Merge tag 'asoc-fix-v5.8' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus

ASoC: Fixes for v5.8

A small pile of fixes that came in during the merge window, the DPCM
fixes from Pierre are the most notable thing here.

15 files changed, 134 insertions, 36 deletions

diff --git a/sound/soc/codecs/max98390.c b/sound/soc/codecs/max98390.c
index b9ce44dda886..0d63ebfbff2f 100644
--- a/sound/soc/codecs/max98390.c
+++ b/sound/soc/codecs/max98390.c
@@ -754,6 +754,7 @@ static struct snd_soc_dai_driver max98390_dai[] = {
 static int max98390_dsm_init(struct snd_soc_component *component)
 {
 	int ret;
+	int param_size, param_start_addr;
 	char filename[128];
 	const char *vendor, *product;
 	struct max98390_priv *max98390 =
@@ -778,16 +779,31 @@ static int max98390_dsm_init(struct snd_soc_component *component)
 	}
 
 	dev_dbg(component->dev,
-		"max98390: param fw size %ld\n",
+		"max98390: param fw size %zd\n",
 		fw->size);
+	if (fw->size < MAX98390_DSM_PARAM_MIN_SIZE) {
+		dev_err(component->dev,
+			"param fw is invalid.\n");
+		goto err_alloc;
+	}
 	dsm_param = (char *)fw->data;
+	param_start_addr = (dsm_param[0] & 0xff) | (dsm_param[1] & 0xff) << 8;
+	param_size = (dsm_param[2] & 0xff) | (dsm_param[3] & 0xff) << 8;
+	if (param_size > MAX98390_DSM_PARAM_MAX_SIZE ||
+		param_start_addr < DSM_STBASS_HPF_B0_BYTE0 ||
+		fw->size < param_size + MAX98390_DSM_PAYLOAD_OFFSET) {
+		dev_err(component->dev,
+			"param fw is invalid.\n");
+		goto err_alloc;
+	}
+	regmap_write(max98390->regmap, MAX98390_R203A_AMP_EN, 0x80);
 	dsm_param += MAX98390_DSM_PAYLOAD_OFFSET;
-	regmap_bulk_write(max98390->regmap, DSM_EQ_BQ1_B0_BYTE0,
-		dsm_param,
-		fw->size - MAX98390_DSM_PAYLOAD_OFFSET);
-	release_firmware(fw);
+	regmap_bulk_write(max98390->regmap, param_start_addr,
+		dsm_param, param_size);
 	regmap_write(max98390->regmap, MAX98390_R23E1_DSP_GLOBAL_EN, 0x01);
 
+err_alloc:
+	release_firmware(fw);
 err:
 	return ret;
 }
diff --git a/sound/soc/codecs/max98390.h b/sound/soc/codecs/max98390.h
index f59cb114d957..5f444e7779b0 100644
--- a/sound/soc/codecs/max98390.h
+++ b/sound/soc/codecs/max98390.h
@@ -650,7 +650,8 @@
 
 /* DSM register offset */
 #define MAX98390_DSM_PAYLOAD_OFFSET 16
-#define MAX98390_DSM_PAYLOAD_OFFSET_2 495
+#define MAX98390_DSM_PARAM_MAX_SIZE 770
+#define MAX98390_DSM_PARAM_MIN_SIZE 670
 
 struct max98390_priv {
 	struct regmap *regmap;
diff --git a/sound/soc/codecs/rl6231.c b/sound/soc/codecs/rl6231.c
index 2586d1cafc0c..8c9daf32bab8 100644
--- a/sound/soc/codecs/rl6231.c
+++ b/sound/soc/codecs/rl6231.c
@@ -80,8 +80,8 @@ int rl6231_calc_dmic_clk(int rate)
 	for (i = 0; i < ARRAY_SIZE(div); i++) {
 		if ((div[i] % 3) == 0)
 			continue;
-		/* find divider that gives DMIC frequency below 3.072MHz */
-		if (3072000 * div[i] >= rate)
+		/* find divider that gives DMIC frequency below 1.536MHz */
+		if (1536000 * div[i] >= rate)
 			return i;
 	}
 
diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c
index 6ba1849a77b0..e2e1d5b03b38 100644
--- a/sound/soc/codecs/rt5645.c
+++ b/sound/soc/codecs/rt5645.c
@@ -3625,6 +3625,12 @@ static const struct rt5645_platform_data asus_t100ha_platform_data = {
 	.inv_jd1_1 = true,
 };
 
