Merge branch 'asoc-5.4' into asoc-next

7 files changed, 151 insertions, 81 deletions

diff --git a/sound/soc/ti/Kconfig b/sound/soc/ti/Kconfig
index 2197f3e1eaed..87a9b9dd4e98 100644
--- a/sound/soc/ti/Kconfig
+++ b/sound/soc/ti/Kconfig
@@ -12,7 +12,7 @@ config SND_SOC_TI_SDMA_PCM
 
 comment "Texas Instruments DAI support for:"
 config SND_SOC_DAVINCI_ASP
-	tristate "daVinci Audio Serial Port (ASP) or McBSP suport"
+	tristate "daVinci Audio Serial Port (ASP) or McBSP support"
 	depends on ARCH_DAVINCI || COMPILE_TEST
 	select SND_SOC_TI_EDMA_PCM
 	help
@@ -33,7 +33,7 @@ config SND_SOC_DAVINCI_MCASP
 	  - Keystone devices
 
 config SND_SOC_DAVINCI_VCIF
-	tristate "daVinci Voice Interface (VCIF) suport"
+	tristate "daVinci Voice Interface (VCIF) support"
 	depends on ARCH_DAVINCI || COMPILE_TEST
 	select SND_SOC_TI_EDMA_PCM
 	help
diff --git a/sound/soc/ti/davinci-evm.c b/sound/soc/ti/davinci-evm.c
index bfd8d1a03ba7..686b23d7a90d 100644
--- a/sound/soc/ti/davinci-evm.c
+++ b/sound/soc/ti/davinci-evm.c
@@ -68,7 +68,7 @@ static int evm_hw_params(struct snd_pcm_substream *substream,
 
 	/* set the CPU system clock */
 	ret = snd_soc_dai_set_sysclk(cpu_dai, 0, sysclk, SND_SOC_CLOCK_OUT);
-	if (ret < 0)
+	if (ret < 0 && ret != -ENOTSUPP)
 		return ret;
 
 	return 0;
diff --git a/sound/soc/ti/davinci-i2s.c b/sound/soc/ti/davinci-i2s.c
index 27afdbb9adf3..d89b5c928c4d 100644
--- a/sound/soc/ti/davinci-i2s.c
+++ b/sound/soc/ti/davinci-i2s.c
@@ -598,6 +598,8 @@ static void davinci_i2s_shutdown(struct snd_pcm_substream *substream,
 }
 
 #define DAVINCI_I2S_RATES	SNDRV_PCM_RATE_8000_96000
+#define DAVINCI_I2S_FORMATS	(SNDRV_PCM_FMTBIT_S16_LE | \
+				 SNDRV_PCM_FMTBIT_S32_LE)
 
 static const struct snd_soc_dai_ops davinci_i2s_dai_ops = {
 	.shutdown	= davinci_i2s_shutdown,
@@ -625,12 +627,14 @@ static struct snd_soc_dai_driver davinci_i2s_dai = {
 		.channels_min = 2,
 		.channels_max = 2,
 		.rates = DAVINCI_I2S_RATES,
-		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
+		.formats = DAVINCI_I2S_FORMATS,
+	},
 	.capture = {
 		.channels_min = 2,
 		.channels_max = 2,
 		.rates = DAVINCI_I2S_RATES,
-		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
+		.formats = DAVINCI_I2S_FORMATS,
+	},
 	.ops = &davinci_i2s_dai_ops,
 
 };
diff --git a/sound/soc/ti/davinci-mcasp.c b/sound/soc/ti/davinci-mcasp.c
index bc7bf15ed7a4..7aa3c32e4a49 100644
--- a/sound/soc/ti/davinci-mcasp.c
+++ b/sound/soc/ti/davinci-mcasp.c
@@ -109,6 +109,8 @@ struct davinci_mcasp {
 
 	/* Used for comstraint setting on the second stream */
 	u32	channels;
+	int	max_format_width;
+	u8	active_serializers[2];
 
