November 2016 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] Applied "ASoC: stac9766: Remove register paging support" to the asoc tree
by Mark Brown 04 Nov '16

04 Nov '16

The patch ASoC: stac9766: Remove register paging support has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 2bea8f97d4c3aded4c71d72e8702aa7dbe9894cf Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen <lars(a)metafoo.de> Date: Fri, 4 Nov 2016 18:26:54 +0100 Subject: [PATCH] ASoC: stac9766: Remove register paging support The AC'97 standard defines paging support for the register range 0x60-0x6f. Meaning registers in this window are mapped to different physical registers depending on the setting of the page select register (0x24). The stac9766 implements support for switching between page 0 and page 1 depending on the addressed register. But the driver never accesses any registers from page 1, in addition page 0 is the page selected by default. Considering the development history it is unlikely that the driver will see any new features that require paging support. Removing the paging support makes transitioning the driver to regmap a bit more straight forward. The default register value table is update to contain the values from page 0, rather than page 1. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/stac9766.c | 16 ++-------------- sound/soc/codecs/stac9766.h | 5 ++--- 2 files changed, 4 insertions(+), 17 deletions(-) diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c index 27f30d352867..e54e4a4ce296 100644 --- a/sound/soc/codecs/stac9766.c +++ b/sound/soc/codecs/stac9766.c @@ -47,8 +47,8 @@ static const u16 stac9766_reg[] = { 0x0000, 0x0000, 0x0003, 0xffff, /* 4e */ 0x0000, 0x0000, 0x0000, 0x0000, /* 56 */ 0x4000, 0x0000, 0x0000, 0x0000, /* 5e */ - 0x1201, 0xFFFF, 0xFFFF, 0x0000, /* 66 */ - 0x0000, 0x0000, 0x0000, 0x0000, /* 6e */ + 0x1201, 0x0000, 0x0000, 0x0000, /* 66 */ + 0x0000, 0x0000, 0x0000, 0x1000, /* 6e */ 0x0000, 0x0000, 0x0000, 0x0006, /* 76 */ 0x0000, 0x0000, 0x0000, 0x0000, /* 7e */ }; @@ -145,12 +145,6 @@ static int stac9766_ac97_write(struct snd_soc_codec *codec, unsigned int reg, struct snd_ac97 *ac97 = snd_soc_codec_get_drvdata(codec); u16 *cache = codec->reg_cache; - if (reg > AC97_STAC_PAGE0) { - stac9766_ac97_write(codec, AC97_INT_PAGING, 0); - soc_ac97_ops->write(ac97, reg, val); - stac9766_ac97_write(codec, AC97_INT_PAGING, 1); - return 0; - } if (reg / 2 >= ARRAY_SIZE(stac9766_reg)) return -EIO; @@ -165,12 +159,6 @@ static unsigned int stac9766_ac97_read(struct snd_soc_codec *codec, struct snd_ac97 *ac97 = snd_soc_codec_get_drvdata(codec); u16 val = 0, *cache = codec->reg_cache; - if (reg > AC97_STAC_PAGE0) { - stac9766_ac97_write(codec, AC97_INT_PAGING, 0); - val = soc_ac97_ops->read(ac97, reg - AC97_STAC_PAGE0); - stac9766_ac97_write(codec, AC97_INT_PAGING, 1); - return val; - } if (reg / 2 >= ARRAY_SIZE(stac9766_reg)) return -EIO; diff --git a/sound/soc/codecs/stac9766.h b/sound/soc/codecs/stac9766.h index cb0d5505d571..e35cee82f416 100644 --- a/sound/soc/codecs/stac9766.h +++ b/sound/soc/codecs/stac9766.h @@ -5,9 +5,8 @@ #ifndef _STAC9766_H #define _STAC9766_H -#define AC97_STAC_PAGE0 0x1000 -#define AC97_STAC_DA_CONTROL (AC97_STAC_PAGE0 | 0x6A) -#define AC97_STAC_ANALOG_SPECIAL (AC97_STAC_PAGE0 | 0x6E) +#define AC97_STAC_DA_CONTROL 0x6A +#define AC97_STAC_ANALOG_SPECIAL 0x6E #define AC97_STAC_STEREO_MIC 0x78 #endif -- 2.10.1

