TW200847133A - System and method for processing an audio signal - Google Patents

System and method for processing an audio signal Download PDF

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Publication number
TW200847133A
TW200847133A TW96144620A TW96144620A TW200847133A TW 200847133 A TW200847133 A TW 200847133A TW 96144620 A TW96144620 A TW 96144620A TW 96144620 A TW96144620 A TW 96144620A TW 200847133 A TW200847133 A TW 200847133A
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Taiwan
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filter
signal
sub
band signals
method
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TW96144620A
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Chinese (zh)
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TWI421858B (en
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Ludger Solback
Lloyd Watts
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Audience Inc
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Abstract

Systems and methods for audio signal processing are provided. In exemplary embodiments, a filter cascade of complex-valued filters are used to decompose an input audio signal into a plurality of frequency components or sub-band signals. These sub-band signals may be processed for phase alignment, amplitude compensation, and time delay prior to summation of real portions of the sub-band signals to generate a reconstructed audio signal.

Description

200847133 IX. DESCRIPTION OF THE INVENTION: TECHNICAL FIELD OF THE INVENTION The specific embodiments of the present invention relate to audio processing, and more particularly to the analysis of audio signals. [Prior Art] There are a number of solutions for dividing an audio signal into sub-bands and deriving frequency-dependent amplitude and phase characteristics that change over time. Examples include a window-type fast Fourier transform/inverse fast Fourier transform (fft/ifft) system and parallel sets of finite impulse response (F J R) filter banks and infinite impulse response (10) filter banks. No. 35, these conventional solutions all suffer from defect 0. The windowed FFT system only provides a single fixed 'width for each frequency band', wide ... pass at the bottom with a high resolution to select a low frequency to the term rate Know the bandwidth. For example, in the workplace (10), a filter (group) with a frequency is required. However, this means that a 5 Hz HZ frequency is used at 8 kHz, and a wider bandwidth (e.g., 400 Hz) at 8 kHz may be more appropriate. Therefore, the soil stems are more than the first line, and can not provide the flexibility to match the human perception. 0 Window type FFT system access + At high frequencies, another disadvantage is the sparse sampling window type FFT system, insufficient fine frequency resolution. Under the condition of applying modification (for example, music I), artificial products (for example, "sound hop I") may not be used. It is possible to reduce the number of overlaps between window frames by significantly reducing the number of FFT products (ie, increasing the number of oversamplings) to some extent. Unfortunately, the computational cost of the FFT system increases with the increase in oversampling by 127033.doc 200847133. Similarly, the FIR subclass of the 遽's wave group is also computationally expensive due to the convolution of the sampled impulse responses in each sub-band (which can result in high latency). For example, a system with a window of 256 samples would require foot multiplication and a potential of 128 samples if the window is symmetrical. The IIR subclass is less expensive to calculate due to its recursive nature, but implementations that only use real-valued filtered H-factors have difficulties in achieving near-perfect reconstructions, especially when modifying sub-band signals. In addition, phase and amplitude compensation and time alignment for each band are required to produce a flat frequency response at the output. It is difficult to perform phase compensation in conjunction with real-valued signals because their loss is used to directly calculate the amplitude and phase orthogonal components with fine time resolution. The most common method for determining amplitude and frequency is to apply a Hilbert transform at each stage of the output. However, an additional computational step for computing the Hilbert transform in a real-valued filter bank is required and is computationally expensive. Therefore, there is a need for a less expensive system and method for analyzing and reconstructing an audio signal while providing low end-to-end latency, as well as the necessary degrees of freedom for time-frequency resolution, as compared to existing systems. SUMMARY OF THE INVENTION Embodiments of the present invention provide systems and methods for audio signal processing. In an exemplary embodiment, a wave cascade of complex value filters is used to decompose an input audio signal into a plurality of sub-band signals. In one embodiment, an input signal is filtered using a filter cascading complex value filter to generate a first filtered signal. The younger filtered 'wave number' is subtracted from the input signal to derive a first frequency band signal. Then, the first 127033.doc 200847133 wave signal is processed by the filter cascade to generate a down-filtered signal. Repeat the procedure until the last complex value filter in the cascade is utilized. In some embodiments, the (IV) complex-valued waver is a single-pole complex-valued waver.

Once the input signal is resolved, the sub-band signals can be processed by a reconstruction module. The reconstruction module is configured to perform a phase alignment on or after the sub-band signals. The reconstruction module can also be configured to perform amplitude compensation on one or more of the sub-band signals. In addition, a time delay can be performed on the plurality of sub-band signals by the reconstruction. The real part of the compensated and/or time delayed subband signals is summed to produce a reconstructed audio signal. [Complex method] This month's specific shell example provides a system and method for near-perfect reconstruction of an audio signal. The exemplary I system utilizes a recursive chopper group to generate quadrature outputs. In an exemplary embodiment, the filter bank includes a plurality of complex value filters. In other embodiments, the filter bank includes a plurality of unipolar complex filters. - An exemplary system 100 is shown with reference to Figure 1 in which a specific embodiment of the present invention can be implemented. (d) Reconciliation (10) may be any device capable of processing audio signals' such as, but not limited to, a cellular telephone, a hearing aid, a speakerphone, a computer, or any other device. (d) System (10) may also indicate an audio path to any of these devices. The system 100 includes an audio processing engine 1() 2...audio source (10), an adjustment module 1〇6, and an audio slot 108. Other components that are important to the audio signal can be provided in the system. In addition, although the system is just described as being logically advanced from the various components of FIG. 1 to the next, alternative embodiments may include various couplings of system 100 via one or more busbars or other components. The components. The example audio processing engine 102 processes the input (audio) 5 hole number entered via the audio source 104. In one embodiment, audio processing engine 102 includes software stored on a device that is operated by a general purpose processor. In various embodiments, the audio processing engine 102 includes an analysis filter bank module 110, a modification module 丨丨2, and a reconstruction module 丨丨4. It should be noted that more, fewer or functional equivalent modules are available in the Audio Processing Engine 1 〇2. For example, one or more of modules 110-114 can be combined into very few modules and still provide the same functionality. Audio source 104 contains any device that receives input (audio) signals. In some embodiments, the audio source 1〇4 is configured to receive analog audio signals. In one example, the audio source 1〇4 is coupled to a microphone of an analog to digital (A/D) converter. The microphone is configured to receive an analog audio signal, and the A/D converter samples the analog audio signals to convert the analog audio signals into digital audio signals suitable for further processing. In other examples, the audio source 1〇4 is configured to receive an analog audio signal, and the adjustment module 1〇6 includes the A/D converter. In an alternate embodiment, the audio source 1〇4 is configured to receive digital audio signals. For example, the audio source 1 is a disk device capable of reading audio signal data stored on a hard disk or other media form. Other embodiments may utilize other forms of audio signal sensing/capturing devices. The adjustment module 106 preprocesses the input signal (i.e., does not require any rounding of the signal 127033.doc 200847133 for any processing). In one embodiment, the adjustment module 丨〇6 includes an automatic gain control. The adjustment module 106 can also perform error correction and noise filtering. Adjustment module 106 can include other components and functions for pre-processing audio signals. The analysis filter bank module 11 decomposes the received input signal into a plurality of sub-band signals. In some embodiments, the output from the analysis filter to the group module 11 can be used directly (e.g., for visual display). The analysis filter bank module 110 will be discussed in greater detail in conjunction with FIG. In an exemplary embodiment, each sub-band signal represents a frequency component. The exemplary modification module 112 receives the sub-band signals from the analysis filter bank group 110 through individual analysis paths. The modification module i丨2 can modify/adjust the sub-band signals based on the individual analysis paths. In one example, the modification module 112 enters the noise in the sub-band signal received through the specific analysis path.

Discuss the reconstruction module 丨丨4 in detail. Line filtering. In another example, a secondary frequency received from a particular analysis path

The unusable part of the signal.

