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[13/13] ASoC: Intel: avs: Rule invalid buffer and period sizes out

Message ID 20240405090929.1184068-14-cezary.rojewski@intel.com (mailing list archive)
State Accepted
Commit 9a385993504e47a0fd6fd34b5384827b4abdee60
Headers show
Series ASoC: Intel: avs: Fixes and cleanups for 6.10 | expand

Commit Message

Cezary Rojewski April 5, 2024, 9:09 a.m. UTC
While HDAudio controller supports buffer packets up to 128 bytes low,
audio format shall be taken into consideration when calculating buffer
and period sizes to avoid undesired xruns.

As *_size in ALSA terms means frames (channels times bit-depth-bytes),
hw_rules can calculate minimal buffer and period sizes solely from
sample rate and the number of milliseconds commonly used on the
AudioDSP firmware side.

Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
---
 sound/soc/intel/avs/pcm.c | 28 ++++++++++++++++++++++++++++
 1 file changed, 28 insertions(+)
diff mbox series

Patch

diff --git a/sound/soc/intel/avs/pcm.c b/sound/soc/intel/avs/pcm.c
index 405de1d58178..77a7e8f93951 100644
--- a/sound/soc/intel/avs/pcm.c
+++ b/sound/soc/intel/avs/pcm.c
@@ -457,6 +457,26 @@  static const struct snd_pcm_hw_constraint_list hw_rates = {
 
 const struct snd_soc_dai_ops avs_dai_fe_ops;
 
+static int hw_rule_param_size(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule)
+{
+	struct snd_interval *interval = hw_param_interval(params, rule->var);
+	struct snd_interval to;
+
+	snd_interval_any(&to);
+	to.integer = interval->integer;
+	to.max = interval->max;
+	/*
+	 * Commonly 2ms buffer size is used in HDA scenarios whereas 4ms is used
+	 * when streaming through GPDMA. Align to the latter to account for both.
+	 */
+	to.min = params_rate(params) / 1000 * 4;
+
+	if (rule->var == SNDRV_PCM_HW_PARAM_PERIOD_SIZE)
+		to.min /= params_periods(params);
+
+	return snd_interval_refine(interval, &to);
+}
+
 static int avs_dai_fe_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai)
 {
 	struct snd_pcm_runtime *runtime = substream->runtime;
@@ -492,6 +512,14 @@  static int avs_dai_fe_startup(struct snd_pcm_substream *substream, struct snd_so
 	if (ret < 0)
 		goto err;
 
+	/* Adjust buffer and period size based on the audio format. */
+	snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, hw_rule_param_size, NULL,
+			    SNDRV_PCM_HW_PARAM_FORMAT, SNDRV_PCM_HW_PARAM_CHANNELS,
+			    SNDRV_PCM_HW_PARAM_RATE, -1);
+	snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, hw_rule_param_size, NULL,
+			    SNDRV_PCM_HW_PARAM_FORMAT, SNDRV_PCM_HW_PARAM_CHANNELS,
+			    SNDRV_PCM_HW_PARAM_RATE, -1);
+
 	snd_pcm_set_sync(substream);
 
 	dev_dbg(dai->dev, "%s fe STARTUP tag %d str %p",