CN111586527A - Intelligent voice processing system - Google Patents
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
- H04R3/00—Circuits for transducers, loudspeakers or microphones
- H04R3/005—Circuits for transducers, loudspeakers or microphones for combining the signals of two or more microphones
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- G—PHYSICS
- G10—MUSICAL INSTRUMENTS; ACOUSTICS
- G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
- G10L21/00—Speech or voice signal processing techniques to produce another audible or non-audible signal, e.g. visual or tactile, in order to modify its quality or its intelligibility
- G10L21/02—Speech enhancement, e.g. noise reduction or echo cancellation
- G10L21/0208—Noise filtering
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- G—PHYSICS
- G10—MUSICAL INSTRUMENTS; ACOUSTICS
- G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
- G10L21/00—Speech or voice signal processing techniques to produce another audible or non-audible signal, e.g. visual or tactile, in order to modify its quality or its intelligibility
- G10L21/02—Speech enhancement, e.g. noise reduction or echo cancellation
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- G10L21/00—Speech or voice signal processing techniques to produce another audible or non-audible signal, e.g. visual or tactile, in order to modify its quality or its intelligibility
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Abstract
The invention relates to the technical field of voice processing, and discloses an intelligent voice processing system, which comprises: the voice noise reduction and mixing elimination device comprises a voice input module, a filtering module, an active noise reduction and mixing elimination module, a howling suppression module and a voice output module; the voice input module is used for converting voice signals collected by the microphone into digital signals and sending the digital signals to the filtering module; the filtering module is used for filtering the digital signals so as to reserve the digital signals with the frequency of 80 Hz-16 kHz; the active noise reduction and mixing elimination module is used for eliminating the ambient sound and the reverberation formed by the reflection of the indoor wall surface of the filtered digital signal; the voice output module is used for converting the processed digital signal into a voice signal and outputting the voice signal. The intelligent voice processing system avoids the problems of howling and noise and reverberation in the voice quality, and improves the voice quality.
Description
Technical Field
The invention relates to the technical field of voice processing, in particular to an intelligent voice processing system.
Background
In the conventional sound amplifying system structure as shown in fig. 1, a sound signal received by a microphone 100 is adjusted by an audio console 200 (e.g., to adjust volume, bass or treble), and then transmitted to a speaker 500 through an audio processor 300 and a power amplifier 400, and the speaker 500 converts the sound signal into voice for playing.
When the sound amplifying system is used in an indoor environment, an audio signal played by the loudspeaker 500 is picked up by the microphone 100 again, and is amplified and played by the power amplifier 400 to form sound wave positive feedback, when the system volume is large, howling is easy to generate, and particularly under the condition that the microphone 100 is used in a long distance, the howling is often caused. Moreover, because of the indoor building structure, material, volume, shape and the like, the reflection and absorption characteristics of the hall for various frequencies can be influenced, the frequency transmission characteristics in the hall are not uniform, and the tone quality of the audio signal has the defects of noise, distortion, overlong or overlong echo time and the like.
Disclosure of Invention
The invention provides an intelligent voice processing system, which solves the problems that howling is easy to generate when a public address system is used in an indoor environment, noise exists in tone quality, distortion exists, and echo time is too long or too short in the prior art.
The invention provides an intelligent voice processing system, comprising: the voice noise reduction and mixing elimination device comprises a voice input module, a filtering module, an active noise reduction and mixing elimination module, a howling suppression module and a voice output module;
the voice input module is connected with the microphone and the filtering module, and is used for converting voice signals collected by the microphone into digital signals and sending the digital signals to the filtering module;
the filtering module is connected with the active noise reduction and demixing module and is used for filtering the digital signal to reserve the digital signal with the frequency of 80 Hz-16 kHz and sending the filtered digital signal to the active noise reduction and demixing module;
the active noise reduction and mixing elimination module is connected with the howling suppression module and used for eliminating ambient sound and reverberation formed by indoor wall surface reflection of the filtered digital signal and sending the processed digital signal to the howling processing module;
the voice output module is used for outputting a voice signal to the voice output module according to the voice signal;
and the voice output module is used for converting the processed digital signal into a voice signal and outputting the voice signal.
