CN106714064A - Artificial cochlea audio real-time processing system and method - Google Patents
Artificial cochlea audio real-time processing system and method Download PDFInfo
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- CN106714064A CN106714064A CN201710111151.8A CN201710111151A CN106714064A CN 106714064 A CN106714064 A CN 106714064A CN 201710111151 A CN201710111151 A CN 201710111151A CN 106714064 A CN106714064 A CN 106714064A
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
- H04R25/00—Deaf-aid sets, i.e. electro-acoustic or electro-mechanical hearing aids; Electric tinnitus maskers providing an auditory perception
- H04R25/50—Customised settings for obtaining desired overall acoustical characteristics
- H04R25/505—Customised settings for obtaining desired overall acoustical characteristics using digital signal processing
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
- H04R2225/00—Details of deaf aids covered by H04R25/00, not provided for in any of its subgroups
- H04R2225/43—Signal processing in hearing aids to enhance the speech intelligibility
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Abstract
The present invention discloses an artificial cochlea audio real-time processing system and method. The system comprises a sound signal collection unit, a DSP processing unit, a DSP signal output unit, a DSP-PC data conversion unit, a PC end signal real-time collection unit, a PC end real-time processing unit, and a PC end real-time debugging unit. The sound signal collection unit is connected with the DSP processing unit and comprises a collection module and an analog-to-digital conversion module. The collection module collects external sound, and then the analog-to-digital conversion module outputs a digital signal. The DSP processing unit is connected with the DSP signal output unit, at least comprises a fast Fourier transform module, a frequency division processing module and an envelope extraction module, and caches a processed signal to the DSP signal output unit. The DSP signal output unit performs outputting in a serial I2S format. Through adoption of the artificial cochlea audio real-time processing system and method, the DSP processing result and process can be output, debugged and demonstrated in real time.
Description
Technical field
The invention belongs to field of signal processing, more particularly to a kind of artificial cochlea's audio real time processing system and method.
Background technology
Artificial cochlea is mainly to aid in severe and pole severe deafness patient recaptures the hearing-aid device of hearing, and it is sayed by external machine
Language processor and et al. Ke body are constituted.Wherein speech processor gathers external sound signal by microphone, then by sound
Sound pretreatment and sound coding strategy etc., voice signal are converted to the pulse signal of each frequency band, finally by the side of radio frequency
The signal that formula transmitting is encoded；After implant decoding chip is collected into radiofrequency signal, by being implanted into the electrod-array of cochlea, bypass
The external hair cell of cochlea directly stimulates auditory nerve, patient is recaptured hearing.
In the signal processing system of artificial cochlea, the quality of sound pretreatment and acoustic coding algorithm is directly determined
Patient's listens acoustic mass, therefore, higher-quality signal processing algorithm is researched and developed and optimizes, it is to improve cochlear implant life
The Main Means of quality.And at present not suitable for the algorithm development system of artificial cochlea, this is also to the work of research staff
With many inconveniences.Therefore, a algorithm development system suitable for artificial cochlea is released, is conducive to promoting relevant industries
Technology development, can preferably lift the quality of life for listening barrier crowd.
The algorithm development system of existing language processing device for artificial cochlea, mainly relies on PC (personal computer
People's computer) simulation software, it is necessary to the one section of voice signal that will be collected to import algorithm simulating as the signal input of algorithm soft
Part, then could be detected by simulation software disposed of in its entirety, after having processed and demonstrate result.This kind of method can not be timely and effective
The influence that the change of ground reaction input signal and surrounding environment is exported to algorithm, without real-time.
Meanwhile, implementation phase is developed in the algorithm DSP of language processing device for artificial cochlea, adjusted without particularly convenient DSP algorithm
Test system.In order to whether the pilot process that debugging algorithm is realized is correct, and the change procedure of some distinctive signals is checked in DSP
In it is whether consistent with the PC algorithm simulating stages, generally will need observation signal store in the internal memory of DSP derive observation.
But due to being limited by its memory storage capacity, the signal of finite length can only be observed, and not have real-time equally.
