Patent Grant
11127390
This invention relates to arrangements for reducing or cancelling ambient noise perceived by a listener using an earphone. In this application, the term “earphone” is intended to relate to a device incorporating a loudspeaker disposed externally of the ear of a listener; for example as part of a “pad-on-ear” or “shell-on-ear” enclosure or as part of an assembly, such as a mobile phone, which is held close to the ear.
The loudspeaker of the earphone may be coupled to a source of speech or other sounds which are to be distinguished from ambient noise, or the loudspeaker may be provided solely for the reduction of ambient noise, but the invention has special application to earphones used with mobile electronic devices such as personal music players and cellular phones.
At present, some earphones are wired directly to their sound source via short leads and connectors, and some are connected via wireless links, such as the “Bluetooth” format, to a local sound generating device, such as a personal music player or cell-phone. The present invention can be used with both wired and wireless formats.
Existing ambient noise-cancellation systems for earphones are based on one or the other of two entirely different principles, namely the “feedback” method, and the “feedforward” method.
The feedback method is based upon the use, inside the cavity that is formed between the ear and the inside of an earphone shell, of a miniature microphone placed directly in front of the earphone loudspeaker. Signals derived from the microphone are coupled back to the loudspeaker via a negative feedback loop (an inverting amplifier), such that it forms a simple servo system in which the loudspeaker is constantly attempting to create a null sound pressure level at the microphone. Although this principle is simple, its implementation presents practical problems which limit the upper frequency of operation, to about 1 kHz or below. Furthermore, effective passive acoustic attenuation must be provided to prevent the ingress of ambient noise above this 1 kHz limit, and this is done by providing an ear-enclosing circumaural seal, designed to block these frequencies. A recent attempt to improve the performance of feedback systems is described in US 2005/0249355 A1.
Still further, if music or speech is to be fed to the user's earphone, then provision must be made to avoid these wanted signals being cancelled out by the feedback system, and this process can introduce undesirable spectral troughs and peaks into the acoustic characteristic of the earphone. Moreover, a feedback system of this type requires that the operating cavity is substantially isolated from the ambient and, although “pad-on-ear” feedback devices were proposed some twenty years ago, it is believed that no earphones of this type are yet commercially available. Feedback systems are susceptible to go into “howl around” oscillation at switch on or when operating conditions change.
Arrangements in accordance with the present invention thus utilize exclusively the feedforward principle, which is shown in basic form in
In feedforward operation, a microphone A is placed on the exterior of an earphone shell B in order to detect the ambient noise signal. The signal detected by the microphone A is inverted at C and added to the drive signal applied to a loudspeaker D, thus creating the “cancellation signal”. The intention is that destructive wave cancellation occurs between the cancellation signal and the incoming ambient acoustic noise signal, adjacent to the earphone loudspeaker outlet port within the cavity between the earphone shell B and the outer ear E of a listener. For this to occur, the cancellation signal must have a magnitude which is substantially equal to that of the incoming noise signal, and it must be of opposite polarity (that is, inverted, or 180° shifted in phase with respect to the noise signal).
The earphone shell B typically carries a foam pad F, or a similar device, in order to provide a comfortable fit to the outer ear E of the listener, and/or to assist in reducing the ambient noise reaching the listener's ear.
Feedforward ambient noise cancellation is, in principle, simple to implement. A basic working system for use with ordinary earphones can be assembled at very low cost using a simple electret microphone capsule and a pair of operational amplifiers to amplify and invert its analogue signal, prior to mixing with the earphone audio drive signal. This is done via an adjustable gain device, such as a potentiometer, in order to adjust the magnitude of the cancellation signal to equal that of the ambient noise. Some measure of noise cancellation can be achieved with this method, but it is far from perfect. Nevertheless, the feedforward principle forms the basis of numerous earphones which are now commercially available. However, even when the cancellation signal is optimally adjusted and balanced, a considerable residual noise signal still remains, and so it is common to observe that most commercially available systems are only claimed to operate below about 1 kHz, thus providing only a slightly greater bandwidth than that of the feedback method. Bearing in mind that the voice spectrum extends to 3.4 kHz, any associated noise-cancellation system demands a bandwidth well in excess of the capabilities of currently available systems in order, for example, to significantly improve the intelligibility of dialogue via a telecommunications link.
