[PATCH AUTOSEL 5.6 07/79] ASoC: topology: Add missing memory checks

[PATCH AUTOSEL 5.6 07/79] ASoC: topology: Add missing memory checks

[Sasha Levin]

From: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>

[ Upstream commit abc3caac24501008465fdb55c5e89e16d58d5a3d ]

kstrdup is an allocation function and it can fail, so its return value
should be checked and handled appropriately.

In order to check all cases, we need to modify set_stream_info to return
a value, so check that everything went correctly when doing kstrdup().
Later add proper checks and error handlers.

Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200327204729.397-2-amadeuszx.slawinski@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
---
 sound/soc/soc-topology.c | 62 +++++++++++++++++++++++++++++++---------
 1 file changed, 49 insertions(+), 13 deletions(-)

diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c
index a152409e8746e..7a7c427de95d6 100644
--- a/sound/soc/soc-topology.c
+++ b/sound/soc/soc-topology.c
@@ -1766,10 +1766,13 @@ static int soc_tplg_dapm_complete(struct soc_tplg *tplg)
 	return 0;
 }
 
-static void set_stream_info(struct snd_soc_pcm_stream *stream,
+static int set_stream_info(struct snd_soc_pcm_stream *stream,
 	struct snd_soc_tplg_stream_caps *caps)
 {
 	stream->stream_name = kstrdup(caps->name, GFP_KERNEL);
+	if (!stream->stream_name)
+		return -ENOMEM;
+
 	stream->channels_min = le32_to_cpu(caps->channels_min);
 	stream->channels_max = le32_to_cpu(caps->channels_max);
 	stream->rates = le32_to_cpu(caps->rates);
@@ -1777,6 +1780,8 @@ static void set_stream_info(struct snd_soc_pcm_stream *stream,
 	stream->rate_max = le32_to_cpu(caps->rate_max);
 	stream->formats = le64_to_cpu(caps->formats);
 	stream->sig_bits = le32_to_cpu(caps->sig_bits);
+
+	return 0;
 }
 
 static void set_dai_flags(struct snd_soc_dai_driver *dai_drv,
@@ -1812,20 +1817,29 @@ static int soc_tplg_dai_create(struct soc_tplg *tplg,
 	if (dai_drv == NULL)
 		return -ENOMEM;
 
-	if (strlen(pcm->dai_name))
+	if (strlen(pcm->dai_name)) {
 		dai_drv->name = kstrdup(pcm->dai_name, GFP_KERNEL);
+		if (!dai_drv->name) {
+			ret = -ENOMEM;
+			goto err;
+		}
+	}
 	dai_drv->id = le32_to_cpu(pcm->dai_id);
 
 	if (pcm->playback) {
 		stream = &dai_drv->playback;
 		caps = &pcm->caps[SND_SOC_TPLG_STREAM_PLAYBACK];
-		set_stream_info(stream, caps);
+		ret = set_stream_info(stream, caps);
+		if (ret < 0)
+			goto err;
 	}
 
 	if (pcm->capture) {
 		stream = &dai_drv->capture;
 		caps = &pcm->caps[SND_SOC_TPLG_STREAM_CAPTURE];
-		set_stream_info(stream, caps);
+		ret = set_stream_info(stream, caps);
+		if (ret < 0)
+			goto err;
 	}
 
 	if (pcm->compress)
@@ -1835,11 +1849,7 @@ static int soc_tplg_dai_create(struct soc_tplg *tplg,
 	ret = soc_tplg_dai_load(tplg, dai_drv, pcm, NULL);
 	if (ret < 0) {
 		dev_err(tplg->comp->dev, "ASoC: DAI loading failed\n");
-		kfree(dai_drv->playback.stream_name);
-		kfree(dai_drv->capture.stream_name);
-		kfree(dai_drv->name);
-		kfree(dai_drv);
-		return ret;
+		goto err;
 	}
 
 	dai_drv->dobj.index = tplg->index;
@@ -1860,6 +1870,14 @@ static int soc_tplg_dai_create(struct soc_tplg *tplg,
 		return ret;
 	}
 
+	return 0;
+
+err:
+	kfree(dai_drv->playback.stream_name);
+	kfree(dai_drv->capture.stream_name);
+	kfree(dai_drv->name);
+	kfree(dai_drv);
+
 	return ret;
 }
 
@@ -1916,11 +1934,20 @@ static int soc_tplg_fe_link_create(struct soc_tplg *tplg,
 	if (strlen(pcm->pcm_name)) {
 		link->name = kstrdup(pcm->pcm_name, GFP_KERNEL);
 		link->stream_name = kstrdup(pcm->pcm_name, GFP_KERNEL);
+		if (!link->name || !link->stream_name) {
+			ret = -ENOMEM;
+			goto err;
+		}
 	}
 	link->id = le32_to_cpu(pcm->pcm_id);
 
-	if (strlen(pcm->dai_name))
+	if (strlen(pcm->dai_name)) {
 		link->cpus->dai_name = kstrdup(pcm->dai_name, GFP_KERNEL);
+		if (!link->cpus->dai_name) {
+			ret = -ENOMEM;
+			goto err;
+		}
+	}
 
 	link->codecs->name = "snd-soc-dummy";
 	link->codecs->dai_name = "snd-soc-dummy-dai";
@@ -2436,13 +2463,17 @@ static int soc_tplg_dai_config(struct soc_tplg *tplg,
 	if (d->playback) {
 		stream = &dai_drv->playback;
 		caps = &d->caps[SND_SOC_TPLG_STREAM_PLAYBACK];
-		set_stream_info(stream, caps);
+		ret = set_stream_info(stream, caps);
+		if (ret < 0)
+			goto err;
 	}
 
 	if (d->capture) {
 		stream = &dai_drv->capture;
 		caps = &d->caps[SND_SOC_TPLG_STREAM_CAPTURE];
-		set_stream_info(stream, caps);
+		ret = set_stream_info(stream, caps);
+		if (ret < 0)
+			goto err;
 	}
 
 	if (d->flag_mask)
@@ -2454,10 +2485,15 @@ static int soc_tplg_dai_config(struct soc_tplg *tplg,
 	ret = soc_tplg_dai_load(tplg, dai_drv, NULL, dai);
 	if (ret < 0) {
 		dev_err(tplg->comp->dev, "ASoC: DAI loading failed\n");
-		return ret;
+		goto err;
 	}
 
 	return 0;
+
+err:
+	kfree(dai_drv->playback.stream_name);
+	kfree(dai_drv->capture.stream_name);
+	return ret;
 }
 
 /* load physical DAI elements */
-- 
2.20.1

[PATCH AUTOSEL 5.6 08/79] ASoC: topology: Check return value of soc_tplg_create_tlv

[Sasha Levin]

From: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>

[ Upstream commit 482db55ae87f3749db05810a38b1d618dfd4407c ]

Function soc_tplg_create_tlv can fail, so we should check if it succeded
or not and proceed appropriately.

Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200327204729.397-3-amadeuszx.slawinski@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
---
 sound/soc/soc-topology.c | 16 ++++++++++++++--
 1 file changed, 14 insertions(+), 2 deletions(-)

diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c
index 7a7c427de95d6..5d974a36cdc92 100644
--- a/sound/soc/soc-topology.c
+++ b/sound/soc/soc-topology.c
@@ -894,7 +894,13 @@ static int soc_tplg_dmixer_create(struct soc_tplg *tplg, unsigned int count,
 		}
 
 		/* create any TLV data */
-		soc_tplg_create_tlv(tplg, &kc, &mc->hdr);
+		err = soc_tplg_create_tlv(tplg, &kc, &mc->hdr);
+		if (err < 0) {
+			dev_err(tplg->dev, "ASoC: failed to create TLV %s\n",
+				mc->hdr.name);
+			kfree(sm);
+			continue;
+		}
 
 		/* pass control to driver for optional further init */
 		err = soc_tplg_init_kcontrol(tplg, &kc,
@@ -1355,7 +1361,13 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_dmixer_create(
 		}
 
 		/* create any TLV data */
-		soc_tplg_create_tlv(tplg, &kc[i], &mc->hdr);
+		err = soc_tplg_create_tlv(tplg, &kc[i], &mc->hdr);
+		if (err < 0) {
+			dev_err(tplg->dev, "ASoC: failed to create TLV %s\n",
+				mc->hdr.name);
+			kfree(sm);
+			continue;
+		}
 
 		/* pass control to driver for optional further init */
 		err = soc_tplg_init_kcontrol(tplg, &kc[i],
-- 
2.20.1

[PATCH AUTOSEL 5.6 09/79] ASoC: topology: Check return value of soc_tplg_*_create

[Sasha Levin]

From: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>

[ Upstream commit 2ae548f30d7f6973388fc3769bb3c2f6fd13652b ]

Functions soc_tplg_denum_create, soc_tplg_dmixer_create,
soc_tplg_dbytes_create can fail, so their return values should be
checked and error should be propagated.

Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200327204729.397-4-amadeuszx.slawinski@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
---
 sound/soc/soc-topology.c | 18 ++++++++++++------
 1 file changed, 12 insertions(+), 6 deletions(-)

diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c
index 5d974a36cdc92..c86c3ea533f68 100644
--- a/sound/soc/soc-topology.c
+++ b/sound/soc/soc-topology.c
@@ -1124,6 +1124,7 @@ static int soc_tplg_kcontrol_elems_load(struct soc_tplg *tplg,
 	struct snd_soc_tplg_hdr *hdr)
 {
 	struct snd_soc_tplg_ctl_hdr *control_hdr;
+	int ret;
 	int i;
 
 	if (tplg->pass != SOC_TPLG_PASS_MIXER) {
@@ -1152,25 +1153,30 @@ static int soc_tplg_kcontrol_elems_load(struct soc_tplg *tplg,
 		case SND_SOC_TPLG_CTL_RANGE:
 		case SND_SOC_TPLG_DAPM_CTL_VOLSW:
 		case SND_SOC_TPLG_DAPM_CTL_PIN:
-			soc_tplg_dmixer_create(tplg, 1,
-					       le32_to_cpu(hdr->payload_size));
+			ret = soc_tplg_dmixer_create(tplg, 1,
+					le32_to_cpu(hdr->payload_size));
 			break;
 		case SND_SOC_TPLG_CTL_ENUM:
 		case SND_SOC_TPLG_CTL_ENUM_VALUE:
 		case SND_SOC_TPLG_DAPM_CTL_ENUM_DOUBLE:
 		case SND_SOC_TPLG_DAPM_CTL_ENUM_VIRT:
 		case SND_SOC_TPLG_DAPM_CTL_ENUM_VALUE:
-			soc_tplg_denum_create(tplg, 1,
-					      le32_to_cpu(hdr->payload_size));
+			ret = soc_tplg_denum_create(tplg, 1,
+					le32_to_cpu(hdr->payload_size));
 			break;
 		case SND_SOC_TPLG_CTL_BYTES:
-			soc_tplg_dbytes_create(tplg, 1,
-					       le32_to_cpu(hdr->payload_size));
+			ret = soc_tplg_dbytes_create(tplg, 1,
+					le32_to_cpu(hdr->payload_size));
 			break;
 		default:
 			soc_bind_err(tplg, control_hdr, i);
 			return -EINVAL;
 		}
+		if (ret < 0) {
+			dev_err(tplg->dev, "ASoC: invalid control\n");
+			return ret;
+		}
+
 	}
 
 	return 0;
-- 
2.20.1

[PATCH AUTOSEL 5.6 10/79] ASoC: topology: Check soc_tplg_add_route return value

[Sasha Levin]

From: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>

[ Upstream commit 6856e887eae3efc0fe56899cb3f969fe063171c5 ]

Function soc_tplg_add_route can propagate error code from callback, we
should check its return value and handle fail in correct way.

Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200327204729.397-5-amadeuszx.slawinski@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
---
 sound/soc/soc-topology.c | 4 +++-
 1 file changed, 3 insertions(+), 1 deletion(-)

diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c
index c86c3ea533f68..aa7714f2b78fd 100644
--- a/sound/soc/soc-topology.c
+++ b/sound/soc/soc-topology.c
@@ -1284,7 +1284,9 @@ static int soc_tplg_dapm_graph_elems_load(struct soc_tplg *tplg,
 		routes[i]->dobj.index = tplg->index;
 		list_add(&routes[i]->dobj.list, &tplg->comp->dobj_list);
 
-		soc_tplg_add_route(tplg, routes[i]);
+		ret = soc_tplg_add_route(tplg, routes[i]);
+		if (ret < 0)
+			break;
 
 		/* add route, but keep going if some fail */
 		snd_soc_dapm_add_routes(dapm, routes[i], 1);
-- 
2.20.1

[PATCH AUTOSEL 5.6 11/79] ASoC: topology: Check return value of pcm_new_ver

[Sasha Levin]

From: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>

[ Upstream commit b3677fc3d68dd942c92de52f0bd9dd8b472a40e6 ]

Function pcm_new_ver can fail, so we should check it's return value and
handle possible error.

Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200327204729.397-6-amadeuszx.slawinski@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
---
 sound/soc/soc-topology.c | 4 +++-
 1 file changed, 3 insertions(+), 1 deletion(-)

diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c
index aa7714f2b78fd..ca0ac5372b293 100644
--- a/sound/soc/soc-topology.c
+++ b/sound/soc/soc-topology.c
@@ -2135,7 +2135,9 @@ static int soc_tplg_pcm_elems_load(struct soc_tplg *tplg,
 			_pcm = pcm;
 		} else {
 			abi_match = false;
-			pcm_new_ver(tplg, pcm, &_pcm);
+			ret = pcm_new_ver(tplg, pcm, &_pcm);
+			if (ret < 0)
+				return ret;
 		}
 
 		/* create the FE DAIs and DAI links */
-- 
2.20.1

[PATCH AUTOSEL 5.6 12/79] ASoC: topology: Check return value of soc_tplg_dai_config

[Sasha Levin]

From: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>

[ Upstream commit dd8e871d4e560eeb8d22af82dde91457ad835a63 ]

Function soc_tplg_dai_config can fail, check for and handle possible
failure.

Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200327204729.397-7-amadeuszx.slawinski@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
---
 sound/soc/soc-topology.c | 9 +++++++--
 1 file changed, 7 insertions(+), 2 deletions(-)

diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c
index ca0ac5372b293..e6a17ab433eea 100644
--- a/sound/soc/soc-topology.c
+++ b/sound/soc/soc-topology.c
@@ -2524,7 +2524,7 @@ static int soc_tplg_dai_elems_load(struct soc_tplg *tplg,
 {
 	struct snd_soc_tplg_dai *dai;
 	int count;
-	int i;
+	int i, ret;
 
 	count = le32_to_cpu(hdr->count);
 
@@ -2539,7 +2539,12 @@ static int soc_tplg_dai_elems_load(struct soc_tplg *tplg,
 			return -EINVAL;
 		}
 
-		soc_tplg_dai_config(tplg, dai);
+		ret = soc_tplg_dai_config(tplg, dai);
+		if (ret < 0) {
+			dev_err(tplg->dev, "ASoC: failed to configure DAI\n");
+			return ret;
+		}
+
 		tplg->pos += (sizeof(*dai) + le32_to_cpu(dai->priv.size));
 	}
 
-- 
2.20.1

[PATCH AUTOSEL 5.6 17/79] ASoC: samsung: s3c24xx-i2s: Fix build after removal of DAI suspend/resume

[Sasha Levin]

From: Krzysztof Kozlowski <krzk@kernel.org>

[ Upstream commit ec21bdc6dd16d74b3674ef1fd12ae8e4e7418603 ]

Commit 450312b640f9 ("ASoC: soc-core: remove DAI suspend/resume")
removed the DAI side suspend/resume hooks and switched entirely to
component suspend/resume.  However the Samsung SoC s3c-i2s-v2 driver was
not updated.

Move the suspend/resume hooks from s3c-i2s-v2.c to s3c2412-i2s.c while
changing dai to component which allows to keep the struct
snd_soc_component_driver const.

This fixes build errors:

    sound/soc/samsung/s3c-i2s-v2.c: In function ‘s3c_i2sv2_register_component’:
    sound/soc/samsung/s3c-i2s-v2.c:730:9: error: ‘struct snd_soc_dai_driver’ has no member named ‘suspend’
      dai_drv->suspend = s3c2412_i2s_suspend;

Reported-by: Arnd Bergmann <arnd@arndb.de>
Fixes: 450312b640f9 ("ASoC: soc-core: remove DAI suspend/resume")
Signed-off-by: Krzysztof Kozlowski <krzk@kernel.org>

Link: https://lore.kernel.org/r/20200413124548.28197-1-krzk@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
---
 sound/soc/samsung/s3c-i2s-v2.c  | 57 ---------------------------------
 sound/soc/samsung/s3c2412-i2s.c | 56 ++++++++++++++++++++++++++++++++
 2 files changed, 56 insertions(+), 57 deletions(-)

diff --git a/sound/soc/samsung/s3c-i2s-v2.c b/sound/soc/samsung/s3c-i2s-v2.c
index 593be1b668d64..b3e12d6a78a19 100644
--- a/sound/soc/samsung/s3c-i2s-v2.c
+++ b/sound/soc/samsung/s3c-i2s-v2.c
@@ -656,60 +656,6 @@ void s3c_i2sv2_cleanup(struct snd_soc_dai *dai,
 }
 EXPORT_SYMBOL_GPL(s3c_i2sv2_cleanup);
 
-#ifdef CONFIG_PM
-static int s3c2412_i2s_suspend(struct snd_soc_dai *dai)
-{
-	struct s3c_i2sv2_info *i2s = to_info(dai);
-	u32 iismod;
-
-	if (dai->active) {
-		i2s->suspend_iismod = readl(i2s->regs + S3C2412_IISMOD);
-		i2s->suspend_iiscon = readl(i2s->regs + S3C2412_IISCON);
-		i2s->suspend_iispsr = readl(i2s->regs + S3C2412_IISPSR);
-
-		/* some basic suspend checks */
-
-		iismod = readl(i2s->regs + S3C2412_IISMOD);
-
-		if (iismod & S3C2412_IISCON_RXDMA_ACTIVE)
-			pr_warn("%s: RXDMA active?\n", __func__);
-
-		if (iismod & S3C2412_IISCON_TXDMA_ACTIVE)
-			pr_warn("%s: TXDMA active?\n", __func__);
-
-		if (iismod & S3C2412_IISCON_IIS_ACTIVE)
-			pr_warn("%s: IIS active\n", __func__);
-	}
-
-	return 0;
-}
-
-static int s3c2412_i2s_resume(struct snd_soc_dai *dai)
-{
-	struct s3c_i2sv2_info *i2s = to_info(dai);
-
-	pr_info("dai_active %d, IISMOD %08x, IISCON %08x\n",
-		dai->active, i2s->suspend_iismod, i2s->suspend_iiscon);
-
-	if (dai->active) {
-		writel(i2s->suspend_iiscon, i2s->regs + S3C2412_IISCON);
-		writel(i2s->suspend_iismod, i2s->regs + S3C2412_IISMOD);
-		writel(i2s->suspend_iispsr, i2s->regs + S3C2412_IISPSR);
-
-		writel(S3C2412_IISFIC_RXFLUSH | S3C2412_IISFIC_TXFLUSH,
-		       i2s->regs + S3C2412_IISFIC);
-
-		ndelay(250);
-		writel(0x0, i2s->regs + S3C2412_IISFIC);
-	}
-
-	return 0;
-}
-#else
-#define s3c2412_i2s_suspend NULL
-#define s3c2412_i2s_resume  NULL
-#endif
-
 int s3c_i2sv2_register_component(struct device *dev, int id,
 			   const struct snd_soc_component_driver *cmp_drv,
 			   struct snd_soc_dai_driver *dai_drv)
@@ -727,9 +673,6 @@ int s3c_i2sv2_register_component(struct device *dev, int id,
 	if (!ops->delay)
 		ops->delay = s3c2412_i2s_delay;
 
-	dai_drv->suspend = s3c2412_i2s_suspend;
-	dai_drv->resume = s3c2412_i2s_resume;
-
 	return devm_snd_soc_register_component(dev, cmp_drv, dai_drv, 1);
 }
 EXPORT_SYMBOL_GPL(s3c_i2sv2_register_component);
diff --git a/sound/soc/samsung/s3c2412-i2s.c b/sound/soc/samsung/s3c2412-i2s.c
index 787a3f6e9f242..b35d828c1cfe9 100644
--- a/sound/soc/samsung/s3c2412-i2s.c
+++ b/sound/soc/samsung/s3c2412-i2s.c
@@ -117,6 +117,60 @@ static int s3c2412_i2s_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
+#ifdef CONFIG_PM
+static int s3c2412_i2s_suspend(struct snd_soc_component *component)
+{
+	struct s3c_i2sv2_info *i2s = snd_soc_component_get_drvdata(component);
+	u32 iismod;
+
+	if (component->active) {
+		i2s->suspend_iismod = readl(i2s->regs + S3C2412_IISMOD);
+		i2s->suspend_iiscon = readl(i2s->regs + S3C2412_IISCON);
+		i2s->suspend_iispsr = readl(i2s->regs + S3C2412_IISPSR);
+
+		/* some basic suspend checks */
+
+		iismod = readl(i2s->regs + S3C2412_IISMOD);
+
+		if (iismod & S3C2412_IISCON_RXDMA_ACTIVE)
+			pr_warn("%s: RXDMA active?\n", __func__);
+
+		if (iismod & S3C2412_IISCON_TXDMA_ACTIVE)
+			pr_warn("%s: TXDMA active?\n", __func__);
+
+		if (iismod & S3C2412_IISCON_IIS_ACTIVE)
+			pr_warn("%s: IIS active\n", __func__);
+	}
+
+	return 0;
+}
+
+static int s3c2412_i2s_resume(struct snd_soc_component *component)
+{
+	struct s3c_i2sv2_info *i2s = snd_soc_component_get_drvdata(component);
+
+	pr_info("component_active %d, IISMOD %08x, IISCON %08x\n",
+		component->active, i2s->suspend_iismod, i2s->suspend_iiscon);
+
+	if (component->active) {
+		writel(i2s->suspend_iiscon, i2s->regs + S3C2412_IISCON);
+		writel(i2s->suspend_iismod, i2s->regs + S3C2412_IISMOD);
+		writel(i2s->suspend_iispsr, i2s->regs + S3C2412_IISPSR);
+
+		writel(S3C2412_IISFIC_RXFLUSH | S3C2412_IISFIC_TXFLUSH,
+		       i2s->regs + S3C2412_IISFIC);
+
+		ndelay(250);
+		writel(0x0, i2s->regs + S3C2412_IISFIC);
+	}
+
+	return 0;
+}
+#else
+#define s3c2412_i2s_suspend NULL
+#define s3c2412_i2s_resume  NULL
+#endif
+
 #define S3C2412_I2S_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 | \
@@ -146,6 +200,8 @@ static struct snd_soc_dai_driver s3c2412_i2s_dai = {
 
