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-rw-r--r--sound/pci/emu10k1/p16v.c2
-rw-r--r--sound/pci/hda/hda_beep.c11
-rw-r--r--sound/pci/hda/hda_codec.c14
-rw-r--r--sound/pci/hda/hda_intel.c29
-rw-r--r--sound/pci/hda/patch_analog.c28
-rw-r--r--sound/pci/hda/patch_ca0110.c2
-rw-r--r--sound/pci/hda/patch_realtek.c36
-rw-r--r--sound/pci/hda/patch_sigmatel.c4
-rw-r--r--sound/pci/hda/patch_via.c89
-rw-r--r--sound/pci/oxygen/virtuoso.c2
-rw-r--r--sound/soc/codecs/wm8753.c6
-rw-r--r--sound/soc/fsl/Kconfig6
-rw-r--r--sound/soc/fsl/mpc5200_dma.c1
-rw-r--r--sound/soc/fsl/mpc5200_dma.h1
-rw-r--r--sound/soc/fsl/mpc5200_psc_ac97.c17
-rw-r--r--sound/soc/omap/omap-pcm.c11
-rw-r--r--sound/soc/pxa/pxa2xx-i2s.c7
-rw-r--r--sound/sound_core.c5
-rw-r--r--sound/usb/caiaq/device.c10
-rw-r--r--sound/usb/usx2y/us122l.c2
-rw-r--r--sound/usb/usx2y/usbusx2y.c2
-rw-r--r--sound/usb/usx2y/usbusx2yaudio.c7
22 files changed, 190 insertions, 102 deletions
diff --git a/sound/pci/emu10k1/p16v.c b/sound/pci/emu10k1/p16v.c
index e617acaf10e..61b8ab39800 100644
--- a/sound/pci/emu10k1/p16v.c
+++ b/sound/pci/emu10k1/p16v.c
@@ -644,7 +644,7 @@ int __devinit snd_p16v_pcm(struct snd_emu10k1 *emu, int device, struct snd_pcm *
int err;
int capture=1;
- /* snd_printk("KERN_DEBUG snd_p16v_pcm called. device=%d\n", device); */
+ /* snd_printk(KERN_DEBUG "snd_p16v_pcm called. device=%d\n", device); */
emu->p16v_device_offset = device;
if (rpcm)
*rpcm = NULL;
diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c
index 29272f2e95a..b0275a05087 100644
--- a/sound/pci/hda/hda_beep.c
+++ b/sound/pci/hda/hda_beep.c
@@ -50,19 +50,22 @@ static void snd_hda_generate_beep(struct work_struct *work)
* The tone frequency of beep generator on IDT/STAC codecs is
* defined from the 8bit tone parameter, in Hz,
* freq = 48000 * (257 - tone) / 1024
- * that is from 12kHz to 93.75kHz in step of 46.875 hz
+ * that is from 12kHz to 93.75Hz in steps of 46.875 Hz
*/
static int beep_linear_tone(struct hda_beep *beep, int hz)
{
+ if (hz <= 0)
+ return 0;
hz *= 1000; /* fixed point */
- hz = hz - DIGBEEP_HZ_MIN;
+ hz = hz - DIGBEEP_HZ_MIN
+ + DIGBEEP_HZ_STEP / 2; /* round to nearest step */
if (hz < 0)
hz = 0; /* turn off PC beep*/
else if (hz >= (DIGBEEP_HZ_MAX - DIGBEEP_HZ_MIN))
- hz = 0xff;
+ hz = 1; /* max frequency */
else {
hz /= DIGBEEP_HZ_STEP;
- hz++;
+ hz = 255 - hz;
}
return hz;
}
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 462e2cedaa6..26d255de6be 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -3470,10 +3470,16 @@ int snd_hda_multi_out_analog_open(struct hda_codec *codec,
}
mutex_lock(&codec->spdif_mutex);
if (mout->share_spdif) {
- runtime->hw.rates &= mout->spdif_rates;
- runtime->hw.formats &= mout->spdif_formats;
- if (mout->spdif_maxbps < hinfo->maxbps)
- hinfo->maxbps = mout->spdif_maxbps;
+ if ((runtime->hw.rates & mout->spdif_rates) &&
+ (runtime->hw.formats & mout->spdif_formats)) {
+ runtime->hw.rates &= mout->spdif_rates;
+ runtime->hw.formats &= mout->spdif_formats;
+ if (mout->spdif_maxbps < hinfo->maxbps)
+ hinfo->maxbps = mout->spdif_maxbps;
+ } else {
+ mout->share_spdif = 0;
+ /* FIXME: need notify? */
+ }
}
mutex_unlock(&codec->spdif_mutex);
}
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 4e9ea708027..77c1b840ca8 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -1454,6 +1454,18 @@ static int azx_pcm_open(struct snd_pcm_substream *substream)
mutex_unlock(&chip->open_mutex);
return err;
}
+ snd_pcm_limit_hw_rates(runtime);
+ /* sanity check */
+ if (snd_BUG_ON(!runtime->hw.channels_min) ||
+ snd_BUG_ON(!runtime->hw.channels_max) ||
+ snd_BUG_ON(!runtime->hw.formats) ||
