CN114333878A - Noise reduction system of wireless microphone - Google Patents
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Abstract
The invention discloses a noise reduction system of a wireless microphone, which comprises: the wireless receiving unit receives voice and transmits the voice to the voice processing unit, the voice processing unit converts the voice into digital audio signals of a fixed frequency band and transmits the digital audio signals to the digital signal processing unit, the digital signal processing unit carries out filtering processing on the digital audio signals, carries out noise reduction processing through echo suppression and noise reduction algorithms and stores the processed digital audio signals in the data control unit, and the data control unit stores the digital audio signals and analyzes noise existing in the digital audio signals. The effect of making an uproar effectively is reduced to the user when having ensured to use wireless microphone, has promoted user experience.
Description
Technical Field
The invention relates to the field of noise reduction processing, in particular to a noise reduction system of a wireless microphone.
Background
With the development of scientific technology, wireless microphones have been widely used in various fields, such as voice communication devices, teleconferencing systems, news interview sites, monitoring, voice recognition, and various consumer electronic products requiring voice communication, but in these applications, wireless microphones are generally present in a single form.
The audio signal picked up by the single wireless microphone is a linear superposition of the sound field in the recognizable range of the wireless microphone at the position of the sound field. Therefore, the sound signal picked up by a single wireless microphone has noise, echo and the like, which affect the quality of sound, and the current wireless microphone cannot distinguish different sound sources, cannot separate environmental noise at the position, and cannot extract only a desired signal, but actually only is interested in one or more sound sources in the signal picked up by the wireless microphone, so that the problem that the current wireless microphone is urgently needed to solve is that the signal picked up by the wireless microphone can distinguish different voices, remove echoes and reduce noise.
Disclosure of Invention
In the summary section, a series of concepts in a simplified form are introduced, which will be described in further detail in the detailed description section; this summary of the invention is not intended to identify key features or essential features of the claimed subject matter, nor is it intended to be used as an aid in determining the scope of the claimed subject matter.
To at least partially solve the above problems, the present invention provides a noise reduction system for a wireless microphone, comprising: the wireless receiving unit receives voice and transmits the voice to the voice processing unit, the voice processing unit converts the voice into digital audio signals of a fixed frequency band and transmits the digital audio signals to the digital signal processing unit, the digital signal processing unit carries out filtering processing on the digital audio signals, carries out noise reduction processing through echo suppression and noise reduction algorithms and stores the processed digital audio signals in the data control unit, and the data control unit stores the digital audio signals and analyzes noise existing in the digital audio signals.
Wherein the voice processing unit includes: the voice is converted into an electric signal through the piezoelectric converter, the electric signal is amplified through the audio amplifier, and the amplified electric signal is converted into a digital audio signal through the audio analog-to-digital converter.
Wherein the digital signal processing unit includes: the device comprises a frequency domain filtering module, a self-adaptive beam module, a post-filtering module and an echo suppression module, wherein the frequency domain filtering module is used for executing a channel noise reduction algorithm on a digital audio signal through statistical parameters to primarily suppress noise; the self-adaptive beam module is used for removing noise from the digital audio signal to obtain a pure digital audio signal; the post-filter module is used for calculating a noise signal by using multiple channels and a filter algorithm to further suppress residual noise; the echo suppression module is used for carrying out echo change estimation on the digital audio signal and suppressing echo.
Wherein the data management unit includes: the device comprises an audio quality sensing module, a noise estimation module and a memory module, wherein the audio quality sensing module is used for calculating the difference operation of a loudness spectrum, acquiring disturbance data and judging the audio quality in a digital audio signal; the noise estimation module is used for smoothing the band noise frequency power spectrum and estimating the noise in the digital audio signal by searching the smoothed band noise frequency power spectrum with the local minimum; the memory module is used for storing and deleting voice data.
Wherein the adaptive beam module comprises: a fixed beam forming sub-module, a blocking matrix sub-module and a self-adaptive cancellation sub-module; and an upper branch is formed by the fixed beam forming submodule, a lower branch is formed by the blocking matrix submodule and the self-adaptive cancellation submodule, and a multi-channel self-adaptive filtering structure is formed by the fixed beam forming submodule, the blocking matrix submodule and the self-adaptive cancellation submodule.
