Merge branch 'asoc-5.2' into asoc-linus

16 files changed, 91 insertions, 29 deletions

diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c
index 05f4514048e2..3ebc0524f4b2 100644
--- a/sound/soc/codecs/ad193x.c
+++ b/sound/soc/codecs/ad193x.c
@@ -240,10 +240,8 @@ static int ad193x_set_dai_fmt(struct snd_soc_dai *codec_dai,
 	}
 
 	/* For DSP_*, LRCLK's polarity must be inverted */
-	if (fmt & SND_SOC_DAIFMT_DSP_A) {
-		change_bit(ffs(AD193X_DAC_LEFT_HIGH) - 1,
-			   (unsigned long *)&dac_fmt);
-	}
+	if (fmt & SND_SOC_DAIFMT_DSP_A)
+		dac_fmt ^= AD193X_DAC_LEFT_HIGH;
 
 	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
 	case SND_SOC_DAIFMT_CBM_CFM: /* codec clk & frm master */
diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c
index 1f57126708e7..911bb6e2a1ac 100644
--- a/sound/soc/codecs/hdac_hdmi.c
+++ b/sound/soc/codecs/hdac_hdmi.c
@@ -539,6 +539,29 @@ static struct hdac_hdmi_port *hdac_hdmi_get_port_from_cvt(
 }
 
 /*
+ * Go through all converters and ensure connection is set to
+ * the correct pin as set via kcontrols.
+ */
+static void hdac_hdmi_verify_connect_sel_all_pins(struct hdac_device *hdev)
+{
+	struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(hdev);
+	struct hdac_hdmi_port *port;
+	struct hdac_hdmi_cvt *cvt;
+	int cvt_idx = 0;
+
+	list_for_each_entry(cvt, &hdmi->cvt_list, head) {
+		port = hdac_hdmi_get_port_from_cvt(hdev, hdmi, cvt);
+		if (port && port->pin) {
+			snd_hdac_codec_write(hdev, port->pin->nid, 0,
+					     AC_VERB_SET_CONNECT_SEL, cvt_idx);
+			dev_dbg(&hdev->dev, "%s: %s set connect %d -> %d\n",
+				__func__, cvt->name, port->pin->nid, cvt_idx);
+		}
+		++cvt_idx;
+	}
+}
+
+/*
  * This tries to get a valid pin and set the HW constraints based on the
  * ELD. Even if a valid pin is not found return success so that device open
  * doesn't fail.
@@ -798,6 +821,14 @@ static int hdac_hdmi_cvt_output_widget_event(struct snd_soc_dapm_widget *w,
 				AC_VERB_SET_CHANNEL_STREAMID, pcm->stream_tag);
 		snd_hdac_codec_write(hdev, cvt->nid, 0,
 				AC_VERB_SET_STREAM_FORMAT, pcm->format);
+
+		/*
+		 * The connection indices are shared by all converters and
+		 * may interfere with each other. Ensure correct
+		 * routing for all converters at stream start.
+		 */
+		hdac_hdmi_verify_connect_sel_all_pins(hdev);
+
 		break;
 
 	case SND_SOC_DAPM_POST_PMD:
diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c
index e5dd05c94f62..9f5aee7de686 100644
--- a/sound/soc/codecs/nau8825.c
+++ b/sound/soc/codecs/nau8825.c
@@ -1880,6 +1880,10 @@ static void nau8825_init_regs(struct nau8825 *nau8825)
 		NAU8825_JACK_EJECT_DEBOUNCE_MASK,
 		nau8825->jack_eject_debounce << NAU8825_JACK_EJECT_DEBOUNCE_SFT);
 
+	/* Pull up IRQ pin */
+	regmap_update_bits(regmap, NAU8825_REG_INTERRUPT_MASK,
+		NAU8825_IRQ_PIN_PULLUP | NAU8825_IRQ_PIN_PULL_EN,
+		NAU8825_IRQ_PIN_PULLUP | NAU8825_IRQ_PIN_PULL_EN);
 	/* Mask unneeded IRQs: 1 - disable, 0 - enable */
 	regmap_update_bits(regmap, NAU8825_REG_INTERRUPT_MASK, 0x7ff, 0x7ff);
 
diff --git a/sound/soc/codecs/nau8825.h b/sound/soc/codecs/nau8825.h
index 5e60696460de..887bbff03ec6 100644
--- a/sound/soc/codecs/nau8825.h
+++ b/sound/soc/codecs/nau8825.h
@@ -168,6 +168,8 @@
 #define NAU8825_JACK_POLARITY	(1 << 1) /* 0 - active low, 1 - active high */
 
