From 0b1f6ec7a5fb3faff1a62afee132dac316eec63d Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 9 Feb 2015 08:05:22 +0000 Subject: ASoC: rsnd: set device data before snd_soc_register_platform/component Set device data before snd_soc_register_platform/component. Otherwise, it will use NULL pointer if user calls unbind -> bind or rmmod -> insmod Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/core.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 75308bbc2ce8..fc227d3bc021 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -1268,6 +1268,8 @@ static int rsnd_probe(struct platform_device *pdev) goto exit_snd_probe; } + dev_set_drvdata(dev, priv); + /* * asoc register */ @@ -1284,8 +1286,6 @@ static int rsnd_probe(struct platform_device *pdev) goto exit_snd_soc; } - dev_set_drvdata(dev, priv); - pm_runtime_enable(dev); dev_info(dev, "probed\n"); -- cgit v1.2.3 From 541b03ad6cfe0e415273f096fd8c47d2879c6c15 Mon Sep 17 00:00:00 2001 From: Nicolin Chen Date: Tue, 10 Feb 2015 21:31:43 -0800 Subject: ASoC: fsl_ssi: Fix the incorrect limitation of the bit clock rate According to i.MX Reference Manual, the bit-clock frequency generated by SSI must be never greater than 1/5 of the peripheral clock frequency. This peripheral clock, however, is not baudclk but the IPG clock (i.e. ssi_private->clk in the fsl_ssi driver). So this patch just simply fixes the incorrect limitation applied to the bit clock (baudclk) rate. Signed-off-by: Nicolin Chen Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_ssi.c | 11 +++++++---- 1 file changed, 7 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 059496ed9ad7..d7365c5d7ec0 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -603,10 +603,6 @@ static int fsl_ssi_set_bclk(struct snd_pcm_substream *substream, factor = (div2 + 1) * (7 * psr + 1) * 2; for (i = 0; i < 255; i++) { - /* The bclk rate must be smaller than 1/5 sysclk rate */ - if (factor * (i + 1) < 5) - continue; - tmprate = freq * factor * (i + 2); if (baudclk_is_used) @@ -614,6 +610,13 @@ static int fsl_ssi_set_bclk(struct snd_pcm_substream *substream, else clkrate = clk_round_rate(ssi_private->baudclk, tmprate); + /* + * Hardware limitation: The bclk rate must be + * never greater than 1/5 IPG clock rate + */ + if (clkrate * 5 > clk_get_rate(ssi_private->clk)) + continue; + clkrate /= factor; afreq = clkrate / (i + 1); -- cgit v1.2.3 From ffa047577127336861d91f3934133f8e8906d1b4 Mon Sep 17 00:00:00 2001 From: Guenter Roeck Date: Wed, 11 Feb 2015 13:13:18 -0800 Subject: ASoC: Fix MAX98357A codec driver dependencies The max98357a driver depends on GPIOLIB. This may cause the following build failure. sound/soc/codecs/max98357a.c: In function 'max98357a_daiops_trigger': sound/soc/codecs/max98357a.c:30:3: error: implicit declaration of function 'gpiod_set_value' sound/soc/codecs/max98357a.c: In function 'max98357a_codec_probe': sound/soc/codecs/max98357a.c:55:2: error: implicit declaration of function 'devm_gpiod_get' sound/soc/codecs/max98357a.c:61:2: error: implicit declaration of function 'gpiod_direction_output' Seen with mips:allmodconfig as well as various randconfig builds. Fixes: af5adf129369 ("ASoC: max98357a: Add MAX98357A codec driver") Cc: Kenneth Westfield Signed-off-by: Guenter Roeck Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 064e6c18e109..ea9f0e31f9d4 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -69,7 +69,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_MAX98088 if I2C select SND_SOC_MAX98090 if I2C select SND_SOC_MAX98095 if I2C - select SND_SOC_MAX98357A + select SND_SOC_MAX98357A if GPIOLIB select SND_SOC_MAX9850 if I2C select SND_SOC_MAX9768 if I2C select SND_SOC_MAX9877 if I2C -- cgit v1.2.3 From 5c8be987d4d9c0262e6229e342fa0da8a5aeee47 Mon Sep 17 00:00:00 2001 From: Vincent Stehlé Date: Wed, 11 Feb 2015 23:08:59 +0100 Subject: ASoC: max98357a: Fix missing include MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit This fixes the following compilation errors: sound/soc/codecs/max98357a.c: In function ‘max98357a_daiops_trigger’: sound/soc/codecs/max98357a.c:30:3: error: implicit declaration of function ‘gpiod_set_value’ [-Werror=implicit-function-declaration] sound/soc/codecs/max98357a.c: In function ‘max98357a_codec_probe’: sound/soc/codecs/max98357a.c:55:2: error: implicit declaration of function ‘devm_gpiod_get’ [-Werror=implicit-function-declaration] sound/soc/codecs/max98357a.c:61:2: error: implicit declaration of function ‘gpiod_direction_output’ [-Werror=implicit-function-declaration] cc1: some warnings being treated as errors Signed-off-by: Vincent Stehlé Cc: Kenneth Westfield Cc: Mark Brown Signed-off-by: Mark Brown --- sound/soc/codecs/max98357a.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/codecs/max98357a.c b/sound/soc/codecs/max98357a.c index 1806333ea29e..f493fb6fd4ea 100644 --- a/sound/soc/codecs/max98357a.c +++ b/sound/soc/codecs/max98357a.c @@ -14,6 +14,7 @@ #include #include +#include #include #define DRV_NAME "max98357a" -- cgit v1.2.3 From fb5ab7296a2bea17c38fae48af2808a07049ac90 Mon Sep 17 00:00:00 2001 From: Kiran Padwal Date: Thu, 12 Feb 2015 14:38:02 +0530 Subject: ASoC: omap-hdmi-audio: Add missing error check for devm_kzalloc This patch add a missing check on the return value of devm_kzalloc, which would cause a NULL pointer dereference in a OOM situation. Signed-off-by: Kiran Padwal Acked-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/omap/omap-hdmi-audio.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/soc/omap/omap-hdmi-audio.c b/sound/soc/omap/omap-hdmi-audio.c index 3f9ac7dbdc80..069ad451d05d 100644 --- a/sound/soc/omap/omap-hdmi-audio.c +++ b/sound/soc/omap/omap-hdmi-audio.c @@ -352,6 +352,9 @@ static int omap_hdmi_audio_probe(struct platform_device *pdev) return ret; card = devm_kzalloc(dev, sizeof(*card), GFP_KERNEL); + if (!card) + return -ENOMEM; + card->name = devm_kasprintf(dev, GFP_KERNEL, "HDMI %s", dev_name(ad->dssdev)); card->owner = THIS_MODULE; -- cgit v1.2.3 From 7bd345c9e87d879d696c6843fe200b60c2051c84 Mon Sep 17 00:00:00 2001 From: Mengdong Lin Date: Fri, 13 Feb 2015 19:21:25 +0800 Subject: ASoC: Intel: set initial runtime PM status to active for ACPI-enumerated ADSP The ADSP on Braswell/Baytrail is an ACPI device. This patch sets its initial runtime PM status to active. Otherwise, its initial status is suspended and runtime_suspend ops will not be called after probe and thus cannot further trigger ACPI _PS3 (D3) method to put the device into low power D3cold state. Signed-off-by: Mengdong Lin Signed-off-by: Mark Brown --- sound/soc/intel/sst/sst.c | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'sound') diff --git a/sound/soc/intel/sst/sst.c b/sound/soc/intel/sst/sst.c index 8a8d56a146e7..d6ea80076ea2 100644 --- a/sound/soc/intel/sst/sst.c +++ b/sound/soc/intel/sst/sst.c @@ -379,6 +379,10 @@ void sst_configure_runtime_pm(struct intel_sst_drv *ctx) * initially active. So change the state to active before * enabling the pm */ + + if (!acpi_disabled) + pm_runtime_set_active(ctx->dev); + pm_runtime_enable(ctx->dev); if (acpi_disabled) -- cgit v1.2.3 From e7a961c9578ce227d3c62c4cce9463b763a1e0c0 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Tue, 17 Feb 2015 13:59:27 +0800 Subject: ASoC: rt5670: Fix the speaker mono output issue We need to set left/right control for the speaker amp to get stereo output on speaker. Signed-off-by: Bard Liao Signed-off-by: Mark Brown --- sound/soc/codecs/rt5670.c | 6 ++++++ 1 file changed, 6 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index 8a0833de1665..d33f33ce865a 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -2591,6 +2591,12 @@ static int rt5670_i2c_probe(struct i2c_client *i2c, regmap_write(rt5670->regmap, RT5670_RESET, 0); + regmap_read(rt5670->regmap, RT5670_VENDOR_ID, &val); + if (val >= 4) + regmap_write(rt5670->regmap, RT5670_GPIO_CTRL3, 0x0980); + else + regmap_write(rt5670->regmap, RT5670_GPIO_CTRL3, 0x0d00); + ret = regmap_register_patch(rt5670->regmap, init_list, ARRAY_SIZE(init_list)); if (ret != 0) -- cgit v1.2.3 From 014c4d637604c9af2f7f2ff4fd91b725a0c58a5c Mon Sep 17 00:00:00 2001 From: Arnd Bergmann Date: Wed, 18 Feb 2015 21:35:08 +0100 Subject: ASoC: Samsung: add missing I2C/SPI dependencies A few sound drivers for the samsung platforms are missing dependencies on I2C or SPI, which can lead to build errors like codecs/rt5631.c:1737:1: warning: data definition has no type or storage class 31_i2c_driver); codecs/rt5631.c:1737:1: error: type defaults to 'int' in declaration of 'module_i2c_driver' [-Werror=implicit-int] codecs/rt5631.c:1737:1: warning: parameter names (without types) in function declaration codecs/rt5631.c:1726:26: warning: 'rt5631_i2c_driver' defined but not used [-Wunused-variable] I have gone through all the ones that did not already have an I2C dependency and added the ones that I found missing, namely arndale, odroid-x2, littlemill, bells and speyside and this patch adds all the dependencies. Signed-off-by: Arnd Bergmann Signed-off-by: Mark Brown --- sound/soc/samsung/Kconfig | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index fc67f97f19f6..e0c4a4ec4280 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -185,7 +185,7 @@ config SND_SOC_SMDK_WM8994_PCM config SND_SOC_SPEYSIDE tristate "Audio support for Wolfson Speyside" - depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 + depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 && I2C && SPI_MASTER select SND_SAMSUNG_I2S select SND_SOC_WM8996 select SND_SOC_WM9081 @@ -200,7 +200,7 @@ config SND_SOC_TOBERMORY config SND_SOC_BELLS tristate "Audio support for Wolfson Bells" - depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 && MFD_ARIZONA + depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 && MFD_ARIZONA && I2C && SPI_MASTER select SND_SAMSUNG_I2S select SND_SOC_WM5102 select SND_SOC_WM5110 @@ -217,7 +217,7 @@ config SND_SOC_LOWLAND config SND_SOC_LITTLEMILL tristate "Audio support for Wolfson Littlemill" - depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 + depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 && I2C select SND_SAMSUNG_I2S select MFD_WM8994 select SND_SOC_WM8994 @@ -234,7 +234,7 @@ config SND_SOC_SNOW config SND_SOC_ODROIDX2 tristate "Audio support for Odroid-X2 and Odroid-U3" - depends on SND_SOC_SAMSUNG + depends on SND_SOC_SAMSUNG && I2C select SND_SOC_MAX98090 select SND_SAMSUNG_I2S help @@ -242,6 +242,6 @@ config SND_SOC_ODROIDX2 config SND_SOC_ARNDALE_RT5631_ALC5631 tristate "Audio support for RT5631(ALC5631) on Arndale Board" - depends on SND_SOC_SAMSUNG + depends on SND_SOC_SAMSUNG && I2C select SND_SAMSUNG_I2S select SND_SOC_RT5631 -- cgit v1.2.3 From 52554fbd2f88a432a16e9e88e14c4b02ccb7cdb6 Mon Sep 17 00:00:00 2001 From: Arnd Bergmann Date: Wed, 18 Feb 2015 21:43:13 +0100 Subject: ASoC: cirrus: tlv320aic23 needs I2C The tlv320aic23 codec is selected by the ep93xx snapper platform, which are missing a dependency on I2C, and that can result in this build error, as found during randconfig builds: .../codecs/tlv320aic23-i2c.c: In function 'tlv320aic23_i2c_probe': .../codecs/tlv320aic23-i2c.c:27:2: error: implicit declaration of function 'i2c_check_functionality' [-Werror=implicit-function-declaration] if (!i2c_check_functionality(i2c->adapter, I2C_FUNC_SMBUS_BYTE_DATA)) ^ This adds the missing dependency. Signed-off-by: Arnd Bergmann Signed-off-by: Mark Brown --- sound/soc/cirrus/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/cirrus/Kconfig b/sound/soc/cirrus/Kconfig index 7b7fbcd49e5e..c7cd60f009e9 100644 --- a/sound/soc/cirrus/Kconfig +++ b/sound/soc/cirrus/Kconfig @@ -16,7 +16,7 @@ config SND_EP93XX_SOC_AC97 config SND_EP93XX_SOC_SNAPPERCL15 tristate "SoC Audio support for Bluewater Systems Snapper CL15 module" - depends on SND_EP93XX_SOC && MACH_SNAPPER_CL15 + depends on SND_EP93XX_SOC && MACH_SNAPPER_CL15 && I2C select SND_EP93XX_SOC_I2S select SND_SOC_TLV320AIC23_I2C help -- cgit v1.2.3 From 08d0a55c33393e6dc838e37b7a8657c28a6de10d Mon Sep 17 00:00:00 2001 From: Kenneth Westfield Date: Tue, 17 Feb 2015 00:53:11 -0800 Subject: ASoC: max98357a: Add missing header files Add missing header files to avoid implicit declarations and indirect inclusions. Signed-off-by: Kenneth Westfield Signed-off-by: Mark Brown --- sound/soc/codecs/max98357a.c | 11 ++++++++++- 1 file changed, 10 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/max98357a.c b/sound/soc/codecs/max98357a.c index f493fb6fd4ea..e9e6efbc21dd 100644 --- a/sound/soc/codecs/max98357a.c +++ b/sound/soc/codecs/max98357a.c @@ -12,10 +12,19 @@ * max98357a.c -- MAX98357A ALSA SoC Codec driver */ -#include +#include +#include #include #include +#include +#include +#include +#include +#include +#include #include +#include +#include #define DRV_NAME "max98357a" -- cgit v1.2.3 From b3ec1c35385a16ddd98fdf104dcf4623a66e042a Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Thu, 12 Feb 2015 09:59:55 +0530 Subject: ASoC: Intel: update MMX ID to 3 The updated firmware expects the MMX ID to be used as 3, so update the driver as well Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/sst-atom-controls.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/intel/sst-atom-controls.h b/sound/soc/intel/sst-atom-controls.h index dfebfdd5eb2a..daecc58f28af 100644 --- a/sound/soc/intel/sst-atom-controls.h +++ b/sound/soc/intel/sst-atom-controls.h @@ -150,7 +150,7 @@ enum sst_cmd_type { enum sst_task { SST_TASK_SBA = 1, - SST_TASK_MMX, + SST_TASK_MMX = 3, }; enum sst_type { -- cgit v1.2.3 From a825ac7678a43f7a22ff19842baebcf4aa14e950 Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Thu, 12 Feb 2015 09:59:59 +0530 Subject: ASoC: Intel: save and restore the CSR register The IPC driver saved only IMR register, we need to save the CSR as well, so add it Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/sst/sst.