Abstract:
A signal processing method in cochlear implant is performed by a speech processor and comprises a noise reduction stage and a signal compression stage. The noise reduction stage can efficiently reduce noise in a electrical speech signal of a normal speech. The signal compression stage can perform good signal compression to enhance signals to stimulate cochlear nerves of a hearing loss patient. The patient who uses a cochlear implant performing the signal processing method of the present invention can understand normal speech.

Description:
BACKGROUND OF THE INVENTION 
       [0001]    1. Field of the Invention 
         [0002]    The present invention relates to a signal processing method, and more particularly to a signal processing method in cochlear implant. 
         [0003]    2. Description of Related Art 
         [0004]    Cochlear implant is a surgically implanted electronic device that provides a sense of sound to hearing loss patients. The tremendous progress of the cochlear implant technologies has enabled many hearing loss patients to enjoy high level of speech understanding quality 
         [0005]    Noise reduction and signal compression are critical stages in the cochlear implant. For example, a conventional cochlear implant comprising multiple microphones can enhance the sensed speech volume. However, noises in the sensed speech are also amplified and compressed so as to affect the speech understanding of the hearing loss patient. Besides, the multiple microphones increase hardware cost. 
       SUMMARY OF THE INVENTION 
       [0006]    An objective of the present invention is to provide a signal processing method in cochlear implant. The signal processing method is performed by a speech processor and comprises a noise reduction stage and a signal compression stage. 
         [0007]    The noise reduction stage comprises:
       receiving a t-th noisy frame y t , wherein the t-th noisy frame y t  is from an electrical speech signal y;   reducing noises in the t-th noisy frame y t  to obtain a t-th clean frame x t ; and   outputting the t-th clean frame x t .       
 
         [0011]    The signal compression stage comprises:
       receiving an amplitude envelope of the t-th clean frame x t ;   compressing the t-th clean frame x t  to form a t-th output frame       
 
         [0000]        z   t =α t ×( x   t   − x     t )+   x     t , wherein
         x   t  is a mean of the amplitude envelope of the t-th clean frame x t ;   α t  is a compression factor;   when the t-th output frame z t  is in a monitoring range between an upper boundary and a lower boundary, α t =α t-1 Δα 1  and Δα 1  is a positive value; and   when the t-th output frame z t  is beyond the monitoring range, α t =α t-1 +Δα 2  and Δα 2  is a negative value; and   outputting the t-th output frame z t .       
 
         [0019]    Another objective of the present invention is to provide a signal processing method in cochlear implant. The signal processing method is performed by a speech processor having a noise reduction unit and a signal compressor. The signal compressor has a compression unit, a boundary calculation unit, and a compression-factor-providing unit. The signal processing method comprises a noise reduction stage and a signal compression stage. 
         [0020]    The noise reduction stage is performed by the noise reduction unit and comprises:
       receiving a t-th noisy frame y t , wherein the t-th noisy frame y t  is from an electrical speech signal y;   reducing noises in the t-th noisy frame y t  to obtain a t-th clean frame x t ; and   outputting the t-th clean frame x t .       
 
         [0024]    The signal compression stage is performed by the signal compressor and comprises:
       receiving an amplitude envelope of the t-th clean frame x t  by the compression unit and the boundary calculation unit;   calculating an upper boundary and a lower boundary and transmitting the upper boundary and the lower boundary to the compression-factor-providing unit by the boundary calculation unit;   compressing the t-th clean frame x t  to obtain a t-th output frame z t  and outputting the t-th output frame z t  by the compression unit; and   calculating a compression factor α t  and transmitting the compression factor α t  to the compression unit by the compression-factor-providing unit according to the t-th output frame z t , the upper boundary, and the lower boundary; wherein       
 
         [0000]        z   t =α t ×( x   t   − x     t )+   x     t ; and
         x   t  is a mean of an amplitude envelope of x t , t=1, . . . T,
 
where T is the lengths of the current utterance;
   when the t-th output frame z t  is in a monitoring range between the upper boundary and the lower boundary, α t =α t-1 +Δα 1  and Δα 1  is a positive value; and   when the t-th output frame z t  is beyond the monitoring range, α t =α t-1 +Δα 2  and Δα 2  is a negative value.       
 
