From 88b5bdfda886e9774b03f02ffe4295be124124f6 Mon Sep 17 00:00:00 2001 From: Wei Yongjun Date: Fri, 27 Sep 2013 16:15:54 +0800 Subject: ASoC: wm8993: drop regulator_bulk_free of devm_ allocated data It's not necessary to free regulator consumers allocated with devm_regulator_bulk_get. Signed-off-by: Wei Yongjun Signed-off-by: Mark Brown --- sound/soc/codecs/wm8993.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index 433d59a0f3ef..2ee23a39622c 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -1562,7 +1562,6 @@ static int wm8993_remove(struct snd_soc_codec *codec) struct wm8993_priv *wm8993 = snd_soc_codec_get_drvdata(codec); wm8993_set_bias_level(codec, SND_SOC_BIAS_OFF); - regulator_bulk_free(ARRAY_SIZE(wm8993->supplies), wm8993->supplies); return 0; } -- cgit v1.2.1 From eaff64705dd5557f8e455563da3d9cbdbdec0204 Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Thu, 30 Jan 2014 11:58:27 +0530 Subject: ASoC: samsung: Remove invalid dependencies These symbols got eliminated when non-DT support for Exynos was removed. Remove them. Signed-off-by: Sachin Kamat Signed-off-by: Mark Brown --- sound/soc/samsung/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 454f41cfc828..33fd3724557b 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -145,7 +145,7 @@ config SND_SOC_SAMSUNG_RX1950_UDA1380 config SND_SOC_SAMSUNG_SMDK_WM9713 tristate "SoC AC97 Audio support for SMDK with WM9713" - depends on SND_SOC_SAMSUNG && (MACH_SMDK6410 || MACH_SMDKC100 || MACH_SMDKV210 || MACH_SMDKC110 || MACH_SMDKV310 || MACH_SMDKC210) + depends on SND_SOC_SAMSUNG && (MACH_SMDK6410 || MACH_SMDKC100 || MACH_SMDKV210 || MACH_SMDKC110) select SND_SOC_WM9713 select SND_SAMSUNG_AC97 help -- cgit v1.2.1 From 25e7e34805a39a1b05b659acbed878b3187a2707 Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Thu, 30 Jan 2014 11:58:28 +0530 Subject: ASoC: samsung: Fix trivial typo Changed Sat -> Say. Signed-off-by: Sachin Kamat Signed-off-by: Mark Brown --- sound/soc/samsung/Kconfig | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 33fd3724557b..350757400391 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -59,7 +59,7 @@ config SND_SOC_SAMSUNG_JIVE_WM8750 select SND_SOC_WM8750 select SND_S3C2412_SOC_I2S help - Sat Y if you want to add support for SoC audio on the Jive. + Say Y if you want to add support for SoC audio on the Jive. config SND_SOC_SAMSUNG_SMDK_WM8580 tristate "SoC I2S Audio support for WM8580 on SMDK" @@ -149,7 +149,7 @@ config SND_SOC_SAMSUNG_SMDK_WM9713 select SND_SOC_WM9713 select SND_SAMSUNG_AC97 help - Sat Y if you want to add support for SoC audio on the SMDK. + Say Y if you want to add support for SoC audio on the SMDK. config SND_SOC_SMARTQ tristate "SoC I2S Audio support for SmartQ board" -- cgit v1.2.1 From 662ffae9ed036bc82ff74c26dc731e2815431fcc Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 30 Jan 2014 15:15:22 +0200 Subject: ASoC: davinci-mcasp: Harmonize the sub hw_params function names Instead of davinci_hw_common_param - for common, I2S/DIT mode settings davinci_hw_dit_param - for DIT protocol configuration davinci_hw_param - for I2S (and compatible protocols) Use the following names: mcasp_common_hw_param, mcasp_dit_hw_param and mcasp_i2s_hw_param. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-mcasp.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index b7858bfa0295..9ec456aaf80b 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -448,7 +448,7 @@ static int davinci_config_channel_size(struct davinci_mcasp *mcasp, return 0; } -static int davinci_hw_common_param(struct davinci_mcasp *mcasp, int stream, +static int mcasp_common_hw_param(struct davinci_mcasp *mcasp, int stream, int channels) { int i; @@ -524,7 +524,7 @@ static int davinci_hw_common_param(struct davinci_mcasp *mcasp, int stream, return 0; } -static void davinci_hw_param(struct davinci_mcasp *mcasp, int stream) +static void mcasp_i2s_hw_param(struct davinci_mcasp *mcasp, int stream) { int i, active_slots; u32 mask = 0; @@ -567,7 +567,7 @@ static void davinci_hw_param(struct davinci_mcasp *mcasp, int stream) } /* S/PDIF */ -static void davinci_hw_dit_param(struct davinci_mcasp *mcasp) +static void mcasp_dit_hw_param(struct davinci_mcasp *mcasp) { /* Set the TX format : 24 bit right rotation, 32 bit slot, Pad 0 and LSB first */ @@ -611,7 +611,7 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, active_serializers = (channels + slots - 1) / slots; - if (davinci_hw_common_param(mcasp, substream->stream, channels) == -EINVAL) + if (mcasp_common_hw_param(mcasp, substream->stream, channels) == -EINVAL) return -EINVAL; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) fifo_level = mcasp->txnumevt * active_serializers; @@ -619,9 +619,9 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, fifo_level = mcasp->rxnumevt * active_serializers; if (mcasp->op_mode == DAVINCI_MCASP_DIT_MODE) - davinci_hw_dit_param(mcasp); + mcasp_dit_hw_param(mcasp); else - davinci_hw_param(mcasp, substream->stream); + mcasp_i2s_hw_param(mcasp, substream->stream); switch (params_format(params)) { case SNDRV_PCM_FORMAT_U8: -- cgit v1.2.1 From 2c56c4c27c59edfaa779da156f6a70a38bb1f2df Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 30 Jan 2014 15:15:23 +0200 Subject: ASoC: davinci-mcasp: Configure xxTDM, xxFMT and xxFMCT registers synchronously These registers can be configured synchronously for playback and capture. Furthermore when McASP is in master and sync mode the capture operation needs the TX path to be configured in order to be able to provide the needed clocks for the bus. xxFMT and xxFMCT registers has been already configured for both TX and RX other places in the driver. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-mcasp.c | 56 +++++++++++++++++++-------------------- 1 file changed, 27 insertions(+), 29 deletions(-) (limited to 'sound') diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 9ec456aaf80b..ae3e40a63e5e 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -524,12 +524,18 @@ static int mcasp_common_hw_param(struct davinci_mcasp *mcasp, int stream, return 0; } -static void mcasp_i2s_hw_param(struct davinci_mcasp *mcasp, int stream) +static int mcasp_i2s_hw_param(struct davinci_mcasp *mcasp, int stream) { int i, active_slots; u32 mask = 0; u32 busel = 0; + if ((mcasp->tdm_slots < 2) || (mcasp->tdm_slots > 32)) { + dev_err(mcasp->dev, "tdm slot %d not supported\n", + mcasp->tdm_slots); + return -EINVAL; + } + active_slots = (mcasp->tdm_slots > 31) ? 32 : mcasp->tdm_slots; for (i = 0; i < active_slots; i++) mask |= (1 << i); @@ -539,35 +545,21 @@ static void mcasp_i2s_hw_param(struct davinci_mcasp *mcasp, int stream) if (!mcasp->dat_port) busel = TXSEL; - if (stream == SNDRV_PCM_STREAM_PLAYBACK) { - /* bit stream is MSB first with no delay */ - /* DSP_B mode */ - mcasp_set_reg(mcasp, DAVINCI_MCASP_TXTDM_REG, mask); - mcasp_set_bits(mcasp, DAVINCI_MCASP_TXFMT_REG, busel | TXORD); - - if ((mcasp->tdm_slots >= 2) && (mcasp->tdm_slots <= 32)) - mcasp_mod_bits(mcasp, DAVINCI_MCASP_TXFMCTL_REG, - FSXMOD(mcasp->tdm_slots), FSXMOD(0x1FF)); - else - printk(KERN_ERR "playback tdm slot %d not supported\n", - mcasp->tdm_slots); - } else { - /* bit stream is MSB first with no delay */ - /* DSP_B mode */ - mcasp_set_bits(mcasp, DAVINCI_MCASP_RXFMT_REG, busel | RXORD); - mcasp_set_reg(mcasp, DAVINCI_MCASP_RXTDM_REG, mask); - - if ((mcasp->tdm_slots >= 2) && (mcasp->tdm_slots <= 32)) - mcasp_mod_bits(mcasp, DAVINCI_MCASP_RXFMCTL_REG, - FSRMOD(mcasp->tdm_slots), FSRMOD(0x1FF)); - else - printk(KERN_ERR "capture tdm slot %d not supported\n", - mcasp->tdm_slots); - } + mcasp_set_reg(mcasp, DAVINCI_MCASP_TXTDM_REG, mask); + mcasp_set_bits(mcasp, DAVINCI_MCASP_TXFMT_REG, busel | TXORD); + mcasp_mod_bits(mcasp, DAVINCI_MCASP_TXFMCTL_REG, + FSXMOD(mcasp->tdm_slots), FSXMOD(0x1FF)); + + mcasp_set_reg(mcasp, DAVINCI_MCASP_RXTDM_REG, mask); + mcasp_set_bits(mcasp, DAVINCI_MCASP_RXFMT_REG, busel | RXORD); + mcasp_mod_bits(mcasp, DAVINCI_MCASP_RXFMCTL_REG, + FSRMOD(mcasp->tdm_slots), FSRMOD(0x1FF)); + + return 0; } /* S/PDIF */ -static void mcasp_dit_hw_param(struct davinci_mcasp *mcasp) +static int mcasp_dit_hw_param(struct davinci_mcasp *mcasp) { /* Set the TX format : 24 bit right rotation, 32 bit slot, Pad 0 and LSB first */ @@ -589,6 +581,8 @@ static void mcasp_dit_hw_param(struct davinci_mcasp *mcasp) /* Enable the DIT */ mcasp_set_bits(mcasp, DAVINCI_MCASP_TXDITCTL_REG, DITEN); + + return 0; } static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, @@ -605,6 +599,7 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, u8 slots = mcasp->tdm_slots; u8 active_serializers; int channels; + int ret; struct snd_interval *pcm_channels = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); channels = pcm_channels->min; @@ -619,9 +614,12 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, fifo_level = mcasp->rxnumevt * active_serializers; if (mcasp->op_mode == DAVINCI_MCASP_DIT_MODE) - mcasp_dit_hw_param(mcasp); + ret = mcasp_dit_hw_param(mcasp); else - mcasp_i2s_hw_param(mcasp, substream->stream); + ret = mcasp_i2s_hw_param(mcasp, substream->stream); + + if (ret) + return ret; switch (params_format(params)) { case SNDRV_PCM_FORMAT_U8: -- cgit v1.2.1 From 1d17a04ef2f2982fb81fde8ca3fe75723204a68a Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 30 Jan 2014 15:21:30 +0200 Subject: ASoC: davinci-mcasp: Consolidate pm_runtime_get/put() use in the driver The use of pm_runtime in trigger() callback is not correct and it will lead to unbalanced power.usage_count. The only place which might need to call pm_runtime is the set_fmt callback. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-mcasp.c | 23 +++++++++-------------- 1 file changed, 9 insertions(+), 14 deletions(-) (limited to 'sound') diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index ae3e40a63e5e..670afa29e30d 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -263,7 +263,9 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) { struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(cpu_dai); + int ret = 0; + pm_runtime_get_sync(mcasp->dev); switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_DSP_B: case SND_SOC_DAIFMT_AC97: @@ -317,7 +319,8 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, break; default: - return -EINVAL; + ret = -EINVAL; + goto out; } switch (fmt & SND_SOC_DAIFMT_INV_MASK) { @@ -354,10 +357,12 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, break; default: - return -EINVAL; + ret = -EINVAL; + break; } - - return 0; +out: + pm_runtime_put_sync(mcasp->dev); + return ret; } static int davinci_mcasp_set_clkdiv(struct snd_soc_dai *dai, int div_id, int div) @@ -676,19 +681,9 @@ static int davinci_mcasp_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - ret = pm_runtime_get_sync(mcasp->dev); - if (IS_ERR_VALUE(ret)) - dev_err(mcasp->dev, "pm_runtime_get_sync() failed\n"); davinci_mcasp_start(mcasp, substream->stream); break; - case SNDRV_PCM_TRIGGER_SUSPEND: - davinci_mcasp_stop(mcasp, substream->stream); - ret = pm_runtime_put_sync(mcasp->dev); - if (IS_ERR_VALUE(ret)) - dev_err(mcasp->dev, "pm_runtime_put_sync() failed\n"); - break; - case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: davinci_mcasp_stop(mcasp, substream->stream); -- cgit v1.2.1 From fb6d208d54de2791d6d361ef258ea7d5d3427d01 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 30 Jan 2014 15:21:31 +0200 Subject: ASoC: davinci-evm: Add pm callbacks to platform driver Set snd_soc_pm_ops for the pm ops. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-evm.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index 70ff3772079f..5e3bc3c6801a 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -399,6 +399,7 @@ static struct platform_driver davinci_evm_driver = { .driver = { .name = "davinci_evm", .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, .of_match_table = of_match_ptr(davinci_evm_dt_ids), }, }; -- cgit v1.2.1 From b31b2b6d5de71c569413d8dc4f7b050cbe25a09e Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Fri, 7 Feb 2014 09:35:16 +0200 Subject: ASoC: rt5640: Add ACPI ID for Intel Baytrail Realtek RT5640 uses ACPI ID "10EC5640" for Intel Baytrail platforms. Signed-off-by: Jarkko Nikula Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/rt5640.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index a3fb41179636..886924934aa5 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -2093,6 +2093,7 @@ MODULE_DEVICE_TABLE(i2c, rt5640_i2c_id); #ifdef CONFIG_ACPI static struct acpi_device_id rt5640_acpi_match[] = { { "INT33CA", 0 }, + { "10EC5640", 0 }, { }, }; MODULE_DEVICE_TABLE(acpi, rt5640_acpi_match); -- cgit v1.2.1 From 47cf84e17ebb79a20e6244b954c4ea4e18a82d43 Mon Sep 17 00:00:00 2001 From: Shawn Guo Date: Sat, 8 Feb 2014 13:20:35 +0800 Subject: ASoC: fsl: fix pm support of machine drivers The commit 1abe729 (ASoC: fsl: Add missing pm to current machine drivers) enables pm support for a few IMX machine drivers. But it does not update dev drvdata to be the pointer to 'card'. This causes the kernel dump below in system suspend, because snd_soc_suspend() expects that the dev drvdata points to 'card', while it still points to the private data of machine driver. This patch fixes imx-sgtl5000 and imx-wm8962 by attaching 'card' to dev drvdata and private data to card drvdata. For imx-mc13783, I simply revert the pm change because it must be broken for the same reason and I don't have hardware to test pm enabling code. $ echo mem > /sys/power/state PM: Syncing filesystems ... done. PM: Preparing system for mem sleep mmc1: card e624 removed Freezing user space processes ... (elapsed 0.002 seconds) done. Freezing remaining freezable tasks ... (elapsed 0.002 seconds) done. PM: Entering mem sleep INFO: trying to register non-static key. the code is fine but needs lockdep annotation. turning off the locking correctness validator. CPU: 0 PID: 1861 Comm: bash Not tainted 3.14.0-rc1+ #1648 Backtrace: [<80012144>] (dump_backtrace) from [<800122e4>] (show_stack+0x18/0x1c) r6:8079c77c r5:00000c5a r4:00000000 r3:00000000 [<800122cc>] (show_stack) from [<80637ac0>] (dump_stack+0x78/0x94) [<80637a48>] (dump_stack) from [<80028918>] (warn_slowpath_common+0x6c/0x8c) r4:bdb21c38 r3:be62df00 [<800288ac>] (warn_slowpath_common) from [<800289dc>] (warn_slowpath_fmt+0x38/0x40) r8:be62e3a8 r7:bf122960 r6:00000005 r5:00000000 r4:00000000 [<800289a8>] (warn_slowpath_fmt) from [<8006518c>] (__lock_acquire+0x1ae0/0x1ce0) r3:8079d598 r2:80799e70 [<800636ac>] (__lock_acquire) from [<80065894>] (lock_acquire+0x68/0x7c) r10:bdb20000 r9:be62df00 r8:00000000 r7:00000000 r6:60000013 r5:bdb20000 r4:00000000 [<8006582c>] (lock_acquire) from [<8063c938>] (mutex_lock_nested+0x5c/0x3b8) r7:00000000 r6:80dfc78c r5:804be444 r4:bf122928 [<8063c8dc>] (mutex_lock_nested) from [<804be444>] (snd_soc_suspend+0x34/0x42c) r10:00000000 r9:00000000 r8:00000000 r7:bf1c4444 r6:bf1c4410 r5:be978150 r4:be978010 [<804be410>] (snd_soc_suspend) from [<8034392c>] (platform_pm_suspend+0x34/0x64) r10:00000000 r8:00000000 r7:bf1c4444 r6:bf1c4410 r5:803438f8 r4:bf1c4410 [<803438f8>] (platform_pm_suspend) from [<80348e18>] (dpm_run_callback.isra.7+0x34/0x6c) [<80348de4>] (dpm_run_callback.isra.7) from [<80349354>] (__device_suspend+0x10c/0x220) r9:808dd974 r8:808c4a5c r6:00000002 r5:80e5001c r4:bf1c4410 [<80349248>] (__device_suspend) from [<8034a338>] (dpm_suspend+0x60/0x220) r7:bf1c4410 r6:808dd90c r5:80e5001c r4:bf1c44c0 [<8034a2d8>] (dpm_suspend) from [<8034a790>] (dpm_suspend_start+0x60/0x68) r10:8079a818 r9:00000000 r8:00000004 r7:80dfbe90 r6:80641eec r5:00000000 r4:00000002 [<8034a730>] (dpm_suspend_start) from [<8006a788>] (suspend_devices_and_enter+0x74/0x318) r4:00000003 r3:80dfbe98 [<8006a714>] (suspend_devices_and_enter) from [<8006abd8>] (pm_suspend+0x1ac/0x244) r10:8079a818 r8:00000004 r7:00000003 r6:80641eec r5:00000000 r4:00000003 [<8006aa2c>] (pm_suspend) from [<80069a4c>] (state_store+0x70/0xc0) r5:00000003 r4:bd85ea40 [<800699dc>] (state_store) from [<80294034>] (kobj_attr_store+0x1c/0x28) r10:beb9fe08 r8:00000000 r7:bdb21f78 r6:bd85ea40 r5:00000004 r4:beb9fe00 [<80294018>] (kobj_attr_store) from [<80140f90>] (sysfs_kf_write+0x54/0x58) [<80140f3c>] (sysfs_kf_write) from [<8014474c>] (kernfs_fop_write+0xc4/0x160) r6:bd85ea40 r5:beb9fe00 r4:00000004 r3:80140f3c [<80144688>] (kernfs_fop_write) from [<800dfa14>] (vfs_write+0xbc/0x184) r10:00000000 r9:00000000 r8:00000000 r7:bdb21f78 r6:00500c08 r5:00000004 r4:be782600 [<800df958>] (vfs_write) from [<800dfe00>] (SyS_write+0x48/0x70) r10:00000000 r8:00000000 r7:00000004 r6:00500c08 r5:00000000 r4:be782600 [<800dfdb8>] (SyS_write) from [<8000e800>] (ret_fast_syscall+0x0/0x48) r9:bdb20000 r8:8000e9c4 r7:00000004 r6:00500c08 r5:00000004 r4:76eb65e0 Fixes: 1abe729 (ASoC: fsl: Add missing pm to current machine drivers) Cc: stable@vger.kernel.org Signed-off-by: Shawn Guo Signed-off-by: Mark Brown --- sound/soc/fsl/imx-mc13783.c | 1 - sound/soc/fsl/imx-sgtl5000.c | 10 ++++++---- sound/soc/fsl/imx-wm8962.c | 11 +++++++---- 3 files changed, 13 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/imx-mc13783.c b/sound/soc/fsl/imx-mc13783.c index 79cee782dbbf..a2fd7321b5a9 100644 --- a/sound/soc/fsl/imx-mc13783.c +++ b/sound/soc/fsl/imx-mc13783.c @@ -160,7 +160,6 @@ static struct platform_driver imx_mc13783_audio_driver = { .driver = { .name = "imx_mc13783", .owner = THIS_MODULE, - .pm = &snd_soc_pm_ops, }, .probe = imx_mc13783_probe, .remove = imx_mc13783_remove diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c index f2beae78969f..1cb22dd034eb 100644 --- a/sound/soc/fsl/imx-sgtl5000.c +++ b/sound/soc/fsl/imx-sgtl5000.c @@ -33,8 +33,7 @@ struct imx_sgtl5000_data { static int imx_sgtl5000_dai_init(struct snd_soc_pcm_runtime *rtd) { - struct imx_sgtl5000_data *data = container_of(rtd->card, - struct imx_sgtl5000_data, card); + struct imx_sgtl5000_data *data = snd_soc_card_get_drvdata(rtd->card); struct device *dev = rtd->card->dev; int ret; @@ -159,13 +158,15 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) data->card.dapm_widgets = imx_sgtl5000_dapm_widgets; data->card.num_dapm_widgets = ARRAY_SIZE(imx_sgtl5000_dapm_widgets); + platform_set_drvdata(pdev, &data->card); + snd_soc_card_set_drvdata(&data->card, data); + ret = devm_snd_soc_register_card(&pdev->dev, &data->card); if (ret) { dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); goto fail; } - platform_set_drvdata(pdev, data); of_node_put(ssi_np); of_node_put(codec_np); @@ -184,7 +185,8 @@ fail: static int imx_sgtl5000_remove(struct platform_device *pdev) { - struct imx_sgtl5000_data *data = platform_get_drvdata(pdev); + struct snd_soc_card *card = platform_get_drvdata(pdev); + struct imx_sgtl5000_data *data = snd_soc_card_get_drvdata(card); clk_put(data->codec_clk); diff --git a/sound/soc/fsl/imx-wm8962.c b/sound/soc/fsl/imx-wm8962.c index 3fd76bc391de..3a3d17ce6ba4 100644 --- a/sound/soc/fsl/imx-wm8962.c +++ b/sound/soc/fsl/imx-wm8962.c @@ -71,7 +71,7 @@ static int imx_wm8962_set_bias_level(struct snd_soc_card *card, { struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; struct imx_priv *priv = &card_priv; - struct imx_wm8962_data *data = platform_get_drvdata(priv->pdev); + struct imx_wm8962_data *data = snd_soc_card_get_drvdata(card); struct device *dev = &priv->pdev->dev; unsigned int pll_out; int ret; @@ -137,7 +137,7 @@ static int imx_wm8962_late_probe(struct snd_soc_card *card) { struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; struct imx_priv *priv = &card_priv; - struct imx_wm8962_data *data = platform_get_drvdata(priv->pdev); + struct imx_wm8962_data *data = snd_soc_card_get_drvdata(card); struct device *dev = &priv->pdev->dev; int ret; @@ -264,13 +264,15 @@ static int imx_wm8962_probe(struct platform_device *pdev) data->card.late_probe = imx_wm8962_late_probe; data->card.set_bias_level = imx_wm8962_set_bias_level; + platform_set_drvdata(pdev, &data->card); + snd_soc_card_set_drvdata(&data->card, data); + ret = devm_snd_soc_register_card(&pdev->dev, &data->card); if (ret) { dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); goto clk_fail; } - platform_set_drvdata(pdev, data); of_node_put(ssi_np); of_node_put(codec_np); @@ -289,7 +291,8 @@ fail: static int imx_wm8962_remove(struct platform_device *pdev) { - struct imx_wm8962_data *data = platform_get_drvdata(pdev); + struct snd_soc_card *card = platform_get_drvdata(pdev); + struct imx_wm8962_data *data = snd_soc_card_get_drvdata(card); if (!IS_ERR(data->codec_clk)) clk_disable_unprepare(data->codec_clk); -- cgit v1.2.1 From 236014ac7a6524f9f466139c2e47af70cb340ba3 Mon Sep 17 00:00:00 2001 From: Xiubo Li Date: Mon, 10 Feb 2014 14:47:17 +0800 Subject: ASoC: fsl-esai: fix ESAI TDM slot setting Signed-off-by: Xiubo Li Acked-by: Nicolin Chen Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_esai.c | 4 ++-- sound/soc/fsl/fsl_esai.h | 2 +- 2 files changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index d0c72ed261e7..c84026c99134 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -326,7 +326,7 @@ static int fsl_esai_set_dai_tdm_slot(struct snd_soc_dai *dai, u32 tx_mask, regmap_update_bits(esai_priv->regmap, REG_ESAI_TSMA, ESAI_xSMA_xS_MASK, ESAI_xSMA_xS(tx_mask)); regmap_update_bits(esai_priv->regmap, REG_ESAI_TSMB, - ESAI_xSMA_xS_MASK, ESAI_xSMB_xS(tx_mask)); + ESAI_xSMB_xS_MASK, ESAI_xSMB_xS(tx_mask)); regmap_update_bits(esai_priv->regmap, REG_ESAI_RCCR, ESAI_xCCR_xDC_MASK, ESAI_xCCR_xDC(slots)); @@ -334,7 +334,7 @@ static int fsl_esai_set_dai_tdm_slot(struct snd_soc_dai *dai, u32 tx_mask, regmap_update_bits(esai_priv->regmap, REG_ESAI_RSMA, ESAI_xSMA_xS_MASK, ESAI_xSMA_xS(rx_mask)); regmap_update_bits(esai_priv->regmap, REG_ESAI_RSMB, - ESAI_xSMA_xS_MASK, ESAI_xSMB_xS(rx_mask)); + ESAI_xSMB_xS_MASK, ESAI_xSMB_xS(rx_mask)); esai_priv->slot_width = slot_width; diff --git a/sound/soc/fsl/fsl_esai.h b/sound/soc/fsl/fsl_esai.h index 9c9f957fcae1..75e14033e8d8 100644 --- a/sound/soc/fsl/fsl_esai.h +++ b/sound/soc/fsl/fsl_esai.h @@ -322,7 +322,7 @@ #define ESAI_xSMB_xS_SHIFT 0 #define ESAI_xSMB_xS_WIDTH 16 #define ESAI_xSMB_xS_MASK (((1 << ESAI_xSMB_xS_WIDTH) - 1) << ESAI_xSMB_xS_SHIFT) -#define ESAI_xSMB_xS(v) (((v) >> ESAI_xSMA_xS_WIDTH) & ESAI_xSMA_xS_MASK) +#define ESAI_xSMB_xS(v) (((v) >> ESAI_xSMA_xS_WIDTH) & ESAI_xSMB_xS_MASK) /* Port C Direction Register -- REG_ESAI_PRRC 0xF8 */ #define ESAI_PRRC_PDC_SHIFT 0 -- cgit v1.2.1 From 07b0e5b10258b48e5edfb6c8ac156f05510eb775 Mon Sep 17 00:00:00 2001 From: Adam Thomson Date: Thu, 6 Feb 2014 18:03:07 +0000 Subject: ASoC: da9055: Fix device registration of PMIC and CODEC devices Currently the I2C device Ids conflict for the MFD and CODEC so cannot be both instantiated on one platform. This patch updates the Ids and names to make them unique from each other. It should be noted that the I2C addresses for both PMIC and CODEC are modifiable so instantiation of the two are kept as separate devices, rather than instantiating the CODEC from the MFD code. Signed-off-by: Adam Thomson Acked-by: Mark Brown Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/da9055.c | 11 +++++++++-- 1 file changed, 9 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/da9055.c b/sound/soc/codecs/da9055.c index 52b79a487ac7..422812613a28 100644 --- a/sound/soc/codecs/da9055.c +++ b/sound/soc/codecs/da9055.c @@ -1523,8 +1523,15 @@ static int da9055_remove(struct i2c_client *client) return 0; } +/* + * DO NOT change the device Ids. The naming is intentionally specific as both + * the CODEC and PMIC parts of this chip are instantiated separately as I2C + * devices (both have configurable I2C addresses, and are to all intents and + * purposes separate). As a result there are specific DA9055 Ids for CODEC + * and PMIC, which must be different to operate together. + */ static const struct i2c_device_id da9055_i2c_id[] = { - { "da9055", 0 }, + { "da9055-codec", 0 }, { } }; MODULE_DEVICE_TABLE(i2c, da9055_i2c_id); @@ -1532,7 +1539,7 @@ MODULE_DEVICE_TABLE(i2c, da9055_i2c_id); /* I2C codec control layer */ static struct i2c_driver da9055_i2c_driver = { .driver = { - .name = "da9055", + .name = "da9055-codec", .owner = THIS_MODULE, }, .probe = da9055_i2c_probe, -- cgit v1.2.1 From e3947ecb4e8adf260b6323eac43b80eeb26911cf Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 12 Feb 2014 17:11:22 +0100 Subject: ASoC: blackfin: Fix machine driver Kconfig dependencies Since the machine driver selects the CODEC driver we need to make sure that the machine driver is only selectable if the CODEC driver can be build. This avoids build errors under some configurations (which typically only result from randconfig). Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/blackfin/Kconfig | 11 +++++------ 1 file changed, 5 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/blackfin/Kconfig b/sound/soc/blackfin/Kconfig index 54f74f8cbb75..4544d8eb1452 100644 --- a/sound/soc/blackfin/Kconfig +++ b/sound/soc/blackfin/Kconfig @@ -11,7 +11,7 @@ config SND_BF5XX_I2S config SND_BF5XX_SOC_SSM2602 tristate "SoC SSM2602 Audio Codec Add-On Card support" - depends on SND_BF5XX_I2S && (SPI_MASTER || I2C) + depends on SND_BF5XX_I2S && SND_SOC_I2C_AND_SPI select SND_BF5XX_SOC_I2S if !BF60x select SND_BF6XX_SOC_I2S if BF60x select SND_SOC_SSM2602 @@ -21,10 +21,9 @@ config SND_BF5XX_SOC_SSM2602 config SND_SOC_BFIN_EVAL_ADAU1701 tristate "Support for the EVAL-ADAU1701MINIZ board on Blackfin eval boards" - depends on SND_BF5XX_I2S + depends on SND_BF5XX_I2S && I2C select SND_BF5XX_SOC_I2S select SND_SOC_ADAU1701 - select I2C help Say Y if you want to add support for the Analog Devices EVAL-ADAU1701MINIZ board connected to one of the Blackfin evaluation boards like the @@ -45,7 +44,7 @@ config SND_SOC_BFIN_EVAL_ADAU1373 config SND_SOC_BFIN_EVAL_ADAV80X tristate "Support for the EVAL-ADAV80X boards on Blackfin eval boards" - depends on SND_BF5XX_I2S && (SPI_MASTER || I2C) + depends on SND_BF5XX_I2S && SND_SOC_I2C_AND_SPI select SND_BF5XX_SOC_I2S select SND_SOC_ADAV80X help @@ -58,7 +57,7 @@ config SND_SOC_BFIN_EVAL_ADAV80X config SND_BF5XX_SOC_AD1836 tristate "SoC AD1836 Audio support for BF5xx" - depends on SND_BF5XX_I2S + depends on SND_BF5XX_I2S && SPI_MASTER select SND_BF5XX_SOC_I2S select SND_SOC_AD1836 help @@ -66,7 +65,7 @@ config SND_BF5XX_SOC_AD1836 config SND_BF5XX_SOC_AD193X tristate "SoC AD193X Audio support for Blackfin" - depends on SND_BF5XX_I2S + depends on SND_BF5XX_I2S && SND_SOC_I2C_AND_SPI select SND_BF5XX_SOC_I2S select SND_SOC_AD193X help -- cgit v1.2.1 From c42c8922c46d33ed769e99618bdfba06866a0c72 Mon Sep 17 00:00:00 2001 From: Dylan Reid Date: Wed, 12 Feb 2014 10:24:54 -0800 Subject: ASoC: max98090: sync regcache on entering STANDBY Sync regcache when entering STANDBY from OFF. ON isn't entered with OFF as the current state, so the registers were not being re-synced after suspend/resume. The 98088 and 98095 already call regcache_sync from STANDBY. Signed-off-by: Dylan Reid Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/max98090.c | 20 ++++++++++---------- 1 file changed, 10 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index 51f9b3d16b41..149b57f6334b 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -1769,16 +1769,6 @@ static int max98090_set_bias_level(struct snd_soc_codec *codec, switch (level) { case SND_SOC_BIAS_ON: - if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { - ret = regcache_sync(max98090->regmap); - - if (ret != 0) { - dev_err(codec->dev, - "Failed to sync cache: %d\n", ret); - return ret; - } - } - if (max98090->jack_state == M98090_JACK_STATE_HEADSET) { /* * Set to normal bias level. @@ -1792,6 +1782,16 @@ static int max98090_set_bias_level(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_STANDBY: + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { + ret = regcache_sync(max98090->regmap); + if (ret != 0) { + dev_err(codec->dev, + "Failed to sync cache: %d\n", ret); + return ret; + } + } + break; + case SND_SOC_BIAS_OFF: /* Set internal pull-up to lowest power mode */ snd_soc_update_bits(codec, M98090_REG_JACK_DETECT, -- cgit v1.2.1 From 9febd494d15c4a351e9c9cae7184643144eea892 Mon Sep 17 00:00:00 2001 From: Alexander Shiyan Date: Sat, 15 Feb 2014 23:28:29 +0400 Subject: ASoC: txx9aclc_ac97: Fix kernel crash on probe This patch fixes a crash caused by commit 3bed3344c826 (ASoC: txx9aclc_ac97: Convert to devm_ioremap_resource()). This is an attempt to assign "drvdata->base" while memory for "drvdata" is not already allocated. Fixes: 3bed3344c826 (ASoC: txx9aclc_ac97: Convert to devm_ioremap_resource()) Signed-off-by: Alexander Shiyan Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/txx9/txx9aclc-ac97.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/txx9/txx9aclc-ac97.c b/sound/soc/txx9/txx9aclc-ac97.c index e0305a148568..9edd68db9f48 100644 --- a/sound/soc/txx9/txx9aclc-ac97.c +++ b/sound/soc/txx9/txx9aclc-ac97.c @@ -183,14 +183,16 @@ static int txx9aclc_ac97_dev_probe(struct platform_device *pdev) irq = platform_get_irq(pdev, 0); if (irq < 0) return irq; + + drvdata = devm_kzalloc(&pdev->dev, sizeof(*drvdata), GFP_KERNEL); + if (!drvdata) + return -ENOMEM; + r = platform_get_resource(pdev, IORESOURCE_MEM, 0); drvdata->base = devm_ioremap_resource(&pdev->dev, r); if (IS_ERR(drvdata->base)) return PTR_ERR(drvdata->base); - drvdata = devm_kzalloc(&pdev->dev, sizeof(*drvdata), GFP_KERNEL); - if (!drvdata) - return -ENOMEM; platform_set_drvdata(pdev, drvdata); drvdata->physbase = r->start; if (sizeof(drvdata->physbase) > sizeof(r->start) && -- cgit v1.2.1 From e126a646f77fdd66978785cb0a3a5e46b07aee2e Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Thu, 13 Feb 2014 16:54:24 -0700 Subject: ASoC: max98090: make REVISION_ID readable The REVISION_ID register is not currently marked readable. snd_soc_read() refuses to read the register, and hence probe() fails. Fixes: d4807ad2c4c0 ("regmap: Check readable regs in _regmap_read") [exposed the bug, by checking for readability] Fixes: 685e42154dcf ("ASoC: Replace max98090 Device Driver") [left out this register from the readable list] Signed-off-by: Stephen Warren Signed-off-by: Mark Brown --- sound/soc/codecs/max98090.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index 149b57f6334b..9f714ea86613 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -336,6 +336,7 @@ static bool max98090_readable_register(struct device *dev, unsigned int reg) case M98090_REG_RECORD_TDM_SLOT: case M98090_REG_SAMPLE_RATE: case M98090_REG_DMIC34_BIQUAD_BASE ... M98090_REG_DMIC34_BIQUAD_BASE + 0x0E: + case M98090_REG_REVISION_ID: return true; default: return false; -- cgit v1.2.1 From 624aef494f86ed0c58056361c06347ad62b26806 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Sun, 16 Feb 2014 17:11:10 +0100 Subject: ALSA: usb-audio: work around KEF X300A firmware bug When the driver tries to access Function Unit 10, the KEF X300A speakers' firmware apparently locks up, making even PCM streaming impossible. Work around this by ignoring this FU. Cc: Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/usb/mixer_maps.c | 9 +++++++++ 1 file changed, 9 insertions(+) (limited to 'sound') diff --git a/sound/usb/mixer_maps.c b/sound/usb/mixer_maps.c index 32af6b741ef5..d1d72ff50347 100644 --- a/sound/usb/mixer_maps.c +++ b/sound/usb/mixer_maps.c @@ -328,6 +328,11 @@ static struct usbmix_name_map gamecom780_map[] = { {} }; +static const struct usbmix_name_map kef_x300a_map[] = { + { 10, NULL }, /* firmware locks up (?) when we try to access this FU */ + { 0 } +}; + /* * Control map entries */ @@ -419,6 +424,10 @@ static struct usbmix_ctl_map usbmix_ctl_maps[] = { .id = USB_ID(0x200c, 0x1018), .map = ebox44_map, }, + { + .id = USB_ID(0x27ac, 0x1000), + .map = kef_x300a_map, + }, { 0 } /* terminator */ }; -- cgit v1.2.1 From 4913e0bf239dafee356bc7fab61806cc2518930c Mon Sep 17 00:00:00 2001 From: Hui Wang Date: Tue, 18 Feb 2014 10:56:46 +0800 Subject: ALSA: hda - add headset mic detect quirks for two Dell laptops When we plug a 3-ring headset on the Dell machines (Vendor ID: 0x10ec0255, Subsystem ID: 0x10280657; Vendor ID: 0x10ec0255, Subsystem ID: 0x1028065f), the headset mic can't be detected, after apply this patch, the headset mic can work well. BugLink: https://bugs.launchpad.net/bugs/1260303 Cc: David Henningsson Tested-by: Cyrus Lien Cc: stable@vger.kernel.org Signed-off-by: Hui Wang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a9a83b85517a..6eb903cc6237 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4308,7 +4308,9 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0651, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0652, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0653, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x0657, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0658, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x065f, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0662, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x15cc, "Dell X5 Precision", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x15cd, "Dell X5 Precision", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), -- cgit v1.2.1 From 30686c350628a68852f8abd67557aecb137789d5 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Tue, 18 Feb 2014 16:05:27 +0000 Subject: ASoC: dapm: Correct regulator bypass error messages The error messages for bypassing/unbypassing a regulator appear to be swapped round, this patch corrects these. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index dc8ff13187f7..fd39cd2827d7 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1218,7 +1218,7 @@ int dapm_regulator_event(struct snd_soc_dapm_widget *w, ret = regulator_allow_bypass(w->regulator, false); if (ret != 0) dev_warn(w->dapm->dev, - "ASoC: Failed to bypass %s: %d\n", + "ASoC: Failed to unbypass %s: %d\n", w->name, ret); } @@ -1228,7 +1228,7 @@ int dapm_regulator_event(struct snd_soc_dapm_widget *w, ret = regulator_allow_bypass(w->regulator, true); if (ret != 0) dev_warn(w->dapm->dev, - "ASoC: Failed to unbypass %s: %d\n", + "ASoC: Failed to bypass %s: %d\n", w->name, ret); } @@ -3248,7 +3248,7 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, ret = regulator_allow_bypass(w->regulator, true); if (ret != 0) dev_warn(w->dapm->dev, - "ASoC: Failed to unbypass %s: %d\n", + "ASoC: Failed to bypass %s: %d\n", w->name, ret); } break; -- cgit v1.2.1 From 28fba95087a7f3d107a3a6728aef7dbfaf3fd782 Mon Sep 17 00:00:00 2001 From: Hsin-Yu Chao Date: Wed, 19 Feb 2014 14:27:07 +0800 Subject: ALSA: hda/ca0132 - setup/cleanup streams When a HDMI stream is opened with the same stream tag as a following opened stream to ca0132, audio will be heard from two ports simultaneously. Fix this issue by change to use snd_hda_codec_setup_stream and snd_hda_codec_cleanup_stream instead, so that an inactive stream can be marked as 'dirty' when found with a conflict stream tag, and then get purified. Signed-off-by: Hsin-Yu Chao Reviewed-by: Chih-Chung Chang Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_ca0132.c | 66 +++++--------------------------------------- 1 file changed, 7 insertions(+), 59 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 54d14793725a..0aa72ee38d03 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -2661,60 +2661,6 @@ static bool dspload_wait_loaded(struct hda_codec *codec) return false; } -/* - * PCM stuffs - */ -static void ca0132_setup_stream(struct hda_codec *codec, hda_nid_t nid, - u32 stream_tag, - int channel_id, int format) -{ - unsigned int oldval, newval; - - if (!nid) - return; - - snd_printdd( - "ca0132_setup_stream: NID=0x%x, stream=0x%x, " - "channel=%d, format=0x%x\n", - nid, stream_tag, channel_id, format); - - /* update the format-id if changed */ - oldval = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_STREAM_FORMAT, - 0); - if (oldval != format) { - msleep(20); - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_STREAM_FORMAT, - format); - } - - oldval = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONV, 0); - newval = (stream_tag << 4) | channel_id; - if (oldval != newval) { - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_CHANNEL_STREAMID, - newval); - } -} - -static void ca0132_cleanup_stream(struct hda_codec *codec, hda_nid_t nid) -{ - unsigned int val; - - if (!nid) - return; - - snd_printdd(KERN_INFO "ca0132_cleanup_stream: NID=0x%x\n", nid); - - val = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONV, 0); - if (!val) - return; - - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_STREAM_FORMAT, 0); - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CHANNEL_STREAMID, 0); -} - /* * PCM callbacks */ @@ -2726,7 +2672,7 @@ static int ca0132_playback_pcm_prepare(struct hda_pcm_stream *hinfo, { struct ca0132_spec *spec = codec->spec; - ca0132_setup_stream(codec, spec->dacs[0], stream_tag, 0, format); + snd_hda_codec_setup_stream(codec, spec->dacs[0], stream_tag, 0, format); return 0; } @@ -2745,7 +2691,7 @@ static int ca0132_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, if (spec->effects_switch[PLAY_ENHANCEMENT - EFFECT_START_NID]) msleep(50); - ca0132_cleanup_stream(codec, spec->dacs[0]); + snd_hda_codec_cleanup_stream(codec, spec->dacs[0]); return 0; } @@ -2824,8 +2770,8 @@ static int ca0132_capture_pcm_prepare(struct hda_pcm_stream *hinfo, { struct ca0132_spec *spec = codec->spec; - ca0132_setup_stream(codec, spec->adcs[substream->number], - stream_tag, 0, format); + snd_hda_codec_setup_stream(codec, spec->adcs[substream->number], + stream_tag, 0, format); return 0; } @@ -2839,7 +2785,7 @@ static int ca0132_capture_pcm_cleanup(struct hda_pcm_stream *hinfo, if (spec->dsp_state == DSP_DOWNLOADING) return 0; - ca0132_cleanup_stream(codec, hinfo->nid); + snd_hda_codec_cleanup_stream(codec, hinfo->nid); return 0; } @@ -4742,6 +4688,8 @@ static int patch_ca0132(struct hda_codec *codec) return err; codec->patch_ops = ca0132_patch_ops; + codec->pcm_format_first = 1; + codec->no_sticky_stream = 1; return 0; } -- cgit v1.2.1 From 13c12dbe3a2ce17227f7ddef652b6a53c78fa51f Mon Sep 17 00:00:00 2001 From: Hsin-Yu Chao Date: Wed, 19 Feb 2014 14:30:35 +0800 Subject: ALSA: hda/ca0132 - Fix recording from mode id 0x8 Incorrect ADC is picked in ca0132_capture_pcm_prepare(), where it assumes multiple streams while there is one stream per ADC. Note that ca0132_capture_pcm_cleanup() already does the right thing. The Chromebook Pixel has a microphone under the keyboard that is attached to node id 0x8. Before this fix, recording would always go to the main internal mic (node id 0x7). Signed-off-by: Hsin-Yu Chao Reviewed-by: Dylan Reid Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_ca0132.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 0aa72ee38d03..46ecdbb9053f 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -2768,9 +2768,7 @@ static int ca0132_capture_pcm_prepare(struct hda_pcm_stream *hinfo, unsigned int format, struct snd_pcm_substream *substream) { - struct ca0132_spec *spec = codec->spec; - - snd_hda_codec_setup_stream(codec, spec->adcs[substream->number], + snd_hda_codec_setup_stream(codec, hinfo->nid, stream_tag, 0, format); return 0; -- cgit v1.2.1 From 025c3fa9256d4c54506b7a29dc3befac54f5c68d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Feb 2014 09:24:12 +0100 Subject: ASoC: sta32x: Fix array access overflow Preset EQ enum of sta32x codec driver declares too many number of items and it may lead to the access over the actual array size. Use SOC_ENUM_SINGLE_DECL() helper and it's automatically fixed. Signed-off-by: Takashi Iwai Acked-by: Liam Girdwood Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/sta32x.c | 72 +++++++++++++++++++++++------------------------ 1 file changed, 36 insertions(+), 36 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c index 06edb396e733..42c5b458b046 100644 --- a/sound/soc/codecs/sta32x.c +++ b/sound/soc/codecs/sta32x.c @@ -187,42 +187,42 @@ static const unsigned int sta32x_limiter_drc_release_tlv[] = { 13, 16, TLV_DB_SCALE_ITEM(-1500, 300, 0), }; -static const struct soc_enum sta32x_drc_ac_enum = - SOC_ENUM_SINGLE(STA32X_CONFD, STA32X_CONFD_DRC_SHIFT, - 2, sta32x_drc_ac); -static const struct soc_enum sta32x_auto_eq_enum = - SOC_ENUM_SINGLE(STA32X_AUTO1, STA32X_AUTO1_AMEQ_SHIFT, - 3, sta32x_auto_eq_mode); -static const struct soc_enum sta32x_auto_gc_enum = - SOC_ENUM_SINGLE(STA32X_AUTO1, STA32X_AUTO1_AMGC_SHIFT, - 4, sta32x_auto_gc_mode); -static const struct soc_enum sta32x_auto_xo_enum = - SOC_ENUM_SINGLE(STA32X_AUTO2, STA32X_AUTO2_XO_SHIFT, - 16, sta32x_auto_xo_mode); -static const struct soc_enum sta32x_preset_eq_enum = - SOC_ENUM_SINGLE(STA32X_AUTO3, STA32X_AUTO3_PEQ_SHIFT, - 32, sta32x_preset_eq_mode); -static const struct soc_enum sta32x_limiter_ch1_enum = - SOC_ENUM_SINGLE(STA32X_C1CFG, STA32X_CxCFG_LS_SHIFT, - 3, sta32x_limiter_select); -static const struct soc_enum sta32x_limiter_ch2_enum = - SOC_ENUM_SINGLE(STA32X_C2CFG, STA32X_CxCFG_LS_SHIFT, - 3, sta32x_limiter_select); -static const struct soc_enum sta32x_limiter_ch3_enum = - SOC_ENUM_SINGLE(STA32X_C3CFG, STA32X_CxCFG_LS_SHIFT, - 3, sta32x_limiter_select); -static const struct soc_enum sta32x_limiter1_attack_rate_enum = - SOC_ENUM_SINGLE(STA32X_L1AR, STA32X_LxA_SHIFT, - 16, sta32x_limiter_attack_rate); -static const struct soc_enum sta32x_limiter2_attack_rate_enum = - SOC_ENUM_SINGLE(STA32X_L2AR, STA32X_LxA_SHIFT, - 16, sta32x_limiter_attack_rate); -static const struct soc_enum sta32x_limiter1_release_rate_enum = - SOC_ENUM_SINGLE(STA32X_L1AR, STA32X_LxR_SHIFT, - 16, sta32x_limiter_release_rate); -static const struct soc_enum sta32x_limiter2_release_rate_enum = - SOC_ENUM_SINGLE(STA32X_L2AR, STA32X_LxR_SHIFT, - 16, sta32x_limiter_release_rate); +static SOC_ENUM_SINGLE_DECL(sta32x_drc_ac_enum, + STA32X_CONFD, STA32X_CONFD_DRC_SHIFT, + sta32x_drc_ac); +static SOC_ENUM_SINGLE_DECL(sta32x_auto_eq_enum, + STA32X_AUTO1, STA32X_AUTO1_AMEQ_SHIFT, + sta32x_auto_eq_mode); +static SOC_ENUM_SINGLE_DECL(sta32x_auto_gc_enum, + STA32X_AUTO1, STA32X_AUTO1_AMGC_SHIFT, + sta32x_auto_gc_mode); +static SOC_ENUM_SINGLE_DECL(sta32x_auto_xo_enum, + STA32X_AUTO2, STA32X_AUTO2_XO_SHIFT, + sta32x_auto_xo_mode); +static SOC_ENUM_SINGLE_DECL(sta32x_preset_eq_enum, + STA32X_AUTO3, STA32X_AUTO3_PEQ_SHIFT, + sta32x_preset_eq_mode); +static SOC_ENUM_SINGLE_DECL(sta32x_limiter_ch1_enum, + STA32X_C1CFG, STA32X_CxCFG_LS_SHIFT, + sta32x_limiter_select); +static SOC_ENUM_SINGLE_DECL(sta32x_limiter_ch2_enum, + STA32X_C2CFG, STA32X_CxCFG_LS_SHIFT, + sta32x_limiter_select); +static SOC_ENUM_SINGLE_DECL(sta32x_limiter_ch3_enum, + STA32X_C3CFG, STA32X_CxCFG_LS_SHIFT, + sta32x_limiter_select); +static SOC_ENUM_SINGLE_DECL(sta32x_limiter1_attack_rate_enum, + STA32X_L1AR, STA32X_LxA_SHIFT, + sta32x_limiter_attack_rate); +static SOC_ENUM_SINGLE_DECL(sta32x_limiter2_attack_rate_enum, + STA32X_L2AR, STA32X_LxA_SHIFT, + sta32x_limiter_attack_rate); +static SOC_ENUM_SINGLE_DECL(sta32x_limiter1_release_rate_enum, + STA32X_L1AR, STA32X_LxR_SHIFT, + sta32x_limiter_release_rate); +static SOC_ENUM_SINGLE_DECL(sta32x_limiter2_release_rate_enum, + STA32X_L2AR, STA32X_LxR_SHIFT, + sta32x_limiter_release_rate); /* byte array controls for setting biquad, mixer, scaling coefficients; * for biquads all five coefficients need to be set in one go, -- cgit v1.2.1 From 7a6c0a58dc824523966f212c76322d47c5b0e6fe Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Feb 2014 09:37:30 +0100 Subject: ASoC: wm8770: Fix wrong number of enum items wm8770 codec driver defines ain_enum with a wrong number of items. Use SOC_ENUM_DOUBLE_DECL() macro and it's automatically fixed. Signed-off-by: Takashi Iwai Acked-by: Liam Girdwood Acked-by: Charles Keepax Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/wm8770.