| lists.openwall.net | lists / announce owl-users owl-dev john-users john-dev passwdqc-users yescrypt popa3d-users / oss-security kernel-hardening musl sabotage tlsify passwords / crypt-dev xvendor / Bugtraq Full-Disclosure linux-kernel linux-netdev linux-ext4 linux-hardening linux-cve-announce PHC | |
|
Open Source and information security mailing list archives
| ||
|
Date: Sun, 12 Jun 2011 09:52:46 +0200
From: Takashi Iwai <tiwai@...e.de>
To: Linus Torvalds <torvalds@...ux-foundation.org>
Cc: Mark Brown <broonie@...nsource.wolfsonmicro.com>,
Liam Girdwood <lrg@...mlogic.co.uk>,
linux-kernel@...r.kernel.org
Subject: [GIT PULL] sound fixes for 3.0-rc3
Linus,
please pull sound fixes for v3.0-rc3 from:
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git for-linus
A few regression fixes and small fixes are included.
Thanks!
Takashi
===
Daniel T Chen (2):
ALSA: hda: Fix quirk for Dell Inspiron 910
ALSA: hda: Fix inaudible internal speakers on CyberpowerPC Gamer Xplorer N57001 laptop
Joachim Eastwood (1):
ASoC: atmel_ssc: Don't try to free ssc if request failed
Lars-Peter Clausen (3):
ASoC: AD1836: Fix setting the PCM format
ASoC: Blackfin: bf5xx-ad1836: Fix codec device name
ASoC: snd_soc_new_{mixer,mux,pga} make sure to use right DAPM context
Mark Brown (5):
ASoC: Only update SYSCLK_ENA when pausing WM8915 SYSCLK
ASoC: Add missing break in WM8915 FLL source selection
ASoC: Check for NULL register bank in snd_soc_get_cache_val()
ASoC: Fix WM8962 headphone volume update for use of advanced caches
ASoC: WM8804 does not support sample rates below 32kHz
Sangbeom Kim (1):
ASoC: SAMSUNG: Fix the incorrect referencing of I2SCON register
Takashi Iwai (3):
ALSA: hda - Fix invalid unsol tag for some alc262 model quirks
ALSA: hda - Fix initialization of hp pins with master_mute in Realtek
ALSA: Use %pV for snd_printk()
Timur Tabi (1):
ASoC: fsl: fix initialization of DMA buffers
---
sound/core/misc.c | 40 +++++++++++++++---------------------
sound/pci/hda/patch_conexant.c | 1 +
sound/pci/hda/patch_realtek.c | 15 ++++++++-----
sound/soc/atmel/atmel_ssc_dai.c | 5 ++-
sound/soc/blackfin/bf5xx-ad1836.c | 4 +-
sound/soc/codecs/ad1836.c | 14 ++++++------
sound/soc/codecs/ad1836.h | 6 +++++
sound/soc/codecs/wm8804.c | 9 ++++++-
sound/soc/codecs/wm8915.c | 3 +-
sound/soc/codecs/wm8962.c | 4 +-
sound/soc/fsl/fsl_dma.c | 9 ++++---
sound/soc/samsung/i2s.c | 4 +-
sound/soc/soc-cache.c | 3 ++
sound/soc/soc-dapm.c | 17 +++++++--------
14 files changed, 74 insertions(+), 60 deletions(-)
diff --git a/sound/core/misc.c b/sound/core/misc.c
index 2c41825..eb9fe2e 100644
--- a/sound/core/misc.c
+++ b/sound/core/misc.c
@@ -58,26 +58,6 @@ static const char *sanity_file_name(const char *path)
else
return path;
}
-
-/* print file and line with a certain printk prefix */
-static int print_snd_pfx(unsigned int level, const char *path, int line,
- const char *format)
-{
- const char *file = sanity_file_name(path);
- char tmp[] = "<0>";
- const char *pfx = level ? KERN_DEBUG : KERN_DEFAULT;
- int ret = 0;
-
- if (format[0] == '<' && format[2] == '>') {
- tmp[1] = format[1];
- pfx = tmp;
- ret = 1;
- }
- printk("%sALSA %s:%d: ", pfx, file, line);
- return ret;
-}
-#else
-#define print_snd_pfx(level, path, line, format) 0
#endif
#if defined(CONFIG_SND_DEBUG) || defined(CONFIG_SND_VERBOSE_PRINTK)
@@ -85,15 +65,29 @@ void __snd_printk(unsigned int level, const char *path, int line,
const char *format, ...)