+ snd_BUG_ON(!runtime->hw.rates)) {
+ azx_release_device(azx_dev);
+ hinfo->ops.close(hinfo, apcm->codec, substream);
+ snd_hda_power_down(apcm->codec);
+ mutex_unlock(&chip->open_mutex);
+ return -EINVAL;
+ }
spin_lock_irqsave(&chip->reg_lock, flags);
azx_dev->substream = substream;
azx_dev->running = 0;
@@ -1462,7 +1474,6 @@ static int azx_pcm_open(struct snd_pcm_substream *substream)
runtime->private_data = azx_dev;
snd_pcm_set_sync(substream);
mutex_unlock(&chip->open_mutex);
-
return 0;
}
@@ -2322,9 +2333,19 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci,
gcap = azx_readw(chip, GCAP);
snd_printdd(SFX "chipset global capabilities = 0x%x\n", gcap);
- /* ATI chips seems buggy about 64bit DMA addresses */
- if (chip->driver_type == AZX_DRIVER_ATI)
- gcap &= ~ICH6_GCAP_64OK;
+ /* disable SB600 64bit support for safety */
+ if ((chip->driver_type == AZX_DRIVER_ATI) ||
+ (chip->driver_type == AZX_DRIVER_ATIHDMI)) {
+ struct pci_dev *p_smbus;
+ p_smbus = pci_get_device(PCI_VENDOR_ID_ATI,
+ PCI_DEVICE_ID_ATI_SBX00_SMBUS,
+ NULL);
+ if (p_smbus) {
+ if (p_smbus->revision < 0x30)
+ gcap &= ~ICH6_GCAP_64OK;
+ pci_dev_put(p_smbus);
+ }
+ }
/* allow 64bit DMA address if supported by H/W */
if ((gcap & ICH6_GCAP_64OK) && !pci_set_dma_mask(pci, DMA_BIT_MASK(64)))
diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index 1988582d1ab..be7d25fa7f3 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -3746,9 +3746,30 @@ static struct snd_kcontrol_new ad1884a_laptop_mixers[] = {
{ } /* end */
};
+static int ad1884a_mobile_master_sw_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ int ret = snd_hda_mixer_amp_switch_put(kcontrol, ucontrol);
+ int mute = (!ucontrol->value.integer.value[0] &&
+ !ucontrol->value.integer.value[1]);
+ /* toggle GPIO1 according to the mute state */
+ snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA,
+ mute ? 0x02 : 0x0);
+ return ret;
+}
+
static struct snd_kcontrol_new ad1884a_mobile_mixers[] = {
HDA_CODEC_VOLUME("Master Playback Volume", 0x21, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT),
+ /*HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT),*/
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Master Playback Switch",
+ .info = snd_hda_mixer_amp_switch_info,
+ .get = snd_hda_mixer_amp_switch_get,
+ .put = ad1884a_mobile_master_sw_put,
+ .private_value = HDA_COMPOSE_AMP_VAL(0x21, 3, 0, HDA_OUTPUT),
+ },
HDA_CODEC_VOLUME("PCM Playback Volume", 0x20, 0x5, HDA_INPUT),
HDA_CODEC_MUTE("PCM Playback Switch", 0x20, 0x5, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Capture Volume", 0x14, 0x0, HDA_INPUT),
@@ -3869,6 +3890,10 @@ static struct hda_verb ad1884a_mobile_verbs[] = {
/* unsolicited event for pin-sense */
{0x11, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_HP_EVENT},
{0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_MIC_EVENT},
+ /* allow to touch GPIO1 (for mute control) */
+ {0x01, AC_VERB_SET_GPIO_MASK, 0x02},
+ {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x02},
+ {0x01, AC_VERB_SET_GPIO_DATA, 0x02}, /* first muted */
{ } /* end */
};
@@ -3978,6 +4003,7 @@ static struct snd_pci_quirk ad1884a_cfg_tbl[] = {
SND_PCI_QUIRK(0x103c, 0x3037, "HP 2230s", AD1884A_LAPTOP),
SND_PCI_QUIRK(0x103c, 0x3056, "HP", AD1884A_MOBILE),
SND_PCI_QUIRK_MASK(0x103c, 0xfff0, 0x3070, "HP", AD1884A_MOBILE),
+ SND_PCI_QUIRK_MASK(0x103c, 0xfff0, 0x30d0, "HP laptop", AD1884A_LAPTOP),
SND_PCI_QUIRK_MASK(0x103c, 0xfff0, 0x30e0, "HP laptop", AD1884A_LAPTOP),
SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x3600, "HP laptop", AD1884A_LAPTOP),
SND_PCI_QUIRK(0x17aa, 0x20ac, "Thinkpad X300", AD1884A_THINKPAD),