In some embodiments, any device of the Dong Jian audio signal is output. An analogy of the reconstructed audio signal 127033.doc 200847133 is output in the frequency bin 108. For example, the audio slot 108 can include a number of "digital" analog-to-digital (D/A) converters with: (d) In this example, the D/A converter is configured to receive:: processing engine 102 Rebuild the audio signal and convert it to an analog reconstruction.曰 Frequency signal. The speaker can then receive and output the analog to reconstruct the audio signal. The audio slot 108 can include any analog output device including: not limited to a headset, earphone, or hearing aid. Alternatively, the audio slot (10) contains the D/A converter and is configured to be used to external audio packs

For example, audio output from speakers, headphones, headphones, hearing aids. In an alternate embodiment, the audio slot 108 outputs a digitally reconstructed audio signal. In another example, the audio slot is a disk device in which the reconstructed audio signal can be stored on a hard disk or other medium. In an alternate embodiment, the audio slot 108 is optional and the audio processing engine 1 2 produces the reconstructed audio signal for further processing (not shown in Figure 1). Referring now to Figure 2, an exemplary analysis filter bank module 110 is shown in greater detail. In an exemplary embodiment, the analysis packet module 11 receives an input signal 202 and processes the input signal 202 through a series of filters 2〇4 to generate a plurality of sub-band signals or components (eg, 1> 1 to 6). A number of filters 204 can include the analysis filter bank module 11(). In an exemplary embodiment, filter 204 is a complex valued filter. In other embodiments, filter 204 is a first order filter (e.g., unipolar complex). The filter 204 is further discussed in FIG. In an exemplary embodiment, filter 204 is organized into a waver cascade, whereby one of the outputs of one filter 204 becomes an input in the next wave 127033.doc -11 - 200847 133 of the cascade. . Therefore, the input signal 2〇2 is fed to a first filter 2 (Ma. The first computing node 2〇6a subtracts one of the output signals ρι of the first filter 204a from the input signal 2〇2 to generate An output D1. The output D1 represents a difference signal between the signal in the first filter 2〇4a and the signal after the first filter 2. In an alternative embodiment, the computing node 2〇6 is not used. The benefits of filter cascading can be achieved by determining the sub-band signal. That is, for example, straight

