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[alsa-devel] [PATCH v2] ASoC: tegra: Use flat regcache.
by Dylan Reid 20 Mar '14

20 Mar '14

When using an rbtree cache, there can be allocations the first time a register is accessed. This can cause an attempt to schedule while atomic in the case that the regmap is using a spinlock. This could be fixed by either initializing all the registers or using a flat cache. The register maps for tegra30_ahub and tegra30_i2s are dense and don't save much from using a tree so convert them to flat. Tegra30 changes tested on Norrin, Tegra20 changes compile. Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- Changes v1 to v2: Fix Tegra20 fix as well. sound/soc/tegra/tegra20_ac97.c | 2 +- sound/soc/tegra/tegra20_das.c | 2 +- sound/soc/tegra/tegra20_i2s.c | 2 +- sound/soc/tegra/tegra20_spdif.c | 2 +- sound/soc/tegra/tegra30_ahub.c | 4 ++-- sound/soc/tegra/tegra30_i2s.c | 2 +- 6 files changed, 7 insertions(+), 7 deletions(-) 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 = { -- 1.8.1.3.605.g02339dd

3 2

[alsa-devel] [PATCH 0/2] ASoC: sirf: Add sirf audio hub driver and use it.
by RongJun Ying 19 Mar '14

19 Mar '14

From: Rongjun Ying <rongjun.ying(a)csr.com> This patchset adds including: 1. SiRF audio hub driver 2. The internal codec and port get regmap form audio hub driver 3. Add SiRF audio hub binding document Rongjun Ying (2): ASoC: sirf: Add sirf audio hub driver for sharing same register address space Documentation: Describe the SiRF audio hub Device Tree bindings .../devicetree/bindings/sound/sirf-audio-hub.txt | 52 +++++++++++++++ sound/soc/codecs/sirf-audio-codec.c | 32 ++-------- sound/soc/sirf/Kconfig | 5 ++ sound/soc/sirf/Makefile | 2 + sound/soc/sirf/sirf-audio-hub.c | 66 ++++++++++++++++++++ sound/soc/sirf/sirf-audio-port.c | 26 +-------- 6 files changed, 131 insertions(+), 52 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/sirf-audio-hub.txt create mode 100644 sound/soc/sirf/sirf-audio-hub.c -- 1.7.5.4

2 4

[alsa-devel] Regression caused by "ASoC: core: Rework SOC_DOUBLE_R_SX_TLV add SOC_SINGLE_SX_TLV"
by Thomas Petazzoni 19 Mar '14

19 Mar '14

Brian, Liam, Mark, I am currently bringing up audio support on the Marvell Armada 370 ARM SoC, for which the development board uses the CS42L51 audio codec. It turns out that the commit 1d99f2436d0d1c7741d6dfd9d27b5376cdbbca40 ("ASoC: core: Rework SOC_DOUBLE_R_SX_TLV add SOC_SINGLE_SX_TLV") causes a problem here, and reverting this commit gets things back to normal. Basically, when this commit is applied, the "PCM", "Analog" and "ADC Mixer" volume controls in alsamixer cannot be changed to any other value than 60 or 61. The volume control doesn't allow to go above 61 and below 60. If I remember correctly, at the codec level, the values configured for PCM in registers 0x10 and 0x11 are either 0x0 or 0x1 depending on whether you select 60 or 61 in alsamixer. Reverting the commit mentioned above makes alsamixer behave normally: the volume can be adjusted between 0 and 100 as expected, the values in register 0x10 and 0x11 are meaningful, and the behavior is as expected: with a volume to 0, I hear nothing in my headphones, with a volume set to 100, the volume is maximum, with a nice progressive volume increase in-between. Therefore, I believe that the commit has a problem. I haven't yet investigated where the problem is, maybe just the values passed to SOC_DOUBLE_R_SX_TLV(), or maybe in the implementation of the volume control functions themselves. I can certainly start investigating, but maybe the author of the commit will immediately realize where the problem could be. Thanks a lot! Thomas -- Thomas Petazzoni, CTO, Free Electrons Embedded Linux, Kernel and Android engineering http://free-electrons.com

4 18

[alsa-devel] [PATCH] Use jack_client_open instead of deprecated jack_client_new.
by Maarten Baert 19 Mar '14

19 Mar '14

Signed-off-by: Maarten Baert <maarten-baert(a)hotmail.com> --- jack/pcm_jack.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/jack/pcm_jack.c b/jack/pcm_jack.c index 4213708..7a8b24d 100644 --- a/jack/pcm_jack.c +++ b/jack/pcm_jack.c @@ -370,7 +370,7 @@ static int snd_pcm_jack_open(snd_pcm_t **pcmp, const char *name, __func__, jack_client_name, (int)strlen(jack_client_name)); } - jack->client = jack_client_new(jack_client_name); + jack->client = jack_client_open(jack_client_name, JackNoStartServer, NULL); if (jack->client == 0) { snd_pcm_jack_free(jack); -- 1.9.0

2 1

[alsa-devel] [PATCH] ALSA: compress: Pass through return value of open ops callback
by Charles Keepax 19 Mar '14

19 Mar '14

The snd_compr_open function would always return 0 even if the compressed ops open function failed, obviously this is incorrect. Looks like this was introduced by a small typo in: commit a0830dbd4e42b38aefdf3fb61ba5019a1a99ea85 ALSA: Add a reference counter to card instance This patch returns the value from the compressed op as it should. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/core/compress_offload.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index 7a20897..7403f34 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -133,7 +133,7 @@ static int snd_compr_open(struct inode *inode, struct file *f) kfree(data); } snd_card_unref(compr->card); - return 0; + return ret; } static int snd_compr_free(struct inode *inode, struct file *f) -- 1.7.2.5

3 2

[alsa-devel] [PATCH] pcm: route: Don't handle no matching chmap as a serious error
by Takashi Iwai 19 Mar '14

19 Mar '14

When find_matching_chmap() returns an error for the non-matching chmap, the caller, snd_pcm_route_open(), also returns an error although it shouldn't be handled as the fatal error. This results in the probe error with PulseAudio and it gives no real output in the end. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- src/pcm/pcm_route.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/src/pcm/pcm_route.c b/src/pcm/pcm_route.c index ab17fa78be2c..ac11bdc8adfd 100644 --- a/src/pcm/pcm_route.c +++ b/src/pcm/pcm_route.c @@ -940,10 +940,8 @@ static int find_matching_chmap(snd_pcm_t *spcm, snd_pcm_chmap_t *tt_chmap, snd_pcm_free_chmaps(chmaps); - if (*found_chmap == NULL) { + if (*found_chmap == NULL) SNDERR("Found no matching channel map"); - return -EINVAL; - } return 0; } -- 1.9.0

2 6

[alsa-devel] [PATCH v2 1/2] remgap: Fix possible sleep-in-atomic in regmap_bulk_write()
by Takashi Iwai 18 Mar '14

18 Mar '14

regmap deploys the spinlock for the protection when set up in fast_io mode. This may lead to sleep-in-atomic by memory allocation with GFP_KERNEL in regmap_bulk_write(). This patch fixes it by moving the allocation out of the lock. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- drivers/base/regmap/regmap.c | 11 ++++++----- 1 file changed, 6 insertions(+), 5 deletions(-) diff --git a/drivers/base/regmap/regmap.c b/drivers/base/regmap/regmap.c index 6a19515f8a45..6d134a3cbfd3 100644 --- a/drivers/base/regmap/regmap.c +++ b/drivers/base/regmap/regmap.c @@ -1520,12 +1520,12 @@ int regmap_bulk_write(struct regmap *map, unsigned int reg, const void *val, if (reg % map->reg_stride) return -EINVAL; - map->lock(map->lock_arg); /* * Some devices don't support bulk write, for * them we have a series of single write operations. */ if (!map->bus || map->use_single_rw) { + map->lock(map->lock_arg); for (i = 0; i < val_count; i++) { unsigned int ival; @@ -1554,24 +1554,25 @@ int regmap_bulk_write(struct regmap *map, unsigned int reg, const void *val, if (ret != 0) goto out; } +out: + map->unlock(map->lock_arg); } else { void *wval; wval = kmemdup(val, val_count * val_bytes, GFP_KERNEL); if (!wval) { - ret = -ENOMEM; dev_err(map->dev, "Error in memory allocation\n"); - goto out; + return -ENOMEM; } + map->lock(map->lock_arg); for (i = 0; i < val_count * val_bytes; i += val_bytes) map->format.parse_inplace(wval + i); ret = _regmap_raw_write(map, reg, wval, val_bytes * val_count); + map->unlock(map->lock_arg); kfree(wval); } -out: - map->unlock(map->lock_arg); return ret; } EXPORT_SYMBOL_GPL(regmap_bulk_write); -- 1.9.0

2 7

[alsa-devel] [PATCH v2] ASoC: simple-card: overwrite cpu_dai->fmt with codec_dai->fmt
by Nicolin Chen 18 Mar '14

18 Mar '14

The current simple-card driver separates the daimft for cpu_dai and codec_dai. So we might get different values for them (0x4003 and 0x1003 for example): asoc-simple-card sound-cs42888.12: cpu : 2024000.esai / 4003 / 132000000 asoc-simple-card sound-cs42888.12: codec : cs42888 / 1003 / 24576000 asoc-simple-card sound-cs42888.12: cs42888 <-> 2024000.esai mapping ok This is not allowed at all as we need to keep the DAIFMT settings identical for both the ends of the link. Thus this patch fixes it by overwriting the cpu_dai->fmt with codec_dai->fmt since we defined the DAIFMT_MASTER basing on CODEC at the first place while the other bits are same. Signed-off-by: Nicolin Chen <Guangyu.Chen(a)freescale.com> --- Changelog v2: * Keep the fmt identical for both CPU and CODEC sides. * Appended warning to binding doc. .../devicetree/bindings/sound/simple-card.txt | 6 ++++++ sound/soc/generic/simple-card.c | 20 ++++++++++++++------ 2 files changed, 20 insertions(+), 6 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/simple-card.txt b/Documentation/devicetree/bindings/sound/simple-card.txt index b30c222..881914b1 100644 --- a/Documentation/devicetree/bindings/sound/simple-card.txt +++ b/Documentation/devicetree/bindings/sound/simple-card.txt @@ -43,6 +43,12 @@ Optional CPU/CODEC subnodes properties: clock node (= common clock), or "system-clock-frequency" (if system doens't support common clock) +Note: + * For 'format', 'frame-master', 'bitclock-master', 'bitclock-inversion' and + 'frame-inversion', the simple card will use the settings of CODEC for both + CPU and CODEC sides as we need to keep the settings identical for both ends + of the link. + Example: sound { diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 5dd4769..09591ab 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -157,6 +157,8 @@ static int asoc_simple_card_parse_of(struct device_node *node, 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; int ret; @@ -189,7 +191,7 @@ static int asoc_simple_card_parse_of(struct device_node *node, np = of_get_child_by_name(node, "simple-audio-card,cpu"); if (np) ret = asoc_simple_card_sub_parse_of(np, priv->daifmt, - &priv->cpu_dai, + cpu_dai, &dai_link->cpu_of_node, &dai_link->cpu_dai_name); if (ret < 0) @@ -200,12 +202,18 @@ static int asoc_simple_card_parse_of(struct device_node *node, np = of_get_child_by_name(node, "simple-audio-card,codec"); if (np) ret = asoc_simple_card_sub_parse_of(np, priv->daifmt, - &priv->codec_dai, + codec_dai, &dai_link->codec_of_node, &dai_link->codec_dai_name); if (ret < 0) return ret; + /* + * 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; @@ -227,12 +235,12 @@ static int asoc_simple_card_parse_of(struct device_node *node, dev_dbg(dev, "platform : %04x\n", priv->daifmt); dev_dbg(dev, "cpu : %s / %04x / %d\n", dai_link->cpu_dai_name, - priv->cpu_dai.fmt, - priv->cpu_dai.sysclk); + cpu_dai->fmt, + cpu_dai->sysclk); dev_dbg(dev, "codec : %s / %04x / %d\n", dai_link->codec_dai_name, - priv->codec_dai.fmt, - priv->codec_dai.sysclk); + codec_dai->fmt, + codec_dai->sysclk); /* * soc_bind_dai_link() will check cpu name -- 1.8.4

3 4

[alsa-devel] [PATCH v3] ASoC: cs42888: Add codec driver support
by Nicolin Chen 18 Mar '14

18 Mar '14

This patch adds support for the Cirrus Logic CS42888 Audio CODEC that has four 24-bit A/D and eight 24-bit D/A converters. [ CS42888 supports both I2C and SPI control ports. As initial patch, this patch only adds the support for I2C. ] Signed-off-by: Nicolin Chen <Guangyu.Chen(a)freescale.com> --- Changelog v3: * Refined document at 'clocks' section. v2: * Separated driver into I2C and main part drivers. * Added error out if failed to get clock. * Enabled Auto-mute function as default. * Disabled runtime-pm in remove(). * Added return value checking for regcache_sync(). * Fixed some trival parts like 'const' as Lars mentioned. .../devicetree/bindings/sound/cs42888.txt | 28 ++ sound/soc/codecs/Kconfig | 10 + sound/soc/codecs/Makefile | 4 + sound/soc/codecs/cs42888-i2c.c | 68 +++ sound/soc/codecs/cs42888.c | 536 +++++++++++++++++++++ sound/soc/codecs/cs42888.h | 213 ++++++++ 6 files changed, 859 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/cs42888.txt create mode 100644 sound/soc/codecs/cs42888-i2c.c create mode 100644 sound/soc/codecs/cs42888.c create mode 100644 sound/soc/codecs/cs42888.h diff --git a/Documentation/devicetree/bindings/sound/cs42888.txt b/Documentation/devicetree/bindings/sound/cs42888.txt new file mode 100644 index 0000000..b20bd78 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cs42888.txt @@ -0,0 +1,28 @@ +CS42888 audio CODEC + +Required properties: + + - compatible : "cirrus,cs42888" + + - reg : the I2C address of the device for I2C + + - clocks : a list of phandles + clock-specifiers, one for each entry in + clock-names + + - clock-names : must contain "mclk" + + - VA-supply, VD-supply, VLS-supply, VLC-supply: power supplies for the device, + as covered in Documentation/devicetree/bindings/regulator/regulator.txt + +Example: + +codec: cs42888@48 { + compatible = "cirrus,cs42888"; + reg = <0x48>; + clocks = <&codec_mclk 0>; + clock-names = "mclk"; + VA-supply = <&reg_audio>; + VD-supply = <&reg_audio>; + VLS-supply = <&reg_audio>; + VLC-supply = <&reg_audio>; +}; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 32d7a6f..1642578 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -44,6 +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_CS42888_I2C if I2C select SND_SOC_CX20442 if TTY select SND_SOC_DA7210 if I2C select SND_SOC_DA7213 if I2C @@ -303,6 +304,15 @@ config SND_SOC_CS4271 tristate "Cirrus Logic CS4271 CODEC" depends on SND_SOC_I2C_AND_SPI +config SND_SOC_CS42888 + tristate + +config SND_SOC_CS42888_I2C + tristate "Cirrus Logic CS42888 CODEC (I2C)" + depends on I2C + select SND_SOC_CS42888 + select REGMAP_I2C + config SND_SOC_CX20442 tristate depends on TTY diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index cb46c4c..f5612fb 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -30,6 +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-cs42888-objs := cs42888.o +snd-soc-cs42888-i2c-objs := cs42888-i2c.o snd-soc-cx20442-objs := cx20442.o snd-soc-da7210-objs := da7210.o snd-soc-da7213-objs := da7213.o @@ -178,6 +180,8 @@ 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-$(CONFIG_SND_SOC_CS42888) += snd-soc-cs42888.o +obj-$(CONFIG_SND_SOC_CS42888_I2C) += snd-soc-cs42888-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 diff --git a/sound/soc/codecs/cs42888-i2c.c b/sound/soc/codecs/cs42888-i2c.c new file mode 100644 index 0000000..54813bc --- /dev/null +++ b/sound/soc/codecs/cs42888-i2c.c @@ -0,0 +1,68 @@ +/* + * Cirrus Logic CS42888 Audio CODEC Digital Audio Interface (DAI) I2C driver + * + * Copyright (C) 2014 Freescale Semiconductor, Inc. + * + * Author: Nicolin Chen <Guangyu.Chen(a)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/i2c.h> +#include <linux/module.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 "cs42888.h" + +static int cs42888_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + u32 ret = cs42888_probe(&i2c->dev, + devm_regmap_init_i2c(i2c, &cs42888_regmap_config)); + if (ret) + return ret; + + pm_runtime_enable(&i2c->dev); + pm_request_idle(&i2c->dev); + + return 0; +} + +static int cs42888_i2c_remove(struct i2c_client *i2c) +{ + snd_soc_unregister_codec(&i2c->dev); + pm_runtime_disable(&i2c->dev); + + return 0; +} + +static struct i2c_device_id cs42888_i2c_id[] = { + {"cs42888", 0}, + {} +}; +MODULE_DEVICE_TABLE(i2c, cs42888_i2c_id); + +static struct i2c_driver cs42888_i2c_driver = { + .driver = { + .name = "cs42888", + .owner = THIS_MODULE, + .pm = &cs42888_pm, + }, + .probe = cs42888_i2c_probe, + .remove = cs42888_i2c_remove, + .id_table = cs42888_i2c_id, +}; + +module_i2c_driver(cs42888_i2c_driver); + +MODULE_DESCRIPTION("Cirrus Logic CS42888 ALSA SoC Codec I2C Driver"); +MODULE_AUTHOR("Freescale Semiconductor, Inc."); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs42888.c b/sound/soc/codecs/cs42888.c new file mode 100644 index 0000000..8cd3d4a --- /dev/null +++ b/sound/soc/codecs/cs42888.c @@ -0,0 +1,536 @@ +/* + * Cirrus Logic CS42888 Audio CODEC Digital Audio Interface (DAI) driver + * + * Copyright (C) 2014 Freescale Semiconductor, Inc. + * + * Author: Nicolin Chen <Guangyu.Chen(a)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/pm_runtime.h> +#include <linux/regulator/consumer.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tlv.h> + +#include "cs42888.h" + +#define CS42888_NUM_SUPPLIES 4 +static const char *const cs42888_supply_names[CS42888_NUM_SUPPLIES] = { + "VA", + "VD", + "VLS", + "VLC", +}; + +#define CS42888_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 cs42888_priv { + struct regulator_bulk_data supplies[CS42888_NUM_SUPPLIES]; + 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 cs42888_adc_single[] = { "Differential", "Single-Ended" }; +static const char *const cs42888_szc[] = { "Immediate Change", "Zero Cross", + "Soft Ramp", "Soft Ramp on Zero Cross" }; + +static const struct soc_enum adc1_single_enum = + SOC_ENUM_SINGLE(CS42888_ADCCTL, 4, 2, cs42888_adc_single); +static const struct soc_enum adc2_single_enum = + SOC_ENUM_SINGLE(CS42888_ADCCTL, 3, 2, cs42888_adc_single); +static const struct soc_enum dac_szc_enum = + SOC_ENUM_SINGLE(CS42888_TXCTL, 5, 4, cs42888_szc); +static const struct soc_enum adc_szc_enum = + SOC_ENUM_SINGLE(CS42888_TXCTL, 0, 4, cs42888_szc); + +static const struct snd_kcontrol_new cs42888_snd_controls[] = { + SOC_DOUBLE_R_TLV("DAC1 Playback Volume", CS42888_VOLAOUT1, + CS42888_VOLAOUT2, 0, 0xff, 1, dac_tlv), + SOC_DOUBLE_R_TLV("DAC2 Playback Volume", CS42888_VOLAOUT3, + CS42888_VOLAOUT4, 0, 0xff, 1, dac_tlv), + SOC_DOUBLE_R_TLV("DAC3 Playback Volume", CS42888_VOLAOUT5, + CS42888_VOLAOUT6, 0, 0xff, 1, dac_tlv), + SOC_DOUBLE_R_TLV("DAC4 Playback Volume", CS42888_VOLAOUT7, + CS42888_VOLAOUT8, 0, 0xff, 1, dac_tlv), + SOC_DOUBLE_R_S_TLV("ADC1 Capture Volume", CS42888_VOLAIN1, + CS42888_VOLAIN2, 0, -0x80, 0x30, 7, 0, adc_tlv), + SOC_DOUBLE_R_S_TLV("ADC2 Capture Volume", CS42888_VOLAIN3, + CS42888_VOLAIN4, 0, -0x80, 0x30, 7, 0, adc_tlv), + SOC_DOUBLE("DAC1 Invert Switch", CS42888_DACINV, 0, 1, 1, 0), + SOC_DOUBLE("DAC2 Invert Switch", CS42888_DACINV, 2, 3, 1, 0), + SOC_DOUBLE("DAC3 Invert Switch", CS42888_DACINV, 4, 5, 1, 0), + SOC_DOUBLE("DAC4 Invert Switch", CS42888_DACINV, 6, 7, 1, 0), + SOC_DOUBLE("ADC1 Invert Switch", CS42888_ADCINV, 0, 1, 1, 0), + SOC_DOUBLE("ADC2 Invert Switch", CS42888_ADCINV, 2, 3, 1, 0), + SOC_SINGLE("ADC High-Pass Filter Switch", CS42888_ADCCTL, 7, 1, 1), + SOC_SINGLE("DAC De-emphasis Switch", CS42888_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", CS42888_TXCTL, 7, 1, 0), + SOC_ENUM("DAC Soft Ramp & Zero Cross Control Switch", dac_szc_enum), + SOC_SINGLE("DAC Auto Mute Switch", CS42888_TXCTL, 4, 1, 0), + SOC_SINGLE("Mute ADC Serial Port Switch", CS42888_TXCTL, 3, 1, 0), + SOC_SINGLE("ADC Single Volume Control Switch", CS42888_TXCTL, 2, 1, 0), + SOC_ENUM("ADC Soft Ramp & Zero Cross Control Switch", adc_szc_enum), +}; + +static const struct snd_soc_dapm_widget cs42888_dapm_widgets[] = { + SND_SOC_DAPM_DAC("DAC1", "Playback", CS42888_PWRCTL, 1, 1), + SND_SOC_DAPM_DAC("DAC2", "Playback", CS42888_PWRCTL, 2, 1), + SND_SOC_DAPM_DAC("DAC3", "Playback", CS42888_PWRCTL, 3, 1), + SND_SOC_DAPM_DAC("DAC4", "Playback", CS42888_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", CS42888_PWRCTL, 5, 1), + SND_SOC_DAPM_ADC("ADC2", "Capture", CS42888_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", CS42888_PWRCTL, 0, 1, NULL, 0), +}; + +static const struct snd_soc_dapm_route cs42888_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" }, +}; + +struct cs42888_ratios { + unsigned int ratio; + unsigned char speed; + unsigned char mclk; +}; + +static const struct cs42888_ratios cs42888_ratios[] = { + { 64, CS42888_FM_QUAD, CS42888_FUNCMOD_MFREQ_256(4) }, + { 96, CS42888_FM_QUAD, CS42888_FUNCMOD_MFREQ_384(4) }, + { 128, CS42888_FM_QUAD, CS42888_FUNCMOD_MFREQ_512(4) }, + { 192, CS42888_FM_QUAD, CS42888_FUNCMOD_MFREQ_768(4) }, + { 256, CS42888_FM_SINGLE, CS42888_FUNCMOD_MFREQ_256(1) }, + { 384, CS42888_FM_SINGLE, CS42888_FUNCMOD_MFREQ_384(1) }, + { 512, CS42888_FM_SINGLE, CS42888_FUNCMOD_MFREQ_512(1) }, + { 768, CS42888_FM_SINGLE, CS42888_FUNCMOD_MFREQ_768(1) }, + { 1024, CS42888_FM_SINGLE, CS42888_FUNCMOD_MFREQ_1024(1) } +}; + +static int cs42888_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 cs42888_priv *cs42888 = snd_soc_codec_get_drvdata(codec); + + cs42888->sysclk = freq; + + return 0; +} + +static int cs42888_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int format) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs42888_priv *cs42888 = 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 = CS42888_INTF_DAC_DIF_LEFTJ | CS42888_INTF_ADC_DIF_LEFTJ; + break; + case SND_SOC_DAIFMT_I2S: + val = CS42888_INTF_DAC_DIF_I2S | CS42888_INTF_ADC_DIF_I2S; + break; + case SND_SOC_DAIFMT_RIGHT_J: + val = CS42888_INTF_DAC_DIF_RIGHTJ | CS42888_INTF_ADC_DIF_RIGHTJ; + break; + default: + dev_err(codec->dev, "unsupported dai format\n"); + return -EINVAL; + } + + regmap_update_bits(cs42888->regmap, CS42888_INTF, + CS42888_INTF_DAC_DIF_MASK | + CS42888_INTF_ADC_DIF_MASK, val); + + /* Set master/slave audio interface */ + switch (format & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + cs42888->slave_mode = true; + break; + case SND_SOC_DAIFMT_CBM_CFM: + cs42888->slave_mode = false; + break; + default: + dev_err(codec->dev, "unsupported master/slave mode\n"); + return -EINVAL; + } + + return 0; +} + +static int cs42888_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 cs42888_priv *cs42888 = snd_soc_codec_get_drvdata(codec); + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + u32 ratio = cs42888->sysclk / params_rate(params); + u32 i, fm, val, mask; + + for (i = 0; i < ARRAY_SIZE(cs42888_ratios); i++) { + if (cs42888_ratios[i].ratio == ratio) + break; + } + + if (i == ARRAY_SIZE(cs42888_ratios)) { + dev_err(codec->dev, "unsupported sysclk ratio\n"); + return -EINVAL; + } + + mask = CS42888_FUNCMOD_MFREQ_MASK; + val = cs42888_ratios[i].mclk; + + fm = cs42888->slave_mode ? CS42888_FM_AUTO : cs42888_ratios[i].speed; + + regmap_update_bits(cs42888->regmap, CS42888_FUNCMOD, + CS42888_FUNCMOD_xC_FM_MASK(tx) | mask, + CS42888_FUNCMOD_xC_FM(tx, fm) | val); + + return 0; +} + +static int cs42888_digital_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + struct cs42888_priv *cs42888 = snd_soc_codec_get_drvdata(codec); + + regmap_update_bits(cs42888->regmap, CS42888_DACMUTE, + CS42888_DACMUTE_ALL, mute ? CS42888_DACMUTE_ALL : 0); + + return 0; +} + +static const struct snd_soc_dai_ops cs42888_dai_ops = { + .set_fmt = cs42888_set_dai_fmt, + .set_sysclk = cs42888_set_dai_sysclk, + .hw_params = cs42888_hw_params, + .digital_mute = cs42888_digital_mute, +}; + +static struct snd_soc_dai_driver cs42888_dai = { + .name = "cs42888", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 8, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = CS42888_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 4, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = CS42888_FORMATS, + }, + .ops = &cs42888_dai_ops, +}; + +static const struct reg_default cs42888_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 */ + { 0x17, 0x00 }, /* ADC Channel Invert */ + { 0x18, 0x00 }, /* Status Control */ + { 0x1a, 0x00 }, /* Status Mask */ + { 0x1b, 0x00 }, /* MUTEC Pin Control */ +}; + +static bool cs42888_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS42888_STATUS: + return true; + default: + return false; + } +} + +static bool cs42888_writeable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS42888_CHIPID: + case CS42888_STATUS: + return false; + default: + return true; + } +} + +const struct regmap_config cs42888_regmap_config = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = CS42888_LASTREG, + .reg_defaults = cs42888_reg, + .num_reg_defaults = ARRAY_SIZE(cs42888_reg), + .volatile_reg = cs42888_volatile_register, + .writeable_reg = cs42888_writeable_register, + .cache_type = REGCACHE_RBTREE, +}; +EXPORT_SYMBOL_GPL(cs42888_regmap_config); + +static int cs42888_codec_probe(struct snd_soc_codec *codec) +{ + struct cs42888_priv *cs42888 = snd_soc_codec_get_drvdata(codec); + + /* Mute all DAC channels */ + regmap_write(cs42888->regmap, CS42888_DACMUTE, CS42888_DACMUTE_ALL); + + return 0; +} + +static const struct snd_soc_codec_driver cs42888_driver = { + .probe = cs42888_codec_probe, + .idle_bias_off = true, + + .controls = cs42888_snd_controls, + .num_controls = ARRAY_SIZE(cs42888_snd_controls), + .dapm_widgets = cs42888_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cs42888_dapm_widgets), + .dapm_routes = cs42888_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(cs42888_dapm_routes), +}; + +int cs42888_probe(struct device *dev, struct regmap *regmap) +{ + struct cs42888_priv *cs42888; + int ret, val, i; + + cs42888 = devm_kzalloc(dev, sizeof(*cs42888), GFP_KERNEL); + if (cs42888 == NULL) + return -ENOMEM; + + dev_set_drvdata(dev, cs42888); + + cs42888->clk = devm_clk_get(dev, "mclk"); + if (IS_ERR(cs42888->clk)) { + dev_err(dev, "failed to get the clock: %ld\n", + PTR_ERR(cs42888->clk)); + return -EINVAL; + } + + cs42888->sysclk = clk_get_rate(cs42888->clk); + + for (i = 0; i < ARRAY_SIZE(cs42888->supplies); i++) + cs42888->supplies[i].supply = cs42888_supply_names[i]; + + ret = devm_regulator_bulk_get(dev, + ARRAY_SIZE(cs42888->supplies), cs42888->supplies); + if (ret) { + dev_err(dev, "failed to request supplies: %d\n", ret); + return ret; + } + + ret = regulator_bulk_enable(ARRAY_SIZE(cs42888->supplies), + cs42888->supplies); + if (ret) { + dev_err(dev, "failed to enable supplies: %d\n", ret); + return ret; + } + + /* Make sure hardware reset done */ + msleep(5); + + cs42888->regmap = regmap; + if (IS_ERR(cs42888->regmap)) { + ret = PTR_ERR(cs42888->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(cs42888->regmap, true); + + /* Validate the chip ID */ + regmap_read(cs42888->regmap, CS42888_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 & CS42888_CHIPID_CHIP_ID_MASK) != 0x00) { + dev_err(dev, "unmatched chip ID: %d\n", + val & CS42888_CHIPID_CHIP_ID_MASK); + ret = -EINVAL; + goto err_enable; + } + + dev_info(dev, "found device, revision %X\n", + val & CS42888_CHIPID_REV_ID_MASK); + + regcache_cache_bypass(cs42888->regmap, false); + + ret = snd_soc_register_codec(dev, &cs42888_driver, &cs42888_dai, 1); + if (ret) { + dev_err(dev, "failed to register codec:%d\n", ret); + goto err_enable; + } + + regcache_cache_only(cs42888->regmap, true); + +err_enable: + regulator_bulk_disable(ARRAY_SIZE(cs42888->supplies), + cs42888->supplies); + + return ret; +} +EXPORT_SYMBOL_GPL(cs42888_probe); + +#ifdef CONFIG_PM_RUNTIME +static int cs42888_runtime_resume(struct device *dev) +{ + struct cs42888_priv *cs42888 = dev_get_drvdata(dev); + int ret; + + ret = clk_prepare_enable(cs42888->clk); + if (ret) { + dev_err(dev, "failed to enable mclk: %d\n", ret); + return ret; + } + + ret = regulator_bulk_enable(ARRAY_SIZE(cs42888->supplies), + cs42888->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(cs42888->regmap, false); + + ret = regcache_sync(cs42888->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(cs42888->supplies), + cs42888->supplies); +err_clk: + clk_disable_unprepare(cs42888->clk); + + return ret; +} + +static int cs42888_runtime_suspend(struct device *dev) +{ + struct cs42888_priv *cs42888 = dev_get_drvdata(dev); + + regcache_cache_only(cs42888->regmap, true); + + regulator_bulk_disable(ARRAY_SIZE(cs42888->supplies), + cs42888->supplies); + + clk_disable_unprepare(cs42888->clk); + + return 0; +} +#endif + +const struct dev_pm_ops cs42888_pm = { + SET_RUNTIME_PM_OPS(cs42888_runtime_suspend, cs42888_runtime_resume, NULL) +}; +EXPORT_SYMBOL_GPL(cs42888_pm); + +const struct of_device_id cs42888_of_match[] = { + { .compatible = "cirrus,cs42888", }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, cs42888_of_match); +EXPORT_SYMBOL_GPL(cs42888_of_match); + +MODULE_DESCRIPTION("Cirrus Logic CS42888 ALSA SoC Codec Driver"); +MODULE_AUTHOR("Freescale Semiconductor, Inc."); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs42888.h b/sound/soc/codecs/cs42888.h new file mode 100644 index 0000000..eda55ac --- /dev/null +++ b/sound/soc/codecs/cs42888.h @@ -0,0 +1,213 @@ +/* + * cs42888.h - Cirrus Logic CS42888 Audio CODEC driver header file + * + * Copyright (C) 2014 Freescale Semiconductor, Inc. + * + * Author: Nicolin Chen <Guangyu.Chen(a)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 _CS42888_H +#define _CS42888_H + +extern const struct dev_pm_ops cs42888_pm; +extern const struct regmap_config cs42888_regmap_config; +int cs42888_probe(struct device *dev, struct regmap *regmap); + +/* CS42888 register map */ +#define CS42888_CHIPID 0x01 /* Chip ID */ +#define CS42888_PWRCTL 0x02 /* Power Control */ +#define CS42888_FUNCMOD 0x03 /* Functional Mode */ +#define CS42888_INTF 0x04 /* Interface Formats */ +#define CS42888_ADCCTL 0x05 /* ADC Control */ +#define CS42888_TXCTL 0x06 /* Transition Control */ +#define CS42888_DACMUTE 0x07 /* DAC Mute Control */ +#define CS42888_VOLAOUT1 0x08 /* Volume Control AOUT1 */ +#define CS42888_VOLAOUT2 0x09 /* Volume Control AOUT2 */ +#define CS42888_VOLAOUT3 0x0A /* Volume Control AOUT3 */ +#define CS42888_VOLAOUT4 0x0B /* Volume Control AOUT4 */ +#define CS42888_VOLAOUT5 0x0C /* Volume Control AOUT5 */ +#define CS42888_VOLAOUT6 0x0D /* Volume Control AOUT6 */ +#define CS42888_VOLAOUT7 0x0E /* Volume Control AOUT7 */ +#define CS42888_VOLAOUT8 0x0F /* Volume Control AOUT8 */ +#define CS42888_DACINV 0x10 /* DAC Channel Invert */ +#define CS42888_VOLAIN1 0x11 /* Volume Control AIN1 */ +#define CS42888_VOLAIN2 0x12 /* Volume Control AIN2 */ +#define CS42888_VOLAIN3 0x13 /* Volume Control AIN3 */ +#define CS42888_VOLAIN4 0x14 /* Volume Control AIN4 */ +#define CS42888_ADCINV 0x17 /* ADC Channel Invert */ +#define CS42888_STATUSCTL 0x18 /* Status Control */ +#define CS42888_STATUS 0x19 /* Status */ +#define CS42888_STATUSM 0x1A /* Status Mask */ +#define CS42888_MUTEC 0x1B /* MUTEC Pin Control */ + +#define CS42888_FIRSTREG CS42888_CHIPID +#define CS42888_LASTREG CS42888_MUTEC +#define CS42888_NUMREGS (CS42888_LASTREG - CS42888_FIRSTREG + 1) +#define CS42888_I2C_INCR 0x80 + +/* Chip I.D. and Revision Register (Address 01h) */ +#define CS42888_CHIPID_CHIP_ID_MASK 0xF0 +#define CS42888_CHIPID_REV_ID_MASK 0x0F + +/* Power Control (Address 02h) */ +#define CS42888_PWRCTL_PDN_ADC2_SHIFT 6 +#define CS42888_PWRCTL_PDN_ADC2_MASK (1 << CS42888_PWRCTL_PDN_ADC2_SHIFT) +#define CS42888_PWRCTL_PDN_ADC2 (1 << CS42888_PWRCTL_PDN_ADC2_SHIFT) +#define CS42888_PWRCTL_PDN_ADC1_SHIFT 5 +#define CS42888_PWRCTL_PDN_ADC1_MASK (1 << CS42888_PWRCTL_PDN_ADC1_SHIFT) +#define CS42888_PWRCTL_PDN_ADC1 (1 << CS42888_PWRCTL_PDN_ADC1_SHIFT) +#define CS42888_PWRCTL_PDN_DAC4_SHIFT 4 +#define CS42888_PWRCTL_PDN_DAC4_MASK (1 << CS42888_PWRCTL_PDN_DAC4_SHIFT) +#define CS42888_PWRCTL_PDN_DAC4 (1 << CS42888_PWRCTL_PDN_DAC4_SHIFT) +#define CS42888_PWRCTL_PDN_DAC3_SHIFT 3 +#define CS42888_PWRCTL_PDN_DAC3_MASK (1 << CS42888_PWRCTL_PDN_DAC3_SHIFT) +#define CS42888_PWRCTL_PDN_DAC3 (1 << CS42888_PWRCTL_PDN_DAC3_SHIFT) +#define CS42888_PWRCTL_PDN_DAC2_SHIFT 2 +#define CS42888_PWRCTL_PDN_DAC2_MASK (1 << CS42888_PWRCTL_PDN_DAC2_SHIFT) +#define CS42888_PWRCTL_PDN_DAC2 (1 << CS42888_PWRCTL_PDN_DAC2_SHIFT) +#define CS42888_PWRCTL_PDN_DAC1_SHIFT 1 +#define CS42888_PWRCTL_PDN_DAC1_MASK (1 << CS42888_PWRCTL_PDN_DAC1_SHIFT) +#define CS42888_PWRCTL_PDN_DAC1 (1 << CS42888_PWRCTL_PDN_DAC1_SHIFT) +#define CS42888_PWRCTL_PDN_SHIFT 0 +#define CS42888_PWRCTL_PDN_MASK (1 << CS42888_PWRCTL_PDN_SHIFT) +#define CS42888_PWRCTL_PDN (1 << CS42888_PWRCTL_PDN_SHIFT) + +/* Functional Mode (Address 03h) */ +#define CS42888_FUNCMOD_DAC_FM_SHIFT 6 +#define CS42888_FUNCMOD_DAC_FM_WIDTH 2 +#define CS42888_FUNCMOD_DAC_FM_MASK (((1 << CS42888_FUNCMOD_DAC_FM_WIDTH) - 1) << CS42888_FUNCMOD_DAC_FM_SHIFT) +#define CS42888_FUNCMOD_DAC_FM(v) ((v) << CS42888_FUNCMOD_DAC_FM_SHIFT) +#define CS42888_FUNCMOD_ADC_FM_SHIFT 4 +#define CS42888_FUNCMOD_ADC_FM_WIDTH 2 +#define CS42888_FUNCMOD_ADC_FM_MASK (((1 << CS42888_FUNCMOD_ADC_FM_WIDTH) - 1) << CS42888_FUNCMOD_ADC_FM_SHIFT) +#define CS42888_FUNCMOD_ADC_FM(v) ((v) << CS42888_FUNCMOD_ADC_FM_SHIFT) +#define CS42888_FUNCMOD_xC_FM_MASK(x) ((x) ? CS42888_FUNCMOD_DAC_FM_MASK : CS42888_FUNCMOD_ADC_FM_MASK) +#define CS42888_FUNCMOD_xC_FM(x, v) ((x) ? CS42888_FUNCMOD_DAC_FM(v) : CS42888_FUNCMOD_ADC_FM(v)) +#define CS42888_FUNCMOD_MFREQ_SHIFT 1 +#define CS42888_FUNCMOD_MFREQ_WIDTH 3 +#define CS42888_FUNCMOD_MFREQ_MASK (((1 << CS42888_FUNCMOD_MFREQ_WIDTH) - 1) << CS42888_FUNCMOD_MFREQ_SHIFT) +#define CS42888_FUNCMOD_MFREQ_256(s) ((0 << CS42888_FUNCMOD_MFREQ_SHIFT) >> (s >> 1)) +#define CS42888_FUNCMOD_MFREQ_384(s) ((1 << CS42888_FUNCMOD_MFREQ_SHIFT) >> (s >> 1)) +#define CS42888_FUNCMOD_MFREQ_512(s) ((2 << CS42888_FUNCMOD_MFREQ_SHIFT) >> (s >> 1)) +#define CS42888_FUNCMOD_MFREQ_768(s) ((3 << CS42888_FUNCMOD_MFREQ_SHIFT) >> (s >> 1)) +#define CS42888_FUNCMOD_MFREQ_1024(s) ((4 << CS42888_FUNCMOD_MFREQ_SHIFT) >> (s >> 1)) + +#define CS42888_FM_SINGLE 0 +#define CS42888_FM_DOUBLE 1 +#define CS42888_FM_QUAD 2 +#define CS42888_FM_AUTO 3 + +/* Interface Formats (Address 04h) */ +#define CS42888_INTF_FREEZE_SHIFT 7 +#define CS42888_INTF_FREEZE_MASK (1 << CS42888_INTF_FREEZE_SHIFT) +#define CS42888_INTF_FREEZE (1 << CS42888_INTF_FREEZE_SHIFT) +#define CS42888_INTF_AUX_DIF_SHIFT 6 +#define CS42888_INTF_AUX_DIF_MASK (1 << CS42888_INTF_AUX_DIF_SHIFT) +#define CS42888_INTF_AUX_DIF (1 << CS42888_INTF_AUX_DIF_SHIFT) +#define CS42888_INTF_DAC_DIF_SHIFT 3 +#define CS42888_INTF_DAC_DIF_WIDTH 3 +#define CS42888_INTF_DAC_DIF_MASK (((1 << CS42888_INTF_DAC_DIF_WIDTH) - 1) << CS42888_INTF_DAC_DIF_SHIFT) +#define CS42888_INTF_DAC_DIF_LEFTJ (0 << CS42888_INTF_DAC_DIF_SHIFT) +#define CS42888_INTF_DAC_DIF_I2S (1 << CS42888_INTF_DAC_DIF_SHIFT) +#define CS42888_INTF_DAC_DIF_RIGHTJ (2 << CS42888_INTF_DAC_DIF_SHIFT) +#define CS42888_INTF_DAC_DIF_RIGHTJ_16 (3 << CS42888_INTF_DAC_DIF_SHIFT) +#define CS42888_INTF_DAC_DIF_ONELINE_20 (4 << CS42888_INTF_DAC_DIF_SHIFT) +#define CS42888_INTF_DAC_DIF_ONELINE_24 (6 << CS42888_INTF_DAC_DIF_SHIFT) +#define CS42888_INTF_DAC_DIF_TDM (7 << CS42888_INTF_DAC_DIF_SHIFT) +#define CS42888_INTF_ADC_DIF_SHIFT 0 +#define CS42888_INTF_ADC_DIF_WIDTH 3 +#define CS42888_INTF_ADC_DIF_MASK (((1 << CS42888_INTF_ADC_DIF_WIDTH) - 1) << CS42888_INTF_ADC_DIF_SHIFT) +#define CS42888_INTF_ADC_DIF_LEFTJ (0 << CS42888_INTF_ADC_DIF_SHIFT) +#define CS42888_INTF_ADC_DIF_I2S (1 << CS42888_INTF_ADC_DIF_SHIFT) +#define CS42888_INTF_ADC_DIF_RIGHTJ (2 << CS42888_INTF_ADC_DIF_SHIFT) +#define CS42888_INTF_ADC_DIF_RIGHTJ_16 (3 << CS42888_INTF_ADC_DIF_SHIFT) +#define CS42888_INTF_ADC_DIF_ONELINE_20 (4 << CS42888_INTF_ADC_DIF_SHIFT) +#define CS42888_INTF_ADC_DIF_ONELINE_24 (6 << CS42888_INTF_ADC_DIF_SHIFT) +#define CS42888_INTF_ADC_DIF_TDM (7 << CS42888_INTF_ADC_DIF_SHIFT) + +/* ADC Control & DAC De-Emphasis (Address 05h) */ +#define CS42888_ADCCTL_ADC_HPF_FREEZE_SHIFT 7 +#define CS42888_ADCCTL_ADC_HPF_FREEZE_MASK (1 << CS42888_ADCCTL_ADC_HPF_FREEZE_SHIFT) +#define CS42888_ADCCTL_ADC_HPF_FREEZE (1 << CS42888_ADCCTL_ADC_HPF_FREEZE_SHIFT) +#define CS42888_ADCCTL_DAC_DEM_SHIFT 5 +#define CS42888_ADCCTL_DAC_DEM_MASK (1 << CS42888_ADCCTL_DAC_DEM_SHIFT) +#define CS42888_ADCCTL_DAC_DEM (1 << CS42888_ADCCTL_DAC_DEM_SHIFT) +#define CS42888_ADCCTL_ADC1_SINGLE_SHIFT 4 +#define CS42888_ADCCTL_ADC1_SINGLE_MASK (1 << CS42888_ADCCTL_ADC1_SINGLE_SHIFT) +#define CS42888_ADCCTL_ADC1_SINGLE (1 << CS42888_ADCCTL_ADC1_SINGLE_SHIFT) +#define CS42888_ADCCTL_ADC2_SINGLE_SHIFT 3 +#define CS42888_ADCCTL_ADC2_SINGLE_MASK (1 << CS42888_ADCCTL_ADC2_SINGLE_SHIFT) +#define CS42888_ADCCTL_ADC2_SINGLE (1 << CS42888_ADCCTL_ADC2_SINGLE_SHIFT) + +/* Transition Control (Address 06h) */ +#define CS42888_TXCTL_DAC_SNGVOL_SHIFT 7 +#define CS42888_TXCTL_DAC_SNGVOL_MASK (1 << CS42888_TXCTL_DAC_SNGVOL_SHIFT) +#define CS42888_TXCTL_DAC_SNGVOL (1 << CS42888_TXCTL_DAC_SNGVOL_SHIFT) +#define CS42888_TXCTL_DAC_SZC_SHIFT 5 +#define CS42888_TXCTL_DAC_SZC_WIDTH 2 +#define CS42888_TXCTL_DAC_SZC_MASK (((1 << CS42888_TXCTL_DAC_SZC_WIDTH) - 1) << CS42888_TXCTL_DAC_SZC_SHIFT) +#define CS42888_TXCTL_DAC_SZC_IC (0 << CS42888_TXCTL_DAC_SZC_SHIFT) +#define CS42888_TXCTL_DAC_SZC_ZC (1 << CS42888_TXCTL_DAC_SZC_SHIFT) +#define CS42888_TXCTL_DAC_SZC_SR (2 << CS42888_TXCTL_DAC_SZC_SHIFT) +#define CS42888_TXCTL_DAC_SZC_SRZC (3 << CS42888_TXCTL_DAC_SZC_SHIFT) +#define CS42888_TXCTL_AMUTE_SHIFT 4 +#define CS42888_TXCTL_AMUTE_MASK (1 << CS42888_TXCTL_AMUTE_SHIFT) +#define CS42888_TXCTL_AMUTE (1 << CS42888_TXCTL_AMUTE_SHIFT) +#define CS42888_TXCTL_MUTE_ADC_SP_SHIFT 3 +#define CS42888_TXCTL_MUTE_ADC_SP_MASK (1 << CS42888_TXCTL_MUTE_ADC_SP_SHIFT) +#define CS42888_TXCTL_MUTE_ADC_SP (1 << CS42888_TXCTL_MUTE_ADC_SP_SHIFT) +#define CS42888_TXCTL_ADC_SNGVOL_SHIFT 2 +#define CS42888_TXCTL_ADC_SNGVOL_MASK (1 << CS42888_TXCTL_ADC_SNGVOL_SHIFT) +#define CS42888_TXCTL_ADC_SNGVOL (1 << CS42888_TXCTL_ADC_SNGVOL_SHIFT) +#define CS42888_TXCTL_ADC_SZC_SHIFT 0 +#define CS42888_TXCTL_ADC_SZC_MASK (((1 << CS42888_TXCTL_ADC_SZC_WIDTH) - 1) << CS42888_TXCTL_ADC_SZC_SHIFT) +#define CS42888_TXCTL_ADC_SZC_IC (0 << CS42888_TXCTL_ADC_SZC_SHIFT) +#define CS42888_TXCTL_ADC_SZC_ZC (1 << CS42888_TXCTL_ADC_SZC_SHIFT) +#define CS42888_TXCTL_ADC_SZC_SR (2 << CS42888_TXCTL_ADC_SZC_SHIFT) +#define CS42888_TXCTL_ADC_SZC_SRZC (3 << CS42888_TXCTL_ADC_SZC_SHIFT) + +/* DAC Channel Mute (Address 07h) */ +#define CS42888_DACMUTE_AOUT(n) (0x1 << n) +#define CS42888_DACMUTE_ALL 0xff + +/* Status Control (Address 18h)*/ +#define CS42888_STATUSCTL_INI_SHIFT 2 +#define CS42888_STATUSCTL_INI_WIDTH 2 +#define CS42888_STATUSCTL_INI_MASK (((1 << CS42888_STATUSCTL_INI_WIDTH) - 1) << CS42888_STATUSCTL_INI_SHIFT) +#define CS42888_STATUSCTL_INT_ACTIVE_HIGH (0 << CS42888_STATUSCTL_INI_SHIFT) +#define CS42888_STATUSCTL_INT_ACTIVE_LOW (1 << CS42888_STATUSCTL_INI_SHIFT) +#define CS42888_STATUSCTL_INT_OPEN_DRAIN (2 << CS42888_STATUSCTL_INI_SHIFT) + +/* Status (Address 19h)*/ +#define CS42888_STATUS_DAC_CLK_ERR_SHIFT 4 +#define CS42888_STATUS_DAC_CLK_ERR_MASK (1 << CS42888_STATUS_DAC_CLK_ERR_SHIFT) +#define CS42888_STATUS_ADC_CLK_ERR_SHIFT 3 +#define CS42888_STATUS_ADC_CLK_ERR_MASK (1 << CS42888_STATUS_ADC_CLK_ERR_SHIFT) +#define CS42888_STATUS_ADC2_OVFL_SHIFT 1 +#define CS42888_STATUS_ADC2_OVFL_MASK (1 << CS42888_STATUS_ADC2_OVFL_SHIFT) +#define CS42888_STATUS_ADC1_OVFL_SHIFT 0 +#define CS42888_STATUS_ADC1_OVFL_MASK (1 << CS42888_STATUS_ADC1_OVFL_SHIFT) + +/* Status Mask (Address 1Ah) */ +#define CS42888_STATUS_DAC_CLK_ERR_M_SHIFT 4 +#define CS42888_STATUS_DAC_CLK_ERR_M_MASK (1 << CS42888_STATUS_DAC_CLK_ERR_M_SHIFT) +#define CS42888_STATUS_ADC_CLK_ERR_M_SHIFT 3 +#define CS42888_STATUS_ADC_CLK_ERR_M_MASK (1 << CS42888_STATUS_ADC_CLK_ERR_M_SHIFT) +#define CS42888_STATUS_ADC2_OVFL_M_SHIFT 1 +#define CS42888_STATUS_ADC2_OVFL_M_MASK (1 << CS42888_STATUS_ADC2_OVFL_M_SHIFT) +#define CS42888_STATUS_ADC1_OVFL_M_SHIFT 0 +#define CS42888_STATUS_ADC1_OVFL_M_MASK (1 << CS42888_STATUS_ADC1_OVFL_M_SHIFT) + +/* MUTEC Pin Control (Address 1Bh) */ +#define CS42888_MUTEC_MCPOLARITY_SHIFT 1 +#define CS42888_MUTEC_MCPOLARITY_MASK (1 << CS42888_MUTEC_MCPOLARITY_SHIFT) +#define CS42888_MUTEC_MCPOLARITY_ACTIVE_LOW (0 << CS42888_MUTEC_MCPOLARITY_SHIFT) +#define CS42888_MUTEC_MCPOLARITY_ACTIVE_HIGH (1 << CS42888_MUTEC_MCPOLARITY_SHIFT) +#define CS42888_MUTEC_MUTEC_ACTIVE_SHIFT 0 +#define CS42888_MUTEC_MUTEC_ACTIVE_MASK (1 << CS42888_MUTEC_MUTEC_ACTIVE_SHIFT) +#define CS42888_MUTEC_MUTEC_ACTIVE (1 << CS42888_MUTEC_MUTEC_ACTIVE_SHIFT) +#endif /* _CS42888_H */ -- 1.8.4

4 5

[alsa-devel] [PATCH] ASoC: cs42l52: Fix mask bits for SOC_VALUE_ENUM_SINGLE
by Brian Austin 18 Mar '14

18 Mar '14

The mask bits values were wrong for the SOC_VALUE_ENUM_SINGLE for the PCM/ADC Swap controls Reported-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Brian Austin <brian.austin(a)cirrus.com> --- sound/soc/codecs/cs42l52.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index be2ba1b..ab3ac7b 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -352,7 +352,7 @@ 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); @@ -361,7 +361,7 @@ 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); @@ -370,7 +370,7 @@ 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); @@ -379,7 +379,7 @@ 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); -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ASoC: cs42l73: Fix mask bits for SOC_VALUE_ENUM_SINGLE
by Brian Austin 18 Mar '14

18 Mar '14

The mask bits values were wrong for the SOC_VALUE_ENUM_SINGLE for the mono mix controls. Reported-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Brian Austin <brian.austin(a)cirrus.com> --- sound/soc/codecs/cs42l73.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c index 61c0f64..4cb7652 100644 --- a/sound/soc/codecs/cs42l73.c +++ b/sound/soc/codecs/cs42l73.c @@ -325,7 +325,7 @@ 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); @@ -343,7 +343,7 @@ 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); @@ -352,7 +352,7 @@ 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); -- 1.7.9.5

2 1

[alsa-devel] [PATCH v2 0/7] Audio support for Armada 370 DB
by Thomas Petazzoni 18 Mar '14

18 Mar '14

Hello, This series of patches enable audio support on the Marvell Armada 370 Development Board. Since both the I2S controller on the SoC side and the I2C audio codec are already supported by the kernel, the amount of work is fairly limited. Changes since v1: * Drop patches "sound: codec: add Device Tree binding to cs42l51" and " sound: soc: enable Kirkwood driver for mvebu platforms" since they have been applied by Mark Brown. * Set DAI format directly in the snd_soc_dai_link structure instead of separately calling snd_soc_dai_set_fmt(). Suggested by Mark Brown. * Do not call snd_soc_dapm_enable_pin(), as Mark Brown mentionned that DAPM widgets default to enabled. It allowed to entirely remove the dai_init operation. * Use devm_snd_soc_register_card() instead of snd_soc_register_card(), as suggested by Mark Brown. * Wrote a Device Tree binding document for the DT binding introduced by this driver. Requested by Mark Brown. * Use the DT to look up the controller and the codec, as suggested by Mark Brown. * Add S/PDIF support, which has been successfully tested. I've kept it as separate patches (for both the ASoC machine driver, and the Device Tree bits), so that it can be reviewed and applied separately. * Modified the commit title so that they contain "ASoC", as requested by Mark Brown. Patches 1 and 2 are to be reviewed/applied by the ASoC maintainer Mark Brown, while patches 3 to 7 are to be reviewed/applied by the ARM mvebu maintainers. Note that the audio support for Armada 370 also needs a fix to the CS42L51, which is being discussed with the author of the change that apparently introduced the problem (see discussion at http://mailman.alsa-project.org/pipermail/alsa-devel/2014-January/071916.ht…). Thanks, Thomas Thomas Petazzoni (7): sound: ASoC: add ASoC board driver for Armada 370 DB sound: ASoC: add S/PDIF support to Armada 370 DB ASoC driver ARM: mvebu: add audio I2S controller to Armada 370 Device Tree ARM: mvebu: add I2C0 muxing option for Armada 370 SoC ARM: mvebu: add audio support to Armada 370 DB ARM: mvebu: enable S/PDIF audio in Armada 370 DB Device Tree ARM: mvebu: enable audio options in mvebu_defconfig .../bindings/sound/armada-370db-audio.txt | 27 ++++ arch/arm/boot/dts/armada-370-db.dts | 56 ++++++++ arch/arm/boot/dts/armada-370.dtsi | 28 ++++ arch/arm/configs/mvebu_defconfig | 5 + sound/soc/kirkwood/Kconfig | 9 ++ sound/soc/kirkwood/Makefile | 2 + sound/soc/kirkwood/armada-370-db.c | 148 +++++++++++++++++++++ 7 files changed, 275 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/armada-370db-audio.txt create mode 100644 sound/soc/kirkwood/armada-370-db.c -- 1.8.3.2

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[alsa-devel] [RFC] ASoC: samsung: don't build pcm and spdif on exynos
by Arnd Bergmann 18 Mar '14

18 Mar '14

In commit d37bdf736d9b "ASoC: samsung: Use ASoC dmaengine code where possible", Mark Brown changed the EXYNOS ASoC code to no longer use the private dma interfaces that are standing in the way of multiplatform enablement. However, in 313367e7bfa "ASoC: Samsung: Update Kconfig for I2S,SPDIF and PCM audio", two new users of that interface appeared, since it is now possible to enable SND_SOC_SAMSUNG_SMDK_SPDIF and SND_SOC_SMDK_WM8994_PCM on all Samsung SoCs including EXYNOS. This patch does a partial revert of 313367e7bfa by allowing these drivers on all samsung platforms except EXYNOS, so we can proceed with the multiplatform patches. If support for these drivers is actually needed on EXYNOS machines, the drivers should probably first be moved over to use the dmaengine code. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Cc: Sangsu Park <sangsu4u.park(a)samsung.com> Cc: Sangbeom Kim <sbkim73(a)samsung.com> Cc: Padmavathi Venna <padma.v(a)samsung.com> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- Padma, could you comment on this? When you did the patch that introduced the usage of these drivers on EXYNOS, did you actually need that for a particular machine that would get broken by this revert? diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 7b610a8..efc01e0 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -167,7 +167,7 @@ config SND_SOC_GONI_AQUILA_WM8994 config SND_SOC_SAMSUNG_SMDK_SPDIF tristate "SoC S/PDIF Audio support for SMDK" - depends on SND_SOC_SAMSUNG + depends on SND_SOC_SAMSUNG && !ARCH_EXYNOS select SND_SAMSUNG_SPDIF help Say Y if you want to add support for SoC S/PDIF audio on the SMDK. @@ -183,7 +183,7 @@ config SND_SOC_SMDK_WM8580_PCM config SND_SOC_SMDK_WM8994_PCM tristate "SoC PCM Audio support for WM8994 on SMDK" - depends on SND_SOC_SAMSUNG + depends on SND_SOC_SAMSUNG && !ARCH_EXYNOS depends on I2C=y select MFD_WM8994 select SND_SOC_WM8994

1 0

[alsa-devel] [PATCH] ASoC: codecs: Replace instances of rtd->codec with dai->codec
by Lars-Peter Clausen 18 Mar '14

18 Mar '14

With CODEC to CODEC links rtd->codec does not necessarily point to the driver's CODEC. CODEC drivers should always use dai->codec and never even look at the PCM runtime. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/isabelle.c | 3 +-- sound/soc/codecs/mc13783.c | 6 ++---- sound/soc/codecs/rt5640.c | 3 +-- sound/soc/codecs/sta529.c | 3 +-- sound/soc/codecs/tlv320aic31xx.c | 5 ++--- sound/soc/codecs/uda134x.c | 3 +-- sound/soc/codecs/uda1380.c | 3 +-- sound/soc/codecs/wm8580.c | 3 +-- 8 files changed, 10 insertions(+), 19 deletions(-) diff --git a/sound/soc/codecs/isabelle.c b/sound/soc/codecs/isabelle.c index 3e264a7..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; diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c index 37d737e..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; diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 13ccee4..0061ae6 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -1594,8 +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; diff --git a/sound/soc/codecs/sta529.c b/sound/soc/codecs/sta529.c index a3c61d3..a40c4b0 100644 --- a/sound/soc/codecs/sta529.c +++ b/sound/soc/codecs/sta529.c @@ -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; diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index d3517a9..cc1c9c2 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -753,10 +753,9 @@ static int aic31xx_setup_pll(struct snd_soc_codec *codec, static int aic31xx_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, - struct snd_soc_dai *tmp) + 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; u8 data = 0; dev_dbg(codec->dev, "## %s: format %d width %d rate %d\n", 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 4dadaa8..e62e707 100644 --- a/sound/soc/codecs/uda1380.c +++ b/sound/soc/codecs/uda1380.c @@ -566,8 +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/wm8580.c b/sound/soc/codecs/wm8580.c index 7558c83..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; -- 1.8.0

4 3

[alsa-devel] [PATCH v2 0/4] ASoC: davinci: edma dmaengine PCM and mcasp preparation
by Peter Ujfalusi 18 Mar '14

18 Mar '14

Hi, This is a cutdown version of the previous [1] series containing only the ASoC part of the edma dmaengine conversion, but without the actual switch to this new edma-pcm platform for the McASP since the edma arch and dmaengine core changes are needed for it to be usable. Changes since v1: - arch and dmaengine patches has been dropped from the ASoC series - Patches has been reordered so only one patch is going to be needed to switch AM335x/AM447x to use the edma-pcm platform driver - all comments from Lars-Peter Clausen has been addressed for the edma-pcm driver - no changes to Kconfig/Makefile - do not yet build the edma-pcm driver. It will be enabled and integrated when we can do the real switch. [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2014-March/074269.html Regards, Peter --- Peter Ujfalusi (4): ASoC: davinci: Add edma dmaengine platform driver ASoC: davinci-mcasp: Provide correct filter_data for dmaengine for non-DT boot ASoC: davinci-mcasp: Constraint on the period and buffer size based on FIFO usage ASoC: davinci-mcasp: Assign the dma_data earlier in dai_probe callback sound/soc/davinci/davinci-mcasp.c | 59 +++++++++++++++++++++++++++++++++------ sound/soc/davinci/edma-pcm.c | 57 +++++++++++++++++++++++++++++++++++++ sound/soc/davinci/edma-pcm.h | 25 +++++++++++++++++ 3 files changed, 132 insertions(+), 9 deletions(-) create mode 100644 sound/soc/davinci/edma-pcm.c create mode 100644 sound/soc/davinci/edma-pcm.h -- 1.9.0

3 13

[alsa-devel] [PATCH v2] ASoC: tlv320aic31xx: Don't call kfree for memory allocated by devm_kzalloc
by Axel Lin 18 Mar '14

18 Mar '14

The kfree call is not necessary, but we need to call snd_soc_unregister_codec() in remove(). Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/tlv320aic31xx.c | 11 +++-------- 1 file changed, 3 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index d3517a9..d5d6213 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -1228,7 +1228,6 @@ static int aic31xx_i2c_probe(struct i2c_client *i2c, 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", @@ -1241,18 +1240,14 @@ static int aic31xx_i2c_probe(struct i2c_client *i2c, aic31xx_device_init(aic31xx); - ret = snd_soc_register_codec(&i2c->dev, &soc_codec_driver_aic31xx, + return snd_soc_register_codec(&i2c->dev, &soc_codec_driver_aic31xx, aic31xx_dai_driver, ARRAY_SIZE(aic31xx_dai_driver)); - - return ret; } static int aic31xx_i2c_remove(struct i2c_client *i2c) { - struct aic31xx_priv *aic31xx = dev_get_drvdata(&i2c->dev); - - kfree(aic31xx); + snd_soc_unregister_codec(&i2c->dev); return 0; } @@ -1274,7 +1269,7 @@ static struct i2c_driver aic31xx_i2c_driver = { .of_match_table = of_match_ptr(tlv320aic31xx_of_match), }, .probe = aic31xx_i2c_probe, - .remove = (aic31xx_i2c_remove), + .remove = aic31xx_i2c_remove, .id_table = aic31xx_i2c_id, }; -- 1.8.1.2

3 2

[alsa-devel] [PATCH] regmap: Fix possible sleep-in-atomic in fast io mode
by Takashi Iwai 18 Mar '14

18 Mar '14

regmap deploys the spinlock for the protection when set up in fast_io mode. But some places are still left with possible sleep in the lock, typically allocating memory with GFP_KERNEL. This patch fixes these calls by moving the allocations out of the lock. The krealloc() call in regmap_register_patch() makes it unsafe for concurrent accesses, however. But such a use case must be very rare, thus we take rather simplicity over complexity, and add a note in the function description mentioning this restriction. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- Only compile tested. drivers/base/regmap/regmap.c | 36 ++++++++++++++++++++---------------- 1 file changed, 20 insertions(+), 16 deletions(-) diff --git a/drivers/base/regmap/regmap.c b/drivers/base/regmap/regmap.c index 6a19515f8a45..e6827cee29a3 100644 --- a/drivers/base/regmap/regmap.c +++ b/drivers/base/regmap/regmap.c @@ -1520,12 +1520,12 @@ int regmap_bulk_write(struct regmap *map, unsigned int reg, const void *val, if (reg % map->reg_stride) return -EINVAL; - map->lock(map->lock_arg); /* * Some devices don't support bulk write, for * them we have a series of single write operations. */ if (!map->bus || map->use_single_rw) { + map->lock(map->lock_arg); for (i = 0; i < val_count; i++) { unsigned int ival; @@ -1554,24 +1554,25 @@ int regmap_bulk_write(struct regmap *map, unsigned int reg, const void *val, if (ret != 0) goto out; } +out: + map->unlock(map->lock_arg); } else { void *wval; wval = kmemdup(val, val_count * val_bytes, GFP_KERNEL); if (!wval) { - ret = -ENOMEM; dev_err(map->dev, "Error in memory allocation\n"); - goto out; + return -ENOMEM; } + map->lock(map->lock_arg); for (i = 0; i < val_count * val_bytes; i += val_bytes) map->format.parse_inplace(wval + i); ret = _regmap_raw_write(map, reg, wval, val_bytes * val_count); + map->unlock(map->lock_arg); kfree(wval); } -out: - map->unlock(map->lock_arg); return ret; } EXPORT_SYMBOL_GPL(regmap_bulk_write); @@ -2173,6 +2174,10 @@ EXPORT_SYMBOL_GPL(regmap_async_complete); * apply them immediately. Typically this is used to apply * corrections to be applied to the device defaults on startup, such * as the updates some vendors provide to undocumented registers. + * + * Note that the function gives no protection over concurrent access + * to map->patch and map->patch_regs. If needed, apply some lock in + * the caller side. */ int regmap_register_patch(struct regmap *map, const struct reg_default *regs, int num_regs) @@ -2185,6 +2190,16 @@ int regmap_register_patch(struct regmap *map, const struct reg_default *regs, num_regs)) return 0; + p = krealloc(map->patch, + sizeof(struct reg_default) * (map->patch_regs + num_regs), + GFP_KERNEL); + if (!p) + return -ENOMEM; + + memcpy(p + map->patch_regs, regs, num_regs * sizeof(*regs)); + map->patch = p; + map->patch_regs += num_regs; + map->lock(map->lock_arg); bypass = map->cache_bypass; @@ -2202,17 +2217,6 @@ int regmap_register_patch(struct regmap *map, const struct reg_default *regs, } } - p = krealloc(map->patch, - sizeof(struct reg_default) * (map->patch_regs + num_regs), - GFP_KERNEL); - if (p) { - memcpy(p + map->patch_regs, regs, num_regs * sizeof(*regs)); - map->patch = p; - map->patch_regs += num_regs; - } else { - ret = -ENOMEM; - } - out: map->async = false; map->cache_bypass = bypass; -- 1.9.0

2 3

[alsa-devel] [PATCH RFC] ALSA: oxygen: Xonar DG(X): fix Stereo Upmixing regression
by Clemens Ladisch 18 Mar '14

18 Mar '14

The code introduced in commit 1f91ecc14dee ("ALSA: oxygen: modify adjust_dg_dac_routing function") accidentally disregarded the old value of the playback routing register, so it broke the "Stereo Upmixing" mixer control. The unmuted parts of the channel routing are the same for all settings of the output destination, so it suffices to revert that part of the patch. Signed-off-by: Clemens Ladisch <clemens(a)ladisch.de> --- sound/pci/oxygen/xonar_dg.c | 30 ++++++++++++++++++++++++------ 1 file changed, 24 insertions(+), 6 deletions(-) Roman, could you please test this patch, and add your Tested-by? diff --git a/sound/pci/oxygen/xonar_dg.c b/sound/pci/oxygen/xonar_dg.c index ed6f199..4cf3200 100644 --- a/sound/pci/oxygen/xonar_dg.c +++ b/sound/pci/oxygen/xonar_dg.c @@ -238,11 +238,21 @@ void set_cs4245_adc_params(struct oxygen *chip, cs4245_write_spi(chip, CS4245_MCLK_FREQ); } +static inline unsigned int shift_bits(unsigned int value, + unsigned int shift_from, + unsigned int shift_to, + unsigned int mask) +{ + if (shift_from < shift_to) + return (value << (shift_to - shift_from)) & mask; + else + return (value >> (shift_from - shift_to)) & mask; +} + unsigned int adjust_dg_dac_routing(struct oxygen *chip, unsigned int play_routing) { struct dg *data = chip->model_data; - unsigned int routing = 0; switch (data->output_sel) { case PLAYBACK_DST_HP: @@ -252,15 +262,23 @@ unsigned int adjust_dg_dac_routing(struct oxygen *chip, OXYGEN_PLAY_MUTE67, OXYGEN_PLAY_MUTE_MASK); break; case PLAYBACK_DST_MULTICH: - routing = (0 << OXYGEN_PLAY_DAC0_SOURCE_SHIFT) | - (2 << OXYGEN_PLAY_DAC1_SOURCE_SHIFT) | - (1 << OXYGEN_PLAY_DAC2_SOURCE_SHIFT) | - (0 << OXYGEN_PLAY_DAC3_SOURCE_SHIFT); oxygen_write8_masked(chip, OXYGEN_PLAY_ROUTING, OXYGEN_PLAY_MUTE01, OXYGEN_PLAY_MUTE_MASK); break; } - return routing; + return (play_routing & OXYGEN_PLAY_DAC0_SOURCE_MASK) | + shift_bits(play_routing, + OXYGEN_PLAY_DAC2_SOURCE_SHIFT, + OXYGEN_PLAY_DAC1_SOURCE_SHIFT, + OXYGEN_PLAY_DAC1_SOURCE_MASK) | + shift_bits(play_routing, + OXYGEN_PLAY_DAC1_SOURCE_SHIFT, + OXYGEN_PLAY_DAC2_SOURCE_SHIFT, + OXYGEN_PLAY_DAC2_SOURCE_MASK) | + shift_bits(play_routing, + OXYGEN_PLAY_DAC0_SOURCE_SHIFT, + OXYGEN_PLAY_DAC3_SOURCE_SHIFT, + OXYGEN_PLAY_DAC3_SOURCE_MASK); } void dump_cs4245_registers(struct oxygen *chip,

3 4

[alsa-devel] Restore default value for alc282
by Kailang 18 Mar '14

18 Mar '14

Hi Takashi, Attachment is the patch for fix below issue for alc282. Restore the registers to prevent the abnormal digital power supply rising ratio/sequence to the codec and causing the incorrect default codec register restoration during initialization. Many Thanks. BR, Kailang

1 0

[alsa-devel] Restore default value for alc283
by Kailang 18 Mar '14

18 Mar '14

Hi Takashi, Attachment is the patch for fix below issue. Restore the registers to prevent the abnormal digital power supply rising ratio/sequence to the codec and causing the incorrect default codec register restoration during initialization. Many Thanks. BR, Kailang

2 4

[alsa-devel] [PATCH] ASoC: tegra: Use flat regcache.
by Dylan Reid 18 Mar '14

18 Mar '14

When using an rbtree cache, there can be allocations the first time a register is accessed. This can cause an attempt to schedule while atomic in the case that the regmap is using a spinlock. This could be fixed by either initializing all the registers or using a flat cache. The register maps for tegra30_ahub and tegra30_i2s are dense and don't save much from using a tree so convert them to flat. Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- sound/soc/tegra/tegra30_ahub.c | 4 ++-- sound/soc/tegra/tegra30_i2s.c | 2 +- 2 files changed, 3 insertions(+), 3 deletions(-) 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 = { -- 1.8.1.3.605.g02339dd

5 7

[alsa-devel] [PATCH 1/2] ASoC: sirf-audio-codec: Remove snd_soc_codec_set_cache_io() call
by Lars-Peter Clausen 18 Mar '14

