CN112397085A - System and method for processing voice and information - Google Patents

System and method for processing voice and information Download PDF

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Publication number
CN112397085A
CN112397085A CN201910760510.1A CN201910760510A CN112397085A CN 112397085 A CN112397085 A CN 112397085A CN 201910760510 A CN201910760510 A CN 201910760510A CN 112397085 A CN112397085 A CN 112397085A
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signal
voice
sub
audio signal
value
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CN112397085B (en
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蓝柏澍
郭俊宏
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C Media Electronics Inc
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C Media Electronics Inc
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    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS OR SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING; SPEECH OR AUDIO CODING OR DECODING
    • G10L25/00—Speech or voice analysis techniques not restricted to a single one of groups G10L15/00 - G10L21/00
    • G10L25/03—Speech or voice analysis techniques not restricted to a single one of groups G10L15/00 - G10L21/00 characterised by the type of extracted parameters
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS OR SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING; SPEECH OR AUDIO CODING OR DECODING
    • G10L21/00—Processing of the speech or voice signal to produce another audible or non-audible signal, e.g. visual or tactile, in order to modify its quality or its intelligibility
    • G10L21/02—Speech enhancement, e.g. noise reduction or echo cancellation
    • G10L21/0316—Speech enhancement, e.g. noise reduction or echo cancellation by changing the amplitude
    • G10L21/0324—Details of processing therefor
    • G10L21/034—Automatic adjustment
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS OR SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING; SPEECH OR AUDIO CODING OR DECODING
    • G10L25/00—Speech or voice analysis techniques not restricted to a single one of groups G10L15/00 - G10L21/00
    • G10L25/03—Speech or voice analysis techniques not restricted to a single one of groups G10L15/00 - G10L21/00 characterised by the type of extracted parameters
    • G10L25/21—Speech or voice analysis techniques not restricted to a single one of groups G10L15/00 - G10L21/00 characterised by the type of extracted parameters the extracted parameters being power information
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS OR SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING; SPEECH OR AUDIO CODING OR DECODING
    • G10L25/00—Speech or voice analysis techniques not restricted to a single one of groups G10L15/00 - G10L21/00
    • G10L25/78—Detection of presence or absence of voice signals
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS OR SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING; SPEECH OR AUDIO CODING OR DECODING
    • G10L25/00—Speech or voice analysis techniques not restricted to a single one of groups G10L15/00 - G10L21/00
    • G10L25/78—Detection of presence or absence of voice signals
    • G10L2025/783—Detection of presence or absence of voice signals based on threshold decision

Abstract

The invention discloses a voice and message processing system and a method. The system comprises a voice and information receiver and a voice and information processing device. The audio-visual receiver comprises an audio-visual interface, an absolute value conversion unit, a sound framing unit and a characteristic value detection unit. The audio-visual interface receives the audio-visual signals of the audio-visual transceiver. The absolute value conversion unit takes the absolute value of the voice signal to output an absolute value voice signal. The sound framing unit divides the absolute value sound signal into a plurality of sub sound frame signals. The feature value detection unit detects a feature value of the sub-frame signal. The voice and information processing device processes the voice and information signals output by the voice and information receiving and transmitting device according to the characteristic value of the sub-frame signals.

Description

System and method for processing voice and information
Technical Field
The present invention relates to a system and a method for processing audio signals, and more particularly, to a system and a method for processing audio signals to improve the operation efficiency of an audio signal processing apparatus.
Background
The common audio and signal processing system acquires audio and signal signals or environmental Noise through an audio and signal transceiver such as a sensor or a microphone, and then executes special algorithm processing through a hardware or Digital Signal Processor (DSP) to realize Automatic Gain processing (Automatic Gain Control), Noise suppression (Noise Gate) and intelligent Volume Control (Smart Volume Control); or monitor the recording/playing audio-visual signal, dynamically adjust the light-emitting device according to the audio-visual content change, so as to achieve the optimal acousto-optic effect. However, the current algorithm processing, which uses dedicated hardware or digital signal processor to implement each algorithm, does not have good processing efficiency and cost effectiveness.
Disclosure of Invention
The present invention is directed to a system for processing audio signals, which is suitable for one or more audio signal transceiver devices, and includes one or more audio signal receivers and an audio signal processing device. The audio-visual receiver comprises an audio-visual interface, an absolute value conversion unit, a sound framing unit and a characteristic value detection unit. The voice interface is connected with one or more voice transceiving devices and configured to receive the voice signals output by the voice transceiving devices. The absolute value conversion unit is connected with the voice and information interface and configured to take an absolute value of the voice and information signal so as to output an absolute value voice and information signal. The sound framing unit is connected with the absolute value conversion unit and configured to divide the absolute value sound signal into a plurality of sub sound frame signals. The feature value detection unit is connected with the voice framing unit and configured to detect one or more feature values of the sub-voice frame signals. The voice and information processing device is connected with the voice and information receiver and the voice and information receiving and transmitting device. The voice signal processing device is configured to output a voice signal processing signal to process the voice signal according to one or more characteristic values of the sub-frame signal of the voice signal transceiver.
