From 0b1f6ec7a5fb3faff1a62afee132dac316eec63d Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Mon, 9 Feb 2015 08:05:22 +0000
Subject: [PATCH 01/26] ASoC: rsnd: set device data before
snd_soc_register_platform/component
Set device data before snd_soc_register_platform/component.
Otherwise, it will use NULL pointer if user calls unbind -> bind or
rmmod -> insmod
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/sh/rcar/core.c | 4 ++--
1 file changed, 2 insertions(+), 2 deletions(-)
diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c
index 75308bbc2ce8..fc227d3bc021 100644
--- a/sound/soc/sh/rcar/core.c
+++ b/sound/soc/sh/rcar/core.c
@@ -1268,6 +1268,8 @@ static int rsnd_probe(struct platform_device *pdev)
goto exit_snd_probe;
}
+ dev_set_drvdata(dev, priv);
+
/*
* asoc register
*/
@@ -1284,8 +1286,6 @@ static int rsnd_probe(struct platform_device *pdev)
goto exit_snd_soc;
}
- dev_set_drvdata(dev, priv);
-
pm_runtime_enable(dev);
dev_info(dev, "probed\n");
From 541b03ad6cfe0e415273f096fd8c47d2879c6c15 Mon Sep 17 00:00:00 2001
From: Nicolin Chen <nicoleotsuka@gmail.com>
Date: Tue, 10 Feb 2015 21:31:43 -0800
Subject: [PATCH 02/26] ASoC: fsl_ssi: Fix the incorrect limitation of the bit
clock rate
According to i.MX Reference Manual, the bit-clock frequency generated
by SSI must be never greater than 1/5 of the peripheral clock frequency.
This peripheral clock, however, is not baudclk but the IPG clock (i.e.
ssi_private->clk in the fsl_ssi driver).
So this patch just simply fixes the incorrect limitation applied to
the bit clock (baudclk) rate.
Signed-off-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/fsl/fsl_ssi.c | 11 +++++++----
1 file changed, 7 insertions(+), 4 deletions(-)
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index 059496ed9ad7..d7365c5d7ec0 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -603,10 +603,6 @@ static int fsl_ssi_set_bclk(struct snd_pcm_substream *substream,
factor = (div2 + 1) * (7 * psr + 1) * 2;
for (i = 0; i < 255; i++) {
- /* The bclk rate must be smaller than 1/5 sysclk rate */
- if (factor * (i + 1) < 5)
- continue;
-
tmprate = freq * factor * (i + 2);
if (baudclk_is_used)
@@ -614,6 +610,13 @@ static int fsl_ssi_set_bclk(struct snd_pcm_substream *substream,
else
clkrate = clk_round_rate(ssi_private->baudclk, tmprate);
+ /*
+ * Hardware limitation: The bclk rate must be
+ * never greater than 1/5 IPG clock rate
+ */
+ if (clkrate * 5 > clk_get_rate(ssi_private->clk))
+ continue;
+
clkrate /= factor;
afreq = clkrate / (i + 1);
From ffa047577127336861d91f3934133f8e8906d1b4 Mon Sep 17 00:00:00 2001
From: Guenter Roeck <linux@roeck-us.net>
Date: Wed, 11 Feb 2015 13:13:18 -0800
Subject: [PATCH 03/26] ASoC: Fix MAX98357A codec driver dependencies
The max98357a driver depends on GPIOLIB. This may cause the following
build failure.
sound/soc/codecs/max98357a.c: In function 'max98357a_daiops_trigger':
sound/soc/codecs/max98357a.c:30:3: error: implicit declaration of function 'gpiod_set_value'
sound/soc/codecs/max98357a.c: In function 'max98357a_codec_probe':
sound/soc/codecs/max98357a.c:55:2: error: implicit declaration of function 'devm_gpiod_get'
sound/soc/codecs/max98357a.c:61:2: error: implicit declaration of function 'gpiod_direction_output'
Seen with mips:allmodconfig as well as various randconfig builds.
Fixes: af5adf129369 ("ASoC: max98357a: Add MAX98357A codec driver")
Cc: Kenneth Westfield <kwestfie@codeaurora.org>
Signed-off-by: Guenter Roeck <linux@roeck-us.net>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/codecs/Kconfig | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 064e6c18e109..ea9f0e31f9d4 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -69,7 +69,7 @@ config SND_SOC_ALL_CODECS
select SND_SOC_MAX98088 if I2C
select SND_SOC_MAX98090 if I2C
select SND_SOC_MAX98095 if I2C
- select SND_SOC_MAX98357A
+ select SND_SOC_MAX98357A if GPIOLIB
select SND_SOC_MAX9850 if I2C
select SND_SOC_MAX9768 if I2C
select SND_SOC_MAX9877 if I2C
From 5c8be987d4d9c0262e6229e342fa0da8a5aeee47 Mon Sep 17 00:00:00 2001
From: =?UTF-8?q?Vincent=20Stehl=C3=A9?= <vincent.stehle@laposte.net>
Date: Wed, 11 Feb 2015 23:08:59 +0100
Subject: [PATCH 04/26] ASoC: max98357a: Fix missing include
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit
This fixes the following compilation errors:
sound/soc/codecs/max98357a.c: In function ‘max98357a_daiops_trigger’:
sound/soc/codecs/max98357a.c:30:3: error: implicit declaration of function ‘gpiod_set_value’ [-Werror=implicit-function-declaration]
sound/soc/codecs/max98357a.c: In function ‘max98357a_codec_probe’:
sound/soc/codecs/max98357a.c:55:2: error: implicit declaration of function ‘devm_gpiod_get’ [-Werror=implicit-function-declaration]
sound/soc/codecs/max98357a.c:61:2: error: implicit declaration of function ‘gpiod_direction_output’ [-Werror=implicit-function-declaration]
cc1: some warnings being treated as errors
Signed-off-by: Vincent Stehlé <vincent.stehle@laposte.net>
Cc: Kenneth Westfield <kwestfie@codeaurora.org>
Cc: Mark Brown <broonie@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/codecs/max98357a.c | 1 +
1 file changed, 1 insertion(+)
diff --git a/sound/soc/codecs/max98357a.c b/sound/soc/codecs/max98357a.c
index 1806333ea29e..f493fb6fd4ea 100644
--- a/sound/soc/codecs/max98357a.c
+++ b/sound/soc/codecs/max98357a.c
@@ -14,6 +14,7 @@
#include <linux/module.h>
#include <linux/gpio.h>
+#include <linux/gpio/consumer.h>
#include <sound/soc.h>
#define DRV_NAME "max98357a"
From fb5ab7296a2bea17c38fae48af2808a07049ac90 Mon Sep 17 00:00:00 2001
From: Kiran Padwal <kiran.padwal@smartplayin.com>
Date: Thu, 12 Feb 2015 14:38:02 +0530
Subject: [PATCH 05/26] ASoC: omap-hdmi-audio: Add missing error check for
devm_kzalloc
This patch add a missing check on the return value of devm_kzalloc,
which would cause a NULL pointer dereference in a OOM situation.
