CN101860774B - Voice equipment and method capable of automatically repairing sound - Google Patents
Voice equipment and method capable of automatically repairing sound Download PDFInfo
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- CN101860774B CN101860774B CN201010191858.2A CN201010191858A CN101860774B CN 101860774 B CN101860774 B CN 101860774B CN 201010191858 A CN201010191858 A CN 201010191858A CN 101860774 B CN101860774 B CN 101860774B
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Abstract
The invention discloses voice equipment and a method capable of automatically repairing sound. The equipment comprises a microphone, a switch, an audio processing module at an input end, an infrared transmitting module, an infrared receiving module and an audio processing module at an output end, wherein the microphone is used for receiving voice signals; the switch is used for selecting a working mode of a system; the audio processing module at the input end is used for processing the input voice; the infrared transmitting module is used for transmitting the processed signals; the infrared receiving module is used for receiving the signals transmitted by the infrared transmitting module; and the audio processing module at the output end is used for reducing the signals received by the infrared receiving module into voice signals, processing the voice signals and outputting the processed voice signals to audio equipment. The technical scheme provided by the invention can supply automatic repair function and automatically adjust the volume so that the sound of a user reaches an optimal state.
Description
Technical field
The present invention relates to signal processing technology field, be specifically related to a kind of speech ciphering equipment and method that can automatically repairing sound.
Background technology
Microphone apparatus is a kind of speech ciphering equipment, as transaudient medium, is applied to widely in every field, and it is day by day important that its tonequality and effect also seem.The use of microphone, seems particularly important to popular singing style singer.Microphone can expand volume, amplify advantage, also can destroy tone color and exposes weakness.Experienced singer in popular singing style, not only can use microphone apparatus to expand and send sound, also will give expression to one's sentiment with microphone apparatus, embodies sound and exercises one's skill.
About volume aspect: it is the key that microphone apparatus is used that the size of volume is controlled.In general volume slightly controls, and when singer is than usual without microphone apparatus, sings and wants volume smaller.The volume of this control will keep the harmony of high, medium and low three sound areas, because accompaniment music is adjusted all the time in a volume intensity, if the volume disunity of three sound areas, just can there is strong or weak sound effect at high pitch or bass, has just affected the integrality of song.Some singers or high pitch are poor, or bass a little less than, also can be according to the situation of oneself, use microphone apparatus to regulate from the distance of mouth.This adjusting also will be depending on the sensitivity of sound equipment.In addition, the song of gas sound singing style, must be by microphone apparatus near mouth, to more can show the characteristic of this singing style.
About breathing aspect: the embodiment of breathing on microphone apparatus is very sensitive.Especially for highly sensitive microphone apparatus, breathing gently is just sent the larger sound by microphone apparatus.In the song of popular singing style, some songs need to have obvious ventilation sound, and to help to express the mood in song, what have is main singing by throat method with gas sound especially.This class song, breathing is not only ventilation, but a part for Song's Sentiment performance.Most of song, all needs to take a breath by light air-breathing method, and the smoothness of destroying song in order to avoid ventilation sound is large is harmonious, affects spectators' appreciation mood.
About stinging the word aspect of enunciating: due to the sensitivity of microphone apparatus, just sting that word is overweight and can send noise, especially " spout " the heavier word of " bang bang bang ", as " back of the body " " fearness " " " etc. word, prefix is overweight will produce this noise.The solution of this problem can take microphone apparatus practise more, finds a distance for angle and microphone apparatus and lip accurately.Be generally microphone apparatus and health angle at 45 °, this attitude is relatively more random and nature also.Get too high arm and send out deadlock, even block nose and face, take too low sound and sting word biography and do not go out, and overcautious.
Above-mentioned several aspect, for singing, GPRS is more difficult, if by solving some problems of above-mentioned existence to microphone apparatus setting, will be greatly user-friendly.
