May 2017 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH -next] ASoC: rt5665: make local symbol rt5665_i2c_driver static
by Wei Yongjun 17 May '17

17 May '17

From: Wei Yongjun <weiyongjun1(a)huawei.com> Fixes the following sparse warnings: sound/soc/codecs/rt5665.c:4928:19: warning: symbol 'rt5665_i2c_driver' was not declared. Should it be static? Signed-off-by: Wei Yongjun <weiyongjun1(a)huawei.com> --- sound/soc/codecs/rt5665.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5665.c b/sound/soc/codecs/rt5665.c index 26bf157..bb440bd 100644 --- a/sound/soc/codecs/rt5665.c +++ b/sound/soc/codecs/rt5665.c @@ -4925,7 +4925,7 @@ static struct acpi_device_id rt5665_acpi_match[] = { MODULE_DEVICE_TABLE(acpi, rt5665_acpi_match); #endif -struct i2c_driver rt5665_i2c_driver = { +static struct i2c_driver rt5665_i2c_driver = { .driver = { .name = "rt5665", .of_match_table = of_match_ptr(rt5665_of_match),

2 2

[alsa-devel] [PATCH] ASoC: fsi: Move inline fsi_stream_is_play() before use
by Geert Uytterhoeven 17 May '17

17 May '17

With gcc 4.1.2: sound/soc/sh/fsi.c:304: warning: ‘fsi_stream_is_play’ declared inline after being called sound/soc/sh/fsi.c:304: warning: previous declaration of ‘fsi_stream_is_play’ was here Move fsi_stream_is_play() up to fix this, removing the need for a forward declaration as well. Signed-off-by: Geert Uytterhoeven <geert(a)linux-m68k.org> --- sound/soc/sh/fsi.c | 13 ++++++------- 1 file changed, 6 insertions(+), 7 deletions(-) diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index ead520182e2684fb..7c4bdd82bb955bbb 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -301,7 +301,12 @@ struct fsi_master { spinlock_t lock; }; -static int fsi_stream_is_play(struct fsi_priv *fsi, struct fsi_stream *io); +static inline int fsi_stream_is_play(struct fsi_priv *fsi, + struct fsi_stream *io) +{ + return &fsi->playback == io; +} + /* * basic read write function @@ -489,12 +494,6 @@ static void fsi_count_fifo_err(struct fsi_priv *fsi) /* * fsi_stream_xx() function */ -static inline int fsi_stream_is_play(struct fsi_priv *fsi, - struct fsi_stream *io) -{ - return &fsi->playback == io; -} - static inline struct fsi_stream *fsi_stream_get(struct fsi_priv *fsi, struct snd_pcm_substream *substream) { -- 2.7.4

2 1

[alsa-devel] [PATCH] ASoC: atmel: Use IS_ENABLED()
by Takashi Iwai 17 May '17

17 May '17

Simplify the ifdef conditions with IS_ENABLED() macro. No functional changes. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/soc/atmel/atmel-pcm.h | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/soc/atmel/atmel-pcm.h b/sound/soc/atmel/atmel-pcm.h index 6eaf081cad50..4b27aed40a51 100644 --- a/sound/soc/atmel/atmel-pcm.h +++ b/sound/soc/atmel/atmel-pcm.h @@ -83,8 +83,7 @@ struct atmel_pcm_dma_params { #define ssc_readx(base, reg) (__raw_readl((base) + (reg))) #define ssc_writex(base, reg, value) __raw_writel((value), (base) + (reg)) -#if defined(CONFIG_SND_ATMEL_SOC_PDC) || \ - defined(CONFIG_SND_ATMEL_SOC_PDC_MODULE) +#if IS_ENABLED(CONFIG_SND_ATMEL_SOC_PDC) int atmel_pcm_pdc_platform_register(struct device *dev); void atmel_pcm_pdc_platform_unregister(struct device *dev); #else @@ -97,8 +96,7 @@ static inline void atmel_pcm_pdc_platform_unregister(struct device *dev) } #endif -#if defined(CONFIG_SND_ATMEL_SOC_DMA) || \ - defined(CONFIG_SND_ATMEL_SOC_DMA_MODULE) +#if IS_ENABLED(CONFIG_SND_ATMEL_SOC_DMA) int atmel_pcm_dma_platform_register(struct device *dev); void atmel_pcm_dma_platform_unregister(struct device *dev); #else -- 2.12.2

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[alsa-devel] [PATCH] ASoC: rsnd: don't use PDTA bit for 24bit on SSI
by Kuninori Morimoto 17 May '17

17 May '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Current SSI uses PDTA bit which indicates data that Input/Output data are Right-Aligned. But, 24bit sound should be Left-Aligned in this HW. Because Linux is using Right-Aligned data, and HW uses Left-Aligned data, current 24bit data is missing lower 8bit. To fix this issue, this patch removes PDTA bit, and shift 8bit in necessary module Reported-by: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx(a)renesas.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Tested-by: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx(a)renesas.com> --- sound/soc/sh/rcar/cmd.c | 1 + sound/soc/sh/rcar/core.c | 51 ++++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/sh/rcar/gen.c | 1 + sound/soc/sh/rcar/rsnd.h | 2 ++ sound/soc/sh/rcar/src.c | 12 ++++++++++-- sound/soc/sh/rcar/ssi.c | 2 +- sound/soc/sh/rcar/ssiu.c | 3 ++- 7 files changed, 68 insertions(+), 4 deletions(-) diff --git a/sound/soc/sh/rcar/cmd.c b/sound/soc/sh/rcar/cmd.c index 7d92a24..d879c01 100644 --- a/sound/soc/sh/rcar/cmd.c +++ b/sound/soc/sh/rcar/cmd.c @@ -89,6 +89,7 @@ static int rsnd_cmd_init(struct rsnd_mod *mod, dev_dbg(dev, "ctu/mix path = 0x%08x", data); rsnd_mod_write(mod, CMD_ROUTE_SLCT, data); + rsnd_mod_write(mod, CMD_BUSIF_MODE, rsnd_get_busif_shift(io, mod) | 1); rsnd_mod_write(mod, CMD_BUSIF_DALIGN, rsnd_get_dalign(mod, io)); rsnd_adg_set_cmd_timsel_gen2(mod, io); diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index e36a6b5..6264b39 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -343,6 +343,57 @@ u32 rsnd_get_dalign(struct rsnd_mod *mod, struct rsnd_dai_stream *io) return 0x76543210; } +u32 rsnd_get_busif_shift(struct rsnd_dai_stream *io, struct rsnd_mod *mod) +{ + enum rsnd_mod_type playback_mods[] = { + RSND_MOD_SRC, + RSND_MOD_CMD, + RSND_MOD_SSIU, + }; + enum rsnd_mod_type capture_mods[] = { + RSND_MOD_CMD, + RSND_MOD_SRC, + RSND_MOD_SSIU, + }; + struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io); + struct rsnd_mod *tmod = NULL; + enum rsnd_mod_type *mods = + rsnd_io_is_play(io) ? + playback_mods : capture_mods; + int i; + + /* + * This is needed for 24bit data + * We need to shift 8bit + * + * Linux 24bit data is located as 0x00****** + * HW 24bit data is located as 0x******00 + * + */ + switch (runtime->sample_bits) { + case 16: + return 0; + case 32: + break; + } + + for (i = 0; i < ARRAY_SIZE(playback_mods); i++) { + tmod = rsnd_io_to_mod(io, mods[i]); + if (tmod) + break; + } + + if (tmod != mod) + return 0; + + if (rsnd_io_is_play(io)) + return (0 << 20) | /* shift to Left */ + (8 << 16); /* 8bit */ + else + return (1 << 20) | /* shift to Right */ + (8 << 16); /* 8bit */ +} + /* * rsnd_dai functions */ diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c index 99f93a17..ee00e35 100644 --- a/sound/soc/sh/rcar/gen.c +++ b/sound/soc/sh/rcar/gen.c @@ -238,6 +238,7 @@ static int rsnd_gen2_probe(struct rsnd_priv *priv) RSND_GEN_M_REG(SRC_ROUTE_MODE0, 0xc, 0x20), RSND_GEN_M_REG(SRC_CTRL, 0x10, 0x20), RSND_GEN_M_REG(SRC_INT_ENABLE0, 0x18, 0x20), + RSND_GEN_M_REG(CMD_BUSIF_MODE, 0x184, 0x20), RSND_GEN_M_REG(CMD_BUSIF_DALIGN,0x188, 0x20), RSND_GEN_M_REG(CMD_ROUTE_SLCT, 0x18c, 0x20), RSND_GEN_M_REG(CMD_CTRL, 0x190, 0x20), diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index af6d57b..cc929fc 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -74,6 +74,7 @@ enum rsnd_reg { RSND_REG_SCU_SYS_INT_EN0, RSND_REG_SCU_SYS_INT_EN1, RSND_REG_CMD_CTRL, + RSND_REG_CMD_BUSIF_MODE, RSND_REG_CMD_BUSIF_DALIGN, RSND_REG_CMD_ROUTE_SLCT, RSND_REG_CMDOUT_TIMSEL, @@ -207,6 +208,7 @@ void rsnd_bset(struct rsnd_priv *priv, struct rsnd_mod *mod, enum rsnd_reg reg, u32 mask, u32 data); u32 rsnd_get_adinr_bit(struct rsnd_mod *mod, struct rsnd_dai_stream *io); u32 rsnd_get_dalign(struct rsnd_mod *mod, struct rsnd_dai_stream *io); +u32 rsnd_get_busif_shift(struct rsnd_dai_stream *io, struct rsnd_mod *mod); /* * R-Car DMA diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c index e395fcb..7aa239e 100644 --- a/sound/soc/sh/rcar/src.c +++ b/sound/soc/sh/rcar/src.c @@ -172,11 +172,13 @@ static void rsnd_src_set_convert_rate(struct rsnd_dai_stream *io, struct rsnd_priv *priv = rsnd_mod_to_priv(mod); struct device *dev = rsnd_priv_to_dev(priv); struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io); + int is_play = rsnd_io_is_play(io); int use_src = 0; u32 fin, fout; u32 ifscr, fsrate, adinr; u32 cr, route; u32 bsdsr, bsisr; + u32 i_busif, o_busif, tmp; uint ratio; if (!runtime) @@ -252,6 +254,11 @@ static void rsnd_src_set_convert_rate(struct rsnd_dai_stream *io, break; } + /* BUSIF_MODE */ + tmp = rsnd_get_busif_shift(io, mod); + i_busif = ( is_play ? tmp : 0) | 1; + o_busif = (!is_play ? tmp : 0) | 1; + rsnd_mod_write(mod, SRC_ROUTE_MODE0, route); rsnd_mod_write(mod, SRC_SRCIR, 1); /* initialize */ @@ -263,8 +270,9 @@ static void rsnd_src_set_convert_rate(struct rsnd_dai_stream *io, rsnd_mod_write(mod, SRC_BSISR, bsisr); rsnd_mod_write(mod, SRC_SRCIR, 0); /* cancel initialize */ - rsnd_mod_write(mod, SRC_I_BUSIF_MODE, 1); - rsnd_mod_write(mod, SRC_O_BUSIF_MODE, 1); + rsnd_mod_write(mod, SRC_I_BUSIF_MODE, i_busif); + rsnd_mod_write(mod, SRC_O_BUSIF_MODE, o_busif); + rsnd_mod_write(mod, SRC_BUSIF_DALIGN, rsnd_get_dalign(mod, io)); rsnd_adg_set_src_timesel_gen2(mod, io, fin, fout); diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index d0602c1..5960042 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -319,7 +319,7 @@ static void rsnd_ssi_config_init(struct rsnd_mod *mod, * always use 32bit system word. * see also rsnd_ssi_master_clk_enable() */ - cr_own = FORCE | SWL_32 | PDTA; + cr_own = FORCE | SWL_32; if (rdai->bit_clk_inv) cr_own |= SCKP; diff --git a/sound/soc/sh/rcar/ssiu.c b/sound/soc/sh/rcar/ssiu.c index 13d648e..bed2c9c 100644 --- a/sound/soc/sh/rcar/ssiu.c +++ b/sound/soc/sh/rcar/ssiu.c @@ -145,7 +145,8 @@ static int rsnd_ssiu_init_gen2(struct rsnd_mod *mod, (rsnd_io_is_play(io) ? rsnd_runtime_channel_after_ctu(io) : rsnd_runtime_channel_original(io))); - rsnd_mod_write(mod, SSI_BUSIF_MODE, 1); + rsnd_mod_write(mod, SSI_BUSIF_MODE, + rsnd_get_busif_shift(io, mod) | 1); rsnd_mod_write(mod, SSI_BUSIF_DALIGN, rsnd_get_dalign(mod, io)); } -- 1.9.1

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[alsa-devel] [PATCH] ASoC: rt5514: Add ACPI match ID
by Oder Chiou 17 May '17

17 May '17

This patch adds the ACPI match ID for rt5514 codec. Signed-off-by: Hsin-Yu Chao <hychao(a)chromium.org> Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5514.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/soc/codecs/rt5514.c b/sound/soc/codecs/rt5514.c index 6856b8c..1b6796c 100644 --- a/sound/soc/codecs/rt5514.c +++ b/sound/soc/codecs/rt5514.c @@ -9,6 +9,7 @@ * published by the Free Software Foundation. */ +#include <linux/acpi.h> #include <linux/fs.h> #include <linux/module.h> #include <linux/moduleparam.h> @@ -1095,6 +1096,14 @@ static const struct of_device_id rt5514_of_match[] = { MODULE_DEVICE_TABLE(of, rt5514_of_match); #endif +#ifdef CONFIG_ACPI +static struct acpi_device_id rt5514_acpi_match[] = { + { "10EC5514", 0}, + {}, +}; +MODULE_DEVICE_TABLE(acpi, rt5514_acpi_match); +#endif + static int rt5514_parse_dt(struct rt5514_priv *rt5514, struct device *dev) { device_property_read_u32(dev, "realtek,dmic-init-delay-ms", @@ -1198,6 +1207,7 @@ static const struct dev_pm_ops rt5514_i2_pm_ops = { static struct i2c_driver rt5514_i2c_driver = { .driver = { .name = "rt5514", + .acpi_match_table = ACPI_PTR(rt5514_acpi_match), .of_match_table = of_match_ptr(rt5514_of_match), .pm = &rt5514_i2_pm_ops, }, -- 2.7.4

2 1

[alsa-devel] [PATCH] ASoC: atmel-classd: sync regcache when resuming
by Quentin Schulz 17 May '17

17 May '17

The PM functions used in this driver are the ones defined in sounc/soc/soc-core.c. When suspending (using snd_soc_suspend), the regcache is marked dirty but is never synced on resume. Sync regcache on resume of Atmel ClassD device. Signed-off-by: Quentin Schulz <quentin.schulz(a)free-electrons.com> --- sound/soc/atmel/atmel-classd.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/atmel/atmel-classd.c b/sound/soc/atmel/atmel-classd.c index 7ae46c2647d4..b7ef8c59b49a 100644 --- a/sound/soc/atmel/atmel-classd.c +++ b/sound/soc/atmel/atmel-classd.c @@ -301,6 +301,14 @@ static int atmel_classd_codec_probe(struct snd_soc_codec *codec) return 0; } +static int atmel_classd_codec_resume(struct snd_soc_codec *codec) +{ + struct snd_soc_card *card = snd_soc_codec_get_drvdata(codec); + struct atmel_classd *dd = snd_soc_card_get_drvdata(card); + + return regcache_sync(dd->regmap); +} + static struct regmap *atmel_classd_codec_get_remap(struct device *dev) { return dev_get_regmap(dev, NULL); @@ -308,6 +316,7 @@ static struct regmap *atmel_classd_codec_get_remap(struct device *dev) static struct snd_soc_codec_driver soc_codec_dev_classd = { .probe = atmel_classd_codec_probe, + .resume = atmel_classd_codec_resume, .get_regmap = atmel_classd_codec_get_remap, .component_driver = { .controls = atmel_classd_snd_controls, -- 2.11.0

4 3

[alsa-devel] [PATCH v2 0/2] ALSA, dmaengine: dw: Remove AVR32 bits in drivers
by Andy Shevchenko 17 May '17

17 May '17

AVR32 is gone. Remove bits in sound/atmel and drivers/dma/dw folders related to it. Sound part hasn't been tested anyhow. Changes since v1: - fix kbuild bot complainings - append Nicolas' Ack - don't forget to send to all TWIMC Andy Shevchenko (2): ALSA: atmel: Remove AVR32 bits from the driver dmaengine: dw: Remove AVR32 bits from the driver drivers/dma/dw/Kconfig | 7 +- drivers/dma/dw/core.c | 332 +-------------------------- drivers/dma/dw/regs.h | 50 +--- include/linux/dma/dw.h | 21 -- sound/atmel/Kconfig | 13 +- sound/atmel/Makefile | 2 - sound/atmel/abdac.c | 610 ------------------------------------------------- sound/atmel/ac97c.c | 447 ++++++------------------------------ 8 files changed, 88 insertions(+), 1394 deletions(-) delete mode 100644 sound/atmel/abdac.c -- 2.11.0

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[alsa-devel] [PATCH] ALSA: pcm: add const qualifier for read-only table for sampling rate
by Takashi Sakamoto 17 May '17

17 May '17

There's a read-only table for each sampling rate, while it doesn't have const qualifier and can be modified. This commit add the qualifier. As a result, a symbol for the table moves from .data section to .rodata. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/core/pcm_native.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index d873f90..f3a3580 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -1987,8 +1987,10 @@ static int snd_pcm_hw_rule_sample_bits(struct snd_pcm_hw_params *params, #error "Change this table" #endif -static unsigned int rates[] = { 5512, 8000, 11025, 16000, 22050, 32000, 44100, - 48000, 64000, 88200, 96000, 176400, 192000 }; +static const unsigned int rates[] = { + 5512, 8000, 11025, 16000, 22050, 32000, 44100, + 48000, 64000, 88200, 96000, 176400, 192000 +}; const struct snd_pcm_hw_constraint_list snd_pcm_known_rates = { .count = ARRAY_SIZE(rates), -- 2.9.3

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[alsa-devel] HDA cs8409 codec support
by Joël Krähemann 17 May '17

17 May '17

Hi all First of all, I am new to kernel programming and an other attempt already failed to do so. However I got familiar with the Intel HDA Codec specification. So I did a start script to log dmesg and mixer controls of different configurations. But I am not sure if I got a functional kernel driver setup to test things. It seems there is something wrong. The codec has 8 GPIOs and first I didn't set any mask within the kernel. Now, I just compile a kernel set it to 0xff. Any help is appreciated. Finally here is my work: https://github.com/joelkraehemann/hda-tool/ Note: I have tried to subscribe using a different email address but didn't get approved message. Thank you, Joël

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[alsa-devel] [PATCH v2 15/53] docs-rst: add sound book to pdf output
by Mauro Carvalho Chehab 16 May '17

16 May '17

The sound subsystem book was added without the bits required to generate PDF output. Add them. Signed-off-by: Mauro Carvalho Chehab <mchehab(a)s-opensource.com> --- Documentation/conf.py | 2 ++ Documentation/sound/conf.py | 10 ++++++++++ 2 files changed, 12 insertions(+) create mode 100644 Documentation/sound/conf.py diff --git a/Documentation/conf.py b/Documentation/conf.py index 1d461f0c1cd0..bca751960dec 100644 --- a/Documentation/conf.py +++ b/Documentation/conf.py @@ -371,6 +371,8 @@ latex_documents = [ 'The kernel development community', 'manual'), ('security/index', 'security.tex', 'The kernel security subsystem manual', 'The kernel development community', 'manual'), + ('sound/index', 'sound.tex', 'Linux Sound Subsystem Documentation', + 'The kernel development community', 'manual'), ] # The name of an image file (relative to this directory) to place at the top of diff --git a/Documentation/sound/conf.py b/Documentation/sound/conf.py new file mode 100644 index 000000000000..3f1fc5e74e7b --- /dev/null +++ b/Documentation/sound/conf.py @@ -0,0 +1,10 @@ +# -*- coding: utf-8; mode: python -*- + +project = "Linux Sound Subsystem Documentation" + +tags.add("subproject") + +latex_documents = [ + ('index', 'sound.tex', project, + 'The kernel development community', 'manual'), +] -- 2.9.3

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[alsa-devel] [PATCH alsa-lib 1/2] build: Define __USE_UNIX98 for old glibc
by Takashi Iwai 16 May '17

16 May '17

Otherwise PTHREAD_MUTEX_RECURSIVE isn't defined and we get an error with old glibc. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- src/pcm/pcm_local.h | 1 + 1 file changed, 1 insertion(+) diff --git a/src/pcm/pcm_local.h b/src/pcm/pcm_local.h index 32e6dcdf6fcd..e4f652187c49 100644 --- a/src/pcm/pcm_local.h +++ b/src/pcm/pcm_local.h @@ -35,6 +35,7 @@ #include "local.h" #ifdef THREAD_SAFE_API +#define __USE_UNIX98 1 /* for old glibc */ #include <pthread.h> #endif -- 2.13.0

