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authorcodeworkx <codeworkx@cyanogenmod.com>2012-09-18 18:51:21 +0200
committercodeworkx <codeworkx@cyanogenmod.com>2012-09-18 19:02:23 +0200
commitf991bd2a427ec6a5e049e19745aba6a5d7f006c4 (patch)
treec6f0d428403c32c95335c34b3f0105c2e4c8087d /sound
parentc28265764ec6ad9995eb0c761a376ffc9f141fcd (diff)
parentbea37381fd9a34c6660e5195d31beea86aa3dda3 (diff)
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Merge linux-3.0.31 from korg into jellybean
Conflicts: arch/arm/mm/proc-v7.S drivers/base/core.c drivers/gpu/drm/i915/i915_gem_execbuffer.c drivers/gpu/drm/i915/intel_display.c drivers/gpu/drm/i915/intel_lvds.c drivers/gpu/drm/radeon/evergreen.c drivers/gpu/drm/radeon/r100.c drivers/gpu/drm/radeon/radeon_connectors.c drivers/gpu/drm/radeon/rs600.c drivers/usb/core/hub.c drivers/usb/host/xhci-pci.c drivers/usb/host/xhci.c drivers/usb/serial/qcserial.c fs/proc/base.c Change-Id: Ia98b35db3f8c0bfd95817867d3acb85be8e5e772
Diffstat (limited to 'sound')
-rw-r--r--sound/pci/hda/hda_codec.c10
-rw-r--r--sound/pci/hda/hda_codec.h3
-rw-r--r--sound/pci/hda/hda_local.h7
-rw-r--r--sound/pci/hda/hda_proc.c2
-rw-r--r--sound/pci/hda/patch_conexant.c42
-rw-r--r--sound/pci/hda/patch_realtek.c11
-rw-r--r--sound/pci/hda/patch_sigmatel.c12
-rw-r--r--sound/pci/ice1712/amp.c7
-rw-r--r--sound/pci/intel8x0.c6
-rw-r--r--sound/pci/oxygen/xonar_wm87x6.c1
-rw-r--r--sound/soc/codecs/ak4642.c2
-rw-r--r--sound/soc/codecs/wm8962.c8
-rw-r--r--sound/soc/codecs/wm_hubs.c6
-rw-r--r--sound/soc/imx/imx-ssi.c2
-rw-r--r--sound/soc/pxa/pxa-ssp.c61
-rw-r--r--sound/soc/samsung/neo1973_wm8753.c4
-rw-r--r--sound/soc/soc-core.c18
-rw-r--r--sound/soc/soc-dapm.c12
-rw-r--r--sound/usb/usx2y/usb_stream.c6
19 files changed, 162 insertions, 58 deletions
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 2195851..39e1a6a 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -1328,7 +1328,7 @@ void snd_hda_codec_setup_stream(struct hda_codec *codec, hda_nid_t nid,
for (i = 0; i < c->cvt_setups.used; i++) {
p = snd_array_elem(&c->cvt_setups, i);
if (!p->active && p->stream_tag == stream_tag &&
- get_wcaps_type(get_wcaps(codec, p->nid)) == type)
+ get_wcaps_type(get_wcaps(c, p->nid)) == type)
p->dirty = 1;
}
}
@@ -1651,7 +1651,11 @@ static void put_vol_mute(struct hda_codec *codec, struct hda_amp_info *info,
parm = ch ? AC_AMP_SET_RIGHT : AC_AMP_SET_LEFT;
parm |= direction == HDA_OUTPUT ? AC_AMP_SET_OUTPUT : AC_AMP_SET_INPUT;
parm |= index << AC_AMP_SET_INDEX_SHIFT;
- parm |= val;
+ if ((val & HDA_AMP_MUTE) && !(info->amp_caps & AC_AMPCAP_MUTE) &&
+ (info->amp_caps & AC_AMPCAP_MIN_MUTE))
+ ; /* set the zero value as a fake mute */
+ else
+ parm |= val;
snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, parm);
info->vol[ch] = val;
}
@@ -1918,7 +1922,7 @@ int snd_hda_mixer_amp_tlv(struct snd_kcontrol *kcontrol, int op_flag,
val1 = -((caps & AC_AMPCAP_OFFSET) >> AC_AMPCAP_OFFSET_SHIFT);
val1 += ofs;
val1 = ((int)val1) * ((int)val2);
- if (min_mute)
+ if (min_mute || (caps & AC_AMPCAP_MIN_MUTE))
