From f5f76ea75dce553631ffb08abc44dcecb68e74d4 Mon Sep 17 00:00:00 2001
From: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Date: Mon, 11 Jan 2016 15:17:23 +0000
Subject: [PATCH 01/47] 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 <srinivas.kandagatla@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
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 79688aa1941a5..4aeb8e1a7160b 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;
From cde6bcd584b1b910d6ee8d6eb968ea5d20815444 Mon Sep 17 00:00:00 2001
From: Dan Carpenter <dan.carpenter@oracle.com>
Date: Wed, 13 Jan 2016 15:20:02 +0300
Subject: [PATCH 02/47] 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 <dan.carpenter@oracle.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
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 3191e0a7d2732..d1fb035f44db8 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;
}
From 1ca2cf8c4167c2016d9716998b4f89c4e79d1f89 Mon Sep 17 00:00:00 2001
From: Axel Lin <axel.lin@ingics.com>
Date: Tue, 12 Jan 2016 15:55:17 +0800
Subject: [PATCH 03/47] 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 <axel.lin@ingics.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
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 820d8fa62b5e5..c166d9394c691 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 8f07ee903eaad..d31c9e5bcec8a 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;
From ec3995da27e782cc407ce48101c98c19c9ce738d Mon Sep 17 00:00:00 2001
From: Arnd Bergmann <arnd@arndb.de>
Date: Wed, 13 Jan 2016 23:14:54 +0100
Subject: [PATCH 04/47] 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 <arnd@arndb.de>
Acked-by: Koro Chen <koro.chen@mediatek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
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 15c04e2eae34a..9769676753878 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
From 2935bf43ef12a8d68b96776ec11155cfa120cb0d Mon Sep 17 00:00:00 2001
From: Xiubo Li <lixiubo@cmss.chinamobile.com>
Date: Fri, 15 Jan 2016 13:33:08 +0800
Subject: [PATCH 05/47] 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 <lixiubo@cmss.chinamobile.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
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 ce55c0a6f7578..4da41bf1888e9 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.
From 6d514c720219a4c0e1c2612c1d830592bfaf5a03 Mon Sep 17 00:00:00 2001
From: Bard Liao <bardliao@realtek.com>
Date: Thu, 21 Jan 2016 14:10:48 +0800
Subject: [PATCH 06/47] 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 <bardliao@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
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 af2ed774b5529..af30b062f57a8 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:
From b28785fa9cede0d4f47310ca0dd2a4e1d50478b5 Mon Sep 17 00:00:00 2001
From: Bard Liao <bardliao@realtek.com>
Date: Thu, 21 Jan 2016 13:13:40 +0800
Subject: [PATCH 07/47] ASoC: rt5645: fix the shift bit of IN1 boost
The shift bit of IN1 boost gain control is 12.
Signed-off-by: Bard Liao <bardliao@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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 28132375e4274..c916c38812596 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),
From 2256b8d2ff6c8e994161ab15b6e6d0314d3174ae Mon Sep 17 00:00:00 2001
From: Arnd Bergmann <arnd@arndb.de>
Date: Wed, 20 Jan 2016 12:46:24 +0100
Subject: [PATCH 08/47] 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 <arnd@arndb.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
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 c166d9394c691..fb8ea05c0de1d 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,
From c14a82c781f8df50c4c5215ab92affdc60d72c01 Mon Sep 17 00:00:00 2001
From: Sudip Mukherjee <sudipm.mukherjee@gmail.com>
Date: Thu, 21 Jan 2016 17:27:59 +0530
Subject: [PATCH 09/47] 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 <sudip@vectorindia.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
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 5315b7422b98b..c7816d52ad085 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;
}
From f5ede8dcc3ec1fe5344f0d30717931a44e630631 Mon Sep 17 00:00:00 2001
From: Richard Fitzgerald <rf@opensource.wolfsonmicro.com>
Date: Thu, 21 Jan 2016 14:41:12 +0000
Subject: [PATCH 10/47] 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 <rf@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
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 c364096018358..cd1b3080a4974 100644
--- a/sound/soc/codecs/wm5110.c
+++ b/sound/soc/codecs/wm5110.c
@@ -2358,6 +2358,7 @@ static int wm5110_probe(struct platform_device *pdev)
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);
From 95826a37991de87659e21b3649f265a049724aa2 Mon Sep 17 00:00:00 2001
From: Stuart Henderson <stuart.henderson@cirrus.com>
Date: Tue, 19 Jan 2016 13:09:08 +0000
Subject: [PATCH 11/47] 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 <stuart.henderson@cirrus.com>
Reviewed-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
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 5380798883b5d..66057f853fae5 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),
From 6bb7451429084cefcb3a18fff809f7992595d2af Mon Sep 17 00:00:00 2001
From: Stuart Henderson <stuart.henderson@cirrus.com>
Date: Tue, 19 Jan 2016 13:09:09 +0000
Subject: [PATCH 12/47] 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 <stuart.henderson@cirrus.com>
Reviewed-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
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 66057f853fae5..4b44013295917 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) {
From 5c408fee254633a5be69505bc86c6b034f871ab4 Mon Sep 17 00:00:00 2001
From: "Maciej S. Szmigiero" <mail@maciej.szmigiero.name>
Date: Mon, 18 Jan 2016 20:07:44 +0100
Subject: [PATCH 13/47] 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 <mail@maciej.szmigiero.name>
Tested-by: Fabio Estevam <fabio.estevam@nxp.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
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 40dfd8a364840..ed8de1035cda1 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");
From 9954859185c6e8359e71121037b627f1e294057d Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Tue, 26 Jan 2016 13:54:15 +0100