Preferably, the feature value detection unit includes a peak value detection unit connected to the sound framing unit and configured to detect a peak value of the sub-sound frame signal, the feature value including the peak value.
Preferably, the feature value detection unit includes an average value calculation unit connected to the sound framing unit and configured to calculate an average value of the sub-sound frame signals, the feature value including the average value.
Preferably, the feature value detecting unit includes a pulse packet filtering unit connected to the sound framing unit and configured to filter the sub-sound frame signal.
Preferably, the sound framing unit divides the absolute value sound signal into a plurality of sub sound frame signals having variable lengths.
Preferably, the audio signal processing system further comprises a light emitting driving module connected to the audio signal processing device and the one or more light emitting modules, and the audio signal processing device is configured to output a light emitting control signal according to the one or more characteristic values to control the light emitting driving module to drive the light emitting module matched with the audio signal transceiver to output a corresponding light emitting signal.
Preferably, the light-emitting driving module comprises a pulse width modulation signal generating unit connected to the one or more light-emitting modules and configured to output a pulse width modulation signal according to the one or more characteristic values to control the light-emitting module matched with the audio signal transceiver to output a corresponding light-emitting signal; wherein the lighting control signal comprises a pulse width modulated signal.
Preferably, when the voice processing device determines that the feature value of the sub-frame signal is smaller than a threshold value and the time during which the feature value is maintained smaller than the threshold value is longer than the hold time, the voice processing device outputs the voice processing signal to turn down the feature value of the voice signal output by the voice signal transceiver to the mute threshold value within the release time.
Preferably, when the voice processing device judges that the feature value of the sub frame signal is greater than the threshold value, the voice processing device outputs the voice processing signal to increase the feature value of the voice signal output by the voice signal transmitting/receiving device within the starting time.
Preferably, the voice processing system further comprises a digital volume controller, and the voice processing apparatus adjusts a gain value of the digital volume controller to eliminate noise of the voice signal when determining that the voice signal has noise according to one or more characteristic values of the sub-frame signal.
In addition, the present invention further provides a voice processing method, which is suitable for one or more voice transceiver devices, and comprises the following steps: receiving the voice and information signal output by the voice and information transceiver by using the voice and information interface of the voice and information receiver; the absolute value conversion unit of the voice and information receiver is utilized to take the absolute value of the voice and information signal so as to output the voice and information signal with the absolute value; dividing the absolute value voice signal into a plurality of sub-voice frame signals by using a voice framing unit of the voice receiver; detecting one or more characteristic values of the sub-frame signal by using a characteristic value detection unit of the voice signal receiver; and outputting the audio signal processing signal to process the audio signal by using the audio signal processing device according to one or more characteristic values of the sub-frame signal.
Preferably, the step of detecting one or more feature values of the sub-frame signal by using the feature value detection unit of the audio signal receiver comprises: a peak value of the sub-frame signal is detected by a peak value detecting unit of the characteristic value detecting unit.
Preferably, the step of detecting one or more feature values of the sub-frame signal by using the feature value detection unit of the audio signal receiver comprises: an average value of the sub-frame signal is detected by an average value calculating unit of the characteristic value detecting unit.
Preferably, the step of detecting one or more feature values of the sub-frame signal by using the feature value detection unit of the audio signal receiver comprises: the sub-frame signal is filtered by a pulse packet filtering unit of the eigenvalue detection unit.
Preferably, the step of dividing into a plurality of sub-frame signals by the frame unit comprises: the speech framing unit divides the absolute value speech signal into a plurality of sub-speech signals having variable lengths.
Preferably, the method for processing the voice message further comprises the following steps: and outputting a luminous control signal by using the voice signal processing device according to one or more characteristic values of the sub-voice frame signal of the voice signal receiving and transmitting device so as to control a luminous driving module to drive a luminous module to output a corresponding luminous signal.
Preferably, the method for processing the voice message further comprises the following steps: and outputting a pulse width modulation signal by a pulse width modulation signal generating unit according to one or more characteristic values to control a light-emitting module to output a corresponding light-emitting signal.
Preferably, the method for processing the voice message further comprises the following steps: when the voice and signal processing device is used for judging that the characteristic value of the sub-voice frame signal is smaller than a threshold value and the time for maintaining the characteristic value smaller than the threshold value is longer than a holding time, the voice and signal processing device outputs the voice and signal processing signal so as to adjust the characteristic value of the voice and signal output by the voice and signal receiving and transmitting device to a mute threshold value within a release time.
Preferably, the method for processing the voice message further comprises the following steps: when the characteristic value of the sub-frame signal is judged to be larger than a threshold value by using the voice signal processing device, the voice signal processing signal is output so as to increase the characteristic value of the voice signal output by the voice signal receiving and transmitting device within the starting time.
