CN111554313A - Digital voice noise reduction device and method for telephone transmitter - Google Patents
Digital voice noise reduction device and method for telephone transmitter Download PDFInfo
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Abstract
The invention belongs to a device and a method for reducing noise of digital voice at a transmitting terminal. The device comprises a voice signal channel and a noise signal channel, wherein the voice signal channel comprises a microphone, a first filter circuit, a first A/D circuit, a DSP circuit and a D/A circuit, and the noise signal channel comprises a reference noise pickup microphone, a second filter circuit and a second A/D circuit; the DSP circuit converts the voice data signal without noise into analog voice signal through D/A circuit and outputs. The method comprises the steps of 1, signal acquisition; step 2, the DSP circuit realizes noise separation; and 3, self-adaptive adjustment of parameters, and error reduction. The invention processes the voice and the noise in the high-noise environment, completes the automatic separation of the noise and the voice signal, realizes the voice communication in the high-noise environment, has higher intelligibility and definition of the voice output after the noise reduction treatment, and has no phenomena of character leakage and noise tail support.
Description
Technical Field
The invention belongs to the technical field of digital voice signal processing, and particularly relates to a digital voice noise reduction device and a digital voice noise reduction method.
Background
In the daily communication process, the voice signal sent by the sender is often influenced by noise from the surrounding environment, so that the voice signal received by the receiver is not a pure voice signal but a noisy voice signal polluted by noise.
The noisy voice signal contaminated by the noise affects the call quality of the whole communication system, and even the call connection cannot be completed in serious cases. In order to reduce the influence of environmental noise and ensure the communication quality, people select a bone conduction type microphone or a second-order pressure difference dual-mode moving coil microphone to improve the signal-to-noise ratio of a signal at a transmitting end. The bone conduction microphone converts the vibration signal of the sounding part into an electric signal by using the vibration principle during speaking, and the noise hardly enables the sounding part to vibrate, so that the aim of reducing the noise is fulfilled. The disadvantage of this noise reduction is that the speech signal picked up by the microphone is poorly intelligible and uncomfortable to wear. The second-order pressure difference dual-mode moving coil microphone adopts the acoustic balance principle, and achieves the purpose of noise reduction according to the near-field and far-field effects of sound, and has the defect that the noise reduction effect is not obvious.
Disclosure of Invention
The invention aims to provide a device and a method for reducing noise of digital voice at a transmitting terminal, which are used for processing voice and noise in a high-noise environment, completing automatic separation of the noise and voice signals, realizing voice communication in the high-noise environment, and outputting voice with higher intelligibility and definition after noise reduction without character leakage and noise tail support.
The technical scheme of the invention is as follows:
a digital voice noise reduction device at a transmitter,
including the passage of a voice signal through the channel,
and a noise signal path,
the voice signal channel comprises a microphone, a first filter circuit, a first A/D circuit, a DSP circuit and a D/A circuit, and the noise signal channel comprises a reference noise pickup microphone, a second filter circuit and a second A/D circuit; pure voice signals and environment noise signals are input through a microphone to complete signal sound-electricity conversion, and then are sent into a DSP circuit through a first filter circuit and a first A/D circuit, and analog signals are digitally collected and encoded by the first A/D circuit and are converted into 12-bit digital signals to enter the DSP circuit; the environmental noise signal is input through a reference noise pickup microphone to complete the sound-electricity conversion of the signal, and then is sent to the DSP circuit through a second filter circuit and a second A/D circuit, and the second A/D circuit carries out digital acquisition and coding on the analog signal, converts the analog signal into a 12-bit digital signal and then enters the DSP circuit; the DSP circuit converts the voice data signal without noise into analog voice signal through the D/A circuit and outputs the analog voice signal, wherein the output signal is the voice analog signal without noise which is expected to be output.
The signal sent to the voice signal channel in the DSP circuit contains voice and noise signals, the signal sent to the noise signal channel in the DSP circuit only contains noise signals, and the noise signals in the two channels have correlation.
