From a28287925555c93984115d37a1a25315ea369764 Mon Sep 17 00:00:00 2001
From: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Date: Wed, 19 Jan 2011 12:55:28 +0000
Subject: ASoC: WM8995: Fix incorrect use of snd_soc_update_bits()
In the wm8995_set_tristate() function when updating the register
bits use the value and not the register index as the value argument.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
---
sound/soc/codecs/wm8995.c | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
(limited to 'sound')
diff --git a/sound/soc/codecs/wm8995.c b/sound/soc/codecs/wm8995.c
index 6045cbde492b..608c84c5aa8e 100644
--- a/sound/soc/codecs/wm8995.c
+++ b/sound/soc/codecs/wm8995.c
@@ -1223,7 +1223,7 @@ static int wm8995_set_tristate(struct snd_soc_dai *codec_dai, int tristate)
else
val = 0;
- return snd_soc_update_bits(codec, reg, mask, reg);
+ return snd_soc_update_bits(codec, reg, mask, val);
}
/* The size in bits of the FLL divide multiplied by 10
--
cgit v1.2.2
From 78b3fb46753872fc79bffecc8d50355a8622b39b Mon Sep 17 00:00:00 2001
From: Qiao Zhou <zhouqiao@marvell.com>
Date: Wed, 19 Jan 2011 19:10:47 +0800
Subject: ASoC: WM8994: fix wrong value in tristate function
fix wrong value in wm8994_set_tristate func. when updating reg bits,
it should use "value", not "reg".
Signed-off-by: Qiao Zhou <zhouqiao@marvell.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
---
sound/soc/codecs/wm8994.c | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
(limited to 'sound')
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index 247a6a99feb8..3351f77607b3 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -2386,7 +2386,7 @@ static int wm8994_set_tristate(struct snd_soc_dai *codec_dai, int tristate)
else
val = 0;
- return snd_soc_update_bits(codec, reg, mask, reg);
+ return snd_soc_update_bits(codec, reg, mask, val);
}
#define WM8994_RATES SNDRV_PCM_RATE_8000_96000
--
cgit v1.2.2
From dc5a460a1bfa44273653700e33d4e7051194cbfd Mon Sep 17 00:00:00 2001
From: "Rajashekhara, Sudhakar" <sudhakar.raj@ti.com>
Date: Fri, 21 Jan 2011 20:10:01 +0530
Subject: ASoC: da8xx/omap-l1xx: match codec_name with i2c ids
The codec_name entry for da8xx evm in sound/soc/davinci/davinci-evm.c
is not matching with the i2c ids in the board file. Without this fix the
soundcard does not get detected on da850/omap-l138/am18x evm.
Signed-off-by: Rajashekhara, Sudhakar <sudhakar.raj@ti.com>
Tested-by: Dan Sharon <dansharon@nanometrics.ca>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org (for 2.6.37)
---
sound/soc/davinci/davinci-evm.c | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
(limited to 'sound')
diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c
index 0c2d6bacc681..b36f0b39b090 100644
--- a/sound/soc/davinci/davinci-evm.c
+++ b/sound/soc/davinci/davinci-evm.c
@@ -223,7 +223,7 @@ static struct snd_soc_dai_link da8xx_evm_dai = {
.stream_name = "AIC3X",
.cpu_dai_name= "davinci-mcasp.0",
.codec_dai_name = "tlv320aic3x-hifi",
- .codec_name = "tlv320aic3x-codec.0-001a",
+ .codec_name = "tlv320aic3x-codec.1-0018",
.platform_name = "davinci-pcm-audio",
.init = evm_aic3x_init,
.ops = &evm_ops,
--
cgit v1.2.2
From 20a4e7fc7e213365ea3771d7bf1e10a6bab853be Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Fri, 21 Jan 2011 12:47:33 +0000
Subject: ASoC: Handle low measured DC offsets for wm_hubs devices
The DC servo codes are actually signed numbers so need to be treated as
such.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Cc: stable@kernel.org
---
sound/soc/codecs/wm_hubs.c | 15 +++++++--------
1 file changed, 7 insertions(+), 8 deletions(-)
(limited to 'sound')
diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c
index c466982eed23..613df5db0b32 100644
