session-android/jni/webrtc/modules/audio_coding/neteq/rtcp.h
Moxie Marlinspike d83a3d71bc Support for Signal calls.
Merge in RedPhone

// FREEBIE
2015-09-30 14:30:09 -07:00

59 lines
2.0 KiB
C++

/*
* 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 WEBRTC_MODULES_AUDIO_CODING_NETEQ_RTCP_H_
#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_RTCP_H_
#include "webrtc/base/constructormagic.h"
#include "webrtc/modules/audio_coding/neteq/interface/neteq.h"
#include "webrtc/typedefs.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.
uint32_t jitter_; // Current jitter value.
int32_t transit_; // Clock difference for previous packet.
DISALLOW_COPY_AND_ASSIGN(Rtcp);
};
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_RTCP_H_