Merge branch 'linux-omap-3.0' into android-omap-3.0

4 files changed, 367 insertions, 156 deletions

diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c
index 508a122..06908dd 100644
--- a/sound/soc/codecs/twl6040.c
+++ b/sound/soc/codecs/twl6040.c
@@ -811,7 +811,23 @@ static int twl6040_hs_dac_event(struct snd_soc_dapm_widget *w,
 	return 0;
 }
 
-static int twl6040_power_mode_event(struct snd_soc_dapm_widget *w,
+static int twl6040_hf_dac_event(struct snd_soc_dapm_widget *w,
+			struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_codec *codec = w->codec;
+	struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec);
+
+	if (SND_SOC_DAPM_EVENT_ON(event))
+		priv->non_lp++;
+	else
+		priv->non_lp--;
+
+	msleep(1);
+
+	return 0;
+}
+
+static int twl6040_ep_event(struct snd_soc_dapm_widget *w,
 			struct snd_kcontrol *kcontrol, int event)
 {
 	struct snd_soc_codec *codec = w->codec;
@@ -820,21 +836,15 @@ static int twl6040_power_mode_event(struct snd_soc_dapm_widget *w,
 
 	if (SND_SOC_DAPM_EVENT_ON(event)) {
 		priv->non_lp++;
-		if (!strcmp(w->name, "Earphone Enable")) {
-			/* Earphone doesn't support low power mode */
-			priv->power_mode_forced = 1;
-			ret = headset_power_mode(codec, 1);
-		}
+		/* Earphone doesn't support low power mode */
+		priv->power_mode_forced = 1;
+		ret = headset_power_mode(codec, 1);
 	} else {
 		priv->non_lp--;
-		if (!strcmp(w->name, "Earphone Enable")) {
-			priv->power_mode_forced = 0;
-			ret = headset_power_mode(codec, priv->headset_mode);
-		}
+		priv->power_mode_forced = 0;
+		ret = headset_power_mode(codec, priv->headset_mode);
 	}
 
-	msleep(1);
-
 	return ret;
 }
 
@@ -1096,6 +1106,11 @@ static const struct soc_enum twl6040_hf_enum[] = {
 			twl6040_hf_texts),
 };
 
+static const char *twl6040_ep_texts[] = {"Off", "HS DAC"};
+
+static const struct soc_enum twl6040_virt_enum =
+	SOC_ENUM_SINGLE(SND_SOC_NOPM, 0, 2, twl6040_ep_texts);
+
 static const struct snd_kcontrol_new amicl_control =
 	SOC_DAPM_ENUM("Route", twl6040_enum[0]);
 
@@ -1109,6 +1124,10 @@ static const struct snd_kcontrol_new hsl_mux_controls =
 static const struct snd_kcontrol_new hsr_mux_controls =
 	SOC_DAPM_ENUM("Route", twl6040_hs_enum[1]);
 
+/* EP playback virtual mux */
+static const struct snd_kcontrol_new ep_virt_mux_controls =
+	SOC_DAPM_ENUM_VIRT("Route", twl6040_virt_enum);
+
 /* Handsfree DAC playback switches */
 static const struct snd_kcontrol_new hfl_mux_controls =
 	SOC_DAPM_ENUM("Route", twl6040_hf_enum[0]);
@@ -1116,9 +1135,6 @@ static const struct snd_kcontrol_new hfl_mux_controls =
 static const struct snd_kcontrol_new hfr_mux_controls =
 	SOC_DAPM_ENUM("Route", twl6040_hf_enum[1]);
 
-static const struct snd_kcontrol_new ep_driver_switch_controls =
-	SOC_DAPM_SINGLE("Switch", TWL6040_REG_EARCTL, 0, 1, 0);
-
 /* Headset power mode */
 static const char *twl6040_headset_power_texts[] = {
 	"Low-Power", "High-Performance",
@@ -1244,11 +1260,11 @@ static const struct snd_soc_dapm_widget twl6040_dapm_widgets[] = {
 			SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
 	SND_SOC_DAPM_DAC_E("HFDAC Left", "Handsfree Playback",
 			TWL6040_REG_HFLCTL, 0, 0,
-			twl6040_power_mode_event,
+			twl6040_hf_dac_event,
 			SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
 	SND_SOC_DAPM_DAC_E("HFDAC Right", "Handsfree Playback",
 			TWL6040_REG_HFRCTL, 0, 0,
-			twl6040_power_mode_event,
+			twl6040_hf_dac_event,
 			SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
 
 	SND_SOC_DAPM_MUX("HF Left Playback",
@@ -1260,6 +1276,8 @@ static const struct snd_soc_dapm_widget twl6040_dapm_widgets[] = {
 			SND_SOC_NOPM, 0, 0, &hsl_mux_controls),
 	SND_SOC_DAPM_MUX("HS Right Playback",
 			SND_SOC_NOPM, 0, 0, &hsr_mux_controls),
+	SND_SOC_DAPM_MUX("Earphone Playback",
+			SND_SOC_NOPM, 0, 0, &ep_virt_mux_controls),
 
 	/* Analog playback drivers */
 	SND_SOC_DAPM_OUT_DRV_E("Handsfree Left Driver",
@@ -1278,15 +1296,16 @@ static const struct snd_soc_dapm_widget twl6040_dapm_widgets[] = {
 			TWL6040_REG_HSRCTL, 2, 0, NULL, 0,
 			pga_event,
 			SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
-	SND_SOC_DAPM_SWITCH_E("Earphone Enable",
-			SND_SOC_NOPM, 0, 0, &ep_driver_switch_controls,
-			twl6040_power_mode_event,
-			SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
 	SND_SOC_DAPM_OUT_DRV_E("Earphone Driver",
-			SND_SOC_NOPM, 0, 0, NULL, 0,
+			TWL6040_REG_EARCTL, 0, 0, NULL, 0,
 			pga_event,
 			SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
 
