commit d4fce5a361a4dba8a7c97e885af898e24344e7ec
author Peter Hanspers <peterhanspers@webrtc.org> Thu Apr 28 18:38:01 2022
committer WebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com> Fri Apr 29 14:07:02 2022
tree d9ef0ed613c634fb679828b0f097cc44ea3344a9
parent 8ec4a2eca3a6bf8966ac07793297752b97a1ba18

Use playout sample rate for audio unit.

Fixing a race condition where session.sampleRate changes before AudioDeviceIOS::HandleValidRouteChange() finishes.

session.sampleRate is read into session_sample_rate at 576 and used at 623 to initialize the audio unit. However, in the call to SetupAudioBuffersForActiveAudioSession() the session.sampleRate is read again and may have changed, resulting in different sample rates used for the buffers and the audio unit. The consequence is a sample rate mismatch with either high pitched or low pitched audio.

The fix is to always use the buffer sample rate for the audio unit.

The DCHECK at 622 would save us in debug, but not in production, hence removed.

Change-Id: I562f1bf7f94d7447139ada2636b02ade7fcd6371
Bug: webrtc:14011
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260329
Reviewed-by: Henrik Andreasson <henrika@google.com>
Commit-Queue: Peter Hanspers <peterhanspers@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36708}