CN113038345B - System for optimizing audio signal in audio acquisition process - Google Patents
System for optimizing audio signal in audio acquisition process Download PDFInfo
- Publication number
- CN113038345B CN113038345B CN201911250388.XA CN201911250388A CN113038345B CN 113038345 B CN113038345 B CN 113038345B CN 201911250388 A CN201911250388 A CN 201911250388A CN 113038345 B CN113038345 B CN 113038345B
- Authority
- CN
- China
- Prior art keywords
- analog
- signal
- voltage
- digital
- level
- Prior art date
- Legal status (The legal status is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the status listed.)
- Active
Links
Images
Classifications
-
- 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
-
- 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
-
- 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
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 digital-to-analog converter DAC and used for receiving the analog signal at the output end of the digital-to-analog converter DAC and outputting the analog signal at the maximum power; the voltage dividing and filtering module is connected between the power amplifier and the loudspeaker and used for reducing the level of a signal sent 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 designed to be filtered; and the analog-to-digital converter ADC is connected with the voltage division and filtering module and is used for converting the signal which passes through the voltage division and filtering module into a digital signal.
Description
Technical Field
The application relates to the technical field of audio processing, in particular to a system for optimizing audio signals in an audio acquisition process.
Background
In the field of artificial intelligent audio entry and real-time call audio, an audio system needs to play and record simultaneously, so that the sound played by the current system can be recorded simultaneously during recording, 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 cancellation system, 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, wherein the digital filter allows a sound spectrum of a speaker to pass through according to real-time change of comparison of the two signals, inhibits background noise, namely the played spectrum, reduces energy of the played spectrum, 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 signal which passes through the voltage division and filtering module into a digital signal.
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, V O+ Is the level value of the playback signal; v ADC Is the proper level value of the analog-digital converter ADC; r is 2 Is the resistance value of the first divider resistor, R 4 The resistance value of the second divider resistor.
The V is O+ The voltage amplitude is usually 5V; according to the voltage range sampled by ADC, the voltage amplitude V should be adjusted ADC Reduced to 1V; calculating to obtain R 2 、R 4 Suitable 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 designs a third resistor (R) 3 ) And a first capacitance (C) 6 ) A component for performing 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 cannot exceed 20Hz to 10 KHz; finally, a DC blocking capacitor (C) is required 5 ) 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 R 3 And C 6 Respectively taking a 6.8K omega resistor and a 2.2nF appropriate capacitor, and filtering out signals with the cut-off frequency higher than 9.6 kHz; while according to cut-off frequencyRate calculation formulaAnd 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 dividing and filtering module is connected between the power amplifier and the loudspeaker and used for reducing the level of a signal sent 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 signals except the frequency response of the microphone are designed to be 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 comprises a voltage divider and a filter, wherein the voltage divider is positioned in front of 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 frequency spectrum of the collected playback signal is basically consistent with the frequency spectrum of the echo signal actually recorded, and the digital filter designed according to the collected signal can eliminate the echo signal in the recording signal more effectively, thereby avoiding the phenomenon that the digital filter changes the sound spectrum of a speaker and the sound distortion caused by 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 V O+ The voltage amplitude is about 5V, and is reduced to about 1V according to the sampling voltage range of the analog-to-digital converter ADC, and the voltage amplitude is divided by a divider resistor R 2 And R 4 To V O+ Performing partial pressure, as shown in FIG. 3, according toCalculating to obtain R 2 、R 4 The resistances of 6.8K omega and 1.8K omega are selected to be appropriate, respectively. And because the frequency response range of the common microphone does not exceed 20Hz to 10KHz, the resistance R is designed to pass 3 And a capacitor C 6 The RC low-pass filter is used for low-pass filtering and the cutoff frequency calculation formula is usedTo give R 3 And C 6 A 6.8K Ω resistor and a 2.2nF capacitor, respectively, were appropriate to filter out signals above this cutoff frequency by about 9.6 kHz. And because ADC chip sampling is to collect variable voltage, and DC is at frequencyThe spectrum is 0Hz, and the collection is meaningless. Finally, a DC blocking capacitor C is required 5 (generally, a capacitor larger than 1uF is selected) to remove the DC level, and then the ADC is used for sampling.