 static const struct snd_soc_component_driver s3c2412_i2s_component = {
 	.name		= "s3c2412-i2s",
+	.suspend	= s3c2412_i2s_suspend,
+	.resume		= s3c2412_i2s_resume,
 };
 
 static int s3c2412_iis_dev_probe(struct platform_device *pdev)
-- 
2.20.1

[PATCH AUTOSEL 5.6 24/79] ASoC: sgtl5000: Fix VAG power-on handling

[Sasha Levin]

From: Sebastian Reichel <sebastian.reichel@collabora.com>

[ Upstream commit aa7812737f2877e192d57626cbe8825cc7cf6de9 ]

As mentioned slightly out of patch context in the code, there
is no reset routine for the chip. On boards where the chip is
supplied by a fixed regulator, it might not even be resetted
during (e.g. watchdog) reboot and can be in any state.

If the device is probed with VAG enabled, the driver's probe
routine will generate a loud pop sound when ANA_POWER is
being programmed. Avoid this by properly disabling just the
VAG bit and waiting the required power down time.

Signed-off-by: Sebastian Reichel <sebastian.reichel@collabora.com>
Reviewed-by: Fabio Estevam <festivem@gmail.com>
Link: https://lore.kernel.org/r/20200414181140.145825-1-sebastian.reichel@collabora.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
---
 sound/soc/codecs/sgtl5000.c | 34 ++++++++++++++++++++++++++++++++++
 sound/soc/codecs/sgtl5000.h |  1 +
 2 files changed, 35 insertions(+)

diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c
index d5130193b4a2f..e8a8bf7b4ffed 100644
--- a/sound/soc/codecs/sgtl5000.c
+++ b/sound/soc/codecs/sgtl5000.c
@@ -1653,6 +1653,40 @@ static int sgtl5000_i2c_probe(struct i2c_client *client,
 		dev_err(&client->dev,
 			"Error %d initializing CHIP_CLK_CTRL\n", ret);
 
+	/* Mute everything to avoid pop from the following power-up */
+	ret = regmap_write(sgtl5000->regmap, SGTL5000_CHIP_ANA_CTRL,
+			   SGTL5000_CHIP_ANA_CTRL_DEFAULT);
+	if (ret) {
+		dev_err(&client->dev,
+			"Error %d muting outputs via CHIP_ANA_CTRL\n", ret);
+		goto disable_clk;
+	}
+
+	/*
+	 * If VAG is powered-on (e.g. from previous boot), it would be disabled
+	 * by the write to ANA_POWER in later steps of the probe code. This
+	 * may create a loud pop even with all outputs muted. The proper way
+	 * to circumvent this is disabling the bit first and waiting the proper
+	 * cool-down time.
+	 */
+	ret = regmap_read(sgtl5000->regmap, SGTL5000_CHIP_ANA_POWER, &value);
+	if (ret) {
+		dev_err(&client->dev, "Failed to read ANA_POWER: %d\n", ret);
+		goto disable_clk;
+	}
+	if (value & SGTL5000_VAG_POWERUP) {
+		ret = regmap_update_bits(sgtl5000->regmap,
+					 SGTL5000_CHIP_ANA_POWER,
+					 SGTL5000_VAG_POWERUP,
+					 0);
+		if (ret) {
+			dev_err(&client->dev, "Error %d disabling VAG\n", ret);
+			goto disable_clk;
+		}
+
+		msleep(SGTL5000_VAG_POWERDOWN_DELAY);
+	}
+
 	/* Follow section 2.2.1.1 of AN3663 */
 	ana_pwr = SGTL5000_ANA_POWER_DEFAULT;
 	if (sgtl5000->num_supplies <= VDDD) {
diff --git a/sound/soc/codecs/sgtl5000.h b/sound/soc/codecs/sgtl5000.h
index a4bf4bca95bf7..56ec5863f2507 100644
--- a/sound/soc/codecs/sgtl5000.h
+++ b/sound/soc/codecs/sgtl5000.h
@@ -233,6 +233,7 @@
 /*
  * SGTL5000_CHIP_ANA_CTRL
  */
+#define SGTL5000_CHIP_ANA_CTRL_DEFAULT		0x0133
 #define SGTL5000_LINE_OUT_MUTE			0x0100
 #define SGTL5000_HP_SEL_MASK			0x0040
 #define SGTL5000_HP_SEL_SHIFT			6
-- 
2.20.1

[PATCH AUTOSEL 5.6 25/79] ASoC: q6dsp6: q6afe-dai: add missing channels to MI2S DAIs

[Sasha Levin]

From: Stephan Gerhold <stephan@gerhold.net>

[ Upstream commit 0c824ec094b5cda766c80d88c2036e28c24a4cb1 ]

For some reason, the MI2S DAIs do not have channels_min/max defined.
This means that snd_soc_dai_stream_valid() returns false,
i.e. the DAIs have neither valid playback nor capture stream.

It's quite surprising that this ever worked correctly,
but in 5.7-rc1 this is now failing badly: :)

Commit 0e9cf4c452ad ("ASoC: pcm: check if cpu-dai supports a given stream")
introduced a check for snd_soc_dai_stream_valid() before calling
hw_params(), which means that the q6i2s_hw_params() function
was never called, eventually resulting in:

    qcom-q6afe aprsvc:q6afe:4:4: no line is assigned

... even though "qcom,sd-lines" is set in the device tree.

Commit 9b5db059366a ("ASoC: soc-pcm: dpcm: Only allow playback/capture if supported")
now even avoids creating PCM devices if the stream is not supported,
which means that it is failing even earlier with e.g.:

    Primary MI2S: ASoC: no backend playback stream

Avoid all that trouble by adding channels_min/max for the MI2S DAIs.

