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Side by Side Diff: webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h

Issue 1335353005: Remove channel ids from various interfaces. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: rebase Created 5 years, 3 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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346 FRIEND_TEST_ALL_PREFIXES(RtpRtcpImplTest, Rtt); 346 FRIEND_TEST_ALL_PREFIXES(RtpRtcpImplTest, Rtt);
347 FRIEND_TEST_ALL_PREFIXES(RtpRtcpImplTest, RttForReceiverOnly); 347 FRIEND_TEST_ALL_PREFIXES(RtpRtcpImplTest, RttForReceiverOnly);
348 int64_t RtcpReportInterval(); 348 int64_t RtcpReportInterval();
349 void SetRtcpReceiverSsrcs(uint32_t main_ssrc); 349 void SetRtcpReceiverSsrcs(uint32_t main_ssrc);
350 350
351 void set_rtt_ms(int64_t rtt_ms); 351 void set_rtt_ms(int64_t rtt_ms);
352 int64_t rtt_ms() const; 352 int64_t rtt_ms() const;
353 353
354 bool TimeToSendFullNackList(int64_t now) const; 354 bool TimeToSendFullNackList(int64_t now) const;
355 355
356 int32_t id_;
357 const bool audio_; 356 const bool audio_;
358 bool collision_detected_; 357 bool collision_detected_;
359 int64_t last_process_time_; 358 int64_t last_process_time_;
360 int64_t last_bitrate_process_time_; 359 int64_t last_bitrate_process_time_;
361 int64_t last_rtt_process_time_; 360 int64_t last_rtt_process_time_;
362 uint16_t packet_overhead_; 361 uint16_t packet_overhead_;
363 362
364 size_t padding_index_; 363 size_t padding_index_;
365 364
366 // Send side 365 // Send side
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380 PacketLossStats receive_loss_stats_; 379 PacketLossStats receive_loss_stats_;
381 380
382 // The processed RTT from RtcpRttStats. 381 // The processed RTT from RtcpRttStats.
383 rtc::scoped_ptr<CriticalSectionWrapper> critical_section_rtt_; 382 rtc::scoped_ptr<CriticalSectionWrapper> critical_section_rtt_;
384 int64_t rtt_ms_; 383 int64_t rtt_ms_;
385 }; 384 };
386 385
387 } // namespace webrtc 386 } // namespace webrtc
388 387
389 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RTCP_IMPL_H_ 388 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RTCP_IMPL_H_
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