Initial implementation of RtpTransportControllerReceive and related interfaces.

webrtc/audio/audio_receive_stream.cc

Index: webrtc/audio/audio_receive_stream.cc
diff --git a/webrtc/audio/audio_receive_stream.cc b/webrtc/audio/audio_receive_stream.cc
index fbd61ccfd4791e3386a9cd68565c053782fe2616..a98e04fa4d216c258946965c1ca8e68d1ff933fc 100644
--- a/webrtc/audio/audio_receive_stream.cc
+++ b/webrtc/audio/audio_receive_stream.cc
@@ -67,6 +67,7 @@ AudioReceiveStream::AudioReceiveStream(
     const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
     webrtc::RtcEventLog* event_log)
     : config_(config),
+      rtp_header_extensions_(config.rtp.extensions),
       audio_state_(audio_state) {
   LOG(LS_INFO) << "AudioReceiveStream: " << config_.ToString();
   RTC_DCHECK_NE(config_.voe_channel_id, -1);
@@ -96,15 +97,6 @@ AudioReceiveStream::AudioReceiveStream(
   channel_proxy_->RegisterExternalTransport(config.rtcp_send_transport);
   channel_proxy_->SetReceiveCodecs(config.decoder_map);
 
-  for (const auto& extension : config.rtp.extensions) {
-    if (extension.uri == RtpExtension::kAudioLevelUri) {
-      channel_proxy_->SetReceiveAudioLevelIndicationStatus(true, extension.id);
-    } else if (extension.uri == RtpExtension::kTransportSequenceNumberUri) {
-      channel_proxy_->EnableReceiveTransportSequenceNumber(extension.id);
-    } else {
-      RTC_NOTREACHED() << "Unsupported RTP extension.";
-    }
-  }
   // Configure bandwidth estimation.
   channel_proxy_->RegisterReceiverCongestionControlObjects(packet_router);
 }
@@ -300,12 +292,14 @@ bool AudioReceiveStream::DeliverRtcp(const uint8_t* packet, size_t length) {
   return channel_proxy_->ReceivedRTCPPacket(packet, length);
 }
 
-void AudioReceiveStream::OnRtpPacket(const RtpPacketReceived& packet) {
+bool AudioReceiveStream::OnRtpPacketReceive(RtpPacketReceived* packet) {
   // TODO(solenberg): Tests call this function on a network thread, libjingle
   // calls on the worker thread. We should move towards always using a network
   // thread. Then this check can be enabled.
   // RTC_DCHECK(!thread_checker_.CalledOnValidThread());
-  channel_proxy_->OnRtpPacket(packet);
+  packet->IdentifyExtensions(rtp_header_extensions_);
+  channel_proxy_->OnRtpPacket(*packet);

[nisse-webrtc, 2017/04/19 08:35:43]

This is where we need channel_proxy_ to implement OnRtpPacket.

Registering the voe::Channel directly doesn't seem practical at the moment, when
registration is done by Call.

Next step on the receive side is to pass an
|RtpTransportControllerReceiveInterface*| from Call to the receive stream
constructors, which would give a bit more flexibility.

+  return true;
 }
 
 const webrtc::AudioReceiveStream::Config& AudioReceiveStream::config() const {