Fix GetReceivedRTCPStatistics and GetSendRTCPStatistics.
Comments where wrong and removed error message when trying to get RTT time from GetReceivedRTCPStatistics. BUG= TEST= Review URL: http://webrtc-codereview.appspot.com/335013 git-svn-id: http://webrtc.googlecode.com/svn/trunk@1312 4adac7df-926f-26a2-2b94-8c16560cd09d
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src/video_engine/main/interface/vie_rtp_rtcp.h
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299
src/video_engine/main/interface/vie_rtp_rtcp.h
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/*
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* Copyright (c) 2011 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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// This sub-API supports the following functionalities:
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// - Callbacks for RTP and RTCP events such as modified SSRC or CSRC.
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// - SSRC handling.
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// - Transmission of RTCP reports.
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// - Obtaining RTCP data from incoming RTCP sender reports.
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// - RTP and RTCP statistics (jitter, packet loss, RTT etc.).
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// - Forward Error Correction (FEC).
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// - RTP Keep‐alive for maintaining the NAT mappings associated to RTP flows.
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// - Writing RTP and RTCP packets to binary files for off‐line analysis of the
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// call quality.
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// - Inserting extra RTP packets into active audio stream.
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#ifndef WEBRTC_VIDEO_ENGINE_MAIN_INTERFACE_VIE_RTP_RTCP_H_
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#define WEBRTC_VIDEO_ENGINE_MAIN_INTERFACE_VIE_RTP_RTCP_H_
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#include "common_types.h"
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namespace webrtc
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{
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class VideoEngine;
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// This enumerator sets the RTCP mode.
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enum ViERTCPMode
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{
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kRtcpNone = 0,
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kRtcpCompound_RFC4585 = 1,
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kRtcpNonCompound_RFC5506 = 2
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};
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// This enumerator describes the key frame request mode.
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enum ViEKeyFrameRequestMethod
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{
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kViEKeyFrameRequestNone = 0,
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kViEKeyFrameRequestPliRtcp = 1,
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kViEKeyFrameRequestFirRtp = 2,
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kViEKeyFrameRequestFirRtcp = 3
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};
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enum StreamType
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{
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kViEStreamTypeNormal = 0, // Normal media stream
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kViEStreamTypeRtx = 1 // Retransmission media stream
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};
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// ----------------------------------------------------------------------------
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// ViERTPObserver
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// ----------------------------------------------------------------------------
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// This class declares an abstract interface for a user defined observer. It is
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// up to the VideoEngine user to implement a derived class which implements the
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// observer class. The observer is registered using RegisterRTPObserver() and
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// deregistered using DeregisterRTPObserver().
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class WEBRTC_DLLEXPORT ViERTPObserver
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{
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public:
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// This method is called if SSRC of the incoming stream is changed.
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virtual void IncomingSSRCChanged(const int videoChannel,
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const unsigned int SSRC) = 0;
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// This method is called if a field in CSRC changes or if the number of
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// CSRCs changes.
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virtual void IncomingCSRCChanged(const int videoChannel,
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const unsigned int CSRC,
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const bool added) = 0;
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protected:
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virtual ~ViERTPObserver() {}
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};
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// ----------------------------------------------------------------------------
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// ViERTCPObserver
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// ----------------------------------------------------------------------------
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// This class declares an abstract interface for a user defined observer. It is
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// up to the VideoEngine user to implement a derived class which implements the
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// observer class. The observer is registered using RegisterRTCPObserver() and
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// deregistered using DeregisterRTCPObserver().
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class WEBRTC_DLLEXPORT ViERTCPObserver
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{
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public:
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// This method is called if a application-defined RTCP packet has been
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// received.
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virtual void OnApplicationDataReceived(
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const int videoChannel, const unsigned char subType,
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const unsigned int name, const char* data,
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const unsigned short dataLengthInBytes) = 0;
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protected:
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virtual ~ViERTCPObserver() {}
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};
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//
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class WEBRTC_DLLEXPORT ViERTP_RTCP
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{
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public:
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// Default values
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enum
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{
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KDefaultDeltaTransmitTimeSeconds = 15
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};
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enum
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{
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KMaxRTCPCNameLength = 256
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};
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// Factory for the ViERTP_RTCP sub‐API and increases an internal reference
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// counter if successful. Returns NULL if the API is not supported or if
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// construction fails.
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static ViERTP_RTCP* GetInterface(VideoEngine* videoEngine);
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// Releases the ViERTP_RTCP sub-API and decreases an internal reference
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// counter. Returns the new reference count. This value should be zero
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// for all sub-API:s before the VideoEngine object can be safely deleted.
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virtual int Release() = 0;
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// This function enables you to specify the RTP synchronization source
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// identifier (SSRC) explicitly.
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virtual int SetLocalSSRC(const int videoChannel,
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const unsigned int SSRC,
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const StreamType usage = kViEStreamTypeNormal,
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const unsigned char simulcastIdx = 0) = 0;
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// This function gets the SSRC for the outgoing RTP stream for the specified
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// channel.