 #ifdef CONFIG_GPIOLIB
 	struct gpio_chip gpio_chip;
@@ -466,6 +468,7 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai,
 		/* FS need to be inverted */
 		inv_fs = true;
 		break;
+	case SND_SOC_DAIFMT_RIGHT_J:
 	case SND_SOC_DAIFMT_LEFT_J:
 		/* configure a full-word SYNC pulse (LRCLK) */
 		mcasp_set_bits(mcasp, DAVINCI_MCASP_TXFMCTL_REG, FSXDUR);
@@ -759,34 +762,30 @@ static int davinci_config_channel_size(struct davinci_mcasp *mcasp,
 				       int sample_width)
 {
 	u32 fmt;
-	u32 tx_rotate = (sample_width / 4) & 0x7;
+	u32 tx_rotate, rx_rotate, slot_width;
 	u32 mask = (1ULL << sample_width) - 1;
-	u32 slot_width = sample_width;
-
-	/*
-	 * For captured data we should not rotate, inversion and masking is
-	 * enoguh to get the data to the right position:
-	 * Format	  data from bus		after reverse (XRBUF)
-	 * S16_LE:	|LSB|MSB|xxx|xxx|	|xxx|xxx|MSB|LSB|
-	 * S24_3LE:	|LSB|DAT|MSB|xxx|	|xxx|MSB|DAT|LSB|
-	 * S24_LE:	|LSB|DAT|MSB|xxx|	|xxx|MSB|DAT|LSB|
-	 * S32_LE:	|LSB|DAT|DAT|MSB|	|MSB|DAT|DAT|LSB|
-	 */
-	u32 rx_rotate = 0;
 
+	if (mcasp->slot_width)
+		slot_width = mcasp->slot_width;
+	else if (mcasp->max_format_width)
+		slot_width = mcasp->max_format_width;
+	else
+		slot_width = sample_width;
 	/*
-	 * Setting the tdm slot width either with set_clkdiv() or
-	 * set_tdm_slot() allows us to for example send 32 bits per
-	 * channel to the codec, while only 16 of them carry audio
-	 * payload.
+	 * TX rotation:
+	 * right aligned formats: rotate w/ slot_width
+	 * left aligned formats: rotate w/ sample_width
+	 *
+	 * RX rotation:
+	 * right aligned formats: no rotation needed
+	 * left aligned formats: rotate w/ (slot_width - sample_width)
 	 */
-	if (mcasp->slot_width) {
-		/*
-		 * When we have more bclk then it is needed for the
-		 * data, we need to use the rotation to move the
-		 * received samples to have correct alignment.
-		 */
-		slot_width = mcasp->slot_width;
+	if ((mcasp->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) ==
+	    SND_SOC_DAIFMT_RIGHT_J) {
+		tx_rotate = (slot_width / 4) & 0x7;
+		rx_rotate = 0;
+	} else {
+		tx_rotate = (sample_width / 4) & 0x7;
 		rx_rotate = (slot_width - sample_width) / 4;
 	}
 
@@ -819,6 +818,7 @@ static int mcasp_common_hw_param(struct davinci_mcasp *mcasp, int stream,
 	u8 rx_ser = 0;
 	u8 slots = mcasp->tdm_slots;
 	u8 max_active_serializers = (channels + slots - 1) / slots;
+	u8 max_rx_serializers, max_tx_serializers;
 	int active_serializers, numevt;
 	u32 reg;
 	/* Default configuration */
@@ -828,22 +828,28 @@ static int mcasp_common_hw_param(struct davinci_mcasp *mcasp, int stream,
 	if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
 		mcasp_set_reg(mcasp, DAVINCI_MCASP_TXSTAT_REG, 0xFFFFFFFF);
 		mcasp_clr_bits(mcasp, DAVINCI_MCASP_XEVTCTL_REG, TXDATADMADIS);
+		max_tx_serializers = max_active_serializers;
+		max_rx_serializers =
+			mcasp->active_serializers[SNDRV_PCM_STREAM_CAPTURE];
 	} else {
 		mcasp_set_reg(mcasp, DAVINCI_MCASP_RXSTAT_REG, 0xFFFFFFFF);
 		mcasp_clr_bits(mcasp, DAVINCI_MCASP_REVTCTL_REG, RXDATADMADIS);
+		max_tx_serializers =
+			mcasp->active_serializers[SNDRV_PCM_STREAM_PLAYBACK];
+		max_rx_serializers = max_active_serializers;
 	}
 