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[alsa-devel] Applied "ASoC: stac9766: Move register defines to main source file" to the asoc tree
by Mark Brown 04 Nov '16

04 Nov '16

The patch ASoC: stac9766: Move register defines to main source file has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From dccb395c268e9f96dfaf3f3bb9933aa9a0ded8cc Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen <lars(a)metafoo.de> Date: Fri, 4 Nov 2016 18:26:55 +0100 Subject: [PATCH] ASoC: stac9766: Move register defines to main source file The stac9766 driver has a header file that defines 3 register locations. Move these to the main source file since it is not really worth it having a separate file for them. The header file is now empty and can be removed. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/stac9766.c | 6 ++++-- sound/soc/codecs/stac9766.h | 12 ------------ sound/soc/fsl/efika-audio-fabric.c | 1 - 3 files changed, 4 insertions(+), 15 deletions(-) delete mode 100644 sound/soc/codecs/stac9766.h diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c index e54e4a4ce296..f675d343b529 100644 --- a/sound/soc/codecs/stac9766.c +++ b/sound/soc/codecs/stac9766.c @@ -26,11 +26,13 @@ #include <sound/soc.h> #include <sound/tlv.h> -#include "stac9766.h" - #define STAC9766_VENDOR_ID 0x83847666 #define STAC9766_VENDOR_ID_MASK 0xffffffff +#define AC97_STAC_DA_CONTROL 0x6A +#define AC97_STAC_ANALOG_SPECIAL 0x6E +#define AC97_STAC_STEREO_MIC 0x78 + /* * STAC9766 register cache */ diff --git a/sound/soc/codecs/stac9766.h b/sound/soc/codecs/stac9766.h deleted file mode 100644 index e35cee82f416..000000000000 --- a/sound/soc/codecs/stac9766.h +++ /dev/null @@ -1,12 +0,0 @@ -/* - * stac9766.h -- STAC9766 Soc Audio driver - */ - -#ifndef _STAC9766_H -#define _STAC9766_H - -#define AC97_STAC_DA_CONTROL 0x6A -#define AC97_STAC_ANALOG_SPECIAL 0x6E -#define AC97_STAC_STEREO_MIC 0x78 - -#endif diff --git a/sound/soc/fsl/efika-audio-fabric.c b/sound/soc/fsl/efika-audio-fabric.c index b2acd3293ea8..f200d1cfc4bd 100644 --- a/sound/soc/fsl/efika-audio-fabric.c +++ b/sound/soc/fsl/efika-audio-fabric.c @@ -27,7 +27,6 @@ #include "mpc5200_dma.h" #include "mpc5200_psc_ac97.h" -#include "../codecs/stac9766.h" #define DRV_NAME "efika-audio-fabric" -- 2.10.1

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[alsa-devel] Applied "ASoC: stac9766: Convert to regmap" to the asoc tree
by Mark Brown 04 Nov '16