Subject: [PATCH 14/47] 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 <fabio.estevam@nxp.com>
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
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 a407e833c6125..fb896b2c9ba32 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);
From f212c6d8c2b21c1e1d0158d38a7c37f4427f3848 Mon Sep 17 00:00:00 2001
From: Mans Rullgard <mans@mansr.com>
Date: Thu, 21 Jan 2016 14:55:56 +0000
Subject: [PATCH 15/47] 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 <mans@mansr.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
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 c866ade28ad0a..a6c7b8d87cd2f 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,
From ee43a1a0cd2a8f33cddfa1323a60b5cfcf865ba0 Mon Sep 17 00:00:00 2001
From: Aaro Koskinen <aaro.koskinen@iki.fi>
Date: Sun, 24 Jan 2016 00:36:40 +0200
Subject: [PATCH 16/47] 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 <aaro.koskinen@iki.fi>
Reviewed-by: Andrew Lunn <andrew@lunn.ch>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
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 1ded8811598ef..2389ab47e25f6 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;
}
From d2f916aaccaf7b3bc27df2fd6cfc00f6cda2f78d Mon Sep 17 00:00:00 2001
From: "Jon Medhurst (Tixy)" <tixy@linaro.org>
Date: Mon, 1 Feb 2016 15:54:37 +0000
Subject: [PATCH 17/47] 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 <tixy@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
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 ce664c239be32..bff258d7bcea1 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 {
From 5e82d2be6ee53275c72e964507518d7964c82753 Mon Sep 17 00:00:00 2001
From: Vinod Koul <vinod.koul@intel.com>
Date: Mon, 1 Feb 2016 22:26:40 +0530
Subject: [PATCH 18/47] 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 <vinod.koul@intel.com>
Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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 e898b427be7ee..1af4f23697a78 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",
From 292d4200a90715ac29f3763df27adb38a243868c Mon Sep 17 00:00:00 2001
From: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Date: Tue, 2 Feb 2016 12:49:49 -0600
Subject: [PATCH 19/47] 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 <alsa-devel@tbspace.de>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
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 55c33dc76ce44..52ed434cbca6a 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 */
From 41d80025a83b9c7a94f97ef25c4cd3345bdc3c5e Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Wed, 3 Feb 2016 21:59:50 +0100
Subject: [PATCH 20/47] 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 <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
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 5a2812fa89460..0d37079879002 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;
From 41556f68d1dd0b6bbf311a220523b034d2a040e7 Mon Sep 17 00:00:00 2001
From: Vinod Koul <vinod.koul@intel.com>
Date: Wed, 3 Feb 2016 17:59:44 +0530
Subject: [PATCH 21/47] 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 <tiwai@suse.de>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
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 c7816d52ad085..c67e3acb81022 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;
From ee564d489cc47b1b6043bbe7e95464306d112cf5 Mon Sep 17 00:00:00 2001
From: Guneshwor Singh <guneshwor.o.singh@intel.com>
Date: Wed, 3 Feb 2016 17:59:45 +0530
Subject: [PATCH 22/47] 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 <guneshwor.o.singh@intel.com>
Signed-off-by: Dharageswari.R <dharageswari.r@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
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 b89ae6f7c0961..f9297dc4b25f2 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,
From 7ca42f5ac5e0d8011086bcfa00e85aede42f0b78 Mon Sep 17 00:00:00 2001
From: Guneshwor Singh <guneshwor.o.singh@intel.com>
Date: Wed, 3 Feb 2016 17:59:46 +0530
Subject: [PATCH 23/47] 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 <guneshwor.o.singh@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
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 c67e3acb81022..86d5323e91846 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;
}
From 9ba8ffef9635c11102bc42d0f2d0a4213de273d5 Mon Sep 17 00:00:00 2001
From: "Dharageswari.R" <dharageswari.r@intel.com>
Date: Wed, 3 Feb 2016 17:59:47 +0530
Subject: [PATCH 24/47] 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 <dharageswari.r@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
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 86d5323e91846..efe001162204c 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;
}
From 9cf3049e21e4e6873aae45df19c11f7243e2f03f Mon Sep 17 00:00:00 2001
From: Jeeja KP <jeeja.kp@intel.com>
Date: Wed, 3 Feb 2016 17:59:48 +0530
Subject: [PATCH 25/47] 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 <jeeja.kp@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
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 de6dac496a0d8..bb5f1d7d0cada 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;
From 0c684c48257bc6033bdd3b942babef22d0a1852a Mon Sep 17 00:00:00 2001
From: Jeeja KP <jeeja.kp@intel.com>
Date: Wed, 3 Feb 2016 17:59:49 +0530
Subject: [PATCH 26/47] 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 <jeeja.kp@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
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 bb5f1d7d0cada..4629372d7c8e0 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;
From 9946f70906eebf2a305d0b189de52eec8ba39649 Mon Sep 17 00:00:00 2001
From: Jeeja KP <jeeja.kp@intel.com>
Date: Wed, 3 Feb 2016 17:59:50 +0530
Subject: [PATCH 27/47] 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 <jeeja.kp@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
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 efe001162204c..a356f3b1dd5ba 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);
}
From 6bd4cf855698312133b7776c77ee78af865608eb Mon Sep 17 00:00:00 2001
From: Jeeja KP <jeeja.kp@intel.com>
Date: Wed, 3 Feb 2016 17:59:51 +0530