 	for (i = 0; i < mcasp->num_serializer; i++) {
 		mcasp_set_bits(mcasp, DAVINCI_MCASP_XRSRCTL_REG(i),
 			       mcasp->serial_dir[i]);
 		if (mcasp->serial_dir[i] == TX_MODE &&
-					tx_ser < max_active_serializers) {
+					tx_ser < max_tx_serializers) {
 			mcasp_mod_bits(mcasp, DAVINCI_MCASP_XRSRCTL_REG(i),
 				       mcasp->dismod, DISMOD_MASK);
 			set_bit(PIN_BIT_AXR(i), &mcasp->pdir);
 			tx_ser++;
 		} else if (mcasp->serial_dir[i] == RX_MODE &&
-					rx_ser < max_active_serializers) {
+					rx_ser < max_rx_serializers) {
 			clear_bit(PIN_BIT_AXR(i), &mcasp->pdir);
 			rx_ser++;
 		} else {
@@ -890,7 +896,8 @@ static int mcasp_common_hw_param(struct davinci_mcasp *mcasp, int stream,
 		} else {
 			dma_data->maxburst = 0;
 		}
-		return 0;
+
+		goto out;
 	}
 
 	if (period_words % active_serializers) {
@@ -920,6 +927,9 @@ static int mcasp_common_hw_param(struct davinci_mcasp *mcasp, int stream,
 		numevt = 0;
 	dma_data->maxburst = numevt;
 
+out:
+	mcasp->active_serializers[stream] = active_serializers;
+
 	return 0;
 }
 
@@ -1159,6 +1169,37 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream,
 	int period_size = params_period_size(params);
 	int ret;
 
+	switch (params_format(params)) {
+	case SNDRV_PCM_FORMAT_U8:
+	case SNDRV_PCM_FORMAT_S8:
+		word_length = 8;
+		break;
+
+	case SNDRV_PCM_FORMAT_U16_LE:
+	case SNDRV_PCM_FORMAT_S16_LE:
+		word_length = 16;
+		break;
+
+	case SNDRV_PCM_FORMAT_U24_3LE:
+	case SNDRV_PCM_FORMAT_S24_3LE:
+		word_length = 24;
+		break;
+
+	case SNDRV_PCM_FORMAT_U24_LE:
+	case SNDRV_PCM_FORMAT_S24_LE:
+		word_length = 24;
+		break;
+
+	case SNDRV_PCM_FORMAT_U32_LE:
+	case SNDRV_PCM_FORMAT_S32_LE:
+		word_length = 32;
+		break;
+
+	default:
+		printk(KERN_WARNING "davinci-mcasp: unsupported PCM format");
+		return -EINVAL;
+	}
+
 	ret = davinci_mcasp_set_dai_fmt(cpu_dai, mcasp->dai_fmt);
 	if (ret)
 		return ret;
@@ -1193,41 +1234,13 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream,
 	if (ret)
 		return ret;
 
-	switch (params_format(params)) {
-	case SNDRV_PCM_FORMAT_U8:
-	case SNDRV_PCM_FORMAT_S8:
-		word_length = 8;
-		break;
-
-	case SNDRV_PCM_FORMAT_U16_LE:
-	case SNDRV_PCM_FORMAT_S16_LE:
-		word_length = 16;
-		break;
-
-	case SNDRV_PCM_FORMAT_U24_3LE:
-	case SNDRV_PCM_FORMAT_S24_3LE:
-		word_length = 24;
-		break;
-
-	case SNDRV_PCM_FORMAT_U24_LE:
-	case SNDRV_PCM_FORMAT_S24_LE:
-		word_length = 24;
-		break;
-
-	case SNDRV_PCM_FORMAT_U32_LE:
-	case SNDRV_PCM_FORMAT_S32_LE:
-		word_length = 32;
-		break;
-
-	default:
-		printk(KERN_WARNING "davinci-mcasp: unsupported PCM format");
-		return -EINVAL;
-	}
-
 	davinci_config_channel_size(mcasp, word_length);
 
