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| author | Stefan Holmer <stefan@webrtc.org> | 2015-12-07 10:26:18 +0100 |
|---|---|---|
| committer | Stefan Holmer <stefan@webrtc.org> | 2015-12-07 09:26:32 +0000 |
| commit | b86d4e4a8dec1eb1b801244a2a97cda66f561d8e (patch) | |
| tree | 97d795f9ebdc3e90bf34ca439d250fcdac1c7a55 /webrtc/call/call.cc | |
| parent | 03f80ebb8310e5f04ced856f7ec8f14b94a0f47e (diff) | |
| download | webrtc-b86d4e4a8dec1eb1b801244a2a97cda66f561d8e.tar.gz | |
Prepare the AudioSendStream to be hooked up to send-side BWE.
This CL contains three changes as a preparation for adding audio send streams
to the send-side BWE:
1. Audio packets are passed through the pacer with high priority. This
is needed to be able to set transport sequence numbers on the packets.
2. A feedback observer is passed to the audio stream's rtcp receiver so
that the BWE can get notified of any BWE feedback being received on the
audio feedback channel.
3. Support for the transport sequence number header extension is added
to audio send streams.
BUG=webrtc:5263,webrtc:5307
R=mflodman@webrtc.org, solenberg@webrtc.org
Review URL: https://codereview.webrtc.org/1479023002 .
Cr-Commit-Position: refs/heads/master@{#10909}
Diffstat (limited to 'webrtc/call/call.cc')
| -rw-r--r-- | webrtc/call/call.cc | 4 |
1 files changed, 2 insertions, 2 deletions
diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc index 4156765d74..9209c7c97a 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); { |