commit 6d5fa001dfc4b8b73c2e16679bafab78548ec74b
author Jakob Ivarsson <jakobi@webrtc.org> Mon Jan 09 15:30:50 2023
committer WebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com> Mon Jan 09 19:16:26 2023
tree 082f8e3cf72dbbcd14fc4ad471a5f28617dcd9d9
parent 075c20fe1626761c5fe545434f1e0cdc9141fea5

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}

audio/channel_receive.cc

diff --git a/audio/channel_receive.cc b/audio/channel_receive.cc
index 363eebf..0562d66 100644
--- a/audio/channel_receive.cc
+++ b/audio/channel_receive.cc
@@ -621,6 +621,7 @@
   RTC_DCHECK_RUN_ON(&worker_thread_checker_);
   playing_ = false;
   _outputAudioLevel.ResetLevelFullRange();
+  acm_receiver_.FlushBuffers();
 }
 
 absl::optional<std::pair<int, SdpAudioFormat>> ChannelReceive::GetReceiveCodec()