U.S. patent application number 11/650966 was filed with the patent office on 2007-07-05 for microphone-tailored equalizing system.
Invention is credited to Stephen R. Schwartz.
Application Number | 20070154037 11/650966 |
Document ID | / |
Family ID | 22107387 |
Filed Date | 2007-07-05 |
United States Patent
Application |
20070154037 |
Kind Code |
A1 |
Schwartz; Stephen R. |
July 5, 2007 |
Microphone-tailored equalizing system
Abstract
A method and system is described to improve the reproduction of
sound of an acoustic musical instrument. According to one
embodiment, a first microphone is placed at a proximate location to
the musical instrument to pick up the sound of the musical
instrument. The sound as picked up by the first microphone is
compared to a reference sound of the instrument (e.g., the sound of
the instrument as perceived at a normal listening position). Based
on this comparison, a tailor-made equalizer is designed to
compensate for the differences between the sounds as picked up by
the first microphone and the reference sounds of the musical
instrument. Accordingly, using the tailor-made equalizer allows the
reproduction of sound from the first microphone to have a quality
similar to that of the reference sound of the musical instrument.
In an implementation of the above system, a filter arrangement is
provided having a low-pass and a high-pass filter that allows
separate control of the frequency and/or gain for each filter.
Inventors: |
Schwartz; Stephen R.;
(Providence, RI) |
Correspondence
Address: |
KENYON & KENYON LLP
1500 K STREET N.W.
SUITE 700
WASHINGTON
DC
20005
US
|
Family ID: |
22107387 |
Appl. No.: |
11/650966 |
Filed: |
January 9, 2007 |
Related U.S. Patent Documents
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Application
Number |
Filing Date |
Patent Number |
|
|
09072412 |
May 4, 1998 |
7162046 |
|
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11650966 |
Jan 9, 2007 |
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Current U.S.
Class: |
381/118 ;
381/122; 381/61; 381/91 |
Current CPC
Class: |
H03G 5/22 20130101; H04R
29/001 20130101; H03G 3/32 20130101 |
Class at
Publication: |
381/118 ;
381/091; 381/122; 381/061 |
International
Class: |
G10H 1/00 20060101
G10H001/00; H04R 1/02 20060101 H04R001/02; H03G 3/00 20060101
H03G003/00 |
Claims
1. A method for providing a system for high fidelity reproduction
of an acoustic signal comprising: (1) placing a microphone at a
selected location proximate to a musical instrument; (2) playing
the musical instrument to produce sounds as picked up by the
microphone; (3) applying a composite algorithm from a processor
coupled to the microphone; and (4) modifying the sounds picked up
by the microphone.
2. The method of claim 1, wherein the composite algorithm is
derived from at least one reference sound of the instrument.
3. The method of claim 1, wherein the composite algorithm of the
processor compensates for the differences between the sounds as
picked up by the microphone and at least one reference sound of the
instrument.
4. The method of claim 1, wherein the composite algorithm is the
average of a plurality of stored algorithms.
5. The method of claim 1, wherein the composite algorithm is
average of a plurality of stored algorithms of a first
instrument.
6. The method of claim 1, wherein the composite algorithm is
average of a stored algorithm of a first instrument and a stored
algorithm of a second instrument.
7. The method of claim 4, wherein the user may select individual
stored algorithms to average.
8. The method of claim 1, wherein the processor contains at least
one plurality of instrument-specific stored algorithms.
9. The method of claim 4, wherein each of the plurality of stored
algorithms compensates for the differences between the musical
instrument's sound at a normal listening spot and the microphone's
placement.
10. The method of claim 1, wherein the processor further comprises
a predetermined number of electronic filter elements, controls, and
control ranges optimized to compensate for differences in sounds
from the musical instrument compared with corresponding reference
sounds of the musical instrument.
11. The method of claim 4, further comprising generating one of the
plurality of stored algorithms, said generating comprising: placing
a first reference microphone and a second reference microphone at a
reference location; playing reference sounds of a reference musical
instrument same as the musical instrument; comparing with a
processor the signals from the first reference microphone and the
second reference microphone; matching the signals from the first
microphone and the second microphone.
Description
BACKGROUND OF THE INVENTION
[0001] 1. Field of the Invention
[0002] The invention relates to microphone pickup and electronic
amplification of musical instruments, particularly acoustic musical
instruments, for concerts or recordings.
[0003] 2. Background Art
[0004] Pickups for electronically reproducing sound from musical
instruments are of two general types, pressure and vibration. A
pressure pickup or microphone has a diaphragm that vibrates in
response to acoustic pressure variations in air. The diaphragm
vibrations are transformed into an electrical signal. Since the
human ear also has a diaphragm that works in the same way, the
acoustic response of a good pressure type microphone located at an
optimal distance from a musical instrument approximates the sound
of the instrument in a given room. Pressure-type microphones
present problems of isolation, placement, and feedback,
however.
[0005] The isolation problem results from pickup sounds from both
desired sources (the instrument or instruments that one wants to
amplify) and undesired sources (e.g., a cough or another musical
instrument that one wishes to amplify separately). The conventional
approach to minimizing the isolation problem is to place the
microphone close to the selected instrument to be picked up and to
use a so-called directional microphone, which attempts to reject
sound from unwanted directions.
[0006] Since sound radiates by the inverse square law, moving the
microphone closer to the instrument reduces the isolation problem
by increasing the amplitude of sound from the selected instrument
relative to the sound from other sources. This solution, however,
increases the placement problem. Musical instruments generate sound
from different parts, such as the strings, sound box, and front and
back surfaces of a violin. At a normal listening distance from a
musical instrument, the characteristic sound of the instrument is
an amalgam of the sounds generated from each part.
