Simplify registration of RTP-header extensions.

webrtc/video/video_send_stream.cc

 /*
  *  Copyright (c) 2013 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/video/video_send_stream.h"
 
 #include <algorithm>
 #include <sstream>
 #include <string>
 #include <vector>
 
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/trace_event.h"
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
 #include "webrtc/modules/bitrate_controller/include/bitrate_controller.h"
 #include "webrtc/modules/congestion_controller/include/congestion_controller.h"
 #include "webrtc/modules/pacing/packet_router.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "webrtc/modules/utility/include/process_thread.h"
 #include "webrtc/video/call_stats.h"
 #include "webrtc/video/video_capture_input.h"
 #include "webrtc/video/vie_remb.h"
 #include "webrtc/video_send_stream.h"
 
 namespace webrtc {
 
 class RtcpIntraFrameObserver;
 class TransportFeedbackObserver;
@@ skipping 186 matching lines @@
   vcm_->RegisterProtectionCallback(vie_channel_.vcm_protection_callback());
 
   call_stats_->RegisterStatsObserver(vie_channel_.GetStatsObserver());
 
   for (size_t i = 0; i < config_.rtp.extensions.size(); ++i) {
     const std::string& extension = config_.rtp.extensions[i].name;
     int id = config_.rtp.extensions[i].id;
     // One-byte-extension local identifiers are in the range 1-14 inclusive.
     RTC_DCHECK_GE(id, 1);
     RTC_DCHECK_LE(id, 14);
-    if (extension == RtpExtension::kTOffset) {
-      RTC_CHECK_EQ(0, vie_channel_.EnableSendTimestampOffset(id));
-    } else if (extension == RtpExtension::kAbsSendTime) {
-      RTC_CHECK_EQ(0, vie_channel_.EnableSendAbsoluteSendTime(id));
-    } else if (extension == RtpExtension::kVideoRotation) {
-      RTC_CHECK_EQ(0, vie_channel_.EnableSendVideoRotation(id));
-    } else if (extension == RtpExtension::kTransportSequenceNumber) {
-      RTC_CHECK_EQ(0, vie_channel_.EnableSendTransportSequenceNumber(id));
-    } else {
-      RTC_NOTREACHED() << "Registering unsupported RTP extension.";
+    for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
+      RTC_CHECK_EQ(0, rtp_rtcp->RegisterSendRtpHeaderExtension(
+                          StringToRtpExtensionType(extension), id));

[stefan-webrtc, 2016/02/26 14:54:11]

Should we no longer fail on trying to set audio extensions on video streams?

[pbos-webrtc, 2016/02/26 15:01:04]

Done.

     }
   }
 
   remb_->AddRembSender(rtp_rtcp_modules_[0]);
   rtp_rtcp_modules_[0]->SetREMBStatus(true);
 
   // Enable NACK, FEC or both.
   const bool enable_protection_nack = config_.rtp.nack.rtp_history_ms > 0;
   bool enable_protection_fec = config_.rtp.fec.red_payload_type != -1;
   // Payload types without picture ID cannot determine that a stream is complete
@@ skipping 366 matching lines @@
     return false;
   }
 
   // Restart the media flow
   vie_encoder_.Restart();
 
   return true;
 }
 }  // namespace internal
 }  // namespace webrtc