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[alsa-devel] [PATCH RFC 00/21] ALSA: pcm: add tracepoints for PCM params operation
by Takashi Sakamoto 16 May '17

16 May '17

Hi, In the last Audio Mini Conference held with Linux Plumber conference 2016[1], I mentioned about tracepoints for PCM params operation. This patchset is for the idea. In ALSA PCM interface, applications can get hardware capability by ioctl(2) with SNDRV_PCM_IOCTL_HW_REFINE/SNDRV_PCM_IOCT_HW_PARAMS in a shape of 'struct snd_pcm_hw_params'. In kernel side, relevant processing is somewhat complicated and developers sometimes have hard time to debug drivers for PCM constraints and rules. This patchset adds tracepoints for hw_params operations. When CONFIG_SND_DEBUG is enabled, you can see 'trace_hw_params_mask/trace_hw_params_interval' events of 'snd_pcm' subsystem. When applications execute ioctl(2) with SNDRV_PCM_IOCTL_HW_REFINE/SNDRV_PCM_IOCTL_HW_PARAMS, these events are probed. Developers can get how many PCM rules are added into runtime of PCM substream and which rule changed which parameters. This patchset also includes some improvements. The last three commits brings small changes to kernel/userspace interface for error handling. I'm happy to receive your comment for this patchset. For your information, low level application of SNDRV_PCM_IOCTL_HW_REFINE operation is available in my github repository[2]. [1] You can see minutes. http://mailman.alsa-project.org/pipermail/alsa-devel/2016-November/114279.h… [2] refine-pcm-param.c https://github.com/takaswie/alsa-ioctl-test Takashi Sakamoto (21): ALSA: pcm: add a helper function to constrain mask-type parameters ALSA: pcm: add a helper function to constrain interval-type parameters ALSA: pcm: add a helper function to apply parameter rules ALSA: pcm: tracepoints for refining PCM parameters ALSA: pcm: enable parameter tracepoints at CONFIG_SND_DEBUG only ALSA: pcm: use goto statement instead of while statement to reduce indentation ALSA: pcm: remove function local variable with alternative evaluation ALSA: pcm: adaption of code formatting ALSA: pcm: add comment about application of rule to PCM parameters ALSA: pcm: constify function local and read-only table ALSA: pcm: localize snd_pcm_hw_params_choose() ALSA: pcm: add tracepoints for selection process of hardware parameters at SNDRV_PCM_IOCTL_HW_PARAMS ALSA: pcm: use helper functions to check whether parameters are determined ALSA: pcm: use helper function to refer parameter as read-only ALSA: pcm: add const qualifier for read-only table for sampling rate ALSA: pcm/oss: refer to parameters instead of copying to reduce usage of kernel stack ALSA: pcm: check type of parameter in added rule ALSA: pcm: calculate non-mask/non-interval parameters always when possible ALSA: pcm: move fixup of info flag after selecting single parameters ALSA: pcm: return error immediately for parameters refinement ALSA: pcm: return error immediately at refine ioctl include/sound/pcm.h | 1 - sound/core/Makefile | 1 + sound/core/oss/pcm_oss.c | 20 +- sound/core/oss/pcm_plugin.c | 5 +- sound/core/oss/pcm_plugin.h | 2 +- sound/core/pcm_drm_eld.c | 8 +- sound/core/pcm_lib.c | 55 +---- sound/core/pcm_native.c | 485 ++++++++++++++++++++++++++++-------------- sound/core/pcm_params_trace.h | 142 +++++++++++++ 9 files changed, 494 insertions(+), 225 deletions(-) create mode 100644 sound/core/pcm_params_trace.h -- 2.9.3

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[alsa-devel] [PATCH 0/6] Cleanup and removal of set_fs() from HD-audio
by Takashi Iwai 16 May '17

16 May '17

Hi, this is the resubmission of a part of the previous RFC patchset. There is no change since then for them, it's just officially submitted for merging to 4.13. Takashi === Takashi Iwai (6): ALSA: hda - Simplify bound-beep mute control for ALC268 ALSA: hda - Move bind-mixer switch codes to generic parser ALSA: hda - Remove the generic bind ctl helpers ALSA: hda - Remove the use of set_fs() ALSA: hda - Fix a typo in comment ALSA: hda - Remove superfluous header inclusions sound/pci/hda/hda_codec.c | 251 ++++++------------------------------------ sound/pci/hda/hda_generic.c | 46 +++++++- sound/pci/hda/hda_local.h | 61 ---------- sound/pci/hda/patch_realtek.c | 37 +++++-- 4 files changed, 103 insertions(+), 292 deletions(-) -- 2.12.2

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[alsa-devel] [PATCH 0/2] Remove set_fs() from drivers
by Takashi Iwai 16 May '17

16 May '17

Hi, this is again a resubmission of a part of the previous RFC patchset. No changes for them, just submitted as the official patches for merging to 4.13. Takashi === Takashi Iwai (2): ALSA: opl3: Kill unused set_fs() ALSA: emu10k1: Get rid of set_fs() usage sound/drivers/opl3/opl3_oss.c | 14 ----- sound/pci/emu10k1/emufx.c | 127 ++++++++++++++++++++++++------------------ 2 files changed, 73 insertions(+), 68 deletions(-) -- 2.12.2

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[alsa-devel] [PATCH 00/13] Get rid of DocBook
by Mauro Carvalho Chehab 16 May '17

16 May '17

As just one book (lsm) was missing conversion, let's convert it and store as if it were a plain text file under Documentation/lsm.txt, adding a notice that it requires update. This way, as everything is now converted, we can get rid of the DocBook building system and update places that were mentioning it. PS.: We could also remove the parts of the kernel-doc script that produce DocBook outputs, but, as this might still be useful for someone, for now let's just keep the functionality there. - This patch series is based on my past 00/05 and 00/36 patch series that convert the other DocBooks to ReST, applied on the top of docs tree (next branch). The full patch series is on this tree is at: https://git.linuxtv.org//mchehab/experimental.git/log/?h=docbook2 And the HTML output at: http://www.infradead.org/~mchehab/kernel_docs/ https://mchehab.fedorapeople.org/kernel_docs/ Mauro Carvalho Chehab (13): docs-rst: convert lsm from DocBook to ReST docs: remove DocBook from the building system docs: update old references for DocBook from the documentation MAINTAINERS: update old references for DocBook directory ata: update references for libata documentation ia64, scsi: update references for the device-io book irq: update genericirq book location fs: update location of filesystems documentation lib: update location of kgdb documentation sound: fix the comments that refers to kernel-doc fs: fix the location of the kernel-api book usb: fix the comment with regards to DocBook docs-rst: get rid of Documentation/sphinx/tmplcvt script Documentation/00-INDEX | 6 +- Documentation/DocBook/.gitignore | 17 -- Documentation/DocBook/Makefile | 276 ----------------------------- Documentation/DocBook/lsm.tmpl | 265 --------------------------- Documentation/DocBook/stylesheet.xsl | 11 -- Documentation/Makefile | 125 +++++++++++++ Documentation/Makefile.sphinx | 130 -------------- Documentation/PCI/MSI-HOWTO.txt | 2 +- Documentation/admin-guide/README.rst | 6 - Documentation/doc-guide/docbook.rst | 90 ---------- Documentation/doc-guide/index.rst | 1 - Documentation/doc-guide/sphinx.rst | 5 - Documentation/fb/api.txt | 4 +- Documentation/gpu/todo.rst | 2 +- Documentation/kernel-doc-nano-HOWTO.txt | 65 ++----- Documentation/lsm.txt | 201 +++++++++++++++++++++ Documentation/process/changes.rst | 26 +-- Documentation/process/howto.rst | 8 - Documentation/process/kernel-docs.rst | 34 +--- Documentation/sphinx/tmplcvt | 28 --- Documentation/translations/ja_JP/howto.rst | 7 - Documentation/translations/ko_KR/howto.rst | 7 - MAINTAINERS | 5 +- Makefile | 11 +- arch/ia64/include/asm/io.h | 2 +- arch/ia64/sn/kernel/iomv.c | 2 +- drivers/ata/acard-ahci.c | 2 +- drivers/ata/ahci.c | 2 +- drivers/ata/ahci.h | 2 +- drivers/ata/ata_piix.c | 2 +- drivers/ata/libahci.c | 2 +- drivers/ata/libata-core.c | 2 +- drivers/ata/libata-eh.c | 2 +- drivers/ata/libata-scsi.c | 2 +- drivers/ata/libata-sff.c | 2 +- drivers/ata/libata.h | 2 +- drivers/ata/pata_pdc2027x.c | 2 +- drivers/ata/pdc_adma.c | 2 +- drivers/ata/sata_nv.c | 2 +- drivers/ata/sata_promise.c | 2 +- drivers/ata/sata_promise.h | 2 +- drivers/ata/sata_qstor.c | 2 +- drivers/ata/sata_sil.c | 2 +- drivers/ata/sata_sis.c | 2 +- drivers/ata/sata_svw.c | 2 +- drivers/ata/sata_sx4.c | 2 +- drivers/ata/sata_uli.c | 2 +- drivers/ata/sata_via.c | 2 +- drivers/ata/sata_vsc.c | 2 +- drivers/scsi/qla1280.c | 2 +- drivers/usb/gadget/Kconfig | 2 +- fs/debugfs/file.c | 2 +- fs/debugfs/inode.c | 2 +- include/linux/ata.h | 2 +- include/linux/debugfs.h | 2 +- include/linux/libata.h | 2 +- include/sound/pcm.h | 2 +- kernel/irq/chip.c | 2 +- kernel/irq/handle.c | 2 +- kernel/irq/irqdesc.c | 2 +- lib/Kconfig.debug | 2 +- lib/Kconfig.kgdb | 2 +- scripts/Makefile | 9 +- scripts/check-lc_ctype.c | 11 -- 64 files changed, 396 insertions(+), 1032 deletions(-) delete mode 100644 Documentation/DocBook/.gitignore delete mode 100644 Documentation/DocBook/Makefile delete mode 100644 Documentation/DocBook/lsm.tmpl delete mode 100644 Documentation/DocBook/stylesheet.xsl delete mode 100644 Documentation/Makefile.sphinx delete mode 100644 Documentation/doc-guide/docbook.rst create mode 100644 Documentation/lsm.txt delete mode 100755 Documentation/sphinx/tmplcvt delete mode 100644 scripts/check-lc_ctype.c -- 2.9.3

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[alsa-devel] future of sounds/oss
by Christoph Hellwig 16 May '17

16 May '17

Ho Jaroslav, hi Takashi, do you know who still uses the sound/oss drivers and why? I've recently been looking into getting rid of set_fs for using copy_{from,to}_user and friends on kernel pointers, and the sound code is a big abuser, both ALSA and the legacy OSS code. But looking at the OSS code it's pretty grotty, and also appears to be pretty much unmaintained except for global cleanups. Is there any chance we could look into getting rid of it over the next few merge windows or are there people that rely on it?

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[alsa-devel] [PATCH] ALSA: hda - No loopback on ALC299 codec
by Takashi Iwai 16 May '17

16 May '17

ALC299 has no loopback mixer, but the driver still tries to add a beep control over the mixer NID which leads to the error at accessing it. This patch fixes it by properly declaring mixer_nid=0 for this codec. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=195775 Fixes: 28f1f9b26cee ("ALSA: hda/realtek - Add new codec ID ALC299") Cc: stable(a)vger.kernel.org Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 58df440013c5..9c22ad694534 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6465,8 +6465,11 @@ static int patch_alc269(struct hda_codec *codec) break; case 0x10ec0225: case 0x10ec0295: + spec->codec_variant = ALC269_TYPE_ALC225; + break; case 0x10ec0299: spec->codec_variant = ALC269_TYPE_ALC225; + spec->gen.mixer_nid = 0; /* no loopback on ALC299 */ break; case 0x10ec0234: case 0x10ec0274: -- 2.12.2

1 0

Re: [alsa-devel] Distorted audio from microphone (Logitech C310)
by Greg KH 16 May '17

16 May '17

On Mon, May 15, 2017 at 03:29:09PM +0200, Domker_ wrote: > My webcam Logitech C310 have a distored microphone audio on kernel 4.10.15-1 > and 4.11.0-1. (on 4.9.27-1 and the older ones is fine). > The distortion effect is as if I spoke very, very fast (like a damn Mickey > Mouse) > > I checked and the bug occurs on both computers I have. (with the same > camera/different sound cards, hardware and different Linux distributions) > The bug always occurs when running a system with a kernel newer than 4.9.x. > > The camera has two internal microphones. > Secound kernel 4.11.0-1 (x86_64) compiled from kernel.org! > > (tested under Linux Arch and Linux Manjaro) Hm, maybe the Linux sound developers can help out here (added to cc:)

2 1

[alsa-devel] [PATCH 0/2] ASoC: sunxi: Add ADC support for A33
by Mylène Josserand 16 May '17

16 May '17

Hello everyone, This is a first version of a serie to add ADC support for Sun8i-A33. Based on asoc/for-next branch, last commit: 20d5c84bef067b7e804a163e2abca16c47125bad. The first patch adds the support of ADC and microphones in the digital part (driver sun8i-codec). The second one adds the route between analog (sun8i-codec-analog) and digital (sun8i-codec) parts to have a functional ADC route on Sun8i-A33 device tree. Tested on Allwinner R16 Parrot board with microphone 1 (external) and microphone 2 (headset). I am not sure about the bias handling so if you have any comments on it, do not hesitate. Thank you in advance! Best regards, Mylène Josserand (2): ASoC: sun8i-codec: Add ADC support for a33 ARM: dts: sun8i: Add ADC routing arch/arm/boot/dts/sun8i-a33.dtsi | 10 ++++- sound/soc/sunxi/sun8i-codec.c | 80 +++++++++++++++++++++++++++++++++++++++- 2 files changed, 87 insertions(+), 3 deletions(-) -- 2.11.0

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[alsa-devel] [PATCH 1/2] ASoC: add es8316 codec driver
by Daniel Drake 15 May '17

