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authorTakashi Iwai <tiwai@suse.de>2009-12-01 08:36:05 +0100
committerTakashi Iwai <tiwai@suse.de>2009-12-01 08:36:05 +0100
commitef47bf386e2042c2e483b4ea0259233202ad3ebf (patch)
tree02baba407d2dbae9d367865c3d52d6e9400c8773 /sound
parent306ecee926cf79f1b3b5f6035be09ef3d83f1b76 (diff)
parent8ee763b9c82c6ca0a59a7271ce4fa29d7baf5c09 (diff)
Merge branch 'fix/misc' into topic/misc
Diffstat (limited to 'sound')
-rw-r--r--sound/arm/aaci.c6
-rw-r--r--sound/pci/hda/patch_conexant.c15
-rw-r--r--sound/pci/hda/patch_nvhdmi.c2
-rw-r--r--sound/pci/hda/patch_realtek.c17
-rw-r--r--sound/pci/hda/patch_sigmatel.c2
-rw-r--r--sound/pci/ice1712/ice1712.h4
-rw-r--r--sound/soc/codecs/tlv320aic23.c5
-rw-r--r--sound/soc/omap/omap3evm.c2
-rw-r--r--sound/soc/omap/omap3pandora.c3
-rw-r--r--sound/soc/soc-dapm.c20
-rw-r--r--sound/usb/usbmixer.c9
11 files changed, 64 insertions, 21 deletions
diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c
index a03fe80a7a73..eb715e732106 100644
--- a/sound/arm/aaci.c
+++ b/sound/arm/aaci.c
@@ -501,6 +501,10 @@ static int aaci_pcm_hw_params(struct snd_pcm_substream *substream,
int err;
aaci_pcm_hw_free(substream);
+ if (aacirun->pcm_open) {
+ snd_ac97_pcm_close(aacirun->pcm);
+ aacirun->pcm_open = 0;
+ }
err = devdma_hw_alloc(NULL, substream,
params_buffer_bytes(params));
@@ -514,7 +518,7 @@ static int aaci_pcm_hw_params(struct snd_pcm_substream *substream,
else
err = snd_ac97_pcm_open(aacirun->pcm, params_rate(params),
params_channels(params),
- aacirun->pcm->r[1].slots);
+ aacirun->pcm->r[0].slots);
if (err)
goto out;
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 6479e65858d3..905859d4f4df 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -110,6 +110,7 @@ struct conexant_spec {
unsigned int dell_automute;
unsigned int port_d_mode;
+ unsigned char ext_mic_bias;
};
static int conexant_playback_pcm_open(struct hda_pcm_stream *hinfo,
@@ -1927,6 +1928,11 @@ static hda_nid_t cxt5066_adc_nids[3] = { 0x14, 0x15, 0x16 };
static hda_nid_t cxt5066_capsrc_nids[1] = { 0x17 };
#define CXT5066_SPDIF_OUT 0x21
+/* OLPC's microphone port is DC coupled for use with external sensors,
+ * therefore we use a 50% mic bias in order to center the input signal with
+ * the DC input range of the codec. */
+#define CXT5066_OLPC_EXT_MIC_BIAS PIN_VREF50
+
static struct hda_channel_mode cxt5066_modes[1] = {
{ 2, NULL },
};
@@ -1980,9 +1986,10 @@ static int cxt5066_hp_master_sw_put(struct snd_kcontrol *kcontrol,
/* toggle input of built-in and mic jack appropriately */
static void cxt5066_automic(struct hda_codec *codec)
{
- static struct hda_verb ext_mic_present[] = {
+ struct conexant_spec *spec = codec->spec;
+ struct hda_verb ext_mic_present[] = {
/* enable external mic, port B */
- {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, spec->ext_mic_bias},
/* switch to external mic input */
{0x17, AC_VERB_SET_CONNECT_SEL, 0},
@@ -2235,7 +2242,7 @@ static struct hda_verb cxt5066_init_verbs_olpc[] = {
{0x19, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC1 */
/* Port B: external microphone */
- {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, CXT5066_OLPC_EXT_MIC_BIAS},
