Merge branch 'asoc-5.1' into asoc-linus

sound/soc/codecs/da7213.c

@@ -1305,7 +1305,10 @@ static int da7213_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
 	/* By default only 64 BCLK per WCLK is supported */
 	dai_clk_mode |= DA7213_DAI_BCLKS_PER_WCLK_64;
 
-	snd_soc_component_write(component, DA7213_DAI_CLK_MODE, dai_clk_mode);
+	snd_soc_component_update_bits(component, DA7213_DAI_CLK_MODE,
+			    DA7213_DAI_BCLKS_PER_WCLK_MASK |
+			    DA7213_DAI_CLK_POL_MASK | DA7213_DAI_WCLK_POL_MASK,
+			    dai_clk_mode);
 	snd_soc_component_update_bits(component, DA7213_DAI_CTRL, DA7213_DAI_FORMAT_MASK,
 			    dai_ctrl);
 	snd_soc_component_write(component, DA7213_DAI_OFFSET, dai_offset);

sound/soc/codecs/da7213.h

@@ -181,7 +181,9 @@
 #define DA7213_DAI_BCLKS_PER_WCLK_256				(0x3 << 0)
 #define DA7213_DAI_BCLKS_PER_WCLK_MASK				(0x3 << 0)
 #define DA7213_DAI_CLK_POL_INV					(0x1 << 2)
+#define DA7213_DAI_CLK_POL_MASK					(0x1 << 2)
 #define DA7213_DAI_WCLK_POL_INV					(0x1 << 3)
+#define DA7213_DAI_WCLK_POL_MASK				(0x1 << 3)
 #define DA7213_DAI_CLK_EN_MASK					(0x1 << 7)
 
 /* DA7213_DAI_CTRL = 0x29 */

sound/soc/codecs/da7219.c

@@ -1583,20 +1583,26 @@ static const struct snd_soc_dai_ops da7219_dai_ops = {
 #define DA7219_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
 			SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
 
+#define DA7219_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
+		      SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\
+		      SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |\
+		      SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 |\
+		      SNDRV_PCM_RATE_96000)
+
 static struct snd_soc_dai_driver da7219_dai = {
 	.name = "da7219-hifi",
 	.playback = {
 		.stream_name = "Playback",
 		.channels_min = 1,
 		.channels_max = DA7219_DAI_CH_NUM_MAX,
-		.rates = SNDRV_PCM_RATE_8000_96000,
+		.rates = DA7219_RATES,
 		.formats = DA7219_FORMATS,
 	},
 	.capture = {
 		.stream_name = "Capture",
 		.channels_min = 1,
 		.channels_max = DA7219_DAI_CH_NUM_MAX,
-		.rates = SNDRV_PCM_RATE_8000_96000,
+		.rates = DA7219_RATES,
 		.formats = DA7219_FORMATS,
 	},
 	.ops = &da7219_dai_ops,

sound/soc/codecs/hdac_hdmi.c

@@ -1854,6 +1854,17 @@ static int hdmi_codec_probe(struct snd_soc_component *component)
 	/* Imp: Store the card pointer in hda_codec */
 	hdmi->card = dapm->card->snd_card;
 
+	/*
+	 * Setup a device_link between card device and HDMI codec device.
+	 * The card device is the consumer and the HDMI codec device is
+	 * the supplier. With this setting, we can make sure that the audio
+	 * domain in display power will be always turned on before operating
+	 * on the HDMI audio codec registers.
+	 * Let's use the flag DL_FLAG_AUTOREMOVE_CONSUMER. This can make
+	 * sure the device link is freed when the machine driver is removed.
+	 */
+	device_link_add(component->card->dev, &hdev->dev, DL_FLAG_RPM_ACTIVE |
+			DL_FLAG_AUTOREMOVE_CONSUMER);
 	/*
 	 * hdac_device core already sets the state to active and calls
 	 * get_noresume. So enable runtime and set the device to suspend.

