Remove RED/RTX workaround from sender/receiver and VideoEngine2.

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 1140 matching lines @@
   // TODO(danilchap): Create rtx packet with extra capacity for SRTP
   // when transport interface would be updated to take buffer class.
   std::unique_ptr<RtpPacketToSend> rtx_packet(new RtpPacketToSend(
       &rtp_header_extension_map_, packet.size() + kRtxHeaderSize));
   // Add original RTP header.
   rtx_packet->CopyHeaderFrom(packet);
   {
     rtc::CritScope lock(&send_critsect_);
     if (!sending_media_)
       return nullptr;
-    // Replace payload type, if a specific type is set for RTX.
+
+    // Replace payload type.
     auto kv = rtx_payload_type_map_.find(packet.PayloadType());
-
-    // Use rtx mapping associated with media codec if we can't find one,
-    // assume it's red.
-    // TODO(holmer): Remove once old Chrome versions don't rely on this.
     if (kv == rtx_payload_type_map_.end())
-      kv = rtx_payload_type_map_.find(payload_type_);
-    if (kv != rtx_payload_type_map_.end())
-      rtx_packet->SetPayloadType(kv->second);
+      return nullptr;
+    rtx_packet->SetPayloadType(kv->second);
 
     // Replace sequence number.
     rtx_packet->SetSequenceNumber(sequence_number_rtx_++);
 
     // Replace SSRC.
     rtx_packet->SetSsrc(ssrc_rtx_);
   }
 
   uint8_t* rtx_payload =
       rtx_packet->AllocatePayload(packet.payload_size() + kRtxHeaderSize);
@@ skipping 57 matching lines @@
   rtc::CritScope lock(&send_critsect_);
 
   RtpState state;
   state.sequence_number = sequence_number_rtx_;
   state.start_timestamp = timestamp_offset_;
 
   return state;
 }
 
 }  // namespace webrtc