webrtc/call/call.cc - Issue 2165743003: Variable audio bitrate.

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Unified Diff: webrtc/call/call.cc

Issue 2165743003: Variable audio bitrate. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master

Patch Set: Rebase Created 4 years, 5 months ago

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Index: webrtc/call/call.cc

diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc

index fa2d1e7e782d977d65de8a64f5c49a56a31683c5..6df86e3c1a0cce8eaf18eee620e5f6acbd62f25f 100644

--- a/webrtc/call/call.cc

+++ b/webrtc/call/call.cc

@@ -348,7 +348,8 @@ webrtc::AudioSendStream* Call::CreateAudioSendStream(

TRACE_EVENT0("webrtc", "Call::CreateAudioSendStream");

RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());

AudioSendStream* send_stream = new AudioSendStream(

- config, config_.audio_state, congestion_controller_.get());

+ config, config_.audio_state, congestion_controller_.get(),

+ bitrate_allocator_.get());

{

WriteLockScoped write_lock(*send_crit_);

RTC_DCHECK(audio_send_ssrcs_.find(config.rtp.ssrc) ==

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