *dapm)
{
- struct snd_soc_dapm_context *dapm = &codec->dapm;
+
+ snd_soc_dapm_mutex_lock(dapm);
if (magician_spk_switch)
- snd_soc_dapm_enable_pin(dapm, "Speaker");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Speaker");
else
- snd_soc_dapm_disable_pin(dapm, "Speaker");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Speaker");
if (magician_hp_switch)
- snd_soc_dapm_enable_pin(dapm, "Headphone Jack");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Headphone Jack");
else
- snd_soc_dapm_disable_pin(dapm, "Headphone Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Headphone Jack");
switch (magician_in_sel) {
case MAGICIAN_MIC:
- snd_soc_dapm_disable_pin(dapm, "Headset Mic");
- snd_soc_dapm_enable_pin(dapm, "Call Mic");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Headset Mic");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Call Mic");
break;
case MAGICIAN_MIC_EXT:
- snd_soc_dapm_disable_pin(dapm, "Call Mic");
- snd_soc_dapm_enable_pin(dapm, "Headset Mic");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Call Mic");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Headset Mic");
break;
}
- snd_soc_dapm_sync(dapm);
+ snd_soc_dapm_sync_unlocked(dapm);
+
+ snd_soc_dapm_mutex_unlock(dapm);
}
static int magician_startup(struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_codec *codec = rtd->codec;
-
- mutex_lock(&codec->mutex);
/* check the jack status at stream startup */
- magician_ext_control(codec);
-
- mutex_unlock(&codec->mutex);
+ magician_ext_control(&rtd->card->dapm);
return 0;
}
@@ -277,13 +275,13 @@ static int magician_get_hp(struct snd_kcontrol *kcontrol,
static int magician_set_hp(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
- struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct snd_soc_card *card = snd_kcontrol_chip(kcontrol);
if (magician_hp_switch == ucontrol->value.integer.value[0])
return 0;
magician_hp_switch = ucontrol->value.integer.value[0];
- magician_ext_control(codec);
+ magician_ext_control(&card->dapm);
return 1;
}
@@ -297,13 +295,13 @@ static int magician_get_spk(struct snd_kcontrol *kcontrol,
static int magician_set_spk(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
- struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct snd_soc_card *card = snd_kcontrol_chip(kcontrol);
if (magician_spk_switch == ucontrol->value.integer.value[0])
return 0;
magician_spk_switch = ucontrol->value.integer.value[0];
- magician_ext_control(codec);
+ magician_ext_control(&card->dapm);
return 1;
}
@@ -400,7 +398,6 @@ static int magician_uda1380_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_codec *codec = rtd->codec;
struct snd_soc_dapm_context *dapm = &codec->dapm;
- int err;
/* NC codec pins */
snd_soc_dapm_nc_pin(dapm, "VOUTLHP");
@@ -410,19 +407,6 @@ static int magician_uda1380_init(struct snd_soc_pcm_runtime *rtd)
snd_soc_dapm_nc_pin(dapm, "VINL");
snd_soc_dapm_nc_pin(dapm, "VINR");
- /* Add magician specific controls */
- err = snd_soc_add_codec_controls(codec, uda1380_magician_controls,
- ARRAY_SIZE(uda1380_magician_controls));
- if (err < 0)
- return err;
-
- /* Add magician specific widgets */
- snd_soc_dapm_new_controls(dapm, uda1380_dapm_widgets,
- ARRAY_SIZE(uda1380_dapm_widgets));
-
- /* Set up magician specific audio path interconnects */
- snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
-
return 0;
}
@@ -456,6 +440,12 @@ static struct snd_soc_card snd_soc_card_magician = {
.dai_link = magician_dai,
.num_links = ARRAY_SIZE(magician_dai),
+ .controls = uda1380_magician_controls,
+ .num_controls = ARRAY_SIZE(uda1380_magician_controls),
+ .dapm_widgets = uda1380_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(uda1380_dapm_widgets),
+ .dapm_routes = audio_map,
+ .num_dapm_routes = ARRAY_SIZE(audio_map),
};
static struct platform_device *magician_snd_device;
diff --git a/sound/soc/pxa/mioa701_wm9713.c b/sound/soc/pxa/mioa701_wm9713.c
index 160c524..595eee3 100644
--- a/sound/soc/pxa/mioa701_wm9713.c
+++ b/sound/soc/pxa/mioa701_wm9713.c
@@ -127,16 +127,8 @@ static const struct snd_soc_dapm_route audio_map[] = {
static int mioa701_wm9713_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_codec *codec = rtd->codec;
- struct snd_soc_dapm_context *dapm = &codec->dapm;
unsigned short reg;
- /* Add mioa701 specific widgets */
- snd_soc_dapm_new_controls(dapm, mioa701_dapm_widgets,
- ARRAY_SIZE(mioa701_dapm_widgets));
-
- /* Set up mioa701 specific audio path audio_mapnects */
- snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
-
/* Prepare GPIO8 for rear speaker amplifier */
reg = codec->driver->read(codec, AC97_GPIO_CFG);
codec->driver->write(codec, AC97_GPIO_CFG, reg | 0x0100);
@@ -145,12 +137,6 @@ static int mioa701_wm9713_init(struct snd_soc_pcm_runtime *rtd)
reg = codec->driver->read(codec, AC97_3D_CONTROL);
codec->driver->write(codec, AC97_3D_CONTROL, reg | 0xc000);
- snd_soc_dapm_enable_pin(dapm, "Front Speaker");
- snd_soc_dapm_enable_pin(dapm, "Rear Speaker");
- snd_soc_dapm_enable_pin(dapm, "Front Mic");
- snd_soc_dapm_enable_pin(dapm, "GSM Line In");
- snd_soc_dapm_enable_pin(dapm, "GSM Line Out");
-
return 0;
}
@@ -183,6 +169,11 @@ static struct snd_soc_card mioa701 = {
.owner = THIS_MODULE,
.dai_link = mioa701_dai,
.num_links = ARRAY_SIZE(mioa701_dai),
+
+ .dapm_widgets = mioa701_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(mioa701_dapm_widgets),
+ .dapm_routes = audio_map,
+ .num_dapm_routes = ARRAY_SIZE(audio_map),
};
static int mioa701_wm9713_probe(struct platform_device *pdev)
diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c
index c93e138..c6bdc6c 100644
--- a/sound/soc/pxa/poodle.c
+++ b/sound/soc/pxa/poodle.c
@@ -74,14 +74,9 @@ static void poodle_ext_control(struct snd_soc_dapm_context *dapm)
static int poodle_startup(struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_codec *codec = rtd->codec;
-
- mutex_lock(&codec->mutex);
/* check the jack status at stream startup */
- poodle_ext_control(&codec->dapm);
-
- mutex_unlock(&codec->mutex);
+ poodle_ext_control(&rtd->card->dapm);
return 0;
}
diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c
index fc052d8..1373b01 100644
--- a/sound/soc/pxa/spitz.c
+++ b/sound/soc/pxa/spitz.c
@@ -46,74 +46,74 @@ static int spitz_mic_gpio;
static void spitz_ext_control(struct snd_soc_dapm_context *dapm)
{
+ snd_soc_dapm_mutex_lock(dapm);
+
if (spitz_spk_func == SPITZ_SPK_ON)
- snd_soc_dapm_enable_pin(dapm, "Ext Spk");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Ext Spk");
else
- snd_soc_dapm_disable_pin(dapm, "Ext Spk");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Ext Spk");
/* set up jack connection */
switch (spitz_jack_func) {
case SPITZ_HP:
/* enable and unmute hp jack, disable mic bias */
- snd_soc_dapm_disable_pin(dapm, "Headset Jack");
- snd_soc_dapm_disable_pin(dapm, "Mic Jack");
- snd_soc_dapm_disable_pin(dapm, "Line Jack");
- snd_soc_dapm_enable_pin(dapm, "Headphone Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Headset Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Mic Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Line Jack");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Headphone Jack");
gpio_set_value(SPITZ_GPIO_MUTE_L, 1);
gpio_set_value(SPITZ_GPIO_MUTE_R, 1);
break;
case SPITZ_MIC:
/* enable mic jack and bias, mute hp */
- snd_soc_dapm_disable_pin(dapm, "Headphone Jack");
- snd_soc_dapm_disable_pin(dapm, "Headset Jack");
- snd_soc_dapm_disable_pin(dapm, "Line Jack");
- snd_soc_dapm_enable_pin(dapm, "Mic Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Headphone Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Headset Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Line Jack");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Mic Jack");
gpio_set_value(SPITZ_GPIO_MUTE_L, 0);
gpio_set_value(SPITZ_GPIO_MUTE_R, 0);
break;
case SPITZ_LINE:
/* enable line jack, disable mic bias and mute hp */
- snd_soc_dapm_disable_pin(dapm, "Headphone Jack");
- snd_soc_dapm_disable_pin(dapm, "Headset Jack");
- snd_soc_dapm_disable_pin(dapm, "Mic Jack");
- snd_soc_dapm_enable_pin(dapm, "Line Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Headphone Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Headset Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Mic Jack");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Line Jack");
gpio_set_value(SPITZ_GPIO_MUTE_L, 0);
gpio_set_value(SPITZ_GPIO_MUTE_R, 0);
break;
case SPITZ_HEADSET:
/* enable and unmute headset jack enable mic bias, mute L hp */
- snd_soc_dapm_disable_pin(dapm, "Headphone Jack");
- snd_soc_dapm_enable_pin(dapm, "Mic Jack");
- snd_soc_dapm_disable_pin(dapm, "Line Jack");
- snd_soc_dapm_enable_pin(dapm, "Headset Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Headphone Jack");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Mic Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Line Jack");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Headset Jack");
gpio_set_value(SPITZ_GPIO_MUTE_L, 0);
gpio_set_value(SPITZ_GPIO_MUTE_R, 1);
break;
case SPITZ_HP_OFF:
/* jack removed, everything off */
- snd_soc_dapm_disable_pin(dapm, "Headphone Jack");
- snd_soc_dapm_disable_pin(dapm, "Headset Jack");
- snd_soc_dapm_disable_pin(dapm, "Mic Jack");
- snd_soc_dapm_disable_pin(dapm, "Line Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Headphone Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Headset Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Mic Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Line Jack");
gpio_set_value(SPITZ_GPIO_MUTE_L, 0);
gpio_set_value(SPITZ_GPIO_MUTE_R, 0);
break;
}
- snd_soc_dapm_sync(dapm);
+
+ snd_soc_dapm_sync_unlocked(dapm);
+
+ snd_soc_dapm_mutex_unlock(dapm);
}
static int spitz_startup(struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_codec *codec = rtd->codec;
-
- mutex_lock(&codec->mutex);
/* check the jack status at stream startup */
- spitz_ext_control(&codec->dapm);
-
- mutex_unlock(&codec->mutex);
+ spitz_ext_control(&rtd->card->dapm);
return 0;
}
diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c
index 1d9c2ed..4a956d1 100644
--- a/sound/soc/pxa/tosa.c
+++ b/sound/soc/pxa/tosa.c
@@ -44,48 +44,46 @@
static int tosa_jack_func;
static int tosa_spk_func;
-static void tosa_ext_control(struct snd_soc_codec *codec)
+static void tosa_ext_control(struct snd_soc_dapm_context *dapm)
{
- struct snd_soc_dapm_context *dapm = &codec->dapm;
+
+ snd_soc_dapm_mutex_lock(dapm);
/* set up jack connection */
switch (tosa_jack_func) {
case TOSA_HP:
- snd_soc_dapm_disable_pin(dapm, "Mic (Internal)");
- snd_soc_dapm_enable_pin(dapm, "Headphone Jack");
- snd_soc_dapm_disable_pin(dapm, "Headset Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Mic (Internal)");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Headphone Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Headset Jack");
break;
case TOSA_MIC_INT:
- snd_soc_dapm_enable_pin(dapm, "Mic (Internal)");
- snd_soc_dapm_disable_pin(dapm, "Headphone Jack");
- snd_soc_dapm_disable_pin(dapm, "Headset Jack");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Mic (Internal)");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Headphone Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Headset Jack");
break;
case TOSA_HEADSET:
- snd_soc_dapm_disable_pin(dapm, "Mic (Internal)");
- snd_soc_dapm_disable_pin(dapm, "Headphone Jack");
- snd_soc_dapm_enable_pin(dapm, "Headset Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Mic (Internal)");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Headphone Jack");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Headset Jack");
break;
}
if (tosa_spk_func == TOSA_SPK_ON)
- snd_soc_dapm_enable_pin(dapm, "Speaker");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Speaker");
else
- snd_soc_dapm_disable_pin(dapm, "Speaker");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Speaker");
+
+ snd_soc_dapm_sync_unlocked(dapm);
- snd_soc_dapm_sync(dapm);
+ snd_soc_dapm_mutex_unlock(dapm);
}
static int tosa_startup(struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_codec *codec = rtd->codec;
-
- mutex_lock(&codec->mutex);
/* check the jack status at stream startup */
- tosa_ext_control(codec);
-
- mutex_unlock(&codec->mutex);
+ tosa_ext_control(&rtd->card->dapm);
return 0;
}
@@ -104,13 +102,13 @@ static int tosa_get_jack(struct snd_kcontrol *kcontrol,
static int tosa_set_jack(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
- struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct snd_soc_card *card = snd_kcontrol_chip(kcontrol);
if (tosa_jack_func == ucontrol->value.integer.value[0])
return 0;
tosa_jack_func = ucontrol->value.integer.value[0];
- tosa_ext_control(codec);
+ tosa_ext_control(&card->dapm);
return 1;
}
@@ -124,13 +122,13 @@ static int tosa_get_spk(struct snd_kcontrol *kcontrol,
static int tosa_set_spk(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
- struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct snd_soc_card *card = snd_kcontrol_chip(kcontrol);
if (tosa_spk_func == ucontrol->value.integer.value[0])
return 0;
tosa_spk_func = ucontrol->value.integer.value[0];
- tosa_ext_control(codec);
+ tosa_ext_control(&card->dapm);
return 1;
}
@@ -191,24 +189,10 @@ static int tosa_ac97_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_codec *codec = rtd->codec;
struct snd_soc_dapm_context *dapm = &codec->dapm;
- int err;
snd_soc_dapm_nc_pin(dapm, "OUT3");
snd_soc_dapm_nc_pin(dapm, "MONOOUT");
- /* add tosa specific controls */
- err = snd_soc_add_codec_controls(codec, tosa_controls,
- ARRAY_SIZE(tosa_controls));
- if (err < 0)
- return err;
-
- /* add tosa specific widgets */
- snd_soc_dapm_new_controls(dapm, tosa_dapm_widgets,
- ARRAY_SIZE(tosa_dapm_widgets));
-
- /* set up tosa specific audio path audio_map */
- snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
-
return 0;
}
@@ -239,6 +223,13 @@ static struct snd_soc_card tosa = {
.owner = THIS_MODULE,
.dai_link = tosa_dai,
.num_links = ARRAY_SIZE(tosa_dai),
+
+ .controls = tosa_controls,
+ .num_controls = ARRAY_SIZE(tosa_controls),
+ .dapm_widgets = tosa_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(tosa_dapm_widgets),
+ .dapm_routes = audio_map,
+ .num_dapm_routes = ARRAY_SIZE(audio_map),
};
static int tosa_probe(struct platform_device *pdev)
diff --git a/sound/soc/pxa/zylonite.c b/sound/soc/pxa/zylonite.c
index db8aadf..23bf991 100644
--- a/sound/soc/pxa/zylonite.c
+++ b/sound/soc/pxa/zylonite.c
@@ -71,22 +71,10 @@ static const struct snd_soc_dapm_route audio_map[] = {
static int zylonite_wm9713_init(struct snd_soc_pcm_runtime *rtd)
{
- struct snd_soc_codec *codec = rtd->codec;
- struct snd_soc_dapm_context *dapm = &codec->dapm;
-
if (clk_pout)
snd_soc_dai_set_pll(rtd->codec_dai, 0, 0,
clk_get_rate(pout), 0);
- snd_soc_dapm_new_controls(dapm, zylonite_dapm_widgets,
- ARRAY_SIZE(zylonite_dapm_widgets));
-
- snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
-
- /* Static setup for now */
- snd_soc_dapm_enable_pin(dapm, "Headphone");
- snd_soc_dapm_enable_pin(dapm, "Headset Earpiece");
-
return 0;
}
@@ -256,6 +244,11 @@ static struct snd_soc_card zylonite = {
.resume_pre = &zylonite_resume_pre,
.dai_link = zylonite_dai,
.num_links = ARRAY_SIZE(zylonite_dai),
+
+ .dapm_widgets = zylonite_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(zylonite_dapm_widgets),
+ .dapm_routes = audio_map,
+ .num_dapm_routes = ARRAY_SIZE(audio_map),
};
static struct platform_device *zylonite_snd_ac97_device;
diff --git a/sound/soc/s6000/s6105-ipcam.c b/sound/soc/s6000/s6105-ipcam.c
index 945e8ab..0b21d1d 100644
--- a/sound/soc/s6000/s6105-ipcam.c
+++ b/sound/soc/s6000/s6105-ipcam.c
@@ -104,8 +104,8 @@ static int output_type_get(struct snd_kcontrol *kcontrol,
static int output_type_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
- struct snd_soc_codec *codec = kcontrol->private_data;
- struct snd_soc_dapm_context *dapm = &codec->dapm;
+ struct snd_soc_card *card = kcontrol->private_data;
+ struct snd_soc_dapm_context *dapm = &card->dapm;
unsigned int val = (ucontrol->value.enumerated.item[0] != 0);
char *differential = "Audio Out Differential";
char *stereo = "Audio Out Stereo";
@@ -137,13 +137,7 @@ static int s6105_aic3x_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_codec *codec = rtd->codec;
struct snd_soc_dapm_context *dapm = &codec->dapm;
-
- /* Add s6105 specific widgets */
- snd_soc_dapm_new_controls(dapm, aic3x_dapm_widgets,
- ARRAY_SIZE(aic3x_dapm_widgets));
-
- /* Set up s6105 specific audio path audio_map */
- snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
+ struct snd_soc_card *card = rtd->card;
/* not present */
snd_soc_dapm_nc_pin(dapm, "MONO_LOUT");
@@ -157,17 +151,10 @@ static int s6105_aic3x_init(struct snd_soc_pcm_runtime *rtd)
snd_soc_dapm_nc_pin(dapm, "RLOUT");
snd_soc_dapm_nc_pin(dapm, "HPRCOM");
- /* always connected */
- snd_soc_dapm_enable_pin(dapm, "Audio In");
-
/* must correspond to audio_out_mux.private_value initializer */
- snd_soc_dapm_disable_pin(dapm, "Audio Out Differential");
- snd_soc_dapm_sync(dapm);
- snd_soc_dapm_enable_pin(dapm, "Audio Out Stereo");
-
- snd_soc_dapm_sync(dapm);
+ snd_soc_dapm_disable_pin(&card->dapm, "Audio Out Differential");
- snd_ctl_add(codec->card->snd_card, snd_ctl_new1(&audio_out_mux, codec));
+ snd_ctl_add(card->snd_card, snd_ctl_new1(&audio_out_mux, card));
return 0;
}
@@ -190,6 +177,11 @@ static struct snd_soc_card snd_soc_card_s6105 = {
.owner = THIS_MODULE,
.dai_link = &s6105_dai,
.num_links = 1,
+
+ .dapm_widgets = aic3x_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(aic3x_dapm_widgets),
+ .dapm_routes = audio_map,
+ .num_dapm_routes = ARRAY_SIZE(audio_map),
};
static struct s6000_snd_platform_data s6105_snd_data __initdata = {
diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig
index 3507574..f2e2891 100644
--- a/sound/soc/samsung/Kconfig
+++ b/sound/soc/samsung/Kconfig
@@ -117,7 +117,7 @@ config SND_SOC_SAMSUNG_SIMTEC_TLV320AIC23
tristate "SoC I2S Audio support for TLV320AIC23 on Simtec boards"
depends on SND_SOC_SAMSUNG && ARCH_S3C24XX
select SND_S3C24XX_I2S
- select SND_SOC_TLV320AIC23
+ select SND_SOC_TLV320AIC23_I2C
select SND_SOC_SAMSUNG_SIMTEC
config SND_SOC_SAMSUNG_SIMTEC_HERMES
diff --git a/sound/soc/samsung/h1940_uda1380.c b/sound/soc/samsung/h1940_uda1380.c
index fbced58..88b09e0 100644
--- a/sound/soc/samsung/h1940_uda1380.c
+++ b/sound/soc/samsung/h1940_uda1380.c
@@ -66,10 +66,6 @@ static int h1940_startup(struct snd_pcm_substream *substream)
{
struct snd_pcm_runtime *runtime = substream->runtime;
- runtime->hw.rate_min = hw_rates.list[0];
- runtime->hw.rate_max = hw_rates.list[hw_rates.count - 1];
- runtime->hw.rates = SNDRV_PCM_RATE_KNOT;
-
return snd_pcm_hw_constraint_list(runtime, 0,
SNDRV_PCM_HW_PARAM_RATE,
&hw_rates);
@@ -94,7 +90,7 @@ static int h1940_hw_params(struct snd_pcm_substream *substream,
div++;
break;
default:
- dev_err(&rtd->dev, "%s: rate %d is not supported\n",
+ dev_err(rtd->dev, "%s: rate %d is not supported\n",
__func__, rate);
return -EINVAL;
}
@@ -181,7 +177,6 @@ static int h1940_uda1380_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_codec *codec = rtd->codec;
struct snd_soc_dapm_context *dapm = &codec->dapm;
- int err;
snd_soc_dapm_enable_pin(dapm, "Headphone Jack");
snd_soc_dapm_enable_pin(dapm, "Speaker");
diff --git a/sound/soc/samsung/neo1973_wm8753.c b/sound/soc/samsung/neo1973_wm8753.c
index 98a04c1..b080033 100644
--- a/sound/soc/samsung/neo1973_wm8753.c
+++ b/sound/soc/samsung/neo1973_wm8753.c
@@ -192,44 +192,6 @@ static struct snd_soc_ops neo1973_voice_ops = {
.hw_free = neo1973_voice_hw_free,
};
-/* Shared routes and controls */
-
-static const struct snd_soc_dapm_widget neo1973_wm8753_dapm_widgets[] = {
- SND_SOC_DAPM_LINE("GSM Line Out", NULL),
- SND_SOC_DAPM_LINE("GSM Line In", NULL),
- SND_SOC_DAPM_MIC("Headset Mic", NULL),
- SND_SOC_DAPM_MIC("Handset Mic", NULL),
-};
-
-static const struct snd_soc_dapm_route neo1973_wm8753_routes[] = {
- /* Connections to the GSM Module */
- {"GSM Line Out", NULL, "MONO1"},
- {"GSM Line Out", NULL, "MONO2"},
- {"RXP", NULL, "GSM Line In"},
- {"RXN", NULL, "GSM Line In"},
-
- /* Connections to Headset */
- {"MIC1", NULL, "Mic Bias"},
- {"Mic Bias", NULL, "Headset Mic"},
-
- /* Call Mic */
- {"MIC2", NULL, "Mic Bias"},
- {"MIC2N", NULL, "Mic Bias"},
- {"Mic Bias", NULL, "Handset Mic"},
-
- /* Connect the ALC pins */
- {"ACIN", NULL, "ACOP"},
-};
-
-static const struct snd_kcontrol_new neo1973_wm8753_controls[] = {
- SOC_DAPM_PIN_SWITCH("GSM Line Out"),
- SOC_DAPM_PIN_SWITCH("GSM Line In"),
- SOC_DAPM_PIN_SWITCH("Headset Mic"),
- SOC_DAPM_PIN_SWITCH("Handset Mic"),
-};
-
-/* GTA02 specific routes and controls */
-
static int gta02_speaker_enabled;
static int lm4853_set_spk(struct snd_kcontrol *kcontrol,
@@ -257,7 +219,34 @@ static int lm4853_event(struct snd_soc_dapm_widget *w,
return 0;
}
-static const struct snd_soc_dapm_route neo1973_gta02_routes[] = {
+static const struct snd_soc_dapm_widget neo1973_wm8753_dapm_widgets[] = {
+ SND_SOC_DAPM_LINE("GSM Line Out", NULL),
+ SND_SOC_DAPM_LINE("GSM Line In", NULL),
+ SND_SOC_DAPM_MIC("Headset Mic", NULL),
+ SND_SOC_DAPM_MIC("Handset Mic", NULL),
+ SND_SOC_DAPM_SPK("Handset Spk", NULL),
+ SND_SOC_DAPM_SPK("Stereo Out", lm4853_event),
+};
+
+static const struct snd_soc_dapm_route neo1973_wm8753_routes[] = {
+ /* Connections to the GSM Module */
+ {"GSM Line Out", NULL, "MONO1"},
+ {"GSM Line Out", NULL, "MONO2"},
+ {"RXP", NULL, "GSM Line In"},
+ {"RXN", NULL, "GSM Line In"},
+
+ /* Connections to Headset */
+ {"MIC1", NULL, "Mic Bias"},
+ {"Mic Bias", NULL, "Headset Mic"},
+
+ /* Call Mic */
+ {"MIC2", NULL, "Mic Bias"},
+ {"MIC2N", NULL, "Mic Bias"},
+ {"Mic Bias", NULL, "Handset Mic"},
+
+ /* Connect the ALC pins */
+ {"ACIN", NULL, "ACOP"},
+
/* Connections to the amp */
{"Stereo Out", NULL, "LOUT1"},
{"Stereo Out", NULL, "ROUT1"},
@@ -267,7 +256,11 @@ static const struct snd_soc_dapm_route neo1973_gta02_routes[] = {
{"Handset Spk", NULL, "ROUT2"},
};
-static const struct snd_kcontrol_new neo1973_gta02_wm8753_controls[] = {
+static const struct snd_kcontrol_new neo1973_wm8753_controls[] = {
+ SOC_DAPM_PIN_SWITCH("GSM Line Out"),
+ SOC_DAPM_PIN_SWITCH("GSM Line In"),
+ SOC_DAPM_PIN_SWITCH("Headset Mic"),
+ SOC_DAPM_PIN_SWITCH("Handset Mic"),
SOC_DAPM_PIN_SWITCH("Handset Spk"),
SOC_DAPM_PIN_SWITCH("Stereo Out"),
@@ -276,86 +269,32 @@ static const struct snd_kcontrol_new neo1973_gta02_wm8753_controls[] = {
lm4853_set_spk),
};
-static const struct snd_soc_dapm_widget neo1973_gta02_wm8753_dapm_widgets[] = {
- SND_SOC_DAPM_SPK("Handset Spk", NULL),
- SND_SOC_DAPM_SPK("Stereo Out", lm4853_event),
-};
-
-static int neo1973_gta02_wm8753_init(struct snd_soc_codec *codec)
-{
- struct snd_soc_dapm_context *dapm = &codec->dapm;
- int ret;
-
