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-rw-r--r--sound/core/misc.c40
-rw-r--r--sound/firewire/isight.c1
-rw-r--r--sound/pci/asihpi/hpidspcd.c2
-rw-r--r--sound/pci/emu10k1/emu10k1_main.c8
-rw-r--r--sound/pci/fm801.c13
-rw-r--r--sound/pci/hda/hda_beep.h9
-rw-r--r--sound/pci/hda/patch_analog.c16
-rw-r--r--sound/pci/hda/patch_conexant.c1
-rw-r--r--sound/pci/hda/patch_realtek.c18
-rw-r--r--sound/pci/hda/patch_via.c11
-rw-r--r--sound/pci/lola/lola.c2
-rw-r--r--sound/pci/rme9652/hdspm.c8
-rw-r--r--sound/soc/atmel/atmel_ssc_dai.c5
-rw-r--r--sound/soc/blackfin/bf5xx-ad1836.c4
-rw-r--r--sound/soc/codecs/ad1836.c14
-rw-r--r--sound/soc/codecs/ad1836.h6
-rw-r--r--sound/soc/codecs/wm8804.c9
-rw-r--r--sound/soc/codecs/wm8915.c3
-rw-r--r--sound/soc/codecs/wm8962.c4
-rw-r--r--sound/soc/codecs/wm_hubs.c8
-rw-r--r--sound/soc/fsl/fsl_dma.c9
-rw-r--r--sound/soc/samsung/i2s.c4
-rw-r--r--sound/soc/soc-cache.c3
-rw-r--r--sound/soc/soc-dapm.c22
-rw-r--r--sound/usb/6fire/firmware.c2
-rw-r--r--sound/usb/6fire/pcm.c4
-rw-r--r--sound/usb/quirks.c2
27 files changed, 137 insertions, 91 deletions
diff --git a/sound/core/misc.c b/sound/core/misc.c
index 2c41825c836e..eb9fe2e1d291 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/firewire/isight.c b/sound/firewire/isight.c
index 86ee16ca365e..440030818db7 100644
--- a/sound/firewire/isight.c
+++ b/sound/firewire/isight.c
@@ -209,6 +209,7 @@ static void isight_packet(struct fw_iso_context *context, u32 cycle,
isight->packet_index = -1;
return;
}
+ fw_iso_context_queue_flush(isight->context);
if (++index >= QUEUE_LENGTH)
index = 0;
diff --git a/sound/pci/asihpi/hpidspcd.c b/sound/pci/asihpi/hpidspcd.c
index fb311d8c05bf..5c6ea113d219 100644
--- a/sound/pci/asihpi/hpidspcd.c
+++ b/sound/pci/asihpi/hpidspcd.c
@@ -60,7 +60,7 @@ struct code_header {
HPI_VER_MINOR(HPI_VER) * 100 + HPI_VER_RELEASE(HPI_VER)))
/***********************************************************************/
-#include "linux/pci.h"
+#include <linux/pci.h>
/*-------------------------------------------------------------------*/
short hpi_dsp_code_open(u32 adapter, struct dsp_code *ps_dsp_code,
u32 *pos_error_code)
diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c
index 5e619a84da06..15f0161ce4a2 100644
--- a/sound/pci/emu10k1/emu10k1_main.c
+++ b/sound/pci/emu10k1/emu10k1_main.c
@@ -1440,6 +1440,14 @@ static struct snd_emu_chip_details emu_chip_details[] = {
.ca0102_chip = 1,
.spk71 = 1,
.emu_model = EMU_MODEL_EMU0404}, /* EMU 0404 */
+ /* EMU0404 PCIe */
+ {.vendor = 0x1102, .device = 0x0008, .subsystem = 0x40051102,
+ .driver = "Audigy2", .name = "E-mu 0404 PCIe [MAEM8984]",
+ .id = "EMU0404",
+ .emu10k2_chip = 1,
+ .ca0108_chip = 1,
+ .spk71 = 1,
+ .emu_model = EMU_MODEL_EMU0404}, /* EMU 0404 PCIe ver_03 */
/* Note that all E-mu cards require kernel 2.6 or newer. */
{.vendor = 0x1102, .device = 0x0008,
.driver = "Audigy2", .name = "SB Audigy 2 Value [Unknown]",
diff --git a/sound/pci/fm801.c b/sound/pci/fm801.c
index eacd4901a308..a7ec7030cf87 100644
