U.S. patent application number 10/129692 was filed with the patent office on 2002-11-07 for audio signal processing with adaptive noise-shaping modulation.
Invention is credited to Nuijten, Petrus Antonius Cornelis Maria, Reefman, Derk.
Application Number | 20020165631 10/129692 |
Document ID | / |
Family ID | 8171989 |
Filed Date | 2002-11-07 |
United States Patent
Application |
20020165631 |
Kind Code |
A1 |
Nuijten, Petrus Antonius Cornelis
Maria ; et al. |
November 7, 2002 |
Audio signal processing with adaptive noise-shaping modulation
Abstract
Processing an audio signal is provided, which processing
comprises conversion of the audio signal into a digital signal by a
noise-shaping modulation, compressive encoding of the digital
signal at a predetermined sampling rate into a compressed digital
signal, and supplying the compressed digital signal, wherein the
noise-shaping modulation is adaptive in response to at least one
parameter.
Inventors: |
Nuijten, Petrus Antonius Cornelis
Maria; (Eindhoven, NL) ; Reefman, Derk;
(Eindhoven, NL) |
Correspondence
Address: |
US Philips Corporation
Intellectual Property Department
580 White Plains Road
Tarrytown
NY
10591
US
|
Family ID: |
8171989 |
Appl. No.: |
10/129692 |
Filed: |
May 8, 2002 |
PCT Filed: |
September 6, 2001 |
PCT NO: |
PCT/EP01/10340 |
Current U.S.
Class: |
700/94 ;
G9B/20.001; G9B/20.014 |
Current CPC
Class: |
H03M 3/43 20130101; H04B
14/064 20130101; H03M 3/406 20130101; H03M 3/482 20130101; G11B
20/00007 20130101; G11B 2020/00065 20130101; H03M 3/49 20130101;
H03M 3/452 20130101; G11B 20/10527 20130101 |
Class at
Publication: |
700/94 |
International
Class: |
G10L 019/04; H03M
003/00; G06F 017/00 |
Foreign Application Data
Date |
Code |
Application Number |
Sep 8, 2000 |
EP |
00203097.1 |
Claims
1. A method of processing an audio signal, the method comprising
the steps of: conversion of the audio signal into a digital signal
by a noise-shaping modulation, compressive encoding of the digital
signal at a predetermined sampling rate into a compressed digital
signal, and supplying the compressed digital signal, the method
being characterized in that the noise-shaping modulation is
adaptive in response to at least one parameter.
2. A method as claimed in claim 1, wherein said conversion step
includes low-pass filtering said audio signal prior to said
adaptive noise-shaping modulation.
3. A method as claimed in claim 2, wherein said low-pass filtering
is adaptive in response to at least one parameter.
4. A method as claimed in claim 3, wherein said adaptive
noise-shaping modulation and/or said adaptive low-pass filtering is
controlled by feed-back control, said at least one parameter
comprising the signal level obtained from said digital signal.
5. A method as claimed in claim 1, wherein the adaptive
noise-shaping modulation comprises an adaptive low-pass filtering
prior to a non-adaptive sigma-delta modulation.
6. A method as claimed in claim 1, wherein said compressive
encoding comprises linear prediction filtering of said digital
signal, and wherein said at least one parameter is based on data
obtained from said prediction filtering.
7. A method as claimed in claim 1, wherein said at least one
parameter comprises a signal power in a selected frequency band of
said digital signal.
8. A method as claimed in claim 7, wherein said selected frequency
band is above 20 kHz.
9. A method as claimed in claim 1, wherein the adaptive
noise-shaping modulation comprises an adaptive sigma-delta
modulation having at least one pole above 20 kHz.
10. A method as claimed in claim 9, wherein said pole is positioned
in the high frequency range from 300 kHz and above.
11. A method as claimed in claim 1, wherein the adaptive
noise-shaping modulation is of a multiple resonator structure with
a loop filter acting as a band-pass filter in parallel with a
low-pass filter.
12. An device for processing an audio signal, the device
comprising: means for conversion of the audio signal into a digital
signal by a noise-shaping modulation, means for compressive
encoding of the digital signal at a predetermined sampling rate
into a compressed digital signal, and means for supplying the
compressed digital signal, the apparatus being characterized in
that the noise-shaping modulation is adaptive in response to at
least one parameter.
13. An apparatus for transmitting or recording an audio signal, the
recording apparatus comprising: an input unit to obtain an audio
signal, an audio signal processing device as claimed in claim 12 to
process the audio signal to obtain a processed audio signal, an
output unit for outputting the processed audio signal.