+	/* Earphone virtual PGA (HS power mode) */
+	SND_SOC_DAPM_PGA_E("Earphone Power Mode", SND_SOC_NOPM,
+			 0, 0, NULL, 0, twl6040_ep_event,
+			SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+
 	/* Analog playback PGAs */
 	SND_SOC_DAPM_PGA("HFDAC Left PGA",
 			TWL6040_REG_HFLCTL, 1, 0, NULL, 0),
@@ -1328,8 +1347,9 @@ static const struct snd_soc_dapm_route intercon[] = {
 	{"HSOR", NULL, "Headset Right Driver"},
 
 	/* Earphone playback path */
-	{"Earphone Enable", "Switch", "HSDAC Left"},
-	{"Earphone Driver", NULL, "Earphone Enable"},
+	{"Earphone Playback", "HS DAC", "HSDAC Left"},
+	{"Earphone Power Mode", NULL, "Earphone Playback"},
+	{"Earphone Driver", NULL, "Earphone Power Mode"},
 	{"EP", NULL, "Earphone Driver"},
 
 	{"HF Left Playback", "HF DAC", "HFDAC Left"},
@@ -1684,8 +1704,10 @@ static int twl6040_probe(struct snd_soc_codec *codec)
 	else
 		priv->ep_step = 1;
 
-	/* default is low-power mode */
+	/* default headset power mode is high-performance */
 	priv->headset_mode = 1;
+
+	/* default is low-power mode */
 	priv->sysclk_constraints = &lp_constraints;
 	priv->workqueue = create_singlethread_workqueue("twl6040-codec");
 
diff --git a/sound/soc/omap/omap-abe-dsp.c b/sound/soc/omap/omap-abe-dsp.c
index fa6964c..c4c8e45 100644
--- a/sound/soc/omap/omap-abe-dsp.c
+++ b/sound/soc/omap/omap-abe-dsp.c
@@ -2057,41 +2057,74 @@ static int aess_set_runtime_opp_level(struct abe_data *abe)
 	return 0;
 }
 
-static int aess_save_context(struct abe_data *abe)
+static void abe_dsp_init_gains(struct abe_data *abe)
 {
-	/* TODO: Find a better way to save/retore gains after OFF mode */
-
-	abe_mute_gain(MIXSDT, MIX_SDT_INPUT_UP_MIXER);
-	abe_mute_gain(MIXSDT, MIX_SDT_INPUT_DL1_MIXER);
+	/* Uplink gains */
 	abe_mute_gain(MIXAUDUL, MIX_AUDUL_INPUT_MM_DL);
 	abe_mute_gain(MIXAUDUL, MIX_AUDUL_INPUT_TONES);
 	abe_mute_gain(MIXAUDUL, MIX_AUDUL_INPUT_UPLINK);
 	abe_mute_gain(MIXAUDUL, MIX_AUDUL_INPUT_VX_DL);
+
 	abe_mute_gain(MIXVXREC, MIX_VXREC_INPUT_TONES);
 	abe_mute_gain(MIXVXREC, MIX_VXREC_INPUT_VX_DL);
 	abe_mute_gain(MIXVXREC, MIX_VXREC_INPUT_MM_DL);
 	abe_mute_gain(MIXVXREC, MIX_VXREC_INPUT_VX_UL);
-	abe_mute_gain(MIXDL1, MIX_DL1_INPUT_MM_DL);
-	abe_mute_gain(MIXDL1, MIX_DL1_INPUT_MM_UL2);
-	abe_mute_gain(MIXDL1, MIX_DL1_INPUT_VX_DL);
-	abe_mute_gain(MIXDL1, MIX_DL1_INPUT_TONES);
-	abe_mute_gain(MIXDL2, MIX_DL2_INPUT_TONES);
-	abe_mute_gain(MIXDL2, MIX_DL2_INPUT_VX_DL);
-	abe_mute_gain(MIXDL2, MIX_DL2_INPUT_MM_DL);
-	abe_mute_gain(MIXDL2, MIX_DL2_INPUT_MM_UL2);
-	abe_mute_gain(MIXECHO, MIX_ECHO_DL1);
-	abe_mute_gain(MIXECHO, MIX_ECHO_DL2);
+
 	abe_mute_gain(GAINS_DMIC1, GAIN_LEFT_OFFSET);
 	abe_mute_gain(GAINS_DMIC1, GAIN_RIGHT_OFFSET);
 	abe_mute_gain(GAINS_DMIC2, GAIN_LEFT_OFFSET);
 	abe_mute_gain(GAINS_DMIC2, GAIN_RIGHT_OFFSET);
 	abe_mute_gain(GAINS_DMIC3, GAIN_LEFT_OFFSET);
 	abe_mute_gain(GAINS_DMIC3, GAIN_RIGHT_OFFSET);
+
 	abe_mute_gain(GAINS_AMIC, GAIN_LEFT_OFFSET);
 	abe_mute_gain(GAINS_AMIC, GAIN_RIGHT_OFFSET);
+
 	abe_mute_gain(GAINS_BTUL, GAIN_LEFT_OFFSET);
 	abe_mute_gain(GAINS_BTUL, GAIN_RIGHT_OFFSET);
 