--- a/sound/pci/fm801.c
+++ b/sound/pci/fm801.c
@@ -1234,9 +1234,12 @@ static int __devinit snd_fm801_create(struct snd_card *card,
sprintf(chip->tea.bus_info, "PCI:%s", pci_name(pci));
if ((tea575x_tuner & TUNER_TYPE_MASK) > 0 &&
(tea575x_tuner & TUNER_TYPE_MASK) < 4) {
- if (snd_tea575x_init(&chip->tea))
+ if (snd_tea575x_init(&chip->tea)) {
snd_printk(KERN_ERR "TEA575x radio not found\n");
- } else if ((tea575x_tuner & TUNER_TYPE_MASK) == 0)
+ snd_fm801_free(chip);
+ return -ENODEV;
+ }
+ } else if ((tea575x_tuner & TUNER_TYPE_MASK) == 0) {
/* autodetect tuner connection */
for (tea575x_tuner = 1; tea575x_tuner <= 3; tea575x_tuner++) {
chip->tea575x_tuner = tea575x_tuner;
@@ -1246,6 +1249,12 @@ static int __devinit snd_fm801_create(struct snd_card *card,
break;
}
}
+ if (tea575x_tuner == 4) {
+ snd_printk(KERN_ERR "TEA575x radio not found\n");
+ snd_fm801_free(chip);
+ return -ENODEV;
+ }
+ }
strlcpy(chip->tea.card, snd_fm801_tea575x_gpios[(tea575x_tuner & TUNER_TYPE_MASK) - 1].name, sizeof(chip->tea.card));
#endif
diff --git a/sound/pci/hda/hda_beep.h b/sound/pci/hda/hda_beep.h
index f1de1bac042c..55f0647458c7 100644
--- a/sound/pci/hda/hda_beep.h
+++ b/sound/pci/hda/hda_beep.h
@@ -50,7 +50,12 @@ int snd_hda_enable_beep_device(struct hda_codec *codec, int enable);
int snd_hda_attach_beep_device(struct hda_codec *codec, int nid);
void snd_hda_detach_beep_device(struct hda_codec *codec);
#else
-#define snd_hda_attach_beep_device(...) 0
-#define snd_hda_detach_beep_device(...)
+static inline int snd_hda_attach_beep_device(struct hda_codec *codec, int nid)
+{
+ return 0;
+}
+static inline void snd_hda_detach_beep_device(struct hda_codec *codec)
+{
+}
#endif
#endif
diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index 696ac2590307..d694e9d4921d 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -506,9 +506,11 @@ static void ad198x_power_eapd_write(struct hda_codec *codec, hda_nid_t front,
hda_nid_t hp)
{
struct ad198x_spec *spec = codec->spec;
- snd_hda_codec_write(codec, front, 0, AC_VERB_SET_EAPD_BTLENABLE,
+ if (snd_hda_query_pin_caps(codec, front) & AC_PINCAP_EAPD)
+ snd_hda_codec_write(codec, front, 0, AC_VERB_SET_EAPD_BTLENABLE,
!spec->inv_eapd ? 0x00 : 0x02);
- snd_hda_codec_write(codec, hp, 0, AC_VERB_SET_EAPD_BTLENABLE,
+ if (snd_hda_query_pin_caps(codec, hp) & AC_PINCAP_EAPD)
+ snd_hda_codec_write(codec, hp, 0, AC_VERB_SET_EAPD_BTLENABLE,
!spec->inv_eapd ? 0x00 : 0x02);
}
@@ -524,6 +526,10 @@ static void ad198x_power_eapd(struct hda_codec *codec)
case 0x11d4184a:
case 0x11d4194a:
case 0x11d4194b:
+ case 0x11d41988:
+ case 0x11d4198b:
+ case 0x11d4989a:
+ case 0x11d4989b:
ad198x_power_eapd_write(codec, 0x12, 0x11);
break;
case 0x11d41981:
@@ -533,12 +539,6 @@ static void ad198x_power_eapd(struct hda_codec *codec)
case 0x11d41986:
ad198x_power_eapd_write(codec, 0x1b, 0x1a);
break;
- case 0x11d41988:
- case 0x11d4198b:
- case 0x11d4989a:
- case 0x11d4989b:
- ad198x_power_eapd_write(codec, 0x29, 0x22);
- break;
}
}