         [0032]    Based on the signal processing method of the present invention, the noise reduction stage can efficiently reduce noise in the electrical speech signal of the normal speech, and the signal compression stage can perform good signal compression to enhance signals to stimulate cochlear nerves of a hearing loss patient, such that the hearing loss patient can well understand the normal speech. The present invention performs the noise reduction stage and the signal compression stage to improve performance of the cochlear implant instead of using multiple microphones. Compared with the conventional cochlear implant with multiple microphones, the present invention would not increase the hardware cost. 
         [0033]    Embodiments of the present invention are described in detail as follows. 
     
    
     
       BRIEF DESCRIPTION OF THE DRAWINGS 
         [0034]      FIG. 1  is a circuit block diagram of a cochlear implant; 
           [0035]      FIG. 2  is a detailed circuit diagram including a speech processor connected to a microphone and pulse generators of the present invention; 
           [0036]      FIG. 3  is a schematic view of a single-layered DAE-based NR structure; 
           [0037]      FIG. 4( a )  shows an amplitude envelope of a clean speech signal; 
           [0038]      FIG. 4( b )  shows an amplitude envelope of a noisy speech signal; 
           [0039]      FIG. 4( c )  shows an amplitude envelope detected by a conventional log-MMSE estimator; 
           [0040]      FIG. 4( d )  shows an amplitude envelope detected by a conventional KLT estimator; 
           [0041]      FIG. 4( e )  shows an amplitude envelope detected by the present invention; 
           [0042]      FIG. 5  is a circuit block diagram of one channel of the speech processor of the present invention; 
           [0043]      FIG. 6  is a waveform diagram of an amplitude envelope detected by an envelope detection unit; and 
           [0044]      FIG. 7  is a waveform diagram of an output frame generated by the signal compressor of the present invention. 
       
    
    
     DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENT 
       [0045]    With reference to  FIG. 1 , a basic and conventional configuration of a circuit block diagram of a cochlear implant comprises a microphone  11 , a speech processor  12 , a transmitter  13 , a receiver  14 , a pulse generator  15 , and an electrode array  16 . The microphone  11  and the speech processor  12  are assembled to be mounted on a patient&#39;s ear. The transmitter  13  is adapted to be attached on an exterior surface of the patient&#39;s head skin. The receiver  14 , the pulse generator  15 , and the electrode array  16  are implanted under the patient&#39;s head skin. 
         [0046]    The microphone  11  is an acoustic-to-electric transducer that converts a normal speech in air into an electrical speech signal. The speech processor  12  receives the electrical speech signal and converts the electrical speech signal into multiple output sub-speech signals in different frequencies. The transmitter  13  receives the output sub-speech signals from the speech processor  12  and wirelessly sends the output sub-speech signals to the receiver  14 . The pulse generator  15  receives the output sub-speech signals from the receiver  14  and generates different electrical pulses based on the output sub-speech signals to the electrode array  16 . The electrode array  16  has multiple electrodes  161  respectively and electrically connected to different cochlear nerves of the hearing loss patient&#39;s inner ear. Therefore, the electrodes  161  respectively output the electrical pulses to stimulate the cochlear nerves, such that the patient can hear something approximating the normal speech. 
         [0047]    In more details, with reference to  FIG. 2 , the speech processor  12  has multiple channels including a first channel, a second channel, . . . , an i-th channel, . . . , and a n-th channel, wherein i and n are positive integrals. Each one of the channels has a band-pass filter  121 , an envelope detection unit  122 , and a signal compressor  123 . The envelope detection unit  122  is used to detect an amplitude envelope of a signal and can have a rectifier  124  and a low-pass filter  125 . In the present invention, a noise reduction unit  126  is added. The noise reduction unit  126  is connected between the microphone  11  and the band-pass filters  121  of each one of the channels. In time domain, when the noise reduction unit  126  receives the electrical speech signal from the microphone  11 , the noise reduction unit  126  segments the electrical speech signal into several continuous frames to reduce noise of the frames. For example, when a time length of the electrical speech signal is 3 seconds, the noise reduction unit  126  can segment the electrical speech signal into 300 continuous frames, wherein a time length of each one of the frames of the electrical speech signal is 10 milliseconds. 
         [0048]    Based on the above configuration, the band-pass filter  121  of each one of the channels sequentially receives the frames of the electrical speech signal from the noise reduction unit  126 . The band-pass filter  121  of each one of the channels can preserve elements of each one of the frames of the electrical speech signal within a specific frequency band and remove elements beyond the specific frequency band from such frame. The specific frequency bands of the band-pass filters  121  of the channels are different from each other. Afterwards, the amplitude envelopes of the frames of the electrical speech signal are detected by the envelope detection units  122  and provided to the signal compressors  123  respectively. 
         [0049]    The present invention relates to a noise reduction stage performed by the noise reduction unit  126  and a signal compression stage performed by the signal compressor  123 . The noise reduction stage and the signal compression stage are respectively described below. 
         [0050]    1. Noise Reduction Stage 
         [0051]    The noise reduction unit  126  can be performed in a DDAE (deep denoising autoencoder)-based NR (noise reduction) structure. The DDAE-based NR structure is widely used in building a deep neural architecture for robust feature extraction and classification. In brief, with reference to  FIG. 3 , a single-layered DAE-based NR structure comprises an input layer  21 , a hidden layer  22 , and an output layer  23 . The DDAE-based NR structure is a multiple-layered DAE-based NR structure comprising the input layer  21 , the output layer  23 , and multiple hidden layers  22 . Because the parameter estimation and speech enhancement procedure of DDAE is the same as those of single-layered DAE, we only present the parameter estimation and speech enhancement for the single-layered DAE for ease of explanation. The same parameter estimation and speech enhancement procedures can be followed for the DDAE. 
         [0052]    The input layer  21  receives an electrical speech signal y from the microphone  11  and segments the electrical speech signal y into a first noisy frame y 1 , a second noisy frame y 2 , . . . , a t-th noisy frame y t , . . . , and a T-th noisy frame y T , wherein T is a length of the current utterance. For the elements in the t-th noisy frame y t , the noise reduction unit  126  reduces noise in the t-th noisy frame y t  to form a t-th clean frame x t . Afterwards, the output layer  23  sends the t-th clean frame x t  to the channels of the speech processor  12  respectively. 
         [0053]    A relationship between the t-th noisy frame y t  and the t-th clean frame x t  can be represented as: 
         [0000]        x   t   =W   2   h ( y   t )+ b   2   (1)
       where   h(y t ) is a function including W 1  and b 1  in time domain;   W 1  and W 2  are default connection weights in time domain; and   b 1  and b 2  are default vectors of biases of the hidden layers  22  of the DDAE-based NR structure in time domain.       
 
         [0058]    Besides, in another embodiment, the relationship between the t-th noisy frame y t  and the t-th clean frame x t  can be represented as 
         [0000]        x   t   =InvF {( W   2   ′h ′( F{y   t })+ b   2 ′)}  (2)
       where   F { } is a Fourier transform function to transfer the t-th noisy frame y t , from time domain to frequency domain;   h′( ) is a function including W 1 ′ and b 1 ′;   W 1 ′ and W 2 ′ are default connection weights in frequency domain;   b 1 ′ and b 2 ′ are default vectors of biases of the hidden layers  22  of the DDAE-based NR structure in frequency domain; and   InvF { } is an inverse Fourier transform function to obtain the t-th clean frame x t .       
 
