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/soc') 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/soc') 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/soc') 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/soc') 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/soc') 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/soc') 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/soc') 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/soc') 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/soc') 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/soc') 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/soc') 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/soc') 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/soc') 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/soc') 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/soc') 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/soc') 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/soc') 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 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/soc') 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/soc') 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/soc') 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 ce9594c6b332fd6fe464e22a83b0e6e0a287aac6 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Fri, 27 Feb 2015 14:13:02 +0100 Subject: ASoC: ak4671: Fix control-less DAPM routes Routes without a control must use NULL for the control name. The ak4671 driver uses "NULL" instead in a few places. Previous to commit 5fe5b767dc6f ("ASoC: dapm: Do not pretend to support controls for non mixer/mux widgets") the DAPM core silently ignored non-NULL controls on non-mixer and non-mux routes. But starting with that commit it will complain and not add the route breaking the ak4671 driver in the process. This patch replaces the incorrect "NULL" control name with NULL to fix the issue. Fixes: 5fe5b767dc6f ("ASoC: dapm: Do not pretend to support controls for non mixer/mux widgets") Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/ak4671.c | 44 ++++++++++++++++++++++---------------------- 1 file changed, 22 insertions(+), 22 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/ak4671.c b/sound/soc/codecs/ak4671.c index 632e89f793a7..2a58b1dccd2f 100644 --- a/sound/soc/codecs/ak4671.c +++ b/sound/soc/codecs/ak4671.c @@ -343,25 +343,25 @@ static const struct snd_soc_dapm_widget ak4671_dapm_widgets[] = { }; static const struct snd_soc_dapm_route ak4671_intercon[] = { - {"DAC Left", "NULL", "PMPLL"}, - {"DAC Right", "NULL", "PMPLL"}, - {"ADC Left", "NULL", "PMPLL"}, - {"ADC Right", "NULL", "PMPLL"}, + {"DAC Left", NULL, "PMPLL"}, + {"DAC Right", NULL, "PMPLL"}, + {"ADC Left", NULL, "PMPLL"}, + {"ADC Right", NULL, "PMPLL"}, /* Outputs */ - {"LOUT1", "NULL", "LOUT1 Mixer"}, - {"ROUT1", "NULL", "ROUT1 Mixer"}, - {"LOUT2", "NULL", "LOUT2 Mix Amp"}, - {"ROUT2", "NULL", "ROUT2 Mix Amp"}, - {"LOUT3", "NULL", "LOUT3 Mixer"}, - {"ROUT3", "NULL", "ROUT3 Mixer"}, + {"LOUT1", NULL, "LOUT1 Mixer"}, + {"ROUT1", NULL, "ROUT1 Mixer"}, + {"LOUT2", NULL, "LOUT2 Mix Amp"}, + {"ROUT2", NULL, "ROUT2 Mix Amp"}, + {"LOUT3", NULL, "LOUT3 Mixer"}, + {"ROUT3", NULL, "ROUT3 Mixer"}, {"LOUT1 Mixer", "DACL", "DAC Left"}, {"ROUT1 Mixer", "DACR", "DAC Right"}, {"LOUT2 Mixer", "DACHL", "DAC Left"}, {"ROUT2 Mixer", "DACHR", "DAC Right"}, - {"LOUT2 Mix Amp", "NULL", "LOUT2 Mixer"}, - {"ROUT2 Mix Amp", "NULL", "ROUT2 Mixer"}, + {"LOUT2 Mix Amp", NULL, "LOUT2 Mixer"}, + {"ROUT2 Mix Amp", NULL, "ROUT2 Mixer"}, {"LOUT3 Mixer", "DACSL", "DAC Left"}, {"ROUT3 Mixer", "DACSR", "DAC Right"}, @@ -381,18 +381,18 @@ static const struct snd_soc_dapm_route ak4671_intercon[] = { {"LIN2", NULL, "Mic Bias"}, {"RIN2", NULL, "Mic Bias"}, - {"ADC Left", "NULL", "LIN MUX"}, - {"ADC Right", "NULL", "RIN MUX"}, + {"ADC Left", NULL, "LIN MUX"}, + {"ADC Right", NULL, "RIN MUX"}, /* Analog Loops */ - {"LIN1 Mixing Circuit", "NULL", "LIN1"}, - {"RIN1 Mixing Circuit", "NULL", "RIN1"}, - {"LIN2 Mixing Circuit", "NULL", "LIN2"}, - {"RIN2 Mixing Circuit", "NULL", "RIN2"}, - {"LIN3 Mixing Circuit", "NULL", "LIN3"}, - {"RIN3 Mixing Circuit", "NULL", "RIN3"}, - {"LIN4 Mixing Circuit", "NULL", "LIN4"}, - {"RIN4 Mixing Circuit", "NULL", "RIN4"}, + {"LIN1 Mixing Circuit", NULL, "LIN1"}, + {"RIN1 Mixing Circuit", NULL, "RIN1"}, + {"LIN2 Mixing Circuit", NULL, "LIN2"}, + {"RIN2 Mixing Circuit", NULL, "RIN2"}, + {"LIN3 Mixing Circuit", NULL, "LIN3"}, + {"RIN3 Mixing Circuit", NULL, "RIN3"}, + {"LIN4 Mixing Circuit", NULL, "LIN4"}, + {"RIN4 Mixing Circuit", NULL, "RIN4"}, {"LOUT1 Mixer", "LINL1", "LIN1 Mixing Circuit"}, {"ROUT1 Mixer", "RINR1", "RIN1 Mixing Circuit"}, -- cgit v1.2.3 From 8e6a75c102f8e232b599a06e06731d8c5d5f2c5d Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Fri, 27 Feb 2015 14:13:03 +0100 Subject: ASoC: da732x: Fix control-less DAPM routes Routes without a control must use NULL for the control name. The da732x driver uses "NULL" instead in a few places. Previous to commit 5fe5b767dc6f ("ASoC: dapm: Do not pretend to support controls for non mixer/mux widgets") the DAPM core silently ignored non-NULL controls on non-mixer and non-mux routes. But starting with that commit it will complain and not add the route breaking the da732x driver in the process. This patch replaces the incorrect "NULL" control name with NULL to fix the issue. Fixes: 5fe5b767dc6f ("ASoC: dapm: Do not pretend to support controls for non mixer/mux widgets") Signed-off-by: Lars-Peter Clausen Acked-by: Adam Thomson Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/da732x.