mirror of
https://github.com/oxen-io/session-android.git
synced 2024-11-24 18:45:19 +00:00
d83a3d71bc
Merge in RedPhone // FREEBIE
59 lines
2.0 KiB
C++
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_
|