CN101866651B - Method for implementing voice integrated circuit - Google Patents
Method for implementing voice integrated circuit Download PDFInfo
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- CN101866651B CN101866651B CN2009101066518A CN200910106651A CN101866651B CN 101866651 B CN101866651 B CN 101866651B CN 2009101066518 A CN2009101066518 A CN 2009101066518A CN 200910106651 A CN200910106651 A CN 200910106651A CN 101866651 B CN101866651 B CN 101866651B
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Abstract
The invention discloses a method for implementing a voice integrated circuit. A noise reduction unit is arranged between a digital-analog conversion unit and a speaker power operator; a main channel of the noise reduction unit is a variable band pass switching capacity filter; a secondary channel of the noise reduction unit is a signal analysis control unit; input ends of the main channel and the secondary channel are connected with an output end of the digital-analog conversion unit respectively; the output end of the main channel is connected with the speaker power amplifier; and the output end of the secondary channel is connected with a control signal input end of the main channel to adjust the bandwidth of the main channel. The method comprises the steps of: performing high-pass filtering on a voice signal which is subjected to analog-to-digital conversion by the secondary channel through the high pass filter; performing analog-digital conversion on the filtered voice signal by using an A/D converter; inputting the signal to a central processing unit; performing peaking operation and average calculating operation on the voice signal which is subjected to analog-to-digital conversion in the central processing unit; controlling the output frequency of a controllable clock source by using the operation result; and controlling the bandwidth of the variable band pass switching capacity filter by the controllable clock source.
Description
Technical field
The present invention discloses a kind of noise-reduction method, the implementation method of decrease of noise functions in particularly a kind of voice integrated circuit.
Background technology
Speech ciphering equipment is ubiquitous in people's daily life, and along with development of science and technology, the integrated level of circuit is increasingly high, and function also from strength to strength.Please referring to accompanying drawing 1; General voice integrated circuit comprises speech signal collection converter unit, signal encoding/decoding unit, D/A conversion unit and central processing unit etc. at present; Microphone or other audio signal source input audio signal are given the signals collecting converter unit; The sound signal that the signals collecting converter unit will be changed inputs to the signal encoding/decoding unit; Carry out inputing to D/A conversion unit behind the coding-decoding operation through the signal encoding/decoding unit, in D/A conversion unit, digital signal converted to and be transferred to loudspeaker (band power amplifier) output after the simulating signal.Many noise reduction unit of not being with in the voice integrated circuit of the prior art; Voice signal after not handling with the voice integrated circuit of noise reduction unit when playing, has noise; Especially when small-signal output or signal gap, noise is particularly evident.Also adopt BPF. to carry out the filtering noise reduction process in some voice integrated circuit at present; BPF. of the prior art has two kinds of implementation methods usually: a kind of is to adopt Active Analog Filter, and low precision, element be many, be difficult to change component parameters, difficulty is integrated; Another kind is the algorithm filtering of adopting the pure digi-tal signal, and cost is also very high.
Summary of the invention
To the above-mentioned voice integrated circuit of mentioning of the prior art do not have that decrease of noise functions or its noise reduction are many with filter element, shortcomings such as low precision, difficulty are integrated, cost height; The present invention provides the implementation method of decrease of noise functions in a kind of new voice integrated circuit; It has additional a noise reduction unit between D/A conversion unit and loudspeaker power amplifier; The logical SCF of variable strip zone is adopted in the noise reduction unit main channel; The noise reduction unit subaisle adopts the input analysis controlling unit, and the input end of main channel and subaisle is connected on the output terminal of D/A conversion unit, and the output terminal of main channel is connected on the power amplifier input end; The output terminal of subaisle is connected with the signal input end of main channel, and the bandwidth of main channel is regulated.Voice signal after subaisle is changed through the Hi-pass filter logarithmic mode carries out high-pass filtering; Through A/D converter filtered voice signal is carried out analog to digital conversion then; Import central processing unit then; In central processing unit, the voice signal after the analog to digital conversion is got peak value and average calculating operation, utilize operation result to control the output frequency in controlled clock source, the bandwidth of the logical SCF of controlled clock source control variable strip zone.