18 Mar '14

There was a overlap between the snd_soc_codec_set_cache_io() cleanup and the addition of the sirf-audio-codec resulting in the sirf-audio-codec driver still using the old signature of snd_soc_codec_set_cache_io(), which will cause a compile error. Since the core is able to automatically setup IO for this driver we can just remove both the snd_soc_set_cache_io() call and the control_data assignment. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/sirf-audio-codec.c | 9 --------- 1 file changed, 9 deletions(-) diff --git a/sound/soc/codecs/sirf-audio-codec.c b/sound/soc/codecs/sirf-audio-codec.c index 90e3a22..58e7c1f 100644 --- a/sound/soc/codecs/sirf-audio-codec.c +++ b/sound/soc/codecs/sirf-audio-codec.c @@ -337,18 +337,9 @@ struct snd_soc_dai_driver sirf_audio_codec_dai = { static int sirf_audio_codec_probe(struct snd_soc_codec *codec) { - int ret; struct snd_soc_dapm_context *dapm = &codec->dapm; - struct sirf_audio_codec *sirf_audio_codec = snd_soc_codec_get_drvdata(codec); pm_runtime_enable(codec->dev); - codec->control_data = sirf_audio_codec->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; - } if (of_device_is_compatible(codec->dev->of_node, "sirf,prima2-audio-codec")) { snd_soc_dapm_new_controls(dapm, -- 1.8.0

3 6

[alsa-devel] [PATCH v2 0/4] ASoC: simple-card: DT fix and multi DAI links extension
by Jean-Francois Moine 17 Mar '14

17 Mar '14

This patch series fixes a small problem about node device locks and also extends the simple card driver to handle many DAI links as this exists in the Cubox audio subsystem. - v2 - change subject/comment about device node reference count (Mark Brown) - use a null size array instead of an implicit area for the DAI links (Li Xiubo) - update the reference count of the device node at end of probe Jean-Francois Moine (4): ASoC: simple-card: Fix the reference count of device nodes ASoC: simple-card: dynamically allocate the DAI link array ASoC: simple-card: accept many DAI links ASoC: simple-card: Add DT documentation for multi-DAI links .../devicetree/bindings/sound/simple-card.txt | 34 +++- sound/soc/generic/simple-card.c | 174 ++++++++++++++------- 2 files changed, 152 insertions(+), 56 deletions(-) -- 1.9.0

4 11

[alsa-devel] [PATCH] ASoC: fsi: fixup compile error for simple-card
by Kuninori Morimoto 17 Mar '14

17 Mar '14

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> This patches fixes c7a507eea1db1430476289f525f9c853d5d485e8 (ASoC: fsi: fixup SND_SOC_DAIFMT_CBx_CFx flags) commit's compie error arch/arm/mach-shmobile/board-mackerel.c:512:2: \ error: unknown field 'fmt' specified in initializer Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- Mark This patch is for mark/fix/rcar arch/arm/mach-shmobile/board-mackerel.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/arch/arm/mach-shmobile/board-mackerel.c b/arch/arm/mach-shmobile/board-mackerel.c index 8c43321..0ff4d8e 100644 --- a/arch/arm/mach-shmobile/board-mackerel.c +++ b/arch/arm/mach-shmobile/board-mackerel.c @@ -509,7 +509,7 @@ static struct asoc_simple_card_info fsi2_hdmi_info = { .card = "FSI2B-HDMI", .codec = "sh-mobile-hdmi", .platform = "sh_fsi2", - .fmt = SND_SOC_DAIFMT_CBS_CFS, + .daifmt = SND_SOC_DAIFMT_CBS_CFS, .cpu_dai = { .name = "fsib-dai", }, -- 1.7.9.5

3 2

[alsa-devel] [PATCH] ASoC: tlv320aic31xx: Turn power off only once.
by Jyri Sarha 17 Mar '14

17 Mar '14

Regulator code keep count of enables and disables. Double disable causes an ugly warning. Signed-off-by: Jyri Sarha <jsarha(a)ti.com> --- sound/soc/codecs/tlv320aic31xx.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index 89f808b..07ddec3 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -1021,7 +1021,8 @@ static int aic31xx_set_bias_level(struct snd_soc_codec *codec, } break; case SND_SOC_BIAS_OFF: - aic31xx_power_off(codec); + if (codec->dapm.bias_level == SND_SOC_BIAS_STANDBY) + aic31xx_power_off(codec); break; } codec->dapm.bias_level = level; -- 1.7.9.5

2 1

[alsa-devel] [PATCH 0/2][RFC] ASoC: fsi/rcar: fixup SND_SOC_DAIFMT_CBx_CFx flags
by Kuninori Morimoto 17 Mar '14

17 Mar '14

Hi Mark, Simon These patches fixup Renesas fsi/rcar driver's SND_SOC_DAIFMT_CBx_CFx flag settings. I had misunderstand these settings... But this is still [RFC], since it modifies DT side code too. >> Mark #1 patch depends on Jyri's "Fix simple-card *-master DT parameter handling" if it was accepted >> Simon Can you check #1 patch which modifys SH-ARM platform ? Kuninori Morimoto (2): 1) ASoC: fsi: fixup SND_SOC_DAIFMT_CBx_CFx flags 2) ASoC: rcar: fixup SND_SOC_DAIFMT_CBx_CFx flags .../boot/dts/r8a7740-armadillo800eva-reference.dts | 3 ++- arch/arm/boot/dts/sh73a0-kzm9g-reference.dts | 2 ++ arch/arm/mach-shmobile/board-armadillo800eva.c | 4 +--- arch/arm/mach-shmobile/board-kzm9g.c | 4 +--- arch/arm/mach-shmobile/board-mackerel.c | 6 ++---- arch/sh/boards/mach-ecovec24/setup.c | 4 +--- arch/sh/boards/mach-se/7724/setup.c | 4 +--- sound/soc/sh/fsi.c | 2 +- sound/soc/sh/rcar/core.c | 4 ++-- 9 files changed, 13 insertions(+), 20 deletions(-) Best regards --- Kuninori Morimoto

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Re: [alsa-devel] compress: adding support for hardware dependent params
by noman pouigt 16 Mar '14

16 Mar '14

> We have a requirement, where we need to set a number of hardware specific > parameters for compressed offload use-cases, before starting a stream. As > these parameters need to be set once for the entire session, we would > like to explore the best way to set these additional parameters somewhere > before calling trigger() function. > Below are few of the proposed solutions that we could think of at this > point for getting this done- As you have not mentioned what kind of metadata you are talking about I am assuming if your meta data can fit in below structure then we already have a code in upstream which takes care of your need. 140 struct snd_compr_metadata { 141 __u32 key; 142 __u32 value[8]; 143 }; Structure snd_compr_ops has set_metadata and get_metadata callbacks which you can use for your need. 88 * struct snd_compr_ops: compressed path DSP operations 89 * @open: Open the compressed stream 90 * This callback is mandatory and shall keep dsp ready to receive the stream 91 * parameter 92 * @free: Close the compressed stream, mandatory 93 * @set_params: Sets the compressed stream parameters, mandatory 94 * This can be called in during stream creation only to set codec params 95 * and the stream properties 96 * @get_params: retrieve the codec parameters, mandatory 97 * @trigger: Trigger operations like start, pause, resume, drain, stop. 98 * This callback is mandatory 99 * @pointer: Retrieve current h/w pointer information. Mandatory 100 * @copy: Copy the compressed data to/from userspace, Optional 101 * Can't be implemented if DSP supports mmap 102 * @mmap: DSP mmap method to mmap DSP memory 103 * @ack: Ack for DSP when data is written to audio buffer, Optional 104 * Not valid if copy is implemented 105 * @get_caps: Retrieve DSP capabilities, mandatory 106 * @get_codec_caps: Retrieve capabilities for a specific codec, mandatory 107 */ It would be nice if someone can send a patch here to describe set and get_metadata callbacks in compress_driver.h > > > 1. Adding new IOCTL call: Add a new ioctl control in compress-offload.c > file to handle hardware specific parameters setting. This new ioctl > control will call into soc-compress.c and eventually call into platform > driver ioctl to set those hardware specific parameters. In this case, we > will add a new structure, let's say "snd_compr_hw_ctrl_params", with the > same fields as "snd_compr_metadata" struct. > > > 2. Adding a function hw_dependant_hw_params(), which will be called after > hw_params() and before prepare() function. The new function will > eventually call into the platform driver to get the hardware specific > parameters set. Once implemented, this would look something like- > hw_params(); > hw_dependant_hw_params(); > prepare(); > > > 3. Add a function "hw_dependant_hw_params()", as _part_ of the hw_params() > function. This new function will be called after the (existing) generic > parameters are set. So once implemented, the proposed function calls would > look like- > hw_params() { > generic_hw_params(); > hw_dependant_hw_params(); > } > prepare(); > > > We would like to get feedback on these proposed solutions and get a > discussion going to see how best these requirements can be fitted in the > current Compressed ALSA framework (for mainlining). Can you explain your meta data? Be the change you want to see in the world:Gandhi

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[alsa-devel] [PATCH] ASoC: tlv320aic31xx: Don't call kfree for memory allocated by devm_kzalloc
by Axel Lin 16 Mar '14

16 Mar '14

Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/tlv320aic31xx.c | 9 --------- 1 file changed, 9 deletions(-) diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index d3517a9..1953184 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -1248,14 +1248,6 @@ static int aic31xx_i2c_probe(struct i2c_client *i2c, return ret; } -static int aic31xx_i2c_remove(struct i2c_client *i2c) -{ - struct aic31xx_priv *aic31xx = dev_get_drvdata(&i2c->dev); - - kfree(aic31xx); - return 0; -} - static const struct i2c_device_id aic31xx_i2c_id[] = { { "tlv320aic310x", AIC3100 }, { "tlv320aic311x", AIC3110 }, @@ -1274,7 +1266,6 @@ static struct i2c_driver aic31xx_i2c_driver = { .of_match_table = of_match_ptr(tlv320aic31xx_of_match), }, .probe = aic31xx_i2c_probe, - .remove = (aic31xx_i2c_remove), .id_table = aic31xx_i2c_id, }; -- 1.8.1.2

1 0

[alsa-devel] [RFC] [PATCH] firewire-lib: add support for AV/C deferred transaction
by Takashi Sakamoto 15 Mar '14

15 Mar '14

Clemens and Stefan, Can I request your comments about attached patch? This is for adding support of deferred transaction into snd-firewire-lib. I confirm some BeBoB devices use this type of transaction. If you have no negative comments, I'll add this patch for my series. For 'undefined interval' between INTERIM and final response, I want to use 'FCP_TIMEOUT_MS' (=125msec) again. In specifications which I can read, there is no limit for this interval. But I need to promise to finish this function for callers. According to AV/C general specification, targets can send INTERIM response one time. But this patch allows to accept several ITERIM responses for device quirks. I don't have devices which have such quirks, but there may be such devices and I want to make codes simpler. Regards Takashi Sakamoto o-takashi(a)sakamocchi.jp

2 2

[alsa-devel] [GIT PULL][PATCH 00/39 v2] Enhancement of support for Firewire devices
by Takashi Sakamoto 14 Mar '14

14 Mar '14

This pull-request is an update for my previous pull-request: [alsa-devel] [GIT PULL][PATCH 00/39] Enhancement of support for Firewire devices http://mailman.alsa-project.org/pipermail/alsa-devel/2014-February/073446.h… Updates since previous pull-request: firewire-lib: - use swap() instead of local macro - check NULL at memory allocation for sort table fireworks: - use bitmap.h for card slots - add 'firewire' subdirectory to card node in procfs - add sanity check of module parameter with clamp() bebob: - use bitmap.h for card slots - add 'firewire' subdirectory to card node in procfs - simplify codes for metering - use swap() instead of local macro - change order of last two commits ---------------------------------------------------------------- The following changes since commit 31278997add61ee13b709e274934c7b0085accce: ALSA: hda/realtek - Add headset quirk for Dell DT (2014-03-03 10:48:55 +0100) are available in the git repository at: https://github.com/takaswie/sound.git request for you to fetch changes up to 5b354e20ec3c64fa0a8fa7fbc7355201d881a8cc: bebob: Send a cue to load firmware for M-Audio Firewire series (2014-03-05 12:00:32 +0900) ---------------------------------------------------------------- Takashi Sakamoto (39): firewire-lib: Rename functions, structure, member for AMDTP firewire-lib: Add macros instead of fixed value for AMDTP firewire-lib: Add 'direction' member to 'amdtp_stream' structure firewire-lib: Split some codes into functions to reuse for both streams firewire-lib: Add support for AMDTP in-stream and PCM capture firewire-lib: Add support for MIDI capture/playback firewire-lib: Give syt value as parameter to handle_out_packet() firewire-lib: Add support for duplex streams synchronization in blocking mode firewire-lib: Add sort function for transmitted packet firewire-lib: Add transfer delay to synchronized duplex streams firewire-lib: Add support for channel mapping firewire-lib: Rename macros, variables and functions for CMP firewire-lib: Add 'direction' member to 'cmp_connection' structure firewire-lib: Add handling output connection by CMP firewire-lib: Add a new function to check others' connection firewire-lib: Add some AV/C general commands firewire-lib: Add quirks for Fireworks firewire-lib: Add a fallback at RCODE_CANCELLED fireworks: Add skelton for Fireworks based devices fireworks: Add transaction and some commands fireworks: Add connection and stream management fireworks: Add proc interface for debugging purpose fireworks: Add MIDI interface fireworks: Add PCM interface fireworks: Add hwdep interface fireworks: Add command/response functionality into hwdep interface bebob: Add skelton for BeBoB based devices bebob: Add commands and connections/streams management bebob: Add proc interface for debugging purpose bebob: Add MIDI interface bebob: Add PCM interface bebob: Add hwdep interface bebob: Prepare for device specific operations bebob: Add support for Terratec PHASE, EWS series and Aureon bebob: Add support for Yamaha GO series bebob: Add support for Focusrite Saffire/SaffirePro series bebob: Add support for M-Audio usual Firewire series bebob: Add support for M-Audio special Firewire series bebob: Send a cue to load firmware for M-Audio Firewire series include/uapi/sound/asound.h | 4 +- include/uapi/sound/firewire.h | 22 +- sound/firewire/Kconfig | 59 ++ sound/firewire/Makefile | 2 + sound/firewire/amdtp.c | 891 +++++++++++++++++------ sound/firewire/amdtp.h | 175 ++++- sound/firewire/bebob/Makefile | 4 + sound/firewire/bebob/bebob.c | 450 ++++++++++++ sound/firewire/bebob/bebob.h | 260 +++++++ sound/firewire/bebob/bebob_command.c | 354 +++++++++ sound/firewire/bebob/bebob_focusrite.c | 287 ++++++++ sound/firewire/bebob/bebob_hwdep.c | 197 +++++ sound/firewire/bebob/bebob_maudio.c | 828 +++++++++++++++++++++ sound/firewire/bebob/bebob_midi.c | 170 +++++ sound/firewire/bebob/bebob_pcm.c | 451 ++++++++++++ sound/firewire/bebob/bebob_proc.c | 201 +++++ sound/firewire/bebob/bebob_stream.c | 869 ++++++++++++++++++++++ sound/firewire/bebob/bebob_terratec.c | 68 ++ sound/firewire/bebob/bebob_yamaha.c | 50 ++ sound/firewire/cmp.c | 229 ++++-- sound/firewire/cmp.h | 14 +- sound/firewire/dice.c | 46 +- sound/firewire/fcp.c | 160 +++- sound/firewire/fcp.h | 21 + sound/firewire/fireworks/Makefile | 4 + sound/firewire/fireworks/fireworks.c | 326 +++++++++ sound/firewire/fireworks/fireworks.h | 223 ++++++ sound/firewire/fireworks/fireworks_command.c | 397 ++++++++++ sound/firewire/fireworks/fireworks_hwdep.c | 294 ++++++++ sound/firewire/fireworks/fireworks_midi.c | 179 +++++ sound/firewire/fireworks/fireworks_pcm.c | 477 ++++++++++++ sound/firewire/fireworks/fireworks_proc.c | 233 ++++++ sound/firewire/fireworks/fireworks_stream.c | 337 +++++++++ sound/firewire/fireworks/fireworks_transaction.c | 327 +++++++++ sound/firewire/lib.c | 6 +- sound/firewire/lib.h | 1 + sound/firewire/speakers.c | 93 +-- 37 files changed, 8307 insertions(+), 402 deletions(-) create mode 100644 sound/firewire/bebob/Makefile create mode 100644 sound/firewire/bebob/bebob.c create mode 100644 sound/firewire/bebob/bebob.h create mode 100644 sound/firewire/bebob/bebob_command.c create mode 100644 sound/firewire/bebob/bebob_focusrite.c create mode 100644 sound/firewire/bebob/bebob_hwdep.c create mode 100644 sound/firewire/bebob/bebob_maudio.c create mode 100644 sound/firewire/bebob/bebob_midi.c create mode 100644 sound/firewire/bebob/bebob_pcm.c create mode 100644 sound/firewire/bebob/bebob_proc.c create mode 100644 sound/firewire/bebob/bebob_stream.c create mode 100644 sound/firewire/bebob/bebob_terratec.c create mode 100644 sound/firewire/bebob/bebob_yamaha.c create mode 100644 sound/firewire/fireworks/Makefile create mode 100644 sound/firewire/fireworks/fireworks.c create mode 100644 sound/firewire/fireworks/fireworks.h create mode 100644 sound/firewire/fireworks/fireworks_command.c create mode 100644 sound/firewire/fireworks/fireworks_hwdep.c create mode 100644 sound/firewire/fireworks/fireworks_midi.c create mode 100644 sound/firewire/fireworks/fireworks_pcm.c create mode 100644 sound/firewire/fireworks/fireworks_proc.c create mode 100644 sound/firewire/fireworks/fireworks_stream.c create mode 100644 sound/firewire/fireworks/fireworks_transaction.c

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[alsa-devel] [PATCH] ASoC: max98090: Remove unused control_data field
by Lars-Peter Clausen 14 Mar '14

14 Mar '14

The driver assigns a value to the control_data field of the driver's state struct, but never reads it again. Which means it is unused and can be removed. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/max98090.c | 1 - sound/soc/codecs/max98090.h | 1 - 2 files changed, 2 deletions(-) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index 390fba1..1baa12c 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -2343,7 +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; -- 1.8.0

2 1

[alsa-devel] [PATCH] ASoC: tlv320aic31xx: Staticise non-exported symbols
by Mark Brown 14 Mar '14

14 Mar '14

From: Mark Brown <broonie(a)linaro.org> Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/tlv320aic31xx.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index dcdc5751048f..d3517a919776 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -129,7 +129,7 @@ static const struct regmap_range_cfg aic31xx_ranges[] = { }, }; -struct regmap_config aic31xx_i2c_regmap = { +static const struct regmap_config aic31xx_i2c_regmap = { .reg_bits = 8, .val_bits = 8, .writeable_reg = aic31xx_writeable, @@ -321,9 +321,9 @@ static const struct snd_kcontrol_new ldac_in_control = static const struct snd_kcontrol_new rdac_in_control = SOC_DAPM_ENUM("DAC Right Input", rdac_in_enum); -int aic31xx_wait_bits(struct aic31xx_priv *aic31xx, unsigned int reg, - unsigned int mask, unsigned int wbits, int sleep, - int count) +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; @@ -1177,7 +1177,7 @@ static void aic31xx_pdata_from_of(struct aic31xx_priv *aic31xx) } #endif /* CONFIG_OF */ -void aic31xx_device_init(struct aic31xx_priv *aic31xx) +static void aic31xx_device_init(struct aic31xx_priv *aic31xx) { int ret, i; -- 1.9.0

3 3

[alsa-devel] [PATCH 00/18] edma/ASoC: dmaengine PCM for AM335x and AM447x
by Peter Ujfalusi 14 Mar '14

14 Mar '14

Hi, With this series AM335x and AM447x will use the dmaengine PCM for audio. The daVinci devices will keep using the davinci-pcm for now since I do not have means to test them but the code is written in a way that they can be switched to use the edma-pcm easily (DA850/OMAP-L138 has been tested and audio was working fine). To be able to use the dmaengine PCM I needed to make changes to the edma code in arch/arm/common/ and in drivers/dma/ : Adding support for DMA pause/resume Possibility to select non default event queue/TC for cyclic (audio) dma channels: all devices using the eDMA via dmaengine was assigned to the default EQ/TC (mmc, i2c, spi, etc, and audio). This is not optimal from system performance point of view since sharing the same EQ/TC can cause latency spikes for cyclic channels (long DMA transfers for MMC for example). While debugging the edma to get things sorted out I noticed that the debug was too verbose and the important information was hidden even when the we did not asked for verbose dmaengine debug. I have included some debug cleanups for the edma dmaengine driver also. On the ASoC side: the series has been tested on AM335x and AM447x. I have report that DA850/OMAP-L138 works with this series (which boots w/o DT). Regards, Peter --- Peter Ujfalusi (18): platform_data: edma: Be precise with the paRAM struct dma: edma: Add support for DMA_PAUSE/RESUME operation dma: edma: Set DMA_CYCLIC capability flag arm: common: edma: Select event queue 1 as default when booted with DT arm: common: edma: Save the number of event queues/TCs arm: common: edma: API to request non default queue for a channel DMA: edma: Use different eventq for cyclic channels dma: edma: Implement device_slave_caps callback dma: edma: Simplify direction configuration in edma_config_pset() dma: edma: Reduce debug print verbosity for non verbose debugging dma: edma: Prefix debug prints where the text were identical in prep callbacks dma: edma: Add channel number to debug prints dma: edma: Print the direction value as well when it is not supported ASoC: davinci: Add edma dmaengine platform driver ASoC: davinci-mcasp: Use dmaengine based platform driver for AM335x/447x ASoC: davinci-mcasp: Provide correct filter_data for dmaengine for non-DT boot ASoC: davinci-mcasp: Assign the dma_data earlier in dai_probe callback ASoC: davinci-mcasp: Place constraint on the period size based on FIFO config arch/arm/common/edma.c | 34 ++++++++++++- drivers/dma/edma.c | 92 ++++++++++++++++++++++++++-------- include/linux/platform_data/edma.h | 20 ++++---- sound/soc/davinci/Kconfig | 1 + sound/soc/davinci/Makefile | 2 +- sound/soc/davinci/davinci-mcasp.c | 100 ++++++++++++++++++++++++++++++------- sound/soc/davinci/edma-pcm.c | 77 ++++++++++++++++++++++++++++ sound/soc/davinci/edma-pcm.h | 26 ++++++++++ 8 files changed, 303 insertions(+), 49 deletions(-) create mode 100644 sound/soc/davinci/edma-pcm.c create mode 100644 sound/soc/davinci/edma-pcm.h -- 1.9.0

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[alsa-devel] Low volume ad1983
by Raymond Yau 14 Mar '14

14 Mar '14

>>sam chan wrote: >> i'm having a problem with sound volume being too low. >> i **think** it's due to the levels of my channel map being >> too low..... >The channel map specifies which channel goes to which >physical output. >> my alsa-info... >> http://www.alsa-project.org/db/?f=3ba6278df3c6c4abe5196ad37f8015de7223ca8e >> Simple mixer control 'Master',0 >> Mono: Playback 37 [59%] [-39.00dB] [on] >That is pretty low. >> Simple mixer control 'Loopback Mixing',0 >> Item0: 'Enabled' >Why is this enabled? seem driver bug 1) Should the volume control and playback switch of line out be named as Front instead of PCM ? Node 0x05 [Pin Complex] wcaps 0x400185: Stereo Amp-Out Control: name="PCM Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="PCM Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Line Out Jack", index=0, device=0 Amp-Out caps: ofs=0x3d, nsteps=0x3f, stepsize=0x05, mute=1 Amp-Out vals: [0x23 0x23] Pincap 0x00000017: OUT Detect Trigger ImpSense Pin Default 0x01014010: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Green DefAssociation = 0x1, Sequence = 0x0 2) dB max of virtual master playback volume is different from the slave when slave dbmax of slaves > 0 control.1 { iface MIXER name 'PCM Playback Volume' value.0 61 value.1 61 comment { access 'read write' type INTEGER count 2 range '0 - 63' dbmin -9150 dbmax 300 dbvalue.0 0 dbvalue.1 0 } } control.3 { iface MIXER name 'Headphone Playback Volume' value.0 61 value.1 61 comment { access 'read write' type INTEGER count 2 range '0 - 63' dbmin -9150 dbmax 300 dbvalue.0 0 dbvalue.1 0 } } control.5 { iface MIXER name 'Speaker Playback Volume' value 61 comment { access 'read write' type INTEGER count 1 range '0 - 63' dbmin -9150 dbmax 300 dbvalue.0 0 } } control.17 { iface MIXER name 'Master Playback Volume' value 37 comment { access 'read write' type INTEGER count 1 range '0 - 63' dbmin -9450 dbmax 0 dbvalue.0 -3900 } }

3 3

[alsa-devel] please ignore my earlier message regarding R16 quirks
by James Miller 14 Mar '14

14 Mar '14

I'm sorry for my ignorance, but I realised that the device has been reporting itself as 1.1 due to the usb hub being 1.1. This is my first week of looking at usb, and its been a steep learning curve, I am finding my way around the code though, and hope to have something working at some point in the near future. I notice that some cards have boot up quirk functions and I am wondering if that is the problem as the R16 can be switched between its audio interface and a card reader. If I have any more questions, is this the correct list to ask them? Many thanks, James

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[alsa-devel] [RFC] ASoC: core: Add support for DAI multicodec
by Benoit Cousson 13 Mar '14

13 Mar '14

From: Misael Lopez Cruz <misael.lopez(a)ti.com> DAI link assumes a one to one mapping between CPU DAI and CODEC. In some cases, the same CPU DAI can be connected to several codecs. This is the case for example, if you connect two mono codecs to the same I2S link in order to have a stereo card. The current ASoC implementation does not allow such setup. Add support for DAI links composed of a single CPU DAI and multiple CODECs. Sound cards have to pass the CODECs array in the corresponding DAI link through a new 'snd_soc_dai_link_codec' struct. Each CODEC in this array is described in the same manner single CODEC DAIs are (either DT/OF node or codec_name). CPU DAI in a multicodec DAI link can have more channels than what each CODEC has. The number of channels each CODEC is responsible for is machine specific, hence it's fixed up in machine drivers in a similar way to what DPCM does. Add few helpers to make the code more readable and fix a trailing space in the header as well. Signed-off-by: Misael Lopez Cruz <misael.lopez(a)ti.com> [bcousson(a)baylibre.com: Adapt the original series to 3.14] Signed-off-by: Benoit Cousson <bcousson(a)baylibre.com> --- Hi Mark and Liam, I'm not sure this is the only way to address that problem, but both Misa and myself were facing the same issue on different platforms, and we didn't find any straighforward way to do that. To avoid breaking all the exiting ASoC implementation, this approach is adding extra attributes for multiple codecs and does not enforce moving every single codec pointer to a multicodec array. The patch is based on 3.14-rc6. Comments are welcome. Thanks, Benoit --- include/sound/soc.h | 20 ++ sound/soc/soc-core.c | 511 ++++++++++++++++++++++++++++++++++----------------- sound/soc/soc-dapm.c | 39 ++-- sound/soc/soc-pcm.c | 412 ++++++++++++++++++++++++++++------------- 4 files changed, 677 insertions(+), 305 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 9a00147..c17351c 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -818,6 +818,19 @@ struct snd_soc_platform_driver { int (*bespoke_trigger)(struct snd_pcm_substream *, int); }; +struct snd_soc_dai_link_codec { + const char *codec_name; + const struct device_node *codec_of_node; + const char *codec_dai_name; + + struct snd_soc_codec *codec; + struct snd_soc_dai *codec_dai; + + int (*hw_params_fixup)(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params); + struct snd_pcm_hw_params hw_params; +}; + struct snd_soc_platform { const char *name; int id; @@ -866,6 +879,10 @@ struct snd_soc_dai_link { const struct device_node *codec_of_node; /* You MUST specify the DAI name within the codec */ const char *codec_dai_name; + + struct snd_soc_dai_link_codec *codecs; + int num_codecs; + /* * You MAY specify the link's platform/PCM/DMA driver, either by * device name, or by DT/OF node, but not both. Some forms of link @@ -1055,6 +1072,9 @@ struct snd_soc_pcm_runtime { struct snd_soc_dai *codec_dai; struct snd_soc_dai *cpu_dai; + struct snd_soc_dai_link_codec *codecs; + int num_codecs; + struct delayed_work delayed_work; #ifdef CONFIG_DEBUG_FS struct dentry *debugfs_dpcm_root; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index fe1df50..1a6c2d0 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -557,8 +557,9 @@ static void codec2codec_close_delayed_work(struct work_struct *work) int snd_soc_suspend(struct device *dev) { struct snd_soc_card *card = dev_get_drvdata(dev); + struct snd_soc_dai_link_codec *codecs; struct snd_soc_codec *codec; - int i; + int i, j; /* If the initialization of this soc device failed, there is no codec * associated with it. Just bail out in this case. @@ -578,14 +579,17 @@ int snd_soc_suspend(struct device *dev) /* mute any active DACs */ for (i = 0; i < card->num_rtd; i++) { - struct snd_soc_dai *dai = card->rtd[i].codec_dai; - struct snd_soc_dai_driver *drv = dai->driver; + codecs = card->rtd[i].codecs; + for (j = 0; j < card->rtd[i].num_codecs; j++) { + struct snd_soc_dai *dai = codecs[j].codec_dai; + struct snd_soc_dai_driver *drv = dai->driver; - if (card->rtd[i].dai_link->ignore_suspend) - continue; + if (card->rtd[i].dai_link->ignore_suspend) + continue; - if (drv->ops->digital_mute && dai->playback_active) - drv->ops->digital_mute(dai, 1); + if (drv->ops->digital_mute && dai->playback_active) + drv->ops->digital_mute(dai, 1); + } } /* suspend all pcms */ @@ -616,8 +620,12 @@ int snd_soc_suspend(struct device *dev) /* close any waiting streams and save state */ for (i = 0; i < card->num_rtd; i++) { + codecs = card->rtd[i].codecs; flush_delayed_work(&card->rtd[i].delayed_work); - card->rtd[i].codec->dapm.suspend_bias_level = card->rtd[i].codec->dapm.bias_level; + for (j = 0; j < card->rtd[i].num_codecs; j++) { + codecs[j].codec->dapm.suspend_bias_level = + codecs[j].codec->dapm.bias_level; + } } for (i = 0; i < card->num_rtd; i++) { @@ -700,8 +708,9 @@ static void soc_resume_deferred(struct work_struct *work) { struct snd_soc_card *card = container_of(work, struct snd_soc_card, deferred_resume_work); + struct snd_soc_dai_link_codec *codecs; struct snd_soc_codec *codec; - int i; + int i, j; /* our power state is still SNDRV_CTL_POWER_D3hot from suspend time, * so userspace apps are blocked from touching us @@ -762,14 +771,17 @@ static void soc_resume_deferred(struct work_struct *work) /* unmute any active DACs */ for (i = 0; i < card->num_rtd; i++) { - struct snd_soc_dai *dai = card->rtd[i].codec_dai; - struct snd_soc_dai_driver *drv = dai->driver; + codecs = card->rtd[i].codecs; + for (j = 0; j < card->rtd[i].num_codecs; j++) { + struct snd_soc_dai *dai = codecs[j].codec_dai; + struct snd_soc_dai_driver *drv = dai->driver; - if (card->rtd[i].dai_link->ignore_suspend) - continue; + if (card->rtd[i].dai_link->ignore_suspend) + continue; - if (drv->ops->digital_mute && dai->playback_active) - drv->ops->digital_mute(dai, 0); + if (drv->ops->digital_mute && dai->playback_active) + drv->ops->digital_mute(dai, 0); + } } for (i = 0; i < card->num_rtd; i++) { @@ -851,14 +863,51 @@ EXPORT_SYMBOL_GPL(snd_soc_resume); static const struct snd_soc_dai_ops null_dai_ops = { }; +static struct snd_soc_codec *soc_find_codec( + const struct device_node *codec_of_node, + const char *codec_name) +{ + struct snd_soc_codec *codec; + + list_for_each_entry(codec, &codec_list, list) { + if (codec_of_node) { + if (codec->dev->of_node != codec_of_node) + continue; + } else { + if (strcmp(codec->name, codec_name)) + continue; + } + + return codec; + } + + return NULL; +} + +static struct snd_soc_dai *soc_find_codec_dai(struct snd_soc_codec *codec, + const char *codec_dai_name) +{ + struct snd_soc_dai *codec_dai; + + list_for_each_entry(codec_dai, &dai_list, list) { + if (codec->dev == codec_dai->dev && + !strcmp(codec_dai->name, codec_dai_name)) { + return codec_dai; + } + } + + return NULL; +} + 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_codec *codec; + struct snd_soc_dai_link_codec *codecs = dai_link->codecs; struct snd_soc_platform *platform; - struct snd_soc_dai *codec_dai, *cpu_dai; + struct snd_soc_dai *cpu_dai; const char *platform_name; + int i; dev_dbg(card->dev, "ASoC: binding %s at idx %d\n", dai_link->name, num); @@ -883,43 +932,34 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num) return -EPROBE_DEFER; } - /* Find CODEC from registered CODECs */ - list_for_each_entry(codec, &codec_list, list) { - if (dai_link->codec_of_node) { - if (codec->dev->of_node != dai_link->codec_of_node) - continue; - } else { - if (strcmp(codec->name, dai_link->codec_name)) - continue; - } - - rtd->codec = codec; - - /* - * 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)) { + rtd->num_codecs = dai_link->num_codecs; + rtd->codecs = dai_link->codecs; - rtd->codec_dai = codec_dai; - } + /* Find CODEC from registered CODECs */ + for (i = 0; i < rtd->num_codecs; i++) { + codecs[i].codec = soc_find_codec(codecs[i].codec_of_node, + codecs[i].codec_name); + if (!codecs[i].codec) { + dev_err(card->dev, "ASoC: CODEC %s not registered\n", + codecs[i].codec_name); + return -EPROBE_DEFER; } + } - if (!rtd->codec_dai) { + for (i = 0; i < rtd->num_codecs; i++) { + codecs[i].codec_dai = soc_find_codec_dai( + codecs[i].codec, + codecs[i].codec_dai_name); + if (!codecs[i].codec_dai) { dev_err(card->dev, "ASoC: CODEC DAI %s not registered\n", - dai_link->codec_dai_name); + codecs[i].codec_dai_name); return -EPROBE_DEFER; } } - if (!rtd->codec) { - dev_err(card->dev, "ASoC: CODEC %s not registered\n", - dai_link->codec_name); - return -EPROBE_DEFER; - } + /* Single codec links expect codec and codec_dai in runtime data */ + rtd->codec = codecs[0].codec; + rtd->codec_dai = codecs[0].codec_dai; /* if there's no platform we match on the empty platform */ platform_name = dai_link->platform_name; @@ -994,8 +1034,9 @@ static void soc_remove_codec(struct snd_soc_codec *codec) static void soc_remove_link_dais(struct snd_soc_card *card, int num, int order) { struct snd_soc_pcm_runtime *rtd = &card->rtd[num]; - struct snd_soc_dai *codec_dai = rtd->codec_dai, *cpu_dai = rtd->cpu_dai; - int err; + struct snd_soc_dai *codec_dai; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + int i, err; /* unregister the rtd device */ if (rtd->dev_registered) { @@ -1006,17 +1047,20 @@ static void soc_remove_link_dais(struct snd_soc_card *card, int num, int order) } /* remove the CODEC DAI */ - if (codec_dai && codec_dai->probed && - codec_dai->driver->remove_order == order) { - if (codec_dai->driver->remove) { - err = codec_dai->driver->remove(codec_dai); - if (err < 0) - dev_err(codec_dai->dev, - "ASoC: failed to remove %s: %d\n", - codec_dai->name, err); + for (i = 0; i < rtd->num_codecs; i++) { + codec_dai = rtd->codecs[i].codec_dai; + if (codec_dai && codec_dai->probed && + codec_dai->driver->remove_order == order) { + if (codec_dai->driver->remove) { + err = codec_dai->driver->remove(codec_dai); + if (err < 0) + dev_err(codec_dai->dev, + "ASoC: failed to remove %s: %d\n", + codec_dai->name, err); + } + codec_dai->probed = 0; + list_del(&codec_dai->card_list); } - codec_dai->probed = 0; - list_del(&codec_dai->card_list); } /* remove the cpu_dai */ @@ -1044,9 +1088,9 @@ static void soc_remove_link_components(struct snd_soc_card *card, int num, { struct snd_soc_pcm_runtime *rtd = &card->rtd[num]; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_platform *platform = rtd->platform; struct snd_soc_codec *codec; + int i; /* remove the platform */ if (platform && platform->probed && @@ -1055,8 +1099,8 @@ static void soc_remove_link_components(struct snd_soc_card *card, int num, } /* remove the CODEC-side CODEC */ - if (codec_dai) { - codec = codec_dai->codec; + for (i = 0; i < rtd->num_codecs; i++) { + codec = rtd->codecs[i].codec; if (codec && codec->probed && codec->driver->remove_order == order) soc_remove_codec(codec); @@ -1239,45 +1283,106 @@ static void rtd_release(struct device *dev) kfree(dev); } +static int soc_aux_dev_init(struct snd_soc_card *card, int num) +{ + struct snd_soc_aux_dev *aux_dev = &card->aux_dev[num]; + struct snd_soc_pcm_runtime *rtd = &card->rtd_aux[num]; + struct snd_soc_codec *codec; + const char *temp; + int ret; + + rtd->card = card; + + codec = soc_find_codec(NULL, aux_dev->codec_name); + if (!codec) + return -EPROBE_DEFER; + + /* Make sure all DAPM widgets are instantiated */ + snd_soc_dapm_new_widgets(codec->dapm.card); + + temp = codec->name_prefix; + codec->name_prefix = NULL; + + /* do machine specific initialization */ + if (aux_dev->init) { + ret = aux_dev->init(&codec->dapm); + if (ret < 0) + return ret; + } + + codec->name_prefix = temp; + + rtd->codec = codec; + + return 0; +} + +static int soc_dai_link_init(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_dai_link_codec *codecs = rtd->codecs; + const char **temp; + int i, ret; + + rtd->card = card; + + temp = devm_kzalloc(card->dev, rtd->num_codecs * sizeof(char *), + GFP_KERNEL); + if (!temp) + return -ENOMEM; + + for (i = 0; i < rtd->num_codecs; i++) { + /* Make sure all DAPM widgets are instantiated */ + snd_soc_dapm_new_widgets(codecs[i].codec->dapm.card); + + /* machine controls, routes and widgets are not prefixed */ + temp[i] = codecs[i].codec->name_prefix; + codecs[i].codec->name_prefix = NULL; + } + + /* do machine specific initialization */ + if (dai_link->init) { + ret = dai_link->init(rtd); + if (ret < 0) + return ret; + } + + for (i = 0; i < rtd->num_codecs; i++) + codecs[i].codec->name_prefix = temp[i]; + + devm_kfree(card->dev, temp); + + return 0; +} + static int soc_post_component_init(struct snd_soc_card *card, - struct snd_soc_codec *codec, int num, int dailess) { struct snd_soc_dai_link *dai_link = NULL; struct snd_soc_aux_dev *aux_dev = NULL; struct snd_soc_pcm_runtime *rtd; - const char *temp, *name; + const char *name; int ret = 0; if (!dailess) { dai_link = &card->dai_link[num]; rtd = &card->rtd[num]; name = dai_link->name; + ret = soc_dai_link_init(card, num); } else { aux_dev = &card->aux_dev[num]; rtd = &card->rtd_aux[num]; name = aux_dev->name; + ret = soc_aux_dev_init(card, num); } - rtd->card = card; - - /* machine controls, routes and widgets are not prefixed */ - temp = codec->name_prefix; - codec->name_prefix = NULL; - /* do machine specific initialization */ - if (!dailess && dai_link->init) - ret = dai_link->init(rtd); - else if (dailess && aux_dev->init) - ret = aux_dev->init(&codec->dapm); if (ret < 0) { dev_err(card->dev, "ASoC: failed to init %s: %d\n", name, ret); return ret; } - codec->name_prefix = temp; /* register the rtd device */ - rtd->codec = codec; - rtd->dev = kzalloc(sizeof(struct device), GFP_KERNEL); if (!rtd->dev) return -ENOMEM; @@ -1295,7 +1400,7 @@ static int soc_post_component_init(struct snd_soc_card *card, if (ret < 0) { /* calling put_device() here to free the rtd->dev */ put_device(rtd->dev); - dev_err(card->dev, + dev_err(rtd->dev, "ASoC: failed to register runtime device: %d\n", ret); return ret; } @@ -1304,13 +1409,13 @@ static int soc_post_component_init(struct snd_soc_card *card, /* add DAPM sysfs entries for this codec */ ret = snd_soc_dapm_sys_add(rtd->dev); if (ret < 0) - dev_err(codec->dev, + dev_err(rtd->dev, "ASoC: failed to add codec dapm sysfs entries: %d\n", ret); /* add codec sysfs entries */ ret = device_create_file(rtd->dev, &dev_attr_codec_reg); if (ret < 0) - dev_err(codec->dev, + dev_err(rtd->dev, "ASoC: failed to add codec sysfs files: %d\n", ret); #ifdef CONFIG_DEBUG_FS @@ -1332,9 +1437,9 @@ static int soc_probe_link_components(struct snd_soc_card *card, int num, { struct snd_soc_pcm_runtime *rtd = &card->rtd[num]; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai_link_codec *codecs = rtd->codecs; struct snd_soc_platform *platform = rtd->platform; - int ret; + int i, ret; /* probe the CPU-side component, if it is a CODEC */ if (cpu_dai->codec && @@ -1345,12 +1450,14 @@ static int soc_probe_link_components(struct snd_soc_card *card, int num, return ret; } - /* probe the CODEC-side component */ - if (!codec_dai->codec->probed && - codec_dai->codec->driver->probe_order == order) { - ret = soc_probe_codec(card, codec_dai->codec); - if (ret < 0) - return ret; + /* probe the CODEC-side components */ + for (i = 0; i < rtd->num_codecs; i++) { + if (!codecs[i].codec->probed && + codecs[i].codec->driver->probe_order == order) { + ret = soc_probe_codec(card, codecs[i].codec); + if (ret < 0) + return ret; + } } /* probe the platform */ @@ -1364,24 +1471,61 @@ static int soc_probe_link_components(struct snd_soc_card *card, int num, return 0; } +static int soc_link_dai_widgets(struct snd_soc_card *card, + struct snd_soc_dai_link *dai_link, + struct snd_soc_dai *cpu_dai, + struct snd_soc_dai *codec_dai) +{ + struct snd_soc_dapm_widget *play_w, *capture_w; + int ret; + + /* link the DAI widgets */ + play_w = codec_dai->playback_widget; + capture_w = cpu_dai->capture_widget; + if (play_w && capture_w) { + ret = snd_soc_dapm_new_pcm(card, dai_link->params, + capture_w, play_w); + if (ret != 0) { + dev_err(card->dev, "ASoC: Can't link %s to %s: %d\n", + play_w->name, capture_w->name, ret); + return ret; + } + } + + play_w = cpu_dai->playback_widget; + capture_w = codec_dai->capture_widget; + if (play_w && capture_w) { + ret = snd_soc_dapm_new_pcm(card, dai_link->params, + capture_w, play_w); + if (ret != 0) { + dev_err(card->dev, "ASoC: Can't link %s to %s: %d\n", + play_w->name, capture_w->name, ret); + return ret; + } + } + + return 0; +} + + static int soc_probe_link_dais(struct snd_soc_card *card, int num, int order) { struct snd_soc_dai_link *dai_link = &card->dai_link[num]; struct snd_soc_pcm_runtime *rtd = &card->rtd[num]; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_dai_link_codec *codecs = rtd->codecs; struct snd_soc_platform *platform = rtd->platform; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dapm_widget *play_w, *capture_w; - int ret; + int i, ret; dev_dbg(card->dev, "ASoC: probe %s dai link %d late %d\n", card->name, num, order); /* config components */ cpu_dai->platform = platform; - codec_dai->card = card; cpu_dai->card = card; + for (i = 0; i < rtd->num_codecs; i++) + codecs[i].codec_dai->card = card; /* set default power off timeout */ rtd->pmdown_time = pmdown_time; @@ -1413,27 +1557,31 @@ static int soc_probe_link_dais(struct snd_soc_card *card, int num, int order) } /* probe the CODEC DAI */ - if (!codec_dai->probed && codec_dai->driver->probe_order == order) { - if (codec_dai->driver->probe) { - ret = codec_dai->driver->probe(codec_dai); - if (ret < 0) { - dev_err(codec_dai->dev, - "ASoC: failed to probe CODEC DAI %s: %d\n", - codec_dai->name, ret); - return ret; + for (i = 0; i < rtd->num_codecs; i++) { + codec_dai = codecs[i].codec_dai; + if (!codec_dai->probed && + codec_dai->driver->probe_order == order) { + if (codec_dai->driver->probe) { + ret = codec_dai->driver->probe(codec_dai); + if (ret < 0) { + dev_err(codec_dai->dev, + "ASoC: failed to probe CODEC DAI %s: %d\n", + codec_dai->name, ret); + return ret; + } } - } - /* mark codec_dai as probed and add to card dai list */ - codec_dai->probed = 1; - list_add(&codec_dai->card_list, &card->dai_dev_list); + /* mark codec_dai as probed and add to card dai list */ + codec_dai->probed = 1; + list_add(&codec_dai->card_list, &card->dai_dev_list); + } } /* complete DAI probe during last probe */ if (order != SND_SOC_COMP_ORDER_LAST) return 0; - ret = soc_post_component_init(card, codec, num, 0); + ret = soc_post_component_init(card, num, 0); if (ret) return ret; @@ -1465,35 +1613,21 @@ static int soc_probe_link_dais(struct snd_soc_card *card, int num, int order) codec2codec_close_delayed_work); /* link the DAI widgets */ - play_w = codec_dai->playback_widget; - capture_w = cpu_dai->capture_widget; - if (play_w && capture_w) { - ret = snd_soc_dapm_new_pcm(card, dai_link->params, - capture_w, play_w); - if (ret != 0) { - dev_err(card->dev, "ASoC: Can't link %s to %s: %d\n", - play_w->name, capture_w->name, ret); + for (i = 0; i < rtd->num_codecs; i++) { + ret = soc_link_dai_widgets(card, dai_link, + cpu_dai, codecs[i].codec_dai); + if (ret) return ret; - } - } - - play_w = cpu_dai->playback_widget; - capture_w = codec_dai->capture_widget; - if (play_w && capture_w) { - ret = snd_soc_dapm_new_pcm(card, dai_link->params, - capture_w, play_w); - if (ret != 0) { - dev_err(card->dev, "ASoC: Can't link %s to %s: %d\n", - play_w->name, capture_w->name, ret); - return ret; - } } } } /* add platform data for AC97 devices */ - if (rtd->codec_dai->driver->ac97_control) - snd_ac97_dev_add_pdata(codec->ac97, rtd->cpu_dai->ac97_pdata); + for (i = 0; i < rtd->num_codecs; i++) { + if (codecs[i].codec_dai->driver->ac97_control) + snd_ac97_dev_add_pdata(codecs[i].codec->ac97, + rtd->cpu_dai->ac97_pdata); + } return 0; } @@ -1501,31 +1635,42 @@ static int soc_probe_link_dais(struct snd_soc_card *card, int num, int order) #ifdef CONFIG_SND_SOC_AC97_BUS static int soc_register_ac97_dai_link(struct snd_soc_pcm_runtime *rtd) { - int ret; + int i, ret; + codec_dai; + codec; - /* Only instantiate AC97 if not already done by the adaptor - * for the generic AC97 subsystem. - */ - if (rtd->codec_dai->driver->ac97_control && !rtd->codec->ac97_registered) { - /* - * It is possible that the AC97 device is already registered to - * the device subsystem. This happens when the device is created - * via snd_ac97_mixer(). Currently only SoC codec that does so - * is the generic AC97 glue but others migh emerge. - * - * In those cases we don't try to register the device again. + for (i = 0; i < rtd->num_codecs; i++) { + codec = rtd->codec; + codec_dai = rtd->codec_dai; + + /* Only instantiate AC97 if not already done by the adaptor + * for the generic AC97 subsystem. */ - if (!rtd->codec->ac97_created) - return 0; + if (codec_dai->driver->ac97_control && + !codec->ac97_registered) { + /* + * It is possible that the AC97 device is already + * registered to the device subsystem. This happens when + * the device is created via snd_ac97_mixer(). Currently + * only SoC codec that does so is the generic AC97 glue + * but others migh emerge. + * + * In those cases we don't try to register the device + * again. + */ + if (!codec->ac97_created) + return 0; + + ret = soc_ac97_dev_register(codec); + if (ret < 0) { + dev_err(codec->dev, + "ASoC: AC97 device register failed: %d\n", + ret); + return ret; + } - ret = soc_ac97_dev_register(rtd->codec); - if (ret < 0) { - dev_err(rtd->codec->dev, - "ASoC: AC97 device register failed: %d\n", ret); - return ret; + codec->ac97_registered = 1; } - - rtd->codec->ac97_registered = 1; } return 0; } @@ -1582,7 +1727,7 @@ found: if (ret < 0) return ret; - ret = soc_post_component_init(card, codec, num, 1); + ret = soc_post_component_init(card, num, 1); out: return ret; @@ -3694,6 +3839,33 @@ int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute, } EXPORT_SYMBOL_GPL(snd_soc_dai_digital_mute); +static int snd_soc_init_multicodec(struct snd_soc_card *card, + struct snd_soc_dai_link *dai_link) +{ + /* Legacy codec/codec_dai link is a single entry in multicodec */ + if (dai_link->codec_name || dai_link->codec_of_node || + dai_link->codec_dai_name) { + dai_link->num_codecs = 1; + + dai_link->codecs = devm_kzalloc(card->dev, + sizeof(struct snd_soc_dai_link_codec), + GFP_KERNEL); + if (!dai_link->codecs) + return -ENOMEM; + + dai_link->codecs[0].codec_name = dai_link->codec_name; + dai_link->codecs[0].codec_of_node = dai_link->codec_of_node; + dai_link->codecs[0].codec_dai_name = dai_link->codec_dai_name; + } + + if (!dai_link->codecs) { + dev_err(card->dev, "ASoC: DAI link has no CODECs\n"); + return -EINVAL; + } + + return 0; +} + /** * snd_soc_register_card - Register a card with the ASoC core * @@ -3702,7 +3874,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_digital_mute); */ int snd_soc_register_card(struct snd_soc_card *card) { - int i, ret; + int i, j, ret; if (!card->name || !card->dev) return -EINVAL; @@ -3710,22 +3882,29 @@ int snd_soc_register_card(struct snd_soc_card *card) for (i = 0; i < card->num_links; i++) { struct snd_soc_dai_link *link = &card->dai_link[i]; - /* - * Codec must be specified by 1 of name or OF node, - * not both or neither. - */ - if (!!link->codec_name == !!link->codec_of_node) { - dev_err(card->dev, - "ASoC: Neither/both codec name/of_node are set for %s\n", - link->name); - return -EINVAL; + ret = snd_soc_init_multicodec(card, link); + if (ret) { + dev_err(card->dev, "ASoC: failed to init multicodec\n"); + return ret; } - /* Codec DAI name must be specified */ - if (!link->codec_dai_name) { - dev_err(card->dev, - "ASoC: codec_dai_name not set for %s\n", - link->name); - return -EINVAL; + + for (j = 0; j < link->num_codecs; j++) { + /* + * Codec must be specified by 1 of name or OF node, + * not both or neither. + */ + if (!!link->codecs[j].codec_name == + !!link->codecs[j].codec_of_node) { + dev_err(card->dev, "ASoC: Neither/both codec name/of_node are set for %s\n", + link->name); + return -EINVAL; + } + /* Codec DAI name must be specified */ + if (!link->codecs[j].codec_dai_name) { + dev_err(card->dev, "ASoC: codec_dai_name not set for %s\n", + link->name); + return -EINVAL; + } } /* diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index b9dc6ac..f4b4e2a 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2192,12 +2192,8 @@ int snd_soc_dapm_mixer_update_power(struct snd_soc_dapm_context *dapm, } EXPORT_SYMBOL_GPL(snd_soc_dapm_mixer_update_power); -/* show dapm widget status in sys fs */ -static ssize_t dapm_widget_show(struct device *dev, - struct device_attribute *attr, char *buf) +static ssize_t dapm_widget_show_codec(struct snd_soc_codec *codec, char *buf) { - struct snd_soc_pcm_runtime *rtd = dev_get_drvdata(dev); - struct snd_soc_codec *codec =rtd->codec; struct snd_soc_dapm_widget *w; int count = 0; char *state = "not set"; @@ -2250,6 +2246,22 @@ static ssize_t dapm_widget_show(struct device *dev, return count; } +/* show dapm widget status in sys fs */ +static ssize_t dapm_widget_show(struct device *dev, + struct device_attribute *attr, char *buf) +{ + struct snd_soc_pcm_runtime *rtd = dev_get_drvdata(dev); + struct snd_soc_codec *codec; + int i, count = 0; + + for (i = 0; i < rtd->num_codecs; i++) { + codec = rtd->codecs[i].codec; + count += dapm_widget_show_codec(codec, buf + count); + } + + return count; +} + static DEVICE_ATTR(dapm_widget, 0444, dapm_widget_show, NULL); int snd_soc_dapm_sys_add(struct device *dev) @@ -3684,13 +3696,12 @@ void snd_soc_dapm_connect_dai_link_widgets(struct snd_soc_card *card) } } -static void soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd, int stream, - int event) +static void soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd, + struct snd_soc_dai *cpu_dai, + struct snd_soc_dai *codec_dai, + int stream, int event) { - struct snd_soc_dapm_widget *w_cpu, *w_codec; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; if (stream == SNDRV_PCM_STREAM_PLAYBACK) { w_cpu = cpu_dai->playback_widget; @@ -3756,9 +3767,15 @@ void snd_soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd, int stream, int event) { struct snd_soc_card *card = rtd->card; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai; + int i; mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); - soc_dapm_stream_event(rtd, stream, event); + for (i = 0; i < rtd->num_codecs; i++) { + codec_dai = rtd->codecs[i].codec_dai; + soc_dapm_stream_event(rtd, cpu_dai, codec_dai, stream, event); + } mutex_unlock(&card->dapm_mutex); } diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 47e1ce7..9d0bb9e 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -7,7 +7,7 @@ * Copyright (C) 2010 Texas Instruments Inc. * * Authors: Liam Girdwood <lrg(a)ti.com> - * Mark Brown <broonie(a)opensource.wolfsonmicro.com> + * Mark Brown <broonie(a)opensource.wolfsonmicro.com> * * This program is free software; you can redistribute it and/or modify it * under the terms of the GNU General Public License as published by the @@ -230,32 +230,67 @@ static void soc_pcm_apply_msb(struct snd_pcm_substream *substream, } } -static void soc_pcm_init_runtime_hw(struct snd_pcm_runtime *runtime, - struct snd_soc_pcm_stream *codec_stream, - struct snd_soc_pcm_stream *cpu_stream) +static void soc_pcm_init_runtime_hw(struct snd_pcm_substream *substream) { + struct snd_pcm_runtime *runtime = substream->runtime; struct snd_pcm_hardware *hw = &runtime->hw; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai_driver *cpu_dai_drv = rtd->cpu_dai->driver; + struct snd_soc_dai_driver *codec_dai_drv; + struct snd_soc_dai_link_codec *codecs = rtd->codecs; + struct snd_soc_pcm_stream *codec_stream; + struct snd_soc_pcm_stream *cpu_stream; + unsigned int chan_min = 0, chan_max = UINT_MAX; + unsigned int rate_min = 0, rate_max = UINT_MAX; + unsigned int rates = UINT_MAX; + unsigned int sig_bits; + u64 formats = ULLONG_MAX; + int i; - hw->channels_min = max(codec_stream->channels_min, - cpu_stream->channels_min); - hw->channels_max = min(codec_stream->channels_max, - cpu_stream->channels_max); - if (hw->formats) - hw->formats &= codec_stream->formats & cpu_stream->formats; + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + cpu_stream = &cpu_dai_drv->playback; else - hw->formats = codec_stream->formats & cpu_stream->formats; - hw->rates = snd_pcm_rate_mask_intersect(codec_stream->rates, - cpu_stream->rates); + cpu_stream = &cpu_dai_drv->capture; - hw->rate_min = 0; - hw->rate_max = UINT_MAX; + /* first calculate min/max only for CODECs in the DAI link */ + for (i = 0; i < rtd->num_codecs; i++) { + codec_dai_drv = codecs[i].codec_dai->driver; + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + codec_stream = &codec_dai_drv->playback; + else + codec_stream = &codec_dai_drv->capture; + chan_min = max(chan_min, codec_stream->channels_min); + chan_max = min(chan_max, codec_stream->channels_max); + rate_min = max(rate_min, codec_stream->rate_min); + rate_max = min_not_zero(rate_max, codec_stream->rate_max); + formats &= codec_stream->formats; + rates = snd_pcm_rate_mask_intersect(codec_stream->rates, rates); + } + + /* + * chan min/max cannot be enforced if there are multiple CODEC DAIs + * connected to a single CPU DAI, use CPU DAI's directly and let + * channel allocation be fixed up later + */ + if (rtd->num_codecs > 1) { + chan_min = cpu_stream->channels_min; + chan_max = cpu_stream->channels_max; + } + + hw->channels_min = max(chan_min, cpu_stream->channels_min); + hw->channels_max = min(chan_max, cpu_stream->channels_max); + if (hw->formats) + hw->formats &= formats & cpu_stream->formats; + else + hw->formats = formats & cpu_stream->formats; + hw->rates = snd_pcm_rate_mask_intersect(rates, cpu_stream->rates); snd_pcm_limit_hw_rates(runtime); hw->rate_min = max(hw->rate_min, cpu_stream->rate_min); - hw->rate_min = max(hw->rate_min, codec_stream->rate_min); + hw->rate_min = max(hw->rate_min, rate_min); hw->rate_max = min_not_zero(hw->rate_max, cpu_stream->rate_max); - hw->rate_max = min_not_zero(hw->rate_max, codec_stream->rate_max); + hw->rate_max = min_not_zero(hw->rate_max, rate_max); } /* @@ -268,16 +303,18 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_platform *platform = rtd->platform; + struct snd_soc_dai_link_codec *codecs = rtd->codecs; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai_driver *cpu_dai_drv = cpu_dai->driver; - struct snd_soc_dai_driver *codec_dai_drv = codec_dai->driver; - int ret = 0; + struct snd_soc_dai *codec_dai; + const char *codec_dai_name = "multicodec"; + int i, ret = 0; pinctrl_pm_select_default_state(cpu_dai->dev); - pinctrl_pm_select_default_state(codec_dai->dev); + for (i = 0; i < rtd->num_codecs; i++) + pinctrl_pm_select_default_state(codecs[i].codec_dai->dev); pm_runtime_get_sync(cpu_dai->dev); - pm_runtime_get_sync(codec_dai->dev); + for (i = 0; i < rtd->num_codecs; i++) + pm_runtime_get_sync(codecs[i].codec_dai->dev); pm_runtime_get_sync(platform->dev); mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); @@ -301,12 +338,17 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) } } - if (codec_dai->driver->ops && codec_dai->driver->ops->startup) { - ret = codec_dai->driver->ops->startup(substream, codec_dai); - if (ret < 0) { - dev_err(codec_dai->dev, "ASoC: can't open codec" - " %s: %d\n", codec_dai->name, ret); - goto codec_dai_err; + for (i = 0; i < rtd->num_codecs; i++) { + codec_dai = codecs[i].codec_dai; + if (codec_dai->driver->ops && codec_dai->driver->ops->startup) { + ret = codec_dai->driver->ops->startup(substream, + codec_dai); + if (ret < 0) { + dev_err(codec_dai->dev, + "ASoC: can't open codec %s: %d\n", + codec_dai->name, ret); + goto codec_dai_err; + } } } @@ -324,13 +366,10 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) goto dynamic; /* Check that the codec and cpu DAIs are compatible */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - soc_pcm_init_runtime_hw(runtime, &codec_dai_drv->playback, - &cpu_dai_drv->playback); - } else { - soc_pcm_init_runtime_hw(runtime, &codec_dai_drv->capture, - &cpu_dai_drv->capture); - } + soc_pcm_init_runtime_hw(substream); + + if (rtd->num_codecs == 1) + codec_dai_name = rtd->codec_dai->name; if (soc_pcm_has_symmetry(substream)) runtime->hw.info |= SNDRV_PCM_INFO_JOINT_DUPLEX; @@ -338,22 +377,22 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) ret = -EINVAL; if (!runtime->hw.rates) { printk(KERN_ERR "ASoC: %s <-> %s No matching rates\n", - codec_dai->name, cpu_dai->name); + codec_dai_name, cpu_dai->name); goto config_err; } if (!runtime->hw.formats) { printk(KERN_ERR "ASoC: %s <-> %s No matching formats\n", - codec_dai->name, cpu_dai->name); + codec_dai_name, cpu_dai->name); goto config_err; } if (!runtime->hw.channels_min || !runtime->hw.channels_max || runtime->hw.channels_min > runtime->hw.channels_max) { printk(KERN_ERR "ASoC: %s <-> %s No matching channels\n", - codec_dai->name, cpu_dai->name); + codec_dai_name, cpu_dai->name); goto config_err; } - soc_pcm_apply_msb(substream, codec_dai); + soc_pcm_apply_msb(substream, codecs[0].codec_dai); soc_pcm_apply_msb(substream, cpu_dai); /* Symmetry only applies if we've already got an active stream. */ @@ -363,14 +402,17 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) goto config_err; } - if (codec_dai->active) { - ret = soc_pcm_apply_symmetry(substream, codec_dai); - if (ret != 0) - goto config_err; + for (i = 0; i < rtd->num_codecs; i++) { + if (codecs[i].codec_dai->active) { + ret = soc_pcm_apply_symmetry(substream, + codecs[i].codec_dai); + if (ret != 0) + goto config_err; + } } pr_debug("ASoC: %s <-> %s info:\n", - codec_dai->name, cpu_dai->name); + codec_dai_name, cpu_dai->name); pr_debug("ASoC: rate mask 0x%x\n", runtime->hw.rates); pr_debug("ASoC: min ch %d max ch %d\n", runtime->hw.channels_min, runtime->hw.channels_max); @@ -380,14 +422,18 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) dynamic: if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { cpu_dai->playback_active++; - codec_dai->playback_active++; + for (i = 0; i < rtd->num_codecs; i++) + codecs[i].codec_dai->playback_active++; } else { cpu_dai->capture_active++; - codec_dai->capture_active++; + for (i = 0; i < rtd->num_codecs; i++) + codecs[i].codec_dai->capture_active++; } cpu_dai->active++; - codec_dai->active++; - rtd->codec->active++; + for (i = 0; i < rtd->num_codecs; i++) { + codecs[i].codec_dai->active++; + codecs[i].codec->active++; + } mutex_unlock(&rtd->pcm_mutex); return 0; @@ -396,10 +442,15 @@ config_err: rtd->dai_link->ops->shutdown(substream); machine_err: - if (codec_dai->driver->ops->shutdown) - codec_dai->driver->ops->shutdown(substream, codec_dai); + i = rtd->num_codecs; codec_dai_err: + while (--i >= 0) { + codec_dai = codecs[i].codec_dai; + if (codec_dai->driver->ops->shutdown) + codec_dai->driver->ops->shutdown(substream, codec_dai); + } + if (platform->driver->ops && platform->driver->ops->close) platform->driver->ops->close(substream); @@ -410,10 +461,13 @@ out: mutex_unlock(&rtd->pcm_mutex); pm_runtime_put(platform->dev); - pm_runtime_put(codec_dai->dev); + for (i = 0; i < rtd->num_codecs; i++) + pm_runtime_put(codecs[i].codec_dai->dev); pm_runtime_put(cpu_dai->dev); - if (!codec_dai->active) - pinctrl_pm_select_sleep_state(codec_dai->dev); + for (i = 0; i < rtd->num_codecs; i++) { + if (!codecs[i].codec_dai->active) + pinctrl_pm_select_sleep_state(codecs[i].codec_dai->dev); + } if (!cpu_dai->active) pinctrl_pm_select_sleep_state(cpu_dai->dev); @@ -429,7 +483,7 @@ static void close_delayed_work(struct work_struct *work) { struct snd_soc_pcm_runtime *rtd = container_of(work, struct snd_soc_pcm_runtime, delayed_work.work); - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = rtd->codecs[0].codec_dai; mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); @@ -457,36 +511,47 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_platform *platform = rtd->platform; + struct snd_soc_dai_link_codec *codecs = rtd->codecs; 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; + struct snd_soc_dai *codec_dai; + int i; mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { cpu_dai->playback_active--; - codec_dai->playback_active--; + for (i = 0; i < rtd->num_codecs; i++) + codecs[i].codec_dai->playback_active--; } else { cpu_dai->capture_active--; - codec_dai->capture_active--; + for (i = 0; i < rtd->num_codecs; i++) + codecs[i].codec_dai->capture_active--; } cpu_dai->active--; - codec_dai->active--; - codec->active--; + for (i = 0; i < rtd->num_codecs; i++) { + codecs[i].codec_dai->active--; + codecs[i].codec->active--; + } /* clear the corresponding DAIs rate when inactive */ if (!cpu_dai->active) cpu_dai->rate = 0; - if (!codec_dai->active) - codec_dai->rate = 0; + for (i = 0; i < rtd->num_codecs; i++) { + codec_dai = codecs[i].codec_dai; + if (!codec_dai->active) + codec_dai->rate = 0; + } if (cpu_dai->driver->ops->shutdown) cpu_dai->driver->ops->shutdown(substream, cpu_dai); - if (codec_dai->driver->ops->shutdown) - codec_dai->driver->ops->shutdown(substream, codec_dai); + for (i = 0; i < rtd->num_codecs; i++) { + codec_dai = codecs[i].codec_dai; + if (codec_dai->driver->ops->shutdown) + codec_dai->driver->ops->shutdown(substream, codec_dai); + } if (rtd->dai_link->ops && rtd->dai_link->ops->shutdown) rtd->dai_link->ops->shutdown(substream); @@ -496,8 +561,13 @@ 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) { + int ignore = rtd->dai_link->ignore_pmdown_time || + !rtd->pmdown_time; + + for (i = 0; (i < rtd->num_codecs) && !ignore; i++) + ignore |= codecs[i].codec->ignore_pmdown_time; + + if (ignore) { /* powered down playback stream now */ snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_PLAYBACK, @@ -518,10 +588,13 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) mutex_unlock(&rtd->pcm_mutex); pm_runtime_put(platform->dev); - pm_runtime_put(codec_dai->dev); + for (i = 0; i < rtd->num_codecs; i++) + pm_runtime_put(codecs[i].codec_dai->dev); pm_runtime_put(cpu_dai->dev); - if (!codec_dai->active) - pinctrl_pm_select_sleep_state(codec_dai->dev); + for (i = 0; i < rtd->num_codecs; i++) { + if (!codecs[i].codec_dai->active) + pinctrl_pm_select_sleep_state(codecs[i].codec_dai->dev); + } if (!cpu_dai->active) pinctrl_pm_select_sleep_state(cpu_dai->dev); @@ -537,9 +610,10 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_platform *platform = rtd->platform; + struct snd_soc_dai_link_codec *codecs = rtd->codecs; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - int ret = 0; + struct snd_soc_dai *codec_dai; + int i, ret = 0; mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); @@ -561,12 +635,16 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) } } - if (codec_dai->driver->ops && codec_dai->driver->ops->prepare) { - ret = codec_dai->driver->ops->prepare(substream, codec_dai); - if (ret < 0) { - dev_err(codec_dai->dev, "ASoC: DAI prepare error: %d\n", - ret); - goto out; + for (i = 0; i < rtd->num_codecs; i++) { + codec_dai = codecs[i].codec_dai; + if (codec_dai->driver->ops && codec_dai->driver->ops->prepare) { + ret = codec_dai->driver->ops->prepare(substream, + codec_dai); + if (ret < 0) { + dev_err(codec_dai->dev, + "ASoC: DAI prepare error: %d\n", ret); + goto out; + } } } @@ -589,7 +667,9 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) snd_soc_dapm_stream_event(rtd, substream->stream, SND_SOC_DAPM_STREAM_START); - snd_soc_dai_digital_mute(codec_dai, 0, substream->stream); + for (i = 0; i < rtd->num_codecs; i++) + snd_soc_dai_digital_mute(codecs[i].codec_dai, 0, + substream->stream); out: mutex_unlock(&rtd->pcm_mutex); @@ -606,9 +686,10 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_platform *platform = rtd->platform; + struct snd_soc_dai_link_codec *codecs = rtd->codecs; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - int ret = 0; + struct snd_soc_dai *codec_dai; + int i, ret = 0; mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); @@ -625,13 +706,38 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, } } - if (codec_dai->driver->ops && codec_dai->driver->ops->hw_params) { - ret = codec_dai->driver->ops->hw_params(substream, params, codec_dai); - if (ret < 0) { - dev_err(codec_dai->dev, "ASoC: can't set %s hw params:" - " %d\n", codec_dai->name, ret); - goto codec_err; + for (i = 0; i < rtd->num_codecs; i++) { + struct snd_pcm_hw_params *codec_params; + + codec_dai = codecs[i].codec_dai; + codec_params = &codecs[i].hw_params; + + /* copy params for each codec */ + memcpy(codec_params, params, sizeof(struct snd_pcm_hw_params)); + + if (codecs[i].hw_params_fixup) { + ret = codecs[i].hw_params_fixup(rtd, codec_params); + if (ret < 0) { + dev_err(codec_dai->dev, + "ASoC: multicodec hw_params fixup failed %d\n", + ret); + goto codec_err; + } } + + if (codec_dai->driver->ops && + codec_dai->driver->ops->hw_params) { + ret = codec_dai->driver->ops->hw_params(substream, + codec_params, codec_dai); + if (ret < 0) { + dev_err(codec_dai->dev, + "ASoC: can't set %s hw params: %d\n", + codec_dai->name, ret); + goto codec_err; + } + } + + codec_dai->rate = params_rate(codec_params); } if (cpu_dai->driver->ops && cpu_dai->driver->ops->hw_params) { @@ -672,10 +778,16 @@ platform_err: cpu_dai->driver->ops->hw_free(substream, cpu_dai); interface_err: - if (codec_dai->driver->ops && codec_dai->driver->ops->hw_free) - codec_dai->driver->ops->hw_free(substream, codec_dai); + i = rtd->num_codecs; codec_err: + while (--i >= 0) { + codec_dai = codecs[i].codec_dai; + if (codec_dai->driver->ops && codec_dai->driver->ops->hw_free) + codec_dai->driver->ops->hw_free(substream, codec_dai); + codec_dai->rate = 0; + } + if (rtd->dai_link->ops && rtd->dai_link->ops->hw_free) rtd->dai_link->ops->hw_free(substream); @@ -690,9 +802,11 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_platform *platform = rtd->platform; + struct snd_soc_dai_link_codec *codecs = rtd->codecs; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai; bool playback = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + int i; mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); @@ -703,16 +817,21 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) cpu_dai->sample_bits = 0; } - if (codec_dai->active == 1) { - codec_dai->rate = 0; - codec_dai->channels = 0; - codec_dai->sample_bits = 0; + for (i = 0; i < rtd->num_codecs; i++) { + codec_dai = codecs[i].codec_dai; + if (codec_dai->active == 1) + codec_dai->rate = 0; + codec_dai->channels = 0; + codec_dai->sample_bits = 0; } /* apply codec digital mute */ - if ((playback && codec_dai->playback_active == 1) || - (!playback && codec_dai->capture_active == 1)) - snd_soc_dai_digital_mute(codec_dai, 1, substream->stream); + for (i = 0; i < rtd->num_codecs; i++) { + if ((playback && codecs[i].codec_dai->playback_active == 1) || + (!playback && codecs[i].codec_dai->capture_active == 1)) + snd_soc_dai_digital_mute(codecs[i].codec_dai, 1, + substream->stream); + } /* free any machine hw params */ if (rtd->dai_link->ops && rtd->dai_link->ops->hw_free) @@ -723,8 +842,11 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) platform->driver->ops->hw_free(substream); /* now free hw params for the DAIs */ - if (codec_dai->driver->ops && codec_dai->driver->ops->hw_free) - codec_dai->driver->ops->hw_free(substream, codec_dai); + for (i = 0; i < rtd->num_codecs; i++) { + codec_dai = codecs[i].codec_dai; + if (codec_dai->driver->ops && codec_dai->driver->ops->hw_free) + codec_dai->driver->ops->hw_free(substream, codec_dai); + } if (cpu_dai->driver->ops && cpu_dai->driver->ops->hw_free) cpu_dai->driver->ops->hw_free(substream, cpu_dai); @@ -737,14 +859,19 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_platform *platform = rtd->platform; + struct snd_soc_dai_link_codec *codecs = rtd->codecs; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - int ret; - - if (codec_dai->driver->ops && codec_dai->driver->ops->trigger) { - ret = codec_dai->driver->ops->trigger(substream, cmd, codec_dai); - if (ret < 0) - return ret; + struct snd_soc_dai *codec_dai; + int i, ret; + + for (i = 0; i < rtd->num_codecs; i++) { + codec_dai = codecs[i].codec_dai; + if (codec_dai->driver->ops && codec_dai->driver->ops->trigger) { + ret = codec_dai->driver->ops->trigger(substream, + cmd, codec_dai); + if (ret < 0) + return ret; + } } if (platform->driver->ops && platform->driver->ops->trigger) { @@ -766,15 +893,20 @@ static int soc_pcm_bespoke_trigger(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_platform *platform = rtd->platform; + struct snd_soc_dai_link_codec *codecs = rtd->codecs; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - int ret; - - if (codec_dai->driver->ops && - codec_dai->driver->ops->bespoke_trigger) { - ret = codec_dai->driver->ops->bespoke_trigger(substream, cmd, codec_dai); - if (ret < 0) - return ret; + struct snd_soc_dai *codec_dai; + int i, ret; + + for (i = 0; i < rtd->num_codecs; i++) { + codec_dai = codecs[i].codec_dai; + if (codec_dai->driver->ops && + codec_dai->driver->ops->bespoke_trigger) { + ret = codec_dai->driver->ops->bespoke_trigger(substream, + cmd, codec_dai); + if (ret < 0) + return ret; + } } if (platform->driver->bespoke_trigger) { @@ -799,11 +931,14 @@ static snd_pcm_uframes_t soc_pcm_pointer(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_platform *platform = rtd->platform; + struct snd_soc_dai_link_codec *codecs = rtd->codecs; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai; struct snd_pcm_runtime *runtime = substream->runtime; snd_pcm_uframes_t offset = 0; snd_pcm_sframes_t delay = 0; + snd_pcm_sframes_t codec_delay = 0; + int i; if (platform->driver->ops && platform->driver->ops->pointer) offset = platform->driver->ops->pointer(substream); @@ -811,11 +946,22 @@ static snd_pcm_uframes_t soc_pcm_pointer(struct snd_pcm_substream *substream) if (cpu_dai->driver->ops && cpu_dai->driver->ops->delay) delay += cpu_dai->driver->ops->delay(substream, cpu_dai); - if (codec_dai->driver->ops && codec_dai->driver->ops->delay) - delay += codec_dai->driver->ops->delay(substream, codec_dai); + for (i = 0; i < rtd->num_codecs; i++) { + codec_dai = codecs[i].codec_dai; + if (codec_dai->driver->ops && codec_dai->driver->ops->delay) + codec_delay = max(codec_delay, + codec_dai->driver->ops->delay(substream, + codec_dai)); + } + delay += codec_delay; + /* + * None of the existing platform drivers implement delay(), so + * for now the codec_dai of first multicodec entry is used + */ if (platform->driver->delay) - delay += platform->driver->delay(substream, codec_dai); + delay += platform->driver->delay(substream, + codecs[0].codec_dai); runtime->delay = delay; @@ -2129,22 +2275,29 @@ static int dpcm_fe_dai_close(struct snd_pcm_substream *fe_substream) int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) { struct snd_soc_platform *platform = rtd->platform; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai_link_codec *codecs = rtd->codecs; + struct snd_soc_dai *codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct snd_pcm *pcm; char new_name[64]; int ret = 0, playback = 0, capture = 0; + int i; if (rtd->dai_link->dynamic || rtd->dai_link->no_pcm) { playback = rtd->dai_link->dpcm_playback; capture = rtd->dai_link->dpcm_capture; } else { - if (codec_dai->driver->playback.channels_min && - cpu_dai->driver->playback.channels_min) - playback = 1; - if (codec_dai->driver->capture.channels_min && - cpu_dai->driver->capture.channels_min) - capture = 1; + for (i = 0; i < rtd->num_codecs; i++) { + codec_dai = codecs[i].codec_dai; + if (codec_dai->driver->playback.channels_min && + cpu_dai->driver->playback.channels_min) + playback++; + if (codec_dai->driver->capture.channels_min && + cpu_dai->driver->capture.channels_min) + capture++; + } + capture = (rtd->num_codecs == capture); + playback = (rtd->num_codecs == playback); } if (rtd->dai_link->playback_only) { @@ -2170,7 +2323,9 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) rtd->dai_link->stream_name); else snprintf(new_name, sizeof(new_name), "%s %s-%d", - rtd->dai_link->stream_name, codec_dai->name, num); + rtd->dai_link->stream_name, + (rtd->num_codecs > 1) ? + "multicodec" : rtd->codec_dai->name, num); ret = snd_pcm_new(rtd->card->snd_card, new_name, num, playback, capture, &pcm); @@ -2243,8 +2398,9 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) pcm->private_free = platform->driver->pcm_free; out: - dev_info(rtd->card->dev, "%s <-> %s mapping ok\n", codec_dai->name, - cpu_dai->name); + dev_info(rtd->card->dev, "%s <-> %s mapping ok\n", + (rtd->num_codecs > 1) ? "multicodec" : rtd->codec_dai->name, + cpu_dai->name); return ret; } -- 1.8.3.2