Summary of the invention
The technical problem to be solved in the present invention is to provide a kind of speech ciphering equipment and method that can automatically repairing sound, and automatic repair function can be provided, and automatic regulating volume, makes user's sound reach best state.
Technical scheme provided by the invention is as follows:
The invention provides a kind of speech ciphering equipment that can automatically repairing sound, comprising:
The audio processing modules of the audio processing modules of microphone, switch, input, infrared transmission module, infrared receiving module, output;
Described microphone, for received speech signal;
Described switch, for selecting the mode of operation of speech ciphering equipment;
The audio processing modules of described input, for processing the voice of input;
Described infrared transmission module, for launching the signal of handling well;
Described infrared receiving module, the signal sending for receiving described infrared transmission module;
The audio processing modules of described output, is reduced to voice signal and processes for the signal that infrared receiving module is received, and outputs to stereo set.
Further, the audio processing modules of described input comprises module digital translation (AD) module, Digital Signal Processing (Digital Signal Processing, DSP) module and microcontroller;
By AD module by Speech input in DSP module, and by DSP module to input sound carry out adaptive-filtering, sound import is optimized and noise reduction, then under the control of microcontroller, send to stereo set.
Further, described AD module is two AD modules, specifically by two AD modules, respectively standard voice and sound import is input in DSP module simultaneously.
Further, described DSP module is first carried out preliminary treatment to voice signal, i.e. primary filtering, and described preliminary treatment comprises: preemphasis, windowing and end-point detection.
Further, in the preprocessing process of described DSP module, the signal of input is the primary speech signal that sampling obtains, and the signal of output is several frame voice signals of having removed non-speech segment.
The present invention also provides a kind of method that can automatically repairing sound, comprising:
1) by switch, select recording mode, can input standard voice by microphone;
2) microphone apparatus deposits standard voice in memory in;
3) select performance pattern, singer carries out Speech input;
4) filtering is carried out to sound import in microphone inside, the sound of being short of breath and causing for filtering;
5) use set adaptive-filtering program to using standard voice as reference, sound import is carried out to secondary filtering, be equivalent to real-time voice signal to calibrate, determine volume gain, and the fine setting of tone;
6) finally export sound after filtering.
Further, described in, carry out filtering and carry out filtering processing by DSP module.
From technique scheme, can find out, the microphone apparatus that the embodiment of the present invention provides, automatic r repair function can be provided, automatic regulating volume, not only can be for on-the-spot concert, simultaneously more can use described microphone singer's sound can be beautified and modifies for popular house for karaoke widely now, can reach better singing effect.
Accompanying drawing explanation
In order to be illustrated more clearly in the embodiment of the present invention or technical scheme of the prior art, to the accompanying drawing of required use in embodiment or description of the Prior Art be briefly described below, apparently, accompanying drawing in the following describes is only some embodiments of the present invention, for those of ordinary skills, do not paying under the prerequisite of creative work, can also obtain according to these accompanying drawings other accompanying drawing.
Fig. 1 is the overall structure schematic diagram of microphone of the present invention;
Fig. 2 is the audio processing modules schematic diagram of input;
Fig. 3 is the audio processing modules schematic diagram of output.
Embodiment
Below in conjunction with the accompanying drawing in the embodiment of the present invention, the technical scheme in the embodiment of the present invention is clearly and completely described, obviously, described embodiment is only the present invention's part embodiment, rather than whole embodiment.Embodiment based in the present invention, those of ordinary skills, not making all other embodiment that obtain under creative work prerequisite, belong to the scope of protection of the invention.
The invention provides a kind of speech ciphering equipment that can automatically repairing sound, there is automatic repair function, can automatic regulating volume, and simultaneously can filtering because user stings the noise of overweight " bang bang bang " of sending of word, thereby can be user-friendly.
Microphone apparatus internal main provided by the invention will comprise memory, filtering chip and amplify output equipment.