04 Nov '16

The patch ASoC: stac9766: Convert to regmap has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 6bbf787bb70c8a16509a2d51182423a6f6977742 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen <lars(a)metafoo.de> Date: Fri, 4 Nov 2016 18:26:56 +0100 Subject: [PATCH] ASoC: stac9766: Convert to regmap Currently the stac9766 driver still uses custom snd_soc_codec_driver IO callbacks. This has been deprecated for a while, so convert the stac9766 driver to use regmap for its IO. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/stac9766.c | 108 ++++++++++++++++++++++++-------------------- 1 file changed, 59 insertions(+), 49 deletions(-) diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c index f675d343b529..62cbeedf93b9 100644 --- a/sound/soc/codecs/stac9766.c +++ b/sound/soc/codecs/stac9766.c @@ -18,6 +18,7 @@ #include <linux/slab.h> #include <linux/module.h> #include <linux/device.h> +#include <linux/regmap.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/ac97_codec.h> @@ -33,26 +34,49 @@ #define AC97_STAC_ANALOG_SPECIAL 0x6E #define AC97_STAC_STEREO_MIC 0x78 -/* - * STAC9766 register cache - */ -static const u16 stac9766_reg[] = { - 0x6A90, 0x8000, 0x8000, 0x8000, /* 6 */ - 0x0000, 0x0000, 0x8008, 0x8008, /* e */ - 0x8808, 0x8808, 0x8808, 0x8808, /* 16 */ - 0x8808, 0x0000, 0x8000, 0x0000, /* 1e */ - 0x0000, 0x0000, 0x0000, 0x000f, /* 26 */ - 0x0a05, 0x0400, 0xbb80, 0x0000, /* 2e */ - 0x0000, 0xbb80, 0x0000, 0x0000, /* 36 */ - 0x0000, 0x2000, 0x0000, 0x0100, /* 3e */ - 0x0000, 0x0000, 0x0080, 0x0000, /* 46 */ - 0x0000, 0x0000, 0x0003, 0xffff, /* 4e */ - 0x0000, 0x0000, 0x0000, 0x0000, /* 56 */ - 0x4000, 0x0000, 0x0000, 0x0000, /* 5e */ - 0x1201, 0x0000, 0x0000, 0x0000, /* 66 */ - 0x0000, 0x0000, 0x0000, 0x1000, /* 6e */ - 0x0000, 0x0000, 0x0000, 0x0006, /* 76 */ - 0x0000, 0x0000, 0x0000, 0x0000, /* 7e */ +static const struct reg_default stac9766_reg_defaults[] = { + { 0x02, 0x8000 }, + { 0x04, 0x8000 }, + { 0x06, 0x8000 }, + { 0x0a, 0x0000 }, + { 0x0c, 0x8008 }, + { 0x0e, 0x8008 }, + { 0x10, 0x8808 }, + { 0x12, 0x8808 }, + { 0x14, 0x8808 }, + { 0x16, 0x8808 }, + { 0x18, 0x8808 }, + { 0x1a, 0x0000 }, + { 0x1c, 0x8000 }, + { 0x20, 0x0000 }, + { 0x22, 0x0000 }, + { 0x28, 0x0a05 }, + { 0x2c, 0xbb80 }, + { 0x32, 0xbb80 }, + { 0x3a, 0x2000 }, + { 0x3e, 0x0100 }, + { 0x4c, 0x0300 }, + { 0x4e, 0xffff }, + { 0x50, 0x0000 }, + { 0x52, 0x0000 }, + { 0x54, 0x0000 }, + { 0x6a, 0x0000 }, + { 0x6e, 0x1000 }, + { 0x72, 0x0000 }, + { 0x78, 0x0000 }, +}; + +static const struct regmap_config stac9766_regmap_config = { + .reg_bits = 16, + .reg_stride = 2, + .val_bits = 16, + .max_register = 0x78, + .cache_type = REGCACHE_RBTREE, + + .volatile_reg = regmap_ac97_default_volatile, + + .reg_defaults = stac9766_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(stac9766_reg_defaults), }; static const char *stac9766_record_mux[] = {"Mic", "CD", "Video", "AUX", @@ -144,34 +168,13 @@ static const struct snd_kcontrol_new stac9766_snd_ac97_controls[] = { static int stac9766_ac97_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int val) { - struct snd_ac97 *ac97 = snd_soc_codec_get_drvdata(codec); - u16 *cache = codec->reg_cache; - - if (reg / 2 >= ARRAY_SIZE(stac9766_reg)) - return -EIO; - - soc_ac97_ops->write(ac97, reg, val); - cache[reg / 2] = val; - return 0; + return snd_soc_write(codec, reg, val); } static unsigned int stac9766_ac97_read(struct snd_soc_codec *codec, unsigned int reg) { - struct snd_ac97 *ac97 = snd_soc_codec_get_drvdata(codec); - u16 val = 0, *cache = codec->reg_cache; - - if (reg / 2 >= ARRAY_SIZE(stac9766_reg)) - return -EIO; - - if (reg == AC97_RESET || reg == AC97_GPIO_STATUS || - reg == AC97_INT_PAGING || reg == AC97_VENDOR_ID1 || - reg == AC97_VENDOR_ID2) { - - val = soc_ac97_ops->read(ac97, reg); - return val; - } - return cache[reg / 2]; + return snd_soc_read(codec, reg); } static int ac97_analog_prepare(struct snd_pcm_substream *substream, @@ -290,21 +293,34 @@ static struct snd_soc_dai_driver stac9766_dai[] = { static int stac9766_codec_probe(struct snd_soc_codec *codec) { struct snd_ac97 *ac97; + struct regmap *regmap; + int ret; ac97 = snd_soc_new_ac97_codec(codec, STAC9766_VENDOR_ID, STAC9766_VENDOR_ID_MASK); if (IS_ERR(ac97)) return PTR_ERR(ac97); + regmap = regmap_init_ac97(ac97, &stac9766_regmap_config); + if (IS_ERR(regmap)) { + ret = PTR_ERR(regmap); + goto err_free_ac97; + } + + snd_soc_codec_init_regmap(codec, regmap); snd_soc_codec_set_drvdata(codec, ac97); return 0; +err_free_ac97: + snd_soc_free_ac97_codec(ac97); + return ret; } static int stac9766_codec_remove(struct snd_soc_codec *codec) { struct snd_ac97 *ac97 = snd_soc_codec_get_drvdata(codec); + snd_soc_codec_exit_regmap(codec); snd_soc_free_ac97_codec(ac97); return 0; } @@ -314,17 +330,11 @@ static struct snd_soc_codec_driver soc_codec_dev_stac9766 = { .controls = stac9766_snd_ac97_controls, .num_controls = ARRAY_SIZE(stac9766_snd_ac97_controls), }, - .write = stac9766_ac97_write, - .read = stac9766_ac97_read, .set_bias_level = stac9766_set_bias_level, .suspend_bias_off = true, .probe = stac9766_codec_probe, .remove = stac9766_codec_remove, .resume = stac9766_codec_resume, - .reg_cache_size = ARRAY_SIZE(stac9766_reg), - .reg_word_size = sizeof(u16), - .reg_cache_step = 2, - .reg_cache_default = stac9766_reg, }; static int stac9766_probe(struct platform_device *pdev) -- 2.10.1