- ret = snd_soc_dapm_new_controls(dapm, neo1973_gta02_wm8753_dapm_widgets,
- ARRAY_SIZE(neo1973_gta02_wm8753_dapm_widgets));
- if (ret)
- return ret;
-
- ret = snd_soc_dapm_add_routes(dapm, neo1973_gta02_routes,
- ARRAY_SIZE(neo1973_gta02_routes));
- if (ret)
- return ret;
-
- ret = snd_soc_add_card_controls(codec->card, neo1973_gta02_wm8753_controls,
- ARRAY_SIZE(neo1973_gta02_wm8753_controls));
- if (ret)
- return ret;
-
- snd_soc_dapm_disable_pin(dapm, "Stereo Out");
- snd_soc_dapm_disable_pin(dapm, "Handset Spk");
- snd_soc_dapm_ignore_suspend(dapm, "Stereo Out");
- snd_soc_dapm_ignore_suspend(dapm, "Handset Spk");
-
- return 0;
-}
-
static int neo1973_wm8753_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_codec *codec = rtd->codec;
- struct snd_soc_dapm_context *dapm = &codec->dapm;
- int ret;
+ struct snd_soc_card *card = rtd->card;
/* set up NC codec pins */
- snd_soc_dapm_nc_pin(dapm, "OUT3");
- snd_soc_dapm_nc_pin(dapm, "OUT4");
- snd_soc_dapm_nc_pin(dapm, "LINE1");
- snd_soc_dapm_nc_pin(dapm, "LINE2");
-
- /* Add neo1973 specific widgets */
- ret = snd_soc_dapm_new_controls(dapm, neo1973_wm8753_dapm_widgets,
- ARRAY_SIZE(neo1973_wm8753_dapm_widgets));
- if (ret)
- return ret;
-
- /* add neo1973 specific controls */
- ret = snd_soc_add_card_controls(rtd->card, neo1973_wm8753_controls,
- ARRAY_SIZE(neo1973_wm8753_controls));
- if (ret)
- return ret;
-
- /* set up neo1973 specific audio routes */
- ret = snd_soc_dapm_add_routes(dapm, neo1973_wm8753_routes,
- ARRAY_SIZE(neo1973_wm8753_routes));
- if (ret)
- return ret;
+ snd_soc_dapm_nc_pin(&codec->dapm, "OUT3");
+ snd_soc_dapm_nc_pin(&codec->dapm, "OUT4");
+ snd_soc_dapm_nc_pin(&codec->dapm, "LINE1");
+ snd_soc_dapm_nc_pin(&codec->dapm, "LINE2");
/* set endpoints to default off mode */
- snd_soc_dapm_disable_pin(dapm, "GSM Line Out");
- snd_soc_dapm_disable_pin(dapm, "GSM Line In");
- snd_soc_dapm_disable_pin(dapm, "Headset Mic");
- snd_soc_dapm_disable_pin(dapm, "Handset Mic");
+ snd_soc_dapm_disable_pin(&card->dapm, "GSM Line Out");
+ snd_soc_dapm_disable_pin(&card->dapm, "GSM Line In");
+ snd_soc_dapm_disable_pin(&card->dapm, "Headset Mic");
+ snd_soc_dapm_disable_pin(&card->dapm, "Handset Mic");
+ snd_soc_dapm_disable_pin(&card->dapm, "Stereo Out");
+ snd_soc_dapm_disable_pin(&card->dapm, "Handset Spk");
/* allow audio paths from the GSM modem to run during suspend */
- snd_soc_dapm_ignore_suspend(dapm, "GSM Line Out");
- snd_soc_dapm_ignore_suspend(dapm, "GSM Line In");
- snd_soc_dapm_ignore_suspend(dapm, "Headset Mic");
- snd_soc_dapm_ignore_suspend(dapm, "Handset Mic");
-
- if (machine_is_neo1973_gta02()) {
- ret = neo1973_gta02_wm8753_init(codec);
- if (ret)
- return ret;
- }
+ snd_soc_dapm_ignore_suspend(&card->dapm, "GSM Line Out");
+ snd_soc_dapm_ignore_suspend(&card->dapm, "GSM Line In");
+ snd_soc_dapm_ignore_suspend(&card->dapm, "Headset Mic");
+ snd_soc_dapm_ignore_suspend(&card->dapm, "Handset Mic");
+ snd_soc_dapm_ignore_suspend(&card->dapm, "Stereo Out");
+ snd_soc_dapm_ignore_suspend(&card->dapm, "Handset Spk");
return 0;
}
@@ -409,6 +348,13 @@ static struct snd_soc_card neo1973 = {
.num_aux_devs = ARRAY_SIZE(neo1973_aux_devs),
.codec_conf = neo1973_codec_conf,
.num_configs = ARRAY_SIZE(neo1973_codec_conf),
+
+ .controls = neo1973_wm8753_controls,
+ .num_controls = ARRAY_SIZE(neo1973_wm8753_controls),
+ .dapm_widgets = neo1973_wm8753_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(neo1973_wm8753_dapm_widgets),
+ .dapm_routes = neo1973_wm8753_routes,
+ .num_dapm_routes = ARRAY_SIZE(neo1973_wm8753_routes),
};
static struct platform_device *neo1973_snd_device;
diff --git a/sound/soc/samsung/rx1950_uda1380.c b/sound/soc/samsung/rx1950_uda1380.c
index 06ebdc0..2982d9e 100644
--- a/sound/soc/samsung/rx1950_uda1380.c
+++ b/sound/soc/samsung/rx1950_uda1380.c
@@ -131,10 +131,6 @@ static int rx1950_startup(struct snd_pcm_substream *substream)
{
struct snd_pcm_runtime *runtime = substream->runtime;
- runtime->hw.rate_min = hw_rates.list[0];
- runtime->hw.rate_max = hw_rates.list[hw_rates.count - 1];
- runtime->hw.rates = SNDRV_PCM_RATE_KNOT;
-
return snd_pcm_hw_constraint_list(runtime, 0,
SNDRV_PCM_HW_PARAM_RATE,
&hw_rates);
@@ -226,7 +222,6 @@ static int rx1950_uda1380_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_codec *codec = rtd->codec;
struct snd_soc_dapm_context *dapm = &codec->dapm;
- int err;
snd_soc_dapm_enable_pin(dapm, "Headphone Jack");
snd_soc_dapm_enable_pin(dapm, "Speaker");
diff --git a/sound/soc/samsung/smdk_wm8994.c b/sound/soc/samsung/smdk_wm8994.c
index d38ae98..682eb4f 100644
--- a/sound/soc/samsung/smdk_wm8994.c
+++ b/sound/soc/samsung/smdk_wm8994.c
@@ -202,7 +202,7 @@ static int smdk_audio_probe(struct platform_device *pdev)
static struct platform_driver smdk_audio_driver = {
.driver = {
- .name = "smdk-audio-wm8894",
+ .name = "smdk-audio-wm8994",
.owner = THIS_MODULE,
.of_match_table = of_match_ptr(samsung_wm8994_of_match),
.pm = &snd_soc_pm_ops,
diff --git a/sound/soc/samsung/tobermory.c b/sound/soc/samsung/tobermory.c
index f21ff60..1807b75 100644
--- a/sound/soc/samsung/tobermory.c
+++ b/sound/soc/samsung/tobermory.c
@@ -44,6 +44,8 @@ static int tobermory_set_bias_level(struct snd_soc_card *card,
SND_SOC_CLOCK_IN);
if (ret < 0) {
pr_err("Failed to set SYSCLK: %d\n", ret);
+ snd_soc_dai_set_pll(codec_dai, WM8962_FLL,
+ 0, 0, 0);
return ret;
}
}
diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c
index 1967f44..710a079 100644
--- a/sound/soc/sh/fsi.c
+++ b/sound/soc/sh/fsi.c
@@ -1711,9 +1711,9 @@ static int fsi_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
/* set master/slave audio interface */
switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
case SND_SOC_DAIFMT_CBM_CFM:
- fsi->clk_master = 1;
break;
case SND_SOC_DAIFMT_CBS_CFS:
+ fsi->clk_master = 1; /* codec is slave, cpu is master */
break;
default:
return -EINVAL;
diff --git a/sound/soc/sh/migor.c b/sound/soc/sh/migor.c
index 5014a88..c58c252 100644
--- a/sound/soc/sh/migor.c
+++ b/sound/soc/sh/migor.c
@@ -136,19 +136,6 @@ static const struct snd_soc_dapm_route audio_map[] = {
{ "Mic Bias", NULL, "External Microphone" },
};
-static int migor_dai_init(struct snd_soc_pcm_runtime *rtd)
-{
- struct snd_soc_codec *codec = rtd->codec;
- struct snd_soc_dapm_context *dapm = &codec->dapm;
-
- snd_soc_dapm_new_controls(dapm, migor_dapm_widgets,
- ARRAY_SIZE(migor_dapm_widgets));
-
- snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
-
- return 0;
-}
-
/* migor digital audio interface glue - connects codec <--> CPU */
static struct snd_soc_dai_link migor_dai = {
.name = "wm8978",
@@ -158,7 +145,6 @@ static struct snd_soc_dai_link migor_dai = {
.platform_name = "siu-pcm-audio",
.codec_name = "wm8978.0-001a",
.ops = &migor_dai_ops,
- .init = migor_dai_init,
};
/* migor audio machine driver */
@@ -167,6 +153,11 @@ static struct snd_soc_card snd_soc_migor = {
.owner = THIS_MODULE,
.dai_link = &migor_dai,
.num_links = 1,
+
+ .dapm_widgets = migor_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(migor_dapm_widgets),
+ .dapm_routes = audio_map,
+ .num_dapm_routes = ARRAY_SIZE(audio_map),
};
static struct platform_device *migor_snd_device;
diff --git a/sound/soc/sh/rcar/Makefile b/sound/soc/sh/rcar/Makefile
index 0ff492d..7d0051c 100644
--- a/sound/soc/sh/rcar/Makefile
+++ b/sound/soc/sh/rcar/Makefile
@@ -1,2 +1,2 @@
-snd-soc-rcar-objs := core.o gen.o scu.o adg.o ssi.o
+snd-soc-rcar-objs := core.o gen.o src.o adg.o ssi.o
obj-$(CONFIG_SND_SOC_RCAR) += snd-soc-rcar.o \ No newline at end of file
diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c
index a53235c..69c4426 100644
--- a/sound/soc/sh/rcar/adg.c
+++ b/sound/soc/sh/rcar/adg.c
@@ -25,15 +25,165 @@ struct rsnd_adg {
};
#define for_each_rsnd_clk(pos, adg, i) \
- for (i = 0, (pos) = adg->clk[i]; \
- i < CLKMAX; \
- i++, (pos) = adg->clk[i])
+ for (i = 0; \
+ (i < CLKMAX) && \
+ ((pos) = adg->clk[i]); \
+ i++)
#define rsnd_priv_to_adg(priv) ((struct rsnd_adg *)(priv)->adg)
-static int rsnd_adg_set_convert_clk_gen1(struct rsnd_priv *priv,
- struct rsnd_mod *mod,
- unsigned int src_rate,
- unsigned int dst_rate)
+
+static u32 rsnd_adg_ssi_ws_timing_gen2(struct rsnd_dai_stream *io)
+{
+ struct rsnd_mod *mod = rsnd_io_to_mod_ssi(io);
+ struct rsnd_priv *priv = rsnd_mod_to_priv(mod);
+ int id = rsnd_mod_id(mod);
+ int ws = id;
+
+ if (rsnd_ssi_is_pin_sharing(rsnd_ssi_mod_get(priv, id))) {
+ switch (id) {
+ case 1:
+ case 2:
+ ws = 0;
+ break;
+ case 4:
+ ws = 3;
+ break;
+ case 8:
+ ws = 7;
+ break;
+ }
+ }
+
+ return (0x6 + ws) << 8;
+}
+
+static int rsnd_adg_set_src_timsel_gen2(struct rsnd_dai *rdai,
+ struct rsnd_mod *mod,
+ struct rsnd_dai_stream *io,
+ u32 timsel)
+{
+ int is_play = rsnd_dai_is_play(rdai, io);
+ int id = rsnd_mod_id(mod);
+ int shift = (id % 2) ? 16 : 0;
+ u32 mask, ws;
+ u32 in, out;
+
+ ws = rsnd_adg_ssi_ws_timing_gen2(io);
+
+ in = (is_play) ? timsel : ws;
+ out = (is_play) ? ws : timsel;
+
+ in = in << shift;
+ out = out << shift;
+ mask = 0xffff << shift;
+
+ switch (id / 2) {
+ case 0:
+ rsnd_mod_bset(mod, SRCIN_TIMSEL0, mask, in);
+ rsnd_mod_bset(mod, SRCOUT_TIMSEL0, mask, out);
+ break;
+ case 1:
+ rsnd_mod_bset(mod, SRCIN_TIMSEL1, mask, in);
+ rsnd_mod_bset(mod, SRCOUT_TIMSEL1, mask, out);
+ break;
+ case 2:
+ rsnd_mod_bset(mod, SRCIN_TIMSEL2, mask, in);
+ rsnd_mod_bset(mod, SRCOUT_TIMSEL2, mask, out);
+ break;
+ case 3:
+ rsnd_mod_bset(mod, SRCIN_TIMSEL3, mask, in);
+ rsnd_mod_bset(mod, SRCOUT_TIMSEL3, mask, out);
+ break;
+ case 4:
+ rsnd_mod_bset(mod, SRCIN_TIMSEL4, mask, in);
+ rsnd_mod_bset(mod, SRCOUT_TIMSEL4, mask, out);
+ break;
+ }
+
+ return 0;
+}
+
+int rsnd_adg_set_convert_clk_gen2(struct rsnd_mod *mod,
+ struct rsnd_dai *rdai,
+ struct rsnd_dai_stream *io,
+ unsigned int src_rate,
+ unsigned int dst_rate)
+{
+ struct rsnd_priv *priv = rsnd_mod_to_priv(mod);
+ struct rsnd_adg *adg = rsnd_priv_to_adg(priv);
+ struct device *dev = rsnd_priv_to_dev(priv);
+ int idx, sel, div, step, ret;
+ u32 val, en;
+ unsigned int min, diff;
+ unsigned int sel_rate [] = {
+ clk_get_rate(adg->clk[CLKA]), /* 0000: CLKA */
+ clk_get_rate(adg->clk[CLKB]), /* 0001: CLKB */
+ clk_get_rate(adg->clk[CLKC]), /* 0010: CLKC */
+ adg->rbga_rate_for_441khz_div_6,/* 0011: RBGA */
+ adg->rbgb_rate_for_48khz_div_6, /* 0100: RBGB */
+ };
+
+ min = ~0;
+ val = 0;
+ en = 0;
+ for (sel = 0; sel < ARRAY_SIZE(sel_rate); sel++) {
+ idx = 0;
+ step = 2;
+
+ if (!sel_rate[sel])
+ continue;
+
+ for (div = 2; div <= 98304; div += step) {
+ diff = abs(src_rate - sel_rate[sel] / div);
+ if (min > diff) {
+ val = (sel << 8) | idx;
+ min = diff;
+ en = 1 << (sel + 1); /* fixme */
+ }
+
+ /*
+ * step of 0_0000 / 0_0001 / 0_1101
+ * are out of order
+ */
+ if ((idx > 2) && (idx % 2))
+ step *= 2;
+ if (idx == 0x1c) {
+ div += step;
+ step *= 2;
+ }
+ idx++;
+ }
+ }
+
+ if (min == ~0) {
+ dev_err(dev, "no Input clock\n");
+ return -EIO;
+ }
+
+ ret = rsnd_adg_set_src_timsel_gen2(rdai, mod, io, val);
+ if (ret < 0) {
+ dev_err(dev, "timsel error\n");
+ return ret;
+ }
+
+ rsnd_mod_bset(mod, DIV_EN, en, en);
+
+ return 0;
+}
+
+int rsnd_adg_set_convert_timing_gen2(struct rsnd_mod *mod,
+ struct rsnd_dai *rdai,
+ struct rsnd_dai_stream *io)
+{
+ u32 val = rsnd_adg_ssi_ws_timing_gen2(io);
+
+ return rsnd_adg_set_src_timsel_gen2(rdai, mod, io, val);
+}
+
+int rsnd_adg_set_convert_clk_gen1(struct rsnd_priv *priv,
+ struct rsnd_mod *mod,
+ unsigned int src_rate,
+ unsigned int dst_rate)
{
struct rsnd_adg *adg = rsnd_priv_to_adg(priv);
struct device *dev = rsnd_priv_to_dev(priv);
@@ -91,18 +241,6 @@ find_rate:
return 0;
}
-int rsnd_adg_set_convert_clk(struct rsnd_priv *priv,
- struct rsnd_mod *mod,
- unsigned int src_rate,
- unsigned int dst_rate)
-{
- if (rsnd_is_gen1(priv))
- return rsnd_adg_set_convert_clk_gen1(priv, mod,
- src_rate, dst_rate);
-
- return -EINVAL;
-}
-
static void rsnd_adg_set_ssi_clk(struct rsnd_mod *mod, u32 val)
{
int id = rsnd_mod_id(mod);
@@ -254,13 +392,14 @@ static void rsnd_adg_ssi_clk_init(struct rsnd_priv *priv, struct rsnd_adg *adg)
}
int rsnd_adg_probe(struct platform_device *pdev,
- struct rcar_snd_info *info,
+ const struct rsnd_of_data *of_data,
struct rsnd_priv *priv)
{
struct rsnd_adg *adg;
struct device *dev = rsnd_priv_to_dev(priv);
- struct clk *clk;
+ struct clk *clk, *clk_orig;
int i;
+ bool use_old_style = false;
adg = devm_kzalloc(dev, sizeof(*adg), GFP_KERNEL);
if (!adg) {
@@ -268,10 +407,39 @@ int rsnd_adg_probe(struct platform_device *pdev,
return -ENOMEM;
}
- adg->clk[CLKA] = clk_get(NULL, "audio_clk_a");
- adg->clk[CLKB] = clk_get(NULL, "audio_clk_b");
- adg->clk[CLKC] = clk_get(NULL, "audio_clk_c");
- adg->clk[CLKI] = clk_get(NULL, "audio_clk_internal");
+ clk_orig = devm_clk_get(dev, NULL);
+ adg->clk[CLKA] = devm_clk_get(dev, "clk_a");
+ adg->clk[CLKB] = devm_clk_get(dev, "clk_b");
+ adg->clk[CLKC] = devm_clk_get(dev, "clk_c");
+ adg->clk[CLKI] = devm_clk_get(dev, "clk_i");
+
+ /*
+ * It request device dependent audio clock.
+ * But above all clks will indicate rsnd module clock
+ * if platform doesn't it
+ */
+ for_each_rsnd_clk(clk, adg, i) {
+ if (clk_orig == clk) {
+ dev_warn(dev,
+ "doesn't have device dependent clock, use independent clock\n");
+ use_old_style = true;
+ break;
+ }
+ }
+
+ /*
+ * note:
+ * these exist in order to keep compatible with
+ * platform which has device independent audio clock,
+ * but will be removed soon
+ */
+ if (use_old_style) {
+ adg->clk[CLKA] = devm_clk_get(NULL, "audio_clk_a");
+ adg->clk[CLKB] = devm_clk_get(NULL, "audio_clk_b");
+ adg->clk[CLKC] = devm_clk_get(NULL, "audio_clk_c");
+ adg->clk[CLKI] = devm_clk_get(NULL, "audio_clk_internal");
+ }
+
for_each_rsnd_clk(clk, adg, i) {
if (IS_ERR(clk)) {
dev_err(dev, "Audio clock failed\n");
@@ -287,14 +455,3 @@ int rsnd_adg_probe(struct platform_device *pdev,
return 0;
}
-
-void rsnd_adg_remove(struct platform_device *pdev,
- struct rsnd_priv *priv)
-{
- struct rsnd_adg *adg = priv->adg;
- struct clk *clk;
- int i;
-
- for_each_rsnd_clk(clk, adg, i)
- clk_put(clk);
-}
diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c
index 743de5e..215b668 100644
--- a/sound/soc/sh/rcar/core.c
+++ b/sound/soc/sh/rcar/core.c
@@ -73,13 +73,13 @@
* | +- ssi[2]
* | ...
* |
- * | ** these control scu
+ * | ** these control src
* |
- * +- scu
+ * +- src
* |
- * +- scu[0]
- * +- scu[1]
- * +- scu[2]
+ * +- src[0]
+ * +- src[1]
+ * +- src[2]
* ...
*
*
@@ -100,6 +100,21 @@
#define RSND_RATES SNDRV_PCM_RATE_8000_96000
#define RSND_FMTS (SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE)
+static struct rsnd_of_data rsnd_of_data_gen1 = {
+ .flags = RSND_GEN1,
+};
+
+static struct rsnd_of_data rsnd_of_data_gen2 = {
+ .flags = RSND_GEN2,
+};
+
+static struct of_device_id rsnd_of_match[] = {
+ { .compatible = "renesas,rcar_sound-gen1", .data = &rsnd_of_data_gen1 },
+ { .compatible = "renesas,rcar_sound-gen2", .data = &rsnd_of_data_gen2 },
+ {},
+};
+MODULE_DEVICE_TABLE(of, rsnd_of_match);
+
/*
* rsnd_platform functions
*/
@@ -107,6 +122,11 @@
(!(priv->info->func) ? 0 : \
priv->info->func(param))
+#define rsnd_is_enable_path(io, name) \
+ ((io)->info ? (io)->info->name : NULL)
+#define rsnd_info_id(priv, io, name) \
+ ((io)->info->name - priv->info->name##_info)
+
/*
* rsnd_mod functions
*/
@@ -121,17 +141,19 @@ char *rsnd_mod_name(struct rsnd_mod *mod)
void rsnd_mod_init(struct rsnd_priv *priv,
struct rsnd_mod *mod,
struct rsnd_mod_ops *ops,
+ enum rsnd_mod_type type,
int id)
{
mod->priv = priv;
mod->id = id;
mod->ops = ops;
- INIT_LIST_HEAD(&mod->list);
+ mod->type = type;
}
/*
* rsnd_dma functions
*/
+static void __rsnd_dma_start(struct rsnd_dma *dma);
static void rsnd_dma_continue(struct rsnd_dma *dma)
{
/* push next A or B plane */
@@ -142,8 +164,9 @@ static void rsnd_dma_continue(struct rsnd_dma *dma)
void rsnd_dma_start(struct rsnd_dma *dma)
{
/* push both A and B plane*/
+ dma->offset = 0;
dma->submit_loop = 2;
- schedule_work(&dma->work);
+ __rsnd_dma_start(dma);
}
void rsnd_dma_stop(struct rsnd_dma *dma)
@@ -156,12 +179,26 @@ void rsnd_dma_stop(struct rsnd_dma *dma)
static void rsnd_dma_complete(void *data)
{
struct rsnd_dma *dma = (struct rsnd_dma *)data;
- struct rsnd_priv *priv = dma->priv;
+ struct rsnd_mod *mod = rsnd_dma_to_mod(dma);
+ struct rsnd_priv *priv = rsnd_mod_to_priv(rsnd_dma_to_mod(dma));
+ struct rsnd_dai_stream *io = rsnd_mod_to_io(mod);
unsigned long flags;
rsnd_lock(priv, flags);
- dma->complete(dma);
+ /*
+ * Renesas sound Gen1 needs 1 DMAC,
+ * Gen2 needs 2 DMAC.
+ * In Gen2 case, it are Audio-DMAC, and Audio-DMAC-peri-peri.
+ * But, Audio-DMAC-peri-peri doesn't have interrupt,
+ * and this driver is assuming that here.
+ *
+ * If Audio-DMAC-peri-peri has interrpt,
+ * rsnd_dai_pointer_update() will be called twice,
+ * ant it will breaks io->byte_pos
+ */
+
+ rsnd_dai_pointer_update(io, io->byte_per_period);
if (dma->submit_loop)
rsnd_dma_continue(dma);
@@ -169,20 +206,23 @@ static void rsnd_dma_complete(void *data)
rsnd_unlock(priv, flags);
}
-static void rsnd_dma_do_work(struct work_struct *work)
+static void __rsnd_dma_start(struct rsnd_dma *dma)
{
- struct rsnd_dma *dma = container_of(work, struct rsnd_dma, work);
- struct rsnd_priv *priv = dma->priv;
+ struct rsnd_mod *mod = rsnd_dma_to_mod(dma);
+ struct rsnd_priv *priv = rsnd_mod_to_priv(mod);
+ struct rsnd_dai_stream *io = rsnd_mod_to_io(mod);
+ struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io);
struct device *dev = rsnd_priv_to_dev(priv);
struct dma_async_tx_descriptor *desc;
dma_addr_t buf;
- size_t len;
+ size_t len = io->byte_per_period;
int i;
for (i = 0; i < dma->submit_loop; i++) {
- if (dma->inquiry(dma, &buf, &len) < 0)
- return;
+ buf = runtime->dma_addr +
+ rsnd_dai_pointer_offset(io, dma->offset + len);
+ dma->offset = len;
desc = dmaengine_prep_slave_single(
dma->chan, buf, len, dma->dir,
@@ -204,16 +244,20 @@ static void rsnd_dma_do_work(struct work_struct *work)
}
}
+static void rsnd_dma_do_work(struct work_struct *work)
+{
+ struct rsnd_dma *dma = container_of(work, struct rsnd_dma, work);
+
+ __rsnd_dma_start(dma);
+}
+
int rsnd_dma_available(struct rsnd_dma *dma)
{
return !!dma->chan;
}
int rsnd_dma_init(struct rsnd_priv *priv, struct rsnd_dma *dma,
- int is_play, int id,
- int (*inquiry)(struct rsnd_dma *dma,
- dma_addr_t *buf, int *len),
- int (*complete)(struct rsnd_dma *dma))
+ int is_play, int id)
{
struct device *dev = rsnd_priv_to_dev(priv);
struct dma_slave_config cfg;
@@ -246,9 +290,6 @@ int rsnd_dma_init(struct rsnd_priv *priv, struct rsnd_dma *dma,
goto rsnd_dma_init_err;
dma->dir = is_play ? DMA_TO_DEVICE : DMA_FROM_DEVICE;
- dma->priv = priv;
- dma->inquiry = inquiry;
- dma->complete = complete;
INIT_WORK(&dma->work, rsnd_dma_do_work);
return 0;
@@ -271,26 +312,42 @@ void rsnd_dma_quit(struct rsnd_priv *priv,
/*
* rsnd_dai functions
*/
-#define rsnd_dai_call(rdai, io, fn) \
-({ \
- struct rsnd_mod *mod, *n; \
- int ret = 0; \
- for_each_rsnd_mod(mod, n, io) { \
- ret = rsnd_mod_call(mod, fn, rdai, io); \
- if (ret < 0) \
- break; \
- } \
- ret; \
+#define __rsnd_mod_call(mod, func, rdai, io) \
+({ \
+ struct rsnd_priv *priv = rsnd_mod_to_priv(mod); \
+ struct device *dev = rsnd_priv_to_dev(priv); \
+ dev_dbg(dev, "%s [%d] %s\n", \
+ rsnd_mod_name(mod), rsnd_mod_id(mod), #func); \
+ (mod)->ops->func(mod, rdai, io); \
+})
+
+#define rsnd_mod_call(mod, func, rdai, io) \
+ (!(mod) ? -ENODEV : \
+ !((mod)->ops->func) ? 0 : \
+ __rsnd_mod_call(mod, func, (rdai), (io)))
+
+#define rsnd_dai_call(rdai, io, fn) \
+({ \
+ struct rsnd_mod *mod; \
+ int ret = 0, i; \
+ for (i = 0; i < RSND_MOD_MAX; i++) { \
+ mod = (io)->mod[i]; \
+ if (!mod) \
+ continue; \
+ ret = rsnd_mod_call(mod, fn, (rdai), (io)); \
+ if (ret < 0) \
+ break; \
+ } \
+ ret; \
})
-int rsnd_dai_connect(struct rsnd_dai *rdai,
- struct rsnd_mod *mod,
- struct rsnd_dai_stream *io)
+static int rsnd_dai_connect(struct rsnd_mod *mod,
+ struct rsnd_dai_stream *io)
{
if (!mod)
return -EIO;
- if (!list_empty(&mod->list)) {
+ if (io->mod[mod->type]) {
struct rsnd_priv *priv = rsnd_mod_to_priv(mod);
struct device *dev = rsnd_priv_to_dev(priv);
@@ -300,14 +357,8 @@ int rsnd_dai_connect(struct rsnd_dai *rdai,
return -EIO;
}
- list_add_tail(&mod->list, &io->head);
-
- return 0;
-}
-
-int rsnd_dai_disconnect(struct rsnd_mod *mod)
-{
- list_del_init(&mod->list);
+ io->mod[mod->type] = mod;
+ mod->io = io;
return 0;
}
@@ -316,7 +367,7 @@ int rsnd_dai_id(struct rsnd_priv *priv, struct rsnd_dai *rdai)
{
int id = rdai - priv->rdai;
- if ((id < 0) || (id >= rsnd_dai_nr(priv)))
+ if ((id < 0) || (id >= rsnd_rdai_nr(priv)))
return -EINVAL;
return id;
@@ -324,7 +375,7 @@ int rsnd_dai_id(struct rsnd_priv *priv, struct rsnd_dai *rdai)
struct rsnd_dai *rsnd_dai_get(struct rsnd_priv *priv, int id)
{
- if ((id < 0) || (id >= rsnd_dai_nr(priv)))
+ if ((id < 0) || (id >= rsnd_rdai_nr(priv)))
return NULL;
return priv->rdai + id;
@@ -382,10 +433,6 @@ static int rsnd_dai_stream_init(struct rsnd_dai_stream *io,
{
struct snd_pcm_runtime *runtime = substream->runtime;
- if (!list_empty(&io->head))
- return -EIO;
-
- INIT_LIST_HEAD(&io->head);
io->substream = substream;
io->byte_pos = 0;
io->period_pos = 0;
@@ -440,10 +487,6 @@ static int rsnd_soc_dai_trigger(struct snd_pcm_substream *substream, int cmd,
if (ret < 0)
goto dai_trigger_end;
- ret = rsnd_gen_path_init(priv, rdai, io);
- if (ret < 0)
- goto dai_trigger_end;
-
ret = rsnd_dai_call(rdai, io, init);
if (ret < 0)
goto dai_trigger_end;
@@ -461,10 +504,6 @@ static int rsnd_soc_dai_trigger(struct snd_pcm_substream *substream, int cmd,
if (ret < 0)
goto dai_trigger_end;
- ret = rsnd_gen_path_exit(priv, rdai, io);
- if (ret < 0)
- goto dai_trigger_end;
-
ret = rsnd_platform_call(priv, dai, stop, ssi_id);
if (ret < 0)
goto dai_trigger_end;
@@ -486,10 +525,10 @@ static int rsnd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
/* set master/slave audio interface */
switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
case SND_SOC_DAIFMT_CBM_CFM:
- rdai->clk_master = 1;
+ rdai->clk_master = 0;
break;
case SND_SOC_DAIFMT_CBS_CFS:
- rdai->clk_master = 0;
+ rdai->clk_master = 1; /* codec is slave, cpu is master */
break;
default:
return -EINVAL;
@@ -540,24 +579,174 @@ static const struct snd_soc_dai_ops rsnd_soc_dai_ops = {
.set_fmt = rsnd_soc_dai_set_fmt,
};
+static int rsnd_path_init(struct rsnd_priv *priv,
+ struct rsnd_dai *rdai,
+ struct rsnd_dai_stream *io)
+{
+ struct rsnd_mod *mod;
+ struct rsnd_dai_platform_info *dai_info = rdai->info;
+ int ret;
+ int ssi_id = -1;
+ int src_id = -1;
+
+ /*
+ * Gen1 is created by SRU/SSI, and this SRU is base module of
+ * Gen2's SCU/SSIU/SSI. (Gen2 SCU/SSIU came from SRU)
+ *
+ * Easy image is..
+ * Gen1 SRU = Gen2 SCU + SSIU + etc
+ *
+ * Gen2 SCU path is very flexible, but, Gen1 SRU (SCU parts) is
+ * using fixed path.
