Merge branch 'asoc-4.18' into asoc-4.19 wmadsp dep

11 files changed, 48 insertions, 19 deletions

diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c
index 38e4a8515709..d00734d31e04 100644
--- a/sound/soc/codecs/hdmi-codec.c
+++ b/sound/soc/codecs/hdmi-codec.c
@@ -291,10 +291,6 @@ static const struct snd_soc_dapm_widget hdmi_widgets[] = {
 	SND_SOC_DAPM_OUTPUT("TX"),
 };
 
-static const struct snd_soc_dapm_route hdmi_routes[] = {
-	{ "TX", NULL, "Playback" },
-};
-
 enum {
 	DAI_ID_I2S = 0,
 	DAI_ID_SPDIF,
@@ -689,9 +685,23 @@ static int hdmi_codec_pcm_new(struct snd_soc_pcm_runtime *rtd,
 	return snd_ctl_add(rtd->card->snd_card, kctl);
 }
 
+static int hdmi_dai_probe(struct snd_soc_dai *dai)
+{
+	struct snd_soc_dapm_context *dapm;
+	struct snd_soc_dapm_route route = {
+		.sink = "TX",
+		.source = dai->driver->playback.stream_name,
+	};
+
+	dapm = snd_soc_component_get_dapm(dai->component);
+
+	return snd_soc_dapm_add_routes(dapm, &route, 1);
+}
+
 static const struct snd_soc_dai_driver hdmi_i2s_dai = {
 	.name = "i2s-hifi",
 	.id = DAI_ID_I2S,
+	.probe = hdmi_dai_probe,
 	.playback = {
 		.stream_name = "I2S Playback",
 		.channels_min = 2,
@@ -707,6 +717,7 @@ static const struct snd_soc_dai_driver hdmi_i2s_dai = {
 static const struct snd_soc_dai_driver hdmi_spdif_dai = {
 	.name = "spdif-hifi",
 	.id = DAI_ID_SPDIF,
+	.probe = hdmi_dai_probe,
 	.playback = {
 		.stream_name = "SPDIF Playback",
 		.channels_min = 2,
@@ -733,8 +744,6 @@ static int hdmi_of_xlate_dai_id(struct snd_soc_component *component,
 static const struct snd_soc_component_driver hdmi_driver = {
 	.dapm_widgets		= hdmi_widgets,
 	.num_dapm_widgets	= ARRAY_SIZE(hdmi_widgets),
-	.dapm_routes		= hdmi_routes,
-	.num_dapm_routes	= ARRAY_SIZE(hdmi_routes),
 	.of_xlate_dai_id	= hdmi_of_xlate_dai_id,
 	.idle_bias_on		= 1,
 	.use_pmdown_time	= 1,
diff --git a/sound/soc/codecs/max98373.c b/sound/soc/codecs/max98373.c
index a92586106932..92b7125ea169 100644
--- a/sound/soc/codecs/max98373.c
+++ b/sound/soc/codecs/max98373.c
@@ -488,6 +488,7 @@ static const DECLARE_TLV_DB_RANGE(max98373_bde_gain_tlv,
 static bool max98373_readable_register(struct device *dev, unsigned int reg)
 {
 	switch (reg) {
+	case MAX98373_R2000_SW_RESET:
 	case MAX98373_R2001_INT_RAW1 ... MAX98373_R200C_INT_EN3:
 	case MAX98373_R2010_IRQ_CTRL:
 	case MAX98373_R2014_THERM_WARN_THRESH
diff --git a/sound/soc/codecs/rt5514.c b/sound/soc/codecs/rt5514.c
index 1570b91bf018..dca82dd6e3bf 100644
--- a/sound/soc/codecs/rt5514.c
+++ b/sound/soc/codecs/rt5514.c
@@ -64,8 +64,8 @@ static const struct reg_sequence rt5514_patch[] = {
 	{RT5514_ANA_CTRL_LDO10,		0x00028604},
 	{RT5514_ANA_CTRL_ADCFED,	0x00000800},
 	{RT5514_ASRC_IN_CTRL1,		0x00000003},
-	{RT5514_DOWNFILTER0_CTRL3,	0x10000362},
-	{RT5514_DOWNFILTER1_CTRL3,	0x10000362},
+	{RT5514_DOWNFILTER0_CTRL3,	0x10000352},
+	{RT5514_DOWNFILTER1_CTRL3,	0x10000352},
 };
 