sound/soc/codecs/hdmi-codec.c

@@ -439,9 +439,13 @@ static int hdmi_codec_startup(struct snd_pcm_substream *substream,
 		if (!ret) {
 			ret = snd_pcm_hw_constraint_eld(substream->runtime,
 							hcp->eld);
-			if (ret)
+			if (ret) {
+				mutex_lock(&hcp->current_stream_lock);
+				hcp->current_stream = NULL;
+				mutex_unlock(&hcp->current_stream_lock);
 				return ret;
 			}
+		}
 		/* Select chmap supported */
 		hdmi_codec_eld_chmap(hcp);
 	}

sound/soc/codecs/rt5682.c

@@ -2588,6 +2588,7 @@ static int rt5682_i2c_probe(struct i2c_client *i2c,
 
 	rt5682_reset(rt5682->regmap);
 
+	mutex_init(&rt5682->calibrate_mutex);
 	rt5682_calibrate(rt5682);
 
 	ret = regmap_multi_reg_write(rt5682->regmap, patch_list,
@@ -2654,7 +2655,6 @@ static int rt5682_i2c_probe(struct i2c_client *i2c,
 	INIT_DELAYED_WORK(&rt5682->jd_check_work,
 				rt5682_jd_check_handler);
 
-	mutex_init(&rt5682->calibrate_mutex);
 
 	if (i2c->irq) {
 		ret = devm_request_threaded_irq(&i2c->dev, i2c->irq, NULL,

sound/soc/mediatek/common/mtk-btcvsd.c

@@ -193,13 +193,13 @@ static const u8 table_msbc_silence[SCO_PACKET_180] = {
 static void mtk_btcvsd_snd_irq_enable(struct mtk_btcvsd_snd *bt)
 {
 	regmap_update_bits(bt->infra, bt->infra_misc_offset,
-			   bt->conn_bt_cvsd_mask, bt->conn_bt_cvsd_mask);
+			   bt->conn_bt_cvsd_mask, 0);
 }
 
 static void mtk_btcvsd_snd_irq_disable(struct mtk_btcvsd_snd *bt)
 {
 	regmap_update_bits(bt->infra, bt->infra_misc_offset,
-			   bt->conn_bt_cvsd_mask, 0);
+			   bt->conn_bt_cvsd_mask, bt->conn_bt_cvsd_mask);
 }
 
 static void mtk_btcvsd_snd_set_state(struct mtk_btcvsd_snd *bt,

sound/soc/sh/rcar/core.c

@@ -300,6 +300,18 @@ int rsnd_runtime_channel_after_ctu_with_params(struct rsnd_dai_stream *io,
 	return chan;
 }
 
+int rsnd_channel_normalization(int chan)
+{
+	if ((chan > 8) || (chan < 0))
+		return 0;
+
+	/* TDM Extend Mode needs 8ch */
+	if (chan == 6)
+		chan = 8;
+
+	return chan;
+}
+
 int rsnd_runtime_channel_for_ssi_with_params(struct rsnd_dai_stream *io,
 					     struct snd_pcm_hw_params *params)
 {
@@ -312,11 +324,7 @@ int rsnd_runtime_channel_for_ssi_with_params(struct rsnd_dai_stream *io,
 	if (rsnd_runtime_is_multi_ssi(io))
 		chan /= rsnd_rdai_ssi_lane_get(rdai);
 
-	/* TDM Extend Mode needs 8ch */
-	if (chan == 6)
-		chan = 8;
-
-	return chan;
+	return rsnd_channel_normalization(chan);
 }
 
 int rsnd_runtime_is_multi_ssi(struct rsnd_dai_stream *io)

sound/soc/sh/rcar/rsnd.h

@@ -446,6 +446,7 @@ void rsnd_parse_connect_common(struct rsnd_dai *rdai,
 		struct device_node *playback,
 		struct device_node *capture);
 
