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WO1991020165A1 - Systeme de traitement audio ameliore et enregistrements effectues a l'aide de celui-ci - Google Patents

Systeme de traitement audio ameliore et enregistrements effectues a l'aide de celui-ci Download PDF

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Publication number
WO1991020165A1
WO1991020165A1 PCT/US1991/004166 US9104166W WO9120165A1 WO 1991020165 A1 WO1991020165 A1 WO 1991020165A1 US 9104166 W US9104166 W US 9104166W WO 9120165 A1 WO9120165 A1 WO 9120165A1
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WO
WIPO (PCT)
Prior art keywords
signal
phase
signals
generating
band
Prior art date
Application number
PCT/US1991/004166
Other languages
English (en)
Inventor
Martin D. Wilde
William M. Martens
Gary S. Kendall
Original Assignee
Auris Corp.
Priority date (The priority date is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the date listed.)
Filing date
Publication date
Application filed by Auris Corp. filed Critical Auris Corp.
Publication of WO1991020165A1 publication Critical patent/WO1991020165A1/fr

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Classifications

    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04S—STEREOPHONIC SYSTEMS 
    • H04S5/00—Pseudo-stereo systems, e.g. in which additional channel signals are derived from monophonic signals by means of phase shifting, time delay or reverberation 
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04S—STEREOPHONIC SYSTEMS 
    • H04S5/00—Pseudo-stereo systems, e.g. in which additional channel signals are derived from monophonic signals by means of phase shifting, time delay or reverberation 
    • H04S5/02—Pseudo-stereo systems, e.g. in which additional channel signals are derived from monophonic signals by means of phase shifting, time delay or reverberation  of the pseudo four-channel type, e.g. in which rear channel signals are derived from two-channel stereo signals

