|
(11) | EP 0 720 146 A1 |
| (12) | EUROPEAN PATENT APPLICATION |
|
|
|
|
|||||||||||||||||||
| (54) | A method for measuring speech masking properties |
| (57) A method measures the masking properties of subband components of a signal and determines
a noise level vector for the signal. In the preferred embodiment, a signal is separated
to yield a set of subband signal components. Bandpass noise components are also generated.
For each combination of bandpass noise and subband signal component, the value of
the noise-to-signal ratio that meets a specified masking criterion is determined.
The values from the combinations are stored. Then, a noise level vector for any other
signal can be determined by filtering the signal into a set of components, accessing
the stored values and combining the values to yield a measure of the masking properties
of the other signal. |
Technical Field
[0001] The invention relates to a method for measuring masking properties of components of a signal and for determining a noise level vector for the signal.
Background of the Invention
[0002] Advances in digital networks such as ISDN (Integrated Services Digital Network) have rekindled interest in the transmission of high quality image and sound. In an age of compact discs and high-definition television, the trend toward higher and higher fidelity has come to include the telephone as well.
[0003] Aside from pure listening pleasure, there is a need for better sounding telephones, especially in the business world. Traditional telephony, with its limited bandwidth of 300-3000 Hz for transmission of narrowband speech, tends to strain listeners over the length of a telephone conversation. Wideband speech in the 50-7000 Hz range, on the other hand, offers listeners a feeling of more presence (by reason of transmission of signals in the 50-300 Hz range) and more intelligibility (by reason of transmission of signals in the 3000-7000 Hz range) and is more easily tolerated over longer periods. Thus, wider bandwidth speech transmission is a natural choice for improving the quality of telephone service.
[0004] In order to transmit speech (either wideband or narrowband) over the telephone network, an input speech signal, which can be characterized as a continuous function of a continuous time variable, must be converted to a digital signal -- a signal that is discrete in both time and amplitude. The conversion is a two step process. First, the input speech signal is sampled periodically in time (i.e. at a particular rate) to produce a sequence of samples where the samples take on a continuum of values. Then the values are quantized to a finite set of values, represented by binary digits (bits), to yield the digital signal. The digital signal is characterized by a bit rate, i.e. a specified number of bits per second that reflects how often the input speech signal was sampled and how many bits were used to quantized the sampled values.
[0005] The improved quality of telephone service made possible through transmission of wideband speech, unfortunately, typically requires higher bit rate transmission unless the wideband signal is properly coded, i.e. such that the wideband signal can be compressed into representation by a fewer number of bits without introducing obvious distortion due to quantization errors. Recently, high fidelity coders of speech and audio have relied on the notion that mean-squared-error measures of distortion (e.g. measures of the energy difference between a signal and the same signal after it is coded and decoded) do not necessarily accurately describe the perceptual quality of a coded signal. In short, not all kinds of distortion are equally perceptible to the human ear. M. R. Schroeder, B. S. Atal and J. L. Hall, "Optimizing Digital Speech Coders by Exploiting Masking Properties of the Human Ear," J. Acous. Soc. Am., Vol. 66, 1647-1652, 1979; N. Jayant, J. Johnston and R. Safranek, "Signal Compression Based on Models of Human Perception," Proc. IEEE, Vol. 81, No. 10, pp. 1385-1422, October 1993; J. D. Johnston, "Transform Coding of Audio Signals Using Perceptual Noise Criteria," IEEE J. Sel. Areas Comm., Vol. 6, pp. 314-323, 1988. Thus, given some knowledge of how the human auditory system tolerates different kinds of noise, it has been possible to design coders that reduce the audibility -- though not necessarily the energy -- of quantization errors. More specifically, these coders exploit a phenomenon of the auditory system known as masking.
[0006] Masking is a term describing the phenomenon of human hearing wherein one sound obscures or drowns out another. A common example is where the sound of a car engine is drowned out if the volume of the car radio is high enough. Similarly, if one is in the shower and misses a telephone call, it is because the sound of the shower masked the sound of the telephone ring; if the shower had not been running, the ring would have been heard.