Fixes: 24c4cbcfac09 ("ASoC: qdsp6: q6afe: Add q6afe dai driver")
Signed-off-by: Stephan Gerhold <stephan@gerhold.net>
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Cc: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20200415150050.616392-1-stephan@gerhold.net
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
---
 sound/soc/qcom/qdsp6/q6afe-dai.c | 16 ++++++++++++++++
 1 file changed, 16 insertions(+)

diff --git a/sound/soc/qcom/qdsp6/q6afe-dai.c b/sound/soc/qcom/qdsp6/q6afe-dai.c
index c1a7624eaf175..2a5302f1db98a 100644
--- a/sound/soc/qcom/qdsp6/q6afe-dai.c
+++ b/sound/soc/qcom/qdsp6/q6afe-dai.c
@@ -902,6 +902,8 @@ static struct snd_soc_dai_driver q6afe_dais[] = {
 				 SNDRV_PCM_RATE_16000,
 			.formats = SNDRV_PCM_FMTBIT_S16_LE |
 				   SNDRV_PCM_FMTBIT_S24_LE,
+			.channels_min = 1,
+			.channels_max = 8,
 			.rate_min =     8000,
 			.rate_max =     48000,
 		},
@@ -917,6 +919,8 @@ static struct snd_soc_dai_driver q6afe_dais[] = {
 				 SNDRV_PCM_RATE_16000,
 			.formats = SNDRV_PCM_FMTBIT_S16_LE |
 				   SNDRV_PCM_FMTBIT_S24_LE,
+			.channels_min = 1,
+			.channels_max = 8,
 			.rate_min =     8000,
 			.rate_max =     48000,
 		},
@@ -931,6 +935,8 @@ static struct snd_soc_dai_driver q6afe_dais[] = {
 			.rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 |
 				 SNDRV_PCM_RATE_16000,
 			.formats = SNDRV_PCM_FMTBIT_S16_LE,
+			.channels_min = 1,
+			.channels_max = 8,
 			.rate_min =     8000,
 			.rate_max =     48000,
 		},
@@ -946,6 +952,8 @@ static struct snd_soc_dai_driver q6afe_dais[] = {
 				 SNDRV_PCM_RATE_16000,
 			.formats = SNDRV_PCM_FMTBIT_S16_LE |
 				   SNDRV_PCM_FMTBIT_S24_LE,
+			.channels_min = 1,
+			.channels_max = 8,
 			.rate_min =     8000,
 			.rate_max =     48000,
 		},
@@ -960,6 +968,8 @@ static struct snd_soc_dai_driver q6afe_dais[] = {
 			.rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 |
 				 SNDRV_PCM_RATE_16000,
 			.formats = SNDRV_PCM_FMTBIT_S16_LE,
+			.channels_min = 1,
+			.channels_max = 8,
 			.rate_min =     8000,
 			.rate_max =     48000,
 		},
@@ -975,6 +985,8 @@ static struct snd_soc_dai_driver q6afe_dais[] = {
 				 SNDRV_PCM_RATE_16000,
 			.formats = SNDRV_PCM_FMTBIT_S16_LE |
 				   SNDRV_PCM_FMTBIT_S24_LE,
+			.channels_min = 1,
+			.channels_max = 8,
 			.rate_min =     8000,
 			.rate_max =     48000,
 		},
@@ -989,6 +1001,8 @@ static struct snd_soc_dai_driver q6afe_dais[] = {
 			.rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 |
 				 SNDRV_PCM_RATE_16000,
 			.formats = SNDRV_PCM_FMTBIT_S16_LE,
+			.channels_min = 1,
+			.channels_max = 8,
 			.rate_min =     8000,
 			.rate_max =     48000,
 		},
@@ -1004,6 +1018,8 @@ static struct snd_soc_dai_driver q6afe_dais[] = {
 				 SNDRV_PCM_RATE_16000,
 			.formats = SNDRV_PCM_FMTBIT_S16_LE |
 				   SNDRV_PCM_FMTBIT_S24_LE,
+			.channels_min = 1,
+			.channels_max = 8,
 			.rate_min =     8000,
 			.rate_max =     48000,
 		},
-- 
2.20.1

[PATCH AUTOSEL 5.6 26/79] ASoC: topology: Fix endianness issue

[Sasha Levin]

From: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>

[ Upstream commit 26d87881590fd55ccdd8f829498d7b3033f81990 ]

As done in already existing cases, we should use le32_to_cpu macro while
accessing hdr->magic. Found with sparse.

Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Link: https://lore.kernel.org/r/20200415162435.31859-2-amadeuszx.slawinski@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
---
 sound/soc/soc-topology.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c
index e6a17ab433eea..009d65a6fb43a 100644
--- a/sound/soc/soc-topology.c
+++ b/sound/soc/soc-topology.c
@@ -2652,7 +2652,7 @@ static int soc_valid_header(struct soc_tplg *tplg,
 	}
 
 	/* big endian firmware objects not supported atm */
-	if (hdr->magic == SOC_TPLG_MAGIC_BIG_ENDIAN) {
+	if (le32_to_cpu(hdr->magic) == SOC_TPLG_MAGIC_BIG_ENDIAN) {
 		dev_err(tplg->dev,
 			"ASoC: pass %d big endian not supported header got %x at offset 0x%lx size 0x%zx.\n",
 			tplg->pass, hdr->magic,
-- 
2.20.1

[PATCH AUTOSEL 5.6 29/79] ASoC: rsnd: Fix parent SSI start/stop in multi-SSI mode

[Sasha Levin]

From: Matthias Blankertz <matthias.blankertz@cetitec.com>

[ Upstream commit a09fb3f28a60ba3e928a1fa94b0456780800299d ]

The parent SSI of a multi-SSI setup must be fully setup, started and
stopped since it is also part of the playback/capture setup. So only
skip the SSI (as per commit 203cdf51f288 ("ASoC: rsnd: SSI parent cares
SWSP bit") and commit 597b046f0d99 ("ASoC: rsnd: control SSICR::EN
correctly")) if the SSI is parent outside of a multi-SSI setup.

Signed-off-by: Matthias Blankertz <matthias.blankertz@cetitec.com>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/20200415141017.384017-2-matthias.blankertz@cetitec.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
---
 sound/soc/sh/rcar/ssi.c | 8 ++++----
 1 file changed, 4 insertions(+), 4 deletions(-)

diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c
index fc5d089868dfc..d51fb3a394486 100644
--- a/sound/soc/sh/rcar/ssi.c
+++ b/sound/soc/sh/rcar/ssi.c
@@ -407,7 +407,7 @@ static void rsnd_ssi_config_init(struct rsnd_mod *mod,
 	 * We shouldn't exchange SWSP after running.
 	 * This means, parent needs to care it.
 	 */
-	if (rsnd_ssi_is_parent(mod, io))
+	if (rsnd_ssi_is_parent(mod, io) && !rsnd_ssi_multi_slaves(io))
 		goto init_end;
 
 	if (rsnd_io_is_play(io))
@@ -559,7 +559,7 @@ static int rsnd_ssi_start(struct rsnd_mod *mod,
 	 * EN is for data output.
 	 * SSI parent EN is not needed.
 	 */
-	if (rsnd_ssi_is_parent(mod, io))
+	if (rsnd_ssi_is_parent(mod, io) && !rsnd_ssi_multi_slaves(io))
 		return 0;
 
 	ssi->cr_en = EN;
@@ -582,7 +582,7 @@ static int rsnd_ssi_stop(struct rsnd_mod *mod,
 	if (!rsnd_ssi_is_run_mods(mod, io))
 		return 0;
 