+int rsnd_channel_normalization(int chan);
 #define rsnd_runtime_channel_original(io) \
 	rsnd_runtime_channel_original_with_params(io, NULL)
 int rsnd_runtime_channel_original_with_params(struct rsnd_dai_stream *io,

sound/soc/sh/rcar/ssi.c

@@ -303,6 +303,8 @@ static int rsnd_ssi_master_clk_start(struct rsnd_mod *mod,
 	if (rsnd_runtime_is_tdm_split(io))
 		chan = rsnd_io_converted_chan(io);
 
+	chan = rsnd_channel_normalization(chan);
+
 	main_rate = rsnd_ssi_clk_query(rdai, rate, chan, &idx);
 	if (!main_rate) {
 		dev_err(dev, "unsupported clock rate\n");

sound/soc/soc-dapm.c

@@ -883,6 +883,7 @@ static int dapm_create_or_share_kcontrol(struct snd_soc_dapm_widget *w,
 			case snd_soc_dapm_switch:
 			case snd_soc_dapm_mixer:
 			case snd_soc_dapm_pga:
+			case snd_soc_dapm_effect:
 			case snd_soc_dapm_out_drv:
 				wname_in_long_name = true;
 				kcname_in_long_name = true;
@@ -2370,6 +2371,7 @@ static ssize_t dapm_widget_show_component(struct snd_soc_component *cmpnt,
 		case snd_soc_dapm_dac:
 		case snd_soc_dapm_adc:
 		case snd_soc_dapm_pga:
+		case snd_soc_dapm_effect:
 		case snd_soc_dapm_out_drv:
 		case snd_soc_dapm_mixer:
 		case snd_soc_dapm_mixer_named_ctl:
@@ -3197,6 +3199,7 @@ int snd_soc_dapm_new_widgets(struct snd_soc_card *card)
 			dapm_new_mux(w);
 			break;
 		case snd_soc_dapm_pga:
+		case snd_soc_dapm_effect:
 		case snd_soc_dapm_out_drv:
 			dapm_new_pga(w);
 			break;

sound/soc/soc-pcm.c

@@ -43,8 +43,8 @@ static bool snd_soc_dai_stream_valid(struct snd_soc_dai *dai, int stream)
 	else
 		codec_stream = &dai->driver->capture;
 
-	/* If the codec specifies any rate at all, it supports the stream. */
-	return codec_stream->rates;
+	/* If the codec specifies any channels at all, it supports the stream */
+	return codec_stream->channels_min;
 }
 
 /**
@@ -1033,6 +1033,9 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream,
 
 codec_err:
 	for_each_rtd_codec_dai_rollback(rtd, i, codec_dai) {
+		if (!snd_soc_dai_stream_valid(codec_dai, substream->stream))
+			continue;
+
 		if (codec_dai->driver->ops->hw_free)
 			codec_dai->driver->ops->hw_free(substream, codec_dai);
 		codec_dai->rate = 0;
@@ -1090,6 +1093,9 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream)
 
 	/* now free hw params for the DAIs  */
 	for_each_rtd_codec_dai(rtd, i, codec_dai) {
+		if (!snd_soc_dai_stream_valid(codec_dai, substream->stream))
+			continue;
+
 		if (codec_dai->driver->ops->hw_free)
 			codec_dai->driver->ops->hw_free(substream, codec_dai);
 	}

sound/soc/stm/stm32_i2s.c

@@ -845,8 +845,9 @@ static int stm32_i2s_parse_dt(struct platform_device *pdev,
 	/* Get irqs */
 	irq = platform_get_irq(pdev, 0);
 	if (irq < 0) {
+		if (irq != -EPROBE_DEFER)
 			dev_err(&pdev->dev, "no irq for node %s\n", pdev->name);
-		return -ENOENT;
+		return irq;
 	}
 
 	ret = devm_request_irq(&pdev->dev, irq, stm32_i2s_isr, IRQF_ONESHOT,