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[alsa-devel] amixer: convert percentage into db wrongly
by Elimar Riesebieter 09 Mar '12

09 Mar '12

severity 663090 normal forwarded 663090 alsa-devel(a)alsa-project.org thanks * Adam Lee <adam8157(a)gmail.com> [2012-03-08 20:36 +0800]: > Package: alsa-utils > Version: 1.0.25-1 > Severity: important > > > db is not linear, but amixer believe it is. > > "amixer get Master" says "Limits: Playback 0 - 74", then everytime I run > "amixer -q sset Master 10%-", there is 8db dec. > > For example, at first Master is 100% and 0db, both alsamixer and amixer > think it is, and after I run "amixer -q sset Master 10%-", both > alsamixer and amixer says Master is -8.00db, but alsamixer says it is > 72%, amixer says it is 89%. > > alsamixer is right, amixer calc and set wrongly. > > Several months ago, they both right, and I upgraded, and amixer messed > up. Please fix it. Thank you. > > FYI, my hardware is: > "HDA-Intel" "Conexant CX20585" "HDA:14f15069,17aa214c,00100302 HDA:14f12c06,17aa2122,00100000" "0x17aa" "0x215e" > > > -- System Information: > Debian Release: wheezy/sid > APT prefers unstable > APT policy: (500, 'unstable') > Architecture: amd64 (x86_64) > > Kernel: Linux 3.2.0-1-amd64 (SMP w/4 CPU cores) > Locale: LANG=en_US.UTF-8, LC_CTYPE=en_US.UTF-8 (charmap=UTF-8) > Shell: /bin/sh linked to /bin/dash > > Versions of packages alsa-utils depends on: > ii dialog 1.1-20120215-1 > ii libasound2 1.0.25-2 > ii libc6 2.13-27 > ii libncursesw5 5.9-4 > ii libsamplerate0 0.1.8-3 > ii libtinfo5 5.9-4 > ii linux-sound-base 1.0.23+dfsg-4 > ii lsb-base 3.2+Debian29 > ii module-init-tools 6-1 > ii whiptail 0.52.14-8 > > Versions of packages alsa-utils recommends: > ii alsa-base 1.0.23+dfsg-4 > ii pciutils 1:3.1.8-2 > > alsa-utils suggests no packages. > > -- no debconf information > > > > _______________________________________________ > Pkg-alsa-devel mailing list > Pkg-alsa-devel(a)lists.alioth.debian.org > http://lists.alioth.debian.org/cgi-bin/mailman/listinfo/pkg-alsa-devel > -- The path to source is always uphill! -unknown-

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[alsa-devel] [PATCH] ASoC: omap-mcbsp: When closing the port select PRCM source for CLKS signal
by Peter Ujfalusi 09 Mar '12

09 Mar '12

If external source for the CLKS signal selection kept after the port is no longer in use the system might refuse to go suspend. There is also a chance that the external clock is not running when next time the McBSP port is started which can result errors when we try to access McBSP registers. Reset the CLKS source back to PRCM source unconditionally. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/omap/mcbsp.c | 10 ++++++++++ 1 files changed, 10 insertions(+), 0 deletions(-) diff --git a/sound/soc/omap/mcbsp.c b/sound/soc/omap/mcbsp.c index d716793..21dbb05 100644 --- a/sound/soc/omap/mcbsp.c +++ b/sound/soc/omap/mcbsp.c @@ -548,6 +548,16 @@ void omap_mcbsp_free(struct omap_mcbsp *mcbsp) reg_cache = mcbsp->reg_cache; + /* + * Select CLKS source from internal source unconditionally before + * marking the McBSP port as free. + * If the external clock source via MCBSP_CLKS pin has been selected the + * system will refuse to enter idle if the CLKS pin source is not reset + * back to internal source. + */ + if (!cpu_class_is_omap1()) + omap2_mcbsp_set_clks_src(mcbsp, MCBSP_CLKS_PRCM_SRC); + spin_lock(&mcbsp->lock); if (mcbsp->free) dev_err(mcbsp->dev, "McBSP%d was not reserved\n", mcbsp->id); -- 1.7.8.5

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[alsa-devel] [PATCH v2 0/3] ARM/ASoC: OMAP McBSP: Correct CLKR/FSR source selection
by Peter Ujfalusi 08 Mar '12

08 Mar '12

Hello, Changes since v1: - Adapted to the moved mcbsp code which helped to reduce the number of patches from 5 to 3. - Clock selection related defines moved from arch to ASoC header. No change in functionality otherwise. This series fixes the CLKR/FSR mux selection (on McBSP ports with 6 pin setup). OMAP2/3 has 6 pin setup on McBSP1 port while OMAP4 has the 6 pin setup on McBSP4 port. Add support for OMAP4 CLKR/FSR source selection, and simplify the ASoC part of CLK mux configuration. This series should go via ASoC with the rest of McBSP changes. Regards, Peter --- Peter Ujfalusi (3): ARM: OMAP2+: McBSP: Correct CLKR/FSR clock source mux configuration ARM/ASoC: OMAP McBSP: Move remainig defines from arch to ASoC header ASoC: omap-mcbsp: Single function CLKR/FSR source mux configuration arch/arm/mach-omap2/mcbsp.c | 46 ++++++++++++++++++++++++++++- arch/arm/plat-omap/include/plat/mcbsp.h | 12 -------- sound/soc/omap/mcbsp.c | 48 ++++++++++++++---------------- sound/soc/omap/mcbsp.h | 13 +++++++- sound/soc/omap/omap-mcbsp.c | 8 ++-- 5 files changed, 81 insertions(+), 46 deletions(-) -- 1.7.8.5

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[alsa-devel] [PATCH] ASoC: add more sample rate for pxa-ssp
by Qiao Zhou 08 Mar '12

08 Mar '12

add more sample rate for pxa-ssp, which are supported, such as 32KHz, 64KHz. Signed-off-by: Qiao Zhou <zhouqiao(a)marvell.com> --- sound/soc/pxa/pxa-ssp.c | 3 ++- 1 files changed, 2 insertions(+), 1 deletions(-) diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index a57cfbc..a16df0f 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -764,7 +764,8 @@ static int pxa_ssp_remove(struct snd_soc_dai *dai) #define PXA_SSP_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | \ - SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | \ + SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_64000 | \ SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) #define PXA_SSP_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\ -- 1.7.4.1

2 1

[alsa-devel] [PATCH] ASoC: da7210: Update for using I2C regmap
by Ashish Chavan 08 Mar '12

08 Mar '12

Current DA7210 driver has I2C support using older register cache methods. This patch updates it for latest regmap framework. This has been tested on DA7210 EVB with Samsung SMDK6410 board. Signed-off-by: Ashish Chavan <ashish.chavan(a)kpitcummins.com> Signed-off-by: David Dajun Chen <dchen(a)diasemi.com> --- sound/soc/codecs/da7210.c | 234 +++++++++++++++++++++++++++++++++++++------- 1 files changed, 196 insertions(+), 38 deletions(-) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index ab38e93..0c23f19 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -17,6 +17,7 @@ #include <linux/delay.h> #include <linux/i2c.h> +#include <linux/regmap.h> #include <linux/slab.h> #include <linux/module.h> #include <sound/pcm.h> @@ -626,41 +627,170 @@ static const struct snd_soc_dapm_route da7210_audio_map[] = { /* Codec private data */ struct da7210_priv { - enum snd_soc_control_type control_type; + struct regmap *regmap; }; -/* - * Register cache - */ -static const u8 da7210_reg[] = { - 0x00, 0x11, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* R0 - R7 */ - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x08, /* R8 - RF */ - 0x00, 0x00, 0x00, 0x00, 0x08, 0x10, 0x10, 0x54, /* R10 - R17 */ - 0x40, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* R18 - R1F */ - 0x00, 0x00, 0x00, 0x02, 0x00, 0x76, 0x00, 0x00, /* R20 - R27 */ - 0x04, 0x00, 0x00, 0x30, 0x2A, 0x00, 0x40, 0x00, /* R28 - R2F */ - 0x40, 0x00, 0x40, 0x00, 0x40, 0x00, 0x40, 0x00, /* R30 - R37 */ - 0x40, 0x00, 0x40, 0x00, 0x40, 0x00, 0x00, 0x00, /* R38 - R3F */ - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* R40 - R4F */ - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* R48 - R4F */ - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* R50 - R57 */ - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* R58 - R5F */ - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* R60 - R67 */ - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* R68 - R6F */ - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* R70 - R77 */ - 0x00, 0x00, 0x00, 0x00, 0x00, 0x54, 0x54, 0x00, /* R78 - R7F */ - 0x00, 0x00, 0x2C, 0x00, 0x00, 0x00, 0x00, 0x00, /* R80 - R87 */ - 0x00, /* R88 */ +static struct reg_default da7210_reg_defaults[] = { + { 0x00, 0x00 }, + { 0x01, 0x11 }, + { 0x02, 0x00 }, + { 0x03, 0x00 }, + { 0x04, 0x00 }, + { 0x05, 0x00 }, + { 0x06, 0x00 }, + { 0x07, 0x00 }, + { 0x08, 0x00 }, + { 0x09, 0x00 }, + { 0x0a, 0x00 }, + { 0x0b, 0x00 }, + { 0x0c, 0x00 }, + { 0x0d, 0x00 }, + { 0x0e, 0x00 }, + { 0x0f, 0x08 }, + { 0x10, 0x00 }, + { 0x11, 0x00 }, + { 0x12, 0x00 }, + { 0x13, 0x00 }, + { 0x14, 0x08 }, + { 0x15, 0x10 }, + { 0x16, 0x10 }, + { 0x17, 0x54 }, + { 0x18, 0x40 }, + { 0x19, 0x00 }, + { 0x1a, 0x00 }, + { 0x1b, 0x00 }, + { 0x1c, 0x00 }, + { 0x1d, 0x00 }, + { 0x1e, 0x00 }, + { 0x1f, 0x00 }, + { 0x20, 0x00 }, + { 0x21, 0x00 }, + { 0x22, 0x00 }, + { 0x23, 0x02 }, + { 0x24, 0x00 }, + { 0x25, 0x76 }, + { 0x26, 0x00 }, + { 0x27, 0x00 }, + { 0x28, 0x04 }, + { 0x29, 0x00 }, + { 0x2a, 0x00 }, + { 0x2b, 0x30 }, + { 0x2c, 0x2A }, + { 0x2d, 0x00 }, + { 0x2e, 0x40 }, + { 0x2f, 0x00 }, + { 0x30, 0x40 }, + { 0x31, 0x00 }, + { 0x32, 0x40 }, + { 0x33, 0x00 }, + { 0x34, 0x40 }, + { 0x35, 0x00 }, + { 0x36, 0x40 }, + { 0x37, 0x00 }, + { 0x38, 0x40 }, + { 0x39, 0x00 }, + { 0x3a, 0x40 }, + { 0x3b, 0x00 }, + { 0x3c, 0x40 }, + { 0x3d, 0x00 }, + { 0x3e, 0x00 }, + { 0x3f, 0x00 }, + { 0x40, 0x00 }, + { 0x41, 0x00 }, + { 0x42, 0x00 }, + { 0x43, 0x00 }, + { 0x44, 0x00 }, + { 0x45, 0x00 }, + { 0x46, 0x00 }, + { 0x47, 0x00 }, + { 0x48, 0x00 }, + { 0x49, 0x00 }, + { 0x4a, 0x00 }, + { 0x4b, 0x00 }, + { 0x4c, 0x00 }, + { 0x4d, 0x00 }, + { 0x4e, 0x00 }, + { 0x4f, 0x00 }, + { 0x50, 0x00 }, + { 0x51, 0x00 }, + { 0x52, 0x00 }, + { 0x53, 0x00 }, + { 0x54, 0x00 }, + { 0x55, 0x00 }, + { 0x56, 0x00 }, + { 0x57, 0x00 }, + { 0x58, 0x00 }, + { 0x59, 0x00 }, + { 0x5a, 0x00 }, + { 0x5b, 0x00 }, + { 0x5c, 0x00 }, + { 0x5d, 0x00 }, + { 0x5e, 0x00 }, + { 0x5f, 0x00 }, + { 0x60, 0x00 }, + { 0x61, 0x00 }, + { 0x62, 0x00 }, + { 0x63, 0x00 }, + { 0x64, 0x00 }, + { 0x65, 0x00 }, + { 0x66, 0x00 }, + { 0x67, 0x00 }, + { 0x68, 0x00 }, + { 0x69, 0x00 }, + { 0x6a, 0x00 }, + { 0x6b, 0x00 }, + { 0x6c, 0x00 }, + { 0x6d, 0x00 }, + { 0x6e, 0x00 }, + { 0x6f, 0x00 }, + { 0x70, 0x00 }, + { 0x71, 0x00 }, + { 0x72, 0x00 }, + { 0x73, 0x00 }, + { 0x74, 0x00 }, + { 0x75, 0x00 }, + { 0x76, 0x00 }, + { 0x77, 0x00 }, + { 0x78, 0x00 }, + { 0x79, 0x00 }, + { 0x7a, 0x00 }, + { 0x7b, 0x00 }, + { 0x7c, 0x00 }, + { 0x7d, 0x54 }, + { 0x7e, 0x54 }, + { 0x7f, 0x00 }, + { 0x80, 0x00 }, + { 0x81, 0x00 }, + { 0x82, 0x2c }, + { 0x83, 0x00 }, + { 0x84, 0x00 }, + { 0x85, 0x00 }, + { 0x86, 0x00 }, + { 0x87, 0x00 }, + { 0x88, 0x00 }, }; -static int da7210_volatile_register(struct snd_soc_codec *codec, +static bool da7210_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case DA7210_A_HID_UNLOCK: + case DA7210_A_TEST_UNLOCK: + case DA7210_A_PLL1: + case DA7210_A_CP_MODE: + return false; + default: + return true; + } +} + +static bool da7210_volatile_register(struct device *dev, unsigned int reg) { switch (reg) { case DA7210_STATUS: - return 1; + return true; default: - return 0; + return false; } } @@ -866,7 +996,8 @@ static int da7210_probe(struct snd_soc_codec *codec) dev_info(codec->dev, "DA7210 Audio Codec %s\n", DA7210_VERSION); - ret = snd_soc_codec_set_cache_io(codec, 8, 8, da7210->control_type); + codec->control_data = da7210->regmap; + ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; @@ -983,12 +1114,14 @@ static int da7210_probe(struct snd_soc_codec *codec) snd_soc_update_bits(codec, DA7210_PLL, DA7210_PLL_EN, DA7210_PLL_EN); /* As suggested by Dialog */ - snd_soc_write(codec, DA7210_A_HID_UNLOCK, 0x8B); /* unlock */ - snd_soc_write(codec, DA7210_A_TEST_UNLOCK, 0xB4); - snd_soc_write(codec, DA7210_A_PLL1, 0x01); - snd_soc_write(codec, DA7210_A_CP_MODE, 0x7C); - snd_soc_write(codec, DA7210_A_HID_UNLOCK, 0x00); /* re-lock */ - snd_soc_write(codec, DA7210_A_TEST_UNLOCK, 0x00); + /* unlock */ + regmap_write(da7210->regmap, DA7210_A_HID_UNLOCK, 0x8B); + regmap_write(da7210->regmap, DA7210_A_TEST_UNLOCK, 0xB4); + regmap_write(da7210->regmap, DA7210_A_PLL1, 0x01); + regmap_write(da7210->regmap, DA7210_A_CP_MODE, 0x7C); + /* re-lock */ + regmap_write(da7210->regmap, DA7210_A_HID_UNLOCK, 0x00); + regmap_write(da7210->regmap, DA7210_A_TEST_UNLOCK, 0x00); /* Activate all enabled subsystem */ snd_soc_write(codec, DA7210_STARTUP1, DA7210_SC_MST_EN); @@ -1000,10 +1133,6 @@ static int da7210_probe(struct snd_soc_codec *codec) static struct snd_soc_codec_driver soc_codec_dev_da7210 = { .probe = da7210_probe, - .reg_cache_size = ARRAY_SIZE(da7210_reg), - .reg_word_size = sizeof(u8), - .reg_cache_default = da7210_reg, - .volatile_register = da7210_volatile_register, .controls = da7210_snd_controls, .num_controls = ARRAY_SIZE(da7210_snd_controls), @@ -1014,6 +1143,17 @@ static struct snd_soc_codec_driver soc_codec_dev_da7210 = { .num_dapm_routes = ARRAY_SIZE(da7210_audio_map), }; +static struct regmap_config da7210_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .reg_defaults = da7210_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(da7210_reg_defaults), + .volatile_reg = da7210_volatile_register, + .readable_reg = da7210_readable_register, + .cache_type = REGCACHE_RBTREE, +}; + #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) static int __devinit da7210_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) @@ -1027,16 +1167,34 @@ static int __devinit da7210_i2c_probe(struct i2c_client *i2c, return -ENOMEM; i2c_set_clientdata(i2c, da7210); - da7210->control_type = SND_SOC_I2C; + + da7210->regmap = regmap_init_i2c(i2c, &da7210_regmap); + if (IS_ERR(da7210->regmap)) { + ret = PTR_ERR(da7210->regmap); + dev_err(&i2c->dev, "regmap_init() failed: %d\n", ret); + return ret; + } ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_da7210, &da7210_dai, 1); + if (ret < 0) { + dev_err(&i2c->dev, "Failed to register codec: %d\n", ret); + goto err_regmap; + } + return ret; + +err_regmap: + regmap_exit(da7210->regmap); + return ret; } static int __devexit da7210_i2c_remove(struct i2c_client *client) { + struct da7210_priv *da7210 = i2c_get_clientdata(client); + snd_soc_unregister_codec(&client->dev); + regmap_exit(da7210->regmap); return 0; } -- 1.7.1

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[alsa-devel] [PATCH] ASoC: wm8996: Remove separate output stage enable step
by Mark Brown 08 Mar '12

08 Mar '12

Marginally improve performance during startup. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8996.c | 31 ++++++++++++------------------- 1 files changed, 12 insertions(+), 19 deletions(-) diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index dd94cd0..a1cfffb 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -890,8 +890,8 @@ static void wm8996_seq_notifier(struct snd_soc_dapm_context *dapm, val = 0; mask = 0; if (wm8996->hpout_pending & HPOUT1L) { - val |= WM8996_HPOUT1L_RMV_SHORT; - mask |= WM8996_HPOUT1L_RMV_SHORT; + val |= WM8996_HPOUT1L_RMV_SHORT | WM8996_HPOUT1L_OUTP; + mask |= WM8996_HPOUT1L_RMV_SHORT | WM8996_HPOUT1L_OUTP; } else { mask |= WM8996_HPOUT1L_RMV_SHORT | WM8996_HPOUT1L_OUTP | @@ -899,8 +899,8 @@ static void wm8996_seq_notifier(struct snd_soc_dapm_context *dapm, } if (wm8996->hpout_pending & HPOUT1R) { - val |= WM8996_HPOUT1R_RMV_SHORT; - mask |= WM8996_HPOUT1R_RMV_SHORT; + val |= WM8996_HPOUT1R_RMV_SHORT | WM8996_HPOUT1R_OUTP; + mask |= WM8996_HPOUT1R_RMV_SHORT | WM8996_HPOUT1R_OUTP; } else { mask |= WM8996_HPOUT1R_RMV_SHORT | WM8996_HPOUT1R_OUTP | @@ -912,8 +912,8 @@ static void wm8996_seq_notifier(struct snd_soc_dapm_context *dapm, val = 0; mask = 0; if (wm8996->hpout_pending & HPOUT2L) { - val |= WM8996_HPOUT2L_RMV_SHORT; - mask |= WM8996_HPOUT2L_RMV_SHORT; + val |= WM8996_HPOUT2L_RMV_SHORT | WM8996_HPOUT2L_OUTP; + mask |= WM8996_HPOUT2L_RMV_SHORT | WM8996_HPOUT2L_OUTP; } else { mask |= WM8996_HPOUT2L_RMV_SHORT | WM8996_HPOUT2L_OUTP | @@ -921,8 +921,8 @@ static void wm8996_seq_notifier(struct snd_soc_dapm_context *dapm, } if (wm8996->hpout_pending & HPOUT2R) { - val |= WM8996_HPOUT2R_RMV_SHORT; - mask |= WM8996_HPOUT2R_RMV_SHORT; + val |= WM8996_HPOUT2R_RMV_SHORT | WM8996_HPOUT2R_OUTP; + mask |= WM8996_HPOUT2R_RMV_SHORT | WM8996_HPOUT2R_OUTP; } else { mask |= WM8996_HPOUT2R_RMV_SHORT | WM8996_HPOUT2R_OUTP | @@ -1214,7 +1214,6 @@ SND_SOC_DAPM_PGA_S("HPOUT2L PGA", 0, WM8996_POWER_MANAGEMENT_1, 7, 0, NULL, 0), SND_SOC_DAPM_PGA_S("HPOUT2L_DLY", 1, WM8996_ANALOGUE_HP_2, 5, 0, NULL, 0), SND_SOC_DAPM_PGA_S("HPOUT2L_DCS", 2, WM8996_DC_SERVO_1, 2, 0, dcs_start, SND_SOC_DAPM_POST_PMU), -SND_SOC_DAPM_PGA_S("HPOUT2L_OUTP", 3, WM8996_ANALOGUE_HP_2, 6, 0, NULL, 0), SND_SOC_DAPM_PGA_S("HPOUT2L_RMV_SHORT", 3, SND_SOC_NOPM, HPOUT2L, 0, rmv_short_event, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD), @@ -1223,7 +1222,6 @@ SND_SOC_DAPM_PGA_S("HPOUT2R PGA", 0, WM8996_POWER_MANAGEMENT_1, 6, 0,NULL, 0), SND_SOC_DAPM_PGA_S("HPOUT2R_DLY", 1, WM8996_ANALOGUE_HP_2, 1, 0, NULL, 0), SND_SOC_DAPM_PGA_S("HPOUT2R_DCS", 2, WM8996_DC_SERVO_1, 3, 0, dcs_start, SND_SOC_DAPM_POST_PMU), -SND_SOC_DAPM_PGA_S("HPOUT2R_OUTP", 3, WM8996_ANALOGUE_HP_2, 2, 0, NULL, 0), SND_SOC_DAPM_PGA_S("HPOUT2R_RMV_SHORT", 3, SND_SOC_NOPM, HPOUT2R, 0, rmv_short_event, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD), @@ -1232,7 +1230,6 @@ SND_SOC_DAPM_PGA_S("HPOUT1L PGA", 0, WM8996_POWER_MANAGEMENT_1, 5, 0, NULL, 0), SND_SOC_DAPM_PGA_S("HPOUT1L_DLY", 1, WM8996_ANALOGUE_HP_1, 5, 0, NULL, 0), SND_SOC_DAPM_PGA_S("HPOUT1L_DCS", 2, WM8996_DC_SERVO_1, 0, 0, dcs_start, SND_SOC_DAPM_POST_PMU), -SND_SOC_DAPM_PGA_S("HPOUT1L_OUTP", 3, WM8996_ANALOGUE_HP_1, 6, 0, NULL, 0), SND_SOC_DAPM_PGA_S("HPOUT1L_RMV_SHORT", 3, SND_SOC_NOPM, HPOUT1L, 0, rmv_short_event, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD), @@ -1434,32 +1431,28 @@ static const struct snd_soc_dapm_route wm8996_dapm_routes[] = { { "HPOUT2L PGA", NULL, "DAC2L" }, { "HPOUT2L_DLY", NULL, "HPOUT2L PGA" }, { "HPOUT2L_DCS", NULL, "HPOUT2L_DLY" }, - { "HPOUT2L_OUTP", NULL, "HPOUT2L_DCS" }, - { "HPOUT2L_RMV_SHORT", NULL, "HPOUT2L_OUTP" }, + { "HPOUT2L_RMV_SHORT", NULL, "HPOUT2L_DCS" }, { "HPOUT2R PGA", NULL, "Charge Pump" }, { "HPOUT2R PGA", NULL, "Bandgap" }, { "HPOUT2R PGA", NULL, "DAC2R" }, { "HPOUT2R_DLY", NULL, "HPOUT2R PGA" }, { "HPOUT2R_DCS", NULL, "HPOUT2R_DLY" }, - { "HPOUT2R_OUTP", NULL, "HPOUT2R_DCS" }, - { "HPOUT2R_RMV_SHORT", NULL, "HPOUT2R_OUTP" }, + { "HPOUT2R_RMV_SHORT", NULL, "HPOUT2R_DCS" }, { "HPOUT1L PGA", NULL, "Charge Pump" }, { "HPOUT1L PGA", NULL, "Bandgap" }, { "HPOUT1L PGA", NULL, "DAC1L" }, { "HPOUT1L_DLY", NULL, "HPOUT1L PGA" }, { "HPOUT1L_DCS", NULL, "HPOUT1L_DLY" }, - { "HPOUT1L_OUTP", NULL, "HPOUT1L_DCS" }, - { "HPOUT1L_RMV_SHORT", NULL, "HPOUT1L_OUTP" }, + { "HPOUT1L_RMV_SHORT", NULL, "HPOUT1L_DCS" }, { "HPOUT1R PGA", NULL, "Charge Pump" }, { "HPOUT1R PGA", NULL, "Bandgap" }, { "HPOUT1R PGA", NULL, "DAC1R" }, { "HPOUT1R_DLY", NULL, "HPOUT1R PGA" }, { "HPOUT1R_DCS", NULL, "HPOUT1R_DLY" }, - { "HPOUT1R_OUTP", NULL, "HPOUT1R_DCS" }, - { "HPOUT1R_RMV_SHORT", NULL, "HPOUT1R_OUTP" }, + { "HPOUT1R_RMV_SHORT", NULL, "HPOUT1R_DCS" }, { "HPOUT2L", NULL, "HPOUT2L_RMV_SHORT" }, { "HPOUT2R", NULL, "HPOUT2R_RMV_SHORT" }, -- 1.7.9.1

1 0

[alsa-devel] [PATCH] USB-Audio: Disable "Logitech USB Headset" iec958 device
by David Henningsson 08 Mar '12

08 Mar '12

Because Logitech USB Headsets do not have any digital devices, prevent them from being opened. This simplifies probing the device in PulseAudio. BugLink: https://bugs.launchpad.net/bugs/940145 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- I did not get any responses to my "digital profile" mail sent to alsa-devel Feb 24th, but it seems the easiest workaround is the below patch for alsa-lib. Alsa-info at: https://bugs.launchpad.net/ubuntu/+source/pulseaudio/+bug/940145/+attachmen… src/conf/cards/USB-Audio.conf | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/src/conf/cards/USB-Audio.conf b/src/conf/cards/USB-Audio.conf index 0726c34..f488284 100644 --- a/src/conf/cards/USB-Audio.conf +++ b/src/conf/cards/USB-Audio.conf @@ -36,6 +36,8 @@ USB-Audio.pcm.surround40_type { # number for the iec958 device can be changed here. USB-Audio.pcm.iec958_device { # "NoiseBlaster 3000" 42 + # Make sure we can't open this device digitally, because it does not exist + "Logitech USB Headset" 999 } -- 1.7.9

4 4

[alsa-devel] [PATCH 0/1] HDSPM: ioctl_compat
by Adrian Knoth 08 Mar '12

08 Mar '12

Hi! There is a one-line patch in my patch-queue that enables 32bit applications that access RME specific ioctls to be run on 64bit kernels. I only had a little test program that worked flawlessly (it solely tested SNDRV_HDSPM_IOCTL_GET_PEAK_RMS), but the IOSONO guys said they've tried the patch and it solves their problem (initial bug report was also from IOSONO). Cheers Adrian Knoth (1): ALSA: hdspm - Provide ioctl_compat sound/pci/rme9652/hdspm.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) -- 1.7.9.1

2 2

[alsa-devel] [PATCH] ASoC: dapm: Allow DAPM registers to be unsigned int
by Mark Brown 08 Mar '12

08 Mar '12

Supports larger register maps. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/soc-dapm.h | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index a4ae018..ab3216b 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -488,7 +488,7 @@ struct snd_soc_dapm_widget { void *priv; /* widget specific data */ /* dapm control */ - short reg; /* negative reg = no direct dapm */ + unsigned int reg; /* negative reg = no direct dapm */ unsigned char shift; /* bits to shift */ unsigned int saved_value; /* widget saved value */ unsigned int value; /* widget current value */ -- 1.7.9.1

3 3

[alsa-devel] [PATCH] ASoC: core: Add platform DAI widget mapping.
by Liam Girdwood 08 Mar '12

08 Mar '12

Add platform driver support for CPU DAI DAPM widgets. Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- include/sound/soc-dai.h | 1 + sound/soc/soc-core.c | 14 ++++++++++++++ 2 files changed, 15 insertions(+), 0 deletions(-) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index c429f24..3248fbc 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -241,6 +241,7 @@ struct snd_soc_dai { struct snd_soc_dapm_widget *playback_widget; struct snd_soc_dapm_widget *capture_widget; + struct snd_soc_dapm_context dapm; /* DAI DMA data */ void *playback_dma_data; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 5aef34a..f466b9d 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1074,6 +1074,7 @@ static int soc_probe_platform(struct snd_soc_card *card, { int ret = 0; const struct snd_soc_platform_driver *driver = platform->driver; + struct snd_soc_dai *dai; platform->card = card; platform->dapm.card = card; @@ -1087,6 +1088,14 @@ static int soc_probe_platform(struct snd_soc_card *card, snd_soc_dapm_new_controls(&platform->dapm, driver->dapm_widgets, driver->num_dapm_widgets); + /* Create DAPM widgets for each DAI stream */ + list_for_each_entry(dai, &dai_list, list) { + if (dai->dev != platform->dev) + continue; + + snd_soc_dapm_new_dai_widgets(&platform->dapm, dai); + } + if (driver->probe) { ret = driver->probe(platform); if (ret < 0) { @@ -1222,9 +1231,12 @@ static int soc_probe_dai_link(struct snd_soc_card *card, int num, int order) /* probe the cpu_dai */ if (!cpu_dai->probed && cpu_dai->driver->probe_order == order) { + cpu_dai->dapm.card = card; if (!try_module_get(cpu_dai->dev->driver->owner)) return -ENODEV; + snd_soc_dapm_new_dai_widgets(&cpu_dai->dapm, cpu_dai); + if (cpu_dai->driver->probe) { ret = cpu_dai->driver->probe(cpu_dai); if (ret < 0) { @@ -3241,6 +3253,7 @@ int snd_soc_register_dai(struct device *dev, dai->dev = dev; dai->driver = dai_drv; + dai->dapm.dev = dev; if (!dai->driver->ops) dai->driver->ops = &null_dai_ops; @@ -3317,6 +3330,7 @@ int snd_soc_register_dais(struct device *dev, dai->id = dai->driver->id; else dai->id = i; + dai->dapm.dev = dev; if (!dai->driver->ops) dai->driver->ops = &null_dai_ops; -- 1.7.5.4

2 5

[alsa-devel] [PATCH] ASoC: dapm: Add API call to query valid DAPM paths.
by Liam Girdwood 08 Mar '12

08 Mar '12

Add a DAPM API call to determine whether a DAPM audio path is valid between source and sink widgets. This also takes into account all kcontrol mux and mixer settings in between the source and sink widgets to validate the audio path. This will be used by the Dynamic PCM core to determine the runtime DAI mappings between FE and BE DAIs in order to run PCM operations. This also adds some extra debug to help with graph debugging. Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- include/sound/soc-dapm.h | 5 ++ sound/soc/soc-dapm.c | 127 ++++++++++++++++++++++++++++++++++++++++++---- 2 files changed, 122 insertions(+), 10 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index d3e45a0..8d58880 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -323,6 +323,7 @@ struct snd_soc_dapm_path; struct snd_soc_dapm_pin; struct snd_soc_dapm_route; struct snd_soc_dapm_context; +struct snd_soc_dapm_widget_list; int dapm_reg_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event); @@ -401,6 +402,10 @@ void snd_soc_dapm_auto_nc_codec_pins(struct snd_soc_codec *codec); /* Mostly internal - should not normally be used */ void dapm_mark_dirty(struct snd_soc_dapm_widget *w, const char *reason); +/* dapm path query */ +int snd_soc_dapm_dai_get_connected_widgets(struct snd_soc_dai *dai, int stream, + struct snd_soc_dapm_widget_list **list); + /* dapm widget types */ enum snd_soc_dapm_type { snd_soc_dapm_input = 0, /* input pin */ diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 8320fd7..c7928db 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -703,11 +703,51 @@ static int snd_soc_dapm_suspend_check(struct snd_soc_dapm_widget *widget) } } +/* add widget to list if it's not already in the list */ +static int dapm_list_add_widget(struct snd_soc_dapm_widget_list **list, + struct snd_soc_dapm_widget *w) +{ + struct snd_soc_dapm_widget_list *wlist; + int wlistsize, wlistentries, i; + + if (*list == NULL) + return -EINVAL; + + wlist = *list; + + /* is this widget already in the list */ + for (i = 0; i < wlist->num_widgets; i++) { + if (wlist->widgets[i] == w) + return 0; + } + + /* allocate some new space */ + wlistentries = wlist->num_widgets + 1; + wlistsize = sizeof(struct snd_soc_dapm_widget_list) + + wlistentries * sizeof(struct snd_soc_dapm_widget *); + *list = krealloc(wlist, wlistsize, GFP_KERNEL); + if (*list == NULL) { + dev_err(w->dapm->dev, "can't allocate widget list for %s\n", + w->name); + return -ENOMEM; + } + wlist = *list; + + /* insert the widget */ + dev_dbg(w->dapm->dev, "added %s in widget list pos %d\n", + w->name, wlist->num_widgets); + + wlist->widgets[wlist->num_widgets] = w; + wlist->num_widgets++; + return 1; +} + /* * Recursively check for a completed path to an active or physically connected * output widget. Returns number of complete paths. */ -static int is_connected_output_ep(struct snd_soc_dapm_widget *widget) +static int is_connected_output_ep(struct snd_soc_dapm_widget *widget, + struct snd_soc_dapm_widget_list **list) { struct snd_soc_dapm_path *path; int con = 0; @@ -763,9 +803,26 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget) if (path->walked) continue; + dev_vdbg(widget->dapm->dev," %c : %s -> %s -> %s\n", + path->sink && path->connect ? '*' : ' ', + widget->name, path->name, path->sink->name); + if (path->sink && path->connect) { path->walked = 1; - con += is_connected_output_ep(path->sink); + + /* do we need to add this widget to the list ? */ + if (list) { + int err; + err = dapm_list_add_widget(list, path->sink); + if (err < 0) { + dev_err(widget->dapm->dev, + "could not add widget %s\n", + widget->name); + return con; + } + } + + con += is_connected_output_ep(path->sink, list); } } @@ -778,7 +835,8 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget) * Recursively check for a completed path to an active or physically connected * input widget. Returns number of complete paths. */ -static int is_connected_input_ep(struct snd_soc_dapm_widget *widget) +static int is_connected_input_ep(struct snd_soc_dapm_widget *widget, + struct snd_soc_dapm_widget_list **list) { struct snd_soc_dapm_path *path; int con = 0; @@ -846,9 +904,26 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget) if (path->walked) continue; + dev_vdbg(widget->dapm->dev," %c : %s <- %s <- %s\n", + path->source && path->connect ? '*' : ' ', + widget->name, path->name, path->source->name); + if (path->source && path->connect) { path->walked = 1; - con += is_connected_input_ep(path->source); + + /* do we need to add this widget to the list ? */ + if (list) { + int err; + err = dapm_list_add_widget(list, path->sink); + if (err < 0) { + dev_err(widget->dapm->dev, + "could not add widget %s\n", + widget->name); + return con; + } + } + + con += is_connected_input_ep(path->source, list); } } @@ -857,6 +932,38 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget) return con; } +/** + * snd_soc_dapm_get_connected_widgets - query audio path and it's widgets. + * @dai: the soc DAI. + * @stream: stream direction. + * @list: list of active widgets for this stream. + * + * Queries DAPM graph as to whether an valid audio stream path exists for + * the initial stream specified by name. This takes into account + * current mixer and mux kcontrol settings. Creates list of valid widgets. + * + * Returns the number of valid paths or negative error. + */ +int snd_soc_dapm_dai_get_connected_widgets(struct snd_soc_dai *dai, int stream, + struct snd_soc_dapm_widget_list **list) +{ + struct snd_soc_card *card = dai->card; + int paths; + + dapm_reset(card); + + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + paths = is_connected_output_ep(dai->playback_widget, list); + else + paths = is_connected_input_ep(dai->playback_widget, list); + + dev_dbg(dai->dev, "%s: found %d paths\n", + stream ? "capture" : "playback", paths); + dapm_clear_walk(&card->dapm); + + return paths; +} + /* * Handler for generic register modifier widget. */ @@ -913,9 +1020,9 @@ static int dapm_generic_check_power(struct snd_soc_dapm_widget *w) DAPM_UPDATE_STAT(w, power_checks); - in = is_connected_input_ep(w); + in = is_connected_input_ep(w, NULL); dapm_clear_walk(w->dapm); - out = is_connected_output_ep(w); + out = is_connected_output_ep(w, NULL); dapm_clear_walk(w->dapm); return out != 0 && in != 0; } @@ -935,7 +1042,7 @@ static int dapm_adc_check_power(struct snd_soc_dapm_widget *w) DAPM_UPDATE_STAT(w, power_checks); if (w->active) { - in = is_connected_input_ep(w); + in = is_connected_input_ep(w, NULL); dapm_clear_walk(w->dapm); return in != 0; } else { @@ -951,7 +1058,7 @@ static int dapm_dac_check_power(struct snd_soc_dapm_widget *w) DAPM_UPDATE_STAT(w, power_checks); if (w->active) { - out = is_connected_output_ep(w); + out = is_connected_output_ep(w, NULL); dapm_clear_walk(w->dapm); return out != 0; } else { @@ -1587,9 +1694,9 @@ static ssize_t dapm_widget_power_read_file(struct file *file, if (!buf) return -ENOMEM; - in = is_connected_input_ep(w); + in = is_connected_input_ep(w, NULL); dapm_clear_walk(w->dapm); - out = is_connected_output_ep(w); + out = is_connected_output_ep(w, NULL); dapm_clear_walk(w->dapm); ret = snprintf(buf, PAGE_SIZE, "%s: %s%s in %d out %d", -- 1.7.5.4

2 3

[alsa-devel] [PATCH] ASoC: omap-mcbsp: Correct clock muxing for CLKR/FSR signals
by Peter Ujfalusi 08 Mar '12

08 Mar '12

Remove the no longer valid check for McBSP1 regarding to signal mux selection (on OMAP4 McBSP4 has 6 pin setup). Only clear the srgr2, pcr0 register configuration if the requested clock configuration will actually going to touch it. In this way we can avoid issues if the CLKR/FSR mux has been configured after the clock selection. We are going to check for the valid McBSP port in the omap_mcbsp_6pin_src_mux() function based on the validity of the mux_signal callback (which is only valid for ports having 6 pin setup). Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- Hello, this patch fixes the remaining issue with the CLKR/FSR mux code. The code in omap-mcbsp.c was checking for McBSP1 to let the mux to be configured, which is no longer valid. In omap-mcbsp.c we are only checking for OMAP version (OMAP1 does not have 6 pin setup), and let the underlying code handle the validity of the McBSP port. This patch depends on my previous CLKR/FSR series. Regards, Peter sound/soc/omap/mcbsp.c | 9 ++++----- sound/soc/omap/omap-mcbsp.c | 34 +++++++++++++++------------------- 2 files changed, 19 insertions(+), 24 deletions(-) diff --git a/sound/soc/omap/mcbsp.c b/sound/soc/omap/mcbsp.c index 95413a1..e5f4444 100644 --- a/sound/soc/omap/mcbsp.c +++ b/sound/soc/omap/mcbsp.c @@ -690,7 +690,9 @@ int omap2_mcbsp_set_clks_src(struct omap_mcbsp *mcbsp, u8 fck_src_id) int omap_mcbsp_6pin_src_mux(struct omap_mcbsp *mcbsp, u8 mux) { const char *signal, *src; - int ret = 0; + + if (mcbsp->pdata->mux_signal) + return -EINVAL; switch (mux) { case CLKR_SRC_CLKR: @@ -713,10 +715,7 @@ int omap_mcbsp_6pin_src_mux(struct omap_mcbsp *mcbsp, u8 mux) return -EINVAL; } - if (mcbsp->pdata->mux_signal) - ret = mcbsp->pdata->mux_signal(mcbsp->dev, signal, src); - - return ret; + return mcbsp->pdata->mux_signal(mcbsp->dev, signal, src); } #define max_thres(m) (mcbsp->pdata->buffer_size) diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index d8409b0..626e2d6 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -511,17 +511,21 @@ static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai, return -EBUSY; } - /* The McBSP signal muxing functions are only available on McBSP1 */ - if (clk_id == OMAP_MCBSP_CLKR_SRC_CLKR || - clk_id == OMAP_MCBSP_CLKR_SRC_CLKX || - clk_id == OMAP_MCBSP_FSR_SRC_FSR || - clk_id == OMAP_MCBSP_FSR_SRC_FSX) - if (cpu_class_is_omap1() || cpu_dai->id != 1) - return -EINVAL; - - mcbsp->in_freq = freq; - regs->srgr2 &= ~CLKSM; - regs->pcr0 &= ~SCLKME; + if (clk_id == OMAP_MCBSP_SYSCLK_CLK || + clk_id == OMAP_MCBSP_SYSCLK_CLKS_FCLK || + clk_id == OMAP_MCBSP_SYSCLK_CLKS_EXT || + clk_id == OMAP_MCBSP_SYSCLK_CLKX_EXT || + clk_id == OMAP_MCBSP_SYSCLK_CLKR_EXT) { + mcbsp->in_freq = freq; + regs->srgr2 &= ~CLKSM; + regs->pcr0 &= ~SCLKME; + } else if (cpu_class_is_omap1()) { + /* + * McBSP CLKR/FSR signal muxing functions are only available on + * OMAP2 or newer versions + */ + return -EINVAL; + } switch (clk_id) { case OMAP_MCBSP_SYSCLK_CLK: @@ -552,23 +556,15 @@ static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai, case OMAP_MCBSP_CLKR_SRC_CLKR: - if (cpu_class_is_omap1()) - break; err = omap_mcbsp_6pin_src_mux(mcbsp, CLKR_SRC_CLKR); break; case OMAP_MCBSP_CLKR_SRC_CLKX: - if (cpu_class_is_omap1()) - break; err = omap_mcbsp_6pin_src_mux(mcbsp, CLKR_SRC_CLKX); break; case OMAP_MCBSP_FSR_SRC_FSR: - if (cpu_class_is_omap1()) - break; err = omap_mcbsp_6pin_src_mux(mcbsp, FSR_SRC_FSR); break; case OMAP_MCBSP_FSR_SRC_FSX: - if (cpu_class_is_omap1()) - break; err = omap_mcbsp_6pin_src_mux(mcbsp, FSR_SRC_FSX); break; default: -- 1.7.8.5

2 1

[alsa-devel] [PATCH 1/2] mfd: wm8994: Remove unused LDO supply field from WM8994 pdata
by Mark Brown 08 Mar '12

08 Mar '12

It's causing confusion with the regulator level field of the same name and serves no useful function. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- arch/arm/mach-s3c64xx/mach-crag6410-module.c | 6 ++++-- include/linux/mfd/wm8994/pdata.h | 1 - 2 files changed, 4 insertions(+), 3 deletions(-) diff --git a/arch/arm/mach-s3c64xx/mach-crag6410-module.c b/arch/arm/mach-s3c64xx/mach-crag6410-module.c index e248581..77c8342 100644 --- a/arch/arm/mach-s3c64xx/mach-crag6410-module.c +++ b/arch/arm/mach-s3c64xx/mach-crag6410-module.c @@ -17,6 +17,8 @@ #include <linux/mfd/wm831x/gpio.h> #include <linux/mfd/wm8994/pdata.h> +#include <linux/regulator/machine.h> + #include <sound/wm0010.h> #include <sound/wm5100.h> #include <sound/wm8996.h> @@ -168,8 +170,8 @@ static struct wm8994_pdata wm8994_pdata = { }, .irq_base = CODEC_IRQ_BASE, .ldo = { - { .supply = "WALLVDD" }, - { .supply = "WALLVDD" }, + { }, + { }, }, }; diff --git a/include/linux/mfd/wm8994/pdata.h b/include/linux/mfd/wm8994/pdata.h index dc3e050..893267b 100644 --- a/include/linux/mfd/wm8994/pdata.h +++ b/include/linux/mfd/wm8994/pdata.h @@ -22,7 +22,6 @@ struct wm8994_ldo_pdata { /** GPIOs to enable regulator, 0 or less if not available */ int enable; - const char *supply; const struct regulator_init_data *init_data; }; -- 1.7.9.1

1 1

[alsa-devel] [PATCH 1/3] ASoC: core: Add card mutex locking subclasses
by Liam Girdwood 08 Mar '12

08 Mar '12

This is the first part of a change that is intended to improve ASoC locking protection for DAPM and PCM operations. This part of the series adds a mutex class for the soc_card mutex. The SND_SOC_CARD_CLASS_INIT class is used for card initialisation only whilst the SND_SOC_CARD_CLASS_PCM class is used for the forth coming Dynamic PCM operations. The new mutex classes are required otherwise we will see a false positive mutex deadlock warning between the card initialisation and the PCM operations (something that would never deadlock in real life). Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- include/sound/soc.h | 5 +++++ sound/soc/soc-core.c | 2 +- 2 files changed, 6 insertions(+), 1 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 2ebf787..70de2f8 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -288,6 +288,11 @@ enum snd_soc_pcm_subclass { SND_SOC_PCM_CLASS_BE = 1, }; +enum snd_soc_card_subclass { + SND_SOC_CARD_CLASS_INIT = 0, + SND_SOC_CARD_CLASS_PCM = 1, +}; + int snd_soc_codec_set_sysclk(struct snd_soc_codec *codec, int clk_id, int source, unsigned int freq, int dir); int snd_soc_codec_set_pll(struct snd_soc_codec *codec, int pll_id, int source, diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index c90bb01..acd0d93 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1416,7 +1416,7 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) struct snd_soc_dai_link *dai_link; int ret, i, order; - mutex_lock(&card->mutex); + mutex_lock_nested(&card->mutex, SND_SOC_CARD_CLASS_INIT); if (card->instantiated) { mutex_unlock(&card->mutex); -- 1.7.5.4

2 6

Re: [alsa-devel] Random process lockup on ARM board: alsa-lib-1.0.25, FUTEX_WAIT_PRIVATE
by Clemens Ladisch 08 Mar '12

08 Mar '12

Jonathan Andrews wrote: > On Wed, 2012-02-22 at 17:23 +0100, Clemens Ladisch wrote: >> Now it doesn't use any pthread functions. > > Thanks, use or require ? Still complains if I static compile my > application but don't link with pthreads. Please try with the patch below; this should remove the last occurence of pthreads in alsa-lib. Regards, Clemens --- alsa-lib/configure.in +++ alsa-lib/configure.in @@ -512,6 +512,7 @@ if test "$HAVE_LIBDL" != "yes"; then fi if test "$HAVE_LIBPTHREAD" != "yes"; then + build_pcm_meter="no" build_pcm_share="no" fi

1 0

[alsa-devel] [PATCH] ASoC: omap McBSP: Clear rx_irq at probe time for OMAP4
by Peter Ujfalusi 08 Mar '12

08 Mar '12

On OMAP4 we have one interrupt line per McBSP port. At proble time tx, and rx irq value will be -ENXIO, and only the tx irq will get corrected. In omap_mcbsp_request if the rx_irq is not 0 we proceed, and try to request the interrupt, which will fail on OMAP4 (rx_irq == -6). To avoid this error, clear the rx_irq at probe time on OMAP4. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/omap/mcbsp.c | 4 +++- 1 files changed, 3 insertions(+), 1 deletions(-) diff --git a/sound/soc/omap/mcbsp.c b/sound/soc/omap/mcbsp.c index 21dbb05..c3e31de 100644 --- a/sound/soc/omap/mcbsp.c +++ b/sound/soc/omap/mcbsp.c @@ -958,8 +958,10 @@ int __devinit omap_mcbsp_init(struct platform_device *pdev) mcbsp->rx_irq = platform_get_irq_byname(pdev, "rx"); /* From OMAP4 there will be a single irq line */ - if (mcbsp->tx_irq == -ENXIO) + if (mcbsp->tx_irq == -ENXIO) { mcbsp->tx_irq = platform_get_irq(pdev, 0); + mcbsp->rx_irq = 0; + } res = platform_get_resource_byname(pdev, IORESOURCE_DMA, "rx"); if (!res) { -- 1.7.8.5

4 3

[alsa-devel] [PATCH] ASoC: dapm: Add platform stream event support.
by Liam Girdwood 07 Mar '12

07 Mar '12

Currently stream events are only perfomed on codec stream widgets only. There is now a need to be able to perform stream events on platform widgets too. e.g. we have the ABE platform driver with several DAI links to dummy codecs. We need to be able to perform stream events on any of the dummy codec DAI links. This patch also removes the snd_soc_dai * parameter since it's already contained within the rtd * parameter. Finally makle stream event return void since no one checks it anyway. Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- include/sound/soc-dapm.h | 4 +- sound/soc/soc-core.c | 8 +--- sound/soc/soc-dapm.c | 79 +++++++++++++++++++++++++++++---------------- sound/soc/soc-pcm.c | 9 ++--- 4 files changed, 59 insertions(+), 41 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 8cd3c74..d3e45a0 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -368,8 +368,8 @@ int snd_soc_dapm_weak_routes(struct snd_soc_dapm_context *dapm, const struct snd_soc_dapm_route *route, int num); /* dapm events */ -int snd_soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd, int stream, - struct snd_soc_dai *dai, int event); +void snd_soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd, int stream, + int event); void snd_soc_dapm_shutdown(struct snd_soc_card *card); /* external DAPM widget events */ diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index f466b9d..cc9f1f1 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -573,19 +573,16 @@ int snd_soc_suspend(struct device *dev) } for (i = 0; i < card->num_rtd; i++) { - struct snd_soc_dai *codec_dai = card->rtd[i].codec_dai; if (card->rtd[i].dai_link->ignore_suspend) continue; snd_soc_dapm_stream_event(&card->rtd[i], SNDRV_PCM_STREAM_PLAYBACK, - codec_dai, SND_SOC_DAPM_STREAM_SUSPEND); snd_soc_dapm_stream_event(&card->rtd[i], SNDRV_PCM_STREAM_CAPTURE, - codec_dai, SND_SOC_DAPM_STREAM_SUSPEND); } @@ -689,17 +686,16 @@ static void soc_resume_deferred(struct work_struct *work) } for (i = 0; i < card->num_rtd; i++) { - struct snd_soc_dai *codec_dai = card->rtd[i].codec_dai; if (card->rtd[i].dai_link->ignore_suspend) continue; snd_soc_dapm_stream_event(&card->rtd[i], - SNDRV_PCM_STREAM_PLAYBACK, codec_dai, + SNDRV_PCM_STREAM_PLAYBACK, SND_SOC_DAPM_STREAM_RESUME); snd_soc_dapm_stream_event(&card->rtd[i], - SNDRV_PCM_STREAM_CAPTURE, codec_dai, + SNDRV_PCM_STREAM_CAPTURE, SND_SOC_DAPM_STREAM_RESUME); } diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index d15a703..8320fd7 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3019,37 +3019,61 @@ int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card) return 0; } -static void soc_dapm_stream_event(struct snd_soc_dapm_context *dapm, - int stream, struct snd_soc_dai *dai, - int event) +static void soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd, int stream, + int event) { - struct snd_soc_dapm_widget *w; - if (stream == SNDRV_PCM_STREAM_PLAYBACK) - w = dai->playback_widget; - else - w = dai->capture_widget; + struct snd_soc_dapm_widget *w_cpu, *w_codec; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; - if (!w) - return; + if (stream == SNDRV_PCM_STREAM_PLAYBACK) { + w_cpu = cpu_dai->playback_widget; + w_codec = codec_dai->playback_widget; + } else { + w_cpu = cpu_dai->capture_widget; + w_codec = codec_dai->capture_widget; + } - dapm_mark_dirty(w, "stream event"); + if (w_cpu) { - switch (event) { - case SND_SOC_DAPM_STREAM_START: - w->active = 1; - break; - case SND_SOC_DAPM_STREAM_STOP: - w->active = 0; - break; - case SND_SOC_DAPM_STREAM_SUSPEND: - case SND_SOC_DAPM_STREAM_RESUME: - case SND_SOC_DAPM_STREAM_PAUSE_PUSH: - case SND_SOC_DAPM_STREAM_PAUSE_RELEASE: - break; + dapm_mark_dirty(w_cpu, "stream event"); + + switch (event) { + case SND_SOC_DAPM_STREAM_START: + w_cpu->active = 1; + break; + case SND_SOC_DAPM_STREAM_STOP: + w_cpu->active = 0; + break; + case SND_SOC_DAPM_STREAM_SUSPEND: + case SND_SOC_DAPM_STREAM_RESUME: + case SND_SOC_DAPM_STREAM_PAUSE_PUSH: + case SND_SOC_DAPM_STREAM_PAUSE_RELEASE: + break; + } + } + + if (w_codec) { + + dapm_mark_dirty(w_codec, "stream event"); + + switch (event) { + case SND_SOC_DAPM_STREAM_START: + w_codec->active = 1; + break; + case SND_SOC_DAPM_STREAM_STOP: + w_codec->active = 0; + break; + case SND_SOC_DAPM_STREAM_SUSPEND: + case SND_SOC_DAPM_STREAM_RESUME: + case SND_SOC_DAPM_STREAM_PAUSE_PUSH: + case SND_SOC_DAPM_STREAM_PAUSE_RELEASE: + break; + } } - dapm_power_widgets(dapm, event); + dapm_power_widgets(&rtd->card->dapm, event); } /** @@ -3063,15 +3087,14 @@ static void soc_dapm_stream_event(struct snd_soc_dapm_context *dapm, * * Returns 0 for success else error. */ -int snd_soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd, int stream, - struct snd_soc_dai *dai, int event) +void snd_soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd, int stream, + int event) { struct snd_soc_card *card = rtd->card; mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_PCM); - soc_dapm_stream_event(&card->dapm, stream, dai, event); + soc_dapm_stream_event(rtd, stream, event); mutex_unlock(&card->dapm_mutex); - return 0; } /** diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 0ad8dca..26a60b4 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -308,7 +308,7 @@ static void close_delayed_work(struct work_struct *work) if (codec_dai->pop_wait == 1) { codec_dai->pop_wait = 0; snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_PLAYBACK, - codec_dai, SND_SOC_DAPM_STREAM_STOP); + SND_SOC_DAPM_STREAM_STOP); } mutex_unlock(&rtd->pcm_mutex); @@ -373,7 +373,6 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) /* powered down playback stream now */ snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_PLAYBACK, - codec_dai, SND_SOC_DAPM_STREAM_STOP); } else { /* start delayed pop wq here for playback streams */ @@ -384,7 +383,7 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) } else { /* capture streams can be powered down now */ snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_CAPTURE, - codec_dai, SND_SOC_DAPM_STREAM_STOP); + SND_SOC_DAPM_STREAM_STOP); } mutex_unlock(&rtd->pcm_mutex); @@ -453,8 +452,8 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) cancel_delayed_work(&rtd->delayed_work); } - snd_soc_dapm_stream_event(rtd, substream->stream, codec_dai, - SND_SOC_DAPM_STREAM_START); + snd_soc_dapm_stream_event(rtd, substream->stream, + SND_SOC_DAPM_STREAM_START); snd_soc_dai_digital_mute(codec_dai, 0); -- 1.7.5.4

2 1

[alsa-devel] [PATCH] ASoC: imx: imx-audmux: Fix section mismatch
by Fabio Estevam 07 Mar '12

07 Mar '12

Fix the following section mismatch warning: WARNING: vmlinux.o(.data+0x35be8): Section mismatch in reference from the variable imx_audmux_driver to the function .init.text:imx_audmux_probe() The variable imx_audmux_driver references the function __init imx_audmux_probe() If the reference is valid then annotate the variable with __init* or __refdata (see linux/init.h) or name the variable: *_template, *_timer, *_sht, *_ops, *_probe, *_probe_one, *_console Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/imx/imx-audmux.c | 8 ++++---- 1 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/imx/imx-audmux.c b/sound/soc/imx/imx-audmux.c index ef425ad..601df80 100644 --- a/sound/soc/imx/imx-audmux.c +++ b/sound/soc/imx/imx-audmux.c @@ -167,7 +167,7 @@ static void __init audmux_debugfs_init(void) } } -static void __exit audmux_debugfs_remove(void) +static void __devexit audmux_debugfs_remove(void) { debugfs_remove_recursive(audmux_debugfs_root); } @@ -249,7 +249,7 @@ int imx_audmux_v2_configure_port(unsigned int port, unsigned int ptcr, } EXPORT_SYMBOL_GPL(imx_audmux_v2_configure_port); -static int __init imx_audmux_probe(struct platform_device *pdev) +static int __devinit imx_audmux_probe(struct platform_device *pdev) { struct resource *res; const struct of_device_id *of_id = @@ -276,7 +276,7 @@ static int __init imx_audmux_probe(struct platform_device *pdev) return 0; } -static int __exit imx_audmux_remove(struct platform_device *pdev) +static int __devexit imx_audmux_remove(struct platform_device *pdev) { if (audmux_type == IMX31_AUDMUX) audmux_debugfs_remove(); @@ -287,7 +287,7 @@ static int __exit imx_audmux_remove(struct platform_device *pdev) static struct platform_driver imx_audmux_driver = { .probe = imx_audmux_probe, - .remove = __exit_p(imx_audmux_remove), + .remove = __devexit_p(imx_audmux_remove), .id_table = imx_audmux_ids, .driver = { .name = DRIVER_NAME, -- 1.7.1

2 1

[alsa-devel] ASOC - Codecs : Renaming of spdif_tranceiver.c
by Nitin PAI 07 Mar '12

07 Mar '12

Hi, In my project I am feeling of need to have generic_codec. This will provide the required codec_dais for the usecases like -HDMI/SPDIF where there is no codec involved. -For audio over MOST, where CPLD/glue logic is used -For complex audio DSPs connected which will be mostly programmed in the user-space. -SOC manufactures who would like to ship their drivers without having to know what will be the final codec that will be present in the system. Currently there is spdif_tranceiver.c which serves the purpose for spdif, but using this for other usecases will create confusion while shipping the code. What is the best way to carry forward? 1) Use spdif_tranceiver.c for all the purpose. disadvantage: this will create more confusion while shipping. 2)Deprecate spdif_transciever.c and call it generic_codec. Change the dai names and the function names. 3)Create another generic_codec.c and keep the spdif_transceiver.c and deprecate the latter in next version. 4) Any other ideas ? Thanks, Nitin

2 14

[alsa-devel] [PATCH] ASoC: imx: imx-audmux: Fix section mismatch
by Fabio Estevam 07 Mar '12

07 Mar '12

Fix the following section mismatch warning: WARNING: vmlinux.o(.data+0x35be8): Section mismatch in reference from the variable imx_audmux_driver to the function .init.text:imx_audmux_probe() The variable imx_audmux_driver references the function __init imx_audmux_probe() If the reference is valid then annotate the variable with __init* or __refdata (see linux/init.h) or name the variable: *_template, *_timer, *_sht, *_ops, *_probe, *_probe_one, *_console Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/imx/imx-audmux.c | 8 ++++---- 1 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/imx/imx-audmux.c b/sound/soc/imx/imx-audmux.c index ef425ad..601df80 100644 --- a/sound/soc/imx/imx-audmux.c +++ b/sound/soc/imx/imx-audmux.c @@ -167,7 +167,7 @@ static void __init audmux_debugfs_init(void) } } -static void __exit audmux_debugfs_remove(void) +static void __devexit audmux_debugfs_remove(void) { debugfs_remove_recursive(audmux_debugfs_root); } @@ -249,7 +249,7 @@ int imx_audmux_v2_configure_port(unsigned int port, unsigned int ptcr, } EXPORT_SYMBOL_GPL(imx_audmux_v2_configure_port); -static int __init imx_audmux_probe(struct platform_device *pdev) +static int __devinit imx_audmux_probe(struct platform_device *pdev) { struct resource *res; const struct of_device_id *of_id = @@ -276,7 +276,7 @@ static int __init imx_audmux_probe(struct platform_device *pdev) return 0; } -static int __exit imx_audmux_remove(struct platform_device *pdev) +static int __devexit imx_audmux_remove(struct platform_device *pdev) { if (audmux_type == IMX31_AUDMUX) audmux_debugfs_remove(); @@ -287,7 +287,7 @@ static int __exit imx_audmux_remove(struct platform_device *pdev) static struct platform_driver imx_audmux_driver = { .probe = imx_audmux_probe, - .remove = __exit_p(imx_audmux_remove), + .remove = __devexit_p(imx_audmux_remove), .id_table = imx_audmux_ids, .driver = { .name = DRIVER_NAME, -- 1.7.1

1 0

[alsa-devel] [PATCH v2] ASoC: DAPM - Use dev_warn for debugfs warning message.
by Liam Girdwood 07 Mar '12

07 Mar '12

Remove printk(KERN_WARNING) and use dev_warn() instead. Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- sound/soc/soc-dapm.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index a4d4aa1..dcd1160 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1667,7 +1667,7 @@ void snd_soc_dapm_debugfs_init(struct snd_soc_dapm_context *dapm, dapm->debugfs_dapm = debugfs_create_dir("dapm", parent); if (!dapm->debugfs_dapm) { - printk(KERN_WARNING + dev_warn(dapm->dev, "Failed to create DAPM debugfs directory\n"); return; } -- 1.7.5.4

3 4

[alsa-devel] [PATCH] ASoC: dapm: Only lock CODEC for I/O if not using regmap
by Mark Brown 07 Mar '12

07 Mar '12

If we do use regmap then regmap will take care of things for us. We actually already have this check at a higher level for the current users but this makes sure we do the right thing in the future too if we need to. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-dapm.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 6f4ea12b..25a395e 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -208,7 +208,7 @@ static int soc_widget_write(struct snd_soc_dapm_widget *w, int reg, int val) static inline void soc_widget_lock(struct snd_soc_dapm_widget *w) { - if (w->codec) + if (w->codec && !w->codec->using_regmap) mutex_lock(&w->codec->mutex); else if (w->platform) mutex_lock(&w->platform->mutex); @@ -216,7 +216,7 @@ static inline void soc_widget_lock(struct snd_soc_dapm_widget *w) static inline void soc_widget_unlock(struct snd_soc_dapm_widget *w) { - if (w->codec) + if (w->codec && !w->codec->using_regmap) mutex_unlock(&w->codec->mutex); else if (w->platform) mutex_unlock(&w->platform->mutex); -- 1.7.9.1

2 1

[alsa-devel] [RFC 0/7] ASoC: Introduce dmaengine pcm helper functions
by Lars-Peter Clausen 07 Mar '12

07 Mar '12

Same series as send out yesterday (sans the cleanup patches which already have been applied), since I forgot to Cc alsa-devel. Sorry for the noice if you got the series twice. This patch series introduces a new set of helper functions for dmaengine based PCM drivers. Currently we have 6 different dmaengine based PCM drivers which are all more or less similar in structure, but all have also individual requirements which can't easily be generalized. So that's why this series does not introduce a generic dmaengine PCM driver but rather a set of helper functions which can be used to implement the common bits between such dmaengine based PCM drivers. The helper functions only provide infrastructure code for dmaengines which provide thed evice_prep_dma_cyclic. While it is possible to add support for dmaengine, which only provide device_prep_slave_sg, it is in my opinion a better idea to either implement the missing functionality in the dmaengine driver or provide a emulation layer for cyclic transfer within the dmaengine framework itself. This has the advantage that it is also available outside of ALSA. This series converts the three PCM drivers which already use the circular dmaengine prepare function. Since I don't have access to hardware using the converted drivers the patches have only been compile tested. But the helper functions introduced in this series have been runtime tested with a platform which is not upstream yet. The first two patches restructure the iMX and MXS drivers to request their DMA channel in the PCM open callback instead of the hwparams callback. This is done to make their structure more similar to what it will be when using the helper functions and make the actual patch converting them to the new dmaengine PCM helper functions less intrusive. The next patch introduces the helper functions and the last three patches respectively convert the imx, mxs and ep39xx dmaengine based PCM drivers to use the new set of helper functions. The set of helper functions consists of a open, a close, a trigger and a pointer callback and also a function to convert hwparams to a dma_slave_config. The trigger and pointer callbacks can usually be used directly for the pcm_ops callbacks. The open callback wants to be called from a driver specific open function which may do some driver specific setup tasks and provide the helper open callback with a dma filter function. If the driver specific open functions allocates resources they need to be freed in a driver specific close function which has to call the helper close callback. If not the helper close callback can be assigned directly to the pcm_ops struct. If a driver needs to keep additional driver specific data around it can be done by using snd_dmaengine_pcm_{get,set}_data. Normally a driver should not require to keep additional data around, but all of the converted drivers need this, because of this horrible abusive pattern where the dma channels private field gets assigned configuration data in the filter callback. We should hopefully be able to get rid of this pattern with Guennadi Liakhovetski's patches[1], which add a slave configuration parameter to dma_request_channel, and with it the need for storing extra private data. While the DMA helper functions only use generic ALSA functions and nothing ASoC specific I've placed the, under ASoC for now since the only users are ASoC driver. - Lars [1] https://lkml.org/lkml/2012/2/1/227 Lars-Peter Clausen (7): ASoC: imx-ssi: Set dma data early ASoC: imx-pcm: Request DMA channel early ASoC: mxs-pcm: Request DMA channel early ASoC: Add dmaengine PCM helper functions ASoC: imx-pcm-dma: Use dmaengine PCM helper functions ASoC: mxs-pcm: Use dmaengine PCM helper functions ASoC: ep93xx-pcm: Use dmaengine PCM helper functions include/sound/dmaengine_pcm.h | 49 +++++++ sound/soc/Kconfig | 3 + sound/soc/Makefile | 3 + sound/soc/ep93xx/Kconfig | 1 + sound/soc/ep93xx/ep93xx-pcm.c | 148 +++----------------- sound/soc/imx/Kconfig | 1 + sound/soc/imx/imx-pcm-dma-mx2.c | 196 ++++---------------------- sound/soc/imx/imx-ssi.c | 28 +++- sound/soc/mxs/Kconfig | 1 + sound/soc/mxs/mxs-pcm.c | 152 ++++----------------- sound/soc/mxs/mxs-pcm.h | 12 -- sound/soc/soc-dmaengine-pcm.c | 287 +++++++++++++++++++++++++++++++++++++++ 12 files changed, 443 insertions(+), 438 deletions(-) create mode 100644 include/sound/dmaengine_pcm.h create mode 100644 sound/soc/soc-dmaengine-pcm.c -- 1.7.9

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[alsa-devel] [Regression] Loss of OPL3 FM Synth in alsa-driver-1.0.25
by Raymond Yau 07 Mar '12

07 Mar '12

alsa-tools/seq/sbiload still work but Loss of OPL3 FM synth for ymf724f in alsa-driver-1.0.25 The OPL3 FM Synth works In alsa-driver-1.0.24 aplaymidi -l Port Client name Port name 14:0 Midi Through Midi Through Port-0 17:0 OPL3 FM synth OPL3 FM Port lsmod Module Size Used by snd_hrtimer 6664 1 snd_opl3_synth 14852 1 snd_seq_midi_emul 9088 1 snd_opl3_synth fuse 49436 2 i915 53764 2 drm 158388 3 i915 vboxnetflt 74200 0 vboxdrv 98632 1 vboxnetflt sunrpc 156436 3 ip6t_REJECT 7296 2 nf_conntrack_ipv6 15864 2 ip6table_filter 6400 1 ip6_tables 14736 1 ip6table_filter ipv6 230516 18 ip6t_REJECT,nf_conntrack_ipv6 cpufreq_ondemand 9996 2 acpi_cpufreq 12172 0 dm_multipath 17292 0 uinput 10624 0 snd_hda_codec_analog 72576 1 snd_hda_intel 30480 0 snd_hda_codec 75520 2 snd_hda_codec_analog,snd_hda_intel snd_seq_dummy 6788 0 snd_seq_oss 28060 0 snd_seq_midi_event 9728 1 snd_seq_oss snd_seq 45756 9 snd_opl3_synth,snd_seq_midi_emul,snd_seq_dummy,snd_seq_oss,snd_seq_midi_event snd_usb_audio 93316 0 snd_ymfpci 34052 2 gameport 13452 1 snd_ymfpci snd_ac97_codec 96548 1 snd_ymfpci ac97_bus 5760 1 snd_ac97_codec snd_opl3_lib 12672 2 snd_opl3_synth,snd_ymfpci snd_hwdep 10500 3 snd_hda_codec,snd_usb_audio,snd_opl3_lib snd_usbmidi_lib 20352 1 snd_usb_audio snd_pcm_oss 36864 0 snd_mpu401_uart 10496 1 snd_ymfpci snd_rawmidi 22016 2 snd_usbmidi_lib,snd_mpu401_uart snd_mixer_oss 16896 1 snd_pcm_oss snd_seq_device 10124 6 snd_opl3_synth,snd_seq_dummy,snd_seq_oss,snd_seq,snd_opl3_lib,snd_rawmidi snd_pcm 64260 7 snd_hda_intel,snd_hda_codec,snd_usb_audio,snd_ymfpci,snd_ac97_codec,snd_pcm_oss snd_timer 21512 5 snd_hrtimer,snd_seq,snd_ymfpci,snd_opl3_lib,snd_pcm snd 52536 23 snd_hrtimer,snd_opl3_synth,snd_hda_codec_analog,snd_hda_intel,snd_hda_codec,snd_seq_dummy,snd_seq_oss,snd_seq,snd_usb_audio,snd_ymfpci,snd_ac97_codec,snd_opl3_lib,snd_hwdep,snd_usbmidi_lib,snd_pcm_oss,snd_mpu401_uart,snd_rawmidi,snd_mixer_oss,snd_seq_device,snd_pcm,snd_timer snd_page_alloc 11400 3 snd_hda_intel,snd_ymfpci,snd_pcm iTCO_wdt 13732 0 iTCO_vendor_support 6916 1 iTCO_wdt i2c_i801 12048 0 i2c_core 21396 2 drm,i2c_i801 skge 38416 0 uvcvideo 51720 0 compat_ioctl32 5120 1 uvcvideo videodev 32000 1 uvcvideo v4l1_compat 15876 2 uvcvideo,videodev firewire_ohci 22532 0 firewire_core 35616 1 firewire_ohci crc_itu_t 5760 1 firewire_core soundcore 9416 1 snd pcspkr 6272 0 pata_jmicron 6912 0 ata_generic 8452 0 But in alsa-driver-1.0.25 aplaymidi -l Port Client name Port name 14:0 Midi Through Midi Through Port-0 Module Size Used by fuse 49436 2 i915 53764 2 drm 158388 3 i915 vboxnetflt 74200 0 vboxdrv 98632 1 vboxnetflt sunrpc 156436 3 ip6t_REJECT 7296 2 nf_conntrack_ipv6 15864 2 ip6table_filter 6400 1 ip6_tables 14736 1 ip6table_filter ipv6 230516 18 ip6t_REJECT,nf_conntrack_ipv6 cpufreq_ondemand 9996 2 acpi_cpufreq 12172 0 dm_multipath 17292 0 uinput 10624 0 snd_hda_codec_analog 72576 1 snd_hda_intel 32016 0 snd_usb_audio 98948 0 snd_hda_codec 78976 2 snd_hda_codec_analog,snd_hda_intel snd_seq_dummy 6788 0 snd_ymfpci 34180 0 gameport 13452 1 snd_ymfpci snd_ac97_codec 96804 1 snd_ymfpci ac97_bus 5760 1 snd_ac97_codec snd_seq_oss 28060 0 snd_seq_midi_event 9728 1 snd_seq_oss snd_seq 45756 5 snd_seq_dummy,snd_seq_oss,snd_seq_midi_event snd_pcm_oss 36864 0 snd_mixer_oss 16768 1 snd_pcm_oss snd_pcm 64516 6 snd_hda_intel,snd_usb_audio,snd_hda_codec,snd_ymfpci,snd_ac97_codec,snd_pcm_oss firewire_ohci 22532 0 snd_opl3_lib 12672 1 snd_ymfpci pata_jmicron 6912 0 snd_timer 21512 4 snd_ymfpci,snd_seq,snd_pcm,snd_opl3_lib firewire_core 35616 1 firewire_ohci uvcvideo 51720 0 compat_ioctl32 5120 1 uvcvideo snd_usbmidi_lib 20608 1 snd_usb_audio pcspkr 6272 0 skge 38416 0 iTCO_wdt 13732 0 snd_hwdep 10500 3 snd_usb_audio,snd_hda_codec,snd_opl3_lib snd_mpu401_uart 10496 1 snd_ymfpci videodev 32000 1 uvcvideo snd_rawmidi 21888 2 snd_usbmidi_lib,snd_mpu401_uart snd_seq_device 10124 5 snd_seq_dummy,snd_seq_oss,snd_seq,snd_opl3_lib,snd_rawmidi crc_itu_t 5760 1 firewire_core iTCO_vendor_support 6916 1 iTCO_wdt snd 54328 19 snd_hda_codec_analog,snd_hda_intel,snd_usb_audio,snd_hda_codec,snd_seq_dummy,snd_ymfpci,snd_ac97_codec,snd_seq_oss,snd_seq,snd_pcm_oss,snd_mixer_oss,snd_pcm,snd_opl3_lib,snd_timer,snd_usbmidi_lib,snd_hwdep,snd_mpu401_uart,snd_rawmidi,snd_seq_device soundcore 9416 1 snd snd_page_alloc 11400 3 snd_hda_intel,snd_ymfpci,snd_pcm v4l1_compat 15876 2 uvcvideo,videodev i2c_i801 12048 0 i2c_core 21396 2 drm,i2c_i801 ata_generic 8452 0 pata_acpi 7680 0

2 2

[alsa-devel] [PATCH 00/20] ASoC: add imx-sgtl5000 machine driver working with fsl_ssi
by Shawn Guo 06 Mar '12

06 Mar '12

It's the full series adding imx-sgtl5000 machine driver which works with fsl_ssi driver on DT based imx platforms. - Patches 1 ~ 5: Convert audmux driver to a platform driver and move it into ASoC folder. - Patches 6 ~ 12: Merge ASoC imx and fsl folders. - Patches 13 ~17: Make fsl_ssi work for imx. - Patches 18 ~ 20: Add imx-sgtl5000 machine driver which supports DT only. It's based on sound/for-next and tested on imx51-babbage board with necessary platform/dts changes, which will be posted separately after the ASoC changes get merged. Regards, Shawn Richard Zhao (2): ARM: imx: convert audmux to a platform driver ASoC: fsl: add dt support for imx-audmux Shawn Guo (18): ASoC: imx: move eukrea audmux call into ASoC machine driver ASoC: imx: move phycore audmux call into ASoC machine driver ARM: imx: merge audmux-v1 and audmux-v2 ASoC: imx: move audmux driver into sound/soc/imx ASoC: imx: rename audmux prefix mxc to imx ASoC: imx: move SND_SOC_AC97_BUS selection down to machine driver ASoC: imx: initialize dma_params burstsize just in imx-ssi ASoC: imx: separate imx-pcm bits from imx-ssi driver ASoC: imx: add an explicit Kconfig option for imx-ssi driver ASoC: fsl: separate SSI and DMA Kconfig options ASoC: imx: merge sound/soc/imx into sound/soc/fsl ASoC: fsl: create fsl_utils to accommodate the common functions ASoC: fsl: check property 'compatible' for the machine name ASoC: fsl: make fsl_ssi driver compilable on ARM/IMX ASoC: fsl: remove the fatal error checking on codec-handle ASoC: fsl: let fsl_ssi work with imx pcm and machine drivers ASoC: rename sgtl5000 device tree binding document ASoC: fsl: add imx-sgtl5000 machine driver .../bindings/sound/imx-audio-sgtl5000.txt | 24 +++ .../devicetree/bindings/sound/imx-audmux.txt | 13 ++ .../{soc/codecs/fsl-sgtl5000.txt => sgtl5000.txt} | 0 arch/arm/mach-imx/Kconfig | 6 - arch/arm/mach-imx/eukrea_mbimx27-baseboard.c | 27 +-- arch/arm/mach-imx/eukrea_mbimxsd-baseboard.c | 1 - arch/arm/mach-imx/eukrea_mbimxsd25-baseboard.c | 24 +-- arch/arm/mach-imx/eukrea_mbimxsd35-baseboard.c | 24 +-- arch/arm/mach-imx/mach-imx27_visstrim_m10.c | 7 + arch/arm/mach-imx/mach-mx31ads.c | 6 + arch/arm/mach-imx/mach-pca100.c | 20 +-- arch/arm/mach-imx/mach-pcm043.c | 20 +-- arch/arm/plat-mxc/Kconfig | 6 - arch/arm/plat-mxc/Makefile | 2 - arch/arm/plat-mxc/audmux-v1.c | 64 ------ arch/arm/plat-mxc/include/mach/audmux.h | 60 ------ sound/soc/Kconfig | 1 - sound/soc/Makefile | 1 - sound/soc/fsl/Kconfig | 124 +++++++++++- sound/soc/fsl/Makefile | 29 +++- sound/soc/{imx => fsl}/eukrea-tlv320.c | 42 ++++- sound/soc/fsl/fsl_ssi.c | 146 ++++++++++---- sound/soc/fsl/fsl_utils.c | 135 +++++++++++++ sound/soc/fsl/fsl_utils.h | 27 +++ .../audmux-v2.c => sound/soc/fsl/imx-audmux.c | 185 +++++++++++++----- sound/soc/fsl/imx-audmux.h | 60 ++++++ sound/soc/{imx => fsl}/imx-pcm-dma-mx2.c | 7 +- sound/soc/{imx => fsl}/imx-pcm-fiq.c | 0 sound/soc/fsl/imx-pcm.c | 105 ++++++++++ sound/soc/fsl/imx-pcm.h | 32 +++ sound/soc/fsl/imx-sgtl5000.c | 210 ++++++++++++++++++++ sound/soc/{imx => fsl}/imx-ssi.c | 92 +--------- sound/soc/{imx => fsl}/imx-ssi.h | 16 +-- sound/soc/fsl/mpc8610_hpcd.c | 159 ++-------------- sound/soc/{imx => fsl}/mx27vis-aic32x4.c | 20 +- sound/soc/fsl/p1022_ds.c | 181 ++---------------- sound/soc/{imx => fsl}/phycore-ac97.c | 27 +++- sound/soc/{imx => fsl}/wm1133-ev1.c | 25 ++-- sound/soc/imx/Kconfig | 60 ------ sound/soc/imx/Makefile | 19 -- 40 files changed, 1163 insertions(+), 844 deletions(-)

9 64

[alsa-devel] [PATCH] ASoC: tegra: Remove unused variable
by Stephen Warren 06 Mar '12

06 Mar '12

From: Stephen Warren <swarren(a)wwwdotorg.org> Fixes the following warning: sound/soc/tegra/tegra_alc5632.c: In function 'tegra_alc5632_asoc_init': sound/soc/tegra/tegra_alc5632.c:118:6: warning: unused variable 'ret' [-Wunused-variable] Signed-off-by: Stephen Warren <swarren(a)wwwdotorg.org> --- sound/soc/tegra/tegra_alc5632.c | 1 - 1 files changed, 0 insertions(+), 1 deletions(-) diff --git a/sound/soc/tegra/tegra_alc5632.c b/sound/soc/tegra/tegra_alc5632.c index 2a27725..e45ccd8 100644 --- a/sound/soc/tegra/tegra_alc5632.c +++ b/sound/soc/tegra/tegra_alc5632.c @@ -115,7 +115,6 @@ static int tegra_alc5632_asoc_init(struct snd_soc_pcm_runtime *rtd) struct snd_soc_dapm_context *dapm = &codec->dapm; struct device_node *np = codec->card->dev->of_node; struct tegra_alc5632 *machine = snd_soc_card_get_drvdata(codec->card); - int ret; snd_soc_jack_new(codec, "Headset Jack", SND_JACK_HEADSET, &tegra_alc5632_hs_jack); -- 1.7.0.4

2 1

[alsa-devel] Drivers from tiwai/sound.git - module snd_seq_midi not compiled?
by Pavel Hofman 06 Mar '12

06 Mar '12

Hi, My midi keyboard is not listed in aconnect -i -l upon installing drivers from Takashi's git using gitcompile/make install. However, when I reinstall the whole kernel package 3.1.5 (from some ubuntu ppa), the midi works ok again. I compared loaded modules and found out for my hand-compiled drivers from git the module snd_seq_midi is missing. I re-run the compilation. Here is what I get at the beginning: utils/mod-deps --basedir /home/pavel/projects/alsa-tiwai/alsa-driver-build/alsa-kernel --hiddendir /home/pavel/projects/alsa-tiwai/alsa-driver-build --versiondep /home/pavel/projects/alsa-tiwai/alsa-driver-build/kconfig-vers --makeconf > toplevel.config.in can't handle 'default SND_SEQUENCER && SND_RAWMIDI', translating as depends on can't handle 'default SND_SEQUENCER && SND_OPL3_LIB', translating as depends on can't handle 'default SND_SEQUENCER && SND_OPL4_LIB', translating as depends on can't handle 'default SND_SEQUENCER && SND_SBAWE', translating as depends on can't handle 'default SND_SEQUENCER && SND_EMU10K1', translating as depends on can't handle 'default SND_FM801 || SND_ES1968 || RADIO_SF16FMR2', translating as depends on can't handle word INPUT=y properly, supposing it's OK can't handle word INPUT=SND_ES1968 properly, supposing it's OK can't handle word VIDEO_V4L2=y properly, supposing it's OK can't handle word VIDEO_V4L2=SND_ES1968 properly, supposing it's OK can't handle word VIDEO_V4L2=y properly, supposing it's OK can't handle word VIDEO_V4L2=SND_FM801 properly, supposing it's OK can't handle word INPUT=y properly, supposing it's OK can't handle word INPUT=SND_HDA_INTEL properly, supposing it's OK can't handle word SND_HDA_INPUT_BEEP=y properly, supposing it's OK can't handle word INPUT=y properly, supposing it's OK can't handle word INPUT=SND properly, supposing it's OK can't handle word INPUT=y properly, supposing it's OK can't handle word INPUT=SND_MAESTRO3 properly, supposing it's OK can't handle word INPUT=y properly, supposing it's OK can't handle word INPUT=SND_USB_CAIAQ properly, supposing it's OK can't handle word I2C=y properly, supposing it's OK can't handle word I2C=y properly, supposing it's OK can't handle word I2C=y properly, supposing it's OK utils/mod-deps --basedir /home/pavel/projects/alsa-tiwai/alsa-driver-build/alsa-kernel --hiddendir /home/pavel/projects/alsa-tiwai/alsa-driver-build --versiondep /home/pavel/projects/alsa-tiwai/alsa-driver-build/kconfig-vers --acinclude > acinclude.m4 can't handle 'default SND_SEQUENCER && SND_RAWMIDI', translating as depends on can't handle 'default SND_SEQUENCER && SND_OPL3_LIB', translating as depends on can't handle 'default SND_SEQUENCER && SND_OPL4_LIB', translating as depends on can't handle 'default SND_SEQUENCER && SND_SBAWE', translating as depends on can't handle 'default SND_SEQUENCER && SND_EMU10K1', translating as depends on can't handle 'default SND_FM801 || SND_ES1968 || RADIO_SF16FMR2', translating as depends on can't handle word INPUT=y properly, supposing it's OK can't handle word INPUT=SND_ES1968 properly, supposing it's OK can't handle word VIDEO_V4L2=y properly, supposing it's OK can't handle word VIDEO_V4L2=SND_ES1968 properly, supposing it's OK can't handle word VIDEO_V4L2=y properly, supposing it's OK can't handle word VIDEO_V4L2=SND_FM801 properly, supposing it's OK can't handle word INPUT=y properly, supposing it's OK can't handle word INPUT=SND_HDA_INTEL properly, supposing it's OK can't handle word SND_HDA_INPUT_BEEP=y properly, supposing it's OK can't handle word INPUT=y properly, supposing it's OK can't handle word INPUT=SND properly, supposing it's OK can't handle word INPUT=y properly, supposing it's OK can't handle word INPUT=SND_MAESTRO3 properly, supposing it's OK can't handle word INPUT=y properly, supposing it's OK can't handle word INPUT=SND_USB_CAIAQ properly, supposing it's OK can't handle word I2C=y properly, supposing it's OK can't handle word I2C=y properly, supposing it's OK can't handle word I2C=y properly, supposing it's OK utils/mod-deps --basedir /home/pavel/projects/alsa-tiwai/alsa-driver-build/alsa-kernel --hiddendir /home/pavel/projects/alsa-tiwai/alsa-driver-build --versiondep /home/pavel/projects/alsa-tiwai/alsa-driver-build/kconfig-vers --include > include/config1.h.in can't handle 'default SND_SEQUENCER && SND_RAWMIDI', translating as depends on can't handle 'default SND_SEQUENCER && SND_OPL3_LIB', translating as depends on can't handle 'default SND_SEQUENCER && SND_OPL4_LIB', translating as depends on can't handle 'default SND_SEQUENCER && SND_SBAWE', translating as depends on can't handle 'default SND_SEQUENCER && SND_EMU10K1', translating as depends on can't handle 'default SND_FM801 || SND_ES1968 || RADIO_SF16FMR2', translating as depends on can't handle word INPUT=y properly, supposing it's OK can't handle word INPUT=SND_ES1968 properly, supposing it's OK In config.status there is S["CONFIG_SND_RAWMIDI_SEQ"]="" I assume there should be "m" instead. And there is no D["CONFIG_SND_RAWMIDI_SEQ_MODULE"]=" 1" in config.status, only: D["CONFIG_SND_RAWMIDI_MODULE"]=" 1" D["CONFIG_SND_SEQUENCER_MODULE"]=" 1" ... As a result, the source acore/seq/seq_midi.c does not get compiled at all. pavel@nahore:~/projects/alsa-tiwai/alsa-driver-build$ ls -l acore/seq/ .... -rw-r--r-- 1 pavel pavel 13124 2012-03-04 21:53 seq_memory.c -rw-r--r-- 1 pavel pavel 12745 2012-03-04 21:53 seq_memory.c.orig -rw-r--r-- 1 pavel pavel 126440 2012-03-04 21:54 seq_memory.o -rw-r--r-- 1 pavel pavel 1112 2012-03-04 21:41 seq_memory.patch -rw-r--r-- 1 pavel pavel 89 2012-03-04 21:41 seq_midi.c <=== -rw-r--r-- 1 pavel pavel 75 2012-03-04 21:41 seq_midi_emul.c -rw-r--r-- 1 pavel pavel 93676 2012-03-04 21:54 seq_midi_emul.o -rw-r--r-- 1 pavel pavel 76 2012-03-04 21:41 seq_midi_event.c -rw-r--r-- 1 pavel pavel 95992 2012-03-04 21:54 seq_midi_event.o -rw-r--r-- 1 pavel pavel 101100 2012-03-04 21:54 seq.o -rw-r--r-- 1 pavel pavel 762 2012-03-04 21:41 seq.patch -rw-r--r-- 1 pavel pavel 94 2012-03-04 21:41 seq_ports.c -rw-r--r-- 1 pavel pavel 125464 2012-03-04 21:54 seq_ports.o .... Thanks a lot for any suggestion or help. Best regards, Pavel.

