Hi,
Thanks to all for your reply to the previous question.
Another question is, is it possible to make CELT scalable? For example, if
I ran the encoder to encode a 320kbps bitstream, is it possible to split
the 320kbps into a valid 160kbps bitstream and the residue? This way, I can
send the 160kbps packet first, and then if the network condition allows,
the residue will be send next.
Thanks
Liz
On Wed, Aug 22, 2012 at 5:14 PM, Benjamin Schwartz <
benjamin.m.schwartz at gmail.com> wrote:
> On Wed, Aug 22, 2012 at 5:09 PM, Gregory Maxwell <gmaxwell at
gmail.com>
> wrote:
> > Flac can have latency as low as you like? e.g. you can code 64 sample
> > frames. You do lose compression efficiency in doing this, however.
>
> For more info on how to do this see encoder_set_blocksize:
> http://flac.sourceforge.net/api/group__flac__stream__encoder.html#ga21
>
> However, for low-latency lossless audio, you may actually want use
> uncompressed PCM (like WAV). With a lossless compressor like FLAC,
> there is no guarantee that the audio will be made smaller (it can even
> become larger), so you will need enough bandwidth to send uncompressed
> audio anyway. if you rely on the compression to work, then your
> system will break on the occasional incompressible block. For
> low-latency lossless audio, compression only makes sense if you have a
> lot of available peak bandwidth, but for some reason (e.g. ISP quota)
> are trying not to use it.
>
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