| OLD | NEW |
| 1 /* | 1 /* |
| 2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2012 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 |
| (...skipping 297 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... |
| 308 uint32_t timeStamp, | 308 uint32_t timeStamp, |
| 309 int64_t capture_time_ms, | 309 int64_t capture_time_ms, |
| 310 const uint8_t* payloadData, | 310 const uint8_t* payloadData, |
| 311 size_t payloadSize, | 311 size_t payloadSize, |
| 312 const RTPFragmentationHeader* fragmentation = NULL, | 312 const RTPFragmentationHeader* fragmentation = NULL, |
| 313 const RTPVideoHeader* rtpVideoHdr = NULL) = 0; | 313 const RTPVideoHeader* rtpVideoHdr = NULL) = 0; |
| 314 | 314 |
| 315 virtual bool TimeToSendPacket(uint32_t ssrc, | 315 virtual bool TimeToSendPacket(uint32_t ssrc, |
| 316 uint16_t sequence_number, | 316 uint16_t sequence_number, |
| 317 int64_t capture_time_ms, | 317 int64_t capture_time_ms, |
| 318 bool retransmission) = 0; | 318 bool retransmission, |
| 319 int probe_cluster_id) = 0; |
| 319 | 320 |
| 320 virtual size_t TimeToSendPadding(size_t bytes) = 0; | 321 virtual size_t TimeToSendPadding(size_t bytes, int probe_cluster_id) = 0; |
| 321 | 322 |
| 322 // Called on generation of new statistics after an RTP send. | 323 // Called on generation of new statistics after an RTP send. |
| 323 virtual void RegisterSendChannelRtpStatisticsCallback( | 324 virtual void RegisterSendChannelRtpStatisticsCallback( |
| 324 StreamDataCountersCallback* callback) = 0; | 325 StreamDataCountersCallback* callback) = 0; |
| 325 virtual StreamDataCountersCallback* | 326 virtual StreamDataCountersCallback* |
| 326 GetSendChannelRtpStatisticsCallback() const = 0; | 327 GetSendChannelRtpStatisticsCallback() const = 0; |
| 327 | 328 |
| 328 /************************************************************************** | 329 /************************************************************************** |
| 329 * | 330 * |
| 330 * RTCP | 331 * RTCP |
| (...skipping 316 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... |
| 647 | 648 |
| 648 /* | 649 /* |
| 649 * send a request for a keyframe | 650 * send a request for a keyframe |
| 650 * | 651 * |
| 651 * return -1 on failure else 0 | 652 * return -1 on failure else 0 |
| 652 */ | 653 */ |
| 653 virtual int32_t RequestKeyFrame() = 0; | 654 virtual int32_t RequestKeyFrame() = 0; |
| 654 }; | 655 }; |
| 655 } // namespace webrtc | 656 } // namespace webrtc |
| 656 #endif // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_H_ | 657 #endif // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_H_ |
| OLD | NEW |