3 4

Re: [alsa-devel] Query about regmap SPI
by Ashish Chavan 06 Mar '12

06 Mar '12

On Tue, 2012-03-06 at 20:15 +0530, Ashish Chavan wrote: > On Tue, Mar 06, 2012 at 12:53:51PM +0500, Ashish Chavan wrote: > > Please always CC the relevant mailing lists on postings. I shouldn't > need to remind you about this stuff. Yes, I agree :-( It's my mistake that I missed linux-kernel(a)vger.kernel.org > No, implementing stuff like this in individual device drivers rather > defeats the point of having frameworks. It sounds like it should be > trivial to add support for this to regmap. I also think that there should be a way to handle this kind of stuff by the framework. May I know if you have any plan to add this to regmap? or should I come up with a patch to extend regmap to support this and propose you?

2 3

[alsa-devel] [PATCH] ASoC: DAPM - Use dev_err for debugfs error message.
by Liam Girdwood 06 Mar '12

06 Mar '12

Remove printk(KERN_ERR) and use dev_err() instead. Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- sound/soc/soc-dapm.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index a4d4aa1..8ad63b3 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1667,7 +1667,7 @@ void snd_soc_dapm_debugfs_init(struct snd_soc_dapm_context *dapm, dapm->debugfs_dapm = debugfs_create_dir("dapm", parent); if (!dapm->debugfs_dapm) { - printk(KERN_WARNING + dev_err(dapm->dev, "Failed to create DAPM debugfs directory\n"); return; } -- 1.7.5.4

2 1

[alsa-devel] [PATCH][RFC] intel-hda: proper support for Asus N55SF/N75SF laptops with external subwoofer
by Andrew 06 Mar '12

06 Mar '12

Hello all, The following patch adds full support for Asus Laptops N55SF/N75SF (Possibly more models) with external subwoofer. It adds a new model (asus-nx5) and proper PCIID detection. Example asoundrc for upmixing and dmix is attached as well. (Thanks to K900 from AgiliaLinux dev. conference for it as well as testing on N55SF) Currently everything is supported except Subwoofer pin autosensing. It is somehow implemented in windows drivers, but I couldn't get it neither via codec's gpio or regular 0x16 unsol event. In 2ch mode the subwoofer is off, so should save a little bit of battery. I couldn't test automic feature and spdif, because I don't have the hardware around. Since alc662_quirks.c looks a little bit overbloated, I moved my quirks to a separate file. If that's not OK, I'll resubmit the patch. (This is my first patch to alsa-devel, so I might be missing something) Regards, Andrew

2 1

[alsa-devel] ASoC updates for 3.4
by Mark Brown 06 Mar '12

06 Mar '12

The following changes since commit ad20ff920c1fd217578e2c637dd50c1878a21c06: ASoC: wm8753: fix initialization (2012-03-04 14:34:44 +0000) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-3.4 for you to fetch changes up to 9d5ef2663fe220a88412a7190942b7d933da0333: ASoC: fsl: add dt support for imx-audmux (2012-03-06 12:44:21 +0000) ---------------------------------------------------------------- A few more ASoC updates, the main one is the move of the audmux driver from arch/arm into sound/soc. There's also some general driver specific tweaks and fixes. ---------------------------------------------------------------- Axel Lin (1): ASoC: Add missing regmap_init_i2c in wm8804_i2c_probe Mark Brown (4): ASoC: dapm: Show if widgets are forced in debugfs ASoC: wm8994: Use audio mode for jack detection when system is active ASoC: wm8962: Remove unneeded pm_runtime_set_active() ASoC: wm8962: Remove defaults for volatile registers Richard Zhao (2): ARM: imx: convert audmux to a platform driver ASoC: fsl: add dt support for imx-audmux Shawn Guo (7): ASoC: imx: move eukrea audmux call into ASoC machine driver ASoC: imx: move phycore audmux call into ASoC machine driver ARM: imx: merge audmux-v1 and audmux-v2 ASoC: imx: move audmux driver into sound/soc/imx ASoC: imx: rename audmux prefix mxc to imx ASoC: imx: separate imx-pcm bits from imx-ssi driver ASoC: imx: add an explicit Kconfig option for imx-ssi driver .../devicetree/bindings/sound/imx-audmux.txt | 13 ++ arch/arm/mach-imx/Kconfig | 6 - arch/arm/mach-imx/eukrea_mbimx27-baseboard.c | 20 -- arch/arm/mach-imx/eukrea_mbimxsd-baseboard.c | 1 - arch/arm/mach-imx/eukrea_mbimxsd25-baseboard.c | 17 -- arch/arm/mach-imx/eukrea_mbimxsd35-baseboard.c | 17 -- arch/arm/mach-imx/mach-pca100.c | 13 -- arch/arm/mach-imx/mach-pcm043.c | 13 -- arch/arm/mach-imx/mm-imx21.c | 6 + arch/arm/mach-imx/mm-imx25.c | 7 + arch/arm/mach-imx/mm-imx27.c | 7 + arch/arm/mach-imx/mm-imx3.c | 13 ++ arch/arm/mach-imx/mm-imx5.c | 22 +++ arch/arm/plat-mxc/Kconfig | 6 - arch/arm/plat-mxc/Makefile | 2 - arch/arm/plat-mxc/audmux-v1.c | 64 ------- arch/arm/plat-mxc/include/mach/audmux.h | 60 ------- sound/soc/codecs/wm8804.c | 6 + sound/soc/codecs/wm8962.c | 18 +-- sound/soc/codecs/wm8994.c | 77 +++++--- sound/soc/codecs/wm8994.h | 1 + sound/soc/imx/Kconfig | 21 ++- sound/soc/imx/Makefile | 15 +- sound/soc/imx/eukrea-tlv320.c | 40 ++++- .../audmux-v2.c => sound/soc/imx/imx-audmux.c | 185 +++++++++++++++----- sound/soc/imx/imx-audmux.h | 60 +++++++ sound/soc/imx/imx-pcm-dma-mx2.c | 2 +- sound/soc/imx/imx-pcm.c | 105 +++++++++++ sound/soc/imx/imx-pcm.h | 32 ++++ sound/soc/imx/imx-ssi.c | 88 --------- sound/soc/imx/imx-ssi.h | 16 +-- sound/soc/imx/mx27vis-aic32x4.c | 20 +- sound/soc/imx/phycore-ac97.c | 27 +++- sound/soc/imx/wm1133-ev1.c | 25 ++-- sound/soc/soc-dapm.c | 5 +- 35 files changed, 581 insertions(+), 449 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/imx-audmux.txt delete mode 100644 arch/arm/plat-mxc/audmux-v1.c delete mode 100644 arch/arm/plat-mxc/include/mach/audmux.h rename arch/arm/plat-mxc/audmux-v2.c => sound/soc/imx/imx-audmux.c (52%) create mode 100644 sound/soc/imx/imx-audmux.h create mode 100644 sound/soc/imx/imx-pcm.c create mode 100644 sound/soc/imx/imx-pcm.h

2 1

[alsa-devel] [PATCH] ASoC: Add missing regmap_init_i2c in wm8804_i2c_probe
by Axel Lin 06 Mar '12

06 Mar '12

commit 891271c "ASoC: Convert wm8804 to direct regmap API usage" only converts wm8804_spi_probe to use regmap_init_spi. This patch adds missing regmap_init_i2c in wm8804_i2c_probe. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/wm8804.c | 6 ++++++ 1 files changed, 6 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8804.c b/sound/soc/codecs/wm8804.c index 7ee8dcf..6bd1b76 100644 --- a/sound/soc/codecs/wm8804.c +++ b/sound/soc/codecs/wm8804.c @@ -755,6 +755,12 @@ static __devinit int wm8804_i2c_probe(struct i2c_client *i2c, if (!wm8804) return -ENOMEM; + wm8804->regmap = regmap_init_i2c(i2c, &wm8804_regmap_config); + if (IS_ERR(wm8804->regmap)) { + ret = PTR_ERR(wm8804->regmap); + return ret; + } + i2c_set_clientdata(i2c, wm8804); ret = snd_soc_register_codec(&i2c->dev, -- 1.7.5.4

2 1

[alsa-devel] Query about regmap SPI
by Ashish Chavan 06 Mar '12

06 Mar '12

Hi, I am planning to add SPI support for da7210 codec driver. SPI for this codec has slightly different format than others. It has 8 bit register address and 8 bit data. Out of 8 bits of register address, one bit (i.e. LSB) is used to indicate read or write. Value of 1 at LSB denotes READ while value of 0 at LSB denotes write operation. Thus in order for SPI READ to work correctly, the register address need to be left shifted by one and then ORed with 0x1. In the same way for SPI WRITE to work correctly, it need to be left shifted by one. I have gone though the regmap sources and found that ORing can be achieved by setting correct read_flag_mask. But I couldn't find anything by which I can tell regmap core to shift the register address before ORing it with read_flag_mask. Am I missing anything here? Is there any other way to achieve this using regmap? The only other way I can think of is providing a wrapper function to read and write in codec driver and doing the shifting job inside that wrapper. TIA!

2 1

Re: [alsa-devel] [PATCH v6] sound/soc/codecs: add LAPIS Semiconductor ML26124
by Mark Brown 06 Mar '12

06 Mar '12

On Tue, Mar 06, 2012 at 12:03:20PM +0900, Tomoya MORINAGA wrote: > 2012年3月2日21:58 Mark Brown <broonie(a)opensource.wolfsonmicro.com>: > >> Let me know why I should use SND_SOC_REGMAP not _I2C. > > Have you checked what _I2C and _REGMAP do? > I've just checked it. > If SND_SOC_REGMAP is used, it seems there is no processing in it. This is not the case. > So, can I delete snd_soc_codec_set_cache_io ? No. > > Use module_i2c_driver(). > I understand. > As a matter of interest, why can I use module_i2c_driver ? It is best practice. > There is no ASoC driver uses module_i2c_driver. This is not the case. As I have *REPEATEDLY* told you you should be sumbitting code against current development versions of the subsystem. > >> +static int ml26124_update_bits(struct snd_soc_codec *codec, unsigned short reg, > >> + unsigned int mask, unsigned int value) > > Why are you open coding this in your driver? There is no point in > > replicating subsystem functionality. > For ML26124 spec, it's necessary. > Because the codec's register read-address and write-address is not the same. > E.G. > Sampling Rate Register > R: offset=0 > W: offset=1 This is not what the above prototype says, the above prototype has only one register in it. You need to make this *much* clearer for the registers it's affecting.

1 0

[alsa-devel] [PATCH] snd-hda-intel: add quirk to detect CD input on Gigabyte EP45-DS3
by Marton Balint 06 Mar '12

06 Mar '12

My CD input got lost in commit 68ef0561efe494143516df38c03a16b837b8e79c. Raymond helped me to add the necessary pin fixup to make it appear again. In fact, this is basically his patch. It fixes alsa bug #5541. Signed-off-by: Marton Balint <cus(a)passwd.hu> --- sound/pci/hda/patch_realtek.c | 9 +++++++++ 1 files changed, 9 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index f286bb8..5e53020 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4367,6 +4367,7 @@ enum { ALC882_FIXUP_PB_M5210, ALC882_FIXUP_ACER_ASPIRE_7736, ALC882_FIXUP_ASUS_W90V, + ALC889_FIXUP_CD, ALC889_FIXUP_VAIO_TT, ALC888_FIXUP_EEE1601, ALC882_FIXUP_EAPD, @@ -4494,6 +4495,13 @@ static const struct alc_fixup alc882_fixups[] = { { } } }, + [ALC889_FIXUP_CD] = { + .type = ALC_FIXUP_PINS, + .v.pins = (const struct alc_pincfg[]) { + { 0x1c, 0x993301f0 }, /* CD */ + { } + } + }, [ALC889_FIXUP_VAIO_TT] = { .type = ALC_FIXUP_PINS, .v.pins = (const struct alc_pincfg[]) { @@ -4650,6 +4658,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x1071, 0x8258, "Evesham Voyaeger", ALC882_FIXUP_EAPD), SND_PCI_QUIRK_VENDOR(0x1462, "MSI", ALC882_FIXUP_GPIO3), + SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte EP45-DS3", ALC889_FIXUP_CD), SND_PCI_QUIRK(0x147b, 0x107a, "Abit AW9D-MAX", ALC882_FIXUP_ABIT_AW9D_MAX), SND_PCI_QUIRK_VENDOR(0x1558, "Clevo laptop", ALC882_FIXUP_EAPD), SND_PCI_QUIRK(0x161f, 0x2054, "Medion laptop", ALC883_FIXUP_EAPD), -- 1.7.3.4

2 1

[alsa-devel] [PATCH - aplay 1/1] aplay - option to dump HW parameters
by pavel.hofman＠ivitera.com 06 Mar '12

06 Mar '12

From: Pavel Hofman <pavel.hofman(a)ivitera.com> Dumping HW parameters of the current device with snd_pcm_hw_params_dump, a new option --dump-hw-params added. Signed-off-by: Pavel Hofman <pavel.hofman(a)ivitera.com> diff --git a/aplay/aplay.1 b/aplay/aplay.1 index 2acaed5..0195322 100644 --- a/aplay/aplay.1 +++ b/aplay/aplay.1 @@ -184,6 +184,13 @@ the month, %H hour, %M minute and %S second. In addition, %v is the file number, starting at 1. When this option is specified, intermediate directories for the output file are created automatically. This option has no effect if \-\-separate\-channels is specified. +.TP +\fI\-\-dump\-hw\-params\fP +Dump hw_params of the device preconfigured status to stderr. The dump +lists capabilities of the selected device such as supported formats, +sampling rates, numbers of channels, period and buffer bytes/sizes/times. +For raw device hw:X this option basically lists hardware capabilities of +the soundcard. .SH SIGNALS When recording, SIGINT, SIGTERM and SIGABRT will close the output diff --git a/aplay/aplay.c b/aplay/aplay.c index 73a9544..0633cbd 100644 --- a/aplay/aplay.c +++ b/aplay/aplay.c @@ -129,6 +129,7 @@ static int max_file_time = 0; static int use_strftime = 0; volatile static int recycle_capture_file = 0; static long term_c_lflag = -1; +static int dump_hw_params = 0; static int fd = -1; static off64_t pbrec_count = LLONG_MAX, fdcount; @@ -223,7 +224,8 @@ _("Usage: %s [OPTION]... [FILE]...\n" " --max-file-time=# start another output file when the old file has recorded\n" " for this many seconds\n" " --process-id-file write the process ID here\n" -" --use-strftime apply the strftime facility to the output file name\n") +" --use-strftime apply the strftime facility to the output file name\n" +" --dump-hw-params dump hw_params of the device\n") , command); printf(_("Recognized sample formats are:")); for (k = 0; k < SND_PCM_FORMAT_LAST; ++k) { @@ -416,7 +418,8 @@ enum { OPT_TEST_NOWAIT, OPT_MAX_FILE_TIME, OPT_PROCESS_ID_FILE, - OPT_USE_STRFTIME + OPT_USE_STRFTIME, + OPT_DUMP_HWPARAMS }; int main(int argc, char *argv[]) @@ -461,6 +464,7 @@ int main(int argc, char *argv[]) {"process-id-file", 1, 0, OPT_PROCESS_ID_FILE}, {"use-strftime", 0, 0, OPT_USE_STRFTIME}, {"interactive", 0, 0, 'i'}, + {"dump-hw-params", 0, 0, OPT_DUMP_HWPARAMS}, {0, 0, 0, 0} }; char *pcm_name = "default"; @@ -662,6 +666,9 @@ int main(int argc, char *argv[]) case OPT_USE_STRFTIME: use_strftime = 1; break; + case OPT_DUMP_HWPARAMS: + dump_hw_params = 1; + break; default: fprintf(stderr, _("Try `%s --help' for more information.\n"), command); return 1; @@ -1059,6 +1066,13 @@ static void set_params(void) error(_("Broken configuration for this PCM: no configurations available")); prg_exit(EXIT_FAILURE); } + if (dump_hw_params) { + fprintf(stderr, _("HW Params of device \"%s\":\n"), + snd_pcm_name(handle)); + fprintf(stderr, "--------------------\n"); + snd_pcm_hw_params_dump(params, log); + fprintf(stderr, "--------------------\n"); + } if (mmap_flag) { snd_pcm_access_mask_t *mask = alloca(snd_pcm_access_mask_sizeof()); snd_pcm_access_mask_none(mask); -- 1.7.0.4

2 1

[alsa-devel] [PATCH] ASoC: wm8962: Remove defaults for volatile registers
by Mark Brown 06 Mar '12

06 Mar '12

Save a little RAM. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8962.c | 17 +++-------------- 1 files changed, 3 insertions(+), 14 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index d5ee3ec..e5aaf24 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -116,11 +116,11 @@ static struct reg_default wm8962_reg[] = { { 1, 0x049F }, /* R1 - Right Input volume */ { 2, 0x0000 }, /* R2 - HPOUTL volume */ { 3, 0x0000 }, /* R3 - HPOUTR volume */ - { 4, 0x0020 }, /* R4 - Clocking1 */ + { 5, 0x0018 }, /* R5 - ADC & DAC Control 1 */ { 6, 0x2008 }, /* R6 - ADC & DAC Control 2 */ { 7, 0x000A }, /* R7 - Audio Interface 0 */ - { 8, 0x01E4 }, /* R8 - Clocking2 */ + { 9, 0x0300 }, /* R9 - Audio Interface 1 */ { 10, 0x00C0 }, /* R10 - Left DAC volume */ { 11, 0x00C0 }, /* R11 - Right DAC volume */ @@ -129,7 +129,7 @@ static struct reg_default wm8962_reg[] = { { 15, 0x6243 }, /* R15 - Software Reset */ { 17, 0x007B }, /* R17 - ALC1 */ - { 18, 0x0000 }, /* R18 - ALC2 */ + { 19, 0x1C32 }, /* R19 - ALC3 */ { 20, 0x3200 }, /* R20 - Noise Gate */ { 21, 0x00C0 }, /* R21 - Left ADC volume */ @@ -153,10 +153,6 @@ static struct reg_default wm8962_reg[] = { { 40, 0x0000 }, /* R40 - SPKOUTL volume */ { 41, 0x0000 }, /* R41 - SPKOUTR volume */ - { 47, 0x0000 }, /* R47 - Thermal Shutdown Status */ - { 48, 0x8027 }, /* R48 - Additional Control (4) */ - { 49, 0x0010 }, /* R49 - Class D Control 1 */ - { 51, 0x0003 }, /* R51 - Class D Control 2 */ { 56, 0x0506 }, /* R56 - Clocking 4 */ @@ -168,8 +164,6 @@ static struct reg_default wm8962_reg[] = { { 64, 0x0810 }, /* R64 - DC Servo 4 */ - { 66, 0x0000 }, /* R66 - DC Servo 6 */ - { 68, 0x001B }, /* R68 - Analogue PGA Bias */ { 69, 0x0000 }, /* R69 - Analogue HP 0 */ @@ -302,9 +296,6 @@ static struct reg_default wm8962_reg[] = { { 516, 0x8100 }, /* R516 - GPIO 5 */ { 517, 0x8100 }, /* R517 - GPIO 6 */ - { 560, 0x0000 }, /* R560 - Interrupt Status 1 */ - { 561, 0x0000 }, /* R561 - Interrupt Status 2 */ - { 568, 0x0030 }, /* R568 - Interrupt Status 1 Mask */ { 569, 0xFFED }, /* R569 - Interrupt Status 2 Mask */ @@ -316,8 +307,6 @@ static struct reg_default wm8962_reg[] = { { 768, 0x1C00 }, /* R768 - DSP2 Power Management */ - { 1037, 0x0000 }, /* R1037 - DSP2_ExecControl */ - { 8192, 0x0000 }, /* R8192 - DSP2 Instruction RAM 0 */ { 9216, 0x0030 }, /* R9216 - DSP2 Address RAM 2 */ -- 1.7.9.1

1 0

[alsa-devel] [PATCH] ASoC: wm8994: Use fast VMID ramp
by Mark Brown 06 Mar '12

06 Mar '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 581049a..a6df001 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -808,7 +808,7 @@ static void vmid_reference(struct snd_soc_codec *codec) WM8994_BIAS_SRC | WM8994_STARTUP_BIAS_ENA | WM8994_VMID_BUF_ENA | - (0x0 << WM8994_VMID_RAMP_SHIFT)); + (0x3 << WM8994_VMID_RAMP_SHIFT)); /* Main bias enable, VMID=2x40k */ snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_1, @@ -816,7 +816,7 @@ static void vmid_reference(struct snd_soc_codec *codec) WM8994_VMID_SEL_MASK, WM8994_BIAS_ENA | 0x2); - msleep(100); + msleep(50); snd_soc_update_bits(codec, WM8994_ANTIPOP_2, WM8994_VMID_RAMP_MASK | WM8994_BIAS_SRC, -- 1.7.9.1

1 0

[alsa-devel] [PATCH] ASoC: wm8994: Use audio mode for jack detection when system is active
by Mark Brown 06 Mar '12

06 Mar '12

When we are out of system sleep always use audio mode for jack detection in order to avoid potential performance issues handing off between modes. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 77 ++++++++++++++++++++++++++++----------------- sound/soc/codecs/wm8994.h | 1 + 2 files changed, 49 insertions(+), 29 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 2417ef9..bc12d09 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -685,8 +685,6 @@ SOC_SINGLE_TLV("MIXINL IN1RP Boost Volume", WM8994_INPUT_MIXER_1, 8, 1, 0, static void wm1811_jackdet_set_mode(struct snd_soc_codec *codec, u16 mode) { struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); - u16 old = snd_soc_read(codec, WM8994_ANTIPOP_2) - & WM1811_JACKDET_MODE_MASK; if (!wm8994->jackdet || !wm8994->jack_cb) return; @@ -694,28 +692,17 @@ static void wm1811_jackdet_set_mode(struct snd_soc_codec *codec, u16 mode) if (wm8994->active_refcount) mode = WM1811_JACKDET_MODE_AUDIO; - if (mode == old) + if (mode == wm8994->jackdet_mode) return; - snd_soc_update_bits(codec, WM8994_ANTIPOP_2, - WM1811_JACKDET_MODE_MASK, mode); - - switch (mode) { - case WM1811_JACKDET_MODE_MIC: - case WM1811_JACKDET_MODE_AUDIO: - switch (old) { - case WM1811_JACKDET_MODE_MIC: - case WM1811_JACKDET_MODE_AUDIO: - break; - default: - msleep(2); - break; - } + wm8994->jackdet_mode = mode; - default: - break; - } + /* Always use audio mode to detect while the system is active */ + if (mode != WM1811_JACKDET_MODE_NONE) + mode = WM1811_JACKDET_MODE_AUDIO; + snd_soc_update_bits(codec, WM8994_ANTIPOP_2, + WM1811_JACKDET_MODE_MASK, mode); } static void active_reference(struct snd_soc_codec *codec) @@ -2749,7 +2736,7 @@ static struct snd_soc_dai_driver wm8994_dai[] = { }; #ifdef CONFIG_PM -static int wm8994_suspend(struct snd_soc_codec *codec) +static int wm8994_codec_suspend(struct snd_soc_codec *codec) { struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); struct wm8994 *control = wm8994->wm8994; @@ -2783,7 +2770,7 @@ static int wm8994_suspend(struct snd_soc_codec *codec) return 0; } -static int wm8994_resume(struct snd_soc_codec *codec) +static int wm8994_codec_resume(struct snd_soc_codec *codec) { struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); struct wm8994 *control = wm8994->wm8994; @@ -2842,8 +2829,8 @@ static int wm8994_resume(struct snd_soc_codec *codec) return 0; } #else -#define wm8994_suspend NULL -#define wm8994_resume NULL +#define wm8994_codec_suspend NULL +#define wm8994_codec_resume NULL #endif static void wm8994_handle_retune_mobile_pdata(struct wm8994_priv *wm8994) @@ -3955,8 +3942,8 @@ static int wm8994_codec_remove(struct snd_soc_codec *codec) static struct snd_soc_codec_driver soc_codec_dev_wm8994 = { .probe = wm8994_codec_probe, .remove = wm8994_codec_remove, - .suspend = wm8994_suspend, - .resume = wm8994_resume, + .suspend = wm8994_codec_suspend, + .resume = wm8994_codec_resume, .set_bias_level = wm8994_set_bias_level, }; @@ -3983,11 +3970,43 @@ static int __devexit wm8994_remove(struct platform_device *pdev) return 0; } +#ifdef CONFIG_PM_SLEEP +static int wm8994_suspend(struct device *dev) +{ + struct wm8994_priv *wm8994 = dev_get_drvdata(dev); + + /* Drop down to power saving mode when system is suspended */ + if (wm8994->jackdet && !wm8994->active_refcount) + regmap_update_bits(wm8994->wm8994->regmap, WM8994_ANTIPOP_2, + WM1811_JACKDET_MODE_MASK, + wm8994->jackdet_mode); + + return 0; +} + +static int wm8994_resume(struct device *dev) +{ + struct wm8994_priv *wm8994 = dev_get_drvdata(dev); + + if (wm8994->jackdet && wm8994->jack_cb) + regmap_update_bits(wm8994->wm8994->regmap, WM8994_ANTIPOP_2, + WM1811_JACKDET_MODE_MASK, + WM1811_JACKDET_MODE_AUDIO); + + return 0; +} +#endif + +static const struct dev_pm_ops wm8994_pm_ops = { + SET_SYSTEM_SLEEP_PM_OPS(wm8994_suspend, wm8994_resume) +}; + static struct platform_driver wm8994_codec_driver = { .driver = { - .name = "wm8994-codec", - .owner = THIS_MODULE, - }, + .name = "wm8994-codec", + .owner = THIS_MODULE, + .pm = &wm8994_pm_ops, + }, .probe = wm8994_probe, .remove = __devexit_p(wm8994_remove), }; diff --git a/sound/soc/codecs/wm8994.h b/sound/soc/codecs/wm8994.h index f996d14..2f4d2d1 100644 --- a/sound/soc/codecs/wm8994.h +++ b/sound/soc/codecs/wm8994.h @@ -122,6 +122,7 @@ struct wm8994_priv { bool jack_mic; int btn_mask; bool jackdet; + int jackdet_mode; wm8958_micdet_cb jack_cb; void *jack_cb_data; -- 1.7.9.1

1 0

[alsa-devel] [PATCH v2 00/17] ASoC: add imx-sgtl5000 machine driver working with fsl_ssi
by Shawn Guo 05 Mar '12

05 Mar '12

Changes since v1: * Address comments from Mark, Timur, and Sascha (Thanks for review). * Drop patches that have been applied by Mark. Richard Zhao (2): ARM: imx: convert audmux to a platform driver ASoC: fsl: add dt support for imx-audmux Shawn Guo (15): ASoC: imx: move eukrea audmux call into ASoC machine driver ASoC: imx: move phycore audmux call into ASoC machine driver ARM: imx: merge audmux-v1 and audmux-v2 ASoC: imx: move audmux driver into sound/soc/imx ASoC: imx: rename audmux prefix mxc to imx ASoC: imx: separate imx-pcm bits from imx-ssi driver ASoC: imx: add an explicit Kconfig option for imx-ssi driver ASoC: fsl: separate SSI and DMA Kconfig options ASoC: imx: merge sound/soc/imx into sound/soc/fsl ASoC: fsl: create fsl_utils to accommodate the common functions ASoC: fsl: check property 'compatible' for the machine name ASoC: fsl: make fsl_ssi driver compilable on ARM/IMX ASoC: fsl: remove the fatal error checking on codec-handle ASoC: fsl: let fsl_ssi work with imx pcm and machine drivers ASoC: fsl: add imx-sgtl5000 machine driver .../bindings/sound/imx-audio-sgtl5000.txt | 24 +++ .../devicetree/bindings/sound/imx-audmux.txt | 13 ++ arch/arm/mach-imx/Kconfig | 6 - arch/arm/mach-imx/eukrea_mbimx27-baseboard.c | 20 -- arch/arm/mach-imx/eukrea_mbimxsd-baseboard.c | 1 - arch/arm/mach-imx/eukrea_mbimxsd25-baseboard.c | 17 -- arch/arm/mach-imx/eukrea_mbimxsd35-baseboard.c | 17 -- arch/arm/mach-imx/mach-pca100.c | 13 -- arch/arm/mach-imx/mach-pcm043.c | 13 -- arch/arm/mach-imx/mm-imx21.c | 6 + arch/arm/mach-imx/mm-imx25.c | 7 + arch/arm/mach-imx/mm-imx27.c | 7 + arch/arm/mach-imx/mm-imx3.c | 13 ++ arch/arm/mach-imx/mm-imx5.c | 22 +++ arch/arm/plat-mxc/Kconfig | 6 - arch/arm/plat-mxc/Makefile | 2 - arch/arm/plat-mxc/audmux-v1.c | 64 ------- arch/arm/plat-mxc/include/mach/audmux.h | 60 ------- sound/soc/Kconfig | 1 - sound/soc/Makefile | 1 - sound/soc/fsl/Kconfig | 124 +++++++++++++- sound/soc/fsl/Makefile | 29 +++- sound/soc/{imx => fsl}/eukrea-tlv320.c | 42 ++++- sound/soc/fsl/fsl_ssi.c | 139 ++++++++++++---- sound/soc/fsl/fsl_utils.c | 135 ++++++++++++++ sound/soc/fsl/fsl_utils.h | 27 +++ .../audmux-v2.c => sound/soc/fsl/imx-audmux.c | 185 +++++++++++++++----- sound/soc/fsl/imx-audmux.h | 60 +++++++ sound/soc/{imx => fsl}/imx-pcm-dma-mx2.c | 2 +- sound/soc/{imx => fsl}/imx-pcm-fiq.c | 0 sound/soc/fsl/imx-pcm.c | 105 +++++++++++ sound/soc/fsl/imx-pcm.h | 32 ++++ sound/soc/fsl/imx-sgtl5000.c | 181 +++++++++++++++++++ sound/soc/{imx => fsl}/imx-ssi.c | 90 +---------- sound/soc/{imx => fsl}/imx-ssi.h | 16 +-- sound/soc/fsl/mpc8610_hpcd.c | 159 ++---------------- sound/soc/{imx => fsl}/mx27vis-aic32x4.c | 20 +- sound/soc/fsl/p1022_ds.c | 181 ++------------------ sound/soc/{imx => fsl}/phycore-ac97.c | 27 +++- sound/soc/{imx => fsl}/wm1133-ev1.c | 25 ++-- sound/soc/imx/Kconfig | 60 ------- sound/soc/imx/Makefile | 19 -- 42 files changed, 1140 insertions(+), 831 deletions(-)

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[alsa-devel] [PATCH 1/3] ASoC: core - Add card mutex locking subclasses
by Liam Girdwood 05 Mar '12

05 Mar '12

This is the first part of a change that is intended to improve ASoC locking protection for DAPM and PCM operations. We currently use the codec->mutex for locking some DAPM operations. This can't guarantee correctness now with the recent multi-component and forth coming Dynamic PCM updates. This part of the series adds a mutex class for the soc_card mutex. The SND_SOC_CARD_CLASS_INIT class is used for card initialisation only whilst the SND_SOC_CARD_CLASS_PCM class is used for all the DAPM and forth coming Dynamic PCM operations. The new mutex classes are required otherwise we will see a false positive mutex deadlock warning between the card initialisation and the DAPM operations (something that would never deadlock in real life). Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- include/sound/soc.h | 5 +++++ sound/soc/soc-core.c | 2 +- 2 files changed, 6 insertions(+), 1 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 82bd773..f2b5ad1 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -288,6 +288,11 @@ enum snd_soc_pcm_subclass { SND_SOC_PCM_CLASS_BE = 1, }; +enum snd_soc_card_subclass { + SND_SOC_CARD_CLASS_INIT = 0, + SND_SOC_CARD_CLASS_PCM = 1, +}; + int snd_soc_codec_set_sysclk(struct snd_soc_codec *codec, int clk_id, int source, unsigned int freq, int dir); int snd_soc_codec_set_pll(struct snd_soc_codec *codec, int pll_id, int source, diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 29dbbd7..2038522 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1416,7 +1416,7 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) struct snd_soc_dai_link *dai_link; int ret, i, order; - mutex_lock(&card->mutex); + mutex_lock_nested(&card->mutex, SND_SOC_CARD_CLASS_INIT); if (card->instantiated) { mutex_unlock(&card->mutex); -- 1.7.5.4

2 9

[alsa-devel] [bug?, 3.1 -> 3.2.7 regression] ALC269 (ThinkPad L412): auto-mute mode disabled by default
by Jonathan Nieder 05 Mar '12

05 Mar '12

Hi Takashi et al, Pavel Yakunin wrote[1]: > There is a problem with a headphones jack detection in the 3.2 kernel for the ALC269 codec: > sound comes from both speaker and headphones simultaneously and never switches between them. > > The problem arised when I upgraded my system to the 3.2 kernel, and when I rolled back to the 3.1 > kernel, everything was ok. It turned out to be due to the auto-mute mode control, which is present on this model in 3.2 and not 3.1, being disabled by default. > On 2012-03-05 05:16, Jonathan Nieder wrote: >> One more test: if you run "alsactl init" as root, does the auto-mute >> mode control end up enabled or disabled? [...] > It ends up disabled. Versions tested: 3.1.y - auto-mute works, no auto-mute control present 3.2.y - auto-mute disabled by default but control works sound/for-next as of 2012-03-03 - likewise Is this behavior intended? More details, including alsa-info.sh logs, at [1]. Thanks, Jonathan [1] http://bugs.debian.org/661830

2 3

[alsa-devel] [PATCH v3 0/3] Amstrad Delta: access MODEM_RESET GPIO pin over a regulator
by Janusz Krzysztofik 05 Mar '12

05 Mar '12

The Amstrad Delta on-board latch2 bit named MODEM_NRESET, now available as a GPIO pin AMS_DELTA_GPIO_PIN_NMODEM_RESET, is used to power up/down (bring into/out of a reset state) two distinct on-board devices simultaneously: the modem, and the voice codec. As a consequence, that bit is, or can be, manipulated concurrently by two drivers, or their platform provided hooks. Instead of updating those drivers to use the gpiolib API as a new method of controlling the MODEM_NRESET pin state, like it was done to other drivers accessing latch2 pins, and still being vulnerable to potential concurrency conflicts, or trying to solve that sharing issue with a custom piece of code, set up a fixed regulator device on top of that GPIO pin and update both drivers to manipulate that regulator, not the GPIO pin directly. This is v3 of this series, with Mark's Ack for ASoC subtree touching changes already collected, rebased on top of linux-omap/ams-delta tip, commit da564a05b3aefe403062411b67d64b7e992718f7. There were initially 4 patches in the series, but patch 2/4, "ASoC: cx20442: add bias control over a platform provided regulator", has been since then applied by Liam to the ASoC tree (thanks!) and is already present in mainline (commit f75a8ff67d161b5166a2c2360bb2ffaefd5eb853), hence omitted from this submission. v2 -> v3 changes: * fix section mismatch issue present in previous versions of patch 1/3. v2 changes against initial version: * in both the codec and the modem callbacks, don't track the regulator enable/disable state, compare new target bias level (the codec case) or power state (the modem case) with the old value instead; thanks to Mark Brown who suggested this solution, * a few other minor changes, mostly stylistic. Janusz Krzysztofik (3): ARM: OMAP1: ams-delta: set up regulator over modem reset GPIO pin ARM: OMAP1: ams-delta: update the modem to use regulator API ASoC: OMAP: ams-delta: drop .set_bias_level callback arch/arm/mach-omap1/Kconfig | 2 + arch/arm/mach-omap1/board-ams-delta.c | 94 +++++++++++++++++++-- arch/arm/plat-omap/include/plat/board-ams-delta.h | 1 - sound/soc/omap/ams-delta.c | 32 ------- 4 files changed, 87 insertions(+), 42 deletions(-) -- 1.7.3.4

2 4

[alsa-devel] [PATCH - aplay 1/1] aplay - option to dump HW parameters
by pavel.hofman＠ivitera.com 05 Mar '12

05 Mar '12

From: Pavel Hofman <pavel.hofman(a)ivitera.com> Dumping HW parameters of the current device with snd_pcm_hw_params_dump, a new option --dump-hw-params added. Signed-off-by: Pavel Hofman <pavel.hofman(a)ivitera.com> diff --git a/aplay/aplay.1 b/aplay/aplay.1 index 2acaed5..218d0b5 100644 --- a/aplay/aplay.1 +++ b/aplay/aplay.1 @@ -184,6 +184,9 @@ the month, %H hour, %M minute and %S second. In addition, %v is the file number, starting at 1. When this option is specified, intermediate directories for the output file are created automatically. This option has no effect if \-\-separate\-channels is specified. +.TP +\fI\-\-dump\-hw\-params\fP +Dump HW parameters of the given device to stderr. .SH SIGNALS When recording, SIGINT, SIGTERM and SIGABRT will close the output diff --git a/aplay/aplay.c b/aplay/aplay.c index 73a9544..0633cbd 100644 --- a/aplay/aplay.c +++ b/aplay/aplay.c @@ -129,6 +129,7 @@ static int max_file_time = 0; static int use_strftime = 0; volatile static int recycle_capture_file = 0; static long term_c_lflag = -1; +static int dump_hw_params = 0; static int fd = -1; static off64_t pbrec_count = LLONG_MAX, fdcount; @@ -223,7 +224,8 @@ _("Usage: %s [OPTION]... [FILE]...\n" " --max-file-time=# start another output file when the old file has recorded\n" " for this many seconds\n" " --process-id-file write the process ID here\n" -" --use-strftime apply the strftime facility to the output file name\n") +" --use-strftime apply the strftime facility to the output file name\n" +" --dump-hw-params dump hw_params of the device\n") , command); printf(_("Recognized sample formats are:")); for (k = 0; k < SND_PCM_FORMAT_LAST; ++k) { @@ -416,7 +418,8 @@ enum { OPT_TEST_NOWAIT, OPT_MAX_FILE_TIME, OPT_PROCESS_ID_FILE, - OPT_USE_STRFTIME + OPT_USE_STRFTIME, + OPT_DUMP_HWPARAMS }; int main(int argc, char *argv[]) @@ -461,6 +464,7 @@ int main(int argc, char *argv[]) {"process-id-file", 1, 0, OPT_PROCESS_ID_FILE}, {"use-strftime", 0, 0, OPT_USE_STRFTIME}, {"interactive", 0, 0, 'i'}, + {"dump-hw-params", 0, 0, OPT_DUMP_HWPARAMS}, {0, 0, 0, 0} }; char *pcm_name = "default"; @@ -662,6 +666,9 @@ int main(int argc, char *argv[]) case OPT_USE_STRFTIME: use_strftime = 1; break; + case OPT_DUMP_HWPARAMS: + dump_hw_params = 1; + break; default: fprintf(stderr, _("Try `%s --help' for more information.\n"), command); return 1; @@ -1059,6 +1066,13 @@ static void set_params(void) error(_("Broken configuration for this PCM: no configurations available")); prg_exit(EXIT_FAILURE); } + if (dump_hw_params) { + fprintf(stderr, _("HW Params of device \"%s\":\n"), + snd_pcm_name(handle)); + fprintf(stderr, "--------------------\n"); + snd_pcm_hw_params_dump(params, log); + fprintf(stderr, "--------------------\n"); + } if (mmap_flag) { snd_pcm_access_mask_t *mask = alloca(snd_pcm_access_mask_sizeof()); snd_pcm_access_mask_none(mask); -- 1.7.0.4

3 6

[alsa-devel] [PATCH] ASoC: dapm: Show if widgets are forced in debugfs
by Mark Brown 05 Mar '12

05 Mar '12

The information was not otherwise visible. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-dapm.c | 5 +++-- 1 files changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 3a5adab..60c851b 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1569,8 +1569,9 @@ static ssize_t dapm_widget_power_read_file(struct file *file, out = is_connected_output_ep(w); dapm_clear_walk(w->dapm); - ret = snprintf(buf, PAGE_SIZE, "%s: %s in %d out %d", - w->name, w->power ? "On" : "Off", in, out); + ret = snprintf(buf, PAGE_SIZE, "%s: %s%s in %d out %d", + w->name, w->power ? "On" : "Off", + w->force ? " (forced)" : "", in, out); if (w->reg >= 0) ret += snprintf(buf + ret, PAGE_SIZE - ret, -- 1.7.9.1

2 1

Re: [alsa-devel] multiple, non-physically accesible, HDMI devices listed for Intel IbexPeak ALC269VB
by David Henningsson 05 Mar '12

05 Mar '12

On 03/04/2012 12:36 AM, Andres Cimmarusti wrote: >> There is active work going on in this area. In fact, I just posted a patch >> to the PA mailinglist [1]. And yes, we already have it in Ubuntu 11.10 (to >> probe multiple hdmi devices for Intel and NVidia), and the main reason it >> took until now to upstream that patch, was the decision to switch jack >> detection method from input devices to kcontrols. > > Thank you for all the references you provided and your work in fixing > this issue for all users. I just looked at the git repository for the > source code of pulseaudio, but I see your patches have not been > included yet. Do you have any estimate of when they will be merged? if > so, do you think they'll be included in the next release (do you know > when this will be?) ? I hope they'll be in PulseAudio 2.0, as they are currently waiting for review. For next release, see [2], but judging from the PulseAudio 1.0 release process - no, I don't know when this will be ;-) > I'm considering reassigning this bug to pulseaudio in debian and > asking them to include the appropriate patches. Which ones would > actually be needed (say, to apply them to pulseaudio 1.1)? would your > 6 patches announced on the mailing list in February be enough? If you want them to apply to PulseAudio 1.1, you can have a look at [1]. The patches currently posted apply to git head. You'll need all of the 06* patches (as well as Linux 3.3 for the kcontrols). A more light-weight version could be what I did in Ubuntu 11.04, where there was no jack detection, but I just exposed all four devices in PulseAudio and let the user choose manually, like this [4]. (I later renamed that file from "nvidia.conf" to "extra-hdmi.conf", and added the same file to be used for Intel chips.) >> Let me also push for the hda-jack-retask [2] application, which is an >> easy-to-use GUI for creating these types of firmware files. I advertised it >> here a while ago [3] but it seems to have gone unnoticed. > > This sounds like a good tool for making this happen. I will submit a > Request For Package in Debian... but this can take time. Would you > consider packaging it there? then it would easily flow into Ubuntu. Certainly, if there is interest from the Debian side to have it. I would also not mind if it became a part of upstream ALSA, I think it would make a nice addition to the hda-analyzer, hda-verb etc tool set. > I've encountered other hardware with the same issue recently. It's an > NVIDIA card HDA MCP89 on a Macbook Pro 7,1. Is there a method I can > follow for crafting my own "patches"? I'm afraid I don't understand > how to find the appropriate HEX values that need to go in the [codec] > and [pincfg] section. I think the easiest way is just to download the hda-jack-retask application and build it yourself. Otherwise, [3]. > Thanks all for your help. You're welcome. -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic [1] http://bazaar.launchpad.net/~ubuntu-audio-dev/pulseaudio/ubuntu.precise/fil… [2] http://pulseaudio.org/wiki/ReleasePlanning [3] http://www.kernel.org/doc/Documentation/sound/alsa/HD-Audio.txt [4] http://bazaar.launchpad.net/~ubuntu-audio-dev/pulseaudio/ubuntu.natty/view/…

2 1

[alsa-devel] [PATCH - ice1724 1/1] ALSA: ice1724 - constrain runtime rates for locked internal rate
by pavel.hofman＠ivitera.com 05 Mar '12

05 Mar '12

From: Pavel Hofman <pavel.hofman(a)ivitera.com> The driver already defines control "Multi Track Rate Locking" which locks the card at current rate if switched to internal clock. This patch limits the runtime rates to this rate only, allowing proper reporting of the card capabilities, and e.g. automatic rate conversion by the plug plugin to the currently locked rate. Signed-off-by: Pavel Hofman <pavel.hofman(a)ivitera.com> diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c index 9236297..812d10e 100644 --- a/sound/pci/ice1712/ice1724.c +++ b/sound/pci/ice1712/ice1724.c @@ -1013,6 +1013,25 @@ static int set_rate_constraints(struct snd_ice1712 *ice, ice->hw_rates); } +/* if the card has the internal rate locked (is_pro_locked), limit runtime + hw rates to the current internal rate only. +*/ +static void constrain_rate_if_locked(struct snd_pcm_substream *substream) +{ + struct snd_ice1712 *ice = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + unsigned int rate; + if (is_pro_rate_locked(ice)) { + rate = ice->get_rate(ice); + if (rate >= runtime->hw.rate_min + && rate <= runtime->hw.rate_max) { + runtime->hw.rate_min = rate; + runtime->hw.rate_max = rate; + } + } +} + + /* multi-channel playback needs alignment 8x32bit regardless of the channels * actually used */ @@ -1046,6 +1065,7 @@ static int snd_vt1724_playback_pro_open(struct snd_pcm_substream *substream) VT1724_BUFFER_ALIGN); snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, VT1724_BUFFER_ALIGN); + constrain_rate_if_locked(substream); if (ice->pro_open) ice->pro_open(ice, substream); return 0; @@ -1066,6 +1086,7 @@ static int snd_vt1724_capture_pro_open(struct snd_pcm_substream *substream) VT1724_BUFFER_ALIGN); snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, VT1724_BUFFER_ALIGN); + constrain_rate_if_locked(substream); if (ice->pro_open) ice->pro_open(ice, substream); return 0; @@ -1215,6 +1236,7 @@ static int snd_vt1724_playback_spdif_open(struct snd_pcm_substream *substream) VT1724_BUFFER_ALIGN); snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, VT1724_BUFFER_ALIGN); + constrain_rate_if_locked(substream); if (ice->spdif.ops.open) ice->spdif.ops.open(ice, substream); return 0; @@ -1251,6 +1273,7 @@ static int snd_vt1724_capture_spdif_open(struct snd_pcm_substream *substream) VT1724_BUFFER_ALIGN); snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, VT1724_BUFFER_ALIGN); + constrain_rate_if_locked(substream); if (ice->spdif.ops.open) ice->spdif.ops.open(ice, substream); return 0; -- 1.7.0.4

2 1

[alsa-devel] [PATCH] ALSA: hda - fix broken automute/autoswitch for Realtek
by David Henningsson 05 Mar '12

05 Mar '12

The recent addition of volume-knob widget in the auto-parser broke automute/autoswitch for some Realtek devices. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 01179d5..7e65168 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -686,7 +686,7 @@ static void alc_sku_unsol_event(struct hda_codec *codec, unsigned int res) else res >>= 26; action = snd_hda_jack_get_action(codec, res); - if (res == ALC_DCVOL_EVENT) { + if (action == ALC_DCVOL_EVENT) { /* Execute the dc-vol event here as it requires the NID * but we don't pass NID to alc_exec_unsol_event(). * Once when we convert all static quirks to the auto-parser, -- 1.7.9

2 1

Re: [alsa-devel] ASoC updates for 3.4
by Mark Brown 05 Mar '12

05 Mar '12

The following changes since commit b01543dfe67bb1d191998e90d20534dc354de059: Linux 3.3-rc4 (2012-02-18 15:53:33 -0800) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-3.4 for you to fetch changes up to ad20ff920c1fd217578e2c637dd50c1878a21c06: ASoC: wm8753: fix initialization (2012-03-04 14:34:44 +0000) This should be the major pull request from my side for 3.4. I did send a version of this yesterday based off topic/asoc but the request came out at 190k or so and will have upset at least some mail software so I've redone it against mainline. You already have some of the changes so the diffstat won't quite reflect what you get when you pull. The summary below doesn't include the OMAP stuff which you've already pulled from Liam, the main thing there that I remember is that we've now got a generic driver for OMAP4 reference boards including Panda which is configured via platform data allowing new boards to be supported without driver changes. ---------------------------------------------------------------- This has been a very active release for ASoC, as well as the usual raft of bugfixes and driver updates there's quite a few framework enhancements. Most are either small or are laying the groundwork for user visible features (especially dynamic PCM), the most directly visible change is the dmaengine library. There's also a bunch of regmap API enhancements pulled into the tree so that either the framework or drivers can take advantage of the new features. Changes include: - Support for widgets not associated with a CODEC, an important part of the dynamic PCM framework. - A library factoring out the common code shared by dmaengine based DMA drivers contributed by Lars-Peter Clausen. This will save a lot of code and make it much easier to deploy enhancements to dmaengine. - Support for binary controls, used for providing runtime configuration of algorithm coefficients. - A new DAPM widget type for regulator supplies allowing drivers for devices that can power down unused supplies while active to do without any per-driver code. - DAPM widgets for DAIs, initially giving a speed boost for playback startup and shutdown and also the basis for CODEC<->CODEC DAI link support. - Support for specifying the number of significant bits on audio interfaces, useful for allowing applications to know how much effort to put into generating data for a larger sample format. - Conversion of the FSI driver used on some SH processors to DMAEngine. - New CODEC drivers for Maxim MAX9768 and Wolfson Microelectronics WM2200. ---------------------------------------------------------------- Andrey Danin (9): ASoC: alc5632: Coding style. Remove two extra empty lines. ASoC: alc5632: Fixed voice DAC volume step. ASoC: alc5632: Add voice DAC playback switch ASoC: alc5632: Rename capture switches to common scheme XXX2REC ASoC: alc5632: Remove unexisting route from Phone Mix to Mono Mix ASoC: alc5632: Refactored DAPM routes to add voice support ASoC: alc5632: Add DMIC switches and controls ASoC: alc5632: Fix I2S digital interface power for recording ASoC: alc5632: Allow 8kHz stream support. Axel Lin (12): ASoC: jz4740: Convert qi_lb60 to use snd_soc_register_card() ASoC: Add __devinit/__devexit annotations at necessary places ASoC: Set idle_bias_off flag in snd_soc_codec_driver ASoC: Use dai_fmt in afeb9260 machine driver ASoC: Convert afeb9260 to table based DAPM init ASoC: Convert kirkwood-t5325 to use snd_soc_register_card() ASoC: Convert kirkwood-openrd to use snd_soc_register_card() ASoC: Fix build error in sound/soc/omap/n810.c ASoC: Get correct revision id for wm2200 ASoC: Show device id in the debug message ASoC: Change spitz_ext_control to take dapm as argument. ASoC: Add __devinit annotation for pxa2xx_ac97_probe Denis 'GNUtoo' Carikli (1): ASoC: wm8753: fix initialization Dimitris Papastamos (1): ASoC: wm8983: Remove useless snd_kcontrol Fabio Estevam (3): ASoC: sgtl5000: Print revision number in hex ASoC: sgtl5000: Convert to table based DAPM and control init ASoC: soc-core: Show the returned values on error messages Felipe Contreras (1): ASoC: tlv320aic3x: remove unused code Jassi Brar (1): ASoC: Samsung: Merge two identical if-else clauses Jaswinder Singh (1): ASoC: Samsung: Update email id of the author Javier Martin (2): ASoC: Route Mic Bias in Visstrim_M10 board. ASoC: Add external amplifier controls for Visstrim_M10. Joachim Eastwood (2): ASoC: use proper defines for stream directions in pcm engines ASoC: replace 0xffffffff with DMA_BIT_MASK macro Julia Lawall (1): ASoC: mxs-saif: use devm_ functions Kuninori Morimoto (18): ASoC: fsi: reduce runtime calculation by using pre-setting ASoC: fsi: tidyup: fsi_stream_xx() functions were gathered ASoC: fsi: data push/pop calculation part was divided ASoC: fsi: rename fsi_dma_soft_xxx() to fsi_pio_xxx() ASoC: fsi: tidyup: move fsi_fifo_init() onto fsi_hw_startup() ASoC: fsi: remove unnecessary parameter from fsi_hw_shutdown() ASoC: fsi: rename fsi_stream_push/pop() to fsi_stream_init/quit() ASoC: fsi: modify fsi_pio_get_area() parameter and using position ASoC: fsi: re-define fsi_is_play() and fsi_stream_is_play() ASoC: fsi: use fsi_stream in fsi_get_current_fifo_samples() parameter ASoC: fsi: add fsi_stream_handler and PIO handler ASoC: fsi: tidyup: fsi_pio_xxx() are gathered ASoC: fsi: don't use is_play as a parameter of fsi functions ASoC: fsi: add .start_stop handler to fsi_stream_handler ASoC: fsi: fsi_stream_is_working() care substream->runtime ASoC: fsi: PortA/B information was controlled by sh_fsi_port_info ASoC: fsi: add .init/.quit handler support ASoC: fsi: Add DMAEngine support Lars-Peter Clausen (10): ASoC: blackfin: Use dai_fmt ASoC: imx-pcm: Remove empty prepare callback ASoC: imx-pcm: Remove unused fields from imx_pcm_runtime_data struct ASoC: mxs-pcm: Remove unused fields from struct mxs_pcm_runtime_data ASoC: imx-ssi: Set dma data early ASoC: imx-pcm: Request DMA channel early ASoC: mxs-pcm: Request DMA channel early ASoC: Add dmaengine PCM helper functions ASoC: imx-pcm-dma: Use dmaengine PCM helper functions ASoC: mxs-pcm: Use dmaengine PCM helper functions Leon Romanovsky (7): ASoC: Tegra+ALC5632: Implement device tree support in board file ASoC: ALC5632: Add device tree binding documentation ASoC: Tegra+ALC5632 machine: Add device tree binding documentation ASoC: Tegra+ALC5632: Enable headset autodetection on PAZ00 board. ASoC: tegra: Remove unused DAPM route structure. ASoC: tegra+alc5632: Added digital microphone DAPM widget. ASoC: DT: Update digital microphone binding documentation to PAZ00 board. Liam Girdwood (13): ASoC: core - Improve card registration error messaging for large DAI links. ASoC: twl6040 - add method to query HS DC offset step size in mV ASoC: utils - Add support for a dummy codec driver. ASoC: core: Add support for DAI and machine kcontrols. ASoC: dapm: Export mixer|mux_update_power() to public API. ASoC: dapm: Clean up header information. ASoC: max9768: Fix build and use new add control API. ALSA: PCM - Add PCM creation API for internal PCMs. ASoC: dapm - Make DAPM reset code a separate function. ASoC: dapm: Notify stream event to all card components. ASoC: core: Convert CODEC debugfs init to use dev_warn() ASoC: spitz: Fix kcontrols to use card instead of codec ASoC: core: cleanup platform debugfs on probe failure. Manjunath Hadli (1): ASoC: CQ93VC: remove machine specific header file inclusion from codec driver Marc Dietrich (1): ASoC: alc5632: Connect HP/HPL/HPR mix'es to HPOut Mix Mark Brown (182): ASoC: Make WM8978 I2C usage unconditional ASoC: Remove unused control type from wm8978 driver ASoC: Convert wm8978 to table based DAPM and control init ASoC: Convert wm8978 to devm_kzalloc() ASoC: Use standard cache sync for WM8978 ASoC: Convert wm8978 to direct regmap API usage ASoC: Push wm8978 reset down into the I2C probe ASoC: Convert WM9090 to use regmap directly ASoC: Move WM9090 device identification and reset to I2C probe ASoC: Convert wm8993 to direct regmap API usage ASoC: Move WM8993 resource acquisition and device reset to bus probe ASoC: Convert WM8753 to direct regmap API usage ASoC: Make WM8971 I2C usage unconditional ASoC: Convert wm8971 to table based DAPM and control init ASoC: Convert wm8971 to devm_kzalloc() ASoC: Make wm8974 I2C usage unconditional ASoC: Convert wm8974 to table based DAPM and control init ASoC: Remove wm8974 private data ASoC: Use standard cache sync for wm8804 ASoC: Convert wm8804 to devm_kzalloc() ASoC: Convert wm8804 to direct regmap API usage ASoC: Make WM8904 I2C usage unconditional ASoC: Convert wm8904 to devm_kzalloc() ASoC: Use standard cache sync for WM8904 ASoC: Convert WM8904 to direct regmap API usage ASoC: Make I2C usage unconditional in WM8940 ASoC: Convert WM8940 to devm_kzalloc() ASoC: Make I2C usage unconditional in WM8955 ASoC: Convert WM8955 to devm_kzalloc() ASoC: Use standard register cache sync for WM8955 ASoC: Convert WM8955 to direct regmap API usage ASoC: Convert WM8955 to table based DAPM and control init ASoC: Optimise performance of WM8904 ADC 128fs OSR mode ASoC: Make WM8962 I2C usage unconditional ASoC: Convert WM8962 register access map to modern style ASoC: Add WM8962 DAC and ADC L/R swap support ASoC: Allow drivers to specify how many bits are significant on a DAI ASoC: 24 bits are significant on the WM8996 audio interfaces ASoC: Implement basic WM8993 interrupt support ASoC: Make WM8993 I2C usage unconditional ASoC: 24 bits are significant on wm_hubs DAIs ASoC: When releasing WM5100 put /RESET into reset ASoC: Say we can't read WM5100 ID register ASoC: Remove redundant set_bias_level() from WM5100 remove() ASoC: Make WM5100 interrupt path use regmap directly ASoC: Push WM5100 interrupt request into I2C probe ASoC: Don't tell applications about msbits unless we're ignoring input ASoC: Use regmap update bits operation for drivers using regmap ASoC: Fix build of tlv320dac33 ASoC: wm8985: Convert to devm_kzalloc() ASoC: wm8985: Convert to table based DAPM and control init ASoC: wm8985: Use standard cache sync implementation ASoC: wm8985 Don't directly reference the cache data structure ASoC: wm8985: Convert to direct regmap API usage ASoC: wm8988: Convert to table based DAPM and control init ASoC: wm8988: Convert to devm_kzalloc() regulator: Implement devm_regulator_free() regulator: Provide devm_regulator_bulk_get() ASoC: sgtl5000: It's sgtl5000 not sgtl500 regmap: Support register patch sets ASoC: Convert WM8731 to direct regmap API usage ASoC: wm5100: Fix mismerge of IRQ frees Merge tag 'regmap-patch-initial' of git://git.kernel.org/.../broonie/regmap into for-3.4 ASoC: Convert the WM5100 revision A updates to a regmap patch ASoC: wm5100: Fix warnings from recent patches ASoC: wm8962: Optimise power consumption for IN4 DC measurement usage Merge tag 'topic/devm' of git://git.kernel.org/.../broonie/regulator into HEAD ASoC: Provide REGULATOR_SUPPLY widget type ASoC: wm5100: Move regulator supplies over to DAPM infrastructure ASoC: wm2200: Add WM2200 CODEC driver ASoC: dapm: Drop runtime PM references asynchronously ASoC: dapm: Ignore isolated signal generators for power purposes ASoC: wm5100: Convert to devm_regulator_bulk_get() ASoC: wm5100: Use pm_runtime for powerdown managment ASoC: wm8996: Handle failures to determine accessory polarity ASoC: wm8994: Report any low impedance accessory as a headphone ARM: S3C64XX: Use IN4 DC measurement mode on Tobermory ASoC: wm8962: Enable idle_bias_off ASoC: wm2200: Remove trailing whitespace ASoC: max9768: Fix up review comments in max9768 driver ASoC: fsi: Remove unneeded empty runtime PM callbacks ASoC: wm8993: Convert to use a regmap patch ASoC: core: Support suspend to disk ASoC: wm_hubs: Push check for idle_bias_off out into drivers Merge tag 'v3.3-rc2' into for-3.4 ASoC: wm8996: Switch to using common code for managing CPVDD supply ASoC: wm8996: Use devm_regulator_bulk_get() ASoC: wm5100: Handle failures to determine accessory polarity ASoC: wm9712: Remove driver specific version ASoC: wm9712: Add TLV information for microphone input ASoC: pcm: Improve error logging ASoC: samsung: Enable accessory detect for WM8994 on Littlemill ASoC: wm8994: Don't bring up to STANDBY by default ASoC: wm8994: We don't need to runtime resume by default ASoC: wm8994: Unsuspend the device while reading GPIO statuses ASoC: wm8994: Bring WM8994 accessory detection up to date ASoC: tlv320aic3x: Remove even more unused code ASoC: wm_hubs: Convert most output drivers to OUT_DRV widgets ASoC: wm8962: Support mono playback and record ASoC: pcm: If pmdown_time is zero then shut down DAPM immediately ASoC: core: Allow CODECs to set ignore_pmdown_time in the driver struct ASoC: twl6040: Move ignore_pmdown_time to driver struct ASoC: wm2200: Ignore pmdown_time Merge branch 'topic/pcm-internal' of git://git.kernel.org/.../tiwai/sound into for-3.4 Merge tag 'v3.3-rc3' as we've got several bugfixes in there which are colliding annoyingly with development. ASoC: wm8993: Correct typos in regmap conversion ASoC: wm8993: Add register default for INPUTS_CLAMP ASoC: wm_hubs: Improve single ended line output enable performance ASoC: wm_hubs: Clamp inputs to VMID while we ramp ASoC: wm8994: VMID management improvements ASoC: wm8994: Actively discharge VMID when not in use ASoC: wm9081: Use module_i2c_driver ASoC: wm8988: Convert to direct regmap API usage ASoC: wm8988: Remove unneded -codec from driver name ASoC: wm9081: Move WM9081 IRQ platform data handling into I2C probe ASoC: wm9081: Use idle_bias_off ASoC: wm8994: Use slow start for VMID ASoC: dapm: Convert pin switches to use snd_soc_card ASoC: wm8962: Update the clocking when setting system clock ASoC: wm8962: Only configure BCLK in hw_params when audio is active ASoC: wm8962: Log the selected SYSCLK ratio ASoC: wm8962: Add new SYSCLK ratios for new device revisions ASoC: wm8962: Clean up register dump cruft ASoC: wm8962: Don't automatically enable and disable FLL ASoC: wm8962: Convert to runtime PM for bias off management ASoC: ak4535: Remove -codec from driver name ASoC: ak4535: Make I2C usage unconditional ASoC: ak4535: Remove bitrotted driver version ASoC: ak4535: Convert to direct regmap API usage ALSA: core: Constify the name in new kcontrols ASoC: dapm: Unexport snd_soc_dapm_new_control() ASoC: dapm: Refactor snd_soc_dapm_new_widget() to return the widget ASoC: dapm: Supply the DAI and substream when calling stream events ASoC: dapm: Constify lots of names that are never modified ASoC: dapm: Implement and instantiate DAI widgets ASoC: dapm: Convert stream events to use DAI widgets ASoC: wm8994: Convert to use DAI widget routing rather than streams ASoC: dapm: Only mark pin widgets as dirty if we actually change state ASoC: wm8962: Convert to module_i2c_driver() ASoC: wm8993: Convert to module_i2c_driver() ASoC: wm8996: Convert to module_i2c_driver() ASoC: wm8996: Make sure we bounce /RESET to reset Merge tag 'v3.3-rc4' into for-3.4 in order to resolve the conflict resolved below within the FSI driver and allow the application of the dmaeengine conversion that depends on this resolution. regmap: Allow users to query the size of register values ASoC: ak4104: Remove uninformative print on probe() ASoC: ak4104: Convert to module_spi_driver() ASoC: wm8776: Add WM8775 device ID to the WM8776 ASoC: wm_hubs: Convert headphone driver to output driver widget ASoC: wm8994: Enable headphone startup mode 1 for WM1811 and WM8958 ASoC: wm8994: Actively discharge idle MICBIAS with jack detect ASoC: wm8994: Support external capacitors on MICBIAS2 with jack detection Merge tag 'topic/introspection' of git://git.kernel.org/.../broonie/regmap into HEAD ASoC: io: Retrieve val_bytes from the regmap API ASoC: core: Add SND_SOC_BYTES control for coefficient blocks ASoC: core: Add support for masking out parts of coefficient blocks ASoC: wm8996: Implement DRC coefficient configuration ASoC: wm5100: Implement DRC coefficient configuration ASoC: wm8996: Convert to use DAPM routes for stream connections ASoC: wm8994: Move wm_hubs callback before we start ramping VMID ASoC: ak4104: Use snd_soc_update_bits() for read/modify/write ASoC: ak4104: Use snd_soc_write() rather than internal write function ASoC: ak4104: Convert to direct regmap API usage ASoC: wm8962: Remove mistakenly committed debug logging ASoC: wm8962: Convert interrupt handler to direct regmap usage ASoC: wm8753: Convert to devm_kzalloc() ASoC: wm8994: Make sure we don't have MICBIAS on during jackdet mode ASoC: wm8996: Fix /RESET bounce ordering ASoC: wm8996: Remove stub register cache ASoC: wm_hubs: Bomb out if we can't read back the DC servo result ASoC: wm8994: Remove stub of register access code ASoC: wm8962: Remove register default for PLL2 ASoC: wm8994: Disable debounce of jack detection on inserted jack ASoC: wm8962: Run the headphone in class G mode when sidetone is enabled ASoC: core: Log a warning when machines use soc-audio ASoC: wm8994: Push wm8994 private data allocation out into device probe ASoC: wm8994: Suppress noop updates of FLL K ASoC: wm_hubs: Disable cache of the DC servo calibration for WM1811 ASoC: wm8994: Make sure we sync DAPM on WM8958 detection mode changes ASoC: wm8994: Disable JACKDET when disabling detecton ASoC: wm8994: Don't bother updating the jackdet mode needlessly ASoC: wm8994: Factor out WM1811A detection mode setting ASoC: core: Split the union for CODEC/platform in the DAI Masanari Iida (1): ASoC: Fix typo in twl4030.c Paul Fertser (2): ASoC: alc5632: Fix Boost Volume TLVs used for the external microphones ASoC: alc5632: Fix Capture/Playback attributes for microphone inputs Paul Gortmaker (1): ASoC: fix trivial build error in mpc5200_dma.c Peter Ujfalusi (6): ASoC: tlv320dac33: Use core to set the msbits constraint ASoC: twl4030: Use core to set the msbits constraint ASoC: omap-dmic: Use core to set the msbits constraint ASoC: omap-mcpdm: Set 24msbits constraint ASoC: soc-pcm: msbits constraint: Drop 8 and 16 bit sample sizes ASoC: twl4030: Debug code cleanup Ryan Mallon (1): ep93xx: Don't use system controller defines in audio drivers Sebastien Guiriec (1): ASoC: core: add platform DAPM debugfs support Shawn Guo (8): ASoC: imx/mxs: remove redundant SND_PCM selection ASoC: imx: let SND_MXC_SOC_FIQ select FIQ ASoC: fsl: correct get_dma_channel parameter name ASoC: fsl: align mpc8610_hpcd with p1022_ds on getting codec node ASoC: Remove unnecessary -codec from cs4270 driver name ASoC: imx: initialize dma_params burstsize just in imx-ssi ASoC: imx: move SND_SOC_AC97_BUS selection down to machine driver ASoC: sgtl5000: rename device tree binding document Stephen Boyd (1): regulator: Add devm_regulator_get() Viresh Kumar (1): ASoC: core: Don't overwrite .poweroff in snd_soc_pm_ops Wolfram Sang (1): ASoC: max9768: add driver for max9768 amplifier .../devicetree/bindings/sound/alc5632.txt | 24 + .../{soc/codecs/fsl-sgtl5000.txt => sgtl5000.txt} | 0 .../bindings/sound/tegra-audio-alc5632.txt | 59 + .../devicetree/bindings/vendor-prefixes.txt | 1 + Documentation/driver-model/devres.txt | 3 + arch/arm/mach-ep93xx/core.c | 19 +- arch/arm/mach-ep93xx/include/mach/platform.h | 2 +- arch/arm/mach-kirkwood/openrd-setup.c | 6 + arch/arm/mach-kirkwood/t5325-setup.c | 6 + arch/arm/mach-s3c64xx/mach-crag6410-module.c | 1 + arch/arm/mach-shmobile/board-ap4evb.c | 30 +- arch/arm/mach-shmobile/board-mackerel.c | 18 +- arch/mips/jz4740/board-qi_lb60.c | 6 + arch/sh/boards/mach-ecovec24/setup.c | 4 +- arch/sh/boards/mach-se/7724/setup.c | 4 +- drivers/base/regmap/internal.h | 3 + drivers/base/regmap/regcache.c | 11 + drivers/base/regmap/regmap.c | 73 + drivers/regulator/core.c | 108 + include/linux/mfd/wm8994/pdata.h | 3 + include/linux/regmap.h | 4 + include/linux/regulator/consumer.h | 12 + include/sound/control.h | 2 +- include/sound/dmaengine_pcm.h | 49 + include/sound/max9768.h | 24 + include/sound/pcm.h | 4 + include/sound/sh_fsi.h | 12 +- include/sound/soc-dai.h | 11 +- include/sound/soc-dapm.h | 33 +- include/sound/soc.h | 44 +- include/sound/wm2200.h | 41 + include/sound/wm8962.h | 6 + sound/core/pcm.c | 99 +- sound/soc/Kconfig | 3 + sound/soc/Makefile | 3 + sound/soc/atmel/atmel-pcm.c | 4 +- sound/soc/atmel/snd-soc-afeb9260.c | 37 +- sound/soc/blackfin/bf5xx-ad1836.c | 17 +- sound/soc/blackfin/bf5xx-ad193x.c | 17 +- sound/soc/blackfin/bf5xx-ad73311.c | 29 +- sound/soc/blackfin/bf5xx-ssm2602.c | 20 +- sound/soc/blackfin/bfin-eval-adau1373.c | 13 +- sound/soc/blackfin/bfin-eval-adau1701.c | 16 +- sound/soc/blackfin/bfin-eval-adav80x.c | 13 +- sound/soc/codecs/Kconfig | 8 + sound/soc/codecs/Makefile | 4 + sound/soc/codecs/ad1836.c | 6 +- sound/soc/codecs/ad1980.c | 2 +- sound/soc/codecs/adau1373.c | 7 +- sound/soc/codecs/adau1701.c | 2 +- sound/soc/codecs/ak4104.c | 174 +- sound/soc/codecs/ak4535.c | 98 +- sound/soc/codecs/ak4535.h | 2 - sound/soc/codecs/ak4642.c | 2 +- sound/soc/codecs/ak4671.c | 2 +- sound/soc/codecs/alc5623.c | 12 +- sound/soc/codecs/alc5632.c | 197 +- sound/soc/codecs/alc5632.h | 1 + sound/soc/codecs/cq93vc.c | 4 +- sound/soc/codecs/cs4270.c | 4 +- sound/soc/codecs/cs4271.c | 2 +- sound/soc/codecs/lm4857.c | 2 +- sound/soc/codecs/max9768.c | 247 ++ sound/soc/codecs/max98088.c | 4 +- sound/soc/codecs/max98095.c | 6 +- sound/soc/codecs/max9877.c | 2 +- sound/soc/codecs/sgtl5000.c | 19 +- sound/soc/codecs/sn95031.c | 5 +- sound/soc/codecs/ssm2602.c | 2 +- sound/soc/codecs/stac9766.c | 2 +- sound/soc/codecs/tlv320aic23.c | 2 +- sound/soc/codecs/tlv320aic26.c | 2 +- sound/soc/codecs/tlv320aic32x4.c | 2 +- sound/soc/codecs/tlv320aic3x.c | 66 +- sound/soc/codecs/tlv320aic3x.h | 9 - sound/soc/codecs/tlv320dac33.c | 10 +- sound/soc/codecs/tpa6130a2.c | 4 +- sound/soc/codecs/twl4030.c | 42 +- sound/soc/codecs/twl6040.c | 15 +- sound/soc/codecs/twl6040.h | 1 + sound/soc/codecs/uda134x.c | 6 +- sound/soc/codecs/wl1273.c | 2 +- sound/soc/codecs/wm2200.c | 2286 ++++++++++++ sound/soc/codecs/wm2200.h | 3674 ++++++++++++++++++++ sound/soc/codecs/wm5100.c | 643 ++-- sound/soc/codecs/wm8731.c | 109 +- sound/soc/codecs/wm8737.c | 2 +- sound/soc/codecs/wm8753.c | 195 +- sound/soc/codecs/wm8770.c | 5 +- sound/soc/codecs/wm8776.c | 8 +- sound/soc/codecs/wm8804.c | 148 +- sound/soc/codecs/wm8904.c | 856 ++--- sound/soc/codecs/wm8904.h | 11 + sound/soc/codecs/wm8940.c | 16 +- sound/soc/codecs/wm8955.c | 247 +- sound/soc/codecs/wm8958-dsp2.c | 14 +- sound/soc/codecs/wm8960.c | 2 +- sound/soc/codecs/wm8961.c | 2 +- sound/soc/codecs/wm8962.c | 2125 +++++------- sound/soc/codecs/wm8971.c | 37 +- sound/soc/codecs/wm8974.c | 45 +- sound/soc/codecs/wm8978.c | 185 +- sound/soc/codecs/wm8978.h | 2 + sound/soc/codecs/wm8983.c | 5 - sound/soc/codecs/wm8985.c | 315 +- sound/soc/codecs/wm8988.c | 171 +- sound/soc/codecs/wm8990.c | 2 +- sound/soc/codecs/wm8991.c | 2 +- sound/soc/codecs/wm8993.c | 649 +++-- sound/soc/codecs/wm8993.h | 9 + sound/soc/codecs/wm8994.c | 304 ++- sound/soc/codecs/wm8994.h | 5 +- sound/soc/codecs/wm8995.c | 4 +- sound/soc/codecs/wm8996.c | 200 +- sound/soc/codecs/wm9081.c | 80 +- sound/soc/codecs/wm9090.c | 272 +- sound/soc/codecs/wm9705.c | 2 +- sound/soc/codecs/wm9712.c | 16 +- sound/soc/codecs/wm9713.c | 2 +- sound/soc/codecs/wm_hubs.c | 148 +- sound/soc/codecs/wm_hubs.h | 12 + sound/soc/davinci/davinci-pcm.c | 4 +- sound/soc/ep93xx/edb93xx.c | 4 +- sound/soc/ep93xx/ep93xx-pcm.c | 4 +- sound/soc/ep93xx/snappercl15.c | 4 +- sound/soc/fsl/fsl_dma.c | 10 +- sound/soc/fsl/mpc5200_dma.c | 17 +- sound/soc/fsl/mpc8610_hpcd.c | 16 +- sound/soc/fsl/p1022_ds.c | 4 +- sound/soc/imx/Kconfig | 6 +- sound/soc/imx/imx-pcm-dma-mx2.c | 221 +- sound/soc/imx/imx-ssi.c | 30 +- sound/soc/imx/mx27vis-aic32x4.c | 104 + sound/soc/jz4740/qi_lb60.c | 56 +- sound/soc/kirkwood/kirkwood-dma.c | 4 +- sound/soc/kirkwood/kirkwood-openrd.c | 46 +- sound/soc/kirkwood/kirkwood-t5325.c | 47 +- sound/soc/mid-x86/mfld_machine.c | 2 +- sound/soc/mxs/Kconfig | 2 +- sound/soc/mxs/mxs-pcm.c | 157 +- sound/soc/mxs/mxs-pcm.h | 16 - sound/soc/mxs/mxs-saif.c | 51 +- sound/soc/omap/ams-delta.c | 2 +- sound/soc/omap/n810.c | 17 +- sound/soc/omap/omap-dmic.c | 7 +- sound/soc/omap/omap-mcbsp.c | 8 +- sound/soc/omap/omap-mcbsp.h | 2 +- sound/soc/omap/omap-mcpdm.c | 2 + sound/soc/omap/rx51.c | 25 +- sound/soc/pxa/corgi.c | 14 +- sound/soc/pxa/magician.c | 2 +- sound/soc/pxa/poodle.c | 14 +- sound/soc/pxa/pxa2xx-ac97.c | 10 +- sound/soc/pxa/raumfeld.c | 2 +- sound/soc/pxa/spitz.c | 14 +- sound/soc/pxa/tosa.c | 2 +- sound/soc/s6000/s6000-pcm.c | 5 +- sound/soc/samsung/ac97.c | 4 +- sound/soc/samsung/dma.c | 2 +- sound/soc/samsung/i2s.c | 14 +- sound/soc/samsung/i2s.h | 2 +- sound/soc/samsung/littlemill.c | 3 + sound/soc/samsung/neo1973_wm8753.c | 4 +- sound/soc/samsung/pcm.c | 4 +- sound/soc/samsung/s3c24xx_simtec.c | 6 +- sound/soc/samsung/smdk_wm8580.c | 4 +- sound/soc/samsung/smdk_wm9713.c | 4 +- sound/soc/sh/fsi.c | 912 ++++-- sound/soc/soc-core.c | 379 ++- sound/soc/soc-dapm.c | 393 ++- sound/soc/soc-dmaengine-pcm.c | 287 ++ sound/soc/soc-io.c | 7 + sound/soc/soc-pcm.c | 104 +- sound/soc/soc-utils.c | 20 +- sound/soc/tegra/tegra_alc5632.c | 130 +- sound/soc/tegra/tegra_pcm.c | 2 +- 176 files changed, 12663 insertions(+), 5230 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/alc5632.txt rename Documentation/devicetree/bindings/sound/{soc/codecs/fsl-sgtl5000.txt => sgtl5000.txt} (100%) create mode 100644 Documentation/devicetree/bindings/sound/tegra-audio-alc5632.txt create mode 100644 include/sound/dmaengine_pcm.h create mode 100644 include/sound/max9768.h create mode 100644 include/sound/wm2200.h create mode 100644 sound/soc/codecs/max9768.c create mode 100644 sound/soc/codecs/wm2200.c create mode 100644 sound/soc/codecs/wm2200.h create mode 100644 sound/soc/soc-dmaengine-pcm.c

