[PATCH v2 0/6] ASoC: improve codec to codec link support

[PATCH v2 0/6] ASoC: improve codec to codec link support

[Jerome Brunet]

As explained in this previous series [0], on Amlogic, we are using codec to
codec links to deal with the glue which is between the i2s backends and the
synopsys hdmi controller.

This worked well until I tried to .get_eld() support in the dw-hdmi-i2s
driver.  Doing so adds channel mapping controls to the codec. This shown
several problem

1) .pcm_new() is not called on codec to codec links.
   struct snd_soc_pcm_runtime do not even have a valid .pcm
2) struct snd_pcm_substream and struct snd_pcm_runtime are ephemeral
   This is a problem if a control needs to access them

The goal of this patchset is to resolve the above issues and improve the
codec to codec link support enough to correctly handle the hdmi-codec

The support of these codec to codec links could probably be improved in the
future to behave like any other link and use soc_pcm_open(),
soc_pcm_hw_params(), etc...

The challenge lies in the dapm mutex. The soc_pcm call dapm function locking
this mutex but the dapm mutex is already held in snd_soc_dai_link_event()

Changes since v1 [1]:
* Fix rebase on Murimoto-san's patches
* Allocate params dynamically again and refactor the pre_pmu code to
  simplify the error handling and rollback a bit

[0]: https://lkml.kernel.org/r/20190515131858.32130-1-jbrunet@baylibre.com
[1]: https://lkml.kernel.org/r/20190724162405.6574-1-jbrunet@baylibre.com

Jerome Brunet (6):
  ASoC: codec2codec: run callbacks in order
  ASoC: codec2codec: name link using stream direction
  ASoC: codec2codec: deal with params when necessary
  ASoC: create pcm for codec2codec links as well
  ASoC: codec2codec: remove ephemeral variables
  ASoC: codec2codec: fill some of the runtime stream parameters

 sound/soc/soc-core.c |  42 +++------
 sound/soc/soc-dapm.c | 220 +++++++++++++++++++++++++++----------------
 sound/soc/soc-pcm.c  |  35 ++++++-
 3 files changed, 182 insertions(+), 115 deletions(-)

-- 
2.21.0


_______________________________________________
linux-amlogic mailing list
linux-amlogic@lists.infradead.org
http://lists.infradead.org/mailman/listinfo/linux-amlogic

[PATCH v2 1/6] ASoC: codec2codec: run callbacks in order

[Jerome Brunet]

When handling dai_link events on codec to codec links, run all .startup()
callbacks on sinks and sources before running any .hw_params(). Same goes
for hw_free() and shutdown(). This is closer to the behavior of regular
dai links

Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
---
 sound/soc/soc-dapm.c | 36 +++++++++++++++++++++++++++---------
 1 file changed, 27 insertions(+), 9 deletions(-)

diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 1d04612601ad..034b31fd2ecb 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -3835,11 +3835,6 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w,
 				goto out;
 			}
 			source->active++;
-			ret = snd_soc_dai_hw_params(source, &substream, params);
-			if (ret < 0)
-				goto out;
-
-			dapm_update_dai_unlocked(&substream, params, source);
 		}
 
 		substream.stream = SNDRV_PCM_STREAM_PLAYBACK;
@@ -3853,6 +3848,23 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w,
 				goto out;
 			}
 			sink->active++;
+		}
+
+		substream.stream = SNDRV_PCM_STREAM_CAPTURE;
+		snd_soc_dapm_widget_for_each_source_path(w, path) {
+			source = path->source->priv;
+
+			ret = snd_soc_dai_hw_params(source, &substream, params);
+			if (ret < 0)
+				goto out;
+
+			dapm_update_dai_unlocked(&substream, params, source);
+		}
+
+		substream.stream = SNDRV_PCM_STREAM_PLAYBACK;
+		snd_soc_dapm_widget_for_each_sink_path(w, path) {
+			sink = path->sink->priv;
+
 			ret = snd_soc_dai_hw_params(sink, &substream, params);
 			if (ret < 0)
 				goto out;
@@ -3889,9 +3901,18 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w,
 		substream.stream = SNDRV_PCM_STREAM_CAPTURE;
 		snd_soc_dapm_widget_for_each_source_path(w, path) {
 			source = path->source->priv;
-
 			snd_soc_dai_hw_free(source, &substream);
+		}
+
+		substream.stream = SNDRV_PCM_STREAM_PLAYBACK;
+		snd_soc_dapm_widget_for_each_sink_path(w, path) {
+			sink = path->sink->priv;
+			snd_soc_dai_hw_free(sink, &substream);
+		}
 
+		substream.stream = SNDRV_PCM_STREAM_CAPTURE;
+		snd_soc_dapm_widget_for_each_source_path(w, path) {
+			source = path->source->priv;
 			source->active--;
 			snd_soc_dai_shutdown(source, &substream);
 		}
@@ -3899,9 +3920,6 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w,
 		substream.stream = SNDRV_PCM_STREAM_PLAYBACK;
 		snd_soc_dapm_widget_for_each_sink_path(w, path) {
 			sink = path->sink->priv;
-
-			snd_soc_dai_hw_free(sink, &substream);
-
 			sink->active--;
 			snd_soc_dai_shutdown(sink, &substream);
 		}
-- 
2.21.0


_______________________________________________
linux-amlogic mailing list
linux-amlogic@lists.infradead.org
http://lists.infradead.org/mailman/listinfo/linux-amlogic

[PATCH v2 2/6] ASoC: codec2codec: name link using stream direction

[Jerome Brunet]

At the moment, codec to codec dai link widgets are named after the
cpu dai and the 1st codec valid on the link. This might be confusing
if there is multiple valid codecs on the link for one stream
direction.

Instead, use the dai link name and the stream direction to name the
the dai link widget

Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
---
 sound/soc/soc-dapm.c | 12 ++++--------
 1 file changed, 4 insertions(+), 8 deletions(-)

diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 034b31fd2ecb..7db4abd9a0a5 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -4056,8 +4056,7 @@ snd_soc_dapm_alloc_kcontrol(struct snd_soc_card *card,
 
 static struct snd_soc_dapm_widget *
 snd_soc_dapm_new_dai(struct snd_soc_card *card, struct snd_soc_pcm_runtime *rtd,
-		     struct snd_soc_dapm_widget *source,
-		     struct snd_soc_dapm_widget *sink)
+		     char *id)
 {
 	struct snd_soc_dapm_widget template;
 	struct snd_soc_dapm_widget *w;
@@ -4067,7 +4066,7 @@ snd_soc_dapm_new_dai(struct snd_soc_card *card, struct snd_soc_pcm_runtime *rtd,
 	int ret;
 
 	link_name = devm_kasprintf(card->dev, GFP_KERNEL, "%s-%s",
-				   source->name, sink->name);
+				   rtd->dai_link->name, id);
 	if (!link_name)
 		return ERR_PTR(-ENOMEM);
 
@@ -4247,15 +4246,13 @@ static void dapm_connect_dai_link_widgets(struct snd_soc_card *card,
 	}
 
 	for_each_rtd_codec_dai(rtd, i, codec_dai) {
-
 		/* connect BE DAI playback if widgets are valid */
 		codec = codec_dai->playback_widget;
 
 		if (playback_cpu && codec) {
 			if (!playback) {
 				playback = snd_soc_dapm_new_dai(card, rtd,
-								playback_cpu,
-								codec);
+								"playback");
 				if (IS_ERR(playback)) {
 					dev_err(rtd->dev,
 						"ASoC: Failed to create DAI %s: %ld\n",
@@ -4284,8 +4281,7 @@ static void dapm_connect_dai_link_widgets(struct snd_soc_card *card,
 		if (codec && capture_cpu) {
 			if (!capture) {
 				capture = snd_soc_dapm_new_dai(card, rtd,
-							       codec,
-							       capture_cpu);
+							       "capture");
 				if (IS_ERR(capture)) {
 					dev_err(rtd->dev,
 						"ASoC: Failed to create DAI %s: %ld\n",
-- 
2.21.0


_______________________________________________
linux-amlogic mailing list
linux-amlogic@lists.infradead.org
http://lists.infradead.org/mailman/listinfo/linux-amlogic

[PATCH v2 3/6] ASoC: codec2codec: deal with params when necessary

[Jerome Brunet]

When there is an event on codec to codec dai_link, we only need to deal
with params if the event is SND_SOC_DAPM_PRE_PMU, when .hw_params() is
called. For the other events, it is useless.

