CN113038345A - System for optimizing audio signal in audio acquisition process - Google Patents
System for optimizing audio signal in audio acquisition process Download PDFInfo
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- CN113038345A CN113038345A CN201911250388.XA CN201911250388A CN113038345A CN 113038345 A CN113038345 A CN 113038345A CN 201911250388 A CN201911250388 A CN 201911250388A CN 113038345 A CN113038345 A CN 113038345A
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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
- H04R9/00—Transducers of moving-coil, moving-strip, or moving-wire type
- H04R9/06—Loudspeakers
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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
- H04R9/00—Transducers of moving-coil, moving-strip, or moving-wire type
- H04R9/02—Details
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- Y—GENERAL TAGGING OF NEW TECHNOLOGICAL DEVELOPMENTS; GENERAL TAGGING OF CROSS-SECTIONAL TECHNOLOGIES SPANNING OVER SEVERAL SECTIONS OF THE IPC; TECHNICAL SUBJECTS COVERED BY FORMER USPC CROSS-REFERENCE ART COLLECTIONS [XRACs] AND DIGESTS
- Y02—TECHNOLOGIES OR APPLICATIONS FOR MITIGATION OR ADAPTATION AGAINST CLIMATE CHANGE
- Y02D—CLIMATE CHANGE MITIGATION TECHNOLOGIES IN INFORMATION AND COMMUNICATION TECHNOLOGIES [ICT], I.E. INFORMATION AND COMMUNICATION TECHNOLOGIES AIMING AT THE REDUCTION OF THEIR OWN ENERGY USE
- Y02D30/00—Reducing energy consumption in communication networks
- Y02D30/70—Reducing energy consumption in communication networks in wireless communication networks
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- Analogue/Digital Conversion (AREA)
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Abstract
The present application relates to a system for optimizing an audio signal during audio acquisition, comprising: the digital-to-analog converter DAC is used for converting the received audio digital signals into analog signals and outputting the analog signals to the power amplifier; the power amplifier is connected with the DAC and used for receiving the analog signal output by the DAC and outputting the analog signal with the maximum power; the voltage division and filtering module is connected between the power amplifier and the loudspeaker and is used for reducing the level of a signal sent out by the output end of the power amplifier to a standard range preset by the level through voltage division, and then according to the frequency response of the microphone, the voltage division and filtering module is designed to filter signals except the frequency response of the microphone; and the analog-to-digital converter ADC is connected with the voltage division and filtering module and is used for converting the signals passing through the voltage division and filtering module into digital signals.
Description
Technical Field
The present application relates to the field of audio processing technologies, and in particular, to a system for optimizing audio signals during audio acquisition.
Background
In the field of artificial intelligence audio entry and real-time communication audio, an audio system needs to play and record simultaneously, so that the recording can be recorded into the sound played by the current system simultaneously, and the recorded and played sound forms echo. To improve audio quality, the system acquires an echo signal and cancels the echo signal from the recorded signal. I.e. the recorded signal is subjected to echo cancellation processing. Furthermore, DAC's are often involved in the art: the digital-to-analog chip is a chip for converting digital signal input into analog signal output; and an ADC: the analog-to-digital chip is a chip for converting an analog signal input into a digital signal output.
In the existing design of an echo eliminating system, an audio processor carries out spectrum analysis on a recording signal and a playback signal, analyzes response intensity and spectrum distribution, and designs a digital filter according to the analysis result, wherein the digital filter enables a speaker to pass through a sound spectrum according to the real-time change of the comparison of the two signals, inhibits background noise, namely the frequency spectrum of playback, reduces the energy of the background noise, and achieves the effect of inhibiting echo. When someone speaks, the audio processor analyzes the signal and analyzes the frequency spectrum of the speaker, thereby inhibiting the echo.
However, in the current design of echo cancellation systems, the echo acquisition part is generally completed at the stage of the original audio digital signal of the playback signal, and the audio processor compares, analyzes and processes the frequency spectrum of the original audio digital signal of the playback signal and the recording signal. In fact, the original audio digital signal passes through the DAC and the amplifier, is played by the loudspeaker and then is recorded and sampled by the microphone, and the frequency spectrum of the playback signal actually recorded into the microphone is changed. Then, a digital filter designed according to the spectrum of the original audio digital signal of the playback signal is used to process the audio signal, so that the sound spectrum of the speaker is changed and the sound is distorted.
