Files
platform-external-webrtc/modules/rtp_rtcp/source/receive_statistics_impl.h
Taylor Brandstetter 84916937b7 Update packetsLost and jitter stats any time a packet is received.
Before this CL, the packetsLost and jitter stats (as returned by
GetStats, at the API level) were only being updated when an RTCP SR or
RR is generated. According to the stats spec, "local" stats like this
should be updated any time a packet is received.

This CL also fixes some minor issues with the calculation of packetsLost
(and fractionLost):
* Packets weren't being count as lost if lost over a sequence number
  rollover.
* Temporary periods of "negative" loss (caused by duplicate or out of
  order packets) weren't being accumulated into the cumulative loss
  counter. Example:
  Period 1: Received packets 1, 2, 4
    Loss over that period: 1 (expected 4 packets, got 3)
    Reported cumulative loss: 1
  Period 2: Received packets 3, 5
    Loss over that period: -1 (expected 1 packet, got 2)
    Reported cumulative loss: 1 (should be 0!)

Landing with NOTRY because Android compile bots are broken for an
unrelated reason.
NOTRY=True

Bug: webrtc:8804
Change-Id: I840ba34de8957b1276f6bdaf93718f805629f5c8
Reviewed-on: https://webrtc-review.googlesource.com/50020
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23731}
2018-06-25 23:56:39 +00:00

137 lines
5.0 KiB
C++

/*
* 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 <math.h>
#include <algorithm>
#include <map>
#include <vector>
#include "rtc_base/criticalsection.h"
#include "rtc_base/rate_statistics.h"
#include "system_wrappers/include/ntp_time.h"
namespace webrtc {
class StreamStatisticianImpl : public StreamStatistician {
public:
StreamStatisticianImpl(uint32_t ssrc,
Clock* clock,
RtcpStatisticsCallback* rtcp_callback,
StreamDataCountersCallback* rtp_callback);
~StreamStatisticianImpl() override;
bool GetStatistics(RtcpStatistics* statistics,
bool update_fraction_lost) 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;
bool IsRetransmitOfOldPacket(const RTPHeader& header) const override;
bool IsPacketInOrder(uint16_t sequence_number) const override;
void IncomingPacket(const RTPHeader& rtp_header,
size_t packet_length,
bool retransmitted);
void FecPacketReceived(const RTPHeader& header, size_t packet_length);
void SetMaxReorderingThreshold(int max_reordering_threshold);
private:
bool InOrderPacketInternal(uint16_t sequence_number) const
RTC_EXCLUSIVE_LOCKS_REQUIRED(stream_lock_);
RtcpStatistics CalculateRtcpStatistics(bool update_fraction_lost)
RTC_EXCLUSIVE_LOCKS_REQUIRED(stream_lock_);
void UpdateJitter(const RTPHeader& header, NtpTime receive_time);
StreamDataCounters UpdateCounters(const RTPHeader& rtp_header,
size_t packet_length,
bool retransmitted)
RTC_EXCLUSIVE_LOCKS_REQUIRED(stream_lock_);
const uint32_t ssrc_;
Clock* const clock_;
rtc::CriticalSection stream_lock_;
RateStatistics incoming_bitrate_;
int max_reordering_threshold_; // In number of packets or sequence numbers.
// Stats on received RTP packets.
uint32_t jitter_q4_;
int64_t last_receive_time_ms_;
NtpTime last_receive_time_ntp_;
uint32_t last_received_timestamp_;
uint16_t received_seq_first_;
uint16_t received_seq_max_;
uint16_t received_seq_wraps_;
// Current counter values.
size_t received_packet_overhead_;
StreamDataCounters receive_counters_ RTC_GUARDED_BY(stream_lock_);
// Used to calculate fraction_lost between reports.
uint32_t last_report_received_packets_ = 0;
uint32_t last_report_extended_seq_max_ = 0;
uint8_t last_fraction_lost_ = 0;
// stream_lock_ shouldn't be held when calling callbacks.
RtcpStatisticsCallback* const rtcp_callback_;
StreamDataCountersCallback* const rtp_callback_;
};
class ReceiveStatisticsImpl : public ReceiveStatistics,
public RtcpStatisticsCallback,
public StreamDataCountersCallback {
public:
explicit ReceiveStatisticsImpl(Clock* clock);
~ReceiveStatisticsImpl() override;
// Implement ReceiveStatisticsProvider.
std::vector<rtcp::ReportBlock> RtcpReportBlocks(size_t max_blocks) override;
// Implement ReceiveStatistics.
void IncomingPacket(const RTPHeader& header,
size_t packet_length,
bool retransmitted) override;
void FecPacketReceived(const RTPHeader& header,
size_t packet_length) override;
StreamStatistician* GetStatistician(uint32_t ssrc) const override;
void SetMaxReorderingThreshold(int max_reordering_threshold) override;
void RegisterRtcpStatisticsCallback(
RtcpStatisticsCallback* callback) override;
void RegisterRtpStatisticsCallback(
StreamDataCountersCallback* callback) override;
private:
void StatisticsUpdated(const RtcpStatistics& statistics,
uint32_t ssrc) override;
void CNameChanged(const char* cname, uint32_t ssrc) override;
void DataCountersUpdated(const StreamDataCounters& counters,
uint32_t ssrc) override;
Clock* const clock_;
rtc::CriticalSection receive_statistics_lock_;
uint32_t last_returned_ssrc_;
std::map<uint32_t, StreamStatisticianImpl*> statisticians_;
RtcpStatisticsCallback* rtcp_stats_callback_;
StreamDataCountersCallback* rtp_stats_callback_;
};
} // namespace webrtc
#endif // MODULES_RTP_RTCP_SOURCE_RECEIVE_STATISTICS_IMPL_H_