[0007] The masking properties of a signal are typically measured as a noise-to-signal ratio determined with respect to a masking criterion. For example, one masking criterion is the just-noticeable-distortion (JND) level, i.e. the noise-to-signal ratio where the noise just becomes audible to a listener. Alternatively, another masking criterion is the audible-but-not-annoying level, i.e. the point where a listener may hear the noise, but the noise level is not sufficiently high as to irritate the listener.
[0008] Experiments in the area of psychoacoustics have focused on the masking properties of pure tones (i.e. single frequencies) and of narrow band noise. See, e.g., J. P. Egan and H. W. Hake, "On the Masking Pattern of a Simple Auditory Stimulus," J. Acous. Soc. Am., Vol. 22, pp. 622-630, 1950; R. L. Wegel and C. E. Lane, "The Masking of One Pure Tone by Another and its Probable Relation to the Dynamics of the Inner Ear," Phys. Rev., Vol. 23, No. 2, pp. 266-285, 1924. Psychoacoustic data gathered during these experiments has demonstrated that: when a first tone is used to mask a second tone, the masking ability of the first tone is maximized when the frequency of the first tone is near the frequency of the second tone and that the ability of narrowband noise to mask the second tone is also maximized when the narrowband noise is centered at a frequency near the second tone a lower frequency tone can mask a higher frequency tone more readily than a higher frequency tone can mask a lower frequency tone.
The masking properties of more complex signals (such as wideband speech), however, are more difficult to determine, in part, because they are not readily decomposed into the tones and narrowband noise whose masking properties have been studied.
[0009] Thus, there is a need for a method to a priori measure the masking properties of complex signals, i.e. to determine a priori the level of noise which may be tolerated based on a selected masking criterion. Such measurements may then be used to improve speech coding as described in our co-pending and commonly assigned application "Method for Noise Weighting Filtering," filed concurrently herewith and incorporated by reference.
Summary of the Invention
[0010] Central to the invention is a recognition that the masking properties of a signal, such as wideband speech, may be determined from the masking properties of its subband components. Accordingly, the invention provides a method for determining the masking properties of a signal in which the signal is decomposed into a set of subband components, as for example by a filterbank. In one embodiment, for a given subband component, the noise power spectrum that can be masked by each subband component is identified and the noise spectra are combined to yield the noise power spectrum that can be masked by the signal. In a further embodiment, output signals are generated based on the power in each subband signal and on a masking matrix. The noise power spectrum that can be masked by the input signal is determined from the output signals.
Brief Description of the Drawings
[0011] Advantages of the present invention will become apparent from the following detailed description taken together with the drawings in which:
FIG. 1 illustrates the inventive method for determining a noise level vector of a speech signal.
FIG. 2A illustrates the elements qi,j of a masking matrix Q.
FIG. 2B illustrates the elements of a noise level vector.
FIG. 3 illustrates a system for determining the values of elements qi,j in masking matrix Q in the inventive method.
FIG. 4 is a flow chart for determining the values of the elements qi,j in masking matrix Q in the inventive method.
Detailed Description
[0012] FIG. 1 illustrates a flow chart of the inventive method in which for a frame (or segment) of an input signal, a noise level vector, i.e. the spectrum of noise which may be added to the frame without exceeding a masking criterion, is determined a priori. The method involves three main steps. In step 120 the input signal frame is broken down, as for example by a filterbank, into subband components whose masking properties are known or can be determined. In step 140 the masking properties for each component are identified or accessed, e.g. from a database or a library, and in step 160 the masking properties are combined to determine the noise level vector, i.e. the spectrum of noise power that can be masked by the input signal.
[0013] Note that the method represents the frame of the input signal as a sum of subband components each of whose masking properties has already been measured. However, in order to determine the noise level vector of an input speech signal, the masking properties of the components required in step 140 must first be determined. Once the library of component masking properties is determined and advantageously stored in a database, the masking components can always be accessed, and optionally adapted, to determine the noise level vector of any input signal.
