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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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75 rtc::Optional<TimingFrameInfo> GetAndResetTimingFrameInfo() override; | 75 rtc::Optional<TimingFrameInfo> GetAndResetTimingFrameInfo() override; |
76 | 76 |
77 // Takes ownership of the file, is responsible for closing it later. | 77 // Takes ownership of the file, is responsible for closing it later. |
78 // Calling this method will close and finalize any current log. | 78 // Calling this method will close and finalize any current log. |
79 // Giving rtc::kInvalidPlatformFileValue disables logging. | 79 // Giving rtc::kInvalidPlatformFileValue disables logging. |
80 // 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 |
81 // 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. |
82 void EnableEncodedFrameRecording(rtc::PlatformFile file, | 82 void EnableEncodedFrameRecording(rtc::PlatformFile file, |
83 size_t byte_limit) override; | 83 size_t byte_limit) override; |
84 | 84 |
| 85 void AddSecondarySink(RtpPacketSinkInterface* sink) override; |
| 86 void RemoveSecondarySink(const RtpPacketSinkInterface* sink) override; |
| 87 |
85 // Implements rtc::VideoSinkInterface<VideoFrame>. | 88 // Implements rtc::VideoSinkInterface<VideoFrame>. |
86 void OnFrame(const VideoFrame& video_frame) override; | 89 void OnFrame(const VideoFrame& video_frame) override; |
87 | 90 |
88 // Implements EncodedImageCallback. | 91 // Implements EncodedImageCallback. |
89 EncodedImageCallback::Result OnEncodedImage( | 92 EncodedImageCallback::Result OnEncodedImage( |
90 const EncodedImage& encoded_image, | 93 const EncodedImage& encoded_image, |
91 const CodecSpecificInfo* codec_specific_info, | 94 const CodecSpecificInfo* codec_specific_info, |
92 const RTPFragmentationHeader* fragmentation) override; | 95 const RTPFragmentationHeader* fragmentation) override; |
93 | 96 |
94 // Implements NackSender. | 97 // Implements NackSender. |
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139 std::unique_ptr<VCMJitterEstimator> jitter_estimator_; | 142 std::unique_ptr<VCMJitterEstimator> jitter_estimator_; |
140 std::unique_ptr<video_coding::FrameBuffer> frame_buffer_; | 143 std::unique_ptr<video_coding::FrameBuffer> frame_buffer_; |
141 | 144 |
142 std::unique_ptr<RtpStreamReceiverInterface> media_receiver_; | 145 std::unique_ptr<RtpStreamReceiverInterface> media_receiver_; |
143 std::unique_ptr<RtpStreamReceiverInterface> rtx_receiver_; | 146 std::unique_ptr<RtpStreamReceiverInterface> rtx_receiver_; |
144 }; | 147 }; |
145 } // namespace internal | 148 } // namespace internal |
146 } // namespace webrtc | 149 } // namespace webrtc |
147 | 150 |
148 #endif // WEBRTC_VIDEO_VIDEO_RECEIVE_STREAM_H_ | 151 #endif // WEBRTC_VIDEO_VIDEO_RECEIVE_STREAM_H_ |
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