From b8edf3e5522735c8ce78b81845f7a1a2d4a08626 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 22 Jun 2012 17:21:17 +0100 Subject: ASoC: wm8994: Ensure there are enough BCLKs for four channels Otherwise if someone tries to use all four channels on AIF1 with the device in master mode we won't be able to clock out all the data. Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/wm8994.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index bb62f4b3d563..235577a3d0e7 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2649,7 +2649,7 @@ static int wm8994_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - bclk_rate = params_rate(params) * 2; + bclk_rate = params_rate(params) * 4; switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: bclk_rate *= 16; -- cgit v1.2.3 From c2e1d9073fc98f471067c0257a31b4818306ebe1 Mon Sep 17 00:00:00 2001 From: Dong Aisheng Date: Fri, 20 Jul 2012 17:20:24 +0800 Subject: ASoC: mxs-saif: fix clock prepare and enable unbalance issue Currently we directly call a clock_enable in trigger function without a clk_prepare as pair first. This will cause system hang immediately when run capture because the clock was not prepared(playback does not hang because the clock was prepared already by get_mclk before), a warning message in clock framework may cause a deadlock to reclaim clock lock (see: pl011_console_write). Here we prepare clock first in hw_param, then enable it in trigger function to guarantee the balance. Signed-off-by: Dong Aisheng Acked-by: Shawn Guo Signed-off-by: Mark Brown --- sound/soc/mxs/mxs-saif.c | 10 ++++++++++ 1 file changed, 10 insertions(+) (limited to 'sound') diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index aba71bfa33b1..fdbb36aa9cf5 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -394,9 +394,14 @@ static int mxs_saif_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai) { struct mxs_saif *saif = snd_soc_dai_get_drvdata(cpu_dai); + struct mxs_saif *master_saif; u32 scr, stat; int ret; + master_saif = mxs_saif_get_master(saif); + if (!master_saif) + return -EINVAL; + /* mclk should already be set */ if (!saif->mclk && saif->mclk_in_use) { dev_err(cpu_dai->dev, "set mclk first\n"); @@ -420,6 +425,11 @@ static int mxs_saif_hw_params(struct snd_pcm_substream *substream, return ret; } + /* prepare clk in hw_param, enable in trigger */ + clk_prepare(saif->clk); + if (saif != master_saif) + clk_prepare(master_saif->clk); + scr = __raw_readl(saif->base + SAIF_CTRL); scr &= ~BM_SAIF_CTRL_WORD_LENGTH; -- cgit v1.2.3 From d0ba4c014934cb56f1eabb481ff8026b6d49d33c Mon Sep 17 00:00:00 2001 From: Dong Aisheng Date: Fri, 20 Jul 2012 17:20:25 +0800 Subject: ASoC: mxs-saif: set a base clock rate for EXTMASTER mode work Set an initial clock rate for the saif internal logic to work properly. This is important when working in EXTMASTER mode that uses the other saif's BITCLK&LRCLK but it still needs a basic clock which should be fast enough for the internal logic. Signed-off-by: Dong Aisheng Acked-by: Shawn Guo Signed-off-by: Mark Brown --- sound/soc/mxs/mxs-saif.c | 16 +++++++++++++++- 1 file changed, 15 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index fdbb36aa9cf5..b3030718c228 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -427,8 +427,22 @@ static int mxs_saif_hw_params(struct snd_pcm_substream *substream, /* prepare clk in hw_param, enable in trigger */ clk_prepare(saif->clk); - if (saif != master_saif) + if (saif != master_saif) { + /* + * Set an initial clock rate for the saif internal logic to work + * properly. This is important when working in EXTMASTER mode + * that uses the other saif's BITCLK&LRCLK but it still needs a + * basic clock which should be fast enough for the internal + * logic. + */ + clk_enable(saif->clk); + ret = clk_set_rate(saif->clk, 24000000); + clk_disable(saif->clk); + if (ret) + return ret; + clk_prepare(master_saif->clk); + } scr = __raw_readl(saif->base + SAIF_CTRL); -- cgit v1.2.3 From a07e8d49e8b48d3dc458e234a15e36f379a36dff Mon Sep 17 00:00:00 2001 From: Dong Aisheng Date: Fri, 20 Jul 2012 17:20:26 +0800 Subject: ASoC: sgtl5000: remove unneeded snd_soc_dapm_new_widgets in probe There's a driver bug that sgtl5000 dapm widget kcontrols do not work. e.g. can not select capture mux with amixer tool(no error info prompted). The root cause is that we still call snd_soc_dapm_new_widgets in codec driver probe function afer converting to table based widgets. This will cause the card dapm widgets are instantiated before the dapm_routes are registered. Then, no available dapm widget pathes can be found during instantiation which finally will cause soc_dapm_mux_update_power to fail(can not find correct path with kcontrol) in snd_soc_dapm_put_enum_double function. Here we remove the unneeded snd_soc_dapm_new_widgets in codec probe and let the soc core to handle the register sequence properly. Then we can fix above issue. Signed-off-by: Dong Aisheng Acked-by: Shawn Guo Signed-off-by: Mark Brown --- sound/soc/codecs/sgtl5000.c | 2 -- 1 file changed, 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 8af6a5245b18..5c54b6f4623a 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -1357,8 +1357,6 @@ static int sgtl5000_probe(struct snd_soc_codec *codec) if (ret) goto err; - snd_soc_dapm_new_widgets(&codec->dapm); - return 0; err: -- cgit v1.2.3 From b8176627b84adfea3a729265a5a0f02c850e9275 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 24 Jul 2012 15:48:57 +0100 Subject: ASoC: wm8994: Hold runtime PM reference while handling mic and jack IRQs Ensures that we don't interact badly with the power management framework, especially in the cases where we're doing deferred work or we're using a direct GPIO for these signals. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 235577a3d0e7..04ef03175c51 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3253,10 +3253,13 @@ static void wm8994_mic_work(struct work_struct *work) int ret; int report; + pm_runtime_get_sync(dev); + ret = regmap_read(regmap, WM8994_INTERRUPT_RAW_STATUS_2, ®); if (ret < 0) { dev_err(dev, "Failed to read microphone status: %d\n", ret); + pm_runtime_put(dev); return; } @@ -3299,6 +3302,8 @@ static void wm8994_mic_work(struct work_struct *work) snd_soc_jack_report(priv->micdet[1].jack, report, SND_JACK_HEADSET | SND_JACK_BTN_0); + + pm_runtime_put(dev); } static irqreturn_t wm8994_mic_irq(int irq, void *data) @@ -3421,12 +3426,15 @@ static irqreturn_t wm1811_jackdet_irq(int irq, void *data) int reg; bool present; + pm_runtime_get_sync(codec->dev); + mutex_lock(&wm8994->accdet_lock); reg = snd_soc_read(codec, WM1811_JACKDET_CTRL); if (reg < 0) { dev_err(codec->dev, "Failed to read jack status: %d\n", reg); mutex_unlock(&wm8994->accdet_lock); + pm_runtime_put(codec->dev); return IRQ_NONE; } @@ -3491,6 +3499,7 @@ static irqreturn_t wm1811_jackdet_irq(int irq, void *data) SND_JACK_MECHANICAL | SND_JACK_HEADSET | wm8994->btn_mask); + pm_runtime_put(codec->dev); return IRQ_HANDLED; } @@ -3602,6 +3611,8 @@ static irqreturn_t wm8958_mic_irq(int irq, void *data) if (!(snd_soc_read(codec, WM8958_MIC_DETECT_1) & WM8958_MICD_ENA)) return IRQ_HANDLED; + pm_runtime_get_sync(codec->dev); + /* We may occasionally read a detection without an impedence * range being provided - if that happens loop again. */ @@ -3612,6 +3623,7 @@ static irqreturn_t wm8958_mic_irq(int irq, void *data) dev_err(codec->dev, "Failed to read mic detect status: %d\n", reg); + pm_runtime_put(codec->dev); return IRQ_NONE; } @@ -3639,6 +3651,7 @@ static irqreturn_t wm8958_mic_irq(int irq, void *data) dev_warn(codec->dev, "Accessory detection with no callback\n"); out: + pm_runtime_put(codec->dev); return IRQ_HANDLED; } -- cgit v1.2.3 From aa50fe55ace7451e6ad6812915db367c8cfd3bb3 Mon Sep 17 00:00:00 2001 From: Lee Jones Date: Thu, 26 Jul 2012 11:28:38 +0100 Subject: ASoC: ux500: Include the correct header files Thought to be another merge error, board-mop500-msp.h has never existed in the upstream kernel, only msp.h. This patch changes the include files to match the existing file name. Signed-off-by: Lee Jones Signed-off-by: Mark Brown --- sound/soc/ux500/ux500_msp_dai.c | 2 +- sound/soc/ux500/ux500_msp_i2s.c | 2 +- sound/soc/ux500/ux500_msp_i2s.h | 2 +- 3 files changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/ux500/ux500_msp_dai.c b/sound/soc/ux500/ux500_msp_dai.c index 62ac0285bfaf..057e28ef770e 100644 --- a/sound/soc/ux500/ux500_msp_dai.c +++ b/sound/soc/ux500/ux500_msp_dai.c @@ -21,7 +21,7 @@ #include #include -#include +#include #include #include diff --git a/sound/soc/ux500/ux500_msp_i2s.c b/sound/soc/ux500/ux500_msp_i2s.c index ee14d2dac2f5..5c472f335a64 100644 --- a/sound/soc/ux500/ux500_msp_i2s.c +++ b/sound/soc/ux500/ux500_msp_i2s.c @@ -19,7 +19,7 @@ #include #include -#include +#include #include diff --git a/sound/soc/ux500/ux500_msp_i2s.h b/sound/soc/ux500/ux500_msp_i2s.h index 7f71b4a0d4bc..2d9136da9865 100644 --- a/sound/soc/ux500/ux500_msp_i2s.h +++ b/sound/soc/ux500/ux500_msp_i2s.h @@ -17,7 +17,7 @@ #include -#include +#include #define MSP_INPUT_FREQ_APB 48000000 -- cgit v1.2.3 From e13ab2aac7a273a890d18bb482849610be178bc5 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Thu, 26 Jul 2012 15:29:03 -0300 Subject: ASoC: mc13783: Provide codec->control_data If codec->control_data is not provided, the following crash happens: Unable to handle kernel NULL pointer dereference at virtual address 00000078 pgd = 80004000 [00000078] *pgd=00000000 Internal error: Oops: 5 [#1] SMP ARM Modules linked in: CPU: 0 Tainted: G W (3.5.0-next-20120725+ #1263) PC is at regmap_read+0x18/0x64 LR is at hw_read+0x50/0x98 pc : [<802bcd90>] lr : [<803cad18>] psr: 60000013 ... Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/codecs/mc13783.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c index 6276e352125f..8f726c063f42 100644 --- a/sound/soc/codecs/mc13783.c +++ b/sound/soc/codecs/mc13783.c @@ -581,6 +581,8 @@ static int mc13783_probe(struct snd_soc_codec *codec) { struct mc13783_priv *priv = snd_soc_codec_get_drvdata(codec); + codec->control_data = priv->mc13xxx; + mc13xxx_lock(priv->mc13xxx); /* these are the reset values */ -- cgit v1.2.3 From 0a88f1f01c5f03f7a21a83aa52454aa930742418 Mon Sep 17 00:00:00 2001 From: Roland Stigge Date: Mon, 18 Jun 2012 18:42:21 +0200 Subject: sound: tegra_wm8903: Adjust to of_get_named_gpio() change of_get_named_gpio() was changed to return -EPROBE_DEFER in case of gpios not probed yet. This patch adjusts tegra_wm8903 to this. Signed-off-by: Roland Stigge Acked-by: Stephen Warren Signed-off-by: Mark Brown --- sound/soc/tegra/tegra_wm8903.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c index 0c5bb33d258e..d4f14e492341 100644 --- a/sound/soc/tegra/tegra_wm8903.c +++ b/sound/soc/tegra/tegra_wm8903.c @@ -284,27 +284,27 @@ static __devinit int tegra_wm8903_driver_probe(struct platform_device *pdev) } else if (np) { pdata->gpio_spkr_en = of_get_named_gpio(np, "nvidia,spkr-en-gpios", 0); - if (pdata->gpio_spkr_en == -ENODEV) + if (pdata->gpio_spkr_en == -EPROBE_DEFER) return -EPROBE_DEFER; pdata->gpio_hp_mute = of_get_named_gpio(np, "nvidia,hp-mute-gpios", 0); - if (pdata->gpio_hp_mute == -ENODEV) + if (pdata->gpio_hp_mute == -EPROBE_DEFER) return -EPROBE_DEFER; pdata->gpio_hp_det = of_get_named_gpio(np, "nvidia,hp-det-gpios", 0); - if (pdata->gpio_hp_det == -ENODEV) + if (pdata->gpio_hp_det == -EPROBE_DEFER) return -EPROBE_DEFER; pdata->gpio_int_mic_en = of_get_named_gpio(np, "nvidia,int-mic-en-gpios", 0); - if (pdata->gpio_int_mic_en == -ENODEV) + if (pdata->gpio_int_mic_en == -EPROBE_DEFER) return -EPROBE_DEFER; pdata->gpio_ext_mic_en = of_get_named_gpio(np, "nvidia,ext-mic-en-gpios", 0); - if (pdata->gpio_ext_mic_en == -ENODEV) + if (pdata->gpio_ext_mic_en == -EPROBE_DEFER) return -EPROBE_DEFER; } -- cgit v1.2.3 From bbfc3280ad6701352383e305112b3b27e5b3106f Mon Sep 17 00:00:00 2001 From: Roland Stigge Date: Mon, 18 Jun 2012 18:42:22 +0200 Subject: sound: tegra_alc5632: Adjust to of_get_named_gpio() change of_get_named_gpio() was changed to return -EPROBE_DEFER in case of gpios not probed yet. This patch adjusts tegra_alc5632 to this. Signed-off-by: Roland Stigge Acked-by: Stephen Warren Signed-off-by: Mark Brown --- sound/soc/tegra/tegra_alc5632.