Index: webrtc/audio/audio_send_stream.cc |
diff --git a/webrtc/audio/audio_send_stream.cc b/webrtc/audio/audio_send_stream.cc |
index 3831b8d27a1997c008dee230bf056bce74922776..cb4c1b041c81bc44f16ce4d3df6d8477fb1bdb8c 100644 |
--- a/webrtc/audio/audio_send_stream.cc |
+++ b/webrtc/audio/audio_send_stream.cc |
@@ -53,7 +53,8 @@ AudioSendStream::AudioSendStream( |
: worker_queue_(worker_queue), |
config_(config), |
audio_state_(audio_state), |
- bitrate_allocator_(bitrate_allocator) { |
+ bitrate_allocator_(bitrate_allocator), |
+ congestion_controller_(congestion_controller) { |
LOG(LS_INFO) << "AudioSendStream: " << config_.ToString(); |
RTC_DCHECK_NE(config_.voe_channel_id, -1); |
RTC_DCHECK(audio_state_.get()); |
@@ -258,6 +259,7 @@ const webrtc::AudioSendStream::Config& AudioSendStream::config() const { |
void AudioSendStream::SetTransportOverhead(int transport_overhead_per_packet) { |
RTC_DCHECK(thread_checker_.CalledOnValidThread()); |
+ congestion_controller_->SetTransportOverhead(transport_overhead_per_packet); |
channel_proxy_->SetTransportOverhead(transport_overhead_per_packet); |
} |