The present invention aims to provide an arrangement capable of achieving significant ambient noise-reduction up to at least 3 kHz.
According to the invention there is provided an ambient noise reducing arrangement comprising a housing, loudspeaker means, supported within said housing, for directing sound energy into an ear of a listener when disposed adjacent an entry location to the auditory canal of the ear; a plurality of microphone means located externally of said housing and positioned to sense ambient noise approaching said entry location; and means for converting the sensed ambient noise into electrical signals for application to said loudspeaker to generate an acoustic signal opposing said ambient noise; the arrangement being such that said acoustic signal is generated by said loudspeaker means in substantial time alignment with the arrival of said ambient noise at said entry location.
By this means, advantage is taken of the time difference between the sensing of ambient noise at the microphone means and its arrival at the entry location to the listener's ear canal to generate a noise-reducing or cancelling signal that is substantially aligned in time with the ambient noise itself as it arrives at the entry point.
In some preferred embodiments, an array of microphone means is provided extending around the perimeter of an ear pad which forms part of a housing for a loudspeaker; the loudspeaker means being disposed within the housing such that there is a known radial distance from the loudspeaker means to each microphone means. In other preferred embodiments, an array of microphone means may be provided around, and radially spaced from, a loudspeaker aperture of a mobile telephone handset. In either event, as will be described in detail hereinafter, the radial path followed by ambient noise from the microphone means to the vicinity of the loudspeaker provides sufficient time for the noise-reducing acoustic signal to be generated such that the required time alignment is achieved.
In particularly preferred embodiments, the relative locations and dispositions of the microphone means and the loudspeaker means relative to incoming ambient noise are chosen to take account of a performance characteristic of the loudspeaker means, so as to ensure the required time alignment.
It is particularly preferred that the microphone means be placed so as to respond, as a whole, substantially uniformly to ambient sound incident from a substantial range of angles.
In some preferred embodiments, at least three, and preferably at least five microphone means are provided to sense incoming ambient noise. Moreover, where such numbers of microphone means are provided, it is preferred that they are disposed substantially equi-angularly around a common locus.
The locus may conveniently carry elements of electrical componentry configured to interconnect the microphone means and/or to convey their outputs to a common location for processing.
The electrical componentry may be provided as a printed circuit, and the processing may comprise combination, phase inversion and amplitude adjustment.
Any or all of the microphone means may be exposed to the ambient noise by way of an aperture and conduit, which may further contain acoustic elements tuned to one or more selected ambient noise features in order to provide enhanced noise reduction in respect of said one or more specific features.
Such acoustic elements as aforesaid may consist of or include Helmholz resonators and/or quarter-wave resonant conduits.
In all embodiments, it is preferred that the acoustic projection axis of the loudspeaker means is in substantial alignment with the longitudinal axis of a listener's ear canal.
In order that the invention may be clearly understood and readily carried into effect, certain embodiments thereof will now be described, by way of example only, with reference to the accompanying drawings, of which:
Prior to describing detailed embodiments of the invention, reference is made, by way of general description, to
When a sound wave SF is incident from the frontal direction, the wave-front arrives at the listener's eardrum G slightly later than at the microphone A because the acoustic path lengths are different, as shown. After travelling through the paths of length X to both the microphone, and also underneath the earphone to a point P of intersection with the longitudinal axis of the auditory canal H, which point lies at an entry location to the canal, the wave must traverse an additional distance Y to reach the tympanic membrane G. The path length Y is approximately equal to the sum of the length of the auditory canal H (typically 22 mm), plus the depth of the concha J (typically 17 mm) plus a small air gap above the ear of about 5 mm, making a total of 44 mm, with a corresponding transit time of 128 μs.
However, if the direction of incidence is from a lateral position (say, 90° azimuth), as shown in
Consequently, there is considerable and significant variation in the relative arrival times of the wave-fronts SF and SL at the microphone A and the point P (and hence the eardrum G), dependent upon the direction of the sound-source relative to the listener; these arrival time differences arising from the difference Z between the two paths.