[0014] The inventive method of FIG. 1 recognizes that the masking property of a speech signal, i.e. the spectrum of noise that the speech signal can mask, can be based on the masking property of components of the speech. For example, in order to determine the masking properties of speech, a segment or frame of a first speech input signal is split into subband components, as for example by using a filterbank comprising a plurality of subband (bandpass) filters. In order to determine the spectrum of noise that can be masked by the first speech input signal in a first embodiment, the spectrum of noise that can be masked by each subband component of the speech input signal is determined and then the spectra for all subband components are combined to find the noise level vector for the first speech input signal.
[0015] In another embodiment, for each subband component a measurement is taken to determine how much narrowband noise in each subband can be masked. Thus, the measurement could be summarized as a method consisting of two nested steps:
for every subband of speech i and for every subband of white noise j: Adjust the noise in subband j to the point where sufficient noise is added so that the masking criterion is met. Measure the noise-to-signal ratio at this point. repeat for next subband j repeat for next subband i.
The noise-to-signal measurements for each combination of i and j, qi,j, represent the ratio of noise power in band j that can be masked by the first speech input signal in band i. The elements qi,j form a matrix Q. An example of such a Q matrix is illustrated in FIG. 2A where, for convenience, the entries have been converted to decibels. The Q matrix of FIG. 2A illustrates the results of an experiment in which narrowband speech masked narrowband noise. The row numbers correspond to noise bands; the column numbers correspond to speech bands. Each element qi,j represents the maximum power ratio that can be maintained between noise in band j and the first speech input signal in band i so that the noise is masked. Note that not all qi,j have an associated value, i.e. some entries in the Q matrix are blank, because, as explained below, it typically is not necessary to determine every value in the Q matrix in order to determine the noise level vector. As explained below, the subbands in the Q matrix are not uniform in bandwidth. Instead, the bandwidth of each subband increases with frequency. For example, as shown in Table 2 below, subband 1 covers a frequency range of 80 Hz, from 0 to 80 Hz, while subband 20 covers a frequency range of 770 Hz, from 6230 Hz to 7000 Hz. If the power in each subband of the input frame of the first speech signal is represented as a column vector, p=[p1,p2,...pn]
, the noise level vector dNLV may be found based on the Q matrix and on the p vector:
, i.e. the noise level vector is also a column vector obtained by multiplying the n×n Q matrix by the n column vector of the power in each subband of the input frame of speech as shown in FIG. 2B.
[0016] In either embodiment, once either the spectrum of noise masked by each subband component or the elements in the Q matrix have been determined for a given input signal, they can be used to determine the spectrum of noise that can be masked not only by the given input signal but also by other input signals. For example, if the power in each subband of a second input signal is p2=[p1,p2,...pn]
, then
with Q as determined by the input signal.
[0017] Note that each qi,j is a power ratio determined for a particular masking criterion. This definition makes sense for stationary stimuli (i.e. signals whose statistical properties are invariant to time translation), but in the case of dynamic stimuli, such as speech, care must be taken in adding noise power to a signal whose level varies rapidly. In this instance, this problem is advantageously avoided by arranging for the noise power level to vary with the speech power level so that within a given segment or frame, the ratio of speech to noise power is a predetermined constant. In other words, the level of the added noise is dynamically adjusted in order to achieve a constant signal-to-noise ratio (SNR) throughout the frame. Measuring the amount of masking between one subband component of speech and another subband of noise therefore consists of listening to an ensemble of frames of bandpassed speech with a range of segmental SNRs to determine which SNR value meets the masking criterion. Different frame sizes may advantageously be used for different subbands as described below.
[0018] In the paragraphs that follow a more rigorous presentation is given of the method described above. A method for determining the masking properties of the component signals required for step 140 is presented below first, and then a method of combining the component masking properties in step 160 is presented. The presentation concludes with a short discussion of other potential uses for the inventive method.
[0019] The more rigorous presentation begins by assuming that an input speech signal, s(n) is divided via a bank of filters into N subbands s1(n),...,sN(n), and that the noise maskee d(n) is similarly split into subband components d1(n),...,dN(n). For each pair of subbands (i,j), measure the maximum segmental noise-to-signal ratio (NSR) between dj(n) and si(n) such that the combination of dj(n)+si(n) meets a given masking threshold, e.g. such that the combination of dj(n)+si(n) is aurally indistinguishable (i.e. meets the just noticeable distortion level) from si(n) alone. Define the NSR to be the reciprocal of the traditional SNR, i.e.
where the summation limits span the current frame of speech.
