Untitled
US 7,184,556 B1
Page 2
6.252,968 B1 * 6/2001 Narasimhan et al. ....... 381/103
6,295,364 B1* 9/2001 Finn et al. ........... .... 38.1/110
6,317, 117 B1 * 11/2001 Goff ........................... 345/156
Toole, Floyd E. and Olive, Sean E., “The Modification of Timbre by
Resonances: Perception and Measurement” J. Audio Eng. Soc., vol.
36, No. 3, Mar, 1988, pp. 122-123.
Fasoldt, A., Jul/Aug. 1999 column in Fanfare Magazine, pp.
6,721,428 B1
383-384.
|U.S. PATENT DOCUMENTS
2003/0132956 A1
4/2004 Allred et al.
7/2003 Duncan
Herschelmann, Russ, “Home Theater Architect–Is It Golden or
OTHER PUBLICATIONS
Goofy?” Stereophile Guide to Home Theater Nov. 1998, pp. 42-44.
Mendonsa, Ruth A., “Laser Vibrometer Heightens Speaker Design”
Photonics Spectra, Jan. 1998, p. 18.
* cited by examiner
U.S. Patent
US 7,184,556 B1
Sheet 4 of 15
Feb. 27, 2007
F/G, 5
| ||
NT
?????
TNT
|N|| ||
Nº
|| |
HLINII]XJ
EXCURSION
?[WITZ
N
NS
ITI-S…
O
|||
NUH
[ XN
=U=
ENCY IN CYCLES PER SECOND
100
U.S. Patent
Feb. 27, 2007
Sheet 6 of 15
`s
s
g . à ???
US 7,184,556 B1
N
U.S. Patent
Feb. 27, 2007
Freq Cut/Boost
334405Hz
Sheet 7 Of 15
US 7,184,556 B1
Freq Cut/Boost
BAD CD NO DONUT
Main Compensation#2
| Main ||
| Smiley Foce
334405Hz
BAD CD NO DONUTT
|| || T
U.S. Patent
Feb. 27, 2007
Sheet 8 of 15
US 7,184,556 B1
Resonance Compensation;2|Standing Wave rejection
| Moin. Tº TT Smiley Foce | Cutoff Response
Main I
TI Similey Fice Eutoff Response
Resonance Compensation #2 istanding Wave Rejection
Compensation
o Cut/Boost
525846
EAD CO NO DONUTI
I
T
U.S. Patent
Feb. 27, 2007
Sheet 9 of 15
US 7,184,556 B1
First Notch
Freq Q Cut/Boost
Freq Cut/Boost
Freq Q Cut/Boost
BAD CD NO DONUT
Main ||
|| Smiley Foce ICutoff Response Response Compensation
Resonance Compensation #2 jStanding Wave Rejection|Allpass|Double Tuned Notch
Allpass
BAD CD NO DONUT
||
T
U.S. Patent
Feb. 27, 2007
Sheet 10 of 15
R
F/G. 9
?
US 7,184,556 B1
R2
?
*p p
OUTPUT
U.S. Patent
Feb. 27, 2007
F/G 11
Sheet 11 of 15
US 7,184,556 B1
U.S. Patent
Feb. 27, 2007
Sheet 12 of 15
US 7,184,556 B1
LOW ··??
BOOST
HIGH "Q”
NOTCH
COMBINED
RESPONSE
INPUT
OUTPUT
LOW Q
F/G, 13
U.S. Patent
Feb. 27, 2007
Sheet 13 of 15
US 7,184,556 B1
------- FREQUENCY ADJUST
DOUBLE TUNED
LOW "Q" BOOSIS
DOUBLE TUNED
HIGH "Q” NOTCHES
––– FREQUENCY ADJUST
COMBINED
|
RESPONSES
|
|
|
H- FREQUENCY ADJUST
INPUT (2)
-.
-
G.) OUTPUT
S S S S S S S S S S S S S S S S S S S S S S S S S S S S S S S S S S S S S S S S S S S S S S S S S S S S S S S S S S S S S S S S S S S S S S S S S S S S S S S S S S S S S S S S S S S S S S S SA S S S S S S S S S S S S S S S S S S S S S S S S S S S S S S S S S S S S S S S S S S S S S S S S S S S S S S S S S S S S S S S S
FREQUENCY
ADJUST
F/G, 14
U.S. Patent
2^ w—
0
|
408|
?GO!09
|
G
07
09
t<º?-=
?||
?:_
?O,
?º000LL00K0L0S0SS00L00LS0SSL0L0S
og
U.S. Patent
Feb. 27, 2007
10?