 /* INTERRUPT_MASK (0xf) */
+#define NAU8825_IRQ_PIN_PULLUP (1 << 14)
+#define NAU8825_IRQ_PIN_PULL_EN (1 << 13)
 #define NAU8825_IRQ_OUTPUT_EN (1 << 11)
 #define NAU8825_IRQ_HEADSET_COMPLETE_EN (1 << 10)
 #define NAU8825_IRQ_RMS_EN (1 << 8)
diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c
index ec7e673ba475..70ed28d97d49 100644
--- a/sound/soc/generic/audio-graph-card.c
+++ b/sound/soc/generic/audio-graph-card.c
@@ -435,9 +435,6 @@ static int graph_for_each_link(struct asoc_simple_priv *priv,
 			codec_ep = of_graph_get_remote_endpoint(cpu_ep);
 			codec_port = of_get_parent(codec_ep);
 
-			of_node_put(codec_ep);
-			of_node_put(codec_port);
-
 			/* get convert-xxx property */
 			memset(&adata, 0, sizeof(adata));
 			graph_parse_convert(dev, codec_ep, &adata);
@@ -457,6 +454,9 @@ static int graph_for_each_link(struct asoc_simple_priv *priv,
 			else
 				ret = func_noml(priv, cpu_ep, codec_ep, li);
 
+			of_node_put(codec_ep);
+			of_node_put(codec_port);
+
 			if (ret < 0)
 				return ret;
 
diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c
index 3343dbcd506f..90d262ebeb10 100644
--- a/sound/soc/intel/boards/sof_rt5682.c
+++ b/sound/soc/intel/boards/sof_rt5682.c
@@ -494,7 +494,7 @@ static int sof_audio_probe(struct platform_device *pdev)
 	int dmic_num, hdmi_num;
 	int ret, ssp_amp, ssp_codec;
 
-	ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_ATOMIC);
+	ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL);
 	if (!ctx)
 		return -ENOMEM;
 
diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c
index 67a4c4e13545..91e5a7753eba 100644
--- a/sound/soc/intel/skylake/skl.c
+++ b/sound/soc/intel/skylake/skl.c
@@ -184,6 +184,25 @@ void skl_update_d0i3c(struct device *dev, bool enable)
 			snd_hdac_chip_readb(bus, VS_D0I3C));
 }
 
+/**
+ * skl_dum_set - set DUM bit in EM2 register
+ * @bus: HD-audio core bus
+ *
+ * Addresses incorrect position reporting for capture streams.
+ * Used on device power up.
+ */
+static void skl_dum_set(struct hdac_bus *bus)
+{
+	/* For the DUM bit to be set, CRST needs to be out of reset state */
+	if (!(snd_hdac_chip_readb(bus, GCTL) & AZX_GCTL_RESET)) {
+		skl_enable_miscbdcge(bus->dev, false);
+		snd_hdac_bus_exit_link_reset(bus);
+		skl_enable_miscbdcge(bus->dev, true);
+	}
+
+	snd_hdac_chip_updatel(bus, VS_EM2, AZX_VS_EM2_DUM, AZX_VS_EM2_DUM);
+}
+
 /* called from IRQ */
 static void skl_stream_update(struct hdac_bus *bus, struct hdac_stream *hstr)
 {
@@ -291,6 +310,7 @@ static int _skl_resume(struct hdac_bus *bus)
 	struct skl *skl = bus_to_skl(bus);
 
 	skl_init_pci(skl);
+	skl_dum_set(bus);
 	skl_init_chip(bus, true);
 
 	return skl_resume_dsp(skl);
@@ -948,6 +968,7 @@ static int skl_first_init(struct hdac_bus *bus)
 
 	/* initialize chip */
 	skl_init_pci(skl);
+	skl_dum_set(bus);
 
 	return skl_init_chip(bus, true);
 }
diff --git a/sound/soc/intel/skylake/skl.h b/sound/soc/intel/skylake/skl.h
index e7870ec81a9b..4e571562261a 100644
--- a/sound/soc/intel/skylake/skl.h
+++ b/sound/soc/intel/skylake/skl.h
@@ -37,6 +37,7 @@
 #define DMA_TRANSMITION_START	2
 #define DMA_TRANSMITION_STOP	3
 