2 5

[alsa-devel] [GIT PULL] ASoC updates for v3.15
by Mark Brown 13 Mar '14

13 Mar '14

The following changes since commit e00447fafbf7daf2cd49205b97e63d9734068a4f: ASoC: tlv320aic31xx: Add basic codec driver implementation (2014-03-13 12:09:51 +0000) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/tlv320aic31xx-fixes for you to fetch changes up to 9296f4da3bafa23d8b9abc5cd271a66ea8f90cd2: ASoC: tlv320aic31xx: Staticise non-exported symbols (2014-03-13 17:44:22 +0000) ---------------------------------------------------------------- ASoC: Fixes for tlv320aic31xx A few fixes for issues not yet identified by the zero day tester due to the last pull request being rushed in order to clean up the problem with the git mismerge. ---------------------------------------------------------------- Jyri Sarha (2): ASoC: tlv320aic31xx: Fix unused variable warning from aic31xx_clk_off ASoC: tlv320aic31xx: Remove snd_soc_codec_set_cache_io() call Mark Brown (1): ASoC: tlv320aic31xx: Staticise non-exported symbols sound/soc/codecs/tlv320aic31xx.c | 20 +++++--------------- 1 file changed, 5 insertions(+), 15 deletions(-)

2 1

[alsa-devel] [PATCH] ASoC: tlv320aic31xx: Remove snd_soc_codec_set_cache_io() call
by Jyri Sarha 13 Mar '14

13 Mar '14

Remove snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP) call and codec->control_data = aic31xx->regmap assignment since that already done by core. Signed-off-by: Jyri Sarha <jsarha(a)ti.com> --- This patch should fix the last minute breakkage of v3.15 ASoC updates build. sound/soc/codecs/tlv320aic31xx.c | 9 --------- 1 file changed, 9 deletions(-) diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index c9f6989..625b184 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -1051,18 +1051,9 @@ static int aic31xx_codec_probe(struct snd_soc_codec *codec) dev_dbg(aic31xx->dev, "## %s\n", __func__); aic31xx = snd_soc_codec_get_drvdata(codec); - codec->control_data = aic31xx->regmap; aic31xx->codec = codec; - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); - - if (ret != 0) { - dev_err(codec->dev, "snd_soc_codec_set_cache_io failed %d\n", - ret); - return ret; - } - for (i = 0; i < ARRAY_SIZE(aic31xx->supplies); i++) { aic31xx->disable_nb[i].nb.notifier_call = aic31xx_regulator_event; -- 1.7.9.5

2 1

[alsa-devel] [PATCH v6] ASoC: tlv320aic31xx: Fix unused variable warning from aic31xx_clk_off
by Jyri Sarha 13 Mar '14

13 Mar '14

Fix "warning: unused variable 'aic31xx'" from function 'aic31xx_clk_off'. Signed-off-by: Jyri Sarha <jsarha(a)ti.com> --- sound/soc/codecs/tlv320aic31xx.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index e626b56..c9f6989 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -945,7 +945,6 @@ static void aic31xx_clk_on(struct snd_soc_codec *codec) static void aic31xx_clk_off(struct snd_soc_codec *codec) { - struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec); u8 mask = AIC31XX_PM_MASK; u8 off = 0; -- 1.7.9.5

2 1

[alsa-devel] [PATCH v6] ASoC: tlv320aic31xx: Add basic codec driver implementation
by Jyri Sarha 13 Mar '14

13 Mar '14

This commit adds a bare bones driver support for TLV320AIC31XX family audio codecs. The driver adds basic stereo playback trough headphone and speaker outputs and mono capture trough microphone inputs. The driver is currently missing support at least for mini DSP features and jack detection. I have tested the driver only on TLV320AIC3111, but based on the data sheets TLV320AIC3100, TLV320AIC3110, and TLV320AIC3120 should work Ok too. The base for the implementation was taken from: git@gitorious.org:ti-codecs/ti-codecs.git ajitk/topics/k3.10.1-aic31xx -branch at commit 77504eba0294764e9e63b4a0c696b44db187cd13. Signed-off-by: Jyri Sarha <jsarha(a)ti.com> --- Since v5 of this patch: - Fix "warning: unused variable 'aic31xx'" from function 'aic31xx_clk_off' .../devicetree/bindings/sound/tlv320aic31xx.txt | 61 + include/dt-bindings/sound/tlv320aic31xx-micbias.h | 8 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tlv320aic31xx.c | 1294 ++++++++++++++++++++ sound/soc/codecs/tlv320aic31xx.h | 258 ++++ 6 files changed, 1627 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/tlv320aic31xx.txt create mode 100644 include/dt-bindings/sound/tlv320aic31xx-micbias.h create mode 100644 sound/soc/codecs/tlv320aic31xx.c create mode 100644 sound/soc/codecs/tlv320aic31xx.h diff --git a/Documentation/devicetree/bindings/sound/tlv320aic31xx.txt b/Documentation/devicetree/bindings/sound/tlv320aic31xx.txt new file mode 100644 index 0000000..74c66de --- /dev/null +++ b/Documentation/devicetree/bindings/sound/tlv320aic31xx.txt @@ -0,0 +1,61 @@ +Texas Instruments - tlv320aic31xx Codec module + +The tlv320aic31xx serial control bus communicates through I2C protocols + +Required properties: + +- compatible - "string" - One of: + "ti,tlv320aic310x" - Generic TLV320AIC31xx with mono speaker amp + "ti,tlv320aic311x" - Generic TLV320AIC31xx with stereo speaker amp + "ti,tlv320aic3100" - TLV320AIC3100 (mono speaker amp, no MiniDSP) + "ti,tlv320aic3110" - TLV320AIC3110 (stereo speaker amp, no MiniDSP) + "ti,tlv320aic3120" - TLV320AIC3120 (mono speaker amp, MiniDSP) + "ti,tlv320aic3111" - TLV320AIC3111 (stereo speaker amp, MiniDSP) + +- reg - <int> - I2C slave address + + +Optional properties: + +- gpio-reset - gpio pin number used for codec reset +- ai31xx-micbias-vg - MicBias Voltage setting + 1 or MICBIAS_2_0V - MICBIAS output is powered to 2.0V + 2 or MICBIAS_2_5V - MICBIAS output is powered to 2.5V + 3 or MICBIAS_AVDD - MICBIAS output is connected to AVDD + If this node is not mentioned or if the value is unknown, then + micbias is set to 2.0V. +- HPVDD-supply, SPRVDD-supply, SPLVDD-supply, AVDD-supply, IOVDD-supply, + DVDD-supply : power supplies for the device as covered in + Documentation/devicetree/bindings/regulator/regulator.txt + +CODEC output pins: + * HPL + * HPR + * SPL, devices with stereo speaker amp + * SPR, devices with stereo speaker amp + * SPK, devices with mono speaker amp + * MICBIAS + +CODEC input pins: + * MIC1LP + * MIC1RP + * MIC1LM + +The pins can be used in referring sound node's audio-routing property. + +Example: +#include <dt-bindings/sound/tlv320aic31xx-micbias.h> + +tlv320aic31xx: tlv320aic31xx@18 { + compatible = "ti,tlv320aic311x"; + reg = <0x18>; + + ai31xx-micbias-vg = <MICBIAS_OFF>; + + HPVDD-supply = <&regulator>; + SPRVDD-supply = <&regulator>; + SPLVDD-supply = <&regulator>; + AVDD-supply = <&regulator>; + IOVDD-supply = <&regulator>; + DVDD-supply = <&regulator>; +}; diff --git a/include/dt-bindings/sound/tlv320aic31xx-micbias.h b/include/dt-bindings/sound/tlv320aic31xx-micbias.h new file mode 100644 index 0000000..f5cb772 --- /dev/null +++ b/include/dt-bindings/sound/tlv320aic31xx-micbias.h @@ -0,0 +1,8 @@ +#ifndef __DT_TLV320AIC31XX_MICBIAS_H +#define __DT_TLV320AIC31XX_MICBIAS_H + +#define MICBIAS_2_0V 1 +#define MICBIAS_2_5V 2 +#define MICBIAS_AVDDV 3 + +#endif /* __DT_TLV320AIC31XX_MICBIAS_H */ diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index e19b64f..af3c049 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -83,6 +83,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_TAS5086 if I2C select SND_SOC_TLV320AIC23 if I2C 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 @@ -428,6 +429,9 @@ config SND_SOC_TLV320AIC26 tristate depends on SPI +config SND_SOC_TLV320AIC31XX + tristate + config SND_SOC_TLV320AIC32X4 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 90ab24b..018e3f3 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -75,6 +75,7 @@ snd-soc-stac9766-objs := stac9766.o snd-soc-tas5086-objs := tas5086.o snd-soc-tlv320aic23-objs := tlv320aic23.o snd-soc-tlv320aic26-objs := tlv320aic26.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 @@ -215,6 +216,7 @@ 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_TLV320AIC26) += snd-soc-tlv320aic26.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 diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c new file mode 100644 index 0000000..bdc0d8b --- /dev/null +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -0,0 +1,1294 @@ +/* + * ALSA SoC TLV320AIC31XX codec driver + * + * Copyright (C) 2014 Texas Instruments, Inc. + * + * Author: Jyri Sarha <jsarha(a)ti.com> + * + * Based on ground work by: Ajit Kulkarni <x0175765(a)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, + }, +}; + +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); + +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 *tmp) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->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: + 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); + codec->control_data = aic31xx->regmap; + + aic31xx->codec = codec; + + ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); + + if (ret != 0) { + dev_err(codec->dev, "snd_soc_codec_set_cache_io failed %d\n", + ret); + return ret; + } + + 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 */ + +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); + + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_driver_aic31xx, + aic31xx_dai_driver, + ARRAY_SIZE(aic31xx_dai_driver)); + + return ret; +} + +static int aic31xx_i2c_remove(struct i2c_client *i2c) +{ + struct aic31xx_priv *aic31xx = dev_get_drvdata(&i2c->dev); + + kfree(aic31xx); + 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 */ -- 1.7.9.5

2 1

[alsa-devel] ASoC updates for v3.15
by Mark Brown 13 Mar '14

13 Mar '14

The following changes since commit deeed33850c8a376addabbf971df433b2a1ba74c: Merge remote-tracking branches 'asoc/topic/wm8996', 'asoc/topic/wm9081' and 'asoc/topic/wm9705' into asoc-next (2014-03-12 23:05:31 +0000) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.15-2 for you to fetch changes up to 56fe48976db38ed3984e0e68b71e503e8179dd89: Merge remote-tracking branch 'asoc/topic/tlv320aic31xx' into asoc-next (2014-03-13 14:19:54 +0000) ---------------------------------------------------------------- ASoC: Updates for v3.15 This is mostly a few additional fixes from Lars-Peter, a new driver and cleaning up a git failure with merging the Intel branch (combined with an xargs failure to pay attention to error codes). The history lists a bunch of additional commits for the branch but the content of those commits is actually present already but not recorded in history due to git failing. Unfortunately xargs is used in the merge script and it doesn't do a good job of noticing errors from the commands it invokes. ---------------------------------------------------------------- Dan Carpenter (4): ASoC: Intel: sst-firmware: missing curly braces (harmless) ASoC: intel: restore IRQs on error ASoC: intel: incorrect sizeof() ASoC: Baytrail: fix error handling in sst_byt_dsp_init() Jarkko Nikula (15): ASoC: Intel: Add common SST driver loader on ACPI systems ASoC: Intel: Move extended fw base and size fields in struct sst_pdata ASoC: Intel: sst-acpi: Request firmware before SST platform driver probing ASoC: Intel: sst-acpi: Add support for multiple machine drivers per platform ASoC: Intel: Add Baytrail SST ID and Baytrail specific register bits ASoC: Intel: Add Intel Baytrail SST DSP support ASoC: Intel: Add Intel Baytrail SST DSP IPC support ASoC: Intel: Add Intel Baytrail SST PCM platform driver ASoC: Intel: Add machine driver for Baytrail SST with RT5640 codec ASoC: Intel: Add Baytrail SST and byt-rt5640 machine driver probing ASoC: Intel: Add build support for Baytrail SST ASoC: Intel: sst-acpi: Fix Oops in case of missing firmware ASoC: Intel: byt-rt5640: Update internal mic and speaker kcontrol names ASoC: Intel: byt-rt5640: Use init time DAI format ASoC: Intel: Baytrail: Fix implicit declaration of function 'memcpy_fromio' Jyri Sarha (1): ASoC: tlv320aic31xx: Add basic codec driver implementation Lars-Peter Clausen (5): ASoC: ams-delta: Fix compile error ASoC: snappercl15: Convert to table based DAPM setup ASoC: pxa: magician: Convert to table based DAPM and control setup ASoC: pxa: tosa: Convert to table based DAPM and control setup ASoC: s6105-ipcam: Convert to table based DAPM setup Liam Girdwood (18): ASoC: Intel: Add a mfld prefix to Intel SST drivers. ASoC: Intel: Add Intel SST audio DSP low level shim driver. ASoC: Intel: Add Intel SST audio DSP Firmware loader. ASoC: Intel: Add trace support for generic SST IPC messages. ASoC: Intel: Add build support for Intel SST DSP core. ASoC: Intel: Add GFP_KERNEL flag to firmware DMA buffer. ASoC: Intel: Rename SST trace event header to be less generic. ASoC: Intel: Fix sparse warnings for firmware loader ASoC: Intel: Fix build for sst-dsp.c on PPC architecture ASoC: Intel: Add support for Haswell/Broadwell DSP ASoC: Intel: Add Haswell/Broadwell IPC ASoC: Intel: Add Haswell and Broadwell PCM platform driver ASoC: Intel: Add trace support for Haswell/Broadwell SST IPC messages. ASoC: Intel: Add build support for Haswell ADSP ASoC: Intel: Add Haswell Machine support ASoC: Intel: Check Haswell IPC process_reply/notification return value. ASoC: Intel: Use .dai_fmt for setting Haswell BE format. ASoC: Intel: Clean up indentation for Haswell machine driver/Kconfig Mark Brown (26): MAINTAINERS: Add entry for Cirrus CODEC drivers ASoC: io: Remove SPI support Merge remote-tracking branches 'asoc/topic/ak4671', 'asoc/topic/cs42l51' and 'asoc/topic/alc5623' into asoc-io ASoC: io: Remove SND_SOC_I2C ASoC: da732x: Replace hw_read usage with snd_soc_read() ASoC: io: Remove hw_read() operation Merge branches 'topic/ad193x', 'topic/tlv320aic23', 'topic/tlv320aic32x4', 'topic/wm8991', 'fix/si476x' and 'fix/88pm860' of git://git.kernel.org/.../broonie/sound into asoc-io Merge branch 'topic/dapm' of git://git.kernel.org/.../broonie/sound into asoc-pxa Merge tag 'asoc-v3.15' into asoc-intel Merge tag 'asoc-v3.14-rc6' into asoc-linus Merge remote-tracking branch 'asoc/fix/spear' into asoc-linus Merge tag 'asoc-v3.15' into asoc-next Merge remote-tracking branch 'asoc/topic/dapm' into asoc-next Merge remote-tracking branch 'asoc/topic/cirrus' into asoc-next Merge remote-tracking branch 'asoc/topic/intel' into asoc-next Merge remote-tracking branch 'asoc/topic/io' into asoc-next Merge remote-tracking branch 'asoc/topic/jack' into asoc-next Merge remote-tracking branch 'asoc/topic/kirkwood' into asoc-next Merge remote-tracking branch 'asoc/topic/lm4857' into asoc-next Merge remote-tracking branch 'asoc/topic/lm49453' into asoc-next Merge remote-tracking branch 'asoc/topic/maintainers' into asoc-next Merge remote-tracking branch 'asoc/topic/max98088' into asoc-next Merge remote-tracking branch 'asoc/topic/max98095' into asoc-next Merge remote-tracking branch 'asoc/topic/pxa' into asoc-next Merge remote-tracking branch 'asoc/topic/s6000' into asoc-next Merge remote-tracking branch 'asoc/topic/tlv320aic31xx' into asoc-next Stephen Warren (1): ASoC: Intel: don't select RT5640 if !I2C Takashi Iwai (4): ASoC: lm4857: Use SOC_ENUM_SINGLE_EXT_DECL() ASoC: max98088: Use SOC_*_ENUM_SINGLE_DECL() ASoC: max98095: Use SOC_ENUM_SINGLE_DECL() ASoC: lm49453: Use SOC_ENUM_SINGLE_DECL() Thomas Petazzoni (2): ASoC: kirkwood: enable Kirkwood driver for mvebu platforms sound: ASoC: add ASoC board driver for Armada 370 DB Xiubo Li (6): ASoC: io: Clean up snd_soc_codec_set_cache_io() ASoC: core: Set the default I/O up try regmap. ASoC: sgtl5000: Simplify ASoC probe code ASoC: codec: Simplify ASoC probe code. ASoC: io: New signature for snd_soc_codec_set_cache_io() ASoC: core: Fix check before setting default I/O up try regmap xiangxiao (2): ASoC: add data field into snd_soc_jack_gpio ASoC: delay the initial jack detect by debounce_time .../bindings/sound/armada-370db-audio.txt | 24 + .../devicetree/bindings/sound/mvebu-audio.txt | 1 + .../devicetree/bindings/sound/tlv320aic31xx.txt | 61 + MAINTAINERS | 7 + include/dt-bindings/sound/tlv320aic31xx-micbias.h | 8 + include/sound/soc.h | 13 +- sound/soc/cirrus/snappercl15.c | 18 +- sound/soc/codecs/88pm860x-codec.c | 3 +- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ad193x.c | 10 +- sound/soc/codecs/adau1373.c | 7 - sound/soc/codecs/adav80x.c | 7 - sound/soc/codecs/ak4535.c | 9 - sound/soc/codecs/ak4641.c | 8 - sound/soc/codecs/ak4642.c | 8 - sound/soc/codecs/ak4671.c | 12 +- sound/soc/codecs/alc5623.c | 7 - sound/soc/codecs/alc5632.c | 8 - sound/soc/codecs/cq93vc.c | 3 +- sound/soc/codecs/cs4270.c | 9 - sound/soc/codecs/cs42l51.c | 6 - sound/soc/codecs/cs42l52.c | 9 +- sound/soc/codecs/cs42l73.c | 11 +- sound/soc/codecs/da7210.c | 8 - sound/soc/codecs/da7213.c | 8 - sound/soc/codecs/da732x.c | 29 +- sound/soc/codecs/da9055.c | 8 - sound/soc/codecs/isabelle.c | 16 - sound/soc/codecs/lm4857.c | 3 +- sound/soc/codecs/lm49453.c | 31 +- sound/soc/codecs/max9768.c | 5 - sound/soc/codecs/max98088.c | 45 +- sound/soc/codecs/max98090.c | 8 - sound/soc/codecs/max98095.c | 56 +- sound/soc/codecs/max9850.c | 8 - sound/soc/codecs/mc13783.c | 4 +- sound/soc/codecs/ml26124.c | 10 - sound/soc/codecs/rt5631.c | 9 - sound/soc/codecs/rt5640.c | 8 - sound/soc/codecs/sgtl5000.c | 8 - sound/soc/codecs/si476x.c | 6 +- sound/soc/codecs/sn95031.c | 2 - sound/soc/codecs/ssm2518.c | 10 - sound/soc/codecs/ssm2602.c | 7 - sound/soc/codecs/sta32x.c | 14 - sound/soc/codecs/sta529.c | 10 - sound/soc/codecs/tlv320aic23.c | 8 - sound/soc/codecs/tlv320aic26.c | 2 - sound/soc/codecs/tlv320aic31xx.c | 1295 ++++++++++++++++++++ sound/soc/codecs/tlv320aic31xx.h | 258 ++++ sound/soc/codecs/tlv320aic32x4.c | 2 - sound/soc/codecs/tlv320aic3x.c | 6 - sound/soc/codecs/tlv320dac33.c | 1 - sound/soc/codecs/wm2000.c | 2 - sound/soc/codecs/wm2200.c | 7 - sound/soc/codecs/wm5100.c | 7 - sound/soc/codecs/wm5102.c | 4 +- sound/soc/codecs/wm5110.c | 3 +- sound/soc/codecs/wm8350.c | 4 +- sound/soc/codecs/wm8400.c | 3 +- sound/soc/codecs/wm8510.c | 10 +- sound/soc/codecs/wm8523.c | 7 - sound/soc/codecs/wm8580.c | 6 - sound/soc/codecs/wm8711.c | 6 - sound/soc/codecs/wm8728.c | 11 +- sound/soc/codecs/wm8731.c | 7 - sound/soc/codecs/wm8737.c | 6 - sound/soc/codecs/wm8741.c | 6 - sound/soc/codecs/wm8750.c | 6 - sound/soc/codecs/wm8753.c | 7 - sound/soc/codecs/wm8770.c | 6 - sound/soc/codecs/wm8776.c | 6 - sound/soc/codecs/wm8804.c | 8 - sound/soc/codecs/wm8900.c | 8 +- sound/soc/codecs/wm8903.c | 10 +- sound/soc/codecs/wm8904.c | 9 - sound/soc/codecs/wm8940.c | 6 - sound/soc/codecs/wm8955.c | 8 - sound/soc/codecs/wm8960.c | 6 - sound/soc/codecs/wm8961.c | 7 - sound/soc/codecs/wm8962.c | 7 - sound/soc/codecs/wm8971.c | 6 - sound/soc/codecs/wm8974.c | 6 - sound/soc/codecs/wm8978.c | 8 +- sound/soc/codecs/wm8983.c | 6 - sound/soc/codecs/wm8985.c | 7 - sound/soc/codecs/wm8988.c | 8 - sound/soc/codecs/wm8990.c | 8 - sound/soc/codecs/wm8991.c | 8 - sound/soc/codecs/wm8993.c | 7 - sound/soc/codecs/wm8994.c | 3 +- sound/soc/codecs/wm8995.c | 7 - sound/soc/codecs/wm8996.c | 12 +- sound/soc/codecs/wm8997.c | 4 +- sound/soc/codecs/wm9081.c | 11 +- sound/soc/codecs/wm9090.c | 10 - sound/soc/intel/Kconfig | 4 +- sound/soc/kirkwood/Kconfig | 10 +- sound/soc/kirkwood/Makefile | 2 + sound/soc/kirkwood/armada-370-db.c | 120 ++ sound/soc/kirkwood/kirkwood-i2s.c | 1 + sound/soc/omap/ams-delta.c | 2 +- sound/soc/pxa/magician.c | 34 +- sound/soc/pxa/tosa.c | 35 +- sound/soc/s6000/s6105-ipcam.c | 28 +- sound/soc/soc-core.c | 14 +- sound/soc/soc-io.c | 64 +- sound/soc/soc-jack.c | 5 +- 109 files changed, 1958 insertions(+), 795 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/armada-370db-audio.txt create mode 100644 Documentation/devicetree/bindings/sound/tlv320aic31xx.txt create mode 100644 include/dt-bindings/sound/tlv320aic31xx-micbias.h create mode 100644 sound/soc/codecs/tlv320aic31xx.c create mode 100644 sound/soc/codecs/tlv320aic31xx.h create mode 100644 sound/soc/kirkwood/armada-370-db.c

4 3

[alsa-devel] [PATCH v5] ASoC: tlv320aic31xx: Add basic codec driver implementation
by Jyri Sarha 13 Mar '14

13 Mar '14

This commit adds a bare bones driver support for TLV320AIC31XX family audio codecs. The driver adds basic stereo playback trough headphone and speaker outputs and mono capture trough microphone inputs. The driver is currently missing support at least for mini DSP features and jack detection. I have tested the driver only on TLV320AIC3111, but based on the data sheets TLV320AIC3100, TLV320AIC3110, and TLV320AIC3120 should work Ok too. The base for the implementation was taken from: git@gitorious.org:ti-codecs/ti-codecs.git ajitk/topics/k3.10.1-aic31xx -branch at commit 77504eba0294764e9e63b4a0c696b44db187cd13. Signed-off-by: Jyri Sarha <jsarha(a)ti.com> --- Since v4 version: - Remove MICBIAS_OFF DT parameter - Remove logging and add missing default: to aic31xx_dapm_power_event() - Take control, widget, and route adding errors into account - Don't try soft reset in power event handler - Remove route to MICBIAS from codec driver damp routing table and add it as output pin to DT doc - Update year in copyright message and change triple newlines to double - Remove internal DAC/ADC routes that were still haunting in the previous patch - Allow PLL reprogramming on the fly as it appears to work just fine BR, Jyri .../devicetree/bindings/sound/tlv320aic31xx.txt | 61 + include/dt-bindings/sound/tlv320aic31xx-micbias.h | 8 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tlv320aic31xx.c | 1295 ++++++++++++++++++++ sound/soc/codecs/tlv320aic31xx.h | 258 ++++ 6 files changed, 1628 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/tlv320aic31xx.txt create mode 100644 include/dt-bindings/sound/tlv320aic31xx-micbias.h create mode 100644 sound/soc/codecs/tlv320aic31xx.c create mode 100644 sound/soc/codecs/tlv320aic31xx.h diff --git a/Documentation/devicetree/bindings/sound/tlv320aic31xx.txt b/Documentation/devicetree/bindings/sound/tlv320aic31xx.txt new file mode 100644 index 0000000..74c66de --- /dev/null +++ b/Documentation/devicetree/bindings/sound/tlv320aic31xx.txt @@ -0,0 +1,61 @@ +Texas Instruments - tlv320aic31xx Codec module + +The tlv320aic31xx serial control bus communicates through I2C protocols + +Required properties: + +- compatible - "string" - One of: + "ti,tlv320aic310x" - Generic TLV320AIC31xx with mono speaker amp + "ti,tlv320aic311x" - Generic TLV320AIC31xx with stereo speaker amp + "ti,tlv320aic3100" - TLV320AIC3100 (mono speaker amp, no MiniDSP) + "ti,tlv320aic3110" - TLV320AIC3110 (stereo speaker amp, no MiniDSP) + "ti,tlv320aic3120" - TLV320AIC3120 (mono speaker amp, MiniDSP) + "ti,tlv320aic3111" - TLV320AIC3111 (stereo speaker amp, MiniDSP) + +- reg - <int> - I2C slave address + + +Optional properties: + +- gpio-reset - gpio pin number used for codec reset +- ai31xx-micbias-vg - MicBias Voltage setting + 1 or MICBIAS_2_0V - MICBIAS output is powered to 2.0V + 2 or MICBIAS_2_5V - MICBIAS output is powered to 2.5V + 3 or MICBIAS_AVDD - MICBIAS output is connected to AVDD + If this node is not mentioned or if the value is unknown, then + micbias is set to 2.0V. +- HPVDD-supply, SPRVDD-supply, SPLVDD-supply, AVDD-supply, IOVDD-supply, + DVDD-supply : power supplies for the device as covered in + Documentation/devicetree/bindings/regulator/regulator.txt + +CODEC output pins: + * HPL + * HPR + * SPL, devices with stereo speaker amp + * SPR, devices with stereo speaker amp + * SPK, devices with mono speaker amp + * MICBIAS + +CODEC input pins: + * MIC1LP + * MIC1RP + * MIC1LM + +The pins can be used in referring sound node's audio-routing property. + +Example: +#include <dt-bindings/sound/tlv320aic31xx-micbias.h> + +tlv320aic31xx: tlv320aic31xx@18 { + compatible = "ti,tlv320aic311x"; + reg = <0x18>; + + ai31xx-micbias-vg = <MICBIAS_OFF>; + + HPVDD-supply = <&regulator>; + SPRVDD-supply = <&regulator>; + SPLVDD-supply = <&regulator>; + AVDD-supply = <&regulator>; + IOVDD-supply = <&regulator>; + DVDD-supply = <&regulator>; +}; diff --git a/include/dt-bindings/sound/tlv320aic31xx-micbias.h b/include/dt-bindings/sound/tlv320aic31xx-micbias.h new file mode 100644 index 0000000..f5cb772 --- /dev/null +++ b/include/dt-bindings/sound/tlv320aic31xx-micbias.h @@ -0,0 +1,8 @@ +#ifndef __DT_TLV320AIC31XX_MICBIAS_H +#define __DT_TLV320AIC31XX_MICBIAS_H + +#define MICBIAS_2_0V 1 +#define MICBIAS_2_5V 2 +#define MICBIAS_AVDDV 3 + +#endif /* __DT_TLV320AIC31XX_MICBIAS_H */ diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index e19b64f..af3c049 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -83,6 +83,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_TAS5086 if I2C select SND_SOC_TLV320AIC23 if I2C 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 @@ -428,6 +429,9 @@ config SND_SOC_TLV320AIC26 tristate depends on SPI +config SND_SOC_TLV320AIC31XX + tristate + config SND_SOC_TLV320AIC32X4 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 90ab24b..018e3f3 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -75,6 +75,7 @@ snd-soc-stac9766-objs := stac9766.o snd-soc-tas5086-objs := tas5086.o snd-soc-tlv320aic23-objs := tlv320aic23.o snd-soc-tlv320aic26-objs := tlv320aic26.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 @@ -215,6 +216,7 @@ 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_TLV320AIC26) += snd-soc-tlv320aic26.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 diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c new file mode 100644 index 0000000..e60e37b --- /dev/null +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -0,0 +1,1295 @@ +/* + * ALSA SoC TLV320AIC31XX codec driver + * + * Copyright (C) 2014 Texas Instruments, Inc. + * + * Author: Jyri Sarha <jsarha(a)ti.com> + * + * Based on ground work by: Ajit Kulkarni <x0175765(a)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, + }, +}; + +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); + +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 *tmp) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->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) +{ + struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(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: + 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); + codec->control_data = aic31xx->regmap; + + aic31xx->codec = codec; + + ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); + + if (ret != 0) { + dev_err(codec->dev, "snd_soc_codec_set_cache_io failed %d\n", + ret); + return ret; + } + + 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 */ + +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); + + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_driver_aic31xx, + aic31xx_dai_driver, + ARRAY_SIZE(aic31xx_dai_driver)); + + return ret; +} + +static int aic31xx_i2c_remove(struct i2c_client *i2c) +{ + struct aic31xx_priv *aic31xx = dev_get_drvdata(&i2c->dev); + + kfree(aic31xx); + 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 */ -- 1.7.9.5

2 2

[alsa-devel] zoom r16 quirks
by James Miller 13 Mar '14

13 Mar '14

Hi, I am trying to get the zoom R16 working with alsa for full duplex operation. I have been able to get the 8 channels of capture working after following jmancine's advice and using the following quirk on ifnum 4: { .ifnum = 4, .type = QUIRK_AUDIO_FIXED_ENDPOINT, .data = & (const struct audioformat) { .formats = SNDRV_PCM_FMTBIT_S24_LE, .channels = 8, .iface = 1, .altsetting = 1, .altset_idx = 1, .attributes = UAC_EP_CS_ATTR_SAMPLE_RATE, .rates = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000, .rate_min = 44100, .rate_max = 96000, .nr_rates = 4, .rate_table = (unsigned int[]) { 44100, 48000, 88200, 96000 } } }, However, what I discovered was that the ifnum 4 in this case is related to the udev matching rules for the device number, so when I plugged in the zoom, it was being assigned to device 4, which was then matched to the ifnum 4. If I unplugged and replugged the device, it was then designated as device 5 in the udev tree, and wasn't working. I read in quirks-table.h the following: (To work with * hotplugging, bDeviceClass must be set to USB_CLASS_PER_INTERFACE.) so I changed the matching rule macro to : USB_DEVICE_INTERFACE_CLASS(0x1686, 0x00dd, USB_CLASS_AUDIO) to attempt to match the device during udev assignment (in case the device class wasn't returned from the device properly) and then set .binterfaceclass = USB_CLASS_PER_INTERFACE below that before the quirk data. I think that I am heading in the correct direction, although I was wondering if you could confirm the following: Am I correct in assuming that the .ifnum in the quirk is related to udev device matching rules? When I write zoom-quirks.h in the way that I am currently (see below), I get a incorrect maxpacket assignment for interface 3 (midi) and I traced this to: dev_warn(ddev, "config %d interface %d altsetting %d " "bulk endpoint 0x%X has invalid maxpacket %d\n", in config.c line 259. The above bit of code is selected in conditions where the device is usb high speed but I have kern.log reporting that the device is class 1.1. Its a 24 bit chip, and jmancine reports that the device is being incorrectly assigned a 32bit le format for playback when it should be 24bit le, so perhaps this is because of a protocol mismatch? When I have the device successfully recognised the next time, I will check in format.c that it is being assigned the correct bit format, but I am uncertain as to how to confirm what bprotocol to use. Any ideas? Here is the zoom-quirks.h that I am currently using. { /* ZOOM R16 in USB 2.0 mode */ USB_DEVICE_INTERFACE_CLASS(0x1686, 0x00dd, USB_CLASS_AUDIO), .bDeviceClass = USB_CLASS_PER_INTERFACE, .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { .ifnum = QUIRK_ANY_INTERFACE, .type = QUIRK_COMPOSITE, .data = (const struct snd_usb_audio_quirk[]) { { .ifnum = 1, /*PLAYBACK*/ .type = QUIRK_AUDIO_FIXED_ENDPOINT, .data = & (const struct audioformat) { .formats = SNDRV_PCM_FMTBIT_S24_LE, .channels = 2, .iface = 1, .altsetting = 1, .altset_idx = 1, .attributes = UAC_EP_CS_ATTR_SAMPLE_RATE, .rates = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000, .endpoint = 0x03, .ep_attr = (USB_ENDPOINT_XFER_ISOC && USB_ENDPOINT_SYNC_ADAPTIVE && SND_USB_ENDPOINT_TYPE_DATA), .rate_min = 44100, .rate_max = 96000, .nr_rates = 4, .rate_table = (unsigned int[]) { 44100, 48000, 88200, 96000 } } }, { .ifnum = 2, /*CAPTURE*/ .type = QUIRK_AUDIO_FIXED_ENDPOINT, .data = & (const struct audioformat) { .formats = SNDRV_PCM_FMTBIT_S24_LE, .channels = 8, .iface = 2, .altsetting = 1, .altset_idx = 1, .attributes = UAC_EP_CS_ATTR_SAMPLE_RATE, .rates = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000, .endpoint = 0x84, .ep_attr = (USB_ENDPOINT_XFER_ISOC && USB_ENDPOINT_SYNC_SYNC && USB_ENDPOINT_USAGE_DATA), .rate_min = 44100, .rate_max = 96000, .nr_rates = 4, .rate_table = (unsigned int[]) { 44100, 48000, 88200, 96000 } } }, { .ifnum = 3, .type = QUIRK_MIDI_FIXED_ENDPOINT, }, { .ifnum = -1, }, } } }, Thanks for any help!! -- James Stewart Miller Bsc(hons) Psych.

1 0

[alsa-devel] [PATCH] ALSA: hda - Fix pop noise on Dell Inspiron 7537
by Takashi Iwai 13 Mar '14

13 Mar '14

Dell Inspiron 7537 shows loud pop noises at booting or shutdown. They can be reduced by avoiding the shutup call that is used for this codec. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=71861 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b50114176736..c31254a20331 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3868,6 +3868,7 @@ enum { ALC269_FIXUP_DELL1_MIC_NO_PRESENCE, ALC269_FIXUP_DELL2_MIC_NO_PRESENCE, ALC269_FIXUP_DELL3_MIC_NO_PRESENCE, + ALC283_FIXUP_DELL1_SHUTUP, ALC269_FIXUP_HEADSET_MODE, ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC, ALC269_FIXUP_ASUS_X101_FUNC, @@ -4087,6 +4088,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC }, + [ALC283_FIXUP_DELL1_SHUTUP] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_no_shutup, + .chained = true, + .chain_id = ALC269_FIXUP_DELL1_MIC_NO_PRESENCE, + }, [ALC269_FIXUP_HEADSET_MODE] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_headset_mode, @@ -4322,7 +4329,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x05f5, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05f6, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05f8, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), - SND_PCI_QUIRK(0x1028, 0x05f9, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05f9, "Dell Inspiron 7537", ALC283_FIXUP_DELL1_SHUTUP), SND_PCI_QUIRK(0x1028, 0x05fb, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0606, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0608, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), -- 1.9.0

1 0

[alsa-devel] [GIT PULL] ASoC updates for v3.15
by Mark Brown 13 Mar '14

13 Mar '14

The following changes since commit fa389e220254c69ffae0d403eac4146171062d08: Linux 3.14-rc6 (2014-03-09 19:41:57 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.15 for you to fetch changes up to deeed33850c8a376addabbf971df433b2a1ba74c: Merge remote-tracking branches 'asoc/topic/wm8996', 'asoc/topic/wm9081' and 'asoc/topic/wm9705' into asoc-next (2014-03-12 23:05:31 +0000) ---------------------------------------------------------------- ASoC: Updates for v3.15 Quite a busy release for ASoC this time, more on janitorial work than exciting new features but welcome nontheless: - Lots of cleanups from Takashi for enumerations; the original API for these was error prone so he's refactored lots of code to use more modern APIs which avoid issues. - Elimination of the ASoC level wrappers for I2C and SPI moving us closer to converting to regmap completely and avoiding some randconfig hassle. - Provide both manually and transparently locked DAPM APIs rather than a mix of the two fixing some concurrency issues. - Start converting CODEC drivers to use separate bus interface drivers rather than having them all in one file helping avoid dependency issues. - DPCM support for Intel Haswell and Bay Trail platforms. - Lots of work on improvements for simple-card, DaVinci and the Renesas rcar drivers. - New drivers for Analog Devices ADAU1977, TI PCM512x and parts of the CSR SiRF SoC. ---------------------------------------------------------------- Aaro Koskinen (1): ASoC: n810: fix init with DT boot Adam Thomson (1): ASoC: da9055: Add DT support for CODEC Alexander Shiyan (1): ASoC: cirrus: Remove excess dependencies on SND_SOC Bo Shen (2): ASoC: atmel_ssc_dai: make option to choose clock Binding: atmel-ssc: add option to choose clock Charles Keepax (33): ASoC: wm5102: Improve EQ coefficient controls ASoC: wm5110: Improve EQ coefficient controls ASoC: wm8997: Improve EQ coefficient controls ASoC: wm_adsp: Add debug print to note that the DSP has shutdown ASoC: wm5102: Correct typo in EQ coefficient sizes ASoC: wm5110: Correct type in EQ coefficient sizes ASoC: wm8997: Correct typo in EQ coefficient sizes ASoC: dapm: Add helpers to lock/unlock DAPM mutex ASoC: dapm: Add unlocked version of snd_soc_dapm_sync ASoC: adav80x: Update locking around use of DAPM pin API ASoC: wm5100: Update locking around use of DAPM pin API ASoC: wm8962: Update locking around use of DAPM pin API ASoC: wm8994: Update locking around use of DAPM pin API ASoC: wm8996: Update locking around use of DAPM pin API ASoC: mfld_machine: Update locking around use of DAPM pin API ASoC: ams-delta: Update locking around use of DAPM pin API ASoC: omap: n810: Update locking around use of DAPM pin API ASoC: omap: rx51: Update locking around use of DAPM pin API ASoC: pxa: corgi: Update locking around use of DAPM pin API ASoC: pxa: magician: Update locking around use of DAPM pin API ASoC: pxa: spitz: Update locking around use of DAPM pin API ASoC: pxa: tosa: Update locking around use of DAPM pin API ASoC: wm_adsp: Split firmware load into smaller chunks ASoC: wm_adsp: Correct type specifier in printf ASoC: arizona: An OUTDIV of 1 is not valid, avoid this ASoC: arizona: Add defines for FLL configuration constants ASoC: arizona: Move set of OUTDIV in to arizona_apply_fll ASoC: arizona: Move calculation of FLL configuration ASoC: arizona: Don't pass Fout into arizona_calc_fll ASoC: arizona: Calculate OUTDIV first ASoC: arizona: Calculate FLL gain last mfd: arizona: Add support for new fratio encoding ASoC: arizona: Support new fratio encoding on the wm5110 rev D Christian Engelmayer (3): ASoC: wm8991: Remove unused pointer in wm8991_probe() ASoC: wm8995: Remove unused pointer in hp_supply_event() ASoC: wm8993: Remove unused pointer in wm8993_remove() Daniel Mack (1): ASoC: cs4271: claim reset GPIO in bus probe function Denis Carikli (1): ASoC: eukrea-tlv320: Add DT support. Fabio Estevam (4): ASoC: wm8737: Use IS_ENABLED() macro ASoC: wm8983: Use IS_ENABLED() macro ASoC: fsl_esai: Check the return value from clk_prepare_enable() ASoC: fsl: imx-pcm-fiq: Remove unneeded 'out' label Jean-Francois Moine (8): ASoC: simple-card: simplify code ASoC: simple-card: simplify code ASoC: simple-card: simplify code ASoC: simple-card: simplify code ASoC: simple-card: simplify code ASoC: simple-card: simplify code ASoC: simple-card: simplify code ASoC: simple-card: simplify code Jyri Sarha (3): ASoC: davinci-evm: Add named clock reference to DT bindings ASoC: davinci-mcasp: Set BCLK divider if McASP is BCLK master ASoC: tlv320aic32x4: Sort Makefile in alphabetic order Kuninori Morimoto (58): ASoC: rsnd: tidyup register naming of BUSIF_MODE ASoC: rsnd: cleanup debug information method ASoC: rsnd: remove meaningless rsnd_ssi_non() ASoC: rsnd: control SCU ops in probe timing ASoC: rsnd: add rsnd_scu_init(), and separate init/start ASoC: rsnd: remove meaningless function parameter ASoC: rsnd: merge rsnd_scu_start/stop() and rsnd_scu_transfer_start/stop() ASoC: rsnd: rsnd_dai_is_clk_master() can be shared ASoC: rsnd: remove pin sync option ASoC: rsnd: SSI_MODE0/1 settings goes to scu.c ASoC: rsnd: remove ssiu from ssi.c ASoC: rsnd: rename rsnd_scu_convert_rate_ctrl() ASoC: rsnd: explain SRC bypass mode settings in comment ASoC: rsnd: remove duplicate *priv from rsnd_dma ASoC: rsnd: non 0 is error on probe ASoC: rsnd: fixup Gen2 module naming ASoC: rsnd: don't use schedule_work() when rsnd_dma_start() ASoC: rsnd: SCU should be called before SSI ASoC: rsnd: clarify scu.c area ASoC: rsnd: Merge macros in scu.c ASoC: rsnd: rsnd_scu_hpbif_is_enable() become macro ASoC: rsnd: merge SRC clock timing/setting ASoC: rsnd: add rsnd_ssi_is_play() ASoC: rsnd: extracts Gen1/Gen2 common parts ASoC: rsnd: remove SSI dependent DMAEngine callback ASoC: rsnd: rsnd_ssi_probe() goes forwarder than rsnd_scu_probe() ASoC: rsnd: add Gen2 SRC and DMAEngine support ASoC: rsnd: use device dependency clock ASoC: rsnd: tidyup original for_each_rsnd_xxx macro ASoC: rsnd: set DIV_EN register on rsnd_adg_set_convert_clk_gen2() ASoC: rsnd: print error if there is SRC settings mismatch ASoC: simple-card: cpu_dai_name creates confusion when DT case ASoC: rsnd: move priv member settings to upper side ASoC: rsnd: move rsnd_mod_call() macro ASoC: rsnd: remove verbose function parameter ASoC: rsnd: remove verbose debug message from scu/ssi ASoC: rsnd: unify rdai naming ASoC: rsnd: tidyup RSND_SSI_xxx flags ASoC: rsnd: run rsnd_path_init() when probe() timing ASoC: rsnd: use mod array instead of list on rdai ASoC: rsnd: get ssi/scu from rsnd_dai_stream ASoC: rsnd: use devm_clk_get() instead of clk_get() ASoC: rsnd: use function pointer for each probe ASoC: rsnd: remove unused SSI_CONTROL ASoC: rsnd: modify rsnd_adg_ssi_ws_timing_gen2() parameter ASoC: rsnd: share reg_field and reduce memory ASoC: rsnd: add struct rsnd_dai_platform_info ASoC: rsnd: Get correct SCU ID ASoC: rsnd: add rsnd_scu_enable_ssi_irq() ASoC: rsnd: call rsnd_scu_ssi_mode_init() from SSI ASoC: rsnd: add probe/remove callback on rsnd_mod_ops ASoC: rsnd: use mod probe method on SCU ASoC: rsnd: use mod probe method on SSI ASoC: rsnd: nothing to do on rsnd_dai_remove() ASoC: rsnd: remove all rsnd_xxx_remove() ASoC: rsnd: rename scu to src ASoC: simple-card: tidyup cpu/codec dai_fmt settings for non-DT ASoC: simple-card: card name can be option Lars-Peter Clausen (58): ASoC: h1940_uda1380: Don't modify runtime->hw ASoC: rx1950_uda1380: Don't modify runtime->hw ASoC: cs4271: Remove outdated comment ASoC: cs42l73: Don't mix SNDRV_PCM_RATE_KNOT with specific rates ASoC: cs42l73: Constify rate constraints ASoC: ssm2602: Constify rate constraints ASoC: twl6040: Constify rate constraints ASoC: wm8741: Constify rate constraints ASoC: wm8988: Constify rate constraints ASoC: dapm: Consistently use unsigned int for register values ASoC: ad193x: Split SPI and I2C code into different modules ASoC: ssm2602: Split SPI and I2C code into different modules ASoC: adav80x: Use devm_kzalloc() ASoC: adav80x: Split SPI and I2C code into different modules ASoC: Add ADAU1977 CODEC driver ASoC: pxa: Remove superfluous locking ASoC: da732x: Use da732x->regmap instead of codec->control_data ASoC: wm8996: Replace codec->control_data with wm8996->regmap ASoC: wm8962: Replace codec->control_data with wm8962->regmap ASoC: Remove unused 'reg2' field from soc_enum struct ASoC: Consolidate enum and value enum controls ASoC: dapm: Consolidate MUXs and value MUXs ASoC: Add macros for defining virtual enums ASoC: adau1373: Use SOC_ENUM_SINGLE_VIRT_DECL() ASoC: max98090: Use SOC_ENUM_SINGLE_VIRT_DECL() ASoC: mc13783: Use SOC_ENUM_SINGLE_VIRT_DECL() ASoC: wm8994: Use SOC_ENUM_SINGLE_VIRT_DECL() ASoC: wm8995: Use SOC_ENUM_SINGLE_VIRT_DECL() ASoC: dapm: Consolidate MUXs and virtual MUXs ASoC: dapm: Break dapm_set_path_status() appart ASoC: neo1973_wm8753: Post gta01 support removal cleanup ASoC: neo1973_wm8753: Convert to table based setup ASoC: pxa: Pass correct DAPM context to {corgi,poodle,spitz}_ext_control ASoC: pxa: e740_wm9705: Convert to table based DAPM setup ASoC: pxa: e750_wm9705: Convert to table based DAPM setup ASoC: pxa: e800_wm9712: Convert to table based DAPM setup ASoC: pxa: mioa701_wm9713: Convert to table based DAPM setup ASoC: pxa: zylonite: Convert to table based DAPM setup ASoC: wm1133-ev1: Convert to table based DAPM setup ASoC: wm8753: Remove superfluous 'codec->cache_sync = 1' ASoC: wm{5102, 5110, 8997}: Replace codec->control_data with arizona->regmap ASoC: ak4104: Remove superfluous codec->control_data initialization ASoC: Handle ignore_pmdown_time for CODEC to CODEC links ASoC: Add helper functions for PCM runtime 'active' management ASoC: Fix active count tracking for CODEC to CODEC links ASoC: Add helper function to check whether a CODEC is active ASoC: Add component pointer to the DAI struct ASoC: Move active count from CODEC to component ASoC: Move ignore_pmdown_time from CODEC to component ASoC: sam9g20_wm8731: Convert to table based DAPM setup ASoC: si476x: Fix IO setup ASoC: 88pm860: Fix IO setup ASoC: Consolidate snd_soc_register_dai() and snd_soc_register_dais() ASoC: Pass CODEC to snd_soc_register_dais() ASoC: Add a per component dai list ASoC: migor: Convert to table based DAPM setup ASoC: spear: spdif_out: Fix mute control ASoC: Fix use after free Mark Brown (60): Merge branch 'fix/davinci' of git://git.kernel.org/.../broonie/sound into HEAD Merge branch 'topic/sign' of git://git.kernel.org/.../broonie/sound into asoc-tlv320aic32x4 ASoC: wm8962: Hold a runtime PM reference while handling interrupts ASoC: wm8962: Check if we runtime resume the device when starting FLL ASoC: wm8962: Clean up error handling for failed FLL start ASoC: wm8962: Reinitialise the IRQ completion rather than just trying it ASoC: ak4554: Add to SND_SOC_ALL_CODECS ASoC: pcm1681: Convert to params_width() ASoC: pcm1792a: Convert to params_width() ASoC: codecs: Make OF supported CODECs visible in Kconfig ASoC: pcm512x: Add PCM512x driver ASoC: pcm512x: More constification ASoC: codecs: Put the CODEC drivers in a menu ASoC: wm8770: Depend on SPI only ASoC: pcm512x: Implement paging support ASoC: pcm512x: Implement analogue volume control Merge branch 'topic/of' of git://git.kernel.org/.../broonie/sound into asoc-simple ASoC: tobermory: Stop the FLL if we fail to switch SYSCLK to it ASoC: cs42l51: Don't log if we fail to allocate memory ASoC: cs42l51: Convert to direct regmap API usage ASoC: ak4671: Convert to table based control init ASoC: ak4671: Convert to direct regmap API usage ASoC: pcm512x: Add regmap select Merge remote-tracking branch 'asoc/topic/adav80x' into asoc-adau1977 ALSA: Export snd_pcm_constraint_mask64() Merge branch 'fix/dapm' of git://git.kernel.org/.../broonie/sound into asoc-dapm ASoC: alc5623: Convert to direct regmap API usage Merge branch 'topic/of' of git://git.kernel.org/.../broonie/sound into asoc-simple ASoC: da732x: Remove leftover cache size setting Merge branches 'topic/dapm', 'topic/sign', 'topic/adau1373', 'topic/max98090', 'topic/mc13783', 'fix/wm8994' and 'topic/wm8995' of git://git.kernel.org/.../broonie/sound into asoc-enum ASoC: fsl: Don't select FIQ from Eukrea Merge branch 'topic/pcm' of git://git.kernel.org/.../broonie/sound into asoc-enum Merge tag 'parse-val' of git://git.kernel.org/.../broonie/regmap into asoc-core Merge branch 'topic/tlv320aic23' of git://git.kernel.org/.../broonie/sound into asoc-samsung Merge branch 'topic/tlv320aic23' of git://git.kernel.org/.../broonie/sound into asoc-fsl ASoC: pcm512x: Split out bus drivers ASoC: pcm512x: Fix duplicate const warning Merge remote-tracking branch 'asoc/fix/pcm' into asoc-linus Merge remote-tracking branches 'asoc/fix/88pm860', 'asoc/fix/omap' and 'asoc/fix/si476x' into asoc-linus Merge branch 'topic/pcm' of git://git.kernel.org/.../broonie/sound into asoc-core ASoC: dapm: Staticise dapm_mark_dirty() ASoC: dapm: Sprinkle lockdep asserts through the code ASoC: tlv320aic23: Remove spurious bits per word setting Merge tag 'asoc-v3.14-rc6' into asoc-linus Merge remote-tracking branch 'asoc/fix/spear' into asoc-linus Merge remote-tracking branch 'asoc/topic/core' into asoc-next Merge remote-tracking branch 'asoc/topic/dapm' into asoc-next Merge remote-tracking branch 'asoc/topic/enum' into asoc-next Merge remote-tracking branch 'asoc/topic/pcm1681' into asoc-next Merge remote-tracking branch 'asoc/topic/pcm1792a' into asoc-next Merge remote-tracking branch 'asoc/topic/pcm512x' into asoc-next Merge remote-tracking branches 'asoc/topic/88pm860x', 'asoc/topic/ad1836', 'asoc/topic/ad193x', 'asoc/topic/adau1977', 'asoc/topic/ak4104', 'asoc/topic/ak4641', 'asoc/topic/ak4671' and 'asoc/topic/alc5623' into asoc-next Merge remote-tracking branches 'asoc/topic/arizona', 'asoc/topic/atmel', 'asoc/topic/cirrus', 'asoc/topic/cs4271', 'asoc/topic/cs42l51', 'asoc/topic/cs42l52', 'asoc/topic/cs42l73' and 'asoc/topic/da7210' into asoc-next Merge remote-tracking branches 'asoc/topic/da7213', 'asoc/topic/da732x', 'asoc/topic/da9055', 'asoc/topic/davinci', 'asoc/topic/fsl', 'asoc/topic/fsl-esai', 'asoc/topic/fsl-sai' and 'asoc/topic/fsl-spdif' into asoc-next Merge remote-tracking branches 'asoc/topic/ml26124', 'asoc/topic/of', 'asoc/topic/omap', 'asoc/topic/pxa' and 'asoc/topic/rcar' into asoc-next Merge remote-tracking branches 'asoc/topic/samsung', 'asoc/topic/sgtl5000', 'asoc/topic/sh', 'asoc/topic/simple', 'asoc/topic/sirf', 'asoc/topic/sn95031', 'asoc/topic/ssm2602' and 'asoc/topic/stac9766' into asoc-next Merge remote-tracking branches 'asoc/topic/tlv320aic23', 'asoc/topic/tlv320aic26', 'asoc/topic/tlv320aic32x4', 'asoc/topic/tlv320aic3x', 'asoc/topic/tlv320dac33', 'asoc/topic/twl4030', 'asoc/topic/twl6040', 'asoc/topic/uda1380', 'asoc/topic/wl1273' and 'asoc/topic/wm2200' into asoc-next Merge remote-tracking branches 'asoc/topic/wm5100', 'asoc/topic/wm8523', 'asoc/topic/wm8731', 'asoc/topic/wm8737', 'asoc/topic/wm8741', 'asoc/topic/wm8753', 'asoc/topic/wm8903', 'asoc/topic/wm8904', 'asoc/topic/wm8940' and 'asoc/topic/wm8955' into asoc-next Merge remote-tracking branches 'asoc/topic/wm8961', 'asoc/topic/wm8962', 'asoc/topic/wm8974', 'asoc/topic/wm8983', 'asoc/topic/wm8988', 'asoc/topic/wm8990', 'asoc/topic/wm8991', 'asoc/topic/wm8993' and 'asoc/topic/wm8994' into asoc-next Merge remote-tracking branches 'asoc/topic/wm8996', 'asoc/topic/wm9081' and 'asoc/topic/wm9705' into asoc-next Markus Pargmann (9): ASoC: core: Add signed register volume control logic ASoC: core: Add SOC_DOUBLE_R_S_TLV ASoC: tlv320aic32x4: Use gpio_is_valid ASoC: tlv320aic32x4: Use signed int mixer controls ASoC: tlv320aic32x4: DT support ASoC: tlv320aic32x4: Support for master clock ASoC: tlv320aic32x4: Support for regulators ASoC: tlv320aic32x4: Rearrange clock tree shutdown ASoC: tlv320aic3x: Remove tlv320aic32x4 from compatibles of tlv320aic3x Max Filippov (3): ASoC: tlv320aic23: add support for SPI control mode ASoC: update Kconfig of AIC23 users to select I2C variant ASoC: tlv320aic23: add missing EXPORT_SYMBOLs Nenghua Cao (3): ASoC: core: fix coccinelle warnings regmap: add regmap_parse_val api ASoC: core: use regmap's parse_val to do endian translation Patrick Lai (1): ASoC: pcm: free path list before exiting from error conditions Paul Bolle (1): ASoC: smdk_wm8994: Fix typo "wm8894" Peter Ujfalusi (6): ASoC: davinci-mcasp: Return value handling cleanup for mcasp_common_hw_param() ASoC: davinci-mcasp: Move pm callbacks from platform device to soc_dai_driver ASoC: davinci-mcasp: Code cleanup in davinci_mcasp_hw_params() ASoC: davinci-mcasp: Rename platform data struct ASoC: davinci-mcasp: Remove long lines from suspend/resume callbacks ASoC: davinci-mcasp: Rename the pointer to davinci dma_params as dma_params Rongjun Ying (3): ASoC: sirf: Add SiRF internal audio codec driver ASoC: sirf: Add SiRF audio port driver is used by SiRF internal audio codec ASoC: sirf: Add SiRF audio card Sachin Kamat (6): ASoC: samsung: h1940_uda1380: Remove unused variable ASoC: samsung: h1940_uda1380: Fix build warning ASoC: samsung: rx1950_uda1380: Remove unused variable ASoC: cs42l51: Remove unused variable ASoC: samsung: Remove invalid file reference ASoC: s3c24xx: Remove invalid file reference Takashi Iwai (72): ASoC: da732x: Remove superfluous DA732X_SOC_ENUM_DOUBLE_R() ASoC: ssm2602: Omit superfluous elements in input select array ASoC: wm8990: Fix the wrong number of enum items ASoC: wm8991: Fix the wrong number of enum items ASoC: ad193x: Use SOC_ENUM_SINGLE_DECL() ASoC: Rename soc_enum.max field with items ASoC: twl4030: Clean up duplicated code ALSA: da732x: Remove superfluous DA732X_SOC_ENUM_DOUBLE_R() ASoC: Add const to SOC_ENUM_*_DECL() macros ASoC: adau1373: Remove superfluous const ASoC: lm49453: Remove superfluous const ASoC: mc13783: Remove superfluous const ASoC: rt5631: Remove superfluous const ASoC: rt5640: Remove superfluous const ASoC: ssm2518: Remove superfluous const ASoC: sta529: Remove superfluous const ASoC: wm8804: Remove superfluous const ASoC: wm8978: Remove superfluous const ASoC: wm8983: Remove superfluous const ASoC: wm8985: Remove superfluous const ASoC: wm8995: Remove superfluous const ASoC: arizona: Fix wrong number of items in enum ctls ASoC: adau1373: Use SOC_ENUM_SINGLE_DECL() ASoC: alc5623: Use SOC_ENUM_SINGLE_DECL() ASoC: cs42l52: Use SOC_ENUM_SINGLE_DECL() ASoC: da7210: Use SOC_ENUM_SINGLE_DECL() ASoC: da7213: Use SOC_ENUM_SINGLE_DECL() ASoC: da9055: Use SOC_ENUM_SINGLE_DECL() ASoC: max98090: Use SOC_ENUM_SINGLE_DECL() ASoC: mc13783: Use SOC_ENUM_SINGLE_DECL() ASoC: sgtl5000: Use SOC_ENUM_SINGLE_DECL() ASoC: sn95031: Use SOC_ENUM_SINGLE_DECL() ASoC: stac9766: Use SOC_ENUM_{SINGLE|DOUBLE}_DECL() ASoC: tlv320aic26: Use SOC_ENUM_SINGLE_DECL() ASoC: twl4030: Use SOC_ENUM_SINGLE_DECL() ASoC: twl6040: Use SOC_ENUM_SINGLE_DECL() ASoC: uda1380: Use SOC_ENUM_SINGLE_DECL() ASoC: wl1273: Use SOC_ENUM_SINGLE_EXT_DECL() ASoC: wm2200: Use SOC_ENUM_SINGLE_DECL() ASoC: wm8523: Use SOC_ENUM_SINGLE_DECL() ASoC: wm8731: Use SOC_ENUM_SINGLE_DECL() ASoC: wm8737: Use SOC_ENUM_SINGLE_DECL() ASoC: wm8903: Use SOC_ENUM_SINGLE_DECL() ASoC: wm8904: Use SOC_ENUM_SINGLE_DECL() ASoC: wm8940: Use SOC_ENUM_SINGLE_DECL() ASoC: wm8961: Use SOC_ENUM_SINGLE_DECL() ASoC: wm8962: Use SOC_ENUM_SINGLE_DECL() ASoC: wm8974: Use SOC_ENUM_SINGLE_DECL() ASoC: wm8993: Use SOC_ENUM_SINGLE_DECL() ASoC: wm8995: Use SOC_ENUM_SINGLE_DECL() ASoC: wm8996: Use SOC_ENUM_SINGLE_DECL() ASoC: wm_hubs: Use SOC_ENUM_SINGLE_DECL() ASoC: omap: Use SOC_ENUM_SINGLE_EXT_DECL() ASoC: 88pm860x: Use SOC_ENUM_SINGLE_DECL() ASoC: ak4641: Use SOC_ENUM_SINGLE_DECL() ASoC: ak4671: Use SOC_ENUM_SINGLE_DECL() ASoC: alc5632: Use SOC_ENUM_SINGLE_DECL() ASoC: wm2200: Use SOC_ENUM_SINGLE_*_DECL() ASoC: da732x: Use SOC_ENUM_SINGLE_DECL() ASoC: ml26124: Use SOC_ENUM_SINGLE_DECL() ASoC: tlv320aic23: Use SOC_ENUM_SINGLE_DECL() ASoC: wm5100: Use SOC_ENUM_SINGLE_DECL() ASoC: wm8955: Use SOC_ENUM_SINGLE_DECL() ASoC: wm8988: Use SOC_ENUM_SINGLE_DECL() ASoC: wm9081: Use SOC_ENUM_SINGLE_DECL() ASoC: wm9705: Use SOC_ENUM_SINGLE_DECL() ASoC: ad1836: Use SOC_ENUM_SINGLE_DECL() ASoC: cs42l73: Use SOC_ENUM_SINGLE_DECL() ASoC: tlv320dac33: Use SOC_ENUM_SINGLE_*_DECL() ASoC: wm5102: Use ARRAY_SIZE() for SOC_VALUE_ENUM_SINGLE() ASoC: wm8997: Use ARRAY_SIZE() for SOC_VALUE_ENUM_SINGLE() ASoC: cs4271: Fix build error without CONFIG_SPI_MASTER Thomas Petazzoni (1): ASoC: cs42l51: add Device Tree binding to cs42l51 Xiang Xiao (2): ASoC: dapm: Power off all widgets in the snd_soc_dapm_shutdown ASoC: dapm: Reorder the bias update sequence Xiubo Li (19): ASoC: core: set_tdm_slot() will return -ENOTSUPP if no operation provided ASoC: fsl: use snd_soc_dai_init_dma_data() ASoC: simple-card: fix __asoc_simple_card_dai_init ASoC: simple-card: simplify the daifmt code ASoC: simple-card: Add snd_card's name parsing from DT node support ASoC: fsl-sai: convert to use regmap API for Freeacale SAI ASoC: add snd_soc_of_parse_audio_simple_widgets for DT ASoC: binding: add widgets.txt ASoC: simple-card: add off-codec widgets supports. ASoC: simple-card: for new properties documenting and usage ASoC: fsl: make sure the regmap-mmio is actually enabled ASoC: fsl-spdif: big-endian support ASoC: fsl-esai: big-endian support ASoC: binding: add tdm-slot.txt ASoC: core: add TDM slot parsing from DT supports ASoC: simple-card: add slot information parsing supports ASoC: fsl-sai: fix Freescale SAI DAI format setting. ASoC: fsl-sai: Add SND_SOC_DAIFMT_DSP_A/B support. ASoC: cache: Do the codec->reg_cache zero pionter check .../devicetree/bindings/i2c/trivial-devices.txt | 1 + .../devicetree/bindings/misc/atmel-ssc.txt | 8 + Documentation/devicetree/bindings/sound/da9055.txt | 22 + .../bindings/sound/davinci-evm-audio.txt | 9 +- .../devicetree/bindings/sound/eukrea-tlv320.txt | 21 + .../devicetree/bindings/sound/fsl,esai.txt | 5 + .../devicetree/bindings/sound/fsl,spdif.txt | 5 + .../devicetree/bindings/sound/pcm512x.txt | 30 + .../devicetree/bindings/sound/simple-card.txt | 17 +- .../devicetree/bindings/sound/sirf-audio-codec.txt | 17 + .../devicetree/bindings/sound/sirf-audio-port.txt | 20 + .../devicetree/bindings/sound/sirf-audio.txt | 41 + .../devicetree/bindings/sound/tdm-slot.txt | 20 + .../devicetree/bindings/sound/tlv320aic32x4.txt | 30 + .../devicetree/bindings/sound/tlv320aic3x.txt | 1 - .../devicetree/bindings/sound/widgets.txt | 20 + drivers/base/regmap/regmap.c | 12 + drivers/misc/atmel-ssc.c | 6 + include/linux/atmel-ssc.h | 1 + include/linux/mfd/arizona/registers.h | 6 +- include/linux/platform_data/adau1977.h | 45 + include/linux/platform_data/asoc-s3c.h | 3 +- include/linux/platform_data/asoc-s3c24xx_simtec.h | 3 +- include/linux/platform_data/davinci_asp.h | 4 +- include/linux/regmap.h | 9 + include/sound/rcar_snd.h | 36 +- include/sound/simple_card.h | 6 +- include/sound/soc-dai.h | 3 + include/sound/soc-dapm.h | 40 +- include/sound/soc.h | 126 +- include/trace/events/hswadsp.h | 384 +++++ include/trace/events/intel-sst.h | 148 ++ sound/core/pcm_lib.c | 1 + sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/atmel/Kconfig | 2 +- sound/soc/atmel/atmel_ssc_dai.c | 13 +- sound/soc/atmel/sam9g20_wm8731.c | 20 +- sound/soc/blackfin/Kconfig | 9 +- sound/soc/cirrus/Kconfig | 4 +- sound/soc/codecs/88pm860x-codec.c | 119 +- sound/soc/codecs/Kconfig | 183 +- sound/soc/codecs/Makefile | 33 +- sound/soc/codecs/ad1836.c | 4 +- sound/soc/codecs/ad193x-i2c.c | 54 + sound/soc/codecs/ad193x-spi.c | 48 + sound/soc/codecs/ad193x.c | 144 +- sound/soc/codecs/ad193x.h | 7 + sound/soc/codecs/adau1373.c | 32 +- sound/soc/codecs/adau1977-i2c.c | 59 + sound/soc/codecs/adau1977-spi.c | 76 + sound/soc/codecs/adau1977.c | 1018 +++++++++++ sound/soc/codecs/adau1977.h | 37 + sound/soc/codecs/adav801.c | 53 + sound/soc/codecs/adav803.c | 50 + sound/soc/codecs/adav80x.c | 152 +- sound/soc/codecs/adav80x.h | 7 + sound/soc/codecs/ak4104.c | 2 - sound/soc/codecs/ak4641.c | 16 +- sound/soc/codecs/ak4671.c | 240 ++- sound/soc/codecs/ak4671.h | 2 - sound/soc/codecs/alc5623.c | 117 +- sound/soc/codecs/alc5632.c | 40 +- sound/soc/codecs/arizona.c | 325 ++-- sound/soc/codecs/cs4271.c | 63 +- sound/soc/codecs/cs42l51.c | 86 +- sound/soc/codecs/cs42l52.c | 92 +- sound/soc/codecs/cs42l73.c | 55 +- sound/soc/codecs/da7210.c | 20 +- sound/soc/codecs/da7213.c | 151 +- sound/soc/codecs/da732x.c | 167 +- sound/soc/codecs/da732x.h | 3 - sound/soc/codecs/da9055.c | 92 +- sound/soc/codecs/lm49453.c | 16 +- sound/soc/codecs/max98088.c | 2 +- sound/soc/codecs/max98090.c | 172 +- sound/soc/codecs/max98095.c | 4 +- sound/soc/codecs/mc13783.c | 20 +- sound/soc/codecs/ml26124.c | 12 +- sound/soc/codecs/pcm1681.c | 15 +- sound/soc/codecs/pcm1792a.c | 33 +- sound/soc/codecs/pcm512x-i2c.c | 71 + sound/soc/codecs/pcm512x-spi.c | 69 + sound/soc/codecs/pcm512x.c | 589 +++++++ sound/soc/codecs/pcm512x.h | 171 ++ sound/soc/codecs/rt5631.c | 75 +- sound/soc/codecs/rt5640.c | 75 +- sound/soc/codecs/sgtl5000.c | 10 +- sound/soc/codecs/si476x.c | 2 +- sound/soc/codecs/sirf-audio-codec.c | 533 ++++++ sound/soc/codecs/sirf-audio-codec.h | 75 + sound/soc/codecs/sn95031.c | 44 +- sound/soc/codecs/ssm2518.c | 14 +- sound/soc/codecs/ssm2602-i2c.c | 57 + sound/soc/codecs/ssm2602-spi.c | 41 + sound/soc/codecs/ssm2602.c | 173 +- sound/soc/codecs/ssm2602.h | 14 + sound/soc/codecs/sta529.c | 2 +- sound/soc/codecs/stac9766.c | 38 +- sound/soc/codecs/tlv320aic23-i2c.c | 59 + sound/soc/codecs/tlv320aic23-spi.c | 56 + sound/soc/codecs/tlv320aic23.c | 71 +- sound/soc/codecs/tlv320aic23.h | 6 + sound/soc/codecs/tlv320aic26.c | 5 +- sound/soc/codecs/tlv320aic32x4.c | 232 ++- sound/soc/codecs/tlv320dac33.c | 34 +- sound/soc/codecs/twl4030.c | 101 +- sound/soc/codecs/twl6040.c | 17 +- sound/soc/codecs/uda1380.c | 40 +- sound/soc/codecs/wl1273.c | 9 +- sound/soc/codecs/wm2200.c | 18 +- sound/soc/codecs/wm5100.c | 36 +- sound/soc/codecs/wm5102.c | 28 +- sound/soc/codecs/wm5110.c | 19 +- sound/soc/codecs/wm8523.c | 4 +- sound/soc/codecs/wm8711.c | 2 +- sound/soc/codecs/wm8731.c | 4 +- sound/soc/codecs/wm8737.c | 50 +- sound/soc/codecs/wm8741.c | 34 +- sound/soc/codecs/wm8753.c | 5 +- sound/soc/codecs/wm8804.c | 2 +- sound/soc/codecs/wm8903.c | 108 +- sound/soc/codecs/wm8904.c | 77 +- sound/soc/codecs/wm8940.c | 20 +- sound/soc/codecs/wm8955.c | 11 +- sound/soc/codecs/wm8958-dsp2.c | 8 +- sound/soc/codecs/wm8961.c | 16 +- sound/soc/codecs/wm8962.c | 80 +- sound/soc/codecs/wm8974.c | 4 +- sound/soc/codecs/wm8978.c | 30 +- sound/soc/codecs/wm8983.c | 45 +- sound/soc/codecs/wm8985.c | 39 +- sound/soc/codecs/wm8988.c | 62 +- sound/soc/codecs/wm8990.c | 41 +- sound/soc/codecs/wm8991.c | 44 +- sound/soc/codecs/wm8993.c | 66 +- sound/soc/codecs/wm8994.c | 52 +- sound/soc/codecs/wm8995.c | 43 +- sound/soc/codecs/wm8996.c | 75 +- sound/soc/codecs/wm8997.c | 25 +- sound/soc/codecs/wm9081.c | 23 +- sound/soc/codecs/wm9705.c | 12 +- sound/soc/codecs/wm_adsp.c | 50 +- sound/soc/codecs/wm_hubs.c | 16 +- sound/soc/davinci/davinci-evm.c | 58 +- sound/soc/davinci/davinci-mcasp.c | 196 ++- sound/soc/fsl/Kconfig | 12 +- sound/soc/fsl/eukrea-tlv320.c | 108 +- sound/soc/fsl/fsl_esai.c | 32 +- sound/soc/fsl/fsl_sai.c | 332 ++-- sound/soc/fsl/fsl_sai.h | 48 +- sound/soc/fsl/fsl_spdif.c | 9 +- sound/soc/fsl/imx-pcm-fiq.c | 7 +- sound/soc/fsl/wm1133-ev1.c | 11 +- sound/soc/generic/simple-card.c | 245 +-- sound/soc/intel/Kconfig | 42 +- sound/soc/intel/Makefile | 27 +- sound/soc/intel/byt-rt5640.c | 187 ++ sound/soc/intel/haswell.c | 235 +++ sound/soc/intel/mfld_machine.c | 65 +- sound/soc/intel/sst-acpi.c | 284 ++++ sound/soc/intel/sst-baytrail-dsp.c | 372 ++++ sound/soc/intel/sst-baytrail-ipc.c | 867 ++++++++++ sound/soc/intel/sst-baytrail-ipc.h | 69 + sound/soc/intel/sst-baytrail-pcm.c | 422 +++++ sound/soc/intel/sst-dsp-priv.h | 309 ++++ sound/soc/intel/sst-dsp.c | 385 +++++ sound/soc/intel/sst-dsp.h | 233 +++ sound/soc/intel/sst-firmware.c | 587 +++++++ sound/soc/intel/sst-haswell-dsp.c | 517 ++++++ sound/soc/intel/sst-haswell-ipc.c | 1785 ++++++++++++++++++++ sound/soc/intel/sst-haswell-ipc.h | 488 ++++++ sound/soc/intel/sst-haswell-pcm.c | 872 ++++++++++ sound/soc/intel/{sst_dsp.h => sst-mfld-dsp.h} | 8 +- .../intel/{sst_platform.c => sst-mfld-platform.c} | 8 +- .../intel/{sst_platform.h => sst-mfld-platform.h} | 4 +- sound/soc/omap/Kconfig | 4 +- sound/soc/omap/ams-delta.c | 55 +- sound/soc/omap/n810.c | 26 +- sound/soc/omap/rx51.c | 22 +- sound/soc/pxa/corgi.c | 49 +- sound/soc/pxa/e740_wm9705.c | 10 +- sound/soc/pxa/e750_wm9705.c | 10 +- sound/soc/pxa/e800_wm9712.c | 19 +- sound/soc/pxa/magician.c | 26 +- sound/soc/pxa/mioa701_wm9713.c | 19 +- sound/soc/pxa/poodle.c | 7 +- sound/soc/pxa/spitz.c | 58 +- sound/soc/pxa/tosa.c | 32 +- sound/soc/pxa/zylonite.c | 17 +- sound/soc/samsung/Kconfig | 2 +- sound/soc/samsung/h1940_uda1380.c | 7 +- sound/soc/samsung/neo1973_wm8753.c | 168 +- sound/soc/samsung/rx1950_uda1380.c | 5 - sound/soc/samsung/smdk_wm8994.c | 2 +- sound/soc/samsung/tobermory.c | 2 + sound/soc/sh/migor.c | 19 +- sound/soc/sh/rcar/Makefile | 2 +- sound/soc/sh/rcar/adg.c | 228 ++- sound/soc/sh/rcar/core.c | 306 ++-- sound/soc/sh/rcar/gen.c | 98 +- sound/soc/sh/rcar/rsnd.h | 195 ++- sound/soc/sh/rcar/scu.c | 384 ----- sound/soc/sh/rcar/src.c | 687 ++++++++ sound/soc/sh/rcar/ssi.c | 332 ++-- sound/soc/sirf/Kconfig | 14 + sound/soc/sirf/Makefile | 5 + sound/soc/sirf/sirf-audio-port.c | 194 +++ sound/soc/sirf/sirf-audio-port.h | 62 + sound/soc/sirf/sirf-audio.c | 156 ++ sound/soc/soc-cache.c | 13 +- sound/soc/soc-compress.c | 65 +- sound/soc/soc-core.c | 582 ++++--- sound/soc/soc-dapm.c | 465 ++--- sound/soc/soc-pcm.c | 112 +- sound/soc/spear/spdif_out.c | 10 +- sound/soc/tegra/Kconfig | 2 +- 217 files changed, 17528 insertions(+), 4570 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/da9055.txt create mode 100644 Documentation/devicetree/bindings/sound/eukrea-tlv320.txt create mode 100644 Documentation/devicetree/bindings/sound/pcm512x.txt create mode 100644 Documentation/devicetree/bindings/sound/sirf-audio-codec.txt create mode 100644 Documentation/devicetree/bindings/sound/sirf-audio-port.txt create mode 100644 Documentation/devicetree/bindings/sound/sirf-audio.txt create mode 100644 Documentation/devicetree/bindings/sound/tdm-slot.txt create mode 100644 Documentation/devicetree/bindings/sound/tlv320aic32x4.txt create mode 100644 Documentation/devicetree/bindings/sound/widgets.txt create mode 100644 include/linux/platform_data/adau1977.h create mode 100644 include/trace/events/hswadsp.h create mode 100644 include/trace/events/intel-sst.h create mode 100644 sound/soc/codecs/ad193x-i2c.c create mode 100644 sound/soc/codecs/ad193x-spi.c create mode 100644 sound/soc/codecs/adau1977-i2c.c create mode 100644 sound/soc/codecs/adau1977-spi.c create mode 100644 sound/soc/codecs/adau1977.c create mode 100644 sound/soc/codecs/adau1977.h create mode 100644 sound/soc/codecs/adav801.c create mode 100644 sound/soc/codecs/adav803.c create mode 100644 sound/soc/codecs/pcm512x-i2c.c create mode 100644 sound/soc/codecs/pcm512x-spi.c create mode 100644 sound/soc/codecs/pcm512x.c create mode 100644 sound/soc/codecs/pcm512x.h create mode 100644 sound/soc/codecs/sirf-audio-codec.c create mode 100644 sound/soc/codecs/sirf-audio-codec.h create mode 100644 sound/soc/codecs/ssm2602-i2c.c create mode 100644 sound/soc/codecs/ssm2602-spi.c create mode 100644 sound/soc/codecs/tlv320aic23-i2c.c create mode 100644 sound/soc/codecs/tlv320aic23-spi.c create mode 100644 sound/soc/intel/byt-rt5640.c create mode 100644 sound/soc/intel/haswell.c create mode 100644 sound/soc/intel/sst-acpi.c create mode 100644 sound/soc/intel/sst-baytrail-dsp.c create mode 100644 sound/soc/intel/sst-baytrail-ipc.c create mode 100644 sound/soc/intel/sst-baytrail-ipc.h create mode 100644 sound/soc/intel/sst-baytrail-pcm.c create mode 100644 sound/soc/intel/sst-dsp-priv.h create mode 100644 sound/soc/intel/sst-dsp.c create mode 100644 sound/soc/intel/sst-dsp.h create mode 100644 sound/soc/intel/sst-firmware.c create mode 100644 sound/soc/intel/sst-haswell-dsp.c create mode 100644 sound/soc/intel/sst-haswell-ipc.c create mode 100644 sound/soc/intel/sst-haswell-ipc.h create mode 100644 sound/soc/intel/sst-haswell-pcm.c rename sound/soc/intel/{sst_dsp.h => sst-mfld-dsp.h} (96%) rename sound/soc/intel/{sst_platform.c => sst-mfld-platform.c} (99%) rename sound/soc/intel/{sst_platform.h => sst-mfld-platform.h} (97%) delete mode 100644 sound/soc/sh/rcar/scu.c create mode 100644 sound/soc/sh/rcar/src.c create mode 100644 sound/soc/sirf/Kconfig create mode 100644 sound/soc/sirf/Makefile create mode 100644 sound/soc/sirf/sirf-audio-port.c create mode 100644 sound/soc/sirf/sirf-audio-port.h create mode 100644 sound/soc/sirf/sirf-audio.c