Definitions

  • the present invention relates to acoustical processing systems and more particularly to processing systems in which the acoustical output from a single sound source is processed to produce a plurality of channels.
  • the processing system could be a stereophonic system in which the sound source is sensed by two microphones whose outputs are then processed to produce left and right channels for eventual playback on left and right speakers or headphones.
  • the output of a single microphone could be electronically processed to produce the left and right channels.
  • the goal of the processing system is to create the illusion of a sound source of a predetermined size located at a specific position relative to the speakers.
  • the perceived locations of the various sound sources generated by the stereophonic signals create for the listener what is known as an acoustic image, i.e., a map of the imaginary physical locations of these sound sources.
  • the apparent location of the sound source is largely determined by the difference in arrival time and the intensity of the relevant component signals generated in the left and right speakers.
  • Shimada U.S. Patent 3,892,624
  • Doi, et al. U.S. Patent 4,069,394
  • R a r (t)-ka j (t) are generated.
  • the apparent distance of the sound source is limited to locations on a line between the speakers.
  • the illusion of a sound source located between the speakers and the listener can not be produced without utilizing additional speak ⁇ ers closer to the listener.
  • the perceived location of the sound source depends critically on the location of the listener relative to the speakers.
  • the component of the acoustic image created by that signal will appear to be located on a line centered between the two speakers. If that signal component arrives fractionally earlier from the left speaker than from the right and/or the intensity of the component from the left speaker is greater than that from the right speaker, its image component will appear to be located left of center. The apparent locations of a set of such image components makes up the composite acoustic image perceived by the listener.
  • listeners When the precedence effect releases, listeners report that the sound image is located in two loudspeakers. When the loudspeakers are separated by a suffi- ciently great distance, listeners report hearing two sound images, one of which being echo-like. As the time delay from the difference in distances to the two loudspeakers increases further, the intensity difference also increases significant ⁇ ly. When the intensity difference is approximately 15 dB, the more distant loudspeaker becomes difficult to hear. At this point, listeners report that the sound image is located in one loudspeaker.
  • a further example of a processing system in which a single sound source is processed for reproduction through a number of loudspeakers is a public address system.
  • a monophonic signal is reproduced through a plurality of loudspeakers to provide a sound field which covers a large area.
  • These systems suffer from problems of a different type. In those areas in which the acoustical signals produced by different loudspeakers overlap, constructive and destructive interference occurs.
  • the particular frequencies at which these different interference patterns occur is determined by the distance from each of the speakers to the location of the listener.
  • the sound field at every point in the room will appear to be filtered by a set of frequency filters whose pass-band frequencies depend on the location relative to the speakers. This is equivalent to timbral shifting the original material.
  • Such added coloration is undesirable, since it reduces the intelligibility of the material being broadcast as well as altering the fidelity of the reproduction.
  • Figure 1 is a block diagram of an audio processing system according to the present invention.
  • FIG. 2 is a block diagram of one embodiment of a phase processor according to the present invention.
  • the present invention comprises an apparatus for audio processing, a method of audio procession and a recording made by said method.
  • An audio processing system according to the present invention generates a plurality of output signals from a sound source input signal.
  • the system comprises circuitry for receiving the sound input signal and for generating a plurality of channel
  • One of said channel signals comprises a signal which is substantially equal to the sum of M band-limited signals, the ith said band-limit ⁇ ed signal having an amplitude substantially equal to that of said input signal in a predetermined frequency range f. ⁇ ⁇ f. and a phase which differs from the phase of said input signal in said predetermined frequency range by an amount
  • the present invention operates on a single input signal to produce two output signals.
  • the output signals may be channels or may be combined with other material to produce the final channels.
  • the manner in which the present invention would operate to produce more than two output signals will be explained in more detail 0 below -
  • the present invention provides its beneficial effects by altering the cross- correlation of the output signals while minimizing any timbral shifts between the input signal and the output signals. 5
  • cross-correlation of two signals, y ⁇ (t) and y 2 (t), is typically measured in terms of a cross-correlation measure which is defined to be the extreme value of the cross-correlation function ⁇ (x), where
  • the cross-correlation measure has a maximum possible value of 1 and a mini ⁇ mum possible value of -1. As will be made clear in the following discussion, it is also important to consider simultaneously both the positive and the negative 5 peaks of the cross-correlation function.
  • the signal input to audio processor 100 is processed by pre-processor 102 to form a plurality of signals which, after further processing, are incorporated into the signals on the output channels.
  • the number of output channels will be denoted by N .
  • pre-processor 102 includes one or more microphones to convert the sound waves to electrical signals.
  • Pre-processor 102 may be as simple as an electrical junction for dividing the input signal into N out signals.
  • Phase-processor 200 converts an input signal x(t) to an phase processed output signal y(t) by altering the phase of various frequency components of x(t) while leaving the amplitude of the signal in the various components substantially unchanged.
  • the output signal is generated by dividing the input signal into M compo- nents, each component matching the intensity of the signal in a specific frequen ⁇ cy band.
  • Apparatus 200 utilizes a plurality of band-pass filters 12 for this pur ⁇ pose.
  • the signal in the ith frequency band is then phase-shifted by an amount ⁇ _. utilizing a phase shifting network 14.