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/tegra/tegra_alc5632.c b/sound/soc/tegra/tegra_alc5632.c index d684df294c0c..e463529b38bb 100644 --- a/sound/soc/tegra/tegra_alc5632.c +++ b/sound/soc/tegra/tegra_alc5632.c @@ -177,7 +177,7 @@ static __devinit int tegra_alc5632_probe(struct platform_device *pdev) } alc5632->gpio_hp_det = of_get_named_gpio(np, "nvidia,hp-det-gpios", 0); - if (alc5632->gpio_hp_det == -ENODEV) + if (alc5632->gpio_hp_det == -EPROBE_DEFER) return -EPROBE_DEFER; ret = snd_soc_of_parse_card_name(card, "nvidia,model"); -- cgit v1.2.3 From 5e70b7fc38a4659540ea5f56f7a1a7da20f4554d Mon Sep 17 00:00:00 2001 From: Guillaume Gardet Date: Thu, 12 Jul 2012 15:08:16 +0200 Subject: ASoC: omap: Add missing modules aliases to get sound working on omap devices This patch add missing modules aliases to get sound working on omap devices. Tested on Beagleboard xM rev. B. This patch is against 3.5-rc6 vanilla. Signed-off-by: Guillaume GARDET Signed-off-by: Mans Rullgard From 18b1ba8becc3dd256bdaad2d825f46b551debda3 Mon Sep 17 00:00:00 2001 From: Guillaume GARDET Date: Tue, 10 Jul 2012 13:47:16 +0200 Subject: [PATCH] Add missing modules aliases to fix audio on omap devices Signed-off-by: Mark Brown --- sound/soc/omap/omap-mcbsp.c | 1 + sound/soc/omap/omap-pcm.c | 1 + 2 files changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 1046083e90a0..acdd3ef14e08 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -820,3 +820,4 @@ module_platform_driver(asoc_mcbsp_driver); MODULE_AUTHOR("Jarkko Nikula "); MODULE_DESCRIPTION("OMAP I2S SoC Interface"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:omap-mcbsp"); diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index 5a649da9122a..f0feb06615f8 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -441,3 +441,4 @@ module_platform_driver(omap_pcm_driver); MODULE_AUTHOR("Jarkko Nikula "); MODULE_DESCRIPTION("OMAP PCM DMA module"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:omap-pcm-audio"); -- cgit v1.2.3 From 58f598ff0bb0c030e026a0738450c6a46248f6a8 Mon Sep 17 00:00:00 2001 From: Lee Jones Date: Tue, 31 Jul 2012 15:45:41 +0100 Subject: ASoC: ab8500: Inform SoC Core that we have our own I/O arrangements If codec->control_data is not populated SoC Core assumes we want to use regmap, which fails catastrophically, as we don't have one: Unable to handle kernel NULL pointer dereference at virtual address 00000080 pgd = c0004000 [00000080] *pgd=00000000 Internal error: Oops: 17 [#1] PREEMPT SMP ARM Modules linked in: CPU: 1 Not tainted (3.5.0-rc6-00884-g0b2419e-dirty #130) PC is at regmap_read+0x10/0x5c LR is at hw_read+0x80/0x90 pc : [] lr : [] psr: 60000013 Signed-off-by: Lee Jones Signed-off-by: Mark Brown --- sound/soc/codecs/ab8500-codec.c | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/ab8500-codec.c b/sound/soc/codecs/ab8500-codec.c index 3c795921c5f6..23b40186f9b8 100644 --- a/sound/soc/codecs/ab8500-codec.c +++ b/sound/soc/codecs/ab8500-codec.c @@ -2406,6 +2406,10 @@ static int ab8500_codec_probe(struct snd_soc_codec *codec) /* Setup AB8500 according to board-settings */ pdata = (struct ab8500_platform_data *)dev_get_platdata(dev->parent); + + /* Inform SoC Core that we have our own I/O arrangements. */ + codec->control_data = (void *)true; + status = ab8500_audio_setup_mics(codec, &pdata->codec->amics); if (status < 0) { pr_err("%s: Failed to setup mics (%d)!\n", __func__, status); -- cgit v1.2.3 From eef69ac7c9672049069a0bb88dae756fdec4de07 Mon Sep 17 00:00:00 2001 From: Dong Aisheng Date: Fri, 27 Jul 2012 19:18:42 +0800 Subject: ASoC: sgtl5000: enable VAG_POWER for LINE_IN LINE_IN also needs VAG_POWER on or we may hear noise when directly route LINE_IN to Headphone Mux. Tested on imx28evk. Signed-off-by: Dong Aisheng Signed-off-by: Mark Brown --- sound/soc/codecs/sgtl5000.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 5c54b6f4623a..df2f99d1d428 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -239,6 +239,7 @@ static const struct snd_soc_dapm_route sgtl5000_dapm_routes[] = { {"Headphone Mux", "DAC", "DAC"}, /* dac --> hp_mux */ {"LO", NULL, "DAC"}, /* dac --> line_out */ + {"LINE_IN", NULL, "VAG_POWER"}, {"Headphone Mux", "LINE_IN", "LINE_IN"},/* line_in --> hp_mux */ {"HP", NULL, "Headphone Mux"}, /* hp_mux --> hp */ -- cgit v1.2.3 From d0e3cce9144eb8bff0852531aadbe221addaa2d5 Mon Sep 17 00:00:00 2001 From: Manuel Lauss Date: Tue, 31 Jul 2012 14:42:27 +0200 Subject: ASoC: AC97 doesn't use regmap by default Since commit 38cbf9598feba97de9f9b43efa9153fd7c1a2ec9 ("ASoC: core: Try to use regmap if the driver doesn't set up any I/O") any ASoC codec which doesn't set codec::control_data is assumed to use regmap. That doesn't work with AC97 so this workaround sets the codec::control_data member to a random value to restore proper behaviour. Tested with WM9712. Signed-off-by: Manuel Lauss Signed-off-by: Mark Brown --- sound/soc/codecs/ad1980.c | 1 + sound/soc/codecs/stac9766.c | 1 + sound/soc/codecs/wm9712.c | 1 + sound/soc/codecs/wm9713.c | 1 + 4 files changed, 4 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c index 8c39dddd7d00..11b1b714b8b5 100644 --- a/sound/soc/codecs/ad1980.c +++ b/sound/soc/codecs/ad1980.c @@ -186,6 +186,7 @@ static int ad1980_soc_probe(struct snd_soc_codec *codec) printk(KERN_INFO "AD1980 SoC Audio Codec\n"); + codec->control_data = codec; /* we don't use regmap! */ ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0); if (ret < 0) { printk(KERN_ERR "ad1980: failed to register AC97 codec\n"); diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c index 982e437799a8..33c0f3d39c87 100644 --- a/sound/soc/codecs/stac9766.c +++ b/sound/soc/codecs/stac9766.c @@ -340,6 +340,7 @@ static int stac9766_codec_probe(struct snd_soc_codec *codec) printk(KERN_INFO "STAC9766 SoC Audio Codec %s\n", STAC9766_VERSION); + codec->control_data = codec; /* we don't use regmap! */ ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0); if (ret < 0) goto codec_err; diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index 099e6ec32125..f16fb361a4eb 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -619,6 +619,7 @@ static int wm9712_soc_probe(struct snd_soc_codec *codec) { int ret = 0; + codec->control_data = codec; /* we don't use regmap! */ ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0); if (ret < 0) { printk(KERN_ERR "wm9712: failed to register AC97 codec\n"); diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index 3eb19fb71d17..d0b8a3287a85 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -1196,6 +1196,7 @@ static int wm9713_soc_probe(struct snd_soc_codec *codec) if (wm9713 == NULL) return -ENOMEM; snd_soc_codec_set_drvdata(codec, wm9713); + codec->control_data = wm9713; /* we don't use regmap! */ ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0); if (ret < 0) -- cgit v1.2.3 From 9d40e5582c9c4cfb6977ba2a0ca9c2ed82c56f21 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 30 Jul 2012 18:24:19 +0100 Subject: ASoC: wm8962: Allow VMID time to fully ramp Required for reliable power up from cold. Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/wm8962.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index eaf65863ec21..aa9ce9dd7d8a 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2501,6 +2501,9 @@ static int wm8962_set_bias_level(struct snd_soc_codec *codec, /* VMID 2*250k */ snd_soc_update_bits(codec, WM8962_PWR_MGMT_1, WM8962_VMID_SEL_MASK, 0x100); + + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) + msleep(100); break; case SND_SOC_BIAS_OFF: -- cgit v1.2.3 From 707fba3fa76a4c8855552f5d4c1a12430c09bce8 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 2 Aug 2012 09:04:39 +0200 Subject: ALSA: hda - Support dock on Lenovo Thinkpad T530 with ALC269VC Lenovo Thinkpad T530 with ALC269VC codec has a dock port but BIOS doesn't set up the pins properly. Enable the pins as well as on Thinkpad X230 Tablet. Reported-and-tested-by: Mario Cc: [v3.2+] 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 344b221d2102..b9a5c4503336 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6206,6 +6206,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x21b8, "Thinkpad Edge 14", ALC269_FIXUP_SKU_IGNORE), SND_PCI_QUIRK(0x17aa, 0x21ca, "Thinkpad L412", ALC269_FIXUP_SKU_IGNORE), SND_PCI_QUIRK(0x17aa, 0x21e9, "Thinkpad Edge 15", ALC269_FIXUP_SKU_IGNORE), + SND_PCI_QUIRK(0x17aa, 0x21f6, "Thinkpad T530", ALC269_FIXUP_LENOVO_DOCK), SND_PCI_QUIRK(0x17aa, 0x2203, "Thinkpad X230 Tablet", ALC269_FIXUP_LENOVO_DOCK), SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_QUANTA_MUTE), SND_PCI_QUIRK(0x17aa, 0x3bf8, "Lenovo Ideapd", ALC269_FIXUP_PCM_44K), -- cgit v1.2.3 From 98d3088e534a2a61f6690b5426909b0c3b57a785 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 1 Aug 2012 20:05:47 +0100 Subject: ASoC: core: Fix check before defaulting to regmap Check if the chip has provided a write operation (which is mandatory for I/O) rather than looking for control data as some of the MFDs use a global for this. Also skip the attempt if there's no regmap available by device in case things get confused by the attempt to default. Signed-off-by: Mark Brown Tested-by: Peter Ujfalusi --- sound/soc/soc-core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index f219b2f7ee68..f81c5976b961 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1096,7 +1096,7 @@ static int soc_probe_codec(struct snd_soc_card *card, } /* If the driver didn't set I/O up try regmap */ - if (!codec->control_data) + if (!codec->write && dev_get_regmap(codec->dev, NULL)) snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP); if (driver->controls) -- cgit v1.2.3 From c810f9039f040681ec9d9f2983b748c193037297 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 3 Aug 2012 12:48:32 +0200 Subject: ALSA: PCM: Fix possible memory leaks in the error path When the first page allocation failed for sgbuf, it leaks the records that have been formerly allocated. Signed-off-by: Takashi Iwai --- sound/core/sgbuf.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/core/sgbuf.c b/sound/core/sgbuf.c index 4e7ec2b49873..d0f00356fc11 100644 --- a/sound/core/sgbuf.c +++ b/sound/core/sgbuf.c @@ -101,7 +101,7 @@ void *snd_malloc_sgbuf_pages(struct device *device, if (snd_dma_alloc_pages_fallback(SNDRV_DMA_TYPE_DEV, device, chunk, &tmpb) < 0) { if (!sgbuf->pages) - return NULL; + goto _failed; if (!res_size) goto _failed; size = sgbuf->pages * PAGE_SIZE; -- cgit v1.2.3 From fcfb7866af9a5d0280b7e51dd772990c636b7dec Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 3 Aug 2012 12:51:21 +0200 Subject: ALSA: emu10k1: Avoid access to invalid pages when period=1 When period=1, the driver tries to allocate a bit bigger buffer than requested by the user due to the irq latency tolerance. This may lead to accesses over the actually allocated pages. This patch adds a check of the page index and assigns the silent page when it's over the given buffer size. Signed-off-by: Takashi Iwai --- sound/pci/emu10k1/memory.