CN100589783C - Electronic cochlea language processing method having S parameter control and based-on Chinese language characteristics - Google Patents
Electronic cochlea language processing method having S parameter control and based-on Chinese language characteristics Download PDFInfo
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- CN100589783C CN100589783C CN200610060581A CN200610060581A CN100589783C CN 100589783 C CN100589783 C CN 100589783C CN 200610060581 A CN200610060581 A CN 200610060581A CN 200610060581 A CN200610060581 A CN 200610060581A CN 100589783 C CN100589783 C CN 100589783C
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Abstract
A speech processing method with S parameter control and based on the characteristics of Chinese language for electronic cochlea includes such steps as pre-treating the speech signals and dividing theminto two channels by external unit, dividing one channel of speech signals into several frequency bands, extracting the envelopes of the signals in each frequency band, extracting the base frequencyof another channel of speech signals, coding, controlling the channel number by S parameter, transmitting to amplitude transmission channel for chosen frequency band, transmitting the information about base frequency and amplitude in the format of command frame to internal unit, receiving them by internal unit, generating electric stimulation pulses, applying them to relative electrode, and stimulating the relative acoustic nerve for restoring the auditognosis of patient.
Description
Technical field:
The present invention relates to cochlear implant, particularly have the cochlear implant method of speech processing based on feature of Chinese language of S parameter control.
Background technology:
Cochlear implant is also referred to as cochlear implant, electronic ear, bionic ear or artificial cochlea, and it is present unique device that can make anacusia patient recover audition.It relies on the physiological function of directly imitating peripheral auditory system with the excited acoustic fibers of weak current, produces the nerve similar to normal human ear and provides pattern, thereby recover patient's audition.
Cochlear implant can help anacusia patient to recover the communication ability, increases that they accept the education, the chance of employment and social communication.Especially no matter deaf youngster is that prelingual or language back are deaf, in the time still can't hearing sound by means of sonifer, will cause them to lose the chance that obtains normal education throughout one's life, increases white elephant for family and society.Cochlear implant can help them to regain tone sense, obtains extraneous information and knowledge, becomes the people useful to society.
The function of cochlear implant voice signal processing section be with acoustical signal be converted to cochlea in the corresponding electrical stimulation signal of electrode, the voice signal Processing Algorithm is the technology of most critical in the cochlear implant system, it has significant effects to the performance of cochlear implant.
Traditional cochlear implant voice signal Processing Algorithm that is applied to clinically can be divided into two big classes, and a class is based on feature extraction, promptly extracts features such as the fundamental frequency of voice signal and formant, produces the stimulus signal of respective electrode then.Mainly comprise F0/F2 algorithm, F0/F1/F2 algorithm and MPEAK (Multi-PEAK) algorithm etc.Developed the second electron-like cochlea voice signal Processing Algorithm afterwards on this basis, promptly voice signal had been carried out the frequency-division section Filtering Processing based on bank of filters.As maximum spectrum peak acoustic processing (Spectral Maxima Sound Processor, abbreviation SMSP) algorithm, spectrum peak extract (SpectralPEAK, abbreviation SPEAK) algorithm, compression simulation (Compressed Analog, abbreviation CA) algorithm, continuous alternately sampling (Continuous Interleaved Sampling, be called for short CIS) multiple algorithms such as algorithm and assembly coding (Advanced CombinedEncoder is called for short ACE) algorithm.
Existing algorithm all is to develop on the basis that English is understood, and does not consider that Chinese etc. has the accent language characteristic.Chinese has the characteristic of oneself, and it is a kind of sound, rhyme, tone language, and wherein tone and intonation have great importance for the correct understanding of Chinese.
Chinese language tone claims the tone of Chinese characters or syllable tone again, and it also has functions such as configuration, boundary, anti-interference, rhetoric except the function with the difference meaning of word and meaning of a word.And the change in pitch of sentence represented in the intonation of Chinese.Different beats makes up the meaning that can change sentence structure and full sentence sometimes in the intonation, and the understanding for semanteme has great importance equally.
Yet the product of clinical use only relies on the extraction of voice amplitude information and transmits the information such as tone that are difficult to effectively transmit Chinese, average recognition rate about 68% under its quiet condition, 99% ± 3.0% mutually far short of what is expected with the normal good hearing experimenter is difficult to satisfy patient's demand.
Therefore can make full use of the Chinese characteristic, more transmission Chinese enrich frequency information, and the cochlear implant speech processing algorithm of more effective adaptation Chinese must be developed again.