Wherein the blocking matrix submodule is used for blocking a desired audio signal in the digital audio signal; by adopting a method of constrained adaptive filtering and only updating the blocking matrix submodule when the expected audio exists, and only updating the coefficient of the adaptive cancellation submodule when the expected voice does not exist, the leakage of the expected voice signal is reduced; and a beam with directivity is formed by the self-adaptive beam module and points to the direction of a desired signal, so that the interference in an undesired direction is eliminated.
Wherein, the echo suppression module includes: the echo detection sub-module establishes an audio model of a far-end signal, estimates an echo and continuously modifies the coefficient of the low-pass filtering sub-module to enable an estimated value to approach to a real echo, and offsets an echo estimated value from an output signal of the wireless microphone, has an acoustic response function under an acoustic environment between a dynamic learning loudspeaker and the wireless microphone, and respectively processes the acoustic response function in a plurality of sub-frequency bands.
The audio quality perception module comprises an audio quality detection submodule and a low-pass filtering submodule, the audio quality detection submodule adjusts an initial audio signal and a noise-reduced signal to the same level of auditory intensity, the low-pass filtering submodule simulates the frequency response of the signal passing through a microphone, the signal is aligned according to time, a loudness spectrum is obtained through auditory conversion processing, the difference of the loudness spectrum is calculated, disturbance data are obtained, and therefore the audio quality is judged.
Wherein the audio quality perception module further comprises: the system comprises a voice positioning submodule and a signal processing submodule, wherein the voice positioning submodule constructs a spatial spectrum according to the orthogonality of a signal subspace and a noise subspace, and carries out average calculation on the spatial spectrums of all sub-bands to obtain a broadband signal spatial spectrum estimation; the signal processing submodule frames the signal, the time domain of the framed signal is converted into the frequency domain through Fourier transform, the signal of each channel under the same frequency is combined into a snapshot, the covariance matrix is decomposed by calculating the covariance matrix of each snapshot to obtain a signal subspace and a noise subspace, the spatial spectrum power is calculated by combining the array manifold, and the peak value is found to be the positioning result.
Wherein the audio amplifier comprises: the system comprises a voice signal enhancer module and a frequency spectrum processing submodule, wherein the voice signal enhancer module is used for windowing, framing and Fourier transform on original voice polluted by noise to obtain an initial voice frequency spectrum; cutting a sound phase spectrum with noise into sub-bands which are not repeated and do not interfere with each other randomly in the process of preprocessing an initial voice frequency spectrum; calculating and solving an over-subtraction factor of each sub-band through the frequency spectrum processing sub-module, independently executing spectral subtraction in each sub-band, obtaining the spectral estimation of the enhanced voice signal of each sub-band by using a frequency spectrum subtraction method, recombining the processed sub-bands, obtaining a complete enhanced acoustic phase spectrum by combining an acoustic phase spectrum with noise, and performing inverse Fourier transform to obtain the enhanced voice.
Compared with the prior art, the invention has the following advantages:
the invention provides a noise reduction system of a wireless microphone, comprising: the wireless receiving unit receives voice and transmits the voice to the voice processing unit, the voice processing unit converts the voice into digital audio signals of a fixed frequency band and transmits the digital audio signals to the digital signal processing unit, the digital signal processing unit carries out filtering processing on the digital audio signals, carries out noise reduction processing through echo suppression and noise reduction algorithms and stores the processed digital audio signals in the data control unit, and the data control unit stores the digital audio signals and analyzes noise existing in the digital audio signals. Therefore, the sensitivity and the precision of the wireless microphone for picking up the voice are improved, the voice effect is enhanced, the distortion rate is reduced, and the effect of inhibiting the environmental noise is achieved.