3 3

[alsa-devel] [PATCH] ASoC: Intel: don't select RT5640 if !I2C
by Stephen Warren 13 Mar '14

13 Mar '14

From: Stephen Warren <swarren(a)nvidia.com> The rt5640 driver won't compile without I2C enabled. Hence, the Intel Haswell and Baytrail+RT5640 ASoC drivers must also depend on I2C, since these select RT5640. This solves: sound/soc/codecs/rt5640.c:2220:1: warning: data definition has no type or storage class [enabled by default] sound/soc/codecs/rt5640.c:2220:1: error: type defaults to ‘int’ in declaration of ‘module_i2c_driver’ [-Werror=implicit-int] sound/soc/codecs/rt5640.c:2220:1: warning: parameter names (without types) in function declaration [enabled by default] sound/soc/codecs/rt5640.c:2210:26: warning: ‘rt5640_i2c_driver’ defined but not used [-Wunused-variable] Reported-by: Jim Davis <jim.epost(a)gmail.com> Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- sound/soc/intel/Kconfig | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 4577b69fcf2c..3c81b3891209 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -32,7 +32,7 @@ config SND_SOC_INTEL_BAYTRAIL config SND_SOC_INTEL_HASWELL_MACH tristate "ASoC Audio DSP support for Intel Haswell Lynxpoint" - depends on SND_SOC_INTEL_SST && X86_INTEL_LPSS + depends on SND_SOC_INTEL_SST && X86_INTEL_LPSS && I2C select SND_SOC_INTEL_HASWELL select SND_SOC_RT5640 help @@ -43,7 +43,7 @@ config SND_SOC_INTEL_HASWELL_MACH 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 + depends on SND_SOC_INTEL_SST && X86_INTEL_LPSS && I2C select SND_SOC_INTEL_BAYTRAIL select SND_SOC_RT5640 help -- 1.8.1.5

2 1

[alsa-devel] [PATCH] ASoC: tlv320aic23: Remove spurious bits per word setting
by Mark Brown 12 Mar '14

12 Mar '14

From: Mark Brown <broonie(a)linaro.org> regmap should handle any byte ordering issues required, it is looking for a byte stream from the bus, so don't set 16 bits per word. This is likely to have tested out OK due to use of an unmerged SPI controller driver. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/tlv320aic23-spi.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/tlv320aic23-spi.c b/sound/soc/codecs/tlv320aic23-spi.c index 585aea436c6a..3b387e41d75d 100644 --- a/sound/soc/codecs/tlv320aic23-spi.c +++ b/sound/soc/codecs/tlv320aic23-spi.c @@ -25,7 +25,6 @@ static int aic23_spi_probe(struct spi_device *spi) dev_dbg(&spi->dev, "probing tlv320aic23 spi device\n"); - spi->bits_per_word = 16; spi->mode = SPI_MODE_0; ret = spi_setup(spi); if (ret < 0) -- 1.9.0

2 1

[alsa-devel] [PATCH] ASoC: wm_adsp: Split firmware load into smaller chunks
by Charles Keepax 12 Mar '14

12 Mar '14

The firmware files can be quite large and allocating the whole firmware a single DMA safe buffer can be problematic if the system is under a high memory load. Ease the requirements slightly by writing the firmware out in page sized chunks. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 50 ++++++++++++++++++++++++++++--------------- 1 files changed, 32 insertions(+), 18 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index f9fd564..937af6f 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 %d 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; } } -- 1.7.2.5

2 3

[alsa-devel] snd_pcm_drain causes repeated playback of last buffer contents
by Sanjeev Chugh 12 Mar '14

12 Mar '14

Hello, I'm not an alsa expert at all but trying to learn alsa. I'm experiencing an issue where on snd_pcm_drain() the slave PCM driven via plugin DSHARE is not filled with silence. Result is endless playback of buffer content until pcm is closed. I'm trying to analyze what could be the reason for this. In my understanding in Alsa, audio data is played as a unit of period. Once playing one period is finished, then next period will be played. If period has lesser frames than the period size, audio data previously played will be re-played until pcm is closed out. Could it be a reason for this continuous playback when snd_pcm_drain is called ? I'm not even sure if it is desired behavior that silence be padded after driver drains the pcm :( I guess it is too big as a bug to be ignored. Thanks Pthread

1 0

[alsa-devel] [PATCH] ASoC: Fix use after free
by Lars-Peter Clausen 12 Mar '14

12 Mar '14

Freeing the current list element while iterating over the list will cause a use after free since the iterator function will still use the current element to look up the next. Use list_for_each_safe() and remove the element from the list before freeing it to avoid this. Fixes: 1438c2f60b ("ASoC: Add a per component dai list") Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/soc-core.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index f67cef4..dac616d 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3928,11 +3928,12 @@ static inline char *fmt_multiple_name(struct device *dev, */ static void snd_soc_unregister_dais(struct snd_soc_component *component) { - struct snd_soc_dai *dai; + struct snd_soc_dai *dai, *_dai; - list_for_each_entry(dai, &component->dai_list, list) { + 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); } -- 1.8.0

2 1

[alsa-devel] [PATCH v5-resend 0/7] ASoC: add CSR SiRFSoC sound drivers
by RongJun Ying 12 Mar '14

12 Mar '14

From: Rongjun Ying <rongjun.ying(a)csr.com> This patchset adds CSR SiRFSoC sound drivers including: 1. The SiRF internal audio codec driver 2. The SiRF audio port driver 3. The SiRF internal audio card driver 4. The SiRF I2S controller driver 5. The SiRF hdmi card driver 6. The SiRF USP controller(DSP mode) driver 7. The SiRF bt-sco card driver -v5: 1. Drop all inlines. 2. Reordering the Kconfig and Makefile 3. Remove the sirf_audio_codec_reg_bits struct, use the new controls instead it. 4. Add some SND_SOC_DAPM_OUT_DRV instead of HP and SPK enable driver 5. Add audio codec clock supply instead of adc event callback 6. Fixed playback and capture can't concurrent work bug. Rongjun Ying (7): ASoC: sirf: Add SiRF internal audio codec driver SoC: sirf: Add SiRF audio port driver is used by SiRF internal audio codec ASoC: sirf: Add SiRF audio card ASoC: sirf: Add SiRF I2S driver ASoC: sirf: Add hdmi card ASoC: sirf: Add usp driver which is used by dsp mode SoC: sirf: Add bt-sco card .../devicetree/bindings/sound/sirf-audio-codec.txt | 17 + .../devicetree/bindings/sound/sirf-audio-port.txt | 20 + .../devicetree/bindings/sound/sirf-audio.txt | 41 ++ .../devicetree/bindings/sound/sirf-bt-sco.txt | 18 + .../devicetree/bindings/sound/sirf-hdmi.txt | 15 + .../devicetree/bindings/sound/sirf-i2s.txt | 27 + .../devicetree/bindings/sound/sirf-usp.txt | 27 + sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 1 + sound/soc/codecs/sirf-audio-codec.c | 533 ++++++++++++++++++++ sound/soc/codecs/sirf-audio-codec.h | 75 +++ sound/soc/sirf/Kconfig | 34 ++ sound/soc/sirf/Makefile | 13 + sound/soc/sirf/sirf-audio-port.c | 194 +++++++ sound/soc/sirf/sirf-audio-port.h | 62 +++ sound/soc/sirf/sirf-audio.c | 156 ++++++ sound/soc/sirf/sirf-bt-sco.c | 99 ++++ sound/soc/sirf/sirf-hdmi.c | 98 ++++ sound/soc/sirf/sirf-i2s.c | 434 ++++++++++++++++ sound/soc/sirf/sirf-i2s.h | 88 ++++ sound/soc/sirf/sirf-usp.c | 448 ++++++++++++++++ sound/soc/sirf/sirf-usp.h | 282 +++++++++++ 24 files changed, 2689 insertions(+), 0 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/sirf-audio-codec.txt create mode 100644 Documentation/devicetree/bindings/sound/sirf-audio-port.txt create mode 100644 Documentation/devicetree/bindings/sound/sirf-audio.txt create mode 100644 Documentation/devicetree/bindings/sound/sirf-bt-sco.txt create mode 100644 Documentation/devicetree/bindings/sound/sirf-hdmi.txt create mode 100644 Documentation/devicetree/bindings/sound/sirf-i2s.txt create mode 100644 Documentation/devicetree/bindings/sound/sirf-usp.txt create mode 100644 sound/soc/codecs/sirf-audio-codec.c create mode 100644 sound/soc/codecs/sirf-audio-codec.h create mode 100644 sound/soc/sirf/Kconfig create mode 100644 sound/soc/sirf/Makefile create mode 100644 sound/soc/sirf/sirf-audio-port.c create mode 100644 sound/soc/sirf/sirf-audio-port.h create mode 100644 sound/soc/sirf/sirf-audio.c create mode 100644 sound/soc/sirf/sirf-bt-sco.c create mode 100644 sound/soc/sirf/sirf-hdmi.c create mode 100644 sound/soc/sirf/sirf-i2s.c create mode 100644 sound/soc/sirf/sirf-i2s.h create mode 100644 sound/soc/sirf/sirf-usp.c create mode 100644 sound/soc/sirf/sirf-usp.h -- 1.7.5.4

2 19

[alsa-devel] HDA jack retask for Debian 7.4 NOT working
by Subodh Chiwate 12 Mar '14

12 Mar '14

Hi, I am trying to get surround 7.1 on debian 7.4 Wheezy. As guided by hda-jack-retask readme I built it from source on my system. But the changes are not reflecting and no errors by utility as well. I am getting Alsa 1.0.25 from the repository. Do I need to build the latest version from source to get hdajackretask as part of alsa-tools ? Any other suggestions to get the 7.1 working are welcome. Thanks in advance. -- Regards Subodh

2 2

[alsa-devel] [PATCH] ASoC: simple-card: overwrite DAIFMT_MASTER of cpu_dai->fmt
by Nicolin Chen 12 Mar '14

12 Mar '14

The current simple-card driver separates the daimft for cpu_dai and codec_dai. So we might get different values for them (0x4003 and 0x1003 for example): asoc-simple-card sound-cs42888.12: cpu : 2024000.esai / 4003 / 132000000 asoc-simple-card sound-cs42888.12: codec : cs42888 / 1003 / 24576000 asoc-simple-card sound-cs42888.12: cs42888 <-> 2024000.esai mapping ok It's pretty fair to do it for DAIFMT_INV since two dais might have differnt definitions in their drivers even though it'd be better to keep it same if we could. But for DAIFMT_MASTER bit, we should keep them identical. Otherwise, it'll make one of dai work in an incorrect mode. Thus this patch fixes it by overwriting the DAIFMT_MASTER bit of cpu_dai->fmt with the one of codec_dai->fmt since we defined DAIFMT_MASTER basing on CODEC at the first place. Signed-off-by: Nicolin Chen <Guangyu.Chen(a)freescale.com> --- sound/soc/generic/simple-card.c | 18 ++++++++++++------ 1 file changed, 12 insertions(+), 6 deletions(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 5dd4769..f9a3c8a 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -157,6 +157,8 @@ static int asoc_simple_card_parse_of(struct device_node *node, 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; int ret; @@ -189,7 +191,7 @@ static int asoc_simple_card_parse_of(struct device_node *node, np = of_get_child_by_name(node, "simple-audio-card,cpu"); if (np) ret = asoc_simple_card_sub_parse_of(np, priv->daifmt, - &priv->cpu_dai, + cpu_dai, &dai_link->cpu_of_node, &dai_link->cpu_dai_name); if (ret < 0) @@ -200,12 +202,16 @@ static int asoc_simple_card_parse_of(struct device_node *node, np = of_get_child_by_name(node, "simple-audio-card,codec"); if (np) ret = asoc_simple_card_sub_parse_of(np, priv->daifmt, - &priv->codec_dai, + codec_dai, &dai_link->codec_of_node, &dai_link->codec_dai_name); if (ret < 0) return ret; + /* overwrite DAIFMT_MASTER of cpu_dai since it's based on CODEC */ + cpu_dai->fmt &= ~SND_SOC_DAIFMT_MASTER_MASK; + cpu_dai->fmt |= codec_dai->fmt & SND_SOC_DAIFMT_MASTER_MASK; + if (!dai_link->cpu_dai_name || !dai_link->codec_dai_name) return -EINVAL; @@ -227,12 +233,12 @@ static int asoc_simple_card_parse_of(struct device_node *node, dev_dbg(dev, "platform : %04x\n", priv->daifmt); dev_dbg(dev, "cpu : %s / %04x / %d\n", dai_link->cpu_dai_name, - priv->cpu_dai.fmt, - priv->cpu_dai.sysclk); + cpu_dai->fmt, + cpu_dai->sysclk); dev_dbg(dev, "codec : %s / %04x / %d\n", dai_link->codec_dai_name, - priv->codec_dai.fmt, - priv->codec_dai.sysclk); + codec_dai->fmt, + codec_dai->sysclk); /* * soc_bind_dai_link() will check cpu name -- 1.8.4

3 6

[alsa-devel] [PATCHv3 resend 0/3] Simplify the CODEC ASoC probe code.
by Xiubo Li 12 Mar '14

12 Mar '14

3 5

[alsa-devel] Creating single-card driver for multi-codec (TLV320AIC34) board - is it possible at the machine level?
by Matt Kleffner 12 Mar '14

12 Mar '14

Hi ALSA developers, I'm developing an ALSA driver for a new board with two TLV320AIC34 chips, which to a driver is the equivalent of 4 TLV320AIC31/2/3 (TLV320AIC3X driver) codecs. The platform is OMAP4, with the OMAP as I2C master. TDM mode with one McBSP port is used, and it is desired that the four codecs appear as one 8-channel codec (clocks are shared and data buses are connected together). It is OK if there are 4 separate mixers, one for each codec. However, when starting/stopping audio all codecs should act together and audio data presented to the ALSA user interface just as with a single 8-channel sound card. My questions have two parts: one on binding together the codecs into a "single card", and another on alterations to support an external sample-clock generator. BONDING CODECS TOGETHER It is desired to write the driver at the "machine" level (ignore the external-clock modification for now), without modifying the ALSA SoC core or the TLV320AIC3X driver. Is this possible? I've searched the internet and looked through the source code, and I have the following gaps in understanding: The following TI link seems to suggest such TLV320AIC34 operation may already be possible with ALSA: http://e2e.ti.com/support/data_converters/audio_converters/w/design_notes/2… However, one of the threads pointed in that link: http://comments.gmane.org/gmane.linux.alsa.devel/84899 suggests such support is not in the ALSA SoC core or the TLV320AIC3X driver. While it is mentioned that if configured appropriately, "you'll end up with a single card", later it is mentioned that "you're probably trying to break new ground here. You're trying to bind all the CODECs together and treat them as one digital interface, currently the code does a series of point to point links multiplexed on a bus only." What does point-to-point link mean in this context? What does "you'll end up with a single card" mean if all 8 channels are not exposed in one card? If it is new ground, then what does the rx51 driver do (it is often referenced as a starting point for this type of binding effort), which interfaces to an AIC34 codec? The threads referenced are a few years old. Is it now possible to write such a driver at the machine level, or must I additionally either modify the core or write a new codec driver (as the original poster did)? ALTERATIONS TO SUPPORT AN EXTERNAL SAMPLE-CLOCK GENERATOR The sample clock is being generated by a single, I2C-configurable clock chip. So any sample-rate configuration must be sent to the clock chip. Can this be handled purely in the machine layer, or do I need to modify the AIC3X driver? I understand I need to use the regmap API to do such things, and ALSA uses this API, but I do not yet know much about it. Any pointers to documentation on doing this besides the source code are appreciated. A key newbie question on regmap (and Linux I2C) as needed by the driver the following: Does the regmap-i2c or i2c driver "own" the I2C bus so that the each device on the bus can be read from/written to via the regmap-i2c or i2c driver without the device's driver having to request-take-release the bus on each access? Thanks, - Matt

1 0

[alsa-devel] [PATCH v2] ASoC: cs42888: Add codec driver support
by Nicolin Chen 12 Mar '14

12 Mar '14

This patch adds support for the Cirrus Logic CS42888 Audio CODEC that has four 24-bit A/D and eight 24-bit D/A converters. [ CS42888 supports both I2C and SPI control ports. As initial patch, this patch only adds the support for I2C. ] Signed-off-by: Nicolin Chen <Guangyu.Chen(a)freescale.com> --- Changelog v2: * Separated driver into I2C and main part drivers. * Added error out if failed to get clock. * Enabled Auto-mute function as default. * Disabled runtime-pm in remove(). * Added return value checking for regcache_sync(). * Fixed some trival parts like 'const' as Lars mentioned. .../devicetree/bindings/sound/cs42888.txt | 27 ++ sound/soc/codecs/Kconfig | 10 + sound/soc/codecs/Makefile | 4 + sound/soc/codecs/cs42888-i2c.c | 68 +++ sound/soc/codecs/cs42888.c | 536 +++++++++++++++++++++ sound/soc/codecs/cs42888.h | 213 ++++++++ 6 files changed, 858 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/cs42888.txt create mode 100644 sound/soc/codecs/cs42888-i2c.c create mode 100644 sound/soc/codecs/cs42888.c create mode 100644 sound/soc/codecs/cs42888.h diff --git a/Documentation/devicetree/bindings/sound/cs42888.txt b/Documentation/devicetree/bindings/sound/cs42888.txt new file mode 100644 index 0000000..7736527 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cs42888.txt @@ -0,0 +1,27 @@ +CS42888 audio CODEC + +Required properties: + + - compatible : "cirrus,cs42888" + + - reg : the I2C address of the device for I2C + + - clocks : phandle to the clock source for MCLK + + - clock-names : must contain "mclk". + + - VA-supply, VD-supply, VLS-supply, VLC-supply: power supplies for the device, + as covered in Documentation/devicetree/bindings/regulator/regulator.txt + +Example: + +codec: cs42888@48 { + compatible = "cirrus,cs42888"; + reg = <0x48>; + clocks = <&codec_mclk 0>; + clock-names = "mclk"; + VA-supply = <&reg_audio>; + VD-supply = <&reg_audio>; + VLS-supply = <&reg_audio>; + VLC-supply = <&reg_audio>; +}; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 32d7a6f..1642578 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -44,6 +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_CS42888_I2C if I2C select SND_SOC_CX20442 if TTY select SND_SOC_DA7210 if I2C select SND_SOC_DA7213 if I2C @@ -303,6 +304,15 @@ config SND_SOC_CS4271 tristate "Cirrus Logic CS4271 CODEC" depends on SND_SOC_I2C_AND_SPI +config SND_SOC_CS42888 + tristate + +config SND_SOC_CS42888_I2C + tristate "Cirrus Logic CS42888 CODEC (I2C)" + depends on I2C + select SND_SOC_CS42888 + select REGMAP_I2C + config SND_SOC_CX20442 tristate depends on TTY diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index cb46c4c..f5612fb 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -30,6 +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-cs42888-objs := cs42888.o +snd-soc-cs42888-i2c-objs := cs42888-i2c.o snd-soc-cx20442-objs := cx20442.o snd-soc-da7210-objs := da7210.o snd-soc-da7213-objs := da7213.o @@ -178,6 +180,8 @@ 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-$(CONFIG_SND_SOC_CS42888) += snd-soc-cs42888.o +obj-$(CONFIG_SND_SOC_CS42888_I2C) += snd-soc-cs42888-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 diff --git a/sound/soc/codecs/cs42888-i2c.c b/sound/soc/codecs/cs42888-i2c.c new file mode 100644 index 0000000..54813bc --- /dev/null +++ b/sound/soc/codecs/cs42888-i2c.c @@ -0,0 +1,68 @@ +/* + * Cirrus Logic CS42888 Audio CODEC Digital Audio Interface (DAI) I2C driver + * + * Copyright (C) 2014 Freescale Semiconductor, Inc. + * + * Author: Nicolin Chen <Guangyu.Chen(a)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/i2c.h> +#include <linux/module.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 "cs42888.h" + +static int cs42888_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + u32 ret = cs42888_probe(&i2c->dev, + devm_regmap_init_i2c(i2c, &cs42888_regmap_config)); + if (ret) + return ret; + + pm_runtime_enable(&i2c->dev); + pm_request_idle(&i2c->dev); + + return 0; +} + +static int cs42888_i2c_remove(struct i2c_client *i2c) +{ + snd_soc_unregister_codec(&i2c->dev); + pm_runtime_disable(&i2c->dev); + + return 0; +} + +static struct i2c_device_id cs42888_i2c_id[] = { + {"cs42888", 0}, + {} +}; +MODULE_DEVICE_TABLE(i2c, cs42888_i2c_id); + +static struct i2c_driver cs42888_i2c_driver = { + .driver = { + .name = "cs42888", + .owner = THIS_MODULE, + .pm = &cs42888_pm, + }, + .probe = cs42888_i2c_probe, + .remove = cs42888_i2c_remove, + .id_table = cs42888_i2c_id, +}; + +module_i2c_driver(cs42888_i2c_driver); + +MODULE_DESCRIPTION("Cirrus Logic CS42888 ALSA SoC Codec I2C Driver"); +MODULE_AUTHOR("Freescale Semiconductor, Inc."); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs42888.c b/sound/soc/codecs/cs42888.c new file mode 100644 index 0000000..8cd3d4a --- /dev/null +++ b/sound/soc/codecs/cs42888.c @@ -0,0 +1,536 @@ +/* + * Cirrus Logic CS42888 Audio CODEC Digital Audio Interface (DAI) driver + * + * Copyright (C) 2014 Freescale Semiconductor, Inc. + * + * Author: Nicolin Chen <Guangyu.Chen(a)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/pm_runtime.h> +#include <linux/regulator/consumer.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tlv.h> + +#include "cs42888.h" + +#define CS42888_NUM_SUPPLIES 4 +static const char *const cs42888_supply_names[CS42888_NUM_SUPPLIES] = { + "VA", + "VD", + "VLS", + "VLC", +}; + +#define CS42888_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 cs42888_priv { + struct regulator_bulk_data supplies[CS42888_NUM_SUPPLIES]; + 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 cs42888_adc_single[] = { "Differential", "Single-Ended" }; +static const char *const cs42888_szc[] = { "Immediate Change", "Zero Cross", + "Soft Ramp", "Soft Ramp on Zero Cross" }; + +static const struct soc_enum adc1_single_enum = + SOC_ENUM_SINGLE(CS42888_ADCCTL, 4, 2, cs42888_adc_single); +static const struct soc_enum adc2_single_enum = + SOC_ENUM_SINGLE(CS42888_ADCCTL, 3, 2, cs42888_adc_single); +static const struct soc_enum dac_szc_enum = + SOC_ENUM_SINGLE(CS42888_TXCTL, 5, 4, cs42888_szc); +static const struct soc_enum adc_szc_enum = + SOC_ENUM_SINGLE(CS42888_TXCTL, 0, 4, cs42888_szc); + +static const struct snd_kcontrol_new cs42888_snd_controls[] = { + SOC_DOUBLE_R_TLV("DAC1 Playback Volume", CS42888_VOLAOUT1, + CS42888_VOLAOUT2, 0, 0xff, 1, dac_tlv), + SOC_DOUBLE_R_TLV("DAC2 Playback Volume", CS42888_VOLAOUT3, + CS42888_VOLAOUT4, 0, 0xff, 1, dac_tlv), + SOC_DOUBLE_R_TLV("DAC3 Playback Volume", CS42888_VOLAOUT5, + CS42888_VOLAOUT6, 0, 0xff, 1, dac_tlv), + SOC_DOUBLE_R_TLV("DAC4 Playback Volume", CS42888_VOLAOUT7, + CS42888_VOLAOUT8, 0, 0xff, 1, dac_tlv), + SOC_DOUBLE_R_S_TLV("ADC1 Capture Volume", CS42888_VOLAIN1, + CS42888_VOLAIN2, 0, -0x80, 0x30, 7, 0, adc_tlv), + SOC_DOUBLE_R_S_TLV("ADC2 Capture Volume", CS42888_VOLAIN3, + CS42888_VOLAIN4, 0, -0x80, 0x30, 7, 0, adc_tlv), + SOC_DOUBLE("DAC1 Invert Switch", CS42888_DACINV, 0, 1, 1, 0), + SOC_DOUBLE("DAC2 Invert Switch", CS42888_DACINV, 2, 3, 1, 0), + SOC_DOUBLE("DAC3 Invert Switch", CS42888_DACINV, 4, 5, 1, 0), + SOC_DOUBLE("DAC4 Invert Switch", CS42888_DACINV, 6, 7, 1, 0), + SOC_DOUBLE("ADC1 Invert Switch", CS42888_ADCINV, 0, 1, 1, 0), + SOC_DOUBLE("ADC2 Invert Switch", CS42888_ADCINV, 2, 3, 1, 0), + SOC_SINGLE("ADC High-Pass Filter Switch", CS42888_ADCCTL, 7, 1, 1), + SOC_SINGLE("DAC De-emphasis Switch", CS42888_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", CS42888_TXCTL, 7, 1, 0), + SOC_ENUM("DAC Soft Ramp & Zero Cross Control Switch", dac_szc_enum), + SOC_SINGLE("DAC Auto Mute Switch", CS42888_TXCTL, 4, 1, 0), + SOC_SINGLE("Mute ADC Serial Port Switch", CS42888_TXCTL, 3, 1, 0), + SOC_SINGLE("ADC Single Volume Control Switch", CS42888_TXCTL, 2, 1, 0), + SOC_ENUM("ADC Soft Ramp & Zero Cross Control Switch", adc_szc_enum), +}; + +static const struct snd_soc_dapm_widget cs42888_dapm_widgets[] = { + SND_SOC_DAPM_DAC("DAC1", "Playback", CS42888_PWRCTL, 1, 1), + SND_SOC_DAPM_DAC("DAC2", "Playback", CS42888_PWRCTL, 2, 1), + SND_SOC_DAPM_DAC("DAC3", "Playback", CS42888_PWRCTL, 3, 1), + SND_SOC_DAPM_DAC("DAC4", "Playback", CS42888_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", CS42888_PWRCTL, 5, 1), + SND_SOC_DAPM_ADC("ADC2", "Capture", CS42888_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", CS42888_PWRCTL, 0, 1, NULL, 0), +}; + +static const struct snd_soc_dapm_route cs42888_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" }, +}; + +struct cs42888_ratios { + unsigned int ratio; + unsigned char speed; + unsigned char mclk; +}; + +static const struct cs42888_ratios cs42888_ratios[] = { + { 64, CS42888_FM_QUAD, CS42888_FUNCMOD_MFREQ_256(4) }, + { 96, CS42888_FM_QUAD, CS42888_FUNCMOD_MFREQ_384(4) }, + { 128, CS42888_FM_QUAD, CS42888_FUNCMOD_MFREQ_512(4) }, + { 192, CS42888_FM_QUAD, CS42888_FUNCMOD_MFREQ_768(4) }, + { 256, CS42888_FM_SINGLE, CS42888_FUNCMOD_MFREQ_256(1) }, + { 384, CS42888_FM_SINGLE, CS42888_FUNCMOD_MFREQ_384(1) }, + { 512, CS42888_FM_SINGLE, CS42888_FUNCMOD_MFREQ_512(1) }, + { 768, CS42888_FM_SINGLE, CS42888_FUNCMOD_MFREQ_768(1) }, + { 1024, CS42888_FM_SINGLE, CS42888_FUNCMOD_MFREQ_1024(1) } +}; + +static int cs42888_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 cs42888_priv *cs42888 = snd_soc_codec_get_drvdata(codec); + + cs42888->sysclk = freq; + + return 0; +} + +static int cs42888_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int format) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs42888_priv *cs42888 = 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 = CS42888_INTF_DAC_DIF_LEFTJ | CS42888_INTF_ADC_DIF_LEFTJ; + break; + case SND_SOC_DAIFMT_I2S: + val = CS42888_INTF_DAC_DIF_I2S | CS42888_INTF_ADC_DIF_I2S; + break; + case SND_SOC_DAIFMT_RIGHT_J: + val = CS42888_INTF_DAC_DIF_RIGHTJ | CS42888_INTF_ADC_DIF_RIGHTJ; + break; + default: + dev_err(codec->dev, "unsupported dai format\n"); + return -EINVAL; + } + + regmap_update_bits(cs42888->regmap, CS42888_INTF, + CS42888_INTF_DAC_DIF_MASK | + CS42888_INTF_ADC_DIF_MASK, val); + + /* Set master/slave audio interface */ + switch (format & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + cs42888->slave_mode = true; + break; + case SND_SOC_DAIFMT_CBM_CFM: + cs42888->slave_mode = false; + break; + default: + dev_err(codec->dev, "unsupported master/slave mode\n"); + return -EINVAL; + } + + return 0; +} + +static int cs42888_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 cs42888_priv *cs42888 = snd_soc_codec_get_drvdata(codec); + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + u32 ratio = cs42888->sysclk / params_rate(params); + u32 i, fm, val, mask; + + for (i = 0; i < ARRAY_SIZE(cs42888_ratios); i++) { + if (cs42888_ratios[i].ratio == ratio) + break; + } + + if (i == ARRAY_SIZE(cs42888_ratios)) { + dev_err(codec->dev, "unsupported sysclk ratio\n"); + return -EINVAL; + } + + mask = CS42888_FUNCMOD_MFREQ_MASK; + val = cs42888_ratios[i].mclk; + + fm = cs42888->slave_mode ? CS42888_FM_AUTO : cs42888_ratios[i].speed; + + regmap_update_bits(cs42888->regmap, CS42888_FUNCMOD, + CS42888_FUNCMOD_xC_FM_MASK(tx) | mask, + CS42888_FUNCMOD_xC_FM(tx, fm) | val); + + return 0; +} + +static int cs42888_digital_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + struct cs42888_priv *cs42888 = snd_soc_codec_get_drvdata(codec); + + regmap_update_bits(cs42888->regmap, CS42888_DACMUTE, + CS42888_DACMUTE_ALL, mute ? CS42888_DACMUTE_ALL : 0); + + return 0; +} + +static const struct snd_soc_dai_ops cs42888_dai_ops = { + .set_fmt = cs42888_set_dai_fmt, + .set_sysclk = cs42888_set_dai_sysclk, + .hw_params = cs42888_hw_params, + .digital_mute = cs42888_digital_mute, +}; + +static struct snd_soc_dai_driver cs42888_dai = { + .name = "cs42888", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 8, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = CS42888_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 4, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = CS42888_FORMATS, + }, + .ops = &cs42888_dai_ops, +}; + +static const struct reg_default cs42888_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 */ + { 0x17, 0x00 }, /* ADC Channel Invert */ + { 0x18, 0x00 }, /* Status Control */ + { 0x1a, 0x00 }, /* Status Mask */ + { 0x1b, 0x00 }, /* MUTEC Pin Control */ +}; + +static bool cs42888_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS42888_STATUS: + return true; + default: + return false; + } +} + +static bool cs42888_writeable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS42888_CHIPID: + case CS42888_STATUS: + return false; + default: + return true; + } +} + +const struct regmap_config cs42888_regmap_config = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = CS42888_LASTREG, + .reg_defaults = cs42888_reg, + .num_reg_defaults = ARRAY_SIZE(cs42888_reg), + .volatile_reg = cs42888_volatile_register, + .writeable_reg = cs42888_writeable_register, + .cache_type = REGCACHE_RBTREE, +}; +EXPORT_SYMBOL_GPL(cs42888_regmap_config); + +static int cs42888_codec_probe(struct snd_soc_codec *codec) +{ + struct cs42888_priv *cs42888 = snd_soc_codec_get_drvdata(codec); + + /* Mute all DAC channels */ + regmap_write(cs42888->regmap, CS42888_DACMUTE, CS42888_DACMUTE_ALL); + + return 0; +} + +static const struct snd_soc_codec_driver cs42888_driver = { + .probe = cs42888_codec_probe, + .idle_bias_off = true, + + .controls = cs42888_snd_controls, + .num_controls = ARRAY_SIZE(cs42888_snd_controls), + .dapm_widgets = cs42888_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cs42888_dapm_widgets), + .dapm_routes = cs42888_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(cs42888_dapm_routes), +}; + +int cs42888_probe(struct device *dev, struct regmap *regmap) +{ + struct cs42888_priv *cs42888; + int ret, val, i; + + cs42888 = devm_kzalloc(dev, sizeof(*cs42888), GFP_KERNEL); + if (cs42888 == NULL) + return -ENOMEM; + + dev_set_drvdata(dev, cs42888); + + cs42888->clk = devm_clk_get(dev, "mclk"); + if (IS_ERR(cs42888->clk)) { + dev_err(dev, "failed to get the clock: %ld\n", + PTR_ERR(cs42888->clk)); + return -EINVAL; + } + + cs42888->sysclk = clk_get_rate(cs42888->clk); + + for (i = 0; i < ARRAY_SIZE(cs42888->supplies); i++) + cs42888->supplies[i].supply = cs42888_supply_names[i]; + + ret = devm_regulator_bulk_get(dev, + ARRAY_SIZE(cs42888->supplies), cs42888->supplies); + if (ret) { + dev_err(dev, "failed to request supplies: %d\n", ret); + return ret; + } + + ret = regulator_bulk_enable(ARRAY_SIZE(cs42888->supplies), + cs42888->supplies); + if (ret) { + dev_err(dev, "failed to enable supplies: %d\n", ret); + return ret; + } + + /* Make sure hardware reset done */ + msleep(5); + + cs42888->regmap = regmap; + if (IS_ERR(cs42888->regmap)) { + ret = PTR_ERR(cs42888->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(cs42888->regmap, true); + + /* Validate the chip ID */ + regmap_read(cs42888->regmap, CS42888_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 & CS42888_CHIPID_CHIP_ID_MASK) != 0x00) { + dev_err(dev, "unmatched chip ID: %d\n", + val & CS42888_CHIPID_CHIP_ID_MASK); + ret = -EINVAL; + goto err_enable; + } + + dev_info(dev, "found device, revision %X\n", + val & CS42888_CHIPID_REV_ID_MASK); + + regcache_cache_bypass(cs42888->regmap, false); + + ret = snd_soc_register_codec(dev, &cs42888_driver, &cs42888_dai, 1); + if (ret) { + dev_err(dev, "failed to register codec:%d\n", ret); + goto err_enable; + } + + regcache_cache_only(cs42888->regmap, true); + +err_enable: + regulator_bulk_disable(ARRAY_SIZE(cs42888->supplies), + cs42888->supplies); + + return ret; +} +EXPORT_SYMBOL_GPL(cs42888_probe); + +#ifdef CONFIG_PM_RUNTIME +static int cs42888_runtime_resume(struct device *dev) +{ + struct cs42888_priv *cs42888 = dev_get_drvdata(dev); + int ret; + + ret = clk_prepare_enable(cs42888->clk); + if (ret) { + dev_err(dev, "failed to enable mclk: %d\n", ret); + return ret; + } + + ret = regulator_bulk_enable(ARRAY_SIZE(cs42888->supplies), + cs42888->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(cs42888->regmap, false); + + ret = regcache_sync(cs42888->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(cs42888->supplies), + cs42888->supplies); +err_clk: + clk_disable_unprepare(cs42888->clk); + + return ret; +} + +static int cs42888_runtime_suspend(struct device *dev) +{ + struct cs42888_priv *cs42888 = dev_get_drvdata(dev); + + regcache_cache_only(cs42888->regmap, true); + + regulator_bulk_disable(ARRAY_SIZE(cs42888->supplies), + cs42888->supplies); + + clk_disable_unprepare(cs42888->clk); + + return 0; +} +#endif + +const struct dev_pm_ops cs42888_pm = { + SET_RUNTIME_PM_OPS(cs42888_runtime_suspend, cs42888_runtime_resume, NULL) +}; +EXPORT_SYMBOL_GPL(cs42888_pm); + +const struct of_device_id cs42888_of_match[] = { + { .compatible = "cirrus,cs42888", }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, cs42888_of_match); +EXPORT_SYMBOL_GPL(cs42888_of_match); + +MODULE_DESCRIPTION("Cirrus Logic CS42888 ALSA SoC Codec Driver"); +MODULE_AUTHOR("Freescale Semiconductor, Inc."); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs42888.h b/sound/soc/codecs/cs42888.h new file mode 100644 index 0000000..eda55ac --- /dev/null +++ b/sound/soc/codecs/cs42888.h @@ -0,0 +1,213 @@ +/* + * cs42888.h - Cirrus Logic CS42888 Audio CODEC driver header file + * + * Copyright (C) 2014 Freescale Semiconductor, Inc. + * + * Author: Nicolin Chen <Guangyu.Chen(a)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 _CS42888_H +#define _CS42888_H + +extern const struct dev_pm_ops cs42888_pm; +extern const struct regmap_config cs42888_regmap_config; +int cs42888_probe(struct device *dev, struct regmap *regmap); + +/* CS42888 register map */ +#define CS42888_CHIPID 0x01 /* Chip ID */ +#define CS42888_PWRCTL 0x02 /* Power Control */ +#define CS42888_FUNCMOD 0x03 /* Functional Mode */ +#define CS42888_INTF 0x04 /* Interface Formats */ +#define CS42888_ADCCTL 0x05 /* ADC Control */ +#define CS42888_TXCTL 0x06 /* Transition Control */ +#define CS42888_DACMUTE 0x07 /* DAC Mute Control */ +#define CS42888_VOLAOUT1 0x08 /* Volume Control AOUT1 */ +#define CS42888_VOLAOUT2 0x09 /* Volume Control AOUT2 */ +#define CS42888_VOLAOUT3 0x0A /* Volume Control AOUT3 */ +#define CS42888_VOLAOUT4 0x0B /* Volume Control AOUT4 */ +#define CS42888_VOLAOUT5 0x0C /* Volume Control AOUT5 */ +#define CS42888_VOLAOUT6 0x0D /* Volume Control AOUT6 */ +#define CS42888_VOLAOUT7 0x0E /* Volume Control AOUT7 */ +#define CS42888_VOLAOUT8 0x0F /* Volume Control AOUT8 */ +#define CS42888_DACINV 0x10 /* DAC Channel Invert */ +#define CS42888_VOLAIN1 0x11 /* Volume Control AIN1 */ +#define CS42888_VOLAIN2 0x12 /* Volume Control AIN2 */ +#define CS42888_VOLAIN3 0x13 /* Volume Control AIN3 */ +#define CS42888_VOLAIN4 0x14 /* Volume Control AIN4 */ +#define CS42888_ADCINV 0x17 /* ADC Channel Invert */ +#define CS42888_STATUSCTL 0x18 /* Status Control */ +#define CS42888_STATUS 0x19 /* Status */ +#define CS42888_STATUSM 0x1A /* Status Mask */ +#define CS42888_MUTEC 0x1B /* MUTEC Pin Control */ + +#define CS42888_FIRSTREG CS42888_CHIPID +#define CS42888_LASTREG CS42888_MUTEC +#define CS42888_NUMREGS (CS42888_LASTREG - CS42888_FIRSTREG + 1) +#define CS42888_I2C_INCR 0x80 + +/* Chip I.D. and Revision Register (Address 01h) */ +#define CS42888_CHIPID_CHIP_ID_MASK 0xF0 +#define CS42888_CHIPID_REV_ID_MASK 0x0F + +/* Power Control (Address 02h) */ +#define CS42888_PWRCTL_PDN_ADC2_SHIFT 6 +#define CS42888_PWRCTL_PDN_ADC2_MASK (1 << CS42888_PWRCTL_PDN_ADC2_SHIFT) +#define CS42888_PWRCTL_PDN_ADC2 (1 << CS42888_PWRCTL_PDN_ADC2_SHIFT) +#define CS42888_PWRCTL_PDN_ADC1_SHIFT 5 +#define CS42888_PWRCTL_PDN_ADC1_MASK (1 << CS42888_PWRCTL_PDN_ADC1_SHIFT) +#define CS42888_PWRCTL_PDN_ADC1 (1 << CS42888_PWRCTL_PDN_ADC1_SHIFT) +#define CS42888_PWRCTL_PDN_DAC4_SHIFT 4 +#define CS42888_PWRCTL_PDN_DAC4_MASK (1 << CS42888_PWRCTL_PDN_DAC4_SHIFT) +#define CS42888_PWRCTL_PDN_DAC4 (1 << CS42888_PWRCTL_PDN_DAC4_SHIFT) +#define CS42888_PWRCTL_PDN_DAC3_SHIFT 3 +#define CS42888_PWRCTL_PDN_DAC3_MASK (1 << CS42888_PWRCTL_PDN_DAC3_SHIFT) +#define CS42888_PWRCTL_PDN_DAC3 (1 << CS42888_PWRCTL_PDN_DAC3_SHIFT) +#define CS42888_PWRCTL_PDN_DAC2_SHIFT 2 +#define CS42888_PWRCTL_PDN_DAC2_MASK (1 << CS42888_PWRCTL_PDN_DAC2_SHIFT) +#define CS42888_PWRCTL_PDN_DAC2 (1 << CS42888_PWRCTL_PDN_DAC2_SHIFT) +#define CS42888_PWRCTL_PDN_DAC1_SHIFT 1 +#define CS42888_PWRCTL_PDN_DAC1_MASK (1 << CS42888_PWRCTL_PDN_DAC1_SHIFT) +#define CS42888_PWRCTL_PDN_DAC1 (1 << CS42888_PWRCTL_PDN_DAC1_SHIFT) +#define CS42888_PWRCTL_PDN_SHIFT 0 +#define CS42888_PWRCTL_PDN_MASK (1 << CS42888_PWRCTL_PDN_SHIFT) +#define CS42888_PWRCTL_PDN (1 << CS42888_PWRCTL_PDN_SHIFT) + +/* Functional Mode (Address 03h) */ +#define CS42888_FUNCMOD_DAC_FM_SHIFT 6 +#define CS42888_FUNCMOD_DAC_FM_WIDTH 2 +#define CS42888_FUNCMOD_DAC_FM_MASK (((1 << CS42888_FUNCMOD_DAC_FM_WIDTH) - 1) << CS42888_FUNCMOD_DAC_FM_SHIFT) +#define CS42888_FUNCMOD_DAC_FM(v) ((v) << CS42888_FUNCMOD_DAC_FM_SHIFT) +#define CS42888_FUNCMOD_ADC_FM_SHIFT 4 +#define CS42888_FUNCMOD_ADC_FM_WIDTH 2 +#define CS42888_FUNCMOD_ADC_FM_MASK (((1 << CS42888_FUNCMOD_ADC_FM_WIDTH) - 1) << CS42888_FUNCMOD_ADC_FM_SHIFT) +#define CS42888_FUNCMOD_ADC_FM(v) ((v) << CS42888_FUNCMOD_ADC_FM_SHIFT) +#define CS42888_FUNCMOD_xC_FM_MASK(x) ((x) ? CS42888_FUNCMOD_DAC_FM_MASK : CS42888_FUNCMOD_ADC_FM_MASK) +#define CS42888_FUNCMOD_xC_FM(x, v) ((x) ? CS42888_FUNCMOD_DAC_FM(v) : CS42888_FUNCMOD_ADC_FM(v)) +#define CS42888_FUNCMOD_MFREQ_SHIFT 1 +#define CS42888_FUNCMOD_MFREQ_WIDTH 3 +#define CS42888_FUNCMOD_MFREQ_MASK (((1 << CS42888_FUNCMOD_MFREQ_WIDTH) - 1) << CS42888_FUNCMOD_MFREQ_SHIFT) +#define CS42888_FUNCMOD_MFREQ_256(s) ((0 << CS42888_FUNCMOD_MFREQ_SHIFT) >> (s >> 1)) +#define CS42888_FUNCMOD_MFREQ_384(s) ((1 << CS42888_FUNCMOD_MFREQ_SHIFT) >> (s >> 1)) +#define CS42888_FUNCMOD_MFREQ_512(s) ((2 << CS42888_FUNCMOD_MFREQ_SHIFT) >> (s >> 1)) +#define CS42888_FUNCMOD_MFREQ_768(s) ((3 << CS42888_FUNCMOD_MFREQ_SHIFT) >> (s >> 1)) +#define CS42888_FUNCMOD_MFREQ_1024(s) ((4 << CS42888_FUNCMOD_MFREQ_SHIFT) >> (s >> 1)) + +#define CS42888_FM_SINGLE 0 +#define CS42888_FM_DOUBLE 1 +#define CS42888_FM_QUAD 2 +#define CS42888_FM_AUTO 3 + +/* Interface Formats (Address 04h) */ +#define CS42888_INTF_FREEZE_SHIFT 7 +#define CS42888_INTF_FREEZE_MASK (1 << CS42888_INTF_FREEZE_SHIFT) +#define CS42888_INTF_FREEZE (1 << CS42888_INTF_FREEZE_SHIFT) +#define CS42888_INTF_AUX_DIF_SHIFT 6 +#define CS42888_INTF_AUX_DIF_MASK (1 << CS42888_INTF_AUX_DIF_SHIFT) +#define CS42888_INTF_AUX_DIF (1 << CS42888_INTF_AUX_DIF_SHIFT) +#define CS42888_INTF_DAC_DIF_SHIFT 3 +#define CS42888_INTF_DAC_DIF_WIDTH 3 +#define CS42888_INTF_DAC_DIF_MASK (((1 << CS42888_INTF_DAC_DIF_WIDTH) - 1) << CS42888_INTF_DAC_DIF_SHIFT) +#define CS42888_INTF_DAC_DIF_LEFTJ (0 << CS42888_INTF_DAC_DIF_SHIFT) +#define CS42888_INTF_DAC_DIF_I2S (1 << CS42888_INTF_DAC_DIF_SHIFT) +#define CS42888_INTF_DAC_DIF_RIGHTJ (2 << CS42888_INTF_DAC_DIF_SHIFT) +#define CS42888_INTF_DAC_DIF_RIGHTJ_16 (3 << CS42888_INTF_DAC_DIF_SHIFT) +#define CS42888_INTF_DAC_DIF_ONELINE_20 (4 << CS42888_INTF_DAC_DIF_SHIFT) +#define CS42888_INTF_DAC_DIF_ONELINE_24 (6 << CS42888_INTF_DAC_DIF_SHIFT) +#define CS42888_INTF_DAC_DIF_TDM (7 << CS42888_INTF_DAC_DIF_SHIFT) +#define CS42888_INTF_ADC_DIF_SHIFT 0 +#define CS42888_INTF_ADC_DIF_WIDTH 3 +#define CS42888_INTF_ADC_DIF_MASK (((1 << CS42888_INTF_ADC_DIF_WIDTH) - 1) << CS42888_INTF_ADC_DIF_SHIFT) +#define CS42888_INTF_ADC_DIF_LEFTJ (0 << CS42888_INTF_ADC_DIF_SHIFT) +#define CS42888_INTF_ADC_DIF_I2S (1 << CS42888_INTF_ADC_DIF_SHIFT) +#define CS42888_INTF_ADC_DIF_RIGHTJ (2 << CS42888_INTF_ADC_DIF_SHIFT) +#define CS42888_INTF_ADC_DIF_RIGHTJ_16 (3 << CS42888_INTF_ADC_DIF_SHIFT) +#define CS42888_INTF_ADC_DIF_ONELINE_20 (4 << CS42888_INTF_ADC_DIF_SHIFT) +#define CS42888_INTF_ADC_DIF_ONELINE_24 (6 << CS42888_INTF_ADC_DIF_SHIFT) +#define CS42888_INTF_ADC_DIF_TDM (7 << CS42888_INTF_ADC_DIF_SHIFT) + +/* ADC Control & DAC De-Emphasis (Address 05h) */ +#define CS42888_ADCCTL_ADC_HPF_FREEZE_SHIFT 7 +#define CS42888_ADCCTL_ADC_HPF_FREEZE_MASK (1 << CS42888_ADCCTL_ADC_HPF_FREEZE_SHIFT) +#define CS42888_ADCCTL_ADC_HPF_FREEZE (1 << CS42888_ADCCTL_ADC_HPF_FREEZE_SHIFT) +#define CS42888_ADCCTL_DAC_DEM_SHIFT 5 +#define CS42888_ADCCTL_DAC_DEM_MASK (1 << CS42888_ADCCTL_DAC_DEM_SHIFT) +#define CS42888_ADCCTL_DAC_DEM (1 << CS42888_ADCCTL_DAC_DEM_SHIFT) +#define CS42888_ADCCTL_ADC1_SINGLE_SHIFT 4 +#define CS42888_ADCCTL_ADC1_SINGLE_MASK (1 << CS42888_ADCCTL_ADC1_SINGLE_SHIFT) +#define CS42888_ADCCTL_ADC1_SINGLE (1 << CS42888_ADCCTL_ADC1_SINGLE_SHIFT) +#define CS42888_ADCCTL_ADC2_SINGLE_SHIFT 3 +#define CS42888_ADCCTL_ADC2_SINGLE_MASK (1 << CS42888_ADCCTL_ADC2_SINGLE_SHIFT) +#define CS42888_ADCCTL_ADC2_SINGLE (1 << CS42888_ADCCTL_ADC2_SINGLE_SHIFT) + +/* Transition Control (Address 06h) */ +#define CS42888_TXCTL_DAC_SNGVOL_SHIFT 7 +#define CS42888_TXCTL_DAC_SNGVOL_MASK (1 << CS42888_TXCTL_DAC_SNGVOL_SHIFT) +#define CS42888_TXCTL_DAC_SNGVOL (1 << CS42888_TXCTL_DAC_SNGVOL_SHIFT) +#define CS42888_TXCTL_DAC_SZC_SHIFT 5 +#define CS42888_TXCTL_DAC_SZC_WIDTH 2 +#define CS42888_TXCTL_DAC_SZC_MASK (((1 << CS42888_TXCTL_DAC_SZC_WIDTH) - 1) << CS42888_TXCTL_DAC_SZC_SHIFT) +#define CS42888_TXCTL_DAC_SZC_IC (0 << CS42888_TXCTL_DAC_SZC_SHIFT) +#define CS42888_TXCTL_DAC_SZC_ZC (1 << CS42888_TXCTL_DAC_SZC_SHIFT) +#define CS42888_TXCTL_DAC_SZC_SR (2 << CS42888_TXCTL_DAC_SZC_SHIFT) +#define CS42888_TXCTL_DAC_SZC_SRZC (3 << CS42888_TXCTL_DAC_SZC_SHIFT) +#define CS42888_TXCTL_AMUTE_SHIFT 4 +#define CS42888_TXCTL_AMUTE_MASK (1 << CS42888_TXCTL_AMUTE_SHIFT) +#define CS42888_TXCTL_AMUTE (1 << CS42888_TXCTL_AMUTE_SHIFT) +#define CS42888_TXCTL_MUTE_ADC_SP_SHIFT 3 +#define CS42888_TXCTL_MUTE_ADC_SP_MASK (1 << CS42888_TXCTL_MUTE_ADC_SP_SHIFT) +#define CS42888_TXCTL_MUTE_ADC_SP (1 << CS42888_TXCTL_MUTE_ADC_SP_SHIFT) +#define CS42888_TXCTL_ADC_SNGVOL_SHIFT 2 +#define CS42888_TXCTL_ADC_SNGVOL_MASK (1 << CS42888_TXCTL_ADC_SNGVOL_SHIFT) +#define CS42888_TXCTL_ADC_SNGVOL (1 << CS42888_TXCTL_ADC_SNGVOL_SHIFT) +#define CS42888_TXCTL_ADC_SZC_SHIFT 0 +#define CS42888_TXCTL_ADC_SZC_MASK (((1 << CS42888_TXCTL_ADC_SZC_WIDTH) - 1) << CS42888_TXCTL_ADC_SZC_SHIFT) +#define CS42888_TXCTL_ADC_SZC_IC (0 << CS42888_TXCTL_ADC_SZC_SHIFT) +#define CS42888_TXCTL_ADC_SZC_ZC (1 << CS42888_TXCTL_ADC_SZC_SHIFT) +#define CS42888_TXCTL_ADC_SZC_SR (2 << CS42888_TXCTL_ADC_SZC_SHIFT) +#define CS42888_TXCTL_ADC_SZC_SRZC (3 << CS42888_TXCTL_ADC_SZC_SHIFT) + +/* DAC Channel Mute (Address 07h) */ +#define CS42888_DACMUTE_AOUT(n) (0x1 << n) +#define CS42888_DACMUTE_ALL 0xff + +/* Status Control (Address 18h)*/ +#define CS42888_STATUSCTL_INI_SHIFT 2 +#define CS42888_STATUSCTL_INI_WIDTH 2 +#define CS42888_STATUSCTL_INI_MASK (((1 << CS42888_STATUSCTL_INI_WIDTH) - 1) << CS42888_STATUSCTL_INI_SHIFT) +#define CS42888_STATUSCTL_INT_ACTIVE_HIGH (0 << CS42888_STATUSCTL_INI_SHIFT) +#define CS42888_STATUSCTL_INT_ACTIVE_LOW (1 << CS42888_STATUSCTL_INI_SHIFT) +#define CS42888_STATUSCTL_INT_OPEN_DRAIN (2 << CS42888_STATUSCTL_INI_SHIFT) + +/* Status (Address 19h)*/ +#define CS42888_STATUS_DAC_CLK_ERR_SHIFT 4 +#define CS42888_STATUS_DAC_CLK_ERR_MASK (1 << CS42888_STATUS_DAC_CLK_ERR_SHIFT) +#define CS42888_STATUS_ADC_CLK_ERR_SHIFT 3 +#define CS42888_STATUS_ADC_CLK_ERR_MASK (1 << CS42888_STATUS_ADC_CLK_ERR_SHIFT) +#define CS42888_STATUS_ADC2_OVFL_SHIFT 1 +#define CS42888_STATUS_ADC2_OVFL_MASK (1 << CS42888_STATUS_ADC2_OVFL_SHIFT) +#define CS42888_STATUS_ADC1_OVFL_SHIFT 0 +#define CS42888_STATUS_ADC1_OVFL_MASK (1 << CS42888_STATUS_ADC1_OVFL_SHIFT) + +/* Status Mask (Address 1Ah) */ +#define CS42888_STATUS_DAC_CLK_ERR_M_SHIFT 4 +#define CS42888_STATUS_DAC_CLK_ERR_M_MASK (1 << CS42888_STATUS_DAC_CLK_ERR_M_SHIFT) +#define CS42888_STATUS_ADC_CLK_ERR_M_SHIFT 3 +#define CS42888_STATUS_ADC_CLK_ERR_M_MASK (1 << CS42888_STATUS_ADC_CLK_ERR_M_SHIFT) +#define CS42888_STATUS_ADC2_OVFL_M_SHIFT 1 +#define CS42888_STATUS_ADC2_OVFL_M_MASK (1 << CS42888_STATUS_ADC2_OVFL_M_SHIFT) +#define CS42888_STATUS_ADC1_OVFL_M_SHIFT 0 +#define CS42888_STATUS_ADC1_OVFL_M_MASK (1 << CS42888_STATUS_ADC1_OVFL_M_SHIFT) + +/* MUTEC Pin Control (Address 1Bh) */ +#define CS42888_MUTEC_MCPOLARITY_SHIFT 1 +#define CS42888_MUTEC_MCPOLARITY_MASK (1 << CS42888_MUTEC_MCPOLARITY_SHIFT) +#define CS42888_MUTEC_MCPOLARITY_ACTIVE_LOW (0 << CS42888_MUTEC_MCPOLARITY_SHIFT) +#define CS42888_MUTEC_MCPOLARITY_ACTIVE_HIGH (1 << CS42888_MUTEC_MCPOLARITY_SHIFT) +#define CS42888_MUTEC_MUTEC_ACTIVE_SHIFT 0 +#define CS42888_MUTEC_MUTEC_ACTIVE_MASK (1 << CS42888_MUTEC_MUTEC_ACTIVE_SHIFT) +#define CS42888_MUTEC_MUTEC_ACTIVE (1 << CS42888_MUTEC_MUTEC_ACTIVE_SHIFT) +#endif /* _CS42888_H */ -- 1.8.4

6 8

[alsa-devel] About snd_dmaengine_pcm_trigger()
by Kuninori Morimoto 12 Mar '14

12 Mar '14

Hi Mark, Lars and all I need your opinion Now, I'm working for ${LINUX}/sound/soc/sh/fsi and ${LINUX}/sound/soc/sh/rcar drirvers. These drivers are supporting DMAEngine transfer, but it is using original DMAEngine method, not using snd_dmaengine_pcm_trigger() or snd_dmaengine_pcm_register(). I was requested to use it from Mark. The reason why I couldn't use it was our DMAEngine didn't have "cyclic" tranfer support which is used on dmaengine_pcm_prepare_and_submit() But now, I created cyclic support on our DMA driver (on my local PC at this point) Then, I noticed our drivers still can't use snd_dmaengine_pcm_xxx() method. 1st, our device needs PIO tranfer support too. Current PIO/DMA transfer are sharing many methods/functions. Code will become difficult to read if it is separated forcibly. (to using snd_dmaengine_pcm_xxx()) 2nd, our device DMA ON/OFF timing has relation ship to other register settings. Unfortunately, our device is picky, it needs like this HW init setting -> DMA ON -> HW start setting It can be solved if snd_dmaengine_pcm_xxx() has callbacks, but many callbacks are needed... 3rd, our device needs special method on snd_pcm_ops int the future. From my point of view, our driver can replace to use "cyclic" DMA transfer instead of current original DMAEngine method. I guess, it is not difficult. But, using snd_dmaengine_pcm_xxx() is difficult. But, what do you think ?