+#define AZX_VS_EM2_DUM			BIT(23)
 #define AZX_REG_VS_EM2_L1SEN		BIT(13)
 
 struct skl_dsp_resource {
diff --git a/sound/soc/meson/axg-tdm.h b/sound/soc/meson/axg-tdm.h
index e578b6f40a07..5774ce0916d4 100644
--- a/sound/soc/meson/axg-tdm.h
+++ b/sound/soc/meson/axg-tdm.h
@@ -40,7 +40,7 @@ struct axg_tdm_iface {
 
 static inline bool axg_tdm_lrclk_invert(unsigned int fmt)
 {
-	return (fmt & SND_SOC_DAIFMT_I2S) ^
+	return ((fmt & SND_SOC_DAIFMT_FORMAT_MASK) == SND_SOC_DAIFMT_I2S) ^
 		!!(fmt & (SND_SOC_DAIFMT_IB_IF | SND_SOC_DAIFMT_NB_IF));
 }
 
diff --git a/sound/soc/meson/axg-tdmin.c b/sound/soc/meson/axg-tdmin.c
index a790f925a4ef..cb87f17f3e95 100644
--- a/sound/soc/meson/axg-tdmin.c
+++ b/sound/soc/meson/axg-tdmin.c
@@ -121,7 +121,6 @@ static int axg_tdmin_prepare(struct regmap *map,
 		break;
 
 	case SND_SOC_DAIFMT_LEFT_J:
-	case SND_SOC_DAIFMT_RIGHT_J:
 	case SND_SOC_DAIFMT_DSP_B:
 		break;
 
diff --git a/sound/soc/meson/axg-tdmout.c b/sound/soc/meson/axg-tdmout.c
index 527bfc4487e0..86537fc0ecb5 100644
--- a/sound/soc/meson/axg-tdmout.c
+++ b/sound/soc/meson/axg-tdmout.c
@@ -137,7 +137,6 @@ static int axg_tdmout_prepare(struct regmap *map,
 		break;
 
 	case SND_SOC_DAIFMT_LEFT_J:
-	case SND_SOC_DAIFMT_RIGHT_J:
 	case SND_SOC_DAIFMT_DSP_B:
 		skew += 1;
 		break;
diff --git a/sound/soc/qcom/common.c b/sound/soc/qcom/common.c
index 5661025e8cec..a612d860ad26 100644
--- a/sound/soc/qcom/common.c
+++ b/sound/soc/qcom/common.c
@@ -97,6 +97,7 @@ int qcom_snd_parse_of(struct snd_soc_card *card)
 			goto err;
 		}
 
+		link->nonatomic = 1;
 		link->dpcm_playback = 1;
 		link->dpcm_capture = 1;
 		link->stream_name = link->name;
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 41c0cfaf2db5..6aeba0d66ec5 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -158,9 +158,10 @@ static void soc_init_component_debugfs(struct snd_soc_component *component)
 				component->card->debugfs_card_root);
 	}
 
-	if (!component->debugfs_root) {
+	if (IS_ERR(component->debugfs_root)) {
 		dev_warn(component->dev,
-			"ASoC: Failed to create component debugfs directory\n");
+			"ASoC: Failed to create component debugfs directory: %ld\n",
+			PTR_ERR(component->debugfs_root));
 		return;
 	}
 
@@ -212,18 +213,21 @@ static void soc_init_card_debugfs(struct snd_soc_card *card)
 
 	card->debugfs_card_root = debugfs_create_dir(card->name,
 						     snd_soc_debugfs_root);
-	if (!card->debugfs_card_root) {
+	if (IS_ERR(card->debugfs_card_root)) {
 		dev_warn(card->dev,
-			 "ASoC: Failed to create card debugfs directory\n");
+			 "ASoC: Failed to create card debugfs directory: %ld\n",
+			 PTR_ERR(card->debugfs_card_root));
+		card->debugfs_card_root = NULL;
 		return;
 	}
 
 	card->debugfs_pop_time = debugfs_create_u32("dapm_pop_time", 0644,
 						    card->debugfs_card_root,
 						    &card->pop_time);
-	if (!card->debugfs_pop_time)
+	if (IS_ERR(card->debugfs_pop_time))
 		dev_warn(card->dev,
-			 "ASoC: Failed to create pop time debugfs file\n");
+			 "ASoC: Failed to create pop time debugfs file: %ld\n",
+			 PTR_ERR(card->debugfs_pop_time));
 }
 
 static void soc_cleanup_card_debugfs(struct snd_soc_card *card)
@@ -2837,14 +2841,12 @@ static void snd_soc_unbind_card(struct snd_soc_card *card, bool unregister)
 		snd_soc_dapm_shutdown(card);
 		snd_soc_flush_all_delayed_work(card);
 
