Hi,
The most obvious solution would be to use an audio buffer of 160 x N
instead of 512 bytes. About your soundcard you should disable that
threshold feature (of which I've never heard).
What kind of a soundcard do you have?
//Best Regards Jonas Tärnström
<p>> -----Ursprungligt meddelande-----> Från: owner-speex-dev@xiph.org [mailto:owner-speex-dev@xiph.org] För
Eric> Ross
> Skickat: den 13 september 2003 02:40
> Till: speex-dev@xiph.org
> Ämne: [speex-dev] framesize v/s audio buffer size
>
> Hello all..
>
> Im new to speex, and im trying to use it in my application to transmit
> speech over multicast.
>
> I've done some test, but i have some questions. First, I'm reading
audio> with a buffer of 512 bytes, but the speex frames are 160 (using
> SPEEX_GET_FRAME_SIZE), so i have to manually divide the 512 bytes (256
> 16bit samples) in blocks of 160 float samples. Using 1 frame, i
dispatch> the first 160 samples, but there are 96 (256-160) samples remaining.
What> can i do with them ? I need to send them as soon as posible, so i
guess i> cant just wait until the next audio buffer comes to push the data
inside> the codec. I dont even know if the sound card will produce more audio
> because before getting here, the audio have to pass some noise level.
If> the audio is not louder enought i will not get any data.
>
> Any ideas ?
> Thanks in advance.
>
>
> --
> The names of their development products, visual this, visual that,
almost> makes one think that you can create software just by looking at your
> computer
>
>
>
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