{
va_list args;
-
+#ifdef CONFIG_SND_VERBOSE_PRINTK
+ struct va_format vaf;
+ char verbose_fmt[] = KERN_DEFAULT "ALSA %s:%d %pV";
+#endif
+
#ifdef CONFIG_SND_DEBUG
if (debug < level)
return;
#endif
+
va_start(args, format);
- if (print_snd_pfx(level, path, line, format))
- format += 3; /* skip the printk level-prefix */
+#ifdef CONFIG_SND_VERBOSE_PRINTK
+ vaf.fmt = format;
+ vaf.va = &args;
+ if (format[0] == '<' && format[2] == '>') {
+ memcpy(verbose_fmt, format, 3);
+ vaf.fmt = format + 3;
+ } else if (level)
+ memcpy(verbose_fmt, KERN_DEBUG, 3);
+ printk(verbose_fmt, sanity_file_name(path), line, &vaf);
+#else
vprintk(format, args);
+#endif
va_end(args);
}
EXPORT_SYMBOL_GPL(__snd_printk);
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 3e6b9a8..694b9daf 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -3102,6 +3102,7 @@ static const struct snd_pci_quirk cxt5066_cfg_tbl[] = {
SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo G560", CXT5066_ASUS),
SND_PCI_QUIRK(0x17aa, 0x3938, "Lenovo G565", CXT5066_AUTO),
SND_PCI_QUIRK_VENDOR(0x17aa, "Lenovo", CXT5066_IDEAPAD), /* Fallback for Lenovos without dock mic */
+ SND_PCI_QUIRK(0x1b0a, 0x2092, "CyberpowerPC Gamer Xplorer N57001", CXT5066_AUTO),
{}
};
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 7a4e100..43fcfbd 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -1141,6 +1141,13 @@ static void update_speakers(struct hda_codec *codec)
struct alc_spec *spec = codec->spec;
int on;
+ /* Control HP pins/amps depending on master_mute state;
+ * in general, HP pins/amps control should be enabled in all cases,
+ * but currently set only for master_mute, just to be safe
+ */
+ do_automute(codec, ARRAY_SIZE(spec->autocfg.hp_pins),
+ spec->autocfg.hp_pins, spec->master_mute, true);
+
if (!spec->automute)
on = 0;
else
@@ -6201,11 +6208,6 @@ static const struct snd_kcontrol_new alc260_input_mixer[] = {
/* update HP, line and mono out pins according to the master switch */
static void alc260_hp_master_update(struct hda_codec *codec)
{
- struct alc_spec *spec = codec->spec;
-
- /* change HP pins */
- do_automute(codec, ARRAY_SIZE(spec->autocfg.hp_pins),
- spec->autocfg.hp_pins, spec->master_mute, true);
update_speakers(codec);
}
@@ -11924,7 +11926,7 @@ static const struct hda_verb alc262_nec_verbs[] = {
* 0x1b = port replicator headphone out
*/
-#define ALC_HP_EVENT 0x37
+#define ALC_HP_EVENT ALC880_HP_EVENT
static const struct hda_verb alc262_fujitsu_unsol_verbs[] = {
{0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
@@ -13860,6 +13862,7 @@ static const struct snd_pci_quirk alc268_cfg_tbl[] = {
SND_PCI_QUIRK(0x1025, 0x015b, "Acer Aspire One",
ALC268_ACER_ASPIRE_ONE),
SND_PCI_QUIRK(0x1028, 0x0253, "Dell OEM", ALC268_DELL),
+ SND_PCI_QUIRK(0x1028, 0x02b0, "Dell Inspiron 910", ALC268_AUTO),
SND_PCI_QUIRK_MASK(0x1028, 0xfff0, 0x02b0,
"Dell Inspiron Mini9/Vostro A90", ALC268_DELL),