 static const struct reg_default rt5514_reg[] = {
@@ -92,10 +92,10 @@ static const struct reg_default rt5514_reg[] = {
 	{RT5514_ASRC_IN_CTRL1,		0x00000003},
 	{RT5514_DOWNFILTER0_CTRL1,	0x00020c2f},
 	{RT5514_DOWNFILTER0_CTRL2,	0x00020c2f},
-	{RT5514_DOWNFILTER0_CTRL3,	0x10000362},
+	{RT5514_DOWNFILTER0_CTRL3,	0x10000352},
 	{RT5514_DOWNFILTER1_CTRL1,	0x00020c2f},
 	{RT5514_DOWNFILTER1_CTRL2,	0x00020c2f},
-	{RT5514_DOWNFILTER1_CTRL3,	0x10000362},
+	{RT5514_DOWNFILTER1_CTRL3,	0x10000352},
 	{RT5514_ANA_CTRL_LDO10,		0x00028604},
 	{RT5514_ANA_CTRL_LDO18_16,	0x02000345},
 	{RT5514_ANA_CTRL_ADC12,		0x0000a2a8},
diff --git a/sound/soc/codecs/wm8988.c b/sound/soc/codecs/wm8988.c
index 62200117444b..6e52c6a8bab3 100644
--- a/sound/soc/codecs/wm8988.c
+++ b/sound/soc/codecs/wm8988.c
@@ -522,7 +522,7 @@ static inline int get_coeff(int mclk, int rate)
 /* The set of rates we can generate from the above for each SYSCLK */
 
 static const unsigned int rates_12288[] = {
-	8000, 12000, 16000, 24000, 24000, 32000, 48000, 96000,
+	8000, 12000, 16000, 24000, 32000, 48000, 96000,
 };
 
 static const struct snd_pcm_hw_constraint_list constraints_12288 = {
@@ -540,7 +540,7 @@ static const struct snd_pcm_hw_constraint_list constraints_112896 = {
 };
 
 static const unsigned int rates_12[] = {
-	8000, 11025, 12000, 16000, 22050, 2400, 32000, 41100, 48000,
+	8000, 11025, 12000, 16000, 22050, 24000, 32000, 41100, 48000,
 	48000, 88235, 96000,
 };
 
diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c
index fbd0515c49d7..1c12c78dbcce 100644
--- a/sound/soc/codecs/wm_adsp.c
+++ b/sound/soc/codecs/wm_adsp.c
@@ -2650,7 +2650,10 @@ int wm_adsp2_preloader_get(struct snd_kcontrol *kcontrol,
 			   struct snd_ctl_elem_value *ucontrol)
 {
 	struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol);
-	struct wm_adsp *dsp = snd_soc_component_get_drvdata(component);
+	struct wm_adsp *dsps = snd_soc_component_get_drvdata(component);
+	struct soc_mixer_control *mc =
+		(struct soc_mixer_control *)kcontrol->private_value;
+	struct wm_adsp *dsp = &dsps[mc->shift - 1];
 
 	ucontrol->value.integer.value[0] = dsp->preloaded;
 
@@ -2662,10 +2665,11 @@ int wm_adsp2_preloader_put(struct snd_kcontrol *kcontrol,
 			   struct snd_ctl_elem_value *ucontrol)
 {
 	struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol);
-	struct wm_adsp *dsp = snd_soc_component_get_drvdata(component);
+	struct wm_adsp *dsps = snd_soc_component_get_drvdata(component);
 	struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component);
 	struct soc_mixer_control *mc =
 		(struct soc_mixer_control *)kcontrol->private_value;
+	struct wm_adsp *dsp = &dsps[mc->shift - 1];
 	char preload[32];
 
 	snprintf(preload, ARRAY_SIZE(preload), "DSP%u Preload", mc->shift);
@@ -2679,6 +2683,8 @@ int wm_adsp2_preloader_put(struct snd_kcontrol *kcontrol,
 
 	snd_soc_dapm_sync(dapm);
 
+	flush_work(&dsp->boot_work);
+
 	return 0;
 }
 EXPORT_SYMBOL_GPL(wm_adsp2_preloader_put);
diff --git a/sound/soc/qcom/qdsp6/q6afe-dai.c b/sound/soc/qcom/qdsp6/q6afe-dai.c
index 139d24be3fb9..60ff4a2d3577 100644
--- a/sound/soc/qcom/qdsp6/q6afe-dai.c
+++ b/sound/soc/qcom/qdsp6/q6afe-dai.c
@@ -313,6 +313,9 @@ static void q6afe_dai_shutdown(struct snd_pcm_substream *substream,
 	struct q6afe_dai_data *dai_data = dev_get_drvdata(dai->dev);
 	int rc;
 
