7bb8f02274
This is accomplished by breaking out RTX and FEC/RED functionality from the RTP module and keeping track of the base payload type, that is the payload type received when not receiving RTX. Enables retransmissions over RTX by default in the loopback test. BUG=1811 TESTS=voe/vie_auto_test --automated and trybots. R=mflodman@webrtc.org, pbos@webrtc.org, xians@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2154004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4692 4adac7df-926f-26a2-2b94-8c16560cd09d
96 lines
3.2 KiB
C++
96 lines
3.2 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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struct Statistics {
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Statistics()
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: fraction_lost(0),
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cumulative_lost(0),
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extended_max_sequence_number(0),
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jitter(0),
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max_jitter(0) {}
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uint8_t fraction_lost;
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uint32_t cumulative_lost;
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uint32_t extended_max_sequence_number;
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uint32_t jitter;
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uint32_t max_jitter;
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};
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virtual ~StreamStatistician();
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virtual bool GetStatistics(Statistics* statistics, bool reset) = 0;
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virtual void GetDataCounters(uint32_t* bytes_received,
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uint32_t* packets_received) 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, size_t bytes,
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bool retransmitted) = 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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};
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class NullReceiveStatistics : public ReceiveStatistics {
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public:
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virtual void IncomingPacket(const RTPHeader& rtp_header, size_t bytes,
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bool retransmitted) 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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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_RTP_RTCP_INTERFACE_RECEIVE_STATISTICS_H_
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