-	if (mcasp->op_mode == DAVINCI_MCASP_IIS_MODE)
+	if (mcasp->op_mode == DAVINCI_MCASP_IIS_MODE) {
 		mcasp->channels = channels;
+		if (!mcasp->max_format_width)
+			mcasp->max_format_width = word_length;
+	}
 
 	return 0;
 }
@@ -1279,6 +1292,28 @@ static int davinci_mcasp_hw_rule_slot_width(struct snd_pcm_hw_params *params,
 	return snd_mask_refine(fmt, &nfmt);
 }
 
+static int davinci_mcasp_hw_rule_format_width(struct snd_pcm_hw_params *params,
+					      struct snd_pcm_hw_rule *rule)
+{
+	struct davinci_mcasp_ruledata *rd = rule->private;
+	struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
+	struct snd_mask nfmt;
+	int i, format_width;
+
+	snd_mask_none(&nfmt);
+	format_width = rd->mcasp->max_format_width;
+
+	for (i = 0; i <= SNDRV_PCM_FORMAT_LAST; i++) {
+		if (snd_mask_test(fmt, i)) {
+			if (snd_pcm_format_width(i) == format_width) {
+				snd_mask_set(&nfmt, i);
+			}
+		}
+	}
+
+	return snd_mask_refine(fmt, &nfmt);
+}
+
 static const unsigned int davinci_mcasp_dai_rates[] = {
 	8000, 11025, 16000, 22050, 32000, 44100, 48000, 64000,
 	88200, 96000, 176400, 192000,
@@ -1433,12 +1468,13 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream,
 	max_channels *= tdm_slots;
 	/*
 	 * If the already active stream has less channels than the calculated
-	 * limnit based on the seirializers * tdm_slots, we need to use that as
-	 * a constraint for the second stream.
-	 * Otherwise (first stream or less allowed channels) we use the
-	 * calculated constraint.
+	 * limit based on the seirializers * tdm_slots, and only one serializer
+	 * is in use we need to use that as a constraint for the second stream.
+	 * Otherwise (first stream or less allowed channels or more than one
+	 * serializer in use) we use the calculated constraint.
 	 */
-	if (mcasp->channels && mcasp->channels < max_channels)
+	if (mcasp->channels && mcasp->channels < max_channels &&
+	    ruledata->serializers == 1)
 		max_channels = mcasp->channels;
 	/*
 	 * But we can always allow channels upto the amount of
@@ -1455,7 +1491,20 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream,
 				   0, SNDRV_PCM_HW_PARAM_CHANNELS,
 				   &mcasp->chconstr[substream->stream]);
 
-	if (mcasp->slot_width) {
+	if (mcasp->max_format_width) {
+		/*
+		 * Only allow formats which require same amount of bits on the
+		 * bus as the currently running stream
+		 */
+		ret = snd_pcm_hw_rule_add(substream->runtime, 0,
+					  SNDRV_PCM_HW_PARAM_FORMAT,
+					  davinci_mcasp_hw_rule_format_width,
+					  ruledata,
+					  SNDRV_PCM_HW_PARAM_FORMAT, -1);
+		if (ret)
+			return ret;
+	}
+	else if (mcasp->slot_width) {
 		/* Only allow formats require <= slot_width bits on the bus */
 		ret = snd_pcm_hw_rule_add(substream->runtime, 0,
 					  SNDRV_PCM_HW_PARAM_FORMAT,
@@ -1501,12 +1550,15 @@ static void davinci_mcasp_shutdown(struct snd_pcm_substream *substream,
 	struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(cpu_dai);
 
 	mcasp->substreams[substream->stream] = NULL;
+	mcasp->active_serializers[substream->stream] = 0;
 
 	if (mcasp->op_mode == DAVINCI_MCASP_DIT_MODE)
 		return;
 
-	if (!cpu_dai->active)
+	if (!cpu_dai->active) {
 		mcasp->channels = 0;
+		mcasp->max_format_width = 0;
+	}
 }
 
 static const struct snd_soc_dai_ops davinci_mcasp_dai_ops = {
@@ -1562,7 +1614,6 @@ static struct snd_soc_dai_driver davinci_mcasp_dai[] = {
 		},
 		.ops 		= &davinci_mcasp_dai_ops,
 
