[linux-audio-dev] linear resampling is crap ? (was: fast linear resampling on ARM - suggestions?)

[torbenh at gmx.de]

On Thu, Mar 30, 2006 at 04:34:43PM +1000, Erik de Castro Lopo wrote:
> torbenh at gmx.de wrote:
> 
> > On Wed, Mar 29, 2006 at 11:47:30PM +1000, Erik de Castro Lopo wrote:
> > > Tobias Scharnberg wrote:
> > > 
> > > However, please do not use linear resampling; its just too crappy.
> > 
> > urgh... i just switched my local copies of alsa_in and _out to linear
> > resampling, because SRC_SINC_FASTEST was eating CPU juice.
> 
> Just out of curiosity, what kind of hardware are you running this
> on and what sort of performance would you find acceptable for your
> application. An estimate that provides:

ok... the program is a jack application, that opens an alsa pcm device
(obviously not the one, jackd is bound to) and either outputs or inputs 
the sound.

> 
>     - input and output sample rates

input is usually 48000 or 44100 (the jackd samplerate)
outputrate depends on the soundcard used. 48k or 44k1 +- 10 Hz varying
with clock drift... and calculation of resampling factor in float.

i will improve the detection algorithm soon again.
so that the samplerate is changing slower...

>     - samples per second per channel throughput wrt the output rate

realtime.

when the jack source clock is not very bursty (ie LAN or local)
2 channels take 7% cpu on an athlon xp 2000


with a bursty clock source from the internet the cpu usage in top is
much higher. like 20% or so.

a little too much for a soundcard IO :)


> 
> would give me the best idea of what you are after.

i hope you get what i am after now. i am heavily exercising libsrc...

> 
> Erik
> -- 
> +-----------------------------------------------------------+
>   Erik de Castro Lopo
> +-----------------------------------------------------------+
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-- 
torben Hohn
http://galan.sourceforge.net -- The graphical Audio language