[0007] Different spots in the area close to an instrument
(especially within a foot or so) yield very different sounds, most
or all of which a listener would consider unnatural. When extremely
close (less than a few inches), the differences become so
exaggerated that one spot sounds very different from another, and
it can be difficult to tell what instrument is being listened to.
Also, if the instrument is not stationary, but is held by the
musician (guitar, violin, flute, etc.), small normal movements of
the performer produce unintended and undesired changes in dynamic
level (volume) and tone quality.
[0008] Feedback is a special circumstance arising from isolation
and placement problems, typically during the types of live
performance where performers hear themselves by listening to
monitor speakers aimed in their direction. These speakers are thus
also aimed at the microphone used to pick up the sound initially.
This can create a positive feedback loop that drives the speaker
amplifier into saturation, producing a loud howl. The usual
corrective for feedback is to use directional microphones, but this
is of limited use. As a last resort, vibration pickups attached to
the instruments themselves have been used. These pickups sense
either the vibration of the instrument at the spot where they are
attached (contact pickup) or the vibration of a metallic string
(magnetic induction pickup) of a stringed instrument. As these
pickups do not respond to the sound in air produced by monitor
speakers or other musical instruments, feedback and isolation
problems are greatly reduced. Also, because they are attached to
and move with the instrument, the problems of changing volume and
tone quality caused by a performer's movement are eliminated.
[0009] The drawback to using contact or induction pickups, however,
is that the result is extremely low fidelity. The vibrations of a
string or sounding board of a violin, for example, are drastically
different from the vibrations of the air around the instrument. But
what is defined as the "acoustic sound" of the instrument is what
the ear hears as the vibration produced in the air in response to
the sum of vibrations of all the instrument's parts, as described
above. Thus, these transducers have been very effective in
developing new electric instruments with their own sound
(especially electric guitar and electric bass). However, their
abilities are limited for the high fidelity reproduction of sound
from acoustic instruments.
[0010] For the above reasons, current practice for electronically
transducing and filtering live music from acoustic instruments is
to use a quality directional microphone or microphones set up near,
and aimed at, a single instrument or group of instruments. These
microphones send their signal via a special cable to a special
pre-amplifier (which sometimes sends power to the mic). This then
connects to general purpose equalizer and mixing circuits. For
example, in a rock band a typical drum set (five drums, one hi-hat,
and two cymbals) may have one directional microphone for each drum
and the hi-hat, mounted on a stand very close to the drum, plus two
"overhead" directional microphones for stereo effect and to pick up
the cymbals. The two overhead microphones must be at least a foot
or so from the cymbals to avoid picking up a loud metallic hum. A
guitar may have one or two of these mics placed between one and
three feet away.
[0011] As previously mentioned, it is also common to mount pickup
devices directly on individual instruments, typically guitars, to
produce a different type of sound from that produced by the
conventional "acoustic" form of the same kind of instrument. These
pickups sense the vibration of some part of the instrument, such as
the front soundboard of a guitar. Examples of such pickups are
described in U.S. Pat. No. 4,051,761 of Nylen, No. 4,143,575 of
Oliver, No. 4,423,654 of Yamagami, No. 4,481,854 of Dugas, No.
4,837,836 of Barcus, No. 5,136,918 of Riboloff, and No. 5,206,449
of McClish. The amplified sound from these vibration sensitive
pickups mounted on either acoustic or so-called electric or
electronic instruments differs intentionally from the sound
produced by acoustic instruments and sensed by pressure type
microphones mounted at a distance from the instrument; so these
vibration pickups are not suitable for high fidelity electronic
reproduction of the sound of an acoustic instrument.
[0012] It has been proposed to mount miniature pressure-sensitive
microphones directly on musical instruments for specific purposes.
For example, U.S. Pat. No. 4,837,836 issued to Barcus on Jun. 6,
1989 addresses the drawbacks of using stationary conventional
microphones to pick up musical instruments in general, and also of
holding a standard full size microphone close to, or attaching a
miniature microphone directly to, an accordion or harmonica in
particular. These drawbacks include feedback from nearby speakers
and undesirable emphasis of the sounds coming from a localized
portion of the elongated reed banks of accordions and harmonicas;
that is, an increased volume of the notes whose reeds are near
where the microphone is attached.
[0013] To overcome the drawbacks of the prior arrangements, Barcus
provides a pickup module in which a miniature pressure-type
microphone capsule is embedded. The module has an elongated narrow
sound guide channel extending between oppositely facing open ends,
and a sound sensitive surface area of the microphone communicates
with the central region of the channel. The narrow sound channel
creates a two-lobed directional sensitivity pattern for the
microphone in an attempt to respond more equally to all notes when
the module is centrally mounted on an elongated reed bank of a
harmonica or accordion. Barcus also suggests that the module may be
used with other musical instruments, and specifically that it can
be attached to a drumhead near the edge of the drumhead to avoid
feedback, pickup of an undesirable amount of room ambience, and
lack of presence that occur with a conventional microphone
stationed in front of the drum.
[0014] This lack of presence, a subjective term often used to
describe a characteristic frequency band (which is different for
each sound source), has not been noticed by the present inventor.
However, in trials by the inventor using a variety of shapes, it
has been noticed that strong, unnatural (and unpleasant) sounding
frequency peaks are created by the shape of the cavity surrounding
the microphone. These shaped microphone enclosures invariably add
more problems than are solved in efforts to replicate the
instrument's acoustic sound. The Barcus patent has a chart that
looks like it may show improved-frequency response, but it only
shows improved evenness of volume from note to note. Test results
of the inventor show each note would have a seriously degraded
frequency response when compared to a high fidelity reference.
