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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: Created 4 years, 5 months ago
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« webrtc/audio/audio_send_stream.cc ('K') | « webrtc/audio_send_stream.h ('k') | webrtc/call/rampup_tests.h » ('j') | webrtc/call/rampup_tests.h » ('J')
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Index: webrtc/call/call.cc
diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc
index 51e0cceac98d84b8eb87774232b54075087d5228..25c5c6634cde39864d83c66378935554dada947b 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) ==
« webrtc/audio/audio_send_stream.cc ('K') | « webrtc/audio_send_stream.h ('k') | webrtc/call/rampup_tests.h » ('j') | webrtc/call/rampup_tests.h » ('J')

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