/* almost compatible with toshiba but with optional digital outs;
diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c
index 7fbfa05..eda955b 100644
--- a/sound/soc/atmel/atmel_ssc_dai.c
+++ b/sound/soc/atmel/atmel_ssc_dai.c
@@ -848,9 +848,10 @@ int atmel_ssc_set_audio(int ssc_id)
if (IS_ERR(ssc))
pr_warn("Unable to parent ASoC SSC DAI on SSC: %ld\n",
PTR_ERR(ssc));
- else
+ else {
ssc_pdev->dev.parent = &(ssc->pdev->dev);
- ssc_free(ssc);
+ ssc_free(ssc);
+ }
ret = platform_device_add(ssc_pdev);
if (ret < 0)
diff --git a/sound/soc/blackfin/bf5xx-ad1836.c b/sound/soc/blackfin/bf5xx-ad1836.c
index ea4951c..f79d165 100644
--- a/sound/soc/blackfin/bf5xx-ad1836.c
+++ b/sound/soc/blackfin/bf5xx-ad1836.c
@@ -75,7 +75,7 @@ static struct snd_soc_dai_link bf5xx_ad1836_dai[] = {
.cpu_dai_name = "bfin-tdm.0",
.codec_dai_name = "ad1836-hifi",
.platform_name = "bfin-tdm-pcm-audio",
- .codec_name = "ad1836.0",
+ .codec_name = "spi0.4",
.ops = &bf5xx_ad1836_ops,
},
{
@@ -84,7 +84,7 @@ static struct snd_soc_dai_link bf5xx_ad1836_dai[] = {
.cpu_dai_name = "bfin-tdm.1",
.codec_dai_name = "ad1836-hifi",
.platform_name = "bfin-tdm-pcm-audio",
- .codec_name = "ad1836.0",
+ .codec_name = "spi0.4",
.ops = &bf5xx_ad1836_ops,
},
};
diff --git a/sound/soc/codecs/ad1836.c b/sound/soc/codecs/ad1836.c
index ab63d52..754c496 100644
--- a/sound/soc/codecs/ad1836.c
+++ b/sound/soc/codecs/ad1836.c
@@ -145,22 +145,22 @@ static int ad1836_hw_params(struct snd_pcm_substream *substream,
/* bit size */
switch (params_format(params)) {
case SNDRV_PCM_FORMAT_S16_LE:
- word_len = 3;
+ word_len = AD1836_WORD_LEN_16;
break;
case SNDRV_PCM_FORMAT_S20_3LE:
- word_len = 1;
+ word_len = AD1836_WORD_LEN_20;
break;
case SNDRV_PCM_FORMAT_S24_LE:
case SNDRV_PCM_FORMAT_S32_LE:
- word_len = 0;
+ word_len = AD1836_WORD_LEN_24;
break;
}
- snd_soc_update_bits(codec, AD1836_DAC_CTRL1,
- AD1836_DAC_WORD_LEN_MASK, word_len);
+ snd_soc_update_bits(codec, AD1836_DAC_CTRL1, AD1836_DAC_WORD_LEN_MASK,
+ word_len << AD1836_DAC_WORD_LEN_OFFSET);
- snd_soc_update_bits(codec, AD1836_ADC_CTRL2,
- AD1836_ADC_WORD_LEN_MASK, word_len);
+ snd_soc_update_bits(codec, AD1836_ADC_CTRL2, AD1836_ADC_WORD_LEN_MASK,
+ word_len << AD1836_ADC_WORD_OFFSET);
return 0;
}
diff --git a/sound/soc/codecs/ad1836.h b/sound/soc/codecs/ad1836.h
index 8455967..9d6a3f8 100644
--- a/sound/soc/codecs/ad1836.h
+++ b/sound/soc/codecs/ad1836.h
@@ -25,6 +25,7 @@
#define AD1836_DAC_SERFMT_PCK256 (0x4 << 5)
#define AD1836_DAC_SERFMT_PCK128 (0x5 << 5)
#define AD1836_DAC_WORD_LEN_MASK 0x18
+#define AD1836_DAC_WORD_LEN_OFFSET 3
#define AD1836_DAC_CTRL2 1
#define AD1836_DACL1_MUTE 0
@@ -51,6 +52,7 @@
#define AD1836_ADCL2_MUTE 2
#define AD1836_ADCR2_MUTE 3
#define AD1836_ADC_WORD_LEN_MASK 0x30
+#define AD1836_ADC_WORD_OFFSET 5
#define AD1836_ADC_SERFMT_MASK (7 << 6)
#define AD1836_ADC_SERFMT_PCK256 (0x4 << 6)
#define AD1836_ADC_SERFMT_PCK128 (0x5 << 6)