         [0065]    According to experimental result, the t-th clean frame x t  deduced from the Fourier transform and the inverse-Fourier transform as mentioned above has better performance than which without the Fourier transform and the inverse-Fourier transform. 
         [0066]    For the time domain based method as shown in equation (1), h(y t ) can be represented as: 
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         [0067]    For the frequency domain based method as shown in equation (2), h′ (F {y t }) can be represented as: 
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         [0068]    Regarding the parameters including W 1 , W 2 , b 1 , and b 2  in time domain or W 1 ′, W 2 ′, b 1 ′, and b 2 ′ in frequency domain, they are preset in the speech processor  12 . 
         [0069]    For example, in time domain, the parameters including W 1 , W 2 , b 1 , and b 2  in equations (1) and (3) are obtained from a training stage. Training data including a clean speech sample u and a corresponding noisy speech sample v. Likewise, the clean speech sample u is segmented into several clean frames u 1 , u 2 , . . . , u T′ , and the noisy speech sample v is segmented into several noisy frames v 1 , v 2 , . . . , v T′ , wherein T′ is a lengths of a training utterance. 
         [0070]    The parameters including W 1 , W 2 , b 1 , and b 2  of equation (1) and equation (3) are optimized based on the following objective function: 
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         [0071]    In equation (5), θ is a parameter set {W 1 , W 2 , b 1 , b 2 }, T′ is a total number of the clean frames u 1 , u 2 , . . . , u T′ , and η is a constant used to control the tradeoff between reconstruction accuracy and regularization on connection weights (for example, η can be set as 0.0002). The training data including the clean frames u 1 , u 2 , . . . , u T′  and the training parameters of W 1-test , W 2-test , b 1-test  and b 2-test  can be substituted into the equation (1) and equation (3) to obtain a reference frame ū t . When the training parameters of W 1-test , W 2-test , b 1-test , and b 2-test  can make the reference frame ū t , mostly approximate the clean frames u t , such training parameters of W 1-test , W 2-test , b 1-test , and b 2-test  are taken as the parameters of W 1 , W 2 , b 1 , and b 2  of equation (1) and equation (3). Besides, when the noisy speech sample v approximates the electrical speech signal y, the training result of the parameters of W 1 , W 2 , b 1 , and b 2  can be optimized. The optimization of equation (5) can be solved by using any unconstrained optimization algorithm. For example, a Hessian-free algorithm can be applied in the present invention. 
         [0072]    After training, optimized parameters including W 1 , W 2 , b 1 , and b 2  are obtained to be applied to equation (1) and equation (3) for real noise reduction application. 
         [0073]    Besides, in frequency domain, the parameters including W 1 ′, W 2 ′, b 1 ′, and b 2 ′ of equation (2) and equation (4) are optimized based on the following objective function: 
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         [0074]    In equation (6), θ is a parameter set {W 1 ′, W 2 ′, b 1 ′, b 2 ′}, T′ is a total number of the clean frames u 1 , u 2 , . . . , u T′ , and η is a constant used to control the tradeoff between reconstruction accuracy and regularization on connection weights (for example, η can be set as 0.0002). The training data including the clean frames u 1 , u 2 , . . . , u T′  and the training parameters of W 1-test ′, W 2-test ′, b 1-test ′, and b 2-test ′ can be substituted into the equation (2) and equation (4) to obtain a reference frame ū t . When the training parameters of W 1-test ′, W 2-test ′, b 1-test ′, and b 2-test ′ can make the reference frame ū t , mostly approximate the clean frames u t , such training parameters of W 1-test ′, W 2-test ′, b 1-test ′, and b 2-test ′ are taken as the parameters of W 1 ′, W 2 ′, b 1 ′, and b 2 ′ of equation (2) and equation (4). Besides, when the noisy speech sample v approximates the electrical speech signal y, the training result of the parameters of W 1 ′, W 2 ′, b 1 ′, and b 2 ′ can be optimized. The optimization of equation (6) can be solved by using any unconstrained optimization algorithm. For example, a Hessian-free algorithm can be applied in the present invention. 
         [0075]    After training, optimized parameters including W 1 ′, W 2 ′, b 1 ′, and b 2 ′ are obtained to be applied to equation (2) and equation (4) for real noise reduction application. 
         [0076]    With reference to  FIGS. 4( a ) and 4( b ) ,  FIG. 4( a )  shows an amplitude envelope of a clean speech signal and  FIG. 4( b )  shows an amplitude envelope of a noisy speech signal.  FIG. 4( c )  shows an amplitude envelope detected by a conventional log-MMSE (minimum mean square error) estimator.  FIG. 4( d )  shows an amplitude envelope detected by a conventional KLT (Karhunen-Loeve transform) estimator.  FIG. 4( e )  shows an amplitude envelope detected by the present invention. Comparing  FIG. 4( e )  with  FIG. 4( a ) , the detection result of the present invention is most approximate to the clean speech signal, which means the noise is removed. Comparing  FIG. 4( b )  with  FIGS. 4( c ) and 4( d ) , the detection results as illustrated in  FIGS. 4( c ) and 4( d )  are still noisy. 
         [0077]    According to experimental result as mentioned above, the signal performances of the conventional log-MMSE estimator and the KLT estimator are not as good as the present invention. The present invention has better noise reducing efficiency. 
         [0078]    2. Signal Compression Stage 
         [0079]    With reference to  FIGS. 2 and 5 , for the i-th channel of the speech processor  12 , the signal compressor  123  receives an amplitude envelope of the t-th clean frame x t  within the specific frequency band from the noise reduction unit  126  through the band-pass filter  121  and the envelope detection unit  122 . The amplitude envelope  30  of the t-th clean frame x t  is illustrated in  FIG. 6 . As shown in  FIG. 6 , the amplitude envelope  30  of t-th clean frame x t  is time-varying. 
         [0080]    The signal compressor  123  of the present invention comprises a compression unit  127 , a boundary calculation unit  128 , and a compression-factor-providing unit  129 . The compression unit  127  and the boundary calculation unit  128  are connected to the envelope detection unit  122  to receive the amplitude envelope  30  of the t-th clean frame x t , real-time. With reference to  FIGS. 5 and 6 , the boundary calculation unit  128  can detect an upper boundary UB and a lower boundary LB in the amplitude envelope of the t-th clean frame x t . The calculation result of the upper boundary UB and the lower boundary LB are transmitted to the compression-factor-providing unit  129 . The upper boundary UB and the lower boundary LB can be calculated by: 
         [0000]        UB= x     t +α 0 ×(max( x   t )−   x     t )  (7)
 