Additional features and advantages of the invention will be set forth in the description which follows, and in part will be obvious from the description, or may be learned by practice of the invention. The objectives and other advantages of the invention will be realized and attained by the structure particularly pointed out in the written description and claims hereof as well as the appended drawings.
The technical solution of the present invention is further described in detail by the accompanying drawings and embodiments.
Drawings
The accompanying drawings, which are included to provide a further understanding of the invention and are incorporated in and constitute a part of this specification, illustrate embodiments of the invention and together with the description serve to explain the principles of the invention and not to limit the invention. In the drawings:
FIG. 1 is a block diagram of a noise reduction system of a wireless microphone according to an embodiment of the present invention;
FIG. 2 is a flow chart of a noise reduction system for a wireless microphone according to an embodiment of the present invention;
fig. 3 is a block diagram of a digital signal processing unit in a noise reduction system of a wireless microphone according to an embodiment of the present invention.
Detailed Description
The preferred embodiments of the present invention will be described in conjunction with the accompanying drawings, and it will be understood that they are described herein for the purpose of illustration and explanation and not limitation.
An embodiment of the present invention provides a noise reduction system for a wireless microphone, referring to fig. 1 to 3, including: the wireless receiving unit receives voice and transmits the voice to the voice processing unit, the voice processing unit converts the voice into digital audio signals of a fixed frequency band and transmits the digital audio signals to the digital signal processing unit, the digital signal processing unit carries out filtering processing on the digital audio signals, carries out noise reduction processing through echo suppression and noise reduction algorithms and stores the processed digital audio signals in the data control unit, and the data control unit stores the digital audio signals and analyzes noise existing in the digital audio signals.
The working principle of the technical scheme is as follows: the wireless receiving unit receives voice and then transmits the voice to the voice processing unit, the voice processing unit converts the voice into digital audio signals of a fixed frequency band and transmits the digital audio signals to the digital signal processing unit, the digital signal processing unit carries out filtering processing on the digital audio signals, carries out noise reduction processing through echo suppression and noise reduction algorithms and stores the processed digital audio signals in the data control unit, and the data control unit stores the digital audio signals and analyzes noise existing in the digital audio signals. Therefore, the sensitivity and the precision of the wireless microphone for picking up the voice are improved, the voice effect is enhanced, the distortion rate is reduced, and the effect of inhibiting the environmental noise is achieved.
The beneficial effects of the above technical scheme are: through with speech transmission extremely after wireless receiving element received pronunciation speech processing unit, speech processing unit converts speech conversion into the digital audio signal of fixed frequency channel, improves the sensitivity and the precision that wireless microphone picked up pronunciation through the processing to pronunciation, digital signal processing unit carries out filtering process and suppresses, falls the algorithm of making an uproar through the echo with digital audio signal and falls the processing of making an uproar, and effectual processing pronunciation plays the effect of suppressing the echo and noise reduction, data management and control unit stores digital audio signal and carries out the analysis to the noise that exists in the digital audio signal, and the later stage user of being convenient for can look over the pronunciation of storage.
In another embodiment, the speech processing unit comprises: the voice is converted into an electric signal through the piezoelectric converter, the electric signal is amplified through the audio amplifier, and the amplified electric signal is converted into a digital audio signal through the audio analog-to-digital converter.
The working principle of the technical scheme is as follows: the voice is converted into the electric signal through the piezoelectric converter, the electric signal is amplified through the audio amplifier, and the amplified electric signal is converted into the digital audio signal through the audio analog-to-digital converter, so that the quantization noise processed by a rear-stage digital end is reduced as much as possible under the condition that the amplitude of the signal is not distorted.
The beneficial effects of the above technical scheme are: the voice is converted into an electric signal through the piezoelectric converter, the electric signal is amplified through the audio amplifier, and the amplified electric signal is converted into a digital audio signal through the audio analog-to-digital converter. The quantization noise processed by the digital end of the later stage is reduced as much as possible under the condition that the amplitude of the signal is not distorted, and the digital signal processing unit is convenient to process the noise.
