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-rw-r--r--sound/core/pcm_native.c4
-rw-r--r--sound/pci/hda/hda_generic.c68
-rw-r--r--sound/pci/hda/hda_generic.h1
-rw-r--r--sound/pci/hda/patch_realtek.c3
-rw-r--r--sound/pci/hda/patch_via.c10
-rw-r--r--sound/pci/sis7019.c3
-rw-r--r--sound/soc/codecs/cs42l52.c12
-rw-r--r--sound/soc/codecs/cs42l52.h2
-rw-r--r--sound/soc/codecs/max98090.c2
-rw-r--r--sound/soc/codecs/tlv320aic3x.c10
-rw-r--r--sound/soc/codecs/wm5102.c3
-rw-r--r--sound/soc/codecs/wm5110.c7
-rw-r--r--sound/soc/codecs/wm8994.c15
-rw-r--r--sound/soc/davinci/davinci-mcasp.c7
-rw-r--r--sound/soc/soc-compress.c8
-rw-r--r--sound/soc/soc-dapm.c49
-rw-r--r--sound/soc/soc-pcm.c13
-rw-r--r--sound/usb/6fire/firmware.c6
-rw-r--r--sound/usb/mixer.c1
-rw-r--r--sound/usb/quirks-table.h14
20 files changed, 151 insertions, 87 deletions
diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c
index ccfa383f1fda..f92818155958 100644
--- a/sound/core/pcm_native.c
+++ b/sound/core/pcm_native.c
@@ -1649,6 +1649,7 @@ static int snd_pcm_link(struct snd_pcm_substream *substream, int fd)
}
if (!snd_pcm_stream_linked(substream)) {
substream->group = group;
+ group = NULL;
spin_lock_init(&substream->group->lock);
INIT_LIST_HEAD(&substream->group->substreams);
list_add_tail(&substream->link_list, &substream->group->substreams);
@@ -1663,8 +1664,7 @@ static int snd_pcm_link(struct snd_pcm_substream *substream, int fd)
_nolock:
snd_card_unref(substream1->pcm->card);
fput_light(file, fput_needed);
- if (res < 0)
- kfree(group);
+ kfree(group);
return res;
}
diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c
index ae85bbd2e6f8..4b1524a861f3 100644
--- a/sound/pci/hda/hda_generic.c
+++ b/sound/pci/hda/hda_generic.c
@@ -788,6 +788,8 @@ static void set_pin_eapd(struct hda_codec *codec, hda_nid_t pin, bool enable)
return;
if (codec->inv_eapd)
enable = !enable;
+ if (spec->keep_eapd_on && !enable)
+ return;
snd_hda_codec_update_cache(codec, pin, 0,
AC_VERB_SET_EAPD_BTLENABLE,
enable ? 0x02 : 0x00);
@@ -1938,17 +1940,7 @@ static int create_speaker_out_ctls(struct hda_codec *codec)
* independent HP controls
*/
-/* update HP auto-mute state too */
-static void update_hp_automute_hook(struct hda_codec *codec)
-{
- struct hda_gen_spec *spec = codec->spec;
-
- if (spec->hp_automute_hook)
- spec->hp_automute_hook(codec, NULL);
- else
- snd_hda_gen_hp_automute(codec, NULL);
-}
-
+static void call_hp_automute(struct hda_codec *codec, struct hda_jack_tbl *jack);
static int indep_hp_info(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
@@ -2009,7 +2001,7 @@ static int indep_hp_put(struct snd_kcontrol *kcontrol,
else
*dacp = spec->alt_dac_nid;
- update_hp_automute_hook(codec);
+ call_hp_automute(codec, NULL);
ret = 1;
}
unlock:
@@ -2305,7 +2297,7 @@ static void update_hp_mic(struct hda_codec *codec, int adc_mux, bool force)