val2 |= TLV_DB_SCALE_MUTE;
if (put_user(SNDRV_CTL_TLVT_DB_SCALE, _tlv))
return -EFAULT;
diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h
index 59c9730..eff1fc5 100644
--- a/sound/pci/hda/hda_codec.h
+++ b/sound/pci/hda/hda_codec.h
@@ -302,6 +302,9 @@ enum {
#define AC_AMPCAP_MUTE (1<<31) /* mute capable */
#define AC_AMPCAP_MUTE_SHIFT 31
+/* driver-specific amp-caps: using bits 24-30 */
+#define AC_AMPCAP_MIN_MUTE (1 << 30) /* min-volume = mute */
+
/* Connection list */
#define AC_CLIST_LENGTH (0x7f<<0)
#define AC_CLIST_LONG (1<<7)
diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h
index 08ec073..e289a13 100644
--- a/sound/pci/hda/hda_local.h
+++ b/sound/pci/hda/hda_local.h
@@ -474,7 +474,12 @@ static inline u32 get_wcaps(struct hda_codec *codec, hda_nid_t nid)
}
/* get the widget type from widget capability bits */
-#define get_wcaps_type(wcaps) (((wcaps) & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT)
+static inline int get_wcaps_type(unsigned int wcaps)
+{
+ if (!wcaps)
+ return -1; /* invalid type */
+ return (wcaps & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT;
+}
static inline unsigned int get_wcaps_channels(u32 wcaps)
{
diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c
index bfe74c2..6fe944a 100644
--- a/sound/pci/hda/hda_proc.c
+++ b/sound/pci/hda/hda_proc.c
@@ -54,6 +54,8 @@ static const char *get_wid_type_name(unsigned int wid_value)
[AC_WID_BEEP] = "Beep Generator Widget",
[AC_WID_VENDOR] = "Vendor Defined Widget",
};
+ if (wid_value == -1)
+ return "UNKNOWN Widget";
wid_value &= 0xf;
if (names[wid_value])
return names[wid_value];
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 3c2381c..4cf3266 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -1917,6 +1917,10 @@ static void cxt5051_init_mic_port(struct hda_codec *codec, hda_nid_t nid,
snd_hda_codec_write(codec, nid, 0,
AC_VERB_SET_UNSOLICITED_ENABLE,
AC_USRSP_EN | event);
+}
+
+static void cxt5051_init_mic_jack(struct hda_codec *codec, hda_nid_t nid)
+{
snd_hda_input_jack_add(codec, nid, SND_JACK_MICROPHONE, NULL);
snd_hda_input_jack_report(codec, nid);
}
@@ -1934,7 +1938,6 @@ static int cxt5051_init(struct hda_codec *codec)
struct conexant_spec *spec = codec->spec;
conexant_init(codec);
- conexant_init_jacks(codec);
if (spec->auto_mic & AUTO_MIC_PORTB)
cxt5051_init_mic_port(codec, 0x17, CXT5051_PORTB_EVENT);
@@ -2067,6 +2070,12 @@ static int patch_cxt5051(struct hda_codec *codec)
if (spec->beep_amp)
snd_hda_attach_beep_device(codec, spec->beep_amp);
+ conexant_init_jacks(codec);
+ if (spec->auto_mic & AUTO_MIC_PORTB)
+ cxt5051_init_mic_jack(codec, 0x17);
+ if (spec->auto_mic & AUTO_MIC_PORTC)
+ cxt5051_init_mic_jack(codec, 0x18);
+
return 0;
}
@@ -3994,9 +4003,14 @@ static void cx_auto_init_output(struct hda_codec *codec)
int i;
mute_outputs(codec, spec->multiout.num_dacs, spec->multiout.dac_nids);
- for (i = 0; i < cfg->hp_outs; i++)
+ for (i = 0; i < cfg->hp_outs; i++) {
+ unsigned int val = PIN_OUT;
+ if (snd_hda_query_pin_caps(codec, cfg->hp_pins[i]) &
+ AC_PINCAP_HP_DRV)
