David wrote:
> Yes, so what is the cleanest : malloc'ing an additionnal frame and not
> filling it or loosing a frame of data that should have come from the
> SRC ?
It realy doesn't make much difference. Feel free to do as you
like :-).
Erik
--
+-
Announcing the 20060122 release of WhySynth, a DSSI softsynth
plugin.
New since the last major release:
* A new oscillator mode, based on Nasca O. Paul's gorgeous
PADsynth algorithm.
* A new filter mode, essentially the low-pass filter from amSynth.
* A new dual delay effect.
* Improved and e
On Tue, 24 Jan 2006 10:56:20 +1100
Erik de Castro Lopo <[EMAIL PROTECTED]> wrote:
> > Is it safe to assume that using floor() instead of ceil() will not
> > lead to a too short output buffer in some cases ?
>
> The most you should ever loose is one sample.
Yes, so what is the cleanest : malloc'i
David wrote:
> I process my input data in one pass using src_simple() and I have to
> compute the length of the output data buffer beforehand. So I did
> somehting like this :
>
> out_len = (long int) ceil((double) in_len * ratio);
>
> It seems that my output buffer is always one frame too big (
fons adriaensen wrote:
> Would it ? It solves the problem, and all other apps will - or
> at least _should_ according to the WAV spec - just ignore it.
> What problems would be created by adding a new chunk ?
> The alternative would be a format that isn't standard at all.
The problems are as foll
On Tue, 2006-01-24 at 00:08 +0100, fons adriaensen wrote:
> On Tue, Jan 24, 2006 at 09:02:23AM +1100, Erik de Castro Lopo wrote:
> > fons adriaensen wrote:
> >
> > > What I need in particular is some way to calibrate the time
> > > axis - i.e. to say frame #N corresponds to t = 0, and some
> > >
On Tue, Jan 24, 2006 at 09:02:23AM +1100, Erik de Castro Lopo wrote:
> fons adriaensen wrote:
>
> > What I need in particular is some way to calibrate the time
> > axis - i.e. to say frame #N corresponds to t = 0, and some
> > other similar info, mostly sample indices.
>
> There is no existing c
Hello everybody !
I've just added a resampling function to my code thanks to the
excellent work of Erik de Castro Lopo (thanks a lot !). Combined with
libsndfile (thanks again) it is really easy to load any sound file I
want. But I'd like to make sure I'm using it correctly.
I process my input da
fons adriaensen wrote:
> Hi all,
>
> is there a recommended way to write / read additional chunks in
> WAV files, using libsndfile (assuming it's possible at all - I
> didn't find any hints to this in the docs) ?
libsndfile already supports the addition of a number of specific
chunk types like P