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virtual int GetLocalSSRC(const int videoChannel,
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unsigned int& SSRC) const = 0;
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// This function map a incoming SSRC to a StreamType so that the engine
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// can know which is the normal stream and which is the RTX
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virtual int SetRemoteSSRCType(const int videoChannel,
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const StreamType usage,
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const unsigned int SSRC) const = 0;
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// This function gets the SSRC for the incoming RTP stream for the specified
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// channel.
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virtual int GetRemoteSSRC(const int videoChannel,
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unsigned int& SSRC) const = 0;
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// This function returns the CSRCs of the incoming RTP packets.
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virtual int GetRemoteCSRCs(const int videoChannel,
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unsigned int CSRCs[kRtpCsrcSize]) const = 0;
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// This function enables manual initialization of the sequence number. The
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// start sequence number is normally a random number.
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virtual int SetStartSequenceNumber(const int videoChannel,
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unsigned short sequenceNumber) = 0;
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// This function sets the RTCP status for the specified channel.
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// Default mode is kRtcpCompound_RFC4585.
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virtual int SetRTCPStatus(const int videoChannel,
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const ViERTCPMode rtcpMode) = 0;
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// This function gets the RTCP status for the specified channel.
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virtual int GetRTCPStatus(const int videoChannel,
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ViERTCPMode& rtcpMode) const = 0;
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// This function sets the RTCP canonical name (CNAME) for the RTCP reports
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// on a specific channel.
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virtual int SetRTCPCName(const int videoChannel,
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const char rtcpCName[KMaxRTCPCNameLength]) = 0;
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// This function gets the RTCP canonical name (CNAME) for the RTCP reports
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// sent the specified channel.
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virtual int GetRTCPCName(const int videoChannel,
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char rtcpCName[KMaxRTCPCNameLength]) const = 0;
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// This function gets the RTCP canonical name (CNAME) for the RTCP reports
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// received on the specified channel.
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virtual int GetRemoteRTCPCName(
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const int videoChannel, char rtcpCName[KMaxRTCPCNameLength]) const = 0;
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// This function sends an RTCP APP packet on a specific channel.
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virtual int SendApplicationDefinedRTCPPacket(
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const int videoChannel, const unsigned char subType,
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unsigned int name, const char* data,
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unsigned short dataLengthInBytes) = 0;
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// This function enables Negative Acknowledgment (NACK) using RTCP,
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// implemented based on RFC 4585. NACK retransmits RTP packets if lost on
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// the network. This creates a lossless transport at the expense of delay.
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// If using NACK, NACK should be enabled on both endpoints in a call.
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virtual int SetNACKStatus(const int videoChannel, const bool enable) = 0;
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// This function enables Forward Error Correction (FEC) using RTCP,
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// implemented based on RFC 5109, to improve packet loss robustness. Extra
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// FEC packets are sent together with the usual media packets, hence
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// part of the bitrate will be used for FEC packets.
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virtual int SetFECStatus(const int videoChannel, const bool enable,
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const unsigned char payloadTypeRED,
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const unsigned char payloadTypeFEC) = 0;
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// This function enables hybrid Negative Acknowledgment using RTCP
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// and Forward Error Correction (FEC) implemented based on RFC 5109,
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// to improve packet loss robustness. Extra
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// FEC packets are sent together with the usual media packets, hence will
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// part of the bitrate be used for FEC packets.
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// The hybrid mode will choose between nack only, fec only and both based on
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// network conditions. When both are applied, only packets that were not
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// recovered by the FEC will be nacked.
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virtual int SetHybridNACKFECStatus(const int videoChannel,
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const bool enable,
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const unsigned char payloadTypeRED,
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const unsigned char payloadTypeFEC) = 0;
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// This function enables RTCP key frame requests.
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virtual int SetKeyFrameRequestMethod(
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const int videoChannel, const ViEKeyFrameRequestMethod method) = 0;
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// This function enables signaling of temporary bitrate constraints using
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// RTCP, implemented based on RFC4585.
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virtual int SetTMMBRStatus(const int videoChannel, const bool enable) = 0;
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// This function returns our localy created statistics of the received
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// RTP stream.
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virtual int GetReceivedRTCPStatistics(
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const int videoChannel, unsigned short& fractionLost,
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unsigned int& cumulativeLost, unsigned int& extendedMax,
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unsigned int& jitter, int& rttMs) const = 0;
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// This function returns statistics reported by the remote client in a
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// RTCP packet.
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virtual int GetSentRTCPStatistics(const int videoChannel,
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unsigned short& fractionLost,
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unsigned int& cumulativeLost,
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unsigned int& extendedMax,
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unsigned int& jitter,
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int& rttMs) const = 0;
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// The function gets statistics from the sent and received RTP streams.
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virtual int GetRTPStatistics(const int videoChannel,
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unsigned int& bytesSent,
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unsigned int& packetsSent,
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unsigned int& bytesReceived,
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unsigned int& packetsReceived) const = 0;
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// The function gets bandwidth usage statistics from the sent RTP streams in
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// bits/s.
