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

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 17da10f35789eb4c864ca6f696d3cfd09da91e9e..4b90473f2aecdb3c5eb4378cf58c131804dea7c4 100644
--- a/webrtc/audio/audio_receive_stream.cc
+++ b/webrtc/audio/audio_receive_stream.cc
@@ -26,10 +26,6 @@
 #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 {
@@ -74,14 +70,12 @@ AudioReceiveStream::AudioReceiveStream(
     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();
 
@@ -112,14 +106,8 @@ AudioReceiveStream::AudioReceiveStream(
   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.";
     }
@@ -325,11 +313,6 @@ bool AudioReceiveStream::DeliverRtp(const uint8_t* packet,
   // 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);
 }