The content of the invention
In view of this, it is an object of the invention to provide a kind of artificial cochlea's audio real time processing system and method, can be real
When output, debugging and demonstration PC end simulation softwares and DSP in algorithm result and process.
To reach above-mentioned purpose, the invention provides a kind of artificial cochlea's audio real time processing system, including voice signal
Collecting unit, DSP processing units, DSP signal output units, DSP to PC Date Conversion Units, PC end signal Real-time Collection lists
Unit, the real-time debugging unit of PC ends real-time processing unit and PC ends, wherein,
The sound signal collecting unit is connected with the DSP processing units, including acquisition module and analog-to-digital conversion module,
After acquisition module sound to external world is acquired, through analog-to-digital conversion module output digit signals；
The DSP processing units are connected with the DSP signal output units, at least including fast Fourier transform module,
Scaling down processing module and envelope extract block, the signal after treatment is cached to the DSP signal output units, and DSP signals are defeated
Go out unit to be exported with serial i 2S forms；
The input of DSP to the PC Date Conversion Units is connected with the output of the DSP signal output units, exports and institute
The connection of PC end signal Real-time Collections unit is stated, at least turns USB hardware module including I2S；
The PC end signals Real-time Collection unit, the PC ends real-time processing unit and the real-time debugging unit in PC ends connect successively
Connect, wherein PC end signals Real-time Collection unit at least includes continuous framing sampling module, the syllable fragment for gathering voice signal is defeated
Go out carries out phonetic algorithm treatment to PC ends real-time processing unit, is debugged in real time by the real-time debugging unit in PC ends after treatment.
Preferably, also it is connected with the DSP signal output units including DSP demonstration units, it is defeated to being buffered in DSP signals
The voice signal gone out in unit synchronizes demonstration.
Preferably, also it is connected with the PC ends real-time processing unit, by real-time processing including PC ends real time demonstration unit
Result or the pilot process synchronous demonstrating for the treatment of.
Preferably, the PC end signals Real-time Collection unit includes the USB module, cache module and the PC framings that are sequentially connected
Acquisition module, the parallel data stream that the USB module will be transmitted is put into cache module, and the data in cache module are constituted
Queue, queue length sets the buffer memory capacity for changing cache module, and last PC framings acquisition module is acquired.
Based on above-mentioned purpose, present invention also offers a kind of artificial cochlea's audio real-time processing side of use said system
Method, comprises the following steps：
After sound signal collecting unit collected sound signal, data signal is converted to through analog-to-digital conversion module, exported to DSP
Processing unit；
DSP signal output units are transferred to after DSP processing unit processes, the data that will debug needs are extracted and examined
Survey, PC ends are transferred to by DSP to PC Date Conversion Units；
The data of the time span required for PC ends carry out continuous framing sampled acquisition；
After carrying out phonetic algorithm treatment, algorithm debugging is carried out.
Preferably, after being transferred to DSP signal output units after the processing unit processes through DSP, by DSP demonstration units pair
Voice signal plays out demonstration.
Preferably, it is described carry out phonetic algorithm treatment after, the pilot process of result or treatment is synchronized into demonstration.
Preferably, the data of the time span required for PC ends carry out continuous framing sampled acquisition, are by USB moulds
Block transmission parallel data stream is placed in cache module, and the data in cache module then are constituted into queue, big by setting queue
The small buffer memory capacity to change cache module, by setting the port number of sample rate, the sampling number of each frame and sound to language
The message number collection of framing in real time.
The beneficial effects of the present invention are：When effectively can solve to be researched and developed in PC simulation softwares, using the conduct that prerecords
Problem of the method for algorithm input signal without real-time；And in the digital information processing system implementation process of algorithm,
Algorithm is caused to be only capable of observing the problem of very limited amount of data-signal when debugging due to DSP data storages low memory；And
When algorithm evaluation is demonstrated, the problem without real-time.
