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| author | Jonas Oreland <jonaso@webrtc.org> | 2022-03-16 13:50:58 +0100 |
|---|---|---|
| committer | WebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com> | 2022-03-17 07:11:44 +0000 |
| commit | a943e730b2acced12fa1028ad43b9c9c91bf631c (patch) | |
| tree | 522f0c36175075429c628052a7b26f4c459df172 /call/call.cc | |
| parent | e72f4bc524b52cb4aaa70005811835698f56785b (diff) | |
| download | webrtc-a943e730b2acced12fa1028ad43b9c9c91bf631c.tar.gz | |
WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 7/inf
Convert audio/ and collateral (audio encoder copy red).
Bug: webrtc:10335
Change-Id: Iac54c0cfd2f62f4402f3deec35ae2725ec35b81a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/255820
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36229}
Diffstat (limited to 'call/call.cc')
| -rw-r--r-- | call/call.cc | 2 |
1 files changed, 1 insertions, 1 deletions
diff --git a/call/call.cc b/call/call.cc index 16f7b9058f..f2101cfc6b 100644 --- a/call/call.cc +++ b/call/call.cc @@ -915,7 +915,7 @@ webrtc::AudioSendStream* Call::CreateAudioSendStream( AudioSendStream* send_stream = new AudioSendStream( clock_, config, config_.audio_state, task_queue_factory_, transport_send_.get(), bitrate_allocator_.get(), event_log_, - call_stats_->AsRtcpRttStats(), suspended_rtp_state); + call_stats_->AsRtcpRttStats(), suspended_rtp_state, trials()); RTC_DCHECK(audio_send_ssrcs_.find(config.rtp.ssrc) == audio_send_ssrcs_.end()); audio_send_ssrcs_[config.rtp.ssrc] = send_stream; |