  • ⁇ _ phase shifting network 14
  • the specific ⁇ . values utilized will depend on the particular application in which audio processor 100 is being utilized.
  • the . are provided by controller
  • each of the band-pass filters preserve the phase of the frequency component of x(t) selected by the filter in question.
  • the phase- shifted signals are then summed by signal adder 16 to form output signal y(t).
  • each phase-processor may be subject ⁇ ed to some form of post-processing.
  • post-processing circuits 121-124 are shown in Figure 1.
  • the post-processing in question may include amplifying the signals and/or mixing the signals with other signals derived from other sound sources. For example, additional stereophonic effects may be obtained by amplifying one channel relative to the remaining channels, thereby creating the illusion of a sound source closer to the speaker through which the corresponding output channel is played.
  • the output signal in the ith output channel will be denoted by y.(t). To simplify the following discussion, it will be assumed that there are only two output channels and that delay circuit 107 is utilized in the second output chan ⁇ nel.
  • the cross-correlation measure of the output signals, y ⁇ (t) and y 2 (t) is determined by the phase shifts ⁇ . that were added to the various frequency components of x(t).
  • the ⁇ . are chosen randomly between two limits which will be defined to be P- ⁇ P and P+ ⁇ P, respectively. Since y 2 (t) is merely x(t) delayed by an amount to be discussed below, the ⁇ . are the phase difference between the V j (t) and y 2 (t) in the various frequency bands. Other methods for choosing the phase shifts will be described below.
  • P (modulo 2x) determines the relative balance between the positive and negative peaks in the cross-correlation function.
  • P is equal to zero ⁇ the positive peak is at its maximum (close to 1) and the negative peak is at its minimum (close to O).
  • P is equal to x, the positive peak is at its minimum (close to O) and the negative peak is at its maximum (close to -1).
  • P is close to x/2 or 3 x/2, the positive and negative peaks are approxi- mately of equal magnitude.
  • ⁇ P determines the magnitude of the positive and/or negative peaks in the cross-correlation function.
  • ⁇ P O
  • the magnitude of the peaks in cross-correlation function are at their maximum (close to +/- 1, but dependent on the value of P).
  • the magnitude of the peaks in the cross-correlation function decrease.
  • ⁇ P is equal to x
  • the magnitude of the peaks in the cross-correlation func ⁇ tion are at their minimum (close to zero regardless of the value of P).
  • ⁇ P determines the perceived image width through control of the magnitude of the peaks in the cross-correlation function.
  • the magnitude of P ceases to have any substantial effect in controlling the apparent location of the image between the listener and the speakers. In this case, the sound is perceived as originating from a broad sound source located between the speakers.
  • the present invention may be utilized to control both the image width and distance. P is selected in order to provide the desired image distance.
  • ⁇ P is selected in order to provide the desired image width. This may be accom ⁇ plished by constructing a two-dimensional calibration curve for P as a function of image distance and ⁇ P as a function of image width, wherein the choice of P and SP are also dependent on each other.
  • phase shifts ⁇ . are chosen between the limits specified by P and ⁇ P is important in determining the quality of the output sig ⁇ nals.
  • the ⁇ . are chosen by generating a sequence of random numbers between the limits in question.
  • phase shifts within the range specified by P and ⁇ P, it is important that the phase shifts change direction frequently from band to band.
  • the phase shifts associated with two bands are said to change direction if the signal to the left speaker lags that to the right speaker in the first band while the signal to the left speaker leads that to the second speaker in the second band, or vice versa.
  • phase shifts because of the random manner in which the phase shifts are chosen, there will be cases for which three consecutive phase shifts will be monotonic. However, on average this condition should be met. To better understand the need for this requirement, consider the case in which one wishes to create the illusion of a physically broad sound source emit- ting sound along its surface between the two speakers. A sound component having a positive phase shift will be perceived as originating from a source which is closer to one speaker. A sound component having a negative phase shift will be perceived as originating from a source which is closer to the other speaker. The exact position at which each of the components is perceived will depend on the magnitude of the phase shift in question.
  • the present invention produces a sound "image" that appears to emanate from a source that is made up of a collection of discrete sound components, each emitting sound in a specific frequency band and being located at a different position relative to the speakers. This requirement assures that, on average, signals from contiguous frequency bands will be perceived as originating from non-contiguous sources between the speakers.
  • phase shifts will determine the spatial distribution of sound components. If the phase shift distribution is not uniform in phase, the spatial distribution will not be uniform in space. A uniform spatial distribution is desired since it is found experimentally that such a distribution remains uni ⁇ form when the listener moves from the center line between the loudspeakers to a point off the center line. For example, when a listener is located left of the center line, sound from the left loudspeaker arrives before sound for the right loudspeaker which introduces a time delay in the arrival sound between the two ears. This time delay affects the phase difference at each frequency differently. A uniform distribution of phase provides the greatest assurance that that sound - • * image is not altered by the time delay, since it results in another uniform distri ⁇ bution of phase.
  • the second source of interference will be referred to as spatial interfer- ence.
  • loudspeakers When loudspeakers are utilized to reproduce the channels, the listener will receive overlapping sound fields, each field being generated by a different loudspeaker. At any given frequency, the signals from the two speakers will be perfectly correlated, since they differ only by a phase shift which depends on the frequency in question. Hence, there will be either constructive or destructive interference between the signals depending upon the phase shift in question.
  • the listener is not located on the center line between the speakers, there will be an additional phase shift added at each frequency.