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/emu10k1/memory.c b/sound/pci/emu10k1/memory.c index 4f502a2bdc3c..0a436626182b 100644 --- a/sound/pci/emu10k1/memory.c +++ b/sound/pci/emu10k1/memory.c @@ -326,7 +326,10 @@ snd_emu10k1_alloc_pages(struct snd_emu10k1 *emu, struct snd_pcm_substream *subst for (page = blk->first_page; page <= blk->last_page; page++, idx++) { unsigned long ofs = idx << PAGE_SHIFT; dma_addr_t addr; - addr = snd_pcm_sgbuf_get_addr(substream, ofs); + if (ofs >= runtime->dma_bytes) + addr = emu->silent_page.addr; + else + addr = snd_pcm_sgbuf_get_addr(substream, ofs); if (! is_valid_page(emu, addr)) { printk(KERN_ERR "emu: failure page = %d\n", idx); mutex_unlock(&hdr->block_mutex); -- cgit v1.2.3 From 4407be6ba217514b1bc01488f8b56467d309e416 Mon Sep 17 00:00:00 2001 From: "Philipp A. Mohrenweiser" Date: Mon, 6 Aug 2012 13:14:18 +0200 Subject: ALSA: hda - add dock support for Thinkpad T430s Add a model/fixup string "lenovo-dock", for Thinkpad T430s, to allow sound in docking station. Tested on Lenovo T430s with ThinkPad Mini Dock Plus Series 3 Cc: stable@kernel.org Signed-off-by: Philipp A. Mohrenweiser 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 b9a5c4503336..bb93f927f7be 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6207,6 +6207,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x21ca, "Thinkpad L412", ALC269_FIXUP_SKU_IGNORE), SND_PCI_QUIRK(0x17aa, 0x21e9, "Thinkpad Edge 15", ALC269_FIXUP_SKU_IGNORE), SND_PCI_QUIRK(0x17aa, 0x21f6, "Thinkpad T530", ALC269_FIXUP_LENOVO_DOCK), + SND_PCI_QUIRK(0x17aa, 0x21fb, "Thinkpad T430s", ALC269_FIXUP_LENOVO_DOCK), SND_PCI_QUIRK(0x17aa, 0x2203, "Thinkpad X230 Tablet", ALC269_FIXUP_LENOVO_DOCK), SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_QUANTA_MUTE), SND_PCI_QUIRK(0x17aa, 0x3bf8, "Lenovo Ideapd", ALC269_FIXUP_PCM_44K), -- cgit v1.2.3 From 8dfaa573918afa34c8eaf8b2120b2e38cc4f651f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 6 Aug 2012 14:49:36 +0200 Subject: ALSA: hda - Fix regression of HDMI codec probing The commit c4bfe94a causes a regression on some codecs at probing. Since this was just a workaround to shut up a kernel warning, it'd be better to revert and fix properly. So we ended up with re-adding the cleanup callback. Tested-and-reported-by: Matt Horan Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_hdmi.c | 12 +++++++++--- 1 file changed, 9 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 69b928449789..8f23374fa642 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -877,8 +877,6 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo, struct hdmi_eld *eld; struct hdmi_spec_per_cvt *per_cvt = NULL; - hinfo->nid = 0; /* clear the leftover value */ - /* Validate hinfo */ pin_idx = hinfo_to_pin_index(spec, hinfo); if (snd_BUG_ON(pin_idx < 0)) @@ -1163,6 +1161,14 @@ static int generic_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo, return hdmi_setup_stream(codec, cvt_nid, pin_nid, stream_tag, format); } +static int generic_hdmi_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream) +{ + snd_hda_codec_cleanup_stream(codec, hinfo->nid); + return 0; +} + static int hdmi_pcm_close(struct hda_pcm_stream *hinfo, struct hda_codec *codec, struct snd_pcm_substream *substream) @@ -1202,6 +1208,7 @@ static const struct hda_pcm_ops generic_ops = { .open = hdmi_pcm_open, .close = hdmi_pcm_close, .prepare = generic_hdmi_playback_pcm_prepare, + .cleanup = generic_hdmi_playback_pcm_cleanup, }; static int generic_hdmi_build_pcms(struct hda_codec *codec) @@ -1220,7 +1227,6 @@ static int generic_hdmi_build_pcms(struct hda_codec *codec) pstr = &info->stream[SNDRV_PCM_STREAM_PLAYBACK]; pstr->substreams = 1; pstr->ops = generic_ops; - pstr->nid = 1; /* FIXME: just for avoiding a debug WARNING */ /* other pstr fields are set in open */ } -- cgit v1.2.3 From c8415a48fcb7a29889f4405d38c57db351e4b50a Mon Sep 17 00:00:00 2001 From: Felix Kaechele Date: Mon, 6 Aug 2012 23:02:01 +0200 Subject: ALSA: hda - add dock support for Thinkpad X230 As with the ThinkPad Models X230 Tablet and T530 the X230 needs a qurik to correctly set up the pins for the dock port. Signed-off-by: Felix Kaechele Cc: [v3.2+] 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 bb93f927f7be..daa5103eb936 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6207,6 +6207,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x21ca, "Thinkpad L412", ALC269_FIXUP_SKU_IGNORE), SND_PCI_QUIRK(0x17aa, 0x21e9, "Thinkpad Edge 15", ALC269_FIXUP_SKU_IGNORE), SND_PCI_QUIRK(0x17aa, 0x21f6, "Thinkpad T530", ALC269_FIXUP_LENOVO_DOCK), + SND_PCI_QUIRK(0x17aa, 0x21fa, "Thinkpad X230", ALC269_FIXUP_LENOVO_DOCK), SND_PCI_QUIRK(0x17aa, 0x21fb, "Thinkpad T430s", ALC269_FIXUP_LENOVO_DOCK), SND_PCI_QUIRK(0x17aa, 0x2203, "Thinkpad X230 Tablet", ALC269_FIXUP_LENOVO_DOCK), SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_QUANTA_MUTE), -- cgit v1.2.3 From e9fc83cb2e5877801a255a37ddbc5be996ea8046 Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Tue, 7 Aug 2012 14:03:29 +0200 Subject: ALSA: hda - remove quirk for Dell Vostro 1015 This computer is confirmed working with model=auto on kernel 3.2. Also, parsing fails with hda-emu with the current model. Cc: stable@kernel.org (3.2+) Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 14361184ae1e..6f538eb3871e 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -2972,7 +2972,6 @@ static const struct snd_pci_quirk cxt5066_cfg_tbl[] = { SND_PCI_QUIRK(0x1028, 0x02d8, "Dell Vostro", CXT5066_DELL_VOSTRO), SND_PCI_QUIRK(0x1028, 0x02f5, "Dell Vostro 320", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x1028, 0x0401, "Dell Vostro 1014", CXT5066_DELL_VOSTRO), - SND_PCI_QUIRK(0x1028, 0x0402, "Dell Vostro", CXT5066_DELL_VOSTRO), SND_PCI_QUIRK(0x1028, 0x0408, "Dell Inspiron One 19T", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x1028, 0x050f, "Dell Inspiron", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x1028, 0x0510, "Dell Vostro", CXT5066_IDEAPAD), -- cgit v1.2.3 From bb10b09a8ea10228ef3e56365fae40f1e24e5589 Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Tue, 7 Aug 2012 14:03:30 +0200 Subject: ALSA: hda - remove redundant auto quirks for conexant 506x Now that the auto model is the default, these quirks are redundant and can be removed. Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 5 ----- 1 file changed, 5 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 6f538eb3871e..5e22a8f43d2e 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -2967,7 +2967,6 @@ static const char * const cxt5066_models[CXT5066_MODELS] = { }; static const struct snd_pci_quirk cxt5066_cfg_tbl[] = { - SND_PCI_QUIRK(0x1025, 0x054c, "Acer Aspire 3830TG", CXT5066_AUTO), SND_PCI_QUIRK_MASK(0x1025, 0xff00, 0x0400, "Acer", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x1028, 0x02d8, "Dell Vostro", CXT5066_DELL_VOSTRO), SND_PCI_QUIRK(0x1028, 0x02f5, "Dell Vostro 320", CXT5066_IDEAPAD), @@ -2987,14 +2986,10 @@ static const struct snd_pci_quirk cxt5066_cfg_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400s", CXT5066_THINKPAD), SND_PCI_QUIRK(0x17aa, 0x21c5, "Thinkpad Edge 13", CXT5066_THINKPAD), SND_PCI_QUIRK(0x17aa, 0x21c6, "Thinkpad Edge 13", CXT5066_ASUS), - SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo T510", CXT5066_AUTO), - SND_PCI_QUIRK(0x17aa, 0x21cf, "Lenovo T520 & W520", CXT5066_AUTO), SND_PCI_QUIRK(0x17aa, 0x21da, "Lenovo X220", CXT5066_THINKPAD), SND_PCI_QUIRK(0x17aa, 0x21db, "Lenovo X220-tablet", CXT5066_THINKPAD), SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo U350", CXT5066_ASUS), SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo G560", CXT5066_ASUS), - SND_PCI_QUIRK(0x17aa, 0x3938, "Lenovo G565", CXT5066_AUTO), - SND_PCI_QUIRK(0x1b0a, 0x2092, "CyberpowerPC Gamer Xplorer N57001", CXT5066_AUTO), {} }; -- cgit v1.2.3 From d0db84e713eaaccea2a435e1625fb3150b335f4a Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 7 Aug 2012 15:37:47 +0300 Subject: ASoC: omap-mcbsp: Fix 6pin mux configuration The check for the mux_signal callback was wrong which prevents us to configure the 6pin port's FSR/CLKR signal mux. Reported-by: CF Adad Signed-off-by: Peter Ujfalusi Acked-by: Jarkko Nikula Signed-off-by: Mark Brown Cc: stable@vger.kernel.org (3.4+) --- sound/soc/omap/mcbsp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/omap/mcbsp.c b/sound/soc/omap/mcbsp.c index 34835e8a9160..d33c48baaf71 100644 --- a/sound/soc/omap/mcbsp.c +++ b/sound/soc/omap/mcbsp.c @@ -745,7 +745,7 @@ int omap_mcbsp_6pin_src_mux(struct omap_mcbsp *mcbsp, u8 mux) { const char *signal, *src; - if (mcbsp->pdata->mux_signal) + if (!mcbsp->pdata->mux_signal) return -EINVAL; switch (mux) { -- cgit v1.2.3 From 709aea6b05f8f6c852c293fb5d47a6461373b4dd Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 7 Aug 2012 18:09:23 +0200 Subject: ALSA: hda - Fix ugly debug prints with CONFIG_SND_VERBOSE_PRINTK=y When CONFIG_SND_VERBOSE_PRINTK=y is set, the debug print in hda_auto_parser.c looks really ugly like: ALSA sound/pci/hda/hda_auto_parser.c:331 mono: mono_out=0x0 ALSA sound/pci/hda/hda_auto_parser.c:334 dig-out=0x12/0x0 ALSA sound/pci/hda/hda_auto_parser.c:335 inputs: ALSA sound/pci/hda/hda_auto_parser.c:339 Mic=0x11ALSA sound/pci/hda/hda_auto_parser.c:339 Line=0x10 ALSA sound/pci/hda/hda_auto_parser.c:341 ALSA sound/pci/hda/hda_auto_parser.c:343 dig-in=0x13 Better to put one item at each line. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_auto_parser.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c index 647218d69f68..4f7d2dfcef7b 100644 --- a/sound/pci/hda/hda_auto_parser.c +++ b/sound/pci/hda/hda_auto_parser.c @@ -332,13 +332,12 @@ int snd_hda_parse_pin_defcfg(struct hda_codec *codec, if (cfg->dig_outs) snd_printd(" dig-out=0x%x/0x%x\n", cfg->dig_out_pins[0], cfg->dig_out_pins[1]); - snd_printd(" inputs:"); + snd_printd(" inputs:\n"); for (i = 0; i < cfg->num_inputs; i++) { - snd_printd(" %s=0x%x", + snd_printd(" %s=0x%x\n", hda_get_autocfg_input_label(codec, cfg, i), cfg->inputs[i].pin); } - snd_printd("\n"); if (cfg->dig_in_pin) snd_printd(" dig-in=0x%x\n", cfg->dig_in_pin); -- cgit v1.2.3 From 012e7eb1e501d0120e0383b81477f63091f5e365 Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Wed, 8 Aug 2012 08:43:37 +0200 Subject: ALSA: hda - Fix double quirk for Quanta FL1 / Lenovo Ideapad The same ID is twice in the quirk table, so the second one is not used. Signed-off-by: David Henningsson Cc: [v3.2+] Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index daa5103eb936..4f81dd44c837 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6099,6 +6099,8 @@ static const struct alc_fixup alc269_fixups[] = { [ALC269_FIXUP_PCM_44K] = { .type = ALC_FIXUP_FUNC, .v.func = alc269_fixup_pcm_44k, + .chained = true, + .chain_id = ALC269_FIXUP_QUANTA_MUTE }, [ALC269_FIXUP_STEREO_DMIC] = { .type = ALC_FIXUP_FUNC, @@ -6210,8 +6212,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x21fa, "Thinkpad X230", ALC269_FIXUP_LENOVO_DOCK), SND_PCI_QUIRK(0x17aa, 0x21fb, "Thinkpad T430s", ALC269_FIXUP_LENOVO_DOCK), SND_PCI_QUIRK(0x17aa, 0x2203, "Thinkpad X230 Tablet", ALC269_FIXUP_LENOVO_DOCK), - SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_QUANTA_MUTE), - SND_PCI_QUIRK(0x17aa, 0x3bf8, "Lenovo Ideapd", ALC269_FIXUP_PCM_44K), + SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K), SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD), #if 0 -- cgit v1.2.3 From 48a08bab3066a9452216f8c52e0d6f35566de04d Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Wed, 8 Aug 2012 00:47:21 -0300 Subject: ASoC: mxs: Fix the name of the SoC family SND_SOC_MXS_SGTL5000 is used on MXS boards, so fix the SoC family name. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/mxs/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/mxs/Kconfig b/sound/soc/mxs/Kconfig index 99a997f19bb9..b6fa77678d97 100644 --- a/sound/soc/mxs/Kconfig +++ b/sound/soc/mxs/Kconfig @@ -10,7 +10,7 @@ menuconfig SND_MXS_SOC if SND_MXS_SOC config SND_SOC_MXS_SGTL5000 - tristate "SoC Audio support for i.MX boards with sgtl5000" + tristate "SoC Audio support for MXS boards with sgtl5000" depends on I2C select SND_SOC_SGTL5000 help -- cgit v1.2.3 From 0865a75d4166bddc533fd50831829ceefb94f9b0 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Tue, 7 Aug 2012 16:51:34 -0300 Subject: ASoC: imx-ssi: Remove mono support MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Playing a mono track results in incorrect playback rate, ie, the audio is played at a faster rate. Remove mono support in the driver by setting 'channes_min' to dual-channel and this allows mono tracks to be played correctly. Reported-by: Gaëtan Carlier Tested-by: Gaëtan Carlier Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/fsl/imx-ssi.