diff --git a/sound/pci/hda/patch_ca0110.c b/sound/pci/hda/patch_ca0110.c
index 392d108c355..019ca7cb56d 100644
--- a/sound/pci/hda/patch_ca0110.c
+++ b/sound/pci/hda/patch_ca0110.c
@@ -510,7 +510,7 @@ static int ca0110_parse_auto_config(struct hda_codec *codec)
}
-int patch_ca0110(struct hda_codec *codec)
+static int patch_ca0110(struct hda_codec *codec)
{
struct ca0110_spec *spec;
int err;
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 3a8e58c483d..bbb9b42e260 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -6919,9 +6919,6 @@ static struct hda_verb alc882_targa_verbs[] = {
{0x1b, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */
{0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
- {0x01, AC_VERB_SET_GPIO_MASK, 0x03},
- {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x03},
- {0x01, AC_VERB_SET_GPIO_DATA, 0x03},
{ } /* end */
};
@@ -7241,7 +7238,8 @@ static struct alc_config_preset alc882_presets[] = {
},
[ALC882_TARGA] = {
.mixers = { alc882_targa_mixer, alc882_chmode_mixer },
- .init_verbs = { alc882_init_verbs, alc882_targa_verbs},
+ .init_verbs = { alc882_init_verbs, alc880_gpio3_init_verbs,
+ alc882_targa_verbs},
.num_dacs = ARRAY_SIZE(alc882_dac_nids),
.dac_nids = alc882_dac_nids,
.dig_out_nid = ALC882_DIGOUT_NID,
@@ -9238,7 +9236,8 @@ static struct alc_config_preset alc883_presets[] = {
},
[ALC883_TARGA_DIG] = {
.mixers = { alc883_targa_mixer, alc883_chmode_mixer },
- .init_verbs = { alc883_init_verbs, alc883_targa_verbs},
+ .init_verbs = { alc883_init_verbs, alc880_gpio3_init_verbs,
+ alc883_targa_verbs},
.num_dacs = ARRAY_SIZE(alc883_dac_nids),
.dac_nids = alc883_dac_nids,
.dig_out_nid = ALC883_DIGOUT_NID,
@@ -9251,7 +9250,8 @@ static struct alc_config_preset alc883_presets[] = {
},
[ALC883_TARGA_2ch_DIG] = {
.mixers = { alc883_targa_2ch_mixer},
- .init_verbs = { alc883_init_verbs, alc883_targa_verbs},
+ .init_verbs = { alc883_init_verbs, alc880_gpio3_init_verbs,
+ alc883_targa_verbs},
.num_dacs = ARRAY_SIZE(alc883_dac_nids),
.dac_nids = alc883_dac_nids,
.adc_nids = alc883_adc_nids_alt,
@@ -12876,20 +12876,11 @@ static struct snd_kcontrol_new alc269_lifebook_mixer[] = {
{ }
};
-/* bind volumes of both NID 0x0c and 0x0d */
-static struct hda_bind_ctls alc269_epc_bind_vol = {
- .ops = &snd_hda_bind_vol,
- .values = {
- HDA_COMPOSE_AMP_VAL(0x02, 3, 0, HDA_OUTPUT),
- HDA_COMPOSE_AMP_VAL(0x03, 3, 0, HDA_OUTPUT),
- 0
- },
-};
-
static struct snd_kcontrol_new alc269_eeepc_mixer[] = {
- HDA_CODEC_MUTE("iSpeaker Playback Switch", 0x14, 0x0, HDA_OUTPUT),
- HDA_BIND_VOL("LineOut Playback Volume", &alc269_epc_bind_vol),
- HDA_CODEC_MUTE("LineOut Playback Switch", 0x15, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT),
{ } /* end */
};
@@ -12902,12 +12893,7 @@ static struct snd_kcontrol_new alc269_epc_capture_mixer[] = {
};
/* FSC amilo */
-static struct snd_kcontrol_new alc269_fujitsu_mixer[] = {
- HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
- HDA_BIND_VOL("PCM Playback Volume", &alc269_epc_bind_vol),
- { } /* end */
-};
+#define alc269_fujitsu_mixer alc269_eeepc_mixer
static struct hda_verb alc269_quanta_fl1_verbs[] = {
{0x15, AC_VERB_SET_CONNECT_SEL, 0x01},
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 14f3c3e0f62..41b5b3a18c1 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -1590,8 +1590,6 @@ static struct snd_pci_quirk stac9200_cfg_tbl[] = {
/* SigmaTel reference board */
SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668,
"DFI LanParty", STAC_REF),
- SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0xfb30,
- "SigmaTel",STAC_9205_REF),
SND_PCI_QUIRK(PCI_VENDOR_ID_DFI, 0x3101,
"DFI LanParty", STAC_REF),
/* Dell laptops have BIOS problem */
