From f5f76ea75dce553631ffb08abc44dcecb68e74d4 Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Mon, 11 Jan 2016 15:17:23 +0000 Subject: ASoC: qcom: use correct device pointer in dma allocation dev pointer in struct snd_soc_pcm_runtime does not have dma_ops set. In v4.4 kernel dma_ops would end up pointing to dummy_dma_ops in such cases. So attempting to use such device in allocating coherent memory on aarch64 would fail. According to commit 1dccb598df549d892b6450c261da54cdd7af44b4 ("arm64: simplify dma_get_ops") The current behavior of dma_get_ops is to fall back to the global dma_ops when a device has not set its own dma_ops, but only for DT based systems. So, this patch fixes the driver to use correct device pointer while allocating coherent memory, and also deletes un-necessary dma_mask setup on soc_runtime->dev. Without this patch lpass driver would fail with below log: ... [ 6.541542] ADV7533: lpass_platform_alloc_buffer: Could not allocate DMA buffer [ 6.541914] apq8016-lpass-cpu 7708000.lpass-cpu: ASoC: pcm constructor failed: -12 [ 6.548216] qcom-apq8016-sbc 7702000.sound: ASoC: can't create pcm ADV7533 :-12 [ 6.555581] qcom-apq8016-sbc 7702000.sound: ASoC: failed to instantiate card -12 [ 6.566072] qcom-apq8016-sbc: probe of 7702000.sound failed with error -12 ... Signed-off-by: Srinivas Kandagatla Signed-off-by: Mark Brown --- sound/soc/qcom/lpass-platform.c | 15 ++++++--------- 1 file changed, 6 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/soc/qcom/lpass-platform.c b/sound/soc/qcom/lpass-platform.c index 79688aa1941a..4aeb8e1a7160 100644 --- a/sound/soc/qcom/lpass-platform.c +++ b/sound/soc/qcom/lpass-platform.c @@ -440,18 +440,18 @@ static irqreturn_t lpass_platform_lpaif_irq(int irq, void *data) } static int lpass_platform_alloc_buffer(struct snd_pcm_substream *substream, - struct snd_soc_pcm_runtime *soc_runtime) + struct snd_soc_pcm_runtime *rt) { struct snd_dma_buffer *buf = &substream->dma_buffer; size_t size = lpass_platform_pcm_hardware.buffer_bytes_max; buf->dev.type = SNDRV_DMA_TYPE_DEV; - buf->dev.dev = soc_runtime->dev; + buf->dev.dev = rt->platform->dev; buf->private_data = NULL; - buf->area = dma_alloc_coherent(soc_runtime->dev, size, &buf->addr, + buf->area = dma_alloc_coherent(rt->platform->dev, size, &buf->addr, GFP_KERNEL); if (!buf->area) { - dev_err(soc_runtime->dev, "%s: Could not allocate DMA buffer\n", + dev_err(rt->platform->dev, "%s: Could not allocate DMA buffer\n", __func__); return -ENOMEM; } @@ -461,12 +461,12 @@ static int lpass_platform_alloc_buffer(struct snd_pcm_substream *substream, } static void lpass_platform_free_buffer(struct snd_pcm_substream *substream, - struct snd_soc_pcm_runtime *soc_runtime) + struct snd_soc_pcm_runtime *rt) { struct snd_dma_buffer *buf = &substream->dma_buffer; if (buf->area) { - dma_free_coherent(soc_runtime->dev, buf->bytes, buf->area, + dma_free_coherent(rt->dev, buf->bytes, buf->area, buf->addr); } buf->area = NULL; @@ -499,9 +499,6 @@ static int lpass_platform_pcm_new(struct snd_soc_pcm_runtime *soc_runtime) snd_soc_pcm_set_drvdata(soc_runtime, data); - soc_runtime->dev->coherent_dma_mask = DMA_BIT_MASK(32); - soc_runtime->dev->dma_mask = &soc_runtime->dev->coherent_dma_mask; - ret = lpass_platform_alloc_buffer(substream, soc_runtime); if (ret) return ret; -- cgit v1.2.1 From cde6bcd584b1b910d6ee8d6eb968ea5d20815444 Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Wed, 13 Jan 2016 15:20:02 +0300 Subject: ASoC: AMD: free memory on error Static checkers complain if we don't free "adata" before returning. Fixes: 7c31335a03b6 ('ASoC: AMD: add AMD ASoC ACP 2.x DMA driver') Signed-off-by: Dan Carpenter Signed-off-by: Mark Brown --- sound/soc/amd/acp-pcm-dma.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index 3191e0a7d273..d1fb035f44db 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -635,6 +635,7 @@ static int acp_dma_open(struct snd_pcm_substream *substream) SNDRV_PCM_HW_PARAM_PERIODS); if (ret < 0) { dev_err(prtd->platform->dev, "set integer constraint failed\n"); + kfree(adata); return ret; } -- cgit v1.2.1 From 1ca2cf8c4167c2016d9716998b4f89c4e79d1f89 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Tue, 12 Jan 2016 15:55:17 +0800 Subject: ASoC: rt5659: Fix irq leak Use devm_request_threaded_irq to ensure the irq is freed when unload the module. The rt5659->i2c is no longer used after this conversion, thus remove it as well. Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/rt5659.c | 16 ++++------------ sound/soc/codecs/rt5659.h | 1 - 2 files changed, 4 insertions(+), 13 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt5659.c b/sound/soc/codecs/rt5659.c index 820d8fa62b5e..c166d9394c69 100644 --- a/sound/soc/codecs/rt5659.c +++ b/sound/soc/codecs/rt5659.c @@ -3985,7 +3985,6 @@ static int rt5659_i2c_probe(struct i2c_client *i2c, if (rt5659 == NULL) return -ENOMEM; - rt5659->i2c = i2c; i2c_set_clientdata(i2c, rt5659); if (pdata) @@ -4157,24 +4156,17 @@ static int rt5659_i2c_probe(struct i2c_client *i2c, INIT_DELAYED_WORK(&rt5659->jack_detect_work, rt5659_jack_detect_work); - if (rt5659->i2c->irq) { - ret = request_threaded_irq(rt5659->i2c->irq, NULL, rt5659_irq, - IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING + if (i2c->irq) { + ret = devm_request_threaded_irq(&i2c->dev, i2c->irq, NULL, + rt5659_irq, IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING | IRQF_ONESHOT, "rt5659", rt5659); if (ret) dev_err(&i2c->dev, "Failed to reguest IRQ: %d\n", ret); } - ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5659, + return snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5659, rt5659_dai, ARRAY_SIZE(rt5659_dai)); - - if (ret) { - if (rt5659->i2c->irq) - free_irq(rt5659->i2c->irq, rt5659); - } - - return 0; } static int rt5659_i2c_remove(struct i2c_client *i2c) diff --git a/sound/soc/codecs/rt5659.h b/sound/soc/codecs/rt5659.h index 8f07ee903eaa..d31c9e5bcec8 100644 --- a/sound/soc/codecs/rt5659.h +++ b/sound/soc/codecs/rt5659.h @@ -1792,7 +1792,6 @@ struct rt5659_priv { struct snd_soc_codec *codec; struct rt5659_platform_data pdata; struct regmap *regmap; - struct i2c_client *i2c; struct gpio_desc *gpiod_ldo1_en; struct gpio_desc *gpiod_reset; struct snd_soc_jack *hs_jack; -- cgit v1.2.1 From ec3995da27e782cc407ce48101c98c19c9ce738d Mon Sep 17 00:00:00 2001 From: Arnd Bergmann Date: Wed, 13 Jan 2016 23:14:54 +0100 Subject: ASoC: mediatek: add i2c dependency The newly added mediatek drivers for mt8173 select codes that depend on I2C, which cuases a build failure if I2C is disabled: warning: (SND_SOC_ADAU1761_I2C && SND_SOC_ADAU1781_I2C && SND_SOC_ADAU1977_I2C && SND_SOC_RT5677 && EXTCON_MAX14577 && EXTCON_MAX77693 && EXTCON_MAX77843 && BMC150_ACCEL_I2C && BMG160_I2C) selects REGMAP_I2C which has unmet direct dependencies (I2C) codecs/rt5645.c:3854:1: warning: data definition has no type or storage class codecs/rt5645.c:3854:1: error: type defaults to 'int' in declaration of 'module_i2c_driver' [-Werror=implicit-int] codecs/rt5677.c:5270:1: warning: data definition has no type or storage class 77_i2c_driver); codecs/rt5677.c:5270:1: error: type defaults to 'int' in declaration of 'module_i2c_driver' [-Werror=implicit-int] This adds an explicit dependency. Signed-off-by: Arnd Bergmann Acked-by: Koro Chen Signed-off-by: Mark Brown --- sound/soc/mediatek/Kconfig | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/mediatek/Kconfig b/sound/soc/mediatek/Kconfig index 15c04e2eae34..976967675387 100644 --- a/sound/soc/mediatek/Kconfig +++ b/sound/soc/mediatek/Kconfig @@ -9,7 +9,7 @@ config SND_SOC_MEDIATEK config SND_SOC_MT8173_MAX98090 tristate "ASoC Audio driver for MT8173 with MAX98090 codec" - depends on SND_SOC_MEDIATEK + depends on SND_SOC_MEDIATEK && I2C select SND_SOC_MAX98090 help This adds ASoC driver for Mediatek MT8173 boards @@ -19,7 +19,7 @@ config SND_SOC_MT8173_MAX98090 config SND_SOC_MT8173_RT5650_RT5676 tristate "ASoC Audio driver for MT8173 with RT5650 RT5676 codecs" - depends on SND_SOC_MEDIATEK + depends on SND_SOC_MEDIATEK && I2C select SND_SOC_RT5645 select SND_SOC_RT5677 help -- cgit v1.2.1 From 6d514c720219a4c0e1c2612c1d830592bfaf5a03 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Thu, 21 Jan 2016 14:10:48 +0800 Subject: ASoC: rt286: fix capture doesn't work at some cases RT286_CBJ_CTRL1(0x4f) bit 10 is needed for headset capture. It will be turned off when "VREF" widget is on and be turned on when bias level is ON. It is odd. And if "VREF" is turned on in bias level is ON, RT286_CBJ_CTRL1(0x4f) bit 10 will be turned off. This patch move the bit control from rt286_set_bias_level and rt298_vref_event to rt286_jack_detect. So it will be turned on once a jack is plugged in. Signed-off-by: Bard Liao Signed-off-by: Mark Brown --- sound/soc/codecs/rt286.c | 26 +++----------------------- 1 file changed, 3 insertions(+), 23 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index af2ed774b552..af30b062f57a 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -266,6 +266,8 @@ static int rt286_jack_detect(struct rt286_priv *rt286, bool *hp, bool *mic) } else { *mic = false; regmap_write(rt286->regmap, RT286_SET_MIC1, 0x20); + regmap_update_bits(rt286->regmap, + RT286_CBJ_CTRL1, 0x0400, 0x0000); } } else { regmap_read(rt286->regmap, RT286_GET_HP_SENSE, &buf); @@ -470,24 +472,6 @@ static int rt286_set_dmic1_event(struct snd_soc_dapm_widget *w, return 0; } -static int rt286_vref_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) -{ - struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); - - switch (event) { - case SND_SOC_DAPM_PRE_PMU: - snd_soc_update_bits(codec, - RT286_CBJ_CTRL1, 0x0400, 0x0000); - mdelay(50); - break; - default: - return 0; - } - - return 0; -} - static int rt286_ldo2_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -536,7 +520,7 @@ static const struct snd_soc_dapm_widget rt286_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY_S("HV", 1, RT286_POWER_CTRL1, 12, 1, NULL, 0), SND_SOC_DAPM_SUPPLY("VREF", RT286_POWER_CTRL1, - 0, 1, rt286_vref_event, SND_SOC_DAPM_PRE_PMU), + 0, 1, NULL, 0), SND_SOC_DAPM_SUPPLY_S("LDO1", 1, RT286_POWER_CTRL2, 2, 0, NULL, 0), SND_SOC_DAPM_SUPPLY_S("LDO2", 2, RT286_POWER_CTRL1, @@ -910,8 +894,6 @@ static int rt286_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_ON: mdelay(10); - snd_soc_update_bits(codec, - RT286_CBJ_CTRL1, 0x0400, 0x0400); snd_soc_update_bits(codec, RT286_DC_GAIN, 0x200, 0x0); @@ -920,8 +902,6 @@ static int rt286_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_STANDBY: snd_soc_write(codec, RT286_SET_AUDIO_POWER, AC_PWRST_D3); - snd_soc_update_bits(codec, - RT286_CBJ_CTRL1, 0x0400, 0x0000); break; default: -- cgit v1.2.1 From b28785fa9cede0d4f47310ca0dd2a4e1d50478b5 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Thu, 21 Jan 2016 13:13:40 +0800 Subject: ASoC: rt5645: fix the shift bit of IN1 boost The shift bit of IN1 boost gain control is 12. Signed-off-by: Bard Liao Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/rt5645.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 28132375e427..c916c3881259 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -600,7 +600,7 @@ static const struct snd_kcontrol_new rt5645_snd_controls[] = { /* IN1/IN2 Control */ SOC_SINGLE_TLV("IN1 Boost", RT5645_IN1_CTRL1, - RT5645_BST_SFT1, 8, 0, bst_tlv), + RT5645_BST_SFT1, 12, 0, bst_tlv), SOC_SINGLE_TLV("IN2 Boost", RT5645_IN2_CTRL, RT5645_BST_SFT2, 8, 0, bst_tlv), -- cgit v1.2.1 From 2256b8d2ff6c8e994161ab15b6e6d0314d3174ae Mon Sep 17 00:00:00 2001 From: Arnd Bergmann Date: Wed, 20 Jan 2016 12:46:24 +0100 Subject: ASoC: rt5659: avoid unused variable warning for rt5659_acpi_match The newly added rt5659 codec driver unconditionally defines an ACPI device match table but then uses ACPI_PTR() to remove the only reference to it, so we get a harmless build warning: sound/soc/codecs/rt5659.c:4200:30: warning: 'rt5659_acpi_match' defined but not used [-Wunused-variable] static struct acpi_device_id rt5659_acpi_match[] = { This changes both the OF match table and the ACPI match table to follow the same style, using ACPI_PTR/of_match_ptr to make the reference conditional, and using an #ifdef to hide the table. This also adds the missing MODULE_DEVICE_TABLE for the OF case and adapts the formatting to the same style. Signed-off-by: Arnd Bergmann Signed-off-by: Mark Brown --- sound/soc/codecs/rt5659.c | 15 ++++++++++----- 1 file changed, 10 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt5659.c b/sound/soc/codecs/rt5659.c index c166d9394c69..fb8ea05c0de1 100644 --- a/sound/soc/codecs/rt5659.c +++ b/sound/soc/codecs/rt5659.c @@ -4183,24 +4183,29 @@ void rt5659_i2c_shutdown(struct i2c_client *client) regmap_write(rt5659->regmap, RT5659_RESET, 0); } +#ifdef CONFIG_OF static const struct of_device_id rt5659_of_match[] = { { .compatible = "realtek,rt5658", }, { .compatible = "realtek,rt5659", }, - {}, + { }, }; +MODULE_DEVICE_TABLE(of, rt5659_of_match); +#endif +#ifdef CONFIG_ACPI static struct acpi_device_id rt5659_acpi_match[] = { - { "10EC5658", 0}, - { "10EC5659", 0}, - { }, + { "10EC5658", 0, }, + { "10EC5659", 0, }, + { }, }; MODULE_DEVICE_TABLE(acpi, rt5659_acpi_match); +#endif struct i2c_driver rt5659_i2c_driver = { .driver = { .name = "rt5659", .owner = THIS_MODULE, - .of_match_table = rt5659_of_match, + .of_match_table = of_match_ptr(rt5659_of_match), .acpi_match_table = ACPI_PTR(rt5659_acpi_match), }, .probe = rt5659_i2c_probe, -- cgit v1.2.1 From c14a82c781f8df50c4c5215ab92affdc60d72c01 Mon Sep 17 00:00:00 2001 From: Sudip Mukherjee Date: Thu, 21 Jan 2016 17:27:59 +0530 Subject: ASoC: Intel: Skylake: Fix memory leak If snd_soc_tplg_component_load() fails we just printed an error message and returned the error code but we missed releasing the firmware. Signed-off-by: Sudip Mukherjee Signed-off-by: Mark Brown --- sound/soc/intel/skylake/skl-topology.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index 5315b7422b98..c7816d52ad08 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -1511,6 +1511,7 @@ int skl_tplg_init(struct snd_soc_platform *platform, struct hdac_ext_bus *ebus) release_firmware(fw); if (ret < 0) { dev_err(bus->dev, "tplg component load failed%d\n", ret); + release_firmware(fw); return -EINVAL; } -- cgit v1.2.1 From f5ede8dcc3ec1fe5344f0d30717931a44e630631 Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Thu, 21 Jan 2016 14:41:12 +0000 Subject: ASoC: wm5110: Unregister compressed platform when driver is removed The driver was not unregistering the compressed platform in wm5110_remove(). If the codec is built as a module, this would lead to a NULL pointer deref if the module was unloaded and then re-probed. Signed-off-by: Richard Fitzgerald Signed-off-by: Mark Brown --- sound/soc/codecs/wm5110.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index c36409601835..cd1b3080a497 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -2358,6 +2358,7 @@ error: static int wm5110_remove(struct platform_device *pdev) { + snd_soc_unregister_platform(&pdev->dev); snd_soc_unregister_codec(&pdev->dev); pm_runtime_disable(&pdev->dev); -- cgit v1.2.1 From 95826a37991de87659e21b3649f265a049724aa2 Mon Sep 17 00:00:00 2001 From: Stuart Henderson Date: Tue, 19 Jan 2016 13:09:08 +0000 Subject: ASoC: wm8960: Fix input boost mixer left/right naming INBMIX1 controls LINPUTs and INBMIX2 controls RINPUTs, so fix the naming accordingly. Signed-off-by: Stuart Henderson Reviewed-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm8960.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 5380798883b5..66057f853fae 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -231,13 +231,13 @@ SOC_DOUBLE_R("Capture Volume ZC Switch", WM8960_LINVOL, WM8960_RINVOL, SOC_DOUBLE_R("Capture Switch", WM8960_LINVOL, WM8960_RINVOL, 7, 1, 1), -SOC_SINGLE_TLV("Right Input Boost Mixer RINPUT3 Volume", +SOC_SINGLE_TLV("Left Input Boost Mixer LINPUT3 Volume", WM8960_INBMIX1, 4, 7, 0, lineinboost_tlv), -SOC_SINGLE_TLV("Right Input Boost Mixer RINPUT2 Volume", +SOC_SINGLE_TLV("Left Input Boost Mixer LINPUT2 Volume", WM8960_INBMIX1, 1, 7, 0, lineinboost_tlv), -SOC_SINGLE_TLV("Left Input Boost Mixer LINPUT3 Volume", +SOC_SINGLE_TLV("Right Input Boost Mixer RINPUT3 Volume", WM8960_INBMIX2, 4, 7, 0, lineinboost_tlv), -SOC_SINGLE_TLV("Left Input Boost Mixer LINPUT2 Volume", +SOC_SINGLE_TLV("Right Input Boost Mixer RINPUT2 Volume", WM8960_INBMIX2, 1, 7, 0, lineinboost_tlv), SOC_SINGLE_TLV("Right Input Boost Mixer RINPUT1 Volume", WM8960_RINPATH, 4, 3, 0, micboost_tlv), -- cgit v1.2.1 From 6bb7451429084cefcb3a18fff809f7992595d2af Mon Sep 17 00:00:00 2001 From: Stuart Henderson Date: Tue, 19 Jan 2016 13:09:09 +0000 Subject: ASoC: wm8960: Fix WM8960_SYSCLK_PLL mode With the introduction of WM8960_SYSCLK_AUTO mode, WM8960_SYSCLK_PLL mode was made unusable. Ensure we're not PLL mode before trying to use MCLK. Fixes: 3176bf2d7ccd ("ASoC: wm8960: update pll and clock setting function") Signed-off-by: Stuart Henderson Reviewed-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm8960.c | 32 +++++++++++++++++--------------- 1 file changed, 17 insertions(+), 15 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 66057f853fae..4b4401329591 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -631,29 +631,31 @@ static int wm8960_configure_clocking(struct snd_soc_codec *codec) return -EINVAL; } - /* check if the sysclk frequency is available. */ - for (i = 0; i < ARRAY_SIZE(sysclk_divs); ++i) { - if (sysclk_divs[i] == -1) - continue; - sysclk = freq_out / sysclk_divs[i]; - for (j = 0; j < ARRAY_SIZE(dac_divs); ++j) { - if (sysclk == dac_divs[j] * lrclk) { + if (wm8960->clk_id != WM8960_SYSCLK_PLL) { + /* check if the sysclk frequency is available. */ + for (i = 0; i < ARRAY_SIZE(sysclk_divs); ++i) { + if (sysclk_divs[i] == -1) + continue; + sysclk = freq_out / sysclk_divs[i]; + for (j = 0; j < ARRAY_SIZE(dac_divs); ++j) { + if (sysclk != dac_divs[j] * lrclk) + continue; for (k = 0; k < ARRAY_SIZE(bclk_divs); ++k) if (sysclk == bclk * bclk_divs[k] / 10) break; if (k != ARRAY_SIZE(bclk_divs)) break; } + if (j != ARRAY_SIZE(dac_divs)) + break; } - if (j != ARRAY_SIZE(dac_divs)) - break; - } - if (i != ARRAY_SIZE(sysclk_divs)) { - goto configure_clock; - } else if (wm8960->clk_id != WM8960_SYSCLK_AUTO) { - dev_err(codec->dev, "failed to configure clock\n"); - return -EINVAL; + if (i != ARRAY_SIZE(sysclk_divs)) { + goto configure_clock; + } else if (wm8960->clk_id != WM8960_SYSCLK_AUTO) { + dev_err(codec->dev, "failed to configure clock\n"); + return -EINVAL; + } } /* get a available pll out frequency and set pll */ for (i = 0; i < ARRAY_SIZE(sysclk_divs); ++i) { -- cgit v1.2.1 From 5c408fee254633a5be69505bc86c6b034f871ab4 Mon Sep 17 00:00:00 2001 From: "Maciej S. Szmigiero" Date: Mon, 18 Jan 2016 20:07:44 +0100 Subject: ASoC: fsl_ssi: remove explicit register defaults There is no guarantee that on fsl_ssi module load SSI registers will have their power-on-reset values. In fact, if the driver is reloaded the values in registers will be whatever they were set to previously. However, the cache needs to be fully populated at probe time to avoid non-atomic allocations during register access. Special case here is imx21-class SSI, since according to datasheet it don't have SACC{ST,EN,DIS} regs. This fixes hard lockup on fsl_ssi module reload, at least in AC'97 mode. Fixes: 05cf237972fe ("ASoC: fsl_ssi: Add driver suspend and resume to support MEGA Fast") Signed-off-by: Maciej S. Szmigiero Tested-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_ssi.c | 42 ++++++++++++++++++++++-------------------- 1 file changed, 22 insertions(+), 20 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 40dfd8a36484..ed8de1035cda 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -112,20 +112,6 @@ struct fsl_ssi_rxtx_reg_val { struct fsl_ssi_reg_val tx; }; -static const struct reg_default fsl_ssi_reg_defaults[] = { - {CCSR_SSI_SCR, 0x00000000}, - {CCSR_SSI_SIER, 0x00003003}, - {CCSR_SSI_STCR, 0x00000200}, - {CCSR_SSI_SRCR, 0x00000200}, - {CCSR_SSI_STCCR, 0x00040000}, - {CCSR_SSI_SRCCR, 0x00040000}, - {CCSR_SSI_SACNT, 0x00000000}, - {CCSR_SSI_STMSK, 0x00000000}, - {CCSR_SSI_SRMSK, 0x00000000}, - {CCSR_SSI_SACCEN, 0x00000000}, - {CCSR_SSI_SACCDIS, 0x00000000}, -}; - static bool fsl_ssi_readable_reg(struct device *dev, unsigned int reg) { switch (reg) { @@ -190,8 +176,7 @@ static const struct regmap_config fsl_ssi_regconfig = { .val_bits = 32, .reg_stride = 4, .val_format_endian = REGMAP_ENDIAN_NATIVE, - .reg_defaults = fsl_ssi_reg_defaults, - .num_reg_defaults = ARRAY_SIZE(fsl_ssi_reg_defaults), + .num_reg_defaults_raw = CCSR_SSI_SACCDIS / sizeof(uint32_t) + 1, .readable_reg = fsl_ssi_readable_reg, .volatile_reg = fsl_ssi_volatile_reg, .precious_reg = fsl_ssi_precious_reg, @@ -201,6 +186,7 @@ static const struct regmap_config fsl_ssi_regconfig = { struct fsl_ssi_soc_data { bool imx; + bool imx21regs; /* imx21-class SSI - no SACC{ST,EN,DIS} regs */ bool offline_config; u32 sisr_write_mask; }; @@ -303,6 +289,7 @@ static struct fsl_ssi_soc_data fsl_ssi_mpc8610 = { static struct fsl_ssi_soc_data fsl_ssi_imx21 = { .imx = true, + .imx21regs = true, .offline_config = true, .sisr_write_mask = 0, }; @@ -586,8 +573,12 @@ static void fsl_ssi_setup_ac97(struct fsl_ssi_private *ssi_private) */ regmap_write(regs, CCSR_SSI_SACNT, CCSR_SSI_SACNT_AC97EN | CCSR_SSI_SACNT_FV); - regmap_write(regs, CCSR_SSI_SACCDIS, 0xff); - regmap_write(regs, CCSR_SSI_SACCEN, 0x300); + + /* no SACC{ST,EN,DIS} regs on imx21-class SSI */ + if (!ssi_private->soc->imx21regs) { + regmap_write(regs, CCSR_SSI_SACCDIS, 0xff); + regmap_write(regs, CCSR_SSI_SACCEN, 0x300); + } /* * Enable SSI, Transmit and Receive. AC97 has to communicate with the @@ -1397,6 +1388,7 @@ static int fsl_ssi_probe(struct platform_device *pdev) struct resource *res; void __iomem *iomem; char name[64]; + struct regmap_config regconfig = fsl_ssi_regconfig; of_id = of_match_device(fsl_ssi_ids, &pdev->dev); if (!of_id || !of_id->data) @@ -1444,15 +1436,25 @@ static int fsl_ssi_probe(struct platform_device *pdev) return PTR_ERR(iomem); ssi_private->ssi_phys = res->start; + if (ssi_private->soc->imx21regs) { + /* + * According to datasheet imx21-class SSI + * don't have SACC{ST,EN,DIS} regs. + */ + regconfig.max_register = CCSR_SSI_SRMSK; + regconfig.num_reg_defaults_raw = + CCSR_SSI_SRMSK / sizeof(uint32_t) + 1; + } + ret = of_property_match_string(np, "clock-names", "ipg"); if (ret < 0) { ssi_private->has_ipg_clk_name = false; ssi_private->regs = devm_regmap_init_mmio(&pdev->dev, iomem, - &fsl_ssi_regconfig); + ®config); } else { ssi_private->has_ipg_clk_name = true; ssi_private->regs = devm_regmap_init_mmio_clk(&pdev->dev, - "ipg", iomem, &fsl_ssi_regconfig); + "ipg", iomem, ®config); } if (IS_ERR(ssi_private->regs)) { dev_err(&pdev->dev, "Failed to init register map\n"); -- cgit v1.2.1 From 9954859185c6e8359e71121037b627f1e294057d Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 26 Jan 2016 13:54:15 +0100 Subject: ASoC: imx-spdif: Fix crash on suspend When registering a ASoC card the driver data of the parent device is set to point to the card. This driver data is used in the snd_soc_suspend()/resume() callbacks. The imx-spdif driver overwrites the driver data with custom data which causes snd_soc_suspend() to crash. Since the custom driver is not used anywhere simply deleting the line which sets the custom driver data fixes the issue. Fixes: 43ac946922b3 ("ASoC: imx-spdif: add snd_soc_pm_ops for spdif machine driver") Tested-by: Fabio Estevam Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/fsl/imx-spdif.c | 2 -- 1 file changed, 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/imx-spdif.c b/sound/soc/fsl/imx-spdif.c index a407e833c612..fb896b2c9ba3 100644 --- a/sound/soc/fsl/imx-spdif.c +++ b/sound/soc/fsl/imx-spdif.c @@ -72,8 +72,6 @@ static int imx_spdif_audio_probe(struct platform_device *pdev) goto end; } - platform_set_drvdata(pdev, data); - end: of_node_put(spdif_np); -- cgit v1.2.1 From f212c6d8c2b21c1e1d0158d38a7c37f4427f3848 Mon Sep 17 00:00:00 2001 From: Mans Rullgard Date: Thu, 21 Jan 2016 14:55:56 +0000 Subject: ASoC: mxs-saif: fix clk_prepare() without matching clk_unprepare() The clk_prepare() call in hw_params() has no matching clk_unprepare(), leaving the clk with an ever-increasing prepare count. Moreover, hw_params() can be called multiple times which would again leave us with a runaway prepare count. Fix this by moving the clk_prepare() call to the startup() function and adding a shutdown() function with a matching clk_unprepare() as these operations are already correctly bracketed by soc-core. Signed-off-by: Mans Rullgard Signed-off-by: Mark Brown --- sound/soc/mxs/mxs-saif.c | 13 +++++++++++-- 1 file changed, 11 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index c866ade28ad0..a6c7b8d87cd2 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -381,9 +381,19 @@ static int mxs_saif_startup(struct snd_pcm_substream *substream, __raw_writel(BM_SAIF_CTRL_CLKGATE, saif->base + SAIF_CTRL + MXS_CLR_ADDR); + clk_prepare(saif->clk); + return 0; } +static void mxs_saif_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) +{ + struct mxs_saif *saif = snd_soc_dai_get_drvdata(cpu_dai); + + clk_unprepare(saif->clk); +} + /* * Should only be called when port is inactive. * although can be called multiple times by upper layers. @@ -424,8 +434,6 @@ 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) { /* * Set an initial clock rate for the saif internal logic to work @@ -611,6 +619,7 @@ static int mxs_saif_trigger(struct snd_pcm_substream *substream, int cmd, static const struct snd_soc_dai_ops mxs_saif_dai_ops = { .startup = mxs_saif_startup, + .shutdown = mxs_saif_shutdown, .trigger = mxs_saif_trigger, .prepare = mxs_saif_prepare, .hw_params = mxs_saif_hw_params, -- cgit v1.2.1 From ee43a1a0cd2a8f33cddfa1323a60b5cfcf865ba0 Mon Sep 17 00:00:00 2001 From: Aaro Koskinen Date: Sun, 24 Jan 2016 00:36:40 +0200 Subject: ASoC: simple-card: don't fail if sysclk setting is not supported Commit e22579713ae1 ("ASoC: simple card: set cpu-dai sysclk with mclk-fs") added sysclk / SND_SOC_CLOCK_OUT setting, that makes asoc_simple_card_hw_params fail if the operation is not supported, although the intention clearly was to ignore ENOTSUPP. Fix it. The patch fixes audio playback on Kirkwood / OpenRD client, where the following errors are seen: asoc-simple-card sound: ASoC: machine hw_params failed: -524 alsa-lib: /alsa-lib-1.0.28/src/pcm/pcm_hw.c:327:(snd_pcm_hw_hw_params) SNDRV_PCM_IOCTL_HW_PARAMS failed (-524): Unknown error 524 Fixes: e22579713ae1 ("ASoC: simple card: set cpu-dai sysclk with mclk-fs") Signed-off-by: Aaro Koskinen Reviewed-by: Andrew Lunn Signed-off-by: Mark Brown --- sound/soc/generic/simple-card.