Remove unnecessary RTPHeaderParser, following https://codereview.webrtc.org/2659563002/

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.
  */
 
 #include "webrtc/audio/audio_receive_stream.h"
 
 #include <string>
 #include <utility>
 
 #include "webrtc/api/call/audio_sink.h"
 #include "webrtc/audio/audio_send_stream.h"
 #include "webrtc/audio/audio_state.h"
 #include "webrtc/audio/conversion.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/timeutils.h"
 #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/voice_engine/channel_proxy.h"
 #include "webrtc/voice_engine/include/voe_base.h"
 #include "webrtc/voice_engine/include/voe_codec.h"
-#include "webrtc/voice_engine/include/voe_neteq_stats.h"
-#include "webrtc/voice_engine/include/voe_rtp_rtcp.h"
-#include "webrtc/voice_engine/include/voe_video_sync.h"
-#include "webrtc/voice_engine/include/voe_volume_control.h"
 #include "webrtc/voice_engine/voice_engine_impl.h"
 
 namespace webrtc {
 
 std::string AudioReceiveStream::Config::Rtp::ToString() const {
   std::stringstream ss;
   ss << "{remote_ssrc: " << remote_ssrc;
   ss << ", local_ssrc: " << local_ssrc;
   ss << ", transport_cc: " << (transport_cc ? "on" : "off");
   ss << ", nack: " << nack.ToString();
@@ skipping 24 matching lines @@
 
 namespace internal {
 AudioReceiveStream::AudioReceiveStream(
     PacketRouter* packet_router,
     RemoteBitrateEstimator* remote_bitrate_estimator,
     const webrtc::AudioReceiveStream::Config& config,
     const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
     webrtc::RtcEventLog* event_log)
     : remote_bitrate_estimator_(remote_bitrate_estimator),
       config_(config),
-      audio_state_(audio_state),
-      rtp_header_parser_(RtpHeaderParser::Create()) {
+      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);
   RTC_DCHECK(remote_bitrate_estimator);
-  RTC_DCHECK(rtp_header_parser_);
 
   module_process_thread_checker_.DetachFromThread();
 
   VoiceEngineImpl* voe_impl = static_cast<VoiceEngineImpl*>(voice_engine());
   channel_proxy_ = voe_impl->GetChannelProxy(config_.voe_channel_id);
   channel_proxy_->SetRtcEventLog(event_log);
   channel_proxy_->SetLocalSSRC(config.rtp.local_ssrc);
   // TODO(solenberg): Config NACK history window (which is a packet count),
   // using the actual packet size for the configured codec.
   channel_proxy_->SetNACKStatus(config_.rtp.nack.rtp_history_ms != 0,
@@ skipping 10 matching lines @@
 
   channel_proxy_->RegisterExternalTransport(config.rtcp_send_transport);
 
   for (const auto& kv : config.decoder_map) {
     channel_proxy_->SetRecPayloadType(kv.first, kv.second);
   }
 
   for (const auto& extension : config.rtp.extensions) {
     if (extension.uri == RtpExtension::kAudioLevelUri) {
       channel_proxy_->SetReceiveAudioLevelIndicationStatus(true, extension.id);
-      bool registered = rtp_header_parser_->RegisterRtpHeaderExtension(
-          kRtpExtensionAudioLevel, extension.id);
-      RTC_DCHECK(registered);
     } else if (extension.uri == RtpExtension::kTransportSequenceNumberUri) {
       channel_proxy_->EnableReceiveTransportSequenceNumber(extension.id);
-      bool registered = rtp_header_parser_->RegisterRtpHeaderExtension(
-          kRtpExtensionTransportSequenceNumber, extension.id);
-      RTC_DCHECK(registered);
     } else {
       RTC_NOTREACHED() << "Unsupported RTP extension.";
     }
   }
   // Configure bandwidth estimation.
   channel_proxy_->RegisterReceiverCongestionControlObjects(packet_router);
 }
 
 AudioReceiveStream::~AudioReceiveStream() {
   RTC_DCHECK_RUN_ON(&worker_thread_checker_);
@@ skipping 185 matching lines @@
   return channel_proxy_->ReceivedRTCPPacket(packet, length);
 }
 
 bool AudioReceiveStream::DeliverRtp(const uint8_t* packet,
                                     size_t length,
                                     const PacketTime& packet_time) {
   // 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());
-  RTPHeader header;
-  if (!rtp_header_parser_->Parse(packet, length, &header)) {
-    return false;
-  }
-

[nisse-webrtc, 2017/02/01 14:50:17]

I would have expected Call to pass on its parsed headers down the pipeline. Are
the headers not needed at all, or are they parsed again further down?

[the sun, 2017/02/01 15:12:06]

They are parsed again inside voe::Channel::ReceivedRTPPacket()

[nisse-webrtc, 2017/02/02 08:10:11]

I see. An opportunity for later improvement...

   return channel_proxy_->ReceivedRTPPacket(packet, length, packet_time);
 }
 
 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();
@@ skipping 10 matching lines @@
 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