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Bob Kozi
2010-09-01
2013-05-23
  • Bob Kozi
    Bob Kozi
    2010-09-01

    Hi!

    I've look your software year ago, and it looks nice then, thank you for adding stereo option yet!

    Could you give more quality options and more buffering options, especially in windows version? I mean worse quality and longer buffering, up to few seconds. As start parameters, for example. It will be usable for other application than jamming (such as transmitting audio from live events to radio broadcasting studio and back).

    Good luck, I will keep my fingers crossed for further development of Licon and Celt of course!

     
  • Volker Fischer
    Volker Fischer
    2010-09-09

    The low quality option (i.e. "Use High Quality Audio" is not checked) gives you already a pretty low bit rate. If you would lower the quality then this would result in audible coder artifacts which will not be pleasant to listen to. Keep in mind that you require a higher bit rate in comparison to, e.g., MP3 because the low latency introduces some overhead.

    To make the delay longer you can use a sound card buffer delay of 512 samples (21.33 ms) and set the Jitter Buffer to maximum. If your ASIO Setup allows you to even use longer delays (e.g. 2048 samples), you can use this and get even longer delays.

     
  • Bob Kozi
    Bob Kozi
    2010-09-09

    I know that lower bitrate always couse worse sound, but buffering time is not sufficient for stable transmitting audio via internet pipes in our country - with current buffering range (with 90 msec too). On the other hand, I think that sometimes very stable ISDN pipes could be used instead, but connection stream must be changed to lower than 128 kbps for stereo mode. What do you mean about it?

     
  • Volker Fischer
    Volker Fischer
    2010-09-18

    Well, I don't think that you ever will be able to produce a 100 % stable audio stream with llcon even if the buffers are very big (note that in this case the latency is high, too) since llcon does not do a sample rate adjustment. Because of sample rate differences in the transmitting and receiving sound card, you will always get audio drop outs from time to time.

    I would prefer to use a software which is designed for your particular purpose like, e.g., http://netjack.sourceforge.net.