else
val = PIN_HP;
set_pin_target(codec, pin, val, true);
- update_hp_automute_hook(codec);
+ call_hp_automute(codec, NULL);
}
}
@@ -2714,7 +2706,7 @@ static int hp_mic_jack_mode_put(struct snd_kcontrol *kcontrol,
val = snd_hda_get_default_vref(codec, nid);
}
snd_hda_set_pin_ctl_cache(codec, nid, val);
- update_hp_automute_hook(codec);
+ call_hp_automute(codec, NULL);
return 1;
}
@@ -3859,20 +3851,42 @@ void snd_hda_gen_mic_autoswitch(struct hda_codec *codec, struct hda_jack_tbl *ja
}
EXPORT_SYMBOL_HDA(snd_hda_gen_mic_autoswitch);
-/* update jack retasking */
-static void update_automute_all(struct hda_codec *codec)
+/* call appropriate hooks */
+static void call_hp_automute(struct hda_codec *codec, struct hda_jack_tbl *jack)
{
struct hda_gen_spec *spec = codec->spec;
+ if (spec->hp_automute_hook)
+ spec->hp_automute_hook(codec, jack);
+ else
+ snd_hda_gen_hp_automute(codec, jack);
+}
- update_hp_automute_hook(codec);
+static void call_line_automute(struct hda_codec *codec,
+ struct hda_jack_tbl *jack)
+{
+ struct hda_gen_spec *spec = codec->spec;
if (spec->line_automute_hook)
- spec->line_automute_hook(codec, NULL);
+ spec->line_automute_hook(codec, jack);
else
- snd_hda_gen_line_automute(codec, NULL);
+ snd_hda_gen_line_automute(codec, jack);
+}
+
+static void call_mic_autoswitch(struct hda_codec *codec,
+ struct hda_jack_tbl *jack)
+{
+ struct hda_gen_spec *spec = codec->spec;
if (spec->mic_autoswitch_hook)
- spec->mic_autoswitch_hook(codec, NULL);
+ spec->mic_autoswitch_hook(codec, jack);
else
- snd_hda_gen_mic_autoswitch(codec, NULL);
+ snd_hda_gen_mic_autoswitch(codec, jack);
+}
+
+/* update jack retasking */
+static void update_automute_all(struct hda_codec *codec)
+{
+ call_hp_automute(codec, NULL);
+ call_line_automute(codec, NULL);
+ call_mic_autoswitch(codec, NULL);
}
/*
@@ -4009,9 +4023,7 @@ static int check_auto_mute_availability(struct hda_codec *codec)
snd_printdd("hda-codec: Enable HP auto-muting on NID 0x%x\n",
nid);
snd_hda_jack_detect_enable_callback(codec, nid, HDA_GEN_HP_EVENT,
- spec->hp_automute_hook ?
- spec->hp_automute_hook :
- snd_hda_gen_hp_automute);
+ call_hp_automute);
spec->detect_hp = 1;
}
@@ -4024,9 +4036,7 @@ static int check_auto_mute_availability(struct hda_codec *codec)
snd_printdd("hda-codec: Enable Line-Out auto-muting on NID 0x%x\n", nid);
snd_hda_jack_detect_enable_callback(codec, nid,
HDA_GEN_FRONT_EVENT,
- spec->line_automute_hook ?
- spec->line_automute_hook :
- snd_hda_gen_line_automute);
+ call_line_automute);
spec->detect_lo = 1;
}
spec->automute_lo_possible = spec->detect_hp;
@@ -4068,9 +4078,7 @@ static bool auto_mic_check_imux(struct hda_codec *codec)
snd_hda_jack_detect_enable_callback(codec,
spec->am_entry[i].pin,
HDA_GEN_MIC_EVENT,
- spec->mic_autoswitch_hook ?
