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 cb90a68a0f72e6898fe11726d31d2b99c9770c98..9f95362f213f645b004ca9b86ebb8a336e9805db 100644
--- a/webrtc/audio/audio_receive_stream.cc
+++ b/webrtc/audio/audio_receive_stream.cc
@@ -23,6 +23,7 @@
 #include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_packet_received.h"
 #include "webrtc/voice_engine/channel_proxy.h"
 #include "webrtc/voice_engine/include/voe_base.h"
 #include "webrtc/voice_engine/voice_engine_impl.h"
@@ -67,6 +68,7 @@ AudioReceiveStream::AudioReceiveStream(
     const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
     webrtc::RtcEventLog* event_log)
     : config_(config),
+      rtp_header_extensions_(config.rtp.extensions),

[pthatcher1, 2017/04/29 00:58:46]

Why not just use the config_.rtp.extensions?  Why make a separate copy of them?

[nisse-webrtc, 2017/05/02 09:34:47]

This is also a type conversion, from a vector to the type RtpHeaderExtensionsMap
used by IdentifyExtensions. 

       audio_state_(audio_state) {
   LOG(LS_INFO) << "AudioReceiveStream: " << config_.ToString();
   RTC_DCHECK_NE(config_.voe_channel_id, -1);
@@ -96,15 +98,6 @@ 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);
 }
@@ -305,12 +298,14 @@ 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);
+  return true;
 }
 
 const webrtc::AudioReceiveStream::Config& AudioReceiveStream::config() const {