From a115c72802c37351b6d87dfb62938d2ad440eef4 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 4 Aug 2011 13:23:38 +0900 Subject: ASoC: Move WM8962 CLKREG_OVD earlier When the clocking registers are not overriden some of the registers are not writable. Signed-off-by: Mark Brown Acked-by: Liam Girdwood Cc: stable@kernel.org --- sound/soc/codecs/wm8962.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 60d740ebeb5b..28650edfdebb 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2927,10 +2927,6 @@ static int wm8962_set_bias_level(struct snd_soc_codec *codec, WM8962_BIAS_ENA | 0x180); msleep(5); - - snd_soc_update_bits(codec, WM8962_CLOCKING2, - WM8962_CLKREG_OVD, - WM8962_CLKREG_OVD); } /* VMID 2*250k */ @@ -3868,6 +3864,10 @@ static int wm8962_probe(struct snd_soc_codec *codec) */ snd_soc_update_bits(codec, WM8962_CLOCKING2, WM8962_SYSCLK_ENA, 0); + /* Ensure we have soft control over all registers */ + snd_soc_update_bits(codec, WM8962_CLOCKING2, + WM8962_CLKREG_OVD, WM8962_CLKREG_OVD); + regulator_bulk_disable(ARRAY_SIZE(wm8962->supplies), wm8962->supplies); if (pdata) { -- cgit v1.2.3 From fd049755636a8b2cc084e088967dd566467ccebc Mon Sep 17 00:00:00 2001 From: Vasily Khoruzhick Date: Fri, 12 Aug 2011 17:52:59 +0300 Subject: ASoC: h1940: Fix compilation error due to missing header MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Add linux/types.h to fix this compilation error: In file included from arch/arm/mach-s3c2410/include/mach/gpio-fns.h:27:0, from arch/arm/mach-s3c2410/include/mach/gpio.h:27, from /home/anarsoul/work/pda-linux/linux-next/arch/arm/include/asm/gpio.h:5, from include/linux/gpio.h:18, from sound/soc/samsung/rx1950_uda1380.c:20: arch/arm/plat-samsung/include/plat/gpio-cfg.h:29:34: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘__attribute__’ before ‘s3c_gpio_pull_t’ arch/arm/plat-samsung/include/plat/gpio-cfg.h:30:34: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘__attribute__’ before ‘s5p_gpio_drvstr_t’ arch/arm/plat-samsung/include/plat/gpio-cfg.h:57:2: error: expected specifier-qualifier-list before ‘s3c_gpio_pull_t’ arch/arm/plat-samsung/include/plat/gpio-cfg.h:148:47: error: expected declaration specifiers or ‘...’ before ‘s3c_gpio_pull_t’ arch/arm/plat-samsung/include/plat/gpio-cfg.h:156:24: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘__attribute__’ before ‘s3c_gpio_getpull’ arch/arm/plat-samsung/include/plat/gpio-cfg.h:175:24: error: expected declaration specifiers or ‘...’ before ‘s3c_gpio_pull_t’ arch/arm/plat-samsung/include/plat/gpio-cfg.h: In function ‘s3c_gpio_cfgrange_nopull’: arch/arm/plat-samsung/include/plat/gpio-cfg.h:180:47: error: ‘s3c_gpio_pull_t’ undeclared (first use in this function) arch/arm/plat-samsung/include/plat/gpio-cfg.h:180:47: note: each undeclared identifier is reported only once for each function it appears in arch/arm/plat-samsung/include/plat/gpio-cfg.h:180:47: error: expected ‘)’ before numeric constant arch/arm/plat-samsung/include/plat/gpio-cfg.h:180:47: error: too many arguments to function ‘s3c_gpio_cfgall_range’ arch/arm/plat-samsung/include/plat/gpio-cfg.h:174:12: note: declared here arch/arm/plat-samsung/include/plat/gpio-cfg.h: At top level: arch/arm/plat-samsung/include/plat/gpio-cfg.h:199:26: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘__attribute__’ before ‘s5p_gpio_get_drvstr’ arch/arm/plat-samsung/include/plat/gpio-cfg.h:210:50: error: expected declaration specifiers or ‘...’ before ‘s5p_gpio_drvstr_t’ Signed-off-by: Vasily Khoruzhick Acked-by: Jassi Brar Signed-off-by: Mark Brown --- sound/soc/samsung/h1940_uda1380.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/samsung/h1940_uda1380.c b/sound/soc/samsung/h1940_uda1380.c index 241f55d00660..c6c65892294e 100644 --- a/sound/soc/samsung/h1940_uda1380.c +++ b/sound/soc/samsung/h1940_uda1380.c @@ -13,6 +13,7 @@ * */ +#include #include #include -- cgit v1.2.3 From b8487928f5ca2976e4cb8d329943af849d2b6197 Mon Sep 17 00:00:00 2001 From: Vasily Khoruzhick Date: Fri, 12 Aug 2011 17:53:00 +0300 Subject: ASoC: rx1950: Fix compilation error due to missing header MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Add linux/types.h to fix this compilation error: In file included from arch/arm/mach-s3c2410/include/mach/gpio-fns.h:27:0, from arch/arm/mach-s3c2410/include/mach/gpio.h:27, from /home/anarsoul/work/pda-linux/linux-next/arch/arm/include/asm/gpio.h:5, from include/linux/gpio.h:18, from sound/soc/samsung/rx1950_uda1380.c:20: arch/arm/plat-samsung/include/plat/gpio-cfg.h:29:34: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘__attribute__’ before ‘s3c_gpio_pull_t’ arch/arm/plat-samsung/include/plat/gpio-cfg.h:30:34: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘__attribute__’ before ‘s5p_gpio_drvstr_t’ arch/arm/plat-samsung/include/plat/gpio-cfg.h:57:2: error: expected specifier-qualifier-list before ‘s3c_gpio_pull_t’ arch/arm/plat-samsung/include/plat/gpio-cfg.h:148:47: error: expected declaration specifiers or ‘...’ before ‘s3c_gpio_pull_t’ arch/arm/plat-samsung/include/plat/gpio-cfg.h:156:24: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘__attribute__’ before ‘s3c_gpio_getpull’ arch/arm/plat-samsung/include/plat/gpio-cfg.h:175:24: error: expected declaration specifiers or ‘...’ before ‘s3c_gpio_pull_t’ arch/arm/plat-samsung/include/plat/gpio-cfg.h: In function ‘s3c_gpio_cfgrange_nopull’: arch/arm/plat-samsung/include/plat/gpio-cfg.h:180:47: error: ‘s3c_gpio_pull_t’ undeclared (first use in this function) arch/arm/plat-samsung/include/plat/gpio-cfg.h:180:47: note: each undeclared identifier is reported only once for each function it appears in arch/arm/plat-samsung/include/plat/gpio-cfg.h:180:47: error: expected ‘)’ before numeric constant arch/arm/plat-samsung/include/plat/gpio-cfg.h:180:47: error: too many arguments to function ‘s3c_gpio_cfgall_range’ arch/arm/plat-samsung/include/plat/gpio-cfg.h:174:12: note: declared here arch/arm/plat-samsung/include/plat/gpio-cfg.h: At top level: arch/arm/plat-samsung/include/plat/gpio-cfg.h:199:26: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘__attribute__’ before ‘s5p_gpio_get_drvstr’ arch/arm/plat-samsung/include/plat/gpio-cfg.h:210:50: error: expected declaration specifiers or ‘...’ before ‘s5p_gpio_drvstr_t’ Signed-off-by: Vasily Khoruzhick Acked-by: Jassi Brar Signed-off-by: Mark Brown --- sound/soc/samsung/rx1950_uda1380.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/samsung/rx1950_uda1380.c b/sound/soc/samsung/rx1950_uda1380.c index 1e574a5d440d..bc8c1676459f 100644 --- a/sound/soc/samsung/rx1950_uda1380.c +++ b/sound/soc/samsung/rx1950_uda1380.c @@ -17,6 +17,7 @@ * */ +#include #include #include -- cgit v1.2.3 From d2b4c7bd7eabfaa2e3e5b8107d5eeb56ac879813 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Sat, 13 Aug 2011 19:15:01 +0800 Subject: ASoC: soc-jack: Fix checking return value of request_any_context_irq request_any_context_irq() returns a negative value on failure. On success, it returns either IRQC_IS_HARDIRQ or IRQC_IS_NESTED. Signed-off-by: Axel Lin Signed-off-by: Mark Brown Cc: stable@kernel.orG --- sound/soc/soc-jack.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index 7c17b98d5846..38b00131b2fe 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -327,7 +327,7 @@ int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count, IRQF_TRIGGER_FALLING, gpios[i].name, &gpios[i]); - if (ret) + if (ret < 0) goto err; if (gpios[i].wake) { -- cgit v1.2.3 From 161d55c3ec4c7e26c96b11dc86caea0b3c9c6b0f Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Sat, 13 Aug 2011 11:33:08 +0800 Subject: ASoC: sta32x: Fix a memory leak if snd_soc_register_codec fails Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/sta32x.