[PATCH v4 0/3] simple-audio-card codec2codec support

[PATCH v4 0/3] simple-audio-card codec2codec support

[Samuel Holland]

We are currently using simple-audio-card on the Allwinner A64 SoC.
The digital audio codec there (sun8i-codec) has 3 AIFs, one each for the
CPU, the modem, and Bluetooth. Adding support for the secondary AIFs
requires adding codec2codec DAI links.

Since the modem and bt-sco codec DAI drivers only have one set of
possible PCM parameters (namely, 8kHz mono S16LE), there's no real
need for a machine driver to specify the DAI link configuration. The
parameters for these "simple" DAI links can be chosen automatically.

This series adds codec2codec DAI link support to simple-audio-card.
Codec to codec links are automatically detected when all DAIs in the
link belong to codec components.

I tried to reuse as much code as possible, so the first two patches
refactor a couple of helper functions to be more generic.

The last patch adds the new feature and its documentation.

Changes in v4:
  - Rebased on top of asoc/for-next, several changes to patch 2
  - Removed unused variable from patch 3
Changes in v3:
  - Update use of for_each_rtd_components for v5.6
Changes in v2:
  - Drop patch 1 as it was merged
  - Automatically detect codec2codec links instead of using a DT property

Samuel Holland (3):
  ALSA: pcm: Add a standalone version of snd_pcm_limit_hw_rates
  ASoC: pcm: Export parameter intersection logic
  ASoC: simple-card: Add support for codec2codec DAI links

 Documentation/sound/soc/codec-to-codec.rst |  9 +++-
 include/sound/pcm.h                        |  9 +++-
 include/sound/soc.h                        |  3 ++
 sound/core/pcm_misc.c                      | 18 +++----
 sound/soc/generic/simple-card-utils.c      | 48 ++++++++++++++++++
 sound/soc/soc-pcm.c                        | 59 ++++++++++++++--------
 6 files changed, 114 insertions(+), 32 deletions(-)

-- 
2.24.1

[PATCH v4 0/3] simple-audio-card codec2codec support

[Samuel Holland]

We are currently using simple-audio-card on the Allwinner A64 SoC.
The digital audio codec there (sun8i-codec) has 3 AIFs, one each for the
CPU, the modem, and Bluetooth. Adding support for the secondary AIFs
requires adding codec2codec DAI links.

Since the modem and bt-sco codec DAI drivers only have one set of
possible PCM parameters (namely, 8kHz mono S16LE), there's no real
need for a machine driver to specify the DAI link configuration. The
parameters for these "simple" DAI links can be chosen automatically.

This series adds codec2codec DAI link support to simple-audio-card.
Codec to codec links are automatically detected when all DAIs in the
link belong to codec components.

I tried to reuse as much code as possible, so the first two patches
refactor a couple of helper functions to be more generic.

The last patch adds the new feature and its documentation.

Changes in v4:
  - Rebased on top of asoc/for-next, several changes to patch 2
  - Removed unused variable from patch 3
Changes in v3:
  - Update use of for_each_rtd_components for v5.6
Changes in v2:
  - Drop patch 1 as it was merged
  - Automatically detect codec2codec links instead of using a DT property

Samuel Holland (3):
  ALSA: pcm: Add a standalone version of snd_pcm_limit_hw_rates
  ASoC: pcm: Export parameter intersection logic
  ASoC: simple-card: Add support for codec2codec DAI links

 Documentation/sound/soc/codec-to-codec.rst |  9 +++-
 include/sound/pcm.h                        |  9 +++-
 include/sound/soc.h                        |  3 ++
 sound/core/pcm_misc.c                      | 18 +++----
 sound/soc/generic/simple-card-utils.c      | 48 ++++++++++++++++++
 sound/soc/soc-pcm.c                        | 59 ++++++++++++++--------
 6 files changed, 114 insertions(+), 32 deletions(-)

-- 
2.24.1

[PATCH v4 1/3] ALSA: pcm: Add a standalone version of snd_pcm_limit_hw_rates

[Samuel Holland]

It can be useful to derive min/max rates of a snd_pcm_hardware without
having a snd_pcm_runtime, such as before constructing an ASoC DAI link.

Create a new helper that takes a pointer to a snd_pcm_hardware directly,
and refactor the original function as a wrapper around it, to avoid
needing to update any call sites.

Signed-off-by: Samuel Holland <samuel@sholland.org>
---
 include/sound/pcm.h   |  9 ++++++++-
 sound/core/pcm_misc.c | 18 +++++++++---------
 2 files changed, 17 insertions(+), 10 deletions(-)

diff --git a/include/sound/pcm.h b/include/sound/pcm.h
index 2628246b76fa..f7a95b711100 100644
--- a/include/sound/pcm.h
+++ b/include/sound/pcm.h
@@ -1127,7 +1127,14 @@ snd_pcm_kernel_readv(struct snd_pcm_substream *substream,
 	return __snd_pcm_lib_xfer(substream, bufs, false, frames, true);
 }
 
-int snd_pcm_limit_hw_rates(struct snd_pcm_runtime *runtime);
+int snd_pcm_hw_limit_rates(struct snd_pcm_hardware *hw);
+
+static inline int
+snd_pcm_limit_hw_rates(struct snd_pcm_runtime *runtime)
+{
+	return snd_pcm_hw_limit_rates(&runtime->hw);
+}
+
 unsigned int snd_pcm_rate_to_rate_bit(unsigned int rate);
 unsigned int snd_pcm_rate_bit_to_rate(unsigned int rate_bit);
 unsigned int snd_pcm_rate_mask_intersect(unsigned int rates_a,
diff --git a/sound/core/pcm_misc.c b/sound/core/pcm_misc.c
index a6a541511534..5dd2e5335900 100644
--- a/sound/core/pcm_misc.c
+++ b/sound/core/pcm_misc.c
@@ -474,32 +474,32 @@ int snd_pcm_format_set_silence(snd_pcm_format_t format, void *data, unsigned int
 EXPORT_SYMBOL(snd_pcm_format_set_silence);
 