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 1ded8811598e..2389ab47e25f 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -99,7 +99,7 @@ static int asoc_simple_card_hw_params(struct snd_pcm_substream *substream, if (ret && ret != -ENOTSUPP) goto err; } - + return 0; err: return ret; } -- cgit v1.2.1 From d2f916aaccaf7b3bc27df2fd6cfc00f6cda2f78d Mon Sep 17 00:00:00 2001 From: "Jon Medhurst (Tixy)" Date: Mon, 1 Feb 2016 15:54:37 +0000 Subject: ASoC: dwc: Ensure i2s_reg_comp{1,2} is always initialised In the case that the driver is configured from device-tree i2s_reg_comp1 and i2s_reg_comp2 aren't initialised, breaking the driver. Fix this by unconditionally setting these values before checking for quirks. Fixes: a242cac1d3aa ("ASoC: dwc: add quirk to override COMP_PARAM_1 register") Signed-off-by: Jon Medhurst Signed-off-by: Mark Brown --- sound/soc/dwc/designware_i2s.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c index ce664c239be3..bff258d7bcea 100644 --- a/sound/soc/dwc/designware_i2s.c +++ b/sound/soc/dwc/designware_i2s.c @@ -645,6 +645,8 @@ static int dw_i2s_probe(struct platform_device *pdev) dev->dev = &pdev->dev; + dev->i2s_reg_comp1 = I2S_COMP_PARAM_1; + dev->i2s_reg_comp2 = I2S_COMP_PARAM_2; if (pdata) { dev->capability = pdata->cap; clk_id = NULL; @@ -652,9 +654,6 @@ static int dw_i2s_probe(struct platform_device *pdev) if (dev->quirks & DW_I2S_QUIRK_COMP_REG_OFFSET) { dev->i2s_reg_comp1 = pdata->i2s_reg_comp1; dev->i2s_reg_comp2 = pdata->i2s_reg_comp2; - } else { - dev->i2s_reg_comp1 = I2S_COMP_PARAM_1; - dev->i2s_reg_comp2 = I2S_COMP_PARAM_2; } ret = dw_configure_dai_by_pd(dev, dw_i2s_dai, res, pdata); } else { -- cgit v1.2.1 From 5e82d2be6ee53275c72e964507518d7964c82753 Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Mon, 1 Feb 2016 22:26:40 +0530 Subject: ASoC: dpcm: fix the BE state on hw_free While performing hw_free, DPCM checks the BE state but leaves out the suspend state. The suspend state needs to be checked as well, as we might be suspended and then usermode closes rather than resuming the audio stream. This was found by a stress testing of system with playback in loop and killed after few seconds running in background and second script running suspend-resume test in loop Signed-off-by: Vinod Koul Acked-by: Liam Girdwood Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/soc-pcm.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index e898b427be7e..1af4f23697a7 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1810,7 +1810,8 @@ int dpcm_be_dai_hw_free(struct snd_soc_pcm_runtime *fe, int stream) (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PREPARE) && (be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_FREE) && (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED) && - (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP)) + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP) && + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_SUSPEND)) continue; dev_dbg(be->dev, "ASoC: hw_free BE %s\n", -- cgit v1.2.1 From 292d4200a90715ac29f3763df27adb38a243868c Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Tue, 2 Feb 2016 12:49:49 -0600 Subject: ASoC: Intel: Atom: fix regression on compress DAI MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Commit a106804 ("ASoC: compress: Fix compress device direction check") added a dependency on the compress-cpu-dai channel_min field which was removed earlier by commit 77095796 ("ASoC: Intel: Atom: clean-up compressed DAI definition") as part of the baytrail cleanups. The net result was a regression at probe on all Atom platforms with no sound card created. Fix by adding explicit initialization for channel_min to 1 for the compress-cpu-dai. Reported-by: Tobias Mädel Signed-off-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/intel/atom/sst-mfld-platform-pcm.c b/sound/soc/intel/atom/sst-mfld-platform-pcm.c index 55c33dc76ce4..52ed434cbca6 100644 --- a/sound/soc/intel/atom/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/atom/sst-mfld-platform-pcm.c @@ -528,6 +528,7 @@ static struct snd_soc_dai_driver sst_platform_dai[] = { .ops = &sst_compr_dai_ops, .playback = { .stream_name = "Compress Playback", + .channels_min = 1, }, }, /* BE CPU Dais */ -- cgit v1.2.1 From 41d80025a83b9c7a94f97ef25c4cd3345bdc3c5e Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 3 Feb 2016 21:59:50 +0100 Subject: ASoC: dapm: Don't prefix autodisable widgets twice When a DAPM context has a prefix the autodisable widgets get prefixed twice, once for the control and once for the widget. To avoid this use the un-prefixed control name to construct the autodisable widget name. This change is purely cosmetic. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 5a2812fa8946..0d3707987900 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -310,7 +310,7 @@ struct dapm_kcontrol_data { }; static int dapm_kcontrol_data_alloc(struct snd_soc_dapm_widget *widget, - struct snd_kcontrol *kcontrol) + struct snd_kcontrol *kcontrol, const char *ctrl_name) { struct dapm_kcontrol_data *data; struct soc_mixer_control *mc; @@ -333,7 +333,7 @@ static int dapm_kcontrol_data_alloc(struct snd_soc_dapm_widget *widget, if (mc->autodisable) { struct snd_soc_dapm_widget template; - name = kasprintf(GFP_KERNEL, "%s %s", kcontrol->id.name, + name = kasprintf(GFP_KERNEL, "%s %s", ctrl_name, "Autodisable"); if (!name) { ret = -ENOMEM; @@ -371,7 +371,7 @@ static int dapm_kcontrol_data_alloc(struct snd_soc_dapm_widget *widget, if (e->autodisable) { struct snd_soc_dapm_widget template; - name = kasprintf(GFP_KERNEL, "%s %s", kcontrol->id.name, + name = kasprintf(GFP_KERNEL, "%s %s", ctrl_name, "Autodisable"); if (!name) { ret = -ENOMEM; @@ -871,7 +871,7 @@ static int dapm_create_or_share_kcontrol(struct snd_soc_dapm_widget *w, kcontrol->private_free = dapm_kcontrol_free; - ret = dapm_kcontrol_data_alloc(w, kcontrol); + ret = dapm_kcontrol_data_alloc(w, kcontrol, name); if (ret) { snd_ctl_free_one(kcontrol); goto exit_free; -- cgit v1.2.1 From 41556f68d1dd0b6bbf311a220523b034d2a040e7 Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Wed, 3 Feb 2016 17:59:44 +0530 Subject: ASoC: Intel: Skylake: Fix the memory overwrite of tlv buffer TLV buffer can be smaller than the module data, so update the size of data to be copied before doing the copy. Also TLV header consists of two unsigned ints, this is also taken into account here and size modified to reflect this Suggested-by: Takashi Iwai Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/skylake/skl-topology.c | 7 +++++++ 1 file changed, 7 insertions(+) (limited to 'sound') diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index c7816d52ad08..c67e3acb8102 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -916,6 +916,13 @@ static int skl_tplg_tlv_control_get(struct snd_kcontrol *kcontrol, skl_get_module_params(skl->skl_sst, (u32 *)bc->params, bc->max, bc->param_id, mconfig); + /* decrement size for TLV header */ + size -= 2 * sizeof(u32); + + /* check size as we don't want to send kernel data */ + if (size > bc->max) + size = bc->max; + if (bc->params) { if (copy_to_user(data, &bc->param_id, sizeof(u32))) return -EFAULT; -- cgit v1.2.1 From ee564d489cc47b1b6043bbe7e95464306d112cf5 Mon Sep 17 00:00:00 2001 From: Guneshwor Singh Date: Wed, 3 Feb 2016 17:59:45 +0530 Subject: ASoC: Intel: Skylake: Fix delay wrap condition When delay reported by HW is equal to buffersize, it means the value is wrapped so we should report as 0. So add the condition to check this while reporting the delay from LPIB. Signed-off-by: Guneshwor Singh Signed-off-by: Dharageswari.R Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/skylake/skl-pcm.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c index b89ae6f7c096..f9297dc4b25f 100644 --- a/sound/soc/intel/skylake/skl-pcm.c +++ b/sound/soc/intel/skylake/skl-pcm.c @@ -829,6 +829,7 @@ static int skl_get_delay_from_lpib(struct hdac_ext_bus *ebus, else delay += hstream->bufsize; } + delay = (hstream->bufsize == delay) ? 0 : delay; if (delay >= hstream->period_bytes) { dev_info(bus->dev, -- cgit v1.2.1 From 7ca42f5ac5e0d8011086bcfa00e85aede42f0b78 Mon Sep 17 00:00:00 2001 From: Guneshwor Singh Date: Wed, 3 Feb 2016 17:59:46 +0530 Subject: ASoC: Intel: Skylake: Fix mcps freeup after module unbind failure While cleaning resources on module pmd event, we check for return of skl_unbind_modules(). On failure this causes leak as all modules attached do not have resources freed. So ignore return value of module unbind and continue freeing resources. This makes dapm state and resources correct. Signed-off-by: Guneshwor Singh Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/skylake/skl-topology.c | 7 ++----- 1 file changed, 2 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index c67e3acb8102..86d5323e9184 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -98,7 +98,7 @@ static bool skl_tplg_alloc_pipe_mcps(struct skl *skl, "%s: module_id %d instance %d\n", __func__, mconfig->id.module_id, mconfig->id.instance_id); dev_err(ctx->dev, - "exceeds ppl memory available %d > mem %d\n", + "exceeds ppl mcps available %d > mem %d\n", skl->resource.max_mcps, skl->resource.mcps); return false; } @@ -773,10 +773,7 @@ static int skl_tplg_mixer_dapm_post_pmd_event(struct snd_soc_dapm_widget *w, continue; } - ret = skl_unbind_modules(ctx, src_module, dst_module); - if (ret < 0) - return ret; - + skl_unbind_modules(ctx, src_module, dst_module); src_module = dst_module; } -- cgit v1.2.1 From 9ba8ffef9635c11102bc42d0f2d0a4213de273d5 Mon Sep 17 00:00:00 2001 From: "Dharageswari.R" Date: Wed, 3 Feb 2016 17:59:47 +0530 Subject: ASoC: Intel: Skylake: Fix pipe memory allocation leak We check and allocate pipeline resources in one shot. That causes leaks if module creation fails later as that is not freed. So split the resource allocation into two, first check if resources are available and then add the resources upon successful creation. So two new functions are added for checking and current functions are re-purposed to only add the resources for memory and MCPS. Signed-off-by: Dharageswari.R Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/skylake/skl-topology.c | 42 +++++++++++++++++++++++----------- 1 file changed, 29 insertions(+), 13 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index 86d5323e9184..efe001162204 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -54,12 +54,9 @@ static int is_skl_dsp_widget_type(struct snd_soc_dapm_widget *w) /* * Each pipelines needs memory to be allocated. Check if we have free memory - * from available pool. Then only add this to pool - * This is freed when pipe is deleted - * Note: DSP does actual memory management we only keep track for complete - * pool + * from available pool. */ -static bool skl_tplg_alloc_pipe_mem(struct skl *skl, +static bool skl_is_pipe_mem_avail(struct skl *skl, struct skl_module_cfg *mconfig) { struct skl_sst *ctx = skl->skl_sst; @@ -74,10 +71,20 @@ static bool skl_tplg_alloc_pipe_mem(struct skl *skl, "exceeds ppl memory available %d mem %d\n", skl->resource.max_mem, skl->resource.mem); return false; + } else { + return true; } +} +/* + * Add the mem to the mem pool. This is freed when pipe is deleted. + * Note: DSP does actual memory management we only keep track for complete + * pool + */ +static void skl_tplg_alloc_pipe_mem(struct skl *skl, + struct skl_module_cfg *mconfig) +{ skl->resource.mem += mconfig->pipe->memory_pages; - return true; } /* @@ -85,10 +92,10 @@ static bool skl_tplg_alloc_pipe_mem(struct skl *skl, * quantified in MCPS (Million Clocks Per Second) required for module/pipe * * Each pipelines needs mcps to be allocated. Check if we have mcps for this - * pipe. This adds the mcps to driver counter - * This is removed on pipeline delete + * pipe. */ -static bool skl_tplg_alloc_pipe_mcps(struct skl *skl, + +static bool skl_is_pipe_mcps_avail(struct skl *skl, struct skl_module_cfg *mconfig) { struct skl_sst *ctx = skl->skl_sst; @@ -101,10 +108,15 @@ static bool skl_tplg_alloc_pipe_mcps(struct skl *skl, "exceeds ppl mcps available %d > mem %d\n", skl->resource.max_mcps, skl->resource.mcps); return false; + } else { + return true; } +} +static void skl_tplg_alloc_pipe_mcps(struct skl *skl, + struct skl_module_cfg *mconfig) +{ skl->resource.mcps += mconfig->mcps; - return true; } /* @@ -411,7 +423,7 @@ skl_tplg_init_pipe_modules(struct skl *skl, struct skl_pipe *pipe) mconfig = w->priv; /* check resource available */ - if (!skl_tplg_alloc_pipe_mcps(skl, mconfig)) + if (!skl_is_pipe_mcps_avail(skl, mconfig)) return -ENOMEM; if (mconfig->is_loadable && ctx->dsp->fw_ops.load_mod) { @@ -435,6 +447,7 @@ skl_tplg_init_pipe_modules(struct skl *skl, struct skl_pipe *pipe) ret = skl_tplg_set_module_params(w, ctx); if (ret < 0) return ret; + skl_tplg_alloc_pipe_mcps(skl, mconfig); } return 0; @@ -477,10 +490,10 @@ static int skl_tplg_mixer_dapm_pre_pmu_event(struct snd_soc_dapm_widget *w, struct skl_sst *ctx = skl->skl_sst; /* check resource available */ - if (!skl_tplg_alloc_pipe_mcps(skl, mconfig)) + if (!skl_is_pipe_mcps_avail(skl, mconfig)) return -EBUSY; - if (!skl_tplg_alloc_pipe_mem(skl, mconfig)) + if (!skl_is_pipe_mem_avail(skl, mconfig)) return -ENOMEM; /* @@ -526,6 +539,9 @@ static int skl_tplg_mixer_dapm_pre_pmu_event(struct snd_soc_dapm_widget *w, src_module = dst_module; } + skl_tplg_alloc_pipe_mem(skl, mconfig); + skl_tplg_alloc_pipe_mcps(skl, mconfig); + return 0; } -- cgit v1.2.1 From 9cf3049e21e4e6873aae45df19c11f7243e2f03f Mon Sep 17 00:00:00 2001 From: Jeeja KP Date: Wed, 3 Feb 2016 17:59:48 +0530 Subject: ASoC: Intel: Skylake: Fix return of skl_get_queue_index In unbind modules, the skl_get_queue_index() can return error if the pin is dynamic and module is not bound yet. So instead of returning error this check should return success as modules is not yet bound. This will let the module be bound when connected pipes are enabled and will bind this as well. So change the return value to 0 Signed-off-by: Jeeja KP Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/skylake/skl-messages.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/skylake/skl-messages.c b/sound/soc/intel/skylake/skl-messages.c index de6dac496a0d..bb5f1d7d0cad 100644 --- a/sound/soc/intel/skylake/skl-messages.c +++ b/sound/soc/intel/skylake/skl-messages.c @@ -688,14 +688,14 @@ int skl_unbind_modules(struct skl_sst *ctx, /* get src queue index */ src_index = skl_get_queue_index(src_mcfg->m_out_pin, dst_id, out_max); if (src_index < 0) - return -EINVAL; + return 0; msg.src_queue = src_index; /* get dst queue index */ dst_index = skl_get_queue_index(dst_mcfg->m_in_pin, src_id, in_max); if (dst_index < 0) - return -EINVAL; + return 0; msg.dst_queue = dst_index; -- cgit v1.2.1 From 0c684c48257bc6033bdd3b942babef22d0a1852a Mon Sep 17 00:00:00 2001 From: Jeeja KP Date: Wed, 3 Feb 2016 17:59:49 +0530 Subject: ASoC: Intel: Skylake: Fix the module state check condition For binding modules we should check if source or destination module is in UNINT state. We canot bind even if one of them is in this state. So update the check from logical AND to logical OR and do not bind modules for this case Signed-off-by: Jeeja KP Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/skylake/skl-messages.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/intel/skylake/skl-messages.c b/sound/soc/intel/skylake/skl-messages.c index bb5f1d7d0cad..4629372d7c8e 100644 --- a/sound/soc/intel/skylake/skl-messages.c +++ b/sound/soc/intel/skylake/skl-messages.c @@ -747,7 +747,7 @@ int skl_bind_modules(struct skl_sst *ctx, skl_dump_bind_info(ctx, src_mcfg, dst_mcfg); - if (src_mcfg->m_state < SKL_MODULE_INIT_DONE && + if (src_mcfg->m_state < SKL_MODULE_INIT_DONE || dst_mcfg->m_state < SKL_MODULE_INIT_DONE) return 0; -- cgit v1.2.1 From 9946f70906eebf2a305d0b189de52eec8ba39649 Mon Sep 17 00:00:00 2001 From: Jeeja KP Date: Wed, 3 Feb 2016 17:59:50 +0530 Subject: ASoC: Intel: Skylake: Fix not to stop sink pipe in pga pmd event We should not stop the sink pipe in it's pmd handler for a mixin module as this module may still be connected to other pipes. This will be stopped and freed by current implementation on last connected pipe unbind. Signed-off-by: Jeeja KP Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/skylake/skl-topology.c | 3 --- 1 file changed, 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index efe001162204..a356f3b1dd5b 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -827,9 +827,6 @@ static int skl_tplg_pga_dapm_post_pmd_event(struct snd_soc_dapm_widget *w, * This is a connecter and if path is found that means * unbind between source and sink has not happened yet */ - ret = skl_stop_pipe(ctx, sink_mconfig->pipe); - if (ret < 0) - return ret; ret = skl_unbind_modules(ctx, src_mconfig, sink_mconfig); } -- cgit v1.2.1 From 6bd4cf855698312133b7776c77ee78af865608eb Mon Sep 17 00:00:00 2001 From: Jeeja KP Date: Wed, 3 Feb 2016 17:59:51 +0530 Subject: ASoC: Intel: Skylake: Fix bind of source with multiple sinks skl_tplg_bind_sinks() takes only the first sink widget. This breaks in case we have multiple sinks for a module. So pass source widget to skl_tplg_bind_sinks() and bind for all sinks by calling this recursively Signed-off-by: Jeeja KP Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/skylake/skl-topology.c | 9 +++++++-- 1 file changed, 7 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index a356f3b1dd5b..77a688d00fc6 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -547,6 +547,7 @@ static int skl_tplg_mixer_dapm_pre_pmu_event(struct snd_soc_dapm_widget *w, static int skl_tplg_bind_sinks(struct snd_soc_dapm_widget *w, struct skl *skl, + struct snd_soc_dapm_widget *src_w, struct skl_module_cfg *src_mconfig) { struct snd_soc_dapm_path *p; @@ -563,6 +564,10 @@ static int skl_tplg_bind_sinks(struct snd_soc_dapm_widget *w, dev_dbg(ctx->dev, "%s: sink widget=%s\n", __func__, p->sink->name); next_sink = p->sink; + + if (!is_skl_dsp_widget_type(p->sink)) + return skl_tplg_bind_sinks(p->sink, skl, src_w, src_mconfig); + /* * here we will check widgets in sink pipelines, so that * can be any widgets type and we are only interested if @@ -592,7 +597,7 @@ static int skl_tplg_bind_sinks(struct snd_soc_dapm_widget *w, } if (!sink) - return skl_tplg_bind_sinks(next_sink, skl, src_mconfig); + return skl_tplg_bind_sinks(next_sink, skl, src_w, src_mconfig); return 0; } @@ -621,7 +626,7 @@ static int skl_tplg_pga_dapm_pre_pmu_event(struct snd_soc_dapm_widget *w, * if sink is not started, start sink pipe first, then start * this pipe */ - ret = skl_tplg_bind_sinks(w, skl, src_mconfig); + ret = skl_tplg_bind_sinks(w, skl, w, src_mconfig); if (ret) return ret; -- cgit v1.2.1 From de1fedf25b075664320010789ede2a0f9f4de07d Mon Sep 17 00:00:00 2001 From: Jeeja KP Date: Wed, 3 Feb 2016 17:59:52 +0530 Subject: ASoC: Intel: Skylake: Add missing PRE/POST_PMU handlers for vmixer Some modules may be directly connected to a pipeline without a mixer module. For these modules, we require PRE_PMU and POST_PMU handler which will do bind between the pipelines, so add these missing handlers. Signed-off-by: Jeeja KP Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/skylake/skl-topology.c | 6 ++++++ 1 file changed, 6 insertions(+) (limited to 'sound') diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index 77a688d00fc6..489848637df5 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -857,6 +857,12 @@ static int skl_tplg_vmixer_event(struct snd_soc_dapm_widget *w, case SND_SOC_DAPM_PRE_PMU: return skl_tplg_mixer_dapm_pre_pmu_event(w, skl); + case SND_SOC_DAPM_POST_PMU: + return skl_tplg_mixer_dapm_post_pmu_event(w, skl); + + case SND_SOC_DAPM_PRE_PMD: + return skl_tplg_mixer_dapm_pre_pmd_event(w, skl); + case SND_SOC_DAPM_POST_PMD: return skl_tplg_mixer_dapm_post_pmd_event(w, skl); } -- cgit v1.2.1 From 6e3ffa00424e198d2f0c628e7575c5adefeda3d7 Mon Sep 17 00:00:00 2001 From: Jeeja KP Date: Wed, 3 Feb 2016 17:59:53 +0530 Subject: ASoC: Intel: Skylake: Fix stereo DMIC record DMIC BE can have 2 or 4 channels supported. The DMIC fixup needs to take this into account. Signed-off-by: Jeeja KP Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/boards/skl_rt286.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/intel/boards/skl_rt286.c b/sound/soc/intel/boards/skl_rt286.c index 7396ddb427d8..2cbcbe412661 100644 --- a/sound/soc/intel/boards/skl_rt286.c +++ b/sound/soc/intel/boards/skl_rt286.c @@ -212,7 +212,10 @@ static int skylake_dmic_fixup(struct snd_soc_pcm_runtime *rtd, { struct snd_interval *channels = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); - channels->min = channels->max = 4; + if (params_channels(params) == 2) + channels->min = channels->max = 2; + else + channels->min = channels->max = 4; return 0; } -- cgit v1.2.1 From 38c079e230f25969e7ce3501fa967b003a2abc39 Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Wed, 3 Feb 2016 17:59:54 +0530 Subject: ASoC: Intel: Skylake: Remove autosuspend delay The driver used autosuspend delay to delay going to D3. But per HW recommendation we should go to D3 soon, so remove the delay from driver Signed-off-by: Subhransu S. Prusty Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/skylake/skl.c | 2 -- 1 file changed, 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index c38bf99ced10..1d36b28d6489 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -558,8 +558,6 @@ static int skl_probe(struct pci_dev *pci, goto out_unregister; /*configure PM */ - pm_runtime_set_autosuspend_delay(bus->dev, SKL_SUSPEND_DELAY); - pm_runtime_use_autosuspend(bus->dev); pm_runtime_put_noidle(bus->dev); pm_runtime_allow(bus->dev); -- cgit v1.2.1 From 360a8245680053619205a3ae10e6bfe624a5da1d Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Fri, 5 Feb 2016 09:05:41 +0100 Subject: ALSA: hda - Fix static checker warning in patch_hdmi.c The static checker warning is: sound/pci/hda/patch_hdmi.c:460 hdmi_eld_ctl_get() error: __memcpy() 'eld->eld_buffer' too small (256 vs 512) I have a hard time figuring out if this can ever cause an information leak (I don't think so), but nonetheless it does not hurt to increase the robustness of the code. Fixes: 68e03de98507 ('ALSA: hda - hdmi: Do not expose eld data when eld is invalid') Reported-by: Dan Carpenter Signed-off-by: David Henningsson Cc: # v3.9+ Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_hdmi.