[0020] To split the speech and noise into subbands a non-uniform, quasi-critical band filterbank is designed. The term quasi-critical is used in recognition that the human cochlea may be represented as a collection of bandpass filters where the bandwidth of each bandpass filter is termed a critical band. See, H. Fletcher, "Auditory Patterns," Rev. Mod. Phy., Vol. 12, pp. 47-65, 1940. Thus, the characteristics and parameters of the filters in the filterbank may incorporate knowledge from auditory experiments as, for example, in determining the bandwidth of the filters in the filterbank. Note that it is advantageous that the filterbank used to produce the library of masking properties of components be the same as the filterbank used in step 120 of FIG. 1. However, some constraints on the filterbank may be advantageously imposed to make measurements obtained with one set of filterbank subbands more readily applicable to filterbanks with other subbands. In particular:
Each filter should be as rectangular as possible, although significant passband ripple can be sacrificed in the name of greater attenuation. Overlap between adjacent filters should be minimized. Thus the filterbank is not completely faithful to the human ear to the extent that experimentally measured cochlear filter responses are not rectangular and tend to overlap a great deal. These conditions are imposed, however, since the ultimate interest is in the problem of coding, and splitting an input signal into (nearly) orthogonal subbands prevents coding the same information twice. The composite response of the filters should have nearly flat frequency response. Although perfect reconstruction is not required, the combined output should advantageously be perceptually indistinguishable from the input. This quality of the filterbank may be verified by listening tests. To avoid audible distortions due to different group delays, linear phase filters may be used, although it should be noted that because of the asymmetry of forward and backward masking it would be preferable to use minimum phase filters. This last point is illustrated by considering the case when the speech signal consists of a single spike. The combined output of a linear-phase filterbank would consist of the same spike delayed by half of the filter length, but the combined filtered noise would be dispersed equally before and after the spike. Since forward masking extends much farther in time than backward masking, it would be preferable if more noise came after the spike instead of before; this might be achieved with a more complicated minimum-phase filter design.
[0021] In order to model the constant-Q, critical band nature of the cochlea, the following constraints may also advantageously be imposed: N=20 total subbands, corresponding roughly to the number of critical bands between 0 and 7KHz as found in prior experimental methods. The bandwidths form an increasing geometric series. Assume that the first band spans the frequencies [0,a] and call b the ratio between successive bandwidths, then these last two conditions may be summarized as
where f20 is the highest frequency to be included, typically 7KHz in a speech case. Setting a = 100, corresponding to previous measurements of the first critical band, and solved for b using Newton's iterative approximation. This value of b is then used to generate an ideal set of band edges as shown in Table 1.
[0022] Using these ideal band edges as a starting point, filters may be designed. In one embodiment of the invention, twenty 512-point, min-max optimal filters using the well-known Remez exchange algorithm were designed. Table 2 lists the parameters for each filter. Typically, it may be necessary to adjust the band edges so that the composite filterbank response would be flatter, but the filterbank's combined output should sound identical to the input.
[0023] Since the human cochlea exhibits increasing time resolution at higher frequencies, the frame size for each band is advantageously chosen according to the length of the impulse response of the band filter. For higher bands, the energy of the impulse response becomes more concentrated in time, leading to a choice of a smaller frame size. Table 3 shows the relationship between the noise band number and frame size.
[0024] Despite the well-known dependence of masking on stimulus level, no precise restrictions on loudness during the experiments typically need be imposed. It is usually sufficient to measure masking effects under the normal operating conditions of an actual speech coder. Thus the volume control may be set to a comfortable level for listening to the full-bandwidth speech and left in the same position when listening to the constituent subbands, which as a result sound much softer than the full speech signal. Listening tests are advantageously be carried out in a soundproof booth using headphones with the same signal is presented to both ears.
