Support multiple rtx codecs.

webrtc/modules/rtp_rtcp/source/rtp_payload_registry.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/modules/rtp_rtcp/include/rtp_payload_registry.h"
 
 #include "webrtc/base/logging.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
 
 namespace webrtc {
 
 RTPPayloadRegistry::RTPPayloadRegistry(RTPPayloadStrategy* rtp_payload_strategy)
     : crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
       rtp_payload_strategy_(rtp_payload_strategy),
       red_payload_type_(-1),
       ulpfec_payload_type_(-1),
       incoming_payload_type_(-1),
       last_received_payload_type_(-1),
       last_received_media_payload_type_(-1),
       rtx_(false),
       rtx_payload_type_(-1),
-      use_rtx_payload_mapping_on_restore_(false),
       ssrc_rtx_(0) {}
 
 RTPPayloadRegistry::~RTPPayloadRegistry() {
   while (!payload_type_map_.empty()) {
     RtpUtility::PayloadTypeMap::iterator it = payload_type_map_.begin();
     delete it->second;
     payload_type_map_.erase(it);
   }
 }
 
@@ skipping 225 matching lines @@
   ByteWriter<uint16_t>::WriteBigEndian(restored_packet + 2,
                                        original_sequence_number);
   ByteWriter<uint32_t>::WriteBigEndian(restored_packet + 8, original_ssrc);
 
   CriticalSectionScoped cs(crit_sect_.get());
   if (!rtx_)
     return true;
 
   int associated_payload_type;
   auto apt_mapping = rtx_payload_type_map_.find(header.payloadType);
-  if (use_rtx_payload_mapping_on_restore_ &&
-      apt_mapping != rtx_payload_type_map_.end()) {
-    associated_payload_type = apt_mapping->second;
-  } else {
-    // In the future, this will be a bug. For now, just assume this RTX packet
-    // matches the last non-RTX payload type we received. There are cases where
-    // this could break, especially where RTX is sent outside of NACKing (e.g.
-    // padding with redundant payloads).
-    if (rtx_payload_type_ == -1 || incoming_payload_type_ == -1) {
-      LOG(LS_WARNING) << "Incorrect RTX configuration, dropping packet.";
-      return false;
-    }
-    associated_payload_type = incoming_payload_type_;
+  if (apt_mapping == rtx_payload_type_map_.end())
+    return false;
+  associated_payload_type = apt_mapping->second;
+  if (red_payload_type_ != -1) {

[pbos-webrtc, 2016/02/01 14:32:19]

Does this path still have test coverage? It should until we decide to remove it.
I couldn't easily see that with the test removal.

[stefan-webrtc, 2016/02/01 15:26:08]

I added a test which specifically tests this path.

+    // Assume red will be used if it's configured.
+    // This is a workaround for a Chrome sdp bug where rtx is associated
+    // with the media codec even though media is sent over red.
+    // TODO(holmer): Remove once the Chrome versions exposing this bug are
+    // old enough.

[pbos-webrtc, 2016/02/01 14:32:19]

Put which Chrome versions these are, so you can remember when it's old enough.

[stefan-webrtc, 2016/02/01 15:26:08]

Done.

+    associated_payload_type = red_payload_type_;
   }
-
   restored_packet[1] = static_cast<uint8_t>(associated_payload_type);
   if (header.markerBit) {
     restored_packet[1] |= kRtpMarkerBitMask;  // Marker bit is set.
   }
   return true;
 }
 
 void RTPPayloadRegistry::SetRtxSsrc(uint32_t ssrc) {
   CriticalSectionScoped cs(crit_sect_.get());
   ssrc_rtx_ = ssrc;
@@ skipping 182 matching lines @@
 RTPPayloadStrategy* RTPPayloadStrategy::CreateStrategy(
     const bool handling_audio) {
   if (handling_audio) {
     return new RTPPayloadAudioStrategy();
   } else {
     return new RTPPayloadVideoStrategy();
   }
 }
 
 }  // namespace webrtc