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Side by Side Diff: webrtc/video/video_receive_stream.h

Issue 2649973005: Inform jitter buffer about FlexFEC protection. (Closed)
Patch Set: Created 3 years, 11 months ago
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1 /* 1 /*
2 * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
3 * 3 *
4 * Use of this source code is governed by a BSD-style license 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 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 6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may 7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree. 8 * be found in the AUTHORS file in the root of the source tree.
9 */ 9 */
10 10
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24 #include "webrtc/video/rtp_stream_receiver.h" 24 #include "webrtc/video/rtp_stream_receiver.h"
25 #include "webrtc/video/rtp_streams_synchronizer.h" 25 #include "webrtc/video/rtp_streams_synchronizer.h"
26 #include "webrtc/video/transport_adapter.h" 26 #include "webrtc/video/transport_adapter.h"
27 #include "webrtc/video/video_stream_decoder.h" 27 #include "webrtc/video/video_stream_decoder.h"
28 #include "webrtc/video_receive_stream.h" 28 #include "webrtc/video_receive_stream.h"
29 29
30 namespace webrtc { 30 namespace webrtc {
31 31
32 class CallStats; 32 class CallStats;
33 class CongestionController; 33 class CongestionController;
34 class FlexfecProtectionMediator;
34 class IvfFileWriter; 35 class IvfFileWriter;
35 class ProcessThread; 36 class ProcessThread;
36 class RTPFragmentationHeader; 37 class RTPFragmentationHeader;
37 class VoiceEngine; 38 class VoiceEngine;
38 class VieRemb; 39 class VieRemb;
39 class VCMTiming; 40 class VCMTiming;
40 class VCMJitterEstimator; 41 class VCMJitterEstimator;
41 42
42 namespace internal { 43 namespace internal {
43 44
44 class VideoReceiveStream : public webrtc::VideoReceiveStream, 45 class VideoReceiveStream : public webrtc::VideoReceiveStream,
45 public rtc::VideoSinkInterface<VideoFrame>, 46 public rtc::VideoSinkInterface<VideoFrame>,
46 public EncodedImageCallback, 47 public EncodedImageCallback,
47 public NackSender, 48 public NackSender,
48 public KeyFrameRequestSender, 49 public KeyFrameRequestSender,
49 public video_coding::OnCompleteFrameCallback { 50 public video_coding::OnCompleteFrameCallback {
50 public: 51 public:
51 VideoReceiveStream(int num_cpu_cores, 52 VideoReceiveStream(int num_cpu_cores,
52 CongestionController* congestion_controller, 53 CongestionController* congestion_controller,
53 PacketRouter* packet_router, 54 PacketRouter* packet_router,
54 VideoReceiveStream::Config config, 55 VideoReceiveStream::Config config,
55 webrtc::VoiceEngine* voice_engine, 56 webrtc::VoiceEngine* voice_engine,
56 ProcessThread* process_thread, 57 ProcessThread* process_thread,
57 CallStats* call_stats, 58 CallStats* call_stats,
58 VieRemb* remb); 59 VieRemb* remb,
60 FlexfecProtectionMediator* flexfec_protection_mediator);
59 ~VideoReceiveStream() override; 61 ~VideoReceiveStream() override;
60 62
61 void SignalNetworkState(NetworkState state); 63 void SignalNetworkState(NetworkState state);
62 bool DeliverRtcp(const uint8_t* packet, size_t length); 64 bool DeliverRtcp(const uint8_t* packet, size_t length);
63 bool DeliverRtp(const uint8_t* packet, 65 bool DeliverRtp(const uint8_t* packet,
64 size_t length, 66 size_t length,
65 const PacketTime& packet_time); 67 const PacketTime& packet_time);
66 68
67 bool OnRecoveredPacket(const uint8_t* packet, size_t length); 69 bool OnRecoveredPacket(const uint8_t* packet, size_t length);
68 70
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99 // Calling this method will close and finalize any current log. 101 // Calling this method will close and finalize any current log.
100 // Giving rtc::kInvalidPlatformFileValue disables logging. 102 // Giving rtc::kInvalidPlatformFileValue disables logging.
