| /* | |
| * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. | |
| * | |
| * Use of this source code is governed by a BSD-style license | |
| * that can be found in the LICENSE file in the root of the source | |
| * tree. An additional intellectual property rights grant can be found | |
| * in the file PATENTS. All contributing project authors may | |
| * be found in the AUTHORS file in the root of the source tree. | |
| */ | |
| #ifndef MODULES_AUDIO_CODING_NETEQ_RTCP_H_ | |
| #define MODULES_AUDIO_CODING_NETEQ_RTCP_H_ | |
| #include "modules/audio_coding/neteq/include/neteq.h" | |
| #include "rtc_base/constructormagic.h" | |
| namespace webrtc { | |
| // Forward declaration. | |
| struct RTPHeader; | |
| class Rtcp { | |
| public: | |
| Rtcp() { Init(0); } | |
| ~Rtcp() {} | |
| // Resets the RTCP statistics, and sets the first received sequence number. | |
| void Init(uint16_t start_sequence_number); | |
| // Updates the RTCP statistics with a new received packet. | |
| void Update(const RTPHeader& rtp_header, uint32_t receive_timestamp); | |
| // Returns the current RTCP statistics. If |no_reset| is true, the statistics | |
| // are not reset, otherwise they are. | |
| void GetStatistics(bool no_reset, RtcpStatistics* stats); | |
| private: | |
| uint16_t cycles_; // The number of wrap-arounds for the sequence number. | |
| uint16_t max_seq_no_; // The maximum sequence number received. Starts over | |
| // from 0 after wrap-around. | |
| uint16_t base_seq_no_; // The sequence number of the first received packet. | |
| uint32_t received_packets_; // The number of packets that have been received. | |
| uint32_t received_packets_prior_; // Number of packets received when last | |
| // report was generated. | |
| uint32_t expected_prior_; // Expected number of packets, at the time of the | |
| // last report. | |
| int64_t jitter_; // Current jitter value in Q4. | |
| int32_t transit_; // Clock difference for previous packet. | |
| RTC_DISALLOW_COPY_AND_ASSIGN(Rtcp); | |
| }; | |
| } // namespace webrtc | |
| #endif // MODULES_AUDIO_CODING_NETEQ_RTCP_H_ |