+ val |= AC_PINCTL_HP_EN;
snd_hda_codec_write(codec, cfg->hp_pins[i], 0,
- AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP);
+ AC_VERB_SET_PIN_WIDGET_CONTROL, val);
+ }
mute_outputs(codec, cfg->hp_outs, cfg->hp_pins);
mute_outputs(codec, cfg->line_outs, cfg->line_out_pins);
mute_outputs(codec, cfg->speaker_outs, cfg->speaker_pins);
@@ -4118,7 +4132,8 @@ static int cx_auto_add_volume_idx(struct hda_codec *codec, const char *basename,
err = snd_hda_ctl_add(codec, nid, kctl);
if (err < 0)
return err;
- if (!(query_amp_caps(codec, nid, hda_dir) & AC_AMPCAP_MUTE))
+ if (!(query_amp_caps(codec, nid, hda_dir) &
+ (AC_AMPCAP_MUTE | AC_AMPCAP_MIN_MUTE)))
break;
}
return 0;
@@ -4363,6 +4378,22 @@ static const struct hda_codec_ops cx_auto_patch_ops = {
.reboot_notify = snd_hda_shutup_pins,
};
+/* add "fake" mute amp-caps to DACs on cx5051 so that mixer mute switches
+ * can be created (bko#42825)
+ */
+static void add_cx5051_fake_mutes(struct hda_codec *codec)
+{
+ static hda_nid_t out_nids[] = {
+ 0x10, 0x11, 0
+ };
+ hda_nid_t *p;
+
+ for (p = out_nids; *p; p++)
+ snd_hda_override_amp_caps(codec, *p, HDA_OUTPUT,
+ AC_AMPCAP_MIN_MUTE |
+ query_amp_caps(codec, *p, HDA_OUTPUT));
+}
+
static int patch_conexant_auto(struct hda_codec *codec)
{
struct conexant_spec *spec;
@@ -4381,6 +4412,9 @@ static int patch_conexant_auto(struct hda_codec *codec)
case 0x14f15045:
spec->single_adc_amp = 1;
break;
+ case 0x14f15051:
+ add_cx5051_fake_mutes(codec);
+ break;
}
err = cx_auto_search_adcs(codec);
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index eb0a141..51412e1 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -16419,6 +16419,7 @@ static const struct alc_config_preset alc861_presets[] = {
/* Pin config fixes */
enum {
PINFIX_FSC_AMILO_PI1505,
+ PINFIX_ASUS_A6RP,
};
static const struct alc_fixup alc861_fixups[] = {
@@ -16430,9 +16431,19 @@ static const struct alc_fixup alc861_fixups[] = {
{ }
}
},
+ [PINFIX_ASUS_A6RP] = {
+ .type = ALC_FIXUP_VERBS,
+ .v.verbs = (const struct hda_verb[]) {
+ /* node 0x0f VREF seems controlling the master output */
+ { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF50 },
+ { }
+ },
+ },
};
static const struct snd_pci_quirk alc861_fixup_tbl[] = {
+ SND_PCI_QUIRK(0x1043, 0x1393, "ASUS A6Rp", PINFIX_ASUS_A6RP),
+ SND_PCI_QUIRK(0x1584, 0x2b01, "Haier W18", PINFIX_ASUS_A6RP),
SND_PCI_QUIRK(0x1734, 0x10c7, "FSC Amilo Pi1505", PINFIX_FSC_AMILO_PI1505),
{}
};
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 5d2e97a..8670682 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -1602,7 +1602,7 @@ static const struct snd_pci_quirk stac92hd73xx_cfg_tbl[] = {
SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x02bd,
"Dell Studio 1557", STAC_DELL_M6_DMIC),
SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x02fe,
- "Dell Studio XPS 1645", STAC_DELL_M6_BOTH),
+ "Dell Studio XPS 1645", STAC_DELL_M6_DMIC),
SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0413,
"Dell Studio 1558", STAC_DELL_M6_DMIC),
{} /* terminator */