- spec->mic_autoswitch_hook :
- snd_hda_gen_mic_autoswitch);
+ call_mic_autoswitch);
return true;
}
diff --git a/sound/pci/hda/hda_generic.h b/sound/pci/hda/hda_generic.h
index 54e665160379..76200314ee95 100644
--- a/sound/pci/hda/hda_generic.h
+++ b/sound/pci/hda/hda_generic.h
@@ -222,6 +222,7 @@ struct hda_gen_spec {
unsigned int multi_cap_vol:1; /* allow multiple capture xxx volumes */
unsigned int inv_dmic_split:1; /* inverted dmic w/a for conexant */
unsigned int own_eapd_ctl:1; /* set EAPD by own function */
+ unsigned int keep_eapd_on:1; /* don't turn off EAPD automatically */
unsigned int vmaster_mute_enum:1; /* add vmaster mute mode enum */
unsigned int indep_hp:1; /* independent HP supported */
unsigned int prefer_hp_amp:1; /* enable HP amp for speaker if any */
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 59d2e91a9ab6..02e22b4458d2 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -3493,6 +3493,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x1028, 0x05f4, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x1028, 0x05f5, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x1028, 0x05f6, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE),
+ SND_PCI_QUIRK(0x1028, 0x05f8, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE),
+ SND_PCI_QUIRK(0x1028, 0x0609, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC2),
SND_PCI_QUIRK(0x103c, 0x18e6, "HP", ALC269_FIXUP_HP_GPIO_LED),
SND_PCI_QUIRK(0x103c, 0x1973, "HP Pavilion", ALC269_FIXUP_HP_MUTE_LED_MIC1),
@@ -3530,6 +3532,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x17aa, 0x21fa, "Thinkpad X230", ALC269_FIXUP_LENOVO_DOCK),
SND_PCI_QUIRK(0x17aa, 0x21f3, "Thinkpad T430", ALC269_FIXUP_LENOVO_DOCK),
SND_PCI_QUIRK(0x17aa, 0x21fb, "Thinkpad T430s", ALC269_FIXUP_LENOVO_DOCK),
+ SND_PCI_QUIRK(0x17aa, 0x2208, "Thinkpad T431s", ALC269_FIXUP_LENOVO_DOCK),
SND_PCI_QUIRK(0x17aa, 0x2203, "Thinkpad X230 Tablet", ALC269_FIXUP_LENOVO_DOCK),
SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K),
SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD),
diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index e0dadcf2030d..e5245544eb52 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -136,6 +136,7 @@ static struct via_spec *via_new_spec(struct hda_codec *codec)
spec->codec_type = VT1708S;
spec->no_pin_power_ctl = 1;
spec->gen.indep_hp = 1;
+ spec->gen.keep_eapd_on = 1;
spec->gen.pcm_playback_hook = via_playback_pcm_hook;
return spec;
}
@@ -231,9 +232,14 @@ static void vt1708_update_hp_work(struct hda_codec *codec)
static void set_widgets_power_state(struct hda_codec *codec)
{
+#if 0 /* FIXME: the assumed connections don't match always with the
+ * actual routes by the generic parser, so better to disable
+ * the control for safety.
+ */
struct via_spec *spec = codec->spec;
if (spec->set_widgets_power_state)
spec->set_widgets_power_state(codec);
+#endif
}
static void update_power_state(struct hda_codec *codec, hda_nid_t nid,
@@ -478,7 +484,9 @@ static int via_suspend(struct hda_codec *codec)
/* Fix pop noise on headphones */
int i;
for (i = 0; i < spec->gen.autocfg.hp_outs; i++)
- snd_hda_set_pin_ctl(codec, spec->gen.autocfg.hp_pins[i], 0);
+ snd_hda_codec_write(codec, spec->gen.autocfg.hp_pins[i],
+ 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
+ 0x00);
}
return 0;
diff --git a/sound/pci/sis7019.c b/sound/pci/sis7019.c
index d59abe1682c5..748e82d4d257 100644
