git-svn-id: http://webrtc.googlecode.com/svn/trunk@168 4adac7df-926f-26a2-2b94-8c16560cd09d
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#ifndef RTCWEB_H
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#define RTCWEB_H
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class StateNotifier
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{
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public:
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// Called when the state of the session changes.
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// INIT->SENT_OFFER->RECEIVED_ANSWER->INPROGRESS->TERMINATED
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virtual void onStateChange(int newState, char * stateInfo)=0;
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};
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class Session
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{
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public:
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static Session * create(char* id, StateNotifier & obj);
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// generates a session description
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virtual int generateLocalDescription(char * desc, int maxLen) = 0;
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// configures the local media options
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virtual int setLocalDescription(char * desc, int maxLenDesc, char * type, int maxLenType) = 0;
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// configures the remote media options
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virtual int setRemoteDescription(char * desc, int maxLenDesc, char * type, int maxLenType) = 0;
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// Starts or stops sending/receiving media.
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virtual int enable(bool enable) = 0;
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// Mutes or unmutes the sending of media.
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virtual int mute(char * media, int maxLen, bool mute) = 0;
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// Sends a DTMF tone (for use telephony situations)
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virtual int sendDTMF(int event) = 0;
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// Adds an additional stream to the session (for multi-user)
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virtual int addStream(char * media, int maxLen, int source) = 0;
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// Removes a stream from the session.
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virtual int removeStream(char * media, int maxLen, int source) = 0;
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// Gets a URL for a given stream that can be used by
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// <video> or another playout destination. The default
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// stream can be obtained by passing “0”.
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virtual int getStreamURL(char * media, int maxLen, int source) = 0;
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};
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#endif // RTCWEB_H
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