Preferably, the method for processing the voice message further comprises the following steps: when the voice signal is judged to have noise by utilizing the voice signal processing device according to one or more characteristic values of the sub-frame signal, the gain value of a digital volume controller is adjusted to eliminate the noise of the voice signal.
As described above, the present invention provides a voice processing system and method, which can receive a voice signal from any signal source, and provide a control parameter for controlling a voice transceiver by a voice processing device by configuring a voice receiver, taking an absolute value of the voice signal, dividing the voice signal into a plurality of sub-frame signals, detecting a feature value of the sub-frame signals, and so on, thereby effectively improving the performance of the voice processing device by simplifying the processing of a single circuit.
For a better understanding of the features and technical content of the present invention, reference should be made to the following detailed description of the invention and accompanying drawings, which are provided for purposes of illustration and description only and are not intended to limit the invention.
Drawings
Fig. 1 is a block diagram of a voice receiver of a voice processing system according to an embodiment of the present invention.
Fig. 2 is a schematic configuration diagram of the audio signal processing system, the audio signal transceiver, the light-emitting module, the light-emitting driving module, the analog-to-digital converter, the audio signal processing device, the digital-to-analog converter, and the peripheral device according to the embodiment of the present invention.
Fig. 3 is a flowchart of the steps of extracting the absolute value of the audio signal, dividing the audio signal, detecting the feature value of the sub-frame signal, and processing the audio signal according to the audio signal processing method of the embodiment of the present invention.
Fig. 4 is a flowchart illustrating steps of adjusting the light emitting state of the light emitting module according to the feature value of the frame signal in the audio signal processing method according to the embodiment of the invention.
Fig. 5 is a flowchart of noise suppression (noise gate) steps of the audio signal processing method according to the embodiment of the present invention.
FIG. 6 is a waveform diagram of the sub-frame filtering signal, the average value signal, the peak value signal, the absolute value audio signal, the sub-frame signal and the audio signal provided to the audio processing apparatus according to the sub-frame length.
Fig. 7 is a waveform diagram of the average signal of the audio signal processing device provided by the audio signal processing system and method for implementing dynamic LED lighting control according to the embodiment of the present invention.
Fig. 8 is a waveform diagram of a pulse width modulation signal for implementing dynamic LED lighting control and finally controlling LED brightness by the audio signal processing system and method according to the embodiment of the present invention.
Fig. 9 is a waveform diagram of an output signal of a sound signal processing system and method for implementing noise suppression according to an embodiment of the present invention.
Fig. 10 is a waveform diagram of an output signal of an audio signal processing system and method for implementing Automatic Gain Control (AGC) according to an embodiment of the present invention.
Detailed Description
The following is a description of embodiments of the present invention with reference to specific embodiments, and those skilled in the art will understand the advantages and effects of the present invention from the disclosure of the present specification. The invention is capable of other and different embodiments and its several details are capable of modifications and various obvious aspects, all without departing from the scope of the invention. The drawings of the present invention are for illustrative purposes only and are not intended to be drawn to scale. The following embodiments will further explain the related art of the present invention in detail, but the disclosure is not intended to limit the scope of the present invention.
It will be understood that, although the terms "first," "second," "third," etc. may be used herein to describe components or signals, these components or signals should not be limited by these terms. These terms are used primarily to distinguish one element from another element or from one signal to another signal. In addition, the term "or" as used herein should be taken to include any one or combination of more of the associated listed items as the case may be.
Referring to fig. 1, fig. 1 is a block diagram of a voice receiver of a voice processing system according to an embodiment of the present invention; fig. 2 is a schematic configuration diagram of a voice signal processing system, a voice signal transceiver, a light-emitting module, a light-emitting driving module, an analog-to-digital converter, a voice signal processing device, a digital-to-analog converter and peripheral devices according to an embodiment of the present invention; fig. 6 is a waveform diagram of generating a sub-frame filtering signal, an average value signal, a peak value signal, an absolute value audio signal, a sub-frame signal and an audio signal provided to an audio processing apparatus according to a sub-frame length set by the audio processing system and method according to the embodiment of the invention.
It should be noted that, as shown in fig. 1, the audio signal processing system of the embodiment of the present invention mainly includes an audio signal receiver SmartRx. The credit receivers SmartRx may comprise credit receivers SmartRx1 and SmartRx2 as shown in fig. 2. For convenience of illustration, the two receivers SmartRx1 and SmartRx2 of the present embodiment are used to detect and process the audio signals from different sources, respectively. It will be understood that the invention is not limited to the necessity of two SmartRx1, SmartRx2 for performing these operations, but may in practice be replaced by a single or any number of receivers performing the operations of detecting and processing the audio signals from different sources.
Each of the credit receivers SmartRx1, SmartRx2 shown in fig. 2 may include the credit interface 111, the absolute value conversion unit 12, the sound framing unit 13, and the feature value detection unit 10 shown in fig. 1.