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/imx-ssi.c b/sound/soc/fsl/imx-ssi.c index 28dd76c7cb1c..81d7728cf67f 100644 --- a/sound/soc/fsl/imx-ssi.c +++ b/sound/soc/fsl/imx-ssi.c @@ -380,13 +380,14 @@ static int imx_ssi_dai_probe(struct snd_soc_dai *dai) static struct snd_soc_dai_driver imx_ssi_dai = { .probe = imx_ssi_dai_probe, .playback = { - .channels_min = 1, + /* The SSI does not support monaural audio. */ + .channels_min = 2, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_96000, .formats = SNDRV_PCM_FMTBIT_S16_LE, }, .capture = { - .channels_min = 1, + .channels_min = 2, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_96000, .formats = SNDRV_PCM_FMTBIT_S16_LE, -- cgit v1.2.3 From ed36081350d2ca4f692f04c6a2d24d1e3a339da1 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 8 Aug 2012 17:12:52 +0200 Subject: ALSA: hda - Add codec->pcm_format_first flag Introduced a new flag to set up the PCM stream format at first before the stream_id and channel tag. Some codecs (e.g. CA0132) seem preferring this over stream_id -> format order. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 69 ++++++++++++++++++++++++++++++----------------- sound/pci/hda/hda_codec.h | 1 + 2 files changed, 46 insertions(+), 24 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 88a9c20eb7a2..598b9e2d85e6 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1386,6 +1386,44 @@ int snd_hda_codec_configure(struct hda_codec *codec) } EXPORT_SYMBOL_HDA(snd_hda_codec_configure); +/* update the stream-id if changed */ +static void update_pcm_stream_id(struct hda_codec *codec, + struct hda_cvt_setup *p, hda_nid_t nid, + u32 stream_tag, int channel_id) +{ + unsigned int oldval, newval; + + if (p->stream_tag != stream_tag || p->channel_id != channel_id) { + 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); + p->stream_tag = stream_tag; + p->channel_id = channel_id; + } +} + +/* update the format-id if changed */ +static void update_pcm_format(struct hda_codec *codec, struct hda_cvt_setup *p, + hda_nid_t nid, int format) +{ + unsigned int oldval; + + if (p->format_id != format) { + oldval = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_STREAM_FORMAT, 0); + if (oldval != format) { + msleep(1); + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_STREAM_FORMAT, + format); + } + p->format_id = format; + } +} + /** * snd_hda_codec_setup_stream - set up the codec for streaming * @codec: the CODEC to set up @@ -1400,7 +1438,6 @@ void snd_hda_codec_setup_stream(struct hda_codec *codec, hda_nid_t nid, { struct hda_codec *c; struct hda_cvt_setup *p; - unsigned int oldval, newval; int type; int i; @@ -1413,29 +1450,13 @@ void snd_hda_codec_setup_stream(struct hda_codec *codec, hda_nid_t nid, p = get_hda_cvt_setup(codec, nid); if (!p) return; - /* update the stream-id if changed */ - if (p->stream_tag != stream_tag || p->channel_id != channel_id) { - 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); - p->stream_tag = stream_tag; - p->channel_id = channel_id; - } - /* update the format-id if changed */ - if (p->format_id != format) { - oldval = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_STREAM_FORMAT, 0); - if (oldval != format) { - msleep(1); - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_STREAM_FORMAT, - format); - } - p->format_id = format; - } + + if (codec->pcm_format_first) + update_pcm_format(codec, p, nid, format); + update_pcm_stream_id(codec, p, nid, stream_tag, channel_id); + if (!codec->pcm_format_first) + update_pcm_format(codec, p, nid, format); + p->active = 1; p->dirty = 0; diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index c422d330ca54..7fbc1bcaf1a9 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -861,6 +861,7 @@ struct hda_codec { unsigned int no_trigger_sense:1; /* don't trigger at pin-sensing */ unsigned int ignore_misc_bit:1; /* ignore MISC_NO_PRESENCE bit */ unsigned int no_jack_detect:1; /* Machine has no jack-detection */ + unsigned int pcm_format_first:1; /* PCM format must be set first */ #ifdef CONFIG_SND_HDA_POWER_SAVE unsigned int power_on :1; /* current (global) power-state */ int power_transition; /* power-state in transition */ -- cgit v1.2.3 From 55cf87fe0e9783e25f442be1e48b8319d86131ea Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 8 Aug 2012 17:15:55 +0200 Subject: ALSA: hda - Fix superfluous "-in" suffix from CA0132 capture items Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_ca0132.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index d0d3540e39e7..2685590925ff 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -988,12 +988,12 @@ static void ca0132_config(struct hda_codec *codec) /* Mic-in */ spec->input_pins[0] = 0x12; - spec->input_labels[0] = "Mic-In"; + spec->input_labels[0] = "Mic"; spec->adcs[0] = 0x07; /* Line-In */ spec->input_pins[1] = 0x11; - spec->input_labels[1] = "Line-In"; + spec->input_labels[1] = "Line"; spec->adcs[1] = 0x08; spec->num_inputs = 2; } -- cgit v1.2.3 From 27ebeb0b1b5bb26908e485a7e8bd2ec30366ffef Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 8 Aug 2012 17:20:18 +0200 Subject: ALSA: hda - Use the standard PCM ops for CA0132 Now with the workaround using codec->pcm_format_first flag, we can clean up the home-baked codes in patch_ca0132.c. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_ca0132.c | 142 +++++++++---------------------------------- 1 file changed, 29 insertions(+), 113 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 2685590925ff..31512a0f1d07 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -464,50 +464,17 @@ exit: } /* - * PCM stuffs + * PCM callbacks */ -static void ca0132_setup_stream(struct hda_codec *codec, hda_nid_t nid, - u32 stream_tag, - int channel_id, int format) +static int ca0132_playback_pcm_open(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream) { - 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) -{ - 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); + struct ca0132_spec *spec = codec->spec; + return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream, + hinfo); } -/* - * PCM callbacks - */ static int ca0132_playback_pcm_prepare(struct hda_pcm_stream *hinfo, struct hda_codec *codec, unsigned int stream_tag, @@ -515,10 +482,8 @@ static int ca0132_playback_pcm_prepare(struct hda_pcm_stream *hinfo, struct snd_pcm_substream *substream) { struct ca0132_spec *spec = codec->spec; - - ca0132_setup_stream(codec, spec->dacs[0], stream_tag, 0, format); - - return 0; + return snd_hda_multi_out_analog_prepare(codec, &spec->multiout, + stream_tag, format, substream); } static int ca0132_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, @@ -526,92 +491,45 @@ static int ca0132_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, struct snd_pcm_substream *substream) { struct ca0132_spec *spec = codec->spec; - - ca0132_cleanup_stream(codec, spec->dacs[0]); - - return 0; + return snd_hda_multi_out_analog_cleanup(codec, &spec->multiout); } /* * Digital out */ -static int ca0132_dig_playback_pcm_prepare(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - unsigned int stream_tag, - unsigned int format, - struct snd_pcm_substream *substream) +static int ca0132_dig_playback_pcm_open(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream) { struct ca0132_spec *spec = codec->spec; - - ca0132_setup_stream(codec, spec->dig_out, stream_tag, 0, format); - - return 0; -} - -static int ca0132_dig_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - struct snd_pcm_substream *substream) -{ - struct ca0132_spec *spec = codec->spec; - - ca0132_cleanup_stream(codec, spec->dig_out); - - return 0; + return snd_hda_multi_out_dig_open(codec, &spec->multiout); } -/* - * Analog capture - */ -static int ca0132_capture_pcm_prepare(struct hda_pcm_stream *hinfo, +static int ca0132_dig_playback_pcm_prepare(struct hda_pcm_stream *hinfo, struct hda_codec *codec, unsigned int stream_tag, unsigned int format, struct snd_pcm_substream *substream) { struct ca0132_spec *spec = codec->spec; - - ca0132_setup_stream(codec, spec->adcs[substream->number], - stream_tag, 0, format); - - return 0; + return snd_hda_multi_out_dig_prepare(codec, &spec->multiout, + stream_tag, format, substream); } -static int ca0132_capture_pcm_cleanup(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - struct snd_pcm_substream *substream) -{ - struct ca0132_spec *spec = codec->spec; - - ca0132_cleanup_stream(codec, spec->adcs[substream->number]); - - return 0; -} - -/* - * Digital capture - */ -static int ca0132_dig_capture_pcm_prepare(struct hda_pcm_stream *hinfo, +static int ca0132_dig_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, struct hda_codec *codec, - unsigned int stream_tag, - unsigned int format, struct snd_pcm_substream *substream) { struct ca0132_spec *spec = codec->spec; - - ca0132_setup_stream(codec, spec->dig_in, stream_tag, 0, format); - - return 0; + return snd_hda_multi_out_dig_cleanup(codec, &spec->multiout); } -static int ca0132_dig_capture_pcm_cleanup(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - struct snd_pcm_substream *substream) +static int ca0132_dig_playback_pcm_close(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream) { struct ca0132_spec *spec = codec->spec; - - ca0132_cleanup_stream(codec, spec->dig_in); - - return 0; + return snd_hda_multi_out_dig_close(codec, &spec->multiout); } /* @@ -621,6 +539,7 @@ static struct hda_pcm_stream ca0132_pcm_analog_playback = { .channels_min = 2, .channels_max = 2, .ops = { + .open = ca0132_playback_pcm_open, .prepare = ca0132_playback_pcm_prepare, .cleanup = ca0132_playback_pcm_cleanup }, @@ -630,10 +549,6 @@ static struct hda_pcm_stream ca0132_pcm_analog_capture = { .substreams = 1, .channels_min = 2, .channels_max = 2, - .ops = { - .prepare = ca0132_capture_pcm_prepare, - .cleanup = ca0132_capture_pcm_cleanup - }, }; static struct hda_pcm_stream ca0132_pcm_digital_playback = { @@ -641,6 +556,8 @@ static struct hda_pcm_stream ca0132_pcm_digital_playback = { .channels_min = 2, .channels_max = 2, .ops = { + .open = ca0132_dig_playback_pcm_open, + .close = ca0132_dig_playback_pcm_close, .prepare = ca0132_dig_playback_pcm_prepare, .cleanup = ca0132_dig_playback_pcm_cleanup }, @@ -650,10 +567,6 @@ static struct hda_pcm_stream ca0132_pcm_digital_capture = { .substreams = 1, .channels_min = 2, .channels_max = 2, - .ops = { - .prepare = ca0132_dig_capture_pcm_prepare, - .cleanup = ca0132_dig_capture_pcm_cleanup - }, }; static int ca0132_build_pcms(struct hda_codec *codec) @@ -961,6 +874,9 @@ static void ca0132_config(struct hda_codec *codec) struct ca0132_spec *spec = codec->spec; struct auto_pin_cfg *cfg = &spec->autocfg; + codec->pcm_format_first = 1; + codec->no_sticky_stream = 1; + /* line-outs */ cfg->line_outs = 1; cfg->line_out_pins[0] = 0x0b; /* front */ -- cgit v1.2.3 From 8e13fc1c5f694a6ae4032c7f94103c137136733f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 8 Aug 2012 17:26:54 +0200 Subject: ALSA: hda - Add missing SPDIF I/O setup for CA0132 CA0132 driver had some codes to handle the S/PDIF I/O, but the actual setups of pins and converters were missing. Now the pins are added. Also, fixed a few points triggering invalid codec verbs and mixer elements since the digital I/O audio widgets on CA0132 have no amp. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_ca0132.c | 20 ++++++++++++++------ 1 file changed, 14 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 31512a0f1d07..9c0ec0a55bef 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -246,7 +246,7 @@ static void init_output(struct hda_codec *codec, hda_nid_t pin, hda_nid_t dac) AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE); } - if (dac) + if (dac && (get_wcaps(codec, dac) & AC_WCAP_OUT_AMP)) snd_hda_codec_write(codec, dac, 0, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO); } @@ -261,7 +261,7 @@ static void init_input(struct hda_codec *codec, hda_nid_t pin, hda_nid_t adc) AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)); } - if (adc) + if (adc && (get_wcaps(codec, adc) & AC_WCAP_IN_AMP)) snd_hda_codec_write(codec, adc, 0, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)); } @@ -841,18 +841,16 @@ static int ca0132_build_controls(struct hda_codec *codec) spec->dig_out); if (err < 0) return err; - err = add_out_volume(codec, spec->dig_out, "IEC958"); + err = snd_hda_create_spdif_share_sw(codec, &spec->multiout); if (err < 0) return err; + /* spec->multiout.share_spdif = 1; */ } if (spec->dig_in) { err = snd_hda_create_spdif_in_ctls(codec, spec->dig_in); if (err < 0) return err; - err = add_in_volume(codec, spec->dig_in, "IEC958"); - if (err < 0) - return err; } return 0; } @@ -912,6 +910,16 @@ static void ca0132_config(struct hda_codec *codec) spec->input_labels[1] = "Line"; spec->adcs[1] = 0x08; spec->num_inputs = 2; + + /* SPDIF I/O */ + spec->dig_out = 0x05; + spec->multiout.dig_out_nid = spec->dig_out; + cfg->dig_out_pins[0] = 0x0c; + cfg->dig_outs = 1; + cfg->dig_out_type[0] = HDA_PCM_TYPE_SPDIF; + spec->dig_in = 0x09; + cfg->dig_in_pin = 0x0e; + cfg->dig_in_type = HDA_PCM_TYPE_SPDIF; } static void ca0132_init_chip(struct hda_codec *codec) -- cgit v1.2.3 From 94c142a160d63edac0e1fca7848960dcf75dd2a9 Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Thu, 9 Aug 2012 10:56:12 +0200 Subject: ALSA: hda - Fix pop noise in headphones on S3 for Asus X55A, X55V To turn off pin control for the pin was tested, and helped against this issue. BugLink: https://bugs.launchpad.net/bugs/1034779 Tested-by: Chih-Hsyuan Ho Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_via.c | 8 ++++++++ 1 file changed, 8 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 80d90cb42853..430771776915 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -1752,6 +1752,14 @@ static int via_suspend(struct hda_codec *codec) { struct via_spec *spec = codec->spec; vt1708_stop_hp_work(spec); + + if (spec->codec_type == VT1802) { + /* Fix pop noise on headphones */ + int i; + for (i = 0; i < spec->autocfg.hp_outs; i++) + snd_hda_set_pin_ctl(codec, spec->autocfg.hp_pins[i], 0); + } + return 0; } #endif -- cgit v1.2.3 From 0d624275720a4b01217693eb80d967a0d5f1f3a3 Mon Sep 17 00:00:00 2001 From: Vaibhav Bedia Date: Wed, 8 Aug 2012 20:40:31 +0530 Subject: ASoC: Davinci: McASP: Flush the FIFO before enabling FIFO should be flushed before it is enabled for the first time. This fixes the I/O errors reported by the ASoC core on a fresh boot Signed-off-by: Vaibhav Bedia Signed-off-by: Hebbar, Gururaja Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-mcasp.c | 10 ++++++++-- 1 file changed, 8 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 95441bfc8190..ce5e5cd254dd 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -380,14 +380,20 @@ static void mcasp_start_tx(struct davinci_audio_dev *dev) static void davinci_mcasp_start(struct davinci_audio_dev *dev, int stream) { if (stream == SNDRV_PCM_STREAM_PLAYBACK) { - if (dev->txnumevt) /* enable FIFO */ + if (dev->txnumevt) { /* enable FIFO */ + mcasp_clr_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, + FIFO_ENABLE); mcasp_set_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, FIFO_ENABLE); + } mcasp_start_tx(dev); } else { - if (dev->rxnumevt) /* enable FIFO */ + if (dev->rxnumevt) { /* enable FIFO */ + mcasp_clr_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, + FIFO_ENABLE); mcasp_set_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, FIFO_ENABLE); + } mcasp_start_rx(dev); } } -- cgit v1.2.3 From 8b5eae137b91cb2db15fe2c5a913cafde4629339 Mon Sep 17 00:00:00 2001 From: Scott Jiang Date: Thu, 9 Aug 2012 18:08:40 -0400 Subject: ASoC: bfin: fix memory leak in sport3 controller driver Signed-off-by: Scott Jiang Signed-off-by: Mark Brown --- sound/soc/blackfin/bf6xx-sport.c | 7 +++++++ 1 file changed, 7 insertions(+) (limited to 'sound') diff --git a/sound/soc/blackfin/bf6xx-sport.c b/sound/soc/blackfin/bf6xx-sport.c index 318c5ba5360f..dfb744381c42 100644 --- a/sound/soc/blackfin/bf6xx-sport.c +++ b/sound/soc/blackfin/bf6xx-sport.c @@ -413,7 +413,14 @@ EXPORT_SYMBOL(sport_create); void sport_delete(struct sport_device *sport) { + if (sport->tx_desc) + dma_free_coherent(NULL, sport->tx_desc_size, + sport->tx_desc, 0); + if (sport->rx_desc) + dma_free_coherent(NULL, sport->rx_desc_size, + sport->rx_desc, 0); sport_free_resource(sport); + kfree(sport); } EXPORT_SYMBOL(sport_delete); -- cgit v1.2.3 From 52c0eee3329b08dfd912a59e0246e21026308301 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 30 Jul 2012 18:23:35 +0100 Subject: ASoC: wm8962: Don't duplicate bias level management in resume The core will bring the bias level up for us since we use idle_bias_off, duplicating this may be harmful. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8962.c | 15 --------------- 1 file changed, 15 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index aa9ce9dd7d8a..ce6720073798 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3733,21 +3733,6 @@ static int wm8962_runtime_resume(struct device *dev) regcache_sync(wm8962->regmap); - regmap_update_bits(wm8962->regmap, WM8962_ANTI_POP, - WM8962_STARTUP_BIAS_ENA | WM8962_VMID_BUF_ENA, - WM8962_STARTUP_BIAS_ENA | WM8962_VMID_BUF_ENA); - - /* Bias enable at 2*50k for ramp */ - regmap_update_bits(wm8962->regmap, WM8962_PWR_MGMT_1, - WM8962_VMID_SEL_MASK | WM8962_BIAS_ENA, - WM8962_BIAS_ENA | 0x180); - - msleep(5); - - /* VMID back to 2x250k for standby */ - regmap_update_bits(wm8962->regmap, WM8962_PWR_MGMT_1, - WM8962_VMID_SEL_MASK, 0x100); - return 0; } -- cgit v1.2.3 From 15676937e6a7e98d37f4c1eaa0e7b3c111627fce Mon Sep 17 00:00:00 2001 From: Chris Rattray Date: Thu, 9 Aug 2012 10:10:54 +0100 Subject: ASoC: wm8994: Add missing dapm routes for WM8958 rev A Signed-off-by: Chris Rattray Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 04ef03175c51..6c9eeca85b95 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -4038,6 +4038,8 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) break; case WM8958: if (wm8994->revision < 1) { + snd_soc_dapm_add_routes(dapm, wm8994_intercon, + ARRAY_SIZE(wm8994_intercon)); snd_soc_dapm_add_routes(dapm, wm8994_revd_intercon, ARRAY_SIZE(wm8994_revd_intercon)); snd_soc_dapm_add_routes(dapm, wm8994_lateclk_revd_intercon, -- cgit v1.2.3 From d34e4e00adbbc91ff9fc96ed9a4e4b65161868da Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 9 Aug 2012 15:47:15 +0200 Subject: ALSA: platform: Check CONFIG_PM_SLEEP instead of CONFIG_PM When CONFIG_PM is set but CONFIG_PM_SLEEP is unset, SIMPLE_DEV_PM_OPS() ignores the given functions, and this leads to compile warnings. For avoiding this, simply check CONFIG_PM_SLEEP instead of CONFIG_PM. Reported-by: Arnd Bergmann Signed-off-by: Takashi Iwai --- sound/arm/pxa2xx-ac97.c | 4 ++-- sound/atmel/abdac.c | 2 +- sound/atmel/ac97c.c | 2 +- sound/drivers/aloop.c | 2 +- sound/drivers/dummy.c | 2 +- sound/drivers/pcsp/pcsp.c | 4 ++-- sound/ppc/powermac.c | 2 +- 7 files changed, 9 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/arm/pxa2xx-ac97.c b/sound/arm/pxa2xx-ac97.c index 0d7b25e81643..4e1fda75c1c9 100644 --- a/sound/arm/pxa2xx-ac97.c +++ b/sound/arm/pxa2xx-ac97.c @@ -106,7 +106,7 @@ static struct pxa2xx_pcm_client pxa2xx_ac97_pcm_client = { .prepare = pxa2xx_ac97_pcm_prepare, }; -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP static int pxa2xx_ac97_do_suspend(struct snd_card *card) { @@ -243,7 +243,7 @@ static struct platform_driver pxa2xx_ac97_driver = { .driver = { .name = "pxa2xx-ac97", .owner = THIS_MODULE, -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP .pm = &pxa2xx_ac97_pm_ops, #endif }, diff --git a/sound/atmel/abdac.c b/sound/atmel/abdac.c index eb4ceb71123e..98554f4882b7 100644 --- a/sound/atmel/abdac.c +++ b/sound/atmel/abdac.c @@ -534,7 +534,7 @@ out_put_pclk: return retval; } -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP static int atmel_abdac_suspend(struct device *pdev) { struct snd_card *card = dev_get_drvdata(pdev); diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c index bf47025bdf45..3c8d3ba7ddfc 100644 --- a/sound/atmel/ac97c.c +++ b/sound/atmel/ac97c.c @@ -1134,7 +1134,7 @@ err_snd_card_new: return retval; } -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP static int atmel_ac97c_suspend(struct device *pdev) { struct snd_card *card = dev_get_drvdata(pdev); diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c index 1128b35b2b05..5a34355e78e8 100644 --- a/sound/drivers/aloop.c +++ b/sound/drivers/aloop.c @@ -1176,7 +1176,7 @@ static int __devexit loopback_remove(struct platform_device *devptr) return 0; } -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP static int loopback_suspend(struct device *pdev) { struct snd_card *card = dev_get_drvdata(pdev); diff --git a/sound/drivers/dummy.c b/sound/drivers/dummy.c index f7d3bfc6bca8..54bb6644a598 100644 --- a/sound/drivers/dummy.c +++ b/sound/drivers/dummy.c @@ -1064,7 +1064,7 @@ static int __devexit snd_dummy_remove(struct platform_device *devptr) return 0; } -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP static int snd_dummy_suspend(struct device *pdev) { struct snd_card *card = dev_get_drvdata(pdev); diff --git a/sound/drivers/pcsp/pcsp.c b/sound/drivers/pcsp/pcsp.c index 6ca59fc6dcb9..ef171295f6d4 100644 --- a/sound/drivers/pcsp/pcsp.c +++ b/sound/drivers/pcsp/pcsp.c @@ -199,7 +199,7 @@ static void pcsp_stop_beep(struct snd_pcsp *chip) pcspkr_stop_sound(); } -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP static int pcsp_suspend(struct device *dev) { struct snd_pcsp *chip = dev_get_drvdata(dev); @@ -212,7 +212,7 @@ static SIMPLE_DEV_PM_OPS(pcsp_pm, pcsp_suspend, NULL); #define PCSP_PM_OPS &pcsp_pm #else #define PCSP_PM_OPS NULL -#endif /* CONFIG_PM */ +#endif /* CONFIG_PM_SLEEP */ static void pcsp_shutdown(struct platform_device *dev) { diff --git a/sound/ppc/powermac.c b/sound/ppc/powermac.c index f5ceb6f282de..210cafe04890 100644 --- a/sound/ppc/powermac.c +++ b/sound/ppc/powermac.c @@ -143,7 +143,7 @@ static int __devexit snd_pmac_remove(struct platform_device *devptr) return 0; } -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP static int snd_pmac_driver_suspend(struct device *dev) { struct snd_card *card = dev_get_drvdata(dev); -- cgit v1.2.3 From 144dad99ef6ad10c8c8ebe787d08157c4a94201f Mon Sep 17 00:00:00 2001 From: James Ralston Date: Thu, 9 Aug 2012 09:38:59 -0700 Subject: ALSA: hda_intel: Add Device IDs for Intel Lynx Point-LP PCH This patch adds the Intel HD Audio Device IDs for the Intel Lynx Point-LP PCH Signed-off-by: James Ralston Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 9 +++++++++ 1 file changed, 9 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index c8aced182fd1..60882c62f180 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -151,6 +151,7 @@ MODULE_SUPPORTED_DEVICE("{{Intel, ICH6}," "{Intel, CPT}," "{Intel, PPT}," "{Intel, LPT}," + "{Intel, LPT_LP}," "{Intel, HPT}," "{Intel, PBG}," "{Intel, SCH}," @@ -3270,6 +3271,14 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = { { PCI_DEVICE(0x8086, 0x8c20), .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP | AZX_DCAPS_BUFSIZE | AZX_DCAPS_POSFIX_COMBO }, + /* Lynx Point-LP */ + { PCI_DEVICE(0x8086, 0x9c20), + .