 /**
- * snd_pcm_limit_hw_rates - determine rate_min/rate_max fields
- * @runtime: the runtime instance
+ * snd_pcm_hw_limit_rates - determine rate_min/rate_max fields
+ * @hw: the pcm hw instance
  *
  * Determines the rate_min and rate_max fields from the rates bits of
- * the given runtime->hw.
+ * the given hw.
  *
  * Return: Zero if successful.
  */
-int snd_pcm_limit_hw_rates(struct snd_pcm_runtime *runtime)
+int snd_pcm_hw_limit_rates(struct snd_pcm_hardware *hw)
 {
 	int i;
 	for (i = 0; i < (int)snd_pcm_known_rates.count; i++) {
-		if (runtime->hw.rates & (1 << i)) {
-			runtime->hw.rate_min = snd_pcm_known_rates.list[i];
+		if (hw->rates & (1 << i)) {
+			hw->rate_min = snd_pcm_known_rates.list[i];
 			break;
 		}
 	}
 	for (i = (int)snd_pcm_known_rates.count - 1; i >= 0; i--) {
-		if (runtime->hw.rates & (1 << i)) {
-			runtime->hw.rate_max = snd_pcm_known_rates.list[i];
+		if (hw->rates & (1 << i)) {
+			hw->rate_max = snd_pcm_known_rates.list[i];
 			break;
 		}
 	}
 	return 0;
 }
-EXPORT_SYMBOL(snd_pcm_limit_hw_rates);
+EXPORT_SYMBOL(snd_pcm_hw_limit_rates);
 
 /**
  * snd_pcm_rate_to_rate_bit - converts sample rate to SNDRV_PCM_RATE_xxx bit
-- 
2.24.1

[PATCH v4 1/3] ALSA: pcm: Add a standalone version of snd_pcm_limit_hw_rates

[Samuel Holland]

It can be useful to derive min/max rates of a snd_pcm_hardware without
having a snd_pcm_runtime, such as before constructing an ASoC DAI link.

Create a new helper that takes a pointer to a snd_pcm_hardware directly,
and refactor the original function as a wrapper around it, to avoid
needing to update any call sites.

Signed-off-by: Samuel Holland <samuel@sholland.org>
---
 include/sound/pcm.h   |  9 ++++++++-
 sound/core/pcm_misc.c | 18 +++++++++---------
 2 files changed, 17 insertions(+), 10 deletions(-)

diff --git a/include/sound/pcm.h b/include/sound/pcm.h
index 2628246b76fa..f7a95b711100 100644
--- a/include/sound/pcm.h
+++ b/include/sound/pcm.h
@@ -1127,7 +1127,14 @@ snd_pcm_kernel_readv(struct snd_pcm_substream *substream,
 	return __snd_pcm_lib_xfer(substream, bufs, false, frames, true);
 }
 
-int snd_pcm_limit_hw_rates(struct snd_pcm_runtime *runtime);
+int snd_pcm_hw_limit_rates(struct snd_pcm_hardware *hw);
+
+static inline int
+snd_pcm_limit_hw_rates(struct snd_pcm_runtime *runtime)
+{
+	return snd_pcm_hw_limit_rates(&runtime->hw);
+}
+
 unsigned int snd_pcm_rate_to_rate_bit(unsigned int rate);
 unsigned int snd_pcm_rate_bit_to_rate(unsigned int rate_bit);
 unsigned int snd_pcm_rate_mask_intersect(unsigned int rates_a,
diff --git a/sound/core/pcm_misc.c b/sound/core/pcm_misc.c
index a6a541511534..5dd2e5335900 100644
--- a/sound/core/pcm_misc.c
+++ b/sound/core/pcm_misc.c
@@ -474,32 +474,32 @@ int snd_pcm_format_set_silence(snd_pcm_format_t format, void *data, unsigned int
 EXPORT_SYMBOL(snd_pcm_format_set_silence);
 
 /**
- * snd_pcm_limit_hw_rates - determine rate_min/rate_max fields
- * @runtime: the runtime instance
+ * snd_pcm_hw_limit_rates - determine rate_min/rate_max fields
+ * @hw: the pcm hw instance
  *
  * Determines the rate_min and rate_max fields from the rates bits of
- * the given runtime->hw.
+ * the given hw.
  *
  * Return: Zero if successful.
  */
-int snd_pcm_limit_hw_rates(struct snd_pcm_runtime *runtime)
+int snd_pcm_hw_limit_rates(struct snd_pcm_hardware *hw)
 {
 	int i;
 	for (i = 0; i < (int)snd_pcm_known_rates.count; i++) {
-		if (runtime->hw.rates & (1 << i)) {
-			runtime->hw.rate_min = snd_pcm_known_rates.list[i];
+		if (hw->rates & (1 << i)) {
+			hw->rate_min = snd_pcm_known_rates.list[i];
 			break;
 		}
 	}
 	for (i = (int)snd_pcm_known_rates.count - 1; i >= 0; i--) {
-		if (runtime->hw.rates & (1 << i)) {
-			runtime->hw.rate_max = snd_pcm_known_rates.list[i];
+		if (hw->rates & (1 << i)) {
+			hw->rate_max = snd_pcm_known_rates.list[i];
 			break;
 		}
 	}
 	return 0;
 }
-EXPORT_SYMBOL(snd_pcm_limit_hw_rates);
+EXPORT_SYMBOL(snd_pcm_hw_limit_rates);
 
 /**
  * snd_pcm_rate_to_rate_bit - converts sample rate to SNDRV_PCM_RATE_xxx bit
-- 
2.24.1

[PATCH v4 2/3] ASoC: pcm: Export parameter intersection logic

[Samuel Holland]

The logic to calculate the subset of stream parameters supported by all
DAIs associated with a PCM stream is nontrivial. Export a helper
function so it can be used to set up simple codec2codec DAI links.

Signed-off-by: Samuel Holland <samuel@sholland.org>
---
 include/sound/soc.h |  3 +++
 sound/soc/soc-pcm.c | 59 ++++++++++++++++++++++++++++++---------------
 2 files changed, 42 insertions(+), 20 deletions(-)

diff --git a/include/sound/soc.h b/include/sound/soc.h
index 81e5d17be935..9543d9246ca4 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -471,6 +471,9 @@ bool snd_soc_runtime_ignore_pmdown_time(struct snd_soc_pcm_runtime *rtd);
 void snd_soc_runtime_activate(struct snd_soc_pcm_runtime *rtd, int stream);
 void snd_soc_runtime_deactivate(struct snd_soc_pcm_runtime *rtd, int stream);
 
+int snd_soc_runtime_calc_hw(struct snd_soc_pcm_runtime *rtd,
+			    struct snd_pcm_hardware *hw, int stream);
+
 int snd_soc_runtime_set_dai_fmt(struct snd_soc_pcm_runtime *rtd,
 	unsigned int dai_fmt);
 