-		mutex_lock(&client_mutex);
 		/* remove all components used by DAI links on this card */
 		for_each_comp_order(order) {
 			for_each_card_rtds(card, rtd) {
 				soc_remove_link_components(card, rtd, order);
 			}
 		}
-		mutex_unlock(&client_mutex);
 
 		soc_cleanup_card_resources(card);
 		if (!unregister)
@@ -2863,7 +2865,9 @@ static void snd_soc_unbind_card(struct snd_soc_card *card, bool unregister)
  */
 int snd_soc_unregister_card(struct snd_soc_card *card)
 {
+	mutex_lock(&client_mutex);
 	snd_soc_unbind_card(card, true);
+	mutex_unlock(&client_mutex);
 	dev_dbg(card->dev, "ASoC: Unregistered card '%s'\n", card->name);
 
 	return 0;
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 55f8278077f4..c91df5a9c840 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -2155,23 +2155,25 @@ void snd_soc_dapm_debugfs_init(struct snd_soc_dapm_context *dapm,
 {
 	struct dentry *d;
 
-	if (!parent)
+	if (!parent || IS_ERR(parent))
 		return;
 
 	dapm->debugfs_dapm = debugfs_create_dir("dapm", parent);
 
-	if (!dapm->debugfs_dapm) {
+	if (IS_ERR(dapm->debugfs_dapm)) {
 		dev_warn(dapm->dev,
-		       "ASoC: Failed to create DAPM debugfs directory\n");
+			 "ASoC: Failed to create DAPM debugfs directory %ld\n",
+			 PTR_ERR(dapm->debugfs_dapm));
 		return;
 	}
 
 	d = debugfs_create_file("bias_level", 0444,
 				dapm->debugfs_dapm, dapm,
 				&dapm_bias_fops);
-	if (!d)
+	if (IS_ERR(d))
 		dev_warn(dapm->dev,
-			 "ASoC: Failed to create bias level debugfs file\n");
+			 "ASoC: Failed to create bias level debugfs file: %ld\n",
+			 PTR_ERR(d));
 }
 
 static void dapm_debugfs_add_widget(struct snd_soc_dapm_widget *w)
@@ -2185,10 +2187,10 @@ static void dapm_debugfs_add_widget(struct snd_soc_dapm_widget *w)
 	d = debugfs_create_file(w->name, 0444,
 				dapm->debugfs_dapm, w,
 				&dapm_widget_power_fops);
-	if (!d)
+	if (IS_ERR(d))
 		dev_warn(w->dapm->dev,
-			"ASoC: Failed to create %s debugfs file\n",
-			w->name);
+			 "ASoC: Failed to create %s debugfs file: %ld\n",
+			 w->name, PTR_ERR(d));
 }
 
 static void dapm_debugfs_cleanup(struct snd_soc_dapm_context *dapm)
diff --git a/sound/soc/ti/davinci-mcasp.c b/sound/soc/ti/davinci-mcasp.c
index 5e8e31743a28..29ca88ba3746 100644
--- a/sound/soc/ti/davinci-mcasp.c
+++ b/sound/soc/ti/davinci-mcasp.c
@@ -2234,7 +2234,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
 		ret = edma_pcm_platform_register(&pdev->dev);
 		break;
 	case PCM_SDMA:
-		ret = sdma_pcm_platform_register(&pdev->dev, NULL, NULL);
+		ret = sdma_pcm_platform_register(&pdev->dev, "tx", "rx");
 		break;
 	default:
 		dev_err(&pdev->dev, "No DMA controller found (%d)\n", ret);
diff --git a/sound/soc/ti/omap-mcbsp.c b/sound/soc/ti/omap-mcbsp.c
index 1ab3c7df4f8b..26b503bbdb5f 100644
--- a/sound/soc/ti/omap-mcbsp.c
+++ b/sound/soc/ti/omap-mcbsp.c
@@ -1424,7 +1424,7 @@ static int asoc_mcbsp_probe(struct platform_device *pdev)
 	if (ret)
 		return ret;
 
-	return sdma_pcm_platform_register(&pdev->dev, NULL, NULL);
+	return sdma_pcm_platform_register(&pdev->dev, "tx", "rx");
 }
 
 static int asoc_mcbsp_remove(struct platform_device *pdev)