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290 lines
6.5 KiB
290 lines
6.5 KiB
/*
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* $Id$
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*
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* Copyright (C) 2002-2003 Fhg Fokus
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*
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* This file is part of sems, a free SIP media server.
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*
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* sems is free software; you can redistribute it and/or modify
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* it under the terms of the GNU General Public License as published by
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* the Free Software Foundation; either version 2 of the License, or
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* (at your option) any later version
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*
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* For a license to use the ser software under conditions
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* other than those described here, or to purchase support for this
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* software, please contact iptel.org by e-mail at the following addresses:
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* info@iptel.org
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*
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* sems is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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* GNU General Public License for more details.
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*
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* You should have received a copy of the GNU General Public License
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* along with this program; if not, write to the Free Software
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* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
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*/
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/** @file AmRtpStream.h */
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#ifndef _RtpStream_h_
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#define _RtpStream_h_
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#include "AmSdp.h"
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#include "AmThread.h"
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#include "SampleArray.h"
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#include "AmRtpPacket.h"
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#include "AmEvent.h"
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#include <netinet/in.h>
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#include <string>
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#include <map>
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#include <memory>
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using std::string;
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using std::auto_ptr;
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// return values of AmRtpStream::receive
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#define RTP_EMPTY 0 // no rtp packet available
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#define RTP_PARSE_ERROR -1 // error while parsing rtp packet
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#define RTP_TIMEOUT -2 // last received packet is too old
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#define RTP_DTMF -3 // dtmf packet has been received
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#define RTP_BUFFER_SIZE -4 // buffer overrun
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#define RTP_UNKNOWN_PL -5 // unknown payload
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struct amci_payload_t;
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class AmAudio;
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class AmSession;
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class SdpPayload;
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typedef std::map<unsigned int, AmRtpPacket*, ts_less> ReceiveBuffer;
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/**
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* This provides the memory for the receive buffer.
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*/
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struct PacketMem {
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#define MAX_PACKETS 32
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AmRtpPacket packets[MAX_PACKETS];
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bool used[MAX_PACKETS];
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PacketMem();
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inline AmRtpPacket* newPacket();
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inline void freePacket(AmRtpPacket* p);
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inline void clear();
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};
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/**
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* \brief RTP implementation
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*
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* Rtp stream high level interface.
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*/
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class AmRtpStream
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{
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protected:
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static int next_port;
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static AmMutex port_mut;
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static int getNextPort();
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/**
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Remote payload (only different from
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int_payload if using dynamic payloads)
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*/
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int payload;
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unsigned int sequence;
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/**
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Payload of last received packet.
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Usefull to detect talk spurt, looking
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for comfort noise packets.
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*/
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int last_payload;
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string r_host;
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unsigned short r_port;
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#ifdef SUPPORT_IPV6
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struct sockaddr_storage r_saddr;
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struct sockaddr_storage l_saddr;
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#else
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struct sockaddr_in r_saddr;
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struct sockaddr_in l_saddr;
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#endif
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unsigned short l_port;
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int l_sd;
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struct timeval last_recv_time;
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unsigned int l_ssrc;
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unsigned int r_ssrc;
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bool r_ssrc_i;
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/** symmetric RTP */
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bool passive; // passive mode ?
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/** Payload type for telephone event */
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auto_ptr<const SdpPayload> telephone_event_pt;
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PacketMem mem;
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ReceiveBuffer receive_buf;
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AmMutex receive_mut;
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/* get next packet in buffer */
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int nextPacket(AmRtpPacket*& p);
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AmSession* session;
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/** Initializes a new random local port, and sets own attributes properly. */
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void setLocalPort();
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public:
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AmRtpPacket* newPacket();
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void freePacket(AmRtpPacket* p);
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/** Mute */
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bool mute;
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/** mute && port == 0 */
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bool hold;
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/** marker flag */
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bool begin_talk;
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/** do check rtp timeout */
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bool monitor_rtp_timeout;
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/** should we receive packets? if not -> drop */
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bool receiving;
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int send( unsigned int ts,
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unsigned char* buffer,
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unsigned int size );
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int receive( unsigned char* buffer, unsigned int size,
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unsigned int& ts, int& payload);
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/** ping the remote side, to open NATs and enable symmetric RTP */
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int ping();
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/** Allocates resources for future use of RTP. */
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AmRtpStream(AmSession* _s=0);
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/** Stops the stream and frees all resources. */
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virtual ~AmRtpStream();
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/**
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* This function must be called before setLocalPort, because
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* setLocalPort will bind the socket and it will be not
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* possible to change the IP later
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*/
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void setLocalIP(const string& ip);
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/**
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* Gets RTP port number. If no RTP port in assigned, assigns a new one.
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* @return local RTP port.
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*/
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int getLocalPort();
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/**
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* Gets remote RTP port.
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* @return remote RTP port.
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*/
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int getRPort();
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/**
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* Gets remote host IP.
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* @return remote host IP.
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*/
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string getRHost();
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/**
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* Set remote IP & port.
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*/
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void setRAddr(const string& addr, unsigned short port);
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/** Symmetric RTP: passive mode ? */
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void setPassiveMode(bool p) { passive = p; }
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bool getPassiveMode() { return passive; }
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/**
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* Set remote telephone event
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* payload type
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*/
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void setTelephoneEventPT(const SdpPayload *pt) {
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telephone_event_pt.reset(pt);
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}
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int getTelephoneEventRate();
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/**
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* Enables RTP stream.
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* @param sdp_payload payload from the SDP message.
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* @warning start() must have been called so that play and record work.
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* @warning It should be called only if the stream has been completly initialized,
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* @warning and only once per session. Use resume() then.
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*/
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virtual void init(const vector<SdpPayload*>& sdp_payloads);
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/**
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* Stops RTP stream.
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*/
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void pause();
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/**
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* Resume a paused RTP stream.
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*/
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void resume();
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/** set the RTP stream on hold (mute && port = 0) */
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void setOnHold(bool on_hold);
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/** get the RTP stream on hold */
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bool getOnHold();
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/** setter for monitor_rtp_timeout */
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void setMonitorRTPTimeout(bool m) { monitor_rtp_timeout = m; }
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/** getter for monitor_rtp_timeout */
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bool getMonitorRTPTimeout() { return monitor_rtp_timeout; }
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/**
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* Report an ICMP error.
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*/
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void icmpError();
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/**
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* Insert an RTP packet to the buffer.
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* Note: memory is owned by this instance.
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*/
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void bufferPacket(AmRtpPacket* p);
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/*
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* clear RTP timeout at time recv_time
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*/
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void clearRTPTimeout(struct timeval* recv_time);
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virtual unsigned int bytes2samples(unsigned int) const = 0;
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};
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/** \brief event fired on RTP timeout */
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class AmRtpTimeoutEvent
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: public AmEvent
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{
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public:
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AmRtpTimeoutEvent()
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: AmEvent(0) { }
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~AmRtpTimeoutEvent() { }
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};
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#endif
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// Local Variables:
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// mode:C++
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// End:
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