--- a/sound/soc/codecs/wm_hubs.c
+++ b/sound/soc/codecs/wm_hubs.c
@@ -91,6 +91,7 @@ static void wait_for_dc_servo(struct snd_soc_codec *codec, unsigned int op)
static void calibrate_dc_servo(struct snd_soc_codec *codec)
{
struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec);
+ s8 offset;
u16 reg, reg_l, reg_r, dcs_cfg;
/* If we're using a digital only path and have a previously
@@ -149,16 +150,14 @@ static void calibrate_dc_servo(struct snd_soc_codec *codec)
hubs->dcs_codes);
/* HPOUT1L */
- if (reg_l + hubs->dcs_codes > 0 &&
- reg_l + hubs->dcs_codes < 0xff)
- reg_l += hubs->dcs_codes;
- dcs_cfg = reg_l << WM8993_DCS_DAC_WR_VAL_1_SHIFT;
+ offset = reg_l;
+ offset += hubs->dcs_codes;
+ dcs_cfg = (u8)offset << WM8993_DCS_DAC_WR_VAL_1_SHIFT;
/* HPOUT1R */
- if (reg_r + hubs->dcs_codes > 0 &&
- reg_r + hubs->dcs_codes < 0xff)
- reg_r += hubs->dcs_codes;
- dcs_cfg |= reg_r;
+ offset = reg_r;
+ offset += hubs->dcs_codes;
+ dcs_cfg |= (u8)offset;
dev_dbg(codec->dev, "DCS result: %x\n", dcs_cfg);
--
cgit v1.2.2
From 233d84c46c2253d13e10b42d88c14748fbb67a98 Mon Sep 17 00:00:00 2001
From: Jesper Juhl <jj@chaosbits.net>
Date: Thu, 20 Jan 2011 22:37:43 +0100
Subject: ALSA: Xonar, CS43xx: Don't overrun static array
'cs4398_regs' in 'struct xonar_cs43xx' is an array of 'u8' with a size of
8. So, this code in sound/pci/oxygen/xonar_cs43xx.c::dump_d1_registers()
for (i = 2; i <= 8; ++i)
snd_iprintf(buffer, " %02x", data->cs4398_regs[i]);
will overrun the array when 'i == 8'.
I guess that what's needed to fix it is the trivial patch below, but I
must admit that I have no idea about this code, so I may very well be
wrong. Additionally, I have no way to actually test this, so all I know is
that the below compiles. Someone who actually knows this code should take
a look before anything is comitted - consider the below (not much more
than) a bug report.
Signed-off-by: Jesper Juhl <jj@chaosbits.net>
Acked-by: Clemens Ladisch <clemens@ladisch.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/pci/oxygen/xonar_cs43xx.c | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
(limited to 'sound')
diff --git a/sound/pci/oxygen/xonar_cs43xx.c b/sound/pci/oxygen/xonar_cs43xx.c
index 9f72d424969c..252719101c42 100644
--- a/sound/pci/oxygen/xonar_cs43xx.c
+++ b/sound/pci/oxygen/xonar_cs43xx.c
@@ -392,7 +392,7 @@ static void dump_d1_registers(struct oxygen *chip,
unsigned int i;
snd_iprintf(buffer, "\nCS4398: 7?");
- for (i = 2; i <= 8; ++i)
+ for (i = 2; i < 8; ++i)
snd_iprintf(buffer, " %02x", data->cs4398_regs[i]);
snd_iprintf(buffer, "\n");
dump_cs4362a_registers(data, buffer);
--
cgit v1.2.2
From c9ba374d24882c04e7cc000d8cf3b0fe56511b84 Mon Sep 17 00:00:00 2001
From: Andreas Mohr <andi@lisas.de>
Date: Tue, 25 Jan 2011 06:46:31 +0100
Subject: ALSA: azt3328 - fix broken AZF_FMT_XLATE macro
Cleanly revert to non-macro implementation of
snd_azf3328_codec_setfmt(), to fix last-minute functionality breakage
induced by following checkpatch.pl recommendations without giving them
their due full share of thought ("revolting computer, ensuing PEBKAC").
I would like to thank Jiri Slaby for his very timely (in -rc1 even)
and unexpected (uncommon hardware) "recognition of the dangerous situation"
due to his very commendable static parser use. :)
Reported-by: Jiri Slaby <jslaby@suse.cz>
Signed-off-by: Andreas Mohr <andi@lisas.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/pci/azt3328.c | 38 ++++++++++++++++----------------------
1 file changed, 16 insertions(+), 22 deletions(-)
(limited to 'sound')
diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c
index 6117595fc075..573594bf3225 100644
--- a/sound/pci/azt3328.c
+++ b/sound/pci/azt3328.c
@@ -979,31 +979,25 @@ snd_azf3328_codec_setfmt(struct snd_azf3328_codec_data *codec,