Subject: [PATCH 28/47] 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 <jeeja.kp@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
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 a356f3b1dd5ba..77a688d00fc6e 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;
From de1fedf25b075664320010789ede2a0f9f4de07d Mon Sep 17 00:00:00 2001
From: Jeeja KP <jeeja.kp@intel.com>
Date: Wed, 3 Feb 2016 17:59:52 +0530
Subject: [PATCH 29/47] 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 <jeeja.kp@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
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 77a688d00fc6e..489848637df5b 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);
}
From 6e3ffa00424e198d2f0c628e7575c5adefeda3d7 Mon Sep 17 00:00:00 2001
From: Jeeja KP <jeeja.kp@intel.com>
Date: Wed, 3 Feb 2016 17:59:53 +0530
Subject: [PATCH 30/47] 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 <jeeja.kp@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
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 7396ddb427d8f..2cbcbe4126611 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;
}
From 38c079e230f25969e7ce3501fa967b003a2abc39 Mon Sep 17 00:00:00 2001
From: Vinod Koul <vinod.koul@intel.com>
Date: Wed, 3 Feb 2016 17:59:54 +0530
Subject: [PATCH 31/47] 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 <subhransu.s.prusty@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
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 c38bf99ced103..1d36b28d64895 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);
From 902c136fe4f72dfc2a616ad755c72f1ee407f79a Mon Sep 17 00:00:00 2001
From: Vinod Koul <vinod.koul@intel.com>
Date: Mon, 8 Feb 2016 10:45:36 +0530
Subject: [PATCH 32/47] 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 <yang.jie@intel.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
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 668fdeee195e2..3b9332e7a094c 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
From 2dcffcee23a2bd491a8c4041db3a8041b23fa4eb Mon Sep 17 00:00:00 2001
From: Vinod Koul <vinod.koul@intel.com>
Date: Mon, 8 Feb 2016 10:45:37 +0530
Subject: [PATCH 33/47] 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 <yang.jie@intel.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
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 803f95e40679d..af7aabbc0977a 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 3b9332e7a094c..fbbb25c2ceed2 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
From cfffcc66a89ab6d9961b2cde6cdab2ba056451ad Mon Sep 17 00:00:00 2001
From: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Date: Mon, 8 Feb 2016 10:45:38 +0530
Subject: [PATCH 34/47] 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 <pierre-louis.bossart@linux.intel.com>
Acked-by: Jie Yang <yang.jie@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
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 af7aabbc0977a..7d7c872c280db 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 7a85c576dad33..2c5eda14d5107 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);
From 8ceffd229f0ef130530c79654e95b5fa007ae639 Mon Sep 17 00:00:00 2001
From: Vinod Koul <vinod.koul@intel.com>
Date: Mon, 8 Feb 2016 10:45:39 +0530
Subject: [PATCH 35/47] 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 <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
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 dd077e116d259..3b4539d214924 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");
From 61c4a1ac4d900e743af0b363fe520405939eab47 Mon Sep 17 00:00:00 2001
From: Pascal Huerst <pascal.huerst@gmail.com>
Date: Wed, 10 Feb 2016 15:59:28 +0100
Subject: [PATCH 36/47] 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 <pascal.huerst@gmail.com>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
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 21ca3a5e9f660..d374c18d4db7f 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,
From 01582a841493f28caf1688b2af4dafbcbee8135e Mon Sep 17 00:00:00 2001
From: Richard Fitzgerald <rf@opensource.wolfsonmicro.com>
Date: Wed, 10 Feb 2016 11:56:13 +0000
Subject: [PATCH 37/47] 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 <rf@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
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 33143fe1de0bd..91785318b2834 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");
From 07d86ca93db7e5cdf4743564d98292042ec21af7 Mon Sep 17 00:00:00 2001
From: Andrey Konovalov <andreyknvl@gmail.com>
Date: Sat, 13 Feb 2016 11:08:06 +0300
Subject: [PATCH 38/47] ALSA: usb-audio: avoid freeing umidi object twice
The 'umidi' object will be free'd on the error path by snd_usbmidi_free()
when tearing down the rawmidi interface. So we shouldn't try to free it
in snd_usbmidi_create() after having registered the rawmidi interface.
Found by KASAN.
Signed-off-by: Andrey Konovalov <andreyknvl@gmail.com>
Acked-by: Clemens Ladisch <clemens@ladisch.de>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/usb/midi.c | 1 -
1 file changed, 1 deletion(-)
diff --git a/sound/usb/midi.c b/sound/usb/midi.c
index cc39f63299ef0..007cf58311215 100644
--- a/sound/usb/midi.c
+++ b/sound/usb/midi.c
@@ -2455,7 +2455,6 @@ int snd_usbmidi_create(struct snd_card *card,
else
err = snd_usbmidi_create_endpoints(umidi, endpoints);
if (err < 0) {
- snd_usbmidi_free(umidi);
return err;
}
From d99a36f4728fcbcc501b78447f625bdcce15b842 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 15 Feb 2016 16:20:24 +0100
Subject: [PATCH 39/47] ALSA: seq: Fix leak of pool buffer at concurrent writes
When multiple concurrent writes happen on the ALSA sequencer device
right after the open, it may try to allocate vmalloc buffer for each
write and leak some of them. It's because the presence check and the
assignment of the buffer is done outside the spinlock for the pool.
The fix is to move the check and the assignment into the spinlock.
(The current implementation is suboptimal, as there can be multiple
unnecessary vmallocs because the allocation is done before the check
in the spinlock. But the pool size is already checked beforehand, so
this isn't a big problem; that is, the only possible path is the
multiple writes before any pool assignment, and practically seen, the
current coverage should be "good enough".)
