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Ant Media Server supports RTMP, RTSP, MP4, HLS, WebRTC, Adaptive Streaming, etc.
package io.antmedia.webrtc.api;
import java.nio.ByteBuffer;
import org.webrtc.IceCandidate;
import org.webrtc.SessionDescription;
public interface IWebRTCClient {
public void setWebRTCAdaptor(IWebRTCAdaptor webRTCAdaptor);
/**
* Send video packet to connected client
* @param videoPacket
* @param isKeyFrame
*/
public void sendVideoPacket(ByteBuffer videoPacket, boolean isKeyFrame, long timestamp, int frameRotation);
/**
* Send audio packet to connected client
* @param audioPacket
*/
public void sendAudioPacket(ByteBuffer audioPacket, long timestamp);
public int getTargetBitrate();
public void start();
public void setRemoteDescription(SessionDescription sdp);
public void addIceCandidate(IceCandidate iceCandidate);
public void setVideoResolution(int width, int height);
public void setWebRTCMuxer(IWebRTCMuxer webRTCMuxer);
public IWebRTCMuxer getWebRTCMuxer();
public void stop();
/**
* Returns the time in milliseconds between the time when start function is called
* and the time when streaming is started
* @return the time in milliseconds
* or -1 if timing is not available yet
*/
long getTimeToStartStreaming();
/**
* Returns the time in milliseconds between the time when stop function is called
* and the time when streaming is fully stopped
* @return the time in milliseconds
* or -1 if timing is not available yet
*/
long getTimeToStop();
/**
* Return the period of send video period in milliseconds
* @return
*/
float getVideoFrameSentPeriod();
/**
* Return the period of send audio period in milliseconds
* @return
*/
float getAudioFrameSentPeriod();
/**
* Return the period of entering audio thread interval in milliseconds
* @return
*/
float getAudioThreadCheckInterval();
/**
* Return the priod of entering video thread interval in milliseconds
* @return
*/
float getVideoThreadCheckInterval();
}
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