The DSP circuit is used as a main processor for noise reduction, a self-adaptive filter is adopted to enable the signal of a noise channel to be consistent with the noise component of a voice signal channel after being filtered, and then the voice signal channel signal is subtracted from the noise source channel signal after the filter, so that only voice data signals without noise are left.
The structure of the self-adaptive filter adopts an FIR (finite impulse response) with a symmetrical transverse structure, and the weight coefficient of the filter adopts a minimum mean square error criterion for iterative estimation.
The iterative estimation method based on the minimum mean square error criterion comprises the following steps: estimating the power spectral density of noise and signals, changing the weight coefficient of a filter, adjusting the signal amplitude, namely, performing square accumulation on 16 values of sampling data codes to obtain an average power value, comparing the average power value with the previous sample point power value, dividing the compared difference value by a set noise threshold value, wherein if the result is greater than 1, the weight coefficient of the adjusting filter is reduced, the signal output amplitude is increased, and if the result is less than or equal to 1, the weight coefficient of the filter is increased, and the signal output amplitude is reduced.
A method for reducing noise of digital voice at a transmitter comprises the following steps:
step 1, signal acquisition;
step 2, the DSP circuit realizes noise separation;
and 3, self-adaptive adjustment of parameters, and error reduction.
In the step 1, the signals collected by the voice signal channel include a pure voice signal s (n) and an environmental noise signal d (n); the signal collected by the ambient noise signal channel comprises an ambient noise signal r (n).
The step 2 comprises the following steps:
step 2.1, adaptively separating the noise in the input signal r (n), and then reversing the separated signal v (n);
and 2.2, comparing the inverted signal with an input signal y (n) through a noise canceller to obtain a signal e (n), wherein e (n) is a voice signal with noise separated.
The step 3 comprises the following steps:
step 3.1, adjusting parameters of the adaptive filter through an adaptive algorithm to enable the filtering effect of the adaptive filter to be better, enabling the output error of the signal e (n) to be smaller, repeating the above processes, enabling the adaptive filter to gradually know the statistical law about the input signal and the noise, and automatically adjusting the parameters according to the statistical law;
and 3.2, the adaptive filter works in an automatic tracking feedback state, once the statistical rule of the input signal changes, the output error signal e (n) becomes large, and at the moment, the step 3.1 is repeated.
In step 3.1, the adaptive algorithm includes the following steps:
setting the input data signal sequence χi(n) the desired output signal is d (n), and the error signal is defined as e (n), wherein the relationship between the three signals is as follows:
wherein, ω isiIs the filter weight coefficient;
the weight coefficient omega of the filter is obtained through derivationiThe final expression is as follows:
ω(k+1)＝ω(k)+2μe(k)χ(k) (2)
u is a convergence factor, and the rate of noise reduction convergence is controlled;
finding the optimal weight coefficient wiMinimizing the error signal e (n);
the derivation process includes the steps of: setting initial values of a filter W (k); calculating an estimated value of actual output of the filter; calculating an estimation error; updating the filter coefficient at the k +1 moment; changing k to k +1, and repeating the steps 2-4.
The invention has the beneficial effects that:
(1) the invention provides a device and a method for reducing noise of digital voice at a transmitter, which apply the modern Digital Signal Processing (DSP) technology and the characteristic of high-speed real-time processing operation thereof, adopt a corresponding algorithm to process voice and noise in a high-noise environment, complete the automatic separation of the noise and the voice signal, realize the voice communication contact in the high-noise environment, and output voice after noise reduction has higher intelligibility and definition and has no phenomena of missing words and noise tail support;
(2) the invention provides a device and a method for reducing noise of digital voice at a transmitter, which are realized by an adaptive filter, wherein the adaptive filter automatically tracks the change of signals and noise, and can achieve the optimal voice noise reduction effect. The device generally has the noise suppression amount of more than 20 decibels, and has stable output voice without character leakage and noise tailing phenomena;
(3) the invention provides a device and a method for reducing noise of digital voice of a transmitter, which are used as a digital anti-noise processing unit and can be used for communication in high-noise environments such as an airborne communication terminal and the like.