15 May '17

Add a codec driver for the Everest ES8316, based on code provided by David Yang from Everest Semi. I limited the functionality to items where the vendor code was clear, and things that can be tested on the Weibu F3C (Intel Cherry Trail). As a result the initial implementation only supports running in slave mode at single speed (up to 48kHz sample rate) using I2S. HPD is not supported. Signed-off-by: Daniel Drake <drake(a)endlessm.com> --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/es8316.c | 639 ++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/es8316.h | 122 +++++++++ 4 files changed, 767 insertions(+) create mode 100644 sound/soc/codecs/es8316.c create mode 100644 sound/soc/codecs/es8316.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 883ed4c8a551..c6286e5ba511 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -72,6 +72,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_DA9055 if I2C select SND_SOC_DIO2125 select SND_SOC_DMIC + select SND_SOC_ES8316 if I2C select SND_SOC_ES8328_SPI if SPI_MASTER select SND_SOC_ES8328_I2C if I2C select SND_SOC_ES7134 @@ -543,6 +544,9 @@ config SND_SOC_HDMI_CODEC config SND_SOC_ES7134 tristate "Everest Semi ES7134 CODEC" +config SND_SOC_ES8316 + tristate "Everest Semi ES8316 CODEC" + config SND_SOC_ES8328 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 28a63fdaf982..e878306ce46e 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -65,6 +65,7 @@ snd-soc-da732x-objs := da732x.o snd-soc-da9055-objs := da9055.o snd-soc-dmic-objs := dmic.o snd-soc-es7134-objs := es7134.o +snd-soc-es8316-objs := es8316.o snd-soc-es8328-objs := es8328.o snd-soc-es8328-i2c-objs := es8328-i2c.o snd-soc-es8328-spi-objs := es8328-spi.o @@ -300,6 +301,7 @@ obj-$(CONFIG_SND_SOC_DA732X) += snd-soc-da732x.o obj-$(CONFIG_SND_SOC_DA9055) += snd-soc-da9055.o obj-$(CONFIG_SND_SOC_DMIC) += snd-soc-dmic.o obj-$(CONFIG_SND_SOC_ES7134) += snd-soc-es7134.o +obj-$(CONFIG_SND_SOC_ES8316) += snd-soc-es8316.o obj-$(CONFIG_SND_SOC_ES8328) += snd-soc-es8328.o obj-$(CONFIG_SND_SOC_ES8328_I2C)+= snd-soc-es8328-i2c.o obj-$(CONFIG_SND_SOC_ES8328_SPI)+= snd-soc-es8328-spi.o diff --git a/sound/soc/codecs/es8316.c b/sound/soc/codecs/es8316.c new file mode 100644 index 000000000000..aea0b981891b --- /dev/null +++ b/sound/soc/codecs/es8316.c @@ -0,0 +1,639 @@ +/* + * es8316.c -- es8316 ALSA SoC audio driver + * Copyright Everest Semiconductor Co.,Ltd + * + * Authors: David Yang <yangxiaohua(a)everest-semi.com>, + * Daniel Drake <drake(a)endlessm.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/acpi.h> +#include <linux/delay.h> +#include <linux/i2c.h> +#include <linux/mod_devicetable.h> +#include <linux/regmap.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/tlv.h> +#include "es8316.h" + +/* In slave mode at single speed, the codec is documented as accepting 5 + * MCLK/LRCK ratios, but we also add ratio 400, which is commonly used on + * Intel Cherry Trail platforms (19.2MHz MCLK, 48kHz LRCK). + */ +#define NR_SUPPORTED_MCLK_LRCK_RATIOS 6 +static const unsigned int supported_mclk_lrck_ratios[] = { + 256, 384, 400, 512, 768, 1024 +}; + +struct es8316_priv { + unsigned int sysclk; + unsigned int allowed_rates[NR_SUPPORTED_MCLK_LRCK_RATIOS]; + struct snd_pcm_hw_constraint_list sysclk_constraints; +}; + +/* + * ES8316 controls + */ +static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -9600, 50, 1); +static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -9600, 50, 1); +static const DECLARE_TLV_DB_SCALE(hpmixer_gain_tlv, -1200, 150, 0); +static const DECLARE_TLV_DB_SCALE(mic_bst_tlv, 0, 1200, 0); + +static unsigned int linin_pga_tlv[] = { + TLV_DB_RANGE_HEAD(12), + 0, 0, TLV_DB_SCALE_ITEM(0, 0, 0), + 1, 1, TLV_DB_SCALE_ITEM(300, 0, 0), + 2, 2, TLV_DB_SCALE_ITEM(600, 0, 0), + 3, 3, TLV_DB_SCALE_ITEM(900, 0, 0), + 4, 4, TLV_DB_SCALE_ITEM(1200, 0, 0), + 5, 5, TLV_DB_SCALE_ITEM(1500, 0, 0), + 6, 6, TLV_DB_SCALE_ITEM(1800, 0, 0), + 7, 7, TLV_DB_SCALE_ITEM(2100, 0, 0), + 8, 8, TLV_DB_SCALE_ITEM(2400, 0, 0), +}; +static unsigned int hpout_vol_tlv[] = { + TLV_DB_RANGE_HEAD(1), + 0, 3, TLV_DB_SCALE_ITEM(-4800, 1200, 0), +}; +static const char * const alc_func_txt[] = { "Off", "On" }; +static const struct soc_enum alc_func = + SOC_ENUM_SINGLE(ES8316_ADC_ALC1, 6, 2, alc_func_txt); + +static const char * const ng_type_txt[] = + { "Constant PGA Gain", "Mute ADC Output"}; +static const struct soc_enum ng_type = + SOC_ENUM_SINGLE(ES8316_ADC_ALC6, 6, 2, ng_type_txt); + +static const char * const adcpol_txt[] = { "Normal", "Invert" }; +static const struct soc_enum adcpol = + SOC_ENUM_SINGLE(ES8316_ADC_MUTE, 1, 2, adcpol_txt); +static const char *const dacpol_txt[] = + { "Normal", "R Invert", "L Invert", "L + R Invert" }; +static const struct soc_enum dacpol = + SOC_ENUM_SINGLE(ES8316_DAC_SET1, 0, 4, dacpol_txt); + +static const struct snd_kcontrol_new es8316_snd_controls[] = { + SOC_DOUBLE_TLV("HP Playback Volume", ES8316_CPHP_ICAL_VOL, + 4, 0, 0, 1, hpout_vol_tlv), + SOC_DOUBLE_TLV("HPMixer Gain", ES8316_HPMIX_VOL, + 0, 4, 7, 0, hpmixer_gain_tlv), + + SOC_DOUBLE_R_TLV("DAC Playback Volume", ES8316_DAC_VOLL, + ES8316_DAC_VOLR, 0, 0xC0, 1, dac_vol_tlv), + + SOC_SINGLE("Enable DAC Soft Ramp", ES8316_DAC_SET1, 4, 1, 1), + SOC_SINGLE("DAC Soft Ramp Rate", ES8316_DAC_SET1, 2, 4, 0), + + SOC_ENUM("Playback Polarity", dacpol), + SOC_SINGLE("DAC Notch Filter", ES8316_DAC_SET2, 6, 1, 0), + SOC_SINGLE("DAC Double Fs Mode", ES8316_DAC_SET2, 7, 1, 0), + SOC_SINGLE("DAC Volume Control-LeR", ES8316_DAC_SET2, 2, 1, 0), + SOC_SINGLE("DAC Stereo Enhancement", ES8316_DAC_SET3, 0, 7, 0), + + /* +20dB D2SE PGA Control */ + SOC_SINGLE_TLV("MIC Boost", ES8316_ADC_D2SEPGA, + 0, 1, 0, mic_bst_tlv), + + /* 0-+24dB Lineinput PGA Control */ + SOC_SINGLE_TLV("Input PGA", ES8316_ADC_PGAGAIN, + 4, 8, 0, linin_pga_tlv), + + SOC_SINGLE_TLV("ADC Capture Volume", ES8316_ADC_VOLUME, + 0, 0xc0, 1, adc_vol_tlv), + SOC_SINGLE("ADC Soft Ramp", ES8316_ADC_MUTE, 4, 1, 0), + SOC_ENUM("Capture Polarity", adcpol), + SOC_SINGLE("ADC Double FS Mode", ES8316_ADC_DMIC, 4, 1, 0), + + SOC_SINGLE("ALC Capture Target Volume", + ES8316_ADC_ALC3, 4, 10, 0), + SOC_SINGLE("ALC Capture Max PGA", ES8316_ADC_ALC1, 0, 28, 0), + SOC_SINGLE("ALC Capture Min PGA", ES8316_ADC_ALC2, 0, 28, 0), + SOC_ENUM("ALC Capture Function", alc_func), + SOC_SINGLE("ALC Capture Hold Time", ES8316_ADC_ALC3, 0, 10, 0), + SOC_SINGLE("ALC Capture Decay Time", ES8316_ADC_ALC4, 4, 10, 0), + SOC_SINGLE("ALC Capture Attack Time", ES8316_ADC_ALC4, 0, 10, 0), + SOC_SINGLE("ALC Capture NG Threshold", ES8316_ADC_ALC6, 0, 31, 0), + SOC_ENUM("ALC Capture NG Type", ng_type), + SOC_SINGLE("ALC Capture NG Switch", ES8316_ADC_ALC6, 5, 1, 0), +}; + +/* Analog Input Mux */ +static const char * const es8316_analog_in_txt[] = { + "lin1-rin1", + "lin2-rin2", + "lin1-rin1 with 20db Boost", + "lin2-rin2 with 20db Boost" +}; +static const unsigned int es8316_analog_in_values[] = { 0, 1, 2, 3 }; +static const struct soc_enum es8316_analog_input_enum = + SOC_VALUE_ENUM_SINGLE(ES8316_ADC_PDN_LINSEL, 4, 3, + ARRAY_SIZE(es8316_analog_in_txt), + es8316_analog_in_txt, + es8316_analog_in_values); +static const struct snd_kcontrol_new es8316_analog_in_mux_controls = + SOC_DAPM_ENUM("Route", es8316_analog_input_enum); + +static const char * const es8316_dmic_txt[] = { + "dmic disable", + "dmic data at high level", + "dmic data at low level", +}; +static const unsigned int es8316_dmic_values[] = { 0, 1, 2 }; +static const struct soc_enum es8316_dmic_src_enum = + SOC_VALUE_ENUM_SINGLE(ES8316_ADC_DMIC, 0, 3, + ARRAY_SIZE(es8316_dmic_txt), + es8316_dmic_txt, + es8316_dmic_values); +static const struct snd_kcontrol_new es8316_dmic_src_controls = + SOC_DAPM_ENUM("Route", es8316_dmic_src_enum); + +/* hp mixer mux */ +static const char * const es8316_hpmux_texts[] = { + "lin1-rin1", + "lin2-rin2", + "lin-rin with Boost", + "lin-rin with Boost and PGA" +}; + +static const unsigned int es8316_hpmux_values[] = { 0, 1, 2, 3 }; + +static SOC_ENUM_SINGLE_DECL(es8316_left_hpmux_enum, ES8316_HPMIX_SEL, + 4, es8316_hpmux_texts); + +static const struct snd_kcontrol_new es8316_left_hpmux_controls = + SOC_DAPM_ENUM("Route", es8316_left_hpmux_enum); + +static SOC_ENUM_SINGLE_DECL(es8316_right_hpmux_enum, ES8316_HPMIX_SEL, + 0, es8316_hpmux_texts); + +static const struct snd_kcontrol_new es8316_right_hpmux_controls = + SOC_DAPM_ENUM("Route", es8316_right_hpmux_enum); + +/* headphone Output Mixer */ +static const struct snd_kcontrol_new es8316_out_left_mix[] = { + SOC_DAPM_SINGLE("LLIN Switch", ES8316_HPMIX_SWITCH, 6, 1, 0), + SOC_DAPM_SINGLE("Left DAC Switch", ES8316_HPMIX_SWITCH, 7, 1, 0), +}; +static const struct snd_kcontrol_new es8316_out_right_mix[] = { + SOC_DAPM_SINGLE("RLIN Switch", ES8316_HPMIX_SWITCH, 2, 1, 0), + SOC_DAPM_SINGLE("Right DAC Switch", ES8316_HPMIX_SWITCH, 3, 1, 0), +}; + +/* DAC data source mux */ +static const char * const es8316_dacsrc_texts[] = { + "LDATA TO LDAC, RDATA TO RDAC", + "LDATA TO LDAC, LDATA TO RDAC", + "RDATA TO LDAC, RDATA TO RDAC", + "RDATA TO LDAC, LDATA TO RDAC", +}; + +static const unsigned int es8316_dacsrc_values[] = { 0, 1, 2, 3 }; + +static SOC_ENUM_SINGLE_DECL(es8316_dacsrc_mux_enum, ES8316_DAC_SET1, + 6, es8316_dacsrc_texts); + +static const struct snd_kcontrol_new es8316_dacsrc_mux_controls = + SOC_DAPM_ENUM("Route", es8316_dacsrc_mux_enum); + +static const struct snd_soc_dapm_widget es8316_dapm_widgets[] = { + SND_SOC_DAPM_INPUT("DMIC"), + SND_SOC_DAPM_INPUT("MIC1"), + SND_SOC_DAPM_INPUT("MIC2"), + SND_SOC_DAPM_MICBIAS("micbias", SND_SOC_NOPM, 0, 0), + + /* Input Mux */ + SND_SOC_DAPM_MUX("Differential Mux", SND_SOC_NOPM, 0, 0, + &es8316_analog_in_mux_controls), + + SND_SOC_DAPM_PGA("Line input PGA", ES8316_ADC_PDN_LINSEL, + 7, 1, NULL, 0), + SND_SOC_DAPM_ADC("Mono ADC", NULL, ES8316_ADC_PDN_LINSEL, 6, 1), + SND_SOC_DAPM_MUX("Digital Mic Mux", SND_SOC_NOPM, 0, 0, + &es8316_dmic_src_controls), + + /* Digital Interface */ + SND_SOC_DAPM_AIF_OUT("I2S OUT", "I2S1 Capture", 1, + ES8316_SERDATA_ADC, 6, 1), + SND_SOC_DAPM_AIF_IN("I2S IN", "I2S1 Playback", 0, + SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_MUX("DAC SRC Mux", SND_SOC_NOPM, 0, 0, + &es8316_dacsrc_mux_controls), + + SND_SOC_DAPM_DAC("Right DAC", NULL, SND_SOC_NOPM, 0, 1), + SND_SOC_DAPM_DAC("Left DAC", NULL, SND_SOC_NOPM, 4, 1), + + /* Headphone Output Side */ + SND_SOC_DAPM_MUX("Left Hp mux", SND_SOC_NOPM, 0, 0, + &es8316_left_hpmux_controls), + SND_SOC_DAPM_MUX("Right Hp mux", SND_SOC_NOPM, 0, 0, + &es8316_right_hpmux_controls), + SND_SOC_DAPM_MIXER("Left Hp mixer", SND_SOC_NOPM, + 4, 1, &es8316_out_left_mix[0], + ARRAY_SIZE(es8316_out_left_mix)), + SND_SOC_DAPM_MIXER("Right Hp mixer", SND_SOC_NOPM, + 0, 1, &es8316_out_right_mix[0], + ARRAY_SIZE(es8316_out_right_mix)), + + /* Output charge pump */ + SND_SOC_DAPM_PGA("HPCP L", SND_SOC_NOPM, 6, 0, NULL, 0), + SND_SOC_DAPM_PGA("HPCP R", SND_SOC_NOPM, 2, 0, NULL, 0), + + /* Output Driver */ + SND_SOC_DAPM_PGA("HPVOL L", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("HPVOL R", SND_SOC_NOPM, 0, 0, NULL, 0), + + SND_SOC_DAPM_OUTPUT("HPOL"), + SND_SOC_DAPM_OUTPUT("HPOR"), +}; + +static const struct snd_soc_dapm_route es8316_dapm_routes[] = { + /* Recording */ + {"MIC1", NULL, "micbias"}, + {"MIC2", NULL, "micbias"}, + + {"Differential Mux", "lin1-rin1", "MIC1"}, + {"Differential Mux", "lin2-rin2", "MIC2"}, + {"Line input PGA", NULL, "Differential Mux"}, + + {"Mono ADC", NULL, "Line input PGA"}, + + {"Digital Mic Mux", "dmic disable", "Mono ADC"}, + + {"I2S OUT", NULL, "Digital Mic Mux"}, + + /* Playback */ + {"DAC SRC Mux", "LDATA TO LDAC, RDATA TO RDAC", "I2S IN"}, + + {"Left DAC", NULL, "DAC SRC Mux"}, + {"Right DAC", NULL, "DAC SRC Mux"}, + + {"Left Hp mux", "lin-rin with Boost and PGA", "Line input PGA"}, + {"Right Hp mux", "lin-rin with Boost and PGA", "Line input PGA"}, + + {"Left Hp mixer", "LLIN Switch", "Left Hp mux"}, + {"Left Hp mixer", "Left DAC Switch", "Left DAC"}, + + {"Right Hp mixer", "RLIN Switch", "Right Hp mux"}, + {"Right Hp mixer", "Right DAC Switch", "Right DAC"}, + + {"HPCP L", NULL, "Left Hp mixer"}, + {"HPCP R", NULL, "Right Hp mixer"}, + + {"HPVOL L", NULL, "HPCP L"}, + {"HPVOL R", NULL, "HPCP R"}, + + {"HPOL", NULL, "HPVOL L"}, + {"HPOR", NULL, "HPVOL R"}, +}; + +static int es8316_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 es8316_priv *es8316 = snd_soc_codec_get_drvdata(codec); + int i; + + if (freq == 0) + return -EINVAL; + + /* Limit supported sample rates to ones that can be autodetected + * by the codec running in slave mode. + */ + for (i = 0; i < NR_SUPPORTED_MCLK_LRCK_RATIOS; i++) { + const unsigned int ratio = supported_mclk_lrck_ratios[i]; + + es8316->allowed_rates[i] = freq / ratio; + } + + es8316->sysclk_constraints.list = es8316->allowed_rates; + es8316->sysclk_constraints.count = NR_SUPPORTED_MCLK_LRCK_RATIOS; + es8316->sysclk = freq; + + return 0; +} + +static int es8316_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u8 serdata1 = 0; + u8 serdata2 = 0; + u8 mask; + + if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS) { + dev_err(codec->dev, "Codec driver only supports slave mode\n"); + return -EINVAL; + } + + if ((fmt & SND_SOC_DAIFMT_FORMAT_MASK) != SND_SOC_DAIFMT_I2S) { + dev_err(codec->dev, "Codec driver only supports I2S format\n"); + return -EINVAL; + } + + /* Clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_IF: + serdata1 |= ES8316_SERDATA1_BCLK_INV; + serdata2 |= ES8316_SERDATA2_ADCLRP; + break; + case SND_SOC_DAIFMT_IB_NF: + serdata1 |= ES8316_SERDATA1_BCLK_INV; + break; + case SND_SOC_DAIFMT_NB_IF: + serdata2 |= ES8316_SERDATA2_ADCLRP; + break; + default: + return -EINVAL; + } + + mask = ES8316_SERDATA1_MASTER | ES8316_SERDATA1_BCLK_INV; + snd_soc_update_bits(codec, ES8316_SERDATA1, mask, serdata1); + + mask = ES8316_SERDATA2_FMT_MASK | ES8316_SERDATA2_ADCLRP; + snd_soc_update_bits(codec, ES8316_SERDATA_ADC, mask, serdata2); + snd_soc_update_bits(codec, ES8316_SERDATA_DAC, mask, serdata2); + + return 0; +} + +static int es8316_pcm_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct es8316_priv *es8316 = snd_soc_codec_get_drvdata(codec); + + /* The set of sample rates that can be supported depends on the + * MCLK supplied to the CODEC. + */ + if (es8316->sysclk) + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &es8316->sysclk_constraints); + + return 0; +} + +static int es8316_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->codec; + struct es8316_priv *es8316 = snd_soc_codec_get_drvdata(codec); + u8 wordlen = 0; + + if (!es8316->sysclk) { + dev_err(codec->dev, "No MCLK configured\n"); + return -EINVAL; + } + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + wordlen = ES8316_SERDATA2_LEN_16; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + wordlen = ES8316_SERDATA2_LEN_20; + break; + case SNDRV_PCM_FORMAT_S24_LE: + wordlen = ES8316_SERDATA2_LEN_24; + break; + case SNDRV_PCM_FORMAT_S32_LE: + wordlen = ES8316_SERDATA2_LEN_32; + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, ES8316_SERDATA_DAC, + ES8316_SERDATA2_LEN_MASK, wordlen); + snd_soc_update_bits(codec, ES8316_SERDATA_ADC, + ES8316_SERDATA2_LEN_MASK, wordlen); + return 0; +} + +static int es8316_mute(struct snd_soc_dai *dai, int mute) +{ + snd_soc_update_bits(dai->codec, ES8316_DAC_SET1, 0x20, + mute ? 0x20 : 0); + return 0; +} + +static int es8316_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_ON: + snd_soc_write(codec, ES8316_RESET, 0xC0); + snd_soc_write(codec, ES8316_CPHP_OUTEN, 0x66); + snd_soc_write(codec, ES8316_DAC_PDN, 0x00); + snd_soc_write(codec, ES8316_CPHP_LDOCTL, 0x30); + snd_soc_write(codec, ES8316_CPHP_PDN2, 0x10); + snd_soc_write(codec, ES8316_CPHP_PDN1, 0x03); + snd_soc_write(codec, ES8316_HPMIX_PDN, 0x00); + snd_soc_update_bits(codec, ES8316_ADC_PDN_LINSEL, 0xc0, 0x00); + snd_soc_write(codec, ES8316_SYS_PDN, 0x00); + snd_soc_write(codec, ES8316_SYS_LP1, 0x04); + snd_soc_write(codec, ES8316_SYS_LP2, 0x0c); + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + break; + case SND_SOC_BIAS_OFF: + snd_soc_write(codec, ES8316_CPHP_OUTEN, 0x00); + snd_soc_write(codec, ES8316_DAC_PDN, 0x11); + snd_soc_write(codec, ES8316_CPHP_LDOCTL, 0x03); + snd_soc_write(codec, ES8316_CPHP_PDN2, 0x22); + snd_soc_write(codec, ES8316_CPHP_PDN1, 0x06); + snd_soc_write(codec, ES8316_HPMIX_PDN, 0x33); + snd_soc_update_bits(codec, ES8316_ADC_PDN_LINSEL, 0xc0, 0xc0); + snd_soc_write(codec, ES8316_SYS_PDN, 0x3f); + snd_soc_write(codec, ES8316_SYS_LP1, 0x3f); + snd_soc_write(codec, ES8316_SYS_LP2, 0x1f); + snd_soc_write(codec, ES8316_RESET, 0x00); + break; + } + + return 0; +} + +#define ES8316_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE) + +static struct snd_soc_dai_ops es8316_ops = { + .startup = es8316_pcm_startup, + .hw_params = es8316_pcm_hw_params, + .set_fmt = es8316_set_dai_fmt, + .set_sysclk = es8316_set_dai_sysclk, + .digital_mute = es8316_mute, +}; + +static struct snd_soc_dai_driver es8316_dai = { + .name = "ES8316 HiFi", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = ES8316_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = ES8316_FORMATS, + }, + .ops = &es8316_ops, + .symmetric_rates = 1, +}; + +static int es8316_probe(struct snd_soc_codec *codec) +{ + snd_soc_write(codec, ES8316_RESET, 0x3f); + usleep_range(5000, 5500); + snd_soc_write(codec, ES8316_RESET, 0x00); + snd_soc_write(codec, ES8316_SYS_VMIDSEL, 0xFF); + msleep(30); + + snd_soc_write(codec, ES8316_CLKMGR_CLKSEL, 0x08); + snd_soc_write(codec, ES8316_CLKMGR_ADCOSR, 0x20); + snd_soc_write(codec, ES8316_CLKMGR_ADCDIV1, 0x11); + snd_soc_write(codec, ES8316_CLKMGR_ADCDIV2, 0x00); + snd_soc_write(codec, ES8316_CLKMGR_DACDIV1, 0x11); + snd_soc_write(codec, ES8316_CLKMGR_DACDIV2, 0x00); + snd_soc_write(codec, ES8316_CLKMGR_CPDIV, 0x00); + snd_soc_write(codec, ES8316_SERDATA1, 0x04); + snd_soc_write(codec, ES8316_CLKMGR_CLKSW, 0x7f); + snd_soc_write(codec, ES8316_CAL_TYPE, 0x0F); + snd_soc_write(codec, ES8316_CAL_HPLIV, 0x90); + snd_soc_write(codec, ES8316_CAL_HPRIV, 0x90); + snd_soc_write(codec, ES8316_ADC_VOLUME, 0x00); + snd_soc_write(codec, ES8316_ADC_D2SEPGA, 0x00); + snd_soc_write(codec, ES8316_ADC_DMIC, 0x08); + snd_soc_write(codec, ES8316_DAC_SET2, 0x00); + snd_soc_write(codec, ES8316_DAC_SET3, 0x00); + snd_soc_write(codec, ES8316_DAC_VOLL, 0x02); + snd_soc_write(codec, ES8316_DAC_VOLR, 0x02); + snd_soc_write(codec, ES8316_SERDATA_ADC, 0x00); + snd_soc_write(codec, ES8316_SERDATA_DAC, 0x00); + snd_soc_write(codec, ES8316_SYS_VMIDLOW, 0x11); + snd_soc_write(codec, ES8316_SYS_VSEL, 0xfc); + snd_soc_write(codec, ES8316_SYS_REF, 0x28); + snd_soc_write(codec, ES8316_HPMIX_SEL, 0x00); + snd_soc_write(codec, ES8316_HPMIX_SWITCH, 0x88); + snd_soc_write(codec, ES8316_HPMIX_VOL, 0x88); + snd_soc_write(codec, ES8316_CPHP_ICAL_VOL, 0x00); + snd_soc_write(codec, ES8316_GPIO_SEL, 0x00); + snd_soc_write(codec, ES8316_GPIO_DEBUNCE_INT, 0x02); + snd_soc_write(codec, ES8316_TESTMODE, 0xa0); + snd_soc_write(codec, ES8316_TEST1, 0x00); + snd_soc_write(codec, ES8316_TEST2, 0x04); + snd_soc_write(codec, ES8316_RESET, 0xc0); + msleep(50); + snd_soc_write(codec, ES8316_ADC_PGAGAIN, 0x60); + snd_soc_write(codec, ES8316_ADC_D2SEPGA, 0x01); + + /* ADC DS mode, HPF enable */ + snd_soc_write(codec, ES8316_ADC_DMIC, 0x08); + snd_soc_write(codec, ES8316_ADC_ALC1, 0xcd); + snd_soc_write(codec, ES8316_ADC_ALC2, 0x08); + snd_soc_write(codec, ES8316_ADC_ALC3, 0xa0); + snd_soc_write(codec, ES8316_ADC_ALC4, 0x05); + snd_soc_write(codec, ES8316_ADC_ALC5, 0x06); + snd_soc_write(codec, ES8316_ADC_ALC6, 0x61); + return 0; +} + +static struct snd_soc_codec_driver soc_codec_dev_es8316 = { + .probe = es8316_probe, + .set_bias_level = es8316_set_bias_level, + .idle_bias_off = true, + + .component_driver = { + .controls = es8316_snd_controls, + .num_controls = ARRAY_SIZE(es8316_snd_controls), + .dapm_widgets = es8316_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(es8316_dapm_widgets), + .dapm_routes = es8316_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(es8316_dapm_routes), + }, +}; + +static const struct regmap_config es8316_regmap = { + .reg_bits = 8, + .val_bits = 8, + .max_register = 0x53, + .cache_type = REGCACHE_RBTREE, +}; + +static int es8316_i2c_probe(struct i2c_client *i2c_client, + const struct i2c_device_id *id) +{ + struct es8316_priv *es8316; + struct regmap *regmap; + + es8316 = devm_kzalloc(&i2c_client->dev, sizeof(struct es8316_priv), + GFP_KERNEL); + if (es8316 == NULL) + return -ENOMEM; + + i2c_set_clientdata(i2c_client, es8316); + + regmap = devm_regmap_init_i2c(i2c_client, &es8316_regmap); + if (IS_ERR(regmap)) + return PTR_ERR(regmap); + + return snd_soc_register_codec(&i2c_client->dev, &soc_codec_dev_es8316, + &es8316_dai, 1); +} + +static int es8316_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static const struct i2c_device_id es8316_i2c_id[] = { + {"ESSX8316", 0 }, + {} +}; +MODULE_DEVICE_TABLE(i2c, es8316_i2c_id); + +static const struct of_device_id es8316_of_match[] = { + { .compatible = "everest,es8316", }, + {}, +}; +MODULE_DEVICE_TABLE(of, es8316_of_match); + +static const struct acpi_device_id es8316_acpi_match[] = { + {"ESSX8316", 0}, + {}, +}; +MODULE_DEVICE_TABLE(acpi, es8316_acpi_match); + +static struct i2c_driver es8316_i2c_driver = { + .driver = { + .name = "es8316", + .acpi_match_table = ACPI_PTR(es8316_acpi_match), + .of_match_table = of_match_ptr(es8316_of_match), + }, + .probe = es8316_i2c_probe, + .remove = es8316_i2c_remove, + .id_table = es8316_i2c_id, +}; +module_i2c_driver(es8316_i2c_driver); + +MODULE_DESCRIPTION("Everest Semi ES8316 ALSA SoC Codec Driver"); +MODULE_AUTHOR("David Yang <yangxiaohua(a)everest-semi.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/es8316.h b/sound/soc/codecs/es8316.h new file mode 100644 index 000000000000..702bebc69ca8 --- /dev/null +++ b/sound/soc/codecs/es8316.h @@ -0,0 +1,122 @@ +/* + * Copyright Everest Semiconductor Co.,Ltd + * + * Author: David Yang <yangxiaohua(a)everest-semi.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + */ + +#ifndef _ES8316_H +#define _ES8316_H + +/* + * ES8316 register space + */ + +/* Reset Control */ +#define ES8316_RESET 0x00 + +/* Clock Management */ +#define ES8316_CLKMGR_CLKSW 0x01 +#define ES8316_CLKMGR_CLKSEL 0x02 +#define ES8316_CLKMGR_ADCOSR 0x03 +#define ES8316_CLKMGR_ADCDIV1 0x04 +#define ES8316_CLKMGR_ADCDIV2 0x05 +#define ES8316_CLKMGR_DACDIV1 0x06 +#define ES8316_CLKMGR_DACDIV2 0x07 +#define ES8316_CLKMGR_CPDIV 0x08 + +/* Serial Data Port Control */ +#define ES8316_SERDATA1 0x09 +#define ES8316_SERDATA_ADC 0x0a +#define ES8316_SERDATA_DAC 0x0b + +/* System Control */ +#define ES8316_SYS_VMIDSEL 0x0c +#define ES8316_SYS_PDN 0x0d +#define ES8316_SYS_LP1 0x0e +#define ES8316_SYS_LP2 0x0f +#define ES8316_SYS_VMIDLOW 0x10 +#define ES8316_SYS_VSEL 0x11 +#define ES8316_SYS_REF 0x12 + +/* Headphone Mixer */ +#define ES8316_HPMIX_SEL 0x13 +#define ES8316_HPMIX_SWITCH 0x14 +#define ES8316_HPMIX_PDN 0x15 +#define ES8316_HPMIX_VOL 0x16 + +/* Charge Pump Headphone driver */ +#define ES8316_CPHP_OUTEN 0x17 +#define ES8316_CPHP_ICAL_VOL 0x18 +#define ES8316_CPHP_PDN1 0x19 +#define ES8316_CPHP_PDN2 0x1a +#define ES8316_CPHP_LDOCTL 0x1b + +/* Calibration */ +#define ES8316_CAL_TYPE 0x1c +#define ES8316_CAL_SET 0x1d +#define ES8316_CAL_HPLIV 0x1e +#define ES8316_CAL_HPRIV 0x1f +#define ES8316_CAL_HPLMV 0x20 +#define ES8316_CAL_HPRMV 0x21 + +/* ADC Control */ +#define ES8316_ADC_PDN_LINSEL 0x22 +#define ES8316_ADC_PGAGAIN 0x23 +#define ES8316_ADC_D2SEPGA 0x24 +#define ES8316_ADC_DMIC 0x25 +#define ES8316_ADC_MUTE 0x26 +#define ES8316_ADC_VOLUME 0x27 +#define ES8316_ADC_ALC1 0x29 +#define ES8316_ADC_ALC2 0x2a +#define ES8316_ADC_ALC3 0x2b +#define ES8316_ADC_ALC4 0x2c +#define ES8316_ADC_ALC5 0x2d +#define ES8316_ADC_ALC6 0x2e + +/* DAC Control */ +#define ES8316_DAC_PDN 0x2f +#define ES8316_DAC_SET1 0x30 +#define ES8316_DAC_SET2 0x31 +#define ES8316_DAC_SET3 0x32 +#define ES8316_DAC_VOLL 0x33 +#define ES8316_DAC_VOLR 0x34 + +/* GPIO */ +#define ES8316_GPIO_SEL 0x4d +#define ES8316_GPIO_DEBUNCE_INT 0x4e +#define ES8316_GPIO_FLAG 0x4f + +/* Test mode */ +#define ES8316_TESTMODE 0x50 +#define ES8316_TEST1 0x51 +#define ES8316_TEST2 0x52 +#define ES8316_TEST3 0x53 + +/* + * Field definitions + */ + +/* ES8316_SERDATA1 */ +#define ES8316_SERDATA1_MASTER 0x80 +#define ES8316_SERDATA1_BCLK_INV 0x20 + +/* ES8316_SERDATA_ADC and _DAC */ +#define ES8316_SERDATA2_FMT_MASK 0x3 +#define ES8316_SERDATA2_FMT_I2S 0x00 +#define ES8316_SERDATA2_FMT_LEFTJ 0x01 +#define ES8316_SERDATA2_FMT_RIGHTJ 0x02 +#define ES8316_SERDATA2_FMT_PCM 0x03 +#define ES8316_SERDATA2_ADCLRP 0x20 +#define ES8316_SERDATA2_LEN_MASK 0x1c +#define ES8316_SERDATA2_LEN_24 0x00 +#define ES8316_SERDATA2_LEN_20 0x04 +#define ES8316_SERDATA2_LEN_18 0x08 +#define ES8316_SERDATA2_LEN_16 0x0c +#define ES8316_SERDATA2_LEN_32 0x10 + +#endif -- 2.11.0

