webrtc/call/call.cc - Issue 1479023002: Prepare the AudioSendStream to be hooked up to send-side BWE.

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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: Remove incorrect thread check. 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 4156765d74c49efc3c541ac3796195cf230d7219..9209c7c97a06ee96b014087a0e42facb3b5e1495 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);

{

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