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[alsa-devel] Applied "ASoC: stac9766: Remove ac97_read/ac97_write wrappers" to the asoc tree
by Mark Brown 04 Nov '16

04 Nov '16

The patch ASoC: stac9766: Remove ac97_read/ac97_write wrappers has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 7aacbc7ff7f6da9ec6deb833154f0883497ab82f Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen <lars(a)metafoo.de> Date: Fri, 4 Nov 2016 18:26:57 +0100 Subject: [PATCH] ASoC: stac9766: Remove ac97_read/ac97_write wrappers Since the regmap conversion ac97_read/ac97_write are just simple wrappers around snd_soc_read/snd_soc_write. Use those instead directly and remove the wrappers. Also use snd_soc_update_bits() where appropriate. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/stac9766.c | 40 +++++++++++----------------------------- 1 file changed, 11 insertions(+), 29 deletions(-) diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c index 62cbeedf93b9..9de7fe8af255 100644 --- a/sound/soc/codecs/stac9766.c +++ b/sound/soc/codecs/stac9766.c @@ -165,38 +165,22 @@ static const struct snd_kcontrol_new stac9766_snd_ac97_controls[] = { SOC_ENUM("Pop Bypass Mux", stac9766_popbypass_enum), }; -static int stac9766_ac97_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int val) -{ - return snd_soc_write(codec, reg, val); -} - -static unsigned int stac9766_ac97_read(struct snd_soc_codec *codec, - unsigned int reg) -{ - return snd_soc_read(codec, reg); -} - static int ac97_analog_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct snd_soc_codec *codec = dai->codec; struct snd_pcm_runtime *runtime = substream->runtime; - unsigned short reg, vra; - - vra = stac9766_ac97_read(codec, AC97_EXTENDED_STATUS); + unsigned short reg; - vra |= 0x1; /* enable variable rate audio */ - vra &= ~0x4; /* disable SPDIF output */ - - stac9766_ac97_write(codec, AC97_EXTENDED_STATUS, vra); + /* enable variable rate audio, disable SPDIF output */ + snd_soc_update_bits(codec, AC97_EXTENDED_STATUS, 0x5, 0x1); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) reg = AC97_PCM_FRONT_DAC_RATE; else reg = AC97_PCM_LR_ADC_RATE; - return stac9766_ac97_write(codec, reg, runtime->rate); + return snd_soc_write(codec, reg, runtime->rate); } static int ac97_digital_prepare(struct snd_pcm_substream *substream, @@ -204,18 +188,16 @@ static int ac97_digital_prepare(struct snd_pcm_substream *substream, { struct snd_soc_codec *codec = dai->codec; struct snd_pcm_runtime *runtime = substream->runtime; - unsigned short reg, vra; - - stac9766_ac97_write(codec, AC97_SPDIF, 0x2002); + unsigned short reg; - vra = stac9766_ac97_read(codec, AC97_EXTENDED_STATUS); - vra |= 0x5; /* Enable VRA and SPDIF out */ + snd_soc_write(codec, AC97_SPDIF, 0x2002); - stac9766_ac97_write(codec, AC97_EXTENDED_STATUS, vra); + /* Enable VRA and SPDIF out */ + snd_soc_update_bits(codec, AC97_EXTENDED_STATUS, 0x5, 0x5); reg = AC97_PCM_FRONT_DAC_RATE; - return stac9766_ac97_write(codec, reg, runtime->rate); + return snd_soc_write(codec, reg, runtime->rate); } static int stac9766_set_bias_level(struct snd_soc_codec *codec, @@ -225,11 +207,11 @@ static int stac9766_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_ON: /* full On */ case SND_SOC_BIAS_PREPARE: /* partial On */ case SND_SOC_BIAS_STANDBY: /* Off, with power */ - stac9766_ac97_write(codec, AC97_POWERDOWN, 0x0000); + snd_soc_write(codec, AC97_POWERDOWN, 0x0000); break; case SND_SOC_BIAS_OFF: /* Off, without power */ /* disable everything including AC link */ - stac9766_ac97_write(codec, AC97_POWERDOWN, 0xffff); + snd_soc_write(codec, AC97_POWERDOWN, 0xffff); break; } return 0; -- 2.10.1

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[alsa-devel] [PATCH v4 00/11] ASoC: topology: Remaining kernel patches
by mengdong.lin＠linux.intel.com 04 Nov '16

04 Nov '16

From: Mengdong Lin <mengdong.lin(a)linux.intel.com> This version can support old topology ABI v4 in a backward compatible way, assuming users start to use topology from ABI v4. This series contains all remaining kernel patches of topology, including some ABI update to PCM (FrontEnds) and link (BackEnds) objects. User will be able to config existing BE DAI links, configure more for FE links. Code are verified and can cover reqeust of Intel pre-release platforms for next year, so ABI should be stable. Current kernel topology code does not really touch Codec-Codec links since there is no user requst atm. We can add support for CC links later by reusing code and data structures for BE links, and no need to revise ABI. History: v2: Add the reason for creating BE DAI & DAI links by topology to commit message. Drop support for configuring DPCM trigger ordering in topology. v3: Topology no longer creates BE DAI or BE DAI links, but only configure existing ones. The API to find a DAI link can also check the name and stream name in case a soc card doesn't use unique ID for DAI links. The user space code is also ready. v4: Support ABI udpate and be backward compatible for topology ABI v4. Code refactoring to configure physical DAIs. Mengdong Lin (11): ASoC: topology: Make manifest backward compatible from ABI v4 ASoC: topology: Make PCM backward compatible from ABI v4 ASoC: topology: Not quit on topology file of ABI version 4 ASoC: topology: ABI - Add flags and private data to PCM ASoC: topology: ABI - Define DAI physical PCM data formats ASoC: topology: ABI - Update physical DAI link configuration for version 5 ASoC: Define API to find a dai link ASoC: topology: Add support to configure existing physical DAI links ASoC: topology: Rename the function to create a FE link ASoC: topology: ABI - Rename struct and type for physical DAIs ASoC: topology: Rename functions & variables for physical links include/sound/soc-dai.h | 15 +- include/sound/soc.h | 3 + include/uapi/sound/asoc.h | 91 +++++++- sound/soc/soc-core.c | 42 ++++ sound/soc/soc-topology.c | 544 +++++++++++++++++++++++++++++++++++++++++----- 5 files changed, 623 insertions(+), 72 deletions(-) -- 2.5.0