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c index 409d89d1f34c..fbd7eb9e61ce 100644 --- a/sound/soc/codecs/sta32x.c +++ b/sound/soc/codecs/sta32x.c @@ -857,6 +857,7 @@ static __devinit int sta32x_i2c_probe(struct i2c_client *i2c, ret = snd_soc_register_codec(&i2c->dev, &sta32x_codec, &sta32x_dai, 1); if (ret != 0) { dev_err(&i2c->dev, "Failed to register codec (%d)\n", ret); + kfree(sta32x); return ret; } -- cgit v1.2.3 From bf545ed72f2eeac664695a8ea2199d9ddaef6020 Mon Sep 17 00:00:00 2001 From: Scott Jiang Date: Fri, 12 Aug 2011 18:04:10 -0400 Subject: ASoC: ad193x: fix registers definition fix dac word len mask and adc tdm fmt shift value Signed-off-by: Scott Jiang Signed-off-by: Mark Brown Cc: stable@kernel.org --- sound/soc/codecs/ad193x.h | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/ad193x.h b/sound/soc/codecs/ad193x.h index 9747b5497877..c1029d242c84 100644 --- a/sound/soc/codecs/ad193x.h +++ b/sound/soc/codecs/ad193x.h @@ -34,7 +34,7 @@ #define AD193X_DAC_LEFT_HIGH (1 << 3) #define AD193X_DAC_BCLK_INV (1 << 7) #define AD193X_DAC_CTRL2 0x804 -#define AD193X_DAC_WORD_LEN_MASK 0xC +#define AD193X_DAC_WORD_LEN_MASK 0x18 #define AD193X_DAC_MASTER_MUTE 1 #define AD193X_DAC_CHNL_MUTE 0x805 #define AD193X_DACL1_MUTE 0 @@ -63,7 +63,7 @@ #define AD193X_ADC_CTRL1 0x80f #define AD193X_ADC_SERFMT_MASK 0x60 #define AD193X_ADC_SERFMT_STEREO (0 << 5) -#define AD193X_ADC_SERFMT_TDM (1 << 2) +#define AD193X_ADC_SERFMT_TDM (1 << 5) #define AD193X_ADC_SERFMT_AUX (2 << 5) #define AD193X_ADC_WORD_LEN_MASK 0x3 #define AD193X_ADC_CTRL2 0x810 -- cgit v1.2.3 From 95c93d8525ebce1024bda7316f602ae45c36cd6f Mon Sep 17 00:00:00 2001 From: Scott Jiang Date: Fri, 12 Aug 2011 18:04:11 -0400 Subject: ASoC: ad193x: fix dac word len setting dac word len value should left shift before setting Signed-off-by: Scott Jiang Acked-by: Barry Song <21cnbao@gmail.com> Signed-off-by: Mark Brown Cc: stable@kernel.org --- sound/soc/codecs/ad193x.c | 3 ++- sound/soc/codecs/ad193x.h | 1 + 2 files changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c index 2374ca5ffe68..f1a8be58255b 100644 --- a/sound/soc/codecs/ad193x.c +++ b/sound/soc/codecs/ad193x.c @@ -307,7 +307,8 @@ static int ad193x_hw_params(struct snd_pcm_substream *substream, snd_soc_write(codec, AD193X_PLL_CLK_CTRL0, reg); reg = snd_soc_read(codec, AD193X_DAC_CTRL2); - reg = (reg & (~AD193X_DAC_WORD_LEN_MASK)) | word_len; + reg = (reg & (~AD193X_DAC_WORD_LEN_MASK)) + | (word_len << AD193X_DAC_WORD_LEN_SHFT); snd_soc_write(codec, AD193X_DAC_CTRL2, reg); reg = snd_soc_read(codec, AD193X_ADC_CTRL1); diff --git a/sound/soc/codecs/ad193x.h b/sound/soc/codecs/ad193x.h index c1029d242c84..cccc2e8e5fbd 100644 --- a/sound/soc/codecs/ad193x.h +++ b/sound/soc/codecs/ad193x.h @@ -34,6 +34,7 @@ #define AD193X_DAC_LEFT_HIGH (1 << 3) #define AD193X_DAC_BCLK_INV (1 << 7) #define AD193X_DAC_CTRL2 0x804 +#define AD193X_DAC_WORD_LEN_SHFT 3 #define AD193X_DAC_WORD_LEN_MASK 0x18 #define AD193X_DAC_MASTER_MUTE 1 #define AD193X_DAC_CHNL_MUTE 0x805 -- cgit v1.2.3 From 25ea524bed0202f823a0adcbbda68e86a22e3670 Mon Sep 17 00:00:00 2001 From: Scott Jiang Date: Fri, 12 Aug 2011 18:04:12 -0400 Subject: ASoC: ad193x: fix system clock system clock is 24.576MHz instead of 12.288MHz Signed-off-by: Scott Jiang Signed-off-by: Mark Brown --- sound/soc/blackfin/bf5xx-ad193x.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/blackfin/bf5xx-ad193x.c b/sound/soc/blackfin/bf5xx-ad193x.c index d6651c033cb7..a118a0fb9d81 100644 --- a/sound/soc/blackfin/bf5xx-ad193x.c +++ b/sound/soc/blackfin/bf5xx-ad193x.c @@ -56,7 +56,7 @@ static int bf5xx_ad193x_hw_params(struct snd_pcm_substream *substream, switch (params_rate(params)) { case 48000: - clk = 12288000; + clk = 24576000; break; } -- cgit v1.2.3 From 396a2e79cdbd562bf7ea48132f8d3ba8304109b2 Mon Sep 17 00:00:00 2001 From: Scott Jiang Date: Fri, 12 Aug 2011 18:04:13 -0400 Subject: ASoC: Add spi hw read function for 16 addr 8 data mode for ad193x fix [This will be used by the ad193x driver to fix the fact that the original author of the driver put a bodge for their particular chip into a the generic ASoC register I/O abstraction layer which looked like an obvious bug which ended up getting fixed in 3.0. Sadly there were no comments documenting what was going on. A minimally invasive correction to the driver is to remove the register cache support and go direct to the hardware all the time so we're adding a new feature -- broonie] Signed-off-by: Scott Jiang Signed-off-by: Mark Brown --- sound/soc/soc-io.c | 23 +++++++++++++++++++++++ 1 file changed, 23 insertions(+) diff --git a/sound/soc/soc-io.c b/sound/soc/soc-io.c index cca490c80589..a62f7dd4ba96 100644 --- a/sound/soc/soc-io.c +++ b/sound/soc/soc-io.c @@ -205,6 +205,25 @@ static unsigned int snd_soc_16_8_read_i2c(struct snd_soc_codec *codec, #define snd_soc_16_8_read_i2c NULL #endif +#if defined(CONFIG_SPI_MASTER) +static unsigned int snd_soc_16_8_read_spi(struct snd_soc_codec *codec, + unsigned int r) +{ + struct spi_device *spi = codec->control_data; + + const u16 reg = cpu_to_be16(r | 0x100); + u8 data; + int ret; + + ret = spi_write_then_read(spi, ®, 2, &data, 1); + if (ret < 0) + return 0; + return data; +} +#else +#define snd_soc_16_8_read_spi NULL +#endif + static int snd_soc_16_8_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int value) { @@ -295,6 +314,7 @@ static struct { int (*write)(struct snd_soc_codec *codec, unsigned int, unsigned int); unsigned int (*read)(struct snd_soc_codec *, unsigned int); unsigned int (*i2c_read)(struct snd_soc_codec *, unsigned int); + unsigned int (*spi_read)(struct snd_soc_codec *, unsigned int); } io_types[] = { { .addr_bits = 4, .data_bits = 12, @@ -318,6 +338,7 @@ static struct { .addr_bits = 16, .data_bits = 8, .write = snd_soc_16_8_write, .i2c_read = snd_soc_16_8_read_i2c, + .spi_read = snd_soc_16_8_read_spi, }, { .addr_bits = 16, .data_bits = 16, @@ -383,6 +404,8 @@ int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec, #ifdef CONFIG_SPI_MASTER codec->hw_write = do_spi_write; #endif + if (io_types[i].spi_read) + codec->hw_read = io_types[i].spi_read; codec->control_data = container_of(codec->dev, struct spi_device, -- cgit v1.2.3 From 0cc62e926324d4f3bd02d378baafbe73164fca35 Mon Sep 17 00:00:00 2001 From: Scott Jiang Date: Fri, 12 Aug 2011 18:04:14 -0400 Subject: ASoC: ad193x: remove cache support asoc cache layer can't support this kind of spi registers well. remove cache support and read/write registers directly Signed-off-by: Scott Jiang Signed-off-by: Mark Brown --- sound/soc/codecs/ad193x.c | 8 -------- 1 file changed, 8 deletions(-) diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c index f1a8be58255b..eedb6f5e5823 100644 --- a/sound/soc/codecs/ad193x.c +++ b/sound/soc/codecs/ad193x.c @@ -27,11 +27,6 @@ struct ad193x_priv { int sysclk; }; -/* ad193x register cache & default register settings */ -static const u8 ad193x_reg[AD193X_NUM_REGS] = { - 0, 0, 0, 0, 0, 0, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0, 0, 0, -}; - /* * AD193X volume/mute/de-emphasis etc. controls */ @@ -390,9 +385,6 @@ static int ad193x_probe(struct snd_soc_codec *codec) static struct snd_soc_codec_driver soc_codec_dev_ad193x = { .probe = ad193x_probe, - .reg_cache_default = ad193x_reg, - .reg_cache_size = AD193X_NUM_REGS, - .reg_word_size = sizeof(u16), }; #if defined(CONFIG_SPI_MASTER) -- cgit v1.2.3 From 25b7679136fd85b1e5197e36a0ca126163e89590 Mon Sep 17 00:00:00 2001 From: Sascha Hauer Date: Wed, 17 Aug 2011 09:20:01 +0200 Subject: ASoC: Fix check for symmetric rate enforcement The ASoC core tries to not enforce symmetric rates when two streams open simultaneously. It does so by checking rtd->rate being zero. This works exactly once after booting because it is not set to zero again when the streams close. Fix this by setting rtd->rate when no active stream is left. [This leads to lots of warnings about not enforcing the symmetry in some situations as there's a race in the userspace API where we know we've got two applications but don't know what rates they want to set. -- broonie ] Signed-off-by: Sascha Hauer Signed-off-by: Mark Brown --- sound/soc/soc-pcm.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index b5759397afa3..2879c883eebc 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -290,6 +290,9 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) codec_dai->active--; codec->active--; + if (!cpu_dai->active && !codec_dai->active) + rtd->rate = 0; + /* Muting the DAC suppresses artifacts caused during digital * shutdown, for example from stopping clocks. */ -- cgit v1.2.3 From e574044acbad7421879270a80acd337459c94cc8 Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Thu, 18 Aug 2011 15:02:47 +0300 Subject: ASoC: omap: Fix build errors in ams-delta Fix "error: too few arguments to function 'ams_delta_set_bias_level'" build errors in ams-delta.c that were introduced after commit d4c6005 ("ASoC: Add context parameter to card DAPM callbacks") by adding dapm context to ams_delta_set_bias_level calls. Signed-off-by: Jarkko Nikula Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/omap/ams-delta.