CN105702260A - MATLAB-based multifunctional voice test filtering system - Google Patents

MATLAB-based multifunctional voice test filtering system Download PDF

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Publication number
CN105702260A
CN105702260A CN201610249869.9A CN201610249869A CN105702260A CN 105702260 A CN105702260 A CN 105702260A CN 201610249869 A CN201610249869 A CN 201610249869A CN 105702260 A CN105702260 A CN 105702260A
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China
Prior art keywords
voice signal
input
filtering system
interface
matlab
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Pending
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CN201610249869.9A
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Chinese (zh)
Inventor
项晓强
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Anhui University
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Anhui University
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Priority to CN201610249869.9A priority Critical patent/CN105702260A/en
Publication of CN105702260A publication Critical patent/CN105702260A/en
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    • GPHYSICS
    • G10MUSICAL INSTRUMENTS; ACOUSTICS
    • G10LSPEECH ANALYSIS OR SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
    • GPHYSICS
    • G10MUSICAL INSTRUMENTS; ACOUSTICS
    • G10LSPEECH ANALYSIS OR SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
    • G10L19/02Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using spectral analysis, e.g. transform vocoders or subband vocoders
    • GPHYSICS
    • G10MUSICAL INSTRUMENTS; ACOUSTICS
    • G10LSPEECH ANALYSIS OR SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
    • G10L19/04Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using predictive techniques
    • G10L19/26Pre-filtering or post-filtering
    • GPHYSICS
    • G10MUSICAL INSTRUMENTS; ACOUSTICS
    • G10LSPEECH ANALYSIS OR SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING; SPEECH OR AUDIO CODING OR DECODING
    • G10L21/00Processing of the speech or voice signal to produce another audible or non-audible signal, e.g. visual or tactile, in order to modify its quality or its intelligibility
    • G10L21/02Speech enhancement, e.g. noise reduction or echo cancellation
    • G10L21/0208Noise filtering

Abstract

The invention discloses an MATLAB-based multifunctional voice test filtering system which comprises a voice signal pickup assembly, a voice signal noise superposing device, an IIR filter, an FIR filter, a display, a reception microphone, an audio data line and an audio interface, wherein the audio interface is arranged on the voice signal pickup assembly, the audio data line is connected with the voice signal pickup assembly through the audio interface, the audio data line is connected with the reception microphone, the voice signal pickup assembly is connected with the voice signal noise superposing device and the display, the voice signal noise superposing device is connected with the FIR filter through the IIR filter, and the FIR filter is connected with the display. The MATLAB-based multifunctional voice test filtering system can analyze time domain and frequency domain waveforms of different voice signals, add different noises to different voice signals, analyze different situations when different noises added, and obtain different effects of adding noises and filtering noises when the IIR filter and the FIR filter are used to filter noises.

