Re: [music-dsp] [OT] vinyl? No, thanks...

2010-11-29 Thread Ralph Glasgal
As a physicist and electrical engineer, I am not one who believes that
analog or vinyl is inherently or mystically better than digital.  In my
experiments, perfecting Ambiophonics and giving demonstrations of
loudspeaker binaural reproduction, I am often able to compare vinyl to
digital recording media using binaural rather than stereo reproduction
methods.  That is, reproducing two channel recordings of differing vintages
and media, using Ambiophonic software to recover and make audible all the
ITD and ILD captured by the original microphones and later console
processing. I also eliminate most pinna angle errors, and in some cases by
using real concert hall IRs to generate signals for surround speakers, I
have a much better chance of hearing all the localization, depth, and
ambience data actually captured and stored on the given media.

Ignoring, considerations of ticks and pops, tape hiss, and sometimes
frequency response, I have been able to judge and compare hundreds of LPs,
CDs, and DVDs just on the basis of how realistic a stage presence they
deliver.  That is, is there clarity, depth, full stage width out to almost
180 degrees (if an orchestra or chorus), and cocktail party effect (so I can
concentrate on just one singer or instrument).  (In the case of vinyl, ticks
and pops are off in left field somewhere and are not frontal as in stereo
reproduction, more like a cough or paper rattling at a live concert so
comparisons to digital are perhaps fairer.)  My remarks do not apply to
recordings of a single vocalist and guitar, etc. since mono localization or
quality is not the issue I am concerned with here.

To make a long story short, in general the older the stereo LP the more
realistic it seems to be, ignoring some frequency range issues.  The reason
seems to be that in the early days, the microphone setups were simpler, just
two or three spaced omnis, coincident figure eights, or cardioids.  Post
processing was minimal with few or no spot mics mixed in.  Today, too many
digital recordings, have a lot of mono soloists or groups and the mic ITD
and ILD is pan potted, spot mic'd, and then mixed to binaural garbage.  They
could not do this in the analog era and I believe it is this lack of such
brutal psychoacoustic manipulation of the ITD and ILD that accounts for much
of the preference for older vinyl exhibited by audiophiles.  Since I use an
ELP laser turntable to do these demos and its tick eliminator output is
digital, the differences in psychoacoustic realism between different
recordings or media cannot be due to analog versus digital.

Of course I also have hundreds of CDs/DVDs that have preserved localization
cues and have not been processed to death.  You can hear some great samples
by downloading them from the Ambiophonic website.  There is no scientific
reason why digital cannot always outperform analog.

Ralph Glasgal
www.ambiophonics.org



-Original Message-
From: music-dsp-boun...@music.columbia.edu
[mailto:music-dsp-boun...@music.columbia.edu] On Behalf Of Ross Bencina
Sent: Sunday, November 28, 2010 10:53 PM
To: A discussion list for music-related DSP
Subject: Re: [music-dsp] [OT] vinyl? No, thanks...

Andrew Reilly wrote:
 On Sun, Nov 28, 2010 at 05:56:17PM +0100, Rainer Buchty wrote:
 On Mon, 22 Nov 2010, Stephen Sinclair wrote:

 (Vinyl just sounds.. different.. better.. but I couldn't tell you
 why.)

 Jumping on this (being a long-time lurker on this list), I never
 believed the above statement until I bought some LPs which I also had on
 CD. Until I had my own kind of revelation playing the old Art of Noise
 LPs and CDs in comparison...

 My own CD-vs-LP revelation came a few years ago when I bought
 some sufficiently high-grade analog/digital IO gear, and had a
 go at digitising some of my favourite LPs.  I noticed two things
 immediately:

 1. replaying the PCM sounded *exactly* like the LP, and

 2. the mean recorded level (in PCM) was *significantly* lower
   than the normal signal level of pre-recorded CDs.

That's a great test :-)

 I could get the signal level back up towards CD-level by using
 compression of various sorts, but in doing so the result wound
 up sounding like the CD version, rather than the LP version.

 The obvious conclusion is that the LP mastering process has
 to use a different paradigm than that for CDs, since the
 limitations of excursion and dynamics are different.

Agreed.