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 3e6b9a8539c2..694b9daf691f 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 7a4e10002f56..61a774b3d3cb 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},
@@ -13314,9 +13316,8 @@ static void alc268_acer_lc_setup(struct hda_codec *codec)
struct alc_spec *spec = codec->spec;
spec->autocfg.hp_pins[0] = 0x15;
spec->autocfg.speaker_pins[0] = 0x14;
- spec->automute_mixer_nid[0] = 0x0f;
spec->automute = 1;
- spec->automute_mode = ALC_AUTOMUTE_MIXER;
+ spec->automute_mode = ALC_AUTOMUTE_AMP;
spec->ext_mic.pin = 0x18;
spec->ext_mic.mux_idx = 0;
spec->int_mic.pin = 0x12;
@@ -13860,6 +13861,7 @@ 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/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index 605c99e1e520..c952582fb218 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -832,10 +832,13 @@ static int via_hp_build(struct hda_codec *codec)
knew->subdevice = HDA_SUBDEV_NID_FLAG | nid;
knew->private_value = nid;
- knew = via_clone_control(spec, &via_hp_mixer[1]);
- if (knew == NULL)
- return -ENOMEM;
- knew->subdevice = side_mute_channel(spec);
+ nid = side_mute_channel(spec);
+ if (nid) {
+ knew = via_clone_control(spec, &via_hp_mixer[1]);
+ if (knew == NULL)
+ return -ENOMEM;
+ knew->subdevice = nid;
+ }
return 0;
}
diff --git a/sound/pci/lola/lola.c b/sound/pci/lola/lola.c
index 34b24286d279..2692e5ae5f2d 100644
--- a/sound/pci/lola/lola.c
+++ b/sound/pci/lola/lola.c
@@ -445,7 +445,7 @@ static void lola_reset_setups(struct lola *chip)
lola_setup_all_analog_gains(chip, PLAY, false); /* output, update */
}
-static int lola_parse_tree(struct lola *chip)
+static int __devinit lola_parse_tree(struct lola *chip)
{
unsigned int val;
int nid, err;
diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c
index 949691a876d3..3f08afc0f0d3 100644
--- a/sound/pci/rme9652/hdspm.c
+++ b/sound/pci/rme9652/hdspm.c
@@ -521,6 +521,7 @@ MODULE_SUPPORTED_DEVICE("{{RME HDSPM-MADI}}");
#define HDSPM_DMA_AREA_KILOBYTES (HDSPM_DMA_AREA_BYTES/1024)
/* revisions >= 230 indicate AES32 card */
+#define HDSPM_MADI_OLD_REV 207
#define HDSPM_MADI_REV 210
#define HDSPM_RAYDAT_REV 211
#define HDSPM_AIO_REV 212
@@ -1143,7 +1144,7 @@ static int hdspm_external_sample_rate(struct hdspm *hdspm)
/* if wordclock has synced freq and wordclock is valid */
if ((status2 & HDSPM_wcLock) != 0 &&
- (status & HDSPM_SelSyncRef0) == 0) {
+ (status2 & HDSPM_SelSyncRef0) == 0) {
rate_bits = status2 & HDSPM_wcFreqMask;
@@ -1639,12 +1640,14 @@ static int snd_hdspm_midi_input_read (struct hdspm_midi *hmidi)
}
}
hmidi->pending = 0;
+ spin_unlock_irqrestore(&hmidi->lock, flags);
+ spin_lock_irqsave(&hmidi->hdspm->lock, flags);
hmidi->hdspm->control_register |= hmidi->ie;
hdspm_write(hmidi->hdspm, HDSPM_controlRegister,
hmidi->hdspm->control_register);
+ spin_unlock_irqrestore(&hmidi->hdspm->lock, flags);
- spin_unlock_irqrestore (&hmidi->lock, flags);
return snd_hdspm_midi_output_write (hmidi);
}
@@ -6377,6 +6380,7 @@ static int __devinit snd_hdspm_create(struct snd_card *card,