Summary of the invention:
The objective of the invention is to the problems referred to above, a kind of cochlear implant method of speech processing based on feature of Chinese language of the S of having parameter control is provided, thereby recover the audition of China's deafness patient more effectively at present cochlear implant existence.
The cochlear implant method of speech processing based on feature of Chinese language that the present invention proposes is a kind of algorithm of introducing the synthetic number of channels of fundamental frequency with the control of S parameter that has, and it may further comprise the steps:
The pretreatment module of a, the external speech processes part of cochlear implant is carried out pretreatment to the voice signal of input, and pretreated voice signal is divided into two-way;
The band filter group of b, external speech processes part is divided into some frequency ranges with pretreated one road voice signal, is extracted the envelope (being amplitude information) of each frequency band signals then respectively by the amplitude information extraction module;
The fundamental frequency information extraction module of c, external speech processes part extracts the fundamental frequency information of pretreated another road voice signal, by coding module the fundamental frequency information that extracts is encoded, then by S parameter control channel number, fundamental frequency information F0 behind the coding is delivered in the amplitude information transmission channel of selected frequency range, and the fundamental frequency information F0 after will encoding with the form of command frame and the amplitude information of selected frequency range send to part in the cochlear implant body by transmitter module;
D, cochlear implant body receive described command frame inner the branch, produce the respective electrode that the electric stimulation pulse sequence acts on part in the cochlear implant body according to the content of command frame, stimulate the acoustic nerve of this electrode institute correspondence position, thereby recover patient's audition.
The pretreatment of voice signal comprises the preemphasis of voice signal and postemphasises, divides functions such as frame, bandpass filtering, preemphasis is in order to keep interchannel spectral balancing, because when acoustical signal is sent by the oral cavity, because lip radiation, have at the high band greater than 1.2kHz-decay of 6dB/Oct, preemphasis can compensate high frequency attenuation.Postemphasising is the preemphasis inverse process.Dividing frame is to be used for realizing cutting apart Audio Processing Unit, thereby makes in the single frames processing unit of voice signal after cutting apart steadily approximate.
The present invention adopts wavelet arithmetic and the method that time domain fundamental frequency Processing Algorithm combines, and has realized accurate extraction and coding to the voice signal fundamental frequency, and its computational speed is fast, the saving internal memory.This method had both solved independent use time domain fundamental frequency Processing Algorithm and had been subjected to interference of noise easily, the shortcoming that extraction effect can descend rapidly along with the decline of signal to noise ratio, though avoided using the method extraction precision of traditional transform domain better again, calculation of complex and the shortcoming big to EMS memory occupation.
The present invention has added S parameter controlling unit in algorithm structure, can be by optionally to the encode redundancy of the information that reduces of fundamental frequency information.This module is controlled the port number that parameter S is the transmission fundamental frequency information by definition, and (begin numeration from low frequency, for example S=1 then has only 1 passage that begins from first passage to transmit frequency information; S=2 then has 2 passages that begin from first passage to transmit frequency information; The rest may be inferred) control and regulate fundamental frequency information and selectively transmit, thus the redundancy of the information of minimizing.Verify that by experiment S=5 both can effectively improve the speech recognition capabilities of cochlear implant patient under noise circumstance, had reduced the complexity of algorithm again.
The present invention utilizes the cochlear implant voice process technology, in conjunction with Matlab and assembler language platform, by extracting and optionally utilizing feature of Chinese language, has realized the cochlear implant speech processing algorithm of suitable China deafness patient use of innovation.Its main effect has:
A) utilize feature of Chinese language, increased the extraction and the transmission of Chinese fundamental frequency information, improved the language identification ability of patient under noise, thus the audition that can better recover China's deafness patient.
B) by the S parameter control, selectively transmit the fundamental frequency information of Chinese, not only fully effectively utilized feature of Chinese language, also greatly reduce the complexity of algorithm.
C) method of utilizing low-pass filtering and Hilbert conversion to be used in combination realizes the extraction of amplitude information, both effectively followed the tracks of the information of voice amplitude of variation, avoid the too much details burr of independent application Hilbert alternative approach again, influenced the weakness that amplitude information is expressed.
D) amplitude information when compression, utilize broken line to fit the assembly algorithm of non-linear compression, simplified the complexity of compression algorithm, the multiplication and the addition that have used when the division in the traditional algorithm has been made into to fit are fit to implement of hardware system more.
Description of drawings:
Fig. 1 is the theory diagram based on the cochlear implant method of speech processing of feature of Chinese language that originally has S parameter control;
Fig. 2 is its minute frequency band bandpass filtering program flow diagram;
Fig. 3 is its amplitude information extraction procedure flow chart;
Fig. 4 is that its broken line fits the condensing routine flow chart.
