audio/hal: Set playback buffer size to integral multiple of msec
- Set the deep-buffer and low-latency output buffer sizes to
integral multiple of msec. This reduces the variations in
the writes.
- Compute the input buffer size based on sample rate. Use 20msec
buffers in capture path.
Bug: 9283911, 9106434
Change-Id: Icbcb653f7f0fd3293dd4b514a54ac91d8311b308
diff --git a/hal/audio_hw.c b/hal/audio_hw.c
index b73c95e..78f38a2 100644
--- a/hal/audio_hw.c
+++ b/hal/audio_hw.c
@@ -73,7 +73,6 @@
struct pcm_config pcm_config_audio_capture = {
.channels = 2,
- .period_size = AUDIO_CAPTURE_PERIOD_SIZE,
.period_count = AUDIO_CAPTURE_PERIOD_COUNT,
.format = PCM_FORMAT_S16_LE,
};
@@ -817,19 +816,19 @@
{
size_t size = 0;
- if (check_input_parameters(sample_rate, format, channel_count) != 0) return 0;
+ if (check_input_parameters(sample_rate, format, channel_count) != 0)
+ return 0;
- if (sample_rate == 8000 || sample_rate == 16000 || sample_rate == 32000) {
- size = (sample_rate * 20) / 1000;
- } else if (sample_rate == 11025 || sample_rate == 12000) {
- size = 256;
- } else if (sample_rate == 22050 || sample_rate == 24000) {
- size = 512;
- } else if (sample_rate == 44100 || sample_rate == 48000) {
- size = 1024;
- }
+ size = (sample_rate * AUDIO_CAPTURE_PERIOD_DURATION_MSEC) / 1000;
+ /* ToDo: should use frame_size computed based on the format and
+ channel_count here. */
+ size *= sizeof(short) * channel_count;
- return size * sizeof(short) * channel_count;
+ /* make sure the size is multiple of 64 */
+ size += 0x3f;
+ size &= ~0x3f;
+
+ return size;
}
static uint32_t out_get_sample_rate(const struct audio_stream *stream)