switch (hdspm->firmware_rev) {
case HDSPM_MADI_REV:
+ case HDSPM_MADI_OLD_REV:
hdspm->io_type = MADI;
hdspm->card_name = "RME MADI";
hdspm->midiPorts = 3;
diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c
index 7fbfa051f6e1..eda955b15834 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 ea4951cf5526..f79d1655e035 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 ab63d52e36e1..754c496412bd 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 845596717fdf..9d6a3f8f8aaf 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 6785688f8806..9a5e67c5a6bd 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 a0b1a7278284..e2ab4fac2819 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 f90ae427242b..5e05eed96c38 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/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c
index e55b298c14a0..9e370d14ad88 100644
--- a/sound/soc/codecs/wm_hubs.c
+++ b/sound/soc/codecs/wm_hubs.c
@@ -215,23 +215,23 @@ static const struct snd_kcontrol_new analogue_snd_controls[] = {
SOC_SINGLE_TLV("IN1L Volume", WM8993_LEFT_LINE_INPUT_1_2_VOLUME, 0, 31, 0,
inpga_tlv),
SOC_SINGLE("IN1L Switch", WM8993_LEFT_LINE_INPUT_1_2_VOLUME, 7, 1, 1),
-SOC_SINGLE("IN1L ZC Switch", WM8993_LEFT_LINE_INPUT_1_2_VOLUME, 7, 1, 0),
+SOC_SINGLE("IN1L ZC Switch", WM8993_LEFT_LINE_INPUT_1_2_VOLUME, 6, 1, 0),
SOC_SINGLE_TLV("IN1R Volume", WM8993_RIGHT_LINE_INPUT_1_2_VOLUME, 0, 31, 0,
inpga_tlv),
SOC_SINGLE("IN1R Switch", WM8993_RIGHT_LINE_INPUT_1_2_VOLUME, 7, 1, 1),
-SOC_SINGLE("IN1R ZC Switch", WM8993_RIGHT_LINE_INPUT_1_2_VOLUME, 7, 1, 0),
+SOC_SINGLE("IN1R ZC Switch", WM8993_RIGHT_LINE_INPUT_1_2_VOLUME, 6, 1, 0),
SOC_SINGLE_TLV("IN2L Volume", WM8993_LEFT_LINE_INPUT_3_4_VOLUME, 0, 31, 0,
inpga_tlv),
SOC_SINGLE("IN2L Switch", WM8993_LEFT_LINE_INPUT_3_4_VOLUME, 7, 1, 1),
-SOC_SINGLE("IN2L ZC Switch", WM8993_LEFT_LINE_INPUT_3_4_VOLUME, 7, 1, 0),
+SOC_SINGLE("IN2L ZC Switch", WM8993_LEFT_LINE_INPUT_3_4_VOLUME, 6, 1, 0),
SOC_SINGLE_TLV("IN2R Volume", WM8993_RIGHT_LINE_INPUT_3_4_VOLUME, 0, 31, 0,
inpga_tlv),
SOC_SINGLE("IN2R Switch", WM8993_RIGHT_LINE_INPUT_3_4_VOLUME, 7, 1, 1),
-SOC_SINGLE("IN2R ZC Switch", WM8993_RIGHT_LINE_INPUT_3_4_VOLUME, 7, 1, 0),
+SOC_SINGLE("IN2R ZC Switch", WM8993_RIGHT_LINE_INPUT_3_4_VOLUME, 6, 1, 0),
SOC_SINGLE_TLV("MIXINL IN2L Volume", WM8993_INPUT_MIXER3, 7, 1, 0,
inmix_sw_tlv),
diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c
index 15dac0f20cd8..6680c0b4d203 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 ffa09b3b2caa..992a732b5211 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 06b7b81a1601..c005ceb70c9d 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 999bb08cdfb1..32ab7fc4579a 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -325,6 +325,7 @@ static int dapm_connect_mixer(struct snd_soc_dapm_context *dapm,
}
static int dapm_is_shared_kcontrol(struct snd_soc_dapm_context *dapm,
+ struct snd_soc_dapm_widget *kcontrolw,
const struct snd_kcontrol_new *kcontrol_new,