snd_azf3328_dbgcallenter();
switch (bitrate) {
-#define AZF_FMT_XLATE(in_freq, out_bits) \
- do { \
- case AZF_FREQ_ ## in_freq: \
- freq = SOUNDFORMAT_FREQ_ ## out_bits; \
- break; \
- } while (0);
- AZF_FMT_XLATE(4000, SUSPECTED_4000)
- AZF_FMT_XLATE(4800, SUSPECTED_4800)
- /* the AZF3328 names it "5510" for some strange reason: */
- AZF_FMT_XLATE(5512, 5510)
- AZF_FMT_XLATE(6620, 6620)
- AZF_FMT_XLATE(8000, 8000)
- AZF_FMT_XLATE(9600, 9600)
- AZF_FMT_XLATE(11025, 11025)
- AZF_FMT_XLATE(13240, SUSPECTED_13240)
- AZF_FMT_XLATE(16000, 16000)
- AZF_FMT_XLATE(22050, 22050)
- AZF_FMT_XLATE(32000, 32000)
+ case AZF_FREQ_4000: freq = SOUNDFORMAT_FREQ_SUSPECTED_4000; break;
+ case AZF_FREQ_4800: freq = SOUNDFORMAT_FREQ_SUSPECTED_4800; break;
+ case AZF_FREQ_5512:
+ /* the AZF3328 names it "5510" for some strange reason */
+ freq = SOUNDFORMAT_FREQ_5510; break;
+ case AZF_FREQ_6620: freq = SOUNDFORMAT_FREQ_6620; break;
+ case AZF_FREQ_8000: freq = SOUNDFORMAT_FREQ_8000; break;
+ case AZF_FREQ_9600: freq = SOUNDFORMAT_FREQ_9600; break;
+ case AZF_FREQ_11025: freq = SOUNDFORMAT_FREQ_11025; break;
+ case AZF_FREQ_13240: freq = SOUNDFORMAT_FREQ_SUSPECTED_13240; break;
+ case AZF_FREQ_16000: freq = SOUNDFORMAT_FREQ_16000; break;
+ case AZF_FREQ_22050: freq = SOUNDFORMAT_FREQ_22050; break;
+ case AZF_FREQ_32000: freq = SOUNDFORMAT_FREQ_32000; break;
default:
snd_printk(KERN_WARNING "unknown bitrate %d, assuming 44.1kHz!\n", bitrate);
/* fall-through */
- AZF_FMT_XLATE(44100, 44100)
- AZF_FMT_XLATE(48000, 48000)
- AZF_FMT_XLATE(66200, SUSPECTED_66200)
-#undef AZF_FMT_XLATE
+ case AZF_FREQ_44100: freq = SOUNDFORMAT_FREQ_44100; break;
+ case AZF_FREQ_48000: freq = SOUNDFORMAT_FREQ_48000; break;
+ case AZF_FREQ_66200: freq = SOUNDFORMAT_FREQ_SUSPECTED_66200; break;
}
/* val = 0xff07; 3m27.993s (65301Hz; -> 64000Hz???) hmm, 66120, 65967, 66123 */
/* val = 0xff09; 17m15.098s (13123,478Hz; -> 12000Hz???) hmm, 13237.2Hz? */
--
cgit v1.2.2
From 81d7da5404aad930a4e4f6111e4f16b752183018 Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Mon, 24 Jan 2011 22:09:22 +0100
Subject: ASoC: Fix codec device id format used by some dai_links
The id part of an I2C device name is created with the "%d-%04x" format string.
So for example for an I2C device which is connected to the adapter with the id 0
and has its address set to 0x1a the id part of the devices name would be
"0-001a".
Currently some sound board drivers have the id part the codec_name field of
their dai_link structures set as if it had been created by a "%d-0x%x" format
string. For example "0-0x1a" instead of "0-001a".
As a result there is no match between the codec device and the dai_link and no
sound card is instantiated.
This patch fixes it.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
---
sound/soc/atmel/snd-soc-afeb9260.c | 2 +-
sound/soc/blackfin/bf5xx-ssm2602.c | 2 +-
sound/soc/samsung/neo1973_gta02_wm8753.c | 4 ++--
sound/soc/samsung/neo1973_wm8753.c | 4 ++--
sound/soc/samsung/s3c24xx_simtec_hermes.c | 2 +-
sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c | 2 +-
6 files changed, 8 insertions(+), 8 deletions(-)
(limited to 'sound')
diff --git a/sound/soc/atmel/snd-soc-afeb9260.c b/sound/soc/atmel/snd-soc-afeb9260.c
index da2208e06b0d..5e4d499d8434 100644
--- a/sound/soc/atmel/snd-soc-afeb9260.c
+++ b/sound/soc/atmel/snd-soc-afeb9260.c
@@ -129,7 +129,7 @@ static struct snd_soc_dai_link afeb9260_dai = {
.cpu_dai_name = "atmel-ssc-dai.0",
.codec_dai_name = "tlv320aic23-hifi",
.platform_name = "atmel_pcm-audio",
- .codec_name = "tlv320aic23-codec.0-0x1a",
+ .codec_name = "tlv320aic23-codec.0-001a",
.init = afeb9260_tlv320aic23_init,
.ops = &afeb9260_ops,
};
diff --git a/sound/soc/blackfin/bf5xx-ssm2602.c b/sound/soc/blackfin/bf5xx-ssm2602.c