These time-of-arrival variations can be measured using an “artificial head” system, which replicates the acoustical properties of a human head and ears, provided that a suitable ear canal simulator or equivalent is incorporated into the acoustical structure in order to ensure correct propagation delay measurement to the eardrum position. For example, the disclosure of U.S. Pat. No. 6,643,375 describes one possible measurement system, developed by the present inventor. The measurements are made by mounting a reference loudspeaker at a distance of about 1 meter from the artificial head, which bears the earphone and microphone system, and in the same horizontal plane as the ears, at a chosen angle of azimuth, and then driving a rapid transient wave, such as a 1 ms rectangular pulse repeated at a frequency of 8 Hz into the loudspeaker. This enables the arrival of the wavefronts to be identified accurately by recording, synchronously and simultaneously, the signal from (a) the microphone in the ear canal in the artificial head, and (b) the microphone mounted externally on the earphone shell.
A typical pair of measurements from a centrally-mounted ambient noise microphone fitted to a 50 mm diameter earphone module, which was mounted on to an artificial head and ear system (with canals), are shown in
Since the time-of-arrival difference varies considerably according to the direction of the sound-source, it is difficult to see how time-alignment of any sort can be achieved with this type of arrangement. Even if the system could be made to work for one particular direction, it would be ineffective for all of the other directions.
Additional problems in implementing simple feedforward arrangements of the kind shown in
Turning now to specific examples of the invention, arrangements in accordance with some embodiments of the invention now to be described utilize a distributed microphone array, formed around the perimeter of an earphone shell, casing or pad, in conjunction with a feedforward system for earphone-related ambient noise-cancellation.
Such arrangements enable improved time-alignment of the cancellation signal to the ambient noise signal at the eardrum, by suitably addressing the two critical problems mentioned above in connection with conventional feedforward systems, namely: (a) the considerable variation in ambient noise to eardrum path length owing to changes in sound-source direction and (b) time-lag associated with the electroacoustic transducer. Consequently, the invention provides feedforward-based arrangements which operate to higher frequencies than hitherto possible, and which also are substantially omnidirectional in nature.
As a first step, plural microphones are used to detect the ambient noise, and these microphones are sited to reduce variations in acoustic path lengths with sound front direction. In practice, even the use of only two microphones affords an improvement on the single-microphone configurations used in the prior-art, but preferably three or more microphones are used. In the immediately following description of a preferred embodiment of the invention, an evenly distributed array of five microphones is used, spaced at 72° intervals around the earphone rim.
The earphone capsule 10 comprises a casing 11 which acts as a chassis for the various components, into which a high-compliance microspeaker 12, typically 34 mm in diameter, is mounted with its diaphragm exposed through a protective grille 13 in the lowermost edge, onto which a foam pad 14 is attached in order to lie comfortably against the outer-ear of a listener. Alternatively, for improved acoustic isolation at higher frequencies (>4 kHz), conventional foam-filled leather-skinned annular rings can be substituted for these. The loudspeaker is provided with a rear cavity 15 in order to provide a high-compliance loading, typically several ml in volume, and preferably this is damped using acoustic foam, in order to minimize the fundamental resonance of the loudspeaker 12. Also, preferably, the rear volume is vented to the ambient through one or more apertures such as 16, in order to maximize the rear loading compliance. It is preferred that the vents are spaced away from the microphone inlet ports such as 26 by 10 mm or more.
With pad-on-ear earphones, the earphone units are acoustically non-transmissive, and so each earphone assembly behaves as an acoustic baffle adjacent to, and in contact with, the pinna of a listener's ear. Typically, a thin foam-rubber pad 14, between 3 mm and 6 mm in thickness, is used to cover the surface of the earphone, in order both to provide a comfortable surface for the listener, and to provide some small measure of acoustic sealing between the outer-ear and the ambient. This latter serves three purposes: (a) to increase the low-frequency response of the earphone; (b) to restrict the outward acoustic emissions from the earphones to the ambient; and (c) to reduce the ingress of ambient noise from the environment; although this is less effective at lower frequencies, below about 4 kHz.
The important feature, in accordance with this embodiment of the invention, is as follows. Because the earphone 10 acts as a baffle, the acoustic leakage pathway from ambient to eardrum is forced to traverse one-half of the diameter of the earphone assembly before reaching the entry location at the axis to the auditory canal. Accordingly, by placing the microphones 21 to 25 at or near the rim 20 of the earphone, the ambient noise signal can be acquired and driven to the electroacoustic transducer 12 in advance of its arrival at the eardrum, thus compensating for the intrinsic response time of the electroacoustic transducer 12. Furthermore, this applies to wavefronts arriving from all directions.
