CN1598927A - Chinese voice signal process method for digital deaf-aid - Google Patents
Chinese voice signal process method for digital deaf-aid Download PDFInfo
- Publication number
- CN1598927A CN1598927A CNA2004100405841A CN200410040584A CN1598927A CN 1598927 A CN1598927 A CN 1598927A CN A2004100405841 A CNA2004100405841 A CN A2004100405841A CN 200410040584 A CN200410040584 A CN 200410040584A CN 1598927 A CN1598927 A CN 1598927A
- Authority
- CN
- China
- Prior art keywords
- signal
- frequency
- digital
- chinese speech
- domain
- Prior art date
- Legal status (The legal status is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the status listed.)
- Granted
Links
Images
Landscapes
- Machine Translation (AREA)
- Complex Calculations (AREA)
Abstract
The invention describes a Chinese speech sound signal processing used in the digital hearing aids and belongs to an speech sound signal processing. The invention aims to the feature of containing large amount of languages and semantic information in the tunes of the Chinese speech, treats the signal's feature of the speech data stream in according to several frequency parts through frequency analysis and outputs the digital signal after the D/A. the speech sound signal got from the invention can make the listeners hear the Chinese speech more clearly, increases the recognition rate of the Chinese speech and semantics and can improve the effect of the hearing aid.
Description
Technical field
The invention belongs to the voice processing technology of digital deaf-aid, it carries out digital signal processing at the pronunciation character of Chinese speech, is applicable to the digital deaf-aid based on Digital Signal Processing.
Background technology
Osophone is as the topmost means of hearing rehabilitation, and the people of the quality of life that always is subjected in the last hundred years striving for improvement pay close attention to, and are accompanied by the continuous development of science and technology, and computer technology and digital signal processing more and more are applied to the osophone field.At present, the osophone new product great majority of releasing in the world all are based on the digital deaf-aid of Digital Signal Processing.Because design, the manufacturer of digital deaf-aid all are western countries basically in the world, therefore various digital hearing aid signal processing technologies that occur for the needs that satisfy the polytype hearing compensation of person under different acoustic enviroments of listening to the barrier, such as: technology such as peak clipping control, automatic gain control, automatic frequency response control are but made with the research object that is characterized as of west voice mostly, so be to handle at the feature of Chinese speech without any a kind of digital deaf-aid signal processing technology also at present.
Research according to Chinese language is characterized in that: the single syllable sounding also has tangible tone features; With respect to western language, tone has comprised a large amount of language and semantic office rank's information.Show based on a large amount of tests, highlight the resolution that the tone feature of amplifying voice can effectively improve Chinese speech.
In the sense of hearing, tonal image or impression are fuzzyyer.We can say that substantially the change in pitch of short time is difficult to differentiate, only an average pitch.The so-called short time, for example half syllable or 1/3rd syllables, or a light tone syllable.The place of syllable beginning and end, the variation of pitch may be to be difficult for differentiating.The fact of significant is, phonetician's psychological feelings, and the credit of making a sound softly behind high and level tone and the rising tone is analysed sometimes the spitting image of falling tone, but is not falling tone.That is because its pitch at the beginning descends rapidly, and meanwhile volume is increasing gradually, and at this moment pleasant what hear mainly is a kind of feature of sinking, rather than high feature of falling.Equally, the syllable that initial consonant l, m, n, r or zero initial are done initial consonant is when going up sound, and pitch at the beginning also is to descend rapidly significantly, and what people's ear was heard also is the feature of sinking mainly.
The present invention is according to the characteristics of tone, sound import decomposed on a plurality of different frequency separations through Fourier transform handle, and the intensity of sound is divided into several grades from low to high on each frequency separation.Characteristic according to Chinese speech, tonal variations shows as the variation of fundamental frequency on signal characteristic, therefore find this to change the gain that also suitably increases fundamental frequency place frequency separation and just can reach the purpose of emphasizing tone, thereby improve the discrimination of user Chinese speech.
Summary of the invention
It is that pronunciation character with the west voice is that research object designs mostly that the present invention is intended at existing digital deaf-aid, and the present situation of incompatibility Chinese pronunciations feature, a kind of Chinese speech signal processing method that is used for digital deaf-aid is provided, handle at the feature of Chinese speech signal by digital signal processor, thereby improve the discrimination of user Chinese speech.
