Re: [music-dsp] A theory of optimal splicing of audio in the time domain.

2011-07-14 Thread robert bristow-johnson


On Jul 13, 2011, at 9:29 AM, Olli Niemitalo wrote:


On Sat, Jul 9, 2011 at 10:53 PM, robert bristow-johnson
r...@audioimagination.com wrote:

On Dec 7, 2010, at 5:27 AM, Olli Niemitalo wrote:


[I] chose that the ratio a(t)/a(-t) [...] should be preserved


by preserved, do you mean constant over all t?


Constant over all r.



i think i figgered that out after hitting the Send button.


what is the fundamental reason for preserving a(t)/a(-t) ?


I'm thinking outside your application of automatic finding of splice
points. Think of crossfades between clips in a multi-track sample
editor. For a cross-fade in which one signal is faded in using a
volume envelope that is a time-reverse of the volume envelope using
which the other signal is faded out, a(t)/a(-t) describes by what
proportions the two signals are mixed at each t. The fundamental
reason then is that I think it is a rather good description of the
shape of the fade, to a user, as it will describe how the second
signal swallows the first by time.


okay, i get it.

so instead of expressing the crossfade envelope as

a(t)  =   e(t)   +   o(t)

i think we could describe it as a constant-voltage crossfade (those  
used for splicing perfectly correlated snippets) bumped up a little by  
an overall loudness function.  an envelope acting on the envelope.   
and, as you correctly observed, for constant-voltage crossfades, the  
even component is always


e(t)  =   1/2

so, pulling another couple of letters outa the alfabet, we can  
represent the crossfade function as


a(t)  =  e(t)  +  o(t)  =  g(t)*( 1/2 + p(t) )

where

g(-t)  =   g(t)  is even
and
p(-t)  =  -p(t)  is odd


g(t) = 1 for constant-voltage crossfades, when r=1.
for constant-power crossfades, r=0, we know that g(0) = sqrt(2)  1

the shape p(t) is preserved for different values of r and we want to  
solve for g(t) given a specified correlation value r and a given  
shape family p(t).  indeed


   a(t)/a(-t)  =  (1/2 + p(t))/(1/2 - p(t))

and remains preserved over r if p(t) remains unchanged.

p(t) can be spec'd initially exactly like o(t) (linear crossfade,  
Hann, Flattened Hann, or whatever odd function your heart desires).  i  
think it should be easy to solve for g(t).  we know that



  e(t)  =  1/2 * g(t)

  o(t)  =  g(t) * p(t)

and recall the result

  e(t)  =  sqrt( (1/2)/(1+r) - (1-r)/(1+r)*(o(t))^2 )

which comes from

  (1+r)*( e(t) )^2  +  (1-r)*( o(t) )^2  =  1/2

so
  (1+r)*( 1/2*g(t) )^2  +  (1-r)*( g(t)*p(t) )^2  =  1/2


  ( g(t) )^2 * ( (1+r)/4 + (1-r)*(p(t))^2 )  =  1/2

and picking the positive square root for g(t) yields

  g(t)  =  1/sqrt( (1+r)/2 + 2*(1-r)*(p(t))^2 )

might this result match what you have?  (assemble a(t) from g(t) and  
p(t) just as we had previously from e(t) and o(t).)


remember that p(t) is odd so p(0)=0  so when

  r=1  ---   g(t) = 1  (constant-voltage crossfade)
and

  r=0  ---   g(0) = sqrt(2)(constant-power crossfade)



The user might choose one shape
for a particular crossfade. Then, depending on the correlation between
the superimposed signals, an appropriate symmetrical volume envelope
could be applied to the mixed signal to ensure that there is no peak
or dip in the contour of the mixed signal. Because the envelope is
symmetrical, applying it preserves a(t)/a(-t). It can also be
incorporated directly into a(t).

All that is not so far off from the application you describe.

but i don't think it is necessary to deal with lags where Rxx(tau)  
 0.  why
splice a waveform to another part of the same waveform that has  
opposite

polarity?  that would create an even a bigger glitch.