Brief description of the drawings
In order that the purpose of the present invention, technical scheme and beneficial effect are clearer, the present invention provides drawings described below and carries out
Explanation：
Fig. 1 is a kind of artificial cochlea's audio real time processing system structural representation of the embodiment of the present invention 1；
Fig. 2 is a kind of artificial cochlea's audio real time processing system structural representation of the embodiment of the present invention 2；
The step of Fig. 3 is a kind of artificial cochlea's audio real-time processing method of embodiment of the present invention flow chart；
The test waves of the 1kHz pure tones that Fig. 4 is gathered for a kind of artificial cochlea's audio real time processing system of the embodiment of the present invention
Shape figure；
The ripple of the vowel " a " that Fig. 5 is gathered for a kind of artificial cochlea's audio real time processing system PC ends of the embodiment of the present invention
Shape figure.
Specific embodiment
Below in conjunction with accompanying drawing, the preferred embodiments of the present invention are described in detail.
Embodiment 1
Referring to Fig. 1, a kind of artificial cochlea's audio real time processing system of the embodiment of the present invention 1, including sound letter are shown
Number collecting unit 10, DSP processing units 20, DSP signal output units 30, DSP to PC Date Conversion Units 40, PC end signal realities
When collecting unit 50, PC ends real-time processings unit 60 and the real-time debugging unit 70 in PC ends, wherein,
Sound signal collecting unit 10 is connected with DSP processing units 20, including acquisition module and analog-to-digital conversion module, collection
After module sound to external world is acquired, through analog-to-digital conversion module output digit signals；
DSP processing units 20 are connected with DSP signal output units 30, at least including fast Fourier transform module, frequency dividing
Processing module and envelope extract block, the signal after treatment is cached to DSP signal output units 30, DSP signal output units
30 are exported with serial i 2S forms；
The input of DSP to PC Date Conversion Units 40 is connected with the output of DSP signal output units 30, and output is believed with PC ends
Number Real-time Collection unit 50 is connected, and at least turns USB hardware module including I2S；
PC end signal Real-time Collections unit 50, PC ends real-time processings unit 60 connect successively with the real-time debugging unit 70 in PC ends
Connect, wherein PC end signals Real-time Collection unit 50 at least includes continuous sampling module and framing sampling module, gather voice signal
Syllable fragment export and carry out phonetic algorithm treatment to PC ends real-time processings unit 60, by the real-time debugging unit 70 in PC ends after treatment
Debugged in real time.
Embodiment 2
On the basis of embodiment 1, referring to Fig. 2, a kind of artificial cochlea's audio for showing the embodiment of the present invention 2 is located in real time
Reason system, also including DSP demonstration units 80, is connected with DSP signal output units 30, will be buffered in DSP signal output units 30
In voice signal demonstration.
Also include PC ends real time demonstrations unit 90, be connected with PC ends real-time processings unit 60, by the result of real-time processing or
The pilot process synchronous demonstrating for the treatment of.
In a particular embodiment, PC end signals Real-time Collection unit 50 include be sequentially connected USB module, cache module and
PC framing acquisition modules, the parallel data stream that USB module will be transmitted is put into cache module, by the data structure in cache module
Into queue, queue length sets the buffer memory capacity for changing cache module, and last PC framings acquisition module is acquired.
Environmental sound signal is transferred in DSP through the collection of sound signal collecting unit 10, in dsp can be by DSP treatment
Unit 20 is processed, or original sound signal directly is transferred into DSP signal output units 30.In DSP signal output units 30,
Signal directly can be carried out output demonstration, or give DSP to PC data conversions by signal transmission in the form of audio or electro photoluminescence
Unit 40, then be acquired by PC end signal Real-time Collections unit 50.If the middle anaplasia of the signal DSP algorithm treatment for collecting
Amount or data, then can carry out real-time algorithm debugging efforts at PC ends；If collection is original sound signal, then can be at PC ends
Real-time processing unit 60 carries out simulation process, and the data and variable in processing procedure are carried out with bug check, or will treatment
Good data carry out output demonstration.