Description
[0001] The present invention relates to processing an audio signal,
e.g. for recording or transmission, the processing comprising the
steps of conversion of the audio signal into a digital signal by a
noise-shaping modulation, compressive encoding of the digital
signal at a predetermined sampling rate into a compressed digital
signal, and supplying the compressed digital signal.
[0002] International Patent Application WO 98/16014 discloses a
data compression apparatus for data compressing an audio signal.
The data compression apparatus comprises an input terminal for
receiving the audio signal, a 1-bit A/D converter for A/D
converting the audio signal so as to obtain a bitstream signal, a
lossless coder for carrying out a lossless data compression step on
the bit-stream signal so as to obtain a data compressed bit-stream
signal, and an output terminal for supplying the data compressed
bit-stream signal. Further, a recording apparatus and a transmitter
apparatus comprising the data compression apparatus are disclosed.
In addition, a data expansion apparatus for data expanding the data
compressed bit-stream signal supplied by the data compression
apparatus is disclosed, as well as a reproducing apparatus and a
receiver apparatus comprising the data expansion apparatus.
[0003] It is an object of the invention to provide advantageous
compression. To this end, the invention provides a signal
processing method and device and an apparatus for recording or
transmission as defined in the independent claims. Advantageous
embodiments are defined in the dependent claims.
[0004] According to a first aspect of the invention, the
noise-shaping modulation is adaptive in response to at least one
parameter. The invention is based on the recognition that by making
the noise-shaping modulation adaptive, the compression gain of the
encoder can be influenced. This is because a change in the
noise-shaping modulation influences the correlation within the
audio signal. Higher correlated signals can be better predicted an
thus be better compressed. This aspect of the invention is
especially advantageous for lossless encoders such as used in the
encoding of Direct Stream Digital (DSD) signals e.g. for storing on
Super Audio Compact Disc (SACD).
[0005] By using adaptive sigma-delta modulation in the
noise-shaping modulation, an increase of compression gain can be
obtained by giving in on dynamic range. Listening tests have
demonstrated that the huge dynamic range of the SACD re-cording
medium appears to be less important in the sense that e.g. a
reduction of the dynamic range from 105 dB to 95 dB would hardly be
perceivable. Particularly at high signal levels a listener will due
to masking effects in general be insensitive to a slight reduction
in dynamic range. Experiments have revealed that several ways
exist, by which the structure of a sigma-delta modulator can be
adapted or modified to provide a higher compression gain from the
encoding algorithm, such as use of a lower order sigma-delta
modulator and/or creating structure in the high frequency noise of
the modulator.
[0006] In an advantageous embodiment of the invention the
conversion of the audio signal into the digital signal includes
low-pass filtering of the audio signal followed by an adaptive
noise-shaping modulation (e.g. sigma-delta modulation). Thereby, a
further increase of compression gain may be obtained, but to a
certain extent at the expense of a signal quality degradation
caused by the bandwidth limitation resulting from the low-pass
filtering.
[0007] The input audio signal may be supplied as an analog signal,
whereby the adaptive sigma-delta modulation is conducted as part of
the noise-shaping modulation, by which the audio signal is
converted into a digital signal such as 1 bit bit-stream signal as
prescribed by the DSD signal format
[0008] The audio signal may alternatively be supplied to the
conversion as a digital signal such as a 1 bit bitstream signal,
which may be obtained by initial oversampling of an analog audio
signal at a rate, which is a multiple of the predetermined sampling
rate for the compressive encoding. In connection with the
above-mentioned preferred embodiment the low pass filtering and
noise-shaping modulation may thereby include downsampling of the 1
bit bitstream signal to the predetermined sampling rate. Thus, with
a predetermined sampling rate of 64 times the sampling frequency of
44.1 kHz the oversampling could be conducted at a rate of 256 times
the sampling frequency. At this sampling level any signal
processing can be effected.
[0009] In the following the invention will be further explained
with reference to the accompanying drawings, in which
[0010] FIGS. 1 and 2 are simplified schematic block diagrams of two
alternative embodiments of a signal processing apparatus according
to the invention,
[0011] FIGS. 3 to 5 are diagrams illustrating alternative ways of
implementing sigma-delta modulation and/or low-pass filtering in
response to a parameter of the audio signal,
[0012] FIG. 6 is a simplified topology diagram of a 5th order
sigma-delta modulator for use in any of the alternative
configurations in FIGS. 1 and 2,
[0013] FIG. 7 is a graphic representation of compression gain for
various orders of sigma-delta modulators,
[0014] FIG. 8 is a graphic representation of the effect of adding
an extra pole in a high frequency range to the sigma-delta
modulator, and
[0015] FIG. 9 is a graphic representation of the relationship
between compression gain and signal power in a selected frequency
band of an audio signal.
