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Unified Diff: webrtc/call/call.cc

Issue 1414743004: Implement AudioSendStream::GetStats(). (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: TODOs Created 5 years, 2 months ago
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Index: webrtc/call/call.cc
diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc
index 3969bc6b517e97c2f51bcd35a64af8cc9a40654b..77741d1158a93fc2808e277aa34a143cc3fe7da5 100644
--- a/webrtc/call/call.cc
+++ b/webrtc/call/call.cc
@@ -203,7 +203,8 @@ 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);
+ AudioSendStream* send_stream =
+ new AudioSendStream(config, config_.voice_engine);
{
rtc::CritScope lock(&network_enabled_crit_);
WriteLockScoped write_lock(*send_crit_);

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