2370b0831f
This reverts commit 84916937b70472715efe5682bc273e91c3a72695. Reason for revert: breaking downstream projects. Original change's description: > 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} TBR=danilchap@webrtc.org,deadbeef@webrtc.org,ossu@webrtc.org # Not skipping CQ checks because original CL landed > 1 day ago. Landing with NOTRY because ios64_sim_ios10_dbg bot is broken. Passing all other bots. NOTRY=True Bug: webrtc:8804 Change-Id: I07bd6b1206d5a8d211792ad392842f9ed6c505e9 Reviewed-on: https://webrtc-review.googlesource.com/95280 Commit-Queue: Qingsi Wang <qingsi@webrtc.org> Reviewed-by: Qingsi Wang <qingsi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24370}
86 lines
2.9 KiB
C++
86 lines
2.9 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_INCLUDE_RECEIVE_STATISTICS_H_
|
|
#define MODULES_RTP_RTCP_INCLUDE_RECEIVE_STATISTICS_H_
|
|
|
|
#include <map>
|
|
#include <vector>
|
|
|
|
#include "modules/include/module.h"
|
|
#include "modules/include/module_common_types.h"
|
|
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
|
|
#include "modules/rtp_rtcp/source/rtcp_packet/report_block.h"
|
|
|
|
namespace webrtc {
|
|
|
|
class Clock;
|
|
|
|
class ReceiveStatisticsProvider {
|
|
public:
|
|
virtual ~ReceiveStatisticsProvider() = default;
|
|
// Collects receive statistic in a form of rtcp report blocks.
|
|
// Returns at most |max_blocks| report blocks.
|
|
virtual std::vector<rtcp::ReportBlock> RtcpReportBlocks(
|
|
size_t max_blocks) = 0;
|
|
};
|
|
|
|
class StreamStatistician {
|
|
public:
|
|
virtual ~StreamStatistician();
|
|
|
|
virtual bool GetStatistics(RtcpStatistics* statistics, bool reset) = 0;
|
|
virtual void GetDataCounters(size_t* bytes_received,
|
|
uint32_t* packets_received) const = 0;
|
|
|
|
// Gets received stream data counters (includes reset counter values).
|
|
virtual void GetReceiveStreamDataCounters(
|
|
StreamDataCounters* data_counters) const = 0;
|
|
|
|
virtual uint32_t BitrateReceived() const = 0;
|
|
|
|
// Returns true if the packet with RTP header |header| is likely to be a
|
|
// retransmitted packet, false otherwise.
|
|
virtual bool IsRetransmitOfOldPacket(const RTPHeader& header) const = 0;
|
|
};
|
|
|
|
class ReceiveStatistics : public ReceiveStatisticsProvider {
|
|
public:
|
|
~ReceiveStatistics() override = default;
|
|
|
|
static ReceiveStatistics* Create(Clock* clock);
|
|
|
|
// Updates the receive statistics with this packet.
|
|
virtual void IncomingPacket(const RTPHeader& rtp_header,
|
|
size_t packet_length,
|
|
bool retransmitted) = 0;
|
|
|
|
// Increment counter for number of FEC packets received.
|
|
virtual void FecPacketReceived(const RTPHeader& header,
|
|
size_t packet_length) = 0;
|
|
|
|
// Returns a pointer to the statistician of an ssrc.
|
|
virtual StreamStatistician* GetStatistician(uint32_t ssrc) const = 0;
|
|
|
|
// Sets the max reordering threshold in number of packets.
|
|
virtual void SetMaxReorderingThreshold(int max_reordering_threshold) = 0;
|
|
|
|
// Called on new RTCP stats creation.
|
|
virtual void RegisterRtcpStatisticsCallback(
|
|
RtcpStatisticsCallback* callback) = 0;
|
|
|
|
// Called on new RTP stats creation.
|
|
virtual void RegisterRtpStatisticsCallback(
|
|
StreamDataCountersCallback* callback) = 0;
|
|
};
|
|
|
|
} // namespace webrtc
|
|
#endif // MODULES_RTP_RTCP_INCLUDE_RECEIVE_STATISTICS_H_
|