For solving the problems of the technologies described above, the technical solution used in the present invention is as follows:
A kind of Chinese speech signal processing method that is used for digital deaf-aid is characterized in that:
A) voice from the apparatus for receiving audio input obtain the time-domain digital input signal by the A/D sample devices;
B), time-domain signal is done time domain/frequency domain conversion obtain signal spectrum through Fourier transform processing;
C) extract the frequency-region signal feature by the spectrum analysis processing, change statistics, analysis of spectrum and calculated gains by the vowel consonant and handle definite signal amplification or decay strategy;
D) signal is handled according to the signal Processing strategy of determining by digital signal processor;
E) digital signal after the processing outputs to D/A converter, signal is reduced into simulating signal outputs to sound-reducing device.
Time domain of the present invention/frequency domain translation process is: adopt the input queue (FIFO) of first in first out that time-domain signal is lined up, and adopt superposition DFT bank of filters processing audio data piece, time-domain signal is transformed into frequency-region signal.
Described bank of filters is decomposed into a plurality of frequency separations to input signal, and through the modulated process of discrete Fourier transform (DFT) (DFT), single prototype filter is replicated to 2N compound filter wave band; This modulated process only produces identical filtering waveform and result in unified bank of filters; Time-domain signal is converted to the frequency-region signal of a plurality of frequency separations through this Fourier transform processing.
Described frequency-region signal characteristic extraction procedure is: to acquired frequency-region signal, extract the feature of each frequency separation in a plurality of frequency separations, judge that this input signal is noise signal or voice signal.
Described vowel consonant changes statistic processes: above-mentioned judgement according to noise or voice signal, if voice signal then changes the vowel consonant and makes statistics, judge the variation of speech pitch.
Described analysis of spectrum and calculated gains process are: the characteristics that change according to speech pitch are determined the gain strategy of each frequency separation, if noise is just given negative gain, if postiive gain just given in voice.
The process that described digital signal processor is handled signal according to the signal Processing strategy of determining is: according to the Chinese speech feature, initial consonant mainly be distributed in high frequency (b, p, m, f, z, c, s), it is short to have a duration, the feature that energy is lower; Simple or compound vowel of a Chinese syllable mainly be distributed in low frequency (a, o, e, i, u ü), has longer duration, the feature that energy is higher.With respect to noise, it is fast that voice have energy variation, the uncertain notable feature of crest frequency.Signal processor is according to these features, input signal is resolved into different frequency bands, add up the energy and the energy variation of each wave band respectively, the energy of all wave bands is compared, find out maximum and minimum wave band and the record of energy, energy distribution with a last period compares again, to find energy variation trend, determine the voice signal distribution situation, different wave bands is decayed respectively and strengthen according to these distribution characteristicss, with the synthetic output of the signal of all wave bands, reach the purpose of outstanding phonetic feature again.
Signal decomposition of the present invention becomes a plurality of frequency bands and is that discrete Fourier transform (DFT) and IDFT are that anti-discrete Fourier transform (DFT) bank of filters is handled with the synthetic method that is adopted of the signal of all frequency bands for superposition DFT, carry out the time-frequency domain conversion of signals, in transfer process, adopt the output queue (FIFO) of first in first out that input/output signal is kept in.
Digital signal processor of the present invention adopts the DSP of 16 fixed point structures can finish all operations.DSP is an IT industry standard assembly, and is the same with resistance capacitance, belongs to the standard term.
The invention has the advantages that:
The present invention is directed to Chinese speech monosyllable sounding and have tangible tone features, tone comprises this feature of a large amount of language messages, highlight the tone feature of amplifying voice, this algorithmic technique is applied to digital deaf-aid, thereby can effectively improve the discrimination of user's Chinese speech.It is simple, convenient that this method is implemented, and digital deaf-aid Chinese user can obtain hearing aid effect preferably.
Description of drawings
The signal processing flow block diagram that Fig. 1 adopts for the present invention
Embodiment
A kind of Chinese speech disposal route that is used for digital deaf-aid is handled at the feature of Chinese speech signal by digital signal processor, thereby improves the discrimination of user to Chinese speech.
The method of the invention is based on the processing to audio frequency.Its process comprises that time domain/frequency domain conversion process, frequency-region signal feature extraction, vowel consonant change statistics, analysis of spectrum gain process, frequency domain/time domain conversion process.See Fig. 1.