3 10

[alsa-devel] [PATCH] ak4642: show error if register write fails
by Ben Dooks 11 Mar '14

11 Mar '14

The calls to snd_soc_update_bits() and snd_soc_write() return error codes if the calls fails. The ak4642 driver discards these errors which means that the user is not informed if there is an issue with the bus providing the register access. Signed-off-by: Ben Dooks <ben.dooks(a)codethink.co.uk> --- sound/soc/codecs/ak4642.c | 62 ++++++++++++++++++++++++++++++++++------------- 1 file changed, 45 insertions(+), 17 deletions(-) diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index 4717fa9..15795f6 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -225,6 +225,34 @@ static const struct reg_default ak4648_reg[] = { { 36, 0x00 }, { 37, 0x88 }, { 38, 0x88 }, { 39, 0x08 }, }; +/* wrapper functions to show any errors to updating register values */ + +static inline int ak4642_update_bits(struct snd_soc_codec *codec, + unsigned int reg, + unsigned int mask, unsigned int val) +{ + int ret = snd_soc_update_bits(codec, reg, mask, val); + + if (ret < 0) { + pr_info("%s: error %d writing %04x (%08x, mask %08x)\n", + codec->name, ret, reg, mask, val); + } + + return ret; +} + +static inline int ak4642_soc_write(struct snd_soc_codec *codec, + unsigned int reg, unsigned int value) +{ + int ret = snd_soc_write(codec, reg, value); + + if (ret < 0) + pr_info("%s: error %d writing %04x to %08x\n", + codec->name, ret, reg, value); + + return ret; +} + static int ak4642_dai_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { @@ -242,8 +270,8 @@ static int ak4642_dai_startup(struct snd_pcm_substream *substream, * This operation came from example code of * "ASAHI KASEI AK4642" (japanese) manual p97. */ - snd_soc_write(codec, L_IVC, 0x91); /* volume */ - snd_soc_write(codec, R_IVC, 0x91); /* volume */ + ak4642_soc_write(codec, L_IVC, 0x91); /* volume */ + ak4642_soc_write(codec, R_IVC, 0x91); /* volume */ } else { /* * start stereo input @@ -258,11 +286,11 @@ static int ak4642_dai_startup(struct snd_pcm_substream *substream, * This operation came from example code of * "ASAHI KASEI AK4642" (japanese) manual p94. */ - snd_soc_update_bits(codec, SG_SL1, PMMP | MGAIN0, PMMP | MGAIN0); - snd_soc_write(codec, TIMER, ZTM(0x3) | WTM(0x3)); - snd_soc_write(codec, ALC_CTL1, ALC | LMTH0); - snd_soc_update_bits(codec, PW_MGMT1, PMADL, PMADL); - snd_soc_update_bits(codec, PW_MGMT3, PMADR, PMADR); + ak4642_update_bits(codec, SG_SL1, PMMP | MGAIN0, PMMP | MGAIN0); + ak4642_soc_write(codec, TIMER, ZTM(0x3) | WTM(0x3)); + ak4642_soc_write(codec, ALC_CTL1, ALC | LMTH0); + ak4642_update_bits(codec, PW_MGMT1, PMADL, PMADL); + ak4642_update_bits(codec, PW_MGMT3, PMADR, PMADR); } return 0; @@ -277,9 +305,9 @@ static void ak4642_dai_shutdown(struct snd_pcm_substream *substream, if (is_play) { } else { /* stop stereo input */ - snd_soc_update_bits(codec, PW_MGMT1, PMADL, 0); - snd_soc_update_bits(codec, PW_MGMT3, PMADR, 0); - snd_soc_update_bits(codec, ALC_CTL1, ALC, 0); + ak4642_update_bits(codec, PW_MGMT1, PMADL, 0); + ak4642_update_bits(codec, PW_MGMT3, PMADR, 0); + ak4642_update_bits(codec, ALC_CTL1, ALC, 0); } } @@ -312,7 +340,7 @@ static int ak4642_dai_set_sysclk(struct snd_soc_dai *codec_dai, default: return -EINVAL; } - snd_soc_update_bits(codec, MD_CTL1, PLL_MASK, pll); + ak4642_update_bits(codec, MD_CTL1, PLL_MASK, pll); return 0; } @@ -339,8 +367,8 @@ static int ak4642_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) default: return -EINVAL; } - snd_soc_update_bits(codec, PW_MGMT2, MS | MCKO | PMPLL, data); - snd_soc_update_bits(codec, MD_CTL1, BCKO_MASK, bcko); + ak4642_update_bits(codec, PW_MGMT2, MS | MCKO | PMPLL, data); + ak4642_update_bits(codec, MD_CTL1, BCKO_MASK, bcko); /* format type */ data = 0; @@ -357,7 +385,7 @@ static int ak4642_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) default: return -EINVAL; } - snd_soc_update_bits(codec, MD_CTL1, DIF_MASK, data); + ak4642_update_bits(codec, MD_CTL1, DIF_MASK, data); return 0; } @@ -411,7 +439,7 @@ static int ak4642_dai_hw_params(struct snd_pcm_substream *substream, default: return -EINVAL; } - snd_soc_update_bits(codec, MD_CTL2, FS_MASK, rate); + ak4642_update_bits(codec, MD_CTL2, FS_MASK, rate); return 0; } @@ -421,10 +449,10 @@ static int ak4642_set_bias_level(struct snd_soc_codec *codec, { switch (level) { case SND_SOC_BIAS_OFF: - snd_soc_write(codec, PW_MGMT1, 0x00); + ak4642_soc_write(codec, PW_MGMT1, 0x00); break; default: - snd_soc_update_bits(codec, PW_MGMT1, PMVCM, PMVCM); + ak4642_update_bits(codec, PW_MGMT1, PMVCM, PMVCM); break; } codec->dapm.bias_level = level; -- 1.9.0

4 9

Re: [alsa-devel] Zoom R16
by Jason Mancine 11 Mar '14

11 Mar '14

Can't seem to compile 3.11 or 3.12 for the distro I am using (AVLInux)... tested it in Ubuntu Studio 13.10 and the quirk didn't seem to have any effect. The output of lsusb is the same, but the device doesn't show up as a sound card. On Sun, Nov 24, 2013 at 9:18 AM, Daniel Mack <daniel(a)zonque.org> wrote: > On Sun, Nov 24, 2013 at 09:03:47AM -0500, Jason Mancine wrote: > > Thanks, I will give it a try. By recent kernel do you mean latest > mainline > > or should my distro's 3.6.11 be new enough? > > Anything >= 3.11 should do. > > In further replies, please do not drop the alsa-devel mailing list. > > > Thanks, > Daniel >

3 15

[alsa-devel] [PATCH] ALSA: sis7019: Simplify dependencies
by Jean Delvare 11 Mar '14

11 Mar '14

Kconfig symbol X86_32 was introduced in October 2005, it's about time to use it :-) Signed-off-by: Jean Delvare <jdelvare(a)suse.de> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> --- sound/pci/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) --- linux-3.14-rc6.orig/sound/pci/Kconfig 2014-03-11 11:29:10.346528331 +0100 +++ linux-3.14-rc6/sound/pci/Kconfig 2014-03-11 14:23:58.996511875 +0100 @@ -797,7 +797,7 @@ config SND_RME9652 config SND_SIS7019 tristate "SiS 7019 Audio Accelerator" - depends on X86 && !X86_64 + depends on X86_32 select SND_AC97_CODEC select ZONE_DMA help -- Jean Delvare SUSE L3 Support

2 1

[alsa-devel] [PATCH] ALSA: cs5535audio: Also needed on MIPS
by Jean Delvare 11 Mar '14

11 Mar '14

The CS5536 companion chipset is not only used on 32-bit x86 systems as I originally thought, it is also used on MIPS Loongson/Lemote 2 systems. So let the snd-cs5535audio driver be built on MIPS too. Signed-off-by: Jean Delvare <jdelvare(a)suse.de> Cc: Jaya Kumar <jayakumar.alsa(a)gmail.com> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: Ralf Baechle <ralf(a)linux-mips.org> --- Takashi: the powerpc reference was irrelevant as the defconfig file I mentioned includes all possible sound drivers. Ralf: do you happen to know if any other CS55xx chip was used on MIPS systems? Or any non-x86 systems for that matter. sound/pci/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) --- linux-3.14-rc6.orig/sound/pci/Kconfig 2014-03-10 21:09:12.142069639 +0100 +++ linux-3.14-rc6/sound/pci/Kconfig 2014-03-11 11:29:10.346528331 +0100 @@ -286,7 +286,7 @@ config SND_CS5530 config SND_CS5535AUDIO tristate "CS5535/CS5536 Audio" - depends on X86_32 || COMPILE_TEST + depends on X86_32 || MIPS || COMPILE_TEST select SND_PCM select SND_AC97_CODEC help -- Jean Delvare SUSE L3 Support

2 1

[alsa-devel] [PATCH] ALSA: cs553*: Fix dependencies
by Jean Delvare 11 Mar '14

11 Mar '14

The CS5530, CS5535 and CS5536 chipsets are companions of the Geode series of processors, which are 32-bit x86 processors. So the snd-cs5530 and snd-cs5535audio drivers are only needed on this architecture, except for build testing purpose. Signed-off-by: Jean Delvare <jdelvare(a)suse.de> Cc: Jaya Kumar <jayakumar.alsa(a)gmail.com> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/Kconfig | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) --- linux-3.14-rc6.orig/sound/pci/Kconfig 2014-02-09 16:53:07.699734368 +0100 +++ linux-3.14-rc6/sound/pci/Kconfig 2014-03-10 20:52:21.908690341 +0100 @@ -276,7 +276,7 @@ config SND_CS46XX_NEW_DSP config SND_CS5530 tristate "CS5530 Audio" - depends on ISA_DMA_API + depends on ISA_DMA_API && (X86_32 || COMPILE_TEST) select SND_SB16_DSP help Say Y here to include support for audio on Cyrix/NatSemi CS5530 chips. @@ -286,6 +286,7 @@ config SND_CS5530 config SND_CS5535AUDIO tristate "CS5535/CS5536 Audio" + depends on X86_32 || COMPILE_TEST select SND_PCM select SND_AC97_CODEC help -- Jean Delvare SUSE L3 Support

2 3

[alsa-devel] [PATCH] sound: sound_core: Reduce checkpatch noise
by Joe Perches 11 Mar '14

11 Mar '14

Mostly whitespace cleanups (spaces around operators, remove trailing spaces and unnecessary blank lines, etc) and o Reduce indent of switch/case labels o Use pr_<level> for printks, add pr_fmt o Move braces to appropriate locations o Wrap arguments to 80 columns o Remove unnecessary braces Mostly done with checkpatch --fix and some typing. Signed-off-by: Joe Perches <joe(a)perches.com> --- sound/sound_core.c | 183 ++++++++++++++++++++++++----------------------------- 1 file changed, 82 insertions(+), 101 deletions(-) diff --git a/sound/sound_core.c b/sound/sound_core.c index 11e953a..952ec83 100644 --- a/sound/sound_core.c +++ b/sound/sound_core.c @@ -4,6 +4,8 @@ * is used OSS or emulation of it. */ +#define pr_fmt(fmt) KBUILD_MODNAME ": " fmt + /* * First, the common part. */ @@ -68,7 +70,7 @@ module_exit(cleanup_soundcore); #ifdef CONFIG_SOUND_OSS_CORE /* * OSS sound core handling. Breaks out sound functions to submodules - * + * * Author: Alan Cox <alan(a)lxorguk.ukuu.org.uk> * * Fixes: @@ -80,9 +82,9 @@ module_exit(cleanup_soundcore); * 2 of the License, or (at your option) any later version. * * -------------------- - * + * * Top level handler for the sound subsystem. Various devices can - * plug into this. The fact they don't all go via OSS doesn't mean + * plug into this. The fact they don't all go via OSS doesn't mean * they don't have to implement the OSS API. There is a lot of logic * to keeping much of the OSS weight out of the code in a compatibility * module, but it's up to the driver to rember to load it... @@ -111,8 +113,7 @@ module_exit(cleanup_soundcore); #define SOUND_STEP 16 -struct sound_unit -{ +struct sound_unit { int unit_minor; const struct file_operations *unit_fops; struct sound_unit *next; @@ -151,15 +152,14 @@ extern int msnd_pinnacle_init(void); #ifdef CONFIG_SOUND_OSS_CORE_PRECLAIM static int preclaim_oss = 1; #else -static int preclaim_oss = 0; +static int preclaim_oss; #endif module_param(preclaim_oss, int, 0444); static int soundcore_open(struct inode *, struct file *); -static const struct file_operations soundcore_fops = -{ +static const struct file_operations soundcore_fops = { /* We must have an owner or the module locking fails */ .owner = THIS_MODULE, .open = soundcore_open, @@ -171,72 +171,72 @@ static const struct file_operations soundcore_fops = * join into it. Called with the lock asserted */ -static int __sound_insert_unit(struct sound_unit * s, struct sound_unit **list, const struct file_operations *fops, int index, int low, int top) +static int __sound_insert_unit(struct sound_unit *s, struct sound_unit **list, + const struct file_operations *fops, int index, + int low, int top) { - int n=low; + int n = low; if (index < 0) { /* first free */ - while (*list && (*list)->unit_minor<n) - list=&((*list)->next); + while (*list && (*list)->unit_minor < n) + list = &((*list)->next); - while(n<top) - { + while (n < top) { /* Found a hole ? */ - if(*list==NULL || (*list)->unit_minor>n) + if (*list == NULL || (*list)->unit_minor > n) break; - list=&((*list)->next); - n+=SOUND_STEP; + list = &((*list)->next); + n += SOUND_STEP; } - if(n>=top) + if (n >= top) return -ENOENT; } else { - n = low+(index*16); + n = low + (index * 16); while (*list) { - if ((*list)->unit_minor==n) + if ((*list)->unit_minor == n) return -EBUSY; - if ((*list)->unit_minor>n) + if ((*list)->unit_minor > n) break; - list=&((*list)->next); + list = &((*list)->next); } - } - + } + /* * Fill it in */ - - s->unit_minor=n; - s->unit_fops=fops; - + + s->unit_minor = n; + s->unit_fops = fops; + /* * Link it */ - - s->next=*list; - *list=s; - - + + s->next = *list; + *list = s; + + return n; } /* * Remove a node from the chain. Called with the lock asserted */ - -static struct sound_unit *__sound_remove_unit(struct sound_unit **list, int unit) + +static struct sound_unit *__sound_remove_unit(struct sound_unit **list, + int unit) { - while(*list) - { - struct sound_unit *p=*list; - if(p->unit_minor==unit) - { - *list=p->next; + while (*list) { + struct sound_unit *p = *list; + if (p->unit_minor == unit) { + *list = p->next; return p; } - list=&(p->next); + list = &(p->next); } - printk(KERN_ERR "Sound device %d went missing!\n", unit); + pr_err("Sound device %d went missing!\n", unit); return NULL; } @@ -251,7 +251,10 @@ static DEFINE_SPINLOCK(sound_loader_lock); * list. Acquires locks as needed */ -static int sound_insert_unit(struct sound_unit **list, const struct file_operations *fops, int index, int low, int top, const char *name, umode_t mode, struct device *dev) +static int sound_insert_unit(struct sound_unit **list, + const struct file_operations *fops, int index, + int low, int top, const char *name, umode_t mode, + struct device *dev) { struct sound_unit *s = kmalloc(sizeof(*s), GFP_KERNEL); int r; @@ -263,7 +266,7 @@ static int sound_insert_unit(struct sound_unit **list, const struct file_operati retry: r = __sound_insert_unit(s, list, fops, index, low, top); spin_unlock(&sound_loader_lock); - + if (r < 0) goto fail; else if (r < SOUND_STEP) @@ -305,7 +308,7 @@ fail: * completed the removal before their file operations become * invalid. */ - + static void sound_remove_unit(struct sound_unit **list, int unit) { struct sound_unit *p; @@ -349,7 +352,7 @@ static struct sound_unit *chains[SOUND_STEP]; * register_sound_special_device - register a special sound node * @fops: File operations for the driver * @unit: Unit number to allocate - * @dev: device pointer + * @dev: device pointer * * Allocate a special sound device by minor number from the sound * subsystem. @@ -357,7 +360,6 @@ static struct sound_unit *chains[SOUND_STEP]; * Return: The allocated number is returned on success. On failure, * a negative error code is returned. */ - int register_sound_special_device(const struct file_operations *fops, int unit, struct device *dev) { @@ -367,69 +369,65 @@ int register_sound_special_device(const struct file_operations *fops, int unit, char _name[16]; switch (chain) { - case 0: + case 0: name = "mixer"; break; - case 1: + case 1: name = "sequencer"; if (unit >= SOUND_STEP) goto __unknown; max_unit = unit + 1; break; - case 2: + case 2: name = "midi"; break; - case 3: + case 3: name = "dsp"; break; - case 4: + case 4: name = "audio"; break; - case 5: + case 5: name = "dspW"; break; - case 8: + case 8: name = "sequencer2"; if (unit >= SOUND_STEP) goto __unknown; max_unit = unit + 1; break; - case 9: + case 9: name = "dmmidi"; break; - case 10: + case 10: name = "dmfm"; break; - case 12: + case 12: name = "adsp"; break; - case 13: + case 13: name = "amidi"; break; - case 14: + case 14: name = "admmidi"; break; - default: - { - __unknown: - sprintf(_name, "unknown%d", chain); - if (unit >= SOUND_STEP) - strcat(_name, "-"); - name = _name; - } +__unknown: + default: + sprintf(_name, "unknown%d", chain); + if (unit >= SOUND_STEP) + strcat(_name, "-"); + name = _name; break; } return sound_insert_unit(&chains[chain], fops, -1, unit, max_unit, name, S_IRUSR | S_IWUSR, dev); } - EXPORT_SYMBOL(register_sound_special_device); int register_sound_special(const struct file_operations *fops, int unit) { return register_sound_special_device(fops, unit, NULL); } - EXPORT_SYMBOL(register_sound_special); /** @@ -443,13 +441,11 @@ EXPORT_SYMBOL(register_sound_special); * Return: On success, the allocated number is returned. On failure, * a negative error code is returned. */ - int register_sound_mixer(const struct file_operations *fops, int dev) { return sound_insert_unit(&chains[0], fops, dev, 0, 128, "mixer", S_IRUSR | S_IWUSR, NULL); } - EXPORT_SYMBOL(register_sound_mixer); /** @@ -469,14 +465,13 @@ int register_sound_midi(const struct file_operations *fops, int dev) return sound_insert_unit(&chains[2], fops, dev, 2, 130, "midi", S_IRUSR | S_IWUSR, NULL); } - EXPORT_SYMBOL(register_sound_midi); /* * DSP's are registered as a triple. Register only one and cheat * in open - see below. */ - + /** * register_sound_dsp - register a DSP device * @fops: File operations for the driver @@ -491,13 +486,11 @@ EXPORT_SYMBOL(register_sound_midi); * Return: On success, the allocated number is returned. On failure, * a negative error code is returned. */ - int register_sound_dsp(const struct file_operations *fops, int dev) { return sound_insert_unit(&chains[3], fops, dev, 3, 131, "dsp", S_IWUSR | S_IRUSR, NULL); } - EXPORT_SYMBOL(register_sound_dsp); /** @@ -508,13 +501,10 @@ EXPORT_SYMBOL(register_sound_dsp); * register_sound_special(). The unit passed is the return value from * the register function. */ - - void unregister_sound_special(int unit) { sound_remove_unit(&chains[unit % SOUND_STEP], unit); } - EXPORT_SYMBOL(unregister_sound_special); /** @@ -524,12 +514,10 @@ EXPORT_SYMBOL(unregister_sound_special); * Release a sound device that was allocated with register_sound_mixer(). * The unit passed is the return value from the register function. */ - void unregister_sound_mixer(int unit) { sound_remove_unit(&chains[0], unit); } - EXPORT_SYMBOL(unregister_sound_mixer); /** @@ -539,12 +527,10 @@ EXPORT_SYMBOL(unregister_sound_mixer); * Release a sound device that was allocated with register_sound_midi(). * The unit passed is the return value from the register function. */ - void unregister_sound_midi(int unit) { sound_remove_unit(&chains[2], unit); } - EXPORT_SYMBOL(unregister_sound_midi); /** @@ -556,25 +542,21 @@ EXPORT_SYMBOL(unregister_sound_midi); * * Both of the allocated units are released together automatically. */ - void unregister_sound_dsp(int unit) { sound_remove_unit(&chains[3], unit); } - - EXPORT_SYMBOL(unregister_sound_dsp); static struct sound_unit *__look_for_unit(int chain, int unit) { struct sound_unit *s; - - s=chains[chain]; - while(s && s->unit_minor <= unit) - { - if(s->unit_minor==unit) + + s = chains[chain]; + while (s && s->unit_minor <= unit) { + if (s->unit_minor == unit) return s; - s=s->next; + s = s->next; } return NULL; } @@ -586,14 +568,13 @@ static int soundcore_open(struct inode *inode, struct file *file) struct sound_unit *s; const struct file_operations *new_fops = NULL; - chain=unit&0x0F; - if(chain==4 || chain==5) /* dsp/audio/dsp16 */ - { - unit&=0xF0; - unit|=3; - chain=3; + chain = unit & 0x0F; + if (chain == 4 || chain == 5) { /* dsp/audio/dsp16 */ + unit &= 0xF0; + unit |= 3; + chain = 3; } - + spin_lock(&sound_loader_lock); s = __look_for_unit(chain, unit); if (s) @@ -636,7 +617,7 @@ static int soundcore_open(struct inode *inode, struct file *file) replace_fops(file, new_fops); if (file->f_op->open) - err = file->f_op->open(inode,file); + err = file->f_op->open(inode, file); return err; } @@ -656,7 +637,7 @@ static int __init init_oss_soundcore(void) { if (preclaim_oss && register_chrdev(SOUND_MAJOR, "sound", &soundcore_fops) == -1) { - printk(KERN_ERR "soundcore: sound device already in use.\n"); + pr_err("sound device already in use\n"); return -EBUSY; }

2 1

[alsa-devel] [PATCH] Revert "ALSA: emu10k1: Fix warning: "CCR" redefined"
by Geert Uytterhoeven 11 Mar '14

11 Mar '14

This reverts commit 83fc3bc09518d42e8f5073e2a65884701dfadf19. sh-specific "CCR" and "CCR2" have been prefixed by "SH_" in commit a5f6ea29f9a918403629f8369ae55fac6b09cb53 ('sh: prefix sh-specific "CCR" and "CCR2" by "SH_"'). Signed-off-by: Geert Uytterhoeven <geert(a)linux-m68k.org> --- include/sound/emu10k1.h | 2 -- 1 file changed, 2 deletions(-) diff --git a/include/sound/emu10k1.h b/include/sound/emu10k1.h index dfb42ca6d043..c46908c1bb3f 100644 --- a/include/sound/emu10k1.h +++ b/include/sound/emu10k1.h @@ -436,8 +436,6 @@ #define CCCA_CURRADDR_MASK 0x00ffffff /* Current address of the selected channel */ #define CCCA_CURRADDR 0x18000008 -/* undefine CCR to avoid conflict with the definition for SH */ -#undef CCR #define CCR 0x09 /* Cache control register */ #define CCR_CACHEINVALIDSIZE 0x07190009 #define CCR_CACHEINVALIDSIZE_MASK 0xfe000000 /* Number of invalid samples cache for this channel */ -- 1.7.9.5

2 1

[alsa-devel] [GIT PULL] ASoC fixes for v3.14
by Mark Brown 11 Mar '14

11 Mar '14

The following changes since commit fa389e220254c69ffae0d403eac4146171062d08: Linux 3.14-rc6 (2014-03-09 19:41:57 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.14-rc6 for you to fetch changes up to 7f35afd44b14a39307757629ebc0a199aade52d9: Merge remote-tracking branches 'asoc/fix/88pm860', 'asoc/fix/omap' and 'asoc/fix/si476x' into asoc-linus (2014-03-10 11:30:56 +0000) ---------------------------------------------------------------- ASoC: Fixes for v3.14 A few things here: - Avoid memory leaks in error cases with DPCM, this code has never been that well tested in mainline due to the lack of mainline drivers but we now have one queued for the merge window! - Fix the N810 audio driver to load when booted with DT since the platform was converted to DT during the merge window. - Fixes for initialisation of some MFD drivers that are probably unused in mainline ---------------------------------------------------------------- Aaro Koskinen (1): ASoC: n810: fix init with DT boot Lars-Peter Clausen (2): ASoC: si476x: Fix IO setup ASoC: 88pm860: Fix IO setup Mark Brown (2): Merge remote-tracking branch 'asoc/fix/pcm' into asoc-linus Merge remote-tracking branches 'asoc/fix/88pm860', 'asoc/fix/omap' and 'asoc/fix/si476x' into asoc-linus Patrick Lai (1): ASoC: pcm: free path list before exiting from error conditions sound/soc/codecs/88pm860x-codec.c | 3 +++ sound/soc/codecs/si476x.c | 2 +- sound/soc/omap/n810.c | 4 +++- sound/soc/soc-pcm.c | 3 +++ 4 files changed, 10 insertions(+), 2 deletions(-)

2 1

[alsa-devel] Playstation Eye won't accept rate greater than 16k, 48k device.
by Benjamin Gorman 10 Mar '14

10 Mar '14

Hello, I'm working with a playstation eye and I'm attempting to grab the raw audio in order to perform some processing using it's 4 channel microphone array. It's listed by the vendor as being a 48khz device. Using Audacity on OSX I can record 4 channels at 44,100hz. However, under linux (raspberry pi running debian) I can only record in 16khz using ALSA. arecord -vv /dev/null -r 16000 -f S16_LE -c 4 -D iec958:CARD=CameraB409241,DEV=0 /dev/null < /dev/null Any higher rate won't be accepted. I tried this using the ALSA C API and also under the PortAudio API. Both without luck. Any ideas on how to patch the driver in order to allow for 44100hz recording? The device is clearly capable of it. Regards, Benjamin The University of Dundee is a registered Scottish Charity, No: SC015096

1 0

[alsa-devel] [PATCH] ASOC: spear: spdif_out: Fix mute control
by Lars-Peter Clausen 10 Mar '14

10 Mar '14

For controls registers with snd_soc_add_dai_controls snd_kcontrol_chip() returns a pointer to the DAI, not to the CODEC. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/spear/spdif_out.c | 10 ++-------- 1 file changed, 2 insertions(+), 8 deletions(-) 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]) -- 1.8.0

2 1

[alsa-devel] [PATCH] ASoC: davinci-evm: Register machine level DAPM elements with the card
by Lars-Peter Clausen 10 Mar '14

10 Mar '14

Machine level DAPM widgets and routes should be registered in the card's DAPM context, rather than in the CODEC's context. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/davinci/davinci-evm.c | 17 +++++++++-------- 1 file changed, 9 insertions(+), 8 deletions(-) diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index 5e3bc3c..8d10244 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -95,23 +95,24 @@ 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 */ @@ -120,10 +121,10 @@ static int evm_aic3x_init(struct snd_soc_pcm_runtime *rtd) 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_enable_pin(&card->dapm, "Headphone Jack"); + snd_soc_dapm_enable_pin(&card->dapm, "Line Out"); + snd_soc_dapm_enable_pin(&card->dapm, "Mic Jack"); + snd_soc_dapm_enable_pin(&card->dapm, "Line In"); return 0; } -- 1.8.0

2 1

[alsa-devel] [PATCH] ASoC: migor: Convert to table based DAPM setup
by Lars-Peter Clausen 10 Mar '14

10 Mar '14

Use table based setup to register the DAPM widgets and routes. This on one hand makes the code a bit shorter and cleaner and on the other hand the board level DAPM elements get registered in the card's DAPM context rather than in the CODEC's DAPM context. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/sh/migor.c | 19 +++++-------------- 1 file changed, 5 insertions(+), 14 deletions(-) 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; -- 1.8.0

2 1

[alsa-devel] integration into ASoC
by Maxime Ripard 10 Mar '14

10 Mar '14

Hi Liam, Mark, I have a sound IP that is part of an SoC that I'm willing to write a driver for. This IP is made of a few registers to control the sampling rate, if we're using mono/stereo, plus two fifos, one for playback, one for capture, that can be seed with data. The data are then taken, go through a DAC, and the outer interface of the IP are directly analog signals (so the DAC/ADC are directly in the SoC, and the only interface we have is plain registers). From what I understood from ASoC, you have mostly three components, the DAI, the codec and the platform that plumbers the two first together. Here, my understanding is that it's pretty much the whole three in a single IP. Should such a hardware block be handled into ASoC, and if yes, how? If not, which other framework should be used? Thanks! Maxime -- Maxime Ripard, Free Electrons Embedded Linux, Kernel and Android engineering http://free-electrons.com

3 7

[alsa-devel] [PATCH] add Steinberg UR44 USB Audio Interface support to quirks-table.
by Cyrus Harmon 10 Mar '14

10 Mar '14

* The patch is based on Clemens Ladisch's patch to support the UR22 (http://linuxaudio.org/mailarchive/lad/2013/2/23/197831), with the device ID modified for the UR44. The patch was tested with kernel versions 3.12.11 and 3.12.12 and a PREEMPT RT-patched 3.12.12-rt kernel. Basic recording and playback appear to work. * NB: On one of the systems that I used to test the patch, I was never able to achieve xrun-free JACK performance at -p256 -n3 as I would invariably see xruns roughly once every minute. But this is a new box (Intel haswell NUC) and it seems there are many possible sources of the unacceptable latency. On an older i3 box, the performance was much better. Signed-off-by: Cyrus Harmon <cyrus(a)bobobeach.com> --- sound/usb/quirks-table.h | 30 ++++++++++++++++++++++++++++++ 1 file changed, 30 insertions(+) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index f652b10..987ca01 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -436,6 +436,36 @@ YAMAHA_DEVICE(0x105d, NULL), } } }, +{ + USB_DEVICE(0x0499, 0x150e), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + /* .vendor_name = "Yamaha", */ + /* .product_name = "Steinberg UR44", */ + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 1, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 2, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 3, + .type = QUIRK_MIDI_YAMAHA + }, + { + .ifnum = 4, + .type = QUIRK_IGNORE_INTERFACE + }, + { + .ifnum = -1 + } + } + } +}, YAMAHA_DEVICE(0x2000, "DGP-7"), YAMAHA_DEVICE(0x2001, "DGP-5"), YAMAHA_DEVICE(0x2002, NULL), -- 1.8.3.2

1 0

[alsa-devel] [PATCHv3 0/3] Simplify the CODEC ASoC probe code.
by Xiubo Li 10 Mar '14

10 Mar '14

5 12

[alsa-devel] [PATCH 0/9] Update FLL calculations on Arizona devices
by Charles Keepax 10 Mar '14

10 Mar '14

Newer versions of the Arizona IP have several differences in the FLL configuration. The reference and synchroniser paths have slightly difference FRATIO encodings and the expanded encoding on the reference path should be used to avoid integer mode operation. This series of patches does some refactoring on the FLL configuration code, previously calculation of the FLL configuration was done before we knew whether this would be applied to reference or synchroniser path. These patches add some checks to make sure the requested clock can be accomodated and then does the calculations once we knew where they will be applied. Additionally, we add support for the encodings on the newer IP. Thanks, Charles Charles Keepax (9): ASoC: arizona: An OUTDIV of 1 is not valid, avoid this ASoC: arizona: Add defines for FLL configuration constants ASoC: arizona: Move set of OUTDIV in to arizona_apply_fll ASoC: arizona: Move calculation of FLL configuration ASoC: arizona: Don't pass Fout into arizona_calc_fll ASoC: arizona: Calculate OUTDIV first ASoC: arizona: Calculate FLL gain last mfd: arizona: Add support for new fratio encoding ASoC: arizona: Support new fratio encoding on the wm5110 rev D include/linux/mfd/arizona/registers.h | 6 +- sound/soc/codecs/arizona.c | 254 ++++++++++++++++++++++----------- 2 files changed, 171 insertions(+), 89 deletions(-) -- 1.7.2.5

2 13

[alsa-devel] [PATCH v2 1/3] ASoC: Consolidate snd_soc_register_dai() and snd_soc_register_dais()
by Lars-Peter Clausen 10 Mar '14

10 Mar '14

snd_soc_register_dais() has basically the same code as snd_soc_register_dai(), but running in a loop. The only difference is that snd_soc_register_dai() calls fmt_single_name() to generate the DAIs name and snd_soc_register_dais() calls fmt_multiple_name(). This patch pushes the check in __snd_soc_register_component() which decides whether to call snd_soc_register_dai() or snd_soc_register_dais() to snd_soc_register_dais() to decide which naming scheme to use. This allows us to remove snd_soc_register_dai(). The patch also updates snd_soc_register_dais() to unregister every DAI it finds for the component rather than trying to unregister one DAI for each DAI that was registered. Both have the same result since there won't be more DAIs than what have been registered, but the former is easier to implement. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- Changes since v1: * Fixed compile error that slipped into patch 1, but was fixed in patch 3 --- sound/soc/soc-core.c | 154 ++++++++++++++------------------------------------- 1 file changed, 42 insertions(+), 112 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 8ddb15c..ae2b7cc 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3897,100 +3897,47 @@ 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 * - * @component: The component the DAIs are registered for - * @dai_drv: DAI driver to use for the DAIs + * @component: The component for which the DAIs should be unregistered */ -static int snd_soc_register_dai(struct snd_soc_component *component, - struct snd_soc_dai_driver *dai_drv) +static void snd_soc_unregister_dais(struct snd_soc_component *component) { - struct device *dev = component->dev; - struct snd_soc_codec *codec; - struct snd_soc_dai *dai; - - dev_dbg(dev, "ASoC: dai register %s\n", dev_name(dev)); - - 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) { - kfree(dai); - return -ENOMEM; - } - - dai->component = component; - dai->dev = dev; - dai->driver = dai_drv; - dai->dapm.dev = dev; - if (!dai->driver->ops) - dai->driver->ops = &null_dai_ops; + struct snd_soc_dai *dai, *_dai; mutex_lock(&client_mutex); + list_for_each_entry_safe(dai, _dai, &dai_list, list) { + if (dai->dev != component->dev) + continue; - 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_del(&dai->list); - list_for_each_entry(dai, &dai_list, list) { - if (dev == dai->dev) - goto found; + dev_dbg(component->dev, "ASoC: Unregistered DAI '%s'\n", + dai->name); + kfree(dai->name); + kfree(dai); } - 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 * * @component: The component the DAIs are registered for * @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 snd_soc_component *component, - struct snd_soc_dai_driver *dai_drv, size_t count) + struct snd_soc_dai_driver *dai_drv, size_t count, + bool legacy_dai_naming) { struct device *dev = component->dev; struct snd_soc_codec *codec; 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); @@ -4002,21 +3949,32 @@ static int snd_soc_register_dais(struct snd_soc_component *component, 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->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; @@ -4025,8 +3983,7 @@ static int snd_soc_register_dais(struct snd_soc_component *component, list_for_each_entry(codec, &codec_list, list) { if (codec->dev == dev) { - dev_dbg(dev, - "ASoC: Mapped DAI %s to CODEC %s\n", + dev_dbg(dev, "ASoC: Mapped DAI %s to CODEC %s\n", dai->name, codec->name); dai->codec = codec; break; @@ -4040,33 +3997,18 @@ static int snd_soc_register_dais(struct snd_soc_component *component, 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 * */ @@ -4097,19 +4039,7 @@ __snd_soc_register_component(struct device *dev, cmpnt->dai_drv = dai_drv; cmpnt->num_dai = num_dai; - /* - * 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(cmpnt, dai_drv); - else - ret = snd_soc_register_dais(cmpnt, dai_drv, num_dai); + ret = snd_soc_register_dais(cmpnt, 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; @@ -4164,7 +4094,7 @@ 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); -- 1.8.0

2 3

[alsa-devel] [PATCH 1/3] ASoC: omap-abe-twl6040: No need to register DMIC routes seperatly
by Lars-Peter Clausen 10 Mar '14

10 Mar '14

When using table based DAPM setup there is no need to register DAPM elements for different sub-components separately. The widgets will be registered before the first sub-component is initialized, the routes are only added after the last sub-component has been initialized, meaning everything will be available when it is needed. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/omap/omap-abe-twl6040.c | 19 ++++--------------- 1 file changed, 4 insertions(+), 15 deletions(-) diff --git a/sound/soc/omap/omap-abe-twl6040.c b/sound/soc/omap/omap-abe-twl6040.c index ebb1390..5011bfa 100644 --- a/sound/soc/omap/omap-abe-twl6040.c +++ b/sound/soc/omap/omap-abe-twl6040.c @@ -163,6 +163,10 @@ static const struct snd_soc_dapm_route audio_map[] = { {"AFML", NULL, "Line In"}, {"AFMR", NULL, "Line In"}, + + /* DMIC routing */ + {"DMic", NULL, "Digital Mic"}, + {"Digital Mic", NULL, "Digital Mic1 Bias"}, }; static int omap_abe_twl6040_init(struct snd_soc_pcm_runtime *rtd) @@ -196,20 +200,6 @@ static int omap_abe_twl6040_init(struct snd_soc_pcm_runtime *rtd) return ret; } -static const struct snd_soc_dapm_route dmic_audio_map[] = { - {"DMic", NULL, "Digital Mic"}, - {"Digital Mic", NULL, "Digital Mic1 Bias"}, -}; - -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; - - return snd_soc_dapm_add_routes(dapm, dmic_audio_map, - ARRAY_SIZE(dmic_audio_map)); -} - /* Digital audio interface glue - connects codec <--> CPU */ static struct snd_soc_dai_link abe_twl6040_dai_links[] = { { @@ -229,7 +219,6 @@ static struct snd_soc_dai_link abe_twl6040_dai_links[] = { .codec_dai_name = "dmic-hifi", .platform_name = "omap-pcm-audio", .codec_name = "dmic-codec", - .init = omap_abe_dmic_init, .ops = &omap_abe_dmic_ops, }, }; -- 1.8.0

3 11

[alsa-devel] [PATCH v2 0/4] AM43xx-ePOS-EVM audio support with TLV320AIC31XX driver
by Jyri Sarha 10 Mar '14

10 Mar '14

Since the RFC version of the patches [1] I have addressed Mark Browns comments [2] as described in my mail here [3]. Besr regards, Jyri [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2014-February/073289.h… [2] http://mailman.alsa-project.org/pipermail/alsa-devel/2014-March/073703.html http://mailman.alsa-project.org/pipermail/alsa-devel/2014-March/073704.html http://mailman.alsa-project.org/pipermail/alsa-devel/2014-March/073705.html [3] http://mailman.alsa-project.org/pipermail/alsa-devel/2014-March/073807.html Jyri Sarha (4): ASoC: tlv320aic31xx: Add basic codec driver implementation ASoC: davinci-evm: Add AM43xx-EPOS-EVM audio support ASoC: davinci: Add SND_AM43XX_SOC_EPOS_EVM build option ASoC: tlv320aic32x4: Sort Makefile in alphabetic order .../bindings/sound/davinci-evm-audio.txt | 9 +- .../devicetree/bindings/sound/tlv320aic31xx.txt | 58 + include/dt-bindings/sound/tlv320aic31xx-micbias.h | 9 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 6 +- sound/soc/codecs/tlv320aic31xx.c | 1345 ++++++++++++++++++++ sound/soc/codecs/tlv320aic31xx.h | 258 ++++ sound/soc/davinci/Kconfig | 12 + sound/soc/davinci/Makefile | 1 + sound/soc/davinci/davinci-evm.c | 41 + 10 files changed, 1738 insertions(+), 5 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/tlv320aic31xx.txt create mode 100644 include/dt-bindings/sound/tlv320aic31xx-micbias.h create mode 100644 sound/soc/codecs/tlv320aic31xx.c create mode 100644 sound/soc/codecs/tlv320aic31xx.h -- 1.7.9.5

4 20

[alsa-devel] [PATCH] ASoC: 88pm860: Fix IO setup
by Lars-Peter Clausen 10 Mar '14

10 Mar '14

The 88pm860 is a MFD device and the CODEC driver is using the regmap struct of the parent device, hence automatic IO setup will not work and we need to manually call snd_soc_codec_set_cache_io(). The issue was introduced in commit f9ded3b2e7 ("ASoC: 88pm860x: Use regmap for I/O"). Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/88pm860x-codec.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/88pm860x-codec.c b/sound/soc/codecs/88pm860x-codec.c index 697d24b..8703244 100644 --- a/sound/soc/codecs/88pm860x-codec.c +++ b/sound/soc/codecs/88pm860x-codec.c @@ -1328,6 +1328,9 @@ 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); + if (ret) + return ret; for (i = 0; i < 4; i++) { ret = request_threaded_irq(pm860x->irq[i], NULL, -- 1.8.0

2 1

[alsa-devel] [PATCH RFC 0/2] Fix simple-card *-master DT parameter handling
by Jyri Sarha 10 Mar '14

10 Mar '14

The schematics of bitclock-master and frame-master DT parameters should depend on whether they are found from a cpu-dai or codec sub-node. - bitclock-master in cpu-dai node means Codec-Bitclock-Slave - frame-master in cpu-dai node means Codec-Frame-Slave - bitclock-master in codec node means Codec-Bitclock-Master - frame-master in codec node means Codec-Frame-Master For example in a cpu-dai mode bitclock-master parameter should produce SND_SOC_DAIFMT_CBS_* daifmt flags and a codec node SND_SOC_DAIFMT_CBM_* flags. Best regards, Jyri Jyri Sarha (2): ASoC: core: Add is_cpu_dai_node-parameter to snd_soc_of_parse_daifmt() ASoC: simple-card: Take snd_soc_of_parse_daifmt() change in to account sound/soc/generic/simple-card.c | 13 ++++++++----- sound/soc/soc-core.c | 8 +++++++- 2 files changed, 15 insertions(+), 6 deletions(-) -- 1.7.9.5

1 2

[alsa-devel] [PATCH] ASoC: si476x: Fix IO setup
by Lars-Peter Clausen 10 Mar '14

10 Mar '14

The si476x is a MFD device and the CODEC driver is using the regmap struct of the parent device, hence automatic IO setup will not work and we need to manually call snd_soc_codec_set_cache_io(). The issue was introduced commit d6173df35f ("ASoC: si476x: Remove custom register I/O implementation") Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/si476x.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/si476x.c b/sound/soc/codecs/si476x.c index 52e7cb0..fa2b8e0 100644 --- a/sound/soc/codecs/si476x.c +++ b/sound/soc/codecs/si476x.c @@ -210,7 +210,7 @@ out: static int si476x_codec_probe(struct snd_soc_codec *codec) { codec->control_data = dev_get_regmap(codec->dev->parent, NULL); - return 0; + return snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP); } static struct snd_soc_dai_ops si476x_dai_ops = { -- 1.8.0

2 1

[alsa-devel] [PATCH] ASoC: sam9g20_wm8731: Convert to table based DAPM setup
by Lars-Peter Clausen 10 Mar '14

10 Mar '14

Use table based setup to register the DAPM widgets and routes. This on one hand makes the code a bit shorter and cleaner and on the other hand the board level DAPM elements get registered in the card's DAPM context rather than in the CODEC's DAPM context. While we are at it also remove the snd_soc_dapm_enable_pin() in the init callback. Pins are enabled by default. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/atmel/sam9g20_wm8731.c | 20 +++++++------------- 1 file changed, 7 insertions(+), 13 deletions(-) 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) -- 1.8.0

3 2

[alsa-devel] [PATCH] ASoC: simple-card: Fix property name for simple-audio-card, format
by Peter Ujfalusi 10 Mar '14

10 Mar '14

The "format" part of the property name is missing from the code. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/generic/simple-card.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 7cabcc5c8703..064b404ed252 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -165,7 +165,7 @@ static int asoc_simple_card_parse_of(struct device_node *node, snd_soc_of_parse_card_name(&priv->snd_card, "simple-audio-card,name"); /* get CPU/CODEC common format via simple-audio-card,format */ - priv->daifmt = snd_soc_of_parse_daifmt(node, "simple-audio-card,") & + priv->daifmt = snd_soc_of_parse_daifmt(node, "simple-audio-card,format") & (SND_SOC_DAIFMT_FORMAT_MASK | SND_SOC_DAIFMT_INV_MASK); /* off-codec widgets */ -- 1.9.0

3 4

[alsa-devel] [PATCH v4] AM43xx-ePOS-EVM audio support with TLV320AIC31XX driver
by Jyri Sarha 10 Mar '14

10 Mar '14

Since the v3 version of the patches: - Dropped "ASoC: tlv320aic32x4: Sort Makefile in alphabetic order" out - Changed DAMP widget "Mic Bias" to "MICBIAS" Best regards, Jyri Jyri Sarha (1): ASoC: tlv320aic31xx: Add basic codec driver implementation .../devicetree/bindings/sound/tlv320aic31xx.txt | 61 + include/dt-bindings/sound/tlv320aic31xx-micbias.h | 9 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tlv320aic31xx.c | 1343 ++++++++++++++++++++ sound/soc/codecs/tlv320aic31xx.h | 258 ++++ 6 files changed, 1677 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/tlv320aic31xx.txt create mode 100644 include/dt-bindings/sound/tlv320aic31xx-micbias.h create mode 100644 sound/soc/codecs/tlv320aic31xx.c create mode 100644 sound/soc/codecs/tlv320aic31xx.h -- 1.7.9.5

4 14

[alsa-devel] 5.1 sound system works as 2.0 on all kernels newer than 3.2.21
by giperon＠land.ru 10 Mar '14

10 Mar '14

Hi, my 5.1 sound system works as 2.0 on all kernels newer than 3.2.21. Control "Channel Mode" is missing in "Alsamixer". I use the option "3stack-6ch" (module snd-hda-intel, Realtetsk codec ALC882) for kernel 3.2.21. "Alsa-info" for kernels 3.2.21 and 3.13.6 attached.

2 1

[alsa-devel] About ALSA SoC cpu/codec asymmetry
by Kuninori Morimoto 10 Mar '14

10 Mar '14

Hi Mark, Stephen, Lars Now, ALSA SoC is using "component" style, and its image is like this. ALSA SoC - platform - component (as CPU) - codec - component It is still not symmetry, and confusable. I guess this is understandable. ALSA SoC - platform - cpu - component - codec - component If we add "cpu", it needs "cpu name", same as current "codec name" on soc_bind_dai_link(). I guess it can keep compatible if it allows "NULL" name on "cpu name" at this point. (but print warning ?) What do you think ? Best regards --- Kuninori Morimoto

2 5

[alsa-devel] [PATCH] ASoC: simple-card: card name can be option
by Kuninori Morimoto 10 Mar '14

10 Mar '14

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> snd_card.name is now option on DT case. non-DT case can follow same style, and it is understandable from platform point of view. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/generic/simple-card.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 7cabcc5..5dd4769 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -287,7 +287,6 @@ static int asoc_simple_card_probe(struct platform_device *pdev) } if (!cinfo->name || - !cinfo->card || !cinfo->codec_dai.name || !cinfo->codec || !cinfo->platform || @@ -296,7 +295,7 @@ static int asoc_simple_card_probe(struct platform_device *pdev) return -EINVAL; } - priv->snd_card.name = cinfo->card; + 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; -- 1.7.9.5

2 1

[alsa-devel] [PATCH 1/3] ASoC: Consolidate snd_soc_register_dai() and snd_soc_register_dais()
by Lars-Peter Clausen 09 Mar '14

09 Mar '14

snd_soc_register_dais() has basically the same code as snd_soc_register_dai(), but running in a loop. The only difference is that snd_soc_register_dai() calls fmt_single_name() to generate the DAIs name and snd_soc_register_dais() calls fmt_multiple_name(). This patch pushes the check in __snd_soc_register_component() which decides whether to call snd_soc_register_dai() or snd_soc_register_dais() to snd_soc_register_dais() to decide which naming scheme to use. This allows us to remove snd_soc_register_dai(). The patch also updates snd_soc_register_dais() to unregister every DAI it finds for the component rather than trying to unregister one DAI for each DAI that was registered. Both have the same result since there won't be more DAIs than what have been registered, but the former is easier to implement. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- Should go on top of topic/pcm --- sound/soc/soc-core.c | 153 ++++++++++++++------------------------------------- 1 file changed, 41 insertions(+), 112 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 8ddb15c..2fe0338 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3897,100 +3897,46 @@ 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 * - * @component: The component the DAIs are registered for - * @dai_drv: DAI driver to use for the DAIs + * @component: The component for which the DAIs should be unregistered */ -static int snd_soc_register_dai(struct snd_soc_component *component, - struct snd_soc_dai_driver *dai_drv) +static void snd_soc_unregister_dais(struct snd_soc_component *component) { - struct device *dev = component->dev; - struct snd_soc_codec *codec; - struct snd_soc_dai *dai; - - dev_dbg(dev, "ASoC: dai register %s\n", dev_name(dev)); - - 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) { - kfree(dai); - return -ENOMEM; - } - - dai->component = component; - dai->dev = dev; - dai->driver = dai_drv; - dai->dapm.dev = dev; - if (!dai->driver->ops) - dai->driver->ops = &null_dai_ops; + struct snd_soc_dai *dai, *_dai; mutex_lock(&client_mutex); + list_for_each_entry_safe(dai, _dai, &dai_list, list) { + if (dai->dev != component->dev) + continue; - 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_del(&dai->list); - list_for_each_entry(dai, &dai_list, list) { - if (dev == dai->dev) - goto found; + dev_dbg(dev, "ASoC: Unregistered DAI '%s'\n", dai->name); + kfree(dai->name); + kfree(dai); } - 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 * * @component: The component the DAIs are registered for * @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 snd_soc_component *component, - struct snd_soc_dai_driver *dai_drv, size_t count) + struct snd_soc_dai_driver *dai_drv, size_t count, + bool legacy_dai_naming) { struct device *dev = component->dev; struct snd_soc_codec *codec; 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); @@ -4002,21 +3948,32 @@ static int snd_soc_register_dais(struct snd_soc_component *component, 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->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; @@ -4025,8 +3982,7 @@ static int snd_soc_register_dais(struct snd_soc_component *component, list_for_each_entry(codec, &codec_list, list) { if (codec->dev == dev) { - dev_dbg(dev, - "ASoC: Mapped DAI %s to CODEC %s\n", + dev_dbg(dev, "ASoC: Mapped DAI %s to CODEC %s\n", dai->name, codec->name); dai->codec = codec; break; @@ -4040,33 +3996,18 @@ static int snd_soc_register_dais(struct snd_soc_component *component, 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 * */ @@ -4097,19 +4038,7 @@ __snd_soc_register_component(struct device *dev, cmpnt->dai_drv = dai_drv; cmpnt->num_dai = num_dai; - /* - * 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(cmpnt, dai_drv); - else - ret = snd_soc_register_dais(cmpnt, dai_drv, num_dai); + ret = snd_soc_register_dais(cmpnt, 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; @@ -4164,7 +4093,7 @@ 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); -- 1.8.0

1 2

[alsa-devel] [PATCH v3 0/2] AM43xx-ePOS-EVM audio support with TLV320AIC31XX driver
by Jyri Sarha 09 Mar '14

09 Mar '14

Since the v2 version of the patches [1] I have dropped davinci-evm related patches since I plan to start using simple-card for machine driver. I have also addressed Mark Brown's comments [2] as I described in my mail here [3]. Best regards, Jyri [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2014-March/073809.html [2] http://mailman.alsa-project.org/pipermail/alsa-devel/2014-March/073814.html [3] http://mailman.alsa-project.org/pipermail/alsa-devel/2014-March/073992.html Jyri Sarha (2): ASoC: tlv320aic32x4: Sort Makefile in alphabetic order ASoC: tlv320aic31xx: Add basic codec driver implementation .../devicetree/bindings/sound/tlv320aic31xx.txt | 61 + include/dt-bindings/sound/tlv320aic31xx-micbias.h | 9 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 6 +- sound/soc/codecs/tlv320aic31xx.c | 1343 ++++++++++++++++++++ sound/soc/codecs/tlv320aic31xx.h | 258 ++++ 6 files changed, 1679 insertions(+), 2 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/tlv320aic31xx.txt create mode 100644 include/dt-bindings/sound/tlv320aic31xx-micbias.h create mode 100644 sound/soc/codecs/tlv320aic31xx.c create mode 100644 sound/soc/codecs/tlv320aic31xx.h -- 1.7.9.5

2 4

[alsa-devel] [PATCH] ASoC: davinci-mcasp: Rename the pointer to davinci dma_params as dma_params
by Peter Ujfalusi 09 Mar '14

09 Mar '14

It is quite confusing to name the pointer to davinci_pcm_dma_params as dma_data. Rename the pointer to avoid confusion. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/davinci/davinci-mcasp.c | 44 +++++++++++++++++++-------------------- 1 file changed, 22 insertions(+), 22 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 35b9b5505f07..b0ae0677f023 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -1025,7 +1025,7 @@ 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 resource *mem, *ioarea, *res, *dat; struct davinci_mcasp_pdata *pdata; struct davinci_mcasp *mcasp; @@ -1094,41 +1094,41 @@ 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_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; + mcasp->dma_data[SNDRV_PCM_STREAM_PLAYBACK].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; + dma_params = &mcasp->dma_params[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; + mcasp->dma_data[SNDRV_PCM_STREAM_CAPTURE].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; @@ -1136,9 +1136,9 @@ 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"; -- 1.9.0

2 1

[alsa-devel] [PATCH] ASoC: tlv320aic23: add missing EXPORT_SYMBOLs
by Max Filippov 09 Mar '14

09 Mar '14

This fixes the following build errors when aic23 is configured as module: >> ERROR: "tlv320aic23_probe" >> [sound/soc/codecs/snd-soc-tlv320aic23-i2c.ko] undefined! >> ERROR: "tlv320aic23_regmap" >> [sound/soc/codecs/snd-soc-tlv320aic23-i2c.ko] undefined! Signed-off-by: Max Filippov <jcmvbkbc(a)gmail.com> --- sound/soc/codecs/tlv320aic23.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index 455250f..a934add 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -59,6 +59,7 @@ 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"}; @@ -634,6 +635,7 @@ int tlv320aic23_probe(struct device *dev, struct regmap *regmap) return snd_soc_register_codec(dev, &soc_codec_dev_tlv320aic23, &tlv320aic23_dai, 1); } +EXPORT_SYMBOL(tlv320aic23_probe); MODULE_DESCRIPTION("ASoC TLV320AIC23 codec driver"); MODULE_AUTHOR("Arun KS <arunks(a)mistralsolutions.com>"); -- 1.8.1.4

2 1

[alsa-devel] [PATCH] ASoC: s6105-ipcam: Convert to table based DAPM setup
by Lars-Peter Clausen 08 Mar '14

08 Mar '14

Use table based setup to register the DAPM widgets and routes. This on one hand makes the code a bit shorter and cleaner and on the other hand the board level DAPM elements get registered in the card's DAPM context rather than in the CODEC's DAPM context. While we are at it also remove the snd_soc_dapm_enable_pin() in the init callback, since pins are enabled by default. Also drop the snd_soc_dapm_sync() calls, since they are ignored by the core anyway until the card has been fully instantiated. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/s6000/s6105-ipcam.c | 28 ++++++++++------------------ 1 file changed, 10 insertions(+), 18 deletions(-) diff --git a/sound/soc/s6000/s6105-ipcam.c b/sound/soc/s6000/s6105-ipcam.c index 945e8ab..b2fb37c 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 = { -- 1.8.0

1 0

[alsa-devel] UCM: How does its api relate to ALSA lib api?
by Kamil Wekalski 07 Mar '14

07 Mar '14

Where will be then best to ask these questions?

1 0

[alsa-devel] compress: dpcm_path_get called in wrong place
by noman pouigt 07 Mar '14

07 Mar '14

Shouldn't dpcm_path_get be called earlier ? I think it should be called earlier as happens in soc-pcm file. static int soc_compr_open_fe(struct snd_compr_stream *cstream) { if (dpcm_path_get(fe, stream, &list) <= 0) { dev_dbg(fe->dev, "ASoC: %s no valid %s route\n", fe->dai_link->name, stream ? "capture" : "playback"); In current code we open platform driver and then machine startup function gets called. After this we check for valid path which i think should happen earlier i.e. in the beginning of soc_compr_open_fe function. If you agree, then i will send a patch.

2 1

Re: [alsa-devel] [GIT PULL v2] Enhancement of support for Firewire devices
by Takashi Iwai 07 Mar '14

07 Mar '14

At Wed, 05 Mar 2014 12:13:03 +0900, Takashi Sakamoto wrote: > > This pull-request is an update for my previous pull-request: > [alsa-devel] [GIT PULL][PATCH 00/39] Enhancement of support for Firewire > devices > http://mailman.alsa-project.org/pipermail/alsa-devel/2014-February/073446.h… > > Updates since previous pull-request: > firewire-lib: > - use swap() instead of local macro > - check NULL at memory allocation for sort table > fireworks: > - use bitmap.h for card slots > - add 'firewire' subdirectory for card node in procfs > - add sanity check of module parameter with clamp() > bebob: > - use bitmap.h for card slots > - add 'firewire' subdirectory for card node in procfs > - simplify codes for metering > - use swap() instead of local macro > - change order of last two commits Clemens, could you review this new series? thanks, Takashi

2 1

[alsa-devel] [TINYCOMPRESS][PATCH 1/1] compress: no need to set metadata before calling next_track
by Richard Fitzgerald 07 Mar '14

07 Mar '14

The metadata is mainly for MP3 gapless playback, since the MP3 audio stream does not contain enough information to enable gapless. Other audio formats do not necessarily require any additional metadata so we should allow next_track to be called without any metadata. Signed-off-by: Zhao Weijia <weijia.zhao(a)capelabs.com> Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> --- compress.c | 2 -- 1 files changed, 0 insertions(+), 2 deletions(-) diff --git a/compress.c b/compress.c index 15dfdb7..0c9ecd2 100644 --- a/compress.c +++ b/compress.c @@ -534,8 +534,6 @@ int compress_next_track(struct compress *compress) if (!is_compress_running(compress)) return oops(compress, ENODEV, "device not ready"); - if (!compress->gapless_metadata) - return oops(compress, EPERM, "metadata not set"); if (ioctl(compress->fd, SNDRV_COMPRESS_NEXT_TRACK)) return oops(compress, errno, "cannot set next track\n"); compress->next_track = 1; -- 1.7.2.5

2 7

[alsa-devel] [PATCH] ALSA: hda - Fix loud click noise with IdeaPad 410Y
by Takashi Iwai 07 Mar '14

07 Mar '14

Lenovo IdeaPad 410Y with ALC282 codec makes loud click noises at boot and shutdown. Also, it wrongly misdetects the acpi_thinkpad hook. This patch adds a device-specific fixup for disabling the shutup callback that is the cause of the click noise and also avoiding the thinpad_helper calls. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=71511 Reported-and-tested-by: Guilherme Amadio <guilherme.amadio(a)gmail.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 19 +++++++++++++++++++ 1 file changed, 19 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 850296a1e0ff..8d0a84436674 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3616,6 +3616,19 @@ static void alc_fixup_auto_mute_via_amp(struct hda_codec *codec, } } +static void alc_no_shutup(struct hda_codec *codec) +{ +} + +static void alc_fixup_no_shutup(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + if (action == HDA_FIXUP_ACT_PRE_PROBE) { + struct alc_spec *spec = codec->spec; + spec->shutup = alc_no_shutup; + } +} + static void alc_fixup_headset_mode_alc668(struct hda_codec *codec, const struct hda_fixup *fix, int action) { @@ -3844,6 +3857,7 @@ enum { ALC269_FIXUP_HP_GPIO_LED, ALC269_FIXUP_INV_DMIC, ALC269_FIXUP_LENOVO_DOCK, + ALC269_FIXUP_NO_SHUTUP, ALC286_FIXUP_SONY_MIC_NO_PRESENCE, ALC269_FIXUP_PINCFG_NO_HP_TO_LINEOUT, ALC269_FIXUP_DELL1_MIC_NO_PRESENCE, @@ -4020,6 +4034,10 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_inv_dmic_0x12, }, + [ALC269_FIXUP_NO_SHUTUP] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_no_shutup, + }, [ALC269_FIXUP_LENOVO_DOCK] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -4405,6 +4423,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x2212, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x2214, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x2215, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), + SND_PCI_QUIRK(0x17aa, 0x3978, "IdeaPad Y410P", ALC269_FIXUP_NO_SHUTUP), SND_PCI_QUIRK(0x17aa, 0x5013, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x501a, "Thinkpad", ALC283_FIXUP_INT_MIC), SND_PCI_QUIRK(0x17aa, 0x5026, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), -- 1.9.0

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Re: [alsa-devel] Call for testing: ALSA driver for Fireworks/BeBoB based devices
by Brand, Mark (Mr) (Summerstrand Campus South) 07 Mar '14

07 Mar '14

Hi Takashi, Thank you once again! Your comments have been very helpful. I now understand how to run ffado-dbus-server/ffado-mixer alongside your ALSA driver. I have pulled your revised Ozonic mixer from ffado-mixer, and when I now invoke ffado-mixer, it properly detects the Ozonic. I can also verify that the mixer control of my audio pass-through (which is plugged into inputs 3 & 4) works as expected. However, I have not yet been able to get audio playback (using "aplay -D plughw:2,0 Desktop/test.wav"). The .wav file is a short, 44.1/16 mono file saved from Audacity, if that's relevant. The terminal output of aplay seems to reflect successful playback, and I hear audible clicks as the file starts and ends its playback. The pass-through is also momentarily muted as aplay starts and stops. I have experimented with every conceivable combination of ffado-mixer settings, ultimately returning to the default settings that it was installed with. When aplay executes, the mixer app shows check marks next to both "Outgoing" and "Incoming". I have been testing the latest git version of the driver (updated this morning), and installing/uninstalling by DKMS. Please, is there anything you can suggest that I should try next? Is there anything obvious that I appear to have overlooked? I will try with a wider range of test materials tomorrow, as well as trying other ALSA apps, but I wanted to get this posted in case that's not the issue. I also recently saw some comments suggesting that pulseaudio be killed. Could that be the issue? I will investigate this tomorrow too. Thank you sincerely, Mark Brand. ________________________________________ From: Takashi Sakamoto [o-takashi(a)sakamocchi.jp] Sent: Sunday, February 23, 2014 9:55 AM To: Brand, Mark (Mr) (Summerstrand Campus South) Cc: alsa-devel(a)alsa-project.org; ffado-devel(a)lists.sf.net Subject: Re: Call for testing: ALSA driver for Fireworks/BeBoB based devices Hi Mark, Thanks for your testing. Ozonic is one of devices for which I test. > My testing is probably not particularly rigorous, and I would > sincerely appreciate your advice on what steps to follow > in order to verify your driver's functionality. Try playback/capture PCM samples/MIDI messages with ALSA applications. > In particular, you mention "(to control mixer channels please use > FFADO upstream)". My drivers don't give ways to control device's internal mixer. So you need the other stuff for this purpose. Please refer to this page. You can see some screenshots of ffado-mixer for 'M-Audio Ozonic'. I already commited these improvement to current FFADO trunk except for FW1814/ProjectMiX: MaudioBebob http://subversion.ffado.org/wiki/MaudioBebob > I have not yet set up FFADO on this machine, but I also see under > "Note": "- Don't use simultaneously 'ALSA PCM/MIDI playback/capture' > and 'jackd with Firewire (FFADO) backend'. Both of them try > connecting to the device when another is running." Would you please > explain? How should one rely on the "upstream" mixer functionality > while keeping jackd out of ALSA's way? Devices which my drivers support handle a pair of streams over IEEE1394 bus. PCM samples/MIDI messages are multiplexed into these streams. And snd-fireworks/snd-bebob uses 'duplex streams synchronization'. As a result: - ALSA PCM/MIDI functionality and jackd with firewire backend conflict for its streaming functionality. But: - ALSA PCM/MIDI fuctionality and ffado-dbus-server/ffado-mixer do not interrupt each other. > I open Audacity, which now lists "Ozonic: Ozonic PCM (hw:2,0)" as > ALSA input and output options. However, no audio input is shown when > I try to monitor the audio input I confirm that Audacity has such bugs to detect ALSA PCM character devices. In detail, its PortAudio v19 codes includes bugs. I have no will to investigate this bug because it's not within my work for these drivers. > $aplay -D hw:2,0 test.wav > (where test.wav has been saved from Audacity) > ... and this responds with: > aplay: main:722: audio open error: Input/output error According to aplay codes, snd_pcm_open() generate this error. I guess that another application still uses PCM character device for Ozonic. > $ aplay -D hw:2,0 Desktop/test.wav > Playing WAVE 'Desktop/test.wav' : Signed 16 bit Little Endian, Rate > 44100 Hz, Mono aplay: set_params:1239: Channels count not available You MUST add 'c' option to indicate the number of channels with 'hw' PCM device. For convinience, use 'plughw' PCM device instead, like: $ aplay -D plughw:2,0 Desktop/test.wav I note an item: - A few days ago, I fixed bug to fail to start stream for Ozonic. See: bebob: fix mappling for MIDI conformant data in stream channels https://github.com/takaswie/snd-firewire-improve/commit/545ab7068f1d53b2b17… Thanks Takashi Sakamoto o-takashi(a)sakamocchi.jp ________________________________ NOTICE: Please note that this eMail, and the contents thereof, is subject to the standard NMMU eMail disclaimer which may be found at: http://www.nmmu.ac.za/disclaimer/email.htm

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[alsa-devel] Controlling the cards order
by Olivier Langlois 07 Mar '14

07 Mar '14

Hi, I have added a second sound device in my system and it unexpectedly took the first card slot. I have found that it is easy to control the order as long as the devices use different drivers by modifying /etc/modprobe.d/alsa.conf ( http://alsa.opensrc.org/MultipleCards ) but in my situation I have 2 different devices using the same driver. I think that there is hope that I find an answer by digging the more general topic of ordering PCI devices drivers loading but if someone has few pointers to share, any help is welcome! Here are my 2 devices that I would like to change the order: lano1106@whippet2 /proc/asound $ cat cards 0 [PCH ]: HDA-Intel - HDA Intel PCH HDA Intel PCH at 0xfbf20000 irq 84 1 [HDMI ]: HDA-Intel - HDA ATI HDMI HDA ATI HDMI at 0xfbe60000 irq 88 thank you for your time, Olivier

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[alsa-devel] [PATCH] ASoC: pcm512x: Split out bus drivers
by Mark Brown 07 Mar '14

07 Mar '14

From: Mark Brown <broonie(a)linaro.org> Move to the new style of defining the bus interfaces in separate modules in order to simplify dependencies. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/Kconfig | 18 +++++- sound/soc/codecs/Makefile | 4 ++ sound/soc/codecs/pcm512x-i2c.c | 71 ++++++++++++++++++++++ sound/soc/codecs/pcm512x-spi.c | 69 +++++++++++++++++++++ sound/soc/codecs/pcm512x.c | 133 +++-------------------------------------- sound/soc/codecs/pcm512x.h | 9 +++ 6 files changed, 175 insertions(+), 129 deletions(-) create mode 100644 sound/soc/codecs/pcm512x-i2c.c create mode 100644 sound/soc/codecs/pcm512x-spi.c diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index f2383eb..3a6046f 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -66,7 +66,8 @@ 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 if SND_SOC_I2C_AND_SPI + 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 @@ -362,8 +363,19 @@ config SND_SOC_PCM3008 tristate config SND_SOC_PCM512x - tristate "Texas Instruments PCM512x CODECs" - select REGMAP + 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 diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 6af7a55..ac7025c 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -54,6 +54,8 @@ 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 @@ -197,6 +199,8 @@ 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 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(a)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(a)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(a)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 index a035ba8..0014d14 100644 --- a/sound/soc/codecs/pcm512x.c +++ b/sound/soc/codecs/pcm512x.c @@ -18,11 +18,9 @@ #include <linux/init.h> #include <linux/module.h> #include <linux/clk.h> -#include <linux/i2c.h> #include <linux/pm_runtime.h> #include <linux/regmap.h> #include <linux/regulator/consumer.h> -#include <linux/spi/spi.h> #include <sound/soc.h> #include <sound/soc-dapm.h> #include <sound/tlv.h> @@ -375,7 +373,7 @@ static const struct regmap_range_cfg pcm512x_range = { .window_start = 0, .window_len = 0x100, }; -static const struct regmap_config pcm512x_regmap = { +const struct regmap_config pcm512x_regmap = { .reg_bits = 8, .val_bits = 8, @@ -391,14 +389,7 @@ static const struct regmap_config pcm512x_regmap = { .cache_type = REGCACHE_RBTREE, }; -static const struct of_device_id pcm512x_of_match[] = { - { .compatible = "ti,pcm5121", }, - { .compatible = "ti,pcm5122", }, - { } -}; -MODULE_DEVICE_TABLE(of, pcm512x_of_match); - -static int pcm512x_probe(struct device *dev, struct regmap *regmap) +int pcm512x_probe(struct device *dev, struct regmap *regmap) { struct pcm512x_priv *pcm512x; int i, ret; @@ -510,8 +501,9 @@ err: pcm512x->supplies); return ret; } +EXPORT_SYMBOL_GPL(pcm512x_probe); -static void pcm512x_remove(struct device *dev) +void pcm512x_remove(struct device *dev) { struct pcm512x_priv *pcm512x = dev_get_drvdata(dev); @@ -522,6 +514,7 @@ static void pcm512x_remove(struct device *dev) regulator_bulk_disable(ARRAY_SIZE(pcm512x->supplies), pcm512x->supplies); } +EXPORT_SYMBOL_GPL(pcm512x_remove); static int pcm512x_suspend(struct device *dev) { @@ -585,122 +578,10 @@ static int pcm512x_resume(struct device *dev) return 0; } -static const struct dev_pm_ops pcm512x_pm_ops = { +const struct dev_pm_ops pcm512x_pm_ops = { SET_RUNTIME_PM_OPS(pcm512x_suspend, pcm512x_resume, NULL) }; - -#if IS_ENABLED(CONFIG_I2C) -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 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, - }, -}; -#endif - -#if defined(CONFIG_SPI_MASTER) -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 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, - }, -}; -#endif - -static int __init pcm512x_modinit(void) -{ - int ret = 0; - -#if IS_ENABLED(CONFIG_I2C) - ret = i2c_add_driver(&pcm512x_i2c_driver); - if (ret) { - printk(KERN_ERR "Failed to register pcm512x I2C driver: %d\n", - ret); - } -#endif -#if defined(CONFIG_SPI_MASTER) - ret = spi_register_driver(&pcm512x_spi_driver); - if (ret != 0) { - printk(KERN_ERR "Failed to register pcm512x SPI driver: %d\n", - ret); - } -#endif - return ret; -} -module_init(pcm512x_modinit); - -static void __exit pcm512x_exit(void) -{ -#if IS_ENABLED(CONFIG_I2C) - i2c_del_driver(&pcm512x_i2c_driver); -#endif -#if defined(CONFIG_SPI_MASTER) - spi_unregister_driver(&pcm512x_spi_driver); -#endif -} -module_exit(pcm512x_exit); +EXPORT_SYMBOL_GPL(pcm512x_pm_ops); MODULE_DESCRIPTION("ASoC PCM512x codec driver"); MODULE_AUTHOR("Mark Brown <broonie(a)linaro.org>"); diff --git a/sound/soc/codecs/pcm512x.h b/sound/soc/codecs/pcm512x.h index ac4a52c..6ee76aa 100644 --- a/sound/soc/codecs/pcm512x.h +++ b/sound/soc/codecs/pcm512x.h @@ -17,6 +17,9 @@ #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)) @@ -159,4 +162,10 @@ #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 -- 1.9.0

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[alsa-devel] [PATCH] ASoC: pcm512x: Fix duplicate const warning
by Mark Brown 07 Mar '14

07 Mar '14

From: Mark Brown <broonie(a)linaro.org> Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/pcm512x.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/pcm512x.c b/sound/soc/codecs/pcm512x.c index 3a0bbb6..a035ba8 100644 --- a/sound/soc/codecs/pcm512x.c +++ b/sound/soc/codecs/pcm512x.c @@ -203,10 +203,10 @@ static const unsigned int pcm512x_dsp_program_values[] = { 7, }; -static const SOC_VALUE_ENUM_SINGLE_DECL(pcm512x_dsp_program, - PCM512x_DSP_PROGRAM, 0, 0x1f, - pcm512x_dsp_program_texts, - pcm512x_dsp_program_values); +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" -- 1.9.0

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[alsa-devel] [PATCH v2 0/2] add support for SPI control mode to tlv320aic23
by Max Filippov 07 Mar '14

07 Mar '14

Hi, this series splits I2C-specific part of tlv320aic23 codec driver into a separate file and adds SPI control mode support. It also updates Kconfigs of existing tlv320aic23 users so that I2C variant is selected. Changes v1->v2: - split i2c interface into separate file; - implement spi interface in a separate file; - update Kconfigs of existing users. Max Filippov (2): ASoC: tlv320aic23: add support for SPI control mode ASoC: update Kconfig of AIC23 users to select I2C variant sound/soc/atmel/Kconfig | 2 +- sound/soc/cirrus/Kconfig | 2 +- sound/soc/codecs/Kconfig | 11 ++++++- sound/soc/codecs/Makefile | 4 +++ sound/soc/codecs/tlv320aic23-i2c.c | 59 ++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/tlv320aic23-spi.c | 57 ++++++++++++++++++++++++++++++++++++ sound/soc/codecs/tlv320aic23.c | 55 +++++++---------------------------- sound/soc/codecs/tlv320aic23.h | 6 ++++ sound/soc/fsl/Kconfig | 2 +- sound/soc/omap/Kconfig | 4 +-- sound/soc/samsung/Kconfig | 2 +- sound/soc/tegra/Kconfig | 2 +- 12 files changed, 154 insertions(+), 52 deletions(-) create mode 100644 sound/soc/codecs/tlv320aic23-i2c.c create mode 100644 sound/soc/codecs/tlv320aic23-spi.c -- 1.8.1.4

2 3

[alsa-devel] [PATCH] ASoC: Intel: Check Haswell IPC process_reply/notification return value.
by Liam Girdwood 07 Mar '14

07 Mar '14

Check the return value for error when processing replies and notifications. The patch 22981243589c: "ASoC: Intel: Add Haswell/Broadwell IPC" from > Feb 20, 2014, leads to the following imaginary static checker warning: > > sound/soc/intel/sst-haswell-ipc.c:898 hsw_irq_thread() > warn: this is always true. > > sound/soc/intel/sst-haswell-ipc.c > 895 /* Handle Immediate reply from DSP Core */ > 896 handled = hsw_process_reply(hsw, ipcx); > ^^^^^^^^^^^^^^^^^ > Returns 1 on success/error and -EIO on error. > > 897 > 898 if (handled) { > 899 /* clear DONE bit - tell DSP we have completed */ > 900 sst_dsp_shim_update_bits_unlocked(sst, SST_IPCX, > 901 SST_IPCX_DONE, 0); > 902 > 903 /* unmask Done interrupt */ > 904 sst_dsp_shim_update_bits_unlocked(sst, SST_IMRX, > 905 SST_IMRX_DONE, 0); > 906 } > Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- sound/soc/intel/sst-haswell-ipc.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/sst-haswell-ipc.c b/sound/soc/intel/sst-haswell-ipc.c index 1f1576a..f46bb4d 100644 --- a/sound/soc/intel/sst-haswell-ipc.c +++ b/sound/soc/intel/sst-haswell-ipc.c @@ -895,7 +895,7 @@ static irqreturn_t hsw_irq_thread(int irq, void *context) /* Handle Immediate reply from DSP Core */ handled = hsw_process_reply(hsw, ipcx); - if (handled) { + if (handled > 0) { /* clear DONE bit - tell DSP we have completed */ sst_dsp_shim_update_bits_unlocked(sst, SST_IPCX, SST_IPCX_DONE, 0); @@ -913,7 +913,7 @@ static irqreturn_t hsw_irq_thread(int irq, void *context) handled = hsw_process_notification(hsw); /* clear BUSY bit and set DONE bit - accept new messages */ - if (handled) { + if (handled > 0) { sst_dsp_shim_update_bits_unlocked(sst, SST_IPCD, SST_IPCD_BUSY | SST_IPCD_DONE, SST_IPCD_DONE); -- 1.8.3.2

2 5

[alsa-devel] [PATCH] ASoC: wm_adsp: Correct type specifier in printf
by Charles Keepax 07 Mar '14

07 Mar '14

Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 937af6f..bb5f7b4 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -706,7 +706,7 @@ static int wm_adsp_load(struct wm_adsp *dsp) to_write); if (ret != 0) { adsp_err(dsp, - "%s.%d: Failed to write %d bytes at %d in %s: %d\n", + "%s.%d: Failed to write %zd bytes at %d in %s: %d\n", file, regions, to_write, offset, region_name, ret); -- 1.7.2.5

2 1

Re: [alsa-devel] ASoC: wm5110: Extend SYSCLK patch file for rev D
by Ben Hutchings 06 Mar '14

06 Mar '14

I'm looking at commit 34354792432b6e0a3b156819a1a19716c50d3473 and include/linux/mfd/arizona/registers.h, and wondering what exactly this the register patch is changing. It looks like there are some registers not named elsewhere (addresses 0x0049, 0x004a, 0x0057, 0x0058). What about the higher addresses? Are these more registers, or is this patching firmware in device memory? Ben. -- Ben Hutchings The world is coming to an end. Please log off.