[0016] In the diagram in FIG. 1 an analog input audio signal is
supplied to a converter 1 comprising a noise-shaping modulator 2,
from which a digital signal is supplied to a lossless encoder 3.
The modulator 2 may typically be a sigma-delta modulator supplying
the digital signal in form of a bit-stream signal such as a 1 bit
bitstream signal in the DSD format.
[0017] The lossless encoder 3 may typically have a structure
incorporating framing, whereby the input signal supplied to it, is
split up in small parts enabling the encoder to exploit the
short-term pseudo-stationary properties of the audio signal as well
as pseudo-stationary properties of the quantization errors of the
sigma-delta modulator 1 and prediction, e.g. by means of a linear
FIR filter 4, to remove the dependencies or redundancy between
successive source samples as much as possible before the coding,
which may be conducted in the form of variable length entropy
encoding, e.g. using Huffman-like coding algorithms, or arithmetic
encoding.
[0018] Thereby, the encoder 3 supplies a compressed digital signal
which as shown may be supplied for re-cording on a record carrier
such as a SACD disc, but may also be used e.g. for transmission via
a transmission medium.
[0019] In the configuration shown in FIG. 1 the compression gain of
the compressed lossless encoded signal supplied by the encoder 2 is
increased in accordance with an embodiment of the invention by
adaptation or modification of the sigma-delta modulator 1 in
response to a parameter P. As will appear from the following
description several approaches can be used, according to the
invention, for such an adaptation or modification of the structure
of the sigma-delta modulator such as use of a lower order modulator
or creating structure in the high frequency noise of the
modulator.
[0020] In the alternative configuration in FIG. 2 a digital audio
input signal is supplied to a converter 5 before being supplied to
the lossless encoder 3. The converter 5 includes a low-pass filter
6, by which the bandwidth of the input signal is limited, e.g. to
100 kHz in conformity with the bandwidth specification of the DSD
format or even to 50 kHz, followed by an adaptive sigma-delta
modulator 7. Although not strictly necessary, also the low-pass
filter 7 is preferably made adaptive in response to at least one
parameter of the audio signal, which would preferably be the same
as, but could also be different from the signal parameter used for
the adaption or modification of the sigma-delta modulator 7. In a
simple embodiment, in the case the low-pass filter 7 is adaptive,
the sigma-delta modulator may be non-adaptive.
[0021] The combination of low-pass filter 6 and adaptive
sigma-delta modulator 7 in the converter 5 provides for
requantisization of the digital input signal. The signal processing
apparatus as shown in FIG. 2 may comprise several successive
pre-processing blocks 5 to achieve a desired increase of the
compression gain.
[0022] The low-pass filter 6 in the converter 5 may e.g. be a 7th
order IIR Chesbyshev type 1 filter and generally the compression
gain increase obtained by one or more pre-processing stages as
shown in FIG. 2 will be higher than for the configuration in FIG.
1, which may also result, however, in some quality degradation of
the signal due to the bandwidth limitation.
[0023] Obviously, one or more converters 5 as shown in FIG. 2 may
also be used in the configuration shown in FIG. 1 between the
modulator 1 and the encoder 2.
[0024] The adaptive sigma-delta modulator 1 or 7 may be of the 3rd,
5th or 7th order to provide compression gains ranging from 3.7 or
higher for a 3rd order modulator down to only 2.3 or lower for a
7th order modulator as illustrated in the graphical representation
in FIG. 7. It should be emphasized, however, that in general the
use of a lower order modulator will result in degradation of audio
quality due to a lower dynamic range in the audio band.
[0025] According to an embodiment of the invention, the sigma-delta
modulation in modulator 2 is adapted or modified in response to at
least one parameter P of the audio signal in order to confine
increase of the compression gain to parts of the lossless encoded
signal, for which this is needed. This would typically be at high
signal levels, where the compression provided by the encoder 3 will
usually drop. As shown in FIG. 3, this may be implemented by means
of a feed-back loop 9 incorporating a signal level detector 10.
Alternatively, the adaption may as shown in FIG. 4 comprise a
control device 11 responding to data obtained from the prediction
filter 4 in the encoder 3 or, as shown in FIG. 5 a control signal
obtained from a signal power extractor and correlator 12, as will
be further explained in the following.