struct snd_kcontrol **kcontrol)
{
@@ -334,6 +335,8 @@ static int dapm_is_shared_kcontrol(struct snd_soc_dapm_context *dapm,
*kcontrol = NULL;
list_for_each_entry(w, &dapm->card->widgets, list) {
+ if (w == kcontrolw || w->dapm != kcontrolw->dapm)
+ continue;
for (i = 0; i < w->num_kcontrols; i++) {
if (&w->kcontrol_news[i] == kcontrol_new) {
if (w->kcontrols)
@@ -347,9 +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;
@@ -447,9 +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;
@@ -468,7 +471,7 @@ static int dapm_new_mux(struct snd_soc_dapm_context *dapm,
return -EINVAL;
}
- shared = dapm_is_shared_kcontrol(dapm, &w->kcontrol_news[0],
+ shared = dapm_is_shared_kcontrol(dapm, w, &w->kcontrol_news[0],
&kcontrol);
if (kcontrol) {
wlist = kcontrol->private_data;
@@ -532,8 +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,
@@ -1823,13 +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:
@@ -1842,7 +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:
diff --git a/sound/usb/6fire/firmware.c b/sound/usb/6fire/firmware.c
index d47beffedb0f..1e3ae3327dd3 100644
--- a/sound/usb/6fire/firmware.c
+++ b/sound/usb/6fire/firmware.c
@@ -227,6 +227,7 @@ static int usb6fire_fw_ezusb_upload(
ret = usb6fire_fw_ihex_init(fw, rec);
if (ret < 0) {
kfree(rec);
+ release_firmware(fw);
snd_printk(KERN_ERR PREFIX "error validating ezusb "
"firmware %s.\n", fwname);
return ret;
@@ -269,7 +270,6 @@ static int usb6fire_fw_ezusb_upload(
data = 0x00; /* resume ezusb cpu */
ret = usb6fire_fw_ezusb_write(device, 0xa0, 0xe600, &data, 1);
if (ret < 0) {
- release_firmware(fw);
snd_printk(KERN_ERR PREFIX "unable to upload ezusb "
"firmware %s: end message.\n", fwname);
return ret;
diff --git a/sound/usb/6fire/pcm.c b/sound/usb/6fire/pcm.c
index b137b25865cc..d144cdb2f159 100644
--- a/sound/usb/6fire/pcm.c
+++ b/sound/usb/6fire/pcm.c
@@ -395,12 +395,12 @@ static int usb6fire_pcm_open(struct snd_pcm_substream *alsa_sub)
alsa_rt->hw = pcm_hw;
if (alsa_sub->stream == SNDRV_PCM_STREAM_PLAYBACK) {
- if (rt->rate >= 0)
+ if (rt->rate < ARRAY_SIZE(rates))
alsa_rt->hw.rates = rates_alsaid[rt->rate];
alsa_rt->hw.channels_max = OUT_N_CHANNELS;
sub = &rt->playback;
} else if (alsa_sub->stream == SNDRV_PCM_STREAM_CAPTURE) {
- if (rt->rate >= 0)
+ if (rt->rate < ARRAY_SIZE(rates))
alsa_rt->hw.rates = rates_alsaid[rt->rate];
alsa_rt->hw.channels_max = IN_N_CHANNELS;
sub = &rt->capture;
diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index 2e969cbb393b..090e1930dfdc 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -403,7 +403,7 @@ static int snd_usb_cm106_boot_quirk(struct usb_device *dev)
static int snd_usb_cm6206_boot_quirk(struct usb_device *dev)
{
int err, reg;
- int val[] = {0x200c, 0x3000, 0xf800, 0x143f, 0x0000, 0x3000};
+ int val[] = {0x2004, 0x3000, 0xf800, 0x143f, 0x0000, 0x3000};
for (reg = 0; reg < ARRAY_SIZE(val); reg++) {
err = snd_usb_cm106_write_int_reg(dev, reg, val[reg]);
generated by cgit v1.2.3 (git 2.39.0) at 2024-11-26 22:43:13 +0300