In another embodiment, referring to fig. 3, the digital signal processing unit includes: the device comprises a frequency domain filtering module, a self-adaptive beam module, a post-filtering module and an echo suppression module, wherein the frequency domain filtering module is used for executing a channel noise reduction algorithm on a digital audio signal through statistical parameters to primarily suppress noise; the self-adaptive beam module is used for removing noise from the digital audio signal to obtain a pure digital audio signal; the post-filter module is used for calculating a noise signal by using multiple channels and a filter algorithm to further suppress residual noise; the echo suppression module is used for carrying out echo change estimation on the digital audio signal and suppressing echo.
The working principle of the technical scheme is as follows: the frequency domain filtering module is used for executing a channel noise reduction algorithm on the digital audio signal through statistical parameters to preliminarily suppress noise; the self-adaptive beam module is used for removing noise from the digital audio signal to obtain a pure digital audio signal; the post-filter module is used for calculating a noise signal by utilizing multiple channels and a filter algorithm, and further inhibiting residual noise so as to achieve a superior noise reduction effect; the echo suppression module is used for carrying out echo change estimation on the digital audio signal and suppressing echo, so that the signal is ensured to be a clean uplink signal.
The generalized cross-correlation method is a method for converting cross-correlation numbers into time domains after weighting in the frequency domain, so that the power of noise can be well reduced, the maximum value of the cross-correlation numbers is accurately and definitely displayed, and the time delay estimation is accurate and correct. Experiments prove that when the independent variable of the cross-correlation number is the time delay between two signals, the value of the cross-correlation number is the largest, and therefore noise is better reduced. Wherein, the cross-power spectrum is:
Gij(w) a cross-power spectrum representing a discrete fourier transform of the ith, j signal; rhoij(w) weights representing the generalized cross-correlation; xi(w) denotes a wireless microphoneA signal i collected by wind; xj(w) represents a signal j acquired by the wireless microphone; e.g. of the type-jwtRepresents a negative exponential signal; | R (w) & gtelectrically2Which represents the modulo square coherence number of the signal acquired by the wireless microphone.
The beneficial effects of the above technical scheme are: the frequency domain filtering module is used for executing a channel noise reduction algorithm on the digital audio signal through statistical parameters to preliminarily suppress noise; the self-adaptive beam module is used for removing noise from the digital audio signal to obtain a pure digital audio signal; the post-filter module is used for calculating a noise signal by utilizing multiple channels and a filter algorithm, and further inhibiting residual noise so as to achieve a superior noise reduction effect; the echo suppression module is used for carrying out echo change estimation on the digital audio signal and suppressing echo, so that the signal is ensured to be a clean uplink signal, and a user is ensured not to hear own voice. Thereby enhancing the target signal and suppressing interference and noise in the audio signal.
In another embodiment, the data manipulating unit includes: the device comprises an audio quality sensing module, a noise estimation module and a memory module, wherein the audio quality sensing module is used for calculating the difference operation of a loudness spectrum, acquiring disturbance data and judging the audio quality in a digital audio signal; the noise estimation module is used for smoothing the band noise frequency power spectrum and estimating the noise in the digital audio signal by searching the smoothed band noise frequency power spectrum with the local minimum; the memory module is used for storing and deleting voice data.
The working principle of the technical scheme is as follows: the audio quality perception module is used for calculating the difference operation of the loudness spectrum, acquiring disturbance data and judging the audio quality in the digital audio signal; the noise estimation module is used for smoothing the band noise frequency power spectrum and estimating the noise in the digital audio signal by searching the smoothed band noise frequency power spectrum with the local minimum; the memory module is used for storing and deleting voice data.
The beneficial effects of the above technical scheme are: the audio quality perception module is used for calculating the difference operation of the loudness spectrum, acquiring disturbance data and judging the audio quality in the digital audio signal; the noise estimation module is used for smoothing the band noise frequency power spectrum and estimating the noise in the digital audio signal by searching the smoothed band noise frequency power spectrum with the local minimum; the memory module is used for storing and deleting voice data, can ensure data reading and writing to the maximum extent, and has the reliability of typical 10-ten-thousand erasing life cycle storage.