2 2

[alsa-devel] [PATCH] ASoC: tlv320aic23: add support for SPI control mode
by Max Filippov 06 Mar '14

06 Mar '14

tlv320aic23 chip control interface may work in either I2C or SPI mode depending on the MODE pin state. Functionality and register layout are independent of the control mode. Provide i2c and spi driver entry points when CONFIG_I2C and CONFIG_SPI_MASTER are enabled respectively. Signed-off-by: Max Filippov <jcmvbkbc(a)gmail.com> --- sound/soc/codecs/Kconfig | 2 +- sound/soc/codecs/tlv320aic23.c | 58 +++++++++++++++++++++++++++++++++++++++--- 2 files changed, 56 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 983d087a..6d82de5 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -71,7 +71,7 @@ config SND_SOC_ALL_CODECS 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 if SND_SOC_I2C_AND_SPI select SND_SOC_TLV320AIC26 if SPI_MASTER select SND_SOC_TLV320AIC32X4 if I2C select SND_SOC_TLV320AIC3X if I2C diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index 5d430cc..2418260 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -26,6 +26,7 @@ #include <linux/i2c.h> #include <linux/regmap.h> #include <linux/slab.h> +#include <linux/spi/spi.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -617,12 +618,13 @@ static struct snd_soc_codec_driver soc_codec_dev_tlv320aic23 = { .num_dapm_routes = ARRAY_SIZE(tlv320aic23_intercon), }; +#ifdef CONFIG_I2C /* * 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) +static int tlv320aic23_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *i2c_id) { struct aic23 *aic23; int ret; @@ -661,12 +663,62 @@ static struct i2c_driver tlv320aic23_i2c_driver = { .driver = { .name = "tlv320aic23-codec", }, - .probe = tlv320aic23_codec_probe, + .probe = tlv320aic23_i2c_probe, .remove = __exit_p(tlv320aic23_i2c_remove), .id_table = tlv320aic23_id, }; module_i2c_driver(tlv320aic23_i2c_driver); +#endif /* CONFIG_I2C */ + +#ifdef CONFIG_SPI_MASTER +static int aic23_spi_probe(struct spi_device *spi) +{ + struct aic23 *aic23; + int ret; + + dev_dbg(&spi->dev, "probing tlv320aic23 spi device\n"); + + spi->bits_per_word = 16; + spi->mode = SPI_MODE_0; + ret = spi_setup(spi); + if (ret < 0) + return ret; + + /* Allocate driver data */ + aic23 = devm_kzalloc(&spi->dev, sizeof(struct aic23), GFP_KERNEL); + if (!aic23) + return -ENOMEM; + + aic23->regmap = devm_regmap_init_spi(spi, &tlv320aic23_regmap); + if (IS_ERR(aic23->regmap)) + return PTR_ERR(aic23->regmap); + + /* Initialize the driver data */ + dev_set_drvdata(&spi->dev, aic23); + + ret = snd_soc_register_codec(&spi->dev, + &soc_codec_dev_tlv320aic23, &tlv320aic23_dai, 1); + return ret; +} + +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); +#endif /* CONFIG_SPI_MASTER */ MODULE_DESCRIPTION("ASoC TLV320AIC23 codec driver"); MODULE_AUTHOR("Arun KS <arunks(a)mistralsolutions.com>"); -- 1.8.1.4

3 5

[alsa-devel] [PATCH 1/7] ASoC: Handle ignore_pmdown_time for CODEC to CODEC links
by Lars-Peter Clausen 06 Mar '14

06 Mar '14

For CODEC to CODEC links we should only immediately power down if both CODECs are configured to ignore the power down delay. Factor the logic for this into a helper function that can be used for both compressed and normal PCMs. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- include/sound/soc.h | 2 ++ sound/soc/soc-compress.c | 3 +-- sound/soc/soc-pcm.c | 27 +++++++++++++++++++++++++-- 3 files changed, 28 insertions(+), 4 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 9a00147..93c31c7 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -413,6 +413,8 @@ struct snd_pcm_substream *snd_soc_get_dai_substream(struct snd_soc_card *card, struct snd_soc_pcm_runtime *snd_soc_get_pcm_runtime(struct snd_soc_card *card, const char *dai_link); +bool snd_soc_runtime_ignore_pmdown_time(struct snd_soc_pcm_runtime *rtd); + /* Utility functions to get clock rates from various things */ int snd_soc_calc_frame_size(int sample_size, int channels, int tdm_slots); int snd_soc_params_to_frame_size(struct snd_pcm_hw_params *params); diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 5e9690c..ef585af 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -235,8 +235,7 @@ 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); diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 47e1ce7..f098c80 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -35,6 +35,30 @@ #define DPCM_MAX_BE_USERS 8 /** + * 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) +{ + bool ignore = true; + + if (!rtd->pmdown_time || rtd->dai_link->ignore_pmdown_time) + return true; + + if (rtd->cpu_dai->codec) + ignore &= rtd->cpu_dai->codec->ignore_pmdown_time; + + ignore &= rtd->codec_dai->codec->ignore_pmdown_time; + + return ignore; +} + +/** * snd_soc_set_runtime_hwparams - set the runtime hardware parameters * @substream: the pcm substream * @hw: the hardware parameters @@ -496,8 +520,7 @@ 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, -- 1.8.0

2 7

[alsa-devel] [PATCH] ASoC: dapm: Sprinkle lockdep asserts through the code
by Mark Brown 06 Mar '14

06 Mar '14

From: Mark Brown <broonie(a)linaro.org> Try to spot locking issues by asserting that the DAPM mutex is held when it should be. There's a bit of fun due to us not requiring the lock to be held prior to the card being instantiated which mean we need to wrap the assert in some paths and this isn't methodical by any stretch of the imagination. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- I've only compile tested this so it's entirely possible there's some oversights here - at this point I'm mostly just throwing it out there for people to play with as a work in progress. sound/soc/soc-dapm.c | 18 ++++++++++++++++++ 1 file changed, 18 insertions(+) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 3728e21..c8a780d 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -111,6 +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) @@ -144,6 +150,8 @@ static bool dapm_dirty_widget(struct snd_soc_dapm_widget *w) 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); @@ -356,6 +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) { @@ -1750,6 +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) { @@ -2045,6 +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]) @@ -2095,6 +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; @@ -2260,6 +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; -- 1.9.0

1 0

[alsa-devel] [PATCH] ASoC: dapm: Staticise dapm_mark_dirty()
by Mark Brown 06 Mar '14

06 Mar '14

From: Mark Brown <broonie(a)linaro.org> The function is not called outside soc-dapm.c so there is no need for it to be exported. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- include/sound/soc-dapm.h | 1 - sound/soc/soc-dapm.c | 3 +-- 2 files changed, 1 insertion(+), 3 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 04d32d8..ef78f56 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -449,7 +449,6 @@ int snd_soc_dapm_ignore_suspend(struct snd_soc_dapm_context *dapm, void snd_soc_dapm_auto_nc_codec_pins(struct snd_soc_codec *codec); /* Mostly internal - should not normally be used */ -void dapm_mark_dirty(struct snd_soc_dapm_widget *w, const char *reason); void dapm_mark_io_dirty(struct snd_soc_dapm_context *dapm); /* dapm path query */ diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 0d8ba30..3728e21 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -142,7 +142,7 @@ 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) { if (!dapm_dirty_widget(w)) { dev_vdbg(w->dapm->dev, "Marking %s dirty due to %s\n", @@ -150,7 +150,6 @@ void dapm_mark_dirty(struct snd_soc_dapm_widget *w, const char *reason) list_add_tail(&w->dirty, &w->dapm->card->dapm_dirty); } } -EXPORT_SYMBOL_GPL(dapm_mark_dirty); void dapm_mark_io_dirty(struct snd_soc_dapm_context *dapm) { -- 1.9.0

1 0

[alsa-devel] [PATCH] ASoC: cs4271: Fix build error without CONFIG_SPI_MASTER
by Takashi Iwai 06 Mar '14

06 Mar '14

cs4271_common_probe() is called from cs4271_i2c_probe() but defined in CONFIG_SPI_MASTER block, thus it results in a build error when CONFIG_SPI_MASTER=n: sound/soc/codecs/cs4271.c:721:2: error: implicit declaration of function ‘cs4271_common_probe’ [-Werror=implicit-function-declaration] Move the function out of #if block. Fixes: d6cf89ee07cb ('ASoC: cs4271: claim reset GPIO in bus probe function') Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/soc/codecs/cs4271.c | 32 ++++++++++++++++---------------- 1 file changed, 16 insertions(+), 16 deletions(-) diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index 96c309777208..aef4965750c7 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -612,22 +612,6 @@ static struct snd_soc_codec_driver soc_codec_dev_cs4271 = { .num_dapm_routes = ARRAY_SIZE(cs4271_dapm_routes), }; -#if defined(CONFIG_SPI_MASTER) - -static const struct regmap_config cs4271_spi_regmap = { - .reg_bits = 16, - .val_bits = 8, - .max_register = CS4271_LASTREG, - .read_flag_mask = 0x21, - .write_flag_mask = 0x20, - - .reg_defaults = cs4271_reg_defaults, - .num_reg_defaults = ARRAY_SIZE(cs4271_reg_defaults), - .cache_type = REGCACHE_RBTREE, - - .volatile_reg = cs4271_volatile_reg, -}; - static int cs4271_common_probe(struct device *dev, struct cs4271_private **c) { @@ -658,6 +642,22 @@ static int cs4271_common_probe(struct device *dev, return 0; } +#if defined(CONFIG_SPI_MASTER) + +static const struct regmap_config cs4271_spi_regmap = { + .reg_bits = 16, + .val_bits = 8, + .max_register = CS4271_LASTREG, + .read_flag_mask = 0x21, + .write_flag_mask = 0x20, + + .reg_defaults = cs4271_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(cs4271_reg_defaults), + .cache_type = REGCACHE_RBTREE, + + .volatile_reg = cs4271_volatile_reg, +}; + static int cs4271_spi_probe(struct spi_device *spi) { struct cs4271_private *cs4271; -- 1.9.0

4 3

[alsa-devel] [patch] ASoC: Baytrail: fix error handling in sst_byt_dsp_init()
by Dan Carpenter 06 Mar '14

06 Mar '14

Calling "kfree(byt)" is a double free because that was allocated with devm_kzalloc(). There were a couple places which leak "byt->msg". That memory is allocated in msg_empty_list_init(). Fixes: f7d01fd6754c ('ASoC: Intel: Add Intel Baytrail SST DSP IPC support') Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/intel/sst-baytrail-ipc.c b/sound/soc/intel/sst-baytrail-ipc.c index c12e194bbc6b..d0eaeee21be4 100644 --- a/sound/soc/intel/sst-baytrail-ipc.c +++ b/sound/soc/intel/sst-baytrail-ipc.c @@ -796,7 +796,7 @@ int sst_byt_dsp_init(struct device *dev, struct sst_pdata *pdata) err = msg_empty_list_init(byt); if (err < 0) - goto list_err; + return -ENOMEM; /* start the IPC message thread */ init_kthread_worker(&byt->kworker); @@ -806,7 +806,7 @@ int sst_byt_dsp_init(struct device *dev, struct sst_pdata *pdata) 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 list_err; + goto err_free_msg; } init_kthread_work(&byt->kwork, sst_byt_ipc_tx_msgs); @@ -816,7 +816,7 @@ int sst_byt_dsp_init(struct device *dev, struct sst_pdata *pdata) byt->dsp = sst_dsp_new(dev, &byt_dev, pdata); if (byt->dsp == NULL) { err = -ENODEV; - goto list_err; + goto err_free_msg; } /* keep the DSP in reset state for base FW loading */ @@ -848,9 +848,9 @@ boot_err: sst_fw_free(byt_sst_fw); fw_err: sst_dsp_free(byt->dsp); +err_free_msg: kfree(byt->msg); -list_err: - kfree(byt); + return err; } EXPORT_SYMBOL_GPL(sst_byt_dsp_init);

3 2

[alsa-devel] UCM: How does its api relate to ALSA lib api?
by kaweka 05 Mar '14

05 Mar '14

Hi, I am a newcomer if it goes to following topics: ALSA, audio in Linux, audio in embedded Linux. However several years of experience in audio development on other platforms. As next a development of audio applications in embedded Linux is the task. However have few gaps in understanding of the audio system architecture. It would be helpful to have answers to following questions. For any hints lot of thanks in advance! * There is well-known handshaking application-ALSA lib based on amonst others snd_pcm_open snd_pcm_hw_params_alloca snd_pcm_hw_params_any snd_pcm_hw_params_set_... snd_pcm_hw_params snd_pcm_hw_params_get_... snd_pcm_hw_params_is_... snd_pcm_hw_params_can_... snd_pcm_close * Does the UCM and its api complement the api and handshaking as above. Or is it rather a question of replacement? * If neither nor, how to describe best the relation between UCM api and ALSA lib api? * How does the entire handshaking application-audio system look like if to use ALSA and UCM api? * An use case seems to be just a container for verb, device, modifier. Is an use case introducing any functionality to be found neither in verb, nor in device, nor in modifier? * For what reason does the term / definition of use case exists at all. Will verb, device, modifier not be enough? * What is the difference between switching to other use case and modifying a use case? Why not to switch to other use case in situations where modifier is used? * QoS api seems to introduce some redundancy. If voice, or music, or other - this information can be gathered from use case. A use case does already read voice, music, other! Why is qos api necessary? Best Regards kaweka

1 0

[alsa-devel] Add "cval->res = 384" quirk for Logitech Webcam C500
by Simon 05 Mar '14

05 Mar '14

Hello, Since some time I am successfully using this workaround http://mailman.alsa-project.org/pipermail/alsa-devel/2011-March/037657.html to solve the problem for the Logitech C500 Webcam (046d:0807). Would it be possible to add this device to the list included in the mixer.c file? Thanks

2 1

[alsa-devel] [patch] ALSA: echoaudio: use after free on error
by Dan Carpenter 05 Mar '14

05 Mar '14

There are some places where we dereference "chip" in the error message but we've already freed it. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c index 166ec0c120d4..9f10c9e0df5e 100644 --- a/sound/pci/echoaudio/echoaudio.c +++ b/sound/pci/echoaudio/echoaudio.c @@ -1990,8 +1990,8 @@ static int snd_echo_create(struct snd_card *card, if ((chip->iores = request_mem_region(chip->dsp_registers_phys, sz, ECHOCARD_NAME)) == NULL) { - snd_echo_free(chip); dev_err(chip->card->dev, "cannot get memory region\n"); + snd_echo_free(chip); return -EBUSY; } chip->dsp_registers = (volatile u32 __iomem *) @@ -1999,8 +1999,8 @@ static int snd_echo_create(struct snd_card *card, if (request_irq(pci->irq, snd_echo_interrupt, IRQF_SHARED, KBUILD_MODNAME, chip)) { - snd_echo_free(chip); dev_err(chip->card->dev, "cannot grab irq\n"); + snd_echo_free(chip); return -EBUSY; } chip->irq = pci->irq; @@ -2012,8 +2012,8 @@ static int snd_echo_create(struct snd_card *card, if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(chip->pci), sizeof(struct comm_page), &chip->commpage_dma_buf) < 0) { - snd_echo_free(chip); dev_err(chip->card->dev, "cannot allocate the comm page\n"); + snd_echo_free(chip); return -ENOMEM; } chip->comm_page_phys = chip->commpage_dma_buf.addr; @@ -2291,8 +2291,8 @@ static int snd_echo_resume(struct device *dev) if (request_irq(pci->irq, snd_echo_interrupt, IRQF_SHARED, KBUILD_MODNAME, chip)) { - snd_echo_free(chip); dev_err(chip->card->dev, "cannot grab irq\n"); + snd_echo_free(chip); return -EBUSY; } chip->irq = pci->irq;

3 3

Re: [alsa-devel] [GIT PULL v2] Enhancement of support for Firewire devices
by Takashi Iwai 05 Mar '14

05 Mar '14

At Wed, 05 Mar 2014 12:13:03 +0900, Takashi Sakamoto wrote: > > This pull-request is an update for my previous pull-request: > [alsa-devel] [GIT PULL][PATCH 00/39] Enhancement of support for Firewire > devices > http://mailman.alsa-project.org/pipermail/alsa-devel/2014-February/073446.h… > > Updates since previous pull-request: > firewire-lib: > - use swap() instead of local macro > - check NULL at memory allocation for sort table > fireworks: > - use bitmap.h for card slots > - add 'firewire' subdirectory for card node in procfs > - add sanity check of module parameter with clamp() > bebob: > - use bitmap.h for card slots > - add 'firewire' subdirectory for card node in procfs > - simplify codes for metering > - use swap() instead of local macro > - change order of last two commits Could you submit v2 patches to ML? The patches need review, and I want get them acked by Clemens before applying. thanks, Takashi > > ---------------------------------------------------------------- > The following changes since commit 31278997add61ee13b709e274934c7b0085accce: > > ALSA: hda/realtek - Add headset quirk for Dell DT (2014-03-03 10:48:55 > +0100) > > are available in the git repository at: > > https://github.com/takaswie/sound.git request > > for you to fetch changes up to 5b354e20ec3c64fa0a8fa7fbc7355201d881a8cc: > > bebob: Send a cue to load firmware for M-Audio Firewire series > (2014-03-05 12:00:32 +0900) > > ---------------------------------------------------------------- > Takashi Sakamoto (39): > firewire-lib: Rename functions, structure, member for AMDTP > firewire-lib: Add macros instead of fixed value for AMDTP > firewire-lib: Add 'direction' member to 'amdtp_stream' structure > firewire-lib: Split some codes into functions to reuse for both > streams > firewire-lib: Add support for AMDTP in-stream and PCM capture > firewire-lib: Add support for MIDI capture/playback > firewire-lib: Give syt value as parameter to handle_out_packet() > firewire-lib: Add support for duplex streams synchronization in > blocking mode > firewire-lib: Add sort function for transmitted packet > firewire-lib: Add transfer delay to synchronized duplex streams > firewire-lib: Add support for channel mapping > firewire-lib: Rename macros, variables and functions for CMP > firewire-lib: Add 'direction' member to 'cmp_connection' structure > firewire-lib: Add handling output connection by CMP > firewire-lib: Add a new function to check others' connection > firewire-lib: Add some AV/C general commands > firewire-lib: Add quirks for Fireworks > firewire-lib: Add a fallback at RCODE_CANCELLED > fireworks: Add skelton for Fireworks based devices > fireworks: Add transaction and some commands > fireworks: Add connection and stream management > fireworks: Add proc interface for debugging purpose > fireworks: Add MIDI interface > fireworks: Add PCM interface > fireworks: Add hwdep interface > fireworks: Add command/response functionality into hwdep interface > bebob: Add skelton for BeBoB based devices > bebob: Add commands and connections/streams management > bebob: Add proc interface for debugging purpose > bebob: Add MIDI interface > bebob: Add PCM interface > bebob: Add hwdep interface > bebob: Prepare for device specific operations > bebob: Add support for Terratec PHASE, EWS series and Aureon > bebob: Add support for Yamaha GO series > bebob: Add support for Focusrite Saffire/SaffirePro series > bebob: Add support for M-Audio usual Firewire series > bebob: Add support for M-Audio special Firewire series > bebob: Send a cue to load firmware for M-Audio Firewire series > > include/uapi/sound/asound.h | 4 +- > include/uapi/sound/firewire.h | 22 +- > sound/firewire/Kconfig | 59 ++ > sound/firewire/Makefile | 2 + > sound/firewire/amdtp.c | 891 > +++++++++++++++++------ > sound/firewire/amdtp.h | 175 ++++- > sound/firewire/bebob/Makefile | 4 + > sound/firewire/bebob/bebob.c | 450 ++++++++++++ > sound/firewire/bebob/bebob.h | 260 +++++++ > sound/firewire/bebob/bebob_command.c | 354 +++++++++ > sound/firewire/bebob/bebob_focusrite.c | 287 ++++++++ > sound/firewire/bebob/bebob_hwdep.c | 197 +++++ > sound/firewire/bebob/bebob_maudio.c | 828 > +++++++++++++++++++++ > sound/firewire/bebob/bebob_midi.c | 170 +++++ > sound/firewire/bebob/bebob_pcm.c | 451 ++++++++++++ > sound/firewire/bebob/bebob_proc.c | 201 +++++ > sound/firewire/bebob/bebob_stream.c | 869 > ++++++++++++++++++++++ > sound/firewire/bebob/bebob_terratec.c | 68 ++ > sound/firewire/bebob/bebob_yamaha.c | 50 ++ > sound/firewire/cmp.c | 229 ++++-- > sound/firewire/cmp.h | 14 +- > sound/firewire/dice.c | 46 +- > sound/firewire/fcp.c | 160 +++- > sound/firewire/fcp.h | 21 + > sound/firewire/fireworks/Makefile | 4 + > sound/firewire/fireworks/fireworks.c | 326 +++++++++ > sound/firewire/fireworks/fireworks.h | 223 ++++++ > sound/firewire/fireworks/fireworks_command.c | 397 ++++++++++ > sound/firewire/fireworks/fireworks_hwdep.c | 294 ++++++++ > sound/firewire/fireworks/fireworks_midi.c | 179 +++++ > sound/firewire/fireworks/fireworks_pcm.c | 477 ++++++++++++ > sound/firewire/fireworks/fireworks_proc.c | 233 ++++++ > sound/firewire/fireworks/fireworks_stream.c | 337 +++++++++ > sound/firewire/fireworks/fireworks_transaction.c | 327 +++++++++ > sound/firewire/lib.c | 6 +- > sound/firewire/lib.h | 1 + > sound/firewire/speakers.c | 93 +-- > 37 files changed, 8307 insertions(+), 402 deletions(-) > create mode 100644 sound/firewire/bebob/Makefile > create mode 100644 sound/firewire/bebob/bebob.c > create mode 100644 sound/firewire/bebob/bebob.h > create mode 100644 sound/firewire/bebob/bebob_command.c > create mode 100644 sound/firewire/bebob/bebob_focusrite.c > create mode 100644 sound/firewire/bebob/bebob_hwdep.c > create mode 100644 sound/firewire/bebob/bebob_maudio.c > create mode 100644 sound/firewire/bebob/bebob_midi.c > create mode 100644 sound/firewire/bebob/bebob_pcm.c > create mode 100644 sound/firewire/bebob/bebob_proc.c > create mode 100644 sound/firewire/bebob/bebob_stream.c > create mode 100644 sound/firewire/bebob/bebob_terratec.c > create mode 100644 sound/firewire/bebob/bebob_yamaha.c > create mode 100644 sound/firewire/fireworks/Makefile > create mode 100644 sound/firewire/fireworks/fireworks.c > create mode 100644 sound/firewire/fireworks/fireworks.h > create mode 100644 sound/firewire/fireworks/fireworks_command.c > create mode 100644 sound/firewire/fireworks/fireworks_hwdep.c > create mode 100644 sound/firewire/fireworks/fireworks_midi.c > create mode 100644 sound/firewire/fireworks/fireworks_pcm.c > create mode 100644 sound/firewire/fireworks/fireworks_proc.c > create mode 100644 sound/firewire/fireworks/fireworks_stream.c > create mode 100644 sound/firewire/fireworks/fireworks_transaction.c >

2 3

[alsa-devel] [PATCH] ALSA: emu10k1: Fix possible NULL dereference
by Takashi Iwai 05 Mar '14

05 Mar '14

The previous dev_err() conversion resulted in a code that may give NULL dereference in snd_emu10k1_ptr_write(). Since it's a sanity check, better to be replaced with a debug macro like other places in this driver. Fixes: 6f002b02166c ('ALSA: emu10k1: Use standard printk helpers') Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/emu10k1/io.c | 5 +---- 1 file changed, 1 insertion(+), 4 deletions(-) diff --git a/sound/pci/emu10k1/io.c b/sound/pci/emu10k1/io.c index 81c75b4528d9..706b4f0c6806 100644 --- a/sound/pci/emu10k1/io.c +++ b/sound/pci/emu10k1/io.c @@ -71,11 +71,8 @@ void snd_emu10k1_ptr_write(struct snd_emu10k1 *emu, unsigned int reg, unsigned i unsigned long flags; unsigned int mask; - if (!emu) { - dev_err(emu->card->dev, "ptr_write: emu is null!\n"); - dump_stack(); + if (snd_BUG_ON(!emu)) return; - } mask = emu->audigy ? A_PTR_ADDRESS_MASK : PTR_ADDRESS_MASK; regptr = ((reg << 16) & mask) | (chn & PTR_CHANNELNUM_MASK); -- 1.9.0

1 0

[alsa-devel] [patch] ALSA: lola: NULL deref on allocation error
by Dan Carpenter 05 Mar '14

05 Mar '14

"chip" is NULL here. We don't need a printk here because kmalloc() has it built in. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/pci/lola/lola.c b/sound/pci/lola/lola.c index d63434d8c3fb..68824cdd137d 100644 --- a/sound/pci/lola/lola.c +++ b/sound/pci/lola/lola.c @@ -586,7 +586,6 @@ static int lola_create(struct snd_card *card, struct pci_dev *pci, chip = kzalloc(sizeof(*chip), GFP_KERNEL); if (!chip) { - dev_err(chip->card->dev, "cannot allocate chip\n"); pci_disable_device(pci); return -ENOMEM; }

2 1

Re: [alsa-devel] ALSA: emu10k1: Use standard printk helpers
by Dan Carpenter 05 Mar '14

05 Mar '14

Hello Takashi Iwai, The patch 6f002b02166c: "ALSA: emu10k1: Use standard printk helpers" from Feb 25, 2014, leads to the following static checker warning: sound/pci/emu10k1/io.c:75 snd_emu10k1_ptr_write() error: we previously assumed 'emu' could be null (see line 74) sound/pci/emu10k1/io.c 68 void snd_emu10k1_ptr_write(struct snd_emu10k1 *emu, unsigned int reg, unsigned int chn, unsigned int data) 69 { 70 unsigned int regptr; 71 unsigned long flags; 72 unsigned int mask; 73 74 if (!emu) { ^^^^ 75 dev_err(emu->card->dev, "ptr_write: emu is null!\n"); ^^^^^^^^^^^^^^ Dereference. 76 dump_stack(); 77 return; 78 } regards, dan carpenter

2 1

[alsa-devel] [PATCH 1/2] ALSA: hda - Add missing loopback merge path for AD1884/1984 codecs
by Takashi Iwai 05 Mar '14

05 Mar '14

The mixer widget (NID 0x20) of AD1884 and AD1984 codecs isn't connected directly to the actual I/O paths but only via another mixer widget (NID 0x21). We need a similar fix as we did for AD1882. Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_analog.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index df3652ad15ef..498971095869 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -1092,6 +1092,7 @@ static int patch_ad1884(struct hda_codec *codec) spec = codec->spec; spec->gen.mixer_nid = 0x20; + spec->gen.mixer_merge_nid = 0x21; spec->gen.beep_nid = 0x10; set_beep_amp(spec, 0x10, 0, HDA_OUTPUT); -- 1.9.0

1 1

[alsa-devel] Xonar DG backports
by Clemens Ladisch 05 Mar '14

05 Mar '14

Hi Roman, I backporting the bugfixes in your patches to the stable kernels. However, I'm unable to see what was changed to enable the headphone outputs. What particular thing did the old driver wrong? Regards, Clemens

2 1

[alsa-devel] Audio-Bug
by Silver Death 05 Mar '14

05 Mar '14

Hi there, many people have this bug: https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/1200925 I would fix it, but I have no knowledge in C and about the kernel internal. I also didn't found any documentation which I could use to translate the datasheet into the source file. So I gentle ask you, if you could as a maintainer fix this bug or at least give me some information/instruction/documentation. Thanks in advance. With best regards, Simon

1 0

Re: [alsa-devel] Question on compress offload framework memory corruption
by Banajit Goswami 05 Mar '14

05 Mar '14

Hi Liam, > On Thu, Dec 19, 2013 at 10:55:08AM +0000, Mark Brown wrote: > > On Wed, Dec 18, 2013 at 01:17:51PM -0000, gsantosh(a)codeaurora.org wrote: > > > Hi, > > > > > > I have following questions in the compressed offload framework API. > > > > > > static int soc_compr_set_params_fe(struct snd_compr_stream *cstream, > > > struct snd_compr_params *params) { ... > > > hw_params = kzalloc(sizeof(*hw_params), GFP_KERNEL); > > > if (hw_params == NULL) > > > return -ENOMEM; > > > /*1st question is, what is the use of above allocated memory I do > > > not see this being used in this function*/ > > > > The code you're quoting there isn't in mainline. I've also added > > Vinod to the CCs, you probably want to include him in any discussion > > of the compressed API. > > That's right, this is from DPCM compressed updated done by Liam, so he is > the best person to comment on this bit, and somehow his address in CC has > gone bad, i have added him in to list > But yes looking at the source, i agree that two are interlinked. > I guess we need to poupulate the hw_params and then use that to copy, that > would be the right thing to do here and yes this is not correct and not > sure how this is working... Continuing the discussion on the "memcpy" function in Santosh's 2nd question- > > > memcpy(&fe->dpcm[fe_substream->stream].hw_params, params, > > > sizeof(struct snd_pcm_hw_params)); > > > > > > /* 2nd question is > > > in above memcpy there is parameter mismatch > > > &fe->dpcm[fe_substream->stream].hw_params is of structure type struct > > > snd_pcm_hw_params > > > > > > params argument is of the structure type struct snd_compr_params > > > the definition of the two structures are different, how this is > > > working? > > > this will over right the destination with junk value which will return > > > in error when this memory is accessed, after memcpy this cpu_dai > > > hw_params > > > returing with EINVAL error, please explain why this is done this way. > > > */ I can see that a later version of the patch adds the function soc_compr_set_params_fe() at- http://permalink.gmane.org/gmane.linux.alsa.devel/117716 with the below code replaces the "memcpy" function with a "memset"- + /* + * Create an empty hw_params for the BE as the machine driver must + * fix this up to match DSP decoder and ASRC configuration. + * I.e. machine driver fixup for compressed BE is mandatory. + */ + memset(&fe->dpcm[fe_substream->stream].hw_params, 0, + sizeof(struct snd_pcm_hw_params)); It's evident that the code relies on machine driver fixup function to populate the parameters of hw_params before propagating those to back-end. I think rather than that, like Vinod also mentioned, the hw_params should have been populated with individual parameters which are relevant. Let me know your comment on this. Thanks, Banajit The Qualcomm Innovation Center, Inc. is a member of Code Aurora Forum, hosted by The Linux Foundation

2 1

[alsa-devel] Update patch for codecs alias name for Dell
by Kailang 05 Mar '14

05 Mar '14

Hi Takashi, Dell assigned more codecs for alias name. I update in attach patch. Thanks. Kailang

3 7

[alsa-devel] [PATCH v5 0/7] ASoC: add CSR SiRFSoC sound drivers
by RongJun Ying 05 Mar '14

05 Mar '14

From: Rongjun Ying <rongjun.ying(a)csr.com> This patchset adds CSR SiRFSoC sound drivers including: 1. The SiRF internal audio codec driver 2. The SiRF audio port driver 3. The SiRF internal audio card driver 4. The SiRF I2S controller driver 5. The SiRF hdmi card driver 6. The SiRF USP controller(DSP mode) driver 7. The SiRF bt-sco card driver -v5: 1. Drop all inlines. 2. Reordering the Kconfig and Makefile 3. Remove the sirf_audio_codec_reg_bits struct, use the new controls instead it. 4. Add some SND_SOC_DAPM_OUT_DRV instead of HP and SPK enable driver 5. Add audio codec clock supply instead of adc event callback 6. Fixed playback and capture can't concurrent work bug. Rongjun Ying (7): ASoC: sirf: Add SiRF internal audio codec driver ASoC: sirf: Add SiRF audio port driver is used by SiRF internal audio codec ASoC: sirf: Add SiRF audio card ASoC: sirf: Add SiRF I2S driver ASoC: sirf: Add hdmi card ASoC: sirf: Add usp driver which is used by dsp mode ASoC: sirf: Add bt-sco card .../devicetree/bindings/sound/sirf-audio-codec.txt | 17 + .../devicetree/bindings/sound/sirf-audio-port.txt | 20 + .../devicetree/bindings/sound/sirf-audio.txt | 41 ++ .../devicetree/bindings/sound/sirf-bt-sco.txt | 18 + .../devicetree/bindings/sound/sirf-hdmi.txt | 15 + .../devicetree/bindings/sound/sirf-i2s.txt | 27 + .../devicetree/bindings/sound/sirf-usp.txt | 27 + sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/codecs/Kconfig | 6 +- sound/soc/codecs/Makefile | 2 + sound/soc/codecs/sirf-audio-codec.c | 534 ++++++++++++++++++++ sound/soc/codecs/sirf-audio-codec.h | 75 +++ sound/soc/sirf/Kconfig | 34 ++ sound/soc/sirf/Makefile | 13 + sound/soc/sirf/sirf-audio-port.c | 194 +++++++ sound/soc/sirf/sirf-audio-port.h | 62 +++ sound/soc/sirf/sirf-audio.c | 156 ++++++ sound/soc/sirf/sirf-bt-sco.c | 99 ++++ sound/soc/sirf/sirf-hdmi.c | 98 ++++ sound/soc/sirf/sirf-i2s.c | 434 ++++++++++++++++ sound/soc/sirf/sirf-i2s.h | 88 ++++ sound/soc/sirf/sirf-usp.c | 448 ++++++++++++++++ sound/soc/sirf/sirf-usp.h | 282 +++++++++++ 24 files changed, 2691 insertions(+), 1 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/sirf-audio-codec.txt create mode 100644 Documentation/devicetree/bindings/sound/sirf-audio-port.txt create mode 100644 Documentation/devicetree/bindings/sound/sirf-audio.txt create mode 100644 Documentation/devicetree/bindings/sound/sirf-bt-sco.txt create mode 100644 Documentation/devicetree/bindings/sound/sirf-hdmi.txt create mode 100644 Documentation/devicetree/bindings/sound/sirf-i2s.txt create mode 100644 Documentation/devicetree/bindings/sound/sirf-usp.txt create mode 100644 sound/soc/codecs/sirf-audio-codec.c create mode 100644 sound/soc/codecs/sirf-audio-codec.h create mode 100644 sound/soc/sirf/Kconfig create mode 100644 sound/soc/sirf/Makefile create mode 100644 sound/soc/sirf/sirf-audio-port.c create mode 100644 sound/soc/sirf/sirf-audio-port.h create mode 100644 sound/soc/sirf/sirf-audio.c create mode 100644 sound/soc/sirf/sirf-bt-sco.c create mode 100644 sound/soc/sirf/sirf-hdmi.c create mode 100644 sound/soc/sirf/sirf-i2s.c create mode 100644 sound/soc/sirf/sirf-i2s.h create mode 100644 sound/soc/sirf/sirf-usp.c create mode 100644 sound/soc/sirf/sirf-usp.h -- 1.7.5.4

2 8

[alsa-devel] [PATCH 00/15] ASoC: fsl-ssi: Driver cleanup
by Markus Pargmann 05 Mar '14

05 Mar '14

Hi, Here is a series of cleanups for the fsl-ssi driver. It would be great if someone with PPC hardware and this fsl-ssi unit could please test this series. I am not sure if the regmap patch works as expected on PPC as well. I tested this series on imx53 quick starter board and a custom board. Both with SSI as I2S slave. Regards, Markus Markus Pargmann (15): ASoC: fsl-ssi: Use regmap ASoC: fsl-ssi: Remove fsl_ssi_setup ASoC: fsl-ssi: Move debugging to seperate file ASoC: fsl-ssi: Use dev_name for DAI driver struct ASoC: fsl-ssi: Move imx-specific probe to seperate function ASoC: fsl-ssi: Remove useless DMA code ASoC: fsl-ssi: baud clock error handling ASoC: fsl-ssi: Cleanup probe function ASoC: fsl-ssi: Remove unnecessary variables from ssi_private ASoC: fsl-ssi: reorder and document fsl_ssi_private ASoC: fsl-ssi: Fix register values when disabling ASoC: fsl-ssi: Only enable baudclk when used ASoC: fsl-ssi: Set default dai-fmts ASoC: fsl-ssi: Transmit enable synchronization ASoC: fsl-ssi: Update binding documentation .../devicetree/bindings/sound/fsl,ssi.txt | 21 +- sound/soc/fsl/Kconfig | 1 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/fsl_ssi.c | 1167 ++++++++------------ sound/soc/fsl/fsl_ssi.h | 112 +- sound/soc/fsl/fsl_ssi_dbg.c | 163 +++ 6 files changed, 750 insertions(+), 716 deletions(-) create mode 100644 sound/soc/fsl/fsl_ssi_dbg.c -- 1.8.5.3

4 21

[alsa-devel] [PATCH 00/10] Simplify the code of TDM slot setting
by Xiubo Li 05 Mar '14

05 Mar '14

Xiubo Li (10): ASoC: core: Add snd_soc_dai_set_tdm_slot_xlate(). ASoC: fsl-utils: Add fsl_asoc_of_xlate_tdm_slot_mask() support. ASoC: fsl-esai: Add .of_xlate_tdm_slot_mask() support. ASoC: fsl-ssi: Add .of_xlate_tdm_slot_mask() support. ASoC: imx-ssi: Add .of_xlate_tdm_slot_mask() support. ASoC: simple-card: Use snd_soc_dai_set_tdm_slot_xlate() ASoC: blackfin: bf5xx-ad1836: Use snd_soc_dai_set_tdm_slot_xlate() ASoC: blackfin: bf5xx-ad193x: Use snd_soc_dai_set_tdm_slot_xlate() ASoC: fsl: eukrea-tlv320: Use snd_soc_dai_set_tdm_slot_xlate() ASoC: fsl: wm1133-ev1: Use snd_soc_dai_set_tdm_slot_xlate() include/sound/soc-dai.h | 3 +++ sound/soc/blackfin/bf5xx-ad1836.c | 2 +- sound/soc/blackfin/bf5xx-ad193x.c | 4 ++-- sound/soc/fsl/Kconfig | 3 +++ sound/soc/fsl/eukrea-tlv320.c | 2 +- sound/soc/fsl/fsl_esai.c | 2 ++ sound/soc/fsl/fsl_ssi.c | 2 ++ sound/soc/fsl/fsl_utils.c | 27 +++++++++++++++++++++++++++ sound/soc/fsl/fsl_utils.h | 4 +++- sound/soc/fsl/imx-ssi.c | 2 ++ sound/soc/fsl/wm1133-ev1.c | 11 +---------- sound/soc/generic/simple-card.c | 7 +++---- sound/soc/soc-core.c | 33 ++++++++++++++++++++++++++++----- 13 files changed, 78 insertions(+), 24 deletions(-) -- 1.8.4

4 14

[alsa-devel] [PATCH] ASoC: simple-card: tidyup cpu/codec dai_fmt settings for non-DT
by Kuninori Morimoto 05 Mar '14

05 Mar '14

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> 30d0341e7da0c012f64fb290dd1153360fb49a8d (ASoC: simple-card: simplify the daifmt code) simplify cpu/codec dai_fmt which consists from dai specific format + common format. But, it didn't care about non-DT case. This patch fixes it Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/generic/simple-card.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 034a2b7..7cabcc5 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -307,6 +307,9 @@ static int asoc_simple_card_probe(struct platform_device *pdev) 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; } /* -- 1.7.9.5

2 1

[alsa-devel] [PATCH v4-resend 0/7] ASoC: add CSR SiRFSoC sound drivers
by RongJun Ying 05 Mar '14

05 Mar '14

From: Rongjun Ying <rongjun.ying(a)csr.com> This patchset adds CSR SiRFSoC sound drivers including: 1. The SiRF internal audio codec driver 2. The SiRF audio port driver 3. The SiRF internal audio card driver 4. The SiRF I2S controller driver 5. The SiRF hdmi card driver 6. The SiRF USP controller(DSP mode) driver 7. The SiRF bt-sco card driver -v4: 1. Use the generic dmaengine pcm codes instead of sirf pcm driver. 2. Add TLV information into internal audio codec driver. 3. Use the MMIO regmap instead of custom read and write callbacks. 4. Remove spin_lock. 5. Change some variable names. 6. The SiRF internal audio driver split into codec, port. 7. Add binding documents. Rongjun Ying (7): ASoC: sirf: Add SiRF internal audio codec driver ASoC: sirf: Add SiRF audio port driver is used by SiRF internal audio codec ASoC: sirf: Add SiRF audio card ASoC: sirf: Add SiRF I2S driver ASoC: sirf: Add hdmi card ASoC: sirf: Add usp driver which is used by dsp mode ASoC: sirf: Add bt-sco card .../devicetree/bindings/sound/sirf-audio-codec.txt | 17 + .../devicetree/bindings/sound/sirf-audio-port.txt | 20 + .../devicetree/bindings/sound/sirf-audio.txt | 41 ++ .../devicetree/bindings/sound/sirf-bt-sco.txt | 18 + .../devicetree/bindings/sound/sirf-hdmi.txt | 15 + .../devicetree/bindings/sound/sirf-i2s.txt | 27 + .../devicetree/bindings/sound/sirf-usp.txt | 27 + sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/sirf-audio-codec.c | 585 ++++++++++++++++++++ sound/soc/codecs/sirf-audio-codec.h | 75 +++ sound/soc/sirf/Kconfig | 34 ++ sound/soc/sirf/Makefile | 13 + sound/soc/sirf/sirf-audio-port.c | 194 +++++++ sound/soc/sirf/sirf-audio-port.h | 62 ++ sound/soc/sirf/sirf-audio.c | 156 ++++++ sound/soc/sirf/sirf-bt-sco.c | 99 ++++ sound/soc/sirf/sirf-hdmi.c | 98 ++++ sound/soc/sirf/sirf-i2s.c | 438 +++++++++++++++ sound/soc/sirf/sirf-i2s.h | 88 +++ sound/soc/sirf/sirf-usp.c | 448 +++++++++++++++ sound/soc/sirf/sirf-usp.h | 282 ++++++++++ 24 files changed, 2745 insertions(+), 0 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/sirf-audio-codec.txt create mode 100644 Documentation/devicetree/bindings/sound/sirf-audio-port.txt create mode 100644 Documentation/devicetree/bindings/sound/sirf-audio.txt create mode 100644 Documentation/devicetree/bindings/sound/sirf-bt-sco.txt create mode 100644 Documentation/devicetree/bindings/sound/sirf-hdmi.txt create mode 100644 Documentation/devicetree/bindings/sound/sirf-i2s.txt create mode 100644 Documentation/devicetree/bindings/sound/sirf-usp.txt create mode 100644 sound/soc/codecs/sirf-audio-codec.c create mode 100644 sound/soc/codecs/sirf-audio-codec.h create mode 100644 sound/soc/sirf/Kconfig create mode 100644 sound/soc/sirf/Makefile create mode 100644 sound/soc/sirf/sirf-audio-port.c create mode 100644 sound/soc/sirf/sirf-audio-port.h create mode 100644 sound/soc/sirf/sirf-audio.c create mode 100644 sound/soc/sirf/sirf-bt-sco.c create mode 100644 sound/soc/sirf/sirf-hdmi.c create mode 100644 sound/soc/sirf/sirf-i2s.c create mode 100644 sound/soc/sirf/sirf-i2s.h create mode 100644 sound/soc/sirf/sirf-usp.c create mode 100644 sound/soc/sirf/sirf-usp.h -- 1.7.5.4

4 18

[alsa-devel] [PATCH] ASoC: pcm: free path list before exiting from error conditions
by Patrick Lai 05 Mar '14

05 Mar '14

dpcm_path_get() allocates dynamic memory to hold path list. Corresponding dpcm_path_put() must be called to free the memory. dpcm_path_put() is not called under several error conditions. This leads to memory leak. Signed-off-by: Patrick Lai <plai(a)codeaurora.org> --- sound/soc/soc-pcm.c | 3 +++ 1 files changed, 3 insertions(+), 0 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 47e1ce7..28522bd 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1989,6 +1989,7 @@ 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); @@ -2018,6 +2019,7 @@ 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); @@ -2082,6 +2084,7 @@ 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"); } -- The Qualcomm Innovation Center, Inc. is a member of the Code Aurora Forum, hosted by The Linux Foundation

2 1

[alsa-devel] [RFC][PATCH 0/3] Remove set_cache_io entirely from ASoC probe
by Xiubo Li 05 Mar '14

05 Mar '14

Xiubo Li (3): ASoC: core: Move the default regmap I/O setting to snd_soc_register_codec() ASoc: codec: remove the set_cache_io() entirely from CODEC ASoC probe. ASoC: core: remove the default regmap I/O setting up from soc_probe_codec() include/sound/soc.h | 3 +++ sound/soc/codecs/cq93vc.c | 6 ++++-- sound/soc/codecs/mc13783.c | 10 ++-------- sound/soc/codecs/wm5102.c | 6 ++---- sound/soc/codecs/wm5110.c | 6 ++---- sound/soc/codecs/wm8350.c | 6 ++++-- sound/soc/codecs/wm8400.c | 6 ++++-- sound/soc/codecs/wm8994.c | 4 ++-- sound/soc/codecs/wm8997.c | 7 ++----- sound/soc/soc-core.c | 21 +++++++++++---------- 10 files changed, 36 insertions(+), 39 deletions(-) -- 1.8.4

3 5

[alsa-devel] [PATCH] ASoC: ak4104: Remove superfluous codec->control_data initialization
by Lars-Peter Clausen 05 Mar '14

05 Mar '14

The driver uses automatic IO setup, which will also initialize the control_data field of the CODEC, no need to do it manually. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/ak4104.c | 2 -- 1 file changed, 2 deletions(-) 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, -- 1.8.0

3 2

[alsa-devel] [PATCH] ASoC: io: Remove hw_read() operation
by Mark Brown 05 Mar '14

05 Mar '14

From: Mark Brown <broonie(a)linaro.org> We now no longer have any users of hw_read() in the kernel so remove the code in order to prevent any new users being added. Users should be using regmap. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- include/sound/soc.h | 1 - sound/soc/soc-io.c | 13 ------------- 2 files changed, 14 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 56c4c71..dc36331 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -706,7 +706,6 @@ struct snd_soc_codec { /* codec IO */ void *control_data; /* codec control (i2c/3wire) data */ hw_write_t hw_write; - unsigned int (*hw_read)(struct snd_soc_codec *, unsigned int); unsigned int (*read)(struct snd_soc_codec *, unsigned int); int (*write)(struct snd_soc_codec *, unsigned int, unsigned int); void *reg_cache; diff --git a/sound/soc/soc-io.c b/sound/soc/soc-io.c index c3b6a0a..3c1de9c 100644 --- a/sound/soc/soc-io.c +++ b/sound/soc/soc-io.c @@ -26,18 +26,6 @@ static int hw_write(struct snd_soc_codec *codec, unsigned int reg, return regmap_write(codec->control_data, reg, value); } -static unsigned int hw_read(struct snd_soc_codec *codec, unsigned int reg) -{ - int ret; - unsigned int val; - - ret = regmap_read(codec->control_data, reg, &val); - if (ret == 0) - return val; - else - return -1; -} - /** * snd_soc_codec_set_cache_io: Set up standard I/O functions. * @@ -64,7 +52,6 @@ int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec, int ret; codec->write = hw_write; - codec->read = hw_read; switch (control) { case SND_SOC_REGMAP: -- 1.9.0

2 6

[alsa-devel] [PATCH] ASoC: fsl: Don't select FIQ from Eukrea
by Mark Brown 05 Mar '14

05 Mar '14

From: Mark Brown <broonie(a)linaro.org> There is no point in using FIQ if DMA is available (it is selected) and selecting FIQ currently breaks the build on non-i.MX platforms. If FIQ is actually required the build will need to be restricted or have a select for the relevant FIQ code adding. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/fsl/Kconfig | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index a7d6ab6..b244830 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -175,7 +175,6 @@ config SND_SOC_EUKREA_TLV320 || (OF && ARM) depends on I2C select SND_SOC_TLV320AIC23 - select SND_SOC_IMX_PCM_FIQ select SND_SOC_IMX_AUDMUX select SND_SOC_IMX_SSI select SND_SOC_FSL_SSI -- 1.9.0

1 0

[alsa-devel] [PATCH] ASoC: rsnd: Fix warnings due to improper printk formats
by Laurent Pinchart 04 Mar '14

04 Mar '14

Use the %pap printk specifier to print resource_size_t variables. This fixes warnings on platforms where resource_size_t has a different size than int. Signed-off-by: Laurent Pinchart <laurent.pinchart+renesas(a)ideasonboard.com> --- sound/soc/sh/rcar/gen.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c index add088b..f505b1f 100644 --- a/sound/soc/sh/rcar/gen.c +++ b/sound/soc/sh/rcar/gen.c @@ -283,13 +283,13 @@ 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, "SRU : %pap => %p\n", &scu_res->start, gen->base[RSND_GEN2_SCU]); - dev_dbg(dev, "ADG : %08x => %p\n", adg_res->start, + dev_dbg(dev, "ADG : %pap => %p\n", &adg_res->start, gen->base[RSND_GEN2_ADG]); - dev_dbg(dev, "SSIU : %08x => %p\n", ssiu_res->start, + dev_dbg(dev, "SSIU : %pap => %p\n", &ssiu_res->start, gen->base[RSND_GEN2_SSIU]); - dev_dbg(dev, "SSI : %08x => %p\n", ssi_res->start, + dev_dbg(dev, "SSI : %pap => %p\n", &ssi_res->start, gen->base[RSND_GEN2_SSI]); return 0; -- Regards, Laurent Pinchart

1 0

[alsa-devel] SNDRV_PCM_INFO_BATCH
by David Henningsson 04 Mar '14

04 Mar '14

So, PulseAudio removed timer scheduling for drivers with SNDRV_PCM_INFO_BATCH, and released 5.0. Meanwhile, USB drivers still have SNDRV_PCM_INFO_BATCH, so this leads to performance regressions for USB sound cards. The question remains - what does SNDRV_PCM_INFO_BATCH really mean? In the documentation, we have this: "int snd_pcm_hw_params_is_batch (const snd_pcm_hw_params_t *params) Check, if hardware does double buffering for data transfers for given configuration." What PulseAudio was looking for was more something like "what accuracy does the pointer callback give us". When we asked this question at the latest audio meeting, SNDRV_PCM_INFO_BATCH was given as the answer, i e, if this flag is set, pointer callback have no more accuracy than period boundaries. At least that's how I understood it. So, either the documentation for snd_pcm_hw_params_is_batch is correct, in which case PulseAudio should revert the tsched patch. Or the answer given at the audio meeting was correct, in which case we need to fix the USB driver to stop sending the SNDRV_PCM_INFO_BATCH flag. -- David Henningsson, Canonical Ltd. https://launchpad.net/~diwic

3 2

[alsa-devel] [GIT PULL][PATCH 00/39] Enhancement of support for Firewire devices
by Takashi Sakamoto 04 Mar '14

04 Mar '14

This is a pull-request of my 39 patches to enhance support for Firewire devices. These patches extend firewire-lib, add new drivers (fireworks/bebob). These drivers can handle playback/capture of PCM samples/MIDI messages and can support 70-80 models totally. These drivers are designed not to interrupt FFADO (user-land driver project) functionality except for playbacking/capturing. These drivers basically give no way to control device's internal mixer. For this purpose, users need to use applications such like ffado-dbus-server/ffado-mixer, or develop software to execute AV/C commands and device specific commands. Updates since RFC v3: [alsa-devel] [RFC v3] [PATCH 00/52] Enhancement for support of firewire devices http://mailman.alsa-project.org/pipermail/alsa-devel/2014-January/071820.ht… firewire-lib: - improve operation for oPCR - add fallbacks at transaction timeout for device's quirk fireworks: - add counter to maintain streams - use memdup_user() for hwdep interface bebob: - add counter to maintain streams - fix failure to handle bus reset ---------------------------------------------------------------- The following changes since commit dde9c38779c1f30253485a160facc211b20ca222: Merge branch 'for-linus' (2014-02-27 12:46:40 +0100) are available in the git repository at: https://github.com/takaswie/sound.git request for you to fetch changes up to 0ed07c65fef2f80fcfc6223b0d689fca56fa6323: bebob: Add support for M-Audio special Firewire series (2014-02-28 11:30:52 +0900) ---------------------------------------------------------------- Takashi Sakamoto (39): firewire-lib: Rename functions, structure, member for AMDTP firewire-lib: Add macros instead of fixed value for AMDTP firewire-lib: Add 'direction' member to 'amdtp_stream' structure firewire-lib: Split some codes into functions to reuse for both streams firewire-lib: Add support for AMDTP in-stream and PCM capture firewire-lib: Add support for MIDI capture/playback firewire-lib: Give syt value as parameter to handle_out_packet() firewire-lib: Add support for duplex streams synchronization in blocking mode firewire-lib: Add sort function for transmitted packet firewire-lib: Add transfer delay to synchronized duplex streams firewire-lib: Add support for channel mapping firewire-lib: Rename macros, variables and functions for CMP firewire-lib: Add 'direction' member to 'cmp_connection' structure firewire-lib: Add handling output connection by CMP firewire-lib: Add a new function to check others' connection firewire-lib: Add some AV/C general commands firewire-lib: Add quirks for Fireworks firewire-lib: Add a fallback at RCODE_CANCELLED fireworks: Add skelton for Fireworks based devices fireworks: Add transaction and some commands fireworks: Add connection and stream management fireworks: Add proc interface for debugging purpose fireworks: Add MIDI interface fireworks: Add PCM interface fireworks: Add hwdep interface fireworks: Add command/response functionality into hwdep interface bebob: Add skelton for BeBoB based devices bebob: Add commands and connections/streams management bebob: Add proc interface for debugging purpose bebob: Add MIDI interface bebob: Add PCM interface bebob: Add hwdep interface bebob: Prepare for device specific operations bebob: Add support for Terratec PHASE, EWS series and Aureon bebob: Add support for Yamaha GO series bebob: Add support for Focusrite Saffire/SaffirePro series bebob: Add support for M-Audio usual Firewire series bebob: Send a cue to load firmware for M-Audio Firewire series bebob: Add support for M-Audio special Firewire series include/uapi/sound/asound.h | 4 +- include/uapi/sound/firewire.h | 22 +- sound/firewire/Kconfig | 59 ++ sound/firewire/Makefile | 2 + sound/firewire/amdtp.c | 891 ++++++++++++++++------ sound/firewire/amdtp.h | 175 ++++- sound/firewire/bebob/Makefile | 4 + sound/firewire/bebob/bebob.c | 450 +++++++++++ sound/firewire/bebob/bebob.h | 262 +++++++ sound/firewire/bebob/bebob_command.c | 354 +++++++++ sound/firewire/bebob/bebob_focusrite.c | 289 +++++++ sound/firewire/bebob/bebob_hwdep.c | 197 +++++ sound/firewire/bebob/bebob_maudio.c | 922 +++++++++++++++++++++++ sound/firewire/bebob/bebob_midi.c | 170 +++++ sound/firewire/bebob/bebob_pcm.c | 451 +++++++++++ sound/firewire/bebob/bebob_proc.c | 179 +++++ sound/firewire/bebob/bebob_stream.c | 869 +++++++++++++++++++++ sound/firewire/bebob/bebob_terratec.c | 68 ++ sound/firewire/bebob/bebob_yamaha.c | 50 ++ sound/firewire/cmp.c | 229 ++++-- sound/firewire/cmp.h | 14 +- sound/firewire/dice.c | 46 +- sound/firewire/fcp.c | 160 +++- sound/firewire/fcp.h | 21 + sound/firewire/fireworks/Makefile | 4 + sound/firewire/fireworks/fireworks.c | 324 ++++++++ sound/firewire/fireworks/fireworks.h | 223 ++++++ sound/firewire/fireworks/fireworks_command.c | 397 ++++++++++ sound/firewire/fireworks/fireworks_hwdep.c | 294 ++++++++ sound/firewire/fireworks/fireworks_midi.c | 179 +++++ sound/firewire/fireworks/fireworks_pcm.c | 477 ++++++++++++ sound/firewire/fireworks/fireworks_proc.c | 208 +++++ sound/firewire/fireworks/fireworks_stream.c | 337 +++++++++ sound/firewire/fireworks/fireworks_transaction.c | 327 ++++++++ sound/firewire/lib.c | 6 +- sound/firewire/lib.h | 1 + sound/firewire/speakers.c | 93 +-- 37 files changed, 8356 insertions(+), 402 deletions(-) create mode 100644 sound/firewire/bebob/Makefile create mode 100644 sound/firewire/bebob/bebob.c create mode 100644 sound/firewire/bebob/bebob.h create mode 100644 sound/firewire/bebob/bebob_command.c create mode 100644 sound/firewire/bebob/bebob_focusrite.c create mode 100644 sound/firewire/bebob/bebob_hwdep.c create mode 100644 sound/firewire/bebob/bebob_maudio.c create mode 100644 sound/firewire/bebob/bebob_midi.c create mode 100644 sound/firewire/bebob/bebob_pcm.c create mode 100644 sound/firewire/bebob/bebob_proc.c create mode 100644 sound/firewire/bebob/bebob_stream.c create mode 100644 sound/firewire/bebob/bebob_terratec.c create mode 100644 sound/firewire/bebob/bebob_yamaha.c create mode 100644 sound/firewire/fireworks/Makefile create mode 100644 sound/firewire/fireworks/fireworks.c create mode 100644 sound/firewire/fireworks/fireworks.h create mode 100644 sound/firewire/fireworks/fireworks_command.c create mode 100644 sound/firewire/fireworks/fireworks_hwdep.c create mode 100644 sound/firewire/fireworks/fireworks_midi.c create mode 100644 sound/firewire/fireworks/fireworks_pcm.c create mode 100644 sound/firewire/fireworks/fireworks_proc.c create mode 100644 sound/firewire/fireworks/fireworks_stream.c create mode 100644 sound/firewire/fireworks/fireworks_transaction.c -- 1.8.3.2

4 67

[alsa-devel] [PATCH RFC 0/5] AM43xx-ePOS-EVM audio support with TLV320AIC31XX driver
by Jyri Sarha 04 Mar '14

04 Mar '14

This patch set adds audio support for AM43xx-ePOS-EVM. I'll mail the dts and defconfig changes once these have been merged. Jyri Sarha (5): ASoC: tlv320aic31xx: Add basic codec driver implementation ASoC: tlv320aic31xx: Add codec driver to Makefile and Kconfig ASoC: tlv320aic31xx: Add DT binding document ASoC: davinci-evm: Add AM43xx-EPOS-EVM audio support ASoC: davinci: Add SND_AM43XX_SOC_EPOS_EVM build option .../bindings/sound/davinci-evm-audio.txt | 9 +- .../devicetree/bindings/sound/tlv320aic31xx.txt | 57 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tlv320aic31xx.c | 1360 ++++++++++++++++++++ sound/soc/codecs/tlv320aic31xx.h | 265 ++++ sound/soc/davinci/Kconfig | 12 + sound/soc/davinci/Makefile | 1 + sound/soc/davinci/davinci-evm.c | 41 + 9 files changed, 1748 insertions(+), 3 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/tlv320aic31xx.txt create mode 100644 sound/soc/codecs/tlv320aic31xx.c create mode 100644 sound/soc/codecs/tlv320aic31xx.h -- 1.7.9.5

3 10

[alsa-devel] [PATCH] Wrap hw_ptr to boundary in pcm_ioplug
by MONTANARO Luciano (MM) 04 Mar '14

04 Mar '14

The function snd_pcm_ioplug_hw_ptr_update() always increased the hw_ptr by delta, without wrapping it to the boundary. This would lead to problems when after many hours, the hw_ptr would overflow. --- src/pcm/pcm_ioplug.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/src/pcm/pcm_ioplug.c b/src/pcm/pcm_ioplug.c index a90c844..c1c3a98 100644 --- a/src/pcm/pcm_ioplug.c +++ b/src/pcm/pcm_ioplug.c @@ -60,7 +60,7 @@ static void snd_pcm_ioplug_hw_ptr_update(snd_pcm_t *pcm) delta = hw - io->last_hw; else delta = pcm->buffer_size + hw - io->last_hw; - io->data->hw_ptr += delta; + snd_pcm_mmap_hw_forward(io->data->pcm, delta); io->last_hw = hw; } else io->data->state = SNDRV_PCM_STATE_XRUN; -- 1.7.0.4 -- Luciano Montanaro EntryNAV - Audio Software Integrator Magneti Marelli Electronic Systems Viale Carlo Emanuele II 150, Venaria Reale (To) Tel. (+39) 011 6879765 VISITA IL NOSTRO NUOVO SITO WEB! - VISIT OUR NEW WEB SITE! www.magnetimarelli.com Confidential Notice: This message - including its attachments - may contain proprietary, confidential and/or legally protected information and is intended solely for the use of the designated addressee(s) above. If you are not the intended recipient be aware that any downloading, copying, disclosure, distribution or use of the contents of the above information is strictly prohibited. If you have received this communication by mistake, please forward the message back to the sender at the email address above, delete the message from all mailboxes and any other electronic storage medium and destroy all copies. Disclaimer Notice: Internet communications cannot be guaranteed to be safe or error-free. Therefore we do not assure that this message is complete or accurate and we do not accept liability for any errors or omissions in the contents of this message.

2 1

[alsa-devel] Fix for busy loop at boundary wrapping with ioplug based plugins
by MONTANARO Luciano (MM) 04 Mar '14

04 Mar '14

Hello everybody, I have found an odd problem during long testing of our device, that suddenly starts misbehaving after several hours of use. What happened is: A client opens a device based on the jack plugin for reading, and keeps reading from it. After 10 hours 55 minutes, the snd_pcm_readi() functions starts to report data to be always available for read, and makes the reader use all the cpu it can. My patch will follow shortly, please tell me what you think of it. Best regards, Luciano -- Luciano Montanaro EntryNAV - Audio Software Integrator Magneti Marelli Electronic Systems Viale Carlo Emanuele II 150, Venaria Reale (To) Tel. (+39) 011 6879765 VISITA IL NOSTRO NUOVO SITO WEB! - VISIT OUR NEW WEB SITE! www.magnetimarelli.com Confidential Notice: This message - including its attachments - may contain proprietary, confidential and/or legally protected information and is intended solely for the use of the designated addressee(s) above. If you are not the intended recipient be aware that any downloading, copying, disclosure, distribution or use of the contents of the above information is strictly prohibited. If you have received this communication by mistake, please forward the message back to the sender at the email address above, delete the message from all mailboxes and any other electronic storage medium and destroy all copies. Disclaimer Notice: Internet communications cannot be guaranteed to be safe or error-free. Therefore we do not assure that this message is complete or accurate and we do not accept liability for any errors or omissions in the contents of this message.