--- a/sound/pci/sis7019.c
+++ b/sound/pci/sis7019.c
@@ -1341,7 +1341,8 @@ static int sis_chip_create(struct snd_card *card,
if (rc)
goto error_out;
- if (pci_set_dma_mask(pci, DMA_BIT_MASK(30)) < 0) {
+ rc = pci_set_dma_mask(pci, DMA_BIT_MASK(30));
+ if (rc < 0) {
dev_err(&pci->dev, "architecture does not support 30-bit PCI busmaster DMA");
goto error_out_enabled;
}
diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c
index 0f6f481cec09..987f728718c5 100644
--- a/sound/soc/codecs/cs42l52.c
+++ b/sound/soc/codecs/cs42l52.c
@@ -86,7 +86,7 @@ static const struct reg_default cs42l52_reg_defaults[] = {
{ CS42L52_BEEP_VOL, 0x00 }, /* r1D Beep Volume off Time */
{ CS42L52_BEEP_TONE_CTL, 0x00 }, /* r1E Beep Tone Cfg. */
{ CS42L52_TONE_CTL, 0x00 }, /* r1F Tone Ctl */
- { CS42L52_MASTERA_VOL, 0x88 }, /* r20 Master A Volume */
+ { CS42L52_MASTERA_VOL, 0x00 }, /* r20 Master A Volume */
{ CS42L52_MASTERB_VOL, 0x00 }, /* r21 Master B Volume */
{ CS42L52_HPA_VOL, 0x00 }, /* r22 Headphone A Volume */
{ CS42L52_HPB_VOL, 0x00 }, /* r23 Headphone B Volume */
@@ -193,6 +193,8 @@ static DECLARE_TLV_DB_SCALE(mic_tlv, 1600, 100, 0);
static DECLARE_TLV_DB_SCALE(pga_tlv, -600, 50, 0);
+static DECLARE_TLV_DB_SCALE(mix_tlv, -50, 50, 0);
+
static const unsigned int limiter_tlv[] = {
TLV_DB_RANGE_HEAD(2),
0, 2, TLV_DB_SCALE_ITEM(-3000, 600, 0),
@@ -225,7 +227,7 @@ static const char * const mic_bias_level_text[] = {
};
static const struct soc_enum mic_bias_level_enum =
- SOC_ENUM_SINGLE(CS42L52_IFACE_CTL1, 0,
+ SOC_ENUM_SINGLE(CS42L52_IFACE_CTL2, 0,
ARRAY_SIZE(mic_bias_level_text), mic_bias_level_text);
static const char * const cs42l52_mic_text[] = { "Single", "Differential" };
@@ -260,7 +262,7 @@ static const char * const hp_gain_num_text[] = {
};
static const struct soc_enum hp_gain_enum =
- SOC_ENUM_SINGLE(CS42L52_PB_CTL1, 4,
+ SOC_ENUM_SINGLE(CS42L52_PB_CTL1, 5,
ARRAY_SIZE(hp_gain_num_text), hp_gain_num_text);
static const char * const beep_pitch_text[] = {
@@ -413,7 +415,7 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = {
SOC_ENUM("Headphone Analog Gain", hp_gain_enum),
SOC_DOUBLE_R_SX_TLV("Speaker Volume", CS42L52_SPKA_VOL,
- CS42L52_SPKB_VOL, 7, 0x1, 0xff, hl_tlv),
+ CS42L52_SPKB_VOL, 0, 0x1, 0xff, hl_tlv),
SOC_DOUBLE_R_SX_TLV("Bypass Volume", CS42L52_PASSTHRUA_VOL,
CS42L52_PASSTHRUB_VOL, 6, 0x18, 0x90, pga_tlv),
@@ -441,7 +443,7 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = {
SOC_DOUBLE_R_SX_TLV("PCM Mixer Volume",
CS42L52_PCMA_MIXER_VOL, CS42L52_PCMB_MIXER_VOL,
- 6, 0x7f, 0x19, hl_tlv),
+ 0, 0x7f, 0x19, mix_tlv),
SOC_DOUBLE_R("PCM Mixer Switch",
CS42L52_PCMA_MIXER_VOL, CS42L52_PCMB_MIXER_VOL, 7, 1, 1),
diff --git a/sound/soc/codecs/cs42l52.h b/sound/soc/codecs/cs42l52.h
index 60985c059071..4277012c4719 100644
--- a/sound/soc/codecs/cs42l52.h
+++ b/sound/soc/codecs/cs42l52.h
@@ -157,7 +157,7 @@
#define CS42L52_PB_CTL1_INV_PCMA (1 << 2)
#define CS42L52_PB_CTL1_MSTB_MUTE (1 << 1)
#define CS42L52_PB_CTL1_MSTA_MUTE (1 << 0)
-#define CS42L52_PB_CTL1_MUTE_MASK 0xFFFD
+#define CS42L52_PB_CTL1_MUTE_MASK 0x03
#define CS42L52_PB_CTL1_MUTE 3
#define CS42L52_PB_CTL1_UNMUTE 0
diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c
index ce0d36412c97..8d14a76c7249 100644
--- a/sound/soc/codecs/max98090.c
+++ b/sound/soc/codecs/max98090.c
@@ -2233,7 +2233,7 @@ static int max98090_probe(struct snd_soc_codec *codec)