+	if (!dai_data->is_port_started[dai->id])
+		return;
+
 	rc = q6afe_port_stop(dai_data->port[dai->id]);
 	if (rc < 0)
 		dev_err(dai->dev, "fail to close AFE port (%d)\n", rc);
@@ -1332,7 +1335,7 @@ static void of_q6afe_parse_dai_data(struct device *dev,
 		int id, i, num_lines;
 
 		ret = of_property_read_u32(node, "reg", &id);
-		if (ret || id > AFE_PORT_MAX) {
+		if (ret || id < 0 || id >= AFE_PORT_MAX) {
 			dev_err(dev, "valid dai id not found:%d\n", ret);
 			continue;
 		}
diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c
index acf96c6549fc..9db9a2944ef2 100644
--- a/sound/soc/qcom/qdsp6/q6asm-dai.c
+++ b/sound/soc/qcom/qdsp6/q6asm-dai.c
@@ -389,7 +389,9 @@ static int q6asm_dai_close(struct snd_pcm_substream *substream)
 	struct q6asm_dai_rtd *prtd = runtime->private_data;
 
 	if (prtd->audio_client) {
-		q6asm_cmd(prtd->audio_client, CMD_CLOSE);
+		if (prtd->state)
+			q6asm_cmd(prtd->audio_client, CMD_CLOSE);
+
 		q6asm_unmap_memory_regions(substream->stream,
 					   prtd->audio_client);
 		q6asm_audio_client_free(prtd->audio_client);
diff --git a/sound/soc/qcom/qdsp6/q6routing.c b/sound/soc/qcom/qdsp6/q6routing.c
index 1d33b00e5b44..dc94c5c53788 100644
--- a/sound/soc/qcom/qdsp6/q6routing.c
+++ b/sound/soc/qcom/qdsp6/q6routing.c
@@ -926,7 +926,7 @@ static int routing_hw_params(struct snd_pcm_substream *substream,
 	else
 		path_type = ADM_PATH_LIVE_REC;
 
-	if (be_id > AFE_MAX_PORTS)
+	if (be_id >= AFE_MAX_PORTS)
 		return -EINVAL;
 
 	session = &data->port_data[be_id];
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index 9833e53754cb..e8b98bfd4cf1 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -1712,6 +1712,14 @@ static void dpcm_runtime_merge_format(struct snd_pcm_substream *substream,
 		int i;
 
 		for (i = 0; i < be->num_codecs; i++) {
+			/*
+			 * Skip CODECs which don't support the current stream
+			 * type. See soc_pcm_init_runtime_hw() for more details
+			 */
+			if (!snd_soc_dai_stream_valid(be->codec_dais[i],
+						      stream))
+				continue;
+
 			codec_dai_drv = be->codec_dais[i]->driver;
 			if (stream == SNDRV_PCM_STREAM_PLAYBACK)
 				codec_stream = &codec_dai_drv->playback;
diff --git a/sound/soc/soc-utils.c b/sound/soc/soc-utils.c
index ea024236c643..e0c93496c0cd 100644
--- a/sound/soc/soc-utils.c
+++ b/sound/soc/soc-utils.c
@@ -375,6 +375,6 @@ int __init snd_soc_util_init(void)
 
 void __exit snd_soc_util_exit(void)
 {
-	platform_device_unregister(soc_dummy_dev);
 	platform_driver_unregister(&soc_dummy_driver);
+	platform_device_unregister(soc_dummy_dev);
 }
diff --git a/sound/soc/tegra/tegra30_i2s.h b/sound/soc/tegra/tegra30_i2s.h
index 774fc6ad2026..2e561e946de2 100644
--- a/sound/soc/tegra/tegra30_i2s.h
+++ b/sound/soc/tegra/tegra30_i2s.h
@@ -173,7 +173,7 @@
 /* Number of slots in frame, minus 1 */
 #define TEGRA30_I2S_SLOT_CTRL_TOTAL_SLOTS_SHIFT		16
 #define TEGRA30_I2S_SLOT_CTRL_TOTAL_SLOTS_MASK_US	7
-#define TEGRA30_I2S_SLOT_CTRL_TOTAL_SLOTS_MASK		(TEGRA30_I2S_SLOT_CTRL_TOTAL_SLOT_MASK_US << TEGRA30_I2S_SLOT_CTRL_TOTAL_SLOT_SHIFT)
+#define TEGRA30_I2S_SLOT_CTRL_TOTAL_SLOTS_MASK		(TEGRA30_I2S_SLOT_CTRL_TOTAL_SLOTS_MASK_US << TEGRA30_I2S_SLOT_CTRL_TOTAL_SLOTS_SHIFT)
 
 /* TDM mode slot enable bitmask */
 #define TEGRA30_I2S_SLOT_CTRL_RX_SLOT_ENABLES_SHIFT	8