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-rw-r--r--third_party/libwebrtc/audio/audio_send_stream.cc3
1 files changed, 2 insertions, 1 deletions
diff --git a/third_party/libwebrtc/audio/audio_send_stream.cc b/third_party/libwebrtc/audio/audio_send_stream.cc
index e7ebb2bf4e..8dc78b18fa 100644
--- a/third_party/libwebrtc/audio/audio_send_stream.cc
+++ b/third_party/libwebrtc/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;