The issue was triggered by syzkaller fuzzer.
BugLink: http://lkml.kernel.org/r/CACT4Y+bSzazpXNvtAr=WXaL8hptqjHwqEyFA+VN2AWEx=aurkg@mail.gmail.com
Reported-by: Dmitry Vyukov <dvyukov@google.com>
Tested-by: Dmitry Vyukov <dvyukov@google.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/core/seq/seq_memory.c | 13 +++++++++----
1 file changed, 9 insertions(+), 4 deletions(-)
diff --git a/sound/core/seq/seq_memory.c b/sound/core/seq/seq_memory.c
index 801076687bb16..c850345c43b53 100644
--- a/sound/core/seq/seq_memory.c
+++ b/sound/core/seq/seq_memory.c
@@ -383,15 +383,20 @@ int snd_seq_pool_init(struct snd_seq_pool *pool)
if (snd_BUG_ON(!pool))
return -EINVAL;
- if (pool->ptr) /* should be atomic? */
- return 0;
- pool->ptr = vmalloc(sizeof(struct snd_seq_event_cell) * pool->size);
- if (!pool->ptr)
+ cellptr = vmalloc(sizeof(struct snd_seq_event_cell) * pool->size);
+ if (!cellptr)
return -ENOMEM;
/* add new cells to the free cell list */
spin_lock_irqsave(&pool->lock, flags);
+ if (pool->ptr) {
+ spin_unlock_irqrestore(&pool->lock, flags);
+ vfree(cellptr);
+ return 0;
+ }
+
+ pool->ptr = cellptr;
pool->free = NULL;
for (cell = 0; cell < pool->size; cell++) {
From 0b8c82190c12e530eb6003720dac103bf63e146e Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 15 Feb 2016 16:37:24 +0100
Subject: [PATCH 40/47] ALSA: hda - Cancel probe work instead of flush at
remove
The commit [991f86d7ae4e: ALSA: hda - Flush the pending probe work at
remove] introduced the sync of async probe work at remove for fixing
the race. However, this may lead to another hangup when the module
removal is performed quickly before starting the probe work, because
it issues flush_work() and it's blocked forever.
The workaround is to use cancel_work_sync() instead of flush_work()
there.
Fixes: 991f86d7ae4e ('ALSA: hda - Flush the pending probe work at remove')
Cc: <stable@vger.kernel.org> # v3.17+
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/pci/hda/hda_intel.c | 4 ++--
1 file changed, 2 insertions(+), 2 deletions(-)
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 4045dca3d699e..ce6b97f313900 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -2168,10 +2168,10 @@ static void azx_remove(struct pci_dev *pci)
struct hda_intel *hda;
if (card) {
- /* flush the pending probing work */
+ /* cancel the pending probing work */
chip = card->private_data;
hda = container_of(chip, struct hda_intel, chip);
- flush_work(&hda->probe_work);
+ cancel_work_sync(&hda->probe_work);
snd_card_free(card);
}
From 13d5e5d4725c64ec06040d636832e78453f477b7 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 16 Feb 2016 14:15:59 +0100
Subject: [PATCH 41/47] ALSA: seq: Fix double port list deletion
The commit [7f0973e973cd: ALSA: seq: Fix lockdep warnings due to
double mutex locks] split the management of two linked lists (source
and destination) into two individual calls for avoiding the AB/BA
deadlock. However, this may leave the possible double deletion of one
of two lists when the counterpart is being deleted concurrently.
It ends up with a list corruption, as revealed by syzkaller fuzzer.
This patch fixes it by checking the list emptiness and skipping the
deletion and the following process.
BugLink: http://lkml.kernel.org/r/CACT4Y+bay9qsrz6dQu31EcGaH9XwfW7o3oBzSQUG9fMszoh=Sg@mail.gmail.com
Fixes: 7f0973e973cd ('ALSA: seq: Fix lockdep warnings due to 'double mutex locks)
Reported-by: Dmitry Vyukov <dvyukov@google.com>
Tested-by: Dmitry Vyukov <dvyukov@google.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/core/seq/seq_ports.c | 13 ++++++++-----
1 file changed, 8 insertions(+), 5 deletions(-)
diff --git a/sound/core/seq/seq_ports.c b/sound/core/seq/seq_ports.c
index 921fb2bd8fadb..fe686ee41c6da 100644
--- a/sound/core/seq/seq_ports.c
+++ b/sound/core/seq/seq_ports.c
@@ -535,19 +535,22 @@ static void delete_and_unsubscribe_port(struct snd_seq_client *client,
bool is_src, bool ack)
{
struct snd_seq_port_subs_info *grp;
+ struct list_head *list;
+ bool empty;
grp = is_src ? &port->c_src : &port->c_dest;
+ list = is_src ? &subs->src_list : &subs->dest_list;
down_write(&grp->list_mutex);
write_lock_irq(&grp->list_lock);
- if (is_src)
- list_del(&subs->src_list);
- else
- list_del(&subs->dest_list);
+ empty = list_empty(list);
+ if (!empty)
+ list_del_init(list);
grp->exclusive = 0;
write_unlock_irq(&grp->list_lock);
up_write(&grp->list_mutex);
- unsubscribe_port(client, port, grp, &subs->info, ack);
+ if (!empty)
+ unsubscribe_port(client, port, grp, &subs->info, ack);
}
/* connect two ports */
From 67ec1072b053c15564e6090ab30127895dc77a89 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 17 Feb 2016 14:30:26 +0100
Subject: [PATCH 42/47] ALSA: pcm: Fix rwsem deadlock for non-atomic PCM stream
A non-atomic PCM stream may take snd_pcm_link_rwsem rw semaphore twice
in the same code path, e.g. one in snd_pcm_action_nonatomic() and
another in snd_pcm_stream_lock(). Usually this is OK, but when a
write lock is issued between these two read locks, the problem
happens: the write lock is blocked due to the first reade lock, and
the second read lock is also blocked by the write lock. This
eventually deadlocks.