@@ -2344,6 +2342,8 @@ static struct snd_pci_quirk stac9205_cfg_tbl[] = {
/* SigmaTel reference board */
SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668,
"DFI LanParty", STAC_9205_REF),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0xfb30,
+ "SigmaTel", STAC_9205_REF),
SND_PCI_QUIRK(PCI_VENDOR_ID_DFI, 0x3101,
"DFI LanParty", STAC_9205_REF),
/* Dell */
diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index 8e004fb6961..9008b4b013a 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -210,7 +210,9 @@ struct via_spec {
/* capture */
unsigned int num_adc_nids;
hda_nid_t *adc_nids;
+ hda_nid_t mux_nids[3];
hda_nid_t dig_in_nid;
+ hda_nid_t dig_in_pin;
/* capture source */
const struct hda_input_mux *input_mux;
@@ -319,6 +321,9 @@ static void via_auto_set_output_and_unmute(struct hda_codec *codec,
pin_type);
snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE,
AMP_OUT_UNMUTE);
+ if (snd_hda_query_pin_caps(codec, nid) & AC_PINCAP_EAPD)
+ snd_hda_codec_write(codec, nid, 0,
+ AC_VERB_SET_EAPD_BTLENABLE, 0x02);
}
@@ -387,27 +392,12 @@ static int via_mux_enum_put(struct snd_kcontrol *kcontrol,
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
struct via_spec *spec = codec->spec;
unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
- unsigned int vendor_id = codec->vendor_id;
-
- /* AIW0 lydia 060801 add for correct sw0 input select */
- if (IS_VT1708_VENDORID(vendor_id) && (adc_idx == 0))
- return snd_hda_input_mux_put(codec, spec->input_mux, ucontrol,
- 0x18, &spec->cur_mux[adc_idx]);
- else if ((IS_VT1709_10CH_VENDORID(vendor_id) ||
- IS_VT1709_6CH_VENDORID(vendor_id)) && (adc_idx == 0))
- return snd_hda_input_mux_put(codec, spec->input_mux, ucontrol,
- 0x19, &spec->cur_mux[adc_idx]);
- else if ((IS_VT1708B_8CH_VENDORID(vendor_id) ||
- IS_VT1708B_4CH_VENDORID(vendor_id)) && (adc_idx == 0))
- return snd_hda_input_mux_put(codec, spec->input_mux, ucontrol,
- 0x17, &spec->cur_mux[adc_idx]);
- else if (IS_VT1702_VENDORID(vendor_id) && (adc_idx == 0))
- return snd_hda_input_mux_put(codec, spec->input_mux, ucontrol,
- 0x13, &spec->cur_mux[adc_idx]);
- else
- return snd_hda_input_mux_put(codec, spec->input_mux, ucontrol,
- spec->adc_nids[adc_idx],
- &spec->cur_mux[adc_idx]);
+
+ if (!spec->mux_nids[adc_idx])
+ return -EINVAL;
+ return snd_hda_input_mux_put(codec, spec->input_mux, ucontrol,
+ spec->mux_nids[adc_idx],
+ &spec->cur_mux[adc_idx]);
}
static int via_independent_hp_info(struct snd_kcontrol *kcontrol,
@@ -998,25 +988,11 @@ static int via_init(struct hda_codec *codec)
/* Lydia Add for EAPD enable */
if (!spec->dig_in_nid) { /* No Digital In connection */
- if (IS_VT1708_VENDORID(codec->vendor_id)) {
- snd_hda_codec_write(codec, VT1708_DIGIN_PIN, 0,
- AC_VERB_SET_PIN_WIDGET_CONTROL,
- PIN_OUT);
- snd_hda_codec_write(codec, VT1708_DIGIN_PIN, 0,
- AC_VERB_SET_EAPD_BTLENABLE, 0x02);
- } else if (IS_VT1709_10CH_VENDORID(codec->vendor_id) ||
- IS_VT1709_6CH_VENDORID(codec->vendor_id)) {
- snd_hda_codec_write(codec, VT1709_DIGIN_PIN, 0,
+ if (spec->dig_in_pin) {
+ snd_hda_codec_write(codec, spec->dig_in_pin, 0,
AC_VERB_SET_PIN_WIDGET_CONTROL,
PIN_OUT);
- snd_hda_codec_write(codec, VT1709_DIGIN_PIN, 0,
- AC_VERB_SET_EAPD_BTLENABLE, 0x02);
- } else if (IS_VT1708B_8CH_VENDORID(codec->vendor_id) ||
- IS_VT1708B_4CH_VENDORID(codec->vendor_id)) {
- snd_hda_codec_write(codec, VT1708B_DIGIN_PIN, 0,
- AC_VERB_SET_PIN_WIDGET_CONTROL,
- PIN_OUT);
- snd_hda_codec_write(codec, VT1708B_DIGIN_PIN, 0,
+ snd_hda_codec_write(codec, spec->dig_in_pin, 0,
AC_VERB_SET_EAPD_BTLENABLE, 0x02);
}
} else /* enable SPDIF-input pin */
@@ -1326,6 +1302,7 @@ static int vt1708_parse_auto_config(struct hda_codec *codec)