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[alsa-devel] [PATCH v5 00/11] ASoC: topology: Remaining kernel patches
by mengdong.lin＠linux.intel.com 04 Nov '16

04 Nov '16

From: Mengdong Lin <mengdong.lin(a)linux.intel.com> This version can support old topology ABI v4 in a backward compatible way, assuming users start to use topology from ABI v4 with alsa-lib v1.1.0. This series contains all remaining kernel patches of topology, including some ABI update to PCM (FrontEnds) and link (BackEnds) objects. Kernel can support topology files generated by ABI v4 without these updates. User will be able to config existing physical DAI links, configure more for FE links. Code are verified and can cover reqeust of Intel pre-release platforms for next year, so ABI should be stable. Current kernel topology code does not really touch Codec-Codec links since there is no user requst atm. We can add support for CC links later by reusing code and data structures for BE links, and no need to revise ABI. History: v2: Add the reason for creating BE DAI & DAI links by topology to commit message. Drop support for configuring DPCM trigger ordering in topology. v3: Topology no longer creates BE DAI or BE DAI links, but only configure existing ones. The API to find a DAI link can also check the name and stream name in case a soc card doesn't use unique ID for DAI links. The user space code is also ready. v4: Support ABI udpate and be backward compatible for topology ABI v4. Code refactoring to configure physical DAIs. v5: Minor code refactoring and modification on commit messages. Mengdong Lin (11): ASoC: topology: Make manifest backward compatible from ABI v4 ASoC: topology: Make PCM backward compatible from ABI v4 ASoC: topology: Support topology file of ABI v4 ASoC: topology: ABI - Add flags and private data to PCM ASoC: topology: ABI - Define DAI physical PCM data formats ASoC: topology: ABI - Update physical DAI link configuration for version 5 ASoC: Define API to find a dai link ASoC: topology: Add support to configure existing physical DAI links ASoC: topology: Rename the function to create a FE link ASoC: topology: ABI - Rename struct and type for physical DAIs ASoC: topology: Rename functions & variables for physical DAIs include/sound/soc-dai.h | 15 +- include/sound/soc.h | 3 + include/uapi/sound/asoc.h | 89 +++++++- sound/soc/soc-core.c | 42 ++++ sound/soc/soc-topology.c | 551 +++++++++++++++++++++++++++++++++++++++++----- 5 files changed, 627 insertions(+), 73 deletions(-) -- 2.5.0

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[alsa-devel] [PATCH 1/5] ASoC: stac9766: Remove unused DAI ID defines
by Lars-Peter Clausen 04 Nov '16

04 Nov '16

The DAI ID defines are back from the time when DAIs were referenced by a numerical ID. These days a string is used instead and the defines are unused. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/stac9766.h | 4 ---- 1 file changed, 4 deletions(-) diff --git a/sound/soc/codecs/stac9766.h b/sound/soc/codecs/stac9766.h index c726f90..cb0d550 100644 --- a/sound/soc/codecs/stac9766.h +++ b/sound/soc/codecs/stac9766.h @@ -10,8 +10,4 @@ #define AC97_STAC_ANALOG_SPECIAL (AC97_STAC_PAGE0 | 0x6E) #define AC97_STAC_STEREO_MIC 0x78 -/* STAC9766 DAI ID's */ -#define STAC9766_DAI_AC97_ANALOG 0 -#define STAC9766_DAI_AC97_DIGITAL 1 - #endif -- 2.1.4

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[alsa-devel] [PATCH v3 0/8] ASoC: topology: Remaining kernel patches
by mengdong.lin＠linux.intel.com 04 Nov '16