Tone color when described microphone apparatus can be selected to input user's optimum state by switch, for example, sound while rehearsing, usings this sound as the reference of standard, and is stored in the middle of memory.When singing, singer arrives internal system by microphone apparatus by Speech input, chip after filtering, sound is processed, can filtering sting that word is overweight and send the sound of " bang bang bang ", use adaptive-filtering to carry out filtering to the standard voice in singer's real-time voice signal and memory simultaneously, to determine the gain of microphone, adjust volume, make to produce best singing effect, the inaccurate of tone once in a while can be corrected simultaneously, singer's fault rate can be reduced.
Below in conjunction with accompanying drawing, the embodiment of the present invention is described in detail.
Fig. 1 is the overall structure schematic diagram of the embodiment of the present invention.
As shown in Figure 1, microphone received speech signal by microphone apparatus, by selector switch, select the mode of operation of speech ciphering equipment, by corresponding audio processing modules, input voice are processed, then by infrared module, the signal of handling well is transferred to audio receiving system.Equally, the infrared module of audio receiving system, for receiving the signal of transmission, is reduced to voice signal by anti-phase process by the signal of reception more afterwards and exports by loud speaker, has reached the function of microphone.
Below introduce correlation module, each module specifically describes as follows:
1, the audio processing modules of input
By the adjustment of comparing of the sound of input and standard voice, to reach the noise reduction of sound import and filtering.
2, infrared transmission module
The sound import of adjusting is mail to the receiving terminal of sound equipment
3, infrared receiving module
Receive the audio signal that infrared transmission module sends
4, the audio processing modules of output
The inverse process that is equivalent to the audio processing modules of input, and audio frequency is carried out to separation and reduction, finally output to stereo set and sound.
As shown in Figure 1, the present invention is usingd infrared ray as the carrier wave of voice signal, and basic principle and common radio system are similar.The characteristic that infrared ray has a common radio system not have: diffracting power, penetration power very a little less than, therefore between the infrared radio microphone equipment in each relatively independent space, be can phase mutual interference, quantitatively do not use restriction, thereby realize any pairing of wireless microphone equipment and receiver.
Infrared ray has straightline propagation in addition, has very strong reflecting properties, and this just needs environment for use to have basic reflecting surface, otherwise the receiver window of receiver infrared receiving tube cannot be received signal.Therefore infrared radio microphone equipment and receiver thereof are generally unsuitable under open environment, using (as open air, the lecture theater that area is very large etc.), in medium and small relatively airtight multi-media classroom, have good effect.
As shown in Figure 2, be the structural representation of the audio processing modules of input of the present invention.
The audio processing modules of input is for processing the sound of input.Two AD modules (AD1 in figure and AD2) are input to standard voice and sound import in the middle of DSP (adopt TMS5320C5590A in the present invention but be not limited to this) simultaneously, and by DSP, the sound of input is carried out to adaptive-filtering, sound import is optimized and noise reduction, then under the control of microcontroller, sends to sound equipment to export.
The major function of the audio processing modules of output is realized by DSP module.
First DSP module must carry out preliminary treatment to the voice signal of being inputted by microphone, i.e. primary filtering.The preliminary treatment of audio frequency comprises: preemphasis, windowing and end-point detection.
The signal of the input of pretreatment unit is the primary speech signal that sampling obtains, and the signal of output is several frame voice signals of having removed non-speech segment.Because voice are jiggly random processes, it is characterized in that slowly changing along with the time, the effect of preemphasis is by high boost, the high frequency loss producing when the lip radiation to make up sound; On signal windowing to avoid the impact at Short Time Speech section edge; The effect of end-point detection is correctly to distinguish voice signal and diversity of settings noise.