In another embodiment, the adaptive beam module comprises: a fixed beam forming sub-module, a blocking matrix sub-module and a self-adaptive cancellation sub-module; and an upper branch is formed by the fixed beam forming submodule, a lower branch is formed by the blocking matrix submodule and the self-adaptive cancellation submodule, and a multi-channel self-adaptive filtering structure is formed by the fixed beam forming submodule, the blocking matrix submodule and the self-adaptive cancellation submodule.
The working principle of the technical scheme is as follows: and an upper branch is formed by the fixed beam forming submodule, a lower branch is formed by the blocking matrix submodule and the self-adaptive cancellation submodule, and a multi-channel self-adaptive filtering structure is formed by the fixed beam forming submodule, the blocking matrix submodule and the self-adaptive cancellation submodule.
The beneficial effects of the above technical scheme are: and an upper branch is formed by the fixed beam forming submodule, a lower branch is formed by the blocking matrix submodule and the self-adaptive cancellation submodule, and a multi-channel self-adaptive filtering structure is formed by the fixed beam forming submodule, the blocking matrix submodule and the self-adaptive cancellation submodule. The self-adaptive beam module uses different distributions of the signal source and the noise source in the space to enable the array beam to point to the direction of the signal source and suppress the direction of the noise source, thereby achieving the effect of improving the signal-to-interference ratio.
In another embodiment, the blocking matrix submodule is configured to block a desired audio signal of the digital audio signals; by adopting a method of constrained adaptive filtering and only updating the blocking matrix submodule when the expected audio exists, and only updating the coefficient of the adaptive cancellation submodule when the expected voice does not exist, the leakage of the expected voice signal is reduced; and a beam with directivity is formed by the self-adaptive beam module and points to the direction of a desired signal, so that the interference in an undesired direction is eliminated.
The working principle of the technical scheme is as follows: the blocking matrix submodule is used for blocking a desired audio signal in the digital audio signal; by adopting a method of constrained adaptive filtering and only updating the blocking matrix submodule when the expected audio exists, and only updating the coefficient of the adaptive cancellation submodule when the expected voice does not exist, the leakage of the expected voice signal is reduced; and a beam with directivity is formed by the self-adaptive beam module and points to the direction of a desired signal, so that the interference in an undesired direction is eliminated. Therefore, the voice signal is better reduced in noise, and a better noise reduction effect is achieved.
The beneficial effects of the above technical scheme are: the blocking matrix submodule is used for blocking a desired audio signal in the digital audio signal; by adopting a method of constrained adaptive filtering and only updating the blocking matrix submodule when the expected audio exists, and only updating the coefficient of the adaptive cancellation submodule when the expected voice does not exist, the leakage of the expected voice signal is reduced; the self-adaptive beam module forms a directional beam which points to the direction of an expected signal and eliminates the interference in an unexpected direction, thereby better finishing the noise reduction of the voice signal and achieving better noise reduction effect.
In another embodiment, the echo suppression module includes: the echo detection sub-module establishes an audio model of a far-end signal, estimates an echo and continuously modifies the coefficient of the low-pass filtering sub-module to enable an estimated value to approach to a real echo, and offsets an echo estimated value from an output signal of the wireless microphone, has an acoustic response function under an acoustic environment between a dynamic learning loudspeaker and the wireless microphone, and respectively processes the acoustic response function in a plurality of sub-frequency bands.
The working principle of the technical scheme is as follows: the echo detection submodule establishes an audio model of a far-end signal, estimates an echo and continuously modifies the coefficient of the low-pass filtering submodule to enable the estimated value to approach a real echo, and offsets the echo estimated value from an output signal of the wireless microphone, has an acoustic response function under the acoustic environment between a dynamic learning loudspeaker and the wireless microphone, and respectively processes the acoustic response function in a plurality of sub-frequency bands.
