From 6d5fa001dfc4b8b73c2e16679bafab78548ec74b Mon Sep 17 00:00:00 2001
From: Jakob Ivarsson <jakobi@webrtc.org>
Date: Mon, 9 Jan 2023 16:30:50 +0100
Subject: [PATCH] Flush buffers when stopping audio receive stream.
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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 | 1 +
 1 file changed, 1 insertion(+)

diff --git a/audio/channel_receive.cc b/audio/channel_receive.cc
index 363eebf0a1..0562d66c78 100644
--- a/audio/channel_receive.cc
+++ b/audio/channel_receive.cc
@@ -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()