For example, and in respect of an arrangement such as that described with reference to
At this stage, for initial clarity of description, the signal path via only one of the microphones (21) will be considered, in order to illustrate and quantify, approximately, the time-delays that are involved.
Referring to
Referring now to
In the foregoing description, the contribution of only a single microphone (21) was considered in order to simplify that stage of the description and to quantify, approximately, the time-delays that are involved. However, it will be appreciated that the process is somewhat more complex. The inventor has observed that, as a wave-front arrives at, and then traverses, the earphone unit, a continuous process of diffraction occurs under the rim of the earphone as depicted in
This phenomenon is direction dependent. If the wave-front comes from a frontal noise source, the acoustic energy is distributed in time related to the period taken for the wave-front to traverse, say, a 60 mm earphone shell, which is about 175 μs. However, if the incoming wave-front is incident normal to the earphone (say from 90° azimuth), then the energy arrives all at once, and it is not so dispersed in time.
Thus, the impulse responses (and associated transfer functions) from the ambient to the eardrum vary considerably with sound source direction, as already shown in
A typical pair of measurements from a 5-microphone distributed array, integrated into a 50 mm diameter earphone module, which was mounted on to an artificial head and ear system (with canals), are shown in
It should be noted that the impulse responses of ambient leakage-to-eardrum (
Conceptually, the total ambient noise leakage into the earphone/outer-ear cavity can be considered to be the sum of a large number of elemental, radial leakage paths, joined at an entry location comprising a central node that is centered on the longitudinal axis through the auditory canal. Thus, the ambient noise signal at the notional center of the radial, elemental leakage paths is the time-varying summation of the elemental contributions after they have propagated from the rim 20 of an earphone 10 to the location P.
If the elemental leakage pathways have similar acoustic impedances, then the ambient noise SPL at the notional center P of the radial elemental leakage paths, after the radial propagation delay, represents the time-varying sum of each SPL at the outer points, around the rim 20 of the earphone, of the elemental leakage paths. This notional, central, ambient noise SPL is what drives the outer-ear and auditory canal, and it is this signal which the distributed ring-microphone array 21 to 25 detects and registers in advance of its occurrence, in accordance with principles of the invention.
The effectiveness of the invention may best be demonstrated by comparing the performance of one of the best commercial, supra-aural noise-cancelling earphones to that of a 5-microphone distributed array of the kind shown in
The shape of the reference response (A), with its large peak at about 2.6 kHz, is caused by the resonant properties of the outer ear and ear canal. With the earphones in place (plot B), the incoming ambient frequencies above 2 kHz are subject to passive attenuation by the foam cushion that partially seals the earphone to the outer ear, as depicted in
In practical terms, in arrangements in accordance with the invention, there is a trade-off between the accuracy of signal-matching (between the cancellation signal and the noise signal) and the chosen number of microphones in terms of cost and complexity. There is also a balance to be sought in terms of the required signal “lead-time” that is required from the microphones, and the physical diameter of the earphone assembly, for it is the diameter of the distributed microphone array that determines this lead-time. The following description is a guide for the practical implementation of the invention in these respects.
In order to achieve correct time-alignment, the time-of-arrival difference between the ambient microphone(s) and the ear canal microphone must be equal (or substantially similar) to the system response-time from the electroacoustic transducer (i.e. the earphone's loudspeaker) to the ear canal. microphone.
Bearing in mind that the respective acoustic pathways share a common path element into the concha and down the auditory canal to the tympanic membrane (shown as Y′ in
The first step is to measure the response time of the chosen electroacoustic transducer for the earphone drive module. If the transducer response time is, for example, 70 μs (a typical value), this corresponds to an acoustic path-length of about 24 mm, and so this mandates that the acoustic centers of the distributed microphone array should be centered, approximately, around a 48 mm diameter circle, or thereabouts.
However, the acoustic paths are not so direct and simple, and it is best to measure the time-of-arrival differences and adjust the radius accordingly, in order to obtain best accuracy. In practice, most transducers that are suitable for this purpose have response times in the range 70 μs to 100 μs, and so distributed microphone array diameters in the range 40 mm to 60 mm are well-suited to these values.