The reason is the way rwsem manages waiters; it's queued like FIFO, so
even if the writer itself doesn't take the lock yet, it blocks all the
waiters (including reads) queued after it.
As a workaround, in this patch, we replace the standard down_write()
with an spinning loop. This is far from optimal, but it's good
enough, as the spinning time is supposed to be relatively short for
normal PCM operations, and the code paths requiring the write lock
aren't called so often.
Reported-by: Vinod Koul <vinod.koul@intel.com>
Tested-by: Ramesh Babu <ramesh.babu@intel.com>
Cc: <stable@vger.kernel.org> # v3.18+
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/core/pcm_native.c | 16 ++++++++++++++--
1 file changed, 14 insertions(+), 2 deletions(-)
diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c
index fadd3eb8e8bb2..9106d8e2300ea 100644
--- a/sound/core/pcm_native.c
+++ b/sound/core/pcm_native.c
@@ -74,6 +74,18 @@ static int snd_pcm_open(struct file *file, struct snd_pcm *pcm, int stream);
static DEFINE_RWLOCK(snd_pcm_link_rwlock);
static DECLARE_RWSEM(snd_pcm_link_rwsem);
+/* Writer in rwsem may block readers even during its waiting in queue,
+ * and this may lead to a deadlock when the code path takes read sem
+ * twice (e.g. one in snd_pcm_action_nonatomic() and another in
+ * snd_pcm_stream_lock()). As a (suboptimal) workaround, let writer to
+ * spin until it gets the lock.
+ */
+static inline void down_write_nonblock(struct rw_semaphore *lock)
+{
+ while (!down_write_trylock(lock))
+ cond_resched();
+}
+
/**
* snd_pcm_stream_lock - Lock the PCM stream
* @substream: PCM substream
@@ -1813,7 +1825,7 @@ static int snd_pcm_link(struct snd_pcm_substream *substream, int fd)
res = -ENOMEM;
goto _nolock;
}
- down_write(&snd_pcm_link_rwsem);
+ down_write_nonblock(&snd_pcm_link_rwsem);
write_lock_irq(&snd_pcm_link_rwlock);
if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN ||
substream->runtime->status->state != substream1->runtime->status->state ||
@@ -1860,7 +1872,7 @@ static int snd_pcm_unlink(struct snd_pcm_substream *substream)
struct snd_pcm_substream *s;
int res = 0;
- down_write(&snd_pcm_link_rwsem);
+ down_write_nonblock(&snd_pcm_link_rwsem);
write_lock_irq(&snd_pcm_link_rwlock);
if (!snd_pcm_stream_linked(substream)) {
res = -EALREADY;
From 7e31a0159461818a1bda49662921b98a29c1187b Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 22 Feb 2016 15:18:13 +0100
Subject: [PATCH 43/47] ALSA: hda - Apply clock gate workaround to Skylake, too
Some Skylake machines show the codec probe errors in certain
situations, e.g. HP Z240 desktop fails to probe the onboard Realtek
codec at reloading the snd-hda-intel module like:
snd_hda_intel 0000:00:1f.3: spurious response 0x200:0x2, last cmd=0x000000
snd_hda_intel 0000:00:1f.3: azx_get_response timeout, switching to polling mode: lastcmd=0x000f0000
snd_hda_intel 0000:00:1f.3: No response from codec, disabling MSI: last cmd=0x000f0000
snd_hda_intel 0000:00:1f.3: Codec #0 probe error; disabling it...
hdaudio hdaudioC0D2: no AFG or MFG node found
snd_hda_intel 0000:00:1f.3: no codecs initialized
Also, HP G470 G3 suffers from the similar problem, as reported in
bugzilla below. On this machine, the codec probe error appears even
at a fresh boot.
As Libin suggested, the same workaround used for Broxton in the commit
[6639484ddaf6: ALSA: hda - disable dynamic clock gating on Broxton
before reset] can be applied for Skylake in order to fix this problem.
The Intel HW team also confirmed that this is needed for SKL.