[0015] U.S. Pat. No. 5,262,586, issued to Oba et al. on Nov. 16,
1993, discloses a sound controller for an acoustic musical
instrument to modify the sound produced by the instrument. In an
example using a piano, the output of a detecting unit having 1)
vibration sensors attached to the bridges and agraffes, 2)
electromagnetic pickup units close to the strings, and 3)
microphones attached to the sound board is delivered to a digital
processing unit. Processors actuated by the various types of
sensors for controlling loudness, delay, equalization, and phase
difference deliver their output to vibration actuators mounted on
the sound board and case boards of the piano. A parameter
determining means adjusts the various processors so that the
actuators create additional vibrations to produce acoustic sounds
with modified qualities. Thus, Oba et al. use microphones as one of
several types of pickups mounted inside a musical instrument to
feed sounds and vibrations generated by playing the instrument to a
vibration unit that alters the acoustic output of the instrument.
Oba et al. do not use these microphones to produce an electronic
signal in response to the acoustic output of the instrument for
recording or amplifying the unmodified output. Consequently, none
of the arrangements of the prior art provides a high fidelity
solution to the problems of microphone isolation and placement
encountered in electronic reproduction of sound from an acoustic
musical instrument. Each case is currently successful only via
careful tailoring by a sound engineer using sophisticated
equipment.
SUMMARY OF THE INVENTION
[0016] The present invention provides a method for designing a
system (microphones, attachment mechanisms, and associated
preamplifiers, equalizers, and processors) to be used with solo or
group musical instruments, and the system as designed by the
method. A principal feature of the system is the use of one or more
microphones placed proximately to, on or inside an instrument. For
example, the microphone can be mounted permanently on or in the
instrument, or it can be attached temporarily to the instrument
with a clip designed for the specific instrument in question. It
may also be held on a stand when feasible and preferable. While any
microphone of suitable quality will do, a miniature microphone (and
particularly when attached to the instrument) has two advantages.
First, it is easier to accurately place, and will go in some places
that a normal microphone will not fit. Second, it will move with a
non-stationary (e.g., hand-held) instrument, and so avoid unwanted
changes of sound quality that arise when an instrument moves
relative to a microphone.
[0017] The system may include a suitably designed microphone preamp
connected in tandem with and closely positioned (less than eight
meters and preferably 3 to 6 meters) to at least one microphone.
The microphone preamp delivers dc power to the microphone (if
needed) as well as receiving, and initially amplifying, the audio
signal from the microphone.
[0018] Another feature of the system is an equalizer unit that is
"tailor-made" for each type of instrument and, more particularly,
for a preselected optimum microphone location on each type of
instrument. The equalizer may include conventional low pass, high
pass, band pass, and/or notch filters, or other processors, as
appropriate. Contrary to conventional general purpose equalizers
having four or more adjustable filters, with up to three controls
for each filter (a total of twelve or more knobs), these units may
have only a minimum number and type of filters needed to compensate
for the differences between the instrument's sound at a normal
listening spot and the microphone attachment spot. Each filter
control can be limited to the smallest useful range that allows
enough flexibility for variations between individual
instruments.
[0019] Each equalizer can be combined with a preamplifier in a
small, lightweight package that can be mounted close to the
performer. This allows the individual musicians to achieve their
own preferred "sound" without needing a skilled audio technician to
make complex multiple adjustments at a master equalizer
console.
[0020] These features of the system sharply reduce the cost of the
audio input equipment for a band or orchestra and dramatically
shorten the time required to set up the equipment for a concert or
recording session. They also enable a musician or other person to
accomplish what is presently achievable only by a sound
engineer.
[0021] As used herein, the term "reference sounds of the
instrument" means sounds produced by the instrument that are
desired to be duplicated in quality by the attached microphone and
tailor-made equalizer. In its simplest form, it means the sound of
an instrument being played and listened to in its normal
environment (but generally exclusive of the room's influence on the
sound). For example, if a guitar player plays a guitar in a
pleasant and dry (non-reverberant) sounding room, the "reference
sounds of the instrument" would be the acoustic signature of the
sound at a good/normal listening position in that room. A second
reference method is when a high quality reference microphone is
used to capture this sound, and a third reference method is when
the microphone signal is recorded on a hi quality storage device
(such as a digital tape recorder).
[0022] When using a microphone reference, the listening site
preferably is spaced from the instrument a sufficient distance to
permit the reference microphone to pick up the optimum sound
quality of the instrument (generally, a distance from the
instrument equal to the width of instrument). This spot also should
avoid the sound of the room. The room itself should be made to
contribute minimally to the sound received at the microphone(s). A
common terminology for this is to say the mic is placed in a
mid-field position, and the room is dry or damped
(literally="discouraged"). An anechoic chamber is an ideal room, as
it would make certain aspects of the design process easier and
perhaps more accurate. However, these rooms are rare and very
expensive, and not necessary to the method.
[0023] Specifically, the present invention provides a method for
designing a system for high fidelity reproduction of the sound of a
selected type of acoustic musical instrument, and also for
providing embodiments of the system, the method comprising:
[0024] (1) placing a first microphone proximately to the acoustic
musical instrument;
[0025] (2) playing the musical instrument to produce sounds as
picked up by the first microphone and playing reference sounds of
the instrument;
[0026] (3) comparing the sounds of the musical instrument as picked
up by the first microphone with the reference sounds of the
instrument; and
[0027] (4) designing a tailor-made equalizer to compensate for the
differences between the sounds as picked up by the microphone and
the reference sounds of the instrument.
