Audio amplification apparatus

0001The present invention relates generally to the audio amplification apparatus for processing sound signals, and in particular to a procedure for fitting of audio amphfication apparatus for use by a listener. The present invention is suitable for adjusting audio amplification apparatus such as hearing aids, and it will be convenient to describe the invention in relation to that exemplary, non-limiting application.

0002<patcit id="pcit0001" dnum="US6201875B"><text>US patent number 6201875</text></patcit> discloses a method for fitting a hearing compensation device. The method comprises selecting a plurality of loudness levels for a plurality of frequencies and comparing each loudness level for each frequency for perceived sameness The loudness levels may then be adjusted as needed to achieve perceived sameness across the frequency spectrum A gain curve for each frequency is calculated from the selected plurality of loudness levels

0003Australian patent number <patcit id="pcit0002" dnum="AU2002300314"><text>2002300314</text></patcit> discloses a frequency transposition arrangement which is adapted to be used in a wearable hearing aid An input sound is processed to produce FFT outputs which are used as the basis for determining the frequencies and amplitude of a set of oscillators Frequency transposition may be performed by altering selectively the frequencies used for the oscillators. Each oscillator in a practical implementation may correspond to the outputs of more than one FFT bin.

0004<nplcit id="ncit0001" npl-type="s"><text>KAMERLING S JANSE K VAN DER MEULEN F: "A new way of Acoustic Feedback Suppression" PREPRINTS OF PAPERS PRESENTED AT THE AES CONVENTION, vol 104, no. 4735, 16 May 1998</text></nplcit>, XP002370645 discloses several systems for acoustic feedback suppression in auditoriums or the like. One system described includes a decorrelator stage and adaptive filter prior to amplification.

0005A theoretical model 1 of a hearing aid is shown in <figref idref="f0001">Figure 1</figref> An external signal received by the hearing aid is amplified along a forward transfer path 2 to provide a signal to an output transducer. The hearing aid amplifier has a forward transfer function H. Feedback in the hearing aid occurs when the acoustic signal from the output transducer finds its way back to the input transducer of the amplifier A feedback path 3 is shown and includes a transfer function G of all combined feedback paths. The feedback signal is added to the forward transfer path 2 in the theoretical model 1 by a summation device 4

0006In hearing aids and like audio amplifiers conforming to this model, feedback can result in audible whistling or howling Under these conditions, the closed loop gain of the amplifier is unstable and approaches infinity at the frequency where certain phase and gain requirements are met. In order to address this problem, a variety of feedback suppression systems have been proposed

0007In one such system, a frequency transposition element is introduced into the audio amplifier to shift the frequency of the input sound signal either upwardly or downwardly, in addition to amplifying the signal, before sending it to the output transducer.

0008One such frequency transposing amplifier is described in pending Australian Patent Application No <patcit id="pcit0003" dnum="AU2002300314"><text>2002300314</text></patcit>, filed 29 July. The manner in which the frequency transposing amplifier operates is illustrated by the theoretical model 5 shown in <figref idref="f0001">Figure 2</figref>. In this model, a frequency transposing element 6 has been added to the forward transfer path 2 of the audio amplifier The frequency of the amplified external signal is transposed to a different frequency. The output, and hence the feedback signal, is at a different frequency from that of the external input signal so that successive summation of a signal at the microphone input at a particular frequency cannot occur. The introduction of such a frequency shifting component can make the closed loop gain of the system stable and avoid spontaneous oscillation under certain conditions.

0009Whilst a frequency transposing amplifier may be stable in terms of its closed loop gain, the amplifier output may be increased to a level which causes unwanted artefacts. Such artefacts are introduced when the output of the amplifier is at a sufficiently high level so that attenuation of the signal via the feedback path results in a feedback signal of sufficiently high level to cause distortion when added to the incoming signal. A feedback suppression amplifier that addresses this problem is described in co-pending Australian Patent Application No <patcit id="pcit0004" dnum="AU2003236382"><text>2003236382, filed 20 August 2003</text></patcit>. The feedback suppression system described in that co-pending patent application acts to remove or compensate for the presence of feedback signals at transposed frequencies in the closed loop system shown in <figref idref="f0001">Figure 2</figref>.

0010In that feedback suppression system, the presence of an undesired feedback signal component resulting from the amplification and frequency transposition of an input sound signal is predicted, and a correction applied to the output signal at each of the transposed frequencies to compensate for the presence of the undesired feedback signal component if the output signal level is greater than a predetermined activation level. The present application should be read in conjunction with Australian Patent Application No <patcit id="pcit0005" dnum="AU2003236382"><text>2003236382</text></patcit>.

0011The frequency-dependent amplification of any hearing aid, including frequency transposing hearing aids, needs to be adjusted to suit each individual wearer's hearing loss. The fitting procedure required to make such adjustments is a time consuming and often inaccurate process, requiring multiple attempts before the hearing aid is adjusted to suit each individual's needs. In particular, it is difficult to adjust the frequency dependent amplification of a hearing aid so that input signals having levels similar to the average level of speech signals are amplified such that they are perceived as equally loud across a range of frequencies by the hearing aid wearer.

0012Moreover, in hearing aids using selectively operable feedback suppression systems, the calculation of the activation levels to determine when output signal correction is required is frequency dependent, and is affected by the characteristics of the acoustic feedback path of the hearing aid when it is worn by a user. Once again, the determination of suitable activation levels is a difficult and time consuming task, and may require several attempts to achieve a result acceptable to a hearing aid user.

