View source for Noise shaping

== Introduction ==

Noise shaping is a technique that is often used in combination with
[[Dithering|dithering]] in the context of [[Quantize|quantization]]. The purpose of noise shaping
is to alter the spectral ''shape'' of the error that is introduced due
to [[Dithering|dithering]] and [[Quantize|quantization]]. Sometimes it may be desirable to
concentrate most of the noise power in a certain frequency band so
that the overall noise becomes less perceivable.

== Applications ==

In the field of digital image processing a popular noise shaping
technique is known as "Floyd Steinberg dithering". This algorithm is
used to reduce the noise power at lower spatial frequencies which
results in an improved perceptual quality.

For word length reductions of audio signals (i.e conversion from a 24bit
signal to a 16bit signal) noise shaping can be used to push quantization
noise out of perceptually important frequency bands to perceptually less
important frequency bands (>20 kHz) in order to reduce the perceived
loudness of the quantization noise.

Another example is the 
[[Super Audio Compact Disc|Super Audio CD]]. The SACD format represents
the most extreme trade-off between word length (1bit/sample) and
sampling frequency (2.8 Mhz). Due to the low word length high power
noise is introduced. With the help of noise shaping this noise is
usually rejected by about 120 dB in the audible band (0-20 kHz) at the
cost of a higher noise power in the ultrasonic range (+3.5dB for
frequencies above 100 kHz).

== Limits & filter design restrictions ==

Noise Shaping without [[Dithering|''dithered'']] quantization is also possible but
less useful because the spectral characteristics of ''undithered''
quantization noise is not easily predictable and depends heavily on the
digital signal that is to be quantized. The SACD case is an example where
[[Dithering|dithering]] can't be properly applied which may lead to a spectral
noise power distribution that deviates from the indented distribution. This has
been demonstrated by Lipshitz and Vanderkooy in
[http://www.sjeng.org/ftp/SACD.pdf their] [1] paper. 

It's not possible to lower the noise power for ''all'' frequencies
via noise shaping because this would be in violation of the
rate/distortion principles. But it ''is'' possible to lower the noise
in certain frequency bands at the cost of increased noise power in other
frequency bands. This property is also known as the
''noise shaping theorem''. It states that the average logarithmic magnitude response of
the corresponding noise shaping ''filter'' over the whole linear
frequency axis cannot drop below 0dB.

The noise shaping application poses some restrictions on the filter
design. The first sample of the filter's impulse response always equals
one. Also, interesting filters for noise shaping are so-called
''minimum phase filters''. This type of filter performs optimal in the sense of
the noise shaping theorem because the average logarithmig magnitude
response ''is'' 0dB.

For the special [[Super Audio Compact Disc|SACD]] case an additional
constraint comes into play: The magnitude response should not exceed
3.5dB. This is enforced to prevent so-called limit cycles.

The design of these kinds of filters is however not trivial. This is not
covered by this article.

== Digital filter reminder ==

A common representation of digital filters is their ''transfer function''
in form of a rational polynomial in z^-1. These polynomials are defined
by their coefficients b0,b1,b2,...,a1,a2,...:

<math>N(z) = \frac{B(z)}{A(z)} = \frac{b_0 + b_1 z^{-1} + b_2 z^{-2} + \ldots + b_m z^{-m}}
{1 + a_1 z^{-1} + a_2 z^{-2} + \ldots + a_m z^{-m}} </math>

where 'N' is the transfer function and 'm' the filter's ''order''.