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/da732x.c b/sound/soc/codecs/da732x.c index ffe96175a8a5..911c26c705fc 100644 --- a/sound/soc/codecs/da732x.c +++ b/sound/soc/codecs/da732x.c @@ -876,11 +876,11 @@ static const struct snd_soc_dapm_widget da732x_dapm_widgets[] = { static const struct snd_soc_dapm_route da732x_dapm_routes[] = { /* Inputs */ - {"AUX1L PGA", "NULL", "AUX1L"}, - {"AUX1R PGA", "NULL", "AUX1R"}, + {"AUX1L PGA", NULL, "AUX1L"}, + {"AUX1R PGA", NULL, "AUX1R"}, {"MIC1 PGA", NULL, "MIC1"}, - {"MIC2 PGA", "NULL", "MIC2"}, - {"MIC3 PGA", "NULL", "MIC3"}, + {"MIC2 PGA", NULL, "MIC2"}, + {"MIC3 PGA", NULL, "MIC3"}, /* Capture Path */ {"ADC1 Left MUX", "MIC1", "MIC1 PGA"}, -- cgit v1.2.3 From cdd3d2a93f08823a0b9802147dc28c99029dfdfd Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Fri, 27 Feb 2015 14:13:04 +0100 Subject: ASoC: sn95031: Fix control-less DAPM routes Routes without a control must use NULL for the control name. The sn95031 driver uses "NULL" instead in a few places. Previous to commit 5fe5b767dc6f ("ASoC: dapm: Do not pretend to support controls for non mixer/mux widgets") the DAPM core silently ignored non-NULL controls on non-mixer and non-mux routes. But starting with that commit it will complain and not add the route breaking the sn95031 driver in the process. This patch replaces the incorrect "NULL" control name with NULL to fix the issue. Fixes: 5fe5b767dc6f ("ASoC: dapm: Do not pretend to support controls for non mixer/mux widgets") Signed-off-by: Lars-Peter Clausen Acked-by: Vinod Koul Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/sn95031.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c index 47b257e41809..82095d6cd070 100644 --- a/sound/soc/codecs/sn95031.c +++ b/sound/soc/codecs/sn95031.c @@ -538,8 +538,8 @@ static const struct snd_soc_dapm_route sn95031_audio_map[] = { /* speaker map */ { "IHFOUTL", NULL, "Speaker Rail"}, { "IHFOUTR", NULL, "Speaker Rail"}, - { "IHFOUTL", "NULL", "Speaker Left Playback"}, - { "IHFOUTR", "NULL", "Speaker Right Playback"}, + { "IHFOUTL", NULL, "Speaker Left Playback"}, + { "IHFOUTR", NULL, "Speaker Right Playback"}, { "Speaker Left Playback", NULL, "Speaker Left Filter"}, { "Speaker Right Playback", NULL, "Speaker Right Filter"}, { "Speaker Left Filter", NULL, "IHFDAC Left"}, -- cgit v1.2.3 From 90aff15b3e0858eaefdcd390e64849542845d489 Mon Sep 17 00:00:00 2001 From: "Maciej S. Szmigiero" Date: Wed, 4 Mar 2015 22:48:30 +0100 Subject: fsl_ssi: fix of_property_read_u32_array return value check of_property_read_u32_array returns 0 on success, so the return value shouldn't be inverted twice, first on assignment then in condition expression. Signed-off-by: Maciej Szmigiero Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_ssi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index d7365c5d7ec0..134388f7d1b8 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1227,7 +1227,7 @@ static int fsl_ssi_imx_probe(struct platform_device *pdev, ssi_private->dma_params_tx.addr = ssi_private->ssi_phys + CCSR_SSI_STX0; ssi_private->dma_params_rx.addr = ssi_private->ssi_phys + CCSR_SSI_SRX0; - ret = !of_property_read_u32_array(np, "dmas", dmas, 4); + ret = of_property_read_u32_array(np, "dmas", dmas, 4); if (ssi_private->use_dma && !ret && dmas[2] == IMX_DMATYPE_SSI_DUAL) { ssi_private->use_dual_fifo = true; /* When using dual fifo mode, we need to keep watermark -- cgit v1.2.3 From 6c8ca30eec7b6f8eb09c957e8dcced89e5f100c7 Mon Sep 17 00:00:00 2001 From: Nicolin Chen Date: Wed, 4 Mar 2015 21:05:04 -0800 Subject: ASoC: fsl_ssi: Don't try to round-up for PM divisor calculation According to i.MX6 Series Reference Manual, the formula to calculate the sys clock is sysclk rate = bclk rate * (div2 + 1) * (7 * psr + 1) * (pm + 1) * 2 Commit aafa85e71a75 ("ASoC: fsl_ssi: Add DAI master mode support for SSI on i.MX series") added the divisor calculation which relies on the clk_round_rate(). However, at that time, clk_round_rate() didn't provide closest clock rates for some cases because it might not use a correct rounding policy. So using the original formula (pm + 1) for PM divisor was not able to give us a desired clock rate. And then we used (pm + 2) to do the trick. However, the clk-divider driver has been refined a lot since commit b11d282dbea2 ("clk: divider: fix rate calculation for fractional rates") Now using (pm + 2) trick would result an incorrect clock rate. So this patch fixes the problem by removing the useless trick. Reported-by: Stephane Cerveau Signed-off-by: Nicolin