[0025] As mentioned above, the level of the noise should be adjusted on a frame-by-frame basis in order to maintain a constant local NSR, qij. FIG. 3 is a block diagram of a system to achieve this for each frame of speech. FIG. 4 is a flowchart illustrating steps carried out by the system of FIG. 3. The operation of the system of FIG. 3 is advantageously described on a step-by-step basis:
Generate a frame of unit variance noise: Unit variance Gaussian random noise generator 305 is used to produce u(n) in step 405, which is then scaled according to
where N is the frame size and m is the number of the current frame, starting from m=0. This ensures noise with unit variance on a frame-by-frame basis. Filter speech: Input the current frame of speech in step 410. In step 415 the speech is filtered through filter j 315 of the filterbank to produce sj(n). Measure energy of bandpass speech: The output of filter 315 is then passed through delay 317. The delay allows the system of FIG. 3 to "look ahead" to maintain a constant local NSR as described below. To compute how much noise to inject in this frame, in step 420 calculate the energy pj of the speech as,
using energy measurer 320 where L s the amount of delay as explained in more detail below. Measure look-ahead energy of bandpass speech: Because of the inherent delay imposed by the filterbank, adjustments to the noise level at the filter input are not immediately registered at the output. Therefore some measure of the speech power is needed in the near future to help decide how to adjust the noise level in the present. The look-ahead energy
j is defined as the energy of one frame of sj(n):
Typically L=320 samples yields the best results for 512 point filters. Note that this problem would be easier to solve if the filters were minimum-phase rather than linear phase. Compute desired narrowband noise power: In step 430 multiply the speech power by the desired noise-to-signal ratio qij in adaptive controller 330 to yield a desired noise power, Δ:
Estimate required broadband noise power: To approximate the desired noise power at the filter output, it is noted that for a filter of bandwidth ωi Hz, the filtered unit-variance noise should have a variance of ωi/S, where S is the Nyquist frequency. Linearity may therefore be exploited to try to achieve the desired noise power Δ at the filter output. Because of the filter delays described above, instead of using the speech power in the current frame to compute Δ, a look-ahead desired noise energy
is defined:
Then the noise is scaled in pre-adjuster 340 in order to try to achieve the look-ahead energy as follows:
Filter the adjusted noise: The adjusted noise e(n) is filtered through band i using filter 350, to yield ei(n), and then applied to delay 355 so that the noise is again synchronous with the input frame of speech. Measure the energy of the bandpass noise: Next measure the actual bandpass noise power, di in measurer 360:
Fine-tune the noise: To adjust the noise so that the desired NSR is achieved exactly, apply at multiplier 380 a time-varying gain gi at the filter output. To minimize smearing in the noise spectrum, it is advantageous to vary gi smoothly so that it takes the form
where A is the final value of gi from the previous frame, W is the length of the smoothing window (which can be thought of as half of a Hann window), and B is the final value of gi. Thus, given A and W, one should be able to solve for B such that
Because gi is linear in B, the above expression becomes a quadratic equation of the form
where
Thus a compromise is forced between a smooth transition using a long window, and a crisp change to the desired noise level using a short window. Making the window too short smears the spectrum of the bandpass noise, an effect that typically is quite noticeable, leading to severe underestimates of masking power. Making the window too long, however, leads to more subtle clicks that emerge when the noise level lags behind the speech. Thus, an initial value of
was chosen.
[0026] The quadratic equation for B usually has two real solutions; typically the solution that minimized |A-B| was chosen in order to avoid drastic changes in gain and reduce spectral smearing. Sometimes, however, there is no real solution. This may occur at transitions from loud to soft frames, when reducing the gain gradually had the effect of including more noise at the beginning of the frame than we wanted in the entire frame. In these cases W may be decremented until the longest possible window that allowed an exact solution was found. In rare cases this search can lead to W=0, but only during very soft passages when both speech and noise were below the threshold of hearing. In the W=0 case, gi has the form
Since there are 20 sub-bands, potentially 400 combinations of i and j need to be measured. However, it is not typically necessary to carry out the experiment for every particular (i,j) combination because masking depends on how closely the signal component and masker are in frequency. Thus, typically measurements should be taken for combinations of i and j such that
. Values for qi,j for
can typically be assumed to be zero, i.e. no masking takes place, with perhaps the exception of small values of i and j where masking may sometimes extend over 3 bands.