index e902b24c1856..ad28663f5bbd 100644
--- a/sound/soc/blackfin/bf5xx-ssm2602.c
+++ b/sound/soc/blackfin/bf5xx-ssm2602.c
@@ -119,7 +119,7 @@ static struct snd_soc_dai_link bf5xx_ssm2602_dai = {
.cpu_dai_name = "bf5xx-i2s",
.codec_dai_name = "ssm2602-hifi",
.platform_name = "bf5xx-pcm-audio",
- .codec_name = "ssm2602-codec.0-0x1b",
+ .codec_name = "ssm2602-codec.0-001b",
.ops = &bf5xx_ssm2602_ops,
};
diff --git a/sound/soc/samsung/neo1973_gta02_wm8753.c b/sound/soc/samsung/neo1973_gta02_wm8753.c
index 3eec610c10f9..9e05e10b86a6 100644
--- a/sound/soc/samsung/neo1973_gta02_wm8753.c
+++ b/sound/soc/samsung/neo1973_gta02_wm8753.c
@@ -401,7 +401,7 @@ static struct snd_soc_dai_link neo1973_gta02_dai[] = {
.codec_dai_name = "wm8753-hifi",
.init = neo1973_gta02_wm8753_init,
.platform_name = "samsung-audio",
- .codec_name = "wm8753-codec.0-0x1a",
+ .codec_name = "wm8753-codec.0-001a",
.ops = &neo1973_gta02_hifi_ops,
},
{ /* Voice via BT */
@@ -410,7 +410,7 @@ static struct snd_soc_dai_link neo1973_gta02_dai[] = {
.cpu_dai_name = "bluetooth-dai",
.codec_dai_name = "wm8753-voice",
.ops = &neo1973_gta02_voice_ops,
- .codec_name = "wm8753-codec.0-0x1a",
+ .codec_name = "wm8753-codec.0-001a",
.platform_name = "samsung-audio",
},
};
diff --git a/sound/soc/samsung/neo1973_wm8753.c b/sound/soc/samsung/neo1973_wm8753.c
index c7a24514beb5..cf69e146890b 100644
--- a/sound/soc/samsung/neo1973_wm8753.c
+++ b/sound/soc/samsung/neo1973_wm8753.c
@@ -561,7 +561,7 @@ static struct snd_soc_dai_link neo1973_dai[] = {
.platform_name = "samsung-audio",
.cpu_dai_name = "s3c24xx-i2s",
.codec_dai_name = "wm8753-hifi",
- .codec_name = "wm8753-codec.0-0x1a",
+ .codec_name = "wm8753-codec.0-001a",
.init = neo1973_wm8753_init,
.ops = &neo1973_hifi_ops,
},
@@ -571,7 +571,7 @@ static struct snd_soc_dai_link neo1973_dai[] = {
.platform_name = "samsung-audio",
.cpu_dai_name = "bluetooth-dai",
.codec_dai_name = "wm8753-voice",
- .codec_name = "wm8753-codec.0-0x1a",
+ .codec_name = "wm8753-codec.0-001a",
.ops = &neo1973_voice_ops,
},
};
diff --git a/sound/soc/samsung/s3c24xx_simtec_hermes.c b/sound/soc/samsung/s3c24xx_simtec_hermes.c
index bb4292e3596c..287a97164ab4 100644
--- a/sound/soc/samsung/s3c24xx_simtec_hermes.c
+++ b/sound/soc/samsung/s3c24xx_simtec_hermes.c
@@ -94,7 +94,7 @@ static int simtec_hermes_init(struct snd_soc_pcm_runtime *rtd)
static struct snd_soc_dai_link simtec_dai_aic33 = {
.name = "tlv320aic33",
.stream_name = "TLV320AIC33",
- .codec_name = "tlv320aic3x-codec.0-0x1a",
+ .codec_name = "tlv320aic3x-codec.0-001a",
.cpu_dai_name = "s3c24xx-i2s",
.codec_dai_name = "tlv320aic3x-hifi",
.platform_name = "samsung-audio",
diff --git a/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c b/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c
index fbba4e367729..d2b14ba6f9e3 100644
--- a/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c
+++ b/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c
@@ -85,7 +85,7 @@ static int simtec_tlv320aic23_init(struct snd_soc_pcm_runtime *rtd)
static struct snd_soc_dai_link simtec_dai_aic23 = {
.name = "tlv320aic23",
.stream_name = "TLV320AIC23",
- .codec_name = "tlv320aic3x-codec.0-0x1a",
+ .codec_name = "tlv320aic3x-codec.0-001a",
.cpu_dai_name = "s3c24xx-i2s",
.codec_dai_name = "tlv320aic3x-hifi",
.platform_name = "samsung-audio",
--
cgit v1.2.2
From 518aa59f6e45b3c90b849187ae1d56757d074b92 Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Mon, 24 Jan 2011 22:12:42 +0100
Subject: ASoC: Samsung: Fix outdated cpu_dai_name for s3c24xx i2s
During the multi-component patch the s3c24xx i2s driver was renamed from
"s3c24xx-i2s" to "s3c24xx-iis", while the sound board drivers were not
updated to reflect this change as well.