3 7

[alsa-devel] [PATCH 00/36] Convert DocBook documents to ReST
by Mauro Carvalho Chehab 15 May '17

15 May '17

This patch series convert the following books from DocBook to ReST: - filesystems - kernel-hacking - kernel-locking - kgdb - libata - networking - rapidio - s390-drivers - scsi - w1 - z8530book It also adjusts some Sphinx-pedantic errors/warnings on some kernel-doc markups. I also added some patches here to add PDF output for all existing ReST books. I did my best to check if what's there is not too outdated, but the best is if the subsystem maintainers could check it. After this series, there are only 4 DocBook remaining conversion: - librs - lsm - mtdnand - sh I'll likely convert those remaining ones during this weekend. - This patch series is based on docs tree (next branch). The full patch series is on this tree: https://git.linuxtv.org//mchehab/experimental.git/log/?h=docbook And the HTML output at: http://www.infradead.org/~mchehab/kernel_docs/ https://mchehab.fedorapeople.org/kernel_docs/ Mauro Carvalho Chehab (36): docs-rst: convert kernel-hacking to ReST kernel-hacking: update document docs-rst: convert kernel-locking to ReST mutex, futex: adjust kernel-doc markups to generate ReST locking.rst: reformat locking table locking.rst: add captions to two tables locking.rst: Update some ReST markups docs-rst: convert kgdb DocBook to ReST kgdb.rst: Adjust ReST markups conf.py: define a color for important markup on PDF output docs-rst: conf.py: sort LaTeX documents in alphabetical order docs-rst: conf.py: remove kernel-documentation from LaTeX docs-rst: add crypto API book to pdf output docs-rst: add dev-tools book to pdf output docs-rst: add sound book to pdf output docs-rst: add userspace API book to pdf output docs-rst: convert filesystems book to ReST docs-rst: filesystems: use c domain references where needed fs: jbd2: make jbd2_journal_start() kernel-doc parseable docs-rst: don't ignore internal functions for jbd2 docs fs: locks: Fix some troubles at kernel-doc comments fs: add a blank lines on some kernel-doc comments fs: eventfd: fix identation on kernel-doc fs: jbd2: escape a string with special chars on a kernel-doc docs-rst: convert libata book to ReST libata.rst: add c function and struct cross-references libata: fix identation on a kernel-doc markup docs-rst: convert s390-drivers DocBook to ReST docs-rst: convert networking book to ReST net: skbuff.h: properly escape a macro name on kernel-doc net: fix some identation issues at kernel-doc markups docs-rst: convert z8530book DocBook to ReST docs-rst: convert scsi DocBook to ReST scsi: fix some kernel-doc markups docs-rst: convert w1 book to ReST docs-rst: convert rapidio book to ReST Documentation/DocBook/Makefile | 11 +- Documentation/DocBook/filesystems.tmpl | 381 ----- Documentation/DocBook/kernel-hacking.tmpl | 1312 ------------------ Documentation/DocBook/kernel-locking.tmpl | 2151 ----------------------------- Documentation/DocBook/kgdb.tmpl | 918 ------------ Documentation/DocBook/libata.tmpl | 1625 ---------------------- Documentation/DocBook/networking.tmpl | 111 -- Documentation/DocBook/rapidio.tmpl | 155 --- Documentation/DocBook/s390-drivers.tmpl | 161 --- Documentation/DocBook/scsi.tmpl | 409 ------ Documentation/DocBook/w1.tmpl | 101 -- Documentation/DocBook/z8530book.tmpl | 371 ----- Documentation/conf.py | 35 +- Documentation/crypto/conf.py | 10 + Documentation/dev-tools/index.rst | 1 + Documentation/dev-tools/kgdb.rst | 907 ++++++++++++ Documentation/driver-api/index.rst | 5 + Documentation/driver-api/libata.rst | 1031 ++++++++++++++ Documentation/driver-api/rapidio.rst | 107 ++ Documentation/driver-api/s390-drivers.rst | 111 ++ Documentation/driver-api/scsi.rst | 344 +++++ Documentation/driver-api/w1.rst | 70 + Documentation/filesystems/conf.py | 10 + Documentation/filesystems/index.rst | 317 +++++ Documentation/index.rst | 3 + Documentation/kernel-hacking/conf.py | 10 + Documentation/kernel-hacking/hacking.rst | 811 +++++++++++ Documentation/kernel-hacking/index.rst | 5 + Documentation/kernel-hacking/locking.rst | 1446 +++++++++++++++++++ Documentation/networking/conf.py | 10 + Documentation/networking/index.rst | 18 + Documentation/networking/kapi.rst | 147 ++ Documentation/networking/z8530book.rst | 256 ++++ Documentation/sound/conf.py | 10 + drivers/ata/libata-scsi.c | 7 +- drivers/net/phy/phy.c | 1 + drivers/scsi/scsi_scan.c | 7 +- drivers/scsi/scsi_transport_fc.c | 18 +- drivers/scsi/scsicam.c | 4 +- fs/eventfd.c | 4 +- fs/fs-writeback.c | 12 +- fs/jbd2/transaction.c | 42 +- fs/locks.c | 18 +- fs/mpage.c | 1 + fs/namei.c | 1 + include/linux/mutex.h | 6 +- include/linux/netdevice.h | 9 +- include/linux/skbuff.h | 2 +- include/net/sock.h | 9 +- kernel/futex.c | 40 +- kernel/locking/mutex.c | 6 +- net/core/datagram.c | 2 +- net/core/sock.c | 7 +- 53 files changed, 5764 insertions(+), 7802 deletions(-) delete mode 100644 Documentation/DocBook/filesystems.tmpl delete mode 100644 Documentation/DocBook/kernel-hacking.tmpl delete mode 100644 Documentation/DocBook/kernel-locking.tmpl delete mode 100644 Documentation/DocBook/kgdb.tmpl delete mode 100644 Documentation/DocBook/libata.tmpl delete mode 100644 Documentation/DocBook/networking.tmpl delete mode 100644 Documentation/DocBook/rapidio.tmpl delete mode 100644 Documentation/DocBook/s390-drivers.tmpl delete mode 100644 Documentation/DocBook/scsi.tmpl delete mode 100644 Documentation/DocBook/w1.tmpl delete mode 100644 Documentation/DocBook/z8530book.tmpl create mode 100644 Documentation/crypto/conf.py create mode 100644 Documentation/dev-tools/kgdb.rst create mode 100644 Documentation/driver-api/libata.rst create mode 100644 Documentation/driver-api/rapidio.rst create mode 100644 Documentation/driver-api/s390-drivers.rst create mode 100644 Documentation/driver-api/scsi.rst create mode 100644 Documentation/driver-api/w1.rst create mode 100644 Documentation/filesystems/conf.py create mode 100644 Documentation/filesystems/index.rst create mode 100644 Documentation/kernel-hacking/conf.py create mode 100644 Documentation/kernel-hacking/hacking.rst create mode 100644 Documentation/kernel-hacking/index.rst create mode 100644 Documentation/kernel-hacking/locking.rst create mode 100644 Documentation/networking/conf.py create mode 100644 Documentation/networking/index.rst create mode 100644 Documentation/networking/kapi.rst create mode 100644 Documentation/networking/z8530book.rst create mode 100644 Documentation/sound/conf.py -- 2.9.3

2 3

[alsa-devel] [PATCH 1/2] ALSA: firewire-tascam: Fix infinite loop in snd_tscm_stream_get_rate()
by Geert Uytterhoeven 15 May '17

15 May '17

Obviously the intention was to put a limit on the maximum number of operations. However, for this to work, the check should be "&& trials++ < 5", not "|| trials++ < 5". Fixes: 35efa5c489de63a9 ("ALSA: firewire-tascam: add streaming functionality") Signed-off-by: Geert Uytterhoeven <geert(a)linux-m68k.org> --- Compile-tested only. Triggered by a false-positive warning from gcc-4.1.2: warning: ‘err’ may be used uninitialized in this function --- sound/firewire/tascam/tascam-stream.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/firewire/tascam/tascam-stream.c b/sound/firewire/tascam/tascam-stream.c index f1657a4e0621ef49..e433b92ac6904db5 100644 --- a/sound/firewire/tascam/tascam-stream.c +++ b/sound/firewire/tascam/tascam-stream.c @@ -84,7 +84,7 @@ int snd_tscm_stream_get_rate(struct snd_tscm *tscm, unsigned int *rate) unsigned int trials = 0; int err; - while (data == 0x0 || trials++ < 5) { + while (data == 0x0 && trials++ < 5) { err = get_clock(tscm, &data); if (err < 0) return err; -- 2.7.4

2 4

[alsa-devel] [PATCH] ASoC: codec: wm9860: avoid maybe-uninitialized warning
by Arnd Bergmann 15 May '17

15 May '17

The new PLL configuration code triggers a harmless warning: sound/soc/codecs/wm8960.c: In function 'wm8960_configure_clocking': sound/soc/codecs/wm8960.c:735:3: error: 'best_freq_out' may be used uninitialized in this function [-Werror=maybe-uninitialized] wm8960_set_pll(codec, freq_in, best_freq_out); ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ sound/soc/codecs/wm8960.c:699:12: note: 'best_freq_out' was declared here I think the warning was introduced by Daniel's bugfix. I've come up with a way to simplify the code in a way that is more readable to both humans and to gcc, which gets us rid of the warning. Fixes: 84fdc00d519f ("ASoC: codec: wm9860: Refactor PLL out freq search") Fixes: 303e8954af8d ("ASoC: codec: wm8960: Stop when a matching PLL freq is found") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/codecs/wm8960.c | 25 ++++++++----------------- 1 file changed, 8 insertions(+), 17 deletions(-) diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index ace69da97cb8..ba0f80504c6b 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -696,14 +696,12 @@ int wm8960_configure_pll(struct snd_soc_codec *codec, int freq_in, { struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); int sysclk, bclk, lrclk, freq_out; - int diff, best_freq_out; + int diff; int i, j, k; bclk = wm8960->bclk; lrclk = wm8960->lrclk; - *bclk_idx = -1; - for (i = 0; i < ARRAY_SIZE(sysclk_divs); ++i) { if (sysclk_divs[i] == -1) continue; @@ -720,22 +718,14 @@ int wm8960_configure_pll(struct snd_soc_codec *codec, int freq_in, *sysclk_idx = i; *dac_idx = j; *bclk_idx = k; - best_freq_out = freq_out; - break; + return freq_out; } } - if (k != ARRAY_SIZE(bclk_divs)) - break; } - if (j != ARRAY_SIZE(dac_divs)) - break; } - - if (*bclk_idx != -1) - wm8960_set_pll(codec, freq_in, best_freq_out); - - return *bclk_idx; + return -EINVAL; } + static int wm8960_configure_clocking(struct snd_soc_codec *codec) { struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); @@ -783,11 +773,12 @@ static int wm8960_configure_clocking(struct snd_soc_codec *codec) } } - ret = wm8960_configure_pll(codec, freq_in, &i, &j, &k); - if (ret < 0) { + freq_out = wm8960_configure_pll(codec, freq_in, &i, &j, &k); + if (freq_out < 0) { dev_err(codec->dev, "failed to configure clock via PLL\n"); - return -EINVAL; + return freq_out; } + wm8960_set_pll(codec, freq_in, freq_out); configure_clock: /* configure sysclk clock */ -- 2.9.0

4 7

[alsa-devel] HDMI+SND_JACK_AVOUT
by Pierre-Louis Bossart 15 May '17

15 May '17

Hi, Does anyone know the reason why the Intel/HDaudio HDMI layers use SND_JACK_AVOUT as a jack type? This value is composite and ends up having the side effect of sending 2 independent input events to userspace (SW_LINEOUT_INSERT and SW_VIDEOOUT_INSERT) and confusing the Android logic which deals with wired accessories. Shouldn't we use just SND_JACK_VIDEOOUT instead? What is the reason or rationale behind the use of LINEOUT for HDMI/DP jack events? Thanks -Pierre http://elixir.free-electrons.com/linux/latest/ident/SND_JACK_AVOUT

2 1

[alsa-devel] [PATCH] ASoC: utils: Add DT support
by Vishal Thanki 15 May '17

15 May '17

Add device tree support for the dummy codec driver. Signed-off-by: Vishal Thanki <vishalthanki(a)gmail.com> --- sound/soc/soc-utils.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/soc/soc-utils.c b/sound/soc/soc-utils.c index 644d9a9..c76b938 100644 --- a/sound/soc/soc-utils.c +++ b/sound/soc/soc-utils.c @@ -19,6 +19,8 @@ #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> +#include <linux/module.h> +#include <linux/of.h> int snd_soc_calc_frame_size(int sample_size, int channels, int tdm_slots) { @@ -359,9 +361,19 @@ static int snd_soc_dummy_remove(struct platform_device *pdev) return 0; } +#ifdef CONFIG_OF +static const struct of_device_id dummy_dt_ids[] = { + { .compatible = "linux,snd-soc-dummy", }, + { } +}; +MODULE_DEVICE_TABLE(of, dummy_dt_ids); +#endif + + static struct platform_driver soc_dummy_driver = { .driver = { .name = "snd-soc-dummy", + .of_match_table = of_match_ptr(dummy_dt_ids), }, .probe = snd_soc_dummy_probe, .remove = snd_soc_dummy_remove, -- 2.4.11

2 3

[alsa-devel] [PATCH 1/3] ALSA: opl4: Use IS_REACHABLE()
by Takashi Iwai 15 May '17

15 May '17

Rewrite the complex ifdef condition with IS_REACHABLE(). The ifdef in opl4_local.h was without defined(MODLE) check, but this is likely the oversight. Use IS_REACHABLE() here as well. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/drivers/opl4/opl4_lib.c | 4 ++-- sound/drivers/opl4/opl4_local.h | 2 +- 2 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/drivers/opl4/opl4_lib.c b/sound/drivers/opl4/opl4_lib.c index 89c7aa04b3bc..240656e54400 100644 --- a/sound/drivers/opl4/opl4_lib.c +++ b/sound/drivers/opl4/opl4_lib.c @@ -153,7 +153,7 @@ static int snd_opl4_detect(struct snd_opl4 *opl4) return 0; } -#if defined(CONFIG_SND_SEQUENCER) || (defined(MODULE) && defined(CONFIG_SND_SEQUENCER_MODULE)) +#if IS_REACHABLE(CONFIG_SND_SEQUENCER) static void snd_opl4_seq_dev_free(struct snd_seq_device *seq_dev) { struct snd_opl4 *opl4 = seq_dev->private_data; @@ -249,7 +249,7 @@ int snd_opl4_create(struct snd_card *card, snd_opl4_create_mixer(opl4); snd_opl4_create_proc(opl4); -#if defined(CONFIG_SND_SEQUENCER) || (defined(MODULE) && defined(CONFIG_SND_SEQUENCER_MODULE)) +#if IS_REACHABLE(CONFIG_SND_SEQUENCER) opl4->seq_client = -1; if (opl4->hardware < OPL3_HW_OPL4_ML) snd_opl4_create_seq_dev(opl4, seq_device); diff --git a/sound/drivers/opl4/opl4_local.h b/sound/drivers/opl4/opl4_local.h index 9a41bdebce6b..d5bac93f8245 100644 --- a/sound/drivers/opl4/opl4_local.h +++ b/sound/drivers/opl4/opl4_local.h @@ -184,7 +184,7 @@ struct snd_opl4 { #endif struct mutex access_mutex; -#if defined(CONFIG_SND_SEQUENCER) || defined(CONFIG_SND_SEQUENCER_MODULE) +#if IS_REACHABLE(CONFIG_SND_SEQUENCER) int used; int seq_dev_num; -- 2.12.2

