Merge branch 'linux-3.0.y' into cm.ics

6 files changed, 41 insertions, 8 deletions

diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 67d341f..39e1a6a 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -1651,7 +1651,11 @@ static void put_vol_mute(struct hda_codec *codec, struct hda_amp_info *info,
 	parm = ch ? AC_AMP_SET_RIGHT : AC_AMP_SET_LEFT;
 	parm |= direction == HDA_OUTPUT ? AC_AMP_SET_OUTPUT : AC_AMP_SET_INPUT;
 	parm |= index << AC_AMP_SET_INDEX_SHIFT;
-	parm |= val;
+	if ((val & HDA_AMP_MUTE) && !(info->amp_caps & AC_AMPCAP_MUTE) &&
+	    (info->amp_caps & AC_AMPCAP_MIN_MUTE))
+		; /* set the zero value as a fake mute */
+	else
+		parm |= val;
 	snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, parm);
 	info->vol[ch] = val;
 }
@@ -1918,7 +1922,7 @@ int snd_hda_mixer_amp_tlv(struct snd_kcontrol *kcontrol, int op_flag,
 	val1 = -((caps & AC_AMPCAP_OFFSET) >> AC_AMPCAP_OFFSET_SHIFT);
 	val1 += ofs;
 	val1 = ((int)val1) * ((int)val2);
-	if (min_mute)
+	if (min_mute || (caps & AC_AMPCAP_MIN_MUTE))
 		val2 |= TLV_DB_SCALE_MUTE;
 	if (put_user(SNDRV_CTL_TLVT_DB_SCALE, _tlv))
 		return -EFAULT;
diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h
index 59c9730..eff1fc5 100644
--- a/sound/pci/hda/hda_codec.h
+++ b/sound/pci/hda/hda_codec.h
@@ -302,6 +302,9 @@ enum {
 #define AC_AMPCAP_MUTE			(1<<31)    /* mute capable */
 #define AC_AMPCAP_MUTE_SHIFT		31
 
+/* driver-specific amp-caps: using bits 24-30 */
+#define AC_AMPCAP_MIN_MUTE		(1 << 30) /* min-volume = mute */
+
 /* Connection list */
 #define AC_CLIST_LENGTH			(0x7f<<0)
 #define AC_CLIST_LONG			(1<<7)
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 81ecd6c..4ad20a6 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -4127,7 +4127,8 @@ static int cx_auto_add_volume_idx(struct hda_codec *codec, const char *basename,
 		err = snd_hda_ctl_add(codec, nid, kctl);
 		if (err < 0)
 			return err;
-		if (!(query_amp_caps(codec, nid, hda_dir) & AC_AMPCAP_MUTE))
+		if (!(query_amp_caps(codec, nid, hda_dir) &
+		      (AC_AMPCAP_MUTE | AC_AMPCAP_MIN_MUTE)))
 			break;
 	}
 	return 0;
@@ -4372,6 +4373,22 @@ static const struct hda_codec_ops cx_auto_patch_ops = {
 	.reboot_notify = snd_hda_shutup_pins,
 };
 
+/* add "fake" mute amp-caps to DACs on cx5051 so that mixer mute switches
+ * can be created (bko#42825)
+ */
+static void add_cx5051_fake_mutes(struct hda_codec *codec)
+{
+	static hda_nid_t out_nids[] = {
+		0x10, 0x11, 0
+	};
+	hda_nid_t *p;
+
+	for (p = out_nids; *p; p++)
+		snd_hda_override_amp_caps(codec, *p, HDA_OUTPUT,
+					  AC_AMPCAP_MIN_MUTE |
+					  query_amp_caps(codec, *p, HDA_OUTPUT));
+}
+
 static int patch_conexant_auto(struct hda_codec *codec)
 {
 	struct conexant_spec *spec;
@@ -4390,6 +4407,9 @@ static int patch_conexant_auto(struct hda_codec *codec)
 	case 0x14f15045:
 		spec->single_adc_amp = 1;
 		break;
+	case 0x14f15051:
+		add_cx5051_fake_mutes(codec);
+		break;
 	}
 
 	err = cx_auto_search_adcs(codec);
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 43d88c7..8670682 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -4589,7 +4589,7 @@ static void stac92xx_hp_detect(struct hda_codec *codec)
 		unsigned int val = AC_PINCTL_OUT_EN | AC_PINCTL_HP_EN;
 		if (no_hp_sensing(spec, i))
 			continue;
-		if (presence)
+		if (1 /*presence*/)
 			stac92xx_set_pinctl(codec, cfg->hp_pins[i], val);
 #if 0 /* FIXME */
 /* Resetting the pinctl like below may lead to (a sort of) regressions
diff --git a/sound/soc/imx/imx-ssi.c b/sound/soc/imx/imx-ssi.c
index 61fceb0..3b56254 100644
--- a/sound/soc/imx/imx-ssi.c
+++ b/sound/soc/imx/imx-ssi.c
@@ -112,7 +112,7 @@ static int imx_ssi_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt)
 		break;
 	case SND_SOC_DAIFMT_DSP_A:
 		/* data on rising edge of bclk, frame high 1clk before data */
-		strcr |= SSI_STCR_TFSL | SSI_STCR_TEFS;
+		strcr |= SSI_STCR_TFSL | SSI_STCR_TXBIT0 | SSI_STCR_TEFS;
 		break;
 	}
 
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 32ab7fc..058c0a8 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -2615,9 +2615,13 @@ static void soc_dapm_shutdown_codec(struct snd_soc_dapm_context *dapm)
 	 * standby.
 	 */
 	if (powerdown) {
-		snd_soc_dapm_set_bias_level(dapm, SND_SOC_BIAS_PREPARE);
+		if (dapm->bias_level == SND_SOC_BIAS_ON)
+			snd_soc_dapm_set_bias_level(dapm,
+						    SND_SOC_BIAS_PREPARE);
 		dapm_seq_run(dapm, &down_list, 0, false);
-		snd_soc_dapm_set_bias_level(dapm, SND_SOC_BIAS_STANDBY);
+		if (dapm->bias_level == SND_SOC_BIAS_PREPARE)
+			snd_soc_dapm_set_bias_level(dapm,
+						    SND_SOC_BIAS_STANDBY);
 	}
 }
 
@@ -2630,7 +2634,9 @@ void snd_soc_dapm_shutdown(struct snd_soc_card *card)
 
 	list_for_each_entry(codec, &card->codec_dev_list, list) {
 		soc_dapm_shutdown_codec(&codec->dapm);
-		snd_soc_dapm_set_bias_level(&codec->dapm, SND_SOC_BIAS_OFF);
+		if (codec->dapm.bias_level == SND_SOC_BIAS_STANDBY)
+			snd_soc_dapm_set_bias_level(&codec->dapm,
+						    SND_SOC_BIAS_OFF);
 	}
 }