The technical scheme that the present invention solves its technical matters employing is: a kind of implementation method of voice integrated circuit; Speech IC comprises signals collecting converter unit, signal encoding/decoding unit, D/A conversion unit, central processing unit and storer; The sound signal input signal of microphone or other audio signal source is gathered in the converter unit; After the signals collecting converter unit is handled, export to the signal encoding/decoding unit and carry out the coding/decoding operation, export to loudspeaker after the process D/A conversion unit converts simulating signal to then, central processing unit is connected on the signal encoding/decoding unit; Storer is connected on the central processing unit; This method has additional a noise reduction unit after being included in the D/A conversion unit in the speech IC, and alterable band-pass filter is adopted in the main channel in the noise reduction unit, and subaisle is the input analysis controlling unit; The main channel signal input end is connected on the D/A conversion unit signal output part in the speech IC; The main channel signal output terminal is signal output, is connected on the power amplifier input end of loudspeaker, and the subaisle signal input part is connected on the D/A conversion unit signal output part in the speech IC; The subaisle control signal output ends is connected on the control end of main channel, and alterable band-pass filter adopts the logical SCF of variable strip zone; The input analysis controlling unit is made up of Hi-pass filter, A/D converter and central processing unit; Wherein the signal input part of Hi-pass filter is connected on the D/A conversion unit signal output part in the speech IC; Voice signal after the logarithmic mode conversion carries out high-pass filtering; The signal output part of Hi-pass filter is connected with the input end of A/D converter; Through A/D converter filtered voice signal is carried out analog to digital conversion, the output terminal of A/D converter is connected on the central processing unit, is connected with controlled clock source on the central processing unit; In central processing unit, the voice signal after the analog to digital conversion is got peak value and average calculating operation; Utilize operation result to control the output frequency in controlled clock source, the bandwidth of the logical SCF of controlled clock source control variable strip zone is carried out noise reduction through the sound signal after the logical SCF logarithmic mode conversion of variable strip zone.
The technical scheme that the present invention solves its technical matters employing further comprises:
It is the Hi-pass filter of 12dB/ octave that described Hi-pass filter adopts 6KHz, slope.
When the voice signal after the analog to digital conversion being got peak value and average calculating operation in the described central processing unit, adopt an interior in short-term voice signal is got peak value and peak value is averaged computing.
Described one is a frame in short-term.
Described one is 10～30mS in short-term.
The invention has the beneficial effects as follows: the present invention has utilized the switching capacity integration time constant to be decided by the ratio of two electric capacity in the circuit; (the absolute value variable effect of electric capacity is little); And the characteristic that is directly proportional with the work clock cycle, precision is high in the process that realizes noise reduction system, realizes simple and convenient; Cost is low, has preferably to be worth.
To combine accompanying drawing and embodiment that the present invention is further specified below.
Description of drawings
Fig. 1 is a voice integrated circuit square frame principle synoptic diagram of the prior art.
Fig. 2 is for adopting voice integrated circuit square frame principle synoptic diagram of the present invention.
Fig. 3 is a circuit square frame principle synoptic diagram of the present invention.
Fig. 4 is a detailed circuit square frame principle synoptic diagram of the present invention.
Fig. 5 is the single order SCF.
Fig. 6 is the equivalent circuit diagram of Fig. 5.
Embodiment
Present embodiment is the preferred embodiment for the present invention, and other all its principles are identical with present embodiment or approximate with basic structure, all within protection domain of the present invention.
The basic structure of using voice integrated circuit of the present invention is with of the prior art identical; Please referring to accompanying drawing 1 and accompanying drawing 2; Use speech IC of the present invention and comprise signals collecting converter unit, signal encoding/decoding unit, D/A conversion unit, central processing unit and storer; The sound signal input signal of microphone or other audio signal source is gathered in the converter unit; After the signals collecting converter unit is handled, export to the signal encoding/decoding unit and carry out the coding/decoding operation; Export to loudspeaker after converting simulating signal to through D/A conversion unit then, central processing unit is connected on the signal encoding/decoding unit, and storer is connected on the central processing unit.Increased a noise reduction unit in the speech IC among the present invention, it can reduce the noise in the source speech signal, well improves the effect of playback.
Noise reduction unit among the present invention is (being principle of work of the present invention) of carrying out work according to following psychoacoustic principle:
1, the output of noise is directly proportional with the bandwidth of system, and (bandwidth that is system is wide more; Incidental noise is just big more in the sound signal of output so); Therefore, if can manage to make controlled the narrowing down of bandwidth of system, then can make corresponding the reducing of noise component in the sound signal of output.
2, people's ear has masking effect, promptly listens in the feeling ability of a sound at people's ear, and stronger component can be covered another kind of more weak component in the sound, make its impression less than.Composite signal for as music and speech has masking effect preferably.In general; As long as the amplitude of signal source (being music and speech or other useful sound signals) is than the big 29dB of background noise (being the noise that comprises in the signal source); Just can provide enough masking effects, make the people can only hear the sound of signal source, and can't hear noise.