-		.symmetric_samplebits	= 1,
 		.symmetric_rates	= 1,
 	},
 	{
diff --git a/sound/soc/ti/edma-pcm.c b/sound/soc/ti/edma-pcm.c
index 3ebea1bd15cb..634b040b65f0 100644
--- a/sound/soc/ti/edma-pcm.c
+++ b/sound/soc/ti/edma-pcm.c
@@ -39,7 +39,22 @@ static const struct snd_dmaengine_pcm_config edma_dmaengine_pcm_config = {
 
 int edma_pcm_platform_register(struct device *dev)
 {
-	return devm_snd_dmaengine_pcm_register(dev, &edma_dmaengine_pcm_config, 0);
+	struct snd_dmaengine_pcm_config *config;
+
+	if (dev->of_node)
+		return devm_snd_dmaengine_pcm_register(dev,
+						&edma_dmaengine_pcm_config, 0);
+
+	config = devm_kzalloc(dev, sizeof(*config), GFP_KERNEL);
+	if (!config)
+		return -ENOMEM;
+
+	*config = edma_dmaengine_pcm_config;
+
+	config->chan_names[0] = "tx";
+	config->chan_names[1] = "rx";
+
+	return devm_snd_dmaengine_pcm_register(dev, config, 0);
 }
 EXPORT_SYMBOL_GPL(edma_pcm_platform_register);
 
diff --git a/sound/soc/ti/n810.c b/sound/soc/ti/n810.c
index 2c3f2a4c1700..3ad2b6daf31e 100644
--- a/sound/soc/ti/n810.c
+++ b/sound/soc/ti/n810.c
@@ -46,6 +46,7 @@ static void n810_ext_control(struct snd_soc_dapm_context *dapm)
 	switch (n810_jack_func) {
 	case N810_JACK_HS:
 		line1l = 1;
+		/* fall through */
 	case N810_JACK_HP:
 		hp = 1;
 		break;
diff --git a/sound/soc/ti/rx51.c b/sound/soc/ti/rx51.c
index bc6046534fa5..588f680a9c24 100644
--- a/sound/soc/ti/rx51.c
+++ b/sound/soc/ti/rx51.c
@@ -55,6 +55,7 @@ static void rx51_ext_control(struct snd_soc_dapm_context *dapm)
 		break;
 	case RX51_JACK_HS:
 		hs = 1;
+		/* fall through */
 	case RX51_JACK_HP:
 		hp = 1;
 		break;
@@ -318,12 +319,10 @@ static struct snd_soc_dai_link rx51_dai[] = {
 
 static struct snd_soc_aux_dev rx51_aux_dev[] = {
 	{
-		.name = "TLV320AIC34b",
-		.codec_name = "tlv320aic3x-codec.2-0019",
+		.dlc = COMP_AUX("tlv320aic3x-codec.2-0019"),
 	},
 	{
-		.name = "TPA61320A2",
-		.codec_name = "tpa6130a2.2-0060",
+		.dlc = COMP_AUX("tpa6130a2.2-0060"),
 	},
 };
 
@@ -396,8 +395,8 @@ static int rx51_soc_probe(struct platform_device *pdev)
 			dev_err(&pdev->dev, "Auxiliary Codec node is not provided\n");
 			return -EINVAL;
 		}
-		rx51_aux_dev[0].codec_name = NULL;
-		rx51_aux_dev[0].codec_of_node = dai_node;
+		rx51_aux_dev[0].dlc.name = NULL;
+		rx51_aux_dev[0].dlc.of_node = dai_node;
 		rx51_codec_conf[0].dev_name = NULL;
 		rx51_codec_conf[0].of_node = dai_node;
 
@@ -406,8 +405,8 @@ static int rx51_soc_probe(struct platform_device *pdev)
 			dev_err(&pdev->dev, "Headphone amplifier node is not provided\n");
 			return -EINVAL;
 		}
-		rx51_aux_dev[1].codec_name = NULL;
-		rx51_aux_dev[1].codec_of_node = dai_node;
+		rx51_aux_dev[1].dlc.name = NULL;
+		rx51_aux_dev[1].dlc.of_node = dai_node;
 		rx51_codec_conf[1].dev_name = NULL;
 		rx51_codec_conf[1].of_node = dai_node;
 	}