if (spec->autocfg.dig_outs)
spec->multiout.dig_out_nid = VT1708_DIGOUT_NID;
+ spec->dig_in_pin = VT1708_DIGIN_PIN;
if (spec->autocfg.dig_in_pin)
spec->dig_in_nid = VT1708_DIGIN_NID;
@@ -1352,6 +1329,34 @@ static int via_auto_init(struct hda_codec *codec)
return 0;
}
+static int get_mux_nids(struct hda_codec *codec)
+{
+ struct via_spec *spec = codec->spec;
+ hda_nid_t nid, conn[8];
+ unsigned int type;
+ int i, n;
+
+ for (i = 0; i < spec->num_adc_nids; i++) {
+ nid = spec->adc_nids[i];
+ while (nid) {
+ type = (get_wcaps(codec, nid) & AC_WCAP_TYPE)
+ >> AC_WCAP_TYPE_SHIFT;
+ if (type == AC_WID_PIN)
+ break;
+ n = snd_hda_get_connections(codec, nid, conn,
+ ARRAY_SIZE(conn));
+ if (n <= 0)
+ break;
+ if (n > 1) {
+ spec->mux_nids[i] = nid;
+ break;
+ }
+ nid = conn[0];
+ }
+ }
+ return 0;
+}
+
static int patch_vt1708(struct hda_codec *codec)
{
struct via_spec *spec;
@@ -1799,6 +1804,7 @@ static int vt1709_parse_auto_config(struct hda_codec *codec)
if (spec->autocfg.dig_outs)
spec->multiout.dig_out_nid = VT1709_DIGOUT_NID;
+ spec->dig_in_pin = VT1709_DIGIN_PIN;
if (spec->autocfg.dig_in_pin)
spec->dig_in_nid = VT1709_DIGIN_NID;
@@ -1859,6 +1865,7 @@ static int patch_vt1709_10ch(struct hda_codec *codec)
if (!spec->adc_nids && spec->input_mux) {
spec->adc_nids = vt1709_adc_nids;
spec->num_adc_nids = ARRAY_SIZE(vt1709_adc_nids);
+ get_mux_nids(codec);
spec->mixers[spec->num_mixers] = vt1709_capture_mixer;
spec->num_mixers++;
}
@@ -1952,6 +1959,7 @@ static int patch_vt1709_6ch(struct hda_codec *codec)
if (!spec->adc_nids && spec->input_mux) {
spec->adc_nids = vt1709_adc_nids;
spec->num_adc_nids = ARRAY_SIZE(vt1709_adc_nids);
+ get_mux_nids(codec);
spec->mixers[spec->num_mixers] = vt1709_capture_mixer;
spec->num_mixers++;
}
@@ -2344,6 +2352,7 @@ static int vt1708B_parse_auto_config(struct hda_codec *codec)
if (spec->autocfg.dig_outs)
spec->multiout.dig_out_nid = VT1708B_DIGOUT_NID;
+ spec->dig_in_pin = VT1708B_DIGIN_PIN;
if (spec->autocfg.dig_in_pin)
spec->dig_in_nid = VT1708B_DIGIN_NID;
@@ -2404,6 +2413,7 @@ static int patch_vt1708B_8ch(struct hda_codec *codec)
if (!spec->adc_nids && spec->input_mux) {
spec->adc_nids = vt1708B_adc_nids;
spec->num_adc_nids = ARRAY_SIZE(vt1708B_adc_nids);
+ get_mux_nids(codec);
spec->mixers[spec->num_mixers] = vt1708B_capture_mixer;
spec->num_mixers++;
}
@@ -2455,6 +2465,7 @@ static int patch_vt1708B_4ch(struct hda_codec *codec)
if (!spec->adc_nids && spec->input_mux) {
spec->adc_nids = vt1708B_adc_nids;
spec->num_adc_nids = ARRAY_SIZE(vt1708B_adc_nids);
+ get_mux_nids(codec);
spec->mixers[spec->num_mixers] = vt1708B_capture_mixer;
spec->num_mixers++;
}
@@ -2889,6 +2900,7 @@ static int patch_vt1708S(struct hda_codec *codec)
if (!spec->adc_nids && spec->input_mux) {
spec->adc_nids = vt1708S_adc_nids;
spec->num_adc_nids = ARRAY_SIZE(vt1708S_adc_nids);
+ get_mux_nids(codec);
spec->mixers[spec->num_mixers] = vt1708S_capture_mixer;
spec->num_mixers++;
}
@@ -3206,6 +3218,7 @@ static int patch_vt1702(struct hda_codec *codec)
if (!spec->adc_nids && spec->input_mux) {
spec->adc_nids = vt1702_adc_nids;
spec->num_adc_nids = ARRAY_SIZE(vt1702_adc_nids);
+ get_mux_nids(codec);
spec->mixers[spec->num_mixers] = vt1702_capture_mixer;
spec->num_mixers++;
}
diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c
index bf971f7cfdc..6ebcb6bdd71 100644
--- a/sound/pci/oxygen/virtuoso.c
+++ b/sound/pci/oxygen/virtuoso.c
@@ -635,6 +635,8 @@ static void xonar_d2_resume(struct oxygen *chip)
static void xonar_d1_resume(struct oxygen *chip)
{
+ oxygen_set_bits8(chip, OXYGEN_FUNCTION, OXYGEN_FUNCTION_RESET_CODEC);
+ msleep(1);
cs43xx_init(chip);
