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}
diff --git a/webrtc/voice_engine/channel_proxy.h b/webrtc/voice_engine/channel_proxy.h
index 3668de4..fa33e6c 100644
--- a/webrtc/voice_engine/channel_proxy.h
+++ b/webrtc/voice_engine/channel_proxy.h
@@ -19,6 +19,11 @@
#include <vector>
namespace webrtc {
+
+class PacketRouter;
+class RtpPacketSender;
+class TransportFeedbackObserver;
+
namespace voe {
class Channel;
@@ -41,8 +46,13 @@
virtual void SetRTCP_CNAME(const std::string& c_name);
virtual void SetSendAbsoluteSenderTimeStatus(bool enable, int id);
virtual void SetSendAudioLevelIndicationStatus(bool enable, int id);
+ virtual void EnableSendTransportSequenceNumber(int id);
virtual void SetReceiveAbsoluteSenderTimeStatus(bool enable, int id);
virtual void SetReceiveAudioLevelIndicationStatus(bool enable, int id);
+ virtual void SetCongestionControlObjects(
+ RtpPacketSender* rtp_packet_sender,
+ TransportFeedbackObserver* transport_feedback_observer,
+ PacketRouter* packet_router);
virtual CallStatistics GetRTCPStatistics() const;
virtual std::vector<ReportBlock> GetRemoteRTCPReportBlocks() const;