This patch makes the workaround applied to both SKL and BXT
platforms. The referred macros are moved and one superfluous macro
(IS_BROXTON()) is another one (IS_BXT()) as well.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=112731
Suggested-by: Libin Yang <libin.yang@linux.intel.com>
Cc: <stable@vger.kernel.org> # v4.4+
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/pci/hda/hda_intel.c | 16 +++++++---------
1 file changed, 7 insertions(+), 9 deletions(-)
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index ce6b97f313900..e5240cb3749f4 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -363,7 +363,10 @@ enum {
((pci)->device == 0x0d0c) || \
((pci)->device == 0x160c))
-#define IS_BROXTON(pci) ((pci)->device == 0x5a98)
+#define IS_SKL(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0xa170)
+#define IS_SKL_LP(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0x9d70)
+#define IS_BXT(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0x5a98)
+#define IS_SKL_PLUS(pci) (IS_SKL(pci) || IS_SKL_LP(pci) || IS_BXT(pci))
static char *driver_short_names[] = {
[AZX_DRIVER_ICH] = "HDA Intel",
@@ -540,13 +543,13 @@ static void hda_intel_init_chip(struct azx *chip, bool full_reset)
if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL)
snd_hdac_set_codec_wakeup(bus, true);
- if (IS_BROXTON(pci)) {
+ if (IS_SKL_PLUS(pci)) {
pci_read_config_dword(pci, INTEL_HDA_CGCTL, &val);
val = val & ~INTEL_HDA_CGCTL_MISCBDCGE;
pci_write_config_dword(pci, INTEL_HDA_CGCTL, val);
}
azx_init_chip(chip, full_reset);
- if (IS_BROXTON(pci)) {
+ if (IS_SKL_PLUS(pci)) {
pci_read_config_dword(pci, INTEL_HDA_CGCTL, &val);
val = val | INTEL_HDA_CGCTL_MISCBDCGE;
pci_write_config_dword(pci, INTEL_HDA_CGCTL, val);
@@ -555,7 +558,7 @@ static void hda_intel_init_chip(struct azx *chip, bool full_reset)
snd_hdac_set_codec_wakeup(bus, false);
/* reduce dma latency to avoid noise */
- if (IS_BROXTON(pci))
+ if (IS_BXT(pci))
bxt_reduce_dma_latency(chip);
}
@@ -977,11 +980,6 @@ static int azx_resume(struct device *dev)
/* put codec down to D3 at hibernation for Intel SKL+;
* otherwise BIOS may still access the codec and screw up the driver
*/
-#define IS_SKL(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0xa170)
-#define IS_SKL_LP(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0x9d70)
-#define IS_BXT(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0x5a98)
-#define IS_SKL_PLUS(pci) (IS_SKL(pci) || IS_SKL_LP(pci) || IS_BXT(pci))
-
static int azx_freeze_noirq(struct device *dev)
{
struct pci_dev *pci = to_pci_dev(dev);
From 3b43b71f05d3ecd01c4116254666d9492301697d Mon Sep 17 00:00:00 2001
From: Kai-Heng Feng <kaihengfeng@gmail.com>
Date: Thu, 25 Feb 2016 15:19:38 +0800
Subject: [PATCH 44/47] ALSA: hda - Fixing background noise on Dell Inspiron
3162
After login to the desktop on Dell Inspiron 3162,
there's a very loud background noise comes from the builtin speaker.
The noise does not go away even if the speaker is muted.
The noise disappears after using the aamix fixup.
Codec: Realtek ALC3234
Address: 0
AFG Function Id: 0x1 (unsol 1)
Vendor Id: 0x10ec0255
Subsystem Id: 0x10280725
Revision Id: 0x100002
No Modem Function Group found
BugLink: http://bugs.launchpad.net/bugs/1549620
Signed-off-by: Kai-Heng Feng <kai.heng.feng@canonical.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/pci/hda/patch_realtek.c | 8 ++++++++
1 file changed, 8 insertions(+)
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index efd4980cffb8a..72fa58dd77239 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -4749,6 +4749,7 @@ enum {
ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE,
ALC293_FIXUP_LENOVO_SPK_NOISE,
ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY,
+ ALC255_FIXUP_DELL_SPK_NOISE,
};
static const struct hda_fixup alc269_fixups[] = {
@@ -5368,6 +5369,12 @@ static const struct hda_fixup alc269_fixups[] = {
.type = HDA_FIXUP_FUNC,
.v.func = alc233_fixup_lenovo_line2_mic_hotkey,
},
+ [ALC255_FIXUP_DELL_SPK_NOISE] = {
+ .type = HDA_FIXUP_FUNC,
+ .v.func = alc_fixup_disable_aamix,
+ .chained = true,
+ .chain_id = ALC255_FIXUP_DELL1_MIC_NO_PRESENCE
+ },
};
static const struct snd_pci_quirk alc269_fixup_tbl[] = {
@@ -5410,6 +5417,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x1028, 0x06df, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK),
SND_PCI_QUIRK(0x1028, 0x06e0, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK),
SND_PCI_QUIRK(0x1028, 0x0704, "Dell XPS 13", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE),
+ SND_PCI_QUIRK(0x1028, 0x0725, "Dell Inspiron 3162", ALC255_FIXUP_DELL_SPK_NOISE),
SND_PCI_QUIRK(0x1028, 0x164a, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x1028, 0x164b, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC2),
From 2ae955774f29bbd7d16149cb0ae8d0319bf2ecc4 Mon Sep 17 00:00:00 2001
From: David Henningsson <david.henningsson@canonical.com>
Date: Thu, 25 Feb 2016 09:37:05 +0100
Subject: [PATCH 45/47] ALSA: hda - Fixup speaker pass-through control for nid
0x14 on ALC225
On one of the machines we enable, we found that the actual speaker volume
did not always correspond to the volume set in alsamixer. This patch
fixes that problem.
This patch was orginally written by Kailang @ Realtek, I've rebased it
to fit sound git master.
