audio hal: fix input buffer size in non real time mode
Fix regression introduced by commit 0e46adf2 in input buffer size
calculation for fast input in non realtime mode.
Bug: 35452939
Test: Verified Skype audio
Change-Id: I76170e8a0c38f42b1be74f045472325d5c93add9
diff --git a/hal/audio_hw.c b/hal/audio_hw.c
index 9a1771c..85216bc 100644
--- a/hal/audio_hw.c
+++ b/hal/audio_hw.c
@@ -3792,13 +3792,13 @@
#endif
in->realtime = may_use_noirq_mode(adev, in->usecase, in->flags);
if (!in->realtime) {
+ in->config = pcm_config_audio_capture;
frame_size = audio_stream_in_frame_size(&in->stream);
buffer_size = get_input_buffer_size(config->sample_rate,
config->format,
channel_count,
is_low_latency);
in->config.period_size = buffer_size / frame_size;
- in->config = pcm_config_audio_capture;
in->config.rate = config->sample_rate;
in->af_period_multiplier = 1;
} else {