2 3

[alsa-devel] [PATCH v2] ASoC: rsnd: tidyup Sampling rate convert example on Document
by Kuninori Morimoto 15 May '17

15 May '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> "renesas,rsrc-card" is exchanged to "simple-scu-audio-card". Let's update Document Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- v1 -> v2 - simple-scu-card -> simple-scu-audio-card on git log .../devicetree/bindings/sound/renesas,rsnd.txt | 30 +++++++++++----------- 1 file changed, 15 insertions(+), 15 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt index 15a7316..3332910 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt @@ -83,11 +83,11 @@ SRC can convert [xx]Hz to [yy]Hz. Then, it has below 2 modes ** Asynchronous mode ------------------ -You need to use "renesas,rsrc-card" sound card for it. +You need to use "simple-scu-audio-card" sound card for it. example) sound { - compatible = "renesas,rsrc-card"; + compatible = "simple-scu-audio-card"; ... /* * SRC Asynchronous mode setting @@ -97,12 +97,12 @@ example) * Inputed 48kHz data will be converted to * system specified Hz */ - convert-rate = <48000>; + simple-audio-card,convert-rate = <48000>; ... - cpu { + simple-audio-card,cpu { sound-dai = <&rcar_sound>; }; - codec { + simple-audio-card,codec { ... }; }; @@ -141,23 +141,23 @@ For more detail information, see below ${LINUX}/sound/soc/sh/rcar/ctu.c - comment of header -You need to use "renesas,rsrc-card" sound card for it. +You need to use "simple-scu-audio-card" sound card for it. example) sound { - compatible = "renesas,rsrc-card"; + compatible = "simple-scu-audio-card"; ... /* * CTU setting * All input data will be converted to 2ch * as output data */ - convert-channels = <2>; + simple-audio-card,convert-channels = <2>; ... - cpu { + simple-audio-card,cpu { sound-dai = <&rcar_sound>; }; - codec { + simple-audio-card,codec { ... }; }; @@ -190,22 +190,22 @@ and these sounds will be merged by MIX. aplay -D plughw:0,0 xxxx.wav & aplay -D plughw:0,1 yyyy.wav -You need to use "renesas,rsrc-card" sound card for it. +You need to use "simple-scu-audio-card" sound card for it. Ex) [MEM] -> [SRC1] -> [CTU02] -+-> [MIX0] -> [DVC0] -> [SSI0] | [MEM] -> [SRC2] -> [CTU03] -+ sound { - compatible = "renesas,rsrc-card"; + compatible = "simple-scu-audio-card"; ... - cpu@0 { + simple-audio-card,cpu@0 { sound-dai = <&rcar_sound 0>; }; - cpu@1 { + simple-audio-card,cpu@1 { sound-dai = <&rcar_sound 1>; }; - codec { + simple-audio-card,codec { ... }; }; -- 1.9.1

2 1

[alsa-devel] [PATCH 1/2] ALSA: Use IS_ENABLED() in common headers
by Takashi Iwai 15 May '17

15 May '17

Simplify the ifdef conditions with IS_ENABLED() macro in the common sound headers. No functional changes. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/sound/core.h | 6 +++--- include/sound/mixer_oss.h | 2 +- include/sound/opl3.h | 4 ++-- include/sound/pcm.h | 10 +++++----- include/sound/rawmidi.h | 4 ++-- 5 files changed, 13 insertions(+), 13 deletions(-) diff --git a/include/sound/core.h b/include/sound/core.h index f7d8c10c4c45..55385588eefa 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -142,7 +142,7 @@ struct snd_card { wait_queue_head_t power_sleep; #endif -#if defined(CONFIG_SND_MIXER_OSS) || defined(CONFIG_SND_MIXER_OSS_MODULE) +#if IS_ENABLED(CONFIG_SND_MIXER_OSS) struct snd_mixer_oss *mixer_oss; int mixer_oss_change_count; #endif @@ -243,7 +243,7 @@ int copy_from_user_toio(volatile void __iomem *dst, const void __user *src, size extern struct snd_card *snd_cards[SNDRV_CARDS]; int snd_card_locked(int card); -#if defined(CONFIG_SND_MIXER_OSS) || defined(CONFIG_SND_MIXER_OSS_MODULE) +#if IS_ENABLED(CONFIG_SND_MIXER_OSS) #define SND_MIXER_OSS_NOTIFY_REGISTER 0 #define SND_MIXER_OSS_NOTIFY_DISCONNECT 1 #define SND_MIXER_OSS_NOTIFY_FREE 2 @@ -394,7 +394,7 @@ static inline void snd_printdd(const char *format, ...) {} #define SNDRV_OSS_VERSION ((3<<16)|(8<<8)|(1<<4)|(0)) /* 3.8.1a */ /* for easier backward-porting */ -#if defined(CONFIG_GAMEPORT) || defined(CONFIG_GAMEPORT_MODULE) +#if IS_ENABLED(CONFIG_GAMEPORT) #define gameport_set_dev_parent(gp,xdev) ((gp)->dev.parent = (xdev)) #define gameport_set_port_data(gp,r) ((gp)->port_data = (r)) #define gameport_get_port_data(gp) (gp)->port_data diff --git a/include/sound/mixer_oss.h b/include/sound/mixer_oss.h index 13cb0b430a1b..930da10fb65b 100644 --- a/include/sound/mixer_oss.h +++ b/include/sound/mixer_oss.h @@ -22,7 +22,7 @@ * */ -#if defined(CONFIG_SND_MIXER_OSS) || defined(CONFIG_SND_MIXER_OSS_MODULE) +#if IS_ENABLED(CONFIG_SND_MIXER_OSS) #define SNDRV_OSS_MAX_MIXERS 32 diff --git a/include/sound/opl3.h b/include/sound/opl3.h index 6ba670707831..5a87641c6ae4 100644 --- a/include/sound/opl3.h +++ b/include/sound/opl3.h @@ -321,7 +321,7 @@ struct snd_opl3 { unsigned char fm_mode; /* OPL mode, see SNDRV_DM_FM_MODE_XXX */ unsigned char rhythm; /* percussion mode flag */ unsigned char max_voices; /* max number of voices */ -#if defined(CONFIG_SND_SEQUENCER) || defined(CONFIG_SND_SEQUENCER_MODULE) +#if IS_ENABLED(CONFIG_SND_SEQUENCER) #define SNDRV_OPL3_MODE_SYNTH 0 /* OSS - voices allocated by application */ #define SNDRV_OPL3_MODE_SEQ 1 /* ALSA - driver handles voice allocation */ int synth_mode; /* synth mode */ @@ -374,7 +374,7 @@ int snd_opl3_release(struct snd_hwdep * hw, struct file *file); void snd_opl3_reset(struct snd_opl3 * opl3); -#if defined(CONFIG_SND_SEQUENCER) || defined(CONFIG_SND_SEQUENCER_MODULE) +#if IS_ENABLED(CONFIG_SND_SEQUENCER) long snd_opl3_write(struct snd_hwdep *hw, const char __user *buf, long count, loff_t *offset); int snd_opl3_load_patch(struct snd_opl3 *opl3, diff --git a/include/sound/pcm.h b/include/sound/pcm.h index eb16912d6046..3f084486868f 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -34,7 +34,7 @@ #define snd_pcm_substream_chip(substream) ((substream)->private_data) #define snd_pcm_chip(pcm) ((pcm)->private_data) -#if defined(CONFIG_SND_PCM_OSS) || defined(CONFIG_SND_PCM_OSS_MODULE) +#if IS_ENABLED(CONFIG_SND_PCM_OSS) #include <sound/pcm_oss.h> #endif @@ -418,7 +418,7 @@ struct snd_pcm_runtime { struct snd_pcm_audio_tstamp_report audio_tstamp_report; struct timespec driver_tstamp; -#if defined(CONFIG_SND_PCM_OSS) || defined(CONFIG_SND_PCM_OSS_MODULE) +#if IS_ENABLED(CONFIG_SND_PCM_OSS) /* -- OSS things -- */ struct snd_pcm_oss_runtime oss; #endif @@ -464,7 +464,7 @@ struct snd_pcm_substream { unsigned int f_flags; void (*pcm_release)(struct snd_pcm_substream *); struct pid *pid; -#if defined(CONFIG_SND_PCM_OSS) || defined(CONFIG_SND_PCM_OSS_MODULE) +#if IS_ENABLED(CONFIG_SND_PCM_OSS) /* -- OSS things -- */ struct snd_pcm_oss_substream oss; #endif @@ -494,7 +494,7 @@ struct snd_pcm_str { unsigned int substream_count; unsigned int substream_opened; struct snd_pcm_substream *substream; -#if defined(CONFIG_SND_PCM_OSS) || defined(CONFIG_SND_PCM_OSS_MODULE) +#if IS_ENABLED(CONFIG_SND_PCM_OSS) /* -- OSS things -- */ struct snd_pcm_oss_stream oss; #endif @@ -526,7 +526,7 @@ struct snd_pcm { void (*private_free) (struct snd_pcm *pcm); bool internal; /* pcm is for internal use only */ bool nonatomic; /* whole PCM operations are in non-atomic context */ -#if defined(CONFIG_SND_PCM_OSS) || defined(CONFIG_SND_PCM_OSS_MODULE) +#if IS_ENABLED(CONFIG_SND_PCM_OSS) struct snd_pcm_oss oss; #endif }; diff --git a/include/sound/rawmidi.h b/include/sound/rawmidi.h index 492a3ca7f17b..6665cb29e1a2 100644 --- a/include/sound/rawmidi.h +++ b/include/sound/rawmidi.h @@ -30,7 +30,7 @@ #include <linux/workqueue.h> #include <linux/device.h> -#if defined(CONFIG_SND_SEQUENCER) || defined(CONFIG_SND_SEQUENCER_MODULE) +#if IS_ENABLED(CONFIG_SND_SEQUENCER) #include <sound/seq_device.h> #endif @@ -144,7 +144,7 @@ struct snd_rawmidi { struct snd_info_entry *proc_entry; -#if defined(CONFIG_SND_SEQUENCER) || defined(CONFIG_SND_SEQUENCER_MODULE) +#if IS_ENABLED(CONFIG_SND_SEQUENCER) struct snd_seq_device *seq_dev; #endif }; -- 2.12.2

1 1

[alsa-devel] [PATCH] ALSA: pcm: Build pcm notifier code conditionally
by Takashi Iwai 15 May '17

15 May '17

The PCM notifier code is used only by OSS emulation layer, so we can build it conditionally for reducing the size. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/sound/pcm.h | 15 ++++++++------- sound/core/pcm.c | 33 ++++++++++++++++++++------------- 2 files changed, 28 insertions(+), 20 deletions(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 361749e60799..eb16912d6046 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -531,13 +531,6 @@ struct snd_pcm { #endif }; -struct snd_pcm_notify { - int (*n_register) (struct snd_pcm * pcm); - int (*n_disconnect) (struct snd_pcm * pcm); - int (*n_unregister) (struct snd_pcm * pcm); - struct list_head list; -}; - /* * Registering */ @@ -552,7 +545,15 @@ int snd_pcm_new_internal(struct snd_card *card, const char *id, int device, struct snd_pcm **rpcm); int snd_pcm_new_stream(struct snd_pcm *pcm, int stream, int substream_count); +#if IS_ENABLED(CONFIG_SND_PCM_OSS) +struct snd_pcm_notify { + int (*n_register) (struct snd_pcm * pcm); + int (*n_disconnect) (struct snd_pcm * pcm); + int (*n_unregister) (struct snd_pcm * pcm); + struct list_head list; +}; int snd_pcm_notify(struct snd_pcm_notify *notify, int nfree); +#endif /* * Native I/O diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 8e980aa678d0..24acbfb5c531 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -36,8 +36,10 @@ MODULE_DESCRIPTION("Midlevel PCM code for ALSA."); MODULE_LICENSE("GPL"); static LIST_HEAD(snd_pcm_devices); -static LIST_HEAD(snd_pcm_notify_list); static DEFINE_MUTEX(register_mutex); +#if IS_ENABLED(CONFIG_SND_PCM_OSS) +static LIST_HEAD(snd_pcm_notify_list); +#endif static int snd_pcm_free(struct snd_pcm *pcm); static int snd_pcm_dev_free(struct snd_device *device); @@ -884,16 +886,23 @@ static void snd_pcm_free_stream(struct snd_pcm_str * pstr) put_device(&pstr->dev); } +#if IS_ENABLED(CONFIG_SND_PCM_OSS) +#define pcm_call_notify(pcm, call) \ + do { \ + struct snd_pcm_notify *_notify; \ + list_for_each_entry(_notify, &snd_pcm_notify_list, list) \ + _notify->call(pcm); \ + } while (0) +#else +#define pcm_call_notify(pcm, call) /* NOP */ +#endif + static int snd_pcm_free(struct snd_pcm *pcm) { - struct snd_pcm_notify *notify; - if (!pcm) return 0; - if (!pcm->internal) { - list_for_each_entry(notify, &snd_pcm_notify_list, list) - notify->n_unregister(pcm); - } + if (!pcm->internal) + pcm_call_notify(pcm, n_unregister); if (pcm->private_free) pcm->private_free(pcm); snd_pcm_lib_preallocate_free_for_all(pcm); @@ -1069,7 +1078,6 @@ static int snd_pcm_dev_register(struct snd_device *device) { int cidx, err; struct snd_pcm_substream *substream; - struct snd_pcm_notify *notify; struct snd_pcm *pcm; if (snd_BUG_ON(!device || !device->device_data)) @@ -1107,8 +1115,7 @@ static int snd_pcm_dev_register(struct snd_device *device) snd_pcm_timer_init(substream); } - list_for_each_entry(notify, &snd_pcm_notify_list, list) - notify->n_register(pcm); + pcm_call_notify(pcm, n_register); unlock: mutex_unlock(&register_mutex); @@ -1118,7 +1125,6 @@ static int snd_pcm_dev_register(struct snd_device *device) static int snd_pcm_dev_disconnect(struct snd_device *device) { struct snd_pcm *pcm = device->device_data; - struct snd_pcm_notify *notify; struct snd_pcm_substream *substream; int cidx; @@ -1138,8 +1144,7 @@ static int snd_pcm_dev_disconnect(struct snd_device *device) } } if (!pcm->internal) { - list_for_each_entry(notify, &snd_pcm_notify_list, list) - notify->n_disconnect(pcm); + pcm_call_notify(pcm, n_disconnect); } for (cidx = 0; cidx < 2; cidx++) { if (!pcm->internal) @@ -1151,6 +1156,7 @@ static int snd_pcm_dev_disconnect(struct snd_device *device) return 0; } +#if IS_ENABLED(CONFIG_SND_PCM_OSS) /** * snd_pcm_notify - Add/remove the notify list * @notify: PCM notify list @@ -1183,6 +1189,7 @@ int snd_pcm_notify(struct snd_pcm_notify *notify, int nfree) return 0; } EXPORT_SYMBOL(snd_pcm_notify); +#endif /* CONFIG_SND_PCM_OSS */ #ifdef CONFIG_SND_PROC_FS /* -- 2.12.2

1 0

[alsa-devel] [PATCH 1/3] ALSA: vx222: Use container_of()
by Takashi Iwai 15 May '17

15 May '17

The vx222 driver is using the explicit cast from the parent class pointer, but it'll be broken when the structure field randomization is applied. Use container_of() in a modern manner, instead. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/vx222/vx222.c | 4 ++-- sound/pci/vx222/vx222.h | 2 ++ sound/pci/vx222/vx222_ops.c | 26 +++++++++++++------------- 3 files changed, 17 insertions(+), 15 deletions(-) diff --git a/sound/pci/vx222/vx222.c b/sound/pci/vx222/vx222.c index ecbaf473fc1e..55861849d7df 100644 --- a/sound/pci/vx222/vx222.c +++ b/sound/pci/vx222/vx222.c @@ -116,7 +116,7 @@ static struct snd_vx_hardware vx222_mic_hw = { */ static int snd_vx222_free(struct vx_core *chip) { - struct snd_vx222 *vx = (struct snd_vx222 *)chip; + struct snd_vx222 *vx = to_vx222(chip); if (chip->irq >= 0) free_irq(chip->irq, (void*)chip); @@ -158,7 +158,7 @@ static int snd_vx222_create(struct snd_card *card, struct pci_dev *pci, pci_disable_device(pci); return -ENOMEM; } - vx = (struct snd_vx222 *)chip; + vx = to_vx222(chip); vx->pci = pci; if ((err = pci_request_regions(pci, CARD_NAME)) < 0) { diff --git a/sound/pci/vx222/vx222.h b/sound/pci/vx222/vx222.h index 2f0d78f609a6..cae355c8ed28 100644 --- a/sound/pci/vx222/vx222.h +++ b/sound/pci/vx222/vx222.h @@ -39,6 +39,8 @@ struct snd_vx222 { int mic_level; /* mic level for vx222 mic */ }; +#define to_vx222(x) container_of(x, struct snd_vx222, core) + /* we use a lookup table with 148 values, see vx_mixer.c */ #define VX2_AKM_LEVEL_MAX 0x93 diff --git a/sound/pci/vx222/vx222_ops.c b/sound/pci/vx222/vx222_ops.c index 7df1663ea510..d4298af6d3ee 100644 --- a/sound/pci/vx222/vx222_ops.c +++ b/sound/pci/vx222/vx222_ops.c @@ -86,7 +86,7 @@ static int vx2_reg_index[VX_REG_MAX] = { static inline unsigned long vx2_reg_addr(struct vx_core *_chip, int reg) { - struct snd_vx222 *chip = (struct snd_vx222 *)_chip; + struct snd_vx222 *chip = to_vx222(_chip); return chip->port[vx2_reg_index[reg]] + vx2_reg_offset[reg]; } @@ -159,7 +159,7 @@ static void vx2_outl(struct vx_core *chip, int offset, unsigned int val) static void vx2_reset_dsp(struct vx_core *_chip) { - struct snd_vx222 *chip = (struct snd_vx222 *)_chip; + struct snd_vx222 *chip = to_vx222(_chip); /* set the reset dsp bit to 0 */ vx_outl(chip, CDSP, chip->regCDSP & ~VX_CDSP_DSP_RESET_MASK); @@ -174,7 +174,7 @@ static void vx2_reset_dsp(struct vx_core *_chip) static int vx2_test_xilinx(struct vx_core *_chip) { - struct snd_vx222 *chip = (struct snd_vx222 *)_chip; + struct snd_vx222 *chip = to_vx222(_chip); unsigned int data; dev_dbg(_chip->card->dev, "testing xilinx...\n"); @@ -479,7 +479,7 @@ static int vx2_test_and_ack(struct vx_core *chip) */ static void vx2_validate_irq(struct vx_core *_chip, int enable) { - struct snd_vx222 *chip = (struct snd_vx222 *)_chip; + struct snd_vx222 *chip = to_vx222(_chip); /* Set the interrupt enable bit to 1 in CDSP register */ if (enable) { @@ -730,7 +730,7 @@ static void vx2_old_write_codec_bit(struct vx_core *chip, int codec, unsigned in */ static void vx2_reset_codec(struct vx_core *_chip) { - struct snd_vx222 *chip = (struct snd_vx222 *)_chip; + struct snd_vx222 *chip = to_vx222(_chip); /* Set the reset CODEC bit to 0. */ vx_outl(chip, CDSP, chip->regCDSP &~ VX_CDSP_CODEC_RESET_MASK); @@ -772,7 +772,7 @@ static void vx2_reset_codec(struct vx_core *_chip) */ static void vx2_change_audio_source(struct vx_core *_chip, int src) { - struct snd_vx222 *chip = (struct snd_vx222 *)_chip; + struct snd_vx222 *chip = to_vx222(_chip); switch (src) { case VX_AUDIO_SRC_DIGITAL: @@ -791,7 +791,7 @@ static void vx2_change_audio_source(struct vx_core *_chip, int src) */ static void vx2_set_clock_source(struct vx_core *_chip, int source) { - struct snd_vx222 *chip = (struct snd_vx222 *)_chip; + struct snd_vx222 *chip = to_vx222(_chip); if (source == INTERNAL_QUARTZ) chip->regCFG &= ~VX_CFG_CLOCKIN_SEL_MASK; @@ -805,7 +805,7 @@ static void vx2_set_clock_source(struct vx_core *_chip, int source) */ static void vx2_reset_board(struct vx_core *_chip, int cold_reset) { - struct snd_vx222 *chip = (struct snd_vx222 *)_chip; + struct snd_vx222 *chip = to_vx222(_chip); /* initialize the register values */ chip->regCDSP = VX_CDSP_CODEC_RESET_MASK | VX_CDSP_DSP_RESET_MASK ; @@ -878,7 +878,7 @@ static int vx_input_level_info(struct snd_kcontrol *kcontrol, struct snd_ctl_ele static int vx_input_level_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct vx_core *_chip = snd_kcontrol_chip(kcontrol); - struct snd_vx222 *chip = (struct snd_vx222 *)_chip; + struct snd_vx222 *chip = to_vx222(_chip); mutex_lock(&_chip->mixer_mutex); ucontrol->value.integer.value[0] = chip->input_level[0]; ucontrol->value.integer.value[1] = chip->input_level[1]; @@ -889,7 +889,7 @@ static int vx_input_level_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem static int vx_input_level_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct vx_core *_chip = snd_kcontrol_chip(kcontrol); - struct snd_vx222 *chip = (struct snd_vx222 *)_chip; + struct snd_vx222 *chip = to_vx222(_chip); if (ucontrol->value.integer.value[0] < 0 || ucontrol->value.integer.value[0] > MIC_LEVEL_MAX) return -EINVAL; @@ -922,7 +922,7 @@ static int vx_mic_level_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_ static int vx_mic_level_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct vx_core *_chip = snd_kcontrol_chip(kcontrol); - struct snd_vx222 *chip = (struct snd_vx222 *)_chip; + struct snd_vx222 *chip = to_vx222(_chip); ucontrol->value.integer.value[0] = chip->mic_level; return 0; } @@ -930,7 +930,7 @@ static int vx_mic_level_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_v static int vx_mic_level_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct vx_core *_chip = snd_kcontrol_chip(kcontrol); - struct snd_vx222 *chip = (struct snd_vx222 *)_chip; + struct snd_vx222 *chip = to_vx222(_chip); if (ucontrol->value.integer.value[0] < 0 || ucontrol->value.integer.value[0] > MIC_LEVEL_MAX) return -EINVAL; @@ -973,7 +973,7 @@ static const struct snd_kcontrol_new vx_control_mic_level = { static int vx2_add_mic_controls(struct vx_core *_chip) { - struct snd_vx222 *chip = (struct snd_vx222 *)_chip; + struct snd_vx222 *chip = to_vx222(_chip); int err; if (_chip->type != VX_TYPE_MIC) -- 2.12.2

