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Merge branch 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6

Browse Source 
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
  ALSA: hda: Fix inaudible internal speakers on CyberpowerPC Gamer Xplorer N57001 laptop
  ALSA: Use %pV for snd_printk()
  ALSA: hda - Fix initialization of hp pins with master_mute in Realtek
  ALSA: hda - Fix invalid unsol tag for some alc262 model quirks
  ASoC: SAMSUNG: Fix the incorrect referencing of I2SCON register
  ASoC: snd_soc_new_{mixer,mux,pga} make sure to use right DAPM context
  ASoC: fsl: fix initialization of DMA buffers
  ASoC: WM8804 does not support sample rates below 32kHz
  ASoC: Fix WM8962 headphone volume update for use of advanced caches
  ASoC: Blackfin: bf5xx-ad1836: Fix codec device name
  ALSA: hda: Fix quirk for Dell Inspiron 910
  ASoC: AD1836: Fix setting the PCM format
  ASoC: Check for NULL register bank in snd_soc_get_cache_val()
  ASoC: Add missing break in WM8915 FLL source selection
  ASoC: Only update SYSCLK_ENA when pausing WM8915 SYSCLK
  ASoC: atmel_ssc: Don't try to free ssc if request failed
This commit is contained in:
Linus Torvalds 2011-06-12 11:04:25 -07:00
commit 9d6fa8fa70
14 changed files with 74 additions and 60 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

40
sound/core/misc.c
View File

@ -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);

1
sound/pci/hda/patch_conexant.c
View File

@ -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),
{}
};

15
sound/pci/hda/patch_realtek.c
View File

@ -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;

5
sound/soc/atmel/atmel_ssc_dai.c
View File

@ -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)

4
sound/soc/blackfin/bf5xx-ad1836.c
View File

@ -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,
},
};

14
sound/soc/codecs/ad1836.c
View File

@ -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;
}

6
sound/soc/codecs/ad1836.h
View File

@ -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

9
sound/soc/codecs/wm8804.c
View File

@ -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,

3
sound/soc/codecs/wm8915.c
View File

@ -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;

4
sound/soc/codecs/wm8962.c
View File

@ -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]);

9
sound/soc/fsl/fsl_dma.c
View File

@ -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);

4
sound/soc/samsung/i2s.c
View File

@ -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;
}

3
sound/soc/soc-cache.c
View File

@ -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;

17
sound/soc/soc-dapm.c
View File

@ -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: