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(-) 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 v1.2.3 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(+) 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 v1.2.3 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(-) 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 v1.2.3 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(-) 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 v1.2.3 From 2935bf43ef12a8d68b96776ec11155cfa120cb0d Mon Sep 17 00:00:00 2001 From: Xiubo Li Date: Fri, 15 Jan 2016 13:33:08 +0800 Subject: ASoC: fsl: document DT compatible string "fsl,imx-audio-wm8960" The devicetree compatible string "fsl,imx-audio-wm8960" for fsl-asoc-card is missing. Signed-off-by: Xiubo Li Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/fsl-asoc-card.txt | 2 ++ 1 file changed, 2 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt b/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt index ce55c0a6f757..4da41bf1888e 100644 --- a/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt +++ b/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt @@ -30,6 +30,8 @@ The compatible list for this generic sound card currently: "fsl,imx-audio-sgtl5000" (compatible with Documentation/devicetree/bindings/sound/imx-audio-sgtl5000.txt) + "fsl,imx-audio-wm8960" + Required properties: - compatible : Contains one of entries in the compatible list. -- cgit v1.2.3 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(-) 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 v1.2.3 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(-) 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 v1.2.3 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(-) 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 v1.2.3 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(+) 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 v1.2.3 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(+) 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 v1.2.3 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(-) 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 v1.2.3 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(-) 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 v1.2.3 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(-) 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 v1.2.3 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(-) 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 v1.2.3 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(-) 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 v1.2.3 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(-) 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 v1.2.3 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(-) 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 v1.2.3 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(-) 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 v1.2.3 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(+) 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 v1.2.3 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(-) 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 v1.2.3 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(+) 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 v1.2.3 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(+) 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 v1.2.3 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(-) 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 v1.2.3 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(-) 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 v1.2.3 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(-) 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 v1.2.3 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(-) 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 v1.2.3 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(-) 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 v1.2.3 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(-) 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 v1.2.3 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(+) 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 v1.2.3 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(-) 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 v1.2.3 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(-) 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 v1.2.3 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(-) 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 v1.2.3 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(+) 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 v1.2.3 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(-) 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 v1.2.3 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(-) 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 v1.2.3 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(-) 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 v1.2.3 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(-) 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 v1.2.3 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(+) 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 v1.2.3 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(-) 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 v1.2.3 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(-) 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 v1.2.3 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(-) 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 v1.2.3 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(-) 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 v1.2.3 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(+) 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 v1.2.3 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(-) 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 v1.2.3 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(+) 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 v1.2.3 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(-) 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 v1.2.3 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(-) 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 v1.2.3 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(-) 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 v1.2.3 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(-) 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 v1.2.3 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(-) 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 v1.2.3