Initial implementation of RtpTransportControllerReceive and related interfaces.

webrtc/video/rtp_video_stream_receiver.cc

 /*
  *  Copyright (c) 2012 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 80 matching lines @@
     ProcessThread* process_thread,
     NackSender* nack_sender,
     KeyFrameRequestSender* keyframe_request_sender,
     video_coding::OnCompleteFrameCallback* complete_frame_callback,
     VCMTiming* timing)
     : clock_(Clock::GetRealTimeClock()),
       config_(*config),
       packet_router_(packet_router),
       process_thread_(process_thread),
       ntp_estimator_(clock_),
-      rtp_header_parser_(RtpHeaderParser::Create()),
+      rtp_header_extensions_(config_.rtp.extensions),
       rtp_receiver_(RtpReceiver::CreateVideoReceiver(clock_,
                                                      this,
                                                      this,
                                                      &rtp_payload_registry_)),
       rtp_receive_statistics_(rtp_receive_statistics),
       ulpfec_receiver_(UlpfecReceiver::Create(config->rtp.remote_ssrc, this)),
       receiving_(false),
       last_packet_log_ms_(-1),
       rtp_rtcp_(CreateRtpRtcpModule(rtp_receive_statistics_,
                                     transport,
@@ skipping 15 matching lines @@
   RTC_DCHECK(config_.rtp.remote_ssrc != 0);
   // TODO(pbos): What's an appropriate local_ssrc for receive-only streams?
   RTC_DCHECK(config_.rtp.local_ssrc != 0);
   RTC_DCHECK(config_.rtp.remote_ssrc != config_.rtp.local_ssrc);
 
   rtp_rtcp_->SetRTCPStatus(config_.rtp.rtcp_mode);
   rtp_rtcp_->SetSSRC(config_.rtp.local_ssrc);
   rtp_rtcp_->SetRemoteSSRC(config_.rtp.remote_ssrc);
   rtp_rtcp_->SetKeyFrameRequestMethod(kKeyFrameReqPliRtcp);
 
-  for (size_t i = 0; i < config_.rtp.extensions.size(); ++i) {
-    EnableReceiveRtpHeaderExtension(config_.rtp.extensions[i].uri,
-                                    config_.rtp.extensions[i].id);
-  }
-
   static const int kMaxPacketAgeToNack = 450;
   const int max_reordering_threshold = (config_.rtp.nack.rtp_history_ms > 0)
                                            ? kMaxPacketAgeToNack
                                            : kDefaultMaxReorderingThreshold;
   rtp_receive_statistics_->SetMaxReorderingThreshold(max_reordering_threshold);
 
   if (config_.rtp.rtx_ssrc) {
     // Needed for rtp_payload_registry_.RtxEnabled().
     rtp_payload_registry_.SetRtxSsrc(config_.rtp.rtx_ssrc);
   }
@@ skipping 122 matching lines @@
 
   packet_buffer_->InsertPacket(&packet);
   return 0;
 }
 
 // TODO(nisse): Try to delete this method. Obstacles: It is used by
 // ParseAndHandleEncapsulatingHeader, for handling Rtx packets, and
 // for callbacks from |ulpfec_receiver_|.
 void RtpVideoStreamReceiver::OnRecoveredPacket(const uint8_t* rtp_packet,
                                                size_t rtp_packet_length) {
+  RtpPacketReceived packet;
+  if (!packet.Parse(rtp_packet, rtp_packet_length))
+    return;
+  packet.IdentifyExtensions(rtp_header_extensions_);
+  packet.set_payload_type_frequency(kVideoPayloadTypeFrequency);
+
   RTPHeader header;
-  if (!rtp_header_parser_->Parse(rtp_packet, rtp_packet_length, &header)) {
-    return;
-  }
-  header.payload_type_frequency = kVideoPayloadTypeFrequency;
+  packet.GetHeader(&header);
   bool in_order = IsPacketInOrder(header);
   ReceivePacket(rtp_packet, rtp_packet_length, header, in_order);
 }
 
 // TODO(pbos): Remove as soon as audio can handle a changing payload type
 // without this callback.
 int32_t RtpVideoStreamReceiver::OnInitializeDecoder(
     const int8_t payload_type,
     const char payload_name[RTP_PAYLOAD_NAME_SIZE],
     const int frequency,
@@ skipping 343 matching lines @@
         "WebRTC.Video.ReceivedFecPacketsInPercent",
         static_cast<int>(counter.num_fec_packets * 100 / counter.num_packets));
   }
   if (counter.num_fec_packets > 0) {
     RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.RecoveredMediaPacketsInPercentOfFec",
                              static_cast<int>(counter.num_recovered_packets *
                                               100 / counter.num_fec_packets));
   }
 }
 
-void RtpVideoStreamReceiver::EnableReceiveRtpHeaderExtension(
-    const std::string& extension, int id) {
-  // One-byte-extension local identifiers are in the range 1-14 inclusive.
-  RTC_DCHECK_GE(id, 1);
-  RTC_DCHECK_LE(id, 14);
-  RTC_DCHECK(RtpExtension::IsSupportedForVideo(extension));
-  RTC_CHECK(rtp_header_parser_->RegisterRtpHeaderExtension(
-      StringToRtpExtensionType(extension), id));
-}
-
 void RtpVideoStreamReceiver::InsertSpsPpsIntoTracker(uint8_t payload_type) {
   auto codec_params_it = pt_codec_params_.find(payload_type);
   if (codec_params_it == pt_codec_params_.end())
     return;
 
   LOG(LS_INFO) << "Found out of band supplied codec parameters for"
                << " payload type: " << static_cast<int>(payload_type);
 
   H264SpropParameterSets sprop_decoder;
   auto sprop_base64_it =
       codec_params_it->second.find(cricket::kH264FmtpSpropParameterSets);
 
   if (sprop_base64_it == codec_params_it->second.end())
     return;
 
   if (!sprop_decoder.DecodeSprop(sprop_base64_it->second.c_str()))
     return;
 
   tracker_.InsertSpsPpsNalus(sprop_decoder.sps_nalu(),
                              sprop_decoder.pps_nalu());
 }
 
 }  // namespace webrtc