@@ -4162,13 +4162,15 @@ static int enable_pin_detect(struct hda_codec *codec, hda_nid_t nid,
return 1;
}
-static int is_nid_hp_pin(struct auto_pin_cfg *cfg, hda_nid_t nid)
+static int is_nid_out_jack_pin(struct auto_pin_cfg *cfg, hda_nid_t nid)
{
int i;
for (i = 0; i < cfg->hp_outs; i++)
if (cfg->hp_pins[i] == nid)
return 1; /* nid is a HP-Out */
-
+ for (i = 0; i < cfg->line_outs; i++)
+ if (cfg->line_out_pins[i] == nid)
+ return 1; /* nid is a line-Out */
return 0; /* nid is not a HP-Out */
};
@@ -4354,7 +4356,7 @@ static int stac92xx_init(struct hda_codec *codec)
continue;
}
- if (is_nid_hp_pin(cfg, nid))
+ if (is_nid_out_jack_pin(cfg, nid))
continue; /* already has an unsol event */
pinctl = snd_hda_codec_read(codec, nid, 0,
@@ -4587,7 +4589,7 @@ static void stac92xx_hp_detect(struct hda_codec *codec)
unsigned int val = AC_PINCTL_OUT_EN | AC_PINCTL_HP_EN;
if (no_hp_sensing(spec, i))
continue;
- if (presence)
+ if (1 /*presence*/)
stac92xx_set_pinctl(codec, cfg->hp_pins[i], val);
#if 0 /* FIXME */
/* Resetting the pinctl like below may lead to (a sort of) regressions
diff --git a/sound/pci/ice1712/amp.c b/sound/pci/ice1712/amp.c
index e328cfb..e525da2 100644
--- a/sound/pci/ice1712/amp.c
+++ b/sound/pci/ice1712/amp.c
@@ -68,8 +68,11 @@ static int __devinit snd_vt1724_amp_init(struct snd_ice1712 *ice)
static int __devinit snd_vt1724_amp_add_controls(struct snd_ice1712 *ice)
{
- /* we use pins 39 and 41 of the VT1616 for left and right read outputs */
- snd_ac97_write_cache(ice->ac97, 0x5a, snd_ac97_read(ice->ac97, 0x5a) & ~0x8000);
+ if (ice->ac97)
+ /* we use pins 39 and 41 of the VT1616 for left and right
+ read outputs */
+ snd_ac97_write_cache(ice->ac97, 0x5a,
+ snd_ac97_read(ice->ac97, 0x5a) & ~0x8000);
return 0;
}
diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c
index 6c896db..2e799a9 100644
--- a/sound/pci/intel8x0.c
+++ b/sound/pci/intel8x0.c
@@ -2076,6 +2076,12 @@ static struct ac97_quirk ac97_quirks[] __devinitdata = {
},
{
.subvendor = 0x161f,
+ .subdevice = 0x202f,
+ .name = "Gateway M520",
+ .type = AC97_TUNE_INV_EAPD
+ },
+ {
+ .subvendor = 0x161f,
.subdevice = 0x203a,
.name = "Gateway 4525GZ", /* AD1981B */
.type = AC97_TUNE_INV_EAPD
diff --git a/sound/pci/oxygen/xonar_wm87x6.c b/sound/pci/oxygen/xonar_wm87x6.c
index 42d1ab1..915546a 100644
--- a/sound/pci/oxygen/xonar_wm87x6.c
+++ b/sound/pci/oxygen/xonar_wm87x6.c
@@ -177,6 +177,7 @@ static void wm8776_registers_init(struct oxygen *chip)
struct xonar_wm87x6 *data = chip->model_data;
wm8776_write(chip, WM8776_RESET, 0);
+ wm8776_write(chip, WM8776_PHASESWAP, WM8776_PH_MASK);
wm8776_write(chip, WM8776_DACCTRL1, WM8776_DZCEN |
WM8776_PL_LEFT_LEFT | WM8776_PL_RIGHT_RIGHT);
wm8776_write(chip, WM8776_DACMUTE, chip->dac_mute ? WM8776_DMUTE : 0);
diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c
index 79c1b3d..7d45197 100644
--- a/sound/soc/codecs/ak4642.c
+++ b/sound/soc/codecs/ak4642.c
@@ -143,7 +143,7 @@
* min : 0xFE : -115.0 dB
* mute: 0xFF
*/
-static const DECLARE_TLV_DB_SCALE(out_tlv, -11500, 50, 1);
+static const DECLARE_TLV_DB_SCALE(out_tlv, -11550, 50, 1);