In language processing device for artificial cochlea outside environmental sounds signal, its dual microphone are gathered using two-microphone array
Position is presented relativeness one in front and one in back, and its relative horizontal position is 2cm.The two-microphone array of voice collecting unit is by sound
Analog electrical signal is converted to through acoustic-electric after the collection of frequency acoustical signal, then carrying out quantization by DSP is sampled as data signal, used as whole
The input signal of individual development system.
In system, setting data transfer (I2S turns USB) hardware module is master chip, and DSP is from chip.Meanwhile,
PC ends set the corresponding data transfer bit wide of sound pick-up outfit and sample rate, interconnect its relevant interface signal.So realize by wheat
The voice signal of gram elegance collection is exported by DSP digital signal output modules with I2S forms, is turned by voice communication converting unit
Change usb signal into and be transferred to PC ends.Completely lost for the speed for preventing PC reading usb datas causes to write less than I2S transmission rates
The problem of data is lost, the caching I2S data of respective stored depth can be set at I2S ends；Simultaneously also can be using regulation USB transmission speed
Rate is accelerated reading speed and is solved.Should be noted that the depth of caching is related to the preclinical problem of voice, depth is bigger, and incubation period is more long,
Speech time time delay is bigger.
The mode for realizing Real-time Collection USB digital audio and video signals at PC ends is divided into：Continuous sampling and framing are sampled.For reality
The debugging function of existing DSP phonetic algorithms, and the signal of observation will be needed to pass through to be used in the traditional DSP algorithm adjustment method of solution
Storage in DSP internal memories derive observation, the method due to limit by its memory capacity be only capable of observation very little data volume problem.
DSP inside coherent signal is exported and at PC ends by the way of continuous sampling in the present system, is observed with this and is judged its phase
The change of OFF signal calculating process in dsp it is whether consistent with PC algorithm simulating softwares come realize DSP algorithm debug work(
Energy.Realized in PC softwares relatively easy from the mode of USB port continuous sampling, its corresponding sampling bit wide, sampling need to be only set
Rate, port number and device id, then set its sampling time length and read the company that sampled data completes usb data again
Continuous collection.
In order that PC phonetic algorithm simulating developers have real-time, need to be transmitted by data conversion module in PC ends framing collection
The usb audio data come.The realization of the real-time of voice in phonetic algorithm treatment, is only capable of believing by by the voice of Time Continuous
Number it is divided into sectional syllable fragment (frame) to realize.Its concrete mode is the framing collection of voice, at framing phonetic algorithm
Reason, framing sound effect play displaying etc..Because each frame (syllable fragment) is very short in time span, need to be by many frame languages
Message continuous processing plays displaying, makes real-time voice.The parallel data stream of physical equipment USB interface transmission is first put
In the buffer, and set its caching data volume size；Then, the data in caching are constituted into queue, and by setting team
Row size changes its buffer memory capacity, but it should be noted that team's queue size directly determines the USB voice signals of real-time transmission
Latent time length in queue, that is, determine whether the voice being input into the phonetic algorithm of PC ends is same with the voice that environment is produced
Step；Finally, adopted by setting the framing to realize real-time voice such as sample rate, the sampling number of each frame, port number of sound
Collection.
To realize Real-time Collection, treatment and synchronous demonstrating, here by the way of framing collection, syllable fragment is by multiframe language
Message number is constituted, and the frame number computational methods that its every section syllable fragment is included are shown below：
In formula fs be framing sampling sample rate, the execution time span that Period is each syllable fragment,
SamplesPerFrame is the sampling number of each frame sampling in framing sampling.
In parallel work-flow, realized respectively by calling correlation function voice signal collection, phonetic algorithm treatment and
The parallel work-flow that the real-time synchronization of result audio is played.And each function is all by function, function is performed and function
Stop three parts composition.The real work of function is specifically completed by constantly repeating function executable portion.In PC algorithms
In reach PC to the delay time that function starts to perform for the first time is performed by setting beginning function corresponding to each function
Speech sample, the treatment of PC phonetic algorithms and PC audios play the time synchronized of displaying.Its specific parallel pipelining process performs flow：
Framing gathers current syllable fragment；The phonetic algorithm treatment of the voice signal of syllable fragment collection is carried out simultaneously；And simultaneously
Result after the voice signal of upper syllable fragment collection is processed through phonetic algorithm in executed in parallel plays displaying.By it is this simultaneously
The mode that row flowing water is performed, and the framing sampling of syllable fragment in short-term, phonetic algorithm treatment and broadcasting displaying, can realize PC
The real-time of end phonetic algorithm exploitation.It can not only be in real time to gather as the defeated of algorithm environment voice by microphone array
Enter；Can also in real time monitor and show the audio result of phonetic algorithm.