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c index 30fe0d0efe1c..0aa475f92efa 100644 --- a/sound/soc/omap/ams-delta.c +++ b/sound/soc/omap/ams-delta.c @@ -514,7 +514,7 @@ static int ams_delta_cx20442_init(struct snd_soc_pcm_runtime *rtd) } /* Set codec bias level */ - ams_delta_set_bias_level(card, SND_SOC_BIAS_STANDBY); + ams_delta_set_bias_level(card, dapm, SND_SOC_BIAS_STANDBY); /* Add hook switch - can be used to control the codec from userspace * even if line discipline fails */ @@ -649,7 +649,9 @@ static void __exit ams_delta_module_exit(void) ams_delta_hook_switch_gpios); /* Keep modem power on */ - ams_delta_set_bias_level(&ams_delta_audio_card, SND_SOC_BIAS_STANDBY); + ams_delta_set_bias_level(&ams_delta_audio_card, + &ams_delta_audio_card.rtd[0].codec->dapm, + SND_SOC_BIAS_STANDBY); platform_device_unregister(cx20442_platform_device); platform_device_unregister(ams_delta_audio_platform_device); -- cgit v1.2.3 From b6acf013bdc6f6ff9643030add85832d44034a28 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sat, 20 Aug 2011 09:14:45 +0200 Subject: ALSA: hda - Don't spew too many ELD errors Currently HD-audio driver shows the all error ELD byte as an error in the kernel message. This is annoying when the video driver doesn't set the correct ELD from the beginning. e.g. radeon sends a zero-byte data, but we still check ELD with the fixed 128 byte as a workaround for some broken devices, it spews 128-times errors. For avoiding this, the driver aborts reading when the first byte is invalid. In such a case, the whole data is certainly invalid. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_eld.c | 31 +++++++++++++++++++------------ 1 file changed, 19 insertions(+), 12 deletions(-) diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c index 28ce17d09c33..c34f730f4815 100644 --- a/sound/pci/hda/hda_eld.c +++ b/sound/pci/hda/hda_eld.c @@ -144,25 +144,17 @@ static int cea_sampling_frequencies[8] = { SNDRV_PCM_RATE_192000, /* 7: 192000Hz */ }; -static unsigned char hdmi_get_eld_byte(struct hda_codec *codec, hda_nid_t nid, +static unsigned int hdmi_get_eld_data(struct hda_codec *codec, hda_nid_t nid, int byte_index) { unsigned int val; val = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_HDMI_ELDD, byte_index); - #ifdef BE_PARANOID printk(KERN_INFO "HDMI: ELD data byte %d: 0x%x\n", byte_index, val); #endif - - if ((val & AC_ELDD_ELD_VALID) == 0) { - snd_printd(KERN_INFO "HDMI: invalid ELD data byte %d\n", - byte_index); - val = 0; - } - - return val & AC_ELDD_ELD_DATA; + return val; } #define GRAB_BITS(buf, byte, lowbit, bits) \ @@ -344,11 +336,26 @@ int snd_hdmi_get_eld(struct hdmi_eld *eld, if (!buf) return -ENOMEM; - for (i = 0; i < size; i++) - buf[i] = hdmi_get_eld_byte(codec, nid, i); + for (i = 0; i < size; i++) { + unsigned int val = hdmi_get_eld_data(codec, nid, i); + if (!(val & AC_ELDD_ELD_VALID)) { + if (!i) { + snd_printd(KERN_INFO + "HDMI: invalid ELD data\n"); + ret = -EINVAL; + goto error; + } + snd_printd(KERN_INFO + "HDMI: invalid ELD data byte %d\n", i); + val = 0; + } else + val &= AC_ELDD_ELD_DATA; + buf[i] = val; + } ret = hdmi_update_eld(eld, buf, size); +error: kfree(buf); return ret; } -- cgit v1.2.3 From 1b004d03d8670bdd871e0f297ed20bc510e404de Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sat, 20 Aug 2011 09:19:59 +0200 Subject: ALSA: hda - Fix error check from snd_hda_get_conn_index() in patch_cirrus.c snd_hda_get_conn_index() returns a negative value while the current code stores it in an unsigned int. It must be stored in a signed integer. Reported-by: Jesper Juhl Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_cirrus.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 47d6ffc9b5b5..d6c93d92b550 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -375,7 +375,7 @@ static int is_ext_mic(struct hda_codec *codec, unsigned int idx) static hda_nid_t get_adc(struct hda_codec *codec, hda_nid_t pin, unsigned int *idxp) { - int i; + int i, idx; hda_nid_t nid; nid = codec->start_nid; @@ -384,9 +384,11 @@ static hda_nid_t get_adc(struct hda_codec *codec, hda_nid_t pin, type = get_wcaps_type(get_wcaps(codec, nid)); if (type != AC_WID_AUD_IN) continue; - *idxp = snd_hda_get_conn_index(codec, nid, pin, false); - if (*idxp >= 0) + idx = snd_hda_get_conn_index(codec, nid, pin, false); + if (idx >= 0) { + *idxp = idx; return nid; + } } return 0; } -- cgit v1.2.3 From de75577c8c3ab733f808c65e1a9d55882efde68e Mon Sep 17 00:00:00 2001 From: Julia Lawall Date: Sat, 20 Aug 2011 08:12:41 +0200 Subject: ALSA: sound/aoa/fabrics/layout.c: remove unneeded kfree The label outnodev is only used when kzalloc has not yet taken place or has failed, so there is no need for the call for kfree under this label. A simplified version of the semantic match that finds this problem is as follows: (http://coccinelle.lip6.fr/) // @@ identifier x; expression E1!=0,E2,E3,E4; statement S; iterator I; @@ ( if (...) { ... when != kfree(x) when != x = E3 when != E3 = x * return ...; } ... when != x = E2 when != I(...,x,...) S if (...) { ... when != x = E4 kfree(x); ... return ...; } ) // Signed-off-by: Julia Lawall Signed-off-by: Takashi Iwai --- sound/aoa/fabrics/layout.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/aoa/fabrics/layout.c b/sound/aoa/fabrics/layout.c index 3fd1a7e24928..552b97afbca5 100644 --- a/sound/aoa/fabrics/layout.c +++ b/sound/aoa/fabrics/layout.c @@ -1073,10 +1073,10 @@ static int aoa_fabric_layout_probe(struct soundbus_dev *sdev) sdev->pcmid = -1; list_del(&ldev->list); layouts_list_items--; + kfree(ldev); outnodev: of_node_put(sound); layout_device = NULL; - kfree(ldev); return -ENODEV; } -- cgit v1.2.3 From 4f41adfd8ce9ff84fa9e968e0fe2854e9bc6fcb0 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 20 Aug 2011 10:23:38 +0100 Subject: ASoC: WM8996 record paths need AIFCLK Make AIFCLK supply the record paths otherwise record will not work unless there is a simultaneous playback. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8996.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index ab8e9d1aaff0..85f43b62743b 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -1213,6 +1213,16 @@ static const struct snd_soc_dapm_route wm8996_dapm_routes[] = { { "AIF2RX0", NULL, "AIFCLK" }, { "AIF2RX1", NULL, "AIFCLK" }, + { "AIF1TX0", NULL, "AIFCLK" }, + { "AIF1TX1", NULL, "AIFCLK" }, + { "AIF1TX2", NULL, "AIFCLK" }, + { "AIF1TX3", NULL, "AIFCLK" }, + { "AIF1TX4", NULL, "AIFCLK" }, + { "AIF1TX5", NULL, "AIFCLK" }, + + { "AIF2TX0", NULL, "AIFCLK" }, + { "AIF2TX1", NULL, "AIFCLK" }, + { "DSP1RXL", NULL, "SYSDSPCLK" }, { "DSP1RXR", NULL, "SYSDSPCLK" }, { "DSP2RXL", NULL, "SYSDSPCLK" }, -- cgit v1.2.3 From 7691cd74c5d6b173e1483277fb70d7798e97d2fa Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 20 Aug 2011 16:59:27 +0100 Subject: ASoC: Fix configuration of WM8996 input enables There's no need for separate widgets for the enables (as the map already shows). Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8996.c | 13 ++++--------- 1 file changed, 4 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index 85f43b62743b..e76d4edb67b2 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -988,15 +988,10 @@ SND_SOC_DAPM_MICBIAS("MICB1", WM8996_POWER_MANAGEMENT_1, 8, 0), SND_SOC_DAPM_PGA("IN1L PGA", WM8996_POWER_MANAGEMENT_2, 5, 0, NULL, 0), SND_SOC_DAPM_PGA("IN1R PGA", WM8996_POWER_MANAGEMENT_2, 4, 0, NULL, 0), -SND_SOC_DAPM_MUX("IN1L Mux", SND_SOC_NOPM, 0, 0, &in1_mux), -SND_SOC_DAPM_MUX("IN1R Mux", SND_SOC_NOPM, 0, 0, &in1_mux), -SND_SOC_DAPM_MUX("IN2L Mux", SND_SOC_NOPM, 0, 0, &in2_mux), -SND_SOC_DAPM_MUX("IN2R Mux", SND_SOC_NOPM, 0, 0, &in2_mux), - -SND_SOC_DAPM_PGA("IN1L", WM8996_POWER_MANAGEMENT_7, 2, 0, NULL, 0), -SND_SOC_DAPM_PGA("IN1R", WM8996_POWER_MANAGEMENT_7, 3, 0, NULL, 0), -SND_SOC_DAPM_PGA("IN2L", WM8996_POWER_MANAGEMENT_7, 6, 0, NULL, 0), -SND_SOC_DAPM_PGA("IN2R", WM8996_POWER_MANAGEMENT_7, 7, 0, NULL, 0), +SND_SOC_DAPM_MUX("IN1L Mux", WM8996_POWER_MANAGEMENT_7, 2, 0, &in1_mux), +SND_SOC_DAPM_MUX("IN1R Mux", WM8996_POWER_MANAGEMENT_7, 3, 0, &in1_mux), +SND_SOC_DAPM_MUX("IN2L Mux", WM8996_POWER_MANAGEMENT_7, 6, 0, &in2_mux), +SND_SOC_DAPM_MUX("IN2R Mux", WM8996_POWER_MANAGEMENT_7, 7, 0, &in2_mux), SND_SOC_DAPM_SUPPLY("DMIC2", WM8996_POWER_MANAGEMENT_7, 9, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("DMIC1", WM8996_POWER_MANAGEMENT_7, 8, 0, NULL, 0), -- cgit v1.2.3 From f79e7ff85223e054c2967820d3be1c125a903bd4 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sun, 21 Aug 2011 12:20:00 +0100 Subject: ASoC: Ensure we only run Speyside WM8962 bias level callbacks once We get called once per DAPM context but only need to run once. When DAPM was serialized this was a series of noops but now it can run in parallel we need to take proper care. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/samsung/speyside_wm8962.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/soc/samsung/speyside_wm8962.c b/sound/soc/samsung/speyside_wm8962.c index 0b9eb5f7ec4c..72535f2daaf2 100644 --- a/sound/soc/samsung/speyside_wm8962.c +++ b/sound/soc/samsung/speyside_wm8962.c @@ -23,6 +23,9 @@ static int speyside_wm8962_set_bias_level(struct snd_soc_card *card, struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; int ret; + if (dapm->dev != codec_dai->dev) + return 0; + switch (level) { case SND_SOC_BIAS_PREPARE: if (dapm->bias_level == SND_SOC_BIAS_STANDBY) { @@ -57,6 +60,9 @@ static int speyside_wm8962_set_bias_level_post(struct snd_soc_card *card, struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; int ret; + if (dapm->dev != codec_dai->dev) + return 0; + switch (level) { case SND_SOC_BIAS_STANDBY: ret = snd_soc_dai_set_sysclk(codec_dai, WM8962_SYSCLK_MCLK, -- cgit v1.2.3 From 4df0cb2fa977f99963b616487a22ebd021ea5463 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sun, 21 Aug 2011 17:18:52 +0100 Subject: ASoC: Clear any outstanding WM8962 FLL lock completions before waiting Ensure that we don't spuriously trigger early. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8962.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 28650edfdebb..1725550c293e 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2221,6 +2221,8 @@ static int sysclk_event(struct snd_soc_dapm_widget *w, switch (event) { case SND_SOC_DAPM_PRE_PMU: if (fll) { + try_wait_for_completion(&wm8962->fll_lock); + snd_soc_update_bits(codec, WM8962_FLL_CONTROL_1, WM8962_FLL_ENA, WM8962_FLL_ENA); if (wm8962->irq) { @@ -3284,6 +3286,8 @@ static int wm8962_set_fll(struct snd_soc_codec *codec, int fll_id, int source, snd_soc_write(codec, WM8962_FLL_CONTROL_7, fll_div.lambda); snd_soc_write(codec, WM8962_FLL_CONTROL_8, fll_div.n); + try_wait_for_completion(&wm8962->fll_lock); + snd_soc_update_bits(codec, WM8962_FLL_CONTROL_1, WM8962_FLL_FRAC | WM8962_FLL_REFCLK_SRC_MASK | WM8962_FLL_ENA, fll1); -- cgit v1.2.3 From a41619455c0e28b6973471e87f1702c6129d3439 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 16 Aug 2011 16:57:58 +0900 Subject: ASoC: Clear completions from late WM8996 FLL lock IRQs In case we have a pending completion, for example due to a problem with the input clock which got corrected after we timed out. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8996.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index e76d4edb67b2..0936ae5e3749 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -2111,6 +2111,9 @@ static int wm8996_set_fll(struct snd_soc_codec *codec, int fll_id, int source, snd_soc_write(codec, WM8996_FLL_EFS_1, fll_div.lambda); + /* Clear any pending completions (eg, from failed startups) */ + try_wait_for_completion(&wm8996->fll_lock); + snd_soc_update_bits(codec, WM8996_FLL_CONTROL_1, WM8996_FLL_ENA, WM8996_FLL_ENA); -- cgit v1.2.3 From 81a081fff7f3c144a0da9ee726906e533f66dd89 Mon Sep 17 00:00:00 2001 From: Timur Tabi Date: Mon, 22 Aug 2011 09:22:41 -0500 Subject: sound/soc/fsl/fsl_dma.c: add missing of_node_put of_parse_phandle increments the reference count of np, so this should be decremented before trying the next possibility. Since we don't actually use np, we can decrement the reference count immediately. Reported-by: Julia Lawall Signed-off-by: Timur Tabi Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_dma.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index 732208c8c0b4..cb50598338e9 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -879,10 +879,12 @@ static struct device_node *find_ssi_node(struct device_node *dma_channel_np) * assume that device_node pointers are a valid comparison. */ np = of_parse_phandle(ssi_np, "fsl,playback-dma", 0); + of_node_put(np); if (np == dma_channel_np) return ssi_np; np = of_parse_phandle(ssi_np, "fsl,capture-dma", 0); + of_node_put(np); if (np == dma_channel_np) return ssi_np; } -- cgit v1.2.3 From 57cf9d4512c86a4a1b58857ca22b18bffbfd6812 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Sat, 20 Aug 2011 11:03:44 +0800 Subject: ASoC: soc-core: use GFP_KERNEL flag for kmalloc in snd_soc_cnew GFP_ATOMIC is not needed here, use GFP_KERNEL instead. Signed-off-by: Axel Lin Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 83ad8ca27490..b085d8e87574 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1913,7 +1913,7 @@ struct snd_kcontrol *snd_soc_cnew(const struct snd_kcontrol_new *_template, if (prefix) { name_len = strlen(long_name) + strlen(prefix) + 2; - name = kmalloc(name_len, GFP_ATOMIC); + name = kmalloc(name_len, GFP_KERNEL); if (!name) return NULL; -- cgit v1.2.3 From 96101bd0bf7974686f6875c065a7a9a83cd2107a Mon Sep 17 00:00:00 2001 From: Julia Lawall Date: Sat, 20 Aug 2011 08:12:38 +0200 Subject: sound/soc/kirkwood/kirkwood-i2s.c: add missing kfree Adjust the goto to jump to the error handling code that includes kfree. A simplified version of the semantic match that finds this problem is as follows: (http://coccinelle.lip6.fr/) // @@ identifier x; expression E1!=0,E2,E3,E4; statement S; iterator I; @@ ( if (...) { ... when != kfree(x) when != x = E3 when != E3 = x * return ...; } ... when != x = E2 when != I(...,x,...) S if (...) { ... when != x = E4 kfree(x); ... return ...; } ) // Signed-off-by: Julia Lawall Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/kirkwood/kirkwood-i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index a33fc51f363b..8f16cd37c2af 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -424,7 +424,7 @@ static __devinit int kirkwood_i2s_dev_probe(struct platform_device *pdev) if (!priv->mem) { dev_err(&pdev->dev, "request_mem_region failed\n"); err = -EBUSY; - goto error; + goto error_alloc; } priv->io = ioremap(priv->mem->start, SZ_16K); -- cgit v1.2.3 From 5006b313283c56cfda498513b7091692bcd74433 Mon Sep 17 00:00:00 2001 From: Julia Lawall Date: Sat, 20 Aug 2011 08:12:39 +0200 Subject: sound/soc/ep93xx/ep93xx-i2s.c: add missing kfree Introduce a new label that includes kfree and jump to that one. A simplified version of the semantic match that finds this problem is as follows: (http://coccinelle.lip6.fr/) // @@ identifier x; expression E1!=0,E2,E3,E4; statement S; iterator I; @@ ( if (...) { ... when != kfree(x) when != x = E3 when != E3 = x * return ...; } ... when != x = E2 when != I(...,x,...) S if (...) { ... when != x = E4 kfree(x); ... return ...; } ) // Signed-off-by: Julia Lawall Acked-by: Alexander Sverdlin Reviewed-by: H Hartley Sweeten Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/ep93xx/ep93xx-i2s.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/ep93xx/ep93xx-i2s.c b/sound/soc/ep93xx/ep93xx-i2s.c index 56efa0c1c9a9..099614e16651 100644 --- a/sound/soc/ep93xx/ep93xx-i2s.c +++ b/sound/soc/ep93xx/ep93xx-i2s.c @@ -385,14 +385,14 @@ static int ep93xx_i2s_probe(struct platform_device *pdev) res = platform_get_resource(pdev, IORESOURCE_MEM, 0); if (!res) { err = -ENODEV; - goto fail; + goto fail_free_info; } info->mem = request_mem_region(res->start, resource_size(res), pdev->name); if (!info->mem) { err = -EBUSY; - goto fail; + goto fail_free_info; } info->regs = ioremap(info->mem->start, resource_size(info->mem)); @@ -435,6 +435,7 @@ fail_unmap_mem: iounmap(info->regs); fail_release_mem: release_mem_region(info->mem->start, resource_size(info->mem)); +fail_free_info: kfree(info); fail: return err; -- cgit v1.2.3 From 178b279b645a14ca8ea01e4ea818c88681a31b07 Mon Sep 17 00:00:00 2001 From: Julia Lawall Date: Sat, 20 Aug 2011 09:02:00 +0200 Subject: sound/soc/fsl/p1022_ds.c: add missing of_node_put dma_channel_np has been accessed at this point, so decrease its reference count before leaving the function. A simplified version of the semantic match that finds this problem is as follows: (http://coccinelle.lip6.fr/) // @@ identifier x; expression E1!