+	/* Downlink gains */
+	abe_write_gain(GAINS_DL1, GAIN_0dB, RAMP_5MS, GAIN_LEFT_OFFSET);
+	abe_write_gain(GAINS_DL1, GAIN_0dB, RAMP_5MS, GAIN_RIGHT_OFFSET);
+	abe_mute_gain(GAINS_DL1, GAIN_LEFT_OFFSET);
+	abe_mute_gain(GAINS_DL1, GAIN_RIGHT_OFFSET);
+
+	abe_write_gain(GAINS_DL2, GAIN_M7dB, RAMP_5MS, GAIN_LEFT_OFFSET);
+	abe_write_gain(GAINS_DL2, GAIN_M7dB, RAMP_5MS, GAIN_RIGHT_OFFSET);
+	abe_mute_gain(GAINS_DL2, GAIN_LEFT_OFFSET);
+	abe_mute_gain(GAINS_DL2, GAIN_RIGHT_OFFSET);
+
+	abe_mute_gain(MIXDL1, MIX_DL1_INPUT_MM_DL);
+	abe_mute_gain(MIXDL1, MIX_DL1_INPUT_MM_UL2);
+	abe_mute_gain(MIXDL1, MIX_DL1_INPUT_VX_DL);
+	abe_mute_gain(MIXDL1, MIX_DL1_INPUT_TONES);
+
+	abe_mute_gain(MIXDL2, MIX_DL2_INPUT_TONES);
+	abe_mute_gain(MIXDL2, MIX_DL2_INPUT_VX_DL);
+	abe_mute_gain(MIXDL2, MIX_DL2_INPUT_MM_DL);
+	abe_mute_gain(MIXDL2, MIX_DL2_INPUT_MM_UL2);
+
+	abe_mute_gain(MIXECHO, MIX_ECHO_DL1);
+	abe_mute_gain(MIXECHO, MIX_ECHO_DL2);
+
+	/* Sidetone gains */
+	abe_mute_gain(MIXSDT, MIX_SDT_INPUT_UP_MIXER);
+	abe_mute_gain(MIXSDT, MIX_SDT_INPUT_DL1_MIXER);
+}
+
+static int aess_save_context(struct abe_data *abe)
+{
+	/* mute gains not associated with FEs/BEs */
+	abe_mute_gain(MIXAUDUL, MIX_AUDUL_INPUT_MM_DL);
+	abe_mute_gain(MIXAUDUL, MIX_AUDUL_INPUT_TONES);
+	abe_mute_gain(MIXAUDUL, MIX_AUDUL_INPUT_VX_DL);
+	abe_mute_gain(MIXVXREC, MIX_VXREC_INPUT_TONES);
+	abe_mute_gain(MIXVXREC, MIX_VXREC_INPUT_VX_DL);
+	abe_mute_gain(MIXVXREC, MIX_VXREC_INPUT_MM_DL);
+	abe_mute_gain(MIXVXREC, MIX_VXREC_INPUT_VX_UL);
+	abe_mute_gain(MIXECHO, MIX_ECHO_DL1);
+	abe_mute_gain(MIXECHO, MIX_ECHO_DL2);
+
 	return 0;
 }
 
@@ -2112,37 +2145,16 @@ static int aess_restore_context(struct abe_data *abe)
 	if (pdata->was_context_lost && pdata->was_context_lost(abe->dev))
 		abe_reload_fw(abe->firmware);
 
-	/* TODO: Find a better way to save/retore gains after dor OFF mode */
-	abe_unmute_gain(MIXSDT, MIX_SDT_INPUT_UP_MIXER);
-	abe_unmute_gain(MIXSDT, MIX_SDT_INPUT_DL1_MIXER);
+	/* unmute gains not associated with FEs/BEs */
 	abe_unmute_gain(MIXAUDUL, MIX_AUDUL_INPUT_MM_DL);
 	abe_unmute_gain(MIXAUDUL, MIX_AUDUL_INPUT_TONES);
-	abe_unmute_gain(MIXAUDUL, MIX_AUDUL_INPUT_UPLINK);
 	abe_unmute_gain(MIXAUDUL, MIX_AUDUL_INPUT_VX_DL);
 	abe_unmute_gain(MIXVXREC, MIX_VXREC_INPUT_TONES);
 	abe_unmute_gain(MIXVXREC, MIX_VXREC_INPUT_VX_DL);
 	abe_unmute_gain(MIXVXREC, MIX_VXREC_INPUT_MM_DL);
 	abe_unmute_gain(MIXVXREC, MIX_VXREC_INPUT_VX_UL);
-	abe_unmute_gain(MIXDL1, MIX_DL1_INPUT_MM_DL);
-	abe_unmute_gain(MIXDL1, MIX_DL1_INPUT_MM_UL2);
-	abe_unmute_gain(MIXDL1, MIX_DL1_INPUT_VX_DL);
-	abe_unmute_gain(MIXDL1, MIX_DL1_INPUT_TONES);
-	abe_unmute_gain(MIXDL2, MIX_DL2_INPUT_TONES);
-	abe_unmute_gain(MIXDL2, MIX_DL2_INPUT_VX_DL);
-	abe_unmute_gain(MIXDL2, MIX_DL2_INPUT_MM_DL);
-	abe_unmute_gain(MIXDL2, MIX_DL2_INPUT_MM_UL2);
 	abe_unmute_gain(MIXECHO, MIX_ECHO_DL1);
 	abe_unmute_gain(MIXECHO, MIX_ECHO_DL2);
-	abe_unmute_gain(GAINS_DMIC1, GAIN_LEFT_OFFSET);
-	abe_unmute_gain(GAINS_DMIC1, GAIN_RIGHT_OFFSET);
-	abe_unmute_gain(GAINS_DMIC2, GAIN_LEFT_OFFSET);
-	abe_unmute_gain(GAINS_DMIC2, GAIN_RIGHT_OFFSET);
-	abe_unmute_gain(GAINS_DMIC3, GAIN_LEFT_OFFSET);
-	abe_unmute_gain(GAINS_DMIC3, GAIN_RIGHT_OFFSET);
-	abe_unmute_gain(GAINS_AMIC, GAIN_LEFT_OFFSET);
-	abe_unmute_gain(GAINS_AMIC, GAIN_RIGHT_OFFSET);
-	abe_unmute_gain(GAINS_BTUL, GAIN_LEFT_OFFSET);
-	abe_unmute_gain(GAINS_BTUL, GAIN_RIGHT_OFFSET);
 
 	abe_set_router_configuration(UPROUTE, 0, (u32 *)abe->router);
 
@@ -2692,6 +2704,9 @@ static int abe_probe(struct snd_soc_platform *platform)
 
