84916937b7
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}
100 lines
3.6 KiB
C++
100 lines
3.6 KiB
C++
/*
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* Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef MODULES_RTP_RTCP_INCLUDE_RECEIVE_STATISTICS_H_
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#define MODULES_RTP_RTCP_INCLUDE_RECEIVE_STATISTICS_H_
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#include <map>
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#include <vector>
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#include "modules/include/module.h"
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#include "modules/include/module_common_types.h"
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#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
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#include "modules/rtp_rtcp/source/rtcp_packet/report_block.h"
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#include "typedefs.h" // NOLINT(build/include)
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namespace webrtc {
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class Clock;
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class ReceiveStatisticsProvider {
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public:
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virtual ~ReceiveStatisticsProvider() = default;
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// Collects receive statistic in a form of rtcp report blocks.
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// Returns at most |max_blocks| report blocks.
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virtual std::vector<rtcp::ReportBlock> RtcpReportBlocks(
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size_t max_blocks) = 0;
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};
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class StreamStatistician {
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public:
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virtual ~StreamStatistician();
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// If |update_fraction_lost| is true, |fraction_lost| will be recomputed
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// between now and the last time |update_fraction_lost| was true. Otherwise
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// the last-computed value of |fraction_lost| will be returned.
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//
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// |update_fraction_lost| should be true any time an RTCP SR or RR is being
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// generated, since RFC3550 defines it as the fraction of packets lost since
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// the previous SR or RR packet was sent.
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//
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// Aside from |fraction_lost|, every other value will be freshly computed.
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virtual bool GetStatistics(RtcpStatistics* statistics,
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bool update_fraction_lost) = 0;
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virtual void GetDataCounters(size_t* bytes_received,
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uint32_t* packets_received) const = 0;
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// Gets received stream data counters (includes reset counter values).
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virtual void GetReceiveStreamDataCounters(
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StreamDataCounters* data_counters) const = 0;
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virtual uint32_t BitrateReceived() const = 0;
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// Returns true if the packet with RTP header |header| is likely to be a
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// retransmitted packet, false otherwise.
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virtual bool IsRetransmitOfOldPacket(const RTPHeader& header) const = 0;
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// Returns true if |sequence_number| is received in order, false otherwise.
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virtual bool IsPacketInOrder(uint16_t sequence_number) const = 0;
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};
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class ReceiveStatistics : public ReceiveStatisticsProvider {
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public:
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~ReceiveStatistics() override = default;
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static ReceiveStatistics* Create(Clock* clock);
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// Updates the receive statistics with this packet.
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virtual void IncomingPacket(const RTPHeader& rtp_header,
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size_t packet_length,
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bool retransmitted) = 0;
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// Increment counter for number of FEC packets received.
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virtual void FecPacketReceived(const RTPHeader& header,
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size_t packet_length) = 0;
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// Returns a pointer to the statistician of an ssrc.
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virtual StreamStatistician* GetStatistician(uint32_t ssrc) const = 0;
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// Sets the max reordering threshold in number of packets.
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virtual void SetMaxReorderingThreshold(int max_reordering_threshold) = 0;
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// Called on new RTCP stats creation.
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virtual void RegisterRtcpStatisticsCallback(
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RtcpStatisticsCallback* callback) = 0;
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// Called on new RTP stats creation.
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virtual void RegisterRtpStatisticsCallback(
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StreamDataCountersCallback* callback) = 0;
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};
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} // namespace webrtc
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#endif // MODULES_RTP_RTCP_INCLUDE_RECEIVE_STATISTICS_H_
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