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 1f52b55d77c9..2191e2359315 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -448,7 +448,8 @@ static int hdmi_eld_ctl_get(struct snd_kcontrol *kcontrol, eld = &per_pin->sink_eld; mutex_lock(&per_pin->lock); - if (eld->eld_size > ARRAY_SIZE(ucontrol->value.bytes.data)) { + if (eld->eld_size > ARRAY_SIZE(ucontrol->value.bytes.data) || + eld->eld_size > ELD_MAX_SIZE) { mutex_unlock(&per_pin->lock); snd_BUG(); return -EINVAL; -- cgit v1.2.1 From 5d2560a427fc7c4050a320be62c4994705ca81b1 Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Fri, 5 Feb 2016 09:56:05 +0900 Subject: ALSA: firewire-tascam: fix NULL pointer dereference when model identification fails When unsupported models are connected, snd-firewire-tascam module causes NULL pointer dereference in fw_core_remove_address_handler() (due to list_del_rcu()). This commit prevents this bug. Signed-off-by: Takashi Sakamoto Signed-off-by: Takashi Iwai --- sound/firewire/tascam/tascam-transaction.c | 6 ++++++ 1 file changed, 6 insertions(+) (limited to 'sound') diff --git a/sound/firewire/tascam/tascam-transaction.c b/sound/firewire/tascam/tascam-transaction.c index 904ce0329fa1..040a96d1ba8e 100644 --- a/sound/firewire/tascam/tascam-transaction.c +++ b/sound/firewire/tascam/tascam-transaction.c @@ -230,6 +230,7 @@ int snd_tscm_transaction_register(struct snd_tscm *tscm) return err; error: fw_core_remove_address_handler(&tscm->async_handler); + tscm->async_handler.callback_data = NULL; return err; } @@ -276,6 +277,9 @@ void snd_tscm_transaction_unregister(struct snd_tscm *tscm) __be32 reg; unsigned int i; + if (tscm->async_handler.callback_data == NULL) + return; + /* Turn off FireWire LED. */ reg = cpu_to_be32(0x0000008e); snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST, @@ -297,6 +301,8 @@ void snd_tscm_transaction_unregister(struct snd_tscm *tscm) ®, sizeof(reg), 0); fw_core_remove_address_handler(&tscm->async_handler); + tscm->async_handler.callback_data = NULL; + for (i = 0; i < TSCM_MIDI_OUT_PORT_MAX; i++) snd_fw_async_midi_port_destroy(&tscm->out_ports[i]); } -- cgit v1.2.1 From 3e78e1518e129407fae75c867e48828262b3ea6d Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Fri, 5 Feb 2016 09:56:06 +0900 Subject: ALSA: firewire-tascam: add support for FW-1804 This model supports: * maximum 12 PCM channels for PCM playback * maximum 18 PCM channels for PCM capture * 4 ports for MIDI playback * 4 ports for MIDI capture * control and status messages in tx isochronous packets * up to 96.0 kHz This commit adds support for the model. As the other supported models, all of available PCM channels are always enabled. As I described in commit c0949b278515da94, Ilya Zimnovich had investigated TASCAM FireWire series in 2011 with his FW-1804. In his report, this model has internal multiplexer and any software implementation can control it. Following to the design of ALSA firewire stack, this commit won't implement it. It should be in userspace via Linux fw character device. Signed-off-by: Takashi Sakamoto Signed-off-by: Takashi Iwai --- sound/firewire/tascam/tascam.c | 11 ++++++++++- 1 file changed, 10 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/firewire/tascam/tascam.c b/sound/firewire/tascam/tascam.c index ee0bc1839508..dcb11c26c225 100644 --- a/sound/firewire/tascam/tascam.c +++ b/sound/firewire/tascam/tascam.c @@ -33,7 +33,16 @@ static struct snd_tscm_spec model_specs[] = { .midi_playback_ports = 2, .is_controller = true, }, - /* FW-1804 may be supported. */ + { + .name = "FW-1804", + .has_adat = true, + .has_spdif = true, + .pcm_capture_analog_channels = 8, + .pcm_playback_analog_channels = 2, + .midi_capture_ports = 2, + .midi_playback_ports = 4, + .is_controller = false, + }, }; static int identify_model(struct snd_tscm *tscm) -- cgit v1.2.1 From 61ebe499643703af517a8253662982f6f4764c92 Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Fri, 5 Feb 2016 09:56:07 +0900 Subject: ALSA: firewire-tascam: remove a flag for controller Currently, 'struct snd_tscm_spec' has a member named as 'is_controller' to identify MIDI controller. This member was originally added to skip parse control and status messages in isochronous packets for non-controller model. As long as I investigate, FW-1804 (non-controller) also transfers the control and status message, thus it becomes meaningless. This commit removes it. Signed-off-by: Takashi Sakamoto Signed-off-by: Takashi Iwai --- sound/firewire/tascam/tascam.c | 3 --- sound/firewire/tascam/tascam.h | 1 - 2 files changed, 4 deletions(-) (limited to 'sound') diff --git a/sound/firewire/tascam/tascam.c b/sound/firewire/tascam/tascam.c index dcb11c26c225..e281c338e562 100644 --- a/sound/firewire/tascam/tascam.c +++ b/sound/firewire/tascam/tascam.c @@ -21,7 +21,6 @@ static struct snd_tscm_spec model_specs[] = { .pcm_playback_analog_channels = 8, .midi_capture_ports = 4, .midi_playback_ports = 4, - .is_controller = true, }, { .name = "FW-1082", @@ -31,7 +30,6 @@ static struct snd_tscm_spec model_specs[] = { .pcm_playback_analog_channels = 2, .midi_capture_ports = 2, .midi_playback_ports = 2, - .is_controller = true, }, { .name = "FW-1804", @@ -41,7 +39,6 @@ static struct snd_tscm_spec model_specs[] = { .pcm_playback_analog_channels = 2, .midi_capture_ports = 2, .midi_playback_ports = 4, - .is_controller = false, }, }; diff --git a/sound/firewire/tascam/tascam.h b/sound/firewire/tascam/tascam.h index 2d028d2bd3bd..66268600c357 100644 --- a/sound/firewire/tascam/tascam.h +++ b/sound/firewire/tascam/tascam.h @@ -39,7 +39,6 @@ struct snd_tscm_spec { unsigned int pcm_playback_analog_channels; unsigned int midi_capture_ports; unsigned int midi_playback_ports; - bool is_controller; }; #define TSCM_MIDI_IN_PORT_MAX 4 -- cgit v1.2.1 From 56661a2ed5348f3d7a3ac8788656654dd50904cd Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Fri, 5 Feb 2016 09:56:08 +0900 Subject: ALSA: firewire-tascam: remove needless member for control and status message Commit 3beab0f844fa added a member for control and status message, while it's planned and not implemented yet. This commit removes it. Fixes: 3beab0f844fa('ALSA: firewire-tascam: add support for outgoing MIDI messages by asynchronous transaction') Signed-off-by: Takashi Sakamoto Signed-off-by: Takashi Iwai --- sound/firewire/tascam/tascam.h | 3 --- 1 file changed, 3 deletions(-) (limited to 'sound') diff --git a/sound/firewire/tascam/tascam.h b/sound/firewire/tascam/tascam.h index 66268600c357..30ab77e924f7 100644 --- a/sound/firewire/tascam/tascam.h +++ b/sound/firewire/tascam/tascam.h @@ -71,9 +71,6 @@ struct snd_tscm { struct snd_fw_async_midi_port out_ports[TSCM_MIDI_OUT_PORT_MAX]; u8 running_status[TSCM_MIDI_OUT_PORT_MAX]; bool on_sysex[TSCM_MIDI_OUT_PORT_MAX]; - - /* For control messages. */ - struct snd_firewire_tascam_status *status; }; #define TSCM_ADDR_BASE 0xffff00000000ull -- cgit v1.2.1 From 6c361d10e0eb859233c71954abcd20d2d8700587 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 5 Feb 2016 20:12:24 +0100 Subject: Revert "ALSA: hda - Fix noise on Gigabyte Z170X mobo" This reverts commit 0c25ad80408e95e0a4fbaf0056950206e95f726f. The original commit disabled the aamixer path due to the noise problem, but it turned out that some mobo with the same PCI SSID doesn't suffer from the issue, and the disabled function (analog loopback) is still demanded by users. Since the recent commit [e7fdd52779a6: ALSA: hda - Implement loopback control switch for Realtek and other codecs], we have the dynamic mixer switch to enable/disable the aamix path, and we don't have to disable the path statically any longer. So, let's revert the disablement, so that only the user suffering from the noise problem can turn off the aamix on the fly. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=108301 Reported-by: Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 8 -------- 1 file changed, 8 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 21992fb7035d..a733e5dc701d 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1787,7 +1787,6 @@ enum { ALC882_FIXUP_NO_PRIMARY_HP, ALC887_FIXUP_ASUS_BASS, ALC887_FIXUP_BASS_CHMAP, - ALC882_FIXUP_DISABLE_AAMIX, }; static void alc889_fixup_coef(struct hda_codec *codec, @@ -1949,8 +1948,6 @@ static void alc882_fixup_no_primary_hp(struct hda_codec *codec, static void alc_fixup_bass_chmap(struct hda_codec *codec, const struct hda_fixup *fix, int action); -static void alc_fixup_disable_aamix(struct hda_codec *codec, - const struct hda_fixup *fix, int action); static const struct hda_fixup alc882_fixups[] = { [ALC882_FIXUP_ABIT_AW9D_MAX] = { @@ -2188,10 +2185,6 @@ static const struct hda_fixup alc882_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_bass_chmap, }, - [ALC882_FIXUP_DISABLE_AAMIX] = { - .type = HDA_FIXUP_FUNC, - .v.func = alc_fixup_disable_aamix, - }, }; static const struct snd_pci_quirk alc882_fixup_tbl[] = { @@ -2259,7 +2252,6 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x1462, 0x7350, "MSI-7350", ALC889_FIXUP_CD), SND_PCI_QUIRK_VENDOR(0x1462, "MSI", ALC882_FIXUP_GPIO3), SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte EP45-DS3/Z87X-UD3H", ALC889_FIXUP_FRONT_HP_NO_PRESENCE), - SND_PCI_QUIRK(0x1458, 0xa182, "Gigabyte Z170X-UD3", ALC882_FIXUP_DISABLE_AAMIX), SND_PCI_QUIRK(0x147b, 0x107a, "Abit AW9D-MAX", ALC882_FIXUP_ABIT_AW9D_MAX), SND_PCI_QUIRK_VENDOR(0x1558, "Clevo laptop", ALC882_FIXUP_EAPD), SND_PCI_QUIRK(0x161f, 0x2054, "Medion laptop", ALC883_FIXUP_EAPD), -- cgit v1.2.1 From c44d9b1181cf34e0860c72cc8a00e0c47417aac0 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sun, 7 Feb 2016 09:38:26 +0100 Subject: ALSA: hda - Fix speaker output from VAIO AiO machines Some Sony VAIO AiO models (VGC-JS4EF and VGC-JS25G, both with PCI SSID 104d:9044) need the same quirk to make the speaker working properly. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=112031 Cc: 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 a733e5dc701d..b43c0f7994eb 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2223,6 +2223,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x104d, 0x9047, "Sony Vaio TT", ALC889_FIXUP_VAIO_TT), SND_PCI_QUIRK(0x104d, 0x905a, "Sony Vaio Z", ALC882_FIXUP_NO_PRIMARY_HP), SND_PCI_QUIRK(0x104d, 0x9043, "Sony Vaio VGC-LN51JGB", ALC882_FIXUP_NO_PRIMARY_HP), + SND_PCI_QUIRK(0x104d, 0x9044, "Sony VAIO AiO", ALC882_FIXUP_NO_PRIMARY_HP), /* All Apple entries are in codec SSIDs */ SND_PCI_QUIRK(0x106b, 0x00a0, "MacBookPro 3,1", ALC889_FIXUP_MBP_VREF), -- cgit v1.2.1 From ddce57a6f0a2d8d1bfacfa77f06043bc760403c2 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 2 Feb 2016 15:27:36 +0100 Subject: ALSA: dummy: Implement timer backend switching more safely Currently the selected timer backend is referred at any moment from the running PCM callbacks. When the backend is switched, it's possible to lead to inconsistency from the running backend. This was pointed by syzkaller fuzzer, and the commit [7ee96216c31a: ALSA: dummy: Disable switching timer backend via sysfs] disabled the dynamic switching for avoiding the crash. This patch improves the handling of timer backend switching. It keeps the reference to the selected backend during the whole operation of an opened stream so that it won't be changed by other streams. Together with this change, the hrtimer parameter is reenabled as writable now. NOTE: this patch also turned out to fix the still remaining race. Namely, ops was still replaced dynamically at dummy_pcm_open: static int dummy_pcm_open(struct snd_pcm_substream *substream) { .... dummy->timer_ops = &dummy_systimer_ops; if (hrtimer) dummy->timer_ops = &dummy_hrtimer_ops; Since dummy->timer_ops is common among all streams, and when the replacement happens during accesses of other streams, it may lead to a crash. This was actually triggered by syzkaller fuzzer and KASAN. This patch rewrites the code not to use the ops shared by all streams any longer, too. BugLink: http://lkml.kernel.org/r/CACT4Y+aZ+xisrpuM6cOXbL21DuM0yVxPYXf4cD4Md9uw0C3dBQ@mail.gmail.com Reported-by: Dmitry Vyukov Cc: Signed-off-by: Takashi Iwai --- sound/drivers/dummy.c | 37 +++++++++++++++++++------------------ 1 file changed, 19 insertions(+), 18 deletions(-) (limited to 'sound') diff --git a/sound/drivers/dummy.c b/sound/drivers/dummy.c index bde33308f0d6..c0f8f613f1f1 100644 --- a/sound/drivers/dummy.c +++ b/sound/drivers/dummy.c @@ -87,7 +87,7 @@ MODULE_PARM_DESC(pcm_substreams, "PCM substreams # (1-128) for dummy driver."); module_param(fake_buffer, bool, 0444); MODULE_PARM_DESC(fake_buffer, "Fake buffer allocations."); #ifdef CONFIG_HIGH_RES_TIMERS -module_param(hrtimer, bool, 0444); +module_param(hrtimer, bool, 0644); MODULE_PARM_DESC(hrtimer, "Use hrtimer as the timer source."); #endif @@ -109,6 +109,9 @@ struct dummy_timer_ops { snd_pcm_uframes_t (*pointer)(struct snd_pcm_substream *); }; +#define get_dummy_ops(substream) \ + (*(const struct dummy_timer_ops **)(substream)->runtime->private_data) + struct dummy_model { const char *name; int (*playback_constraints)(struct snd_pcm_runtime *runtime); @@ -137,7 +140,6 @@ struct snd_dummy { int iobox; struct snd_kcontrol *cd_volume_ctl; struct snd_kcontrol *cd_switch_ctl; - const struct dummy_timer_ops *timer_ops; }; /* @@ -231,6 +233,8 @@ static struct dummy_model *dummy_models[] = { */ struct dummy_systimer_pcm { + /* ops must be the first item */ + const struct dummy_timer_ops *timer_ops; spinlock_t lock; struct timer_list timer; unsigned long base_time; @@ -366,6 +370,8 @@ static const struct dummy_timer_ops dummy_systimer_ops = { */ struct dummy_hrtimer_pcm { + /* ops must be the first item */ + const struct dummy_timer_ops *timer_ops; ktime_t base_time; ktime_t period_time; atomic_t running; @@ -492,31 +498,25 @@ static const struct dummy_timer_ops dummy_hrtimer_ops = { static int dummy_pcm_trigger(struct snd_pcm_substream *substream, int cmd) { - struct snd_dummy *dummy = snd_pcm_substream_chip(substream); - switch (cmd) { case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: - return dummy->timer_ops->start(substream); + return get_dummy_ops(substream)->start(substream); case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: - return dummy->timer_ops->stop(substream); + return get_dummy_ops(substream)->stop(substream); } return -EINVAL; } static int dummy_pcm_prepare(struct snd_pcm_substream *substream) { - struct snd_dummy *dummy = snd_pcm_substream_chip(substream); - - return dummy->timer_ops->prepare(substream); + return get_dummy_ops(substream)->prepare(substream); } static snd_pcm_uframes_t dummy_pcm_pointer(struct snd_pcm_substream *substream) { - struct snd_dummy *dummy = snd_pcm_substream_chip(substream); - - return dummy->timer_ops->pointer(substream); + return get_dummy_ops(substream)->pointer(substream); } static struct snd_pcm_hardware dummy_pcm_hardware = { @@ -562,17 +562,19 @@ static int dummy_pcm_open(struct snd_pcm_substream *substream) struct snd_dummy *dummy = snd_pcm_substream_chip(substream); struct dummy_model *model = dummy->model; struct snd_pcm_runtime *runtime = substream->runtime; + const struct dummy_timer_ops *ops; int err; - dummy->timer_ops = &dummy_systimer_ops; + ops = &dummy_systimer_ops; #ifdef CONFIG_HIGH_RES_TIMERS if (hrtimer) - dummy->timer_ops = &dummy_hrtimer_ops; + ops = &dummy_hrtimer_ops; #endif - err = dummy->timer_ops->create(substream); + err = ops->create(substream); if (err < 0) return err; + get_dummy_ops(substream) = ops; runtime->hw = dummy->pcm_hw; if (substream->pcm->device & 1) { @@ -594,7 +596,7 @@ static int dummy_pcm_open(struct snd_pcm_substream *substream) err = model->capture_constraints(substream->runtime); } if (err < 0) { - dummy->timer_ops->free(substream); + get_dummy_ops(substream)->free(substream); return err; } return 0; @@ -602,8 +604,7 @@ static int dummy_pcm_open(struct snd_pcm_substream *substream) static int dummy_pcm_close(struct snd_pcm_substream *substream) { - struct snd_dummy *dummy = snd_pcm_substream_chip(substream); - dummy->timer_ops->free(substream); + get_dummy_ops(substream)->free(substream); return 0; } -- cgit v1.2.1 From 902c136fe4f72dfc2a616ad755c72f1ee407f79a Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Mon, 8 Feb 2016 10:45:36 +0530 Subject: ASoC: Intel: Revert "ASoC: Intel: fix ACPI probe regression with Atom DPCM driver" This reverts commit dc901a354171 ("ASoC: Intel: fix ACPI probe regression with Atom DPCM driver") as the fix prevented the probe on HSW/BDW if Atom-DPCM was selected Acked-by: Jie Yang Acked-by: Pierre-Louis Bossart Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/common/Makefile | 5 ----- 1 file changed, 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/common/Makefile b/sound/soc/intel/common/Makefile index 668fdeee195e..3b9332e7a094 100644 --- a/sound/soc/intel/common/Makefile +++ b/sound/soc/intel/common/Makefile @@ -1,10 +1,5 @@ snd-soc-sst-dsp-objs := sst-dsp.o -ifneq ($(CONFIG_SND_SST_IPC_ACPI),) -snd-soc-sst-acpi-objs := sst-match-acpi.o -else snd-soc-sst-acpi-objs := sst-acpi.o sst-match-acpi.o -endif - snd-soc-sst-ipc-objs := sst-ipc.o snd-soc-sst-dsp-$(CONFIG_DW_DMAC_CORE) += sst-firmware.o -- cgit v1.2.1 From 2dcffcee23a2bd491a8c4041db3a8041b23fa4eb Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Mon, 8 Feb 2016 10:45:37 +0530 Subject: ASoC: Intel: Create independent acpi match module The ACPI match module is common to all three drivers, HSW, SKL and Atom-DPCM driver. But Atom-DPCM driver does not use common sst code so we cannot include the common SST module in Atom-DPCM driver. So the solution is to have a independent sst-match-acpi module which helps in matching for all the three drivers. Now all driver can be inbuilt in a single image This patch really fixes the regression introduced by the commit 95f098014815 ("ASoC: Intel: Move apci find machine routines") Acked-by: Jie Yang Acked-by: Pierre-Louis Bossart Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/Kconfig | 9 +++++++++ sound/soc/intel/common/Makefile | 4 +++- 2 files changed, 12 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 803f95e40679..af7aabbc0977 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -30,11 +30,15 @@ config SND_SST_IPC_ACPI config SND_SOC_INTEL_SST tristate select SND_SOC_INTEL_SST_ACPI if ACPI + select SND_SOC_INTEL_SST_MATCH if ACPI depends on (X86 || COMPILE_TEST) config SND_SOC_INTEL_SST_ACPI tristate +config SND_SOC_INTEL_SST_MATCH + tristate + config SND_SOC_INTEL_HASWELL tristate @@ -97,6 +101,7 @@ config SND_SOC_INTEL_BYTCR_RT5640_MACH select SND_SOC_RT5640 select SND_SST_MFLD_PLATFORM select SND_SST_IPC_ACPI + select SND_SOC_INTEL_SST_MATCH if ACPI help This adds support for ASoC machine driver for Intel(R) Baytrail and Baytrail-CR platforms with RT5640 audio codec. @@ -109,6 +114,7 @@ config SND_SOC_INTEL_BYTCR_RT5651_MACH select SND_SOC_RT5651 select SND_SST_MFLD_PLATFORM select SND_SST_IPC_ACPI + select SND_SOC_INTEL_SST_MATCH if ACPI help This adds support for ASoC machine driver for Intel(R) Baytrail and Baytrail-CR platforms with RT5651 audio codec. @@ -121,6 +127,7 @@ config SND_SOC_INTEL_CHT_BSW_RT5672_MACH select SND_SOC_RT5670 select SND_SST_MFLD_PLATFORM select SND_SST_IPC_ACPI + select SND_SOC_INTEL_SST_MATCH if ACPI help This adds support for ASoC machine driver for Intel(R) Cherrytrail & Braswell platforms with RT5672 audio codec. @@ -133,6 +140,7 @@ config SND_SOC_INTEL_CHT_BSW_RT5645_MACH select SND_SOC_RT5645 select SND_SST_MFLD_PLATFORM select SND_SST_IPC_ACPI + select SND_SOC_INTEL_SST_MATCH if ACPI help This adds support for ASoC machine driver for Intel(R) Cherrytrail & Braswell platforms with RT5645/5650 audio codec. @@ -145,6 +153,7 @@ config SND_SOC_INTEL_CHT_BSW_MAX98090_TI_MACH select SND_SOC_TS3A227E select SND_SST_MFLD_PLATFORM select SND_SST_IPC_ACPI + select SND_SOC_INTEL_SST_MATCH if ACPI help This adds support for ASoC machine driver for Intel(R) Cherrytrail & Braswell platforms with MAX98090 audio codec it also can support TI jack chip as aux device. diff --git a/sound/soc/intel/common/Makefile b/sound/soc/intel/common/Makefile index 3b9332e7a094..fbbb25c2ceed 100644 --- a/sound/soc/intel/common/Makefile +++ b/sound/soc/intel/common/Makefile @@ -1,8 +1,10 @@ snd-soc-sst-dsp-objs := sst-dsp.o -snd-soc-sst-acpi-objs := sst-acpi.o sst-match-acpi.o +snd-soc-sst-acpi-objs := sst-acpi.o +snd-soc-sst-match-objs := sst-match-acpi.o snd-soc-sst-ipc-objs := sst-ipc.o snd-soc-sst-dsp-$(CONFIG_DW_DMAC_CORE) += sst-firmware.o obj-$(CONFIG_SND_SOC_INTEL_SST) += snd-soc-sst-dsp.o snd-soc-sst-ipc.o obj-$(CONFIG_SND_SOC_INTEL_SST_ACPI) += snd-soc-sst-acpi.o +obj-$(CONFIG_SND_SOC_INTEL_SST_MATCH) += snd-soc-sst-match.o -- cgit v1.2.1 From cfffcc66a89ab6d9961b2cde6cdab2ba056451ad Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Mon, 8 Feb 2016 10:45:38 +0530 Subject: ASoC: Intel: Load the atom DPCM driver only DPCM driver is recommended for BYT, CHT based platforms, so if CONFIG_SND_SST_IPC_ACPI is selected then don't compile the BYT Device IDs in common ACPI driver to avoid probe conflicts. Signed-off-by: Pierre-Louis Bossart Acked-by: Jie Yang Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/Kconfig | 4 ++-- sound/soc/intel/common/sst-acpi.c | 4 ++++ 2 files changed, 6 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index af7aabbc0977..7d7c872c280d 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -61,7 +61,7 @@ config SND_SOC_INTEL_HASWELL_MACH config SND_SOC_INTEL_BYT_RT5640_MACH tristate "ASoC Audio driver for Intel Baytrail with RT5640 codec" depends on X86_INTEL_LPSS && I2C - depends on DW_DMAC_CORE=y && (SND_SOC_INTEL_BYTCR_RT5640_MACH = n) + depends on DW_DMAC_CORE=y && (SND_SST_IPC_ACPI = n) select SND_SOC_INTEL_SST select SND_SOC_INTEL_BAYTRAIL select SND_SOC_RT5640 @@ -73,7 +73,7 @@ config SND_SOC_INTEL_BYT_RT5640_MACH config SND_SOC_INTEL_BYT_MAX98090_MACH tristate "ASoC Audio driver for Intel Baytrail with MAX98090 codec" depends on X86_INTEL_LPSS && I2C - depends on DW_DMAC_CORE=y + depends on DW_DMAC_CORE=y && (SND_SST_IPC_ACPI = n) select SND_SOC_INTEL_SST select SND_SOC_INTEL_BAYTRAIL select SND_SOC_MAX98090 diff --git a/sound/soc/intel/common/sst-acpi.c b/sound/soc/intel/common/sst-acpi.c index 7a85c576dad3..2c5eda14d510 100644 --- a/sound/soc/intel/common/sst-acpi.c +++ b/sound/soc/intel/common/sst-acpi.c @@ -215,6 +215,7 @@ static struct sst_acpi_desc sst_acpi_broadwell_desc = { .dma_size = SST_LPT_DSP_DMA_SIZE, }; +#if !IS_ENABLED(CONFIG_SND_SST_IPC_ACPI) static struct sst_acpi_mach baytrail_machines[] = { { "10EC5640", "byt-rt5640", "intel/fw_sst_0f28.bin-48kHz_i2s_master", NULL, NULL, NULL }, { "193C9890", "byt-max98090", "intel/fw_sst_0f28.bin-48kHz_i2s_master", NULL, NULL, NULL }, @@ -231,11 +232,14 @@ static struct sst_acpi_desc sst_acpi_baytrail_desc = { .sst_id = SST_DEV_ID_BYT, .resindex_dma_base = -1, }; +#endif static const struct acpi_device_id sst_acpi_match[] = { { "INT33C8", (unsigned long)&sst_acpi_haswell_desc }, { "INT3438", (unsigned long)&sst_acpi_broadwell_desc }, +#if !IS_ENABLED(CONFIG_SND_SST_IPC_ACPI) { "80860F28", (unsigned long)&sst_acpi_baytrail_desc }, +#endif { } }; MODULE_DEVICE_TABLE(acpi, sst_acpi_match); -- cgit v1.2.1 From 8ceffd229f0ef130530c79654e95b5fa007ae639 Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Mon, 8 Feb 2016 10:45:39 +0530 Subject: ASoC: Intel: Add module tags for common match module The match module lacked module license and description, so add it Acked-by: Pierre-Louis Bossart Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/common/sst-match-acpi.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/soc/intel/common/sst-match-acpi.c b/sound/soc/intel/common/sst-match-acpi.c index dd077e116d25..3b4539d21492 100644 --- a/sound/soc/intel/common/sst-match-acpi.c +++ b/sound/soc/intel/common/sst-match-acpi.c @@ -41,3 +41,6 @@ struct sst_acpi_mach *sst_acpi_find_machine(struct sst_acpi_mach *machines) return NULL; } EXPORT_SYMBOL_GPL(sst_acpi_find_machine); + +MODULE_LICENSE("GPL v2"); +MODULE_DESCRIPTION("Intel Common ACPI Match module"); -- cgit v1.2.1 From 117159f0b9d392fb433a7871426fad50317f06f7 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 8 Feb 2016 17:36:25 +0100 Subject: ALSA: timer: Fix wrong instance passed to slave callbacks In snd_timer_notify1(), the wrong timer instance was passed for slave ccallback function. This leads to the access to the wrong data when an incompatible master is handled (e.g. the master is the sequencer timer and the slave is a user timer), as spotted by syzkaller fuzzer. This patch fixes that wrong assignment. BugLink: http://lkml.kernel.org/r/CACT4Y+Y_Bm+7epAb=8Wi=AaWd+DYS7qawX52qxdCfOfY49vozQ@mail.gmail.com Reported-by: Dmitry Vyukov Cc: Signed-off-by: Takashi Iwai --- sound/core/timer.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/core/timer.c b/sound/core/timer.c index 9b513a05765a..dea932ac6165 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -422,7 +422,7 @@ static void snd_timer_notify1(struct snd_timer_instance *ti, int event) spin_lock_irqsave(&timer->lock, flags); list_for_each_entry(ts, &ti->slave_active_head, active_list) if (ts->ccallback) - ts->ccallback(ti, event + 100, &tstamp, resolution); + ts->ccallback(ts, event + 100, &tstamp, resolution); spin_unlock_irqrestore(&timer->lock, flags); } -- cgit v1.2.1 From ed8b1d6d2c741ab26d60d499d7fbb7ac801f0f51 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 9 Feb 2016 12:02:32 +0100 Subject: ALSA: timer: Fix race between stop and interrupt A slave timer element also unlinks at snd_timer_stop() but it takes only slave_active_lock. When a slave is assigned to a master, however, this may become a race against the master's interrupt handling, eventually resulting in a list corruption. The actual bug could be seen with a syzkaller fuzzer test case in BugLink below. As a fix, we need to take timeri->timer->lock when timer isn't NULL, i.e. assigned to a master, while the assignment to a master itself is protected by slave_active_lock. BugLink: http://lkml.kernel.org/r/CACT4Y+Y_Bm+7epAb=8Wi=AaWd+DYS7qawX52qxdCfOfY49vozQ@mail.gmail.com Cc: Signed-off-by: Takashi Iwai --- sound/core/timer.c | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'sound') diff --git a/sound/core/timer.c b/sound/core/timer.c index dea932ac6165..a0405b0078c6 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -518,9 +518,13 @@ static int _snd_timer_stop(struct snd_timer_instance *timeri, int event) spin_unlock_irqrestore(&slave_active_lock, flags); return -EBUSY; } + if (timeri->timer) + spin_lock(&timeri->timer->lock); timeri->flags &= ~SNDRV_TIMER_IFLG_RUNNING; list_del_init(&timeri->ack_list); list_del_init(&timeri->active_list); + if (timeri->timer) + spin_unlock(&timeri->timer->lock); spin_unlock_irqrestore(&slave_active_lock, flags); goto __end; } -- cgit v1.2.1 From 2ebab40eb74a0225d5dfba72bfae317dd948fa2d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 9 Feb 2016 10:23:52 +0100 Subject: ALSA: hda - Fix bad dereference of jack object The hda_jack_tbl entries are managed by snd_array for allowing multiple jacks. It's good per se, but the problem is that struct hda_jack_callback keeps the hda_jack_tbl pointer. Since snd_array doesn't preserve each pointer at resizing the array, we can't keep the original pointer but have to deduce the pointer at each time via snd_array_entry() instead. Actually, this resulted in the deference to the wrong pointer on codecs that have many pins such as CS4208. This patch replaces the pointer to the NID value as the search key. As an unexpected good side effect, this even simplifies the code, as only NID is needed in most cases. Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_generic.c | 4 ++-- sound/pci/hda/hda_jack.c | 2 +- sound/pci/hda/hda_jack.h | 2 +- sound/pci/hda/patch_ca0132.c | 5 ++++- sound/pci/hda/patch_hdmi.c | 2 +- sound/pci/hda/patch_realtek.c | 2 +- sound/pci/hda/patch_sigmatel.c | 6 +++--- 7 files changed, 13 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 30c8efe0f80a..7ca5b89f088a 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -4028,9 +4028,9 @@ static void pin_power_callback(struct hda_codec *codec, struct hda_jack_callback *jack, bool on) { - if (jack && jack->tbl->nid) + if (jack && jack->nid) sync_power_state_change(codec, - set_pin_power_jack(codec, jack->tbl->nid, on)); + set_pin_power_jack(codec, jack->nid, on)); } /* callback only doing power up -- called at first */ diff --git a/sound/pci/hda/hda_jack.c b/sound/pci/hda/hda_jack.c index c945e257d368..a33234e04d4f 100644 --- a/sound/pci/hda/hda_jack.c +++ b/sound/pci/hda/hda_jack.c @@ -259,7 +259,7 @@ snd_hda_jack_detect_enable_callback(struct hda_codec *codec, hda_nid_t nid, if (!callback) return ERR_PTR(-ENOMEM); callback->func = func; - callback->tbl = jack; + callback->nid = jack->nid; callback->next = jack->callback; jack->callback = callback; } diff --git a/sound/pci/hda/hda_jack.h b/sound/pci/hda/hda_jack.h index 858708a044f5..e9814c0168ea 100644 --- a/sound/pci/hda/hda_jack.h +++ b/sound/pci/hda/hda_jack.h @@ -21,7 +21,7 @@ struct hda_jack_callback; typedef void (*hda_jack_callback_fn) (struct hda_codec *, struct hda_jack_callback *); struct hda_jack_callback { - struct hda_jack_tbl *tbl; + hda_nid_t nid; hda_jack_callback_fn func; unsigned int private_data; /* arbitrary data */ struct hda_jack_callback *next; diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 4ef2259f88ca..9ceb2bc36e68 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -4427,13 +4427,16 @@ static void ca0132_process_dsp_response(struct hda_codec *codec, static void hp_callback(struct hda_codec *codec, struct hda_jack_callback *cb) { struct ca0132_spec *spec = codec->spec; + struct hda_jack_tbl *tbl; /* Delay enabling the HP amp, to let the mic-detection * state machine run. */ cancel_delayed_work_sync(&spec->unsol_hp_work); schedule_delayed_work(&spec->unsol_hp_work, msecs_to_jiffies(500)); - cb->tbl->block_report = 1; + tbl = snd_hda_jack_tbl_get(codec, cb->nid); + if (tbl) + tbl->block_report = 1; } static void amic_callback(struct hda_codec *codec, struct hda_jack_callback *cb) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 2191e2359315..8ee78dbd4c60 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1194,7 +1194,7 @@ static void check_presence_and_report(struct hda_codec *codec, hda_nid_t nid) static void jack_callback(struct hda_codec *codec, struct hda_jack_callback *jack) { - check_presence_and_report(codec, jack->tbl->nid); + check_presence_and_report(codec, jack->nid); } static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b43c0f7994eb..efd4980cffb8 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -282,7 +282,7 @@ static void alc_update_knob_master(struct hda_codec *codec, uctl = kzalloc(sizeof(*uctl), GFP_KERNEL); if (!uctl) return; - val = snd_hda_codec_read(codec, jack->tbl->nid, 0, + val = snd_hda_codec_read(codec, jack->nid, 0, AC_VERB_GET_VOLUME_KNOB_CONTROL, 0); val &= HDA_AMP_VOLMASK; uctl->value.integer.value[0] = val; diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 2c7c5eb8b1e9..37b70f8e878f 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -493,9 +493,9 @@ static void jack_update_power(struct hda_codec *codec, if (!spec->num_pwrs) return; - if (jack && jack->tbl->nid) { - stac_toggle_power_map(codec, jack->tbl->nid, - snd_hda_jack_detect(codec, jack->tbl->nid), + if (jack && jack->nid) { + stac_toggle_power_map(codec, jack->nid, + snd_hda_jack_detect(codec, jack->nid), true); return; } -- cgit v1.2.1 From b8cb3750ce94d7610934465263850dcf40736bca Mon Sep 17 00:00:00 2001 From: Geert Uytterhoeven Date: Sun, 7 Feb 2016 15:14:15 +0100 Subject: ALSA: firewire-digi00x: Drop bogus const type qualifier on dot_scrt() sound/firewire/digi00x/amdtp-dot.c:67: warning: type qualifiers ignored on function return type Drop the bogus "const" type qualifier on the return type of dot_scrt() to fix this. Signed-off-by: Geert Uytterhoeven Reviewed-by: Takashi Sakamoto Signed-off-by: Takashi Iwai --- sound/firewire/digi00x/amdtp-dot.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/firewire/digi00x/amdtp-dot.c b/sound/firewire/digi00x/amdtp-dot.c index b02a5e8cad44..0ac92aba5bc1 100644 --- a/sound/firewire/digi00x/amdtp-dot.c +++ b/sound/firewire/digi00x/amdtp-dot.c @@ -63,7 +63,7 @@ struct amdtp_dot { #define BYTE_PER_SAMPLE (4) #define MAGIC_DOT_BYTE (2) #define MAGIC_BYTE_OFF(x) (((x) * BYTE_PER_SAMPLE) + MAGIC_DOT_BYTE) -static const u8 dot_scrt(const u8 idx, const unsigned int off) +static u8 dot_scrt(const u8 idx, const unsigned int off) { /* * the length of the added pattern only depends on the lower nibble -- cgit v1.2.1 From 4dff5c7b7093b19c19d3a100f8a3ad87cb7cd9e7 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 8 Feb 2016 17:26:58 +0100 Subject: ALSA: timer: Fix race at concurrent reads snd_timer_user_read() has a potential race among parallel reads, as qhead and qused are updated outside the critical section due to copy_to_user() calls. Move them into the critical section, and also sanitize the relevant code a bit. Cc: Signed-off-by: Takashi Iwai --- sound/core/timer.c | 34 +++++++++++++++------------------- 1 file changed, 15 insertions(+), 19 deletions(-) (limited to 'sound') diff --git a/sound/core/timer.c b/sound/core/timer.c index a0405b0078c6..dca817fc7894 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -1933,6 +1933,7 @@ static ssize_t snd_timer_user_read(struct file *file, char __user *buffer, { struct snd_timer_user *tu; long result = 0, unit; + int qhead; int err = 0; tu = file->private_data; @@ -1944,7 +1945,7 @@ static ssize_t snd_timer_user_read(struct file *file, char __user *buffer, if ((file->f_flags & O_NONBLOCK) != 0 || result > 0) { err = -EAGAIN; - break; + goto _error; } set_current_state(TASK_INTERRUPTIBLE); @@ -1959,42 +1960,37 @@ static ssize_t snd_timer_user_read(struct file *file, char __user *buffer, if (tu->disconnected) { err = -ENODEV; - break; + goto _error; } if (signal_pending(current)) { err = -ERESTARTSYS; - break; + goto _error; } } + qhead = tu->qhead++; + tu->qhead %= tu->queue_size; spin_unlock_irq(&tu->qlock); - if (err < 0) - goto _error; if (tu->tread) { - if (copy_to_user(buffer, &tu->tqueue[tu->qhead++], - sizeof(struct snd_timer_tread))) { + if (copy_to_user(buffer, &tu->tqueue[qhead], + sizeof(struct snd_timer_tread))) err = -EFAULT; - goto _error; - } } else { - if (copy_to_user(buffer, &tu->queue[tu->qhead++], - sizeof(struct snd_timer_read))) { + if (copy_to_user(buffer, &tu->queue[qhead], + sizeof(struct snd_timer_read))) err = -EFAULT; - goto _error; - } } - tu->qhead %= tu->queue_size; - - result += unit; - buffer += unit; - spin_lock_irq(&tu->qlock); tu->qused--; + if (err < 0) + goto _error; + result += unit; + buffer += unit; } - spin_unlock_irq(&tu->qlock); _error: + spin_unlock_irq(&tu->qlock); return result > 0 ? result : err; } -- cgit v1.2.1 From 61c4a1ac4d900e743af0b363fe520405939eab47 Mon Sep 17 00:00:00 2001 From: Pascal Huerst Date: Wed, 10 Feb 2016 15:59:28 +0100 Subject: ASoC: sigmadsp: Fix missleading return value Forwarding the return value of i2c_master_send, leads to errors later on, since i2c_master_send returns the number of bytes transmittet. Check for ret < 0 instead and return 0 otherwise. Signed-off-by: Pascal Huerst Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/sigmadsp-i2c.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/sigmadsp-i2c.c b/sound/soc/codecs/sigmadsp-i2c.c index 21ca3a5e9f66..d374c18d4db7 100644 --- a/sound/soc/codecs/sigmadsp-i2c.c +++ b/sound/soc/codecs/sigmadsp-i2c.c @@ -31,7 +31,10 @@ static int sigmadsp_write_i2c(void *control_data, kfree(buf); - return ret; + if (ret < 0) + return ret; + + return 0; } static int sigmadsp_read_i2c(void *control_data, -- cgit v1.2.1 From 01582a841493f28caf1688b2af4dafbcbee8135e Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Wed, 10 Feb 2016 11:56:13 +0000 Subject: ASoC: arizona: fref must be limited in pseudo-fractional mode When the FLL is in pseudo-fractional mode there is an additional limit on fref based on the fratio, to prevent aliasing around the Nyquist frequency. If fref exceeds this limit the refclk divider must be increased and the calculation tried again until a suitable combination of fref and fratio is found or we have to fall back to integer mode. This patch also adds some debug log prints around this code. Signed-off-by: Richard Fitzgerald Signed-off-by: Mark Brown --- sound/soc/codecs/arizona.c | 43 ++++++++++++++++++++++++++++++++++++++++++- 1 file changed, 42 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 33143fe1de0b..91785318b283 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1929,6 +1929,25 @@ static struct { { 1000000, 13500000, 0, 1 }, }; +static const unsigned int pseudo_fref_max[ARIZONA_FLL_MAX_FRATIO] = { + 13500000, + 6144000, + 6144000, + 3072000, + 3072000, + 2822400, + 2822400, + 1536000, + 1536000, + 1536000, + 1536000, + 1536000, + 1536000, + 1536000, + 1536000, + 768000, +}; + static struct { unsigned int min; unsigned int max; @@ -2042,16 +2061,32 @@ static int arizona_calc_fratio(struct arizona_fll *fll, /* Adjust FRATIO/refdiv to avoid integer mode if possible */ refdiv = cfg->refdiv; + arizona_fll_dbg(fll, "pseudo: initial ratio=%u fref=%u refdiv=%u\n", + init_ratio, Fref, refdiv); + while (div <= ARIZONA_FLL_MAX_REFDIV) { for (ratio = init_ratio; ratio <= ARIZONA_FLL_MAX_FRATIO; ratio++) { if ((ARIZONA_FLL_VCO_CORNER / 2) / - (fll->vco_mult * ratio) < Fref) + (fll->vco_mult * ratio) < Fref) { + arizona_fll_dbg(fll, "pseudo: hit VCO corner\n"); break; + } + + if (Fref > pseudo_fref_max[ratio - 1]) { + arizona_fll_dbg(fll, + "pseudo: exceeded max fref(%u) for ratio=%u\n", + pseudo_fref_max[ratio - 1], + ratio); + break; + } if (target % (ratio * Fref)) { cfg->refdiv = refdiv; cfg->fratio = ratio - 1; + arizona_fll_dbg(fll, + "pseudo: found fref=%u refdiv=%d(%d) ratio=%d\n", + Fref, refdiv, div, ratio); return ratio; } } @@ -2060,6 +2095,9 @@ static int arizona_calc_fratio(struct arizona_fll *fll, if (target % (ratio * Fref)) { cfg->refdiv = refdiv; cfg->fratio = ratio - 1; + arizona_fll_dbg(fll, + "pseudo: found fref=%u refdiv=%d(%d) ratio=%d\n", + Fref, refdiv, div, ratio); return ratio; } } @@ -2068,6 +2106,9 @@ static int arizona_calc_fratio(struct arizona_fll *fll, Fref /= 2; refdiv++; init_ratio = arizona_find_fratio(Fref, NULL); + arizona_fll_dbg(fll, + "pseudo: change fref=%u refdiv=%d(%d) ratio=%u\n", + Fref, refdiv, div, init_ratio); } arizona_fll_warn(fll, "Falling back to integer mode operation\n"); -- cgit v1.2.1 From a947b724069a25eae86d8dfed905374d04c3f93c Mon Sep 17 00:00:00 2001 From: Stephen Boyd Date: Fri, 12 Feb 2016 09:30:17 -0800 Subject: ASoC: qcom: Don't specify LE device endianness This reverts commit 18560a4e3 (ASoC: qcom: Specify LE device endianness). The commit that caused us to specify LE device endianness here, 29bb45f25ff3 (regmap-mmio: Use native endianness for read/write, 2015-10-29), has been reverted in mainline so now when we specify LE it actively breaks big endian kernels because the byte swapping in regmap-mmio is incorrect. Let's revert this change because it will 1) fix the big endian kernels and 2) be redundant to specify LE because that will become the default soon. Signed-off-by: Stephen Boyd Signed-off-by: Mark Brown --- sound/soc/qcom/lpass-cpu.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c index 00b6c9d039cf..e5101e0d2d37 100644 --- a/sound/soc/qcom/lpass-cpu.c +++ b/sound/soc/qcom/lpass-cpu.c @@ -355,7 +355,6 @@ static struct regmap_config lpass_cpu_regmap_config = { .readable_reg = lpass_cpu_regmap_readable, .volatile_reg = lpass_cpu_regmap_volatile, .cache_type = REGCACHE_FLAT, - .val_format_endian = REGMAP_ENDIAN_LITTLE, }; int asoc_qcom_lpass_cpu_platform_probe(struct platform_device *pdev) -- cgit v1.2.1 From 07d86ca93db7e5cdf4743564d98292042ec21af7 Mon Sep 17 00:00:00 2001 From: Andrey Konovalov Date: Sat, 13 Feb 2016 11:08:06 +0300 Subject: ALSA: usb-audio: avoid freeing umidi object twice The 'umidi' object will be free'd on the error path by snd_usbmidi_free() when tearing down the rawmidi interface. So we shouldn't try to free it in snd_usbmidi_create() after having registered the rawmidi interface. Found by KASAN. Signed-off-by: Andrey Konovalov Acked-by: Clemens Ladisch Cc: Signed-off-by: Takashi Iwai --- sound/usb/midi.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/usb/midi.c b/sound/usb/midi.c index cc39f63299ef..007cf5831121 100644 --- a/sound/usb/midi.c +++ b/sound/usb/midi.c @@ -2455,7 +2455,6 @@ int snd_usbmidi_create(struct snd_card *card, else err = snd_usbmidi_create_endpoints(umidi, endpoints); if (err < 0) { - snd_usbmidi_free(umidi); return err; } -- cgit v1.2.1 From d99a36f4728fcbcc501b78447f625bdcce15b842 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 15 Feb 2016 16:20:24 +0100 Subject: ALSA: seq: Fix leak of pool buffer at concurrent writes When multiple concurrent writes happen on the ALSA sequencer device right after the open, it may try to allocate vmalloc buffer for each write and leak some of them. It's because the presence check and the assignment of the buffer is done outside the spinlock for the pool. The fix is to move the check and the assignment into the spinlock. (The current implementation is suboptimal, as there can be multiple unnecessary vmallocs because the allocation is done before the check in the spinlock. But the pool size is already checked beforehand, so this isn't a big problem; that is, the only possible path is the multiple writes before any pool assignment, and practically seen, the current coverage should be "good enough".) The issue was triggered by syzkaller fuzzer. BugLink: http://lkml.kernel.org/r/CACT4Y+bSzazpXNvtAr=WXaL8hptqjHwqEyFA+VN2AWEx=aurkg@mail.gmail.com Reported-by: Dmitry Vyukov Tested-by: Dmitry Vyukov Cc: Signed-off-by: Takashi Iwai --- sound/core/seq/seq_memory.c | 13 +++++++++---- 1 file changed, 9 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/core/seq/seq_memory.c b/sound/core/seq/seq_memory.c index 801076687bb1..c850345c43b5 100644 --- a/sound/core/seq/seq_memory.c +++ b/sound/core/seq/seq_memory.c @@ -383,15 +383,20 @@ int snd_seq_pool_init(struct snd_seq_pool *pool) if (snd_BUG_ON(!pool)) return -EINVAL; - if (pool->ptr) /* should be atomic? */ - return 0; - pool->ptr = vmalloc(sizeof(struct snd_seq_event_cell) * pool->size); - if (!pool->ptr) + cellptr = vmalloc(sizeof(struct snd_seq_event_cell) * pool->size); + if (!cellptr) return -ENOMEM; /* add new cells to the free cell list */ spin_lock_irqsave(&pool->lock, flags); + if (pool->ptr) { + spin_unlock_irqrestore(&pool->lock, flags); + vfree(cellptr); + return 0; + } + + pool->ptr = cellptr; pool->free = NULL; for (cell = 0; cell < pool->size; cell++) { -- cgit v1.2.1 From 0b8c82190c12e530eb6003720dac103bf63e146e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 15 Feb 2016 16:37:24 +0100 Subject: ALSA: hda - Cancel probe work instead of flush at remove The commit [991f86d7ae4e: ALSA: hda - Flush the pending probe work at remove] introduced the sync of async probe work at remove for fixing the race. However, this may lead to another hangup when the module removal is performed quickly before starting the probe work, because it issues flush_work() and it's blocked forever. The workaround is to use cancel_work_sync() instead of flush_work() there. Fixes: 991f86d7ae4e ('ALSA: hda - Flush the pending probe work at remove') Cc: # v3.17+ Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 4045dca3d699..ce6b97f31390 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2168,10 +2168,10 @@ static void azx_remove(struct pci_dev *pci) struct hda_intel *hda; if (card) { - /* flush the pending probing work */ + /* cancel the pending probing work */ chip = card->private_data; hda = container_of(chip, struct hda_intel, chip); - flush_work(&hda->probe_work); + cancel_work_sync(&hda->probe_work); snd_card_free(card); } -- cgit v1.2.1 From 13d5e5d4725c64ec06040d636832e78453f477b7 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 16 Feb 2016 14:15:59 +0100 Subject: ALSA: seq: Fix double port list deletion The commit [7f0973e973cd: ALSA: seq: Fix lockdep warnings due to double mutex locks] split the management of two linked lists (source and destination) into two individual calls for avoiding the AB/BA deadlock. However, this may leave the possible double deletion of one of two lists when the counterpart is being deleted concurrently. It ends up with a list corruption, as revealed by syzkaller fuzzer. This patch fixes it by checking the list emptiness and skipping the deletion and the following process. BugLink: http://lkml.kernel.org/r/CACT4Y+bay9qsrz6dQu31EcGaH9XwfW7o3oBzSQUG9fMszoh=Sg@mail.gmail.com Fixes: 7f0973e973cd ('ALSA: seq: Fix lockdep warnings due to 'double mutex locks) Reported-by: Dmitry Vyukov Tested-by: Dmitry Vyukov Cc: Signed-off-by: Takashi Iwai --- sound/core/seq/seq_ports.c | 13 ++++++++----- 1 file changed, 8 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/core/seq/seq_ports.c b/sound/core/seq/seq_ports.c index 921fb2bd8fad..fe686ee41c6d 100644 --- a/sound/core/seq/seq_ports.c +++ b/sound/core/seq/seq_ports.c @@ -535,19 +535,22 @@ static void delete_and_unsubscribe_port(struct snd_seq_client *client, bool is_src, bool ack) { struct snd_seq_port_subs_info *grp; + struct list_head *list; + bool empty; grp = is_src ? &port->c_src : &port->c_dest; + list = is_src ? &subs->src_list : &subs->dest_list; down_write(&grp->list_mutex); write_lock_irq(&grp->list_lock); - if (is_src) - list_del(&subs->src_list); - else - list_del(&subs->dest_list); + empty = list_empty(list); + if (!empty) + list_del_init(list); grp->exclusive = 0; write_unlock_irq(&grp->list_lock); up_write(&grp->list_mutex); - unsubscribe_port(client, port, grp, &subs->info, ack); + if (!empty) + unsubscribe_port(client, port, grp, &subs->info, ack); } /* connect two ports */ -- cgit v1.2.1 From 67ec1072b053c15564e6090ab30127895dc77a89 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 17 Feb 2016 14:30:26 +0100 Subject: ALSA: pcm: Fix rwsem deadlock for non-atomic PCM stream A non-atomic PCM stream may take snd_pcm_link_rwsem rw semaphore twice in the same code path, e.g. one in snd_pcm_action_nonatomic() and another in snd_pcm_stream_lock(). Usually this is OK, but when a write lock is issued between these two read locks, the problem happens: the write lock is blocked due to the first reade lock, and the second read lock is also blocked by the write lock. This eventually deadlocks. The reason is the way rwsem manages waiters; it's queued like FIFO, so even if the writer itself doesn't take the lock yet, it blocks all the waiters (including reads) queued after it. As a workaround, in this patch, we replace the standard down_write() with an spinning loop. This is far from optimal, but it's good enough, as the spinning time is supposed to be relatively short for normal PCM operations, and the code paths requiring the write lock aren't called so often. Reported-by: Vinod Koul Tested-by: Ramesh Babu Cc: # v3.18+ Signed-off-by: Takashi Iwai --- sound/core/pcm_native.c | 16 ++++++++++++++-- 1 file changed, 14 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index fadd3eb8e8bb..9106d8e2300e 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -74,6 +74,18 @@ static int snd_pcm_open(struct file *file, struct snd_pcm *pcm, int stream); static DEFINE_RWLOCK(snd_pcm_link_rwlock); static DECLARE_RWSEM(snd_pcm_link_rwsem); +/* Writer in rwsem may block readers even during its waiting in queue, + * and this may lead to a deadlock when the code path takes read sem + * twice (e.g. one in snd_pcm_action_nonatomic() and another in + * snd_pcm_stream_lock()). As a (suboptimal) workaround, let writer to + * spin until it gets the lock. + */ +static inline void down_write_nonblock(struct rw_semaphore *lock) +{ + while (!down_write_trylock(lock)) + cond_resched(); +} + /** * snd_pcm_stream_lock - Lock the PCM stream * @substream: PCM substream @@ -1813,7 +1825,7 @@ static int snd_pcm_link(struct snd_pcm_substream *substream, int fd) res = -ENOMEM; goto _nolock; } - down_write(&snd_pcm_link_rwsem); + down_write_nonblock(&snd_pcm_link_rwsem); write_lock_irq(&snd_pcm_link_rwlock); if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN || substream->runtime->status->state != substream1->runtime->status->state || @@ -1860,7 +1872,7 @@ static int snd_pcm_unlink(struct snd_pcm_substream *substream) struct snd_pcm_substream *s; int res = 0; - down_write(&snd_pcm_link_rwsem); + down_write_nonblock(&snd_pcm_link_rwsem); write_lock_irq(&snd_pcm_link_rwlock); if (!snd_pcm_stream_linked(substream)) { res = -EALREADY; -- cgit v1.2.1 From 316fa9e09ad76e095b9d7e9350c628b918370a22 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 18 Feb 2016 15:47:13 +0000 Subject: ASoC: samsung: Use IRQ safe spin lock calls Lockdep warns of a potential lock inversion, i2s->lock is held numerous times whilst we are under the substream lock (snd_pcm_stream_lock). If we use the IRQ unsafe spin lock calls, you can also end up locking snd_pcm_stream_lock whilst under i2s->lock (if an IRQ happens whilst we are holding i2s->lock). This could result in deadlock. [ 18.147001] CPU0 CPU1 [ 18.151509] ---- ---- [ 18.156022] lock(&(&pri_dai->spinlock)->rlock); [ 18.160701] local_irq_disable(); [ 18.166622] lock(&(&substream->self_group.lock)->rlock); [ 18.174595] lock(&(&pri_dai->spinlock)->rlock); [ 18.181806] [ 18.184408] lock(&(&substream->self_group.lock)->rlock); [ 18.190045] [ 18.190045] *** DEADLOCK *** This patch changes to using the irq safe spinlock calls, to avoid this issue. Fixes: ce8bcdbb61d9 ("ASoC: samsung: i2s: Protect more registers with a spinlock") Signed-off-by: Charles Keepax Tested-by: Anand Moon Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/samsung/i2s.c | 21 ++++++++++++--------- 1 file changed, 12 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 84d9e77c0fbe..70a2559b63f9 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -481,10 +481,11 @@ static int i2s_set_sysclk(struct snd_soc_dai *dai, unsigned int cdcon_mask = 1 << i2s_regs->cdclkcon_off; unsigned int rsrc_mask = 1 << i2s_regs->rclksrc_off; u32 mod, mask, val = 0; + unsigned long flags; - spin_lock(i2s->lock); + spin_lock_irqsave(i2s->lock, flags); mod = readl(i2s->addr + I2SMOD); - spin_unlock(i2s->lock); + spin_unlock_irqrestore(i2s->lock, flags); switch (clk_id) { case SAMSUNG_I2S_OPCLK: @@ -575,11 +576,11 @@ static int i2s_set_sysclk(struct snd_soc_dai *dai, return -EINVAL; } - spin_lock(i2s->lock); + spin_lock_irqsave(i2s->lock, flags); mod = readl(i2s->addr + I2SMOD); mod = (mod & ~mask) | val; writel(mod, i2s->addr + I2SMOD); - spin_unlock(i2s->lock); + spin_unlock_irqrestore(i2s->lock, flags); return 0; } @@ -590,6 +591,7 @@ static int i2s_set_fmt(struct snd_soc_dai *dai, struct i2s_dai *i2s = to_info(dai); int lrp_shift, sdf_shift, sdf_mask, lrp_rlow, mod_slave; u32 mod, tmp = 0; + unsigned long flags; lrp_shift = i2s->variant_regs->lrp_off; sdf_shift = i2s->variant_regs->sdf_off; @@ -649,7 +651,7 @@ static int i2s_set_fmt(struct snd_soc_dai *dai, return -EINVAL; } - spin_lock(i2s->lock); + spin_lock_irqsave(i2s->lock, flags); mod = readl(i2s->addr + I2SMOD); /* * Don't change the I2S mode if any controller is active on this @@ -657,7 +659,7 @@ static int i2s_set_fmt(struct snd_soc_dai *dai, */ if (any_active(i2s) && ((mod & (sdf_mask | lrp_rlow | mod_slave)) != tmp)) { - spin_unlock(i2s->lock); + spin_unlock_irqrestore(i2s->lock, flags); dev_err(&i2s->pdev->dev, "%s:%d Other DAI busy\n", __func__, __LINE__); return -EAGAIN; @@ -666,7 +668,7 @@ static int i2s_set_fmt(struct snd_soc_dai *dai, mod &= ~(sdf_mask | lrp_rlow | mod_slave); mod |= tmp; writel(mod, i2s->addr + I2SMOD); - spin_unlock(i2s->lock); + spin_unlock_irqrestore(i2s->lock, flags); return 0; } @@ -676,6 +678,7 @@ static int i2s_hw_params(struct snd_pcm_substream *substream, { struct i2s_dai *i2s = to_info(dai); u32 mod, mask = 0, val = 0; + unsigned long flags; if (!is_secondary(i2s)) mask |= (MOD_DC2_EN | MOD_DC1_EN); @@ -744,11 +747,11 @@ static int i2s_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - spin_lock(i2s->lock); + spin_lock_irqsave(i2s->lock, flags); mod = readl(i2s->addr + I2SMOD); mod = (mod & ~mask) | val; writel(mod, i2s->addr + I2SMOD); - spin_unlock(i2s->lock); + spin_unlock_irqrestore(i2s->lock, flags); samsung_asoc_init_dma_data(dai, &i2s->dma_playback, &i2s->dma_capture); -- cgit v1.2.1 From 65b4bcb82967fd5a60694c3477e58a04a9170aea Mon Sep 17 00:00:00 2001 From: Alan Date: Fri, 19 Feb 2016 11:42:32 +0530 Subject: ASoC: Intel: Skylake: fix pointer scaling skl_tplg_tlv_control_set does pointer maths on data but forgets that data is not uint8_t so the maths is already scaled in the pointer type. Signed-off-by: Alan Cox Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/skylake/skl-topology.