 	/* "tick" of the audio engine */
 	abe_write_event_generator(EVENT_TIMER);
+
+	abe_dsp_init_gains(abe);
+
 	/* Stop the engine */
 	abe_stop_event_generator();
 	abe_disable_irq();
diff --git a/sound/soc/omap/omap-abe.c b/sound/soc/omap/omap-abe.c
index 6cb7d9f..95cfe86 100644
--- a/sound/soc/omap/omap-abe.c
+++ b/sound/soc/omap/omap-abe.c
@@ -61,8 +61,6 @@ struct omap_abe_data {
 	int suspended_dais;
 };
 
-static int dl1_gain;	/* dynamically set by machine drivers */
-
 /*
  * Stream DMA parameters
  */
@@ -152,7 +150,7 @@ static int modem_get_dai(struct snd_pcm_substream *substream,
 
 	abe_priv->modem_substream[substream->stream] =
 			snd_soc_get_dai_substream(rtd->card,
-					OMAP_ABE_BE_MM_EXT1, substream->stream);
+					OMAP_ABE_BE_MM_EXT1, !substream->stream);
 
 	if (abe_priv->modem_substream[substream->stream] == NULL)
 		return -ENODEV;
@@ -166,64 +164,106 @@ static int modem_get_dai(struct snd_pcm_substream *substream,
 
 int omap_abe_set_dl1_output(int output)
 {
+	int gain;
+
 	/*
 	 * the output itself is not important, but the DL1 gain
 	 * to use when each output is active
 	 */
 	switch (output) {
 	case OMAP_ABE_DL1_HEADSET_LP:
-		dl1_gain = GAIN_M8dB;
+		gain = GAIN_M8dB;
 		break;
 	case OMAP_ABE_DL1_HEADSET_HP:
 	case OMAP_ABE_DL1_EARPIECE:
-		dl1_gain = GAIN_M1dB;
+		gain = GAIN_M1dB;
 		break;
 	case OMAP_ABE_DL1_NO_PDM:
-		dl1_gain = GAIN_0dB;
+		gain = GAIN_0dB;
 		break;
 	default:
 		return -EINVAL;
 	}
 
+	abe_write_gain(GAINS_DL1, gain, RAMP_5MS, GAIN_LEFT_OFFSET);
+	abe_write_gain(GAINS_DL1, gain, RAMP_5MS, GAIN_RIGHT_OFFSET);
+
 	return 0;
 }
 EXPORT_SYMBOL(omap_abe_set_dl1_output);
 
+static int omap_abe_dl1_enabled(struct omap_abe_data *abe_priv)
+{
+	/* DL1 path is common for PDM_DL1, BT_VX_DL and MM_EXT_DL */
+	return omap_abe_port_is_enabled(abe_priv->abe,
+				abe_priv->port[OMAP_ABE_BE_PORT_PDM_DL1]) +
+		omap_abe_port_is_enabled(abe_priv->abe,
+				abe_priv->port[OMAP_ABE_BE_PORT_BT_VX_DL]) +
+		omap_abe_port_is_enabled(abe_priv->abe,
+				abe_priv->port[OMAP_ABE_BE_PORT_MM_EXT_DL]);
+}
+
+static int omap_abe_dl2_enabled(struct omap_abe_data *abe_priv)
+{
+	return omap_abe_port_is_enabled(abe_priv->abe,
+				abe_priv->port[OMAP_ABE_BE_PORT_PDM_DL2]);
+}
+
 static void mute_be(struct snd_soc_pcm_runtime *be,
 		struct snd_soc_dai *dai, int stream)
 {
+	struct omap_abe_data *abe_priv = snd_soc_dai_get_drvdata(dai);
+
 	dev_dbg(&be->dev, "%s: %s %d\n", __func__, be->cpu_dai->name, stream);
 
