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

[torbenh at gmx.de]

On Sat, Apr 01, 2006 at 10:49:20AM +0100, Simon Jenkins wrote:
> On Thu, 2006-03-30 at 22:33 +0200, torbenh at gmx.de wrote:
> 
> > i need all features of libsrc (slowly changing samplerate, and resample
> > factors of 1.00001 or so)
> 
> Lets assume - unrealistically, but for the convenience of the
> maths - a sample rate of 50kHz and a buffer size of 1000. Your
> resample factor of 1.00001 or so means you need to "find" or
> "lose" a sample about once every 2 seconds, and 99% of your
> buffers will be precisely the same size going out as they are
> coming in.
> 
> Is putting the entire stream through a resampler (especially
> a "crap" one) the thing to do here? I'm wondering if there
> might be a better way to gain or lose the occasional sample.

well... there are some soundcards which can only do one samplerate
(48k or 44k1 namely) so for this case its real resampling...

i have just found that there was a timeout value in the select()
in net_driver_wait() resulting in kernel busy waiting... 
this changed the cpu usage in some unfavourable manner... i am off for
some other tests... and going back to SINC_FASTEST now...

also for the internet case i am getting quite bursty clocking an
currently dont have a way to get timestamps etc from the jack_driver to
the client. so while experimenting with this stuff, i am experiencing 
resampling of 5% or so...

i am currently experimenting with the coefficients and function for the
delay locked loop controlling the resample factor...


> 
> erm... Anyone?
> 
> Simon Jenkins
> Bristol, UK
> 
> 

-- 
torben Hohn
http://galan.sourceforge.net -- The graphical Audio language