97d0489058
- total bitrate ("WebRTC.Video.BitrateSentInKbps")
- media bitrate ("WebRTC.Video.MediaBitrateSentInKbps")
- rtx bitrate ("WebRTC.Video.RtxBitrateSentInKbps")
- padding bitrate ("WebRTC.Video.PaddingBitrateSentInKbps")
- retransmitted bitrate ("WebRTC.Video.RetransmittedBitrateInKbps")
Add retransmitted bytes to StreamDataCounters.
Change in UpdateRtpStats to also update counters for retransmitted packet.
BUG=crbug/419657
R=mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/30199004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7838 4adac7df-926f-26a2-2b94-8c16560cd09d
104 lines
3.7 KiB
C++
104 lines
3.7 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 WEBRTC_MODULES_RTP_RTCP_INTERFACE_RECEIVE_STATISTICS_H_
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#define WEBRTC_MODULES_RTP_RTCP_INTERFACE_RECEIVE_STATISTICS_H_
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#include <map>
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#include "webrtc/modules/interface/module.h"
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#include "webrtc/modules/interface/module_common_types.h"
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#include "webrtc/typedefs.h"
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namespace webrtc {
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class Clock;
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class StreamStatistician {
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public:
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virtual ~StreamStatistician();
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virtual bool GetStatistics(RtcpStatistics* statistics, bool reset) = 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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// Resets all statistics.
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virtual void ResetStatistics() = 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,
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int min_rtt) 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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typedef std::map<uint32_t, StreamStatistician*> StatisticianMap;
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class ReceiveStatistics : public Module {
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public:
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virtual ~ReceiveStatistics() {}
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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(uint32_t ssrc) = 0;
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// Returns a map of all statisticians which have seen an incoming packet
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// during the last two seconds.
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virtual StatisticianMap GetActiveStatisticians() const = 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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class NullReceiveStatistics : public ReceiveStatistics {
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public:
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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) OVERRIDE;
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virtual void FecPacketReceived(uint32_t ssrc) OVERRIDE;
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virtual StatisticianMap GetActiveStatisticians() const OVERRIDE;
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virtual StreamStatistician* GetStatistician(uint32_t ssrc) const OVERRIDE;
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virtual int32_t TimeUntilNextProcess() OVERRIDE;
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virtual int32_t Process() OVERRIDE;
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virtual void SetMaxReorderingThreshold(int max_reordering_threshold) OVERRIDE;
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virtual void RegisterRtcpStatisticsCallback(RtcpStatisticsCallback* callback)
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OVERRIDE;
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virtual void RegisterRtpStatisticsCallback(
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StreamDataCountersCallback* callback) OVERRIDE;
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_RTP_RTCP_INTERFACE_RECEIVE_STATISTICS_H_
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