1 2

[alsa-devel] Headphones not detected when using Dell Dock WD15
by Sergio Villar Senin 15 May '17

15 May '17

Hi, I reported this bug to pulseaudio[1] but after some investigation it looks like the alsa mixer is not working for the dock. I'll be happy to provide any additional info which might be required (attaching alsa-info.sh BTW). Please add me to the Cc as I am not subscribed to this list BR [1] https://bugs.freedesktop.org/show_bug.cgi?id=100711

3 11

[alsa-devel] [PATCH 0/4] ALSA/ASoC: fix after merging ASoC PR for v4.12-rc1
by Takashi Sakamoto 15 May '17

15 May '17

Hi, This patchset is for current 'for-linus' remote branch[1], after merging PR from ALSA SoC part for v4.12-rc1[2]. The first two patch includes bug fixes introduced at former kernel releases, the rest is to suppress sparse. [1] https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git/commit/?h=f… [2] http://mailman.alsa-project.org/pipermail/alsa-devel/2017-May/120308.html Takashi Sakamoto (4): ASoC: codecs: da7213: fix invalid usage of bitwise operation in loop condition ASoC: codecs: msm8916: fix invalid cast to bool type ASoC: hisilicon: localize functions without external linkage ASoC: intel: atom: localize variable without external linkage sound/soc/codecs/da7213.c | 14 +++++--------- sound/soc/codecs/msm8916-wcd-analog.c | 6 +++--- sound/soc/hisilicon/hi6210-i2s.c | 11 ++++++----- sound/soc/intel/atom/sst/sst.c | 2 +- 4 files changed, 15 insertions(+), 18 deletions(-) -- 2.9.3

4 12

[alsa-devel] [PATCH] ASoC: davinci-mcasp: Support for one channel (mono) audio
by Peter Ujfalusi 15 May '17

15 May '17

Mono audio can be achieved by configuring McASP to transmit/receive only during one timeslot. McASP will still going to generate clocks for the other slot(s), but will only use the single slot to transmit/receive. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/davinci/davinci-mcasp.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 5e02c7a15a83..31b4eac87aa2 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -650,7 +650,7 @@ static int davinci_mcasp_ch_constraint(struct davinci_mcasp *mcasp, int stream, if (mcasp->tdm_mask[stream]) slots = hweight32(mcasp->tdm_mask[stream]); - for (i = 2; i <= slots; i++) + for (i = 1; i <= slots; i++) list[count++] = i; for (i = 2; i <= serializers; i++) @@ -1318,7 +1318,7 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream, snd_pcm_hw_constraint_minmax(substream->runtime, SNDRV_PCM_HW_PARAM_CHANNELS, - 2, max_channels); + 0, max_channels); snd_pcm_hw_constraint_list(substream->runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, @@ -1480,13 +1480,13 @@ static struct snd_soc_dai_driver davinci_mcasp_dai[] = { .suspend = davinci_mcasp_suspend, .resume = davinci_mcasp_resume, .playback = { - .channels_min = 2, + .channels_min = 1, .channels_max = 32 * 16, .rates = DAVINCI_MCASP_RATES, .formats = DAVINCI_MCASP_PCM_FMTS, }, .capture = { - .channels_min = 2, + .channels_min = 1, .channels_max = 32 * 16, .rates = DAVINCI_MCASP_RATES, .formats = DAVINCI_MCASP_PCM_FMTS, @@ -1992,12 +1992,12 @@ static int davinci_mcasp_probe(struct platform_device *pdev) */ mcasp->chconstr[SNDRV_PCM_STREAM_PLAYBACK].list = devm_kzalloc(mcasp->dev, sizeof(unsigned int) * - (32 + mcasp->num_serializer - 2), + (32 + mcasp->num_serializer - 1), GFP_KERNEL); mcasp->chconstr[SNDRV_PCM_STREAM_CAPTURE].list = devm_kzalloc(mcasp->dev, sizeof(unsigned int) * - (32 + mcasp->num_serializer - 2), + (32 + mcasp->num_serializer - 1), GFP_KERNEL); if (!mcasp->chconstr[SNDRV_PCM_STREAM_PLAYBACK].list || -- 2.12.2

2 1

[alsa-devel] [PATCH 1/2] ASoC: rt5514: fix gcc-7 warning
by Arnd Bergmann 15 May '17

15 May '17

gcc-7 warns that there is a duplicate 'const' specifier in some variables that are declared using the SOC_ENUM_SINGLE_DECL macro: sound/soc/codecs/rt5514.c:398:14: error: duplicate 'const' declaration specifier [-Werror=duplicate-decl-specifier] static const SOC_ENUM_SINGLE_DECL( sound/soc/codecs/rt5514.c:405:14: error: duplicate 'const' declaration specifier [-Werror=duplicate-decl-specifier] static const SOC_ENUM_SINGLE_DECL( This removes one to fix the warning. Fixes: 4a6180ea7399 ("ASoC: rt5514: add rt5514 codec driver") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/codecs/rt5514.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5514.c b/sound/soc/codecs/rt5514.c index f91221b1ddf0..28ab9e2bb051 100644 --- a/sound/soc/codecs/rt5514.c +++ b/sound/soc/codecs/rt5514.c @@ -395,14 +395,14 @@ static const char * const rt5514_dmic_src[] = { "DMIC1", "DMIC2" }; -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5514_stereo1_dmic_enum, RT5514_DIG_SOURCE_CTRL, RT5514_AD0_DMIC_INPUT_SEL_SFT, rt5514_dmic_src); static const struct snd_kcontrol_new rt5514_sto1_dmic_mux = SOC_DAPM_ENUM("Stereo1 DMIC Source", rt5514_stereo1_dmic_enum); -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5514_stereo2_dmic_enum, RT5514_DIG_SOURCE_CTRL, RT5514_AD1_DMIC_INPUT_SEL_SFT, rt5514_dmic_src); -- 2.9.0

2 3

[alsa-devel] [PATCH v2] ASoC: ak4613: disable asymmetric audio interface format
by Kuninori Morimoto 15 May '17

15 May '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Asymmetric audio interface formats exist in ak4613 by same register settings. Capture Playback 24bit LEFT_J 16bit RIGHT_J 24bit LEFT_J 20bit RIGHT_J 24bit LEFT_J 24bit RIGHT_J 24bit LEFT_J 24bit LEFT_J 24bit I2S 24bit I2S These asymmetric formats makes driver / behavior difficult. It is not HW limitation, but SW limitation. To makes code reading easy, this patch removes asymmetric format support. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- v1 -> v2 - just remove code instead of comment out sound/soc/codecs/ak4613.c | 18 ++++-------------- 1 file changed, 4 insertions(+), 14 deletions(-) diff --git a/sound/soc/codecs/ak4613.c b/sound/soc/codecs/ak4613.c index b2dfdde..557ac16 100644 --- a/sound/soc/codecs/ak4613.c +++ b/sound/soc/codecs/ak4613.c @@ -139,9 +139,7 @@ struct ak4613_priv { #define AUDIO_IFACE(b, fmt) { b, SND_SOC_DAIFMT_##fmt } static const struct ak4613_interface ak4613_iface[] = { /* capture */ /* playback */ - [0] = { AUDIO_IFACE(24, LEFT_J), AUDIO_IFACE(16, RIGHT_J) }, - [1] = { AUDIO_IFACE(24, LEFT_J), AUDIO_IFACE(20, RIGHT_J) }, - [2] = { AUDIO_IFACE(24, LEFT_J), AUDIO_IFACE(24, RIGHT_J) }, + /* [0] - [2] are not supported */ [3] = { AUDIO_IFACE(24, LEFT_J), AUDIO_IFACE(24, LEFT_J) }, [4] = { AUDIO_IFACE(24, I2S), AUDIO_IFACE(24, I2S) }, }; @@ -262,11 +260,9 @@ static int ak4613_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) fmt &= SND_SOC_DAIFMT_FORMAT_MASK; switch (fmt) { - case SND_SOC_DAIFMT_RIGHT_J: case SND_SOC_DAIFMT_LEFT_J: case SND_SOC_DAIFMT_I2S: priv->fmt = fmt; - break; default: return -EINVAL; @@ -286,13 +282,8 @@ static bool ak4613_dai_fmt_matching(const struct ak4613_interface *iface, if (fmts->fmt != fmt) return false; - if (fmt == SND_SOC_DAIFMT_RIGHT_J) { - if (fmts->width != width) - return false; - } else { - if (fmts->width < width) - return false; - } + if (fmts->width != width) + return false; return true; } @@ -420,8 +411,7 @@ static int ak4613_set_bias_level(struct snd_soc_codec *codec, SNDRV_PCM_RATE_96000 |\ SNDRV_PCM_RATE_176400 |\ SNDRV_PCM_RATE_192000) -#define AK4613_PCM_FMTBIT (SNDRV_PCM_FMTBIT_S16_LE |\ - SNDRV_PCM_FMTBIT_S24_LE) +#define AK4613_PCM_FMTBIT (SNDRV_PCM_FMTBIT_S24_LE) static struct snd_soc_dai_driver ak4613_dai = { .name = "ak4613-hifi", -- 1.9.1

2 1

[alsa-devel] [PATCH 1/2] ASoC: Intel: Skylake: Move i915 registration to worker thread
by Subhransu S. Prusty 15 May '17

15 May '17

From: Vinod Koul <vinod.koul(a)intel.com> The i915 component framework expects the caller to be invoking snd_hdac_i915_init() from a thread context. Otherwise it results in lockups on drm side. So move the registering of component interface and probing of codecs on this bus to a worker thread. init_failed in skl structure is not used currently, so renamed to init_done and used to track the initialization done in worker thread. Reported-by: Imre Deak <imre.deak(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> Signed-off-by: Sodhi, VunnyX <vunnyx.sodhi(a)intel.com> Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty(a)intel.com> --- sound/soc/intel/skylake/skl.c | 162 +++++++++++++++++++++++------------------- sound/soc/intel/skylake/skl.h | 4 +- 2 files changed, 91 insertions(+), 75 deletions(-) diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index 6df3b317a476..4c9b5781282b 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -410,7 +410,7 @@ static int skl_free(struct hdac_ext_bus *ebus) struct skl *skl = ebus_to_skl(ebus); struct hdac_bus *bus = ebus_to_hbus(ebus); - skl->init_failed = 1; /* to be sure */ + skl->init_done = 0; /* to be sure */ snd_hdac_ext_stop_streams(ebus); @@ -428,8 +428,10 @@ static int skl_free(struct hdac_ext_bus *ebus) snd_hdac_ext_bus_exit(ebus); + cancel_work_sync(&skl->probe_work); if (IS_ENABLED(CONFIG_SND_SOC_HDAC_HDMI)) snd_hdac_i915_exit(&ebus->bus); + return 0; } @@ -566,6 +568,84 @@ static int skl_codec_create(struct hdac_ext_bus *ebus) .get_response = snd_hdac_bus_get_response, }; +static int skl_i915_init(struct hdac_bus *bus) +{ + int err; + + /* + * The HDMI codec is in GPU so we need to ensure that it is powered + * up and ready for probe + */ + err = snd_hdac_i915_init(bus); + if (err < 0) + return err; + + err = snd_hdac_display_power(bus, true); + if (err < 0) + dev_err(bus->dev, "Cannot turn on display power on i915\n"); + + return err; +} + +static void skl_probe_work(struct work_struct *work) +{ + struct skl *skl = container_of(work, struct skl, probe_work); + struct hdac_ext_bus *ebus = &skl->ebus; + struct hdac_bus *bus = ebus_to_hbus(ebus); + struct hdac_ext_link *hlink = NULL; + int err; + + if (IS_ENABLED(CONFIG_SND_SOC_HDAC_HDMI)) { + err = skl_i915_init(bus); + if (err < 0) + return; + } + + err = skl_init_chip(bus, true); + if (err < 0) { + dev_err(bus->dev, "Init chip failed with err: %d\n", err); + goto out_err; + } + + /* codec detection */ + if (!bus->codec_mask) + dev_info(bus->dev, "no hda codecs found!\n"); + + /* create codec instances */ + err = skl_codec_create(ebus); + if (err < 0) + goto out_err; + + if (IS_ENABLED(CONFIG_SND_SOC_HDAC_HDMI)) { + err = snd_hdac_display_power(bus, false); + if (err < 0) { + dev_err(bus->dev, "Cannot turn off display power on i915\n"); + return; + } + } + + /* register platform dai and controls */ + err = skl_platform_register(bus->dev); + if (err < 0) + return; + /* + * we are done probing so decrement link counts + */ + list_for_each_entry(hlink, &ebus->hlink_list, list) + snd_hdac_ext_bus_link_put(ebus, hlink); + + /* configure PM */ + pm_runtime_put_noidle(bus->dev); + pm_runtime_allow(bus->dev); + skl->init_done = 1; + + return; + +out_err: + if (IS_ENABLED(CONFIG_SND_SOC_HDAC_HDMI)) + err = snd_hdac_display_power(bus, false); +} + /* * constructor */ @@ -593,6 +673,7 @@ static int skl_create(struct pci_dev *pci, snd_hdac_ext_bus_init(ebus, &pci->dev, &bus_core_ops, io_ops); ebus->bus.use_posbuf = 1; skl->pci = pci; + INIT_WORK(&skl->probe_work, skl_probe_work); ebus->bus.bdl_pos_adj = 0; @@ -601,27 +682,6 @@ static int skl_create(struct pci_dev *pci, return 0; } -static int skl_i915_init(struct hdac_bus *bus) -{ - int err; - - /* - * The HDMI codec is in GPU so we need to ensure that it is powered - * up and ready for probe - */ - err = snd_hdac_i915_init(bus); - if (err < 0) - return err; - - err = snd_hdac_display_power(bus, true); - if (err < 0) { - dev_err(bus->dev, "Cannot turn on display power on i915\n"); - return err; - } - - return err; -} - static int skl_first_init(struct hdac_ext_bus *ebus) { struct skl *skl = ebus_to_skl(ebus); @@ -684,20 +744,7 @@ static int skl_first_init(struct hdac_ext_bus *ebus) /* initialize chip */ skl_init_pci(skl); - if (IS_ENABLED(CONFIG_SND_SOC_HDAC_HDMI)) { - err = skl_i915_init(bus); - if (err < 0) - return err; - } - - skl_init_chip(bus, true); - - /* codec detection */ - if (!bus->codec_mask) { - dev_info(bus->dev, "no hda codecs found!\n"); - } - - return 0; + return skl_init_chip(bus, true); } static int skl_probe(struct pci_dev *pci, @@ -706,7 +753,6 @@ static int skl_probe(struct pci_dev *pci, struct skl *skl; struct hdac_ext_bus *ebus = NULL; struct hdac_bus *bus = NULL; - struct hdac_ext_link *hlink = NULL; int err; /* we use ext core ops, so provide NULL for ops here */ @@ -729,7 +775,7 @@ static int skl_probe(struct pci_dev *pci, if (skl->nhlt == NULL) { err = -ENODEV; - goto out_display_power_off; + goto out_free; } err = skl_nhlt_create_sysfs(skl); @@ -760,56 +806,24 @@ static int skl_probe(struct pci_dev *pci, if (bus->mlcap) snd_hdac_ext_bus_get_ml_capabilities(ebus); + snd_hdac_bus_stop_chip(bus); + /* create device for soc dmic */ err = skl_dmic_device_register(skl); if (err < 0) goto out_dsp_free; - /* register platform dai and controls */ - err = skl_platform_register(bus->dev); - if (err < 0) - goto out_dmic_free; - - /* create codec instances */ - err = skl_codec_create(ebus); - if (err < 0) - goto out_unregister; - - if (IS_ENABLED(CONFIG_SND_SOC_HDAC_HDMI)) { - err = snd_hdac_display_power(bus, false); - if (err < 0) { - dev_err(bus->dev, "Cannot turn off display power on i915\n"); - return err; - } - } - - /* - * we are done probling so decrement link counts - */ - list_for_each_entry(hlink, &ebus->hlink_list, list) - snd_hdac_ext_bus_link_put(ebus, hlink); - - /* configure PM */ - pm_runtime_put_noidle(bus->dev); - pm_runtime_allow(bus->dev); + schedule_work(&skl->probe_work); return 0; -out_unregister: - skl_platform_unregister(bus->dev); -out_dmic_free: - skl_dmic_device_unregister(skl); out_dsp_free: skl_free_dsp(skl); out_mach_free: skl_machine_device_unregister(skl); out_nhlt_free: skl_nhlt_free(skl->nhlt); -out_display_power_off: - if (IS_ENABLED(CONFIG_SND_SOC_HDAC_HDMI)) - snd_hdac_display_power(bus, false); out_free: - skl->init_failed = 1; skl_free(ebus); return err; @@ -828,7 +842,7 @@ static void skl_shutdown(struct pci_dev *pci) skl = ebus_to_skl(ebus); - if (skl->init_failed) + if (!skl->init_done) return; snd_hdac_ext_stop_streams(ebus); diff --git a/sound/soc/intel/skylake/skl.h b/sound/soc/intel/skylake/skl.h index 5b3fa4b91691..c1968a06174a 100644 --- a/sound/soc/intel/skylake/skl.h +++ b/sound/soc/intel/skylake/skl.h @@ -67,7 +67,7 @@ struct skl { struct hdac_ext_bus ebus; struct pci_dev *pci; - unsigned int init_failed:1; /* delayed init failed */ + unsigned int init_done:1; /* delayed init status */ struct platform_device *dmic_dev; struct platform_device *i2s_dev; struct snd_soc_platform *platform; @@ -85,6 +85,8 @@ struct skl { const struct firmware *tplg; int supend_active; + + struct work_struct probe_work; }; #define skl_to_ebus(s) (&(s)->ebus) -- 1.9.1

3 5

[alsa-devel] [PATCH v2 1/2] ASoC: Intel: Skylake: Move i915 registration to worker thread
by Subhransu S. Prusty 15 May '17