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virtual int GetBandwidthUsage(const int videoChannel,
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unsigned int& totalBitrateSent,
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unsigned int& videoBitrateSent,
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unsigned int& fecBitrateSent,
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unsigned int& nackBitrateSent) const = 0;
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// This function enables or disables an RTP keep-alive mechanism which can
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// be used to maintain an existing Network Address Translator (NAT) mapping
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// while regular RTP is no longer transmitted.
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virtual int SetRTPKeepAliveStatus(
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const int videoChannel, bool enable, const char unknownPayloadType,
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const unsigned int deltaTransmitTimeSeconds =
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KDefaultDeltaTransmitTimeSeconds) = 0;
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// This function gets the RTP keep-alive status.
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virtual int GetRTPKeepAliveStatus(
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const int videoChannel, bool& enabled, char& unkownPayloadType,
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unsigned int& deltaTransmitTimeSeconds) const = 0;
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// This function enables capturing of RTP packets to a binary file on a
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// specific channel and for a given direction. The file can later be
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// replayed using e.g. RTP Tools rtpplay since the binary file format is
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// compatible with the rtpdump format.
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virtual int StartRTPDump(const int videoChannel,
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const char fileNameUTF8[1024],
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RTPDirections direction) = 0;
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// This function disables capturing of RTP packets to a binary file on a
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// specific channel and for a given direction.
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virtual int StopRTPDump(const int videoChannel,
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RTPDirections direction) = 0;
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// Registers an instance of a user implementation of the ViERTPObserver.
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virtual int RegisterRTPObserver(const int videoChannel,
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ViERTPObserver& observer) = 0;
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// Removes a registered instance of ViERTPObserver.
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virtual int DeregisterRTPObserver(const int videoChannel) = 0;
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// Registers an instance of a user implementation of the ViERTCPObserver.
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virtual int RegisterRTCPObserver(const int videoChannel,
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ViERTCPObserver& observer) = 0;
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// Removes a registered instance of ViERTCPObserver.
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virtual int DeregisterRTCPObserver(const int videoChannel) = 0;
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protected:
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ViERTP_RTCP() {};
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virtual ~ViERTP_RTCP() {};
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};
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} // namespace webrtc
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#endif // WEBRTC_VIDEO_ENGINE_MAIN_INTERFACE_VIE_RTP_RTCP_H_
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@ -929,7 +929,7 @@ WebRtc_Word32 ViEChannel::GetSendRtcpStatistics(WebRtc_UWord16& fraction_lost,
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WebRtc_UWord32 remoteSSRC = rtp_rtcp_.RemoteSSRC();
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RTCPReportBlock remote_stat;
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if (rtp_rtcp_.RemoteRTCPStat(remoteSSRC, &remote_stat) != 0) {
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WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
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WEBRTC_TRACE(kTraceWarning, kTraceVideo, ViEId(engine_id_, channel_id_),
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"%s: Could not get remote stats", __FUNCTION__);
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return -1;
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}
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@ -941,7 +941,7 @@ WebRtc_Word32 ViEChannel::GetSendRtcpStatistics(WebRtc_UWord16& fraction_lost,
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WebRtc_UWord16 dummy;
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WebRtc_UWord16 rtt = 0;
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if (rtp_rtcp_.RTT(remoteSSRC, &rtt, &dummy, &dummy, &dummy) != 0) {
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WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
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WEBRTC_TRACE(kTraceWarning, kTraceVideo, ViEId(engine_id_, channel_id_),
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"%s: Could not get RTT", __FUNCTION__);
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return -1;
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}
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@ -971,9 +971,8 @@ WebRtc_Word32 ViEChannel::GetReceivedRtcpStatistics(
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WebRtc_UWord16 dummy = 0;
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WebRtc_UWord16 rtt = 0;
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if (rtp_rtcp_.RTT(remoteSSRC, &rtt, &dummy, &dummy, &dummy) != 0) {
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WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
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WEBRTC_TRACE(kTraceStateInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
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"%s: Could not get RTT", __FUNCTION__);
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return -1;
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}
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rtt_ms = rtt;
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return 0;
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@ -139,14 +139,14 @@ class ViEChannel
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const WebRtc_UWord8* data,
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WebRtc_UWord16 data_length_in_bytes);
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// Gets statistics sent in RTCP packets to remote side.
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// Returns statistics reported by the remote client in an RTCP packet.
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WebRtc_Word32 GetSendRtcpStatistics(WebRtc_UWord16& fraction_lost,
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WebRtc_UWord32& cumulative_lost,
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WebRtc_UWord32& extended_max,
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WebRtc_UWord32& jitter_samples,
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WebRtc_Word32& rtt_ms);
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// Gets statistics received in RTCP packets from remote side.
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// Returns our localy created statistics of the received RTP stream.
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WebRtc_Word32 GetReceivedRtcpStatistics(WebRtc_UWord16& fraction_lost,
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WebRtc_UWord32& cumulative_lost,
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WebRtc_UWord32& extended_max,
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