+ */
+ if (dai_info) {
+ if (rsnd_is_enable_path(io, ssi))
+ ssi_id = rsnd_info_id(priv, io, ssi);
+ if (rsnd_is_enable_path(io, src))
+ src_id = rsnd_info_id(priv, io, src);
+ } else {
+ /* get SSI's ID */
+ mod = rsnd_ssi_mod_get_frm_dai(priv,
+ rsnd_dai_id(priv, rdai),
+ rsnd_dai_is_play(rdai, io));
+ if (!mod)
+ return 0;
+ ssi_id = src_id = rsnd_mod_id(mod);
+ }
+
+ ret = 0;
+
+ /* SRC */
+ if (src_id >= 0) {
+ mod = rsnd_src_mod_get(priv, src_id);
+ ret = rsnd_dai_connect(mod, io);
+ if (ret < 0)
+ return ret;
+ }
+
+ /* SSI */
+ if (ssi_id >= 0) {
+ mod = rsnd_ssi_mod_get(priv, ssi_id);
+ ret = rsnd_dai_connect(mod, io);
+ if (ret < 0)
+ return ret;
+ }
+
+ return ret;
+}
+
+static void rsnd_of_parse_dai(struct platform_device *pdev,
+ const struct rsnd_of_data *of_data,
+ struct rsnd_priv *priv)
+{
+ struct device_node *dai_node, *dai_np;
+ struct device_node *ssi_node, *ssi_np;
+ struct device_node *src_node, *src_np;
+ struct device_node *playback, *capture;
+ struct rsnd_dai_platform_info *dai_info;
+ struct rcar_snd_info *info = rsnd_priv_to_info(priv);
+ struct device *dev = &pdev->dev;
+ int nr, i;
+ int dai_i, ssi_i, src_i;
+
+ if (!of_data)
+ return;
+
+ dai_node = of_get_child_by_name(dev->of_node, "rcar_sound,dai");
+ if (!dai_node)
+ return;
+
+ nr = of_get_child_count(dai_node);
+ if (!nr)
+ return;
+
+ dai_info = devm_kzalloc(dev,
+ sizeof(struct rsnd_dai_platform_info) * nr,
+ GFP_KERNEL);
+ if (!dai_info) {
+ dev_err(dev, "dai info allocation error\n");
+ return;
+ }
+
+ info->dai_info_nr = nr;
+ info->dai_info = dai_info;
+
+ ssi_node = of_get_child_by_name(dev->of_node, "rcar_sound,ssi");
+ src_node = of_get_child_by_name(dev->of_node, "rcar_sound,src");
+
+#define mod_parse(name) \
+if (name##_node) { \
+ struct rsnd_##name##_platform_info *name##_info; \
+ \
+ name##_i = 0; \
+ for_each_child_of_node(name##_node, name##_np) { \
+ name##_info = info->name##_info + name##_i; \
+ \
+ if (name##_np == playback) \
+ dai_info->playback.name = name##_info; \
+ if (name##_np == capture) \
+ dai_info->capture.name = name##_info; \
+ \
+ name##_i++; \
+ } \
+}
+
+ /*
+ * parse all dai
+ */
+ dai_i = 0;
+ for_each_child_of_node(dai_node, dai_np) {
+ dai_info = info->dai_info + dai_i;
+
+ for (i = 0;; i++) {
+
+ playback = of_parse_phandle(dai_np, "playback", i);
+ capture = of_parse_phandle(dai_np, "capture", i);
+
+ if (!playback && !capture)
+ break;
+
+ mod_parse(ssi);
+ mod_parse(src);
+
+ if (playback)
+ of_node_put(playback);
+ if (capture)
+ of_node_put(capture);
+ }
+
+ dai_i++;
+ }
+}
+
static int rsnd_dai_probe(struct platform_device *pdev,
- struct rcar_snd_info *info,
+ const struct rsnd_of_data *of_data,
struct rsnd_priv *priv)
{
struct snd_soc_dai_driver *drv;
+ struct rcar_snd_info *info = rsnd_priv_to_info(priv);
struct rsnd_dai *rdai;
struct rsnd_mod *pmod, *cmod;
struct device *dev = rsnd_priv_to_dev(priv);
int dai_nr;
int i;
- /* get max dai nr */
- for (dai_nr = 0; dai_nr < 32; dai_nr++) {
- pmod = rsnd_ssi_mod_get_frm_dai(priv, dai_nr, 1);
- cmod = rsnd_ssi_mod_get_frm_dai(priv, dai_nr, 0);
+ rsnd_of_parse_dai(pdev, of_data, priv);
- if (!pmod && !cmod)
- break;
+ /*
+ * dai_nr should be set via dai_info_nr,
+ * but allow it to keeping compatible
+ */
+ dai_nr = info->dai_info_nr;
+ if (!dai_nr) {
+ /* get max dai nr */
+ for (dai_nr = 0; dai_nr < 32; dai_nr++) {
+ pmod = rsnd_ssi_mod_get_frm_dai(priv, dai_nr, 1);
+ cmod = rsnd_ssi_mod_get_frm_dai(priv, dai_nr, 0);
+
+ if (!pmod && !cmod)
+ break;
+ }
}
if (!dai_nr) {
@@ -572,7 +761,13 @@ static int rsnd_dai_probe(struct platform_device *pdev,
return -ENOMEM;
}
+ priv->rdai_nr = dai_nr;
+ priv->daidrv = drv;
+ priv->rdai = rdai;
+
for (i = 0; i < dai_nr; i++) {
+ if (info->dai_info)
+ rdai[i].info = &info->dai_info[i];
pmod = rsnd_ssi_mod_get_frm_dai(priv, i, 1);
cmod = rsnd_ssi_mod_get_frm_dai(priv, i, 0);
@@ -580,9 +775,6 @@ static int rsnd_dai_probe(struct platform_device *pdev,
/*
* init rsnd_dai
*/
- INIT_LIST_HEAD(&rdai[i].playback.head);
- INIT_LIST_HEAD(&rdai[i].capture.head);
-
snprintf(rdai[i].name, RSND_DAI_NAME_SIZE, "rsnd-dai.%d", i);
/*
@@ -595,12 +787,20 @@ static int rsnd_dai_probe(struct platform_device *pdev,
drv[i].playback.formats = RSND_FMTS;
drv[i].playback.channels_min = 2;
drv[i].playback.channels_max = 2;
+
+ if (info->dai_info)
+ rdai[i].playback.info = &info->dai_info[i].playback;
+ rsnd_path_init(priv, &rdai[i], &rdai[i].playback);
}
if (cmod) {
drv[i].capture.rates = RSND_RATES;
drv[i].capture.formats = RSND_FMTS;
drv[i].capture.channels_min = 2;
drv[i].capture.channels_max = 2;
+
+ if (info->dai_info)
+ rdai[i].capture.info = &info->dai_info[i].capture;
+ rsnd_path_init(priv, &rdai[i], &rdai[i].capture);
}
dev_dbg(dev, "%s (%s/%s)\n", rdai[i].name,
@@ -608,18 +808,9 @@ static int rsnd_dai_probe(struct platform_device *pdev,
cmod ? "capture" : " -- ");
}
- priv->dai_nr = dai_nr;
- priv->daidrv = drv;
- priv->rdai = rdai;
-
return 0;
}
-static void rsnd_dai_remove(struct platform_device *pdev,
- struct rsnd_priv *priv)
-{
-}
-
/*
* pcm ops
*/
@@ -713,9 +904,30 @@ static int rsnd_probe(struct platform_device *pdev)
struct rcar_snd_info *info;
struct rsnd_priv *priv;
struct device *dev = &pdev->dev;
- int ret;
+ struct rsnd_dai *rdai;
+ const struct of_device_id *of_id = of_match_device(rsnd_of_match, dev);
+ const struct rsnd_of_data *of_data;
+ int (*probe_func[])(struct platform_device *pdev,
+ const struct rsnd_of_data *of_data,
+ struct rsnd_priv *priv) = {
+ rsnd_gen_probe,
+ rsnd_ssi_probe,
+ rsnd_src_probe,
+ rsnd_adg_probe,
+ rsnd_dai_probe,
+ };
+ int ret, i;
+
+ info = NULL;
+ of_data = NULL;
+ if (of_id) {
+ info = devm_kzalloc(&pdev->dev,
+ sizeof(struct rcar_snd_info), GFP_KERNEL);
+ of_data = of_id->data;
+ } else {
+ info = pdev->dev.platform_data;
+ }
- info = pdev->dev.platform_data;
if (!info) {
dev_err(dev, "driver needs R-Car sound information\n");
return -ENODEV;
@@ -737,25 +949,21 @@ static int rsnd_probe(struct platform_device *pdev)
/*
* init each module
*/
- ret = rsnd_gen_probe(pdev, info, priv);
- if (ret < 0)
- return ret;
-
- ret = rsnd_scu_probe(pdev, info, priv);
- if (ret < 0)
- return ret;
+ for (i = 0; i < ARRAY_SIZE(probe_func); i++) {
+ ret = probe_func[i](pdev, of_data, priv);
+ if (ret)
+ return ret;
+ }
- ret = rsnd_adg_probe(pdev, info, priv);
- if (ret < 0)
- return ret;
+ for_each_rsnd_dai(rdai, priv, i) {
+ ret = rsnd_dai_call(rdai, &rdai->playback, probe);
+ if (ret)
+ return ret;
- ret = rsnd_ssi_probe(pdev, info, priv);
- if (ret < 0)
- return ret;
-
- ret = rsnd_dai_probe(pdev, info, priv);
- if (ret < 0)
- return ret;
+ ret = rsnd_dai_call(rdai, &rdai->capture, probe);
+ if (ret)
+ return ret;
+ }
/*
* asoc register
@@ -767,7 +975,7 @@ static int rsnd_probe(struct platform_device *pdev)
}
ret = snd_soc_register_component(dev, &rsnd_soc_component,
- priv->daidrv, rsnd_dai_nr(priv));
+ priv->daidrv, rsnd_rdai_nr(priv));
if (ret < 0) {
dev_err(dev, "cannot snd dai register\n");
goto exit_snd_soc;
@@ -789,17 +997,20 @@ exit_snd_soc:
static int rsnd_remove(struct platform_device *pdev)
{
struct rsnd_priv *priv = dev_get_drvdata(&pdev->dev);
+ struct rsnd_dai *rdai;
+ int ret, i;
pm_runtime_disable(&pdev->dev);
- /*
- * remove each module
- */
- rsnd_ssi_remove(pdev, priv);
- rsnd_adg_remove(pdev, priv);
- rsnd_scu_remove(pdev, priv);
- rsnd_dai_remove(pdev, priv);
- rsnd_gen_remove(pdev, priv);
+ for_each_rsnd_dai(rdai, priv, i) {
+ ret = rsnd_dai_call(rdai, &rdai->playback, remove);
+ if (ret)
+ return ret;
+
+ ret = rsnd_dai_call(rdai, &rdai->capture, remove);
+ if (ret)
+ return ret;
+ }
return 0;
}
@@ -807,6 +1018,7 @@ static int rsnd_remove(struct platform_device *pdev)
static struct platform_driver rsnd_driver = {
.driver = {
.name = "rcar_sound",
+ .of_match_table = rsnd_of_match,
},
.probe = rsnd_probe,
.remove = rsnd_remove,
diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c
index add088b..50a1ef3 100644
--- a/sound/soc/sh/rcar/gen.c
+++ b/sound/soc/sh/rcar/gen.c
@@ -155,62 +155,6 @@ static int rsnd_gen_regmap_init(struct rsnd_priv *priv,
return 0;
}
-int rsnd_gen_path_init(struct rsnd_priv *priv,
- struct rsnd_dai *rdai,
- struct rsnd_dai_stream *io)
-{
- struct rsnd_mod *mod;
- int ret;
- int id;
-
- /*
- * Gen1 is created by SRU/SSI, and this SRU is base module of
- * Gen2's SCU/SSIU/SSI. (Gen2 SCU/SSIU came from SRU)
- *
- * Easy image is..
- * Gen1 SRU = Gen2 SCU + SSIU + etc
- *
- * Gen2 SCU path is very flexible, but, Gen1 SRU (SCU parts) is
- * using fixed path.
- *
- * Then, SSI id = SCU id here
- */
-
- /* get SSI's ID */
- mod = rsnd_ssi_mod_get_frm_dai(priv,
- rsnd_dai_id(priv, rdai),
- rsnd_dai_is_play(rdai, io));
- id = rsnd_mod_id(mod);
-
- /* SSI */
- mod = rsnd_ssi_mod_get(priv, id);
- ret = rsnd_dai_connect(rdai, mod, io);
- if (ret < 0)
- return ret;
-
- /* SCU */
- mod = rsnd_scu_mod_get(priv, id);
- ret = rsnd_dai_connect(rdai, mod, io);
-
- return ret;
-}
-
-int rsnd_gen_path_exit(struct rsnd_priv *priv,
- struct rsnd_dai *rdai,
- struct rsnd_dai_stream *io)
-{
- struct rsnd_mod *mod, *n;
- int ret = 0;
-
- /*
- * remove all mod from rdai
- */
- for_each_rsnd_mod(mod, n, io)
- ret |= rsnd_dai_disconnect(mod);
-
- return ret;
-}
-
/*
* Gen2
*/
@@ -229,14 +173,40 @@ static int rsnd_gen2_regmap_init(struct rsnd_priv *priv, struct rsnd_gen *gen)
RSND_GEN2_S_REG(gen, SSIU, SSI_MODE0, 0x800),
RSND_GEN2_S_REG(gen, SSIU, SSI_MODE1, 0x804),
/* FIXME: it needs SSI_MODE2/3 in the future */
+ RSND_GEN2_M_REG(gen, SSIU, SSI_BUSIF_MODE, 0x0, 0x80),
+ RSND_GEN2_M_REG(gen, SSIU, SSI_BUSIF_ADINR,0x4, 0x80),
+ RSND_GEN2_M_REG(gen, SSIU, SSI_CTRL, 0x10, 0x80),
RSND_GEN2_M_REG(gen, SSIU, INT_ENABLE, 0x18, 0x80),
+ RSND_GEN2_M_REG(gen, SCU, SRC_BUSIF_MODE, 0x0, 0x20),
+ RSND_GEN2_M_REG(gen, SCU, SRC_ROUTE_MODE0,0xc, 0x20),
+ RSND_GEN2_M_REG(gen, SCU, SRC_CTRL, 0x10, 0x20),
+ RSND_GEN2_M_REG(gen, SCU, SRC_SWRSR, 0x200, 0x40),
+ RSND_GEN2_M_REG(gen, SCU, SRC_SRCIR, 0x204, 0x40),
+ RSND_GEN2_M_REG(gen, SCU, SRC_ADINR, 0x214, 0x40),
+ RSND_GEN2_M_REG(gen, SCU, SRC_IFSCR, 0x21c, 0x40),
+ RSND_GEN2_M_REG(gen, SCU, SRC_IFSVR, 0x220, 0x40),
+ RSND_GEN2_M_REG(gen, SCU, SRC_SRCCR, 0x224, 0x40),
+ RSND_GEN2_M_REG(gen, SCU, SRC_BSDSR, 0x22c, 0x40),
+ RSND_GEN2_M_REG(gen, SCU, SRC_BSISR, 0x238, 0x40),
+
RSND_GEN2_S_REG(gen, ADG, BRRA, 0x00),
RSND_GEN2_S_REG(gen, ADG, BRRB, 0x04),
RSND_GEN2_S_REG(gen, ADG, SSICKR, 0x08),
RSND_GEN2_S_REG(gen, ADG, AUDIO_CLK_SEL0, 0x0c),
RSND_GEN2_S_REG(gen, ADG, AUDIO_CLK_SEL1, 0x10),
RSND_GEN2_S_REG(gen, ADG, AUDIO_CLK_SEL2, 0x14),
+ RSND_GEN2_S_REG(gen, ADG, DIV_EN, 0x30),
+ RSND_GEN2_S_REG(gen, ADG, SRCIN_TIMSEL0, 0x34),
+ RSND_GEN2_S_REG(gen, ADG, SRCIN_TIMSEL1, 0x38),
+ RSND_GEN2_S_REG(gen, ADG, SRCIN_TIMSEL2, 0x3c),
+ RSND_GEN2_S_REG(gen, ADG, SRCIN_TIMSEL3, 0x40),
+ RSND_GEN2_S_REG(gen, ADG, SRCIN_TIMSEL4, 0x44),
+ RSND_GEN2_S_REG(gen, ADG, SRCOUT_TIMSEL0, 0x48),
+ RSND_GEN2_S_REG(gen, ADG, SRCOUT_TIMSEL1, 0x4c),
+ RSND_GEN2_S_REG(gen, ADG, SRCOUT_TIMSEL2, 0x50),
+ RSND_GEN2_S_REG(gen, ADG, SRCOUT_TIMSEL3, 0x54),
+ RSND_GEN2_S_REG(gen, ADG, SRCOUT_TIMSEL4, 0x58),
RSND_GEN2_M_REG(gen, SSI, SSICR, 0x00, 0x40),
RSND_GEN2_M_REG(gen, SSI, SSISR, 0x04, 0x40),
@@ -249,7 +219,6 @@ static int rsnd_gen2_regmap_init(struct rsnd_priv *priv, struct rsnd_gen *gen)
}
static int rsnd_gen2_probe(struct platform_device *pdev,
- struct rcar_snd_info *info,
struct rsnd_priv *priv)
{
struct device *dev = rsnd_priv_to_dev(priv);
@@ -283,7 +252,7 @@ static int rsnd_gen2_probe(struct platform_device *pdev,
return ret;
dev_dbg(dev, "Gen2 device probed\n");
- dev_dbg(dev, "SRU : %08x => %p\n", scu_res->start,
+ dev_dbg(dev, "SCU : %08x => %p\n", scu_res->start,
gen->base[RSND_GEN2_SCU]);
dev_dbg(dev, "ADG : %08x => %p\n", adg_res->start,
gen->base[RSND_GEN2_ADG]);
@@ -317,7 +286,7 @@ static int rsnd_gen1_regmap_init(struct rsnd_priv *priv, struct rsnd_gen *gen)
RSND_GEN1_S_REG(gen, SRU, SRC_ROUTE_CTRL, 0xc0),
RSND_GEN1_S_REG(gen, SRU, SSI_MODE0, 0xD0),
RSND_GEN1_S_REG(gen, SRU, SSI_MODE1, 0xD4),
- RSND_GEN1_M_REG(gen, SRU, BUSIF_MODE, 0x20, 0x4),
+ RSND_GEN1_M_REG(gen, SRU, SRC_BUSIF_MODE, 0x20, 0x4),
RSND_GEN1_M_REG(gen, SRU, SRC_ROUTE_MODE0,0x50, 0x8),
RSND_GEN1_M_REG(gen, SRU, SRC_SWRSR, 0x200, 0x40),
RSND_GEN1_M_REG(gen, SRU, SRC_SRCIR, 0x204, 0x40),
@@ -347,7 +316,6 @@ static int rsnd_gen1_regmap_init(struct rsnd_priv *priv, struct rsnd_gen *gen)
}
static int rsnd_gen1_probe(struct platform_device *pdev,
- struct rcar_snd_info *info,
struct rsnd_priv *priv)
{
struct device *dev = rsnd_priv_to_dev(priv);
@@ -391,14 +359,28 @@ static int rsnd_gen1_probe(struct platform_device *pdev,
/*
* Gen
*/
+static void rsnd_of_parse_gen(struct platform_device *pdev,
+ const struct rsnd_of_data *of_data,
+ struct rsnd_priv *priv)
+{
+ struct rcar_snd_info *info = priv->info;
+
+ if (!of_data)
+ return;
+
+ info->flags = of_data->flags;
+}
+
int rsnd_gen_probe(struct platform_device *pdev,
- struct rcar_snd_info *info,
+ const struct rsnd_of_data *of_data,
struct rsnd_priv *priv)
{
struct device *dev = rsnd_priv_to_dev(priv);
struct rsnd_gen *gen;
int ret;
+ rsnd_of_parse_gen(pdev, of_data, priv);
+
gen = devm_kzalloc(dev, sizeof(*gen), GFP_KERNEL);
if (!gen) {
dev_err(dev, "GEN allocate failed\n");
@@ -409,17 +391,12 @@ int rsnd_gen_probe(struct platform_device *pdev,
ret = -ENODEV;
if (rsnd_is_gen1(priv))
- ret = rsnd_gen1_probe(pdev, info, priv);
+ ret = rsnd_gen1_probe(pdev, priv);
else if (rsnd_is_gen2(priv))
- ret = rsnd_gen2_probe(pdev, info, priv);
+ ret = rsnd_gen2_probe(pdev, priv);
if (ret < 0)
dev_err(dev, "unknown generation R-Car sound device\n");
return ret;
}
-
-void rsnd_gen_remove(struct platform_device *pdev,
- struct rsnd_priv *priv)
-{
-}
diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h
index 4ca66cd..619d198 100644
--- a/sound/soc/sh/rcar/rsnd.h
+++ b/sound/soc/sh/rcar/rsnd.h
@@ -17,6 +17,8 @@
#include <linux/io.h>
#include <linux/list.h>
#include <linux/module.h>
+#include <linux/of_device.h>
+#include <linux/of_irq.h>
#include <linux/sh_dma.h>
#include <linux/workqueue.h>
#include <sound/rcar_snd.h>
@@ -32,15 +34,9 @@
*/
enum rsnd_reg {
/* SRU/SCU/SSIU */
- RSND_REG_SRC_ROUTE_SEL, /* for Gen1 */
- RSND_REG_SRC_TMG_SEL0, /* for Gen1 */
- RSND_REG_SRC_TMG_SEL1, /* for Gen1 */
- RSND_REG_SRC_TMG_SEL2, /* for Gen1 */
- RSND_REG_SRC_ROUTE_CTRL, /* for Gen1 */
RSND_REG_SSI_MODE0,
RSND_REG_SSI_MODE1,
- RSND_REG_BUSIF_MODE,
- RSND_REG_INT_ENABLE, /* for Gen2 */
+ RSND_REG_SRC_BUSIF_MODE,
RSND_REG_SRC_ROUTE_MODE0,
RSND_REG_SRC_SWRSR,
RSND_REG_SRC_SRCIR,
@@ -48,7 +44,6 @@ enum rsnd_reg {
RSND_REG_SRC_IFSCR,
RSND_REG_SRC_IFSVR,
RSND_REG_SRC_SRCCR,
- RSND_REG_SRC_MNFSR,
/* ADG */
RSND_REG_BRRA,
@@ -56,10 +51,6 @@ enum rsnd_reg {
RSND_REG_SSICKR,
RSND_REG_AUDIO_CLK_SEL0,
RSND_REG_AUDIO_CLK_SEL1,
- RSND_REG_AUDIO_CLK_SEL2,
- RSND_REG_AUDIO_CLK_SEL3, /* for Gen1 */
- RSND_REG_AUDIO_CLK_SEL4, /* for Gen1 */
- RSND_REG_AUDIO_CLK_SEL5, /* for Gen1 */
/* SSI */
RSND_REG_SSICR,
@@ -68,9 +59,63 @@ enum rsnd_reg {
RSND_REG_SSIRDR,
RSND_REG_SSIWSR,
+ /* SHARE see below */
+ RSND_REG_SHARE01,
+ RSND_REG_SHARE02,
+ RSND_REG_SHARE03,
+ RSND_REG_SHARE04,
+ RSND_REG_SHARE05,
+ RSND_REG_SHARE06,
+ RSND_REG_SHARE07,
+ RSND_REG_SHARE08,
+ RSND_REG_SHARE09,
+ RSND_REG_SHARE10,
+ RSND_REG_SHARE11,
+ RSND_REG_SHARE12,
+ RSND_REG_SHARE13,
+ RSND_REG_SHARE14,
+ RSND_REG_SHARE15,
+ RSND_REG_SHARE16,
+ RSND_REG_SHARE17,
+ RSND_REG_SHARE18,
+ RSND_REG_SHARE19,
+
RSND_REG_MAX,
};
+/* Gen1 only */
+#define RSND_REG_SRC_ROUTE_SEL RSND_REG_SHARE01
+#define RSND_REG_SRC_TMG_SEL0 RSND_REG_SHARE02
+#define RSND_REG_SRC_TMG_SEL1 RSND_REG_SHARE03
+#define RSND_REG_SRC_TMG_SEL2 RSND_REG_SHARE04
+#define RSND_REG_SRC_ROUTE_CTRL RSND_REG_SHARE05
+#define RSND_REG_SRC_MNFSR RSND_REG_SHARE06
+#define RSND_REG_AUDIO_CLK_SEL3 RSND_REG_SHARE07
+#define RSND_REG_AUDIO_CLK_SEL4 RSND_REG_SHARE08
+#define RSND_REG_AUDIO_CLK_SEL5 RSND_REG_SHARE09
+
+/* Gen2 only */
+#define RSND_REG_SRC_CTRL RSND_REG_SHARE01
+#define RSND_REG_SSI_CTRL RSND_REG_SHARE02
+#define RSND_REG_SSI_BUSIF_MODE RSND_REG_SHARE03
+#define RSND_REG_SSI_BUSIF_ADINR RSND_REG_SHARE04
+#define RSND_REG_INT_ENABLE RSND_REG_SHARE05
+#define RSND_REG_SRC_BSDSR RSND_REG_SHARE06
+#define RSND_REG_SRC_BSISR RSND_REG_SHARE07
+#define RSND_REG_DIV_EN RSND_REG_SHARE08
+#define RSND_REG_SRCIN_TIMSEL0 RSND_REG_SHARE09
+#define RSND_REG_SRCIN_TIMSEL1 RSND_REG_SHARE10
+#define RSND_REG_SRCIN_TIMSEL2 RSND_REG_SHARE11
+#define RSND_REG_SRCIN_TIMSEL3 RSND_REG_SHARE12
+#define RSND_REG_SRCIN_TIMSEL4 RSND_REG_SHARE13
+#define RSND_REG_SRCOUT_TIMSEL0 RSND_REG_SHARE14
+#define RSND_REG_SRCOUT_TIMSEL1 RSND_REG_SHARE15
+#define RSND_REG_SRCOUT_TIMSEL2 RSND_REG_SHARE16
+#define RSND_REG_SRCOUT_TIMSEL3 RSND_REG_SHARE17
+#define RSND_REG_SRCOUT_TIMSEL4 RSND_REG_SHARE18
+#define RSND_REG_AUDIO_CLK_SEL2 RSND_REG_SHARE19
+
+struct rsnd_of_data;
struct rsnd_priv;
struct rsnd_mod;
struct rsnd_dai;
@@ -96,24 +141,20 @@ void rsnd_bset(struct rsnd_priv *priv, struct rsnd_mod *mod, enum rsnd_reg reg,
* R-Car DMA
*/
struct rsnd_dma {
- struct rsnd_priv *priv;
struct sh_dmae_slave slave;
struct work_struct work;
struct dma_chan *chan;
enum dma_data_direction dir;
- int (*inquiry)(struct rsnd_dma *dma, dma_addr_t *buf, int *len);
- int (*complete)(struct rsnd_dma *dma);
int submit_loop;
+ int offset; /* it cares A/B plane */
};
void rsnd_dma_start(struct rsnd_dma *dma);
void rsnd_dma_stop(struct rsnd_dma *dma);
int rsnd_dma_available(struct rsnd_dma *dma);
int rsnd_dma_init(struct rsnd_priv *priv, struct rsnd_dma *dma,
- int is_play, int id,
- int (*inquiry)(struct rsnd_dma *dma, dma_addr_t *buf, int *len),
- int (*complete)(struct rsnd_dma *dma));
+ int is_play, int id);
void rsnd_dma_quit(struct rsnd_priv *priv,
struct rsnd_dma *dma);
@@ -121,9 +162,20 @@ void rsnd_dma_quit(struct rsnd_priv *priv,
/*
* R-Car sound mod
*/
+enum rsnd_mod_type {
+ RSND_MOD_SRC = 0,
+ RSND_MOD_SSI,
+ RSND_MOD_MAX,
+};
struct rsnd_mod_ops {
char *name;
+ int (*probe)(struct rsnd_mod *mod,
+ struct rsnd_dai *rdai,
+ struct rsnd_dai_stream *io);
+ int (*remove)(struct rsnd_mod *mod,
+ struct rsnd_dai *rdai,
+ struct rsnd_dai_stream *io);
int (*init)(struct rsnd_mod *mod,
struct rsnd_dai *rdai,
struct rsnd_dai_stream *io);
@@ -138,28 +190,26 @@ struct rsnd_mod_ops {
struct rsnd_dai_stream *io);
};
+struct rsnd_dai_stream;
struct rsnd_mod {
int id;
+ enum rsnd_mod_type type;
struct rsnd_priv *priv;
struct rsnd_mod_ops *ops;
- struct list_head list; /* connect to rsnd_dai playback/capture */
struct rsnd_dma dma;
+ struct rsnd_dai_stream *io;
};
#define rsnd_mod_to_priv(mod) ((mod)->priv)
#define rsnd_mod_to_dma(mod) (&(mod)->dma)
#define rsnd_dma_to_mod(_dma) container_of((_dma), struct rsnd_mod, dma)
+#define rsnd_mod_to_io(mod) ((mod)->io)
#define rsnd_mod_id(mod) ((mod)->id)
-#define for_each_rsnd_mod(pos, n, io) \
- list_for_each_entry_safe(pos, n, &(io)->head, list)
-#define rsnd_mod_call(mod, func, rdai, io) \
- (!(mod) ? -ENODEV : \
- !((mod)->ops->func) ? 0 : \
- (mod)->ops->func(mod, rdai, io))
void rsnd_mod_init(struct rsnd_priv *priv,
struct rsnd_mod *mod,
struct rsnd_mod_ops *ops,
+ enum rsnd_mod_type type,
int id);
char *rsnd_mod_name(struct rsnd_mod *mod);
@@ -168,13 +218,16 @@ char *rsnd_mod_name(struct rsnd_mod *mod);
*/
#define RSND_DAI_NAME_SIZE 16
struct rsnd_dai_stream {
- struct list_head head; /* head of rsnd_mod list */
struct snd_pcm_substream *substream;
+ struct rsnd_mod *mod[RSND_MOD_MAX];
+ struct rsnd_dai_path_info *info; /* rcar_snd.h */
int byte_pos;
int period_pos;
int byte_per_period;
int next_period_byte;
};
+#define rsnd_io_to_mod_ssi(io) ((io)->mod[RSND_MOD_SSI])
+#define rsnd_io_to_mod_src(io) ((io)->mod[RSND_MOD_SRC])
struct rsnd_dai {
char name[RSND_DAI_NAME_SIZE];
@@ -189,16 +242,14 @@ struct rsnd_dai {
unsigned int data_alignment:1;
};
-#define rsnd_dai_nr(priv) ((priv)->dai_nr)
+#define rsnd_rdai_nr(priv) ((priv)->rdai_nr)
#define for_each_rsnd_dai(rdai, priv, i) \
- for (i = 0, (rdai) = rsnd_dai_get(priv, i); \
- i < rsnd_dai_nr(priv); \
- i++, (rdai) = rsnd_dai_get(priv, i))
+ for (i = 0; \
+ (i < rsnd_rdai_nr(priv)) && \
+ ((rdai) = rsnd_dai_get(priv, i)); \
+ i++)
struct rsnd_dai *rsnd_dai_get(struct rsnd_priv *priv, int id);
-int rsnd_dai_disconnect(struct rsnd_mod *mod);
-int rsnd_dai_connect(struct rsnd_dai *rdai, struct rsnd_mod *mod,
- struct rsnd_dai_stream *io);
int rsnd_dai_is_play(struct rsnd_dai *rdai, struct rsnd_dai_stream *io);
int rsnd_dai_id(struct rsnd_priv *priv, struct rsnd_dai *rdai);
#define rsnd_dai_get_platform_info(rdai) ((rdai)->info)
@@ -206,21 +257,14 @@ int rsnd_dai_id(struct rsnd_priv *priv, struct rsnd_dai *rdai);
void rsnd_dai_pointer_update(struct rsnd_dai_stream *io, int cnt);
int rsnd_dai_pointer_offset(struct rsnd_dai_stream *io, int additional);
+#define rsnd_dai_is_clk_master(rdai) ((rdai)->clk_master)
/*
* R-Car Gen1/Gen2
*/
int rsnd_gen_probe(struct platform_device *pdev,
- struct rcar_snd_info *info,
+ const struct rsnd_of_data *of_data,
struct rsnd_priv *priv);
-void rsnd_gen_remove(struct platform_device *pdev,
- struct rsnd_priv *priv);
-int rsnd_gen_path_init(struct rsnd_priv *priv,
- struct rsnd_dai *rdai,
- struct rsnd_dai_stream *io);
-int rsnd_gen_path_exit(struct rsnd_priv *priv,
- struct rsnd_dai *rdai,
- struct rsnd_dai_stream *io);
void __iomem *rsnd_gen_reg_get(struct rsnd_priv *priv,
struct rsnd_mod *mod,
enum rsnd_reg reg);
@@ -233,18 +277,28 @@ void __iomem *rsnd_gen_reg_get(struct rsnd_priv *priv,
int rsnd_adg_ssi_clk_stop(struct rsnd_mod *mod);
int rsnd_adg_ssi_clk_try_start(struct rsnd_mod *mod, unsigned int rate);
int rsnd_adg_probe(struct platform_device *pdev,
- struct rcar_snd_info *info,
+ const struct rsnd_of_data *of_data,
struct rsnd_priv *priv);
-void rsnd_adg_remove(struct platform_device *pdev,
- struct rsnd_priv *priv);
-int rsnd_adg_set_convert_clk(struct rsnd_priv *priv,
- struct rsnd_mod *mod,
- unsigned int src_rate,
- unsigned int dst_rate);
+int rsnd_adg_set_convert_clk_gen1(struct rsnd_priv *priv,
+ struct rsnd_mod *mod,
+ unsigned int src_rate,
+ unsigned int dst_rate);
+int rsnd_adg_set_convert_clk_gen2(struct rsnd_mod *mod,
+ struct rsnd_dai *rdai,
+ struct rsnd_dai_stream *io,
+ unsigned int src_rate,
+ unsigned int dst_rate);
+int rsnd_adg_set_convert_timing_gen2(struct rsnd_mod *mod,
+ struct rsnd_dai *rdai,
+ struct rsnd_dai_stream *io);
/*
* R-Car sound priv
*/
+struct rsnd_of_data {
+ u32 flags;
+};
+
struct rsnd_priv {
struct device *dev;
@@ -257,10 +311,10 @@ struct rsnd_priv {
void *gen;
/*
- * below value will be filled on rsnd_scu_probe()
+ * below value will be filled on rsnd_src_probe()
*/
- void *scu;
- int scu_nr;
+ void *src;
+ int src_nr;
/*
* below value will be filled on rsnd_adg_probe()
@@ -270,46 +324,64 @@ struct rsnd_priv {
/*
* below value will be filled on rsnd_ssi_probe()
*/
- void *ssiu;
+ void *ssi;
+ int ssi_nr;
/*
* below value will be filled on rsnd_dai_probe()
*/
struct snd_soc_dai_driver *daidrv;
struct rsnd_dai *rdai;
- int dai_nr;
+ int rdai_nr;
};
#define rsnd_priv_to_dev(priv) ((priv)->dev)
+#define rsnd_priv_to_info(priv) ((priv)->info)
#define rsnd_lock(priv, flags) spin_lock_irqsave(&priv->lock, flags)
#define rsnd_unlock(priv, flags) spin_unlock_irqrestore(&priv->lock, flags)
+#define rsnd_info_is_playback(priv, type) \
+({ \
+ struct rcar_snd_info *info = rsnd_priv_to_info(priv); \
+ int i, is_play = 0; \
+ for (i = 0; i < info->dai_info_nr; i++) { \
+ if (info->dai_info[i].playback.type == (type)->info) { \
+ is_play = 1; \
+ break; \
+ } \
+ } \
+ is_play; \
+})
+
/*
- * R-Car SCU
+ * R-Car SRC
*/
-int rsnd_scu_probe(struct platform_device *pdev,
- struct rcar_snd_info *info,
+int rsnd_src_probe(struct platform_device *pdev,
+ const struct rsnd_of_data *of_data,
struct rsnd_priv *priv);
-void rsnd_scu_remove(struct platform_device *pdev,
- struct rsnd_priv *priv);
-struct rsnd_mod *rsnd_scu_mod_get(struct rsnd_priv *priv, int id);
-bool rsnd_scu_hpbif_is_enable(struct rsnd_mod *mod);
-unsigned int rsnd_scu_get_ssi_rate(struct rsnd_priv *priv,
- struct rsnd_mod *ssi_mod,
+struct rsnd_mod *rsnd_src_mod_get(struct rsnd_priv *priv, int id);
+unsigned int rsnd_src_get_ssi_rate(struct rsnd_priv *priv,
+ struct rsnd_dai_stream *io,
struct snd_pcm_runtime *runtime);
+int rsnd_src_ssi_mode_init(struct rsnd_mod *ssi_mod,
+ struct rsnd_dai *rdai,
+ struct rsnd_dai_stream *io);
+int rsnd_src_enable_ssi_irq(struct rsnd_mod *ssi_mod,
+ struct rsnd_dai *rdai,
+ struct rsnd_dai_stream *io);
-#define rsnd_scu_nr(priv) ((priv)->scu_nr)
+#define rsnd_src_nr(priv) ((priv)->src_nr)
/*
* R-Car SSI
*/
int rsnd_ssi_probe(struct platform_device *pdev,
- struct rcar_snd_info *info,
- struct rsnd_priv *priv);
-void rsnd_ssi_remove(struct platform_device *pdev,
+ const struct rsnd_of_data *of_data,
struct rsnd_priv *priv);
struct rsnd_mod *rsnd_ssi_mod_get(struct rsnd_priv *priv, int id);
struct rsnd_mod *rsnd_ssi_mod_get_frm_dai(struct rsnd_priv *priv,
int dai_id, int is_play);
+int rsnd_ssi_is_pin_sharing(struct rsnd_mod *mod);
+int rsnd_ssi_is_play(struct rsnd_mod *mod);
#endif
diff --git a/sound/soc/sh/rcar/scu.c b/sound/soc/sh/rcar/scu.c
deleted file mode 100644
index 9bb08bb..0000000
--- a/sound/soc/sh/rcar/scu.c
+++ /dev/null
@@ -1,384 +0,0 @@
-/*
- * Renesas R-Car SCU support
- *
- * Copyright (C) 2013 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 "rsnd.h"
-
-struct rsnd_scu {
- struct rsnd_scu_platform_info *info; /* rcar_snd.h */
- struct rsnd_mod mod;
- struct clk *clk;
-};
-
-#define rsnd_scu_mode_flags(p) ((p)->info->flags)
-#define rsnd_scu_convert_rate(p) ((p)->info->convert_rate)
-
-#define RSND_SCU_NAME_SIZE 16
-
-/*
- * ADINR
- */
-#define OTBL_24 (0 << 16)
-#define OTBL_22 (2 << 16)
-#define OTBL_20 (4 << 16)
-#define OTBL_18 (6 << 16)
-#define OTBL_16 (8 << 16)
-
-/*
- * image of SRC (Sampling Rate Converter)
- *
- * 96kHz <-> +-----+ 48kHz +-----+ 48kHz +-------+
- * 48kHz <-> | SRC | <------> | SSI | <-----> | codec |
- * 44.1kHz <-> +-----+ +-----+ +-------+
- * ...