[0028] The method of the present invention may additionally
include: [0029] selecting an attachment location in step 1 by
locating the first microphone successively at a plurality of
possible attachment locations that do not interfere with playing
the instrument, [0030] playing the instrument to produce reference
sounds of the instrument, [0031] comparing sounds as picked up by
the first microphone at each attachment location with the reference
sounds of the instrument, and [0032] selecting the attachment
location at which the amplified microphone sound is closest to the
reference sound of the instrument.
[0033] Although the step of comparing the sounds picked up by the
first microphone with reference sounds of the instrument can be
made by listening directly to the two sounds, a preferred
embodiment of the method comprises:
[0034] (1) placing a first microphone proximately to the acoustic
musical instrument;
[0035] (2) positioning a high quality reference second microphone
at an appropriate listening site (normally mid-field, as discussed
above) for the acoustic musical instrument;
[0036] (3) playing the musical instrument to produce reference
sounds of the instrument as picked up by the reference second
microphone;
[0037] (4) making simultaneous first and second audio recordings of
the sounds of the musical instrument as picked up by the respective
first and second microphones;
[0038] (5) comparing the first and second audio recordings to
determine the audio differences between the recordings; and
[0039] (6) designing a tailor-made equalizer for the first
microphone to compensate for the differences of the first sound
recording from the second sound recording.
[0040] The method of the invention may further include repeating
the above steps (1) through (5) using different musical instruments
of the same type to determine adjustment ranges for sections of the
equalizer designed in step (6).
[0041] The fourth step of making simultaneous first and second
audio recordings preferably can include making multi-track
recordings on a digital or other high quality recording medium.
[0042] The fifth step of comparing the first and second audio
recordings preferably includes displaying acoustic waveforms of the
first and second recordings; equalizing one of the first and second
waveforms to substantially conform to the other waveform; and using
the equalization values to design the tailor-made equalizer for the
first microphone in step (6).
[0043] The present invention also provides a system for high
fidelity electronic reproduction of the sound of an acoustic
musical instrument, the system comprising:
[0044] a microphone element or elements;
[0045] microphone attachment devices, where suitable or
advantageous;
[0046] an equalizer having an input coupled to the microphone, the
equalizer including a predetermined minimum number of electronic
filter circuits, controls, and control ranges optimized to
compensate for differences in the electronic reproduction by the
microphone element, of sounds from the preselected type of acoustic
musical instrument compared with corresponding reference sounds
from the type of musical instrument.
[0047] The mounting device may also include a device for removably
attaching the microphone to the instrument, the device being
specifically designed for attachment at the preselected location on
the particular instrument so as to avoid or minimize altering the
sounds produced by the instrument and to enable a performer to play
the instrument unencumbered.
[0048] The equalizer may comprise one or more electronic filter
types, depending on the type of musical instrument. For example, a
tailor-made equalizer for acoustic guitars has a high-pass filter
and two notch (band-reject) filters. A tom-tom drum equalizer has
one high-pass filter and one low-pass filter. A bass drum equalizer
has three high-pass filters and one low-pass filter, with a total
of five controls. Hi-hat and cymbal equalizers have only a
high-pass filter in series with a single notch filter, but with
three controls.
[0049] The above-described and other features and advantages of the
invention are presented in the drawings in connection with the
detailed description of the preferred embodiments.
BRIEF DESCRIPTION OF THE DRAWINGS
[0050] FIG. 1 is a general block diagram of an equalizer system
designed according to an embodiment of the present invention.
[0051] FIG. 2A is a general block diagram of an equalizer for a
miniature microphone attached to an acoustic guitar;
[0052] FIG. 2B is a schematic diagram of the circuit of FIG.
2A;
[0053] FIG. 3 is a block diagram of an equalizer circuit designed
according to an embodiment of the invention for a bass drum;
[0054] FIG. 4 is a block diagram of an equalizer circuit designed
according to an embodiment of the invention for a snare drum;
[0055] FIG. 5 is a block diagram of an equalizer circuit designed
according to an embodiment of the invention for a tom-tom.
[0056] FIG. 6 is a graphic diagram showing how a low-pass and a
high pass filter are combined in series as known in the art.
[0057] FIG. 7 is a graphic diagram showing how a low-pass and a
high-pass filter are combined in parallel according to an
embodiment of the present invention.
[0058] FIG. 8 is graphic diagram of a second embodiment of the
present invention combining a low-pass filter and a high-pass
filter in parallel.
DETAILED DESCRIPTION
[0059] According to the present invention, the microphone assembly
includes a microphone such as an omnidirectional or unidirectional
microphone element, such as a Model No. DPA 4060 microphone
manufactured by Bruel and Kjaer. The microphone element can be
attached to a clip or housing to permit for temporarily attaching
the microphone assembly to a pre-selected spot on or in an acoustic
musical instrument. Other temporary attachment arrangements may be
provided when necessary. The clips or other attachment devices are
selected to minimize mass loading of the instrument structure to
which the microphone is attached. In some cases, permanent
attachment of the microphone assembly to a specific musical
instrument by suitable means may be acceptable. Alternatively, the
microphone can be placed proximately to the musical instrument
rather than being attached to the musical instrument.
[0060] Referring to FIG. 1, a general block diagram of an equalizer
system is shown constructed according to an embodiment of the
present invention. Block 10 represents an instrument such as the
instruments described below. Sounds generated by instrument 10 are
picked up by a microphone 11 (e.g., attached to the instrument).
The sounds picked up by microphone 11 are provided to a microphone
power supply (if needed) and amplifier 12 and a specialized
processor 13 (e.g., a tailor-made equalizer as described below).
The output of the specialized processor 13 is then provided to an
appropriate output device 14 such as a recording device, a public
address (PA) system, a speaker system, etc.