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/intel/sst/sst.c b/sound/soc/intel/sst/sst.c index d6ea80076ea2..97234ec4e416 100644 --- a/sound/soc/intel/sst/sst.c +++ b/sound/soc/intel/sst/sst.c @@ -350,7 +350,9 @@ static inline void sst_save_shim64(struct intel_sst_drv *ctx, spin_lock_irqsave(&ctx->ipc_spin_lock, irq_flags); - shim_regs->imrx = sst_shim_read64(shim, SST_IMRX), + shim_regs->imrx = sst_shim_read64(shim, SST_IMRX); + shim_regs->csr = sst_shim_read64(shim, SST_CSR); + spin_unlock_irqrestore(&ctx->ipc_spin_lock, irq_flags); } @@ -367,6 +369,7 @@ static inline void sst_restore_shim64(struct intel_sst_drv *ctx, */ spin_lock_irqsave(&ctx->ipc_spin_lock, irq_flags); sst_shim_write64(shim, SST_IMRX, shim_regs->imrx), + sst_shim_write64(shim, SST_CSR, shim_regs->csr), spin_unlock_irqrestore(&ctx->ipc_spin_lock, irq_flags); } -- cgit v1.2.3 From de251d773bb214fa5e7666a0da1225528e07da5e Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Thu, 12 Feb 2015 10:00:00 +0530 Subject: ASoC: Intel: reset the DSP while suspending The manual recommends that we reset the DSP when we suspend so add that in runtime suspend handler Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/sst/sst.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/intel/sst/sst.c b/sound/soc/intel/sst/sst.c index 97234ec4e416..11c578651c1c 100644 --- a/sound/soc/intel/sst/sst.c +++ b/sound/soc/intel/sst/sst.c @@ -416,6 +416,7 @@ static int intel_sst_runtime_suspend(struct device *dev) synchronize_irq(ctx->irq_num); flush_workqueue(ctx->post_msg_wq); + ctx->ops->reset(ctx); /* save the shim registers because PMC doesn't save state */ sst_save_shim64(ctx, ctx->shim, ctx->shim_regs64); -- cgit v1.2.3 From 850529249d7cce02e9bfae9476d09c8c51410d28 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Mon, 16 Feb 2015 13:06:45 +0800 Subject: ASoC: rt5670: Set RT5670_IRQ_CTRL1 non volatile RT5670_IRQ_CTRL1(0xbd) is a non volatile register. And we need to restore its value after suspend/resume. Signed-off-by: Bard Liao Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/rt5670.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index d33f33ce865a..b651bc06cfdf 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -223,7 +223,6 @@ static bool rt5670_volatile_register(struct device *dev, unsigned int reg) case RT5670_ADC_EQ_CTRL1: case RT5670_EQ_CTRL1: case RT5670_ALC_CTRL_1: - case RT5670_IRQ_CTRL1: case RT5670_IRQ_CTRL2: case RT5670_INT_IRQ_ST: case RT5670_IL_CMD: -- cgit v1.2.3 From 148388f375394ac1afed543cb653c94be5faa810 Mon Sep 17 00:00:00 2001 From: Thomas Niederprüm Date: Sat, 21 Feb 2015 17:22:38 +0100 Subject: ASoC: sta32x: fix register range in regmap. MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The STA32X_AUTO3 is a writable register that currently does not appear in the regmap ranges(neither read nor write). By adding this register to the register ranges there is no gap anymore and the existing register ranges can be joined. This fixes a regression introduced in commit a1be4cead9b9504aa6fc93b624975601cec8c188 where the driver was moved to direct regmap usage and the STA32X_AUTO3 register was missed. That made it impossible to choose the preset EQ mode set through the STA32X_AUTO3 register. Fixes: a1be4cead9 (ASoC: sta32x: Convert to direct regmap API usage) Signed-off-by: Thomas Niederprüm Signed-off-by: Mark Brown --- sound/soc/codecs/sta32x.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c index 3a1343fa109b..007a0e3bc273 100644 --- a/sound/soc/codecs/sta32x.c +++ b/sound/soc/codecs/sta32x.c @@ -106,13 +106,11 @@ static const struct reg_default sta32x_regs[] = { }; static const struct regmap_range sta32x_write_regs_range[] = { - regmap_reg_range(STA32X_CONFA, STA32X_AUTO2), - regmap_reg_range(STA32X_C1CFG, STA32X_FDRC2), + regmap_reg_range(STA32X_CONFA, STA32X_FDRC2), }; static const struct regmap_range sta32x_read_regs_range[] = { - regmap_reg_range(STA32X_CONFA, STA32X_AUTO2), - regmap_reg_range(STA32X_C1CFG, STA32X_FDRC2), + regmap_reg_range(STA32X_CONFA, STA32X_FDRC2), }; static const struct regmap_range sta32x_volatile_regs_range[] = { -- cgit v1.2.3 From 70068776c49b37fe0c8f9115cec068d07375c6fb Mon Sep 17 00:00:00 2001 From: Oder Chiou Date: Wed, 25 Feb 2015 17:36:13 +0800 Subject: ASoC: rt5677: Correct the routing paths of that after IF1/2 DACx Mux The patch corrects the routing paths of that after IF1/2 DACx Mux Signed-off-by: Oder Chiou Signed-off-by: Mark Brown --- sound/soc/codecs/rt5677.c | 32 ++++++++++++++++---------------- 1 file changed, 16 insertions(+), 16 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 5d0bb8748dd1..fb9c20eace3f 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -3284,8 +3284,8 @@ static const struct snd_soc_dapm_route rt5677_dapm_routes[] = { { "IB45 Bypass Mux", "Bypass", "IB45 Mux" }, { "IB45 Bypass Mux", "Pass SRC", "IB45 Mux" }, - { "IB6 Mux", "IF1 DAC 6", "IF1 DAC6" }, - { "IB6 Mux", "IF2 DAC 6", "IF2 DAC6" }, + { "IB6 Mux", "IF1 DAC 6", "IF1 DAC6 Mux" }, + { "IB6 Mux", "IF2 DAC 6", "IF2 DAC6 Mux" }, { "IB6 Mux", "SLB DAC 6", "SLB DAC6" }, { "IB6 Mux", "STO4 ADC MIX L", "Stereo4 ADC MIXL" }, { "IB6 Mux", "IF4 DAC L", "IF4 DAC L" }, @@ -3293,8 +3293,8 @@ static const struct snd_soc_dapm_route rt5677_dapm_routes[] = { { "IB6 Mux", "STO2 ADC MIX L", "Stereo2 ADC MIXL" }, { "IB6 Mux", "STO3 ADC MIX L", "Stereo3 ADC MIXL" }, - { "IB7 Mux", "IF1 DAC 7", "IF1 DAC7" }, - { "IB7 Mux", "IF2 DAC 7", "IF2 DAC7" }, + { "IB7 Mux", "IF1 DAC 7", "IF1 DAC7 Mux" }, + { "IB7 Mux", "IF2 DAC 7", "IF2 DAC7 Mux" }, { "IB7 Mux", "SLB DAC 7", "SLB DAC7" }, { "IB7 Mux", "STO4 ADC MIX R", "Stereo4 ADC MIXR" }, { "IB7 Mux", "IF4 DAC R", "IF4 DAC R" }, @@ -3635,15 +3635,15 @@ static const struct snd_soc_dapm_route rt5677_dapm_routes[] = { { "DAC1 FS", NULL, "DAC1 MIXL" }, { "DAC1 FS", NULL, "DAC1 MIXR" }, - { "DAC2 L Mux", "IF1 DAC 2", "IF1 DAC2" }, - { "DAC2 L Mux", "IF2 DAC 2", "IF2 DAC2" }, + { "DAC2 L Mux", "IF1 DAC 2", "IF1 DAC2 Mux" }, + { "DAC2 L Mux", "IF2 DAC 2", "IF2 DAC2 Mux" }, { "DAC2 L Mux", "IF3 DAC L", "IF3 DAC L" }, { "DAC2 L Mux", "IF4 DAC L", "IF4 DAC L" }, { "DAC2 L Mux", "SLB DAC 2", "SLB DAC2" }, { "DAC2 L Mux", "OB 2", "OutBound2" }, - { "DAC2 R Mux", "IF1 DAC 3", "IF1 DAC3" }, - { "DAC2 R Mux", "IF2 DAC 3", "IF2 DAC3" }, + { "DAC2 R Mux", "IF1 DAC 3", "IF1 DAC3 Mux" }, + { "DAC2 R Mux", "IF2 DAC 3", "IF2 DAC3 Mux" }, { "DAC2 R Mux", "IF3 DAC R", "IF3 DAC R" }, { "DAC2 R Mux", "IF4 DAC R", "IF4 DAC R" }, { "DAC2 R Mux", "SLB DAC 3", "SLB DAC3" }, @@ -3651,29 +3651,29 @@ static const struct snd_soc_dapm_route rt5677_dapm_routes[] = { { "DAC2 R Mux", "Haptic Generator", "Haptic Generator" }, { "DAC2 R Mux", "VAD ADC", "VAD ADC Mux" }, - { "DAC3 L Mux", "IF1 DAC 4", "IF1 DAC4" }, - { "DAC3 L Mux", "IF2 DAC 4", "IF2 DAC4" }, + { "DAC3 L Mux", "IF1 DAC 4", "IF1 DAC4 Mux" }, + { "DAC3 L Mux", "IF2 DAC 4", "IF2 DAC4 Mux" }, { "DAC3 L Mux", "IF3 DAC L", "IF3 DAC L" }, { "DAC3 L Mux", "IF4 DAC L", "IF4 DAC L" }, { "DAC3 L Mux", "SLB DAC 4", "SLB DAC4" }, { "DAC3 L Mux", "OB 4", "OutBound4" }, - { "DAC3 R Mux", "IF1 DAC 5", "IF1 DAC4" }, - { "DAC3 R Mux", "IF2 DAC 5", "IF2 DAC4" }, + { "DAC3 R Mux", "IF1 DAC 5", "IF1 DAC5 Mux" }, + { "DAC3 R Mux", "IF2 DAC 5", "IF2 DAC5 Mux" }, { "DAC3 R Mux", "IF3 DAC R", "IF3 DAC R" }, { "DAC3 R