10K
Sheet 15 of 15
10K
US 7,184,556 B1
10K
R, @ CW, CCW
US 7,184,556 B1
1
2
for a wider frequency range can be designed and imple
mented in this manner. Several components are needed to
COMPENSATION SYSTEM AND METHOD
FOR SOUND REPRODUCTION
match the resonant behavior, but all interact with each other
RELATED APPLICATION
This application claims the benefit of co-pending provi
sional application Ser. No. 60/148,412 filed Aug. 11, 1999.
BACKGROUND OF THE INVENTION
10
1. Field of the Invention
This invention relates generally to a compensation
method and system for use in sonic transmission and repro
duction systems and more particularly to a compensation
method and system that uses parametric values to control or
adjust processes having transforms or models with proper
ties or responses like the components or elements used in the
transmission or reproduction system.
2. Description of Related Art
Most audio reproduction systems use electromechanical
loudspeakers to acoustically reproduce audio signals. The
electrical, mechanical, and acoustical properties of the loud
speakers are often less than ideal, causing distortions,
response anomalies, and other coloration of the sound. Many
techniques are used to compensate for the loudspeaker’s
characteristics in order to improve perceived audio quality.
Functional or behavioral models of loudspeakers are used
in practice and found in literature to develop such compen
sations. A good example of models and how the modeling
process works is described in “Active Equalization of Loud
speakers”, Speaker Builder, February 1997. Models consoli
date technical languages and are usually intended to imitate
or simulate the acoustic responses of the speaker system
from electrical stimulus. Model creation or synthesis fre
quently begins by making functional groupings of elements
which collectively represent or behave like all or part of the
speaker. Coil and magnet parts become motors, which are
represented by resistors, inductors, capacitors, back EMF
generators and other transformed parts. A combination of
factors such as air volume, moving mass, acoustic loading,
magnetic-braking, and mechanical losses might be analyzed
and simplified to LCR resonator networks or circuits. Most
15
20
25
30
35
40
often, the transformed electromechanical, acoustic, and
mechanical representations expressed in the model are fur
ther simplified or reduced to fewer elements. The model still
responds like the speaker, but the parts making the model no
longer have exact behavioral equivalence to the parts mak
ing the speaker. Consequently, traditional models are neither
intended, nor capable of making parametrically addressed
zero-phase compensations when speaker parts are changed.
One could characterize and invert the frequency response,
as well as other properties, of a well-conceived model and
achieve linear-phase correction of the loudspeaker. The
technique does work to a fashion, but its dedicated, inflex
ible circuitry or specialized process tied to the traditional
model limits its use to a one-speaker design. Some high
quality crossover networks constructed to divide the signal
spectrum amongst multiple drivers may have some conju
gate response correction like this.
A low-frequency resonant boost is intentionally designed
for most speakers. Frequently, traditional models are made
to represent quantifiable and predictable acoustic behavior
as well as other speaker design factors affecting bass
response. Mechanical construction and properties of air
determine frequency, resonant losses and the configuration’s
effect on acoustic output from the speaker. A good approxi
mation to a zero-phase conjugate or same-order correction
45
50
when adjustments are made for a different speaker of similar
concept and design. Therefore, the operation is not strictly
parametrically controlled, as the adjustments must be re
calculated from the model to create the minimum-phase or
exact match needed for best fidelity with the new speaker.
When more corrections are added the interaction problem
becomes formidable. The system must be tuned experimen
tally or the model analyzed each time an adjustment is made.
Consequently, lumped model processes for response flatten
ing are inherently designed for a specific speaker. The
process must be redesigned for other speakers.
Traditional curve-fitting methods can require hundreds of
data points and corresponding adjustments to set up and
many components or much processing power to match the
acquired frequency response. Analog methods are imprac
tical and digital processes require much computation and
extensive architectures to do this. Neither can provide phase
accurate responses or the hidden corrections described later
without having knowledge of the speaker and its operation.
Without a model, the effort to combine amplitude, time and
phase corrections together from measured responses
becomes formidable.
Some of the most important behaviors of loudspeakers
(with respect to acoustically perceptible effect) cannot be
modeled or implemented from traditional methods. Such
behaviors include standing wave interference, modal
breakup, and coupled resonance as well as nonlinear con
sequences from such potentially interacting acoustic and
mechanical behaviors. Counterproductive random motion or
breakup may occur. Even when the average response
remains flat or is the same as other frequencies being
reproduced, energy can build up during signal stimulus and
be released when the signal changes or ceases. In addition,
other spatial factors related to stiffness of moving parts and
high frequency de-coupling for motions away from a driving
voice coil need to be considered. Any of these can create
source movements, delayed energy release, and phase error
to binaural hearing. Often, such destructive responses can be
invisible or very difficult to interpret from traditional micro
phone-and-spectrum-analyzer calibration methods.