Splicing at quiet regions with negative correlation can give a smaller
glitch than splicing at louder regions with positive correlation.


okay.  i would still like to hunt for a splice displacement around  
that quiet region that would have correlation better than zero.  and,  
if both x(t) and y(t) have no DC, it should be possible to find  
something.



This
applies particularly to rhythmic material like drum loops, where the
time lag between the splice points is constrained, and it may make
most sense to look for quiet spots. However, if it's already so quiet
in there, I don't know how much it matters what you use for a
cross-fade.

Apart from it's so quiet it doesn't matter, I can think of one other
objection against using cross-fades tailored for r  0: For example,
let's imagine that our signal is white noise generated from a Gaussian
distribution, and we are dealing with given splice points for which
Rxx(tau)  0 (slightly).


but you should also be able to find a tau where Rxx(tau) is slightly  
greater than zero because Rxx(tau) should be DC free (if x(t) is DC  
free).  if it were true noise, it should not be far from zero so you  
would likely use the r=0 crossfade function.



Now, while the samples of the signal were
generated independently, there is by accident a bit of 

Re: [music-dsp] A theory of optimal splicing of audio in the time domain.

2011-07-14 Thread Olli Niemitalo
On Thu, Jul 14, 2011 at 9:22 PM, robert bristow-johnson
r...@audioimagination.com wrote:

      g(t)  =  1/sqrt( (1+r)/2 + 2*(1-r)*(p(t))^2 )

 might this result match what you have?

Yes! I only derived the formula for the linear ramp, p(t) = t/2,
because one can get the other shapes by warping time and I didn't want
to bloat the cumbersome equations. With the linear ramp our results
match exactly.

 okay.  i would still like to hunt for a splice displacement around that
 quiet region that would have correlation better than zero

Sometimes you are stuck with a certain displacement. Think drum loops;
changing tau would change tempo.

 i think it's better to define p(t) (with the same restrictions as o(t)) and 
 find g(t) as a
 function of r than it is to do it with o(t) and e(t).

I agree, even though the theory was quite elegant with o(t) and e(t)...

-olli
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Re: [music-dsp] A theory of optimal splicing of audio in the time domain.

2011-07-14 Thread robert bristow-johnson


On Jul 14, 2011, at 5:36 PM, Olli Niemitalo wrote:


On Thu, Jul 14, 2011 at 9:22 PM, robert bristow-johnson
r...@audioimagination.com wrote:


 g(t)  =  1/sqrt( (1+r)/2 + 2*(1-r)*(p(t))^2 )

might this result match what you have?


Yes! I only derived the formula for the linear ramp, p(t) = t/2,
because one can get the other shapes by warping time and I didn't want
to bloat the cumbersome equations. With the linear ramp our results
match exactly.

okay.  i would still like to hunt for a splice displacement  
around that

quiet region that would have correlation better than zero


Sometimes you are stuck with a certain displacement. Think drum loops;
changing tau would change tempo.

i think it's better to define p(t) (with the same restrictions as  
o(t)) and find g(t) as a

function of r than it is to do it with o(t) and e(t).


I agree, even though the theory was quite elegant with o(t) and  
e(t)...




do you have any of this in a document?  i wonder if one of us should  
put this down in a pdf and put it in the music-dsp code archive.



--

r b-j  r...@audioimagination.com

Imagination is more important than knowledge.




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Re: [music-dsp] A theory of optimal splicing of audio in the time domain.

2011-07-13 Thread Olli Niemitalo
On Sat, Jul 9, 2011 at 10:53 PM, robert bristow-johnson
r...@audioimagination.com wrote:
 On Dec 7, 2010, at 5:27 AM, Olli Niemitalo wrote:

  [I] chose that the ratio a(t)/a(-t) [...] should be preserved

 by preserved, do you mean constant over all t?

Constant over all r.

 what is the fundamental reason for preserving a(t)/a(-t) ?