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index 90857138c823..2ad501aaa4f9 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -587,11 +587,18 @@ static void soc_pcm_apply_msb(struct snd_pcm_substream *substream)
 	soc_pcm_set_msb(substream, cpu_bits);
 }
 
-static void soc_pcm_init_runtime_hw(struct snd_pcm_substream *substream)
+/**
+ * snd_soc_runtime_calc_hw() - Calculate hw limits for a PCM stream
+ * @rtd: ASoC PCM runtime
+ * @hw: PCM hardware parameters (output)
+ * @stream: Direction of the PCM stream
+ *
+ * Calculates the subset of stream parameters supported by all DAIs
+ * associated with the PCM stream.
+ */
+int snd_soc_runtime_calc_hw(struct snd_soc_pcm_runtime *rtd,
+			    struct snd_pcm_hardware *hw, int stream)
 {
-	struct snd_pcm_runtime *runtime = substream->runtime;
-	struct snd_pcm_hardware *hw = &runtime->hw;
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_dai *codec_dai;
 	struct snd_soc_dai *cpu_dai;
 	struct snd_soc_pcm_stream *codec_stream;
@@ -602,7 +609,6 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_substream *substream)
 	unsigned int cpu_rate_min = 0, cpu_rate_max = UINT_MAX;
 	unsigned int rates = UINT_MAX, cpu_rates = UINT_MAX;
 	u64 formats = ULLONG_MAX;
-	int stream = substream->stream;
 	int i;
 
 	/* first calculate min/max only for CPUs in the DAI link */
@@ -613,12 +619,8 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_substream *substream)
 		 * Otherwise, since the rate, channel, and format values will
 		 * zero in that case, we would have no usable settings left,
 		 * causing the resulting setup to fail.
-		 * At least one CPU should match, otherwise we should have
-		 * bailed out on a higher level, since there would be no
-		 * CPU to support the transfer direction in that case.
 		 */
-		if (!snd_soc_dai_stream_valid(cpu_dai,
-					      substream->stream))
+		if (!snd_soc_dai_stream_valid(cpu_dai, stream))
 			continue;
 
 		cpu_stream = snd_soc_dai_get_pcm_stream(cpu_dai, stream);
@@ -640,12 +642,8 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_substream *substream)
 		 * Otherwise, since the rate, channel, and format values will
 		 * zero in that case, we would have no usable settings left,
 		 * causing the resulting setup to fail.
-		 * At least one CODEC should match, otherwise we should have
-		 * bailed out on a higher level, since there would be no
-		 * CODEC to support the transfer direction in that case.
 		 */
-		if (!snd_soc_dai_stream_valid(codec_dai,
-					      substream->stream))
+		if (!snd_soc_dai_stream_valid(codec_dai, stream))
 			continue;
 
 		codec_stream = snd_soc_dai_get_pcm_stream(codec_dai, stream);
@@ -658,6 +656,10 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_substream *substream)
 		rates = snd_pcm_rate_mask_intersect(codec_stream->rates, rates);
 	}
 
+	/* Verify both a valid CPU DAI and a valid CODEC DAI were found */
+	if (!chan_min || !cpu_chan_min)
+		return -EINVAL;
+
 	/*
 	 * chan min/max cannot be enforced if there are multiple CODEC DAIs
 	 * connected to CPU DAI(s), use CPU DAI's directly and let
@@ -671,18 +673,35 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_substream *substream)
 	/* finally find a intersection between CODECs and CPUs */
 	hw->channels_min = max(chan_min, cpu_chan_min);
 	hw->channels_max = min(chan_max, cpu_chan_max);
-	if (hw->formats)
-		hw->formats &= formats;
-	else
-		hw->formats = formats;
+	hw->formats = formats;
 	hw->rates = snd_pcm_rate_mask_intersect(rates, cpu_rates);
 
-	snd_pcm_limit_hw_rates(runtime);
+	snd_pcm_hw_limit_rates(hw);
 
 	hw->rate_min = max(hw->rate_min, cpu_rate_min);
 	hw->rate_min = max(hw->rate_min, rate_min);
 	hw->rate_max = min_not_zero(hw->rate_max, cpu_rate_max);
 	hw->rate_max = min_not_zero(hw->rate_max, rate_max);
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_runtime_calc_hw);
+
+static void soc_pcm_init_runtime_hw(struct snd_pcm_substream *substream)
+{
+	struct snd_pcm_hardware *hw = &substream->runtime->hw;
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	u64 formats = hw->formats;
+
+	/*
+	 * At least one CPU and one CODEC should match. Otherwise, we should
+	 * have bailed out on a higher level, since there would be no CPU or
+	 * CODEC to support the transfer direction in that case.
+	 */
+	snd_soc_runtime_calc_hw(rtd, hw, substream->stream);
+
+	if (formats)
+		hw->formats &= formats;
 }
 
 static int soc_pcm_components_open(struct snd_pcm_substream *substream)
-- 
2.24.1

[PATCH v4 2/3] ASoC: pcm: Export parameter intersection logic

[Samuel Holland]

The logic to calculate the subset of stream parameters supported by all
DAIs associated with a PCM stream is nontrivial. Export a helper
function so it can be used to set up simple codec2codec DAI links.

Signed-off-by: Samuel Holland <samuel@sholland.org>
---
 include/sound/soc.h |  3 +++
 sound/soc/soc-pcm.c | 59 ++++++++++++++++++++++++++++++---------------
 2 files changed, 42 insertions(+), 20 deletions(-)

diff --git a/include/sound/soc.h b/include/sound/soc.h
index 81e5d17be935..9543d9246ca4 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -471,6 +471,9 @@ bool snd_soc_runtime_ignore_pmdown_time(struct snd_soc_pcm_runtime *rtd);
 void snd_soc_runtime_activate(struct snd_soc_pcm_runtime *rtd, int stream);
 void snd_soc_runtime_deactivate(struct snd_soc_pcm_runtime *rtd, int stream);
 
+int snd_soc_runtime_calc_hw(struct snd_soc_pcm_runtime *rtd,
+			    struct snd_pcm_hardware *hw, int stream);
+
 int snd_soc_runtime_set_dai_fmt(struct snd_soc_pcm_runtime *rtd,
 	unsigned int dai_fmt);
 
