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Unified Diff: webrtc/call/call.cc

Issue 1479023002: Prepare the AudioSendStream to be hooked up to send-side BWE. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Reverted back to using a proxy and fixed an issue related to audio nack. Created 5 years ago
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Index: webrtc/call/call.cc
diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc
index 441de87ef4983fc7ee0fd0c88486d3828bbf04b5..bfde12a273ba84540584f11b35c6028885c7b846 100644
--- a/webrtc/call/call.cc
+++ b/webrtc/call/call.cc
@@ -300,8 +300,8 @@ webrtc::AudioSendStream* Call::CreateAudioSendStream(
const webrtc::AudioSendStream::Config& config) {
TRACE_EVENT0("webrtc", "Call::CreateAudioSendStream");
RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
- AudioSendStream* send_stream =
- new AudioSendStream(config, config_.audio_state);
+ AudioSendStream* send_stream = new AudioSendStream(
+ config, config_.audio_state, congestion_controller_.get());
if (!network_enabled_)
send_stream->SignalNetworkState(kNetworkDown);
{
« no previous file with comments | « webrtc/audio_send_stream.h ('k') | webrtc/call/call_perf_tests.cc » ('j') | webrtc/modules/rtp_rtcp/source/rtp_sender.cc » ('J')

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