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[alsa-devel] Query About regmap SPI
by Ashish Chavan 05 Mar '12

05 Mar '12

Hi, I am planning to add SPI support for da7210 codec driver. SPI for this codec has slightly different format than others. It has 8 bit register address and 8 bit data. Out of 8 bits of register address, one bit (i.e. LSB) is used to indicate read or write. Value of 1 at LSB denotes READ while value of 0 at LSB denotes write operation. Thus in order for SPI READ to work correctly, the register address need to be left shifted by one and then ORed with 0x1. In the same way for SPI WRITE to work correctly, it need to be left shifted by one. I have gone though the regmap sources and found that ORing can be achieved by setting correct read_flag_mask. But I couldn't find anything by which I can tell regmap core to shift the register address before ORing it with read_flag_mask. Am I missing anything here? Is there any other way to achieve this using regmap? The only other way I can think of is providing a wrapper function to read and write in codec driver and doing the shifting job inside that wrapper. This message contains information that may be privileged or confidential and is the property of the KPIT Cummins Infosystems Ltd. It is intended only for the person to whom it is addressed. If you are not the intended recipient, you are not authorized to read, print, retain copy, disseminate, distribute, or use this message or any part thereof. If you receive this message in error, please notify the sender immediately and delete all copies of this message. KPIT Cummins Infosystems Ltd. does not accept any liability for virus infected mails.

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Re: [alsa-devel] Support new HDA codec
by Mark Brown 05 Mar '12

05 Mar '12

On Thu, Mar 01, 2012 at 10:44:23AM +0100, Jorge Fernandez Monteagudo wrote: > Attached it's the patch applied to alsa 1.0.23. A new patch_wolfson.c file > is created. I would like to note that no Wolfson support was received to > develop to this code, and this code is useful for us because we only need Hrm, that's not strictly accurate - we didn't help with the code but I know there's been some help (possibly via the distributor). > the outputs differential and SPDIF. This code must be revisited in order to > support the different inputs, the stereo normal output and the I/O. Looking at the code it seems like most of this is things that I'd really expect the generic driver to be able to cope with assuming a suitable BIOS setup, but looking at the generic driver it seems like it might need some enhancement to do so. Things like the S/PIDF output should really be able to Just Work. Hrm.

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[alsa-devel] [PATCH] ASoC: omap-mcbsp: Configure wakeup in later phase
by Peter Ujfalusi 05 Mar '12

05 Mar '12

Configure the WAKEUPEN register at the same time we configure the rest of the McBSP registers. In case of OMAP3+, if the sysclock has been reconfigured we are going to disable McBSP for the duration of the clock change, which will reset the McBSP registers. The WAKEUPEN register need to be configured later, so the changes will be effective during runtime. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/omap/mcbsp.c | 7 +++---- 1 files changed, 3 insertions(+), 4 deletions(-) diff --git a/sound/soc/omap/mcbsp.c b/sound/soc/omap/mcbsp.c index 5f6c21d..d716793 100644 --- a/sound/soc/omap/mcbsp.c +++ b/sound/soc/omap/mcbsp.c @@ -173,6 +173,9 @@ void omap_mcbsp_config(struct omap_mcbsp *mcbsp, MCBSP_WRITE(mcbsp, XCCR, config->xccr); MCBSP_WRITE(mcbsp, RCCR, config->rccr); } + /* Enable wakeup behavior */ + if (mcbsp->pdata->has_wakeup) + MCBSP_WRITE(mcbsp, WAKEUPEN, XRDYEN | RRDYEN); } /** @@ -479,10 +482,6 @@ int omap_mcbsp_request(struct omap_mcbsp *mcbsp) if (mcbsp->pdata && mcbsp->pdata->ops && mcbsp->pdata->ops->request) mcbsp->pdata->ops->request(mcbsp->id - 1); - /* Enable wakeup behavior */ - if (mcbsp->pdata->has_wakeup) - MCBSP_WRITE(mcbsp, WAKEUPEN, XRDYEN | RRDYEN); - /* * Make sure that transmitter, receiver and sample-rate generator are * not running before activating IRQs. -- 1.7.8.5

2 1

[alsa-devel] [PATCH 0/6] snd-usb endpoint rework, version 6
by Aurélien Leblond 05 Mar '12

05 Mar '12

Hello everyone, My laptop has been in reparation for a while and I just got it back now, so I have been out of sync with any update on the snd-usb module, the support of the M-Audio Fast Track Ultra, and I now need to re-setup my laptop. As far as I can see, the kernel 3.2.5 doesn't have the update of Alsa 1.0.25, so the first thing I did was to compile those two separately (compile first the kernel without any sound support, then compile and install 1.0.25), but the midi was then not working on the M-Audio FTU, the Roland UA-25Ex and my E-Mu USB keyboard. All modules are probed, no messages in dmesg, but jack doesn't see any alsa midi driver. In addition, I cannot make out if Daniel's changes are part of this release of Alsa! My next step was to get a git version of the kernel, add Takashi's branch and Daniel's one on top of it (process I used to do 2 months ago), but I get a conflicting error when pulling/merging Daniel's patch, so I guess it is not in sync with the latest kernel anymore... So my questions are: - Did I do something wrong with the kernel 3.2.5/alsa 1.0.25 that broke the USB Midi or is it a bug? I ran through the mailing list but couldn't see anything mentionned about USB Midi. As far as I can tell, I have all the necessary modules. - What is the status of Daniel's work on the snd-usb module? Again I ran through the changelog of alsa 1.0.25 but couldn't make out if it was integrated or not. So as usual, thanks in advance for your help and if there is anymore information I can provide, please let me know. Aurélien

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Re: [alsa-devel] [PATCH v6 0/2] patch set about the MXS-DMA
by Vinod Koul 05 Mar '12

05 Mar '12

On Wed, 2012-02-29 at 08:13 -0500, Huang Shijie wrote: > Hi Artem: > > On Wed, Feb 29, 2012 at 7:00 AM, Bityutskiy, Artem > <artem.bityutskiy(a)intel.com> wrote: > > On Wed, 2012-02-29 at 16:25 +0800, Huang Shijie wrote: > >> Hi Artem: > >> > On Wed, 2012-02-29 at 11:10 +0800, Huang Shijie wrote: > >> >> Hi: > >> >> > >> >> ping. > >> > Is this patch-set independent or requires something like gpma patches? > >> > > >> This patch set is independent, and it does not depend upon any other > >> patches. > > > > Pushed both to l2-mtd.git tree, thanks. I am worried that it touches > > other subsystems like dma, sound, and mmc and there are no acks for > > those. Would be nice to get them somehow. > > > > I ever thought Vinod could accept this patch set. > > To Vinod & Chris: > Could you give the ack to the first patch? Both, Acked-by: Vinod Koul <vinod.koul(a)linux.intel.com> Although the second patch overloads the dmaengine flags but for lack of any other suitable alternative, this looks okay... > > Does the second patch need extra acks beside Shawn's? > > BR > Huang Shijie > -- > To unsubscribe from this list: send the line "unsubscribe linux-kernel" in > the body of a message to majordomo(a)vger.kernel.org > More majordomo info at http://vger.kernel.org/majordomo-info.html > Please read the FAQ at http://www.tux.org/lkml/ -- ~Vinod

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[alsa-devel] [PATCH 1/2] ASoC: ak4641: Convert to module_i2c_driver()
by Mark Brown 04 Mar '12

04 Mar '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/ak4641.c | 18 +----------------- 1 files changed, 1 insertions(+), 17 deletions(-) diff --git a/sound/soc/codecs/ak4641.c b/sound/soc/codecs/ak4641.c index c4d165a..611e8f0 100644 --- a/sound/soc/codecs/ak4641.c +++ b/sound/soc/codecs/ak4641.c @@ -641,23 +641,7 @@ static struct i2c_driver ak4641_i2c_driver = { .id_table = ak4641_i2c_id, }; -static int __init ak4641_modinit(void) -{ - int ret; - - ret = i2c_add_driver(&ak4641_i2c_driver); - if (ret != 0) - pr_err("Failed to register AK4641 I2C driver: %d\n", ret); - - return ret; -} -module_init(ak4641_modinit); - -static void __exit ak4641_exit(void) -{ - i2c_del_driver(&ak4641_i2c_driver); -} -module_exit(ak4641_exit); +module_i2c_driver(ak4641_i2c_driver); MODULE_DESCRIPTION("SoC AK4641 driver"); MODULE_AUTHOR("Harald Welte <laforge(a)gnufiish.org>"); -- 1.7.9.1

1 1

[alsa-devel] [PATCH v4 0/2] MFD: twl6040: Conversion to i2c driver
by Peter Ujfalusi 04 Mar '12

04 Mar '12

Hello, Changes since v3: - Combined the main patch (conversion to i2c driver) with changes in Kconfig files to avoid build breakage on other platforms - Added the received Acks from maintainers Samuel: is it possible for you to stop carrying the twl6040 changes you have at the moment in your for-next branch (can you drop the patches)? This series will make sure I will not brake x86_64 allmodconfig build. As we agreed this series will go via the audio tree (Liam) since it has dependency on earlier work of mine going through ASoC tree. Thank you. ------ Intro mail from the original series: This series will convert the twl6040 MFD driver to an i2c driver. Compared to older twl4030/5030/TPS the twl6040 is a standalone audio IC. It is better if the twl6040-core (and all of it's child devices) does not depend on the twl-core since it has nothing to with it. With this conversion the dependency on twl can be dropped from the twl6040 driver stack (core, vibra, audio). Between the first and second patch the audio will not probe on OMAP4, but I felt it is better this way at least for the first RFC series to not have too big change within one patch. The two patch can be squashed together later if no objections. Regards, Peter --- Peter Ujfalusi (2): MFD: twl-core: Detach twl6040 from the pmic mfd driver MFD: twl6040: Convert to i2c driver, and separate it from twl core arch/arm/mach-omap2/board-4430sdp.c | 12 ++-- arch/arm/mach-omap2/board-generic.c | 2 +- arch/arm/mach-omap2/board-omap4panda.c | 13 ++-- arch/arm/mach-omap2/twl-common.c | 37 +++++++++-- arch/arm/mach-omap2/twl-common.h | 10 +-- drivers/input/misc/Kconfig | 3 +- drivers/input/misc/twl6040-vibra.c | 4 +- drivers/mfd/Kconfig | 11 +++- drivers/mfd/twl-core.c | 58 +++++++++------- drivers/mfd/twl6040-core.c | 114 +++++++++++++++++++------------- include/linux/i2c/twl.h | 12 ---- include/linux/mfd/twl6040.h | 27 ++++++++ sound/soc/codecs/Kconfig | 3 +- sound/soc/codecs/twl6040.c | 3 +- sound/soc/omap/Kconfig | 2 +- 15 files changed, 192 insertions(+), 119 deletions(-) -- 1.7.8.4

2 3

[alsa-devel] ASoC fix for 3.3
by Mark Brown 04 Mar '12

04 Mar '12

The following changes since commit 5ed80a75b248bfaf840ea6b38f941edcf6ee7dc7: ASoC: i.MX SSI: Fix DSP_A format. (2012-02-23 18:58:17 +0000) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-3.3 for you to fetch changes up to b2ccf065f7b23147ed135a41b01d05a332ca6b7e: ASoC: neo1973: fix neo1973 wm8753 initialization (2012-03-04 14:30:18 +0000) ---------------------------------------------------------------- A driver specific fix that wasn't noticed as the OpenMoko guys have been stuck on 2.6.39 for a very long time now and are just starting to catch up again. ---------------------------------------------------------------- Denis 'GNUtoo' Carikli (1): ASoC: neo1973: fix neo1973 wm8753 initialization sound/soc/samsung/neo1973_wm8753.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-)

2 1

[alsa-devel] [PATCH 1/2] ASoC: fix WM8753 initialization.
by Denis 'GNUtoo' Carikli 04 Mar '12

04 Mar '12

Without that fix the wm8753 SPI initialization fails, and then produces a kernel panic during boot with the following call trace: Unable to handle kernel paging request at virtual address 37386d9b [<c01ccafc>] (regmap_get_val_bytes+0x0/0x14) from [<c0243dfc>] (snd_soc_codec_set_cache_io+0x9c/0xcc) [<c0243dfc>] (snd_soc_codec_set_cache_io+0x9c/0xcc) from [<c0244a4c>] (wm8753_probe+0x5c/0x1c4) [<c0244a4c>] (wm8753_probe+0x5c/0x1c4) from [<c023bb24>] (soc_probe_codec+0x174/0x284) [<c023bb24>] (soc_probe_codec+0x174/0x284) from [<c023c2c0>] (snd_soc_instantiate_cards+0x68c/0xe28) [<c023c2c0>] (snd_soc_instantiate_cards+0x68c/0xe28) from [<c023d278>] (snd_soc_register_card+0x240/0x2d4) [<c023d278>] (snd_soc_register_card+0x240/0x2d4) from [<c023d330>] (soc_probe+0x24/0x40) [<c023d330>] (soc_probe+0x24/0x40) from [<c01c3900>] (platform_drv_probe+0x14/0x18) [...] The commit d3398ff05907167f463e119421b053ce043741d1 ( ASoC: Convert WM8753 to direct regmap API usage ) introduced the problem. Thanks to Lars-Peter Clausen for helping me a bit during the debugging. Signed-off-by: Denis 'GNUtoo' Carikli <GNUtoo(a)no-log.org> --- sound/soc/codecs/wm8753.c | 6 ++++++ 1 files changed, 6 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index 21ed75d..98fb921 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -1577,6 +1577,9 @@ static int __devinit wm8753_spi_probe(struct spi_device *spi) dev_err(&spi->dev, "Failed to register CODEC: %d\n", ret); goto err_regmap; } + + return 0; + err_regmap: regmap_exit(wm8753->regmap); err: @@ -1632,6 +1635,9 @@ static __devinit int wm8753_i2c_probe(struct i2c_client *i2c, dev_err(&i2c->dev, "Failed to register CODEC: %d\n", ret); goto err_regmap; } + + return 0; + err_regmap: regmap_exit(wm8753->regmap); err: -- 1.7.4.1

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[alsa-devel] [PATCH] ASoC: wm8994: Suppres noop updates of FLL K
by Mark Brown 03 Mar '12

03 Mar '12

Using snd_soc_write() means we always write to the register even if it already contains the newly calculated value. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 3 ++- 1 files changed, 2 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 3009541..6d351b9 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -1920,7 +1920,8 @@ static int _wm8994_set_fll(struct snd_soc_codec *codec, int id, int src, WM8994_FLL1_OUTDIV_MASK | WM8994_FLL1_FRATIO_MASK, reg); - snd_soc_write(codec, WM8994_FLL1_CONTROL_3 + reg_offset, fll.k); + snd_soc_update_bits(codec, WM8994_FLL1_CONTROL_3 + reg_offset, + WM8994_FLL1_K_MASK, fll.k); snd_soc_update_bits(codec, WM8994_FLL1_CONTROL_4 + reg_offset, WM8994_FLL1_N_MASK, -- 1.7.9.1

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Re: [alsa-devel] ALC892, Clevo P150HM: Input Source gets reset to "Mic" every resume
by Julian Sikorski 03 Mar '12

03 Mar '12

W dniu 02.03.2012 23:08, Greg KH pisze: > On Fri, Mar 02, 2012 at 11:01:36PM +0100, Julian Sikorski wrote: >> W dniu 02.03.2012 19:43, Greg KH pisze: >>> On Fri, Mar 02, 2012 at 07:37:32PM +0100, Julian Sikorski wrote: >>>> W dniu 25.02.2012 11:16, Takashi Iwai pisze: >>>>> At Fri, 24 Feb 2012 23:06:26 +0100, >>>>> Julian Sikorski wrote: >>>>>> >>>>>> W dniu 24.02.2012 10:43, Takashi Iwai pisze: >>>>>>> At Thu, 23 Feb 2012 20:11:44 +0100, >>>>>>> Julian Sikorski wrote: >>>>>>>> >>>>>>>> W dniu 09.12.2011 14:32, Raymond Yau pisze: >>>>>>>>> 2011/12/4 Julian Sikorski <belegdol(a)gmail.com>: >>>>>>>>>> Hi, >>>>>>>>>> >>>>>>>>>> whenever I resume the machine, the Input Source would get changed back >>>>>>>>>> from "Internal Mic" to "Mic". This is a bit annoying, because pulseaudio >>>>>>>>>> is unaware of that change. As a result, everything looks fine in >>>>>>>>>> pavucontrol but no sound gets recorded. A quick look at alsamixer -c0 >>>>>>>>>> revealed the problem. My alsa info: >>>>>>>>>> >>>>>>>>>> http://www.alsa-project.org/db/?f=c5776b615a5f3b9ae624e4250fb0c2b69118ce26 >>>>>>>>>> >>>>>>>>>> This is unrelated to the other problems I have been reporting, since >>>>>>>>>> this happens on a mostly-stock Fedora 3.1.2 kernel (the patch I added >>>>>>>>>> deals with xhci, not alsa). >>>>>>>>> >>>>>>>>> The problem seem can be reproduced by using hda-emu, >>>>>>>>> it is init_capsrc_for_pin() and alc_auto_init_input_src() write value >>>>>>>>> to cache before snd_hda_codec_resume_app() from cache >>>>>>>>> >>>>>>>>> >>>>>>>>>> get 28 >>>>>>>>> 28 Input Source:0 >>>>>>>>> ITEM: 0:Mic, 1:Internal Mic, 2:Line, VAL: [Mic] >>>>>>>>>> set 28 1 >>>>>>>>> send: NID=0x23, VERB=0x360(set_amp_gain_mute,I:L#0), PARM=0x80 >>>>>>>>> send: NID=0x23, VERB=0x350(set_amp_gain_mute,I:R#0), PARM=0x80 >>>>>>>>> send: NID=0x23, VERB=0x361(set_amp_gain_mute,I:L#1), PARM=0x0 >>>>>>>>> send: NID=0x23, VERB=0x351(set_amp_gain_mute,I:R#1), PARM=0x0 >>>>>>>>>> get 28 >>>>>>>>> 28 Input Source:0 >>>>>>>>> ITEM: 0:Mic, 1:Internal Mic, 2:Line, VAL: [Internal Mic] >>>>>>>>> >>>>>>>>> pm >>>>>>>>> ** SUSPENDING ** >>>>>>>>> ... >>>>>>>>> ** RESUMING ** >>>>>>>>> PM-Notified >>>>>>>>> ... >>>>>>>>> exec_init_verbs >>>>>>>>> ... >>>>>>>>> >>>>>>>>> init_capsrc_for_pin 18 >>>>>>>>> send: NID=0x23, VERB=0xb20(get_amp_gain_mute,I:L#0), PARM=0x0 >>>>>>>>> receive: 0x0 >>>>>>>>> send: NID=0x23, VERB=0xb00(get_amp_gain_mute,I:R#0), PARM=0x0 >>>>>>>>> receive: 0x0 >>>>>>>>> init capsrc for pin 19 >>>>>>>>> send: NID=0x23, VERB=0xb20(get_amp_gain_mute,I:L#0), PARM=0x1 >>>>>>>>> receive: 0x80 >>>>>>>>> send: NID=0x23, VERB=0x361(set_amp_gain_mute,I:L#1), PARM=0x0 >>>>>>>>> send: NID=0x23, VERB=0xb00(get_amp_gain_mute,I:R#0), PARM=0x1 >>>>>>>>> receive: 0x80 >>>>>>>>> send: NID=0x23, VERB=0x351(set_amp_gain_mute,I:R#1), PARM=0x0 >>>>>>>>> init capsrc for pin 1a >>>>>>>>> send: NID=0x23, VERB=0xb20(get_amp_gain_mute,I:L#0), PARM=0x2 >>>>>>>>> receive: 0x80 >>>>>>>>> send: NID=0x23, VERB=0x362(set_amp_gain_mute,I:L#2), PARM=0x0 >>>>>>>>> >>>>>>>>> alc_auto_init_input_src >>>>>>>>> send: NID=0x23, VERB=0x361(set_amp_gain_mute,I:L#1), PARM=0x80 >>>>>>>>> send: NID=0x23, VERB=0x351(set_amp_gain_mute,I:R#1), PARM=0x80 >>>>>>>>> send: NID=0x23, VERB=0x362(set_amp_gain_mute,I:L#2), PARM=0x80 >>>>>>>>> send: NID=0x23, VERB=0x352(set_amp_gain_mute,I:R#2), PARM=0x80 >>>>>>>>> >>>>>>>>> snd_hda_codec_resume_app >>>>>>>>> send: NID=0x23, VERB=0x363(set_amp_gain_mute,I:L#3), PARM=0x80 >>>>>>>>> send: NID=0x23, VERB=0x353(set_amp_gain_mute,I:R#3), PARM=0x80 >>>>>>>>> send: NID=0x23, VERB=0x364(set_amp_gain_mute,I:L#4), PARM=0x80 >>>>>>>>> send: NID=0x23, VERB=0x354(set_amp_gain_mute,I:R#4), PARM=0x80 >>>>>>>>> send: NID=0x23, VERB=0x365(set_amp_gain_mute,I:L#5), PARM=0x80 >>>>>>>>> send: NID=0x23, VERB=0x355(set_amp_gain_mute,I:R#5), PARM=0x80 >>>>>>>>> send: NID=0x23, VERB=0x366(set_amp_gain_mute,I:L#6), PARM=0x80 >>>>>>>>> send: NID=0x23, VERB=0x356(set_amp_gain_mute,I:R#6), PARM=0x80 >>>>>>>>> send: NID=0x23, VERB=0x367(set_amp_gain_mute,I:L#7), PARM=0x80 >>>>>>>>> send: NID=0x23, VERB=0x357(set_amp_gain_mute,I:R#7), PARM=0x80 >>>>>>>>> send: NID=0x23, VERB=0x368(set_amp_gain_mute,I:L#8), PARM=0x80 >>>>>>>>> send: NID=0x23, VERB=0x358(set_amp_gain_mute,I:R#8), PARM=0x80 >>>>>>>>> send: NID=0x23, VERB=0x369(set_amp_gain_mute,I:L#9), PARM=0x80 >>>>>>>>> send: NID=0x23, VERB=0x359(set_amp_gain_mute,I:R#9), PARM=0x80 >>>>>>>>> send: NID=0x23, VERB=0x36a(set_amp_gain_mute,I:L#10), PARM=0x80 >>>>>>>>> send: NID=0x23, VERB=0x35a(set_amp_gain_mute,I:R#10), PARM=0x80 >>>>>>>>> ... >>>>>>>>> >>>>>>>>>> get 28 >>>>>>>>> 28 Input Source:0 >>>>>>>>> ITEM: 0:Mic, 1:Internal Mic, 2:Line, VAL: [Mic] >>>>>>>> The problem still exists in 3.2.7-1.fc16.x86_64 kernel, is there any >>>>>>>> chance for a fix? >>>>>>> >>>>>>> Does the patch below fix? >>>>>>> >>>>>>> >>>>>>> Takashi >>>>>> >>>>>> Yes, it seems so. I don't want to get ahead of myself, but two >>>>>> suspend-resume cycles have passed and inputs are at Internal Mic and >>>>>> Mic, as expected. >>>>> >>>>> Thanks for checking. I committed the fix to git tree, also Cc'ed to >>>>> stable. >>>>> >>>>> >>>>> Takashi >>>> I cannot seem to find this patch in the 3.2 stable tree, was it rejected? >>> >>> What is the git commit id of it in Linus's tree? >> >> Hmm, it does not seem to be in Linus' tree yet. It is in for-linus >> branch of sound tree under commit 068b9394 though. > > Then there is nothing I can do about it at this point in time. > > greg k-h Dear Greg, it seems the patch has been merged to the Linus' tree: http://git.kernel.org/?p=linux/kernel/git/torvalds/linux.git;a=commit;h=068… Regards, Julian

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[alsa-devel] [PATCH] ASoC: wm8994: Don't bother updating the jackdet mode needlessly
by Mark Brown 03 Mar '12

03 Mar '12

If we're not doing jackdet it's not needed. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 3 +++ 1 files changed, 3 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 89d1819..57acf52 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -686,6 +686,9 @@ static void wm1811_jackdet_set_mode(struct snd_soc_codec *codec, u16 mode) { struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + if (!wm8994->jackdet || !wm8994->jack_cb) + return; + if (wm8994->active_refcount) mode = WM1811_JACKDET_MODE_AUDIO; -- 1.7.9.1

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Re: [alsa-devel] ALC892, Clevo P150HM: Input Source gets reset to "Mic" every resume
by Julian Sikorski 03 Mar '12

03 Mar '12

W dniu 2012-03-02 23:08, Greg KH pisze: > On Fri, Mar 02, 2012 at 11:01:36PM +0100, Julian Sikorski wrote: >> W dniu 02.03.2012 19:43, Greg KH pisze: >>> On Fri, Mar 02, 2012 at 07:37:32PM +0100, Julian Sikorski wrote: >>>> W dniu 25.02.2012 11:16, Takashi Iwai pisze: >>>>> At Fri, 24 Feb 2012 23:06:26 +0100, >>>>> Julian Sikorski wrote: >>>>>> >>>>>> W dniu 24.02.2012 10:43, Takashi Iwai pisze: >>>>>>> At Thu, 23 Feb 2012 20:11:44 +0100, >>>>>>> Julian Sikorski wrote: >>>>>>>> >>>>>>>> W dniu 09.12.2011 14:32, Raymond Yau pisze: >>>>>>>>> 2011/12/4 Julian Sikorski<belegdol(a)gmail.com>: >>>>>>>>>> Hi, >>>>>>>>>> >>>>>>>>>> whenever I resume the machine, the Input Source would get changed back >>>>>>>>>> from "Internal Mic" to "Mic". This is a bit annoying, because pulseaudio >>>>>>>>>> is unaware of that change. As a result, everything looks fine in >>>>>>>>>> pavucontrol but no sound gets recorded. A quick look at alsamixer -c0 >>>>>>>>>> revealed the problem. My alsa info: >>>>>>>>>> >>>>>>>>>> http://www.alsa-project.org/db/?f=c5776b615a5f3b9ae624e4250fb0c2b69118ce26 >>>>>>>>>> >>>>>>>>>> This is unrelated to the other problems I have been reporting, since >>>>>>>>>> this happens on a mostly-stock Fedora 3.1.2 kernel (the patch I added >>>>>>>>>> deals with xhci, not alsa). >>>>>>>>> >>>>>>>>> The problem seem can be reproduced by using hda-emu, >>>>>>>>> it is init_capsrc_for_pin() and alc_auto_init_input_src() write value >>>>>>>>> to cache before snd_hda_codec_resume_app() from cache >>>>>>>>> >>>>>>>>> >>>>>>>>>> get 28 >>>>>>>>> 28 Input Source:0 >>>>>>>>> ITEM: 0:Mic, 1:Internal Mic, 2:Line, VAL: [Mic] >>>>>>>>>> set 28 1 >>>>>>>>> send: NID=0x23, VERB=0x360(set_amp_gain_mute,I:L#0), PARM=0x80 >>>>>>>>> send: NID=0x23, VERB=0x350(set_amp_gain_mute,I:R#0), PARM=0x80 >>>>>>>>> send: NID=0x23, VERB=0x361(set_amp_gain_mute,I:L#1), PARM=0x0 >>>>>>>>> send: NID=0x23, VERB=0x351(set_amp_gain_mute,I:R#1), PARM=0x0 >>>>>>>>>> get 28 >>>>>>>>> 28 Input Source:0 >>>>>>>>> ITEM: 0:Mic, 1:Internal Mic, 2:Line, VAL: [Internal Mic] >>>>>>>>> >>>>>>>>> pm >>>>>>>>> ** SUSPENDING ** >>>>>>>>> ... >>>>>>>>> ** RESUMING ** >>>>>>>>> PM-Notified >>>>>>>>> ... >>>>>>>>> exec_init_verbs >>>>>>>>> ... >>>>>>>>> >>>>>>>>> init_capsrc_for_pin 18 >>>>>>>>> send: NID=0x23, VERB=0xb20(get_amp_gain_mute,I:L#0), PARM=0x0 >>>>>>>>> receive: 0x0 >>>>>>>>> send: NID=0x23, VERB=0xb00(get_amp_gain_mute,I:R#0), PARM=0x0 >>>>>>>>> receive: 0x0 >>>>>>>>> init capsrc for pin 19 >>>>>>>>> send: NID=0x23, VERB=0xb20(get_amp_gain_mute,I:L#0), PARM=0x1 >>>>>>>>> receive: 0x80 >>>>>>>>> send: NID=0x23, VERB=0x361(set_amp_gain_mute,I:L#1), PARM=0x0 >>>>>>>>> send: NID=0x23, VERB=0xb00(get_amp_gain_mute,I:R#0), PARM=0x1 >>>>>>>>> receive: 0x80 >>>>>>>>> send: NID=0x23, VERB=0x351(set_amp_gain_mute,I:R#1), PARM=0x0 >>>>>>>>> init capsrc for pin 1a >>>>>>>>> send: NID=0x23, VERB=0xb20(get_amp_gain_mute,I:L#0), PARM=0x2 >>>>>>>>> receive: 0x80 >>>>>>>>> send: NID=0x23, VERB=0x362(set_amp_gain_mute,I:L#2), PARM=0x0 >>>>>>>>> >>>>>>>>> alc_auto_init_input_src >>>>>>>>> send: NID=0x23, VERB=0x361(set_amp_gain_mute,I:L#1), PARM=0x80 >>>>>>>>> send: NID=0x23, VERB=0x351(set_amp_gain_mute,I:R#1), PARM=0x80 >>>>>>>>> send: NID=0x23, VERB=0x362(set_amp_gain_mute,I:L#2), PARM=0x80 >>>>>>>>> send: NID=0x23, VERB=0x352(set_amp_gain_mute,I:R#2), PARM=0x80 >>>>>>>>> >>>>>>>>> snd_hda_codec_resume_app >>>>>>>>> send: NID=0x23, VERB=0x363(set_amp_gain_mute,I:L#3), PARM=0x80 >>>>>>>>> send: NID=0x23, VERB=0x353(set_amp_gain_mute,I:R#3), PARM=0x80 >>>>>>>>> send: NID=0x23, VERB=0x364(set_amp_gain_mute,I:L#4), PARM=0x80 >>>>>>>>> send: NID=0x23, VERB=0x354(set_amp_gain_mute,I:R#4), PARM=0x80 >>>>>>>>> send: NID=0x23, VERB=0x365(set_amp_gain_mute,I:L#5), PARM=0x80 >>>>>>>>> send: NID=0x23, VERB=0x355(set_amp_gain_mute,I:R#5), PARM=0x80 >>>>>>>>> send: NID=0x23, VERB=0x366(set_amp_gain_mute,I:L#6), PARM=0x80 >>>>>>>>> send: NID=0x23, VERB=0x356(set_amp_gain_mute,I:R#6), PARM=0x80 >>>>>>>>> send: NID=0x23, VERB=0x367(set_amp_gain_mute,I:L#7), PARM=0x80 >>>>>>>>> send: NID=0x23, VERB=0x357(set_amp_gain_mute,I:R#7), PARM=0x80 >>>>>>>>> send: NID=0x23, VERB=0x368(set_amp_gain_mute,I:L#8), PARM=0x80 >>>>>>>>> send: NID=0x23, VERB=0x358(set_amp_gain_mute,I:R#8), PARM=0x80 >>>>>>>>> send: NID=0x23, VERB=0x369(set_amp_gain_mute,I:L#9), PARM=0x80 >>>>>>>>> send: NID=0x23, VERB=0x359(set_amp_gain_mute,I:R#9), PARM=0x80 >>>>>>>>> send: NID=0x23, VERB=0x36a(set_amp_gain_mute,I:L#10), PARM=0x80 >>>>>>>>> send: NID=0x23, VERB=0x35a(set_amp_gain_mute,I:R#10), PARM=0x80 >>>>>>>>> ... >>>>>>>>> >>>>>>>>>> get 28 >>>>>>>>> 28 Input Source:0 >>>>>>>>> ITEM: 0:Mic, 1:Internal Mic, 2:Line, VAL: [Mic] >>>>>>>> The problem still exists in 3.2.7-1.fc16.x86_64 kernel, is there any >>>>>>>> chance for a fix? >>>>>>> >>>>>>> Does the patch below fix? >>>>>>> >>>>>>> >>>>>>> Takashi >>>>>> >>>>>> Yes, it seems so. I don't want to get ahead of myself, but two >>>>>> suspend-resume cycles have passed and inputs are at Internal Mic and >>>>>> Mic, as expected. >>>>> >>>>> Thanks for checking. I committed the fix to git tree, also Cc'ed to >>>>> stable. >>>>> >>>>> >>>>> Takashi >>>> I cannot seem to find this patch in the 3.2 stable tree, was it rejected? >>> >>> What is the git commit id of it in Linus's tree? >> >> Hmm, it does not seem to be in Linus' tree yet. It is in for-linus >> branch of sound tree under commit 068b9394 though. > > Then there is nothing I can do about it at this point in time. > > greg k-h I see, thanks for clarification. Regards, Julian

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Re: [alsa-devel] ALC892, Clevo P150HM: Input Source gets reset to "Mic" every resume
by Julian Sikorski 03 Mar '12

03 Mar '12

W dniu 02.03.2012 19:43, Greg KH pisze: > On Fri, Mar 02, 2012 at 07:37:32PM +0100, Julian Sikorski wrote: >> W dniu 25.02.2012 11:16, Takashi Iwai pisze: >>> At Fri, 24 Feb 2012 23:06:26 +0100, >>> Julian Sikorski wrote: >>>> >>>> W dniu 24.02.2012 10:43, Takashi Iwai pisze: >>>>> At Thu, 23 Feb 2012 20:11:44 +0100, >>>>> Julian Sikorski wrote: >>>>>> >>>>>> W dniu 09.12.2011 14:32, Raymond Yau pisze: >>>>>>> 2011/12/4 Julian Sikorski <belegdol(a)gmail.com>: >>>>>>>> Hi, >>>>>>>> >>>>>>>> whenever I resume the machine, the Input Source would get changed back >>>>>>>> from "Internal Mic" to "Mic". This is a bit annoying, because pulseaudio >>>>>>>> is unaware of that change. As a result, everything looks fine in >>>>>>>> pavucontrol but no sound gets recorded. A quick look at alsamixer -c0 >>>>>>>> revealed the problem. My alsa info: >>>>>>>> >>>>>>>> http://www.alsa-project.org/db/?f=c5776b615a5f3b9ae624e4250fb0c2b69118ce26 >>>>>>>> >>>>>>>> This is unrelated to the other problems I have been reporting, since >>>>>>>> this happens on a mostly-stock Fedora 3.1.2 kernel (the patch I added >>>>>>>> deals with xhci, not alsa). >>>>>>> >>>>>>> The problem seem can be reproduced by using hda-emu, >>>>>>> it is init_capsrc_for_pin() and alc_auto_init_input_src() write value >>>>>>> to cache before snd_hda_codec_resume_app() from cache >>>>>>> >>>>>>> >>>>>>>> get 28 >>>>>>> 28 Input Source:0 >>>>>>> ITEM: 0:Mic, 1:Internal Mic, 2:Line, VAL: [Mic] >>>>>>>> set 28 1 >>>>>>> send: NID=0x23, VERB=0x360(set_amp_gain_mute,I:L#0), PARM=0x80 >>>>>>> send: NID=0x23, VERB=0x350(set_amp_gain_mute,I:R#0), PARM=0x80 >>>>>>> send: NID=0x23, VERB=0x361(set_amp_gain_mute,I:L#1), PARM=0x0 >>>>>>> send: NID=0x23, VERB=0x351(set_amp_gain_mute,I:R#1), PARM=0x0 >>>>>>>> get 28 >>>>>>> 28 Input Source:0 >>>>>>> ITEM: 0:Mic, 1:Internal Mic, 2:Line, VAL: [Internal Mic] >>>>>>> >>>>>>> pm >>>>>>> ** SUSPENDING ** >>>>>>> ... >>>>>>> ** RESUMING ** >>>>>>> PM-Notified >>>>>>> ... >>>>>>> exec_init_verbs >>>>>>> ... >>>>>>> >>>>>>> init_capsrc_for_pin 18 >>>>>>> send: NID=0x23, VERB=0xb20(get_amp_gain_mute,I:L#0), PARM=0x0 >>>>>>> receive: 0x0 >>>>>>> send: NID=0x23, VERB=0xb00(get_amp_gain_mute,I:R#0), PARM=0x0 >>>>>>> receive: 0x0 >>>>>>> init capsrc for pin 19 >>>>>>> send: NID=0x23, VERB=0xb20(get_amp_gain_mute,I:L#0), PARM=0x1 >>>>>>> receive: 0x80 >>>>>>> send: NID=0x23, VERB=0x361(set_amp_gain_mute,I:L#1), PARM=0x0 >>>>>>> send: NID=0x23, VERB=0xb00(get_amp_gain_mute,I:R#0), PARM=0x1 >>>>>>> receive: 0x80 >>>>>>> send: NID=0x23, VERB=0x351(set_amp_gain_mute,I:R#1), PARM=0x0 >>>>>>> init capsrc for pin 1a >>>>>>> send: NID=0x23, VERB=0xb20(get_amp_gain_mute,I:L#0), PARM=0x2 >>>>>>> receive: 0x80 >>>>>>> send: NID=0x23, VERB=0x362(set_amp_gain_mute,I:L#2), PARM=0x0 >>>>>>> >>>>>>> alc_auto_init_input_src >>>>>>> send: NID=0x23, VERB=0x361(set_amp_gain_mute,I:L#1), PARM=0x80 >>>>>>> send: NID=0x23, VERB=0x351(set_amp_gain_mute,I:R#1), PARM=0x80 >>>>>>> send: NID=0x23, VERB=0x362(set_amp_gain_mute,I:L#2), PARM=0x80 >>>>>>> send: NID=0x23, VERB=0x352(set_amp_gain_mute,I:R#2), PARM=0x80 >>>>>>> >>>>>>> snd_hda_codec_resume_app >>>>>>> send: NID=0x23, VERB=0x363(set_amp_gain_mute,I:L#3), PARM=0x80 >>>>>>> send: NID=0x23, VERB=0x353(set_amp_gain_mute,I:R#3), PARM=0x80 >>>>>>> send: NID=0x23, VERB=0x364(set_amp_gain_mute,I:L#4), PARM=0x80 >>>>>>> send: NID=0x23, VERB=0x354(set_amp_gain_mute,I:R#4), PARM=0x80 >>>>>>> send: NID=0x23, VERB=0x365(set_amp_gain_mute,I:L#5), PARM=0x80 >>>>>>> send: NID=0x23, VERB=0x355(set_amp_gain_mute,I:R#5), PARM=0x80 >>>>>>> send: NID=0x23, VERB=0x366(set_amp_gain_mute,I:L#6), PARM=0x80 >>>>>>> send: NID=0x23, VERB=0x356(set_amp_gain_mute,I:R#6), PARM=0x80 >>>>>>> send: NID=0x23, VERB=0x367(set_amp_gain_mute,I:L#7), PARM=0x80 >>>>>>> send: NID=0x23, VERB=0x357(set_amp_gain_mute,I:R#7), PARM=0x80 >>>>>>> send: NID=0x23, VERB=0x368(set_amp_gain_mute,I:L#8), PARM=0x80 >>>>>>> send: NID=0x23, VERB=0x358(set_amp_gain_mute,I:R#8), PARM=0x80 >>>>>>> send: NID=0x23, VERB=0x369(set_amp_gain_mute,I:L#9), PARM=0x80 >>>>>>> send: NID=0x23, VERB=0x359(set_amp_gain_mute,I:R#9), PARM=0x80 >>>>>>> send: NID=0x23, VERB=0x36a(set_amp_gain_mute,I:L#10), PARM=0x80 >>>>>>> send: NID=0x23, VERB=0x35a(set_amp_gain_mute,I:R#10), PARM=0x80 >>>>>>> ... >>>>>>> >>>>>>>> get 28 >>>>>>> 28 Input Source:0 >>>>>>> ITEM: 0:Mic, 1:Internal Mic, 2:Line, VAL: [Mic] >>>>>> The problem still exists in 3.2.7-1.fc16.x86_64 kernel, is there any >>>>>> chance for a fix? >>>>> >>>>> Does the patch below fix? >>>>> >>>>> >>>>> Takashi >>>> >>>> Yes, it seems so. I don't want to get ahead of myself, but two >>>> suspend-resume cycles have passed and inputs are at Internal Mic and >>>> Mic, as expected. >>> >>> Thanks for checking. I committed the fix to git tree, also Cc'ed to >>> stable. >>> >>> >>> Takashi >> I cannot seem to find this patch in the 3.2 stable tree, was it rejected? > > What is the git commit id of it in Linus's tree? Hmm, it does not seem to be in Linus' tree yet. It is in for-linus branch of sound tree under commit 068b9394 though. Julian

1 0

[alsa-devel] ALC892, Clevo P150HM: Input Source gets reset to "Mic" every resume
by Julian Sikorski 02 Mar '12

02 Mar '12

Hi, whenever I resume the machine, the Input Source would get changed back from "Internal Mic" to "Mic". This is a bit annoying, because pulseaudio is unaware of that change. As a result, everything looks fine in pavucontrol but no sound gets recorded. A quick look at alsamixer -c0 revealed the problem. My alsa info: http://www.alsa-project.org/db/?f=c5776b615a5f3b9ae624e4250fb0c2b69118ce26 This is unrelated to the other problems I have been reporting, since this happens on a mostly-stock Fedora 3.1.2 kernel (the patch I added deals with xhci, not alsa). Regards, Julian

3 8

[alsa-devel] [PATCH] ASoC: core: Split the union for CODEC/platform in the DAI
by Mark Brown 02 Mar '12

02 Mar '12

There's now core code which falls back to global CODEC operations for DAI calls that needs to be able to tell if it's dealing with a CPU or CODEC DAI and given the small number of DAIs in a typical system and overall memory usage pattern saving a pointer per DAI is really not worth the effort. Reported-by: Ian Lartey <ian(a)opensource.wolfsonmicro.com> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/soc-dai.h | 7 +++---- 1 files changed, 3 insertions(+), 4 deletions(-) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index d474128..868f4b6 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -250,10 +250,9 @@ struct snd_soc_dai { unsigned int rate; /* parent platform/codec */ - union { - struct snd_soc_platform *platform; - struct snd_soc_codec *codec; - }; + struct snd_soc_platform *platform; + struct snd_soc_codec *codec; + struct snd_soc_card *card; struct list_head list; -- 1.7.9.1

2 1

[alsa-devel] [PATCH v6] sound/soc/codecs: add LAPIS Semiconductor ML26124
by Tomoya MORINAGA 02 Mar '12

02 Mar '12

ML26124-01HB/ML26124-02GD is 16bit monaural audio CODEC which has high resistance to voltage noise. On chip regulator realizes power supply rejection ratio be over 90dB so more than 50dB is improved than ever. ML26124-01HB/ ML26124-02GD can deliver stable audio performance without being affected by noise from the power supply circuit and peripheral components. The chip also includes a composite video signal output, which can be applied to various portable device requirements. The ML26124 is realized these functions into very small package the size is only 2.56mm x 2.46mm therefore can be construct high quality sound system easily. ML26124-01HB is 25pin WCSP package; ML26124-02GD is 32pin WQFN package. Signed-off-by: Tomoya MORINAGA <tomoya.rohm(a)gmail.com> --- V6 - Add break at "case SND_SOC_BIAS_ON:" - Use switch() statement - Delete ASoC cache code. --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ml26124.c | 742 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/ml26124.h | 132 ++++++++ 4 files changed, 880 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/ml26124.c create mode 100644 sound/soc/codecs/ml26124.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index fa787d4..321ea08 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -39,6 +39,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_MAX98095 if I2C select SND_SOC_MAX9850 if I2C select SND_SOC_MAX9877 if I2C + select SND_SOC_ML26124 if I2C select SND_SOC_PCM3008 select SND_SOC_RT5631 if I2C select SND_SOC_SGTL5000 if I2C @@ -411,6 +412,9 @@ config SND_SOC_LM4857 config SND_SOC_MAX9877 tristate +config SND_SOC_ML26124 + tristate + config SND_SOC_TPA6130A2 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index a2c7842..b28f48e 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -25,6 +25,7 @@ snd-soc-l3-objs := l3.o snd-soc-max98088-objs := max98088.o snd-soc-max98095-objs := max98095.o snd-soc-max9850-objs := max9850.o +snd-soc-ml26124-objs := ml26124.o snd-soc-pcm3008-objs := pcm3008.o snd-soc-rt5631-objs := rt5631.o snd-soc-sgtl5000-objs := sgtl5000.o @@ -126,6 +127,7 @@ obj-$(CONFIG_SND_SOC_JZ4740_CODEC) += snd-soc-jz4740-codec.o obj-$(CONFIG_SND_SOC_MAX98088) += snd-soc-max98088.o obj-$(CONFIG_SND_SOC_MAX98095) += snd-soc-max98095.o obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o +obj-$(CONFIG_SND_SOC_ML26124) += snd-soc-ml26124.o obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o obj-$(CONFIG_SND_SOC_RT5631) += snd-soc-rt5631.o obj-$(CONFIG_SND_SOC_SGTL5000) += snd-soc-sgtl5000.o diff --git a/sound/soc/codecs/ml26124.c b/sound/soc/codecs/ml26124.c new file mode 100644 index 0000000..058b831 --- /dev/null +++ b/sound/soc/codecs/ml26124.c @@ -0,0 +1,742 @@ +/* + * Copyright (C) 2011 LAPIS Semiconductor Co., Ltd. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2 of the License. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307, USA. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/slab.h> +#include <linux/platform_device.h> +#include <linux/regmap.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tlv.h> +#include "ml26124.h" + +#define DVOL_CTL_DVMUTE_ON BIT(4) /* Digital volume MUTE On */ +#define DVOL_CTL_DVMUTE_OFF 0 /* Digital volume MUTE Off */ +#define ML26124_SAI_NO_DELAY BIT(1) +#define ML26124_SAI_FRAME_SYNC (BIT(5) | BIT(0)) /* For mono (Telecodec) */ +#define ML26134_CACHESIZE 79 +#define ML26124_VMID BIT(0) +#define ML26124_RATES (SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_32000 |\ + SNDRV_PCM_RATE_48000) +#define ML26124_FORMATS SNDRV_PCM_FMTBIT_S16_LE +#define ML26124_NUM_REGISTER ML26134_CACHESIZE + +struct ml26124_priv { + u32 mclk; + u32 channels; + struct regmap *regmap; +}; + +struct clk_coeff { + u32 mclk; + u32 rate; + u8 pllnl; + u8 pllnh; + u8 pllml; + u8 pllmh; + u8 plldiv; +}; + +/* ML26124 configuration */ +static const DECLARE_TLV_DB_SCALE(digital_tlv, -7150, 50, 0); + +static const DECLARE_TLV_DB_SCALE(alclvl, -2250, 150, 0); +static const DECLARE_TLV_DB_SCALE(mingain, -1200, 600, 0); +static const DECLARE_TLV_DB_SCALE(maxgain, -675, 600, 0); +static const DECLARE_TLV_DB_SCALE(boost_vol, -1200, 75, 0); +static const DECLARE_TLV_DB_SCALE(ngth, -7650, 150, 0); + +static const char * const ml26124_companding[] = {"16bit PCM", "u-law", + "A-law"}; + +static const struct soc_enum ml26124_adc_companding_enum + = SOC_ENUM_SINGLE(ML26124_SAI_TRANS_CTL, 6, 3, ml26124_companding); + +static const struct soc_enum ml26124_dac_companding_enum + = SOC_ENUM_SINGLE(ML26124_SAI_RCV_CTL, 6, 3, ml26124_companding); + +static const struct snd_kcontrol_new ml26124_snd_controls[] = { + SOC_SINGLE_TLV("Capture Digital Volume", ML26124_RECORD_DIG_VOL, 0, + 0xff, 1, digital_tlv), + SOC_SINGLE_TLV("Playback Digital Volume", ML26124_PLBAK_DIG_VOL, 0, + 0xff, 1, digital_tlv), + SOC_SINGLE_TLV("Digital Boost Volume", ML26124_DIGI_BOOST_VOL, 0, + 0x3f, 0, boost_vol), + SOC_SINGLE_TLV("EQ Band0 Volume", ML26124_EQ_GAIN_BRAND0, 0, + 0xff, 1, digital_tlv), + SOC_SINGLE_TLV("EQ Band1 Volume", ML26124_EQ_GAIN_BRAND1, 0, + 0xff, 1, digital_tlv), + SOC_SINGLE_TLV("EQ Band2 Volume", ML26124_EQ_GAIN_BRAND2, 0, + 0xff, 1, digital_tlv), + SOC_SINGLE_TLV("EQ Band3 Volume", ML26124_EQ_GAIN_BRAND3, 0, + 0xff, 1, digital_tlv), + SOC_SINGLE_TLV("EQ Band4 Volume", ML26124_EQ_GAIN_BRAND4, 0, + 0xff, 1, digital_tlv), + SOC_SINGLE_TLV("ALC Target Level", ML26124_ALC_TARGET_LEV, 0, + 0xf, 1, alclvl), + SOC_SINGLE_TLV("ALC Min Input Volume", ML26124_ALC_MAXMIN_GAIN, 0, + 7, 0, mingain), + SOC_SINGLE_TLV("ALC Max Input Volume", ML26124_ALC_MAXMIN_GAIN, 4, + 7, 1, maxgain), + SOC_SINGLE_TLV("Playback Limitter Min Input Volume", + ML26124_PL_MAXMIN_GAIN, 0, 7, 0, mingain), + SOC_SINGLE_TLV("Playback Limitter Max Input Volume", + ML26124_PL_MAXMIN_GAIN, 4, 7, 1, maxgain), + SOC_SINGLE_TLV("Playback Boost Volume", ML26124_PLYBAK_BOST_VOL, 0, + 0x3f, 0, boost_vol), + SOC_SINGLE("DC High Pass Filter Switch", ML26124_FILTER_EN, 0, 1, 0), + SOC_SINGLE("Noise High Pass Filter Switch", ML26124_FILTER_EN, 1, 1, 0), + SOC_SINGLE("Zero Cross Comparator Switch", ML26124_PW_ZCCMP_PW_MNG, 1, + 1, 0), + SOC_SINGLE("EQ Band0 Switch", ML26124_FILTER_EN, 2, 1, 0), + SOC_SINGLE("EQ Band1 Switch", ML26124_FILTER_EN, 3, 1, 0), + SOC_SINGLE("EQ Band2 Switch", ML26124_FILTER_EN, 4, 1, 0), + SOC_SINGLE("EQ Band3 Switch", ML26124_FILTER_EN, 5, 1, 0), + SOC_SINGLE("EQ Band4 Switch", ML26124_FILTER_EN, 6, 1, 0), + SOC_SINGLE("Play Limitter", ML26124_DVOL_CTL, 0, 1, 0), + SOC_SINGLE("Capture Limitter", ML26124_DVOL_CTL, 1, 1, 0), + SOC_SINGLE("Digital Volume Fade Switch", ML26124_DVOL_CTL, 3, 1, 0), + SOC_SINGLE("Digital Switch", ML26124_DVOL_CTL, 4, 1, 0), + SOC_ENUM("DAC Companding", ml26124_dac_companding_enum), + SOC_ENUM("ADC Companding", ml26124_adc_companding_enum), +}; + +static const struct snd_kcontrol_new ml26124_output_mixer_controls[] = { + SOC_DAPM_SINGLE("DAC Switch", ML26124_SPK_AMP_OUT, 1, 1, 0), + SOC_DAPM_SINGLE("Line in Switch", ML26124_SPK_AMP_OUT, 3, 1, 0), + SOC_DAPM_SINGLE("PGA Switch", ML26124_SPK_AMP_OUT, 5, 1, 0), +}; + +/* Input mux */ +static const char * const ml26124_input_select[] = {"Analog MIC SingleEnded in", + "Analog MIC Differential in", + "Digital MIC in"}; + +static const struct soc_enum ml26124_insel_enum = + SOC_ENUM_SINGLE(ML26124_MIC_IF_CTL, 0, 2, ml26124_input_select); + +static const struct snd_kcontrol_new ml26124_input_mux_controls = + SOC_DAPM_ENUM("Input Select", ml26124_insel_enum); + +static const struct snd_kcontrol_new ml26124_line_control = + SOC_DAPM_SINGLE("Switch", ML26124_PW_LOUT_PW_MNG, 1, 1, 0); + +static const struct snd_soc_dapm_widget ml26124_dapm_widgets[] = { + SND_SOC_DAPM_SUPPLY("MCLK", ML26124_CLK_EN, 0, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("PLL", ML26124_CLK_EN, 1, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("MICBIAS", ML26124_PW_REF_PW_MNG, 2, 0, NULL, 0), + SND_SOC_DAPM_MIXER("Output Mixer", ML26124_PW_SPAMP_PW_MNG, 0, 0, + &ml26124_output_mixer_controls[0], + ARRAY_SIZE(ml26124_output_mixer_controls)), + SND_SOC_DAPM_DAC("DAC", "Playback", ML26124_PW_DAC_PW_MNG, 1, 0), + SND_SOC_DAPM_ADC("ADC", "Capture", ML26124_PW_IN_PW_MNG, 1, 0), + SND_SOC_DAPM_PGA("PGA", ML26124_PW_IN_PW_MNG, 3, 0, NULL, 0), + SND_SOC_DAPM_MUX("Input Mux", SND_SOC_NOPM, 0, 0, + &ml26124_input_mux_controls), + SND_SOC_DAPM_SWITCH("Line Out Enable", SND_SOC_NOPM, 0, 0, + &ml26124_line_control), + SND_SOC_DAPM_INPUT("MDIN"), + SND_SOC_DAPM_INPUT("MIN"), + SND_SOC_DAPM_INPUT("LIN"), + SND_SOC_DAPM_OUTPUT("SPOUT"), + SND_SOC_DAPM_OUTPUT("LOUT"), +}; + +static const struct snd_soc_dapm_route ml26124_intercon[] = { + /* Supply */ + {"DAC", NULL, "MCLK"}, + {"ADC", NULL, "MCLK"}, + {"DAC", NULL, "PLL"}, + {"ADC", NULL, "PLL"}, + + /* output mixer */ + {"Output Mixer", "PGA Switch", "PGA"}, + {"Output Mixer", "DAC Switch", "DAC"}, + {"Output Mixer", "Line in Switch", "LIN"}, + + /* outputs */ + {"LOUT", NULL, "Output Mixer"}, + {"SPOUT", NULL, "Output Mixer"}, + + /* input */ + {"Input Mux", "Analog MIC SingleEnded in", "MIN"}, + {"Input Mux", "Analog MIC Differential in", "MIN"}, + {"ADC", NULL, "Input Mux"}, +}; + +/* PLLOutputFreq(Hz) = InputMclkFreq(Hz) * PLLM / (PLLN * PLLDIV) */ +static const struct clk_coeff coeff_div[] = { + {12288800, 16000, 0xc, 0x0, 0x20, 0x0, 0x4}, + {12288800, 32000, 0xc, 0x0, 0x20, 0x0, 0x4}, + {12288800, 48000, 0xc, 0x0, 0x30, 0x0, 0x4}, +}; + +static struct reg_default ml26124_reg[] = { + /* CLOCK control Register */ + {0x00, 0x00 }, /* Sampling Rate */ + {0x02, 0x00}, /* PLL NL */ + {0x04, 0x00}, /* PLLNH */ + {0x06, 0x00}, /* PLLML */ + {0x08, 0x00}, /* MLLMH */ + {0x0a, 0x00}, /* PLLDIV */ + {0x0c, 0x00}, /* Clock Enable */ + {0x0e, 0x00}, /* CLK Input/Output Control */ + + /* System Control Register */ + {0x10, 0x00}, /* Software RESET */ + {0x12, 0x00}, /* Record/Playback Run */ + {0x14, 0x00}, /* Mic Input/Output control */ + + /* Power Management Register */ + {0x20, 0x00}, /* Reference Power Management */ + {0x22, 0x00}, /* Input Power Management */ + {0x24, 0x00}, /* DAC Power Management */ + {0x26, 0x00}, /* SP-AMP Power Management */ + {0x28, 0x00}, /* LINEOUT Power Management */ + {0x2a, 0x00}, /* VIDEO Power Management */ + {0x2e, 0x00}, /* AC-CMP Power Management */ + + /* Analog reference Control Register */ + {0x30, 0x04}, /* MICBIAS Voltage Control */ + + /* Input/Output Amplifier Control Register */ + {0x32, 0x10}, /* MIC Input Volume */ + {0x38, 0x00}, /* Mic Boost Volume */ + {0x3a, 0x33}, /* Speaker AMP Volume */ + {0x48, 0x00}, /* AMP Volume Control Function Enable */ + {0x4a, 0x00}, /* Amplifier Volume Fader Control */ + + /* Analog Path Control Register */ + {0x54, 0x00}, /* Speaker AMP Output Control */ + {0x5a, 0x00}, /* Mic IF Control */ + {0xe8, 0x01}, /* Mic Select Control */ + + /* Audio Interface Control Register */ + {0x60, 0x00}, /* SAI-Trans Control */ + {0x62, 0x00}, /* SAI-Receive Control */ + {0x64, 0x00}, /* SAI Mode select */ + + /* DSP Control Register */ + {0x66, 0x01}, /* Filter Func Enable */ + {0x68, 0x00}, /* Volume Control Func Enable */ + {0x6A, 0x00}, /* Mixer & Volume Control*/ + {0x6C, 0xff}, /* Record Digital Volume */ + {0x70, 0xff}, /* Playback Digital Volume */ + {0x72, 0x10}, /* Digital Boost Volume */ + {0x74, 0xe7}, /* EQ gain Band0 */ + {0x76, 0xe7}, /* EQ gain Band1 */ + {0x78, 0xe7}, /* EQ gain Band2 */ + {0x7A, 0xe7}, /* EQ gain Band3 */ + {0x7C, 0xe7}, /* EQ gain Band4 */ + {0x7E, 0x00}, /* HPF2 CutOff*/ + {0x80, 0x00}, /* EQ Band0 Coef0L */ + {0x82, 0x00}, /* EQ Band0 Coef0H */ + {0x84, 0x00}, /* EQ Band0 Coef0L */ + {0x86, 0x00}, /* EQ Band0 Coef0H */ + {0x88, 0x00}, /* EQ Band1 Coef0L */ + {0x8A, 0x00}, /* EQ Band1 Coef0H */ + {0x8C, 0x00}, /* EQ Band1 Coef0L */ + {0x8E, 0x00}, /* EQ Band1 Coef0H */ + {0x90, 0x00}, /* EQ Band2 Coef0L */ + {0x92, 0x00}, /* EQ Band2 Coef0H */ + {0x94, 0x00}, /* EQ Band2 Coef0L */ + {0x96, 0x00}, /* EQ Band2 Coef0H */ + {0x98, 0x00}, /* EQ Band3 Coef0L */ + {0x9A, 0x00}, /* EQ Band3 Coef0H */ + {0x9C, 0x00}, /* EQ Band3 Coef0L */ + {0x9E, 0x00}, /* EQ Band3 Coef0H */ + {0xA0, 0x00}, /* EQ Band4 Coef0L */ + {0xA2, 0x00}, /* EQ Band4 Coef0H */ + {0xA4, 0x00}, /* EQ Band4 Coef0L */ + {0xA6, 0x00}, /* EQ Band4 Coef0H */ + + /* ALC Control Register */ + {0xb0, 0x00}, /* ALC Mode */ + {0xb2, 0x02}, /* ALC Attack Time */ + {0xb4, 0x03}, /* ALC Decay Time */ + {0xb6, 0x00}, /* ALC Hold Time */ + {0xb8, 0x0b}, /* ALC Target Level */ + {0xba, 0x70}, /* ALC Max/Min Gain */ + {0xbc, 0x00}, /* Noise Gate Threshold */ + {0xbe, 0x00}, /* ALC ZeroCross TimeOut */ + + /* Playback Limiter Control Register */ + {0xc0, 0x04}, /* PL Attack Time */ + {0xc2, 0x05}, /* PL Decay Time */ + {0xc4, 0x0d}, /* PL Target Level */ + {0xc6, 0x70}, /* PL Max/Min Gain */ + {0xc8, 0x10}, /* Playback Boost Volume */ + {0xca, 0x00}, /* PL ZeroCross TimeOut */ + + /* Video Amplifier Control Register */ + {0xd0, 0x01}, /* VIDEO AMP Gain Control */ + {0xd2, 0x01}, /* VIDEO AMP Setup 1 */ + {0xd4, 0x01}, /* VIDEO AMP Control2 */ +}; + +static int ml26124_update_bits(struct snd_soc_codec *codec, unsigned short reg, + unsigned int mask, unsigned int value) +{ + int change; + unsigned int old, new; + int ret; + + ret = snd_soc_read(codec, reg); + if (ret < 0) + return ret; + + old = ret; + new = (old & ~mask) | (value & mask); + change = old != new; + if (change) { + ret = snd_soc_write(codec, reg + 1, new); + if (ret < 0) + return ret; + } + + return change; +} + +/* Get sampling rate value of sampling rate setting register (0x0) */ +static inline int get_srate(int rate) +{ + int srate; + + switch (rate) { + case 16000: + srate = 3; + break; + case 32000: + srate = 6; + break; + case 48000: + srate = 8; + break; + default: + return -EINVAL; + } + return srate; +} + +static inline int get_coeff(int mclk, int rate) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(coeff_div); i++) { + if (coeff_div[i].rate == rate && coeff_div[i].mclk == mclk) + return i; + } + return -EINVAL; +} + +static int ml26124_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct ml26124_priv *priv = snd_soc_codec_get_drvdata(codec); + int i = get_coeff(priv->mclk, params_rate(hw_params)); + + priv->channels = params_channels(hw_params); + + switch (params_rate(hw_params)) { + case 16000: + ml26124_update_bits(codec, ML26124_SMPLING_RATE, 0xf, + get_srate(coeff_div[i].rate)); + ml26124_update_bits(codec, ML26124_PLLNL, 0xff, + coeff_div[i].pllnl); + ml26124_update_bits(codec, ML26124_PLLNH, 0x1, + coeff_div[i].pllnh); + ml26124_update_bits(codec, ML26124_PLLML, 0xff, + coeff_div[i].pllml); + ml26124_update_bits(codec, ML26124_PLLMH, 0x3f, + coeff_div[i].pllmh); + ml26124_update_bits(codec, ML26124_PLLDIV, 0x1f, + coeff_div[i].plldiv); + break; + case 32000: + ml26124_update_bits(codec, ML26124_SMPLING_RATE, 0xf, + get_srate(coeff_div[i].rate)); + ml26124_update_bits(codec, ML26124_PLLNL, 0xff, + coeff_div[i].pllnl); + ml26124_update_bits(codec, ML26124_PLLNH, 0x1, + coeff_div[i].pllnh); + ml26124_update_bits(codec, ML26124_PLLML, 0xff, + coeff_div[i].pllml); + ml26124_update_bits(codec, ML26124_PLLMH, 0x3f, + coeff_div[i].pllmh); + ml26124_update_bits(codec, ML26124_PLLDIV, 0x1f, + coeff_div[i].plldiv); + break; + case 48000: + ml26124_update_bits(codec, ML26124_SMPLING_RATE, 0xf, + get_srate(coeff_div[i].rate)); + ml26124_update_bits(codec, ML26124_PLLNL, 0xff, + coeff_div[i].pllnl); + ml26124_update_bits(codec, ML26124_PLLNH, 0x1, + coeff_div[i].pllnh); + ml26124_update_bits(codec, ML26124_PLLML, 0xff, + coeff_div[i].pllml); + ml26124_update_bits(codec, ML26124_PLLMH, 0x3f, + coeff_div[i].pllmh); + ml26124_update_bits(codec, ML26124_PLLDIV, 0x1f, + coeff_div[i].plldiv); + break; + default: + pr_err("%s:this rate is no support for ml26124\n", __func__); + return -EINVAL; + } + + return 0; +} + +static int ml26124_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + if (mute) + ml26124_update_bits(codec, ML26124_DVOL_CTL, BIT(4), + DVOL_CTL_DVMUTE_ON); + else + ml26124_update_bits(codec, ML26124_DVOL_CTL, BIT(4), + DVOL_CTL_DVMUTE_OFF); + + return 0; +} + +static int ml26124_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + unsigned char mode; + struct snd_soc_codec *codec = codec_dai->codec; + struct ml26124_priv *priv = snd_soc_codec_get_drvdata(codec); + unsigned char mask = ML26124_SAI_NO_DELAY | ML26124_SAI_FRAME_SYNC; + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + mode = 1; + break; + case SND_SOC_DAIFMT_CBS_CFS: + mode = 0; + break; + default: + return -EINVAL; + } + ml26124_update_bits(codec, ML26124_SAI_MODE_SEL, BIT(0), mode); + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + break; + default: + return -EINVAL; + } + + /* clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + default: + return -EINVAL; + } +if (!priv) + return -1; + + /* SAI configuration */ + if (priv->channels == 1) { + ml26124_update_bits(codec, ML26124_SAI_TRANS_CTL, mask, + ML26124_SAI_NO_DELAY | ML26124_SAI_FRAME_SYNC); + ml26124_update_bits(codec, ML26124_SAI_RCV_CTL, mask, + ML26124_SAI_NO_DELAY | ML26124_SAI_FRAME_SYNC); + } else { + ml26124_update_bits(codec, ML26124_SAI_TRANS_CTL, mask, + ML26124_SAI_NO_DELAY); + ml26124_update_bits(codec, ML26124_SAI_RCV_CTL, mask, + ML26124_SAI_NO_DELAY); + } + + + return 0; +} + +static int ml26124_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 ml26124_priv *priv = snd_soc_codec_get_drvdata(codec); + + switch (clk_id) { + case ML26124_USE_PLL: + break; + default: + return -EINVAL; + } + + switch (freq) { + case 12288000: + priv->mclk = freq; + break; + default: + return -EINVAL; + } + return 0; +} + +static int ml26124_set_dai_clkdiv(struct snd_soc_dai *codec_dai, + int div_id, int div) +{ + struct snd_soc_codec *codec = codec_dai->codec; + + switch (div_id) { + case ML26124_MCLK: + ml26124_update_bits(codec, ML26124_CLK_CTL, + BIT(0) | BIT(1), div); + break; + default: + return -EINVAL; + } + return 0; +} + +static int ml26124_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct ml26124_priv *priv = snd_soc_codec_get_drvdata(codec); + switch (level) { + case SND_SOC_BIAS_ON: + pr_debug("%s: level=ON priv=%p\n", __func__, priv); + /* VMID ON */ + ml26124_update_bits(codec, ML26124_PW_REF_PW_MNG, + ML26124_VMID, ML26124_VMID); + msleep(500); + break; + case SND_SOC_BIAS_PREPARE: + pr_debug("%s: level=PREPARE\n", __func__); + break; + case SND_SOC_BIAS_STANDBY: + pr_debug("%s: level=STANDBY\n", __func__); + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) + regcache_sync(priv->regmap); + break; + case SND_SOC_BIAS_OFF: + pr_debug("%s: level=OFF\n", __func__); + /* VMID OFF */ + ml26124_update_bits(codec, ML26124_PW_REF_PW_MNG, + ML26124_VMID, 0); + break; + } + codec->dapm.bias_level = level; + return 0; +} + +static int ml26124_pcm_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *codec_dai) +{ + struct snd_soc_codec *codec = codec_dai->codec; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + ml26124_update_bits(codec, ML26124_REC_PLYBAK_RUN, 0x3, + 0); + return 0; + break; + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + ml26124_update_bits(codec, + ML26124_REC_PLYBAK_RUN, 0x3, 1); + else + ml26124_update_bits(codec, + ML26124_REC_PLYBAK_RUN, 0x3, 2); + break; + } + + return 0; +} + +static const struct snd_soc_dai_ops ml26124_dai_ops = { + .hw_params = ml26124_hw_params, + .digital_mute = ml26124_mute, + .set_fmt = ml26124_set_dai_fmt, + .set_sysclk = ml26124_set_dai_sysclk, + .set_clkdiv = ml26124_set_dai_clkdiv, + .trigger = ml26124_pcm_trigger, +}; + +static struct snd_soc_dai_driver ml26124_dai = { + .name = "ml26124-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = ML26124_RATES, + .formats = ML26124_FORMATS,}, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = ML26124_RATES, + .formats = ML26124_FORMATS,}, + .ops = &ml26124_dai_ops, + .symmetric_rates = 1, +}; + +#ifdef CONFIG_PM +static int ml26124_suspend(struct snd_soc_codec *codec, pm_message_t state) +{ + ml26124_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +static int ml26124_resume(struct snd_soc_codec *codec) +{ + struct ml26124_priv *priv = snd_soc_codec_get_drvdata(codec); + + regcache_sync(priv->regmap); + ml26124_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + return 0; +} +#else +#define ml26124_suspend NULL +#define ml26124_resume NULL +#endif + +static int ml26124_probe(struct snd_soc_codec *codec) +{ + int ret; + struct ml26124_priv *priv = snd_soc_codec_get_drvdata(codec); + codec->control_data = priv->regmap; + + ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_I2C); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } + + /* Software Reset */ + ml26124_update_bits(codec, ML26124_SW_RST, 0x01, 1); + ml26124_update_bits(codec, ML26124_SW_RST, 0x01, 0); + + ml26124_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + return 0; +} + +/* power down chip */ +static struct snd_soc_codec_driver soc_codec_dev_ml26124 = { + .probe = ml26124_probe, + .suspend = ml26124_suspend, + .resume = ml26124_resume, + .set_bias_level = ml26124_set_bias_level, + .dapm_widgets = ml26124_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(ml26124_dapm_widgets), + .dapm_routes = ml26124_intercon, + .num_dapm_routes = ARRAY_SIZE(ml26124_intercon), + .controls = ml26124_snd_controls, + .num_controls = ARRAY_SIZE(ml26124_snd_controls), +}; + +static const struct regmap_config ml26124_i2c_regmap = { + .val_bits = 8, + .reg_bits = 8, + .max_register = ML26124_NUM_REGISTER, + .reg_defaults = ml26124_reg, + .num_reg_defaults = ARRAY_SIZE(ml26124_reg), + .cache_type = REGCACHE_RBTREE, +}; + +static __devinit int ml26124_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct ml26124_priv *priv; + int ret; + + priv = devm_kzalloc(&i2c->dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + i2c_set_clientdata(i2c, priv); + + priv->regmap = regmap_init_i2c(i2c, &ml26124_i2c_regmap); + if (IS_ERR(priv->regmap)) { + ret = PTR_ERR(priv->regmap); + dev_err(&i2c->dev, "regmap_init_i2c() failed: %d\n", ret); + return ret; + } + + return snd_soc_register_codec(&i2c->dev, + &soc_codec_dev_ml26124, &ml26124_dai, 1); +} + +static __devexit int ml26124_i2c_remove(struct i2c_client *client) +{ + struct ml26124_priv *priv = i2c_get_clientdata(client); + + snd_soc_unregister_codec(&client->dev); + regmap_exit(priv->regmap); + return 0; +} + +static const struct i2c_device_id ml26124_i2c_id[] = { + { "ml26124", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, ml26124_i2c_id); + +static struct i2c_driver ml26124_i2c_driver = { + .driver = { + .name = "ml26124-codec", + .owner = THIS_MODULE, + }, + .probe = ml26124_i2c_probe, + .remove = __devexit_p(ml26124_i2c_remove), + .id_table = ml26124_i2c_id, +}; + +static int __init ml26124_modinit(void) +{ + int ret; + + ret = i2c_add_driver(&ml26124_i2c_driver); + if (ret != 0) + pr_err("Failed to register ML26124 I2C driver: %d\n", ret); + + return ret; +} +module_init(ml26124_modinit); + +static void __exit ml26124_exit(void) +{ + i2c_del_driver(&ml26124_i2c_driver); +} +module_exit(ml26124_exit); + +MODULE_AUTHOR("Tomoya MORINAGA <tomoya.rohm(a)gmail.com>"); +MODULE_DESCRIPTION("LAPIS Semiconductor ML26124 ALSA SoC codec driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/ml26124.h b/sound/soc/codecs/ml26124.h new file mode 100644 index 0000000..d4a8df5 --- /dev/null +++ b/sound/soc/codecs/ml26124.h @@ -0,0 +1,132 @@ +/* + * Copyright (C) 2011 LAPIS Semiconductor Co., Ltd. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2 of the License. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307, USA. + */ + +#ifndef ML26124_H +#define ML26124_H + +/* Clock Control Register */ +#define ML26124_SMPLING_RATE 0x00 +#define ML26124_PLLNL 0x02 +#define ML26124_PLLNH 0x04 +#define ML26124_PLLML 0x06 +#define ML26124_PLLMH 0x08 +#define ML26124_PLLDIV 0x0a +#define ML26124_CLK_EN 0x0c +#define ML26124_CLK_CTL 0x0e + +/* System Control Register */ +#define ML26124_SW_RST 0x10 +#define ML26124_REC_PLYBAK_RUN 0x12 +#define ML26124_MIC_TIM 0x14 + +/* Power Mnagement Register */ +#define ML26124_PW_REF_PW_MNG 0x20 +#define ML26124_PW_IN_PW_MNG 0x22 +#define ML26124_PW_DAC_PW_MNG 0x24 +#define ML26124_PW_SPAMP_PW_MNG 0x26 +#define ML26124_PW_LOUT_PW_MNG 0x28 +#define ML26124_PW_VOUT_PW_MNG 0x2a +#define ML26124_PW_ZCCMP_PW_MNG 0x2e + +/* Analog Reference Control Register */ +#define ML26124_PW_MICBIAS_VOL 0x30 + +/* Input/Output Amplifier Control Register */ +#define ML26124_PW_MIC_IN_VOL 0x32 +#define ML26124_PW_MIC_BOST_VOL 0x38 +#define ML26124_PW_SPK_AMP_VOL 0x3a +#define ML26124_PW_AMP_VOL_FUNC 0x48 +#define ML26124_PW_AMP_VOL_FADE 0x4a + +/* Analog Path Control Register */ +#define ML26124_SPK_AMP_OUT 0x54 +#define ML26124_MIC_IF_CTL 0x5a +#define ML26124_MIC_SELECT 0xe8 + +/* Audio Interface Control Register */ +#define ML26124_SAI_TRANS_CTL 0x60 +#define ML26124_SAI_RCV_CTL 0x62 +#define ML26124_SAI_MODE_SEL 0x64 + +/* DSP Control Register */ +#define ML26124_FILTER_EN 0x66 +#define ML26124_DVOL_CTL 0x68 +#define ML26124_MIXER_VOL_CTL 0x6a +#define ML26124_RECORD_DIG_VOL 0x6c +#define ML26124_PLBAK_DIG_VOL 0x70 +#define ML26124_DIGI_BOOST_VOL 0x72 +#define ML26124_EQ_GAIN_BRAND0 0x74 +#define ML26124_EQ_GAIN_BRAND1 0x76 +#define ML26124_EQ_GAIN_BRAND2 0x78 +#define ML26124_EQ_GAIN_BRAND3 0x7a +#define ML26124_EQ_GAIN_BRAND4 0x7c +#define ML26124_HPF2_CUTOFF 0x7e +#define ML26124_EQBRAND0_F0L 0x80 +#define ML26124_EQBRAND0_F0H 0x82 +#define ML26124_EQBRAND0_F1L 0x84 +#define ML26124_EQBRAND0_F1H 0x86 +#define ML26124_EQBRAND1_F0L 0x88 +#define ML26124_EQBRAND1_F0H 0x8a +#define ML26124_EQBRAND1_F1L 0x8c +#define ML26124_EQBRAND1_F1H 0x8e +#define ML26124_EQBRAND2_F0L 0x90 +#define ML26124_EQBRAND2_F0H 0x92 +#define ML26124_EQBRAND2_F1L 0x94 +#define ML26124_EQBRAND2_F1H 0x96 +#define ML26124_EQBRAND3_F0L 0x98 +#define ML26124_EQBRAND3_F0H 0x9a +#define ML26124_EQBRAND3_F1L 0x9c +#define ML26124_EQBRAND3_F1H 0x9e +#define ML26124_EQBRAND4_F0L 0xa0 +#define ML26124_EQBRAND4_F0H 0xa2 +#define ML26124_EQBRAND4_F1L 0xa4 +#define ML26124_EQBRAND4_F1H 0xa6 + +/* ALC Control Register */ +#define ML26124_ALC_MODE 0xb0 +#define ML26124_ALC_ATTACK_TIM 0xb2 +#define ML26124_ALC_DECAY_TIM 0xb4 +#define ML26124_ALC_HOLD_TIM 0xb6 +#define ML26124_ALC_TARGET_LEV 0xb8 +#define ML26124_ALC_MAXMIN_GAIN 0xba +#define ML26124_NOIS_GATE_THRSH 0xbc +#define ML26124_ALC_ZERO_TIMOUT 0xbe + +/* Playback Limiter Control Register */ +#define ML26124_PL_ATTACKTIME 0xc0 +#define ML26124_PL_DECAYTIME 0xc2 +#define ML26124_PL_TARGETTIME 0xc4 +#define ML26124_PL_MAXMIN_GAIN 0xc6 +#define ML26124_PLYBAK_BOST_VOL 0xc8 +#define ML26124_PL_0CROSS_TIMOUT 0xca + +/* Video Amplifer Control Register */ +#define ML26124_VIDEO_AMP_GAIN_CTL 0xd0 +#define ML26124_VIDEO_AMP_SETUP1 0xd2 +#define ML26124_VIDEO_AMP_CTL2 0xd4 + +/* Clock select for machine driver */ +#define ML26124_USE_PLL 0 +#define ML26124_USE_MCLKI_256FS 1 +#define ML26124_USE_MCLKI_512FS 2 +#define ML26124_USE_MCLKI_1024FS 3 + +enum ml26124_regs { + ML26124_MCLK = 0, +}; + +#endif -- 1.7.7.6