15 May '17

From: Vinod Koul <vinod.koul(a)intel.com> The i915 component framework expects the caller to be invoking snd_hdac_i915_init() from a thread context. Otherwise it results in lockups on drm side. So move the registering of component interface and probing of codecs on this bus to a worker thread. init_failed in skl structure is not used currently, so renamed to init_done and used to track the initialization done in worker thread. Reported-by: Imre Deak <imre.deak(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> Signed-off-by: Sodhi, VunnyX <vunnyx.sodhi(a)intel.com> Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty(a)intel.com> --- v1->v2 - no change sound/soc/intel/skylake/skl.c | 162 +++++++++++++++++++++++------------------- sound/soc/intel/skylake/skl.h | 4 +- 2 files changed, 91 insertions(+), 75 deletions(-) diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index 6df3b317a476..4c9b5781282b 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -410,7 +410,7 @@ static int skl_free(struct hdac_ext_bus *ebus) struct skl *skl = ebus_to_skl(ebus); struct hdac_bus *bus = ebus_to_hbus(ebus); - skl->init_failed = 1; /* to be sure */ + skl->init_done = 0; /* to be sure */ snd_hdac_ext_stop_streams(ebus); @@ -428,8 +428,10 @@ static int skl_free(struct hdac_ext_bus *ebus) snd_hdac_ext_bus_exit(ebus); + cancel_work_sync(&skl->probe_work); if (IS_ENABLED(CONFIG_SND_SOC_HDAC_HDMI)) snd_hdac_i915_exit(&ebus->bus); + return 0; } @@ -566,6 +568,84 @@ static int skl_codec_create(struct hdac_ext_bus *ebus) .get_response = snd_hdac_bus_get_response, }; +static int skl_i915_init(struct hdac_bus *bus) +{ + int err; + + /* + * The HDMI codec is in GPU so we need to ensure that it is powered + * up and ready for probe + */ + err = snd_hdac_i915_init(bus); + if (err < 0) + return err; + + err = snd_hdac_display_power(bus, true); + if (err < 0) + dev_err(bus->dev, "Cannot turn on display power on i915\n"); + + return err; +} + +static void skl_probe_work(struct work_struct *work) +{ + struct skl *skl = container_of(work, struct skl, probe_work); + struct hdac_ext_bus *ebus = &skl->ebus; + struct hdac_bus *bus = ebus_to_hbus(ebus); + struct hdac_ext_link *hlink = NULL; + int err; + + if (IS_ENABLED(CONFIG_SND_SOC_HDAC_HDMI)) { + err = skl_i915_init(bus); + if (err < 0) + return; + } + + err = skl_init_chip(bus, true); + if (err < 0) { + dev_err(bus->dev, "Init chip failed with err: %d\n", err); + goto out_err; + } + + /* codec detection */ + if (!bus->codec_mask) + dev_info(bus->dev, "no hda codecs found!\n"); + + /* create codec instances */ + err = skl_codec_create(ebus); + if (err < 0) + goto out_err; + + if (IS_ENABLED(CONFIG_SND_SOC_HDAC_HDMI)) { + err = snd_hdac_display_power(bus, false); + if (err < 0) { + dev_err(bus->dev, "Cannot turn off display power on i915\n"); + return; + } + } + + /* register platform dai and controls */ + err = skl_platform_register(bus->dev); + if (err < 0) + return; + /* + * we are done probing so decrement link counts + */ + list_for_each_entry(hlink, &ebus->hlink_list, list) + snd_hdac_ext_bus_link_put(ebus, hlink); + + /* configure PM */ + pm_runtime_put_noidle(bus->dev); + pm_runtime_allow(bus->dev); + skl->init_done = 1; + + return; + +out_err: + if (IS_ENABLED(CONFIG_SND_SOC_HDAC_HDMI)) + err = snd_hdac_display_power(bus, false); +} + /* * constructor */ @@ -593,6 +673,7 @@ static int skl_create(struct pci_dev *pci, snd_hdac_ext_bus_init(ebus, &pci->dev, &bus_core_ops, io_ops); ebus->bus.use_posbuf = 1; skl->pci = pci; + INIT_WORK(&skl->probe_work, skl_probe_work); ebus->bus.bdl_pos_adj = 0; @@ -601,27 +682,6 @@ static int skl_create(struct pci_dev *pci, return 0; } -static int skl_i915_init(struct hdac_bus *bus) -{ - int err; - - /* - * The HDMI codec is in GPU so we need to ensure that it is powered - * up and ready for probe - */ - err = snd_hdac_i915_init(bus); - if (err < 0) - return err; - - err = snd_hdac_display_power(bus, true); - if (err < 0) { - dev_err(bus->dev, "Cannot turn on display power on i915\n"); - return err; - } - - return err; -} - static int skl_first_init(struct hdac_ext_bus *ebus) { struct skl *skl = ebus_to_skl(ebus); @@ -684,20 +744,7 @@ static int skl_first_init(struct hdac_ext_bus *ebus) /* initialize chip */ skl_init_pci(skl); - if (IS_ENABLED(CONFIG_SND_SOC_HDAC_HDMI)) { - err = skl_i915_init(bus); - if (err < 0) - return err; - } - - skl_init_chip(bus, true); - - /* codec detection */ - if (!bus->codec_mask) { - dev_info(bus->dev, "no hda codecs found!\n"); - } - - return 0; + return skl_init_chip(bus, true); } static int skl_probe(struct pci_dev *pci, @@ -706,7 +753,6 @@ static int skl_probe(struct pci_dev *pci, struct skl *skl; struct hdac_ext_bus *ebus = NULL; struct hdac_bus *bus = NULL; - struct hdac_ext_link *hlink = NULL; int err; /* we use ext core ops, so provide NULL for ops here */ @@ -729,7 +775,7 @@ static int skl_probe(struct pci_dev *pci, if (skl->nhlt == NULL) { err = -ENODEV; - goto out_display_power_off; + goto out_free; } err = skl_nhlt_create_sysfs(skl); @@ -760,56 +806,24 @@ static int skl_probe(struct pci_dev *pci, if (bus->mlcap) snd_hdac_ext_bus_get_ml_capabilities(ebus); + snd_hdac_bus_stop_chip(bus); + /* create device for soc dmic */ err = skl_dmic_device_register(skl); if (err < 0) goto out_dsp_free; - /* register platform dai and controls */ - err = skl_platform_register(bus->dev); - if (err < 0) - goto out_dmic_free; - - /* create codec instances */ - err = skl_codec_create(ebus); - if (err < 0) - goto out_unregister; - - if (IS_ENABLED(CONFIG_SND_SOC_HDAC_HDMI)) { - err = snd_hdac_display_power(bus, false); - if (err < 0) { - dev_err(bus->dev, "Cannot turn off display power on i915\n"); - return err; - } - } - - /* - * we are done probling so decrement link counts - */ - list_for_each_entry(hlink, &ebus->hlink_list, list) - snd_hdac_ext_bus_link_put(ebus, hlink); - - /* configure PM */ - pm_runtime_put_noidle(bus->dev); - pm_runtime_allow(bus->dev); + schedule_work(&skl->probe_work); return 0; -out_unregister: - skl_platform_unregister(bus->dev); -out_dmic_free: - skl_dmic_device_unregister(skl); out_dsp_free: skl_free_dsp(skl); out_mach_free: skl_machine_device_unregister(skl); out_nhlt_free: skl_nhlt_free(skl->nhlt); -out_display_power_off: - if (IS_ENABLED(CONFIG_SND_SOC_HDAC_HDMI)) - snd_hdac_display_power(bus, false); out_free: - skl->init_failed = 1; skl_free(ebus); return err; @@ -828,7 +842,7 @@ static void skl_shutdown(struct pci_dev *pci) skl = ebus_to_skl(ebus); - if (skl->init_failed) + if (!skl->init_done) return; snd_hdac_ext_stop_streams(ebus); diff --git a/sound/soc/intel/skylake/skl.h b/sound/soc/intel/skylake/skl.h index 5b3fa4b91691..c1968a06174a 100644 --- a/sound/soc/intel/skylake/skl.h +++ b/sound/soc/intel/skylake/skl.h @@ -67,7 +67,7 @@ struct skl { struct hdac_ext_bus ebus; struct pci_dev *pci; - unsigned int init_failed:1; /* delayed init failed */ + unsigned int init_done:1; /* delayed init status */ struct platform_device *dmic_dev; struct platform_device *i2s_dev; struct snd_soc_platform *platform; @@ -85,6 +85,8 @@ struct skl { const struct firmware *tplg; int supend_active; + + struct work_struct probe_work; }; #define skl_to_ebus(s) (&(s)->ebus) -- 1.9.1

3 3

[alsa-devel] [PATCH] rt286: add Thinkpad Helix 2 to force_combo_jack_table
by Yifeng Li 15 May '17

15 May '17

Thinkpad Helix 2 is a tablet PC, the audio is powered by Core M broadwell-audio and rt286 codec. For all versions of Linux kernel, the stereo output doesn't work properly when earphones are plugged in, the sound was coming out from both channels even if the audio contains only the left or right channel. Furthermore, if a music recorded in stereo is played, the two channels cancle out each other out, as a result, no voice but only distorted background music can be heard, like a sound card with builtin a Karaoke sount effect. Apparently this tablet uses a combo jack with polarity incorrectly set by rt286 driver. This patch adds DMI information of Thinkpad Helix 2 to force_combo_jack_table[] and the issue is resolved. The microphone input doesn't work regardless to the presence of this patch and still needs help from other developers to investigate. This is my first patch to LKML directly, sorry for CC-ing too many people here. Link: https://bugzilla.kernel.org/show_bug.cgi?id=93841 Cc: Bard Liao <bardliao(a)realtek.com> Cc: Oder Chiou <oder_chiou(a)realtek.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Keyon <yang.jie(a)linux.intel.com> Cc: Jinliang Wang <jinliangx.wang(a)intel.com> Cc: stable(a)vger.kernel.org Signed-off-by: Yifeng Li <tomli(a)tomli.me> --- sound/soc/codecs/rt286.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index 9c365a7f758d..7899a2cdeb42 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -1108,6 +1108,13 @@ static const struct dmi_system_id force_combo_jack_table[] = { DMI_MATCH(DMI_PRODUCT_NAME, "Kabylake Client platform") } }, + { + .ident = "Thinkpad Helix 2nd", + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "LENOVO"), + DMI_MATCH(DMI_PRODUCT_VERSION, "ThinkPad Helix 2nd") + } + }, { } }; -- 2.12.1

2 1

[alsa-devel] [PATCH] ASoC: da7213: Fix incorrect usage of bitwise '&' operator for SRM check
by Adam Thomson 15 May '17

15 May '17

In the SRM lock check section of code the '&' bitwise operator is used as part of checking lock status. Functionally the code works as intended, but the conditional statement is a boolean comparison so should really use '&&' logical operator instead. This commit rectifies this discrepancy. Signed-off-by: Adam Thomson <Adam.Thomson.Opensource(a)diasemi.com> --- sound/soc/codecs/da7213.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/da7213.c b/sound/soc/codecs/da7213.c index 12da558..d114f33 100644 --- a/sound/soc/codecs/da7213.c +++ b/sound/soc/codecs/da7213.c @@ -771,7 +771,7 @@ static int da7213_dai_event(struct snd_soc_dapm_widget *w, ++i; msleep(50); } - } while ((i < DA7213_SRM_CHECK_RETRIES) & (!srm_lock)); + } while ((i < DA7213_SRM_CHECK_RETRIES) && (!srm_lock)); if (!srm_lock) dev_warn(codec->dev, "SRM failed to lock\n"); -- 1.9.1

2 1

[alsa-devel] Applied "ASoC: tlv320dac31xx: Fix mistype in tlv320dac31xx codec" to the asoc tree
by Mark Brown 15 May '17

15 May '17

The patch ASoC: tlv320dac31xx: Fix mistype in tlv320dac31xx codec 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 51827c41c9ce07293b094691673e6ec23dfdc5e8 Mon Sep 17 00:00:00 2001 From: Tomas Vilda <tomas.vilda(a)gmail.com> Date: Sat, 13 May 2017 00:29:37 +0300 Subject: [PATCH] ASoC: tlv320dac31xx: Fix mistype in tlv320dac31xx codec Signed-off-by: Tomas Vilda <tomas.vilda(a)gmail.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/tlv320aic31xx.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index f8a90ba8cd71..d7d03c92cb8a 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -1210,7 +1210,7 @@ static const struct snd_soc_dai_ops aic31xx_dai_ops = { static struct snd_soc_dai_driver dac31xx_dai_driver[] = { { - .name = "tlv32dac31xx-hifi", + .name = "tlv320dac31xx-hifi", .playback = { .stream_name = "Playback", .channels_min = 2, -- 2.11.0

1 0

[alsa-devel] [PATCH][V2] ASoC: Intel: sst: fix spelling mistake: "allocationf" -> "allocation"
by Colin King 15 May '17

15 May '17

From: Colin Ian King <colin.king(a)canonical.com> Trivial fix to spelling mistake in dev_err message. Also replace "fail" with "failure". Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/atom/sst-mfld-platform-pcm.c b/sound/soc/intel/atom/sst-mfld-platform-pcm.c index 21cac1c8dd4c..e74119113713 100644 --- a/sound/soc/intel/atom/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/atom/sst-mfld-platform-pcm.c @@ -690,7 +690,7 @@ static int sst_pcm_new(struct snd_soc_pcm_runtime *rtd) snd_dma_continuous_data(GFP_DMA), SST_MIN_BUFFER, SST_MAX_BUFFER); if (retval) { - dev_err(rtd->dev, "dma buffer allocationf fail\n"); + dev_err(rtd->dev, "dma buffer allocation failure\n"); return retval; } } -- 2.11.0

3 2

[alsa-devel] [PATCH] ASoC: nau8824: TDM support
by John Hsu 15 May '17

15 May '17

Support TDM format for NAU88L24. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> Signed-off-by: John Hsu <supercraig0719(a)gmail.com> --- sound/soc/codecs/nau8824.c | 52 ++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/nau8824.h | 12 +++++++++++ 2 files changed, 64 insertions(+) diff --git a/sound/soc/codecs/nau8824.c b/sound/soc/codecs/nau8824.c index cca974d..3a309b1 100644 --- a/sound/soc/codecs/nau8824.c +++ b/sound/soc/codecs/nau8824.c @@ -1125,6 +1125,57 @@ static int nau8824_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) } /** + * nau8824_set_tdm_slot - configure DAI TDM. + * @dai: DAI + * @tx_mask: Bitmask representing active TX slots. Ex. + * 0xf for normal 4 channel TDM. + * 0xf0 for shifted 4 channel TDM + * @rx_mask: Bitmask [0:1] representing active DACR RX slots. + * Bitmask [2:3] representing active DACL RX slots. + * 00=CH0,01=CH1,10=CH2,11=CH3. Ex. + * 0xf for DACL/R selecting TDM CH3. + * 0xf0 for DACL/R selecting shifted TDM CH3. + * @slots: Number of slots in use. + * @slot_width: Width in bits for each slot. + * + * Configures a DAI for TDM operation. Only support 4 slots TDM. + */ +static int nau8824_set_tdm_slot(struct snd_soc_dai *dai, + unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width) +{ + struct snd_soc_codec *codec = dai->codec; + struct nau8824 *nau8824 = snd_soc_codec_get_drvdata(codec); + unsigned int tslot_l = 0, ctrl_val = 0; + + if (slots > 4 || ((tx_mask & 0xf0) && (tx_mask & 0xf)) || + ((rx_mask & 0xf0) && (rx_mask & 0xf)) || + ((rx_mask & 0xf0) && (tx_mask & 0xf)) || + ((rx_mask & 0xf) && (tx_mask & 0xf0))) + return -EINVAL; + + ctrl_val |= (NAU8824_TDM_MODE | NAU8824_TDM_OFFSET_EN); + if (tx_mask & 0xf0) { + tslot_l = 4 * slot_width; + ctrl_val |= (tx_mask >> 4); + } else { + ctrl_val |= tx_mask; + } + if (rx_mask & 0xf0) + ctrl_val |= ((rx_mask >> 4) << NAU8824_TDM_DACR_RX_SFT); + else + ctrl_val |= (rx_mask << NAU8824_TDM_DACR_RX_SFT); + + regmap_update_bits(nau8824->regmap, NAU8824_REG_TDM_CTRL, + NAU8824_TDM_MODE | NAU8824_TDM_OFFSET_EN | + NAU8824_TDM_DACL_RX_MASK | NAU8824_TDM_DACR_RX_MASK | + NAU8824_TDM_TX_MASK, ctrl_val); + regmap_update_bits(nau8824->regmap, NAU8824_REG_PORT0_LEFT_TIME_SLOT, + NAU8824_TSLOT_L_MASK, tslot_l); + + return 0; +} + +/** * nau8824_calc_fll_param - Calculate FLL parameters. * @fll_in: external clock provided to codec. * @fs: sampling rate. @@ -1440,6 +1491,7 @@ static struct snd_soc_codec_driver nau8824_codec_driver = { static const struct snd_soc_dai_ops nau8824_dai_ops = { .hw_params = nau8824_hw_params, .set_fmt = nau8824_set_fmt, + .set_tdm_slot = nau8824_set_tdm_slot, }; #define NAU8824_RATES SNDRV_PCM_RATE_8000_192000 diff --git a/sound/soc/codecs/nau8824.h b/sound/soc/codecs/nau8824.h index 87ac9a3..21eae24 100644 --- a/sound/soc/codecs/nau8824.h +++ b/sound/soc/codecs/nau8824.h @@ -258,6 +258,18 @@ #define NAU8824_I2S_MS_SLAVE (0 << NAU8824_I2S_MS_SFT) #define NAU8824_I2S_BLK_DIV_MASK 0x7 +/* PORT0_LEFT_TIME_SLOT (0x1E) */ +#define NAU8824_TSLOT_L_MASK 0x3ff + +/* TDM_CTRL (0x20) */ +#define NAU8824_TDM_MODE (0x1 << 15) +#define NAU8824_TDM_OFFSET_EN (0x1 << 14) +#define NAU8824_TDM_DACL_RX_SFT 6 +#define NAU8824_TDM_DACL_RX_MASK (0x3 << NAU8824_TDM_DACL_RX_SFT) +#define NAU8824_TDM_DACR_RX_SFT 4 +#define NAU8824_TDM_DACR_RX_MASK (0x3 << NAU8824_TDM_DACR_RX_SFT) +#define NAU8824_TDM_TX_MASK 0xf + /* ADC_FILTER_CTRL (0x24) */ #define NAU8824_ADC_SYNC_DOWN_MASK 0x3 #define NAU8824_ADC_SYNC_DOWN_32 0 -- 2.6.4

2 1

[alsa-devel] [PATCH] ASoC: rt5514: Add more width and channels support in the TDM mode
by Oder Chiou 15 May '17

15 May '17

This patch adds more width and channels support in the TDM mode. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5514.c | 23 +++++++++++++++++++++-- sound/soc/codecs/rt5514.h | 6 ++++++ 2 files changed, 27 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5514.c b/sound/soc/codecs/rt5514.c index f91221b..ff97360 100644 --- a/sound/soc/codecs/rt5514.c +++ b/sound/soc/codecs/rt5514.c @@ -906,9 +906,23 @@ static int rt5514_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, if (rx_mask || tx_mask) val |= RT5514_TDM_MODE; - if (slots == 4) + switch (slots) { + case 4: val |= RT5514_TDMSLOT_SEL_RX_4CH | RT5514_TDMSLOT_SEL_TX_4CH; + break; + + case 6: + val |= RT5514_TDMSLOT_SEL_RX_6CH | RT5514_TDMSLOT_SEL_TX_6CH; + break; + + case 8: + val |= RT5514_TDMSLOT_SEL_RX_8CH | RT5514_TDMSLOT_SEL_TX_8CH; + break; + case 2: + default: + break; + } switch (slot_width) { case 20: @@ -919,6 +933,10 @@ static int rt5514_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, val |= RT5514_CH_LEN_RX_24 | RT5514_CH_LEN_TX_24; break; + case 25: + val |= RT5514_TDM_MODE2; + break; + case 32: val |= RT5514_CH_LEN_RX_32 | RT5514_CH_LEN_TX_32; break; @@ -930,7 +948,8 @@ static int rt5514_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, regmap_update_bits(rt5514->regmap, RT5514_I2S_CTRL1, RT5514_TDM_MODE | RT5514_TDMSLOT_SEL_RX_MASK | RT5514_TDMSLOT_SEL_TX_MASK | - RT5514_CH_LEN_RX_MASK | RT5514_CH_LEN_TX_MASK, val); + RT5514_CH_LEN_RX_MASK | RT5514_CH_LEN_TX_MASK | + RT5514_TDM_MODE2, val); return 0; } diff --git a/sound/soc/codecs/rt5514.h b/sound/soc/codecs/rt5514.h index 5d343fb..02bc212 100644 --- a/sound/soc/codecs/rt5514.h +++ b/sound/soc/codecs/rt5514.h @@ -117,6 +117,8 @@ #define RT5514_POW_ADCFEDL_BIT 0 /* RT5514_I2S_CTRL1 (0x2010) */ +#define RT5514_TDM_MODE2 (0x1 << 30) +#define RT5514_TDM_MODE2_SFT 30 #define RT5514_TDM_MODE (0x1 << 28) #define RT5514_TDM_MODE_SFT 28 #define RT5514_I2S_LR_MASK (0x1 << 26) @@ -136,6 +138,8 @@ #define RT5514_TDMSLOT_SEL_RX_MASK (0x3 << 10) #define RT5514_TDMSLOT_SEL_RX_SFT 10 #define RT5514_TDMSLOT_SEL_RX_4CH (0x1 << 10) +#define RT5514_TDMSLOT_SEL_RX_6CH (0x2 << 10) +#define RT5514_TDMSLOT_SEL_RX_8CH (0x3 << 10) #define RT5514_CH_LEN_RX_MASK (0x3 << 8) #define RT5514_CH_LEN_RX_SFT 8 #define RT5514_CH_LEN_RX_16 (0x0 << 8) @@ -145,6 +149,8 @@ #define RT5514_TDMSLOT_SEL_TX_MASK (0x3 << 6) #define RT5514_TDMSLOT_SEL_TX_SFT 6 #define RT5514_TDMSLOT_SEL_TX_4CH (0x1 << 6) +#define RT5514_TDMSLOT_SEL_TX_6CH (0x2 << 6) +#define RT5514_TDMSLOT_SEL_TX_8CH (0x3 << 6) #define RT5514_CH_LEN_TX_MASK (0x3 << 4) #define RT5514_CH_LEN_TX_SFT 4 #define RT5514_CH_LEN_TX_16 (0x0 << 4) -- 2.7.4

