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Side by Side Diff: webrtc/modules/rtp_rtcp/include/rtp_rtcp.h

Issue 2005313003: Propagate probing cluster id to SendTimeHistory. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Test probe_cluster_id in PacketRouterTest. Created 4 years, 6 months ago
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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
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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
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_
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