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index 90857138c823..2ad501aaa4f9 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -587,11 +587,18 @@ static void soc_pcm_apply_msb(struct snd_pcm_substream *substream)
 	soc_pcm_set_msb(substream, cpu_bits);
 }
 
-static void soc_pcm_init_runtime_hw(struct snd_pcm_substream *substream)
+/**
+ * snd_soc_runtime_calc_hw() - Calculate hw limits for a PCM stream
+ * @rtd: ASoC PCM runtime
+ * @hw: PCM hardware parameters (output)
+ * @stream: Direction of the PCM stream
+ *
+ * Calculates the subset of stream parameters supported by all DAIs
+ * associated with the PCM stream.
+ */
+int snd_soc_runtime_calc_hw(struct snd_soc_pcm_runtime *rtd,
+			    struct snd_pcm_hardware *hw, int stream)
 {
-	struct snd_pcm_runtime *runtime = substream->runtime;
-	struct snd_pcm_hardware *hw = &runtime->hw;
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_dai *codec_dai;
 	struct snd_soc_dai *cpu_dai;
 	struct snd_soc_pcm_stream *codec_stream;
@@ -602,7 +609,6 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_substream *substream)
 	unsigned int cpu_rate_min = 0, cpu_rate_max = UINT_MAX;
 	unsigned int rates = UINT_MAX, cpu_rates = UINT_MAX;
 	u64 formats = ULLONG_MAX;
-	int stream = substream->stream;
 	int i;
 
 	/* first calculate min/max only for CPUs in the DAI link */
@@ -613,12 +619,8 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_substream *substream)
 		 * Otherwise, since the rate, channel, and format values will
 		 * zero in that case, we would have no usable settings left,
 		 * causing the resulting setup to fail.
-		 * At least one CPU should match, otherwise we should have
-		 * bailed out on a higher level, since there would be no
-		 * CPU to support the transfer direction in that case.
 		 */
-		if (!snd_soc_dai_stream_valid(cpu_dai,
-					      substream->stream))
+		if (!snd_soc_dai_stream_valid(cpu_dai, stream))
 			continue;
 
 		cpu_stream = snd_soc_dai_get_pcm_stream(cpu_dai, stream);
@@ -640,12 +642,8 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_substream *substream)
 		 * Otherwise, since the rate, channel, and format values will
 		 * zero in that case, we would have no usable settings left,
 		 * causing the resulting setup to fail.
-		 * At least one CODEC should match, otherwise we should have
-		 * bailed out on a higher level, since there would be no
-		 * CODEC to support the transfer direction in that case.
 		 */
-		if (!snd_soc_dai_stream_valid(codec_dai,
-					      substream->stream))
+		if (!snd_soc_dai_stream_valid(codec_dai, stream))
 			continue;
 
 		codec_stream = snd_soc_dai_get_pcm_stream(codec_dai, stream);
@@ -658,6 +656,10 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_substream *substream)
 		rates = snd_pcm_rate_mask_intersect(codec_stream->rates, rates);
 	}
 
+	/* Verify both a valid CPU DAI and a valid CODEC DAI were found */
+	if (!chan_min || !cpu_chan_min)
+		return -EINVAL;
+
 	/*
 	 * chan min/max cannot be enforced if there are multiple CODEC DAIs
 	 * connected to CPU DAI(s), use CPU DAI's directly and let
@@ -671,18 +673,35 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_substream *substream)
 	/* finally find a intersection between CODECs and CPUs */
 	hw->channels_min = max(chan_min, cpu_chan_min);
 	hw->channels_max = min(chan_max, cpu_chan_max);
-	if (hw->formats)
-		hw->formats &= formats;
-	else
-		hw->formats = formats;
+	hw->formats = formats;
 	hw->rates = snd_pcm_rate_mask_intersect(rates, cpu_rates);
 
-	snd_pcm_limit_hw_rates(runtime);
+	snd_pcm_hw_limit_rates(hw);
 
 	hw->rate_min = max(hw->rate_min, cpu_rate_min);
 	hw->rate_min = max(hw->rate_min, rate_min);
 	hw->rate_max = min_not_zero(hw->rate_max, cpu_rate_max);
 	hw->rate_max = min_not_zero(hw->rate_max, rate_max);
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_runtime_calc_hw);
+
+static void soc_pcm_init_runtime_hw(struct snd_pcm_substream *substream)
+{
+	struct snd_pcm_hardware *hw = &substream->runtime->hw;
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	u64 formats = hw->formats;
+
+	/*
+	 * At least one CPU and one CODEC should match. Otherwise, we should
+	 * have bailed out on a higher level, since there would be no CPU or
+	 * CODEC to support the transfer direction in that case.
+	 */
+	snd_soc_runtime_calc_hw(rtd, hw, substream->stream);
+
+	if (formats)
+		hw->formats &= formats;
 }
 
 static int soc_pcm_components_open(struct snd_pcm_substream *substream)
-- 
2.24.1

[PATCH v4 3/3] ASoC: simple-card: Add support for codec2codec DAI links

[Samuel Holland]

Following the example in cb2cf0de1174 ("ASoC: soc-core: care Codec <->
Codec case by non_legacy_dai_naming"), determine if a DAI link contains
only codec DAIs by examining the non_legacy_dai_naming flag in each
DAI's component.

For now, we assume there is only one or a small set of valid PCM stream
parameters, so num_params == 1 is good enough. We also assume that the
same params are valid for all supported streams. params is set to the
subset of parameters common among all DAIs, and then the existing code
automatically chooses the highest quality of the remaining values when
the link is brought up.

Signed-off-by: Samuel Holland <samuel@sholland.org>
---
 Documentation/sound/soc/codec-to-codec.rst |  9 +++-
 sound/soc/generic/simple-card-utils.c      | 48 ++++++++++++++++++++++
 2 files changed, 55 insertions(+), 2 deletions(-)

diff --git a/Documentation/sound/soc/codec-to-codec.rst b/Documentation/sound/soc/codec-to-codec.rst
index 810109d7500d..4eaa9a0c41fc 100644
--- a/Documentation/sound/soc/codec-to-codec.rst
+++ b/Documentation/sound/soc/codec-to-codec.rst
@@ -104,5 +104,10 @@ Make sure to name your corresponding cpu and codec playback and capture
 dai names ending with "Playback" and "Capture" respectively as dapm core
 will link and power those dais based on the name.
 
