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authorTakashi Iwai <tiwai@suse.de>2009-07-03 23:50:45 +0200
committerTakashi Iwai <tiwai@suse.de>2009-07-03 23:50:45 +0200
commit854ace9c40d2b121191e1644aa4b0b68c4a226d3 (patch)
treefdc43a62eacc95a1276003a4d7d46287312407ab
parentdbe45d0ce394732cc06187e929697fc0fb16aa53 (diff)
parentc470331e69bd54d11a9ea3c27a0e4ad783d02d6b (diff)
downloadlinux-stericsson-854ace9c40d2b121191e1644aa4b0b68c4a226d3.tar.gz
Merge branch 'fix/hda' into for-linus
* 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
-rw-r--r--sound/pci/hda/hda_codec.c14
-rw-r--r--sound/pci/hda/hda_intel.c7
-rw-r--r--sound/pci/hda/patch_analog.c27
-rw-r--r--sound/pci/hda/patch_realtek.c24
4 files changed, 48 insertions, 24 deletions
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 462e2cedaa6a..26d255de6beb 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 4e9ea7080270..1877d95d4aa6 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -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;
@@ -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;
}
diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index ad700761a561..be7d25fa7f35 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 */
};
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 3a8e58c483df..e661b21354be 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -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 */
};
@@ -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},