Re: [LAU] jamin (was: Albums under a label recorded and/or mixed with Linux)

On Thu, Sep 30, 2010 at 05:40:46PM +0200, torbenh wrote:

> wouldnt a linkwitz-riley fit best here ?

It would, but it would probably be overkill. We are not driving
a set of speakers here, but three or four compressors, and there
are good reasons to make the crossover quite gentle. That is my
gripe with the FFT-based xover: it is far too steep. It could
result in one note of a melody being processed by on compressor
and the next, just a few semitones higher or lower, by another.
This is probably not what you want, certainly not if those
compressors are behaving quite differently which is the whole
point of multi-band processing.

I'd go for a filter type that is close to first order in the
xover region and becomes more steep an octave or so higher or
lower. This would produce some minimal (< 1 dB) ripple in the
FR, but who cares if you are going to multiband compress anyway ?
 
> i am still a bit confused.
> i always thought, that a mastering processor should be linear phase.
> so what came to my mind first, when i saw the comments about the
> problems in the lower bands, was to just increase the number of FIR
> taps. we would lower cpu usage also.

CPU usage would increase actually, if the filter is implemented
really as a filter. For the linear phase thing, see below.

> fons... do you think a 16384 tap fir could do a decent crossover and EQ ?

16k taps (at 48 kHz) will allow you to make most wanted types of
audio filter (except steep highpass at low F), and zillions of
very bad ones... I'd never advocate the use of such filters except
if they are derived e.g. from an inversion procedure, as for room
correction.

> (assuming this linear phase mumbo jumbo actually makes sense)

Some of it. But it's being taken way over the top. There are (in
audio) almost no good reasons to make a filter linear-phase, and
many filters will sound very bad in their linear-phase form while
being perfectly acceptable in their minimum-phase form.

It's often claimed but not true that a linear-phase filter will
'ring' less. It will ring just as much as any other filter with
the same amplitude response, and what is worse, half of the ringing
will precede the main peak of the IR, which makes it much more
audible.

Let me give an simple example. You have a FIR of 4801 samples,
all of them are zero except:

Linear-phase form: F[0] = 0.02, F[2400] = 1, F[4800] = 0.02

Min-phase form: F[0] = 0, F[2400] = 1, F[4800] = 0.04

Max-phase form: F[0] = 0.04, F[2400] = 1, F[4800] = 0

All three have an amplitude response that oscillates around 0dB
by about 0.35 dB. It is *very unlikely* that anyone would actually
hear the effect of the minimum-phase form. The two others will
have a 50 ms pre-echo of resp. -34 and -28 dB, which is easy to
hear on some types of signal (anything percussive).

Our hearing is very sensitive to anything preceding the onset
or main peak of a sound: this is almost never a property of the
original sound source but it indicates very often the presence
of more than one of them. Which is essential info for any life
form trying to avoid its predators, so this sensitivity may
have its origins in evolution.

Also all percussive sounds are 'minimum-phase' - they sound as
'ping' and not as 'gnip' or 'gnipping'. In cases where you need
a long filter (that is for a FR that has much fine detail), the
minimum-phase form will affect the envelope of a sound the least,
and linear-phase is to be avoided. For 'smooth' filters it's OK,
but I doubt very much if it actually improves the sound at all.

Ciao,

-- 
FA
There are three of them, and Alleline.
_______________________________________________
Linux-audio-user mailing list
Linux-audio-user@email-addr-hidden
http://lists.linuxaudio.org/listinfo/linux-audio-user