1 0

[alsa-devel] [PATCH 2/4] ARM: dove: prepare new Dove DT Kconfig variable
by Sebastian Hesselbarth 04 Mar '14

04 Mar '14

DT-enabled Dove will move over from ARCH_DOVE in mach-dove to MACH_DOVE in mach-mvebu. As non-DT ARCH_DOVE will stay to rot for a while, add a new DT-only MACH_DOVE to mmc, thermal, wdt, and asoc Kconfigs. Signed-off-by: Sebastian Hesselbarth <sebastian.hesselbarth(a)gmail.com> --- Cc: Chris Ball <chris(a)printf.net> Cc: Zhang Rui <rui.zhang(a)intel.com> Cc: Eduardo Valentin <eduardo.valentin(a)ti.com> Cc: Wim Van Sebroeck <wim(a)iguana.be> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jason Cooper <jason(a)lakedaemon.net> Cc: Andrew Lunn <andrew(a)lunn.ch> Cc: Gregory Clement <gregory.clement(a)free-electrons.com> Cc: alsa-devel(a)alsa-project.org Cc: linux-watchdog(a)vger.kernel.org Cc: linux-pm(a)vger.kernel.org Cc: linux-mmc(a)vger.kernel.org Cc: linux-arm-kernel(a)lists.infradead.org Cc: linux-kernel(a)vger.kernel.org --- drivers/mmc/host/Kconfig | 2 +- drivers/thermal/Kconfig | 2 +- drivers/watchdog/Kconfig | 2 +- sound/soc/kirkwood/Kconfig | 2 +- 4 files changed, 4 insertions(+), 4 deletions(-) diff --git a/drivers/mmc/host/Kconfig b/drivers/mmc/host/Kconfig index 1384f67abe21..a69e48b582e5 100644 --- a/drivers/mmc/host/Kconfig +++ b/drivers/mmc/host/Kconfig @@ -168,7 +168,7 @@ config MMC_SDHCI_ESDHC_IMX config MMC_SDHCI_DOVE tristate "SDHCI support on Marvell's Dove SoC" - depends on ARCH_DOVE + depends on ARCH_DOVE || MACH_DOVE depends on MMC_SDHCI_PLTFM select MMC_SDHCI_IO_ACCESSORS help diff --git a/drivers/thermal/Kconfig b/drivers/thermal/Kconfig index 35c066489a19..07b5e56e9a97 100644 --- a/drivers/thermal/Kconfig +++ b/drivers/thermal/Kconfig @@ -150,7 +150,7 @@ config KIRKWOOD_THERMAL config DOVE_THERMAL tristate "Temperature sensor on Marvell Dove SoCs" - depends on ARCH_DOVE + depends on ARCH_DOVE || MACH_DOVE depends on OF help Support for the Dove thermal sensor driver in the Linux thermal diff --git a/drivers/watchdog/Kconfig b/drivers/watchdog/Kconfig index 79d25894343a..5659c62a175c 100644 --- a/drivers/watchdog/Kconfig +++ b/drivers/watchdog/Kconfig @@ -292,7 +292,7 @@ config DAVINCI_WATCHDOG config ORION_WATCHDOG tristate "Orion watchdog" - depends on ARCH_ORION5X || ARCH_KIRKWOOD || ARCH_DOVE + depends on ARCH_ORION5X || ARCH_KIRKWOOD || ARCH_DOVE || MACH_DOVE select WATCHDOG_CORE help Say Y here if to include support for the watchdog timer diff --git a/sound/soc/kirkwood/Kconfig b/sound/soc/kirkwood/Kconfig index 78ed4a42ad21..a1506d57e646 100644 --- a/sound/soc/kirkwood/Kconfig +++ b/sound/soc/kirkwood/Kconfig @@ -1,6 +1,6 @@ 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 || MACH_DOVE || 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 -- 1.8.5.3

4 9

[alsa-devel] [PATCH] ASoC: da732x: Replace hw_read usage with snd_soc_read()
by Mark Brown 04 Mar '14

04 Mar '14

From: Mark Brown <broonie(a)linaro.org> Pre-merge code was using direct hw_read() calls as an out of framework way of doing volatile register I/O when not using regmap. This has never functioned correctly in mainline due to the regmap conversion, the hw_read() implementation still does caching. In order to facilitate removal of the subsystem level I/O code convert to use snd_soc_read(), there should be no functional impact. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/da732x.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/da732x.c b/sound/soc/codecs/da732x.c index 4d1c302..14acb3a 100644 --- a/sound/soc/codecs/da732x.c +++ b/sound/soc/codecs/da732x.c @@ -1297,9 +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) */ @@ -1316,10 +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; @@ -1360,9 +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 | @@ -1383,10 +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; -- 1.9.0

2 1

[alsa-devel] [PATCH] ALSA: hda - add automute fix for another dell AIO model
by Hui Wang 04 Mar '14

04 Mar '14

When plugging a headphone or headset, lots of noise is heard from internal speaker, after changing the automute via amp instead of pinctl, the noise disappears. BugLink: https://bugs.launchpad.net/bugs/1268468 Cc: David Henningsson <david.henningsson(a)canonical.com> Cc: stable(a)vger.kernel.org Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 2056b7e..b501141 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5263,7 +5263,7 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0625, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0626, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0628, "Dell", ALC668_FIXUP_AUTO_MUTE), - SND_PCI_QUIRK(0x1028, 0x064e, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x064e, "Dell", ALC668_FIXUP_AUTO_MUTE), SND_PCI_QUIRK(0x103c, 0x1632, "HP RP5800", ALC662_FIXUP_HP_RP5800), SND_PCI_QUIRK(0x1043, 0x11cd, "Asus N550", ALC662_FIXUP_BASS_1A), SND_PCI_QUIRK(0x1043, 0x1477, "ASUS N56VZ", ALC662_FIXUP_BASS_MODE4_CHMAP), -- 1.8.1.2

2 1

[alsa-devel] [PATCH v2] ASoC: snappercl15: Convert to table based DAPM setup
by Lars-Peter Clausen 04 Mar '14

04 Mar '14

Use table based setup to register the DAPM widgets and routes. This on one hand makes the code a bit shorter and cleaner and on the other hand the board level DAPM elements get registered in the card's DAPM context rather than in the CODEC's DAPM context. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- Changes since v1: * Tab align the right side of the field initializers --- sound/soc/cirrus/snappercl15.c | 18 +++++------------- 1 file changed, 5 insertions(+), 13 deletions(-) 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) -- 1.8.0

1 0

Re: [alsa-devel] change kmalloc into vmalloc for large memory allocations
by 'gregkh＠linuxfoundation.org' 04 Mar '14

04 Mar '14

On Mon, Mar 03, 2014 at 10:51:23AM +0800, Wang, Yalin wrote: > Hi greg, > > I am sorry, > I make a mistake , > You are right , > https://git.kernel.org/cgit/linux/kernel/git/will/linux.git/tree/drivers/us… > > this one should not changed to use vmalloc, > the buffer will be used by DMA， Which is why a wrapper function will never work. > others should be safe to change: > > https://git.kernel.org/cgit/linux/kernel/git/will/linux.git/tree/sound/soc/… > > https://git.kernel.org/cgit/linux/kernel/git/will/linux.git/tree/net/netfil… > https://git.kernel.org/cgit/linux/kernel/git/will/linux.git/tree/net/netfil… > https://git.kernel.org/cgit/linux/kernel/git/will/linux.git/tree/net/netfil… > https://git.kernel.org/cgit/linux/kernel/git/will/linux.git/tree/net/netfil… > > https://git.kernel.org/cgit/linux/kernel/git/will/linux.git/tree/drivers/in… Then send individual patches for these and see what happens. greg k-h

2 3

[alsa-devel] [PATCH] ASoC: cache: Do the codec->reg_cache zero pionter check
by Xiubo Li 04 Mar '14

04 Mar '14

For the snd_soc_cache_init(), the reg_size maybe zero and then the value of codec->reg_cache, which is alloced via kzalloc, maybe equal to ZERO_SIZE_PTR. If the reg parameter of snd_soc_cache_write() is large enough, the cache[idx] = val maybe cause the kernel crash... So this patch fix this via doing the zero pionter check of it. Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- sound/soc/soc-cache.c | 13 +++++++------ 1 file changed, 7 insertions(+), 6 deletions(-) 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; -- 1.8.4

3 2

[alsa-devel] [PATCH RESEND] sound: omap: n810: fix init with DT boot
by Aaro Koskinen 04 Mar '14

04 Mar '14

Since 3.14-rc1 only DT boot has been supported on N810, so this file fails to init. Make a minimal fix to retain functionality. This file should be properly converted to DT in longer term. There seems to be still other unresolved issues with N810 audio support, but this patch is needed at minimum as otherwise the machine driver probing would completely fail. Signed-off-by: Aaro Koskinen <aaro.koskinen(a)iki.fi> Acked-by: Jarkko Nikula <jarkko.nikula(a)linux.intel.com> --- Previous discussion: http://marc.info/?l=linux-omap&m=139387208426122&w=2 sound/soc/omap/n810.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c index 3fde9e4..d163e18 100644 --- a/sound/soc/omap/n810.c +++ b/sound/soc/omap/n810.c @@ -305,7 +305,9 @@ static int __init n810_soc_init(void) int err; struct device *dev; - if (!(machine_is_nokia_n810() || machine_is_nokia_n810_wimax())) + if (!of_have_populated_dt() || + (!of_machine_is_compatible("nokia,n810") && + !of_machine_is_compatible("nokia,n810-wimax"))) return -ENODEV; n810_snd_device = platform_device_alloc("soc-audio", -1); -- 1.9.0

2 1

[alsa-devel] [PATCH] ASoC: wm{5102, 5110, 8997}: Replace codec->control_data with arizona->regmap
by Lars-Peter Clausen 04 Mar '14

04 Mar '14

With the ongoing component-ization of the ASoC framework and the continuing migration to using regmap for IO the control_data field of the snd_soc_codec struct will eventually be removed. Prepare the wm5192, wm5110 and wm8997 drivers for this by using arizona->regmap instead of accessing the CODEC's control_data field. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/wm5102.c | 2 +- sound/soc/codecs/wm5110.c | 2 +- sound/soc/codecs/wm8997.c | 2 +- 3 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index 293dffc..3410905 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; diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 4de2bf1..d7bf884 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; diff --git a/sound/soc/codecs/wm8997.c b/sound/soc/codecs/wm8997.c index 4e6442c..e10f44d 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; -- 1.8.0

3 2

[alsa-devel] [PATCH] ASoC: wm8753: Remove superfluous 'codec->cache_sync = 1'
by Lars-Peter Clausen 04 Mar '14

04 Mar '14

The wm8763 driver uses regmap for IO which means that codec->cache_sync is not used. The line was added in commit d3398ff ('ASoC: Convert WM8753 to direct regmap API usage'). Presumably this was meant to be regcache_mark_dirty(), but since we already call regcache_mark_dirty() in the core when suspending the CODEC it is safe to just remove the line. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/wm8753.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index be85da9..a02e76c 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -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; } -- 1.8.0

3 2

[alsa-devel] change kmalloc into vmalloc for large memory allocations
by Wang, Yalin 04 Mar '14

04 Mar '14

Hi I find there is some drivers use kmalloc to allocate large Memorys during module_init, some can be changed to use vmalloc To save some low mem, I add log in kernel to track , And list them here: https://git.kernel.org/cgit/linux/kernel/git/will/linux.git/tree/drivers/us… https://git.kernel.org/cgit/linux/kernel/git/will/linux.git/tree/sound/soc/… https://git.kernel.org/cgit/linux/kernel/git/will/linux.git/tree/net/netfil… https://git.kernel.org/cgit/linux/kernel/git/will/linux.git/tree/net/netfil… https://git.kernel.org/cgit/linux/kernel/git/will/linux.git/tree/net/netfil… https://git.kernel.org/cgit/linux/kernel/git/will/linux.git/tree/net/netfil… https://git.kernel.org/cgit/linux/kernel/git/will/linux.git/tree/drivers/in… they allocate large memory from 10k~64K , this will use lots of low mem, instead if we use vmalloc for these drivers , it will be good for some devices like smart phone, we often encounter some errors like kmalloc failed because there is not enough low mem , especially when the device has physical memory less than 1GB . could this module change the memory allocation into vmalloc ? I was thinking that if we can introduce a helper function in kmalloc.h like this : Kmalloc(size, flags) { If (size > PAGE_SIZE && flags&CAN_USE_VMALLOC_FLAG) return vmalloc(size); Else return real_kmalloc(size); } Kfree(ptr) { If (is_vmalloc_addr(ptr)) Vfree(ptr); Else Kfree(ptr); } But we need add some flags to ensure always use kmalloc for Some special use (dma etc..) How do you think of it ? Thanks

7 18

[alsa-devel] [PATCH] ASoC: snappercl15: Convert to table based DAPM setup
by Lars-Peter Clausen 03 Mar '14

03 Mar '14

Use table based setup to register the DAPM widgets and routes. This on one hand makes the code a bit shorter and cleaner and on the other hand the board level DAPM elements get registered in the card's DAPM context rather than in the CODEC's DAPM context. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/cirrus/snappercl15.c | 18 +++++------------- 1 file changed, 5 insertions(+), 13 deletions(-) diff --git a/sound/soc/cirrus/snappercl15.c b/sound/soc/cirrus/snappercl15.c index 29238a7..bdc88e5 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) -- 1.8.0

2 2

[alsa-devel] [PATCH] sound: omap: n810: fix init with DT boot
by Aaro Koskinen 03 Mar '14

03 Mar '14

Since 3.14-rc1 only DT boot has been supported on N810. Make a minimal fix to retain functionality. This file should be properly converted to DT in longer term. Signed-off-by: Aaro Koskinen <aaro.koskinen(a)iki.fi> --- sound/soc/omap/n810.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c index 3fde9e4..d163e18 100644 --- a/sound/soc/omap/n810.c +++ b/sound/soc/omap/n810.c @@ -305,7 +305,9 @@ static int __init n810_soc_init(void) int err; struct device *dev; - if (!(machine_is_nokia_n810() || machine_is_nokia_n810_wimax())) + if (!of_have_populated_dt() || + (!of_machine_is_compatible("nokia,n810") && + !of_machine_is_compatible("nokia,n810-wimax"))) return -ENODEV; n810_snd_device = platform_device_alloc("soc-audio", -1); -- 1.8.3.3

2 1

[alsa-devel] Enable headset Mic for Dell DT
by Kailang 03 Mar '14

03 Mar '14

Hi Takashi, I created patch below for Dell DT for support headset MIC. Many Thanks. Kailang. >From 2f448d6f9ad7023befa91b80f86cb267311d3f8a Mon Sep 17 00:00:00 2001 From: Kailang Yang <kailang(a)realtek.com> Date: Mon, 3 Mar 2014 15:27:22 +0800 Subject: [PATCH] ALSA: hda/realtek - Add headset quirk for Dell DT This quirk is needed for the headset microphone to work. Signed-off-by: Kailang Yang <kailang(a)realtek.com> diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 534796e..74456bd 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3594,21 +3594,38 @@ static void alc_fixup_headset_mode_no_hp_mic(struct hda_codec *codec, alc_fixup_headset_mode(codec, fix, action); } +static void alc255_set_default_jack_type(struct hda_codec *codec) +{ + /* Set to iphone type */ + alc_write_coef_idx(codec, 0x1b, 0x880b); + alc_write_coef_idx(codec, 0x45, 0xd089); + alc_write_coef_idx(codec, 0x1b, 0x080b); + alc_write_coef_idx(codec, 0x46, 0x0004); + alc_write_coef_idx(codec, 0x1b, 0x0c0b); + msleep(30); +} + static void alc_fixup_headset_mode_alc255(struct hda_codec *codec, const struct hda_fixup *fix, int action) { if (action == HDA_FIXUP_ACT_PRE_PROBE) { - /* Set to iphone type */ - alc_write_coef_idx(codec, 0x1b, 0x880b); - alc_write_coef_idx(codec, 0x45, 0xd089); - alc_write_coef_idx(codec, 0x1b, 0x080b); - alc_write_coef_idx(codec, 0x46, 0x0004); - alc_write_coef_idx(codec, 0x1b, 0x0c0b); - msleep(30); + alc255_set_default_jack_type(codec); } alc_fixup_headset_mode(codec, fix, action); } +static void alc_fixup_headset_mode_alc255_no_hp_mic(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + if (action == HDA_FIXUP_ACT_PRE_PROBE) { + struct alc_spec *spec = codec->spec; + spec->parse_flags |= HDA_PINCFG_HEADSET_MIC; + alc255_set_default_jack_type(codec); + } + else + alc_fixup_headset_mode(codec, fix, action); +} + static void alc_fixup_auto_mute_via_amp(struct hda_codec *codec, const struct hda_fixup *fix, int action) { @@ -3875,7 +3892,9 @@ enum { ALC290_FIXUP_SUBWOOFER_HSJACK, ALC269_FIXUP_THINKPAD_ACPI, ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, + ALC255_FIXUP_DELL2_MIC_NO_PRESENCE, ALC255_FIXUP_HEADSET_MODE, + ALC255_FIXUP_HEADSET_MODE_NO_HP_MIC, }; static const struct hda_fixup alc269_fixups[] = { @@ -4248,10 +4267,23 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC255_FIXUP_HEADSET_MODE }, + [ALC255_FIXUP_DELL2_MIC_NO_PRESENCE] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x19, 0x01a1913c }, /* use as headset mic, without its own jack detect */ + { } + }, + .chained = true, + .chain_id = ALC255_FIXUP_HEADSET_MODE_NO_HP_MIC + }, [ALC255_FIXUP_HEADSET_MODE] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_headset_mode_alc255, }, + [ALC255_FIXUP_HEADSET_MODE_NO_HP_MIC] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_headset_mode_alc255_no_hp_mic, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -4314,6 +4346,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0658, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x065f, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0662, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x0668, "Dell", ALC255_FIXUP_DELL2_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x0669, "Dell", ALC255_FIXUP_DELL2_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x15cc, "Dell X5 Precision", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x15cd, "Dell X5 Precision", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC2),

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[alsa-devel] [PATCH] ALSA: hda - Added inverted digital-mic handling for Acer TravelMate 8371
by Marius Knaust 03 Mar '14

03 Mar '14

Signed-off-by: Marius Knaust <marius.knaust(a)gmail.com> diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 534796e..9fc7bb6 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4255,6 +4255,7 @@ static const struct hda_fixup alc269_fixups[] = { }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { + SND_PCI_QUIRK(0x1025, 0x0283, "Acer TravelMate 8371", ALC269_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x1025, 0x029b, "Acer 1810TZ", ALC269_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x1025, 0x0349, "Acer AOD260", ALC269_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x1025, 0x047c, "Acer AC700", ALC269_FIXUP_ACER_AC700),

2 1

[alsa-devel] [PATCH 1/3] ALSA: hda - Make azx_attach_pcm_stream static
by Dylan Reid 03 Mar '14

03 Mar '14

It is only used in hda_controller.c now. Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- sound/pci/hda/hda_controller.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index 6156d0a..97993e1 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -971,8 +971,8 @@ static void azx_pcm_free(struct snd_pcm *pcm) #define MAX_PREALLOC_SIZE (32 * 1024 * 1024) -int azx_attach_pcm_stream(struct hda_bus *bus, struct hda_codec *codec, - struct hda_pcm *cpcm) +static int azx_attach_pcm_stream(struct hda_bus *bus, struct hda_codec *codec, + struct hda_pcm *cpcm) { struct azx *chip = bus->private_data; struct snd_pcm *pcm; -- 1.8.1.3.605.g02339dd

2 3

[alsa-devel] [PATCH] ASoC: wm1133-ev1: Convert to table based DAPM setup
by Lars-Peter Clausen 03 Mar '14

03 Mar '14

Use table based setup to register the DAPM widgets and routes. This on one hand makes the code a bit shorter and cleaner and on the other hand the board level DAPM elements get registered in the card's DAPM context rather than in the CODEC's DAPM context. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/fsl/wm1133-ev1.c | 11 +++++------ 1 file changed, 5 insertions(+), 6 deletions(-) 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; -- 1.8.0

2 1

Re: [alsa-devel] 7.1 surround not working on ALC892/ Ubuntu 13.04
by Raymond Yau 03 Mar '14

03 Mar '14

> I am unable to get 7.1 surround sound option on my system. > Alsamixer is not giving me any option to choose 8 channels. > Surround 5.1 is working correctly. > I am not sure how to configure the system to get the 8 channel setup working. > >Distributor ID: Ubuntu >Description: Ubuntu 13.04 >Release: 13.04 >Codename: raring >Attaching all the sound-card information I could gather from the system. >Please let me know in case any additional information is needed to debug the issue. > user1 <at> user1-MI970VF:~/ufouser$ cat /proc/asound/card0/codec#* | grep Codec > Codec: Realtek ALC892 For motherboard with 3 audio jacks at rear panel and 10 channels HDA codec you need to retask headphone at front panel as side line out the easy way is to use hdajackretask (advanced override) or early patching to retask if green line out at rear panel has sequence 1 in pin default. 1) blue line jack as Line Out with sequence 2 2) pink mic jack as Line Out with sequence 3 3) front panel headphone to Line out with sequence 4 the three retasked line out should have same default association as the green line out in theory, it is possible to add the retask of headphone to line out into the existing channel mode (8ch) when the codec has extra DAC and volume control using multi_Ios pin butt need to hack the jack detection control, disable auto mute and independent headphone when the headphone is retasked

2 1

[alsa-devel] [PATCHv2 0/3] Simplify the CODEC ASoC probe code.
by Xiubo Li 03 Mar '14

03 Mar '14

4 11

[alsa-devel] [PATCH 1/2] ASoC: dapm: Power off all widgets in the snd_soc_dapm_shutdown
by Xiang Xiao 03 Mar '14

03 Mar '14

The widgets generated by the machine driver need to power off too. Signed-off-by: Xiang Xiao <xiaoxiang(a)xiaomi.com> --- sound/soc/soc-dapm.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 3651a37..00ed0e9 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3899,7 +3899,7 @@ 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; @@ -3939,12 +3939,12 @@ 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, + list_for_each_entry(dapm, &card->dapm_list, list) { + soc_dapm_shutdown_dapm(dapm); + if (dapm->bias_level == SND_SOC_BIAS_STANDBY) + snd_soc_dapm_set_bias_level(dapm, SND_SOC_BIAS_OFF); } } -- 1.8.1.2

2 3

[alsa-devel] [PATCHv9][ 1/2] ASoC: eukrea-tlv320: Add DT support.
by Denis Carikli 03 Mar '14

03 Mar '14

Cc: Eric Bénard <eric(a)eukrea.com> Cc: Ian Campbell <ijc+devicetree(a)hellion.org.uk> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Mark Rutland <mark.rutland(a)arm.com> Cc: Pawel Moll <pawel.moll(a)arm.com> Cc: Rob Herring <robh+dt(a)kernel.org> Cc: alsa-devel(a)alsa-project.org Cc: devicetree(a)vger.kernel.org Signed-off-by: Denis Carikli <denis(a)eukrea.com> --- ChangeLog v9->v10: - Better Cc list - Some non-arm architecture that have CONFIG_OF could select that driver, but they lacked asm/mach-types.h, so the compilation failed. That has been fixed by permitting, when CONFIG_OF is selected, to select the driver only if CONFIG_ARM is set. ChangeLog v7->v8: - fixed the vendor prefixes in the dts bindings. - updated the code and the documentation accordingly. - removed the need of an audio-codec property, since it's available in the ssi bindings. - the dai codec is now retrived trough the dt in order to support machines that have its codec on another i2c bus number. - driver compatible is now similar to the ones of the other drivers. ChangeLog v6->v7: - The cleanups went into another patch. - The support for fsl_ssi doesn't depend on dt anymore. - platform_name = "imx-ssi.0" is still set when in non-dt mode. --- .../devicetree/bindings/sound/eukrea-tlv320.txt | 21 ++++ sound/soc/fsl/Kconfig | 5 +- sound/soc/fsl/eukrea-tlv320.c | 108 +++++++++++++++++--- 3 files changed, 121 insertions(+), 13 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/eukrea-tlv320.txt diff --git a/Documentation/devicetree/bindings/sound/eukrea-tlv320.txt b/Documentation/devicetree/bindings/sound/eukrea-tlv320.txt new file mode 100644 index 0000000..0d7985c --- /dev/null +++ b/Documentation/devicetree/bindings/sound/eukrea-tlv320.txt @@ -0,0 +1,21 @@ +Audio complex for Eukrea boards with tlv320aic23 codec. + +Required properties: +- compatible : "eukrea,asoc-tlv320" +- eukrea,model : The user-visible name of this sound complex. +- ssi-controller : The phandle of the SSI controller. +- fsl,mux-int-port : The internal port of the i.MX audio muxer (AUDMUX). +- fsl,mux-ext-port : The external port of the i.MX audio muxer. + +Note: The AUDMUX port numbering should start at 1, which is consistent with +hardware manual. + +Example: + + sound { + compatible = "eukrea,asoc-tlv320"; + eukrea,model = "imx51-eukrea-tlv320aic23"; + ssi-controller = <&ssi2>; + fsl,mux-int-port = <2>; + fsl,mux-ext-port = <3>; + }; diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 07f8f14..b8f8703 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -168,12 +168,15 @@ 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_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 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, -- 1.7.9.5

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[alsa-devel] [PATCH 1/8] ASoC: pxa: Pass correct DAPM context to {corgi, poodle, spitz}_ext_control
by Lars-Peter Clausen 03 Mar '14

03 Mar '14

When calling {corgi,poodle,spitz}_ext_control() from the startup callback we pass the CODEC's DAPM context instead of the card's DAPM context. This is not a problem per se since all the DAPM functions in ext_control() fallback to widgets from other DAPM contexts, but passing the card's context is more consistent. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/pxa/corgi.c | 3 +-- sound/soc/pxa/poodle.c | 3 +-- sound/soc/pxa/spitz.c | 3 +-- 3 files changed, 3 insertions(+), 6 deletions(-) diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c index b4db3b9..5a88136 100644 --- a/sound/soc/pxa/corgi.c +++ b/sound/soc/pxa/corgi.c @@ -101,10 +101,9 @@ static void corgi_ext_control(struct snd_soc_dapm_context *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; /* check the jack status at stream startup */ - corgi_ext_control(&codec->dapm); + corgi_ext_control(&rtd->card->dapm); return 0; } diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c index 27c6c03..c6bdc6c 100644 --- a/sound/soc/pxa/poodle.c +++ b/sound/soc/pxa/poodle.c @@ -74,10 +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; /* check the jack status at stream startup */ - poodle_ext_control(&codec->dapm); + poodle_ext_control(&rtd->card->dapm); return 0; } diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c index a3a13c9..1373b01 100644 --- a/sound/soc/pxa/spitz.c +++ b/sound/soc/pxa/spitz.c @@ -111,10 +111,9 @@ static void spitz_ext_control(struct snd_soc_dapm_context *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; /* check the jack status at stream startup */ - spitz_ext_control(&codec->dapm); + spitz_ext_control(&rtd->card->dapm); return 0; } -- 1.8.0

2 16

[alsa-devel] [PATCH 1/2] ASoC: neo1973_wm8753: Post gta01 support removal cleanup
by Lars-Peter Clausen 03 Mar '14

03 Mar '14

With GTA01 support gone from the driver there is no need to keep a separate initilization routine for the GTA02 specific bits. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/samsung/neo1973_wm8753.c | 116 ++++++++++++------------------------- 1 file changed, 37 insertions(+), 79 deletions(-) diff --git a/sound/soc/samsung/neo1973_wm8753.c b/sound/soc/samsung/neo1973_wm8753.c index 98a04c1..0125d9d 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"), @@ -276,39 +269,6 @@ 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; @@ -344,18 +304,16 @@ static int neo1973_wm8753_init(struct snd_soc_pcm_runtime *rtd) 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(dapm, "Stereo Out"); + snd_soc_dapm_disable_pin(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(dapm, "Stereo Out"); + snd_soc_dapm_ignore_suspend(dapm, "Handset Spk"); return 0; } -- 1.8.0

2 2

[alsa-devel] [PATCH] ASoC: pcm: free path list before exiting from error conditions
by Patrick Lai 02 Mar '14

02 Mar '14

dpcm_path_get() allocates dynamic memory to hold path list. Corresponding dpcm_get_path() must be called to free the memory. dpcm_get_path() is not called under several error conditions. This leads to memory leak. Signed-off-by: Patrick Lai <plai(a)codeaurora.org> --- sound/soc/soc-pcm.c | 3 +++ 1 files changed, 3 insertions(+), 0 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 47e1ce7..28522bd 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1989,6 +1989,7 @@ 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); @@ -2018,6 +2019,7 @@ 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); @@ -2082,6 +2084,7 @@ 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"); } -- The Qualcomm Innovation Center, Inc. is a member of the Code Aurora Forum, hosted by The Linux Foundation

1 0

[alsa-devel] [RFC 00/19] Enable platform HDA drivers
by Dylan Reid 01 Mar '14

01 Mar '14

This series rearranges some code in the HDA driver to break dependencies on PCI and allow for most of the HDA code to be reused for non-PCI based HDA controllers. I tried to break it up to make it less scary and to keep changes away from copies of code. It is mostly copies with a few modifications to make more of the code usable by both HDA drivers. The main changes were to avoid using pci to get a device pointer, and to make the bus reads/writes into callbacks so that the Tegra driver could ensure 32-bit access to the HDA registers. The final patch is a work in progress included here to provide context for the preceding patches. The Tegra HDA driver still needs some cleaning up, but nothing that will affect the other changes. The last infrastructure issue is that the Tegra driver uses DT, causing it to be probed before the codec patches. I haven't determined how to fix that yet. I didn't include the patch to move the hda directory from under pci because of its size and how easy it will be to regenerate later. Thanks for looking. Let me know if you think any of these should be squashed or re-ordered and if the changes and breakup of code makes sense. Change since v1: - Rename hda_shared to hda_controller - Add more 'ops' to break out hda_intel specific functions, such as postponed interrupt handling and page allocation - Re order a few patches to make rebases easier - Make bdl_pos_adj optional - I left the position fix code in hda_controller, I don't have a good way to test a refactoring of that, there are a few combinations of settings. It will probably be useful to other controllers as well, but moving it wouldn't be hard if we really want to. Thanks again, Dylan -- Dylan Reid (19): ALSA: hda - Move some definitions to new hda_priv.h ALSA: hda - Allow different ops to read/write registers ALSA: hda - Keep pointer to bdl_pos_fix in chip struct ALSA: hda - Store device pointer in chip structure ALSA: hda - Move pcm ops and support funcs to shared file ALSA: hda - Pull pages allocation to shared file ALSA: hda - Move the dsp loader to hda_shared ALSA: hda - Add function pointer for disabling MSI ALSA: hda - Relocate RIRB/CORB interface to hda_shared ALSA: hda - move alloc_cmd_io to hda_shared. ALSA: hda - Move low level functions to hda_shared ALSA: hda - remove unused clear of STATESTS ALSA: hda - Move azx_interrupt to hda_shared ALSA: hda - Add jackpoll_ms to struct azx ALSA: hda - Pass max_slots and power_save to codec_create ALSA: hda - Move codec create to hda_shared ALSA: core - Define snd_pci_quirk without CONFIG_PCI ALSA: hda - remove PCI dependency in Kconfig WIP: ALSA: hda - Add driver for Tegra SoC HDA include/sound/core.h | 15 +- sound/pci/Kconfig | 4 +- sound/pci/hda/Kconfig | 55 +- sound/pci/hda/Makefile | 8 +- sound/pci/hda/hda_intel.c | 2832 +++----------------------------------------- sound/pci/hda/hda_priv.h | 446 +++++++ sound/pci/hda/hda_shared.c | 2175 ++++++++++++++++++++++++++++++++++ sound/pci/hda/hda_shared.h | 52 + sound/pci/hda/hda_tegra.c | 858 ++++++++++++++ 9 files changed, 3778 insertions(+), 2667 deletions(-) create mode 100644 sound/pci/hda/hda_priv.h create mode 100644 sound/pci/hda/hda_shared.c create mode 100644 sound/pci/hda/hda_shared.h create mode 100644 sound/pci/hda/hda_tegra.c -- 1.8.1.3.605.g02339dd

2 25

[alsa-devel] [PATCH 0/3] To simplify the code for ASoC probe.
by Xiubo Li 01 Mar '14

01 Mar '14

Xiubo Li (3): ASoC: io: Clean up snd_soc_codec_set_cache_io() ASoC: core: Set the default I/O up try regmap. ASoC: SGTL5000: Simplify ASoC probe code. sound/soc/codecs/sgtl5000.c | 8 -------- sound/soc/soc-core.c | 8 ++++---- sound/soc/soc-io.c | 5 ----- 3 files changed, 4 insertions(+), 17 deletions(-) -- 1.8.4

4 18

[alsa-devel] [PATCH 14/28] Remove MACH_SMDKC210
by Richard Weinberger 01 Mar '14

01 Mar '14

The symbol is an orphan, get rid of it. Signed-off-by: Richard Weinberger <richard(a)nod.at> --- sound/soc/samsung/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 454f41c..7ea576c 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -145,7 +145,7 @@ config SND_SOC_SAMSUNG_RX1950_UDA1380 config SND_SOC_SAMSUNG_SMDK_WM9713 tristate "SoC AC97 Audio support for SMDK with WM9713" - depends on SND_SOC_SAMSUNG && (MACH_SMDK6410 || MACH_SMDKC100 || MACH_SMDKV210 || MACH_SMDKC110 || MACH_SMDKV310 || MACH_SMDKC210) + depends on SND_SOC_SAMSUNG && (MACH_SMDK6410 || MACH_SMDKC100 || MACH_SMDKV210 || MACH_SMDKC110 || MACH_SMDKV310) select SND_SOC_WM9713 select SND_SAMSUNG_AC97 help -- 1.8.4.2

3 9

[alsa-devel] [PATCH v2 00/11] ASoC: Consolidate enum controls and DAPM MUXs
by Lars-Peter Clausen 01 Mar '14

01 Mar '14

Changes since v1: * Rename the xmax parameter for the SOC_ENUM_SINGLE_VIRT() macro to xitems. Original cover letter below. We currently have 5 nearly identical implementations of enum controls. Two for normal controls and three for DAPM controls. The only difference between them is how and if the selected enum item is mapped to a register value. For normal enums there is a direct mapping of the selected item to the register value, for value enums there is an additional look-up table that maps the selected item to a register value. DAPM also as a virtual enum which is not backed by any hardware register and the enum is solely used to influence DAPM routing decisions. The different implementations can easily be consolidating by making a few small changes. The series first introduces two helper functions to translate between the selected item and the register value. These functions will do the right thing depending on whether the enum has a direct mapping or is a value enum. These functions will be used to consolidate enums and value enums as well as MUXs and value MUXs. The next step is consolidate MUXs and virtual MUXs. This requires enums that are only used as virtual enums to be marked as such. This is done the same way as for virtual mixer controls by setting reg to SND_SOC_NOPM. All users of virtual MUXs are updated accordingly and then virtual MUXs and MUXs are updated to share the same implementation. The last patch in the series does cleanup of dapm_set_path_status() on top of all this. Once this has been merged I'll send a second patch-set replacing and removing the now deprecated ENUM and MUX macros. The series depends on topic/dapm, topic/enum, topic/sign, topic/adau1373, topic/max98090, topic/mc13783, fix/wm8994, topic/wm899 Lars-Peter Clausen (11): ASoC: Remove unused 'reg2' field from soc_enum struct ASoC: Consolidate enum and value enum controls ASoC: dapm: Consolidate MUXs and value MUXs ASoC: Add macros for defining virtual enums ASoC: adau1373: Use SOC_ENUM_SINGLE_VIRT_DECL() ASoC: max98090: Use SOC_ENUM_SINGLE_VIRT_DECL() ASoC: mc13783: Use SOC_ENUM_SINGLE_VIRT_DECL() ASoC: wm8994: Use SOC_ENUM_SINGLE_VIRT_DECL() ASoC: wm8995: Use SOC_ENUM_SINGLE_VIRT_DECL() ASoC: dapm: Consolidate MUXs and virtual MUXs ASoC: dapm: Break dapm_set_path_status() appart include/sound/soc-dapm.h | 38 +---- include/sound/soc.h | 41 +++-- sound/soc/codecs/adau1373.c | 4 +- sound/soc/codecs/max98090.c | 2 +- sound/soc/codecs/mc13783.c | 6 +- sound/soc/codecs/wm8994.c | 3 +- sound/soc/codecs/wm8995.c | 2 +- sound/soc/soc-core.c | 101 ++---------- sound/soc/soc-dapm.c | 369 +++++++++----------------------------------- 9 files changed, 136 insertions(+), 430 deletions(-) -- 1.8.0

2 12

[alsa-devel] Audio passthrough on Pandaboard using ALSA API
by Vadim Izmalkov 28 Feb '14

28 Feb '14

Hello I created simple application that makes audio pass-through (from analog input to analog output), After 2-10 seconds get error like: AlsaPassthrough: pcm.c:693: snd_pcm_close: Assertion `pcm' failed. or write to audio interface failed (Broken pipe) Seems I have problem with buffer size/time or somethink like that. Parameters used: Buffer - 100 ms / 4800 frames / 19200 bytes Period - 10 ms / 480 frames / 1920 bytes Buffer length used 19200 bytes. char buffer[19200]; Write/read function used as follow snd_pcm_readi(pcm, buffer, frames); snd_pcm_writei(pcm, buffer, frames); Have somebody any suggestion? Segments of the code you can find bellow. Alsamixer settings: amixer cset name='Analog Left Capture Route' 'Headset Mic' amixer cset name='Analog Right Capture Route' 'Headset Mic' amixer cset name='MUX_UL00' 'AMic0' amixer cset name='MUX_UL01' 'AMic1' amixer cset name='Headset Left Playback' 'HS DAC' amixer cset name='Headset Right Playback' 'HS DAC' amixer cset name='Headset Playback Volume' 100 amixer cset name='DL1 Media' 120 Structure for the devices: typedef struct { char *deviceName; snd_pcm_t *pcm; snd_pcm_hw_params_t *hwParams; snd_pcm_stream_t stream; snd_pcm_access_t access; snd_pcm_format_t format; snd_pcm_uframes_t frames; int mode; unsigned int sampleRate; unsigned int bufferTime; unsigned int periodTime; unsigned int channels; } pcmDevice_t; Parameters for capture: pcmDevice_t* initRecordDevice() { pcmDevice_t *device; device = malloc(sizeof(pcmDevice_t)); device->deviceName = strdup("plughw:0,0"); device->stream = SND_PCM_STREAM_CAPTURE; device->access = SND_PCM_ACCESS_RW_INTERLEAVED; device->format = SND_PCM_FORMAT_S16_LE; device->mode = 0; //SND_PCM_ASYNC; //SND_PCM_NONBLOCK; device->sampleRate = 48000; device->bufferTime = 100 * 1000; // 100 ms device->periodTime = 10 * 1000; // 20ms device->channels = 2; return device; } Parameters for playback: pcmDevice_t* initPlaybackDevice() { pcmDevice_t *device; device = malloc(sizeof(pcmDevice_t)); device->deviceName = strdup("plughw:0,0"); device->stream = SND_PCM_STREAM_PLAYBACK; device->access = SND_PCM_ACCESS_RW_INTERLEAVED; device->format = SND_PCM_FORMAT_S16_LE; device->mode = 0; //SND_PCM_ASYNC; //SND_PCM_NONBLOCK; device->sampleRate = 48000; device->bufferTime = 100 * 1000; // 100 ms device->periodTime = 10 * 1000; // 20ms device->channels = 2; return device; } Setup devices: int setupDevice(pcmDevice_t *device) { int err; /* Open PCM. The last parameter of this function is the mode. */ /* If this is set to 0, the standard mode is used. Possible */ /* other values are SND_PCM_NONBLOCK and SND_PCM_ASYNC. */ /* If SND_PCM_NONBLOCK is used, read / write access to the */ /* PCM device will return immediately. If SND_PCM_ASYNC is */ /* specified, SIGIO will be emitted whenever a period has */ /* been completely processed by the soundcard. */ if ((err = snd_pcm_open(&device->pcm, device->deviceName, device->stream, device->mode)) < 0) { fprintf (stderr, "cannot open audio device %s (%s)\n", device->deviceName, snd_strerror (err)); exit (1); } if ((err = snd_pcm_hw_params_malloc (&device->hwParams)) < 0) { fprintf (stderr, "cannot allocate hardware parameter structure (%s)\n", snd_strerror (err)); exit (1); } if ((err = snd_pcm_hw_params_any (device->pcm, device->hwParams)) < 0) { fprintf (stderr, "cannot initialize hardware parameter structure (%s)\n", snd_strerror (err)); exit (1); } if ((err = snd_pcm_hw_params_set_access (device->pcm, device->hwParams, device->access)) < 0) { fprintf (stderr, "cannot set access type (%s)\n", snd_strerror (err)); exit (1); } if ((err = snd_pcm_hw_params_set_format (device->pcm, device->hwParams, device->format)) < 0) { fprintf (stderr, "cannot set sample format (%s)\n", snd_strerror (err)); exit (1); } if ((err = snd_pcm_hw_params_set_buffer_time_near (device->pcm, device->hwParams, &device->bufferTime, NULL)) < 0) { fprintf (stderr, "cannot set buffer size (%s)\n", snd_strerror (err)); exit (1); } else { /* Set buffer size (in frames). The resulting latency is given by */ /* latency = periodsize * periods / (rate * bytes_per_frame) */ snd_pcm_hw_params_get_buffer_size(device->hwParams, &device->frames); } if ((err = snd_pcm_hw_params_set_period_time_near (device->pcm, device->hwParams, &device->periodTime, NULL)) < 0) { fprintf (stderr, "cannot set period size (%s)\n", snd_strerror (err)); exit (1); } if ((err = snd_pcm_hw_params_set_rate_near (device->pcm, device->hwParams, &device->sampleRate, NULL)) < 0) { fprintf (stderr, "cannot set sample rate (%s)\n", snd_strerror (err)); exit (1); } if ((err = snd_pcm_hw_params_set_channels (device->pcm, device->hwParams, device->channels)) < 0) { fprintf (stderr, "cannot set channel count (%s)\n", snd_strerror (err)); exit (1); } /* Apply HW parameter settings to */ /* PCM device and prepare device */ if ((err = snd_pcm_hw_params (device->pcm, device->hwParams)) < 0) { fprintf (stderr, "cannot set parameters (%s)\n", snd_strerror (err)); exit (1); } snd_pcm_hw_params_free (device->hwParams); if ((err = snd_pcm_prepare (device->pcm)) < 0) { fprintf (stderr, "cannot prepare audio interface for use (%s)\n", snd_strerror (err)); exit (1); } return 0; } Main: main (int argc, char *argv[]) { int i; int err; char buf[4800*4]; snd_pcm_t *capture_handle; snd_pcm_hw_params_t *hw_params; snd_pcm_uframes_t frames; pcmDevice_t *pcmRecord; pcmDevice_t *pcmPlayback; pcmRecord = initRecordDevice(); pcmPlayback = initPlaybackDevice(); printf("\n----------- RECORD DEVICE -----------\n"); setupDevice(pcmRecord); showConfig(pcmRecord); printf("\n---------- PLAYBACK DEVICE ----------\n"); setupDevice(pcmPlayback); showConfig(pcmPlayback); for (i = 0; i < 100; ++i) { /* Read num_frames frames from the PCM device */ /* pointed to by pcm_handle to buffer capdata. */ /* Returns the number of frames actually read. */ //if((err = snd_pcm_readi (pcmRecord->pcm, buf, pcmRecord->frames)) != pcmRecord->frames) { if((err = snd_pcm_readi (pcmRecord->pcm, buf, 480)) != 480) { fprintf (stderr, "read from audio interface failed (%s)\n", snd_strerror (err)); exit (1); } /* Write num_frames frames from buffer data to */ /* the PCM device pointed to by pcm_handle. */ /* Returns the number of frames actually written. */ if((err = snd_pcm_writei (pcmPlayback->pcm, buf, pcmRecord->frames)) != pcmRecord->frames) { fprintf (stderr, "write to audio interface failed (%s)\n", snd_strerror (err)); exit (1); } } snd_pcm_close (capture_handle); exit (0); }? Best

2 1

[alsa-devel] [alsa-plugins] [PATCH 1/1] Allow number of channels
by Corfu, ValentinX 28 Feb '14

28 Feb '14

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1 0

Re: [alsa-devel] Griffin FireWave stream format information
by Stefan Richter 28 Feb '14

28 Feb '14

On Feb 27 Takashi Sakamoto wrote: > Stefan or Jay, > > Could I request you to get outputs against command in this message with > FireWave? Jay already responded, but [...] > Based on this guess and according to its manual, FireWave should have 7 > entries: > $ firewire-request /dev/fw8 fcp 0x01ffbfc100000000ffff01ff > 0c ff bf c1 00 00 00 00 ff 00 01 90 40 02 02 01 06 06 $ firewire-request /dev/fw8 fcp 0x01ffbfc100000000ffff01ff response: 000: 0c ff bf c1 00 00 00 00 ff 00 01 90 40 02 02 01 ............@... response: 010: 06 06 .. > $ firewire-request /dev/fw8 fcp 0x01ffbfc100000000ffff02ff > 0c ff bf c1 00 00 00 00 ff 00 02 90 40 03 02 01 06 06 $ firewire-request /dev/fw8 fcp 0x01ffbfc100000000ffff02ff response: 000: 0c ff bf c1 00 00 00 00 ff 00 02 90 40 03 02 01 ............@... response: 010: 06 06 .. > $ firewire-request /dev/fw8 fcp 0x01ffbfc100000000ffff03ff > 0c ff bf c1 00 00 00 00 ff 00 03 90 40 04 02 01 06 06 $ firewire-request /dev/fw8 fcp 0x01ffbfc100000000ffff03ff response: 000: 0c ff bf c1 00 00 00 00 ff 00 03 90 40 05 02 01 ............@... response: 010: 06 06 .. > $ firewire-request /dev/fw8 fcp 0x01ffbfc100000000ffff04ff > 0c ff bf c1 00 00 00 00 ff 00 04 90 40 05 02 01 06 06 $ firewire-request /dev/fw8 fcp 0x01ffbfc100000000ffff04ff response: 000: 0c ff bf c1 00 00 00 00 ff 00 04 90 40 02 02 01 ............@... response: 010: 02 00 .. > (These four is for 6ch sound for usual 5.1ch surround) > $ firewire-request /dev/fw8 fcp 0x01ffbfc100000000ffff05ff > 0c ff bf c1 00 00 00 00 ff 00 05 90 40 02 02 01 02 06 $ firewire-request /dev/fw8 fcp 0x01ffbfc100000000ffff05ff response: 000: 0c ff bf c1 00 00 00 00 ff 00 05 90 40 03 02 01 ............@... response: 010: 02 00 .. > $ firewire-request /dev/fw8 fcp 0x01ffbfc100000000ffff06ff > 0c ff bf c1 00 00 00 00 ff 00 06 90 40 03 02 01 02 06 $ firewire-request /dev/fw8 fcp 0x01ffbfc100000000ffff06ff response: 000: 0c ff bf c1 00 00 00 00 ff 00 06 90 40 04 02 01 ............@... response: 010: 02 00 .. > $ firewire-request /dev/fw8 fcp 0x01ffbfc100000000ffff07ff > 0c ff bf c1 00 00 00 00 ff 00 07 90 40 04 02 01 02 06 $ firewire-request /dev/fw8 fcp 0x01ffbfc100000000ffff07ff response: 000: 0a ff bf c1 00 00 00 00 ff ff 07 ff ............ > (These three is for 2ch stereo sound for Dolby Pro Logic II) > > Would you please confirm these output? So this seems to be the same as from Jay's FireWave specimen. Here are results of the other requests from your subsequent mail: $ firewire-request /dev/fw8 fcp 0x01ffbfc000000000ffffffff response: 000: 0c ff bf c0 00 00 00 00 ff 00 90 40 04 02 01 06 ...........@.... response: 010: 06 . $ firewire-request /dev/fw8 fcp 0x00ffbfc100000000ff0090400202010200 response: 000: 08 ff bf c1 00 00 00 00 ff 00 90 40 02 02 01 02 ...........@.... response: 010: 00 . $ firewire-request /dev/fw8 fcp 0x00ffbfc100000000ff0090400302010200 response: 000: 08 ff bf c1 00 00 00 00 ff 00 90 40 03 02 01 02 ...........@.... response: 010: 00 . $ firewire-request /dev/fw8 fcp 0x00ffbfc100000000ff0090400402010200 response: 000: 08 ff bf c1 00 00 00 00 ff 00 90 40 04 02 01 02 ...........@.... response: 010: 00 . -- Stefan Richter -=====-====- --=- ===-- http://arcgraph.de/sr/

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[alsa-devel] [PATCH v2 0/3] alsa-lib: Improve 2.1 surround support
by David Henningsson 28 Feb '14

28 Feb '14

Thanks Takashi for your encouraging review, Diff since last version is only the things you remarked about in the first patch series. > Besides that, I find it also OK to just return the channel (>= 0) or a > negative error. But it's no big matter in which form is. This one I didn't change, just because then I would just have had to change it back in the second patch anyway. David Henningsson (3): route: Allow chmap syntax for slave channels in ttable route: Select slave chmap based on ttable information conf: Allow 2.1 surround to use different number of channels src/conf/pcm/surround21.conf | 7 +- src/pcm/pcm_route.c | 323 +++++++++++++++++++++++++++++++++++++----- 2 files changed, 290 insertions(+), 40 deletions(-) -- 1.7.9.5

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[alsa-devel] hda: Bug - (timing?) problem causes DP monitor to not get activated
by Daniel Martin 28 Feb '14

28 Feb '14

Hi, I'm having a bug here, where the loaded hda module causes some (timing?) problem resulting in a monitor connected to the DP not to get active. I can reproduce this on a Lenovo X201 and T410 (Ironlake) sitting on a docking station, which has a monitor (Dell U2410) connected to a DP. Steps to reproduce: 1. boot with the laptop at the docking station and the monitor plugged in 2. start the X-Server ... output on LVDS and DP is okay 3. disable the DP: xrandr --output LVDS1 --preferred --output DP1 --off ... output on LVDS - okay 4. switch to DP: xrandr --output LVDS1 --off --output DP1 --preferred ... DP _usually_ doesn't gets active, it turn off into suspend state Why I think that it is related to the hda module? If I boot a kernel without the hda module step 4 always works. When having the hda module loaded I need some luck and some iterations of step 3 and 4 to activate the monitor at the DP. I verified that the problem exists in kernel v3.13.1 and v3.14-rc4. The logs I have attached have been created on the T410 with kernel v3.14-rc4. The first alsa-info....txt has been created right after booting, the second one after step 4 failed. The kernel has been build with all alsa debug options I could find, the config is attached as well, and the kernel has been booted with drm.debug=0xe - this might be to much, but it shows pretty clear when the outputs have been switched. I hope that anyone can read something usefull - something that went wrong - out of the log files. My unexercised eye can't spot anything obvious. If more logs are necessary, I'll send them. Cheers, Daniel Martin

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[alsa-devel] [PATCH] ALSA: hda - Fix registration of beep input device
by Takashi Iwai 28 Feb '14

28 Feb '14

The beep input device is registered via input_register_device(), but this is called in snd_hda_attach_beep_device() where the sound devices aren't registered yet. This leads to the binding to non-existing object, thus results in failure. And, even if the binding worked (against the PCI object), it's still racy; the input device appears before the sound objects. For fixing this, register the input device properly at dev_register ops of the codec object it's bound with. Also, call snd_hda_detach_beep_device() at dev_disconnection so that it's detached at the right timing. As a bonus, since it's called in the codec's ops, we can get rid of the further call from the other codec drivers. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_beep.c | 27 ++++++++++++++++++++------- sound/pci/hda/hda_beep.h | 5 +++++ sound/pci/hda/hda_codec.c | 7 ++++++- sound/pci/hda/hda_generic.c | 1 - sound/pci/hda/patch_conexant.c | 4 +--- 5 files changed, 32 insertions(+), 12 deletions(-) diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c index 88bb08486f77..b9d3d0cd75c9 100644 --- a/sound/pci/hda/hda_beep.c +++ b/sound/pci/hda/hda_beep.c @@ -148,7 +148,6 @@ static int snd_hda_do_attach(struct hda_beep *beep) { struct input_dev *input_dev; struct hda_codec *codec = beep->codec; - int err; input_dev = input_allocate_device(); if (!input_dev) @@ -169,12 +168,6 @@ static int snd_hda_do_attach(struct hda_beep *beep) input_dev->dev.parent = &codec->dev; input_set_drvdata(input_dev, beep); - err = input_register_device(input_dev); - if (err < 0) { - input_free_device(input_dev); - codec_err(codec, "hda_beep: unable to register input device\n"); - return err; - } beep->dev = input_dev; return 0; } @@ -244,6 +237,26 @@ void snd_hda_detach_beep_device(struct hda_codec *codec) } EXPORT_SYMBOL_GPL(snd_hda_detach_beep_device); +int snd_hda_register_beep_device(struct hda_codec *codec) +{ + struct hda_beep *beep = codec->beep; + int err; + + if (!beep || !beep->dev) + return 0; + + err = input_register_device(beep->dev); + if (err < 0) { + codec_err(codec, "hda_beep: unable to register input device\n"); + input_free_device(beep->dev); + codec->beep = NULL; + kfree(beep); + return err; + } + return 0; +} +EXPORT_SYMBOL_GPL(snd_hda_register_beep_device); + static bool ctl_has_mute(struct snd_kcontrol *kcontrol) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); diff --git a/sound/pci/hda/hda_beep.h b/sound/pci/hda/hda_beep.h index cb88464676b6..76eb8589fb9e 100644 --- a/sound/pci/hda/hda_beep.h +++ b/sound/pci/hda/hda_beep.h @@ -45,6 +45,7 @@ struct hda_beep { int snd_hda_enable_beep_device(struct hda_codec *codec, int enable); int snd_hda_attach_beep_device(struct hda_codec *codec, int nid); void snd_hda_detach_beep_device(struct hda_codec *codec); +int snd_hda_register_beep_device(struct hda_codec *codec); #else static inline int snd_hda_attach_beep_device(struct hda_codec *codec, int nid) { @@ -53,5 +54,9 @@ static inline int snd_hda_attach_beep_device(struct hda_codec *codec, int nid) static inline void snd_hda_detach_beep_device(struct hda_codec *codec) { } +static inline int snd_hda_register_beep_device(struct hda_codec *codec) +{ + return 0; +} #endif #endif diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 6db2dbcbf4d3..555e8c771cfd 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1379,14 +1379,19 @@ static unsigned int hda_set_power_state(struct hda_codec *codec, static int snd_hda_codec_dev_register(struct snd_device *device) { struct hda_codec *codec = device->device_data; + int err = device_add(&codec->dev); - return device_add(&codec->dev); + if (err < 0) + return err; + snd_hda_register_beep_device(codec); + return 0; } static int snd_hda_codec_dev_disconnect(struct snd_device *device) { struct hda_codec *codec = device->device_data; + snd_hda_detach_beep_device(codec); device_del(&codec->dev); return 0; } diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 9e0609a4b2ba..16133881e967 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -5350,7 +5350,6 @@ EXPORT_SYMBOL_GPL(snd_hda_gen_init); void snd_hda_gen_free(struct hda_codec *codec) { snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_FREE); - snd_hda_detach_beep_device(codec); snd_hda_gen_spec_free(codec->spec); kfree(codec->spec); codec->spec = NULL; diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 09d6d0db2b06..1dc7e974f3b1 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -445,9 +445,7 @@ static int conexant_init(struct hda_codec *codec) static void conexant_free(struct hda_codec *codec) { - struct conexant_spec *spec = codec->spec; - snd_hda_detach_beep_device(codec); - kfree(spec); + kfree(codec->spec); } static const struct snd_kcontrol_new cxt_capture_mixers[] = { -- 1.9.0

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Re: [alsa-devel] PCM rules for FireWave in speaker.c
by Clemens Ladisch 28 Feb '14

28 Feb '14

Takashi Sakamoto wrote: > Current speaker.c have a rule for PCM channels/rates below: > - 6ch PCM at 32.0/44.1kHz > - 2ch PCM at 48.0/96.0kHz The current rules in speakers.c disallow rates less than 48 kHz if there are less than six channels, and disallow two channels if the rate is less than 48 kHz. This means that the allowed parameter combinations are 2ch at 48/96 kHz, and 6ch at 32/44.1/48/96 kHz. > Recently, with Jay Fenlason's help, I got some information about its > stream format. I'll show them in the end of this message. > > As a result, I realized FireWave has stream formats below: > - 6ch PCM at 32.0/44.1/48.0/96.0kHz > - 2ch IEC 60958-3 at 32.0/44.1/48.0kHz > > Although there are no stream formats for 2ch PCM at 48.0/96.0kHz, the > driver has it. Do you know the reason? Are there any model quirks? I don't have any of the AV/C audio specifications; these rules were determined experimentally. In other words, when the sample rate is set to 48 or 96 kHz, sending a stream with two PCM channels appears to work. If you wanted to support these undocumented formats in your driver, you would have to add some kind of quirk. Regards, Clemens

2 1

[alsa-devel] [PATCH 0/2] Simplify the CODEC ASoC probe code.
by Xiubo Li 28 Feb '14

28 Feb '14

3 4

[alsa-devel] [RFC 00/19] Enable platform HDA drivers
by Dylan Reid 28 Feb '14

28 Feb '14

This series rearranges some code in the HDA driver to break dependencies on PCI and allow for most of the HDA code to be reused for non-PCI based HDA controllers. I tried to break it up to make it less scary and to keep changes away from copies of code. It is mostly copies with a few modifications to make more of the code usable by both HDA drivers. The main changes were to avoid using pci to get a device pointer, and to make the bus reads/writes into callbacks so that the Tegra driver could ensure 32-bit access to the HDA registers. The final patch is a work in progress included here to provide context for the preceding patches. The Tegra HDA driver still needs some cleaning up, but nothing that will affect the other changes. The last infrastructure issue is that the Tegra driver uses DT, causing it to be probed before the codec patches. I haven't determined how to fix that yet. I didn't include the patch to move the hda directory from under pci because of its size and how easy it will be to regenerate later. Thanks for looking. Let me know if you think any of these should be squashed or re-ordered and if the changes and breakup of code makes sense. Dylan -- Dylan Reid (19): ALSA: hda - Move some definitions to new hda_priv.h ALSA: hda - Allow different ops to read/write registers ALSA: hda - Keep pointer to bdl_pos_fix in chip struct ALSA: hda - Store device pointer in chip structure ALSA: hda - Move pcm ops and support funcs to shared file ALSA: hda - Pull pages allocation to shared file ALSA: hda - Move the dsp loader to hda_shared ALSA: hda - Add function pointer for disabling MSI ALSA: hda - Relocate RIRB/CORB interface to hda_shared ALSA: hda - move alloc_cmd_io to hda_shared. ALSA: hda - Move low level functions to hda_shared ALSA: hda - remove unused clear of STATESTS ALSA: hda - Move azx_interrupt to hda_shared ALSA: hda - Add jackpoll_ms to struct azx ALSA: hda - Pass max_slots and power_save to codec_create ALSA: hda - Move codec create to hda_shared ALSA: core - Define snd_pci_quirk without CONFIG_PCI ALSA: hda - remove PCI dependency in Kconfig WIP: ALSA: hda - Add driver for Tegra SoC HDA include/sound/core.h | 15 +- sound/pci/Kconfig | 4 +- sound/pci/hda/Kconfig | 55 +- sound/pci/hda/Makefile | 8 +- sound/pci/hda/hda_intel.c | 2832 +++----------------------------------------- sound/pci/hda/hda_priv.h | 446 +++++++ sound/pci/hda/hda_shared.c | 2175 ++++++++++++++++++++++++++++++++++ sound/pci/hda/hda_shared.h | 52 + sound/pci/hda/hda_tegra.c | 858 ++++++++++++++ 9 files changed, 3778 insertions(+), 2667 deletions(-) create mode 100644 sound/pci/hda/hda_priv.h create mode 100644 sound/pci/hda/hda_shared.c create mode 100644 sound/pci/hda/hda_shared.h create mode 100644 sound/pci/hda/hda_tegra.c -- 1.8.1.3.605.g02339dd

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[alsa-devel] [PATCH 00/11] ASoC: Consolidate enum controls and DAPM MUXs
by Lars-Peter Clausen 28 Feb '14

28 Feb '14

We currently have 5 nearly identical implementations of enum controls. Two for normal controls and three for DAPM controls. The only difference between them is how and if the selected enum item is mapped to a register value. For normal enums there is a direct mapping of the selected item to the register value, for value enums there is an additional look-up table that maps the selected item to a register value. DAPM also as a virtual enum which is not backed by any hardware register and the enum is solely used to influence DAPM routing decisions. The different implementations can easily be consolidating by making a few small changes. The series first introduces two helper functions to translate between the selected item and the register value. These functions will do the right thing depending on whether the enum has a direct mapping or is a value enum. These functions will be used to consolidate enums and value enums as well as MUXs and value MUXs. The next step is consolidate MUXs and virtual MUXs. This requires enums that are only used as virtual enums to be marked as such. This is done the same way as for virtual mixer controls by setting reg to SND_SOC_NOPM. All users of virtual MUXs are updated accordingly and then virtual MUXs and MUXs are updated to share the same implementation. The last patch in the series does cleanup of dapm_set_path_status() on top of all this. Once this has been merged I'll send a second patch-set replacing and removing the now deprecated ENUM and MUX macros. The series depends on topic/dapm, topic/enum, topic/sign, topic/adau1373, topic/max98090, topic/mc13783, fix/wm8994, topic/wm8995 - Lars Lars-Peter Clausen (11): ASoC: Remove unused 'reg2' field from soc_enum struct ASoC: Consolidate enum and value enum controls ASoC: dapm: Consolidate MUXs and value MUXs ASoC: Add macros for defining virtual enums ASoC: adau1373: Use SOC_ENUM_SINGLE_VIRT_DECL() ASoC: max98090: Use SOC_ENUM_SINGLE_VIRT_DECL() ASoC: mc13783: Use SOC_ENUM_SINGLE_VIRT_DECL() ASoC: wm8994: Use SOC_ENUM_SINGLE_VIRT_DECL() ASoC: wm8995: Use SOC_ENUM_SINGLE_VIRT_DECL() ASoC: dapm: Consolidate MUXs and virtual MUXs ASoC: dapm: Break dapm_set_path_status() appart include/sound/soc-dapm.h | 38 +---- include/sound/soc.h | 41 +++-- sound/soc/codecs/adau1373.c | 4 +- sound/soc/codecs/max98090.c | 2 +- sound/soc/codecs/mc13783.c | 6 +- sound/soc/codecs/wm8994.c | 3 +- sound/soc/codecs/wm8995.c | 2 +- sound/soc/soc-core.c | 101 ++---------- sound/soc/soc-dapm.c | 369 +++++++++----------------------------------- 9 files changed, 136 insertions(+), 430 deletions(-) -- 1.8.0

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[alsa-devel] [RFC PATCH] ALSA: hda - Fix CORB reset to follow specification
by David Henningsson 28 Feb '14

28 Feb '14

According to the HDA spec, we must write 1 to bit 15 on a CORBRP reset, read back 1, then write 0, then read back 0. This must be done while the DMA is not running. We accidentaly ended up writing back the 0 by using a writel instead of a writew to CORBWP. This caused occasional controller failure on Bay Trail hardware. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/hda_intel.c | 24 +++++++++++++++++++++++- 1 file changed, 23 insertions(+), 1 deletion(-) Hi Takashi, A while ago I researched a spurious response on some Bay Trail hardware, so I proofread some of our CORB/RIRB code and found the below. When applied, the spurious responses disappeared but other problems might have appeared instead. I don't have the hardware here so I'm not 100% sure. So I'm hesitant to actually apply it, especially as the existing non-compliant way seems to work for all other chips. Still wanted to show the issue to you. Also the timeout (1 ms) is completely arbitrarily as there are no time limits written in the HDA specification. diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index d8d9bf3..59156f1 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -775,6 +775,8 @@ static int azx_alloc_cmd_io(struct azx *chip) static void azx_init_cmd_io(struct azx *chip) { + int timeout; + spin_lock_irq(&chip->reg_lock); /* CORB set up */ chip->corb.addr = chip->rb.addr; @@ -786,8 +788,28 @@ static void azx_init_cmd_io(struct azx *chip) azx_writeb(chip, CORBSIZE, 0x02); /* set the corb write pointer to 0 */ azx_writew(chip, CORBWP, 0); + /* reset the corb hw read pointer */ azx_writew(chip, CORBRP, ICH6_CORBRP_RST); + for (timeout = 1000; timeout > 0; timeout--) { + if ((azx_readw(chip, CORBRP) & ICH6_CORBRP_RST) == ICH6_CORBRP_RST) + break; + udelay(1); + } + if (timeout <= 0) + snd_printk(KERN_ERR SFX "%s: CORB reset timeout - CORBRP = %d\n", + pci_name(chip->pci), (int) azx_readw(chip, CORBRP)); + + azx_writew(chip, CORBRP, 0); + for (timeout = 1000; timeout > 0; timeout--) { + if (azx_readw(chip, CORBRP) == 0) + break; + udelay(1); + } + if (timeout <= 0) + snd_printk(KERN_ERR SFX "%s: CORB reset timeout, CORBRP = %d\n", + pci_name(chip->pci), azx_readw(chip, CORBRP)); + /* enable corb dma */ azx_writeb(chip, CORBCTL, ICH6_CORBCTL_RUN); @@ -862,7 +884,7 @@ static int azx_corb_send_cmd(struct hda_bus *bus, u32 val) chip->rirb.cmds[addr]++; chip->corb.buf[wp] = cpu_to_le32(val); - azx_writel(chip, CORBWP, wp); + azx_writew(chip, CORBWP, wp); spin_unlock_irq(&chip->reg_lock); -- 1.7.9.5

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Re: [alsa-devel] [PATCH 00/48] Rewrite kernel messages with standard printk helpers
by Takashi Iwai 28 Feb '14

28 Feb '14

At Thu, 27 Feb 2014 15:32:53 +0800, Raymond Yau wrote: > > 2014-02-26 23:53 GMT+08:00 Takashi Iwai <tiwai(a)suse.de>: > > > Hi, > > > > this is another long series of patches for trivial cleanups. > > This time, they replace the printk and snd_print*() calls in ALSA PCI > > and USB drivers with dev_err() and co. This will give nicer and more > > consistent outputs. > > > > The only drawback by these replacements is that alsa-info.sh won't be > > able to catch the dmesg lines because it doesn't contain any longer > > the word "ALSA", but only show the module name like "snd_intel8x0". > > > > I'm going to fix alsa-info.sh on git soon later, but if you guys think > > "oh no it's disaster!", let me know. We need to reconsider a better > > way to put the known markers (like explicitly putting "ALSA" at the > > beginning.) > > > > > seem missing au88x0 Yes, because some driver codes are horrible to work on. Patches are welcome. > can the driver build as modules in kernel 3.11 (ubuntu 13.10) ? The dev_err() and co exist since ages ago, so it shouldn't be a problem. These patches, however, need the previous change of snd_card_new() to make the device pointer available in time. No idea about Ubuntu. Takashi

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[alsa-devel] [PATCH] ASoC: simple-card: cpu_dai_name creates confusion when DT case
by Kuninori Morimoto 28 Feb '14

28 Feb '14

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Basically, soc_bind_dai_link() checks cpu_dai->dev->of_node and dai_link->cpu_of_node in DT case. But after that it will check cpu_dai->name and dai_link->cpu_dai_name too. On the other hand, snd_soc_dai :: name is created by fmt_single_name() or fmt_multiple_name(). There is no confusion if dai name is created by fmt_multiple_name(), since cpu_dai->name is same as dai_link->cpu_dai_name. but, if dai name is created by fmt_single_name(), CPU DAI never match. Thus, simple-card not set dai_link->cpu_dai_name if DT case to skip naming match on soc_bind_dai_link() Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/generic/simple-card.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 2a1b1b5..4a0e06e 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -242,7 +242,8 @@ static int asoc_simple_card_probe(struct platform_device *pdev) */ cinfo->snd_link.name = cinfo->name; cinfo->snd_link.stream_name = cinfo->name; - cinfo->snd_link.cpu_dai_name = cinfo->cpu_dai.name; + if (!of_cpu) + 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; -- 1.7.9.5

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[alsa-devel] [PATCH 0/2] ASoC: ARM Cirrus Logic CLPS711X DAI driver
by Alexander Shiyan 28 Feb '14

28 Feb '14

This series adds sound support for the Cirrus Logic CLPS711X CPUs. Driver is designed (and tested) for using with generic sound card. Alexander Shiyan (2): ASoC: Add driver for CLPS711X DAI interface ASoC: cirrus: clps711x: Add bindings documentation for the CLPS711X DAI .../bindings/sound/cirrus,clps711x-dai.txt | 20 + sound/soc/cirrus/Kconfig | 8 + sound/soc/cirrus/Makefile | 5 + sound/soc/cirrus/clps711x-dai.c | 522 +++++++++++++++++++++ sound/soc/cirrus/clps711x-dai.h | 61 +++ sound/soc/cirrus/clps711x-fiq.S | 56 +++ 6 files changed, 672 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/cirrus,clps711x-dai.txt create mode 100644 sound/soc/cirrus/clps711x-dai.c create mode 100644 sound/soc/cirrus/clps711x-dai.h create mode 100644 sound/soc/cirrus/clps711x-fiq.S -- 1.8.3.2

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[alsa-devel] [PATCH v2] ASOC: dapm: add params_select callback
by nikesh＠opensource.wolfsonmicro.com 28 Feb '14

28 Feb '14

From: nikesh <Nikesh.Oswal(a)wolfsonmicro.com> dai-link params for codec-codec links were fixed. The fixed link between codec and another chip which may be another codec, baseband, bluetooth codec etc may require run time configuaration changes. This change provides an optional callback to select one of the param from a list of params. Signed-off-by: nikesh <Nikesh.Oswal(a)wolfsonmicro.com> --- Change since v1: Used an index to select from a list of params, instead of overwriting the param. Thanks, Nikesh include/sound/soc-dapm.h | 3 ++- include/sound/soc.h | 3 +++ sound/soc/soc-core.c | 4 ++-- sound/soc/soc-dapm.c | 20 ++++++++++++++++++-- 4 files changed, 25 insertions(+), 5 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 68d92e3..12b0d9e 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -416,7 +416,8 @@ void snd_soc_dapm_connect_dai_link_widgets(struct snd_soc_card *card); int snd_soc_dapm_new_pcm(struct snd_soc_card *card, const struct snd_soc_pcm_stream *params, struct snd_soc_dapm_widget *source, - struct snd_soc_dapm_widget *sink); + struct snd_soc_dapm_widget *sink, + struct snd_soc_dai_link *dai_link); /* dapm path setup */ int snd_soc_dapm_new_widgets(struct snd_soc_card *card); diff --git a/include/sound/soc.h b/include/sound/soc.h index 9a00147..20110aa 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -876,6 +876,9 @@ struct snd_soc_dai_link { int be_id; /* optional ID for machine driver BE identification */ const struct snd_soc_pcm_stream *params; + /* optional params selection for dai links */ + int (*params_select)(const struct snd_soc_pcm_stream *params, + unsigned int *param_index, int event); unsigned int dai_fmt; /* format to set on init */ diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index fe1df50..4f03e88 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1469,7 +1469,7 @@ static int soc_probe_link_dais(struct snd_soc_card *card, int num, int order) capture_w = cpu_dai->capture_widget; if (play_w && capture_w) { ret = snd_soc_dapm_new_pcm(card, dai_link->params, - capture_w, play_w); + capture_w, play_w, dai_link); if (ret != 0) { dev_err(card->dev, "ASoC: Can't link %s to %s: %d\n", play_w->name, capture_w->name, ret); @@ -1481,7 +1481,7 @@ static int soc_probe_link_dais(struct snd_soc_card *card, int num, int order) capture_w = codec_dai->capture_widget; if (play_w && capture_w) { ret = snd_soc_dapm_new_pcm(card, dai_link->params, - capture_w, play_w); + capture_w, play_w, dai_link); if (ret != 0) { dev_err(card->dev, "ASoC: Can't link %s to %s: %d\n", play_w->name, capture_w->name, ret); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index dc8ff13..2e2f633 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3379,10 +3379,12 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, struct snd_soc_dapm_path *source_p, *sink_p; struct snd_soc_dai *source, *sink; const struct snd_soc_pcm_stream *config = w->params; + struct snd_soc_dai_link *dai_link = w->priv; struct snd_pcm_substream substream; struct snd_pcm_hw_params *params = NULL; u64 fmt; - int ret; + int ret = 0; + unsigned int param_index; if (WARN_ON(!config) || WARN_ON(list_empty(&w->sources) || list_empty(&w->sinks))) @@ -3402,6 +3404,18 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, source = source_p->source->priv; sink = sink_p->sink->priv; + if (dai_link && dai_link->params_select) { + ret = dai_link->params_select(config, &param_index, event); + if (ret < 0) { + dev_err(w->dapm->dev, + "ASoC: params_select for dai link widget failed %d\n", + ret); + goto out; + } + if (param_index > 0) + config += param_index; + } + /* Be a little careful as we don't want to overflow the mask array */ if (config->formats) { fmt = ffs(config->formats) - 1; @@ -3485,7 +3499,8 @@ out: int snd_soc_dapm_new_pcm(struct snd_soc_card *card, const struct snd_soc_pcm_stream *params, struct snd_soc_dapm_widget *source, - struct snd_soc_dapm_widget *sink) + struct snd_soc_dapm_widget *sink, + struct snd_soc_dai_link *dai_link) { struct snd_soc_dapm_route routes[2]; struct snd_soc_dapm_widget template; @@ -3517,6 +3532,7 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, } w->params = params; + w->priv = (void *)dai_link; memset(&routes, 0, sizeof(routes)); -- 1.7.9.5

2 1

[alsa-devel] [PATCH 1/2] ASoC: fsl: Separation of the main audio options and boards
by Alexander Shiyan 28 Feb '14

28 Feb '14

This patch provide separation of the main audio options and boards. Signed-off-by: Alexander Shiyan <shc_work(a)mail.ru> --- sound/soc/fsl/Kconfig | 36 ++++++++++++++++++++++-------------- 1 file changed, 22 insertions(+), 14 deletions(-) diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 07f8f14..ecc1092 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -1,3 +1,5 @@ +menu "SoC Audio for Freescale CPUs" + config SND_SOC_FSL_SAI tristate select SND_SOC_GENERIC_DMAENGINE_PCM @@ -14,13 +16,27 @@ config SND_SOC_FSL_ESAI config SND_SOC_FSL_UTILS tristate -menuconfig SND_POWERPC_SOC +config SND_SOC_IMX_PCM_DMA + tristate + select SND_SOC_GENERIC_DMAENGINE_PCM + +config SND_SOC_IMX_AUDMUX + tristate + +config SND_POWERPC_SOC tristate "SoC Audio for Freescale PowerPC CPUs" depends on FSL_SOC || PPC_MPC52xx help Say Y or M if you want to add support for codecs attached to the PowerPC CPUs. +config SND_IMX_SOC + tristate "SoC Audio for Freescale i.MX CPUs" + depends on ARCH_MXC || COMPILE_TEST + help + Say Y or M if you want to add support for codecs attached to + the i.MX CPUs. + if SND_POWERPC_SOC config SND_MPC52xx_DMA @@ -29,6 +45,8 @@ config SND_MPC52xx_DMA config SND_SOC_POWERPC_DMA tristate +comment "SoC Audio support for Freescale PPC boards:" + config SND_SOC_MPC8610_HPCD tristate "ALSA SoC support for the Freescale MPC8610 HPCD board" # I2C is necessary for the CS4270 driver @@ -106,13 +124,6 @@ config SND_MPC52xx_SOC_EFIKA endif # SND_POWERPC_SOC -menuconfig SND_IMX_SOC - tristate "SoC Audio for Freescale i.MX CPUs" - depends on ARCH_MXC || COMPILE_TEST - help - Say Y or M if you want to add support for codecs attached to - the i.MX CPUs. - if SND_IMX_SOC config SND_SOC_IMX_SSI @@ -122,12 +133,7 @@ config SND_SOC_IMX_PCM_FIQ tristate select FIQ -config SND_SOC_IMX_PCM_DMA - tristate - select SND_SOC_GENERIC_DMAENGINE_PCM - -config SND_SOC_IMX_AUDMUX - tristate +comment "SoC Audio support for Freescale i.MX boards:" config SND_MXC_SOC_WM1133_EV1 tristate "Audio on the i.MX31ADS with WM1133-EV1 fitted" @@ -219,3 +225,5 @@ config SND_SOC_IMX_MC13783 select SND_SOC_IMX_PCM_DMA endif # SND_IMX_SOC + +endmenu -- 1.8.3.2

2 7

[alsa-devel] [PATCH 1/3] ALSA: hda - Remove dependency on bus->pci in hda_beep.c
by Takashi Iwai 27 Feb '14

27 Feb '14

The default parent device can be obtained directly via card object, so we don't need to rely on pci->dev.parent. Since there is no access to pci_dev, we can reduce the inclusion of linux/pci.h, too. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_beep.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c index 0589b39cda6e..d8c437a94559 100644 --- a/sound/pci/hda/hda_beep.c +++ b/sound/pci/hda/hda_beep.c @@ -20,7 +20,6 @@ */ #include <linux/input.h> -#include <linux/pci.h> #include <linux/slab.h> #include <linux/workqueue.h> #include <linux/export.h> @@ -167,7 +166,7 @@ static int snd_hda_do_attach(struct hda_beep *beep) input_dev->evbit[0] = BIT_MASK(EV_SND); input_dev->sndbit[0] = BIT_MASK(SND_BELL) | BIT_MASK(SND_TONE); input_dev->event = snd_hda_beep_event; - input_dev->dev.parent = &codec->bus->pci->dev; + input_dev->dev.parent = codec->bus->card->dev; input_set_drvdata(input_dev, beep); err = input_register_device(input_dev); -- 1.8.5.2

3 13

[alsa-devel] NUC D34010WYK HDMI Audio does not work...
by Robert Krakora 27 Feb '14

27 Feb '14

Hi All, We have a NUC D34010WYK on which I have installed several distros/kernels in a, so far, vain attempt to get HDMI audio to function via ALSA. I upload and alsainfo snap shot here: http://www.alsa-project.org/db/?f=d5f75cecee1195591d85be99c62db09bcd2bbdaf The latest distro I tried was an Ubuntu 14.04 daily from Feburary 18th. I also found this link where someone had tried unsuccessfully to get HDMI audio from this same NUC model unsuccessfully with Ubuntu 13.10: http://ubuntuforums.org/showthread.php?t=2202727 Any and all help will be greatly appreciated! :-) Best Regards, Rob -- Rob Krakora, Senior Software Engineer MessageNet Systems 101 E Carmel Dr, Suite 105 Carmel, IN 46032 MessageNetSystems.com<http://www.messagenetcommunicationsystems.com/?utm_source=email+signature&u…> Rob.Krakora(a)MessageNetSystems.com <rob.krakora(a)messagenetsystems.com> P: 317.566.1677, 212 F: 317.663.0808 For the latest news, information, and blogs, please be sure to visit, follow, and like us... <http://www.messagenetcommunicationsystems.com/get-the-message-out-blog/?utm…> <http://www.youtube.com/user/MessageNetConnection/feed> <http://www.linkedin.com/company/messagenet-systems> <http://twitter.com/MessageNet> <http://www.facebook.com/MessageNetsystems>