Next, the number of microphones to be used in the array must be chosen. Ideally, of course, a larger number is better than a smaller one, because there might be a risk of some quantization effects if a very small number is used. If we wish to mandate a reasonable criterion that time-alignment of better than 40 μs is desirable (with a corresponding propagation distance of about 14 mm), it is possible to inspect the geometry of a wave passing over a circular microphone array of radius R, and derive a simple, approximate relationship for the effective distance, D, for a transverse wave to pass between the individual microphones, according to their angular separation .theta., as follows:
This indicates that, for a microphone-to-microphone time interval of less than 40 μs (D=14 mm), and if R=30 mm, then θ=.about 60°, and hence 6 microphones should be used. However, this is only a rough guideline.
It is inevitable in all systems of this sort that there is considerable variability in both the acoustic leakage properties and also in the various acoustic path lengths, when the earphone is located in slightly different positions when applied to the listener's ears. This, together with the effects of any small design compromises that have been made, tend to limit the performance of the system, and so the noise suppression characteristic will still feature a “finite” suppression crossover point. However, this is usually observed well above 3 kHz, in contrast to the sub-1 kHz crossover frequencies measured in prior art devices.
The correct orientation of the individual microphones is important but not critical. In order to best represent the SPL at the entrance to the leakage pathways, the microphone inlets (e.g. 26) should be positioned close to the rim edge 20 adjacent the listener's head. This ensures, for example, that the back-diffracted wave at the trailing edge of the earphone (
In terms of defining the microphone array, the most suitable transducers are miniature electret microphones, as will be familiar to those skilled in the art. The inventor has used a variety of sub-miniature electret microphones from various manufacturers, ranging in size from 6 mm diameter x.5 mm length, to 3 mm diameter x.1.5 mm in length. The microphones should have a relatively flat frequency response (±3 dB between 200 Hz and 10 kHz), and the sensitivity variation between microphones should be less than ±3 dB at 1 kHz. (These specifications are typical of the 3 mm diameter x.1.5 mm long microphones used by the inventor.)
In terms of configuring the microphone array electronically, each microphone contains an integral FET buffer amplifier, and therefore an output impedance of only several k.
However, the microphone signals are relatively small (several mV in amplitude), and therefore still require amplification. It is expedient to arrange for a single amplification stage to serve all of the microphones simultaneously, rather than to use separate pre-amplification stages for each microphone, followed by a voltage-summing stage. One way to achieve this is to connect all of the microphones in parallel. However, it is essential in this specific type of construction that all of the microphones are operated in their saturation regions, otherwise inter-modulation will occur, in that the change in current in one microphone would change their common node voltage, which would, in turn, change the current in the integral FETs of the other microphones. For example, in
In order to avoid this inter-modulation phenomenon, the ID/VDs characteristics of the chosen microphone type should be measured, as shown in
A simple, basic embodiment of the invention has already been described with reference to
Another practical embodiment of the invention is shown in
In general reference to the departure of the present invention from conventional feedforward concepts, as discussed with reference to
Previous proposals for feedforward arrangements appear to have been based on the principle that both the incoming ambient noise signal and the signal driven via the earphone loudspeaker undergo various transformations, such as by acoustic resonance in the earphone shell cavity, for example. These transformations were considered to modify the amplitude responses of the signals, and to prevent total cancellation from occurring. However, no similar significance was attributed to the phase of the two signals and it was proposed that, if these various transfer functions were to be combined mathematically, an ideal electronic filter could be created to take account of, and anticipate, all of these effects.
In accordance with the present invention, it will be appreciated that the relative phase of the cancellation signal with respect to the ambient noise signal is attributed at least equal importance with the relative amplitudes of the two signals.
Whilst various prior-art disclosures in respect of ambient noise-cancellation refer to the use of electronic filters to modify the amplitude response, there are no explicit descriptions about dealing with the timing, or phase, response. For example, U.S. Pat. No. 6,069,959 describes a complex filtering arrangement for use with a feedforward noise-cancellation system, and discloses many graphs depicting the amplitude response, but there are no accounts of, or references to, timing or phase response.
There are also some major practical difficulties in implementing the above methods in terms of measuring various transfer functions and then combining them to form the requisite filter function.
The inventor of the present invention considers that the directionality of the above transfer functions is important, and believe that this factor has not been observed previously.
The inventor of the present invention further considers that it is not valid to use a transfer function that has been obtained from a single-angle measurement for use with a diffuse sound-field, as would be predominant in everyday usage.