It is corresponding with said system, present invention also offers a kind of artificial cochlea's audio real-time processing method, its flow chart
Referring to Fig. 3, comprise the following steps：
S101, after sound signal collecting unit collected sound signal, data signal is converted to through analog-to-digital conversion module, output
Give DSP processing units；
S102, is transferred to DSP signal output units after DSP processing unit processes, and the data that will debug needs are carried
Take and detect, PC ends are transferred to by DSP to PC Date Conversion Units；
S103, the data of the time span required for PC ends carry out continuous framing sampled acquisition；
S104, after carrying out phonetic algorithm treatment, carries out algorithm debugging.
Further, after S102 is transferred to DSP signal output units after DSP processing unit processes, by DSP demonstration units
Demonstration is played out to voice signal.
After S104 carries out phonetic algorithm treatment, the pilot process of result or treatment is synchronized into demonstration.
The data of time spans of the S103 required for PC ends carry out continuous framing sampled acquisition, are transmitted by USB module
Parallel data stream is placed in cache module, and the data in cache module then are constituted into queue, is changed by setting queue size
Become the buffer memory capacity of cache module, by setting the port number of sample rate, the sampling number of each frame and sound to voice signal
Real-time framing collection.
Specific embodiment will not be described here with reference to said system embodiment.
In order to illustrate beneficial effects of the present invention, judge whether it receives device during audio digital signals transmitting and converting
Intrinsic noise is disturbed.Two kinds of experimental programs are devised to be verified.
It is a kind of signal acquisition test waves of artificial cochlea's audio real time processing system of the embodiment of the present invention referring to Fig. 4
Shape figure；Solid line and circle are all 1kHz tonal signals.
First, it is that 0.3, frequency is 1kHz, the pure tone sinusoidal signal that sample rate is 96kHz to produce the amplitude of standard at PC ends
DSP audio input interfaces are connected to by tone frequency channel wire, are used to replace the ambient sound input of dual microphone collection.Then, pass through
The audio digital signals that the system is passed in another PC ends collection.
Solid line is originally inputted sinusoidal tonal signal for what PC ends produced, and its sample rate is 96kHz, frequency is 1kHz and amplitude
It is 0.3.Circle is by after the analog-to-digital conversion sampling of the 16kHz sample rates of DSP, exporting what is finally gathered at PC ends by I2S
Sinusoidal signal.As can be seen that sound signal collecting system is in the main true in data acquisition transmittance process, the drift without frequency and
Distortion.
It is a kind of vowel " a " test waveform of artificial cochlea's audio real time processing system of the embodiment of the present invention referring to Fig. 5
Figure.
For the ambient noise in preventing microphone from collecting environment, DSP audio inputs are connected to using by tone frequency channel wire
Interface substitutes microphone as signal input sources.Vowel " a " two sound is played at PC ends, the sample rate of its raw tone is 16kHz,
Duration is about 0.5s.Finally, the signal that will be gathered is exported by the I2S interfaces of DSP, and digital language is gathered at PC ends by the system
Message number, and using PC related software comparative analysis raw tones and the time domain beamformer of the voice of collection.
Curve 1 is the voice signal by being collected in PC after system acquisition transmitting and converting under 16kHz sample rates
Time domain beamformer, curve 2 is primary standard mandarin vowel " a " two sound, the time domain of the voice input signal that sample rate is 16kHz
Oscillogram.Correlation curve 1,2 understands, by the voice signal being collected after the system voice signal timely collection system,
It is consistent substantially with primary signal, but can be influenceed by the transmitted noise of very low circuit intrinsic noise and audio transmission line.This
It is a little to influence to judge from acoustically, Chinese Character Recognition rate will not be reduced.