Cc: stable@vger.kernel.org
BugLink: https://bugs.launchpad.net/bugs/1549660
Co-Authored-By: Kailang <kailang@realtek.com>
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/pci/hda/patch_realtek.c | 23 +++++++++++++++++++++--
1 file changed, 21 insertions(+), 2 deletions(-)
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 72fa58dd77239..7fded69fb58ed 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -3801,6 +3801,10 @@ static void alc_headset_mode_mic_in(struct hda_codec *codec, hda_nid_t hp_pin,
static void alc_headset_mode_default(struct hda_codec *codec)
{
+ static struct coef_fw coef0225[] = {
+ UPDATE_COEF(0x45, 0x3f<<10, 0x34<<10),
+ {}
+ };
static struct coef_fw coef0255[] = {
WRITE_COEF(0x45, 0xc089),
WRITE_COEF(0x45, 0xc489),
@@ -3842,6 +3846,9 @@ static void alc_headset_mode_default(struct hda_codec *codec)
};
switch (codec->core.vendor_id) {
+ case 0x10ec0225:
+ alc_process_coef_fw(codec, coef0225);
+ break;
case 0x10ec0255:
case 0x10ec0256:
alc_process_coef_fw(codec, coef0255);
@@ -4750,6 +4757,7 @@ enum {
ALC293_FIXUP_LENOVO_SPK_NOISE,
ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY,
ALC255_FIXUP_DELL_SPK_NOISE,
+ ALC225_FIXUP_DELL1_MIC_NO_PRESENCE,
};
static const struct hda_fixup alc269_fixups[] = {
@@ -5375,6 +5383,17 @@ static const struct hda_fixup alc269_fixups[] = {
.chained = true,
.chain_id = ALC255_FIXUP_DELL1_MIC_NO_PRESENCE
},
+ [ALC225_FIXUP_DELL1_MIC_NO_PRESENCE] = {
+ .type = HDA_FIXUP_VERBS,
+ .v.verbs = (const struct hda_verb[]) {
+ /* Disable pass-through path for FRONT 14h */
+ { 0x20, AC_VERB_SET_COEF_INDEX, 0x36 },
+ { 0x20, AC_VERB_SET_PROC_COEF, 0x57d7 },
+ {}
+ },
+ .chained = true,
+ .chain_id = ALC269_FIXUP_DELL1_MIC_NO_PRESENCE
+ },
};
static const struct snd_pci_quirk alc269_fixup_tbl[] = {
@@ -5646,10 +5665,10 @@ static const struct hda_model_fixup alc269_fixup_models[] = {
{0x21, 0x03211020}
static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = {
- SND_HDA_PIN_QUIRK(0x10ec0225, 0x1028, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE,
+ SND_HDA_PIN_QUIRK(0x10ec0225, 0x1028, "Dell", ALC225_FIXUP_DELL1_MIC_NO_PRESENCE,
ALC225_STANDARD_PINS,
{0x14, 0x901701a0}),
- SND_HDA_PIN_QUIRK(0x10ec0225, 0x1028, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE,
+ SND_HDA_PIN_QUIRK(0x10ec0225, 0x1028, "Dell", ALC225_FIXUP_DELL1_MIC_NO_PRESENCE,
ALC225_STANDARD_PINS,
{0x14, 0x901701b0}),
SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL2_MIC_NO_PRESENCE,
From f883982dc1b117f04579f0896821cd9f2e397f94 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 25 Feb 2016 14:31:59 +0100
Subject: [PATCH 46/47] ALSA: hda - Fix headset support and noise on HP
EliteBook 755 G2
HP EliteBook 755 G2 with ALC3228 (ALC280) codec [103c:221c] requires
the known fixup (ALC269_FIXUP_HEADSET_MIC) for making the headset mic
working. Also, it suffers from the loopback noise problem, so we
should disable aamix path as well.
Reported-by: Derick Eddington <derick.eddington@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/pci/hda/patch_realtek.c | 8 ++++++++
1 file changed, 8 insertions(+)
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 7fded69fb58ed..1f357cd72d9c3 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -4758,6 +4758,7 @@ enum {
ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY,
ALC255_FIXUP_DELL_SPK_NOISE,
ALC225_FIXUP_DELL1_MIC_NO_PRESENCE,
+ ALC280_FIXUP_HP_HEADSET_MIC,
};
static const struct hda_fixup alc269_fixups[] = {
@@ -5394,6 +5395,12 @@ static const struct hda_fixup alc269_fixups[] = {
.chained = true,
.chain_id = ALC269_FIXUP_DELL1_MIC_NO_PRESENCE
},
+ [ALC280_FIXUP_HP_HEADSET_MIC] = {
+ .type = HDA_FIXUP_FUNC,
+ .v.func = alc_fixup_disable_aamix,
+ .chained = true,
+ .chain_id = ALC269_FIXUP_HEADSET_MIC,
+ },
};
static const struct snd_pci_quirk alc269_fixup_tbl[] = {
@@ -5497,6 +5504,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x103c, 0x2335, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1),
SND_PCI_QUIRK(0x103c, 0x2336, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1),
SND_PCI_QUIRK(0x103c, 0x2337, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1),
+ SND_PCI_QUIRK(0x103c, 0x221c, "HP EliteBook 755 G2", ALC280_FIXUP_HP_HEADSET_MIC),
SND_PCI_QUIRK(0x1043, 0x103f, "ASUS TX300", ALC282_FIXUP_ASUS_TX300),
SND_PCI_QUIRK(0x1043, 0x106d, "Asus K53BE", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
SND_PCI_QUIRK(0x1043, 0x115d, "Asus 1015E", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
From 473f414564528a819f0c2bb6b4bf26366b64c9ab Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 23 Feb 2016 15:54:47 +0100
Subject: [PATCH 47/47] ALSA: hda - Loop interrupt handling until really
cleared
Currently the interrupt handler of HD-audio driver assumes that no irq
update is needed while processing the irq. But in reality, it has
been confirmed that the HW irq is issued even during the irq
handling. Since we clear the irq status at the beginning, process the
interrupt, then exits from the handler, the lately issued interrupt is
left untouched without being properly processed.
This patch changes the interrupt handler code to loop over the
check-and-process. The handler tries repeatedly as long as the IRQ
status are turned on, and either stream or CORB/RIRB is handled.