  • the additional phase shift results from the difference in distances between the listen- er and each speaker. For example, if the listener is closer to the right speaker, the signal from the left speaker will be delayed by a time equal to the difference in distance divided by the speed of sound. This time delay is equivalent to a frequency dependent phase shift being added to the output of one of the speak ⁇ ers. This added phase shift changes as the listener moves relative to the loud- speakers.
  • the listener is located in a sound field consisting of the sum of two signals having phase shifts which depend on the location of the listener and sound frequency. These signals will interfere with one another and produce a second timbral shift pattern which depends on the location of the listener.
  • the critical bandwidth depends on frequency, varying from approximately 100 Hz at low frequencies ( ⁇ 2000 Hz) to approximately one seventh the center frequency of the band in question at high frequencies (>2000 Hz).
  • the critical frequency band in question will be made-up of a plurality of sub-bands, each with a different phase shift, ⁇ ..
  • the intensity of the sound in the band will be the average of the intensities of each of the sub-bands.
  • Each sub-band will have an intensity which has been modified by the construc ⁇ tive or destructive interference resulting from the combining of the sound fields from the two speakers. This intensity will vary from 0 to 100 percent of the intensity that would have been present had the interference not taken place.
  • the undesired coloration results when the average intensity from band to band changes as a result of the interferences occurring at the sub-band level in each band. If the sub-bands were so small that there is a very large number of sub-bands in each band, then the change in average intensity from band to band would be negligible.
  • the average intensity of each band is the average of the intensities of each sub-band.
  • the intensity of each sub-band is reduced by a factor which is a function of a randomly selected variable, i.e., the ⁇ r It is well known in the statistical arts that the standard deviation of the average of a function of a random variable for N values of the function goes to zero as N is increased to infinity. Thus, the variation form band to band is reduced as the number of sub-bands is increased.
  • the bandwidth is chosen experimentally between about 50 Hz and twice the critical bandwidth. However, bandwidths as large as 4 critical bandwidths will function. If spatial interference coloration is small, then the larger bandwidth is found experimen- tally to be more desirable. This will be the case when the listener is equidistant from the loudspeakers or wears headphones.
  • phase shifts for each band can also be prescribed. For example, a 5 vivid stereo separation for frequencies below 1,500 Hz may be achieved by selecting a phase value such as +x (or-x) for each band.
  • the phase shifts can be selected by choosing a random phase shift between 0 and fx for the L bands and 0 and -fx for the R bands, where f determines the apparent width of the image. If the image is to appear to emanate from a location be- 10 tween the speakers and the listener, a constant can be added to each phase shift in a manner analogous to that described above.
  • the transformation function h(z) provides the phase shifting of the individual o frequency bands.
  • the present invention preferably utilizes a digital input signal. If the signal source consists of an analog signal, it may be converted to digital form via a conventional analog-to-digital converter.
  • each output signal 5 consists of a sequence of digital values. The ith value for each output signal corresponds to the value of the output signal at a time iT, where T is the time between digital samples.
  • the convolution operation given in Eq. (2) reduces to
  • k runs from 0 to N-l
  • w 2x/N
  • exp( ⁇ ) eJ ⁇
  • N is the total number of frequency samples.
  • only one of the output signals is obtained from the input signal by processing the input signal, the other output signal being identical to the input signal.
  • the output signal that is identical to the input signal can be delayed in time to compensate for the overall delay introduced by the processing. In the case that the processing is performed by convolution, this delay will be approximately equal to half the length of the convolution sequence.
  • the cross-correlation measure value is determined by the relationship of the processed output channel to the unproc ⁇ essed output channel. That is, one of the output channels is not phase-proc ⁇ essed. It is fqund experimentally that the presence of an unprocessed channel reduces the perceived effect of any small timbral shifts. Those skilled in the art will also recognize that the same interchannel relationship can be achieved in an implementation in which both output signals are processed. In such an imple ⁇ mentation, the phase characteristics we have described for the processed signal in the preferred embodiment are implemented such that the interchannel phase differences satisfy the conditions in question.
  • each of the output channels can be processed by a phase-processor according to the present invention.
  • Each phase- processor would utilize its own set of phase shifts, ⁇ .. Each such set of phase shifts would be different from those used by other said phase-processors.
  • the perceptual effects obtained with the present invention are resilient in loudspeaker reproduction, even when the listeners are far off the line equidistant between the two loudspeakers and even when the reproduction environment is reverberant.
  • the effect is present even when the distance between the listener and each of the loudspeakers differs by as much as 15 meters in typical reproduction settings.
  • the output signals provided by the present invention may be played through conventional speakers or headphones. These signals may also be re ⁇ corded onto conventional stereophonic recording media for subsequent play-
  • each sound track can be viewed as being composed of two signals Q(t) and R(t).
  • the Q(t) signal is the result of the processing by the present invention.
  • a 1 (f) gA 2 (f) where g is a gain related constant.
  • the phase of Q j (t) at any given frequency f will differ from that of Q 2 (t) by an amount ⁇ (f), where ⁇ (f) varies between P- ⁇ P and P+ ⁇ P, and ⁇ (f) is a rapidly changing function of frequency.
  • ⁇ (f) is defined to be rapidly varying if the following criteria are met.
  • the frequency spectrum as being broken into bands of width no larger than four critical bandwidths.
  • ⁇ (f) is said to be a rapidly varying c function of f if, on average, the sign of the difference in the average value of ⁇ (f) between bands is zero. If this criterion is met, then, on average, adjacent frequency bands will lead through different speakers when P is 0.