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8770.c b/sound/soc/codecs/wm8770.c index 89a18d82f303..5bce21013485 100644 --- a/sound/soc/codecs/wm8770.c +++ b/sound/soc/codecs/wm8770.c @@ -196,8 +196,8 @@ static const char *ain_text[] = { "AIN5", "AIN6", "AIN7", "AIN8" }; -static const struct soc_enum ain_enum = - SOC_ENUM_DOUBLE(WM8770_ADCMUX, 0, 4, 8, ain_text); +static SOC_ENUM_DOUBLE_DECL(ain_enum, + WM8770_ADCMUX, 0, 4, ain_text); static const struct snd_kcontrol_new ain_mux = SOC_DAPM_ENUM("Capture Mux", ain_enum); -- cgit v1.2.1 From 9d1663143652d579dd5555e60fd8c78362ffb8c6 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Feb 2014 09:39:46 +0100 Subject: ASoC: wm8900: Fix the wrong number of enum items wm8900 codec driver has a few places wrongly defining the number of enum items. Use SOC_ENUM_SINGLE_DECL() macro and they are automatically fixed. Acked-by: Liam Girdwood Acked-by: Charles Keepax Acked-by: Lars-Peter Clausen Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/codecs/wm8900.c | 44 ++++++++++++++++++++++---------------------- 1 file changed, 22 insertions(+), 22 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c index e98bc7038a08..43c2201cb901 100644 --- a/sound/soc/codecs/wm8900.c +++ b/sound/soc/codecs/wm8900.c @@ -304,53 +304,53 @@ static const DECLARE_TLV_DB_SCALE(adc_tlv, -7200, 75, 1); static const char *mic_bias_level_txt[] = { "0.9*AVDD", "0.65*AVDD" }; -static const struct soc_enum mic_bias_level = -SOC_ENUM_SINGLE(WM8900_REG_INCTL, 8, 2, mic_bias_level_txt); +static SOC_ENUM_SINGLE_DECL(mic_bias_level, + WM8900_REG_INCTL, 8, mic_bias_level_txt); static const char *dac_mute_rate_txt[] = { "Fast", "Slow" }; -static const struct soc_enum dac_mute_rate = -SOC_ENUM_SINGLE(WM8900_REG_DACCTRL, 7, 2, dac_mute_rate_txt); +static SOC_ENUM_SINGLE_DECL(dac_mute_rate, + WM8900_REG_DACCTRL, 7, dac_mute_rate_txt); static const char *dac_deemphasis_txt[] = { "Disabled", "32kHz", "44.1kHz", "48kHz" }; -static const struct soc_enum dac_deemphasis = -SOC_ENUM_SINGLE(WM8900_REG_DACCTRL, 4, 4, dac_deemphasis_txt); +static SOC_ENUM_SINGLE_DECL(dac_deemphasis, + WM8900_REG_DACCTRL, 4, dac_deemphasis_txt); static const char *adc_hpf_cut_txt[] = { "Hi-fi mode", "Voice mode 1", "Voice mode 2", "Voice mode 3" }; -static const struct soc_enum adc_hpf_cut = -SOC_ENUM_SINGLE(WM8900_REG_ADCCTRL, 5, 4, adc_hpf_cut_txt); +static SOC_ENUM_SINGLE_DECL(adc_hpf_cut, + WM8900_REG_ADCCTRL, 5, adc_hpf_cut_txt); static const char *lr_txt[] = { "Left", "Right" }; -static const struct soc_enum aifl_src = -SOC_ENUM_SINGLE(WM8900_REG_AUDIO1, 15, 2, lr_txt); +static SOC_ENUM_SINGLE_DECL(aifl_src, + WM8900_REG_AUDIO1, 15, lr_txt); -static const struct soc_enum aifr_src = -SOC_ENUM_SINGLE(WM8900_REG_AUDIO1, 14, 2, lr_txt); +static SOC_ENUM_SINGLE_DECL(aifr_src, + WM8900_REG_AUDIO1, 14, lr_txt); -static const struct soc_enum dacl_src = -SOC_ENUM_SINGLE(WM8900_REG_AUDIO2, 15, 2, lr_txt); +static SOC_ENUM_SINGLE_DECL(dacl_src, + WM8900_REG_AUDIO2, 15, lr_txt); -static const struct soc_enum dacr_src = -SOC_ENUM_SINGLE(WM8900_REG_AUDIO2, 14, 2, lr_txt); +static SOC_ENUM_SINGLE_DECL(dacr_src, + WM8900_REG_AUDIO2, 14, lr_txt); static const char *sidetone_txt[] = { "Disabled", "Left ADC", "Right ADC" }; -static const struct soc_enum dacl_sidetone = -SOC_ENUM_SINGLE(WM8900_REG_SIDETONE, 2, 3, sidetone_txt); +static SOC_ENUM_SINGLE_DECL(dacl_sidetone, + WM8900_REG_SIDETONE, 2, sidetone_txt); -static const struct soc_enum dacr_sidetone = -SOC_ENUM_SINGLE(WM8900_REG_SIDETONE, 0, 3, sidetone_txt); +static SOC_ENUM_SINGLE_DECL(dacr_sidetone, + WM8900_REG_SIDETONE, 0, sidetone_txt); static const struct snd_kcontrol_new wm8900_snd_controls[] = { SOC_ENUM("Mic Bias Level", mic_bias_level), @@ -496,8 +496,8 @@ SOC_DAPM_SINGLE("RINPUT3 Switch", WM8900_REG_INCTL, 0, 1, 0), static const char *wm8900_lp_mux[] = { "Disabled", "Enabled" }; -static const struct soc_enum wm8900_lineout2_lp_mux = -SOC_ENUM_SINGLE(WM8900_REG_LOUTMIXCTL1, 1, 2, wm8900_lp_mux); +static SOC_ENUM_SINGLE_DECL(wm8900_lineout2_lp_mux, + WM8900_REG_LOUTMIXCTL1, 1, wm8900_lp_mux); static const struct snd_kcontrol_new wm8900_lineout2_lp = SOC_DAPM_ENUM("Route", wm8900_lineout2_lp_mux); -- cgit v1.2.1 From e61a35b798497a85725ce10c07049fa1b9ca014c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Feb 2014 09:45:30 +0100 Subject: ASoC: wm8994: Fix the wrong number of enum items wm8994 codec driver has a few places wrongly defining the number of enum items. Use SOC_ENUM_SINGLE_DECL() macro and they are automatically fixed. Signed-off-by: Takashi Iwai Acked-by: Liam Girdwood Acked-by: Charles Keepax Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 135 ++++++++++++++++++++++++---------------------- 1 file changed, 70 insertions(+), 65 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index b9be9cbc4603..adb72063d44e 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -265,21 +265,21 @@ static const char *sidetone_hpf_text[] = { "2.7kHz", "1.35kHz", "675Hz", "370Hz", "180Hz", "90Hz", "45Hz" }; -static const struct soc_enum sidetone_hpf = - SOC_ENUM_SINGLE(WM8994_SIDETONE, 7, 7, sidetone_hpf_text); +static SOC_ENUM_SINGLE_DECL(sidetone_hpf, + WM8994_SIDETONE, 7, sidetone_hpf_text); static const char *adc_hpf_text[] = { "HiFi", "Voice 1", "Voice 2", "Voice 3" }; -static const struct soc_enum aif1adc1_hpf = - SOC_ENUM_SINGLE(WM8994_AIF1_ADC1_FILTERS, 13, 4, adc_hpf_text); +static SOC_ENUM_SINGLE_DECL(aif1adc1_hpf, + WM8994_AIF1_ADC1_FILTERS, 13, adc_hpf_text); -static const struct soc_enum aif1adc2_hpf = - SOC_ENUM_SINGLE(WM8994_AIF1_ADC2_FILTERS, 13, 4, adc_hpf_text); +static SOC_ENUM_SINGLE_DECL(aif1adc2_hpf, + WM8994_AIF1_ADC2_FILTERS, 13, adc_hpf_text); -static const struct soc_enum aif2adc_hpf = - SOC_ENUM_SINGLE(WM8994_AIF2_ADC_FILTERS, 13, 4, adc_hpf_text); +static SOC_ENUM_SINGLE_DECL(aif2adc_hpf, + WM8994_AIF2_ADC_FILTERS, 13, adc_hpf_text); static const DECLARE_TLV_DB_SCALE(aif_tlv, 0, 600, 0); static const DECLARE_TLV_DB_SCALE(digital_tlv, -7200, 75, 1); @@ -501,39 +501,39 @@ static const char *aif_chan_src_text[] = { "Left", "Right" }; -static const struct soc_enum aif1adcl_src = - SOC_ENUM_SINGLE(WM8994_AIF1_CONTROL_1, 15, 2, aif_chan_src_text); +static SOC_ENUM_SINGLE_DECL(aif1adcl_src, + WM8994_AIF1_CONTROL_1, 15, aif_chan_src_text); -static const struct soc_enum aif1adcr_src = - SOC_ENUM_SINGLE(WM8994_AIF1_CONTROL_1, 14, 2, aif_chan_src_text); +static SOC_ENUM_SINGLE_DECL(aif1adcr_src, + WM8994_AIF1_CONTROL_1, 14, aif_chan_src_text); -static const struct soc_enum aif2adcl_src = - SOC_ENUM_SINGLE(WM8994_AIF2_CONTROL_1, 15, 2, aif_chan_src_text); +static SOC_ENUM_SINGLE_DECL(aif2adcl_src, + WM8994_AIF2_CONTROL_1, 15, aif_chan_src_text); -static const struct soc_enum aif2adcr_src = - SOC_ENUM_SINGLE(WM8994_AIF2_CONTROL_1, 14, 2, aif_chan_src_text); +static SOC_ENUM_SINGLE_DECL(aif2adcr_src, + WM8994_AIF2_CONTROL_1, 14, aif_chan_src_text); -static const struct soc_enum aif1dacl_src = - SOC_ENUM_SINGLE(WM8994_AIF1_CONTROL_2, 15, 2, aif_chan_src_text); +static SOC_ENUM_SINGLE_DECL(aif1dacl_src, + WM8994_AIF1_CONTROL_2, 15, aif_chan_src_text); -static const struct soc_enum aif1dacr_src = - SOC_ENUM_SINGLE(WM8994_AIF1_CONTROL_2, 14, 2, aif_chan_src_text); +static SOC_ENUM_SINGLE_DECL(aif1dacr_src, + WM8994_AIF1_CONTROL_2, 14, aif_chan_src_text); -static const struct soc_enum aif2dacl_src = - SOC_ENUM_SINGLE(WM8994_AIF2_CONTROL_2, 15, 2, aif_chan_src_text); +static SOC_ENUM_SINGLE_DECL(aif2dacl_src, + WM8994_AIF2_CONTROL_2, 15, aif_chan_src_text); -static const struct soc_enum aif2dacr_src = - SOC_ENUM_SINGLE(WM8994_AIF2_CONTROL_2, 14, 2, aif_chan_src_text); +static SOC_ENUM_SINGLE_DECL(aif2dacr_src, + WM8994_AIF2_CONTROL_2, 14, aif_chan_src_text); static const char *osr_text[] = { "Low Power", "High Performance", }; -static const struct soc_enum dac_osr = - SOC_ENUM_SINGLE(WM8994_OVERSAMPLING, 0, 2, osr_text); +static SOC_ENUM_SINGLE_DECL(dac_osr, + WM8994_OVERSAMPLING, 0, osr_text); -static const struct soc_enum adc_osr = - SOC_ENUM_SINGLE(WM8994_OVERSAMPLING, 1, 2, osr_text); +static SOC_ENUM_SINGLE_DECL(adc_osr, + WM8994_OVERSAMPLING, 1, osr_text); static const struct snd_kcontrol_new wm8994_snd_controls[] = { SOC_DOUBLE_R_TLV("AIF1ADC1 Volume", WM8994_AIF1_ADC1_LEFT_VOLUME, @@ -690,17 +690,20 @@ static const char *wm8958_ng_text[] = { "30ms", "125ms", "250ms", "500ms", }; -static const struct soc_enum wm8958_aif1dac1_ng_hold = - SOC_ENUM_SINGLE(WM8958_AIF1_DAC1_NOISE_GATE, - WM8958_AIF1DAC1_NG_THR_SHIFT, 4, wm8958_ng_text); +static SOC_ENUM_SINGLE_DECL(wm8958_aif1dac1_ng_hold, + WM8958_AIF1_DAC1_NOISE_GATE, + WM8958_AIF1DAC1_NG_THR_SHIFT, + wm8958_ng_text); -static const struct soc_enum wm8958_aif1dac2_ng_hold = - SOC_ENUM_SINGLE(WM8958_AIF1_DAC2_NOISE_GATE, - WM8958_AIF1DAC2_NG_THR_SHIFT, 4, wm8958_ng_text); +static SOC_ENUM_SINGLE_DECL(wm8958_aif1dac2_ng_hold, + WM8958_AIF1_DAC2_NOISE_GATE, + WM8958_AIF1DAC2_NG_THR_SHIFT, + wm8958_ng_text); -static const struct soc_enum wm8958_aif2dac_ng_hold = - SOC_ENUM_SINGLE(WM8958_AIF2_DAC_NOISE_GATE, - WM8958_AIF2DAC_NG_THR_SHIFT, 4, wm8958_ng_text); +static SOC_ENUM_SINGLE_DECL(wm8958_aif2dac_ng_hold, + WM8958_AIF2_DAC_NOISE_GATE, + WM8958_AIF2DAC_NG_THR_SHIFT, + wm8958_ng_text); static const struct snd_kcontrol_new wm8958_snd_controls[] = { SOC_SINGLE_TLV("AIF3 Boost Volume", WM8958_AIF3_CONTROL_2, 10, 3, 0, aif_tlv), @@ -1341,8 +1344,8 @@ static const char *adc_mux_text[] = { "DMIC", }; -static const struct soc_enum adc_enum = - SOC_ENUM_SINGLE(0, 0, 2, adc_mux_text); +static SOC_ENUM_SINGLE_DECL(adc_enum, + 0, 0, adc_mux_text); static const struct snd_kcontrol_new adcl_mux = SOC_DAPM_ENUM_VIRT("ADCL Mux", adc_enum); @@ -1478,14 +1481,14 @@ static const char *sidetone_text[] = { "ADC/DMIC1", "DMIC2", }; -static const struct soc_enum sidetone1_enum = - SOC_ENUM_SINGLE(WM8994_SIDETONE, 0, 2, sidetone_text); +static SOC_ENUM_SINGLE_DECL(sidetone1_enum, + WM8994_SIDETONE, 0, sidetone_text); static const struct snd_kcontrol_new sidetone1_mux = SOC_DAPM_ENUM("Left Sidetone Mux", sidetone1_enum); -static const struct soc_enum sidetone2_enum = - SOC_ENUM_SINGLE(WM8994_SIDETONE, 1, 2, sidetone_text); +static SOC_ENUM_SINGLE_DECL(sidetone2_enum, + WM8994_SIDETONE, 1, sidetone_text); static const struct snd_kcontrol_new sidetone2_mux = SOC_DAPM_ENUM("Right Sidetone Mux", sidetone2_enum); @@ -1498,22 +1501,24 @@ static const char *loopback_text[] = { "None", "ADCDAT", }; -static const struct soc_enum aif1_loopback_enum = - SOC_ENUM_SINGLE(WM8994_AIF1_CONTROL_2, WM8994_AIF1_LOOPBACK_SHIFT, 2, - loopback_text); +static SOC_ENUM_SINGLE_DECL(aif1_loopback_enum, + WM8994_AIF1_CONTROL_2, + WM8994_AIF1_LOOPBACK_SHIFT, + loopback_text); static const struct snd_kcontrol_new aif1_loopback = SOC_DAPM_ENUM("AIF1 Loopback", aif1_loopback_enum); -static const struct soc_enum aif2_loopback_enum = - SOC_ENUM_SINGLE(WM8994_AIF2_CONTROL_2, WM8994_AIF2_LOOPBACK_SHIFT, 2, - loopback_text); +static SOC_ENUM_SINGLE_DECL(aif2_loopback_enum, + WM8994_AIF2_CONTROL_2, + WM8994_AIF2_LOOPBACK_SHIFT, + loopback_text); static const struct snd_kcontrol_new aif2_loopback = SOC_DAPM_ENUM("AIF2 