[0027] Recall that a noise level vector for a speech signal. i.e. the spectrum of noise masked by the input signal, may be calculated according to a three step process. Already demonstrated is that speech might best be analyzed in terms of its constituent critical bands, and determining the masking properties of each band. Now the third step of the process, namely, superposing the masking properties of the subbands to form a noise level vector, is discussed.
[0028] Given a vector of speech powers
, where pi corresponds to the power of the speech in band i in the current frame, a noise level vector
can be determined such that noise added at these levels or below does not exceed the masking threshold.
[0029] This calculation requires knowledge of how to add the masking effects of two or more maskers and the effects are combined simple addition; or, more formally:
Linear superposition of noise power: If a signal S masks a noise power vector
, i.e., where dj is the power of the noise in band j in the current frame and "T" indicates the transpose; and another signal S', uncorrelated with S, masks a noise power vector d' = (d
,...,d
)T; then the combined signal S + S' will mask the noise power vector
Simple addition is advantageously used instead of non-linear superpositions rules because it typically leads to more conservative estimates of the masking properties of the signal.
[0030] Note generally that the superposition idea assumes that consecutive bands in the filterbank do not overlap, so that the noise level in one band can be adjusted without affecting the level in another, and so that the speech may be decomposed into uncorrelated subbands. Thus high-order, nearly rectangular filters in the filterbank were used.
[0031] Accordingly the total spectrum of the noise level vector, dNLV can be found in a given frame if we know the masking property di for every band of speech i = 1,...,20 is known. This involves a simple sum of noise powers:
To find the masked noise vector di for speech band i, use the measured threshold NSRs qij. Since the speech power pi and the minimum ratio of speech to noise power qij are known, then the maximum masked power in bands 1-20 using one column of the qij matrix can be computed:
In other words, the threshold noise power in each band is equal to the product of the signal power and the threshold noise-to-signal ratio.
[0032] Combining equations 4.2 and 4.3 to summarize the method as one matrix equation yield.
where
. (Note that whenever qij has not been measured, assume that there is zero masking; qij = 0.) Equation 4.4 thus describes how the noise level vector for a given frame of speech can be determined based on the input power in the speech frame and on the masking properties of speech as represented by the masking matrix Q.
[0033] The above method is flexible in that new knowledge about masking effects in the human auditory system may be readily incorporated. The choice of a linear superposition rule, for example, can be easily changed to a more complex function based on future auditory experiments. The values in the Q matrix, moreover, need not be fixed. Each element in the matrix could be adaptive, e.g. a function of loudness since masking properties have been shown to change at high volume levels. It would also be easy to use different Q matrices depending on whether the current frame of speech consisted of voiced or unvoiced speech.
[0034] This disclosure describes a method for measuring the masking properties of components of speech signals and for determining the masking threshold of the speech signals. The method disclosed herein has been described without reference to specific hardware or software. Instead the method has been described in such a manner that those skilled in the art can readily adapt such hardware or software as may be available or preferable.
[0035] While the above teaching of the present invention has been in terms of determining the masking properties of speech signals, those skilled in the art of digital signal processing will recognize the applicability of these teachings to other specific contexts. Thus, for example, the masking properties of music, other audio signals, images and other signals may be determined using the present invention.