In the embodiment, the eigenvalue detection unit 10 may include a pulse packet filtering unit 14, a peak value detection unit 15 and an average value calculation unit 16 configured to filter the audio signal and detect the peak value and the average value of the audio signal, respectively, but the invention is not limited thereto, and other eigenvalues of the audio signal may be actually calculated.
The audio interface 111 is connected to the absolute value conversion unit 12. The absolute value conversion unit 12 is connected to the sound framing unit 13. The sound framing unit 13 is connected to the feature value detection unit 10.
As shown in fig. 2, the audio transceiving apparatus may include a microphone m1, a microphone m2, a speaker L1, a speaker L2, or other apparatuses capable of playing or providing audio signals, which is only exemplary and not intended to limit the present invention.
As shown in fig. 1 and 2, an analog-to-digital converter ADCL may be provided between the voice interface 111 of the voice receiver SmartRx1 and the programmable gain amplifier PGA of the microphone m 1. The programmable gain amplifier PGA of the microphone m1 multiplies the parameter value of the original audio signal of the microphone m1 by a programmable gain, and outputs the amplified audio signal to the analog-to-digital converter ADCL.
When the analog-to-digital converter ADCL receives the analog audio signal from the programmable gain amplifier PGA of the microphone m1, the analog-to-digital converter ADCL may convert the analog audio signal into a digital audio signal and output the digital audio signal to the audio receiver SmartRx 1.
Similarly, an analog-to-digital converter ADCR may be provided between the voice interface 111 of the voice receiver SmartRx1 and the programmable gain amplifier PGA of the microphone m 2. The programmable gain amplifier PGA of the microphone m2 multiplies the parameter value of the original audio signal of the microphone m2 by a programmable gain, and outputs the amplified audio signal to the analog-to-digital converter ADCR.
When the analog-to-digital converter ADCR receives an analog audio signal from the programmable gain amplifier PGA of the microphone m2, the analog-to-digital converter ADCR converts the analog audio signal into a digital audio signal and outputs the digital audio signal to the audio signal receiver SmartRx 1.
However, if the format of the digital audio signal is different from the format of the signal processed by the audio receiver SmartRx1, the audio interface 111 may convert the signal format first. Further, as shown in fig. 1, the sound signal interface 111 of the sound signal receiver SmartRx1 supplies the converted digital sound signal of the signal format to the absolute value conversion unit 12. The absolute value conversion unit 12 is configured to take absolute values of the audio signal to output an absolute value audio signal respectively. As shown in fig. 6, the audio signal AU before the absolute value is taken has positive and negative half waves, while the audio signal ABU of an absolute value generated by taking the absolute value of the audio signal AU has only positive half waves.
The sound framing unit 13 is configured to divide the absolute value sound signal ABU output from the absolute value conversion unit 12 into a plurality of sub sound frame signals FR as shown in fig. 6. The number and length of the sub-frame signals FR can be individually adjusted according to actual requirements, for example, the lengths of the sub-frame signals FR can be the same or different from each other.
After the division of the plurality of sub frame signals FR, the plurality of sub frame signals FR are processed or detected by the pulse packet filtering unit 14, the peak detecting unit 15, and the average value calculating unit 16 of the tone signal receiver SmartRx 1.
For example, the pulse packet filtering unit 14 may filter each sub-frame signal FR according to the filtering parameter matrix h (z) shown in fig. 1 and fig. 6 to output a sub-frame filtered signal. The peak detecting unit 15 may detect a peak of each waveform of the sub frame signal FR, such as the peak signal Vm shown in fig. 6. The average value calculating unit 16 may detect an average value of each waveform of the sub frame signal FR, such as an average value signal Vave shown in fig. 6.
Further, as shown in fig. 1, the tone receiver SmartRx1 may output the sub-tone frame signal FR, the sub-tone frame filtered signal, and the detected and calculated peak and average values to the tone processing device 20.
As shown in fig. 2, the digital volume controller ST1 is connected to the audio signal processing device 20, the analog-to-digital converter ADCL, and the peripheral device 30. The audio signal processing device 20 obtains the feature values of the plurality of sub frame signals FR detected by the audio signal receiver SmartRx1, and determines whether or not there is noise in the audio signal AU received from the analog-to-digital converter ADCL based on the feature values. If there is noise in the audio signal AU, the gain of the digital volume controller ST1 is adjusted to remove noise, and then the audio signal AU is provided to the peripheral device 30.
Similarly, the digital volume controller ST2 is connected to the audio signal processing device 20, the analog-to-digital converter ADCR, and the peripheral device 30. The audio signal processing device 20 obtains the feature values of the plurality of sub frame signals FR detected by the audio signal receiver SmartRx1, and determines whether or not there is noise in the audio signal AU received from the analog-to-digital converter ADCR based on the feature values. If there is noise in the audio signal AU, the gain of the digital volume controller ST2 is adjusted to remove noise, and then the audio signal AU is provided to the peripheral device 30.