The beneficial effects of the above technical scheme are: the echo detection sub-module establishes an audio model of a far-end signal, estimates an echo and continuously modifies the coefficient of the low-pass filter sub-module to enable the estimated value to approach to a real echo, and offsets the echo estimated value from the output signal of the wireless microphone, has an acoustic response function under the acoustic environment between a dynamic learning loudspeaker and the wireless microphone, and respectively processes the acoustic response function in a plurality of sub-frequency bands.
In another embodiment, the audio quality sensing module includes an audio quality detection sub-module and a low-pass filtering sub-module, the audio quality detection sub-module adjusts an initial audio signal and a noise-reduced signal to a level with the same hearing intensity, the low-pass filtering sub-module simulates a frequency response of the signal passing through a microphone, the signals are aligned according to time, a loudness spectrum is obtained through hearing transformation processing, a difference of the loudness spectrum is calculated, and disturbance data is obtained, so that the audio quality is judged.
The working principle of the technical scheme is as follows: the audio quality detection submodule adjusts an initial audio signal and a noise-reduced signal to the level of the same hearing intensity, the low-pass filtering submodule simulates the frequency response of the signal passing through a microphone, the signals are aligned according to time, a loudness spectrum is obtained through hearing transformation processing, the difference of the loudness spectrum is calculated, disturbance data are obtained, and therefore the audio quality is judged.
The beneficial effects of the above technical scheme are: the audio quality detection submodule adjusts an initial audio signal and a noise-reduced signal to be at the same level of auditory intensity, the low-pass filtering submodule simulates the frequency response of the signal passing through a microphone, the signals are aligned according to time, a loudness spectrum is obtained through auditory transformation processing, the difference of the loudness spectrum is calculated, disturbance data is obtained, and therefore audio quality is judged, and the audio quality can be accurately evaluated according to the occurring voice distortion condition to obtain the best objective audio quality evaluation value.
In another embodiment, the audio quality perception module further comprises: the system comprises a voice positioning submodule and a signal processing submodule, wherein the voice positioning submodule constructs a spatial spectrum according to the orthogonality of a signal subspace and a noise subspace, and carries out average calculation on the spatial spectrums of all sub-bands to obtain a broadband signal spatial spectrum estimation; the signal processing submodule frames the signal, the time domain of the framed signal is converted into the frequency domain through Fourier transform, the signal of each channel under the same frequency is combined into a snapshot, the covariance matrix is decomposed by calculating the covariance matrix of each snapshot to obtain a signal subspace and a noise subspace, the spatial spectrum power is calculated by combining the array manifold, and the peak value is found to be the positioning result.
The working principle of the technical scheme is as follows: the voice positioning submodule constructs a spatial spectrum according to the orthogonality of the signal subspace and the noise subspace, and carries out average calculation on the spatial spectrums of all the sub-bands to obtain the estimation of the spatial spectrum of the broadband signal; the signal processing submodule frames the signal, the time domain of the framed signal is converted into the frequency domain through Fourier transform, the signal of each channel under the same frequency is combined into a snapshot, the covariance matrix is decomposed by calculating the covariance matrix of each snapshot to obtain a signal subspace and a noise subspace, the spatial spectrum power is calculated by combining the array manifold, and the peak value is found to be the positioning result.
The beneficial effects of the above technical scheme are: the voice positioning submodule constructs a spatial spectrum according to the orthogonality of the signal subspace and the noise subspace, and carries out average calculation on the spatial spectrums of all the sub-bands to obtain the estimation of the spatial spectrum of the broadband signal; the signal processing submodule frames the signal, the time domain of the framed signal is converted into the frequency domain through Fourier transform, the signal of each channel under the same frequency is combined into a snapshot, the covariance matrix is decomposed by calculating the covariance matrix of each snapshot to obtain a signal subspace and a noise subspace, the spatial spectrum power is calculated by combining the array manifold, the peak value is found to be the positioning result, and therefore the position of the sound source is accurately judged.
