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())

[danilchap, 2016/11/02 13:56:52]

this look like hack 'If no payload type mapping existed, the payload type of the
underlying media would be used.'
Is it still needed, or is it safe to reject sending rtx packet:
if (kv == rtx_map_.end())
  return nullptr;

[brandtr, 2016/11/03 12:02:48]

It should be safe to reject sending it, because all associated payload types are
set here:
https://cs.chromium.org/chromium/src/third_party/webrtc/video/video_send_stre...

+holmer: can you verify this?

       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 =
@@ skipping 58 matching lines @@
   rtc::CritScope lock(&send_critsect_);
 
   RtpState state;
   state.sequence_number = sequence_number_rtx_;
   state.start_timestamp = timestamp_offset_;
 
   return state;
 }
 
 }  // namespace webrtc