-	if (rsnd_ssi_is_parent(mod, io))
+	if (rsnd_ssi_is_parent(mod, io) && !rsnd_ssi_multi_slaves(io))
 		return 0;
 
 	cr  =	ssi->cr_own	|
@@ -620,7 +620,7 @@ static int rsnd_ssi_irq(struct rsnd_mod *mod,
 	if (rsnd_is_gen1(priv))
 		return 0;
 
-	if (rsnd_ssi_is_parent(mod, io))
+	if (rsnd_ssi_is_parent(mod, io) && !rsnd_ssi_multi_slaves(io))
 		return 0;
 
 	if (!rsnd_ssi_is_run_mods(mod, io))
-- 
2.20.1

[PATCH AUTOSEL 5.6 30/79] ASoC: rsnd: Fix HDMI channel mapping for multi-SSI mode

[Sasha Levin]

From: Matthias Blankertz <matthias.blankertz@cetitec.com>

[ Upstream commit b94e164759b82d0c1c80d4b1c8f12c9bee83f11d ]

The HDMI?_SEL register maps up to four stereo SSI data lanes onto the
sdata[0..3] inputs of the HDMI output block. The upper half of the
register contains four blocks of 4 bits, with the most significant
controlling the sdata3 line and the least significant the sdata0 line.

The shift calculation has an off-by-one error, causing the parent SSI to
be mapped to sdata3, the first multi-SSI child to sdata0 and so forth.
As the parent SSI transmits the stereo L/R channels, and the HDMI core
expects it on the sdata0 line, this causes no audio to be output when
playing stereo audio on a multichannel capable HDMI out, and
multichannel audio has permutated channels.

Fix the shift calculation to map the parent SSI to sdata0, the first
child to sdata1 etc.

Signed-off-by: Matthias Blankertz <matthias.blankertz@cetitec.com>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/20200415141017.384017-3-matthias.blankertz@cetitec.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
---
 sound/soc/sh/rcar/ssiu.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/sound/soc/sh/rcar/ssiu.c b/sound/soc/sh/rcar/ssiu.c
index f35d882118874..9c7c3e7539c93 100644
--- a/sound/soc/sh/rcar/ssiu.c
+++ b/sound/soc/sh/rcar/ssiu.c
@@ -221,7 +221,7 @@ static int rsnd_ssiu_init_gen2(struct rsnd_mod *mod,
 			i;
 
 		for_each_rsnd_mod_array(i, pos, io, rsnd_ssi_array) {
-			shift	= (i * 4) + 16;
+			shift	= (i * 4) + 20;
 			val	= (val & ~(0xF << shift)) |
 				rsnd_mod_id(pos) << shift;
 		}
-- 
2.20.1

[PATCH AUTOSEL 5.6 31/79] ASoC: codecs: hdac_hdmi: Fix incorrect use of list_for_each_entry

[Sasha Levin]

From: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>

[ Upstream commit 326b509238171d37402dbe308e154cc234ed1960 ]

If we don't find any pcm, pcm will point at address at an offset from
the the list head and not a meaningful structure. Fix this by returning
correct pcm if found and NULL if not. Found with coccinelle.

Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Link: https://lore.kernel.org/r/20200415162849.308-1-amadeuszx.slawinski@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
---
 sound/soc/codecs/hdac_hdmi.c | 6 +++---
 1 file changed, 3 insertions(+), 3 deletions(-)

diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c
index e6558475e006d..f0f689ddbefe8 100644
--- a/sound/soc/codecs/hdac_hdmi.c
+++ b/sound/soc/codecs/hdac_hdmi.c
@@ -142,14 +142,14 @@ static struct hdac_hdmi_pcm *
 hdac_hdmi_get_pcm_from_cvt(struct hdac_hdmi_priv *hdmi,
 			   struct hdac_hdmi_cvt *cvt)
 {
-	struct hdac_hdmi_pcm *pcm = NULL;
+	struct hdac_hdmi_pcm *pcm;
 
 	list_for_each_entry(pcm, &hdmi->pcm_list, head) {
 		if (pcm->cvt == cvt)
-			break;
+			return pcm;
 	}
 
-	return pcm;
+	return NULL;
 }
 
 static void hdac_hdmi_jack_report(struct hdac_hdmi_pcm *pcm,
-- 
2.20.1

[PATCH AUTOSEL 5.6 36/79] ASoC: stm32: sai: fix sai probe

[Sasha Levin]

From: Olivier Moysan <olivier.moysan@st.com>

[ Upstream commit e2bcb65782f91390952e849e21b82ed7cb05697f ]

pcm config must be set before snd_dmaengine_pcm_register() call.

Fixes: 0d6defc7e0e4 ("ASoC: stm32: sai: manage rebind issue")

Signed-off-by: Olivier Moysan <olivier.moysan@st.com>
Link: https://lore.kernel.org/r/20200417142122.10212-1-olivier.moysan@st.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
---
 sound/soc/stm/stm32_sai_sub.c | 12 +++++-------
 1 file changed, 5 insertions(+), 7 deletions(-)

diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c
index d3259de43712b..7e965848796c3 100644
--- a/sound/soc/stm/stm32_sai_sub.c
+++ b/sound/soc/stm/stm32_sai_sub.c
@@ -1543,6 +1543,9 @@ static int stm32_sai_sub_probe(struct platform_device *pdev)
 		return ret;
 	}
 
+	if (STM_SAI_PROTOCOL_IS_SPDIF(sai))
+		conf = &stm32_sai_pcm_config_spdif;
+
 	ret = snd_dmaengine_pcm_register(&pdev->dev, conf, 0);
 	if (ret) {
 		dev_err(&pdev->dev, "Could not register pcm dma\n");
@@ -1551,15 +1554,10 @@ static int stm32_sai_sub_probe(struct platform_device *pdev)
 
 	ret = snd_soc_register_component(&pdev->dev, &stm32_component,
 					 &sai->cpu_dai_drv, 1);
-	if (ret) {
+	if (ret)
 		snd_dmaengine_pcm_unregister(&pdev->dev);
-		return ret;
-	}
-
-	if (STM_SAI_PROTOCOL_IS_SPDIF(sai))
-		conf = &stm32_sai_pcm_config_spdif;
 
-	return 0;
+	return ret;
 }
 
 static int stm32_sai_sub_remove(struct platform_device *pdev)
-- 
2.20.1

[PATCH AUTOSEL 5.6 37/79] ASoC: SOF: Intel: add min/max channels for SSP on Baytrail/Broadwell

[Sasha Levin]

From: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>

[ Upstream commit 8c05246c0b58cbe80580ea4be05f6d51228af8a9 ]

Major regressions were detected by SOF CI on CherryTrail and Broadwell:

[   25.705750]  SSP2-Codec: ASoC: no backend playback stream
[   27.923378]  SSP2-Codec: ASoC: no users playback at close - state

This is root-caused to the introduction of the DAI capability checks
with snd_soc_dai_stream_valid(). Its use in soc-pcm.c makes it a
requirement for all DAIs to report at least a non-zero min_channels
field.

For some reason the SSP structures used for SKL+ did provide this
information but legacy platforms didn't.