[0026] The diagram in FIG. 6 shows a preferred topology of a 5th
order sigma-delta modulator for use in any of the configurations in
FIGS. 1-5. The illustrated topology is based on a multiple
resonator structure, in which the coefficients c1, c2, . . . c5 in
the feed-back loops of resonators R1, R2, . . . R5 determine the
poles of the loop filters (or zeroes of the noise transfer
function). Whereas the illustrated topology is for a 5th order
modulator the same topology may be used for a 7th order modulator
just by adding another resonator structure.
[0027] As mentioned above, FIG. 7 shows a graphic representation of
compression gain cg for various orders of sigma-delta modulators as
a function of amplitude swa for a 10 kHz audio sine wave signal to
illustrate the in-crease in compression gain for lower order
modulators, which is obtained, however, at the expense of an
increased quantisization noise in the audio band.
[0028] In ordinary design of a modulator the poles will normally be
positioned in the audio band, According to a further embodiment of
the invention it is preferred, however, as shown in the graphic
representation in FIG. 8 of compression gain for various signal as
a function of the pole position pp for a 5th order sigma-delta
modulator, to have at least one pole positioned outside the audio
band to create additional structure in the--otherwise almost
flat--high frequency part of the sigma-delta spectrum.
[0029] In standard designs of sigma-delta modulators the poles are
typically positioned at 8.7, 15.7 and 19.5 kHz, whereas in
accordance with the invention the last pole is preferably shifted
from the 20 kHz region to higher frequencies. As will appear from
the diagram, positioning of the pole around 200 kHz may result in a
rather bad compression gain, because this pole position is too
close to the point where the modulator will change from 5th to 1st
order behavior, whereby the modulator becomes almost unstable.
[0030] On the other hand positioning of this pole around 300 kHz or
higher may lead to a significant increase of compression gain. This
may be accompanied by a slight decrease of the signal-to-noise
performance, which will be quite acceptable for the adapted
modulator, however, because the extra noise is introduced on the
high side of the frequency band, where the human ear is less
sensitive.
[0031] The shifting of the pole position from the 20 kHz region
towards higher frequencies can be effected by addition of a
separate extra band pass filter to the-existing modulator
structure, e.g. in parallel to the low-pass loop filter. By use of
a 2nd order Butter-worth band pass filter for such a parallel
filter a significant increase of compression gain can be realized
with the resulting modulator remaining stable for large inputs and
the signal-to-noise performance in the audio band remaining
virtually unchanged with respect to an unmodified modulator
[0032] According to the invention a further approach as shown in
FIG. 5 for the adaptation of the adaptive sigma-delta modulator
and/or the adaptive low-pass filter in the pre-processing device is
to provide an estimate of the amount of data that can be stored on
the recording medium such as a SACD disc and use such an estimate
for the adaptive control of the sigma-delta modulator and/or the
low-pass filter.
[0033] In theory, to provide such an estimate it could be chosen to
determine compression gains only for, e.g. randomly, selected
subset of music recordings and use this estimate as an average gain
indication for a whole piece of music.
[0034] In view of the fact, however, that typical pieces of music
have a very wide coverage of gains with significant short-time
correlations, a very significant fraction of the piece of music
would have to be used to obtain an estimate by this approach with
the required precision. Due to the amount of computation that would
inevitably be required for such an operation this approach could
not be seen as an acceptable solution.
[0035] According to the invention a correlation between the signal
power of the bitstream signal in the DSD format and the compression
gain is used to provide the desired estimate.
[0036] Whereas investigations have demonstrated that in the audio
signal band, e.g. up to 20 kHz, itself the correlation is very weak
due to a very flat response curve for the compression gain as
function of signal power, a fully usable correlation resulting from
a very steep response curve as illustrated in the graphic
representation in FIG. 9 can be observed by shifting to a frequency
band just above the normal audible range, e.g. from 20 to 50 kHz.
Preliminary limited experiments have revealed that in this way
estimates with an accuracy within 1% can be obtained.
[0037] It should be noted that the above-mentioned embodiments
illustrate rather than limit the invention, and that those skilled
in the art will be able to design many alternative embodiments
without departing from the scope of the appended claims. In the
claims, any reference signs placed between parentheses shall not be
construed as limiting the claim. The word `comprising` does not
exclude the presence of other elements or steps than those listed
in a claim. The invention can be implemented by means of hardware
comprising several distinct elements, and by means of a suitably
programmed computer. In a device claim enumerating several means,
several of these means can be embodied by one and the same item of
hardware. The mere fact that certain measures are recited in
mutually different dependent claims does not indicate that a
combination of these measures cannot be used to advantage.
* * * * *