Simplify registration of RTP-header extensions.

webrtc/video/vie_channel.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 359 matching lines @@
   }
 }
 
 int ViEChannel::GetRequiredNackListSize(int target_delay_ms) {
   // The max size of the nack list should be large enough to accommodate the
   // the number of packets (frames) resulting from the increased delay.
   // Roughly estimating for ~40 packets per frame @ 30fps.
   return target_delay_ms * 40 * 30 / 1000;
 }
 
-int ViEChannel::EnableSendTimestampOffset(int id) {
-  // Enable the extension.
-  int error = 0;
-  for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
-    error |= rtp_rtcp->RegisterSendRtpHeaderExtension(
-        kRtpExtensionTransmissionTimeOffset, id);
-  }
-  return error;
-}
-
-int ViEChannel::EnableSendAbsoluteSendTime(int id) {
-  // Enable the extension.
-  int error = 0;
-  for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
-    error |= rtp_rtcp->RegisterSendRtpHeaderExtension(
-        kRtpExtensionAbsoluteSendTime, id);
-  }
-  return error;
-}
-
-int ViEChannel::EnableSendVideoRotation(int id) {
-  // Enable the extension.
-  int error = 0;
-  for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
-    error |= rtp_rtcp->RegisterSendRtpHeaderExtension(
-        kRtpExtensionVideoRotation, id);
-  }
-  return error;
-}
-
-int ViEChannel::EnableSendTransportSequenceNumber(int id) {
-  RTC_DCHECK(sender_);
-  // Enable the extension.
-  int error = 0;
-  for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
-    error |= rtp_rtcp->RegisterSendRtpHeaderExtension(
-        kRtpExtensionTransportSequenceNumber, id);
-  }
-  return error;
-}
-
 RtpState ViEChannel::GetRtpStateForSsrc(uint32_t ssrc) const {
   RTC_DCHECK(!rtp_rtcp_modules_[0]->Sending());
   RtpState rtp_state;
   for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
     if (rtp_rtcp->GetRtpStateForSsrc(ssrc, &rtp_state))
       return rtp_state;
   }
   LOG(LS_ERROR) << "Couldn't get RTP state for ssrc: " << ssrc;
   return rtp_state;
 }
@@ skipping 274 matching lines @@
   rtc::CritScope lock(&crit_);
   receive_stats_callback_ = receive_statistics_proxy;
 }
 
 void ViEChannel::SetIncomingVideoStream(
     IncomingVideoStream* incoming_video_stream) {
   rtc::CritScope lock(&crit_);
   incoming_video_stream_ = incoming_video_stream;
 }
 }  // namespace webrtc