The specific embodiment:
Below in conjunction with description of drawings the specific embodiment of the present invention.
Fig. 1 is a total algorithm structured flowchart of the present invention.With reference to Fig. 1, through after the pretreatment, the one tunnel carries out frequency band division and envelope extraction to the external speech processes part of cochlear implant with voice signal; Another road utilizes small echo to promote and extracts fundamental frequency.The fundamental frequency information that extracts is under the regulation and control of S parameter, only be passed to the part low-frequency range (for example only being passed to 5 frequency ranges during S=5) of selection near cupula cochleae, fundamental frequency information F0 behind the amplitude information of selected low-frequency range and the coding sends donor internal stimulus circuit with the form of command frame by transmitter module, and body internal stimulus circuit produces electric stimulation pulse in view of the above.Each command frame occlusion body internal stimulus circuit produces the full detail of an electric stimulation pulse, when body internal stimulus circuit produces electric stimulation pulse according to command frame, amplitude information in the command frame is used for modulating the amplitude of the electric stimulation pulse that is produced, the speed that the fundamental frequency information F0 in the command frame is used for modulating the electric stimulation pulse that is produced.Originally the cochlear implant method of speech processing based on feature of Chinese language that has the control of S parameter specifically may further comprise the steps:
The pretreatment module of a, the external speech processes part of cochlear implant is carried out pretreatment to the voice signal of input, and pretreated voice signal is divided into two-way;
The band filter group of b, external speech processes part is divided into some frequency ranges with pretreated one road voice signal, is extracted the envelope (being amplitude information) of each frequency band signals then respectively by the amplitude information extraction module;
The fundamental frequency information extraction module of c, external speech processes part extracts the fundamental frequency information of pretreated another road voice signal, by coding module the fundamental frequency information that extracts is encoded, then by S parameter control channel number, fundamental frequency information F0 behind the coding is delivered in the amplitude information transmission channel of selected frequency range, and the fundamental frequency information F0 after will encoding with the form of command frame and the amplitude information of selected frequency range send to part in the cochlear implant body by transmitter module:
D, cochlear implant body receive described command frame inner the branch, produce the respective electrode that the electric stimulation pulse sequence acts on part in the cochlear implant body according to the content of command frame, stimulate the acoustic nerve of this electrode institute correspondence position, thereby recover patient's audition.
Described command frame adopts long-frame structure, body internal stimulus circuit can produce an electric stimulation pulse according to each frame data, each command frame comprises amplitude, pulsewidth and the rate information of the electric stimulation pulse that will produce, and also comprises information such as electrode that this electric stimulation pulse will stimulate and stimulation mode.
The amplitude information that extracts among the step b is used for, the amplitude of modulating the electric stimulation pulse that is produced when body internal stimulus circuit produces electric stimulation pulse; Fundamental frequency information F0 among the step c behind the coding is used for, the speed of modulating the electric stimulation pulse that is produced when body internal stimulus circuit produces electric stimulation pulse.
The pretreatment of voice signal comprises the preemphasis of voice signal and postemphasises, divides functions such as frame, bandpass filtering, preemphasis is in order to keep interchannel spectral balancing, because when acoustical signal is sent by the oral cavity, because lip radiation, have at high band-decay of 6dB/Oct greater than 1.2kHz, preemphasis can compensate high frequency attenuation, and preemphasis adopts the first-order difference formula: y (n)=x (n)-ax (n-1) realizes, the coefficient a=0.9 of first-order difference.Postemphasising is the preemphasis inverse process.Dividing frame is to be used for realizing cutting apart Audio Processing Unit, is similar in the single frames processing unit of voice signal after cutting apart steadily thereby make, and divides frame processing and utilizing sliding window function y (n)=x (n) w (n) to realize, window function is selected Hamming window
Frame length is 16 milliseconds, and it is 8 milliseconds that frame moves.Only in voice signal length is 10-20 millisecond scope, can think that just this signal is to be similar to stably.X in the described formula (n) is an input speech signal, and y (n) is the output voice signal.
Fig. 2 is that the present invention divides frequency band bandpass filtering program flow diagram.The present invention selects in Butterworth method design IIR band filter group or with window function method design FIR band filter group, in order to reduce the nonlinear phase link as far as possible, the signal that is different frequency is through filtered phase shift difference, and the present invention finally adopts Hamming window design FIR band filter group.In order to guarantee that the bank of filters parameter conforms to the cochlea frequency characteristic, promptly keep permanent Q characteristic, the filter order of selecting for use has well been finished the function of dividing the frequency band bandpass filtering up to 40.Also can adopt the IIR band filter group of Butterworth method design.