Description

Multifunction speech based on MATLAB tests filtering system
Technical field
The present invention relates to a kind of audio filtering system, particularly relate to a kind of multifunction speech based on MATLAB and test filtering system。
Background technology
Wave filter is a kind of frequency selective device, it is possible to make specific frequency content in signal pass through, and other frequency content that greatly decays。In test device, utilize this frequency-selecting effect of wave filter, it is possible to filtering interfering noise or carry out spectrum analysis。
In a broad aspect, the passage (medium) of any information transmission is all considered as a kind of wave filter。Because the response characteristic of any device is all the function of driving frequency, all its transmission characteristic of available frequency domain function representation。Therefore, constituting any one link of test system, such as mechanical system, electrical network, instrument and meter even connect wire etc., all within the scope of certain frequency, by its frequency domain characteristic, will carry out converting to the signal passed through and process。
The kind of wave filter is a lot, and sorting technique is also different, as can be functionally divided, it is also possible to from realizing method point, or grade method for designing。But generally speaking, wave filter can be divided into two big classes, i.e. classical filter device and Modern Filter。Classical filter device assumes that the useful signal in input signal x (n) and noise (or interference) signal component are respectively at different frequency bands, after x (n) is by a linear filtering system, it is possible to be intended to noise signal composition and effectively remove。, if the frequency band of useful signal and noise signal is overlapped, then classical wave filter is by helpless。The present wave filter in Ground Penetrating Radar Signal process mainly adopts the wave filter of classics to process。Therefore filter effect is better sometimes, sometimes poor。
Summary of the invention
The purpose of the present invention: provide a kind of multifunction speech based on MATLAB to test filtering system, design based on MATLAB, adopt multi-functional window interface, it is possible to analyze time domain and the frequency-domain waveform of different phonetic signal。
To achieve these goals, the technical scheme is that
A kind of multifunction speech based on MATLAB tests filtering system, including voice signal acquisition device, pronunciation signal noise stacking apparatus, iir filter, FIR filter, display, radio reception microphone, Audio Data Line and audio interface;Described audio interface is arranged in described voice signal acquisition device, one end of described Audio Data Line is connected by the input of described audio interface with described voice signal acquisition device, and the other end of described Audio Data Line is connected with the outfan of described radio reception microphone;The outfan of described voice signal acquisition device is connected with the input of described pronunciation signal noise stacking apparatus and display respectively, the outfan of described pronunciation signal noise stacking apparatus is connected with the input of described iir filter, the outfan of described iir filter is connected with the input of described FIR filter, and the outfan of described FIR filter is connected with the input of described display。
The above-mentioned multifunction speech based on MATLAB tests filtering system, and wherein, described display is touching-type monitor。
The above-mentioned multifunction speech based on MATLAB tests filtering system, wherein, described voice signal acquisition device includes input interface, buffer storage, calibration equipment and output interface, one end of described input interface is connected with described audio interface, the other end of described input interface is connected with one end of described buffer storage, the other end of described buffer storage is connected with one end of described calibration equipment, the other end of described calibration equipment is connected with one end of described output interface, the other end of described output interface is connected with described pronunciation signal noise stacking apparatus。
The above-mentioned multifunction speech based on MATLAB tests filtering system, wherein, described iir filter application Butterworth Bilinear transformation method filtering, described FIR filter application hamming window filtering。
The above-mentioned multifunction speech based on MATLAB tests filtering system, wherein, being provided with format conversion apparatus in described radio reception microphone, described format conversion apparatus is connected with described Audio Data Line, and voice data converts to the file output of WAV and MP3 format。
The present invention is possible not only to analyze time domain and the frequency-domain waveform of different phonetic signal, different noises can also be added for voice signal, such as white noise, single-frequency noise, multifrequency noise etc., then to the speech signal analysis time domain after interpolation noise and frequency-domain waveform, the different situations added under different noise situations are analyzed;IIR and FIR filter filtering noise can also be used, such that it is able to obtain adding the different-effect before and after noise and filtering noise。
Accompanying drawing explanation
Fig. 1 is the present invention connection block diagram based on the multifunction speech test filtering system of MATLAB。
Detailed description of the invention
Embodiments of the invention are further illustrated below in conjunction with accompanying drawing。
Refer to shown in accompanying drawing 1, a kind of multifunction speech based on MATLAB tests filtering system, including voice signal acquisition device 1, pronunciation signal noise stacking apparatus 2, iir filter 3, FIR filter 4, display 5, radio reception microphone 6, Audio Data Line 7 and audio interface 8;Described audio interface 8 is arranged in described voice signal acquisition device 1, one end of described Audio Data Line 7 is connected by the input of described audio interface 8 with described voice signal acquisition device 1, and the other end of described Audio Data Line 7 is connected with the outfan of described radio reception microphone 6;The outfan of described voice signal acquisition device 1 is connected with the input of described pronunciation signal noise stacking apparatus 2 and display 5 respectively, the outfan of described pronunciation signal noise stacking apparatus 2 is connected with the input of described iir filter 3, the outfan of described iir filter 3 is connected with the input of described FIR filter 4, and the outfan of described FIR filter 4 is connected with the input of described display 5。
Described display 5 is touching-type monitor, it is simple to control and operation。
Described voice signal acquisition device 1 includes input interface 11, buffer storage 12, calibration equipment 13 and output interface 14, one end of described input interface 11 is connected with described audio interface 8, the other end of described input interface 11 is connected with one end of described buffer storage 12, the other end of described buffer storage 12 is connected with one end of described calibration equipment 13, the other end of described calibration equipment 13 is connected with one end of described output interface 14, and the other end of described output interface 14 is connected with described pronunciation signal noise stacking apparatus 2。
Described iir filter 3 applies the filtering of Butterworth Bilinear transformation method, and described FIR filter 4 applies hamming window filtering。
Being provided with format conversion apparatus 61 in described radio reception microphone 6, described format conversion apparatus 61 is connected with described Audio Data Line 7, and voice data converts to the file output of WAV and MP3 format。
The design of the present invention is comparatively perfect, and its program is all tested on MATLAB2013a and MATLAB2014b and passed through, and program (includes annotating) 2045 row altogether。The present invention can also set up some little functions, it is possible to extends efficient help for simple analyzing speech signal。
In sum, the present invention is possible not only to analyze time domain and the frequency-domain waveform of different phonetic signal, different noises can also be added for voice signal, such as white noise, single-frequency noise, multifrequency noises etc., then to the speech signal analysis time domain after interpolation noise and frequency-domain waveform, analyze the different situations added under different noise situations;IIR and FIR filter filtering noise can also be used, such that it is able to obtain adding the different-effect before and after noise and filtering noise。
The foregoing is only the preferred embodiments of the present invention; not thereby the scope of the claims of the present invention is limited; every equivalent structure transformation utilizing description of the present invention to make; or directly or indirectly use the technical field being attached to other Related products, all in like manner include in the scope of patent protection of the present invention。