dev_dbg(codec->dev, "irq = %d\n", max98090->irq);
ret = request_threaded_irq(max98090->irq, NULL,
- max98090_interrupt, IRQF_TRIGGER_FALLING,
+ max98090_interrupt, IRQF_TRIGGER_FALLING | IRQF_ONESHOT,
"max98090_interrupt", codec);
if (ret < 0) {
dev_err(codec->dev, "request_irq failed: %d\n",
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index 65d09d60b7c6..1514bf845e4b 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -187,14 +187,14 @@ static int snd_soc_dapm_put_volsw_aic3x(struct snd_kcontrol *kcontrol,
break;
}
-
- if (found)
- snd_soc_dapm_sync(widget->dapm);
}
- ret = snd_soc_update_bits(widget->codec, reg, val_mask, val);
-
mutex_unlock(&widget->codec->mutex);
+
+ if (found)
+ snd_soc_dapm_sync(widget->dapm);
+
+ ret = snd_soc_update_bits_locked(widget->codec, reg, val_mask, val);
return ret;
}
diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c
index e895d3939eef..100fdadda56a 100644
--- a/sound/soc/codecs/wm5102.c
+++ b/sound/soc/codecs/wm5102.c
@@ -1120,7 +1120,8 @@ SND_SOC_DAPM_AIF_IN("AIF3RX2", NULL, 0,
ARIZONA_DSP_WIDGETS(DSP1, "DSP1"),
SND_SOC_DAPM_VALUE_MUX("AEC Loopback", ARIZONA_DAC_AEC_CONTROL_1,
- ARIZONA_AEC_LOOPBACK_ENA, 0, &wm5102_aec_loopback_mux),
+ ARIZONA_AEC_LOOPBACK_ENA_SHIFT, 0,
+ &wm5102_aec_loopback_mux),
SND_SOC_DAPM_PGA_E("OUT1L", SND_SOC_NOPM,
ARIZONA_OUT1L_ENA_SHIFT, 0, NULL, 0, arizona_hp_ev,
diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c
index 731884e04776..88ad7db52dde 100644
--- a/sound/soc/codecs/wm5110.c
+++ b/sound/soc/codecs/wm5110.c
@@ -190,7 +190,7 @@ ARIZONA_MIXER_CONTROLS("DSP2R", ARIZONA_DSP2RMIX_INPUT_1_SOURCE),
ARIZONA_MIXER_CONTROLS("DSP3L", ARIZONA_DSP3LMIX_INPUT_1_SOURCE),
ARIZONA_MIXER_CONTROLS("DSP3R", ARIZONA_DSP3RMIX_INPUT_1_SOURCE),
ARIZONA_MIXER_CONTROLS("DSP4L", ARIZONA_DSP4LMIX_INPUT_1_SOURCE),
-ARIZONA_MIXER_CONTROLS("DSP5R", ARIZONA_DSP4RMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("DSP4R", ARIZONA_DSP4RMIX_INPUT_1_SOURCE),
ARIZONA_MIXER_CONTROLS("Mic", ARIZONA_MICMIX_INPUT_1_SOURCE),
ARIZONA_MIXER_CONTROLS("Noise", ARIZONA_NOISEMIX_INPUT_1_SOURCE),
@@ -503,7 +503,8 @@ SND_SOC_DAPM_PGA("ASRC2R", ARIZONA_ASRC_ENABLE, ARIZONA_ASRC2R_ENA_SHIFT, 0,
NULL, 0),
SND_SOC_DAPM_VALUE_MUX("AEC Loopback", ARIZONA_DAC_AEC_CONTROL_1,
- ARIZONA_AEC_LOOPBACK_ENA, 0, &wm5110_aec_loopback_mux),
+ ARIZONA_AEC_LOOPBACK_ENA_SHIFT, 0,
+ &wm5110_aec_loopback_mux),
SND_SOC_DAPM_AIF_OUT("AIF1TX1", NULL, 0,
ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX1_ENA_SHIFT, 0),
@@ -976,6 +977,8 @@ static int wm5110_codec_probe(struct snd_soc_codec *codec)
if (ret != 0)
return ret;
+ arizona_init_spk(codec);
+
snd_soc_dapm_disable_pin(&codec->dapm, "HAPTICS");
priv->core.arizona->dapm = &codec->dapm;
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index 1eb152cb1097..29e95f93d482 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -383,6 +383,8 @@ static int wm8994_get_drc_enum(struct snd_kcontrol *kcontrol,
struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
int drc = wm8994_get_drc(kcontrol->id.name);
+ if (drc < 0)
+ return drc;
ucontrol->value.enumerated.item[0] = wm8994->drc_cfg[drc];
return 0;
@@ -488,6 +490,9 @@ static int wm8994_get_retune_mobile_enum(struct snd_kcontrol *kcontrol,
struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
int block = wm8994_get_retune_mobile_block(kcontrol->id.name);
+ if (block < 0)
+ return block;
+