- *
- */
-
-#define rsnd_mod_to_scu(_mod) \
- container_of((_mod), struct rsnd_scu, mod)
-
-#define for_each_rsnd_scu(pos, priv, i) \
- for ((i) = 0; \
- ((i) < rsnd_scu_nr(priv)) && \
- ((pos) = (struct rsnd_scu *)(priv)->scu + i); \
- i++)
-
-/* Gen1 only */
-static int rsnd_src_set_route_if_gen1(struct rsnd_priv *priv,
- struct rsnd_mod *mod,
- struct rsnd_dai *rdai,
- struct rsnd_dai_stream *io)
-{
- struct scu_route_config {
- u32 mask;
- int shift;
- } routes[] = {
- { 0xF, 0, }, /* 0 */
- { 0xF, 4, }, /* 1 */
- { 0xF, 8, }, /* 2 */
- { 0x7, 12, }, /* 3 */
- { 0x7, 16, }, /* 4 */
- { 0x7, 20, }, /* 5 */
- { 0x7, 24, }, /* 6 */
- { 0x3, 28, }, /* 7 */
- { 0x3, 30, }, /* 8 */
- };
- struct rsnd_scu *scu = rsnd_mod_to_scu(mod);
- u32 mask;
- u32 val;
- int shift;
- int id;
-
- /*
- * Gen1 only
- */
- if (!rsnd_is_gen1(priv))
- return 0;
-
- id = rsnd_mod_id(mod);
- if (id < 0 || id >= ARRAY_SIZE(routes))
- return -EIO;
-
- /*
- * SRC_ROUTE_SELECT
- */
- val = rsnd_dai_is_play(rdai, io) ? 0x1 : 0x2;
- val = val << routes[id].shift;
- mask = routes[id].mask << routes[id].shift;
-
- rsnd_mod_bset(mod, SRC_ROUTE_SEL, mask, val);
-
- /*
- * SRC_TIMING_SELECT
- */
- shift = (id % 4) * 8;
- mask = 0x1F << shift;
-
- /*
- * ADG is used as source clock if SRC was used,
- * then, SSI WS is used as destination clock.
- * SSI WS is used as source clock if SRC is not used
- * (when playback, source/destination become reverse when capture)
- */
- if (rsnd_scu_convert_rate(scu)) /* use ADG */
- val = 0;
- else if (8 == id) /* use SSI WS, but SRU8 is special */
- val = id << shift;
- else /* use SSI WS */
- val = (id + 1) << shift;
-
- switch (id / 4) {
- case 0:
- rsnd_mod_bset(mod, SRC_TMG_SEL0, mask, val);
- break;
- case 1:
- rsnd_mod_bset(mod, SRC_TMG_SEL1, mask, val);
- break;
- case 2:
- rsnd_mod_bset(mod, SRC_TMG_SEL2, mask, val);
- break;
- }
-
- return 0;
-}
-
-unsigned int rsnd_scu_get_ssi_rate(struct rsnd_priv *priv,
- struct rsnd_mod *ssi_mod,
- struct snd_pcm_runtime *runtime)
-{
- struct rsnd_scu *scu;
- unsigned int rate;
-
- /* this function is assuming SSI id = SCU id here */
- scu = rsnd_mod_to_scu(rsnd_scu_mod_get(priv, rsnd_mod_id(ssi_mod)));
-
- /*
- * return convert rate if SRC is used,
- * otherwise, return runtime->rate as usual
- */
- rate = rsnd_scu_convert_rate(scu);
- if (!rate)
- rate = runtime->rate;
-
- return rate;
-}
-
-static int rsnd_scu_convert_rate_ctrl(struct rsnd_priv *priv,
- struct rsnd_mod *mod,
- struct rsnd_dai *rdai,
- struct rsnd_dai_stream *io)
-{
- struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io);
- struct rsnd_scu *scu = rsnd_mod_to_scu(mod);
- u32 convert_rate = rsnd_scu_convert_rate(scu);
- u32 adinr = runtime->channels;
-
- /* set/clear soft reset */
- rsnd_mod_write(mod, SRC_SWRSR, 0);
- rsnd_mod_write(mod, SRC_SWRSR, 1);
-
- /* Initialize the operation of the SRC internal circuits */
- rsnd_mod_write(mod, SRC_SRCIR, 1);
-
- /* Set channel number and output bit length */
- switch (runtime->sample_bits) {
- case 16:
- adinr |= OTBL_16;
- break;
- case 32:
- adinr |= OTBL_24;
- break;
- default:
- return -EIO;
- }
- rsnd_mod_write(mod, SRC_ADINR, adinr);
-
- if (convert_rate) {
- u32 fsrate = 0x0400000 / convert_rate * runtime->rate;
- int ret;
-
- /* Enable the initial value of IFS */
- rsnd_mod_write(mod, SRC_IFSCR, 1);
-
- /* Set initial value of IFS */
- rsnd_mod_write(mod, SRC_IFSVR, fsrate);
-
- /* Select SRC mode (fixed value) */
- rsnd_mod_write(mod, SRC_SRCCR, 0x00010110);
-
- /* Set the restriction value of the FS ratio (98%) */
- rsnd_mod_write(mod, SRC_MNFSR, fsrate / 100 * 98);
-
- if (rsnd_is_gen1(priv)) {
- /* no SRC_BFSSR settings, since SRC_SRCCR::BUFMD is 0 */
- }
-
- /* set convert clock */
- ret = rsnd_adg_set_convert_clk(priv, mod,
- runtime->rate,
- convert_rate);
- if (ret < 0)
- return ret;
- }
-
- /* Cancel the initialization and operate the SRC function */
- rsnd_mod_write(mod, SRC_SRCIR, 0);
-
- /* use DMA transfer */
- rsnd_mod_write(mod, BUSIF_MODE, 1);
-
- return 0;
-}
-
-static int rsnd_scu_transfer_start(struct rsnd_priv *priv,
- struct rsnd_mod *mod,
- struct rsnd_dai *rdai,
- struct rsnd_dai_stream *io)
-{
- struct rsnd_scu *scu = rsnd_mod_to_scu(mod);
- int id = rsnd_mod_id(mod);
- u32 val;
-
- if (rsnd_is_gen1(priv)) {
- val = (1 << id);
- rsnd_mod_bset(mod, SRC_ROUTE_CTRL, val, val);
- }
-
- if (rsnd_scu_convert_rate(scu))
- rsnd_mod_write(mod, SRC_ROUTE_MODE0, 1);
-
- return 0;
-}
-
-static int rsnd_scu_transfer_stop(struct rsnd_priv *priv,
- struct rsnd_mod *mod,
- struct rsnd_dai *rdai,
- struct rsnd_dai_stream *io)
-{
- struct rsnd_scu *scu = rsnd_mod_to_scu(mod);
- int id = rsnd_mod_id(mod);
- u32 mask;
-
- if (rsnd_is_gen1(priv)) {
- mask = (1 << id);
- rsnd_mod_bset(mod, SRC_ROUTE_CTRL, mask, 0);
- }
-
- if (rsnd_scu_convert_rate(scu))
- rsnd_mod_write(mod, SRC_ROUTE_MODE0, 0);
-
- return 0;
-}
-
-bool rsnd_scu_hpbif_is_enable(struct rsnd_mod *mod)
-{
- struct rsnd_scu *scu = rsnd_mod_to_scu(mod);
- u32 flags = rsnd_scu_mode_flags(scu);
-
- return !!(flags & RSND_SCU_USE_HPBIF);
-}
-
-static int rsnd_scu_start(struct rsnd_mod *mod,
- struct rsnd_dai *rdai,
- struct rsnd_dai_stream *io)
-{
- struct rsnd_priv *priv = rsnd_mod_to_priv(mod);
- struct rsnd_scu *scu = rsnd_mod_to_scu(mod);
- struct device *dev = rsnd_priv_to_dev(priv);
- int ret;
-
- /*
- * SCU will be used if it has RSND_SCU_USE_HPBIF flags
- */
- if (!rsnd_scu_hpbif_is_enable(mod)) {
- /* it use PIO transter */
- dev_dbg(dev, "%s%d is not used\n",
- rsnd_mod_name(mod), rsnd_mod_id(mod));
-
- return 0;
- }
-
- clk_enable(scu->clk);
-
- /* it use DMA transter */
-
- ret = rsnd_src_set_route_if_gen1(priv, mod, rdai, io);
- if (ret < 0)
- return ret;
-
- ret = rsnd_scu_convert_rate_ctrl(priv, mod, rdai, io);
- if (ret < 0)
- return ret;
-
- ret = rsnd_scu_transfer_start(priv, mod, rdai, io);
- if (ret < 0)
- return ret;
-
- dev_dbg(dev, "%s%d start\n", rsnd_mod_name(mod), rsnd_mod_id(mod));
-
- return 0;
-}
-
-static int rsnd_scu_stop(struct rsnd_mod *mod,
- struct rsnd_dai *rdai,
- struct rsnd_dai_stream *io)
-{
- struct rsnd_priv *priv = rsnd_mod_to_priv(mod);
- struct rsnd_scu *scu = rsnd_mod_to_scu(mod);
-
- if (!rsnd_scu_hpbif_is_enable(mod))
- return 0;
-
- rsnd_scu_transfer_stop(priv, mod, rdai, io);
-
- clk_disable(scu->clk);
-
- return 0;
-}
-
-static struct rsnd_mod_ops rsnd_scu_ops = {
- .name = "scu",
- .start = rsnd_scu_start,
- .stop = rsnd_scu_stop,
-};
-
-struct rsnd_mod *rsnd_scu_mod_get(struct rsnd_priv *priv, int id)
-{
- if (WARN_ON(id < 0 || id >= rsnd_scu_nr(priv)))
- id = 0;
-
- return &((struct rsnd_scu *)(priv->scu) + id)->mod;
-}
-
-int rsnd_scu_probe(struct platform_device *pdev,
- struct rcar_snd_info *info,
- struct rsnd_priv *priv)
-{
- struct device *dev = rsnd_priv_to_dev(priv);
- struct rsnd_scu *scu;
- struct clk *clk;
- char name[RSND_SCU_NAME_SIZE];
- int i, nr;
-
- /*
- * init SCU
- */
- nr = info->scu_info_nr;
- scu = devm_kzalloc(dev, sizeof(*scu) * nr, GFP_KERNEL);
- if (!scu) {
- dev_err(dev, "SCU allocate failed\n");
- return -ENOMEM;
- }
-
- priv->scu_nr = nr;
- priv->scu = scu;
-
- for_each_rsnd_scu(scu, priv, i) {
- snprintf(name, RSND_SCU_NAME_SIZE, "scu.%d", i);
-
- clk = devm_clk_get(dev, name);
- if (IS_ERR(clk))
- return PTR_ERR(clk);
-
- rsnd_mod_init(priv, &scu->mod,
- &rsnd_scu_ops, i);
- scu->info = &info->scu_info[i];
- scu->clk = clk;
-
- dev_dbg(dev, "SCU%d probed\n", i);
- }
- dev_dbg(dev, "scu probed\n");
-
- return 0;
-}
-
-void rsnd_scu_remove(struct platform_device *pdev,
- struct rsnd_priv *priv)
-{
-}
diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c
new file mode 100644
index 0000000..eee75eb
--- /dev/null
+++ b/sound/soc/sh/rcar/src.c
@@ -0,0 +1,723 @@
+/*
+ * Renesas R-Car SRC support
+ *
+ * Copyright (C) 2013 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 "rsnd.h"
+
+struct rsnd_src {
+ struct rsnd_src_platform_info *info; /* rcar_snd.h */
+ struct rsnd_mod mod;
+ struct clk *clk;
+};
+
+#define RSND_SRC_NAME_SIZE 16
+
+/*
+ * ADINR
+ */
+#define OTBL_24 (0 << 16)
+#define OTBL_22 (2 << 16)
+#define OTBL_20 (4 << 16)
+#define OTBL_18 (6 << 16)
+#define OTBL_16 (8 << 16)
+
+#define rsnd_src_mode_flags(p) ((p)->info->flags)
+#define rsnd_src_convert_rate(p) ((p)->info->convert_rate)
+#define rsnd_mod_to_src(_mod) \
+ container_of((_mod), struct rsnd_src, mod)
+#define rsnd_src_hpbif_is_enable(src) \
+ (rsnd_src_mode_flags(src) & RSND_SCU_USE_HPBIF)
+#define rsnd_src_dma_available(src) \
+ rsnd_dma_available(rsnd_mod_to_dma(&(src)->mod))
+
+#define for_each_rsnd_src(pos, priv, i) \
+ for ((i) = 0; \
+ ((i) < rsnd_src_nr(priv)) && \
+ ((pos) = (struct rsnd_src *)(priv)->src + i); \
+ i++)
+
+
+/*
+ * image of SRC (Sampling Rate Converter)
+ *
+ * 96kHz <-> +-----+ 48kHz +-----+ 48kHz +-------+
+ * 48kHz <-> | SRC | <------> | SSI | <-----> | codec |
+ * 44.1kHz <-> +-----+ +-----+ +-------+
+ * ...
+ *
+ */
+
+/*
+ * src.c is caring...
+ *
+ * Gen1
+ *
+ * [mem] -> [SRU] -> [SSI]
+ * |--------|
+ *
+ * Gen2
+ *
+ * [mem] -> [SRC] -> [SSIU] -> [SSI]
+ * |-----------------|
+ */
+
+/*
+ * How to use SRC bypass mode for debugging
+ *
+ * SRC has bypass mode, and it is useful for debugging.
+ * In Gen2 case,
+ * SRCm_MODE controls whether SRC is used or not
+ * SSI_MODE0 controls whether SSIU which receives SRC data
+ * is used or not.
+ * Both SRCm_MODE/SSI_MODE0 settings are needed if you use SRC,
+ * but SRC bypass mode needs SSI_MODE0 only.
+ *
+ * This driver request
+ * struct rsnd_src_platform_info {
+ * u32 flags;
+ * u32 convert_rate;
+ * }
+ *
+ * rsnd_src_hpbif_is_enable() will be true
+ * if flags had RSND_SRC_USE_HPBIF,
+ * and it controls whether SSIU is used or not.
+ *
+ * rsnd_src_convert_rate() indicates
+ * above convert_rate, and it controls
+ * whether SRC is used or not.
+ *
+ * ex) doesn't use SRC
+ * struct rsnd_src_platform_info info = {
+ * .flags = 0,
+ * .convert_rate = 0,
+ * };
+ *
+ * ex) uses SRC
+ * struct rsnd_src_platform_info info = {
+ * .flags = RSND_SRC_USE_HPBIF,
+ * .convert_rate = 48000,
+ * };
+ *
+ * ex) uses SRC bypass mode
+ * struct rsnd_src_platform_info info = {
+ * .flags = RSND_SRC_USE_HPBIF,
+ * .convert_rate = 0,
+ * };
+ *
+ */
+
+/*
+ * Gen1/Gen2 common functions
+ */
+int rsnd_src_ssi_mode_init(struct rsnd_mod *ssi_mod,
+ struct rsnd_dai *rdai,
+ struct rsnd_dai_stream *io)
+{
+ struct rsnd_priv *priv = rsnd_mod_to_priv(ssi_mod);
+ struct rsnd_mod *src_mod = rsnd_io_to_mod_src(io);
+ struct rcar_snd_info *info = rsnd_priv_to_info(priv);
+ int ssi_id = rsnd_mod_id(ssi_mod);
+ int has_src = 0;
+
+ /*
+ * SSI_MODE0
+ */
+ if (info->dai_info) {
+ has_src = !!src_mod;
+ } else {
+ struct rsnd_src *src = rsnd_mod_to_src(src_mod);
+ has_src = rsnd_src_hpbif_is_enable(src);
+ }
+
+ rsnd_mod_bset(ssi_mod, SSI_MODE0, (1 << ssi_id),
+ has_src ? 0 : (1 << ssi_id));
+
+ /*
+ * SSI_MODE1
+ */
+ if (rsnd_ssi_is_pin_sharing(ssi_mod)) {
+ int shift = -1;
+ switch (ssi_id) {
+ case 1:
+ shift = 0;
+ break;
+ case 2:
+ shift = 2;
+ break;
+ case 4:
+ shift = 16;
+ break;
+ }
+
+ if (shift >= 0)
+ rsnd_mod_bset(ssi_mod, SSI_MODE1,
+ 0x3 << shift,
+ rsnd_dai_is_clk_master(rdai) ?
+ 0x2 << shift : 0x1 << shift);
+ }
+
+ return 0;
+}
+
+int rsnd_src_enable_ssi_irq(struct rsnd_mod *ssi_mod,
+ struct rsnd_dai *rdai,
+ struct rsnd_dai_stream *io)
+{
+ struct rsnd_priv *priv = rsnd_mod_to_priv(ssi_mod);
+
+ /* enable PIO interrupt if Gen2 */
+ if (rsnd_is_gen2(priv))
+ rsnd_mod_write(ssi_mod, INT_ENABLE, 0x0f000000);
+
+ return 0;
+}
+
+unsigned int rsnd_src_get_ssi_rate(struct rsnd_priv *priv,
+ struct rsnd_dai_stream *io,
+ struct snd_pcm_runtime *runtime)
+{
+ struct rsnd_src *src;
+ unsigned int rate;
+
+ src = rsnd_mod_to_src(rsnd_io_to_mod_src(io));
+
+ /*
+ * return convert rate if SRC is used,
+ * otherwise, return runtime->rate as usual
+ */
+ rate = rsnd_src_convert_rate(src);
+ if (!rate)
+ rate = runtime->rate;
+
+ return rate;
+}
+
+static int rsnd_src_set_convert_rate(struct rsnd_mod *mod,
+ struct rsnd_dai *rdai,
+ struct rsnd_dai_stream *io)
+{
+ struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io);
+ struct rsnd_src *src = rsnd_mod_to_src(mod);
+ u32 convert_rate = rsnd_src_convert_rate(src);
+ u32 adinr = runtime->channels;
+ u32 fsrate = 0;
+
+ if (convert_rate)
+ fsrate = 0x0400000 / convert_rate * runtime->rate;
+
+ /* set/clear soft reset */
+ rsnd_mod_write(mod, SRC_SWRSR, 0);
+ rsnd_mod_write(mod, SRC_SWRSR, 1);
+
+ /*
+ * Initialize the operation of the SRC internal circuits
+ * see rsnd_src_start()
+ */
+ rsnd_mod_write(mod, SRC_SRCIR, 1);
+
+ /* Set channel number and output bit length */
+ switch (runtime->sample_bits) {
+ case 16:
+ adinr |= OTBL_16;
+ break;
+ case 32:
+ adinr |= OTBL_24;
+ break;
+ default:
+ return -EIO;
+ }
+ rsnd_mod_write(mod, SRC_ADINR, adinr);
+
+ /* Enable the initial value of IFS */
+ if (fsrate) {
+ rsnd_mod_write(mod, SRC_IFSCR, 1);
+
+ /* Set initial value of IFS */
+ rsnd_mod_write(mod, SRC_IFSVR, fsrate);
+ }
+
+ /* use DMA transfer */
+ rsnd_mod_write(mod, SRC_BUSIF_MODE, 1);
+
+ return 0;
+}
+
+static int rsnd_src_init(struct rsnd_mod *mod,
+ struct rsnd_dai *rdai,
+ struct rsnd_dai_stream *io)
+{
+ struct rsnd_src *src = rsnd_mod_to_src(mod);
+
+ clk_enable(src->clk);
+
+ return 0;
+}
+
+static int rsnd_src_quit(struct rsnd_mod *mod,
+ struct rsnd_dai *rdai,
+ struct rsnd_dai_stream *io)
+{
+ struct rsnd_src *src = rsnd_mod_to_src(mod);
+
+ clk_disable(src->clk);
+
+ return 0;
+}
+
+static int rsnd_src_start(struct rsnd_mod *mod,
+ struct rsnd_dai *rdai,
+ struct rsnd_dai_stream *io)
+{
+ struct rsnd_src *src = rsnd_mod_to_src(mod);
+
+ /*
+ * Cancel the initialization and operate the SRC function
+ * see rsnd_src_set_convert_rate()
+ */
+ rsnd_mod_write(mod, SRC_SRCIR, 0);
+
+ if (rsnd_src_convert_rate(src))
+ rsnd_mod_write(mod, SRC_ROUTE_MODE0, 1);
+
+ return 0;
+}
+
+
+static int rsnd_src_stop(struct rsnd_mod *mod,
+ struct rsnd_dai *rdai,
+ struct rsnd_dai_stream *io)
+{
+ struct rsnd_src *src = rsnd_mod_to_src(mod);
+
+ if (rsnd_src_convert_rate(src))
+ rsnd_mod_write(mod, SRC_ROUTE_MODE0, 0);
+
+ return 0;
+}
+
+static struct rsnd_mod_ops rsnd_src_non_ops = {
+ .name = "src (non)",
+};
+
+/*
+ * Gen1 functions
+ */
+static int rsnd_src_set_route_gen1(struct rsnd_mod *mod,
+ struct rsnd_dai *rdai,
+ struct rsnd_dai_stream *io)
+{
+ struct src_route_config {
+ u32 mask;
+ int shift;
+ } routes[] = {
+ { 0xF, 0, }, /* 0 */
+ { 0xF, 4, }, /* 1 */
+ { 0xF, 8, }, /* 2 */
+ { 0x7, 12, }, /* 3 */
+ { 0x7, 16, }, /* 4 */
+ { 0x7, 20, }, /* 5 */
+ { 0x7, 24, }, /* 6 */
+ { 0x3, 28, }, /* 7 */
+ { 0x3, 30, }, /* 8 */
+ };
+ u32 mask;
+ u32 val;
+ int id;
+
+ id = rsnd_mod_id(mod);
+ if (id < 0 || id >= ARRAY_SIZE(routes))
+ return -EIO;
+
+ /*
+ * SRC_ROUTE_SELECT
+ */
+ val = rsnd_dai_is_play(rdai, io) ? 0x1 : 0x2;
+ val = val << routes[id].shift;
+ mask = routes[id].mask << routes[id].shift;
+
+ rsnd_mod_bset(mod, SRC_ROUTE_SEL, mask, val);
+
+ return 0;
+}
+
+static int rsnd_src_set_convert_timing_gen1(struct rsnd_mod *mod,
+ struct rsnd_dai *rdai,
+ struct rsnd_dai_stream *io)
+{
+ struct rsnd_priv *priv = rsnd_mod_to_priv(mod);
+ struct rsnd_src *src = rsnd_mod_to_src(mod);
+ struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io);
+ u32 convert_rate = rsnd_src_convert_rate(src);
+ u32 mask;
+ u32 val;
+ int shift;
+ int id = rsnd_mod_id(mod);
+ int ret;
+
+ /*
+ * SRC_TIMING_SELECT
+ */
+ shift = (id % 4) * 8;
+ mask = 0x1F << shift;
+
+ /*
+ * ADG is used as source clock if SRC was used,
+ * then, SSI WS is used as destination clock.