The difference of the playback signal acquired by the echo cancellation method and the actually recorded echo signal in the frequency spectrum causes low efficiency in echo cancellation processing and loss of effective signals recorded by a microphone.
Technical content
In order to solve the above problem, the present application provides a system for optimizing an audio signal in an audio acquisition process, comprising:
the digital-to-analog converter DAC is used for converting the received audio digital signals into analog signals and outputting the analog signals to the power amplifier;
the power amplifier is connected with the DAC and used for receiving the analog signal output by the DAC and outputting the analog signal with the maximum power;
the voltage division and filtering module is connected between the power amplifier and the loudspeaker and is used for reducing the level of a signal sent out by the output end of the power amplifier to a level preset standard range through voltage division, and then according to the frequency response of the microphone, the voltage division and filtering module is designed to filter signals except the frequency response of the microphone;
and the analog-to-digital converter ADC is connected with the voltage division and filtering module and is used for converting the signals passing through the voltage division and filtering module into digital signals.
The voltage dividing and filtering module includes a voltage divider and a filter, the voltage divider being located before the filter to divide the level of the obtained signal.
The predetermined standard range is a standard range of levels that conforms to the analog-to-digital converter ADC.
The voltage divider at least comprises two divider resistors which are respectively a first divider resistor (R)2) And a second voltage dividing resistor (R)4) According toCalculating to obtain a first divider resistance (R)2) A second voltage dividing resistor (R)4) Suitable two resistance values; wherein, VO+Is the level value of the playback signal; vADCIs a proper level value of the analog-to-digital converter ADC; r2Is the resistance value of the first divider resistor, R4Is the resistance value of the second divider resistor.
The V isO+The voltage amplitude is usually 5V; according to the sampling voltage range of the analog-to-digital converter ADC, the voltage amplitude V should be adjustedADCReducing to 1V; calculating to obtain R2、R4Suitable resistance values of 6.8K omega and 1.8K omega, respectively, may be selected.
The filter comprises an RC low-pass filter and/or an RC high-pass filter.
The RC low-pass filter is formed by designing a third resistor (R)3) And a first capacitance (C)6) A component for low-pass filtering; the RC high-pass filter designs a third resistor (R)3) And a second capacitor (C)5) And (c) means for performing high-pass filtering.
The frequency response range of the microphone is not beyond 20 Hz-10 KHz; finally, a DC blocking capacitor (C) is required5) And removing the direct current level, and then entering a second analog-to-digital converter (ADC) for sampling.
According to the formula of cut-off frequency calculationTo obtain R3And C6Respectively taking a 6.8K omega resistor and a 2.2nF appropriate capacitor, and filtering out signals with the cutoff frequency higher than 9.6 kHz; while calculating formula based on cut-off frequencyAnd obtaining a proper resistance value and a proper capacitance value, and filtering out signals with the cut-off frequency lower than 21.1 Hz.
Drawings
Fig. 1 is a block diagram schematic diagram of a prior art system.
Fig. 2 is a block diagram schematic of an embodiment of the present application.
Fig. 3 is a circuit diagram of a system according to an embodiment of the present application.
Detailed Description
As shown in fig. 1, in the current design of eliminating echo, an audio processor performs spectrum analysis on a recording signal and a playback signal, analyzes response intensity and spectrum distribution, and designs a digital filter, which passes the sound spectrum of a speaker according to the real-time change of the comparison of the two signals, suppresses background noise, i.e., the playback spectrum, reduces the energy thereof, and achieves the effect of suppressing echo. When someone speaks, the audio processor analyzes the signal and analyzes the frequency spectrum of the speaker, so as to suppress the echo.
The present application relates to a new system for eliminating echo and improving audio quality, as shown in fig. 2, the system includes:
the digital-to-analog converter DAC is used for converting the received audio digital signals into analog signals and outputting the analog signals to the power amplifier;
the power amplifier is connected with the DAC and used for receiving the analog signal output by the DAC and outputting the analog signal with the maximum power;
the voltage division and filtering module is connected between the power amplifier and the loudspeaker and is used for reducing the level of a signal sent out by the output end of the power amplifier to a standard range preset by the level through voltage division, and then according to the frequency response of the microphone, the signal except the frequency response of the microphone is filtered;
and the analog-to-digital converter ADC is connected with the voltage division and filtering module and is used for converting the signals passing through the voltage division and filtering module into digital signals.