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index 4624556f486d..b77c253394d7 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -950,7 +950,7 @@ static int skl_tplg_tlv_control_set(struct snd_kcontrol *kcontrol, return -EFAULT; } else { if (copy_from_user(ac->params, - data + 2 * sizeof(u32), size)) + data + 2, size)) return -EFAULT; } -- cgit v1.2.1 From 56e5fd8feb286ab71f4ca7674505b0d17967376d Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Sun, 21 Feb 2016 11:35:00 -0300 Subject: ASoC: fsl_ssi: Go back to explicit register defaults MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Commit 5c408fee2546 ("ASoC: fsl_ssi: remove explicit register defaults") causes the driver to fail to probe: fsl-ssi-dai 2028000.ssi: No cache defaults, reading back from HW fsl-ssi-dai 2028000.ssi: Failed to init register map fsl-ssi-dai: probe of 2028000.ssi failed with error -22 , so revert this commit. Reported-by: Mika Penttilä Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_ssi.c | 42 ++++++++++++++++++++---------------------- 1 file changed, 20 insertions(+), 22 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index ed8de1035cda..40dfd8a36484 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -112,6 +112,20 @@ struct fsl_ssi_rxtx_reg_val { struct fsl_ssi_reg_val tx; }; +static const struct reg_default fsl_ssi_reg_defaults[] = { + {CCSR_SSI_SCR, 0x00000000}, + {CCSR_SSI_SIER, 0x00003003}, + {CCSR_SSI_STCR, 0x00000200}, + {CCSR_SSI_SRCR, 0x00000200}, + {CCSR_SSI_STCCR, 0x00040000}, + {CCSR_SSI_SRCCR, 0x00040000}, + {CCSR_SSI_SACNT, 0x00000000}, + {CCSR_SSI_STMSK, 0x00000000}, + {CCSR_SSI_SRMSK, 0x00000000}, + {CCSR_SSI_SACCEN, 0x00000000}, + {CCSR_SSI_SACCDIS, 0x00000000}, +}; + static bool fsl_ssi_readable_reg(struct device *dev, unsigned int reg) { switch (reg) { @@ -176,7 +190,8 @@ static const struct regmap_config fsl_ssi_regconfig = { .val_bits = 32, .reg_stride = 4, .val_format_endian = REGMAP_ENDIAN_NATIVE, - .num_reg_defaults_raw = CCSR_SSI_SACCDIS / sizeof(uint32_t) + 1, + .reg_defaults = fsl_ssi_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(fsl_ssi_reg_defaults), .readable_reg = fsl_ssi_readable_reg, .volatile_reg = fsl_ssi_volatile_reg, .precious_reg = fsl_ssi_precious_reg, @@ -186,7 +201,6 @@ static const struct regmap_config fsl_ssi_regconfig = { struct fsl_ssi_soc_data { bool imx; - bool imx21regs; /* imx21-class SSI - no SACC{ST,EN,DIS} regs */ bool offline_config; u32 sisr_write_mask; }; @@ -289,7 +303,6 @@ static struct fsl_ssi_soc_data fsl_ssi_mpc8610 = { static struct fsl_ssi_soc_data fsl_ssi_imx21 = { .imx = true, - .imx21regs = true, .offline_config = true, .sisr_write_mask = 0, }; @@ -573,12 +586,8 @@ static void fsl_ssi_setup_ac97(struct fsl_ssi_private *ssi_private) */ regmap_write(regs, CCSR_SSI_SACNT, CCSR_SSI_SACNT_AC97EN | CCSR_SSI_SACNT_FV); - - /* no SACC{ST,EN,DIS} regs on imx21-class SSI */ - if (!ssi_private->soc->imx21regs) { - regmap_write(regs, CCSR_SSI_SACCDIS, 0xff); - regmap_write(regs, CCSR_SSI_SACCEN, 0x300); - } + regmap_write(regs, CCSR_SSI_SACCDIS, 0xff); + regmap_write(regs, CCSR_SSI_SACCEN, 0x300); /* * Enable SSI, Transmit and Receive. AC97 has to communicate with the @@ -1388,7 +1397,6 @@ static int fsl_ssi_probe(struct platform_device *pdev) struct resource *res; void __iomem *iomem; char name[64]; - struct regmap_config regconfig = fsl_ssi_regconfig; of_id = of_match_device(fsl_ssi_ids, &pdev->dev); if (!of_id || !of_id->data) @@ -1436,25 +1444,15 @@ static int fsl_ssi_probe(struct platform_device *pdev) return PTR_ERR(iomem); ssi_private->ssi_phys = res->start; - if (ssi_private->soc->imx21regs) { - /* - * According to datasheet imx21-class SSI - * don't have SACC{ST,EN,DIS} regs. - */ - regconfig.max_register = CCSR_SSI_SRMSK; - regconfig.num_reg_defaults_raw = - CCSR_SSI_SRMSK / sizeof(uint32_t) + 1; - } - ret = of_property_match_string(np, "clock-names", "ipg"); if (ret < 0) { ssi_private->has_ipg_clk_name = false; ssi_private->regs = devm_regmap_init_mmio(&pdev->dev, iomem, - ®config); + &fsl_ssi_regconfig); } else { ssi_private->has_ipg_clk_name = true; ssi_private->regs = devm_regmap_init_mmio_clk(&pdev->dev, - "ipg", iomem, ®config); + "ipg", iomem, &fsl_ssi_regconfig); } if (IS_ERR(ssi_private->regs)) { dev_err(&pdev->dev, "Failed to init register map\n"); -- cgit v1.2.1 From 7e31a0159461818a1bda49662921b98a29c1187b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 22 Feb 2016 15:18:13 +0100 Subject: ALSA: hda - Apply clock gate workaround to Skylake, too Some Skylake machines show the codec probe errors in certain situations, e.g. HP Z240 desktop fails to probe the onboard Realtek codec at reloading the snd-hda-intel module like: snd_hda_intel 0000:00:1f.3: spurious response 0x200:0x2, last cmd=0x000000 snd_hda_intel 0000:00:1f.3: azx_get_response timeout, switching to polling mode: lastcmd=0x000f0000 snd_hda_intel 0000:00:1f.3: No response from codec, disabling MSI: last cmd=0x000f0000 snd_hda_intel 0000:00:1f.3: Codec #0 probe error; disabling it... hdaudio hdaudioC0D2: no AFG or MFG node found snd_hda_intel 0000:00:1f.3: no codecs initialized Also, HP G470 G3 suffers from the similar problem, as reported in bugzilla below. On this machine, the codec probe error appears even at a fresh boot. As Libin suggested, the same workaround used for Broxton in the commit [6639484ddaf6: ALSA: hda - disable dynamic clock gating on Broxton before reset] can be applied for Skylake in order to fix this problem. The Intel HW team also confirmed that this is needed for SKL. This patch makes the workaround applied to both SKL and BXT platforms. The referred macros are moved and one superfluous macro (IS_BROXTON()) is another one (IS_BXT()) as well. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=112731 Suggested-by: Libin Yang Cc: # v4.4+ Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 16 +++++++--------- 1 file changed, 7 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index ce6b97f31390..e5240cb3749f 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -363,7 +363,10 @@ enum { ((pci)->device == 0x0d0c) || \ ((pci)->device == 0x160c)) -#define IS_BROXTON(pci) ((pci)->device == 0x5a98) +#define IS_SKL(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0xa170) +#define IS_SKL_LP(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0x9d70) +#define IS_BXT(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0x5a98) +#define IS_SKL_PLUS(pci) (IS_SKL(pci) || IS_SKL_LP(pci) || IS_BXT(pci)) static char *driver_short_names[] = { [AZX_DRIVER_ICH] = "HDA Intel", @@ -540,13 +543,13 @@ static void hda_intel_init_chip(struct azx *chip, bool full_reset) if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) snd_hdac_set_codec_wakeup(bus, true); - if (IS_BROXTON(pci)) { + if (IS_SKL_PLUS(pci)) { pci_read_config_dword(pci, INTEL_HDA_CGCTL, &val); val = val & ~INTEL_HDA_CGCTL_MISCBDCGE; pci_write_config_dword(pci, INTEL_HDA_CGCTL, val); } azx_init_chip(chip, full_reset); - if (IS_BROXTON(pci)) { + if (IS_SKL_PLUS(pci)) { pci_read_config_dword(pci, INTEL_HDA_CGCTL, &val); val = val | INTEL_HDA_CGCTL_MISCBDCGE; pci_write_config_dword(pci, INTEL_HDA_CGCTL, val); @@ -555,7 +558,7 @@ static void hda_intel_init_chip(struct azx *chip, bool full_reset) snd_hdac_set_codec_wakeup(bus, false); /* reduce dma latency to avoid noise */ - if (IS_BROXTON(pci)) + if (IS_BXT(pci)) bxt_reduce_dma_latency(chip); } @@ -977,11 +980,6 @@ static int azx_resume(struct device *dev) /* put codec down to D3 at hibernation for Intel SKL+; * otherwise BIOS may still access the codec and screw up the driver */ -#define IS_SKL(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0xa170) -#define IS_SKL_LP(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0x9d70) -#define IS_BXT(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0x5a98) -#define IS_SKL_PLUS(pci) (IS_SKL(pci) || IS_SKL_LP(pci) || IS_BXT(pci)) - static int azx_freeze_noirq(struct device *dev) { struct pci_dev *pci = to_pci_dev(dev); -- cgit v1.2.1 From da9b9303ed8d1673a89a4bdd85464e33614775e3 Mon Sep 17 00:00:00 2001 From: Robert Jarzmik Date: Mon, 22 Feb 2016 23:35:44 +0100 Subject: ASoC: wm9713: fix regmap free path In the conversion to regmap, I assumed that the devm_() variant could be used in the soc probe function. As a mater of fact with the current code the regmap is freed twice because of the devm_() call: (mutex_lock) from [] (debugfs_remove_recursive+0x50/0x1d0) (debugfs_remove_recursive) from [] (regmap_debugfs_exit+0x1c/0xd4) (regmap_debugfs_exit) from [] (regmap_exit+0x28/0xc8) (regmap_exit) from [] (release_nodes+0x18c/0x204) (release_nodes) from [] (device_release+0x18/0x90) (device_release) from [] (kobject_release+0x90/0x1bc) (kobject_release) from [] (wm9713_soc_remove+0x1c/0x24) (wm9713_soc_remove) from [] (soc_remove_component+0x50/0x7c) (soc_remove_component) from [] (soc_remove_dai_links+0x118/0x228) (soc_remove_dai_links) from [] (snd_soc_register_card+0x4e4/0xdd4) (snd_soc_register_card) from [] (devm_snd_soc_register_card+0x34/0x70) Fix this by replacing the devm_regmap initialization code with the non devm_() variant. Fixes: 700dadfefc3d ASoC: wm9713: convert to regmap Signed-off-by: Robert Jarzmik Acked-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm9713.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index 79e143625ac3..9849643ef809 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -1212,7 +1212,7 @@ static int wm9713_soc_probe(struct snd_soc_codec *codec) if (IS_ERR(wm9713->ac97)) return PTR_ERR(wm9713->ac97); - regmap = devm_regmap_init_ac97(wm9713->ac97, &wm9713_regmap_config); + regmap = regmap_init_ac97(wm9713->ac97, &wm9713_regmap_config); if (IS_ERR(regmap)) { snd_soc_free_ac97_codec(wm9713->ac97); return PTR_ERR(regmap); -- cgit v1.2.1 From 7c139db2e579669c3313f92d2dd2256b255fcc07 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Andreas=20Irest=C3=A5l?= Date: Tue, 16 Feb 2016 13:56:41 +0100 Subject: ASoC: adau17x1: Fix incorrect BCLK ratio definitions MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Signed-off-by: Andreas Irestål Signed-off-by: Mark Brown --- sound/soc/codecs/adau17x1.h | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/adau17x1.h b/sound/soc/codecs/adau17x1.h index e13583e6ff56..5ae87a084d97 100644 --- a/sound/soc/codecs/adau17x1.h +++ b/sound/soc/codecs/adau17x1.h @@ -103,9 +103,9 @@ bool adau17x1_has_dsp(struct adau *adau); #define ADAU17X1_CLOCK_CONTROL_CORECLK_SRC_PLL BIT(3) #define ADAU17X1_CLOCK_CONTROL_SYSCLK_EN BIT(0) -#define ADAU17X1_SERIAL_PORT1_BCLK32 (0x0 << 5) -#define ADAU17X1_SERIAL_PORT1_BCLK48 (0x1 << 5) -#define ADAU17X1_SERIAL_PORT1_BCLK64 (0x2 << 5) +#define ADAU17X1_SERIAL_PORT1_BCLK64 (0x0 << 5) +#define ADAU17X1_SERIAL_PORT1_BCLK32 (0x1 << 5) +#define ADAU17X1_SERIAL_PORT1_BCLK48 (0x2 << 5) #define ADAU17X1_SERIAL_PORT1_BCLK128 (0x3 << 5) #define ADAU17X1_SERIAL_PORT1_BCLK256 (0x4 << 5) #define ADAU17X1_SERIAL_PORT1_BCLK_MASK (0x7 << 5) -- cgit v1.2.1 From 3b43b71f05d3ecd01c4116254666d9492301697d Mon Sep 17 00:00:00 2001 From: Kai-Heng Feng Date: Thu, 25 Feb 2016 15:19:38 +0800 Subject: ALSA: hda - Fixing background noise on Dell Inspiron 3162 After login to the desktop on Dell Inspiron 3162, there's a very loud background noise comes from the builtin speaker. The noise does not go away even if the speaker is muted. The noise disappears after using the aamix fixup. Codec: Realtek ALC3234 Address: 0 AFG Function Id: 0x1 (unsol 1) Vendor Id: 0x10ec0255 Subsystem Id: 0x10280725 Revision Id: 0x100002 No Modem Function Group found BugLink: http://bugs.launchpad.net/bugs/1549620 Signed-off-by: Kai-Heng Feng Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 8 ++++++++ 1 file changed, 8 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index efd4980cffb8..72fa58dd7723 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4749,6 +4749,7 @@ enum { ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE, ALC293_FIXUP_LENOVO_SPK_NOISE, ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY, + ALC255_FIXUP_DELL_SPK_NOISE, }; static const struct hda_fixup alc269_fixups[] = { @@ -5368,6 +5369,12 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc233_fixup_lenovo_line2_mic_hotkey, }, + [ALC255_FIXUP_DELL_SPK_NOISE] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_disable_aamix, + .chained = true, + .chain_id = ALC255_FIXUP_DELL1_MIC_NO_PRESENCE + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -5410,6 +5417,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x06df, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK), SND_PCI_QUIRK(0x1028, 0x06e0, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK), SND_PCI_QUIRK(0x1028, 0x0704, "Dell XPS 13", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE), + SND_PCI_QUIRK(0x1028, 0x0725, "Dell Inspiron 3162", ALC255_FIXUP_DELL_SPK_NOISE), SND_PCI_QUIRK(0x1028, 0x164a, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x164b, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC2), -- cgit v1.2.1 From 2ae955774f29bbd7d16149cb0ae8d0319bf2ecc4 Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Thu, 25 Feb 2016 09:37:05 +0100 Subject: ALSA: hda - Fixup speaker pass-through control for nid 0x14 on ALC225 On one of the machines we enable, we found that the actual speaker volume did not always correspond to the volume set in alsamixer. This patch fixes that problem. This patch was orginally written by Kailang @ Realtek, I've rebased it to fit sound git master. Cc: stable@vger.kernel.org BugLink: https://bugs.launchpad.net/bugs/1549660 Co-Authored-By: Kailang Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 23 +++++++++++++++++++++-- 1 file changed, 21 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 72fa58dd7723..7fded69fb58e 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3801,6 +3801,10 @@ static void alc_headset_mode_mic_in(struct hda_codec *codec, hda_nid_t hp_pin, static void alc_headset_mode_default(struct hda_codec *codec) { + static struct coef_fw coef0225[] = { + UPDATE_COEF(0x45, 0x3f<<10, 0x34<<10), + {} + }; static struct coef_fw coef0255[] = { WRITE_COEF(0x45, 0xc089), WRITE_COEF(0x45, 0xc489), @@ -3842,6 +3846,9 @@ static void alc_headset_mode_default(struct hda_codec *codec) }; switch (codec->core.vendor_id) { + case 0x10ec0225: + alc_process_coef_fw(codec, coef0225); + break; case 0x10ec0255: case 0x10ec0256: alc_process_coef_fw(codec, coef0255); @@ -4750,6 +4757,7 @@ enum { ALC293_FIXUP_LENOVO_SPK_NOISE, ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY, ALC255_FIXUP_DELL_SPK_NOISE, + ALC225_FIXUP_DELL1_MIC_NO_PRESENCE, }; static const struct hda_fixup alc269_fixups[] = { @@ -5375,6 +5383,17 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC255_FIXUP_DELL1_MIC_NO_PRESENCE }, + [ALC225_FIXUP_DELL1_MIC_NO_PRESENCE] = { + .type = HDA_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + /* Disable pass-through path for FRONT 14h */ + { 0x20, AC_VERB_SET_COEF_INDEX, 0x36 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x57d7 }, + {} + }, + .chained = true, + .chain_id = ALC269_FIXUP_DELL1_MIC_NO_PRESENCE + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -5646,10 +5665,10 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {0x21, 0x03211020} static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { - SND_HDA_PIN_QUIRK(0x10ec0225, 0x1028, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE, + SND_HDA_PIN_QUIRK(0x10ec0225, 0x1028, "Dell", ALC225_FIXUP_DELL1_MIC_NO_PRESENCE, ALC225_STANDARD_PINS, {0x14, 0x901701a0}), - SND_HDA_PIN_QUIRK(0x10ec0225, 0x1028, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE, + SND_HDA_PIN_QUIRK(0x10ec0225, 0x1028, "Dell", ALC225_FIXUP_DELL1_MIC_NO_PRESENCE, ALC225_STANDARD_PINS, {0x14, 0x901701b0}), SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL2_MIC_NO_PRESENCE, -- cgit v1.2.1 From f883982dc1b117f04579f0896821cd9f2e397f94 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 25 Feb 2016 14:31:59 +0100 Subject: ALSA: hda - Fix headset support and noise on HP EliteBook 755 G2 HP EliteBook 755 G2 with ALC3228 (ALC280) codec [103c:221c] requires the known fixup (ALC269_FIXUP_HEADSET_MIC) for making the headset mic working. Also, it suffers from the loopback noise problem, so we should disable aamix path as well. Reported-by: Derick Eddington Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 8 ++++++++ 1 file changed, 8 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 7fded69fb58e..1f357cd72d9c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4758,6 +4758,7 @@ enum { ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY, ALC255_FIXUP_DELL_SPK_NOISE, ALC225_FIXUP_DELL1_MIC_NO_PRESENCE, + ALC280_FIXUP_HP_HEADSET_MIC, }; static const struct hda_fixup alc269_fixups[] = { @@ -5394,6 +5395,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_DELL1_MIC_NO_PRESENCE }, + [ALC280_FIXUP_HP_HEADSET_MIC] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_disable_aamix, + .chained = true, + .chain_id = ALC269_FIXUP_HEADSET_MIC, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -5497,6 +5504,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x2335, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x2336, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x2337, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x221c, "HP EliteBook 755 G2", ALC280_FIXUP_HP_HEADSET_MIC), SND_PCI_QUIRK(0x1043, 0x103f, "ASUS TX300", ALC282_FIXUP_ASUS_TX300), SND_PCI_QUIRK(0x1043, 0x106d, "Asus K53BE", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x1043, 0x115d, "Asus 1015E", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), -- cgit v1.2.1 From c8560b7c917f8738f5d80dd516930edc1d05e4e4 Mon Sep 17 00:00:00 2001 From: Carlo Caione Date: Tue, 23 Feb 2016 09:50:20 +0100 Subject: ASoC: cht_bsw_rt5645: Fix writing to string literal We cannot use strcpy() to write to a const char * location. This is causing a 'BUG: unable to handle kernel paging request' error at boot when using the cht-bsw-rt5645 driver. With this patch we also fix a wrong indexing in the driver where the codec_name of the wrong dai_link is being overwritten. Signed-off-by: Carlo Caione Signed-off-by: Mark Brown --- sound/soc/intel/boards/cht_bsw_rt5645.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/intel/boards/cht_bsw_rt5645.c b/sound/soc/intel/boards/cht_bsw_rt5645.c index 2d3afddb0a2e..a7b96a9a4e0e 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5645.c +++ b/sound/soc/intel/boards/cht_bsw_rt5645.c @@ -367,8 +367,12 @@ static int snd_cht_mc_probe(struct platform_device *pdev) } card->dev = &pdev->dev; sprintf(codec_name, "i2c-%s:00", drv->acpi_card->codec_id); + /* set correct codec name */ - strcpy((char *)card->dai_link[2].codec_name, codec_name); + for (i = 0; i < ARRAY_SIZE(cht_dailink); i++) + if (!strcmp(card->dai_link[i].codec_name, "i2c-10EC5645:00")) + card->dai_link[i].codec_name = kstrdup(codec_name, GFP_KERNEL); + snd_soc_card_set_drvdata(card, drv); ret_val = devm_snd_soc_register_card(&pdev->dev, card); if (ret_val) { -- cgit v1.2.1 From 473f414564528a819f0c2bb6b4bf26366b64c9ab Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 23 Feb 2016 15:54:47 +0100 Subject: ALSA: hda - Loop interrupt handling until really cleared Currently the interrupt handler of HD-audio driver assumes that no irq update is needed while processing the irq. But in reality, it has been confirmed that the HW irq is issued even during the irq handling. Since we clear the irq status at the beginning, process the interrupt, then exits from the handler, the lately issued interrupt is left untouched without being properly processed. This patch changes the interrupt handler code to loop over the check-and-process. The handler tries repeatedly as long as the IRQ status are turned on, and either stream or CORB/RIRB is handled. For checking the stream handling, snd_hdac_bus_handle_stream_irq() returns a value indicating the stream indices bits. Other than that, the change is only in the irq handler itself. Reported-by: Libin Yang Cc: Signed-off-by: Takashi Iwai --- sound/hda/hdac_controller.c | 7 ++++++- sound/pci/hda/hda_controller.c | 47 +++++++++++++++++++++++------------------- 2 files changed, 32 insertions(+), 22 deletions(-) (limited to 'sound') diff --git a/sound/hda/hdac_controller.c b/sound/hda/hdac_controller.c index b5a17cb510a0..8c486235c905 100644 --- a/sound/hda/hdac_controller.c +++ b/sound/hda/hdac_controller.c @@ -426,18 +426,22 @@ EXPORT_SYMBOL_GPL(snd_hdac_bus_stop_chip); * @bus: HD-audio core bus * @status: INTSTS register value * @ask: callback to be called for woken streams + * + * Returns the bits of handled streams, or zero if no stream is handled. */ -void snd_hdac_bus_handle_stream_irq(struct hdac_bus *bus, unsigned int status, +int snd_hdac_bus_handle_stream_irq(struct hdac_bus *bus, unsigned int status, void (*ack)(struct hdac_bus *, struct hdac_stream *)) { struct hdac_stream *azx_dev; u8 sd_status; + int handled = 0; list_for_each_entry(azx_dev, &bus->stream_list, list) { if (status & azx_dev->sd_int_sta_mask) { sd_status = snd_hdac_stream_readb(azx_dev, SD_STS); snd_hdac_stream_writeb(azx_dev, SD_STS, SD_INT_MASK); + handled |= 1 << azx_dev->index; if (!azx_dev->substream || !azx_dev->running || !(sd_status & SD_INT_COMPLETE)) continue; @@ -445,6 +449,7 @@ void snd_hdac_bus_handle_stream_irq(struct hdac_bus *bus, unsigned int status, ack(bus, azx_dev); } } + return handled; } EXPORT_SYMBOL_GPL(snd_hdac_bus_handle_stream_irq); diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index 37cf9cee9835..27de8015717d 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -930,6 +930,8 @@ irqreturn_t azx_interrupt(int irq, void *dev_id) struct azx *chip = dev_id; struct hdac_bus *bus = azx_bus(chip); u32 status; + bool active, handled = false; + int repeat = 0; /* count for avoiding endless loop */ #ifdef CONFIG_PM if (azx_has_pm_runtime(chip)) @@ -939,33 +941,36 @@ irqreturn_t azx_interrupt(int irq, void *dev_id) spin_lock(&bus->reg_lock); - if (chip->disabled) { - spin_unlock(&bus->reg_lock); - return IRQ_NONE; - } - - status = azx_readl(chip, INTSTS); - if (status == 0 || status == 0xffffffff) { - spin_unlock(&bus->reg_lock); - return IRQ_NONE; - } + if (chip->disabled) + goto unlock; - snd_hdac_bus_handle_stream_irq(bus, status, stream_update); + do { + status = azx_readl(chip, INTSTS); + if (status == 0 || status == 0xffffffff) + break; - /* clear rirb int */ - status = azx_readb(chip, RIRBSTS); - if (status & RIRB_INT_MASK) { - if (status & RIRB_INT_RESPONSE) { - if (chip->driver_caps & AZX_DCAPS_CTX_WORKAROUND) - udelay(80); - snd_hdac_bus_update_rirb(bus); + handled = true; + active = false; + if (snd_hdac_bus_handle_stream_irq(bus, status, stream_update)) + active = true; + + /* clear rirb int */ + status = azx_readb(chip, RIRBSTS); + if (status & RIRB_INT_MASK) { + active = true; + if (status & RIRB_INT_RESPONSE) { + if (chip->driver_caps & AZX_DCAPS_CTX_WORKAROUND) + udelay(80); + snd_hdac_bus_update_rirb(bus); + } + azx_writeb(chip, RIRBSTS, RIRB_INT_MASK); } - azx_writeb(chip, RIRBSTS, RIRB_INT_MASK); - } + } while (active && ++repeat < 10); + unlock: spin_unlock(&bus->reg_lock); - return IRQ_HANDLED; + return IRQ_RETVAL(handled); } EXPORT_SYMBOL_GPL(azx_interrupt); -- cgit v1.2.1 From 6236d8bb2afcfe71b88ecea554e0dc638090a45f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sat, 27 Feb 2016 17:52:42 +0100 Subject: ALSA: ctl: Fix ioctls for X32 ABI The X32 ABI takes the same alignment like x86-64, and this may result in the incompatible struct size from ia32. Unfortunately, we hit this in some control ABI: struct snd_ctl_elem_value differs between them due to the position of 64bit variable array. This ends up with the unknown ioctl (ENOTTY) error. The fix is to add the compat entries for the new aligned struct. Reported-and-tested-by: Steven Newbury Cc: # v3.4+ Signed-off-by: Takashi Iwai --- sound/core/control_compat.c | 90 +++++++++++++++++++++++++++++++++++++-------- 1 file changed, 74 insertions(+), 16 deletions(-) (limited to 'sound') diff --git a/sound/core/control_compat.c b/sound/core/control_compat.c index b9c0910fb8c4..0608f216f359 100644 --- a/sound/core/control_compat.c +++ b/sound/core/control_compat.c @@ -170,6 +170,19 @@ struct snd_ctl_elem_value32 { unsigned char reserved[128]; }; +#ifdef CONFIG_X86_X32 +/* x32 has a different alignment for 64bit values from ia32 */ +struct snd_ctl_elem_value_x32 { + struct snd_ctl_elem_id id; + unsigned int indirect; /* bit-field causes misalignment */ + union { + s32 integer[128]; + unsigned char data[512]; + s64 integer64[64]; + } value; + unsigned char reserved[128]; +}; +#endif /* CONFIG_X86_X32 */ /* get the value type and count of the control */ static int get_ctl_type(struct snd_card *card, struct snd_ctl_elem_id *id, @@ -219,9 +232,11 @@ static int get_elem_size(int type, int count) static int copy_ctl_value_from_user(struct snd_card *card, struct snd_ctl_elem_value *data, - struct snd_ctl_elem_value32 __user *data32, + void __user *userdata, + void __user *valuep, int *typep, int *countp) { + struct snd_ctl_elem_value32 __user *data32 = userdata; int i, type, size; int uninitialized_var(count); unsigned int indirect; @@ -239,8 +254,9 @@ static int copy_ctl_value_from_user(struct snd_card *card, if (type == SNDRV_CTL_ELEM_TYPE_BOOLEAN || type == SNDRV_CTL_ELEM_TYPE_INTEGER) { for (i = 0; i < count; i++) { + s32 __user *intp = valuep; int val; - if (get_user(val, &data32->value.integer[i])) + if (get_user(val, &intp[i])) return -EFAULT; data->value.integer.value[i] = val; } @@ -250,8 +266,7 @@ static int copy_ctl_value_from_user(struct snd_card *card, dev_err(card->dev, "snd_ioctl32_ctl_elem_value: unknown type %d\n", type); return -EINVAL; } - if (copy_from_user(data->value.bytes.data, - data32->value.data, size)) + if (copy_from_user(data->value.bytes.data, valuep, size)) return -EFAULT; } @@ -261,7 +276,8 @@ static int copy_ctl_value_from_user(struct snd_card *card, } /* restore the value to 32bit */ -static int copy_ctl_value_to_user(struct snd_ctl_elem_value32 __user *data32, +static int