Finally illustrate, preferred embodiment above is merely illustrative of the technical solution of the present invention and unrestricted, although logical
Cross above preferred embodiment to be described in detail the present invention, it is to be understood by those skilled in the art that can be
Various changes are made to it in form and in details, without departing from claims of the present invention limited range.
Claims (8)
1. a kind of artificial cochlea's audio real time processing system, it is characterised in that single including sound signal collecting unit, DSP treatment
Unit, DSP signal output units, DSP to PC Date Conversion Units, PC end signal Real-time Collections unit, PC ends real-time processing unit
With the real-time debugging unit in PC ends, wherein,
The sound signal collecting unit is connected with the DSP processing units, including acquisition module and analog-to-digital conversion module, collection
After module sound to external world is acquired, through analog-to-digital conversion module output digit signals；
The DSP processing units are connected with the DSP signal output units, at least including fast Fourier transform module, frequency dividing
Processing module and envelope extract block, the signal after treatment is cached to the DSP signal output units, DSP signal output lists
Unit is exported with serial i 2S forms；
The input of DSP to the PC Date Conversion Units is connected with the output of the DSP signal output units, exports and the PC
End signal Real-time Collection unit is connected, and at least turns USB hardware module including I2S；
The PC end signals Real-time Collection unit, the PC ends real-time processing unit and the real-time debugging unit in PC ends are sequentially connected,
Wherein PC end signals Real-time Collection unit at least include continuous framing sampling module, gather voice signal syllable fragment export to
PC ends real-time processing unit carries out phonetic algorithm treatment, is debugged in real time by the real-time debugging unit in PC ends after treatment.
2. artificial cochlea's audio real time processing system according to claim 1, it is characterised in that also demonstrate single including DSP
Unit, is connected with the DSP signal output units, and the voice signal to being buffered in DSP signal output units synchronizes demonstration.
3. artificial cochlea's audio real time processing system according to claim 1, it is characterised in that also drilled in real time including PC ends
Show unit, be connected with the PC ends real-time processing unit, by the result of real-time processing or the pilot process synchronous demonstrating for the treatment of.
4. artificial cochlea's audio real time processing system according to claim 1, it is characterised in that the PC end signals are real-time
Collecting unit includes the USB module, cache module and the PC framing acquisition modules that are sequentially connected, and the USB module will be transmitted
Parallel data stream be put into cache module, by cache module data constitute queue, queue length set determine cache module
Buffer memory capacity, be finally acquired by PC framing sampling modules.
5. artificial cochlea's audio real-time processing method of the system of one of a kind of use claim 1-4, it is characterised in that including
Following steps：
After sound signal collecting unit collected sound signal, data signal is converted to through analog-to-digital conversion module, exports and give DSP treatment
Unit；
DSP signal output units are transferred to after DSP processing unit processes, it would be desirable to which the data of debugging are extracted and detected,
PC ends are transferred to by DSP to PC Date Conversion Units；
The data of the time span required for PC ends carry out continuous framing sampled acquisition；
After carrying out phonetic algorithm treatment, algorithm debugging is carried out.
6. method according to claim 5, it is characterised in that be transferred to DSP signals after the processing unit processes through DSP
After output unit, demonstration is played out to voice signal by DSP demonstration units.
7. method according to claim 5, it is characterised in that it is described carry out phonetic algorithm treatment after, by result or
The pilot process for the treatment of synchronizes demonstration.
8. method according to claim 5, it is characterised in that described required for PC ends carry out continuous framing sampled acquisition
Time span data, be by USB module transmission parallel data stream be placed in cache module, then by the number in cache module
According to queue is constituted, the buffer memory capacity of cache module is changed by setting queue size, adopted by setting sample rate, each frame
The port number of number of samples and sound is to the real-time framing collection of voice signal.