Also, dealing with the codec to codec params just before calling
.hw_params() callbacks give change to either party on the link to alter
params content in .startup(), which might be useful in some cases

Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
---
 sound/soc/soc-dapm.c | 159 +++++++++++++++++++++++++------------------
 1 file changed, 92 insertions(+), 67 deletions(-)

diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 7db4abd9a0a5..6dcaf9ff6eb5 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -3764,25 +3764,59 @@ int snd_soc_dapm_new_controls(struct snd_soc_dapm_context *dapm,
 }
 EXPORT_SYMBOL_GPL(snd_soc_dapm_new_controls);
 
-static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w,
-				  struct snd_kcontrol *kcontrol, int event)
+static int
+snd_soc_dai_link_event_pre_pmu(struct snd_soc_dapm_widget *w,
+			       struct snd_pcm_substream *substream)
 {
 	struct snd_soc_dapm_path *path;
 	struct snd_soc_dai *source, *sink;
-	struct snd_soc_pcm_runtime *rtd = w->priv;
-	const struct snd_soc_pcm_stream *config;
-	struct snd_pcm_substream substream;
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_pcm_hw_params *params = NULL;
-	struct snd_pcm_runtime *runtime = NULL;
+	const struct snd_soc_pcm_stream *config = NULL;
 	unsigned int fmt;
-	int ret = 0;
+	int ret;
 
-	config = rtd->dai_link->params + rtd->params_select;
+	params = kzalloc(sizeof(*params), GFP_KERNEL);
+	if (!params)
+		return -ENOMEM;
 
-	if (WARN_ON(!config) ||
-	    WARN_ON(list_empty(&w->edges[SND_SOC_DAPM_DIR_OUT]) ||
-		    list_empty(&w->edges[SND_SOC_DAPM_DIR_IN])))
-		return -EINVAL;
+	substream->stream = SNDRV_PCM_STREAM_CAPTURE;
+	snd_soc_dapm_widget_for_each_source_path(w, path) {
+		source = path->source->priv;
+
+		ret = snd_soc_dai_startup(source, substream);
+		if (ret < 0) {
+			dev_err(source->dev,
+				"ASoC: startup() failed: %d\n", ret);
+			goto out;
+		}
+		source->active++;
+	}
+
+	substream->stream = SNDRV_PCM_STREAM_PLAYBACK;
+	snd_soc_dapm_widget_for_each_sink_path(w, path) {
+		sink = path->sink->priv;
+
+		ret = snd_soc_dai_startup(sink, substream);
+		if (ret < 0) {
+			dev_err(sink->dev,
+				"ASoC: startup() failed: %d\n", ret);
+			goto out;
+		}
+		sink->active++;
+	}
+
+	/*
+	 * Note: getting the config after .startup() gives a chance to
+	 * either party on the link to alter the configuration if
+	 * necessary
+	 */
+	config = rtd->dai_link->params + rtd->params_select;
+	if (WARN_ON(!config)) {
+		dev_err(w->dapm->dev, "ASoC: link config missing\n");
+		ret = -EINVAL;
+		goto out;
+	}
 
 	/* Be a little careful as we don't want to overflow the mask array */
 	if (config->formats) {
@@ -3790,27 +3824,62 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w,
 	} else {
 		dev_warn(w->dapm->dev, "ASoC: Invalid format %llx specified\n",
 			 config->formats);
-		fmt = 0;
-	}
 
-	/* Currently very limited parameter selection */
-	params = kzalloc(sizeof(*params), GFP_KERNEL);
-	if (!params) {
-		ret = -ENOMEM;
+		ret = -EINVAL;
 		goto out;
 	}
-	snd_mask_set(hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT), fmt);
 
+	snd_mask_set(hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT), fmt);
 	hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE)->min =
 		config->rate_min;
 	hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE)->max =
 		config->rate_max;
-
 	hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS)->min
 		= config->channels_min;
 	hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS)->max
 		= config->channels_max;
 
+	substream->stream = SNDRV_PCM_STREAM_CAPTURE;
+	snd_soc_dapm_widget_for_each_source_path(w, path) {
+		source = path->source->priv;
+
+		ret = snd_soc_dai_hw_params(source, substream, params);
+		if (ret < 0)
+			goto out;
+
+		dapm_update_dai_unlocked(substream, params, source);
+	}
+
+	substream->stream = SNDRV_PCM_STREAM_PLAYBACK;
+	snd_soc_dapm_widget_for_each_sink_path(w, path) {
+		sink = path->sink->priv;
+
+		ret = snd_soc_dai_hw_params(sink, substream, params);
+		if (ret < 0)
+			goto out;
+
+		dapm_update_dai_unlocked(substream, params, sink);
+	}
+
+out:
+	kfree(params);
+	return 0;
+}
+
+static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w,
+				  struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_dapm_path *path;
+	struct snd_soc_dai *source, *sink;
+	struct snd_soc_pcm_runtime *rtd = w->priv;
+	struct snd_pcm_substream substream;
+	struct snd_pcm_runtime *runtime = NULL;
+	int ret = 0;
+
+	if (WARN_ON(list_empty(&w->edges[SND_SOC_DAPM_DIR_OUT]) ||
+		    list_empty(&w->edges[SND_SOC_DAPM_DIR_IN])))
+		return -EINVAL;
+
 	memset(&substream, 0, sizeof(substream));
 
 	/* Allocate a dummy snd_pcm_runtime for startup() and other ops() */
@@ -3824,53 +3893,10 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w,
 
 	switch (event) {
 	case SND_SOC_DAPM_PRE_PMU:
-		substream.stream = SNDRV_PCM_STREAM_CAPTURE;
-		snd_soc_dapm_widget_for_each_source_path(w, path) {
-			source = path->source->priv;
-
-			ret = snd_soc_dai_startup(source, &substream);
-			if (ret < 0) {
-				dev_err(source->dev,
-					"ASoC: startup() failed: %d\n", ret);
-				goto out;
-			}
-			source->active++;
-		}
-
-		substream.stream = SNDRV_PCM_STREAM_PLAYBACK;
-		snd_soc_dapm_widget_for_each_sink_path(w, path) {
-			sink = path->sink->priv;
-
-			ret = snd_soc_dai_startup(sink, &substream);
-			if (ret < 0) {
-				dev_err(sink->dev,
-					"ASoC: startup() failed: %d\n", ret);
-				goto out;
-			}
-			sink->active++;
-		}
-
-		substream.stream = SNDRV_PCM_STREAM_CAPTURE;
-		snd_soc_dapm_widget_for_each_source_path(w, path) {
-			source = path->source->priv;
-
-			ret = snd_soc_dai_hw_params(source, &substream, params);
-			if (ret < 0)
-				goto out;
-
-			dapm_update_dai_unlocked(&substream, params, source);
-		}
-
-		substream.stream = SNDRV_PCM_STREAM_PLAYBACK;
-		snd_soc_dapm_widget_for_each_sink_path(w, path) {
-			sink = path->sink->priv;
-
-			ret = snd_soc_dai_hw_params(sink, &substream, params);
-			if (ret < 0)
-				goto out;
+		ret = snd_soc_dai_link_event_pre_pmu(w, &substream);
+		if (ret < 0)
+			goto out;
 
-			dapm_update_dai_unlocked(&substream, params, sink);
-		}
 		break;
 
 	case SND_SOC_DAPM_POST_PMU:
@@ -3932,7 +3958,6 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w,
 
 out:
 	kfree(runtime);
-	kfree(params);
 	return ret;
 }
 
-- 
2.21.0


_______________________________________________
linux-amlogic mailing list
linux-amlogic@lists.infradead.org
http://lists.infradead.org/mailman/listinfo/linux-amlogic

[PATCH v2 4/6] ASoC: create pcm for codec2codec links as well

[Jerome Brunet]

At the moment, codec to codec links uses an ephemeral variable for
the struct snd_pcm_substream. Also the struct snd_soc_pcm_runtime
does not have real struct snd_pcm.

This might a problem if the functions used by a codec on codec to
codec link expect these structures to exist, and keep on existing
during the life of the codec.

