diff options
| author | Jakob Ivarsson <jakobi@webrtc.org> | 2023-12-12 10:58:18 +0100 |
|---|---|---|
| committer | WebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com> | 2023-12-12 10:43:47 +0000 |
| commit | 871af9225ff348c28e3df0560a840e444807533f (patch) | |
| tree | fb52590016affeeba6eff47d221dcc2f285f70f7 | |
| parent | 776fe6d9230fb9175a406b4b415bd5f6a379f4e0 (diff) | |
| download | webrtc-871af9225ff348c28e3df0560a840e444807533f.tar.gz | |
Log audio stream start/stop.
Bug: None
Change-Id: I3f97672bc7d29dd6023fe8b6bdf98d699622841d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/331160
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41362}
| -rw-r--r-- | audio/audio_receive_stream.cc | 2 | ||||
| -rw-r--r-- | audio/audio_send_stream.cc | 3 |
2 files changed, 4 insertions, 1 deletions
diff --git a/audio/audio_receive_stream.cc b/audio/audio_receive_stream.cc index 978bbb25b2..415ad0640a 100644 --- a/audio/audio_receive_stream.cc +++ b/audio/audio_receive_stream.cc @@ -183,6 +183,7 @@ void AudioReceiveStreamImpl::Start() { if (playing_) { return; } + RTC_LOG(LS_INFO) << "AudioReceiveStreamImpl::Start: " << remote_ssrc(); channel_receive_->StartPlayout(); playing_ = true; audio_state()->AddReceivingStream(this); @@ -193,6 +194,7 @@ void AudioReceiveStreamImpl::Stop() { if (!playing_) { return; } + RTC_LOG(LS_INFO) << "AudioReceiveStreamImpl::Stop: " << remote_ssrc(); channel_receive_->StopPlayout(); playing_ = false; audio_state()->RemoveReceivingStream(this); diff --git a/audio/audio_send_stream.cc b/audio/audio_send_stream.cc index c9dc42c04e..3701eafd6f 100644 --- a/audio/audio_send_stream.cc +++ b/audio/audio_send_stream.cc @@ -355,6 +355,7 @@ void AudioSendStream::Start() { if (sending_) { return; } + RTC_LOG(LS_INFO) << "AudioSendStream::Start: " << config_.rtp.ssrc; if (!config_.has_dscp && config_.min_bitrate_bps != -1 && config_.max_bitrate_bps != -1 && (allocate_audio_without_feedback_ || TransportSeqNumId(config_) != 0)) { @@ -376,7 +377,7 @@ void AudioSendStream::Stop() { if (!sending_) { return; } - + RTC_LOG(LS_INFO) << "AudioSendStream::Stop: " << config_.rtp.ssrc; RemoveBitrateObserver(); channel_send_->StopSend(); sending_ = false; |