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A
Anonymous
July 24, 2010
Question

Audio Delay FMS 3.5

  • July 24, 2010
  • 1 reply
  • 1308 views

I have an application that uses FMS 3.5 to stream live audio between 2 computers.  Currently, the delay between when the person speaks and the time the other person hears it is about 2 - 3 seconds.  This makes it difficult to have a conversation because one person regularly steps on the other.

I am using a NetStream object that connects to the fms server.  I then attach audio.  What configurations can be set so that the least amount of delay is experienced between the two computers?  Can fms ever relay audio as quick as something like  Google Talk?

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    1 reply

    A
    Anonymous
    July 26, 2010

    While latency across networks is always a challenge, you should be able to achieve much better results assuming network conditions allow.

    What codec and sample rate (microphone.rate) are you using, and what do you have the bufferLength property set to on the netstreams?

    A
    Anonymous
    July 26, 2010

    I'm not sure about the codec or how to set that.  To experiment and try to get the best performance, I've set the microphone rate to 5 and the setSilenceLevel(0, 20000).  I have also set the buffertime to 0.

    A
    Anonymous
    July 26, 2010

    You'll definitely want to set the mic rate higher. Some devices can't decode 5kHz or 8kHz audio with a zero buffer (I recall Flashplayer for Mac having a problem with that), and my understanding is that the server will actually send the data faster at higher rates.

    If you haven't changed the audio codec, then you're using the default (Nellymoser Asao). I'm fairly certain that Asao is faster than Speex, but perhaps someone else with real test results on that will chime in here.

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