For example, the audio signal processing device 20 may be a microcontroller (mcu) or a Digital Signal Processor (DSP); the peripheral device 30 may comprise a device having a USB, I2S, which is only exemplary and not limiting.
The voice processing device 20 can be connected with a voice receiver SmartRx1 and voice transceiving devices such as microphones m1 and m 2. The audio signal processing device 20 is configured to output an audio signal processing signal to process the audio signal AU output by the audio signal transceiver according to the characteristic value.
For example, the audio signal processing device 20 can directly adjust or instruct the programmable gain amplifier PGA of the audio signal transceiver to select or set the waveform parameters such as the peak value, amplitude, average value, frequency or duty cycle of the audio signal outputted by the audio signal processing device 20.
The light-emitting drive module LTD1 is connected to the audio/video processing device 20 and the light-emitting module LED 1. The voice signal processing device 20 may be configured to control the light-emitting driving module LTD1 to drive the light-emitting module LED1 to output a light-emitting signal corresponding to the characteristic value of the sub-frame signal of the microphone m 1.
Similarly, the smart rx2 may detect the feature value of the sub-frame signal FR provided by the peripheral device 30, and the audio processing device 20 may be configured to control the lighting driving module LTD2 to drive the lighting module LED2 to output a lighting signal corresponding to the feature value of the sub-frame signal of the microphone m 2.
The light driving modules LTD1 and LTD2 may drive the light modules LED1 and LED2 in various manners. For example, the light driving modules LTD1 and LTD2 may include pulse width modulation signal generating units respectively connected to the light modules LED1 and LED 2. The PWM signal generating unit is configured to output a PWM signal according to the feature values of the sub-frame signals of the microphones m1 and m2, so as to control the light emitting modules LED1 and LED2 respectively matched with the microphones m1 and m2 to output corresponding light emitting signals.
It should be understood that the parameters of the waveform, the frequency and the period of the continuous light or flash, and the intensity and the color of the light, etc. of the light emitting signals output by the light emitting modules LED1 and LED2 may vary with the characteristic value of the sub-sound frame signal FR.
The peripheral device 30 may provide the audio signal to the digital volume controllers ST3, ST 4. The audio signal processing device 20 obtains the feature values of the plurality of sub frame signals FR detected by the audio signal receiver SmartRx2, and determines whether or not there is noise in the audio signal AU received from the peripheral device 30 based on the feature values. If there is noise in the audio signal AU, the gain values of the digital volume controllers ST3, ST4 are adjusted to eliminate noise, and then the audio signal AU is supplied to the digital-to-analog converters DACL, DACR respectively after the noise is eliminated.
If the signal format of the audio signal indicated by the audio signal processing signal does not match the format of the audio signal playable by the audio signal transceiver L1 or L2, the digital audio signal is converted into an analog audio signal by a digital-to-analog converter DACL or DACR, and the analog audio signal is outputted to the speaker L1 or L2 for playing.
Please refer to fig. 3, which is a flowchart illustrating steps of extracting an absolute value of a voice signal, dividing the voice signal, detecting a feature value of a sub-frame signal, and processing the voice signal according to the voice processing method of the embodiment of the present invention. As shown in fig. 3, the audio signal processing method according to the embodiment of the present invention includes the following steps S301 to S311, and is applied to the audio signal processing system.
For convenience of explanation, the present embodiment will be described below by way of example with two audio signal receivers SmartRx1 and SmartRx2 respectively performing operations of detecting and processing audio signal from different sources. It will be understood that the invention is not limited to the necessity of two SmartRx1, SmartRx2 for performing these operations, but may in practice be replaced by a single or any number of receivers performing the operations of detecting and processing the audio signals from different sources.
In step S301, the audio/video transceiver plays audio/video signals through the microphones m1 and m2 shown in fig. 2.
In step S303, the audio signal output from the microphone m1 or m2 is acquired by the audio interface 111 of the audio receiver SmartRx 1. Further, the audio signal is acquired from the peripheral device 30 by the audio interface 111 of the audio receiver SmartRx 2.
In step S305, the absolute value conversion unit 12 of the tone signal receivers SmartRx1, SmartRx2 takes the absolute value of the obtained tone signal to output an absolute value tone signal.
In step S307, the absolute value voice signal is divided into a plurality of sub-frame signals by the voice framing units 13 of the voice signal receivers SmartRx1, SmartRx 2.
In step S309, the feature value of the sub frame signal is detected or calculated by the feature value detection unit 10 of the tone signal receivers SmartRx1, SmartRx 2. For example, all sub-frame signals are filtered by the pulse packet filtering unit 14 of the tone signal receivers SmartRx1, SmartRx 2. For another example, the peak detection means 15 of the audio signal receivers SmartRx1 and SmartRx2 detects the peak of the sub frame signal. For another example, the average value of the sub frame signals is calculated by the average value calculating unit 16 of the sound signal receivers SmartRx1 and SmartRx 2.