+ * SSI WS is used as source clock if SRC is not used
+ * (when playback, source/destination become reverse when capture)
+ */
+ ret = 0;
+ if (convert_rate) {
+ /* use ADG */
+ val = 0;
+ ret = rsnd_adg_set_convert_clk_gen1(priv, mod,
+ runtime->rate,
+ convert_rate);
+ } else if (8 == id) {
+ /* use SSI WS, but SRU8 is special */
+ val = id << shift;
+ } else {
+ /* use SSI WS */
+ val = (id + 1) << shift;
+ }
+
+ if (ret < 0)
+ return ret;
+
+ switch (id / 4) {
+ case 0:
+ rsnd_mod_bset(mod, SRC_TMG_SEL0, mask, val);
+ break;
+ case 1:
+ rsnd_mod_bset(mod, SRC_TMG_SEL1, mask, val);
+ break;
+ case 2:
+ rsnd_mod_bset(mod, SRC_TMG_SEL2, mask, val);
+ break;
+ }
+
+ return 0;
+}
+
+static int rsnd_src_set_convert_rate_gen1(struct rsnd_mod *mod,
+ struct rsnd_dai *rdai,
+ struct rsnd_dai_stream *io)
+{
+ int ret;
+
+ ret = rsnd_src_set_convert_rate(mod, rdai, io);
+ if (ret < 0)
+ return ret;
+
+ /* Select SRC mode (fixed value) */
+ rsnd_mod_write(mod, SRC_SRCCR, 0x00010110);
+
+ /* Set the restriction value of the FS ratio (98%) */
+ rsnd_mod_write(mod, SRC_MNFSR,
+ rsnd_mod_read(mod, SRC_IFSVR) / 100 * 98);
+
+ /* no SRC_BFSSR settings, since SRC_SRCCR::BUFMD is 0 */
+
+ return 0;
+}
+
+static int rsnd_src_init_gen1(struct rsnd_mod *mod,
+ struct rsnd_dai *rdai,
+ struct rsnd_dai_stream *io)
+{
+ int ret;
+
+ ret = rsnd_src_init(mod, rdai, io);
+ if (ret < 0)
+ return ret;
+
+ ret = rsnd_src_set_route_gen1(mod, rdai, io);
+ if (ret < 0)
+ return ret;
+
+ ret = rsnd_src_set_convert_rate_gen1(mod, rdai, io);
+ if (ret < 0)
+ return ret;
+
+ ret = rsnd_src_set_convert_timing_gen1(mod, rdai, io);
+ if (ret < 0)
+ return ret;
+
+ return 0;
+}
+
+static int rsnd_src_start_gen1(struct rsnd_mod *mod,
+ struct rsnd_dai *rdai,
+ struct rsnd_dai_stream *io)
+{
+ int id = rsnd_mod_id(mod);
+
+ rsnd_mod_bset(mod, SRC_ROUTE_CTRL, (1 << id), (1 << id));
+
+ return rsnd_src_start(mod, rdai, io);
+}
+
+static int rsnd_src_stop_gen1(struct rsnd_mod *mod,
+ struct rsnd_dai *rdai,
+ struct rsnd_dai_stream *io)
+{
+ int id = rsnd_mod_id(mod);
+
+ rsnd_mod_bset(mod, SRC_ROUTE_CTRL, (1 << id), 0);
+
+ return rsnd_src_stop(mod, rdai, io);
+}
+
+static struct rsnd_mod_ops rsnd_src_gen1_ops = {
+ .name = "sru (gen1)",
+ .init = rsnd_src_init_gen1,
+ .quit = rsnd_src_quit,
+ .start = rsnd_src_start_gen1,
+ .stop = rsnd_src_stop_gen1,
+};
+
+/*
+ * Gen2 functions
+ */
+static int rsnd_src_set_convert_rate_gen2(struct rsnd_mod *mod,
+ struct rsnd_dai *rdai,
+ struct rsnd_dai_stream *io)
+{
+ int ret;
+
+ ret = rsnd_src_set_convert_rate(mod, rdai, io);
+ if (ret < 0)
+ return ret;
+
+ rsnd_mod_write(mod, SSI_BUSIF_ADINR, rsnd_mod_read(mod, SRC_ADINR));
+ rsnd_mod_write(mod, SSI_BUSIF_MODE, rsnd_mod_read(mod, SRC_BUSIF_MODE));
+
+ rsnd_mod_write(mod, SRC_SRCCR, 0x00011110);
+
+ rsnd_mod_write(mod, SRC_BSDSR, 0x01800000);
+ rsnd_mod_write(mod, SRC_BSISR, 0x00100060);
+
+ return 0;
+}
+
+static int rsnd_src_set_convert_timing_gen2(struct rsnd_mod *mod,
+ struct rsnd_dai *rdai,
+ struct rsnd_dai_stream *io)
+{
+ struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io);
+ struct rsnd_src *src = rsnd_mod_to_src(mod);
+ u32 convert_rate = rsnd_src_convert_rate(src);
+ int ret;
+
+ if (convert_rate)
+ ret = rsnd_adg_set_convert_clk_gen2(mod, rdai, io,
+ runtime->rate,
+ convert_rate);
+ else
+ ret = rsnd_adg_set_convert_timing_gen2(mod, rdai, io);
+
+ return ret;
+}
+
+static int rsnd_src_probe_gen2(struct rsnd_mod *mod,
+ struct rsnd_dai *rdai,
+ struct rsnd_dai_stream *io)
+{
+ struct rsnd_priv *priv = rsnd_mod_to_priv(mod);
+ struct rcar_snd_info *info = rsnd_priv_to_info(priv);
+ struct rsnd_src *src = rsnd_mod_to_src(mod);
+ struct rsnd_mod *ssi = rsnd_ssi_mod_get(priv, rsnd_mod_id(mod));
+ struct device *dev = rsnd_priv_to_dev(priv);
+ int ret;
+ int is_play;
+
+ if (info->dai_info)
+ is_play = rsnd_info_is_playback(priv, src);
+ else
+ is_play = rsnd_ssi_is_play(ssi);
+
+ ret = rsnd_dma_init(priv,
+ rsnd_mod_to_dma(mod),
+ is_play,
+ src->info->dma_id);
+ if (ret < 0)
+ dev_err(dev, "SRC DMA failed\n");
+
+ return ret;
+}
+
+static int rsnd_src_remove_gen2(struct rsnd_mod *mod,
+ struct rsnd_dai *rdai,
+ struct rsnd_dai_stream *io)
+{
+ rsnd_dma_quit(rsnd_mod_to_priv(mod), rsnd_mod_to_dma(mod));
+
+ return 0;
+}
+
+static int rsnd_src_init_gen2(struct rsnd_mod *mod,
+ struct rsnd_dai *rdai,
+ struct rsnd_dai_stream *io)
+{
+ int ret;
+
+ ret = rsnd_src_init(mod, rdai, io);
+ if (ret < 0)
+ return ret;
+
+ ret = rsnd_src_set_convert_rate_gen2(mod, rdai, io);
+ if (ret < 0)
+ return ret;
+
+ ret = rsnd_src_set_convert_timing_gen2(mod, rdai, io);
+ if (ret < 0)
+ return ret;
+
+ return 0;
+}
+
+static int rsnd_src_start_gen2(struct rsnd_mod *mod,
+ struct rsnd_dai *rdai,
+ struct rsnd_dai_stream *io)
+{
+ struct rsnd_src *src = rsnd_mod_to_src(mod);
+
+ rsnd_dma_start(rsnd_mod_to_dma(&src->mod));
+
+ rsnd_mod_write(mod, SSI_CTRL, 0x1);
+ rsnd_mod_write(mod, SRC_CTRL, 0x11);
+
+ return rsnd_src_start(mod, rdai, io);
+}
+
+static int rsnd_src_stop_gen2(struct rsnd_mod *mod,
+ struct rsnd_dai *rdai,
+ struct rsnd_dai_stream *io)
+{
+ struct rsnd_src *src = rsnd_mod_to_src(mod);
+
+ rsnd_mod_write(mod, SSI_CTRL, 0);
+ rsnd_mod_write(mod, SRC_CTRL, 0);
+
+ rsnd_dma_stop(rsnd_mod_to_dma(&src->mod));
+
+ return rsnd_src_stop(mod, rdai, io);
+}
+
+static struct rsnd_mod_ops rsnd_src_gen2_ops = {
+ .name = "src (gen2)",
+ .probe = rsnd_src_probe_gen2,
+ .remove = rsnd_src_remove_gen2,
+ .init = rsnd_src_init_gen2,
+ .quit = rsnd_src_quit,
+ .start = rsnd_src_start_gen2,
+ .stop = rsnd_src_stop_gen2,
+};
+
+struct rsnd_mod *rsnd_src_mod_get(struct rsnd_priv *priv, int id)
+{
+ if (WARN_ON(id < 0 || id >= rsnd_src_nr(priv)))
+ id = 0;
+
+ return &((struct rsnd_src *)(priv->src) + id)->mod;
+}
+
+static void rsnd_of_parse_src(struct platform_device *pdev,
+ const struct rsnd_of_data *of_data,
+ struct rsnd_priv *priv)
+{
+ struct device_node *src_node;
+ struct rcar_snd_info *info = rsnd_priv_to_info(priv);
+ struct rsnd_src_platform_info *src_info;
+ struct device *dev = &pdev->dev;
+ int nr;
+
+ if (!of_data)
+ return;
+
+ src_node = of_get_child_by_name(dev->of_node, "rcar_sound,src");
+ if (!src_node)
+ return;
+
+ nr = of_get_child_count(src_node);
+ if (!nr)
+ return;
+
+ src_info = devm_kzalloc(dev,
+ sizeof(struct rsnd_src_platform_info) * nr,
+ GFP_KERNEL);
+ if (!src_info) {
+ dev_err(dev, "src info allocation error\n");
+ return;
+ }
+
+ info->src_info = src_info;
+ info->src_info_nr = nr;
+}
+
+int rsnd_src_probe(struct platform_device *pdev,
+ const struct rsnd_of_data *of_data,
+ struct rsnd_priv *priv)
+{
+ struct rcar_snd_info *info = rsnd_priv_to_info(priv);
+ struct device *dev = rsnd_priv_to_dev(priv);
+ struct rsnd_src *src;
+ struct rsnd_mod_ops *ops;
+ struct clk *clk;
+ char name[RSND_SRC_NAME_SIZE];
+ int i, nr;
+
+ rsnd_of_parse_src(pdev, of_data, priv);
+
+ /*
+ * init SRC
+ */
+ nr = info->src_info_nr;
+ if (!nr)
+ return 0;
+
+ src = devm_kzalloc(dev, sizeof(*src) * nr, GFP_KERNEL);
+ if (!src) {
+ dev_err(dev, "SRC allocate failed\n");
+ return -ENOMEM;
+ }
+
+ priv->src_nr = nr;
+ priv->src = src;
+
+ for_each_rsnd_src(src, priv, i) {
+ snprintf(name, RSND_SRC_NAME_SIZE, "src.%d", i);
+
+ clk = devm_clk_get(dev, name);
+ if (IS_ERR(clk)) {
+ snprintf(name, RSND_SRC_NAME_SIZE, "scu.%d", i);
+ clk = devm_clk_get(dev, name);
+ }
+
+ if (IS_ERR(clk))
+ return PTR_ERR(clk);
+
+ src->info = &info->src_info[i];
+ src->clk = clk;
+
+ ops = &rsnd_src_non_ops;
+ if (rsnd_src_hpbif_is_enable(src)) {
+ if (rsnd_is_gen1(priv))
+ ops = &rsnd_src_gen1_ops;
+ if (rsnd_is_gen2(priv))
+ ops = &rsnd_src_gen2_ops;
+ }
+
+ rsnd_mod_init(priv, &src->mod, ops, RSND_MOD_SRC, i);
+
+ dev_dbg(dev, "SRC%d probed\n", i);
+ }
+
+ return 0;
+}
diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c
index 4b8cf7c..4b7e206 100644
--- a/sound/soc/sh/rcar/ssi.c
+++ b/sound/soc/sh/rcar/ssi.c
@@ -64,108 +64,29 @@ struct rsnd_ssi {
struct rsnd_mod mod;
struct rsnd_dai *rdai;
- struct rsnd_dai_stream *io;
u32 cr_own;
u32 cr_clk;
u32 cr_etc;
int err;
- int dma_offset;
unsigned int usrcnt;
unsigned int rate;
};
-struct rsnd_ssiu {
- u32 ssi_mode0;
- u32 ssi_mode1;
-
- int ssi_nr;
- struct rsnd_ssi *ssi;
-};
-
#define for_each_rsnd_ssi(pos, priv, i) \
for (i = 0; \
(i < rsnd_ssi_nr(priv)) && \
- ((pos) = ((struct rsnd_ssiu *)((priv)->ssiu))->ssi + i); \
+ ((pos) = ((struct rsnd_ssi *)(priv)->ssi + i)); \
i++)
-#define rsnd_ssi_nr(priv) (((struct rsnd_ssiu *)((priv)->ssiu))->ssi_nr)
+#define rsnd_ssi_nr(priv) ((priv)->ssi_nr)
#define rsnd_mod_to_ssi(_mod) container_of((_mod), struct rsnd_ssi, mod)
#define rsnd_dma_to_ssi(dma) rsnd_mod_to_ssi(rsnd_dma_to_mod(dma))
#define rsnd_ssi_pio_available(ssi) ((ssi)->info->pio_irq > 0)
#define rsnd_ssi_dma_available(ssi) \
rsnd_dma_available(rsnd_mod_to_dma(&(ssi)->mod))
#define rsnd_ssi_clk_from_parent(ssi) ((ssi)->parent)
-#define rsnd_rdai_is_clk_master(rdai) ((rdai)->clk_master)
#define rsnd_ssi_mode_flags(p) ((p)->info->flags)
#define rsnd_ssi_dai_id(ssi) ((ssi)->info->dai_id)
-#define rsnd_ssi_to_ssiu(ssi)\
- (((struct rsnd_ssiu *)((ssi) - rsnd_mod_id(&(ssi)->mod))) - 1)
-
-static void rsnd_ssi_mode_set(struct rsnd_priv *priv,
- struct rsnd_dai *rdai,
- struct rsnd_ssi *ssi)
-{
- struct device *dev = rsnd_priv_to_dev(priv);
- struct rsnd_mod *scu;
- struct rsnd_ssiu *ssiu = rsnd_ssi_to_ssiu(ssi);
- int id = rsnd_mod_id(&ssi->mod);
- u32 flags;
- u32 val;
-
- scu = rsnd_scu_mod_get(priv, rsnd_mod_id(&ssi->mod));
-
- /*
- * SSI_MODE0
- */
-
- /* see also BUSIF_MODE */
- if (rsnd_scu_hpbif_is_enable(scu)) {
- ssiu->ssi_mode0 &= ~(1 << id);
- dev_dbg(dev, "SSI%d uses DEPENDENT mode\n", id);
- } else {
- ssiu->ssi_mode0 |= (1 << id);
- dev_dbg(dev, "SSI%d uses INDEPENDENT mode\n", id);
- }
-
- /*
- * SSI_MODE1
- */
-#define ssi_parent_set(p, sync, adg, ext) \
- do { \
- ssi->parent = ssiu->ssi + p; \
- if (rsnd_rdai_is_clk_master(rdai)) \
- val = adg; \
- else \
- val = ext; \
- if (flags & RSND_SSI_SYNC) \
- val |= sync; \
- } while (0)
-
- flags = rsnd_ssi_mode_flags(ssi);
- if (flags & RSND_SSI_CLK_PIN_SHARE) {
-
- val = 0;
- switch (id) {
- case 1:
- ssi_parent_set(0, (1 << 4), (0x2 << 0), (0x1 << 0));
- break;
- case 2:
- ssi_parent_set(0, (1 << 4), (0x2 << 2), (0x1 << 2));
- break;
- case 4:
- ssi_parent_set(3, (1 << 20), (0x2 << 16), (0x1 << 16));
- break;
- case 8:
- ssi_parent_set(7, 0, 0, 0);
- break;
- }
-
- ssiu->ssi_mode1 |= val;
- }
-
- rsnd_mod_write(&ssi->mod, SSI_MODE0, ssiu->ssi_mode0);
- rsnd_mod_write(&ssi->mod, SSI_MODE1, ssiu->ssi_mode1);
-}
static void rsnd_ssi_status_check(struct rsnd_mod *mod,
u32 bit)
@@ -200,7 +121,7 @@ static int rsnd_ssi_master_clk_start(struct rsnd_ssi *ssi,
1, 2, 4, 8, 16, 6, 12,
};
unsigned int main_rate;
- unsigned int rate = rsnd_scu_get_ssi_rate(priv, &ssi->mod, runtime);
+ unsigned int rate = rsnd_src_get_ssi_rate(priv, io, runtime);
/*
* Find best clock, and try to start ADG
@@ -252,7 +173,7 @@ static void rsnd_ssi_hw_start(struct rsnd_ssi *ssi,
if (0 == ssi->usrcnt) {
clk_enable(ssi->clk);
- if (rsnd_rdai_is_clk_master(rdai)) {
+ if (rsnd_dai_is_clk_master(rdai)) {
if (rsnd_ssi_clk_from_parent(ssi))
rsnd_ssi_hw_start(ssi->parent, rdai, io);
else
@@ -302,7 +223,7 @@ static void rsnd_ssi_hw_stop(struct rsnd_ssi *ssi,
rsnd_mod_write(&ssi->mod, SSICR, cr); /* disabled all */
rsnd_ssi_status_check(&ssi->mod, IIRQ);
- if (rsnd_rdai_is_clk_master(rdai)) {
+ if (rsnd_dai_is_clk_master(rdai)) {
if (rsnd_ssi_clk_from_parent(ssi))
rsnd_ssi_hw_stop(ssi->parent, rdai);
else
@@ -323,8 +244,6 @@ static int rsnd_ssi_init(struct rsnd_mod *mod,
struct rsnd_dai_stream *io)
{
struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod);
- struct rsnd_priv *priv = rsnd_mod_to_priv(mod);
- struct device *dev = rsnd_priv_to_dev(priv);
struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io);
u32 cr;
@@ -365,13 +284,10 @@ static int rsnd_ssi_init(struct rsnd_mod *mod,
* set ssi parameter
*/
ssi->rdai = rdai;
- ssi->io = io;
ssi->cr_own = cr;
ssi->err = -1; /* ignore 1st error */
- rsnd_ssi_mode_set(priv, rdai, ssi);
-
- dev_dbg(dev, "%s.%d init\n", rsnd_mod_name(mod), rsnd_mod_id(mod));
+ rsnd_src_ssi_mode_init(mod, rdai, io);
return 0;
}
@@ -384,13 +300,10 @@ static int rsnd_ssi_quit(struct rsnd_mod *mod,
struct rsnd_priv *priv = rsnd_mod_to_priv(mod);
struct device *dev = rsnd_priv_to_dev(priv);
- dev_dbg(dev, "%s.%d quit\n", rsnd_mod_name(mod), rsnd_mod_id(mod));
-
if (ssi->err > 0)
dev_warn(dev, "ssi under/over flow err = %d\n", ssi->err);
ssi->rdai = NULL;
- ssi->io = NULL;
ssi->cr_own = 0;
ssi->err = 0;
@@ -414,8 +327,9 @@ static void rsnd_ssi_record_error(struct rsnd_ssi *ssi, u32 status)
static irqreturn_t rsnd_ssi_pio_interrupt(int irq, void *data)
{
struct rsnd_ssi *ssi = data;
- struct rsnd_dai_stream *io = ssi->io;
- u32 status = rsnd_mod_read(&ssi->mod, SSISR);
+ struct rsnd_mod *mod = &ssi->mod;
+ struct rsnd_dai_stream *io = rsnd_mod_to_io(mod);
+ u32 status = rsnd_mod_read(mod, SSISR);
irqreturn_t ret = IRQ_NONE;
if (io && (status & DIRQ)) {
@@ -432,9 +346,9 @@ static irqreturn_t rsnd_ssi_pio_interrupt(int irq, void *data)
* see rsnd_ssi_init()
*/
if (rsnd_dai_is_play(rdai, io))
- rsnd_mod_write(&ssi->mod, SSITDR, *buf);
+ rsnd_mod_write(mod, SSITDR, *buf);
else
- *buf = rsnd_mod_read(&ssi->mod, SSIRDR);
+ *buf = rsnd_mod_read(mod, SSIRDR);
rsnd_dai_pointer_update(io, sizeof(*buf));
@@ -444,25 +358,39 @@ static irqreturn_t rsnd_ssi_pio_interrupt(int irq, void *data)
return ret;
}
-static int rsnd_ssi_pio_start(struct rsnd_mod *mod,
+static int rsnd_ssi_pio_probe(struct rsnd_mod *mod,
struct rsnd_dai *rdai,
struct rsnd_dai_stream *io)
{
struct rsnd_priv *priv = rsnd_mod_to_priv(mod);
- struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod);
struct device *dev = rsnd_priv_to_dev(priv);
+ struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod);
+ int irq = ssi->info->pio_irq;
+ int ret;
+
+ ret = devm_request_irq(dev, irq,
+ rsnd_ssi_pio_interrupt,
+ IRQF_SHARED,
+ dev_name(dev), ssi);
+ if (ret)
+ dev_err(dev, "SSI request interrupt failed\n");
+
+ return ret;
+}
+
+static int rsnd_ssi_pio_start(struct rsnd_mod *mod,
+ struct rsnd_dai *rdai,
+ struct rsnd_dai_stream *io)
+{
+ struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod);
/* enable PIO IRQ */
ssi->cr_etc = UIEN | OIEN | DIEN;
- /* enable PIO interrupt if gen2 */
- if (rsnd_is_gen2(priv))
- rsnd_mod_write(&ssi->mod, INT_ENABLE, 0x0f000000);
+ rsnd_src_enable_ssi_irq(mod, rdai, io);
rsnd_ssi_hw_start(ssi, rdai, io);
- dev_dbg(dev, "%s.%d start\n", rsnd_mod_name(mod), rsnd_mod_id(mod));
-
return 0;
}
@@ -470,12 +398,8 @@ static int rsnd_ssi_pio_stop(struct rsnd_mod *mod,
struct rsnd_dai *rdai,
struct rsnd_dai_stream *io)
{
- struct rsnd_priv *priv = rsnd_mod_to_priv(mod);
- struct device *dev = rsnd_priv_to_dev(priv);
struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod);
- dev_dbg(dev, "%s.%d stop\n", rsnd_mod_name(mod), rsnd_mod_id(mod));
-
ssi->cr_etc = 0;
rsnd_ssi_hw_stop(ssi, rdai);
@@ -485,35 +409,46 @@ static int rsnd_ssi_pio_stop(struct rsnd_mod *mod,
static struct rsnd_mod_ops rsnd_ssi_pio_ops = {
.name = "ssi (pio)",
+ .probe = rsnd_ssi_pio_probe,
.init = rsnd_ssi_init,
.quit = rsnd_ssi_quit,
.start = rsnd_ssi_pio_start,
.stop = rsnd_ssi_pio_stop,
};
-static int rsnd_ssi_dma_inquiry(struct rsnd_dma *dma, dma_addr_t *buf, int *len)
+static int rsnd_ssi_dma_probe(struct rsnd_mod *mod,
+ struct rsnd_dai *rdai,
+ struct rsnd_dai_stream *io)
{
- struct rsnd_ssi *ssi = rsnd_dma_to_ssi(dma);
- struct rsnd_dai_stream *io = ssi->io;
- struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io);
+ struct rsnd_priv *priv = rsnd_mod_to_priv(mod);
+ struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod);
+ struct rcar_snd_info *info = rsnd_priv_to_info(priv);
+ struct device *dev = rsnd_priv_to_dev(priv);
+ int dma_id = ssi->info->dma_id;
+ int is_play;
+ int ret;
- *len = io->byte_per_period;
- *buf = runtime->dma_addr +
- rsnd_dai_pointer_offset(io, ssi->dma_offset + *len);
- ssi->dma_offset = *len; /* it cares A/B plane */
+ if (info->dai_info)
+ is_play = rsnd_info_is_playback(priv, ssi);
+ else
+ is_play = rsnd_ssi_is_play(&ssi->mod);
- return 0;
-}
+ ret = rsnd_dma_init(
+ priv, rsnd_mod_to_dma(mod),
+ is_play,
+ dma_id);
-static int rsnd_ssi_dma_complete(struct rsnd_dma *dma)
-{
- struct rsnd_ssi *ssi = rsnd_dma_to_ssi(dma);
- struct rsnd_dai_stream *io = ssi->io;
- u32 status = rsnd_mod_read(&ssi->mod, SSISR);
+ if (ret < 0)
+ dev_err(dev, "SSI DMA failed\n");
- rsnd_ssi_record_error(ssi, status);
+ return ret;
+}
- rsnd_dai_pointer_update(ssi->io, io->byte_per_period);
+static int rsnd_ssi_dma_remove(struct rsnd_mod *mod,
+ struct rsnd_dai *rdai,
+ struct rsnd_dai_stream *io)
+{
+ rsnd_dma_quit(rsnd_mod_to_priv(mod), rsnd_mod_to_dma(mod));
return 0;
}
@@ -527,14 +462,13 @@ static int rsnd_ssi_dma_start(struct rsnd_mod *mod,
/* enable DMA transfer */
ssi->cr_etc = DMEN;
- ssi->dma_offset = 0;
rsnd_dma_start(dma);
rsnd_ssi_hw_start(ssi, ssi->rdai, io);
/* enable WS continue */
- if (rsnd_rdai_is_clk_master(rdai))
+ if (rsnd_dai_is_clk_master(rdai))
rsnd_mod_write(&ssi->mod, SSIWSR, CONT);
return 0;
@@ -549,6 +483,8 @@ static int rsnd_ssi_dma_stop(struct rsnd_mod *mod,
ssi->cr_etc = 0;
+ rsnd_ssi_record_error(ssi, rsnd_mod_read(mod, SSISR));
+
rsnd_ssi_hw_stop(ssi, rdai);
rsnd_dma_stop(dma);
@@ -558,6 +494,8 @@ static int rsnd_ssi_dma_stop(struct rsnd_mod *mod,
static struct rsnd_mod_ops rsnd_ssi_dma_ops = {
.name = "ssi (dma)",
+ .probe = rsnd_ssi_dma_probe,
+ .remove = rsnd_ssi_dma_remove,
.init = rsnd_ssi_init,
.quit = rsnd_ssi_quit,
.start = rsnd_ssi_dma_start,
@@ -567,24 +505,8 @@ static struct rsnd_mod_ops rsnd_ssi_dma_ops = {
/*
* Non SSI
*/
-static int rsnd_ssi_non(struct rsnd_mod *mod,
- struct rsnd_dai *rdai,
- struct rsnd_dai_stream *io)
-{
- struct rsnd_priv *priv = rsnd_mod_to_priv(mod);
- struct device *dev = rsnd_priv_to_dev(priv);
-
- dev_dbg(dev, "%s\n", __func__);
-
- return 0;
-}
-
static struct rsnd_mod_ops rsnd_ssi_non_ops = {
.name = "ssi (non)",
- .init = rsnd_ssi_non,
- .quit = rsnd_ssi_non,
- .start = rsnd_ssi_non,
- .stop = rsnd_ssi_non,
};
/*
@@ -593,16 +515,30 @@ static struct rsnd_mod_ops rsnd_ssi_non_ops = {
struct rsnd_mod *rsnd_ssi_mod_get_frm_dai(struct rsnd_priv *priv,
int dai_id, int is_play)
{
+ struct rsnd_dai_platform_info *dai_info = NULL;
+ struct rsnd_dai_path_info *path_info = NULL;
+ struct rsnd_ssi_platform_info *target_info = NULL;
struct rsnd_ssi *ssi;
int i, has_play;
+ if (priv->rdai)
+ dai_info = priv->rdai[dai_id].info;
+ if (dai_info)
+ path_info = (is_play) ? &dai_info->playback : &dai_info->capture;
+ if (path_info)
+ target_info = path_info->ssi;
+
is_play = !!is_play;
for_each_rsnd_ssi(ssi, priv, i) {
+ if (target_info == ssi->info)
+ return &ssi->mod;
+
+ /* for compatible */
if (rsnd_ssi_dai_id(ssi) != dai_id)
continue;
- has_play = !!(rsnd_ssi_mode_flags(ssi) & RSND_SSI_PLAY);
+ has_play = rsnd_ssi_is_play(&ssi->mod);
if (is_play == has_play)
return &ssi->mod;
@@ -616,36 +552,122 @@ struct rsnd_mod *rsnd_ssi_mod_get(struct rsnd_priv *priv, int id)
if (WARN_ON(id < 0 || id >= rsnd_ssi_nr(priv)))
id = 0;
- return &(((struct rsnd_ssiu *)(priv->ssiu))->ssi + id)->mod;
+ return &((struct rsnd_ssi *)(priv->ssi) + id)->mod;
+}
+
+int rsnd_ssi_is_pin_sharing(struct rsnd_mod *mod)
+{
+ struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod);
+
+ return !!(rsnd_ssi_mode_flags(ssi) & RSND_SSI_CLK_PIN_SHARE);
+}
+
+int rsnd_ssi_is_play(struct rsnd_mod *mod)
+{
+ struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod);
+
+ return !!(rsnd_ssi_mode_flags(ssi) & RSND_SSI_PLAY);
+}
+
+static void rsnd_ssi_parent_clk_setup(struct rsnd_priv *priv, struct rsnd_ssi *ssi)
+{
+ if (!rsnd_ssi_is_pin_sharing(&ssi->mod))
+ return;
+
+ switch (rsnd_mod_id(&ssi->mod)) {
+ case 1:
+ case 2:
+ ssi->parent = rsnd_mod_to_ssi(rsnd_ssi_mod_get(priv, 0));
+ break;
+ case 4:
+ ssi->parent = rsnd_mod_to_ssi(rsnd_ssi_mod_get(priv, 3));
+ break;
+ case 8:
+ ssi->parent = rsnd_mod_to_ssi(rsnd_ssi_mod_get(priv, 7));
+ break;
+ }
+}
+
+
+static void rsnd_of_parse_ssi(struct platform_device *pdev,
+ const struct rsnd_of_data *of_data,
+ struct rsnd_priv *priv)
+{
+ struct device_node *node;
+ struct device_node *np;
+ struct rsnd_ssi_platform_info *ssi_info;
+ struct rcar_snd_info *info = rsnd_priv_to_info(priv);
+ struct device *dev = &pdev->dev;
+ int nr, i;
+
+ if (!of_data)
+ return;
+
+ node = of_get_child_by_name(dev->of_node, "rcar_sound,ssi");
+ if (!node)
+ return;
+
+ nr = of_get_child_count(node);
+ if (!nr)
+ return;
+
+ ssi_info = devm_kzalloc(dev,
+ sizeof(struct rsnd_ssi_platform_info) * nr,
+ GFP_KERNEL);
+ if (!ssi_info) {
+ dev_err(dev, "ssi info allocation error\n");
+ return;
+ }
+
+ info->ssi_info = ssi_info;
+ info->ssi_info_nr = nr;
+
+ i = -1;
+ for_each_child_of_node(node, np) {
+ i++;
+
+ ssi_info = info->ssi_info + i;
+
+ /*
+ * pin settings
+ */
+ if (of_get_property(np, "shared-pin", NULL))
+ ssi_info->flags |= RSND_SSI_CLK_PIN_SHARE;
+
+ /*
+ * irq
+ */
+ ssi_info->pio_irq = irq_of_parse_and_map(np, 0);
+ }
}
int rsnd_ssi_probe(struct platform_device *pdev,
- struct rcar_snd_info *info,
+ const struct rsnd_of_data *of_data,
struct rsnd_priv *priv)
{
+ struct rcar_snd_info *info = rsnd_priv_to_info(priv);
struct rsnd_ssi_platform_info *pinfo;
struct device *dev = rsnd_priv_to_dev(priv);
struct rsnd_mod_ops *ops;
struct clk *clk;
- struct rsnd_ssiu *ssiu;
struct rsnd_ssi *ssi;
char name[RSND_SSI_NAME_SIZE];
- int i, nr, ret;
+ int i, nr;
+
+ rsnd_of_parse_ssi(pdev, of_data, priv);
/*
* init SSI
*/
nr = info->ssi_info_nr;
- ssiu = devm_kzalloc(dev, sizeof(*ssiu) + (sizeof(*ssi) * nr),
- GFP_KERNEL);
- if (!ssiu) {
+ ssi = devm_kzalloc(dev, sizeof(*ssi) * nr, GFP_KERNEL);
+ if (!ssi) {
dev_err(dev, "SSI allocate failed\n");
return -ENOMEM;
}
- priv->ssiu = ssiu;
- ssiu->ssi = (struct rsnd_ssi *)(ssiu + 1);
- ssiu->ssi_nr = nr;
+ priv->ssi = ssi;
+ priv->ssi_nr = nr;
for_each_rsnd_ssi(ssi, priv, i) {
pinfo = &info->ssi_info[i];
@@ -660,61 +682,15 @@ int rsnd_ssi_probe(struct platform_device *pdev,
ssi->clk = clk;
ops = &rsnd_ssi_non_ops;
+ if (pinfo->dma_id > 0)
+ ops = &rsnd_ssi_dma_ops;
+ else if (rsnd_ssi_pio_available(ssi))
+ ops = &rsnd_ssi_pio_ops;
- /*
- * SSI DMA case
- */
- if (pinfo->dma_id > 0) {
- ret = rsnd_dma_init(
- priv, rsnd_mod_to_dma(&ssi->mod),
- (rsnd_ssi_mode_flags(ssi) & RSND_SSI_PLAY),
- pinfo->dma_id,
- rsnd_ssi_dma_inquiry,
- rsnd_ssi_dma_complete);
- if (ret < 0)
- dev_info(dev, "SSI DMA failed. try PIO transter\n");
- else
- ops = &rsnd_ssi_dma_ops;
-
- dev_dbg(dev, "SSI%d use DMA transfer\n", i);
- }
-
- /*
- * SSI PIO case
- */
- if (!rsnd_ssi_dma_available(ssi) &&
- rsnd_ssi_pio_available(ssi)) {
- ret = devm_request_irq(dev, pinfo->pio_irq,
- &rsnd_ssi_pio_interrupt,
- IRQF_SHARED,
- dev_name(dev), ssi);
- if (ret) {
- dev_err(dev, "SSI request interrupt failed\n");
- return ret;
- }
-
- ops = &rsnd_ssi_pio_ops;
-
- dev_dbg(dev, "SSI%d use PIO transfer\n", i);
- }
+ rsnd_mod_init(priv, &ssi->mod, ops, RSND_MOD_SSI, i);
- rsnd_mod_init(priv, &ssi->mod, ops, i);
+ rsnd_ssi_parent_clk_setup(priv, ssi);
}
- dev_dbg(dev, "ssi probed\n");
-
return 0;
}
-
-void rsnd_ssi_remove(struct platform_device *pdev,
- struct rsnd_priv *priv)
-{
- struct rsnd_ssi *ssi;
- int i;
-
- for_each_rsnd_ssi(ssi, priv, i) {
- if (rsnd_ssi_dma_available(ssi))
- rsnd_dma_quit(priv, rsnd_mod_to_dma(&ssi->mod));
- }
-
-}
diff --git a/sound/soc/sirf/Kconfig b/sound/soc/sirf/Kconfig
new file mode 100644
index 0000000..89e8942
--- /dev/null
+++ b/sound/soc/sirf/Kconfig
@@ -0,0 +1,14 @@
+config SND_SOC_SIRF
+ tristate "SoC Audio for the SiRF SoC chips"
+ depends on ARCH_SIRF || COMPILE_TEST
+ select SND_SOC_GENERIC_DMAENGINE_PCM
+
+config SND_SOC_SIRF_AUDIO
+ tristate "SoC Audio support for SiRF internal audio codec"
+ depends on SND_SOC_SIRF
+ select SND_SOC_SIRF_AUDIO_CODEC
+ select SND_SOC_SIRF_AUDIO_PORT
+
+config SND_SOC_SIRF_AUDIO_PORT
+ select REGMAP_MMIO
+ tristate
diff --git a/sound/soc/sirf/Makefile b/sound/soc/sirf/Makefile
new file mode 100644
index 0000000..913b932
--- /dev/null
+++ b/sound/soc/sirf/Makefile
@@ -0,0 +1,5 @@
+snd-soc-sirf-audio-objs := sirf-audio.o
+snd-soc-sirf-audio-port-objs := sirf-audio-port.o
+
+obj-$(CONFIG_SND_SOC_SIRF_AUDIO) += snd-soc-sirf-audio.o
+obj-$(CONFIG_SND_SOC_SIRF_AUDIO_PORT) += snd-soc-sirf-audio-port.o
diff --git a/sound/soc/sirf/sirf-audio-port.c b/sound/soc/sirf/sirf-audio-port.c
new file mode 100644
index 0000000..b04a53f
--- /dev/null
+++ b/sound/soc/sirf/sirf-audio-port.c
@@ -0,0 +1,194 @@
+/*
+ * SiRF Audio port driver
+ *
+ * Copyright (c) 2011 Cambridge Silicon Radio Limited, a CSR plc group company.
+ *
+ * Licensed under GPLv2 or later.
