CN1477900A - Sound effect treatment method for microphone and its device - Google Patents
Sound effect treatment method for microphone and its device Download PDFInfo
- Publication number
- CN1477900A CN1477900A CNA021302723A CN02130272A CN1477900A CN 1477900 A CN1477900 A CN 1477900A CN A021302723 A CNA021302723 A CN A021302723A CN 02130272 A CN02130272 A CN 02130272A CN 1477900 A CN1477900 A CN 1477900A
- Authority
- CN
- China
- Prior art keywords
- frequency
- signal
- sound
- voice signal
- microphone
- 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
Abstract
The present invention provides a sound effect treatment method for microphone and its device. The speech signal picked by microphone uses first frequency as sampling frequency, and is digitilized to form original speech signal, then the sampling frequency of the original speed signal can be down-converted, undergone the process of sound effect treatment to restore to first frequency, then the signal and the original speech signal are mixed, and undergone the process of analogue conversion and outputted to loudspeake for broadcasting.
Description
Technical field
The present invention relates to a kind of sound effect treatment method and device, particularly a kind of utilization adjustment sampling frequency (sampling rate) reduces audio and handles sound effect treatment method and the device that the microphone of required memory span is used.
Background technology
General microphone audio is handled two kinds of audio processing modes that roughly can be divided into analog and digital signal.Compared to simulation audio processing mode, the digital audio processing mode can obtain better audio effect at present.
Because microphone 1 will convert analog electrical signal to sound wave output for the conversion of signals analog electrical signal and the loud speaker 2 of sound wave, handle so digital audio processing unit 3 domestic demands will be carried out audio again behind microphone 1 signal digitalized, and after after audio is handled, needing again with signal imitationization to be suitable for for loud speaker 2 outputs.Therefore, as Fig. 1, show the calcspar of the digital audio processing unit 3 that a kind of existing microphone uses.This device 3 comprises one and receives the analog signal of microphone 1 output and with its digitized analog/digital converter (analog-to-digital converter, be called for short ADC) 31, the memory 32 of one temporary digitized voice signal, one carries out digital signal processor (the digital signal processor of signal processing from memory 32 acquisition voice signals, hereinafter to be referred as DSP) 33, and one receive through DSP 33 processing signals and with its simulated digital/analog converter (digital-to-analogconverter is called for short DAC) 34 to be suitable for being sent to loud speaker 2.By this, when microphone 1 converts sound wave to analog electrical signal, then can be through ADC 31 earlier with the signal digitalized memory 32 that is stored to, then DSP 33 is also handled to make various effects after memory 32 reads signal according to need, for example echo (echo), synaeresis (chorus), trill (flanger), lifting sound (pitch shift), equalizer (equalizer), the change of voice, compression and decompression, speech recognition or the like, again by DAC 34 conversion of signals become analog signal, play for loud speaker 2 thereafter.
In recent years along with the rise of multimedia recreation, allow the user also improve for the tonequality of microphone requirement thereupon, handle afterwards that a wherein key factor of tonequality is a sampling frequency and influence audio.This sampling frequency means at analog/digital converter 31 analog signal conversion is become in the digital signal processes, with the technology of periodic fixed frequency acquisition continuous data.In general, sampling frequency need be higher than the frequency of source of sound (being voice concerning microphone), and then tonequality is better for height along with sampling frequency heals.
Today, electronic component trended towards integrated and trend microminiaturization, made existing sound-effect processing equipment 3 also can be formed in single IC for both with the system integration, and very the person can be integrated on the one chip.But, make memory 32 capacity MIPS (the million instructions per second required in the sound-effect processing equipment 3 with corresponding DSP along with the increase of sampling frequency; 1,000,000 instructions of per second) amount also significantly increases thereupon, for instance, handle required memory span according to requiring the required bank bit (bit) of maximum delay (delay) time, sampling frequency and each sampling to determine (bank bit of each sampling of memory span=maximum delay time * sampling frequency *) such as the audio of echo, suppose that maximum delay time is that 300ms, sampling frequency are that the bank bit of 40,008 KHz (Hz) and each sampling is 16, then memory 32 capacity just need 28800 bytes (being 0.3*48000*16=230400bits=28800bytes).So, sound-effect processing equipment 1 requires the capacity of so high memory 32 and the operational capability of DSP 33, therefore need take too much circuit area, causes existing sound-effect processing equipment 3 difficult integrated, even becomes the significant burden that is integrated on the chip.