Drawings
FIG. 1 is a block diagram of a digital voice noise reduction apparatus for a transmitting terminal according to the present invention;
fig. 2 is a schematic diagram of adaptive noise cancellation of a digital voice noise reduction apparatus at a transmitter according to the present invention.
Detailed Description
The following describes a device and a method for reducing noise of a digital voice at a transmitting end in further detail with reference to the accompanying drawings and specific embodiments.
As shown in fig. 1, a digital voice noise reduction device at a transmitting end comprises a voice signal channel and a noise signal channel, wherein the voice signal channel comprises a transmitter, a first filter circuit, a first a/D circuit, a DSP circuit and a D/a circuit, and the noise signal channel comprises a reference noise pickup microphone, a second filter circuit and a second a/D circuit;
the speech coding and decoding of the voice signal channel and the noise signal channel adopt a universal audio coding and decoding chip, and the filtering and A/D, D/A functions of the voice signal channel and the noise signal channel are respectively realized by utilizing the stereo left and right channels of the chip;
pure voice signals and environment noise signals are input through a microphone to complete signal sound-electricity conversion, and then are sent into a DSP circuit through a first filter circuit and a first A/D circuit, and analog signals are digitally collected and encoded by the first A/D circuit and are converted into 12-bit digital signals to enter the DSP circuit;
the environmental noise signal is input through a reference noise pickup microphone to complete the sound-electricity conversion of the signal, and then is sent to the DSP circuit through a second filter circuit and a second A/D circuit, and the second A/D circuit carries out digital acquisition and coding on the analog signal, converts the analog signal into a 12-bit digital signal and then enters the DSP circuit;
the signal sent to the voice signal channel in the DSP circuit contains voice and noise signals, the signal sent to the noise signal channel in the DSP circuit only contains noise signals, and the noise signals in the two channels have correlation;
the DSP circuit is used as a main processor for noise reduction, a self-adaptive filter is adopted to ensure that the signal of a noise channel is consistent with the noise component of a voice signal channel after being filtered, and then the voice signal channel signal is subtracted from the noise source channel signal after the filter, so that only voice data signals without noise are left;
the DSP circuit converts the voice data signal without noise into analog voice signal through the D/A circuit and outputs the analog voice signal, wherein the output signal is the voice analog signal without noise which is expected to be output.
The structure of the self-adaptive filter adopts the FIR of the traditional symmetrical transverse structure, the weight coefficient of the filter adopts the Minimum Mean Square Error (MMSE) criterion for iterative estimation, and the method is as follows: estimating the power spectral density of noise and signals, changing the weight coefficient of a filter, adjusting the signal amplitude, namely, performing square accumulation on 16 values of sampling data codes to obtain an average power value, comparing the average power value with the previous sample point power value, dividing the compared difference value by a set noise threshold value, wherein if the result is greater than 1, the weight coefficient of the adjusting filter is reduced, the signal output amplitude is increased, and if the result is less than or equal to 1, the weight coefficient of the filter is increased, and the signal output amplitude is reduced.
As shown in fig. 2, a method for reducing noise of digital voice at a sender is based on the following principle: the microphone and the reference noise pickup microphone receive an input signal y (n) and a signal r (n), wherein the input signal y (n) comprises an original voice signal s (n) and an environment noise signal d (n), the input signal r (n) is a reference noise input, the signal v (n) is a noise signal separated by an adaptive filter, and the signal e (n) is a voice signal expected after noise is filtered; the adaptive filter of the DSP circuit automatically adjusts the filter parameters at the current moment by using the filter parameters obtained at the previous moment so as to adapt to the unknown or time-varying statistical characteristics of the voice signal and the noise signal, thereby realizing the noise filtering separation.