Is digital signal from the voice of apparatus for receiving audio input through the A/D analog to digital conversion, the audio data stream that obtains is converted to the required audio data stream of system through time domain/frequency domain, with bank of filters this audio data stream is decomposed into N frequency separation, modulated process through DFT, single prototype filter is replicated to 2N compound filter wave band, handle the signal characteristic that extracts this 2N frequency separation respectively by spectrum analysis, determine that signal amplifies the decay strategy, each interval digital signal is extracted through signal characteristic, the vowel consonant changes statistical treatment, the analysis of spectrum gain process, the signal of this 2N frequency separation is handled according to the signal Processing strategy of determining by digital signal processor then, output to input/output device by frequency domain/time domain conversion, again by outputing to sound-reducing device after the A/D conversion.
Described time domain/frequency domain translation process is: adopt first in first out input queue (FIFO) that time-domain signal is lined up, and employing superposition DFT bank of filters processing audio data piece, time-domain signal is transformed into frequency-region signal, utilize above-mentioned bank of filters that input signal is decomposed into N frequency separation, through the modulated process of DFT, single prototype filter is replicated to 2N combination frequency interval.This modulated process only produces identical filtering waveform and result in unified bank of filters.Time-domain signal is converted to the frequency-region signal of N frequency separation through this conversion process.
Described frequency-region signal characteristic extraction procedure is: to the frequency-region signal that obtains above, extract the feature of each frequency separation in 16 frequency separations, judge that this input signal is noise signal or voice signal.
Described vowel consonant changes statistic processes: according to the judgement of previous processed, if voice signal then changes the vowel consonant and makes statistics, judge the variation of speech pitch.
Described spectrum analysis gain process process is: determine the gain strategy of each frequency separation according to the characteristics that change, if noise is just given negative gain, if postiive gain just given in voice.
The process that described digital signal processor is handled signal according to the signal Processing strategy of determining is: according to the Chinese speech feature, initial consonant mainly be distributed in high frequency (b, p, m, f, z, c, s), it is short to have a duration, the feature that energy is lower; Simple or compound vowel of a Chinese syllable mainly be distributed in low frequency (a, o, e, i, u ü), has longer duration, the feature that energy is higher.With respect to noise, it is fast that voice have energy variation, the uncertain notable feature of crest frequency.Signal processor is according to these features, input signal is resolved into different frequency bands, add up the energy and the energy variation of each wave band respectively, the energy of all wave bands is compared, find out maximum and minimum wave band and the record of energy, energy distribution with a last period compares again, to find energy variation trend, determine the voice signal distribution situation, different wave bands is decayed respectively and strengthen according to these distribution characteristicss, with the synthetic output of the signal of all wave bands, reach the purpose of outstanding phonetic feature again.
Signal decomposition of the present invention becomes a plurality of frequency bands and is that discrete Fourier transform (DFT) and IDFT are that anti-discrete Fourier transform (DFT) bank of filters is handled with the synthetic method that is adopted of the signal of all frequency bands for superposition DFT, carry out the time-frequency domain conversion of signals, in transfer process, adopt the output queue (FIFO) of first in first out that input/output signal is kept in.
Digital signal processor of the present invention adopts the DSP of 16 fixed point structures can finish all operations.DSP is an IT industry standard assembly, and is the same with resistance capacitance, belongs to the standard term.
Frequency domain of the present invention/time domain transfer process is the inverse process of time domain/frequency domain conversion, adopt superposition DFT bank of filters processing audio data piece, time-domain signal is transformed into frequency-region signal, adopts first in first out output queue (FIFO) that the output time-domain signal is lined up.Amplify the Chinese speech tone thereby finish, reach the purpose that improves user's Chinese speech discrimination.
Claims (9)
1, a kind of Chinese speech signal processing method that is used for digital deaf-aid is characterized in that:
A) voice from the apparatus for receiving audio input obtain the time domain number by the A/D sample devices
The word input signal;
B), time-domain signal is done time domain/frequency domain conversion obtain letter through Fourier transform processing
Number frequency spectrum;
C) handle extraction frequency-region signal feature by spectrum analysis, change by the vowel consonant
Statistics, analysis of spectrum and calculated gains are handled and are determined that signal amplifies or the decay strategy;
D) signal is carried out according to the signal Processing strategy of determining by digital signal processor
Handle;
E) digital signal after the processing outputs to D/A converter, and signal is reduced into mould
Analog signal outputs to sound-reducing device.
2, a kind of Chinese speech signal processing method that is used for digital deaf-aid according to claim 1, it is characterized in that: described time domain/frequency domain translation process is: adopt the input queue (FIFO) of first in first out that time-domain signal is lined up, and adopt superposition DFT bank of filters processing audio data piece, time-domain signal is transformed into frequency-region signal.
