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

Issue 2061423003: Refactor NACK bitrate allocation (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Fixed data race 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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279 * get send status 279 * get send status
280 */ 280 */
281 virtual bool SendingMedia() const = 0; 281 virtual bool SendingMedia() const = 0;
282 282
283 /* 283 /*
284 * get sent bitrate in Kbit/s 284 * get sent bitrate in Kbit/s
285 */ 285 */
286 virtual void BitrateSent(uint32_t* totalRate, 286 virtual void BitrateSent(uint32_t* totalRate,
287 uint32_t* videoRate, 287 uint32_t* videoRate,
288 uint32_t* fecRate, 288 uint32_t* fecRate,
289 uint32_t* nackRate) const = 0; 289 uint32_t* nackRate) = 0;
danilchap 2016/06/20 16:53:46 this would break downstream code.
sprang_webrtc 2016/06/23 07:54:29 Acknowledged.
290 290
291 /* 291 /*
292 * Used by the codec module to deliver a video or audio frame for 292 * Used by the codec module to deliver a video or audio frame for
293 * packetization. 293 * packetization.
294 * 294 *
295 * frameType - type of frame to send 295 * frameType - type of frame to send
296 * payloadType - payload type of frame to send 296 * payloadType - payload type of frame to send
297 * timestamp - timestamp of frame to send 297 * timestamp - timestamp of frame to send
298 * payloadData - payload buffer of frame to send 298 * payloadData - payload buffer of frame to send
299 * payloadSize - size of payload buffer to send 299 * payloadSize - size of payload buffer to send
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647 647
648 /* 648 /*
649 * send a request for a keyframe 649 * send a request for a keyframe
650 * 650 *
651 * return -1 on failure else 0 651 * return -1 on failure else 0
652 */ 652 */
653 virtual int32_t RequestKeyFrame() = 0; 653 virtual int32_t RequestKeyFrame() = 0;
654 }; 654 };
655 } // namespace webrtc 655 } // namespace webrtc
656 #endif // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_H_ 656 #endif // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_H_
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