=== FIR filters ===

In case a1=a2=a3=...=0 the filter's [http://en.wikipedia.org/wiki/Impulse_response impulse response] is finite which is
why these filters are called FIR ([[Finite Impulse Response Filter|finite impulse response]])
filters as opposed to IIR
[[Infinite Impulse Response Filter|infinite impulse response]] filters. Note that in case of
FIR filters the coefficients b0, b1, ... correspond to the impulse
response. The impulse response is another way a filter can be
characterized and describes what the filter does to a unit pulse. Let's
assume for the moment that a1=a2=a3=...=0. The the impulse response
would be:

 x:  0  0  1  0  0  0  0  ... 0  0  0 ... (input signal)
 y:  0  0  b0 b1 b2 b3 b4 ... 0  0  0 ... (output signal)

It helps to think of negative powers of z in transfer functions as a
delay: z^-1 = delay of one sample. z^-2 = delay of two samples etc.
Since a digital filter is a
[http://en.wikipedia.org/wiki/Discrete_system discrete-time]
[http://en.wikipedia.org/wiki/LTI_system_theory linear and time-invariant]
system it is
possible to determine the filter's output by a superposition of scaled
and shifted impulse responses. This leads to the following relationship
between input and output in the case of FIR filters:

 time domain                                     Z-transform
 -----------                                     -----------
 y[k] = b0 x[k] + b1 x[k-1] + b2 x[k-2] + ...    Y(z) = X(z) * B(z)

where x[] is the input signal, y[] the output signal and k the current
sample index. This is a simple convolution directly computed using the
given impulse response. The convolution in the time domain corresponds
to a multiplication in the Z-domain.

=== IIR filters ===

IIR filters are a simple extension to the FIR case:

 time domain                                Z-transform
 -----------                                -----------
      y[k] + a1 y[k-1] + a2 y[k-2] + ...    Y(z) * A(Z) = X(Z) * B(Z)
 = b0 x[k] + b1 x[k-1] + b2 x[k-2] + ...

Solving for Y we get

 time domain                                     Z-transform
 -----------                                     -----------
 y[k] = b0 x[k] + b1 x[k-1] + b2 x[k-2] + ...    Y(Z) = X(Z) * N(z)
                - a1 y[k-1] - a2 y[k-2] + ...

which is the well known ''difference formula''. Its signal flow graph
representation is known as
[http://ccrma.stanford.edu/~jos/filters/Direct_Form_I.html direct form I].

== Noise Shaper Implementation ==

Usually a filter's transfer function is called 'H'. In this article 'N'
has been used because this filter is also known as noise transfer
function (NTF) in the context of noise shaping. Later, another filter
'H' is introduced that has a special relation to 'N'.

=== Direct Form I ===

For convenience let us substitute the actual difference in the
difference formula for 'u'. Also, b0 must be equal to one for noise
shaping filters as mentioned above. This leads to

 y[k] = x[k] + u[k]   with u[k] = b1 x[k-1] + b2 x[k-2] + ...
                                - a1 y[k-1] - a2 y[k-2] + ...

Note that since the filter is supposed to be applied on the quantization
noise y[] corresponds to the filtered noise and x[] to the unfiltered
noise. Suppose s[] is the signal we're trying to quantize. The following
pseudo code snippet shows the inner loop over k of ''one possible''
noise shaping implementation:

 (1) compute u[k]
 (2) wanted[k] = s[k] + u[k]
 (3) quantized[k] = quantize( wanted[k] + dither() )
 (4) y[k] = quantized[k] - s[k]
 (5) x[k] = y[k] - u[k]

Note that the following is true

 time domain                          Z-transform
 -----------                          -----------
 quantized[k] = s[k] + u[k] + x[k]    Q(z) = S(z) + X(z) * N(z)
              = s[k] + y[k]                = S(z) + Y(z)

u[k], wanted[k] are temporary variables. The values are not needed
outside this iteration. But the past filtered and unfiltered error
samples need to be remembered (y[] & x[]) upto some amount of samples
determined by the filter order.

