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

Issue 2974453002: Protected streams report RTP messages directly to the FlexFec streams (Closed)
Patch Set: Appease lint. Created 3 years, 4 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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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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