Mux", "IF4 DAC R", "IF4 DAC R" }, { "DAC3 R Mux", "SLB DAC 5", "SLB DAC5" }, { "DAC3 R Mux", "OB 5", "OutBound5" }, - { "DAC4 L Mux", "IF1 DAC 6", "IF1 DAC6" }, - { "DAC4 L Mux", "IF2 DAC 6", "IF2 DAC6" }, + { "DAC4 L Mux", "IF1 DAC 6", "IF1 DAC6 Mux" }, + { "DAC4 L Mux", "IF2 DAC 6", "IF2 DAC6 Mux" }, { "DAC4 L Mux", "IF3 DAC L", "IF3 DAC L" }, { "DAC4 L Mux", "IF4 DAC L", "IF4 DAC L" }, { "DAC4 L Mux", "SLB DAC 6", "SLB DAC6" }, { "DAC4 L Mux", "OB 6", "OutBound6" }, - { "DAC4 R Mux", "IF1 DAC 7", "IF1 DAC7" }, - { "DAC4 R Mux", "IF2 DAC 7", "IF2 DAC7" }, + { "DAC4 R Mux", "IF1 DAC 7", "IF1 DAC7 Mux" }, + { "DAC4 R Mux", "IF2 DAC 7", "IF2 DAC7 Mux" }, { "DAC4 R Mux", "IF3 DAC R", "IF3 DAC R" }, { "DAC4 R Mux", "IF4 DAC R", "IF4 DAC R" }, { "DAC4 R Mux", "SLB DAC 7", "SLB DAC7" }, -- cgit v1.2.3 From 8af4baa7087a0ae74c6ee29d4d979a60e14b119e Mon Sep 17 00:00:00 2001 From: Thomas Niederprüm Date: Sat, 21 Feb 2015 18:11:29 +0100 Subject: ASoC: OMAP: mcbsp: Fix CLKX and CLKR pinmux when used as inputs MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit This patch fixes faulty behaviour in a setup where the input clock for the SRG is fed through the CLKR/CLKX pin but the McBSP is configured to be master (SND_SOC_DAIFMT_CBS_CFS). In that case of course CLKR/CLKX must not be configured as output pin. Otherwise the input clock is messed up horribly. This patch makes it possible to use the CLKR/CLKX pin rather than CLKS to inject a reference clock in setups where McBSP is master and not both rx and tx are used. However for this to work it has to be ensured that set_dai_sysclk() is called after set_dai_fmt(). This was tested on a beagleboard-xm using McBSP1 to drive a i2s DAC through the tx lines (CLKX,FSX,DX). Using this patch the CLKR pin is used to inject an external reference clock. Signed-off-by: Thomas Niederprüm Acked-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/omap/omap-mcbsp.c | 11 +++++++++++ 1 file changed, 11 insertions(+) (limited to 'sound') diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index c7eb9dd67f60..fd99d89de6a8 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -530,8 +530,19 @@ static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai, case OMAP_MCBSP_SYSCLK_CLKX_EXT: regs->srgr2 |= CLKSM; + regs->pcr0 |= SCLKME; + /* + * If McBSP is master but yet the CLKX/CLKR pin drives the SRG, + * disable output on those pins. This enables to inject the + * reference clock through CLKX/CLKR. For this to work + * set_dai_sysclk() _needs_ to be called after set_dai_fmt(). + */ + regs->pcr0 &= ~CLKXM; + break; case OMAP_MCBSP_SYSCLK_CLKR_EXT: regs->pcr0 |= SCLKME; + /* Disable ouput on CLKR pin in master mode */ + regs->pcr0 &= ~CLKRM; break; default: err = -ENODEV; -- cgit v1.2.3 From f2b14c0bc510c6a8f67a4f36049deefe5d99a537 Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Fri, 27 Feb 2015 09:39:32 +0900 Subject: ALSA: oxfw: fix a condition and return code in start_stream() The amdtp_stream_wait_callback() doesn't return minus value and the return code is not for error code. This commit fixes with a propper condition and an error code. Fixes: f3699e2c7745 ('ALSA: oxfw: Change the way to start stream') Reported-by: Dan Carpenter Signed-off-by: Takashi Sakamoto Cc: # 3.19+ Signed-off-by: Takashi Iwai --- sound/firewire/oxfw/oxfw-stream.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/firewire/oxfw/oxfw-stream.c b/sound/firewire/oxfw/oxfw-stream.c index 29ccb3637164..e6757cd85724 100644 --- a/sound/firewire/oxfw/oxfw-stream.c +++ b/sound/firewire/oxfw/oxfw-stream.c @@ -171,9 +171,10 @@ static int start_stream(struct snd_oxfw *oxfw, struct amdtp_stream *stream, } /* Wait first packet */ - err = amdtp_stream_wait_callback(stream, CALLBACK_TIMEOUT); - if (err < 0) + if (!amdtp_stream_wait_callback(stream, CALLBACK_TIMEOUT)) { stop_stream(oxfw, stream); + err = -ETIMEDOUT; + } end: return err; } -- cgit v1.2.3 From 8cdebf71098c07168ef6335e2f1f35d85dbe3049 Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Sun, 1 Mar 2015 18:12:16 +0900 Subject: ALSA: dice: fix wrong offsets for Dice interface For received packet stream, the offset of 'RX_SEQ_START' locates after the offset of 'RX_NUMBER_MIDI', although current macro and proc output includes wrong offsets. Fortunately, this bug doesn't affect streaming functionality because these macro is not used. This commit fixes these wrong macro and outputs. Signed-off-by: Takashi Sakamoto Signed-off-by: Takashi Iwai --- sound/firewire/dice/dice-interface.h | 18 +++++++++--------- sound/firewire/dice/dice-proc.c | 4 ++-- 2 files changed, 11 insertions(+), 11 deletions(-) (limited to 'sound') diff --git a/sound/firewire/dice/dice-interface.h b/sound/firewire/dice/dice-interface.h index 27b044f84c81..de7602bd69b5 100644 --- a/sound/firewire/dice/dice-interface.h +++ b/sound/firewire/dice/dice-interface.h @@ -298,24 +298,24 @@ */ #define RX_ISOCHRONOUS 0x008 -/* - * Index of first quadlet to be interpreted; read/write. If > 0, that many - * quadlets at the beginning of each data block will be ignored, and all the - * audio and MIDI quadlets will follow. - */ -#define RX_SEQ_START 0x00c - /* * The number of audio channels; read-only. There will be one quadlet per * channel. */ -#define RX_NUMBER_AUDIO 0x010 +#define RX_NUMBER_AUDIO 0x00c /* * The number of MIDI ports, 0-8; read-only. If > 0, there will be one * additional quadlet in each data block, following the audio quadlets. */ -#define RX_NUMBER_MIDI 0x014 +#define RX_NUMBER_MIDI 0x010 + +/* + * Index of first quadlet to be interpreted; read/write. If > 0, that many + * quadlets at the beginning of each data block will be ignored, and all the + * audio and MIDI quadlets will follow. + */ +#define RX_SEQ_START 0x014 /* * Names of all audio channels; read-only. Quadlets are byte-swapped. Names diff --git a/sound/firewire/dice/dice-proc.c b/sound/firewire/dice/dice-proc.c index f5c1d1bced59..ecfe20fd4de5 100644 --- a/sound/firewire/dice/dice-proc.c +++ b/sound/firewire/dice/dice-proc.c @@ -99,9 +99,9 @@ static void dice_proc_read(struct snd_info_entry *entry, } tx; struct { u32 iso; - u32 seq_start; u32 number_audio; u32 number_midi; + u32 seq_start; char names[RX_NAMES_SIZE]; u32 ac3_caps; u32 ac3_enable; @@ -204,10 +204,10 @@ static void dice_proc_read(struct snd_info_entry *entry, break; snd_iprintf(buffer, "rx %u:\n", stream); snd_iprintf(buffer, " iso channel: %d\n", (int)buf.rx.iso); - snd_iprintf(buffer, " sequence start: %u\n", buf.rx.seq_start); snd_iprintf(buffer, " audio channels: %u\n", buf.rx.number_audio); snd_iprintf(buffer, " midi ports: %u\n", buf.rx.number_midi); + snd_iprintf(buffer, " sequence start: %u\n", buf.rx.seq_start); if (quadlets >= 68) { dice_proc_fixup_string(buf.rx.names, RX_NAMES_SIZE); snd_iprintf(buffer, " names: %s\n", buf.rx.names); -- cgit v1.2.3 From d7a6fe015b2abe33565538a3faf757e095e094e7 Mon Sep 17 00:00:00 2001 From: Alexandre Belloni Date: Tue, 6 Jan 2015 12:14:32 +0100 Subject: ASoC: sam9g20_wm8731: drop machine_is_xxx Atmel based boards can now only be used with device tree. Drop non DT initialization. Signed-off-by: Alexandre Belloni Signed-off-by: Mark Brown --- sound/soc/atmel/sam9g20_wm8731.c | 68 ++++++++++++++++++---------------------- 1 file changed, 31 insertions(+), 37 deletions(-) (limited to 'sound') diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c index f5ad214663f9..8de836165cf2 100644 --- a/sound/soc/atmel/sam9g20_wm8731.c +++ b/sound/soc/atmel/sam9g20_wm8731.c @@ -46,8 +46,6 @@ #include #include -#include - #include "../codecs/wm8731.h" #include "atmel-pcm.h" #include "atmel_ssc_dai.h" @@ -171,9 +169,7 @@ static int at91sam9g20ek_audio_probe(struct platform_device *pdev) int ret; if (!np) { - if (!