Chen Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_ssi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 134388f7d1b8..7eebc0889c9d 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -603,7 +603,7 @@ static int fsl_ssi_set_bclk(struct snd_pcm_substream *substream, factor = (div2 + 1) * (7 * psr + 1) * 2; for (i = 0; i < 255; i++) { - tmprate = freq * factor * (i + 2); + tmprate = freq * factor * (i + 1); if (baudclk_is_used) clkrate = clk_get_rate(ssi_private->baudclk); -- cgit v1.2.3 From c7d910b87d3c8e9fcf4077089ca4327c12eee099 Mon Sep 17 00:00:00 2001 From: Eric Nelson Date: Fri, 27 Feb 2015 08:06:45 -0700 Subject: ASoC: sgtl5000: remove useless register write clearing CHRGPUMP_POWERUP The SGTL5000_CHIP_ANA_POWER register is cached. Update the cached value instead of writing it directly. Patch inspired by Russell King's more colorful remarks in this patch: https://github.com/SolidRun/linux-imx6-3.14/commit/dd4bf6a Signed-off-by: Eric Nelson Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/sgtl5000.c | 8 +------- 1 file changed, 1 insertion(+), 7 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index e182e6569bbd..3593a1496056 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -1151,13 +1151,7 @@ static int sgtl5000_set_power_regs(struct snd_soc_codec *codec) /* Enable VDDC charge pump */ ana_pwr |= SGTL5000_VDDC_CHRGPMP_POWERUP; } else if (vddio >= 3100 && vdda >= 3100) { - /* - * if vddio and vddd > 3.1v, - * charge pump should be clean before set ana_pwr - */ - snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER, - SGTL5000_VDDC_CHRGPMP_POWERUP, 0); - + ana_pwr &= ~SGTL5000_VDDC_CHRGPMP_POWERUP; /* VDDC use VDDIO rail */ lreg_ctrl |= SGTL5000_VDDC_ASSN_OVRD; lreg_ctrl |= SGTL5000_VDDC_MAN_ASSN_VDDIO << -- cgit v1.2.3 From a4ee556137a5bb4b542c5023e6fead4b7cf33495 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Fri, 6 Mar 2015 10:12:58 +0800 Subject: ASoC: rt286: Change the DMI mapping for Dino The board ID will be changed between revisions. So, it is better to map it by project name. Signed-off-by: Bard Liao Signed-off-by: Mark Brown --- sound/soc/codecs/rt286.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index f374840a5a7c..9b541e52da8c 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -1198,7 +1198,7 @@ static struct dmi_system_id dmi_dell_dino[] = { .ident = "Dell Dino", .matches = { DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc."), - DMI_MATCH(DMI_BOARD_NAME, "0144P8") + DMI_MATCH(DMI_PRODUCT_NAME, "XPS 13 9343") } }, { } -- cgit v1.2.3 From 3fe0607a04ed7deea7c048052fd63b8670e7a176 Mon Sep 17 00:00:00 2001 From: "Lu, Han" Date: Wed, 25 Feb 2015 08:26:21 +0800 Subject: ASoC: Intel: remove conflicts when load/unload multiple firmware images Details: 1. Unload all modules on fw_list of dsp when suspend, and reload all modules on fw_list when resume. 2. A DSP expects only one scratch, but hsw_parse_fw_image() allocates scratch blocks for each firmware image it parses. Move the allocate function sst_block_alloc_scratch() out of hsw_parse_fw_image() to make sure a scratch be allocated only after all firmware images be parsed. Signed-off-by: Lu, Han Signed-off-by: Mark Brown --- sound/soc/intel/sst-haswell-dsp.c | 3 --- sound/soc/intel/sst-haswell-ipc.c | 32 ++++++++++++++++++++++++-------- 2 files changed, 24 insertions(+), 11 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/intel/sst-haswell-dsp.c b/sound/soc/intel/sst-haswell-dsp.c index 57039b00efc2..f6e1e6b2b18e 100644 --- a/sound/soc/intel/sst-haswell-dsp.c +++ b/sound/soc/intel/sst-haswell-dsp.c @@ -207,9 +207,6 @@ static int hsw_parse_fw_image(struct sst_fw *sst_fw) module = (void *)module + sizeof(*module) + module->mod_size; } - /* allocate scratch mem regions */ - sst_block_alloc_scratch(dsp); - return 0; } diff --git a/sound/soc/intel/sst-haswell-ipc.c b/sound/soc/intel/sst-haswell-ipc.c index 8156cc1accb7..6c7052a40e36 100644 --- a/sound/soc/intel/sst-haswell-ipc.c +++ b/sound/soc/intel/sst-haswell-ipc.c @@ -1870,6 +1870,7 @@ static void sst_hsw_drop_all(struct sst_hsw *hsw) int sst_hsw_dsp_load(struct sst_hsw *hsw) { struct sst_dsp *dsp = hsw->dsp; + struct sst_fw *sst_fw, *t; int ret; dev_dbg(hsw->dev, "loading audio DSP...."); @@ -1886,12 +1887,17 @@ int sst_hsw_dsp_load(struct sst_hsw *hsw) return ret; } - ret = sst_fw_reload(hsw->sst_fw); - if (ret < 0) { - dev_err(hsw->dev, "error: SST FW reload failed\n"); - sst_dsp_dma_put_channel(dsp); - return -ENOMEM; + list_for_each_entry_safe_reverse(sst_fw, t, &dsp->fw_list, list) { + ret = sst_fw_reload(sst_fw); + if (ret < 0) { + dev_err(hsw->dev, "error: SST FW reload failed\n"); + sst_dsp_dma_put_channel(dsp); + return -ENOMEM; + } } + ret = sst_block_alloc_scratch(hsw->dsp); + if (ret < 0) + return -EINVAL; sst_dsp_dma_put_channel(dsp); return 0; @@ -1947,12 +1953,17 @@ int sst_hsw_dsp_runtime_suspend(struct sst_hsw *hsw) int sst_hsw_dsp_runtime_sleep(struct sst_hsw *hsw) { - sst_fw_unload(hsw->sst_fw); - sst_block_free_scratch(hsw->dsp); + struct sst_fw *sst_fw, *t; + struct sst_dsp *dsp = hsw->dsp; + + list_for_each_entry_safe(sst_fw, t, &dsp->fw_list, list) { + sst_fw_unload(sst_fw); + } + sst_block_free_scratch(dsp); hsw->boot_complete = false; - sst_dsp_sleep(hsw->dsp); + sst_dsp_sleep(dsp); return 0; } @@ -2081,6 +2092,11 @@ int sst_hsw_dsp_init(struct device *dev, struct sst_pdata *pdata) goto fw_err; } + /* allocate scratch mem regions */ + ret = sst_block_alloc_scratch(hsw->dsp); + if (ret < 0) + goto boot_err; + /* wait for DSP boot completion */ sst_dsp_boot(hsw->dsp); ret = wait_event_timeout(hsw->boot_wait, hsw->boot_complete, -- cgit v1.2.3 From 34e81ab4556f3b1371763861e74e3600818924b5 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sat, 7 Mar 2015 19:34:03 +0100 Subject: ASoC: Fix component lists locking Any access to the component_list, codec_list and platform_list needs to be properly locked by the client_mutex. Otherwise undefined behavior can occur if the list is modified in one thread and concurrently accessed from another thread. This patch adds the missing locking to the debugfs file handlers that display the registered components, as well as the various components unregister functions. Furthermore the client_lock is now held for the whole snd_soc_instantiate_card() sequence to make sure that component removal does not race against the card registration. Reported-by: Takashi Iwai Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 41 ++++++++++++++++++++++++++++++----------- 1 file changed, 30 insertions(+), 11 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 30579ca5bacb..e5c990889dcc 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -347,6 +347,8 @@ static ssize_t codec_list_read_file(struct file *file, char __user *user_buf, if (!buf) return -ENOMEM; + mutex_lock(&client_mutex); + list_for_each_entry(codec, &codec_list, list) { len = snprintf(buf + ret, PAGE_SIZE - ret, "%s\n", codec->component.name); @@ -358,6 +360,8 @@ static ssize_t codec_list_read_file(struct file *file, char __user *user_buf, } } + mutex_unlock(&client_mutex); + if (ret >= 0) ret = simple_read_from_buffer(user_buf, count, ppos, buf, ret); @@ -382,6 +386,8 @@ static ssize_t dai_list_read_file(struct file *file, char __user *user_buf, if (!buf) return -ENOMEM; + mutex_lock(&client_mutex); + list_for_each_entry(component, &component_list, list) { list_for_each_entry(dai, &component->dai_list, list) { len = snprintf(buf + ret, PAGE_SIZE - ret, "%s\n", @@ -395,6 +401,8 @@ static ssize_t dai_list_read_file(struct file *file, char __user *user_buf, } } + mutex_unlock(&client_mutex); + ret = simple_read_from_buffer(user_buf, count, ppos, buf, ret); kfree(buf); @@ -418,6 +426,8 @@ static ssize_t platform_list_read_file(struct file *file, if (!buf) return -ENOMEM; + mutex_lock(&client_mutex); + list_for_each_entry(platform, &platform_list, list) { len = snprintf(buf + ret, PAGE_SIZE - ret, "%s\n", platform->component.name); @@ -429,6 +439,8 @@ static ssize_t platform_list_read_file(struct file *file, } } + mutex_unlock(&client_mutex); + ret = simple_read_from_buffer(user_buf, count, ppos, buf, ret); kfree(buf); @@ -836,6 +848,8 @@ static struct snd_soc_component *soc_find_component( { struct snd_soc_component *component; + lockdep_assert_held(&client_mutex); + list_for_each_entry(component, &component_list, list) { if (of_node) { if (component->dev->of_node == of_node) @@ -854,6 +868,8 @@ static struct snd_soc_dai *snd_soc_find_dai( struct snd_soc_component *component; struct snd_soc_dai *dai; + lockdep_assert_held(&client_mutex); + /* Find CPU DAI from registered DAIs*/ list_for_each_entry(component, &component_list, list) { if (dlc->of_node && component->dev->of_node != dlc->of_node) @@ -1508,6 +1524,7 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) struct snd_soc_codec *codec; int ret, i, order; + mutex_lock(&client_mutex); mutex_lock_nested(&card->mutex, SND_SOC_CARD_CLASS_INIT); /* bind DAIs */ @@ -1662,6 +1679,7 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) card->instantiated = 1; snd_soc_dapm_sync(&card->dapm); mutex_unlock(&card->mutex); + mutex_unlock(&client_mutex); return 0; @@ -1680,6 +1698,7 @@ card_probe_error: base_error: mutex_unlock(&card->mutex); + mutex_unlock(&client_mutex); return ret; } @@ -2713,13 +2732,6 @@ static void snd_soc_component_del_unlocked(struct snd_soc_component *component) list_del(&component->list); } -static void snd_soc_component_del(struct snd_soc_component *component) -{ - mutex_lock(&client_mutex); - snd_soc_component_del_unlocked(component); - mutex_unlock(&client_mutex); -} - int snd_soc_register_component(struct device *dev, const struct snd_soc_component_driver *cmpnt_drv, struct snd_soc_dai_driver *dai_drv, @@ -2767,14 +2779,17 @@ void snd_soc_unregister_component(struct device *dev) { struct snd_soc_component *cmpnt; + mutex_lock(&client_mutex); list_for_each_entry(cmpnt, &component_list, list) { if (dev == cmpnt->dev && cmpnt->registered_as_component) goto found; } + mutex_unlock(&client_mutex); return; found: - snd_soc_component_del(cmpnt); + snd_soc_component_del_unlocked(cmpnt); + mutex_unlock(&client_mutex); snd_soc_component_cleanup(cmpnt); kfree(cmpnt); } @@ -2882,10 +2897,14 @@ struct