Signed-off-by: Kiran Padwal <kiran.padwal@smartplayin.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/omap/omap-hdmi-audio.c | 3 +++
1 file changed, 3 insertions(+)
diff --git a/sound/soc/omap/omap-hdmi-audio.c b/sound/soc/omap/omap-hdmi-audio.c
index 3f9ac7dbdc80..069ad451d05d 100644
--- a/sound/soc/omap/omap-hdmi-audio.c
+++ b/sound/soc/omap/omap-hdmi-audio.c
@@ -352,6 +352,9 @@ static int omap_hdmi_audio_probe(struct platform_device *pdev)
return ret;
card = devm_kzalloc(dev, sizeof(*card), GFP_KERNEL);
+ if (!card)
+ return -ENOMEM;
+
card->name = devm_kasprintf(dev, GFP_KERNEL,
"HDMI %s", dev_name(ad->dssdev));
card->owner = THIS_MODULE;
From 7bd345c9e87d879d696c6843fe200b60c2051c84 Mon Sep 17 00:00:00 2001
From: Mengdong Lin <mengdong.lin@intel.com>
Date: Fri, 13 Feb 2015 19:21:25 +0800
Subject: [PATCH 06/26] ASoC: Intel: set initial runtime PM status to active
for ACPI-enumerated ADSP
The ADSP on Braswell/Baytrail is an ACPI device. This patch sets its initial
runtime PM status to active. Otherwise, its initial status is suspended and
runtime_suspend ops will not be called after probe and thus cannot further
trigger ACPI _PS3 (D3) method to put the device into low power D3cold state.
Signed-off-by: Mengdong Lin <mengdong.lin@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/sst/sst.c | 4 ++++
1 file changed, 4 insertions(+)
diff --git a/sound/soc/intel/sst/sst.c b/sound/soc/intel/sst/sst.c
index 8a8d56a146e7..d6ea80076ea2 100644
--- a/sound/soc/intel/sst/sst.c
+++ b/sound/soc/intel/sst/sst.c
@@ -379,6 +379,10 @@ void sst_configure_runtime_pm(struct intel_sst_drv *ctx)
* initially active. So change the state to active before
* enabling the pm
*/
+
+ if (!acpi_disabled)
+ pm_runtime_set_active(ctx->dev);
+
pm_runtime_enable(ctx->dev);
if (acpi_disabled)
From e7a961c9578ce227d3c62c4cce9463b763a1e0c0 Mon Sep 17 00:00:00 2001
From: Bard Liao <bardliao@realtek.com>
Date: Tue, 17 Feb 2015 13:59:27 +0800
Subject: [PATCH 07/26] ASoC: rt5670: Fix the speaker mono output issue
We need to set left/right control for the speaker amp to get stereo
output on speaker.
Signed-off-by: Bard Liao <bardliao@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/codecs/rt5670.c | 6 ++++++
1 file changed, 6 insertions(+)
diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c
index 8a0833de1665..d33f33ce865a 100644
--- a/sound/soc/codecs/rt5670.c
+++ b/sound/soc/codecs/rt5670.c
@@ -2591,6 +2591,12 @@ static int rt5670_i2c_probe(struct i2c_client *i2c,
regmap_write(rt5670->regmap, RT5670_RESET, 0);
+ regmap_read(rt5670->regmap, RT5670_VENDOR_ID, &val);
+ if (val >= 4)
+ regmap_write(rt5670->regmap, RT5670_GPIO_CTRL3, 0x0980);
+ else
+ regmap_write(rt5670->regmap, RT5670_GPIO_CTRL3, 0x0d00);
+
ret = regmap_register_patch(rt5670->regmap, init_list,
ARRAY_SIZE(init_list));
if (ret != 0)
From 014c4d637604c9af2f7f2ff4fd91b725a0c58a5c Mon Sep 17 00:00:00 2001
From: Arnd Bergmann <arnd@arndb.de>
Date: Wed, 18 Feb 2015 21:35:08 +0100
Subject: [PATCH 08/26] ASoC: Samsung: add missing I2C/SPI dependencies
A few sound drivers for the samsung platforms are missing dependencies
on I2C or SPI, which can lead to build errors like
codecs/rt5631.c:1737:1: warning: data definition has no type or storage class
31_i2c_driver);
codecs/rt5631.c:1737:1: error: type defaults to 'int' in declaration of 'module_i2c_driver' [-Werror=implicit-int]
codecs/rt5631.c:1737:1: warning: parameter names (without types) in function declaration
codecs/rt5631.c:1726:26: warning: 'rt5631_i2c_driver' defined but not used [-Wunused-variable]
I have gone through all the ones that did not already have
an I2C dependency and added the ones that I found missing,
namely arndale, odroid-x2, littlemill, bells and speyside
and this patch adds all the dependencies.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/samsung/Kconfig | 10 +++++-----
1 file changed, 5 insertions(+), 5 deletions(-)
diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig
index fc67f97f19f6..e0c4a4ec4280 100644
--- a/sound/soc/samsung/Kconfig
+++ b/sound/soc/samsung/Kconfig
@@ -185,7 +185,7 @@ config SND_SOC_SMDK_WM8994_PCM
config SND_SOC_SPEYSIDE
tristate "Audio support for Wolfson Speyside"
- depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410
+ depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 && I2C && SPI_MASTER
select SND_SAMSUNG_I2S
select SND_SOC_WM8996
select SND_SOC_WM9081
@@ -200,7 +200,7 @@ config SND_SOC_TOBERMORY
config SND_SOC_BELLS
tristate "Audio support for Wolfson Bells"
- depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 && MFD_ARIZONA
+ depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 && MFD_ARIZONA && I2C && SPI_MASTER
select SND_SAMSUNG_I2S
select SND_SOC_WM5102
select SND_SOC_WM5110
@@ -217,7 +217,7 @@ config SND_SOC_LOWLAND
config SND_SOC_LITTLEMILL
tristate "Audio support for Wolfson Littlemill"
- depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410
+ depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 && I2C
select SND_SAMSUNG_I2S
select MFD_WM8994
select SND_SOC_WM8994
@@ -234,7 +234,7 @@ config SND_SOC_SNOW
config SND_SOC_ODROIDX2
tristate "Audio support for Odroid-X2 and Odroid-U3"
- depends on SND_SOC_SAMSUNG
+ depends on SND_SOC_SAMSUNG && I2C
select SND_SOC_MAX98090
select SND_SAMSUNG_I2S
help
@@ -242,6 +242,6 @@ config SND_SOC_ODROIDX2
config SND_SOC_ARNDALE_RT5631_ALC5631
tristate "Audio support for RT5631(ALC5631) on Arndale Board"
- depends on SND_SOC_SAMSUNG
+ depends on SND_SOC_SAMSUNG && I2C
select SND_SAMSUNG_I2S
select SND_SOC_RT5631
From 52554fbd2f88a432a16e9e88e14c4b02ccb7cdb6 Mon Sep 17 00:00:00 2001
From: Arnd Bergmann <arnd@arndb.de>
Date: Wed, 18 Feb 2015 21:43:13 +0100
Subject: [PATCH 09/26] ASoC: cirrus: tlv320aic23 needs I2C
The tlv320aic23 codec is selected by the ep93xx snapper platform,
which are missing a dependency on I2C, and that can result in this
build error, as found during randconfig builds:
.../codecs/tlv320aic23-i2c.c: In function 'tlv320aic23_i2c_probe':
.../codecs/tlv320aic23-i2c.c:27:2: error: implicit declaration of function 'i2c_check_functionality' [-Werror=implicit-function-declaration]
if (!i2c_check_functionality(i2c->adapter, I2C_FUNC_SMBUS_BYTE_DATA))
^
This adds the missing dependency.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/cirrus/Kconfig | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/sound/soc/cirrus/Kconfig b/sound/soc/cirrus/Kconfig
index 7b7fbcd49e5e..c7cd60f009e9 100644
--- a/sound/soc/cirrus/Kconfig
+++ b/sound/soc/cirrus/Kconfig
@@ -16,7 +16,7 @@ config SND_EP93XX_SOC_AC97
config SND_EP93XX_SOC_SNAPPERCL15
tristate "SoC Audio support for Bluewater Systems Snapper CL15 module"
- depends on SND_EP93XX_SOC && MACH_SNAPPER_CL15
+ depends on SND_EP93XX_SOC && MACH_SNAPPER_CL15 && I2C
select SND_EP93XX_SOC_I2S
select SND_SOC_TLV320AIC23_I2C
help
From 08d0a55c33393e6dc838e37b7a8657c28a6de10d Mon Sep 17 00:00:00 2001
From: Kenneth Westfield <kwestfie@codeaurora.org>
Date: Tue, 17 Feb 2015 00:53:11 -0800
Subject: [PATCH 10/26] ASoC: max98357a: Add missing header files
Add missing header files to avoid implicit
declarations and indirect inclusions.
Signed-off-by: Kenneth Westfield <kwestfie@codeaurora.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/codecs/max98357a.c | 11 ++++++++++-
1 file changed, 10 insertions(+), 1 deletion(-)
diff --git a/sound/soc/codecs/max98357a.c b/sound/soc/codecs/max98357a.c
index f493fb6fd4ea..e9e6efbc21dd 100644
--- a/sound/soc/codecs/max98357a.c
+++ b/sound/soc/codecs/max98357a.c
@@ -12,10 +12,19 @@
* max98357a.c -- MAX98357A ALSA SoC Codec driver
*/
-#include <linux/module.h>
+#include <linux/device.h>
+#include <linux/err.h>
#include <linux/gpio.h>
#include <linux/gpio/consumer.h>
+#include <linux/kernel.h>
+#include <linux/mod_devicetable.h>
+#include <linux/module.h>
+#include <linux/of.h>
+#include <linux/platform_device.h>
+#include <sound/pcm.h>
#include <sound/soc.h>
+#include <sound/soc-dai.h>
+#include <sound/soc-dapm.h>
#define DRV_NAME "max98357a"
From b3ec1c35385a16ddd98fdf104dcf4623a66e042a Mon Sep 17 00:00:00 2001
From: Vinod Koul <vinod.koul@intel.com>
Date: Thu, 12 Feb 2015 09:59:55 +0530
Subject: [PATCH 11/26] ASoC: Intel: update MMX ID to 3
The updated firmware expects the MMX ID to be used as 3, so update the
driver as well
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/sst-atom-controls.h | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/sound/soc/intel/sst-atom-controls.h b/sound/soc/intel/sst-atom-controls.h
index dfebfdd5eb2a..daecc58f28af 100644
--- a/sound/soc/intel/sst-atom-controls.h
+++ b/sound/soc/intel/sst-atom-controls.h
@@ -150,7 +150,7 @@ enum sst_cmd_type {
enum sst_task {
SST_TASK_SBA = 1,
- SST_TASK_MMX,
+ SST_TASK_MMX = 3,
};
enum sst_type {
From a825ac7678a43f7a22ff19842baebcf4aa14e950 Mon Sep 17 00:00:00 2001
From: Vinod Koul <vinod.koul@intel.com>
Date: Thu, 12 Feb 2015 09:59:59 +0530
Subject: [PATCH 12/26] ASoC: Intel: save and restore the CSR register
The IPC driver saved only IMR register, we need to save the CSR as well, so
add it
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/sst/sst.c | 5 ++++-
