RtpRtcp: Remove the SetSendREDPayloadType and SendREDPayloadType methods

webrtc/modules/rtp_rtcp/source/rtp_sender.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 374 matching lines @@
 }
 
 int32_t RTPSender::CheckPayloadType(int8_t payload_type,
                                     RtpVideoCodecTypes* video_type) {
   rtc::CritScope lock(&send_critsect_);
 
   if (payload_type < 0) {
     LOG(LS_ERROR) << "Invalid payload_type " << payload_type;
     return -1;
   }
-  if (audio_configured_) {
-    int8_t red_pl_type = -1;
-    if (audio_->RED(&red_pl_type) == 0) {
-      // We have configured RED.
-      if (red_pl_type == payload_type) {
-        // And it's a match...
-        return 0;
-      }
-    }
-  }
   if (payload_type_ == payload_type) {
     if (!audio_configured_) {
       *video_type = video_->VideoCodecType();
     }
     return 0;
   }
   std::map<int8_t, RtpUtility::Payload*>::iterator it =
       payload_type_map_.find(payload_type);
   if (it == payload_type_map_.end()) {
     LOG(LS_WARNING) << "Payload type " << static_cast<int>(payload_type)
@@ skipping 1202 matching lines @@
   if (!audio_configured_) {
     return -1;
   }
   return audio_->SetAudioPacketSize(packet_size_samples);
 }
 
 int32_t RTPSender::SetAudioLevel(uint8_t level_d_bov) {
   return audio_->SetAudioLevel(level_d_bov);
 }
 
-int32_t RTPSender::SetRED(int8_t payload_type) {
-  if (!audio_configured_) {
-    return -1;
-  }
-  return audio_->SetRED(payload_type);
-}
-
-int32_t RTPSender::RED(int8_t *payload_type) const {
-  if (!audio_configured_) {
-    return -1;
-  }
-  return audio_->RED(payload_type);
-}
-
 RtpVideoCodecTypes RTPSender::VideoCodecType() const {
   assert(!audio_configured_ && "Sender is an audio stream!");
   return video_->VideoCodecType();
 }
 
 void RTPSender::SetGenericFECStatus(bool enable,
                                     uint8_t payload_type_red,
                                     uint8_t payload_type_fec) {
   RTC_DCHECK(!audio_configured_);
   video_->SetGenericFECStatus(enable, payload_type_red, payload_type_fec);
@@ skipping 107 matching lines @@
   rtc::CritScope lock(&send_critsect_);
 
   RtpState state;
   state.sequence_number = sequence_number_rtx_;
   state.start_timestamp = start_timestamp_;
 
   return state;
 }
 
 }  // namespace webrtc