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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: Comments addressed. 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 4d758d99a62662e25fccafa1e496a1fdd621fe6b..a1ee9a3c7e00cd8eff2336ab387076a9ac2d4b7e 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/test/mock_voe_channel_proxy.h » ('J')

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