I have friends who press new LPs and dub plates pretty regularly -- although

this is indie and dance music, I imagine similar same processes would apply 
to audiophile material:

When the masters are cut, the signal is compressed/tweaked to squeeze it in 
to the available dynamics of the medium and the cutting lathe -- this is 
done at the lathe, often under direction of the producer to get a decent 
dynamics/compression trade off. This is quite different from producing a 
digital master in a mastering studio and sending it off to the CD plant for 

Re: [music-dsp] Algorithms for finding seamless loops in audio

2010-11-29 Thread Element Green
On Thu, Nov 25, 2010 at 9:33 PM, robert bristow-johnson
r...@audioimagination.com wrote:

 depending on how big your window is, i think a better term for this is
 *cross-correlation* not autocorrelation.  it's a single stream of audio so
 in a sense of the word, it *is* autocorrelation, but what i normally think
 of, with that semantic is something where the lag is no bigger or not much
 bigger than the analysis window of either loop-end region of the audio and
 the loop-begin.

 if the loop points are separated by a much longer time (number of samples)
 than the size (in samples) of the two slices of audio being correlated, it's
 really cross-correlation.  and you might find poor correlation given all
 lags that you're looking at.  in fact, doing cross-correlation from one part
 of the tone or sound to another part that has a rapid change in amplitude
 envelope might fool your correlation into thinking there is a good match
 when there really isn't (because the amplitude is increasing, then the
 cross-correlation increases, but not necessarily because of a good match).

 so, instead of either cross or autocorrelation, you might want to consider
 AMDF between the loop end and potential candidates to loop back to.  instead
 of looking for a maximum, you're looking for a minimum and a very low
 minimum means a good match (or a bad match during a very low signal level).

Looking at the equation here for AMDF:
http://mi.eng.cam.ac.uk/~ajr/SpeechAnalysis/node72.html

It seems like the algorithm I came up with independently is very
similar.  The absolute value of the difference of the sample points is
taken as with AMDF.  Prior to summing the values together though, I'm
multiplying by the window I described before (with a peak in the
center where the loop point is), giving samples closer to the loop
point more weight.

In practice this seems to work quite well and I'm going to leave it as
is for now.  It seems reasonably fast and straight forward.


 find good loop points, then crossfade.

 another thing about cross fading is that there is something you can do to
 adapt a little to better or poor loop points.  if the loop points (and the
 window surrounding them) match well, then you're doing a crossfade between
 coherent audio and a constant voltage crossfade is indicated (when the
 crossfade is half done, both the fade out and fade in envelopes are at 50%).
  if the loop points are not well matched (but it's the best loop points your
 correlation function can find), then you want to do a crossfade that is
 closer to a constant power crossfade where both fade in and fade out
 envelopes are at 70.7% at the midpoint of the crossfade.  there is a way to
 define the optimal crossfade function for any correlation between 0 (when
 it's like crossfading white noise to white noise) to 100% (like crossfading
 a perfectly periodic waveform to a similarly appearing portion of the
 waveform at loop start).

 does any of this make any sense?


I'm not sure I'm following you.  From what I can understand it sounds
like you are saying that the degree to which the two loop point signal
windows match could be used to select different cross fade envelope
curves, for a better perceptual cross fade.  I hadn't given this much
thought and just assumed a linear cross fade (0-100%) would be the way
to do it (that is from a limited DSP background mind you).  I am
intrigued by this idea though.  Any tips on how to generate the
envelope functions and what sort of equation could be used for
selecting the optimal envelope based on the signal correlation?

 can i ask what the application is? (i may have missed it, but i'll look at
 earlier posts.)  if it's looping for sound/instrument samples, this is an
 analysis thing that is not real-time and we can consider finding the best
 loop-begin points for a large variety of possible loop-end points.  then
 pick the pair that looks  best, given whatever your measure of good is.  but
 in a (time-domain) real-time pitch shifter, having so many choices may not
 be available to you.  you might find yourself in a situation where your
 loop-end is pretty well defined, you have to find a place to splice to and
 take the best that you can get from that.


Its a sample/instrument editor, so its all non-realtime.

 --

 r b-j                  ...@audioimagination.com

 Imagination is more important than knowledge.


Thanks for the helpful info!
Element Green
--
dupswapdrop -- the music-dsp mailing list and website:
subscription info, FAQ, source code archive, list archive, book reviews, dsp 
links
http://music.columbia.edu/cmc/music-dsp
http://music.columbia.edu/mailman/listinfo/music-dsp