RtpPacketReceiver base class and OnRtpPacket, with a pre-parsed RTP packet.

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 05d6edfa4c1fccd33b953ec2e38c3c456eefcb83..e071b69f6cebd5fd0a323e0d4ec28b8ba52b1ae5 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_config_(config.rtp.extensions, config.rtp.transport_cc),
       audio_state_(audio_state) {
   LOG(LS_INFO) << "AudioReceiveStream: " << config_.ToString();
   RTC_DCHECK_NE(config_.voe_channel_id, -1);
@@ -302,14 +303,16 @@ bool AudioReceiveStream::DeliverRtcp(const uint8_t* packet, size_t length) {
   return channel_proxy_->ReceivedRTCPPacket(packet, length);
 }
 
-bool AudioReceiveStream::DeliverRtp(const uint8_t* packet,
-                                    size_t length,
-                                    const PacketTime& packet_time) {
+const RtpPacketReceiver::RtpConfig& AudioReceiveStream::rtp_config() const {
+  return rtp_config_;
+}
+
+bool AudioReceiveStream::OnRtpPacket(const 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());
-  return channel_proxy_->ReceivedRTPPacket(packet, length, packet_time);
+  return channel_proxy_->OnRtpPacket(packet);
 }
 
 const webrtc::AudioReceiveStream::Config& AudioReceiveStream::config() const {