1 file changed, 4 insertions(+), 1 deletion(-)
diff --git a/sound/soc/intel/sst/sst.c b/sound/soc/intel/sst/sst.c
index d6ea80076ea2..97234ec4e416 100644
--- a/sound/soc/intel/sst/sst.c
+++ b/sound/soc/intel/sst/sst.c
@@ -350,7 +350,9 @@ static inline void sst_save_shim64(struct intel_sst_drv *ctx,
spin_lock_irqsave(&ctx->ipc_spin_lock, irq_flags);
- shim_regs->imrx = sst_shim_read64(shim, SST_IMRX),
+ shim_regs->imrx = sst_shim_read64(shim, SST_IMRX);
+ shim_regs->csr = sst_shim_read64(shim, SST_CSR);
+
spin_unlock_irqrestore(&ctx->ipc_spin_lock, irq_flags);
}
@@ -367,6 +369,7 @@ static inline void sst_restore_shim64(struct intel_sst_drv *ctx,
*/
spin_lock_irqsave(&ctx->ipc_spin_lock, irq_flags);
sst_shim_write64(shim, SST_IMRX, shim_regs->imrx),
+ sst_shim_write64(shim, SST_CSR, shim_regs->csr),
spin_unlock_irqrestore(&ctx->ipc_spin_lock, irq_flags);
}
From de251d773bb214fa5e7666a0da1225528e07da5e Mon Sep 17 00:00:00 2001
From: Vinod Koul <vinod.koul@intel.com>
Date: Thu, 12 Feb 2015 10:00:00 +0530
Subject: [PATCH 13/26] ASoC: Intel: reset the DSP while suspending
The manual recommends that we reset the DSP when we suspend so add that in
runtime suspend handler
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/sst/sst.c | 1 +
1 file changed, 1 insertion(+)
diff --git a/sound/soc/intel/sst/sst.c b/sound/soc/intel/sst/sst.c
index 97234ec4e416..11c578651c1c 100644
--- a/sound/soc/intel/sst/sst.c
+++ b/sound/soc/intel/sst/sst.c
@@ -416,6 +416,7 @@ static int intel_sst_runtime_suspend(struct device *dev)
synchronize_irq(ctx->irq_num);
flush_workqueue(ctx->post_msg_wq);
+ ctx->ops->reset(ctx);
/* save the shim registers because PMC doesn't save state */
sst_save_shim64(ctx, ctx->shim, ctx->shim_regs64);
From 850529249d7cce02e9bfae9476d09c8c51410d28 Mon Sep 17 00:00:00 2001
From: Bard Liao <bardliao@realtek.com>
Date: Mon, 16 Feb 2015 13:06:45 +0800
Subject: [PATCH 14/26] ASoC: rt5670: Set RT5670_IRQ_CTRL1 non volatile
RT5670_IRQ_CTRL1(0xbd) is a non volatile register. And we need to
restore its value after suspend/resume.
Signed-off-by: Bard Liao <bardliao@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
---
sound/soc/codecs/rt5670.c | 1 -
1 file changed, 1 deletion(-)
diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c
index d33f33ce865a..b651bc06cfdf 100644
--- a/sound/soc/codecs/rt5670.c
+++ b/sound/soc/codecs/rt5670.c
@@ -223,7 +223,6 @@ static bool rt5670_volatile_register(struct device *dev, unsigned int reg)
case RT5670_ADC_EQ_CTRL1:
case RT5670_EQ_CTRL1:
case RT5670_ALC_CTRL_1:
- case RT5670_IRQ_CTRL1:
case RT5670_IRQ_CTRL2:
case RT5670_INT_IRQ_ST:
case RT5670_IL_CMD:
From 148388f375394ac1afed543cb653c94be5faa810 Mon Sep 17 00:00:00 2001
From: =?UTF-8?q?Thomas=20Niederpr=C3=BCm?= <niederp@physik.uni-kl.de>
Date: Sat, 21 Feb 2015 17:22:38 +0100
Subject: [PATCH 15/26] ASoC: sta32x: fix register range in regmap.
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit
The STA32X_AUTO3 is a writable register that currently does not appear
in the regmap ranges(neither read nor write). By adding this register
to the register ranges there is no gap anymore and the existing
register ranges can be joined. This fixes a regression introduced in
commit a1be4cead9b9504aa6fc93b624975601cec8c188 where the driver was
moved to direct regmap usage and the STA32X_AUTO3 register was missed.
That made it impossible to choose the preset EQ mode set through the
STA32X_AUTO3 register.
Fixes: a1be4cead9 (ASoC: sta32x: Convert to direct regmap API usage)
Signed-off-by: Thomas Niederprüm <niederp@physik.uni-kl.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/codecs/sta32x.c | 6 ++----
1 file changed, 2 insertions(+), 4 deletions(-)
diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c
index 3a1343fa109b..007a0e3bc273 100644
--- a/sound/soc/codecs/sta32x.c
+++ b/sound/soc/codecs/sta32x.c
@@ -106,13 +106,11 @@ static const struct reg_default sta32x_regs[] = {
};
static const struct regmap_range sta32x_write_regs_range[] = {
- regmap_reg_range(STA32X_CONFA, STA32X_AUTO2),
- regmap_reg_range(STA32X_C1CFG, STA32X_FDRC2),
+ regmap_reg_range(STA32X_CONFA, STA32X_FDRC2),
};
static const struct regmap_range sta32x_read_regs_range[] = {
- regmap_reg_range(STA32X_CONFA, STA32X_AUTO2),
- regmap_reg_range(STA32X_C1CFG, STA32X_FDRC2),
+ regmap_reg_range(STA32X_CONFA, STA32X_FDRC2),
};
static const struct regmap_range sta32x_volatile_regs_range[] = {
From 70068776c49b37fe0c8f9115cec068d07375c6fb Mon Sep 17 00:00:00 2001
From: Oder Chiou <oder_chiou@realtek.com>