/* Port C: internal microphone */
{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
@@ -2353,6 +2360,7 @@ static int patch_cxt5066(struct hda_codec *codec)
spec->input_mux = &cxt5066_capture_source;
spec->port_d_mode = PIN_HP;
+ spec->ext_mic_bias = PIN_VREF80;
spec->num_init_verbs = 1;
spec->init_verbs[0] = cxt5066_init_verbs;
@@ -2384,6 +2392,7 @@ static int patch_cxt5066(struct hda_codec *codec)
spec->mixers[spec->num_mixers++] = cxt5066_mixer_master_olpc;
spec->mixers[spec->num_mixers++] = cxt5066_mixers;
spec->port_d_mode = 0;
+ spec->ext_mic_bias = CXT5066_OLPC_EXT_MIC_BIAS;
/* no S/PDIF out */
spec->multiout.dig_out_nid = 0;
diff --git a/sound/pci/hda/patch_nvhdmi.c b/sound/pci/hda/patch_nvhdmi.c
index 9fb60276f5c9..6afdab09bab7 100644
--- a/sound/pci/hda/patch_nvhdmi.c
+++ b/sound/pci/hda/patch_nvhdmi.c
@@ -397,6 +397,7 @@ static int patch_nvhdmi_2ch(struct hda_codec *codec)
static struct hda_codec_preset snd_hda_preset_nvhdmi[] = {
{ .id = 0x10de0002, .name = "MCP78 HDMI", .patch = patch_nvhdmi_8ch },
{ .id = 0x10de0003, .name = "MCP78 HDMI", .patch = patch_nvhdmi_8ch },
+ { .id = 0x10de0005, .name = "MCP78 HDMI", .patch = patch_nvhdmi_8ch },
{ .id = 0x10de0006, .name = "MCP78 HDMI", .patch = patch_nvhdmi_8ch },
{ .id = 0x10de0007, .name = "MCP7A HDMI", .patch = patch_nvhdmi_8ch },
{ .id = 0x10de0067, .name = "MCP67 HDMI", .patch = patch_nvhdmi_2ch },
@@ -406,6 +407,7 @@ static struct hda_codec_preset snd_hda_preset_nvhdmi[] = {
MODULE_ALIAS("snd-hda-codec-id:10de0002");
MODULE_ALIAS("snd-hda-codec-id:10de0003");
+MODULE_ALIAS("snd-hda-codec-id:10de0005");
MODULE_ALIAS("snd-hda-codec-id:10de0006");
MODULE_ALIAS("snd-hda-codec-id:10de0007");
MODULE_ALIAS("snd-hda-codec-id:10de0067");
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index ff20048504b6..70583719282b 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -4684,9 +4684,9 @@ static int alc880_parse_auto_config(struct hda_codec *codec)
spec->multiout.dig_out_nid = dig_nid;
else {
spec->multiout.slave_dig_outs = spec->slave_dig_outs;
- spec->slave_dig_outs[i - 1] = dig_nid;
- if (i == ARRAY_SIZE(spec->slave_dig_outs) - 1)
+ if (i >= ARRAY_SIZE(spec->slave_dig_outs) - 1)
break;
+ spec->slave_dig_outs[i - 1] = dig_nid;
}
}
if (spec->autocfg.dig_in_pin)
@@ -6249,7 +6249,7 @@ static struct snd_pci_quirk alc260_cfg_tbl[] = {
SND_PCI_QUIRK(0x1025, 0x008f, "Acer", ALC260_ACER),
SND_PCI_QUIRK(0x1509, 0x4540, "Favorit 100XS", ALC260_FAVORIT100),
SND_PCI_QUIRK(0x103c, 0x2808, "HP d5700", ALC260_HP_3013),
- SND_PCI_QUIRK(0x103c, 0x280a, "HP d5750", ALC260_HP_3013),
+ SND_PCI_QUIRK(0x103c, 0x280a, "HP d5750", ALC260_AUTO), /* no quirk */
SND_PCI_QUIRK(0x103c, 0x3010, "HP", ALC260_HP_3013),
SND_PCI_QUIRK(0x103c, 0x3011, "HP", ALC260_HP_3013),
SND_PCI_QUIRK(0x103c, 0x3012, "HP", ALC260_HP_DC7600),
@@ -8911,10 +8911,11 @@ static struct snd_pci_quirk alc882_ssid_cfg_tbl[] = {
SND_PCI_QUIRK(0x106b, 0x3800, "MacbookPro 4,1", ALC885_MBP3),
SND_PCI_QUIRK(0x106b, 0x3e00, "iMac 24 Aluminum", ALC885_IMAC24),
SND_PCI_QUIRK(0x106b, 0x3f00, "Macbook 5,1", ALC885_MB5),
- /* FIXME: HP jack sense seems not working for MBP 5,1, so apparently