04 Nov '16

From: Mengdong Lin <mengdong.lin(a)linux.intel.com> This series contains all remaining kernel patches of topology, including some ABI update to PCM (FrontEnds) and link (BackEnds) objects. User will be able to config existing BE DAI links, configure more for FE links. Code are verified and can cover reqeust of Intel pre-release platforms for next year, so ABI should be stable. Current kernel topology code does not really touch Codec-Codec links since there is no user requst atm. We can add support for CC links later by reusing code and data structures for BE links, and no need to revise ABI. History: v2: Add the reason for creating BE DAI & DAI links by topology to commit message. Drop support for configuring DPCM trigger ordering in topology. v3: Topology no longer creates BE DAI or BE DAI links, but only configure existing ones. The API to find a DAI link can also check the name and stream name in case a soc card doesn't use unique ID for DAI links. The user space code is also ready. Mengdong Lin (8): ASoC: topology: ABI - Add flags to PCM ASoC: topology: ABI - Add private data to PCM ASoC: topology: ABI - Define DAI physical PCM data formats ASoC: topology: ABI - Add HW configurations to BE DAI links ASoC: topology: ABI - Add flags and private data to BE DAI links ASoC: topology: ABI - Add name and stream name to BE DAI links ASoC: Define API to find a dai link ASoC: topology: Add support to configure existing BE DAI links include/sound/soc-dai.h | 15 +++-- include/sound/soc.h | 3 + include/uapi/sound/asoc.h | 67 +++++++++++++++++++ sound/soc/soc-core.c | 42 ++++++++++++ sound/soc/soc-topology.c | 166 ++++++++++++++++++++++++++++++++++++++++++++-- 5 files changed, 282 insertions(+), 11 deletions(-) -- 2.5.0

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[alsa-devel] Applied "ASoC: codecs: msm8916-wcd-analog: clean parse_dt()" to the asoc tree
by Mark Brown 04 Nov '16

04 Nov '16

The patch ASoC: codecs: msm8916-wcd-analog: clean parse_dt() has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 4323ec250cbc675c7fc40edf5f791a736210747b Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Date: Fri, 4 Nov 2016 14:45:39 +0000 Subject: [PATCH] ASoC: codecs: msm8916-wcd-analog: clean parse_dt() Move the code which is not parsing dt from pm8916_wcd_analog_parse_dt() to make it clear to reader. No functional changes done. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/msm8916-wcd-analog.c | 34 +++++++++++++++++----------------- 1 file changed, 17 insertions(+), 17 deletions(-) diff --git a/sound/soc/codecs/msm8916-wcd-analog.c b/sound/soc/codecs/msm8916-wcd-analog.c index d9f999b14b28..50ad75ab10ea 100644 --- a/sound/soc/codecs/msm8916-wcd-analog.c +++ b/sound/soc/codecs/msm8916-wcd-analog.c @@ -802,7 +802,6 @@ static struct snd_soc_codec_driver pm8916_wcd_analog = { static int pm8916_wcd_analog_parse_dt(struct device *dev, struct pm8916_wcd_analog_priv *priv) { - int ret, i; if (of_property_read_bool(dev->of_node, "qcom,micbias1-ext-cap")) priv->micbias1_cap_mode = MICB_1_EN_EXT_BYP_CAP; @@ -814,21 +813,6 @@ static int pm8916_wcd_analog_parse_dt(struct device *dev, else priv->micbias2_cap_mode = MICB_1_EN_NO_EXT_BYP_CAP; - priv->mclk = devm_clk_get(dev, "mclk"); - if (IS_ERR(priv->mclk)) { - dev_err(dev, "failed to get mclk\n"); - return PTR_ERR(priv->mclk); - } - for (i = 0; i < ARRAY_SIZE(supply_names); i++) - priv->supplies[i].supply = supply_names[i]; - - ret = devm_regulator_bulk_get(dev, ARRAY_SIZE(priv->supplies), - priv->supplies); - if (ret) { - dev_err(dev, "Failed to get regulator supplies %d\n", ret); - return ret; - } - return 0; } @@ -836,7 +820,7 @@ static int pm8916_wcd_analog_spmi_probe(struct platform_device *pdev) { struct pm8916_wcd_analog_priv *priv; struct device *dev = &pdev->dev; - int ret; + int ret, i; priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); if (!priv) @@ -846,6 +830,22 @@ static int pm8916_wcd_analog_spmi_probe(struct platform_device *pdev) if (ret < 0) return ret; + priv->mclk = devm_clk_get(dev, "mclk"); + if (IS_ERR(priv->mclk)) { + dev_err(dev, "failed to get mclk\n"); + return PTR_ERR(priv->mclk); + } + + for (i = 0; i < ARRAY_SIZE(supply_names); i++) + priv->supplies[i].supply = supply_names[i]; + + ret = devm_regulator_bulk_get(dev, ARRAY_SIZE(priv->supplies), + priv->supplies); + if (ret) { + dev_err(dev, "Failed to get regulator supplies %d\n", ret); + return ret; + } + ret = clk_prepare_enable(priv->mclk); if (ret < 0) { dev_err(dev, "failed to enable mclk %d\n", ret); -- 2.10.1