Date: Wed, 25 Feb 2015 17:36:13 +0800
Subject: [PATCH 16/26] ASoC: rt5677: Correct the routing paths of that after
IF1/2 DACx Mux
The patch corrects the routing paths of that after IF1/2 DACx Mux
Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/codecs/rt5677.c | 32 ++++++++++++++++----------------
1 file changed, 16 insertions(+), 16 deletions(-)
diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c
index 5d0bb8748dd1..fb9c20eace3f 100644
--- a/sound/soc/codecs/rt5677.c
+++ b/sound/soc/codecs/rt5677.c
@@ -3284,8 +3284,8 @@ static const struct snd_soc_dapm_route rt5677_dapm_routes[] = {
{ "IB45 Bypass Mux", "Bypass", "IB45 Mux" },
{ "IB45 Bypass Mux", "Pass SRC", "IB45 Mux" },
- { "IB6 Mux", "IF1 DAC 6", "IF1 DAC6" },
- { "IB6 Mux", "IF2 DAC 6", "IF2 DAC6" },
+ { "IB6 Mux", "IF1 DAC 6", "IF1 DAC6 Mux" },
+ { "IB6 Mux", "IF2 DAC 6", "IF2 DAC6 Mux" },
{ "IB6 Mux", "SLB DAC 6", "SLB DAC6" },
{ "IB6 Mux", "STO4 ADC MIX L", "Stereo4 ADC MIXL" },
{ "IB6 Mux", "IF4 DAC L", "IF4 DAC L" },
@@ -3293,8 +3293,8 @@ static const struct snd_soc_dapm_route rt5677_dapm_routes[] = {
{ "IB6 Mux", "STO2 ADC MIX L", "Stereo2 ADC MIXL" },
{ "IB6 Mux", "STO3 ADC MIX L", "Stereo3 ADC MIXL" },
- { "IB7 Mux", "IF1 DAC 7", "IF1 DAC7" },
- { "IB7 Mux", "IF2 DAC 7", "IF2 DAC7" },
+ { "IB7 Mux", "IF1 DAC 7", "IF1 DAC7 Mux" },
+ { "IB7 Mux", "IF2 DAC 7", "IF2 DAC7 Mux" },
{ "IB7 Mux", "SLB DAC 7", "SLB DAC7" },
{ "IB7 Mux", "STO4 ADC MIX R", "Stereo4 ADC MIXR" },
{ "IB7 Mux", "IF4 DAC R", "IF4 DAC R" },
@@ -3635,15 +3635,15 @@ static const struct snd_soc_dapm_route rt5677_dapm_routes[] = {
{ "DAC1 FS", NULL, "DAC1 MIXL" },
{ "DAC1 FS", NULL, "DAC1 MIXR" },
- { "DAC2 L Mux", "IF1 DAC 2", "IF1 DAC2" },
- { "DAC2 L Mux", "IF2 DAC 2", "IF2 DAC2" },
+ { "DAC2 L Mux", "IF1 DAC 2", "IF1 DAC2 Mux" },
+ { "DAC2 L Mux", "IF2 DAC 2", "IF2 DAC2 Mux" },
{ "DAC2 L Mux", "IF3 DAC L", "IF3 DAC L" },
{ "DAC2 L Mux", "IF4 DAC L", "IF4 DAC L" },
{ "DAC2 L Mux", "SLB DAC 2", "SLB DAC2" },
{ "DAC2 L Mux", "OB 2", "OutBound2" },
- { "DAC2 R Mux", "IF1 DAC 3", "IF1 DAC3" },
- { "DAC2 R Mux", "IF2 DAC 3", "IF2 DAC3" },
+ { "DAC2 R Mux", "IF1 DAC 3", "IF1 DAC3 Mux" },
+ { "DAC2 R Mux", "IF2 DAC 3", "IF2 DAC3 Mux" },
{ "DAC2 R Mux", "IF3 DAC R", "IF3 DAC R" },
{ "DAC2 R Mux", "IF4 DAC R", "IF4 DAC R" },
{ "DAC2 R Mux", "SLB DAC 3", "SLB DAC3" },
@@ -3651,29 +3651,29 @@ static const struct snd_soc_dapm_route rt5677_dapm_routes[] = {
{ "DAC2 R Mux", "Haptic Generator", "Haptic Generator" },
{ "DAC2 R Mux", "VAD ADC", "VAD ADC Mux" },
- { "DAC3 L Mux", "IF1 DAC 4", "IF1 DAC4" },
- { "DAC3 L Mux", "IF2 DAC 4", "IF2 DAC4" },
+ { "DAC3 L Mux", "IF1 DAC 4", "IF1 DAC4 Mux" },
+ { "DAC3 L Mux", "IF2 DAC 4", "IF2 DAC4 Mux" },
{ "DAC3 L Mux", "IF3 DAC L", "IF3 DAC L" },
{ "DAC3 L Mux", "IF4 DAC L", "IF4 DAC L" },
{ "DAC3 L Mux", "SLB DAC 4", "SLB DAC4" },
{ "DAC3 L Mux", "OB 4", "OutBound4" },
- { "DAC3 R Mux", "IF1 DAC 5", "IF1 DAC4" },
- { "DAC3 R Mux", "IF2 DAC 5", "IF2 DAC4" },
+ { "DAC3 R Mux", "IF1 DAC 5", "IF1 DAC5 Mux" },
+ { "DAC3 R Mux", "IF2 DAC 5", "IF2 DAC5 Mux" },
{ "DAC3 R Mux", "IF3 DAC R", "IF3 DAC R" },
{ "DAC3 R Mux", "IF4 DAC R", "IF4 DAC R" },
{ "DAC3 R Mux", "SLB DAC 5", "SLB DAC5" },
{ "DAC3 R Mux", "OB 5", "OutBound5" },
- { "DAC4 L Mux", "IF1 DAC 6", "IF1 DAC6" },
- { "DAC4 L Mux", "IF2 DAC 6", "IF2 DAC6" },
+ { "DAC4 L Mux", "IF1 DAC 6", "IF1 DAC6 Mux" },
+ { "DAC4 L Mux", "IF2 DAC 6", "IF2 DAC6 Mux" },
{ "DAC4 L Mux", "IF3 DAC L", "IF3 DAC L" },
{ "DAC4 L Mux", "IF4 DAC L", "IF4 DAC L" },
{ "DAC4 L Mux", "SLB DAC 6", "SLB DAC6" },
{ "DAC4 L Mux", "OB 6", "OutBound6" },
- { "DAC4 R Mux", "IF1 DAC 7", "IF1 DAC7" },
- { "DAC4 R Mux", "IF2 DAC 7", "IF2 DAC7" },
+ { "DAC4 R Mux", "IF1 DAC 7", "IF1 DAC7 Mux" },
+ { "DAC4 R Mux", "IF2 DAC 7", "IF2 DAC7 Mux" },
{ "DAC4 R Mux", "IF3 DAC R", "IF3 DAC R" },
{ "DAC4 R Mux", "IF4 DAC R", "IF4 DAC R" },
{ "DAC4 R Mux", "SLB DAC 7", "SLB DAC7" },
From 8af4baa7087a0ae74c6ee29d4d979a60e14b119e Mon Sep 17 00:00:00 2001
From: =?UTF-8?q?Thomas=20Niederpr=C3=BCm?= <niederp@physik.uni-kl.de>
Date: Sat, 21 Feb 2015 18:11:29 +0100
Subject: [PATCH 17/26] ASoC: OMAP: mcbsp: Fix CLKX and CLKR pinmux when used
as inputs
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit
This patch fixes faulty behaviour in a setup where the input clock for the
SRG is fed through the CLKR/CLKX pin but the McBSP is configured to be
master (SND_SOC_DAIFMT_CBS_CFS). In that case of course CLKR/CLKX must
not be configured as output pin. Otherwise the input clock is messed up
horribly.