ucontrol->value.enumerated.item[0] = wm8994->retune_mobile_cfg[block];
return 0;
@@ -1031,7 +1036,7 @@ static int aif1clk_ev(struct snd_soc_dapm_widget *w,
{
struct snd_soc_codec *codec = w->codec;
struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
- struct wm8994 *control = codec->control_data;
+ struct wm8994 *control = wm8994->wm8994;
int mask = WM8994_AIF1DAC1L_ENA | WM8994_AIF1DAC1R_ENA;
int i;
int dac;
@@ -3831,8 +3836,14 @@ static irqreturn_t wm8958_mic_irq(int irq, void *data)
ret);
} else if (!(ret & WM1811_JACKDET_LVL)) {
dev_dbg(codec->dev, "Ignoring removed jack\n");
- return IRQ_HANDLED;
+ goto out;
}
+ } else if (!(reg & WM8958_MICD_STS)) {
+ snd_soc_jack_report(wm8994->micdet[0].jack, 0,
+ SND_JACK_MECHANICAL | SND_JACK_HEADSET |
+ wm8994->btn_mask);
+ wm8994->mic_detecting = true;
+ goto out;
}
if (wm8994->mic_detecting)
diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index 56ecfc72f2e9..81490febac6d 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -631,7 +631,8 @@ static int davinci_config_channel_size(struct davinci_audio_dev *dev,
int word_length)
{
u32 fmt;
- u32 rotate = (word_length / 4) & 0x7;
+ u32 tx_rotate = (word_length / 4) & 0x7;
+ u32 rx_rotate = (32 - word_length) / 4;
u32 mask = (1ULL << word_length) - 1;
/*
@@ -655,9 +656,9 @@ static int davinci_config_channel_size(struct davinci_audio_dev *dev,
mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMT_REG,
TXSSZ(fmt), TXSSZ(0x0F));
mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMT_REG,
- TXROT(rotate), TXROT(7));
+ TXROT(tx_rotate), TXROT(7));
mcasp_mod_bits(dev->base + DAVINCI_MCASP_RXFMT_REG,
- RXROT(rotate), RXROT(7));
+ RXROT(rx_rotate), RXROT(7));
mcasp_set_reg(dev->base + DAVINCI_MCASP_RXMASK_REG,
mask);
}
diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c
index 3853f7eb3f28..06a8000aa07b 100644
--- a/sound/soc/soc-compress.c
+++ b/sound/soc/soc-compress.c
@@ -220,8 +220,12 @@ static int soc_compr_set_params(struct snd_compr_stream *cstream,
goto err;
}
- snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_PLAYBACK,
- SND_SOC_DAPM_STREAM_START);
+ if (cstream->direction == SND_COMPRESS_PLAYBACK)
+ snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_PLAYBACK,
+ SND_SOC_DAPM_STREAM_START);
+ else
+ snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_CAPTURE,
+ SND_SOC_DAPM_STREAM_START);
/* cancel any delayed stream shutdown that is pending */
rtd->pop_wait = 0;
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index a80c883bb8be..c7051c457b75 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -55,7 +55,8 @@ static int dapm_up_seq[] = {
[snd_soc_dapm_clock_supply] = 1,
[snd_soc_dapm_micbias] = 2,
[snd_soc_dapm_dai_link] = 2,
- [snd_soc_dapm_dai] = 3,
+ [snd_soc_dapm_dai_in] = 3,
+ [snd_soc_dapm_dai_out] = 3,
[snd_soc_dapm_aif_in] = 3,
[snd_soc_dapm_aif_out] = 3,
[snd_soc_dapm_mic] = 4,
@@ -92,7 +93,8 @@ static int dapm_down_seq[] = {
[snd_soc_dapm_value_mux] = 9,
[snd_soc_dapm_aif_in] = 10,
[snd_soc_dapm_aif_out] = 10,
- [snd_soc_dapm_dai] = 10,
+ [snd_soc_dapm_dai_in] = 10,
+ [snd_soc_dapm_dai_out] = 10,
[snd_soc_dapm_dai_link] = 11,
[snd_soc_dapm_clock_supply] = 12,
[snd_soc_dapm_regulator_supply] = 12,
@@ -419,7 +421,8 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w,
case snd_soc_dapm_clock_supply:
case snd_soc_dapm_aif_in:
case snd_soc_dapm_aif_out:
- case snd_soc_dapm_dai:
+ case snd_soc_dapm_dai_in:
+ case snd_soc_dapm_dai_out:
case snd_soc_dapm_hp:
case snd_soc_dapm_mic:
case snd_soc_dapm_spk:
@@ -820,7 +823,7 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget,
switch (widget->id) {
case snd_soc_dapm_adc:
case snd_soc_dapm_aif_out:
- case snd_soc_dapm_dai:
+ case snd_soc_dapm_dai_out:
if (widget->active) {
widget->outputs = snd_soc_dapm_suspend_check(widget);
return widget->outputs;
@@ -916,7 +919,7 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget,
switch (widget->id) {
case snd_soc_dapm_dac:
case snd_soc_dapm_aif_in:
- case snd_soc_dapm_dai:
+ case snd_soc_dapm_dai_in:
if (widget->active) {
widget->inputs = snd_soc_dapm_suspend_check(widget);
return widget->inputs;
@@ -1135,16 +1138,6 @@ static int dapm_generic_check_power(struct snd_soc_dapm_widget *w)
return out != 0 && in != 0;
}
-static int dapm_dai_check_power(struct snd_soc_dapm_widget *w)
-{
- DAPM_UPDATE_STAT(w, power_checks);
-
- if (w->active)
- return w->active;
-
- return dapm_generic_check_power(w);
-}
-
/* Check to see if an ADC has power */
static int dapm_adc_check_power(struct snd_soc_dapm_widget *w)
{
@@ -2318,7 +2311,8 @@ static int snd_soc_dapm_add_route(struct snd_soc_dapm_context *dapm,
case snd_soc_dapm_clock_supply:
case snd_soc_dapm_aif_in:
case snd_soc_dapm_aif_out:
- case snd_soc_dapm_dai:
+ case snd_soc_dapm_dai_in:
+ case snd_soc_dapm_dai_out:
case snd_soc_dapm_dai_link:
list_add(&path->list, &dapm->card->paths);
list_add(&path->list_sink, &wsink->sources);
@@ -3129,10 +3123,12 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm,
break;
case snd_soc_dapm_adc:
case snd_soc_dapm_aif_out:
+ case snd_soc_dapm_dai_out:
w->power_check = dapm_adc_check_power;
break;
case snd_soc_dapm_dac:
case snd_soc_dapm_aif_in:
+ case snd_soc_dapm_dai_in:
w->power_check = dapm_dac_check_power;
break;
case snd_soc_dapm_pga:
@@ -3152,9 +3148,6 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm,
case snd_soc_dapm_clock_supply:
w->power_check = dapm_supply_check_power;
break;
- case snd_soc_dapm_dai:
- w->power_check = dapm_dai_check_power;
- break;
default:
w->power_check = dapm_always_on_check_power;
break;
@@ -3375,7 +3368,7 @@ int snd_soc_dapm_new_dai_widgets(struct snd_soc_dapm_context *dapm,
template.reg = SND_SOC_NOPM;
if (dai->driver->playback.stream_name) {
- template.id = snd_soc_dapm_dai;
+ template.id = snd_soc_dapm_dai_in;
template.name = dai->driver->playback.stream_name;
template.sname = dai->driver->playback.stream_name;
@@ -3393,7 +3386,7 @@ int snd_soc_dapm_new_dai_widgets(struct snd_soc_dapm_context *dapm,
}
if (dai->driver->capture.stream_name) {
- template.id = snd_soc_dapm_dai;
+ template.id = snd_soc_dapm_dai_out;
template.name = dai->driver->capture.stream_name;
template.sname = dai->driver->capture.stream_name;
@@ -3423,8 +3416,13 @@ int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card)
/* For each DAI widget... */
list_for_each_entry(dai_w, &card->widgets, list) {
- if (dai_w->id != snd_soc_dapm_dai)
+ switch (dai_w->id) {
+ case snd_soc_dapm_dai_in:
+ case snd_soc_dapm_dai_out:
+ break;
+ default:
continue;
+ }
dai = dai_w->priv;
@@ -3433,8 +3431,13 @@ int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card)
if (w->dapm != dai_w->dapm)
continue;
- if (w->id == snd_soc_dapm_dai)
+ switch (w->id) {
+ case snd_soc_dapm_dai_in:
+ case snd_soc_dapm_dai_out:
continue;
+ default:
+ break;
+ }
if (!w->sname)
continue;