For example, unwanted responses arising from nodal and
standing wave behavior affect the settling time, directional
behavior, and radiated output of a speaker. Frequently, these
responses cause perceptual changes to intelligence signals
yet may not be visible or recognizable from response plots.
Mechanical motions having large stored energy can be out
of phase at different parts of the transducer. The acoustic
output might appear to be flat, but human binaural hearing
can localize the behavior to its source and the altered
ss
60
65
perception can degrade stereo imaging.
Often, mechanical disturbances are audible yet invisible
or hard to interpret from response measurements made using
a frequency sweep and microphone. Parts of a radiating
surface can vibrate with different phase relationships to
other parts, so that their additive acoustic output is low
compared to motions within the transducer and the energy
storage involved. When signals at the node frequency
change and suddenly stop the release of stored energy can
interact with other signals at different frequencies. The
resulting beat sounds between the two frequencies can be
audible and very objectionable. Sounds with spectra in the
interference frequency range may appear louder and granu
lar. Human binaural hearing can localize the disturbance to
the driver or surfaces from which directional lobes might
US 7,184,556 B1
3
bounce, thereby imparting further damage to the stage
illusion from multi-speaker stereo reproduction. For this
situation, frequencies creating the mechanical disturbance
must be sufficiently attenuated to prevent unmasked repro
duction of consequential responses. Experience has shown
that a sharp, deep notch needed to do this removes enough
energy around the correction frequencies to cause a nasal
sound. If this inappropriate correction is modified to achieve
flat response, then the mechanical sounds remain along with
a potential undesirable balance aberration.
Many small loudspeakers are constructed with a trans
5
10
ducer, enclosure, and some resonant means to extend bass
response such as a port or passive radiator. Usually, these
parts are designed to achieve a practical and economical
compromise between efficiency, frequency response accu
racy, bass extension, and acceptable distortion. Designers of
inexpensive, low-powered systems generally opt for higher
efficiency to reduce amplifier requirements along with
related costs of power supplies and packaging. The com
promise situation exposes many undesirable behavioral
15
nodal distortion corrections could be made from such a
20
aspects.
Most traditional speaker correction methods apply some
variation of amplitude equalization to flatten and extend
response from speakers. Adjustments are sometimes done by
ear. To be quantitative, one must acquire relevant data. The
most common techniques to do this use spectral analysis
from noise stimulus. Then, response plots or displays indi
cate how an equalizer is to be adjusted. More sophisticated
techniques based on delayed acceptance or sampled win
dows can measure first-arrival responses from the speaker
and remove higher-frequency room disturbance to create
anechoic-like data. The intent is to capture information
25
30
relevant either to a listener in a room or to standard mea
surement practice where a test microphone is usually speci
fied and placed one meter from the speaker. Such technique
creates a response that may sound balanced to the single
point test microphone. One or more known systems go
slightly beyond this by adjusting path lengths, or time delays
to align multiple speakers to a listening position.
Other techniques provide transient response waveforms,
waterfall or successions of spectral plots after an event.
Group delay and time-related information is acquired. Such
data needs interpretation and has limited use for frequency
response leveling practice. Some behavioral responses can
be recognized but much more information must be known
about the speaker. Measurement devices such as accelerom
eters, differential acoustic probes, as well as microphones,
are needed for this. Instrumentation may be placed near a
suspected behavior site and moved to explore how a
response changes with position. Weighted notches can be
tuned or slowly swept through suspected frequencies while
subjectively observing noise production. More information
is needed about dimensions of parts, listening positions, as
well as floor, shelf, possibly a computer monitor, or other
interceding objects that may be part of the listening envi
ronment. Other technical specifications or expressions are
needed to complete the conjugate model capable of time
phase-accurate correction.
A human operator can assume an alignment role by
adjusting a graphic equalizer, manually tuning a parametric
filter, or changing settings to a crossover device. Commer
cial analog components perform these functions, but they
have limitations. Graphic equalizers have up to 31 bands or
resonators, parametric devices include several adjustable
filters and a few have variable crossover and shelf functions.
Many more filters are needed. Combinations of graphic and
parametric equalizers are incapable of providing a large
4
enough number of points, nor the exact phase and time
response to effectively compensate complex behavior from
a loudspeaker. Either the corrections do not match specific
frequencies, thereby creating phase error, or the number of
filters is inadequate to deal with settling time and standing
wave issues. Group delay distortion, time-phase error,
incomplete correction and other shortcomings are likely to
outweigh other improvements.
