[PATCH v3 0/3] simple-audio-card codec2codec support
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 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 | 49 +++++++++++++++++++ sound/soc/soc-pcm.c | 55 +++++++++++++++------- 6 files changed, 115 insertions(+), 28 deletions(-)
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 f657ff08f317..89529dfffaac 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -1122,7 +1122,14 @@ 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
On Sat, Feb 22, 2020 at 09:45:31PM -0600, 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.
Takashi, are you OK with me taking this patch?
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 | 55 ++++++++++++++++++++++++++++++++------------- 2 files changed, 42 insertions(+), 16 deletions(-)
diff --git a/include/sound/soc.h b/include/sound/soc.h index f0e4f36f83bf..8724928d02c5 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 ff1b7c7078e5..6680d31eece3 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -402,11 +402,18 @@ 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_driver *cpu_dai_drv = rtd->cpu_dai->driver; struct snd_soc_dai_driver *codec_dai_drv; @@ -418,7 +425,7 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_substream *substream) u64 formats = ULLONG_MAX; int i;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (stream == SNDRV_PCM_STREAM_PLAYBACK) cpu_stream = &cpu_dai_drv->playback; else cpu_stream = &cpu_dai_drv->capture; @@ -431,16 +438,12 @@ 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_dai_drv = codec_dai->driver; - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (stream == SNDRV_PCM_STREAM_PLAYBACK) codec_stream = &codec_dai_drv->playback; else codec_stream = &codec_dai_drv->capture; @@ -452,6 +455,9 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_substream *substream) rates = snd_pcm_rate_mask_intersect(codec_stream->rates, rates); }
+ if (!chan_min) + return -EINVAL; + /* * chan min/max cannot be enforced if there are multiple CODEC DAIs * connected to a single CPU DAI, use CPU DAI's directly and let @@ -464,18 +470,35 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_substream *substream)
hw->channels_min = max(chan_min, cpu_stream->channels_min); hw->channels_max = min(chan_max, cpu_stream->channels_max); - if (hw->formats) - hw->formats &= formats & cpu_stream->formats; - else - hw->formats = formats & cpu_stream->formats; + hw->formats = formats & cpu_stream->formats; hw->rates = snd_pcm_rate_mask_intersect(rates, cpu_stream->rates);
- snd_pcm_limit_hw_rates(runtime); + snd_pcm_hw_limit_rates(hw);
hw->rate_min = max(hw->rate_min, cpu_stream->rate_min); hw->rate_min = max(hw->rate_min, rate_min); hw->rate_max = min_not_zero(hw->rate_max, cpu_stream->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 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. + */ + 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,
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 | 49 ++++++++++++++++++++++ 2 files changed, 56 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..54294367a40f 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -331,6 +331,51 @@ 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_rtdcom_list *rtdcom; + 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 +392,10 @@ 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);
On 2/22/20 9:45 PM, Samuel Holland wrote:
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 | 49 ++++++++++++++++++++++ 2 files changed, 56 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..54294367a40f 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -331,6 +331,51 @@ 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_rtdcom_list *rtdcom;
This variable is unused in v3. I can send a v4.
- 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 +392,10 @@ 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);
On Tue, Mar 03, 2020 at 07:49:33AM -0600, Samuel Holland wrote:
On 2/22/20 9:45 PM, Samuel Holland wrote:
+{
- struct snd_soc_dai_link *dai_link = rtd->dai_link;
- struct snd_soc_component *component;
- struct snd_soc_rtdcom_list *rtdcom;
This variable is unused in v3. I can send a v4.
Please.
Please delete unneeded context from mails when replying. Doing this makes it much easier to find your reply in the message, helping ensure it won't be missed by people scrolling through the irrelevant quoted material.
participants (2)
-
Mark Brown -
Samuel Holland