Voice signal is in acquisition process; because the various objective impacts because of rope are often mixed with noise in the data that detect; after A/D conversion, increased again quantization noise; simultaneously; the noise of the generations such as unvoiced segments or the artificial breathing in pronunciation front and back makes the end points of voice fuzzyyer; therefore before extracting characteristic parameter, carry out preliminary treatment to data, remove as far as possible these noises to improve signal to noise ratio.Can take the multiresolution feature of wavelet transformation that primary speech signal is decomposed into a plurality of frames, calculate energy and the variance of each frame, meanwhile, change wavelet conversion coefficient and carry out noise reduction process.
Noise rejection method of wavelets mainly selects suitable wavelet function and the wavelet decomposition number of plies that the initial data output sequence that contains noise is carried out to orthogonal wavelet transformation, determine number of transitions, pass through settings, data are carried out to wavelet transform, calculate all wavelet coefficients, then reject and be considered to the relevant wavelet coefficient of noise, and wavelet coefficient is revised, revised wavelet coefficient is carried out to inverse transformation, obtain the data after noise reduction.The present invention uses the method for wavelet transformation, based on DSP, primary speech signal is processed.
In addition, for the voice signal of having eliminated background sound, must compare and proofread and correct to reach best output effect with standard signal.Here can adopt the method for adaptive-filtering to proofread and correct.Conventional adaptive algorithm has least-mean-square error algorithm (LMS), direct inversion algorithms (DMI) and the recursive least squares (RLS) of sampling covariance matrix.Native system selects least-squares algorithm as adaptive filter algorithm.The weight coefficient of this algorithm by continuous adjustment self adaptation part is to reach the output of the signal that is near the mark most.This part algorithm is also by realizing DSP.
Audio signal by twice processing has met output requirement, audio signal is processed to can better be exported by infrared module using also needing before the input signal as infrared output system.Under the effect of praying for controller, to the storage addressing of audio signal, coding, modulation etc., coordinate radio frequency to send out the functional modules such as large simultaneously, audio signal is outwards sent by infrared module.So far, the function of input audio frequency processing capacity module is just complete.
Figure 3 shows that the audio processing modules schematic diagram of output, this module, for the sound receiving is exported to processing, is equivalent to the inverse process of Fig. 2, needs equally the control of microcontroller, and audio frequency is carried out to separation and reduction.
Foregoing description the function of system and module, in use, method flow of the present invention is:
1) by switch, select recording mode, can input standard voice by microphone.
2) microphone apparatus deposits standard voice in memory in.
3) select performance pattern, singer carries out Speech input.
4) filtering is carried out to sound import in microphone inside, and this step is carried out filtering processing for the be short of breath sound of " bang bang bang " that cause of filtering by DSP module, and TMS5320C5509A chip is processed
5) use set adaptive-filtering program to using standard voice as reference, sound import is carried out to secondary filtering, be equivalent to real-time voice signal to calibrate, determine volume gain, and the fine setting of tone.
Standard voice and sound to be calibrated are converted to by AD converter can be for the digital signal of DSP resume module, and secondary filtering is processed via DSP module equally, be that with difference for the first time filtering is for the first time only the noise higher to the low-pass filtering rejection frequency of sound to be calibrated, filtering is for the second time to take the secondary calibration that standard voice carries out as benchmark.
6) finally export sound after filtering.
In sum, the application scenario of microphone very extensively, can be used as amusement in KTV box and use, also can be used as to sing and use, and hosting activity etc., the microphone apparatus that the embodiment of the present invention provides, can provide automatic repair function, automatic regulating volume, not only can be for on-the-spot concert, simultaneously more can use described microphone singer's sound can be beautified and modifies for popular house for karaoke widely now, can reach better singing effect.
A kind of speech ciphering equipment and the method that can automatically repairing sound that above the embodiment of the present invention are provided, be described in detail, applied specific case herein principle of the present invention and execution mode are set forth, the explanation of above embodiment is just for helping to understand method of the present invention and core concept thereof; , for one of ordinary skill in the art, according to thought of the present invention, all will change in specific embodiments and applications, in sum, this description should not be construed as limitation of the present invention meanwhile.