[0061] In practicing the method of the invention for designing a
tailor-made microphone and equalizer system for a specific type of
acoustic musical instrument, the first step is to place a
microphone proximately to the instrument. Alternatively, the
microphone can be attached to a location on or in an instrument of
the selected type. The placement location is preselected with a
view to sensing an airborne acoustic signature that is as little
different as possible from the acoustic signature of the instrument
when heard or sensed at a normal listening site spaced from the
instrument. The term "listening site" refers either to a location
for a human auditor or to a location for placement of a reference
microphone to pick up sound for amplification.
[0062] What constitutes a normal listening site includes how much,
if any, "room sound" (i.e. reverberation) is desired. The sound of
an instrument differs in each room, and even varies significantly
from place to place in a given room. Acoustic engineers designate a
location where the sound of an instrument is augmented by room
sound as the "far field" and a location where the sound of an
instrument is substantially unaffected by room sound as the "mid
field" or "near field."
[0063] In some recording or broadcast situations, with an
instrument ensemble or orchestra in an exceptionally nice sounding
environment such as Carnegie Hall, using microphones in the far
field to involve the sound of the room can produce excellent
results, so long as no sound reinforcement is required. Most often,
however, room effect is undesirable, since a poorly selected
listening site can produce results worse than having no room sound
at all. In an anechoic chamber, or even a heavily draped and
carpeted room, reverberation is essentially eliminated and there is
no room sound, regardless of listening location. To eliminate the
effect of room sound on the sound of an acoustic instrument without
the expense of an anechoic chamber, it has become the norm in both
the sound-reinforcement and recording and broadcast industries to
use unidirectional microphones placed in what is called the mid
field. The microphone is placed far enough away to get the natural
sound of the entire instrument, but as close as possible to avoid
the sound of the room. This distance is generally about the same as
the average dimension of the instrument or a group of
instruments.
[0064] Because unidirectional microphones reject only a portion of
sound coming from unwanted directions, and because some unwanted
sound comes from behind the instrument, a significant amount of
room and other sound may be still picked up at a mid field
listening position. To eliminate this, a microphone is placed in
the near field, spaced about zero to twelve inches away from the
instrument. This is known as "close micing." This strongly
increases the ratio of instrument sound to room and other sound,
because of the inverse square relation of distance from source to
energy of sound. Close micing produces a less natural sound than
listeners are used to, however; even the ears of the musician
playing the instrument are further away than the close mic in most
cases. Various processing equipment is usually needed to improve
the nature of close-miced sound. The present invention improves and
optimizes this, as shown below. It should be noted that for some
instruments, however, particularly drumset components and certain
vocal styles, a "close mic" sound has become the musical norm
because the most common listening experience of these instruments
has been a close-miced recording or performance. If close micing
becomes more widely used, other instruments may have their `sound`
defined this way.
[0065] For most purposes, it is desirable to have a listening room
comparable in equipment and sound characteristics to a professional
recording studio. The following equipment list is given by way of
example: [0066] Unidirectional dynamic reference microphone (RefMic
1)--Sennheiser Model MD441 super-cardioid dynamic; [0067]
Unidirectional condenser reference microphone (RefMic 2)--Neumann
K150 hyper-cardioid condenser; [0068] Omnidirectional dynamic
reference microphone (RefMic 3)--Sennheiser Model MD211 dynamic;
[0069] Microphone preamplifier (Mic Preamp 1)--John Hardy Model M1;
[0070] Microphone preamplifier (Mic Preamp 2)--Symetrix Model 201;
[0071] Parametric equalizer (EQ 1)--Orban Model 621B (four bands
per channel); [0072] Parametric equalizer (EQ 2)--Symetrix Model
SX201 (three bands per channel); [0073] Graphic equalizer (EQ
3)--DOD Model R-231 (one-third octave per band, 31 bands per
channel); [0074] Monitor amplifier (Amp)--Macintosh Model 6200;
[0075] Small near field monitor speaker (Spkr 1)--Rogers Model
LS3/5A, BBC near field reference standard; [0076] Large monitor
speaker (Spkr 2)--Tannoy Dual-Concentric 12-inch, tuned to room
with UREI Model 539 Room Equalizer; [0077] Mixer (Mxr)--Hill Model
B3 24.times.8.times.2 (uses 5532 operational amplifiers); [0078]
Multi-channel audio tape recorder--Alesis Digital Audio Tape (ADAT)
multi-channel recorder, run at 48 kHz; [0079] Real Time Analyzer
(RTA)--Audio Control Model SA-3050A one-third octave with
calibrated microphone (ANSI Class S1.11-1971)
[0080] The above list is not exhaustive, and the choice of
manufacturer and model in each case is not intended to be
exclusive. Other makes and models of comparable or better quality
may be used.
[0081] The choice of a reference microphone depends on the choice
of listening site. Repeated comparisons between a condenser
microphone (RefMic 2) and several dynamic microphones (RefMic 1)
rarely showed significant differences, however, and it is usually
sufficient to use a RefMic 1.
[0082] As stated in the summary of the invention, the term,
"reference sounds of the instrument," means sounds produced by the
instrument that are desired to be duplicated in quality by the
attached microphone and tailor-made equalizer. The quality or
nature of sounds produced by the instrument will be different,
however, for different rooms and for different placements of
instrument and listener in a given room. Thus, the "reference"
sounds used as a standard of comparison necessarily will involve
subjective choice (this is true for all musical reproduction), but
this subjectiveness is minimized by using a high quality (usually
directional) microphone at a proper (usually mid-field) distance
from the instrument in a room with minimal reverberation or other
sound components. An anechoic chamber (rarely available) would be
an ideal place for this purpose.