=0,E2,E3,E4; statement S; iterator I; @@ ( if (...) { ... when != of_node_put(x) when != x = E3 when != E3 = x * return ...; } ... when != x = E2 when != I(...,x,...) S if (...) { ... when != x = E4 of_node_put(x); ... return ...; } ) // Signed-off-by: Julia Lawall Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/fsl/p1022_ds.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/fsl/p1022_ds.c b/sound/soc/fsl/p1022_ds.c index 8fa4d5f8eda1..fcb862eb0c73 100644 --- a/sound/soc/fsl/p1022_ds.c +++ b/sound/soc/fsl/p1022_ds.c @@ -297,8 +297,10 @@ static int get_dma_channel(struct device_node *ssi_np, * dai->platform name should already point to an allocated buffer. */ ret = of_address_to_resource(dma_channel_np, 0, &res); - if (ret) + if (ret) { + of_node_put(dma_channel_np); return ret; + } snprintf((char *)dai->platform_name, DAI_NAME_SIZE, "%llx.%s", (unsigned long long) res.start, dma_channel_np->name); -- cgit v1.2.3 From c09f5ca7bdc9a82c5f721bc28c46d65452240cfa Mon Sep 17 00:00:00 2001 From: Julia Lawall Date: Sat, 20 Aug 2011 09:02:01 +0200 Subject: sound/soc/fsl/mpc8610_hpcd.c: add missing of_node_put The first change is to add an of_node_put, since codec_np has previously been allocated. The rest of the patch reorganizes the error handling code so the only code executed is that which is needed. A simplified version of the semantic match that finds this problem is as follows: (http://coccinelle.lip6.fr/) // @@ identifier x; expression E1!=0,E2,E3,E4; statement S; iterator I; @@ ( if (...) { ... when != of_node_put(x) when != x = E3 when != E3 = x * return ...; } ... when != x = E2 when != I(...,x,...) S if (...) { ... when != x = E4 of_node_put(x); ... return ...; } ) // Signed-off-by: Julia Lawall Acked-by: Timur Tabi Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/fsl/mpc8610_hpcd.c | 18 +++++++++--------- 1 file changed, 9 insertions(+), 9 deletions(-) diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c index a19297959587..358f0baaf71b 100644 --- a/sound/soc/fsl/mpc8610_hpcd.c +++ b/sound/soc/fsl/mpc8610_hpcd.c @@ -345,8 +345,10 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev) } machine_data = kzalloc(sizeof(struct mpc8610_hpcd_data), GFP_KERNEL); - if (!machine_data) - return -ENOMEM; + if (!machine_data) { + ret = -ENOMEM; + goto error_alloc; + } machine_data->dai[0].cpu_dai_name = dev_name(&ssi_pdev->dev); machine_data->dai[0].ops = &mpc8610_hpcd_ops; @@ -494,7 +496,7 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev) ret = platform_device_add(sound_device); if (ret) { dev_err(&pdev->dev, "platform device add failed\n"); - goto error; + goto error_sound; } dev_set_drvdata(&pdev->dev, sound_device); @@ -502,14 +504,12 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev) return 0; +error_sound: + platform_device_unregister(sound_device); error: - of_node_put(codec_np); - - if (sound_device) - platform_device_unregister(sound_device); - kfree(machine_data); - +error_alloc: + of_node_put(codec_np); return ret; } -- cgit v1.2.3 From 675c1aa3c4a7290e537e854d0af7cdf9692bd396 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 23 Aug 2011 12:36:28 +0200 Subject: ALSA: hda - Fix output-path initialization for Realtek auto-parser When the headphone or speaker output has no own DAC, initialize the path using the primary DAC. Otherwise the path won't be set properly and can result in the silence. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 20 +++++++++++++------- 1 file changed, 13 insertions(+), 7 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index fcb11af9ad24..0fefc1088d11 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3083,16 +3083,22 @@ static void alc_auto_init_multi_out(struct hda_codec *codec) static void alc_auto_init_extra_out(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - hda_nid_t pin; + hda_nid_t pin, dac; pin = spec->autocfg.hp_pins[0]; - if (pin) - alc_auto_set_output_and_unmute(codec, pin, PIN_HP, - spec->multiout.hp_nid); + if (pin) { + dac = spec->multiout.hp_nid; + if (!dac) + dac = spec->multiout.dac_nids[0]; + alc_auto_set_output_and_unmute(codec, pin, PIN_HP, dac); + } pin = spec->autocfg.speaker_pins[0]; - if (pin) - alc_auto_set_output_and_unmute(codec, pin, PIN_OUT, - spec->multiout.extra_out_nid[0]); + if (pin) { + dac = spec->multiout.extra_out_nid[0]; + if (!dac) + dac = spec->multiout.dac_nids[0]; + alc_auto_set_output_and_unmute(codec, pin, PIN_OUT, dac); + } } /* -- cgit v1.2.3 From 3c715a98844f72cec0fa3ef2b68232b8f751468b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 23 Aug 2011 12:41:09 +0200 Subject: ALSA: hda - Update jack-sense info even when no automute is set The internal states, jack_present and line_jack_present should be updated upon unsolicited events even if no automute is set. Otherwise the wrong state is referred when the automute behavior is changed by the mixer control. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 0fefc1088d11..7cabd7317163 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -565,11 +565,11 @@ static void alc_hp_automute(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - if (!spec->automute) - return; spec->jack_present = detect_jacks(codec, ARRAY_SIZE(spec->autocfg.hp_pins), spec->autocfg.hp_pins); + if (!spec->automute) + return; update_speakers(codec); } @@ -578,11 +578,11 @@ static void alc_line_automute(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - if (!spec->automute || !spec->detect_line) - return; spec->line_jack_present = detect_jacks(codec, ARRAY_SIZE(spec->autocfg.line_out_pins), spec->autocfg.line_out_pins); + if (!spec->automute || !spec->detect_line) + return; update_speakers(codec); } -- cgit v1.2.3 From 1f015f5fdc4003f3f2a7c66efdb1acf7a2d230bf Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 23 Aug 2011 14:57:08 +0200 Subject: ALSA: hda - Fix double-headphone/speaker paths for Cxt auto-parser When multiple headphones or speakers are assigned but no individual DACs are available, the driver should take the first HP/SPK DAC instead of another primary output. The patch adds a bit-flag to dac field of struct pin_dac_pair indicating that it's a slave DAC. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 13 ++++++++++--- 1 file changed, 10 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 502fc9499453..4c462c3d6462 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3348,6 +3348,8 @@ static hda_nid_t get_unassigned_dac(struct hda_codec *codec, hda_nid_t pin, #define MAX_AUTO_DACS 5 +#define DAC_SLAVE_FLAG 0x8000 /* filled dac is a slave */ + /* fill analog DAC list from the widget tree */ static int fill_cx_auto_dacs(struct hda_codec *codec, hda_nid_t *dacs) { @@ -3379,6 +3381,8 @@ static int fill_dacs_for_pins(struct hda_codec *codec, hda_nid_t *pins, filled[nums].pin = pins[i]; filled[nums].type = type; filled[nums].dac = get_unassigned_dac(codec, pins[i], dacs, rest); + if (!filled[nums].dac && i > 0 && filled[0].dac) + filled[nums].dac = filled[0].dac | DAC_SLAVE_FLAG; nums++; } return nums; @@ -3407,7 +3411,7 @@ static void cx_auto_parse_output(struct hda_codec *codec) /* fill multiout struct */ for (i = 0; i < nums; i++) { hda_nid_t dac = spec->dac_info[i].dac; - if (!dac) + if (!dac || (dac & DAC_SLAVE_FLAG)) continue; switch (spec->dac_info[i].type) { case AUTO_PIN_LINE_OUT: @@ -4035,6 +4039,8 @@ static void cx_auto_init_output(struct hda_codec *codec) nid = spec->dac_info[i].dac; if (!nid) nid = spec->multiout.dac_nids[0]; + else if (nid & DAC_SLAVE_FLAG) + nid &= ~DAC_SLAVE_FLAG; select_connection(codec, spec->dac_info[i].pin, nid); } if (spec->auto_mute) { @@ -4191,7 +4197,8 @@ static int cx_auto_build_output_controls(struct hda_codec *codec) for (i = 0; i < spec->dac_info_filled; i++) { const char *label; int idx, type; - if (!spec->dac_info[i].dac) + hda_nid_t dac = spec->dac_info[i].dac; + if (!dac || (dac & DAC_SLAVE_FLAG)) continue; type = spec->dac_info[i].type; if (type == AUTO_PIN_LINE_OUT) @@ -4211,7 +4218,7 @@ static int cx_auto_build_output_controls(struct hda_codec *codec) idx = num_spk++; break; } - err = try_add_pb_volume(codec, spec->dac_info[i].dac, + err = try_add_pb_volume(codec, dac, spec->dac_info[i].pin, label, idx); if (err < 0) -- cgit v1.2.3 From 7675535958175b85b8117bcee245d9ecbc4d3d74 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 24 Aug 2011 10:53:10 +0200 Subject: ALSA: hda/conexant - Enable ADC-switching for auto-mic mode, too The ADC-switching can work also in the auto-mic mode, too. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 4c462c3d6462..5616444a8ed7 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3866,7 +3866,7 @@ static void cx_auto_parse_input(struct hda_codec *codec) } if (imux->num_items >= 2 && cfg->num_inputs == imux->num_items) cx_auto_check_auto_mic(codec); - if (imux->num_items > 1 && !spec->auto_mic) { + if (imux->num_items > 1) { for (i = 1; i < imux->num_items; i++) { if (spec->imux_info[i].adc != spec->imux_info[0].adc) { spec->adc_switching = 1; -- cgit v1.2.3 From 52c49e0156e167fa65bbc3dd87a3a2f651af03fb Mon Sep 17 00:00:00 2001 From: Joseph Pentland Date: Tue, 23 Aug 2011 10:41:50 +0100 Subject: ASoC: Add Springbank I/O card to Speyside Kconfig Signed-off-by: Joseph Pentland Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/samsung/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index b99091fc34eb..65f980ef2870 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -185,6 +185,7 @@ config SND_SOC_SPEYSIDE select SND_SAMSUNG_I2S select SND_SOC_WM8996 select SND_SOC_WM9081 + select SND_SOC_WM1250_EV1 config SND_SOC_SPEYSIDE_WM8962 tristate "Audio support for Wolfson Speyside with WM8962" -- cgit v1.2.3 From 250b68512dd7e7d31a8c85a740a4b085bade4ba0 Mon Sep 17 00:00:00 2001 From: Sangbeom Kim Date: Tue, 23 Aug 2011 19:36:59 +0900 Subject: ASoC: Add samsung maintainer Signed-off-by: Sangbeom Kim Acked-by: Jassi Brar Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- MAINTAINERS | 1 + 1 file changed, 1 insertion(+) diff --git a/MAINTAINERS b/MAINTAINERS index 46e3e6b99220..4f555d8e5346 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -5532,6 +5532,7 @@ F: include/media/*7146* SAMSUNG AUDIO (ASoC) DRIVERS M: Jassi Brar +M: Sangbeom Kim L: alsa-devel@alsa-project.org (moderated for non-subscribers) S: Supported F: sound/soc/samsung -- cgit v1.2.3 From ee1a4d4b7fcfce31dade9f2ad333b34159cee799 Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Tue, 23 Aug 2011 11:16:28 -0600 Subject: ASoC: Tegra: wm8903 machine driver: Drop Ventana support Board file support for Ventana is not yet mainlined, and probably won't ever be given the move to Device-Tree. Consequently, the Ventana entry is being removed from arch/arm/tools/mach-types in the next merge window, since it was registered over a year ago. This will also remove function machine_is_ventana(), which is used by the ASoC Tegra WM8903 machine driver. This will cause compilation failures. Drop Ventana support to resolve this. Hopefully, in the not-too-distant future, tegra_wm8903.c will be able to configure itself from Device-Tree, and hence we'll be able to re-instate Ventana support just by creating a .dts file for the board. Also note that Aebl support is in a similar boat. However, that board isn't scheduled for deprecation for at least another 5 months, and perhaps we will have completely removed non-Device-Tree support from tegra_wm8903.c by then and/or adjusted mach-types policy. Signed-off-by: Stephen Warren Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/tegra/tegra_wm8903.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c index 661373c2352a..be27f1d229af 100644 --- a/sound/soc/tegra/tegra_wm8903.c +++ b/sound/soc/tegra/tegra_wm8903.c @@ -319,7 +319,7 @@ static int tegra_wm8903_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_force_enable_pin(dapm, "Mic Bias"); /* FIXME: Calculate automatically based on DAPM routes? */ - if (!machine_is_harmony() && !machine_is_ventana()) + if (!machine_is_harmony()) snd_soc_dapm_nc_pin(dapm, "IN1L"); if (!machine_is_seaboard() && !machine_is_aebl()) snd_soc_dapm_nc_pin(dapm, "IN1R"); @@ -395,7 +395,7 @@ static __devinit int tegra_wm8903_driver_probe(struct platform_device *pdev) platform_set_drvdata(pdev, card); snd_soc_card_set_drvdata(card, machine); - if (machine_is_harmony() || machine_is_ventana()) { + if (machine_is_harmony()) { card->dapm_routes = harmony_audio_map; card->num_dapm_routes = ARRAY_SIZE(harmony_audio_map); } else if (machine_is_seaboard()) { -- cgit v1.2.3 From 18036b5866b5e407a28f444a80de186a5d7df767 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 24 Aug 2011 16:35:32 +0100 Subject: ASoC: Correct element count for WM8996 sidetone HPF I can count. Honest. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8996.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index 0936ae5e3749..0cdb9d105671 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -420,7 +420,7 @@ static const char *sidetone_hpf_text[] = { }; static const struct soc_enum sidetone_hpf = - SOC_ENUM_SINGLE(WM8996_SIDETONE, 7, 6, sidetone_hpf_text); + SOC_ENUM_SINGLE(WM8996_SIDETONE, 7, 7, sidetone_hpf_text); static const char *hpf_mode_text[] = { "HiFi", "Custom", "Voice" -- cgit v1.2.3 From 3bdf28feafc52864bd7f17b39deec64833a89d19 Mon Sep 17 00:00:00 2001 From: Timur Tabi Date: Tue, 23 Aug 2011 16:48:26 -0500 Subject: ASoC: MPC5200: replace of_device with platform_device 'struct of_device' no longer exists, and its functionality has been merged into platform_device. Update the MPC5200 audio DMA driver (mpc5200_dma) accordingly. This fixes a build break. Signed-off-by: Timur Tabi Acked-by: Liam Girdwood Signed-off-by: Mark Brown Cc: stable@kernel.org --- sound/soc/fsl/mpc5200_dma.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/fsl/mpc5200_dma.c b/sound/soc/fsl/mpc5200_dma.c index fd0dc46afc34..5c6c2457386e 100644 --- a/sound/soc/fsl/mpc5200_dma.c +++ b/sound/soc/fsl/mpc5200_dma.c @@ -369,7 +369,7 @@ static struct snd_soc_platform_driver mpc5200_audio_dma_platform = { .pcm_free = &psc_dma_free, }; -static int mpc5200_hpcd_probe(struct of_device *op) +static int mpc5200_hpcd_probe(struct platform_device *op) { phys_addr_t fifo; struct psc_dma *psc_dma; @@ -487,7 +487,7 @@ out_unmap: return ret; } -static int mpc5200_hpcd_remove(struct of_device *op) +static int mpc5200_hpcd_remove(struct platform_device *op) { struct psc_dma *psc_dma = dev_get_drvdata(&op->dev); @@ -519,7 +519,7 @@ MODULE_DEVICE_TABLE(of, mpc5200_hpcd_match); static struct platform_driver mpc5200_hpcd_of_driver = { .probe = mpc5200_hpcd_probe, .remove = mpc5200_hpcd_remove, - .dev = { + .driver = { .owner = THIS_MODULE, .name = "mpc5200-pcm-audio", .of_match_table = mpc5200_hpcd_match, -- cgit v1.2.3 From 468c5458856236cde6df1b0654d32bf6625349a5 Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Thu, 25 Aug 2011 13:16:02 +0200 Subject: ALSA: hda: Conexant: Allow different output types to share DAC Headphones has stopped working for the original reported (a regression compared to 2.6.38). This is because Speaker and Headphones share the same DAC, in which case no Headphones volume control was created. This patch fixes so that both Speaker and Headphones volume controls are created in such scenario. BugLink: http://bugs.launchpad.net/bugs/817943 Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 46 +++++++++++++++++++++++++----------------- 1 file changed, 27 insertions(+), 19 deletions(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 5616444a8ed7..7696d05b9356 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3372,18 +3372,26 @@ static int fill_cx_auto_dacs(struct hda_codec *codec, hda_nid_t *dacs) /* fill pin_dac_pair list from the pin and dac list */ static int fill_dacs_for_pins(struct hda_codec *codec, hda_nid_t *pins, int num_pins, hda_nid_t *dacs, int *rest, - struct pin_dac_pair *filled, int type) + struct pin_dac_pair *filled, int nums, + int type) { - int i, nums; + int i, start = nums; - nums = 0; - for (i = 0; i < num_pins; i++) { + for (i = 0; i < num_pins; i++, nums++) { filled[nums].pin = pins[i]; filled[nums].type = type; filled[nums].dac = get_unassigned_dac(codec, pins[i], dacs, rest); - if (!filled[nums].dac && i > 0 && filled[0].dac) + if (filled[nums].dac) + continue; + if (filled[start].dac && get_connection_index(codec, pins[i], filled[start].dac) >= 0) { + filled[nums].dac = filled[start].dac | DAC_SLAVE_FLAG; + continue; + } + if (filled[0].dac && get_connection_index(codec, pins[i], filled[0].dac) >= 