This patch makes it possible to use the CLKR/CLKX pin rather than CLKS to
inject a reference clock in setups where McBSP is master and not both
rx and tx are used. However for this to work it has to be ensured that
set_dai_sysclk() is called after set_dai_fmt().
This was tested on a beagleboard-xm using McBSP1 to drive a i2s DAC through
the tx lines (CLKX,FSX,DX). Using this patch the CLKR pin is used to inject
an external reference clock.
Signed-off-by: Thomas Niederprüm <niederp@physik.uni-kl.de>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/omap/omap-mcbsp.c | 11 +++++++++++
1 file changed, 11 insertions(+)
diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c
index c7eb9dd67f60..fd99d89de6a8 100644
--- a/sound/soc/omap/omap-mcbsp.c
+++ b/sound/soc/omap/omap-mcbsp.c
@@ -530,8 +530,19 @@ static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
case OMAP_MCBSP_SYSCLK_CLKX_EXT:
regs->srgr2 |= CLKSM;
+ regs->pcr0 |= SCLKME;
+ /*
+ * If McBSP is master but yet the CLKX/CLKR pin drives the SRG,
+ * disable output on those pins. This enables to inject the
+ * reference clock through CLKX/CLKR. For this to work
+ * set_dai_sysclk() _needs_ to be called after set_dai_fmt().
+ */
+ regs->pcr0 &= ~CLKXM;
+ break;
case OMAP_MCBSP_SYSCLK_CLKR_EXT:
regs->pcr0 |= SCLKME;
+ /* Disable ouput on CLKR pin in master mode */
+ regs->pcr0 &= ~CLKRM;
break;
default:
err = -ENODEV;
From f2b14c0bc510c6a8f67a4f36049deefe5d99a537 Mon Sep 17 00:00:00 2001
From: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Date: Fri, 27 Feb 2015 09:39:32 +0900
Subject: [PATCH 18/26] ALSA: oxfw: fix a condition and return code in
start_stream()
The amdtp_stream_wait_callback() doesn't return minus value and
the return code is not for error code.
This commit fixes with a propper condition and an error code.
Fixes: f3699e2c7745 ('ALSA: oxfw: Change the way to start stream')
Reported-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Cc: <stable@vger.kernel.org> # 3.19+
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/firewire/oxfw/oxfw-stream.c | 5 +++--
1 file changed, 3 insertions(+), 2 deletions(-)
diff --git a/sound/firewire/oxfw/oxfw-stream.c b/sound/firewire/oxfw/oxfw-stream.c
index 29ccb3637164..e6757cd85724 100644
--- a/sound/firewire/oxfw/oxfw-stream.c
+++ b/sound/firewire/oxfw/oxfw-stream.c
@@ -171,9 +171,10 @@ static int start_stream(struct snd_oxfw *oxfw, struct amdtp_stream *stream,
}
/* Wait first packet */
- err = amdtp_stream_wait_callback(stream, CALLBACK_TIMEOUT);
- if (err < 0)
+ if (!amdtp_stream_wait_callback(stream, CALLBACK_TIMEOUT)) {
stop_stream(oxfw, stream);
+ err = -ETIMEDOUT;
+ }
end:
return err;
}
From 8cdebf71098c07168ef6335e2f1f35d85dbe3049 Mon Sep 17 00:00:00 2001
From: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Date: Sun, 1 Mar 2015 18:12:16 +0900
Subject: [PATCH 19/26] ALSA: dice: fix wrong offsets for Dice interface
For received packet stream, the offset of 'RX_SEQ_START' locates after
the offset of 'RX_NUMBER_MIDI', although current macro and proc output
includes wrong offsets.
Fortunately, this bug doesn't affect streaming functionality because
these macro is not used.
This commit fixes these wrong macro and outputs.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/firewire/dice/dice-interface.h | 18 +++++++++---------
sound/firewire/dice/dice-proc.c | 4 ++--
2 files changed, 11 insertions(+), 11 deletions(-)
diff --git a/sound/firewire/dice/dice-interface.h b/sound/firewire/dice/dice-interface.h
index 27b044f84c81..de7602bd69b5 100644
--- a/sound/firewire/dice/dice-interface.h
+++ b/sound/firewire/dice/dice-interface.h
@@ -298,24 +298,24 @@
*/
#define RX_ISOCHRONOUS 0x008
-/*
- * Index of first quadlet to be interpreted; read/write. If > 0, that many
- * quadlets at the beginning of each data block will be ignored, and all the
- * audio and MIDI quadlets will follow.
- */
-#define RX_SEQ_START 0x00c
-
/*
* The number of audio channels; read-only. There will be one quadlet per
* channel.
*/
-#define RX_NUMBER_AUDIO 0x010
+#define RX_NUMBER_AUDIO 0x00c
/*
* The number of MIDI ports, 0-8; read-only. If > 0, there will be one
* additional quadlet in each data block, following the audio quadlets.
*/
-#define RX_NUMBER_MIDI 0x014
+#define RX_NUMBER_MIDI 0x010
+
+/*
+ * Index of first quadlet to be interpreted; read/write. If > 0, that many
+ * quadlets at the beginning of each data block will be ignored, and all the
+ * audio and MIDI quadlets will follow.
+ */
+#define RX_SEQ_START 0x014
/*
* Names of all audio channels; read-only. Quadlets are byte-swapped. Names
diff --git a/sound/firewire/dice/dice-proc.c b/sound/firewire/dice/dice-proc.c
index f5c1d1bced59..ecfe20fd4de5 100644
--- a/sound/firewire/dice/dice-proc.c
+++ b/sound/firewire/dice/dice-proc.c
@@ -99,9 +99,9 @@ static void dice_proc_read(struct snd_info_entry *entry,
} tx;
struct {
u32 iso;
- u32 seq_start;
u32 number_audio;
u32 number_midi;
+ u32 seq_start;
char names[RX_NAMES_SIZE];
u32 ac3_caps;
u32 ac3_enable;
@@ -204,10 +204,10 @@ static void dice_proc_read(struct snd_info_entry *entry,
break;
snd_iprintf(buffer, "rx %u:\n", stream);
snd_iprintf(buffer, " iso channel: %d\n", (int)buf.rx.iso);
- snd_iprintf(buffer, " sequence start: %u\n", buf.rx.seq_start);
snd_iprintf(buffer, " audio channels: %u\n",
buf.rx.number_audio);
snd_iprintf(buffer, " midi ports: %u\n", buf.rx.number_midi);
+ snd_iprintf(buffer, " sequence start: %u\n", buf.rx.seq_start);
if (quadlets >= 68) {
dice_proc_fixup_string(buf.rx.names, RX_NAMES_SIZE);
snd_iprintf(buffer, " names: %s\n", buf.rx.names);
From d7a6fe015b2abe33565538a3faf757e095e094e7 Mon Sep 17 00:00:00 2001
From: Alexandre Belloni <alexandre.belloni@free-electrons.com>
Date: Tue, 6 Jan 2015 12:14:32 +0100
Subject: [PATCH 20/26] ASoC: sam9g20_wm8731: drop machine_is_xxx
Atmel based boards can now only be used with device tree. Drop non DT
initialization.
