Flush buffers when stopping audio receive stream.

Bug: chromium:1400642
Change-Id: I19f22ca2fcf04d5e973d0e49fda841c9d40b12a1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290723
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39045}
This commit is contained in:
Jakob Ivarsson 2023-01-09 16:30:50 +01:00 committed by WebRTC LUCI CQ
parent 075c20fe16
commit 6d5fa001df

View File

@ -621,6 +621,7 @@ void ChannelReceive::StopPlayout() {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
playing_ = false;
_outputAudioLevel.ResetLevelFullRange();
acm_receiver_.FlushBuffers();
}
absl::optional<std::pair<int, SdpAudioFormat>> ChannelReceive::GetReceiveCodec()