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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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| 55 VideoReceiveStream::Config config, | 55 VideoReceiveStream::Config config, |
| 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_; } | 61 const Config& config() const { return config_; } |
| 62 | 62 |
| 63 void SignalNetworkState(NetworkState state); | 63 void SignalNetworkState(NetworkState state); |
| 64 bool DeliverRtcp(const uint8_t* packet, size_t length); | 64 bool DeliverRtcp(const uint8_t* packet, size_t length); |
| 65 bool DeliverRtp(const uint8_t* packet, | |
| 66 size_t length, | |
| 67 const PacketTime& packet_time); | |
| 68 | 65 |
| 69 bool OnRecoveredPacket(const uint8_t* packet, size_t length); | 66 bool OnRecoveredPacket(const uint8_t* packet, size_t length); |
| 70 | 67 |
| 71 void SetSync(Syncable* audio_syncable); | 68 void SetSync(Syncable* audio_syncable); |
| 72 | 69 |
| 73 // Implements webrtc::VideoReceiveStream. | 70 // Implements webrtc::VideoReceiveStream. |
| 74 void Start() override; | 71 void Start() override; |
| 75 void Stop() override; | 72 void Stop() override; |
| 76 | 73 |
| 77 webrtc::VideoReceiveStream::Stats GetStats() const override; | 74 webrtc::VideoReceiveStream::Stats GetStats() const override; |
| 78 | 75 |
| 79 // Takes ownership of the file, is responsible for closing it later. | 76 // Takes ownership of the file, is responsible for closing it later. |
| 80 // Calling this method will close and finalize any current log. | 77 // Calling this method will close and finalize any current log. |
| 81 // Giving rtc::kInvalidPlatformFileValue disables logging. | 78 // Giving rtc::kInvalidPlatformFileValue disables logging. |
| 82 // If a frame to be written would make the log too large the write fails and | 79 // If a frame to be written would make the log too large the write fails and |
| 83 // the log is closed and finalized. A |byte_limit| of 0 means no limit. | 80 // the log is closed and finalized. A |byte_limit| of 0 means no limit. |
| 84 void EnableEncodedFrameRecording(rtc::PlatformFile file, | 81 void EnableEncodedFrameRecording(rtc::PlatformFile file, |
| 85 size_t byte_limit) override; | 82 size_t byte_limit) override; |
| 86 | 83 |
| 84 // TODO(nisse): Intended to be part of an RtpPacketReceiver interface. |
| 85 void OnRtpPacket(const RtpPacketReceived& packet); |
| 86 |
| 87 // Implements rtc::VideoSinkInterface<VideoFrame>. | 87 // Implements rtc::VideoSinkInterface<VideoFrame>. |
| 88 void OnFrame(const VideoFrame& video_frame) override; | 88 void OnFrame(const VideoFrame& video_frame) override; |
| 89 | 89 |
| 90 // Implements EncodedImageCallback. | 90 // Implements EncodedImageCallback. |
| 91 EncodedImageCallback::Result OnEncodedImage( | 91 EncodedImageCallback::Result OnEncodedImage( |
| 92 const EncodedImage& encoded_image, | 92 const EncodedImage& encoded_image, |
| 93 const CodecSpecificInfo* codec_specific_info, | 93 const CodecSpecificInfo* codec_specific_info, |
| 94 const RTPFragmentationHeader* fragmentation) override; | 94 const RTPFragmentationHeader* fragmentation) override; |
| 95 | 95 |
| 96 // Implements NackSender. | 96 // Implements NackSender. |
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| 140 | 140 |
| 141 // Members for the new jitter buffer experiment. | 141 // Members for the new jitter buffer experiment. |
| 142 const bool jitter_buffer_experiment_; | 142 const bool jitter_buffer_experiment_; |
| 143 std::unique_ptr<VCMJitterEstimator> jitter_estimator_; | 143 std::unique_ptr<VCMJitterEstimator> jitter_estimator_; |
| 144 std::unique_ptr<video_coding::FrameBuffer> frame_buffer_; | 144 std::unique_ptr<video_coding::FrameBuffer> frame_buffer_; |
| 145 }; | 145 }; |
| 146 } // namespace internal | 146 } // namespace internal |
| 147 } // namespace webrtc | 147 } // namespace webrtc |
| 148 | 148 |
| 149 #endif // WEBRTC_VIDEO_VIDEO_RECEIVE_STREAM_H_ | 149 #endif // WEBRTC_VIDEO_VIDEO_RECEIVE_STREAM_H_ |
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