+static const struct rt5645_platform_data asus_t101ha_platform_data = {
+	.dmic1_data_pin = RT5645_DMIC_DATA_IN2N,
+	.dmic2_data_pin = RT5645_DMIC2_DISABLE,
+	.jd_mode = 3,
+};
+
 static const struct rt5645_platform_data lenovo_ideapad_miix_310_pdata = {
 	.jd_mode = 3,
 	.in2_diff = true,
@@ -3709,6 +3715,14 @@ static const struct dmi_system_id dmi_platform_data[] = {
 		.driver_data = (void *)&asus_t100ha_platform_data,
 	},
 	{
+		.ident = "ASUS T101HA",
+		.matches = {
+			DMI_MATCH(DMI_SYS_VENDOR, "ASUSTeK COMPUTER INC."),
+			DMI_MATCH(DMI_PRODUCT_NAME, "T101HA"),
+		},
+		.driver_data = (void *)&asus_t101ha_platform_data,
+	},
+	{
 		.ident = "MINIX Z83-4",
 		.matches = {
 			DMI_EXACT_MATCH(DMI_SYS_VENDOR, "MINIX"),
diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c
index cf4feb835743..00be73900888 100644
--- a/sound/soc/fsl/fsl-asoc-card.c
+++ b/sound/soc/fsl/fsl-asoc-card.c
@@ -581,7 +581,7 @@ static int fsl_asoc_card_probe(struct platform_device *pdev)
 
 	if (!fsl_asoc_card_is_ac97(priv) && !codec_dev) {
 		dev_err(&pdev->dev, "failed to find codec device\n");
-		ret = -EINVAL;
+		ret = -EPROBE_DEFER;
 		goto asrc_fail;
 	}
 
diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c
index 30f70bbdf89c..1fdb70b9e478 100644
--- a/sound/soc/intel/boards/bytcr_rt5640.c
+++ b/sound/soc/intel/boards/bytcr_rt5640.c
@@ -754,6 +754,18 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = {
 					BYT_RT5640_JD_NOT_INV |
 					BYT_RT5640_MCLK_EN),
 	},
+	{	/* Toshiba Encore WT10-A */
+		.matches = {
+			DMI_EXACT_MATCH(DMI_SYS_VENDOR, "TOSHIBA"),
+			DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "TOSHIBA WT10-A-103"),
+		},
+		.driver_data = (void *)(BYT_RT5640_DMIC1_MAP |
+					BYT_RT5640_JD_SRC_JD1_IN4P |
+					BYT_RT5640_OVCD_TH_2000UA |
+					BYT_RT5640_OVCD_SF_0P75 |
+					BYT_RT5640_SSP0_AIF2 |
+					BYT_RT5640_MCLK_EN),
+	},
 	{	/* Catch-all for generic Insyde tablets, must be last */
 		.matches = {
 			DMI_MATCH(DMI_SYS_VENDOR, "Insyde"),
diff --git a/sound/soc/intel/boards/glk_rt5682_max98357a.c b/sound/soc/intel/boards/glk_rt5682_max98357a.c
index 48eda1a8aa6c..954ab01f695b 100644
--- a/sound/soc/intel/boards/glk_rt5682_max98357a.c
+++ b/sound/soc/intel/boards/glk_rt5682_max98357a.c
@@ -407,7 +407,7 @@ static struct snd_soc_dai_link geminilake_dais[] = {
 		.name = "Glk Audio Echo Reference cap",
 		.stream_name = "Echoreference Capture",
 		.init = NULL,
-		.capture_only = 1,
+		.dpcm_capture = 1,
 		.nonatomic = 1,
 		.dynamic = 1,
 		SND_SOC_DAILINK_REG(echoref, dummy, platform),
diff --git a/sound/soc/intel/boards/kbl_da7219_max98927.c b/sound/soc/intel/boards/kbl_da7219_max98927.c
index cc9b5eab8b4a..e29c31ffd241 100644
--- a/sound/soc/intel/boards/kbl_da7219_max98927.c
+++ b/sound/soc/intel/boards/kbl_da7219_max98927.c
@@ -692,7 +692,7 @@ static struct snd_soc_dai_link kabylake_dais[] = {
 		.name = "Kbl Audio Echo Reference cap",
 		.stream_name = "Echoreference Capture",
 		.init = NULL,
-		.capture_only = 1,
+		.dpcm_capture = 1,
 		.nonatomic = 1,
 		SND_SOC_DAILINK_REG(echoref, dummy, platform),
 	},
@@ -858,7 +858,7 @@ static struct snd_soc_dai_link kabylake_max98_927_373_dais[] = {
 		.name = "Kbl Audio Echo Reference cap",
 		.stream_name = "Echoreference Capture",
 		.init = NULL,
-		.capture_only = 1,
+		.dpcm_capture = 1,
 		.nonatomic = 1,
 		SND_SOC_DAILINK_REG(echoref, dummy, platform),
 	},
diff --git a/sound/soc/intel/boards/kbl_rt5663_max98927.c b/sound/soc/intel/boards/kbl_rt5663_max98927.c
index 658a9da3a40f..09ba55fc36d5 100644
--- a/sound/soc/intel/boards/kbl_rt5663_max98927.c
+++ b/sound/soc/intel/boards/kbl_rt5663_max98927.c
@@ -672,7 +672,7 @@ static struct snd_soc_dai_link kabylake_dais[] = {
 		.name = "Kbl Audio Echo Reference cap",
 		.stream_name = "Echoreference Capture",
 		.init = NULL,
-		.capture_only = 1,
+		.dpcm_capture = 1,
 		.nonatomic = 1,
 		SND_SOC_DAILINK_REG(echoref, dummy, platform),
 	},
diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c
index 1b1f8d7a4ea3..b34cf6cf1139 100644
--- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c
+++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c
@@ -566,7 +566,7 @@ static struct snd_soc_dai_link kabylake_dais[] = {
 		.name = "Kbl Audio Echo Reference cap",
 		.stream_name = "Echoreference Capture",
 		.init = NULL,
-		.capture_only = 1,
+		.dpcm_capture = 1,
 		.nonatomic = 1,
 		SND_SOC_DAILINK_REG(echoref, dummy, platform),
 	},
diff --git a/sound/soc/meson/axg-fifo.c b/sound/soc/meson/axg-fifo.c
index 2e9b56b29d31..b2e867113226 100644
--- a/sound/soc/meson/axg-fifo.c
+++ b/sound/soc/meson/axg-fifo.c
@@ -249,7 +249,7 @@ int axg_fifo_pcm_open(struct snd_soc_component *component,
 	/* Enable pclk to access registers and clock the fifo ip */
 	ret = clk_prepare_enable(fifo->pclk);
 	if (ret)
-		return ret;
+		goto free_irq;
 