Claims (5)

1. the multifunction speech based on MATLAB tests filtering system, it is characterised in that: include voice signal acquisition device, pronunciation signal noise stacking apparatus, iir filter, FIR filter, display, radio reception microphone, Audio Data Line and audio interface;Described audio interface is arranged in described voice signal acquisition device, one end of described Audio Data Line is connected by the input of described audio interface with described voice signal acquisition device, and the other end of described Audio Data Line is connected with the outfan of described radio reception microphone;The outfan of described voice signal acquisition device is connected with the input of described pronunciation signal noise stacking apparatus and display respectively, the outfan of described pronunciation signal noise stacking apparatus is connected with the input of described iir filter, the outfan of described iir filter is connected with the input of described FIR filter, and the outfan of described FIR filter is connected with the input of described display。
2. the multifunction speech based on MATLAB according to claim 1 tests filtering system, it is characterised in that: described display is touching-type monitor。
3. the multifunction speech based on MATLAB according to claim 1 tests filtering system, it is characterized in that: described voice signal acquisition device includes input interface, buffer storage, calibration equipment and output interface, one end of described input interface is connected with described audio interface, the other end of described input interface is connected with one end of described buffer storage, the other end of described buffer storage is connected with one end of described calibration equipment, the other end of described calibration equipment is connected with one end of described output interface, the other end of described output interface is connected with described pronunciation signal noise stacking apparatus。
4. the multifunction speech based on MATLAB according to claim 1 tests filtering system, it is characterised in that: described iir filter application Butterworth Bilinear transformation method filtering, described FIR filter application hamming window filtering。
5. the multifunction speech based on MATLAB according to claim 1 tests filtering system, it is characterized in that: in described radio reception microphone, be provided with format conversion apparatus, described format conversion apparatus is connected with described Audio Data Line, and voice data converts to the file output of WAV and MP3 format。
CN201610249869.9A 2016-04-18 2016-04-18 MATLAB-based multifunctional voice test filtering system Pending CN105702260A (en)

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Cited By (3)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
CN107846691A (en) * 2016-09-18 2018-03-27 中兴通讯股份有限公司 A kind of MOS measuring methods, device and analyzer
CN111292748A (en) * 2020-02-07 2020-06-16 普强时代(珠海横琴)信息技术有限公司 Voice input system capable of adapting to various frequencies
CN111341335A (en) * 2020-03-07 2020-06-26 深圳市十盏灯科技有限责任公司 Audio processing method and system for sound card

Cited By (5)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
CN107846691A (en) * 2016-09-18 2018-03-27 中兴通讯股份有限公司 A kind of MOS measuring methods, device and analyzer
CN107846691B (en) * 2016-09-18 2022-08-02 中兴通讯股份有限公司 MOS (Metal oxide semiconductor) measuring method and device and analyzer
CN111292748A (en) * 2020-02-07 2020-06-16 普强时代(珠海横琴)信息技术有限公司 Voice input system capable of adapting to various frequencies
CN111292748B (en) * 2020-02-07 2023-07-28 普强时代(珠海横琴)信息技术有限公司 Voice input system adaptable to multiple frequencies
CN111341335A (en) * 2020-03-07 2020-06-26 深圳市十盏灯科技有限责任公司 Audio processing method and system for sound card

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Application publication date: 20160622