(machine_is_at91sam9g20ek() || - machine_is_at91sam9g20ek_2mmc())) - return -ENODEV; + return -ENODEV; } ret = atmel_ssc_set_audio(0); @@ -210,39 +206,37 @@ static int at91sam9g20ek_audio_probe(struct platform_device *pdev) card->dev = &pdev->dev; /* Parse device node info */ - if (np) { - ret = snd_soc_of_parse_card_name(card, "atmel,model"); - if (ret) - goto err; - - ret = snd_soc_of_parse_audio_routing(card, - "atmel,audio-routing"); - if (ret) - goto err; - - /* Parse codec info */ - at91sam9g20ek_dai.codec_name = NULL; - codec_np = of_parse_phandle(np, "atmel,audio-codec", 0); - if (!codec_np) { - dev_err(&pdev->dev, "codec info missing\n"); - return -EINVAL; - } - at91sam9g20ek_dai.codec_of_node = codec_np; - - /* Parse dai and platform info */ - at91sam9g20ek_dai.cpu_dai_name = NULL; - at91sam9g20ek_dai.platform_name = NULL; - cpu_np = of_parse_phandle(np, "atmel,ssc-controller", 0); - if (!cpu_np) { - dev_err(&pdev->dev, "dai and pcm info missing\n"); - return -EINVAL; - } - at91sam9g20ek_dai.cpu_of_node = cpu_np; - at91sam9g20ek_dai.platform_of_node = cpu_np; - - of_node_put(codec_np); - of_node_put(cpu_np); + ret = snd_soc_of_parse_card_name(card, "atmel,model"); + if (ret) + goto err; + + ret = snd_soc_of_parse_audio_routing(card, + "atmel,audio-routing"); + if (ret) + goto err; + + /* Parse codec info */ + at91sam9g20ek_dai.codec_name = NULL; + codec_np = of_parse_phandle(np, "atmel,audio-codec", 0); + if (!codec_np) { + dev_err(&pdev->dev, "codec info missing\n"); + return -EINVAL; + } + at91sam9g20ek_dai.codec_of_node = codec_np; + + /* Parse dai and platform info */ + at91sam9g20ek_dai.cpu_dai_name = NULL; + at91sam9g20ek_dai.platform_name = NULL; + cpu_np = of_parse_phandle(np, "atmel,ssc-controller", 0); + if (!cpu_np) { + dev_err(&pdev->dev, "dai and pcm info missing\n"); + return -EINVAL; } + at91sam9g20ek_dai.cpu_of_node = cpu_np; + at91sam9g20ek_dai.platform_of_node = cpu_np; + + of_node_put(codec_np); + of_node_put(cpu_np); ret = snd_soc_register_card(card); if (ret) { -- cgit v1.2.3 From 31f3032c1a5504259f6fa8e0c7f8d2d3e2f5db48 Mon Sep 17 00:00:00 2001 From: Vishal Thanki Date: Tue, 3 Mar 2015 18:59:00 +0530 Subject: ASoC: simple-card: Add a NULL pointer check in asoc_simple_card_dai_link_of Make sure devm_kzalloc() succeeds. Signed-off-by: Vishal Thanki Signed-off-by: Mark Brown --- sound/soc/generic/simple-card.c | 5 +++++ 1 file changed, 5 insertions(+) (limited to 'sound') diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index f7c6734bd5da..fb550b5869d2 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -372,6 +372,11 @@ static int asoc_simple_card_dai_link_of(struct device_node *node, strlen(dai_link->cpu_dai_name) + strlen(dai_link->codec_dai_name) + 2, GFP_KERNEL); + if (!name) { + ret = -ENOMEM; + goto dai_link_of_err; + } + sprintf(name, "%s-%s", dai_link->cpu_dai_name, dai_link->codec_dai_name); dai_link->name = dai_link->stream_name = name; -- cgit v1.2.3 From d51199a83a2cf82a291d19ee852c44caa511427d Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 3 Mar 2015 13:38:14 +0200 Subject: ASoC: omap-pcm: Correct dma mask DMA_BIT_MASK of 64 is not valid dma address mask for OMAPs, it should be set to 32. The 64 was introduced by commit (in 2009): a152ff24b978 ASoC: OMAP: Make DMA 64 aligned But the dma_mask and coherent_dma_mask can not be used to specify alignment. Fixes: a152ff24b978 (ASoC: OMAP: Make DMA 64 aligned) Reported-by: Grygorii Strashko Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/omap/omap-pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index f4b05bc23e4b..1343ecbf0bd5 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -201,7 +201,7 @@ static int omap_pcm_new(struct snd_soc_pcm_runtime *rtd) struct snd_pcm *pcm = rtd->pcm; int ret; - ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(64)); + ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(32)); if (ret) return ret; -- cgit v1.2.3 From 096a020a9ef5c947577d3b57199bfc9b7e686b49 Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Thu, 5 Mar 2015 14:26:37 +0300 Subject: ALSA: msnd: add some missing curly braces There were some curly braces intended here. Signed-off-by: Dan Carpenter Signed-off-by: Takashi Iwai --- sound/isa/msnd/msnd_pinnacle_mixer.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/isa/msnd/msnd_pinnacle_mixer.c b/sound/isa/msnd/msnd_pinnacle_mixer.c index 17e49a071af4..b408540798c1 100644 --- a/sound/isa/msnd/msnd_pinnacle_mixer.c +++ b/sound/isa/msnd/msnd_pinnacle_mixer.c @@ -306,11 +306,12 @@ int snd_msndmix_new(struct snd_card *card) spin_lock_init(&chip->mixer_lock); strcpy(card->mixername, "MSND Pinnacle Mixer"); - for (idx = 0; idx < ARRAY_SIZE(snd_msnd_controls); idx++) + for (idx = 0; idx < ARRAY_SIZE(snd_msnd_controls); idx++) { err = snd_ctl_add(card, snd_ctl_new1(snd_msnd_controls + idx, chip)); if (err < 0) return err; + } return 0; } -- cgit v1.2.3 From f44f07cf3910f84b15b2a78c4933d5946bf409cf Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 5 Mar 2015 13:03:28 +0100 Subject: ALSA: line6: Clamp values correctly The usages of clamp() macro in sound/usb/line6/playback.c are just wrong, the low and high values are swapped. Reported-by: Dan Carpenter Signed-off-by: Takashi Iwai --- sound/usb/line6/playback.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/usb/line6/playback.c b/sound/usb/line6/playback.c index 05dee690f487..97ed593f6010 100644 --- a/sound/usb/line6/playback.c +++ b/sound/usb/line6/playback.c @@ -39,7 +39,7 @@ static void change_volume(struct urb *urb_out, int volume[], for (; p < buf_end; ++p) { short pv = le16_to_cpu(*p); int val = (pv * volume[chn & 1]) >> 8; - pv = clamp(val, 0x7fff, -0x8000); + pv = clamp(val, -0x8000, 0x7fff); *p = cpu_to_le16(pv); ++chn; } @@ -54,7 +54,7 @@ static void change_volume(struct urb *urb_out, int volume[], val = p[0] + (p[1] << 8) + ((signed char)p[2] << 16); val = (val * volume[chn & 1]) >> 8; - val = clamp(val, 0x7fffff, -0x800000); + val = clamp(val, -0x800000, 0x7fffff); p[0] = val; p[1] = val >> 8; p[2] = val >> 16; @@ -126,7 +126,7 @@ static void add_monitor_signal(struct urb *urb_out, unsigned char *signal, short pov = le16_to_cpu(*po); short piv = le16_to_cpu(*pi); int val = pov + ((piv * volume) >> 8); - pov = clamp(val, 0x7fff, -0x8000); + pov = clamp(val, -0x8000, 0x7fff); *po = cpu_to_le16(pov); } } -- cgit v1.2.3 From d124380674b58f62d0ef974630d74d67bb8afeb0 Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Thu, 5 Mar 2015 20:49:06 +0300 Subject: ALSA: opl3: small array underflow There is a missing lower bound check on "pitchbend" so it means we can read up to 6 elements before the start of the opl3_note_table[] array. Thanks to Clemens Ladisch for his help with this patch. Signed-off-by: Dan Carpenter Signed-off-by: Takashi Iwai --- sound/drivers/opl3/opl3_midi.