static const struct snd_kcontrol_new ak4642_snd_controls[] = {
diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index 4a0f666..c850e3d 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -2373,7 +2373,7 @@ static int out_pga_event(struct snd_soc_dapm_widget *w,
}
}
-static const char *st_text[] = { "None", "Right", "Left" };
+static const char *st_text[] = { "None", "Left", "Right" };
static const struct soc_enum str_enum =
SOC_ENUM_SINGLE(WM8962_DAC_DSP_MIXING_1, 2, 3, st_text);
@@ -2975,13 +2975,13 @@ static int wm8962_hw_params(struct snd_pcm_substream *substream,
case SNDRV_PCM_FORMAT_S16_LE:
break;
case SNDRV_PCM_FORMAT_S20_3LE:
- aif0 |= 0x40;
+ aif0 |= 0x4;
break;
case SNDRV_PCM_FORMAT_S24_LE:
- aif0 |= 0x80;
+ aif0 |= 0x8;
break;
case SNDRV_PCM_FORMAT_S32_LE:
- aif0 |= 0xc0;
+ aif0 |= 0xc;
break;
default:
return -EINVAL;
diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c
index 54cbfe4..92c5cda 100644
--- a/sound/soc/codecs/wm_hubs.c
+++ b/sound/soc/codecs/wm_hubs.c
@@ -652,6 +652,8 @@ SND_SOC_DAPM_INPUT("IN2RP:VXRP"),
SND_SOC_DAPM_MICBIAS("MICBIAS2", WM8993_POWER_MANAGEMENT_1, 5, 0),
SND_SOC_DAPM_MICBIAS("MICBIAS1", WM8993_POWER_MANAGEMENT_1, 4, 0),
+SND_SOC_DAPM_SUPPLY("LINEOUT_VMID_BUF", WM8993_ANTIPOP1, 7, 0, NULL, 0),
+
SND_SOC_DAPM_MIXER("IN1L PGA", WM8993_POWER_MANAGEMENT_2, 6, 0,
in1l_pga, ARRAY_SIZE(in1l_pga)),
SND_SOC_DAPM_MIXER("IN1R PGA", WM8993_POWER_MANAGEMENT_2, 4, 0,
@@ -876,9 +878,11 @@ static const struct snd_soc_dapm_route lineout1_diff_routes[] = {
};
static const struct snd_soc_dapm_route lineout1_se_routes[] = {
+ { "LINEOUT1N Mixer", NULL, "LINEOUT_VMID_BUF" },
{ "LINEOUT1N Mixer", "Left Output Switch", "Left Output PGA" },
{ "LINEOUT1N Mixer", "Right Output Switch", "Right Output PGA" },
+ { "LINEOUT1P Mixer", NULL, "LINEOUT_VMID_BUF" },
{ "LINEOUT1P Mixer", "Left Output Switch", "Left Output PGA" },
{ "LINEOUT1N Driver", NULL, "LINEOUT1N Mixer" },
@@ -895,9 +899,11 @@ static const struct snd_soc_dapm_route lineout2_diff_routes[] = {
};
static const struct snd_soc_dapm_route lineout2_se_routes[] = {
+ { "LINEOUT2N Mixer", NULL, "LINEOUT_VMID_BUF" },
{ "LINEOUT2N Mixer", "Left Output Switch", "Left Output PGA" },
{ "LINEOUT2N Mixer", "Right Output Switch", "Right Output PGA" },
+ { "LINEOUT2P Mixer", NULL, "LINEOUT_VMID_BUF" },
{ "LINEOUT2P Mixer", "Right Output Switch", "Right Output PGA" },
{ "LINEOUT2N Driver", NULL, "LINEOUT2N Mixer" },
diff --git a/sound/soc/imx/imx-ssi.c b/sound/soc/imx/imx-ssi.c
index 61fceb0..3b56254 100644
--- a/sound/soc/imx/imx-ssi.c
+++ b/sound/soc/imx/imx-ssi.c
@@ -112,7 +112,7 @@ static int imx_ssi_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt)
break;
case SND_SOC_DAIFMT_DSP_A:
/* data on rising edge of bclk, frame high 1clk before data */
- strcr |= SSI_STCR_TFSL | SSI_STCR_TEFS;
+ strcr |= SSI_STCR_TFSL | SSI_STCR_TXBIT0 | SSI_STCR_TEFS;
break;
}
diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c
index 8ad93ee..b583e60 100644
--- a/sound/soc/pxa/pxa-ssp.c
+++ b/sound/soc/pxa/pxa-ssp.c
