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

webrtc/audio/audio_receive_stream.cc

 /*
  *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 49 matching lines @@
   return ss.str();
 }
 
 namespace internal {
 AudioReceiveStream::AudioReceiveStream(
     PacketRouter* packet_router,
     const webrtc::AudioReceiveStream::Config& config,
     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);
   RTC_DCHECK(audio_state_.get());
   RTC_DCHECK(packet_router);
 
   module_process_thread_checker_.DetachFromThread();
 
   VoiceEngineImpl* voe_impl = static_cast<VoiceEngineImpl*>(voice_engine());
   channel_proxy_ = voe_impl->GetChannelProxy(config_.voe_channel_id);
@@ skipping 215 matching lines @@
 }
 
 bool AudioReceiveStream::DeliverRtcp(const uint8_t* packet, size_t length) {
   // 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_->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 {
   RTC_DCHECK_RUN_ON(&worker_thread_checker_);
   return config_;
 }
 
 VoiceEngine* AudioReceiveStream::voice_engine() const {
   auto* voice_engine = audio_state()->voice_engine();
   RTC_DCHECK(voice_engine);
   return voice_engine;
 }
 
 internal::AudioState* AudioReceiveStream::audio_state() const {
   auto* audio_state = static_cast<internal::AudioState*>(audio_state_.get());
   RTC_DCHECK(audio_state);
   return audio_state;
 }
 
 int AudioReceiveStream::SetVoiceEnginePlayout(bool playout) {
   ScopedVoEInterface<VoEBase> base(voice_engine());
   if (playout) {
     return base->StartPlayout(config_.voe_channel_id);
   } else {
     return base->StopPlayout(config_.voe_channel_id);
   }
 }
 }  // namespace internal
 }  // namespace webrtc