As a result there is no match between the dai_link and the i2s driver and no
sound card is instantiated.
This patch fixes the problem by updating the sound board drivers to use
"s3c24xx-iis" for the cpu_dai_name.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
---
sound/soc/samsung/neo1973_gta02_wm8753.c | 2 +-
sound/soc/samsung/neo1973_wm8753.c | 2 +-
sound/soc/samsung/s3c24xx_simtec_hermes.c | 2 +-
sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c | 2 +-
sound/soc/samsung/s3c24xx_uda134x.c | 2 +-
5 files changed, 5 insertions(+), 5 deletions(-)
(limited to 'sound')
diff --git a/sound/soc/samsung/neo1973_gta02_wm8753.c b/sound/soc/samsung/neo1973_gta02_wm8753.c
index 9e05e10b86a6..0d0ae2b9eef6 100644
--- a/sound/soc/samsung/neo1973_gta02_wm8753.c
+++ b/sound/soc/samsung/neo1973_gta02_wm8753.c
@@ -397,7 +397,7 @@ static struct snd_soc_dai_link neo1973_gta02_dai[] = {
{ /* Hifi Playback - for similatious use with voice below */
.name = "WM8753",
.stream_name = "WM8753 HiFi",
- .cpu_dai_name = "s3c24xx-i2s",
+ .cpu_dai_name = "s3c24xx-iis",
.codec_dai_name = "wm8753-hifi",
.init = neo1973_gta02_wm8753_init,
.platform_name = "samsung-audio",
diff --git a/sound/soc/samsung/neo1973_wm8753.c b/sound/soc/samsung/neo1973_wm8753.c
index cf69e146890b..d20815d5ab2e 100644
--- a/sound/soc/samsung/neo1973_wm8753.c
+++ b/sound/soc/samsung/neo1973_wm8753.c
@@ -559,7 +559,7 @@ static struct snd_soc_dai_link neo1973_dai[] = {
.name = "WM8753",
.stream_name = "WM8753 HiFi",
.platform_name = "samsung-audio",
- .cpu_dai_name = "s3c24xx-i2s",
+ .cpu_dai_name = "s3c24xx-iis",
.codec_dai_name = "wm8753-hifi",
.codec_name = "wm8753-codec.0-001a",
.init = neo1973_wm8753_init,
diff --git a/sound/soc/samsung/s3c24xx_simtec_hermes.c b/sound/soc/samsung/s3c24xx_simtec_hermes.c
index 287a97164ab4..08fcaaa66907 100644
--- a/sound/soc/samsung/s3c24xx_simtec_hermes.c
+++ b/sound/soc/samsung/s3c24xx_simtec_hermes.c
@@ -95,7 +95,7 @@ static struct snd_soc_dai_link simtec_dai_aic33 = {
.name = "tlv320aic33",
.stream_name = "TLV320AIC33",
.codec_name = "tlv320aic3x-codec.0-001a",
- .cpu_dai_name = "s3c24xx-i2s",
+ .cpu_dai_name = "s3c24xx-iis",
.codec_dai_name = "tlv320aic3x-hifi",
.platform_name = "samsung-audio",
.init = simtec_hermes_init,
diff --git a/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c b/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c
index d2b14ba6f9e3..116e3e670167 100644
--- a/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c
+++ b/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c
@@ -86,7 +86,7 @@ static struct snd_soc_dai_link simtec_dai_aic23 = {
.name = "tlv320aic23",
.stream_name = "TLV320AIC23",
.codec_name = "tlv320aic3x-codec.0-001a",
- .cpu_dai_name = "s3c24xx-i2s",
+ .cpu_dai_name = "s3c24xx-iis",
.codec_dai_name = "tlv320aic3x-hifi",
.platform_name = "samsung-audio",
.init = simtec_tlv320aic23_init,
diff --git a/sound/soc/samsung/s3c24xx_uda134x.c b/sound/soc/samsung/s3c24xx_uda134x.c
index cdc8ecbcb8ef..2c09e93dd566 100644
--- a/sound/soc/samsung/s3c24xx_uda134x.c
+++ b/sound/soc/samsung/s3c24xx_uda134x.c
@@ -228,7 +228,7 @@ static struct snd_soc_dai_link s3c24xx_uda134x_dai_link = {
.stream_name = "UDA134X",
.codec_name = "uda134x-hifi",
.codec_dai_name = "uda134x-hifi",
- .cpu_dai_name = "s3c24xx-i2s",
+ .cpu_dai_name = "s3c24xx-iis",
.ops = &s3c24xx_uda134x_ops,
.platform_name = "samsung-audio",
};
--
cgit v1.2.2
From a3adfa00e8089aa72826c6ba04bcb18cfceaf0a9 Mon Sep 17 00:00:00 2001
From: Dmitry Eremin-Solenikov <dbaryshkov@gmail.com>
Date: Fri, 21 Jan 2011 22:14:17 +0300
Subject: ASoC: correct link specifications for corgi, poodle and spitz
ASoC DAI link descriptions for Corgi, Poodle and Spitz platforms
contained incorrect names for cpu_dai and codec, which effectievly disabled sound
on theese platforms. Fix that errors.