@@ -668,6 +668,38 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream,
return 0;
}
+static void pxa_ssp_set_running_bit(struct snd_pcm_substream *substream,
+ struct ssp_device *ssp, int value)
+{
+ uint32_t sscr0 = pxa_ssp_read_reg(ssp, SSCR0);
+ uint32_t sscr1 = pxa_ssp_read_reg(ssp, SSCR1);
+ uint32_t sspsp = pxa_ssp_read_reg(ssp, SSPSP);
+ uint32_t sssr = pxa_ssp_read_reg(ssp, SSSR);
+
+ if (value && (sscr0 & SSCR0_SSE))
+ pxa_ssp_write_reg(ssp, SSCR0, sscr0 & ~SSCR0_SSE);
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ if (value)
+ sscr1 |= SSCR1_TSRE;
+ else
+ sscr1 &= ~SSCR1_TSRE;
+ } else {
+ if (value)
+ sscr1 |= SSCR1_RSRE;
+ else
+ sscr1 &= ~SSCR1_RSRE;
+ }
+
+ pxa_ssp_write_reg(ssp, SSCR1, sscr1);
+
+ if (value) {
+ pxa_ssp_write_reg(ssp, SSSR, sssr);
+ pxa_ssp_write_reg(ssp, SSPSP, sspsp);
+ pxa_ssp_write_reg(ssp, SSCR0, sscr0 | SSCR0_SSE);
+ }
+}
+
static int pxa_ssp_trigger(struct snd_pcm_substream *substream, int cmd,
struct snd_soc_dai *cpu_dai)
{
@@ -681,42 +713,21 @@ static int pxa_ssp_trigger(struct snd_pcm_substream *substream, int cmd,
pxa_ssp_enable(ssp);
break;
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
- val = pxa_ssp_read_reg(ssp, SSCR1);
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
- val |= SSCR1_TSRE;
- else
- val |= SSCR1_RSRE;
- pxa_ssp_write_reg(ssp, SSCR1, val);
+ pxa_ssp_set_running_bit(substream, ssp, 1);
val = pxa_ssp_read_reg(ssp, SSSR);
pxa_ssp_write_reg(ssp, SSSR, val);
break;
case SNDRV_PCM_TRIGGER_START:
- val = pxa_ssp_read_reg(ssp, SSCR1);
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
- val |= SSCR1_TSRE;
- else
- val |= SSCR1_RSRE;
- pxa_ssp_write_reg(ssp, SSCR1, val);
- pxa_ssp_enable(ssp);
+ pxa_ssp_set_running_bit(substream, ssp, 1);
break;
case SNDRV_PCM_TRIGGER_STOP:
- val = pxa_ssp_read_reg(ssp, SSCR1);
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
- val &= ~SSCR1_TSRE;
- else
- val &= ~SSCR1_RSRE;
- pxa_ssp_write_reg(ssp, SSCR1, val);
+ pxa_ssp_set_running_bit(substream, ssp, 0);
break;
case SNDRV_PCM_TRIGGER_SUSPEND:
pxa_ssp_disable(ssp);
break;
case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
- val = pxa_ssp_read_reg(ssp, SSCR1);
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
- val &= ~SSCR1_TSRE;
- else
- val &= ~SSCR1_RSRE;
- pxa_ssp_write_reg(ssp, SSCR1, val);
+ pxa_ssp_set_running_bit(substream, ssp, 0);
break;
default:
diff --git a/sound/soc/samsung/neo1973_wm8753.c b/sound/soc/samsung/neo1973_wm8753.c
index 16152ed..c1290da 100644
--- a/sound/soc/samsung/neo1973_wm8753.c
+++ b/sound/soc/samsung/neo1973_wm8753.c
@@ -425,7 +425,7 @@ static struct snd_soc_dai_link neo1973_dai[] = {
.platform_name = "samsung-audio",
.cpu_dai_name = "s3c24xx-iis",
.codec_dai_name = "wm8753-hifi",
- .codec_name = "wm8753-codec.0-001a",
+ .codec_name = "wm8753.0-001a",
.init = neo1973_wm8753_init,
.ops = &neo1973_hifi_ops,
},
@@ -434,7 +434,7 @@ static struct snd_soc_dai_link neo1973_dai[] = {
.stream_name = "Voice",
.cpu_dai_name = "dfbmcs320-pcm",
.codec_dai_name = "wm8753-voice",
- .codec_name = "wm8753-codec.0-001a",