If the sound of microphone 1 is handled without any audio and promptly is sent to loud speaker 2 broadcasts, then sampling frequency should be high as much as possible, and then play with preferable tonequality.Yet, in fact sample rate is faster, the then human otherness that more can't discover between two sub-samplings, and owing to sound will change through the audio processing originally, for instance, echo is simulated sound reverberation via wall bounce-back back passback under large space, and such echo is through the attenuation by absorption of wall and air, most tonequality own are just not good, so people do not seek for the directly high tone quality of output of onomatopoeia sound on the contrary for the sound of handling through audio.Use with microphone 1, it is more than sufficient for people's sound that sampling frequency is about eight KHz.Therefore do not influencing under the microphone output tonequality, if reduce the sampling frequency that audio is handled, the requirement of memory 32 capacity and DSP33 operational capability also can decrease thereupon, and then solves the problem that in the past existed.
Summary of the invention
Therefore, one object of the present invention aims to provide a kind of being used in and downgrades sampling frequency when audio is handled to save memory span and to reduce the sound effect treatment method that the microphone of operational capability requirement is used.
Another object of the present invention, aim to provide the sound effect treatment method that a kind of microphone is used, when audio is handled, downgrade sampling frequency and after audio is handled, revert to originally sampling frequency with originally without audio processing signals audio mixing, handle the effect of required memory span to reach keeping the high tone quality low audio that descends.
A further object of the present invention aims to provide a kind of reduction audio that reaches and handles the sound-effect processing equipment that the microphone of required memory span is used.
A further object of the present invention aims to provide a kind of sound-effect processing equipment that reduction is used the microphone of operational capability requirement that reaches.
So, the invention provides the sound effect treatment method that a kind of microphone is used, be suitable for the audio processing and behind the voice signal of microphone acquisition, be sent to a loudspeaker plays, described method comprises following steps:
A) will be to export after sampling frequency is digitized into a primary speech signal with a first frequency from the voice signal of described microphone;
B) with steps A) in the sampling frequency of described primary speech signal carry out audio after downgrading and handle, thereafter, the sampling frequency with treated signal reverts to described first frequency again, to form a sound signal; And
C) primary speech signal receiving step A) and step B) sound signal and audio mixing become a voice signal, convert sound wave to and play to be suitable for exporting to described loud speaker voice signal is simulated thereafter.
The sound-effect processing equipment that the present invention also provides a kind of microphone to use is suitable for the audio processing and is sent to a loudspeaker plays behind the voice signal of microphone acquisition, and described device comprises:
One analog/digital converter receives the voice signal of described microphone, and this analog/digital converter forms a primary speech signal with a first frequency as the described voice signal of sampling frequency digitlization;
One audio processing unit receives described primary speech signal, and this audio processing unit downgrades the sampling frequency of described primary speech signal to carry out the audio processing and to handle the sampling frequency of recovering described voice signal by audio, to form a sound signal;
One mixer receives described primary speech signal and described sound signal with a synthetic voice signal; And
One digital/analog converter receives described voice signal and it is simulated to be suitable for exporting to described loudspeaker plays.
Description of drawings
Further feature of the present invention and advantage in the following detailed description that cooperates with reference to graphic preferred embodiment, can clearly be understood, in graphic:
Fig. 1 is the circuit block diagram of the digital audio processing unit used of existing microphone, this device be suitable for receiving the signal of a microphone and it is handled after export a loudspeaker plays to;
Fig. 2 is the circuit block diagram of preferred embodiment of the present invention, and this embodiment is in conjunction with a microphone and a loud speaker; And
Fig. 3 is the flow chart of preferred embodiment among Fig. 2.
Each description of reference numerals in the accompanying drawing is as follows:
4 microphones, 5 loud speakers
6 sound-effect processing equipments, 61 analog/digital converters/ADC
62 frequency reducing unit, 63 memories
64 digital signal processors/DSP 65 raising frequency unit
66 mixers, 67 digital/analog converters/DAC
Embodiment
Consulting Fig. 2, is a preferred embodiment of the sound-effect processing equipment 6 that shows that microphone of the present invention is used.This sound-effect processing equipment 6 is suitable for receiving the analoging sound signal of a microphone 4 acquisitions and is sent to a loud speaker 5 after treatment to be play.The sound-effect processing equipment 6 of present embodiment comprises an analog/digital converter (ADC) 61, frequency reducing unit 62, a memory 63, a digital signal processing unit (DSP) 64, raising frequency unit 65, a mixer 66 and a digital/analog converter 67.