For example, it is the case of the hdmi-codec, which uses
snd_pcm_add_chmap_ctls(). For the controls to works, the pcm and
substream must to exist.

This change is first step, it create pcm (and substreams) for codec
to codec links, in the same way as dpcm backend links.

Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
---
 sound/soc/soc-core.c | 42 ++++++++++++------------------------------
 sound/soc/soc-pcm.c  | 35 ++++++++++++++++++++++++++++++++---
 2 files changed, 44 insertions(+), 33 deletions(-)

diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 7ecfe641ca46..9e8eb93c8a3f 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -447,16 +447,6 @@ static void snd_soc_flush_all_delayed_work(struct snd_soc_card *card)
 		flush_delayed_work(&rtd->delayed_work);
 }
 
-static void codec2codec_close_delayed_work(struct work_struct *work)
-{
-	/*
-	 * Currently nothing to do for c2c links
-	 * Since c2c links are internal nodes in the DAPM graph and
-	 * don't interface with the outside world or application layer
-	 * we don't have to do any special handling on close.
-	 */
-}
-
 #ifdef CONFIG_PM_SLEEP
 /* powers down audio subsystem for suspend */
 int snd_soc_suspend(struct device *dev)
@@ -1552,27 +1542,19 @@ static int soc_probe_link_dais(struct snd_soc_card *card,
 		return ret;
 	}
 
-	if (!dai_link->params) {
-		/* create the pcm */
-		ret = soc_new_pcm(rtd, num);
-		if (ret < 0) {
-			dev_err(card->dev, "ASoC: can't create pcm %s :%d\n",
-				dai_link->stream_name, ret);
-			return ret;
-		}
-		ret = soc_link_dai_pcm_new(&cpu_dai, 1, rtd);
-		if (ret < 0)
-			return ret;
-		ret = soc_link_dai_pcm_new(rtd->codec_dais,
-					   rtd->num_codecs, rtd);
-		if (ret < 0)
-			return ret;
-	} else {
-		INIT_DELAYED_WORK(&rtd->delayed_work,
-				  codec2codec_close_delayed_work);
+	/* create the pcm */
+	ret = soc_new_pcm(rtd, num);
+	if (ret < 0) {
+		dev_err(card->dev, "ASoC: can't create pcm %s :%d\n",
+			dai_link->stream_name, ret);
+		return ret;
 	}
-
-	return 0;
+	ret = soc_link_dai_pcm_new(&cpu_dai, 1, rtd);
+	if (ret < 0)
+		return ret;
+	ret = soc_link_dai_pcm_new(rtd->codec_dais,
+				   rtd->num_codecs, rtd);
+	return ret;
 }
 
 static int soc_bind_aux_dev(struct snd_soc_card *card, int num)
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index fabeac164a6c..30264bc592f6 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -678,6 +678,16 @@ static void close_delayed_work(struct work_struct *work)
 	mutex_unlock(&rtd->pcm_mutex);
 }
 
+static void codec2codec_close_delayed_work(struct work_struct *work)
+{
+	/*
+	 * Currently nothing to do for c2c links
+	 * Since c2c links are internal nodes in the DAPM graph and
+	 * don't interface with the outside world or application layer
+	 * we don't have to do any special handling on close.
+	 */
+}
+
 /*
  * Called by ALSA when a PCM substream is closed. Private data can be
  * freed here. The cpu DAI, codec DAI, machine and components are also
@@ -3011,6 +3021,12 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num)
 		playback = rtd->dai_link->dpcm_playback;
 		capture = rtd->dai_link->dpcm_capture;
 	} else {
+		/* Adapt stream for codec2codec links */
+		struct snd_soc_pcm_stream *cpu_capture = rtd->dai_link->params ?
+			&cpu_dai->driver->playback : &cpu_dai->driver->capture;
+		struct snd_soc_pcm_stream *cpu_playback = rtd->dai_link->params ?
+			&cpu_dai->driver->capture : &cpu_dai->driver->playback;
+
 		for_each_rtd_codec_dai(rtd, i, codec_dai) {
 			if (snd_soc_dai_stream_valid(codec_dai, SNDRV_PCM_STREAM_PLAYBACK) &&
 			    snd_soc_dai_stream_valid(cpu_dai,   SNDRV_PCM_STREAM_PLAYBACK))
@@ -3019,6 +3035,9 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num)
 			    snd_soc_dai_stream_valid(cpu_dai,   SNDRV_PCM_STREAM_CAPTURE))
 				capture = 1;
 		}
+
+		capture = capture && cpu_capture->channels_min;
+		playback = playback && cpu_playback->channels_min;
 	}
 
 	if (rtd->dai_link->playback_only) {
@@ -3032,7 +3051,13 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num)
 	}
 
 	/* create the PCM */
-	if (rtd->dai_link->no_pcm) {
+	if (rtd->dai_link->params) {
+		snprintf(new_name, sizeof(new_name), "codec2codec(%s)",
+			 rtd->dai_link->stream_name);
+
+		ret = snd_pcm_new_internal(rtd->card->snd_card, new_name, num,
+					   playback, capture, &pcm);
+	} else if (rtd->dai_link->no_pcm) {
 		snprintf(new_name, sizeof(new_name), "(%s)",
 			rtd->dai_link->stream_name);
 
@@ -3059,13 +3084,17 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num)
 	dev_dbg(rtd->card->dev, "ASoC: registered pcm #%d %s\n",num, new_name);
 
 	/* DAPM dai link stream work */
-	INIT_DELAYED_WORK(&rtd->delayed_work, close_delayed_work);
+	if (rtd->dai_link->params)
+		INIT_DELAYED_WORK(&rtd->delayed_work,
+				  codec2codec_close_delayed_work);
+	else
+		INIT_DELAYED_WORK(&rtd->delayed_work, close_delayed_work);
 
 	pcm->nonatomic = rtd->dai_link->nonatomic;
 	rtd->pcm = pcm;
 	pcm->private_data = rtd;
 
-	if (rtd->dai_link->no_pcm) {
+	if (rtd->dai_link->no_pcm || rtd->dai_link->params) {
 		if (playback)
 			pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream->private_data = rtd;
 		if (capture)
-- 
2.21.0


_______________________________________________
linux-amlogic mailing list
linux-amlogic@lists.infradead.org
http://lists.infradead.org/mailman/listinfo/linux-amlogic

[PATCH v2 5/6] ASoC: codec2codec: remove ephemeral variables

[Jerome Brunet]

Now that codec to codec links struct snd_soc_pcm_runtime have lasting pcm
and substreams, let's use them. Alsa allocate and keep the
struct snd_pcm_runtime as long as the link is powered.

Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
---
 sound/soc/soc-dapm.c | 72 ++++++++++++++++++++++++++------------------
 1 file changed, 42 insertions(+), 30 deletions(-)

diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 6dcaf9ff6eb5..3e3b81ae87dd 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -3773,6 +3773,7 @@ snd_soc_dai_link_event_pre_pmu(struct snd_soc_dapm_widget *w,
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_pcm_hw_params *params = NULL;
 	const struct snd_soc_pcm_stream *config = NULL;
+	struct snd_pcm_runtime *runtime = NULL;
 	unsigned int fmt;
 	int ret;
 
@@ -3780,6 +3781,14 @@ snd_soc_dai_link_event_pre_pmu(struct snd_soc_dapm_widget *w,
 	if (!params)
 		return -ENOMEM;
 
+	runtime = kzalloc(sizeof(*runtime), GFP_KERNEL);
+	if (!runtime) {
+		ret = -ENOMEM;
+		goto out;
+	}
+
+	substream->runtime = runtime;
+
 	substream->stream = SNDRV_PCM_STREAM_CAPTURE;
 	snd_soc_dapm_widget_for_each_source_path(w, path) {
 		source = path->source->priv;
@@ -3806,6 +3815,8 @@ snd_soc_dai_link_event_pre_pmu(struct snd_soc_dapm_widget *w,
 		sink->active++;
 	}
 