+ .codec_name = "wm8753.0-001a",
.ops = &neo1973_voice_ops,
},
};
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 7ed17b7..2fb57cf 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -30,6 +30,7 @@
#include <linux/bitops.h>
#include <linux/debugfs.h>
#include <linux/platform_device.h>
+#include <linux/ctype.h>
#include <linux/slab.h>
#include <sound/ac97_codec.h>
#include <sound/core.h>
@@ -1934,9 +1935,20 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card)
"%s", card->name);
snprintf(card->snd_card->longname, sizeof(card->snd_card->longname),
"%s", card->long_name ? card->long_name : card->name);
- if (card->driver_name)
- strlcpy(card->snd_card->driver, card->driver_name,
- sizeof(card->snd_card->driver));
+ snprintf(card->snd_card->driver, sizeof(card->snd_card->driver),
+ "%s", card->driver_name ? card->driver_name : card->name);
+ for (i = 0; i < ARRAY_SIZE(card->snd_card->driver); i++) {
+ switch (card->snd_card->driver[i]) {
+ case '_':
+ case '-':
+ case '\0':
+ break;
+ default:
+ if (!isalnum(card->snd_card->driver[i]))
+ card->snd_card->driver[i] = '_';
+ break;
+ }
+ }
if (card->late_probe) {
ret = card->late_probe(card);
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 3e79714..d34c862 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -2647,9 +2647,13 @@ static void soc_dapm_shutdown_codec(struct snd_soc_dapm_context *dapm)
* standby.
*/
if (powerdown) {
- snd_soc_dapm_set_bias_level(dapm, SND_SOC_BIAS_PREPARE);
+ if (dapm->bias_level == SND_SOC_BIAS_ON)
+ snd_soc_dapm_set_bias_level(dapm,
+ SND_SOC_BIAS_PREPARE);
dapm_seq_run(dapm, &down_list, 0, false);
- snd_soc_dapm_set_bias_level(dapm, SND_SOC_BIAS_STANDBY);
+ if (dapm->bias_level == SND_SOC_BIAS_PREPARE)
+ snd_soc_dapm_set_bias_level(dapm,
+ SND_SOC_BIAS_STANDBY);
}
}
@@ -2662,7 +2666,9 @@ void snd_soc_dapm_shutdown(struct snd_soc_card *card)
list_for_each_entry(codec, &card->codec_dev_list, list) {
soc_dapm_shutdown_codec(&codec->dapm);
- snd_soc_dapm_set_bias_level(&codec->dapm, SND_SOC_BIAS_OFF);
+ if (codec->dapm.bias_level == SND_SOC_BIAS_STANDBY)
+ snd_soc_dapm_set_bias_level(&codec->dapm,
+ SND_SOC_BIAS_OFF);
}
}
diff --git a/sound/usb/usx2y/usb_stream.c b/sound/usb/usx2y/usb_stream.c
index c400ade..1e7a47a 100644
--- a/sound/usb/usx2y/usb_stream.c
+++ b/sound/usb/usx2y/usb_stream.c
@@ -674,7 +674,7 @@ dotry:
inurb->transfer_buffer_length =
inurb->number_of_packets *
inurb->iso_frame_desc[0].length;
- preempt_disable();
+
if (u == 0) {
int now;
struct usb_device *dev = inurb->dev;
@@ -686,19 +686,17 @@ dotry:
}
err = usb_submit_urb(inurb, GFP_ATOMIC);
if (err < 0) {
- preempt_enable();
snd_printk(KERN_ERR"usb_submit_urb(sk->inurb[%i])"
" returned %i\n", u, err);
return err;
}
err = usb_submit_urb(outurb, GFP_ATOMIC);
if (err < 0) {
- preempt_enable();
snd_printk(KERN_ERR"usb_submit_urb(sk->outurb[%i])"
" returned %i\n", u, err);
return err;
}
- preempt_enable();
+
if (inurb->start_frame != outurb->start_frame) {
snd_printd(KERN_DEBUG
"u[%i] start_frames differ in:%u out:%u\n",