The howling suppression module comprises:
the intelligent reflection spectrum screening module is used for improving the microphone sound amplification loudness in the digital signal by 8-12 dB before feedback;
the automatic noise suppression module is used for suppressing environmental noise in the digital signal;
the dodge device module is used for automatically reducing the input volume of the circuit when a digital signal corresponding to voice is input;
the matrix adjusting module is used for adjusting the routing of the multi-path input channel and the multi-path output channel;
the parameter equalization module is used for dividing the 80 Hz-16 kHz digital signal into a plurality of frequency segments and respectively carrying out different boosting or attenuation equalization processing on the signals of the frequency segments;
and the voltage limiter module is used for limiting the range of the maximum value and the minimum value of the volume level adjustment value in the digital signal.
The parameter equalizing module divides the digital signals of 80 Hz-16 kHz into 80 Hz-1000 Hz, 1001 Hz-2000 Hz, 2001 Hz-4000 Hz, 4001 Hz-8000 Hz and 8001 Hz-16000 Hz.
The device comprises a storage module, wherein acoustic compensation parameters of specialized sound recorders are preset in the storage module, and the intelligent reflection spectrum screening module, the automatic noise suppression module, the dodge device module, the matrix adjustment module, the parameter equalization module and the limiter module are used for reading corresponding acoustic compensation parameters from the storage module so as to realize the automatic processing of the digital signals.
The voice input module is used for collecting voice of at least one of a gooseneck microphone, an interface microphone, an embedded button microphone, a gun-type super-directional microphone and a ceiling microphone.
And the voice output module is used for sending the converted voice signal to the Dante network digital interface.
According to the intelligent voice processing system, before a voice signal enters a sound console, the digital signal is preprocessed through the filtering module, the active noise reduction and mixing elimination module and the howling suppression module in a digital signal mode, the main frequency range of speaking and sounding of a person is picked up, echo formed by ambient sound and indoor wall reflection is eliminated, howling is suppressed, the preprocessed digital signal is converted into the voice signal, and then the voice signal enters a traditional sound amplification system, so that the problems of noise, distortion and overlong or overlong echo time in the howling and tone quality are solved, and the voice quality is improved.
Drawings
In order to more clearly illustrate the embodiments of the present invention or the technical solutions in the prior art, the drawings used in the description of the embodiments or the prior art will be briefly described below, it is obvious that the drawings in the following description are only some embodiments of the present invention, and for those skilled in the art, other drawings can be obtained according to the drawings without creative efforts.
FIG. 1 is a schematic diagram of a prior art loudspeaker system;
FIG. 2 is a diagram illustrating an intelligent speech processing system according to an embodiment of the present invention;
FIG. 3 is a schematic diagram of the speaker system of the intelligent speech processing system of FIG. 2.
Detailed Description
The technical solutions in the embodiments of the present invention will be clearly and completely described below with reference to the drawings in the embodiments of the present invention, and it is obvious that the described embodiments are only a part of the embodiments of the present invention, and not all of the embodiments. All other embodiments, which can be derived by a person skilled in the art from the embodiments given herein without making any creative effort, shall fall within the protection scope of the present invention.
The intelligent speech processing system of the present embodiment is shown in fig. 2, and includes: the voice input module 1, the filtering module 2, the active noise reduction and mixing elimination module 3, the howling suppression module 4 and the voice output module 5.
The voice input module 1 is used for connecting the microphone and the filtering module 2, converting voice signals collected by the microphone into digital signals and sending the digital signals to the filtering module 2. The voice input module 1 can adopt voice signals through wired or 2.4G transmission, and is suitable for wired and 2.4G wireless transmission microphones.
The filtering module 2 is connected with the active noise reduction and demixing module 3, and is used for filtering the digital signal to reserve the digital signal with the frequency of 80 Hz-16 kHz, and sending the filtered digital signal to the active noise reduction and demixing module 3. Because the voice of the person speaking is mainly in the interval, even if the voice is higher than 16kHz, the voice cannot be collected by the microphone and cannot enter the subsequent signal processing process after the voice is processed. Meanwhile, after the filtering, the range of the processed signal can be reduced, and the high-frequency signal can be prevented from generating howling to a certain extent (the high-frequency signal is easier to cause the howling).