3, a kind of Chinese speech signal processing method that is used for digital deaf-aid according to claim 2, it is characterized in that: described bank of filters is decomposed into a plurality of frequency separations to input signal, through the modulated process of discrete Fourier transform (DFT) (DFT), single prototype filter is replicated to 2N compound filter wave band; This modulated process only produces identical filtering waveform and result in unified bank of filters; Time-domain signal is converted to the frequency-region signal of a plurality of frequency separations through this Fourier transform processing.
4, a kind of Chinese speech signal processing method that is used for digital deaf-aid according to claim 1, it is characterized in that: described frequency-region signal characteristic extraction procedure is: to acquired frequency-region signal, extract the feature of each frequency separation in a plurality of frequency separations, judge that this input signal is noise signal or voice signal.
5, a kind of Chinese speech signal processing method that is used for digital deaf-aid according to claim 1, it is characterized in that: described vowel consonant changes statistic processes and is: above-mentioned judgement according to noise or voice signal, if voice signal, then the vowel consonant is changed and make statistics, judge the variation of speech pitch.
6, a kind of Chinese speech signal processing method that is used for digital deaf-aid according to claim 1, it is characterized in that: described analysis of spectrum and calculated gains process are: the gain strategy of determining each frequency separation according to the characteristics of speech pitch variation, if noise is just given negative gain, if postiive gain just given in voice.
7, a kind of Chinese speech signal processing method that is used for digital deaf-aid according to claim 1, it is characterized in that: digital signal processor is according to the Chinese speech feature, input signal is resolved into different frequency bands, add up the energy and the energy variation of each wave band respectively, the energy of all wave bands is compared, find out maximum and minimum wave band and the record of energy, energy distribution with a last period compares again, to find energy variation trend, determine the voice signal distribution situation, different wave bands is decayed respectively and strengthen according to these distribution characteristicss, again with the synthetic output of the signal of all wave bands, with this outstanding phonetic feature.
8, according to claim 2 or 3 described a kind of Chinese speech signal processing methods that are used for digital deaf-aid, it is characterized in that: signal decomposition of the present invention becomes a plurality of frequency bands and is that discrete Fourier transform (DFT) and IDFT are that anti-discrete Fourier transform (DFT) bank of filters is handled with the synthetic method that is adopted of the signal of all frequency bands for superposition DFT, carry out the time-frequency domain conversion of signals, in transfer process, adopt the output queue (FIFO) of first in first out that input/output signal is kept in.
9, according to claim 1 or 7 described a kind of Chinese speech signal processing methods that are used for digital deaf-aid, it is characterized in that: digital signal processor adopts the DSP of 16 fixed point structures to finish all operations.
Priority Applications (1)
| Application Number | Priority Date | Filing Date | Title |
|---|---|---|---|
| CNB2004100405841A CN1269106C (en) | 2004-08-31 | 2004-08-31 | Chinese voice signal process method for digital deaf-aid |
Applications Claiming Priority (1)
| Application Number | Priority Date | Filing Date | Title |
|---|---|---|---|
| CNB2004100405841A CN1269106C (en) | 2004-08-31 | 2004-08-31 | Chinese voice signal process method for digital deaf-aid |
Publications (2)
| Publication Number | Publication Date |
|---|---|
| CN1598927A true CN1598927A (en) | 2005-03-23 |
| CN1269106C CN1269106C (en) | 2006-08-09 |
Family
ID=34664725
Family Applications (1)
| Application Number | Title | Priority Date | Filing Date |