0013<patcit id="pcit0006" dnum="US6201875B"><text>US 6201875 to Davis et al.</text></patcit> discloses a method of fitting a hearing compensation device (e.g. a hearing aid) which involves a user selecting a plurality of loudness levels for a plurality of frequencies and comparing each loudness level for each frequency for perceived sameness. A gain curve for each frequency is calculated from the selected plurality of loudness levels.

0014<patcit id="pcit0007" dnum="US4471171A"><text>US 4471171 Koepke et al.</text></patcit> discloses a method of matching the response of a hearing aid to individual's requirements of a user which implements a form of subjective audiometry in which the user is asked to select a point when they can perceive a sound of a given frequency. This data is then used to set a correction factor at that frequency to be used by the hearing aid.

0015It would be desirable to provide a method of adjusting an audio amplification apparatus such as a hearing aid in a manner that ameliorates or overcomes one or more disadvantages of known hearing aid adjustment techniques.

0016It would also be desirable to provide a method of adjusting frequency dependency amplification in an audio amplification apparatus that ameliorates or overcomes one or more known disadvantages of the prior art.

0017It would furthermore be desirable to provide a method of determining activation levels for feedback suppression in an audio amplification apparatus that ameliorates or overcomes one or more disadvantages of the prior art.

0018One aspect of the present invention provides a method of adjusting frequency-dependent amplification in an audio amplification apparatus, as given in claim 1.

0019The predefined loudness perception level of the listener may be selected from the group of: very soft, soft, comfortable but slightly soft, comfortable, comfortable but slightly loud, loud but OK, uncomfortably loud and extremely uncomfortable. In one embodiment, the predefined loudness perception of the listener level is comfortable but slightly soft.

0020The predefined loudness perception level of the listener may be determined by: <ul id="ul0001" list-style="none"><li>the listener choosing a descriptor of the stimulus level at each frequency; and</li><li>adjusting the stimulus level until the chosen descriptor meets the predefined loudness perception level of the listener.</li></ul>

0021Alternatively, the predefined loudness perception level of the listener may be determined by: <ul id="ul0002" list-style="none"><li>presenting a reference stimulus to the output transducer at one of the plurality of frequencies, the stimulus level meeting the predefined loudness perception level of the listener;</li><li>presenting stimuli to the output transducer at the other frequencies; and</li><li>adjusting the stimulus level at the other frequencies to match the listener's loudness perception level to that of the reference stimulus.</li></ul>

0022The magnitude of the input signal component at each frequency may be the level of international long-term average speech spectrum (ILTASS) at an overall level of 70 dB SPL at that frequency.

0023Another aspect of the invention provides an audio amplification apparatus as given in claim 7.

0024A still further aspect of the invention provides an audio amplification apparatus comprising a signal processing device, the device acting to receive digitised sound signals and generate output transducer signals, the sound processing device including: <ul id="ul0003" list-style="none" compact="compact"><li>processing means for use in performing a method according to any one of the preceding claims.</li></ul>

0025Conveniently, the processing means may be implemented in digital signal processing (DSP) technology.

0026In the drawings: <ul id="ul0004" list-style="none"><li><figref idref="f0001">Figure 1</figref> is a schematic diagram illustrating a model of an acoustic amplification device including a forward transfer path and a feedback path;</li><li><figref idref="f0001">Figure 2</figref> is a schematic diagram illustrating a model of an acoustic amplification device using frequency translation to minimise the effect of feedback;</li><li><figref idref="f0002">Figure 3</figref> is a schematic diagram of an embodiment of a sound processing device using frequency translation in accordance with one embodiment of the present invention;</li><li><figref idref="f0003">Figure 4</figref> is a flow chart showing functional steps performed by part of the sound processing device of <figref idref="f0002">Figure 3</figref>; and</li><li><figref idref="f0004">Figure 5</figref> is a schematic diagram illustrating a model of an acoustic amplification device and a stimulus generator for use in the fitting of the acoustic amplification device to a user;</li><li><figref idref="f0005">Figure 6</figref> is a graphical representation of the stimulus signal components and ILTASS standardised input signal levels from the acoustic amplification device shown in <figref idref="f0004">Figure 5</figref> together with derived amplifications;</li><li><figref idref="f0006">Figure 7</figref> is a schematic diagram illustrating another model of an acoustic amplification device for use in the fitting of the acoustic amplification device to the user for determining frequency-dependent amplifications during fitting; and</li><li><figref idref="f0007">Figure 8</figref> is a graphical representation of the feedback signal components, amplifications, stimulus levels and hearing threshold levels from the audio amplification device shown in <figref idref="f0004">Figure 5</figref> as used to determine activation levels used in the flow chart shown in <figref idref="f0003">Figure 4</figref>.</li></ul>

0027Referring now to <figref idref="f0002">Figure 3</figref>, there is shown generally a sound processing device 10 in which input signals from a microphone are sampled, converted to a digital representation, and then periodically subject to a windowing operation followed by a Fast Fourier Transform (FFT). The result of the FFT is analysed to estimate the magnitude and phase of each frequency component of the input signal. The magnitudes are processed to produce amplitude control signals which are assigned to a number of oscillators. These oscillators are tuned to appropriate frequencies using information derived from the changes over time in the phase estimates. The final output signal is constructed by summing the output signals for the oscillators, and subsequently converting the composite signal from digital to analogue form. The composite output signal is then conveyed to a suitable transducer, such as the earphone (receiver) of a hearing aid.