[0083] The step of playing the musical instrument to produce sounds
as picked up by the first microphone and also reference sounds
requires a skilled musician to play, with consistent volume and
tone quality, a series of notes, chords, and musical phrases, as
appropriate, to produce musical sounds representative of the full
range of the instrument. As the musician plays, each note and chord
is picked up by the first microphone, which may be coupled through
a suitable preamplifier (e.g., Mic Preamp 1 or Mic Preamp 2), a
conventional professional quality equalizer (e.g., EQ 1, EQ 2, or
EQ 3), and an amplifier (Amp) to a monitor loudspeaker (e.g., Spkr
1 or Spkr 2) or to headphones, as desired. The musician produces
the reference sounds either simultaneously or alternately with the
corresponding sounds picked up by the first microphone, depending
on which of several possible ways the comparing step is to be
performed. As indicated above, the reference sound may be the
acoustic sound of the instrument heard at the listening site or may
be picked up by a reference microphone at the listening site that
approximates an average "ideal" acoustic sound. (For comparison
purposes, it may be processed through amplifying and equalizing
equipment similar to that used with the first microphone.)
[0084] The step of comparing the sounds of the musical instrument
as picked up by the first microphone with the reference sounds of
the instrument may be performed in several ways. Each way
preferably entails a skilled audio engineer, or equivalent in
training, making the comparison between the two sounds and
adjusting filters, equalizers, etc. to bring one sound into
conformance with the other. It is well recognized in the audio
engineering art that a skilled engineer or audio technician can
discriminate between the acoustic signatures of similar sounds at
least as well as any currently available audio test equipment. The
following excerpts from articles in "Handbook for Sound
Engineers--The New Audio Cyclopedia," 2nd edition, Glen Ballou, ed.
(1991, H. W. Sams and Co., Div. Of Macmillan, Carmel Ind.) make
this clear: [0085] (Pg. 253) F. Miller: "The very best piece of
test equipment you own is your set of ears and good judgement."
[0086] (Pg. 501) C. Hendrickson: "Some users or evaluators of sound
equipment are actually capable of making judgements merely by
listening to the sound quality of a loudspeaker with music or voice
signals being the input. This is actually a learnable art and
discipline." [0087] (Pg. 1408) D. & C. Davis: "walking the
audience areas while using the most sophisticated analyzer
available, namely the trained ear-brain system, determine the best
areas and the worst areas. Then measure with equipment in the best
areas for reference use and the worst areas for correction
purposes. Watching an engineer place a measuring microphone
relative to a given situation is more revealing than any resume of
his experience."
[0088] In the simplest performance of the comparison step, the
sound signal delivered by the first microphone to either a monitor
loudspeaker or an earphone can be compared directly and
simultaneously with the sound received acoustically by an audio
engineer stationed at an equalizer located at the listening site.
During repeated playings of a specific note or chord, the engineer
adjusts the equalizer to bring the sound from the first microphone
into coincidence with the reference sound heard directly.
[0089] An advantage of this way of comparing the sounds is that the
reference sound is the true acoustic sound transmitted from the
instrument to the listening site, unaffected by translation to and
from electronic form. This way also presents several disadvantages,
however. These include: [0090] If two sounds are played together,
they produce a single combined sound, and thus color each other, so
it is nearly impossible to compare and equalize two simultaneous
sounds. They must be listened to one at a time, which is somewhat
less accurate than a direct comparison back-and-forth. [0091] It
may be desirable, for some applications, to provide a reference
sound that differs from the pure acoustic signal delivered from the
instrument to the listening site. The sound of a drumset and the
sound of certain vocal styles are common examples where the
placement of a microphone combined with the process of electronic
modification has become the musically accepted standard of
sound.
[0092] The preferred way of comparing the two sounds, therefore, is
to simultaneously pick up and record the acoustic sound of the
instrument with a second, reference microphone placed at the
listening site. To obtain a complete record, the instrument should
be played through a succession of notes and chords covering its
full range, along with representative musical excerpts. This entire
process is repeated with several instruments of the same type to be
sure that differences in individual instruments are accounted for
and will work within the final design parameters. The output of the
reference microphone may be passed through a conventional
studio-quality mixer or equalizer bank that is adjusted to create a
compensated reference sound that, when fed to an amplifier and
monitor speaker or headphones, produces an audio output that is
identical, or as close as possible, to the direct unamplified sound
of the musical instrument, or when different, is close to a
reference sound desired in common practice. The equalizer for the
first microphone may next be adjusted, through successive playings
of the instrument, to bring the sound from the first microphone
into conformance with the reference sound.
[0093] The settings of the equalizer for the first microphone then
provide data for designing the tailor-made equalizer in the final
step. Alternatively, after noting the initial settings of the
equalizer for the second microphone, the technician may adjust that
equalizer to bring the reference sound into conformance with the
sound from the first microphone. The change in settings of the
second equalizer then provides the data for designing the
tailor-made equalizer of the invention. As further described below,
this adjustment can also be done automatically.
[0094] The use of a second, reference microphone also permits a
further improvement in the comparing step. Since both the sound
from the first microphone and the reference sound have been
converted to electronic form, they can be simultaneously recorded
on separate tracks of a multi-channel tape recorder (e.g., ADAT).
This has two advantages. First, the test data (notes, chord, and
musical phrases) need be played only once, and then can be repeated
identically again and again from the tape, as adjustments are made
to the equalizer controls. This assures that the same sounds are
being compared each time. Secondly, the sound from the first
microphone and the reference sound can be separated and played back
sequentially, which makes the task of comparing the sounds much
easier. Depending whether the differences are relatively uniform
through the high end or low end of the audio frequency spectrum or
whether they are in one or more relatively narrow frequency ranges,
appropriate high-pass, low-pass, band-pass, or notch filter
circuits can be selected and combined and the component values
determined by a competent technician to achieve the desired
equalizer that is to be tailor-made for the selected type of
instrument.
