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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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