For checking the stream handling, snd_hdac_bus_handle_stream_irq()
returns a value indicating the stream indices bits. Other than that,
the change is only in the irq handler itself.
Reported-by: Libin Yang <libin.yang@linux.intel.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
include/sound/hdaudio.h | 2 +-
sound/hda/hdac_controller.c | 7 ++++-
sound/pci/hda/hda_controller.c | 47 +++++++++++++++++++---------------
3 files changed, 33 insertions(+), 23 deletions(-)
diff --git a/include/sound/hdaudio.h b/include/sound/hdaudio.h
index e2b712c90d3f2..c21c38ce74501 100644
--- a/include/sound/hdaudio.h
+++ b/include/sound/hdaudio.h
@@ -343,7 +343,7 @@ void snd_hdac_bus_enter_link_reset(struct hdac_bus *bus);
void snd_hdac_bus_exit_link_reset(struct hdac_bus *bus);
void snd_hdac_bus_update_rirb(struct hdac_bus *bus);
-void snd_hdac_bus_handle_stream_irq(struct hdac_bus *bus, unsigned int status,
+int snd_hdac_bus_handle_stream_irq(struct hdac_bus *bus, unsigned int status,
void (*ack)(struct hdac_bus *,
struct hdac_stream *));
diff --git a/sound/hda/hdac_controller.c b/sound/hda/hdac_controller.c
index b5a17cb510a0b..8c486235c9052 100644
--- a/sound/hda/hdac_controller.c
+++ b/sound/hda/hdac_controller.c
@@ -426,18 +426,22 @@ EXPORT_SYMBOL_GPL(snd_hdac_bus_stop_chip);
* @bus: HD-audio core bus
* @status: INTSTS register value
* @ask: callback to be called for woken streams
+ *
+ * Returns the bits of handled streams, or zero if no stream is handled.
*/
-void snd_hdac_bus_handle_stream_irq(struct hdac_bus *bus, unsigned int status,
+int snd_hdac_bus_handle_stream_irq(struct hdac_bus *bus, unsigned int status,
void (*ack)(struct hdac_bus *,
struct hdac_stream *))
{
struct hdac_stream *azx_dev;
u8 sd_status;
+ int handled = 0;
list_for_each_entry(azx_dev, &bus->stream_list, list) {
if (status & azx_dev->sd_int_sta_mask) {
sd_status = snd_hdac_stream_readb(azx_dev, SD_STS);
snd_hdac_stream_writeb(azx_dev, SD_STS, SD_INT_MASK);
+ handled |= 1 << azx_dev->index;
if (!azx_dev->substream || !azx_dev->running ||
!(sd_status & SD_INT_COMPLETE))
continue;
@@ -445,6 +449,7 @@ void snd_hdac_bus_handle_stream_irq(struct hdac_bus *bus, unsigned int status,
ack(bus, azx_dev);
}
}
+ return handled;
}
EXPORT_SYMBOL_GPL(snd_hdac_bus_handle_stream_irq);
diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c
index 37cf9cee98355..27de8015717d9 100644
--- a/sound/pci/hda/hda_controller.c
+++ b/sound/pci/hda/hda_controller.c
@@ -930,6 +930,8 @@ irqreturn_t azx_interrupt(int irq, void *dev_id)
struct azx *chip = dev_id;
struct hdac_bus *bus = azx_bus(chip);
u32 status;
+ bool active, handled = false;
+ int repeat = 0; /* count for avoiding endless loop */
#ifdef CONFIG_PM
if (azx_has_pm_runtime(chip))
@@ -939,33 +941,36 @@ irqreturn_t azx_interrupt(int irq, void *dev_id)
spin_lock(&bus->reg_lock);
- if (chip->disabled) {
- spin_unlock(&bus->reg_lock);
- return IRQ_NONE;
- }
-
- status = azx_readl(chip, INTSTS);
- if (status == 0 || status == 0xffffffff) {
- spin_unlock(&bus->reg_lock);
- return IRQ_NONE;
- }
+ if (chip->disabled)
+ goto unlock;
- snd_hdac_bus_handle_stream_irq(bus, status, stream_update);
+ do {
+ status = azx_readl(chip, INTSTS);
+ if (status == 0 || status == 0xffffffff)
+ break;
- /* clear rirb int */
- status = azx_readb(chip, RIRBSTS);
- if (status & RIRB_INT_MASK) {
- if (status & RIRB_INT_RESPONSE) {
- if (chip->driver_caps & AZX_DCAPS_CTX_WORKAROUND)
- udelay(80);
- snd_hdac_bus_update_rirb(bus);
+ handled = true;
+ active = false;
+ if (snd_hdac_bus_handle_stream_irq(bus, status, stream_update))
+ active = true;
+
+ /* clear rirb int */
+ status = azx_readb(chip, RIRBSTS);
+ if (status & RIRB_INT_MASK) {
+ active = true;
+ if (status & RIRB_INT_RESPONSE) {
+ if (chip->driver_caps & AZX_DCAPS_CTX_WORKAROUND)
+ udelay(80);
+ snd_hdac_bus_update_rirb(bus);
+ }
+ azx_writeb(chip, RIRBSTS, RIRB_INT_MASK);
}
- azx_writeb(chip, RIRBSTS, RIRB_INT_MASK);
- }
+ } while (active && ++repeat < 10);
+ unlock:
spin_unlock(&bus->reg_lock);
- return IRQ_HANDLED;
+ return IRQ_RETVAL(handled);
}
EXPORT_SYMBOL_GPL(azx_interrupt);