0028In more detail, an input sound signal received at a microphone 11 is preamplified and filtered to limit its bandwidth in the preamplifier and anti aliasing filter. An analogue to digital converter 13 samples the band limited signal at a constant rate and converts the sampled signal into digital form. In the exemplary implementation of the present invention, a block of sequential input samples is placed in the memory of a suitable digital signal processing (DSP) unit. These samples are windowed by a windowing block 14 which multiplies each sample by a corresponding coefficient. Various windowing functions defining suitable sets of coefficients have been described in the literature readily available to those skilled in the art. The purpose of the window is to ensure that the subsequent FFT operation performed by an FFT block 15 produces an acceptable estimate of the short term spectrum of the input signal without noticeable distortion or other undesirable side-effects.

0029A 256 point window with coefficients defined by the product of a hamming window and a mathematical sinc function is suitable when an input sampling rate of 14.4 kHz is used. The window of outputs are stacked and added (using a standard numerical operation known as folding) to produce a set of windowed input samples. This set of data is then processed by the 128 point FFT block 15.

0030The FFT and subsequent processing performed by the sound processing device of <figref idref="f0001">Figure 1</figref> are executed every time a new set of 32 samples has been obtained from the input transducer. Thus, with the sampling rate of 14.4 kHz, the FFT and subsequent processing steps are repeated at intervals of approximately 2.2 ms. However, it will be appreciated that differing sampling rates, different types and links of the window function and Fourier transform, and different extents of FFT overlap may be envisaged.

0031The outputs of the FFT block 15 comprise a set of complex numbers which together represent approximately a short term spectrum of the input signal. With a 128 point FFT, the first 64 bins contain spectral estimates covering the frequency range of zero to 7.2 kHz, approximately (for a sampling rate of 14.4 kHz). Ignoring the first and last of these bins, which generally do not contain signals of interest in the present exemplary hearing aid implementation of the sound processing device, the remaining bins each provide information about a substantially contiguous sub band of the input frequency range, each bin extending over a bandwidth of approximately 112.5 Hz. For example, the first bin of interest contains a complex number which describes the real and imaginary components of the input signal-within a bandwidth of approximately 112.5 Hz centred on a frequency of 112.5 Hz. The power of each component of the input signal is estimated for each frequency bin by summing the squares of the real and imaginary parts of the complex estimate.

0032A well known deficiency for the FFT for spectral analysis in general is that the output bins are spaced at constant frequency intervals (e.g. 112.5 Hz in the present case, and have a constant band width, e.g. approximately 112.5 Hz). For the purposes of frequency transposition as outlined above, it is desirable to obtain a more precise estimate of the frequency content of the input spectrum than is possible using FFT alone, especially at relatively low frequencies. As is described in co-pending Australian Patent Application No <patcit id="pcit0008" dnum="AU2002300314"><text>2002300314</text></patcit>, filed 29 July, this can be achieved by making use of information contained in the phase value represented in each frequency bin at the output of the FFT block 15. This extension of the standard FFT process is embodied in an algorithm described as a phase vocoder.

0033Firstly, the phase angle is estimated by calculating the inverse tangent of the quotient of the imaginary and real parts of the complex number in each FFT bin. A look-up table is provided containing the pre-calculated tangents of a relatively small number (e.g. 64) of phase values. This table contains discrete samples of the range of possible phase values over any two quadrants (e.g. for phase values between -π/2 and +π/2 radians) These values correspond to the case where the real part of the complex number from the FFT bin is positive. If the real part is in fact negative, it is firstly treated as positive, and later the phase estimate is corrected by adding an appropriate constant to the phase angle initially calculated

0034The phase value for each FFT bin is estimated by a process of successive approximation. A starting value for the phase angle being sought is selected, and the tangent of that value is obtained from the look-up table. The tangent of the candidate phase value is then multiplied by the imaginary part of the complex number in the FFT bin The product is then compared with the corresponding real part, and the candidate phase value is adjusted up or down according to the difference between the estimated and actual real part.

0035Next the new candidate value is used to obtain the corresponding tangent from the look-up table This process is repeated until the candidate phase value has the desired accuracy. It has been found that adequate precision can be obtained with a 64 entry look-up table encompassing a phase range of -π/2 to +π/2. Because multiplication and table look-ups can be carried out very rapidly and efficiently in current DSP devices, the above described algorithm is particularly suitable for use in a wearable, digital hearing aid.

0036To use the phase estimates to improve the resolution of the frequency analysis provided by the FFT, the rate of change of the phase in each FFT bin over time is estimated This is because the rate of phase change in a particular bin is known to be proportional to the difference in frequency between the dominant component contained in that bin and the nominal centre frequency of the bin. In this implementation, the rate of phase change for each bin is calculated by subtracting the phase estimates obtained from the immediately previous FFT operation from the current phase estimates. Phase differences are accumulated over time, and then multiplied by a suitable scaling factor to represent the frequency off-set between the input signal component dominating the content of each FFT bin and the corresponding centre frequency for that bin It will be appreciated that alternative processes to determine the phase estimates may be used, for example, a direct calculation process.

0037The processing described thus far results in a set of power estimates representing the square of the magnitude spectrum of the input signal, and a set of precise frequency estimates representing the dominant components present in the input signal. These sets comprise one power value and one frequency value for each FFT bin. These sets normally contain 62 power and frequency values assuming that a 128 point FFT is employed.

0038In the present example, a bank of 24 oscillators is used in the sound processing device 10. In <figref idref="f0002">Figure 3</figref>, the bank of oscillators is indicated by the reference 21. The information contained in the 62 FFT bins is reduced to 24 bands in the reduction block 16, with each band assigned to a corresponding oscillator. The frequency range covered by the 24 bands are normally, but not necessarily, contiguous. The reduction of the FFT bins to a smaller number of bands may be accomplished in various ways. One practical technique is to exploit the fact that less frequency resolution is generally needed in an assistive hearing device at high frequencies than at low frequencies. Thus the contents of several relatively high frequency FFT bins can be combined into a single processing band. The combining operation is performed by summing powers of the FFT bins, and by obtaining the required precise frequency estimate from only one of the combined bins. The bin selected for this purpose is the one containing the highest power out of the set of combined bins. For low frequency FFT bins, each bin is usually assigned separately to a corresponding band for further processing.