 	if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
 		switch (be->dai_link->be_id) {
 		case OMAP_ABE_DAI_PDM_DL1:
-			abe_write_gain(GAINS_DL1, MUTE_GAIN, RAMP_5MS,
-				GAIN_LEFT_OFFSET);
-			abe_write_gain(GAINS_DL1, MUTE_GAIN, RAMP_5MS,
-				GAIN_RIGHT_OFFSET);
+		case OMAP_ABE_DAI_BT_VX:
+		case OMAP_ABE_DAI_MM_FM:
+			/*
+			 * DL1 Mixer->SDT Mixer and DL1 gain are common for
+			 * PDM_DL1, BT_VX_DL and MM_EXT_DL, mute those gains
+			 * only if the last active BE
+			 */
+			if (omap_abe_dl1_enabled(abe_priv) == 1) {
+				abe_mute_gain(GAINS_DL1, GAIN_LEFT_OFFSET);
+				abe_mute_gain(GAINS_DL1, GAIN_RIGHT_OFFSET);
+				abe_mute_gain(MIXSDT, MIX_SDT_INPUT_DL1_MIXER);
+			}
 			break;
 		case OMAP_ABE_DAI_PDM_DL2:
-			abe_write_gain(GAINS_DL2, MUTE_GAIN, RAMP_5MS,
-				GAIN_LEFT_OFFSET);
-			abe_write_gain(GAINS_DL2, MUTE_GAIN, RAMP_5MS,
-				GAIN_RIGHT_OFFSET);
+			abe_mute_gain(GAINS_DL2, GAIN_LEFT_OFFSET);
+			abe_mute_gain(GAINS_DL2, GAIN_RIGHT_OFFSET);
 			break;
 		case OMAP_ABE_DAI_PDM_VIB:
-		case OMAP_ABE_DAI_BT_VX:
-		case OMAP_ABE_DAI_MM_FM:
 		case OMAP_ABE_DAI_MODEM:
 			break;
 		}
 	} else {
 		switch (be->dai_link->be_id) {
 		case OMAP_ABE_DAI_PDM_UL:
+			abe_mute_gain(GAINS_AMIC, GAIN_LEFT_OFFSET);
+			abe_mute_gain(GAINS_AMIC, GAIN_RIGHT_OFFSET);
 			break;
 		case OMAP_ABE_DAI_BT_VX:
+			abe_mute_gain(GAINS_BTUL, GAIN_LEFT_OFFSET);
+			abe_mute_gain(GAINS_BTUL, GAIN_RIGHT_OFFSET);
+			break;
 		case OMAP_ABE_DAI_MM_FM:
 		case OMAP_ABE_DAI_MODEM:
+			break;
 		case OMAP_ABE_DAI_DMIC0:
+			abe_mute_gain(GAINS_DMIC1, GAIN_LEFT_OFFSET);
+			abe_mute_gain(GAINS_DMIC1, GAIN_RIGHT_OFFSET);
+			break;
 		case OMAP_ABE_DAI_DMIC1:
+			abe_mute_gain(GAINS_DMIC2, GAIN_LEFT_OFFSET);
+			abe_mute_gain(GAINS_DMIC2, GAIN_RIGHT_OFFSET);
+			break;
 		case OMAP_ABE_DAI_DMIC2:
+			abe_mute_gain(GAINS_DMIC3, GAIN_LEFT_OFFSET);
+			abe_mute_gain(GAINS_DMIC3, GAIN_RIGHT_OFFSET);
 			break;
 		}
 	}
@@ -239,20 +279,22 @@ static void unmute_be(struct snd_soc_pcm_runtime *be,
 	if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
 		switch (be->dai_link->be_id) {
 		case OMAP_ABE_DAI_PDM_DL1:
-			abe_write_gain(GAINS_DL1, dl1_gain, RAMP_5MS,
-				GAIN_LEFT_OFFSET);
-			abe_write_gain(GAINS_DL1, dl1_gain, RAMP_5MS,
-				GAIN_RIGHT_OFFSET);
+		case OMAP_ABE_DAI_BT_VX:
+		case OMAP_ABE_DAI_MM_FM:
+			/*
+			 * DL1 Mixer->SDT Mixer and DL1 gain are common for
+			 * PDM_DL1, BT_VX_DL and MM_EXT_DL, unmute when any
+			 * of them becomes active
+			 */
+			abe_unmute_gain(GAINS_DL1, GAIN_LEFT_OFFSET);
+			abe_unmute_gain(GAINS_DL1, GAIN_RIGHT_OFFSET);
+			abe_unmute_gain(MIXSDT, MIX_SDT_INPUT_DL1_MIXER);
 			break;
 		case OMAP_ABE_DAI_PDM_DL2:
-			abe_write_gain(GAINS_DL2, GAIN_M7dB, RAMP_5MS,
-				GAIN_LEFT_OFFSET);
-			abe_write_gain(GAINS_DL2, GAIN_M7dB, RAMP_5MS,
-				GAIN_RIGHT_OFFSET);
+			abe_unmute_gain(GAINS_DL2, GAIN_LEFT_OFFSET);
+			abe_unmute_gain(GAINS_DL2, GAIN_RIGHT_OFFSET);
 			break;
 		case OMAP_ABE_DAI_PDM_VIB:
-		case OMAP_ABE_DAI_BT_VX:
-		case OMAP_ABE_DAI_MM_FM:
 			break;
 		case OMAP_ABE_DAI_MODEM:
 			if (omap_abe_port_is_enabled(abe_priv->abe,
@@ -270,14 +312,32 @@ static void unmute_be(struct snd_soc_pcm_runtime *be,
 
 		switch (be->dai_link->be_id) {
 		case OMAP_ABE_DAI_PDM_UL:
+			abe_unmute_gain(GAINS_AMIC, GAIN_LEFT_OFFSET);
+			abe_unmute_gain(GAINS_AMIC, GAIN_RIGHT_OFFSET);
 			break;
 		case OMAP_ABE_DAI_BT_VX:
+			abe_reset_mic_ul_src_filters();
+			abe_unmute_gain(GAINS_BTUL, GAIN_LEFT_OFFSET);
+			abe_unmute_gain(GAINS_BTUL, GAIN_RIGHT_OFFSET);
+			break;
 		case OMAP_ABE_DAI_MM_FM:
 		case OMAP_ABE_DAI_MODEM:
+			abe_reset_mic_ul_src_filters();
+			break;
 		case OMAP_ABE_DAI_DMIC0:
+			abe_reset_mic_ul_src_filters();
+			abe_unmute_gain(GAINS_DMIC1, GAIN_LEFT_OFFSET);
+			abe_unmute_gain(GAINS_DMIC1, GAIN_RIGHT_OFFSET);
+			break;
 		case OMAP_ABE_DAI_DMIC1:
+			abe_reset_mic_ul_src_filters();
+			abe_unmute_gain(GAINS_DMIC2, GAIN_LEFT_OFFSET);
+			abe_unmute_gain(GAINS_DMIC2, GAIN_RIGHT_OFFSET);
+			break;
 		case OMAP_ABE_DAI_DMIC2:
 			abe_reset_mic_ul_src_filters();
+			abe_unmute_gain(GAINS_DMIC3, GAIN_LEFT_OFFSET);
+			abe_unmute_gain(GAINS_DMIC3, GAIN_RIGHT_OFFSET);
 			break;
 		}
 	}
@@ -511,88 +571,184 @@ static void disable_fe_port(struct snd_pcm_substream *substream,
 	}
 }
 
-static void mute_fe_port(struct snd_pcm_substream *substream,
-		struct snd_soc_dai *dai, int stream)
+static void mute_fe_port_capture(struct snd_soc_pcm_runtime *fe,
+				struct snd_soc_pcm_runtime *be, int mute)
 {
-	struct snd_soc_pcm_runtime *fe = substream->private_data;
-	struct omap_abe_data *abe_priv = snd_soc_dai_get_drvdata(dai);
+	struct omap_abe_data *abe_priv = snd_soc_dai_get_drvdata(fe->cpu_dai);
 