+ */
+#include <linux/module.h>
+#include <linux/io.h>
+#include <linux/regmap.h>
+#include <sound/soc.h>
+#include <sound/dmaengine_pcm.h>
+
+#include "sirf-audio-port.h"
+
+struct sirf_audio_port {
+ struct regmap *regmap;
+ struct snd_dmaengine_dai_dma_data playback_dma_data;
+ struct snd_dmaengine_dai_dma_data capture_dma_data;
+};
+
+static void sirf_audio_port_tx_enable(struct sirf_audio_port *port)
+{
+ regmap_update_bits(port->regmap, AUDIO_PORT_IC_TXFIFO_OP,
+ AUDIO_FIFO_RESET, AUDIO_FIFO_RESET);
+ regmap_write(port->regmap, AUDIO_PORT_IC_TXFIFO_INT_MSK, 0);
+ regmap_write(port->regmap, AUDIO_PORT_IC_TXFIFO_OP, 0);
+ regmap_update_bits(port->regmap, AUDIO_PORT_IC_TXFIFO_OP,
+ AUDIO_FIFO_START, AUDIO_FIFO_START);
+ regmap_update_bits(port->regmap, AUDIO_PORT_IC_CODEC_TX_CTRL,
+ IC_TX_ENABLE, IC_TX_ENABLE);
+}
+
+static void sirf_audio_port_tx_disable(struct sirf_audio_port *port)
+{
+ regmap_write(port->regmap, AUDIO_PORT_IC_TXFIFO_OP, 0);
+ regmap_update_bits(port->regmap, AUDIO_PORT_IC_CODEC_TX_CTRL,
+ IC_TX_ENABLE, ~IC_TX_ENABLE);
+}
+
+static void sirf_audio_port_rx_enable(struct sirf_audio_port *port,
+ int channels)
+{
+ regmap_update_bits(port->regmap, AUDIO_PORT_IC_RXFIFO_OP,
+ AUDIO_FIFO_RESET, AUDIO_FIFO_RESET);
+ regmap_write(port->regmap, AUDIO_PORT_IC_RXFIFO_INT_MSK, 0);
+ regmap_write(port->regmap, AUDIO_PORT_IC_RXFIFO_OP, 0);
+ regmap_update_bits(port->regmap, AUDIO_PORT_IC_RXFIFO_OP,
+ AUDIO_FIFO_START, AUDIO_FIFO_START);
+ if (channels == 1)
+ regmap_update_bits(port->regmap, AUDIO_PORT_IC_CODEC_RX_CTRL,
+ IC_RX_ENABLE_MONO, IC_RX_ENABLE_MONO);
+ else
+ regmap_update_bits(port->regmap, AUDIO_PORT_IC_CODEC_RX_CTRL,
+ IC_RX_ENABLE_STEREO, IC_RX_ENABLE_STEREO);
+}
+
+static void sirf_audio_port_rx_disable(struct sirf_audio_port *port)
+{
+ regmap_update_bits(port->regmap, AUDIO_PORT_IC_CODEC_RX_CTRL,
+ IC_RX_ENABLE_STEREO, ~IC_RX_ENABLE_STEREO);
+}
+
+static int sirf_audio_port_dai_probe(struct snd_soc_dai *dai)
+{
+ struct sirf_audio_port *port = snd_soc_dai_get_drvdata(dai);
+ snd_soc_dai_init_dma_data(dai, &port->playback_dma_data,
+ &port->capture_dma_data);
+ return 0;
+}
+
+static int sirf_audio_port_trigger(struct snd_pcm_substream *substream, int cmd,
+ struct snd_soc_dai *dai)
+{
+ struct sirf_audio_port *port = snd_soc_dai_get_drvdata(dai);
+ int playback = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ if (playback)
+ sirf_audio_port_tx_disable(port);
+ else
+ sirf_audio_port_rx_disable(port);
+ break;
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ if (playback)
+ sirf_audio_port_tx_enable(port);
+ else
+ sirf_audio_port_rx_enable(port,
+ substream->runtime->channels);
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static const struct snd_soc_dai_ops sirf_audio_port_dai_ops = {
+ .trigger = sirf_audio_port_trigger,
+};
+
+static struct snd_soc_dai_driver sirf_audio_port_dai = {
+ .probe = sirf_audio_port_dai_probe,
+ .name = "sirf-audio-port",
+ .id = 0,
+ .playback = {
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_48000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ },
+ .capture = {
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_48000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ },
+ .ops = &sirf_audio_port_dai_ops,
+};
+
+static const struct snd_soc_component_driver sirf_audio_port_component = {
+ .name = "sirf-audio-port",
+};
+
+static const struct regmap_config sirf_audio_port_regmap_config = {
+ .reg_bits = 32,
+ .reg_stride = 4,
+ .val_bits = 32,
+ .max_register = AUDIO_PORT_IC_RXFIFO_INT_MSK,
+ .cache_type = REGCACHE_NONE,
+};
+
+static int sirf_audio_port_probe(struct platform_device *pdev)
+{
+ int ret;
+ struct sirf_audio_port *port;
+ void __iomem *base;
+ struct resource *mem_res;
+
+ port = devm_kzalloc(&pdev->dev,
+ sizeof(struct sirf_audio_port), GFP_KERNEL);
+ if (!port)
+ return -ENOMEM;
+
+ mem_res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+ if (!mem_res) {
+ dev_err(&pdev->dev, "no mem resource?\n");
+ return -ENODEV;
+ }
+
+ base = devm_ioremap(&pdev->dev, mem_res->start,
+ resource_size(mem_res));
+ if (base == NULL)
+ return -ENOMEM;
+
+ port->regmap = devm_regmap_init_mmio(&pdev->dev, base,
+ &sirf_audio_port_regmap_config);
+ if (IS_ERR(port->regmap))
+ return PTR_ERR(port->regmap);
+
+ ret = devm_snd_soc_register_component(&pdev->dev,
+ &sirf_audio_port_component, &sirf_audio_port_dai, 1);
+ if (ret)
+ return ret;
+
+ platform_set_drvdata(pdev, port);
+ return devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0);
+}
+
+static const struct of_device_id sirf_audio_port_of_match[] = {
+ { .compatible = "sirf,audio-port", },
+ {}
+};
+MODULE_DEVICE_TABLE(of, sirf_audio_port_of_match);
+
+static struct platform_driver sirf_audio_port_driver = {
+ .driver = {
+ .name = "sirf-audio-port",
+ .owner = THIS_MODULE,
+ .of_match_table = sirf_audio_port_of_match,
+ },
+ .probe = sirf_audio_port_probe,
+};
+
+module_platform_driver(sirf_audio_port_driver);
+
+MODULE_DESCRIPTION("SiRF Audio Port driver");
+MODULE_AUTHOR("RongJun Ying <Rongjun.Ying@csr.com>");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/sirf/sirf-audio-port.h b/sound/soc/sirf/sirf-audio-port.h
new file mode 100644
index 0000000..f32dc54
--- /dev/null
+++ b/sound/soc/sirf/sirf-audio-port.h
@@ -0,0 +1,62 @@
+/*
+ * SiRF Audio port controllers define
+ *
+ * Copyright (c) 2011 Cambridge Silicon Radio Limited, a CSR plc group company.
+ *
+ * Licensed under GPLv2 or later.
+ */
+
+#ifndef _SIRF_AUDIO_PORT_H
+#define _SIRF_AUDIO_PORT_H
+
+#define AUDIO_PORT_TX_FIFO_LEVEL_CHECK_MASK 0x3F
+#define AUDIO_PORT_TX_FIFO_SC_OFFSET 0
+#define AUDIO_PORT_TX_FIFO_LC_OFFSET 10
+#define AUDIO_PORT_TX_FIFO_HC_OFFSET 20
+
+#define TX_FIFO_SC(x) (((x) & AUDIO_PORT_TX_FIFO_LEVEL_CHECK_MASK) \
+ << AUDIO_PORT_TX_FIFO_SC_OFFSET)
+#define TX_FIFO_LC(x) (((x) & AUDIO_PORT_TX_FIFO_LEVEL_CHECK_MASK) \
+ << AUDIO_PORT_TX_FIFO_LC_OFFSET)
+#define TX_FIFO_HC(x) (((x) & AUDIO_PORT_TX_FIFO_LEVEL_CHECK_MASK) \
+ << AUDIO_PORT_TX_FIFO_HC_OFFSET)
+
+#define AUDIO_PORT_RX_FIFO_LEVEL_CHECK_MASK 0x0F
+#define AUDIO_PORT_RX_FIFO_SC_OFFSET 0
+#define AUDIO_PORT_RX_FIFO_LC_OFFSET 10
+#define AUDIO_PORT_RX_FIFO_HC_OFFSET 20
+
+#define RX_FIFO_SC(x) (((x) & AUDIO_PORT_RX_FIFO_LEVEL_CHECK_MASK) \
+ << AUDIO_PORT_RX_FIFO_SC_OFFSET)
+#define RX_FIFO_LC(x) (((x) & AUDIO_PORT_RX_FIFO_LEVEL_CHECK_MASK) \
+ << AUDIO_PORT_RX_FIFO_LC_OFFSET)
+#define RX_FIFO_HC(x) (((x) & AUDIO_PORT_RX_FIFO_LEVEL_CHECK_MASK) \
+ << AUDIO_PORT_RX_FIFO_HC_OFFSET)
+#define AUDIO_PORT_IC_CODEC_TX_CTRL (0x00F4)
+#define AUDIO_PORT_IC_CODEC_RX_CTRL (0x00F8)
+
+#define AUDIO_PORT_IC_TXFIFO_OP (0x00FC)
+#define AUDIO_PORT_IC_TXFIFO_LEV_CHK (0x0100)
+#define AUDIO_PORT_IC_TXFIFO_STS (0x0104)
+#define AUDIO_PORT_IC_TXFIFO_INT (0x0108)
+#define AUDIO_PORT_IC_TXFIFO_INT_MSK (0x010C)
+
+#define AUDIO_PORT_IC_RXFIFO_OP (0x0110)
+#define AUDIO_PORT_IC_RXFIFO_LEV_CHK (0x0114)
+#define AUDIO_PORT_IC_RXFIFO_STS (0x0118)
+#define AUDIO_PORT_IC_RXFIFO_INT (0x011C)
+#define AUDIO_PORT_IC_RXFIFO_INT_MSK (0x0120)
+
+#define AUDIO_FIFO_START (1 << 0)
+#define AUDIO_FIFO_RESET (1 << 1)
+
+#define AUDIO_FIFO_FULL (1 << 0)
+#define AUDIO_FIFO_EMPTY (1 << 1)
+#define AUDIO_FIFO_OFLOW (1 << 2)
+#define AUDIO_FIFO_UFLOW (1 << 3)
+
+#define IC_TX_ENABLE (0x03)
+#define IC_RX_ENABLE_MONO (0x01)
+#define IC_RX_ENABLE_STEREO (0x03)
+
+#endif /*__SIRF_AUDIO_PORT_H*/
diff --git a/sound/soc/sirf/sirf-audio.c b/sound/soc/sirf/sirf-audio.c
new file mode 100644
index 0000000..ecef510
--- /dev/null
+++ b/sound/soc/sirf/sirf-audio.c
@@ -0,0 +1,156 @@
+/*
+ * SiRF audio card driver
+ *
+ * Copyright (c) 2011 Cambridge Silicon Radio Limited, a CSR plc group company.
+ *
+ * Licensed under GPLv2 or later.
+ */
+
+#include <linux/platform_device.h>
+#include <linux/module.h>
+#include <linux/of.h>
+#include <linux/gpio.h>
+#include <linux/of_gpio.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+
+struct sirf_audio_card {
+ unsigned int gpio_hp_pa;
+ unsigned int gpio_spk_pa;
+};
+
+static int sirf_audio_hp_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *ctrl, int event)
+{
+ struct snd_soc_dapm_context *dapm = w->dapm;
+ struct snd_soc_card *card = dapm->card;
+ struct sirf_audio_card *sirf_audio_card = snd_soc_card_get_drvdata(card);
+ int on = !SND_SOC_DAPM_EVENT_OFF(event);
+ if (gpio_is_valid(sirf_audio_card->gpio_hp_pa))
+ gpio_set_value(sirf_audio_card->gpio_hp_pa, on);
+ return 0;
+}
+
+static int sirf_audio_spk_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *ctrl, int event)
+{
+ struct snd_soc_dapm_context *dapm = w->dapm;
+ struct snd_soc_card *card = dapm->card;
+ struct sirf_audio_card *sirf_audio_card = snd_soc_card_get_drvdata(card);
+ int on = !SND_SOC_DAPM_EVENT_OFF(event);
+
+ if (gpio_is_valid(sirf_audio_card->gpio_spk_pa))
+ gpio_set_value(sirf_audio_card->gpio_spk_pa, on);
+
+ return 0;
+}
+static const struct snd_soc_dapm_widget sirf_audio_dapm_widgets[] = {
+ SND_SOC_DAPM_HP("Hp", sirf_audio_hp_event),
+ SND_SOC_DAPM_SPK("Ext Spk", sirf_audio_spk_event),
+ SND_SOC_DAPM_MIC("Ext Mic", NULL),
+};
+
+static const struct snd_soc_dapm_route intercon[] = {
+ {"Hp", NULL, "HPOUTL"},
+ {"Hp", NULL, "HPOUTR"},
+ {"Ext Spk", NULL, "SPKOUT"},
+ {"MICIN1", NULL, "Mic Bias"},
+ {"Mic Bias", NULL, "Ext Mic"},
+};
+
+/* Digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link sirf_audio_dai_link[] = {
+ {
+ .name = "SiRF audio card",
+ .stream_name = "SiRF audio HiFi",
+ .codec_dai_name = "sirf-audio-codec",
+ },
+};
+
+/* Audio machine driver */
+static struct snd_soc_card snd_soc_sirf_audio_card = {
+ .name = "SiRF audio card",
+ .owner = THIS_MODULE,
+ .dai_link = sirf_audio_dai_link,
+ .num_links = ARRAY_SIZE(sirf_audio_dai_link),
+ .dapm_widgets = sirf_audio_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(sirf_audio_dapm_widgets),
+ .dapm_routes = intercon,
+ .num_dapm_routes = ARRAY_SIZE(intercon),
+};
+
+static int sirf_audio_probe(struct platform_device *pdev)
+{
+ struct snd_soc_card *card = &snd_soc_sirf_audio_card;
+ struct sirf_audio_card *sirf_audio_card;
+ int ret;
+
+ sirf_audio_card = devm_kzalloc(&pdev->dev, sizeof(struct sirf_audio_card),
+ GFP_KERNEL);
+ if (sirf_audio_card == NULL)
+ return -ENOMEM;
+
+ sirf_audio_dai_link[0].cpu_of_node =
+ of_parse_phandle(pdev->dev.of_node, "sirf,audio-platform", 0);
+ sirf_audio_dai_link[0].platform_of_node =
+ of_parse_phandle(pdev->dev.of_node, "sirf,audio-platform", 0);
+ sirf_audio_dai_link[0].codec_of_node =
+ of_parse_phandle(pdev->dev.of_node, "sirf,audio-codec", 0);
+ sirf_audio_card->gpio_spk_pa = of_get_named_gpio(pdev->dev.of_node,
+ "spk-pa-gpios", 0);
+ sirf_audio_card->gpio_hp_pa = of_get_named_gpio(pdev->dev.of_node,
+ "hp-pa-gpios", 0);
+ if (gpio_is_valid(sirf_audio_card->gpio_spk_pa)) {
+ ret = devm_gpio_request_one(&pdev->dev,
+ sirf_audio_card->gpio_spk_pa,
+ GPIOF_OUT_INIT_LOW, "SPA_PA_SD");
+ if (ret) {
+ dev_err(&pdev->dev,
+ "Failed to request GPIO_%d for reset: %d\n",
+ sirf_audio_card->gpio_spk_pa, ret);
+ return ret;
+ }
+ }
+ if (gpio_is_valid(sirf_audio_card->gpio_hp_pa)) {
+ ret = devm_gpio_request_one(&pdev->dev,
+ sirf_audio_card->gpio_hp_pa,
+ GPIOF_OUT_INIT_LOW, "HP_PA_SD");
+ if (ret) {
+ dev_err(&pdev->dev,
+ "Failed to request GPIO_%d for reset: %d\n",
+ sirf_audio_card->gpio_hp_pa, ret);
+ return ret;
+ }
+ }
+
+ card->dev = &pdev->dev;
+ snd_soc_card_set_drvdata(card, sirf_audio_card);
+
+ ret = devm_snd_soc_register_card(&pdev->dev, card);
+ if (ret)
+ dev_err(&pdev->dev, "snd_soc_register_card() failed:%d\n", ret);
+
+ return ret;
+}
+
+static const struct of_device_id sirf_audio_of_match[] = {
+ {.compatible = "sirf,sirf-audio-card", },
+ { },
+};
+MODULE_DEVICE_TABLE(of, sirf_audio_of_match);
+
+static struct platform_driver sirf_audio_driver = {
+ .driver = {
+ .name = "sirf-audio-card",
+ .owner = THIS_MODULE,
+ .pm = &snd_soc_pm_ops,
+ .of_match_table = sirf_audio_of_match,
+ },
+ .probe = sirf_audio_probe,
+};
+module_platform_driver(sirf_audio_driver);
+
+MODULE_AUTHOR("RongJun Ying <RongJun.Ying@csr.com>");
+MODULE_DESCRIPTION("ALSA SoC SIRF audio card driver");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c
index 375dc6d..bfed3e4 100644
--- a/sound/soc/soc-cache.c
+++ b/sound/soc/soc-cache.c
@@ -96,8 +96,7 @@ int snd_soc_cache_exit(struct snd_soc_codec *codec)
{
dev_dbg(codec->dev, "ASoC: Destroying cache for %s codec\n",
codec->name);
- if (!codec->reg_cache)
- return 0;
+
kfree(codec->reg_cache);
codec->reg_cache = NULL;
return 0;
@@ -117,8 +116,9 @@ int snd_soc_cache_read(struct snd_soc_codec *codec,
return -EINVAL;
mutex_lock(&codec->cache_rw_mutex);
- *value = snd_soc_get_cache_val(codec->reg_cache, reg,
- codec->driver->reg_word_size);
+ if (!ZERO_OR_NULL_PTR(codec->reg_cache))
+ *value = snd_soc_get_cache_val(codec->reg_cache, reg,
+ codec->driver->reg_word_size);
mutex_unlock(&codec->cache_rw_mutex);
return 0;
@@ -136,8 +136,9 @@ int snd_soc_cache_write(struct snd_soc_codec *codec,
unsigned int reg, unsigned int value)
{
mutex_lock(&codec->cache_rw_mutex);
- snd_soc_set_cache_val(codec->reg_cache, reg, value,
- codec->driver->reg_word_size);
+ if (!ZERO_OR_NULL_PTR(codec->reg_cache))
+ snd_soc_set_cache_val(codec->reg_cache, reg, value,
+ codec->driver->reg_word_size);
mutex_unlock(&codec->cache_rw_mutex);
return 0;
diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c
index 5e9690c..91083e6 100644
--- a/sound/soc/soc-compress.c
+++ b/sound/soc/soc-compress.c
@@ -30,8 +30,6 @@ static int soc_compr_open(struct snd_compr_stream *cstream)
{
struct snd_soc_pcm_runtime *rtd = cstream->private_data;
struct snd_soc_platform *platform = rtd->platform;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
int ret = 0;
mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass);
@@ -52,17 +50,7 @@ static int soc_compr_open(struct snd_compr_stream *cstream)
}
}
- if (cstream->direction == SND_COMPRESS_PLAYBACK) {
- cpu_dai->playback_active++;
- codec_dai->playback_active++;
- } else {
- cpu_dai->capture_active++;
- codec_dai->capture_active++;
- }
-
- cpu_dai->active++;
- codec_dai->active++;
- rtd->codec->active++;
+ snd_soc_runtime_activate(rtd, cstream->direction);
mutex_unlock(&rtd->pcm_mutex);
@@ -81,8 +69,6 @@ static int soc_compr_open_fe(struct snd_compr_stream *cstream)
struct snd_soc_pcm_runtime *fe = cstream->private_data;
struct snd_pcm_substream *fe_substream = fe->pcm->streams[0].substream;
struct snd_soc_platform *platform = fe->platform;
- struct snd_soc_dai *cpu_dai = fe->cpu_dai;
- struct snd_soc_dai *codec_dai = fe->codec_dai;
struct snd_soc_dpcm *dpcm;
struct snd_soc_dapm_widget_list *list;
int stream;
@@ -140,17 +126,7 @@ static int soc_compr_open_fe(struct snd_compr_stream *cstream)
fe->dpcm[stream].state = SND_SOC_DPCM_STATE_OPEN;
fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO;
- if (cstream->direction == SND_COMPRESS_PLAYBACK) {
- cpu_dai->playback_active++;
- codec_dai->playback_active++;
- } else {
- cpu_dai->capture_active++;
- codec_dai->capture_active++;
- }
-
- cpu_dai->active++;
- codec_dai->active++;
- fe->codec->active++;
+ snd_soc_runtime_activate(fe, stream);
mutex_unlock(&fe->card->mutex);
@@ -202,23 +178,18 @@ static int soc_compr_free(struct snd_compr_stream *cstream)
struct snd_soc_platform *platform = rtd->platform;
struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
struct snd_soc_dai *codec_dai = rtd->codec_dai;
- struct snd_soc_codec *codec = rtd->codec;
+ int stream;
mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass);
- if (cstream->direction == SND_COMPRESS_PLAYBACK) {
- cpu_dai->playback_active--;
- codec_dai->playback_active--;
- } else {
- cpu_dai->capture_active--;
- codec_dai->capture_active--;
- }
+ if (cstream->direction == SND_COMPRESS_PLAYBACK)
+ stream = SNDRV_PCM_STREAM_PLAYBACK;
+ else
+ stream = SNDRV_PCM_STREAM_CAPTURE;
- snd_soc_dai_digital_mute(codec_dai, 1, cstream->direction);
+ snd_soc_runtime_deactivate(rtd, stream);
- cpu_dai->active--;
- codec_dai->active--;
- codec->active--;
+ snd_soc_dai_digital_mute(codec_dai, 1, cstream->direction);
if (!cpu_dai->active)
cpu_dai->rate = 0;
@@ -235,8 +206,7 @@ static int soc_compr_free(struct snd_compr_stream *cstream)
cpu_dai->runtime = NULL;
if (cstream->direction == SND_COMPRESS_PLAYBACK) {
- if (!rtd->pmdown_time || codec->ignore_pmdown_time ||
- rtd->dai_link->ignore_pmdown_time) {
+ if (snd_soc_runtime_ignore_pmdown_time(rtd)) {
snd_soc_dapm_stream_event(rtd,
SNDRV_PCM_STREAM_PLAYBACK,
SND_SOC_DAPM_STREAM_STOP);
@@ -261,26 +231,17 @@ static int soc_compr_free_fe(struct snd_compr_stream *cstream)
{
struct snd_soc_pcm_runtime *fe = cstream->private_data;
struct snd_soc_platform *platform = fe->platform;
- struct snd_soc_dai *cpu_dai = fe->cpu_dai;
- struct snd_soc_dai *codec_dai = fe->codec_dai;
struct snd_soc_dpcm *dpcm;
int stream, ret;
mutex_lock_nested(&fe->card->mutex, SND_SOC_CARD_CLASS_RUNTIME);
- if (cstream->direction == SND_COMPRESS_PLAYBACK) {
+ if (cstream->direction == SND_COMPRESS_PLAYBACK)
stream = SNDRV_PCM_STREAM_PLAYBACK;
- cpu_dai->playback_active--;
- codec_dai->playback_active--;
- } else {
+ else
stream = SNDRV_PCM_STREAM_CAPTURE;
- cpu_dai->capture_active--;
- codec_dai->capture_active--;
- }
- cpu_dai->active--;
- codec_dai->active--;
- fe->codec->active--;
+ snd_soc_runtime_deactivate(fe, stream);
fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_FE;
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index fe1df50..b65ca10 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -56,7 +56,6 @@ EXPORT_SYMBOL_GPL(snd_soc_debugfs_root);
#endif
static DEFINE_MUTEX(client_mutex);
-static LIST_HEAD(dai_list);
static LIST_HEAD(platform_list);
static LIST_HEAD(codec_list);
static LIST_HEAD(component_list);
@@ -370,18 +369,22 @@ static ssize_t dai_list_read_file(struct file *file, char __user *user_buf,
{
char *buf = kmalloc(PAGE_SIZE, GFP_KERNEL);
ssize_t len, ret = 0;
+ struct snd_soc_component *component;
struct snd_soc_dai *dai;
if (!buf)
return -ENOMEM;
- list_for_each_entry(dai, &dai_list, list) {
- len = snprintf(buf + ret, PAGE_SIZE - ret, "%s\n", dai->name);
- if (len >= 0)
- ret += len;
- if (ret > PAGE_SIZE) {
- ret = PAGE_SIZE;
- break;
+ list_for_each_entry(component, &component_list, list) {
+ list_for_each_entry(dai, &component->dai_list, list) {
+ len = snprintf(buf + ret, PAGE_SIZE - ret, "%s\n",
+ dai->name);
+ if (len >= 0)
+ ret += len;
+ if (ret > PAGE_SIZE) {
+ ret = PAGE_SIZE;
+ break;
+ }
}
}
@@ -855,6 +858,7 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num)
{
struct snd_soc_dai_link *dai_link = &card->dai_link[num];
struct snd_soc_pcm_runtime *rtd = &card->rtd[num];
+ struct snd_soc_component *component;
struct snd_soc_codec *codec;
struct snd_soc_platform *platform;
struct snd_soc_dai *codec_dai, *cpu_dai;
@@ -863,18 +867,20 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num)
dev_dbg(card->dev, "ASoC: binding %s at idx %d\n", dai_link->name, num);
/* Find CPU DAI from registered DAIs*/
- list_for_each_entry(cpu_dai, &dai_list, list) {
+ list_for_each_entry(component, &component_list, list) {
if (dai_link->cpu_of_node &&
- (cpu_dai->dev->of_node != dai_link->cpu_of_node))
+ component->dev->of_node != dai_link->cpu_of_node)
continue;
if (dai_link->cpu_name &&
- strcmp(dev_name(cpu_dai->dev), dai_link->cpu_name))
- continue;
- if (dai_link->cpu_dai_name &&
- strcmp(cpu_dai->name, dai_link->cpu_dai_name))
+ strcmp(dev_name(component->dev), dai_link->cpu_name))
continue;
+ list_for_each_entry(cpu_dai, &component->dai_list, list) {
+ if (dai_link->cpu_dai_name &&
+ strcmp(cpu_dai->name, dai_link->cpu_dai_name))
+ continue;
- rtd->cpu_dai = cpu_dai;
+ rtd->cpu_dai = cpu_dai;
+ }
}
if (!rtd->cpu_dai) {
@@ -899,12 +905,10 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num)
* CODEC found, so find CODEC DAI from registered DAIs from
* this CODEC
*/
- list_for_each_entry(codec_dai, &dai_list, list) {
- if (codec->dev == codec_dai->dev &&
- !strcmp(codec_dai->name,
- dai_link->codec_dai_name)) {
-
+ list_for_each_entry(codec_dai, &codec->component.dai_list, list) {
+ if (!strcmp(codec_dai->name, dai_link->codec_dai_name)) {
rtd->codec_dai = codec_dai;
+ break;
}
}
@@ -1128,15 +1132,21 @@ static int soc_probe_codec(struct snd_soc_card *card,
driver->num_dapm_widgets);
/* Create DAPM widgets for each DAI stream */
- list_for_each_entry(dai, &dai_list, list) {
- if (dai->dev != codec->dev)
- continue;
-
+ list_for_each_entry(dai, &codec->component.dai_list, list)
snd_soc_dapm_new_dai_widgets(&codec->dapm, dai);
- }
codec->dapm.idle_bias_off = driver->idle_bias_off;
+ if (!codec->write && dev_get_regmap(codec->dev, NULL)) {
+ /* Set the default I/O up try regmap */
+ ret = snd_soc_codec_set_cache_io(codec, NULL);
+ if (ret < 0) {
+ dev_err(codec->dev,
+ "Failed to set cache I/O: %d\n", ret);
+ goto err_probe;
+ }
+ }
+
if (driver->probe) {
ret = driver->probe(codec);
if (ret < 0) {
@@ -1150,10 +1160,6 @@ static int soc_probe_codec(struct snd_soc_card *card,
codec->name);
}
- /* If the driver didn't set I/O up try regmap */
- if (!codec->write && dev_get_regmap(codec->dev, NULL))
- snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP);
-
if (driver->controls)
snd_soc_add_codec_controls(codec, driver->controls,
driver->num_controls);
@@ -1180,6 +1186,7 @@ static int soc_probe_platform(struct snd_soc_card *card,
{
int ret = 0;
const struct snd_soc_platform_driver *driver = platform->driver;
+ struct snd_soc_component *component;
struct snd_soc_dai *dai;
platform->card = card;
@@ -1195,11 +1202,11 @@ static int soc_probe_platform(struct snd_soc_card *card,
driver->dapm_widgets, driver->num_dapm_widgets);
/* Create DAPM widgets for each DAI stream */
- list_for_each_entry(dai, &dai_list, list) {
- if (dai->dev != platform->dev)
+ list_for_each_entry(component, &component_list, list) {
+ if (component->dev != platform->dev)
continue;
-
- snd_soc_dapm_new_dai_widgets(&platform->dapm, dai);
+ list_for_each_entry(dai, &component->dai_list, list)
+ snd_soc_dapm_new_dai_widgets(&platform->dapm, dai);
}
platform->dapm.idle_bias_off = 1;
@@ -1654,7 +1661,7 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card)
}
/* card bind complete so register a sound card */
- ret = snd_card_create(SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1,
+ ret = snd_card_new(card->dev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1,
card->owner, 0, &card->snd_card);
if (ret < 0) {
dev_err(card->dev,
@@ -1662,7 +1669,6 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card)
card->name, ret);
goto base_error;
}
- card->snd_card->dev = card->dev;
card->dapm.bias_level = SND_SOC_BIAS_OFF;
card->dapm.dev = card->dev;
@@ -2571,10 +2577,10 @@ int snd_soc_info_enum_double(struct snd_kcontrol *kcontrol,
uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
uinfo->count = e->shift_l == e->shift_r ? 1 : 2;
- uinfo->value.enumerated.items = e->max;
+ uinfo->value.enumerated.items = e->items;
- if (uinfo->value.enumerated.item > e->max - 1)
- uinfo->value.enumerated.item = e->max - 1;
+ if (uinfo->value.enumerated.item >= e->items)
+ uinfo->value.enumerated.item = e->items - 1;
strlcpy(uinfo->value.enumerated.name,
e->texts[uinfo->value.enumerated.item],
sizeof(uinfo->value.enumerated.name));
@@ -2596,14 +2602,18 @@ int snd_soc_get_enum_double(struct snd_kcontrol *kcontrol,
{
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
- unsigned int val;
+ unsigned int val, item;
+ unsigned int reg_val;
- val = snd_soc_read(codec, e->reg);
- ucontrol->value.enumerated.item[0]
- = (val >> e->shift_l) & e->mask;
- if (e->shift_l != e->shift_r)
- ucontrol->value.enumerated.item[1] =
- (val >> e->shift_r) & e->mask;
+ reg_val = snd_soc_read(codec, e->reg);
+ val = (reg_val >> e->shift_l) & e->mask;
+ item = snd_soc_enum_val_to_item(e, val);
+ ucontrol->value.enumerated.item[0] = item;
+ if (e->shift_l != e->shift_r) {
+ val = (reg_val >> e->shift_l) & e->mask;
+ item = snd_soc_enum_val_to_item(e, val);
+ ucontrol->value.enumerated.item[1] = item;
+ }
return 0;
}
@@ -2623,17 +2633,18 @@ int snd_soc_put_enum_double(struct snd_kcontrol *kcontrol,
{
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
+ unsigned int *item = ucontrol->value.enumerated.item;
unsigned int val;
unsigned int mask;
- if (ucontrol->value.enumerated.item[0] > e->max - 1)
+ if (item[0] >= e->items)
return -EINVAL;
- val = ucontrol->value.enumerated.item[0] << e->shift_l;
+ val = snd_soc_enum_item_to_val(e, item[0]) << e->shift_l;
mask = e->mask << e->shift_l;
if (e->shift_l != e->shift_r) {
- if (ucontrol->value.enumerated.item[1] > e->max - 1)
+ if (item[1] >= e->items)
return -EINVAL;
- val |= ucontrol->value.enumerated.item[1] << e->shift_r;
+ val |= snd_soc_enum_item_to_val(e, item[1]) << e->shift_r;
mask |= e->mask << e->shift_r;
}
@@ -2642,78 +2653,46 @@ int snd_soc_put_enum_double(struct snd_kcontrol *kcontrol,
EXPORT_SYMBOL_GPL(snd_soc_put_enum_double);
/**
- * snd_soc_get_value_enum_double - semi enumerated double mixer get callback
- * @kcontrol: mixer control
- * @ucontrol: control element information
+ * snd_soc_read_signed - Read a codec register and interprete as signed value
+ * @codec: codec
+ * @reg: Register to read
+ * @mask: Mask to use after shifting the register value
+ * @shift: Right shift of register value
+ * @sign_bit: Bit that describes if a number is negative or not.
*
- * Callback to get the value of a double semi enumerated mixer.
+ * This functions reads a codec register. The register value is shifted right
+ * by 'shift' bits and masked with the given 'mask'. Afterwards it translates
+ * the given registervalue into a signed integer if sign_bit is non-zero.
*
- * Semi enumerated mixer: the enumerated items are referred as values. Can be
- * used for handling bitfield coded enumeration for example.
- *
- * Returns 0 for success.
+ * Returns the register value as signed int.
*/
-int snd_soc_get_value_enum_double(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
+static int snd_soc_read_signed(struct snd_soc_codec *codec, unsigned int reg,
+ unsigned int mask, unsigned int shift, unsigned int sign_bit)
{
- struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
- struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
- unsigned int reg_val, val, mux;
+ int ret;
+ unsigned int val;
- reg_val = snd_soc_read(codec, e->reg);
- val = (reg_val >> e->shift_l) & e->mask;
- for (mux = 0; mux < e->max; mux++) {
- if (val == e->values[mux])
- break;
- }
- ucontrol->value.enumerated.item[0] = mux;
- if (e->shift_l != e->shift_r) {
- val = (reg_val >> e->shift_r) & e->mask;
- for (mux = 0; mux < e->max; mux++) {
- if (val == e->values[mux])
- break;
- }
- ucontrol->value.enumerated.item[1] = mux;
- }
+ val = (snd_soc_read(codec, reg) >> shift) & mask;
- return 0;
-}
-EXPORT_SYMBOL_GPL(snd_soc_get_value_enum_double);
+ if (!sign_bit)
+ return val;
-/**
- * snd_soc_put_value_enum_double - semi enumerated double mixer put callback
- * @kcontrol: mixer control
- * @ucontrol: control element information
- *
- * Callback to set the value of a double semi enumerated mixer.
- *
- * Semi enumerated mixer: the enumerated items are referred as values. Can be
- * used for handling bitfield coded enumeration for example.
- *
- * Returns 0 for success.