xonar_enable_output(chip);
}
diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c
index d28eeaceb85..49c4b2898af 100644
--- a/sound/soc/codecs/wm8753.c
+++ b/sound/soc/codecs/wm8753.c
@@ -79,7 +79,7 @@ static const u16 wm8753_reg[] = {
0x0097, 0x0097, 0x0000, 0x0004,
0x0000, 0x0083, 0x0024, 0x01ba,
0x0000, 0x0083, 0x0024, 0x01ba,
- 0x0000, 0x0000
+ 0x0000, 0x0000, 0x0000
};
/* codec private data */
@@ -1660,11 +1660,11 @@ static int wm8753_register(struct wm8753_priv *wm8753)
codec->set_bias_level = wm8753_set_bias_level;
codec->dai = wm8753_dai;
codec->num_dai = 2;
- codec->reg_cache_size = ARRAY_SIZE(wm8753->reg_cache);
+ codec->reg_cache_size = ARRAY_SIZE(wm8753->reg_cache) + 1;
codec->reg_cache = &wm8753->reg_cache;
codec->private_data = wm8753;
- memcpy(codec->reg_cache, wm8753_reg, sizeof(codec->reg_cache));
+ memcpy(codec->reg_cache, wm8753_reg, sizeof(wm8753->reg_cache));
INIT_DELAYED_WORK(&codec->delayed_work, wm8753_work);
ret = wm8753_reset(codec);
diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig
index 5dbebf82249..8cb65ccad35 100644
--- a/sound/soc/fsl/Kconfig
+++ b/sound/soc/fsl/Kconfig
@@ -33,7 +33,7 @@ config SND_SOC_MPC5200_I2S
config SND_SOC_MPC5200_AC97
tristate "Freescale MPC5200 PSC in AC97 mode driver"
depends on PPC_MPC52xx && PPC_BESTCOMM
- select AC97_BUS
+ select SND_SOC_AC97_BUS
select SND_MPC52xx_DMA
select PPC_BESTCOMM_GEN_BD
help
@@ -41,7 +41,7 @@ config SND_SOC_MPC5200_AC97
config SND_MPC52xx_SOC_PCM030
tristate "SoC AC97 Audio support for Phytec pcm030 and WM9712"
- depends on PPC_MPC5200_SIMPLE && BROKEN
+ depends on PPC_MPC5200_SIMPLE
select SND_SOC_MPC5200_AC97
select SND_SOC_WM9712
help
@@ -50,7 +50,7 @@ config SND_MPC52xx_SOC_PCM030
config SND_MPC52xx_SOC_EFIKA
tristate "SoC AC97 Audio support for bbplan Efika and STAC9766"
- depends on PPC_EFIKA && BROKEN
+ depends on PPC_EFIKA
select SND_SOC_MPC5200_AC97
select SND_SOC_STAC9766
help
diff --git a/sound/soc/fsl/mpc5200_dma.c b/sound/soc/fsl/mpc5200_dma.c
index efec33a1c5b..f0a2d407199 100644
--- a/sound/soc/fsl/mpc5200_dma.c
+++ b/sound/soc/fsl/mpc5200_dma.c
@@ -456,6 +456,7 @@ int mpc5200_audio_dma_create(struct of_device *op)
return -ENODEV;
spin_lock_init(&psc_dma->lock);
+ mutex_init(&psc_dma->mutex);
psc_dma->id = be32_to_cpu(*prop);
psc_dma->irq = irq;
psc_dma->psc_regs = regs;
diff --git a/sound/soc/fsl/mpc5200_dma.h b/sound/soc/fsl/mpc5200_dma.h
index 2000803f06a..8d396bb9d9f 100644
--- a/sound/soc/fsl/mpc5200_dma.h
+++ b/sound/soc/fsl/mpc5200_dma.h
@@ -55,6 +55,7 @@ struct psc_dma {
unsigned int irq;
struct device *dev;
spinlock_t lock;
+ struct mutex mutex;
u32 sicr;
uint sysclk;
int imr;
diff --git a/sound/soc/fsl/mpc5200_psc_ac97.c b/sound/soc/fsl/mpc5200_psc_ac97.c
index 794a247b3eb..7eb549985d4 100644
--- a/sound/soc/fsl/mpc5200_psc_ac97.c
+++ b/sound/soc/fsl/mpc5200_psc_ac97.c
@@ -34,13 +34,20 @@ static unsigned short psc_ac97_read(struct snd_ac97 *ac97, unsigned short reg)
int status;
unsigned int val;
+ mutex_lock(&psc_dma->mutex);
+
/* Wait for command send status zero = ready */
status = spin_event_timeout(!(in_be16(&psc_dma->psc_regs->sr_csr.status) &
MPC52xx_PSC_SR_CMDSEND), 100, 0);
if (status == 0) {
pr_err("timeout on ac97 bus (rdy)\n");
+ mutex_unlock(&psc_dma->mutex);
return -ENODEV;
}
+
+ /* Force clear the data valid bit */
+ in_be32(&psc_dma->psc_regs->ac97_data);
+
/* Send the read */
out_be32(&psc_dma->psc_regs->ac97_cmd, (1<<31) | ((reg & 0x7f) << 24));
@@ -50,16 +57,19 @@ static unsigned short psc_ac97_read(struct snd_ac97 *ac97, unsigned short reg)
if (status == 0) {
pr_err("timeout on ac97 read (val) %x\n",
in_be16(&psc_dma->psc_regs->sr_csr.status));