TABLE 1
| Band number | Lower edge Hz | Upper edge Hz |
| 1 | 0 | 100 |
| 2 | 100 | 212 |
| 3 | 212 | 337 |
| 4 | 337 | 476 |
| 5 | 476 | 632 |
| 6 | 632 | 806 |
| 7 | 806 | 1001 |
| 8 | 1001 | 1219 |
| 9 | 1219 | 1462 |
| 10 | 1462 | 1734 |
| 11 | 1734 | 2038 |
| 12 | 2038 | 2377 |
| 13 | 2377 | 2756 |
| 14 | 2756 | 3180 |
| 15 | 3180 | 3654 |
| 16 | 3654 | 4183 |
| 17 | 4183 | 4775 |
| 18 | 4775 | 5436 |
| 19 | 5436 | 6174 |
| 20 | 6174 | 7000 |
TABLE 2
| Band number | Lower edge Hz | Upper edge Hz | Δflow Hz | Δfhigh | W | Scale factor |
| 1 | 0 | 80 | 70 | 80 | 200.0 | 1.0 |
| 2 | 120 | 195 | 75 | 75 | 450.0 | 0.9 |
| 3 | 228 | 300 | 80 | 80 | 300.0 | 0.9 |
| 4 | 337 | 435 | 75 | 75 | 300.0 | 0.9 |
| 5 | 485 | 600 | 90 | 90 | 150.0 | 1.0 |
| 6 | 660 | 806 | 85 | 85 | 150.0 | 1.0 |
| 7 | 860 | 1000 | 85 | 85 | 150.0 | 1.0 |
| 8 | 1060 | 1210 | 85 | 85 | 150.0 | 1.0 |
| 9 | 1265 | 1460 | 85 | 85 | 150.0 | 1.0 |
| 10 | 1515 | 1735 | 85 | 85 | 150.0 | 1.0 |
| 11 | 1790 | 2038 | 85 | 85 | 150.0 | 1.0 |
| 12 | 2095 | 2377 | 85 | 85 | 150.0 | 1.0 |
| 13 | 2435 | 2756 | 85 | 85 | 150.0 | 1.0 |
| 14 | 2815 | 3180 | 85 | 85 | 150.0 | 1.0 |
| 15 | 3239 | 3654 | 85 | 85 | 150.0 | 1.0 |
| 16 | 3712 | 4183 | 85 | 85 | 150.0 | 1.0 |
| 17 | 4242 | 4775 | 85 | 85 | 150.0 | 1.0 |
| 18 | 4835 | 5437 | 85 | 85 | 150.0 | 1.0 |
| 19 | 5495 | 6174 | 85 | 85 | 150.0 | 1.0 |
| 20 | 6230 | 7000 | 85 | 85 | 150.0 | 1.0 |
TABLE 3
| Noise band# | Frame size (samples) |
| 1-5 | 512 |
| 6-14 | 256 |
| 15-20 | 128 |
1. A method of determining the noise power spectrum that can be masked by a signal, the
method comprising the steps of:
separating said signal into a set of subband components,
identifying the noise power spectrum that can be masked by each subband component in said set of subband components, and
combining the identified noise power spectrum masked by each subband component to yield the noise power spectrum that can be masked by said signal.
separating said signal into a set of subband components,
identifying the noise power spectrum that can be masked by each subband component in said set of subband components, and
combining the identified noise power spectrum masked by each subband component to yield the noise power spectrum that can be masked by said signal.
2. The method of claim 1 wherein the step of separating comprises the step of:
applying said signal to a filterbank comprising a set of filters wherein the output of each filter in said set of filters is a subband component of the signal.
applying said signal to a filterbank comprising a set of filters wherein the output of each filter in said set of filters is a subband component of the signal.
3. The method of claim 1 wherein the step of combining comprises the step of:
adding the noise power spectra masked by each subband component to yield the noise power spectrum masked by said signal.
adding the noise power spectra masked by each subband component to yield the noise power spectrum masked by said signal.
4. The method of claim 1 wherein said signal is wideband speech.
5. A method comprising the steps of:
separating an input signal to a set of subband signal components, and
generating output signals based on the power in each subband signal component and on a masking matrix.
separating an input signal to a set of subband signal components, and
generating output signals based on the power in each subband signal component and on a masking matrix.
6. The method of claim 5 wherein said masking matrix Q is an n×n matrix wherein each element qi,j of said masking matrix is the ratio of the noise power in band j that can be masked by the power of the subband signal component in band i.
7. The method of claim 5 wherein the input signal is a speech signal.
8. The method of claim 5 wherein the step of separating comprises the step of:
applying said input signal to a filterbank comprising a set of filters wherein the output of each filter in said set of filters is a subband component of the signal.
applying said input signal to a filterbank comprising a set of filters wherein the output of each filter in said set of filters is a subband component of the signal.
