Line data    Source code

       1             : /*
       2             :  *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
       3             :  *
       4             :  *  Use of this source code is governed by a BSD-style license
       5             :  *  that can be found in the LICENSE file in the root of the source
       6             :  *  tree. An additional intellectual property rights grant can be found
       7             :  *  in the file PATENTS.  All contributing project authors may
       8             :  *  be found in the AUTHORS file in the root of the source tree.
       9             :  */
      10             : 
      11             : #ifndef WEBRTC_VIDEO_VIDEO_RECEIVE_STREAM_H_
      12             : #define WEBRTC_VIDEO_VIDEO_RECEIVE_STREAM_H_
      13             : 
      14             : #include <memory>
      15             : #include <vector>
      16             : 
      17             : #include "webrtc/common_video/include/incoming_video_stream.h"
      18             : #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
      19             : #include "webrtc/modules/rtp_rtcp/include/flexfec_receiver.h"
      20             : #include "webrtc/modules/video_coding/frame_buffer2.h"
      21             : #include "webrtc/modules/video_coding/video_coding_impl.h"
      22             : #include "webrtc/system_wrappers/include/clock.h"
      23             : #include "webrtc/video/receive_statistics_proxy.h"
      24             : #include "webrtc/video/rtp_stream_receiver.h"
      25             : #include "webrtc/video/rtp_streams_synchronizer.h"
      26             : #include "webrtc/video/transport_adapter.h"
      27             : #include "webrtc/video/video_stream_decoder.h"
      28             : #include "webrtc/video_receive_stream.h"
      29             : 
      30             : namespace webrtc {
      31             : 
      32             : class CallStats;
      33             : class CongestionController;
      34             : class IvfFileWriter;
      35             : class ProcessThread;
      36             : class RTPFragmentationHeader;
      37             : class VoiceEngine;
      38             : class VieRemb;
      39             : class VCMTiming;
      40             : class VCMJitterEstimator;
      41             : 
      42             : namespace internal {
      43             : 
      44             : class VideoReceiveStream : public webrtc::VideoReceiveStream,
      45             :                            public rtc::VideoSinkInterface<VideoFrame>,
      46             :                            public EncodedImageCallback,
      47             :                            public NackSender,
      48             :                            public KeyFrameRequestSender,
      49             :                            public video_coding::OnCompleteFrameCallback {
      50             :  public:
      51             :   VideoReceiveStream(int num_cpu_cores,
      52             :                      CongestionController* congestion_controller,
      53             :                      PacketRouter* packet_router,
      54             :                      VideoReceiveStream::Config config,
      55             :                      webrtc::VoiceEngine* voice_engine,
      56             :                      ProcessThread* process_thread,
      57             :                      CallStats* call_stats,
      58             :                      VieRemb* remb);
      59             :   ~VideoReceiveStream() override;
      60             : 
      61             :   void SignalNetworkState(NetworkState state);
      62             :   bool DeliverRtcp(const uint8_t* packet, size_t length);
      63             :   bool DeliverRtp(const uint8_t* packet,
      64             :                   size_t length,
      65             :                   const PacketTime& packet_time);
      66             : 
      67             :   bool OnRecoveredPacket(const uint8_t* packet, size_t length);
      68             : 
      69             :   // webrtc::VideoReceiveStream implementation.
      70             :   void Start() override;
      71             :   void Stop() override;
      72             : 
      73             :   webrtc::VideoReceiveStream::Stats GetStats() const override;
      74             : 
      75             :   // Overrides rtc::VideoSinkInterface<VideoFrame>.
      76             :   void OnFrame(const VideoFrame& video_frame) override;
      77             : 
      78             :   // Implements video_coding::OnCompleteFrameCallback.
      79             :   void OnCompleteFrame(
      80             :       std::unique_ptr<video_coding::FrameObject> frame) override;
      81             : 
      82             :   // Overrides EncodedImageCallback.
      83             :   EncodedImageCallback::Result OnEncodedImage(
      84             :       const EncodedImage& encoded_image,
      85             :       const CodecSpecificInfo* codec_specific_info,
      86             :       const RTPFragmentationHeader* fragmentation) override;
      87             : 
      88           0 :   const Config& config() const { return config_; }
      89             : 
      90             :   void SetSyncChannel(VoiceEngine* voice_engine, int audio_channel_id) override;
      91             : 
      92             :   // Implements NackSender.
      93             :   void SendNack(const std::vector<uint16_t>& sequence_numbers) override;
      94             : 
      95             :   // Implements KeyFrameRequestSender.
      96             :   void RequestKeyFrame() override;
      97             : 
      98             :   // Takes ownership of the file, is responsible for closing it later.
      99             :   // Calling this method will close and finalize any current log.
     100             :   // Giving rtc::kInvalidPlatformFileValue disables logging.
     101             :   // If a frame to be written would make the log too large the write fails and
     102             :   // the log is closed and finalized. A |byte_limit| of 0 means no limit.
     103             :   void EnableEncodedFrameRecording(rtc::PlatformFile file,
     104             :                                    size_t byte_limit) override;
     105             : 
     106             : 
     107             :   bool GetRemoteRTCPSenderInfo(RTCPSenderInfo* sender_info) const override;
     108             :  private:
     109             :   static bool DecodeThreadFunction(void* ptr);
     110             :   void Decode();
     111             : 
     112             :   TransportAdapter transport_adapter_;
     113             :   const VideoReceiveStream::Config config_;
     114             :   const int num_cpu_cores_;
     115             :   ProcessThread* const process_thread_;
     116             :   Clock* const clock_;
     117             : 
     118             :   rtc::PlatformThread decode_thread_;
     119             : 
     120             :   CongestionController* const congestion_controller_;
     121             :   CallStats* const call_stats_;
     122             : 
     123             :   std::unique_ptr<VCMTiming> timing_;  // Jitter buffer experiment.
     124             :   vcm::VideoReceiver video_receiver_;
     125             :   std::unique_ptr<rtc::VideoSinkInterface<VideoFrame>> incoming_video_stream_;
     126             :   ReceiveStatisticsProxy stats_proxy_;
     127             :   RtpStreamReceiver rtp_stream_receiver_;
     128             :   std::unique_ptr<VideoStreamDecoder> video_stream_decoder_;
     129             :   RtpStreamsSynchronizer rtp_stream_sync_;
     130             : 
     131             :   rtc::CriticalSection ivf_writer_lock_;
     132             :   std::unique_ptr<IvfFileWriter> ivf_writer_ GUARDED_BY(ivf_writer_lock_);
     133             : 
     134             :   // Members for the new jitter buffer experiment.
     135             :   const bool jitter_buffer_experiment_;
     136             :   std::unique_ptr<VCMJitterEstimator> jitter_estimator_;
     137             :   std::unique_ptr<video_coding::FrameBuffer> frame_buffer_;
     138             : };
     139             : }  // namespace internal
     140             : }  // namespace webrtc
     141             : 
     142             : #endif  // WEBRTC_VIDEO_VIDEO_RECEIVE_STREAM_H_