Ultimate principle of the present invention is through the check and analysis to signal intensity and frequency component; When small-signal, no signal, make the bandwidth of system minimum, output noise is minimum; Increase and variation along with signal; The bandwidth of Adjustment System (promptly reducing noise to the full extent) is dynamically utilized the masking effect of people's ear simultaneously, reaches the purpose of noise reduction.
Noise reduction unit among the present invention is made up of main channel (being the noise reduction passage) and subaisle (being control channel); The input end of main channel and subaisle all is connected on the output terminal of digital to analog converter; The output terminal of main channel is connected on the loudspeaker, and the output terminal of subaisle is connected on the control input end of main channel.In the present embodiment, the noise reduction passage is made up of alterable band-pass filter, controlled clock circuit (clock frequency is variable), is used to change the bandwidth of system; Control channel is 6KHz (also selecting other value again between 4KHz～6KHz for use) by corner frequency; Slope is that Hi-pass filter, A/D converter, the central processing unit of 12dB/ octave formed; Wherein central processing unit can be used to the detection peak level and peak value is averaged computing, and control channel can be controlled the frequency of clock circuit according to the situation of signal.
Alterable band-pass filter among the present invention adopts the logical SCF of variable strip zone; Utilized the time constant of switched-capacitor integrator in the integrated circuit to be decided by the ratio of two electric capacity, and the characteristic that is directly proportional with the clock period of control signal, in the CMOS integrated circuit, realize switch-capacitor filtering; Can accomplish that precision is very high; And available clock carry out tuning, the adjustment filter time constant, just adjust its bandwidth.Realize the wave filter of bandwidth varying, and then the noise reduction of the system of realization.
The noise reduction unit course of work of the present invention is following:
Sound signal after the digital to analog converter conversion is an input signal; Input signal is (to pass through 6KHz before the input analysis earlier after 6KHz, slope are the high pass filter filters of 12dB/ octave through corner frequency; The Hi-pass filter of 12dB slope; Be in order to keep the high fdrequency component in the sound), input to A/D converter,, A/D converter inputs to central controller after converting digital signal to; Through central controller it is carried out producing a control signal by incoming signal level and frequency response decision after peak value detection and the average calculating operation again; Be used to control the output clock frequency of controlled clock source circuit, thereby the variable strip zone in the control noise reduction passage is led to the bandwidth of operation of SCF, reach the purpose of noise reduction.In the present embodiment, central controller to through A/D converter conversion back digital signal carry out that peak value detects and during average calculating operation, normally interior in short-term signal calculated, promptly be that a frame signal (or also can become 10～30Ms signal) is calculated.
The present invention is when no signal input (being signal gap) or small-signal input, and the wave filter that switching capacity is formed is operated in lowest-bandwidth state (about 600Hz), and this moment, high frequency background noise was suppressed (because signal to noise ratio (S/N ratio) is decided by bandwidth).
When signal is imported; Through central processing unit the peak value detection of input signal and the value after the average calculating operation are raise; Make controlled clock source output frequency raise, the broadened bandwidth of SCF (its bandwidth and useful signal are adapted), but enhancing because of useful signal the time; Be enough to use the useful signal masking noise, so imperceptible noise.
Please referring to accompanying drawing 5 and accompanying drawing 6, providing the simplest a kind of single order low pass switching capacity in the present embodiment is that example specifies, and wherein Φ 1 is work clock (promptly being the clock by controlled clock source circuit output) with Φ 2.In the present embodiment; (about 300Hz～3.4KHz) is an example to be operated in voice band; About 50KHz～the 400KHz of the work clock of the SCF of selecting for use; When the amplitude of signal and frequency response changed, clock frequency changed between 50KHz～400KHz, and the low pass corner frequency of SCF changes to 4.5KHz from 600Hz.
Method of the present invention has utilized the switching capacity integration time constant to be decided by the ratio of two electric capacity in the circuit; (the absolute value variable effect of electric capacity is little); And the characteristic that is directly proportional with the work clock cycle, precision is high in the process that realizes noise reduction system, realizes simple and convenient; Cost is low, has preferably to be worth.