[0095] FIGS. 2A and 2B show the results of the above-described
design process as applied to an acoustic guitar in which the
miniature microphone assembly was attached by a special clip (not
shown) to the sound-hole. In the equalizer of these figures, an
input gain circuit 60 connects to a high-pass filter circuit 70
that, in turn feeds two band-reject filter circuits in series 110
and 210, which finally connect to an output amp circuit 310. Since
the filters are typical of conventional textbook circuits, no
further explanation of the their operation is needed.
[0096] FIG. 3 shows a block diagram of a tailor-made equalizer,
designed by the method of the invention, for a bass drum. The
output from a microphone (not shown) attached to a selected
location on a bass drum would be fed through a preamplifier (not
shown) to a high-pass filter 32 having an adjustable lower
frequency roll-off of from 16 to 160 Hz. This circuit permits
cutting off the strong low frequency component of the sound from a
drum of this type, which could otherwise saturate the amplifier
system. From high-pass filter 32, the signal passes through
high-pass filter 33 (having a low-frequency roll-off at 10 kHz),
through high-pass filter 34 (with an adjustable low frequency
roll-off between 160 Hz and 12 kHz), and through low-pass filter 35
(having an adjustable high-frequency roll-off ranging from 31.5 to
500 Hz). Block 36, labeled "DRY" denotes a selectable bypass path
around filters 33-35, to allow a comparison with the original
sound.
[0097] FIG. 4 is a block diagram of an equalizer tailor-made for a
snare drum. This equalizer also has a high-pass input filter 42
with an adjustable low frequency filter leading to high-pass filter
43 (with an adjustable low frequency roll-off between 160 Hz and 16
kHz), and a low-pass filter 44 (having an adjustable low frequency
roll-off between 40 and 2 kHz). As in the bass drum equalizer of
FIG. 3, there is a bypass "DRY" path 45.
[0098] FIG. 5 illustrates a tailor-made equalizer for a tom-tom.
This is a simple circuit having a high-pass filter 53 (with a fixed
low frequency roll-off at 3.15 kHz) and a low-pass filter 54
(having an adjustable high frequency roll-off between 40 Hz and 2
kHz). As in the preceding drum equalizers, there is a bypass "DRY"
path 55.
[0099] The foregoing examples of equalizers tailor-made for
specific types of acoustic musical instruments demonstrate the
simple, and therefore inexpensive, solution of the present
invention for providing high fidelity audio reproduction of these
instruments when combined with a microphone proximately placed or
directly attached to the instrument.
[0100] A further advantage of the tailored equalizer system is a
marked increase in the rejection of unwanted sounds, functionally
equivalent to a sharp increase in the directional characteristics
of the microphone. Also, this increase is accomplished without
adding any added coloration (inaccuracies) to the sound. This
coloration is very typical of directional microphones, especially
as they become extremely directional, because they work by using
phase cancellations, which are highly complex and unpredictable.
"Off-axis" response (sound from the unwanted directions) is
particularly problematic. The arrangement of tailored filter
elements according to an embodiment of the present invention allows
a natural sound to be produced from a spot much closer to the
instrument than is usually possible. Because of this, the
inverse-square relationship between distance and energy gives a
marked increase in the ratio of wanted-to-unwanted sound, which is
what a directional microphone attempts to do via phase
cancellation. This increase happens without any additional phase
cancellation, so the coloration of whatever microphone is used
remains constant.
[0101] A further advantage of the method of the present invention
is that it allows the use of omni-directional microphones in
circumstances where previously not feasible. This is advantageous
because an omni-directional microphone is inherently smaller, more
accurate, and easier to make than a similar quality directional
microphone. Even when using an omni-directional microphone, the
amount of rejection provided by the method of the present invention
equals or betters that provided by even highly directional
microphones, in many circumstances. Since omni-directional
microphones are inherently more natural sounding (no phase
cancellations) and can be made significantly smaller
(omni-directional microphones do not need the housing which
directional microphones use to provide the desired cancellation),
the system has the following advantageous effects with an
omni-directional microphone:
[0102] 1--The system allows closer placement of the microphone to
the instrument.
[0103] 2--The placement effects an increase in rejection.
[0104] 3--The increase reduces or eliminates the need for a
directional microphone, and thus allows the use of an
omni-directional microphone.
[0105] 4--The omni-directional microphone can be made smaller than
a directional microphone allowing even closer placement to the
instrument.
[0106] 5--The new placement provides a further increase in
rejection.
[0107] Also, at very close ranges, an omni-directional microphone
is likely to pick up a more accurate acoustic signature than a
uni-directional microphone (because it "sees" more of the
instrument at a close distance). Because the microphone can be
placed closer to the instrument, the signal-to-noise ratio of the
omni-directional microphone with a tailored equalizing system of
the present invention is higher than for a microphone placed
further away from the instrument.
[0108] A system (especially a digital one) can be built that
automatically accomplishes the method of the present invention. The
hardware required is not specific to a particular instrument or
even instrument type (except for the microphone and attachment
mechanism). An embodiment of this method includes the following
steps:
[0109] 1--Place 2 microphones (Mic1=system microphone on or near
instrument, Mic2=reference microphone at a reference location as
previously described).
[0110] 2--Play reference sounds of the instrument.
[0111] 3--Have a processor compare (e.g., via a fast fourier
transform) signals from both microphones.
[0112] 4--Have the processor create a digital filter algorithm
(e.g., FIR) to match the Mic1 signal to the Mic2 signal and store
the algorithm.