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index 73bb8eefa491..ccb6be4d658d 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -928,8 +928,13 @@ static int dpcm_add_paths(struct snd_soc_pcm_runtime *fe, int stream,
/* Create any new FE <--> BE connections */
for (i = 0; i < list->num_widgets; i++) {
- if (list->widgets[i]->id != snd_soc_dapm_dai)
+ switch (list->widgets[i]->id) {
+ case snd_soc_dapm_dai_in:
+ case snd_soc_dapm_dai_out:
+ break;
+ default:
continue;
+ }
/* is there a valid BE rtd for this widget */
be = dpcm_get_be(card, list->widgets[i], stream);
@@ -2011,9 +2016,11 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num)
if (cpu_dai->driver->capture.channels_min)
capture = 1;
} else {
- if (codec_dai->driver->playback.channels_min)
+ if (codec_dai->driver->playback.channels_min &&
+ cpu_dai->driver->playback.channels_min)
playback = 1;
- if (codec_dai->driver->capture.channels_min)
+ if (codec_dai->driver->capture.channels_min &&
+ cpu_dai->driver->capture.channels_min)
capture = 1;
}
diff --git a/sound/usb/6fire/firmware.c b/sound/usb/6fire/firmware.c
index a1d9b0792a1e..b9defcdeb7ef 100644
--- a/sound/usb/6fire/firmware.c
+++ b/sound/usb/6fire/firmware.c
@@ -42,8 +42,8 @@ static const u8 ep_w_max_packet_size[] = {
0x94, 0x01, 0x5c, 0x02 /* alt 3: 404 EP2 and 604 EP6 (25 fpp) */
};
-static const u8 known_fw_versions[][4] = {
- { 0x03, 0x01, 0x0b, 0x00 }
+static const u8 known_fw_versions[][2] = {
+ { 0x03, 0x01 }
};
struct ihex_record {
@@ -343,7 +343,7 @@ static int usb6fire_fw_check(u8 *version)
int i;
for (i = 0; i < ARRAY_SIZE(known_fw_versions); i++)
- if (!memcmp(version, known_fw_versions + i, 4))
+ if (!memcmp(version, known_fw_versions + i, 2))
return 0;
snd_printk(KERN_ERR PREFIX "invalid fimware version in device: %*ph. "
diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c
index ca4739c3f650..e5c7f9f20fdd 100644
--- a/sound/usb/mixer.c
+++ b/sound/usb/mixer.c
@@ -886,6 +886,7 @@ static void volume_control_quirks(struct usb_mixer_elem_info *cval,
case USB_ID(0x046d, 0x0808):
case USB_ID(0x046d, 0x0809):
case USB_ID(0x046d, 0x081d): /* HD Webcam c510 */
+ case USB_ID(0x046d, 0x0825): /* HD Webcam c270 */
case USB_ID(0x046d, 0x0991):
/* Most audio usb devices lie about volume resolution.
* Most Logitech webcams have res = 384.
diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h
index 7f1722f82c89..8b75bcf136f6 100644
--- a/sound/usb/quirks-table.h
+++ b/sound/usb/quirks-table.h
@@ -215,7 +215,13 @@
.bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL
},
{
- USB_DEVICE(0x046d, 0x0990),
+ .match_flags = USB_DEVICE_ID_MATCH_DEVICE |
+ USB_DEVICE_ID_MATCH_INT_CLASS |
+ USB_DEVICE_ID_MATCH_INT_SUBCLASS,
+ .idVendor = 0x046d,
+ .idProduct = 0x0990,
+ .bInterfaceClass = USB_CLASS_AUDIO,
+ .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL,
.driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
.vendor_name = "Logitech, Inc.",
.product_name = "QuickCam Pro 9000",
@@ -1792,7 +1798,11 @@ YAMAHA_DEVICE(0x7010, "UB99"),
USB_DEVICE_VENDOR_SPEC(0x0582, 0x0108),
.driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
.ifnum = 0,
- .type = QUIRK_MIDI_STANDARD_INTERFACE
+ .type = QUIRK_MIDI_FIXED_ENDPOINT,
+ .data = & (const struct snd_usb_midi_endpoint_info) {
+ .out_cables = 0x0007,
+ .in_cables = 0x0007
+ }
}
},
{
generated by cgit v1.2.3 (git 2.25.1) at 2024-11-24 05:29:06 +0000