 	/* Setup status2 so it reports the memory pointer */
 	regmap_update_bits(fifo->map, FIFO_CTRL1,
@@ -269,8 +269,14 @@ int axg_fifo_pcm_open(struct snd_soc_component *component,
 	/* Take memory arbitror out of reset */
 	ret = reset_control_deassert(fifo->arb);
 	if (ret)
-		clk_disable_unprepare(fifo->pclk);
+		goto free_clk;
+
+	return 0;
 
+free_clk:
+	clk_disable_unprepare(fifo->pclk);
+free_irq:
+	free_irq(fifo->irq, ss);
 	return ret;
 }
 EXPORT_SYMBOL_GPL(axg_fifo_pcm_open);
diff --git a/sound/soc/meson/meson-card-utils.c b/sound/soc/meson/meson-card-utils.c
index 2ca8c98e204f..5a4a91c88734 100644
--- a/sound/soc/meson/meson-card-utils.c
+++ b/sound/soc/meson/meson-card-utils.c
@@ -49,19 +49,26 @@ int meson_card_reallocate_links(struct snd_soc_card *card,
 	links = krealloc(priv->card.dai_link,
 			 num_links * sizeof(*priv->card.dai_link),
 			 GFP_KERNEL | __GFP_ZERO);
+	if (!links)
+		goto err_links;
+
 	ldata = krealloc(priv->link_data,
 			 num_links * sizeof(*priv->link_data),
 			 GFP_KERNEL | __GFP_ZERO);
-
-	if (!links || !ldata) {
-		dev_err(priv->card.dev, "failed to allocate links\n");
-		return -ENOMEM;
-	}
+	if (!ldata)
+		goto err_ldata;
 
 	priv->card.dai_link = links;
 	priv->link_data = ldata;
 	priv->card.num_links = num_links;
 	return 0;
+
+err_ldata:
+	kfree(links);
+err_links:
+	dev_err(priv->card.dev, "failed to allocate links\n");
+	return -ENOMEM;
+
 }
 EXPORT_SYMBOL_GPL(meson_card_reallocate_links);
 