-	dev_dbg(&fe->dev, "%s: %s %d\n", __func__, dai->name, stream);
+	dev_dbg(&fe->dev, "%s: %s FE %s BE %s\n",
+			__func__, mute ? "mute" : "unmute",
+			fe->dai_link->name, be->dai_link->name);
 
-	switch(dai->id) {
-	case ABE_FRONTEND_DAI_MEDIA:
-	case ABE_FRONTEND_DAI_LP_MEDIA:
-		if (omap_abe_port_is_enabled(abe_priv->abe,
-						abe_priv->port[OMAP_ABE_BE_PORT_PDM_DL2]))
-			abe_mute_gain(MIXDL2, MIX_DL2_INPUT_MM_DL);
-		if (omap_abe_port_is_enabled(abe_priv->abe,
-						abe_priv->port[OMAP_ABE_BE_PORT_PDM_DL1]))
-			abe_mute_gain(MIXDL1, MIX_DL1_INPUT_MM_DL);
+	switch (fe->cpu_dai->id) {
+	case ABE_FRONTEND_DAI_MEDIA_CAPTURE:
+		if (omap_abe_dl1_enabled(abe_priv)) {
+			if (mute)
+				abe_mute_gain(MIXDL1, MIX_DL1_INPUT_MM_UL2);
+			else
+				abe_unmute_gain(MIXDL1, MIX_DL1_INPUT_MM_UL2);
+		}
+		if (omap_abe_dl2_enabled(abe_priv)) {
+			if (mute)
+				abe_mute_gain(MIXDL2, MIX_DL2_INPUT_MM_UL2);
+			else
+				abe_unmute_gain(MIXDL2, MIX_DL2_INPUT_MM_UL2);
+		}
 		break;
-	case ABE_FRONTEND_DAI_VOICE:
 	case ABE_FRONTEND_DAI_MODEM:
-		if (omap_abe_port_is_enabled(abe_priv->abe,
-						abe_priv->port[OMAP_ABE_BE_PORT_PDM_DL2]))
-			abe_mute_gain(MIXDL2, MIX_DL2_INPUT_VX_DL);
-		if (omap_abe_port_is_enabled(abe_priv->abe,
-						abe_priv->port[OMAP_ABE_BE_PORT_PDM_DL1]))
-			abe_mute_gain(MIXDL1, MIX_DL1_INPUT_VX_DL);
-		break;
-	case ABE_FRONTEND_DAI_TONES:
-			if (omap_abe_port_is_enabled(abe_priv->abe,
-						abe_priv->port[OMAP_ABE_BE_PORT_PDM_DL2]))
-			abe_mute_gain(MIXDL2, MIX_DL2_INPUT_TONES);
-		if (omap_abe_port_is_enabled(abe_priv->abe,
-						abe_priv->port[OMAP_ABE_BE_PORT_PDM_DL1]))
-			abe_mute_gain(MIXDL1, MIX_DL1_INPUT_TONES);
+	case ABE_FRONTEND_DAI_VOICE:
+		if (mute) {
+			abe_mute_gain(MIXSDT, MIX_SDT_INPUT_UP_MIXER);
+			abe_mute_gain(MIXAUDUL, MIX_AUDUL_INPUT_UPLINK);
+		} else {
+			abe_unmute_gain(MIXSDT, MIX_SDT_INPUT_UP_MIXER);
+			abe_unmute_gain(MIXAUDUL, MIX_AUDUL_INPUT_UPLINK);
+		}
 		break;
-	case ABE_FRONTEND_DAI_VIBRA:
-	case ABE_FRONTEND_DAI_MEDIA_CAPTURE:
+	case ABE_FRONTEND_DAI_MEDIA:
+	default:
 		break;
 	}
 }
 
-static void unmute_fe_port(struct snd_pcm_substream *substream,
-		struct snd_soc_dai *dai, int stream)
+static void mute_fe_port_playback(struct snd_soc_pcm_runtime *fe,
+				struct snd_soc_pcm_runtime *be, int mute)
 {
-	struct snd_soc_pcm_runtime *fe = substream->private_data;
-	struct omap_abe_data *abe_priv = snd_soc_dai_get_drvdata(dai);
+	struct omap_abe_data *abe_priv = snd_soc_dai_get_drvdata(fe->cpu_dai);
 
-	dev_dbg(&fe->dev, "%s: %s %d\n", __func__, dai->name, stream);
+	dev_dbg(&fe->dev, "%s: %s FE %s BE %s\n",
+			__func__, mute ? "mute" : "unmute",
+			fe->dai_link->name, be->dai_link->name);
 