In another embodiment, the audio amplifier comprises: the system comprises a voice signal enhancer module and a frequency spectrum processing submodule, wherein the voice signal enhancer module is used for windowing, framing and Fourier transform on original voice polluted by noise to obtain an initial voice frequency spectrum; cutting a sound phase spectrum with noise into sub-bands which are not repeated and do not interfere with each other randomly in the process of preprocessing an initial voice frequency spectrum; calculating and solving an over-subtraction factor of each sub-band through the frequency spectrum processing sub-module, independently executing spectral subtraction in each sub-band, obtaining the spectral estimation of the enhanced voice signal of each sub-band by using a frequency spectrum subtraction method, recombining the processed sub-bands, obtaining a complete enhanced acoustic phase spectrum by combining an acoustic phase spectrum with noise, and performing inverse Fourier transform to obtain the enhanced voice.
The working principle of the technical scheme is as follows: windowing, framing and Fourier transforming the original voice polluted by noise through the voice signal enhancer module to obtain an initial voice frequency spectrum; cutting a sound phase spectrum with noise into sub-bands which are not repeated and do not interfere with each other randomly in the process of preprocessing an initial voice frequency spectrum; calculating and solving an over-subtraction factor of each sub-band through the frequency spectrum processing sub-module, independently executing spectral subtraction in each sub-band, obtaining the spectrum estimation of the enhanced voice signal of each sub-band by using a frequency spectrum subtraction method, recombining the processed sub-bands, obtaining a complete enhanced acoustic phase spectrum by combining an acoustic phase spectrum with noise, and performing inverse Fourier transform to obtain the enhanced voice, so that the sensitivity and the accuracy of the wireless microphone for picking up the voice are improved, and the voice effect is enhanced.
The beneficial effects of the above technical scheme are: windowing, framing and Fourier transforming the original voice polluted by noise through the voice signal enhancer module to obtain an initial voice frequency spectrum; cutting a sound phase spectrum with noise into sub-bands which are not repeated and do not interfere with each other randomly in the process of preprocessing an initial voice frequency spectrum; calculating and solving an over-subtraction factor of each sub-band through the frequency spectrum processing sub-module, independently executing spectral subtraction in each sub-band, obtaining the spectral estimation of the enhanced voice signal of each sub-band by using a frequency spectrum subtraction method, recombining the processed sub-bands, obtaining a complete enhanced acoustic phase spectrum by combining an acoustic phase spectrum with noise, and performing inverse Fourier transform to obtain the enhanced voice. The sensitivity and the precision of the voice are enhanced, and the high-quality voice is obtained.
It will be apparent to those skilled in the art that various changes and modifications may be made in the present invention without departing from the spirit and scope of the invention. Thus, if such modifications and variations of the present invention fall within the scope of the claims of the present invention and their equivalents, the present invention is also intended to include such modifications and variations.
Claims (10)
1. A noise reduction system for a wireless microphone, comprising: the wireless receiving unit receives voice and transmits the voice to the voice processing unit, the voice processing unit converts the voice into digital audio signals of a fixed frequency band and transmits the digital audio signals to the digital signal processing unit, the digital signal processing unit carries out filtering processing on the digital audio signals, carries out noise reduction processing through echo suppression and noise reduction algorithms and stores the processed digital audio signals in the data control unit, and the data control unit stores the digital audio signals and analyzes noise existing in the digital audio signals.
2. The noise reduction system of claim 1, wherein the speech processing unit comprises: the voice is converted into an electric signal through the piezoelectric converter, the electric signal is amplified through the audio amplifier, and the amplified electric signal is converted into a digital audio signal through the audio analog-to-digital converter.
3. The noise reduction system of claim 1, wherein the digital signal processing unit comprises: the device comprises a frequency domain filtering module, a self-adaptive beam module, a post-filtering module and an echo suppression module, wherein the frequency domain filtering module is used for executing a channel noise reduction algorithm on a digital audio signal through statistical parameters to primarily suppress noise; the self-adaptive beam module is used for removing noise from the digital audio signal to obtain a pure digital audio signal; the post-filter module is used for calculating a noise signal by using multiple channels and a filter algorithm to further suppress residual noise; the echo suppression module is used for carrying out echo change estimation on the digital audio signal and suppressing echo.
