Hi all,
I come up with the following piece of code that streams the microphone signal.
But for some reason each audio RTMP has 80 ms of sound. Does anyone if it's
possible to reduce this time to 20 ms, and where I could find out more
information about how to do it?
Thanks,
Andre
public function handleConnectionStatus(event:NetStatusEvent):void
{
if(event.info.code == "NetConnection.Connect.Success")
{
mic = Microphone.getMicrophone();
if (mic != null)
{
mic.setSilenceLevel(20);
mic.setUseEchoSuppression(true);
mic.codec = SoundCodec.SPEEX;
mic.encodeQuality = 8;
stream = new NetStream(netConnection);
stream.attachAudio(mic);
stream.publish("testing");
}
}
}
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