-Note that in current device tree there is no way to mark a dai_link
-as codec to codec. However, it may change in future.
+A dai_link in a "simple-audio-card" will automatically be detected as
+codec to codec when all DAIs on the link belong to codec components.
+The dai_link will be initialized with the subset of stream parameters
+(channels, format, sample rate) supported by all DAIs on the link. Since
+there is no way to provide these parameters in the device tree, this is
+mostly useful for communication with simple fixed-function codecs, such
+as a Bluetooth controller or cellular modem.
diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c
index 9b794775df53..320e648f7499 100644
--- a/sound/soc/generic/simple-card-utils.c
+++ b/sound/soc/generic/simple-card-utils.c
@@ -331,6 +331,50 @@ static int asoc_simple_init_dai(struct snd_soc_dai *dai,
 	return 0;
 }
 
+static int asoc_simple_init_dai_link_params(struct snd_soc_pcm_runtime *rtd,
+					    struct simple_dai_props *dai_props)
+{
+	struct snd_soc_dai_link *dai_link = rtd->dai_link;
+	struct snd_soc_component *component;
+	struct snd_soc_pcm_stream *params;
+	struct snd_pcm_hardware hw;
+	int i, ret, stream;
+
+	/* Only codecs should have non_legacy_dai_naming set. */
+	for_each_rtd_components(rtd, i, component) {
+		if (!component->driver->non_legacy_dai_naming)
+			return 0;
+	}
+
+	/* Assumes the capabilities are the same for all supported streams */
+	for (stream = 0; stream < 2; stream++) {
+		ret = snd_soc_runtime_calc_hw(rtd, &hw, stream);
+		if (ret == 0)
+			break;
+	}
+
+	if (ret < 0) {
+		dev_err(rtd->dev, "simple-card: no valid dai_link params\n");
+		return ret;
+	}
+
+	params = devm_kzalloc(rtd->dev, sizeof(*params), GFP_KERNEL);
+	if (!params)
+		return -ENOMEM;
+
+	params->formats = hw.formats;
+	params->rates = hw.rates;
+	params->rate_min = hw.rate_min;
+	params->rate_max = hw.rate_max;
+	params->channels_min = hw.channels_min;
+	params->channels_max = hw.channels_max;
+
+	dai_link->params = params;
+	dai_link->num_params = 1;
+
+	return 0;
+}
+
 int asoc_simple_dai_init(struct snd_soc_pcm_runtime *rtd)
 {
 	struct asoc_simple_priv *priv = snd_soc_card_get_drvdata(rtd->card);
@@ -347,6 +391,10 @@ int asoc_simple_dai_init(struct snd_soc_pcm_runtime *rtd)
 	if (ret < 0)
 		return ret;
 
+	ret = asoc_simple_init_dai_link_params(rtd, dai_props);
+	if (ret < 0)
+		return ret;
+
 	return 0;
 }
 EXPORT_SYMBOL_GPL(asoc_simple_dai_init);
-- 
2.24.1

[PATCH v4 3/3] ASoC: simple-card: Add support for codec2codec DAI links

[Samuel Holland]

Following the example in cb2cf0de1174 ("ASoC: soc-core: care Codec <->
Codec case by non_legacy_dai_naming"), determine if a DAI link contains
only codec DAIs by examining the non_legacy_dai_naming flag in each
DAI's component.

For now, we assume there is only one or a small set of valid PCM stream
parameters, so num_params == 1 is good enough. We also assume that the
same params are valid for all supported streams. params is set to the
subset of parameters common among all DAIs, and then the existing code
automatically chooses the highest quality of the remaining values when
the link is brought up.

Signed-off-by: Samuel Holland <samuel@sholland.org>
---
 Documentation/sound/soc/codec-to-codec.rst |  9 +++-
 sound/soc/generic/simple-card-utils.c      | 48 ++++++++++++++++++++++
 2 files changed, 55 insertions(+), 2 deletions(-)

diff --git a/Documentation/sound/soc/codec-to-codec.rst b/Documentation/sound/soc/codec-to-codec.rst
index 810109d7500d..4eaa9a0c41fc 100644
--- a/Documentation/sound/soc/codec-to-codec.rst
+++ b/Documentation/sound/soc/codec-to-codec.rst
@@ -104,5 +104,10 @@ Make sure to name your corresponding cpu and codec playback and capture
 dai names ending with "Playback" and "Capture" respectively as dapm core
 will link and power those dais based on the name.
 
-Note that in current device tree there is no way to mark a dai_link
-as codec to codec. However, it may change in future.
+A dai_link in a "simple-audio-card" will automatically be detected as
+codec to codec when all DAIs on the link belong to codec components.
+The dai_link will be initialized with the subset of stream parameters
+(channels, format, sample rate) supported by all DAIs on the link. Since
+there is no way to provide these parameters in the device tree, this is
+mostly useful for communication with simple fixed-function codecs, such
+as a Bluetooth controller or cellular modem.
diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c
index 9b794775df53..320e648f7499 100644
--- a/sound/soc/generic/simple-card-utils.c
+++ b/sound/soc/generic/simple-card-utils.c
@@ -331,6 +331,50 @@ static int asoc_simple_init_dai(struct snd_soc_dai *dai,
 	return 0;
 }
 
+static int asoc_simple_init_dai_link_params(struct snd_soc_pcm_runtime *rtd,
+					    struct simple_dai_props *dai_props)
+{
+	struct snd_soc_dai_link *dai_link = rtd->dai_link;
+	struct snd_soc_component *component;
+	struct snd_soc_pcm_stream *params;
+	struct snd_pcm_hardware hw;
+	int i, ret, stream;
+
+	/* Only codecs should have non_legacy_dai_naming set. */
+	for_each_rtd_components(rtd, i, component) {
+		if (!component->driver->non_legacy_dai_naming)
+			return 0;
+	}
+
+	/* Assumes the capabilities are the same for all supported streams */
+	for (stream = 0; stream < 2; stream++) {
+		ret = snd_soc_runtime_calc_hw(rtd, &hw, stream);
+		if (ret == 0)
+			break;
+	}
+
+	if (ret < 0) {
+		dev_err(rtd->dev, "simple-card: no valid dai_link params\n");
+		return ret;
+	}
+
+	params = devm_kzalloc(rtd->dev, sizeof(*params), GFP_KERNEL);
+	if (!params)
+		return -ENOMEM;
+
+	params->formats = hw.formats;
+	params->rates = hw.rates;
+	params->rate_min = hw.rate_min;
+	params->rate_max = hw.rate_max;
+	params->channels_min = hw.channels_min;
+	params->channels_max = hw.channels_max;
+
+	dai_link->params = params;
+	dai_link->num_params = 1;
+
+	return 0;
+}
+
 int asoc_simple_dai_init(struct snd_soc_pcm_runtime *rtd)
 {
 	struct asoc_simple_priv *priv = snd_soc_card_get_drvdata(rtd->card);
@@ -347,6 +391,10 @@ int asoc_simple_dai_init(struct snd_soc_pcm_runtime *rtd)
 	if (ret < 0)
 		return ret;
 
