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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 MODULES_RTP_RTCP_SOURCE_RECEIVE_STATISTICS_IMPL_H_
#define MODULES_RTP_RTCP_SOURCE_RECEIVE_STATISTICS_IMPL_H_
#include "modules/rtp_rtcp/include/receive_statistics.h"
#include <algorithm>
#include <map>
#include <vector>
#include "absl/types/optional.h"
#include "modules/include/module_common_types_public.h"
#include "rtc_base/critical_section.h"
#include "rtc_base/rate_statistics.h"
#include "rtc_base/thread_annotations.h"
namespace webrtc {
class StreamStatisticianImpl : public StreamStatistician,
public RtpPacketSinkInterface {
public:
StreamStatisticianImpl(uint32_t ssrc,
Clock* clock,
bool enable_retransmit_detection,
int max_reordering_threshold,
RtcpStatisticsCallback* rtcp_callback,
StreamDataCountersCallback* rtp_callback);
~StreamStatisticianImpl() override;
// |reset| here and in next method restarts calculation of fraction_lost stat.
bool GetStatistics(RtcpStatistics* statistics, bool reset) override;
bool GetActiveStatisticsAndReset(RtcpStatistics* statistics);
void GetDataCounters(size_t* bytes_received,
uint32_t* packets_received) const override;
void GetReceiveStreamDataCounters(
StreamDataCounters* data_counters) const override;
uint32_t BitrateReceived() const override;
// Implements RtpPacketSinkInterface
void OnRtpPacket(const RtpPacketReceived& packet) override;
void FecPacketReceived(const RtpPacketReceived& packet);
void SetMaxReorderingThreshold(int max_reordering_threshold);
void EnableRetransmitDetection(bool enable);
private:
bool IsRetransmitOfOldPacket(const RtpPacketReceived& packet,
int64_t now_ms) const
RTC_EXCLUSIVE_LOCKS_REQUIRED(stream_lock_);
RtcpStatistics CalculateRtcpStatistics()
RTC_EXCLUSIVE_LOCKS_REQUIRED(stream_lock_);
void UpdateJitter(const RtpPacketReceived& packet, int64_t receive_time_ms)
RTC_EXCLUSIVE_LOCKS_REQUIRED(stream_lock_);
// Updates StreamStatistician for out of order packets.
// Returns true if packet considered to be out of order.
bool UpdateOutOfOrder(const RtpPacketReceived& packet,
int64_t sequence_number,
int64_t now_ms)
RTC_EXCLUSIVE_LOCKS_REQUIRED(stream_lock_);
// Updates StreamStatistician for incoming packets.
StreamDataCounters UpdateCounters(const RtpPacketReceived& packet);
// Checks if this StreamStatistician received any rtp packets.
bool ReceivedRtpPacket() const RTC_EXCLUSIVE_LOCKS_REQUIRED(stream_lock_) {
return received_seq_max_ >= 0;
}
const uint32_t ssrc_;
Clock* const clock_;
rtc::CriticalSection stream_lock_;
RateStatistics incoming_bitrate_ RTC_GUARDED_BY(&stream_lock_);
// In number of packets or sequence numbers.
int max_reordering_threshold_ RTC_GUARDED_BY(&stream_lock_);
bool enable_retransmit_detection_ RTC_GUARDED_BY(&stream_lock_);
// Stats on received RTP packets.
uint32_t jitter_q4_ RTC_GUARDED_BY(&stream_lock_);
uint32_t cumulative_loss_ RTC_GUARDED_BY(&stream_lock_);
int64_t last_receive_time_ms_ RTC_GUARDED_BY(&stream_lock_);
uint32_t last_received_timestamp_ RTC_GUARDED_BY(&stream_lock_);
SequenceNumberUnwrapper seq_unwrapper_ RTC_GUARDED_BY(&stream_lock_);
int64_t received_seq_first_ RTC_GUARDED_BY(&stream_lock_);
int64_t received_seq_max_ RTC_GUARDED_BY(&stream_lock_);
// Assume that the other side restarted when there are two sequential packets
// with large jump from received_seq_max_.
absl::optional<uint16_t> received_seq_out_of_order_
RTC_GUARDED_BY(&stream_lock_);
// Current counter values.
StreamDataCounters receive_counters_ RTC_GUARDED_BY(&stream_lock_);
// Counter values when we sent the last report.
uint32_t last_report_inorder_packets_ RTC_GUARDED_BY(&stream_lock_);
uint32_t last_report_old_packets_ RTC_GUARDED_BY(&stream_lock_);
int64_t last_report_seq_max_ RTC_GUARDED_BY(&stream_lock_);
RtcpStatistics last_reported_statistics_ RTC_GUARDED_BY(&stream_lock_);
// stream_lock_ shouldn't be held when calling callbacks.
RtcpStatisticsCallback* const rtcp_callback_;
StreamDataCountersCallback* const rtp_callback_;
};
class ReceiveStatisticsImpl : public ReceiveStatistics {
public:
ReceiveStatisticsImpl(Clock* clock,
RtcpStatisticsCallback* rtcp_callback,
StreamDataCountersCallback* rtp_callback);
~ReceiveStatisticsImpl() override;
// Implements ReceiveStatisticsProvider.
std::vector<rtcp::ReportBlock> RtcpReportBlocks(size_t max_blocks) override;
// Implements RtpPacketSinkInterface
void OnRtpPacket(const RtpPacketReceived& packet) override;
// Implements ReceiveStatistics.
void FecPacketReceived(const RtpPacketReceived& packet) override;
StreamStatistician* GetStatistician(uint32_t ssrc) const override;
void SetMaxReorderingThreshold(int max_reordering_threshold) override;
void EnableRetransmitDetection(uint32_t ssrc, bool enable) override;
private:
Clock* const clock_;
rtc::CriticalSection receive_statistics_lock_;
uint32_t last_returned_ssrc_;
int max_reordering_threshold_ RTC_GUARDED_BY(receive_statistics_lock_);
std::map<uint32_t, StreamStatisticianImpl*> statisticians_
RTC_GUARDED_BY(receive_statistics_lock_);
RtcpStatisticsCallback* const rtcp_stats_callback_;
StreamDataCountersCallback* const rtp_stats_callback_;
};
} // namespace webrtc
#endif // MODULES_RTP_RTCP_SOURCE_RECEIVE_STATISTICS_IMPL_H_