0) { filled[nums].dac = filled[0].dac | DAC_SLAVE_FLAG; - nums++; + continue; + } + snd_printdd("Failed to find a DAC for pin 0x%x", pins[i]); } return nums; } @@ -3399,14 +3407,14 @@ static void cx_auto_parse_output(struct hda_codec *codec) rest = fill_cx_auto_dacs(codec, dacs); /* parse all analog output pins */ nums = fill_dacs_for_pins(codec, cfg->line_out_pins, cfg->line_outs, - dacs, &rest, spec->dac_info, - AUTO_PIN_LINE_OUT); - nums += fill_dacs_for_pins(codec, cfg->hp_pins, cfg->hp_outs, - dacs, &rest, spec->dac_info + nums, - AUTO_PIN_HP_OUT); - nums += fill_dacs_for_pins(codec, cfg->speaker_pins, cfg->speaker_outs, - dacs, &rest, spec->dac_info + nums, - AUTO_PIN_SPEAKER_OUT); + dacs, &rest, spec->dac_info, 0, + AUTO_PIN_LINE_OUT); + nums = fill_dacs_for_pins(codec, cfg->hp_pins, cfg->hp_outs, + dacs, &rest, spec->dac_info, nums, + AUTO_PIN_HP_OUT); + nums = fill_dacs_for_pins(codec, cfg->speaker_pins, cfg->speaker_outs, + dacs, &rest, spec->dac_info, nums, + AUTO_PIN_SPEAKER_OUT); spec->dac_info_filled = nums; /* fill multiout struct */ for (i = 0; i < nums; i++) { @@ -4173,9 +4181,11 @@ static int try_add_pb_volume(struct hda_codec *codec, hda_nid_t dac, hda_nid_t pin, const char *name, int idx) { unsigned int caps; - caps = query_amp_caps(codec, dac, HDA_OUTPUT); - if (caps & AC_AMPCAP_NUM_STEPS) - return cx_auto_add_pb_volume(codec, dac, name, idx); + if (dac && !(dac & DAC_SLAVE_FLAG)) { + caps = query_amp_caps(codec, dac, HDA_OUTPUT); + if (caps & AC_AMPCAP_NUM_STEPS) + return cx_auto_add_pb_volume(codec, dac, name, idx); + } caps = query_amp_caps(codec, pin, HDA_OUTPUT); if (caps & AC_AMPCAP_NUM_STEPS) return cx_auto_add_pb_volume(codec, pin, name, idx); @@ -4198,8 +4208,6 @@ static int cx_auto_build_output_controls(struct hda_codec *codec) const char *label; int idx, type; hda_nid_t dac = spec->dac_info[i].dac; - if (!dac || (dac & DAC_SLAVE_FLAG)) - continue; type = spec->dac_info[i].type; if (type == AUTO_PIN_LINE_OUT) type = spec->autocfg.line_out_type; -- cgit v1.2.3 From 59ec6da2e3e53a3e4f3cb4d35b6449e05f1bcc18 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 19 Aug 2011 17:53:12 +0900 Subject: MAINTAINERS: Add some missed Wolfson files Mostly input related. Signed-off-by: Mark Brown --- MAINTAINERS | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/MAINTAINERS b/MAINTAINERS index 4f555d8e5346..0b4ccdd35bbb 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -7200,6 +7200,9 @@ W: http://opensource.wolfsonmicro.com/content/linux-drivers-wolfson-devices S: Supported F: Documentation/hwmon/wm83?? F: drivers/leds/leds-wm83*.c +F: drivers/input/misc/wm831x-on.c +F: drivers/input/touchscreen/wm831x-ts.c +F: drivers/input/touchscreen/wm97*.c F: drivers/mfd/wm8*.c F: drivers/power/wm83*.c F: drivers/rtc/rtc-wm83*.c @@ -7209,6 +7212,7 @@ F: drivers/watchdog/wm83*_wdt.c F: include/linux/mfd/wm831x/ F: include/linux/mfd/wm8350/ F: include/linux/mfd/wm8400* +F: include/linux/wm97xx.h F: include/sound/wm????.h F: sound/soc/codecs/wm* -- cgit v1.2.3 From 728a52222461a8cf0d5c375da1ef514a72d5194b Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 26 Aug 2011 16:33:52 +0300 Subject: ASoC: soc-dapm: Fix parameter comment for snd_soc_dapm_free We have dapm_context instead of codec parameter. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 7e15914b3633..d67c637557a7 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2763,7 +2763,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_ignore_suspend); /** * snd_soc_dapm_free - free dapm resources - * @card: SoC device + * @dapm: DAPM context * * Free all dapm widgets and resources. */ -- cgit v1.2.3 From 63fa0a288cfedca681175fe13cf15677e944cdb2 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sat, 27 Aug 2011 18:24:12 +0200 Subject: ASoC: snd_soc_codec_{readable,writable}_register change default to true Change the default return value of snd_soc_codec_{readable,writable}_register to true when no codec specific callback for this function is given. Otherwise all registers of that codec will neither be readable nor writable, which is most certainly not what we want. Signed-off-by: Lars-Peter Clausen Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index b085d8e87574..d2ef014af215 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1633,7 +1633,7 @@ int snd_soc_codec_readable_register(struct snd_soc_codec *codec, if (codec->readable_register) return codec->readable_register(codec, reg); else - return 0; + return 1; } EXPORT_SYMBOL_GPL(snd_soc_codec_readable_register); @@ -1651,7 +1651,7 @@ int snd_soc_codec_writable_register(struct snd_soc_codec *codec, if (codec->writable_register) return codec->writable_register(codec, reg); else - return 0; + return 1; } EXPORT_SYMBOL_GPL(snd_soc_codec_writable_register); -- cgit v1.2.3 From 6c5b756aaa3e4a20c8d4b47b9dc4799b9cfdfcb8 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sat, 27 Aug 2011 18:24:13 +0200 Subject: ASoC: Fix register cache sync register_writable WARN_ONs Currently the condition for these WARN_ONs is reversed and they are placed before the actual check whether we are going to write to that register. So if the codec implements the register_writable callback we'll get a warning for each writable register when syncing the register cache. While we are at it change the check to use snd_soc_codec_writable_register instead of open-coding it. Signed-off-by: Lars-Peter Clausen Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/soc-cache.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index d9f8aded51f3..20b7f3b003a3 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -203,14 +203,14 @@ static int snd_soc_rbtree_cache_sync(struct snd_soc_codec *codec) rbnode = rb_entry(node, struct snd_soc_rbtree_node, node); for (i = 0; i < rbnode->blklen; ++i) { regtmp = rbnode->base_reg + i; - WARN_ON(codec->writable_register && - codec->writable_register(codec, regtmp)); val = snd_soc_rbtree_get_register(rbnode, i); def = snd_soc_get_cache_val(codec->reg_def_copy, i, rbnode->word_size); if (val == def) continue; + WARN_ON(!snd_soc_codec_writable_register(codec, regtmp)); + codec->cache_bypass = 1; ret = snd_soc_write(codec, regtmp, val); codec->cache_bypass = 0; @@ -563,8 +563,7 @@ static int snd_soc_lzo_cache_sync(struct snd_soc_codec *codec) lzo_blocks = codec->reg_cache; for_each_set_bit(i, lzo_blocks[0]->sync_bmp, lzo_blocks[0]->sync_bmp_nbits) { - WARN_ON(codec->writable_register && - codec->writable_register(codec, i)); + WARN_ON(!snd_soc_codec_writable_register(codec, i)); ret = snd_soc_cache_read(codec, i, &val); if (ret) return ret; @@ -823,8 +822,6 @@ static int snd_soc_flat_cache_sync(struct snd_soc_codec *codec) codec_drv = codec->driver; for (i = 0; i < codec_drv->reg_cache_size; ++i) { - WARN_ON(codec->writable_register && - codec->writable_register(codec, i)); ret = snd_soc_cache_read(codec, i, &val); if (ret) return ret; @@ -832,6 +829,9 @@ static int snd_soc_flat_cache_sync(struct snd_soc_codec *codec) if (snd_soc_get_cache_val(codec->reg_def_copy, i, codec_drv->reg_word_size) == val) continue; + + WARN_ON(!snd_soc_codec_writable_register(codec, i)); + ret = snd_soc_write(codec, i, val); if (ret) return ret; -- cgit v1.2.3 From 117ef9570bf3a332eca443231f41dd3a52fcfb2c Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Tue, 30 Aug 2011 00:28:42 -0300 Subject: ASoC: imx: Fix build warning of unused 'card' variable Fixes the following warning: CC sound/soc/imx/imx-pcm-fiq.o sound/soc/imx/imx-pcm-fiq.c: In function 'imx_pcm_fiq_new': sound/soc/imx/imx-pcm-fiq.c:243: warning: unused variable 'card' CC sound/soc/imx/imx-pcm-dma-mx2.o Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/imx/imx-pcm-fiq.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/imx/imx-pcm-fiq.c b/sound/soc/imx/imx-pcm-fiq.c index 309c59e6fb6c..7945625e0e08 100644 --- a/sound/soc/imx/imx-pcm-fiq.c +++ b/sound/soc/imx/imx-pcm-fiq.c @@ -240,7 +240,6 @@ static int ssi_irq = 0; static int imx_pcm_fiq_new(struct snd_soc_pcm_runtime *rtd) { - struct snd_card *card = rtd->card->snd_card; struct snd_soc_dai *dai = rtd->cpu_dai; struct snd_pcm *pcm = rtd->pcm; int ret; -- cgit v1.2.3 From 747da0f80e566500421bd7760b2e050fea3fde5e Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sun, 4 Sep 2011 08:18:18 -0700 Subject: ASoC: Fix reporting of partial jack updates We need to report the entire jack state to the core jack code, not just the bits that were being updated by the caller, otherwise the status reported by other detection methods will be omitted from the state seen by userspace. Signed-off-by: Mark Brown Acked-by: Liam Girdwood Cc: stable@kernel.org --- sound/soc/soc-jack.