2 4

[alsa-devel] [PATCH] ALSA: hda/sigmatel - Allow auto-switching for dock line-in of HP laptops
by Takashi Iwai 27 Feb '14

27 Feb '14

Many HP laptops with STAC codecs have a docking station port and BIOS sets the pins for the input on the dock as a line in. Because the generic parser doesn't handle a line in pin as auto-switchable, this resulted in the manual capture source selection on these laptops. However, from the usability POV, the automatic switching is easier. This patch adds the line_in_auto_switch hint in the fixup function for these laptops. Even if no dock port is present, this should be harmless as the generic parser allows the auto-switching only in a limited situation (all three pins are located in different positions). Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_sigmatel.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index f3784808758e..75515b494034 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -2089,6 +2089,9 @@ static void stac92hd83xxx_fixup_hp(struct hda_codec *codec, codec_dbg(codec, "mute LED gpio %d polarity %d\n", spec->gpio_led, spec->gpio_led_polarity); + + /* allow auto-switching of dock line-in */ + spec->gen.line_in_auto_switch = true; } static void stac92hd83xxx_fixup_hp_zephyr(struct hda_codec *codec, -- 1.9.0

1 0

[alsa-devel] [PATCH] ALSA: Move EXPORT_SYMBOL() in appropriate places
by Takashi Iwai 27 Feb '14

27 Feb '14

Just a cleanup to follow the standard coding style. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/hwdep.c | 3 +-- sound/core/rawmidi.c | 35 +++++++++++++++++------------------ 2 files changed, 18 insertions(+), 20 deletions(-) diff --git a/sound/core/hwdep.c b/sound/core/hwdep.c index d6eb3ef3e3c7..69459e5f712e 100644 --- a/sound/core/hwdep.c +++ b/sound/core/hwdep.c @@ -395,6 +395,7 @@ int snd_hwdep_new(struct snd_card *card, char *id, int device, *rhwdep = hwdep; return 0; } +EXPORT_SYMBOL(snd_hwdep_new); static int snd_hwdep_free(struct snd_hwdep *hwdep) { @@ -567,5 +568,3 @@ static void __exit alsa_hwdep_exit(void) module_init(alsa_hwdep_init) module_exit(alsa_hwdep_exit) - -EXPORT_SYMBOL(snd_hwdep_new); diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c index 801c86133937..6fc71a4c8a51 100644 --- a/sound/core/rawmidi.c +++ b/sound/core/rawmidi.c @@ -172,6 +172,7 @@ int snd_rawmidi_drop_output(struct snd_rawmidi_substream *substream) spin_unlock_irqrestore(&runtime->lock, flags); return 0; } +EXPORT_SYMBOL(snd_rawmidi_drop_output); int snd_rawmidi_drain_output(struct snd_rawmidi_substream *substream) { @@ -203,6 +204,7 @@ int snd_rawmidi_drain_output(struct snd_rawmidi_substream *substream) } return err; } +EXPORT_SYMBOL(snd_rawmidi_drain_output); int snd_rawmidi_drain_input(struct snd_rawmidi_substream *substream) { @@ -217,6 +219,7 @@ int snd_rawmidi_drain_input(struct snd_rawmidi_substream *substream) spin_unlock_irqrestore(&runtime->lock, flags); return 0; } +EXPORT_SYMBOL(snd_rawmidi_drain_input); /* look for an available substream for the given stream direction; * if a specific subdevice is given, try to assign it @@ -354,6 +357,7 @@ int snd_rawmidi_kernel_open(struct snd_card *card, int device, int subdevice, module_put(rmidi->card->module); return err; } +EXPORT_SYMBOL(snd_rawmidi_kernel_open); static int snd_rawmidi_open(struct inode *inode, struct file *file) { @@ -532,6 +536,7 @@ int snd_rawmidi_kernel_release(struct snd_rawmidi_file *rfile) module_put(rmidi->card->module); return 0; } +EXPORT_SYMBOL(snd_rawmidi_kernel_release); static int snd_rawmidi_release(struct inode *inode, struct file *file) { @@ -608,6 +613,7 @@ int snd_rawmidi_info_select(struct snd_card *card, struct snd_rawmidi_info *info } return -ENXIO; } +EXPORT_SYMBOL(snd_rawmidi_info_select); static int snd_rawmidi_info_select_user(struct snd_card *card, struct snd_rawmidi_info __user *_info) @@ -655,6 +661,7 @@ int snd_rawmidi_output_params(struct snd_rawmidi_substream *substream, substream->active_sensing = !params->no_active_sensing; return 0; } +EXPORT_SYMBOL(snd_rawmidi_output_params); int snd_rawmidi_input_params(struct snd_rawmidi_substream *substream, struct snd_rawmidi_params * params) @@ -680,6 +687,7 @@ int snd_rawmidi_input_params(struct snd_rawmidi_substream *substream, runtime->avail_min = params->avail_min; return 0; } +EXPORT_SYMBOL(snd_rawmidi_input_params); static int snd_rawmidi_output_status(struct snd_rawmidi_substream *substream, struct snd_rawmidi_status * status) @@ -935,6 +943,7 @@ int snd_rawmidi_receive(struct snd_rawmidi_substream *substream, spin_unlock_irqrestore(&runtime->lock, flags); return result; } +EXPORT_SYMBOL(snd_rawmidi_receive); static long snd_rawmidi_kernel_read1(struct snd_rawmidi_substream *substream, unsigned char __user *userbuf, @@ -977,6 +986,7 @@ long snd_rawmidi_kernel_read(struct snd_rawmidi_substream *substream, snd_rawmidi_input_trigger(substream, 1); return snd_rawmidi_kernel_read1(substream, NULL/*userbuf*/, buf, count); } +EXPORT_SYMBOL(snd_rawmidi_kernel_read); static ssize_t snd_rawmidi_read(struct file *file, char __user *buf, size_t count, loff_t *offset) @@ -1052,6 +1062,7 @@ int snd_rawmidi_transmit_empty(struct snd_rawmidi_substream *substream) spin_unlock_irqrestore(&runtime->lock, flags); return result; } +EXPORT_SYMBOL(snd_rawmidi_transmit_empty); /** * snd_rawmidi_transmit_peek - copy data from the internal buffer @@ -1108,6 +1119,7 @@ int snd_rawmidi_transmit_peek(struct snd_rawmidi_substream *substream, spin_unlock_irqrestore(&runtime->lock, flags); return result; } +EXPORT_SYMBOL(snd_rawmidi_transmit_peek); /** * snd_rawmidi_transmit_ack - acknowledge the transmission @@ -1143,6 +1155,7 @@ int snd_rawmidi_transmit_ack(struct snd_rawmidi_substream *substream, int count) spin_unlock_irqrestore(&runtime->lock, flags); return count; } +EXPORT_SYMBOL(snd_rawmidi_transmit_ack); /** * snd_rawmidi_transmit - copy from the buffer to the device @@ -1164,6 +1177,7 @@ int snd_rawmidi_transmit(struct snd_rawmidi_substream *substream, return count; return snd_rawmidi_transmit_ack(substream, count); } +EXPORT_SYMBOL(snd_rawmidi_transmit); static long snd_rawmidi_kernel_write1(struct snd_rawmidi_substream *substream, const unsigned char __user *userbuf, @@ -1225,6 +1239,7 @@ long snd_rawmidi_kernel_write(struct snd_rawmidi_substream *substream, { return snd_rawmidi_kernel_write1(substream, NULL, buf, count); } +EXPORT_SYMBOL(snd_rawmidi_kernel_write); static ssize_t snd_rawmidi_write(struct file *file, const char __user *buf, size_t count, loff_t *offset) @@ -1504,6 +1519,7 @@ int snd_rawmidi_new(struct snd_card *card, char *id, int device, *rrawmidi = rmidi; return 0; } +EXPORT_SYMBOL(snd_rawmidi_new); static void snd_rawmidi_free_substreams(struct snd_rawmidi_str *stream) { @@ -1687,6 +1703,7 @@ void snd_rawmidi_set_ops(struct snd_rawmidi *rmidi, int stream, list_for_each_entry(substream, &rmidi->streams[stream].substreams, list) substream->ops = ops; } +EXPORT_SYMBOL(snd_rawmidi_set_ops); /* * ENTRY functions @@ -1725,21 +1742,3 @@ static void __exit alsa_rawmidi_exit(void) module_init(alsa_rawmidi_init) module_exit(alsa_rawmidi_exit) - -EXPORT_SYMBOL(snd_rawmidi_output_params); -EXPORT_SYMBOL(snd_rawmidi_input_params); -EXPORT_SYMBOL(snd_rawmidi_drop_output); -EXPORT_SYMBOL(snd_rawmidi_drain_output); -EXPORT_SYMBOL(snd_rawmidi_drain_input); -EXPORT_SYMBOL(snd_rawmidi_receive); -EXPORT_SYMBOL(snd_rawmidi_transmit_empty); -EXPORT_SYMBOL(snd_rawmidi_transmit_peek); -EXPORT_SYMBOL(snd_rawmidi_transmit_ack); -EXPORT_SYMBOL(snd_rawmidi_transmit); -EXPORT_SYMBOL(snd_rawmidi_new); -EXPORT_SYMBOL(snd_rawmidi_set_ops); -EXPORT_SYMBOL(snd_rawmidi_info_select); -EXPORT_SYMBOL(snd_rawmidi_kernel_open); -EXPORT_SYMBOL(snd_rawmidi_kernel_release); -EXPORT_SYMBOL(snd_rawmidi_kernel_read); -EXPORT_SYMBOL(snd_rawmidi_kernel_write); -- 1.9.0

1 0

[alsa-devel] [GIT PUL] ASoC updates for v3.14
by Mark Brown 27 Feb '14

27 Feb '14

The following changes since commit 57374bb491facf716c172126b7e9b99c9e3f8b45: Merge remote-tracking branches 'asoc/fix/sta32x', 'asoc/fix/wm8400', 'asoc/fix/wm8770', 'asoc/fix/wm8900' and 'asoc/fix/wm8994' into asoc-linus (2014-02-23 12:20:34 +0900) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.14-rc4-2 for you to fetch changes up to 23308e88201c2c0036da23450f7fed6fd64ae1ef: Merge remote-tracking branch 'asoc/fix/wm8958' into asoc-linus (2014-02-27 20:26:10 +0900) ---------------------------------------------------------------- ASoC: Updates for v3.14 A few more driver specific bug fixes, all driver specific things that only affect users of those devices. ---------------------------------------------------------------- Lars-Peter Clausen (1): ASoC: wm8958-dsp: Fix firmware block loading Mark Brown (5): ASoC: da732x: Mark DC offset control registers volatile Merge tag 'asoc-v3.14-rc3' into asoc-linus Merge tag 'asoc-v3.14-rc4' into asoc-linus Merge remote-tracking branches 'asoc/fix/da732x' and 'asoc/fix/sta32x' into asoc-linus Merge remote-tracking branch 'asoc/fix/wm8958' into asoc-linus Takashi Iwai (1): ASoC: sta32x: Fix wrong enum for limiter2 release rate sound/soc/codecs/da732x.c | 12 ++++++++++++ sound/soc/codecs/sta32x.c | 2 +- sound/soc/codecs/wm8958-dsp2.c | 2 +- 3 files changed, 14 insertions(+), 2 deletions(-)

2 1

[alsa-devel] [PATCH] ASoC: io: Clean up snd_soc_codec_set_cache_io()
by Xiubo Li 27 Feb '14

27 Feb '14

Now that all users have been converted to regmap and the config.reg_bits and config.val_bits can be setted by each user through regmap core API. So these two params are redundant here. Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- @Mark and All I'm also thinking could we just discard snd_soc_codec_set_cache_io() calling from each individual driver to simply the code? And just bind it to devm_regmap_init_i2c() and devm_regmap_init_spi()... Is there any other limitations for snd_soc_codec_set_cache_io() usage? include/sound/soc.h | 1 - sound/soc/codecs/ad193x.c | 2 +- sound/soc/codecs/adau1373.c | 2 +- sound/soc/codecs/adav80x.c | 2 +- sound/soc/codecs/ak4535.c | 2 +- sound/soc/codecs/ak4641.c | 2 +- sound/soc/codecs/ak4642.c | 2 +- sound/soc/codecs/ak4671.c | 2 +- sound/soc/codecs/alc5623.c | 2 +- sound/soc/codecs/alc5632.c | 2 +- sound/soc/codecs/cq93vc.c | 2 +- sound/soc/codecs/cs4270.c | 2 +- sound/soc/codecs/cs42l51.c | 2 +- sound/soc/codecs/cs42l52.c | 2 +- sound/soc/codecs/cs42l73.c | 2 +- sound/soc/codecs/da7210.c | 2 +- sound/soc/codecs/da7213.c | 2 +- sound/soc/codecs/da732x.c | 2 +- sound/soc/codecs/da9055.c | 2 +- sound/soc/codecs/isabelle.c | 2 +- sound/soc/codecs/lm49453.c | 2 +- sound/soc/codecs/max9768.c | 2 +- sound/soc/codecs/max98088.c | 2 +- sound/soc/codecs/max98090.c | 2 +- sound/soc/codecs/max98095.c | 2 +- sound/soc/codecs/max9850.c | 2 +- sound/soc/codecs/mc13783.c | 2 +- sound/soc/codecs/ml26124.c | 2 +- sound/soc/codecs/rt5631.c | 2 +- sound/soc/codecs/rt5640.c | 2 +- sound/soc/codecs/sgtl5000.c | 2 +- sound/soc/codecs/sn95031.c | 2 +- sound/soc/codecs/ssm2518.c | 2 +- sound/soc/codecs/ssm2602.c | 2 +- sound/soc/codecs/sta32x.c | 2 +- sound/soc/codecs/sta529.c | 2 +- sound/soc/codecs/tlv320aic23.c | 2 +- sound/soc/codecs/tlv320aic26.c | 2 +- sound/soc/codecs/tlv320aic32x4.c | 2 +- sound/soc/codecs/tlv320aic3x.c | 2 +- sound/soc/codecs/wm2000.c | 2 +- sound/soc/codecs/wm2200.c | 2 +- sound/soc/codecs/wm5100.c | 2 +- sound/soc/codecs/wm5102.c | 2 +- sound/soc/codecs/wm5110.c | 2 +- sound/soc/codecs/wm8350.c | 2 +- sound/soc/codecs/wm8400.c | 2 +- sound/soc/codecs/wm8510.c | 2 +- sound/soc/codecs/wm8523.c | 2 +- sound/soc/codecs/wm8580.c | 2 +- sound/soc/codecs/wm8711.c | 2 +- sound/soc/codecs/wm8728.c | 2 +- sound/soc/codecs/wm8731.c | 2 +- sound/soc/codecs/wm8737.c | 2 +- sound/soc/codecs/wm8741.c | 2 +- sound/soc/codecs/wm8750.c | 2 +- sound/soc/codecs/wm8753.c | 2 +- sound/soc/codecs/wm8770.c | 2 +- sound/soc/codecs/wm8776.c | 2 +- sound/soc/codecs/wm8804.c | 2 +- sound/soc/codecs/wm8900.c | 2 +- sound/soc/codecs/wm8903.c | 2 +- sound/soc/codecs/wm8904.c | 2 +- sound/soc/codecs/wm8940.c | 2 +- sound/soc/codecs/wm8955.c | 2 +- sound/soc/codecs/wm8960.c | 2 +- sound/soc/codecs/wm8961.c | 2 +- sound/soc/codecs/wm8962.c | 2 +- sound/soc/codecs/wm8971.c | 2 +- sound/soc/codecs/wm8974.c | 2 +- sound/soc/codecs/wm8978.c | 2 +- sound/soc/codecs/wm8983.c | 2 +- sound/soc/codecs/wm8985.c | 2 +- sound/soc/codecs/wm8988.c | 2 +- sound/soc/codecs/wm8990.c | 2 +- sound/soc/codecs/wm8991.c | 2 +- sound/soc/codecs/wm8993.c | 2 +- sound/soc/codecs/wm8994.c | 2 +- sound/soc/codecs/wm8995.c | 2 +- sound/soc/codecs/wm8996.c | 2 +- sound/soc/codecs/wm8997.c | 2 +- sound/soc/codecs/wm9081.c | 2 +- sound/soc/codecs/wm9090.c | 2 +- sound/soc/soc-core.c | 2 +- sound/soc/soc-io.c | 9 --------- 85 files changed, 83 insertions(+), 93 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index ecfb334..0a1d732 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -404,7 +404,6 @@ int snd_soc_codec_readable_register(struct snd_soc_codec *codec, int snd_soc_codec_writable_register(struct snd_soc_codec *codec, unsigned int reg); int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec, - int addr_bits, int data_bits, enum snd_soc_control_type control); int snd_soc_cache_sync(struct snd_soc_codec *codec); int snd_soc_cache_init(struct snd_soc_codec *codec); diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c index 9381a76..052495f 100644 --- a/sound/soc/codecs/ad193x.c +++ b/sound/soc/codecs/ad193x.c @@ -325,7 +325,7 @@ static int ad193x_codec_probe(struct snd_soc_codec *codec) int ret; codec->control_data = ad193x->regmap; - ret = snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/adau1373.c b/sound/soc/codecs/adau1373.c index 5765c22..69b434c 100644 --- a/sound/soc/codecs/adau1373.c +++ b/sound/soc/codecs/adau1373.c @@ -1379,7 +1379,7 @@ static int adau1373_probe(struct snd_soc_codec *codec) int ret; int i; - ret = snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret) { dev_err(codec->dev, "failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/adav80x.c b/sound/soc/codecs/adav80x.c index 9eb53c8..0066feb 100644 --- a/sound/soc/codecs/adav80x.c +++ b/sound/soc/codecs/adav80x.c @@ -804,7 +804,7 @@ 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); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret) { dev_err(codec->dev, "failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c index 684fe91..9d4f878 100644 --- a/sound/soc/codecs/ak4535.c +++ b/sound/soc/codecs/ak4535.c @@ -392,7 +392,7 @@ static int ak4535_probe(struct snd_soc_codec *codec) int ret; codec->control_data = ak4535->regmap; - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/ak4641.c b/sound/soc/codecs/ak4641.c index 684b56f..5d81cbd 100644 --- a/sound/soc/codecs/ak4641.c +++ b/sound/soc/codecs/ak4641.c @@ -521,7 +521,7 @@ static int ak4641_probe(struct snd_soc_codec *codec) { int ret; - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret != 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index 1f646c6..7e7621a 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -467,7 +467,7 @@ static int ak4642_probe(struct snd_soc_codec *codec) { int ret; - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/ak4671.c b/sound/soc/codecs/ak4671.c index deb2b44..46fd465 100644 --- a/sound/soc/codecs/ak4671.c +++ b/sound/soc/codecs/ak4671.c @@ -615,7 +615,7 @@ static int ak4671_probe(struct snd_soc_codec *codec) { int ret; - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/alc5623.c b/sound/soc/codecs/alc5623.c index ed50625..030dcda 100644 --- a/sound/soc/codecs/alc5623.c +++ b/sound/soc/codecs/alc5623.c @@ -905,7 +905,7 @@ static int alc5623_probe(struct snd_soc_codec *codec) int ret; codec->control_data = alc5623->regmap; - ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/alc5632.c b/sound/soc/codecs/alc5632.c index d885056..2cd5d27 100644 --- a/sound/soc/codecs/alc5632.c +++ b/sound/soc/codecs/alc5632.c @@ -1065,7 +1065,7 @@ static int alc5632_probe(struct snd_soc_codec *codec) codec->control_data = alc5632->regmap; - ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret != 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/cq93vc.c b/sound/soc/codecs/cq93vc.c index 43737a27..d4f8714 100644 --- a/sound/soc/codecs/cq93vc.c +++ b/sound/soc/codecs/cq93vc.c @@ -140,7 +140,7 @@ static int cq93vc_probe(struct snd_soc_codec *codec) 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, SND_SOC_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..988a00a 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -509,7 +509,7 @@ static int cs4270_probe(struct snd_soc_codec *codec) /* 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); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "failed to set cache I/O (ret=%i)\n", ret); return ret; diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index 5caf75b..42564fb 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -488,7 +488,7 @@ static int cs42l51_probe(struct snd_soc_codec *codec) { int ret, reg; - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index be455ea..d7a5967 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -1112,7 +1112,7 @@ static int cs42l52_probe(struct snd_soc_codec *codec) int ret; codec->control_data = cs42l52->regmap; - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c index 06f4291..efa3b20 100644 --- a/sound/soc/codecs/cs42l73.c +++ b/sound/soc/codecs/cs42l73.c @@ -1350,7 +1350,7 @@ static int cs42l73_probe(struct snd_soc_codec *codec) codec->control_data = cs42l73->regmap; - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index 01e55fc..3cfbab6 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -1076,7 +1076,7 @@ static int da7210_probe(struct snd_soc_codec *codec) 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); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/da7213.c b/sound/soc/codecs/da7213.c index 439d103..bf6b840 100644 --- a/sound/soc/codecs/da7213.c +++ b/sound/soc/codecs/da7213.c @@ -1398,7 +1398,7 @@ static int da7213_probe(struct snd_soc_codec *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); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/da732x.c b/sound/soc/codecs/da732x.c index 05f07fb..f32f219 100644 --- a/sound/soc/codecs/da732x.c +++ b/sound/soc/codecs/da732x.c @@ -1508,7 +1508,7 @@ static int da732x_probe(struct snd_soc_codec *codec) codec->control_data = da732x->regmap; - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret != 0) { dev_err(codec->dev, "Failed to register codec.\n"); goto err; diff --git a/sound/soc/codecs/da9055.c b/sound/soc/codecs/da9055.c index f118daa..187d8ce 100644 --- a/sound/soc/codecs/da9055.c +++ b/sound/soc/codecs/da9055.c @@ -1387,7 +1387,7 @@ static int da9055_probe(struct snd_soc_codec *codec) 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); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/isabelle.c b/sound/soc/codecs/isabelle.c index cb736dd..54ea3cf 100644 --- a/sound/soc/codecs/isabelle.c +++ b/sound/soc/codecs/isabelle.c @@ -1096,7 +1096,7 @@ static int isabelle_probe(struct snd_soc_codec *codec) codec->control_data = dev_get_regmap(codec->dev, NULL); - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/lm49453.c b/sound/soc/codecs/lm49453.c index e533594..8061103 100644 --- a/sound/soc/codecs/lm49453.c +++ b/sound/soc/codecs/lm49453.c @@ -1414,7 +1414,7 @@ static int lm49453_probe(struct snd_soc_codec *codec) codec->control_data = lm49453->regmap; - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/max9768.c b/sound/soc/codecs/max9768.c index 31f9156..896ae28 100644 --- a/sound/soc/codecs/max9768.c +++ b/sound/soc/codecs/max9768.c @@ -136,7 +136,7 @@ static int max9768_probe(struct snd_soc_codec *codec) int ret; codec->control_data = max9768->regmap; - ret = snd_soc_codec_set_cache_io(codec, 2, 6, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret) return ret; diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index 4b9274d..5355e1a 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -1914,7 +1914,7 @@ 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); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret != 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index a75752f..d01ac5d 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -2220,7 +2220,7 @@ static int max98090_probe(struct snd_soc_codec *codec) codec->control_data = max98090->regmap; - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret != 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/max98095.c b/sound/soc/codecs/max98095.c index 4c22582..02666ce 100644 --- a/sound/soc/codecs/max98095.c +++ b/sound/soc/codecs/max98095.c @@ -2238,7 +2238,7 @@ 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); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret != 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/max9850.c b/sound/soc/codecs/max9850.c index 82757eb..b7951b3 100644 --- a/sound/soc/codecs/max9850.c +++ b/sound/soc/codecs/max9850.c @@ -314,7 +314,7 @@ static int max9850_probe(struct snd_soc_codec *codec) { int ret; - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c index c605036..16725e9 100644 --- a/sound/soc/codecs/mc13783.c +++ b/sound/soc/codecs/mc13783.c @@ -615,7 +615,7 @@ static int mc13783_probe(struct snd_soc_codec *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, SND_SOC_REGMAP); 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 577fb87..3202eee 100644 --- a/sound/soc/codecs/ml26124.c +++ b/sound/soc/codecs/ml26124.c @@ -590,7 +590,7 @@ static int ml26124_probe(struct snd_soc_codec *codec) 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); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/rt5631.c b/sound/soc/codecs/rt5631.c index ce199d3..3287fe7 100644 --- a/sound/soc/codecs/rt5631.c +++ b/sound/soc/codecs/rt5631.c @@ -1574,7 +1574,7 @@ static int rt5631_probe(struct snd_soc_codec *codec) codec->control_data = rt5631->regmap; - ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret != 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 1a1e115..26b9988 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -1941,7 +1941,7 @@ static int rt5640_probe(struct snd_soc_codec *codec) rt5640->codec = codec; codec->control_data = rt5640->regmap; - ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret != 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index ab4754a..8758c3b 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -1354,7 +1354,7 @@ static int sgtl5000_probe(struct snd_soc_codec *codec) /* setup i2c data ops */ codec->control_data = sgtl5000->regmap; - ret = snd_soc_codec_set_cache_io(codec, 16, 16, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c index bca7d02..671c45b 100644 --- a/sound/soc/codecs/sn95031.c +++ b/sound/soc/codecs/sn95031.c @@ -825,7 +825,7 @@ 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); + snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); /* PCM interface config * This sets the pcm rx slot conguration to max 6 slots diff --git a/sound/soc/codecs/ssm2518.c b/sound/soc/codecs/ssm2518.c index 806f3d8..ce4f890 100644 --- a/sound/soc/codecs/ssm2518.c +++ b/sound/soc/codecs/ssm2518.c @@ -652,7 +652,7 @@ static int ssm2518_probe(struct snd_soc_codec *codec) int ret; codec->control_data = ssm2518->regmap; - ret = snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index 1294709..912fa5d 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -563,7 +563,7 @@ static int ssm260x_codec_probe(struct snd_soc_codec *codec) int ret; codec->control_data = ssm2602->regmap; - ret = snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c index ea78c17..93e67c8 100644 --- a/sound/soc/codecs/sta32x.c +++ b/sound/soc/codecs/sta32x.c @@ -876,7 +876,7 @@ static int sta32x_probe(struct snd_soc_codec *codec) * then do the I2C transactions itself. */ codec->control_data = sta32x->regmap; - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "failed to set cache I/O (ret=%i)\n", ret); goto err; diff --git a/sound/soc/codecs/sta529.c b/sound/soc/codecs/sta529.c index f15b0e3..9530e079 100644 --- a/sound/soc/codecs/sta529.c +++ b/sound/soc/codecs/sta529.c @@ -326,7 +326,7 @@ static int sta529_probe(struct snd_soc_codec *codec) int ret; codec->control_data = sta529->regmap; - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index 139f11f..70cb324 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -561,7 +561,7 @@ static int tlv320aic23_probe(struct snd_soc_codec *codec) { int ret; - ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c index ff5f23d..41186ef 100644 --- a/sound/soc/codecs/tlv320aic26.c +++ b/sound/soc/codecs/tlv320aic26.c @@ -296,7 +296,7 @@ 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); + snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); aic26->codec = codec; diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index c6bd7e7..d79a9ec 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -614,7 +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); + snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (gpio_is_valid(aic32x4->rstn_gpio)) { ndelay(10); diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 470fbfb..598625b 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1344,7 +1344,7 @@ 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); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret != 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c index 8ae5027..a8dde7b 100644 --- a/sound/soc/codecs/wm2000.c +++ b/sound/soc/codecs/wm2000.c @@ -786,7 +786,7 @@ 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); + snd_soc_codec_set_cache_io(codec, 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 1e0a083..58b9fb0 100644 --- a/sound/soc/codecs/wm2200.c +++ b/sound/soc/codecs/wm2200.c @@ -1557,7 +1557,7 @@ static int wm2200_probe(struct snd_soc_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); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret != 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index d3fa65f..5874238 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -2345,7 +2345,7 @@ static int wm5100_probe(struct snd_soc_codec *codec) wm5100->codec = codec; codec->control_data = wm5100->regmap; - ret = snd_soc_codec_set_cache_io(codec, 16, 16, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret != 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index 293dffc..50593bb 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -1762,7 +1762,7 @@ static int wm5102_codec_probe(struct snd_soc_codec *codec) 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, SND_SOC_REGMAP); if (ret != 0) return ret; diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 4de2bf1..9cb4257 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -1590,7 +1590,7 @@ static int wm5110_codec_probe(struct snd_soc_codec *codec) 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, SND_SOC_REGMAP); if (ret != 0) return ret; diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index a183dcf..35c30a9 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -1507,7 +1507,7 @@ static int wm8350_codec_probe(struct snd_soc_codec *codec) codec->control_data = wm8350->regmap; - snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); + snd_soc_codec_set_cache_io(codec, SND_SOC_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..10b1172 100644 --- a/sound/soc/codecs/wm8400.c +++ b/sound/soc/codecs/wm8400.c @@ -1319,7 +1319,7 @@ static int wm8400_codec_probe(struct snd_soc_codec *codec) 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, SND_SOC_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..e531b9d 100644 --- a/sound/soc/codecs/wm8510.c +++ b/sound/soc/codecs/wm8510.c @@ -591,7 +591,7 @@ static int wm8510_probe(struct snd_soc_codec *codec) { int ret; - ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret < 0) { printk(KERN_ERR "wm8510: failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/wm8523.c b/sound/soc/codecs/wm8523.c index 5dfd571..f5c17a0 100644 --- a/sound/soc/codecs/wm8523.c +++ b/sound/soc/codecs/wm8523.c @@ -398,7 +398,7 @@ static int wm8523_probe(struct snd_soc_codec *codec) wm8523->rate_constraint.count = ARRAY_SIZE(wm8523->rate_constraint_list); - ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret != 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index 318989a..249961b 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -869,7 +869,7 @@ 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); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/wm8711.c b/sound/soc/codecs/wm8711.c index d99f948..3673b90 100644 --- a/sound/soc/codecs/wm8711.c +++ b/sound/soc/codecs/wm8711.c @@ -367,7 +367,7 @@ static int wm8711_probe(struct snd_soc_codec *codec) { int ret; - ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/wm8728.c b/sound/soc/codecs/wm8728.c index cd89033..c2d2835 100644 --- a/sound/soc/codecs/wm8728.c +++ b/sound/soc/codecs/wm8728.c @@ -230,7 +230,7 @@ static int wm8728_probe(struct snd_soc_codec *codec) { int ret; - ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret < 0) { printk(KERN_ERR "wm8728: failed to configure cache I/O: %d\n", ret); diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index d9655f9..3f877d2 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -584,7 +584,7 @@ static int wm8731_probe(struct snd_soc_codec *codec) int ret = 0, i; codec->control_data = wm8731->regmap; - ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/wm8737.c b/sound/soc/codecs/wm8737.c index ecc4e87..8f14ed8 100644 --- a/sound/soc/codecs/wm8737.c +++ b/sound/soc/codecs/wm8737.c @@ -570,7 +570,7 @@ 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); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret != 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c index dd02ebf..6a40489 100644 --- a/sound/soc/codecs/wm8741.c +++ b/sound/soc/codecs/wm8741.c @@ -429,7 +429,7 @@ 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); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret != 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); goto err_enable; diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index 78616a6..886d618 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -702,7 +702,7 @@ static int wm8750_probe(struct snd_soc_codec *codec) { int ret; - ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret < 0) { printk(KERN_ERR "wm8750: failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index be85da9..f5e4d1a 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -1472,7 +1472,7 @@ 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); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/wm8770.c b/sound/soc/codecs/wm8770.c index 5bce210..ff2e087 100644 --- a/sound/soc/codecs/wm8770.c +++ b/sound/soc/codecs/wm8770.c @@ -580,7 +580,7 @@ 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); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/wm8776.c b/sound/soc/codecs/wm8776.c index ef82467..70276d7 100644 --- a/sound/soc/codecs/wm8776.c +++ b/sound/soc/codecs/wm8776.c @@ -430,7 +430,7 @@ 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); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/wm8804.c b/sound/soc/codecs/wm8804.c index 72d12bb..ae7f031 100644 --- a/sound/soc/codecs/wm8804.c +++ b/sound/soc/codecs/wm8804.c @@ -548,7 +548,7 @@ static int wm8804_probe(struct snd_soc_codec *codec) codec->control_data = wm8804->regmap; - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache i/o: %d\n", ret); return ret; diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c index 43c2201..651908f 100644 --- a/sound/soc/codecs/wm8900.c +++ b/sound/soc/codecs/wm8900.c @@ -1180,7 +1180,7 @@ 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); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret != 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index b82b70a..0a92e99 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -1902,7 +1902,7 @@ static int wm8903_probe(struct snd_soc_codec *codec) wm8903->codec = codec; codec->control_data = wm8903->regmap; - ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret != 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 27299cd..f524481 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -2064,7 +2064,7 @@ static int wm8904_probe(struct snd_soc_codec *codec) return -EINVAL; } - ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret != 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c index 87f032d..f53033b 100644 --- a/sound/soc/codecs/wm8940.c +++ b/sound/soc/codecs/wm8940.c @@ -712,7 +712,7 @@ 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); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/wm8955.c b/sound/soc/codecs/wm8955.c index d4dcaec..5762e3c 100644 --- a/sound/soc/codecs/wm8955.c +++ b/sound/soc/codecs/wm8955.c @@ -897,7 +897,7 @@ static int wm8955_probe(struct snd_soc_codec *codec) codec->control_data = wm8955->regmap; - ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret != 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index f156010..1b90049 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -976,7 +976,7 @@ 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); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/wm8961.c b/sound/soc/codecs/wm8961.c index ce8fa6e..f18950c 100644 --- a/sound/soc/codecs/wm8961.c +++ b/sound/soc/codecs/wm8961.c @@ -839,7 +839,7 @@ static int wm8961_probe(struct snd_soc_codec *codec) int ret = 0; u16 reg; - ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret != 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 62af9dc..4199e0e 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3426,7 +3426,7 @@ static int wm8962_probe(struct snd_soc_codec *codec) wm8962->codec = codec; codec->control_data = wm8962->regmap; - ret = snd_soc_codec_set_cache_io(codec, 16, 16, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret != 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c index 67aba78..7dfee38 100644 --- a/sound/soc/codecs/wm8971.c +++ b/sound/soc/codecs/wm8971.c @@ -648,7 +648,7 @@ 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); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret < 0) { printk(KERN_ERR "wm8971: failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c index 6e16c43..3aa5821 100644 --- a/sound/soc/codecs/wm8974.c +++ b/sound/soc/codecs/wm8974.c @@ -593,7 +593,7 @@ 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); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c index a9e2f46..4564695 100644 --- a/sound/soc/codecs/wm8978.c +++ b/sound/soc/codecs/wm8978.c @@ -983,7 +983,7 @@ static int wm8978_probe(struct snd_soc_codec *codec) */ wm8978->sysclk = WM8978_PLL; codec->control_data = wm8978->regmap; - ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/wm8983.c b/sound/soc/codecs/wm8983.c index 58f0551..ff0f005 100644 --- a/sound/soc/codecs/wm8983.c +++ b/sound/soc/codecs/wm8983.c @@ -995,7 +995,7 @@ 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); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache i/o: %d\n", ret); return ret; diff --git a/sound/soc/codecs/wm8985.c b/sound/soc/codecs/wm8985.c index d786f2b..677253c 100644 --- a/sound/soc/codecs/wm8985.c +++ b/sound/soc/codecs/wm8985.c @@ -997,7 +997,7 @@ static int wm8985_probe(struct snd_soc_codec *codec) 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); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache i/o: %d\n", ret); return ret; diff --git a/sound/soc/codecs/wm8988.c b/sound/soc/codecs/wm8988.c index 0277a76..88fec3c 100644 --- a/sound/soc/codecs/wm8988.c +++ b/sound/soc/codecs/wm8988.c @@ -814,7 +814,7 @@ static int wm8988_probe(struct snd_soc_codec *codec) int ret = 0; codec->control_data = wm8988->regmap; - ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index 33f53ab..8acfa9a 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -1291,7 +1291,7 @@ static int wm8990_probe(struct snd_soc_codec *codec) { int ret; - ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret < 0) { printk(KERN_ERR "wm8990: failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/wm8991.c b/sound/soc/codecs/wm8991.c index 32d2195..d288976 100644 --- a/sound/soc/codecs/wm8991.c +++ b/sound/soc/codecs/wm8991.c @@ -1250,7 +1250,7 @@ static int wm8991_probe(struct snd_soc_codec *codec) { int ret; - ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache i/o: %d\n", ret); return ret; diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index 7b0630a..abc78d2 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -1494,7 +1494,7 @@ static int wm8993_probe(struct snd_soc_codec *codec) wm8993->hubs_data.series_startup = 1; codec->control_data = wm8993->regmap; - ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret != 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 699b527..cef8fdc 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -4001,7 +4001,7 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) 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, SND_SOC_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 33ff361..f50b937 100644 --- a/sound/soc/codecs/wm8995.c +++ b/sound/soc/codecs/wm8995.c @@ -2043,7 +2043,7 @@ static int wm8995_probe(struct snd_soc_codec *codec) wm8995->codec = codec; codec->control_data = wm8995->regmap; - ret = snd_soc_codec_set_cache_io(codec, 16, 16, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache i/o: %d\n", ret); return ret; diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index c8244af..14f27f25 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -2635,7 +2635,7 @@ static int wm8996_probe(struct snd_soc_codec *codec) codec->control_data = wm8996->regmap; - ret = snd_soc_codec_set_cache_io(codec, 16, 16, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret != 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); goto err; diff --git a/sound/soc/codecs/wm8997.c b/sound/soc/codecs/wm8997.c index 4e6442c..4a36d1a 100644 --- a/sound/soc/codecs/wm8997.c +++ b/sound/soc/codecs/wm8997.c @@ -1055,7 +1055,7 @@ static int wm8997_codec_probe(struct snd_soc_codec *codec) 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, SND_SOC_REGMAP); if (ret != 0) return ret; diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index 721cee7..d4b8a0c 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -1264,7 +1264,7 @@ static int wm9081_probe(struct snd_soc_codec *codec) codec->control_data = wm9081->regmap; - ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret != 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/wm9090.c b/sound/soc/codecs/wm9090.c index a07fe16..19285b0 100644 --- a/sound/soc/codecs/wm9090.c +++ b/sound/soc/codecs/wm9090.c @@ -526,7 +526,7 @@ static int wm9090_probe(struct snd_soc_codec *codec) int ret; codec->control_data = wm9090->regmap; - ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret != 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index e5a535b..86c5733 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1152,7 +1152,7 @@ static int soc_probe_codec(struct snd_soc_card *card, /* 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); + snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (driver->controls) snd_soc_add_codec_controls(codec, driver->controls, diff --git a/sound/soc/soc-io.c b/sound/soc/soc-io.c index add99e2..4b4f3d9 100644 --- a/sound/soc/soc-io.c +++ b/sound/soc/soc-io.c @@ -69,8 +69,6 @@ static unsigned int hw_read(struct snd_soc_codec *codec, unsigned int reg) * 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. * * Register formats are frequently shared between many I2C and SPI @@ -85,19 +83,13 @@ 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_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; - switch (control) { case SND_SOC_REGMAP: /* Device has made its own regmap arrangements */ @@ -123,7 +115,6 @@ int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec, 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) { return -ENOTSUPP; -- 1.8.4

3 2

[alsa-devel] [PATCH] ASoC: sta32x: Fix wrong enum for limiter2 release rate
by Takashi Iwai 27 Feb '14

27 Feb '14

There is a typo in the Limiter2 Release Rate control, a wrong enum for Limiter1 is assigned. It must point to Limiter2. Spotted by a compile warning: In file included from sound/soc/codecs/sta32x.c:34:0: sound/soc/codecs/sta32x.c:223:29: warning: ‘sta32x_limiter2_release_rate_enum’ defined but not used [-Wunused-variable] static SOC_ENUM_SINGLE_DECL(sta32x_limiter2_release_rate_enum, ^ include/sound/soc.h:275:18: note: in definition of macro ‘SOC_ENUM_DOUBLE_DECL’ struct soc_enum name = SOC_ENUM_DOUBLE(xreg, xshift_l, xshift_r, \ ^ sound/soc/codecs/sta32x.c:223:8: note: in expansion of macro ‘SOC_ENUM_SINGLE_DECL’ static SOC_ENUM_SINGLE_DECL(sta32x_limiter2_release_rate_enum, ^ Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- Mark, this comes up after merging your pull request. Interestingly, the warning is triggered because const was temporarily reduced along my changes. Since my whole patch series added the const in the macro definition, I didn't see this. Ditto for linux-next, where no relevant warning is seen. I'd like to put this as a quick fix for the next 3.14 pull request. So, if you don't mind, I'm going to commit to my tree for-linus branch. Takashi sound/soc/codecs/sta32x.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c index ea78c172538c..2735361a4c3c 100644 --- a/sound/soc/codecs/sta32x.c +++ b/sound/soc/codecs/sta32x.c @@ -434,7 +434,7 @@ SOC_SINGLE_TLV("Treble Tone Control", STA32X_TONE, STA32X_TONE_TTC_SHIFT, 15, 0, SOC_ENUM("Limiter1 Attack Rate (dB/ms)", sta32x_limiter1_attack_rate_enum), SOC_ENUM("Limiter2 Attack Rate (dB/ms)", sta32x_limiter2_attack_rate_enum), SOC_ENUM("Limiter1 Release Rate (dB/ms)", sta32x_limiter1_release_rate_enum), -SOC_ENUM("Limiter2 Release Rate (dB/ms)", sta32x_limiter1_release_rate_enum), +SOC_ENUM("Limiter2 Release Rate (dB/ms)", sta32x_limiter2_release_rate_enum), /* depending on mode, the attack/release thresholds have * two different enum definitions; provide both -- 1.9.0

2 3

[alsa-devel] Unsupported Valley View 'LPE' adapter?
by Adam Williamson 27 Feb '14

27 Feb '14

Hi, folks. I recently bought a Dell Venue 8 Pro - a Bay Trail / Valley View-based Intel tablet - to try and get Fedora running on it. Among the many things I'm currently fighting :) , the sound doesn't work. snd-hda-intel is not loaded at all, and there are no audio devices present at any level. In Windows' Device Manager I see an 'Intel SST Audio Device' with PCI ID 0x86:0f28 . Indeed, the PCI ID database has this listed as "ValleyView LPE Audio Controller" - http://pci-ids.ucw.cz/read/PC/8086/0f28 . But hda-intel does not have 0f28 as a supported device. The commit for 'Bay Trail' audio support added the ID 0f04: https://lkml.org/lkml/2013/5/16/38 So...what is this 0f28? Is support for it going to be added? Is there any info I can provide to help? (I can boot Linux on the tablet and get info out of it _just barely_, but it's pretty painful - there's no wifi and an install to 'disk' (really a usb stick) doesn't work too well, so I'm stuck working in a live image with no network). -- Adam Williamson Fedora QA Community Monkey IRC: adamw | Twitter: AdamW_Fedora | XMPP: adamw AT happyassassin . net http://www.happyassassin.net

6 13

[alsa-devel] [PATCH v4 0/7] ASoC: add CSR SiRFSoC sound drivers
by RongJun Ying 27 Feb '14

27 Feb '14

From: Rongjun Ying <rongjun.ying(a)csr.com> This patchset adds CSR SiRFSoC sound drivers including: 1. The SiRF internal audio codec driver 2. The SiRF audio port driver 3. The SiRF internal audio card driver 4. The SiRF I2S controller driver 5. The SiRF hdmi card driver 6. The SiRF USP controller(DSP mode) driver 7. The SiRF bt-sco card driver -v4: 1. Use the generic dmaengine pcm codes instead of sirf pcm driver. 2. Add TLV information into internal audio codec driver. 3. Use the MMIO regmap instead of custom read and write callbacks. 4. Remove spin_lock. 5. Change some variable names. 6. The SiRF internal audio driver split into codec, port. 7. Add binding documents. Rongjun Ying (7): ASoC: sirf: Add SiRF internal audio codec driver ASoC: sirf: Add SiRF audio port driver is used by SiRF internal audio codec ASoC: sirf: Add SiRF audio card ASoC: sirf: Add SiRF I2S driver ASoC: sirf: Add hdmi card ASoC: sirf: Add usp driver which is used by dsp mode ASoC: sirf: Add bt-sco card .../devicetree/bindings/sound/sirf-audio-codec.txt | 17 + .../devicetree/bindings/sound/sirf-audio-port.txt | 20 + .../devicetree/bindings/sound/sirf-audio.txt | 41 ++ .../devicetree/bindings/sound/sirf-bt-sco.txt | 18 + .../devicetree/bindings/sound/sirf-hdmi.txt | 15 + .../devicetree/bindings/sound/sirf-i2s.txt | 27 + .../devicetree/bindings/sound/sirf-usp.txt | 27 + sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/sirf-audio-codec.c | 585 ++++++++++++++++++++ sound/soc/codecs/sirf-audio-codec.h | 75 +++ sound/soc/sirf/Kconfig | 34 ++ sound/soc/sirf/Makefile | 13 + sound/soc/sirf/sirf-audio-port.c | 194 +++++++ sound/soc/sirf/sirf-audio-port.h | 62 ++ sound/soc/sirf/sirf-audio.c | 156 ++++++ sound/soc/sirf/sirf-bt-sco.c | 99 ++++ sound/soc/sirf/sirf-hdmi.c | 98 ++++ sound/soc/sirf/sirf-i2s.c | 438 +++++++++++++++ sound/soc/sirf/sirf-i2s.h | 88 +++ sound/soc/sirf/sirf-usp.c | 448 +++++++++++++++ sound/soc/sirf/sirf-usp.h | 282 ++++++++++ 24 files changed, 2745 insertions(+), 0 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/sirf-audio-codec.txt create mode 100644 Documentation/devicetree/bindings/sound/sirf-audio-port.txt create mode 100644 Documentation/devicetree/bindings/sound/sirf-audio.txt create mode 100644 Documentation/devicetree/bindings/sound/sirf-bt-sco.txt create mode 100644 Documentation/devicetree/bindings/sound/sirf-hdmi.txt create mode 100644 Documentation/devicetree/bindings/sound/sirf-i2s.txt create mode 100644 Documentation/devicetree/bindings/sound/sirf-usp.txt create mode 100644 sound/soc/codecs/sirf-audio-codec.c create mode 100644 sound/soc/codecs/sirf-audio-codec.h create mode 100644 sound/soc/sirf/Kconfig create mode 100644 sound/soc/sirf/Makefile create mode 100644 sound/soc/sirf/sirf-audio-port.c create mode 100644 sound/soc/sirf/sirf-audio-port.h create mode 100644 sound/soc/sirf/sirf-audio.c create mode 100644 sound/soc/sirf/sirf-bt-sco.c create mode 100644 sound/soc/sirf/sirf-hdmi.c create mode 100644 sound/soc/sirf/sirf-i2s.c create mode 100644 sound/soc/sirf/sirf-i2s.h create mode 100644 sound/soc/sirf/sirf-usp.c create mode 100644 sound/soc/sirf/sirf-usp.h -- 1.7.5.4

3 10

[alsa-devel] [PATCH] ASoC: fsl-sai: Add SND_SOC_DAIFMT_DSP_A/B support.
by Xiubo Li 27 Feb '14

27 Feb '14

o Add SND_SOC_DAIFMT_DSP_A support. o Add SND_SOC_DAIFMT_DSP_B support. Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- sound/soc/fsl/fsl_sai.c | 45 ++++++++++++++++++++++++++++++++++++++++++--- sound/soc/fsl/fsl_sai.h | 1 + 2 files changed, 43 insertions(+), 3 deletions(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 26d9f5e..c4a4231 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -108,15 +108,44 @@ static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai, /* DAI mode */ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: - /* Data on rising edge of bclk, frame low, 1clk before data */ + /* + * 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: - /* Data on rising edge of bclk, frame high, 0clk before data */ + /* + * 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: @@ -219,7 +248,9 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, 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); @@ -245,6 +276,10 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd, struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); u32 tcsr, rcsr; + /* + * 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, @@ -261,6 +296,10 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd, tcsr &= ~FSL_SAI_CSR_FRDE; } + /* + * 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: diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h index 1571459..e432260 100644 --- a/sound/soc/fsl/fsl_sai.h +++ b/sound/soc/fsl/fsl_sai.h @@ -103,6 +103,7 @@ struct fsl_sai { 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; -- 1.8.4

2 1

[alsa-devel] [PATCH] Asoc: codecs: Remove tlv320aic32x4 from compatibles of tlv320aic3x
by Markus Pargmann 27 Feb '14

27 Feb '14

This reverts tlv320aic32x4 as compatible for tlv320aic3x as it has its own bindings now. Signed-off-by: Markus Pargmann <mpa(a)pengutronix.de> --- Documentation/devicetree/bindings/sound/tlv320aic3x.txt | 1 - 1 file changed, 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/tlv320aic3x.txt b/Documentation/devicetree/bindings/sound/tlv320aic3x.txt index 9d8ea14..5e6040c 100644 --- a/Documentation/devicetree/bindings/sound/tlv320aic3x.txt +++ b/Documentation/devicetree/bindings/sound/tlv320aic3x.txt @@ -6,7 +6,6 @@ Required properties: - compatible - "string" - One of: "ti,tlv320aic3x" - Generic TLV320AIC3x device - "ti,tlv320aic32x4" - TLV320AIC32x4 "ti,tlv320aic33" - TLV320AIC33 "ti,tlv320aic3007" - TLV320AIC3007 "ti,tlv320aic3106" - TLV320AIC3106 -- 1.8.5.3

2 1

[alsa-devel] [GIT PULL] ASoC fixes for v3.14
by Mark Brown 27 Feb '14

27 Feb '14

The following changes since commit 895be5b31e5175bef575008aadb4f0a27b850daa: Merge remote-tracking branch 'asoc/fix/wm8993' into asoc-linus (2014-02-19 13:13:58 +0900) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.14-rc4 for you to fetch changes up to 57374bb491facf716c172126b7e9b99c9e3f8b45: Merge remote-tracking branches 'asoc/fix/sta32x', 'asoc/fix/wm8400', 'asoc/fix/wm8770', 'asoc/fix/wm8900' and 'asoc/fix/wm8994' into asoc-linus (2014-02-23 12:20:34 +0900) Getting a bit large but they're all actual bug fixes unfortunately. ---------------------------------------------------------------- ASoC: Fixes for v3.14 A somewhat large set of fixes here due to the identification of some systematic problems with hard to use APIs in the subsystem. Takashi did a lot of work to address the enumeration API which uncovered a number of off by one bugs caused by confusing APIs while Charles addressed issues in the locking around DAPM. ---------------------------------------------------------------- Charles Keepax (3): ASoC: dapm: Correct regulator bypass error messages Input - arizona-haptics: Fix double lock of dapm_mutex ASoC: dapm: Add locking to snd_soc_dapm_xxxx_pin functions Lars-Peter Clausen (1): ASoC: sta32x: Fix cache sync Mark Brown (4): Merge tag 'asoc-v3.14-rc3' into asoc-linus Merge remote-tracking branch 'asoc/fix/dapm' into asoc-linus Merge remote-tracking branches 'asoc/fix/ad1980' and 'asoc/fix/isabelle' into asoc-linus Merge remote-tracking branches 'asoc/fix/sta32x', 'asoc/fix/wm8400', 'asoc/fix/wm8770', 'asoc/fix/wm8900' and 'asoc/fix/wm8994' into asoc-linus Takashi Iwai (7): ASoC: sta32x: Fix array access overflow ASoC: wm8770: Fix wrong number of enum items ASoC: wm8900: Fix the wrong number of enum items ASoC: wm8994: Fix the wrong number of enum items ASoC: ad1980: Fix wrong number of items for capture source ASoC: isabelle: Fix the wrong number of items in enum ctls ASoC: wm8400: Fix the wrong number of enum items drivers/extcon/extcon-arizona.c | 12 --- drivers/input/misc/arizona-haptics.c | 19 ----- include/sound/soc-dapm.h | 8 ++ sound/soc/codecs/ad1980.c | 4 +- sound/soc/codecs/isabelle.c | 52 +++++++------ sound/soc/codecs/sta32x.c | 74 +++++++++---------- sound/soc/codecs/wm8400.c | 34 +++++---- sound/soc/codecs/wm8770.c | 4 +- sound/soc/codecs/wm8900.c | 44 +++++------ sound/soc/codecs/wm8994.c | 135 ++++++++++++++++++---------------- sound/soc/soc-dapm.c | 139 +++++++++++++++++++++++++++++++---- 11 files changed, 317 insertions(+), 208 deletions(-)

1 0

[alsa-devel] tlv320aic3x regcache issue
by Jack Mitchell 26 Feb '14

26 Feb '14

Hi, I have come across an issue where the regcache mechanism used in the tlv320aic3x driver is causing problems. When switching from 44.1khz to 48khz some of the registers are not written due to the default register cache holding the same value as being written. However, the codec requires these registers to be written in order to trigger the change, this has been proven by changing the register default to something different, therefore forcing a write at the correct time and it triggers the change as expected. What would be the best way to force these registers to be written? Is there some sort of force write flag, or a way to turn off the cache just during this portion of the configuration? Any direction in the proper way of doing this would be much appreciated. Cheers, Jack -- Jack Mitchell (jack(a)embed.me.uk) Embedded Systems Engineer Cambridgeshire, UK http://www.embed.me.uk --

1 0

[alsa-devel] tlv320aic3x regcache issue
by Jack Mitchell 26 Feb '14

26 Feb '14

Hi, I have come across an issue where the regcache mechanism used in the tlv320aic3x driver is causing problems. When switching from 44.1khz to 48khz some of the registers are not written due to the default register cache holding the same value as being written. However, the codec requires these registers to be written in order to trigger the change, this has been proven by changing the register default to something different, therefore forcing a write at the correct time and it triggers the change as expected. What would be the best way to force these registers to be written? Is there some sort of force write flag, or a way to turn off the cache just during this portion of the configuration? Any direction in the proper way of doing this would be much appreciated. Cheers, Jack -- Jack Mitchell (jack(a)embed.me.uk) Embedded Systems Engineer Cambridgeshire, UK http://www.embed.me.uk --

1 0

[alsa-devel] [PATCH 00/48] Rewrite kernel messages with standard printk helpers
by Takashi Iwai 26 Feb '14

26 Feb '14

Hi, this is another long series of patches for trivial cleanups. This time, they replace the printk and snd_print*() calls in ALSA PCI and USB drivers with dev_err() and co. This will give nicer and more consistent outputs. The only drawback by these replacements is that alsa-info.sh won't be able to catch the dmesg lines because it doesn't contain any longer the word "ALSA", but only show the module name like "snd_intel8x0". I'm going to fix alsa-info.sh on git soon later, but if you guys think "oh no it's disaster!", let me know. We need to reconsider a better way to put the known markers (like explicitly putting "ALSA" at the beginning.) Takashi

1 49

[alsa-devel] [PATCH 2/2] ASoC: cirrus: clps711x: Add bindings documentation for the CLPS711X DAI
by Alexander Shiyan 26 Feb '14

26 Feb '14

This patch adds the devicetree documentation for the Cirrus Logic CLPS711X Digital Audio Interface (DAI). Signed-off-by: Alexander Shiyan <shc_work(a)mail.ru> --- .../bindings/sound/cirrus,clps711x-dai.txt | 20 ++++++++++++++++++++ 1 file changed, 20 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/cirrus,clps711x-dai.txt diff --git a/Documentation/devicetree/bindings/sound/cirrus,clps711x-dai.txt b/Documentation/devicetree/bindings/sound/cirrus,clps711x-dai.txt new file mode 100644 index 0000000..718599e --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cirrus,clps711x-dai.txt @@ -0,0 +1,20 @@ +* Cirrus Logic CLPS711X Digital Audio Interface (DAI) + +Required properties: +- compatible: Should be "cirrus,clps711x-dai". +- reg: Should contain registers location and length. +- clocks: A clock specifier for the CPU PLL clock. +- clock-names: The CPU PLL clock should be named "pll". +- interrupts: Should contain DAI interrupt number. +- syscon: phandle to syscon device node which control DAI (SYSCON3). + +Example: + dai: dai@80002000 { + #sound-dai-cells = <0>; + compatible = "cirrus,clps711x-dai"; + reg = <0x80002000 0x604>; + clocks = <&clks 3>; + clock-names = "pll"; + interrupts = <32>; + syscon = <&syscon3>; + }; -- 1.8.3.2

1 0

[alsa-devel] [PATCH] ALSA: rme96: Convert to the new pm_ops
by Takashi Iwai 26 Feb '14

26 Feb '14

This driver slipped from the last rewrite. Just convert to the new standard pm ops. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/rme96.c | 34 ++++++++++++++++++---------------- 1 file changed, 18 insertions(+), 16 deletions(-) diff --git a/sound/pci/rme96.c b/sound/pci/rme96.c index 00d46116e04a..f5014a8e69ad 100644 --- a/sound/pci/rme96.c +++ b/sound/pci/rme96.c @@ -240,7 +240,7 @@ struct rme96 { u8 rev; /* card revision number */ -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP u32 playback_pointer; u32 capture_pointer; void *playback_suspend_buffer; @@ -1570,7 +1570,7 @@ snd_rme96_free(void *private_data) pci_release_regions(rme96->pci); rme96->port = 0; } -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP vfree(rme96->playback_suspend_buffer); vfree(rme96->capture_suspend_buffer); #endif @@ -2372,13 +2372,12 @@ snd_rme96_create_switches(struct snd_card *card, * Card initialisation */ -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP -static int -snd_rme96_suspend(struct pci_dev *pci, - pm_message_t state) +static int rme96_suspend(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct rme96 *rme96 = card->private_data; snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); @@ -2407,10 +2406,10 @@ snd_rme96_suspend(struct pci_dev *pci, return 0; } -static int -snd_rme96_resume(struct pci_dev *pci) +static int rme96_resume(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct rme96 *rme96 = card->private_data; pci_restore_state(pci); @@ -2451,7 +2450,11 @@ snd_rme96_resume(struct pci_dev *pci) return 0; } -#endif +static SIMPLE_DEV_PM_OPS(rme96_pm, rme96_suspend, rme96_resume); +#define RME96_PM_OPS &rme96_pm +#else +#define RME96_PM_OPS NULL +#endif /* CONFIG_PM_SLEEP */ static void snd_rme96_card_free(struct snd_card *card) { @@ -2488,7 +2491,7 @@ snd_rme96_probe(struct pci_dev *pci, return err; } -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP rme96->playback_suspend_buffer = vmalloc(RME96_BUFFER_SIZE); if (!rme96->playback_suspend_buffer) { snd_printk(KERN_ERR @@ -2547,10 +2550,9 @@ static struct pci_driver rme96_driver = { .id_table = snd_rme96_ids, .probe = snd_rme96_probe, .remove = snd_rme96_remove, -#ifdef CONFIG_PM - .suspend = snd_rme96_suspend, - .resume = snd_rme96_resume, -#endif + .driver = { + .pm = RME96_PM_OPS, + }, }; module_pci_driver(rme96_driver); -- 1.9.0

1 0

[alsa-devel] [alsa-plugins] [PATCH 1/2] Allow to specify a client name as parameters
by Corfu, ValentinX 26 Feb '14

26 Feb '14

Hi Takashi, I reworked the patches and hope to be the final ones. J You will find attached at each of the emails. Kind Regards, Valentin Corfu Intel GmbH Dornacher Strasse 1 85622 Feldkirchen/Muenchen, Deutschland Sitz der Gesellschaft: Feldkirchen bei Muenchen Geschaeftsfuehrer: Christian Lamprechter, Hannes Schwaderer, Douglas Lusk Registergericht: Muenchen HRB 47456 Ust.-IdNr./VAT Registration No.: DE129385895 Citibank Frankfurt a.M. (BLZ 502 109 00) 600119052

2 1

[alsa-devel] [PATCH - speaker-test 1/1] speaker-test: add --force-frequency option to allow hz outside range
by erik＠caspar.alsa-project.org 26 Feb '14

26 Feb '14

From: Erik Ackermann <domohawk(a)gmail.com> Signed-off-by: Erik Ackermann <domohawk(a)gmail.com> diff --git a/speaker-test/speaker-test.c b/speaker-test/speaker-test.c index 261f399..22927f2 100644 --- a/speaker-test/speaker-test.c +++ b/speaker-test/speaker-test.c @@ -106,6 +106,7 @@ static snd_pcm_uframes_t period_size; static const char *given_test_wav_file = NULL; static char *wav_file_dir = SOUNDSDIR; static int debug = 0; +static int force_frequency = 0; static int in_aborting = 0; static snd_pcm_t *pcm_handle = NULL; @@ -1015,6 +1016,7 @@ static void help(void) "-w,--wavfile Use the given WAV file as a test sound\n" "-W,--wavdir Specify the directory containing WAV files\n" "-m,--chmap Specify the channel map to override\n" + "-X,--force-frequency force frequencies outside the 30-5000hz range\n" "\n")); printf(_("Recognized sample formats are:")); for (fmt = supported_formats; *fmt >= 0; fmt++) { @@ -1057,6 +1059,7 @@ int main(int argc, char *argv[]) { {"wavfile", 1, NULL, 'w'}, {"wavdir", 1, NULL, 'W'}, {"debug", 0, NULL, 'd'}, + {"force-frequncy", 1, NULL, 'X'}, #ifdef CONFIG_SUPPORT_CHMAP {"chmap", 1, NULL, 'm'}, #endif @@ -1078,7 +1081,7 @@ int main(int argc, char *argv[]) { while (1) { int c; - if ((c = getopt_long(argc, argv, "hD:r:c:f:F:b:p:P:t:l:s:w:W:d" + if ((c = getopt_long(argc, argv, "hD:r:c:f:F:b:p:P:t:l:s:w:W:d:X" #ifdef CONFIG_SUPPORT_CHMAP "m:" #endif @@ -1114,8 +1117,6 @@ int main(int argc, char *argv[]) { break; case 'f': freq = atof(optarg); - freq = freq < 30.0 ? 30.0 : freq; - freq = freq > 5000.0 ? 5000.0 : freq; break; case 'b': buffer_time = atoi(optarg); @@ -1173,6 +1174,9 @@ int main(int argc, char *argv[]) { case 'd': debug = 1; break; + case 'X': + force_frequency = 1; + break; #ifdef CONFIG_SUPPORT_CHMAP case 'm': chmap = optarg; @@ -1190,6 +1194,11 @@ int main(int argc, char *argv[]) { exit(EXIT_SUCCESS); } + if (!force_frequency) { + freq = freq < 30.0 ? 30.0 : freq; + freq = freq > 5000.0 ? 5000.0 : freq; + } + if (test_type == TEST_WAV) format = SND_PCM_FORMAT_S16_LE; /* fixed format */ -- 1.8.3.2

4 7

[alsa-devel] MAYA44 USB audio device no inputs control (again)
by Leo Combes 26 Feb '14

26 Feb '14

Hi, I have bought a Maya 44 USB, with 4 output and 4 input channels. http://www.esi-audio.com/products/maya44usb/ The soundcard works fine in playback, but in recording has some problems, specificly there is no way to control the volume of the recording on inputs. It seems to be default setted to 50%. Alsamixer, in playback section shows 3 controls, "speaker", "line" and "line1". But, if I choose capture, it shows "This sound device does not have any capture controls". It seems like there are controls, and they work fine, running on Windows. I searched the problem on Google and I found a lot of users with the same problem. The most interesting thread that I found, is one on this mailist, sadly after some messages, user didn't catch up with the conversation: http://mailman.alsa-project.org/pipermail/alsa-devel/2011-November/046242.h… I have a hardware mixer connected to 2 inputs of the Maya. If I try to recording with Audacity or any other software, the volume of recording is never greater than -6dB. If I try increase the volume in the mixer to get louder recording, the only thing I get is a completely distorted sound not louder than -6dB, what tells me that Maya inputs have a presetted volume at 50%. Besides, if I increase the volume in the mixer, Audicity's vumeter confirm that it never go further than -6dB. Following the thread that the user left unfinished, here goes the information that I get with my system, to see if could find a solution. leo-ThinkPad leo # aplay -L null Discard all samples (playback) or generate zero samples (capture) pulse PulseAudio Sound Server sysdefault:CARD=MID HDA Intel MID, CX20585 Analog Default Audio Device front:CARD=MID,DEV=0 HDA Intel MID, CX20585 Analog Front speakers surround40:CARD=MID,DEV=0 HDA Intel MID, CX20585 Analog 4.0 Surround output to Front and Rear speakers surround41:CARD=MID,DEV=0 HDA Intel MID, CX20585 Analog 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=MID,DEV=0 HDA Intel MID, CX20585 Analog 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=MID,DEV=0 HDA Intel MID, CX20585 Analog 5.1 Surround output to Front, Center, Rear and Subwoofer speakers surround71:CARD=MID,DEV=0 HDA Intel MID, CX20585 Analog 7.1 Surround output to Front, Center, Side, Rear and Woofer speakers dmix:CARD=MID,DEV=0 HDA Intel MID, CX20585 Analog Direct sample mixing device dsnoop:CARD=MID,DEV=0 HDA Intel MID, CX20585 Analog Direct sample snooping device hw:CARD=MID,DEV=0 HDA Intel MID, CX20585 Analog Direct hardware device without any conversions plughw:CARD=MID,DEV=0 HDA Intel MID, CX20585 Analog Hardware device with all software conversions hdmi:CARD=NVidia,DEV=0 HDA NVidia, HDMI 0 HDMI Audio Output hdmi:CARD=NVidia,DEV=1 HDA NVidia, HDMI 0 HDMI Audio Output hdmi:CARD=NVidia,DEV=2 HDA NVidia, HDMI 0 HDMI Audio Output hdmi:CARD=NVidia,DEV=3 HDA NVidia, HDMI 0 HDMI Audio Output dmix:CARD=NVidia,DEV=3 HDA NVidia, HDMI 0 Direct sample mixing device dmix:CARD=NVidia,DEV=7 HDA NVidia, HDMI 0 Direct sample mixing device dmix:CARD=NVidia,DEV=8 HDA NVidia, HDMI 0 Direct sample mixing device dmix:CARD=NVidia,DEV=9 HDA NVidia, HDMI 0 Direct sample mixing device dsnoop:CARD=NVidia,DEV=3 HDA NVidia, HDMI 0 Direct sample snooping device dsnoop:CARD=NVidia,DEV=7 HDA NVidia, HDMI 0 Direct sample snooping device dsnoop:CARD=NVidia,DEV=8 HDA NVidia, HDMI 0 Direct sample snooping device dsnoop:CARD=NVidia,DEV=9 HDA NVidia, HDMI 0 Direct sample snooping device hw:CARD=NVidia,DEV=3 HDA NVidia, HDMI 0 Direct hardware device without any conversions hw:CARD=NVidia,DEV=7 HDA NVidia, HDMI 0 Direct hardware device without any conversions hw:CARD=NVidia,DEV=8 HDA NVidia, HDMI 0 Direct hardware device without any conversions hw:CARD=NVidia,DEV=9 HDA NVidia, HDMI 0 Direct hardware device without any conversions plughw:CARD=NVidia,DEV=3 HDA NVidia, HDMI 0 Hardware device with all software conversions plughw:CARD=NVidia,DEV=7 HDA NVidia, HDMI 0 Hardware device with all software conversions plughw:CARD=NVidia,DEV=8 HDA NVidia, HDMI 0 Hardware device with all software conversions plughw:CARD=NVidia,DEV=9 HDA NVidia, HDMI 0 Hardware device with all software conversions sysdefault:CARD=USB MAYA44 USB+, USB Audio Default Audio Device front:CARD=USB,DEV=0 MAYA44 USB+, USB Audio Front speakers surround40:CARD=USB,DEV=0 MAYA44 USB+, USB Audio 4.0 Surround output to Front and Rear speakers surround41:CARD=USB,DEV=0 MAYA44 USB+, USB Audio 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=USB,DEV=0 MAYA44 USB+, USB Audio 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=USB,DEV=0 MAYA44 USB+, USB Audio 5.1 Surround output to Front, Center, Rear and Subwoofer speakers surround71:CARD=USB,DEV=0 MAYA44 USB+, USB Audio 7.1 Surround output to Front, Center, Side, Rear and Woofer speakers iec958:CARD=USB,DEV=0 MAYA44 USB+, USB Audio IEC958 (S/PDIF) Digital Audio Output dmix:CARD=USB,DEV=0 MAYA44 USB+, USB Audio Direct sample mixing device dsnoop:CARD=USB,DEV=0 MAYA44 USB+, USB Audio Direct sample snooping device hw:CARD=USB,DEV=0 MAYA44 USB+, USB Audio Direct hardware device without any conversions plughw:CARD=USB,DEV=0 MAYA44 USB+, USB Audio Hardware device with all software conversions leo-ThinkPad leo # lsmod | grep snd snd_hda_codec_hdmi 41684 4 snd_usb_audio 155972 3 snd_hda_codec_conexant 57181 1 snd_usbmidi_lib 29576 1 snd_usb_audio snd_hda_intel 57183 2 snd_hda_codec 194881 3 snd_hda_codec_hdmi,snd_hda_codec_conexant,snd_hda_intel snd_hwdep 13613 2 snd_usb_audio,snd_hda_codec snd_pcm 107140 6 snd_usb_audio,snd_hda_codec_hdmi,snd_hda_codec,snd_hda_intel snd_page_alloc 18798 2 snd_pcm,snd_hda_intel snd_seq_midi 13324 0 snd_seq_midi_event 14899 1 snd_seq_midi snd_rawmidi 30465 2 snd_usbmidi_lib,snd_seq_midi snd_seq 66061 2 snd_seq_midi_event,snd_seq_midi snd_seq_device 14497 3 snd_seq,snd_rawmidi,snd_seq_midi snd_timer 30038 2 snd_pcm,snd_seq snd 73802 22 snd_usb_audio,snd_hwdep,snd_timer,snd_hda_codec_hdmi,snd_hda_codec_conexant,snd_pcm,snd_seq,snd_rawmidi,snd_usbmidi_lib,snd_hda_codec,snd_hda_intel,thinkpad_acpi,snd_seq_device,snd_seq_midi soundcore 12680 1 snd leo-ThinkPad leo # amixer contents numid=2,iface=MIXER,name='Line Playback Switch' ; type=BOOLEAN,access=rw------,values=1 : values=off numid=4,iface=MIXER,name='Line Playback Switch',index=1 ; type=BOOLEAN,access=rw------,values=1 : values=off numid=3,iface=MIXER,name='Line Playback Volume' ; type=INTEGER,access=rw---R--,values=2,min=0,max=255,step=0 : values=0,0 | dBminmax-min=0.00dB,max=0.99dB numid=5,iface=MIXER,name='Line Playback Volume',index=1 ; type=INTEGER,access=rw---R--,values=2,min=0,max=255,step=0 : values=0,0 | dBminmax-min=0.00dB,max=0.99dB numid=6,iface=MIXER,name='Speaker Playback Switch' ; type=BOOLEAN,access=rw------,values=1 : values=on numid=7,iface=MIXER,name='Speaker Playback Volume' ; type=INTEGER,access=rw---R--,values=2,min=0,max=255,step=0 : values=255,255 | dBminmax-min=0.00dB,max=0.99dB numid=1,iface=PCM,name='Playback Channel Map' ; type=INTEGER,access=r----R--,values=4,min=0,max=36,step=0 : values=3,4,0,0 | | TLV size error (257, 8, 0)! leo-ThinkPad leo # amixer -c2 Simple mixer control 'Speaker',0 Capabilities: pvolume pswitch pswitch-joined penum Playback channels: Front Left - Front Right Limits: Playback 0 - 255 Mono: Front Left: Playback 255 [100%] [0.99dB] [on] Front Right: Playback 255 [100%] [0.99dB] [on] Simple mixer control 'Line',0 Capabilities: pvolume pswitch pswitch-joined penum Playback channels: Front Left - Front Right Limits: Playback 0 - 255 Mono: Front Left: Playback 0 [0%] [0.00dB] [off] Front Right: Playback 0 [0%] [0.00dB] [off] Simple mixer control 'Line',1 Capabilities: pvolume pswitch pswitch-joined penum Playback channels: Front Left - Front Right Limits: Playback 0 - 255 Mono: Front Left: Playback 0 [0%] [0.00dB] [off] Front Right: Playback 0 [0%] [0.00dB] [off] Thanks in advance!

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