This ADC 61 is electrically connected to microphone 4, frequency reducing unit 62 and mixer 66.In the present embodiment, microphone 4 conversion sound wave voice become an analog voice signal, and ADC 61 exports frequency reducing unit 62 and mixer 66 respectively to after to be sampling frequency (for example 48kHz) with the first frequency with the analog voice signal of microphone 4 outputs convert digital primary speech signal to.
This frequency reducing unit 62 exports memory 63 in order to downgrade to after sampling frequency from the primary speech signal of ADC61 becomes one to be lower than the second frequency of first frequency.In the present embodiment, because sound-effect processing equipment 6 is used for microphone 4, so only being enough to deal with human sound, second sampling frequency gets final product, in other words, second frequency only need equal eight KHz at least and get final product, second frequency is can 20,004 hertz or 10,002 hertz or the like for instance, and decides the numerical value of second frequency according to designer's demand.
This memory 63 is in order to the temporary voice signal that is downconverted to second frequency through frequency reducing unit 62, captures from memory 63 in for follow-up DSP 64 that voice signal is suitably handled and can supply DSP 64 carry out data access in the processes voice signals process.
This DSP 64 electrically connects memory 63 and raising frequency unit 64 and has the voice signal processing capacity, carry out the signal data processing to use according to required audio, and the audio processing comprises echo, synaeresis, trill, lifting sound, equalizer, the change of voice, compression and decompression, speech recognition or the like from acquisition audio digital signals in the memory 63.Because required memory can come access because of DSP 64 still needing in the audio processing procedure to carry out memory 63 when DSP 64 carried out the voice signal processing, make that DSP 64 desired memory 63 capacity can be greater than desired memory 63 capacity of the voice signal in frequency reducing unit 62 memories 63, so required memory 63 capacity of sound-effect processing equipment 6 are decided by the DSP 64 of need larger capacity.For instance, handle required memory span by requiring the required bank bit (bit) of maximum delay (delay) time, sampling frequency and each sampling to be determined (bank bit of each sampling of memory span=maximum delay time * sampling frequency *) such as the audio of echo, suppose that maximum delay time is that 300ms, sampling frequency are that the bank bit of 20,004 hertz and each sampling is 16, then memory 63 capacity only need 14400 bytes (being 0.3*24000*16=115200bits=14400bytes).Because no matter be which kind of audio processing determines that a key factor of required memory span is a sampling frequency, so compared to existing sound-effect processing equipment, the sound-effect processing equipment 6 of present embodiment utilizes frequency reducing unit 62 that the sampling frequency of voice signal is reduced to second frequency (as 20,004 hertz) by first frequency (as 40,008 hertz) before audio is handled in advance, making audio handle required memory 63 capacity can significantly reduce thereupon, to reach the effect that reduces required memory 63 capacity, and along with the sampling frequency of voice signal is reduced to second frequency, then the data volume of voice signal also reduces thereupon, allow DSP64 also can reach the effect that significantly reduces compared to needed operational capability requirement (referring to MIPS) in the past, so shared circuit area and the cost of memory 63 and DSP64 all can significantly reduce, to reach the trend and easier being integrated on single IC for both or the one chip that more meets the electronic component microminiaturization.
This raising frequency unit 65 electrically connects DSP 64 and mixer 66.This raising frequency unit 65 comes with primary speech signal integrated in order to receive the voice signal of handling through DSP 64 audios and to increase the sampling frequency of this voice signal so that the sampling frequency of voice signal returns to first frequency by second frequency to form a sound signal.
This mixer 66 is electrically connected to raising frequency unit 65 and ADC 61 and in order to the primary speech signal that is received from ADC 61 outputs (promptly refer to handle without frequency reducing unit 62, DSP 64, raising frequency unit 65 voice signal) and the sound signal (voice signal of promptly refer to through frequency reducing unit 62, DSP 64, raising frequency unit 65 handling) that is received from 65 outputs of raising frequency unit, to export the synthetic voice signal of two signals to DAC 67.Because primary speech signal without frequency reducing, in the present embodiment, in the analog/digital conversion process with first frequency same as the prior art as sampling frequency, because of its sampling frequency so high, make the tonequality of primary speech signal can be considered former sound reproducing, so that mix the effect that the voice signal of this primary speech signal can reach high tone quality.In the present embodiment, this mixer 66 is an accumulator.