-	switch(dai->id) {
+	switch (fe->cpu_dai->id) {
 	case ABE_FRONTEND_DAI_MEDIA:
 	case ABE_FRONTEND_DAI_LP_MEDIA:
-		if (omap_abe_port_is_enabled(abe_priv->abe,
-						abe_priv->port[OMAP_ABE_BE_PORT_PDM_DL2]))
-			abe_unmute_gain(MIXDL2, MIX_DL2_INPUT_MM_DL);
-		if (omap_abe_port_is_enabled(abe_priv->abe,
-						abe_priv->port[OMAP_ABE_BE_PORT_PDM_DL1]))
-			abe_unmute_gain(MIXDL1, MIX_DL1_INPUT_MM_DL);
+		switch (be->dai_link->be_id) {
+		case OMAP_ABE_DAI_PDM_DL1:
+		case OMAP_ABE_DAI_BT_VX:
+		case OMAP_ABE_DAI_MM_FM:
+			if (mute) {
+				/* mute if last running DL1-related BE */
+				if (omap_abe_dl1_enabled(abe_priv) == 1)
+					abe_mute_gain(MIXDL1,
+							MIX_DL1_INPUT_MM_DL);
+			} else {
+				abe_unmute_gain(MIXDL1, MIX_DL1_INPUT_MM_DL);
+			}
+			break;
+		case OMAP_ABE_DAI_PDM_DL2:
+			if (mute)
+				abe_mute_gain(MIXDL2, MIX_DL2_INPUT_MM_DL);
+			else
+				abe_unmute_gain(MIXDL2, MIX_DL2_INPUT_MM_DL);
+			break;
+		case OMAP_ABE_DAI_MODEM:
+		case OMAP_ABE_DAI_PDM_VIB:
+		default:
+			break;
+		}
 		break;
 	case ABE_FRONTEND_DAI_VOICE:
 	case ABE_FRONTEND_DAI_MODEM:
-		if (omap_abe_port_is_enabled(abe_priv->abe,
-						abe_priv->port[OMAP_ABE_BE_PORT_PDM_DL2]))
-			abe_unmute_gain(MIXDL2, MIX_DL2_INPUT_VX_DL);
-		if (omap_abe_port_is_enabled(abe_priv->abe,
-						abe_priv->port[OMAP_ABE_BE_PORT_PDM_DL1]))
-			abe_unmute_gain(MIXDL1, MIX_DL1_INPUT_VX_DL);
+		switch (be->dai_link->be_id) {
+		case OMAP_ABE_DAI_PDM_DL1:
+		case OMAP_ABE_DAI_BT_VX:
+		case OMAP_ABE_DAI_MM_FM:
+			if (mute) {
+				/* mute if last running DL1-related BE */
+				if (omap_abe_dl1_enabled(abe_priv) == 1)
+					abe_mute_gain(MIXDL1,
+							MIX_DL1_INPUT_VX_DL);
+			} else {
+				abe_unmute_gain(MIXDL1, MIX_DL1_INPUT_VX_DL);
+			}
+			break;
+		case OMAP_ABE_DAI_PDM_DL2:
+			if (mute)
+				abe_mute_gain(MIXDL2, MIX_DL2_INPUT_VX_DL);
+			else
+				abe_unmute_gain(MIXDL2, MIX_DL2_INPUT_VX_DL);
+			break;
+		case OMAP_ABE_DAI_MODEM:
+		case OMAP_ABE_DAI_PDM_VIB:
+		default:
+			break;
+		}
 		break;
 	case ABE_FRONTEND_DAI_TONES:
-			if (omap_abe_port_is_enabled(abe_priv->abe,
-						abe_priv->port[OMAP_ABE_BE_PORT_PDM_DL2]))
-			abe_unmute_gain(MIXDL2, MIX_DL2_INPUT_TONES);
-		if (omap_abe_port_is_enabled(abe_priv->abe,
-						abe_priv->port[OMAP_ABE_BE_PORT_PDM_DL1]))
-			abe_unmute_gain(MIXDL1, MIX_DL1_INPUT_TONES);
+		switch (be->dai_link->be_id) {
+		case OMAP_ABE_DAI_PDM_DL1:
+		case OMAP_ABE_DAI_BT_VX:
+		case OMAP_ABE_DAI_MM_FM:
+			if (mute) {
+				/* mute if last running DL1-related BE */
+				if (omap_abe_dl1_enabled(abe_priv) == 1)
+					abe_mute_gain(MIXDL1,
+							MIX_DL1_INPUT_TONES);
+			} else{
+				abe_unmute_gain(MIXDL1, MIX_DL1_INPUT_TONES);
+			}
+			break;
+		case OMAP_ABE_DAI_PDM_DL2:
+			if (mute)
+				abe_mute_gain(MIXDL2, MIX_DL2_INPUT_TONES);
+			else
+				abe_unmute_gain(MIXDL2, MIX_DL2_INPUT_TONES);
+			break;
+		case OMAP_ABE_DAI_MODEM:
+		case OMAP_ABE_DAI_PDM_VIB:
+		default:
+			break;
+		}
 		break;
 	case ABE_FRONTEND_DAI_VIBRA:
-	case ABE_FRONTEND_DAI_MEDIA_CAPTURE:
+	default:
 		break;
 	}
 }
 
+static void mute_fe_port(struct snd_pcm_substream *substream,
+		struct snd_soc_dai *dai, int stream)
+{
+	struct snd_soc_pcm_runtime *fe = substream->private_data;
+	struct snd_soc_dsp_params *dsp_params;
+
+	dev_dbg(&fe->dev, "%s: %s %d\n", __func__, dai->name, stream);
+
+	list_for_each_entry(dsp_params, &fe->dsp[stream].be_clients, list_be) {
+		struct snd_soc_pcm_runtime *be = dsp_params->be;
+
+		if (!snd_soc_dsp_is_op_for_be(fe, be, stream))
+			continue;
+
+		if (stream == SNDRV_PCM_STREAM_PLAYBACK)
+			mute_fe_port_playback(fe, be, 1);
+		else
+			mute_fe_port_capture(fe, be, 1);
+	}
+}
+
+static void unmute_fe_port(struct snd_pcm_substream *substream,
+		struct snd_soc_dai *dai, int stream)
+{
+	struct snd_soc_pcm_runtime *fe = substream->private_data;
+	struct snd_soc_dsp_params *dsp_params;
+
+	dev_dbg(&fe->dev, "%s: %s %d\n", __func__, dai->name, stream);
+
+	list_for_each_entry(dsp_params, &fe->dsp[stream].be_clients, list_be) {
+		struct snd_soc_pcm_runtime *be = dsp_params->be;
+
+		if (!snd_soc_dsp_is_op_for_be(fe, be, stream))
+			continue;
+
+		if (stream == SNDRV_PCM_STREAM_PLAYBACK)
+			mute_fe_port_playback(fe, be, 0);
+		else
+			mute_fe_port_capture(fe, be, 0);
+	}
+}
+
 static void capture_trigger(struct snd_pcm_substream *substream,
 		struct snd_soc_dai *dai, int cmd)
 {
@@ -640,6 +796,9 @@ static void capture_trigger(struct snd_pcm_substream *substream,
 			/* Enable Frontend sDMA  */
 			snd_soc_dsp_platform_trigger(substream, cmd, fe->platform);
 			enable_fe_port(substream, dai, stream);
+
+			/* unmute FE port */
+			unmute_fe_port(substream, dai, stream);
 		}
 
