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

Issue 2646343005: Sort method declarations/definitions in VideoReceiveStream. (Closed)
Patch Set: Rebase. 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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51 VideoReceiveStream(int num_cpu_cores, 51 VideoReceiveStream(int num_cpu_cores,
52 CongestionController* congestion_controller, 52 CongestionController* congestion_controller,
53 PacketRouter* packet_router, 53 PacketRouter* packet_router,
54 VideoReceiveStream::Config config, 54 VideoReceiveStream::Config config,
55 webrtc::VoiceEngine* voice_engine, 55 webrtc::VoiceEngine* voice_engine,
56 ProcessThread* process_thread, 56 ProcessThread* process_thread,
57 CallStats* call_stats, 57 CallStats* call_stats,
58 VieRemb* remb); 58 VieRemb* remb);
59 ~VideoReceiveStream() override; 59 ~VideoReceiveStream() override;
60 60
61 const Config& config() const { return config_; }
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
69 // webrtc::VideoReceiveStream implementation. 71 void SetSyncChannel(VoiceEngine* voice_engine, int audio_channel_id);
72
73 // Implements webrtc::VideoReceiveStream.
70 void Start() override; 74 void Start() override;
71 void Stop() override; 75 void Stop() override;
72 76
73 webrtc::VideoReceiveStream::Stats GetStats() const override; 77 webrtc::VideoReceiveStream::Stats GetStats() const override;
74 78
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 const Config& config() const { return config_; }
89
90 void SetSyncChannel(VoiceEngine* voice_engine, int audio_channel_id);
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. 79 // Takes ownership of the file, is responsible for closing it later.
99 // Calling this method will close and finalize any current log. 80 // Calling this method will close and finalize any current log.
100 // Giving rtc::kInvalidPlatformFileValue disables logging. 81 // Giving rtc::kInvalidPlatformFileValue disables logging.
101 // If a frame to be written would make the log too large the write fails and 82 // 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. 83 // the log is closed and finalized. A |byte_limit| of 0 means no limit.
103 void EnableEncodedFrameRecording(rtc::PlatformFile file, 84 void EnableEncodedFrameRecording(rtc::PlatformFile file,
104 size_t byte_limit) override; 85 size_t byte_limit) override;
105 86
87 // Implements rtc::VideoSinkInterface<VideoFrame>.
88 void OnFrame(const VideoFrame& video_frame) override;
89
90 // Implements EncodedImageCallback.
91 EncodedImageCallback::Result OnEncodedImage(
92 const EncodedImage& encoded_image,
93 const CodecSpecificInfo* codec_specific_info,
94 const RTPFragmentationHeader* fragmentation) override;
95
96 // Implements NackSender.
97 void SendNack(const std::vector<uint16_t>& sequence_numbers) override;
98
99 // Implements KeyFrameRequestSender.
100 void RequestKeyFrame() override;
101
102 // Implements video_coding::OnCompleteFrameCallback.
103 void OnCompleteFrame(
104 std::unique_ptr<video_coding::FrameObject> frame) override;
105
106 private: 106 private:
107 static bool DecodeThreadFunction(void* ptr); 107 static bool DecodeThreadFunction(void* ptr);
108 void Decode(); 108 void Decode();
109 109
110 TransportAdapter transport_adapter_; 110 TransportAdapter transport_adapter_;
111 const VideoReceiveStream::Config config_; 111 const VideoReceiveStream::Config config_;
112 const int num_cpu_cores_; 112 const int num_cpu_cores_;
113 ProcessThread* const process_thread_; 113 ProcessThread* const process_thread_;
114 Clock* const clock_; 114 Clock* const clock_;
115 115
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131 131
132 // Members for the new jitter buffer experiment. 132 // Members for the new jitter buffer experiment.
133 const bool jitter_buffer_experiment_; 133 const bool jitter_buffer_experiment_;
134 std::unique_ptr<VCMJitterEstimator> jitter_estimator_; 134 std::unique_ptr<VCMJitterEstimator> jitter_estimator_;
135 std::unique_ptr<video_coding::FrameBuffer> frame_buffer_; 135 std::unique_ptr<video_coding::FrameBuffer> frame_buffer_;
136 }; 136 };
137 } // namespace internal 137 } // namespace internal
138 } // namespace webrtc 138 } // namespace webrtc
139 139
140 #endif // WEBRTC_VIDEO_VIDEO_RECEIVE_STREAM_H_ 140 #endif // WEBRTC_VIDEO_VIDEO_RECEIVE_STREAM_H_
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