Signed-off-by: Dmitry Eremin-Solenikov <dbaryshkov@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
---
sound/soc/pxa/corgi.c | 4 ++--
sound/soc/pxa/poodle.c | 2 +-
sound/soc/pxa/spitz.c | 4 ++--
3 files changed, 5 insertions(+), 5 deletions(-)
(limited to 'sound')
diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c
index fc592f0d5fc7..784cff5f67e8 100644
--- a/sound/soc/pxa/corgi.c
+++ b/sound/soc/pxa/corgi.c
@@ -307,10 +307,10 @@ static int corgi_wm8731_init(struct snd_soc_pcm_runtime *rtd)
static struct snd_soc_dai_link corgi_dai = {
.name = "WM8731",
.stream_name = "WM8731",
- .cpu_dai_name = "pxa-is2-dai",
+ .cpu_dai_name = "pxa2xx-i2s",
.codec_dai_name = "wm8731-hifi",
.platform_name = "pxa-pcm-audio",
- .codec_name = "wm8731-codec-0.001a",
+ .codec_name = "wm8731-codec-0.001b",
.init = corgi_wm8731_init,
.ops = &corgi_ops,
};
diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c
index 6298ee115e27..a7d4999f9b24 100644
--- a/sound/soc/pxa/poodle.c
+++ b/sound/soc/pxa/poodle.c
@@ -276,7 +276,7 @@ static struct snd_soc_dai_link poodle_dai = {
.cpu_dai_name = "pxa2xx-i2s",
.codec_dai_name = "wm8731-hifi",
.platform_name = "pxa-pcm-audio",
- .codec_name = "wm8731-codec.0-001a",
+ .codec_name = "wm8731-codec.0-001b",
.init = poodle_wm8731_init,
.ops = &poodle_ops,
};
diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c
index c2acb69b957a..8e1571350630 100644
--- a/sound/soc/pxa/spitz.c
+++ b/sound/soc/pxa/spitz.c
@@ -315,10 +315,10 @@ static int spitz_wm8750_init(struct snd_soc_pcm_runtime *rtd)
static struct snd_soc_dai_link spitz_dai = {
.name = "wm8750",
.stream_name = "WM8750",
- .cpu_dai_name = "pxa-is2",
+ .cpu_dai_name = "pxa2xx-i2s",
.codec_dai_name = "wm8750-hifi",
.platform_name = "pxa-pcm-audio",
- .codec_name = "wm8750-codec.0-001a",
+ .codec_name = "wm8750-codec.0-001b",
.init = spitz_wm8750_init,
.ops = &spitz_ops,
};
--
cgit v1.2.2
From fd76804f3f5484b35e6a51214c91e916ebba05aa Mon Sep 17 00:00:00 2001
From: Hans-Christian Egtvedt <hans-christian.egtvedt@atmel.com>
Date: Mon, 24 Jan 2011 16:09:56 +0100
Subject: ALSA: fix invalid hardware.h include in ac97c for AVR32 architecture
This patch fixes the non-compiling AC97C driver for AVR32 architecture by
include mach/hardware.h only for AT91 architecture. The AVR32 architecture does
not supply the hardware.h include file.
Signed-off-by: Hans-Christian Egtvedt <hans-christian.egtvedt@atmel.com>
CC: stable@kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/atmel/ac97c.c | 5 ++++-
1 file changed, 4 insertions(+), 1 deletion(-)
(limited to 'sound')
diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c
index 10c3a871a12d..b310702c646e 100644
--- a/sound/atmel/ac97c.c
+++ b/sound/atmel/ac97c.c
@@ -33,9 +33,12 @@
#include <linux/dw_dmac.h>
#include <mach/cpu.h>
-#include <mach/hardware.h>
#include <mach/gpio.h>
+#ifdef CONFIG_ARCH_AT91
+#include <mach/hardware.h>
+#endif
+
#include "ac97c.h"
enum {
--
cgit v1.2.2
From d757534ed15387202e322854cd72dc58bbb975de Mon Sep 17 00:00:00 2001
From: David Henningsson <david.henningsson@canonical.com>
Date: Tue, 25 Jan 2011 19:44:26 +0100
Subject: ALSA: HDA: Fix dmesg output of HDMI supported bits
This typo caused the dmesg output of the supported bits of HDMI
to be cut off early.