=== Decoupling filtering and quantization ===

[[Image:Noiseshaper3.png|thumb|right|200px]]
As mentioned, the previous example is ''one possible'' implementation.
It's not the best possible one. But it is the first version that comes to mind.
There's another common formulation which requires
some more transfer function fiddeling:

<math> N(z) = \frac{B(z)}{A(z)} = 1 - \frac{A(z)}{A(z)} + \frac{B(z)}{A(z)}</math>

The two fractions have a common denominator so they can be easily
combined to one fraction. It's also possible to factor z^-1 out of the fraction:

<math>N(z) = 1 - \frac{A(z)-B(z)}{A(z)} = 1 - z^{-1} \frac{z ( A(z)-B(z) )}{A(z)} =: 1 - z^{-1} H(z)</math>

Substituting z*(A(Z)-B(Z)) for H(z) leads to a new filter that is related to N.
Note that with b0=1 it follows that z*(1-b0)=0. So, H(z) can be written as

<math>H(z) = \frac{c_0 + c_1 z^{-1} + c_2 z^{-2} + \ldots}{1 + a_1 z^{-1} + a_2 z^{-2} + \ldots} </math>

with <math>c_{k-1} = a_k - b_k</math> for <math>k = 1...m</math>. H(z)
is some ''arbitrary'' filter. The big advantage now is that it is
possible to plug nearly any digital filter implementation for H(z) into
the noise shaping loop. One obvious approach is the difference formula
from above but there are other alternatives with desirable properties.
However, the purpose of this article is not to discuss digital filters
in general but their application as noise shaping filters. So,
throughout the rest of this article H(z) is simply treated as a black
box. This black box has a certain state and gives us an output sample
for each input sample. The modified pseudo code loop looks like this:

 (1) h_out[k] = H.filter(last_error)
 (2) wanted[k] = s[k] - h_out[k]
 (3) quantized[k] = quantize( wanted[k] + dither() )
 (4) last_error = quantized[k] - wanted[k]

which is equivalent to the shown signal flow graph where the unit
sample delay z^-1 is implemented by the state variable 'last_error':
Remember that this black box filter H(z) doesn't correspond the the
actual noise shaping filter N(z). Depending on the context published
coefficients for noise shaping correspond either to N(z), H(z), or
-H(z).

== Example filters ==

The following two filters have been designed by Sebastian Gesemann
via non-linear optimization to match a slightly modified version of
the LAME MP3 encoder's ATH (absolute threshold of hearing) curve. They
were first published in the
[http://www.hydrogenaudio.org/forums/index.php?showtopic=47980&pid=551558&st=0 Hydrogenaudio forum]
including the noise transfer function's second order factorizations.

=== Sampling rate of 44100 Hz, derived from LAME's ATH formula ===

The noise transfer function is given by

<math>N(z) = \frac{1 -1.1474 z^{-1} +0.5383 z^{-2} -0.3520 z^{-3} +0.3475 z^{-4}}
{1 +1.0587 z^{-1} +0.0676 z^{-2} -0.6054 z^{-3} -0.2738 z^{-4}}</math>

The corresponding filter H is given by

<math>H(z) = \frac{2.2061 -0.4706 z^{-1} -0.2534 z^{-2} -0.6214 z^{-3}}
{1 +1.0587 z^{-1} +0.0676 z^{-2} -0.6054 z^{-3} -0.2738 z^{-4}}</math>

=== Sampling rate of 48000 Hz, derived from LAME's ATH formula ===

The noise transfer function is given by

<math>N(z) = \frac{1 -1.3344 z^{-1} +0.7455 z^{-2} -0.4602 z^{-3} +0.3463 z^{-4}}
{1 +0.9030 z^{-1} +0.0116 z^{-2} -0.5853 z^{-3} -0.2571 z^{-4}}</math>

The corresponding filter H is given by

<math>H(z) = \frac{2.2374 -0.7339 z^{-1} -0.1251 z^{-2} -0.6033 z^{-3}}
{1 +0.9030 z^{-1} +0.0116 z^{-2} -0.5853 z^{-3} -0.2571 z^{-4}}</math>

==Notes==

==References== 
* Lipshitz. P, Stanley and John Vanderkooy. "Why 1-bit Sigma-Delta Conversion is Unsuitable for High-Quality Applications". 110th Convention.  <i>Audio Engineering Society</i> (May 2001). 2-4.


[[Category:Signal Processing]]