The output of each filter is used to represent the energy at the output or the signal to be displayed. Due to the cascading structure of the analysis filter bank module 110, the output signal is now an input signal entering the servant of the next filter 20 in the cascade. Similar to the procedure associated with the first filter 204a, the next compute node 2 subtracts the output of the down-chopper 2_ from the input signal pi (ie, p2) to obtain the down-band or channel. (ie output). This lower channel emphasizes the frequency between the local wave 204b and the cutoff frequency of the pre-filter 204a. This program continues through the rest of the cascaded filter 204. ^ In a specific embodiment, the 级, 古 τ < 宜渡杰集 in the cascading is separated into eight groups. Therefore, filter parameters and coefficients can be shared between corresponding data filters (located in a similar position) in different groups of eight. This procedure is described in detail in the official patent application Serial No. 09/534,682. , is a unipolar complex value of the Boqin. For example, 'filter 2G4 can include an order-order or analog filter that operates with complex values. Overall, the output table of the filter 2〇4 is not the sub-frequency V component of the audio signal. Due to the calculation node 200, each output #^, the return table does not have a frequency band, and the sum of 127033.doc -12-200847133 has the entire input signal 202. Since the cascading filter 2 〇 4 is a first order, the pain λ 々々 is comparable to the cascading filter 204 as a second or second order. Ten different costs are much less. In addition, each frequency band captured by the audio signal can be easily changed by changing the first-order filter. In other embodiments, filter 204 is a complex valued filter and does not have to be a single pole. In other embodiments, the modification module 112 (FIG. 1) can process the output of the juice node 206 as necessary. For example, the modification module 112 can half-wave rectify the wave band. In addition, the gain of the output can be adjusted. To compress or extend the motion "range. In some embodiments, the output of any filter 2〇4 can be reduced before being processed by another filter 2〇4 chain/cascade. In an exemplary implementation In the example, filter 2〇4 has infinite impulse response (IIR) filtering designed to produce a cutoff frequency of the desired channel resolution: Filter 204 can perform continuous Hilbert transform using various coefficients on the complex audio signal so that The signal is suppressed or outputted in a particular sub-band. Figure 3 is a block diagram illustrating the signal stream in an exemplary embodiment of the present invention. The output of the pass filter 204 is ^ι[η]^_[η Used as the input of the next filter 2〇4 in the cascade and X1 mag[η+1 ]. The item η identifies the sub-band to be extracted from the audio signal, where "n" is assumed to be an integer. „R filter 2〇4 is recursive, so filter The output of the device can be changed based on the previous output. The imaginary component of the input signal (eg, Ximag[[]] can be summed after, before, or during the summation of the real components of the signal. In a particular embodiment, The complex first-order difference equation is used to describe the filter 2〇4, where b = r_Z*exp(i*theta_P) and a=_r_p*exp(i*theta_p)*, " is the same as this 127033.doc -13 - 200847133 Index. In the present embodiment 'MgM is a gain factor. It should be noted that this gain factor can be applied anywhere without affecting the pole and zero positions. In an alternative embodiment, the audio signal has been decomposed into times. After the band signal, the gain can be applied by modifying module 112 (FIG. 1). Referring now to Figure 4, an exemplary magnitude and phase logarithmic display is displayed for every six sub-bands of an audio signal. The phase information is based on the output from the analysis filter bank module ^ (^ 图). That is, the amplitude shown in Figure 4 is from the output of the compute node 206 (Figure 2) (ie, outputs 〇1 to 〇6). In the example, 'for the filter bank module 1 i 〇 is combining one from 8 to this edge The frequency range of 23 sub-bands operates at a sampling rate of 16 kHz. The end-to-end latency of the analysis filter and wave bank module 110 is 17 3 ms. In some embodiments, at high frequencies It has a wide frequency response and a narrow frequency response at low frequencies. Since the specific embodiment of the present invention can be adapted to many audio sources 1G4 (Fig. 所以, different bandwidths of different frequencies can be used. The fast response with wide bandwidth at high frequencies and the slow response with narrow and short bandwidth at low frequencies have relatively low latency (eg 12ms) and are more suitable for human ear response. - Figure 5, showing - analyzing the magnitude and phase of each level of the deaf design to P6) & / ° The amplitude shown in Figure 5 is the output of the waver 204 of Figure 2 (eg, P1 is said to be in accordance with the present invention) In the specific example of the reconstruction module 114 of the example of the ~1 丹 腹 她 她 她 她 她 她 她 她 她 她 她 她 她 她 她 她 她 她 她 她 她 她 她 她 她 她 她 她 她 她 她 她 她 她 她 她 她 她 她 她At 127033.doc -14- 200847133, the time is adjusted by delaying the signal of each sub-band to obtain a flat reconstructed spectrum and reduce the impulse response dispersion. The port uses a composite signal (such as real and imaginary parts) for filtering. The phase is derived for any sample. In addition, the amplitude can also be calculated by A ==. Therefore, the audio