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/drivers/opl3/opl3_midi.c b/sound/drivers/opl3/opl3_midi.c index f62780ed64ad..7821b07415a7 100644 --- a/sound/drivers/opl3/opl3_midi.c +++ b/sound/drivers/opl3/opl3_midi.c @@ -105,6 +105,8 @@ static void snd_opl3_calc_pitch(unsigned char *fnum, unsigned char *blocknum, int pitchbend = chan->midi_pitchbend; int segment; + if (pitchbend < -0x2000) + pitchbend = -0x2000; if (pitchbend > 0x1FFF) pitchbend = 0x1FFF; -- cgit v1.2.3 From 70658b99490dd86cfdbf4fca117bbe2ef9a80d03 Mon Sep 17 00:00:00 2001 From: Hui Wang Date: Fri, 6 Mar 2015 14:03:57 +0800 Subject: ALSA: hda - One more Dell macine needs DELL1_MIC_NO_PRESENCE quirk Cc: BugLink: https://bugs.launchpad.net/bugs/1428947 Signed-off-by: Hui Wang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 7 +++++++ 1 file changed, 7 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b2b24a8b3dac..526398a4a442 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5209,6 +5209,13 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { {0x17, 0x40000000}, {0x1d, 0x40700001}, {0x21, 0x02211040}), + SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, + ALC255_STANDARD_PINS, + {0x12, 0x90a60170}, + {0x14, 0x90170140}, + {0x17, 0x40000000}, + {0x1d, 0x40700001}, + {0x21, 0x02211050}), SND_HDA_PIN_QUIRK(0x10ec0280, 0x103c, "HP", ALC280_FIXUP_HP_GPIO4, {0x12, 0x90a60130}, {0x13, 0x40000000}, -- cgit v1.2.3 From a1f3f1ca66bd12c339b17a0c2ef93a093f90a277 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sun, 8 Mar 2015 18:29:50 +0100 Subject: ALSA: hda - Fix regression of HD-audio controller fallback modes The commit [63e51fd708f5: ALSA: hda - Don't take unresponsive D3 transition too serious] introduced a conditional fallback behavior to the HD-audio controller depending on the flag set. However, it introduced a silly bug, too, that the flag was evaluated in a reverse way. This resulted in a regression of HD-audio controller driver where it can't go to the fallback mode at communication errors. Unfortunately (or fortunately?) this didn't come up until recently because the affected code path is an error handling that happens only on an unstable hardware chip. Most of recent chips work stably, thus they didn't hit this problem. Now, we've got a regression report with a VIA chip, and this seems indeed requiring the fallback to the polling mode, and finally the bug was revealed. The fix is a oneliner to remove the wrong logical NOT in the check. (Lesson learned - be careful about double negation.) The bug should be backported to stable, but the patch won't be applicable to 3.13 or earlier because of the code splits. The stable fix patches for earlier kernels will be posted later manually. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=94021 Fixes: 63e51fd708f5 ('ALSA: hda - Don't take unresponsive D3 transition too serious') Cc: # v3.14+ Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_controller.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index a2ce773bdc62..17c2637d842c 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -1164,7 +1164,7 @@ static unsigned int azx_rirb_get_response(struct hda_bus *bus, } } - if (!bus->no_response_fallback) + if (bus->no_response_fallback) return -1; if (!chip->polling_mode && chip->poll_count < 2) { -- cgit v1.2.3 From 5b1274efe2a24eb5a85a00cc48c334b1cdfc75aa Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Tue, 10 Mar 2015 21:58:48 +0900 Subject: Revert "ALSA: dice: fix wrong offsets for Dice interface" This reverts commit 8cdebf71098c07168ef6335e2f1f35d85dbe3049. The reverted commit breaks out-stream functionality of Dice driver. Signed-off-by: Takashi Sakamoto Signed-off-by: Takashi Iwai --- sound/firewire/dice/dice-interface.h | 18 +++++++++--------- sound/firewire/dice/dice-proc.c | 4 ++-- 2 files changed, 11 insertions(+), 11 deletions(-) (limited to 'sound') diff --git a/sound/firewire/dice/dice-interface.h b/sound/firewire/dice/dice-interface.h index de7602bd69b5..27b044f84c81 100644 --- a/sound/firewire/dice/dice-interface.h +++ b/sound/firewire/dice/dice-interface.h @@ -298,24 +298,24 @@ */ #define RX_ISOCHRONOUS 0x008 +/* + * Index of first quadlet to be interpreted; read/write. If > 0, that many + * quadlets at the beginning of each data block will be ignored, and all the + * audio and MIDI quadlets will follow. + */ +#define RX_SEQ_START 0x00c + /* * The number of audio channels; read-only. There will be one quadlet per * channel. */ -#define RX_NUMBER_AUDIO 0x00c +#define RX_NUMBER_AUDIO 0x010 /* * The number of MIDI ports, 0-8; read-only. If > 0, there will be one * additional quadlet in each data block, following the audio quadlets. */ -#define RX_NUMBER_MIDI 0x010 - -/* - * Index of first quadlet to be interpreted; read/write. If > 0, that many - * quadlets at the beginning of each data block will be ignored, and all the - * audio and MIDI quadlets will follow. - */ -#define RX_SEQ_START 0x014 +#define RX_NUMBER_MIDI 0x014 /* * Names of all audio channels; read-only. Quadlets are byte-swapped. Names diff --git a/sound/firewire/dice/dice-proc.c b/sound/firewire/dice/dice-proc.c index ecfe20fd4de5..f5c1d1bced59 100644 --- a/sound/firewire/dice/dice-proc.c +++ b/sound/firewire/dice/dice-proc.c @@ -99,9 +99,9 @@ static void dice_proc_read(struct snd_info_entry *entry, } tx; struct { u32 iso; + u32 seq_start; u32 number_audio; u32 number_midi; - u32 seq_start; char names[RX_NAMES_SIZE]; u32 ac3_caps; u32 ac3_enable; @@ -204,10 +204,10 @@ static void dice_proc_read(struct snd_info_entry *entry, break; snd_iprintf(buffer, "rx %u:\n", stream); snd_iprintf(buffer, " iso channel: %d\n", (int)buf.rx.iso); + snd_iprintf(buffer, " sequence start: %u\n", buf.rx.seq_start); snd_iprintf(buffer, " audio channels: %u\n", buf.rx.number_audio); snd_iprintf(buffer, " midi ports: %u\n", buf.rx.number_midi); - snd_iprintf(buffer, " sequence start: %u\n", buf.rx.seq_start); if (quadlets >= 68) { dice_proc_fixup_string(buf.rx.names, RX_NAMES_SIZE); snd_iprintf(buffer, " names: %s\n", buf.rx.names); -- cgit v1.2.3 From 59294a01d7037f63fb8bf994af10ce63c618770a Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Tue, 10 Mar 2015 21:54:35 +0900 Subject: ALSA: firewire-lib: leave unit reference counting completely With previous commit, this module managed to leave the counting to each drivers, but the isochronous resources functionality still increment/decrement the count. This commit purge such codes to leave the responsibility to each drivers. Fix: c6f224dc20ad ('ALSA: firewire-lib: remove reference counting') Signed-off-by: Takashi Sakamoto Signed-off-by: Takashi Iwai --- sound/firewire/iso-resources.