Cc: stable@kernel.org
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/pci/hda/hda_eld.c | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
(limited to 'sound')
diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c
index 4a663471dadc..74b0560289c0 100644
--- a/sound/pci/hda/hda_eld.c
+++ b/sound/pci/hda/hda_eld.c
@@ -381,7 +381,7 @@ static void hdmi_show_short_audio_desc(struct cea_sad *a)
snd_print_pcm_rates(a->rates, buf, sizeof(buf));
if (a->format == AUDIO_CODING_TYPE_LPCM)
- snd_print_pcm_bits(a->sample_bits, buf2 + 8, sizeof(buf2 - 8));
+ snd_print_pcm_bits(a->sample_bits, buf2 + 8, sizeof(buf2) - 8);
else if (a->max_bitrate)
snprintf(buf2, sizeof(buf2),
", max bitrate = %d", a->max_bitrate);
--
cgit v1.2.2
From 69058cd6d10423126ab6aeb568f4af2bd34c49fe Mon Sep 17 00:00:00 2001
From: Russell King <rmk+kernel@arm.linux.org.uk>
Date: Wed, 12 Jan 2011 23:17:24 +0000
Subject: ALSA: AACI: fix timeout condition checking
Ensure that a timeout coincident with the condition being waited for
results in success rather than failure. This helps avoid timeout
conditions being inappropriately flagged.
Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk>
---
sound/arm/aaci.c | 6 +++---
1 file changed, 3 insertions(+), 3 deletions(-)
(limited to 'sound')
diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c
index 91acc9a243ec..21ff6296d160 100644
--- a/sound/arm/aaci.c
+++ b/sound/arm/aaci.c
@@ -88,7 +88,7 @@ static void aaci_ac97_write(struct snd_ac97 *ac97, unsigned short reg,
v = readl(aaci->base + AACI_SLFR);
} while ((v & (SLFR_1TXB|SLFR_2TXB)) && --timeout);
- if (!timeout)
+ if (v & (SLFR_1TXB|SLFR_2TXB))
dev_err(&aaci->dev->dev,
"timeout waiting for write to complete\n");
@@ -124,7 +124,7 @@ static unsigned short aaci_ac97_read(struct snd_ac97 *ac97, unsigned short reg)
v = readl(aaci->base + AACI_SLFR);
} while ((v & SLFR_1TXB) && --timeout);
- if (!timeout) {
+ if (v & SLFR_1TXB) {
dev_err(&aaci->dev->dev, "timeout on slot 1 TX busy\n");
v = ~0;
goto out;
@@ -145,7 +145,7 @@ static unsigned short aaci_ac97_read(struct snd_ac97 *ac97, unsigned short reg)
v = readl(aaci->base + AACI_SLFR) & (SLFR_1RXV|SLFR_2RXV);
} while ((v != (SLFR_1RXV|SLFR_2RXV)) && --timeout);
- if (!timeout) {
+ if (v != (SLFR_1RXV|SLFR_2RXV)) {
dev_err(&aaci->dev->dev, "timeout on RX valid\n");
v = ~0;
goto out;
--
cgit v1.2.2
From 250c7a61c35a258e2422b3d55c61bfbad33326be Mon Sep 17 00:00:00 2001
From: Russell King <rmk+kernel@arm.linux.org.uk>
Date: Wed, 12 Jan 2011 23:42:57 +0000
Subject: ALSA: AACI: fix timeout duration
Relying on the access time of peripherals is unreliable - it depends
on the speed of the CPU and the bus. On Versatile Express, these
timeouts were expiring, causing the driver to fail.
Add udelay(1) to ensure that they don't expire early, and adjust
timeouts to give a reasonable margin over the response times.
Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk>
---
sound/arm/aaci.c | 42 +++++++++++++++++++++++-------------------
1 file changed, 23 insertions(+), 19 deletions(-)
(limited to 'sound')
diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c
index 21ff6296d160..24d3013c0231 100644
--- a/sound/arm/aaci.c
+++ b/sound/arm/aaci.c
@@ -30,6 +30,8 @@
#define DRIVER_NAME "aaci-pl041"
+#define FRAME_PERIOD_US 21
+
/*
* PM support is not complete. Turn it off.
*/
@@ -64,8 +66,8 @@ static void aaci_ac97_write(struct snd_ac97 *ac97, unsigned short reg,
unsigned short val)
{
struct aaci *aaci = ac97->private_data;
+ int timeout;
u32 v;
- int timeout = 5000;
if (ac97->num >= 4)
return;
@@ -81,10 +83,13 @@ static void aaci_ac97_write(struct snd_ac97 *ac97, unsigned short reg,
writel(val << 4, aaci->base + AACI_SL2TX);
writel(reg << 12, aaci->base + AACI_SL1TX);
- /*
- * Wait for the transmission of both slots to complete.
