Unwrap picture ids in the RtpFrameReferencerFinder.

webrtc/video/rtp_video_stream_receiver.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 75 matching lines @@
   int GetCsrcs(uint32_t* csrcs) const;
 
   RtpReceiver* GetRtpReceiver() const;
   RtpRtcp* rtp_rtcp() const { return rtp_rtcp_.get(); }
 
   void StartReceive();
   void StopReceive();
 
   bool DeliverRtcp(const uint8_t* rtcp_packet, size_t rtcp_packet_length);
 
-  void FrameContinuous(uint16_t seq_num);
+  void FrameContinuous(int64_t seq_num);
 
-  void FrameDecoded(uint16_t seq_num);
+  void FrameDecoded(int64_t seq_num);
 
   void SignalNetworkState(NetworkState state);
 
   // Implements RtpPacketSinkInterface.
   void OnRtpPacket(const RtpPacketReceived& packet) override;
 
   // Implements RtpData.
   int32_t OnReceivedPayloadData(const uint8_t* payload_data,
                                 size_t payload_size,
                                 const WebRtcRTPHeader* rtp_header) override;
@@ skipping 83 matching lines @@
   const std::unique_ptr<RtpRtcp> rtp_rtcp_;
 
   // Members for the new jitter buffer experiment.
   video_coding::OnCompleteFrameCallback* complete_frame_callback_;
   KeyFrameRequestSender* keyframe_request_sender_;
   VCMTiming* timing_;
   std::unique_ptr<NackModule> nack_module_;
   rtc::scoped_refptr<video_coding::PacketBuffer> packet_buffer_;
   std::unique_ptr<video_coding::RtpFrameReferenceFinder> reference_finder_;
   rtc::CriticalSection last_seq_num_cs_;
-  std::map<uint16_t, uint16_t, DescendingSeqNumComp<uint16_t>>
-      last_seq_num_for_pic_id_ GUARDED_BY(last_seq_num_cs_);
+  std::map<int64_t, uint16_t> last_seq_num_for_pic_id_
+      GUARDED_BY(last_seq_num_cs_);
   video_coding::H264SpsPpsTracker tracker_;
   // TODO(johan): Remove pt_codec_params_ once
   // https://bugs.chromium.org/p/webrtc/issues/detail?id=6883 is resolved.
   // Maps a payload type to a map of out-of-band supplied codec parameters.
   std::map<uint8_t, std::map<std::string, std::string>> pt_codec_params_;
   int16_t last_payload_type_ = -1;
 
   bool has_received_frame_;
 
   std::vector<RtpPacketSinkInterface*> secondary_sinks_
       GUARDED_BY(worker_task_checker_);
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_VIDEO_RTP_VIDEO_STREAM_RECEIVER_H_