generic-library/webrtc
1
0
Fork 0
Code Issues Pull Requests Releases Wiki Activity

Removed file duplicate added in r1312.

Browse Source 
Review URL: https://webrtc-codereview.appspot.com/357003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1495 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
mflodman@webrtc.org 2012-01-20 12:30:42 +00:00
parent 6a8147519c
commit c672d34ac7
1 changed files with 0 additions and 299 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

299
src/video_engine/main/interface/vie_rtp_rtcp.h
View File

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