Delete RtpData::OnRecoveredPacket, use RecoveredPacketReceiver instead.

webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h

 /*
  *  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 177 matching lines @@
   bool media_has_been_sent;
 };
 
 class RtpData {
  public:
   virtual ~RtpData() {}
 
   virtual int32_t OnReceivedPayloadData(const uint8_t* payload_data,
                                         size_t payload_size,
                                         const WebRtcRTPHeader* rtp_header) = 0;
+};
 
-  virtual bool OnRecoveredPacket(const uint8_t* packet,
-                                 size_t packet_length) = 0;
+// Callback interface for packets recovered by FlexFEC or ULPFEC. In
+// the FlexFEC case, the implementation should be able to demultiplex
+// the recovered RTP packets based on SSRC.
+class RecoveredPacketReceiver {
+ public:
+  virtual void OnRecoveredPacket(const uint8_t* packet, size_t length) = 0;
+
+ protected:
+  virtual ~RecoveredPacketReceiver() = default;
 };
 
 class RtpFeedback {
  public:
   virtual ~RtpFeedback() {}
 
   // Receiving payload change or SSRC change. (return success!)
   /*
   *   channels    - number of channels in codec (1 = mono, 2 = stereo)
   */
@@ skipping 182 matching lines @@
 // Null object version of RtpData.
 class NullRtpData : public RtpData {
  public:
   virtual ~NullRtpData() {}
 
   int32_t OnReceivedPayloadData(const uint8_t* payload_data,
                                 size_t payload_size,
                                 const WebRtcRTPHeader* rtp_header) override {
     return 0;
   }
-
-  bool OnRecoveredPacket(const uint8_t* packet, size_t packet_length) override {
-    return true;
-  }
 };
 
 // Statistics about packet loss for a single directional connection. All values
 // are totals since the connection initiated.
 struct RtpPacketLossStats {
   // The number of packets lost in events where no adjacent packets were also
   // lost.
   uint64_t single_packet_loss_count;
   // The number of events in which more than one adjacent packet was lost.
   uint64_t multiple_packet_loss_event_count;
@@ skipping 28 matching lines @@
 class TransportSequenceNumberAllocator {
  public:
   TransportSequenceNumberAllocator() {}
   virtual ~TransportSequenceNumberAllocator() {}
 
   virtual uint16_t AllocateSequenceNumber() = 0;
 };
 
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_DEFINES_H_