DSP filters can create many more filter sections than is
practical from analog circuits. Graphic equalizers made up
with parametrically controlled sections have been used with
specialized control-generating software to create room
response leveling. Such processes are difficult to set up
because the room interferes with the identification of impor
tant behavioral indicators. Without their input, conjugate
response corrections are not possible. Standing-wave and
35
40
45
50
system. However, the awkward compiling and processing
needed to parametrically move the compensated notches
would be difficult. Most likely, a single point response
pickup and FFT has been used for data input to the system.
Such methods cannot respond to or provide the time-phase
information needed to create a true conjugate response to the
speaker. Analysis systems, such as MLSSA, can remove
room interference from measurements, and can produce
frequency, transient, and settling response data from a
loudspeaker system. However, the large amount of data from
these measurements must be interpreted. The multiple-band
graphic equalizer is not a good choice to install the correc
tion.
DSP systems can economically create many parametric
filters and time-related processes that are impractical with
analog circuitry. Traditional large-scale DSP systems have
little means to identify and cull out speaker behavior from
other measurement anomalies. Their frequency-domain
responses are likely to add phase errors and to overlook
delayed settling energy. The sound might improve for one
listening position but it will degrade for all others. More
likely, the reproduced sound will change without definitive
improvement.
Hence, those concerned with the reproduction of sound
have recognized the need for a system and method of
modeling the complete behavior of sound reproduction
devices such that conjugate responses to the sound repro
duction device responses may be created. The need for a
system and method employing modifiable conjugate
response has also been recognized. Furthermore, the need
has also been recognized for a system and method that
compensates the reproduction of sound independent of the
environment in which the sound is to be heard. The present
invention fulfills these needs and others.
SUMMARY OF THE INVENTION
ss
60
65
Briefly, and in general terms, the present invention pro
vides a system and method for modeling individual response
characteristics of a sonic reproduction device to create a
conjugate model for improving frequency, time, phase, and
amplitude performance of the device and to provide
improved sonic balance, sound clarity, reduced distortion
and improved stereo imaging.
In a first aspect, the invention relates to an apparatus for
modifying an electrical audio signal for input to a sonic
reproduction device characterized by a plurality of indi
vidual responses. The individual responses of the device
combine to define an overall response. Each individual
response includes one or more of a frequency, time, phase or
US 7,184,556 B1
17
provides a dead-band or band-reject capability to accom
modate manufacturing tolerances from one speaker to
another. Side frequency boost is still needed and double
tuned resonators are best used.
Other variations and simplifications can be made when
adjacent behavioral modes have similar properties, as is
likely with most speakers. Two or more rejection notches
can share Q and amplitude settings as well as compensation
boost. Combinations include two notches with three boosts,
two notches with two asymmetrical boosts, three with two,
etc. A single low-Q boost with a frequency halfway between
two notches can be used. Three low-Q boosts with frequen
10
cies below, above, and between are a better variation. For all
of these implementations, the notch depth is often great and
the side frequency boost is usually small. Usually, the
overall energy response to random noise averaged about the
compensation region is made to be the same or slightly
higher than without correction.
Delayed Interference or All-Pass—A hybrid analog-digi
tal CCD device can create a small, convenient tunable delay.
Though performance may be poor, they can be connected or
configured like the example in FIG. 16 to provide interfer
ence-like behavior. The circuit can create approximate con
jugate responses to wavelength related reflection and trans
mission behavior from walls, tables and the insides of
speaker enclosures or other parts of the transmission path or
system. The circuit can be set to create an inverted comb
filter or additive interference like response which would be
opposite in time, phase and amplitude to subtractive inter
ference loss from reflecting surfaces. The correction boosts
where interference takes away. The circuit can also be
adjusted to have a comb filter like response to cancel
additive energy from reflections within the speaker enclo
sure. Better time delay interference filters or comb filter like
responses can be made from DSP processes. Both the analog
and the DSP can be configured to be relevant to the physical
reflection model and like other parts of the correction
system, are controlled by parametric adjustments related to
physical behavior. The delay interference path filter has
controls relating to dimensions, surface absorption, and the
15
20
different corrections are needed from one installation to
25
30
35
40
amount of interference correction needed.