0039The outputs of each of the 24 bands are then analysed by a frequency estimation block 17 and a magnitude estimation block 18 to derive an estimate respectively of the frequency and magnitude of each of the 24 bands of the input signal. The frequency estimation is derived from phase information provided by the reduction block 16.

0040Frequency and magnitude data for each analysis band are provided to a frequency transposition block 19 and magnitude processing block 20. Each of the 24 oscillators in the sound processing device 10 generates a sine wave that can be controlled in both amplitude and frequency. The desired amplitude is determined by the magnitude processing block 20 from the magnitude data for the corresponding band. The conversion between the power value and the desired oscillator amplitude may be specified by a look-up table or calculated from an appropriate equation. Accordingly, any desired amount of amplification or attenuation of the input signal may be achieved at each frequency (i.e. within the frequency range associated with each band).

0041The desired oscillation frequency of each oscillator is set by the frequency translation block 19 and may be specified by a look-up table or calculated from an equation. For example, if no change to the frequencies present in the input signal is required, each of the oscillators is merely tuned to generate the same frequency as that estimated from input signal in the corresponding band as determined by the frequency estimation block 17. However, if frequency translation is required to be formed by the frequency translation block 19 (for example, lowering of one or more input frequencies by 1 octave), then the frequency estimated from the input signal in each band is multiplied by an appropriate factor (for example, 0.5) before applying it to tune the corresponding oscillator. It should be noted that both the amplitude control and the frequency control for each oscillator can be specified completely independently of the operation of all other oscillators. Thus it is possible to lower some input frequencies and not others, or to lower each input frequency by a different amount. It will be appreciated that it is also possible to raise input frequencies in the same manner.

0042Accordingly, amplitude control signals are provided from the magnitude processing block 20 to each of the 24 oscillators in the bank of oscillators, whilst frequency control information is provided from the frequency translation block 19 to that same bank of oscillators.

0043The composite output signal is produced by summing the output signals from the bank of all 24 oscillators. The composite signal is then converted to analogue form by the digital to analogue converter 22 and amplified by amplifier 23 to drive a suitable transducer 24 (such as the earphone of a hearing aid or other receiver).

0044Feedback artefacts resulting from the frequency translation carried out in the sound processing device 10 are compensated for or removed. Given the input to output frequency mapping employed by the sound processing device 10, it is possible to predict the frequency of the feedback signal produced by any given external signal. The time delay between the original external signal and its corresponding frequency lowered feedback signal can also be accurately predicted and is directly related to the signal processing delay of one complete loop around the system.

0045The output signal level at the input frequency of each of the 24 bands is accordingly monitored by a feedback prediction block 25 to determine if it is above or below a predefined activation level. If the output signal level is above the activation level, a feedback correction block 26 computes the difference between the output signal level and the predetermined activation level in terms of acoustic power. In alternative embodiments of the invention, the difference may be computed in terms of decibels.

0046In the transposed frequency computation block 27, the transposed frequency at which the undesired feedback signal component will occur is calculated, and the calculated difference is used to effectively "correct" the output signal at that transposed frequency to compensate for the presence of the undesired feedback signal component. In the context of the present invention, "translation" is to be understood as encompassing any form of frequency modification including, for example, frequency shifting, frequency compression and any shift in frequency from a first to a second value.

0047The activation level is an estimate of the output signal level which will result in a feedback signal which, when amplified and transposed, will be audible or otherwise create a perceptual disturbance to the listener. A set of activation levels are required by the feedback detection block 25 to activate the feedback suppression at the frequency of each of the 24 bands. The characteristics of the feedback path may be different for each situation, and may change over time. Accordingly, the activation levels may be fixed or may be adaptable to change according to changes in the characteristics of the feedback path over time.

0048<figref idref="f0003">Figure 4</figref> illustrates in more detail the operation of the sound processing device 10 during suppression of an undesired feedback signal component resulting from frequency translation. At step 30, a first frequency of an output signal intended to drive one of the oscillators in the bank is analysed. At step 31, the output signal level at that output frequency is compared with the activation level. If the output signal level is below the activation level, there is no need to perform any feedback suppression at that frequency, and processing moves on to the next output frequency. If however, the output signal level is above the activation level, the difference between them is calculated at step 32 in terms of acoustic power. At step 33, the transposed frequency of the undesired feedback signal component is computed using input to output frequency mapping. This computation determines the frequency at which the undesired feedback signal component is effectively applied as an additional input signal to one of the oscillators in the bank.

0049In step 34, at the computed transposed frequency, the feedback correction value is subtracted from the output signal level after an appropriate delay dependent on the processing delay of the amplifier. At step 35, a determination is made as to whether all output frequencies have been analysed, and if so, processing is continued by other elements of the sound processing device 10 at step 36. The quantity that is subtracted from the output signal level is best done in terms of acoustic power (squared linear amplitude). However, due to programming efficiency, it may be more advantageous to perform computations in terms of decibels in some situations, for example when the total signal level is not greatly above the audibility threshold at the expected feedback frequency.

0050If the activation level is set to low, feedback suppression will cause the amplifier to reduce the output level at a given transposed frequency, even when no feedback signal is present. This may result in a reduction of the wanted signal even if there was one present at that frequency. If the activation level is set to high, feedback artefacts will be present at the transposed frequency, and may be audible.