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index 38b00131b2fe..fa31d9c2abd8 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -105,7 +105,7 @@ void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask) snd_soc_dapm_sync(dapm); - snd_jack_report(jack->jack, status); + snd_jack_report(jack->jack, jack->status); out: mutex_unlock(&codec->mutex); -- cgit v1.2.3 From c5d2e650bd805a00ff9af537d5b5dede598a198c Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 5 Sep 2011 13:49:57 +0200 Subject: ASoC: Blackfin: bf5xx-ad193x: Fix codec device name Fix the codec_name field of the dai_link to match the actual device name of the codec. Otherwise the card won't be instantiated. Signed-off-by: Lars-Peter Clausen Acked-by: Liam Girdwood Signed-off-by: Mark Brown Cc: stable@kernel.org --- sound/soc/blackfin/bf5xx-ad193x.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/blackfin/bf5xx-ad193x.c b/sound/soc/blackfin/bf5xx-ad193x.c index a118a0fb9d81..5956584ea3a4 100644 --- a/sound/soc/blackfin/bf5xx-ad193x.c +++ b/sound/soc/blackfin/bf5xx-ad193x.c @@ -103,7 +103,7 @@ static struct snd_soc_dai_link bf5xx_ad193x_dai[] = { .cpu_dai_name = "bfin-tdm.0", .codec_dai_name ="ad193x-hifi", .platform_name = "bfin-tdm-pcm-audio", - .codec_name = "ad193x.5", + .codec_name = "spi0.5", .ops = &bf5xx_ad193x_ops, }, { @@ -112,7 +112,7 @@ static struct snd_soc_dai_link bf5xx_ad193x_dai[] = { .cpu_dai_name = "bfin-tdm.1", .codec_dai_name ="ad193x-hifi", .platform_name = "bfin-tdm-pcm-audio", - .codec_name = "ad193x.5", + .codec_name = "spi0.5", .ops = &bf5xx_ad193x_ops, }, }; -- cgit v1.2.3 From 5013951be88e136d9990ef55303276e2779ce8d8 Mon Sep 17 00:00:00 2001 From: Arnd Bergmann Date: Sun, 11 Sep 2011 20:07:30 +0200 Subject: ASoC: Fix trivial build regression in Kirkwood I2S A fix merged in 3.1-rc2 introduced a small regression, this should get it to build again. Signed-off-by: Arnd Bergmann Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/kirkwood/kirkwood-i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index 8f16cd37c2af..d0bcf3fcea01 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -424,7 +424,7 @@ static __devinit int kirkwood_i2s_dev_probe(struct platform_device *pdev) if (!priv->mem) { dev_err(&pdev->dev, "request_mem_region failed\n"); err = -EBUSY; - goto error_alloc; + goto err_alloc; } priv->io = ioremap(priv->mem->start, SZ_16K); -- cgit v1.2.3 From 99e14c9d4140e9ed2c8810322a377e2a51b356eb Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 13 Sep 2011 10:33:16 +0200 Subject: ALSA: hda - Terminate the recursive connection search properly The recursive search of widget connections in snd_hda_get_conn_index() must be terminated at the pin and the audio-out widgets. Otherwise you'll get "too deep connection" warnings unnecessarily. Reported-by: Francis Moreau Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 3e7850c238c3..f3aefef37216 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -579,9 +579,13 @@ int snd_hda_get_conn_index(struct hda_codec *codec, hda_nid_t mux, return -1; } recursive++; - for (i = 0; i < nums; i++) + for (i = 0; i < nums; i++) { + unsigned int type = get_wcaps_type(get_wcaps(codec, conn[i])); + if (type == AC_WID_PIN || type == AC_WID_AUD_OUT) + continue; if (snd_hda_get_conn_index(codec, conn[i], nid, recursive) >= 0) return i; + } return -1; } EXPORT_SYMBOL_HDA(snd_hda_get_conn_index); -- cgit v1.2.3 From 2e1210bc3d065a6e26ff5fef228a9a7e08921d2c Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Wed, 14 Sep 2011 13:22:54 +0200 Subject: ALSA: HDA: Cirrus - fix "Surround Speaker" volume control name This patch fixes "Surround Speaker Playback Volume" being cut off. (Commit b4dabfc452a10 was probably meant to fix this, but it fixed only the "Switch" name, not the "Volume" name.) Signed-off-by: David Henningsson Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_cirrus.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index d6c93d92b550..c45f3e69bcf0 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -535,7 +535,7 @@ static int add_volume(struct hda_codec *codec, const char *name, int index, unsigned int pval, int dir, struct snd_kcontrol **kctlp) { - char tmp[32]; + char tmp[44]; struct snd_kcontrol_new knew = HDA_CODEC_VOLUME_IDX(tmp, index, 0, 0, HDA_OUTPUT); knew.private_value = pval; -- cgit v1.2.3 From 763437a9e7737535b2fc72175ad4974048769be6 Mon Sep 17 00:00:00 2001 From: Arjan van de Ven Date: Thu, 15 Sep 2011 08:49:25 +0200 Subject: ALSA: pcm - fix race condition in wait_for_avail() wait_for_avail() in pcm_lib.c has a race in it (observed in practice by an Intel validation group). The function is supposed to return once space in the buffer has become available, or if some timeout happens. The entity that creates space (irq handler of sound driver and some such) will do a wake up on a waitqueue that this function registers for. However there are two races in the existing code 1) If space became available between the caller noticing there was no space and this function actually sleeping, the wakeup is missed and the timeout condition will happen instead 2) If a wakeup happened but not sufficient space became available, the code will loop again and wait for more space. However, if the second wake comes in prior to hitting the schedule_timeout_interruptible(), it will be missed, and potentially you'll wait out until the timeout happens. The fix consists of using more careful setting of the current state (so that if a wakeup happens in the main loop window, the schedule_timeout() falls through) and by checking for available space prior to going into the schedule_timeout() loop, but after being on the waitqueue and having the state set to interruptible. [tiwai: the following changes have been added to Arjan's original patch: - merged akpm's fix for waitqueue adding order into a single patch - reduction of duplicated code of avail check ] Signed-off-by: Arjan van de Ven Signed-off-by: Andrew Morton Cc: Signed-off-by: Takashi Iwai --- sound/core/pcm_lib.c | 33 ++++++++++++++++++++++++--------- 1 file changed, 24 insertions(+), 9 deletions(-) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 86d0caf91b35..62e90b862a0d 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -1761,6 +1761,10 @@ static int wait_for_avail(struct snd_pcm_substream *substream, snd_pcm_uframes_t avail = 0; long wait_time, tout; + init_waitqueue_entry(&wait, current); + set_current_state(TASK_INTERRUPTIBLE); + add_wait_queue(&runtime->tsleep, &wait); + if (runtime->no_period_wakeup) wait_time = MAX_SCHEDULE_TIMEOUT; else { @@ -1771,16 +1775,32 @@ static int wait_for_avail(struct snd_pcm_substream *substream, } wait_time = msecs_to_jiffies(wait_time * 1000); } - init_waitqueue_entry(&wait, current); - add_wait_queue(&runtime->tsleep, &wait); + for (;;) { if (signal_pending(current)) { err = -ERESTARTSYS; break; } + + /* + * We need to check if space became available already + * (and thus the wakeup happened already) first to close + * the race of space already having become available. + * This check must happen after been added to the waitqueue + * and having current state be INTERRUPTIBLE. + */ + if (is_playback) + avail = snd_pcm_playback_avail(runtime); + else + avail = snd_pcm_capture_avail(runtime); + if (avail >= runtime->twake) + break; snd_pcm_stream_unlock_irq(substream); - tout = schedule_timeout_interruptible(wait_time); + + tout = schedule_timeout(wait_time); + snd_pcm_stream_lock_irq(substream); + set_current_state(TASK_INTERRUPTIBLE); switch (runtime->status->state) { case SNDRV_PCM_STATE_SUSPENDED: err = -ESTRPIPE; @@ -1806,14 +1826,9 @@ static int wait_for_avail(struct snd_pcm_substream *substream, err = -EIO; break; } - if (is_playback) - avail = snd_pcm_playback_avail(runtime); - else - avail = snd_pcm_capture_avail(runtime); - if (avail >= runtime->twake) - break; } _endloop: + set_current_state(TASK_RUNNING); remove_wait_queue(&runtime->tsleep, &wait); *availp = avail; return err; -- cgit v1.2.3