In step S311, the audio signal processing device 20 outputs an audio signal processing signal according to the feature value of the sub-frame signal to process the sound source signal output by the audio signal transceiver.
Please refer to fig. 4, which is a flowchart illustrating a step of adjusting a lighting state of the lighting module according to a feature value of the frame signal in the audio signal processing method according to an embodiment of the present invention. As shown in fig. 4, the audio signal processing method according to the embodiment of the present invention includes the following steps S401 to S409, and is applied to the audio signal processing system.
For convenience of explanation, the present embodiment will be described below by way of example with two audio signal receivers SmartRx1 and SmartRx2 respectively performing operations of detecting and processing audio signal from different sources. It will be understood that the invention is not limited to the necessity of two SmartRx1, SmartRx2 for performing these operations, but may in practice be replaced by a single or any number of receivers performing the operations of detecting and processing the audio signals from different sources.
In step S401, the characteristic values of the audio signal AU of the microphones m1 and m2 are acquired by the audio receiver SmartRx1, and the characteristic values of the audio signal AU supplied from the peripheral device 30 are acquired by the audio receiver SmartRx 2.
In step S403, the audio signal processing apparatus 20 determines whether the feature value of the sub-audio frame signal is greater than a threshold value, such as whether the average value of the sub-audio frame signal is greater than an average threshold value, whether the peak value is greater than a peak threshold value, whether the amplitude is greater than an amplitude threshold value, whether the frequency is greater than a frequency threshold value, whether the period is greater than a period threshold value, whether the length of the sampled sub-audio frame signal is greater than a frame length threshold value, and so on.
In step S405, the audio signal processing apparatus 20 may determine the parameter value of the light-emitting signal according to the determination result of the feature value and the threshold value of the sub-frame signal. For example, the larger the peak value of the sub-sound frame signal (the larger the sound volume), the larger the peak value of the light emitting signal output by the light emitting module (the stronger the brightness), which is only exemplified herein, and is not limited thereto.
In step S407, a threshold value corresponding to full silence or low volume may be set. For example, when any one or more characteristic peaks of the sub-frame signal of the audio signal transceiver is not greater than the threshold, the audio signal processing device 20 determines that the volume output by the audio signal transceiver is too low (possibly lower than the volume audible to human ears), and may control the light-emitting driving module LTD1 to turn off the light-emitting module LED1, and stop outputting the light-emitting signal to alert the user of the volume level.
In step S409, when any one or more of the characteristic values of the sub-frame signals of the audio signal transceiver is greater than the threshold value, the parameter value of the light-emitting signal may be determined according to the characteristic value of the sub-frame signal, and the light-emitting signal output by the light-emitting module LED1 may be adjusted in real time.
Referring to fig. 5, fig. 9 and fig. 10, fig. 5 is a flowchart illustrating a noise gate (noise gate) step of a voice signal processing method according to an embodiment of the present invention; fig. 9 is a waveform diagram of an output signal of a sound signal processing system and method for implementing noise suppression according to an embodiment of the present invention; fig. 10 is a waveform diagram of an output signal of a voice signal processing system and method for implementing Automatic Gain Control (AGC) according to an embodiment of the present invention. As shown in fig. 5, the audio/video processing method according to the embodiment of the present invention includes the following steps S501 to S525.
Step S501: the characteristic value of the sub-frame signal is detected by the voice receiver.
Step S503: the first mode is entered.
Step S505: the characteristic value of the sub-frame signal detected or calculated by the voice receiver, such as the volume peak value of the sub-frame signal detected by the peak value detecting unit or the volume average value of the sub-frame signal calculated by the average value calculating unit, is obtained by the voice processing device. And judging whether the volume of the sub-sound frame signal is smaller than a volume threshold value or not by using the sound signal processing device according to the characteristic value of the sub-sound frame signal. If not, the original feature value of the sub-frame signal is maintained, for example, the original volume is maintained, and the light emitting module is controlled to maintain the original light emitting state, and step S525 is performed. If yes, go to step S507.
Step S507: whether the time for which the volume of the sub-frame signal is maintained less than the volume threshold value is longer than the holding time is judged by the audio signal processing device, and the holding time is shown in fig. 10. If not, the original feature value of the sub-frame signal is maintained, for example, the original volume is maintained, and the light emitting module is controlled to maintain the original light emitting state, and step S525 is performed. If yes, go to step S509.
Step S509: a second mode is entered.
Step S511: the voice processing device controls the voice receiving and sending device to decrease the volume of the voice signal output within the release time (release time).
Step S513: the characteristic value of the sub-frame signal output by the voice signal receiver is used by the voice signal processing device to judge whether the characteristic value of the sub-frame signal reaches a mute threshold value, such as 0 dB. If not, the original feature value of the sub-frame signal is maintained, for example, the original volume is maintained, and the light emitting module is controlled to maintain the original light emitting state, and step S525 is performed. If yes, the audio signal AU shown in fig. 9 enters the mute level from the first trigger point, and step S515 is executed.