0051In the described embodiment, the undesired feedback signal component is subtracted from the output signal at each of the transposed frequencies to compensate for the pressure of the undesired feedback signal component. However, it will be appreciated by those skilled in the art that in alternative embodiments, the undesired feedback signal component may be subtracted from the input sound signal, prior to amplification and frequency translation, in order to achieve the same connection of the output signal.

0052In yet other alternative arrangements, the amplification of the input sound signal at each of the transposed frequencies may be reduced to compensate for the undesired feedback signal component.

0053In a preferred embodiment of the invention, the sound processing device is implemented according to digital signal processing techniques. As described above, the input signal is windowed and processed as a block of data every 2.2 ms which corresponds to 32 input data samples at a sampling rate of 14.4 kHz. The output signal of the amplifier 23 is generated by summing together the outputs of the 24 oscillators in the bank. The amplitude and frequency controls of the oscillators are determined by pre-processing of the input signal and are updated once for every block of data analysed.

0054One example of a practical fitting procedure to determine the frequency-dependent amplifications of the amplifier involves obtaining a subjective rating of loudness from the listener. The subjective loudness of a stimulus can be judged using a loudness rating scale, such as one containing nine loudness descriptors: <i>Very soft, Soft, Comfortable but slightly soft, Comfortable, Comfortable but slightly loud, Loud but OK, Uncomfortably loud and Extremely uncomfortable.</i> In the following example, a set of <i>comfortable but slightly soft</i> levels are measured and used to determine the desired amplification versus frequency of the amplifier. An appropriate set of stimuli, each of narrow bandwidth, is chosen to be presented to the listener. In one preferred procedure, these stimuli are narrow-band noises, each of one-third octave bandwidth, centred at standard frequencies (i.e. 100, 125, 160, 200, 250, 315, 400, 500, 630, 800, 1000, 1250, 1600 Hz, etc).

0055In one example, the stimuli are presented to the listener using an audio amplifier apparatus having a theoretical model 40 shown in <figref idref="f0004">Figure 5</figref>. It will be noted from this Figure that transducer output is temporarily disconnected from the amplifier circuit to break the feedback loop, thus eliminating any feedback signal from the output. A stimulus generator 41 is connected to the receiver to deliver the desired stimulus to the listener via the transducer. At each stimulus frequency, the stimulus is generated and presented to the listener. The listener responds by describing the perceived loudness. The level of the stimulus is increased and decreased during the fitting procedure so as to converge upon a level which is perceived by the listener to be comfortable but slightly soft.

0056In one practical implementation, each stimulus is presented to the listener, and the listener chooses the most appropriate loudness descriptor from a written list of loudness categories. The stimuli may be presented with varying level and frequency in an appropriate sequence until the loudness at each frequency is perceived as comfortable but slightly soft.

0057An alternative practical implementation involves comparing the loudness of each stimulus with that of a reference stimulus at a selected, fixed frequency. The level of the reference stimulus is set to evoke a loudness perceived as approximately comfortable but slightly soft, and its frequency is chosen to be relatively close to the frequencies of the other stimuli to be presented. In this loudness balancing procedure, each such stimulus maybe presented to the listener with the reference stimulus in a sequential pair (i.e. reference stimulus followed by comparison stimulus, or <i>vice versa</i>), and the listener is asked which of each pair of sounds is louder. The level of the reference stimulus is kept constant. The level of each comparison stimulus is adjusted until its loudness is perceived to be equal to that of the reference stimulus.

0058In one preferred embodiment, the system shown in the attached drawings is implemented partly in a digital signal processor. The level of signals is digitally measured and recorded without the need for separate sound measurement equipment. The level of the stimulus output is recorded for each level found to produce comfortable but slightly soft loudness at each frequency.

0059The following notation is introduced to help describe the relationship between the different levels measured and the frequencies at which they are defined: <ul id="ul0005" list-style="bullet"><li>f with subscript 'o' is an amplifier output frequency;</li><li>f with subscript 'i' is an amplifier input frequency;</li><li>when f_o and f_i are used as arguments in the following formulas, they define whether the function is referred to an output frequency or an input frequency, respectively;</li><li>in the following formulas, f denotes the frequency-shifted version of f. (Similarly, f" denotes the frequency-shifted version of f.)</li></ul>

0060The following functions of frequency are defined: <ul id="ul0006" list-style="bullet"><li>I(f_i) is the level of international long-term average speech spectrum (ILTASS, defined in [<nplcit id="ncit0002" npl-type="s"><text>D. Byrne, H. Dillon, K. Tran, S. Arlinger, K. Wilbraham, R.M. Cox, B. Hagerman, R. Hetu, J. Kei, C. Lui, and J. Kiessling, "An international comparison of long-term average speech spectra," Journal of the Acoustical Society of America, vol. 96, pp. 2108-2120, 1994</text></nplcit>]) at each frequency for speech at an overall level of 70dB SPL, and is a function of input frequency;</li><li>C(f_o) is the comfortable but slightly soft level as a function of output frequency</li><li>H(f_i) is the forward amplification of the amplifier, and is a function of input frequency</li></ul>

0061The amplification required at each input frequency can be defined as the difference (in dB) between the amplifier input ILTASS level and the comfortable but slightly soft level at the shifted (output) frequency. This can be described by the equation:<maths id="math0001"><math display="block"><mfenced open="|" close="|"><mi>H</mi><mfenced><msub><mi>f</mi><mi>i</mi></msub></mfenced></mfenced><mo>=</mo><mfenced open="|" close="|"><mi>C</mi><mfenced><msubsup><mi>f</mi><mi>o</mi><mi>ʹ</mi></msubsup></mfenced></mfenced><mo>-</mo><mfenced open="|" close="|"><mi>I</mi><mfenced><msub><mi>f</mi><mi>i</mi></msub></mfenced></mfenced><mn>.</mn></math><img file="EP1538868B1_D0001.tif" /></maths>

0062This means the set of required amplification can be calculated from the set of comfortable but slightly soft levels and the known ILTASS input levels.

