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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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| 59 CallStats* call_stats); | 59 CallStats* call_stats); |
| 60 ~VideoReceiveStream() override; | 60 ~VideoReceiveStream() override; |
| 61 | 61 |
| 62 const Config& config() const { return config_; } | 62 const Config& config() const { return config_; } |
| 63 | 63 |
| 64 void SignalNetworkState(NetworkState state); | 64 void SignalNetworkState(NetworkState state); |
| 65 bool DeliverRtcp(const uint8_t* packet, size_t length); | 65 bool DeliverRtcp(const uint8_t* packet, size_t length); |
| 66 | 66 |
| 67 void SetSync(Syncable* audio_syncable); | 67 void SetSync(Syncable* audio_syncable); |
| 68 | 68 |
| 69 // Implements SecondaryRtpSinksContainer (from webrtc::VideoReceiveStream). |
| 70 void AddSecondarySink(RtpPacketSinkInterface* sink) override; |
| 71 void RemoveSecondarySink(const RtpPacketSinkInterface* sink) override; |
| 72 |
| 69 // Implements webrtc::VideoReceiveStream. | 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 rtc::Optional<TimingFrameInfo> GetAndResetTimingFrameInfo() override; | 79 rtc::Optional<TimingFrameInfo> GetAndResetTimingFrameInfo() override; |
| 76 | 80 |
| 77 // Takes ownership of the file, is responsible for closing it later. | 81 // Takes ownership of the file, is responsible for closing it later. |
| 78 // Calling this method will close and finalize any current log. | 82 // Calling this method will close and finalize any current log. |
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| 139 std::unique_ptr<VCMJitterEstimator> jitter_estimator_; | 143 std::unique_ptr<VCMJitterEstimator> jitter_estimator_; |
| 140 std::unique_ptr<video_coding::FrameBuffer> frame_buffer_; | 144 std::unique_ptr<video_coding::FrameBuffer> frame_buffer_; |
| 141 | 145 |
| 142 std::unique_ptr<RtpStreamReceiverInterface> media_receiver_; | 146 std::unique_ptr<RtpStreamReceiverInterface> media_receiver_; |
| 143 std::unique_ptr<RtpStreamReceiverInterface> rtx_receiver_; | 147 std::unique_ptr<RtpStreamReceiverInterface> rtx_receiver_; |
| 144 }; | 148 }; |
| 145 } // namespace internal | 149 } // namespace internal |
| 146 } // namespace webrtc | 150 } // namespace webrtc |
| 147 | 151 |
| 148 #endif // WEBRTC_VIDEO_VIDEO_RECEIVE_STREAM_H_ | 152 #endif // WEBRTC_VIDEO_VIDEO_RECEIVE_STREAM_H_ |
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