signal reconstruction is mathematically made easier. As a result of this method, the amplitude and phase of any sample can be easily used for further processing ( That is, to modify the module 112 (Fig. )). (Because the impulse response of the sub-band signal can have different group delays, only the summation of the output of the analysis filter bank module 110 (Fig. 1) may not be mentioned. Accurate reconstruction of the audio signal. Therefore, the output of the primary band can be delayed by the pulse response peak time of the sub-band so that all sub-band filters have their impulse response envelope maximum at the same time instant. In a specific embodiment where the desired group delay is later, the filter output is multiplied by a complex constant such that the real portion of the impulse response has a local maximum at the desired group delay. As shown, re-modeling Group 114 receives subband signal 602 (e.g., SG, Sj Sm) from modification module 112 (Fig. 1). Coefficients 604 (e.g., known, ^, and am) are applied to the subband signal. The coefficients include a fixed complex factor (i.e., Include a real and imaginary part.) Alternatively, the coefficient 604 can be applied to the sub-band signal in the analysis filter bank module. The application of the coefficient to each sub-band signal will align the phase of the sub-band signal and compensate for the amplitude. In an exemplary embodiment, the coefficients are predetermined. After the application of the coefficients, the imaginary part is discarded by a real-valued module 606 (ie, Re{ }). The real portion of the sub-band signal is delayed by 127033.doc -15-200847133 by a delay Z·1 608. This delay provides cross-subband alignment. In one embodiment, the delay 提供·ι 608 provides a tap delay. After the delay, the individual sub-band signals are summed in a summing node 610, resulting in a value. The partial reconstruction signal is then carried to the next summing node 61 and applied to the next delayed sub-band signal. The program continues until all sub-band signals are summed, resulting in the reconstruction of the audio signal. Therefore, the reconstructed audio signal is adapted to the audio slot 1 Figure 1). Although the display delay Ζ·ι 608 is described after summing the sub-band signals, However, the order of operation of the reconstruction module 114 is interchangeable. FIG. 7 illustrates a reconstruction graph based on the examples of FIGS. 4 and 5. After the phase alignment, amplitude compensation, and delay for the sub-band alignment performed by the reconstruction module 114 (FIG. 1), reconstruction is achieved by combining the rotations of the filters 2〇4 (FIG. 2) (ie, reconstruction). Audio signal). Therefore, the reconstructed graph phase = flatter. Referring now to Figure 8, a flow chart 800 of an exemplary method for audio signal processing is provided. In step 802, an audio signal is decomposed into a number of sub-band signals. In an exemplary embodiment, # is processed by the analysis filter bank module 11 (Fig. 1). The process includes filtering the audio signal through a cascade of filters 2〇4 (Fig. 2), and the output of each filter 2〇4 results in a primary band signal at the individual output 206. In a specific embodiment, filter 204 is a complex valued filter. In another embodiment, the filter benefit is a unipolar complex value filter. After the sub-band decomposition, the sub-band signal is processed by the modification module 112 (Fig. 804) in step 804. In an exemplary embodiment, the module 112 is modified (the gain of the output is adjusted to compress or expand a dynamic range. In some embodiments 127033.doc -16-200847133, the modified module m can be suppressed from being adopted. Sub-band signal. In step _, a reconstruction module 114 (Fig. 仃 then performs phase and amplitude compensation in each sub-band signal. In a specific embodiment, by applying a coefficient to the sub-band signal Phase and amplitude compensation. The imaginary part of the compensated sub-band signal is then discarded in step 808. In other embodiments, the imaginary part of the compensated sub-band signal is retained. Using the real part of the compensated sub-band signal, In step 81A, the sub-band signal is delayed for the sub-band alignment. In a specific embodiment, the delay is obtained by using a delay line in the reconstruction module 114. In step 812, the delayed sub-band is used. Signal summation to obtain a reconstruction nickname. In an exemplary embodiment, each sub-band signal/segment represents a frequency. The invention has been described above with reference to exemplary embodiments. The embodiments are well known to those skilled in the art, and various modifications may be made and other specific embodiments may be employed without departing from the scope of the invention. And other variations are encompassed by the present invention. Χ [Simplified illustration of the drawings] FIG. 1 is an exemplary block diagram of a system embodying a specific embodiment of the present invention; FIG. 2 is an analysis of an exemplary embodiment of the present invention. An exemplary block diagram of a filter bank module; FIG. 3 illustrates a filter of the analysis filter bank module in accordance with an embodiment; 127033.doc • 17- 200847133 FIG. 4 for every six (6) sub-bands Explain the logarithm of the magnitude and phase of the subband transfer function; Figure 5 illustrates the logarithm of the magnitude and phase of the cumulative filter transfer function for every six (6) stages; Figure 6 illustrates the operation of the exemplary reconstruction module; and

