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authorXin Li <delphij@google.com>2021-06-21 05:49:07 +0000
committerGerrit Code Review <noreply-gerritcodereview@google.com>2021-06-21 05:49:07 +0000
commit87f5e5c4fc8f794990728a02e6f29fe0472c5785 (patch)
tree803766e59f6e780eabd3d7d2ba39d8e8f47bb2f7
parent9cfbee19697b606ad115b5556f6fcbed9e50a21e (diff)
parent97a1c3d09a696d5226b9ec37bbbe8312c420a190 (diff)
downloadaudio-87f5e5c4fc8f794990728a02e6f29fe0472c5785.tar.gz
Merge "DO NOT MERGE - Merge RQ3A.210605.005"android-s-beta-4android-s-beta-3android-s-beta-4
-rw-r--r--hal/audio_hw.c26
1 files changed, 12 insertions, 14 deletions
diff --git a/hal/audio_hw.c b/hal/audio_hw.c
index a3abb8a..28faffc 100644
--- a/hal/audio_hw.c
+++ b/hal/audio_hw.c
@@ -2738,23 +2738,21 @@ static size_t get_stream_buffer_size(size_t duration_ms,
int channel_count,
bool is_low_latency)
{
- size_t size = 0;
+ // Compute target frames based on time or period size.
+ size_t target_frames = is_low_latency
+ ? configured_low_latency_capture_period_size // record only
+ : (sample_rate * duration_ms) / 1000;
- size = (sample_rate * duration_ms) / 1000;
- if (is_low_latency)
- size = configured_low_latency_capture_period_size;
-
- size *= channel_count * audio_bytes_per_sample(format);
+ // Round up to a multiple of 16 frames in case sizing for the MixerThread.
+ if (!is_low_latency) { // low latency flag set for record only
+ target_frames = (target_frames + 0xf) & ~0xf;
+ }
- /* make sure the size is multiple of 32 bytes
- * At 48 kHz mono 16-bit PCM:
- * 5.000 ms = 240 frames = 15*16*1*2 = 480, a whole multiple of 32 (15)
- * 3.333 ms = 160 frames = 10*16*1*2 = 320, a whole multiple of 32 (10)
- */
- size += 0x1f;
- size &= ~0x1f;
+ // Buffer size is the target frames multiplied by the frame size in bytes.
+ const size_t frame_size = channel_count * audio_bytes_per_sample(format);
+ const size_t buffer_size = target_frames * frame_size;
- return size;
+ return buffer_size;
}
static uint32_t out_get_sample_rate(const struct audio_stream *stream)