+	ret = asoc_simple_init_dai_link_params(rtd, dai_props);
+	if (ret < 0)
+		return ret;
+
 	return 0;
 }
 EXPORT_SYMBOL_GPL(asoc_simple_dai_init);
-- 
2.24.1

Re: [PATCH v4 1/3] ALSA: pcm: Add a standalone version of snd_pcm_limit_hw_rates

[Takashi Iwai]

On Thu, 05 Mar 2020 06:11:41 +0100,
Samuel Holland wrote:
> 
> It can be useful to derive min/max rates of a snd_pcm_hardware without
> having a snd_pcm_runtime, such as before constructing an ASoC DAI link.
> 
> Create a new helper that takes a pointer to a snd_pcm_hardware directly,
> and refactor the original function as a wrapper around it, to avoid
> needing to update any call sites.
> 
> Signed-off-by: Samuel Holland <samuel@sholland.org>

The code change looks OK to me.

Reviewed-by: Takashi Iwai <tiwai@suse.de>


thanks,

Takashi


> ---
>  include/sound/pcm.h   |  9 ++++++++-
>  sound/core/pcm_misc.c | 18 +++++++++---------
>  2 files changed, 17 insertions(+), 10 deletions(-)
> 
> diff --git a/include/sound/pcm.h b/include/sound/pcm.h
> index 2628246b76fa..f7a95b711100 100644
> --- a/include/sound/pcm.h
> +++ b/include/sound/pcm.h
> @@ -1127,7 +1127,14 @@ snd_pcm_kernel_readv(struct snd_pcm_substream *substream,
>  	return __snd_pcm_lib_xfer(substream, bufs, false, frames, true);
>  }
>  
> -int snd_pcm_limit_hw_rates(struct snd_pcm_runtime *runtime);
> +int snd_pcm_hw_limit_rates(struct snd_pcm_hardware *hw);
> +
> +static inline int
> +snd_pcm_limit_hw_rates(struct snd_pcm_runtime *runtime)
> +{
> +	return snd_pcm_hw_limit_rates(&runtime->hw);
> +}
> +
>  unsigned int snd_pcm_rate_to_rate_bit(unsigned int rate);
>  unsigned int snd_pcm_rate_bit_to_rate(unsigned int rate_bit);
>  unsigned int snd_pcm_rate_mask_intersect(unsigned int rates_a,
> diff --git a/sound/core/pcm_misc.c b/sound/core/pcm_misc.c
> index a6a541511534..5dd2e5335900 100644
> --- a/sound/core/pcm_misc.c
> +++ b/sound/core/pcm_misc.c
> @@ -474,32 +474,32 @@ int snd_pcm_format_set_silence(snd_pcm_format_t format, void *data, unsigned int
>  EXPORT_SYMBOL(snd_pcm_format_set_silence);
>  
>  /**
> - * snd_pcm_limit_hw_rates - determine rate_min/rate_max fields
> - * @runtime: the runtime instance
> + * snd_pcm_hw_limit_rates - determine rate_min/rate_max fields
> + * @hw: the pcm hw instance
>   *
>   * Determines the rate_min and rate_max fields from the rates bits of
> - * the given runtime->hw.
> + * the given hw.
>   *
>   * Return: Zero if successful.
>   */
> -int snd_pcm_limit_hw_rates(struct snd_pcm_runtime *runtime)
> +int snd_pcm_hw_limit_rates(struct snd_pcm_hardware *hw)
>  {
>  	int i;
>  	for (i = 0; i < (int)snd_pcm_known_rates.count; i++) {
> -		if (runtime->hw.rates & (1 << i)) {
> -			runtime->hw.rate_min = snd_pcm_known_rates.list[i];
> +		if (hw->rates & (1 << i)) {
> +			hw->rate_min = snd_pcm_known_rates.list[i];
>  			break;
>  		}
>  	}
>  	for (i = (int)snd_pcm_known_rates.count - 1; i >= 0; i--) {
> -		if (runtime->hw.rates & (1 << i)) {
> -			runtime->hw.rate_max = snd_pcm_known_rates.list[i];
> +		if (hw->rates & (1 << i)) {
> +			hw->rate_max = snd_pcm_known_rates.list[i];
>  			break;
>  		}
>  	}
>  	return 0;
>  }
> -EXPORT_SYMBOL(snd_pcm_limit_hw_rates);
> +EXPORT_SYMBOL(snd_pcm_hw_limit_rates);
>  
>  /**
>   * snd_pcm_rate_to_rate_bit - converts sample rate to SNDRV_PCM_RATE_xxx bit
> -- 
> 2.24.1
> 

Re: [PATCH v4 1/3] ALSA: pcm: Add a standalone version of snd_pcm_limit_hw_rates

[Takashi Iwai]

On Thu, 05 Mar 2020 06:11:41 +0100,
Samuel Holland wrote:
> 
> It can be useful to derive min/max rates of a snd_pcm_hardware without
> having a snd_pcm_runtime, such as before constructing an ASoC DAI link.
> 
> Create a new helper that takes a pointer to a snd_pcm_hardware directly,
> and refactor the original function as a wrapper around it, to avoid
> needing to update any call sites.
> 
> Signed-off-by: Samuel Holland <samuel@sholland.org>

The code change looks OK to me.