         [0000]        LB= x     t +α 0 ×(min( x   t )−   x     t )  (8)
 
         [0081]    where α 0  is an initial value. 
         [0082]    The compression unit  127  receives the amplitude envelope  30  of the t-th clean frame x t  and outputs a t-th output frame z t . Inputs of the compression-factor-providing unit  129  is connected to an input of the compression unit  127 , an output of the compression unit  127 , and an output of the boundary calculation unit  128  to receive a calculating result of the upper boundary UB, the lower boundary LB, and the t-th output frame z t . An output of the compression-factor-providing unit  129  is connected to the input of the compression unit  127 , such that the compression-factor-providing unit  129  provides a compression factor α t  to the compression unit  127 . The compression factor α t  is determined according to a previous compression factor α t-1 , the upper boundary UB, the lower boundary LB, and the t-th output frame z t . When the t-th output frame z t  is in a monitoring range between the upper boundary UB and the lower boundary LB, the compression factor α t  is expressed as: 
         [0000]      α t =α t-1 +Δα 1   (9)
 
         [0083]    where Δα 1  is a positive value (i.e., Δα 1 =1). 
         [0084]    In contrast, when the t-th output frame z t  is beyond the monitoring range, the compression factor α t  is expressed as: 
         [0000]      α t =α t-1 +Δα 2   (10)
 
         [0085]    where Δα 2  is a negative value (i.e., Δα 2 =−0.1). 
         [0086]    The t-th output frame z t  can be expressed as: 
         [0000]        z   t =α t ×( x   t   − x     t )+   x     t   (11)
 
         [0087]    where  x   t  is a mean of the amplitude envelope of the t-th clean frame x t . 
         [0088]    According to equation (11), the t-th output frame z t  is repeatedly adjusted by the t-th clean frame x t  and the calculation result of UB, LB, and α t . According to experimental result, the signal compression capability is good. As illustrated in  FIG. 7 , speech components A in the t-th output frame z t  are amplified. The speech components A even reach the upper boundary UB. In contrast, noise components B are not exactly amplified. Therefore, the t-th output frame z t  is enhanced to stimulate the cochlear nerves. The user can accurately understand a conversation.