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Merge branch 'fix/hda' into for-linus
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* fix/hda:
  ALSA: hda - Add sanity check in PCM open callback
  ALSA: hda - Call snd_pcm_lib_hw_rates() again after codec open callback
  ALSA: hda - Avoid invalid formats and rates with shared SPDIF
  ALSA: hda - Improve ASUS eeePC 1000 mixer
  ALSA: hda - Add GPIO1 control at muting with HP laptops
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Takashi Iwai committed Jul 3, 2009
2 parents dbe45d0 + c470331 commit 854ace9
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Showing 4 changed files with 48 additions and 24 deletions.
14 changes: 10 additions & 4 deletions sound/pci/hda/hda_codec.c
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Original file line number Diff line number Diff line change
Expand Up @@ -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);
}
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7 changes: 7 additions & 0 deletions sound/pci/hda/hda_intel.c
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Original file line number Diff line number Diff line change
Expand Up @@ -1454,6 +1454,7 @@ static int azx_pcm_open(struct snd_pcm_substream *substream)
mutex_unlock(&chip->open_mutex);
return err;
}
snd_pcm_limit_hw_rates(runtime);
spin_lock_irqsave(&chip->reg_lock, flags);
azx_dev->substream = substream;
azx_dev->running = 0;
Expand All @@ -1463,6 +1464,12 @@ static int azx_pcm_open(struct snd_pcm_substream *substream)
snd_pcm_set_sync(substream);
mutex_unlock(&chip->open_mutex);

if (snd_BUG_ON(!runtime->hw.channels_min || !runtime->hw.channels_max))
return -EINVAL;
if (snd_BUG_ON(!runtime->hw.formats))
return -EINVAL;
if (snd_BUG_ON(!runtime->hw.rates))
return -EINVAL;
return 0;
}

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27 changes: 26 additions & 1 deletion sound/pci/hda/patch_analog.c
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Original file line number Diff line number Diff line change
Expand Up @@ -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),
Expand Down Expand Up @@ -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 */
};

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24 changes: 5 additions & 19 deletions sound/pci/hda/patch_realtek.c
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Original file line number Diff line number Diff line change
Expand Up @@ -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_MUTE("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT),
{ } /* end */
};

Expand All @@ -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},
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