Specifically, the method comprises the following steps:
step 1, inputting a pure voice signal s (n) and an environmental noise signal D (n) through a microphone to complete signal sound-electricity conversion, and then sending the signals into a DSP circuit through a first filter circuit and a first A/D circuit, wherein the first A/D circuit carries out digital acquisition and coding on an analog signal, converts the analog signal into a 12-bit digital signal and then sends the digital signal into the DSP circuit;
step 2, an environmental noise signal r (n) is input through a reference noise pickup microphone to complete signal sound-electricity conversion, and then is sent to a DSP circuit through a second filter circuit and a second A/D circuit, and analog signals are digitally collected and encoded by the second A/D circuit and are converted into 12-bit digital signals to enter the DSP circuit;
and 3, the DSP circuit realizes noise separation through the self-adaptive filter.
Step 3.1,
Adaptively separating noise in an input signal r (n), and then reversing the separated signal v (n);
step 3.2,
v (n) the inverted signal is compared with the input signal y (n) by a noise canceller to obtain a signal e (n), wherein e (n) is the voice signal with the noise separated;
step 3.3,
The parameters of the adaptive filter are adjusted through an adaptive algorithm by the signals e (n) and r (n), so that the filtering effect of the adaptive filter is better, the output error of the signals e (n) is smaller, the process is repeated, the adaptive filter gradually learns the statistical rule of the input signals and the noise, and the parameters of the adaptive filter are automatically adjusted according to the statistical rule;
and 3.4, the adaptive filter works in an automatic tracking feedback state, once the statistical rule of the input signal changes, the output error signal e (n) becomes large, and at the moment, the step 3.3 is repeated.
In step 3.3, the basic principle of adaptive noise reduction of the adaptive filter is to utilize the filter parameters obtained at the previous moment to automatically adjust the current filter parameters so as to adapt to the unknown or randomly changed statistical characteristics of signals and noise, thereby realizing optimal filtering,
the algorithm criterion comprises the following steps:
input data signal sequence xi(n) the desired output signal is d (n), and the error signal is defined as e (n), wherein the relationship between the three signals is as follows:
wherein wiFor the filter weight coefficients, the essence of the algorithm is to find the optimal weight coefficient wiSo that the error signal e (n) is minimized.
Obtaining the filter weight coefficient w through a series of derivationiThe final expression is as follows:
ω(k+1)＝ω(k)+2μe(k)χ(k) (2)
u is a convergence factor, controlling the rate of noise reduction convergence.
The derivation process comprises the following steps:
1) setting initial values of a filter W (k);
2) calculating an estimated value of actual output of the filter;
3) calculating an estimation error;
4) updating the filter coefficient at the k +1 moment;
5) changing k into k +1, and repeating the steps 2-4.
Claims (10)
1. A device for reducing noise of digital voice at a transmitter is characterized in that:
including the passage of a voice signal through the channel,
and a noise signal path,
the voice signal channel comprises a microphone, a first filter circuit, a first A/D circuit, a DSP circuit and a D/A circuit, and the noise signal channel comprises a reference noise pickup microphone, a second filter circuit and a second A/D circuit; pure voice signals and environment noise signals are input through a microphone to complete signal sound-electricity conversion, and then are sent into a DSP circuit through a first filter circuit and a first A/D circuit, and analog signals are digitally collected and encoded by the first A/D circuit and are converted into 12-bit digital signals to enter the DSP circuit; the environmental noise signal is input through a reference noise pickup microphone to complete the sound-electricity conversion of the signal, and then is sent to the DSP circuit through a second filter circuit and a second A/D circuit, and the second A/D circuit carries out digital acquisition and coding on the analog signal, converts the analog signal into a 12-bit digital signal and then enters the DSP circuit; the DSP circuit converts the voice data signal without noise into analog voice signal through the D/A circuit and outputs the analog voice signal, wherein the output signal is the voice analog signal without noise which is expected to be output.
