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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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200 // Starts stream activity. | 200 // Starts stream activity. |
201 // When a stream is active, it can receive, process and deliver packets. | 201 // When a stream is active, it can receive, process and deliver packets. |
202 virtual void Start() = 0; | 202 virtual void Start() = 0; |
203 // Stops stream activity. | 203 // Stops stream activity. |
204 // When a stream is stopped, it can't receive, process or deliver packets. | 204 // When a stream is stopped, it can't receive, process or deliver packets. |
205 virtual void Stop() = 0; | 205 virtual void Stop() = 0; |
206 | 206 |
207 // TODO(pbos): Add info on currently-received codec to Stats. | 207 // TODO(pbos): Add info on currently-received codec to Stats. |
208 virtual Stats GetStats() const = 0; | 208 virtual Stats GetStats() const = 0; |
209 | 209 |
| 210 virtual rtc::Optional<TimingFrameInfo> GetAndResetTimingFrameInfo() = 0; |
| 211 |
210 // Takes ownership of the file, is responsible for closing it later. | 212 // Takes ownership of the file, is responsible for closing it later. |
211 // Calling this method will close and finalize any current log. | 213 // Calling this method will close and finalize any current log. |
212 // Giving rtc::kInvalidPlatformFileValue disables logging. | 214 // Giving rtc::kInvalidPlatformFileValue disables logging. |
213 // If a frame to be written would make the log too large the write fails and | 215 // If a frame to be written would make the log too large the write fails and |
214 // the log is closed and finalized. A |byte_limit| of 0 means no limit. | 216 // the log is closed and finalized. A |byte_limit| of 0 means no limit. |
215 virtual void EnableEncodedFrameRecording(rtc::PlatformFile file, | 217 virtual void EnableEncodedFrameRecording(rtc::PlatformFile file, |
216 size_t byte_limit) = 0; | 218 size_t byte_limit) = 0; |
217 inline void DisableEncodedFrameRecording() { | 219 inline void DisableEncodedFrameRecording() { |
218 EnableEncodedFrameRecording(rtc::kInvalidPlatformFileValue, 0); | 220 EnableEncodedFrameRecording(rtc::kInvalidPlatformFileValue, 0); |
219 } | 221 } |
220 | 222 |
221 protected: | 223 protected: |
222 virtual ~VideoReceiveStream() {} | 224 virtual ~VideoReceiveStream() {} |
223 }; | 225 }; |
224 | 226 |
225 } // namespace webrtc | 227 } // namespace webrtc |
226 | 228 |
227 #endif // WEBRTC_VIDEO_RECEIVE_STREAM_H_ | 229 #endif // WEBRTC_VIDEO_RECEIVE_STREAM_H_ |
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