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

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;
+			if ((runtime->hw.rates & mout->spdif_rates) &&
-			runtime->hw.formats &= mout->spdif_formats;
+			    (runtime->hw.formats & mout->spdif_formats)) {
-			if (mout->spdif_maxbps < hinfo->maxbps)
+				runtime->hw.rates &= mout->spdif_rates;
-				hinfo->maxbps = mout->spdif_maxbps;
+				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);
 	}

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;
 }
 

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 */
 };
 

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_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT),
-	HDA_BIND_VOL("LineOut Playback Volume", &alc269_epc_bind_vol),
+	HDA_CODEC_MUTE("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("LineOut Playback Switch", 0x15, 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[] = {
+#define alc269_fujitsu_mixer	alc269_eeepc_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 */
-};
 
 static struct hda_verb alc269_quanta_fl1_verbs[] = {
 	{0x15, AC_VERB_SET_CONNECT_SEL, 0x01},