2 5

[alsa-devel] [PATCH] ASoC: core - cleanup platform debugfs on probe failure.
by Liam Girdwood 02 Mar '12

02 Mar '12

Make sure we cleanup the platform debugfs when probe fails. Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- sound/soc/soc-core.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 1bdc67e..29dbbd7 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1112,6 +1112,7 @@ static int soc_probe_platform(struct snd_soc_card *card, return 0; err_probe: + soc_cleanup_platform_debugfs(platform); module_put(platform->dev->driver->owner); return ret; -- 1.7.5.4

2 1

[alsa-devel] [PATCH] ASoC: core: Log a warning when machines use soc-audio
by Mark Brown 02 Mar '12

02 Mar '12

snd_soc_register_card() has been available and strongly preferred since 2.6.38 but we're still seeing new drivers using it and the conversion rate for older machines has been low. Help address both issues by logging a warning when the soc-audio device probes. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-core.c | 4 ++++ 1 files changed, 4 insertions(+), 0 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index f52bf5b..8d2d066 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1644,6 +1644,10 @@ static int soc_probe(struct platform_device *pdev) if (!card) return -EINVAL; + dev_warn(&pdev->dev, + "ASoC machine %s should use snd_soc_register_card()\n", + card->name); + /* Bodge while we unpick instantiation */ card->dev = &pdev->dev; -- 1.7.9.1

2 1

[alsa-devel] [PATCH v2 00/10] OMAP/ASoC: Move and merge McBSP driver under ASoC
by Peter Ujfalusi 02 Mar '12

02 Mar '12

Hello, Changes since v1 (RFC series): - removed pm_runtime_get/put_sync calls since core takes care of this. Fixes the initial issue with OMAP3 Pandora setup reported by Grazvydas Ignotas - Dropped my version of snd_soc_add_dai_controls from the series since Liam implementation adds the same functionality. - Added Tested-by tags from Janusz (OMAP1), and Grazvydas (OMAP3 pandora) - Added Ack from Tony for the arch/arm/ patches - I have not added Ack from Mark, and Jarkko although they said it looks OK, but not given explicit ack for the series. NOTE: this series has been generated agains Takashi's topic/asoc branch merged with Mark's for-next branch since this series depends on changes in Marks' branch, but not yet pulled by Takashi (snd_soc_add_dai_controls implementation from Liam). Intro mail from thre RFC series: The aim of this series is to move the plat-omap/mcbsp.c driver out to sound/soc/omap/ By doing this we can combine the two levels of McBSP driver stack into a single one. Formally we had omap-mcbsp.[1-X] drivers (plat-omap/mcbsp.c), and omap-mcbsp-dai.[0-(X-1) drivers (sound/soc/omap/omap-mcbsp.c). We had global, shared structures, variables, pointers. The API between the two driver was a set of exported function calls where one of the parameter was the McBSP port id. With this series we will have one driver for McBSP (omap-mcbsp.[1-X]) which is basically the ASoC dai driver. All exported/shared global structures are gone since we can use driver data for all ports in this way. With the code in one place we can do further clean up in the McBSP driver, and adding new features will be much easier in the future. Also the DT support for McBSP will be significantly easier if we have one set of driver. McBSP is only used by audio, there are no known other users of the OMAP McBSP (well, Nokia n700 has the WLAN connected through McBSP but we do not have driver for that anyway). The series was created against Takashi's topic/asoc branch. I have tested it on Beagle xM, and compile tested it for OMAP1, OMAP2, and OMAP4 as well. I'm going to do more work on the cleanup, and the plan is to merge the two files into one (mcbsp.c -> omap-mcbsp.c) eventually. Regards, Peter --- Peter Ujfalusi (10): ARM: OMAP: mcbsp: Convert core driver to proper platform driver OMAP: mcbsp: Move core driver under sound/soc/omap ASoC: OMAP: McBSP: Consolidate plat/mcbsp.h content ASoC: omap-mcbsp: Create a single driver for McBSP ARM: OMAP: Do not register omap-mcbsp-dai device ASoC: OMAP: mcbsp.h: Clean up bit definitions ASoC: OMAP McBSP: Remove redundant accessors ASoC: omap-mcbsp: Cleanup of module probe/remove code ASoC: omap-mcbsp: Merge the omap_mcbsp_data into omap_mcbsp structure ASoC: omap-mcbsp: Simplify DMA configuration arch/arm/mach-omap1/devices.c | 9 - arch/arm/mach-omap1/mcbsp.c | 14 +- arch/arm/mach-omap2/devices.c | 22 -- arch/arm/mach-omap2/mcbsp.c | 8 +- arch/arm/plat-omap/Kconfig | 8 - arch/arm/plat-omap/Makefile | 2 - arch/arm/plat-omap/include/plat/mcbsp.h | 321 ---------------- sound/soc/omap/Kconfig | 4 + sound/soc/omap/Makefile | 1 + sound/soc/omap/am3517evm.c | 2 +- sound/soc/omap/ams-delta.c | 2 +- sound/soc/omap/igep0020.c | 2 +- {arch/arm/plat-omap => sound/soc/omap}/mcbsp.c | 470 ++++-------------------- sound/soc/omap/mcbsp.h | 337 +++++++++++++++++ sound/soc/omap/n810.c | 2 +- sound/soc/omap/omap-mcbsp.c | 287 ++++++++------- sound/soc/omap/omap-mcbsp.h | 2 +- sound/soc/omap/omap-pcm.h | 2 + sound/soc/omap/omap3beagle.c | 2 +- sound/soc/omap/omap3evm.c | 2 +- sound/soc/omap/omap3pandora.c | 4 +- sound/soc/omap/osk5912.c | 2 +- sound/soc/omap/overo.c | 2 +- sound/soc/omap/rx51.c | 4 +- sound/soc/omap/sdp3430.c | 4 +- sound/soc/omap/zoom2.c | 4 +- 26 files changed, 598 insertions(+), 921 deletions(-) rename {arch/arm/plat-omap => sound/soc/omap}/mcbsp.c (68%) create mode 100644 sound/soc/omap/mcbsp.h -- 1.7.8.4

3 19

Re: [alsa-devel] [PATCH v6] sound/soc/codecs: add LAPIS Semiconductor ML26124
by Mark Brown 02 Mar '12

02 Mar '12

On Fri, Mar 02, 2012 at 05:16:04PM +0900, Tomoya MORINAGA wrote: > 2012年3月1日8:51 Mark Brown <broonie(a)opensource.wolfsonmicro.com>: > >> +static int ml26124_set_dai_clkdiv(struct snd_soc_dai *codec_dai, > >> + int div_id, int div) > >> +{ > >> + struct snd_soc_codec *codec = codec_dai->codec; > >> + switch (div_id) { > >> + case ML26124_MCLK: > >> + ml26124_update_bits(codec, ML26124_CLK_CTL, > >> + BIT(0) | BIT(1), div); > >> + break; > > Why can't the driver calculate this automatically given the MCLK? > Sorry, I can't understand your saying. > Why does this driver need to calculate ? > I think this driver can use "div" value directly. Why does the user have to set this divider by hand at all? There's already a set_sysclk() function, just have the user specify the clock they're providing. > >> + ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_I2C); > > You're mixing regmap API usage and ASoC level I2C, this should be > > _REGMAP. > Currently, it seems codec control via i2c works well. > Let me know why I should use SND_SOC_REGMAP not _I2C. Have you checked what _I2C and _REGMAP do? > >> +static int __init ml26124_modinit(void) > >> +{ > >> + int ret; > >> + > >> + ret = i2c_add_driver(&ml26124_i2c_driver); > >> + if (ret != 0) > > module_i2c_driver(). > Sorry, I can't understand your saying. Use module_i2c_driver().

1 0

Re: [alsa-devel] [PATCH 2/2] ALSA: Integrate control based jack reporting with core jack reporting
by Takashi Iwai 02 Mar '12

02 Mar '12

So, a question of "Line Out" to be hyphened or not to be hyphened. The problem is less serious than Hamlet, and I think we can get rid of the redundant hyphen. Not only because many women dislike hyphened names nowadays (as you can find easily in Google), the naming rule with a direction suffix looks more flexible, and I saw that both or Mark and David don't matter much (so written in their posts). The fix patch is below. David, if you think it's too bad (that makes your code too messy), please speak up. Or, this is utterly broken, also let me know. thanks, Takashi --- From: Takashi Iwai <tiwai(a)suse.de> Subject: [PATCH] ALSA: hda/realtek - Kill hyphnated names Kill hyphens from "Line-Out" name strings, as suggested by Mark Brown. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_codec.c | 4 ++-- sound/pci/hda/patch_realtek.c | 4 ++-- 2 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 0ae6eb2..6843073 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -5118,7 +5118,7 @@ static int fill_audio_out_name(struct hda_codec *codec, hda_nid_t nid, const char *pfx = "", *sfx = ""; /* handle as a speaker if it's a fixed line-out */ - if (!strcmp(name, "Line-Out") && attr == INPUT_PIN_ATTR_INT) + if (!strcmp(name, "Line Out") && attr == INPUT_PIN_ATTR_INT) name = "Speaker"; /* check the location */ switch (attr) { @@ -5177,7 +5177,7 @@ int snd_hda_get_pin_label(struct hda_codec *codec, hda_nid_t nid, switch (get_defcfg_device(def_conf)) { case AC_JACK_LINE_OUT: - return fill_audio_out_name(codec, nid, cfg, "Line-Out", + return fill_audio_out_name(codec, nid, cfg, "Line Out", label, maxlen, indexp); case AC_JACK_SPEAKER: return fill_audio_out_name(codec, nid, cfg, "Speaker", diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 4fe2d59..a5be866 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1856,7 +1856,7 @@ static const char * const alc_slave_vols[] = { "Headphone Playback Volume", "Speaker Playback Volume", "Mono Playback Volume", - "Line-Out Playback Volume", + "Line Out Playback Volume", "CLFE Playback Volume", "Bass Speaker Playback Volume", "PCM Playback Volume", @@ -1873,7 +1873,7 @@ static const char * const alc_slave_sws[] = { "Speaker Playback Switch", "Mono Playback Switch", "IEC958 Playback Switch", - "Line-Out Playback Switch", + "Line Out Playback Switch", "CLFE Playback Switch", "Bass Speaker Playback Switch", "PCM Playback Switch", -- 1.7.9.2

3 3

[alsa-devel] tlv320aic3104
by rohith R 02 Mar '12

02 Mar '12

Beginner query. I'm modifying the codec driver for tlv320aic3104 and wanted to control inputs to a mixer.One of them being page 0/register 17. It basically has input level control(0-0x8) and then 0xf(which is off) and the values in-between 0x8-0xf as invalid. How to give the input to mixer?As a switch or volume/bias control and what is the most likely macro to be used. DAPM_SINGLE? --rohith

1 0

[alsa-devel] [PATCH] ASoC: wm8994: Disable debounce of jack detection on inserted jack
by Mark Brown 01 Mar '12

01 Mar '12

Don't debounce jack detection for inserted jacks, giving improved responsiveness. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 8 ++++++++ 1 files changed, 8 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index bc1f119..81d62a3 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3235,6 +3235,10 @@ static irqreturn_t wm1811_jackdet_irq(int irq, void *data) snd_soc_update_bits(codec, WM8958_MICBIAS2, WM8958_MICB2_DISCH, 0); + /* Disable debounce while inserted */ + snd_soc_update_bits(codec, WM1811_JACKDET_CTRL, + WM1811_JACKDET_DB, 0); + /* * Start off measument of microphone impedence to find * out what's actually there. @@ -3270,6 +3274,10 @@ static irqreturn_t wm1811_jackdet_irq(int irq, void *data) SND_JACK_MECHANICAL | SND_JACK_HEADSET | wm8994->btn_mask); + /* Enable debounce while removed */ + snd_soc_update_bits(codec, WM1811_JACKDET_CTRL, + WM1811_JACKDET_DB, WM1811_JACKDET_DB); + wm8994->mic_detecting = false; wm8994->jack_mic = false; snd_soc_update_bits(codec, WM8958_MIC_DETECT_1, -- 1.7.9.1

1 0

[alsa-devel] [PATCH] ASoC: wm8962: Run the headphone in class G mode when sidetone is enabled
by Mark Brown 01 Mar '12

01 Mar '12

Class W mode with sidetone is not fully supported. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8962.c | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index e73c857..d5ee3ec 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2242,9 +2242,11 @@ static const struct snd_soc_dapm_route wm8962_intercon[] = { { "STL", "Left", "ADCL" }, { "STL", "Right", "ADCR" }, + { "STL", NULL, "Class G" }, { "STR", "Left", "ADCL" }, { "STR", "Right", "ADCR" }, + { "STR", NULL, "Class G" }, { "DACL", NULL, "SYSCLK" }, { "DACL", NULL, "TOCLK" }, -- 1.7.9.1

1 0

[alsa-devel] ASoC updates for 3.3
by Mark Brown 01 Mar '12

01 Mar '12

The following changes since commit e555cf363167f09efae96d32a363e24c4de16b7b: ASoC: ak4642: fixup HeadPhone L/R dapm settings (2012-02-21 10:00:24 +0000) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-3.3 for you to fetch changes up to 5ed80a75b248bfaf840ea6b38f941edcf6ee7dc7: ASoC: i.MX SSI: Fix DSP_A format. (2012-02-23 18:58:17 +0000) ---------------------------------------------------------------- A small fix for the SSI driver and a fix for system shutdown with modern devices. Most of the modern devices will never get shut down normally with a visible kernel log as the systems they're in tend not to shut down often and when they do it's usually in form factors that don't have a user visible console. ---------------------------------------------------------------- Javier Martin (1): ASoC: i.MX SSI: Fix DSP_A format. Mark Brown (1): ASoC: dapm: Check for bias level when powering down sound/soc/imx/imx-ssi.c | 2 +- sound/soc/soc-dapm.c | 12 +++++++++--- 2 files changed, 10 insertions(+), 4 deletions(-)

2 1

[alsa-devel] [PATCH] ASoC: wm8962: Remove register default for PLL2
by Mark Brown 01 Mar '12

01 Mar '12

The initial value can be changed depending on system configuration. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8962.c | 2 -- 1 files changed, 0 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 890796f..e73c857 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -208,8 +208,6 @@ static struct reg_default wm8962_reg[] = { { 126, 0x000D }, /* R126 - Analogue Clocking3 */ { 127, 0x0000 }, /* R127 - PLL Software Reset */ - { 129, 0x0000 }, /* R129 - PLL2 */ - { 131, 0x0000 }, /* R131 - PLL 4 */ { 136, 0x0067 }, /* R136 - PLL 9 */ -- 1.7.9.1

1 0

[alsa-devel] oss plugin: SNDCTL_DSP_SETFRAGMENT problems
by Andriy Gapon 01 Mar '12

01 Mar '12

First, a disclaimer: what I'd like to discuss seems to affect a FreeBSD OSS implementation. After looking at the code in oss_pointer() and in snd_pcm_ioplug_hw_ptr_update() it seems that the code depends on OSS driver using exactly the same buffer size as specified by the buffer_size parameter. Otherwise the code that handles hw pointer cycling would calculate a delta incorrectly. Then, it seems that oss_hw_params() tries to enforce the driver's buffer size using SNDCTL_DSP_SETFRAGMENT. But in the OSS specification there are oh so many warnings about it: http://manuals.opensound.com/developer/SNDCTL_DSP_SETFRAGMENT.html > It's important to understand that this call just sets the size hint. There is > no guarantee that the requested size gets used. ... > It's very important to make this ioctl call as early as possible after > opening the device since otherwise the call may not have any effect. In > particular this call must be made before calling read, write or some other > ioctl calls. ... > This ioctl call will return an error only if the argument value is incorrect > or if the call is made in a wrong place. Even if the call returned OK there > is no guarantee that the requested buffer size will be used. This ioctl call > sets just the size hint and the driver may or may not honor the request. I see at least two issues with the oss plugin code: o Actual OSS driver buffer size value is never queried/verified if the SNDCTL_DSP_SETFRAGMENT call is successful o Apparently oss_hw_constraint() function is called before oss_hw_params() and the former makes SNDCTL_DSP_CHANNELS ioctl calls which change a configuration of the OSS driver and may affect the future SNDCTL_DSP_SETFRAGMENT call. This also violates the order described here: http://manuals.opensound.com/developer/callorder.html There is a retry loop in oss_hw_params() which closes and re-opens an OSS device, but that loop is only activated on SNDCTL_DSP_SETFRAGMENT hard failure; there is no handling for the case where the ioctl succeeds but the resulting buffer size differs from the requested one. I think that to nullify any effects of oss_hw_constraint() on the OSS driver configuration there should be a call to SNDCTL_DSP_RESET at the end of the function. Alternatively, there could be a call to this ioctl before SNDCTL_DSP_SETFRAGMENT. Or perhaps there could be resets at both places... Ideally, of course, the OSS plugin should query the exact OSS driver buffer size and alter the buffer_size parameter accordingly. But this is a bit more work. What do you think? Thank you. -- Andriy Gapon

1 0

[alsa-devel] [PATCH] ALSA: hda - Add a fake mute feature
by Takashi Iwai 01 Mar '12

01 Mar '12

David, I thought you have a machine with Cx20561 (such as Lenovo X200, T400 or W500)? Could you check whether this really doesn't break? Thanks, Takashi === Some codecs don't supply the mute amp-capabilities although the lowest volume gives the mute. It'd be handy if the parser provides the mute mixers in such a case. This patch adds an extension amp-cap bit (which is used only in the driver) to represent the min volume = mute state. Also modified the amp cache code to support the fake mute feature when this bit is set but the real mute bit is unset. In addition, conexant cx5051 parser uses this new feature to implement the missing mute controls. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=42825 Cc: <stable(a)kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_codec.c | 8 ++++++-- sound/pci/hda/hda_codec.h | 3 +++ sound/pci/hda/patch_conexant.c | 22 +++++++++++++++++++++- 3 files changed, 30 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index c2c65f6..0ae6eb2 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1759,7 +1759,11 @@ static void put_vol_mute(struct hda_codec *codec, struct hda_amp_info *info, parm = ch ? AC_AMP_SET_RIGHT : AC_AMP_SET_LEFT; parm |= direction == HDA_OUTPUT ? AC_AMP_SET_OUTPUT : AC_AMP_SET_INPUT; parm |= index << AC_AMP_SET_INDEX_SHIFT; - parm |= val; + if ((val & HDA_AMP_MUTE) && !(info->amp_caps & AC_AMPCAP_MUTE) && + (info->amp_caps & AC_AMPCAP_MIN_MUTE)) + ; /* set the zero value as a fake mute */ + else + parm |= val; snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, parm); info->vol[ch] = val; } @@ -2026,7 +2030,7 @@ int snd_hda_mixer_amp_tlv(struct snd_kcontrol *kcontrol, int op_flag, val1 = -((caps & AC_AMPCAP_OFFSET) >> AC_AMPCAP_OFFSET_SHIFT); val1 += ofs; val1 = ((int)val1) * ((int)val2); - if (min_mute) + if (min_mute || (caps & AC_AMPCAP_MIN_MUTE)) val2 |= TLV_DB_SCALE_MUTE; if (put_user(SNDRV_CTL_TLVT_DB_SCALE, _tlv)) return -EFAULT; diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index e9f71dc..f0f1943 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -298,6 +298,9 @@ enum { #define AC_AMPCAP_MUTE (1<<31) /* mute capable */ #define AC_AMPCAP_MUTE_SHIFT 31 +/* driver-specific amp-caps: using bits 24-30 */ +#define AC_AMPCAP_MIN_MUTE (1 << 30) /* min-volume = mute */ + /* Connection list */ #define AC_CLIST_LENGTH (0x7f<<0) #define AC_CLIST_LONG (1<<7) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index a7a5733..ca117cf 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -4079,7 +4079,8 @@ static int cx_auto_add_volume_idx(struct hda_codec *codec, const char *basename, err = snd_hda_ctl_add(codec, nid, kctl); if (err < 0) return err; - if (!(query_amp_caps(codec, nid, hda_dir) & AC_AMPCAP_MUTE)) + if (!(query_amp_caps(codec, nid, hda_dir) & + (AC_AMPCAP_MUTE | AC_AMPCAP_MIN_MUTE))) break; } return 0; @@ -4379,6 +4380,22 @@ static const struct snd_pci_quirk cxt_fixups[] = { {} }; +/* add "fake" mute amp-caps to DACs on cx5051 so that mixer mute switches + * can be created (bko#42825) + */ +static void add_cx5051_fake_mutes(struct hda_codec *codec) +{ + static hda_nid_t out_nids[] = { + 0x10, 0x11, 0 + }; + hda_nid_t *p; + + for (p = out_nids; *p; p++) + snd_hda_override_amp_caps(codec, *p, HDA_OUTPUT, + AC_AMPCAP_MIN_MUTE | + query_amp_caps(codec, *p, HDA_OUTPUT)); +} + static int patch_conexant_auto(struct hda_codec *codec) { struct conexant_spec *spec; @@ -4397,6 +4414,9 @@ static int patch_conexant_auto(struct hda_codec *codec) case 0x14f15045: spec->single_adc_amp = 1; break; + case 0x14f15051: + add_cx5051_fake_mutes(codec); + break; } apply_pin_fixup(codec, cxt_fixups, cxt_pincfg_tbl); -- 1.7.9

2 1

Re: [alsa-devel] Cranking sound coming out of speakers with the new 3.1 kernel
by Takashi Iwai 01 Mar '12

01 Mar '12

At Thu, 1 Mar 2012 08:44:33 +0100, Jan Prunk wrote: > > Hello ! > > The sound has been stable for the last days without noises, > after applying % amixer -c0 set "Auto-Mute Mode" "Disabled" OK, another question is whether the auto-mute feature works in hardware, i.e. even when this mixer is set off, the hardware switches the speaker upon plugging the headphone? If yes, we can simply remove this switch and software feature from the driver. thanks, Takashi > > It seems this has solved it, until further notice. > You might want to keep the bug open for a while, > so I can check with all the kernels (it works with latest). > > Kind regards, > Jan > > On Tue, Feb 28, 2012 at 4:38 PM, Takashi Iwai <tiwai(a)suse.de> wrote: > > At Tue, 28 Feb 2012 16:16:48 +0100, > > Jan Prunk wrote: > >> > >> Hello ! > >> > >> On Sat, Jan 14, 2012 at 12:04 AM, Jonathan Nieder <jrnieder(a)gmail.com> wrote: > >> > Hi again, > >> > > >> > Jan Prunk wrote: > >> > > >> >> I tried booting into 3.2.0-rc7-686-pae, but the crunching sound > >> >> is still present, I am able to play music file properly, and > >> >> crunching is in the background, and also while not processing > >> >> any sounds, the crunching appears out of the speakers. > >> > > >> > Ok, thanks for testing. Please send a summary of this regression to > >> > alsa-devel(a)alsa-project.org, cc-ing Takashi Iwai <tiwai(a)suse.de> and > >> > either me or this bug log so we can track it. > >> > > >> > Be sure to mention: > >> > > >> > - steps to reproduce, what you expect versus what actually happens, > >> > and how that indicates a bug > >> > >> The sound is ok in kernel version "3.0.0-1-686-pae" but on all newer > >> versions there are sparkling/farting noises coming out of the speaker. > >> The noise is not very loud, its barely noticeable, but its repetative > >> and doesn't go away. And I think it happens while typing on the > >> keyboard and a bit later, then it dissapears for a few seconds, > >> or by the time that I type again. > >> > >> > - which kernel versions you tried, and what happens with each > >> I tried running alsa-info.sh as a "user". > >> > >> I tried 3.0.0-1-686-pae here is the debug: > >> http://www.alsa-project.org/db/?f=c1149b6466b2068d79c7517a8b0808d8b5c26e6b > >> --- > >> Linux vaio 3.2.0-1-686-pae #1 SMP Fri Feb 17 06:27:21 UTC 2012 i686 GNU/Linux > >> ii linux-image-3.2.0-1-686-pae 3.2.6-1 > >> http://www.alsa-project.org/db/?f=fddbfbdc4fc44555a6dc851ecdc70760da6dd840 > >> --- > >> Linux vaio 3.2.0-rc7-686-pae #1 SMP Wed Dec 28 21:26:25 UTC 2011 i686 GNU/Linux > >> ii linux-image-3.2.0-rc7-686-pae 3.2~rc7-1~experimental.1 > >> http://www.alsa-project.org/db/?f=78c8919e87dd03294484f83b5f7ae44331b5edfe > >> Being booted into this version for 15 minutes, it seems to not make any noises, > >> but I will resubmit the info later if the noises will start to appear. > > > > 3.2.0-rc7 alsa-info shows that the headphone jack is plugged, thus the > > speaker is muted. It's natural that the noise goes away in this > > state. > > > > Apart from that, I see no obvious problem in 3.2.0 output. > > > > Does the problem still happen when you disable "Auto-Mute Mode" mixer > > enum? > > % amixer -c0 set "Auto-Mute Mode" "Disabled" > > > > Also, try to pass algin_buffer_size=0 option to snd-hda-intel. > > I don't expect much that this will influence, but at least, it's a > > difference between 3.0 and 3.2. > > > > > > Takashi > > > > > >> > - alsa-info.sh output, as an attachment > >> > > >> > - if you can make a recording of the strange sound available > >> > somewhere online, that would be ideal > >> > >> Because the noise is barely heard, I would need to find a good mic. > >> to be able to record it, but I could do that in the next bug submission. > >> > >> > - whether you are able to bisect or test patches if needed > >> > >> I might try that, if there are any manuals around how to do it, > >> but my response might take some time. > >> > >> > [1] may have more hints. > >> > > >> > Thanks and good luck, > >> > Jonathan > >> > > >> > [1] http://alsa-project.org/main/index.php/Help_To_Debug_Intel_HDA > >> > >> Kind regards, > >> Jan Prunk > >> -- > >> Jan Prunk http://www.prunk.si > >> 0x00E80E86 http://pgp.prunk.si > >> http://AS50763.peeringdb.com > >> > > > > -- > Jan Prunk http://www.prunk.si > 0x00E80E86 http://pgp.prunk.si > http://AS50763.peeringdb.com >

1 0

[alsa-devel] [PATCH] ASoC: Samsung: Update email id of the author
by Jaswinder Singh 01 Mar '12

01 Mar '12

I moved on from a great employer and the email-id no longer exists. Update email-id to a personal one, assuming I don't move on from myself anytime soon. And when I do, people don't get the eulogies bounced. Signed-off-by: Jaswinder Singh <jassisinghbrar(a)gmail.com> --- sound/soc/samsung/ac97.c | 4 ++-- sound/soc/samsung/i2s.c | 4 ++-- sound/soc/samsung/i2s.h | 2 +- sound/soc/samsung/pcm.c | 4 ++-- sound/soc/samsung/smdk_wm8580.c | 4 ++-- sound/soc/samsung/smdk_wm9713.c | 4 ++-- 6 files changed, 11 insertions(+), 11 deletions(-) diff --git a/sound/soc/samsung/ac97.c b/sound/soc/samsung/ac97.c index 7b9bf93..e8883d4d 100644 --- a/sound/soc/samsung/ac97.c +++ b/sound/soc/samsung/ac97.c @@ -4,7 +4,7 @@ * Evolved from s3c2443-ac97.c * * Copyright (c) 2010 Samsung Electronics Co. Ltd - * Author: Jaswinder Singh <jassi.brar(a)samsung.com> + * Author: Jaswinder Singh <jassisinghbrar(a)gmail.com> * Credits: Graeme Gregory, Sean Choi * * This program is free software; you can redistribute it and/or modify @@ -511,7 +511,7 @@ static struct platform_driver s3c_ac97_driver = { module_platform_driver(s3c_ac97_driver); -MODULE_AUTHOR("Jaswinder Singh, <jassi.brar(a)samsung.com>"); +MODULE_AUTHOR("Jaswinder Singh, <jassisinghbrar(a)gmail.com>"); MODULE_DESCRIPTION("AC97 driver for the Samsung SoC"); MODULE_LICENSE("GPL"); MODULE_ALIAS("platform:samsung-ac97"); diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 87a874d..7ccb075 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -3,7 +3,7 @@ * ALSA SoC Audio Layer - Samsung I2S Controller driver * * Copyright (c) 2010 Samsung Electronics Co. Ltd. - * Jaswinder Singh <jassi.brar(a)samsung.com> + * Jaswinder Singh <jassisinghbrar(a)gmail.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 @@ -1143,7 +1143,7 @@ static struct platform_driver samsung_i2s_driver = { module_platform_driver(samsung_i2s_driver); /* Module information */ -MODULE_AUTHOR("Jaswinder Singh, <jassi.brar(a)samsung.com>"); +MODULE_AUTHOR("Jaswinder Singh, <jassisinghbrar(a)gmail.com>"); MODULE_DESCRIPTION("Samsung I2S Interface"); MODULE_ALIAS("platform:samsung-i2s"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/samsung/i2s.h b/sound/soc/samsung/i2s.h index 8e15f6a..d420a7c 100644 --- a/sound/soc/samsung/i2s.h +++ b/sound/soc/samsung/i2s.h @@ -3,7 +3,7 @@ * ALSA SoC Audio Layer - Samsung I2S Controller driver * * Copyright (c) 2010 Samsung Electronics Co. Ltd. - * Jaswinder Singh <jassi.brar(a)samsung.com> + * Jaswinder Singh <jassisinghbrar(a)gmail.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 diff --git a/sound/soc/samsung/pcm.c b/sound/soc/samsung/pcm.c index 5678020..b7b2a1f 100644 --- a/sound/soc/samsung/pcm.c +++ b/sound/soc/samsung/pcm.c @@ -3,7 +3,7 @@ * ALSA SoC Audio Layer - S3C PCM-Controller driver * * Copyright (c) 2009 Samsung Electronics Co. Ltd - * Author: Jaswinder Singh <jassi.brar(a)samsung.com> + * Author: Jaswinder Singh <jassisinghbrar(a)gmail.com> * based upon I2S drivers by Ben Dooks. * * This program is free software; you can redistribute it and/or modify @@ -639,7 +639,7 @@ static struct platform_driver s3c_pcm_driver = { module_platform_driver(s3c_pcm_driver); /* Module information */ -MODULE_AUTHOR("Jaswinder Singh, <jassi.brar(a)samsung.com>"); +MODULE_AUTHOR("Jaswinder Singh, <jassisinghbrar(a)gmail.com>"); MODULE_DESCRIPTION("S3C PCM Controller Driver"); MODULE_LICENSE("GPL"); MODULE_ALIAS("platform:samsung-pcm"); diff --git a/sound/soc/samsung/smdk_wm8580.c b/sound/soc/samsung/smdk_wm8580.c index bff8758..ade2809 100644 --- a/sound/soc/samsung/smdk_wm8580.c +++ b/sound/soc/samsung/smdk_wm8580.c @@ -2,7 +2,7 @@ * smdk_wm8580.c * * Copyright (c) 2009 Samsung Electronics Co. Ltd - * Author: Jaswinder Singh <jassi.brar(a)samsung.com> + * Author: Jaswinder Singh <jassisinghbrar(a)gmail.com> * * This program is free software; you can redistribute it and/or modify it * under the terms of the GNU General Public License as published by the @@ -253,6 +253,6 @@ static void __exit smdk_audio_exit(void) } module_exit(smdk_audio_exit); -MODULE_AUTHOR("Jaswinder Singh, jassi.brar(a)samsung.com"); +MODULE_AUTHOR("Jaswinder Singh, jassisinghbrar(a)gmail.com"); MODULE_DESCRIPTION("ALSA SoC SMDK WM8580"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/samsung/smdk_wm9713.c b/sound/soc/samsung/smdk_wm9713.c index 8e26a73..55b2ca7 100644 --- a/sound/soc/samsung/smdk_wm9713.c +++ b/sound/soc/samsung/smdk_wm9713.c @@ -2,7 +2,7 @@ * smdk_wm9713.c -- SoC audio for SMDK * * Copyright 2010 Samsung Electronics Co. Ltd. - * Author: Jaswinder Singh Brar <jassi.brar(a)samsung.com> + * Author: Jaswinder Singh Brar <jassisinghbrar(a)gmail.com> * * This program is free software; you can redistribute it and/or * modify it under the terms of the GNU General Public License as @@ -103,6 +103,6 @@ module_init(smdk_init); module_exit(smdk_exit); /* Module information */ -MODULE_AUTHOR("Jaswinder Singh Brar, jassi.brar(a)samsung.com"); +MODULE_AUTHOR("Jaswinder Singh Brar, jassisinghbrar(a)gmail.com"); MODULE_DESCRIPTION("ALSA SoC SMDK+WM9713"); MODULE_LICENSE("GPL"); -- 1.7.4.1

2 1

[alsa-devel] [PATCH] ASoC: Samsung: Merge two identical if-else clauses
by Jassi Brar 01 Mar '12

01 Mar '12

Saves two lines and a hell of a lot of embarrassment looking at the code. Signed-off-by: Jassi Brar <jaswinder.singh(a)linaro.org> --- sound/soc/samsung/i2s.c | 10 ++++------ 1 files changed, 4 insertions(+), 6 deletions(-) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 7ccb075..6553b19 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -761,15 +761,13 @@ static int i2s_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_PAUSE_PUSH: local_irq_save(flags); - if (capture) + if (capture) { i2s_rxctrl(i2s, 0); - else - i2s_txctrl(i2s, 0); - - if (capture) i2s_fifo(i2s, FIC_RXFLUSH); - else + } else { + i2s_txctrl(i2s, 0); i2s_fifo(i2s, FIC_TXFLUSH); + } local_irq_restore(flags); break; -- 1.7.4.1

2 1

[alsa-devel] [PATCH] ASoC: wm8996: Fix /RESET bounce ordering
by Mark Brown 29 Feb '12

29 Feb '12

We want to leave the device out of rather than in reset. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8996.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index c1c7ef0..9376b19 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -1719,8 +1719,8 @@ static bool wm8996_volatile_register(struct device *dev, unsigned int reg) static int wm8996_reset(struct wm8996_priv *wm8996) { if (wm8996->pdata.ldo_ena > 0) { - gpio_set_value_cansleep(wm8996->pdata.ldo_ena, 1); gpio_set_value_cansleep(wm8996->pdata.ldo_ena, 0); + gpio_set_value_cansleep(wm8996->pdata.ldo_ena, 1); return 0; } else { return regmap_write(wm8996->regmap, WM8996_SOFTWARE_RESET, -- 1.7.9.1

1 0

[alsa-devel] [PATCH] ASoC: wm_hubs: Bomb out if we can't read back the DC servo result
by Mark Brown 29 Feb '12

29 Feb '12

Should have no practical impact but it's safer than trying to soldier on. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_hubs.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index 689311c..9d60a6e 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -172,7 +172,7 @@ static void calibrate_dc_servo(struct snd_soc_codec *codec) break; default: WARN(1, "Unknown DCS readback method\n"); - break; + return; } dev_dbg(codec->dev, "DCS input: %x %x\n", reg_l, reg_r); -- 1.7.9.1

1 0

[alsa-devel] [PATCH] ASoC: wm8994: Make sure we don't have MICBIAS on during jackdet mode
by Mark Brown 29 Feb '12

29 Feb '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 25 +++++++++++++------------ 1 files changed, 13 insertions(+), 12 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 33bc718..691e897 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3170,9 +3170,6 @@ static void wm8958_default_micdet(u16 status, void *data) snd_soc_update_bits(codec, WM8958_MIC_DETECT_1, WM8958_MICD_ENA, 0); - wm1811_jackdet_set_mode(codec, - WM1811_JACKDET_MODE_JACK); - if (wm8994->pdata->jd_ext_cap) { mutex_lock(&codec->mutex); snd_soc_dapm_disable_pin(&codec->dapm, @@ -3180,6 +3177,9 @@ static void wm8958_default_micdet(u16 status, void *data) snd_soc_dapm_sync(&codec->dapm); mutex_unlock(&codec->mutex); } + + wm1811_jackdet_set_mode(codec, + WM1811_JACKDET_MODE_JACK); } } @@ -3235,23 +3235,24 @@ static irqreturn_t wm1811_jackdet_irq(int irq, void *data) snd_soc_update_bits(codec, WM8958_MICBIAS2, WM8958_MICB2_DISCH, 0); - /* If required for an external cap force MICBIAS on */ - if (wm8994->pdata->jd_ext_cap) { - mutex_lock(&codec->mutex); - snd_soc_dapm_force_enable_pin(&codec->dapm, - "MICBIAS2"); - snd_soc_dapm_sync(&codec->dapm); - mutex_unlock(&codec->mutex); - } - /* * Start off measument of microphone impedence to find * out what's actually there. */ wm8994->mic_detecting = true; wm1811_jackdet_set_mode(codec, WM1811_JACKDET_MODE_MIC); + snd_soc_update_bits(codec, WM8958_MIC_DETECT_1, WM8958_MICD_ENA, WM8958_MICD_ENA); + + /* If required for an external cap force MICBIAS on */ + if (wm8994->pdata->jd_ext_cap) { + mutex_lock(&codec->mutex); + snd_soc_dapm_force_enable_pin(&codec->dapm, + "MICBIAS2"); + snd_soc_dapm_sync(&codec->dapm); + mutex_unlock(&codec->mutex); + } } else { dev_dbg(codec->dev, "Jack not detected\n"); -- 1.7.9.1

1 0

[alsa-devel] [PATCH 0/2] ASoC: omap: mcbsp: small cleanups for the sidetone code
by Peter Ujfalusi 28 Feb '12

28 Feb '12

Hello, to make the code a bit nicer to look at. This series is on top of the McBSP move changeset: http://mailman.alsa-project.org/pipermail/alsa-devel/2012-February/049628.h… Regards, Peter --- Peter Ujfalusi (2): ASoC: omap: mcbsp: Use uniform st_data pointer initialization ASoC: omap: mcbsp: Remove redundant checks for the st_data pointer sound/soc/omap/mcbsp.c | 25 +++++++------------------ 1 files changed, 7 insertions(+), 18 deletions(-) -- 1.7.8.4

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[alsa-devel] [PATCH] ALSA: hda - Add position_fix=4 (COMBO) option
by Takashi Iwai 28 Feb '12

28 Feb '12

This patch adds a new position_fix option value, 4, as a combo mode to use LPIB for playbacks and POSBUF for captures. It's the way recommended by Intel hardware guys. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- Seth, this is the patch to add the mode you mentioned. It'll be queued for 3.4 kernel. This mode isn't chosen as default yet, but if this is confirmed to work more reliably on Intel controllers, we can set these as default for them. Documentation/sound/alsa/ALSA-Configuration.txt | 8 +++++++- Documentation/sound/alsa/HD-Audio.txt | 7 ++++++- sound/pci/hda/hda_intel.c | 10 +++++++++- 3 files changed, 22 insertions(+), 3 deletions(-) diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index 936699e..9af64c5 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -860,7 +860,8 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. [Multiple options for each card instance] model - force the model name - position_fix - Fix DMA pointer (0 = auto, 1 = use LPIB, 2 = POSBUF) + position_fix - Fix DMA pointer (0 = auto, 1 = use LPIB, 2 = POSBUF, + 3 = VIACOMBO, 4 = COMBO) probe_mask - Bitmask to probe codecs (default = -1, meaning all slots) When the bit 8 (0x100) is set, the lower 8 bits are used as the "fixed" codec slots; i.e. the driver probes the @@ -925,6 +926,11 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. (Usually SD_LPIB register is more accurate than the position buffer.) + position_fix=3 is specific to VIA devices. The position + of the capture stream is checked from both LPIB and POSBUF + values. position_fix=4 is a combination mode, using LPIB + for playback and POSBUF for capture. + NB: If you get many "azx_get_response timeout" messages at loading, it's likely a problem of interrupts (e.g. ACPI irq routing). Try to boot with options like "pci=noacpi". Also, you diff --git a/Documentation/sound/alsa/HD-Audio.txt b/Documentation/sound/alsa/HD-Audio.txt index 91fee3b..7813c06 100644 --- a/Documentation/sound/alsa/HD-Audio.txt +++ b/Documentation/sound/alsa/HD-Audio.txt @@ -59,7 +59,12 @@ a case, you can change the default method via `position_fix` option. `position_fix=1` means to use LPIB method explicitly. `position_fix=2` means to use the position-buffer. `position_fix=3` means to use a combination of both methods, needed -for some VIA and ATI controllers. 0 is the default value for all other +for some VIA controllers. The capture stream position is corrected +by comparing both LPIB and position-buffer values. +`position_fix=4` is another combination available for all controllers, +and uses LPIB for the playback and the position-buffer for the capture +streams. +0 is the default value for all other controllers, the automatic check and fallback to LPIB as described in the above. If you get a problem of repeated sounds, this option might help. diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index e354c16..6e958bf 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -84,7 +84,7 @@ module_param_array(model, charp, NULL, 0444); MODULE_PARM_DESC(model, "Use the given board model."); module_param_array(position_fix, int, NULL, 0444); MODULE_PARM_DESC(position_fix, "DMA pointer read method." - "(0 = auto, 1 = LPIB, 2 = POSBUF, 3 = VIACOMBO)."); + "(0 = auto, 1 = LPIB, 2 = POSBUF, 3 = VIACOMBO, 4 = COMBO)."); module_param_array(bdl_pos_adj, int, NULL, 0644); MODULE_PARM_DESC(bdl_pos_adj, "BDL position adjustment offset."); module_param_array(probe_mask, int, NULL, 0444); @@ -330,6 +330,7 @@ enum { POS_FIX_LPIB, POS_FIX_POSBUF, POS_FIX_VIACOMBO, + POS_FIX_COMBO, }; /* Defines for ATI HD Audio support in SB450 south bridge */ @@ -2520,6 +2521,7 @@ static int __devinit check_position_fix(struct azx *chip, int fix) case POS_FIX_LPIB: case POS_FIX_POSBUF: case POS_FIX_VIACOMBO: + case POS_FIX_COMBO: return fix; } @@ -2699,6 +2701,12 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci, chip->position_fix[0] = chip->position_fix[1] = check_position_fix(chip, position_fix[dev]); + /* combo mode uses LPIB for playback */ + if (chip->position_fix[0] == POS_FIX_COMBO) { + chip->position_fix[0] = POS_FIX_LPIB; + chip->position_fix[1] = POS_FIX_AUTO; + } + check_probe_mask(chip, dev); chip->single_cmd = single_cmd; -- 1.7.9

2 2

[alsa-devel] [PATCH 1/4] ASoC: imx: let SND_MXC_SOC_FIQ select FIQ
by Shawn Guo 28 Feb '12

28 Feb '12

CONFIG_FIQ is only needed when CONFIG_SND_MXC_SOC_FIQ is selected to build imx-pcm-fiq.c, so let SND_MXC_SOC_FIQ select FIQ. Signed-off-by: Shawn Guo <shawn.guo(a)linaro.org> --- sound/soc/imx/Kconfig | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/imx/Kconfig b/sound/soc/imx/Kconfig index 91b66ef..17810d9 100644 --- a/sound/soc/imx/Kconfig +++ b/sound/soc/imx/Kconfig @@ -1,7 +1,6 @@ menuconfig SND_IMX_SOC tristate "SoC Audio for Freescale i.MX CPUs" depends on ARCH_MXC - select FIQ select SND_SOC_AC97_BUS help Say Y or M if you want to add support for codecs attached to @@ -12,6 +11,7 @@ if SND_IMX_SOC config SND_MXC_SOC_FIQ tristate + select FIQ config SND_MXC_SOC_MX2 tristate -- 1.7.5.4

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Re: [alsa-devel] Cranking sound coming out of speakers with the new 3.1 kernel
by Takashi Iwai 28 Feb '12

28 Feb '12

At Tue, 28 Feb 2012 16:16:48 +0100, Jan Prunk wrote: > > Hello ! > > On Sat, Jan 14, 2012 at 12:04 AM, Jonathan Nieder <jrnieder(a)gmail.com> wrote: > > Hi again, > > > > Jan Prunk wrote: > > > >> I tried booting into 3.2.0-rc7-686-pae, but the crunching sound > >> is still present, I am able to play music file properly, and > >> crunching is in the background, and also while not processing > >> any sounds, the crunching appears out of the speakers. > > > > Ok, thanks for testing. Please send a summary of this regression to > > alsa-devel(a)alsa-project.org, cc-ing Takashi Iwai <tiwai(a)suse.de> and > > either me or this bug log so we can track it. > > > > Be sure to mention: > > > > - steps to reproduce, what you expect versus what actually happens, > > and how that indicates a bug > > The sound is ok in kernel version "3.0.0-1-686-pae" but on all newer > versions there are sparkling/farting noises coming out of the speaker. > The noise is not very loud, its barely noticeable, but its repetative > and doesn't go away. And I think it happens while typing on the > keyboard and a bit later, then it dissapears for a few seconds, > or by the time that I type again. > > > - which kernel versions you tried, and what happens with each > I tried running alsa-info.sh as a "user". > > I tried 3.0.0-1-686-pae here is the debug: > http://www.alsa-project.org/db/?f=c1149b6466b2068d79c7517a8b0808d8b5c26e6b > --- > Linux vaio 3.2.0-1-686-pae #1 SMP Fri Feb 17 06:27:21 UTC 2012 i686 GNU/Linux > ii linux-image-3.2.0-1-686-pae 3.2.6-1 > http://www.alsa-project.org/db/?f=fddbfbdc4fc44555a6dc851ecdc70760da6dd840 > --- > Linux vaio 3.2.0-rc7-686-pae #1 SMP Wed Dec 28 21:26:25 UTC 2011 i686 GNU/Linux > ii linux-image-3.2.0-rc7-686-pae 3.2~rc7-1~experimental.1 > http://www.alsa-project.org/db/?f=78c8919e87dd03294484f83b5f7ae44331b5edfe > Being booted into this version for 15 minutes, it seems to not make any noises, > but I will resubmit the info later if the noises will start to appear. 3.2.0-rc7 alsa-info shows that the headphone jack is plugged, thus the speaker is muted. It's natural that the noise goes away in this state. Apart from that, I see no obvious problem in 3.2.0 output. Does the problem still happen when you disable "Auto-Mute Mode" mixer enum? % amixer -c0 set "Auto-Mute Mode" "Disabled" Also, try to pass algin_buffer_size=0 option to snd-hda-intel. I don't expect much that this will influence, but at least, it's a difference between 3.0 and 3.2. Takashi > > - alsa-info.sh output, as an attachment > > > > - if you can make a recording of the strange sound available > > somewhere online, that would be ideal > > Because the noise is barely heard, I would need to find a good mic. > to be able to record it, but I could do that in the next bug submission. > > > - whether you are able to bisect or test patches if needed > > I might try that, if there are any manuals around how to do it, > but my response might take some time. > > > [1] may have more hints. > > > > Thanks and good luck, > > Jonathan > > > > [1] http://alsa-project.org/main/index.php/Help_To_Debug_Intel_HDA > > Kind regards, > Jan Prunk > -- > Jan Prunk http://www.prunk.si > 0x00E80E86 http://pgp.prunk.si > http://AS50763.peeringdb.com >

1 0

[alsa-devel] [PATCH] ASoC: DT: Update digital microphone binding documentation to PAZ00 board.
by Leon Romanovsky 28 Feb '12

28 Feb '12

This patch updates device tree binding documentation to add digital microphone to PAZ00 board. Signed-off-by: Leon Romanovsky <leon(a)leon.nu> --- .../bindings/sound/tegra-audio-alc5632.txt | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/tegra-audio-alc5632.txt b/Documentation/devicetree/bindings/sound/tegra-audio-alc5632.txt index 300f2da..9e305bb 100644 --- a/Documentation/devicetree/bindings/sound/tegra-audio-alc5632.txt +++ b/Documentation/devicetree/bindings/sound/tegra-audio-alc5632.txt @@ -25,12 +25,14 @@ Required properties: * MIC2_P * MIC2_N * MICBIAS1 + * DMICDAT Board connectors: * Headset Stereophone * Int Spk * Headset Mic + * Digital Mic - nvidia,i2s-controller : The phandle of the Tegra I2S controller - nvidia,audio-codec : The phandle of the ALC5632 audio codec -- 1.7.3.4

2 1

[alsa-devel] [PATCH v2 0/9] imx patches when I enable imx6q sabrelite audio
by Richard Zhao 28 Feb '12

28 Feb '12

Changes in v2: - change dt peroperty/label to lower case as much as possible keep XXX-supply property upper case. - convert audmux-v2 to platform device driver - move audmux-v2 driver to sound/soc/imx [PATCH v2 1/9] ARM: mxc: make imx_dma_is_general_purpose more [PATCH v2 2/9] ARM: imx6q: add cko1 clock [PATCH v2 3/9] ARM: dts: imx6q-sabrelite: add 2p5v and 3p3v [PATCH v2 4/9] ARM: dts: imx6q-sabrelite: add sgtl5000 audio codec [PATCH v2 5/9] ARM: mxc: convert audmux-v2 to a platform driver [PATCH v2 6/9] ARM: mx31ads: add audmux device [PATCH v2 7/9] ARM: mxc: move audmux-v2 to sound/soc/imx [PATCH v2 8/9] ASoC: imx: add dt support for audmux-v2 [PATCH v2 9/9] ARM: imx6q-sabrelite: add audmux support Thanks Richard

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[alsa-devel] [GIT PULL FOR v3.4] tea575x update to latest V4L2 framework requirements
by Hans Verkuil 28 Feb '12

28 Feb '12

Hi Mauro, This patch series updates the tea575x drivers to the latest V4L2 framework requirements, converts radio-maxiradio to tea575x as well and adds hwseek support to these drivers. All changes are V4L related, even though some of the patches touch sound drivers (but only the V4L parts of those sound drivers). Tested with SF16-FMR2, SF64-PCR, SF64-PCE2, SF256-PCP (Ondrej) and maxiradio (Hans). Regards, Hans The following changes since commit a3db60bcf7671cc011ab4f848cbc40ff7ab52c1e: [media] xc5000: declare firmware configuration structures as static const (2012-02-14 17:22:46 -0200) are available in the git repository at: git://linuxtv.org/hverkuil/media_tree.git radio-pci Hans Verkuil (4): tea575x-tuner: update to latest V4L2 framework requirements. tea575x: fix HW seek radio-maxiradio: use the tea575x framework. V4L2 Spec: return -EINVAL on unsupported wrap_around value. .../DocBook/media/v4l/vidioc-s-hw-freq-seek.xml | 6 +- drivers/media/radio/Kconfig | 2 +- drivers/media/radio/radio-maxiradio.c | 379 ++++---------------- drivers/media/radio/radio-sf16fmr2.c | 61 +++- include/sound/tea575x-tuner.h | 6 +- sound/i2c/other/tea575x-tuner.c | 169 ++++++--- sound/pci/es1968.c | 15 + sound/pci/fm801.c | 20 +- 8 files changed, 273 insertions(+), 385 deletions(-)

1 0

[alsa-devel] [PATCH] ASoC: omap3beagle: Add dai link for McBSP2 master configuration (CLKS source)
by Peter Ujfalusi 28 Feb '12

28 Feb '12

Create new dai link so we can choose between two different modes of operation: PCM0: twl4030 master, McBSP2 slave (current configuration) PCM1: twl4030 slave, McBSP2 slave (SRG source is from CLKS pin) This will help us tracking down, and test different setups on Beagle board. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- Hello, While tracking down an issue regported by Grazvydas Ignotas regarding to the McBSP move series (Pandora setup was not working) I have noticed that we can use the Beagle board in a similar way as Pandora uses it (McBSP master, twl4030 slave, CLKS pin is connected to twl4030 256*FS clock output). With this patch we can switch between the two modes by playing on hw:0,0 or on hw:0,1. I will add the third option later where McBSP is the master, but the clock for SRG is coming from internal source. This patch depends on the McBSP rewrite series (OMAP/ASoC: Move and merge McBSP driver under ASoC) http://mailman.alsa-project.org/pipermail/alsa-devel/2012-February/049628.h… Regards, Peter sound/soc/omap/omap3beagle.c | 110 ++++++++++++++++++++++++++++++++++++++---- 1 files changed, 100 insertions(+), 10 deletions(-) diff --git a/sound/soc/omap/omap3beagle.c b/sound/soc/omap/omap3beagle.c index 2830dfd..a2f9130 100644 --- a/sound/soc/omap/omap3beagle.c +++ b/sound/soc/omap/omap3beagle.c @@ -34,6 +34,7 @@ #include "omap-mcbsp.h" #include "omap-pcm.h" +/* McBSP2 slave, TWL4030 master */ static int omap3beagle_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -80,6 +81,14 @@ static int omap3beagle_hw_params(struct snd_pcm_substream *substream, return ret; } + /* Set McBSP clock to PER_96M_FCLK */ + ret = snd_soc_dai_set_sysclk(cpu_dai, OMAP_MCBSP_SYSCLK_CLKS_FCLK, + 96000000, SND_SOC_CLOCK_IN); + if (ret < 0) { + printk(KERN_ERR "can't set cpu system clock\n"); + return ret; + } + return 0; } @@ -87,23 +96,104 @@ static struct snd_soc_ops omap3beagle_ops = { .hw_params = omap3beagle_hw_params, }; +/* McBSP2 master, TWL4030 slave */ +static int omap3beagle_slave_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + unsigned int fmt; + int ret; + + switch (params_channels(params)) { + case 2: /* Stereo I2S mode */ + fmt = SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS; + break; + case 4: /* Four channel TDM mode */ + fmt = SND_SOC_DAIFMT_DSP_A | + SND_SOC_DAIFMT_IB_NF | + SND_SOC_DAIFMT_CBS_CFS; + break; + default: + return -EINVAL; + } + + /* Set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, fmt); + if (ret < 0) { + printk(KERN_ERR "can't set codec DAI configuration\n"); + return ret; + } + + /* Set cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, fmt); + if (ret < 0) { + printk(KERN_ERR "can't set cpu DAI configuration\n"); + return ret; + } + + /* Set the codec system clock for DAC and ADC */ + ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000, + SND_SOC_CLOCK_IN); + if (ret < 0) { + printk(KERN_ERR "can't set codec system clock\n"); + return ret; + } + + /* Set McBSP clock to external (CLKS) */ + ret = snd_soc_dai_set_sysclk(cpu_dai, OMAP_MCBSP_SYSCLK_CLKS_EXT, + 256 * params_rate(params), + SND_SOC_CLOCK_IN); + if (ret < 0) { + printk(KERN_ERR "can't set cpu system clock\n"); + return ret; + } + + ret = snd_soc_dai_set_clkdiv(cpu_dai, OMAP_MCBSP_CLKGDV, 8); + if (ret < 0) { + printk(KERN_ERR "can't set SRG clock divider\n"); + return ret; + } + + return 0; +} + +static struct snd_soc_ops omap3beagle_slave_ops = { + .hw_params = omap3beagle_slave_hw_params, +}; + /* Digital audio interface glue - connects codec <--> CPU */ -static struct snd_soc_dai_link omap3beagle_dai = { - .name = "TWL4030", - .stream_name = "TWL4030", - .cpu_dai_name = "omap-mcbsp.2", - .platform_name = "omap-pcm-audio", - .codec_dai_name = "twl4030-hifi", - .codec_name = "twl4030-codec", - .ops = &omap3beagle_ops, +static struct snd_soc_dai_link omap3beagle_dai[] = { + { + /* McBSP2 slave, TWL4030 master configuration */ + .name = "TWL4030 master", + .stream_name = "TWL4030 master", + .cpu_dai_name = "omap-mcbsp.2", + .platform_name = "omap-pcm-audio", + .codec_dai_name = "twl4030-hifi", + .codec_name = "twl4030-codec", + .ops = &omap3beagle_ops, + }, { + /* McBSP2 master, TWL4030 slave configuration */ + .name = "TWL4030 slave", + .stream_name = "TWL4030 slave", + .cpu_dai_name = "omap-mcbsp.2", + .platform_name = "omap-pcm-audio", + .codec_dai_name = "twl4030-hifi", + .codec_name = "twl4030-codec", + .ops = &omap3beagle_slave_ops, + }, }; /* Audio machine driver */ static struct snd_soc_card snd_soc_omap3beagle = { .name = "omap3beagle", .owner = THIS_MODULE, - .dai_link = &omap3beagle_dai, - .num_links = 1, + .dai_link = omap3beagle_dai, + .num_links = ARRAY_SIZE(omap3beagle_dai), }; static struct platform_device *omap3beagle_snd_device; -- 1.7.8.4

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Re: [alsa-devel] alsa usb audio class 2
by Daniel Mack 28 Feb '12

28 Feb '12

On 28.02.2012 09:57, Markus Bollinger wrote: > OK, thank you Takashi. > I attached the log file of the 'lsusb -v' output Thanks for the dump. Are there any kernel messages that are generated when plugging in the device? Check with 'dmesg'. Also, from the dump I can see that some string descriptors are really broken in this device. Don't know if that causes your trouble though. Daniel

1 0

[alsa-devel] [PATCH 00/38] Realtek HD-audio auto-parser
by Takashi Iwai 28 Feb '12

28 Feb '12

Hi, for 3.4 kernel, finally I got rid of all static quirks from HD-audio Realtek codec parser. This means that the all necessary quirks have been converted to alc_fixup either the pin-config overrides, the additional verbs, or the extra init functions. Since some machines have really weird pin mappings, I introduced the new mechanism in the parser to evaluate the best DAC->pin connections, which is based on the badness calculation. You won't notice on most of machines but some machines that have fewer DACs than necessary output pins will get different mappings by this change. The whole changes can be seen in sound git tree topic/hda branch, as usual. git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git Now it's merged to main branch, so the snapshot tarball contains these changes, too. The rest of the thread will contain the changes in patch_realtek.c. It's just for refernces, and I omitted the other files. For the complete changes, take a look at the git tree. thanks, Takashi

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[alsa-devel] [PATCH] ALSA : Au88x0 - Support 4 channels playback when AC97 codec has SDAC bit , Reduce the number of playback devices of au8830 from 32 to 16
by Raymond Yau 28 Feb '12

28 Feb '12

[PATCH - ALSA 1/2] Au88x0 - Support 4 channels playback when AC97 codecs has SDAC bit - Check SDAC bit of AC97 codec for supporting 4 channels playback. [PATCH - ALSA 2/2] Au88x0 - Reduce the number of playback subdevices of au8830 from 32 to 16 - The maximum number of playback streams depend on the number of sample rate conveters (16) and the number of DMA channels (32).