copy_ctl_value_to_user(void __user *userdata, + void __user *valuep, struct snd_ctl_elem_value *data, int type, int count) { @@ -270,22 +286,22 @@ static int copy_ctl_value_to_user(struct snd_ctl_elem_value32 __user *data32, if (type == SNDRV_CTL_ELEM_TYPE_BOOLEAN || type == SNDRV_CTL_ELEM_TYPE_INTEGER) { for (i = 0; i < count; i++) { + s32 __user *intp = valuep; int val; val = data->value.integer.value[i]; - if (put_user(val, &data32->value.integer[i])) + if (put_user(val, &intp[i])) return -EFAULT; } } else { size = get_elem_size(type, count); - if (copy_to_user(data32->value.data, - data->value.bytes.data, size)) + if (copy_to_user(valuep, data->value.bytes.data, size)) return -EFAULT; } return 0; } -static int snd_ctl_elem_read_user_compat(struct snd_card *card, - struct snd_ctl_elem_value32 __user *data32) +static int ctl_elem_read_user(struct snd_card *card, + void __user *userdata, void __user *valuep) { struct snd_ctl_elem_value *data; int err, type, count; @@ -294,7 +310,9 @@ static int snd_ctl_elem_read_user_compat(struct snd_card *card, if (data == NULL) return -ENOMEM; - if ((err = copy_ctl_value_from_user(card, data, data32, &type, &count)) < 0) + err = copy_ctl_value_from_user(card, data, userdata, valuep, + &type, &count); + if (err < 0) goto error; snd_power_lock(card); @@ -303,14 +321,15 @@ static int snd_ctl_elem_read_user_compat(struct snd_card *card, err = snd_ctl_elem_read(card, data); snd_power_unlock(card); if (err >= 0) - err = copy_ctl_value_to_user(data32, data, type, count); + err = copy_ctl_value_to_user(userdata, valuep, data, + type, count); error: kfree(data); return err; } -static int snd_ctl_elem_write_user_compat(struct snd_ctl_file *file, - struct snd_ctl_elem_value32 __user *data32) +static int ctl_elem_write_user(struct snd_ctl_file *file, + void __user *userdata, void __user *valuep) { struct snd_ctl_elem_value *data; struct snd_card *card = file->card; @@ -320,7 +339,9 @@ static int snd_ctl_elem_write_user_compat(struct snd_ctl_file *file, if (data == NULL) return -ENOMEM; - if ((err = copy_ctl_value_from_user(card, data, data32, &type, &count)) < 0) + err = copy_ctl_value_from_user(card, data, userdata, valuep, + &type, &count); + if (err < 0) goto error; snd_power_lock(card); @@ -329,12 +350,39 @@ static int snd_ctl_elem_write_user_compat(struct snd_ctl_file *file, err = snd_ctl_elem_write(card, file, data); snd_power_unlock(card); if (err >= 0) - err = copy_ctl_value_to_user(data32, data, type, count); + err = copy_ctl_value_to_user(userdata, valuep, data, + type, count); error: kfree(data); return err; } +static int snd_ctl_elem_read_user_compat(struct snd_card *card, + struct snd_ctl_elem_value32 __user *data32) +{ + return ctl_elem_read_user(card, data32, &data32->value); +} + +static int snd_ctl_elem_write_user_compat(struct snd_ctl_file *file, + struct snd_ctl_elem_value32 __user *data32) +{ + return ctl_elem_write_user(file, data32, &data32->value); +} + +#ifdef CONFIG_X86_X32 +static int snd_ctl_elem_read_user_x32(struct snd_card *card, + struct snd_ctl_elem_value_x32 __user *data32) +{ + return ctl_elem_read_user(card, data32, &data32->value); +} + +static int snd_ctl_elem_write_user_x32(struct snd_ctl_file *file, + struct snd_ctl_elem_value_x32 __user *data32) +{ + return ctl_elem_write_user(file, data32, &data32->value); +} +#endif /* CONFIG_X86_X32 */ + /* add or replace a user control */ static int snd_ctl_elem_add_compat(struct snd_ctl_file *file, struct snd_ctl_elem_info32 __user *data32, @@ -393,6 +441,10 @@ enum { SNDRV_CTL_IOCTL_ELEM_WRITE32 = _IOWR('U', 0x13, struct snd_ctl_elem_value32), SNDRV_CTL_IOCTL_ELEM_ADD32 = _IOWR('U', 0x17, struct snd_ctl_elem_info32), SNDRV_CTL_IOCTL_ELEM_REPLACE32 = _IOWR('U', 0x18, struct snd_ctl_elem_info32), +#ifdef CONFIG_X86_X32 + SNDRV_CTL_IOCTL_ELEM_READ_X32 = _IOWR('U', 0x12, struct snd_ctl_elem_value_x32), + SNDRV_CTL_IOCTL_ELEM_WRITE_X32 = _IOWR('U', 0x13, struct snd_ctl_elem_value_x32), +#endif /* CONFIG_X86_X32 */ }; static inline long snd_ctl_ioctl_compat(struct file *file, unsigned int cmd, unsigned long arg) @@ -431,6 +483,12 @@ static inline long snd_ctl_ioctl_compat(struct file *file, unsigned int cmd, uns return snd_ctl_elem_add_compat(ctl, argp, 0); case SNDRV_CTL_IOCTL_ELEM_REPLACE32: return snd_ctl_elem_add_compat(ctl, argp, 1); +#ifdef CONFIG_X86_X32 + case SNDRV_CTL_IOCTL_ELEM_READ_X32: + return snd_ctl_elem_read_user_x32(ctl->card, argp); + case SNDRV_CTL_IOCTL_ELEM_WRITE_X32: + return snd_ctl_elem_write_user_x32(ctl, argp); +#endif /* CONFIG_X86_X32 */ } down_read(&snd_ioctl_rwsem); -- cgit v1.2.1 From 513ace79b657e2022a592e77f24074e088681ecc Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sun, 28 Feb 2016 11:23:09 +0100 Subject: ALSA: pcm: Fix ioctls for X32 ABI X32 ABI uses the 64bit timespec in addition to 64bit alignment of 64bit values. This leads to incompatibilities in some PCM ioctls involved with snd_pcm_channel_info, snd_pcm_status and snd_pcm_sync_ptr structs. Fix the PCM compat ABI for these ioctls like the previous commit for ctl API. Reported-by: Steven Newbury Cc: # v3.4+ Signed-off-by: Takashi Iwai --- sound/core/pcm_compat.c | 177 +++++++++++++++++++++++++++++++++++++++++++++++- 1 file changed, 176 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/core/pcm_compat.c b/sound/core/pcm_compat.c index 9630e9f72b7b..1f64ab0c2a95 100644 --- a/sound/core/pcm_compat.c +++ b/sound/core/pcm_compat.c @@ -183,6 +183,14 @@ static int snd_pcm_ioctl_channel_info_compat(struct snd_pcm_substream *substream return err; } +#ifdef CONFIG_X86_X32 +/* X32 ABI has the same struct as x86-64 for snd_pcm_channel_info */ +static int snd_pcm_channel_info_user(struct snd_pcm_substream *substream, + struct snd_pcm_channel_info __user *src); +#define snd_pcm_ioctl_channel_info_x32(s, p) \ + snd_pcm_channel_info_user(s, p) +#endif /* CONFIG_X86_X32 */ + struct snd_pcm_status32 { s32 state; struct compat_timespec trigger_tstamp; @@ -243,6 +251,71 @@ static int snd_pcm_status_user_compat(struct snd_pcm_substream *substream, return err; } +#ifdef CONFIG_X86_X32 +/* X32 ABI has 64bit timespec and 64bit alignment */ +struct snd_pcm_status_x32 { + s32 state; + u32 rsvd; /* alignment */ + struct timespec trigger_tstamp; + struct timespec tstamp; + u32 appl_ptr; + u32 hw_ptr; + s32 delay; + u32 avail; + u32 avail_max; + u32 overrange; + s32 suspended_state; + u32 audio_tstamp_data; + struct timespec audio_tstamp; + struct timespec driver_tstamp; + u32 audio_tstamp_accuracy; + unsigned char reserved[52-2*sizeof(struct timespec)]; +} __packed; + +#define put_timespec(src, dst) copy_to_user(dst, src, sizeof(*dst)) + +static int snd_pcm_status_user_x32(struct snd_pcm_substream *substream, + struct snd_pcm_status_x32 __user *src, + bool ext) +{ + struct snd_pcm_status status; + int err; + + memset(&status, 0, sizeof(status)); + /* + * with extension, parameters are read/write, + * get audio_tstamp_data from user, + * ignore rest of status structure + */ + if (ext && get_user(status.audio_tstamp_data, + (u32 __user *)(&src->audio_tstamp_data))) + return -EFAULT; + err = snd_pcm_status(substream, &status); + if (err < 0) + return err; + + if (clear_user(src, sizeof(*src))) + return -EFAULT; + if (put_user(status.state, &src->state) || + put_timespec(&status.trigger_tstamp, &src->trigger_tstamp) || + put_timespec(&status.tstamp, &src->tstamp) || + put_user(status.appl_ptr, &src->appl_ptr) || + put_user(status.hw_ptr, &src->hw_ptr) || + put_user(status.delay, &src->delay) || + put_user(status.avail, &src->avail) || + put_user(status.avail_max, &src->avail_max) || + put_user(status.overrange, &src->overrange) || + put_user(status.suspended_state, &src->suspended_state) || + put_user(status.audio_tstamp_data, &src->audio_tstamp_data) || + put_timespec(&status.audio_tstamp, &src->audio_tstamp) || + put_timespec(&status.driver_tstamp, &src->driver_tstamp) || + put_user(status.audio_tstamp_accuracy, &src->audio_tstamp_accuracy)) + return -EFAULT; + + return err; +} +#endif /* CONFIG_X86_X32 */ + /* both for HW_PARAMS and HW_REFINE */ static int snd_pcm_ioctl_hw_params_compat(struct snd_pcm_substream *substream, int refine, @@ -469,6 +542,93 @@ static int snd_pcm_ioctl_sync_ptr_compat(struct snd_pcm_substream *substream, return 0; } +#ifdef CONFIG_X86_X32 +/* X32 ABI has 64bit timespec and 64bit alignment */ +struct snd_pcm_mmap_status_x32 { + s32 state; + s32 pad1; + u32 hw_ptr; + u32 pad2; /* alignment */ + struct timespec tstamp; + s32 suspended_state; + struct timespec audio_tstamp; +} __packed; + +struct snd_pcm_mmap_control_x32 { + u32 appl_ptr; + u32 avail_min; +}; + +struct snd_pcm_sync_ptr_x32 { + u32 flags; + u32 rsvd; /* alignment */ + union { + struct snd_pcm_mmap_status_x32 status; + unsigned char reserved[64]; + } s; + union { + struct snd_pcm_mmap_control_x32 control; + unsigned char reserved[64]; + } c; +} __packed; + +static int snd_pcm_ioctl_sync_ptr_x32(struct snd_pcm_substream *substream, + struct snd_pcm_sync_ptr_x32 __user *src) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + volatile struct snd_pcm_mmap_status *status; + volatile struct snd_pcm_mmap_control *control; + u32 sflags; + struct snd_pcm_mmap_control scontrol; + struct snd_pcm_mmap_status sstatus; + snd_pcm_uframes_t boundary; + int err; + + if (snd_BUG_ON(!runtime)) + return -EINVAL; + + if (get_user(sflags, &src->flags) || + get_user(scontrol.appl_ptr, &src->c.control.appl_ptr) || + get_user(scontrol.avail_min, &src->c.control.avail_min)) + return -EFAULT; + if (sflags & SNDRV_PCM_SYNC_PTR_HWSYNC) { + err = snd_pcm_hwsync(substream); + if (err < 0) + return err; + } + status = runtime->status; + control = runtime->control; + boundary = recalculate_boundary(runtime); + if (!boundary) + boundary = 0x7fffffff; + snd_pcm_stream_lock_irq(substream); + /* FIXME: we should consider the boundary for the sync from app */ + if (!(sflags & SNDRV_PCM_SYNC_PTR_APPL)) + control->appl_ptr = scontrol.appl_ptr; + else + scontrol.appl_ptr = control->appl_ptr % boundary; + if (!(sflags & SNDRV_PCM_SYNC_PTR_AVAIL_MIN)) + control->avail_min = scontrol.avail_min; + else + scontrol.avail_min = control->avail_min; + sstatus.state = status->state; + sstatus.hw_ptr = status->hw_ptr % boundary; + sstatus.tstamp = status->tstamp; + sstatus.suspended_state = status->suspended_state; + sstatus.audio_tstamp = status->audio_tstamp; + snd_pcm_stream_unlock_irq(substream); + if (put_user(sstatus.state, &src->s.status.state) || + put_user(sstatus.hw_ptr, &src->s.status.hw_ptr) || + put_timespec(&sstatus.tstamp, &src->s.status.tstamp) || + put_user(sstatus.suspended_state, &src->s.status.suspended_state) || + put_timespec(&sstatus.audio_tstamp, &src->s.status.audio_tstamp) || + put_user(scontrol.appl_ptr, &src->c.control.appl_ptr) || + put_user(scontrol.avail_min, &src->c.control.avail_min)) + return -EFAULT; + + return 0; +} +#endif /* CONFIG_X86_X32 */ /* */ @@ -487,7 +647,12 @@ enum { SNDRV_PCM_IOCTL_WRITEN_FRAMES32 = _IOW('A', 0x52, struct snd_xfern32), SNDRV_PCM_IOCTL_READN_FRAMES32 = _IOR('A', 0x53, struct snd_xfern32), SNDRV_PCM_IOCTL_SYNC_PTR32 = _IOWR('A', 0x23, struct snd_pcm_sync_ptr32), - +#ifdef CONFIG_X86_X32 + SNDRV_PCM_IOCTL_CHANNEL_INFO_X32 = _IOR('A', 0x32, struct snd_pcm_channel_info), + SNDRV_PCM_IOCTL_STATUS_X32 = _IOR('A', 0x20, struct snd_pcm_status_x32), + SNDRV_PCM_IOCTL_STATUS_EXT_X32 = _IOWR('A', 0x24, struct snd_pcm_status_x32), + SNDRV_PCM_IOCTL_SYNC_PTR_X32 = _IOWR('A', 0x23, struct snd_pcm_sync_ptr_x32), +#endif /* CONFIG_X86_X32 */ }; static long snd_pcm_ioctl_compat(struct file *file, unsigned int cmd, unsigned long arg) @@ -559,6 +724,16 @@ static long snd_pcm_ioctl_compat(struct file *file, unsigned int cmd, unsigned l return snd_pcm_ioctl_rewind_compat(substream, argp); case SNDRV_PCM_IOCTL_FORWARD32: return snd_pcm_ioctl_forward_compat(substream, argp); +#ifdef CONFIG_X86_X32 + case SNDRV_PCM_IOCTL_STATUS_X32: + return snd_pcm_status_user_x32(substream, argp, false); + case SNDRV_PCM_IOCTL_STATUS_EXT_X32: + return snd_pcm_status_user_x32(substream, argp, true); + case SNDRV_PCM_IOCTL_SYNC_PTR_X32: + return snd_pcm_ioctl_sync_ptr_x32(substream, argp); + case SNDRV_PCM_IOCTL_CHANNEL_INFO_X32: + return snd_pcm_ioctl_channel_info_x32(substream, argp); +#endif /* CONFIG_X86_X32 */ } return -ENOIOCTLCMD; -- cgit v1.2.1 From dd7e3f805231cf83317c39bd346d5e4839f5cbb0 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sun, 28 Feb 2016 11:30:53 +0100 Subject: ALSA: rawmidi: Use comapt_put_timespec() Instead of open-coding, use the existing helper to copy a 32bit timespec from/to 64bit. Signed-off-by: Takashi Iwai --- sound/core/rawmidi_compat.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/core/rawmidi_compat.c b/sound/core/rawmidi_compat.c index 5268c1f58c25..836d1c9fc2cc 100644 --- a/sound/core/rawmidi_compat.c +++ b/sound/core/rawmidi_compat.c @@ -85,8 +85,7 @@ static int snd_rawmidi_ioctl_status_compat(struct snd_rawmidi_file *rfile, if (err < 0) return err; - if (put_user(status.tstamp.tv_sec, &src->tstamp.tv_sec) || - put_user(status.tstamp.tv_nsec, &src->tstamp.tv_nsec) || + if (compat_put_timespec(&status.tstamp, &src->tstamp) || put_user(status.avail, &src->avail) || put_user(status.xruns, &src->xruns)) return -EFAULT; -- cgit v1.2.1 From 2251fbbc1539f05b0b206b37a602d5776be37252 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sun, 28 Feb 2016 11:28:08 +0100 Subject: ALSA: rawmidi: Fix ioctls X32 ABI Like the previous fixes for ctl and PCM, we need a fix for incompatible X32 ABI regarding the rawmidi: namely, struct snd_rawmidi_status has the timespec, and the size and the alignment on X32 differ from IA32. This patch fixes the incompatible ioctl for X32. Cc: # v3.4+ Signed-off-by: Takashi Iwai --- sound/core/rawmidi_compat.c | 53 +++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 53 insertions(+) (limited to 'sound') diff --git a/sound/core/rawmidi_compat.c b/sound/core/rawmidi_compat.c index 836d1c9fc2cc..f69764d7cdd7 100644 --- a/sound/core/rawmidi_compat.c +++ b/sound/core/rawmidi_compat.c @@ -93,9 +93,58 @@ static int snd_rawmidi_ioctl_status_compat(struct snd_rawmidi_file *rfile, return 0; } +#ifdef CONFIG_X86_X32 +/* X32 ABI has 64bit timespec and 64bit alignment */ +struct snd_rawmidi_status_x32 { + s32 stream; + u32 rsvd; /* alignment */ + struct timespec tstamp; + u32 avail; + u32 xruns; + unsigned char reserved[16]; +} __attribute__((packed)); + +#define put_timespec(src, dst) copy_to_user(dst, src, sizeof(*dst)) + +static int snd_rawmidi_ioctl_status_x32(struct snd_rawmidi_file *rfile, + struct snd_rawmidi_status_x32 __user *src) +{ + int err; + struct snd_rawmidi_status status; + + if (rfile->output == NULL) + return -EINVAL; + if (get_user(status.stream, &src->stream)) + return -EFAULT; + + switch (status.stream) { + case SNDRV_RAWMIDI_STREAM_OUTPUT: + err = snd_rawmidi_output_status(rfile->output, &status); + break; + case SNDRV_RAWMIDI_STREAM_INPUT: + err = snd_rawmidi_input_status(rfile->input, &status); + break; + default: + return -EINVAL; + } + if (err < 0) + return err; + + if (put_timespec(&status.tstamp, &src->tstamp) || + put_user(status.avail, &src->avail) || + put_user(status.xruns, &src->xruns)) + return -EFAULT; + + return 0; +} +#endif /* CONFIG_X86_X32 */ + enum { SNDRV_RAWMIDI_IOCTL_PARAMS32 = _IOWR('W', 0x10, struct snd_rawmidi_params32), SNDRV_RAWMIDI_IOCTL_STATUS32 = _IOWR('W', 0x20, struct snd_rawmidi_status32), +#ifdef CONFIG_X86_X32 + SNDRV_RAWMIDI_IOCTL_STATUS_X32 = _IOWR('W', 0x20, struct snd_rawmidi_status_x32), +#endif /* CONFIG_X86_X32 */ }; static long snd_rawmidi_ioctl_compat(struct file *file, unsigned int cmd, unsigned long arg) @@ -114,6 +163,10 @@ static long snd_rawmidi_ioctl_compat(struct file *file, unsigned int cmd, unsign return snd_rawmidi_ioctl_params_compat(rfile, argp); case SNDRV_RAWMIDI_IOCTL_STATUS32: return snd_rawmidi_ioctl_status_compat(rfile, argp); +#ifdef CONFIG_X86_X32 + case SNDRV_RAWMIDI_IOCTL_STATUS_X32: + return snd_rawmidi_ioctl_status_x32(rfile, argp); +#endif /* CONFIG_X86_X32 */ } return -ENOIOCTLCMD; } -- cgit v1.2.1 From 3a72494ac2a3bd229db941d51e7efe2f6ccd947b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sun, 28 Feb 2016 11:36:14 +0100 Subject: ALSA: timer: Fix broken compat timer user status ioctl The timer user status compat ioctl returned the bogus struct used for 64bit architectures instead of the 32bit one. This patch addresses it to return the proper struct. Cc: Signed-off-by: Takashi Iwai --- sound/core/timer_compat.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/core/timer_compat.c b/sound/core/timer_compat.c index e05802ae6e1b..8e7eddf35c6a 100644 --- a/sound/core/timer_compat.c +++ b/sound/core/timer_compat.c @@ -70,13 +70,14 @@ static int snd_timer_user_status_compat(struct file *file, struct snd_timer_status32 __user *_status) { struct snd_timer_user *tu; - struct snd_timer_status status; + struct snd_timer_status32 status; tu = file->private_data; if (snd_BUG_ON(!tu->timeri)) return -ENXIO; memset(&status, 0, sizeof(status)); - status.tstamp = tu->tstamp; + status.tstamp.tv_sec = tu->tstamp.tv_sec; + status.tstamp.tv_nsec = tu->tstamp.tv_nsec; status.resolution = snd_timer_resolution(tu->timeri); status.lost = tu->timeri->lost; status.overrun = tu->overrun; -- cgit v1.2.1 From b24e7ad1fdc22177eb3e51584e1cfcb45d818488 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sun, 28 Feb 2016 11:41:47 +0100 Subject: ALSA: timer: Fix ioctls for X32 ABI X32 ABI takes the 64bit timespec, thus the timer user status ioctl becomes incompatible with IA32. This results in NOTTY error when the ioctl is issued. Meanwhile, this struct in X32 is essentially identical with the one in X86-64, so we can just bypassing to the existing code for this specific compat ioctl. Cc: # v3.4+ Signed-off-by: Takashi Iwai --- sound/core/timer_compat.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) (limited to 'sound') diff --git a/sound/core/timer_compat.c b/sound/core/timer_compat.c index 8e7eddf35c6a..2e908225d754 100644 --- a/sound/core/timer_compat.c +++ b/sound/core/timer_compat.c @@ -89,12 +89,21 @@ static int snd_timer_user_status_compat(struct file *file, return 0; } +#ifdef CONFIG_X86_X32 +/* X32 ABI has the same struct as x86-64 */ +#define snd_timer_user_status_x32(file, s) \ + snd_timer_user_status(file, s) +#endif /* CONFIG_X86_X32 */ + /* */ enum { SNDRV_TIMER_IOCTL_INFO32 = _IOR('T', 0x11, struct snd_timer_info32), SNDRV_TIMER_IOCTL_STATUS32 = _IOW('T', 0x14, struct snd_timer_status32), +#ifdef CONFIG_X86_X32 + SNDRV_TIMER_IOCTL_STATUS_X32 = _IOW('T', 0x14, struct snd_timer_status), +#endif /* CONFIG_X86_X32 */ }; static long snd_timer_user_ioctl_compat(struct file *file, unsigned int cmd, unsigned long arg) @@ -123,6 +132,10 @@ static long snd_timer_user_ioctl_compat(struct file *file, unsigned int cmd, uns return snd_timer_user_info_compat(file, argp); case SNDRV_TIMER_IOCTL_STATUS32: return snd_timer_user_status_compat(file, argp); +#ifdef CONFIG_X86_X32 + case SNDRV_TIMER_IOCTL_STATUS_X32: + return snd_timer_user_status_x32(file, argp); +#endif /* CONFIG_X86_X32 */ } return -ENOIOCTLCMD; } -- cgit v1.2.1 From 537e48136295c5860a92138c5ea3959b9542868b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 29 Feb 2016 14:25:16 +0100 Subject: ALSA: hdspm: Fix wrong boolean ctl value accesses snd-hdspm driver accesses enum item values (int) instead of boolean values (long) wrongly for some ctl elements. This patch fixes them. Cc: Signed-off-by: Takashi Iwai --- sound/pci/rme9652/hdspm.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 8bc8016c173d..b047b1ba48fd 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -2261,7 +2261,7 @@ static int snd_hdspm_put_system_sample_rate(struct snd_kcontrol *kcontrol, { struct hdspm *hdspm = snd_kcontrol_chip(kcontrol); - hdspm_set_dds_value(hdspm, ucontrol->value.enumerated.item[0]); + hdspm_set_dds_value(hdspm, ucontrol->value.integer.value[0]); return 0; } @@ -4449,7 +4449,7 @@ static int snd_hdspm_get_tco_word_term(struct snd_kcontrol *kcontrol, { struct hdspm *hdspm = snd_kcontrol_chip(kcontrol); - ucontrol->value.enumerated.item[0] = hdspm->tco->term; + ucontrol->value.integer.value[0] = hdspm->tco->term; return 0; } @@ -4460,8 +4460,8 @@ static int snd_hdspm_put_tco_word_term(struct snd_kcontrol *kcontrol, { struct hdspm *hdspm = snd_kcontrol_chip(kcontrol); - if (hdspm->tco->term != ucontrol->value.enumerated.item[0]) { - hdspm->tco->term = ucontrol->value.enumerated.item[0]; + if (hdspm->tco->term != ucontrol->value.integer.value[0]) { + hdspm->tco->term = ucontrol->value.integer.value[0]; hdspm_tco_write(hdspm); -- cgit v1.2.1 From c1099c3294c2344110085a38c50e478a5992b368 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 29 Feb 2016 14:32:42 +0100 Subject: ALSA: hdspm: Fix zero-division HDSPM driver contains a code issuing zero-division potentially in system sample rate ctl code. This patch fixes it by not processing a zero or invalid rate value as a divisor, as well as excluding the invalid value to be passed via the given ctl element. Cc: Signed-off-by: Takashi Iwai --- sound/pci/rme9652/hdspm.c | 8 ++++++++ 1 file changed, 8 insertions(+) (limited to 'sound') diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index b047b1ba48fd..a4a999a0317e 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -1601,6 +1601,9 @@ static void hdspm_set_dds_value(struct hdspm *hdspm, int rate) { u64 n; + if (snd_BUG_ON(rate <= 0)) + return; + if (rate >= 112000) rate /= 4; else if (rate >= 56000) @@ -2215,6 +2218,8 @@ static int hdspm_get_system_sample_rate(struct hdspm *hdspm) } else { /* slave mode, return external sample rate */ rate = hdspm_external_sample_rate(hdspm); + if (!rate) + rate = hdspm->system_sample_rate; } } @@ -2260,7 +2265,10 @@ static int snd_hdspm_put_system_sample_rate(struct snd_kcontrol *kcontrol, ucontrol) { struct hdspm *hdspm = snd_kcontrol_chip(kcontrol); + int rate = ucontrol->value.integer.value[0]; + if (rate < 27000 || rate > 207000) + return -EINVAL; hdspm_set_dds_value(hdspm, ucontrol->value.integer.value[0]); return 0; } -- cgit v1.2.1 From eab3c4db193f5fcccf70e884de9a922ca2c63d80 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 29 Feb 2016 14:26:43 +0100 Subject: ALSA: hdsp: Fix wrong boolean ctl value accesses snd-hdsp driver accesses enum item values (int) instead of boolean values (long) wrongly for some ctl elements. This patch fixes them. Cc: Signed-off-by: Takashi Iwai --- sound/pci/rme9652/hdsp.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index 2875b4f6d8c9..7c8941b8b2de 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -2879,7 +2879,7 @@ static int snd_hdsp_get_dds_offset(struct snd_kcontrol *kcontrol, struct snd_ctl { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - ucontrol->value.enumerated.item[0] = hdsp_dds_offset(hdsp); + ucontrol->value.integer.value[0] = hdsp_dds_offset(hdsp); return 0; } @@ -2891,7 +2891,7 @@ static int snd_hdsp_put_dds_offset(struct snd_kcontrol *kcontrol, struct snd_ctl if (!snd_hdsp_use_is_exclusive(hdsp)) return -EBUSY; - val = ucontrol->value.enumerated.item[0]; + val = ucontrol->value.integer.value[0]; spin_lock_irq(&hdsp->lock); if (val != hdsp_dds_offset(hdsp)) change = (hdsp_set_dds_offset(hdsp, val) == 0) ? 1 : 0; -- cgit v1.2.1 From 741338f99f16dc24d2d01ac777b0798ae9d10a90 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 29 Feb 2016 17:20:48 +0100 Subject: ASoC: dapm: Fix ctl value accesses in a wrong type snd_soc_dapm_dai_link_get() and _put() access the associated ctl values as value.integer.value[]. However, this is an enum ctl, and it has to be accessed via value.enumerated.item[]. The former is long while the latter is unsigned int, so they don't align. Fixes: c66150824b8a ('ASoC: dapm: add code to configure dai link parameters') Cc: Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 5a2812fa8946..335c7de2b046 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3573,7 +3573,7 @@ static int snd_soc_dapm_dai_link_get(struct snd_kcontrol *kcontrol, { struct snd_soc_dapm_widget *w = snd_kcontrol_chip(kcontrol); - ucontrol->value.integer.value[0] = w->params_select; + ucontrol->value.enumerated.item[0] = w->params_select; return 0; } @@ -3587,13 +3587,13 @@ static int snd_soc_dapm_dai_link_put(struct snd_kcontrol *kcontrol, if (w->power) return -EBUSY; - if (ucontrol->value.integer.value[0] == w->params_select) + if (ucontrol->value.enumerated.item[0] == w->params_select) return 0; - if (ucontrol->value.integer.value[0] >= w->num_params) + if (ucontrol->value.enumerated.item[0] >= w->num_params) return -EINVAL; - w->params_select = ucontrol->value.integer.value[0]; + w->params_select = ucontrol->value.enumerated.item[0]; return 0; } -- cgit v1.2.1 From 1457ad0e99dc5b1fe3fe3123d5135c95c00ed74b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 29 Feb 2016 17:23:49 +0100 Subject: ASoC: pxa: corgi: Fix enum ctl accesses in a wrong type "Jack Function" and "Speaker Function" ctls in corgi are enum, while the current driver accesses wrongly via value.integer.value[]. They have to be via value.enumerated.item[] instead. Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/pxa/corgi.