+	substream->hw_opened = 1;
+
 	/*
 	 * Note: getting the config after .startup() gives a chance to
 	 * either party on the link to alter the configuration if
@@ -3862,6 +3873,9 @@ snd_soc_dai_link_event_pre_pmu(struct snd_soc_dapm_widget *w,
 	}
 
 out:
+	if (ret < 0)
+		kfree(runtime);
+
 	kfree(params);
 	return 0;
 }
@@ -3871,29 +3885,16 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w,
 {
 	struct snd_soc_dapm_path *path;
 	struct snd_soc_dai *source, *sink;
-	struct snd_soc_pcm_runtime *rtd = w->priv;
-	struct snd_pcm_substream substream;
-	struct snd_pcm_runtime *runtime = NULL;
-	int ret = 0;
+	struct snd_pcm_substream *substream = w->priv;
+	int ret = 0, saved_stream = substream->stream;
 
 	if (WARN_ON(list_empty(&w->edges[SND_SOC_DAPM_DIR_OUT]) ||
 		    list_empty(&w->edges[SND_SOC_DAPM_DIR_IN])))
 		return -EINVAL;
 
-	memset(&substream, 0, sizeof(substream));
-
-	/* Allocate a dummy snd_pcm_runtime for startup() and other ops() */
-	runtime = kzalloc(sizeof(*runtime), GFP_KERNEL);
-	if (!runtime) {
-		ret = -ENOMEM;
-		goto out;
-	}
-	substream.runtime = runtime;
-	substream.private_data = rtd;
-
 	switch (event) {
 	case SND_SOC_DAPM_PRE_PMU:
-		ret = snd_soc_dai_link_event_pre_pmu(w, &substream);
+		ret = snd_soc_dai_link_event_pre_pmu(w, substream);
 		if (ret < 0)
 			goto out;
 
@@ -3924,40 +3925,45 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w,
 			ret = 0;
 		}
 
-		substream.stream = SNDRV_PCM_STREAM_CAPTURE;
+		substream->stream = SNDRV_PCM_STREAM_CAPTURE;
 		snd_soc_dapm_widget_for_each_source_path(w, path) {
 			source = path->source->priv;
-			snd_soc_dai_hw_free(source, &substream);
+			snd_soc_dai_hw_free(source, substream);
 		}
 
-		substream.stream = SNDRV_PCM_STREAM_PLAYBACK;
+		substream->stream = SNDRV_PCM_STREAM_PLAYBACK;
 		snd_soc_dapm_widget_for_each_sink_path(w, path) {
 			sink = path->sink->priv;
-			snd_soc_dai_hw_free(sink, &substream);
+			snd_soc_dai_hw_free(sink, substream);
 		}
 
-		substream.stream = SNDRV_PCM_STREAM_CAPTURE;
+		substream->stream = SNDRV_PCM_STREAM_CAPTURE;
 		snd_soc_dapm_widget_for_each_source_path(w, path) {
 			source = path->source->priv;
 			source->active--;
-			snd_soc_dai_shutdown(source, &substream);
+			snd_soc_dai_shutdown(source, substream);
 		}
 
-		substream.stream = SNDRV_PCM_STREAM_PLAYBACK;
+		substream->stream = SNDRV_PCM_STREAM_PLAYBACK;
 		snd_soc_dapm_widget_for_each_sink_path(w, path) {
 			sink = path->sink->priv;
 			sink->active--;
-			snd_soc_dai_shutdown(sink, &substream);
+			snd_soc_dai_shutdown(sink, substream);
 		}
 		break;
 
+	case SND_SOC_DAPM_POST_PMD:
+		kfree(substream->runtime);
+		break;
+
 	default:
 		WARN(1, "Unknown event %d\n", event);
 		ret = -EINVAL;
 	}
 
 out:
-	kfree(runtime);
+	/* Restore the substream direction */
+	substream->stream = saved_stream;
 	return ret;
 }
 
@@ -4080,9 +4086,11 @@ snd_soc_dapm_alloc_kcontrol(struct snd_soc_card *card,
 }
 
 static struct snd_soc_dapm_widget *
-snd_soc_dapm_new_dai(struct snd_soc_card *card, struct snd_soc_pcm_runtime *rtd,
+snd_soc_dapm_new_dai(struct snd_soc_card *card,
+		     struct snd_pcm_substream *substream,
 		     char *id)
 {
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_dapm_widget template;
 	struct snd_soc_dapm_widget *w;
 	const char **w_param_text;
@@ -4101,7 +4109,7 @@ snd_soc_dapm_new_dai(struct snd_soc_card *card, struct snd_soc_pcm_runtime *rtd,
 	template.name = link_name;
 	template.event = snd_soc_dai_link_event;
 	template.event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
-		SND_SOC_DAPM_PRE_PMD;
+		SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD;
 	template.kcontrol_news = NULL;
 
 	/* allocate memory for control, only in case of multiple configs */
@@ -4136,7 +4144,7 @@ snd_soc_dapm_new_dai(struct snd_soc_card *card, struct snd_soc_pcm_runtime *rtd,
 		goto outfree_kcontrol_news;
 	}
 
-	w->priv = rtd;
+	w->priv = substream;
 
 	return w;
 
@@ -4258,6 +4266,8 @@ static void dapm_connect_dai_link_widgets(struct snd_soc_card *card,
 	struct snd_soc_dai *codec_dai;
 	struct snd_soc_dapm_widget *playback = NULL, *capture = NULL;
 	struct snd_soc_dapm_widget *codec, *playback_cpu, *capture_cpu;
+	struct snd_pcm_substream *substream;
+	struct snd_pcm_str *streams = rtd->pcm->streams;
 	int i;
 
 	if (rtd->dai_link->params) {
@@ -4276,7 +4286,8 @@ static void dapm_connect_dai_link_widgets(struct snd_soc_card *card,
 
 		if (playback_cpu && codec) {
 			if (!playback) {
-				playback = snd_soc_dapm_new_dai(card, rtd,
+				substream = streams[SNDRV_PCM_STREAM_PLAYBACK].substream;
+				playback = snd_soc_dapm_new_dai(card, substream,
 								"playback");
 				if (IS_ERR(playback)) {
 					dev_err(rtd->dev,
@@ -4305,7 +4316,8 @@ static void dapm_connect_dai_link_widgets(struct snd_soc_card *card,
 
 		if (codec && capture_cpu) {
 			if (!capture) {
-				capture = snd_soc_dapm_new_dai(card, rtd,
+				substream = streams[SNDRV_PCM_STREAM_CAPTURE].substream;
+				capture = snd_soc_dapm_new_dai(card, substream,
 							       "capture");
 				if (IS_ERR(capture)) {
 					dev_err(rtd->dev,
-- 
2.21.0


_______________________________________________
linux-amlogic mailing list
linux-amlogic@lists.infradead.org
http://lists.infradead.org/mailman/listinfo/linux-amlogic

[PATCH v2 6/6] ASoC: codec2codec: fill some of the runtime stream parameters

[Jerome Brunet]

Set the information provided struct snd_soc_pcm_stream in the
struct snd_pcm_runtime of the codec to codec link.

Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
---
 sound/soc/soc-dapm.c | 5 +++++
 1 file changed, 5 insertions(+)

diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 3e3b81ae87dd..fadc804be89c 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -3872,6 +3872,11 @@ snd_soc_dai_link_event_pre_pmu(struct snd_soc_dapm_widget *w,
 		dapm_update_dai_unlocked(substream, params, sink);
 	}
 
+	runtime->format = params_format(params);
+	runtime->subformat = params_subformat(params);
+	runtime->channels = params_channels(params);
+	runtime->rate = params_rate(params);
+
 out:
 	if (ret < 0)
 		kfree(runtime);
-- 
2.21.0


_______________________________________________
linux-amlogic mailing list
linux-amlogic@lists.infradead.org
http://lists.infradead.org/mailman/listinfo/linux-amlogic

Applied "ASoC: codec2codec: run callbacks in order" to the asoc tree

[Mark Brown]

The patch

   ASoC: codec2codec: run callbacks in order

has been applied to the asoc tree at

   https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-5.4

All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.  

You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.

If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.

Please add any relevant lists and maintainers to the CCs when replying
to this mail.