The voltage dividing and filtering module includes a voltage divider and a filter, the voltage divider being located before the filter to divide the level of the obtained signal. The predetermined standard range is a standard range of levels that conforms to the analog-to-digital converter ADC.
The new echo eliminating method is that the sound playing signal collected in the front end of the loudspeaker has frequency spectrum identical to that of the loudspeaker sound, the level of the signal is first lowered to the level standard range of ADC chip through voltage division, and the filter is designed to filter out the signal except the microphone frequency response based on the microphone frequency response. The acquired playback signal is basically consistent with the actually recorded echo signal frequency spectrum, and the digital filter designed according to the acquired signal can eliminate the echo signal in the recording signal more effectively, thereby avoiding the sound spectrum and sound distortion of the speaker caused by the digital filter due to the frequency spectrum difference.
In the circuit shown in FIG. 3, the rear end of the power amplifier and the front end of the horn collect the playback signal VO+The voltage amplitude is about 5V, and the voltage amplitude is reduced to about 1V according to the sampling voltage range of the analog-to-digital converter ADC, and the voltage amplitude passes through the divider resistor R2And R4To VO+Performing partial pressure, as shown in FIG. 3, according toCalculating to obtain R2、R4The resistances of 6.8K omega and 1.8K omega are respectively selected to be appropriate. And because the frequency response range of the common microphone does not exceed 20Hz to 10KHz, the resistance R is designed to pass3And a capacitor C6The RC low-pass filter is used for low-pass filtering and the cutoff frequency calculation formula is usedTo obtain R3And C6A6.8K omega resistor and a 2.2nF capacitor are respectively taken as appropriate, and signals higher than the cut-off frequency by about 9.6kHz are filtered. And because the ADC chip samples the voltage of collecting the change, and the direct current is 0Hz on the frequency spectrum, it has no meaning to the collection. Finally, a DC blocking capacitor C is required5(generally, a capacitor larger than 1uF is selected) to remove the DC level, and then the ADC is used for sampling.
Claims (9)
1. A system for optimizing an audio signal during audio acquisition, comprising:
the digital-to-analog converter DAC is used for converting the received audio digital signals into analog signals and outputting the analog signals to the power amplifier;
the power amplifier is connected with the DAC and used for receiving the analog signal output by the DAC and outputting the analog signal with the maximum power;
the voltage division and filtering module is connected between the power amplifier and the loudspeaker and is used for reducing the level of a signal sent out by the output end of the power amplifier to a standard range preset by the level through voltage division, and then according to the frequency response of the microphone, the voltage division and filtering module is designed to filter signals except the frequency response of the microphone;
and the analog-to-digital converter ADC is connected with the voltage division and filtering module and is used for converting the signals passing through the voltage division and filtering module into digital signals.
2. The system of claim 1, wherein the voltage divider and filter module comprises a voltage divider and a filter, the voltage divider being located before the filter to divide the level of the obtained signal.
3. The system of claim 1, wherein the predetermined standard range is a standard range of levels for an analog-to-digital converter (ADC).
4. A system for optimizing audio signals during audio acquisition as claimed in claim 2, wherein the voltage divider comprises at least two divider resistors, each of which is a first divider resistor (R)2) And a second voltage dividing resistor (R)4) According toCalculating to obtain a first divider resistance (R)2) A second voltage dividing resistor (R)4) Suitable two resistance values; wherein, VO+Is the level value of the playback signal; vADCIs the proper level value of the analog-to-digital converter ADC; r2Is the resistance value of the first divider resistor, R4Is the resistance value of the second divider resistor.
5. The system of claim 4, wherein V is a function of the audio signal being optimized during the audio acquisitionO+The voltage amplitude is usually 5V; according to the sampling voltage range of the analog-to-digital converter ADC, the voltage amplitude V should be adjustedADCReducing to 1V; calculating to obtain R2、R4Suitable resistance values of 6.8K omega and 1.8K omega, respectively, may be selected.