I'm thinking outside your application of automatic finding of splice
points. Think of crossfades between clips in a multi-track sample
editor. For a cross-fade in which one signal is faded in using a
volume envelope that is a time-reverse of the volume envelope using
which the other signal is faded out, a(t)/a(-t) describes by what
proportions the two signals are mixed at each t. The fundamental
reason then is that I think it is a rather good description of the
shape of the fade, to a user, as it will describe how the second
signal swallows the first by time. The user might choose one shape
for a particular crossfade. Then, depending on the correlation between
the superimposed signals, an appropriate symmetrical volume envelope
could be applied to the mixed signal to ensure that there is no peak
or dip in the contour of the mixed signal. Because the envelope is
symmetrical, applying it preserves a(t)/a(-t). It can also be
incorporated directly into a(t).

All that is not so far off from the application you describe.

 but i don't think it is necessary to deal with lags where Rxx(tau)  0.  why
 splice a waveform to another part of the same waveform that has opposite
 polarity?  that would create an even a bigger glitch.

Splicing at quiet regions with negative correlation can give a smaller
glitch than splicing at louder regions with positive correlation. This
applies particularly to rhythmic material like drum loops, where the
time lag between the splice points is constrained, and it may make
most sense to look for quiet spots. However, if it's already so quiet
in there, I don't know how much it matters what you use for a
cross-fade.

Apart from it's so quiet it doesn't matter, I can think of one other
objection against using cross-fades tailored for r  0: For example,
let's imagine that our signal is white noise generated from a Gaussian
distribution, and we are dealing with given splice points for which
Rxx(tau)  0 (slightly). Now, while the samples of the signal were
generated independently, there is by accident a bit of negative
correlation in the instantiation of the noise, between those splice
points. Knowing all this, shouldn't we simply use a constant-power
fade, rather than a fade tailored for r  0, because random deviations
in noise power are to be expected, and only a constant-power fade will
produce noise that is statistically identical to the original. I would
imagine that noise with long-time non-zero autocorrelation (all the
way across the splice points) is a very rare occurrence. Then again,
do we really know all this, or even that we are dealing with noise.

I should note that Rxx(tau)  0 does not imply opposite polarity, in
the fullest sense of the adjective. Two equal sinusoids that have
phases 91 degrees apart have a correlation coefficient of about
-0.009.

RBJ, I'd like to return the favor and let you know that I have great
respect for you in these matters (and absolutely no disrespect in any
others :-) ). Hey, I wonder if you missed also my other post in the
parent thread? You can search for
AANLkTim=eM_kgPeibOqFGEr2FdKyL5uCCB_wJhz1Vne

-olli
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Re: [music-dsp] A theory of optimal splicing of audio in the time domain.

2011-01-22 Thread Victor Lazzarini

OK, so explain a bit more.

On 21 Jan 2011, at 22:55, Sampo Syreeni wrote:

My best bet? Go into the cepstral domain to find the most likely  
loop duration


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[music-dsp] A theory of optimal splicing of audio in the time domain.

2010-12-06 Thread robert bristow-johnson


 a few mistakes are spotted and corrected before i forget 


This is a continuation of the thread started by Element Green titled:  
Algorithms for finding seamless loops in audio


As far as I know, it is not published anywhere.  A few years ago, I  
was thinking of writing this up and publishing it (or submitting it  
for publication, probably to JAES), and had let it fall by the  
wayside.  I'm publishing the main ideas here on music-dsp because of  
some possible interest here (and the hope it might be helpful to  
somebody), and so that prior art is established in case of anyone  
like IVL is thinking of claiming it as their own.  I really do not  
know how useful it will be in practice.  It might not make any  
difference.  It's just a theory.


__

Section 0:

This is about the generalization of the different ways we can splice  
and crossfade audio that has these two extremes:


  (1)  Splicing perfectly coherent and correlated signals
  (2)  Splicing completely uncorrelated signals

I sometimes call the first case the constant-voltage crossfade  
because the crossfade envelopes of the two signals being spliced add  
up to one.  The two envelopes meet when both have a value of 1/2.  In  
the second case, we use a constant-power crossfade, the square of  
the two envelopes add to one and they meet when both have a value of  
sqrt(1/2)=0.707.