0063In other words, by presenting stimuli to the transducer output at a number of frequencies, and then adjusting the stimulus level at each frequency to meet the predefined loudness perception levels or detection threshold of the listener, an equal loudness contour of transducer output levels can be derived from the adjusted stimuli levels. The equal loudness contour is derived by interpolating between each of the adjusted stimuli levels at the frequencies at which the stimuli are presented to the receiver. The set of required amplifications are then derived by subtracting the magnitude of the known ILTASS level at a particular frequency from the magnitude of the stimulus level defined by the equal loudness contour at the transposed frequency resulting from the frequency transposing element in the forward transfer path.

0064An example of this is shown in <figref idref="f0005">Figure 6</figref>. This Figure illustrates the relationship between input level, output level, and amplification for a downward frequency shift performed by the frequency transposing block 6 of one octave across all frequencies. The amplifications of the amplifier 40 make the output signal comfortable but slightly soft for each one-third octave frequency component when the input frequency components have levels corresponding to those of an average speech signal at 70dB SPL overall level. It will be appreciated from the foregoing that in the absence of any external signal applied to the amplifier 40, the input signal applied to the forward transfer path of the amplifier 40 corresponds to the feedback signal applied to the summation device 4 from the feedback path 3.

0065Stimuli are presented to the transducer from the stimulus generator 41 at a series of frequencies separated by one third of an octave. The stimulus level at each frequency is adjusted to meet the predefined loudness perception level of the listener, in this case the comfortable but slightly soft level. The adjusted stimulus levels are then used to derive an equal loudness contour of transducer output levels - referenced 51 in <figref idref="f0005">Figure 6</figref> - by interpolating between the stimulus levels.

0066In the present example of an audio amplification apparatus involving a feedback suppression system with frequency transposition, the amplification of the amplifier at each desired frequency is determined by subtracting the magnitude of the ILTASS standardised input signal component at that frequency from the magnitude of the stimulus at a transposed frequency as read from the equal loudness contour 51.

0067In the example shown in <figref idref="f0005">Figure 6</figref>, a 4000 Hz stimulus with a level producing comfortable but slightly soft loudness is referenced 52. In post-fitting operation, such a stimulus will have arisen from an input signal in the forward transfer path of an amplifier at 8000 Hz, which will then be frequency transposed downward by one octave to 4000 Hz. Accordingly, the ILTASS standardised input signal component level at 8000 Hz, referenced 53, is used to determine the amplification at 8000 Hz. The difference between the amplitudes of the ILTASS standardised input signal component level 53 at 8000 Hz and stimulus level 52 at 4000 Hz is then used to derive the amplification to be applied at 8000 Hz, referenced 54 in <figref idref="f0005">Figure 6</figref>.

0068It will be appreciated that in the case of audio amplification apparatus that does not include frequency transposition, the difference between the amplitudes of the ILTASS standardised input signal component level and stimulus level at the same frequency are used to derive the amplification to be applied atthat same frequency.

0069It is to be understood that other methods may be used for adjusting the frequency dependent amplification of the audio amplification apparatus. For example, assuming that the feedback path gain is small and has little influence on the closed loop system shown in <figref idref="f0001">Figure 2</figref>, the output signal equal loudness level contour C(f) can be defined in units of dB by:<maths id="math0002"><math display="block"><mfenced open="|" close="|"><mi mathvariant="normal">I</mi><mfenced><msub><mi mathvariant="normal">f</mi><mi mathvariant="normal">i</mi></msub></mfenced></mfenced><mo mathvariant="normal">+</mo><mfenced open="|" close="|"><mi mathvariant="normal">H</mi><mfenced><msub><mi mathvariant="normal">f</mi><mi mathvariant="normal">i</mi></msub></mfenced></mfenced><mo mathvariant="normal">=</mo><mfenced open="|" close="|"><mi mathvariant="normal">C</mi><mfenced><msup><msub><mi mathvariant="normal">f</mi><mi mathvariant="normal">o</mi></msub><mi mathvariant="normal">ʹ</mi></msup></mfenced></mfenced></math><img file="EP1538868B1_D0002.tif" /></maths>

0070The value of |H(f)| can be derived from any loudness contour, as described above. Accordingly, the frequency dependent amplification can be obtained by presenting stimuli to the receiver at a number of frequencies, adjusting the stimulus level at each frequency to meet a predefined loudness perception level of the listener, deriving an equal loudness contour of transducer output levels from the adjusted stimuli levels, and deriving the frequency dependent amplification of levels of input signals at each frequency from the equal loudness contour at the corresponding transposed frequencies.

0071Methods of measuring equal loudness contours sometimes use headphones or other apparatus that require conversion of measured levels to those used in a hearing aid. Individual ear canal shape and hearing aid type influence these conversions which are based on population averages rather than individual characteristics. One way of adjusting the frequency dependent amplification in an audio amplification apparatus which addresses this issue is to use the hearing aid itself to effectively obtain an equal loudness contour.