@@ -60,4 +62,8 @@
#define AD1836_NUM_REGS 16
+#define AD1836_WORD_LEN_24 0x0
+#define AD1836_WORD_LEN_20 0x1
+#define AD1836_WORD_LEN_16 0x2
+
#endif
diff --git a/sound/soc/codecs/wm8804.c b/sound/soc/codecs/wm8804.c
index 6785688..9a5e67c 100644
--- a/sound/soc/codecs/wm8804.c
+++ b/sound/soc/codecs/wm8804.c
@@ -680,20 +680,25 @@ static struct snd_soc_dai_ops wm8804_dai_ops = {
#define WM8804_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
SNDRV_PCM_FMTBIT_S24_LE)
+#define WM8804_RATES (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \
+ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_64000 | \
+ SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 | \
+ SNDRV_PCM_RATE_176400 | SNDRV_PCM_RATE_192000)
+
static struct snd_soc_dai_driver wm8804_dai = {
.name = "wm8804-spdif",
.playback = {
.stream_name = "Playback",
.channels_min = 2,
.channels_max = 2,
- .rates = SNDRV_PCM_RATE_8000_192000,
+ .rates = WM8804_RATES,
.formats = WM8804_FORMATS,
},
.capture = {
.stream_name = "Capture",
.channels_min = 2,
.channels_max = 2,
- .rates = SNDRV_PCM_RATE_8000_192000,
+ .rates = WM8804_RATES,
.formats = WM8804_FORMATS,
},
.ops = &wm8804_dai_ops,
diff --git a/sound/soc/codecs/wm8915.c b/sound/soc/codecs/wm8915.c
index a0b1a72..e2ab4fa 100644
--- a/sound/soc/codecs/wm8915.c
+++ b/sound/soc/codecs/wm8915.c
@@ -1839,7 +1839,7 @@ static int wm8915_set_sysclk(struct snd_soc_dai *dai,
int old;
/* Disable SYSCLK while we reconfigure */
- old = snd_soc_read(codec, WM8915_AIF_CLOCKING_1);
+ old = snd_soc_read(codec, WM8915_AIF_CLOCKING_1) & WM8915_SYSCLK_ENA;
snd_soc_update_bits(codec, WM8915_AIF_CLOCKING_1,
WM8915_SYSCLK_ENA, 0);
@@ -2038,6 +2038,7 @@ static int wm8915_set_fll(struct snd_soc_codec *codec, int fll_id, int source,
break;
case WM8915_FLL_MCLK2:
reg = 1;
+ break;
case WM8915_FLL_DACLRCLK1:
reg = 2;
break;
diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index f90ae42..5e05eed 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -1999,12 +1999,12 @@ static int wm8962_put_hp_sw(struct snd_kcontrol *kcontrol,
return 0;
/* If the left PGA is enabled hit that VU bit... */
- if (reg_cache[WM8962_PWR_MGMT_2] & WM8962_HPOUTL_PGA_ENA)
+ if (snd_soc_read(codec, WM8962_PWR_MGMT_2) & WM8962_HPOUTL_PGA_ENA)
return snd_soc_write(codec, WM8962_HPOUTL_VOLUME,
reg_cache[WM8962_HPOUTL_VOLUME]);
/* ...otherwise the right. The VU is stereo. */
- if (reg_cache[WM8962_PWR_MGMT_2] & WM8962_HPOUTR_PGA_ENA)
+ if (snd_soc_read(codec, WM8962_PWR_MGMT_2) & WM8962_HPOUTR_PGA_ENA)
return snd_soc_write(codec, WM8962_HPOUTR_VOLUME,
reg_cache[WM8962_HPOUTR_VOLUME]);
diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c
index 15dac0f..6680c0b 100644
--- a/sound/soc/fsl/fsl_dma.c
+++ b/sound/soc/fsl/fsl_dma.c
@@ -310,7 +310,7 @@ static int fsl_dma_new(struct snd_card *card, struct snd_soc_dai *dai,
* should allocate a DMA buffer only for the streams that are valid.