The questions I wanted to answer are: What does one do for cases in  
between, and how does one know from the audio, which crossfade  
function to use?  How does one quantify the answers to these  
questions?  How much can we generalize the answer?


__

Section 1: Set up the problem.

We have two continuous-time audio signals, x(t) and y(t), and we want  
to splice from one to the other at time t=0.  In pitch-shifting or  
time-scaling or any other looping, y(t) can be some delayed or  
advanced version of x(t).


   e.g.y(t) = x(t-P)

   where P is a period length or some other good splice  
displacement.  We get that value from an algorithm we call a pitch  
detector.


Also, it doesn't matter whether x(t) is getting spliced to y(t) or the  
other way around, it should work just as well for the audio played in  
reverse.  And it should be no loss of generality that the splice  
happens at t=0, we define our coordinate system any damn way we damn  
well please.


The signal resulting from the splice is

   v(t)  =  a(t)*x(t) + a(-t)*y(t)

By restricting our result to be equivalent if run either forward or  
backward in time, we can conclude that fade-out function (say that's  
a(t)) is the time-reversed copy of the fade-in function, a(-t).


For the correlated case   (1):   a(t)+  a(-t)= 1   for all t

For the uncorrelated case (2):  (a(t))^2 + (a(-t))^2 = 1   for all t

This crossfade function, a(t), has well-defined even and odd symmetry  
components:


   a(t)  =  e(t) + o(t)
where

   even part:  e(t) =  e(-t)  =  ( a(t) + a(-t) )/2
   odd part:   o(t) = -o(-t)  =  ( a(t) - a(-t) )/2

And it's clear that

   a(-t)  =  e(t) - o(t)  .


For example, if it's a simple linear crossfade (equivalent to splicing  
analog tape with a diagonally-oriented razor blade):


 { 0 for   t = -1
 {
  a(t) = { 1/2 + t/2 for  -1  t  1
 {
 { 1 for   t = 1

This is represented simply, in the even and odd components, as:

  e(t) = 1/2

 { t/2   for  |t|  1
  o(t) = {
 { sgn(t)/2  for  |t| = 1


   where  sgn(t) is the sign function:  sgn(t) = t/|t| .

This is a constant voltage-crossfade, appropriate for perfectly  
correlated signals; x(t) and y(t).  There is no loss of generality by  
defining the crossfade to take place around t=0 and have two time  
units in length.  Both are simply a matter of offset and scaling of  
time.


Another constant-voltage crossfade would be what I might call a Hann  
crossfade (after the Hann window):


  e(t) = 1/2

 { (1/2)*sin(pi/2 * t) for  |t|  1
  o(t) = {
 { sgn(t)/2for  |t| = 1


Some might like that better because the derivative is continuous  
everywhere.  Extending this idea, one more constant-voltage crossfade  
is what I might call a Flattened Hann crossfade:


  e(t) = 1/2

 { (9/16)*sin(pi/2 * t) + (1/16)*sin(3*pi/2 * t) for |t|  1
  o(t) = {
 { sgn(t)/2 for |t| = 1

This splice is everywhere continuous in the zeroth, first, and second  
derivative.  A very smooth crossfade.


As another example, a constant-power crossfade would be the same as  
any of the above, but where the above a(t) is square rooted:


 { 0   for   t = -1
 {
  a(t) = { sqrt(1/2 + t/2) for  -1  t  1
 {
 { 1   for   t = 1

This 

Re: [music-dsp] A theory of optimal splicing of audio in the time domain.

2010-12-06 Thread Stefan Stenzel
On 06.12.2010 08:59, robert bristow-johnson wrote:
 
 This is a continuation of the thread started by Element Green titled: 
 Algorithms for finding seamless loops in audio

I suspect it works better to *construct* a seamless loop instead trying find 
one where there is none.

Stefan
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