2 3

[alsa-devel] question on Asoc tlv320aic3104
by rohith R 28 Feb '12

28 Feb '12

Hi, I'm modifying Asoc tlv320aic3104 codec driver to support MIC3L/R inputs(currently not supported in the code.) I'm a newbie to ALSA programming and had these questions: 1.Is there any documentation for ASOC apart from the one in kernel sources? 2.ALSA api reference guide link on the website is giving a 404 error.Any updated link? 3.Trying to program Page0 register17/18 to add as a "route".Which is the best way to do it.Still new to the range of SOC_** #defines available. --Rohith

1 0

[alsa-devel] [PATCH] OMAP4: panda: Correct cpu version check for 4430
by Peter Ujfalusi 28 Feb '12

28 Feb '12

The cpu_is_omap4430() macro always return with 0. Use the correct cpu_is_omap443x() to check for Panda revision. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- Hi, Since the depending patches are going via Liam's tree it would be best if we can queue this via audio as well. Without this patch the audio on Panda 4430 is not working correctly. Regards, Peter arch/arm/mach-omap2/board-omap4panda.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/arch/arm/mach-omap2/board-omap4panda.c b/arch/arm/mach-omap2/board-omap4panda.c index eada955..74db647 100644 --- a/arch/arm/mach-omap2/board-omap4panda.c +++ b/arch/arm/mach-omap2/board-omap4panda.c @@ -520,7 +520,7 @@ void omap4_panda_display_init(void) static void omap4_panda_init_rev(void) { - if (cpu_is_omap4430()) { + if (cpu_is_omap443x()) { /* PandaBoard 4430 */ /* ASoC audio configuration */ panda_abe_audio_data.card_name = "PandaBoard"; -- 1.7.8.4

2 2

Re: [alsa-devel] [PATCH] Sony Vaio with stac9872
by Takashi Iwai 28 Feb '12

28 Feb '12

At Tue, 28 Feb 2012 07:26:25 +0300, 107(a)mail.ru wrote: > > 2012/2/27 Takashi Iwai <tiwai(a)suse.de>: > > At Fri, 24 Feb 2012 16:41:43 +0400, > > 107 wrote: > >> > >> Hello, I have no sound on my Sony Vaio SZ4XRN laptop with newest kernels. > >> Sound: hda-intel, sigmatel - STAC9872 > >> > >> To solve this issue I make this patch for sound/pci/hda/patch_sigmatel.c: > >> > >> --- patch_sigmatel.c.bak 2012-02-23 21:01:28.577515317 +0400 > >> +++ patch_sigmatel.c 2012-02-23 21:53:21.035605216 +0400 > >> @@ -6357,7 +6357,16 @@ > >> */ > >> > >> static const struct hda_verb stac9872_core_init[] = { > >> - {0x15, AC_VERB_SET_CONNECT_SEL, 0x1}, /* mic-sel: 0a,0d,14,02 */ > >> + {0x0a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP }, /* HP <- 0x2 */ > >> + {0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, /* Speaker <- 0x5 */ > >> + {0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, /* Mic? (<- 0x2) */ > >> + {0x0e, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, /* CD */ > >> + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, /* Mic? */ > >> + {0x15, AC_VERB_SET_CONNECT_SEL, 0x2 /*0x1*/ }, /* mic-sel: 0a,0d,14,02 */ > >> + {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* HP */ > >> + {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* Speaker */ > >> + {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* capture sw/vol -> 0x8 */ > >> + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, /* CD-in -> 0x6 */ > >> {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Mic-in -> 0x9 */ > >> {} > >> }; > >> > >> > >> Constants got from old kernel (2.6.22). > >> With this patch sound works fine. Thank you. > > > > Most of verbs above must be superfluous, but some of them might be > > needed in your case. Could you give alsa-info.sh outputs (run with > > --no-upload option) on both working and non-working cases? > > > > > > thanks, > > > > Takashi > > > > Yes of course, here is my alsa-info-nosound.txt on default kernel and > alsa-info-work.txt with patch (see attach). Thanks. Could you check whether the one-liner below fixes your problem? Takashi --- diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 8c346ac..5988dbd 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -4638,7 +4638,7 @@ static void stac92xx_hp_detect(struct hda_codec *codec) unsigned int val = AC_PINCTL_OUT_EN | AC_PINCTL_HP_EN; if (no_hp_sensing(spec, i)) continue; - if (presence) + if (1 /*presence*/) stac92xx_set_pinctl(codec, cfg->hp_pins[i], val); #if 0 /* FIXME */ /* Resetting the pinctl like below may lead to (a sort of) regressions

1 0

[alsa-devel] [PATCH] tea575x: fix HW seek
by Ondrej Zary 27 Feb '12

27 Feb '12

Fix HW seek in TEA575x to work properly: - a delay must be present after search start and before first register read or the seek does weird things - when the search stops, the new frequency is not available immediately, we must wait until it appears in the register (fortunately, we can clear the frequency bits when starting the search as it starts at the frequency currently set, not from the value written) - sometimes, seek remains on the current frequency (or moves only a little), so repeat it until it moves by at least 50 kHz Signed-off-by: Ondrej Zary <linux(a)rainbow-software.org> --- a/sound/i2c/other/tea575x-tuner.c +++ b/sound/i2c/other/tea575x-tuner.c @@ -89,7 +89,7 @@ static void snd_tea575x_write(struct snd_tea575x *tea, unsigned int val) tea->ops->set_pins(tea, 0); } -static unsigned int snd_tea575x_read(struct snd_tea575x *tea) +static u32 snd_tea575x_read(struct snd_tea575x *tea) { u16 l, rdata; u32 data = 0; @@ -120,6 +120,27 @@ static unsigned int snd_tea575x_read(struct snd_tea575x *tea) return data; } +static void snd_tea575x_get_freq(struct snd_tea575x *tea) +{ + u32 freq = snd_tea575x_read(tea) & TEA575X_BIT_FREQ_MASK; + + if (freq == 0) { + tea->freq = 0; + return; + } + + /* freq *= 12.5 */ + freq *= 125; + freq /= 10; + /* crystal fixup */ + if (tea->tea5759) + freq += TEA575X_FMIF; + else + freq -= TEA575X_FMIF; + + tea->freq = clamp(freq * 16, FREQ_LO, FREQ_HI); /* from kHz */ +} + static void snd_tea575x_set_freq(struct snd_tea575x *tea) { u32 freq = tea->freq; @@ -203,6 +224,8 @@ static int vidioc_g_frequency(struct file *file, void *priv, if (f->tuner != 0) return -EINVAL; f->type = V4L2_TUNER_RADIO; + if (!tea->cannot_read_data) + snd_tea575x_get_freq(tea); f->frequency = tea->freq; return 0; } @@ -225,36 +248,50 @@ static int vidioc_s_hw_freq_seek(struct file *file, void *fh, struct v4l2_hw_freq_seek *a) { struct snd_tea575x *tea = video_drvdata(file); + int i, old_freq; + unsigned long timeout; if (tea->cannot_read_data) return -ENOTTY; + + snd_tea575x_get_freq(tea); + old_freq = tea->freq; + /* clear the frequency, HW will fill it in */ + tea->val &= ~TEA575X_BIT_FREQ_MASK; tea->val |= TEA575X_BIT_SEARCH; - tea->val &= ~TEA575X_BIT_UPDOWN; if (a->seek_upward) tea->val |= TEA575X_BIT_UPDOWN; + else + tea->val &= ~TEA575X_BIT_UPDOWN; snd_tea575x_write(tea, tea->val); + timeout = jiffies + msecs_to_jiffies(10000); for (;;) { - unsigned val = snd_tea575x_read(tea); - - if (!(val & TEA575X_BIT_SEARCH)) { - /* Found a frequency */ - val &= TEA575X_BIT_FREQ_MASK; - val = (val * 10) / 125; - if (tea->tea5759) - val += TEA575X_FMIF; - else - val -= TEA575X_FMIF; - tea->freq = clamp(val * 16, FREQ_LO, FREQ_HI); - return 0; - } + if (time_after(jiffies, timeout)) + break; if (schedule_timeout_interruptible(msecs_to_jiffies(10))) { /* some signal arrived, stop search */ tea->val &= ~TEA575X_BIT_SEARCH; snd_tea575x_write(tea, tea->val); return -ERESTARTSYS; } + if (!(snd_tea575x_read(tea) & TEA575X_BIT_SEARCH)) { + /* Found a frequency, wait until it can be read */ + for (i = 0; i < 100; i++) { + msleep(10); + snd_tea575x_get_freq(tea); + if (tea->freq != 0) /* available */ + break; + } + /* if we moved by less than 50 kHz, continue seeking */ + if (abs(old_freq - tea->freq) < 16 * 500) { + snd_tea575x_write(tea, tea->val); + continue; + } + tea->val &= ~TEA575X_BIT_SEARCH; + return 0; + } } - return 0; + return -ETIMEDOUT; } static int tea575x_s_ctrl(struct v4l2_ctrl *ctrl) @@ -318,7 +355,7 @@ int snd_tea575x_init(struct snd_tea575x *tea) return -ENODEV; } - tea->val = TEA575X_BIT_BAND_FM | TEA575X_BIT_SEARCH_10_40; + tea->val = TEA575X_BIT_BAND_FM | TEA575X_BIT_SEARCH_5_28; tea->freq = 90500 * 16; /* 90.5Mhz default */ snd_tea575x_set_freq(tea); -- Ondrej Zary

2 6

[alsa-devel] [PATCH 1/2] ALSA: Use a define for the number of jack switch types
by Mark Brown 27 Feb '12

27 Feb '12

This is intended to facilitate the merge of the two jack detection mechanisms. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/jack.h | 3 +++ sound/core/jack.c | 4 ++-- 2 files changed, 5 insertions(+), 2 deletions(-) diff --git a/include/sound/jack.h b/include/sound/jack.h index 63c7907..5891657 100644 --- a/include/sound/jack.h +++ b/include/sound/jack.h @@ -53,6 +53,9 @@ enum snd_jack_types { SND_JACK_BTN_5 = 0x0200, }; +/* Keep in sync with definitions above */ +#define SND_JACK_SWITCH_TYPES 6 + struct snd_jack { struct input_dev *input_dev; int registered; diff --git a/sound/core/jack.c b/sound/core/jack.c index 26edf63..471e1e3 100644 --- a/sound/core/jack.c +++ b/sound/core/jack.c @@ -25,7 +25,7 @@ #include <sound/jack.h> #include <sound/core.h> -static int jack_switch_types[] = { +static int jack_switch_types[SND_JACK_SWITCH_TYPES] = { SW_HEADPHONE_INSERT, SW_MICROPHONE_INSERT, SW_LINEOUT_INSERT, @@ -128,7 +128,7 @@ int snd_jack_new(struct snd_card *card, const char *id, int type, jack->type = type; - for (i = 0; i < ARRAY_SIZE(jack_switch_types); i++) + for (i = 0; i < SND_JACK_SWITCH_TYPES; i++) if (type & (1 << i)) input_set_capability(jack->input_dev, EV_SW, jack_switch_types[i]); -- 1.7.9.rc1

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[alsa-devel] [PATCH] Add default config for tegra-alc5632 (as found on paz00)
by Paul Fertser 27 Feb '12

27 Feb '12

This makes paz00 (aka Toshiba ac100, a popular arm-based netbook) use dmix/dsnoop by default since it doesn't support hw mixing. Signed-off-by: Paul Fertser <fercerpav(a)gmail.com> --- Hi, I haven't seen any other ASoC configs here so probably i'm missing something obvious. My reasoning is that when a person uses a regular notebook with HDA he gets DMix by default so it should be the case for this less common netbook too. Rate is set to 44100 because it's unlikely a person will listen to Hi-Fi music and hence it avoids resampling for the most common usecase. Feel free to drop if that's inappropriate, the device works fine with 48000 too. src/conf/cards/Makefile.am | 1 + src/conf/cards/tegra-alc5632.conf | 23 +++++++++++++++++++++++ 2 files changed, 24 insertions(+), 0 deletions(-) create mode 100644 src/conf/cards/tegra-alc5632.conf diff --git a/src/conf/cards/Makefile.am b/src/conf/cards/Makefile.am index b7190e7..efed590 100644 --- a/src/conf/cards/Makefile.am +++ b/src/conf/cards/Makefile.am @@ -44,6 +44,7 @@ cfg_files = aliases.conf \ RME9652.conf \ SI7018.conf \ SB-XFi.conf \ + tegra-alc5632.conf \ TRID4DWAVENX.conf \ USB-Audio.conf \ YMF744.conf \ diff --git a/src/conf/cards/tegra-alc5632.conf b/src/conf/cards/tegra-alc5632.conf new file mode 100644 index 0000000..371fe2a --- /dev/null +++ b/src/conf/cards/tegra-alc5632.conf @@ -0,0 +1,23 @@ +# default with dmix/dsnoop +tegra-alc5632.pcm.default { + @args [ CARD ] + @args.CARD { + type string + } + type asym + playback.pcm { + type plug + slave.pcm { + @func concat + strings [ "dmix:CARD=" $CARD ",RATE=44100" ] + } + } + capture.pcm { + type plug + slave.pcm { + @func concat + strings [ "dsnoop:CARD=" $CARD ",RATE=44100" ] + } + } +} + -- 1.7.3.4

5 8

[alsa-devel] [PATCH] ASoC: wm8753: Convert to devm_kzalloc()
by Mark Brown 27 Feb '12

27 Feb '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8753.c | 10 +++++----- 1 files changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index 21ed75d..59c28db 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -1557,7 +1557,8 @@ static int __devinit wm8753_spi_probe(struct spi_device *spi) struct wm8753_priv *wm8753; int ret; - wm8753 = kzalloc(sizeof(struct wm8753_priv), GFP_KERNEL); + wm8753 = devm_kzalloc(&spi->dev, sizeof(struct wm8753_priv), + GFP_KERNEL); if (wm8753 == NULL) return -ENOMEM; @@ -1580,7 +1581,6 @@ static int __devinit wm8753_spi_probe(struct spi_device *spi) err_regmap: regmap_exit(wm8753->regmap); err: - kfree(wm8753); return ret; } @@ -1612,7 +1612,8 @@ static __devinit int wm8753_i2c_probe(struct i2c_client *i2c, struct wm8753_priv *wm8753; int ret; - wm8753 = kzalloc(sizeof(struct wm8753_priv), GFP_KERNEL); + wm8753 = devm_kzalloc(&i2c->dev, sizeof(struct wm8753_priv), + GFP_KERNEL); if (wm8753 == NULL) return -ENOMEM; @@ -1632,10 +1633,10 @@ static __devinit int wm8753_i2c_probe(struct i2c_client *i2c, dev_err(&i2c->dev, "Failed to register CODEC: %d\n", ret); goto err_regmap; } + err_regmap: regmap_exit(wm8753->regmap); err: - kfree(wm8753); return ret; } @@ -1645,7 +1646,6 @@ static __devexit int wm8753_i2c_remove(struct i2c_client *client) snd_soc_unregister_codec(&client->dev); regmap_exit(wm8753->regmap); - kfree(wm8753); return 0; } -- 1.7.9.1

1 0

[alsa-devel] [PATCH] ASoC: imx: let SND_MXC_SOC_FIQ select FIQ
by Shawn Guo 27 Feb '12

27 Feb '12

CONFIG_FIQ is only needed when CONFIG_SND_MXC_SOC_FIQ is selected to build imx-pcm-fiq.c, so let SND_MXC_SOC_FIQ select FIQ. Signed-off-by: Shawn Guo <shawn.guo(a)linaro.org> --- sound/soc/imx/Kconfig | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/imx/Kconfig b/sound/soc/imx/Kconfig index 91b66ef..e89139f 100644 --- a/sound/soc/imx/Kconfig +++ b/sound/soc/imx/Kconfig @@ -1,7 +1,6 @@ menuconfig SND_IMX_SOC tristate "SoC Audio for Freescale i.MX CPUs" depends on ARCH_MXC - select FIQ select SND_SOC_AC97_BUS help Say Y or M if you want to add support for codecs attached to @@ -11,6 +10,7 @@ menuconfig SND_IMX_SOC if SND_IMX_SOC config SND_MXC_SOC_FIQ + select FIQ tristate config SND_MXC_SOC_MX2 -- 1.7.5.4

2 1

Re: [alsa-devel] Support new HDA codec
by Mark Brown 27 Feb '12

27 Feb '12

On Mon, Feb 27, 2012 at 03:01:11PM +0100, Jorge Fernandez Monteagudo wrote: > > Mark Brown wrote > > It should be very standards confirming and not require any magic Wolfson > > specific verbs to use the functionality, a generic HDA driver should do > > fine with an appropriate setup from the BIOS. > With the generic HDA the only enabled path is DAC1 and PORT-A, the first > it founds. In our carrier design we only use the stereo differential and > SPIDF outputs. I've implemented a 'patch_wolfson.c' in order to enable these > outputs and share the stream through both outputs, because I've no access > to the BIOS code. The inputs are not used in our carrier, then I've not > implemented anything. Please share this code. Whatever is being done to work around the BIOS probably ought to be board specific, even if the BIOS is not configurable hopefully there's enough information been added to distinguish the board.

1 0

[alsa-devel] [PATCH] Sony Vaio with stac9872
by 107 27 Feb '12

27 Feb '12

Hello, I have no sound on my Sony Vaio SZ4XRN laptop with newest kernels. Sound: hda-intel, sigmatel - STAC9872 To solve this issue I make this patch for sound/pci/hda/patch_sigmatel.c: --- patch_sigmatel.c.bak 2012-02-23 21:01:28.577515317 +0400 +++ patch_sigmatel.c 2012-02-23 21:53:21.035605216 +0400 @@ -6357,7 +6357,16 @@ */ static const struct hda_verb stac9872_core_init[] = { - {0x15, AC_VERB_SET_CONNECT_SEL, 0x1}, /* mic-sel: 0a,0d,14,02 */ + {0x0a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP }, /* HP <- 0x2 */ + {0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, /* Speaker <- 0x5 */ + {0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, /* Mic? (<- 0x2) */ + {0x0e, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, /* CD */ + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, /* Mic? */ + {0x15, AC_VERB_SET_CONNECT_SEL, 0x2 /*0x1*/ }, /* mic-sel: 0a,0d,14,02 */ + {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* HP */ + {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* Speaker */ + {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* capture sw/vol -> 0x8 */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, /* CD-in -> 0x6 */ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Mic-in -> 0x9 */ {} }; Constants got from old kernel (2.6.22). With this patch sound works fine. Thank you.

2 1

Re: [alsa-devel] [PATCH] ALSA: azt3328 - Fix NULL ptr dereference on cards without OPL3
by Takashi Iwai 25 Feb '12

25 Feb '12

At Sat, 25 Feb 2012 16:15:57 +0100, Alban Bedel wrote: > > opl3->private_data was set even if opl3 could not be created. > > Signed-off-by: Alban Bedel <albeu(a)free.fr> Thanks, applied now. Takashi > --- > sound/pci/azt3328.c | 3 +-- > 1 files changed, 1 insertions(+), 2 deletions(-) > > diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c > index 95ffa6a..496f14c 100644 > --- a/sound/pci/azt3328.c > +++ b/sound/pci/azt3328.c > @@ -2684,10 +2684,9 @@ snd_azf3328_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) > err = snd_opl3_hwdep_new(opl3, 0, 1, NULL); > if (err < 0) > goto out_err; > + opl3->private_data = chip; > } > > - opl3->private_data = chip; > - > sprintf(card->longname, "%s at 0x%lx, irq %i", > card->shortname, chip->ctrl_io, chip->irq); > > -- > 1.7.1 >

1 0

Re: [alsa-devel] linux-next: Tree for Feb 24 (sound/usb/6fire)
by Randy Dunlap 25 Feb '12

25 Feb '12

On 02/23/2012 10:05 PM, Stephen Rothwell wrote: > Hi all, > > Changes since 20120223: on x86_64: ERROR: "_snd_ctl_add_slave" [sound/usb/6fire/snd-usb-6fire.ko] undefined! ERROR: "snd_ctl_make_virtual_master" [sound/usb/6fire/snd-usb-6fire.ko] undefined! Full randconfig file is attached. CONFIG_SND_VMASTER is not enabled. Looks like this driver should either depend on or select SND_VMASTER. -- ~Randy

3 2

[alsa-devel] USB Headset, digital profile problem
by David Henningsson 25 Feb '12

25 Feb '12

Hi, I'm researching I problem I'm seeing here. As you might know PulseAudio tries to open devices in different ways, and so it is trying to open my USB headset (at card index 1) with the device string "iec958:1", and I want this to fail (to make sure this device is not shown in the user interface). So far it seems like a cold boot will cause the issue to disappear (i e, the call fails), whereas a suspend/resume can cause it to appear; but I'm not completely sure this is the actual trigger. When the call fails (that's what I want), the PulseAudio log looks like: D: [pulseaudio] alsa-util.c: Trying iec958:1 with SND_PCM_NO_AUTO_FORMAT ... I: [pulseaudio] (alsa-lib)confmisc.c: Unable to find definition 'cards.USB-Audio.pcm.iec958.0:CARD=1,AES0=4,AES1=130,AES2=0,AES3=2' I: [pulseaudio] (alsa-lib)conf.c: function snd_func_refer returned error: Filen eller katalogen finns inte I: [pulseaudio] (alsa-lib)conf.c: Evaluate error: Filen eller katalogen finns inte I: [pulseaudio] (alsa-lib)pcm.c: Unknown PCM iec958:1 I: [pulseaudio] alsa-util.c: Error opening PCM device iec958:1: Filen eller katalogen finns inte ("Filen eller katalogen finns inte" means "File or directory not found") When the call succeeds, the PulseAudio log looks like: D: [pulseaudio] alsa-util.c: Trying iec958:1 with SND_PCM_NO_AUTO_FORMAT ... D: [pulseaudio] alsa-util.c: Managed to open iec958:1 I have yet to master the alsa-lib configuration language fully, but at least I'm not changing the files between runs. Do you have an idea of what the problem could be, or how I can debug it a little deeper? For reference, all logs are at https://bugs.launchpad.net/ubuntu/+source/pulseaudio/+bug/940145 With alsa-infos at: https://bugs.launchpad.net/ubuntu/+source/pulseaudio/+bug/940145/+attachmen… https://bugs.launchpad.net/ubuntu/+source/pulseaudio/+bug/940145/+attachmen… -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

2 1

Re: [alsa-devel] [PATCH 4/4] ASoC: fsl: make fsl_ssi driver compilable on ARM/IMX
by Timur Tabi 25 Feb '12

25 Feb '12

Sergei Shtylyov wrote: > These functions are not equivalent concerning endianness. You should > probably have used readl_be() instead, else it won't work on PPC anymore. readl_be() is not good enough, either. The in/out_be functions have specific syncronization instructions in them, so that the operations conform to the PowerPC instruction ordering architecture. You probably need to do something like this: #ifdef PPC #define read_ssi(x) in_be32(x) #else #if defined ARM #define read_ssi(x) readl(x) #endif -- Timur Tabi Linux kernel developer at Freescale

3 13

Re: [alsa-devel] [PATCH] sound/soc-core: Don't overwrite .poweroff in snd_soc_pm_ops
by Mark Brown 24 Feb '12

24 Feb '12

On Fri, Feb 24, 2012 at 04:25:49PM +0530, Viresh Kumar wrote: > SET_SYSTEM_SLEEP_PM_OPS writes .poweroff = *_resume once. Then we overwrite it > again explicitly as .poweroff = snd_soc_poweroff. Even though it works, as the Applied but please do try to use subject lines appropriate for the subsystem you're submitting against. I'd also suggest creating a mailing list for ST internal people if you need to CC so many people on your team, the vger lists have a fairly low limit on the number of CCs they will accept which I think is about twenty or so, you're using a fair proportion of that.

2 2

[alsa-devel] [Q] ASoC: What's the real meaning of SND_SOC_DAIFMT_NB_NF?
by javier Martin 24 Feb '12

24 Feb '12

Hi, I am trying to configure a DAI interface between a i.MX27 SSI and a wm8974 codec. The definition of SND_SOC_DAIFMT_NB_NF specifies 'normal' bit clock and frame for the specified format. http://lxr.linux.no/#linux+v3.2.7/include/sound/soc-dai.h#L48 As I wasn't sure about the meaning of 'normal' I've dug into the code of some platforms to see what they do. What I've found, however, is that it doesn't seem to be an agreement about it. Let me expose some examples: 1. Frame clock is active High: i.MX: http://lxr.linux.no/#linux+v3.2.7/sound/soc/imx/imx-ssi.c#L131 + MCIMX27RM.pdf (p 1606) (default) PXA: http://lxr.linux.no/#linux+v3.2.7/sound/soc/pxa/pxa-ssp.c#L498 pxa-ssp.c + PXA3xx_DM_Vol_IV.pdf (p 332) (not default) 2. Frame clock is active Low: OMAP: http://lxr.linux.no/#linux+v3.2.7/sound/soc/omap/omap-mcbsp.c#L478 + sprugn4h.pdf (p 3165) (not default) Some chips like s6000 seem to provide a more original use for this flag: http://lxr.linux.no/#linux+v3.2.7/sound/soc/s6000/s6000-i2s.c#L255 Similar behavior has been found for bit clock. ALso, it seems that there is no possibility to specify different configurations for RX and TX parts. -- Javier Martin Vista Silicon S.L. CDTUC - FASE C - Oficina S-345 Avda de los Castros s/n 39005- Santander. Cantabria. Spain +34 942 25 32 60 www.vista-silicon.com

3 9

[PATCH] Fix internal mic for Lenovo Ideapad U300s
by David Henningsson 24 Feb '12

24 Feb '12

The internal mic input is phase inverted on one channel. To avoid people in userspace summing the channels together and get zero result, mute the right channel and make sure they can't unmute it. BugLink: https://bugs.launchpad.net/bugs/903853 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_conexant.c | 41 +++++++++++++++++++++++++++++---------- 1 files changed, 30 insertions(+), 11 deletions(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 266e5a6..4a05817 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -140,6 +140,7 @@ struct conexant_spec { unsigned int asus:1; unsigned int pin_eapd_ctrls:1; unsigned int single_adc_amp:1; + unsigned int fixup_stereo_dmic:1; unsigned int adc_switching:1; @@ -4044,7 +4045,7 @@ static int cx_auto_init(struct hda_codec *codec) static int cx_auto_add_volume_idx(struct hda_codec *codec, const char *basename, const char *dir, int cidx, - hda_nid_t nid, int hda_dir, int amp_idx) + hda_nid_t nid, int hda_dir, int amp_idx, int chs) { static char name[32]; static struct snd_kcontrol_new knew[] = { @@ -4056,7 +4057,7 @@ static int cx_auto_add_volume_idx(struct hda_codec *codec, const char *basename, for (i = 0; i < 2; i++) { struct snd_kcontrol *kctl; - knew[i].private_value = HDA_COMPOSE_AMP_VAL(nid, 3, amp_idx, + knew[i].private_value = HDA_COMPOSE_AMP_VAL(nid, chs, amp_idx, hda_dir); knew[i].subdevice = HDA_SUBDEV_AMP_FLAG; knew[i].index = cidx; @@ -4074,7 +4075,7 @@ static int cx_auto_add_volume_idx(struct hda_codec *codec, const char *basename, } #define cx_auto_add_volume(codec, str, dir, cidx, nid, hda_dir) \ - cx_auto_add_volume_idx(codec, str, dir, cidx, nid, hda_dir, 0) + cx_auto_add_volume_idx(codec, str, dir, cidx, nid, hda_dir, 0, 3) #define cx_auto_add_pb_volume(codec, nid, str, idx) \ cx_auto_add_volume(codec, str, " Playback", idx, nid, HDA_OUTPUT) @@ -4152,14 +4153,22 @@ static int cx_auto_add_capture_volume(struct hda_codec *codec, hda_nid_t nid, int i; for (i = 0; i < spec->num_adc_nids; i++) { + int chs = 3; hda_nid_t adc_nid = spec->adc_nids[i]; int idx = get_input_connection(codec, adc_nid, nid); if (idx < 0) continue; if (spec->single_adc_amp) idx = 0; + + if (spec->fixup_stereo_dmic && (strcmp("Internal Mic", label) == 0)) { + /* Make left channel controllable and right channel muted */ + chs = 1; + snd_hda_codec_write_cache(codec, adc_nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, + AC_AMP_SET_RIGHT | AC_AMP_SET_INPUT | (idx << AC_AMP_SET_INDEX_SHIFT) | AC_AMP_MUTE); + } return cx_auto_add_volume_idx(codec, label, pfx, - cidx, adc_nid, HDA_INPUT, idx); + cidx, adc_nid, HDA_INPUT, idx, chs); } return 0; } @@ -4335,22 +4344,30 @@ static void apply_pincfg(struct hda_codec *codec, const struct cxt_pincfg *cfg) } -static void apply_pin_fixup(struct hda_codec *codec, +enum { + CXT_PINCFG_LENOVO_X200, + CXT_FIXUP_STEREO_DMIC, +}; + +static void apply_fixup(struct hda_codec *codec, const struct snd_pci_quirk *quirk, const struct cxt_pincfg **table) { + struct conexant_spec *spec = codec->spec; + quirk = snd_pci_quirk_lookup(codec->bus->pci, quirk); - if (quirk) { + if (quirk && table[quirk->value]) { snd_printdd(KERN_INFO "hda_codec: applying pincfg for %s\n", quirk->name); apply_pincfg(codec, table[quirk->value]); } + if (quirk->value == CXT_FIXUP_STEREO_DMIC) { + snd_printdd(KERN_INFO "hda_codec: applying internal mic workaround for %s\n", + quirk->name); + spec->fixup_stereo_dmic = 1; + } } -enum { - CXT_PINCFG_LENOVO_X200, -}; - static const struct cxt_pincfg cxt_pincfg_lenovo_x200[] = { { 0x16, 0x042140ff }, /* HP (seq# overridden) */ { 0x17, 0x21a11000 }, /* dock-mic */ @@ -4360,10 +4377,12 @@ static const struct cxt_pincfg cxt_pincfg_lenovo_x200[] = { static const struct cxt_pincfg *cxt_pincfg_tbl[] = { [CXT_PINCFG_LENOVO_X200] = cxt_pincfg_lenovo_x200, + [CXT_FIXUP_STEREO_DMIC] = NULL, }; static const struct snd_pci_quirk cxt_fixups[] = { SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo X200", CXT_PINCFG_LENOVO_X200), + SND_PCI_QUIRK(0x17aa, 0x3975, "Lenovo U300s", CXT_FIXUP_STEREO_DMIC), {} }; @@ -4387,7 +4406,7 @@ static int patch_conexant_auto(struct hda_codec *codec) break; } - apply_pin_fixup(codec, cxt_fixups, cxt_pincfg_tbl); + apply_fixup(codec, cxt_fixups, cxt_pincfg_tbl); err = cx_auto_search_adcs(codec); if (err < 0) -- 1.7.9 --------------080406020609030400030200--

1 0

Re: [alsa-devel] Problem with "set_period_size"
by Clemens Ladisch 24 Feb '12

24 Feb '12

Alicia Romero wrote: > I forgot to write that at the end I activate all the params using: > > /* Activate the parameters */ > result = snd_pcm_hw_params (stream->pb_pcm, params); > if (result < 0) { If set_period_size fails, params has not been changed. Regards, Clemens

1 0

[alsa-devel] [PATCH] hda - add id for Atom Cedar Trail HDMI codec
by Wu Fengguang 24 Feb '12

24 Feb '12

Signed-off-by: Wu Fengguang <fengguang.wu(a)intel.com> --- sound/pci/hda/patch_hdmi.c | 2 ++ 1 file changed, 2 insertions(+) --- sound.orig/sound/pci/hda/patch_hdmi.c 2012-02-23 14:51:19.562540743 +0800 +++ sound/sound/pci/hda/patch_hdmi.c 2012-02-23 15:05:51.774522844 +0800 @@ -1913,6 +1913,7 @@ static const struct hda_codec_preset snd { .id = 0x80862805, .name = "CougarPoint HDMI", .patch = patch_generic_hdmi }, { .id = 0x80862806, .name = "PantherPoint HDMI", .patch = patch_generic_hdmi }, { .id = 0x808629fb, .name = "Crestline HDMI", .patch = patch_generic_hdmi }, +{ .id = 0x80862880, .name = "CedarTrail HDMI", .patch = patch_generic_hdmi }, {} /* terminator */ }; @@ -1959,6 +1960,7 @@ MODULE_ALIAS("snd-hda-codec-id:80862804" MODULE_ALIAS("snd-hda-codec-id:80862805"); MODULE_ALIAS("snd-hda-codec-id:80862806"); MODULE_ALIAS("snd-hda-codec-id:808629fb"); +MODULE_ALIAS("snd-hda-codec-id:80862880"); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("HDMI HD-audio codec");

4 4

[alsa-devel] How to trigger a widget?
by Wang, Kuirong 24 Feb '12

24 Feb '12

As I saw many examples to use the snd_soc_dapm_enable_pin() to trigger a particular widget. But I saw the widget has to be like a speaker, mic, lineout, etc. Could I use the snd_soc_dapm_enable_pin() to trigger any type of widget I want? I tried to use it to trigger a supply widget, but I never see it works even I did snd_soc_dapm_sync() after the enable_pin. Could someone provide some inputs? Appreciated your time. Thanks, Kuirong Wang

3 5

[alsa-devel] [PATCH] ASoC: wm8962: Convert interrupt handler to direct regmap usage
by Mark Brown 23 Feb '12

23 Feb '12

Avoids potential locking issues with anything that needs the CODEC lock. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8962.c | 59 +++++++++++++++++++++++++++++--------------- 1 files changed, 39 insertions(+), 20 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 6ff7513..eb9da3a 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2981,54 +2981,73 @@ static void wm8962_mic_work(struct work_struct *work) static irqreturn_t wm8962_irq(int irq, void *data) { - struct snd_soc_codec *codec = data; - struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec); - int mask; - int active; - int reg; + struct device *dev = data; + struct wm8962_priv *wm8962 = dev_get_drvdata(dev); + unsigned int mask; + unsigned int active; + int reg, ret; + + ret = regmap_read(wm8962->regmap, WM8962_INTERRUPT_STATUS_2_MASK, + &mask); + if (ret != 0) { + dev_err(dev, "Failed to read interrupt mask: %d\n", + ret); + return IRQ_NONE; + } - mask = snd_soc_read(codec, WM8962_INTERRUPT_STATUS_2_MASK); + ret = regmap_read(wm8962->regmap, WM8962_INTERRUPT_STATUS_2, &active); + if (ret != 0) { + dev_err(dev, "Failed to read interrupt: %d\n", ret); + return IRQ_NONE; + } - active = snd_soc_read(codec, WM8962_INTERRUPT_STATUS_2); active &= ~mask; if (!active) return IRQ_NONE; /* Acknowledge the interrupts */ - snd_soc_write(codec, WM8962_INTERRUPT_STATUS_2, active); + ret = regmap_write(wm8962->regmap, WM8962_INTERRUPT_STATUS_2, active); + if (ret != 0) + dev_warn(dev, "Failed to ack interrupt: %d\n", ret); if (active & WM8962_FLL_LOCK_EINT) { - dev_dbg(codec->dev, "FLL locked\n"); + dev_dbg(dev, "FLL locked\n"); complete(&wm8962->fll_lock); } if (active & WM8962_FIFOS_ERR_EINT) - dev_err(codec->dev, "FIFO error\n"); + dev_err(dev, "FIFO error\n"); if (active & WM8962_TEMP_SHUT_EINT) { - dev_crit(codec->dev, "Thermal shutdown\n"); + dev_crit(dev, "Thermal shutdown\n"); - reg = snd_soc_read(codec, WM8962_THERMAL_SHUTDOWN_STATUS); + ret = regmap_read(wm8962->regmap, + WM8962_THERMAL_SHUTDOWN_STATUS, &reg); + if (ret != 0) { + dev_warn(dev, "Failed to read thermal status: %d\n", + ret); + reg = 0; + } if (reg & WM8962_TEMP_ERR_HP) - dev_crit(codec->dev, "Headphone thermal error\n"); + dev_crit(dev, "Headphone thermal error\n"); if (reg & WM8962_TEMP_WARN_HP) - dev_crit(codec->dev, "Headphone thermal warning\n"); + dev_crit(dev, "Headphone thermal warning\n"); if (reg & WM8962_TEMP_ERR_SPK) - dev_crit(codec->dev, "Speaker thermal error\n"); + dev_crit(dev, "Speaker thermal error\n"); if (reg & WM8962_TEMP_WARN_SPK) - dev_crit(codec->dev, "Speaker thermal warning\n"); + dev_crit(dev, "Speaker thermal warning\n"); } if (active & (WM8962_MICSCD_EINT | WM8962_MICD_EINT)) { - dev_dbg(codec->dev, "Microphone event detected\n"); + dev_dbg(dev, "Microphone event detected\n"); #ifndef CONFIG_SND_SOC_WM8962_MODULE - trace_snd_soc_jack_irq(dev_name(codec->dev)); + trace_snd_soc_jack_irq(dev_name(dev)); #endif - pm_wakeup_event(codec->dev, 300); + pm_wakeup_event(dev, 300); schedule_delayed_work(&wm8962->mic_work, msecs_to_jiffies(250)); @@ -3534,7 +3553,7 @@ static int wm8962_probe(struct snd_soc_codec *codec) ret = request_threaded_irq(wm8962->irq, NULL, handler, trigger | IRQF_ONESHOT, - "wm8962", codec); + "wm8962", codec->dev); if (ret != 0) { dev_err(codec->dev, "Failed to request IRQ %d: %d\n", wm8962->irq, ret); -- 1.7.9.1

1 0

[alsa-devel] [PATCH] ASoC: i.MX SSI: Fix DSP_A format.
by Javier Martin 23 Feb '12

23 Feb '12

According to i.MX27 Reference Manual (p 1593) TXBIT0 bit selects whether the most significant or the less significant part of the data word written to the FIFO is transmitted. As DSP_A is the same as DSP_B with a data offset of 1 bit, it doesn't make any sense to remove TXBIT0 bit here. Signed-off-by: Javier Martin <javier.martin(a)vista-silicon.com> --- sound/soc/imx/imx-ssi.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/imx/imx-ssi.c b/sound/soc/imx/imx-ssi.c index 01d1f74..b6adbed 100644 --- a/sound/soc/imx/imx-ssi.c +++ b/sound/soc/imx/imx-ssi.c @@ -112,7 +112,7 @@ static int imx_ssi_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) break; case SND_SOC_DAIFMT_DSP_A: /* data on rising edge of bclk, frame high 1clk before data */ - strcr |= SSI_STCR_TFSL | SSI_STCR_TEFS; + strcr |= SSI_STCR_TFSL | SSI_STCR_TXBIT0 | SSI_STCR_TEFS; break; } -- 1.7.0.4

3 2

Re: [alsa-devel] [PATCH] [trivial] sound: Fix typo in ctvmem.c
by Takashi Iwai 23 Feb '12

23 Feb '12

At Thu, 23 Feb 2012 23:23:06 +0900, Masanari Iida wrote: > > Correct spelling "virtural" to "virtual" in > sound/pci/ctxfi/ctvmem.c > > Signed-off-by: Masanari Iida <standby24x7(a)gmail.com> Thanks, applied now. (BTW, a typo in my address in your original post...) Takashi > --- > sound/pci/ctxfi/ctvmem.c | 2 +- > 1 files changed, 1 insertions(+), 1 deletions(-) > > diff --git a/sound/pci/ctxfi/ctvmem.c b/sound/pci/ctxfi/ctvmem.c > index b78f3fc..6109490 100644 > --- a/sound/pci/ctxfi/ctvmem.c > +++ b/sound/pci/ctxfi/ctvmem.c > @@ -36,7 +36,7 @@ get_vm_block(struct ct_vm *vm, unsigned int size) > > size = CT_PAGE_ALIGN(size); > if (size > vm->size) { > - printk(KERN_ERR "ctxfi: Fail! No sufficient device virtural " > + printk(KERN_ERR "ctxfi: Fail! No sufficient device virtual " > "memory space available!\n"); > return NULL; > } > -- > 1.7.6.5 > > -- > To unsubscribe from this list: send the line "unsubscribe linux-kernel" in > the body of a message to majordomo(a)vger.kernel.org > More majordomo info at http://vger.kernel.org/majordomo-info.html > Please read the FAQ at http://www.tux.org/lkml/ >

1 0

Re: [alsa-devel] multiple, non-physically accesible, HDMI devices listed for Intel IbexPeak ALC269VB
by David Henningsson 23 Feb '12

23 Feb '12

On 02/23/2012 01:01 AM, Andres Cimmarusti wrote: >> If the new two pins can be never used, i.e. physically unreachable, >> we may disable these pins by giving the proper default pin-config >> values. Usually it's a job of BIOS. But if BIOS doesn't do it, user >> need to do it manually. >> >> Build your kernel with CONFIG_SND_HDA_HWDEP=y, >> CONFIG_SND_HDA_RECONFIG=y, CONFIG_SND_HDA_PATCH_LOADER=y. >> I guess most of distro kernels are built with them. >> Then create a file containing below in /lib/firmware, such as, >> /lib/firmware/ibx-hdmi: >> >> ================================================================ >> [codec] >> 0x80862804 0x80860101 3 >> [pincfg] >> 0x04 0x411111f0 >> 0x06 0x411111f0 >> ================================================================ >> >> Now pass this file to "patch" module option for snd-hda-intel. >> For example, create a file in /etc/modprobe.d/, >> e.g. /etc/modprobe.d/50-hdmi.conf, containing the line >> >> options snd-hda-intel patch="ibx-hdmi" >> >> Then reload the driver or reboot. This will disable pins 0x04 and >> 0x06 so that only the pin 0x05 will be used. > > I've tested this workaround and it works well. I don't suppose this > could be added as a quirk to the kernel for this particular device? > (when and only if there's only one physically accessible HDMI > connector). Stephen, could you comment on this? Are we guaranteed a certain board configuration (i e which pins are physically connected or not) given a PCI SSID, or anything else that can be used as a key to a quirk table? I think I've asked you before and that the answer was "no", but my memory could be failing me. >> There are ways to configure pulseaudio to allow the user to select which >> PCM device to use on a given sound card. David Henningsson made this work >> for NVIDIA GPUs at least in Ubuntu, and I imagine the same technique >> could be applied to Intel devices too. > > Mmm.. just in Ubuntu? was this work submitted upstream? It appears > there are some related fixes shown in the Ubuntu pulseaudio changelog: > > http://changelogs.ubuntu.com/changelogs/pool/main/p/pulseaudio/pulseaudio_1… > > I found a thread related to this issue here: > http://www.mail-archive.com/pulseaudio-discuss@mail.0pointer.de/msg07433.ht… > Started by yourself Stephen Warren! but it doesn't seem like it got anywhere... > >> As Takashi mentions, from a kernel perspective, this isn't really a >> regression at all, but simply exposing all the features of the HW that >> were previously hidden. Without that change, others can't use some HW >> usefully at all. Unfortunately, pulseaudio makes some rather simplistic >> assumptions about how HW works by default, and can be confused by the >> additional features that are exposed. > > Agreed. But in the case of laptops, I don't think I've ever seen one > that actually has more than one physical connector. I'm a little > puzzled as to how all these outputs (in my case 3) make sense for my > hardware... Even if there is only one HDMI output, it's it not that uncommon, actually. Many DVI outputs can output HDMI audio as well (when connected through an HDMI monitor through DVI->HDMI adapter), and DisplayPort also has audio support. -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

2 1

[alsa-devel] [PATCH 1/8] ALSA: core: Constify the name in new kcontrols
by Mark Brown 23 Feb '12

23 Feb '12

We never modify it and this lets us use a const string as the name without warnings. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/control.h | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/include/sound/control.h b/include/sound/control.h index b2796e8..57815f6 100644 --- a/include/sound/control.h +++ b/include/sound/control.h @@ -40,7 +40,7 @@ struct snd_kcontrol_new { snd_ctl_elem_iface_t iface; /* interface identifier */ unsigned int device; /* device/client number */ unsigned int subdevice; /* subdevice (substream) number */ - unsigned char *name; /* ASCII name of item */ + const unsigned char *name; /* ASCII name of item */ unsigned int index; /* index of item */ unsigned int access; /* access rights */ unsigned int count; /* count of same elements */ -- 1.7.9.rc1

3 16

[alsa-devel] [PATCH] Wrong latency in pulseaudio plugin breaks Adobe Flash Player
by Philip Spencer 23 Feb '12

23 Feb '12

When Adobe Flash player is playing a live stream in "live" (no internal latency) mode[1], through alsa's pulseaudio plugin, the audio is a half-second out of sync with the video. This is with alsa-plugins version 1.0.25.[2] I see the problem is one already reported on the bug tracker several years ago (issue #3944). The plugin sets pulseaudio's target buffer length to match the IO buffer length, and sets the prebuffering attribute to match the IO buffer length less one period. In our case, the IO buffer is 500ms long, the IO period length is 20ms (for 16KHz Speex soud packets), and the application requests that playback start immediately (by setting the start_threshold software parameter very low). However, the plugin ignores start_threshold and sets the pulseaudio target buffer length to 500ms, meaning pulseaudio plays the sound with a half-second delay. The fix initially suggested in the bug report was to implement the start_threshold parameter and use it to lower the prebuffering attribute but this does not work: playback STARTS sooner, but when dropouts and underruns occur pulseaudio increases the latency to match the target buffer length. It is necessary to lower the target buffer length too. Below is a proposed patch (my apologies if I've missed some subtleties as I've never looked at the alsa source code before). The patch changes the plugin to do the following: -- decouple the length of the fake ring buffer seen by ALSA (which needs to match the IO buffer length) from the "target length" parameter sent to pulseaudio (which needs to reflect the desired latency), adding a new variable to the structure to store the former value. Currently they are forced to be the same value. -- when the start_threshold software parameter is seen, lower both the target length and prebuffering attributes to match it, except don't set them less than one IO period or longer than the IO buffer length -- when hardware parameters are seen, don't mess with the target length or prebuffering attributes if already set by software. It doesn't currently tell pulseaudio to change its parameters if the start_threshold is enountered after the stream is already started, but that could be added too if necessary with if (pcm->stream) pa_stream_set_buffer_attr( ... ) I hope this patch (or something like it) can get put into an alsa release since otherwise the Flash Player 2-way interaction app we're developing won't be usable by Linux users unless it runs in about 750ms time-delayed mode! Thank you, Philip [1] Normally Flash Player plays prerecorded streams with a slight internal buffering delay, and in this mode it requests the audio be buffered for 500ms in addition to its own internal delay, and it automatically delays video frames for 500ms beyond its own internal delay to compensate, so there are no out-of-sync problems with prerecorded streams or live streams played in regular buffered mode. [2] In 1.0.23 with low-sample-rate streams there was no sync problem but each sound sample was truncated, resulting in distorted and crackly sound; this problem has been fixed in 1.0.25 -- thanks! --------------------------------------------+------------------------------- Philip Spencer pspencer(a)fields.utoronto.ca | Director of Computing Services Room 336 (416)-348-9710 ext3036 | The Fields Institute for 222 College St, Toronto ON M5T 3J1 Canada | Research in Mathematical Sciences Signed-Of-By: Philip Spencer <pspencer(a)fields.utoronto.ca> --- alsa-plugins-1.0.25.orig/pulse/pcm_pulse.c 2012-01-25 02:57:07.000000000 -0500 +++ alsa-plugins-1.0.25/pulse/pcm_pulse.c 2012-02-15 16:23:08.000000000 -0500 @@ -49,6 +49,9 @@ pa_sample_spec ss; size_t frame_size; pa_buffer_attr buffer_attr; + uint32_t ring_buffer_length; + int tlength_set_by_sw; + } snd_pcm_pulse_t; static int check_stream(snd_pcm_pulse_t *pcm) @@ -93,12 +96,12 @@ size -= pcm->offset; /* Prevent accidental overrun of the fake ringbuffer */ - if (size > pcm->buffer_attr.tlength - pcm->frame_size) - size = pcm->buffer_attr.tlength - pcm->frame_size; + if (size > pcm->ring_buffer_length - pcm->frame_size) + size = pcm->ring_buffer_length - pcm->frame_size; if (size > pcm->last_size) { pcm->ptr += size - pcm->last_size; - pcm->ptr %= pcm->buffer_attr.tlength; + pcm->ptr %= pcm->ring_buffer_length; } pcm->last_size = size; @@ -830,12 +833,16 @@ pcm->ss.rate = io->rate; pcm->ss.channels = io->channels; + pcm->ring_buffer_length = + io->buffer_size * pcm->frame_size; pcm->buffer_attr.maxlength = 4 * 1024 * 1024; - pcm->buffer_attr.tlength = - io->buffer_size * pcm->frame_size; - pcm->buffer_attr.prebuf = - (io->buffer_size - io->period_size) * pcm->frame_size; + if (! pcm->tlength_set_by_sw) { + pcm->buffer_attr.tlength = + io->buffer_size * pcm->frame_size; + pcm->buffer_attr.prebuf = + (io->buffer_size - io->period_size) * pcm->frame_size; + } pcm->buffer_attr.minreq = io->period_size * pcm->frame_size; pcm->buffer_attr.fragsize = io->period_size * pcm->frame_size; @@ -845,6 +852,36 @@ return err; } +static int pulse_sw_params(snd_pcm_ioplug_t * io, + snd_pcm_sw_params_t * params) +{ + snd_pcm_pulse_t *pcm = io->private_data; + int err = 0; + + assert(pcm); + + if (!pcm->p || !pcm->p->mainloop) + return -EBADFD; + + pa_threaded_mainloop_lock(pcm->p->mainloop); + snd_pcm_uframes_t start_thresh; + err = snd_pcm_sw_params_get_start_threshold(params, &start_thresh); + if (! err) { + if (start_thresh < io->period_size) + start_thresh = io->period_size; + if (start_thresh > io->buffer_size) + start_thresh = io->buffer_size; + + pcm->buffer_attr.tlength = start_thresh * pcm->frame_size; + pcm->buffer_attr.prebuf = start_thresh * pcm->frame_size; + pcm->tlength_set_by_sw = 1; + } + + pa_threaded_mainloop_unlock(pcm->p->mainloop); + return err; + +} + static int pulse_close(snd_pcm_ioplug_t * io) { snd_pcm_pulse_t *pcm = io->private_data; @@ -911,6 +948,7 @@ .poll_revents = pulse_pcm_poll_revents, .prepare = pulse_prepare, .hw_params = pulse_hw_params, + .sw_params = pulse_sw_params, .close = pulse_close, .pause = pulse_pause }; @@ -1080,6 +1118,7 @@ pcm->io.mmap_rw = 0; pcm->io.callback = stream == SND_PCM_STREAM_PLAYBACK ? &pulse_playback_callback : &pulse_capture_callback; + pcm->tlength_set_by_sw = 0; pcm->io.private_data = pcm; err = snd_pcm_ioplug_create(&pcm->io, name, stream, mode);

3 10

[alsa-devel] Problem with "set_period_size"
by Alicia Romero 23 Feb '12

23 Feb '12

Hi list! I am new using an ALSA device and I have a problem when setting the period size. In my code I try to set the period size to 640 as follow: snd_pcm_hw_params_set_access (stream->pb_pcm, params, SND_PCM_ACCESS_RW_INTERLEAVED); /* Set format */ snd_pcm_hw_params_set_format (stream->pb_pcm, params, SND_PCM_FORMAT_S16_LE); /* Set number of channels */ snd_pcm_hw_params_set_channels (stream->pb_pcm, params, 2); /* Set clock rate */ rate =32000; snd_pcm_hw_params_set_rate_near (stream->pb_pcm, params, &rate, NULL); /* Set period size to samples_per_frame frames. */ stream->pb_frames = (snd_pcm_uframes_t) 640; snd_pcm_hw_params_set_period_size_near (stream->pb_pcm, params, &stream->pb_frames, NULL); /* Set the sound device buffer size and latency */ if (param->flags & PJMEDIA_AUD_DEV_CAP_OUTPUT_LATENCY) tmp_buf_size = (rate / 1000) * param->output_latency_ms; else tmp_buf_size = (rate / 1000) * PJMEDIA_SND_DEFAULT_PLAY_LATENCY; snd_pcm_hw_params_set_buffer_size_near (stream->pb_pcm, params, &tmp_buf_size); stream->param.output_latency_ms = tmp_buf_size / (rate / 1000); /* Activate the parameters */ snd_pcm_hw_params (stream->pb_pcm, params); The problem here is that the period size is not set to 640, but to 512. Otherwise if I use instead: stream->pb_frames = (snd_pcm_uframes_t) 640; snd_pcm_hw_params_set_period_size (stream->pb_pcm, params, &stream->pb_frames, NULL); the period size is set to 640 and the device is working correctly. Why is not set to 640 if I use the "*_near*" version? Thanks!!! :)

2 1

[alsa-devel] [PATCH] ASoC: dapm: Check for bias level when powering down
by Mark Brown 23 Feb '12

23 Feb '12

Recent enhancements in the bias management means that we might not be in standby when the CODEC is idle and can have active widgets without being in full power mode but the shutdown functionality assumes these things. Add checks for the bias level at each stage so that we don't do transitions other than the ON->PREPARE->STANDBY->OFF ones that the drivers are expecting. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-dapm.c | 12 +++++++++--- 1 files changed, 9 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 9029aad..3a5adab 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3237,9 +3237,13 @@ static void soc_dapm_shutdown_codec(struct snd_soc_dapm_context *dapm) * standby. */ if (powerdown) { - snd_soc_dapm_set_bias_level(dapm, SND_SOC_BIAS_PREPARE); + if (dapm->bias_level == SND_SOC_BIAS_ON) + snd_soc_dapm_set_bias_level(dapm, + SND_SOC_BIAS_PREPARE); dapm_seq_run(dapm, &down_list, 0, false); - snd_soc_dapm_set_bias_level(dapm, SND_SOC_BIAS_STANDBY); + if (dapm->bias_level == SND_SOC_BIAS_PREPARE) + snd_soc_dapm_set_bias_level(dapm, + SND_SOC_BIAS_STANDBY); } } @@ -3252,7 +3256,9 @@ void snd_soc_dapm_shutdown(struct snd_soc_card *card) list_for_each_entry(codec, &card->codec_dev_list, list) { soc_dapm_shutdown_codec(&codec->dapm); - snd_soc_dapm_set_bias_level(&codec->dapm, SND_SOC_BIAS_OFF); + if (codec->dapm.bias_level == SND_SOC_BIAS_STANDBY) + snd_soc_dapm_set_bias_level(&codec->dapm, + SND_SOC_BIAS_OFF); } } -- 1.7.9.1

2 1

Re: [alsa-devel] multiple, non-physically accesible, HDMI devices listed for Intel IbexPeak ALC269VB
by Takashi Iwai 23 Feb '12

23 Feb '12

At Wed, 22 Feb 2012 01:43:44 -0500, Andres Cimmarusti wrote: > > [1 <text/plain; UTF-8 (7bit)>] > Dear Mr. Warren, > > I recently upgraded my laptop to Debian testing (from Debian stable + > the longterm stable 3.0.x kernel). The newer kernel 3.2.x came with a > regression that git bisect has traced down to one of your commits in > the early 3.1.x kernel development stage (git bisect output and git > bisect log attached). > > Under kernel 3.0.x HDMI sound works out-of-the-box as tested with > pulse audio (choosing the option Digital > Stereo (HDMI) Output) and by the command: > > $ aplay -D plughw:0,3 /usr/share/sounds/alsa/Front_Center.wav > > Alsa's device list reveals: > > $ aplay -l > **** List of PLAYBACK Hardware Devices **** > card 0: Intel [HDA Intel], device 0: ALC269VB Analog [ALC269VB Analog] > Subdevices: 0/1 > Subdevice #0: subdevice #0 > card 0: Intel [HDA Intel], device 3: HDMI 0 [HDMI 0] > Subdevices: 1/1 > Subdevice #0: subdevice #0 > > Unfortunately with kernel 3.2.x and 3.1.x I get no sound out choosing the same > configuration in pulseaudio. Device is advertised correctly but > there's a bizarre multiplicity advertised: > > $ aplay -l > **** List of PLAYBACK Hardware Devices **** > card 0: Intel [HDA Intel], device 0: ALC269VB Analog [ALC269VB Analog] > Subdevices: 1/1 > Subdevice #0: subdevice #0 > card 0: Intel [HDA Intel], device 3: HDMI 0 [HDMI 0] > Subdevices: 1/1 > Subdevice #0: subdevice #0 > card 0: Intel [HDA Intel], device 7: HDMI 1 [HDMI 1] > Subdevices: 1/1 > Subdevice #0: subdevice #0 > card 0: Intel [HDA Intel], device 8: HDMI 2 [HDMI 2] > Subdevices: 1/1 > Subdevice #0: subdevice #0 > > Using aplay with device 3 says device is busy. Device 7 works > correctly (but is not available in pulseaudio unless forced by > default, which then renders internal speakers disabled) and device 8 > produces no sound out. > > This appears to be a regression in the kernel about this device, > advertising non-physically connected HDMI sound devices that cause > pulse audio to get > confused (Pulseaudio only seems to be able to handle one HDMI output > by default device 0,3). The biggest problem is that PA checks only the first HDMI device. In that sense, this is no regression in the kernel side, although I know it's annoying. If the new two pins can be never used, i.e. physically unreachable, we may disable these pins by giving the proper default pin-config values. Usually it's a job of BIOS. But if BIOS doesn't do it, user need to do it manually. Build your kernel with CONFIG_SND_HDA_HWDEP=y, CONFIG_SND_HDA_RECONFIG=y, CONFIG_SND_HDA_PATCH_LOADER=y. I guess most of distro kernels are built with them. Then create a file containing below in /lib/firmware, such as, /lib/firmware/ibx-hdmi: ================================================================ [codec] 0x80862804 0x80860101 3 [pincfg] 0x04 0x411111f0 0x06 0x411111f0 ================================================================ Now pass this file to "patch" module option for snd-hda-intel. For example, create a file in /etc/modprobe.d/, e.g. /etc/modprobe.d/50-hdmi.conf, containing the line options snd-hda-intel patch="ibx-hdmi" Then reload the driver or reboot. This will disable pins 0x04 and 0x06 so that only the pin 0x05 will be used. Takashi > > Output of the alsa-info script for kernel 3.0.20 and 3.2.4 is attached. > > Can you please advice how to fix this problem? I see there was a patch > about the same time meant to address the multiplicity of port > advertised and restrict to the ones being used, but this doesn't seem > to be working here (see > http://git.kernel.org/?p=linux/kernel/git/stable/linux-stable.git;a=commit;…) > > For reference, this is the original Debian bug report: > http://bugs.debian.org/660111 > > Best Regards, > > Andres Cimmarusti > [2 git_bisect_1st_bad.txt <text/plain; US-ASCII (base64)>] > 384a48d71520ca569a63f1e61e51a538bedb16df is the first bad commit > commit 384a48d71520ca569a63f1e61e51a538bedb16df > Author: Stephen Warren <swarren(a)nvidia.com> > Date: Wed Jun 1 11:14:21 2011 -0600 > > ALSA: hda: HDMI: Support codecs with fewer cvts than pins > > The general concept of this change is to create a PCM device for each > pin widget instead of each converter widget. Whenever a PCM is opened, > a converter is dynamically selected to drive that pin based on those > available for muxing into the pin. > > The one thing this model doesn't support is a single PCM/converter > sending audio to multiple pin widgets at once. > > Note that this means that a struct hda_pcm_stream's nid variable is > set to 0 except between a stream's open and cleanup calls. The dynamic > de-assignment of converters to PCMs occurs within cleanup, not close, > in order for it to co-incide with when controller stream IDs are > cleaned up from converters. > > While the PCM for a pin is not open, the pin is disabled (its widget > control's PIN_OUT bit is cleared) so that if the currently routed > converter is used to drive a different PCM/pin, that audio does not > leak out over a disabled pin. > > We use the recently added SPDIF virtualization feature in order to > create SPDIF controls for each pin widget instead of each converter > widget, so that state is specific to a PCM. > > In order to support this, a number of more mechanical changes are made: > > * s/nid/pin_nid/ or s/nid/cvt_nid/ in many places in order to make it > clear exactly what the code is dealing with. > > * We now have per_pin and per_cvt arrays in hdmi_spec to store relevant > data. In particular, we store a converter's capabilities in the per_cvt > entry, rather than relying on a combination of codec_pcm_pars and > the struct hda_pcm_stream. > > * ELD-related workarounds were removed from hdmi_channel_allocation > into hdmi_instrinsic in order to simplifiy infoframe calculations and > remove HW dependencies. > > * Various functions only apply to a single pin, since there is now > only 1 pin per PCM. For example, hdmi_setup_infoframe, > hdmi_setup_stream. > > * hdmi_add_pin and hdmi_add_cvt are more oriented at pure codec parsing > and data retrieval, rather than determining which pins/converters > are to be used for creating PCMs. > > This is quite a large change; it may be appropriate to simply read the > result of the patch rather than the diffs. Some small parts of the change > might be separable into different patches, but I think the bulk of the > change will probably always be one large patch. Hopefully the change > isn't too opaque! > > This has been tested on: > > * NVIDIA GeForce 400 series discrete graphics card. This model has the > classical 1:1:1 codec:converter:pcm widget model. Tested stereo PCM > audio to a PC monitor that supports audio. > > * NVIDIA GeForce 520 discrete graphics card. This model is the new > 1 codec n converters m pins m>n model. Tested stereo PCM audio to a > PC monitor that supports audio. > > * NVIDIA GeForce 400 series laptop graphics chip. This model has the > classical 1:1:1 codec:converter:pcm widget model. Tested stereo PCM, > multi-channel PCM, and AC3 pass-through to an AV receiver. > > * Intel Ibex Peak laptop. This model is the new 1 codec n converters m > pins m>n model. Tested stereo PCM, multi-channel PCM, and AC3 pass- > through to an AV receiver. > > Note that I'm not familiar at all with AC3 pass-through. Hence, I may > not have covered all possible mechanisms that are applicable here. I do > know that my receiver definitely received AC3, not decoded PCM. I tested > with mplayer's "-afm hwac3" and/or "-af lavcac3enc" options, and alsa a > WAV file that I believe has AC3 content rather than PCM. > > I also tested: > * Play a stream > * Mute while playing > * Stop stream > * Play some other streams to re-assign the converter to a different > pin, PCM, set of SPDIF controls, ... hence hopefully triggering > cleanup for the original PCM. > * Unmute original stream while not playing > * Play a stream on the original pin/PCM. > > This was to test SPDIF control virtualization. > > Signed-off-by: Stephen Warren <swarren(a)nvidia.com> > Signed-off-by: Takashi Iwai <tiwai(a)suse.de> > > :040000 040000 894370c6534b1bf03df9a8a8c7d85c2eeffc7555 98cb8a73a0ed46f034e25bd35002930bc22376ef M sound > [3 git_bisect_log.txt <text/plain; US-ASCII (base64)>] > git bisect log > git bisect start > # bad: [76531d4166fb620375ff3c1ac24753265216d579] Merge branch 'topic/hda' into for-linus > git bisect bad 76531d4166fb620375ff3c1ac24753265216d579 > # good: [7d339ae99758bc21033d4a19bcd4f7b55f96e24e] Merge branch 'topic/misc' into for-linus > git bisect good 7d339ae99758bc21033d4a19bcd4f7b55f96e24e > # bad: [39fa84e94a7df64a6ba27669ef98b51994fb6894] ALSA: hda - Simplify EAPD control in patch_realtek.c > git bisect bad 39fa84e94a7df64a6ba27669ef98b51994fb6894 > # bad: [096a885494f6b89a9962c6faf18e1c6092e7919c] ALSA: hda - Initialize input-path dynamically in patch_via.c > git bisect bad 096a885494f6b89a9962c6faf18e1c6092e7919c > # bad: [e06e5a297474c8027beffe10541981845ca0c98b] ALSA: hda - Defer mixer element creation to the right time in patch_via.c > git bisect bad e06e5a297474c8027beffe10541981845ca0c98b > # bad: [b4a655e81d4d1d12abc92d29dfb7550e66a08799] ALSA: hda - Judge playback stream from stream id in azx_via_get_position() > git bisect bad b4a655e81d4d1d12abc92d29dfb7550e66a08799 > # good: [3aaf898025b1f75f30457e00e890c9f7c43567ab] ALSA: hda: Separate generic and non-generic implementations > git bisect good 3aaf898025b1f75f30457e00e890c9f7c43567ab > # bad: [384a48d71520ca569a63f1e61e51a538bedb16df] ALSA: hda: HDMI: Support codecs with fewer cvts than pins > git bisect bad 384a48d71520ca569a63f1e61e51a538bedb16df > # good: [2def8172c6611f2577260287ebf5dd3b63f7ef55] ALSA: hda: hdmi_eld_update_pcm_info: update a stream in place > git bisect good 2def8172c6611f2577260287ebf5dd3b63f7ef55 > [4 alsa-info_3.0.20.txt <text/plain; US-ASCII (base64)>] > !!################################ > !!ALSA Information Script v 0.4.60 > !!################################ > > !!Script ran on: Thu Feb 16 05:24:37 UTC 2012 > > > !!Linux Distribution > !!------------------ > > Debian GNU/Linux wheezy/sid \n \l > > > !!DMI Information > !!--------------- > > Manufacturer: To be filled by O.E.M. > Product Name: To be filled by O.E.M. > Product Version: AC5G3N-S8_F6 > > > !!Kernel Information > !!------------------ > > Kernel release: 3.0.20-c2d.1-amd64 > Operating System: GNU/Linux > Architecture: x86_64 > Processor: unknown > SMP Enabled: Yes > > > !!ALSA Version > !!------------ > > Driver version: 1.0.24 > Library version: 1.0.24.1 > Utilities version: 1.0.24.2 > > > !!Loaded ALSA modules > !!------------------- > > snd_hda_intel > > > !!Sound Servers on this system > !!---------------------------- > > Pulseaudio: > Installed - Yes (/usr/bin/pulseaudio) > Running - Yes > > ESound Daemon: > Installed - Yes (/usr/bin/esd) > Running - No > > > !!Soundcards recognised by ALSA > !!----------------------------- > > 0 [Intel ]: HDA-Intel - HDA Intel > HDA Intel at 0xfe500000 irq 46 > > > !!PCI Soundcards installed in the system > !!-------------------------------------- > > 00:1b.0 Audio device: Intel Corporation 5 Series/3400 Series Chipset > High Definition Audio (rev 06) > > > !!Advanced information - PCI Vendor/Device/Subsystem ID's > !!-------------------------------------------------------- > > 00:1b.0 0403: 8086:3b56 (rev 06) > Subsystem: 1458:a950 > > > !!Modprobe options (Sound related) > !!-------------------------------- > > snd-atiixp-modem: index=-2 > snd-intel8x0m: index=-2 > snd-via82xx-modem: index=-2 > snd-pcsp: index=-2 > snd-usb-audio: index=-2 > > > !!Loaded sound module options > !!-------------------------- > > !!Module: snd_hda_intel > bdl_pos_adj : 1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 > beep_mode : 1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1 > enable : Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y > enable_msi : -1 > id : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) > index : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 > model : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) > patch : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) > position_fix : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 > power_save : 0 > power_save_controller : Y > probe_mask : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 > probe_only : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 > single_cmd : N > > > !!HDA-Intel Codec information > !!--------------------------- > --startcollapse-- > > Codec: Realtek ALC269VB > Address: 0 > AFG Function Id: 0x1 (unsol 1) > Vendor Id: 0x10ec0269 > Subsystem Id: 0x1458a950 > Revision Id: 0x100100 > No Modem Function Group found > Default PCM: > rates [0x560]: 44100 48000 96000 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Default Amp-In caps: N/A > Default Amp-Out caps: N/A > GPIO: io=2, o=0, i=0, unsolicited=1, wake=0 > IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 > IO[1]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 > Node 0x02 [Audio Output] wcaps 0x1d: Stereo Amp-Out > Control: name="Speaker Playback Volume", index=0, device=0 > ControlAmp: chs=3, dir=Out, idx=0, ofs=0 > Amp-Out caps: ofs=0x57, nsteps=0x57, stepsize=0x02, mute=0 > Amp-Out vals: [0x50 0x50] > Converter: stream=5, channel=0 > PCM: > rates [0x560]: 44100 48000 96000 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Node 0x03 [Audio Output] wcaps 0x1d: Stereo Amp-Out > Control: name="Front Playback Volume", index=0, device=0 > ControlAmp: chs=3, dir=Out, idx=0, ofs=0 > Device: name="ALC269VB Analog", type="Audio", device=0 > Amp-Out caps: ofs=0x57, nsteps=0x57, stepsize=0x02, mute=0 > Amp-Out vals: [0x50 0x50] > Converter: stream=5, channel=0 > PCM: > rates [0x560]: 44100 48000 96000 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Node 0x04 [Vendor Defined Widget] wcaps 0xf00000: Mono > Node 0x05 [Vendor Defined Widget] wcaps 0xf00000: Mono > Node 0x06 [Audio Output] wcaps 0x211: Stereo Digital > Converter: stream=0, channel=0 > Digital: > Digital category: 0x0 > PCM: > rates [0x5e0]: 44100 48000 88200 96000 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Node 0x07 [Vendor Defined Widget] wcaps 0xf00000: Mono > Node 0x08 [Audio Input] wcaps 0x10011b: Stereo Amp-In > Amp-In caps: ofs=0x0b, nsteps=0x1f, stepsize=0x05, mute=1 > Amp-In vals: [0x8b 0x8b] > Converter: stream=0, channel=0 > SDI-Select: 0 > PCM: > rates [0x560]: 44100 48000 96000 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Connection: 1 > 0x23 > Node 0x09 [Audio Input] wcaps 0x10011b: Stereo Amp-In > Control: name="Capture Switch", index=0, device=0 > Control: name="Capture Volume", index=0, device=0 > Device: name="ALC269VB Analog", type="Audio", device=0 > Amp-In caps: ofs=0x0b, nsteps=0x1f, stepsize=0x05, mute=1 > Amp-In vals: [0x8f 0x8f] > Converter: stream=1, channel=0 > SDI-Select: 0 > PCM: > rates [0x560]: 44100 48000 96000 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Connection: 1 > 0x22 > Node 0x0a [Vendor Defined Widget] wcaps 0xf00000: Mono > Node 0x0b [Audio Mixer] wcaps 0x20010b: Stereo Amp-In > Control: name="Beep Playback Volume", index=0, device=0 > ControlAmp: chs=3, dir=In, idx=4, ofs=0 > Control: name="Beep Playback Switch", index=0, device=0 > ControlAmp: chs=3, dir=In, idx=4, ofs=0 > Amp-In caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 > Amp-In vals: [0x97 0x97] [0x97 0x97] [0x97 0x97] [0x97 0x97] [0x97 0x97] > Connection: 5 > 0x18 0x19 0x1a 0x1b 0x1d > Node 0x0c [Audio Mixer] wcaps 0x20010b: Stereo Amp-In > Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-In vals: [0x00 0x00] [0x00 0x00] > Connection: 2 > 0x02 0x0b > Node 0x0d [Audio Mixer] wcaps 0x20010b: Stereo Amp-In > Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-In vals: [0x00 0x00] [0x00 0x00] > Connection: 2 > 0x03 0x0b > Node 0x0e [Vendor Defined Widget] wcaps 0xf00000: Mono > Node 0x0f [Audio Mixer] wcaps 0x20010a: Mono Amp-In > Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-In vals: [0x00] [0x00] > Connection: 2 > 0x02 0x0b > Node 0x10 [Vendor Defined Widget] wcaps 0xf00000: Mono > Node 0x11 [Vendor Defined Widget] wcaps 0xf00000: Mono > Node 0x12 [Pin Complex] wcaps 0x40000b: Stereo Amp-In > Control: name="Internal Mic Boost Volume", index=0, device=0 > ControlAmp: chs=3, dir=In, idx=0, ofs=0 > Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x2f, mute=0 > Amp-In vals: [0x03 0x03] > Pincap 0x00000020: IN > Pin Default 0x99a30940: [Fixed] Mic at Int ATAPI > Conn = ATAPI, Color = Unknown > DefAssociation = 0x4, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x20: IN > Node 0x13 [Vendor Defined Widget] wcaps 0xf00000: Mono > Node 0x14 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out > Control: name="Speaker Playback Switch", index=0, device=0 > ControlAmp: chs=3, dir=Out, idx=0, ofs=0 > Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-Out vals: [0x00 0x00] > Pincap 0x00010014: OUT EAPD Detect > EAPD 0x2: EAPD > Pin Default 0x99130110: [Fixed] Speaker at Int ATAPI > Conn = ATAPI, Color = Unknown > DefAssociation = 0x1, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x40: OUT > Unsolicited: tag=00, enabled=0 > Connection: 2 > 0x0c* 0x0d > Node 0x15 [Vendor Defined Widget] wcaps 0xf00000: Mono > Node 0x16 [Vendor Defined Widget] wcaps 0xf00000: Mono > Node 0x17 [Pin Complex] wcaps 0x40010c: Mono Amp-Out > Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-Out vals: [0x80] > Pincap 0x00000010: OUT > Pin Default 0x411111f0: [N/A] Speaker at Ext Rear > Conn = 1/8, Color = Black > DefAssociation = 0xf, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x00: > Connection: 1 > 0x0f > Node 0x18 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out > Control: name="Rear Mic Boost Volume", index=0, device=0 > ControlAmp: chs=3, dir=In, idx=0, ofs=0 > Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x2f, mute=0 > Amp-In vals: [0x02 0x02] > Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-Out vals: [0x80 0x80] > Pincap 0x00001734: IN OUT Detect > Vref caps: HIZ 50 GRD 80 > Pin Default 0x01a11830: [Jack] Mic at Ext Rear > Conn = 1/8, Color = Black > DefAssociation = 0x3, Sequence = 0x0 > Pin-ctls: 0x24: IN VREF_80 > Unsolicited: tag=00, enabled=0 > Connection: 1 > 0x0d > Node 0x19 [Pin Complex] wcaps 0x40008b: Stereo Amp-In > Control: name="Front Mic Boost Volume", index=0, device=0 > ControlAmp: chs=3, dir=In, idx=0, ofs=0 > Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x2f, mute=0 > Amp-In vals: [0x00 0x00] > Pincap 0x00001724: IN Detect > Vref caps: HIZ 50 GRD 80 > Pin Default 0x02a1183f: [Jack] Mic at Ext Front > Conn = 1/8, Color = Black > DefAssociation = 0x3, Sequence = 0xf > Pin-ctls: 0x24: IN VREF_80 > Unsolicited: tag=00, enabled=0 > Node 0x1a [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out > Control: name="Front Playback Switch", index=0, device=0 > ControlAmp: chs=3, dir=Out, idx=0, ofs=0 > Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x2f, mute=0 > Amp-In vals: [0x00 0x00] > Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-Out vals: [0x00 0x00] > Pincap 0x0000003c: IN OUT HP Detect > Pin Default 0x02211020: [Jack] HP Out at Ext Front > Conn = 1/8, Color = Black > DefAssociation = 0x2, Sequence = 0x0 > Pin-ctls: 0x40: OUT > Unsolicited: tag=01, enabled=1 > Connection: 2 > 0x0c 0x0d* > Node 0x1b [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out > Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x2f, mute=0 > Amp-In vals: [0x00 0x00] > Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-Out vals: [0x80 0x80] > Pincap 0x00000034: IN OUT Detect > Pin Default 0x411111f0: [N/A] Speaker at Ext Rear > Conn = 1/8, Color = Black > DefAssociation = 0xf, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x20: IN > Unsolicited: tag=00, enabled=0 > Connection: 2 > 0x0c* 0x0d > Node 0x1c [Vendor Defined Widget] wcaps 0xf00000: Mono > Node 0x1d [Pin Complex] wcaps 0x400000: Mono > Pincap 0x00000020: IN > Pin Default 0x598301f0: [N/A] Line In at Int ATAPI > Conn = ATAPI, Color = Unknown > DefAssociation = 0xf, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x20: IN > Node 0x1e [Pin Complex] wcaps 0x400381: Stereo Digital > Pincap 0x00000014: OUT Detect > Pin Default 0x411111f0: [N/A] Speaker at Ext Rear > Conn = 1/8, Color = Black > DefAssociation = 0xf, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x40: OUT > Unsolicited: tag=00, enabled=0 > Connection: 1 > 0x06 > Node 0x1f [Vendor Defined Widget] wcaps 0xf00000: Mono > Node 0x20 [Vendor Defined Widget] wcaps 0xf00040: Mono > Processing caps: benign=0, ncoeff=25 > Node 0x21 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out > Control: name="Headphone Playback Switch", index=0, device=0 > ControlAmp: chs=3, dir=Out, idx=0, ofs=0 > Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-Out vals: [0x00 0x00] > Pincap 0x0000001c: OUT HP Detect > Pin Default 0x0121101f: [Jack] HP Out at Ext Rear > Conn = 1/8, Color = Black > DefAssociation = 0x1, Sequence = 0xf > Pin-ctls: 0xc0: OUT HP > Unsolicited: tag=04, enabled=1 > Connection: 2 > 0x0c 0x0d* > Node 0x22 [Audio Selector] wcaps 0x30010b: Stereo Amp-In > Control: name="Input Source", index=0, device=0 > Amp-In caps: N/A > Amp-In vals: [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 > 0x00] [0x00 0x00] [0x00 0x00] > Connection: 7 > 0x18* 0x19 0x1a 0x1b 0x1d 0x0b 0x12 > Node 0x23 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In > Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-In vals: [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 > 0x80] [0x80 0x80] > Connection: 6 > 0x18 0x19 0x1a 0x1b 0x1d 0x0b > Codec: Intel IbexPeak HDMI > Address: 3 > AFG Function Id: 0x1 (unsol 0) > Vendor Id: 0x80862804 > Subsystem Id: 0x80860101 > Revision Id: 0x100000 > No Modem Function Group found > Default PCM: > rates [0x0]: > bits [0x0]: > formats [0x0]: > Default Amp-In caps: N/A > Default Amp-Out caps: N/A > GPIO: io=0, o=0, i=0, unsolicited=0, wake=0 > Node 0x02 [Audio Output] wcaps 0x6611: 8-Channels Digital > Control: name="IEC958 Playback Con Mask", index=0, device=0 > Control: name="IEC958 Playback Pro Mask", index=0, device=0 > Control: name="IEC958 Playback Default", index=0, device=0 > Control: name="IEC958 Playback Switch", index=0, device=0 > Device: name="HDMI 0", type="HDMI", device=3 > Converter: stream=8, channel=0 > Digital: Enabled > Digital category: 0x0 > PCM: > rates [0x7f0]: 32000 44100 48000 88200 96000 176400 192000 > bits [0x1e]: 16 20 24 32 > formats [0x5]: PCM AC3 > Power states: D0 D3 > Power: setting=D0, actual=D0 > Node 0x03 [Audio Output] wcaps 0x6611: 8-Channels Digital > Converter: stream=0, channel=0 > Digital: Enabled > Digital category: 0x0 > PCM: > rates [0x7f0]: 32000 44100 48000 88200 96000 176400 192000 > bits [0x1e]: 16 20 24 32 > formats [0x5]: PCM AC3 > Power states: D0 D3 > Power: setting=D0, actual=D0 > Node 0x04 [Pin Complex] wcaps 0x40778d: 8-Channels Digital Amp-Out CP > Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-Out vals: [0x00 0x00] > Pincap 0x09000094: OUT Detect HBR HDMI DP > Pin Default 0x18560010: [Jack] Digital Out at Int HDMI > Conn = Digital, Color = Unknown > DefAssociation = 0x1, Sequence = 0x0 > Pin-ctls: 0x40: OUT > Unsolicited: tag=04, enabled=1 > Power states: D0 D3 > Power: setting=D0, actual=D0 > Connection: 2 > 0x02* 0x03 > Node 0x05 [Pin Complex] wcaps 0x40778d: 8-Channels Digital Amp-Out CP > Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-Out vals: [0x00 0x00] > Pincap 0x09000094: OUT Detect HBR HDMI DP > Pin Default 0x18560010: [Jack] Digital Out at Int HDMI > Conn = Digital, Color = Unknown > DefAssociation = 0x1, Sequence = 0x0 > Pin-ctls: 0x40: OUT > Unsolicited: tag=05, enabled=1 > Power states: D0 D3 > Power: setting=D0, actual=D0 > Connection: 2 > 0x02* 0x03 > Node 0x06 [Pin Complex] wcaps 0x40778d: 8-Channels Digital Amp-Out CP > Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-Out vals: [0x00 0x00] > Pincap 0x09000094: OUT Detect HBR HDMI DP > Pin Default 0x18560010: [Jack] Digital Out at Int HDMI > Conn = Digital, Color = Unknown > DefAssociation = 0x1, Sequence = 0x0 > Pin-ctls: 0x40: OUT > Unsolicited: tag=06, enabled=1 > Power states: D0 D3 > Power: setting=D0, actual=D0 > Connection: 2 > 0x02* 0x03 > Node 0x07 [Vendor Defined Widget] wcaps 0xf00000: Mono > --endcollapse-- > > [5 alsa-info_3.2.4.txt <text/plain; US-ASCII (base64)>] > !!################################ > !!ALSA Information Script v 0.4.60 > !!################################ > > !!Script ran on: Thu Feb 16 05:29:09 UTC 2012 > > > !!Linux Distribution > !!------------------ > > Debian GNU/Linux wheezy/sid \n \l > > > !!DMI Information > !!--------------- > > Manufacturer: To be filled by O.E.M. > Product Name: To be filled by O.E.M. > Product Version: AC5G3N-S8_F6 > > > !!Kernel Information > !!------------------ > > Kernel release: 3.2.0-1-amd64 > Operating System: GNU/Linux > Architecture: x86_64 > Processor: unknown > SMP Enabled: Yes > > > !!ALSA Version > !!------------ > > Driver version: 1.0.24 > Library version: 1.0.24.1 > Utilities version: 1.0.24.2 > > > !!Loaded ALSA modules > !!------------------- > > snd_hda_intel > > > !!Sound Servers on this system > !!---------------------------- > > Pulseaudio: > Installed - Yes (/usr/bin/pulseaudio) > Running - Yes > > ESound Daemon: > Installed - Yes (/usr/bin/esd) > Running - No > > > !!Soundcards recognised by ALSA > !!----------------------------- > > 0 [Intel ]: HDA-Intel - HDA Intel > HDA Intel at 0xfe500000 irq 46 > > > !!PCI Soundcards installed in the system > !!-------------------------------------- > > 00:1b.0 Audio device: Intel Corporation 5 Series/3400 Series Chipset > High Definition Audio (rev 06) > > > !!Advanced information - PCI Vendor/Device/Subsystem ID's > !!-------------------------------------------------------- > > 00:1b.0 0403: 8086:3b56 (rev 06) > Subsystem: 1458:a950 > > > !!Modprobe options (Sound related) > !!-------------------------------- > > snd-atiixp-modem: index=-2 > snd-intel8x0m: index=-2 > snd-via82xx-modem: index=-2 > snd-pcsp: index=-2 > snd-usb-audio: index=-2 > > > !!Loaded sound module options > !!-------------------------- > > !!Module: snd_hda_intel > align_buffer_size : Y > bdl_pos_adj : 1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 > beep_mode : 1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1 > enable : Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y > enable_msi : -1 > id : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) > index : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 > model : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) > patch : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) > position_fix : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 > power_save : 0 > power_save_controller : Y > probe_mask : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 > probe_only : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 > single_cmd : N > snoop : Y > > > !!HDA-Intel Codec information > !!--------------------------- > --startcollapse-- > > Codec: Realtek ALC269VB > Address: 0 > AFG Function Id: 0x1 (unsol 1) > Vendor Id: 0x10ec0269 > Subsystem Id: 0x1458a950 > Revision Id: 0x100100 > No Modem Function Group found > Default PCM: > rates [0x560]: 44100 48000 96000 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Default Amp-In caps: N/A > Default Amp-Out caps: N/A > GPIO: io=2, o=0, i=0, unsolicited=1, wake=0 > IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 > IO[1]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 > Node 0x02 [Audio Output] wcaps 0x1d: Stereo Amp-Out > Control: name="PCM Playback Volume", index=0, device=0 > ControlAmp: chs=3, dir=Out, idx=0, ofs=0 > Device: name="ALC269VB Analog", type="Audio", device=0 > Amp-Out caps: ofs=0x57, nsteps=0x57, stepsize=0x02, mute=0 > Amp-Out vals: [0x50 0x50] > Converter: stream=8, channel=0 > PCM: > rates [0x560]: 44100 48000 96000 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Node 0x03 [Audio Output] wcaps 0x1d: Stereo Amp-Out > Control: name="Headphone Playback Volume", index=0, device=0 > ControlAmp: chs=3, dir=Out, idx=0, ofs=0 > Amp-Out caps: ofs=0x57, nsteps=0x57, stepsize=0x02, mute=0 > Amp-Out vals: [0x50 0x50] > Converter: stream=8, channel=0 > PCM: > rates [0x560]: 44100 48000 96000 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Node 0x04 [Vendor Defined Widget] wcaps 0xf00000: Mono > Node 0x05 [Vendor Defined Widget] wcaps 0xf00000: Mono > Node 0x06 [Audio Output] wcaps 0x211: Stereo Digital > Converter: stream=0, channel=0 > Digital: > Digital category: 0x0 > PCM: > rates [0x5e0]: 44100 48000 88200 96000 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Node 0x07 [Vendor Defined Widget] wcaps 0xf00000: Mono > Node 0x08 [Audio Input] wcaps 0x10011b: Stereo Amp-In > Amp-In caps: ofs=0x0b, nsteps=0x1f, stepsize=0x05, mute=1 > Amp-In vals: [0x8b 0x8b] > Converter: stream=0, channel=0 > SDI-Select: 0 > PCM: > rates [0x560]: 44100 48000 96000 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Connection: 1 > 0x23 > Node 0x09 [Audio Input] wcaps 0x10011b: Stereo Amp-In > Control: name="Capture Switch", index=0, device=0 > Control: name="Capture Volume", index=0, device=0 > Device: name="ALC269VB Analog", type="Audio", device=0 > Amp-In caps: ofs=0x0b, nsteps=0x1f, stepsize=0x05, mute=1 > Amp-In vals: [0x8f 0x8f] > Converter: stream=4, channel=0 > SDI-Select: 0 > PCM: > rates [0x560]: 44100 48000 96000 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Connection: 1 > 0x22 > Node 0x0a [Vendor Defined Widget] wcaps 0xf00000: Mono > Node 0x0b [Audio Mixer] wcaps 0x20010b: Stereo Amp-In > Control: name="Rear Mic Playback Volume", index=0, device=0 > ControlAmp: chs=3, dir=In, idx=0, ofs=0 > Control: name="Rear Mic Playback Switch", index=0, device=0 > ControlAmp: chs=3, dir=In, idx=0, ofs=0 > Control: name="Front Mic Playback Volume", index=0, device=0 > ControlAmp: chs=3, dir=In, idx=1, ofs=0 > Control: name="Front Mic Playback Switch", index=0, device=0 > ControlAmp: chs=3, dir=In, idx=1, ofs=0 > Control: name="Beep Playback Volume", index=0, device=0 > ControlAmp: chs=3, dir=In, idx=4, ofs=0 > Control: name="Beep Playback Switch", index=0, device=0 > ControlAmp: chs=3, dir=In, idx=4, ofs=0 > Amp-In caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 > Amp-In vals: [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x97 0x97] > Connection: 5 > 0x18 0x19 0x1a 0x1b 0x1d > Node 0x0c [Audio Mixer] wcaps 0x20010b: Stereo Amp-In > Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-In vals: [0x00 0x00] [0x00 0x00] > Connection: 2 > 0x02 0x0b > Node 0x0d [Audio Mixer] wcaps 0x20010b: Stereo Amp-In > Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-In vals: [0x00 0x00] [0x00 0x00] > Connection: 2 > 0x03 0x0b > Node 0x0e [Vendor Defined Widget] wcaps 0xf00000: Mono > Node 0x0f [Audio Mixer] wcaps 0x20010a: Mono Amp-In > Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-In vals: [0x00] [0x80] > Connection: 2 > 0x02 0x0b > Node 0x10 [Vendor Defined Widget] wcaps 0xf00000: Mono > Node 0x11 [Vendor Defined Widget] wcaps 0xf00000: Mono > Node 0x12 [Pin Complex] wcaps 0x40000b: Stereo Amp-In > Control: name="Internal Mic Boost Volume", index=0, device=0 > ControlAmp: chs=3, dir=In, idx=0, ofs=0 > Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x2f, mute=0 > Amp-In vals: [0x03 0x03] > Pincap 0x00000020: IN > Pin Default 0x99a30940: [Fixed] Mic at Int ATAPI > Conn = ATAPI, Color = Unknown > DefAssociation = 0x4, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x20: IN > Node 0x13 [Vendor Defined Widget] wcaps 0xf00000: Mono > Node 0x14 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out > Control: name="Speaker Playback Switch", index=0, device=0 > ControlAmp: chs=3, dir=Out, idx=0, ofs=0 > Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-Out vals: [0x00 0x00] > Pincap 0x00010014: OUT EAPD Detect > EAPD 0x2: EAPD > Pin Default 0x99130110: [Fixed] Speaker at Int ATAPI > Conn = ATAPI, Color = Unknown > DefAssociation = 0x1, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x40: OUT > Unsolicited: tag=00, enabled=0 > Connection: 2 > 0x0c* 0x0d > Node 0x15 [Vendor Defined Widget] wcaps 0xf00000: Mono > Node 0x16 [Vendor Defined Widget] wcaps 0xf00000: Mono > Node 0x17 [Pin Complex] wcaps 0x40010c: Mono Amp-Out > Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-Out vals: [0x80] > Pincap 0x00000010: OUT > Pin Default 0x411111f0: [N/A] Speaker at Ext Rear > Conn = 1/8, Color = Black > DefAssociation = 0xf, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x00: > Connection: 1 > 0x0f > Node 0x18 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out > Control: name="Rear Mic Boost Volume", index=0, device=0 > ControlAmp: chs=3, dir=In, idx=0, ofs=0 > Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x2f, mute=0 > Amp-In vals: [0x02 0x02] > Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-Out vals: [0x80 0x80] > Pincap 0x00001734: IN OUT Detect > Vref caps: HIZ 50 GRD 80 > Pin Default 0x01a11830: [Jack] Mic at Ext Rear > Conn = 1/8, Color = Black > DefAssociation = 0x3, Sequence = 0x0 > Pin-ctls: 0x24: IN VREF_80 > Unsolicited: tag=00, enabled=0 > Connection: 1 > 0x0d > Node 0x19 [Pin Complex] wcaps 0x40008b: Stereo Amp-In > Control: name="Front Mic Boost Volume", index=0, device=0 > ControlAmp: chs=3, dir=In, idx=0, ofs=0 > Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x2f, mute=0 > Amp-In vals: [0x00 0x00] > Pincap 0x00001724: IN Detect > Vref caps: HIZ 50 GRD 80 > Pin Default 0x02a1183f: [Jack] Mic at Ext Front > Conn = 1/8, Color = Black > DefAssociation = 0x3, Sequence = 0xf > Pin-ctls: 0x24: IN VREF_80 > Unsolicited: tag=00, enabled=0 > Node 0x1a [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out > Control: name="PCM Playback Switch", index=0, device=0 > ControlAmp: chs=3, dir=Out, idx=0, ofs=0 > Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x2f, mute=0 > Amp-In vals: [0x00 0x00] > Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-Out vals: [0x00 0x00] > Pincap 0x0000003c: IN OUT HP Detect > Pin Default 0x02211020: [Jack] HP Out at Ext Front > Conn = 1/8, Color = Black > DefAssociation = 0x2, Sequence = 0x0 > Pin-ctls: 0x40: OUT > Unsolicited: tag=01, enabled=1 > Connection: 2 > 0x0c* 0x0d > Node 0x1b [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out > Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x2f, mute=0 > Amp-In vals: [0x00 0x00] > Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-Out vals: [0x80 0x80] > Pincap 0x00000034: IN OUT Detect > Pin Default 0x411111f0: [N/A] Speaker at Ext Rear > Conn = 1/8, Color = Black > DefAssociation = 0xf, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x20: IN > Unsolicited: tag=00, enabled=0 > Connection: 2 > 0x0c* 0x0d > Node 0x1c [Vendor Defined Widget] wcaps 0xf00000: Mono > Node 0x1d [Pin Complex] wcaps 0x400000: Mono > Pincap 0x00000020: IN > Pin Default 0x598301f0: [N/A] Line In at Int ATAPI > Conn = ATAPI, Color = Unknown > DefAssociation = 0xf, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x20: IN > Node 0x1e [Pin Complex] wcaps 0x400381: Stereo Digital > Pincap 0x00000014: OUT Detect > Pin Default 0x411111f0: [N/A] Speaker at Ext Rear > Conn = 1/8, Color = Black > DefAssociation = 0xf, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x40: OUT > Unsolicited: tag=00, enabled=0 > Connection: 1 > 0x06 > Node 0x1f [Vendor Defined Widget] wcaps 0xf00000: Mono > Node 0x20 [Vendor Defined Widget] wcaps 0xf00040: Mono > Processing caps: benign=0, ncoeff=25 > Node 0x21 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out > Control: name="Headphone Playback Switch", index=0, device=0 > ControlAmp: chs=3, dir=Out, idx=0, ofs=0 > Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-Out vals: [0x00 0x00] > Pincap 0x0000001c: OUT HP Detect > Pin Default 0x0121101f: [Jack] HP Out at Ext Rear > Conn = 1/8, Color = Black > DefAssociation = 0x1, Sequence = 0xf > Pin-ctls: 0xc0: OUT HP > Unsolicited: tag=04, enabled=1 > Connection: 2 > 0x0c 0x0d* > Node 0x22 [Audio Selector] wcaps 0x30010b: Stereo Amp-In > Control: name="Input Source", index=0, device=0 > Amp-In caps: N/A > Amp-In vals: [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 > 0x00] [0x00 0x00] [0x00 0x00] > Connection: 7 > 0x18* 0x19 0x1a 0x1b 0x1d 0x0b 0x12 > Node 0x23 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In > Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-In vals: [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 > 0x80] [0x80 0x80] > Connection: 6 > 0x18 0x19 0x1a 0x1b 0x1d 0x0b > Codec: Intel IbexPeak HDMI > Address: 3 > AFG Function Id: 0x1 (unsol 0) > Vendor Id: 0x80862804 > Subsystem Id: 0x80860101 > Revision Id: 0x100000 > No Modem Function Group found > Default PCM: > rates [0x0]: > bits [0x0]: > formats [0x0]: > Default Amp-In caps: N/A > Default Amp-Out caps: N/A > GPIO: io=0, o=0, i=0, unsolicited=0, wake=0 > Node 0x02 [Audio Output] wcaps 0x6611: 8-Channels Digital > Converter: stream=0, channel=0 > Digital: Enabled GenLevel > Digital category: 0x2 > PCM: > rates [0x7f0]: 32000 44100 48000 88200 96000 176400 192000 > bits [0x1e]: 16 20 24 32 > formats [0x5]: PCM AC3 > Power states: D0 D3 > Power: setting=D0, actual=D0 > Node 0x03 [Audio Output] wcaps 0x6611: 8-Channels Digital > Converter: stream=0, channel=0 > Digital: Enabled > Digital category: 0x0 > PCM: > rates [0x7f0]: 32000 44100 48000 88200 96000 176400 192000 > bits [0x1e]: 16 20 24 32 > formats [0x5]: PCM AC3 > Power states: D0 D3 > Power: setting=D0, actual=D0 > Node 0x04 [Pin Complex] wcaps 0x40778d: 8-Channels Digital Amp-Out CP > Control: name="IEC958 Playback Con Mask", index=0, device=0 > Control: name="IEC958 Playback Pro Mask", index=0, device=0 > Control: name="IEC958 Playback Default", index=0, device=0 > Control: name="IEC958 Playback Switch", index=0, device=0 > Control: name="ELD", index=0, device=3 > Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-Out vals: [0x00 0x00] > Pincap 0x09000094: OUT Detect HBR HDMI DP > Pin Default 0x18560010: [Jack] Digital Out at Int HDMI > Conn = Digital, Color = Unknown > DefAssociation = 0x1, Sequence = 0x0 > Pin-ctls: 0x00: > Unsolicited: tag=04, enabled=1 > Power states: D0 D3 > Power: setting=D0, actual=D0 > Connection: 2 > 0x02* 0x03 > Node 0x05 [Pin Complex] wcaps 0x40778d: 8-Channels Digital Amp-Out CP > Control: name="IEC958 Playback Con Mask", index=1, device=0 > Control: name="IEC958 Playback Pro Mask", index=1, device=0 > Control: name="IEC958 Playback Default", index=1, device=0 > Control: name="IEC958 Playback Switch", index=1, device=0 > Control: name="ELD", index=0, device=7 > Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-Out vals: [0x00 0x00] > Pincap 0x09000094: OUT Detect HBR HDMI DP > Pin Default 0x18560010: [Jack] Digital Out at Int HDMI > Conn = Digital, Color = Unknown > DefAssociation = 0x1, Sequence = 0x0 > Pin-ctls: 0x00: > Unsolicited: tag=05, enabled=1 > Power states: D0 D3 > Power: setting=D0, actual=D0 > Connection: 2 > 0x02* 0x03 > Node 0x06 [Pin Complex] wcaps 0x40778d: 8-Channels Digital Amp-Out CP > Control: name="IEC958 Playback Con Mask", index=2, device=0 > Control: name="IEC958 Playback Pro Mask", index=2, device=0 > Control: name="IEC958 Playback Default", index=2, device=0 > Control: name="IEC958 Playback Switch", index=2, device=0 > Control: name="ELD", index=0, device=8 > Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-Out vals: [0x00 0x00] > Pincap 0x09000094: OUT Detect HBR HDMI DP > Pin Default 0x18560010: [Jack] Digital Out at Int HDMI > Conn = Digital, Color = Unknown > DefAssociation = 0x1, Sequence = 0x0 > Pin-ctls: 0x00: > Unsolicited: tag=06, enabled=1 > Power states: D0 D3 > Power: setting=D0, actual=D0 > Connection: 2 > 0x02* 0x03 > Node 0x07 [Vendor Defined Widget] wcaps 0xf00000: Mono > --endcollapse-- >