- */
-int snd_soc_put_value_enum_double(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
- struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
- unsigned int val;
- unsigned int mask;
+ /* non-negative number */
+ if (!(val & BIT(sign_bit)))
+ return val;
- if (ucontrol->value.enumerated.item[0] > e->max - 1)
- return -EINVAL;
- val = e->values[ucontrol->value.enumerated.item[0]] << e->shift_l;
- mask = e->mask << e->shift_l;
- if (e->shift_l != e->shift_r) {
- if (ucontrol->value.enumerated.item[1] > e->max - 1)
- return -EINVAL;
- val |= e->values[ucontrol->value.enumerated.item[1]] << e->shift_r;
- mask |= e->mask << e->shift_r;
- }
+ ret = val;
- return snd_soc_update_bits_locked(codec, e->reg, mask, val);
+ /*
+ * The register most probably does not contain a full-sized int.
+ * Instead we have an arbitrary number of bits in a signed
+ * representation which has to be translated into a full-sized int.
+ * This is done by filling up all bits above the sign-bit.
+ */
+ ret |= ~((int)(BIT(sign_bit) - 1));
+
+ return ret;
}
-EXPORT_SYMBOL_GPL(snd_soc_put_value_enum_double);
/**
* snd_soc_info_volsw - single mixer info callback
@@ -2743,7 +2722,7 @@ int snd_soc_info_volsw(struct snd_kcontrol *kcontrol,
uinfo->count = snd_soc_volsw_is_stereo(mc) ? 2 : 1;
uinfo->value.integer.min = 0;
- uinfo->value.integer.max = platform_max;
+ uinfo->value.integer.max = platform_max - mc->min;
return 0;
}
EXPORT_SYMBOL_GPL(snd_soc_info_volsw);
@@ -2769,11 +2748,16 @@ int snd_soc_get_volsw(struct snd_kcontrol *kcontrol,
unsigned int shift = mc->shift;
unsigned int rshift = mc->rshift;
int max = mc->max;
+ int min = mc->min;
+ int sign_bit = mc->sign_bit;
unsigned int mask = (1 << fls(max)) - 1;
unsigned int invert = mc->invert;
- ucontrol->value.integer.value[0] =
- (snd_soc_read(codec, reg) >> shift) & mask;
+ if (sign_bit)
+ mask = BIT(sign_bit + 1) - 1;
+
+ ucontrol->value.integer.value[0] = snd_soc_read_signed(codec, reg, mask,
+ shift, sign_bit) - min;
if (invert)
ucontrol->value.integer.value[0] =
max - ucontrol->value.integer.value[0];
@@ -2781,10 +2765,12 @@ int snd_soc_get_volsw(struct snd_kcontrol *kcontrol,
if (snd_soc_volsw_is_stereo(mc)) {
if (reg == reg2)
ucontrol->value.integer.value[1] =
- (snd_soc_read(codec, reg) >> rshift) & mask;
+ snd_soc_read_signed(codec, reg, mask, rshift,
+ sign_bit) - min;
else
ucontrol->value.integer.value[1] =
- (snd_soc_read(codec, reg2) >> shift) & mask;
+ snd_soc_read_signed(codec, reg2, mask, shift,
+ sign_bit) - min;
if (invert)
ucontrol->value.integer.value[1] =
max - ucontrol->value.integer.value[1];
@@ -2815,20 +2801,25 @@ int snd_soc_put_volsw(struct snd_kcontrol *kcontrol,
unsigned int shift = mc->shift;
unsigned int rshift = mc->rshift;
int max = mc->max;
+ int min = mc->min;
+ unsigned int sign_bit = mc->sign_bit;
unsigned int mask = (1 << fls(max)) - 1;
unsigned int invert = mc->invert;
int err;
- bool type_2r = 0;
+ bool type_2r = false;
unsigned int val2 = 0;
unsigned int val, val_mask;
- val = (ucontrol->value.integer.value[0] & mask);
+ if (sign_bit)
+ mask = BIT(sign_bit + 1) - 1;
+
+ val = ((ucontrol->value.integer.value[0] + min) & mask);
if (invert)
val = max - val;
val_mask = mask << shift;
val = val << shift;
if (snd_soc_volsw_is_stereo(mc)) {
- val2 = (ucontrol->value.integer.value[1] & mask);
+ val2 = ((ucontrol->value.integer.value[1] + min) & mask);
if (invert)
val2 = max - val2;
if (reg == reg2) {
@@ -2836,7 +2827,7 @@ int snd_soc_put_volsw(struct snd_kcontrol *kcontrol,
val |= val2 << rshift;
} else {
val2 = val2 << shift;
- type_2r = 1;
+ type_2r = true;
}
}
err = snd_soc_update_bits_locked(codec, reg, val_mask, val);
@@ -3234,7 +3225,7 @@ int snd_soc_bytes_put(struct snd_kcontrol *kcontrol,
struct soc_bytes *params = (void *)kcontrol->private_value;
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
int ret, len;
- unsigned int val;
+ unsigned int val, mask;
void *data;
if (!codec->using_regmap)
@@ -3264,12 +3255,36 @@ int snd_soc_bytes_put(struct snd_kcontrol *kcontrol,
((u8 *)data)[0] |= val;
break;
case 2:
- ((u16 *)data)[0] &= cpu_to_be16(~params->mask);
- ((u16 *)data)[0] |= cpu_to_be16(val);
+ mask = ~params->mask;
+ ret = regmap_parse_val(codec->control_data,
+ &mask, &mask);
+ if (ret != 0)
+ goto out;
+
+ ((u16 *)data)[0] &= mask;
+
+ ret = regmap_parse_val(codec->control_data,
+ &val, &val);
+ if (ret != 0)
+ goto out;
+
+ ((u16 *)data)[0] |= val;
break;
case 4:
- ((u32 *)data)[0] &= cpu_to_be32(~params->mask);
- ((u32 *)data)[0] |= cpu_to_be32(val);
+ mask = ~params->mask;
+ ret = regmap_parse_val(codec->control_data,
+ &mask, &mask);
+ if (ret != 0)
+ goto out;
+
+ ((u32 *)data)[0] &= mask;
+
+ ret = regmap_parse_val(codec->control_data,
+ &val, &val);
+ if (ret != 0)
+ goto out;
+
+ ((u32 *)data)[0] |= val;
break;
default:
ret = -EINVAL;
@@ -3609,6 +3624,30 @@ int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
EXPORT_SYMBOL_GPL(snd_soc_dai_set_fmt);
/**
+ * snd_soc_of_xlate_tdm_slot - generate tx/rx slot mask.
+ * @slots: Number of slots in use.
+ * @tx_mask: bitmask representing active TX slots.
+ * @rx_mask: bitmask representing active RX slots.
+ *
+ * Generates the TDM tx and rx slot default masks for DAI.
+ */
+static int snd_soc_of_xlate_tdm_slot_mask(unsigned int slots,
+ unsigned int *tx_mask,
+ unsigned int *rx_mask)
+{
+ if (*tx_mask || *rx_mask)
+ return 0;
+
+ if (!slots)
+ return -EINVAL;
+
+ *tx_mask = (1 << slots) - 1;
+ *rx_mask = (1 << slots) - 1;
+
+ return 0;
+}
+
+/**
* snd_soc_dai_set_tdm_slot - configure DAI TDM.
* @dai: DAI
* @tx_mask: bitmask representing active TX slots.
@@ -3622,11 +3661,17 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_fmt);
int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai,
unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width)
{
+ if (dai->driver && dai->driver->ops->of_xlate_tdm_slot_mask)
+ dai->driver->ops->of_xlate_tdm_slot_mask(slots,
+ &tx_mask, &rx_mask);
+ else
+ snd_soc_of_xlate_tdm_slot_mask(slots, &tx_mask, &rx_mask);
+
if (dai->driver && dai->driver->ops->set_tdm_slot)
return dai->driver->ops->set_tdm_slot(dai, tx_mask, rx_mask,
slots, slot_width);
else
- return -EINVAL;
+ return -ENOTSUPP;
}
EXPORT_SYMBOL_GPL(snd_soc_dai_set_tdm_slot);
@@ -3882,95 +3927,42 @@ static inline char *fmt_multiple_name(struct device *dev,
}
/**
- * snd_soc_register_dai - Register a DAI with the ASoC core
+ * snd_soc_unregister_dai - Unregister DAIs from the ASoC core
*
- * @dai: DAI to register
+ * @component: The component for which the DAIs should be unregistered
*/
-static int snd_soc_register_dai(struct device *dev,
- struct snd_soc_dai_driver *dai_drv)
+static void snd_soc_unregister_dais(struct snd_soc_component *component)
{
- struct snd_soc_codec *codec;
- struct snd_soc_dai *dai;
-
- dev_dbg(dev, "ASoC: dai register %s\n", dev_name(dev));
+ struct snd_soc_dai *dai, *_dai;
- dai = kzalloc(sizeof(struct snd_soc_dai), GFP_KERNEL);
- if (dai == NULL)
- return -ENOMEM;
-
- /* create DAI component name */
- dai->name = fmt_single_name(dev, &dai->id);
- if (dai->name == NULL) {
+ list_for_each_entry_safe(dai, _dai, &component->dai_list, list) {
+ dev_dbg(component->dev, "ASoC: Unregistered DAI '%s'\n",
+ dai->name);
+ list_del(&dai->list);
+ kfree(dai->name);
kfree(dai);
- return -ENOMEM;
}
-
- dai->dev = dev;
- dai->driver = dai_drv;
- dai->dapm.dev = dev;
- if (!dai->driver->ops)
- dai->driver->ops = &null_dai_ops;
-
- mutex_lock(&client_mutex);
-
- list_for_each_entry(codec, &codec_list, list) {
- if (codec->dev == dev) {
- dev_dbg(dev, "ASoC: Mapped DAI %s to CODEC %s\n",
- dai->name, codec->name);
- dai->codec = codec;
- break;
- }
- }
-
- if (!dai->codec)
- dai->dapm.idle_bias_off = 1;
-
- list_add(&dai->list, &dai_list);
-
- mutex_unlock(&client_mutex);
-
- dev_dbg(dev, "ASoC: Registered DAI '%s'\n", dai->name);
-
- return 0;
-}
-
-/**
- * snd_soc_unregister_dai - Unregister a DAI from the ASoC core
- *
- * @dai: DAI to unregister
- */
-static void snd_soc_unregister_dai(struct device *dev)
-{
- struct snd_soc_dai *dai;
-
- list_for_each_entry(dai, &dai_list, list) {
- if (dev == dai->dev)
- goto found;
- }
- return;
-
-found:
- mutex_lock(&client_mutex);
- list_del(&dai->list);
- mutex_unlock(&client_mutex);
-
- dev_dbg(dev, "ASoC: Unregistered DAI '%s'\n", dai->name);
- kfree(dai->name);
- kfree(dai);
}
/**
- * snd_soc_register_dais - Register multiple DAIs with the ASoC core
+ * snd_soc_register_dais - Register a DAI with the ASoC core
*
- * @dai: Array of DAIs to register
+ * @component: The component the DAIs are registered for
+ * @codec: The CODEC that the DAIs are registered for, NULL if the component is
+ * not a CODEC.
+ * @dai_drv: DAI driver to use for the DAIs
* @count: Number of DAIs
+ * @legacy_dai_naming: Use the legacy naming scheme and let the DAI inherit the
+ * parent's name.
*/
-static int snd_soc_register_dais(struct device *dev,
- struct snd_soc_dai_driver *dai_drv, size_t count)
+static int snd_soc_register_dais(struct snd_soc_component *component,
+ struct snd_soc_codec *codec, struct snd_soc_dai_driver *dai_drv,
+ size_t count, bool legacy_dai_naming)
{
- struct snd_soc_codec *codec;
+ struct device *dev = component->dev;
struct snd_soc_dai *dai;
- int i, ret = 0;
+ unsigned int i;
+ int ret;
dev_dbg(dev, "ASoC: dai register %s #%Zu\n", dev_name(dev), count);
@@ -3982,70 +3974,54 @@ static int snd_soc_register_dais(struct device *dev,
goto err;
}
- /* create DAI component name */
- dai->name = fmt_multiple_name(dev, &dai_drv[i]);
+ /*
+ * Back in the old days when we still had component-less DAIs,
+ * instead of having a static name, component-less DAIs would
+ * inherit the name of the parent device so it is possible to
+ * register multiple instances of the DAI. We still need to keep
+ * the same naming style even though those DAIs are not
+ * component-less anymore.
+ */
+ if (count == 1 && legacy_dai_naming) {
+ dai->name = fmt_single_name(dev, &dai->id);
+ } else {
+ dai->name = fmt_multiple_name(dev, &dai_drv[i]);
+ if (dai_drv[i].id)
+ dai->id = dai_drv[i].id;
+ else
+ dai->id = i;
+ }
if (dai->name == NULL) {
kfree(dai);
- ret = -EINVAL;
+ ret = -ENOMEM;
goto err;
}
+ dai->component = component;
+ dai->codec = codec;
dai->dev = dev;
dai->driver = &dai_drv[i];
- if (dai->driver->id)
- dai->id = dai->driver->id;
- else
- dai->id = i;
dai->dapm.dev = dev;
if (!dai->driver->ops)
dai->driver->ops = &null_dai_ops;
- mutex_lock(&client_mutex);
-
- list_for_each_entry(codec, &codec_list, list) {
- if (codec->dev == dev) {
- dev_dbg(dev,
- "ASoC: Mapped DAI %s to CODEC %s\n",
- dai->name, codec->name);
- dai->codec = codec;
- break;
- }
- }
-
if (!dai->codec)
dai->dapm.idle_bias_off = 1;
- list_add(&dai->list, &dai_list);
+ list_add(&dai->list, &component->dai_list);
- mutex_unlock(&client_mutex);
-
- dev_dbg(dai->dev, "ASoC: Registered DAI '%s'\n", dai->name);
+ dev_dbg(dev, "ASoC: Registered DAI '%s'\n", dai->name);
}
return 0;
err:
- for (i--; i >= 0; i--)
- snd_soc_unregister_dai(dev);
+ snd_soc_unregister_dais(component);
return ret;
}
/**
- * snd_soc_unregister_dais - Unregister multiple DAIs from the ASoC core
- *
- * @dai: Array of DAIs to unregister
- * @count: Number of DAIs
- */
-static void snd_soc_unregister_dais(struct device *dev, size_t count)
-{
- int i;
-
- for (i = 0; i < count; i++)
- snd_soc_unregister_dai(dev);
-}
-
-/**
* snd_soc_register_component - Register a component with the ASoC core
*
*/
@@ -4053,6 +4029,7 @@ static int
__snd_soc_register_component(struct device *dev,
struct snd_soc_component *cmpnt,
const struct snd_soc_component_driver *cmpnt_drv,
+ struct snd_soc_codec *codec,
struct snd_soc_dai_driver *dai_drv,
int num_dai, bool allow_single_dai)
{
@@ -4075,20 +4052,10 @@ __snd_soc_register_component(struct device *dev,
cmpnt->driver = cmpnt_drv;
cmpnt->dai_drv = dai_drv;
cmpnt->num_dai = num_dai;
+ INIT_LIST_HEAD(&cmpnt->dai_list);
- /*
- * snd_soc_register_dai() uses fmt_single_name(), and
- * snd_soc_register_dais() uses fmt_multiple_name()
- * for dai->name which is used for name based matching
- *
- * this function is used from cpu/codec.
- * allow_single_dai flag can ignore "codec" driver reworking
- * since it had been used snd_soc_register_dais(),
- */
- if ((1 == num_dai) && allow_single_dai)
- ret = snd_soc_register_dai(dev, dai_drv);
- else
- ret = snd_soc_register_dais(dev, dai_drv, num_dai);
+ ret = snd_soc_register_dais(cmpnt, codec, dai_drv, num_dai,
+ allow_single_dai);
if (ret < 0) {
dev_err(dev, "ASoC: Failed to regster DAIs: %d\n", ret);
goto error_component_name;
@@ -4121,7 +4088,9 @@ int snd_soc_register_component(struct device *dev,
return -ENOMEM;
}
- return __snd_soc_register_component(dev, cmpnt, cmpnt_drv,
+ cmpnt->ignore_pmdown_time = true;
+
+ return __snd_soc_register_component(dev, cmpnt, cmpnt_drv, NULL,
dai_drv, num_dai, true);
}
EXPORT_SYMBOL_GPL(snd_soc_register_component);
@@ -4141,7 +4110,7 @@ void snd_soc_unregister_component(struct device *dev)
return;
found:
- snd_soc_unregister_dais(dev, cmpnt->num_dai);
+ snd_soc_unregister_dais(cmpnt);
mutex_lock(&client_mutex);
list_del(&cmpnt->list);
@@ -4319,7 +4288,7 @@ int snd_soc_register_codec(struct device *dev,
codec->volatile_register = codec_drv->volatile_register;
codec->readable_register = codec_drv->readable_register;
codec->writable_register = codec_drv->writable_register;
- codec->ignore_pmdown_time = codec_drv->ignore_pmdown_time;
+ codec->component.ignore_pmdown_time = codec_drv->ignore_pmdown_time;
codec->dapm.bias_level = SND_SOC_BIAS_OFF;
codec->dapm.dev = dev;
codec->dapm.codec = codec;
@@ -4342,7 +4311,7 @@ int snd_soc_register_codec(struct device *dev,
/* register component */
ret = __snd_soc_register_component(dev, &codec->component,
&codec_drv->component_driver,
- dai_drv, num_dai, false);
+ codec, dai_drv, num_dai, false);
if (ret < 0) {
dev_err(codec->dev, "ASoC: Failed to regster component: %d\n", ret);
goto fail_codec_name;
@@ -4417,6 +4386,122 @@ int snd_soc_of_parse_card_name(struct snd_soc_card *card,
}
EXPORT_SYMBOL_GPL(snd_soc_of_parse_card_name);
+static const struct snd_soc_dapm_widget simple_widgets[] = {
+ SND_SOC_DAPM_MIC("Microphone", NULL),
+ SND_SOC_DAPM_LINE("Line", NULL),
+ SND_SOC_DAPM_HP("Headphone", NULL),
+ SND_SOC_DAPM_SPK("Speaker", NULL),
+};
+
+int snd_soc_of_parse_audio_simple_widgets(struct snd_soc_card *card,
+ const char *propname)
+{
+ struct device_node *np = card->dev->of_node;
+ struct snd_soc_dapm_widget *widgets;
+ const char *template, *wname;
+ int i, j, num_widgets, ret;
+
+ num_widgets = of_property_count_strings(np, propname);
+ if (num_widgets < 0) {
+ dev_err(card->dev,
+ "ASoC: Property '%s' does not exist\n", propname);
+ return -EINVAL;
+ }
+ if (num_widgets & 1) {
+ dev_err(card->dev,
+ "ASoC: Property '%s' length is not even\n", propname);
+ return -EINVAL;
+ }
+
+ num_widgets /= 2;
+ if (!num_widgets) {
+ dev_err(card->dev, "ASoC: Property '%s's length is zero\n",
+ propname);
+ return -EINVAL;
+ }
+
+ widgets = devm_kcalloc(card->dev, num_widgets, sizeof(*widgets),
+ GFP_KERNEL);
+ if (!widgets) {
+ dev_err(card->dev,
+ "ASoC: Could not allocate memory for widgets\n");
+ return -ENOMEM;
+ }
+
+ for (i = 0; i < num_widgets; i++) {
+ ret = of_property_read_string_index(np, propname,
+ 2 * i, &template);
+ if (ret) {
+ dev_err(card->dev,
+ "ASoC: Property '%s' index %d read error:%d\n",
+ propname, 2 * i, ret);
+ return -EINVAL;
+ }
+
+ for (j = 0; j < ARRAY_SIZE(simple_widgets); j++) {
+ if (!strncmp(template, simple_widgets[j].name,
+ strlen(simple_widgets[j].name))) {
+ widgets[i] = simple_widgets[j];
+ break;
+ }
+ }
+
+ if (j >= ARRAY_SIZE(simple_widgets)) {
+ dev_err(card->dev,
+ "ASoC: DAPM widget '%s' is not supported\n",
+ template);
+ return -EINVAL;
+ }
+
+ ret = of_property_read_string_index(np, propname,
+ (2 * i) + 1,
+ &wname);
+ if (ret) {
+ dev_err(card->dev,
+ "ASoC: Property '%s' index %d read error:%d\n",
+ propname, (2 * i) + 1, ret);
+ return -EINVAL;
+ }
+
+ widgets[i].name = wname;
+ }
+
+ card->dapm_widgets = widgets;
+ card->num_dapm_widgets = num_widgets;
+
+ return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_of_parse_audio_simple_widgets);
+
+int snd_soc_of_parse_tdm_slot(struct device_node *np,
+ unsigned int *slots,
+ unsigned int *slot_width)
+{
+ u32 val;
+ int ret;
+
+ if (of_property_read_bool(np, "dai-tdm-slot-num")) {
+ ret = of_property_read_u32(np, "dai-tdm-slot-num", &val);
+ if (ret)
+ return ret;
+
+ if (slots)
+ *slots = val;
+ }
+
+ if (of_property_read_bool(np, "dai-tdm-slot-width")) {
+ ret = of_property_read_u32(np, "dai-tdm-slot-width", &val);
+ if (ret)
+ return ret;
+
+ if (slot_width)
+ *slot_width = val;
+ }
+
+ return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_of_parse_tdm_slot);
+
int snd_soc_of_parse_audio_routing(struct snd_soc_card *card,
const char *propname)
{
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index b9dc6ac..c8a780d 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -70,8 +70,6 @@ static int dapm_up_seq[] = {
[snd_soc_dapm_aif_out] = 4,
[snd_soc_dapm_mic] = 5,
[snd_soc_dapm_mux] = 6,
- [snd_soc_dapm_virt_mux] = 6,
- [snd_soc_dapm_value_mux] = 6,
[snd_soc_dapm_dac] = 7,
[snd_soc_dapm_switch] = 8,
[snd_soc_dapm_mixer] = 8,
@@ -102,8 +100,6 @@ static int dapm_down_seq[] = {
[snd_soc_dapm_mic] = 7,
[snd_soc_dapm_micbias] = 8,
[snd_soc_dapm_mux] = 9,
- [snd_soc_dapm_virt_mux] = 9,
- [snd_soc_dapm_value_mux] = 9,
[snd_soc_dapm_aif_in] = 10,
[snd_soc_dapm_aif_out] = 10,
[snd_soc_dapm_dai_in] = 10,
@@ -115,6 +111,12 @@ static int dapm_down_seq[] = {
[snd_soc_dapm_post] = 14,
};
+static void dapm_assert_locked(struct snd_soc_dapm_context *dapm)
+{
+ if (dapm->card && dapm->card->instantiated)
+ lockdep_assert_held(&dapm->card->dapm_mutex);
+}
+
static void pop_wait(u32 pop_time)
{
if (pop_time)
@@ -146,15 +148,16 @@ static bool dapm_dirty_widget(struct snd_soc_dapm_widget *w)
return !list_empty(&w->dirty);
}
-void dapm_mark_dirty(struct snd_soc_dapm_widget *w, const char *reason)
+static void dapm_mark_dirty(struct snd_soc_dapm_widget *w, const char *reason)
{
+ dapm_assert_locked(w->dapm);
+
if (!dapm_dirty_widget(w)) {
dev_vdbg(w->dapm->dev, "Marking %s dirty due to %s\n",
w->name, reason);
list_add_tail(&w->dirty, &w->dapm->card->dapm_dirty);
}
}
-EXPORT_SYMBOL_GPL(dapm_mark_dirty);
void dapm_mark_io_dirty(struct snd_soc_dapm_context *dapm)
{
@@ -361,6 +364,8 @@ static void dapm_reset(struct snd_soc_card *card)
{
struct snd_soc_dapm_widget *w;
+ lockdep_assert_held(&card->dapm_mutex);
+
memset(&card->dapm_stats, 0, sizeof(card->dapm_stats));
list_for_each_entry(w, &card->widgets, list) {
@@ -386,7 +391,8 @@ static int soc_widget_read(struct snd_soc_dapm_widget *w, int reg,
return -1;
}
-static int soc_widget_write(struct snd_soc_dapm_widget *w, int reg, int val)
+static int soc_widget_write(struct snd_soc_dapm_widget *w, int reg,
+ unsigned int val)
{
if (w->codec)
return snd_soc_write(w->codec, reg, val);
@@ -498,131 +504,40 @@ out:
return ret;
}
-/* set up initial codec paths */
-static void dapm_set_path_status(struct snd_soc_dapm_widget *w,
- struct snd_soc_dapm_path *p, int i)
+/* connect mux widget to its interconnecting audio paths */
+static int dapm_connect_mux(struct snd_soc_dapm_context *dapm,
+ struct snd_soc_dapm_widget *src, struct snd_soc_dapm_widget *dest,
+ struct snd_soc_dapm_path *path, const char *control_name,
+ const struct snd_kcontrol_new *kcontrol)
{
- switch (w->id) {
- case snd_soc_dapm_switch:
- case snd_soc_dapm_mixer:
- case snd_soc_dapm_mixer_named_ctl: {
- int val;
- struct soc_mixer_control *mc = (struct soc_mixer_control *)
- w->kcontrol_news[i].private_value;
- int reg = mc->reg;
- unsigned int shift = mc->shift;
- int max = mc->max;
- unsigned int mask = (1 << fls(max)) - 1;
- unsigned int invert = mc->invert;
-
- if (reg != SND_SOC_NOPM) {
- soc_widget_read(w, reg, &val);
- val = (val >> shift) & mask;
- if (invert)
- val = max - val;
- p->connect = !!val;
- } else {
- p->connect = 0;
- }
-
- }
- break;
- case snd_soc_dapm_mux: {
- struct soc_enum *e = (struct soc_enum *)
- w->kcontrol_news[i].private_value;
- int val, item;
-
- soc_widget_read(w, e->reg, &val);
- item = (val >> e->shift_l) & e->mask;
-
- if (item < e->max && !strcmp(p->name, e->texts[item]))
- p->connect = 1;
- else
- p->connect = 0;
- }
- break;
- case snd_soc_dapm_virt_mux: {
- struct soc_enum *e = (struct soc_enum *)
- w->kcontrol_news[i].private_value;
+ struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
+ unsigned int val, item;
+ int i;
- p->connect = 0;
+ if (e->reg != SND_SOC_NOPM) {
+ soc_widget_read(dest, e->reg, &val);
+ val = (val >> e->shift_l) & e->mask;
+ item = snd_soc_enum_val_to_item(e, val);
+ } else {
/* since a virtual mux has no backing registers to
* decide which path to connect, it will try to match
* with the first enumeration. This is to ensure
* that the default mux choice (the first) will be
* correctly powered up during initialization.
*/
- if (!strcmp(p->name, e->texts[0]))
- p->connect = 1;
+ item = 0;
}
- break;
- case snd_soc_dapm_value_mux: {
- struct soc_enum *e = (struct soc_enum *)
- w->kcontrol_news[i].private_value;
- int val, item;
- soc_widget_read(w, e->reg, &val);
- val = (val >> e->shift_l) & e->mask;
- for (item = 0; item < e->max; item++) {
- if (val == e->values[item])
- break;
- }
-
- if (item < e->max && !strcmp(p->name, e->texts[item]))
- p->connect = 1;
- else
- p->connect = 0;
- }
- break;
- /* does not affect routing - always connected */
- case snd_soc_dapm_pga:
- case snd_soc_dapm_out_drv:
- case snd_soc_dapm_output:
- case snd_soc_dapm_adc:
- case snd_soc_dapm_input:
- case snd_soc_dapm_siggen:
- case snd_soc_dapm_dac:
- case snd_soc_dapm_micbias:
- case snd_soc_dapm_vmid:
- case snd_soc_dapm_supply:
- case snd_soc_dapm_regulator_supply:
- case snd_soc_dapm_clock_supply:
- case snd_soc_dapm_aif_in:
- case snd_soc_dapm_aif_out:
- case snd_soc_dapm_dai_in:
- case snd_soc_dapm_dai_out:
- case snd_soc_dapm_hp:
- case snd_soc_dapm_mic:
- case snd_soc_dapm_spk:
- case snd_soc_dapm_line:
- case snd_soc_dapm_dai_link:
- case snd_soc_dapm_kcontrol:
- p->connect = 1;
- break;
- /* does affect routing - dynamically connected */
- case snd_soc_dapm_pre:
- case snd_soc_dapm_post:
- p->connect = 0;
- break;
- }
-}
-
-/* connect mux widget to its interconnecting audio paths */
-static int dapm_connect_mux(struct snd_soc_dapm_context *dapm,
- struct snd_soc_dapm_widget *src, struct snd_soc_dapm_widget *dest,
- struct snd_soc_dapm_path *path, const char *control_name,
- const struct snd_kcontrol_new *kcontrol)
-{
- struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
- int i;
-
- for (i = 0; i < e->max; i++) {
+ for (i = 0; i < e->items; i++) {
if (!(strcmp(control_name, e->texts[i]))) {
list_add(&path->list, &dapm->card->paths);
list_add(&path->list_sink, &dest->sources);
list_add(&path->list_source, &src->sinks);
path->name = (char*)e->texts[i];
- dapm_set_path_status(dest, path, 0);
+ if (i == item)
+ path->connect = 1;
+ else
+ path->connect = 0;
return 0;
}
}
@@ -630,6 +545,30 @@ static int dapm_connect_mux(struct snd_soc_dapm_context *dapm,
return -ENODEV;
}
+/* set up initial codec paths */
+static void dapm_set_mixer_path_status(struct snd_soc_dapm_widget *w,
+ struct snd_soc_dapm_path *p, int i)
+{
+ struct soc_mixer_control *mc = (struct soc_mixer_control *)
+ w->kcontrol_news[i].private_value;
+ unsigned int reg = mc->reg;
+ unsigned int shift = mc->shift;
+ unsigned int max = mc->max;
+ unsigned int mask = (1 << fls(max)) - 1;
+ unsigned int invert = mc->invert;
+ unsigned int val;
+
+ if (reg != SND_SOC_NOPM) {
+ soc_widget_read(w, reg, &val);
+ val = (val >> shift) & mask;
+ if (invert)
+ val = max - val;
+ p->connect = !!val;
+ } else {
+ p->connect = 0;
+ }
+}
+
/* connect mixer widget to its interconnecting audio paths */
static int dapm_connect_mixer(struct snd_soc_dapm_context *dapm,
struct snd_soc_dapm_widget *src, struct snd_soc_dapm_widget *dest,
@@ -644,7 +583,7 @@ static int dapm_connect_mixer(struct snd_soc_dapm_context *dapm,
list_add(&path->list_sink, &dest->sources);
list_add(&path->list_source, &src->sinks);
path->name = dest->kcontrol_news[i].name;
- dapm_set_path_status(dest, path, i);
+ dapm_set_mixer_path_status(dest, path, i);
return 0;
}
}
@@ -723,8 +662,6 @@ static int dapm_create_or_share_mixmux_kcontrol(struct snd_soc_dapm_widget *w,
kcname_in_long_name = true;
break;
case snd_soc_dapm_mux:
- case snd_soc_dapm_virt_mux:
- case snd_soc_dapm_value_mux:
wname_in_long_name = true;
kcname_in_long_name = false;
break;
@@ -1823,6 +1760,8 @@ static int dapm_power_widgets(struct snd_soc_card *card, int event)
ASYNC_DOMAIN_EXCLUSIVE(async_domain);
enum snd_soc_bias_level bias;
+ lockdep_assert_held(&card->dapm_mutex);
+
trace_snd_soc_dapm_start(card);
list_for_each_entry(d, &card->dapm_list, list) {
@@ -1897,10 +1836,14 @@ static int dapm_power_widgets(struct snd_soc_card *card, int event)
trace_snd_soc_dapm_walk_done(card);
- /* Run all the bias changes in parallel */
- list_for_each_entry(d, &card->dapm_list, list)
- async_schedule_domain(dapm_pre_sequence_async, d,
- &async_domain);
+ /* Run card bias changes at first */
+ dapm_pre_sequence_async(&card->dapm, 0);
+ /* Run other bias changes in parallel */
+ list_for_each_entry(d, &card->dapm_list, list) {
+ if (d != &card->dapm)
+ async_schedule_domain(dapm_pre_sequence_async, d,
+ &async_domain);
+ }
async_synchronize_full_domain(&async_domain);
list_for_each_entry(w, &down_list, power_list) {
@@ -1920,10 +1863,14 @@ static int dapm_power_widgets(struct snd_soc_card *card, int event)
dapm_seq_run(card, &up_list, event, true);
/* Run all the bias changes in parallel */
- list_for_each_entry(d, &card->dapm_list, list)
- async_schedule_domain(dapm_post_sequence_async, d,
- &async_domain);
+ list_for_each_entry(d, &card->dapm_list, list) {
+ if (d != &card->dapm)
+ async_schedule_domain(dapm_post_sequence_async, d,
+ &async_domain);
+ }
async_synchronize_full_domain(&async_domain);
+ /* Run card bias changes at last */
+ dapm_post_sequence_async(&card->dapm, 0);
/* do we need to notify any clients that DAPM event is complete */
list_for_each_entry(d, &card->dapm_list, list) {
@@ -2110,6 +2057,8 @@ static int soc_dapm_mux_update_power(struct snd_soc_card *card,
struct snd_soc_dapm_path *path;
int found = 0;
+ lockdep_assert_held(&card->dapm_mutex);
+
/* find dapm widget path assoc with kcontrol */
dapm_kcontrol_for_each_path(path, kcontrol) {
if (!path->name || !e->texts[mux])
@@ -2160,6 +2109,8 @@ static int soc_dapm_mixer_update_power(struct snd_soc_card *card,
struct snd_soc_dapm_path *path;
int found = 0;
+ lockdep_assert_held(&card->dapm_mutex);
+
/* find dapm widget path assoc with kcontrol */
dapm_kcontrol_for_each_path(path, kcontrol) {
found = 1;
@@ -2325,6 +2276,8 @@ static int snd_soc_dapm_set_pin(struct snd_soc_dapm_context *dapm,
{
struct snd_soc_dapm_widget *w = dapm_find_widget(dapm, pin, true);
+ dapm_assert_locked(dapm);
+
if (!w) {
dev_err(dapm->dev, "ASoC: DAPM unknown pin %s\n", pin);
return -EINVAL;
@@ -2341,18 +2294,18 @@ static int snd_soc_dapm_set_pin(struct snd_soc_dapm_context *dapm,
}
/**
- * snd_soc_dapm_sync - scan and power dapm paths
+ * snd_soc_dapm_sync_unlocked - scan and power dapm paths
* @dapm: DAPM context
*
* Walks all dapm audio paths and powers widgets according to their
* stream or path usage.