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c index c97dc13d3608..dcbb7aa9830c 100644 --- a/sound/soc/pxa/corgi.c +++ b/sound/soc/pxa/corgi.c @@ -163,7 +163,7 @@ static struct snd_soc_ops corgi_ops = { static int corgi_get_jack(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - ucontrol->value.integer.value[0] = corgi_jack_func; + ucontrol->value.enumerated.item[0] = corgi_jack_func; return 0; } @@ -172,10 +172,10 @@ static int corgi_set_jack(struct snd_kcontrol *kcontrol, { struct snd_soc_card *card = snd_kcontrol_chip(kcontrol); - if (corgi_jack_func == ucontrol->value.integer.value[0]) + if (corgi_jack_func == ucontrol->value.enumerated.item[0]) return 0; - corgi_jack_func = ucontrol->value.integer.value[0]; + corgi_jack_func = ucontrol->value.enumerated.item[0]; corgi_ext_control(&card->dapm); return 1; } @@ -183,7 +183,7 @@ static int corgi_set_jack(struct snd_kcontrol *kcontrol, static int corgi_get_spk(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - ucontrol->value.integer.value[0] = corgi_spk_func; + ucontrol->value.enumerated.item[0] = corgi_spk_func; return 0; } @@ -192,10 +192,10 @@ static int corgi_set_spk(struct snd_kcontrol *kcontrol, { struct snd_soc_card *card = snd_kcontrol_chip(kcontrol); - if (corgi_spk_func == ucontrol->value.integer.value[0]) + if (corgi_spk_func == ucontrol->value.enumerated.item[0]) return 0; - corgi_spk_func = ucontrol->value.integer.value[0]; + corgi_spk_func = ucontrol->value.enumerated.item[0]; corgi_ext_control(&card->dapm); return 1; } -- cgit v1.2.1 From 127ee199d5d232f29ad64bace3b441127fce6cd3 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 29 Feb 2016 17:23:50 +0100 Subject: ASoC: pxa: magician: Fix enum ctl accesses in a wrong type "Input Select" ctl in magician driver is an enum, while the current driver accesses wrongly via value.integer.value[]. They have to be via value.enumerated.item[] instead. (Meanwhile "Headphone Switch" and "Speaker Switch" are boolean, so they should stay to access via value.integer.value[] as is.) Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/pxa/magician.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/pxa/magician.c b/sound/soc/pxa/magician.c index 241d0be42d7a..62b8377a9d2b 100644 --- a/sound/soc/pxa/magician.c +++ b/sound/soc/pxa/magician.c @@ -308,17 +308,17 @@ static int magician_set_spk(struct snd_kcontrol *kcontrol, static int magician_get_input(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - ucontrol->value.integer.value[0] = magician_in_sel; + ucontrol->value.enumerated.item[0] = magician_in_sel; return 0; } static int magician_set_input(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - if (magician_in_sel == ucontrol->value.integer.value[0]) + if (magician_in_sel == ucontrol->value.enumerated.item[0]) return 0; - magician_in_sel = ucontrol->value.integer.value[0]; + magician_in_sel = ucontrol->value.enumerated.item[0]; switch (magician_in_sel) { case MAGICIAN_MIC: -- cgit v1.2.1 From 7a3f4b488ce366de6ab2cf19d95353da9814bdd9 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 29 Feb 2016 17:23:51 +0100 Subject: ASoC: pxa: poodle: Fix enum ctl accesses in a wrong type "Jack Function" and "Speaker Function" ctls in poodle are enum, while the current driver accesses wrongly via value.integer.value[]. They have to be via value.enumerated.item[] instead. Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/pxa/poodle.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c index 84d0e2e50808..4b3b714f5ee7 100644 --- a/sound/soc/pxa/poodle.c +++ b/sound/soc/pxa/poodle.c @@ -138,7 +138,7 @@ static struct snd_soc_ops poodle_ops = { static int poodle_get_jack(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - ucontrol->value.integer.value[0] = poodle_jack_func; + ucontrol->value.enumerated.item[0] = poodle_jack_func; return 0; } @@ -147,10 +147,10 @@ static int poodle_set_jack(struct snd_kcontrol *kcontrol, { struct snd_soc_card *card = snd_kcontrol_chip(kcontrol); - if (poodle_jack_func == ucontrol->value.integer.value[0]) + if (poodle_jack_func == ucontrol->value.enumerated.item[0]) return 0; - poodle_jack_func = ucontrol->value.integer.value[0]; + poodle_jack_func = ucontrol->value.enumerated.item[0]; poodle_ext_control(&card->dapm); return 1; } @@ -158,7 +158,7 @@ static int poodle_set_jack(struct snd_kcontrol *kcontrol, static int poodle_get_spk(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - ucontrol->value.integer.value[0] = poodle_spk_func; + ucontrol->value.enumerated.item[0] = poodle_spk_func; return 0; } @@ -167,10 +167,10 @@ static int poodle_set_spk(struct snd_kcontrol *kcontrol, { struct snd_soc_card *card = snd_kcontrol_chip(kcontrol); - if (poodle_spk_func == ucontrol->value.integer.value[0]) + if (poodle_spk_func == ucontrol->value.enumerated.item[0]) return 0; - poodle_spk_func = ucontrol->value.integer.value[0]; + poodle_spk_func = ucontrol->value.enumerated.item[0]; poodle_ext_control(&card->dapm); return 1; } -- cgit v1.2.1 From 49a1ba16ab15b9a572c188fe0d9d79017d2fef99 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 29 Feb 2016 17:23:52 +0100 Subject: ASoC: pxa: spitz: Fix enum ctl accesses in a wrong type "Jack Function" and "Speaker Function" ctls in spitz are enum, while the current driver accesses wrongly via value.integer.value[]. They have to be via value.enumerated.item[] instead. Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/pxa/spitz.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c index b00222620fd0..0e02634c8b7f 100644 --- a/sound/soc/pxa/spitz.c +++ b/sound/soc/pxa/spitz.c @@ -164,7 +164,7 @@ static struct snd_soc_ops spitz_ops = { static int spitz_get_jack(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - ucontrol->value.integer.value[0] = spitz_jack_func; + ucontrol->value.enumerated.item[0] = spitz_jack_func; return 0; } @@ -173,10 +173,10 @@ static int spitz_set_jack(struct snd_kcontrol *kcontrol, { struct snd_soc_card *card = snd_kcontrol_chip(kcontrol); - if (spitz_jack_func == ucontrol->value.integer.value[0]) + if (spitz_jack_func == ucontrol->value.enumerated.item[0]) return 0; - spitz_jack_func = ucontrol->value.integer.value[0]; + spitz_jack_func = ucontrol->value.enumerated.item[0]; spitz_ext_control(&card->dapm); return 1; } @@ -184,7 +184,7 @@ static int spitz_set_jack(struct snd_kcontrol *kcontrol, static int spitz_get_spk(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - ucontrol->value.integer.value[0] = spitz_spk_func; + ucontrol->value.enumerated.item[0] = spitz_spk_func; return 0; } @@ -193,10 +193,10 @@ static int spitz_set_spk(struct snd_kcontrol *kcontrol, { struct snd_soc_card *card = snd_kcontrol_chip(kcontrol); - if (spitz_spk_func == ucontrol->value.integer.value[0]) + if (spitz_spk_func == ucontrol->value.enumerated.item[0]) return 0; - spitz_spk_func = ucontrol->value.integer.value[0]; + spitz_spk_func = ucontrol->value.enumerated.item[0]; spitz_ext_control(&card->dapm); return 1; } -- cgit v1.2.1 From 419396d5a1d12003a18a58785687697800d2f02a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 29 Feb 2016 17:23:53 +0100 Subject: ASoC: pxa: tosa: Fix enum ctl accesses in a wrong type "Jack Function" and "Speaker Function" ctls in tosa are enum, while the current driver accesses wrongly via value.integer.value[]. They have to be via value.enumerated.item[] instead. Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/pxa/tosa.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c index 49518dd642aa..c508f024ecfb 100644 --- a/sound/soc/pxa/tosa.c +++ b/sound/soc/pxa/tosa.c @@ -95,7 +95,7 @@ static struct snd_soc_ops tosa_ops = { static int tosa_get_jack(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - ucontrol->value.integer.value[0] = tosa_jack_func; + ucontrol->value.enumerated.item[0] = tosa_jack_func; return 0; } @@ -104,10 +104,10 @@ static int tosa_set_jack(struct snd_kcontrol *kcontrol, { struct snd_soc_card *card = snd_kcontrol_chip(kcontrol); - if (tosa_jack_func == ucontrol->value.integer.value[0]) + if (tosa_jack_func == ucontrol->value.enumerated.item[0]) return 0; - tosa_jack_func = ucontrol->value.integer.value[0]; + tosa_jack_func = ucontrol->value.enumerated.item[0]; tosa_ext_control(&card->dapm); return 1; } @@ -115,7 +115,7 @@ static int tosa_set_jack(struct snd_kcontrol *kcontrol, static int tosa_get_spk(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - ucontrol->value.integer.value[0] = tosa_spk_func; + ucontrol->value.enumerated.item[0] = tosa_spk_func; return 0; } @@ -124,10 +124,10 @@ static int tosa_set_spk(struct snd_kcontrol *kcontrol, { struct snd_soc_card *card = snd_kcontrol_chip(kcontrol); - if (tosa_spk_func == ucontrol->value.integer.value[0]) + if (tosa_spk_func == ucontrol->value.enumerated.item[0]) return 0; - tosa_spk_func = ucontrol->value.integer.value[0]; + tosa_spk_func = ucontrol->value.enumerated.item[0]; tosa_ext_control(&card->dapm); return 1; } -- cgit v1.2.1 From dd90533cd6bbfe075a72ab789b6e9d6f280ba476 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 29 Feb 2016 17:26:21 +0100 Subject: ASoC: omap: n810: Fix enum ctl accesses in a wrong type "Jack Function", "Speaker Function" and "Input Select" ctls in n810 driver are enum, while the current driver accesses wrongly via value.integer.value[]. They have to be via value.enumerated.item[] instead. Signed-off-by: Takashi Iwai Acked-by: Jarkko Nikula Signed-off-by: Mark Brown --- sound/soc/omap/n810.c | 18 +++++++++--------- 1 file changed, 9 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c index 190f868e78b2..fdecb7043174 100644 --- a/sound/soc/omap/n810.c +++ b/sound/soc/omap/n810.c @@ -133,7 +133,7 @@ static struct snd_soc_ops n810_ops = { static int n810_get_spk(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - ucontrol->value.integer.value[0] = n810_spk_func; + ucontrol->value.enumerated.item[0] = n810_spk_func; return 0; } @@ -143,10 +143,10 @@ static int n810_set_spk(struct snd_kcontrol *kcontrol, { struct snd_soc_card *card = snd_kcontrol_chip(kcontrol); - if (n810_spk_func == ucontrol->value.integer.value[0]) + if (n810_spk_func == ucontrol->value.enumerated.item[0]) return 0; - n810_spk_func = ucontrol->value.integer.value[0]; + n810_spk_func = ucontrol->value.enumerated.item[0]; n810_ext_control(&card->dapm); return 1; @@ -155,7 +155,7 @@ static int n810_set_spk(struct snd_kcontrol *kcontrol, static int n810_get_jack(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - ucontrol->value.integer.value[0] = n810_jack_func; + ucontrol->value.enumerated.item[0] = n810_jack_func; return 0; } @@ -165,10 +165,10 @@ static int n810_set_jack(struct snd_kcontrol *kcontrol, { struct snd_soc_card *card = snd_kcontrol_chip(kcontrol); - if (n810_jack_func == ucontrol->value.integer.value[0]) + if (n810_jack_func == ucontrol->value.enumerated.item[0]) return 0; - n810_jack_func = ucontrol->value.integer.value[0]; + n810_jack_func = ucontrol->value.enumerated.item[0]; n810_ext_control(&card->dapm); return 1; @@ -177,7 +177,7 @@ static int n810_set_jack(struct snd_kcontrol *kcontrol, static int n810_get_input(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - ucontrol->value.integer.value[0] = n810_dmic_func; + ucontrol->value.enumerated.item[0] = n810_dmic_func; return 0; } @@ -187,10 +187,10 @@ static int n810_set_input(struct snd_kcontrol *kcontrol, { struct snd_soc_card *card = snd_kcontrol_chip(kcontrol); - if (n810_dmic_func == ucontrol->value.integer.value[0]) + if (n810_dmic_func == ucontrol->value.enumerated.item[0]) return 0; - n810_dmic_func = ucontrol->value.integer.value[0]; + n810_dmic_func = ucontrol->value.enumerated.item[0]; n810_ext_control(&card->dapm); return 1; -- cgit v1.2.1 From f4d438eec7fe56d55998195e08dfaa5aa3e08f0c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 29 Feb 2016 17:26:22 +0100 Subject: ASoC: omap: rx51: Fix enum ctl accesses in a wrong type "Speaker Function", "Input Select" and "Jack Function" ctls in rx51 driver are enum, while the current driver accesses wrongly via value.integer.value[]. They have to be via value.enumerated.item[] instead. Signed-off-by: Takashi Iwai Acked-by: Jarkko Nikula Signed-off-by: Mark Brown --- sound/soc/omap/rx51.c | 18 +++++++++--------- 1 file changed, 9 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/rx51.c b/sound/soc/omap/rx51.c index 5e21f08579d8..54949242bc70 100644 --- a/sound/soc/omap/rx51.c +++ b/sound/soc/omap/rx51.c @@ -132,7 +132,7 @@ static struct snd_soc_ops rx51_ops = { static int rx51_get_spk(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - ucontrol->value.integer.value[0] = rx51_spk_func; + ucontrol->value.enumerated.item[0] = rx51_spk_func; return 0; } @@ -142,10 +142,10 @@ static int rx51_set_spk(struct snd_kcontrol *kcontrol, { struct snd_soc_card *card = snd_kcontrol_chip(kcontrol); - if (rx51_spk_func == ucontrol->value.integer.value[0]) + if (rx51_spk_func == ucontrol->value.enumerated.item[0]) return 0; - rx51_spk_func = ucontrol->value.integer.value[0]; + rx51_spk_func = ucontrol->value.enumerated.item[0]; rx51_ext_control(&card->dapm); return 1; @@ -180,7 +180,7 @@ static int rx51_hp_event(struct snd_soc_dapm_widget *w, static int rx51_get_input(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - ucontrol->value.integer.value[0] = rx51_dmic_func; + ucontrol->value.enumerated.item[0] = rx51_dmic_func; return 0; } @@ -190,10 +190,10 @@ static int rx51_set_input(struct snd_kcontrol *kcontrol, { struct snd_soc_card *card = snd_kcontrol_chip(kcontrol); - if (rx51_dmic_func == ucontrol->value.integer.value[0]) + if (rx51_dmic_func == ucontrol->value.enumerated.item[0]) return 0; - rx51_dmic_func = ucontrol->value.integer.value[0]; + rx51_dmic_func = ucontrol->value.enumerated.item[0]; rx51_ext_control(&card->dapm); return 1; @@ -202,7 +202,7 @@ static int rx51_set_input(struct snd_kcontrol *kcontrol, static int rx51_get_jack(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - ucontrol->value.integer.value[0] = rx51_jack_func; + ucontrol->value.enumerated.item[0] = rx51_jack_func; return 0; } @@ -212,10 +212,10 @@ static int rx51_set_jack(struct snd_kcontrol *kcontrol, { struct snd_soc_card *card = snd_kcontrol_chip(kcontrol); - if (rx51_jack_func == ucontrol->value.integer.value[0]) + if (rx51_jack_func == ucontrol->value.enumerated.item[0]) return 0; - rx51_jack_func = ucontrol->value.integer.value[0]; + rx51_jack_func = ucontrol->value.enumerated.item[0]; rx51_ext_control(&card->dapm); return 1; -- cgit v1.2.1 From 508ddfba37cad76d774da8f8d8e2387f0eca5fa4 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 29 Feb 2016 17:29:25 +0100 Subject: ASoC: intel: mfld: Fix enum ctl accesses in a wrong type "Playback Switch" and "Lineout Mux" ctls in medfld machine driver are enum, while the current driver accesses wrongly via value.integer.value[]. They have to be via value.enumerated.item[] instead. Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/intel/boards/mfld_machine.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/boards/mfld_machine.c b/sound/soc/intel/boards/mfld_machine.c index 49c09a0add79..34f46c72a0e2 100644 --- a/sound/soc/intel/boards/mfld_machine.c +++ b/sound/soc/intel/boards/mfld_machine.c @@ -94,7 +94,7 @@ static const struct soc_enum lo_enum = static int headset_get_switch(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - ucontrol->value.integer.value[0] = hs_switch; + ucontrol->value.enumerated.item[0] = hs_switch; return 0; } @@ -104,12 +104,12 @@ static int headset_set_switch(struct snd_kcontrol *kcontrol, struct snd_soc_card *card = snd_kcontrol_chip(kcontrol); struct snd_soc_dapm_context *dapm = &card->dapm; - if (ucontrol->value.integer.value[0] == hs_switch) + if (ucontrol->value.enumerated.item[0] == hs_switch) return 0; snd_soc_dapm_mutex_lock(dapm); - if (ucontrol->value.integer.value[0]) { + if (ucontrol->value.enumerated.item[0]) { pr_debug("hs_set HS path\n"); snd_soc_dapm_enable_pin_unlocked(dapm, "Headphones"); snd_soc_dapm_disable_pin_unlocked(dapm, "EPOUT"); @@ -123,7 +123,7 @@ static int headset_set_switch(struct snd_kcontrol *kcontrol, snd_soc_dapm_mutex_unlock(dapm); - hs_switch = ucontrol->value.integer.value[0]; + hs_switch = ucontrol->value.enumerated.item[0]; return 0; } @@ -148,7 +148,7 @@ static void lo_enable_out_pins(struct snd_soc_dapm_context *dapm) static int lo_get_switch(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - ucontrol->value.integer.value[0] = lo_dac; + ucontrol->value.enumerated.item[0] = lo_dac; return 0; } @@ -158,7 +158,7 @@ static int lo_set_switch(struct snd_kcontrol *kcontrol, struct snd_soc_card *card = snd_kcontrol_chip(kcontrol); struct snd_soc_dapm_context *dapm = &card->dapm; - if (ucontrol->value.integer.value[0] == lo_dac) + if (ucontrol->value.enumerated.item[0] == lo_dac) return 0; snd_soc_dapm_mutex_lock(dapm); @@ -168,7 +168,7 @@ static int lo_set_switch(struct snd_kcontrol *kcontrol, */ lo_enable_out_pins(dapm); - switch (ucontrol->value.integer.value[0]) { + switch (ucontrol->value.enumerated.item[0]) { case 0: pr_debug("set vibra path\n"); snd_soc_dapm_disable_pin_unlocked(dapm, "VIB1OUT"); @@ -202,7 +202,7 @@ static int lo_set_switch(struct snd_kcontrol *kcontrol, snd_soc_dapm_mutex_unlock(dapm); - lo_dac = ucontrol->value.integer.value[0]; + lo_dac = ucontrol->value.enumerated.item[0]; return 0; } -- cgit v1.2.1 From 89300b4e5a5b5b5c145318f3b257291239522da6 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 29 Feb 2016 18:02:59 +0100 Subject: ASoC: cs42l51: Fix enum ctl accesses in a wrong type "PCM channel mixer" ctl in cs42l51 codec driver is enum, while the current driver accesses wrongly via value.integer.value[]. They have to be via value.enumerated.item[] instead. Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l51.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index b3951524339f..35488f14e237 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -60,15 +60,15 @@ static int cs42l51_get_chan_mix(struct snd_kcontrol *kcontrol, switch (value) { default: case 0: - ucontrol->value.integer.value[0] = 0; + ucontrol->value.enumerated.item[0] = 0; break; /* same value : (L+R)/2 and (R+L)/2 */ case 1: case 2: - ucontrol->value.integer.value[0] = 1; + ucontrol->value.enumerated.item[0] = 1; break; case 3: - ucontrol->value.integer.value[0] = 2; + ucontrol->value.enumerated.item[0] = 2; break; } @@ -85,7 +85,7 @@ static int cs42l51_set_chan_mix(struct snd_kcontrol *kcontrol, struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); unsigned char val; - switch (ucontrol->value.integer.value[0]) { + switch (ucontrol->value.enumerated.item[0]) { default: case 0: val = CHAN_MIX_NORMAL; -- cgit v1.2.1 From fe9aba13108ee8e15ae59009a26db02460dbb04f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 29 Feb 2016 18:04:42 +0100 Subject: ASoC: da732x: Fix enum ctl accesses in a wrong type "DAC1 High Pass Filter Mode" & co in da732x codec driver are enum, while the current driver accesses wrongly via value.integer.value[]. They have to be via value.enumerated.item[] instead. Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/codecs/da732x.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/da732x.c b/sound/soc/codecs/da732x.c index 1d5a89c5164b..461506a4ca6a 100644 --- a/sound/soc/codecs/da732x.c +++ b/sound/soc/codecs/da732x.c @@ -334,7 +334,7 @@ static int da732x_hpf_set(struct snd_kcontrol *kcontrol, struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); struct soc_enum *enum_ctrl = (struct soc_enum *)kcontrol->private_value; unsigned int reg = enum_ctrl->reg; - unsigned int sel = ucontrol->value.integer.value[0]; + unsigned int sel = ucontrol->value.enumerated.item[0]; unsigned int bits; switch (sel) { @@ -368,13 +368,13 @@ static int da732x_hpf_get(struct snd_kcontrol *kcontrol, switch (val) { case DA732X_HPF_VOICE_EN: - ucontrol->value.integer.value[0] = DA732X_HPF_VOICE; + ucontrol->value.enumerated.item[0] = DA732X_HPF_VOICE; break; case DA732X_HPF_MUSIC_EN: - ucontrol->value.integer.value[0] = DA732X_HPF_MUSIC; + ucontrol->value.enumerated.item[0] = DA732X_HPF_MUSIC; break; default: - ucontrol->value.integer.value[0] = DA732X_HPF_DISABLED; + ucontrol->value.enumerated.item[0] = DA732X_HPF_DISABLED; break; } -- cgit v1.2.1 From 4b606316129571c46381430852c149855ac50477 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 29 Feb 2016 18:08:00 +0100 Subject: ASoC: ab8500: Fix enum ctl accesses in a wrong type "Sidetone Status" and "ANC Status" ctls in ab8500 codec driver are enum, while the current driver accesses wrongly via value.integer.value[]. They have to be via value.enumerated.item[] instead. Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/codecs/ab8500-codec.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/ab8500-codec.c b/sound/soc/codecs/ab8500-codec.c index affb192238a4..faae6936bae4 100644 --- a/sound/soc/codecs/ab8500-codec.c +++ b/sound/soc/codecs/ab8500-codec.c @@ -1130,7 +1130,7 @@ static int sid_status_control_get(struct snd_kcontrol *kcontrol, struct ab8500_codec_drvdata *drvdata = dev_get_drvdata(codec->dev); mutex_lock(&drvdata->ctrl_lock); - ucontrol->value.integer.value[0] = drvdata->sid_status; + ucontrol->value.enumerated.item[0] = drvdata->sid_status; mutex_unlock(&drvdata->ctrl_lock); return 0; @@ -1147,7 +1147,7 @@ static int sid_status_control_put(struct snd_kcontrol *kcontrol, dev_dbg(codec->dev, "%s: Enter\n", __func__); - if (ucontrol->value.integer.value[0] != SID_APPLY_FIR) { + if (ucontrol->value.enumerated.item[0] != SID_APPLY_FIR) { dev_err(codec->dev, "%s: ERROR: This control supports '%s' only!\n", __func__, enum_sid_state[SID_APPLY_FIR]); @@ -1199,7 +1199,7 @@ static int anc_status_control_get(struct snd_kcontrol *kcontrol, struct ab8500_codec_drvdata *drvdata = dev_get_drvdata(codec->dev); mutex_lock(&drvdata->ctrl_lock); - ucontrol->value.integer.value[0] = drvdata->anc_status; + ucontrol->value.enumerated.item[0] = drvdata->anc_status; mutex_unlock(&drvdata->ctrl_lock); return 0; @@ -1220,7 +1220,7 @@ static int anc_status_control_put(struct snd_kcontrol *kcontrol, mutex_lock(&drvdata->ctrl_lock); - req = ucontrol->value.integer.value[0]; + req = ucontrol->value.enumerated.item[0]; if (req >= ARRAY_SIZE(enum_anc_state)) { status = -EINVAL; goto cleanup; -- cgit v1.2.1 From 9af39044753d7280fa795528598636fe9c58a54e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 29 Feb 2016 18:08:01 +0100 Subject: ASoC: max98088: Fix enum ctl accesses in a wrong type "EQ1 Mode" and "EQ2 Mode" ctls in max98088 codec driver are enum, while the current driver accesses wrongly via value.integer.value[]. They have to be via value.enumerated.item[] instead. Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/codecs/max98088.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index 20dcc496d39c..fc22804cabc5 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -1496,7 +1496,7 @@ static int max98088_put_eq_enum(struct snd_kcontrol *kcontrol, struct max98088_pdata *pdata = max98088->pdata; int channel = max98088_get_channel(codec, kcontrol->id.name); struct max98088_cdata *cdata; - int sel = ucontrol->value.integer.value[0]; + int sel = ucontrol->value.enumerated.item[0]; if (channel < 0) return channel; -- cgit v1.2.1 From 58c0213872816bf24d14364989c82dc4acd58103 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 29 Feb 2016 18:08:02 +0100 Subject: ASoC: max98095: Fix enum ctl accesses in a wrong type "Biquad1 Mode" and "Biquad2 Mode" ctls in max98095 codec driver are enum, while the current driver accesses wrongly via value.integer.value[]. They have to be via value.enumerated.item[] instead. Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/codecs/max98095.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/max98095.c b/sound/soc/codecs/max98095.c index 1fedac50355e..3577003f39cf 100644 --- a/sound/soc/codecs/max98095.c +++ b/sound/soc/codecs/max98095.c @@ -1499,7 +1499,7 @@ static int max98095_put_eq_enum(struct snd_kcontrol *kcontrol, struct max98095_pdata *pdata = max98095->pdata; int channel = max98095_get_eq_channel(kcontrol->id.name); struct max98095_cdata *cdata; - unsigned int sel = ucontrol->value.integer.value[0]; + unsigned int sel = ucontrol->value.enumerated.item[0]; struct max98095_eq_cfg *coef_set; int fs, best, best_val, i; int regmask, regsave; @@ -1653,7 +1653,7 @@ static int max98095_put_bq_enum(struct snd_kcontrol *kcontrol, struct max98095_pdata *pdata = max98095->pdata; int channel = max98095_get_bq_channel(codec, kcontrol->id.name); struct max98095_cdata *cdata; - unsigned int sel = ucontrol->value.integer.value[0]; + unsigned int sel = ucontrol->value.enumerated.item[0]; struct max98095_biquad_cfg *coef_set; int fs, best, best_val, i; int regmask, regsave; -- cgit v1.2.1 From 8733f99c23726532918034f4ae599d9e6d27bd1e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 29 Feb 2016 18:08:03 +0100 Subject: ASoC: tlv320dac33: Fix enum ctl accesses in a wrong type "FIFO Mode" ctl in tlv320dac33 codec driver is enum, while the current driver accesses wrongly via value.integer.value[]. They have to be via value.enumerated.item[] instead. Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320dac33.c | 9 ++++----- 1 file changed, 4 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index 781398fb2841..f7a6ce7e5fb1 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -446,7 +446,7 @@ static int dac33_get_fifo_mode(struct snd_kcontrol *kcontrol, struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec); - ucontrol->value.integer.value[0] = dac33->fifo_mode; + ucontrol->value.enumerated.item[0] = dac33->fifo_mode; return 0; } @@ -458,17 +458,16 @@ static int dac33_set_fifo_mode(struct snd_kcontrol *kcontrol, struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec); int ret = 0; - if (dac33->fifo_mode == ucontrol->value.integer.value[0]) + if (dac33->fifo_mode == ucontrol->value.enumerated.item[0]) return 0; /* Do not allow changes while stream is running*/ if (snd_soc_codec_is_active(codec)) return -EPERM; - if (ucontrol->value.integer.value[0] < 0 || - ucontrol->value.integer.value[0] >= DAC33_FIFO_LAST_MODE) + if (ucontrol->value.enumerated.item[0] >= DAC33_FIFO_LAST_MODE) ret = -EINVAL; else - dac33->fifo_mode = ucontrol->value.integer.value[0]; + dac33->fifo_mode = ucontrol->value.enumerated.item[0]; return ret; } -- cgit v1.2.1 From 154a0fddd4275aa93f4683705579a7d8ec3d177b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 29 Feb 2016 18:08:04 +0100 Subject: ASoC: wl1273: Fix enum ctl accesses in a wrong type "Codec Mode" and "Audio Switch" ctls in wl1273 codec driver are enum, while the current driver accesses wrongly via value.integer.value[]. They have to be via value.enumerated.item[] instead. Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/codecs/wl1273.c | 13 ++++++------- 1 file changed, 6 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wl1273.c b/sound/soc/codecs/wl1273.c index 7693c1129bab..1b79778098d2 100644 --- a/sound/soc/codecs/wl1273.c +++ b/sound/soc/codecs/wl1273.c @@ -175,7 +175,7 @@ static int snd_wl1273_get_audio_route(struct snd_kcontrol *kcontrol, struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); struct wl1273_priv *wl1273 = snd_soc_codec_get_drvdata(codec); - ucontrol->value.integer.value[0] = wl1273->mode; + ucontrol->value.enumerated.item[0] = wl1273->mode; return 0; } @@ -193,18 +193,17 @@ static int snd_wl1273_set_audio_route(struct snd_kcontrol *kcontrol, struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); struct wl1273_priv *wl1273 = snd_soc_codec_get_drvdata(codec); - if (wl1273->mode == ucontrol->value.integer.value[0]) + if (wl1273->mode == ucontrol->value.enumerated.item[0]) return 0; /* Do not allow changes while stream is running */ if (snd_soc_codec_is_active(codec)) return -EPERM; - if (ucontrol->value.integer.value[0] < 0 || - ucontrol->value.integer.value[0] >= ARRAY_SIZE(wl1273_audio_route)) + if (ucontrol->value.enumerated.item[0] >= ARRAY_SIZE(wl1273_audio_route)) return -EINVAL; - wl1273->mode = ucontrol->value.integer.value[0]; + wl1273->mode = ucontrol->value.enumerated.item[0]; return 1; } @@ -219,7 +218,7 @@ static int snd_wl1273_fm_audio_get(struct snd_kcontrol *kcontrol, dev_dbg(codec->dev, "%s: enter.\n", __func__); - ucontrol->value.integer.value[0] = wl1273->core->audio_mode; + ucontrol->value.enumerated.item[0] = wl1273->core->audio_mode; return 0; } @@ -233,7 +232,7 @@ static int snd_wl1273_fm_audio_put(struct snd_kcontrol *kcontrol, dev_dbg(codec->dev, "%s: enter.\n", __func__); - val = ucontrol->value.integer.value[0]; + val = ucontrol->value.enumerated.item[0]; if (wl1273->core->audio_mode == val) return 0; -- cgit v1.2.1 From 0cad10539b281e88335afd765afaf9885dcfc3ef Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 29 Feb 2016 18:01:10 +0100 Subject: ASoC: wm8753: Fix enum ctl accesses in a wrong type "DAI Mode" ctl in wm8753 codec driver is enum, while the current driver accesses wrongly via value.integer.value[]. They have to be via value.enumerated.item[] instead. Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/codecs/wm8753.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index 61299ca372ff..6f1024f48b19 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -233,7 +233,7 @@ static int wm8753_get_dai(struct snd_kcontrol *kcontrol, struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); struct wm8753_priv *wm8753 = snd_soc_codec_get_drvdata(codec); - ucontrol->value.integer.value[0] = wm8753->dai_func; + ucontrol->value.enumerated.item[0] = wm8753->dai_func; return 0; } @@ -244,7 +244,7 @@ static int wm8753_set_dai(struct snd_kcontrol *kcontrol, struct wm8753_priv *wm8753 = snd_soc_codec_get_drvdata(codec); u16 ioctl; - if (wm8753->dai_func == ucontrol->value.integer.value[0]) + if (wm8753->dai_func == ucontrol->value.enumerated.item[0]) return 0; if (snd_soc_codec_is_active(codec)) @@ -252,7 +252,7 @@ static int wm8753_set_dai(struct snd_kcontrol *kcontrol, ioctl = snd_soc_read(codec, WM8753_IOCTL); - wm8753->dai_func = ucontrol->value.integer.value[0]; + wm8753->dai_func = ucontrol->value.enumerated.item[0]; if (((ioctl >> 2) & 0x3) == wm8753->dai_func) return 1; -- cgit v1.2.1 From c41a024c4e770fff999f4164cc4d1696e5f17437 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 29 Feb 2016 18:01:11 +0100 Subject: ASoC: wm8904: Fix enum ctl accesses in a wrong type "DRC Mode" and "EQ Mode" ctls in wm8904 codec driver are enum, while the current driver accesses wrongly via value.integer.value[]. They have to be via value.enumerated.item[] instead. Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/codecs/wm8904.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 8172e499e6ed..edd7a7709194 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -396,7 +396,7 @@ static int wm8904_put_drc_enum(struct snd_kcontrol *kcontrol, struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); struct wm8904_priv *wm8904 = snd_soc_codec_get_drvdata(codec); struct wm8904_pdata *pdata = wm8904->pdata; - int value = ucontrol->value.integer.value[0]; + int value = ucontrol->value.enumerated.item[0]; if (value >= pdata->num_drc_cfgs) return -EINVAL; @@ -467,7 +467,7 @@ static int wm8904_put_retune_mobile_enum(struct snd_kcontrol *kcontrol, struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); struct wm8904_priv *wm8904 = snd_soc_codec_get_drvdata(codec); struct wm8904_pdata *pdata = wm8904->pdata; - int value = ucontrol->value.integer.value[0]; + int value = ucontrol->value.enumerated.item[0]; if (value >= pdata->num_retune_mobile_cfgs) return -EINVAL; -- cgit v1.2.1 From d0784829ae3b0beeb69b476f017d5c8a2eb95198 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 29 Feb 2016 18:01:12 +0100 Subject: ASoC: wm8958: Fix enum ctl accesses in a wrong type "MBC Mode", "VSS Mode", "VSS HPF Mode" and "Enhanced EQ Mode" ctls in wm8958 codec driver are enum, while the current driver accesses wrongly via value.integer.value[]. They have to be via value.enumerated.item[] instead. Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/wm8958-dsp2.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8958-dsp2.c b/sound/soc/codecs/wm8958-dsp2.c index c799cca5abeb..6b864c0fc2b6 100644 --- a/sound/soc/codecs/wm8958-dsp2.c +++ b/sound/soc/codecs/wm8958-dsp2.c @@ -459,7 +459,7 @@ static int wm8958_put_mbc_enum(struct snd_kcontrol *kcontrol, struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); struct wm8994 *control = wm8994->wm8994; - int value = ucontrol->value.integer.value[0]; + int value = ucontrol->value.enumerated.item[0]; int reg; /* Don't allow on the fly reconfiguration */ @@ -549,7 +549,7 @@ static int wm8958_put_vss_enum(struct snd_kcontrol *kcontrol, struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); struct wm8994 *control = wm8994->wm8994; - int value = ucontrol->value.integer.value[0]; + int value = ucontrol->value.enumerated.item[0]; int reg; /* Don't allow on the fly reconfiguration */ @@ -582,7 +582,7 @@ static int wm8958_put_vss_hpf_enum(struct snd_kcontrol *kcontrol, struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); struct wm8994 *control = wm8994->wm8994; - int value = ucontrol->value.integer.value[0]; + int value = ucontrol->value.enumerated.item[0]; int reg; /* Don't allow on the fly reconfiguration */ @@ -749,7 +749,7 @@ static int wm8958_put_enh_eq_enum(struct snd_kcontrol *kcontrol, struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); struct wm8994 *control = wm8994->wm8994; - int value = ucontrol->value.integer.value[0]; + int value = ucontrol->value.enumerated.item[0]; int reg; /* Don't allow on the fly reconfiguration */ -- cgit v1.2.1 From b5ab265905b3e07ad9dc7d553a074404b25e9200 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 29 Feb 2016 18:01:13 +0100 Subject: ASoC: wm8983: Fix enum ctl accesses in a wrong type "Equalizer Function" ctl in wm8983 codec driver is enum, while the current driver accesses wrongly via value.integer.value[]. They have to be via value.enumerated.item[] instead. Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/codecs/wm8983.c | 14 +++++++------- 1 file changed, 7 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8983.c b/sound/soc/codecs/wm8983.c index 7350ff654bbf..0c002a5712cb 100644 --- a/sound/soc/codecs/wm8983.c +++ b/sound/soc/codecs/wm8983.c @@ -497,9 +497,9 @@ static int eqmode_get(struct snd_kcontrol *kcontrol, reg = snd_soc_read(codec, WM8983_EQ1_LOW_SHELF); if (reg & WM8983_EQ3DMODE) - ucontrol->value.integer.value[0] = 1; + ucontrol->value.enumerated.item[0] = 1; else - ucontrol->value.integer.value[0] = 0; + ucontrol->value.enumerated.item[0] = 0; return 0; } @@ -511,18 +511,18 @@ static int eqmode_put(struct snd_kcontrol *kcontrol, unsigned int regpwr2, regpwr3; unsigned int reg_eq; - if (ucontrol->value.integer.value[0] != 0 - && ucontrol->value.integer.value[0] != 1) + if (ucontrol->value.enumerated.item[0] != 0 + && ucontrol->value.enumerated.item[0] != 1) return -EINVAL; reg_eq = snd_soc_read(codec, WM8983_EQ1_LOW_SHELF); switch ((reg_eq & WM8983_EQ3DMODE) >> WM8983_EQ3DMODE_SHIFT) { case 0: - if (!ucontrol->value.integer.value[0]) + if (!ucontrol->value.enumerated.item[0]) return 0; break; case 1: - if (ucontrol->value.integer.value[0]) + if (ucontrol->value.enumerated.item[0]) return 0; break; } @@ -537,7 +537,7 @@ static int eqmode_put(struct snd_kcontrol *kcontrol, /* set the desired eqmode */ snd_soc_update_bits(codec, WM8983_EQ1_LOW_SHELF, WM8983_EQ3DMODE_MASK, - ucontrol->value.integer.value[0] + ucontrol->value.enumerated.item[0] << WM8983_EQ3DMODE_SHIFT); /* restore DAC/ADC configuration */ snd_soc_write(codec, WM8983_POWER_MANAGEMENT_2, regpwr2); -- cgit v1.2.1 From 251d604778f1f7bbf29672a79cccc4663a7efd62 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 29 Feb 2016 18:01:14 +0100 Subject: ASoC: wm8985: Fix enum ctl accesses in a wrong type "Equalizer Function" ctl in wm8985 codec driver is enum, while the current driver accesses wrongly via value.integer.value[]. They have to be via value.enumerated.item[] instead. Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/codecs/wm8985.c | 14 +++++++------- 1 file changed, 7 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8985.c b/sound/soc/codecs/wm8985.c index 9918152a03c7..6ac76fe116b0 100644 --- a/sound/soc/codecs/wm8985.c +++ b/sound/soc/codecs/wm8985.c @@ -531,9 +531,9 @@ static int eqmode_get(struct snd_kcontrol *kcontrol, reg = snd_soc_read(codec, WM8985_EQ1_LOW_SHELF); if (reg & WM8985_EQ3DMODE) - ucontrol->value.integer.value[0] = 1; + ucontrol->value.enumerated.item[0] = 1; else - ucontrol->value.integer.value[0] = 0; + ucontrol->value.enumerated.item[0] = 0; return 0; } @@ -545,18 +545,18 @@ static int eqmode_put(struct snd_kcontrol *kcontrol, unsigned int regpwr2, regpwr3; unsigned int reg_eq; - if (ucontrol->value.integer.value[0] != 0 - && ucontrol->value.integer.value[0] != 1) + if (ucontrol->value.enumerated.item[0] != 0 + && ucontrol->value.enumerated.item[0] != 1) return -EINVAL; reg_eq = snd_soc_read(codec, WM8985_EQ1_LOW_SHELF); switch ((reg_eq & WM8985_EQ3DMODE) >> WM8985_EQ3DMODE_SHIFT) { case 0: - if (!ucontrol->value.integer.value[0]) + if (!ucontrol->value.enumerated.item[0]) return 0; break; case 1: - if (ucontrol->value.integer.value[0]) + if (ucontrol->value.enumerated.item[0]) return 0; break; } @@ -573,7 +573,7 @@ static int eqmode_put(struct snd_kcontrol *kcontrol, /* set the desired eqmode */ snd_soc_update_bits(codec, WM8985_EQ1_LOW_SHELF, WM8985_EQ3DMODE_MASK, - ucontrol->value.integer.value[0] + ucontrol->value.enumerated.item[0] << WM8985_EQ3DMODE_SHIFT); /* restore DAC/ADC configuration */ snd_soc_write(codec, WM8985_POWER_MANAGEMENT_2, regpwr2); -- cgit v1.2.1 From 8019c0b37cd5a87107808300a496388b777225bf Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 29 Feb 2016 18:01:15 +0100 Subject: ASoC: wm8994: Fix enum ctl accesses in a wrong type The DRC Mode like "AIF1DRC1 Mode" and EQ Mode like "AIF1.1 EQ Mode" in wm8994 codec driver are enum ctls, while the current driver accesses wrongly via value.integer.value[]. They have to be via value.enumerated.item[] instead. Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/wm8994.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 2ccbb322df77..a18aecb49935 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -362,7 +362,7 @@ static int wm8994_put_drc_enum(struct snd_kcontrol *kcontrol, struct wm8994 *control = wm8994->wm8994; struct wm8994_pdata *pdata = &control->pdata; int drc = wm8994_get_drc(kcontrol->id.name); - int value = ucontrol->value.integer.value[0]; + int value = ucontrol->value.enumerated.item[0]; if (drc < 0) return drc; @@ -469,7 +469,7 @@ static int wm8994_put_retune_mobile_enum(struct snd_kcontrol *kcontrol, struct wm8994 *control = wm8994->wm8994; struct wm8994_pdata *pdata = &control->pdata; int block = wm8994_get_retune_mobile_block(kcontrol->id.name); - int value = ucontrol->value.integer.value[0]; + int value = ucontrol->value.enumerated.item[0]; if (block < 0) return block; -- cgit v1.2.1 From 8293004c81dd33e6e91afe2f9a773fe6796893cc Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 29 Feb 2016 18:01:16 +0100 Subject: ASoC: wm8996: Fix enum ctl accesses in a wrong type "DSP1 EQ Mode" and "DSP2 EQ Mode" ctls in wm8996 codec driver are enum, while the current driver accesses wrongly via value.integer.value[]. They have to be via value.enumerated.item[] instead. Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/codecs/wm8996.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index 8d7d6c01a2f7..f99b34f7647b 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -416,7 +416,7 @@ static int wm8996_put_retune_mobile_enum(struct snd_kcontrol *kcontrol, struct wm8996_priv *wm8996 = snd_soc_codec_get_drvdata(codec); struct wm8996_pdata *pdata = &wm8996->pdata; int block = wm8996_get_retune_mobile_block(kcontrol->id.name); - int value = ucontrol->value.integer.value[0]; + int value = ucontrol->value.enumerated.item[0]; if (block < 0) return block; -- cgit v1.2.1 From 39a79fe10482572ce76fd724b7915b3ee4cbd81e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 29 Feb 2016 18:01:17 +0100 Subject: ASoC: wm9081: Fix enum ctl accesses in a wrong type "Speaker Mode "ctl in wm9081 codec driver is enum, while the current driver accesses wrongly via value.integer.value[]. They have to be via value.enumerated.item[] instead. Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/codecs/wm9081.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index ccb3b15139ad..363b3b667616 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -344,9 +344,9 @@ static int speaker_mode_get(struct snd_kcontrol *kcontrol, reg = snd_soc_read(codec, WM9081_ANALOGUE_SPEAKER_2); if (reg & WM9081_SPK_MODE) - ucontrol->value.integer.value[0] = 1; + ucontrol->value.enumerated.item[0] = 1; else - ucontrol->value.integer.value[0] = 0; + ucontrol->value.enumerated.item[0] = 0; return 0; } @@ -365,7 +365,7 @@ static int speaker_mode_put(struct snd_kcontrol *kcontrol, unsigned int reg2 = snd_soc_read(codec, WM9081_ANALOGUE_SPEAKER_2); /* Are we changing anything? */ - if (ucontrol->value.integer.value[0] == + if (ucontrol->value.enumerated.item[0] == ((reg2 & WM9081_SPK_MODE) != 0)) return 0; @@ -373,7 +373,7 @@ static int speaker_mode_put(struct snd_kcontrol *kcontrol, if (reg_pwr & WM9081_SPK_ENA) return -EINVAL; - if (ucontrol->value.integer.value[0]) { + if (ucontrol->value.enumerated.item[0]) { /* Class AB */ reg2 &= ~(WM9081_SPK_INV_MUTE | WM9081_OUT_SPK_CTRL); reg2 |= WM9081_SPK_MODE; -- cgit v1.2.1 From 15c665700bf6f4543f003ac0fbb1e9ec692e93f2 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 29 Feb 2016 18:01:18 +0100 Subject: ASoC: wm_adsp: Fix enum ctl accesses in a wrong type The firmware ctls like "DSP1 Firmware" in wm_adsp codec driver are enum, while the current driver accesses wrongly via value.integer.value[]. They have to be via value.enumerated.item[] instead. Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/wm_adsp.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 33806d487b8a..b9195b9c2b05 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -586,7 +586,7 @@ static int wm_adsp_fw_get(struct snd_kcontrol *kcontrol, struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; struct wm_adsp *dsp = snd_soc_codec_get_drvdata(codec); - ucontrol->value.integer.value[0] = dsp[e->shift_l].fw; + ucontrol->value.enumerated.item[0] = dsp[e->shift_l].fw; return 0; } @@ -599,10 +599,10 @@ static int wm_adsp_fw_put(struct snd_kcontrol *kcontrol, struct wm_adsp *dsp = snd_soc_codec_get_drvdata(codec); int ret = 0; - if (ucontrol->value.integer.value[0] == dsp[e->shift_l].fw) + if (ucontrol->value.enumerated.item[0] == dsp[e->shift_l].fw) return 0; - if (ucontrol->value.integer.value[0] >= WM_ADSP_NUM_FW) + if (ucontrol->value.enumerated.item[0] >= WM_ADSP_NUM_FW) return -EINVAL; mutex_lock(&dsp[e->shift_l].pwr_lock); @@ -610,7 +610,7 @@ static int wm_adsp_fw_put(struct snd_kcontrol *kcontrol, if (dsp[e->shift_l].running || dsp[e->shift_l].compr) ret = -EBUSY; else - dsp[e->shift_l].fw = ucontrol->value.integer.value[0]; + dsp[e->shift_l].fw = ucontrol->value.enumerated.item[0]; mutex_unlock(&dsp[e->shift_l].pwr_lock); -- cgit v1.2.1 From 17e2df4613be57d0fab68df749f6b8114e453152 Mon Sep 17 00:00:00 2001 From: Dennis Kadioglu Date: Tue, 1 Mar 2016 14:23:29 +0100 Subject: ALSA: usb-audio: Add a quirk for Plantronics DA45 Plantronics DA45 does not support reading the sample rate which leads to many lines of "cannot get freq at ep 0x4" and "cannot get freq at ep 0x84". This patch adds the USB ID of the DA45 to quirks.c and avoids those error messages. Signed-off-by: Dennis Kadioglu Cc: Signed-off-by: Takashi Iwai --- sound/usb/quirks.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 4f6ce1cac8e2..c458d60d5030 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1124,6 +1124,7 @@ bool snd_usb_get_sample_rate_quirk(struct snd_usb_audio *chip) case USB_ID(0x045E, 0x076F): /* MS Lifecam HD-6000 */ case USB_ID(0x045E, 0x0772): /* MS Lifecam Studio */ case USB_ID(0x045E, 0x0779): /* MS Lifecam HD-3000 */ + case USB_ID(0x047F, 0xAA05): /* Plantronics DA45 */ case USB_ID(0x04D8, 0xFEEA): /* Benchmark DAC1 Pre */ case USB_ID(0x074D, 0x3553): /* Outlaw RR2150 (Micronas UAC3553B) */ case USB_ID(0x21B4, 0x0081): /* AudioQuest DragonFly */ -- cgit v1.2.1 From 197b958c1e76a575d77038cc98b4bebc2134279f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 1 Mar 2016 18:30:18 +0100 Subject: ALSA: seq: oss: Don't drain at closing a client The OSS sequencer client tries to drain the pending events at releasing. Unfortunately, as spotted by syzkaller fuzzer, this may lead to an unkillable process state when the event has been queued at the far future. Since the process being released can't be signaled any longer, it remains and waits for the echo-back event in that far future. Back to history, the draining feature was implemented at the time we misinterpreted POSIX definition for blocking file operation. Actually, such a behavior is superfluous at release, and we should just release the device as is instead of keeping it up forever. This patch just removes the draining call that may block the release for too long time unexpectedly. BugLink: http://lkml.kernel.org/r/CACT4Y+Y4kD-aBGj37rf-xBw9bH3GMU6P+MYg4W1e-s-paVD2pg@mail.gmail.com Reported-by: Dmitry Vyukov Cc: Signed-off-by: Takashi Iwai --- sound/core/seq/oss/seq_oss.c | 2 -- sound/core/seq/oss/seq_oss_device.h | 1 - sound/core/seq/oss/seq_oss_init.c | 16 ---------------- 3 files changed, 19 deletions(-) (limited to 'sound') diff --git a/sound/core/seq/oss/seq_oss.c b/sound/core/seq/oss/seq_oss.c index 8db156b207f1..8cdf489df80e 100644 --- a/sound/core/seq/oss/seq_oss.c +++ b/sound/core/seq/oss/seq_oss.c @@ -149,8 +149,6 @@ odev_release(struct inode *inode, struct file *file) if ((dp = file->private_data) == NULL) return 0; - snd_seq_oss_drain_write(dp); - mutex_lock(®ister_mutex); snd_seq_oss_release(dp); mutex_unlock(®ister_mutex); diff --git a/sound/core/seq/oss/seq_oss_device.h b/sound/core/seq/oss/seq_oss_device.h index b43924325249..d7b4d016b547 100644 --- a/sound/core/seq/oss/seq_oss_device.h +++ b/sound/core/seq/oss/seq_oss_device.h @@ -127,7 +127,6 @@ int snd_seq_oss_write(struct seq_oss_devinfo *dp, const char __user *buf, int co unsigned int snd_seq_oss_poll(struct seq_oss_devinfo *dp, struct file *file, poll_table * wait); void snd_seq_oss_reset(struct seq_oss_devinfo *dp); -void snd_seq_oss_drain_write(struct seq_oss_devinfo *dp); /* */ void snd_seq_oss_process_queue(struct seq_oss_devinfo *dp, abstime_t time); diff --git a/sound/core/seq/oss/seq_oss_init.c b/sound/core/seq/oss/seq_oss_init.c index 6779e82b46dd..92c96a95a903 100644 --- a/sound/core/seq/oss/seq_oss_init.c +++ b/sound/core/seq/oss/seq_oss_init.c @@ -435,22 +435,6 @@ snd_seq_oss_release(struct seq_oss_devinfo *dp) } -/* - * Wait until the queue is empty (if we don't have nonblock) - */ -void -snd_seq_oss_drain_write(struct seq_oss_devinfo *dp) -{ - if (! dp->timer->running) - return; - if (is_write_mode(dp->file_mode) && !is_nonblock_mode(dp->file_mode) && - dp->writeq) { - while (snd_seq_oss_writeq_sync(dp->writeq)) - ; - } -} - - /* * reset sequencer devices */ -- cgit v1.2.1 From 02322ac9dee9aff8d8862e8d6660ebe102f492ea Mon Sep 17 00:00:00 2001 From: Simon South Date: Wed, 2 Mar 2016 23:10:44 -0500 Subject: ALSA: hda - Fix mic issues on Acer Aspire E1-472 This patch applies the microphone-related fix created for the Acer Aspire E1-572 to the E1-472 as well, as it uses the same Realtek ALC282 CODEC and demonstrates the same issues. This patch allows an external, headset microphone to be used and limits the gain on the (quite noisy) internal microphone. Signed-off-by: Simon South Cc: 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 1f357cd72d9c..93d2156b6241 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5412,6 +5412,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x080d, "Acer Aspire V5-122P", ALC269_FIXUP_ASPIRE_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x0740, "Acer AO725", ALC271_FIXUP_HP_GATE_MIC_JACK), SND_PCI_QUIRK(0x1025, 0x0742, "Acer AO756", ALC271_FIXUP_HP_GATE_MIC_JACK), + SND_PCI_QUIRK(0x1025, 0x0762, "Acer Aspire E1-472", ALC271_FIXUP_HP_GATE_MIC_JACK_E1_572), SND_PCI_QUIRK(0x1025, 0x0775, "Acer Aspire E1-572", ALC271_FIXUP_HP_GATE_MIC_JACK_E1_572), SND_PCI_QUIRK(0x1025, 0x079b, "Acer Aspire V5-573G", ALC282_FIXUP_ASPIRE_V5_PINS), SND_PCI_QUIRK(0x1025, 0x106d, "Acer Cloudbook 14", ALC283_FIXUP_CHROME_BOOK), -- cgit v1.2.1 From ec75a940b1037e877efd9a5a9e94eab1e464f73b Mon Sep 17 00:00:00 2001 From: Libin Yang Date: Fri, 4 Mar 2016 14:33:06 +0800 Subject: ALSA: hda - hdmi add wmb barrier for audio component To make sure audio_ptr is set before intel_audio_codec_enable() or intel_audio_codec_disable() calling pin_eld_notify(), this patch adds wmb barrier to prevent optimizing. Signed-off-by: Libin Yang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_hdmi.c | 5 +++++ 1 file changed, 5 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 8ee78dbd4c60..6858e88c7326 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2480,6 +2480,11 @@ static int patch_generic_hdmi(struct hda_codec *codec) if (codec_has_acomp(codec)) { codec->depop_delay = 0; spec->i915_audio_ops.audio_ptr = codec; + /* intel_audio_codec_enable() or intel_audio_codec_disable() + * will call pin_eld_notify with using audio_ptr pointer + * We need make sure audio_ptr is really setup + */ + wmb(); spec->i915_audio_ops.pin_eld_notify = intel_pin_eld_notify; snd_hdac_i915_register_notifier(&spec->i915_audio_ops); } -- cgit v1.2.1 From 790b415c98de62602810b0eedce26f0f9d6ddd78 Mon Sep 17 00:00:00 2001 From: Libin Yang Date: Fri, 4 Mar 2016 14:33:43 +0800 Subject: ALSA: hda - hdmi defer to register acomp eld notifier Defer to register acomp eld notifier until hdmi audio driver is fully ready. After registering eld notifier, gfx driver can use this callback function to notify audio driver the monitor connection event. However this action may happen when audio driver is adding the pins or doing other initialization. This is not always safe, however. For example, using per_pin->lock before the lock is initialized. Let's register the eld notifier after the initialization is done. Signed-off-by: Libin Yang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_hdmi.c | 24 ++++++++++++------------ 1 file changed, 12 insertions(+), 12 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 6858e88c7326..bcbc4ee10130 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2477,18 +2477,6 @@ static int patch_generic_hdmi(struct hda_codec *codec) is_broxton(codec)) codec->core.link_power_control = 1; - if (codec_has_acomp(codec)) { - codec->depop_delay = 0; - spec->i915_audio_ops.audio_ptr = codec; - /* intel_audio_codec_enable() or intel_audio_codec_disable() - * will call pin_eld_notify with using audio_ptr pointer - * We need make sure audio_ptr is really setup - */ - wmb(); - spec->i915_audio_ops.pin_eld_notify = intel_pin_eld_notify; - snd_hdac_i915_register_notifier(&spec->i915_audio_ops); - } - if (hdmi_parse_codec(codec) < 0) { if (spec->i915_bound) snd_hdac_i915_exit(&codec->bus->core); @@ -2510,6 +2498,18 @@ static int patch_generic_hdmi(struct hda_codec *codec) init_channel_allocations(); + if (codec_has_acomp(codec)) { + codec->depop_delay = 0; + spec->i915_audio_ops.audio_ptr = codec; + /* intel_audio_codec_enable() or intel_audio_codec_disable() + * will call pin_eld_notify with using audio_ptr pointer + * We need make sure audio_ptr is really setup + */ + wmb(); + spec->i915_audio_ops.pin_eld_notify = intel_pin_eld_notify; + snd_hdac_i915_register_notifier(&spec->i915_audio_ops); + } + return 0; } -- cgit v1.2.1