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

webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h

 /*
  *  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.
  */
 
 #ifndef WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_PAYLOAD_REGISTRY_H_
 #define WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_PAYLOAD_REGISTRY_H_
 
 #include <map>
 #include <memory>
+#include <set>
 
 #include "webrtc/base/criticalsection.h"
+#include "webrtc/base/deprecation.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
 
 namespace webrtc {
 
 // This strategy deals with the audio/video-specific aspects
 // of payload handling.
 class RTPPayloadStrategy {
  public:
   virtual ~RTPPayloadStrategy() {}
@@ skipping 52 matching lines @@
   bool GetRtxSsrc(uint32_t* ssrc) const;
 
   void SetRtxPayloadType(int payload_type, int associated_payload_type);
 
   bool IsRtx(const RTPHeader& header) const;
 
   bool RestoreOriginalPacket(uint8_t* restored_packet,
                              const uint8_t* packet,
                              size_t* packet_length,
                              uint32_t original_ssrc,
-                             const RTPHeader& header) const;
+                             const RTPHeader& header);
 
   bool IsRed(const RTPHeader& header) const;
 
   // Returns true if the media of this RTP packet is encapsulated within an
   // extra header, such as RTX or RED.
   bool IsEncapsulated(const RTPHeader& header) const;
 
   bool GetPayloadSpecifics(uint8_t payload_type, PayloadUnion* payload) const;
 
   int GetPayloadTypeFrequency(uint8_t payload_type) const;
@@ skipping 29 matching lines @@
   void set_last_received_payload_type(int8_t last_received_payload_type) {
     rtc::CritScope cs(&crit_sect_);
     last_received_payload_type_ = last_received_payload_type;
   }
 
   int8_t last_received_media_payload_type() const {
     rtc::CritScope cs(&crit_sect_);
     return last_received_media_payload_type_;
   }
 
-  bool use_rtx_payload_mapping_on_restore() const {
-    rtc::CritScope cs(&crit_sect_);
-    return use_rtx_payload_mapping_on_restore_;
-  }
-
-  void set_use_rtx_payload_mapping_on_restore(bool val) {
-    rtc::CritScope cs(&crit_sect_);
-    use_rtx_payload_mapping_on_restore_ = val;
-  }
+  RTC_DEPRECATED void set_use_rtx_payload_mapping_on_restore(bool val) {}
 
  private:
   // Prunes the payload type map of the specific payload type, if it exists.
   void DeregisterAudioCodecOrRedTypeRegardlessOfPayloadType(
       const char payload_name[RTP_PAYLOAD_NAME_SIZE],
       size_t payload_name_length,
       uint32_t frequency,
       size_t channels,
       uint32_t rate);
 
   bool IsRtxInternal(const RTPHeader& header) const;
 
   rtc::CriticalSection crit_sect_;
   RtpUtility::PayloadTypeMap payload_type_map_;
   std::unique_ptr<RTPPayloadStrategy> rtp_payload_strategy_;
   int8_t red_payload_type_;
   int8_t ulpfec_payload_type_;
   int8_t incoming_payload_type_;
   int8_t last_received_payload_type_;
   int8_t last_received_media_payload_type_;
   bool rtx_;
-  // TODO(changbin): Remove rtx_payload_type_ once interop with old clients that
-  // only understand one RTX PT is no longer needed.
-  int rtx_payload_type_;
   // Mapping rtx_payload_type_map_[rtx] = associated.
   std::map<int, int> rtx_payload_type_map_;
-  // When true, use rtx_payload_type_map_ when restoring RTX packets to get the
-  // correct payload type.
-  bool use_rtx_payload_mapping_on_restore_;
   uint32_t ssrc_rtx_;
+  // Only warn once per payload type, if an RTX packet is received but
+  // no associated payload type found in |rtx_payload_type_map_|.
+  std::set<int> payload_types_with_suppressed_warnings_ GUARDED_BY(crit_sect_);
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_PAYLOAD_REGISTRY_H_