Step S515: a third mode is entered.
Step S517: and judging whether the characteristic value of the sub-sound frame signal reaches the threshold value by using the sound signal processing device according to the characteristic value of the sub-sound frame signal output by the sound signal receiver. If not, the original volume of the sub-frame signal is maintained, and the light-emitting module is controlled to maintain the original light-emitting state, and step S525 is performed. If yes, go to step S519.
Step S519: the fourth mode is entered.
Step S521: starting from the second trigger point as shown in fig. 9, the gain of the zero-crossing point is updated, and the sound volume of the audio signal AU returns to the original level. Then, as shown in fig. 10, the audio signal transmitting/receiving apparatus increases the volume of the audio signal AU to be output within the attack time (attack time).
Step S523: the voice processing device is used for judging the change amplitude and the volume descending trend according to the characteristic value of the sub-frame signal output by the voice receiver, such as the peak value, the average value or the slope of the waveform, so as to judge whether the characteristic value of the sub-frame signal returns to 0 dB. If not, the original volume of the sub-frame signal is maintained, and the light-emitting module is controlled to maintain the original light-emitting state, and step S525 is performed. If yes, go back to step S503.
Step S525: waiting for the characteristic value of the audio signal to be updated.
Referring to fig. 7 and 8 together, fig. 7 is a waveform diagram of the audio signal processing system and method for implementing dynamic LED lighting control to provide an average signal of the audio signal processing device according to the embodiment of the present invention; fig. 8 is a waveform diagram of the audio signal processing system and method according to the embodiment of the present invention for implementing dynamic LED lighting control and finally controlling LED brightness.
The audio signal processing system and method of the embodiment of the present invention generates the average value signal AUP shown in fig. 7 using the audio signal receiver SmartRx shown in fig. 1. The audio signal receiver SmartRx as shown in fig. 2 comprises an audio signal receiver SmartRx1 configured to detect an average value of each waveform of a sub frame signal divided from an audio signal output from an audio signal transceiving means such as the microphone m1 shown in fig. 2 to generate an average value signal AUP as shown in fig. 7.
The audio signal receiver SmartRx1 outputs the average value signal AUP of the microphone m1 to the audio signal processing device 20. The audio signal processing device 20 outputs a light-emitting control signal according to the average value signal AUP to control the light-emitting driving module LTD1 to output the pulse width modulation signal PWM as shown in fig. 8.
The light emitting driving module LTD1 drives the light emitting module LED1 matched with the microphone m1 to output a light emitting signal corresponding to the average value signal AUP by a pulse width modulation signal PWM. The larger the volume average value of the average value signal AUP shown in fig. 7 is, the larger the width of the pulse width modulation signal PWM shown in fig. 8 is. In this case, the greater the light intensity of the light emitting signal may be.
[ advantageous effects of the embodiments ]
In summary, the present invention provides a voice processing system and method, which can receive a voice signal from any signal source, and provide a control parameter for a voice processing apparatus to control a voice transceiver by configuring a voice receiver to obtain an absolute value of the voice signal, dividing the voice signal into a plurality of sub-frame signals, filtering the sub-frame signals, and detecting a peak value and an average value of the sub-frame signals.
The disclosure is only a preferred embodiment of the invention and is not intended to limit the scope of the invention, which is defined by the claims and their equivalents.

Claims (20)

1. A voice signal processing system adapted for use with one or more voice signal transceiving devices, the voice signal processing system comprising:
one or more audio signal receivers, the audio signal receivers comprising:
the voice and information interface is connected with the one or more voice and information transceiving devices and configured to receive voice and information signals output by the voice and information transceiving devices;
the absolute value conversion unit is connected with the voice and information interface and is configured to take an absolute value of the voice and information signal so as to output an absolute value voice and information signal;
a sound framing unit connected with the absolute value conversion unit and configured to divide the absolute value sound signal into a plurality of sub sound frame signals;
a feature value detection unit, connected to the frame unit, configured to detect one or more feature values of the sub-frame signal; and
the voice and information processing device is connected with the voice and information receiver and the voice and information receiving and sending device and is configured to output a voice and information processing signal to process the voice and information signal according to the one or more characteristic values of the sub frame signal of the voice and information receiving and sending device.
2. The audio signal processing system of claim 1, wherein the feature value detection unit comprises a peak value detection unit, connected to the framing unit, configured to detect a peak value of the sub-frame signal, the feature value comprising the peak value.
3. The audio signal processing system according to claim 1, wherein said feature value detecting unit comprises an average value calculating unit connected to said framing unit and configured to calculate an average value of said sub-frame signals, said feature value comprising said average value.
4. The audio signal processing system of claim 1, wherein the eigenvalue detection unit comprises a pulse packet filtering unit connected to the framing unit and configured to filter the sub-frame signals.