snd_soc_platform *snd_soc_lookup_platform(struct device *dev) { struct snd_soc_platform *platform; + mutex_lock(&client_mutex); list_for_each_entry(platform, &platform_list, list) { - if (dev == platform->dev) + if (dev == platform->dev) { + mutex_unlock(&client_mutex); return platform; + } } + mutex_unlock(&client_mutex); return NULL; } @@ -3090,15 +3109,15 @@ void snd_soc_unregister_codec(struct device *dev) { struct snd_soc_codec *codec; + mutex_lock(&client_mutex); list_for_each_entry(codec, &codec_list, list) { if (dev == codec->dev) goto found; } + mutex_unlock(&client_mutex); return; found: - - mutex_lock(&client_mutex); list_del(&codec->list); snd_soc_component_del_unlocked(&codec->component); mutex_unlock(&client_mutex); -- cgit v1.2.3 From 2bf4c1d483d911cda5dd385527194d23e5cea73d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 10 Mar 2015 12:39:03 +0100 Subject: ASoC: adav80x: Fix wrong value references for boolean kctl The correct values referred by a boolean control are value.integer.value[], not value.enumerated.item[]. The former is long while the latter is int, so it's even incompatible on 64bit architectures. Signed-off-by: Takashi Iwai Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown Cc: --- sound/soc/codecs/adav80x.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/adav80x.c b/sound/soc/codecs/adav80x.c index b67480f1b1aa..4373ada95648 100644 --- a/sound/soc/codecs/adav80x.c +++ b/sound/soc/codecs/adav80x.c @@ -317,7 +317,7 @@ static int adav80x_put_deemph(struct snd_kcontrol *kcontrol, { struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); - unsigned int deemph = ucontrol->value.enumerated.item[0]; + unsigned int deemph = ucontrol->value.integer.value[0]; if (deemph > 1) return -EINVAL; @@ -333,7 +333,7 @@ static int adav80x_get_deemph(struct snd_kcontrol *kcontrol, struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); - ucontrol->value.enumerated.item[0] = adav80x->deemph; + ucontrol->value.integer.value[0] = adav80x->deemph; return 0; }; -- cgit v1.2.3 From 08641d9b7bf915144a57a736b42642e13eb1167f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 10 Mar 2015 12:39:04 +0100 Subject: ASoC: ak4641: Fix wrong value references for boolean kctl The correct values referred by a boolean control are value.integer.value[], not value.enumerated.item[]. The former is long while the latter is int, so it's even incompatible on 64bit architectures. Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown Cc: --- sound/soc/codecs/ak4641.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/ak4641.c b/sound/soc/codecs/ak4641.c index 70861c7b1631..81b54a270bd8 100644 --- a/sound/soc/codecs/ak4641.c +++ b/sound/soc/codecs/ak4641.c @@ -76,7 +76,7 @@ static int ak4641_put_deemph(struct snd_kcontrol *kcontrol, { struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); struct ak4641_priv *ak4641 = snd_soc_codec_get_drvdata(codec); - int deemph = ucontrol->value.enumerated.item[0]; + int deemph = ucontrol->value.integer.value[0]; if (deemph > 1) return -EINVAL; @@ -92,7 +92,7 @@ static int ak4641_get_deemph(struct snd_kcontrol *kcontrol, struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); struct ak4641_priv *ak4641 = snd_soc_codec_get_drvdata(codec); - ucontrol->value.enumerated.item[0] = ak4641->deemph; + ucontrol->value.integer.value[0] = ak4641->deemph; return 0; }; -- cgit v1.2.3 From e8371aa0fecb73fb8a4b2e0296b025b11e7d6229 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 10 Mar 2015 12:39:05 +0100 Subject: ASoC: cs4271: Fix wrong value references for boolean kctl The correct values referred by a boolean control are value.integer.value[], not value.enumerated.item[]. The former is long while the latter is int, so it's even incompatible on 64bit architectures. Signed-off-by: Takashi Iwai Acked-by: Paul Handrigan Signed-off-by: Mark Brown Cc: --- sound/soc/codecs/cs4271.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index 79a4efcb894c..7d3a6accaf9a 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -286,7 +286,7 @@ static int cs4271_get_deemph(struct snd_kcontrol *kcontrol, struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); - ucontrol->value.enumerated.item[0] = cs4271->deemph; + ucontrol->value.integer.value[0] = cs4271->deemph; return 0; } @@ -296,7 +296,7 @@ static int cs4271_put_deemph(struct snd_kcontrol *kcontrol, struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); - cs4271->deemph = ucontrol->value.enumerated.item[0]; + cs4271->deemph = ucontrol->value.integer.value[0]; return cs4271_set_deemph(codec); } -- cgit v1.2.3 From d223b0e7fcfecc23380e7de45eb6a0e7b328c17c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 10 Mar 2015 12:39:06 +0100 Subject: ASoC: es8238: Fix wrong value references for boolean kctl The correct values referred by a boolean control are value.integer.value[], not value.enumerated.item[]. The former is long while the latter is int, so it's even incompatible on 64bit architectures. Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown Cc: --- sound/soc/codecs/es8328.