2 1

[alsa-devel] [PATCH] ASoC: rt5665: add ADC STO2 ASRC support
by Bard Liao 15 May '17

15 May '17

"ADC Stereo2 Filter" is with ASRC supported. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5665.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/rt5665.c b/sound/soc/codecs/rt5665.c index 8cd2230..ece77b8 100644 --- a/sound/soc/codecs/rt5665.c +++ b/sound/soc/codecs/rt5665.c @@ -2684,6 +2684,8 @@ static const struct snd_soc_dapm_widget rt5665_dapm_widgets[] = { RT5665_DAC_MONO_R_ASRC_SFT, 0, NULL, 0), SND_SOC_DAPM_SUPPLY_S("ADC STO1 ASRC", 1, RT5665_ASRC_1, RT5665_ADC_STO1_ASRC_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("ADC STO2 ASRC", 1, RT5665_ASRC_1, + RT5665_ADC_STO2_ASRC_SFT, 0, NULL, 0), SND_SOC_DAPM_SUPPLY_S("ADC Mono L ASRC", 1, RT5665_ASRC_1, RT5665_ADC_MONO_L_ASRC_SFT, 0, NULL, 0), SND_SOC_DAPM_SUPPLY_S("ADC Mono R ASRC", 1, RT5665_ASRC_1, @@ -3227,6 +3229,7 @@ static const struct snd_soc_dapm_route rt5665_dapm_routes[] = { /*ASRC*/ {"ADC Stereo1 Filter", NULL, "ADC STO1 ASRC", is_using_asrc}, + {"ADC Stereo2 Filter", NULL, "ADC STO2 ASRC", is_using_asrc}, {"ADC Mono Left Filter", NULL, "ADC Mono L ASRC", is_using_asrc}, {"ADC Mono Right Filter", NULL, "ADC Mono R ASRC", is_using_asrc}, {"DAC Mono Left Filter", NULL, "DAC Mono L ASRC", is_using_asrc}, -- 2.7.4

2 1

[alsa-devel] [PATCH] ASoC: rsnd: tidyup Sampling rate convert example on Document
by Kuninori Morimoto 15 May '17

15 May '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> "renesas,rsrc-card" is exchanged to "simple-scu-card". Let's update Document Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- .../devicetree/bindings/sound/renesas,rsnd.txt | 30 +++++++++++----------- 1 file changed, 15 insertions(+), 15 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt index 15a7316..3332910 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt @@ -83,11 +83,11 @@ SRC can convert [xx]Hz to [yy]Hz. Then, it has below 2 modes ** Asynchronous mode ------------------ -You need to use "renesas,rsrc-card" sound card for it. +You need to use "simple-scu-audio-card" sound card for it. example) sound { - compatible = "renesas,rsrc-card"; + compatible = "simple-scu-audio-card"; ... /* * SRC Asynchronous mode setting @@ -97,12 +97,12 @@ example) * Inputed 48kHz data will be converted to * system specified Hz */ - convert-rate = <48000>; + simple-audio-card,convert-rate = <48000>; ... - cpu { + simple-audio-card,cpu { sound-dai = <&rcar_sound>; }; - codec { + simple-audio-card,codec { ... }; }; @@ -141,23 +141,23 @@ For more detail information, see below ${LINUX}/sound/soc/sh/rcar/ctu.c - comment of header -You need to use "renesas,rsrc-card" sound card for it. +You need to use "simple-scu-audio-card" sound card for it. example) sound { - compatible = "renesas,rsrc-card"; + compatible = "simple-scu-audio-card"; ... /* * CTU setting * All input data will be converted to 2ch * as output data */ - convert-channels = <2>; + simple-audio-card,convert-channels = <2>; ... - cpu { + simple-audio-card,cpu { sound-dai = <&rcar_sound>; }; - codec { + simple-audio-card,codec { ... }; }; @@ -190,22 +190,22 @@ and these sounds will be merged by MIX. aplay -D plughw:0,0 xxxx.wav & aplay -D plughw:0,1 yyyy.wav -You need to use "renesas,rsrc-card" sound card for it. +You need to use "simple-scu-audio-card" sound card for it. Ex) [MEM] -> [SRC1] -> [CTU02] -+-> [MIX0] -> [DVC0] -> [SSI0] | [MEM] -> [SRC2] -> [CTU03] -+ sound { - compatible = "renesas,rsrc-card"; + compatible = "simple-scu-audio-card"; ... - cpu@0 { + simple-audio-card,cpu@0 { sound-dai = <&rcar_sound 0>; }; - cpu@1 { + simple-audio-card,cpu@1 { sound-dai = <&rcar_sound 1>; }; - codec { + simple-audio-card,codec { ... }; }; -- 1.9.1

2 2

[alsa-devel] [PATCH][RESEND] ASoC: rsnd: add missing clocks/clock-names on Document
by Kuninori Morimoto 15 May '17

15 May '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- Documentation/devicetree/bindings/sound/renesas,rsnd.txt | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt index 3332910..13cb21d5 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt @@ -368,6 +368,10 @@ Required properties: see below for detail. - #sound-dai-cells : it must be 0 if your system is using single DAI it must be 1 if your system is using multi DAI +- clocks : References to SSI/SRC/MIX/CTU/DVC/AUDIO_CLK clocks. +- clock-names : List of necessary clock names. + "ssi-all", "ssi.X", "src.X", "mix.X", "ctu.X", + "dvc.X", "clk_a", "clk_b", "clk_c", "clk_i" Optional properties: - #clock-cells : it must be 0 if your system has audio_clkout -- 1.9.1

1 0

[alsa-devel] [PATCH v2][RESEND] ASoC: rsnd: control kctrl items acceptance anytime/runtime
by Kuninori Morimoto 15 May '17

15 May '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Current SRC/DVC/CTU adds kctrl for each device, and SRC can adjust its sampling rate during playback, thus, this feature should be enabled only *during* playback. This patch controls it more clearly Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/core.c | 17 +++++++++++++++++ sound/soc/sh/rcar/ctu.c | 6 ++++++ sound/soc/sh/rcar/dvc.c | 5 +++++ sound/soc/sh/rcar/rsnd.h | 16 ++++++++++------ sound/soc/sh/rcar/src.c | 2 ++ 5 files changed, 40 insertions(+), 6 deletions(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 1744015..d3c9078 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -982,6 +982,9 @@ static int rsnd_kctrl_put(struct snd_kcontrol *kctrl, struct rsnd_kctrl_cfg *cfg = kcontrol_to_cfg(kctrl); int i, change = 0; + if (!cfg->accept(cfg->io)) + return 0; + for (i = 0; i < cfg->size; i++) { if (cfg->texts) { change |= (uc->value.enumerated.item[i] != cfg->val[i]); @@ -998,6 +1001,18 @@ static int rsnd_kctrl_put(struct snd_kcontrol *kctrl, return change; } +int rsnd_kctrl_accept_anytime(struct rsnd_dai_stream *io) +{ + return 1; +} + +int rsnd_kctrl_accept_runtime(struct rsnd_dai_stream *io) +{ + struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io); + + return !!runtime; +} + struct rsnd_kctrl_cfg *rsnd_kctrl_init_m(struct rsnd_kctrl_cfg_m *cfg) { cfg->cfg.val = cfg->val; @@ -1016,6 +1031,7 @@ int rsnd_kctrl_new(struct rsnd_mod *mod, struct rsnd_dai_stream *io, struct snd_soc_pcm_runtime *rtd, const unsigned char *name, + int (*accept)(struct rsnd_dai_stream *io), void (*update)(struct rsnd_dai_stream *io, struct rsnd_mod *mod), struct rsnd_kctrl_cfg *cfg, @@ -1050,6 +1066,7 @@ int rsnd_kctrl_new(struct rsnd_mod *mod, cfg->texts = texts; cfg->max = max; cfg->size = size; + cfg->accept = accept; cfg->update = update; cfg->card = card; cfg->kctrl = kctrl; diff --git a/sound/soc/sh/rcar/ctu.c b/sound/soc/sh/rcar/ctu.c index 9dcc1f9..4ba8f2f 100644 --- a/sound/soc/sh/rcar/ctu.c +++ b/sound/soc/sh/rcar/ctu.c @@ -279,12 +279,14 @@ static int rsnd_ctu_pcm_new(struct rsnd_mod *mod, /* CTU Pass */ ret = rsnd_kctrl_new_m(mod, io, rtd, "CTU Pass", + rsnd_kctrl_accept_anytime, NULL, &ctu->pass, RSND_MAX_CHANNELS, 0xC); /* ROW0 */ ret = rsnd_kctrl_new_m(mod, io, rtd, "CTU SV0", + rsnd_kctrl_accept_anytime, NULL, &ctu->sv0, RSND_MAX_CHANNELS, 0x00FFFFFF); @@ -293,6 +295,7 @@ static int rsnd_ctu_pcm_new(struct rsnd_mod *mod, /* ROW1 */ ret = rsnd_kctrl_new_m(mod, io, rtd, "CTU SV1", + rsnd_kctrl_accept_anytime, NULL, &ctu->sv1, RSND_MAX_CHANNELS, 0x00FFFFFF); @@ -301,6 +304,7 @@ static int rsnd_ctu_pcm_new(struct rsnd_mod *mod, /* ROW2 */ ret = rsnd_kctrl_new_m(mod, io, rtd, "CTU SV2", + rsnd_kctrl_accept_anytime, NULL, &ctu->sv2, RSND_MAX_CHANNELS, 0x00FFFFFF); @@ -309,6 +313,7 @@ static int rsnd_ctu_pcm_new(struct rsnd_mod *mod, /* ROW3 */ ret = rsnd_kctrl_new_m(mod, io, rtd, "CTU SV3", + rsnd_kctrl_accept_anytime, NULL, &ctu->sv3, RSND_MAX_CHANNELS, 0x00FFFFFF); @@ -317,6 +322,7 @@ static int rsnd_ctu_pcm_new(struct rsnd_mod *mod, /* Reset */ ret = rsnd_kctrl_new_s(mod, io, rtd, "CTU Reset", + rsnd_kctrl_accept_anytime, rsnd_ctu_value_reset, &ctu->reset, 1); diff --git a/sound/soc/sh/rcar/dvc.c b/sound/soc/sh/rcar/dvc.c index 463de83..75af6e7 100644 --- a/sound/soc/sh/rcar/dvc.c +++ b/sound/soc/sh/rcar/dvc.c @@ -257,6 +257,7 @@ static int rsnd_dvc_pcm_new(struct rsnd_mod *mod, ret = rsnd_kctrl_new_m(mod, io, rtd, is_play ? "DVC Out Playback Volume" : "DVC In Capture Volume", + rsnd_kctrl_accept_anytime, rsnd_dvc_volume_update, &dvc->volume, slots, 0x00800000 - 1); @@ -267,6 +268,7 @@ static int rsnd_dvc_pcm_new(struct rsnd_mod *mod, ret = rsnd_kctrl_new_m(mod, io, rtd, is_play ? "DVC Out Mute Switch" : "DVC In Mute Switch", + rsnd_kctrl_accept_anytime, rsnd_dvc_volume_update, &dvc->mute, slots, 1); @@ -277,6 +279,7 @@ static int rsnd_dvc_pcm_new(struct rsnd_mod *mod, ret = rsnd_kctrl_new_s(mod, io, rtd, is_play ? "DVC Out Ramp Switch" : "DVC In Ramp Switch", + rsnd_kctrl_accept_anytime, rsnd_dvc_volume_update, &dvc->ren, 1); if (ret < 0) @@ -285,6 +288,7 @@ static int rsnd_dvc_pcm_new(struct rsnd_mod *mod, ret = rsnd_kctrl_new_e(mod, io, rtd, is_play ? "DVC Out Ramp Up Rate" : "DVC In Ramp Up Rate", + rsnd_kctrl_accept_anytime, rsnd_dvc_volume_update, &dvc->rup, dvc_ramp_rate); @@ -294,6 +298,7 @@ static int rsnd_dvc_pcm_new(struct rsnd_mod *mod, ret = rsnd_kctrl_new_e(mod, io, rtd, is_play ? "DVC Out Ramp Down Rate" : "DVC In Ramp Down Rate", + rsnd_kctrl_accept_anytime, rsnd_dvc_volume_update, &dvc->rdown, dvc_ramp_rate); diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index dbf4163..f68aecb 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -594,6 +594,7 @@ struct rsnd_kctrl_cfg { unsigned int size; u32 *val; const char * const *texts; + int (*accept)(struct rsnd_dai_stream *io); void (*update)(struct rsnd_dai_stream *io, struct rsnd_mod *mod); struct rsnd_dai_stream *io; struct snd_card *card; @@ -611,12 +612,15 @@ struct rsnd_kctrl_cfg_s { u32 val; }; +int rsnd_kctrl_accept_anytime(struct rsnd_dai_stream *io); +int rsnd_kctrl_accept_runtime(struct rsnd_dai_stream *io); struct rsnd_kctrl_cfg *rsnd_kctrl_init_m(struct rsnd_kctrl_cfg_m *cfg); struct rsnd_kctrl_cfg *rsnd_kctrl_init_s(struct rsnd_kctrl_cfg_s *cfg); int rsnd_kctrl_new(struct rsnd_mod *mod, struct rsnd_dai_stream *io, struct snd_soc_pcm_runtime *rtd, const unsigned char *name, + int (*accept)(struct rsnd_dai_stream *io), void (*update)(struct rsnd_dai_stream *io, struct rsnd_mod *mod), struct rsnd_kctrl_cfg *cfg, @@ -624,16 +628,16 @@ int rsnd_kctrl_new(struct rsnd_mod *mod, int size, u32 max); -#define rsnd_kctrl_new_m(mod, io, rtd, name, update, cfg, size, max) \ - rsnd_kctrl_new(mod, io, rtd, name, update, rsnd_kctrl_init_m(cfg), \ +#define rsnd_kctrl_new_m(mod, io, rtd, name, accept, update, cfg, size, max) \ + rsnd_kctrl_new(mod, io, rtd, name, accept, update, rsnd_kctrl_init_m(cfg), \ NULL, size, max) -#define rsnd_kctrl_new_s(mod, io, rtd, name, update, cfg, max) \ - rsnd_kctrl_new(mod, io, rtd, name, update, rsnd_kctrl_init_s(cfg), \ +#define rsnd_kctrl_new_s(mod, io, rtd, name, accept, update, cfg, max) \ + rsnd_kctrl_new(mod, io, rtd, name, accept, update, rsnd_kctrl_init_s(cfg), \ NULL, 1, max) -#define rsnd_kctrl_new_e(mod, io, rtd, name, update, cfg, texts) \ - rsnd_kctrl_new(mod, io, rtd, name, update, rsnd_kctrl_init_s(cfg), \ +#define rsnd_kctrl_new_e(mod, io, rtd, name, accept, update, cfg, texts) \ + rsnd_kctrl_new(mod, io, rtd, name, accept, update, rsnd_kctrl_init_s(cfg), \ texts, 1, ARRAY_SIZE(texts)) /* diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c index e4bde0c..e395fcb 100644 --- a/sound/soc/sh/rcar/src.c +++ b/sound/soc/sh/rcar/src.c @@ -489,6 +489,7 @@ static int rsnd_src_pcm_new(struct rsnd_mod *mod, rsnd_io_is_play(io) ? "SRC Out Rate Switch" : "SRC In Rate Switch", + rsnd_kctrl_accept_anytime, rsnd_src_set_convert_rate, &src->sen, 1); if (ret < 0) @@ -498,6 +499,7 @@ static int rsnd_src_pcm_new(struct rsnd_mod *mod, rsnd_io_is_play(io) ? "SRC Out Rate" : "SRC In Rate", + rsnd_kctrl_accept_runtime, rsnd_src_set_convert_rate, &src->sync, 192000); -- 1.9.1

1 0

[alsa-devel] linux-4.12-rc1/sound/soc/codecs/cs35l35.c: false comparison ?
by David Binderman 14 May '17

14 May '17

hello there, [linux-4.12-rc1/sound/soc/codecs/cs35l35.c:495] -> [linux-4.12-rc1/sound/soc/codecs/cs35l35.c:498]: (style) Mismatching assignment and comparison, comparison 'errata_chk==1' is always false. [linux-4.12-rc1/sound/soc/codecs/cs35l35.c:495] -> [linux-4.12-rc1/sound/soc/codecs/cs35l35.c:498]: (style) Mismatching assignment and comparison, comparison 'errata_chk==3' is always false. Source code is errata_chk = clk_ctl & CS35L35_SP_RATE_MASK; if (classh->classh_wk_fet_disable == 0x00 && (errata_chk == 0x01 || errata_chk == 0x03)) { but ./sound/soc/codecs/cs35l35.h:#define CS35L35_SP_RATE_MASK 0xC0 Regards David Binderman

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[alsa-devel] [sound:topic/for-4.13 22/30] sound/core/pcm.c:905:3: error: implicit declaration of function 'pcm_call_notify'
by kbuild test robot 12 May '17

12 May '17

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git topic/for-4.13 head: c56407e6d958efda0d2a587ed89e02a251975e06 commit: e9a0fa885872c92e6ada08fb1cf3433945d235a3 [22/30] ALSA: pcm: Build pcm notifier code conditionally config: x86_64-kexec (attached as .config) compiler: gcc-6 (Debian 6.2.0-3) 6.2.0 20160901 reproduce: git checkout e9a0fa885872c92e6ada08fb1cf3433945d235a3 # save the attached .config to linux build tree make ARCH=x86_64 All errors (new ones prefixed by >>): sound/core/pcm.c: In function 'snd_pcm_free': >> sound/core/pcm.c:905:3: error: implicit declaration of function 'pcm_call_notify' [-Werror=implicit-function-declaration] pcm_call_notify(pcm, n_unregister); ^~~~~~~~~~~~~~~ >> sound/core/pcm.c:905:24: error: 'n_unregister' undeclared (first use in this function) pcm_call_notify(pcm, n_unregister); ^~~~~~~~~~~~ sound/core/pcm.c:905:24: note: each undeclared identifier is reported only once for each function it appears in sound/core/pcm.c: In function 'snd_pcm_dev_register': >> sound/core/pcm.c:1118:23: error: 'n_register' undeclared (first use in this function) pcm_call_notify(pcm, n_register); ^~~~~~~~~~ sound/core/pcm.c: In function 'snd_pcm_dev_disconnect': >> sound/core/pcm.c:1147:24: error: 'n_disconnect' undeclared (first use in this function) pcm_call_notify(pcm, n_disconnect); ^~~~~~~~~~~~ cc1: some warnings being treated as errors vim +/pcm_call_notify +905 sound/core/pcm.c 899 900 static int snd_pcm_free(struct snd_pcm *pcm) 901 { 902 if (!pcm) 903 return 0; 904 if (!pcm->internal) > 905 pcm_call_notify(pcm, n_unregister); 906 if (pcm->private_free) 907 pcm->private_free(pcm); 908 snd_pcm_lib_preallocate_free_for_all(pcm); --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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[alsa-devel] [PATCH 0/3] ALSA: USB-descriptor endianess fixes
by Johan Hovold 12 May '17

12 May '17

These patches add missing endianness conversions when accessing the USB device-descriptor fields. In the process, clean up the us122l driver which had product-id conditionals sprinkled throughout. The final patch drops the Kconfig dependency on x86 since it's a USB driver that compiles just fine on non-x86. Perhaps this should be a (X86 || COMPILE_TEST) dependency instead in case there are external dependencies that warrants this limitation. Note that these patches have only been compile-tested. Johan Johan Hovold (3): ALSA: usb-audio: fix Amanero Combo384 quirk on big-endian hosts ALSA: us122l: clean up US144 handling ALSA: us122l: drop x86 dependency sound/usb/Kconfig | 1 - sound/usb/quirks.c | 2 +- sound/usb/usx2y/us122l.c | 36 ++++++++++++++++++------------------ sound/usb/usx2y/us122l.h | 2 ++ 4 files changed, 21 insertions(+), 20 deletions(-) -- 2.13.0

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