Claims (5)
1. the implementation method of a voice integrated circuit; Described speech IC comprises signals collecting converter unit, signal encoding/decoding unit, D/A conversion unit, central processing unit and storer; The sound signal input signal of microphone or other audio signal source is gathered in the converter unit; After the signals collecting converter unit is handled, export to the signal encoding/decoding unit and carry out the coding/decoding operation, export to loudspeaker after the process D/A conversion unit converts simulating signal to then, central processing unit is connected on the signal encoding/decoding unit; Storer is connected on the central processing unit; It is characterized in that: described method has additional a noise reduction unit after being included in the D/A conversion unit in the speech IC, and alterable band-pass filter is adopted in the main channel in the noise reduction unit, and subaisle is the input analysis controlling unit; The main channel signal input end is connected on the D/A conversion unit signal output part in the speech IC; The main channel signal output terminal is signal output, is connected on the power amplifier input end of loudspeaker, and the subaisle signal input part is connected on the D/A conversion unit signal output part in the speech IC; The subaisle control signal output ends is connected on the control end of main channel
Described alterable band-pass filter adopts the logical SCF of variable strip zone;
Described input analysis controlling unit is made up of Hi-pass filter, A/D converter and central processing unit; Wherein the signal input part of Hi-pass filter is connected on the D/A conversion unit signal output part in the speech IC; Voice signal after the logarithmic mode conversion carries out high-pass filtering; The signal output part of Hi-pass filter is connected with the input end of A/D converter; Through A/D converter filtered voice signal is carried out analog to digital conversion, the output terminal of A/D converter is connected on the central processing unit, is connected with controlled clock source on the central processing unit; In central processing unit, the voice signal after the analog to digital conversion is got peak value and average calculating operation; Utilize operation result to control the output frequency in controlled clock source, the bandwidth of the logical SCF of controlled clock source control variable strip zone is carried out noise reduction through the sound signal after the logical SCF logarithmic mode conversion of variable strip zone.
2. the implementation method of voice integrated circuit according to claim 1 is characterized in that: it is the Hi-pass filter of 12dB/ octave that described Hi-pass filter adopts 6KHZ, slope.
3. the implementation method of voice integrated circuit according to claim 1 and 2; It is characterized in that: when the voice signal after the analog to digital conversion being got peak value and average calculating operation in the described central processing unit, adopt an interior in short-term voice signal is got peak value and peak value is averaged computing.
4. the implementation method of voice integrated circuit according to claim 3, it is characterized in that: described one is a frame in short-term.
5. the implementation method of voice integrated circuit according to claim 3, it is characterized in that: described one is 10～30mS in short-term.
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| CN2009101066518A CN101866651B (en) | 2009-04-15 | 2009-04-15 | Method for implementing voice integrated circuit |
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| CN2009101066518A CN101866651B (en) | 2009-04-15 | 2009-04-15 | Method for implementing voice integrated circuit |
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| CN101866651B true CN101866651B (en) | 2012-02-22 |
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| CN106328153B (en) * | 2016-08-24 | 2020-05-08 | 青岛歌尔声学科技有限公司 | Electronic communication equipment voice signal processing system and method and electronic communication equipment |
| CN108075745B (en) * | 2016-11-17 | 2022-01-07 | 比亚迪半导体股份有限公司 | Digital filtering method and device |
| CN113225046B (en) * | 2021-04-14 | 2022-12-20 | 华南理工大学 | Dual-channel adaptive filter model order determination method for digital stethoscope |
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| CN1964471A (en) * | 2006-01-05 | 2007-05-16 | 宁波大学 | 2.4G public frequency band digital visual interphone and method to process audio image |
| CN1998265A (en) * | 2003-12-23 | 2007-07-11 | 奥迪吉康姆有限责任公司 | Digital cell phone with hearing aid functionality |
| CN101017428A (en) * | 2006-12-22 | 2007-08-15 | 广东电子工业研究院有限公司 | Embedded voice interaction device and interaction method thereof |
| CN101064875A (en) * | 2006-04-27 | 2007-10-31 | 叶波 | Vehicle-mounted handfree communication chip |
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2009
- 2009-04-15 CN CN2009101066518A patent/CN101866651B/en not_active Expired - Fee Related
Patent Citations (5)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| US5493617A (en) * | 1991-10-09 | 1996-02-20 | Waller, Jr.; James K. | Frequency bandwidth dependent exponential release for dynamic filter |
| CN1998265A (en) * | 2003-12-23 | 2007-07-11 | 奥迪吉康姆有限责任公司 | Digital cell phone with hearing aid functionality |
| CN1964471A (en) * | 2006-01-05 | 2007-05-16 | 宁波大学 | 2.4G public frequency band digital visual interphone and method to process audio image |
| CN101064875A (en) * | 2006-04-27 | 2007-10-31 | 叶波 | Vehicle-mounted handfree communication chip |
| CN101017428A (en) * | 2006-12-22 | 2007-08-15 | 广东电子工业研究院有限公司 | Embedded voice interaction device and interaction method thereof |
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