9. A method comprising the steps of:
separating a signal into a set of n subband signal components, wherein each subband signal component is characterized by a power level,
generating a set of n subband noise components, and
for combinations of one subband signal component i,i=1,2,...n and one subband noise component j,j=1,2,...n, measuring the ratio of the power level of the jth subband noise component that can be masked by the ith subband signal component to the power level of the ith subband signal component.
separating a signal into a set of n subband signal components, wherein each subband signal component is characterized by a power level,
generating a set of n subband noise components, and
for combinations of one subband signal component i,i=1,2,...n and one subband noise component j,j=1,2,...n, measuring the ratio of the power level of the jth subband noise component that can be masked by the ith subband signal component to the power level of the ith subband signal component.
10. The method of claim 9 wherein the power level of each subband noise component that
can be masked by each subband signal component is determined according to a masking
criterion.
11. The method of claim 10 wherein said masking criterion is a just-noticeable-distortion
level.
12. The method of claim 10 wherein said masking criterion is an audible-but-not-annoying
level.
13. The method of claim 9 wherein said step of separating a signal into a set of n subband signal components comprises the step of applying said signal to a first filterbank
comprising a first set of n filters, wherein the outputs of said first set of filters in said first filterbank
are the set of n subband signal components.
14. The method of claim 13 wherein said step of generating a set of n subband noise components comprises applying a wideband noise signal to a second filterbank
comprising a second set of filters, said second filterbank having the same filter
characteristics as said first filterbank, wherein the outputs of said second set of
filters in the second filterbank are said set of n subband noise components.
15. The method of claim 10 wherein
the measured ratio is an element qi,j of a masking matrix Q.
the measured ratio is an element qi,j of a masking matrix Q.
16. The method of claim 15 further comprising the steps of:
multiplying the masking matrix by a vector p whose elements pi are the power in each subband component of an input signal, to yield the noise power spectrum that can be masked by the signal.
multiplying the masking matrix by a vector p whose elements pi are the power in each subband component of an input signal, to yield the noise power spectrum that can be masked by the signal.
17. A method of determining the power of a filtered noise signal that can be masked by
a filtered frame of speech, said method comprising the steps of:
delaying said filtered frame of speech by a specified time,
determining the power of said filtered frame of speech,
measuring the power of said filtered noise signal,
delaying said filtered noise signal by said specified time, and
adjusting the power of said filtered noise signal as a function of the power of said filtered frame of speech and of a desired noise-to-signal ratio to yield the power of the filtered noise signal that is masked by the filtered frame of speech.
delaying said filtered frame of speech by a specified time,
determining the power of said filtered frame of speech,
measuring the power of said filtered noise signal,
delaying said filtered noise signal by said specified time, and
adjusting the power of said filtered noise signal as a function of the power of said filtered frame of speech and of a desired noise-to-signal ratio to yield the power of the filtered noise signal that is masked by the filtered frame of speech.
18. The method of claim 17 further comprising the step of multiplying said filtered noise
signal by a gain signal so as to achieve the desired noise-to-signal ratio.
19. The method of claim 17 wherein said specified time is a function of the impulse response
of said first filter.
20. The method of claim 17 wherein said desired noise-to-signal ratio is determined according
to a masking criterion.
21. The method of claim 17 further comprising the steps of:
generating a noise signal, said noise signal having unit variance; and
applying said noise signal to a second filter to generate said filtered noise signal.
generating a noise signal, said noise signal having unit variance; and
applying said noise signal to a second filter to generate said filtered noise signal.
22. A method comprising the steps of:
applying an input speech signal to a filterbank, said filterbank comprising a set of n filters wherein the output of each filter is a respective subband signal component in a set of n subband signal components, and
generating output signals based on the product of a masking matrix Q and a vector p, wherein said masking matrix Q is an n×n matrix in which each element qi,j of said masking matrix is the ratio of power of the noise in filter j that can be masked by the power of the subband signal component in band i and wherein said vector p is a vector of length n in which each element pi is the power of the ith signal component.
applying an input speech signal to a filterbank, said filterbank comprising a set of n filters wherein the output of each filter is a respective subband signal component in a set of n subband signal components, and
generating output signals based on the product of a masking matrix Q and a vector p, wherein said masking matrix Q is an n×n matrix in which each element qi,j of said masking matrix is the ratio of power of the noise in filter j that can be masked by the power of the subband signal component in band i and wherein said vector p is a vector of length n in which each element pi is the power of the ith signal component.