6. A system for optimizing audio signals during audio acquisition as recited in claim 2, wherein the filter comprises an RC low pass filter and/or an RC high pass filter.
7. A system for optimizing audio signals during audio acquisition as claimed in claim 6, wherein the RC low pass filter is implemented by designing a third resistor (R)3) And a first capacitance (C)6) A component for low-pass filtering; the RC high-pass filter designs a third resistor (R)3) And a second capacitance (C)5) And (c) means for performing high-pass filtering.
8. The system for optimizing audio signals during audio acquisition of claim 7 wherein the frequency response range of said microphone does not exceed 20Hz to 10 KHz; finally, a DC blocking capacitor (C) is required5) And removing the direct current level, and then entering a second analog-to-digital converter (ADC) for sampling.
9. The system of claim 8, wherein the formula is calculated based on a cutoff frequencyTo obtain R3And C6Respectively taking a 6.8K omega resistor and a 2.2nF appropriate capacitor, and filtering out signals with the cutoff frequency higher than 9.6 kHz; while calculating formula according to cut-off frequencyAnd obtaining a proper resistance value and a proper capacitance value, and filtering out signals with the cut-off frequency lower than 21.1 Hz.
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Citations (8)
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US4670903A (en) * | 1981-06-30 | 1987-06-02 | Nippon Electric Co., Ltd. | Echo canceller for attenuating acoustic echo signals on a frequency divisional manner |
US6173056B1 (en) * | 1998-08-25 | 2001-01-09 | Ericsson Inc. | Methods for adjusting audio signals responsive to changes in a power supply level and related communications devices |
US20010038697A1 (en) * | 2000-04-17 | 2001-11-08 | Wilder Kevin Dean | Echo/noise canceling device for use with personal computers |
US20080170515A1 (en) * | 2004-11-10 | 2008-07-17 | Matech, Inc. | Single transducer full duplex talking circuit |
CN101373960A (en) * | 2007-08-20 | 2009-02-25 | 罗姆股份有限公司 | Output limiting circuit, class d power amplifier and audio equipment |
US20110150209A1 (en) * | 2008-06-24 | 2011-06-23 | Wilhelm Ernst Riedl | Full duplex telephone system employing automatic level control for improved digital signal processing of audio signals |
CN105825862A (en) * | 2015-01-05 | 2016-08-03 | 沈阳新松机器人自动化股份有限公司 | Robot man-machine dialogue echo cancellation system |
WO2018211759A1 (en) * | 2017-05-19 | 2018-11-22 | 株式会社Jvcケンウッド | Noise elimination device, noise elimination method and noise elimination program |
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2019
- 2019-12-09 CN CN201911250388.XA patent/CN113038345B/en active Active
Patent Citations (8)
Publication number | Priority date | Publication date | Assignee | Title |
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US4670903A (en) * | 1981-06-30 | 1987-06-02 | Nippon Electric Co., Ltd. | Echo canceller for attenuating acoustic echo signals on a frequency divisional manner |
US6173056B1 (en) * | 1998-08-25 | 2001-01-09 | Ericsson Inc. | Methods for adjusting audio signals responsive to changes in a power supply level and related communications devices |
US20010038697A1 (en) * | 2000-04-17 | 2001-11-08 | Wilder Kevin Dean | Echo/noise canceling device for use with personal computers |
US20080170515A1 (en) * | 2004-11-10 | 2008-07-17 | Matech, Inc. | Single transducer full duplex talking circuit |
CN101373960A (en) * | 2007-08-20 | 2009-02-25 | 罗姆股份有限公司 | Output limiting circuit, class d power amplifier and audio equipment |
US20110150209A1 (en) * | 2008-06-24 | 2011-06-23 | Wilhelm Ernst Riedl | Full duplex telephone system employing automatic level control for improved digital signal processing of audio signals |
CN105825862A (en) * | 2015-01-05 | 2016-08-03 | 沈阳新松机器人自动化股份有限公司 | Robot man-machine dialogue echo cancellation system |
WO2018211759A1 (en) * | 2017-05-19 | 2018-11-22 | 株式会社Jvcケンウッド | Noise elimination device, noise elimination method and noise elimination program |
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