Fixes: 9b5db059366ae2 ("ASoC: soc-pcm: dpcm: Only allow playback/capture if supported")
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20200417172014.11760-1-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
---
 sound/soc/sof/intel/bdw.c | 16 +++++++++++++
 sound/soc/sof/intel/byt.c | 48 +++++++++++++++++++++++++++++++++++++++
 2 files changed, 64 insertions(+)

diff --git a/sound/soc/sof/intel/bdw.c b/sound/soc/sof/intel/bdw.c
index 6c23c57693309..a32a3ef78ec5d 100644
--- a/sound/soc/sof/intel/bdw.c
+++ b/sound/soc/sof/intel/bdw.c
@@ -567,9 +567,25 @@ static void bdw_set_mach_params(const struct snd_soc_acpi_mach *mach,
 static struct snd_soc_dai_driver bdw_dai[] = {
 {
 	.name = "ssp0-port",
+	.playback = {
+		.channels_min = 1,
+		.channels_max = 8,
+	},
+	.capture = {
+		.channels_min = 1,
+		.channels_max = 8,
+	},
 },
 {
 	.name = "ssp1-port",
+	.playback = {
+		.channels_min = 1,
+		.channels_max = 8,
+	},
+	.capture = {
+		.channels_min = 1,
+		.channels_max = 8,
+	},
 },
 };
 
diff --git a/sound/soc/sof/intel/byt.c b/sound/soc/sof/intel/byt.c
index f84391294f12c..29fd1d86156ce 100644
--- a/sound/soc/sof/intel/byt.c
+++ b/sound/soc/sof/intel/byt.c
@@ -459,21 +459,69 @@ static void byt_set_mach_params(const struct snd_soc_acpi_mach *mach,
 static struct snd_soc_dai_driver byt_dai[] = {
 {
 	.name = "ssp0-port",
+	.playback = {
+		.channels_min = 1,
+		.channels_max = 8,
+	},
+	.capture = {
+		.channels_min = 1,
+		.channels_max = 8,
+	},
 },
 {
 	.name = "ssp1-port",
+	.playback = {
+		.channels_min = 1,
+		.channels_max = 8,
+	},
+	.capture = {
+		.channels_min = 1,
+		.channels_max = 8,
+	},
 },
 {
 	.name = "ssp2-port",
+	.playback = {
+		.channels_min = 1,
+		.channels_max = 8,
+	},
+	.capture = {
+		.channels_min = 1,
+		.channels_max = 8,
+	}
 },
 {
 	.name = "ssp3-port",
+	.playback = {
+		.channels_min = 1,
+		.channels_max = 8,
+	},
+	.capture = {
+		.channels_min = 1,
+		.channels_max = 8,
+	},
 },
 {
 	.name = "ssp4-port",
+	.playback = {
+		.channels_min = 1,
+		.channels_max = 8,
+	},
+	.capture = {
+		.channels_min = 1,
+		.channels_max = 8,
+	},
 },
 {
 	.name = "ssp5-port",
+	.playback = {
+		.channels_min = 1,
+		.channels_max = 8,
+	},
+	.capture = {
+		.channels_min = 1,
+		.channels_max = 8,
+	},
 },
 };
 
-- 
2.20.1

[PATCH AUTOSEL 5.6 45/79] ASoC: meson: axg-card: fix codec-to-codec link setup

[Sasha Levin]

From: Jerome Brunet <jbrunet@baylibre.com>

[ Upstream commit 1164284270779e1865cc2046a2a01b58a1e858a9 ]

Since the addition of commit 9b5db059366a ("ASoC: soc-pcm: dpcm: Only allow
playback/capture if supported"), meson-axg cards which have codec-to-codec
links fail to init and Oops:

  Unable to handle kernel NULL pointer dereference at virtual address 0000000000000128
  Internal error: Oops: 96000044 [#1] PREEMPT SMP
  CPU: 3 PID: 1582 Comm: arecord Not tainted 5.7.0-rc1
  pc : invalidate_paths_ep+0x30/0xe0
  lr : snd_soc_dapm_dai_get_connected_widgets+0x170/0x1a8
  Call trace:
   invalidate_paths_ep+0x30/0xe0
   snd_soc_dapm_dai_get_connected_widgets+0x170/0x1a8
   dpcm_path_get+0x38/0xd0
   dpcm_fe_dai_open+0x70/0x920
   snd_pcm_open_substream+0x564/0x840
   snd_pcm_open+0xfc/0x228
   snd_pcm_capture_open+0x4c/0x78
   snd_open+0xac/0x1a8
   ...

While initiliazing the links, ASoC treats the codec-to-codec links of this
card type as a DPCM backend. This error eventually leads to the Oops.

Most of the card driver code is shared between DPCM backends and
codec-to-codec links. The property "no_pcm" marking DCPM BE was left set on
codec-to-codec links, leading to this problem. This commit fixes that.

Fixes: 0a8f1117a680 ("ASoC: meson: axg-card: add basic codec-to-codec link support")
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Link: https://lore.kernel.org/r/20200420114511.450560-2-jbrunet@baylibre.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
---
 sound/soc/meson/axg-card.c | 4 +++-
 1 file changed, 3 insertions(+), 1 deletion(-)

diff --git a/sound/soc/meson/axg-card.c b/sound/soc/meson/axg-card.c
index 1f698adde506c..2b04ac3d8fd3b 100644
--- a/sound/soc/meson/axg-card.c
+++ b/sound/soc/meson/axg-card.c
@@ -586,8 +586,10 @@ static int axg_card_add_link(struct snd_soc_card *card, struct device_node *np,
 
 	if (axg_card_cpu_is_tdm_iface(dai_link->cpus->of_node))
 		ret = axg_card_parse_tdm(card, np, index);
-	else if (axg_card_cpu_is_codec(dai_link->cpus->of_node))
+	else if (axg_card_cpu_is_codec(dai_link->cpus->of_node)) {
 		dai_link->params = &codec_params;
+		dai_link->no_pcm = 0; /* link is not a DPCM BE */
+	}
 
 	return ret;
 }
-- 
2.20.1

[PATCH AUTOSEL 5.6 46/79] ASoC: rsnd: Don't treat master SSI in multi SSI setup as parent

[Sasha Levin]

From: Matthias Blankertz <matthias.blankertz@cetitec.com>

[ Upstream commit 0c258657ddfe81b4fc0183378d800c97ba0b7cdd ]

The master SSI of a multi-SSI setup was attached both to the
RSND_MOD_SSI slot and the RSND_MOD_SSIP slot of the rsnd_dai_stream.
This is not correct wrt. the meaning of being "parent" in the rest of
the SSI code, where it seems to indicate an SSI that provides clock and
word sync but is not transmitting/receiving audio data.

Not treating the multi-SSI master as parent allows removal of various
special cases to the rsnd_ssi_is_parent conditions introduced in commit
a09fb3f28a60 ("ASoC: rsnd: Fix parent SSI start/stop in multi-SSI mode").
It also fixes the issue that operations performed via rsnd_dai_call()
were performed twice for the master SSI. This caused some "status check
failed" spam when stopping a multi-SSI stream as the driver attempted to
stop the master SSI twice.