The active noise reduction and mixing elimination module 3 is connected with the howling suppression module 4, and is used for eliminating environmental sounds (such as mobile phone ring tones, fan sounds, noise transmitted from the outdoor and the like suddenly imagined in a conference room) and echoes formed by indoor wall reflection from the filtered digital signals, and sending the processed digital signals to the howling processing module 4.
The howling suppression module 4 is connected with the voice output module 5, and is used for suppressing howling according to the digital signal and sending the digital signal with the howling suppressed to the voice output module 5.
The voice output module 5 is used for converting the processed digital signal into a voice signal and outputting the voice signal.
As shown in fig. 3, the intelligent speech processing system 600 of the embodiment is connected between the microphone 100 and the sound mixing console 200, the sound mixing console 200 is connected with the audio processor 300, the audio processor 300 is connected with the power amplifier 400, and the power amplifier 400 is connected with the speaker 500. The voice output module 5 of the intelligent voice processing system 600 of the present embodiment transmits the preprocessed voice signal to the sound console 200. The intelligent voice processing system 600 of the embodiment converts the voice signals into digital signals before entering the sound console 200, preprocesses the digital signals through the filtering module 2, the active noise reduction and mixing elimination module 3 and the howling suppression module 4 respectively, picks up the main frequency range of the speaking voice of a person, eliminates the environmental sounds and echoes formed by indoor wall reflection, suppresses the howling, converts the preprocessed digital signals into the voice signals, and then enters the traditional sound amplification system, thereby avoiding the problems of noise, distortion and overlong or overlong echo time in the howling and tone quality, and improving the voice quality.
In this embodiment, the howling suppression module 4 includes:
and the intelligent reflection spectrum screening module 41 is used for improving the microphone sound amplification loudness in the digital signal by 8-12 dB before feedback.
And an automatic noise suppression module 42 for suppressing ambient noise in the digital signal.
And a ducker module 43 for automatically reducing the line input volume when a digital signal corresponding to voice is input.
And a matrix adjusting module 44 for adjusting the routing of the multiple input channels and the multiple output channels.
And the parameter equalization module 45 is configured to divide the 80 Hz-16 kHz digital signal into a plurality of frequency segments, and perform different boosting or attenuation equalization processing on the signals of the plurality of frequency segments.
And a limiter block 46 for limiting the range of maximum and minimum volume level adjustments in the digital signal.
The intelligent reflection spectrum screening module 41, the automatic noise suppression module 42, the dodge device module 43, the matrix adjustment module 44, the parameter equalization module 45 and the limiter module 46 can be adjusted according to the external input parameters of the user.
In this embodiment, the specific way of suppressing the environmental noise in the digital signal by the automatic noise suppression module 42 is as follows:
the data sampled by the ADC is divided into 1024 points, and the more points, the higher the resolution.
Each set is transformed by a Fast Fourier Transform (FFT) of 1024 points, and 1024 instantaneous energies, i.e., the modulus, are calculated for 1024 complex numbers in the frequency domain.
Comparing the instantaneous energy with a threshold value, and determining the sequence number of the howling point exceeding the threshold value;
mapping the howling point to a frequency value;
generating a phase shift function using a waveform table lookup method or cordic;
multiplying a frequency value and the phase shift function;
and performing IFFT (inverse fast Fourier transform) operation on the multiplication result to obtain a digital signal after compensating the howling point.
The howling point can be accurately detected and suppressed in the state of the digital signal by adopting the steps, so that the voice quality is improved.
In this embodiment, the parameter equalization module 45 divides the digital signal of 80Hz to 16kHz into 80Hz to 1000Hz, 1001Hz to 2000Hz, 2001Hz to 4000Hz, 4001Hz to 8000Hz, and 8001Hz to 16000Hz, and performs the adjustment in sections. The parametric equalization module 45 mainly functions to compensate or attenuate a loss portion (mainly, loss of frequency) in the audio signal, so that the spectrum in the audio frequency band is balanced, and the sectional adjustment is more favorable for balancing the spectrum in the audio frequency band.
The intelligent speech processing system of this embodiment further includes a storage module 6, wherein acoustic compensation parameters for teaching by professional sound recorders are preset in the storage module 6, and the intelligent reflection spectrum screening module 41, the automatic noise suppression module 42, the dodging device module 43, the matrix adjustment module 44, the parameter equalization module 45, and the limiter module 46 are configured to read corresponding acoustic compensation parameters from the storage module 6, so as to implement automatic processing of digital signals. Acoustic compensation parameters for teaching of professional sound recorders are preset, so that non-sound engineering and recording professionals can conveniently finish installation and debugging.