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP | + AZX_DCAPS_BUFSIZE | AZX_DCAPS_POSFIX_COMBO }, + /* Lynx Point-LP */ + { PCI_DEVICE(0x8086, 0x9c21), + .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP | + AZX_DCAPS_BUFSIZE | AZX_DCAPS_POSFIX_COMBO }, /* Haswell */ { PCI_DEVICE(0x8086, 0x0c0c), .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_SCH_SNOOP | -- cgit v1.2.3 From fb099cb712e878b9eb4e78dd6b268312a0b2b50f Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 9 Aug 2012 18:44:37 +0100 Subject: ASoC: core: Upgrade the severity of probe deferral errors to dev_err() In the past when ASoC had a custom probe deferral mechanism people complained about the logspam it generated and didn't want to know about the fact that we were doing probe deferral so all the error messages for it were at dev_dbg(), making diagnostics hard. Now that we have probe deferral as an accepted thing and it's generating log messages anyway there's no need to worry about this so upgrade the severity of all the probe deferral sources to dev_err() so that they are displayed by default. Also add one for missing aux_devs since there wasn't one. Reported-by: Russell King Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 10 ++++++---- 1 file changed, 6 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index f81c5976b961..c501af6d8dbe 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -826,7 +826,7 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num) } if (!rtd->cpu_dai) { - dev_dbg(card->dev, "CPU DAI %s not registered\n", + dev_err(card->dev, "CPU DAI %s not registered\n", dai_link->cpu_dai_name); return -EPROBE_DEFER; } @@ -857,14 +857,14 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num) } if (!rtd->codec_dai) { - dev_dbg(card->dev, "CODEC DAI %s not registered\n", + dev_err(card->dev, "CODEC DAI %s not registered\n", dai_link->codec_dai_name); return -EPROBE_DEFER; } } if (!rtd->codec) { - dev_dbg(card->dev, "CODEC %s not registered\n", + dev_err(card->dev, "CODEC %s not registered\n", dai_link->codec_name); return -EPROBE_DEFER; } @@ -888,7 +888,7 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num) rtd->platform = platform; } if (!rtd->platform) { - dev_dbg(card->dev, "platform %s not registered\n", + dev_err(card->dev, "platform %s not registered\n", dai_link->platform_name); return -EPROBE_DEFER; } @@ -1481,6 +1481,8 @@ static int soc_check_aux_dev(struct snd_soc_card *card, int num) return 0; } + dev_err(card->dev, "%s not registered\n", aux_dev->codec_name); + return -EPROBE_DEFER; } -- cgit v1.2.3 From de64c0ee7dbcbfbbe63bd9ea45783d87babc6452 Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Fri, 10 Aug 2012 12:22:58 +0300 Subject: ALSA: cs46xx - signedness bug in snd_cs46xx_codec_read() This function returns its own error codes instead of normal negative error codes. Signed-off-by: Dan Carpenter Signed-off-by: Takashi Iwai --- sound/pci/cs46xx/cs46xx_lib.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/cs46xx/cs46xx_lib.c b/sound/pci/cs46xx/cs46xx_lib.c index f75f5ffdfdfb..a71d1c14a0f6 100644 --- a/sound/pci/cs46xx/cs46xx_lib.c +++ b/sound/pci/cs46xx/cs46xx_lib.c @@ -94,7 +94,7 @@ static unsigned short snd_cs46xx_codec_read(struct snd_cs46xx *chip, if (snd_BUG_ON(codec_index != CS46XX_PRIMARY_CODEC_INDEX && codec_index != CS46XX_SECONDARY_CODEC_INDEX)) - return -EINVAL; + return 0xffff; chip->active_ctrl(chip, 1); -- cgit v1.2.3 From 14bc9c6dc694e2d7930802f7afd275de25ef8394 Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Fri, 10 Aug 2012 13:29:32 +0200 Subject: ALSA: hda - Fix panned "Beep Playback Switch" When "Beep Playback Switch" had a different value on left and right channels (such as muting left but not right, or vice versa), this could result in the right channel being ignored. This patch enables beep to be sounding from right channel only, and also give correct result back to userspace (e g amixer). Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_beep.c | 20 ++++++++++++++------ 1 file changed, 14 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c index 0bc2315b181d..d26ae65b43b7 100644 --- a/sound/pci/hda/hda_beep.c +++ b/sound/pci/hda/hda_beep.c @@ -237,10 +237,9 @@ int snd_hda_mixer_amp_switch_get_beep(struct snd_kcontrol *kcontrol, { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct hda_beep *beep = codec->beep; - if (beep) { + if (beep && !beep->enabled) { ucontrol->value.integer.value[0] = - ucontrol->value.integer.value[1] = - beep->enabled; + ucontrol->value.integer.value[1] = 0; return 0; } return snd_hda_mixer_amp_switch_get(kcontrol, ucontrol); @@ -252,9 +251,18 @@ int snd_hda_mixer_amp_switch_put_beep(struct snd_kcontrol *kcontrol, { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct hda_beep *beep = codec->beep; - if (beep) - snd_hda_enable_beep_device(codec, - *ucontrol->value.integer.value); + if (beep) { + u8 chs = get_amp_channels(kcontrol); + int enable = 0; + long *valp = ucontrol->value.integer.value; + if (chs & 1) { + enable |= *valp; + valp++; + } + if (chs & 2) + enable |= *valp; + snd_hda_enable_beep_device(codec, enable); + } return snd_hda_mixer_amp_switch_put(kcontrol, ucontrol); } EXPORT_SYMBOL_HDA(snd_hda_mixer_amp_switch_put_beep); -- cgit v1.2.3 From e037cb4a54e26b5f55f856e0e7445cfcfb2f3d31 Mon Sep 17 00:00:00 2001 From: Mengdong Lin Date: Fri, 10 Aug 2012 14:11:58 +0200 Subject: ALSA : hda - bug fix on checking the supported power states of a codec The return value of snd_hda_param_read() is -1 for an error, otherwise it's the supported power states of a codec. The supported power states is a 32-bit value. Bit 31 will be set to 1 if the codec supports EPSS, thus making "sup" negative. And the bit 28:5 is reserved as "0". So a negative value other than -1 shall be further checked. Please refer to High-Definition spec 7.3.4.12 "Supported Power States", thanks! Signed-off-by: Mengdong Lin Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 88a9c20eb7a2..629131ad7b8b 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3497,7 +3497,7 @@ static bool snd_hda_codec_get_supported_ps(struct hda_codec *codec, hda_nid_t fg { int sup = snd_hda_param_read(codec, fg, AC_PAR_POWER_STATE); - if (sup < 0) + if (sup == -1) return false; if (sup & power_state) return true; -- cgit v1.2.3 From 61f5d61ef94d7082d96494e2a6dd79de2b4437d2 Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Wed, 8 Aug 2012 11:34:43 +0530 Subject: ASoC: Samsung: Fix build error MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Fixes the following build error: In file included from arch/arm/mach-exynos/include/mach/dma.h:24:0, from arch/arm/plat-samsung/include/plat/dma-ops.h:17, from arch/arm/plat-samsung/include/plat/dma.h:128, from sound/soc/samsung/pcm.c:23: arch/arm/plat-samsung/include/plat/dma-pl330.h:106:8: error: redefinition of ‘struct s3c2410_dma_client’ arch/arm/plat-samsung/include/plat/dma.h:40:8: note: originally defined here make[3]: *** [sound/soc/samsung/pcm.o] Error 1 Signed-off-by: Sachin Kamat Signed-off-by: Sachin Kamat Acked-by: Kukjin Kim Signed-off-by: Mark Brown --- sound/soc/samsung/pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/samsung/pcm.c b/sound/soc/samsung/pcm.c index b7b2a1f91425..89b064650f14 100644 --- a/sound/soc/samsung/pcm.c +++ b/sound/soc/samsung/pcm.c @@ -20,7 +20,7 @@ #include #include -#include +#include #include "dma.h" #include "pcm.h" -- cgit v1.2.3 From 088c820b732dbfd515fc66d459d5f5777f79b406 Mon Sep 17 00:00:00 2001 From: Wang Xingchao Date: Mon, 13 Aug 2012 14:11:10 +0800 Subject: ALSA: hda - fix Copyright debug message As spec said, 1 indicates no copyright is asserted. Signed-off-by: Wang Xingchao Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_proc.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index 7e46258fc700..6894ec66258c 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -412,7 +412,7 @@ static void print_digital_conv(struct snd_info_buffer *buffer, if (digi1 & AC_DIG1_EMPHASIS) snd_iprintf(buffer, " Preemphasis"); if (digi1 & AC_DIG1_COPYRIGHT) - snd_iprintf(buffer, " Copyright"); + snd_iprintf(buffer, " Non-Copyright"); if (digi1 & AC_DIG1_NONAUDIO) snd_iprintf(buffer, " Non-Audio"); if (digi1 & AC_DIG1_PROFESSIONAL) -- cgit v1.2.3 From 14ebd8a8c15e9fed638120bdb93f1a814e13d6a9 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 10 Aug 2012 15:40:12 +0100 Subject: ASoC: wm5102: Add missing input PGA routes Signed-off-by: Mark Brown --- sound/soc/codecs/wm5102.c | 9 +++++++++ 1 file changed, 9 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index 6537f16d383e..496ce9a9d8be 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -639,6 +639,15 @@ static const struct snd_soc_dapm_route wm5102_dapm_routes[] = { { "AIF2 Capture", NULL, "SYSCLK" }, { "AIF3 Capture", NULL, "SYSCLK" }, + { "IN1L PGA", NULL, "IN1L" }, + { "IN1R PGA", NULL, "IN1R" }, + + { "IN2L PGA", NULL, "IN2L" }, + { "IN2R PGA", NULL, "IN2R" }, + + { "IN3L PGA", NULL, "IN3L" }, + { "IN3R PGA", NULL, "IN3R" }, + ARIZONA_MIXER_ROUTES("OUT1L", "HPOUT1L"), ARIZONA_MIXER_ROUTES("OUT1R", "HPOUT1R"), ARIZONA_MIXER_ROUTES("OUT2L", "HPOUT2L"), -- cgit v1.2.3 From 17c3f7e8f3ef796a9db3b22f3797188d0e7ac88c Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 10 Aug 2012 15:40:22 +0100 Subject: ASoC: wm5110: Add missing input PGA routes Signed-off-by: Mark Brown --- sound/soc/codecs/wm5110.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 8033f7065189..01ebbcc5c6a4 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -681,6 +681,18 @@ static const struct snd_soc_dapm_route wm5110_dapm_routes[] = { { "AIF2 Capture", NULL, "SYSCLK" }, { "AIF3 Capture", NULL, "SYSCLK" }, + { "IN1L PGA", NULL, "IN1L" }, + { "IN1R PGA", NULL, "IN1R" }, + + { "IN2L PGA", NULL, "IN2L" }, + { "IN2R PGA", NULL, "IN2R" }, + + { "IN3L PGA", NULL, "IN3L" }, + { "IN3R PGA", NULL, "IN3R" }, + + { "IN4L PGA", NULL, "IN4L" }, + { "IN4R PGA", NULL, "IN4R" }, + ARIZONA_MIXER_ROUTES("OUT1L", "HPOUT1L"), ARIZONA_MIXER_ROUTES("OUT1R", "HPOUT1R"), ARIZONA_MIXER_ROUTES("OUT2L", "HPOUT2L"), -- cgit v1.2.3 From 12022a785328fdf61a3e1a4bc34db0098dabe839 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 13 Aug 2012 16:28:36 +0100 Subject: ASoC: jack: Always notify full jack status Don't just notify for the bits we've updated, notify the full state of the jack otherwise users might get confused by misleading reports. Signed-off-by: Mark Brown --- sound/soc/soc-jack.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index 7f8b3b7428bb..0c172938b82a 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -103,7 +103,7 @@ void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask) } /* Report before the DAPM sync to help users updating micbias status */ - blocking_notifier_call_chain(&jack->notifier, status, jack); + blocking_notifier_call_chain(&jack->notifier, jack->status, jack); snd_soc_dapm_sync(dapm); -- cgit v1.2.3 From 265d931a9e9a7e290faa5e2145f4b2ebf38ea84c Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Mon, 13 Aug 2012 17:10:46 +0200 Subject: ALSA: hda - Fix 'Beep Playback Switch' with no underlying mute switch Some Conexant devices (e g CX20590) have no mute capability on their Beep widgets. This patch makes sure we don't try setting mutes on those widgets. Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_beep.c | 13 +++++++++++-- 1 file changed, 11 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c index d26ae65b43b7..0849aac449f2 100644 --- a/sound/pci/hda/hda_beep.c +++ b/sound/pci/hda/hda_beep.c @@ -231,15 +231,22 @@ void snd_hda_detach_beep_device(struct hda_codec *codec) } EXPORT_SYMBOL_HDA(snd_hda_detach_beep_device); +static bool ctl_has_mute(struct snd_kcontrol *kcontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + return query_amp_caps(codec, get_amp_nid(kcontrol), + get_amp_direction(kcontrol)) & AC_AMPCAP_MUTE; +} + /* get/put callbacks for beep mute mixer switches */ int snd_hda_mixer_amp_switch_get_beep(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct hda_beep *beep = codec->beep; - if (beep && !beep->enabled) { + if (beep && (!beep->enabled || !ctl_has_mute(kcontrol))) { ucontrol->value.integer.value[0] = - ucontrol->value.integer.value[1] = 0; + ucontrol->value.integer.value[1] = beep->enabled; return 0; } return snd_hda_mixer_amp_switch_get(kcontrol, ucontrol); @@ -263,6 +270,8 @@ int snd_hda_mixer_amp_switch_put_beep(struct snd_kcontrol *kcontrol, enable |= *valp; snd_hda_enable_beep_device(codec, enable); } + if (!ctl_has_mute(kcontrol)) + return 0; return snd_hda_mixer_amp_switch_put(kcontrol, ucontrol); } EXPORT_SYMBOL_HDA(snd_hda_mixer_amp_switch_put_beep); -- cgit v1.2.3 From 3bdcff70b6cd049e6f4437b955850f5db83653cc Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 14 Aug 2012 17:42:11 +0200 Subject: ALSA: lx6464es: Add a missing error check Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=44541 Signed-off-by: Takashi Iwai --- sound/pci/lx6464es/lx6464es.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/pci/lx6464es/lx6464es.c b/sound/pci/lx6464es/lx6464es.c index d1ab43706735..5579b08bb35b 100644 --- a/sound/pci/lx6464es/lx6464es.c +++ b/sound/pci/lx6464es/lx6464es.c @@ -851,6 +851,8 @@ static int __devinit lx_pcm_create(struct lx6464es *chip) /* hardcoded device name & channel count */ err = snd_pcm_new(chip->card, (char *)card_name, 0, 1, 1, &pcm); + if (err < 0) + return err; pcm->private_data = chip; -- cgit v1.2.3 From e9ba389c5ffc4dd29dfe17e00e48877302111135 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 15 Aug 2012 12:32:00 +0200 Subject: ALSA: usb-audio: Fix scheduling-while-atomic bug in PCM capture stream A PCM capture stream on usb-audio causes a scheduling-while-atomic BUG, as reported in the bugzilla entry below. It's because snd_usb_endpoint_start() is called at first at trigger START for a capture stream, and this function contains the left-over EP deactivation codes. The problem doesn't happen for a playback stream because the function is called at PCM prepare time, which can sleep. This patch fixes the BUG by moving the EP deactivation code into the PCM prepare callback. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=46011 Cc: [v3.5+] Signed-off-by: Takashi Iwai --- sound/usb/endpoint.c | 4 ---- sound/usb/pcm.c | 3 +++ 2 files changed, 3 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 0f647d22cb4a..c41181202688 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -821,10 +821,6 @@ int snd_usb_endpoint_start(struct snd_usb_endpoint *ep) if (++ep->use_count != 1) return 0; - /* just to be sure */ - deactivate_urbs(ep, 0, 1); - wait_clear_urbs(ep); - ep->active_mask = 0; ep->unlink_mask = 0; ep->phase = 0; diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index a1298f379428..62ec808ed792 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -544,6 +544,9 @@ static int snd_usb_pcm_prepare(struct snd_pcm_substream *substream) subs->last_frame_number = 0; runtime->delay = 0; + /* clear the pending deactivation on the target EPs */ + deactivate_endpoints(subs); + /* for playback, submit the URBs now; otherwise, the first hwptr_done * updates for all URBs would happen at the same time when starting */ if (subs->direction == SNDRV_PCM_STREAM_PLAYBACK) -- cgit v1.2.3 From 5e68fb3cab23b327e9f15803607e697d7eea1966 Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Thu, 16 Aug 2012 14:11:09 +0200 Subject: ALSA: hda - Don't send invalid volume knob command on IDT 92hd75bxx Instead of blindly initializing a volume knob widget, first check that there actually is a volume knob widget. Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 94040ccf8e8f..ea5775a1a7db 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -4272,7 +4272,8 @@ static int stac92xx_init(struct hda_codec *codec) unsigned int gpio; int i; - snd_hda_sequence_write(codec, spec->init); + if (spec->init) + snd_hda_sequence_write(codec, spec->init); /* power down adcs initially */ if (spec->powerdown_adcs) @@ -5748,7 +5749,6 @@ again: /* fallthru */ case 0x111d76b4: /* 6 Port without Analog Mixer */ case 0x111d76b5: - spec->init = stac92hd71bxx_core_init; codec->slave_dig_outs = stac92hd71bxx_slave_dig_outs; spec->num_dmics = stac92xx_connected_ports(codec, stac92hd71bxx_dmic_nids, @@ -5773,7 +5773,6 @@ again: spec->stream_delay = 40; /* 40 milliseconds */ /* disable VSW */ - spec->init = stac92hd71bxx_core_init; unmute_init++; snd_hda_codec_set_pincfg(codec, 0x0f, 0x40f000f0); snd_hda_codec_set_pincfg(codec, 0x19, 0x40f000f3); @@ -5788,7 +5787,6 @@ again: /* fallthru */ default: - spec->init = stac92hd71bxx_core_init; codec->slave_dig_outs = stac92hd71bxx_slave_dig_outs; spec->num_dmics = stac92xx_connected_ports(codec, stac92hd71bxx_dmic_nids, @@ -5796,6 +5794,9 @@ again: break; } + if (get_wcaps_type(get_wcaps(codec, 0x28)) == AC_WID_VOL_KNB) + spec->init = stac92hd71bxx_core_init; + if (get_wcaps(codec, 0xa) & AC_WCAP_IN_AMP) snd_hda_sequence_write_cache(codec, unmute_init); -- cgit v1.2.3 From 939d5044b117302cabdd30833685d9f214e9bff6 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 16 Aug 2012 13:08:23 +0100 Subject: ASoC: wm5102: Remove DRC2 It will be removed from future device revisions. Signed-off-by: Mark Brown --- sound/soc/codecs/wm5102.c | 16 ---------------- 1 file changed, 16 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index 496ce9a9d8be..e33d327396ad 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -128,13 +128,9 @@ SOC_SINGLE_TLV("EQ4 B5 Volume", ARIZONA_EQ4_2, ARIZONA_EQ4_B5_GAIN_SHIFT, ARIZONA_MIXER_CONTROLS("DRC1L", ARIZONA_DRC1LMIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("DRC1R", ARIZONA_DRC1RMIX_INPUT_1_SOURCE), -ARIZONA_MIXER_CONTROLS("DRC2L", ARIZONA_DRC2LMIX_INPUT_1_SOURCE), -ARIZONA_MIXER_CONTROLS("DRC2R", ARIZONA_DRC2RMIX_INPUT_1_SOURCE), SND_SOC_BYTES_MASK("DRC1", ARIZONA_DRC1_CTRL1, 5, ARIZONA_DRC1R_ENA | ARIZONA_DRC1L_ENA), -SND_SOC_BYTES_MASK("DRC2", ARIZONA_DRC2_CTRL1, 5, - ARIZONA_DRC2R_ENA | ARIZONA_DRC2L_ENA), ARIZONA_MIXER_CONTROLS("LHPF1", ARIZONA_HPLP1MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("LHPF2", ARIZONA_HPLP2MIX_INPUT_1_SOURCE), @@ -236,8 +232,6 @@ ARIZONA_MIXER_ENUMS(EQ4, ARIZONA_EQ4MIX_INPUT_1_SOURCE); ARIZONA_MIXER_ENUMS(DRC1L, ARIZONA_DRC1LMIX_INPUT_1_SOURCE); ARIZONA_MIXER_ENUMS(DRC1R, ARIZONA_DRC1RMIX_INPUT_1_SOURCE); -ARIZONA_MIXER_ENUMS(DRC2L, ARIZONA_DRC2LMIX_INPUT_1_SOURCE); -ARIZONA_MIXER_ENUMS(DRC2R, ARIZONA_DRC2RMIX_INPUT_1_SOURCE); ARIZONA_MIXER_ENUMS(LHPF1, ARIZONA_HPLP1MIX_INPUT_1_SOURCE); ARIZONA_MIXER_ENUMS(LHPF2, ARIZONA_HPLP2MIX_INPUT_1_SOURCE); @@ -349,10 +343,6 @@ SND_SOC_DAPM_PGA("DRC1L", ARIZONA_DRC1_CTRL1, ARIZONA_DRC1L_ENA_SHIFT, 0, NULL, 0), SND_SOC_DAPM_PGA("DRC1R", ARIZONA_DRC1_CTRL1, ARIZONA_DRC1R_ENA_SHIFT, 0, NULL, 0), -SND_SOC_DAPM_PGA("DRC2L", ARIZONA_DRC2_CTRL1, ARIZONA_DRC2L_ENA_SHIFT, 0, - NULL, 0), -SND_SOC_DAPM_PGA("DRC2R", ARIZONA_DRC2_CTRL1, ARIZONA_DRC2R_ENA_SHIFT, 0, - NULL, 0), SND_SOC_DAPM_PGA("LHPF1", ARIZONA_HPLPF1_1, ARIZONA_LHPF1_ENA_SHIFT, 0, NULL, 0), @@ -466,8 +456,6 @@ ARIZONA_MIXER_WIDGETS(EQ4, "EQ4"), ARIZONA_MIXER_WIDGETS(DRC1L, "DRC1L"), ARIZONA_MIXER_WIDGETS(DRC1R, "DRC1R"), -ARIZONA_MIXER_WIDGETS(DRC2L, "DRC2L"), -ARIZONA_MIXER_WIDGETS(DRC2R, "DRC2R"), ARIZONA_MIXER_WIDGETS(LHPF1, "LHPF1"), ARIZONA_MIXER_WIDGETS(LHPF2, "LHPF2"), @@ -553,8 +541,6 @@ SND_SOC_DAPM_OUTPUT("SPKDAT1R"), { name, "EQ4", "EQ4" }, \ { name, "DRC1L", "DRC1L" }, \ { name, "DRC1R", "DRC1R" }, \ - { name, "DRC2L", "DRC2L" }, \ - { name, "DRC2R", "DRC2R" }, \ { name, "LHPF1", "LHPF1" }, \ { name, "LHPF2", "LHPF2" }, \ { name, "LHPF3", "LHPF3" }, \ @@ -684,8 +670,6 @@ static const struct snd_soc_dapm_route wm5102_dapm_routes[] = { ARIZONA_MIXER_ROUTES("DRC1L", "DRC1L"), ARIZONA_MIXER_ROUTES("DRC1R", "DRC1R"), - ARIZONA_MIXER_ROUTES("DRC2L", "DRC2L"), - ARIZONA_MIXER_ROUTES("DRC2R", "DRC2R"), ARIZONA_MIXER_ROUTES("LHPF1", "LHPF1"), ARIZONA_MIXER_ROUTES("LHPF2", "LHPF2"), -- cgit v1.2.3 From ccf795847a38235ee4a56a24129ce75147d6ba8f Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 16 Aug 2012 22:36:04 +0100 Subject: ASoC: wm9712: Fix microphone source selection Currently the microphone input source is not selectable as while there is a DAPM widget it's not connected to anything so it won't be properly instantiated. Add something more correct for the input structure to get things going, even though it's not hooked into the rest of the routing map and so won't actually achieve anything except allowing the relevant register bits to be written. Reported-by: Christop Fritz Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/wm9712.c | 19 +++++++++++++++++-- 1 file changed, 17 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index f16fb361a4eb..fd74b8843d34 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -272,7 +272,7 @@ SOC_DAPM_ENUM("Route", wm9712_enum[9]); /* Mic select */ static const struct snd_kcontrol_new wm9712_mic_src_controls = -SOC_DAPM_ENUM("Route", wm9712_enum[7]); +SOC_DAPM_ENUM("Mic Source Select", wm9712_enum[7]); /* diff select */ static const struct snd_kcontrol_new wm9712_diff_sel_controls = @@ -291,7 +291,9 @@ SND_SOC_DAPM_MUX("Left Capture Select", SND_SOC_NOPM, 0, 0, &wm9712_capture_selectl_controls), SND_SOC_DAPM_MUX("Right Capture Select", SND_SOC_NOPM, 0, 0, &wm9712_capture_selectr_controls), -SND_SOC_DAPM_MUX("Mic Select Source", SND_SOC_NOPM, 0, 0, +SND_SOC_DAPM_MUX("Left Mic Select Source", SND_SOC_NOPM, 0, 0, + &wm9712_mic_src_controls), +SND_SOC_DAPM_MUX("Right Mic Select Source", SND_SOC_NOPM, 0, 0, &wm9712_mic_src_controls), SND_SOC_DAPM_MUX("Differential Source", SND_SOC_NOPM, 0, 0, &wm9712_diff_sel_controls), @@ -319,6 +321,7 @@ SND_SOC_DAPM_PGA("Out 3 PGA", AC97_INT_PAGING, 5, 1, NULL, 0), SND_SOC_DAPM_PGA("Line PGA", AC97_INT_PAGING, 2, 1, NULL, 0), SND_SOC_DAPM_PGA("Phone PGA", AC97_INT_PAGING, 1, 1, NULL, 0), SND_SOC_DAPM_PGA("Mic PGA", AC97_INT_PAGING, 0, 1, NULL, 0), +SND_SOC_DAPM_PGA("Differential Mic", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_MICBIAS("Mic Bias", AC97_INT_PAGING, 10, 1), SND_SOC_DAPM_OUTPUT("MONOOUT"), SND_SOC_DAPM_OUTPUT("HPOUTL"), @@ -379,6 +382,18 @@ static const struct snd_soc_dapm_route wm9712_audio_map[] = { {"Mic PGA", NULL, "MIC1"}, {"Mic PGA", NULL, "MIC2"}, + /* microphones */ + {"Differential Mic", NULL, "MIC1"}, + {"Differential Mic", NULL, "MIC2"}, + {"Left Mic Select Source", "Mic 1", "MIC1"}, + {"Left Mic Select Source", "Mic 2", "MIC2"}, + {"Left Mic Select Source", "Stereo", "MIC1"}, + {"Left Mic Select Source", "Differential", "Differential Mic"}, + {"Right Mic Select Source", "Mic 1", "MIC1"}, + {"Right Mic Select Source", "Mic 2", "MIC2"}, + {"Right Mic Select Source", "Stereo", "MIC2"}, + {"Right Mic Select Source", "Differential", "Differential Mic"}, + /* left capture selector */ {"Left Capture Select", "Mic", "MIC1"}, {"Left Capture Select", "Speaker Mixer", "Speaker Mixer"}, -- cgit v1.2.3 From 28c42c28309244d0b15d1b385e33429d59997679 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 31 Jul 2012 18:37:28 +0100 Subject: ASoC: wm9712: Fix inverted capture volume The capture volume increases with the register value so it shouldn't be flagged as inverted. Reported-by: Christoph Fritz Signed-off-by: Mark Brown --- sound/soc/codecs/wm9712.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index fd74b8843d34..c6d2076a796b 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -148,7 +148,7 @@ SOC_SINGLE("Treble Volume", AC97_MASTER_TONE, 0, 15, 1), SOC_SINGLE("Capture ADC Switch", AC97_REC_GAIN, 15, 1, 1), SOC_ENUM("Capture Volume Steps", wm9712_enum[6]), -SOC_DOUBLE("Capture Volume", AC97_REC_GAIN, 8, 0, 63, 1), +SOC_DOUBLE("Capture Volume", AC97_REC_GAIN, 8, 0, 63, 0), SOC_SINGLE("Capture ZC Switch", AC97_REC_GAIN, 7, 1, 0), SOC_SINGLE_TLV("Mic 1 Volume", AC97_MIC, 8, 31, 1, main_tlv), -- cgit v1.2.3 From 94f3ec6b2222eb5c0af0c784f0656ff5b909d870 Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Sat, 18 Aug 2012 18:55:15 +0300 Subject: sound: oss/sb_audio: prevent divide by zero bug Speed comes from get_user() in audio_ioctl(). We use it to set the "s" variable before clamping it to valid values so it could lead to a divide by zero bug. Signed-off-by: Dan Carpenter Signed-off-by: Takashi Iwai --- sound/oss/sb_audio.