7 is a flow chart showing an exemplary reconstruction of the audio signal. FIG. 8 is a flow chart of an exemplary method for reconstructing an audio signal. [Main element symbol description] 100 System 102 104 106 108 110 112 114 204 204a 204b 206 206a 206b 606 610 Audio Processing Engine Audio Source Adjustment Module Audio Slot Analysis Filter Bank Module Modification Module Reconstruction Module Filter First Filter Next Filter Computation Node/Turn Out First Computation Node Ding One Computation Node Value module summation node 127033.doc -18-

Claims (1)

  1. 200847133 X. Patent application scope: 1 · A method for processing an audio signal, comprising: filtering a input signal by using a filter cascaded complex value filter to generate a first filtered and wave signal; Subtracting the first filtered signal from the input signal to derive a first-order frequency band signal; using the filter cascade to filter the first filtered signal to generate a next filtered signal And subtracting the next filtered signal from the first filtered signal to derive a next frequency band signal. 2. The method of claim 1, wherein the complex value filter and the next complex value generator are unipolar complex value filters and waves. 3. The method of claim 1, further comprising performing phase alignment on one or more of the sub-band signals. 4. The method of claim 3, further comprising arranging an imaginary portion of the one or more phase aligned sub-band signals. 5. The method of claim 1, further comprising performing amplitude compensation on one or more of the sub-band signals. 6. The method of claim 1, further comprising performing a time delay on one or more of the sub-band signals, such as δH, for the cross-subband alignment. 7) The method of claim 6, further comprising summing the delayed one or more sub-band signals to generate a reconstructed audio signal. 8. The method of claim 1, further comprising preprocessing the input 127033.doc 200847133 signal prior to filtering the input signal using the complex cascade filter of the filter cascade. 9. The method of claim 1, further comprising modifying one or more of the sub-band signals based on an analysis path from the filter cascade. 10. The method of claim 1, wherein the sub-band signals are frequency components of the input signal. 11. A system for processing an audio signal, comprising: an audio processing engine comprising a filter cascade of complex-valued filters; a complex-valued filter such as 4 is configured to derive a complex number from an input signal The personal frequency f signal 'the complex value filter set is configured in the filter cascade, whereby one of the complex value filters is passed to the next complex value filter in the filter cascade. 12. The system of claim 11, wherein the complex value filters are unipolar complex filters. The system of claim 11, wherein the audio processing engine further comprises a reconstruction module configured to perform phase alignment on one or more of the sub-band signals. 14. The system of claim 1, wherein the audio processing engine further comprises a reconstruction module configured to perform amplitude compensation on one or more of the sub-band signals. 15. The system of claim 1, wherein the audio processing engine further comprises a reconstruction module configured to perform a time delay on one or more of the sub-band signals. 16. The system of claim 1, wherein the audio processing engine further comprises a modification module configured to modify the path based on a filter from the filter cascade 127033.doc 200847133 One or more of the sub-band signals. 17. The system of claim 11, wherein the step-by-step includes an adjustment module configured to preprocess the input prior to filtering the wheel input filter using the filter cascade Signal. ^ The machine executing the program can read the medium, and the program can be executed by the machine to execute a method for processing the audio signal, the method includes: the capturing one of the filter cascades The complex value filter filters an input signal to generate a first filtered signal; Subtracting the first filtered signal from the input port to derive a first frequency band signal; capturing a fourth filter cascade to filter the first crossed wave to generate a lower _ The filtered signal is subtracted from the n-th filtered signal to subtract the next chopped signal to derive the next band signal. 19. The machine readable medium of claim 18, wherein the complex money filter and the next complex value filter are unipolar complex value filters. 2. The machine readable medium of claim 18, wherein the method further comprises performing phase alignment on the one or more of the sub-band signals. 21. The machine readable medium of claim 18, wherein the method further comprises performing amplitude compensation on the one or more of the sub-band signals. For example, the machine of claim 18 can read the medium, wherein the method further comprises performing a time delay on one or more of the sub-band signals. 23. The machine readable medium of claim 18, wherein the method further comprises 127033.doc 200847133 preprocessing the input signal prior to filtering the input signal using the filter cascade. 127033.doc
TW96144620A 2006-05-25 2007-11-23 System and method for processing an audio signal TWI421858B (en)

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US9536540B2 (en) 2013-07-19 2017-01-03 Knowles Electronics, Llc Speech signal separation and synthesis based on auditory scene analysis and speech modeling
US9820042B1 (en) 2016-05-02 2017-11-14 Knowles Electronics, Llc Stereo separation and directional suppression with omni-directional microphones
US9838784B2 (en) 2009-12-02 2017-12-05 Knowles Electronics, Llc Directional audio capture
US9978388B2 (en) 2014-09-12 2018-05-22 Knowles Electronics, Llc Systems and methods for restoration of speech components

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US4137510A (en) * 1976-01-22 1979-01-30 Victor Company Of Japan, Ltd. Frequency band dividing filter
US6496795B1 (en) * 1999-05-05 2002-12-17 Microsoft Corporation Modulated complex lapped transform for integrated signal enhancement and coding
US20050228518A1 (en) * 2002-02-13 2005-10-13 Applied Neurosystems Corporation Filter set for frequency analysis

Cited By (4)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US9838784B2 (en) 2009-12-02 2017-12-05 Knowles Electronics, Llc Directional audio capture
US9536540B2 (en) 2013-07-19 2017-01-03 Knowles Electronics, Llc Speech signal separation and synthesis based on auditory scene analysis and speech modeling
US9978388B2 (en) 2014-09-12 2018-05-22 Knowles Electronics, Llc Systems and methods for restoration of speech components
US9820042B1 (en) 2016-05-02 2017-11-14 Knowles Electronics, Llc Stereo separation and directional suppression with omni-directional microphones

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