In this embodiment, the voice input module 1 is configured to collect voice of at least one of a gooseneck microphone, an interface microphone, an embedded button microphone, a gun-type super-directional microphone, and a ceiling microphone, and may collect voice signals by multiple microphones simultaneously or in a time-sharing manner.
The voice output module 5 is configured to send the converted voice signal to the Dante network digital interface, and the Dante network digital interface can output the converted voice signal to a plurality of speakers simultaneously.
The present invention is not limited to the above preferred embodiments, and any modifications, equivalent replacements, improvements, etc. within the spirit and principle of the present invention should be included in the protection scope of the present invention.
Claims (6)
1. An intelligent speech processing system, comprising: the voice noise reduction and mixing elimination device comprises a voice input module, a filtering module, an active noise reduction and mixing elimination module, a howling suppression module and a voice output module;
the voice input module is connected with the microphone and the filtering module, and is used for converting voice signals collected by the microphone into digital signals and sending the digital signals to the filtering module;
the filtering module is connected with the active noise reduction and demixing module and is used for filtering the digital signal to reserve the digital signal with the frequency of 80 Hz-16 kHz and sending the filtered digital signal to the active noise reduction and demixing module;
the active noise reduction and mixing elimination module is connected with the howling suppression module and used for eliminating environmental sounds and echoes formed by indoor wall reflection of the filtered digital signals and sending the processed digital signals to the howling processing module;
the voice output module is used for outputting a voice signal to the voice output module according to the voice signal;
and the voice output module is used for converting the processed digital signal into a voice signal and outputting the voice signal.
2. The intelligent speech processing system of claim 1 wherein the howling suppression module comprises:
the intelligent reflection spectrum screening module is used for improving the microphone sound amplification loudness in the digital signal by 8-12 dB before feedback;
the automatic noise suppression module is used for suppressing environmental noise in the digital signal;
the dodge device module is used for automatically reducing the input volume of the circuit when a digital signal corresponding to voice is input;
the matrix adjusting module is used for adjusting the routing of the multi-path input channel and the multi-path output channel;
the parameter equalization module is used for dividing the 80 Hz-16 kHz digital signal into a plurality of frequency segments and respectively carrying out different boosting or attenuation equalization processing on the signals of the frequency segments;
and the voltage limiter module is used for limiting the range of the maximum value and the minimum value of the volume level adjustment value in the digital signal.
3. The intelligent speech processing system of claim 2 wherein the parametric equalization module divides the digital signal between 80Hz and 16kHz into 80Hz to 1000Hz, 1001Hz to 2000Hz, 2001Hz to 4000Hz, 4001Hz to 8000Hz, 8001Hz to 16000 Hz.
4. The intelligent speech processing system according to claim 2, further comprising a storage module, wherein the storage module is preset with acoustic compensation parameters for specialized audio recorders, and the intelligent reflection spectrum screening module, the automatic noise suppression module, the ducker module, the matrix adjustment module, the parametric equalization module, and the limiter module are configured to read corresponding acoustic compensation parameters from the storage module to automatically process the digital signals.
5. The intelligent speech processing system of claim 1 wherein the speech input module is configured to capture speech from at least one of a gooseneck microphone, an interface microphone, an embedded button microphone, a gun-type hyper-directional microphone, and a ceiling-mounted microphone.
6. The intelligent speech processing system according to any one of claims 1-5, wherein the speech output module is configured to send the converted speech signal to a Dante network digital interface.
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Cited By (3)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| CN113891217A (en) * | 2021-11-08 | 2022-01-04 | 易兆微电子(杭州)股份有限公司 | Howling suppression method, howling suppression device, electronic equipment and storage medium |
| CN114449428A (en) * | 2022-01-10 | 2022-05-06 | 惠州Tcl移动通信有限公司 | Howling prevention method, howling prevention device, storage medium and hearing aid |
| CN114696920A (en) * | 2022-04-15 | 2022-07-01 | 深圳市湖山科技有限公司 | On-site remote free sound pickup and amplification device and method |
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