Loopback", aif2_loopback_enum); -static const struct soc_enum aif1dac_enum = - SOC_ENUM_SINGLE(WM8994_POWER_MANAGEMENT_6, 0, 2, aif1dac_text); +static SOC_ENUM_SINGLE_DECL(aif1dac_enum, + WM8994_POWER_MANAGEMENT_6, 0, aif1dac_text); static const struct snd_kcontrol_new aif1dac_mux = SOC_DAPM_ENUM("AIF1DAC Mux", aif1dac_enum); @@ -1522,8 +1527,8 @@ static const char *aif2dac_text[] = { "AIF2DACDAT", "AIF3DACDAT", }; -static const struct soc_enum aif2dac_enum = - SOC_ENUM_SINGLE(WM8994_POWER_MANAGEMENT_6, 1, 2, aif2dac_text); +static SOC_ENUM_SINGLE_DECL(aif2dac_enum, + WM8994_POWER_MANAGEMENT_6, 1, aif2dac_text); static const struct snd_kcontrol_new aif2dac_mux = SOC_DAPM_ENUM("AIF2DAC Mux", aif2dac_enum); @@ -1532,8 +1537,8 @@ static const char *aif2adc_text[] = { "AIF2ADCDAT", "AIF3DACDAT", }; -static const struct soc_enum aif2adc_enum = - SOC_ENUM_SINGLE(WM8994_POWER_MANAGEMENT_6, 2, 2, aif2adc_text); +static SOC_ENUM_SINGLE_DECL(aif2adc_enum, + WM8994_POWER_MANAGEMENT_6, 2, aif2adc_text); static const struct snd_kcontrol_new aif2adc_mux = SOC_DAPM_ENUM("AIF2ADC Mux", aif2adc_enum); @@ -1542,14 +1547,14 @@ static const char *aif3adc_text[] = { "AIF1ADCDAT", "AIF2ADCDAT", "AIF2DACDAT", "Mono PCM", }; -static const struct soc_enum wm8994_aif3adc_enum = - SOC_ENUM_SINGLE(WM8994_POWER_MANAGEMENT_6, 3, 3, aif3adc_text); +static SOC_ENUM_SINGLE_DECL(wm8994_aif3adc_enum, + WM8994_POWER_MANAGEMENT_6, 3, aif3adc_text); static const struct snd_kcontrol_new wm8994_aif3adc_mux = SOC_DAPM_ENUM("AIF3ADC Mux", wm8994_aif3adc_enum); -static const struct soc_enum wm8958_aif3adc_enum = - SOC_ENUM_SINGLE(WM8994_POWER_MANAGEMENT_6, 3, 4, aif3adc_text); +static SOC_ENUM_SINGLE_DECL(wm8958_aif3adc_enum, + WM8994_POWER_MANAGEMENT_6, 3, aif3adc_text); static const struct snd_kcontrol_new wm8958_aif3adc_mux = SOC_DAPM_ENUM("AIF3ADC Mux", wm8958_aif3adc_enum); @@ -1558,8 +1563,8 @@ static const char *mono_pcm_out_text[] = { "None", "AIF2ADCL", "AIF2ADCR", }; -static const struct soc_enum mono_pcm_out_enum = - SOC_ENUM_SINGLE(WM8994_POWER_MANAGEMENT_6, 9, 3, mono_pcm_out_text); +static SOC_ENUM_SINGLE_DECL(mono_pcm_out_enum, + WM8994_POWER_MANAGEMENT_6, 9, mono_pcm_out_text); static const struct snd_kcontrol_new mono_pcm_out_mux = SOC_DAPM_ENUM("Mono PCM Out Mux", mono_pcm_out_enum); @@ -1569,14 +1574,14 @@ static const char *aif2dac_src_text[] = { }; /* Note that these two control shouldn't be simultaneously switched to AIF3 */ -static const struct soc_enum aif2dacl_src_enum = - SOC_ENUM_SINGLE(WM8994_POWER_MANAGEMENT_6, 7, 2, aif2dac_src_text); +static SOC_ENUM_SINGLE_DECL(aif2dacl_src_enum, + WM8994_POWER_MANAGEMENT_6, 7, aif2dac_src_text); static const struct snd_kcontrol_new aif2dacl_src_mux = SOC_DAPM_ENUM("AIF2DACL Mux", aif2dacl_src_enum); -static const struct soc_enum aif2dacr_src_enum = - SOC_ENUM_SINGLE(WM8994_POWER_MANAGEMENT_6, 8, 2, aif2dac_src_text); +static SOC_ENUM_SINGLE_DECL(aif2dacr_src_enum, + WM8994_POWER_MANAGEMENT_6, 8, aif2dac_src_text); static const struct snd_kcontrol_new aif2dacr_src_mux = SOC_DAPM_ENUM("AIF2DACR Mux", aif2dacr_src_enum); -- cgit v1.2.1 From 901bb6c55d2a7ef66bc0bdf0ba410c417f16b7cf Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Feb 2014 09:56:40 +0100 Subject: ASoC: ad1980: Fix wrong number of items for capture source The number of capture sources is 8, not 7. Use SOC_ENUM_DOUBLE_DECL() macro and it's automatically fixed. Signed-off-by: Takashi Iwai Acked-by: Liam Girdwood Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/ad1980.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c index 7257a8885f42..34d965a4a040 100644 --- a/sound/soc/codecs/ad1980.c +++ b/sound/soc/codecs/ad1980.c @@ -57,8 +57,8 @@ static const u16 ad1980_reg[] = { static const char *ad1980_rec_sel[] = {"Mic", "CD", "NC", "AUX", "Line", "Stereo Mix", "Mono Mix", "Phone"}; -static const struct soc_enum ad1980_cap_src = - SOC_ENUM_DOUBLE(AC97_REC_SEL, 8, 0, 7, ad1980_rec_sel); +static SOC_ENUM_DOUBLE_DECL(ad1980_cap_src, + AC97_REC_SEL, 8, 0, ad1980_rec_sel); static const struct snd_kcontrol_new ad1980_snd_ac97_controls[] = { SOC_DOUBLE("Master Playback Volume", AC97_MASTER, 8, 0, 31, 1), -- cgit v1.2.1 From cdbb492557112bddad26f0ba4ba9ba87d919ebfc Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Feb 2014 10:07:56 +0100 Subject: ASoC: isabelle: Fix the wrong number of items in enum ctls isabelle codec driver has a few places wrongly defining the number of enum items. Use SOC_ENUM_SINGLE_DECL() macro and they are automatically fixed. Signed-off-by: Takashi Iwai Acked-by: Liam Girdwood Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/isabelle.c | 52 ++++++++++++++++++++++++++------------------- 1 file changed, 30 insertions(+), 22 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/isabelle.c b/sound/soc/codecs/isabelle.c index 5839048ec467..cb736ddc446d 100644 --- a/sound/soc/codecs/isabelle.c +++ b/sound/soc/codecs/isabelle.c @@ -140,13 +140,17 @@ static const char *isabelle_rx1_texts[] = {"VRX1", "ARX1"}; static const char *isabelle_rx2_texts[] = {"VRX2", "ARX2"}; static const struct soc_enum isabelle_rx1_enum[] = { - SOC_ENUM_SINGLE(ISABELLE_VOICE_HPF_CFG_REG, 3, 1, isabelle_rx1_texts), - SOC_ENUM_SINGLE(ISABELLE_AUDIO_HPF_CFG_REG, 5, 1, isabelle_rx1_texts), + SOC_ENUM_SINGLE(ISABELLE_VOICE_HPF_CFG_REG, 3, + ARRAY_SIZE(isabelle_rx1_texts), isabelle_rx1_texts), + SOC_ENUM_SINGLE(ISABELLE_AUDIO_HPF_CFG_REG, 5, + ARRAY_SIZE(isabelle_rx1_texts), isabelle_rx1_texts), }; static const struct soc_enum isabelle_rx2_enum[] = { - SOC_ENUM_SINGLE(ISABELLE_VOICE_HPF_CFG_REG, 2, 1, isabelle_rx2_texts), - SOC_ENUM_SINGLE(ISABELLE_AUDIO_HPF_CFG_REG, 4, 1, isabelle_rx2_texts), + SOC_ENUM_SINGLE(ISABELLE_VOICE_HPF_CFG_REG, 2, + ARRAY_SIZE(isabelle_rx2_texts), isabelle_rx2_texts), + SOC_ENUM_SINGLE(ISABELLE_AUDIO_HPF_CFG_REG, 4, + ARRAY_SIZE(isabelle_rx2_texts), isabelle_rx2_texts), }; /* Headset DAC playback switches */ @@ -161,13 +165,17 @@ static const char *isabelle_atx_texts[] = {"AMIC1", "DMIC"}; static const char *isabelle_vtx_texts[] = {"AMIC2", "DMIC"}; static const struct soc_enum isabelle_atx_enum[] = { - SOC_ENUM_SINGLE(ISABELLE_AMIC_CFG_REG, 7, 1, isabelle_atx_texts), - SOC_ENUM_SINGLE(ISABELLE_DMIC_CFG_REG, 0, 1, isabelle_atx_texts), + SOC_ENUM_SINGLE(ISABELLE_AMIC_CFG_REG, 7, + ARRAY_SIZE(isabelle_atx_texts), isabelle_atx_texts), + SOC_ENUM_SINGLE(ISABELLE_DMIC_CFG_REG, 0, + ARRAY_SIZE(isabelle_atx_texts), isabelle_atx_texts), }; static const struct soc_enum isabelle_vtx_enum[] = { - SOC_ENUM_SINGLE(ISABELLE_AMIC_CFG_REG, 6, 1, isabelle_vtx_texts), - SOC_ENUM_SINGLE(ISABELLE_DMIC_CFG_REG, 0, 1, isabelle_vtx_texts), + SOC_ENUM_SINGLE(ISABELLE_AMIC_CFG_REG, 6, + ARRAY_SIZE(isabelle_vtx_texts), isabelle_vtx_texts), + SOC_ENUM_SINGLE(ISABELLE_DMIC_CFG_REG, 0, + ARRAY_SIZE(isabelle_vtx_texts), isabelle_vtx_texts), }; static const struct snd_kcontrol_new atx_mux_controls = @@ -183,17 +191,13 @@ static const char *isabelle_amic1_texts[] = { /* Left analog microphone selection */ static const char *isabelle_amic2_texts[] = {"Sub Mic", "Aux/FM Right"}; -static const struct soc_enum isabelle_amic1_enum[] = { - SOC_ENUM_SINGLE(ISABELLE_AMIC_CFG_REG, 5, - ARRAY_SIZE(isabelle_amic1_texts), - isabelle_amic1_texts), -}; +static SOC_ENUM_SINGLE_DECL(isabelle_amic1_enum, + ISABELLE_AMIC_CFG_REG, 5, + isabelle_amic1_texts); -static const struct soc_enum isabelle_amic2_enum[] = { - SOC_ENUM_SINGLE(ISABELLE_AMIC_CFG_REG, 4, - ARRAY_SIZE(isabelle_amic2_texts), - isabelle_amic2_texts), -}; +static SOC_ENUM_SINGLE_DECL(isabelle_amic2_enum, + ISABELLE_AMIC_CFG_REG, 4, + isabelle_amic2_texts); static const struct snd_kcontrol_new amic1_control = SOC_DAPM_ENUM("Route", isabelle_amic1_enum); @@ -206,16 +210,20 @@ static const char *isabelle_st_audio_texts[] = {"ATX1", "ATX2"}; static const char *isabelle_st_voice_texts[] = {"VTX1", "VTX2"}; static const struct soc_enum isabelle_st_audio_enum[] = { - SOC_ENUM_SINGLE(ISABELLE_ATX_STPGA1_CFG_REG, 7, 1, + SOC_ENUM_SINGLE(ISABELLE_ATX_STPGA1_CFG_REG, 7, + ARRAY_SIZE(isabelle_st_audio_texts), isabelle_st_audio_texts), - SOC_ENUM_SINGLE(ISABELLE_ATX_STPGA2_CFG_REG, 7, 1, + SOC_ENUM_SINGLE(ISABELLE_ATX_STPGA2_CFG_REG, 7, + ARRAY_SIZE(isabelle_st_audio_texts), isabelle_st_audio_texts), }; static const struct soc_enum isabelle_st_voice_enum[] = { - SOC_ENUM_SINGLE(ISABELLE_VTX_STPGA1_CFG_REG, 7, 1, + SOC_ENUM_SINGLE(ISABELLE_VTX_STPGA1_CFG_REG, 7, + ARRAY_SIZE(isabelle_st_voice_texts), isabelle_st_voice_texts), - SOC_ENUM_SINGLE(ISABELLE_VTX2_STPGA2_CFG_REG, 7, 1, + SOC_ENUM_SINGLE(ISABELLE_VTX2_STPGA2_CFG_REG, 7, + ARRAY_SIZE(isabelle_st_voice_texts), isabelle_st_voice_texts), }; -- cgit v1.2.1 From 898b48eb88bff3a7a49590a08bee546c2d26bd91 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Feb 2014 09:34:44 +0100 Subject: ASoC: wm8400: Fix the wrong number of enum items wm8400 codec driver has a few places wrongly defining the number of enum items. Use SOC_ENUM_SINGLE_DECL() macro and they are automatically fixed. Signed-off-by: Takashi Iwai Acked-by: Liam Girdwood Acked-by: Charles Keepax Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/wm8400.c | 34 ++++++++++++++++++++-------------- 1 file changed, 20 insertions(+), 14 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c index 48dc7d2fee36..6d684d934f4d 100644 --- a/sound/soc/codecs/wm8400.c +++ b/sound/soc/codecs/wm8400.c @@ -117,19 +117,23 @@ static int wm8400_outpga_put_volsw_vu(struct snd_kcontrol *kcontrol, static const char *wm8400_digital_sidetone[] = {"None", "Left ADC", "Right ADC", "Reserved"}; -static const struct soc_enum wm8400_left_digital_sidetone_enum = -SOC_ENUM_SINGLE(WM8400_DIGITAL_SIDE_TONE, - WM8400_ADC_TO_DACL_SHIFT, 2, wm8400_digital_sidetone); +static SOC_ENUM_SINGLE_DECL(wm8400_left_digital_sidetone_enum, + WM8400_DIGITAL_SIDE_TONE, + WM8400_ADC_TO_DACL_SHIFT, + wm8400_digital_sidetone); -static const struct soc_enum wm8400_right_digital_sidetone_enum = -SOC_ENUM_SINGLE(WM8400_DIGITAL_SIDE_TONE, - WM8400_ADC_TO_DACR_SHIFT, 2, wm8400_digital_sidetone); +static SOC_ENUM_SINGLE_DECL(wm8400_right_digital_sidetone_enum, + WM8400_DIGITAL_SIDE_TONE, + WM8400_ADC_TO_DACR_SHIFT, + wm8400_digital_sidetone); static const char *wm8400_adcmode[] = {"Hi-fi mode", "Voice mode 1", "Voice mode 2", "Voice mode 3"}; -static const struct soc_enum wm8400_right_adcmode_enum = -SOC_ENUM_SINGLE(WM8400_ADC_CTRL, WM8400_ADC_HPF_CUT_SHIFT, 3, wm8400_adcmode); +static SOC_ENUM_SINGLE_DECL(wm8400_right_adcmode_enum, + WM8400_ADC_CTRL, + WM8400_ADC_HPF_CUT_SHIFT, + wm8400_adcmode); static const struct snd_kcontrol_new wm8400_snd_controls[] = { /* INMIXL */ @@ -422,9 +426,10 @@ SOC_DAPM_SINGLE("RINPGA34 Switch", WM8400_INPUT_MIXER3, WM8400_L34MNB_SHIFT, static const char *wm8400_ainlmux[] = {"INMIXL Mix", "RXVOICE Mix", "DIFFINL Mix"}; -static const struct soc_enum wm8400_ainlmux_enum = -SOC_ENUM_SINGLE( WM8400_INPUT_MIXER1, WM8400_AINLMODE_SHIFT, - ARRAY_SIZE(wm8400_ainlmux), wm8400_ainlmux); +static