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/es8328.c b/sound/soc/codecs/es8328.c index f27325155ace..c5f35a07e8e4 100644 --- a/sound/soc/codecs/es8328.c +++ b/sound/soc/codecs/es8328.c @@ -120,7 +120,7 @@ static int es8328_get_deemph(struct snd_kcontrol *kcontrol, struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); struct es8328_priv *es8328 = snd_soc_codec_get_drvdata(codec); - ucontrol->value.enumerated.item[0] = es8328->deemph; + ucontrol->value.integer.value[0] = es8328->deemph; return 0; } @@ -129,7 +129,7 @@ static int es8328_put_deemph(struct snd_kcontrol *kcontrol, { struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); struct es8328_priv *es8328 = snd_soc_codec_get_drvdata(codec); - int deemph = ucontrol->value.enumerated.item[0]; + int deemph = ucontrol->value.integer.value[0]; int ret; if (deemph > 1) -- cgit v1.2.3 From d7f58db49d9ad92bdb12d21fdc2308b76bc2ed38 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 10 Mar 2015 12:39:07 +0100 Subject: ASoC: pcm1681: Fix wrong value references for boolean kctl The correct values referred by a boolean control are value.integer.value[], not value.enumerated.item[]. The former is long while the latter is int, so it's even incompatible on 64bit architectures. Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown Cc: --- sound/soc/codecs/pcm1681.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/pcm1681.c b/sound/soc/codecs/pcm1681.c index a722a023c262..477e13d30971 100644 --- a/sound/soc/codecs/pcm1681.c +++ b/sound/soc/codecs/pcm1681.c @@ -118,7 +118,7 @@ static int pcm1681_get_deemph(struct snd_kcontrol *kcontrol, struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); struct pcm1681_private *priv = snd_soc_codec_get_drvdata(codec); - ucontrol->value.enumerated.item[0] = priv->deemph; + ucontrol->value.integer.value[0] = priv->deemph; return 0; } @@ -129,7 +129,7 @@ static int pcm1681_put_deemph(struct snd_kcontrol *kcontrol, struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); struct pcm1681_private *priv = snd_soc_codec_get_drvdata(codec); - priv->deemph = ucontrol->value.enumerated.item[0]; + priv->deemph = ucontrol->value.integer.value[0]; return pcm1681_set_deemph(codec); } -- cgit v1.2.3 From 4c523ef61160b7d478371ddc9f48c8ce0a00d675 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 10 Mar 2015 12:39:08 +0100 Subject: ASoC: tas5086: Fix wrong value references for boolean kctl The correct values referred by a boolean control are value.integer.value[], not value.enumerated.item[]. The former is long while the latter is int, so it's even incompatible on 64bit architectures. Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown Cc: --- sound/soc/codecs/tas5086.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/tas5086.c b/sound/soc/codecs/tas5086.c index 249ef5c4c762..32942bed34b1 100644 --- a/sound/soc/codecs/tas5086.c +++ b/sound/soc/codecs/tas5086.c @@ -281,7 +281,7 @@ static int tas5086_get_deemph(struct snd_kcontrol *kcontrol, struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec); - ucontrol->value.enumerated.item[0] = priv->deemph; + ucontrol->value.integer.value[0] = priv->deemph; return 0; } @@ -292,7 +292,7 @@ static int tas5086_put_deemph(struct snd_kcontrol *kcontrol, struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec); - priv->deemph = ucontrol->value.enumerated.item[0]; + priv->deemph = ucontrol->value.integer.value[0]; return tas5086_set_deemph(codec); } -- cgit v1.2.3 From 00a14c2968e3d55817e0fa35c78106ca840537bf Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 10 Mar 2015 12:39:09 +0100 Subject: ASoC: wm2000: Fix wrong value references for boolean kctl The correct values referred by a boolean control are value.integer.value[], not value.enumerated.item[]. The former is long while the latter is int, so it's even incompatible on 64bit architectures. Signed-off-by: Takashi Iwai Acked-by: Charles Keepax Signed-off-by: Mark Brown Cc: --- sound/soc/codecs/wm2000.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c index 8d9de49a5052..21d5402e343f 100644 --- a/sound/soc/codecs/wm2000.c +++ b/sound/soc/codecs/wm2000.c @@ -610,7 +610,7 @@ static int wm2000_anc_mode_get(struct snd_kcontrol *kcontrol, struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); struct wm2000_priv *wm2000 = dev_get_drvdata(codec->dev); - ucontrol->value.enumerated.item[0] = wm2000->anc_active; + ucontrol->value.integer.value[0] = wm2000->anc_active; return 0; } @@ -620,7 +620,7 @@ static int wm2000_anc_mode_put(struct snd_kcontrol *kcontrol, { struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); struct wm2000_priv *wm2000 = dev_get_drvdata(codec->dev); - int anc_active = ucontrol->value.enumerated.item[0]; + int anc_active = ucontrol->value.integer.value[0]; int ret; if (anc_active > 1) @@ -643,7 +643,7 @@ static int wm2000_speaker_get(struct snd_kcontrol *kcontrol, struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); struct wm2000_priv *wm2000 = dev_get_drvdata(codec->dev); - ucontrol->value.enumerated.item[0] = wm2000->spk_ena; + ucontrol->value.integer.value[0] = wm2000->spk_ena; return 0; } @@ -653,7 +653,7 @@ static int