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 diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index 433d59a..2ee23a3 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 v0.10.2 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 diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 454f41c..33fd372 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 v0.10.2 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 diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 33fd372..3507574 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 v0.10.2 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 diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index b7858bf..9ec456a 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 v0.10.2 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 diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 9ec456a..ae3e40a 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 v0.10.2 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 diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index ae3e40a..670afa2 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 v0.10.2 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 diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index 70ff377..5e3bc3c 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 v0.10.2 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 diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index a3fb411..8869249 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 v0.10.2 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 diff --git a/sound/soc/fsl/imx-mc13783.c b/sound/soc/fsl/imx-mc13783.c index 79cee78..a2fd732 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 f2beae7..1cb22dd 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 3fd76bc..3a3d17c 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 v0.10.2 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 diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index d0c72ed..c84026c 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 9c9f957..75e1403 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 v0.10.2 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 diff --git a/drivers/mfd/da9055-i2c.c b/drivers/mfd/da9055-i2c.c index 13af7e5..8103e43 100644 --- a/drivers/mfd/da9055-i2c.c +++ b/drivers/mfd/da9055-i2c.c @@ -53,17 +53,25 @@ static int da9055_i2c_remove(struct i2c_client *i2c) return 0; } +/* + * DO NOT change the device Ids. The naming is intentionally specific as both + * the PMIC and CODEC 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 PMIC + * and CODEC, which must be different to operate together. + */ static struct i2c_device_id da9055_i2c_id[] = { - {"da9055", 0}, + {"da9055-pmic", 0}, { } }; +MODULE_DEVICE_TABLE(i2c, da9055_i2c_id); static struct i2c_driver da9055_i2c_driver = { .probe = da9055_i2c_probe, .remove = da9055_i2c_remove, .id_table = da9055_i2c_id, .driver = { - .name = "da9055", + .name = "da9055-pmic", .owner = THIS_MODULE, }, }; diff --git a/sound/soc/codecs/da9055.c b/sound/soc/codecs/da9055.c index 52b79a4..4228126 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 v0.10.2 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 diff --git a/sound/soc/blackfin/Kconfig b/sound/soc/blackfin/Kconfig index 54f74f8..4544d8e 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 v0.10.2 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 diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index 51f9b3d..149b57f 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 v0.10.2 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 diff --git a/sound/soc/txx9/txx9aclc-ac97.c b/sound/soc/txx9/txx9aclc-ac97.c index e0305a1..9edd68d 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 v0.10.2 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 diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index 149b57f..9f714ea 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 v0.10.2 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 diff --git a/sound/usb/mixer_maps.c b/sound/usb/mixer_maps.c index 32af6b7..d1d72ff 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 v0.10.2 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 diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a9a83b8..6eb903c 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 v0.10.2 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 diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 54d1479..0aa72ee 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -2662,60 +2662,6 @@ static bool dspload_wait_loaded(struct hda_codec *codec) } /* - * 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 */ static int ca0132_playback_pcm_prepare(struct hda_pcm_stream *hinfo, @@ -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 v0.10.2 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 diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 0aa72ee..46ecdbb 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 v0.10.2 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 diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 7311bad..a2f11bf 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 v0.10.2