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/*
* Copyright (c) 2013 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_RTP_RTCP_SOURCE_RECEIVE_STATISTICS_IMPL_H_
#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RECEIVE_STATISTICS_IMPL_H_
#include "webrtc/modules/rtp_rtcp/interface/receive_statistics.h"
#include <algorithm>
#include "webrtc/modules/rtp_rtcp/source/bitrate.h"
#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
namespace webrtc {
class CriticalSectionWrapper;
class StreamStatisticianImpl : public StreamStatistician {
public:
explicit StreamStatisticianImpl(Clock* clock);
virtual ~StreamStatisticianImpl() {}
virtual bool GetStatistics(Statistics* statistics, bool reset) OVERRIDE;
virtual void GetDataCounters(uint32_t* bytes_received,
uint32_t* packets_received) const OVERRIDE;
virtual uint32_t BitrateReceived() const OVERRIDE;
virtual void ResetStatistics() OVERRIDE;
virtual bool IsRetransmitOfOldPacket(const RTPHeader& header,
int min_rtt) const OVERRIDE;
virtual bool IsPacketInOrder(uint16_t sequence_number) const OVERRIDE;
void IncomingPacket(const RTPHeader& rtp_header, size_t bytes,
bool retransmitted);
void SetMaxReorderingThreshold(int max_reordering_threshold);
void ProcessBitrate();
virtual void LastReceiveTimeNtp(uint32_t* secs, uint32_t* frac) const;
private:
bool InOrderPacketInternal(uint16_t sequence_number) const;
Clock* clock_;
scoped_ptr<CriticalSectionWrapper> crit_sect_;
Bitrate incoming_bitrate_;
uint32_t ssrc_;
int max_reordering_threshold_; // In number of packets or sequence numbers.
// Stats on received RTP packets.
uint32_t jitter_q4_;
uint32_t jitter_max_q4_;
uint32_t cumulative_loss_;
uint32_t jitter_q4_transmission_time_offset_;
int64_t last_receive_time_ms_;
uint32_t last_receive_time_secs_;
uint32_t last_receive_time_frac_;
uint32_t last_received_timestamp_;
int32_t last_received_transmission_time_offset_;
uint16_t received_seq_first_;
uint16_t received_seq_max_;
uint16_t received_seq_wraps_;
bool first_packet_;
// Current counter values.
uint16_t received_packet_overhead_;
uint32_t received_byte_count_;
uint32_t received_retransmitted_packets_;
uint32_t received_inorder_packet_count_;
// Counter values when we sent the last report.
uint32_t last_report_inorder_packets_;
uint32_t last_report_old_packets_;
uint16_t last_report_seq_max_;
Statistics last_reported_statistics_;
};
class ReceiveStatisticsImpl : public ReceiveStatistics {
public:
explicit ReceiveStatisticsImpl(Clock* clock);
~ReceiveStatisticsImpl();
// Implement ReceiveStatistics.
virtual void IncomingPacket(const RTPHeader& header, size_t bytes,
bool old_packet) OVERRIDE;
virtual StatisticianMap GetActiveStatisticians() const OVERRIDE;
virtual StreamStatistician* GetStatistician(uint32_t ssrc) const OVERRIDE;
virtual void SetMaxReorderingThreshold(int max_reordering_threshold) OVERRIDE;
// Implement Module.
virtual int32_t Process() OVERRIDE;
virtual int32_t TimeUntilNextProcess() OVERRIDE;
void ChangeSsrc(uint32_t from_ssrc, uint32_t to_ssrc);
private:
typedef std::map<uint32_t, StreamStatisticianImpl*> StatisticianImplMap;
Clock* clock_;
scoped_ptr<CriticalSectionWrapper> crit_sect_;
int64_t last_rate_update_ms_;
StatisticianImplMap statisticians_;
};
} // namespace webrtc
#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RECEIVE_STATISTICS_IMPL_H_