wm2000_speaker_put(struct snd_kcontrol *kcontrol, { struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); struct wm2000_priv *wm2000 = dev_get_drvdata(codec->dev); - int val = ucontrol->value.enumerated.item[0]; + int val = ucontrol->value.integer.value[0]; int ret; if (val > 1) -- cgit v1.2.3 From bd14016fbf31aa199026f1e2358eab695f374eb1 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 10 Mar 2015 12:39:10 +0100 Subject: ASoC: wm8731: Fix wrong value references for boolean kctl The correct values referred by a boolean control are value.integer.value[], not value.enumerated.item[]. The former is long while the latter is int, so it's even incompatible on 64bit architectures. Signed-off-by: Takashi Iwai Acked-by: Charles Keepax Signed-off-by: Mark Brown Cc: --- sound/soc/codecs/wm8731.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 098c143f44d6..c6d10533e2bd 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -125,7 +125,7 @@ static int wm8731_get_deemph(struct snd_kcontrol *kcontrol, struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); struct wm8731_priv *wm8731 = snd_soc_codec_get_drvdata(codec); - ucontrol->value.enumerated.item[0] = wm8731->deemph; + ucontrol->value.integer.value[0] = wm8731->deemph; return 0; } @@ -135,7 +135,7 @@ static int wm8731_put_deemph(struct snd_kcontrol *kcontrol, { struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); struct wm8731_priv *wm8731 = snd_soc_codec_get_drvdata(codec); - int deemph = ucontrol->value.enumerated.item[0]; + int deemph = ucontrol->value.integer.value[0]; int ret = 0; if (deemph > 1) -- cgit v1.2.3 From 24cc883c1fd16df34211ae41624aa6d3cd906693 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 10 Mar 2015 12:39:11 +0100 Subject: ASoC: wm8903: Fix wrong value references for boolean kctl The correct values referred by a boolean control are value.integer.value[], not value.enumerated.item[]. The former is long while the latter is int, so it's even incompatible on 64bit architectures. Signed-off-by: Takashi Iwai Acked-by: Charles Keepax Signed-off-by: Mark Brown Cc: --- sound/soc/codecs/wm8903.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index dde462c082be..04b04f8e147c 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -442,7 +442,7 @@ static int wm8903_get_deemph(struct snd_kcontrol *kcontrol, struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec); - ucontrol->value.enumerated.item[0] = wm8903->deemph; + ucontrol->value.integer.value[0] = wm8903->deemph; return 0; } @@ -452,7 +452,7 @@ static int wm8903_put_deemph(struct snd_kcontrol *kcontrol, { struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec); - int deemph = ucontrol->value.enumerated.item[0]; + int deemph = ucontrol->value.integer.value[0]; int ret = 0; if (deemph > 1) -- cgit v1.2.3 From eaddf6fd959074f6a6e71deffe079c71eef35da6 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 10 Mar 2015 12:39:12 +0100 Subject: ASoC: wm8904: Fix wrong value references for boolean kctl The correct values referred by a boolean control are value.integer.value[], not value.enumerated.item[]. The former is long while the latter is int, so it's even incompatible on 64bit architectures. Signed-off-by: Takashi Iwai Acked-by: Charles Keepax Signed-off-by: Mark Brown Cc: --- sound/soc/codecs/wm8904.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index d3b3f57668cc..215e93c1ddf0 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -525,7 +525,7 @@ static int wm8904_get_deemph(struct snd_kcontrol *kcontrol, struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); struct wm8904_priv *wm8904 = snd_soc_codec_get_drvdata(codec); - ucontrol->value.enumerated.item[0] = wm8904->deemph; + ucontrol->value.integer.value[0] = wm8904->deemph; return 0; } @@ -534,7 +534,7 @@ static int wm8904_put_deemph(struct snd_kcontrol *kcontrol, { struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); struct wm8904_priv *wm8904 = snd_soc_codec_get_drvdata(codec); - int deemph = ucontrol->value.enumerated.item[0]; + int deemph = ucontrol->value.integer.value[0]; if (deemph > 1) return -EINVAL; -- cgit v1.2.3 From 07892b10356f17717abdc578acbef72db86c880e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 10 Mar 2015 12:39:13 +0100 Subject: ASoC: wm8955: Fix wrong value references for boolean kctl The correct values referred by a boolean control are value.integer.value[], not value.enumerated.item[]. The former is long while the latter is int, so it's even incompatible on 64bit architectures. Signed-off-by: Takashi Iwai Acked-by: Charles Keepax Signed-off-by: Mark Brown Cc: --- sound/soc/codecs/wm8955.