+ mutex_unlock(&psc_dma->mutex);
return -ENODEV;
}
/* Get the data */
val = in_be32(&psc_dma->psc_regs->ac97_data);
if (((val >> 24) & 0x7f) != reg) {
pr_err("reg echo error on ac97 read\n");
+ mutex_unlock(&psc_dma->mutex);
return -ENODEV;
}
val = (val >> 8) & 0xffff;
+ mutex_unlock(&psc_dma->mutex);
return (unsigned short) val;
}
@@ -68,16 +78,21 @@ static void psc_ac97_write(struct snd_ac97 *ac97,
{
int status;
+ mutex_lock(&psc_dma->mutex);
+
/* Wait for command status zero = ready */
status = spin_event_timeout(!(in_be16(&psc_dma->psc_regs->sr_csr.status) &
MPC52xx_PSC_SR_CMDSEND), 100, 0);
if (status == 0) {
pr_err("timeout on ac97 bus (write)\n");
- return;
+ goto out;
}
/* Write data */
out_be32(&psc_dma->psc_regs->ac97_cmd,
((reg & 0x7f) << 24) | (val << 8));
+
+ out:
+ mutex_unlock(&psc_dma->mutex);
}
static void psc_ac97_warm_reset(struct snd_ac97 *ac97)
diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c
index 6454e15f7d2..84a1950880e 100644
--- a/sound/soc/omap/omap-pcm.c
+++ b/sound/soc/omap/omap-pcm.c
@@ -216,12 +216,15 @@ static snd_pcm_uframes_t omap_pcm_pointer(struct snd_pcm_substream *substream)
dma_addr_t ptr;
snd_pcm_uframes_t offset;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
- ptr = omap_get_dma_src_pos(prtd->dma_ch);
- else
+ if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
ptr = omap_get_dma_dst_pos(prtd->dma_ch);
+ offset = bytes_to_frames(runtime, ptr - runtime->dma_addr);
+ } else if (!(cpu_is_omap1510())) {
+ ptr = omap_get_dma_src_pos(prtd->dma_ch);
+ offset = bytes_to_frames(runtime, ptr - runtime->dma_addr);
+ } else
+ offset = prtd->period_index * runtime->period_size;
- offset = bytes_to_frames(runtime, ptr - runtime->dma_addr);
if (offset >= runtime->buffer_size)
offset = 0;
diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c
index 4743e262895..6b8f655d1ad 100644
--- a/sound/soc/pxa/pxa2xx-i2s.c
+++ b/sound/soc/pxa/pxa2xx-i2s.c
@@ -167,6 +167,7 @@ static int pxa2xx_i2s_hw_params(struct snd_pcm_substream *substream,
BUG_ON(IS_ERR(clk_i2s));
clk_enable(clk_i2s);
+ dai->private_data = dai;
pxa_i2s_wait();
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
@@ -255,7 +256,10 @@ static void pxa2xx_i2s_shutdown(struct snd_pcm_substream *substream,
if ((SACR1 & (SACR1_DREC | SACR1_DRPL)) == (SACR1_DREC | SACR1_DRPL)) {
SACR0 &= ~SACR0_ENB;
pxa_i2s_wait();
- clk_disable(clk_i2s);
+ if (dai->private_data != NULL) {
+ clk_disable(clk_i2s);
+ dai->private_data = NULL;
+ }
}
}
@@ -336,6 +340,7 @@ static int pxa2xx_i2s_probe(struct platform_device *dev)
return PTR_ERR(clk_i2s);
pxa_i2s_dai.dev = &dev->dev;
+ pxa_i2s_dai.private_data = NULL;
ret = snd_soc_register_dai(&pxa_i2s_dai);
if (ret != 0)
clk_put(clk_i2s);
diff --git a/sound/sound_core.c b/sound/sound_core.c
index 12522e6913d..a41f8b127f4 100644
--- a/sound/sound_core.c
+++ b/sound/sound_core.c
@@ -10,6 +10,8 @@
#include <linux/module.h>
#include <linux/device.h>
#include <linux/err.h>
+#include <linux/kdev_t.h>
+#include <linux/major.h>
#include <sound/core.h>
#ifdef CONFIG_SOUND_OSS_CORE
@@ -29,6 +31,8 @@ MODULE_LICENSE("GPL");
static char *sound_nodename(struct device *dev)
{
+ if (MAJOR(dev->devt) == SOUND_MAJOR)
+ return NULL;
return kasprintf(GFP_KERNEL, "snd/%s", dev_name(dev));
}
@@ -104,7 +108,6 @@ module_exit(cleanup_soundcore);
#include <linux/types.h>
#include <linux/kernel.h>
#include <linux/sound.h>
-#include <linux/major.h>
#include <linux/kmod.h>
#define SOUND_STEP 16
diff --git a/sound/usb/caiaq/device.c b/sound/usb/caiaq/device.c
index 0e5db719de2..de38108f0b2 100644
--- a/sound/usb/caiaq/device.c
+++ b/sound/usb/caiaq/device.c
@@ -35,7 +35,7 @@
