hal: Fix input buffer size for fractional rate 24bit capture
The input buffer size from ADSP must be a multiple of 32 bytes and a
multiple of the frame_size. For fractional sample rates (eg. 44.1kHz)
with 24bit packed samples, 32 is not the smallest common multiple.
Change-Id: Ifcef1daa4f43e90fdb30873192c0047677ae3c00
diff --git a/hal/audio_hw.c b/hal/audio_hw.c
index 9993c33..0b66246 100644
--- a/hal/audio_hw.c
+++ b/hal/audio_hw.c
@@ -3289,6 +3289,8 @@
int channel_count,
bool is_low_latency)
{
+ int i = 0;
+ size_t frame_size = 0;
size_t size = 0;
if (check_input_parameters(sample_rate, format, channel_count) != 0)
@@ -3298,15 +3300,23 @@
if (is_low_latency)
size = configured_low_latency_capture_period_size;
- size *= audio_bytes_per_sample(format) * channel_count;
+ frame_size = audio_bytes_per_sample(format) * channel_count;
+ size *= frame_size;
- /* make sure the size is multiple of 32 bytes
+ /* make sure the size is multiple of 32 bytes and additionally multiple of
+ * the frame_size (required for 24bit samples and non-power-of-2 channel counts)
* 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)
+ *
+ * The loop reaches result within 32 iterations, as initial size is
+ * already a multiple of frame_size
*/
- size += 0x1f;
- size &= ~0x1f;
+ for (i=0; i<32; i++) {
+ if ((size & 0x1f) == 0)
+ break;
+ size += frame_size;
+ }
return size;
}