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8955.c b/sound/soc/codecs/wm8955.c index 1ab2d462afad..00bec915d652 100644 --- a/sound/soc/codecs/wm8955.c +++ b/sound/soc/codecs/wm8955.c @@ -393,7 +393,7 @@ static int wm8955_get_deemph(struct snd_kcontrol *kcontrol, struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); struct wm8955_priv *wm8955 = snd_soc_codec_get_drvdata(codec); - ucontrol->value.enumerated.item[0] = wm8955->deemph; + ucontrol->value.integer.value[0] = wm8955->deemph; return 0; } @@ -402,7 +402,7 @@ static int wm8955_put_deemph(struct snd_kcontrol *kcontrol, { struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); struct wm8955_priv *wm8955 = snd_soc_codec_get_drvdata(codec); - int deemph = ucontrol->value.enumerated.item[0]; + int deemph = ucontrol->value.integer.value[0]; if (deemph > 1) return -EINVAL; -- cgit v1.2.3 From b4a18c8b1af15ebfa9054a3d2aef7b0a7e6f2a05 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 10 Mar 2015 12:39:14 +0100 Subject: ASoC: wm8960: Fix wrong value references for boolean kctl The correct values referred by a boolean control are value.integer.value[], not value.enumerated.item[]. The former is long while the latter is int, so it's even incompatible on 64bit architectures. Signed-off-by: Takashi Iwai Acked-by: Charles Keepax Signed-off-by: Mark Brown Cc: --- sound/soc/codecs/wm8960.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index cf8fecf97f2c..3035d9856415 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -184,7 +184,7 @@ static int wm8960_get_deemph(struct snd_kcontrol *kcontrol, struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); - ucontrol->value.enumerated.item[0] = wm8960->deemph; + ucontrol->value.integer.value[0] = wm8960->deemph; return 0; } @@ -193,7 +193,7 @@ static int wm8960_put_deemph(struct snd_kcontrol *kcontrol, { struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); - int deemph = ucontrol->value.enumerated.item[0]; + int deemph = ucontrol->value.integer.value[0]; if (deemph > 1) return -EINVAL; -- cgit v1.2.3 From 4b0b669b86a963f71feaa1a694e881832fdf4f86 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 10 Mar 2015 12:39:15 +0100 Subject: ASoC: wm9712: Fix wrong value references for boolean kctl The correct values referred by a boolean control are value.integer.value[], not value.enumerated.item[]. The former is long while the latter is int, so it's even incompatible on 64bit architectures. Signed-off-by: Takashi Iwai Acked-by: Charles Keepax Signed-off-by: Mark Brown Cc: --- sound/soc/codecs/wm9712.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index 9517571e820d..98c9525bd751 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -180,7 +180,7 @@ static int wm9712_hp_mixer_put(struct snd_kcontrol *kcontrol, struct snd_soc_dapm_context *dapm = snd_soc_dapm_kcontrol_dapm(kcontrol); struct snd_soc_codec *codec = snd_soc_dapm_to_codec(dapm); struct wm9712_priv *wm9712 = snd_soc_codec_get_drvdata(codec); - unsigned int val = ucontrol->value.enumerated.item[0]; + unsigned int val = ucontrol->value.integer.value[0]; struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; unsigned int mixer, mask, shift, old; @@ -193,7 +193,7 @@ static int wm9712_hp_mixer_put(struct snd_kcontrol *kcontrol, mutex_lock(&wm9712->lock); old = wm9712->hp_mixer[mixer]; - if (ucontrol->value.enumerated.item[0]) + if (ucontrol->value.integer.value[0]) wm9712->hp_mixer[mixer] |= mask; else wm9712->hp_mixer[mixer] &= ~mask; @@ -231,7 +231,7 @@ static int wm9712_hp_mixer_get(struct snd_kcontrol *kcontrol, mixer = mc->shift >> 8; shift = mc->shift & 0xff; - ucontrol->value.enumerated.item[0] = + ucontrol->value.integer.value[0] = (wm9712->hp_mixer[mixer] >> shift) & 1; return 0; -- cgit v1.2.3 From 87a8b286e2f63c048a586dc677140d4a5b5808aa Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 10 Mar 2015 12:39:16 +0100 Subject: ASoC: wm9713: Fix wrong value references for boolean kctl The correct values referred by a boolean control are value.integer.value[], not value.enumerated.item[]. The former is long while the latter is int, so it's even incompatible on 64bit architectures. Signed-off-by: Takashi Iwai Acked-by: Charles Keepax Signed-off-by: Mark Brown Cc: --- sound/soc/codecs/wm9713.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index 68222917b396..79552953e1bd 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -255,7 +255,7 @@ static int wm9713_hp_mixer_put(struct snd_kcontrol *kcontrol, struct snd_soc_dapm_context *dapm = snd_soc_dapm_kcontrol_dapm(kcontrol); struct snd_soc_codec *codec = snd_soc_dapm_to_codec(dapm); struct wm9713_priv *wm9713 = snd_soc_codec_get_drvdata(codec); - unsigned int val = ucontrol->value.enumerated.item[0]; + unsigned int val = ucontrol->value.integer.value[0]; struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; unsigned int mixer, mask, shift, old; @@ -268,7 +268,7 @@ static int wm9713_hp_mixer_put(struct snd_kcontrol *kcontrol, mutex_lock(&wm9713->lock); old = wm9713->hp_mixer[mixer]; - if (ucontrol->value.enumerated.item[0]) + if (ucontrol->value.integer.value[0]) wm9713->hp_mixer[mixer] |= mask; else wm9713->hp_mixer[mixer] &= ~mask; @@ -306,7 +306,7 @@ static int wm9713_hp_mixer_get(struct snd_kcontrol *kcontrol, mixer = mc->shift >> 8; shift = mc->shift & 0xff; - ucontrol->value.enumerated.item[0] = + ucontrol->value.integer.value[0] = (wm9713->hp_mixer[mixer] >> shift) & 1; return 0; -- 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/soc') 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/soc') 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