2 1

Re: [alsa-devel] multiple, non-physically accesible, HDMI devices listed for Intel IbexPeak ALC269VB
by Stephen Warren 23 Feb '12

23 Feb '12

Andres Cimmarusti wrote at Wednesday, February 22, 2012 5:02 PM: > First of all, thank you for your feedback. It's been quite helpful / insightful ... > > There are ways to configure pulseaudio to allow the user to select which > > PCM device to use on a given sound card. David Henningsson made this work > > for NVIDIA GPUs at least in Ubuntu, and I imagine the same technique > > could be applied to Intel devices too. > > Mmm.. just in Ubuntu? was this work submitted upstream? It appears > there are some related fixes shown in the Ubuntu pulseaudio changelog: I think the following Ubuntu bug covers the patch: https://bugs.launchpad.net/ubuntu/+source/pulseaudio/+bug/716371 See the patch in comment #2. Hopefully David will chime in on upstreaming status etc. The following might be interesting; at least some of it will apply to non-NVIDIA graphics cards: ftp://download.nvidia.com/XFree86/gpu-hdmi-audio-document/gpu-hdmi-audio.ht… -- nvpublic

1 0

Re: [alsa-devel] multiple, non-physically accesible, HDMI devices listed for Intel IbexPeak ALC269VB
by Stephen Warren 22 Feb '12

22 Feb '12

Andres Cimmarusti wrote at Tuesday, February 21, 2012 11:44 PM: ... > Unfortunately with kernel 3.2.x and 3.1.x I get no sound out choosing the same > configuration in pulseaudio. Device is advertised correctly but > there's a bizarre multiplicity advertised: > > $ aplay -l > **** List of PLAYBACK Hardware Devices **** > card 0: Intel [HDA Intel], device 0: ALC269VB Analog [ALC269VB Analog] > Subdevices: 1/1 > Subdevice #0: subdevice #0 > card 0: Intel [HDA Intel], device 3: HDMI 0 [HDMI 0] > Subdevices: 1/1 > Subdevice #0: subdevice #0 > card 0: Intel [HDA Intel], device 7: HDMI 1 [HDMI 1] > Subdevices: 1/1 > Subdevice #0: subdevice #0 > card 0: Intel [HDA Intel], device 8: HDMI 2 [HDMI 2] > Subdevices: 1/1 > Subdevice #0: subdevice #0 > > Using aplay with device 3 says device is busy. pulseaudio probably has that device open, hence aplay can't open it again. To validate this, run the following: pasuspender bash Then, run your aplay within that bash. At least aplay should run now, even if you still can't hear audio; it's possible you don't (or can't) have anything hooked up to that audio output > Device 7 works > correctly (but is not available in pulseaudio unless forced by > default, which then renders internal speakers disabled) There are ways to configure pulseaudio to allow the user to select which PCM device to use on a given sound card. David Henningsson made this work for NVIDIA GPUs at least in Ubuntu, and I imagine the same technique could be applied to Intel devices too. In the meantime, the WAR that Takashi mentioned in his response sounds like it'd work very well, albeit it's a bit manual to set up. > and device 8 produces no sound out. It's plausible that isn't connected, or is connected to a graphics port you simply don't have hooked up to a monitor or audio-capable monitor. As Takashi mentions, from a kernel perspective, this isn't really a regression at all, but simply exposing all the features of the HW that were previously hidden. Without that change, others can't use some HW usefully at all. Unfortunately, pulseaudio makes some rather simplistic assumptions about how HW works by default, and can be confused by the additional features that are exposed. Sorry this caused problems for you. -- nvpublic

2 1

[alsa-devel] [PATCH 0/5] Refresh ak4104 driver
by Mark Brown 22 Feb '12

22 Feb '12

This patch series updates the ak4104 driver to more modern idioms, mainly converting to regmap. I've absolutely no ability to test this so there's a reasonable chance this will just explode but hopefully it should work fine. Since all the read modify write, I/O and cache code was open coded in the driver we end up factoring out a reasonable amount. Mark Brown (5): ASoC: ak4104: Remove uninformative print on probe() ASoC: ak4104: Convert to module_spi_driver() ASoC: ak4104: Use snd_soc_update_bits() for read/modify/write ASoC: ak4104: Use snd_soc_write() rather than internal write function ASoC: ak4104: Convert to direct regmap API usage sound/soc/codecs/ak4104.c | 174 ++++++++++++++++----------------------------- 1 files changed, 62 insertions(+), 112 deletions(-)

2 6

Re: [alsa-devel] Random process lockup on ARM board: alsa-lib-1.0.25, FUTEX_WAIT_PRIVATE
by Clemens Ladisch 22 Feb '12

22 Feb '12

Jonathan Andrews wrote: > On Mon, 2012-02-20 at 08:54 +0100, Clemens Ladisch wrote: >> Jonathan Andrews wrote: >>> I've tried building alsalib in many ways, at the moment its : >>> $ ./configure --target=arm-linux --host=i686-linux --enable-shared --disable-static --without-pthread >> >> Try adding --disable-ucm and --disable-python. > > Err ok, alsa-lib still builds ok. Now it doesn't use any pthread functions. > My application runs for longer, but still freezes in the end. With the > same futex locking problem. I'd guess that this is not an alsa-lib problem. Now you really need a stack trace. Regards, Clemens

1 0

[alsa-devel] v3.1.8: hda-intel broken
by Russell King - ARM Linux 22 Feb '12

22 Feb '12

v3.1.8: <4>[1641072.853409] WARNING: at sound/pci/hda/hda_codec.c:5152 snd_array_new+0x47/0x9b [snd_hda_codec]() <4>[1641072.853412] Hardware name: 2081CTO <4>[1641072.853413] BUG? (num >= 4096) <4>[1641072.853415] Modules linked in: lp cdc_ether usbnet mii cdc_acm cdc_wdm usblp ftdi_sio usbserial parport_jtag fuse joydev cpufreq_ondemand acpi_cpufreq mperf nf_conntrack_netbios_ns nf_conntrack_broadcast ip6t_REJECT nf_conntrack_ipv6 nf_defrag_ipv6 ip6table_filter ip6_tables ipv6 uinput snd_hda_codec_conexant snd_hda_intel snd_hda_codec snd_hwdep snd_seq snd_seq_device arc4 snd_pcm snd_timer r852 sm_common nand nand_ids nand_ecc mtd iwlagn mac80211 snd_page_alloc ppdev iTCO_wdt parport_pc cfg80211 iTCO_vendor_support i2c_i801 thinkpad_acpi rfkill e1000e snd soundcore parport pcspkr wmi microcode firewire_ohci sdhci_pci sdhci mmc_core firewire_core crc_itu_t yenta_socket i915 drm_kms_helper drm i2c_algo_bit i2c_core video [last unloaded: ftdi_sio] <4>[1641072.853459] Pid: 3089, comm: bash Tainted: G W 3.1.8 #2 <4>[1641072.853461] Call Trace: <4>[1641072.853466] [<c04387eb>] warn_slowpath_common+0x6a/0x7f <4>[1641072.853472] [<f861d5d0>] ? snd_array_new+0x47/0x9b [snd_hda_codec] <4>[1641072.853475] [<c0438873>] warn_slowpath_fmt+0x2b/0x2f <4>[1641072.853481] [<f861d5d0>] snd_array_new+0x47/0x9b [snd_hda_codec] <4>[1641072.853488] [<f861d65c>] snd_hda_input_jack_add+0x38/0xa2 [snd_hda_codec] <4>[1641072.853493] [<f872dcc4>] cxt5051_init+0x62/0xea [snd_hda_codec_conexant] <4>[1641072.853500] [<f8621c51>] hda_call_codec_resume+0xa5/0xba [snd_hda_codec] <4>[1641072.853506] [<f861f369>] snd_hda_power_up+0x46/0x5c [snd_hda_codec] <4>[1641072.853513] [<f861f3b9>] codec_exec_verb+0x3a/0xc7 [snd_hda_codec] <4>[1641072.853519] [<f861f482>] snd_hda_codec_write+0x3c/0x43 [snd_hda_codec] <4>[1641072.853526] [<f861f4b9>] set_pincfg+0x30/0x45 [snd_hda_codec] <4>[1641072.853532] [<f861f507>] restore_pincfgs+0x39/0x46 [snd_hda_codec] <4>[1641072.853539] [<f861fa0a>] snd_hda_codec_free+0x6c/0x1f8 [snd_hda_codec] <4>[1641072.853545] [<f861fbdb>] snd_hda_bus_free+0x45/0x7c [snd_hda_codec] <4>[1641072.853552] [<f861fdb9>] snd_hda_bus_dev_free+0x17/0x19 [snd_hda_codec]<4>[1641072.853560] [<f80979a4>] snd_device_free+0x91/0xde [snd] <4>[1641072.853566] [<f8097c9c>] snd_device_free_all+0x68/0x80 [snd] <4>[1641072.853572] [<f80936e3>] snd_card_do_free+0x4e/0xdf [snd] <4>[1641072.853577] [<f8094160>] snd_card_free+0x89/0x94 [snd] <4>[1641072.853581] [<c044fcee>] ? wake_up_bit+0x20/0x20 <4>[1641072.853585] [<f86ccddf>] azx_remove+0x13/0x1f [snd_hda_intel] <4>[1641072.853589] [<c05de80d>] pci_device_remove+0x2c/0x74 <4>[1641072.853593] [<c066371d>] __device_release_driver+0x66/0x9c <4>[1641072.853596] [<c0663770>] device_release_driver+0x1d/0x28 <4>[1641072.853599] [<c0662caf>] driver_unbind+0x4d/0x7b <4>[1641072.853601] [<c0662c62>] ? store_drivers_probe+0x33/0x33 <4>[1641072.853604] [<c06627fb>] drv_attr_store+0x24/0x28 <4>[1641072.853607] [<c05263a6>] sysfs_write_file+0xb3/0xec <4>[1641072.853609] [<c05262f3>] ? sysfs_poll+0x73/0x73 <4>[1641072.853612] [<c04df254>] vfs_write+0x87/0xde <4>[1641072.853615] [<c04e678e>] ? path_put+0x1a/0x1d <4>[1641072.853617] [<c04df428>] sys_write+0x40/0x62 <4>[1641072.853621] [<c07c751f>] sysenter_do_call+0x12/0x28 <4>[1641072.853623] ---[ end trace 3d590d4bbb62d735 ]--- _From what I can see, this uses the suspend/resume methods on device open/ close. The suspend method does nothing apart from "shutting up pins". The resume method adds new jacks via snd_array_new(), which eventually ends up getting (after many many days of uptime) to 4096 entries. And then it starts exploding with warnings every so often, filling my screen from top to bottom with abrtd boxes warning me that my kernel is doing this. Then, if I try to unbind the PCI device associated with this to shut the thing up, it oopses and takes the machine out: <1>[1641072.855748] BUG: unable to handle kernel NULL pointer dereference at 00000018 <1>[1641072.855796] IP: [<c06cf531>] is_event_supported+0xf/0x20 <4>[1641072.855830] *pde = 00000000 <0>[1641072.855849] Oops: 0000 [#1] SMP <4>[1641072.855873] Modules linked in: lp cdc_ether usbnet mii cdc_acm cdc_wdm usblp ftdi_sio usbserial parport_jtag fuse joydev cpufreq_ondemand acpi_cpufreq mperf nf_conntrack_netbios_ns nf_conntrack_broadcast ip6t_REJECT nf_conntrack_ipv6 nf_defrag_ipv6 ip6table_filter ip6_tables ipv6 uinput snd_hda_codec_conexant snd_hda_intel snd_hda_codec snd_hwdep snd_seq snd_seq_device arc4 snd_pcm snd_timer r852 sm_common nand nand_ids nand_ecc mtd iwlagn mac80211 snd_page_alloc ppdev iTCO_wdt parport_pc cfg80211 iTCO_vendor_support i2c_i801 thinkpad_acpi rfkill e1000e snd soundcore parport pcspkr wmi microcode firewire_ohci sdhci_pci sdhci mmc_core firewire_core crc_itu_t yenta_socket i915 drm_kms_helper drm i2c_algo_bit i2c_core video [last unloaded: ftdi_sio] <4>[1641072.856004] <4>[1641072.856004] Pid: 3089, comm: bash Tainted: G W 3.1.8 #2 LENOVO 2081CTO/2081CTO <4>[1641072.856004] EIP: 0060:[<c06cf531>] EFLAGS: 00210293 CPU: 1 <4>[1641072.856004] EIP is at is_event_supported+0xf/0x20 <4>[1641072.856004] EAX: 00000005 EBX: 00000000 ECX: 0000001f EDX: 00000018 <4>[1641072.856004] ESI: 00000005 EDI: 00000000 EBP: c42a3d50 ESP: c42a3d4c <4>[1641072.856004] DS: 007b ES: 007b FS: 00d8 GS: 00e0 SS: 0068 <0>[1641072.856004] Process bash (pid: 3089, ti=c42a2000 task=c8009920 task.ti=c42a2000) <0>[1641072.856004] Stack: <4>[1641072.856004] 00000000 c42a3d6c c06d075f 00000000 00000002 f1a47780 00000000 00000000 <4>[1641072.856004] c42a3d84 f8098c00 00000000 c13005a0 f46c6000 00000078 c42a3da4 f862132e <4>[1641072.856004] 00000000 00000016 00160016 f8731ce8 f46c6000 00000000 c42a3dc8 f872dcd1 <0>[1641072.856004] Call Trace: <4>[1641072.856004] [<c06d075f>] input_event+0x24/0x69 <4>[1641072.856004] [<f8098c00>] snd_jack_report+0x70/0x8d [snd] <4>[1641072.856004] [<f862132e>] snd_hda_input_jack_report+0x7a/0x8e [snd_hda_codec] <4>[1641072.856004] [<f872dcd1>] cxt5051_init+0x6f/0xea [snd_hda_codec_conexant] <4>[1641072.856004] [<f8621c51>] hda_call_codec_resume+0xa5/0xba [snd_hda_codec] <4>[1641072.856004] [<f861f369>] snd_hda_power_up+0x46/0x5c [snd_hda_codec] <4>[1641072.856004] [<f861f3b9>] codec_exec_verb+0x3a/0xc7 [snd_hda_codec] <4>[1641072.856004] [<f861f482>] snd_hda_codec_write+0x3c/0x43 [snd_hda_codec] <4>[1641072.856004] [<f861f4b9>] set_pincfg+0x30/0x45 [snd_hda_codec] <4>[1641072.856004] [<f861f507>] restore_pincfgs+0x39/0x46 [snd_hda_codec] <4>[1641072.856004] [<f861fa0a>] snd_hda_codec_free+0x6c/0x1f8 [snd_hda_codec] <4>[1641072.856004] [<f861fbdb>] snd_hda_bus_free+0x45/0x7c [snd_hda_codec] <4>[1641072.856004] [<f861fdb9>] snd_hda_bus_dev_free+0x17/0x19 [snd_hda_codec]<4>[1641072.856004] [<f80979a4>] snd_device_free+0x91/0xde [snd] <4>[1641072.856004] [<f8097c9c>] snd_device_free_all+0x68/0x80 [snd] <4>[1641072.856004] [<f80936e3>] snd_card_do_free+0x4e/0xdf [snd] <4>[1641072.856004] [<f8094160>] snd_card_free+0x89/0x94 [snd] <4>[1641072.856004] [<c044fcee>] ? wake_up_bit+0x20/0x20 <4>[1641072.856004] [<f86ccddf>] azx_remove+0x13/0x1f [snd_hda_intel] <4>[1641072.856004] [<c05de80d>] pci_device_remove+0x2c/0x74 <4>[1641072.856004] [<c066371d>] __device_release_driver+0x66/0x9c <4>[1641072.856004] [<c0663770>] device_release_driver+0x1d/0x28 <4>[1641072.856004] [<c0662caf>] driver_unbind+0x4d/0x7b <4>[1641072.856004] [<c0662c62>] ? store_drivers_probe+0x33/0x33 <4>[1641072.856004] [<c06627fb>] drv_attr_store+0x24/0x28 <4>[1641072.856004] [<c05263a6>] sysfs_write_file+0xb3/0xec <4>[1641072.856004] [<c05262f3>] ? sysfs_poll+0x73/0x73 <4>[1641072.856004] [<c04df254>] vfs_write+0x87/0xde <4>[1641072.856004] [<c04e678e>] ? path_put+0x1a/0x1d <4>[1641072.856004] [<c04df428>] sys_write+0x40/0x62 <4>[1641072.856004] [<c07c751f>] sysenter_do_call+0x12/0x28 <0>[1641072.856004] Code: d0 fe ff ff eb 05 bf f0 ff ff ff b8 14 ce 9e c0 e8 a2 19 0f 00 89 f8 5b 5e 5f 5d c3 55 89 e5 53 3e 8d 74 26 00 31 db 39 c8 77 0c <0f> a3 02 19 c0 31 db 85 c0 0f 95 c3 89 d8 5b 5d c3 55 89 e5 57 <0>[1641072.856004] EIP: [<c06cf531>] is_event_supported+0xf/0x20 SS:ESP 0068:c42a3d4c <0>[1641072.856004] CR2: 0000000000000018

2 14

[alsa-devel] [PATCH v3 0/5] snd-usb-6fire: volume control updates
by Torsten Schenk 22 Feb '12

22 Feb '12

Changes in v3: - update only channels that need updating - indices for controls now start from 0 - snd_ctl_make_virtual_master now also gets a tlv Thanks, Torsten

2 1

[alsa-devel] [PATCH v3 5/5] snd-usb-6fire: add analog input volume control
by Torsten Schenk 22 Feb '12

22 Feb '12

Add a stereo volume control for analog input channel pair 1/2. Signed-off-by: Torsten Schenk <torsten.schenk(a)zoho.com> diff -Nur a/sound/usb/6fire/control.c b/sound/usb/6fire/control.c --- a/sound/usb/6fire/control.c 2012-02-22 15:05:47.206526365 +0100 +++ b/sound/usb/6fire/control.c 2012-02-22 15:11:26.846504694 +0100 @@ -55,6 +55,7 @@ static const u16 rates_6fire_vh[] = {0x11, 0x11, 0x10, 0x10, 0x00, 0x00}; static DECLARE_TLV_DB_MINMAX(tlv_output, -9000, 0); +static DECLARE_TLV_DB_MINMAX(tlv_input, -1500, 1500); enum { DIGITAL_THRU_ONLY_SAMPLERATE = 3 @@ -82,6 +83,20 @@ comm_rt->write8(comm_rt, 0x12, 0x0e, ~rt->output_mute); } +static void usb6fire_control_input_vol_update(struct control_runtime *rt) +{ + struct comm_runtime *comm_rt = rt->chip->comm; + int i; + + if (comm_rt) + for (i = 0; i < 2; i++) + if (!(rt->ivol_updated & (1 << i))) { + comm_rt->write8(comm_rt, 0x12, 0x1c + i, + rt->input_vol[i] & 0x3f); + rt->ivol_updated |= 1 << i; + } +} + static void usb6fire_control_line_phono_update(struct control_runtime *rt) { struct comm_runtime *comm_rt = rt->chip->comm; @@ -261,6 +276,50 @@ return 0; } +static int usb6fire_control_input_vol_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 2; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 30; + return 0; +} + +static int usb6fire_control_input_vol_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct control_runtime *rt = snd_kcontrol_chip(kcontrol); + int changed = 0; + + if (rt->input_vol[0] != ucontrol->value.integer.value[0]) { + rt->input_vol[0] = ucontrol->value.integer.value[0] - 15; + rt->ivol_updated &= ~(1 << 0); + changed = 1; + } + if (rt->input_vol[1] != ucontrol->value.integer.value[1]) { + rt->input_vol[1] = ucontrol->value.integer.value[1] - 15; + rt->ivol_updated &= ~(1 << 1); + changed = 1; + } + + if (changed) + usb6fire_control_input_vol_update(rt); + + return changed; +} + +static int usb6fire_control_input_vol_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct control_runtime *rt = snd_kcontrol_chip(kcontrol); + + ucontrol->value.integer.value[0] = rt->input_vol[0] + 15; + ucontrol->value.integer.value[1] = rt->input_vol[1] + 15; + + return 0; +} + static int usb6fire_control_line_phono_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { @@ -454,6 +513,17 @@ .get = usb6fire_control_digital_thru_get, .put = usb6fire_control_digital_thru_put }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Analog Capture Volume", + .index = 0, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ, + .info = usb6fire_control_input_vol_info, + .get = usb6fire_control_input_vol_get, + .put = usb6fire_control_input_vol_put, + .tlv = { .p = tlv_input } + }, {} }; @@ -518,6 +588,7 @@ usb6fire_control_line_phono_update(rt); usb6fire_control_output_vol_update(rt); usb6fire_control_output_mute_update(rt); + usb6fire_control_input_vol_update(rt); usb6fire_control_streaming_update(rt); ret = usb6fire_control_add_virtual(rt, chip->card, diff -Nur a/sound/usb/6fire/control.h b/sound/usb/6fire/control.h --- a/sound/usb/6fire/control.h 2012-02-22 15:03:44.370251101 +0100 +++ b/sound/usb/6fire/control.h 2012-02-22 15:10:05.055517247 +0100 @@ -46,6 +46,8 @@ u8 output_vol[6]; u8 ovol_updated; u8 output_mute; + s8 input_vol[2]; + u8 ivol_updated; }; int __devinit usb6fire_control_init(struct sfire_chip *chip);

1 0

[alsa-devel] [PATCH v3 4/5] snd-usb-6fire: add mute control for analog outputs
by Torsten Schenk 22 Feb '12

22 Feb '12

Add a mute control for every analog output channel. Signed-off-by: Torsten Schenk <torsten.schenk(a)zoho.com> diff -Nur a/sound/usb/6fire/control.c b/sound/usb/6fire/control.c --- a/sound/usb/6fire/control.c 2012-02-22 15:02:08.594916052 +0100 +++ b/sound/usb/6fire/control.c 2012-02-22 15:05:47.206526365 +0100 @@ -74,6 +74,14 @@ } } +static void usb6fire_control_output_mute_update(struct control_runtime *rt) +{ + struct comm_runtime *comm_rt = rt->chip->comm; + + if (comm_rt) + comm_rt->write8(comm_rt, 0x12, 0x0e, ~rt->output_mute); +} + static void usb6fire_control_line_phono_update(struct control_runtime *rt) { struct comm_runtime *comm_rt = rt->chip->comm; @@ -208,6 +216,51 @@ return 0; } +static int usb6fire_control_output_mute_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct control_runtime *rt = snd_kcontrol_chip(kcontrol); + unsigned int ch = kcontrol->private_value; + u8 old = rt->output_mute; + u8 value = 0; + + if (ch > 4) { + snd_printk(KERN_ERR PREFIX "Invalid channel in volume control."); + return -EINVAL; + } + + rt->output_mute &= ~(3 << ch); + if (ucontrol->value.integer.value[0]) + value |= 1; + if (ucontrol->value.integer.value[1]) + value |= 2; + rt->output_mute |= value << ch; + + if (rt->output_mute != old) + usb6fire_control_output_mute_update(rt); + + return rt->output_mute != old; +} + +static int usb6fire_control_output_mute_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct control_runtime *rt = snd_kcontrol_chip(kcontrol); + unsigned int ch = kcontrol->private_value; + u8 value = rt->output_mute >> ch; + + if (ch > 4) { + snd_printk(KERN_ERR PREFIX "Invalid channel in volume control."); + return -EINVAL; + } + + ucontrol->value.integer.value[0] = 1 & value; + value >>= 1; + ucontrol->value.integer.value[1] = 1 & value; + + return 0; +} + static int usb6fire_control_line_phono_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { @@ -339,6 +392,40 @@ {} }; +static struct __devinitdata snd_kcontrol_new mute_elements[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Analog Playback Switch", + .index = 0, + .private_value = 0, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_ctl_boolean_stereo_info, + .get = usb6fire_control_output_mute_get, + .put = usb6fire_control_output_mute_put, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Analog Playback Switch", + .index = 1, + .private_value = 2, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_ctl_boolean_stereo_info, + .get = usb6fire_control_output_mute_get, + .put = usb6fire_control_output_mute_put, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Analog Playback Switch", + .index = 2, + .private_value = 4, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_ctl_boolean_stereo_info, + .get = usb6fire_control_output_mute_get, + .put = usb6fire_control_output_mute_put, + }, + {} +}; + static struct __devinitdata snd_kcontrol_new elements[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -430,13 +517,21 @@ usb6fire_control_opt_coax_update(rt); usb6fire_control_line_phono_update(rt); usb6fire_control_output_vol_update(rt); + usb6fire_control_output_mute_update(rt); usb6fire_control_streaming_update(rt); ret = usb6fire_control_add_virtual(rt, chip->card, "Master Playback Volume", vol_elements); if (ret) { + snd_printk(KERN_ERR PREFIX "cannot add control.\n"); kfree(rt); + return ret; + } + ret = usb6fire_control_add_virtual(rt, chip->card, + "Master Playback Switch", mute_elements); + if (ret) { snd_printk(KERN_ERR PREFIX "cannot add control.\n"); + kfree(rt); return ret; } diff -Nur a/sound/usb/6fire/control.h b/sound/usb/6fire/control.h --- a/sound/usb/6fire/control.h 2012-02-22 15:00:10.034785200 +0100 +++ b/sound/usb/6fire/control.h 2012-02-22 15:03:44.370251101 +0100 @@ -45,6 +45,7 @@ bool usb_streaming; u8 output_vol[6]; u8 ovol_updated; + u8 output_mute; }; int __devinit usb6fire_control_init(struct sfire_chip *chip);

1 0

[alsa-devel] [PATCH v3 3/5] snd-usb-6fire: add individual volume control for analog channels
by Torsten Schenk 22 Feb '12

22 Feb '12

Add a stereo volume control for every analog output pair 1/2, 3/4, 5/6. Signed-off-by: Torsten Schenk <torsten.schenk(a)zoho.com> diff -Nur a/sound/usb/6fire/control.c b/sound/usb/6fire/control.c --- a/sound/usb/6fire/control.c 2012-01-17 18:36:14.480508754 +0100 +++ b/sound/usb/6fire/control.c 2012-02-22 15:02:08.594916052 +0100 @@ -7,6 +7,10 @@ * Created: Jan 01, 2011 * Copyright: (C) Torsten Schenk * + * Thanks to: + * - Holger Ruckdeschel: he found out how to control individual channel + * volumes and introduced mute switch + * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by * the Free Software Foundation; either version 2 of the License, or @@ -39,7 +43,7 @@ { 0x22, 0x03, 0x00 }, { 0x20, 0x03, 0x08 }, { 0x22, 0x04, 0x00 }, { 0x20, 0x04, 0x08 }, { 0x22, 0x05, 0x01 }, { 0x20, 0x05, 0x08 }, { 0x22, 0x04, 0x01 }, { 0x12, 0x04, 0x00 }, { 0x12, 0x05, 0x00 }, - { 0x12, 0x0d, 0x78 }, { 0x12, 0x21, 0x82 }, { 0x12, 0x22, 0x80 }, + { 0x12, 0x0d, 0x38 }, { 0x12, 0x21, 0x82 }, { 0x12, 0x22, 0x80 }, { 0x12, 0x23, 0x00 }, { 0x12, 0x06, 0x02 }, { 0x12, 0x03, 0x00 }, { 0x12, 0x02, 0x00 }, { 0x22, 0x03, 0x01 }, { 0 } /* TERMINATING ENTRY */ @@ -56,15 +60,18 @@ DIGITAL_THRU_ONLY_SAMPLERATE = 3 }; -static void usb6fire_control_master_vol_update(struct control_runtime *rt) +static void usb6fire_control_output_vol_update(struct control_runtime *rt) { struct comm_runtime *comm_rt = rt->chip->comm; - if (comm_rt) { - /* set volume */ - comm_rt->write8(comm_rt, 0x12, 0x0f, 180 - rt->master_vol); - /* unmute */ - comm_rt->write8(comm_rt, 0x12, 0x0e, 0x00); - } + int i; + + if (comm_rt) + for (i = 0; i < 6; i++) + if (!(rt->ovol_updated & (1 << i))) { + comm_rt->write8(comm_rt, 0x12, 0x0f + i, + 180 - rt->output_vol[i]); + rt->ovol_updated |= 1 << i; + } } static void usb6fire_control_line_phono_update(struct control_runtime *rt) @@ -146,34 +153,58 @@ return -EINVAL; } -static int usb6fire_control_master_vol_info(struct snd_kcontrol *kcontrol, +static int usb6fire_control_output_vol_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; - uinfo->count = 1; + uinfo->count = 2; uinfo->value.integer.min = 0; uinfo->value.integer.max = 180; return 0; } -static int usb6fire_control_master_vol_put(struct snd_kcontrol *kcontrol, +static int usb6fire_control_output_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct control_runtime *rt = snd_kcontrol_chip(kcontrol); + unsigned int ch = kcontrol->private_value; int changed = 0; - if (rt->master_vol != ucontrol->value.integer.value[0]) { - rt->master_vol = ucontrol->value.integer.value[0]; - usb6fire_control_master_vol_update(rt); + + if (ch > 4) { + snd_printk(KERN_ERR PREFIX "Invalid channel in volume control."); + return -EINVAL; + } + + if (rt->output_vol[ch] != ucontrol->value.integer.value[0]) { + rt->output_vol[ch] = ucontrol->value.integer.value[0]; + rt->ovol_updated &= ~(1 << ch); changed = 1; } + if (rt->output_vol[ch + 1] != ucontrol->value.integer.value[1]) { + rt->output_vol[ch + 1] = ucontrol->value.integer.value[1]; + rt->ovol_updated &= ~(2 << ch); + changed = 1; + } + + if (changed) + usb6fire_control_output_vol_update(rt); + return changed; } -static int usb6fire_control_master_vol_get(struct snd_kcontrol *kcontrol, +static int usb6fire_control_output_vol_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct control_runtime *rt = snd_kcontrol_chip(kcontrol); - ucontrol->value.integer.value[0] = rt->master_vol; + unsigned int ch = kcontrol->private_value; + + if (ch > 4) { + snd_printk(KERN_ERR PREFIX "Invalid channel in volume control."); + return -EINVAL; + } + + ucontrol->value.integer.value[0] = rt->output_vol[ch]; + ucontrol->value.integer.value[1] = rt->output_vol[ch + 1]; return 0; } @@ -268,18 +299,47 @@ return 0; } -static struct __devinitdata snd_kcontrol_new elements[] = { +static struct __devinitdata snd_kcontrol_new vol_elements[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Master Playback Volume", + .name = "Analog Playback Volume", .index = 0, + .private_value = 0, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ, + .info = usb6fire_control_output_vol_info, + .get = usb6fire_control_output_vol_get, + .put = usb6fire_control_output_vol_put, + .tlv = { .p = tlv_output } + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Analog Playback Volume", + .index = 1, + .private_value = 2, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ, + .info = usb6fire_control_output_vol_info, + .get = usb6fire_control_output_vol_get, + .put = usb6fire_control_output_vol_put, + .tlv = { .p = tlv_output } + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Analog Playback Volume", + .index = 2, + .private_value = 4, .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ, - .info = usb6fire_control_master_vol_info, - .get = usb6fire_control_master_vol_get, - .put = usb6fire_control_master_vol_put, + .info = usb6fire_control_output_vol_info, + .get = usb6fire_control_output_vol_get, + .put = usb6fire_control_output_vol_put, .tlv = { .p = tlv_output } }, + {} +}; + +static struct __devinitdata snd_kcontrol_new elements[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Line/Phono Capture Route", @@ -310,6 +370,40 @@ {} }; +static int usb6fire_control_add_virtual( + struct control_runtime *rt, + struct snd_card *card, + char *name, + struct snd_kcontrol_new *elems) +{ + int ret; + int i; + struct snd_kcontrol *vmaster = + snd_ctl_make_virtual_master(name, tlv_output); + struct snd_kcontrol *control; + + if (!vmaster) + return -ENOMEM; + ret = snd_ctl_add(card, vmaster); + if (ret < 0) + return ret; + + i = 0; + while (elems[i].name) { + control = snd_ctl_new1(&elems[i], rt); + if (!control) + return -ENOMEM; + ret = snd_ctl_add(card, control); + if (ret < 0) + return ret; + ret = snd_ctl_add_slave(vmaster, control); + if (ret < 0) + return ret; + i++; + } + return 0; +} + int __devinit usb6fire_control_init(struct sfire_chip *chip) { int i; @@ -335,9 +429,17 @@ usb6fire_control_opt_coax_update(rt); usb6fire_control_line_phono_update(rt); - usb6fire_control_master_vol_update(rt); + usb6fire_control_output_vol_update(rt); usb6fire_control_streaming_update(rt); + ret = usb6fire_control_add_virtual(rt, chip->card, + "Master Playback Volume", vol_elements); + if (ret) { + kfree(rt); + snd_printk(KERN_ERR PREFIX "cannot add control.\n"); + return ret; + } + i = 0; while (elements[i].name) { ret = snd_ctl_add(chip->card, snd_ctl_new1(&elements[i], rt)); diff -Nur a/sound/usb/6fire/control.h b/sound/usb/6fire/control.h --- a/sound/usb/6fire/control.h 2012-01-09 22:32:56.088002206 +0100 +++ b/sound/usb/6fire/control.h 2012-02-22 15:00:10.034785200 +0100 @@ -43,7 +43,8 @@ bool line_phono_switch; bool digital_thru_switch; bool usb_streaming; - u8 master_vol; + u8 output_vol[6]; + u8 ovol_updated; }; int __devinit usb6fire_control_init(struct sfire_chip *chip);

1 0

[alsa-devel] [PATCH v3 2/5] snd-usb-6fire: add tlv to controls
by Torsten Schenk 22 Feb '12

22 Feb '12

Remove the soft log-conversion and add a dB scale according to the DAC documentation instead. Signed-off-by: Torsten Schenk <torsten.schenk(a)zoho.com> diff -Nur a/sound/usb/6fire/control.c b/sound/usb/6fire/control.c --- a/sound/usb/6fire/control.c 2012-01-09 22:33:01.512182294 +0100 +++ b/sound/usb/6fire/control.c 2012-01-17 18:36:14.480508754 +0100 @@ -15,6 +15,7 @@ #include <linux/interrupt.h> #include <sound/control.h> +#include <sound/tlv.h> #include "control.h" #include "comm.h" @@ -24,26 +25,6 @@ static char *line_phono_texts[2] = { "Line", "Phono" }; /* - * calculated with $value\[i\] = 128 \cdot sqrt[3]{\frac{i}{128}}$ - * this is done because the linear values cause rapid degredation - * of volume in the uppermost region. - */ -static const u8 log_volume_table[128] = { - 0x00, 0x19, 0x20, 0x24, 0x28, 0x2b, 0x2e, 0x30, 0x32, 0x34, - 0x36, 0x38, 0x3a, 0x3b, 0x3d, 0x3e, 0x40, 0x41, 0x42, 0x43, - 0x44, 0x46, 0x47, 0x48, 0x49, 0x4a, 0x4b, 0x4c, 0x4d, 0x4e, - 0x4e, 0x4f, 0x50, 0x51, 0x52, 0x53, 0x53, 0x54, 0x55, 0x56, - 0x56, 0x57, 0x58, 0x58, 0x59, 0x5a, 0x5b, 0x5b, 0x5c, 0x5c, - 0x5d, 0x5e, 0x5e, 0x5f, 0x60, 0x60, 0x61, 0x61, 0x62, 0x62, - 0x63, 0x63, 0x64, 0x65, 0x65, 0x66, 0x66, 0x67, 0x67, 0x68, - 0x68, 0x69, 0x69, 0x6a, 0x6a, 0x6b, 0x6b, 0x6c, 0x6c, 0x6c, - 0x6d, 0x6d, 0x6e, 0x6e, 0x6f, 0x6f, 0x70, 0x70, 0x70, 0x71, - 0x71, 0x72, 0x72, 0x73, 0x73, 0x73, 0x74, 0x74, 0x75, 0x75, - 0x75, 0x76, 0x76, 0x77, 0x77, 0x77, 0x78, 0x78, 0x78, 0x79, - 0x79, 0x7a, 0x7a, 0x7a, 0x7b, 0x7b, 0x7b, 0x7c, 0x7c, 0x7c, - 0x7d, 0x7d, 0x7d, 0x7e, 0x7e, 0x7e, 0x7f, 0x7f }; - -/* * data that needs to be sent to device. sets up card internal stuff. * values dumped from windows driver and filtered by trial'n'error. */ @@ -69,6 +50,8 @@ static const u16 rates_6fire_vl[] = {0x00, 0x01, 0x00, 0x01, 0x00, 0x01}; static const u16 rates_6fire_vh[] = {0x11, 0x11, 0x10, 0x10, 0x00, 0x00}; +static DECLARE_TLV_DB_MINMAX(tlv_output, -9000, 0); + enum { DIGITAL_THRU_ONLY_SAMPLERATE = 3 }; @@ -78,8 +61,7 @@ struct comm_runtime *comm_rt = rt->chip->comm; if (comm_rt) { /* set volume */ - comm_rt->write8(comm_rt, 0x12, 0x0f, 0x7f - - log_volume_table[rt->master_vol]); + comm_rt->write8(comm_rt, 0x12, 0x0f, 180 - rt->master_vol); /* unmute */ comm_rt->write8(comm_rt, 0x12, 0x0e, 0x00); } @@ -170,7 +152,7 @@ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; uinfo->count = 1; uinfo->value.integer.min = 0; - uinfo->value.integer.max = 127; + uinfo->value.integer.max = 180; return 0; } @@ -291,10 +273,12 @@ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Volume", .index = 0, - .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ, .info = usb6fire_control_master_vol_info, .get = usb6fire_control_master_vol_get, - .put = usb6fire_control_master_vol_put + .put = usb6fire_control_master_vol_put, + .tlv = { .p = tlv_output } }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER,

1 0

[alsa-devel] [PATCH v3 1/5] snd-usb-6fire: remove driver version information
by Torsten Schenk 22 Feb '12

22 Feb '12

Remove unused driver version information from the individual files. Signed-off-by: Torsten Schenk <torsten.schenk(a)zoho.com> diff -Nur a/sound/usb/6fire/chip.c b/sound/usb/6fire/chip.c --- a/sound/usb/6fire/chip.c 2012-01-09 21:55:55.863579333 +0100 +++ b/sound/usb/6fire/chip.c 2012-01-09 22:33:49.857786986 +0100 @@ -5,7 +5,6 @@ * * Author: Torsten Schenk <torsten.schenk(a)zoho.com> * Created: Jan 01, 2011 - * Version: 0.3.0 * Copyright: (C) Torsten Schenk * * This program is free software; you can redistribute it and/or modify @@ -29,7 +28,7 @@ #include <sound/initval.h> MODULE_AUTHOR("Torsten Schenk <torsten.schenk(a)zoho.com>"); -MODULE_DESCRIPTION("TerraTec DMX 6Fire USB audio driver, version 0.3.0"); +MODULE_DESCRIPTION("TerraTec DMX 6Fire USB audio driver"); MODULE_LICENSE("GPL v2"); MODULE_SUPPORTED_DEVICE("{{TerraTec, DMX 6Fire USB}}"); diff -Nur a/sound/usb/6fire/chip.h b/sound/usb/6fire/chip.h --- a/sound/usb/6fire/chip.h 2012-01-09 21:55:55.000000000 +0100 +++ b/sound/usb/6fire/chip.h 2012-01-09 22:33:38.469409108 +0100 @@ -3,7 +3,6 @@ * * Author: Torsten Schenk <torsten.schenk(a)zoho.com> * Created: Jan 01, 2011 - * Version: 0.3.0 * Copyright: (C) Torsten Schenk * * This program is free software; you can redistribute it and/or modify diff -Nur a/sound/usb/6fire/comm.c b/sound/usb/6fire/comm.c --- a/sound/usb/6fire/comm.c 2012-01-09 21:55:55.000000000 +0100 +++ b/sound/usb/6fire/comm.c 2012-01-09 22:33:11.620517902 +0100 @@ -5,7 +5,6 @@ * * Author: Torsten Schenk <torsten.schenk(a)zoho.com> * Created: Jan 01, 2011 - * Version: 0.3.0 * Copyright: (C) Torsten Schenk * * This program is free software; you can redistribute it and/or modify diff -Nur a/sound/usb/6fire/comm.h b/sound/usb/6fire/comm.h --- a/sound/usb/6fire/comm.h 2012-01-09 21:55:55.000000000 +0100 +++ b/sound/usb/6fire/comm.h 2012-01-09 22:33:16.900693132 +0100 @@ -3,7 +3,6 @@ * * Author: Torsten Schenk <torsten.schenk(a)zoho.com> * Created: Jan 01, 2011 - * Version: 0.3.0 * Copyright: (C) Torsten Schenk * * This program is free software; you can redistribute it and/or modify diff -Nur a/sound/usb/6fire/common.h b/sound/usb/6fire/common.h --- a/sound/usb/6fire/common.h 2012-01-09 21:55:55.000000000 +0100 +++ b/sound/usb/6fire/common.h 2012-01-09 22:33:24.636949983 +0100 @@ -3,7 +3,6 @@ * * Author: Torsten Schenk <torsten.schenk(a)zoho.com> * Created: Jan 01, 2011 - * Version: 0.3.0 * Copyright: (C) Torsten Schenk * * This program is free software; you can redistribute it and/or modify diff -Nur a/sound/usb/6fire/control.c b/sound/usb/6fire/control.c --- a/sound/usb/6fire/control.c 2012-01-09 21:55:55.000000000 +0100 +++ b/sound/usb/6fire/control.c 2012-01-09 22:33:01.512182294 +0100 @@ -5,7 +5,6 @@ * * Author: Torsten Schenk <torsten.schenk(a)zoho.com> * Created: Jan 01, 2011 - * Version: 0.3.0 * Copyright: (C) Torsten Schenk * * This program is free software; you can redistribute it and/or modify diff -Nur a/sound/usb/6fire/control.h b/sound/usb/6fire/control.h --- a/sound/usb/6fire/control.h 2012-01-09 21:55:55.000000000 +0100 +++ b/sound/usb/6fire/control.h 2012-01-09 22:32:56.088002206 +0100 @@ -3,7 +3,6 @@ * * Author: Torsten Schenk <torsten.schenk(a)zoho.com> * Created: Jan 01, 2011 - * Version: 0.3.0 * Copyright: (C) Torsten Schenk * * This program is free software; you can redistribute it and/or modify diff -Nur a/sound/usb/6fire/firmware.c b/sound/usb/6fire/firmware.c --- a/sound/usb/6fire/firmware.c 2012-01-09 21:55:55.000000000 +0100 +++ b/sound/usb/6fire/firmware.c 2012-01-09 22:33:30.933158868 +0100 @@ -5,7 +5,6 @@ * * Author: Torsten Schenk <torsten.schenk(a)zoho.com> * Created: Jan 01, 2011 - * Version: 0.3.0 * Copyright: (C) Torsten Schenk * * This program is free software; you can redistribute it and/or modify diff -Nur a/sound/usb/6fire/midi.c b/sound/usb/6fire/midi.c --- a/sound/usb/6fire/midi.c 2012-01-09 21:55:55.000000000 +0100 +++ b/sound/usb/6fire/midi.c 2012-01-09 22:35:07.356357430 +0100 @@ -5,7 +5,6 @@ * * Author: Torsten Schenk <torsten.schenk(a)zoho.com> * Created: Jan 01, 2011 - * Version: 0.3.0 * Copyright: (C) Torsten Schenk * * This program is free software; you can redistribute it and/or modify diff -Nur a/sound/usb/6fire/midi.h b/sound/usb/6fire/midi.h --- a/sound/usb/6fire/midi.h 2012-01-09 21:55:55.000000000 +0100 +++ b/sound/usb/6fire/midi.h 2012-01-09 22:33:58.866085888 +0100 @@ -3,7 +3,6 @@ * * Author: Torsten Schenk <torsten.schenk(a)zoho.com> * Created: Jan 01, 2011 - * Version: 0.3.0 * Copyright: (C) Torsten Schenk * * This program is free software; you can redistribute it and/or modify diff -Nur a/sound/usb/6fire/pcm.c b/sound/usb/6fire/pcm.c --- a/sound/usb/6fire/pcm.c 2012-01-09 21:55:55.000000000 +0100 +++ b/sound/usb/6fire/pcm.c 2012-01-09 22:34:04.546274285 +0100 @@ -5,7 +5,6 @@ * * Author: Torsten Schenk <torsten.schenk(a)zoho.com> * Created: Jan 01, 2011 - * Version: 0.3.0 * Copyright: (C) Torsten Schenk * * This program is free software; you can redistribute it and/or modify diff -Nur a/sound/usb/6fire/pcm.h b/sound/usb/6fire/pcm.h --- a/sound/usb/6fire/pcm.h 2012-01-09 21:55:55.000000000 +0100 +++ b/sound/usb/6fire/pcm.h 2012-01-09 22:34:07.798382278 +0100 @@ -3,7 +3,6 @@ * * Author: Torsten Schenk <torsten.schenk(a)zoho.com> * Created: Jan 01, 2011 - * Version: 0.3.0 * Copyright: (C) Torsten Schenk * * This program is free software; you can redistribute it and/or modify

1 0

[alsa-devel] [PATCH v2 0/5] snd-usb-6fire: volume control updates
by Torsten Schenk 22 Feb '12

22 Feb '12

Changes in v2: - removal of log-conversion now in separate patch - tlv added - invalid channels in put and get routines now return -EINVAL - index numbers used for controls instead of stating the index in the .name-string - list terminator now added when list is added, not later in patchset - input volume info now doesn't use negative values any longer; shifts the value to start at zero and adds tlv to indicate the negative value - removed index of single input volume element Thanks, Torsten

2 1

[alsa-devel] [PATCH v2 3/5] snd-usb-6fire: add individual volume control for analog channels
by Torsten Schenk 22 Feb '12

22 Feb '12

Add a stereo volume control for every analog output pair 1/2, 3/4, 5/6. Signed-off-by: Torsten Schenk <torsten.schenk(a)zoho.com> diff -Nur a/sound/usb/6fire/control.c b/sound/usb/6fire/control.c --- a/sound/usb/6fire/control.c 2012-01-17 18:36:14.480508754 +0100 +++ b/sound/usb/6fire/control.c 2012-01-17 19:46:16.278171321 +0100 @@ -7,6 +7,10 @@ * Created: Jan 01, 2011 * Copyright: (C) Torsten Schenk * + * Thanks to: + * - Holger Ruckdeschel: he found out how to control individual channel + * volumes and introduced mute switch + * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by * the Free Software Foundation; either version 2 of the License, or @@ -39,7 +43,7 @@ { 0x22, 0x03, 0x00 }, { 0x20, 0x03, 0x08 }, { 0x22, 0x04, 0x00 }, { 0x20, 0x04, 0x08 }, { 0x22, 0x05, 0x01 }, { 0x20, 0x05, 0x08 }, { 0x22, 0x04, 0x01 }, { 0x12, 0x04, 0x00 }, { 0x12, 0x05, 0x00 }, - { 0x12, 0x0d, 0x78 }, { 0x12, 0x21, 0x82 }, { 0x12, 0x22, 0x80 }, + { 0x12, 0x0d, 0x38 }, { 0x12, 0x21, 0x82 }, { 0x12, 0x22, 0x80 }, { 0x12, 0x23, 0x00 }, { 0x12, 0x06, 0x02 }, { 0x12, 0x03, 0x00 }, { 0x12, 0x02, 0x00 }, { 0x22, 0x03, 0x01 }, { 0 } /* TERMINATING ENTRY */ @@ -56,15 +60,15 @@ DIGITAL_THRU_ONLY_SAMPLERATE = 3 }; -static void usb6fire_control_master_vol_update(struct control_runtime *rt) +static void usb6fire_control_output_vol_update(struct control_runtime *rt) { struct comm_runtime *comm_rt = rt->chip->comm; - if (comm_rt) { - /* set volume */ - comm_rt->write8(comm_rt, 0x12, 0x0f, 180 - rt->master_vol); - /* unmute */ - comm_rt->write8(comm_rt, 0x12, 0x0e, 0x00); - } + int i; + + if (comm_rt) + for (i = 0; i < 6; i++) + comm_rt->write8(comm_rt, 0x12, 0x0f + i, + 180 - rt->output_vol[i]); } static void usb6fire_control_line_phono_update(struct control_runtime *rt) @@ -146,34 +150,56 @@ return -EINVAL; } -static int usb6fire_control_master_vol_info(struct snd_kcontrol *kcontrol, +static int usb6fire_control_output_vol_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; - uinfo->count = 1; + uinfo->count = 2; uinfo->value.integer.min = 0; uinfo->value.integer.max = 180; return 0; } -static int usb6fire_control_master_vol_put(struct snd_kcontrol *kcontrol, +static int usb6fire_control_output_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct control_runtime *rt = snd_kcontrol_chip(kcontrol); + unsigned int ch = kcontrol->private_value; int changed = 0; - if (rt->master_vol != ucontrol->value.integer.value[0]) { - rt->master_vol = ucontrol->value.integer.value[0]; - usb6fire_control_master_vol_update(rt); + + if (ch > 4) { + snd_printk(KERN_ERR PREFIX "Invalid channel in volume control."); + return -EINVAL; + } + + if (rt->output_vol[ch] != ucontrol->value.integer.value[0]) { + rt->output_vol[ch] = ucontrol->value.integer.value[0]; changed = 1; } + if (rt->output_vol[ch + 1] != ucontrol->value.integer.value[1]) { + rt->output_vol[ch + 1] = ucontrol->value.integer.value[1]; + changed = 1; + } + + if (changed) + usb6fire_control_output_vol_update(rt); + return changed; } -static int usb6fire_control_master_vol_get(struct snd_kcontrol *kcontrol, +static int usb6fire_control_output_vol_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct control_runtime *rt = snd_kcontrol_chip(kcontrol); - ucontrol->value.integer.value[0] = rt->master_vol; + unsigned int ch = kcontrol->private_value; + + if (ch > 4) { + snd_printk(KERN_ERR PREFIX "Invalid channel in volume control."); + return -EINVAL; + } + + ucontrol->value.integer.value[0] = rt->output_vol[ch]; + ucontrol->value.integer.value[1] = rt->output_vol[ch + 1]; return 0; } @@ -268,20 +294,49 @@ return 0; } -static struct __devinitdata snd_kcontrol_new elements[] = { +static struct __devinitdata snd_kcontrol_new vol_elements[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Master Playback Volume", - .index = 0, + .name = "Analog Playback Volume", + .index = 1, + .private_value = 0, .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ, - .info = usb6fire_control_master_vol_info, - .get = usb6fire_control_master_vol_get, - .put = usb6fire_control_master_vol_put, + .info = usb6fire_control_output_vol_info, + .get = usb6fire_control_output_vol_get, + .put = usb6fire_control_output_vol_put, .tlv = { .p = tlv_output } }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Analog Playback Volume", + .index = 2, + .private_value = 2, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ, + .info = usb6fire_control_output_vol_info, + .get = usb6fire_control_output_vol_get, + .put = usb6fire_control_output_vol_put, + .tlv = { .p = tlv_output } + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Analog Playback Volume", + .index = 3, + .private_value = 4, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ, + .info = usb6fire_control_output_vol_info, + .get = usb6fire_control_output_vol_get, + .put = usb6fire_control_output_vol_put, + .tlv = { .p = tlv_output } + }, + {} +}; + +static struct __devinitdata snd_kcontrol_new elements[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Line/Phono Capture Route", .index = 0, .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, @@ -310,6 +365,39 @@ {} }; +static int usb6fire_control_add_virtual( + struct control_runtime *rt, + struct snd_card *card, + char *name, + struct snd_kcontrol_new *elems) +{ + int ret; + int i; + struct snd_kcontrol *vmaster = snd_ctl_make_virtual_master(name, NULL); + struct snd_kcontrol *control; + + if (!vmaster) + return -ENOMEM; + ret = snd_ctl_add(card, vmaster); + if (ret < 0) + return ret; + + i = 0; + while (elems[i].name) { + control = snd_ctl_new1(&elems[i], rt); + if (!control) + return -ENOMEM; + ret = snd_ctl_add(card, control); + if (ret < 0) + return ret; + ret = snd_ctl_add_slave(vmaster, control); + if (ret < 0) + return ret; + i++; + } + return 0; +} + int __devinit usb6fire_control_init(struct sfire_chip *chip) { int i; @@ -335,9 +423,17 @@ usb6fire_control_opt_coax_update(rt); usb6fire_control_line_phono_update(rt); - usb6fire_control_master_vol_update(rt); + usb6fire_control_output_vol_update(rt); usb6fire_control_streaming_update(rt); + ret = usb6fire_control_add_virtual(rt, chip->card, + "Master Playback Volume", vol_elements); + if (ret) { + kfree(rt); + snd_printk(KERN_ERR PREFIX "cannot add control.\n"); + return ret; + } + i = 0; while (elements[i].name) { ret = snd_ctl_add(chip->card, snd_ctl_new1(&elements[i], rt)); diff -Nur a/sound/usb/6fire/control.h b/sound/usb/6fire/control.h --- a/sound/usb/6fire/control.h 2012-01-09 22:32:56.088002206 +0100 +++ b/sound/usb/6fire/control.h 2012-01-09 22:36:07.698544501 +0100 @@ -43,7 +43,7 @@ bool line_phono_switch; bool digital_thru_switch; bool usb_streaming; - u8 master_vol; + u8 output_vol[6]; }; int __devinit usb6fire_control_init(struct sfire_chip *chip);

2 1

[alsa-devel] [PATCH] ASoC: imx: move SND_SOC_AC97_BUS selection down to machine driver
by Shawn Guo 22 Feb '12

22 Feb '12

SND_SOC_AC97_BUS is selected to enable the AC97 support in soc-core. Rather than selecting the option under SND_IMX_SOC, it's better to leave the selection to individual machine driver which knows if AC97 support is needed or not. Signed-off-by: Shawn Guo <shawn.guo(a)linaro.org> --- sound/soc/imx/Kconfig | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/imx/Kconfig b/sound/soc/imx/Kconfig index e89139f..7c4aa42 100644 --- a/sound/soc/imx/Kconfig +++ b/sound/soc/imx/Kconfig @@ -1,7 +1,6 @@ menuconfig SND_IMX_SOC tristate "SoC Audio for Freescale i.MX CPUs" depends on ARCH_MXC - select SND_SOC_AC97_BUS help Say Y or M if you want to add support for codecs attached to the i.MX SSI interface. @@ -37,6 +36,7 @@ config SND_SOC_MX27VIS_AIC32X4 config SND_SOC_PHYCORE_AC97 tristate "SoC Audio support for Phytec phyCORE (and phyCARD) boards" depends on MACH_PCM043 || MACH_PCA100 + select SND_SOC_AC97_BUS select SND_SOC_WM9712 select SND_MXC_SOC_FIQ help -- 1.7.5.4

2 1

[alsa-devel] [PATCH v2 5/5] snd-usb-6fire: add analog input volume control
by Torsten Schenk 22 Feb '12

22 Feb '12

Add a stereo volume control for analog input channel pair 1/2. Signed-off-by: Torsten Schenk <torsten.schenk(a)zoho.com> diff -Nur a/sound/usb/6fire/control.c b/sound/usb/6fire/control.c --- a/sound/usb/6fire/control.c 2012-01-30 11:57:10.557379497 +0100 +++ b/sound/usb/6fire/control.c 2012-01-30 11:56:47.260578549 +0100 @@ -55,6 +55,7 @@ static const u16 rates_6fire_vh[] = {0x11, 0x11, 0x10, 0x10, 0x00, 0x00}; static DECLARE_TLV_DB_MINMAX(tlv_output, -9000, 0); +static DECLARE_TLV_DB_MINMAX(tlv_input, -1500, 1500); enum { DIGITAL_THRU_ONLY_SAMPLERATE = 3 @@ -79,6 +80,17 @@ comm_rt->write8(comm_rt, 0x12, 0x0e, ~rt->output_mute); } +static void usb6fire_control_input_vol_update(struct control_runtime *rt) +{ + struct comm_runtime *comm_rt = rt->chip->comm; + int i; + + if (comm_rt) + for (i = 0; i < 2; i++) + comm_rt->write8(comm_rt, 0x12, 0x1c + i, + rt->input_vol[i] & 0x3f); +} + static void usb6fire_control_line_phono_update(struct control_runtime *rt) { struct comm_runtime *comm_rt = rt->chip->comm; @@ -256,6 +268,48 @@ return 0; } +static int usb6fire_control_input_vol_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 2; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 30; + return 0; +} + +static int usb6fire_control_input_vol_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct control_runtime *rt = snd_kcontrol_chip(kcontrol); + int changed = 0; + + if (rt->input_vol[0] != ucontrol->value.integer.value[0]) { + rt->input_vol[0] = ucontrol->value.integer.value[0] - 15; + changed = 1; + } + if (rt->input_vol[1] != ucontrol->value.integer.value[1]) { + rt->input_vol[1] = ucontrol->value.integer.value[1] - 15; + changed = 1; + } + + if (changed) + usb6fire_control_input_vol_update(rt); + + return changed; +} + +static int usb6fire_control_input_vol_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct control_runtime *rt = snd_kcontrol_chip(kcontrol); + + ucontrol->value.integer.value[0] = rt->input_vol[0] + 15; + ucontrol->value.integer.value[1] = rt->input_vol[1] + 15; + + return 0; +} + static int usb6fire_control_line_phono_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { @@ -449,6 +503,17 @@ .get = usb6fire_control_digital_thru_get, .put = usb6fire_control_digital_thru_put }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Analog Capture Volume", + .index = 0, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ, + .info = usb6fire_control_input_vol_info, + .get = usb6fire_control_input_vol_get, + .put = usb6fire_control_input_vol_put, + .tlv = { .p = tlv_input } + }, {} }; @@ -512,6 +577,7 @@ usb6fire_control_line_phono_update(rt); usb6fire_control_output_vol_update(rt); usb6fire_control_output_mute_update(rt); + usb6fire_control_input_vol_update(rt); usb6fire_control_streaming_update(rt); ret = usb6fire_control_add_virtual(rt, chip->card, diff -Nur a/sound/usb/6fire/control.h b/sound/usb/6fire/control.h --- a/sound/usb/6fire/control.h 2012-01-11 19:54:50.651190007 +0100 +++ b/sound/usb/6fire/control.h 2012-02-05 12:28:10.314770911 +0100 @@ -45,6 +45,7 @@ bool usb_streaming; u8 output_vol[6]; u8 output_mute; + s8 input_vol[2]; }; int __devinit usb6fire_control_init(struct sfire_chip *chip);