0072<figref idref="f0006">Figure 7</figref> shows a theoretical model 55 of an audio amplification device including a forward transfer path 56 and a feedback path 57. In this arrangement however, the feedback path is disconnected from the forward transfer path during hearing aid fitting. In this instance, the feedback loop is broken at the microphone output 58, and a desired input signal is applied at the input 59 to the amplifier 60 in the forward transfer path 56. At each frequency, the relevant ILTASS standardised input signal component is applied, and the amplification H(f) is adjusted until the listener indicates that the receiver output meets a predefined loudness perception level of the listener. In this instance, it is not necessary to measure each output level or output frequency of the hearing aid, nor measure the value at which the amplification is set, as long as the set of known input stimuli at the various frequencies applied to the amplifier illicit an equally loud perception level by the user.

0073The manner in which the activation levels described in relation to <figref idref="f0003">Figure 4</figref> are derived will now be discussed. An appropriate set of stimuli, each of narrow bandwidth, is again chosen to be presented to the listener by the stimulus generator 41. In one preferred embodiment, these stimuli are narrow-band noises, each of one-third octave bandwidth, centred at standard frequencies (i.e. 100, 125, 160, 200, 250, 315, 400, 500, 630, 800, 1000, 1250, 1600 Hz, etc).

0074Using the apparatus shown in <figref idref="f0004">Figure 5</figref>, the stimuli are presented one at a time at an appropriate level. The level chosen may be the listener's hearing threshold level, their comfortable but slightly soft level, or some other predefined loudness perception level which elicits a detectable amount of feedback. There is a time-saving advantage in choosing the listener's hearing threshold level, or their comfortable but slightly soft level, since these levels are presented to obtain other data required for aid fitting, as described above.

0075Ideally, the apparatus and listener are situated in a quiet environment (e.g. a sound-proof, or semi sound-proof booth), where background noise levels are low. In order to make reliable and repeatable measurements in the presence of any ambient background noise, the level of the stimuli (and feedback signals) should be high enough to ensure a relatively high signal-to-noise ratio. This needs to be considered when choosing the level of the stimuli to be presented. To further reduce the influence of background noise on the reliability of measurements, it is possible to obtain a set of activation levels from more than one set of stimulus levels; for example, from both the threshold levels and the comfortable but slightly soft levels. These independently measured activation levels could then be combined to produce a single, more accurate set of data.

0076For each stimulus that is presented, the following data are recorded: <ul id="ul0007" list-style="bullet" compact="compact"><li>the output level of the stimulus that is presented;</li><li>the input level of the feedback signal that results from delivery of the stimulus.</li></ul>

0077The following functions of frequency are defined: <ul id="ul0008" list-style="bullet"><li>S(fo) is the level of the stimulus, and is a function of output frequency;</li><li>D(fo) is the listener's hearing threshold level as a function of output frequency;</li><li>F_S(fi) is the level of the feedback signal created by S(fo), and is a function of input frequency;</li><li>A_S(fo) is the activation level of the feedback suppressor and is a function of output frequency. The subscript S indicates that the activation levels were obtained in response to the set of stimuli, S(fo).</li></ul>

0078The transfer function of the feedback path can be determined by comparing the stimulus level S(fo) with the feedback signal level F_S(fi). The feedback suppressor should become active once the output level is above the activation level. The activation level is defined as the output level which causes a feedback signal which, when amplified and optionally shifted in frequency by the hearing aid, will be just audible (or just disturbing) to the listener.

0079An alternative definition for the activation level is the output level that causes a feedback signal which, when amplified and shifted in frequency, would be just above the inherent noise level of the amplifier. When present at the input, the feedback signal is amplified and optionally shifted in frequency, and the corresponding output level is determined by the amplification, H(fi). The relationship between activation level, stimulus level, feedback level, and threshold level in the case of the amplifier shown in <figref idref="f0004">Figure 5</figref>, is defined as follows:<maths id="math0003"><math display="block"><msub><mi>A</mi><mi>S</mi></msub><mfenced><msubsup><mi>f</mi><mi>o</mi><mi>ʹ</mi></msubsup></mfenced><mo>=</mo><mi>S</mi><mfenced><msubsup><mi>f</mi><mi>o</mi><mi>ʹ</mi></msubsup></mfenced><mo>+</mo><mfenced open="{" close="}"><mi>D</mi><mfenced><msubsup><mi>f</mi><mi>o</mi><mi>ʹʹ</mi></msubsup></mfenced><mo>-</mo><mfenced open="[" close="]"><msub><mi>F</mi><mi>S</mi></msub><mfenced><msubsup><mi>f</mi><mi>i</mi><mi>ʹ</mi></msubsup></mfenced><mo>+</mo><mi>H</mi><mfenced><msubsup><mi>f</mi><mi>i</mi><mi>ʹ</mi></msubsup></mfenced></mfenced></mfenced></math><img file="EP1538868B1_D0003.tif" /></maths>

0080It will be appreciated that in audio amplification apparatus having feedback suppression schemes that do not use frequency transposition, the magnitude of the stimulus, listener's hearing threshold level, feedback signal component and amplification may all be taken at the same frequency to determine the activation level at which the undesired feedback signal component is suppressed at that same frequency.

0081The example in <figref idref="f0007">Figure 8</figref> shows the activation levels calculated from a stimulus set of comfortable but slightly soft levels, C(fo), and the corresponding feedback signals. The subscript 'C' has been used to indicate that the set of activation levels, A_C(fo), and the set of feedback levels, F_C(fo), are in response to the stimulus set of comfortable but slightly soft levels, C(f_o).