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index b07eca2c6ccc..7b387202c5db 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1648,9 +1648,25 @@ match:
 			dai_link->platforms->name = component->name;
 
 			/* convert non BE into BE */
-			dai_link->no_pcm = 1;
-			dai_link->dpcm_playback = 1;
-			dai_link->dpcm_capture = 1;
+			if (!dai_link->no_pcm) {
+				dai_link->no_pcm = 1;
+
+				if (dai_link->dpcm_playback)
+					dev_warn(card->dev,
+						 "invalid configuration, dailink %s has flags no_pcm=0 and dpcm_playback=1\n",
+						 dai_link->name);
+				if (dai_link->dpcm_capture)
+					dev_warn(card->dev,
+						 "invalid configuration, dailink %s has flags no_pcm=0 and dpcm_capture=1\n",
+						 dai_link->name);
+
+				/* convert normal link into DPCM one */
+				if (!(dai_link->dpcm_playback ||
+				      dai_link->dpcm_capture)) {
+					dai_link->dpcm_playback = !dai_link->capture_only;
+					dai_link->dpcm_capture = !dai_link->playback_only;
+				}
+			}
 
 			/*
 			 * override any BE fixups
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index 276505fb9d50..2c114b4542ce 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -2789,20 +2789,44 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num)
 	struct snd_pcm *pcm;
 	char new_name[64];
 	int ret = 0, playback = 0, capture = 0;
+	int stream;
 	int i;
 
+	if (rtd->dai_link->dynamic && rtd->num_cpus > 1) {
+		dev_err(rtd->dev,
+			"DPCM doesn't support Multi CPU for Front-Ends yet\n");
+		return -EINVAL;
+	}
+
 	if (rtd->dai_link->dynamic || rtd->dai_link->no_pcm) {
-		cpu_dai = asoc_rtd_to_cpu(rtd, 0);
-		if (rtd->num_cpus > 1) {
-			dev_err(rtd->dev,
-				"DPCM doesn't support Multi CPU yet\n");
-			return -EINVAL;
+		if (rtd->dai_link->dpcm_playback) {
+			stream = SNDRV_PCM_STREAM_PLAYBACK;
+
+			for_each_rtd_cpu_dais(rtd, i, cpu_dai)
+				if (!snd_soc_dai_stream_valid(cpu_dai,
+							      stream)) {
+					dev_err(rtd->card->dev,
+						"CPU DAI %s for rtd %s does not support playback\n",
+						cpu_dai->name,
+						rtd->dai_link->stream_name);
+					return -EINVAL;
+				}
+			playback = 1;
+		}
+		if (rtd->dai_link->dpcm_capture) {
+			stream = SNDRV_PCM_STREAM_CAPTURE;
+
+			for_each_rtd_cpu_dais(rtd, i, cpu_dai)
+				if (!snd_soc_dai_stream_valid(cpu_dai,
+							      stream)) {
+					dev_err(rtd->card->dev,
+						"CPU DAI %s for rtd %s does not support capture\n",
+						cpu_dai->name,
+						rtd->dai_link->stream_name);
+					return -EINVAL;
+				}
+			capture = 1;
 		}
-
-		playback = rtd->dai_link->dpcm_playback &&
-			   snd_soc_dai_stream_valid(cpu_dai, SNDRV_PCM_STREAM_PLAYBACK);
-		capture = rtd->dai_link->dpcm_capture &&
-			  snd_soc_dai_stream_valid(cpu_dai, SNDRV_PCM_STREAM_CAPTURE);
 	} else {
 		/* Adapt stream for codec2codec links */
 		int cpu_capture = rtd->dai_link->params ?
diff --git a/sound/soc/sof/nocodec.c b/sound/soc/sof/nocodec.c
index ce053ba8f2e8..d03b5be31255 100644
--- a/sound/soc/sof/nocodec.c
+++ b/sound/soc/sof/nocodec.c
@@ -52,8 +52,10 @@ static int sof_nocodec_bes_setup(struct device *dev,
 		links[i].platforms->name = dev_name(dev);
 		links[i].codecs->dai_name = "snd-soc-dummy-dai";
 		links[i].codecs->name = "snd-soc-dummy";
-		links[i].dpcm_playback = 1;
-		links[i].dpcm_capture = 1;
+		if (ops->drv[i].playback.channels_min)
+			links[i].dpcm_playback = 1;
+		if (ops->drv[i].capture.channels_min)
+			links[i].dpcm_capture = 1;
 	}
 
 	card->dai_link = links;