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Side by Side Diff: webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.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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380 // while we are paused. 380 // while we are paused.
381 381
382 // Returns true if we send the packet now, else it will add the packet 382 // Returns true if we send the packet now, else it will add the packet
383 // information to the queue and call TimeToSendPacket when it's time to send. 383 // information to the queue and call TimeToSendPacket when it's time to send.
384 virtual void InsertPacket(Priority priority, 384 virtual void InsertPacket(Priority priority,
385 uint32_t ssrc, 385 uint32_t ssrc,
386 uint16_t sequence_number, 386 uint16_t sequence_number,
387 int64_t capture_time_ms, 387 int64_t capture_time_ms,
388 size_t bytes, 388 size_t bytes,
389 bool retransmission) = 0; 389 bool retransmission) = 0;
390
391 // Try to allocate bitrate for sending a retransmission. This may fail if too
392 // much retransmission is requested in relation to payload.
393 virtual bool AllocateRetransmissionBitrate(size_t bytes) = 0;
394
395 virtual int CurrentRetransmissionBitrate() = 0;
danilchap 2016/06/23 12:46:12 maybe call it CurrentRetransmisisonBitrateBps to p
sprang_webrtc 2016/06/28 09:12:33 Done. And removed. :)
390 }; 396 };
391 397
392 class TransportSequenceNumberAllocator { 398 class TransportSequenceNumberAllocator {
393 public: 399 public:
394 TransportSequenceNumberAllocator() {} 400 TransportSequenceNumberAllocator() {}
395 virtual ~TransportSequenceNumberAllocator() {} 401 virtual ~TransportSequenceNumberAllocator() {}
396 402
397 virtual uint16_t AllocateSequenceNumber() = 0; 403 virtual uint16_t AllocateSequenceNumber() = 0;
398 }; 404 };
399 405
400 } // namespace webrtc 406 } // namespace webrtc
401 #endif // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_DEFINES_H_ 407 #endif // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_DEFINES_H_
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