Reviewed-by: Takashi Iwai <tiwai@suse.de>


thanks,

Takashi


> ---
>  include/sound/pcm.h   |  9 ++++++++-
>  sound/core/pcm_misc.c | 18 +++++++++---------
>  2 files changed, 17 insertions(+), 10 deletions(-)
> 
> diff --git a/include/sound/pcm.h b/include/sound/pcm.h
> index 2628246b76fa..f7a95b711100 100644
> --- a/include/sound/pcm.h
> +++ b/include/sound/pcm.h
> @@ -1127,7 +1127,14 @@ snd_pcm_kernel_readv(struct snd_pcm_substream *substream,
>  	return __snd_pcm_lib_xfer(substream, bufs, false, frames, true);
>  }
>  
> -int snd_pcm_limit_hw_rates(struct snd_pcm_runtime *runtime);
> +int snd_pcm_hw_limit_rates(struct snd_pcm_hardware *hw);
> +
> +static inline int
> +snd_pcm_limit_hw_rates(struct snd_pcm_runtime *runtime)
> +{
> +	return snd_pcm_hw_limit_rates(&runtime->hw);
> +}
> +
>  unsigned int snd_pcm_rate_to_rate_bit(unsigned int rate);
>  unsigned int snd_pcm_rate_bit_to_rate(unsigned int rate_bit);
>  unsigned int snd_pcm_rate_mask_intersect(unsigned int rates_a,
> diff --git a/sound/core/pcm_misc.c b/sound/core/pcm_misc.c
> index a6a541511534..5dd2e5335900 100644
> --- a/sound/core/pcm_misc.c
> +++ b/sound/core/pcm_misc.c
> @@ -474,32 +474,32 @@ int snd_pcm_format_set_silence(snd_pcm_format_t format, void *data, unsigned int
>  EXPORT_SYMBOL(snd_pcm_format_set_silence);
>  
>  /**
> - * snd_pcm_limit_hw_rates - determine rate_min/rate_max fields
> - * @runtime: the runtime instance
> + * snd_pcm_hw_limit_rates - determine rate_min/rate_max fields
> + * @hw: the pcm hw instance
>   *
>   * Determines the rate_min and rate_max fields from the rates bits of
> - * the given runtime->hw.
> + * the given hw.
>   *
>   * Return: Zero if successful.
>   */
> -int snd_pcm_limit_hw_rates(struct snd_pcm_runtime *runtime)
> +int snd_pcm_hw_limit_rates(struct snd_pcm_hardware *hw)
>  {
>  	int i;
>  	for (i = 0; i < (int)snd_pcm_known_rates.count; i++) {
> -		if (runtime->hw.rates & (1 << i)) {
> -			runtime->hw.rate_min = snd_pcm_known_rates.list[i];
> +		if (hw->rates & (1 << i)) {
> +			hw->rate_min = snd_pcm_known_rates.list[i];
>  			break;
>  		}
>  	}
>  	for (i = (int)snd_pcm_known_rates.count - 1; i >= 0; i--) {
> -		if (runtime->hw.rates & (1 << i)) {
> -			runtime->hw.rate_max = snd_pcm_known_rates.list[i];
> +		if (hw->rates & (1 << i)) {
> +			hw->rate_max = snd_pcm_known_rates.list[i];
>  			break;
>  		}
>  	}
>  	return 0;
>  }
> -EXPORT_SYMBOL(snd_pcm_limit_hw_rates);
> +EXPORT_SYMBOL(snd_pcm_hw_limit_rates);
>  
>  /**
>   * snd_pcm_rate_to_rate_bit - converts sample rate to SNDRV_PCM_RATE_xxx bit
> -- 
> 2.24.1
> 

Applied "ASoC: simple-card: Add support for codec2codec DAI links" to the asoc tree

[Mark Brown]

The patch

   ASoC: simple-card: Add support for codec2codec DAI links

has been applied to the asoc tree at

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

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 95cfc0a0aaf575207152dd7601e782702565a6f1 Mon Sep 17 00:00:00 2001
From: Samuel Holland <samuel@sholland.org>
Date: Wed, 4 Mar 2020 23:11:43 -0600
Subject: [PATCH] ASoC: simple-card: Add support for codec2codec DAI links

Following the example in cb2cf0de1174 ("ASoC: soc-core: care Codec <->
Codec case by non_legacy_dai_naming"), determine if a DAI link contains
only codec DAIs by examining the non_legacy_dai_naming flag in each
DAI's component.

For now, we assume there is only one or a small set of valid PCM stream
parameters, so num_params == 1 is good enough. We also assume that the
same params are valid for all supported streams. params is set to the
subset of parameters common among all DAIs, and then the existing code
automatically chooses the highest quality of the remaining values when
the link is brought up.

Signed-off-by: Samuel Holland <samuel@sholland.org>
Link: https://lore.kernel.org/r/20200305051143.60691-4-samuel@sholland.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 Documentation/sound/soc/codec-to-codec.rst |  9 +++-
 sound/soc/generic/simple-card-utils.c      | 48 ++++++++++++++++++++++
 2 files changed, 55 insertions(+), 2 deletions(-)

diff --git a/Documentation/sound/soc/codec-to-codec.rst b/Documentation/sound/soc/codec-to-codec.rst
index 810109d7500d..4eaa9a0c41fc 100644
--- a/Documentation/sound/soc/codec-to-codec.rst
+++ b/Documentation/sound/soc/codec-to-codec.rst
@@ -104,5 +104,10 @@ Make sure to name your corresponding cpu and codec playback and capture
 dai names ending with "Playback" and "Capture" respectively as dapm core
 will link and power those dais based on the name.
 
-Note that in current device tree there is no way to mark a dai_link
-as codec to codec. However, it may change in future.
+A dai_link in a "simple-audio-card" will automatically be detected as
+codec to codec when all DAIs on the link belong to codec components.
+The dai_link will be initialized with the subset of stream parameters
+(channels, format, sample rate) supported by all DAIs on the link. Since
+there is no way to provide these parameters in the device tree, this is
+mostly useful for communication with simple fixed-function codecs, such
+as a Bluetooth controller or cellular modem.
diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c
index 9b794775df53..320e648f7499 100644
--- a/sound/soc/generic/simple-card-utils.c
+++ b/sound/soc/generic/simple-card-utils.c
@@ -331,6 +331,50 @@ static int asoc_simple_init_dai(struct snd_soc_dai *dai,
 	return 0;
 }
 