1 0

[alsa-devel] [PATCH v2 4/5] snd-usb-6fire: add mute control for analog outputs
by Torsten Schenk 22 Feb '12

22 Feb '12

Add a mute control for every analog output channel. Signed-off-by: Torsten Schenk <torsten.schenk(a)zoho.com> diff -Nur a/sound/usb/6fire/control.c b/sound/usb/6fire/control.c --- a/sound/usb/6fire/control.c 2012-01-17 19:46:16.278171321 +0100 +++ b/sound/usb/6fire/control.c 2012-01-30 11:57:10.557379497 +0100 @@ -71,6 +71,14 @@ 180 - rt->output_vol[i]); } +static void usb6fire_control_output_mute_update(struct control_runtime *rt) +{ + struct comm_runtime *comm_rt = rt->chip->comm; + + if (comm_rt) + comm_rt->write8(comm_rt, 0x12, 0x0e, ~rt->output_mute); +} + static void usb6fire_control_line_phono_update(struct control_runtime *rt) { struct comm_runtime *comm_rt = rt->chip->comm; @@ -203,6 +211,51 @@ return 0; } +static int usb6fire_control_output_mute_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct control_runtime *rt = snd_kcontrol_chip(kcontrol); + unsigned int ch = kcontrol->private_value; + u8 old = rt->output_mute; + u8 value = 0; + + if (ch > 4) { + snd_printk(KERN_ERR PREFIX "Invalid channel in volume control."); + return -EINVAL; + } + + rt->output_mute &= ~(3 << ch); + if (ucontrol->value.integer.value[0]) + value |= 1; + if (ucontrol->value.integer.value[1]) + value |= 2; + rt->output_mute |= value << ch; + + if (rt->output_mute != old) + usb6fire_control_output_mute_update(rt); + + return rt->output_mute != old; +} + +static int usb6fire_control_output_mute_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct control_runtime *rt = snd_kcontrol_chip(kcontrol); + unsigned int ch = kcontrol->private_value; + u8 value = rt->output_mute >> ch; + + if (ch > 4) { + snd_printk(KERN_ERR PREFIX "Invalid channel in volume control."); + return -EINVAL; + } + + ucontrol->value.integer.value[0] = 1 & value; + value >>= 1; + ucontrol->value.integer.value[1] = 1 & value; + + return 0; +} + static int usb6fire_control_line_phono_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { @@ -334,6 +387,40 @@ {} }; +static struct __devinitdata snd_kcontrol_new mute_elements[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Analog Playback Switch", + .index = 1, + .private_value = 0, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_ctl_boolean_stereo_info, + .get = usb6fire_control_output_mute_get, + .put = usb6fire_control_output_mute_put, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Analog Playback Switch", + .index = 2, + .private_value = 2, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_ctl_boolean_stereo_info, + .get = usb6fire_control_output_mute_get, + .put = usb6fire_control_output_mute_put, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Analog Playback Switch", + .index = 3, + .private_value = 4, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_ctl_boolean_stereo_info, + .get = usb6fire_control_output_mute_get, + .put = usb6fire_control_output_mute_put, + }, + {} +}; + static struct __devinitdata snd_kcontrol_new elements[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -424,13 +511,21 @@ usb6fire_control_opt_coax_update(rt); usb6fire_control_line_phono_update(rt); usb6fire_control_output_vol_update(rt); + usb6fire_control_output_mute_update(rt); usb6fire_control_streaming_update(rt); ret = usb6fire_control_add_virtual(rt, chip->card, "Master Playback Volume", vol_elements); if (ret) { + snd_printk(KERN_ERR PREFIX "cannot add control.\n"); kfree(rt); + return ret; + } + ret = usb6fire_control_add_virtual(rt, chip->card, + "Master Playback Switch", mute_elements); + if (ret) { snd_printk(KERN_ERR PREFIX "cannot add control.\n"); + kfree(rt); return ret; } diff -Nur a/sound/usb/6fire/control.h b/sound/usb/6fire/control.h --- a/sound/usb/6fire/control.h 2012-01-09 22:36:07.698544501 +0100 +++ b/sound/usb/6fire/control.h 2012-01-11 19:54:50.651190007 +0100 @@ -44,6 +44,7 @@ bool digital_thru_switch; bool usb_streaming; u8 output_vol[6]; + u8 output_mute; }; int __devinit usb6fire_control_init(struct sfire_chip *chip);

1 0

[alsa-devel] [PATCH v2 2/5] snd-usb-6fire: add tlv to controls
by Torsten Schenk 22 Feb '12

22 Feb '12

Remove the soft log-conversion and add a dB scale according to the DAC documentation instead. Signed-off-by: Torsten Schenk <torsten.schenk(a)zoho.com> diff -Nur a/sound/usb/6fire/control.c b/sound/usb/6fire/control.c --- a/sound/usb/6fire/control.c 2012-01-09 22:33:01.512182294 +0100 +++ b/sound/usb/6fire/control.c 2012-01-17 18:36:14.480508754 +0100 @@ -15,6 +15,7 @@ #include <linux/interrupt.h> #include <sound/control.h> +#include <sound/tlv.h> #include "control.h" #include "comm.h" @@ -24,26 +25,6 @@ static char *line_phono_texts[2] = { "Line", "Phono" }; /* - * calculated with $value\[i\] = 128 \cdot sqrt[3]{\frac{i}{128}}$ - * this is done because the linear values cause rapid degredation - * of volume in the uppermost region. - */ -static const u8 log_volume_table[128] = { - 0x00, 0x19, 0x20, 0x24, 0x28, 0x2b, 0x2e, 0x30, 0x32, 0x34, - 0x36, 0x38, 0x3a, 0x3b, 0x3d, 0x3e, 0x40, 0x41, 0x42, 0x43, - 0x44, 0x46, 0x47, 0x48, 0x49, 0x4a, 0x4b, 0x4c, 0x4d, 0x4e, - 0x4e, 0x4f, 0x50, 0x51, 0x52, 0x53, 0x53, 0x54, 0x55, 0x56, - 0x56, 0x57, 0x58, 0x58, 0x59, 0x5a, 0x5b, 0x5b, 0x5c, 0x5c, - 0x5d, 0x5e, 0x5e, 0x5f, 0x60, 0x60, 0x61, 0x61, 0x62, 0x62, - 0x63, 0x63, 0x64, 0x65, 0x65, 0x66, 0x66, 0x67, 0x67, 0x68, - 0x68, 0x69, 0x69, 0x6a, 0x6a, 0x6b, 0x6b, 0x6c, 0x6c, 0x6c, - 0x6d, 0x6d, 0x6e, 0x6e, 0x6f, 0x6f, 0x70, 0x70, 0x70, 0x71, - 0x71, 0x72, 0x72, 0x73, 0x73, 0x73, 0x74, 0x74, 0x75, 0x75, - 0x75, 0x76, 0x76, 0x77, 0x77, 0x77, 0x78, 0x78, 0x78, 0x79, - 0x79, 0x7a, 0x7a, 0x7a, 0x7b, 0x7b, 0x7b, 0x7c, 0x7c, 0x7c, - 0x7d, 0x7d, 0x7d, 0x7e, 0x7e, 0x7e, 0x7f, 0x7f }; - -/* * data that needs to be sent to device. sets up card internal stuff. * values dumped from windows driver and filtered by trial'n'error. */ @@ -69,6 +50,8 @@ static const u16 rates_6fire_vl[] = {0x00, 0x01, 0x00, 0x01, 0x00, 0x01}; static const u16 rates_6fire_vh[] = {0x11, 0x11, 0x10, 0x10, 0x00, 0x00}; +static DECLARE_TLV_DB_MINMAX(tlv_output, -9000, 0); + enum { DIGITAL_THRU_ONLY_SAMPLERATE = 3 }; @@ -78,8 +61,7 @@ struct comm_runtime *comm_rt = rt->chip->comm; if (comm_rt) { /* set volume */ - comm_rt->write8(comm_rt, 0x12, 0x0f, 0x7f - - log_volume_table[rt->master_vol]); + comm_rt->write8(comm_rt, 0x12, 0x0f, 180 - rt->master_vol); /* unmute */ comm_rt->write8(comm_rt, 0x12, 0x0e, 0x00); } @@ -170,7 +152,7 @@ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; uinfo->count = 1; uinfo->value.integer.min = 0; - uinfo->value.integer.max = 127; + uinfo->value.integer.max = 180; return 0; } @@ -291,10 +273,12 @@ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Volume", .index = 0, - .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ, .info = usb6fire_control_master_vol_info, .get = usb6fire_control_master_vol_get, - .put = usb6fire_control_master_vol_put + .put = usb6fire_control_master_vol_put, + .tlv = { .p = tlv_output } }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER,

1 0

[alsa-devel] [PATCH v2 1/5] snd-usb-6fire: remove driver version information
by Torsten Schenk 22 Feb '12

22 Feb '12

Remove unused driver version information from the individual files. Signed-off-by: Torsten Schenk <torsten.schenk(a)zoho.com> diff -Nur a/sound/usb/6fire/chip.c b/sound/usb/6fire/chip.c --- a/sound/usb/6fire/chip.c 2012-01-09 21:55:55.863579333 +0100 +++ b/sound/usb/6fire/chip.c 2012-01-09 22:33:49.857786986 +0100 @@ -5,7 +5,6 @@ * * Author: Torsten Schenk <torsten.schenk(a)zoho.com> * Created: Jan 01, 2011 - * Version: 0.3.0 * Copyright: (C) Torsten Schenk * * This program is free software; you can redistribute it and/or modify @@ -29,7 +28,7 @@ #include <sound/initval.h> MODULE_AUTHOR("Torsten Schenk <torsten.schenk(a)zoho.com>"); -MODULE_DESCRIPTION("TerraTec DMX 6Fire USB audio driver, version 0.3.0"); +MODULE_DESCRIPTION("TerraTec DMX 6Fire USB audio driver"); MODULE_LICENSE("GPL v2"); MODULE_SUPPORTED_DEVICE("{{TerraTec, DMX 6Fire USB}}"); diff -Nur a/sound/usb/6fire/chip.h b/sound/usb/6fire/chip.h --- a/sound/usb/6fire/chip.h 2012-01-09 21:55:55.000000000 +0100 +++ b/sound/usb/6fire/chip.h 2012-01-09 22:33:38.469409108 +0100 @@ -3,7 +3,6 @@ * * Author: Torsten Schenk <torsten.schenk(a)zoho.com> * Created: Jan 01, 2011 - * Version: 0.3.0 * Copyright: (C) Torsten Schenk * * This program is free software; you can redistribute it and/or modify diff -Nur a/sound/usb/6fire/comm.c b/sound/usb/6fire/comm.c --- a/sound/usb/6fire/comm.c 2012-01-09 21:55:55.000000000 +0100 +++ b/sound/usb/6fire/comm.c 2012-01-09 22:33:11.620517902 +0100 @@ -5,7 +5,6 @@ * * Author: Torsten Schenk <torsten.schenk(a)zoho.com> * Created: Jan 01, 2011 - * Version: 0.3.0 * Copyright: (C) Torsten Schenk * * This program is free software; you can redistribute it and/or modify diff -Nur a/sound/usb/6fire/comm.h b/sound/usb/6fire/comm.h --- a/sound/usb/6fire/comm.h 2012-01-09 21:55:55.000000000 +0100 +++ b/sound/usb/6fire/comm.h 2012-01-09 22:33:16.900693132 +0100 @@ -3,7 +3,6 @@ * * Author: Torsten Schenk <torsten.schenk(a)zoho.com> * Created: Jan 01, 2011 - * Version: 0.3.0 * Copyright: (C) Torsten Schenk * * This program is free software; you can redistribute it and/or modify diff -Nur a/sound/usb/6fire/common.h b/sound/usb/6fire/common.h --- a/sound/usb/6fire/common.h 2012-01-09 21:55:55.000000000 +0100 +++ b/sound/usb/6fire/common.h 2012-01-09 22:33:24.636949983 +0100 @@ -3,7 +3,6 @@ * * Author: Torsten Schenk <torsten.schenk(a)zoho.com> * Created: Jan 01, 2011 - * Version: 0.3.0 * Copyright: (C) Torsten Schenk * * This program is free software; you can redistribute it and/or modify diff -Nur a/sound/usb/6fire/control.c b/sound/usb/6fire/control.c --- a/sound/usb/6fire/control.c 2012-01-09 21:55:55.000000000 +0100 +++ b/sound/usb/6fire/control.c 2012-01-09 22:33:01.512182294 +0100 @@ -5,7 +5,6 @@ * * Author: Torsten Schenk <torsten.schenk(a)zoho.com> * Created: Jan 01, 2011 - * Version: 0.3.0 * Copyright: (C) Torsten Schenk * * This program is free software; you can redistribute it and/or modify diff -Nur a/sound/usb/6fire/control.h b/sound/usb/6fire/control.h --- a/sound/usb/6fire/control.h 2012-01-09 21:55:55.000000000 +0100 +++ b/sound/usb/6fire/control.h 2012-01-09 22:32:56.088002206 +0100 @@ -3,7 +3,6 @@ * * Author: Torsten Schenk <torsten.schenk(a)zoho.com> * Created: Jan 01, 2011 - * Version: 0.3.0 * Copyright: (C) Torsten Schenk * * This program is free software; you can redistribute it and/or modify diff -Nur a/sound/usb/6fire/firmware.c b/sound/usb/6fire/firmware.c --- a/sound/usb/6fire/firmware.c 2012-01-09 21:55:55.000000000 +0100 +++ b/sound/usb/6fire/firmware.c 2012-01-09 22:33:30.933158868 +0100 @@ -5,7 +5,6 @@ * * Author: Torsten Schenk <torsten.schenk(a)zoho.com> * Created: Jan 01, 2011 - * Version: 0.3.0 * Copyright: (C) Torsten Schenk * * This program is free software; you can redistribute it and/or modify diff -Nur a/sound/usb/6fire/midi.c b/sound/usb/6fire/midi.c --- a/sound/usb/6fire/midi.c 2012-01-09 21:55:55.000000000 +0100 +++ b/sound/usb/6fire/midi.c 2012-01-09 22:35:07.356357430 +0100 @@ -5,7 +5,6 @@ * * Author: Torsten Schenk <torsten.schenk(a)zoho.com> * Created: Jan 01, 2011 - * Version: 0.3.0 * Copyright: (C) Torsten Schenk * * This program is free software; you can redistribute it and/or modify diff -Nur a/sound/usb/6fire/midi.h b/sound/usb/6fire/midi.h --- a/sound/usb/6fire/midi.h 2012-01-09 21:55:55.000000000 +0100 +++ b/sound/usb/6fire/midi.h 2012-01-09 22:33:58.866085888 +0100 @@ -3,7 +3,6 @@ * * Author: Torsten Schenk <torsten.schenk(a)zoho.com> * Created: Jan 01, 2011 - * Version: 0.3.0 * Copyright: (C) Torsten Schenk * * This program is free software; you can redistribute it and/or modify diff -Nur a/sound/usb/6fire/pcm.c b/sound/usb/6fire/pcm.c --- a/sound/usb/6fire/pcm.c 2012-01-09 21:55:55.000000000 +0100 +++ b/sound/usb/6fire/pcm.c 2012-01-09 22:34:04.546274285 +0100 @@ -5,7 +5,6 @@ * * Author: Torsten Schenk <torsten.schenk(a)zoho.com> * Created: Jan 01, 2011 - * Version: 0.3.0 * Copyright: (C) Torsten Schenk * * This program is free software; you can redistribute it and/or modify diff -Nur a/sound/usb/6fire/pcm.h b/sound/usb/6fire/pcm.h --- a/sound/usb/6fire/pcm.h 2012-01-09 21:55:55.000000000 +0100 +++ b/sound/usb/6fire/pcm.h 2012-01-09 22:34:07.798382278 +0100 @@ -3,7 +3,6 @@ * * Author: Torsten Schenk <torsten.schenk(a)zoho.com> * Created: Jan 01, 2011 - * Version: 0.3.0 * Copyright: (C) Torsten Schenk * * This program is free software; you can redistribute it and/or modify

1 0

Re: [alsa-devel] [RFC 00/10] ASoC: Introduce dmaengine pcm helper functions
by Shawn Guo 22 Feb '12

22 Feb '12

Nice consolidation ... On Tue, Feb 21, 2012 at 05:47:54PM +0100, Lars-Peter Clausen wrote: > This patch series introduces a new set of helper functions for dmaengine based > PCM drivers. Currently we have 6 different dmaengine based PCM drivers which are > all more or less similar in structure, but all have also individual requirements > which can't easily be generalized. So that's why this series does not introduce > a generic dmaengine PCM driver but rather a set of helper functions which can be > used to implement the common bits between such dmaengine based PCM drivers. > > The helper functions only provide infrastructure code for dmaengines which > provide thed evice_prep_dma_cyclic. While it is possible to add support for > dmaengine, which only provide device_prep_slave_sg, it is in my opinion a better > idea to either implement the missing functionality in the dmaengine driver or > provide a emulation layer for cyclic transfer within the dmaengine framework > itself. This has the advantage that it is also available outside of ALSA. > > This series converts the three PCM drivers which already use the circular > dmaengine prepare function. Since I don't have access to hardware using the > converted drivers the patches have only been compile tested. But the helper > functions introduced in this series have been runtime tested with a platform > which is not upstream yet. > > The first three patches are just cleanup patches which remove unused functions > and struct fields and can probably applied without waiting for the rest of the > series. The next two patches restructure the iMX and MXS drivers to request > their DMA channel in the PCM open callback instead of the hwparams callback. > This is done to make their structure more similar to what it will be when using > the helper functions and make the actual patch converting them to the new > dmaengine PCM helper functions less intrusive. The next patch introduces the > helper functions and the last three patches respectively convert the imx, mxs > and ep39xx dmaengine based PCM drivers to use the new set of helper functions. > > The set of helper functions consists of a open, a close, a trigger and a pointer > callback and also a function to convert hwparams to a dma_slave_config. The > trigger and pointer callbacks can usually be used directly for the pcm_ops > callbacks. The open callback wants to be called from a driver specific open > function which may do some driver specific setup tasks and provide the helper > open callback with a dma filter function. If the driver specific open functions > allocates resources they need to be freed in a driver specific close function > which has to call the helper close callback. If not the helper close callback > can be assigned directly to the pcm_ops struct. > > If a driver needs to keep additional driver specific data around it can be done > by using snd_dmaengine_pcm_{get,set}_data. Normally a driver should not require > to keep additional data around, but all of the converted drivers need this, > because of this horrible abusive pattern where the dma channels private field > gets assigned configuration data in the filter callback. We should hopefully be > able to get rid of this pattern with Guennadi Liakhovetski's patches[1], which > add a slave configuration parameter to dma_request_channel, and with it the need > for storing extra private data. > > While the DMA helper functions only use generic ALSA functions and nothing ASoC > specific I've placed the, under ASoC for now since the only users are ASoC > driver. > > - Lars > > [1] https://lkml.org/lkml/2012/2/1/227 > > Lars-Peter Clausen (10): > ASoC: imx-pcm: Remove empty prepare callback > ASoC: imx-pcm: Remove unused fields from imx_pcm_runtime_data struct > ASoC: mxs-pcm: Remove unused fields from struct mxs_pcm_runtime_data > ASoC: imx-ssi: Set dma data early > ASoC: imx-pcm: Request DMA channel early > ASoC: mxs-pxm: Request DMA channel early s/mxs-pxm/mxs-pcm > ASoC: Add dmaengine PCM helper functions > ASoC: imx-dma: Use dmaengine PCM helper functions s/imx-dma/imx-pcm > ASoC: mxs-pcm: Use dmaengine PCM helper functions > ASoC: ep93xx-pcm: Use dmaengine PCM helper functions > > include/sound/dmaengine_pcm.h | 49 +++++++ > sound/soc/Kconfig | 3 + > sound/soc/Makefile | 3 + > sound/soc/ep93xx/Kconfig | 1 + > sound/soc/ep93xx/ep93xx-pcm.c | 148 +++----------------- > sound/soc/imx/Kconfig | 1 + > sound/soc/imx/imx-pcm-dma-mx2.c | 216 ++++------------------------- > sound/soc/imx/imx-ssi.c | 28 +++- > sound/soc/mxs/Kconfig | 1 + > sound/soc/mxs/mxs-pcm.c | 157 ++++------------------ > sound/soc/mxs/mxs-pcm.h | 16 -- > sound/soc/soc-dmaengine-pcm.c | 287 +++++++++++++++++++++++++++++++++++++++ > 12 files changed, 443 insertions(+), 467 deletions(-) > create mode 100644 include/sound/dmaengine_pcm.h > create mode 100644 sound/soc/soc-dmaengine-pcm.c > I tested it on both MXS (i.MX28) and IMX (i.MX51), it works fine, so Tested-by: Shawn Guo <shawn.guo(a)linaro.org> However, when I play the series on today's linux-next, I found a patch from Russell (Cc-ed) below doing something overlapping with the work here. commit 4c815a835e9b0231028e9cc303406b381bfff7a2 Author: Russell King <rmk+kernel(a)arm.linux.org.uk> Date: Wed Jan 18 13:01:26 2012 +0000 ALSA: ASoC: add generic slave-only DMA engine support Add a generic DMA engine platform driver to ASoC. This allows a range of DMA engines to be used to transfer data to a CPUs audio interface without having to rewrite this code many times. This driver only supports DMA_SLAVE channels, not DMA_CYCLIC channels. DMA_CYCLIC channels could be added at a later date. Signed-off-by: Russell King <rmk+kernel(a)arm.linux.org.uk> I did not notice this patch on any list yet, so am not sure about Russell's plan about it. But it seems this series does not reach alsa-devel list either. (forgot?) -- Regards, Shawn

4 4

[alsa-devel] [PATCH] ASoC: pandora: switch clock back to internal on stop
by Grazvydas Ignotas 22 Feb '12

22 Feb '12

For some reason, OMAP doesn't enter lower power states with functional clock (CLKS) source set to external, so switch it back to internal when done playing. Signed-off-by: Grazvydas Ignotas <notasas(a)gmail.com> --- sound/soc/omap/omap3pandora.c | 22 ++++++++++++++++++++++ 1 files changed, 22 insertions(+), 0 deletions(-) diff --git a/sound/soc/omap/omap3pandora.c b/sound/soc/omap/omap3pandora.c index 07794bd..1be26df 100644 --- a/sound/soc/omap/omap3pandora.c +++ b/sound/soc/omap/omap3pandora.c @@ -77,6 +77,27 @@ static int omap3pandora_hw_params(struct snd_pcm_substream *substream, return 0; } +static int omap3pandora_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + int ret; + + /* + * For some reason, if we don't set clock back to internal, OMAP power + * saving features don't work properly (per power domain doesn't enter + * lower power states). Switch it back until better solution is found. + */ + ret = snd_soc_dai_set_sysclk(cpu_dai, OMAP_MCBSP_SYSCLK_CLKS_FCLK, + 0, SND_SOC_CLOCK_IN); + if (ret < 0) { + pr_err(PREFIX "can't set cpu system clock\n"); + return ret; + } + + return 0; +} + static int omap3pandora_dac_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *k, int event) { @@ -201,6 +222,7 @@ static int omap3pandora_in_init(struct snd_soc_pcm_runtime *rtd) static struct snd_soc_ops omap3pandora_ops = { .hw_params = omap3pandora_hw_params, + .hw_free = omap3pandora_hw_free, }; /* Digital audio interface glue - connects codec <--> CPU */ -- 1.7.0.4

4 7

Re: [alsa-devel] [PATCH v3 0/3] MFD: twl6040: Conversion to i2c driver
by Liam Girdwood 22 Feb '12

22 Feb '12

On Tue, 2012-02-21 at 12:05 +0100, Samuel Ortiz wrote: > Hi Peter, > > On Tue, Feb 21, 2012 at 12:33:07PM +0200, Peter Ujfalusi wrote: > > > I'm not sure hwo we could handle that properly. Either by letting Tony > > > carrying this patchset, or by sending me the panda patch that adds those > > > structures. As you prefer. > > > > If Liam would take this series that would be probably the way forward. > > If he does not have objections. > > Liam: I can create a branch for you to pull from as soon as all other > > patch receives the needed acks... > > > > Samuel: if Liam agrees is it possible for you to drop the twl6040 > > related patches from your for-next branch? > That would make sense to me, yes. > Liam, if you're ok with taking it, please add my: > > Acked-by: Samuel Ortiz <sameo(a)linux.intel.com> > Ok, sounds fine by me. Tony is this fine wrt the OMAP board file updates ? Thanks Liam

3 2

[alsa-devel] ASoC updates for 3.3
by Mark Brown 21 Feb '12

21 Feb '12

The following changes since commit 1987877d869027ab63dc9df515e11f19279a8091: ASoC: fsi: fixup fsi_pointer() calculation method (2012-02-09 10:50:53 +0000) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-3.3 for you to fetch changes up to e555cf363167f09efae96d32a363e24c4de16b7b: ASoC: ak4642: fixup HeadPhone L/R dapm settings (2012-02-21 10:00:24 +0000) ---------------------------------------------------------------- A couple of small, driver specific fixes - nothing too exciting going on. ---------------------------------------------------------------- Kuninori Morimoto (1): ASoC: ak4642: fixup HeadPhone L/R dapm settings Mark Brown (1): ASoC: wm8962: Fix sidetone enumeration texts sound/soc/codecs/ak4642.c | 31 ++++++++++++++++--------------- sound/soc/codecs/wm8962.c | 2 +- 2 files changed, 17 insertions(+), 16 deletions(-)

2 1

[alsa-devel] [PATCH v3 0/3] MFD: twl6040: Conversion to i2c driver
by Peter Ujfalusi 21 Feb '12

21 Feb '12

Hello, Changes since v2: - soc/codec/Kconfig: make twl6040 depend on I2C - Regulator related patches has been removed (to be sent as separate series) ------ Intro mail from the original series: This series will convert the twl6040 MFD driver to an i2c driver. Compared to older twl4030/5030/TPS the twl6040 is a standalone audio IC. It is better if the twl6040-core (and all of it's child devices) does not depend on the twl-core since it has nothing to with it. With this conversion the dependency on twl can be dropped from the twl6040 driver stack (core, vibra, audio). Between the first and second patch the audio will not probe on OMAP4, but I felt it is better this way at least for the first RFC series to not have too big change within one patch. The two patch can be squashed together later if no objections. Regards, Peter --- Peter Ujfalusi (3): MFD: twl-core: Detach twl6040 from the pmic mfd driver MFD: twl6040: Convert to i2c driver, and separate it from twl core ASoC: twl6040: Remove dependency on twl4030 from Kconfig arch/arm/mach-omap2/board-4430sdp.c | 12 ++-- arch/arm/mach-omap2/board-generic.c | 2 +- arch/arm/mach-omap2/board-omap4panda.c | 13 ++-- arch/arm/mach-omap2/twl-common.c | 37 +++++++++-- arch/arm/mach-omap2/twl-common.h | 10 +-- drivers/input/misc/Kconfig | 1 - drivers/input/misc/twl6040-vibra.c | 4 +- drivers/mfd/Kconfig | 2 +- drivers/mfd/twl-core.c | 58 +++++++++------- drivers/mfd/twl6040-core.c | 114 +++++++++++++++++++------------- include/linux/i2c/twl.h | 12 ---- include/linux/mfd/twl6040.h | 27 ++++++++ sound/soc/codecs/Kconfig | 2 +- sound/soc/codecs/twl6040.c | 3 +- 14 files changed, 182 insertions(+), 115 deletions(-) -- 1.7.8.4

3 6

[alsa-devel] [PATCH] ASoC: wm8994: Actively discharge idle MICBIAS with jack detect
by Mark Brown 21 Feb '12

21 Feb '12

This minimises the chance of any external capacitors that are fitted being discharged into headphones as they insert. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 9 +++++++++ 1 files changed, 9 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index ee102d6..c20f376 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3201,6 +3201,9 @@ static irqreturn_t wm1811_jackdet_irq(int irq, void *data) snd_soc_jack_report(wm8994->micdet[0].jack, SND_JACK_MECHANICAL, SND_JACK_MECHANICAL); + snd_soc_update_bits(codec, WM8958_MICBIAS2, + WM8958_MICB2_DISCH, 0); + /* * Start off measument of microphone impedence to find * out what's actually there. @@ -3212,6 +3215,9 @@ static irqreturn_t wm1811_jackdet_irq(int irq, void *data) } else { dev_dbg(codec->dev, "Jack not detected\n"); + snd_soc_update_bits(codec, WM8958_MICBIAS2, + WM8958_MICB2_DISCH, WM8958_MICB2_DISCH); + snd_soc_jack_report(wm8994->micdet[0].jack, 0, SND_JACK_MECHANICAL | SND_JACK_HEADSET | wm8994->btn_mask); @@ -3297,6 +3303,9 @@ int wm8958_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack, * otherwise jump straight to microphone detection. */ if (wm8994->jackdet) { + snd_soc_update_bits(codec, WM8958_MICBIAS2, + WM8958_MICB2_DISCH, + WM8958_MICB2_DISCH); snd_soc_update_bits(codec, WM8994_LDO_1, WM8994_LDO1_DISCH, 0); wm1811_jackdet_set_mode(codec, -- 1.7.9.1

1 0

[alsa-devel] [PATCH] ASoC: wm8994: Enable headphone startup mode 1 for WM1811 and WM8958
by Mark Brown 21 Feb '12

21 Feb '12

The latest recommendation for optimal performance. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 19de7d5..aa16477 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3478,11 +3478,13 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) case WM8958: wm8994->hubs.dcs_readback_mode = 1; + wm8994->hubs.hp_startup_mode = 1; break; case WM1811: wm8994->hubs.dcs_readback_mode = 2; wm8994->hubs.no_series_update = 1; + wm8994->hubs.hp_startup_mode = 1; switch (wm8994->revision) { case 0: -- 1.7.9

1 0

[alsa-devel] [PATCH] MFD/Input/ASoC: twl6040 kconfig fixes
by Peter Ujfalusi 21 Feb '12

21 Feb '12

To avoid breaking x86_64 allmodconfig build the Kconfig files need to be changed. Change the dependency graph of the twl6040 stack. Stephen Rothwell wrote: > Hi Samuel, > > After merging the mfd tree, today's linux-next build (x86_64 allmodconfig) > failed like this: > > drivers/built-in.o: In function `twl6040_driver_init': > twl6040-core.c:(.init.text+0xb147): undefined reference to `i2c_register_driver' > drivers/built-in.o: In function `twl6040_probe': > twl6040-core.c:(.devinit.text+0x7f22): undefined reference to `regmap_init_i2c' > drivers/built-in.o: In function `twl6040_driver_exit': > twl6040-core.c:(.exit.text+0x67a): undefined reference to `i2c_del_driver' > > > Caused by commit ded0df851077 ("mfd: Convert twl6040 to an i2c driver, > and separate it from twl core"). Reported-by: Stephen Rothwell <sfr(a)canb.auug.org.au> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- Hi Stephen, Samuel, This patch will fix the build failure with x86_64 allmodconfig, and it will still work on the target platform. As a note: this series might not be going via the mfd tree at the end since some of the dependencies are going via sound tree. We are discussing the way forward. Regards, Peter drivers/input/misc/Kconfig | 2 +- drivers/mfd/Kconfig | 11 +++++++++-- sound/soc/codecs/Kconfig | 3 +-- 3 files changed, 11 insertions(+), 5 deletions(-) diff --git a/drivers/input/misc/Kconfig b/drivers/input/misc/Kconfig index 6a9ee24..9450d4c 100644 --- a/drivers/input/misc/Kconfig +++ b/drivers/input/misc/Kconfig @@ -368,7 +368,7 @@ config INPUT_TWL4030_VIBRA config INPUT_TWL6040_VIBRA tristate "Support for TWL6040 Vibrator" - select TWL6040_CORE + depends on TWL6040_CORE select INPUT_FF_MEMLESS help This option enables support for TWL6040 Vibrator Driver. diff --git a/drivers/mfd/Kconfig b/drivers/mfd/Kconfig index 0e48997..ea7c36f 100644 --- a/drivers/mfd/Kconfig +++ b/drivers/mfd/Kconfig @@ -266,10 +266,17 @@ config TWL6030_PWM This is used to control charging LED brightness. config TWL6040_CORE - bool - depends on I2C && GENERIC_HARDIRQS + bool "Support for TWL6040 audio codec" + depends on I2C=y && GENERIC_HARDIRQS select MFD_CORE + select REGMAP_I2C default n + help + Say yes here if you want support for Texas Instruments TWL6040 audio + codec. + This driver provides common support for accessing the device, + additional drivers must be enabled in order to use the + functionality of the device (audio, vibra). config MFD_STMPE bool "Support STMicroelectronics STMPE" diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index f1a1197..6b4d86a 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -56,7 +56,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_TPA6130A2 if I2C select SND_SOC_TLV320DAC33 if I2C select SND_SOC_TWL4030 if TWL4030_CORE - select SND_SOC_TWL6040 if I2C + select SND_SOC_TWL6040 if TWL6040_CORE select SND_SOC_UDA134X select SND_SOC_UDA1380 if I2C select SND_SOC_WL1273 if MFD_WL1273_CORE @@ -274,7 +274,6 @@ config SND_SOC_TWL4030 tristate config SND_SOC_TWL6040 - select TWL6040_CORE tristate config SND_SOC_UDA134X -- 1.7.8.4

1 0

Re: [alsa-devel] [PATCH v3 0/3] MFD: twl6040: Conversion to i2c driver
by Peter Ujfalusi 21 Feb '12

21 Feb '12

Hi Samuel, On 02/20/2012 09:47 PM, Samuel Ortiz wrote: > Hi Peter, > > On Fri, Feb 10, 2012 at 12:00:15PM +0200, Peter Ujfalusi wrote: >> Hello, >> >> Changes since v2: >> - soc/codec/Kconfig: make twl6040 depend on I2C >> - Regulator related patches has been removed (to be sent as separate series) > I applied all 3 patches. Patch #2 did not apply cleanly, due to the fact that > several struct twl4030_codec_data omap4panda.c references are not upstream > yet. This is my .rej: The Panda audio support is going via sound tree. It is in linux-next already for couple of weeks now. The reason you have this reject is that you don't have the audio support for Panda series, and a such upstream does not have it either :( It is for 3.4 also. > --- arch/arm/mach-omap2/board-omap4panda.c > +++ arch/arm/mach-omap2/board-omap4panda.c > @@ -278,7 +279,7 @@ > return 0; > } > > -static struct twl4030_codec_data twl6040_codec = { > +static struct twl6040_codec_data twl6040_codec = { > /* single-step ramp for headset and handsfree */ > .hs_left_step = 0x0f, > .hs_right_step = 0x0f, > @@ -286,17 +287,14 @@ > .hf_right_step = 0x1d, > }; > > -static struct twl4030_audio_data twl6040_audio = { > +static struct twl6040_platform_data twl6040_data = { > .codec = &twl6040_codec, > .audpwron_gpio = 127, > - .naudint_irq = OMAP44XX_IRQ_SYS_2N, > .irq_base = TWL6040_CODEC_IRQ_BASE, > }; > > /* Panda board uses the common PMIC configuration */ > -static struct twl4030_platform_data omap4_panda_twldata = { > - .audio = &twl6040_audio, > -}; > +static struct twl4030_platform_data omap4_panda_twldata; > > I'm not sure hwo we could handle that properly. Either by letting Tony > carrying this patchset, or by sending me the panda patch that adds those > structures. As you prefer. If Liam would take this series that would be probably the way forward. If he does not have objections. Liam: I can create a branch for you to pull from as soon as all other patch receives the needed acks... Samuel: if Liam agrees is it possible for you to drop the twl6040 related patches from your for-next branch? Thank you, Péter

1 0

[alsa-devel] [PATCH] ASoC: ak4642: fixup HeadPhone L/R dapm settings
by Kuninori Morimoto 21 Feb '12

21 Feb '12

Current ak4642 driver had wrong dapm settings for headphone L/R. If you select headphone L, and select R after that, headphone L setting was removed by R settings. This patch fixes it up. It provides just "Headphone Enable" to user side Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/codecs/ak4642.c | 31 ++++++++++++++++--------------- 1 files changed, 16 insertions(+), 15 deletions(-) diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index 16bd1e7..f8e10ce 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -146,13 +146,10 @@ static const struct snd_kcontrol_new ak4642_snd_controls[] = { SOC_DOUBLE_R_TLV("Digital Playback Volume", L_DVC, R_DVC, 0, 0xFF, 1, out_tlv), - - SOC_SINGLE("Headphone Switch", PW_MGMT2, 6, 1, 0), }; -static const struct snd_kcontrol_new ak4642_hpout_mixer_controls[] = { - SOC_DAPM_SINGLE("DACH", MD_CTL4, 0, 1, 0), -}; +static const struct snd_kcontrol_new ak4642_headphone_control = + SOC_DAPM_SINGLE("Switch", PW_MGMT2, 6, 1, 0); static const struct snd_kcontrol_new ak4642_lout_mixer_controls[] = { SOC_DAPM_SINGLE("DACL", SG_SL1, 4, 1, 0), @@ -165,13 +162,12 @@ static const struct snd_soc_dapm_widget ak4642_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("HPOUTR"), SND_SOC_DAPM_OUTPUT("LINEOUT"), - SND_SOC_DAPM_MIXER("HPOUTL Mixer", PW_MGMT2, 5, 0, - &ak4642_hpout_mixer_controls[0], - ARRAY_SIZE(ak4642_hpout_mixer_controls)), + SND_SOC_DAPM_PGA("HPL Out", PW_MGMT2, 5, 0, NULL, 0), + SND_SOC_DAPM_PGA("HPR Out", PW_MGMT2, 4, 0, NULL, 0), + SND_SOC_DAPM_SWITCH("Headphone Enable", SND_SOC_NOPM, 0, 0, + &ak4642_headphone_control), - SND_SOC_DAPM_MIXER("HPOUTR Mixer", PW_MGMT2, 4, 0, - &ak4642_hpout_mixer_controls[0], - ARRAY_SIZE(ak4642_hpout_mixer_controls)), + SND_SOC_DAPM_PGA("DACH", MD_CTL4, 0, 0, NULL, 0), SND_SOC_DAPM_MIXER("LINEOUT Mixer", PW_MGMT1, 3, 0, &ak4642_lout_mixer_controls[0], @@ -184,12 +180,17 @@ static const struct snd_soc_dapm_widget ak4642_dapm_widgets[] = { static const struct snd_soc_dapm_route ak4642_intercon[] = { /* Outputs */ - {"HPOUTL", NULL, "HPOUTL Mixer"}, - {"HPOUTR", NULL, "HPOUTR Mixer"}, + {"HPOUTL", NULL, "HPL Out"}, + {"HPOUTR", NULL, "HPR Out"}, {"LINEOUT", NULL, "LINEOUT Mixer"}, - {"HPOUTL Mixer", "DACH", "DAC"}, - {"HPOUTR Mixer", "DACH", "DAC"}, + {"HPL Out", NULL, "Headphone Enable"}, + {"HPR Out", NULL, "Headphone Enable"}, + + {"Headphone Enable", "Switch", "DACH"}, + + {"DACH", NULL, "DAC"}, + {"LINEOUT Mixer", "DACL", "DAC"}, }; -- 1.7.5.4

2 1

[alsa-devel] [PATCH v2] ASoC: twl4030: Debug code cleanup
by Peter Ujfalusi 21 Feb '12

21 Feb '12

Replace the printk(KERN_ERR* instances with dev_err in the driver. While we are here clean up some of the debug messages as well. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- Hi Mark, your branch was ahead of Liam's, and Takashi's asoc tree that is why it failed to apply cleanly. I have created this patch against your for-next branch. There were one typo fix which caused the git am failure. No other chganges since v1. Regards, Peter sound/soc/codecs/twl4030.c | 33 +++++++++++++++++---------------- 1 files changed, 17 insertions(+), 16 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index d7eee0d..170cf9a 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -1002,8 +1002,8 @@ static int snd_soc_put_twl4030_opmode_enum_double(struct snd_kcontrol *kcontrol, unsigned short mask, bitmask; if (twl4030->configured) { - printk(KERN_ERR "twl4030 operation mode cannot be " - "changed on-the-fly\n"); + dev_err(codec->dev, + "operation mode cannot be changed on-the-fly\n"); return -EBUSY; } @@ -1800,7 +1800,7 @@ static int twl4030_hw_params(struct snd_pcm_substream *substream, mode |= TWL4030_APLL_RATE_96000; break; default: - printk(KERN_ERR "TWL4030 hw params: unknown rate %d\n", + dev_err(codec->dev, "%s: unknown rate %d\n", __func__, params_rate(params)); return -EINVAL; } @@ -1817,7 +1817,7 @@ static int twl4030_hw_params(struct snd_pcm_substream *substream, format |= TWL4030_DATA_WIDTH_32S_24W; break; default: - printk(KERN_ERR "TWL4030 hw params: unknown format %d\n", + dev_err(codec->dev, "%s: unknown format %d\n", __func__, params_format(params)); return -EINVAL; } @@ -1867,13 +1867,13 @@ static int twl4030_set_dai_sysclk(struct snd_soc_dai *codec_dai, case 38400000: break; default: - dev_err(codec->dev, "Unsupported APLL mclk: %u\n", freq); + dev_err(codec->dev, "Unsupported HFCLKIN: %u\n", freq); return -EINVAL; } if ((freq / 1000) != twl4030->sysclk) { dev_err(codec->dev, - "Mismatch in APLL mclk: %u (configured: %u)\n", + "Mismatch in HFCLKIN: %u (configured: %u)\n", freq, twl4030->sysclk * 1000); return -EINVAL; } @@ -1983,9 +1983,9 @@ static int twl4030_voice_startup(struct snd_pcm_substream *substream, * not available. */ if (twl4030->sysclk != 26000) { - dev_err(codec->dev, "The board is configured for %u Hz, while" - "the Voice interface needs 26MHz APLL mclk\n", - twl4030->sysclk * 1000); + dev_err(codec->dev, + "%s: HFCLKIN is %u KHz, voice interface needs 26MHz\n", + __func__, twl4030->sysclk); return -EINVAL; } @@ -1996,8 +1996,8 @@ static int twl4030_voice_startup(struct snd_pcm_substream *substream, & TWL4030_OPT_MODE; if (mode != TWL4030_OPTION_2) { - printk(KERN_ERR "TWL4030 voice startup: " - "the codec mode is not option2\n"); + dev_err(codec->dev, "%s: the codec mode is not option2\n", + __func__); return -EINVAL; } @@ -2038,7 +2038,7 @@ static int twl4030_voice_hw_params(struct snd_pcm_substream *substream, mode |= TWL4030_SEL_16K; break; default: - printk(KERN_ERR "TWL4030 voice hw params: unknown rate %d\n", + dev_err(codec->dev, "%s: unknown rate %d\n", __func__, params_rate(params)); return -EINVAL; } @@ -2067,13 +2067,14 @@ static int twl4030_voice_set_dai_sysclk(struct snd_soc_dai *codec_dai, struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec); if (freq != 26000000) { - dev_err(codec->dev, "Unsupported APLL mclk: %u, the Voice" - "interface needs 26MHz APLL mclk\n", freq); + dev_err(codec->dev, + "%s: HFCLKIN is %u KHz, voice interface needs 26MHz\n", + __func__, freq / 1000); return -EINVAL; } if ((freq / 1000) != twl4030->sysclk) { dev_err(codec->dev, - "Mismatch in APLL mclk: %u (configured: %u)\n", + "Mismatch in HFCLKIN: %u (configured: %u)\n", freq, twl4030->sysclk * 1000); return -EINVAL; } @@ -2221,7 +2222,7 @@ static int twl4030_soc_probe(struct snd_soc_codec *codec) twl4030 = kzalloc(sizeof(struct twl4030_priv), GFP_KERNEL); if (twl4030 == NULL) { - printk("Can not allocate memory\n"); + dev_err(codec->dev, "Can not allocate memory\n"); return -ENOMEM; } snd_soc_codec_set_drvdata(codec, twl4030); -- 1.7.8.4

2 1

[alsa-devel] [PATCH] ASoC: wm_hubs: Convert headphone driver to output driver widget
by Mark Brown 21 Feb '12

21 Feb '12

Mostly for neatness, though it may help with sequencing in some situations. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_hubs.c | 5 ++--- 1 files changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index 9742c66..82b7e9d 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -670,9 +670,8 @@ SND_SOC_DAPM_PGA("Right Output PGA", WM8993_POWER_MANAGEMENT_3, 6, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("Headphone Supply", SND_SOC_NOPM, 0, 0, hp_supply_event, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD), -SND_SOC_DAPM_PGA_E("Headphone PGA", SND_SOC_NOPM, 0, 0, - NULL, 0, - hp_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), +SND_SOC_DAPM_OUT_DRV_E("Headphone PGA", SND_SOC_NOPM, 0, 0, NULL, 0, + hp_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), SND_SOC_DAPM_MIXER("Earpiece Mixer", SND_SOC_NOPM, 0, 0, earpiece_mixer, ARRAY_SIZE(earpiece_mixer)), -- 1.7.9

1 0

[alsa-devel] [PATCH] ASoC: wm8776: Add WM8775 device ID to the WM8776
by Mark Brown 21 Feb '12

21 Feb '12

The WM8775 is register compatible with the WM8776 so can be supported with the same driver though it is an ADC only part. Add the device ID to the WM8776 driver, further updates will be added in the future. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8776.c | 8 +++++++- 1 files changed, 7 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm8776.c b/sound/soc/codecs/wm8776.c index 33e97d1..a19db5a 100644 --- a/sound/soc/codecs/wm8776.c +++ b/sound/soc/codecs/wm8776.c @@ -30,6 +30,11 @@ #include "wm8776.h" +enum wm8776_chip_type { + WM8775 = 1, + WM8776, +}; + /* codec private data */ struct wm8776_priv { enum snd_soc_control_type control_type; @@ -512,7 +517,8 @@ static __devexit int wm8776_i2c_remove(struct i2c_client *client) } static const struct i2c_device_id wm8776_i2c_id[] = { - { "wm8776", 0 }, + { "wm8775", WM8775 }, + { "wm8776", WM8776 }, { } }; MODULE_DEVICE_TABLE(i2c, wm8776_i2c_id); -- 1.7.9

1 0

[alsa-devel] [RFC 00/11]O MAP/ASoC: Move and merge McBSP driver under ASoC
by Peter Ujfalusi 20 Feb '12

20 Feb '12

Hello, The aim of this series is to move the plat-omap/mcbsp.c driver out to sound/soc/omap/ By doing this we can combine the two levels of McBSP driver stack into a single one. Formally we had omap-mcbsp.[1-X] drivers (plat-omap/mcbsp.c), and omap-mcbsp-dai.[0-(X-1) drivers (sound/soc/omap/omap-mcbsp.c). We had global, shared structures, variables, pointers. The API between the two driver was a set of exported function calls where one of the parameter was the McBSP port id. With this series we will have one driver for McBSP (omap-mcbsp.[1-X]) which is basically the ASoC dai driver. All exported/shared global structures are gone since we can use driver data for all ports in this way. With the code in one place we can do further clean up in the McBSP driver, and adding new features will be much easier in the future. Also the DT support for McBSP will be significantly easier if we have one set of driver. McBSP is only used by audio, there are no known other users of the OMAP McBSP (well, Nokia n700 has the WLAN connected through McBSP but we do not have driver for that anyway). The series was created against Takashi's topic/asoc branch. I have tested it on Beagle xM, and compile tested it for OMAP1, OMAP2, and OMAP4 as well. I'm going to do more work on the cleanup, and the plan is to merge the two files into one (mcbsp.c -> omap-mcbsp.c) eventually. Comments, testers are welcome... CC-ing Samuel as well since he is the original author of the plat-omap/mcbsp.c driver. Regards, Peter --- Peter Ujfalusi (11): ARM: OMAP: mcbsp: Convert core driver to proper platform driver OMAP: mcbsp: Move core driver under sound/soc/omap ASoC: OMAP: McBSP: Consolidate plat/mcbsp.h content ASoC: core - Add API call to register dai kcontrols. ASoC: omap-mcbsp: Create a single driver for McBSP ARM: OMAP: Do not register omap-mcbsp-dai device ASoC: OMAP: mcbsp.h: Clean up bit definitions ASoC: OMAP McBSP: Remove redundant accessors ASoC: omap-mcbsp: Cleanup of module probe/remove code ASoC: omap-mcbsp: Merge the omap_mcbsp_data into omap_mcbsp structure ASoC: omap-mcbsp: Simplify DMA configuration arch/arm/mach-omap1/devices.c | 9 - arch/arm/mach-omap1/mcbsp.c | 14 +- arch/arm/mach-omap2/devices.c | 22 -- arch/arm/mach-omap2/mcbsp.c | 8 +- arch/arm/plat-omap/Kconfig | 8 - arch/arm/plat-omap/Makefile | 2 - arch/arm/plat-omap/include/plat/mcbsp.h | 321 ---------------- include/sound/soc.h | 2 + sound/soc/omap/Kconfig | 4 + sound/soc/omap/Makefile | 1 + sound/soc/omap/am3517evm.c | 2 +- sound/soc/omap/ams-delta.c | 2 +- sound/soc/omap/igep0020.c | 2 +- {arch/arm/plat-omap => sound/soc/omap}/mcbsp.c | 463 ++++-------------------- sound/soc/omap/mcbsp.h | 337 +++++++++++++++++ sound/soc/omap/n810.c | 2 +- sound/soc/omap/omap-mcbsp.c | 287 ++++++++------- sound/soc/omap/omap-mcbsp.h | 2 +- sound/soc/omap/omap-pcm.h | 2 + sound/soc/omap/omap3beagle.c | 2 +- sound/soc/omap/omap3evm.c | 2 +- sound/soc/omap/omap3pandora.c | 4 +- sound/soc/omap/osk5912.c | 2 +- sound/soc/omap/overo.c | 2 +- sound/soc/omap/rx51.c | 4 +- sound/soc/omap/sdp3430.c | 4 +- sound/soc/omap/zoom2.c | 4 +- sound/soc/soc-core.c | 31 ++ 28 files changed, 631 insertions(+), 914 deletions(-) rename {arch/arm/plat-omap => sound/soc/omap}/mcbsp.c (69%) create mode 100644 sound/soc/omap/mcbsp.h -- 1.7.8.4

2 3

[alsa-devel] [RFC 00/11]O MAP/ASoC: Move and merge McBSP driver under ASoC
by Peter Ujfalusi 20 Feb '12

20 Feb '12

Hello, The aim of this series is to move the plat-omap/mcbsp.c driver out to sound/soc/omap/ By doing this we can combine the two levels of McBSP driver stack into a single one. Formally we had omap-mcbsp.[1-X] drivers (plat-omap/mcbsp.c), and omap-mcbsp-dai.[0-(X-1) drivers (sound/soc/omap/omap-mcbsp.c). We had global, shared structures, variables, pointers. The API between the two driver was a set of exported function calls where one of the parameter was the McBSP port id. With this series we will have one driver for McBSP (omap-mcbsp.[1-X]) which is basically the ASoC dai driver. All exported/shared global structures are gone since we can use driver data for all ports in this way. With the code in one place we can do further clean up in the McBSP driver, and adding new features will be much easier in the future. Also the DT support for McBSP will be significantly easier if we have one set of driver. McBSP is only used by audio, there are no known other users of the OMAP McBSP (well, Nokia n700 has the WLAN connected through McBSP but we do not have driver for that anyway). The series was created against Takashi's topic/asoc branch. I have tested it on Beagle xM, and compile tested it for OMAP1, OMAP2, and OMAP4 as well. I'm going to do more work on the cleanup, and the plan is to merge the two files into one (mcbsp.c -> omap-mcbsp.c) eventually. Comments, testers are welcome... CC-ing Samuel as well since he is the original author of the plat-omap/mcbsp.c driver. Regards, Peter --- Peter Ujfalusi (11): ARM: OMAP: mcbsp: Convert core driver to proper platform driver OMAP: mcbsp: Move core driver under sound/soc/omap ASoC: OMAP: McBSP: Consolidate plat/mcbsp.h content ASoC: core - Add API call to register dai kcontrols. ASoC: omap-mcbsp: Create a single driver for McBSP ARM: OMAP: Do not register omap-mcbsp-dai device ASoC: OMAP: mcbsp.h: Clean up bit definitions ASoC: OMAP McBSP: Remove redundant accessors ASoC: omap-mcbsp: Cleanup of module probe/remove code ASoC: omap-mcbsp: Merge the omap_mcbsp_data into omap_mcbsp structure ASoC: omap-mcbsp: Simplify DMA configuration arch/arm/mach-omap1/devices.c | 9 - arch/arm/mach-omap1/mcbsp.c | 14 +- arch/arm/mach-omap2/devices.c | 22 -- arch/arm/mach-omap2/mcbsp.c | 8 +- arch/arm/plat-omap/Kconfig | 8 - arch/arm/plat-omap/Makefile | 2 - arch/arm/plat-omap/include/plat/mcbsp.h | 321 ---------------- include/sound/soc.h | 2 + sound/soc/omap/Kconfig | 4 + sound/soc/omap/Makefile | 1 + sound/soc/omap/am3517evm.c | 2 +- sound/soc/omap/ams-delta.c | 2 +- sound/soc/omap/igep0020.c | 2 +- {arch/arm/plat-omap => sound/soc/omap}/mcbsp.c | 463 ++++-------------------- sound/soc/omap/mcbsp.h | 337 +++++++++++++++++ sound/soc/omap/n810.c | 2 +- sound/soc/omap/omap-mcbsp.c | 287 ++++++++------- sound/soc/omap/omap-mcbsp.h | 2 +- sound/soc/omap/omap-pcm.h | 2 + sound/soc/omap/omap3beagle.c | 2 +- sound/soc/omap/omap3evm.c | 2 +- sound/soc/omap/omap3pandora.c | 4 +- sound/soc/omap/osk5912.c | 2 +- sound/soc/omap/overo.c | 2 +- sound/soc/omap/rx51.c | 4 +- sound/soc/omap/sdp3430.c | 4 +- sound/soc/omap/zoom2.c | 4 +- sound/soc/soc-core.c | 31 ++ 28 files changed, 631 insertions(+), 914 deletions(-) rename {arch/arm/plat-omap => sound/soc/omap}/mcbsp.c (69%) create mode 100644 sound/soc/omap/mcbsp.h -- 1.7.8.4

7 19

[alsa-devel] [PATCH] ASoC: twl4030: Debug code cleanup
by Peter Ujfalusi 20 Feb '12

20 Feb '12

Replace the printk(KERN_ERR* instances with dev_err in the driver. While we are here clean up some of the debug messages as well. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/codecs/twl4030.c | 33 +++++++++++++++++---------------- 1 files changed, 17 insertions(+), 16 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index a193f5f..df2dd19 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -1002,8 +1002,8 @@ static int snd_soc_put_twl4030_opmode_enum_double(struct snd_kcontrol *kcontrol, unsigned short mask, bitmask; if (twl4030->configured) { - printk(KERN_ERR "twl4030 operation mode cannot be " - "changed on-the-fly\n"); + dev_err(codec->dev, + "operation mode cannot be changed on-the-fly\n"); return -EBUSY; } @@ -1800,7 +1800,7 @@ static int twl4030_hw_params(struct snd_pcm_substream *substream, mode |= TWL4030_APLL_RATE_96000; break; default: - printk(KERN_ERR "TWL4030 hw params: unknown rate %d\n", + dev_err(codec->dev, "%s: unknown rate %d\n", __func__, params_rate(params)); return -EINVAL; } @@ -1817,7 +1817,7 @@ static int twl4030_hw_params(struct snd_pcm_substream *substream, format |= TWL4030_DATA_WIDTH_32S_24W; break; default: - printk(KERN_ERR "TWL4030 hw params: unknown format %d\n", + dev_err(codec->dev, "%s: unknown format %d\n", __func__, params_format(params)); return -EINVAL; } @@ -1867,13 +1867,13 @@ static int twl4030_set_dai_sysclk(struct snd_soc_dai *codec_dai, case 38400000: break; default: - dev_err(codec->dev, "Unsupported APLL mclk: %u\n", freq); + dev_err(codec->dev, "Unsupported HFCLKIN: %u\n", freq); return -EINVAL; } if ((freq / 1000) != twl4030->sysclk) { dev_err(codec->dev, - "Mismatch in APLL mclk: %u (configured: %u)\n", + "Mismatch in HFCLKIN: %u (configured: %u)\n", freq, twl4030->sysclk * 1000); return -EINVAL; } @@ -1983,9 +1983,9 @@ static int twl4030_voice_startup(struct snd_pcm_substream *substream, * not available. */ if (twl4030->sysclk != 26000) { - dev_err(codec->dev, "The board is configured for %u Hz, while" - "the Voice interface needs 26MHz APLL mclk\n", - twl4030->sysclk * 1000); + dev_err(codec->dev, + "%s: HFCLKIN is %u KHz, voice interface needs 26MHz\n", + __func__, twl4030->sysclk); return -EINVAL; } @@ -1996,8 +1996,8 @@ static int twl4030_voice_startup(struct snd_pcm_substream *substream, & TWL4030_OPT_MODE; if (mode != TWL4030_OPTION_2) { - printk(KERN_ERR "TWL4030 voice startup: " - "the codec mode is not option2\n"); + dev_err(codec->dev, "%s: the codec mode is not option2\n", + __func__); return -EINVAL; } @@ -2038,7 +2038,7 @@ static int twl4030_voice_hw_params(struct snd_pcm_substream *substream, mode |= TWL4030_SEL_16K; break; default: - printk(KERN_ERR "TWL4030 voice hw params: unknown rate %d\n", + dev_err(codec->dev, "%s: unknown rate %d\n", __func__, params_rate(params)); return -EINVAL; } @@ -2067,13 +2067,14 @@ static int twl4030_voice_set_dai_sysclk(struct snd_soc_dai *codec_dai, struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec); if (freq != 26000000) { - dev_err(codec->dev, "Unsupported APLL mclk: %u, the Voice" - "interface needs 26MHz APLL mclk\n", freq); + dev_err(codec->dev, + "%s: HFCLKIN is %u KHz, voice interface needs 26MHz\n", + __func__, freq / 1000); return -EINVAL; } if ((freq / 1000) != twl4030->sysclk) { dev_err(codec->dev, - "Mismatch in APLL mclk: %u (configured: %u)\n", + "Mismatch in HFCLKIN: %u (configured: %u)\n", freq, twl4030->sysclk * 1000); return -EINVAL; } @@ -2221,7 +2222,7 @@ static int twl4030_soc_probe(struct snd_soc_codec *codec) twl4030 = kzalloc(sizeof(struct twl4030_priv), GFP_KERNEL); if (twl4030 == NULL) { - printk("Can not allocate memroy\n"); + dev_err(codec->dev, "Can not allocate memroy\n"); return -ENOMEM; } snd_soc_codec_set_drvdata(codec, twl4030); -- 1.7.8.4

2 1

[alsa-devel] [PATCH] ASoC: imx/mxs: remove redundant SND_PCM selection
by Shawn Guo 20 Feb '12

20 Feb '12

SND_PCM is already selected by SND_SOC, there is no need for SND_IMX_SOC and SND_MXS_SOC to select it again. Signed-off-by: Shawn Guo <shawn.guo(a)linaro.org> --- sound/soc/imx/Kconfig | 1 - sound/soc/mxs/Kconfig | 1 - 2 files changed, 0 insertions(+), 2 deletions(-) diff --git a/sound/soc/imx/Kconfig b/sound/soc/imx/Kconfig index 7383917..91b66ef 100644 --- a/sound/soc/imx/Kconfig +++ b/sound/soc/imx/Kconfig @@ -1,7 +1,6 @@ menuconfig SND_IMX_SOC tristate "SoC Audio for Freescale i.MX CPUs" depends on ARCH_MXC - select SND_PCM select FIQ select SND_SOC_AC97_BUS help diff --git a/sound/soc/mxs/Kconfig b/sound/soc/mxs/Kconfig index e4ba8d5..21d20f3 100644 --- a/sound/soc/mxs/Kconfig +++ b/sound/soc/mxs/Kconfig @@ -1,7 +1,6 @@ menuconfig SND_MXS_SOC tristate "SoC Audio for Freescale MXS CPUs" depends on ARCH_MXS - select SND_PCM help Say Y or M if you want to add support for codecs attached to the MXS SAIF interface. -- 1.7.5.4

2 1

[alsa-devel] [PATCH] Remove the cvscompile file
by Vikram Narayanan 20 Feb '12

20 Feb '12

package: alsa-utils Removed the cvscompile file. Signed-off-by: Vikram Narayanan <vikram186(a)gmail.com> --- cvscompile | 25 ------------------------- 1 files changed, 0 insertions(+), 25 deletions(-) delete mode 100755 cvscompile diff --git a/cvscompile b/cvscompile deleted file mode 100755 index 64ade4f..0000000 --- a/cvscompile +++ /dev/null @@ -1,25 +0,0 @@ -#!/bin/bash - -if test -d ../alsa-lib/utils && ! test -r `aclocal --print-ac-dir`/alsa.m4; then - alsa_m4_flags="-I ../alsa-lib/utils" -fi -aclocal $alsa_m4_flags $ACLOCAL_FLAGS -# save original files to avoid stupid modifications by gettextize -cp Makefile.am Makefile.am.ok -cp configure.in configure.in.ok -gettextize -c -f --no-changelog -echo "EXTRA_DIST = gettext.m4" > m4/Makefile.am -cp Makefile.am.ok Makefile.am -cp configure.in.ok configure.in -autoheader -automake --foreign --copy --add-missing -touch depcomp # for older automake -autoconf -export CFLAGS='-O2 -Wall -pipe -g' -echo "CFLAGS=$CFLAGS" -echo "./configure $@" -./configure $@ || exit 1 -unset CFLAGS -if [ -z "$CVSCOMPILE_NO_MAKE" ]; then - make -fi -- 1.7.4.1

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Re: [alsa-devel] Direct use of the DMA buffer?
by Pavel Hofman 20 Feb '12

20 Feb '12

Dne 20.2.2012 10:21, Jaroslav Kysela napsal(a): >> >> Clemens, thanks a lot for the information. How does it work when the >> alsa-lib chain contains plugins, such as plug, rate, etc.? I would >> assume the application circumvents these. > > Nope. The mmap commit mechanism was introduced to keep the plugin > interface functional. The direct mmap access to the DMA buffer is only > applied for the direct hw: devices. If a plugin is used inside alsa-lib, > the application talks to the plugin chain (it means that an intermediate > buffer for samples can be allocated to let things working). > Jaroslav, thanks a lot for the clarification. Pavel.

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Re: [alsa-devel] Direct use of the DMA buffer?
by Jaroslav Kysela 20 Feb '12

20 Feb '12

Date 20.2.2012 09:38, Pavel Hofman wrote: > > Dne 20.2.2012 09:11, Clemens Ladisch napsal(a): >> Pavel Hofman wrote: >>> Please excuse my ignorance, but is it possible for an application to >>> directly use/output to the DMA buffer allocated by the kernel in >>> hw_params? >> >> Set one of the SND_PCM_ACCESS_MMAP_* access types and then use >> snd_pcm_mmap_begin/commit. >> > > Clemens, thanks a lot for the information. How does it work when the > alsa-lib chain contains plugins, such as plug, rate, etc.? I would > assume the application circumvents these. Nope. The mmap commit mechanism was introduced to keep the plugin interface functional. The direct mmap access to the DMA buffer is only applied for the direct hw: devices. If a plugin is used inside alsa-lib, the application talks to the plugin chain (it means that an intermediate buffer for samples can be allocated to let things working). Jaroslav -- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project; Red Hat, Inc.

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[alsa-devel] [PATCH] ASoC: io: Retrieve val_bytes from the regmap API
by Mark Brown 20 Feb '12

20 Feb '12

Allow us to build infrastructure which needs to know the size of a value without requiring regmap based drivers to supply this information to both ASoC and regmap by asking regmap for the value. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-io.c | 6 ++++++ 1 files changed, 6 insertions(+), 0 deletions(-) diff --git a/sound/soc/soc-io.c b/sound/soc/soc-io.c index 39ba507..4d8dc6a 100644 --- a/sound/soc/soc-io.c +++ b/sound/soc/soc-io.c @@ -114,6 +114,7 @@ int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec, enum snd_soc_control_type control) { struct regmap_config config; + int ret; memset(&config, 0, sizeof(config)); codec->write = hw_write; @@ -141,6 +142,11 @@ int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec, case SND_SOC_REGMAP: /* Device has made its own regmap arrangements */ codec->using_regmap = true; + + ret = regmap_get_val_bytes(codec->control_data); + /* Errors are legitimate for non-integer byte multiples */ + if (ret > 0) + codec->val_bytes = ret; break; default: -- 1.7.9

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[alsa-devel] [PATCH 0/3] Coefficient configuration support
by Mark Brown 20 Feb '12

20 Feb '12

This series is only compile tested as yet, I'll probably get a chance to test it shortly - it should work out so hopefully the review can overlap usefully with the testing. Mark Brown (3): ASoC: core: Add SND_SOC_BYTES control for coefficient blocks ASoC: core: Add support for masking out parts of coefficient blocks ASoC: wm5100: Add DRC coefficient configuration include/sound/soc.h | 27 +++++++++++ sound/soc/codecs/wm5100.c | 2 + sound/soc/soc-core.c | 110 +++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 139 insertions(+), 0 deletions(-)

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[alsa-devel] Direct use of the DMA buffer?
by Pavel Hofman 20 Feb '12

20 Feb '12

Hello, Please excuse my ignorance, but is it possible for an application to directly use/output to the DMA buffer allocated by the kernel in hw_params? Something like using the actual DMA buffer in the snd_pcm_writei function, without copying the data in alsa-lib/driver PCM midle layer (?), just managing the pointers. Only a theoretical question though, no plans on my side :) Thanks a lot for even a short No :-) Best regards, Pavel.