This DAC 67 electrically connects this mixer 66 and in order to convert this voice signal to analog signal by digital signal, plays to export this loud speaker 5 to.
According to preceding described,, with reference to Fig. 3 the method for present embodiment is explained hereinafter in the lump in order to allow the easier quilt of the present invention understand.
At first, in step 71, receive by the voice signal of microphone 4 acquisitions and with first frequency (as 40,000 eight hertz) by ADC 61 and digitization of speech signals (be about to analog signal and transfer digital signal to) to be become primary speech signal, then ADC 61 and primary speech signal is sent to frequency reducing unit 62 and mixer 66 respectively as sampling frequency.
Thereafter, ground step 72, the sampling frequency of primary speech signal that is downgraded reception by frequency reducing unit 62 is to the second frequency that is lower than first frequency.
Then, in step 73, the voice signal after frequency reducing unit 62 will downgrade is sent in the memory 63 temporary.
And then in step 74, handle to carry out audio on demand by DSP 64 access voice signal in memory 63, because the sampling frequency (being second frequency) of the voice signal of audio significantly reduces, make memory 63 demand capacity are also significantly reduced with DSP 64 being required operational capability thereupon in the present embodiment.
, in step 75, increase the sampling frequency of the voice signal through DSP 64 audios handled, make sampling frequency return to first frequency to form sound signal by second frequency with voice signal thereafter.
Then, in step 76, be script sampling frequency (being first frequency) owing in step 75, heightening through the signal at frequency reducing unit 62 frequency reducings carrying out audio place, make sound signal and primary speech signal be all same sampling frequency, two signals like this carry out audio mixing to form voice signal through mixer 66, and this voice signal is owing to exist with the first frequency of high frequency (promptly such as the 40008 hertz) primary speech signal as sampling frequency, makes the tonequality of voice signal can reach the effect of identical with known sound-effect processing equipment at least high tone quality.And, sound-effect processing equipment 6 is in the sampling frequency (being first frequency) of decision ADC61 in the present embodiment, high sampling frequency causes memory 63 capacity and DSP64 operational capability also to need the problem that significantly promotes in the follow-up audio processing owing to more not taking into account, make the sampling frequency of ADC61 to improve as best one can (for example 90,000 six hertz), so make the present invention even can reach tonequality and be higher than known effect according to designer's demand.
At last, in step 77, receive voice signal and this voice signal is simulated from mixer 66, play usefulness to be suitable for being sent to loud speaker 5 by DAC 67.
Described before combining, the present invention utilizes high sampling frequency (first frequency) to come digitized voice signal to become primary speech signal, and only downgrade the sampling frequency of desiring the voice signal handled through audio and audio handle the back recover through the sampling frequency of the voice signal of audio processing with the primary speech signal audio mixing, so can reach the effect of taking into account high tone quality and low memory 63 capacity and DSP 64 operational capabilities, and then the cost of reduction sound-effect processing equipment 6, so that product has more the market competitiveness.
Content described above only is preferred embodiment of the present invention, can not limit scope of the invention process with this, all in the present patent application claim and the specification scope in the equivalence done change and modify, all should still belong in the scope that patent of the present invention contains.
Claims (7)
1. the sound effect treatment method that microphone is used is suitable for the audio processing and is sent to a loudspeaker plays behind the voice signal of microphone acquisition, and described method comprises following steps:
A) will be to export after sampling frequency is digitized into a primary speech signal with a first frequency from the voice signal of described microphone;
B) with steps A) in the sampling frequency of described primary speech signal carry out audio after downgrading and handle, thereafter, the sampling frequency with treated signal reverts to described first frequency again, to form a sound signal; And
C) primary speech signal receiving step A) and step B) sound signal and audio mixing become a voice signal, convert sound wave to and play to be suitable for exporting to described loud speaker voice signal is simulated thereafter.
2. the sound effect treatment method that microphone as claimed in claim 1 is used, wherein, described step B) comprise following inferior step:
B-1) with steps A) in the sampling frequency of primary speech signal be downconverted to a second frequency that is lower than described first frequency by described first frequency;
B-2) be that the voice signal of described second frequency carries out audio according to need and handles to sampling frequency; And
B-3) again with described step B-2) the sampling frequency of voice signal heighten into described first frequency from described second frequency, to form described sound signal.
