[Ffmpeg-devel] alsa input / output

[Michael Niedermayer]

Hi

On Wed, Mar 21, 2007 at 04:19:12AM +0300, Vladimir Mosgalin wrote:
> Hi Michael Niedermayer!
> 
>  On 2007.03.21 at 00:57:20 +0100, Michael Niedermayer wrote next:
> 
> > so the low quality resampler from libsamplerate sounds better to you then
> 
> What do you mean, "low quality"? Does it sound THAT low to you? It
> sounds perfectly fine to me.

you said "(believe it or not, but SRC_SINC_FASTEST sounds better than
SRC_SINC_BEST_QUALITY and all other SRC_SINC libsamplerate filters to
me)"
which of course you cliped away from the quote


[...]
> > the high quality ones from libsamplerate, while i dont have much faith
> > in libsamplerate i cant belive that their fast low quality code sounds
> > better then their hq code, maybe your audio hardware is broken and the
> > additional low pass filtering done by low quality resamplers mask that
> > your furher comment below points strongly in that direction, only very
> > odd hardware is limited to 24bit/sample
> 

[audiophile trash]

> > besides this it would be interresting to run proper double blind tests
> > between libsamplerate and the lavc resampler, if libsamplerate beats us
> 
> Well, as a biased way of doing this, with special fragments of sound
> that expose some of the resamplers' problems the difference should be
> clear for everyone. (it's not that these fragments are artificial, it's
> just that human's ear is really sensitive to distortion in their sound).
> They won't show you very subtle difference between resamplers on real
> content, but you will be able to hear which resamplers produce more
> distortion. Google for Chenoa_16KHz.wav, 

2 links from thailand no files


> Androgyny_16KHz.wav, 

no single hit


> udial.wav,

~500 hits first is from someone who too searches for the file and then
happily finds .ape of it

sorry but you will have to upload these files to upload.mplayerhq.hu if
you want us to look at them, also dont forget to tell us after you uploaded
them ...



[...]
> > it would be trivial to use their filter, after all samplerate conversation
> > is just a trivial linear equation with coefficients choosen to please
> 

[more audiophile trash]


> 
> If you have soundcard that supports 44100hz output, I can send you a
> patch which adds dithered (24 bit to 16) output to mplayer's mad decoder
> (I guess no explanation needed why other decoders are much worse). Of

ffmp3 has dithered output by default you just have to belive it for
the placebo effect to work that is unless you use artificial tests with
the volume turned up and the content being very silent


[...]
> > the latest audophile trend
> 
> Very funny. You really believe that you have to choose between
> "pleasant-sounding" and "mathematically correct" when it comes to this
> kind of digital processing?

there is no matematical correct way known, real world samples are not
bandlimited, sure bandlimited theory works fine normally but considering
that you want samples to have errors smaller then 1/2^16 i can assure you 
that at that level the bandlimited approximation does no longer hold

[...]
-- 
Michael     GnuPG fingerprint: 9FF2128B147EF6730BADF133611EC787040B0FAB

it is not once nor twice but times without number that the same ideas make
their appearance in the world. -- Aristotle
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