|---|---|---|---|
| CNB2004100405841A Expired - Fee Related CN1269106C (en) | 2004-08-31 | 2004-08-31 | Chinese voice signal process method for digital deaf-aid |
Country Status (1)
| Country | Link |
|---|---|
| CN (1) | CN1269106C (en) |
Cited By (14)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| CN100589185C (en) * | 2006-04-13 | 2010-02-10 | 北京中星微电子有限公司 | Device for processing voice signal |
| WO2011035626A1 (en) * | 2009-09-24 | 2011-03-31 | 华为终端有限公司 | Audio playing method and audio playing apparatus |
| CN102222507A (en) * | 2011-06-07 | 2011-10-19 | 中国科学院声学研究所 | Method and equipment for compensating hearing loss of Chinese language |
| CN101939784B (en) * | 2009-01-29 | 2012-11-21 | 松下电器产业株式会社 | Hearing aid and hearing-aid processing method |
| CN103310791A (en) * | 2012-03-07 | 2013-09-18 | 精工爱普生株式会社 | Speech recognition processing device and speech recognition processing method |
| CN104093111A (en) * | 2014-03-25 | 2014-10-08 | 嘉兴益尔电子科技有限公司 | Digital hearing aid with Chinese tone enhancing method |
| CN105338462A (en) * | 2015-12-12 | 2016-02-17 | 中国计量科学研究院 | Implementation method for reproducing hearing-aid insertion gain |
| CN105765654A (en) * | 2013-11-28 | 2016-07-13 | 弗劳恩霍夫应用研究促进协会 | Hearing assistance device with fundamental frequency modification |
| CN105989834A (en) * | 2015-02-05 | 2016-10-05 | 宏碁股份有限公司 | Voice recognition apparatus and voice recognition method |
| CN110798789A (en) * | 2018-08-03 | 2020-02-14 | 张伟明 | Hearing aid and method of use |
| CN110830897A (en) * | 2018-08-08 | 2020-02-21 | 塞舌尔商元鼎音讯股份有限公司 | Hearing aid and method for adjusting output voice of hearing aid |
| CN112738701A (en) * | 2021-01-07 | 2021-04-30 | 湖南芯海聆半导体有限公司 | Full-digital PWM audio output method for hearing aid chip and hearing aid chip |
| CN113286243A (en) * | 2021-04-29 | 2021-08-20 | 佛山博智医疗科技有限公司 | Error correction system and method for self-testing speech recognition |
| CN114584908A (en) * | 2022-03-04 | 2022-06-03 | 科大讯飞股份有限公司 | Acoustic testing method, device and equipment for hearing aid |
-
2004
- 2004-08-31 CN CNB2004100405841A patent/CN1269106C/en not_active Expired - Fee Related
Cited By (18)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| CN100589185C (en) * | 2006-04-13 | 2010-02-10 | 北京中星微电子有限公司 | Device for processing voice signal |
| CN101939784B (en) * | 2009-01-29 | 2012-11-21 | 松下电器产业株式会社 | Hearing aid and hearing-aid processing method |
| WO2011035626A1 (en) * | 2009-09-24 | 2011-03-31 | 华为终端有限公司 | Audio playing method and audio playing apparatus |
| CN102222507A (en) * | 2011-06-07 | 2011-10-19 | 中国科学院声学研究所 | Method and equipment for compensating hearing loss of Chinese language |
| CN103310791A (en) * | 2012-03-07 | 2013-09-18 | 精工爱普生株式会社 | Speech recognition processing device and speech recognition processing method |
| CN105765654A (en) * | 2013-11-28 | 2016-07-13 | 弗劳恩霍夫应用研究促进协会 | Hearing assistance device with fundamental frequency modification |
| CN104093111A (en) * | 2014-03-25 | 2014-10-08 | 嘉兴益尔电子科技有限公司 | Digital hearing aid with Chinese tone enhancing method |
| CN105989834A (en) * | 2015-02-05 | 2016-10-05 | 宏碁股份有限公司 | Voice recognition apparatus and voice recognition method |
| CN105338462A (en) * | 2015-12-12 | 2016-02-17 | 中国计量科学研究院 | Implementation method for reproducing hearing-aid insertion gain |
| CN105338462B (en) * | 2015-12-12 | 2018-11-27 | 中国计量科学研究院 | A kind of implementation method for reappearing hearing aid insertion gain |
| CN110798789A (en) * | 2018-08-03 | 2020-02-14 | 张伟明 | Hearing aid and method of use |
| CN110830897A (en) * | 2018-08-08 | 2020-02-21 | 塞舌尔商元鼎音讯股份有限公司 | Hearing aid and method for adjusting output voice of hearing aid |
| CN110830897B (en) * | 2018-08-08 | 2021-04-09 | 原相科技股份有限公司 | Hearing aid and method for adjusting output voice of hearing aid |
| CN112738701A (en) * | 2021-01-07 | 2021-04-30 | 湖南芯海聆半导体有限公司 | Full-digital PWM audio output method for hearing aid chip and hearing aid chip |
| CN112738701B (en) * | 2021-01-07 | 2022-02-25 | 湖南芯海聆半导体有限公司 | Full-digital PWM audio output method for hearing aid chip and hearing aid chip |
| CN113286243A (en) * | 2021-04-29 | 2021-08-20 | 佛山博智医疗科技有限公司 | Error correction system and method for self-testing speech recognition |