+static int asoc_simple_init_dai_link_params(struct snd_soc_pcm_runtime *rtd,
+					    struct simple_dai_props *dai_props)
+{
+	struct snd_soc_dai_link *dai_link = rtd->dai_link;
+	struct snd_soc_component *component;
+	struct snd_soc_pcm_stream *params;
+	struct snd_pcm_hardware hw;
+	int i, ret, stream;
+
+	/* Only codecs should have non_legacy_dai_naming set. */
+	for_each_rtd_components(rtd, i, component) {
+		if (!component->driver->non_legacy_dai_naming)
+			return 0;
+	}
+
+	/* Assumes the capabilities are the same for all supported streams */
+	for (stream = 0; stream < 2; stream++) {
+		ret = snd_soc_runtime_calc_hw(rtd, &hw, stream);
+		if (ret == 0)
+			break;
+	}
+
+	if (ret < 0) {
+		dev_err(rtd->dev, "simple-card: no valid dai_link params\n");
+		return ret;
+	}
+
+	params = devm_kzalloc(rtd->dev, sizeof(*params), GFP_KERNEL);
+	if (!params)
+		return -ENOMEM;
+
+	params->formats = hw.formats;
+	params->rates = hw.rates;
+	params->rate_min = hw.rate_min;
+	params->rate_max = hw.rate_max;
+	params->channels_min = hw.channels_min;
+	params->channels_max = hw.channels_max;
+
+	dai_link->params = params;
+	dai_link->num_params = 1;
+
+	return 0;
+}
+
 int asoc_simple_dai_init(struct snd_soc_pcm_runtime *rtd)
 {
 	struct asoc_simple_priv *priv = snd_soc_card_get_drvdata(rtd->card);
@@ -347,6 +391,10 @@ int asoc_simple_dai_init(struct snd_soc_pcm_runtime *rtd)
 	if (ret < 0)
 		return ret;
 
+	ret = asoc_simple_init_dai_link_params(rtd, dai_props);
+	if (ret < 0)
+		return ret;
+
 	return 0;
 }
 EXPORT_SYMBOL_GPL(asoc_simple_dai_init);
-- 
2.20.1

Applied "ASoC: simple-card: Add support for codec2codec DAI links" to the asoc tree

[Mark Brown]

The patch

   ASoC: simple-card: Add support for codec2codec DAI links

has been applied to the asoc tree at

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

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 95cfc0a0aaf575207152dd7601e782702565a6f1 Mon Sep 17 00:00:00 2001
From: Samuel Holland <samuel@sholland.org>
Date: Wed, 4 Mar 2020 23:11:43 -0600
Subject: [PATCH] ASoC: simple-card: Add support for codec2codec DAI links

Following the example in cb2cf0de1174 ("ASoC: soc-core: care Codec <->
Codec case by non_legacy_dai_naming"), determine if a DAI link contains
only codec DAIs by examining the non_legacy_dai_naming flag in each
DAI's component.

For now, we assume there is only one or a small set of valid PCM stream
parameters, so num_params == 1 is good enough. We also assume that the
same params are valid for all supported streams. params is set to the
subset of parameters common among all DAIs, and then the existing code
automatically chooses the highest quality of the remaining values when
the link is brought up.

Signed-off-by: Samuel Holland <samuel@sholland.org>
Link: https://lore.kernel.org/r/20200305051143.60691-4-samuel@sholland.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 Documentation/sound/soc/codec-to-codec.rst |  9 +++-
 sound/soc/generic/simple-card-utils.c      | 48 ++++++++++++++++++++++
 2 files changed, 55 insertions(+), 2 deletions(-)

diff --git a/Documentation/sound/soc/codec-to-codec.rst b/Documentation/sound/soc/codec-to-codec.rst
index 810109d7500d..4eaa9a0c41fc 100644
--- a/Documentation/sound/soc/codec-to-codec.rst
+++ b/Documentation/sound/soc/codec-to-codec.rst
@@ -104,5 +104,10 @@ Make sure to name your corresponding cpu and codec playback and capture
 dai names ending with "Playback" and "Capture" respectively as dapm core
 will link and power those dais based on the name.
 
-Note that in current device tree there is no way to mark a dai_link
-as codec to codec. However, it may change in future.
+A dai_link in a "simple-audio-card" will automatically be detected as
+codec to codec when all DAIs on the link belong to codec components.
+The dai_link will be initialized with the subset of stream parameters
+(channels, format, sample rate) supported by all DAIs on the link. Since
+there is no way to provide these parameters in the device tree, this is
+mostly useful for communication with simple fixed-function codecs, such
+as a Bluetooth controller or cellular modem.
diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c
index 9b794775df53..320e648f7499 100644
--- a/sound/soc/generic/simple-card-utils.c
+++ b/sound/soc/generic/simple-card-utils.c
@@ -331,6 +331,50 @@ static int asoc_simple_init_dai(struct snd_soc_dai *dai,
 	return 0;
 }
 
+static int asoc_simple_init_dai_link_params(struct snd_soc_pcm_runtime *rtd,
+					    struct simple_dai_props *dai_props)
+{
+	struct snd_soc_dai_link *dai_link = rtd->dai_link;
+	struct snd_soc_component *component;
+	struct snd_soc_pcm_stream *params;
+	struct snd_pcm_hardware hw;
+	int i, ret, stream;
+
+	/* Only codecs should have non_legacy_dai_naming set. */
+	for_each_rtd_components(rtd, i, component) {
+		if (!component->driver->non_legacy_dai_naming)
+			return 0;
+	}
+
+	/* Assumes the capabilities are the same for all supported streams */
+	for (stream = 0; stream < 2; stream++) {
+		ret = snd_soc_runtime_calc_hw(rtd, &hw, stream);
+		if (ret == 0)
+			break;
+	}
+
+	if (ret < 0) {
+		dev_err(rtd->dev, "simple-card: no valid dai_link params\n");
+		return ret;
+	}
+
+	params = devm_kzalloc(rtd->dev, sizeof(*params), GFP_KERNEL);
+	if (!params)
+		return -ENOMEM;
+
+	params->formats = hw.formats;
+	params->rates = hw.rates;
+	params->rate_min = hw.rate_min;
+	params->rate_max = hw.rate_max;
+	params->channels_min = hw.channels_min;
+	params->channels_max = hw.channels_max;
+
+	dai_link->params = params;
+	dai_link->num_params = 1;
+
+	return 0;
+}
+
 int asoc_simple_dai_init(struct snd_soc_pcm_runtime *rtd)
 {
 	struct asoc_simple_priv *priv = snd_soc_card_get_drvdata(rtd->card);
@@ -347,6 +391,10 @@ int asoc_simple_dai_init(struct snd_soc_pcm_runtime *rtd)
 	if (ret < 0)
 		return ret;
 
+	ret = asoc_simple_init_dai_link_params(rtd, dai_props);
+	if (ret < 0)
+		return ret;
+
 	return 0;
 }
 EXPORT_SYMBOL_GPL(asoc_simple_dai_init);
-- 
2.20.1