Signed-off-by: Alexandre Belloni <alexandre.belloni@free-electrons.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/atmel/sam9g20_wm8731.c | 64 +++++++++++++++-----------------
1 file changed, 29 insertions(+), 35 deletions(-)
diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c
index f5ad214663f9..8de836165cf2 100644
--- a/sound/soc/atmel/sam9g20_wm8731.c
+++ b/sound/soc/atmel/sam9g20_wm8731.c
@@ -46,8 +46,6 @@
#include <sound/pcm_params.h>
#include <sound/soc.h>
-#include <asm/mach-types.h>
-
#include "../codecs/wm8731.h"
#include "atmel-pcm.h"
#include "atmel_ssc_dai.h"
@@ -171,9 +169,7 @@ static int at91sam9g20ek_audio_probe(struct platform_device *pdev)
int ret;
if (!np) {
- if (!(machine_is_at91sam9g20ek() ||
- machine_is_at91sam9g20ek_2mmc()))
- return -ENODEV;
+ return -ENODEV;
}
ret = atmel_ssc_set_audio(0);
@@ -210,39 +206,37 @@ static int at91sam9g20ek_audio_probe(struct platform_device *pdev)
card->dev = &pdev->dev;
/* Parse device node info */
- if (np) {
- ret = snd_soc_of_parse_card_name(card, "atmel,model");
- if (ret)
- goto err;
+ ret = snd_soc_of_parse_card_name(card, "atmel,model");
+ if (ret)
+ goto err;
- ret = snd_soc_of_parse_audio_routing(card,
- "atmel,audio-routing");
- if (ret)
- goto err;
+ ret = snd_soc_of_parse_audio_routing(card,
+ "atmel,audio-routing");
+ if (ret)
+ goto err;
- /* Parse codec info */
- at91sam9g20ek_dai.codec_name = NULL;
- codec_np = of_parse_phandle(np, "atmel,audio-codec", 0);
- if (!codec_np) {
- dev_err(&pdev->dev, "codec info missing\n");
- return -EINVAL;
- }
- at91sam9g20ek_dai.codec_of_node = codec_np;
-
- /* Parse dai and platform info */
- at91sam9g20ek_dai.cpu_dai_name = NULL;
- at91sam9g20ek_dai.platform_name = NULL;
- cpu_np = of_parse_phandle(np, "atmel,ssc-controller", 0);
- if (!cpu_np) {
- dev_err(&pdev->dev, "dai and pcm info missing\n");
- return -EINVAL;
- }
- at91sam9g20ek_dai.cpu_of_node = cpu_np;
- at91sam9g20ek_dai.platform_of_node = cpu_np;
-
- of_node_put(codec_np);
- of_node_put(cpu_np);
+ /* Parse codec info */
+ at91sam9g20ek_dai.codec_name = NULL;
+ codec_np = of_parse_phandle(np, "atmel,audio-codec", 0);
+ if (!codec_np) {
+ dev_err(&pdev->dev, "codec info missing\n");
+ return -EINVAL;
}
+ at91sam9g20ek_dai.codec_of_node = codec_np;
+
+ /* Parse dai and platform info */
+ at91sam9g20ek_dai.cpu_dai_name = NULL;
+ at91sam9g20ek_dai.platform_name = NULL;
+ cpu_np = of_parse_phandle(np, "atmel,ssc-controller", 0);
+ if (!cpu_np) {
+ dev_err(&pdev->dev, "dai and pcm info missing\n");
+ return -EINVAL;
+ }
+ at91sam9g20ek_dai.cpu_of_node = cpu_np;
+ at91sam9g20ek_dai.platform_of_node = cpu_np;
+
+ of_node_put(codec_np);
+ of_node_put(cpu_np);
ret = snd_soc_register_card(card);
if (ret) {
From 31f3032c1a5504259f6fa8e0c7f8d2d3e2f5db48 Mon Sep 17 00:00:00 2001
From: Vishal Thanki <vishalthanki@gmail.com>
Date: Tue, 3 Mar 2015 18:59:00 +0530
Subject: [PATCH 21/26] ASoC: simple-card: Add a NULL pointer check in
asoc_simple_card_dai_link_of
Make sure devm_kzalloc() succeeds.
Signed-off-by: Vishal Thanki <vishalthanki@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/generic/simple-card.c | 5 +++++
1 file changed, 5 insertions(+)
diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c
index f7c6734bd5da..fb550b5869d2 100644
--- a/sound/soc/generic/simple-card.c
+++ b/sound/soc/generic/simple-card.c
@@ -372,6 +372,11 @@ static int asoc_simple_card_dai_link_of(struct device_node *node,
strlen(dai_link->cpu_dai_name) +
strlen(dai_link->codec_dai_name) + 2,
GFP_KERNEL);
+ if (!name) {
+ ret = -ENOMEM;
+ goto dai_link_of_err;
+ }
+
sprintf(name, "%s-%s", dai_link->cpu_dai_name,
dai_link->codec_dai_name);
dai_link->name = dai_link->stream_name = name;
From d51199a83a2cf82a291d19ee852c44caa511427d Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@ti.com>
Date: Tue, 3 Mar 2015 13:38:14 +0200
Subject: [PATCH 22/26] ASoC: omap-pcm: Correct dma mask
DMA_BIT_MASK of 64 is not valid dma address mask for OMAPs, it should be
set to 32.
The 64 was introduced by commit (in 2009):
a152ff24b978 ASoC: OMAP: Make DMA 64 aligned
But the dma_mask and coherent_dma_mask can not be used to specify alignment.