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| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| CN109966022A (en) * | 2017-12-27 | 2019-07-05 | 上海力声特医学科技有限公司 | Artificial cochlea's voice processing apparatus, mobile terminal and artificial cochlea |
Citations (9)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| US6665409B1 (en) * | 1999-04-12 | 2003-12-16 | Cirrus Logic, Inc. | Methods for surround sound simulation and circuits and systems using the same |
| CA2653767A1 (en) * | 2006-04-04 | 2007-10-11 | Cleartone Technologies Limited | Calibrated digital headset and audiometric test methods therewith |
| WO2009023784A1 (en) * | 2007-08-14 | 2009-02-19 | Personics Holdings Inc. | Method and device for linking matrix control of an earpiece ii |
| CN201290806Y (en) * | 2008-11-27 | 2009-08-19 | 上海力声特医学科技有限公司 | Artificial cochlea MAP wireless debugging device |
| US20110019846A1 (en) * | 2009-07-23 | 2011-01-27 | Dean Robert Gary Anderson As Trustee Of The D/L Anderson Family Trust | Hearing aids configured for directional acoustic fitting |
| CN203506973U (en) * | 2013-08-15 | 2014-04-02 | 中国科学院声学研究所 | External voice processor in artificial cochlea |
| CN104811883A (en) * | 2015-02-12 | 2015-07-29 | 杭州爱听科技有限公司 | Autonomously fitting hearing aid debugging system and data programming line |
| US20160249835A1 (en) * | 2013-10-31 | 2016-09-01 | Jiangsu Betterlife Medical Co., Ltd. | Interactive hearing diagnosis and treatment system based on wireless mobile communication platform |
| CN106075720A (en) * | 2016-05-30 | 2016-11-09 | 沈阳弘鼎康医疗器械有限公司 | A kind of artificial cochlea's external device (ED) debugging instrument |
-
2017
- 2017-02-28 CN CN201710111151.8A patent/CN106714064B/en active Active
Patent Citations (9)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| US6665409B1 (en) * | 1999-04-12 | 2003-12-16 | Cirrus Logic, Inc. | Methods for surround sound simulation and circuits and systems using the same |
| CA2653767A1 (en) * | 2006-04-04 | 2007-10-11 | Cleartone Technologies Limited | Calibrated digital headset and audiometric test methods therewith |
| WO2009023784A1 (en) * | 2007-08-14 | 2009-02-19 | Personics Holdings Inc. | Method and device for linking matrix control of an earpiece ii |
| CN201290806Y (en) * | 2008-11-27 | 2009-08-19 | 上海力声特医学科技有限公司 | Artificial cochlea MAP wireless debugging device |
| US20110019846A1 (en) * | 2009-07-23 | 2011-01-27 | Dean Robert Gary Anderson As Trustee Of The D/L Anderson Family Trust | Hearing aids configured for directional acoustic fitting |
| CN203506973U (en) * | 2013-08-15 | 2014-04-02 | 中国科学院声学研究所 | External voice processor in artificial cochlea |
| US20160249835A1 (en) * | 2013-10-31 | 2016-09-01 | Jiangsu Betterlife Medical Co., Ltd. | Interactive hearing diagnosis and treatment system based on wireless mobile communication platform |
| CN104811883A (en) * | 2015-02-12 | 2015-07-29 | 杭州爱听科技有限公司 | Autonomously fitting hearing aid debugging system and data programming line |
| CN106075720A (en) * | 2016-05-30 | 2016-11-09 | 沈阳弘鼎康医疗器械有限公司 | A kind of artificial cochlea's external device (ED) debugging instrument |
Non-Patent Citations (3)
| Title |
|---|
| 《微机接口实用技术教程》编委会: "《微机接口实用技术教程》", 31 August 2007 * |
| 刘托等: "无线便携式人工耳蜗调试系统的设计", 《北京生物医学工程》 * |
| 江丕栋: "《生物医学工程 保障人类健康的高新技术》", 31 July 2001 * |
Cited By (1)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| CN109966022A (en) * | 2017-12-27 | 2019-07-05 | 上海力声特医学科技有限公司 | Artificial cochlea's voice processing apparatus, mobile terminal and artificial cochlea |
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