| CN114584908A (en) * | 2022-03-04 | 2022-06-03 | 科大讯飞股份有限公司 | Acoustic testing method, device and equipment for hearing aid |
| CN114584908B (en) * | 2022-03-04 | 2023-12-01 | 科大讯飞股份有限公司 | Acoustic testing method, device and equipment for hearing aid |
Also Published As
| Publication number | Publication date |
|---|---|
| CN1269106C (en) | 2006-08-09 |
Similar Documents
| Publication | Publication Date | Title |
|---|---|---|
| CN1269106C (en) | Chinese voice signal process method for digital deaf-aid | |
| Kingsbury et al. | Recognizing reverberant speech with RASTA-PLP | |
| Shrawankar et al. | Techniques for feature extraction in speech recognition system: A comparative study | |
| CN101625857B (en) | Self-adaptive voice endpoint detection method | |
| CN1167045C (en) | Speech recongition method and device | |
| Shete et al. | Zero crossing rate and Energy of the Speech Signal of Devanagari Script | |
| Hu et al. | Pitch‐based gender identification with two‐stage classification | |
| CN108198545B (en) | Speech recognition method based on wavelet transformation | |
| CN1991976A (en) | Phoneme based voice recognition method and system | |
| EP1250700A1 (en) | Speech parameter compression | |
| JPH08263097A (en) | Method for recognition of word of speech and system for discrimination of word of speech | |
| JPH09500223A (en) | Multilingual speech recognition system | |
| Venter et al. | Automatic detection of African elephant (Loxodonta africana) infrasonic vocalisations from recordings | |
| Dua et al. | Performance evaluation of Hindi speech recognition system using optimized filterbanks | |
| EP1744303A3 (en) | Method and apparatus for extracting pitch information from audio signal using morphology | |
| CN1300049A (en) | Method and apparatus for identifying speech sound of chinese language common speech | |
| Li et al. | Energy-efficient MFCC extraction architecture in mixed-signal domain for automatic speech recognition | |
| CN101281747A (en) | Method for recognizing Chinese language whispered pectoriloquy intonation based on acoustic channel parameter | |
| Mishra et al. | An Overview of Hindi Speech Recognition | |
| Ravindran et al. | Improving the noise-robustness of mel-frequency cepstral coefficients for speech processing | |
| CN103035237B (en) | Chinese speech signal processing method, device and hearing aid device | |
| Clapper | Automatic word recognition | |
| Rasetshwane et al. | Identification of speech transients using variable frame rate analysis and wavelet packets | |
| Biswas et al. | Articulation based admissible wavelet packet feature based on human cochlear frequency response for TIMIT speech recognition | |
| CN113611326B (en) | Real-time voice emotion recognition method and device |
Legal Events
| Date | Code | Title | Description |
|---|---|---|---|
| C06 | Publication | ||
| PB01 | Publication | ||
| C10 | Entry into substantive examination | ||
| SE01 | Entry into force of request for substantive examination | ||
| C14 | Grant of patent or utility model | ||
| GR01 | Patent grant | ||
| ASS | Succession or assignment of patent right |
Owner name: NANJING GUOGUANG MEDICAL INSTRUMENT CO., LTD. Free format text: FORMER OWNER: SICHUAN WEIDI DIGITAL TECHNOLOGY CO., LTD. Effective date: 20140515 |
|
| C41 | Transfer of patent application or patent right or utility model | ||
| COR | Change of bibliographic data |
Free format text: CORRECT: ADDRESS; FROM: 610021 CHENGDU, SICHUAN PROVINCE TO: 210000 NANJING, JIANGSU PROVINCE |
|
| TR01 | Transfer of patent right |
Effective date of registration: 20140515 Address after: Qinhuai city water Laifeng Street District of Nanjing City, Jiangsu province 210000 No. 66 national leading talent Park 40 building 203 Patentee after: Nanjing Guoguang Medical Instrument Co., Ltd. Address before: 610021 B, Weihai International Commercial Affairs Building, No. 136, Binjiang East Road, Sichuan, Chengdu, Chengdu, 16F Patentee before: Sichuan Weidi Digital Technology Co., Ltd. |
|
| CF01 | Termination of patent right due to non-payment of annual fee |
Granted publication date: 20060809 Termination date: 20200831 |
|
| CF01 | Termination of patent right due to non-payment of annual fee |