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/firewire/iso-resources.c b/sound/firewire/iso-resources.c index 5f17b77ee152..f0e4d502d604 100644 --- a/sound/firewire/iso-resources.c +++ b/sound/firewire/iso-resources.c @@ -26,7 +26,7 @@ int fw_iso_resources_init(struct fw_iso_resources *r, struct fw_unit *unit) { r->channels_mask = ~0uLL; - r->unit = fw_unit_get(unit); + r->unit = unit; mutex_init(&r->mutex); r->allocated = false; @@ -42,7 +42,6 @@ void fw_iso_resources_destroy(struct fw_iso_resources *r) { WARN_ON(r->allocated); mutex_destroy(&r->mutex); - fw_unit_put(r->unit); } EXPORT_SYMBOL(fw_iso_resources_destroy); -- cgit v1.2.3 From ddb6ca75b5671b8fbf1909bc588c449ee74b34f9 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 11 Mar 2015 16:05:19 +0100 Subject: ALSA: hda - Fix built-in mic on Compaq Presario CQ60 Compaq Presario CQ60 laptop with CX20561 gives a wrong pin for the built-in mic NID 0x17 instead of NID 0x1d, and it results in the non-working mic. This patch just remaps the pin correctly via fixup. Bugzilla: https://bugzilla.opensuse.org/show_bug.cgi?id=920604 Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 11 +++++++++++ 1 file changed, 11 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index fd3ed18670e9..da67ea8645a6 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -223,6 +223,7 @@ enum { CXT_PINCFG_LENOVO_TP410, CXT_PINCFG_LEMOTE_A1004, CXT_PINCFG_LEMOTE_A1205, + CXT_PINCFG_COMPAQ_CQ60, CXT_FIXUP_STEREO_DMIC, CXT_FIXUP_INC_MIC_BOOST, CXT_FIXUP_HEADPHONE_MIC_PIN, @@ -660,6 +661,15 @@ static const struct hda_fixup cxt_fixups[] = { .type = HDA_FIXUP_PINS, .v.pins = cxt_pincfg_lemote, }, + [CXT_PINCFG_COMPAQ_CQ60] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + /* 0x17 was falsely set up as a mic, it should 0x1d */ + { 0x17, 0x400001f0 }, + { 0x1d, 0x97a70120 }, + { } + } + }, [CXT_FIXUP_STEREO_DMIC] = { .type = HDA_FIXUP_FUNC, .v.func = cxt_fixup_stereo_dmic, @@ -769,6 +779,7 @@ static const struct hda_model_fixup cxt5047_fixup_models[] = { }; static const struct snd_pci_quirk cxt5051_fixups[] = { + SND_PCI_QUIRK(0x103c, 0x360b, "Compaq CQ60", CXT_PINCFG_COMPAQ_CQ60), SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo X200", CXT_PINCFG_LENOVO_X200), {} }; -- cgit v1.2.3 From 81efec851477957f964f9978921d5ae36d521d45 Mon Sep 17 00:00:00 2001 From: Shawn Guo Date: Wed, 25 Feb 2015 22:53:37 +0800 Subject: ASoC: fsl_spdif: fix struct clk pointer comparing Since commit 035a61c314eb ("clk: Make clk API return per-user struct clk instances"), clk API users can no longer check if two struct clk pointers are pointing to the same hardware clock, i.e. struct clk_hw, by simply comparing two pointers. That's because with the per-user clk change, a brand new struct clk is created whenever clients try to look up the clock by calling clk_get() or sister functions like clk_get_sys() and of_clk_get(). This changes the original behavior where the struct clk is only created for once when clock driver registers the clock to CCF in the first place. The net change here is before commit 035a61c314eb the struct clk pointer is unique for given hardware clock, while after the commit the pointers returned by clk lookup calls become different for the same hardware clock. That said, the struct clk pointer comparing in the code doesn't work any more. Call helper function clk_is_match() instead to fix the problem. Signed-off-by: Shawn Guo Signed-off-by: Michael Turquette Signed-off-by: Stephen Boyd --- sound/soc/fsl/fsl_spdif.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index 75870c0ea2c9..91eb3aef7f02 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -1049,7 +1049,7 @@ static u32 fsl_spdif_txclk_caldiv(struct fsl_spdif_priv *spdif_priv, enum spdif_txrate index, bool round) { const u32 rate[] = { 32000, 44100, 48000, 96000, 192000 }; - bool is_sysclk = clk == spdif_priv->sysclk; + bool is_sysclk = clk_is_match(clk, spdif_priv->sysclk); u64 rate_ideal, rate_actual, sub; u32 sysclk_dfmin, sysclk_dfmax; u32 txclk_df, sysclk_df, arate; @@ -1143,7 +1143,7 @@ static int fsl_spdif_probe_txclk(struct fsl_spdif_priv *spdif_priv, spdif_priv->txclk_src[index], rate[index]); dev_dbg(&pdev->dev, "use txclk df %d for %dHz sample rate\n", spdif_priv->txclk_df[index], rate[index]); - if (spdif_priv->txclk[index] == spdif_priv->sysclk) + if (clk_is_match(spdif_priv->txclk[index], spdif_priv->sysclk)) dev_dbg(&pdev->dev, "use sysclk df %d for %dHz sample rate\n", spdif_priv->sysclk_df[index], rate[index]); dev_dbg(&pdev->dev, "the best rate for %dHz sample rate is %dHz\n", -- cgit v1.2.3 From aaa6d06282a749d0df8e5e22e73f8a3372f96853 Mon Sep 17 00:00:00 2001 From: Shawn Guo Date: Wed, 25 Feb 2015 22:53:38 +0800 Subject: ASoC: kirkwood: fix struct clk pointer comparing Since commit 035a61c314eb ("clk: Make clk API return per-user struct clk instances"), clk API users can no longer check if two struct clk pointers are pointing to the same hardware clock, i.e. struct clk_hw, by simply comparing two pointers. That's because with the per-user clk change, a brand new struct clk is created whenever clients try to look up the clock by calling clk_get() or sister functions like clk_get_sys() and of_clk_get(). This changes the original behavior where the struct clk is only created for once when clock driver registers the clock to CCF in the first place. The net change here is before commit 035a61c314eb the struct clk pointer is unique for given hardware clock, while after the commit the pointers returned by clk lookup calls become different for the same hardware clock. That said, the struct clk pointer comparing in the code doesn't work any more. Call helper function clk_is_match() instead to fix the problem. Signed-off-by: Shawn Guo Signed-off-by: Michael Turquette Signed-off-by: Stephen Boyd --- sound/soc/kirkwood/kirkwood-i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index def7d8260c4e..d19483081f9b 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -579,7 +579,7 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) if (PTR_ERR(priv->extclk) == -EPROBE_DEFER) return -EPROBE_DEFER; } else { - if (priv->extclk == priv->clk) { + if (clk_is_match(priv->extclk, priv->clk)) { devm_clk_put(&pdev->dev, priv->extclk); priv->extclk = ERR_PTR(-EINVAL); } else { -- cgit v1.2.3 From be3bb8236db2d0fcd705062ae2e2a9d75131222f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 11 Mar 2015 18:12:49 +0100 Subject: ALSA: control: Add sanity checks for user ctl id name string There was no check about the id string of user control elements, so we accepted even a control element with an empty string, which is obviously bogus. This patch adds more sanity checks of id strings. Cc: Signed-off-by: Takashi Iwai --- sound/core/control.c | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'sound') diff --git a/sound/core/control.c b/sound/core/control.c index 35324a8e83c8..eeb691d1911f 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -1170,6 +1170,10 @@ static int snd_ctl_elem_add(struct snd_ctl_file *file, if (info->count < 1) return -EINVAL; + if (!