[0113] 5--Repeat steps 2, 3, and 4 with different reference sounds
(and store each algorithm).
[0114] 6--Have the processor "average" the algorithms into a final
algorithm (e.g. FIR).
[0115] 7--Apply the final algorithm to a real time processor, such
as a Digital Signal Processor (DSP).
[0116] The foregoing steps can be applied to a specific individual
instrument. To eliminate the need for a user to do this, steps 1-7
may be repeated using several different instruments of the same
type (e.g., several violins), and storing the results of the
several averaged algorithms (in step 6), The results of the several
algorithms can then be "averaged" into a composite final algorithm,
which can be used "off the shelf" by a user not interested in the
effort required to customize the system to a particular instrument.
Since it is useful to provide the same number and range of controls
as has been determined by the empirical version of the method,
these controls would allow the user to tune either version to
taste. The hardware required to run these algorithms is not
specific to a particular instrument or even instrument type (except
for the microphone and attachment mechanism). All that is required
is that there be enough controls (e.g., knobs that are data wheels)
for an instrument which may require the most manual tuning
controls. Thus, various software (e.g., programs, data sets, etc.)
for different instruments and instrument types, and even different
reference sounds for instruments or instrument types, can be
supplied on suitable media (ROM, diskette, etc.). The above
described system of the present invention is a viable
implementation of the method, regardless of whether the algorithms
were derived as in the steps above, or whether an algorithm is
created (separately) to imitate the filter elements derived using
other methods, such as those described elsewhere herein.
[0117] It is advantageous, for both accuracy of result and ease of
use, to be able to separately adjust the two sides of both a
band-pass filter and a band-reject filter. This is because in many
circumstances, the choices one makes about the placement of the two
"sides" of a band-pass or band-reject filter are independent.
[0118] In the case of a band-pass filter in audio work, for
example, where you choose to start rejecting low frequency sounds
(like hum) has nothing to do with where you want to start rejecting
high frequency sounds (like hiss). While a single parametric
band-pass filter will create the shape needed, any single control
will effect both sides; only by countering the change in one
control with a simultaneous appropriate change in another will the
one side that has been set remain steady, so the other side can be
adjusted independently. In practice, this is impossible. Some
available equalizers allow this by providing separate low pass and
high pass filters in series. FIG. 6 shows how the typical
combination of low pass and high pass in series yields a band pass
transfer function.
[0119] Typically a band-reject filter is used to eliminate a single
frequency or a relatively narrow range, such as feedback ringing.
It is also used to reduce a wide, shallow range. In most cases the
rejection is placed around a center frequency conceptually.
However, in instances where a large chunk of mid-range frequencies
are rejected, it is sometimes more desirable to hear and think of
two pass-band regions rather than one stop band. An example is a
snare drum, which has a sound largely made of two portions. The
first is a fairly low skin and shell resonance somewhere below 1
kHz (the sound of the drum with the snares `off`), and the second
is the high end of the metal snares buzzing against the bottom
skin, perhaps above 4 kHz. Trying to tune these 2 independent areas
with a parametric band-reject filter poses the same problems of
interdependence as in the case of the band-pass filter above. In
this case separate high and low frequency controls are more useful
in shaping a signal. FIG. 7 shows how a low pass and high pass may
be used in parallel to create an easily controlled notch
function.
[0120] An added advantage of this arrangement is that separate gain
controls can be added to each of the filters, so that the balance
of the high and low pass bands can be varied. FIG. 8 shows a
circumstance where a snare drum has been tuned to emphasize more of
the high end snare-buzz sound. This is extremely difficult or
impossible with a parametric equalizer.
[0121] Referring back to FIG. 4, a block diagram is shown that
includes an example of this arrangement in a filter designed for a
snare drum. Element 43 is a tunable high-pass filter with a volume
control. It is shown in series with a tunable low-pass filter 44 (a
separate volume control is not shown). The schematic for this is a
combination of straightforward textbook circuits, and is thus not
shown in detail here. It could be imitated on some mixing consoles
by sending the original signal to two separate channels, then using
a tunable hi-pass filter on channel 1, a tunable low-pass filter on
channel 2, and then mixing the results. This is a very cumbersome
process, and the nature of the acoustical signals would have to be
well understood to begin with for the user to attempt to try.
According to an embodiment of the present invention, this circuit
teaches the user to obtain these superior results without the need
to understand any of these issues beforehand.
[0122] It is true that some digital equalizers (especially on
sophisticated computer programs and synthesizer-workstations) allow
the `drawing` of virtually any shape of filter or filter
combination. But these do not `teach` the use of any proper
combination and control of elements for any situation, and thus do
not specifically imply what is accomplished here. In a digital
implementation of the current invention, this special filter
combination would be made available, with proper control
parameters, as part of a `preset` to the user for whatever
circumstance was requested.
[0123] As stated above, a further advantage of the present
invention is a large increase in signal-to-noise ratio obtained by
the close micing made possible by the invention. The noise floor
(inherent noise) of electrical equipment (microphones, amplifiers,
equalizers, etc.) is constant. As the method allows the microphone
to be placed closer to the sound source than other methods, the
signal level becomes appreciably higher, whereas the noise of the
components stays constant. This yields a significantly improved
signal to noise ratio. In particular, it allows the use of a
microphone and associated microphone pre-amp (and other associated
electronics) that can have a much higher noise level than is
usually tolerated using other methods. This allows the microphone
itself to be made smaller and more cheaply (cheaper and perhaps
fewer electronics need be included in the microphone capsule
itself), and allows for less rigorous specifications for all the
equipment described herein.
* * * * *