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

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[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

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[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

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[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

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[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

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