SOC_ENUM_SINGLE_DECL(wm8400_ainlmux_enum, + WM8400_INPUT_MIXER1, + WM8400_AINLMODE_SHIFT, + wm8400_ainlmux); static const struct snd_kcontrol_new wm8400_dapm_ainlmux_controls = SOC_DAPM_ENUM("Route", wm8400_ainlmux_enum); @@ -435,9 +440,10 @@ SOC_DAPM_ENUM("Route", wm8400_ainlmux_enum); static const char *wm8400_ainrmux[] = {"INMIXR Mix", "RXVOICE Mix", "DIFFINR Mix"}; -static const struct soc_enum wm8400_ainrmux_enum = -SOC_ENUM_SINGLE( WM8400_INPUT_MIXER1, WM8400_AINRMODE_SHIFT, - ARRAY_SIZE(wm8400_ainrmux), wm8400_ainrmux); +static SOC_ENUM_SINGLE_DECL(wm8400_ainrmux_enum, + WM8400_INPUT_MIXER1, + WM8400_AINRMODE_SHIFT, + wm8400_ainrmux); static const struct snd_kcontrol_new wm8400_dapm_ainrmux_controls = SOC_DAPM_ENUM("Route", wm8400_ainrmux_enum); -- cgit v1.2.1 From 1de7ca5e844866f56bebb2fc47fa18e090677e88 Mon Sep 17 00:00:00 2001 From: Hui Wang Date: Thu, 20 Feb 2014 11:47:21 +0800 Subject: ALSA: hda - Enable front audio jacks on one HP desktop model The front headphone and mic jackes on a HP desktop model (Vendor Id: 0x111d76c7 Subsystem Id: 0x103c2b17) can not work, the codec on this machine has 8 physical ports, 6 of them are routed to rear jackes and all of them work very well, while the remaining 2 ports are routed to front headphone and mic jackes, but the corresponding pin complex node are not defined correctly. After apply this fix, the front audio jackes can work very well. [trivial fix of enum definition by tiwai] BugLink: https://bugs.launchpad.net/bugs/1282369 Cc: David Henningsson Tested-by: Gerald Yang Cc: stable@vger.kernel.org Signed-off-by: Hui Wang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 7311badf6a94..a2f11bf8155c 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -83,6 +83,7 @@ enum { STAC_DELL_M6_BOTH, STAC_DELL_EQ, STAC_ALIENWARE_M17X, + STAC_92HD89XX_HP_FRONT_JACK, STAC_92HD73XX_MODELS }; @@ -1795,6 +1796,12 @@ static const struct hda_pintbl intel_dg45id_pin_configs[] = { {} }; +static const struct hda_pintbl stac92hd89xx_hp_front_jack_pin_configs[] = { + { 0x0a, 0x02214030 }, + { 0x0b, 0x02A19010 }, + {} +}; + static void stac92hd73xx_fixup_ref(struct hda_codec *codec, const struct hda_fixup *fix, int action) { @@ -1913,6 +1920,10 @@ static const struct hda_fixup stac92hd73xx_fixups[] = { [STAC_92HD73XX_NO_JD] = { .type = HDA_FIXUP_FUNC, .v.func = stac92hd73xx_fixup_no_jd, + }, + [STAC_92HD89XX_HP_FRONT_JACK] = { + .type = HDA_FIXUP_PINS, + .v.pins = stac92hd89xx_hp_front_jack_pin_configs, } }; @@ -1973,6 +1984,8 @@ static const struct snd_pci_quirk stac92hd73xx_fixup_tbl[] = { "Alienware M17x", STAC_ALIENWARE_M17X), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0490, "Alienware M17x R3", STAC_DELL_EQ), + SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x2b17, + "unknown HP", STAC_92HD89XX_HP_FRONT_JACK), {} /* terminator */ }; -- cgit v1.2.1 From 113911006442a36c2b4669faf1699d9042ef80ab Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Tue, 18 Feb 2014 15:22:14 +0000 Subject: ASoC: dapm: Add locking to snd_soc_dapm_xxxx_pin functions The snd_soc_dapm_xxxx_pin all require the dapm_mutex to be held when they are called as they edit the dirty list, however very few of the callers do so. This patch adds unlocked versions of all the functions replacing the existing implementations with one that holds the lock internally. We also fix up the places where the lock was actually held on the caller side. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/soc-dapm.c | 133 +++++++++++++++++++++++++++++++++++++++++++++++---- 1 file changed, 123 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index fd39cd2827d7..b9dc6acbba8c 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3210,15 +3210,11 @@ int snd_soc_dapm_put_pin_switch(struct snd_kcontrol *kcontrol, struct snd_soc_card *card = snd_kcontrol_chip(kcontrol); const char *pin = (const char *)kcontrol->private_value; - mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); - if (ucontrol->value.integer.value[0]) snd_soc_dapm_enable_pin(&card->dapm, pin); else snd_soc_dapm_disable_pin(&card->dapm, pin); - mutex_unlock(&card->dapm_mutex); - snd_soc_dapm_sync(&card->dapm); return 0; } @@ -3766,6 +3762,26 @@ void snd_soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd, int stream, mutex_unlock(&card->dapm_mutex); } +/** + * snd_soc_dapm_enable_pin_unlocked - enable pin. + * @dapm: DAPM context + * @pin: pin name + * + * Enables input/output pin and its parents or children widgets iff there is + * a valid audio route and active audio stream. + * + * Requires external locking. + * + * NOTE: snd_soc_dapm_sync() needs to be called after this for DAPM to + * do any widget power switching. + */ +int snd_soc_dapm_enable_pin_unlocked(struct snd_soc_dapm_context *dapm, + const char *pin) +{ + return snd_soc_dapm_set_pin(dapm, pin, 1); +} +EXPORT_SYMBOL_GPL(snd_soc_dapm_enable_pin_unlocked); + /** * snd_soc_dapm_enable_pin - enable pin. * @dapm: DAPM context @@ -3773,17 +3789,26 @@ void snd_soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd, int stream, * * Enables input/output pin and its parents or children widgets iff there is * a valid audio route and active audio stream. + * * NOTE: snd_soc_dapm_sync() needs to be called after this for DAPM to * do any widget power switching. */ int snd_soc_dapm_enable_pin(struct snd_soc_dapm_context *dapm, const char *pin) { - return snd_soc_dapm_set_pin(dapm, pin, 1); + int ret; + + mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); + + ret = snd_soc_dapm_set_pin(dapm, pin, 1); + + mutex_unlock(&dapm->card->dapm_mutex); + + return ret; } EXPORT_SYMBOL_GPL(snd_soc_dapm_enable_pin); /** - * snd_soc_dapm_force_enable_pin - force a pin to be enabled + * snd_soc_dapm_force_enable_pin_unlocked - force a pin to be enabled * @dapm: DAPM context * @pin: pin name * @@ -3791,11 +3816,13 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_enable_pin); * intended for use with microphone bias supplies used in microphone * jack detection. * + * Requires external locking. + * * NOTE: snd_soc_dapm_sync() needs to be called after this for DAPM to * do any widget power switching. */ -int snd_soc_dapm_force_enable_pin(struct snd_soc_dapm_context *dapm, - const char *pin) +int snd_soc_dapm_force_enable_pin_unlocked(struct snd_soc_dapm_context *dapm, + const char *pin) { struct snd_soc_dapm_widget *w = dapm_find_widget(dapm, pin, true); @@ -3811,24 +3838,102 @@ int snd_soc_dapm_force_enable_pin(struct snd_soc_dapm_context *dapm, return 0; } +EXPORT_SYMBOL_GPL(snd_soc_dapm_force_enable_pin_unlocked); + +/** + * snd_soc_dapm_force_enable_pin - force a pin to be enabled + * @dapm: DAPM context + * @pin: pin name + * + * Enables input/output pin regardless of any other state. This is + * intended for use with microphone bias supplies used in microphone + * jack detection. + * + * NOTE: snd_soc_dapm_sync() needs to be called after this for DAPM to + * do any widget power switching. + */ +int snd_soc_dapm_force_enable_pin(struct snd_soc_dapm_context *dapm, + const char *pin) +{ + int ret; + + mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); + + ret = snd_soc_dapm_force_enable_pin_unlocked(dapm, pin); + + mutex_unlock(&dapm->card->dapm_mutex); + + return ret; +} EXPORT_SYMBOL_GPL(snd_soc_dapm_force_enable_pin); +/** + * snd_soc_dapm_disable_pin_unlocked - disable pin. + * @dapm: DAPM context + * @pin: pin name + * + * Disables input/output pin and its parents or children widgets. + * + * Requires external locking. + * + * NOTE: snd_soc_dapm_sync() needs to be called after this for DAPM to + * do any widget power switching. + */ +int snd_soc_dapm_disable_pin_unlocked(struct snd_soc_dapm_context *dapm, + const char *pin) +{ + return snd_soc_dapm_set_pin(dapm, pin, 0); +} +EXPORT_SYMBOL_GPL(snd_soc_dapm_disable_pin_unlocked); + /** * snd_soc_dapm_disable_pin - disable pin. * @dapm: DAPM context * @pin: pin name * * Disables input/output pin and its parents or children widgets. + * * NOTE: snd_soc_dapm_sync() needs to be called after this for DAPM to * do any widget power switching. */ int snd_soc_dapm_disable_pin(struct snd_soc_dapm_context *dapm, const char *pin) { - return snd_soc_dapm_set_pin(dapm, pin, 0); + int ret; + + mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); + + ret = snd_soc_dapm_set_pin(dapm, pin, 0); + + mutex_unlock(&dapm->card->dapm_mutex); + + return ret; } EXPORT_SYMBOL_GPL(snd_soc_dapm_disable_pin); +/** + * snd_soc_dapm_nc_pin_unlocked - permanently disable pin. + * @dapm: DAPM context + * @pin: pin name + * + * Marks the specified pin as being not connected, disabling it along + * any parent or child widgets. At present this is identical to + * snd_soc_dapm_disable_pin() but in future it will be extended to do + * additional things such as disabling controls which only affect + * paths through the pin. + * + * Requires external locking. + * + * NOTE: snd_soc_dapm_sync() needs to be called after this for DAPM to + * do any widget power switching. + */ +int snd_soc_dapm_nc_pin_unlocked(struct snd_soc_dapm_context *dapm, + const char *pin) +{ + return snd_soc_dapm_set_pin(dapm, pin, 0); +} +EXPORT_SYMBOL_GPL(snd_soc_dapm_nc_pin_unlocked); + /** * snd_soc_dapm_nc_pin - permanently disable pin. * @dapm: DAPM context @@ -3845,7 +3950,15 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_disable_pin); */ int snd_soc_dapm_nc_pin(struct snd_soc_dapm_context *dapm, const char *pin) { - return snd_soc_dapm_set_pin(dapm, pin, 0); + int ret; + + mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); + + ret = snd_soc_dapm_set_pin(dapm, pin, 0); + + mutex_unlock(&dapm->card->dapm_mutex); + + return ret; } EXPORT_SYMBOL_GPL(snd_soc_dapm_nc_pin); -- cgit v1.2.1 From c60666bd2224a32606364fdbc620e4cb9446a1d1 Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Fri, 21 Feb 2014 16:23:35 +0800 Subject: ALSA: hda/realtek - Add more entry for enable HP mute led More HP machine need mute led support. Signed-off-by: Kailang Yang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 47 +++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 47 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 6eb903cc6237..0b4ea6c7eca9 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4319,6 +4319,53 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x1973, "HP Pavilion", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x1983, "HP Pavilion", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x218b, "HP", ALC269_FIXUP_LIMIT_INT_MIC_BOOST_MUTE_LED), + /* ALC282 */ + SND_PCI_QUIRK(0x103c, 