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.cc b/webrtc/voice_engine/channel_proxy.cc
index 1772ad5..68fbf38 100644
--- a/webrtc/voice_engine/channel_proxy.cc
+++ b/webrtc/voice_engine/channel_proxy.cc
@@ -52,6 +52,11 @@
   RTC_DCHECK_EQ(0, error);
 }
 
+void ChannelProxy::EnableSendTransportSequenceNumber(int id) {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  channel()->EnableSendTransportSequenceNumber(id);
+}
+
 void ChannelProxy::SetReceiveAbsoluteSenderTimeStatus(bool enable, int id) {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   int error = channel()->SetReceiveAbsoluteSenderTimeStatus(enable, id);
@@ -64,6 +69,15 @@
   RTC_DCHECK_EQ(0, error);
 }
 
+void ChannelProxy::SetCongestionControlObjects(
+    RtpPacketSender* rtp_packet_sender,
+    TransportFeedbackObserver* transport_feedback_observer,
+    PacketRouter* packet_router) {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  channel()->SetCongestionControlObjects(
+      rtp_packet_sender, transport_feedback_observer, packet_router);
+}
+
 CallStatistics ChannelProxy::GetRTCPStatistics() const {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   CallStatistics stats = {0};
@@ -124,5 +138,6 @@
   RTC_DCHECK(channel_owner_.channel());
   return channel_owner_.channel();
 }
+
 }  // namespace voe
 }  // namespace webrtc