With reference to FIG. 16, Ta relates to the difference
between the direct path from the speaker to listener and the
longer bounce path also from speaker to listener. To also
relates to the out and return path between the speaker and an
opposite surface inside an enclosure. A wall behind the
speaker can be characterized the same way. Larger Ta gives
a larger distance. RC relates to surface roughness or absorp
tion at high frequencies. Larger RC product for greater loss
or faster attenuation of upper frequency comb filter response
and correction. The control R1 adjusts the magnitude of the
response or correction. CW direction increases subtractive
responses while the CCW position near the +input to the op
amp gives maximum additive responses. The circuit pro
duces an interference response whose amplitude decreases
with frequency. This matches or simulates losses of absorp
tion materials of practical speakers. Much of the irregular
response from small speakers can be experimentally
changed to something that appears to be more easily pro
cessed by the compensation system. Usually, the delay
setting to do this matches the back arrival wave relationship
expected from the speaker enclosure. When it does, this one
adjustable parameter equals a multitude of conventional
response-leveling processes.
FIG. 17 is an all pass or phase shift network. Its frequency
response is flat however its output is in phase and high
frequencies and out of phase at low frequencies. The circuit
18
alters transient response without changing frequency
response. The variable control increases the transition fre
quency as it is turned CW. This element is useful to correct
group delay and other transient related responses.
As previously mentioned, mechanical disturbances pro
duced by speakers are often audible, yet invisible or hard to
interpret from response measurements made using a fre
quency sweep and microphone. Traditional compensation
methods using a deep notch usually result in either a nasal
sound or undesirable balance aberration. The weighted com
pensated notch filter of the present invention solves this
problem and yields some other advantages as well. When
two drivers (woofer and tweeter) are crossed over by just a
capacitor or by circuits that overlap frequencies, one or both
drivers can have interference compensation without percep
tual loss to the other. The same applies to different listening
positions. One position having a bad response can be com
pensated without compromising the sound for other listen
ing positions. The correction is hidden by the weighted side
energy. Since good listening positions are not compromised
by the correction, a wide range of listening positions can
have good sonics. This feature is particularly useful to
horn-type loudspeakers for theater sound, where slightly
45
50
ss
60
65
another. By using default optimization for a class of speakers
or construction features, parametric control and adjustment
is simple and intuitive.
Some of the most important behaviors of loudspeakers
(with respect to acoustically perceptible effect) cannot be
modeled or implemented from traditional methods. Such
behaviors include interference and resonant coupling, as
well as nonlinear consequences from such behaviors. In
addition, other spatial factors related to stiffness of moving
parts and high frequency de-coupling for motions away from
a driving voice coil need to be considered. Any of these can
create source movements, delayed energy release and phase
error to binaural hearing. Often, such destructive responses
can be invisible or very difficult to interpret from traditional
microphone-and spectrum-analyzer calibration methods.
An example of a parametrically addressed compensation
for a specific physical effect is compensation for mechanical
de-coupling in large full-range speakers. Such speakers are
usually designed to have the entire cone move at low
frequencies. At high frequencies, only the inner part of the
driver is the primary active radiator. The rest of the cone is
intentionally de-coupled to attenuate its nodal breakup. This
design choice changes the position ofan equivalent radiation
source. A linear correction to move high-frequency radiation
forward in time uses complex pass filters to create band
limited delay of the lower frequencies. Then, as frequency
increases, a latent delay decreases thereby maintaining a
phase match to the response order of the speaker. Some
group delay distortions can be removed this way. A physical
dimension from the speaker along with attenuation and
speed-of-sound properties of the cone can yield information
to specify a correction from a dedicated filter process.
However, the delay effect and de-coupling frequencies can
be experimentally determined to yield parameter values. If
some of the speaker parts change size or attenuation prop
erties, new values can be extrapolated.
An additional example of a parametrically addressed
compensation, which is difficult or impossible to compen
sate using traditional methods, is compensation for Doppler
modulation in small speakers. Computer and multi-channel
sound systems require small speakers to play very loud. The
resulting combination of fast, high-displacement cone
motions can impart an additive or subtractive velocity to the
US 7,184,556 B1
25
mass, air volume, mechanical compliance, radiating area,
damping, moving mass and motor characteristics.
7. The system of claim 1 wherein the sonic reproduction
device comprises a speaker and at least one of the plurality
offilters comprises at least one associated adjustable param
eter and the value of the parameter is derived from a
standard speaker model.
8. The system of claim 1 wherein at least one of the
plurality of filters has at least one associated adjustable
26
parameter and the value of the parameter is determined
experimentally using standard test measurements.
9. The system of claim 1 wherein the sonic reproduction
device comprises a speaker and the one parameter that
modules the at least one other parameter relates to the
magnet structure and voice coil of the speaker.
Was this manual useful for you? yes no
Thank you for your participation!

* Your assessment is very important for improving the work of artificial intelligence, which forms the content of this project

Download PDF

advertisement