3. the sound-effect processing equipment that microphone is used is suitable for the audio processing and is sent to a loudspeaker plays behind the voice signal of microphone acquisition, and described device comprises:
One analog/digital converter receives the voice signal of described microphone, and this analog/digital converter forms a primary speech signal with a first frequency as the described voice signal of sampling frequency digitlization;
One audio processing unit receives described primary speech signal, and this audio processing unit downgrades the sampling frequency of described primary speech signal to carry out the audio processing and to handle the sampling frequency of recovering described voice signal by audio, to form a sound signal;
One mixer receives described primary speech signal and described sound signal with a synthetic voice signal; And
One digital/analog converter receives described voice signal and it is simulated to be suitable for exporting to described loudspeaker plays.
4. the sound-effect processing equipment that microphone as claimed in claim 3 is used, wherein, described analog/digital converter, described audio processing unit, described mixer and described digital/analog converter are construed as single IC for both.
5. the sound-effect processing equipment that microphone as claimed in claim 3 is used, wherein, described analog/digital converter, described audio processing unit, described mixer and described digital/analog converter are construed as one chip.
6. the sound-effect processing equipment that microphone as claimed in claim 3 is used, wherein, described sound-effect processing equipment comprises:
One frequency reducing unit receives described primary speech signal and the sampling frequency of this primary speech signal is downconverted to one by described first frequency and exports after being lower than the second frequency of described first frequency;
One memory is stored described voice signal through frequency reducing;
One digital signal processor is got described voice signal by described memory and is handled back output this voice signal is carried out according to need audio; And
One raising frequency unit receives the described voice signal of handling through audio and also the sampling frequency of this voice signal is heightened into described first frequency from described second frequency, to form described sound signal.
7. the sound-effect processing equipment that microphone as claimed in claim 3 is used, wherein, described mixer is an accumulator.
Priority Applications (1)
| Application Number | Priority Date | Filing Date | Title |
|---|---|---|---|
| CNB021302723A CN100433938C (en) | 2002-08-22 | 2002-08-22 | Sound effect treatment method for microphone and its device |
Applications Claiming Priority (1)
| Application Number | Priority Date | Filing Date | Title |
|---|---|---|---|
| CNB021302723A CN100433938C (en) | 2002-08-22 | 2002-08-22 | Sound effect treatment method for microphone and its device |
Publications (2)
| Publication Number | Publication Date |
|---|---|
| CN1477900A true CN1477900A (en) | 2004-02-25 |
| CN100433938C CN100433938C (en) | 2008-11-12 |
Family
ID=34144433
Family Applications (1)
| Application Number | Title | Priority Date | Filing Date |
|---|---|---|---|
| CNB021302723A Expired - Fee Related CN100433938C (en) | 2002-08-22 | 2002-08-22 | Sound effect treatment method for microphone and its device |
Country Status (1)
| Country | Link |
|---|---|
| CN (1) | CN100433938C (en) |
Cited By (8)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| CN101790120A (en) * | 2009-01-23 | 2010-07-28 | 瑞昱半导体股份有限公司 | Sound effect wave-mixing device and method |
| CN102296761A (en) * | 2010-06-09 | 2011-12-28 | 施米茨-工厂有限及两合公司 | Awning |
| CN101221767B (en) * | 2008-01-23 | 2012-05-30 | 晨星半导体股份有限公司 | Voice boosting device and method used on the same |
| US8396230B2 (en) | 2008-01-16 | 2013-03-12 | Mstar Semiconductor, Inc. | Speech enhancement device and method for the same |
| US8565907B2 (en) | 2008-07-21 | 2013-10-22 | Realtek Semiconductor Corp. | Audio mixing device and method |
| CN105118514A (en) * | 2015-08-17 | 2015-12-02 | 惠州Tcl移动通信有限公司 | A method and earphone for playing lossless quality sound |
| CN105472507A (en) * | 2015-12-07 | 2016-04-06 | 蔡亮明 | Method and system for audio optimization |
| CN111526466A (en) * | 2020-04-30 | 2020-08-11 | 成都千立网络科技有限公司 | Real-time audio signal processing method for sound amplification system |