*info->id.name) + return -EINVAL; + if (strnlen(info->id.name, sizeof(info->id.name)) >= sizeof(info->id.name)) + return -EINVAL; access = info->access == 0 ? SNDRV_CTL_ELEM_ACCESS_READWRITE : (info->access & (SNDRV_CTL_ELEM_ACCESS_READWRITE| SNDRV_CTL_ELEM_ACCESS_INACTIVE| -- cgit v1.2.3 From fcdcd1dec6d2c7b718385ec743ae5a9a233edad4 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Thu, 12 Mar 2015 09:41:32 +0100 Subject: ALSA: snd-usb: add quirks for Roland UA-22 MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The device complies to the UAC1 standard but hides that fact with proprietary descriptors. The autodetect quirk for Roland devices catches the audio interface but misses the MIDI part, so a specific quirk is needed. Signed-off-by: Daniel Mack Reported-by: Rafa Lafuente Tested-by: Raphaël Doursenaud Cc: Signed-off-by: Takashi Iwai --- sound/usb/quirks-table.h | 30 ++++++++++++++++++++++++++++++ 1 file changed, 30 insertions(+) (limited to 'sound') diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 67d476548dcf..07f984d5f516 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -1773,6 +1773,36 @@ YAMAHA_DEVICE(0x7010, "UB99"), } } }, +{ + USB_DEVICE(0x0582, 0x0159), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + /* .vendor_name = "Roland", */ + /* .product_name = "UA-22", */ + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 1, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 2, + .type = QUIRK_MIDI_FIXED_ENDPOINT, + .data = & (const struct snd_usb_midi_endpoint_info) { + .out_cables = 0x0001, + .in_cables = 0x0001 + } + }, + { + .ifnum = -1 + } + } + } +}, /* this catches most recent vendor-specific Roland devices */ { .match_flags = USB_DEVICE_ID_MATCH_VENDOR | -- cgit v1.2.3 From bad994f5b4ab57eec8d56c180edca00505c3eeb2 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 12 Mar 2015 20:28:04 +0100 Subject: ALSA: hda - Set single_adc_amp flag for CS420x codecs CS420x codecs seem to deal only the single amps of ADC nodes even though the nodes receive multiple inputs. This leads to the inconsistent amp value after S3/S4 resume, for example. The fix is just to set codec->single_adc_amp flag. Then the driver handles these ADC amps as if single connections. Reported-and-tested-by: Vasil Zlatanov Cc: # 3.9+ Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_cirrus.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 1589c9bcce3e..ab687ffb28c2 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -584,6 +584,7 @@ static int patch_cs420x(struct hda_codec *codec) return -ENOMEM; spec->gen.automute_hook = cs_automute; + codec->single_adc_amp = 1; snd_hda_pick_fixup(codec, cs420x_models, cs420x_fixup_tbl, cs420x_fixups); -- cgit v1.2.3 From 2ddee91abe9cc34ddb6294ee14702b46ae07d460 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 12 Mar 2015 20:47:15 +0100 Subject: ALSA: hda - Add workaround for MacBook Air 5,2 built-in mic MacBook Air 5,2 has the same problem as MacBook Pro 8,1 where the built-in mic records only the right channel. Apply the same workaround as MBP8,1 to spread the mono channel via a Cirrus codec vendor-specific COEF setup. Reported-and-tested-by: Vasil Zlatanov Cc: # 3.9+ Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_cirrus.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index ab687ffb28c2..dd2b3d92071f 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -393,6 +393,7 @@ static const struct snd_pci_quirk cs420x_fixup_tbl[] = { SND_PCI_QUIRK(0x106b, 0x1c00, "MacBookPro 8,1", CS420X_MBP81), SND_PCI_QUIRK(0x106b, 0x2000, "iMac 12,2", CS420X_IMAC27_122), SND_PCI_QUIRK(0x106b, 0x2800, "MacBookPro 10,1", CS420X_MBP101), + SND_PCI_QUIRK(0x106b, 0x5600, "MacBookAir 5,2", CS420X_MBP81), SND_PCI_QUIRK(0x106b, 0x5b00, "MacBookAir 4,2", CS420X_MBA42), SND_PCI_QUIRK_VENDOR(0x106b, "Apple", CS420X_APPLE), {} /* terminator */ -- cgit v1.2.3 From ef403edb75580a3ec5d155f5de82155f0419c621 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 12 Mar 2015 08:30:11 +0100 Subject: ALSA: hda - Don't access stereo amps for mono channel widgets The current HDA generic parser initializes / modifies the amp values always in stereo, but this seems causing the problem on ALC3229 codec that has a few mono channel widgets: namely, these mono widgets react to actions for both channels equally. In the driver code, we do care the mono channel and create a control only for the left channel (as defined in HD-audio spec) for such a node. When the control is updated, only the left channel value is changed. However, in the resume, the right channel value is also restored from the initial value we took as stereo, and this overwrites the left channel value. This ends up being the silent output as the right channel has been never touched and remains muted. This patch covers the places where unconditional stereo amp accesses are done and converts to the conditional accesses. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=94581 Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_generic.c | 30 ++++++++++++++++++++++-------- 1 file changed, 22 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index b680b4ec6331..fe18071bf93a 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -692,7 +692,23 @@ static void init_amp(struct hda_codec *codec, hda_nid_t nid, int dir, int idx) { unsigned int caps = query_amp_caps(codec, nid, dir); int val = get_amp_val_to_activate(codec, nid, dir, caps, false); - snd_hda_codec_amp_init_stereo(codec, nid, dir, idx, 0xff, val); + + if (get_wcaps(codec, nid) & AC_WCAP_STEREO) + snd_hda_codec_amp_init_stereo(codec, nid, dir, idx, 0xff, val); + else + snd_hda_codec_amp_init(codec, nid, 0, dir, idx, 0xff, val); +} + +/* update the amp, doing in stereo or mono depending on NID */ +static int update_amp(struct hda_codec *codec, hda_nid_t nid, int dir, int idx, + unsigned int mask, unsigned int val) +{ + if (get_wcaps(codec, nid) & AC_WCAP_STEREO) + return snd_hda_codec_amp_stereo(codec, nid, dir, idx, + mask, val); + else + return snd_hda_codec_amp_update(codec, nid, 0, dir, idx, + mask, val); } /* calculate amp value mask we can modify; @@ -732,7 +748,7 @@ static void activate_amp(struct hda_codec *codec, hda_nid_t nid, int dir, return; val &= mask; - snd_hda_codec_amp_stereo(codec, nid, dir, idx, mask, val); + update_amp(codec, nid, dir, idx, mask, val); } static void activate_amp_out(struct hda_codec *codec, struct nid_path *path, @@ -4424,13 +4440,11 @@ static void mute_all_mixer_nid(struct hda_codec *codec, hda_nid_t mix) has_amp = nid_has_mute(codec, mix, HDA_INPUT); for (i = 0; i < nums; i++) { if (has_amp) - snd_hda_codec_amp_stereo(codec, mix, - HDA_INPUT, i, - 0xff, HDA_AMP_MUTE); + update_amp(codec, mix, HDA_INPUT, i, + 0xff, HDA_AMP_MUTE); else if (nid_has_volume(codec, conn[i], HDA_OUTPUT)) - snd_hda_codec_amp_stereo(codec, conn[i], - HDA_OUTPUT, 0, - 0xff, HDA_AMP_MUTE); + update_amp(codec, conn[i], HDA_OUTPUT, 0, + 0xff, HDA_AMP_MUTE); } } -- cgit v1.2.3