4. The noise reduction system of claim 1, wherein the data management unit comprises: the device comprises an audio quality sensing module, a noise estimation module and a memory module, wherein the audio quality sensing module is used for calculating the difference operation of a loudness spectrum, acquiring disturbance data and judging the audio quality in a digital audio signal; the noise estimation module is used for smoothing the band noise frequency power spectrum and estimating the noise in the digital audio signal by searching the smoothed band noise frequency power spectrum with the local minimum; the memory module is used for storing and deleting voice data.
5. The system of claim 3, wherein the adaptive beam module comprises: a fixed beam forming sub-module, a blocking matrix sub-module and a self-adaptive cancellation sub-module; and an upper branch is formed by the fixed beam forming submodule, a lower branch is formed by the blocking matrix submodule and the self-adaptive cancellation submodule, and a multi-channel self-adaptive filtering structure is formed by the fixed beam forming submodule, the blocking matrix submodule and the self-adaptive cancellation submodule.
6. The noise reduction system of claim 5, wherein the blocking matrix submodule is configured to block a desired audio signal of the digital audio signals; by adopting a method of constrained adaptive filtering and only updating the blocking matrix submodule when the expected audio exists, and only updating the coefficient of the adaptive cancellation submodule when the expected voice does not exist, the leakage of the expected voice signal is reduced; and a beam with directivity is formed by the self-adaptive beam module and points to the direction of a desired signal, so that the interference in an undesired direction is eliminated.
7. The noise reduction system of claim 3, wherein the echo suppression module comprises: the echo detection sub-module establishes an audio model of a far-end signal, estimates an echo and continuously modifies the coefficient of the low-pass filtering sub-module to enable an estimated value to approach to a real echo, and offsets an echo estimated value from an output signal of the wireless microphone, has an acoustic response function under an acoustic environment between a dynamic learning loudspeaker and the wireless microphone, and respectively processes the acoustic response function in a plurality of sub-frequency bands.
8. The noise reduction system of claim 4, wherein the audio quality perception module comprises an audio quality detection sub-module and a low-pass filtering sub-module, the audio quality detection sub-module adjusts an initial audio signal and a noise-reduced signal to a level with the same hearing intensity, the low-pass filtering sub-module simulates a frequency response of the signal passing through the microphone, the signals are aligned according to time, a response spectrum is obtained through hearing transformation processing, a difference of the response spectrum is calculated, and disturbance data is obtained, so that the audio quality is judged.
9. The noise reduction system of claim 4, wherein the audio quality perception module further comprises: the system comprises a voice positioning submodule and a signal processing submodule, wherein the voice positioning submodule constructs a spatial spectrum according to the orthogonality of a signal subspace and a noise subspace, and carries out average calculation on the spatial spectrums of all sub-bands to obtain a broadband signal spatial spectrum estimation; the signal processing submodule frames the signal, the time domain of the framed signal is converted into the frequency domain through Fourier transform, the signal of each channel under the same frequency is combined into a snapshot, the covariance matrix is decomposed by calculating the covariance matrix of each snapshot to obtain a signal subspace and a noise subspace, the spatial spectrum power is calculated by combining the array manifold, and the peak value is found to be the positioning result.
10. The noise reduction system of claim 2, wherein the audio amplifier comprises: the system comprises a voice signal enhancer module and a frequency spectrum processing submodule, wherein the voice signal enhancer module is used for windowing, framing and Fourier transform on original voice polluted by noise to obtain an initial voice frequency spectrum; cutting a sound phase spectrum with noise into sub-bands which are not repeated and do not interfere with each other randomly in the process of preprocessing an initial voice frequency spectrum; calculating and solving an over-subtraction factor of each sub-band through the frequency spectrum processing sub-module, independently executing spectral subtraction in each sub-band, obtaining the spectral estimation of the enhanced voice signal of each sub-band by using a frequency spectrum subtraction method, recombining the processed sub-bands, obtaining a complete enhanced acoustic phase spectrum by combining an acoustic phase spectrum with noise, and performing inverse Fourier transform to obtain the enhanced voice.
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