 		/* Restore ABE GAINS AMIC */
@@ -665,9 +824,11 @@ static void capture_trigger(struct snd_pcm_substream *substream,
 		snd_soc_dsp_platform_trigger(substream, cmd, fe->platform);
 		break;
 	case SNDRV_PCM_TRIGGER_STOP:
-
 		/* does this trigger() apply to the FE ? */
 		if (snd_soc_dsp_is_trigger_for_fe(fe, stream)) {
+			/* mute FE port */
+			mute_fe_port(substream, dai, stream);
+
 			/* Disable sDMA */
 			disable_fe_port(substream, dai, stream);
 			snd_soc_dsp_platform_trigger(substream, cmd, fe->platform);
@@ -691,6 +852,9 @@ static void capture_trigger(struct snd_pcm_substream *substream,
 
 			be_substream = snd_soc_dsp_get_substream(be, stream);
 
+			/* mute the BE port */
+			mute_be(be, dai, stream);
+
 			/* disable the BE port */
 			disable_be_port(be, dai, stream);
 
@@ -756,14 +920,14 @@ static void playback_trigger(struct snd_pcm_substream *substream,
 
 		/* does this trigger() apply to the FE ? */
 		if (snd_soc_dsp_is_trigger_for_fe(fe, stream)) {
-
 			/* Enable Frontend sDMA  */
 			snd_soc_dsp_platform_trigger(substream, cmd, fe->platform);
 			enable_fe_port(substream, dai, stream);
-
-			/* unmute FE port */
-			unmute_fe_port(substream, dai, stream);
 		}
+
+		/* unmute FE port (sensitive to runtime udpates) */
+		unmute_fe_port(substream, dai, stream);
+
 		break;
 	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
 		/* Enable Frontend sDMA  */
@@ -784,11 +948,11 @@ static void playback_trigger(struct snd_pcm_substream *substream,
 	case SNDRV_PCM_TRIGGER_STOP:
 	case SNDRV_PCM_TRIGGER_SUSPEND:
 
+		/* mute FE port (sensitive to runtime udpates) */
+		mute_fe_port(substream, dai, stream);
+
 		/* does this trigger() apply to the FE ? */
 		if (snd_soc_dsp_is_trigger_for_fe(fe, stream)) {
-			/* mute FE port */
-			mute_fe_port(substream, dai, stream);
-
 			/* disable the transfer */
 			disable_fe_port(substream, dai, stream);
 			snd_soc_dsp_platform_trigger(substream, cmd, fe->platform);
@@ -812,6 +976,9 @@ static void playback_trigger(struct snd_pcm_substream *substream,
 
 			be_substream = snd_soc_dsp_get_substream(be, stream);
 
+			/* mute the BE port */
+			mute_be(be, dai, stream);
+
 			/* disable the BE */
 			disable_be_port(be, dai, stream);
 
@@ -1061,12 +1228,14 @@ static int omap_abe_dai_trigger(struct snd_pcm_substream *substream,
 static int omap_abe_dai_bespoke_trigger(struct snd_pcm_substream *substream,
 				  int cmd, struct snd_soc_dai *dai)
 {
+	struct snd_soc_pcm_runtime *fe = substream->private_data;
 	struct omap_abe_data *abe_priv = snd_soc_dai_get_drvdata(dai);
 	int ret = 0;
 
 	dev_dbg(dai->dev, "%s: %s cmd %d\n", __func__, dai->name, cmd);
 
-	if (dai->id == ABE_FRONTEND_DAI_MODEM) {
+	if ((dai->id == ABE_FRONTEND_DAI_MODEM) &&
+			snd_soc_dsp_is_trigger_for_fe(fe, substream->stream)) {
 
 		dev_dbg(abe_priv->modem_dai->dev, "%s: MODEM stream %d cmd %d\n",
 				__func__, substream->stream, cmd);
@@ -1235,9 +1404,6 @@ static int omap_abe_dai_probe(struct snd_soc_dai *dai)
 	}
 
 	snd_soc_dai_set_drvdata(dai, abe_priv);
-
-	dl1_gain = GAIN_0dB;
-
 	return 0;
 
 err_port:
diff --git a/sound/soc/soc-dsp.c b/sound/soc/soc-dsp.c
index ebb3dd7..24c8e5e 100644
--- a/sound/soc/soc-dsp.c
+++ b/sound/soc/soc-dsp.c
@@ -1124,6 +1124,10 @@ int soc_dsp_runtime_update(struct snd_soc_dapm_widget *widget)
 		/* DAPM sync will call this to update DSP paths */
 		dev_dbg(card->dev, "DSP runtime update for FE %s\n", fe->dai_link->name);
 
+		/* skip if FE doesn't have playback capability */
+		if (!fe->cpu_dai->driver->playback.channels_min)
+			goto capture;
+
 		/* update any playback paths */
 		start = dsp_add_new_paths(fe, SNDRV_PCM_STREAM_PLAYBACK, 1);
 		stop = dsp_prune_old_paths(fe, SNDRV_PCM_STREAM_PLAYBACK, 1);
@@ -1142,6 +1146,10 @@ int soc_dsp_runtime_update(struct snd_soc_dapm_widget *widget)
 		fe_clear_pending(fe, SNDRV_PCM_STREAM_PLAYBACK);
 
 capture:
+		/* skip if FE doesn't have capture capability */
+		if (!fe->cpu_dai->driver->capture.channels_min)
+			continue;
+
 		/* update any capture paths */
 		start = dsp_add_new_paths(fe, SNDRV_PCM_STREAM_CAPTURE, 1);
 		stop = dsp_prune_old_paths(fe, SNDRV_PCM_STREAM_CAPTURE, 1);