Changes between Version 25 and Version 26 of StreamingGuide


Ignore:
Timestamp:
Sep 14, 2012, 11:06:55 PM (4 years ago)
Author:
rogerdpack
Comment:

audio latency

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  • StreamingGuide

    v25 v26  
    5252 
    5353You may be able to decrease latency by specifing that I-frames come "more frequently" (or basically always, in the case of [[x264EncodingGuide|x264]]'s zerolatency setting), though this can increase frame size and decrease quality, see [http://mewiki.project357.com/wiki/X264_Encoding_Suggestions here] for some more background.  Basically for typical x264 streams, it inserts an I-frame every 250 frames.  This means that new clients that connect to the stream may have to wait up to 250 frames before they can start receiving the stream (or start with old data).  So increasing I-frame frequency (makes the stream larger, but might decrease latency).  For real time captures you can also decrease latency of audio in windows dshow by using the dshow audio_buffer_size [http://ffmpeg.org/ffmpeg.html#Options setting].  You can also decrease latency by tuning any broadcast server you are using to minimize latency, and finally by tuning the client that receives the stream to not "cache" any incoming data, which, if it does, increases latency. 
     54 
     55Sometimes audio codecs also introduce some latency of their own.  You may be able to get less latency by using speex, for example, or opus, in place of libmp3lame. 
    5456 
    5557== Cpu usage/File size ==