New jitter buffer experiment.

webrtc/video/rtp_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.
  */
 
 #ifndef WEBRTC_VIDEO_RTP_STREAM_RECEIVER_H_
 #define WEBRTC_VIDEO_RTP_STREAM_RECEIVER_H_
 
 #include <list>
+#include <map>
 #include <memory>
 #include <string>
 #include <vector>
 
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/criticalsection.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/rtp_rtcp/include/receive_statistics.h"
 #include "webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "webrtc/modules/video_coding/h264_sps_pps_tracker.h"
 #include "webrtc/modules/video_coding/include/video_coding_defines.h"
+#include "webrtc/modules/video_coding/packet_buffer.h"
+#include "webrtc/modules/video_coding/rtp_frame_reference_finder.h"
+#include "webrtc/modules/video_coding/sequence_number_util.h"
 #include "webrtc/typedefs.h"
 #include "webrtc/video_receive_stream.h"
 
 namespace webrtc {
 
+class NackModule;
 class PacedSender;
 class PacketRouter;
 class ProcessThread;
-class RemoteNtpTimeEstimator;
 class ReceiveStatistics;
 class ReceiveStatisticsProxy;
 class RemoteBitrateEstimator;
+class RemoteNtpTimeEstimator;
 class RtcpRttStats;
 class RtpHeaderParser;
 class RTPPayloadRegistry;
 class RtpReceiver;
 class Transport;
 class UlpfecReceiver;
+class VCMTiming;
 class VieRemb;
 
 namespace vcm {
 class VideoReceiver;
 }  // namespace vcm
 
-class RtpStreamReceiver : public RtpData, public RtpFeedback,
+class RtpStreamReceiver : public RtpData,
+                          public RtpFeedback,
                           public VCMFrameTypeCallback,
-                          public VCMPacketRequestCallback {
+                          public VCMPacketRequestCallback,
+                          public video_coding::OnReceivedFrameCallback,
+                          public video_coding::OnCompleteFrameCallback,
+                          public CallStatsObserver {
  public:
-  RtpStreamReceiver(vcm::VideoReceiver* video_receiver,
-                    RemoteBitrateEstimator* remote_bitrate_estimator,
-                    Transport* transport,
-                    RtcpRttStats* rtt_stats,
-                    PacedSender* paced_sender,
-                    PacketRouter* packet_router,
-                    VieRemb* remb,
-                    const VideoReceiveStream::Config* config,
-                    ReceiveStatisticsProxy* receive_stats_proxy,
-                    ProcessThread* process_thread,
-                    RateLimiter* retransmission_rate_limiter);
+  RtpStreamReceiver(
+      vcm::VideoReceiver* video_receiver,
+      RemoteBitrateEstimator* remote_bitrate_estimator,
+      Transport* transport,
+      RtcpRttStats* rtt_stats,
+      PacedSender* paced_sender,
+      PacketRouter* packet_router,
+      VieRemb* remb,
+      const VideoReceiveStream::Config* config,
+      ReceiveStatisticsProxy* receive_stats_proxy,
+      ProcessThread* process_thread,
+      RateLimiter* retransmission_rate_limiter,
+      NackSender* nack_sender,
+      KeyFrameRequestSender* keyframe_request_sender,
+      video_coding::OnCompleteFrameCallback* complete_frame_callback,
+      VCMTiming* timing);
   ~RtpStreamReceiver();
 
   bool SetReceiveCodec(const VideoCodec& video_codec);
 
   uint32_t GetRemoteSsrc() const;
   int GetCsrcs(uint32_t* csrcs) const;
 
   RtpReceiver* GetRtpReceiver() const;
   RtpRtcp* rtp_rtcp() const { return rtp_rtcp_.get(); }
 
   void StartReceive();
   void StopReceive();
 
   bool DeliverRtp(const uint8_t* rtp_packet,
                   size_t rtp_packet_length,
                   const PacketTime& packet_time);
   bool DeliverRtcp(const uint8_t* rtcp_packet, size_t rtcp_packet_length);
 
+  void FrameContinuous(uint16_t seq_num);
+
+  void FrameDecoded(uint16_t seq_num);
+
   void SignalNetworkState(NetworkState state);
 
   // Implements RtpData.
   int32_t OnReceivedPayloadData(const uint8_t* payload_data,
                                 size_t payload_size,
                                 const WebRtcRTPHeader* rtp_header) override;
   bool OnRecoveredPacket(const uint8_t* packet, size_t packet_length) override;
 
   // Implements RtpFeedback.
   int32_t OnInitializeDecoder(int8_t payload_type,
@@ skipping 10 matching lines @@
 
   bool IsFecEnabled() const;
   bool IsRetransmissionsEnabled() const;
   // Don't use, still experimental.
   void RequestPacketRetransmit(const std::vector<uint16_t>& sequence_numbers);
 
   // Implements VCMPacketRequestCallback.
   int32_t ResendPackets(const uint16_t* sequenceNumbers,
                         uint16_t length) override;
 
+  // Implements OnReceivedFrameCallback.
+  void OnReceivedFrame(
+      std::unique_ptr<video_coding::RtpFrameObject> frame) override;
+
+  // Implements OnCompleteFrameCallback.
+  void OnCompleteFrame(
+      std::unique_ptr<video_coding::FrameObject> frame) override;
+
+  void OnRttUpdate(int64_t avg_rtt_ms, int64_t max_rtt_ms) override;
+
  private:
   bool ReceivePacket(const uint8_t* packet,
                      size_t packet_length,
                      const RTPHeader& header,
                      bool in_order);
   // Parses and handles for instance RTX and RED headers.
   // This function assumes that it's being called from only one thread.
   bool ParseAndHandleEncapsulatingHeader(const uint8_t* packet,
                                          size_t packet_length,
                                          const RTPHeader& header);
@@ skipping 20 matching lines @@
   const std::unique_ptr<ReceiveStatistics> rtp_receive_statistics_;
   std::unique_ptr<UlpfecReceiver> ulpfec_receiver_;
 
   rtc::CriticalSection receive_cs_;
   bool receiving_ GUARDED_BY(receive_cs_);
   uint8_t restored_packet_[IP_PACKET_SIZE] GUARDED_BY(receive_cs_);
   bool restored_packet_in_use_ GUARDED_BY(receive_cs_);
   int64_t last_packet_log_ms_ GUARDED_BY(receive_cs_);
 
   const std::unique_ptr<RtpRtcp> rtp_rtcp_;
+
+  // Members for the new jitter buffer experiment.
+  bool 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_);
+  video_coding::H264SpsPpsTracker tracker_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_VIDEO_RTP_STREAM_RECEIVER_H_