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  • Physics & Mathematics (AREA)
  • Engineering & Computer Science (AREA)
  • Acoustics & Sound (AREA)
  • Signal Processing (AREA)
  • Stereophonic System (AREA)

Abstract

Système de traitement audio dans lequel une pluralité de signaux de sortie (131, 132, 133, 134) sont générés à partir d'un signal sonore par des traitements de ce signal (104 à 107, 121 à 124) visant à produire une pluralité de signaux de la même intensité dans n'importe quelle bande de fréqunce donnée que celle du signal sonore. Toutefois, les signaux en question présentent des relations de phase modifiées par rapport au signal sonore.
PCT/US1991/004166 1990-06-15 1991-06-11 Systeme de traitement audio ameliore et enregistrements effectues a l'aide de celui-ci WO1991020165A1 (fr)

Applications Claiming Priority (2)

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US53854890A 1990-06-15 1990-06-15
US538,548 1990-06-15

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Cited By (3)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
EP1021063A3 (fr) * 1998-12-24 2002-08-14 Bose Corporation Traitement de signal audio
WO2003065761A1 (fr) * 2002-01-29 2003-08-07 Bang & Olufsen A/S Haut-parleur modulaire
WO2015078597A1 (fr) * 2013-11-29 2015-06-04 Harman Becker Automotive Systems Gmbh Production d'un signal audio avec une indication de distance configurable

Citations (3)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US3670106A (en) * 1970-04-06 1972-06-13 Parasound Inc Stereo synthesizer
JPS58190199A (ja) * 1982-04-30 1983-11-07 Nippon Hoso Kyokai <Nhk> 擬似ステレオ方式
US4908858A (en) * 1987-03-13 1990-03-13 Matsuo Ohno Stereo processing system

Patent Citations (3)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US3670106A (en) * 1970-04-06 1972-06-13 Parasound Inc Stereo synthesizer
JPS58190199A (ja) * 1982-04-30 1983-11-07 Nippon Hoso Kyokai <Nhk> 擬似ステレオ方式
US4908858A (en) * 1987-03-13 1990-03-13 Matsuo Ohno Stereo processing system

Cited By (7)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
EP1021063A3 (fr) * 1998-12-24 2002-08-14 Bose Corporation Traitement de signal audio
US6928169B1 (en) 1998-12-24 2005-08-09 Bose Corporation Audio signal processing
WO2003065761A1 (fr) * 2002-01-29 2003-08-07 Bang & Olufsen A/S Haut-parleur modulaire
WO2015078597A1 (fr) * 2013-11-29 2015-06-04 Harman Becker Automotive Systems Gmbh Production d'un signal audio avec une indication de distance configurable
CN105764774A (zh) * 2013-11-29 2016-07-13 哈曼贝克自动系统股份有限公司 以可配置的距离线索生成音频信号
US9855893B2 (en) 2013-11-29 2018-01-02 Harman Becker Automotive Systems Gmbh Generating an audio signal with a configurable distance cue
CN105764774B (zh) * 2013-11-29 2019-01-01 哈曼贝克自动系统股份有限公司 以可配置的距离线索生成音频信号

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