*/
- if (dai->driver->playback.channels_min) {
+ if (pcm->streams[0].substream) {
ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, card->dev,
fsl_dma_hardware.buffer_bytes_max,
&pcm->streams[0].substream->dma_buffer);
@@ -320,13 +320,13 @@ static int fsl_dma_new(struct snd_card *card, struct snd_soc_dai *dai,
}
}
- if (dai->driver->capture.channels_min) {
+ if (pcm->streams[1].substream) {
ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, card->dev,
fsl_dma_hardware.buffer_bytes_max,
&pcm->streams[1].substream->dma_buffer);
if (ret) {
- snd_dma_free_pages(&pcm->streams[0].substream->dma_buffer);
dev_err(card->dev, "can't alloc capture dma buffer\n");
+ snd_dma_free_pages(&pcm->streams[0].substream->dma_buffer);
return ret;
}
}
@@ -449,7 +449,8 @@ static int fsl_dma_open(struct snd_pcm_substream *substream)
dma_private->ld_buf_phys = ld_buf_phys;
dma_private->dma_buf_phys = substream->dma_buffer.addr;
- ret = request_irq(dma_private->irq, fsl_dma_isr, 0, "DMA", dma_private);
+ ret = request_irq(dma_private->irq, fsl_dma_isr, 0, "fsldma-audio",
+ dma_private);
if (ret) {
dev_err(dev, "can't register ISR for IRQ %u (ret=%i)\n",
dma_private->irq, ret);
diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c
index ffa09b3..992a732 100644
--- a/sound/soc/samsung/i2s.c
+++ b/sound/soc/samsung/i2s.c
@@ -191,7 +191,7 @@ static inline bool tx_active(struct i2s_dai *i2s)
if (!i2s)
return false;
- active = readl(i2s->addr + I2SMOD);
+ active = readl(i2s->addr + I2SCON);
if (is_secondary(i2s))
active &= CON_TXSDMA_ACTIVE;
@@ -223,7 +223,7 @@ static inline bool rx_active(struct i2s_dai *i2s)
if (!i2s)
return false;
- active = readl(i2s->addr + I2SMOD) & CON_RXDMA_ACTIVE;
+ active = readl(i2s->addr + I2SCON) & CON_RXDMA_ACTIVE;
return active ? true : false;
}
diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c
index 06b7b81..c005ceb 100644
--- a/sound/soc/soc-cache.c
+++ b/sound/soc/soc-cache.c
@@ -466,6 +466,9 @@ static bool snd_soc_set_cache_val(void *base, unsigned int idx,
static unsigned int snd_soc_get_cache_val(const void *base, unsigned int idx,
unsigned int word_size)
{
+ if (!base)
+ return -1;
+
switch (word_size) {
case 1: {
const u8 *cache = base;
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 776e6f4..32ab7fc 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -350,9 +350,9 @@ static int dapm_is_shared_kcontrol(struct snd_soc_dapm_context *dapm,
}
/* create new dapm mixer control */
-static int dapm_new_mixer(struct snd_soc_dapm_context *dapm,
- struct snd_soc_dapm_widget *w)
+static int dapm_new_mixer(struct snd_soc_dapm_widget *w)
{
+ struct snd_soc_dapm_context *dapm = w->dapm;
int i, ret = 0;
size_t name_len, prefix_len;
struct snd_soc_dapm_path *path;
@@ -450,9 +450,9 @@ static int dapm_new_mixer(struct snd_soc_dapm_context *dapm,
}
/* create new dapm mux control */
-static int dapm_new_mux(struct snd_soc_dapm_context *dapm,
- struct snd_soc_dapm_widget *w)
+static int dapm_new_mux(struct snd_soc_dapm_widget *w)
{
+ struct snd_soc_dapm_context *dapm = w->dapm;
struct snd_soc_dapm_path *path = NULL;
struct snd_kcontrol *kcontrol;
struct snd_card *card = dapm->card->snd_card;
@@ -535,8 +535,7 @@ static int dapm_new_mux(struct snd_soc_dapm_context *dapm,
}
/* create new dapm volume control */
-static int dapm_new_pga(struct snd_soc_dapm_context *dapm,
- struct snd_soc_dapm_widget *w)
+static int dapm_new_pga(struct snd_soc_dapm_widget *w)
{
if (w->num_kcontrols)
dev_err(w->dapm->dev,
@@ -1826,13 +1825,13 @@ int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm)
case snd_soc_dapm_mixer:
case snd_soc_dapm_mixer_named_ctl:
w->power_check = dapm_generic_check_power;
- dapm_new_mixer(dapm, w);
+ dapm_new_mixer(w);
break;
case snd_soc_dapm_mux:
case snd_soc_dapm_virt_mux:
case snd_soc_dapm_value_mux:
w->power_check = dapm_generic_check_power;
- dapm_new_mux(dapm, w);
+ dapm_new_mux(w);
break;
case snd_soc_dapm_adc:
case snd_soc_dapm_aif_out:
@@ -1845,7 +1844,7 @@ int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm)
case snd_soc_dapm_pga:
case snd_soc_dapm_out_drv:
w->power_check = dapm_generic_check_power;
- dapm_new_pga(dapm, w);
+ dapm_new_pga(w);
break;
case snd_soc_dapm_input:
case snd_soc_dapm_output:
--
To unsubscribe from this list: send the line "unsubscribe linux-kernel" in
the body of a message to majordomo@...r.kernel.org
More majordomo info at http://vger.kernel.org/majordomo-info.html
Please read the FAQ at http://www.tux.org/lkml/
Powered by blists - more mailing lists