- * no perfect solution yet
+ /* FIXME: HP jack sense seems not working for MBP 5,1 or 5,2,
+ * so apparently no perfect solution yet
*/
SND_PCI_QUIRK(0x106b, 0x4000, "MacbookPro 5,1", ALC885_MB5),
+ SND_PCI_QUIRK(0x106b, 0x4600, "MacbookPro 5,2", ALC885_MB5),
{} /* terminator */
};
@@ -9813,9 +9814,9 @@ static int alc882_parse_auto_config(struct hda_codec *codec)
spec->multiout.dig_out_nid = dig_nid;
else {
spec->multiout.slave_dig_outs = spec->slave_dig_outs;
- spec->slave_dig_outs[i - 1] = dig_nid;
- if (i == ARRAY_SIZE(spec->slave_dig_outs) - 1)
+ if (i >= ARRAY_SIZE(spec->slave_dig_outs) - 1)
break;
+ spec->slave_dig_outs[i - 1] = dig_nid;
}
}
if (spec->autocfg.dig_in_pin)
@@ -11460,6 +11461,8 @@ static struct snd_pci_quirk alc262_cfg_tbl[] = {
SND_PCI_QUIRK(0x104d, 0x820f, "Sony ASSAMD", ALC262_SONY_ASSAMD),
SND_PCI_QUIRK(0x104d, 0x9016, "Sony VAIO", ALC262_AUTO), /* dig-only */
SND_PCI_QUIRK(0x104d, 0x9025, "Sony VAIO Z21MN", ALC262_TOSHIBA_S06),
+ SND_PCI_QUIRK(0x104d, 0x9035, "Sony VAIO VGN-FW170J", ALC262_AUTO),
+ SND_PCI_QUIRK(0x104d, 0x9047, "Sony VAIO Type G", ALC262_AUTO),
SND_PCI_QUIRK_MASK(0x104d, 0xff00, 0x9000, "Sony VAIO",
ALC262_SONY_ASSAMD),
SND_PCI_QUIRK(0x1179, 0x0001, "Toshiba dynabook SS RX1",
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 8eb6508cd991..86de305fc9f2 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -1590,6 +1590,8 @@ static struct snd_pci_quirk stac92hd73xx_cfg_tbl[] = {
"Dell Studio 17", STAC_DELL_M6_DMIC),
SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x02be,
"Dell Studio 1555", STAC_DELL_M6_DMIC),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x02bd,
+ "Dell Studio 1557", STAC_DELL_M6_DMIC),
{} /* terminator */
};
diff --git a/sound/pci/ice1712/ice1712.h b/sound/pci/ice1712/ice1712.h
index 9da2dae64c5b..d063149e7047 100644
--- a/sound/pci/ice1712/ice1712.h
+++ b/sound/pci/ice1712/ice1712.h
@@ -382,8 +382,8 @@ struct snd_ice1712 {
#ifdef CONFIG_PM
int (*pm_suspend)(struct snd_ice1712 *);
int (*pm_resume)(struct snd_ice1712 *);
- int pm_suspend_enabled:1;
- int pm_saved_is_spdif_master:1;
+ unsigned int pm_suspend_enabled:1;
+ unsigned int pm_saved_is_spdif_master:1;
unsigned int pm_saved_spdif_ctrl;
unsigned char pm_saved_spdif_cfg;
unsigned int pm_saved_route;
diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c
index 35606ae60868..58ffb6de400f 100644
--- a/sound/soc/codecs/tlv320aic23.c
+++ b/sound/soc/codecs/tlv320aic23.c
@@ -265,8 +265,8 @@ static const int bosr_usb_divisor_table[] = {
#define UPPER_GROUP ((1<<8) | (1<<9) | (1<<10) | (1<<11) | (1<<15))
static const unsigned short sr_valid_mask[] = {
LOWER_GROUP|UPPER_GROUP, /* Normal, bosr - 0*/
- LOWER_GROUP|UPPER_GROUP, /* Normal, bosr - 1*/
LOWER_GROUP, /* Usb, bosr - 0*/
+ LOWER_GROUP|UPPER_GROUP, /* Normal, bosr - 1*/
UPPER_GROUP, /* Usb, bosr - 1*/
};
/*
@@ -625,11 +625,10 @@ static int tlv320aic23_resume(struct platform_device *pdev)
{
struct snd_soc_device *socdev = platform_get_drvdata(pdev);
struct snd_soc_codec *codec = socdev->card->codec;
- int i;
u16 reg;
/* Sync reg_cache with the hardware */
- for (reg = 0; reg < ARRAY_SIZE(tlv320aic23_reg); i++) {
