Simplify registration of RTP-header extensions.

webrtc/video/vie_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 34 matching lines @@
                                            rtp_feedback,
                                            rtp_payload_registry_.get())),
       rtp_receive_statistics_(ReceiveStatistics::Create(clock_)),
       fec_receiver_(FecReceiver::Create(this)),
       rtp_rtcp_(NULL),
       vcm_(module_vcm),
       remote_bitrate_estimator_(remote_bitrate_estimator),
       ntp_estimator_(new RemoteNtpTimeEstimator(clock_)),
       receiving_(false),
       restored_packet_in_use_(false),
-      receiving_ast_enabled_(false),
-      receiving_cvo_enabled_(false),
-      receiving_tsn_enabled_(false),
       last_packet_log_ms_(-1) {}
 
 ViEReceiver::~ViEReceiver() {
   UpdateHistograms();
 }
 
 void ViEReceiver::UpdateHistograms() {
   FecPacketCounter counter = fec_receiver_->GetPacketCounter();
   if (counter.num_packets > 0) {
     RTC_HISTOGRAM_PERCENTAGE(
@@ skipping 81 matching lines @@
 void ViEReceiver::RegisterRtpRtcpModules(
     const std::vector<RtpRtcp*>& rtp_modules) {
   rtc::CritScope lock(&receive_cs_);
   // Only change the "simulcast" modules, the base module can be accessed
   // without a lock whereas the simulcast modules require locking as they can be
   // changed in runtime.
   rtp_rtcp_simulcast_ =
       std::vector<RtpRtcp*>(rtp_modules.begin() + 1, rtp_modules.end());
 }
 
-bool ViEReceiver::EnableReceiveTimestampOffset(int id) {
-  return rtp_header_parser_->RegisterRtpHeaderExtension(
-      kRtpExtensionTransmissionTimeOffset, id);
-}
-
-bool ViEReceiver::EnableReceiveAbsoluteSendTime(int id) {
-  if (rtp_header_parser_->RegisterRtpHeaderExtension(
-      kRtpExtensionAbsoluteSendTime, id)) {
-    receiving_ast_enabled_ = true;
-    return true;
-  } else {
-    return false;
-  }
-}
-
-bool ViEReceiver::EnableReceiveVideoRotation(int id) {
-  if (rtp_header_parser_->RegisterRtpHeaderExtension(
-          kRtpExtensionVideoRotation, id)) {
-    receiving_cvo_enabled_ = true;
-    return true;
-  } else {
-    return false;
-  }
-}
-
-bool ViEReceiver::EnableReceiveTransportSequenceNumber(int id) {
-  if (rtp_header_parser_->RegisterRtpHeaderExtension(
-          kRtpExtensionTransportSequenceNumber, id)) {
-    receiving_tsn_enabled_ = true;
-    return true;
-  } else {
-    return false;
-  }
+void ViEReceiver::EnableReceiveRtpHeaderExtension(const std::string& extension,
+                                                  int id) {
+  RTC_CHECK(rtp_header_parser_->RegisterRtpHeaderExtension(
+      StringToRtpExtensionType(extension), id));

[stefan-webrtc, 2016/02/26 15:19:18]

Verify that this is a valid video extension.

[pbos-webrtc, 2016/02/26 15:20:48]

Done.

 }
 
 int32_t ViEReceiver::OnReceivedPayloadData(const uint8_t* payload_data,
                                            const size_t payload_size,
                                            const WebRtcRTPHeader* rtp_header) {
   RTC_DCHECK(vcm_);
   WebRtcRTPHeader rtp_header_with_ntp = *rtp_header;
   rtp_header_with_ntp.ntp_time_ms =
       ntp_estimator_->Estimate(rtp_header->header.timestamp);
   if (vcm_->IncomingPacket(payload_data,
@@ skipping 233 matching lines @@
       rtp_receive_statistics_->GetStatistician(header.ssrc);
   if (!statistician)
     return false;
   // Check if this is a retransmission.
   int64_t min_rtt = 0;
   rtp_rtcp_->RTT(rtp_receiver_->SSRC(), NULL, NULL, &min_rtt, NULL);
   return !in_order &&
       statistician->IsRetransmitOfOldPacket(header, min_rtt);
 }
 }  // namespace webrtc