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[alsa-devel] [PATCH 1/2] ASoC: codecs: msm8916-wcd-analog: clean parse_dt()
by Srinivas Kandagatla 04 Nov '16

04 Nov '16

Move the code which is not parsing dt from pm8916_wcd_analog_parse_dt() to make it clear to reader. No functional changes done. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- This change was suggested by Mark http://www.spinics.net/lists/alsa-devel/msg55878.html thanks, srini sound/soc/codecs/msm8916-wcd-analog.c | 34 +++++++++++++++++----------------- 1 file changed, 17 insertions(+), 17 deletions(-) diff --git a/sound/soc/codecs/msm8916-wcd-analog.c b/sound/soc/codecs/msm8916-wcd-analog.c index d9f999b..50ad75a 100644 --- a/sound/soc/codecs/msm8916-wcd-analog.c +++ b/sound/soc/codecs/msm8916-wcd-analog.c @@ -802,7 +802,6 @@ static struct snd_soc_codec_driver pm8916_wcd_analog = { static int pm8916_wcd_analog_parse_dt(struct device *dev, struct pm8916_wcd_analog_priv *priv) { - int ret, i; if (of_property_read_bool(dev->of_node, "qcom,micbias1-ext-cap")) priv->micbias1_cap_mode = MICB_1_EN_EXT_BYP_CAP; @@ -814,21 +813,6 @@ static int pm8916_wcd_analog_parse_dt(struct device *dev, else priv->micbias2_cap_mode = MICB_1_EN_NO_EXT_BYP_CAP; - priv->mclk = devm_clk_get(dev, "mclk"); - if (IS_ERR(priv->mclk)) { - dev_err(dev, "failed to get mclk\n"); - return PTR_ERR(priv->mclk); - } - for (i = 0; i < ARRAY_SIZE(supply_names); i++) - priv->supplies[i].supply = supply_names[i]; - - ret = devm_regulator_bulk_get(dev, ARRAY_SIZE(priv->supplies), - priv->supplies); - if (ret) { - dev_err(dev, "Failed to get regulator supplies %d\n", ret); - return ret; - } - return 0; } @@ -836,7 +820,7 @@ static int pm8916_wcd_analog_spmi_probe(struct platform_device *pdev) { struct pm8916_wcd_analog_priv *priv; struct device *dev = &pdev->dev; - int ret; + int ret, i; priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); if (!priv) @@ -846,6 +830,22 @@ static int pm8916_wcd_analog_spmi_probe(struct platform_device *pdev) if (ret < 0) return ret; + priv->mclk = devm_clk_get(dev, "mclk"); + if (IS_ERR(priv->mclk)) { + dev_err(dev, "failed to get mclk\n"); + return PTR_ERR(priv->mclk); + } + + for (i = 0; i < ARRAY_SIZE(supply_names); i++) + priv->supplies[i].supply = supply_names[i]; + + ret = devm_regulator_bulk_get(dev, ARRAY_SIZE(priv->supplies), + priv->supplies); + if (ret) { + dev_err(dev, "Failed to get regulator supplies %d\n", ret); + return ret; + } + ret = clk_prepare_enable(priv->mclk); if (ret < 0) { dev_err(dev, "failed to enable mclk %d\n", ret); -- 2.10.1

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