5. The voice signal processing system according to claim 1, wherein the voice framing unit divides the absolute value voice signal into the plurality of sub voice frame signals having variable lengths.
6. The audio signal processing system according to claim 1, further comprising a light-emitting driving module connected to the audio signal processing apparatus and one or more light-emitting modules, wherein the audio signal processing apparatus is configured to output a light-emitting control signal according to the one or more characteristic values to control the light-emitting driving module to drive the light-emitting module matched with the audio signal transceiver to output a corresponding light-emitting signal.
7. The audio signal processing system according to claim 6, wherein the light-emitting driving module comprises a pulse-width modulation signal generating unit connected to the one or more light-emitting modules and configured to output a pulse-width modulation signal according to the one or more characteristic values to control the light-emitting module matched with the audio signal transceiver to output the corresponding light-emitting signal;
wherein the lighting control signal comprises the pulse width modulation signal.
8. The audio signal processing system according to claim 1, wherein when said audio signal processing means determines that said feature value of said sub-frame signal is smaller than a threshold value and that the time during which said feature value remains smaller than said threshold value is longer than a hold time, said audio signal processing means outputs said audio signal processing signal to turn down said feature value of said audio signal outputted from said audio signal transmitting/receiving means to a mute threshold value during a release time.
9. The audio signal processing system according to claim 1, wherein said audio signal processing means outputs said audio signal processing signal to raise said feature value of said audio signal output from said audio signal transmitting/receiving means during a sound-on time when said audio signal processing means determines that said feature value of said sub-frame signal is greater than a threshold value.
10. The audio signal processing system of claim 1, further comprising a digital volume controller, wherein the audio signal processing apparatus adjusts a gain value of the digital volume controller to eliminate noise of the audio signal when the audio signal is determined to have noise according to the one or more characteristic values of the sub-frame signal.
11. A method for processing audio signals, adapted to one or more audio signal transceiving devices, the method comprising:
receiving the voice and information signal output by the voice and information transceiver by using a voice and information interface of a voice and information receiver;
the absolute value conversion unit of the voice signal receiver is utilized to take the absolute value of the voice signal so as to output the voice signal with the absolute value;
dividing the absolute value voice signal into a plurality of sub-voice frame signals by a voice framing unit of the voice receiver;
detecting one or more feature values of the sub-frame signal by using a feature value detection unit of the voice receiver; and
and outputting a voice signal processing signal by utilizing a voice signal processing device according to the one or more characteristic values of the sub-frame signal so as to process the voice signal.
12. The method according to claim 11, wherein the step of detecting the one or more feature values of the sub-frame signal by the feature value detection unit of the audio receiver comprises:
and detecting the peak value of the sub sound frame signal by using a peak value detection unit of the characteristic value detection unit.
13. The method according to claim 11, wherein the step of detecting the one or more feature values of the sub-frame signal by the feature value detection unit of the audio receiver comprises:
and detecting the average value of the sub-sound frame signals by using the average value calculating unit of the characteristic value detecting unit.
14. The method according to claim 11, wherein the step of detecting the one or more feature values of the sub-frame signal by the feature value detection unit of the audio receiver comprises:
and filtering the sub-sound frame signal by utilizing a pulse packet filtering unit of the characteristic value detection unit.
15. The method according to claim 11, wherein the step of dividing the frame signal into a plurality of sub-frame signals by the framing unit comprises:
the sound framing unit divides the absolute value sound signal into the plurality of sub sound frame signals having variable lengths.
16. The audio signal processing method according to claim 10, further comprising the steps of:
and outputting a light-emitting control signal by using the sound-information processing device according to the one or more characteristic values of the sub-sound frame signal of the sound-information receiving and transmitting device so as to control a light-emitting driving module to drive a light-emitting module to output a corresponding light-emitting signal.
17. The audio signal processing method according to claim 11, further comprising the steps of:
and outputting a pulse width modulation signal by using a pulse width modulation signal generating unit according to the one or more characteristic values to control the light emitting module to output a corresponding light emitting signal.
18. The audio signal processing method according to claim 11, further comprising the steps of:
and when the voice and information processing device is used for judging that the characteristic value of the sub voice frame signal is smaller than a threshold value and the time for maintaining the characteristic value smaller than the threshold value is longer than the holding time, outputting the voice and information processing signal to turn down the characteristic value of the voice and information signal output by the voice and information receiving and transmitting device to a mute threshold value within the release time.
19. The audio signal processing method according to claim 11, further comprising the steps of:
and when the characteristic value of the sub-voice frame signal is judged to be larger than a threshold value by using the voice and information processing device, outputting the voice and information processing signal to turn up the characteristic value of the voice and information signal output by the voice and information receiving and transmitting device within the starting time.
20. The audio signal processing method according to claim 11, further comprising the steps of:
and adjusting the gain value of a digital volume controller to eliminate the noise of the voice signal when the voice signal is judged to have the noise by utilizing the voice processing device according to the one or more characteristic values of the sub-voice frame signal.
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