Fig. 3 is an amplitude information extraction procedure flow chart.The method that the present invention adopts low-pass filtering and Hilbert conversion to be used in combination realizes the extraction of amplitude information, specifically may further comprise the steps: with each frequency band signals all wave rectification; Constitute analytic signal after the signal process Hilbert conversion after the rectification, the mould of analytic signal is the envelope of this signal; High-frequency Interference composition with in the mould of the described analytic signal of low pass filter filters out draws amplitude information.The cut-off frequency of low pass filter is generally got 400Hz, can require to do rationally to adjust according to reality.Method for designing mainly contains two kinds of IIR and FIR, and the present invention has adopted the FIR method when the band filter group designs, very big to the whole system resource overhead, in order to save resource, has selected 18 rank IIR Butterworth low pass filters for use when extracting envelope.
After amplitude information extracts, because the patient is limited for the dynamic range of electricity irritation, after selecting the frequency range that will stimulate, need carry out the compression of dynamic range to the amplitude of this frequency band signals, the present invention adopts log compression function Y=Alog (x)+B the envelope signal of each passage to be carried out the non-linear compression of dynamic range, x in the formula is the preceding voice signal of compression, and Y is determined by patient's the threshold of audibility and threshold of pain parameter for compression back signal, A, B.After compression function is determined, when on DSP, realizing with assembler language, present embodiment adopts 16 sections piecewise linear approximation logarithmic functions, and the parameter with simple linear list is preserved turning point fits the envelope signal that calculates after each passage compresses according to broken line rule linearity during compression.For 8 channel electron cochlea, need the memory space of no more than 0.5K word.When needs are adjusted mapping curve, only need revise the parameter of look-up table, can finish the adjustment of condensing routine.Figure 4 shows that the present embodiment broken line fits the condensing routine flow chart.
Among the above-mentioned steps c, the fundamental frequency information extraction module adopts average magnitude difference function
In the formula, (i=1～N) be the windowing voice signal, j is for postponing (j=1～N-1), extract the fundamental frequency information of pretreated another road voice signal for x (i).If voice signal is a cyclical signal, the average magnitude difference function that obtains after then postponing to subtract each other significantly " paddy " value will occur at the integral multiple place of its pitch period.The size of pitch period is tried to achieve in the position that the present invention puts by calculating " paddy ", and the inverse of pitch period is fundamental frequency.
Before the fundamental frequency information that extracts voice signal, at first utilize wavelet arithmetic that voice signal is carried out frequency band division and denoising, the wavelet transformation that the present invention is based on wavelet arithmetic is called second filial generation wavelet transformation, it comprises signal is divided, predicts and upgrades, through after the iteration of certain number of times, just calculate the multistage decomposition of signal.The fundamental frequency that the profile information that the present invention adopts the 4th to promote enters next step extracts, and has made full use of the calculating on the throne of wavelet arithmetic, characteristics such as efficient, parallel.
Described S parameter control channel number, the method that fundamental frequency information F0 behind the coding is delivered in the amplitude information transmission channel of selected frequency range is: definition control parameter S is for transmitting the port number of fundamental frequency information, and the fundamental frequency information behind the coding is delivered in the amplitude information transmission channel near S frequency range of cupula cochleae.Verify that by experiment S=5 both can effectively improve the speech recognition capabilities of cochlear implant patient under noise circumstance, had reduced the complexity of algorithm again.
Claims (10)
1, a kind of external method of speech processing of the cochlear implant based on feature of Chinese language with the control of S parameter is characterized in that may further comprise the steps:
The pretreatment module of a, the external speech processes part of cochlear implant is carried out pretreatment to the voice signal of input, and pretreated voice signal is divided into two-way;
The band filter group of b, described external speech processes part is divided into some frequency ranges with pretreated one road voice signal, is extracted the amplitude information of each frequency band signals then respectively by the amplitude information extraction module;
The fundamental frequency information extraction module of c, described external speech processes part extracts the fundamental frequency information of pretreated another road voice signal, by coding module the fundamental frequency information that extracts is encoded, then by S parameter control channel number, fundamental frequency information F0 behind the coding is delivered in the amplitude information transmission channel of selected frequency range, and the fundamental frequency information F0 after will encoding with the form of command frame and the amplitude information of selected frequency range send by transmitter module.
2, the external method of speech processing of the cochlear implant based on feature of Chinese language with the control of S parameter according to claim 1 is characterized in that: described band filter group is to adopt the FIR band filter group of Hamming window design or the IIR band filter group that designs with the Butterworth method.