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Re: [alsa-devel] Random process lockup on ARM board: alsa-lib-1.0.25, FUTEX_WAIT_PRIVATE
by Clemens Ladisch 20 Feb '12

20 Feb '12

Jonathan Andrews wrote: > I've tried building alsalib in many ways, at the moment its : > $ ./configure --target=arm-linux --host=i686-linux --enable-shared --disable-static --without-pthread Try adding --disable-ucm and --disable-python. Regards, Clemens

1 0

[alsa-devel] ASoC: fsi: Add FSI DMAEngine support
by Kuninori Morimoto 20 Feb '12

20 Feb '12

Hi Mark, Liam, and Paul These are DMAEngine support for FSI driver I created a lot of patches since FSI DMAEngine needed much modification. I hope it was good for reviewing. But some patches are just cleanup/tidyup/rename. Last patch is main DMAEngine patch. it support Tx case only at this point. FSI will use PIO version if CPU/platform doesn't have DMAEngine support. >> Paul Could you please check #16 patch which modify arch/arm/mach-shmobile and arch/sh Kuninori Morimoto (19): ASoC: fsi: reduce runtime calculation by using pre-setting ASoC: fsi: tidyup: fsi_stream_xx() functions were gathered ASoC: fsi: data push/pop calculation part was divided ASoC: fsi: rename fsi_dma_soft_xxx() to fsi_pio_xxx() ASoC: fsi: tidyup: move fsi_fifo_init() onto fsi_hw_startup() ASoC: fsi: remove unnecessary parameter from fsi_hw_shutdown() ASoC: fsi: rename fsi_stream_push/pop() to fsi_stream_init/quit() ASoC: fsi: modify fsi_pio_get_area() parameter and using position ASoC: fsi: re-define fsi_is_play() and fsi_stream_is_play() ASoC: fsi: use fsi_stream in fsi_get_current_fifo_samples() parameter ASoC: fsi: add fsi_stream_handler and PIO handler ASoC: fsi: tidyup: fsi_pio_xxx() are gathered ASoC: fsi: don't use is_play as a parameter of fsi functions ASoC: fsi: add .start_stop handler to fsi_stream_handler ASoC: fsi: fsi_stream_is_working() care substream->runtime ASoC: fsi: PortA/B information was controlled by sh_fsi_port_info ASoC: fsi: add .init/.quit handler support ASoC: fsi: fixup fsi_pointer() calculation method ASoC: fsi: Add DMAEngine support Best regards --- Kuninori Morimoto

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[alsa-devel] [PATCH] ASoC: Add __devinit annotation for pxa2xx_ac97_probe
by Axel Lin 20 Feb '12

20 Feb '12

This fixes below build warning: WARNING: vmlinux.o(.text+0x1e632c): Section mismatch in reference from the function pxa2xx_ac97_probe() to the function .devinit.text:pxa2xx_ac97_hw_probe() The function pxa2xx_ac97_probe() references the function __devinit pxa2xx_ac97_hw_probe(). This is often because pxa2xx_ac97_probe lacks a __devinit annotation or the annotation of pxa2xx_ac97_hw_probe is wrong. Also rename pxa_ac97_dai to pxa_ac97_dai_driver to fix below build warning: LD sound/soc/pxa/built-in.o WARNING: sound/soc/pxa/built-in.o(.data+0x18c): Section mismatch in reference from the variable pxa_ac97_dai to the function .devinit.text:pxa2xx_ac97_probe() The variable pxa_ac97_dai references the function __devinit pxa2xx_ac97_probe() If the reference is valid then annotate the variable with __init* or __refdata (see linux/init.h) or name the variable: *driver, *_template, *_timer, *_sht, *_ops, *_probe, *_probe_one, *_console Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/pxa/pxa2xx-ac97.c | 10 +++++----- 1 files changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index 837ff34..4800d5f 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -103,7 +103,7 @@ static int pxa2xx_ac97_resume(struct snd_soc_dai *dai) #define pxa2xx_ac97_resume NULL #endif -static int pxa2xx_ac97_probe(struct snd_soc_dai *dai) +static int __devinit pxa2xx_ac97_probe(struct snd_soc_dai *dai) { return pxa2xx_ac97_hw_probe(to_platform_device(dai->dev)); } @@ -179,7 +179,7 @@ static const struct snd_soc_dai_ops pxa_ac97_mic_dai_ops = { * There is only 1 physical AC97 interface for pxa2xx, but it * has extra fifo's that can be used for aux DACs and ADCs. */ -static struct snd_soc_dai_driver pxa_ac97_dai[] = { +static struct snd_soc_dai_driver pxa_ac97_dai_driver[] = { { .name = "pxa2xx-ac97", .ac97_control = 1, @@ -244,13 +244,13 @@ static __devinit int pxa2xx_ac97_dev_probe(struct platform_device *pdev) * driver to do interesting things with the clocking to get us up * and running. */ - return snd_soc_register_dais(&pdev->dev, pxa_ac97_dai, - ARRAY_SIZE(pxa_ac97_dai)); + return snd_soc_register_dais(&pdev->dev, pxa_ac97_dai_driver, + ARRAY_SIZE(pxa_ac97_dai_driver)); } static int __devexit pxa2xx_ac97_dev_remove(struct platform_device *pdev) { - snd_soc_unregister_dais(&pdev->dev, ARRAY_SIZE(pxa_ac97_dai)); + snd_soc_unregister_dais(&pdev->dev, ARRAY_SIZE(pxa_ac97_dai_driver)); return 0; } -- 1.7.5.4

2 1

[alsa-devel] [PATCH 1/2] ALSA: Remove __devinit annotation for pxa2xx_ac97_hw_probe to fix section mismatch warning
by Axel Lin 20 Feb '12

20 Feb '12

This fixes below build warning: WARNING: vmlinux.o(.text+0x1e632c): Section mismatch in reference from the function pxa2xx_ac97_probe() to the function .devinit.text:pxa2xx_ac97_hw_probe() The function pxa2xx_ac97_probe() references the function __devinit pxa2xx_ac97_hw_probe(). This is often because pxa2xx_ac97_probe lacks a __devinit annotation or the annotation of pxa2xx_ac97_hw_probe is wrong. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/arm/pxa2xx-ac97-lib.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/arm/pxa2xx-ac97-lib.c b/sound/arm/pxa2xx-ac97-lib.c index 0e9e822..5e926c4 100644 --- a/sound/arm/pxa2xx-ac97-lib.c +++ b/sound/arm/pxa2xx-ac97-lib.c @@ -313,7 +313,7 @@ int pxa2xx_ac97_hw_resume(void) EXPORT_SYMBOL_GPL(pxa2xx_ac97_hw_resume); #endif -int __devinit pxa2xx_ac97_hw_probe(struct platform_device *dev) +int pxa2xx_ac97_hw_probe(struct platform_device *dev) { int ret; pxa2xx_audio_ops_t *pdata = dev->dev.platform_data; -- 1.7.5.4

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[alsa-devel] [PATCH] checkpatch fixes in ASoC
by Vikram Narayanan 19 Feb '12

19 Feb '12

Various checkpatch fixes in the ASoC tree Signed-off-by: Vikram Narayanan <vikram186(a)gmail.com> --- sound/soc/soc-cache.c | 5 ++--- sound/soc/soc-core.c | 17 ++++++++--------- sound/soc/soc-dapm.c | 14 ++++++++------ sound/soc/soc-pcm.c | 4 ++-- sound/soc/soc-utils.c | 2 +- 5 files changed, 21 insertions(+), 21 deletions(-) diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index 9077aa4..6b6c226 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -73,7 +73,7 @@ struct snd_soc_rbtree_node { unsigned int word_size; /* number of bytes needed to represent the register index */ void *block; /* block of adjacent registers */ unsigned int blklen; /* number of registers available in the block */ -} __attribute__ ((packed)); +} __attribute__ ((__packed)); struct snd_soc_rbtree_ctx { struct rb_root root; @@ -1017,9 +1017,8 @@ int snd_soc_cache_sync(struct snd_soc_codec *codec) int ret; const char *name; - if (!codec->cache_sync) { + if (!codec->cache_sync) return 0; - } if (!codec->cache_ops || !codec->cache_ops->sync) return -ENOSYS; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index a25fa63..c13d414 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -195,7 +195,7 @@ static ssize_t pmdown_time_set(struct device *dev, container_of(dev, struct snd_soc_pcm_runtime, dev); int ret; - ret = strict_strtol(buf, 10, &rtd->pmdown_time); + ret = kstrtol(buf, 10, &rtd->pmdown_time); if (ret) return ret; @@ -254,10 +254,10 @@ static ssize_t codec_reg_write_file(struct file *file, while (*start == ' ') start++; - reg = simple_strtoul(start, &start, 16); + reg = kstrtoul(start, &start, 16); while (*start == ' ') start++; - if (strict_strtoul(start, 16, &value)) + if (kstrtoul(start, 16, &value)) return -EINVAL; /* Userspace has been fiddling around behind the kernel's back */ @@ -458,7 +458,7 @@ static int soc_ac97_dev_unregister(struct snd_soc_codec *codec) } /* stop no dev release warning */ -static void soc_ac97_device_release(struct device *dev){} +static void soc_ac97_device_release(struct device *dev) {} /* register ac97 codec to bus */ static int soc_ac97_dev_register(struct snd_soc_codec *codec) @@ -758,9 +758,9 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num) dev_dbg(card->dev, "binding %s at idx %d\n", dai_link->name, num); /* do we already have the CPU DAI for this link ? */ - if (rtd->cpu_dai) { + if (rtd->cpu_dai) goto find_codec; - } + /* no, then find CPU DAI from registered DAIs*/ list_for_each_entry(cpu_dai, &dai_list, list) { if (!strcmp(cpu_dai->name, dai_link->cpu_dai_name)) { @@ -773,9 +773,8 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num) find_codec: /* do we already have the CODEC for this link ? */ - if (rtd->codec) { + if (rtd->codec) goto find_platform; - } /* no, then find CODEC from registered CODECs*/ list_for_each_entry(codec, &codec_list, list) { @@ -2028,7 +2027,7 @@ int snd_soc_add_platform_controls(struct snd_soc_platform *platform, err = snd_ctl_add(card, snd_soc_cnew(control, platform, control->name, NULL)); if (err < 0) { - dev_err(platform->dev, "Failed to add %s %d\n",control->name, err); + dev_err(platform->dev, "Failed to add %s %d\n", control->name, err); return err; } } diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index fcd4cb5..07abd58 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -379,7 +379,7 @@ static int dapm_connect_mux(struct snd_soc_dapm_context *dapm, list_add(&path->list, &dapm->card->paths); list_add(&path->list_sink, &dest->sources); list_add(&path->list_source, &src->sinks); - path->name = (char*)e->texts[i]; + path->name = (char *)e->texts[i]; dapm_set_path_status(dest, path, 0); return 0; } @@ -1769,7 +1769,7 @@ static ssize_t dapm_widget_show(struct device *dev, { struct snd_soc_pcm_runtime *rtd = container_of(dev, struct snd_soc_pcm_runtime, dev); - struct snd_soc_codec *codec =rtd->codec; + struct snd_soc_codec *codec = rtd->codec; struct snd_soc_dapm_widget *w; int count = 0; char *state = "not set"; @@ -1795,7 +1795,7 @@ static ssize_t dapm_widget_show(struct device *dev, case snd_soc_dapm_regulator_supply: if (w->name) count += sprintf(buf + count, "%s: %s\n", - w->name, w->power ? "On":"Off"); + w->name, w->power ? "On" : "Off"); break; default: break; @@ -2209,7 +2209,7 @@ int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm) return -ENOMEM; } - switch(w->id) { + switch (w->id) { case snd_soc_dapm_switch: case snd_soc_dapm_mixer: case snd_soc_dapm_mixer_named_ctl: @@ -2700,7 +2700,9 @@ int snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, size_t name_len; int ret; - if ((w = dapm_cnew_widget(widget)) == NULL) + w = dapm_cnew_widget(widget); + + if (!w) return -ENOMEM; switch (w->id) { @@ -2829,7 +2831,7 @@ static void soc_dapm_stream_event(struct snd_soc_dapm_context *dapm, w->name, w->sname, stream, event); if (strstr(w->sname, stream)) { dapm_mark_dirty(w, "stream event"); - switch(event) { + switch (event) { case SND_SOC_DAPM_STREAM_START: w->active = 1; break; diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index ee15337..174fdbb 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -7,7 +7,7 @@ * Copyright (C) 2010 Texas Instruments Inc. * * Authors: Liam Girdwood <lrg(a)ti.com> - * Mark Brown <broonie(a)opensource.wolfsonmicro.com> + * Mark Brown <broonie(a)opensource.wolfsonmicro.com> * * This program is free software; you can redistribute it and/or modify it * under the terms of the GNU General Public License as published by the @@ -613,7 +613,7 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) if (codec_dai->driver->capture.channels_min) capture = 1; - dev_dbg(rtd->card->dev, "registered pcm #%d %s\n",num,new_name); + dev_dbg(rtd->card->dev, "registered pcm #%d %s\n", num, new_name); ret = snd_pcm_new(rtd->card->snd_card, new_name, num, playback, capture, &pcm); if (ret < 0) { diff --git a/sound/soc/soc-utils.c b/sound/soc/soc-utils.c index 4220bb0..2eac245 100644 --- a/sound/soc/soc-utils.c +++ b/sound/soc/soc-utils.c @@ -5,7 +5,7 @@ * * Author: Mark Brown <broonie(a)opensource.wolfsonmicro.com> * Liam Girdwood <lrg(a)slimlogic.co.uk> - * + * * * This program is free software; you can redistribute it and/or modify it * under the terms of the GNU General Public License as published by the -- 1.7.4.1

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[alsa-devel] Random process lockup on ARM board: alsa-lib-1.0.25, FUTEX_WAIT_PRIVATE
by Jonathan Andrews 18 Feb '12

18 Feb '12

Sorry all me again. I've written a program that picks up UDP broadcast packets containing audio from a number of sources, mixes it together and presents it to alsa for playback. I'm experiencing random freezes. I cant reproduce then running the process under strace as the process on the ARM board slows down to the point it underruns with ALSA all the time and the symptom doesn't show. If I attach strace after the process has frozen I get this and nothing else: # strace -p 443 Process 443 attached - interrupt to quit futex(0x4030e144, FUTEX_WAIT_PRIVATE, 2, NULL This stop happens at random at any point typically after minutes but run time can be over an hour. I need to work around this as I need the board for a demo. Im using, alsa-lib-1.0.25 kernel 2.6.36.4 BusyBox v1.19.2 I also compiled strace for arm, but this and a busybox shell is the only environment I have on the board. The problem showed when I starting using this function to poll alsa: The same code on my PC dual core AMD does not seem to suffer the problem : alsa-lib-1.0.23, Linux jonspc 2.6.32.26-175.fc12.i686 #1 SMP Before anyone points it out I want to poll as alsa is not the only sound output supported by the code. int sound_system_needs_data_now() { int framesleft; snd_pcm_status_t *status; snd_pcm_status_alloca( &status ); // If the alsa playback has stalled then it definately needs data snd_pcm_status(playback_handle,status); if( snd_pcm_status_get_state( status ) != SND_PCM_STATE_RUNNING ) return(TRUE); framesleft=snd_pcm_avail (playback_handle); //printf("FL=%d\n",framesleft); fflush(stdout); if (framesleft>1000) return(TRUE); else return(FALSE); } Stuck process # cat /proc/443/status Name: udp-many-way-au State: S (sleeping) Tgid: 443 Pid: 443 PPid: 392 TracerPid: 0 Uid: 0 0 0 0 Gid: 0 0 0 0 FDSize: 32 Groups: VmPeak: 4356 kB VmSize: 4292 kB VmLck: 132 kB VmHWM: 1788 kB VmRSS: 1788 kB VmData: 396 kB VmStk: 136 kB VmExe: 28 kB VmLib: 3348 kB VmPTE: 12 kB VmSwap: 0 kB Threads: 1 SigQ: 0/471 SigPnd: 0000000000000000 ShdPnd: 0000000000000000 SigBlk: 0000000000002000 SigIgn: 0000000000000006 SigCgt: 0000000180002000 CapInh: 0000000000000000 CapPrm: ffffffffffffffff CapEff: ffffffffffffffff CapBnd: ffffffffffffffff Cpus_allowed: 1 Cpus_allowed_list: 0 Mems_allowed: 1 Mems_allowed_list: 0 voluntary_ctxt_switches: 5908 nonvoluntary_ctxt_switches: 1766 Any useful suggestions welcome. Thanks, Jon

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[alsa-devel] [PATCH] ASoC: soc-core: Show the returned values on error messages
by Fabio Estevam 17 Feb '12

17 Feb '12

Showing the returned values on error messages is useful information. While at it, use pr_err/pr_warn whenever possible. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/soc-core.c | 39 ++++++++++++++++++++++----------------- 1 files changed, 22 insertions(+), 17 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 6bad7cd..7001fb3 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -931,7 +931,8 @@ static void soc_remove_dai_link(struct snd_soc_card *card, int num, int order) if (codec_dai->driver->remove) { err = codec_dai->driver->remove(codec_dai); if (err < 0) - printk(KERN_ERR "asoc: failed to remove %s\n", codec_dai->name); + pr_err("asoc: failed to remove %s: %d\n", + codec_dai->name, err); } codec_dai->probed = 0; list_del(&codec_dai->card_list); @@ -943,7 +944,8 @@ static void soc_remove_dai_link(struct snd_soc_card *card, int num, int order) if (platform->driver->remove) { err = platform->driver->remove(platform); if (err < 0) - printk(KERN_ERR "asoc: failed to remove %s\n", platform->name); + pr_err("asoc: failed to remove %s: %d\n", + platform->name, err); } /* Make sure all DAPM widgets are freed */ @@ -966,7 +968,8 @@ static void soc_remove_dai_link(struct snd_soc_card *card, int num, int order) if (cpu_dai->driver->remove) { err = cpu_dai->driver->remove(cpu_dai); if (err < 0) - printk(KERN_ERR "asoc: failed to remove %s\n", cpu_dai->name); + pr_err("asoc: failed to remove %s: %d\n", + cpu_dai->name, err); } cpu_dai->probed = 0; list_del(&cpu_dai->card_list); @@ -1213,8 +1216,8 @@ static int soc_probe_dai_link(struct snd_soc_card *card, int num, int order) if (cpu_dai->driver->probe) { ret = cpu_dai->driver->probe(cpu_dai); if (ret < 0) { - printk(KERN_ERR "asoc: failed to probe CPU DAI %s\n", - cpu_dai->name); + pr_err("asoc: failed to probe CPU DAI %s: %d\n", + cpu_dai->name, ret); module_put(cpu_dai->dev->driver->owner); return ret; } @@ -1245,8 +1248,8 @@ static int soc_probe_dai_link(struct snd_soc_card *card, int num, int order) if (codec_dai->driver->probe) { ret = codec_dai->driver->probe(codec_dai); if (ret < 0) { - printk(KERN_ERR "asoc: failed to probe CODEC DAI %s\n", - codec_dai->name); + pr_err("asoc: failed to probe CODEC DAI %s: %d\n", + codec_dai->name, ret); return ret; } } @@ -1266,12 +1269,13 @@ static int soc_probe_dai_link(struct snd_soc_card *card, int num, int order) ret = device_create_file(rtd->dev, &dev_attr_pmdown_time); if (ret < 0) - printk(KERN_WARNING "asoc: failed to add pmdown_time sysfs\n"); + pr_warn("asoc: failed to add pmdown_time sysfs:%d\n", ret); /* create the pcm */ ret = soc_new_pcm(rtd, num); if (ret < 0) { - printk(KERN_ERR "asoc: can't create pcm %s\n", dai_link->stream_name); + pr_err("asoc: can't create pcm %s :%d\n", + dai_link->stream_name, ret); return ret; } @@ -1304,7 +1308,7 @@ static int soc_register_ac97_dai_link(struct snd_soc_pcm_runtime *rtd) ret = soc_ac97_dev_register(rtd->codec); if (ret < 0) { - printk(KERN_ERR "asoc: AC97 device register failed\n"); + pr_err("asoc: AC97 device register failed:%d\n", ret); return ret; } @@ -1444,8 +1448,8 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) ret = snd_card_create(SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1, card->owner, 0, &card->snd_card); if (ret < 0) { - printk(KERN_ERR "asoc: can't create sound card for card %s\n", - card->name); + pr_err("asoc: can't create sound card for card %s: %d\n", + card->name, ret); mutex_unlock(&card->mutex); return; } @@ -1563,7 +1567,8 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) ret = snd_card_register(card->snd_card); if (ret < 0) { - printk(KERN_ERR "asoc: failed to register soundcard for %s\n", card->name); + pr_err("asoc: failed to register soundcard for %s: %d\n", + card->name, ret); goto probe_aux_dev_err; } @@ -1572,7 +1577,8 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) for (i = 0; i < card->num_rtd; i++) { ret = soc_register_ac97_dai_link(&card->rtd[i]); if (ret < 0) { - printk(KERN_ERR "asoc: failed to register AC97 %s\n", card->name); + pr_err("asoc: failed to register AC97 %s: %d\n", + card->name, ret); while (--i >= 0) soc_unregister_ac97_dai_link(card->rtd[i].codec); goto probe_aux_dev_err; @@ -3070,7 +3076,7 @@ static inline char *fmt_multiple_name(struct device *dev, struct snd_soc_dai_driver *dai_drv) { if (dai_drv->name == NULL) { - printk(KERN_ERR "asoc: error - multiple DAI %s registered with no name\n", + pr_err("asoc: error - multiple DAI %s registered with no name\n", dev_name(dev)); return NULL; } @@ -3542,8 +3548,7 @@ static int __init snd_soc_init(void) #ifdef CONFIG_DEBUG_FS snd_soc_debugfs_root = debugfs_create_dir("asoc", NULL); if (IS_ERR(snd_soc_debugfs_root) || !snd_soc_debugfs_root) { - printk(KERN_WARNING - "ASoC: Failed to create debugfs directory\n"); + pr_warn("ASoC: Failed to create debugfs directory\n"); snd_soc_debugfs_root = NULL; } -- 1.7.1

2 1

Re: [alsa-devel] tea575x-tuner improvements & use in maxiradio
by Ondrej Zary 17 Feb '12

17 Feb '12

On Wednesday 08 February 2012 08:29:25 Hans Verkuil wrote: > On Tuesday, February 07, 2012 23:20:19 Ondrej Zary wrote: > > On Sunday 05 February 2012 14:17:05 Hans Verkuil wrote: > > > These patches improve the tea575x-tuner module to make it up to date > > > with the latest V4L2 frameworks. > > > > > > The maxiradio driver has also been converted to use the tea575x-tuner > > > and I've used that card to test it. > > > > > > Unfortunately, this card can't read the data pin, so the new hardware > > > seek functionality has been tested only partially (yes, it seeks, but > > > when it finds a channel I can't read back the frequency). > > > > > > Ondrej, are you able to test these patches for the sound cards that use > > > this tea575x tuner? > > > > > > Note that these two patches rely on other work that I did and that > > > hasn't been merged yet. So it is best to pull from my git tree: > > > > > > http://git.linuxtv.org/hverkuil/media_tree.git/shortlog/refs/heads/radi > > >o-pc i2 > > > > > > You can use the v4l-utils repository > > > (http://git.linuxtv.org/v4l-utils.git) to test the drivers: the > > > v4l2-compliance test should succeed and with v4l2-ctl you can test the > > > hardware seek: > > > > > > To seek down: > > > > > > v4l2-ctl -d /dev/radio0 --freq-seek=dir=0 > > > > > > To seek up: > > > > > > v4l2-ctl -d /dev/radio0 --freq-seek=dir=1 > > > > > > To do the compliance test: > > > > > > v4l2-compliance -r /dev/radio0 > > > > It seems to work (tested with SF64-PCR - snd_fm801) but the seek is > > severely broken. Reading the frequency immediately after seek does not > > work, it always returns the old value (haven't found a delay that works). > > Reading it later (copied back snd_tea575x_get_freq function) works. The > > chip seeks randomly up or down, ignoring UP/DOWN flag and often stops at > > wrong place (only noise) or even outside the FM range. > > > > So I strongly suggest not to enable this (mis-)feature. The HW seems to > > be completely broken (unless there's some weird bug in the code). > > Well, it seemed like a good idea at the time :-) I'll remove this > 'feature', it's really not worth our time to try and make this work for > these old cards. I fixed it somehow. Now it works most of the time. The important things: - a delay must be present after search start and before first register read or the seek does weird things - when the search stops, the new frequency is not available immediately, we must wait until it appears in the register (fortunately, we can clear the frequency bits when starting the search as it starts at the frequency currently set, not from the value written) - sometimes, seek remains on the current frequency (or moves only a little), so repeat it until it moves by at least 50 kHz --- a/sound/i2c/other/tea575x-tuner.c +++ b/sound/i2c/other/tea575x-tuner.c @@ -89,7 +89,7 @@ static void snd_tea575x_write(struct snd_tea575x *tea, unsigned int val) tea->ops->set_pins(tea, 0); } -static unsigned int snd_tea575x_read(struct snd_tea575x *tea) +static u32 snd_tea575x_read(struct snd_tea575x *tea) { u16 l, rdata; u32 data = 0; @@ -120,6 +120,27 @@ static unsigned int snd_tea575x_read(struct snd_tea575x *tea) return data; } +static void snd_tea575x_get_freq(struct snd_tea575x *tea) +{ + u32 freq = snd_tea575x_read(tea) & TEA575X_BIT_FREQ_MASK; + + if (freq == 0) { + tea->freq = 0; + return; + } + + /* freq *= 12.5 */ + freq *= 125; + freq /= 10; + /* crystal fixup */ + if (tea->tea5759) + freq += TEA575X_FMIF; + else + freq -= TEA575X_FMIF; + + tea->freq = clamp(freq * 16, FREQ_LO, FREQ_HI); /* from kHz */ +} + static void snd_tea575x_set_freq(struct snd_tea575x *tea) { u32 freq = tea->freq; @@ -203,6 +224,8 @@ static int vidioc_g_frequency(struct file *file, void *priv, if (f->tuner != 0) return -EINVAL; f->type = V4L2_TUNER_RADIO; + if (!tea->cannot_read_data) + snd_tea575x_get_freq(tea); f->frequency = tea->freq; return 0; } @@ -225,36 +248,50 @@ static int vidioc_s_hw_freq_seek(struct file *file, void *fh, struct v4l2_hw_freq_seek *a) { struct snd_tea575x *tea = video_drvdata(file); + int i, old_freq; + unsigned long timeout; if (tea->cannot_read_data) return -ENOTTY; + + snd_tea575x_get_freq(tea); + old_freq = tea->freq; + /* clear the frequency, HW will fill it in */ + tea->val &= ~TEA575X_BIT_FREQ_MASK; tea->val |= TEA575X_BIT_SEARCH; - tea->val &= ~TEA575X_BIT_UPDOWN; if (a->seek_upward) tea->val |= TEA575X_BIT_UPDOWN; + else + tea->val &= ~TEA575X_BIT_UPDOWN; snd_tea575x_write(tea, tea->val); + timeout = jiffies + msecs_to_jiffies(10000); for (;;) { - unsigned val = snd_tea575x_read(tea); - - if (!(val & TEA575X_BIT_SEARCH)) { - /* Found a frequency */ - val &= TEA575X_BIT_FREQ_MASK; - val = (val * 10) / 125; - if (tea->tea5759) - val += TEA575X_FMIF; - else - val -= TEA575X_FMIF; - tea->freq = clamp(val * 16, FREQ_LO, FREQ_HI); - return 0; - } + if (time_after(jiffies, timeout)) + break; if (schedule_timeout_interruptible(msecs_to_jiffies(10))) { /* some signal arrived, stop search */ tea->val &= ~TEA575X_BIT_SEARCH; snd_tea575x_write(tea, tea->val); return -ERESTARTSYS; } + if (!(snd_tea575x_read(tea) & TEA575X_BIT_SEARCH)) { + /* Found a frequency, wait until it can be read */ + for (i = 0; i < 100; i++) { + msleep(10); + snd_tea575x_get_freq(tea); + if (tea->freq != 0) /* available */ + break; + } + /* if we moved by less than 50 kHz, continue seeking */ + if (abs(old_freq - tea->freq) < 16 * 500) { + snd_tea575x_write(tea, tea->val); + continue; + } + tea->val &= ~TEA575X_BIT_SEARCH; + return 0; + } } - return 0; + return -ETIMEDOUT; } static int tea575x_s_ctrl(struct v4l2_ctrl *ctrl) @@ -318,7 +355,7 @@ int snd_tea575x_init(struct snd_tea575x *tea) return -ENODEV; } - tea->val = TEA575X_BIT_BAND_FM | TEA575X_BIT_SEARCH_10_40; + tea->val = TEA575X_BIT_BAND_FM | TEA575X_BIT_SEARCH_5_28; tea->freq = 90500 * 16; /* 90.5Mhz default */ snd_tea575x_set_freq(tea); -- Ondrej Zary

2 1

[alsa-devel] [PATCH] ASoC: wm8996: Make sure we bounce /RESET to reset
by Mark Brown 17 Feb '12

17 Feb '12

While it matches the current code only bringing the device out of reset isn't actually doing what the function says so make sure we set the GPIO high before we pull it low. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8996.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index e3e3a37..03d191a 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -1713,6 +1713,7 @@ static bool wm8996_volatile_register(struct device *dev, unsigned int reg) static int wm8996_reset(struct wm8996_priv *wm8996) { if (wm8996->pdata.ldo_ena > 0) { + gpio_set_value_cansleep(wm8996->pdata.ldo_ena, 1); gpio_set_value_cansleep(wm8996->pdata.ldo_ena, 0); return 0; } else { -- 1.7.9

1 0

[alsa-devel] Bluetooth: alsa says connection refused?
by Fabrice DELENTE 17 Feb '12

17 Feb '12

Hello. I have a bluetooth headset that I have successfully used with a cell phone and an Android tablet, and that I could use in Ubuntu on a friend's machine. I'd like to use it with mplayer on my laptop (under Slackware 13.37). I have the BT address, and the test-audio script in bluez-4.98/test works (at least, I hear two beeps in the headphone, and hcitool con shows # hcitool con Connections: < ACL 00:18:91:D0:7A:24 handle 42 state 1 lm MASTER AUTH ENCRYPT so I guess it is possible to connnect to the device. However, if I try hcitool cc $HEADSET, the connection is closed immediately...) I created an ~/.asoundrc with: pcm.btheadset { type plug slave { pcm { type bluetooth device "00:18:91:D0:7A:24" profile "auto" } } hint { show on description "BlueTooth Halterrego H.EAR 01 Headset" } } ctl.btheadset { type bluetooth } but when I try to use it with mplayer: # mplayer -ao alsa:device=btheadset file.mp3 I get bt_audio_service_open: connect() failed: Connection refused (111) [AO_ALSA] Playback open error: Connection refused Failed to initialize audio driver 'alsa:device=btheadset' Could not open/initialize audio device -> no sound. Audio: no sound Video: no video Exiting... (End of file) What did I miss? (I recompiled bluez with support for alsa and alsa with support for bluetooth) Thanks! -- F. Delente

1 0

[alsa-devel] [PATCH] ASoC: Change spitz_ext_control to take dapm as argument.
by Axel Lin 17 Feb '12

17 Feb '12

This fixes below build warning: CC sound/soc/pxa/spitz.o sound/soc/pxa/spitz.c: In function 'spitz_startup': sound/soc/pxa/spitz.c:116: warning: passing argument 1 of 'spitz_ext_control' from incompatible pointer type sound/soc/pxa/spitz.c:47: note: expected 'struct snd_soc_card *' but argument is of type 'struct snd_soc_codec *' Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/pxa/spitz.c | 12 +++++------- 1 files changed, 5 insertions(+), 7 deletions(-) diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c index 4d108dc..fc052d8 100644 --- a/sound/soc/pxa/spitz.c +++ b/sound/soc/pxa/spitz.c @@ -44,10 +44,8 @@ static int spitz_jack_func; static int spitz_spk_func; static int spitz_mic_gpio; -static void spitz_ext_control(struct snd_soc_card *card) +static void spitz_ext_control(struct snd_soc_dapm_context *dapm) { - struct snd_soc_dapm_context *dapm = &card->dapm; - if (spitz_spk_func == SPITZ_SPK_ON) snd_soc_dapm_enable_pin(dapm, "Ext Spk"); else @@ -113,7 +111,7 @@ static int spitz_startup(struct snd_pcm_substream *substream) mutex_lock(&codec->mutex); /* check the jack status at stream startup */ - spitz_ext_control(codec); + spitz_ext_control(&codec->dapm); mutex_unlock(&codec->mutex); @@ -179,7 +177,7 @@ static int spitz_set_jack(struct snd_kcontrol *kcontrol, return 0; spitz_jack_func = ucontrol->value.integer.value[0]; - spitz_ext_control(card); + spitz_ext_control(&card->dapm); return 1; } @@ -193,13 +191,13 @@ static int spitz_get_spk(struct snd_kcontrol *kcontrol, static int spitz_set_spk(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_card *card = snd_kcontrol_chip(kcontrol); + struct snd_soc_card *card = snd_kcontrol_chip(kcontrol); if (spitz_spk_func == ucontrol->value.integer.value[0]) return 0; spitz_spk_func = ucontrol->value.integer.value[0]; - spitz_ext_control(card); + spitz_ext_control(&card->dapm); return 1; } -- 1.7.5.4

3 2

[alsa-devel] [PATCH 1/2] ASoC: core - Add direct mapping between DAIs and DAPM
by Liam Girdwood 17 Feb '12

17 Feb '12

Currently we map DAI streams to DAPM widgets based on a string stream name. This can lead to some limitations when working with lots of DAI links as there is a risk of naming collision with similar stream names. This patch creates a 1:N mapping between a DAI and it's DAPM widgets. It's still string based, but we now directly reference the widget names with the DAI and bind the DAI to the widget ptrs at initialisation. The binding also tales into consideration the parent codec or platform that the DAI and widget belong to so that we can easily support multiple components with the same widget names. This patch still works in conjunction with the stream based events, but it is intended to eventually replace the DAI stream name. e.g. to map twl6040 headset DACs to headset DAI static const char *headset_widgets[] = {"HSDAC Left", "HSDAC Right"}; static struct snd_soc_dai_driver twl6040_dai[] = { ... { .name = "twl6040-dl1", .playback = { .stream_name = "Headset Playback", .channel_widgets = headset_widgets, .channels_min = 1, .channels_max = ARRAY_SIZE(headset_widgets), .rates = TWL6040_RATES, .formats = TWL6040_FORMATS, }, .ops = &twl6040_dai_ops, }, This patch will allow DAI connection to any widget type as not all DAIs connect directly to DACs or ADCs, some can connect via a mux that can swap channels etc. This patch is also intended to be the starting point for future DAPM channel mapping support as we now have a direct link between PCMs and DAPM widgets. e.g. In the example above we map channel 0 (left) to "HSDAC Left" and channel 1 (right) to "HSDAC Right". Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- include/sound/soc-dai.h | 11 ++- include/sound/soc.h | 1 + sound/soc/soc-core.c | 42 +++++++++++++ sound/soc/soc-dapm.c | 155 +++++++++++++++++++++++++++++++++++++++++++++++ 4 files changed, 205 insertions(+), 4 deletions(-) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 2413acc..ce5beea 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -246,14 +246,17 @@ struct snd_soc_dai { unsigned int rate; /* parent platform/codec */ - union { - struct snd_soc_platform *platform; - struct snd_soc_codec *codec; - }; + struct snd_soc_platform *platform; + struct snd_soc_codec *codec; struct snd_soc_card *card; struct list_head list; struct list_head card_list; + + /* runtime widgets */ + struct snd_soc_dapm_widget **playback_widgets; + struct snd_soc_dapm_widget **capture_widgets; + bool dapm_bind_complete; }; static inline void *snd_soc_dai_get_dma_data(const struct snd_soc_dai *dai, diff --git a/include/sound/soc.h b/include/sound/soc.h index 8fa4dca..bace9b8 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -507,6 +507,7 @@ struct snd_soc_pcm_stream { unsigned int rates; /* SNDRV_PCM_RATE_* */ unsigned int rate_min; /* min rate */ unsigned int rate_max; /* max rate */ + const char **channel_widgets; /* DAPM channel widget names */ unsigned int channels_min; /* min channels */ unsigned int channels_max; /* max channels */ unsigned int sig_bits; /* number of bits of content */ diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 6bad7cd..733f5de 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -68,6 +68,8 @@ static int pmdown_time = 5000; module_param(pmdown_time, int, 0); MODULE_PARM_DESC(pmdown_time, "DAPM stream powerdown time (msecs)"); +int dapm_bind_dai_widgets(struct snd_soc_dai *dai, int stream); + /* returns the minimum number of bytes needed to represent * a particular given value */ static int min_bytes_needed(unsigned long val) @@ -1392,6 +1394,41 @@ static int snd_soc_init_codec_cache(struct snd_soc_codec *codec, return 0; } +static int soc_init_card_dai_widgets(struct snd_soc_card *card) +{ + struct snd_soc_dai *dai; + int ret = 0; + + list_for_each_entry(dai, &card->dai_dev_list, card_list) { + + /* only process DAIs that use the new API until + * the old "stream name" API is fully deprecated */ + if (!dai->driver->playback.channel_widgets && + !dai->driver->capture.channel_widgets) + continue; + + /* skip if already instanciated */ + if (dai->dapm_bind_complete) + continue; + + /* bind DAI widget name to component widget */ + if (dai->driver->playback.channels_max) { + ret = dapm_bind_dai_widgets(dai, SNDRV_PCM_STREAM_PLAYBACK); + if (ret < 0) + return ret; + } + if (dai->driver->capture.channels_max) { + ret = dapm_bind_dai_widgets(dai, SNDRV_PCM_STREAM_CAPTURE); + if (ret < 0) + return ret; + } + + dai->dapm_bind_complete = 1; + } + + return ret; +} + static void snd_soc_instantiate_card(struct snd_soc_card *card) { struct snd_soc_codec *codec; @@ -1556,6 +1593,9 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) } snd_soc_dapm_new_widgets(&card->dapm); + ret = soc_init_card_dai_widgets(card); + if (ret < 0) + goto probe_aux_dev_err; if (card->fully_routed) list_for_each_entry(codec, &card->codec_dev_list, card_list) @@ -3139,6 +3179,8 @@ found: pr_debug("Unregistered DAI '%s'\n", dai->name); kfree(dai->name); + kfree(dai->playback_widgets); + kfree(dai->capture_widgets); kfree(dai); } EXPORT_SYMBOL_GPL(snd_soc_unregister_dai); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 63a5614..5f99842 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -829,6 +829,161 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget) return con; } +static struct snd_soc_dapm_widget * + dapm_get_codec_widget(struct snd_soc_codec *codec, const char *name) +{ + struct snd_soc_card *card = codec->card; + struct snd_soc_dapm_widget *w; + + /* find widget for this codec and widget name */ + list_for_each_entry(w, &card->widgets, list) { + + /* make sure the widget belongs to the DAI codec */ + if (w->codec && w->codec != codec) + continue; + + if (!strcmp(w->name, name)) + return w; + } + + return NULL; +} + +static struct snd_soc_dapm_widget * + dapm_get_platform_widget(struct snd_soc_platform *platform, + const char *name) +{ + struct snd_soc_card *card = platform->card; + struct snd_soc_dapm_widget *w; + + /* find widget for this platform and widget name */ + list_for_each_entry(w, &card->widgets, list) { + + /* make sure the widget belongs to the DAI platform */ + if (w->platform && w->platform != platform) + continue; + + if (!strcmp(w->name, name)) + return w; + } + + return NULL; +} + +int dapm_bind_dai_playback_widgets(struct snd_soc_dai *dai) +{ + struct snd_soc_card *card = dai->card; + struct snd_soc_dapm_widget *w = NULL; + const char *widget_name; + int i; + + if (!dai->driver->playback.channel_widgets) + return 0; + + dai->playback_widgets = + kzalloc(sizeof(struct snd_soc_dapm_widget *) * + dai->driver->playback.channels_max, GFP_KERNEL); + if (!dai->playback_widgets) + return -ENOMEM; + + for (i = 0; i < dai->driver->playback.channels_max; i++) { + widget_name = dai->driver->playback.channel_widgets[i]; + + if (dai->codec) { + struct snd_soc_codec *codec; + + list_for_each_entry(codec, &card->codec_dev_list, card_list) { + w = dapm_get_codec_widget(codec, widget_name); + if (w) { + dev_dbg(dai->dev, "bind %s to %s\n", w->name, codec->name); + break; + } + } + } else if (dai->platform) { + struct snd_soc_platform *platform; + + list_for_each_entry(platform, &card->platform_dev_list, card_list) { + w = dapm_get_platform_widget(platform, widget_name); + if (w) { + dev_dbg(dai->dev, "bind %s to %s\n", w->name, platform->name); + break; + } + } + } + + if (w) + dai->playback_widgets[i] = w; + else { + dev_err(dai->dev, "unable to find playback DAI AIF %s\n", + widget_name); + return -EINVAL; + } + } + + return 0; +} + +int dapm_bind_dai_capture_widgets(struct snd_soc_dai *dai) +{ + struct snd_soc_card *card = dai->card; + struct snd_soc_dapm_widget *w = NULL; + const char *widget_name; + int i; + + if (!dai->driver->capture.channel_widgets) + return 0; + + dai->capture_widgets = + kzalloc(sizeof(struct snd_soc_dapm_widget *) * + dai->driver->capture.channels_max, GFP_KERNEL); + if (!dai->capture_widgets) + return -ENOMEM; + + for (i = 0; i < dai->driver->capture.channels_max; i++) { + widget_name = dai->driver->capture.channel_widgets[i]; + + if (dai->codec) { + struct snd_soc_codec *codec; + + list_for_each_entry(codec, &card->codec_dev_list, card_list) { + w = dapm_get_codec_widget(codec, widget_name); + if (w) { + dev_dbg(dai->dev, "bind %s to %s\n", w->name, codec->name); + break; + } + } + } else if (dai->platform) { + struct snd_soc_platform *platform; + + list_for_each_entry(platform, &card->platform_dev_list, card_list) { + w = dapm_get_platform_widget(platform, widget_name); + if (w) { + dev_dbg(dai->dev, "bind %s to %s\n", w->name, platform->name); + break; + } + } + } + + if (w) + dai->capture_widgets[i] = w; + else { + dev_err(dai->dev, "unable to find playback DAI AIF %s\n", + widget_name); + return -EINVAL; + } + } + + return 0; +} + +int dapm_bind_dai_widgets(struct snd_soc_dai *dai, int stream) +{ + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + return dapm_bind_dai_playback_widgets(dai); + else + return dapm_bind_dai_capture_widgets(dai); +} + /* * Handler for generic register modifier widget. */ -- 1.7.5.4

3 7

[alsa-devel] Realtime controlling of ALSA parameters from UI
by Aleksey Midenkov 17 Feb '12

17 Feb '12

Hello guys! I want to suggest you an idea of what I am thinking for a long time. And want to know of what you are thinking on it from perspective of how hard it would be to implement. I'm pretty sure the feature will be well popular because for me as an ordinary user it would be great stuff! So to the point. What I am saying about is the ability to control parameters from asound.conf at user level. This would give us such great things like switching sound output realtime or control each channel volume level in mixer GUI. If you will suggest userpace daemons like Pulse or Jack. There are some situations (like playing piano f.ex.) where realtime really matters, where such daemons put unacceptable latency. Another point is that they are not such feature rich. They cannot downmix f.ex. with controlling of each channel volume level (please tell me if I am wrong here). Every opinion is appreciated!

1 0

[alsa-devel] [PATCH] ALSA: core: Constify the name in new kcontrols
by Mark Brown 17 Feb '12

17 Feb '12

We never modify it and this lets us use a const string as the name without warnings. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- I've got some ASoC patches which need this to avoid warnings, if you're OK with it can I apply it via there please? include/sound/control.h | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/include/sound/control.h b/include/sound/control.h index b2796e8..57815f6 100644 --- a/include/sound/control.h +++ b/include/sound/control.h @@ -40,7 +40,7 @@ struct snd_kcontrol_new { snd_ctl_elem_iface_t iface; /* interface identifier */ unsigned int device; /* device/client number */ unsigned int subdevice; /* subdevice (substream) number */ - unsigned char *name; /* ASCII name of item */ + const unsigned char *name; /* ASCII name of item */ unsigned int index; /* index of item */ unsigned int access; /* access rights */ unsigned int count; /* count of same elements */ -- 1.7.9.rc1

2 1

[alsa-devel] Usage of jack_status_check in ASoC GPIO
by Omair Mohammed Abdullah 17 Feb '12

17 Feb '12

If I need to call a codec driver function whenever Headset status changes (to set various registers, turn off micbias etc.). In such a case, can the jack_status_check callback in snd_soc_jack_gpio structure be used? If I use it, I would need to modify the callback to pass certain parameters (codec pointer, status). Is this the intended use of the callback?

2 1

[alsa-devel] [PATCH 1/4] ASoC: ak4535: Remove -codec from driver name
by Mark Brown 17 Feb '12

17 Feb '12

Redundant, the device is only a CODEC. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/ak4535.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c index dd15516..29d4dec 100644 --- a/sound/soc/codecs/ak4535.c +++ b/sound/soc/codecs/ak4535.c @@ -443,7 +443,7 @@ MODULE_DEVICE_TABLE(i2c, ak4535_i2c_id); static struct i2c_driver ak4535_i2c_driver = { .driver = { - .name = "ak4535-codec", + .name = "ak4535", .owner = THIS_MODULE, }, .probe = ak4535_i2c_probe, -- 1.7.9.rc1

1 3

[alsa-devel] [PATCH 1/3] ASoC: wm8962: Convert to module_i2c_driver()
by Mark Brown 17 Feb '12

17 Feb '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8962.c | 12 +----------- 1 files changed, 1 insertions(+), 11 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 2f06886..bc04b49 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3793,17 +3793,7 @@ static struct i2c_driver wm8962_i2c_driver = { .id_table = wm8962_i2c_id, }; -static int __init wm8962_modinit(void) -{ - return i2c_add_driver(&wm8962_i2c_driver); -} -module_init(wm8962_modinit); - -static void __exit wm8962_exit(void) -{ - i2c_del_driver(&wm8962_i2c_driver); -} -module_exit(wm8962_exit); +module_i2c_driver(wm8962_i2c_driver); MODULE_DESCRIPTION("ASoC WM8962 driver"); MODULE_AUTHOR("Mark Brown <broonie(a)opensource.wolfsonmicro.com>"); -- 1.7.9.rc1

1 2

[alsa-devel] [PATCH 1/7] ASoC: wm8962: Update the clocking when setting system clock
by Mark Brown 17 Feb '12

17 Feb '12

Make sure we update for any changes in cases where we reconfigure while live (eg, for analogue bypass). Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8962.c | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 4a081dd..3f37a91 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2705,6 +2705,8 @@ static int wm8962_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, wm8962->sysclk_rate = freq; + wm8962_configure_bclk(codec); + return 0; } -- 1.7.9.rc1

1 6

[alsa-devel] snd_emu10k1 with E-MU 1616m PCIe
by tilt 16 Feb '12

16 Feb '12

Hello! I use the Creative E-MU 1616m PCIe: 03:04.0 Multimedia audio controller: Creative Labs SB0400 Audigy2 Value Subsystem: Creative Labs Device 4007 I am on Linux kernel 3.2.4, -driver and -firmware from git; -utils from 1.0.24.2-5 (Debian amd_64). I have manually applied the patch from [1] and initialized the device successfully: [ 3.387211] snd_emu10k1 0000:03:04.0: PCI INT A -> GSI 16 (level, low) -> IRQ 16 [ 3.387341] ALSA emu10k1_main.c:823 emu1010: Special config. [ 3.387479] ALSA emu10k1_main.c:862 emu1010: EMU_HANA_ID = 0x3f [ 3.387530] ALSA emu10k1_main.c:881 emu1010: filename emu/emu1010b.fw testing [ 3.393206] ALSA emu10k1_main.c:686 firmware size = 0x11bdc [ 6.294376] ALSA emu10k1_main.c:898 emu1010: Hana Firmware loaded [ 6.294475] ALSA emu10k1_main.c:901 emu1010: Hana version: 1.6 [ 6.294583] ALSA emu10k1_main.c:906 emu1010: Card options = 0x8 [ 6.294656] ALSA emu10k1_main.c:908 emu1010: Card options = 0x8 [ 6.295222] ALSA emu10k1_main.c:947 emu1010: Card options3 = 0x8 [ 6.314806] ALSA emu10k1_main.c:227 Audigy2 value: Special config. [ 6.315231] ALSA emufx.c:1545 EMU outputs on [ 6.315280] ALSA emufx.c:1593 EMU2 inputs on There are irregularities & limitations, but I do have 48kHz capture and playback. I have two questions: 1.: I only use Dock ports, and I notice many irregularities in DAC and ADC assignment; so I wonder: should /usr/local/lib/firmware/emu/micro_dock.fw not also be loaded? 2.: I use JACK with input: hw:0,2 (Mutichannel Capture/PT Playback) output: hw:0,3 (Multichannel Playback) One thing that I would like to fix if at all possible: I can not assign anything to Dock Phones L/R; instead they always repeat Dock DAC 1L/R, no matter what I route there. Is this a limitation of the firmware, or are the controls misleading me? Kind regards & thanks in advance for any suggestions, Tilman. [1] ALSA bugtracking system 0005250: E-MU 1010 PCI-Express not recognized https://bugtrack.alsa-project.org/alsa-bug/view.php?id=5250

1 2

[alsa-devel] [PATCH] ASoC: dapm: Only mark pin widgets as dirty if we actually change state
by Mark Brown 16 Feb '12

16 Feb '12

Small optimisation for noop state updates. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-dapm.c | 4 +++- 1 files changed, 3 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 2844879..a2a9c11 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1915,10 +1915,12 @@ static int snd_soc_dapm_set_pin(struct snd_soc_dapm_context *dapm, return -EINVAL; } + if (w->connected != status) + dapm_mark_dirty(w, "pin configuration"); + w->connected = status; if (status == 0) w->force = 0; - dapm_mark_dirty(w, "pin configuration"); return 0; } -- 1.7.9.rc1

1 0

[alsa-devel] [PATCH] ASoC: spitz - Fix kcontrols to use card instead of codec.
by Liam Girdwood 16 Feb '12

16 Feb '12

Machine kcontrols now use card instead of codec for thier "chip". Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- sound/soc/pxa/spitz.c | 12 ++++++------ 1 files changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c index 90c5245..4d108dc 100644 --- a/sound/soc/pxa/spitz.c +++ b/sound/soc/pxa/spitz.c @@ -44,9 +44,9 @@ static int spitz_jack_func; static int spitz_spk_func; static int spitz_mic_gpio; -static void spitz_ext_control(struct snd_soc_codec *codec) +static void spitz_ext_control(struct snd_soc_card *card) { - struct snd_soc_dapm_context *dapm = &codec->dapm; + struct snd_soc_dapm_context *dapm = &card->dapm; if (spitz_spk_func == SPITZ_SPK_ON) snd_soc_dapm_enable_pin(dapm, "Ext Spk"); @@ -173,13 +173,13 @@ static int spitz_get_jack(struct snd_kcontrol *kcontrol, static int spitz_set_jack(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct snd_soc_card *card = snd_kcontrol_chip(kcontrol); if (spitz_jack_func == ucontrol->value.integer.value[0]) return 0; spitz_jack_func = ucontrol->value.integer.value[0]; - spitz_ext_control(codec); + spitz_ext_control(card); return 1; } @@ -193,13 +193,13 @@ static int spitz_get_spk(struct snd_kcontrol *kcontrol, static int spitz_set_spk(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct snd_soc_card *card = snd_kcontrol_chip(kcontrol); if (spitz_spk_func == ucontrol->value.integer.value[0]) return 0; spitz_spk_func = ucontrol->value.integer.value[0]; - spitz_ext_control(codec); + spitz_ext_control(card); return 1; } -- 1.7.5.4

2 1

[alsa-devel] Strange behaviour on audio chipset VIA VT2021 codec
by Marco De Maio - BETTINI Srl 16 Feb '12

16 Feb '12

Hello, i need to develop for my work an audio recorder application. I have to develop the software on Linux OS (Ubuntu 8.04 with kernel 2.6.23.13), and the motherboard is Gigabyte GA-G41MT-S2P (rev. 1.3) where is installed the integrated audio chipset VIA VT2021 codec. I upgraded the alsa driver to 1.0.25 version for support of audio chipset VIA VT2021. I downloaded the driver and compiled it with the following option ./configure --with-cards=hda-intel --with-sequencer=yes and i done some experiments with the following simple application example with alsa lib: #include <alsa/asoundlib.h> int main() { int rc; int size; snd_pcm_t *handle; snd_pcm_hw_params_t *params; unsigned int val; int dir; snd_pcm_uframes_t frames; char *buffer; /* Open PCM device for playback. */ rc = snd_pcm_open(&handle, "default", SND_PCM_STREAM_CAPTURE, 0); if (rc < 0) { fprintf(stderr, "unable to open pcm device: %s\n", snd_strerror(rc)); exit(1); } /* Allocate a hardware parameters object. */ snd_pcm_hw_params_alloca(&params); /* Fill it in with default values. */ snd_pcm_hw_params_any(handle, params); /* Set the desired hardware parameters. */ /* Interleaved mode */ snd_pcm_hw_params_set_access(handle, params, SND_PCM_ACCESS_RW_INTERLEAVED); /* Signed 16-bit little-endian format */ snd_pcm_hw_params_set_format(handle, params, SND_PCM_FORMAT_S16_LE); /* Two channels (stereo) */ snd_pcm_hw_params_set_channels(handle, params, 2); /* 44100 bits/second sampling rate (CD quality) */ val = 44100; snd_pcm_hw_params_set_rate_near(handle, params, &val, &dir); /* Set period size to 32 frames. */ frames = 32; snd_pcm_hw_params_set_period_size_near(handle, params, &frames, &dir); /* Write the parameters to the driver */ rc = snd_pcm_hw_params(handle, params); if (rc < 0) { fprintf(stderr, "unable to set hw parameters: %s\n", snd_strerror(rc)); exit(1); } /* Use a buffer large enough to hold one period */ snd_pcm_hw_params_get_period_size(params, &frames, &dir); size = frames * 4; /* 2 bytes/sample, 2 channels */ buffer = (char *) malloc(size); snd_pcm_hw_params_get_period_time(params, &val, &dir); while (1) { printf ("Capture data\n"); rc = snd_pcm_readi(handle, buffer, size); if (rc == -EAGAIN) { fprintf(stderr, "EAGAIN occurred\n"); snd_pcm_prepare(handle); } else if (rc == -EPIPE) { fprintf(stderr, "EPIPE occurred\n"); snd_pcm_prepare(handle); } else if (rc == -ESTRPIPE) { fprintf(stderr, "ESTRPIPE occurred\n"); snd_pcm_prepare(handle); } else if (rc > 0) { fprintf(stderr, "read %d ok\n", rc); } else { fprintf(stderr, "error occurred %s\n", snd_strerror(rc)); exit (1); } } snd_pcm_drain(handle); snd_pcm_close(handle); free(buffer); return 0; } Running the example above, I found the following behaviors: 1) Until there is no mic plugged, the application stop on snd_pcm_readi. When mic is plugged, the application starts again. So when i unplug mic the application stops on snd_pcm_readi, when i plug mic the application starts again, and so on. I want to point out that i achieve the same result with a simple stand alone jack. 2) If i reboot my machine with mic plugged, after restart, I run the application but it stop immediately on snd_pcm_readi and I need to unplug/plug the mic to stats again the application. This behavior seems to me quite strange. Please someone could help me to figure out why this behavior ? Maybe something wrong with my example ? Maybe something wrong with compiling options of the driver or maybe into the drivers itself ? I tried the same application on a different mainboard with Realtek ALC889 codec, and the applications never stops on snd_pcm_readi (I suppose this is the correct behavior), and run correctly. Let me know if you need more information about. Thank to all in advance. Marco

1 0

Re: [alsa-devel] Bug report - patch_realtek.c - Laptop HP COMPAQ B1900 Series
by Takashi Iwai 16 Feb '12

16 Feb '12

At Wed, 15 Feb 2012 17:08:45 +0800, joey.jiaojg wrote: > > Well, now I have fully fixed the speaker/headphone issue for HP COMPAQ > B1900 by adding a new model=b1900 into patch_realtek.c which also can > automute by detect HP state. That's great. > As I cannot compile alsa-driver-1.0.25 on my ubuntu 11.10, so I modified > based on kernel-3.0.13 (alsa version 1.0.24) from ubuntu src. > Attached with stocked source and my updated one, please consider adding > into next alsa-driver. Could you simply give a patch file (with "diff -up")? Then I can check your changes more easily. thanks, Takashi

3 18

[alsa-devel] hda_intel:azx_get_response timeout, switching to single_cmd mode: last cmd=0x00070503
by Sirus Jolt 16 Feb '12

16 Feb '12

Hi, I've got some problems to start session, because i've got the message like subject: hda_intel: azx_get_response timeout, switching to single_cmd mode: last cmd=0x00070503 I have no idea about the problem, I need your help. In some threat mention it but don't have the answer. thanks

4 5

[alsa-devel] ASoC: A missing entry in wm2200_bclk_rates_cd array?
by Axel Lin 16 Feb '12

16 Feb '12

Hi Mark, WM2200_NUM_BCLK_RATES is 12. It looks like there is a missing entry in wm2200_bclk_rates_cd array. static int wm2200_bclk_rates_cd[WM2200_NUM_BCLK_RATES] = { 5644800, 2882400, 1881600, 1411200, 705600, 470400, 352800, 176400, 117600, 88200, 58800, }; Axel

1 0

[alsa-devel] [PATCH] ASoC: Show device id in the debug message
by Axel Lin 16 Feb '12

16 Feb '12

Show the id we read when the id mismatch is detected. This is useful for debugging. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/wm8962.c | 2 +- sound/soc/codecs/wm8996.c | 2 +- sound/soc/codecs/wm9090.c | 2 +- 3 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 172dcac..5a6516e 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3718,7 +3718,7 @@ static __devinit int wm8962_i2c_probe(struct i2c_client *i2c, } if (reg != 0x6243) { dev_err(&i2c->dev, - "Device is not a WM8962, ID %x != 0x6243\n", ret); + "Device is not a WM8962, ID %x != 0x6243\n", reg); ret = -EINVAL; goto err_regmap; } diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index 759ea69..c3bde4a 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -3185,7 +3185,7 @@ static __devinit int wm8996_i2c_probe(struct i2c_client *i2c, goto err_regmap; } if (reg != 0x8915) { - dev_err(&i2c->dev, "Device is not a WM8996, ID %x\n", ret); + dev_err(&i2c->dev, "Device is not a WM8996, ID %x\n", reg); ret = -EINVAL; goto err_regmap; } diff --git a/sound/soc/codecs/wm9090.c b/sound/soc/codecs/wm9090.c index e8280ee..4b263b6 100644 --- a/sound/soc/codecs/wm9090.c +++ b/sound/soc/codecs/wm9090.c @@ -639,7 +639,7 @@ static int wm9090_i2c_probe(struct i2c_client *i2c, if (ret < 0) goto err; if (reg != 0x9093) { - dev_err(&i2c->dev, "Device is not a WM9090, ID=%x\n", ret); + dev_err(&i2c->dev, "Device is not a WM9090, ID=%x\n", reg); ret = -ENODEV; goto err; } -- 1.7.5.4

2 1

[alsa-devel] [PATCH] ASoC: Get correct revision id for wm2200
by Axel Lin 16 Feb '12

16 Feb '12

Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/wm2200.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c index a9388df..acbdc5f 100644 --- a/sound/soc/codecs/wm2200.c +++ b/sound/soc/codecs/wm2200.c @@ -2084,7 +2084,7 @@ static __devinit int wm2200_i2c_probe(struct i2c_client *i2c, goto err_reset; } - wm2200->rev = ret & WM2200_DEVICE_REVISION_MASK; + wm2200->rev = reg & WM2200_DEVICE_REVISION_MASK; dev_info(&i2c->dev, "revision %c\n", wm2200->rev + 'A'); -- 1.7.5.4

2 1

Re: [alsa-devel] [PATCH v5 2/2] mxs-dma : rewrite the last parameter of mxs_dma_prep_slave_sg()
by Shawn Guo 16 Feb '12

16 Feb '12

On Thu, Feb 16, 2012 at 10:05:26AM +0800, Huang Shijie wrote: > Hi Shawn: > >I'm trying to give it a test. But it does not apply on either mainline > >or next tree. You probably had a wrong base for this patch. > > > thanks for your work. > > As the previous email said, this patch set bases on :[PATCH v2] > mtd/gpmi : add gpmi support for mx6q. > Then you are doing something wrong. This series should be based on mainline, and then you add imx6q support on top of this series. -- Regards, Shawn

1 0

Re: [alsa-devel] [PATCH v5 1/2] mxs-dma : move the mxs dma.h to a more common place
by Shawn Guo 15 Feb '12

15 Feb '12

On Mon, Feb 13, 2012 at 01:25:55PM +0800, Huang Shijie wrote: > Move the header to a more common place. > The mxs dma engine is not only used in mx23/mx28, but also used > in mx50/mx6q. It will also be used in the future chips. > > Rename it to mxs-dma.h, and create a new folder include/linux/fsl/ to > store the Freescale's header files. > > change mxs-dma driver, mxs-mmc driver, gpmi-nand driver, mxs-saif driver > to the new header file. > > Signed-off-by: Huang Shijie <b32955(a)freescale.com> Acked-by: Shawn Guo <shawn.guo(a)linaro.org>

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Re: [alsa-devel] [PATCH v5 2/2] mxs-dma : rewrite the last parameter of mxs_dma_prep_slave_sg()
by Shawn Guo 15 Feb '12

15 Feb '12

On Mon, Feb 13, 2012 at 01:25:56PM +0800, Huang Shijie wrote: > [1] Background : > The GPMI does ECC read page operation with a DMA chain consist of three DMA > Command Structures. The middle one of the chain is used to enable the BCH, > and read out the NAND page. > > The WAIT4END(wait for command end) is a comunication signal between > the GPMI and MXS-DMA. > > [2] The current DMA code sets the WAIT4END bit at the last one, such as: > > +-----+ +-----+ +-----+ > | cmd | ------------> | cmd | ------------------> | cmd | > +-----+ +-----+ +-----+ > ^ > | > | > set WAIT4END here > > This chain works fine in the mx23/mx28. > > [3] But in the new GPMI version (used in MX50/MX60), the WAIT4END bit should > be set not only at the last DMA Command Structure, > but also at the middle one, such as: > > +-----+ +-----+ +-----+ > | cmd | ------------> | cmd | ------------------> | cmd | > +-----+ +-----+ +-----+ > ^ ^ > | | > | | > set WAIT4END here too set WAIT4END here > > If we do not set WAIT4END, the BCH maybe stalls in "ECC reading page" state. > In the next ECC write page operation, a DMA-timeout occurs. > This has been catched in the MX6Q board. > > [4] In order to fix the bug, rewrite the last parameter of mxs_dma_prep_slave_sg(), > and use the dma_ctrl_flags: > --------------------------------------------------------- > DMA_PREP_INTERRUPT : append a new DMA Command Structrue. > DMA_CTRL_ACK : set the WAIT4END bit for this DMA Command Structure. > --------------------------------------------------------- > > [5] changes to the relative drivers: > <1> For mxs-mmc driver, just use the new flags, do not change any logic. > <2> For gpmi-nand driver, and use the new flags to set the DMA chain for > ecc read page. > > Signed-off-by: Huang Shijie <b32955(a)freescale.com> > --- > drivers/dma/mxs-dma.c | 32 ++++++++++++++++++++++++++++---- > drivers/mmc/host/mxs-mmc.c | 10 +++++----- > drivers/mtd/nand/gpmi-nand/gpmi-lib.c | 19 +++++++++++++------ > 3 files changed, 46 insertions(+), 15 deletions(-) > I'm trying to give it a test. But it does not apply on either mainline or next tree. You probably had a wrong base for this patch. > diff --git a/drivers/dma/mxs-dma.c b/drivers/dma/mxs-dma.c > index 0afcedb..0ddfd30 100644 > --- a/drivers/dma/mxs-dma.c > +++ b/drivers/dma/mxs-dma.c > @@ -349,10 +349,32 @@ static void mxs_dma_free_chan_resources(struct dma_chan *chan) > clk_disable_unprepare(mxs_dma->clk); > } > > +/* > + * How to use the flags for ->device_prep_slave_sg() : > + * [1] If there is only one DMA command in the DMA chain, the code should be: > + * ...... > + * ->device_prep_slave_sg(DMA_CTRL_ACK); > + * ...... > + * [2] If there are two DMA commands in the DMA chain, the code should be > + * ...... > + * ->device_prep_slave_sg(0); > + * ...... > + * ->device_prep_slave_sg(DMA_PREP_INTERRUPT | DMA_CTRL_ACK); > + * ...... > + * [3] If there are more than two DMA commands in the DMA chain, the code > + * should be: > + * ...... > + * ->device_prep_slave_sg(0); // First > + * ...... > + * ->device_prep_slave_sg(DMA_PREP_INTERRUPT [| DMA_CTRL_ACK]); > + * ...... > + * ->device_prep_slave_sg(DMA_PREP_INTERRUPT | DMA_CTRL_ACK); // Last > + * ...... > + */ > static struct dma_async_tx_descriptor *mxs_dma_prep_slave_sg( > struct dma_chan *chan, struct scatterlist *sgl, > unsigned int sg_len, enum dma_transfer_direction direction, > - unsigned long append) > + unsigned long flags) > { > struct mxs_dma_chan *mxs_chan = to_mxs_dma_chan(chan); > struct mxs_dma_engine *mxs_dma = mxs_chan->mxs_dma; > @@ -360,6 +382,7 @@ static struct dma_async_tx_descriptor *mxs_dma_prep_slave_sg( > struct scatterlist *sg; > int i, j; > u32 *pio; > + bool append = flags & DMA_PREP_INTERRUPT; > int idx = append ? mxs_chan->desc_count : 0; > > if (mxs_chan->status == DMA_IN_PROGRESS && !append) > @@ -386,7 +409,6 @@ static struct dma_async_tx_descriptor *mxs_dma_prep_slave_sg( > ccw->bits |= CCW_CHAIN; > ccw->bits &= ~CCW_IRQ; > ccw->bits &= ~CCW_DEC_SEM; > - ccw->bits &= ~CCW_WAIT4END; > } else { > idx = 0; > } > @@ -401,7 +423,8 @@ static struct dma_async_tx_descriptor *mxs_dma_prep_slave_sg( > ccw->bits = 0; > ccw->bits |= CCW_IRQ; > ccw->bits |= CCW_DEC_SEM; > - ccw->bits |= CCW_WAIT4END; > + if (flags & DMA_CTRL_ACK) > + ccw->bits |= CCW_WAIT4END; > ccw->bits |= CCW_HALT_ON_TERM; > ccw->bits |= CCW_TERM_FLUSH; > ccw->bits |= BF_CCW(sg_len, PIO_NUM); > @@ -432,7 +455,8 @@ static struct dma_async_tx_descriptor *mxs_dma_prep_slave_sg( > ccw->bits &= ~CCW_CHAIN; > ccw->bits |= CCW_IRQ; > ccw->bits |= CCW_DEC_SEM; > - ccw->bits |= CCW_WAIT4END; > + if (flags & DMA_CTRL_ACK) > + ccw->bits |= CCW_WAIT4END; > } > } > } > diff --git a/drivers/mmc/host/mxs-mmc.c b/drivers/mmc/host/mxs-mmc.c > index e5ea2b1..4062812 100644 > --- a/drivers/mmc/host/mxs-mmc.c > +++ b/drivers/mmc/host/mxs-mmc.c > @@ -305,7 +305,7 @@ static irqreturn_t mxs_mmc_irq_handler(int irq, void *dev_id) > } > > static struct dma_async_tx_descriptor *mxs_mmc_prep_dma( > - struct mxs_mmc_host *host, unsigned int append) > + struct mxs_mmc_host *host, unsigned long flags) > { > struct dma_async_tx_descriptor *desc; > struct mmc_data *data = host->data; > @@ -325,7 +325,7 @@ static struct dma_async_tx_descriptor *mxs_mmc_prep_dma( > } > > desc = host->dmach->device->device_prep_slave_sg(host->dmach, > - sgl, sg_len, host->slave_dirn, append); > + sgl, sg_len, host->slave_dirn, flags); > if (desc) { > desc->callback = mxs_mmc_dma_irq_callback; > desc->callback_param = host; > @@ -358,7 +358,7 @@ static void mxs_mmc_bc(struct mxs_mmc_host *host) > host->ssp_pio_words[2] = cmd1; > host->dma_dir = DMA_NONE; > host->slave_dirn = DMA_TRANS_NONE; > - desc = mxs_mmc_prep_dma(host, 0); > + desc = mxs_mmc_prep_dma(host, DMA_CTRL_ACK); > if (!desc) > goto out; > > @@ -398,7 +398,7 @@ static void mxs_mmc_ac(struct mxs_mmc_host *host) > host->ssp_pio_words[2] = cmd1; > host->dma_dir = DMA_NONE; > host->slave_dirn = DMA_TRANS_NONE; > - desc = mxs_mmc_prep_dma(host, 0); > + desc = mxs_mmc_prep_dma(host, DMA_CTRL_ACK); > if (!desc) > goto out; > > @@ -526,7 +526,7 @@ static void mxs_mmc_adtc(struct mxs_mmc_host *host) > host->data = data; > host->dma_dir = dma_data_dir; > host->slave_dirn = slave_dirn; > - desc = mxs_mmc_prep_dma(host, 1); > + desc = mxs_mmc_prep_dma(host, DMA_PREP_INTERRUPT | DMA_CTRL_ACK); > if (!desc) > goto out; > > diff --git a/drivers/mtd/nand/gpmi-nand/gpmi-lib.c b/drivers/mtd/nand/gpmi-nand/gpmi-lib.c > index cbcf022..420ca08 100644 > --- a/drivers/mtd/nand/gpmi-nand/gpmi-lib.c > +++ b/drivers/mtd/nand/gpmi-nand/gpmi-lib.c > @@ -850,7 +850,9 @@ int gpmi_send_command(struct gpmi_nand_data *this) > sg_init_one(sgl, this->cmd_buffer, this->command_length); > dma_map_sg(this->dev, sgl, 1, DMA_TO_DEVICE); > desc = channel->device->device_prep_slave_sg(channel, > - sgl, 1, DMA_MEM_TO_DEV, 1); > + sgl, 1, DMA_MEM_TO_DEV, > + DMA_PREP_INTERRUPT | DMA_CTRL_ACK); > + > if (!desc) { > pr_err("step 2 error\n"); > return -1; > @@ -892,7 +894,8 @@ int gpmi_send_data(struct gpmi_nand_data *this) > /* [2] send DMA request */ > prepare_data_dma(this, DMA_TO_DEVICE); > desc = channel->device->device_prep_slave_sg(channel, &this->data_sgl, > - 1, DMA_MEM_TO_DEV, 1); > + 1, DMA_MEM_TO_DEV, > + DMA_PREP_INTERRUPT | DMA_CTRL_ACK); > if (!desc) { > pr_err("step 2 error\n"); > return -1; > @@ -928,7 +931,8 @@ int gpmi_read_data(struct gpmi_nand_data *this) > /* [2] : send DMA request */ > prepare_data_dma(this, DMA_FROM_DEVICE); > desc = channel->device->device_prep_slave_sg(channel, &this->data_sgl, > - 1, DMA_DEV_TO_MEM, 1); > + 1, DMA_DEV_TO_MEM, > + DMA_PREP_INTERRUPT | DMA_CTRL_ACK); > if (!desc) { > pr_err("step 2 error\n"); > return -1; > @@ -975,7 +979,8 @@ int gpmi_send_page(struct gpmi_nand_data *this, > > desc = channel->device->device_prep_slave_sg(channel, > (struct scatterlist *)pio, > - ARRAY_SIZE(pio), DMA_TRANS_NONE, 0); > + ARRAY_SIZE(pio), DMA_TRANS_NONE, > + DMA_CTRL_ACK); > if (!desc) { > pr_err("step 2 error\n"); > return -1; > @@ -1039,7 +1044,8 @@ int gpmi_read_page(struct gpmi_nand_data *this, > pio[5] = auxiliary; > desc = channel->device->device_prep_slave_sg(channel, > (struct scatterlist *)pio, > - ARRAY_SIZE(pio), DMA_TRANS_NONE, 1); > + ARRAY_SIZE(pio), DMA_TRANS_NONE, > + DMA_PREP_INTERRUPT | DMA_CTRL_ACK); > if (!desc) { > pr_err("step 2 error\n"); > return -1; > @@ -1059,7 +1065,8 @@ int gpmi_read_page(struct gpmi_nand_data *this, > pio[2] = 0; /* Set GPMI_HW_GPMI_ECCCTRL, disable the BCH. */ > desc = channel->device->device_prep_slave_sg(channel, > (struct scatterlist *)pio, 3, I have the mainline code show above line as (struct scatterlist *)pio, 2, That's why I think you got a wrong base. But I tested it with manually applying it. So other than the base issue, Acked-by: Shawn Guo <shawn.guo(a)linaro.org> Regards, Shawn > - DMA_TRANS_NONE, 1); > + DMA_TRANS_NONE, > + DMA_PREP_INTERRUPT | DMA_CTRL_ACK); > if (!desc) { > pr_err("step 3 error\n"); > return -1; > -- > 1.7.0.4 > >

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[alsa-devel] [PATCH] ASoC: dapm: Convert pin switches to use snd_soc_card
by Mark Brown 15 Feb '12

15 Feb '12

Since the addition of the non-CODEC control adds card controls like the DAPM pin switch have been broken as they are expecting the private data for the control to be the CODEC but it's now the card. Fix that for the pin switches, an audit of other drivers is required. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-dapm.c | 20 ++++++++++---------- 1 files changed, 10 insertions(+), 10 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 6fc488e..9b796e3 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2644,15 +2644,15 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_info_pin_switch); int snd_soc_dapm_get_pin_switch(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct snd_soc_card *card = snd_kcontrol_chip(kcontrol); const char *pin = (const char *)kcontrol->private_value; - mutex_lock(&codec->mutex); + mutex_lock(&card->mutex); ucontrol->value.integer.value[0] = - snd_soc_dapm_get_pin_status(&codec->dapm, pin); + snd_soc_dapm_get_pin_status(&card->dapm, pin); - mutex_unlock(&codec->mutex); + mutex_unlock(&card->mutex); return 0; } @@ -2667,19 +2667,19 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_get_pin_switch); int snd_soc_dapm_put_pin_switch(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct snd_soc_card *card = snd_kcontrol_chip(kcontrol); const char *pin = (const char *)kcontrol->private_value; - mutex_lock(&codec->mutex); + mutex_lock(&card->mutex); if (ucontrol->value.integer.value[0]) - snd_soc_dapm_enable_pin(&codec->dapm, pin); + snd_soc_dapm_enable_pin(&card->dapm, pin); else - snd_soc_dapm_disable_pin(&codec->dapm, pin); + snd_soc_dapm_disable_pin(&card->dapm, pin); - snd_soc_dapm_sync(&codec->dapm); + snd_soc_dapm_sync(&card->dapm); - mutex_unlock(&codec->mutex); + mutex_unlock(&card->mutex); return 0; } -- 1.7.9.rc1

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[alsa-devel] [PATCH v2] ASoC: core - add platform DAPM debugfs support
by Liam Girdwood 15 Feb '12

15 Feb '12

From: Sebastien Guiriec <s-guiriec(a)ti.com> Allow platform widgets to be visible in debugfs like codec widgets. Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- include/sound/soc.h | 5 +++++ sound/soc/soc-core.c | 32 ++++++++++++++++++++++++++++++++ 2 files changed, 37 insertions(+), 0 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 4f67e01..8fa4dca 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -706,6 +706,11 @@ struct snd_soc_platform { struct list_head card_list; struct snd_soc_dapm_context dapm; + +#ifdef CONFIG_DEBUG_FS + struct dentry *debugfs_platform_root; + struct dentry *debugfs_dapm; +#endif }; struct snd_soc_dai_link { diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 8bd9995..276b68a 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -302,6 +302,27 @@ static void soc_cleanup_codec_debugfs(struct snd_soc_codec *codec) debugfs_remove_recursive(codec->debugfs_codec_root); } +static void soc_init_platform_debugfs(struct snd_soc_platform *platform) +{ + struct dentry *debugfs_card_root = platform->card->debugfs_card_root; + + platform->debugfs_platform_root = debugfs_create_dir(platform->name, + debugfs_card_root); + if (!platform->debugfs_platform_root) { + dev_warn(platform->dev, + "Failed to create platform debugfs directory\n"); + return; + } + + snd_soc_dapm_debugfs_init(&platform->dapm, + platform->debugfs_platform_root); +} + +static void soc_cleanup_platform_debugfs(struct snd_soc_platform *platform) +{ + debugfs_remove_recursive(platform->debugfs_platform_root); +} + static ssize_t codec_list_read_file(struct file *file, char __user *user_buf, size_t count, loff_t *ppos) { @@ -435,6 +456,14 @@ static inline void soc_cleanup_codec_debugfs(struct snd_soc_codec *codec) { } +static inline void soc_init_platform_debugfs(struct snd_soc_platform *platform) +{ +} + +static inline void soc_cleanup_platform_debugfs(struct snd_soc_platform *platform) +{ +} + static inline void soc_init_card_debugfs(struct snd_soc_card *card) { } @@ -922,6 +951,7 @@ static void soc_remove_dai_link(struct snd_soc_card *card, int num, int order) /* Make sure all DAPM widgets are freed */ snd_soc_dapm_free(&platform->dapm); + soc_cleanup_platform_debugfs(platform); platform->probed = 0; list_del(&platform->card_list); module_put(platform->dev->driver->owner); @@ -1039,6 +1069,8 @@ static int soc_probe_platform(struct snd_soc_card *card, if (!try_module_get(platform->dev->driver->owner)) return -ENODEV; + soc_init_platform_debugfs(platform); + if (driver->dapm_widgets) snd_soc_dapm_new_controls(&platform->dapm, driver->dapm_widgets, driver->num_dapm_widgets); -- 1.7.5.4

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[alsa-devel] [PATCH] ASoC: dapm - Make DAPM reset code a separate function.
by Liam Girdwood 15 Feb '12

15 Feb '12

It's useful to export the DAPM reset as a static function for future use by other DAPM functions. e.g. The dynamic PCM query widgets resets the DAPM graph before working out active paths. Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- sound/soc/soc-dapm.c | 21 ++++++++++++++------- 1 files changed, 14 insertions(+), 7 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 0c94027..227887e 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -169,6 +169,19 @@ static inline struct snd_soc_card *dapm_get_soc_card( return NULL; } +static void dapm_reset(struct snd_soc_card *card) +{ + struct snd_soc_dapm_widget *w; + + memset(&card->dapm_stats, 0, sizeof(card->dapm_stats)); + + list_for_each_entry(w, &card->widgets, list) { + w->power_checked = false; + w->inputs = -1; + w->outputs = -1; + } +} + static int soc_widget_read(struct snd_soc_dapm_widget *w, int reg) { if (w->codec) @@ -1402,13 +1415,7 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event) } } - memset(&card->dapm_stats, 0, sizeof(card->dapm_stats)); - - list_for_each_entry(w, &card->widgets, list) { - w->power_checked = false; - w->inputs = -1; - w->outputs = -1; - } + dapm_reset(card); /* Check which widgets we need to power and store them in * lists indicating if they should be powered up or down. We -- 1.7.5.4

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[alsa-devel] [PATCH] ALSA: usb-audio: avoid integer overflow in create_fixed_stream_quirk()
by Xi Wang 15 Feb '12

15 Feb '12

A malicious USB device could feed in a large nr_rates value. This would cause the subsequent call to kmemdup() to allocate a smaller buffer than expected, leading to out-of-bounds access. This patch validates the nr_rates value and reuses the limit introduced in commit 4fa0e81b ("ALSA: usb-audio: fix possible hang and overflow in parse_uac2_sample_rate_range()"). Signed-off-by: Xi Wang <xi.wang(a)gmail.com> --- sound/usb/card.h | 1 + sound/usb/format.c | 4 +--- sound/usb/quirks.c | 6 +++++- 3 files changed, 7 insertions(+), 4 deletions(-) diff --git a/sound/usb/card.h b/sound/usb/card.h index a39edcc..da5fa1a 100644 --- a/sound/usb/card.h +++ b/sound/usb/card.h @@ -1,6 +1,7 @@ #ifndef __USBAUDIO_CARD_H #define __USBAUDIO_CARD_H +#define MAX_NR_RATES 1024 #define MAX_PACKS 20 #define MAX_PACKS_HS (MAX_PACKS * 8) /* in high speed mode */ #define MAX_URBS 8 diff --git a/sound/usb/format.c b/sound/usb/format.c index e09aba1..ddfef57 100644 --- a/sound/usb/format.c +++ b/sound/usb/format.c @@ -209,8 +209,6 @@ static int parse_audio_format_rates_v1(struct snd_usb_audio *chip, struct audiof return 0; } -#define MAX_UAC2_NR_RATES 1024 - /* * Helper function to walk the array of sample rate triplets reported by * the device. The problem is that we need to parse whole array first to @@ -255,7 +253,7 @@ static int parse_uac2_sample_rate_range(struct audioformat *fp, int nr_triplets, fp->rates |= snd_pcm_rate_to_rate_bit(rate); nr_rates++; - if (nr_rates >= MAX_UAC2_NR_RATES) { + if (nr_rates >= MAX_NR_RATES) { snd_printk(KERN_ERR "invalid uac2 rates\n"); break; } diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index a3ddac0..2781726 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -132,10 +132,14 @@ static int create_fixed_stream_quirk(struct snd_usb_audio *chip, unsigned *rate_table = NULL; fp = kmemdup(quirk->data, sizeof(*fp), GFP_KERNEL); - if (! fp) { + if (!fp) { snd_printk(KERN_ERR "cannot memdup\n"); return -ENOMEM; } + if (fp->nr_rates > MAX_NR_RATES) { + kfree(fp); + return -EINVAL; + } if (fp->nr_rates > 0) { rate_table = kmemdup(fp->rate_table, sizeof(int) * fp->nr_rates, GFP_KERNEL); -- 1.7.5.4

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Re: [alsa-devel] [PATCH v2] [trivial] usx2y: Fix typo in usbusx2yaudio.c and usx2yhwdeppcm.c
by Takashi Iwai 15 Feb '12

15 Feb '12

At Wed, 15 Feb 2012 00:38:55 +0900, Masanari Iida wrote: > > Correct spelling "propably" to "probably" and "activ" to "active" > in sound/usb/usx2y/usbusx2yaudio.c and usx2yhwdeppcm.c > > Signed-off-by: Masanari Iida <standby24x7(a)gmail.com> Thanks, applied now to topic/misc branch of sound git tree. Takashi > --- > sound/usb/usx2y/usbusx2yaudio.c | 4 ++-- > sound/usb/usx2y/usx2yhwdeppcm.c | 2 +- > 2 files changed, 3 insertions(+), 3 deletions(-) > > diff --git a/sound/usb/usx2y/usbusx2yaudio.c b/sound/usb/usx2y/usbusx2yaudio.c > index 6ffb371..520ef96 100644 > --- a/sound/usb/usx2y/usbusx2yaudio.c > +++ b/sound/usb/usx2y/usbusx2yaudio.c > @@ -80,7 +80,7 @@ static int usX2Y_urb_capt_retire(struct snd_usX2Y_substream *subs) > cp = (unsigned char*)urb->transfer_buffer + urb->iso_frame_desc[i].offset; > if (urb->iso_frame_desc[i].status) { /* active? hmm, skip this */ > snd_printk(KERN_ERR "active frame status %i. " > - "Most propably some hardware problem.\n", > + "Most probably some hardware problem.\n", > urb->iso_frame_desc[i].status); > return urb->iso_frame_desc[i].status; > } > @@ -300,7 +300,7 @@ static void usX2Y_error_sequence(struct usX2Ydev *usX2Y, > { > snd_printk(KERN_ERR > "Sequence Error!(hcd_frame=%i ep=%i%s;wait=%i,frame=%i).\n" > -"Most propably some urb of usb-frame %i is still missing.\n" > +"Most probably some urb of usb-frame %i is still missing.\n" > "Cause could be too long delays in usb-hcd interrupt handling.\n", > usb_get_current_frame_number(usX2Y->dev), > subs->endpoint, usb_pipein(urb->pipe) ? "in" : "out", > diff --git a/sound/usb/usx2y/usx2yhwdeppcm.c b/sound/usb/usx2y/usx2yhwdeppcm.c > index a51340f..8e40b6e 100644 > --- a/sound/usb/usx2y/usx2yhwdeppcm.c > +++ b/sound/usb/usx2y/usx2yhwdeppcm.c > @@ -74,7 +74,7 @@ static int usX2Y_usbpcm_urb_capt_retire(struct snd_usX2Y_substream *subs) > } > for (i = 0; i < nr_of_packs(); i++) { > if (urb->iso_frame_desc[i].status) { /* active? hmm, skip this */ > - snd_printk(KERN_ERR "activ frame status %i. Most propably some hardware problem.\n", urb->iso_frame_desc[i].status); > + snd_printk(KERN_ERR "active frame status %i. Most probably some hardware problem.\n", urb->iso_frame_desc[i].status); > return urb->iso_frame_desc[i].status; > } > lens += urb->iso_frame_desc[i].actual_length / usX2Y->stride; > -- > 1.7.6.5 >

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