*
+ * Requires external locking.
+ *
* Returns 0 for success.
*/
-int snd_soc_dapm_sync(struct snd_soc_dapm_context *dapm)
+int snd_soc_dapm_sync_unlocked(struct snd_soc_dapm_context *dapm)
{
- int ret;
-
/*
* Suppress early reports (eg, jacks syncing their state) to avoid
* silly DAPM runs during card startup.
@@ -2360,8 +2313,25 @@ int snd_soc_dapm_sync(struct snd_soc_dapm_context *dapm)
if (!dapm->card || !dapm->card->instantiated)
return 0;
+ return dapm_power_widgets(dapm->card, SND_SOC_DAPM_STREAM_NOP);
+}
+EXPORT_SYMBOL_GPL(snd_soc_dapm_sync_unlocked);
+
+/**
+ * snd_soc_dapm_sync - scan and power dapm paths
+ * @dapm: DAPM context
+ *
+ * Walks all dapm audio paths and powers widgets according to their
+ * stream or path usage.
+ *
+ * Returns 0 for success.
+ */
+int snd_soc_dapm_sync(struct snd_soc_dapm_context *dapm)
+{
+ int ret;
+
mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME);
- ret = dapm_power_widgets(dapm->card, SND_SOC_DAPM_STREAM_NOP);
+ ret = snd_soc_dapm_sync_unlocked(dapm);
mutex_unlock(&dapm->card->dapm_mutex);
return ret;
}
@@ -2444,8 +2414,6 @@ static int snd_soc_dapm_add_path(struct snd_soc_dapm_context *dapm,
path->connect = 1;
return 0;
case snd_soc_dapm_mux:
- case snd_soc_dapm_virt_mux:
- case snd_soc_dapm_value_mux:
ret = dapm_connect_mux(dapm, wsource, wsink, path, control,
&wsink->kcontrol_news[0]);
if (ret != 0)
@@ -2772,8 +2740,6 @@ int snd_soc_dapm_new_widgets(struct snd_soc_card *card)
dapm_new_mixer(w);
break;
case snd_soc_dapm_mux:
- case snd_soc_dapm_virt_mux:
- case snd_soc_dapm_value_mux:
dapm_new_mux(w);
break;
case snd_soc_dapm_pga:
@@ -2935,213 +2901,75 @@ int snd_soc_dapm_get_enum_double(struct snd_kcontrol *kcontrol,
{
struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol);
struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
- unsigned int val;
-
- val = snd_soc_read(codec, e->reg);
- ucontrol->value.enumerated.item[0] = (val >> e->shift_l) & e->mask;
- if (e->shift_l != e->shift_r)
- ucontrol->value.enumerated.item[1] =
- (val >> e->shift_r) & e->mask;
-
- return 0;
-}
-EXPORT_SYMBOL_GPL(snd_soc_dapm_get_enum_double);
-
-/**
- * snd_soc_dapm_put_enum_double - dapm enumerated double mixer set callback
- * @kcontrol: mixer control
- * @ucontrol: control element information
- *
- * Callback to set the value of a dapm enumerated double mixer control.
- *
- * Returns 0 for success.
- */
-int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol);
- struct snd_soc_card *card = codec->card;
- struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
- unsigned int val, mux, change;
- unsigned int mask;
- struct snd_soc_dapm_update update;
- int ret = 0;
-
- if (ucontrol->value.enumerated.item[0] > e->max - 1)
- return -EINVAL;
- mux = ucontrol->value.enumerated.item[0];
- val = mux << e->shift_l;
- mask = e->mask << e->shift_l;
- if (e->shift_l != e->shift_r) {
- if (ucontrol->value.enumerated.item[1] > e->max - 1)
- return -EINVAL;
- val |= ucontrol->value.enumerated.item[1] << e->shift_r;
- mask |= e->mask << e->shift_r;
- }
-
- mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME);
-
- change = snd_soc_test_bits(codec, e->reg, mask, val);
- if (change) {
- update.kcontrol = kcontrol;
- update.reg = e->reg;
- update.mask = mask;
- update.val = val;
- card->update = &update;
-
- ret = soc_dapm_mux_update_power(card, kcontrol, mux, e);
-
- card->update = NULL;
- }
-
- mutex_unlock(&card->dapm_mutex);
-
- if (ret > 0)
- soc_dpcm_runtime_update(card);
-
- return change;
-}
-EXPORT_SYMBOL_GPL(snd_soc_dapm_put_enum_double);
-
-/**
- * snd_soc_dapm_get_enum_virt - Get virtual DAPM mux
- * @kcontrol: mixer control
- * @ucontrol: control element information
- *
- * Returns 0 for success.
- */
-int snd_soc_dapm_get_enum_virt(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- ucontrol->value.enumerated.item[0] = dapm_kcontrol_get_value(kcontrol);
- return 0;
-}
-EXPORT_SYMBOL_GPL(snd_soc_dapm_get_enum_virt);
-
-/**
- * snd_soc_dapm_put_enum_virt - Set virtual DAPM mux
- * @kcontrol: mixer control
- * @ucontrol: control element information
- *
- * Returns 0 for success.
- */
-int snd_soc_dapm_put_enum_virt(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol);
- struct snd_soc_card *card = codec->card;
- unsigned int value;
- struct soc_enum *e =
- (struct soc_enum *)kcontrol->private_value;
- int change;
- int ret = 0;
-
- if (ucontrol->value.enumerated.item[0] >= e->max)
- return -EINVAL;
+ unsigned int reg_val, val;
- mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME);
-
- value = ucontrol->value.enumerated.item[0];
- change = dapm_kcontrol_set_value(kcontrol, value);
- if (change)
- ret = soc_dapm_mux_update_power(card, kcontrol, value, e);
-
- mutex_unlock(&card->dapm_mutex);
-
- if (ret > 0)
- soc_dpcm_runtime_update(card);
-
- return change;
-}
-EXPORT_SYMBOL_GPL(snd_soc_dapm_put_enum_virt);
-
-/**
- * snd_soc_dapm_get_value_enum_double - dapm semi enumerated double mixer get
- * callback
- * @kcontrol: mixer control
- * @ucontrol: control element information
- *
- * Callback to get the value of a dapm semi enumerated double mixer control.
- *
- * Semi enumerated mixer: the enumerated items are referred as values. Can be
- * used for handling bitfield coded enumeration for example.
- *
- * Returns 0 for success.
- */
-int snd_soc_dapm_get_value_enum_double(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol);
- struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
- unsigned int reg_val, val, mux;
+ if (e->reg != SND_SOC_NOPM)
+ reg_val = snd_soc_read(codec, e->reg);
+ else
+ reg_val = dapm_kcontrol_get_value(kcontrol);
- reg_val = snd_soc_read(codec, e->reg);
val = (reg_val >> e->shift_l) & e->mask;
- for (mux = 0; mux < e->max; mux++) {
- if (val == e->values[mux])
- break;
- }
- ucontrol->value.enumerated.item[0] = mux;
+ ucontrol->value.enumerated.item[0] = snd_soc_enum_val_to_item(e, val);
if (e->shift_l != e->shift_r) {
val = (reg_val >> e->shift_r) & e->mask;
- for (mux = 0; mux < e->max; mux++) {
- if (val == e->values[mux])
- break;
- }
- ucontrol->value.enumerated.item[1] = mux;
+ val = snd_soc_enum_val_to_item(e, val);
+ ucontrol->value.enumerated.item[1] = val;
}
return 0;
}
-EXPORT_SYMBOL_GPL(snd_soc_dapm_get_value_enum_double);
+EXPORT_SYMBOL_GPL(snd_soc_dapm_get_enum_double);
/**
- * snd_soc_dapm_put_value_enum_double - dapm semi enumerated double mixer set
- * callback
+ * snd_soc_dapm_put_enum_double - dapm enumerated double mixer set callback
* @kcontrol: mixer control
* @ucontrol: control element information
*
- * Callback to set the value of a dapm semi enumerated double mixer control.
- *
- * Semi enumerated mixer: the enumerated items are referred as values. Can be
- * used for handling bitfield coded enumeration for example.
+ * Callback to set the value of a dapm enumerated double mixer control.
*
* Returns 0 for success.
*/
-int snd_soc_dapm_put_value_enum_double(struct snd_kcontrol *kcontrol,
+int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol);
struct snd_soc_card *card = codec->card;
struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
- unsigned int val, mux, change;
+ unsigned int *item = ucontrol->value.enumerated.item;
+ unsigned int val, change;
unsigned int mask;
struct snd_soc_dapm_update update;
int ret = 0;
- if (ucontrol->value.enumerated.item[0] > e->max - 1)
+ if (item[0] >= e->items)
return -EINVAL;
- mux = ucontrol->value.enumerated.item[0];
- val = e->values[ucontrol->value.enumerated.item[0]] << e->shift_l;
+
+ val = snd_soc_enum_item_to_val(e, item[0]) << e->shift_l;
mask = e->mask << e->shift_l;
if (e->shift_l != e->shift_r) {
- if (ucontrol->value.enumerated.item[1] > e->max - 1)
+ if (item[1] > e->items)
return -EINVAL;
- val |= e->values[ucontrol->value.enumerated.item[1]] << e->shift_r;
+ val |= snd_soc_enum_item_to_val(e, item[1]) << e->shift_l;
mask |= e->mask << e->shift_r;
}
mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME);
- change = snd_soc_test_bits(codec, e->reg, mask, val);
+ if (e->reg != SND_SOC_NOPM)
+ change = snd_soc_test_bits(codec, e->reg, mask, val);
+ else
+ change = dapm_kcontrol_set_value(kcontrol, val);
+
if (change) {
- update.kcontrol = kcontrol;
- update.reg = e->reg;
- update.mask = mask;
- update.val = val;
- card->update = &update;
+ if (e->reg != SND_SOC_NOPM) {
+ update.kcontrol = kcontrol;
+ update.reg = e->reg;
+ update.mask = mask;
+ update.val = val;
+ card->update = &update;
+ }
- ret = soc_dapm_mux_update_power(card, kcontrol, mux, e);
+ ret = soc_dapm_mux_update_power(card, kcontrol, item[0], e);
card->update = NULL;
}
@@ -3153,7 +2981,7 @@ int snd_soc_dapm_put_value_enum_double(struct snd_kcontrol *kcontrol,
return change;
}
-EXPORT_SYMBOL_GPL(snd_soc_dapm_put_value_enum_double);
+EXPORT_SYMBOL_GPL(snd_soc_dapm_put_enum_double);
/**
* snd_soc_dapm_info_pin_switch - Info for a pin switch
@@ -3283,8 +3111,6 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm,
w->power_check = dapm_generic_check_power;
break;
case snd_soc_dapm_mux:
- case snd_soc_dapm_virt_mux:
- case snd_soc_dapm_value_mux:
w->power_check = dapm_generic_check_power;
break;
case snd_soc_dapm_dai_out:
@@ -4098,7 +3924,7 @@ void snd_soc_dapm_free(struct snd_soc_dapm_context *dapm)
}
EXPORT_SYMBOL_GPL(snd_soc_dapm_free);
-static void soc_dapm_shutdown_codec(struct snd_soc_dapm_context *dapm)
+static void soc_dapm_shutdown_dapm(struct snd_soc_dapm_context *dapm)
{
struct snd_soc_card *card = dapm->card;
struct snd_soc_dapm_widget *w;
@@ -4138,14 +3964,21 @@ static void soc_dapm_shutdown_codec(struct snd_soc_dapm_context *dapm)
*/
void snd_soc_dapm_shutdown(struct snd_soc_card *card)
{
- struct snd_soc_codec *codec;
+ struct snd_soc_dapm_context *dapm;
- list_for_each_entry(codec, &card->codec_dev_list, card_list) {
- soc_dapm_shutdown_codec(&codec->dapm);
- if (codec->dapm.bias_level == SND_SOC_BIAS_STANDBY)
- snd_soc_dapm_set_bias_level(&codec->dapm,
- SND_SOC_BIAS_OFF);
+ list_for_each_entry(dapm, &card->dapm_list, list) {
+ if (dapm != &card->dapm) {
+ soc_dapm_shutdown_dapm(dapm);
+ if (dapm->bias_level == SND_SOC_BIAS_STANDBY)
+ snd_soc_dapm_set_bias_level(dapm,
+ SND_SOC_BIAS_OFF);
+ }
}
+
+ soc_dapm_shutdown_dapm(&card->dapm);
+ if (card->dapm.bias_level == SND_SOC_BIAS_STANDBY)
+ snd_soc_dapm_set_bias_level(&card->dapm,
+ SND_SOC_BIAS_OFF);
}
/* Module information */
diff --git a/sound/soc/soc-io.c b/sound/soc/soc-io.c
index aa886cc..260efc8 100644
--- a/sound/soc/soc-io.c
+++ b/sound/soc/soc-io.c
@@ -23,21 +23,6 @@
static int hw_write(struct snd_soc_codec *codec, unsigned int reg,
unsigned int value)
{
- int ret;
-
- if (!snd_soc_codec_volatile_register(codec, reg) &&
- reg < codec->driver->reg_cache_size &&
- !codec->cache_bypass) {
- ret = snd_soc_cache_write(codec, reg, value);
- if (ret < 0)
- return -1;
- }
-
- if (codec->cache_only) {
- codec->cache_sync = 1;
- return 0;
- }
-
return regmap_write(codec->control_data, reg, value);
}
@@ -46,32 +31,18 @@ static unsigned int hw_read(struct snd_soc_codec *codec, unsigned int reg)
int ret;
unsigned int val;
- if (reg >= codec->driver->reg_cache_size ||
- snd_soc_codec_volatile_register(codec, reg) ||
- codec->cache_bypass) {
- if (codec->cache_only)
- return -1;
-
- ret = regmap_read(codec->control_data, reg, &val);
- if (ret == 0)
- return val;
- else
- return -1;
- }
-
- ret = snd_soc_cache_read(codec, reg, &val);
- if (ret < 0)
+ ret = regmap_read(codec->control_data, reg, &val);
+ if (ret == 0)
+ return val;
+ else
return -1;
- return val;
}
/**
* snd_soc_codec_set_cache_io: Set up standard I/O functions.
*
* @codec: CODEC to configure.
- * @addr_bits: Number of bits of register address data.
- * @data_bits: Number of bits of data per register.
- * @control: Control bus used.
+ * @map: Register map to write to
*
* Register formats are frequently shared between many I2C and SPI
* devices. In order to promote code reuse the ASoC core provides
@@ -85,60 +56,36 @@ static unsigned int hw_read(struct snd_soc_codec *codec, unsigned int reg)
* volatile registers.
*/
int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec,
- int addr_bits, int data_bits,
- enum snd_soc_control_type control)
+ struct regmap *regmap)
{
- struct regmap_config config;
int ret;
- memset(&config, 0, sizeof(config));
- codec->write = hw_write;
- codec->read = hw_read;
-
- config.reg_bits = addr_bits;
- config.val_bits = data_bits;
+ /* Device has made its own regmap arrangements */
+ if (!regmap)
+ codec->control_data = dev_get_regmap(codec->dev, NULL);
+ else
+ codec->control_data = regmap;
- switch (control) {
-#if IS_ENABLED(CONFIG_REGMAP_I2C)
- case SND_SOC_I2C:
- codec->control_data = regmap_init_i2c(to_i2c_client(codec->dev),
- &config);
- break;
-#endif
+ if (IS_ERR(codec->control_data))
+ return PTR_ERR(codec->control_data);
-#if IS_ENABLED(CONFIG_REGMAP_SPI)
- case SND_SOC_SPI:
- codec->control_data = regmap_init_spi(to_spi_device(codec->dev),
- &config);
- break;
-#endif
-
- case SND_SOC_REGMAP:
- /* Device has made its own regmap arrangements */
- codec->using_regmap = true;
- if (!codec->control_data)
- codec->control_data = dev_get_regmap(codec->dev, NULL);
+ codec->write = hw_write;
+ codec->read = hw_read;
- if (codec->control_data) {
- ret = regmap_get_val_bytes(codec->control_data);
- /* Errors are legitimate for non-integer byte
- * multiples */
- if (ret > 0)
- codec->val_bytes = ret;
- }
- break;
+ ret = regmap_get_val_bytes(codec->control_data);
+ /* Errors are legitimate for non-integer byte
+ * multiples */
+ if (ret > 0)
+ codec->val_bytes = ret;
- default:
- return -EINVAL;
- }
+ codec->using_regmap = true;
- return PTR_ERR_OR_ZERO(codec->control_data);
+ return 0;
}
EXPORT_SYMBOL_GPL(snd_soc_codec_set_cache_io);
#else
int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec,
- int addr_bits, int data_bits,
- enum snd_soc_control_type control)
+ struct regmap *regmap)
{
return -ENOTSUPP;
}
diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c
index 23d43da..b903f82 100644
--- a/sound/soc/soc-jack.c
+++ b/sound/soc/soc-jack.c
@@ -250,7 +250,7 @@ static void snd_soc_jack_gpio_detect(struct snd_soc_jack_gpio *gpio)
report = 0;
if (gpio->jack_status_check)
- report = gpio->jack_status_check();
+ report = gpio->jack_status_check(gpio->data);
snd_soc_jack_report(jack, report, gpio->report);
}
@@ -342,7 +342,8 @@ int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count,
gpio_export(gpios[i].gpio, false);
/* Update initial jack status */
- snd_soc_jack_gpio_detect(&gpios[i]);
+ schedule_delayed_work(&gpios[i].work,
+ msecs_to_jiffies(gpios[i].debounce_time));
}
return 0;
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index 28522bd..2cedf09 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -35,6 +35,86 @@
#define DPCM_MAX_BE_USERS 8
/**
+ * snd_soc_runtime_activate() - Increment active count for PCM runtime components
+ * @rtd: ASoC PCM runtime that is activated
+ * @stream: Direction of the PCM stream
+ *
+ * Increments the active count for all the DAIs and components attached to a PCM
+ * runtime. Should typically be called when a stream is opened.
+ *
+ * Must be called with the rtd->pcm_mutex being held
+ */
+void snd_soc_runtime_activate(struct snd_soc_pcm_runtime *rtd, int stream)
+{
+ struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *codec_dai = rtd->codec_dai;
+
+ lockdep_assert_held(&rtd->pcm_mutex);
+
+ if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ cpu_dai->playback_active++;
+ codec_dai->playback_active++;
+ } else {
+ cpu_dai->capture_active++;
+ codec_dai->capture_active++;
+ }
+
+ cpu_dai->active++;
+ codec_dai->active++;
+ cpu_dai->component->active++;
+ codec_dai->component->active++;
+}
+
+/**
+ * snd_soc_runtime_deactivate() - Decrement active count for PCM runtime components
+ * @rtd: ASoC PCM runtime that is deactivated
+ * @stream: Direction of the PCM stream
+ *
+ * Decrements the active count for all the DAIs and components attached to a PCM
+ * runtime. Should typically be called when a stream is closed.
+ *
+ * Must be called with the rtd->pcm_mutex being held
+ */
+void snd_soc_runtime_deactivate(struct snd_soc_pcm_runtime *rtd, int stream)
+{
+ struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *codec_dai = rtd->codec_dai;
+
+ lockdep_assert_held(&rtd->pcm_mutex);
+
+ if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ cpu_dai->playback_active--;
+ codec_dai->playback_active--;
+ } else {
+ cpu_dai->capture_active--;
+ codec_dai->capture_active--;
+ }
+
+ cpu_dai->active--;
+ codec_dai->active--;
+ cpu_dai->component->active--;
+ codec_dai->component->active--;
+}
+
+/**
+ * snd_soc_runtime_ignore_pmdown_time() - Check whether to ignore the power down delay
+ * @rtd: The ASoC PCM runtime that should be checked.
+ *
+ * This function checks whether the power down delay should be ignored for a
+ * specific PCM runtime. Returns true if the delay is 0, if it the DAI link has
+ * been configured to ignore the delay, or if none of the components benefits
+ * from having the delay.
+ */
+bool snd_soc_runtime_ignore_pmdown_time(struct snd_soc_pcm_runtime *rtd)
+{
+ if (!rtd->pmdown_time || rtd->dai_link->ignore_pmdown_time)
+ return true;
+
+ return rtd->cpu_dai->component->ignore_pmdown_time &&
+ rtd->codec_dai->component->ignore_pmdown_time;
+}
+
+/**
* snd_soc_set_runtime_hwparams - set the runtime hardware parameters
* @substream: the pcm substream
* @hw: the hardware parameters
@@ -378,16 +458,9 @@ static int soc_pcm_open(struct snd_pcm_substream *substream)
runtime->hw.rate_max);
dynamic:
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
- cpu_dai->playback_active++;
- codec_dai->playback_active++;
- } else {
- cpu_dai->capture_active++;
- codec_dai->capture_active++;
- }
- cpu_dai->active++;
- codec_dai->active++;
- rtd->codec->active++;
+
+ snd_soc_runtime_activate(rtd, substream->stream);
+
mutex_unlock(&rtd->pcm_mutex);
return 0;
@@ -459,21 +532,10 @@ static int soc_pcm_close(struct snd_pcm_substream *substream)
struct snd_soc_platform *platform = rtd->platform;
struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
struct snd_soc_dai *codec_dai = rtd->codec_dai;
- struct snd_soc_codec *codec = rtd->codec;
mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass);
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
- cpu_dai->playback_active--;
- codec_dai->playback_active--;
- } else {
- cpu_dai->capture_active--;
- codec_dai->capture_active--;
- }
-
- cpu_dai->active--;
- codec_dai->active--;
- codec->active--;
+ snd_soc_runtime_deactivate(rtd, substream->stream);
/* clear the corresponding DAIs rate when inactive */
if (!cpu_dai->active)
@@ -496,8 +558,7 @@ static int soc_pcm_close(struct snd_pcm_substream *substream)
cpu_dai->runtime = NULL;
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
- if (!rtd->pmdown_time || codec->ignore_pmdown_time ||
- rtd->dai_link->ignore_pmdown_time) {
+ if (snd_soc_runtime_ignore_pmdown_time(rtd)) {
/* powered down playback stream now */
snd_soc_dapm_stream_event(rtd,
SNDRV_PCM_STREAM_PLAYBACK,
@@ -1989,7 +2050,6 @@ int soc_dpcm_runtime_update(struct snd_soc_card *card)
paths = dpcm_path_get(fe, SNDRV_PCM_STREAM_PLAYBACK, &list);
if (paths < 0) {
- dpcm_path_put(&list);
dev_warn(fe->dev, "ASoC: %s no valid %s path\n",
fe->dai_link->name, "playback");
mutex_unlock(&card->mutex);
@@ -2019,7 +2079,6 @@ capture:
paths = dpcm_path_get(fe, SNDRV_PCM_STREAM_CAPTURE, &list);
if (paths < 0) {
- dpcm_path_put(&list);
dev_warn(fe->dev, "ASoC: %s no valid %s path\n",
fe->dai_link->name, "capture");
mutex_unlock(&card->mutex);
@@ -2084,7 +2143,6 @@ static int dpcm_fe_dai_open(struct snd_pcm_substream *fe_substream)
fe->dpcm[stream].runtime = fe_substream->runtime;
if (dpcm_path_get(fe, stream, &list) <= 0) {
- dpcm_path_put(&list);
dev_dbg(fe->dev, "ASoC: %s no valid %s route\n",
fe->dai_link->name, stream ? "capture" : "playback");
}
diff --git a/sound/soc/spear/spdif_out.c b/sound/soc/spear/spdif_out.c
index fe99f46..19cca04 100644
--- a/sound/soc/spear/spdif_out.c
+++ b/sound/soc/spear/spdif_out.c
@@ -213,10 +213,7 @@ static int spdif_digital_mute(struct snd_soc_dai *dai, int mute)
static int spdif_mute_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
- struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
- struct snd_soc_card *card = codec->card;
- struct snd_soc_pcm_runtime *rtd = card->rtd;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kcontrol);
struct spdif_out_dev *host = snd_soc_dai_get_drvdata(cpu_dai);
ucontrol->value.integer.value[0] = host->saved_params.mute;
@@ -226,10 +223,7 @@ static int spdif_mute_get(struct snd_kcontrol *kcontrol,
static int spdif_mute_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
- struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
- struct snd_soc_card *card = codec->card;
- struct snd_soc_pcm_runtime *rtd = card->rtd;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kcontrol);
struct spdif_out_dev *host = snd_soc_dai_get_drvdata(cpu_dai);
if (host->saved_params.mute == ucontrol->value.integer.value[0])
diff --git a/sound/soc/tegra/Kconfig b/sound/soc/tegra/Kconfig
index 9f9c185..31198cf7 100644
--- a/sound/soc/tegra/Kconfig
+++ b/sound/soc/tegra/Kconfig
@@ -105,7 +105,7 @@ config SND_SOC_TEGRA_TRIMSLICE
tristate "SoC Audio support for TrimSlice board"
depends on SND_SOC_TEGRA && I2C
select SND_SOC_TEGRA20_I2S if ARCH_TEGRA_2x_SOC
- select SND_SOC_TLV320AIC23
+ select SND_SOC_TLV320AIC23_I2C
help
Say Y or M here if you want to add support for SoC audio on the
TrimSlice platform.
diff --git a/sound/soc/tegra/tegra20_ac97.c b/sound/soc/tegra/tegra20_ac97.c
index cf5e1cf..0a59e23 100644
--- a/sound/soc/tegra/tegra20_ac97.c
+++ b/sound/soc/tegra/tegra20_ac97.c
@@ -306,7 +306,7 @@ static const struct regmap_config tegra20_ac97_regmap_config = {
.readable_reg = tegra20_ac97_wr_rd_reg,
.volatile_reg = tegra20_ac97_volatile_reg,
.precious_reg = tegra20_ac97_precious_reg,
- .cache_type = REGCACHE_RBTREE,
+ .cache_type = REGCACHE_FLAT,
};
static int tegra20_ac97_platform_probe(struct platform_device *pdev)
diff --git a/sound/soc/tegra/tegra20_das.c b/sound/soc/tegra/tegra20_das.c
index e723929..a634f13 100644
--- a/sound/soc/tegra/tegra20_das.c
+++ b/sound/soc/tegra/tegra20_das.c
@@ -128,7 +128,7 @@ static const struct regmap_config tegra20_das_regmap_config = {
.max_register = LAST_REG(DAC_INPUT_DATA_CLK_SEL),
.writeable_reg = tegra20_das_wr_rd_reg,
.readable_reg = tegra20_das_wr_rd_reg,
- .cache_type = REGCACHE_RBTREE,
+ .cache_type = REGCACHE_FLAT,
};
static int tegra20_das_probe(struct platform_device *pdev)
diff --git a/sound/soc/tegra/tegra20_i2s.c b/sound/soc/tegra/tegra20_i2s.c
index 42c1f6b..79a9932 100644
--- a/sound/soc/tegra/tegra20_i2s.c
+++ b/sound/soc/tegra/tegra20_i2s.c
@@ -333,7 +333,7 @@ static const struct regmap_config tegra20_i2s_regmap_config = {
.readable_reg = tegra20_i2s_wr_rd_reg,
.volatile_reg = tegra20_i2s_volatile_reg,
.precious_reg = tegra20_i2s_precious_reg,
- .cache_type = REGCACHE_RBTREE,
+ .cache_type = REGCACHE_FLAT,
};
static int tegra20_i2s_platform_probe(struct platform_device *pdev)
diff --git a/sound/soc/tegra/tegra20_spdif.c b/sound/soc/tegra/tegra20_spdif.c
index 8c7c102..a0ce924 100644
--- a/sound/soc/tegra/tegra20_spdif.c
+++ b/sound/soc/tegra/tegra20_spdif.c
@@ -259,7 +259,7 @@ static const struct regmap_config tegra20_spdif_regmap_config = {
.readable_reg = tegra20_spdif_wr_rd_reg,
.volatile_reg = tegra20_spdif_volatile_reg,
.precious_reg = tegra20_spdif_precious_reg,
- .cache_type = REGCACHE_RBTREE,
+ .cache_type = REGCACHE_FLAT,
};
static int tegra20_spdif_platform_probe(struct platform_device *pdev)
diff --git a/sound/soc/tegra/tegra30_ahub.c b/sound/soc/tegra/tegra30_ahub.c
index d6f4c99..0db68f4 100644
--- a/sound/soc/tegra/tegra30_ahub.c
+++ b/sound/soc/tegra/tegra30_ahub.c
@@ -471,7 +471,7 @@ static const struct regmap_config tegra30_ahub_apbif_regmap_config = {
.readable_reg = tegra30_ahub_apbif_wr_rd_reg,
.volatile_reg = tegra30_ahub_apbif_volatile_reg,
.precious_reg = tegra30_ahub_apbif_precious_reg,
- .cache_type = REGCACHE_RBTREE,
+ .cache_type = REGCACHE_FLAT,
};
static bool tegra30_ahub_ahub_wr_rd_reg(struct device *dev, unsigned int reg)
@@ -490,7 +490,7 @@ static const struct regmap_config tegra30_ahub_ahub_regmap_config = {
.max_register = LAST_REG(AUDIO_RX),
.writeable_reg = tegra30_ahub_ahub_wr_rd_reg,
.readable_reg = tegra30_ahub_ahub_wr_rd_reg,
- .cache_type = REGCACHE_RBTREE,
+ .cache_type = REGCACHE_FLAT,
};
static struct tegra30_ahub_soc_data soc_data_tegra30 = {
diff --git a/sound/soc/tegra/tegra30_i2s.c b/sound/soc/tegra/tegra30_i2s.c
index 49ad936..f146c41 100644
--- a/sound/soc/tegra/tegra30_i2s.c
+++ b/sound/soc/tegra/tegra30_i2s.c
@@ -357,7 +357,7 @@ static const struct regmap_config tegra30_i2s_regmap_config = {
.writeable_reg = tegra30_i2s_wr_rd_reg,
.readable_reg = tegra30_i2s_wr_rd_reg,
.volatile_reg = tegra30_i2s_volatile_reg,
- .cache_type = REGCACHE_RBTREE,
+ .cache_type = REGCACHE_FLAT,
};
static const struct tegra30_i2s_soc_data tegra30_i2s_config = {
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