Thanks,
Mark

From 68c907f10cd816cad2287167a1a1d77914a6d466 Mon Sep 17 00:00:00 2001
From: Jerome Brunet <jbrunet@baylibre.com>
Date: Thu, 25 Jul 2019 18:59:44 +0200
Subject: [PATCH] ASoC: codec2codec: run callbacks in order

When handling dai_link events on codec to codec links, run all .startup()
callbacks on sinks and sources before running any .hw_params(). Same goes
for hw_free() and shutdown(). This is closer to the behavior of regular
dai links

Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Link: https://lore.kernel.org/r/20190725165949.29699-2-jbrunet@baylibre.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/soc-dapm.c | 36 +++++++++++++++++++++++++++---------
 1 file changed, 27 insertions(+), 9 deletions(-)

diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 1d04612601ad..034b31fd2ecb 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -3835,11 +3835,6 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w,
 				goto out;
 			}
 			source->active++;
-			ret = snd_soc_dai_hw_params(source, &substream, params);
-			if (ret < 0)
-				goto out;
-
-			dapm_update_dai_unlocked(&substream, params, source);
 		}
 
 		substream.stream = SNDRV_PCM_STREAM_PLAYBACK;
@@ -3853,6 +3848,23 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w,
 				goto out;
 			}
 			sink->active++;
+		}
+
+		substream.stream = SNDRV_PCM_STREAM_CAPTURE;
+		snd_soc_dapm_widget_for_each_source_path(w, path) {
+			source = path->source->priv;
+
+			ret = snd_soc_dai_hw_params(source, &substream, params);
+			if (ret < 0)
+				goto out;
+
+			dapm_update_dai_unlocked(&substream, params, source);
+		}
+
+		substream.stream = SNDRV_PCM_STREAM_PLAYBACK;
+		snd_soc_dapm_widget_for_each_sink_path(w, path) {
+			sink = path->sink->priv;
+
 			ret = snd_soc_dai_hw_params(sink, &substream, params);
 			if (ret < 0)
 				goto out;
@@ -3889,9 +3901,18 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w,
 		substream.stream = SNDRV_PCM_STREAM_CAPTURE;
 		snd_soc_dapm_widget_for_each_source_path(w, path) {
 			source = path->source->priv;
-
 			snd_soc_dai_hw_free(source, &substream);
+		}
+
+		substream.stream = SNDRV_PCM_STREAM_PLAYBACK;
+		snd_soc_dapm_widget_for_each_sink_path(w, path) {
+			sink = path->sink->priv;
+			snd_soc_dai_hw_free(sink, &substream);
+		}
 
+		substream.stream = SNDRV_PCM_STREAM_CAPTURE;
+		snd_soc_dapm_widget_for_each_source_path(w, path) {
+			source = path->source->priv;
 			source->active--;
 			snd_soc_dai_shutdown(source, &substream);
 		}
@@ -3899,9 +3920,6 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w,
 		substream.stream = SNDRV_PCM_STREAM_PLAYBACK;
 		snd_soc_dapm_widget_for_each_sink_path(w, path) {
 			sink = path->sink->priv;
-
-			snd_soc_dai_hw_free(sink, &substream);
-
 			sink->active--;
 			snd_soc_dai_shutdown(sink, &substream);
 		}
-- 
2.20.1


_______________________________________________
linux-amlogic mailing list
linux-amlogic@lists.infradead.org
http://lists.infradead.org/mailman/listinfo/linux-amlogic

Applied "ASoC: codec2codec: name link using stream direction" to the asoc tree

[Mark Brown]

The patch

   ASoC: codec2codec: name link using stream direction

has been applied to the asoc tree at

   https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-5.4

All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.  

You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.

If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.

Please add any relevant lists and maintainers to the CCs when replying
to this mail.

Thanks,
Mark

From 054d65004c6a008dfefbdae4fc1b46a3ad4e94c1 Mon Sep 17 00:00:00 2001
From: Jerome Brunet <jbrunet@baylibre.com>
Date: Thu, 25 Jul 2019 18:59:45 +0200
Subject: [PATCH] ASoC: codec2codec: name link using stream direction

At the moment, codec to codec dai link widgets are named after the
cpu dai and the 1st codec valid on the link. This might be confusing
if there is multiple valid codecs on the link for one stream
direction.

Instead, use the dai link name and the stream direction to name the
the dai link widget

Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Link: https://lore.kernel.org/r/20190725165949.29699-3-jbrunet@baylibre.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/soc-dapm.c | 12 ++++--------
 1 file changed, 4 insertions(+), 8 deletions(-)

diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 034b31fd2ecb..7db4abd9a0a5 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -4056,8 +4056,7 @@ snd_soc_dapm_alloc_kcontrol(struct snd_soc_card *card,
 
 static struct snd_soc_dapm_widget *
 snd_soc_dapm_new_dai(struct snd_soc_card *card, struct snd_soc_pcm_runtime *rtd,
-		     struct snd_soc_dapm_widget *source,
-		     struct snd_soc_dapm_widget *sink)
+		     char *id)
 {
 	struct snd_soc_dapm_widget template;
 	struct snd_soc_dapm_widget *w;
@@ -4067,7 +4066,7 @@ snd_soc_dapm_new_dai(struct snd_soc_card *card, struct snd_soc_pcm_runtime *rtd,
 	int ret;
 
 	link_name = devm_kasprintf(card->dev, GFP_KERNEL, "%s-%s",
-				   source->name, sink->name);
+				   rtd->dai_link->name, id);
 	if (!link_name)
 		return ERR_PTR(-ENOMEM);
 
@@ -4247,15 +4246,13 @@ static void dapm_connect_dai_link_widgets(struct snd_soc_card *card,
 	}
 
 	for_each_rtd_codec_dai(rtd, i, codec_dai) {
-
 		/* connect BE DAI playback if widgets are valid */
 		codec = codec_dai->playback_widget;
 
 		if (playback_cpu && codec) {
 			if (!playback) {
 				playback = snd_soc_dapm_new_dai(card, rtd,
-								playback_cpu,
-								codec);
+								"playback");
 				if (IS_ERR(playback)) {
 					dev_err(rtd->dev,
 						"ASoC: Failed to create DAI %s: %ld\n",
@@ -4284,8 +4281,7 @@ static void dapm_connect_dai_link_widgets(struct snd_soc_card *card,
 		if (codec && capture_cpu) {
 			if (!capture) {
 				capture = snd_soc_dapm_new_dai(card, rtd,
-							       codec,
-							       capture_cpu);
+							       "capture");
 				if (IS_ERR(capture)) {
 					dev_err(rtd->dev,
 						"ASoC: Failed to create DAI %s: %ld\n",
-- 
2.20.1


_______________________________________________
linux-amlogic mailing list
linux-amlogic@lists.infradead.org
http://lists.infradead.org/mailman/listinfo/linux-amlogic

Applied "ASoC: codec2codec: deal with params when necessary" to the asoc tree

[Mark Brown]

The patch

   ASoC: codec2codec: deal with params when necessary

has been applied to the asoc tree at

   https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-5.4

All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.  

You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.

If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.

Please add any relevant lists and maintainers to the CCs when replying
to this mail.

Thanks,
Mark

From 3dcfb397dad2ad55bf50de3c5d5a57090d35a18a Mon Sep 17 00:00:00 2001
From: Jerome Brunet <jbrunet@baylibre.com>
Date: Thu, 25 Jul 2019 18:59:46 +0200
Subject: [PATCH] ASoC: codec2codec: deal with params when necessary

When there is an event on codec to codec dai_link, we only need to deal
with params if the event is SND_SOC_DAPM_PRE_PMU, when .hw_params() is
called. For the other events, it is useless.