Family Cites Families (5)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| CN2099345U (en) * | 1991-08-02 | 1992-03-18 | 王佩青 | Universal pronunciation machine |
| JPH07199998A (en) * | 1993-12-27 | 1995-08-04 | Rohm Co Ltd | Compressing and expanding device for speech signal |
| CN1159700C (en) * | 1995-03-24 | 2004-07-28 | 华邦电子股份有限公司 | Phonetics synthesizer |
| WO1998018203A1 (en) * | 1996-10-22 | 1998-04-30 | Kakumoto Jun Ichi | Acoustic signal waveform intensifier and intensifying method |
| CN1137601C (en) * | 1996-12-11 | 2004-02-04 | Gn奈康有限公司 | Power supply for microphone |
-
2002
- 2002-08-22 CN CNB021302723A patent/CN100433938C/en not_active Expired - Fee Related
Cited By (9)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| US8396230B2 (en) | 2008-01-16 | 2013-03-12 | Mstar Semiconductor, Inc. | Speech enhancement device and method for the same |
| CN101221767B (en) * | 2008-01-23 | 2012-05-30 | 晨星半导体股份有限公司 | Voice boosting device and method used on the same |
| US8565907B2 (en) | 2008-07-21 | 2013-10-22 | Realtek Semiconductor Corp. | Audio mixing device and method |
| CN101790120A (en) * | 2009-01-23 | 2010-07-28 | 瑞昱半导体股份有限公司 | Sound effect wave-mixing device and method |
| CN102296761A (en) * | 2010-06-09 | 2011-12-28 | 施米茨-工厂有限及两合公司 | Awning |
| CN105118514A (en) * | 2015-08-17 | 2015-12-02 | 惠州Tcl移动通信有限公司 | A method and earphone for playing lossless quality sound |
| CN105472507A (en) * | 2015-12-07 | 2016-04-06 | 蔡亮明 | Method and system for audio optimization |
| CN111526466A (en) * | 2020-04-30 | 2020-08-11 | 成都千立网络科技有限公司 | Real-time audio signal processing method for sound amplification system |
| CN111526466B (en) * | 2020-04-30 | 2022-07-22 | 成都千立网络科技有限公司 | Real-time audio signal processing method for sound amplification system |
Also Published As
| Publication number | Publication date |
|---|---|
| CN100433938C (en) | 2008-11-12 |
Similar Documents
| Publication | Publication Date | Title |
|---|---|---|
| CN1065378C (en) | Bass enhancement circuit | |
| US20090098902A1 (en) | Method and system for processing multi-rate audio from a plurality of audio processing sources | |
| CN1992513A (en) | Effect adding method and effect adding apparatus | |
| US20120140952A1 (en) | Audio reproducing apparatus | |
| US8935157B2 (en) | Audio decoding system and an audio decoding method thereof for compressing and storing decoded audio data in a first time interval and decompressing the stored audio data in a second time interval | |
| CN1092420C (en) | Audio data decoding apparatus and audio data coding/decoding system | |
| JP2007512733A (en) | System and method for controlling audio output | |
| KR100664961B1 (en) | Mobile device for supporting multi-audio output | |
| CN1669357A (en) | Signal processing apparatus, signal processing method, program and recording medium | |
| CN1477900A (en) | Sound effect treatment method for microphone and its device | |
| CN1819452A (en) | Method and device for adjusting to adapt speech inputting distance | |
| JP3081378B2 (en) | Coding method for audio frequency signal of 32 kb / sec | |
| CN1716764A (en) | System and method for adjusting an audio signal | |
| CN1135525C (en) | Device for processing microphone signal of karaoke equipment | |
| CN1787069A (en) | Audio-frequency processing apparatus with voice mixing function and portable terminal | |
| EP1249837A2 (en) | A method for decompressing a compressed audio signal | |
| JP2002533790A (en) | Adaptive bit allocator and audio encoder | |
| CN1822092A (en) | Method and its device for elliminating background noise in speech input | |
| CN101080764A (en) | Sound generation assignment device, music sound generator and sound generation assignment process program | |
| CN1082305C (en) | Telephone answering machine with audio signal compression/expansion circuit | |
| CN1313718A (en) | Device for shaping signal, especially speech signal | |
| CN1065400C (en) | Compatible AC-3 and MPEG-2 audio-frequency code-decode device and its computing method | |
| CN1798451A (en) | Sound effect process device for wireless microphone | |
| CN219658385U (en) | Audio playing equipment | |
| CN1208984C (en) | Self-adapting ringing unit cooperate with chord ring |
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 | ||
| CF01 | Termination of patent right due to non-payment of annual fee |
Granted publication date: 20081112 Termination date: 20200822 |
|
| CF01 | Termination of patent right due to non-payment of annual fee |