0x220f, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x2213, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x2266, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x2267, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x2268, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x2269, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x226a, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x226b, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x227a, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x227b, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x229e, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x22a0, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x22b2, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x22b7, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x22bf, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x22c0, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x22c1, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x22c2, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x22cd, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x22ce, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x22cf, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x22d0, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + /* ALC290 */ + SND_PCI_QUIRK(0x103c, 0x2260, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x2261, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x2262, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x2263, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x2264, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x2265, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x227d, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x227e, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x227f, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x2280, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x2281, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x2282, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x2289, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x228a, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x228b, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x228c, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x228e, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x22c5, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x22c6, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x22c7, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x22c8, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x22c3, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x22c4, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK_VENDOR(0x103c, "HP", ALC269_FIXUP_HP_MUTE_LED), SND_PCI_QUIRK(0x1043, 0x103f, "ASUS TX300", ALC282_FIXUP_ASUS_TX300), SND_PCI_QUIRK(0x1043, 0x106d, "Asus K53BE", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), -- cgit v1.2.1 From 70ff00f82a6af0ff68f8f7b411738634ce2f20d0 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sat, 22 Feb 2014 18:27:17 +0100 Subject: ASoC: sta32x: Fix cache sync codec->control_data contains a pointer to the regmap struct of the device, not to the device private data. Use snd_soc_codec_get_drvdata() instead. The issue was introduced in commit 29fdf4fbbe ("ASoC: sta32x: Convert to regmap"). Fixes: 29fdf4fbbe (ASoC: sta32x: Convert to regmap) Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/sta32x.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c index 42c5b458b046..ea78c172538c 100644 --- a/sound/soc/codecs/sta32x.c +++ b/sound/soc/codecs/sta32x.c @@ -331,7 +331,7 @@ static int sta32x_sync_coef_shadow(struct snd_soc_codec *codec) static int sta32x_cache_sync(struct snd_soc_codec *codec) { - struct sta32x_priv *sta32x = codec->control_data; + struct sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec); unsigned int mute; int rc; -- cgit v1.2.1 From 548da08fc1e245faf9b0d7c41ecd8e07984fc332 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sat, 22 Feb 2014 18:30:13 +0100 Subject: ASoC: wm8958-dsp: Fix firmware block loading The codec->control_data contains a pointer to the device's regmap struct. But wm8994_bulk_write() expects a pointer to the parent wm8998 device. The issue was introduced in commit d9a7666f ("ASoC: Remove ASoC-specific WM8994 I/O code"). Fixes: d9a7666f ("ASoC: Remove ASoC-specific WM8994 I/O code") Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/wm8958-dsp2.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8958-dsp2.c b/sound/soc/codecs/wm8958-dsp2.c index b7488f190d2b..d4248e00160e 100644 --- a/sound/soc/codecs/wm8958-dsp2.c +++ b/sound/soc/codecs/wm8958-dsp2.c @@ -153,7 +153,7 @@ static int wm8958_dsp2_fw(struct snd_soc_codec *codec, const char *name, data32 &= 0xffffff; - wm8994_bulk_write(codec->control_data, + wm8994_bulk_write(wm8994->wm8994, data32 & 0xffffff, block_len / 2, (void *)(data + 8)); -- cgit v1.2.1 From 37c367ecdb9a01c9acc980e6e17913570a1788a7 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 24 Feb 2014 15:23:10 +0100 Subject: ALSA: hda - Add a fixup for HP Folio 13 mute LED HP Folio 13 may have a broken BIOS that doesn't set up the mute LED GPIO properly, and the driver guesses it wrongly, too. Add a new fixup entry for setting the GPIO pin statically for this laptop. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=70991 Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 20 ++++++++++++++++++++ 1 file changed, 20 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index a2f11bf8155c..3bc29c9b2529 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -98,6 +98,7 @@ enum { STAC_92HD83XXX_HP_LED, STAC_92HD83XXX_HP_INV_LED, STAC_92HD83XXX_HP_MIC_LED, + STAC_HP_LED_GPIO10, STAC_92HD83XXX_HEADSET_JACK, STAC_92HD83XXX_HP, STAC_HP_ENVY_BASS, @@ -2130,6 +2131,17 @@ static void stac92hd83xxx_fixup_hp_mic_led(struct hda_codec *codec, } } +static void stac92hd83xxx_fixup_hp_led_gpio10(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct sigmatel_spec *spec = codec->spec; + + if (action == HDA_FIXUP_ACT_PRE_PROBE) { + spec->gpio_led = 0x10; /* GPIO4 */ + spec->default_polarity = 0; + } +} + static void stac92hd83xxx_fixup_headset_jack(struct hda_codec *codec, const struct hda_fixup *fix, int action) { @@ -2624,6 +2636,12 @@ static const struct hda_fixup stac92hd83xxx_fixups[] = { .chained = true, .chain_id = STAC_92HD83XXX_HP, }, + [STAC_HP_LED_GPIO10] = { + .type = HDA_FIXUP_FUNC, + .v.func = stac92hd83xxx_fixup_hp_led_gpio10, + .chained = true, + .chain_id = STAC_92HD83XXX_HP, + }, [STAC_92HD83XXX_HEADSET_JACK] = { .type = HDA_FIXUP_FUNC, .v.func = stac92hd83xxx_fixup_headset_jack, @@ -2702,6 +2720,8 @@ static const struct snd_pci_quirk stac92hd83xxx_fixup_tbl[] = { "HP", STAC_92HD83XXX_HP_cNB11_INTQUAD), SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x1888, "HP Envy Spectre", STAC_HP_ENVY_BASS), + SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x1899, + "HP Folio 13", STAC_HP_LED_GPIO10), SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x18df, "HP Folio", STAC_HP_BNB13_EQ), SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x18F8, -- cgit v1.2.1 From fce0a0c72618e021e29ed4e051ce6b42b218c5e6 Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Wed, 26 Feb 2014 15:23:19 +0800 Subject: ALSA: hda/realtek - Add more entry for enable HP mute led I lost this SSID. Add it into the fixup table. Signed-off-by: Kailang Yang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 0b4ea6c7eca9..ec304f3ae3b4 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4359,6 +4359,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x228a, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x228b, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x228c, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x228d, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x228e, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x22c5, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x22c6, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), -- cgit v1.2.1 From 75306820248e26d15d84acf4e297b9fb27dd3bb2 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 24 Feb 2014 11:59:14 +0900 Subject: ASoC: da732x: Mark DC offset control registers volatile The driver reads from the DC offset control registers during callibration but since the registers are marked as volatile and there is a register cache the values will not be read from the hardware after the first reading rendering the callibration ineffective. It appears that the driver was originally written for the ASoC level register I/O code but converted to regmap prior to merge and this issue was missed during the conversion as the framework level volatile register functionality was not being used. Signed-off-by: Mark Brown Acked-by: Adam Thomson Cc: stable@vger.kernel.org --- sound/soc/codecs/da732x.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/da732x.c b/sound/soc/codecs/da732x.c index f295b6569910..f4d965ebc29e 100644 --- a/sound/soc/codecs/da732x.c +++ b/sound/soc/codecs/da732x.c @@ -1268,11 +1268,23 @@ static struct snd_soc_dai_driver da732x_dai[] = { }, }; +static bool da732x_volatile(struct device *dev, unsigned int reg) +{ + switch (reg) { + case DA732X_REG_HPL_DAC_OFF_CNTL: + case DA732X_REG_HPR_DAC_OFF_CNTL: + return true; + default: + return false; + } +} + static const struct regmap_config da732x_regmap = { .reg_bits = 8, .val_bits = 8, .max_register = DA732X_MAX_REG, + .volatile_reg = da732x_volatile, .reg_defaults = da732x_reg_cache, .num_reg_defaults = ARRAY_SIZE(da732x_reg_cache), .cache_type = REGCACHE_RBTREE, -- cgit v1.2.1 From b3619b288b621e63f66908045f48495869a996a6 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 27 Feb 2014 07:41:32 +0100 Subject: ASoC: sta32x: Fix wrong enum for limiter2 release rate MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit There is a typo in the Limiter2 Release Rate control, a wrong enum for Limiter1 is assigned. It must point to Limiter2. Spotted by a compile warning: In file included from sound/soc/codecs/sta32x.c:34:0: sound/soc/codecs/sta32x.c:223:29: warning: ‘sta32x_limiter2_release_rate_enum’ defined but not used [-Wunused-variable] static SOC_ENUM_SINGLE_DECL(sta32x_limiter2_release_rate_enum, ^ include/sound/soc.h:275:18: note: in definition of macro ‘SOC_ENUM_DOUBLE_DECL’ struct soc_enum name = SOC_ENUM_DOUBLE(xreg, xshift_l, xshift_r, \ ^ sound/soc/codecs/sta32x.c:223:8: note: in expansion of macro ‘SOC_ENUM_SINGLE_DECL’ static SOC_ENUM_SINGLE_DECL(sta32x_limiter2_release_rate_enum, ^ Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown Cc: --- sound/soc/codecs/sta32x.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c index ea78c172538c..2735361a4c3c 100644 --- a/sound/soc/codecs/sta32x.c +++ b/sound/soc/codecs/sta32x.c @@ -434,7 +434,7 @@ SOC_SINGLE_TLV("Treble Tone Control", STA32X_TONE, STA32X_TONE_TTC_SHIFT, 15, 0, SOC_ENUM("Limiter1 Attack Rate (dB/ms)", sta32x_limiter1_attack_rate_enum), SOC_ENUM("Limiter2 Attack Rate (dB/ms)", sta32x_limiter2_attack_rate_enum), SOC_ENUM("Limiter1 Release Rate (dB/ms)", sta32x_limiter1_release_rate_enum), -SOC_ENUM("Limiter2 Release Rate (dB/ms)", sta32x_limiter1_release_rate_enum), +SOC_ENUM("Limiter2 Release Rate (dB/ms)", sta32x_limiter2_release_rate_enum), /* depending on mode, the attack/release thresholds have * two different enum definitions; provide both -- cgit v1.2.1