Also, dealing with the codec to codec params just before calling
.hw_params() callbacks give change to either party on the link to alter
params content in .startup(), which might be useful in some cases

Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Link: https://lore.kernel.org/r/20190725165949.29699-4-jbrunet@baylibre.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/soc-dapm.c | 159 +++++++++++++++++++++++++------------------
 1 file changed, 92 insertions(+), 67 deletions(-)

diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 7db4abd9a0a5..6dcaf9ff6eb5 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -3764,25 +3764,59 @@ int snd_soc_dapm_new_controls(struct snd_soc_dapm_context *dapm,
 }
 EXPORT_SYMBOL_GPL(snd_soc_dapm_new_controls);
 
-static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w,
-				  struct snd_kcontrol *kcontrol, int event)
+static int
+snd_soc_dai_link_event_pre_pmu(struct snd_soc_dapm_widget *w,
+			       struct snd_pcm_substream *substream)
 {
 	struct snd_soc_dapm_path *path;
 	struct snd_soc_dai *source, *sink;
-	struct snd_soc_pcm_runtime *rtd = w->priv;
-	const struct snd_soc_pcm_stream *config;
-	struct snd_pcm_substream substream;
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_pcm_hw_params *params = NULL;
-	struct snd_pcm_runtime *runtime = NULL;
+	const struct snd_soc_pcm_stream *config = NULL;
 	unsigned int fmt;
-	int ret = 0;
+	int ret;
 
-	config = rtd->dai_link->params + rtd->params_select;
+	params = kzalloc(sizeof(*params), GFP_KERNEL);
+	if (!params)
+		return -ENOMEM;
 
-	if (WARN_ON(!config) ||
-	    WARN_ON(list_empty(&w->edges[SND_SOC_DAPM_DIR_OUT]) ||
-		    list_empty(&w->edges[SND_SOC_DAPM_DIR_IN])))
-		return -EINVAL;
+	substream->stream = SNDRV_PCM_STREAM_CAPTURE;
+	snd_soc_dapm_widget_for_each_source_path(w, path) {
+		source = path->source->priv;
+
+		ret = snd_soc_dai_startup(source, substream);
+		if (ret < 0) {
+			dev_err(source->dev,
+				"ASoC: startup() failed: %d\n", ret);
+			goto out;
+		}
+		source->active++;
+	}
+
+	substream->stream = SNDRV_PCM_STREAM_PLAYBACK;
+	snd_soc_dapm_widget_for_each_sink_path(w, path) {
+		sink = path->sink->priv;
+
+		ret = snd_soc_dai_startup(sink, substream);
+		if (ret < 0) {
+			dev_err(sink->dev,
+				"ASoC: startup() failed: %d\n", ret);
+			goto out;
+		}
+		sink->active++;
+	}
+
+	/*
+	 * Note: getting the config after .startup() gives a chance to
+	 * either party on the link to alter the configuration if
+	 * necessary
+	 */
+	config = rtd->dai_link->params + rtd->params_select;
+	if (WARN_ON(!config)) {
+		dev_err(w->dapm->dev, "ASoC: link config missing\n");
+		ret = -EINVAL;
+		goto out;
+	}
 
 	/* Be a little careful as we don't want to overflow the mask array */
 	if (config->formats) {
@@ -3790,27 +3824,62 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w,
 	} else {
 		dev_warn(w->dapm->dev, "ASoC: Invalid format %llx specified\n",
 			 config->formats);
-		fmt = 0;
-	}
 
-	/* Currently very limited parameter selection */
-	params = kzalloc(sizeof(*params), GFP_KERNEL);
-	if (!params) {
-		ret = -ENOMEM;
+		ret = -EINVAL;
 		goto out;
 	}
-	snd_mask_set(hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT), fmt);
 
+	snd_mask_set(hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT), fmt);
 	hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE)->min =
 		config->rate_min;
 	hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE)->max =
 		config->rate_max;
-
 	hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS)->min
 		= config->channels_min;
 	hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS)->max
 		= config->channels_max;
 
+	substream->stream = SNDRV_PCM_STREAM_CAPTURE;
+	snd_soc_dapm_widget_for_each_source_path(w, path) {
+		source = path->source->priv;
+
+		ret = snd_soc_dai_hw_params(source, substream, params);
+		if (ret < 0)
+			goto out;
+
+		dapm_update_dai_unlocked(substream, params, source);
+	}
+
+	substream->stream = SNDRV_PCM_STREAM_PLAYBACK;
+	snd_soc_dapm_widget_for_each_sink_path(w, path) {
+		sink = path->sink->priv;
+
+		ret = snd_soc_dai_hw_params(sink, substream, params);
+		if (ret < 0)
+			goto out;
+
+		dapm_update_dai_unlocked(substream, params, sink);
+	}
+
+out:
+	kfree(params);
+	return 0;
+}
+
+static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w,
+				  struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_dapm_path *path;
+	struct snd_soc_dai *source, *sink;
+	struct snd_soc_pcm_runtime *rtd = w->priv;
+	struct snd_pcm_substream substream;
+	struct snd_pcm_runtime *runtime = NULL;
+	int ret = 0;
+
+	if (WARN_ON(list_empty(&w->edges[SND_SOC_DAPM_DIR_OUT]) ||
+		    list_empty(&w->edges[SND_SOC_DAPM_DIR_IN])))
+		return -EINVAL;
+
 	memset(&substream, 0, sizeof(substream));
 
 	/* Allocate a dummy snd_pcm_runtime for startup() and other ops() */
@@ -3824,53 +3893,10 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w,
 
 	switch (event) {
 	case SND_SOC_DAPM_PRE_PMU:
-		substream.stream = SNDRV_PCM_STREAM_CAPTURE;
-		snd_soc_dapm_widget_for_each_source_path(w, path) {
-			source = path->source->priv;
-
-			ret = snd_soc_dai_startup(source, &substream);
-			if (ret < 0) {
-				dev_err(source->dev,
-					"ASoC: startup() failed: %d\n", ret);
-				goto out;
-			}
-			source->active++;
-		}
-
-		substream.stream = SNDRV_PCM_STREAM_PLAYBACK;
-		snd_soc_dapm_widget_for_each_sink_path(w, path) {
-			sink = path->sink->priv;
-
-			ret = snd_soc_dai_startup(sink, &substream);
-			if (ret < 0) {
-				dev_err(sink->dev,
-					"ASoC: startup() failed: %d\n", ret);
-				goto out;
-			}
-			sink->active++;
-		}
-
-		substream.stream = SNDRV_PCM_STREAM_CAPTURE;
-		snd_soc_dapm_widget_for_each_source_path(w, path) {
-			source = path->source->priv;
-
-			ret = snd_soc_dai_hw_params(source, &substream, params);
-			if (ret < 0)
-				goto out;
-
-			dapm_update_dai_unlocked(&substream, params, source);
-		}
-
-		substream.stream = SNDRV_PCM_STREAM_PLAYBACK;
-		snd_soc_dapm_widget_for_each_sink_path(w, path) {
-			sink = path->sink->priv;
-
-			ret = snd_soc_dai_hw_params(sink, &substream, params);
-			if (ret < 0)
-				goto out;
+		ret = snd_soc_dai_link_event_pre_pmu(w, &substream);
+		if (ret < 0)
+			goto out;
 
-			dapm_update_dai_unlocked(&substream, params, sink);
-		}
 		break;
 
 	case SND_SOC_DAPM_POST_PMU:
@@ -3932,7 +3958,6 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w,
 
 out:
 	kfree(runtime);
-	kfree(params);
 	return ret;
 }
 
-- 
2.20.1


_______________________________________________
linux-amlogic mailing list
linux-amlogic@lists.infradead.org
http://lists.infradead.org/mailman/listinfo/linux-amlogic

Applied "ASoC: create pcm for codec2codec links as well" to the asoc tree

[Mark Brown]

The patch

   ASoC: create pcm for codec2codec links as well

has been applied to the asoc tree at

   https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-5.4

All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.  

You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.

If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.

Please add any relevant lists and maintainers to the CCs when replying
to this mail.

Thanks,
Mark

From a342031cdd0818cb0fbcb44798211c7a02c7ca27 Mon Sep 17 00:00:00 2001
From: Jerome Brunet <jbrunet@baylibre.com>
Date: Thu, 25 Jul 2019 18:59:47 +0200
Subject: [PATCH] ASoC: create pcm for codec2codec links as well

At the moment, codec to codec links uses an ephemeral variable for
the struct snd_pcm_substream. Also the struct snd_soc_pcm_runtime
does not have real struct snd_pcm.

This might a problem if the functions used by a codec on codec to
codec link expect these structures to exist, and keep on existing
during the life of the codec.

For example, it is the case of the hdmi-codec, which uses
snd_pcm_add_chmap_ctls(). For the controls to works, the pcm and
substream must to exist.

This change is first step, it create pcm (and substreams) for codec
to codec links, in the same way as dpcm backend links.

Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Link: https://lore.kernel.org/r/20190725165949.29699-5-jbrunet@baylibre.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/soc-core.c | 42 ++++++++++++------------------------------
 sound/soc/soc-pcm.c  | 35 ++++++++++++++++++++++++++++++++---
 2 files changed, 44 insertions(+), 33 deletions(-)

diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 06697b2d96b1..da11e44b01aa 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -447,16 +447,6 @@ static void snd_soc_flush_all_delayed_work(struct snd_soc_card *card)
 		flush_delayed_work(&rtd->delayed_work);
 }
 
-static void codec2codec_close_delayed_work(struct work_struct *work)
-{
-	/*
-	 * Currently nothing to do for c2c links
-	 * Since c2c links are internal nodes in the DAPM graph and
-	 * don't interface with the outside world or application layer
-	 * we don't have to do any special handling on close.
-	 */
-}
-
 #ifdef CONFIG_PM_SLEEP
 /* powers down audio subsystem for suspend */
 int snd_soc_suspend(struct device *dev)
@@ -1555,27 +1545,19 @@ static int soc_probe_link_dais(struct snd_soc_card *card,
 		return ret;
 	}
 
-	if (!dai_link->params) {
-		/* create the pcm */
-		ret = soc_new_pcm(rtd, num);
-		if (ret < 0) {
-			dev_err(card->dev, "ASoC: can't create pcm %s :%d\n",
-				dai_link->stream_name, ret);
-			return ret;
-		}
-		ret = soc_link_dai_pcm_new(&cpu_dai, 1, rtd);
-		if (ret < 0)
-			return ret;
-		ret = soc_link_dai_pcm_new(rtd->codec_dais,
-					   rtd->num_codecs, rtd);
-		if (ret < 0)
-			return ret;
-	} else {
-		INIT_DELAYED_WORK(&rtd->delayed_work,
-				  codec2codec_close_delayed_work);
+	/* create the pcm */
+	ret = soc_new_pcm(rtd, num);
+	if (ret < 0) {
+		dev_err(card->dev, "ASoC: can't create pcm %s :%d\n",
+			dai_link->stream_name, ret);
+		return ret;
 	}
-
-	return 0;
+	ret = soc_link_dai_pcm_new(&cpu_dai, 1, rtd);
+	if (ret < 0)
+		return ret;
+	ret = soc_link_dai_pcm_new(rtd->codec_dais,
+				   rtd->num_codecs, rtd);
+	return ret;
 }
 
 static int soc_bind_aux_dev(struct snd_soc_card *card, int num)
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index fabeac164a6c..30264bc592f6 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -678,6 +678,16 @@ static void close_delayed_work(struct work_struct *work)
 	mutex_unlock(&rtd->pcm_mutex);
 }
 
+static void codec2codec_close_delayed_work(struct work_struct *work)
+{
+	/*
+	 * Currently nothing to do for c2c links
+	 * Since c2c links are internal nodes in the DAPM graph and
+	 * don't interface with the outside world or application layer
+	 * we don't have to do any special handling on close.
+	 */
+}
+
 /*
  * Called by ALSA when a PCM substream is closed. Private data can be
  * freed here. The cpu DAI, codec DAI, machine and components are also
@@ -3011,6 +3021,12 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num)
 		playback = rtd->dai_link->dpcm_playback;
 		capture = rtd->dai_link->dpcm_capture;
 	} else {
+		/* Adapt stream for codec2codec links */
+		struct snd_soc_pcm_stream *cpu_capture = rtd->dai_link->params ?
+			&cpu_dai->driver->playback : &cpu_dai->driver->capture;
+		struct snd_soc_pcm_stream *cpu_playback = rtd->dai_link->params ?
+			&cpu_dai->driver->capture : &cpu_dai->driver->playback;
+
 		for_each_rtd_codec_dai(rtd, i, codec_dai) {
 			if (snd_soc_dai_stream_valid(codec_dai, SNDRV_PCM_STREAM_PLAYBACK) &&
 			    snd_soc_dai_stream_valid(cpu_dai,   SNDRV_PCM_STREAM_PLAYBACK))
@@ -3019,6 +3035,9 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num)
 			    snd_soc_dai_stream_valid(cpu_dai,   SNDRV_PCM_STREAM_CAPTURE))
 				capture = 1;
 		}
+
+		capture = capture && cpu_capture->channels_min;
+		playback = playback && cpu_playback->channels_min;
 	}
 
 	if (rtd->dai_link->playback_only) {
@@ -3032,7 +3051,13 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num)
 	}
 
 	/* create the PCM */
-	if (rtd->dai_link->no_pcm) {
+	if (rtd->dai_link->params) {
+		snprintf(new_name, sizeof(new_name), "codec2codec(%s)",
+			 rtd->dai_link->stream_name);
+
+		ret = snd_pcm_new_internal(rtd->card->snd_card, new_name, num,
+					   playback, capture, &pcm);
+	} else if (rtd->dai_link->no_pcm) {
 		snprintf(new_name, sizeof(new_name), "(%s)",
 			rtd->dai_link->stream_name);
 
@@ -3059,13 +3084,17 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num)
 	dev_dbg(rtd->card->dev, "ASoC: registered pcm #%d %s\n",num, new_name);
 
 	/* DAPM dai link stream work */
-	INIT_DELAYED_WORK(&rtd->delayed_work, close_delayed_work);
+	if (rtd->dai_link->params)
+		INIT_DELAYED_WORK(&rtd->delayed_work,
+				  codec2codec_close_delayed_work);
+	else
+		INIT_DELAYED_WORK(&rtd->delayed_work, close_delayed_work);
 
 	pcm->nonatomic = rtd->dai_link->nonatomic;
 	rtd->pcm = pcm;
 	pcm->private_data = rtd;
 
-	if (rtd->dai_link->no_pcm) {
+	if (rtd->dai_link->no_pcm || rtd->dai_link->params) {
 		if (playback)
 			pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream->private_data = rtd;
 		if (capture)
-- 
2.20.1


_______________________________________________
linux-amlogic mailing list
linux-amlogic@lists.infradead.org
http://lists.infradead.org/mailman/listinfo/linux-amlogic

Applied "ASoC: codec2codec: fill some of the runtime stream parameters" to the asoc tree

[Mark Brown]

The patch

   ASoC: codec2codec: fill some of the runtime stream parameters

has been applied to the asoc tree at

   https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-5.4

All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.  

You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.

If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.

Please add any relevant lists and maintainers to the CCs when replying
to this mail.

Thanks,
Mark

From 9de98628c895d15427138073986eab1e3ce39cb4 Mon Sep 17 00:00:00 2001
From: Jerome Brunet <jbrunet@baylibre.com>
Date: Thu, 25 Jul 2019 18:59:49 +0200
Subject: [PATCH] ASoC: codec2codec: fill some of the runtime stream parameters

Set the information provided struct snd_soc_pcm_stream in the
struct snd_pcm_runtime of the codec to codec link.

Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Link: https://lore.kernel.org/r/20190725165949.29699-7-jbrunet@baylibre.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/soc-dapm.c | 5 +++++
 1 file changed, 5 insertions(+)

diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 1c953a1b46ce..e16838e1bda2 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -3874,6 +3874,11 @@ snd_soc_dai_link_event_pre_pmu(struct snd_soc_dapm_widget *w,
 		dapm_update_dai_unlocked(substream, params, sink);
 	}
 
+	runtime->format = params_format(params);
+	runtime->subformat = params_subformat(params);
+	runtime->channels = params_channels(params);
+	runtime->rate = params_rate(params);
+
 out:
 	if (ret < 0)
 		kfree(runtime);
-- 
2.20.1


_______________________________________________
linux-amlogic mailing list
linux-amlogic@lists.infradead.org
http://lists.infradead.org/mailman/listinfo/linux-amlogic

Applied "ASoC: codec2codec: remove ephemeral variables" to the asoc tree

[Mark Brown]

The patch

   ASoC: codec2codec: remove ephemeral variables

has been applied to the asoc tree at

   https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-5.4

All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.  

You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.

If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.

Please add any relevant lists and maintainers to the CCs when replying
to this mail.

Thanks,
Mark

From a72706ed8208ac3f72d1c3ebbc6509e368b0dcb0 Mon Sep 17 00:00:00 2001
From: Jerome Brunet <jbrunet@baylibre.com>
Date: Thu, 25 Jul 2019 18:59:48 +0200
Subject: [PATCH] ASoC: codec2codec: remove ephemeral variables

Now that codec to codec links struct snd_soc_pcm_runtime have lasting pcm
and substreams, let's use them. Alsa allocate and keep the
struct snd_pcm_runtime as long as the link is powered.

Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Link: https://lore.kernel.org/r/20190725165949.29699-6-jbrunet@baylibre.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/soc-dapm.c | 72 ++++++++++++++++++++++++++------------------
 1 file changed, 42 insertions(+), 30 deletions(-)

diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 2d183e2d23de..1c953a1b46ce 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -3775,6 +3775,7 @@ snd_soc_dai_link_event_pre_pmu(struct snd_soc_dapm_widget *w,
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_pcm_hw_params *params = NULL;
 	const struct snd_soc_pcm_stream *config = NULL;
+	struct snd_pcm_runtime *runtime = NULL;
 	unsigned int fmt;
 	int ret = 0;
 
@@ -3782,6 +3783,14 @@ snd_soc_dai_link_event_pre_pmu(struct snd_soc_dapm_widget *w,
 	if (!params)
 		return -ENOMEM;
 
+	runtime = kzalloc(sizeof(*runtime), GFP_KERNEL);
+	if (!runtime) {
+		ret = -ENOMEM;
+		goto out;
+	}
+
+	substream->runtime = runtime;
+
 	substream->stream = SNDRV_PCM_STREAM_CAPTURE;
 	snd_soc_dapm_widget_for_each_source_path(w, path) {
 		source = path->source->priv;
@@ -3808,6 +3817,8 @@ snd_soc_dai_link_event_pre_pmu(struct snd_soc_dapm_widget *w,
 		sink->active++;
 	}
 
+	substream->hw_opened = 1;
+
 	/*
 	 * Note: getting the config after .startup() gives a chance to
 	 * either party on the link to alter the configuration if
@@ -3864,6 +3875,9 @@ snd_soc_dai_link_event_pre_pmu(struct snd_soc_dapm_widget *w,
 	}
 
 out:
+	if (ret < 0)
+		kfree(runtime);
+
 	kfree(params);
 	return ret;
 }
@@ -3873,29 +3887,16 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w,
 {
 	struct snd_soc_dapm_path *path;
 	struct snd_soc_dai *source, *sink;
-	struct snd_soc_pcm_runtime *rtd = w->priv;
-	struct snd_pcm_substream substream;
-	struct snd_pcm_runtime *runtime = NULL;
-	int ret = 0;
+	struct snd_pcm_substream *substream = w->priv;
+	int ret = 0, saved_stream = substream->stream;
 
 	if (WARN_ON(list_empty(&w->edges[SND_SOC_DAPM_DIR_OUT]) ||
 		    list_empty(&w->edges[SND_SOC_DAPM_DIR_IN])))
 		return -EINVAL;
 
-	memset(&substream, 0, sizeof(substream));
-
-	/* Allocate a dummy snd_pcm_runtime for startup() and other ops() */
-	runtime = kzalloc(sizeof(*runtime), GFP_KERNEL);
-	if (!runtime) {
-		ret = -ENOMEM;
-		goto out;
-	}
-	substream.runtime = runtime;
-	substream.private_data = rtd;
-
 	switch (event) {
 	case SND_SOC_DAPM_PRE_PMU:
-		ret = snd_soc_dai_link_event_pre_pmu(w, &substream);
+		ret = snd_soc_dai_link_event_pre_pmu(w, substream);
 		if (ret < 0)
 			goto out;
 
@@ -3926,40 +3927,45 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w,
 			ret = 0;
 		}
 
-		substream.stream = SNDRV_PCM_STREAM_CAPTURE;
+		substream->stream = SNDRV_PCM_STREAM_CAPTURE;
 		snd_soc_dapm_widget_for_each_source_path(w, path) {
 			source = path->source->priv;
-			snd_soc_dai_hw_free(source, &substream);
+			snd_soc_dai_hw_free(source, substream);
 		}
 
-		substream.stream = SNDRV_PCM_STREAM_PLAYBACK;
+		substream->stream = SNDRV_PCM_STREAM_PLAYBACK;
 		snd_soc_dapm_widget_for_each_sink_path(w, path) {
 			sink = path->sink->priv;
-			snd_soc_dai_hw_free(sink, &substream);
+			snd_soc_dai_hw_free(sink, substream);
 		}
 
-		substream.stream = SNDRV_PCM_STREAM_CAPTURE;
+		substream->stream = SNDRV_PCM_STREAM_CAPTURE;
 		snd_soc_dapm_widget_for_each_source_path(w, path) {
 			source = path->source->priv;
 			source->active--;
-			snd_soc_dai_shutdown(source, &substream);
+			snd_soc_dai_shutdown(source, substream);
 		}
 
-		substream.stream = SNDRV_PCM_STREAM_PLAYBACK;
+		substream->stream = SNDRV_PCM_STREAM_PLAYBACK;
 		snd_soc_dapm_widget_for_each_sink_path(w, path) {
 			sink = path->sink->priv;
 			sink->active--;
-			snd_soc_dai_shutdown(sink, &substream);
+			snd_soc_dai_shutdown(sink, substream);
 		}
 		break;
 
+	case SND_SOC_DAPM_POST_PMD:
+		kfree(substream->runtime);
+		break;
+
 	default:
 		WARN(1, "Unknown event %d\n", event);
 		ret = -EINVAL;
 	}
 
 out:
-	kfree(runtime);
+	/* Restore the substream direction */
+	substream->stream = saved_stream;
 	return ret;
 }
 
@@ -4082,9 +4088,11 @@ snd_soc_dapm_alloc_kcontrol(struct snd_soc_card *card,
 }
 
 static struct snd_soc_dapm_widget *
-snd_soc_dapm_new_dai(struct snd_soc_card *card, struct snd_soc_pcm_runtime *rtd,
+snd_soc_dapm_new_dai(struct snd_soc_card *card,
+		     struct snd_pcm_substream *substream,
 		     char *id)
 {
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_dapm_widget template;
 	struct snd_soc_dapm_widget *w;
 	const char **w_param_text;
@@ -4103,7 +4111,7 @@ snd_soc_dapm_new_dai(struct snd_soc_card *card, struct snd_soc_pcm_runtime *rtd,
 	template.name = link_name;
 	template.event = snd_soc_dai_link_event;
 	template.event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
-		SND_SOC_DAPM_PRE_PMD;
+		SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD;
 	template.kcontrol_news = NULL;
 
 	/* allocate memory for control, only in case of multiple configs */
@@ -4138,7 +4146,7 @@ snd_soc_dapm_new_dai(struct snd_soc_card *card, struct snd_soc_pcm_runtime *rtd,
 		goto outfree_kcontrol_news;
 	}
 
-	w->priv = rtd;
+	w->priv = substream;
 
 	return w;
 
@@ -4260,6 +4268,8 @@ static void dapm_connect_dai_link_widgets(struct snd_soc_card *card,
 	struct snd_soc_dai *codec_dai;
 	struct snd_soc_dapm_widget *playback = NULL, *capture = NULL;
 	struct snd_soc_dapm_widget *codec, *playback_cpu, *capture_cpu;
+	struct snd_pcm_substream *substream;
+	struct snd_pcm_str *streams = rtd->pcm->streams;
 	int i;
 
 	if (rtd->dai_link->params) {
@@ -4278,7 +4288,8 @@ static void dapm_connect_dai_link_widgets(struct snd_soc_card *card,
 
 		if (playback_cpu && codec) {
 			if (!playback) {
-				playback = snd_soc_dapm_new_dai(card, rtd,
+				substream = streams[SNDRV_PCM_STREAM_PLAYBACK].substream;
+				playback = snd_soc_dapm_new_dai(card, substream,
 								"playback");
 				if (IS_ERR(playback)) {
 					dev_err(rtd->dev,
@@ -4307,7 +4318,8 @@ static void dapm_connect_dai_link_widgets(struct snd_soc_card *card,
 
 		if (codec && capture_cpu) {
 			if (!capture) {
-				capture = snd_soc_dapm_new_dai(card, rtd,
+				substream = streams[SNDRV_PCM_STREAM_CAPTURE].substream;
+				capture = snd_soc_dapm_new_dai(card, substream,
 							       "capture");
 				if (IS_ERR(capture)) {
 					dev_err(rtd->dev,
-- 
2.20.1


_______________________________________________
linux-amlogic mailing list
linux-amlogic@lists.infradead.org
http://lists.infradead.org/mailman/listinfo/linux-amlogic