- */
+ /* Initially, wait one frame period */
+ udelay(FRAME_PERIOD_US);
+
+ /* And then wait an additional eight frame periods for it to be sent */
+ timeout = FRAME_PERIOD_US * 8;
do {
+ udelay(1);
v = readl(aaci->base + AACI_SLFR);
} while ((v & (SLFR_1TXB|SLFR_2TXB)) && --timeout);
@@ -101,9 +106,8 @@ static void aaci_ac97_write(struct snd_ac97 *ac97, unsigned short reg,
static unsigned short aaci_ac97_read(struct snd_ac97 *ac97, unsigned short reg)
{
struct aaci *aaci = ac97->private_data;
+ int timeout, retries = 10;
u32 v;
- int timeout = 5000;
- int retries = 10;
if (ac97->num >= 4)
return ~0;
@@ -117,10 +121,13 @@ static unsigned short aaci_ac97_read(struct snd_ac97 *ac97, unsigned short reg)
*/
writel((reg << 12) | (1 << 19), aaci->base + AACI_SL1TX);
- /*
- * Wait for the transmission to complete.
- */
+ /* Initially, wait one frame period */
+ udelay(FRAME_PERIOD_US);
+
+ /* And then wait an additional eight frame periods for it to be sent */
+ timeout = FRAME_PERIOD_US * 8;
do {
+ udelay(1);
v = readl(aaci->base + AACI_SLFR);
} while ((v & SLFR_1TXB) && --timeout);
@@ -130,17 +137,13 @@ static unsigned short aaci_ac97_read(struct snd_ac97 *ac97, unsigned short reg)
goto out;
}
- /*
- * Give the AC'97 codec more than enough time
- * to respond. (42us = ~2 frames at 48kHz.)
- */
- udelay(42);
+ /* Now wait for the response frame */
+ udelay(FRAME_PERIOD_US);
- /*
- * Wait for slot 2 to indicate data.
- */
- timeout = 5000;
+ /* And then wait an additional eight frame periods for data */
+ timeout = FRAME_PERIOD_US * 8;
do {
+ udelay(1);
cond_resched();
v = readl(aaci->base + AACI_SLFR) & (SLFR_1RXV|SLFR_2RXV);
} while ((v != (SLFR_1RXV|SLFR_2RXV)) && --timeout);
@@ -179,6 +182,7 @@ aaci_chan_wait_ready(struct aaci_runtime *aacirun, unsigned long mask)
int timeout = 5000;
do {
+ udelay(1);
val = readl(aacirun->base + AACI_SR);
} while (val & mask && timeout--);
}
@@ -874,7 +878,7 @@ static int __devinit aaci_probe_ac97(struct aaci *aaci)
* Give the AC'97 codec more than enough time
* to wake up. (42us = ~2 frames at 48kHz.)
*/
- udelay(42);
+ udelay(FRAME_PERIOD_US * 2);
ret = snd_ac97_bus(aaci->card, 0, &aaci_bus_ops, aaci, &ac97_bus);
if (ret)
--
cgit v1.2.2
From ded9f5238bb719737f82b0b5b957937cb0203804 Mon Sep 17 00:00:00 2001
From: David Henningsson <david.henningsson@canonical.com>
Date: Wed, 26 Jan 2011 11:46:12 +0100
Subject: ALSA: HDA: Fix automute on Thinkpad L412/L512
BugLink: http://bugs.launchpad.net/bugs/707902
More Thinkpad machines with invalid SKU found, that disables
automute between speakers and headphones on these machines.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/pci/hda/patch_realtek.c | 6 ++++--
1 file changed, 4 insertions(+), 2 deletions(-)
(limited to 'sound')
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index be4df4c6fd56..2fa9ed99c32f 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -14954,9 +14954,11 @@ static struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x104d, 0x9084, "Sony VAIO", ALC275_FIXUP_SONY_HWEQ),
SND_PCI_QUIRK_VENDOR(0x104d, "Sony VAIO", ALC269_FIXUP_SONY_VAIO),
SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z),
- SND_PCI_QUIRK(0x17aa, 0x21b8, "Thinkpad Edge 14", ALC269_FIXUP_SKU_IGNORE),
- SND_PCI_QUIRK(0x17aa, 0x21e9, "Thinkpad Edge 14", ALC269_FIXUP_SKU_IGNORE),
SND_PCI_QUIRK(0x17aa, 0x20f2, "Thinkpad SL410/510", ALC269_FIXUP_SKU_IGNORE),
+ SND_PCI_QUIRK(0x17aa, 0x215e, "Thinkpad L512", ALC269_FIXUP_SKU_IGNORE),
+ SND_PCI_QUIRK(0x17aa, 0x21b8, "Thinkpad Edge 14", ALC269_FIXUP_SKU_IGNORE),
+ SND_PCI_QUIRK(0x17aa, 0x21ca, "Thinkpad L412", ALC269_FIXUP_SKU_IGNORE),
+ SND_PCI_QUIRK(0x17aa, 0x21e9, "Thinkpad Edge 15", ALC269_FIXUP_SKU_IGNORE),
SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW),
SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD),
{}
--
cgit v1.2.2