In light of the poor results of prior-art attempts to improve ambient noise cancellation systems, many have turned to very sophisticated methods, such as the use of adaptive filters. A paper summarising the state-of-the-art and entitled “Adaptive feedback active noise control headset: implementation, evaluation and its extensions” by W S Gan, S Mitra and S M Kuo has been published in IEEE Transactions on Consumer Electronics, 51, (3), Aug. 2005. This approach attempts to analyse and identify the various components of the incoming noise, primarily for repetitive noises, using a digital signal-processor (DSP), and then to modify an electronic filter in real-time to provide an optimal cancellation signal. However, despite considerable mathematical and engineering effort, this approach has met with limited success. For example, the paper “Analogue active noise control” by M Pawelczyk, published in Applied Acoustics, 63, (2002), pp. 1193-1213 includes a review of the state-of-the-art in this area. From FIG. 15 of that paper, it can be seen that the cancellation bandwidth of a state-of-the-art adaptive system is limited to frequencies below about 500 Hz. Also, Pawelczyk notes that such systems cannot suppress impulsive, non-repetitive noise.
Thus it is clear that prior-art disclosures have either omitted or neglected the importance of the phase response of the cancellation signal with respect to the incoming ambient noise signal. Furthermore, the resultant effect of incorrectly matching the amplitudes of the two signals has not been quantified.
In order to discover how sensitive the noise-cancellation process is to variations in amplitude and phase, simultaneously, above and below the optimum values, the inventor has conducted an analysis intended to define the effectiveness of the noise-cancellation process in terms of the remaining amount of (non-cancelled) noise—the “residual” signal—both as fraction (percentage), and also in terms of a logarithmic reduction of the noise sound pressure level (SPL), in dB units.
The somewhat surprising result is to discover the very tight tolerances which are needed for even a modest amount of noise cancellation. If 65% cancellation (−9 dB) is to be achieved (residual signal=35%), the amplitude of the cancellation signal must be matched to that of the noise signal within +3 dB, and, simultaneously, the phase of the signals must lie within +20° (0.35 radian).
The present invention provides an improved ambient noise-cancellation arrangement for an earphone user, which is effective to frequencies up to, and beyond, 3 kHz, in contrast to the sub-1 kHz limit of presently available commercial products. Further advantages of the invention are that it is both comfortable in use, and that the amount of noise-cancellation may be electronically controllable; both of these features being very desirable for use with mobile electronic devices.
In contrast to the various prior-art feedforward signal-processing disclosures, in which emphasis has been placed exclusively on the amplitude response of the signals as a function of frequency, the present invention recognizes the critical importance of the relative phase of the signals.
In contrast to various prior-art methods which incorporate signal-processing based on various fixed transfer functions, each measured from a single, chosen spatial direction, and where it was assumed that these were valid for use with a diffuse sound-field (omni-directional), arrangements in accordance with the present invention accommodate variations in transfer function with sound-source direction, thereby providing an omni-directional, diffuse sound-field noise-reduction or cancellation means.
The invention is based on the new principle that the cancellation signal should be arranged so as to be substantially “time-aligned” with the incoming ambient noise signal at the eardrum of the listener, and provides an arrangement which not only ensures the correct time-alignment of the signals at the eardrum of the listener, but also ensures directional-independent matching of the amplitudes of the two signals.
Following the aforementioned analysis conducted by the inventor in respect of the sensitivity of the residual signal on both the amplitudes and relative phase of the noise signal and the cancellation signal, the conclusion was reached that the correct phase relationship cannot be attained or adjusted by electronic filtering, or by adaptive feedback or adaptive filtering means, and that the only means to achieve the correct phase relationship is to provide a “time-aligned” system. By this, it is meant that the cancellation signal is engineered such that it is substantially time-aligned to the incoming ambient noise signal.
However, this is not straightforward, because the ambient noise signal itself is an acoustic one, not an electronic one, and therefore it is not available for modification using signal-processing means.
At present, and as mentioned previously, the various commercially available active noise-cancellation systems are not effective above 1 kHz, at best, and rely on passive attenuation by their ear-pads to reduce noise ingress above 1 kHz. The second plot of
Although the aforementioned analysis was based upon sinusoidal waveforms, it will be clear that it is also directly applicable to random, non-repetitive waveforms, in the sense that correct time-alignment will result in total cancellation of the noise signal.