0082By way of illustration, a 4000-Hz stimulus with a level producing comfortable but slightly soft loudness is shown in <figref idref="f0007">Figure 8</figref> at position 61. Position 61 is equivalent to the level S(fo) in the above equation, where f'_o=4000 Hz. This stimulus will cause a feedback signal at the microphone with frequency of 4000 Hz at the level shown at position 62, F_S(f'_i). This input signal will then be amplified and optionally shifted in frequency. In this example, a downward shift of one octave produces an output signal with frequency f'_o=2000 Hz, and the level shown at position 63, [F_S(f'_i) + H(f'_i)]. From the difference in dB between position 63, the threshold level, D(f"o) - [F_S(f'i) + H(f'i)], and the level of the original stimulus, it is possible to calculate the activation level. This is the maximum level that limits the feedback signal to below the threshold of hearing, position 64.

0083The foregoing is a specific example of the more general case in which the activation levels for feedback suppression in an audio amplification apparatus are determined from listener disturbance threshold levels, amplification and feedback path transfer function. In the example of an audio amplification apparatus using frequency transposition as a means of feedback suppression, the activation levels A(f) can be determined by the following:<maths id="math0004"><math display="block"><mfenced open="|" close="|"><mi mathvariant="normal">A</mi><mfenced><mi mathvariant="normal">f</mi></mfenced></mfenced><mo mathvariant="normal">=</mo><mfenced open="|" close="|"><mi mathvariant="normal">D</mi><mfenced><mi mathvariant="normal">fʹ</mi></mfenced></mfenced><mo mathvariant="normal">-</mo><mfenced open="|" close="|"><mi mathvariant="normal">H</mi><mfenced><mi mathvariant="normal">f</mi></mfenced></mfenced><mo mathvariant="normal">-</mo><mfenced open="|" close="|"><mi mathvariant="normal">G</mi><mfenced><mi mathvariant="normal">f</mi></mfenced></mfenced></math><img file="EP1538868B1_D0004.tif" /></maths> where D(f) are the listener disturbance threshold levels at which a disturbance is detected by a listener. As an example, the listener disturbance threshold levels may be the hearing threshold levels of a listener. It will be appreciated from the foregoing that where the forward transfer path of the audio amplification apparatus includes a frequency transposing element, the activation level at each frequency will be determined from the amplification and feedback path transfer function at that frequency and from the disturbance threshold level at a transposed frequency. However, where the forward transfer path does not include a frequency transposing element, the activation level at each frequency will be determined from the amplifiication, feedback path transfer function and disturbance threshold level at that frequency.

0084Referring once again to <figref idref="f0004">Figure 5</figref>, the stimulus generator 41 can be used to generate a set of output stimuli, which can be used to directly calculate the feedback path transfer function by measuring and recording output transducer signal components and feedback signal components at various frequencies. A suitable set of stimuli may be the set of comfortable but slightly soft equal loudness levels, but other sets of stimuli may equally be used. The characteristics of the feedback path change with disturbances to the sound field around the ear and head. It is therefore possible to construct several different feedback path situations which reflect typical changes in the feedback path that may be expected during every day use of the hearing aid.

0085Assuming an external signal is not provided to the audio amplification apparatus shown in <figref idref="f0004">Figure 5</figref>, we can therefore calculate the feedback path transfer function from the following:<maths id="math0005"><math display="block"><mfenced open="|" close="|"><mi mathvariant="normal">G</mi><mfenced><mi mathvariant="normal">f</mi></mfenced></mfenced><mo mathvariant="normal">=</mo><mfenced open="|" close="|"><msub><mi mathvariant="normal">F</mi><mi mathvariant="normal">O</mi></msub><mfenced><mi mathvariant="normal">f</mi></mfenced></mfenced><mo mathvariant="normal">-</mo><mfenced open="|" close="|"><mi mathvariant="normal">O</mi><mfenced><mi mathvariant="normal">f</mi></mfenced></mfenced></math><img file="EP1538868B1_D0005.tif" /></maths>

0086In this way, the feedback path transfer function G(f) at each frequency is determined by presenting stimuli to the receiver at various frequencies, recording output transducer signal components O(f) and feedback signal components F_O(f) resulting from those stimuli, and deriving the feedback path transfer function at each frequency from the transducer output signal components O(f) and the feedback signal components F_O(f).

0087The feedback path transfer function may also be estimated by detecting the onset of feedback oscillation in a conventional, non-shifting audio amplification apparatus, as shown in <figref idref="f0001">Figure 1</figref>. Without the use of any external apparatus, the amplification at each frequency can be adjusted, and the feedback path transfer function at that frequency derived from the lowest amplification at which feedback oscillation is detected. This onset indicates that the magnitude of the feedback path and the magnitude of the amplification are equal in units of dB, and also that the phase of the loop gain is a whole multiple of 360°. Typically, the onset of feedback oscillation is heard by an audiologist but may also be measured at the output transducer. Once the feedback path transfer function has been estimated, the activation levels may be derived, as described above.

0088It will be appreciated that the above general method of determining activation levels for a feedback suppressor during fitting of a hearing aid to each user is also applicable when no frequency shifting is applied.

0089The above-described embodiment of the sound processor 10 may be implemented by digital signal processing techniques, using processing means to perform the various computations and control the operation of the various other elements of the sound processor 10. It will be appreciated that although a substantially digital implementation of the sound processing device and method has been described above, some or all of the elements or processing stages may be implemented using other techniques, such as by use of analogue electronic circuits. For example, the oscillators may be implemented using appropriate analogue circuits, resulting in a reduction in the electrical power requirements of the processing system, and therefore providing benefits for a practical implementation in a wearable hearing aid.

0090Many other variations may be made to the above described method and device for processing sound signals without departing from the scope of the invention. For example, although no detailed implementation has been described, the present invention may have application to areas of sound processing other than hearing aids.