101 // If a frame to be written would make the log too large the write fails and 103 // 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. 104 // the log is closed and finalized. A |byte_limit| of 0 means no limit.
103 void EnableEncodedFrameRecording(rtc::PlatformFile file, 105 void EnableEncodedFrameRecording(rtc::PlatformFile file,
104 size_t byte_limit) override; 106 size_t byte_limit) override;
105 107
106 private: 108 private:
107 static bool DecodeThreadFunction(void* ptr); 109 static bool DecodeThreadFunction(void* ptr);
108 void Decode(); 110 void Decode();
109 111
brandtr 2017/01/24 13:02:49 The methods below are not strictly needed, but I t
112 // The return value of this method depends on what FlexfecReceiveStreams are
113 // registered in Call, which might change during the lifetime of |this|.
114 bool IsFlexfecEnabled() const;
115
116 // The return value of this method is constant during the lifetime of |this|.
117 bool IsUlpfecEnabled() const;
118
110 TransportAdapter transport_adapter_; 119 TransportAdapter transport_adapter_;
111 const VideoReceiveStream::Config config_; 120 const VideoReceiveStream::Config config_;
112 const int num_cpu_cores_; 121 const int num_cpu_cores_;
113 ProcessThread* const process_thread_; 122 ProcessThread* const process_thread_;
114 Clock* const clock_; 123 Clock* const clock_;
115 124
116 rtc::PlatformThread decode_thread_; 125 rtc::PlatformThread decode_thread_;
117 126
118 CongestionController* const congestion_controller_; 127 CongestionController* const congestion_controller_;
119 CallStats* const call_stats_; 128 CallStats* const call_stats_;
129 FlexfecProtectionMediator* const flexfec_protection_mediator_;
120 130
121 std::unique_ptr<VCMTiming> timing_; // Jitter buffer experiment. 131 std::unique_ptr<VCMTiming> timing_; // Jitter buffer experiment.
122 vcm::VideoReceiver video_receiver_; 132 vcm::VideoReceiver video_receiver_;
123 std::unique_ptr<rtc::VideoSinkInterface<VideoFrame>> incoming_video_stream_; 133 std::unique_ptr<rtc::VideoSinkInterface<VideoFrame>> incoming_video_stream_;
124 ReceiveStatisticsProxy stats_proxy_; 134 ReceiveStatisticsProxy stats_proxy_;
125 RtpStreamReceiver rtp_stream_receiver_; 135 RtpStreamReceiver rtp_stream_receiver_;
126 std::unique_ptr<VideoStreamDecoder> video_stream_decoder_; 136 std::unique_ptr<VideoStreamDecoder> video_stream_decoder_;
127 RtpStreamsSynchronizer rtp_stream_sync_; 137 RtpStreamsSynchronizer rtp_stream_sync_;
128 138
129 rtc::CriticalSection ivf_writer_lock_; 139 rtc::CriticalSection ivf_writer_lock_;
130 std::unique_ptr<IvfFileWriter> ivf_writer_ GUARDED_BY(ivf_writer_lock_); 140 std::unique_ptr<IvfFileWriter> ivf_writer_ GUARDED_BY(ivf_writer_lock_);
131 141
132 // Members for the new jitter buffer experiment. 142 // Members for the new jitter buffer experiment.
133 const bool jitter_buffer_experiment_; 143 const bool jitter_buffer_experiment_;
134 std::unique_ptr<VCMJitterEstimator> jitter_estimator_; 144 std::unique_ptr<VCMJitterEstimator> jitter_estimator_;
135 std::unique_ptr<video_coding::FrameBuffer> frame_buffer_; 145 std::unique_ptr<video_coding::FrameBuffer> frame_buffer_;
136 }; 146 };
137 } // namespace internal 147 } // namespace internal
138 } // namespace webrtc 148 } // namespace webrtc
139 149
140 #endif // WEBRTC_VIDEO_VIDEO_RECEIVE_STREAM_H_ 150 #endif // WEBRTC_VIDEO_VIDEO_RECEIVE_STREAM_H_
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