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/oss/sb_audio.c b/sound/oss/sb_audio.c index 733b014ec7d1..b2b3c014221a 100644 --- a/sound/oss/sb_audio.c +++ b/sound/oss/sb_audio.c @@ -575,13 +575,15 @@ static int jazz16_audio_set_speed(int dev, int speed) if (speed > 0) { int tmp; - int s = speed * devc->channels; + int s; if (speed < 5000) speed = 5000; if (speed > 44100) speed = 44100; + s = speed * devc->channels; + devc->tconst = (256 - ((1000000 + s / 2) / s)) & 0xff; tmp = 256 - devc->tconst; -- cgit v1.2.3 From aaf265c22e48f10c94ad04d23b6ab0c88f554d35 Mon Sep 17 00:00:00 2001 From: Julia Lawall Date: Sun, 19 Aug 2012 09:02:58 +0200 Subject: ALSA: sound/atmel/abdac.c: fix error return code Initialize retval before returning from a failed call to ioremap. A simplified version of the semantic match that finds this problem is as follows: (http://coccinelle.lip6.fr/) // ( if@p1 (\(ret < 0\|ret != 0\)) { ... return ret; } | ret@p1 = 0 ) ... when != ret = e1 when != &ret *if(...) { ... when != ret = e2 when forall return ret; } // Signed-off-by: Julia Lawall Signed-off-by: Takashi Iwai --- sound/atmel/abdac.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/atmel/abdac.c b/sound/atmel/abdac.c index 98554f4882b7..277ebce23a45 100644 --- a/sound/atmel/abdac.c +++ b/sound/atmel/abdac.c @@ -452,6 +452,7 @@ static int __devinit atmel_abdac_probe(struct platform_device *pdev) dac->regs = ioremap(regs->start, resource_size(regs)); if (!dac->regs) { dev_dbg(&pdev->dev, "could not remap register memory\n"); + retval = -ENOMEM; goto out_free_card; } -- cgit v1.2.3 From 0c23e46eb4878422c25351ff54ab0fe80c643809 Mon Sep 17 00:00:00 2001 From: Julia Lawall Date: Sun, 19 Aug 2012 09:02:57 +0200 Subject: ALSA: sound/atmel/ac97c.c: fix error return code In the first case, the second test of whether retval is negative is redundant. It is dropped and the previous and subsequent tests are combined. In the second case, add an initialization of retval on failure of ioremap. A simplified version of the semantic match that finds this problem is as follows: (http://coccinelle.lip6.fr/) // ( if@p1 (\(ret < 0\|ret != 0\)) { ... return ret; } | ret@p1 = 0 ) ... when != ret = e1 when != &ret *if(...) { ... when != ret = e2 when forall return ret; } // Signed-off-by: Julia Lawall Signed-off-by: Takashi Iwai --- sound/atmel/ac97c.c | 12 ++++-------- 1 file changed, 4 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c index 3c8d3ba7ddfc..9052aff37f64 100644 --- a/sound/atmel/ac97c.c +++ b/sound/atmel/ac97c.c @@ -278,14 +278,9 @@ static int atmel_ac97c_capture_hw_params(struct snd_pcm_substream *substream, if (retval < 0) return retval; /* snd_pcm_lib_malloc_pages returns 1 if buffer is changed. */ - if (cpu_is_at32ap7000()) { - if (retval < 0) - return retval; - /* snd_pcm_lib_malloc_pages returns 1 if buffer is changed. */ - if (retval == 1) - if (test_and_clear_bit(DMA_RX_READY, &chip->flags)) - dw_dma_cyclic_free(chip->dma.rx_chan); - } + if (cpu_is_at32ap7000() && retval == 1) + if (test_and_clear_bit(DMA_RX_READY, &chip->flags)) + dw_dma_cyclic_free(chip->dma.rx_chan); /* Set restrictions to params. */ mutex_lock(&opened_mutex); @@ -980,6 +975,7 @@ static int __devinit atmel_ac97c_probe(struct platform_device *pdev) if (!chip->regs) { dev_dbg(&pdev->dev, "could not remap register memory\n"); + retval = -ENOMEM; goto err_ioremap; } -- cgit v1.2.3 From 4d8ce1c9966663bad69e738952179f3cc52710bf Mon Sep 17 00:00:00 2001 From: Julia Lawall Date: Sun, 19 Aug 2012 09:02:56 +0200 Subject: ALSA: sound/pci/ctxfi/ctatc.c: fix error return code Initialize err before returning on failure, as done elsewhere in the function. A simplified version of the semantic match that finds this problem is as follows: (http://coccinelle.lip6.fr/) // ( if@p1 (\(ret < 0\|ret != 0\)) { ... return ret; } | ret@p1 = 0 ) ... when != ret = e1 when != &ret *if(...) { ... when != ret = e2 when forall return ret; } // Signed-off-by: Julia Lawall Signed-off-by: Takashi Iwai --- sound/pci/ctxfi/ctatc.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/ctxfi/ctatc.c b/sound/pci/ctxfi/ctatc.c index 8e40262d4117..2f6e9c762d3f 100644 --- a/sound/pci/ctxfi/ctatc.c +++ b/sound/pci/ctxfi/ctatc.c @@ -1725,8 +1725,10 @@ int __devinit ct_atc_create(struct snd_card *card, struct pci_dev *pci, atc_connect_resources(atc); atc->timer = ct_timer_new(atc); - if (!atc->timer) + if (!atc->timer) { + err = -ENOMEM; goto error1; + } err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, atc, &ops); if (err < 0) -- cgit v1.2.3 From ae970eb45d8a1ea4506be23c3f776225b9575d0e Mon Sep 17 00:00:00 2001 From: Julia Lawall Date: Sun, 19 Aug 2012 09:02:55 +0200 Subject: ALSA: sound/pci/sis7019.c: fix error return code Initialize rc before returning on failure, as done elsewhere in the function. A simplified version of the semantic match that finds this problem is as follows: (http://coccinelle.lip6.fr/) // ( if@p1 (\(ret < 0\|ret != 0\)) { ... return ret; } | ret@p1 = 0 ) ... when != ret = e1 when != &ret *if(...) { ... when != ret = e2 when forall return ret; } // Signed-off-by: Julia Lawall Signed-off-by: Takashi Iwai --- sound/pci/sis7019.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/sis7019.c b/sound/pci/sis7019.c index 512434efcc31..805ab6e9a78f 100644 --- a/sound/pci/sis7019.c +++ b/sound/pci/sis7019.c @@ -1377,8 +1377,9 @@ static int __devinit sis_chip_create(struct snd_card *card, if (rc) goto error_out_cleanup; - if (request_irq(pci->irq, sis_interrupt, IRQF_SHARED, KBUILD_MODNAME, - sis)) { + rc = request_irq(pci->irq, sis_interrupt, IRQF_SHARED, KBUILD_MODNAME, + sis); + if (rc) { dev_err(&pci->dev, "unable to allocate irq %d\n", sis->irq); goto error_out_cleanup; } -- cgit v1.2.3 From b17cbdd85f84c8323189416da6e9701d2793b0e5 Mon Sep 17 00:00:00 2001 From: Julia Lawall Date: Sun, 19 Aug 2012 09:02:54 +0200 Subject: ALSA: sound/pci/rme9652/hdspm.c: fix error return code Convert a nonnegative error return code to a negative one, as returned elsewhere in the function. A simplified version of the semantic match that finds this problem is as follows: (http://coccinelle.lip6.fr/) // ( if@p1 (\(ret < 0\|ret != 0\)) { ... return ret; } | ret@p1 = 0 ) ... when != ret = e1 when != &ret *if(...) { ... when != ret = e2 when forall return ret; } // Signed-off-by: Julia Lawall Signed-off-by: Takashi Iwai --- sound/pci/rme9652/hdspm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index b8ac8710f47f..b12308b5ba2a 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -6585,7 +6585,7 @@ static int __devinit snd_hdspm_create(struct snd_card *card, snd_printk(KERN_ERR "HDSPM: " "unable to kmalloc Mixer memory of %d Bytes\n", (int)sizeof(struct hdspm_mixer)); - return err; + return -ENOMEM; } hdspm->port_names_in = NULL; -- cgit v1.2.3 From c86b93628e5649fd7bb0574b570a51b2b02d586c Mon Sep 17 00:00:00 2001 From: Julia Lawall Date: Sun, 19 Aug 2012 09:02:59 +0200 Subject: ALSA: sound/ppc/snd_ps3.c: fix error return code Initialize ret before returning on failure, as done elsewhere in the function. A simplified version of the semantic match that finds this problem is as follows: (http://coccinelle.lip6.fr/) // ( if@p1 (\(ret < 0\|ret != 0\)) { ... return ret; } | ret@p1 = 0 ) ... when != ret = e1 when != &ret *if(...) { ... when != ret = e2 when forall return ret; } // Signed-off-by: Julia Lawall Signed-off-by: Takashi Iwai --- sound/ppc/snd_ps3.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/ppc/snd_ps3.c b/sound/ppc/snd_ps3.c index 1aa52eff526a..9b18b5243a56 100644 --- a/sound/ppc/snd_ps3.c +++ b/sound/ppc/snd_ps3.c @@ -1040,6 +1040,7 @@ static int __devinit snd_ps3_driver_probe(struct ps3_system_bus_device *dev) GFP_KERNEL); if (!the_card.null_buffer_start_vaddr) { pr_info("%s: nullbuffer alloc failed\n", __func__); + ret = -ENOMEM; goto clean_preallocate; } pr_debug("%s: null vaddr=%p dma=%#llx\n", __func__, -- cgit v1.2.3 From c41999a23929f30808bae6009d8065052d4d73fd Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Mon, 20 Aug 2012 11:17:00 +0200 Subject: ALSA: hda - don't create dysfunctional mixer controls for ca0132 It's possible that these amps are settable somehow, e g through secret codec verbs, but for now, don't create the controls (as they won't be working anyway, and cause errors in amixer). Cc: stable@kernel.org BugLink: https://bugs.launchpad.net/bugs/1038651 Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_ca0132.c | 8 ++++++++ 1 file changed, 8 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 9c0ec0a55bef..49750a96d649 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -275,6 +275,10 @@ static int _add_switch(struct hda_codec *codec, hda_nid_t nid, const char *pfx, int type = dir ? HDA_INPUT : HDA_OUTPUT; struct snd_kcontrol_new knew = HDA_CODEC_MUTE_MONO(namestr, nid, chan, 0, type); + if ((query_amp_caps(codec, nid, type) & AC_AMPCAP_MUTE) == 0) { + snd_printdd("Skipping '%s %s Switch' (no mute on node 0x%x)\n", pfx, dirstr[dir], nid); + return 0; + } sprintf(namestr, "%s %s Switch", pfx, dirstr[dir]); return snd_hda_ctl_add(codec, nid, snd_ctl_new1(&knew, codec)); } @@ -286,6 +290,10 @@ static int _add_volume(struct hda_codec *codec, hda_nid_t nid, const char *pfx, int type = dir ? HDA_INPUT : HDA_OUTPUT; struct snd_kcontrol_new knew = HDA_CODEC_VOLUME_MONO(namestr, nid, chan, 0, type); + if ((query_amp_caps(codec, nid, type) & AC_AMPCAP_NUM_STEPS) == 0) { + snd_printdd("Skipping '%s %s Volume' (no amp on node 0x%x)\n", pfx, dirstr[dir], nid); + return 0; + } sprintf(namestr, "%s %s Volume", pfx, dirstr[dir]); return snd_hda_ctl_add(codec, nid, snd_ctl_new1(&knew, codec)); } -- cgit v1.2.3 From 535b6c51fe8293c88ce919cdfc4390c67a1cb6d1 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 20 Aug 2012 21:25:22 +0200 Subject: ALSA: hda - Fix leftover codec->power_transition When the codec turn-on operation is canceled by the immediate power-on, the driver left the power_transition flag as is. This caused the persistent avoidance of power-save behavior. Cc: [v3.5+] Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index c3077d5dec6e..f560051a949e 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -4454,6 +4454,8 @@ static void __snd_hda_power_up(struct hda_codec *codec, bool wait_power_down) * then there is no need to go through power up here. */ if (codec->power_on) { + if (codec->power_transition < 0) + codec->power_transition = 0; spin_unlock(&codec->power_lock); return; } -- cgit v1.2.3 From 53e1719f3da0f095b8db1461bd12dd79f3246b84 Mon Sep 17 00:00:00 2001 From: Ondrej Zary Date: Mon, 20 Aug 2012 21:50:13 +0200 Subject: ALSA: snd-als100: fix suspend/resume snd_card_als100_probe() does not set pcm field in struct snd_sb. As a result, PCM is not suspended and applications don't know that they need to resume the playback. Tested with Labway A381-F20 card (ALS120). Signed-off-by: Ondrej Zary Signed-off-by: Takashi Iwai --- sound/isa/als100.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/isa/als100.c b/sound/isa/als100.c index 2d67c78c9f4b..f7cdaf51512d 100644 --- a/sound/isa/als100.c +++ b/sound/isa/als100.c @@ -233,7 +233,7 @@ static int __devinit snd_card_als100_probe(int dev, irq[dev], dma8[dev], dma16[dev]); } - if ((error = snd_sb16dsp_pcm(chip, 0, NULL)) < 0) { + if ((error = snd_sb16dsp_pcm(chip, 0, &chip->pcm)) < 0) { snd_card_free(card); return error; } -- cgit v1.2.3