Signed-off-by: Matthias Blankertz <matthias.blankertz@cetitec.com>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/20200417153017.1744454-2-matthias.blankertz@cetitec.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
---
 sound/soc/sh/rcar/ssi.c | 11 +++++++----
 1 file changed, 7 insertions(+), 4 deletions(-)

diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c
index d51fb3a394486..9900a4f6f4e53 100644
--- a/sound/soc/sh/rcar/ssi.c
+++ b/sound/soc/sh/rcar/ssi.c
@@ -407,7 +407,7 @@ static void rsnd_ssi_config_init(struct rsnd_mod *mod,
 	 * We shouldn't exchange SWSP after running.
 	 * This means, parent needs to care it.
 	 */
-	if (rsnd_ssi_is_parent(mod, io) && !rsnd_ssi_multi_slaves(io))
+	if (rsnd_ssi_is_parent(mod, io))
 		goto init_end;
 
 	if (rsnd_io_is_play(io))
@@ -559,7 +559,7 @@ static int rsnd_ssi_start(struct rsnd_mod *mod,
 	 * EN is for data output.
 	 * SSI parent EN is not needed.
 	 */
-	if (rsnd_ssi_is_parent(mod, io) && !rsnd_ssi_multi_slaves(io))
+	if (rsnd_ssi_is_parent(mod, io))
 		return 0;
 
 	ssi->cr_en = EN;
@@ -582,7 +582,7 @@ static int rsnd_ssi_stop(struct rsnd_mod *mod,
 	if (!rsnd_ssi_is_run_mods(mod, io))
 		return 0;
 
-	if (rsnd_ssi_is_parent(mod, io) && !rsnd_ssi_multi_slaves(io))
+	if (rsnd_ssi_is_parent(mod, io))
 		return 0;
 
 	cr  =	ssi->cr_own	|
@@ -620,7 +620,7 @@ static int rsnd_ssi_irq(struct rsnd_mod *mod,
 	if (rsnd_is_gen1(priv))
 		return 0;
 
-	if (rsnd_ssi_is_parent(mod, io) && !rsnd_ssi_multi_slaves(io))
+	if (rsnd_ssi_is_parent(mod, io))
 		return 0;
 
 	if (!rsnd_ssi_is_run_mods(mod, io))
@@ -737,6 +737,9 @@ static void rsnd_ssi_parent_attach(struct rsnd_mod *mod,
 	if (!rsnd_rdai_is_clk_master(rdai))
 		return;
 
+	if (rsnd_ssi_is_multi_slave(mod, io))
+		return;
+
 	switch (rsnd_mod_id(mod)) {
 	case 1:
 	case 2:
-- 
2.20.1

[PATCH AUTOSEL 5.6 47/79] ASoC: rsnd: Fix "status check failed" spam for multi-SSI

[Sasha Levin]

From: Matthias Blankertz <matthias.blankertz@cetitec.com>

[ Upstream commit 54cb6221688660670a2e430892d7f4e6370263b8 ]

Fix the rsnd_ssi_stop function to skip disabling the individual SSIs of
a multi-SSI setup, as the actual stop is performed by rsnd_ssiu_stop_gen2
- the same logic as in rsnd_ssi_start. The attempt to disable these SSIs
was harmless, but caused a "status check failed" message to be printed
for every SSI in the multi-SSI setup.
The disabling of interrupts is still performed, as they are enabled for
all SSIs in rsnd_ssi_init, but care is taken to not accidentally set the
EN bit for an SSI where it was not set by rsnd_ssi_start.

Signed-off-by: Matthias Blankertz <matthias.blankertz@cetitec.com>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/20200417153017.1744454-3-matthias.blankertz@cetitec.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
---
 sound/soc/sh/rcar/ssi.c | 8 +++++++-
 1 file changed, 7 insertions(+), 1 deletion(-)

diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c
index 9900a4f6f4e53..4a7d3413917fc 100644
--- a/sound/soc/sh/rcar/ssi.c
+++ b/sound/soc/sh/rcar/ssi.c
@@ -594,10 +594,16 @@ static int rsnd_ssi_stop(struct rsnd_mod *mod,
 	 * Capture:  It might not receave data. Do nothing
 	 */
 	if (rsnd_io_is_play(io)) {
-		rsnd_mod_write(mod, SSICR, cr | EN);
+		rsnd_mod_write(mod, SSICR, cr | ssi->cr_en);
 		rsnd_ssi_status_check(mod, DIRQ);
 	}
 
+	/* In multi-SSI mode, stop is performed by setting ssi0129 in
+	 * SSI_CONTROL to 0 (in rsnd_ssio_stop_gen2). Do nothing here.
+	 */
+	if (rsnd_ssi_multi_slaves_runtime(io))
+		return 0;
+
 	/*
 	 * disable SSI,
 	 * and, wait idle state
-- 
2.20.1

[PATCH AUTOSEL 5.6 56/79] ASoC: wm8960: Fix wrong clock after suspend & resume

[Sasha Levin]

From: Shengjiu Wang <shengjiu.wang@nxp.com>

[ Upstream commit 1e060a453c8604311fb45ae2f84f67ed673329b4 ]

After suspend & resume, wm8960_hw_params may be called when
bias_level is not SND_SOC_BIAS_ON, then wm8960_configure_clocking
is not called. But if sample rate is changed at that time, then
the output clock rate will be not correct.

So judgement of bias_level is SND_SOC_BIAS_ON in wm8960_hw_params
is not necessary and it causes above issue.

Fixes: 3176bf2d7ccd ("ASoC: wm8960: update pll and clock setting function")
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Acked-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/1587468525-27514-1-git-send-email-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
---
 sound/soc/codecs/wm8960.c | 3 +--
 1 file changed, 1 insertion(+), 2 deletions(-)

diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c
index 55112c1bba5e7..6cf0f6612bdaf 100644
--- a/sound/soc/codecs/wm8960.c
+++ b/sound/soc/codecs/wm8960.c
@@ -860,8 +860,7 @@ static int wm8960_hw_params(struct snd_pcm_substream *substream,
 
 	wm8960->is_stream_in_use[tx] = true;
 
-	if (snd_soc_component_get_bias_level(component) == SND_SOC_BIAS_ON &&
-	    !wm8960->is_stream_in_use[!tx])
+	if (!wm8960->is_stream_in_use[!tx])
 		return wm8960_configure_clocking(component);
 
 	return 0;
-- 
2.20.1

Re: [PATCH AUTOSEL 5.6 45/79] ASoC: meson: axg-card: fix codec-to-codec link setup

[Mark Brown]

[-- Attachment #1: Type: text/plain, Size: 349 bytes --]

On Thu, Apr 30, 2020 at 09:50:09AM -0400, Sasha Levin wrote:

> Since the addition of commit 9b5db059366a ("ASoC: soc-pcm: dpcm: Only allow
> playback/capture if supported"), meson-axg cards which have codec-to-codec
> links fail to init and Oops:

This clearly describes the issue as only being present after the above
commit which is not in v5.6.

[-- Attachment #2: signature.asc --]
[-- Type: application/pgp-signature, Size: 488 bytes --]

Re: [PATCH AUTOSEL 5.6 45/79] ASoC: meson: axg-card: fix codec-to-codec link setup

[Sasha Levin]

On Thu, Apr 30, 2020 at 02:56:10PM +0100, Mark Brown wrote:
>On Thu, Apr 30, 2020 at 09:50:09AM -0400, Sasha Levin wrote:
>
>> Since the addition of commit 9b5db059366a ("ASoC: soc-pcm: dpcm: Only allow
>> playback/capture if supported"), meson-axg cards which have codec-to-codec
>> links fail to init and Oops:
>
>This clearly describes the issue as only being present after the above
>commit which is not in v5.6.

Uh, I thought that the fixes tag points to the same commit as mentioned
in the description... I'll drop it, sorry.

-- 
Thanks,
Sasha