+ for (reg = 0; reg < TLV320AIC23_RESET; reg++) {
u16 val = tlv320aic23_read_reg_cache(codec, reg);
tlv320aic23_write(codec, reg, val);
}
diff --git a/sound/soc/omap/omap3evm.c b/sound/soc/omap/omap3evm.c
index 9114c263077b..13aa380de162 100644
--- a/sound/soc/omap/omap3evm.c
+++ b/sound/soc/omap/omap3evm.c
@@ -144,4 +144,4 @@ module_exit(omap3evm_soc_exit);
MODULE_AUTHOR("Anuj Aggarwal <anuj.aggarwal@ti.com>");
MODULE_DESCRIPTION("ALSA SoC OMAP3 EVM");
-MODULE_LICENSE("GPLv2");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/omap/omap3pandora.c b/sound/soc/omap/omap3pandora.c
index ad219aaf7cb8..0cd06f5dd356 100644
--- a/sound/soc/omap/omap3pandora.c
+++ b/sound/soc/omap/omap3pandora.c
@@ -134,7 +134,7 @@ static int omap3pandora_hp_event(struct snd_soc_dapm_widget *w,
* |P| <--- TWL4030 <--------- Line In and MICs
*/
static const struct snd_soc_dapm_widget omap3pandora_out_dapm_widgets[] = {
- SND_SOC_DAPM_DAC("PCM DAC", "Playback", SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_DAC("PCM DAC", "HiFi Playback", SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_PGA_E("Headphone Amplifier", SND_SOC_NOPM,
0, 0, NULL, 0, omap3pandora_hp_event,
SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
@@ -181,6 +181,7 @@ static int omap3pandora_out_init(struct snd_soc_codec *codec)
snd_soc_dapm_nc_pin(codec, "CARKITR");
snd_soc_dapm_nc_pin(codec, "HFL");
snd_soc_dapm_nc_pin(codec, "HFR");
+ snd_soc_dapm_nc_pin(codec, "VIBRA");
ret = snd_soc_dapm_new_controls(codec, omap3pandora_out_dapm_widgets,
ARRAY_SIZE(omap3pandora_out_dapm_widgets));
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index d89f6dc00908..66d4c165f99b 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -973,9 +973,19 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event)
if (!w->power_check)
continue;
- power = w->power_check(w);
- if (power)
- sys_power = 1;
+ /* If we're suspending then pull down all the
+ * power. */
+ switch (event) {
+ case SND_SOC_DAPM_STREAM_SUSPEND:
+ power = 0;
+ break;
+
+ default:
+ power = w->power_check(w);
+ if (power)
+ sys_power = 1;
+ break;
+ }
if (w->power == power)
continue;
@@ -999,8 +1009,12 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event)
case SND_SOC_DAPM_STREAM_RESUME:
sys_power = 1;
break;
+ case SND_SOC_DAPM_STREAM_SUSPEND:
+ sys_power = 0;
+ break;
case SND_SOC_DAPM_STREAM_NOP:
sys_power = codec->bias_level != SND_SOC_BIAS_STANDBY;
+ break;
default:
break;
}
diff --git a/sound/usb/usbmixer.c b/sound/usb/usbmixer.c
index 9efcfd08d747..c998220b99c6 100644
--- a/sound/usb/usbmixer.c
+++ b/sound/usb/usbmixer.c
@@ -1071,6 +1071,15 @@ static int parse_audio_feature_unit(struct mixer_build *state, int unitid, unsig
channels = (ftr[0] - 7) / csize - 1;
master_bits = snd_usb_combine_bytes(ftr + 6, csize);
+ /* master configuration quirks */
+ switch (state->chip->usb_id) {
+ case USB_ID(0x08bb, 0x2702):
+ snd_printk(KERN_INFO
+ "usbmixer: master volume quirk for PCM2702 chip\n");
+ /* disable non-functional volume control */
+ master_bits &= ~(1 << (USB_FEATURE_VOLUME - 1));
+ break;
+ }
if (channels > 0)
first_ch_bits = snd_usb_combine_bytes(ftr + 6 + csize, csize);
else