Problems also arise with conventional feedforward systems as a result of ignoring the intrinsic time-lag of the electroacoustic transducer. Many assume that the response times of electroacoustic transducers used for earphone applications are virtually negligible, in that the acceleration of the voice coil (and diaphragm) is proportional to the current flowing in the coil (dependent upon applied voltage), and hence that the sound pressure level (force per unit area) is directly proportional to this.
In practice, however, the air which is coupled to the diaphragm presents a complex acoustic load to the diaphragm, in terms of its acoustic inertance, acoustic mass and acoustic resistance. This results in a finite response time which is dependent on many factors. In the inventor's experience, this is usually greater than 70 μs, even for microspeakers of very small diameter (16 mm), and typically about 100 μs for a 38 mm diameter earphone-type loudspeaker.
The response time of a small loudspeaker can be measured by mounting the speaker on to a baffle plate, with a reference grade microphone (B&K type 4003) mounted on-axis to the speaker diaphragm, and very closely, at a distance of about 2 mm. By driving the speaker with a rectangular waveform, as above, an oscilloscope can be used to observe the microphone signal and drive signal synchronously and simultaneously, and measure the rise-time and response-time of the speaker. The propagation delay across the 2 mm separation distance is about 6 μs, and this can be subtracted from the measurements to yield the intrinsic loudspeaker response time. For one 34 mm loudspeaker, used by the inventor, the measured response time is about 76 μs, and hence the intrinsic response time is about 70 μs, which corresponds to a sound wave path-length distance of 24 mm.
This creates a further major conceptual problem for the feedforward system of
In general, the system response-time is the sum of (a) the intrinsic loudspeaker response (described above), and (b) the propagation time from loudspeaker diaphragm to the concha outer edge, then into the depth of the concha cavity, and finally down the ear canal to the microphone at the tympanic membrane position (path Y in
As regards amplitude matching of the cancellation signal to the noise signal, by the time the ambient noise signal reaches the eardrum, it has travelled through a complex acoustic path represented by the various leakage paths between the earphone pad and outer ear, the outer ear cavities and the auditory canal, terminated by the tympanic membrane. This network of conduits and cavities forms, in effect, an acoustic filter that modifies the spectral properties of the noise signal prior to its arrival at the tympanic membrane. Both the frequency response and the phase characteristics are changed, as has been noted in the prior-art. However, the inventor has discovered that, because the earphone/outer-ear acoustic structure is common to both the ambient noise signal pathway to eardrum, and also to the earphone loudspeaker to eardrum, then the spectral modifications that occur to both signals are surprisingly similar. In fact, the inventor has discovered that, provided that the microphones exhibit a reasonably flat frequency response and the earphone loudspeaker also has a relatively flat frequency response, little or no amplitude shaping is required.
This observation is in contrast to some prior-art disclosures, in which signal-processing based on the various frequency domain transfer functions is advocated. Instead, the present inventor uses time-domain methodology.
| Number | Date | Country | Kind |
|---|---|---|---|
| 0601536 | Jan 2006 | GB | national |
This is a continuation of U.S. application Ser. No. 15/695,610, filed Sep. 5, 2017, which is a divisional of U.S. application Ser. No. 13/919,743, filed on Jun. 17, 2013, now U.S. Pat. No. 9,786,264, which is a continuation of U.S. application Ser. No. 12/160,986, filed on Jul. 15, 2008, now U.S. Pat. No. 8,472,636, which is a 371 of International Application No. PCT/GB2007/000120, filed on Jan. 17, 2007, which claims priority to UK Application No. 0601536.6, filed on Jan. 26, 2006.
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| 2000-059876 | Feb 2000 | JP |
| 2000-112483 | Apr 2000 | JP |
| WO 2005052911 | Jun 2005 | WO |
| Number | Date | Country | |
|---|---|---|---|
| 20200020312 A1 | Jan 2020 | US |
| Number | Date | Country | |
|---|---|---|---|
| Parent | 13919743 | Jun 2013 | US |
| Child | 15695610 | US |
| Number | Date | Country | |
|---|---|---|---|
| Parent | 15695610 | Sep 2017 | US |
| Child | 16584190 | US | |
| Parent | 12160986 | US | |
| Child | 13919743 | US |