Claims (5)
1. a speech ciphering equipment that can automatically repairing sound, is characterized in that, comprising:
The audio processing modules of the audio processing modules of microphone, switch, input, infrared transmission module, infrared receiving module, output;
Described microphone, for received speech signal;
Described switch, for selecting the mode of operation of speech ciphering equipment;
The audio processing modules of described input, for processing the voice of input;
Described infrared transmission module, for launching the signal of handling well;
Described infrared receiving module, the signal sending for receiving described infrared transmission module;
The audio processing modules of described output, is reduced to voice signal and processes for the signal that infrared receiving module is received, and outputs to stereo set;
Wherein, the audio processing modules of described input comprises analog digital conversion AD module, Digital Signal Processing DSP module and microcontroller;
By AD module by Speech input in DSP module, and by DSP module to input sound carry out adaptive-filtering, sound import is optimized and noise reduction, then under the control of microcontroller, send to stereo set.
2. speech ciphering equipment that can automatically repairing sound according to claim 1, is characterized in that:
Described AD module is two AD modules, specifically by two AD modules, respectively standard voice and sound import is input in DSP module simultaneously.
3. speech ciphering equipment that can automatically repairing sound according to claim 2, is characterized in that:
Described DSP module is first carried out preliminary treatment to voice signal, i.e. primary filtering, and described preliminary treatment comprises: preemphasis, windowing and end-point detection.
4. speech ciphering equipment that can automatically repairing sound according to claim 3, is characterized in that:
In the preprocessing process of described DSP module, the signal of input is the primary speech signal that sampling obtains, and the signal of output is several frame voice signals of having removed non-speech segment.
5. a method that can automatically repairing sound, is characterized in that, comprising:
1) by switch, select recording mode, can input standard voice by microphone;
2) microphone apparatus deposits standard voice in memory in;
3) select performance pattern, singer carries out Speech input;
4) filtering is carried out to sound import in microphone inside, the sound of being short of breath and causing for filtering;
5) use set adaptive-filtering program to using standard voice as reference, sound import is carried out to secondary filtering, be equivalent to real-time voice signal to calibrate, determine volume gain, and the fine setting of tone;
6) finally export sound after filtering;
Wherein, described in, carry out filtering and carry out filtering processing by DSP module.
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| CN110634462B (en) * | 2019-09-30 | 2024-01-09 | 深圳市通世海精密机械有限公司 | Sound adjusting system and adjusting method |
| CN111696564B (en) * | 2020-06-05 | 2023-08-18 | 北京搜狗科技发展有限公司 | Voice processing method, device and medium |
| CN111696566B (en) * | 2020-06-05 | 2023-10-13 | 北京搜狗智能科技有限公司 | Voice processing method, device and medium |
| CN111696565B (en) * | 2020-06-05 | 2023-10-10 | 北京搜狗科技发展有限公司 | Voice processing method, device and medium |
Citations (2)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| CN1311891A (en) * | 1998-05-27 | 2001-09-05 | 艾利森电话股份有限公司 | Signal noise reduction by spectral substration using linear convolution and causal filtering |
| CN2755892Y (en) * | 2004-06-17 | 2006-02-01 | 佛山市公信智能会议设备制造有限公司 | Talking microphone without contact card |
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| US7054268B1 (en) * | 2000-02-04 | 2006-05-30 | Nokia Mobile Phones, Inc. | Method and arrangement for transferring information in a packet radio service with application-based choice of release mode |
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| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| CN1311891A (en) * | 1998-05-27 | 2001-09-05 | 艾利森电话股份有限公司 | Signal noise reduction by spectral substration using linear convolution and causal filtering |
| CN2755892Y (en) * | 2004-06-17 | 2006-02-01 | 佛山市公信智能会议设备制造有限公司 | Talking microphone without contact card |
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