2. A handset digital speech noise reduction apparatus as defined in claim 1, wherein: the signal sent to the voice signal channel in the DSP circuit contains voice and noise signals, the signal sent to the noise signal channel in the DSP circuit only contains noise signals, and the noise signals in the two channels have correlation.
3. A handset digital speech noise reduction apparatus as defined in claim 2, wherein: the DSP circuit is used as a main processor for noise reduction, a self-adaptive filter is adopted to enable the signal of a noise channel to be consistent with the noise component of a voice signal channel after being filtered, and then the voice signal channel signal is subtracted from the noise source channel signal after the filter, so that only voice data signals without noise are left.
4. A handset digital speech noise reduction apparatus as defined in claim 3, wherein: the structure of the self-adaptive filter adopts an FIR (finite impulse response) with a symmetrical transverse structure, and the weight coefficient of the filter adopts a minimum mean square error criterion for iterative estimation.
5. A handset digital speech noise reduction apparatus as defined in claim 4, wherein: the iterative estimation method based on the minimum mean square error criterion comprises the following steps: estimating the power spectral density of noise and signals, changing the weight coefficient of a filter, adjusting the signal amplitude, namely, performing square accumulation on 16 values of sampling data codes to obtain an average power value, comparing the average power value with the previous sample point power value, dividing the compared difference value by a set noise threshold value, wherein if the result is greater than 1, the weight coefficient of the adjusting filter is reduced, the signal output amplitude is increased, and if the result is less than or equal to 1, the weight coefficient of the filter is increased, and the signal output amplitude is reduced.
6. A method of reducing noise in a digital speech noise reducing device at a telephone transmitter as claimed in any one of claims 1 to 5, characterized by: the method comprises the following steps:
step 1, signal acquisition;
step 2, the DSP circuit realizes noise separation;
and 3, self-adaptive adjustment of parameters, and error reduction.
7. The method of claim 6, wherein the method further comprises: in the step 1, the signals collected by the voice signal channel include a pure voice signal s (n) and an environmental noise signal d (n); the signal collected by the ambient noise signal channel comprises an ambient noise signal r (n).
8. The method of claim 7, wherein the method further comprises: the step 2 comprises the following steps:
step 2.1, adaptively separating the noise in the input signal r (n), and then reversing the separated signal v (n);
and 2.2, comparing the inverted signal with an input signal y (n) through a noise canceller to obtain a signal e (n), wherein e (n) is a voice signal with noise separated.
9. The method of claim 8, wherein the method further comprises: the step 3 comprises the following steps:
step 3.1, adjusting parameters of the adaptive filter through an adaptive algorithm to enable the filtering effect of the adaptive filter to be better, enabling the output error of the signal e (n) to be smaller, repeating the above processes, enabling the adaptive filter to gradually know the statistical law about the input signal and the noise, and automatically adjusting the parameters according to the statistical law;
and 3.2, the adaptive filter works in an automatic tracking feedback state, once the statistical rule of the input signal changes, the output error signal e (n) becomes large, and at the moment, the step 3.1 is repeated.
10. The method of claim 9, wherein the method further comprises: in step 3.1, the adaptive algorithm includes the following steps:
setting the input data signal sequence χi(n) the desired output signal is d (n), and the error signal is defined as e (n), wherein the relationship between the three signals is as follows:
wherein, ω isiIs the filter weight coefficient;
the weight coefficient omega of the filter is obtained through derivationiThe final expression is as follows:
ω(k+1)＝ω(k)+2μe(k)χ(k) (2)
u is a convergence factor, and the rate of noise reduction convergence is controlled;
finding the optimal weight coefficient wiMinimizing the error signal e (n);
the derivation process includes the steps of: setting initial values of a filter W (k); calculating an estimated value of actual output of the filter; calculating an estimation error; updating the filter coefficient at the k +1 moment; changing k to k +1, and repeating the steps 2-4.
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