#include "input.h"
MODULE_AUTHOR("Daniel Mack <daniel@caiaq.de>");
-MODULE_DESCRIPTION("caiaq USB audio, version 1.3.17");
+MODULE_DESCRIPTION("caiaq USB audio, version 1.3.18");
MODULE_LICENSE("GPL");
MODULE_SUPPORTED_DEVICE("{{Native Instruments, RigKontrol2},"
"{Native Instruments, RigKontrol3},"
@@ -349,7 +349,9 @@ static void __devinit setup_card(struct snd_usb_caiaqdev *dev)
log("Unable to set up control system (ret=%d)\n", ret);
}
-static int create_card(struct usb_device* usb_dev, struct snd_card **cardp)
+static int create_card(struct usb_device *usb_dev,
+ struct usb_interface *intf,
+ struct snd_card **cardp)
{
int devnum;
int err;
@@ -374,7 +376,7 @@ static int create_card(struct usb_device* usb_dev, struct snd_card **cardp)
dev->chip.usb_id = USB_ID(le16_to_cpu(usb_dev->descriptor.idVendor),
le16_to_cpu(usb_dev->descriptor.idProduct));
spin_lock_init(&dev->spinlock);
- snd_card_set_dev(card, &usb_dev->dev);
+ snd_card_set_dev(card, &intf->dev);
*cardp = card;
return 0;
@@ -461,7 +463,7 @@ static int __devinit snd_probe(struct usb_interface *intf,
struct snd_card *card;
struct usb_device *device = interface_to_usbdev(intf);
- ret = create_card(device, &card);
+ ret = create_card(device, intf, &card);
if (ret < 0)
return ret;
diff --git a/sound/usb/usx2y/us122l.c b/sound/usb/usx2y/us122l.c
index a5aae9d67f3..fd44946ce4b 100644
--- a/sound/usb/usx2y/us122l.c
+++ b/sound/usb/usx2y/us122l.c
@@ -514,7 +514,6 @@ static int usx2y_create_card(struct usb_device *device, struct snd_card **cardp)
US122L(card)->chip.dev->bus->busnum,
US122L(card)->chip.dev->devnum
);
- snd_card_set_dev(card, &device->dev);
*cardp = card;
return 0;
}
@@ -531,6 +530,7 @@ static int us122l_usb_probe(struct usb_interface *intf,
if (err < 0)
return err;
+ snd_card_set_dev(card, &intf->dev);
if (!us122l_create_card(card)) {
snd_card_free(card);
return -EINVAL;
diff --git a/sound/usb/usx2y/usbusx2y.c b/sound/usb/usx2y/usbusx2y.c
index 5ce0da23ee9..cb4bb8373ca 100644
--- a/sound/usb/usx2y/usbusx2y.c
+++ b/sound/usb/usx2y/usbusx2y.c
@@ -364,7 +364,6 @@ static int usX2Y_create_card(struct usb_device *device, struct snd_card **cardp)
0,//us428(card)->usbmidi.ifnum,
usX2Y(card)->chip.dev->bus->busnum, usX2Y(card)->chip.dev->devnum
);
- snd_card_set_dev(card, &device->dev);
*cardp = card;
return 0;
}
@@ -388,6 +387,7 @@ static int usX2Y_usb_probe(struct usb_device *device,
err = usX2Y_create_card(device, &card);
if (err < 0)
return err;
+ snd_card_set_dev(card, &intf->dev);
if ((err = usX2Y_hwdep_new(card, device)) < 0 ||
(err = snd_card_register(card)) < 0) {
snd_card_free(card);
diff --git a/sound/usb/usx2y/usbusx2yaudio.c b/sound/usb/usx2y/usbusx2yaudio.c
index dd1ab617784..9efd27f6b52 100644
--- a/sound/usb/usx2y/usbusx2yaudio.c
+++ b/sound/usb/usx2y/usbusx2yaudio.c
@@ -296,9 +296,10 @@ static void usX2Y_error_urb_status(struct usX2Ydev *usX2Y,
static void usX2Y_error_sequence(struct usX2Ydev *usX2Y,
struct snd_usX2Y_substream *subs, struct urb *urb)
{
- snd_printk(KERN_ERR "Sequence Error!(hcd_frame=%i ep=%i%s;wait=%i,frame=%i).\n"
- KERN_ERR "Most propably some urb of usb-frame %i is still missing.\n"
- KERN_ERR "Cause could be too long delays in usb-hcd interrupt handling.\n",
+ snd_printk(KERN_ERR
+"Sequence Error!(hcd_frame=%i ep=%i%s;wait=%i,frame=%i).\n"
+"Most propably some urb of usb-frame %i is still missing.\n"
+"Cause could be too long delays in usb-hcd interrupt handling.\n",
usb_get_current_frame_number(usX2Y->chip.dev),
subs->endpoint, usb_pipein(urb->pipe) ? "in" : "out",
usX2Y->wait_iso_frame, urb->start_frame, usX2Y->wait_iso_frame);
generated by cgit (git 2.34.1) at 2024-09-28 05:22:31 +0000