Fixes: a152ff24b978 (ASoC: OMAP: Make DMA 64 aligned)
Reported-by: Grygorii Strashko <Grygorii.Strashko@linaro.org>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
---
sound/soc/omap/omap-pcm.c | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c
index f4b05bc23e4b..1343ecbf0bd5 100644
--- a/sound/soc/omap/omap-pcm.c
+++ b/sound/soc/omap/omap-pcm.c
@@ -201,7 +201,7 @@ static int omap_pcm_new(struct snd_soc_pcm_runtime *rtd)
struct snd_pcm *pcm = rtd->pcm;
int ret;
- ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(64));
+ ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(32));
if (ret)
return ret;
From 096a020a9ef5c947577d3b57199bfc9b7e686b49 Mon Sep 17 00:00:00 2001
From: Dan Carpenter <dan.carpenter@oracle.com>
Date: Thu, 5 Mar 2015 14:26:37 +0300
Subject: [PATCH 23/26] ALSA: msnd: add some missing curly braces
There were some curly braces intended here.
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/isa/msnd/msnd_pinnacle_mixer.c | 3 ++-
1 file changed, 2 insertions(+), 1 deletion(-)
diff --git a/sound/isa/msnd/msnd_pinnacle_mixer.c b/sound/isa/msnd/msnd_pinnacle_mixer.c
index 17e49a071af4..b408540798c1 100644
--- a/sound/isa/msnd/msnd_pinnacle_mixer.c
+++ b/sound/isa/msnd/msnd_pinnacle_mixer.c
@@ -306,11 +306,12 @@ int snd_msndmix_new(struct snd_card *card)
spin_lock_init(&chip->mixer_lock);
strcpy(card->mixername, "MSND Pinnacle Mixer");
- for (idx = 0; idx < ARRAY_SIZE(snd_msnd_controls); idx++)
+ for (idx = 0; idx < ARRAY_SIZE(snd_msnd_controls); idx++) {
err = snd_ctl_add(card,
snd_ctl_new1(snd_msnd_controls + idx, chip));
if (err < 0)
return err;
+ }
return 0;
}
From f44f07cf3910f84b15b2a78c4933d5946bf409cf Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 5 Mar 2015 13:03:28 +0100
Subject: [PATCH 24/26] ALSA: line6: Clamp values correctly
The usages of clamp() macro in sound/usb/line6/playback.c are just
wrong, the low and high values are swapped.
Reported-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/usb/line6/playback.c | 6 +++---
1 file changed, 3 insertions(+), 3 deletions(-)
diff --git a/sound/usb/line6/playback.c b/sound/usb/line6/playback.c
index 05dee690f487..97ed593f6010 100644
--- a/sound/usb/line6/playback.c
+++ b/sound/usb/line6/playback.c
@@ -39,7 +39,7 @@ static void change_volume(struct urb *urb_out, int volume[],
for (; p < buf_end; ++p) {
short pv = le16_to_cpu(*p);
int val = (pv * volume[chn & 1]) >> 8;
- pv = clamp(val, 0x7fff, -0x8000);
+ pv = clamp(val, -0x8000, 0x7fff);
*p = cpu_to_le16(pv);
++chn;
}
@@ -54,7 +54,7 @@ static void change_volume(struct urb *urb_out, int volume[],
val = p[0] + (p[1] << 8) + ((signed char)p[2] << 16);
val = (val * volume[chn & 1]) >> 8;
- val = clamp(val, 0x7fffff, -0x800000);
+ val = clamp(val, -0x800000, 0x7fffff);
p[0] = val;
p[1] = val >> 8;
p[2] = val >> 16;
@@ -126,7 +126,7 @@ static void add_monitor_signal(struct urb *urb_out, unsigned char *signal,
short pov = le16_to_cpu(*po);
short piv = le16_to_cpu(*pi);
int val = pov + ((piv * volume) >> 8);
- pov = clamp(val, 0x7fff, -0x8000);
+ pov = clamp(val, -0x8000, 0x7fff);
*po = cpu_to_le16(pov);
}
}
From d124380674b58f62d0ef974630d74d67bb8afeb0 Mon Sep 17 00:00:00 2001
From: Dan Carpenter <dan.carpenter@oracle.com>
Date: Thu, 5 Mar 2015 20:49:06 +0300
Subject: [PATCH 25/26] ALSA: opl3: small array underflow
There is a missing lower bound check on "pitchbend" so it means we can
read up to 6 elements before the start of the opl3_note_table[] array.
Thanks to Clemens Ladisch for his help with this patch.
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/drivers/opl3/opl3_midi.c | 2 ++
1 file changed, 2 insertions(+)
diff --git a/sound/drivers/opl3/opl3_midi.c b/sound/drivers/opl3/opl3_midi.c
index f62780ed64ad..7821b07415a7 100644
--- a/sound/drivers/opl3/opl3_midi.c
+++ b/sound/drivers/opl3/opl3_midi.c
@@ -105,6 +105,8 @@ static void snd_opl3_calc_pitch(unsigned char *fnum, unsigned char *blocknum,
int pitchbend = chan->midi_pitchbend;
int segment;
+ if (pitchbend < -0x2000)
+ pitchbend = -0x2000;
if (pitchbend > 0x1FFF)
pitchbend = 0x1FFF;
From 70658b99490dd86cfdbf4fca117bbe2ef9a80d03 Mon Sep 17 00:00:00 2001
From: Hui Wang <hui.wang@canonical.com>
Date: Fri, 6 Mar 2015 14:03:57 +0800
Subject: [PATCH 26/26] ALSA: hda - One more Dell macine needs
DELL1_MIC_NO_PRESENCE quirk
Cc: <stable@vger.kernel.org>
BugLink: https://bugs.launchpad.net/bugs/1428947
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/pci/hda/patch_realtek.c | 7 +++++++
1 file changed, 7 insertions(+)
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index b2b24a8b3dac..526398a4a442 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -5209,6 +5209,13 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = {
{0x17, 0x40000000},
{0x1d, 0x40700001},
{0x21, 0x02211040}),
+ SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE,
+ ALC255_STANDARD_PINS,
+ {0x12, 0x90a60170},
+ {0x14, 0x90170140},
+ {0x17, 0x40000000},
+ {0x1d, 0x40700001},
+ {0x21, 0x02211050}),
SND_HDA_PIN_QUIRK(0x10ec0280, 0x103c, "HP", ALC280_FIXUP_HP_GPIO4,
{0x12, 0x90a60130},
{0x13, 0x40000000},