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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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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 webrtc::VideoReceiveStream. | 69 // Implements webrtc::VideoReceiveStream. |
70 void Start() override; | 70 void Start() override; |
71 void Stop() override; | 71 void Stop() override; |
72 | 72 |
73 webrtc::VideoReceiveStream::Stats GetStats() const override; | 73 webrtc::VideoReceiveStream::Stats GetStats() const override; |
74 | 74 |
| 75 rtc::Optional<TimingFrameInfo> GetAndResetTimingFrameInfo() override; |
| 76 |
75 // Takes ownership of the file, is responsible for closing it later. | 77 // Takes ownership of the file, is responsible for closing it later. |
76 // Calling this method will close and finalize any current log. | 78 // Calling this method will close and finalize any current log. |
77 // Giving rtc::kInvalidPlatformFileValue disables logging. | 79 // Giving rtc::kInvalidPlatformFileValue disables logging. |
78 // If a frame to be written would make the log too large the write fails and | 80 // If a frame to be written would make the log too large the write fails and |
79 // the log is closed and finalized. A |byte_limit| of 0 means no limit. | 81 // the log is closed and finalized. A |byte_limit| of 0 means no limit. |
80 void EnableEncodedFrameRecording(rtc::PlatformFile file, | 82 void EnableEncodedFrameRecording(rtc::PlatformFile file, |
81 size_t byte_limit) override; | 83 size_t byte_limit) override; |
82 | 84 |
83 // Implements rtc::VideoSinkInterface<VideoFrame>. | 85 // Implements rtc::VideoSinkInterface<VideoFrame>. |
84 void OnFrame(const VideoFrame& video_frame) override; | 86 void OnFrame(const VideoFrame& video_frame) override; |
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137 std::unique_ptr<VCMJitterEstimator> jitter_estimator_; | 139 std::unique_ptr<VCMJitterEstimator> jitter_estimator_; |
138 std::unique_ptr<video_coding::FrameBuffer> frame_buffer_; | 140 std::unique_ptr<video_coding::FrameBuffer> frame_buffer_; |
139 | 141 |
140 std::unique_ptr<RtpStreamReceiverInterface> media_receiver_; | 142 std::unique_ptr<RtpStreamReceiverInterface> media_receiver_; |
141 std::unique_ptr<RtpStreamReceiverInterface> rtx_receiver_; | 143 std::unique_ptr<RtpStreamReceiverInterface> rtx_receiver_; |
142 }; | 144 }; |
143 } // namespace internal | 145 } // namespace internal |
144 } // namespace webrtc | 146 } // namespace webrtc |
145 | 147 |
146 #endif // WEBRTC_VIDEO_VIDEO_RECEIVE_STREAM_H_ | 148 #endif // WEBRTC_VIDEO_VIDEO_RECEIVE_STREAM_H_ |
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