3, the external method of speech processing of the cochlear implant based on feature of Chinese language with the control of S parameter according to claim 1, it is characterized in that the method that fundamental frequency information F0 behind the described S parameter control coding is delivered in the amplitude information transmission channel of selected frequency range is: definition control parameter S is for transmitting the port number of fundamental frequency information F0, and the fundamental frequency information F0 behind the coding is delivered in the amplitude information transmission channel near S frequency range of cupula cochleae.
4, the external method of speech processing of the cochlear implant based on feature of Chinese language with the control of S parameter according to claim 1, it is characterized in that: described pretreatment comprises preemphasis and divides frame, preemphasis adopts the first-order difference formula: y (n)=x (n)-ax (n-1) realizes, the coefficient a=0.9 of first-order difference; Divide frame processing and utilizing sliding window function y (n)=x (n) ... w (n) realizes that window function is selected Hamming window
Frame length is 16 milliseconds, and it is 8 milliseconds that frame moves, and described x (n) is an input speech signal, and y (n) is the output voice signal.
5, the external method of speech processing of the cochlear implant based on feature of Chinese language with the control of S parameter according to claim 1, it is characterized in that: in step b, the method that the amplitude information extraction module extracts the amplitude information of each frequency band signals may further comprise the steps:
With each frequency band signals all wave rectification;
Constitute analytic signal after the signal process Hilbert conversion after the rectification, the mould of analytic signal is the envelope of this frequency band signals;
With the High-frequency Interference in the mould of the described analytic signal of low pass filter filters out.
6, the external method of speech processing of the cochlear implant based on feature of Chinese language with the control of S parameter according to claim 1, it is characterized in that: in step b, after the amplitude information extraction module extracts the amplitude information of each frequency band signals, adopt log compression function Y=Alog (x)+B the envelope of each frequency band signals to be carried out the non-linear compression of dynamic range, x in the formula is the preceding voice signal of compression, Y is determined by patient's the threshold of audibility and threshold of pain parameter for compression back signal, A, B.
7, the external method of speech processing of the cochlear implant based on feature of Chinese language with the control of S parameter according to claim 1 is characterized in that: in step c, the fundamental frequency information extraction module adopts average magnitude difference function
Extract the fundamental frequency information of pretreated another road voice signal, (i=1～N) is the windowing voice signal to x in the formula (i), and j is for postponing (j=1～N-1).
8, the external method of speech processing of the cochlear implant based on feature of Chinese language with the control of S parameter according to claim 1, it is characterized in that: before the fundamental frequency information that extracts voice signal, at first utilize wavelet arithmetic that voice signal is carried out frequency band division and denoising, described wavelet arithmetic comprises signal is divided, predicts and upgrades, through after the iteration of certain number of times, calculate the multistage decomposition of signal.
9, the external method of speech processing of the cochlear implant based on feature of Chinese language with the control of S parameter according to claim 1, it is characterized in that: described command frame adopts long-frame structure, each command frame comprises amplitude, pulsewidth and the rate information of the electric stimulation pulse that will produce, and also comprises electrode and stimulation mode that this electric stimulation pulse will stimulate.
10, the external method of speech processing of the cochlear implant based on feature of Chinese language with the control of S parameter according to claim 1, it is characterized in that: the amplitude information that extracts among the step b is used for, the amplitude of modulating the electric stimulation pulse that is produced when body internal stimulus circuit produces electric stimulation pulse; Fundamental frequency information F0 among the step c behind the coding is used for, the speed of modulating the electric stimulation pulse that is produced when body internal stimulus circuit produces electric stimulation pulse.
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Families Citing this family (5)
| Publication number | Priority date | Publication date | Assignee | Title |
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| CN101582260B (en) * | 2009-05-20 | 2012-01-11 | 王卫东 | Non-amplitude modulation speech coding method and device in electrical cochlea |
| CN103892939B (en) * | 2014-04-11 | 2016-03-23 | 浙江诺尔康神经电子科技股份有限公司 | Improve language processing device for artificial cochlea and the method for Chinese tone recognition rate |
| CN105513604B (en) * | 2016-01-05 | 2022-11-18 | 浙江诺尔康神经电子科技股份有限公司 | Fundamental frequency contour extraction artificial cochlea speech processing method and system |
| CN109859768A (en) * | 2019-03-12 | 2019-06-07 | 上海力声特医学科技有限公司 | Artificial cochlea's sound enhancement method |
| CN110111800B (en) * | 2019-04-04 | 2021-05-07 | 深圳信息职业技术学院 | Frequency band division method and device of electronic cochlea and electronic cochlea equipment |
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