Claims (1)
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;
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 division and filtering module comprises a voltage divider and a filter, wherein the voltage divider is positioned in front of the filter to divide the level of the obtained signal;
the preset standard range is a level standard range which accords with an analog-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, V O+ Is the level value of the playback signal; v ADC Is the proper level value of the analog-to-digital converter ADC; r 2 Is the resistance value of the first divider resistor, R 4 Is the resistance value of the second divider resistor;
the V is O+ 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 adjusted ADC Reducing to 1V; calculating to obtain R 2 、R 4 Selecting appropriate resistance values of 6.8K omega and 1.8K omega respectively;
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 capacitor C 6 A component for low-pass filtering; the RC high-pass filter is formed by designing a third resistor R 3 And a second capacitor C 5 Forming for high pass filtering;
the frequency response range of the microphone cannot exceed 20Hz to 10KHz; finally, a DC blocking capacitor C is required 5 Removing DC level, and sampling in a second ADC (analog-to-digital converter), wherein the second capacitor C 5 And the blocking capacitor C 5 Are the same capacitance.
Priority Applications (1)
Application Number | Priority Date | Filing Date | Title |
---|---|---|---|
CN201911250388.XA CN113038345B (en) | 2019-12-09 | 2019-12-09 | System for optimizing audio signal in audio acquisition process |
Applications Claiming Priority (1)
Application Number | Priority Date | Filing Date | Title |
---|---|---|---|
CN201911250388.XA CN113038345B (en) | 2019-12-09 | 2019-12-09 | System for optimizing audio signal in audio acquisition process |
Publications (2)
Publication Number | Publication Date |
---|---|
CN113038345A CN113038345A (en) | 2021-06-25 |
CN113038345B true CN113038345B (en) | 2023-03-14 |
Family
ID=76451161
Family Applications (1)
Application Number | Title | Priority Date | Filing Date |
---|---|---|---|
CN201911250388.XA Active CN113038345B (en) | 2019-12-09 | 2019-12-09 | System for optimizing audio signal in audio acquisition process |
Country Status (1)
Country | Link |
---|---|
CN (1) | CN113038345B (en) |
Citations (5)
Publication number | Priority date | Publication date | Assignee | Title |
---|---|---|---|---|
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 |
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 |
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 |
Family Cites Families (3)
Publication number | Priority date | Publication date | Assignee | Title |
---|---|---|---|---|
US6963648B2 (en) * | 2000-04-17 | 2005-11-08 | Harold D. Wilder | Echo/noise canceling device for use with personal computers |
US8315379B2 (en) * | 2004-11-10 | 2012-11-20 | Matech, Inc. | Single transducer full duplex talking circuit |
US8229104B2 (en) * | 2008-06-24 | 2012-07-24 | Thomson Licensing | Full duplex telephone system employing automatic level control for improved digital signal processing of audio signals |
-
2019
- 2019-12-09 CN CN201911250388.XA patent/CN113038345B/en active Active
Patent Citations (5)
Publication number | Priority date | Publication date | Assignee | Title |
---|---|---|---|---|
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 |
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 |
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 |
Also Published As
Publication number | Publication date |
---|---|
CN113038345A (en) | 2021-06-25 |
Similar Documents
Publication | Publication Date | Title |
---|---|---|
CN101267223B (en) | Bass enhancing method, signal processing device, and audio reproducing system | |
EP2712207A1 (en) | Audio signal broadcasting system and electronic apparatus | |
CN108401204A (en) | A kind of novel active noise reduction earphone | |
CN111182431A (en) | Howling suppression method for conference sound reinforcement system | |
CN103796136A (en) | Equipment and method for ensuring output loudness and tone quality of different sound effect modes | |
CN101808260A (en) | Audio dynamic feedback suppression method | |
CN113038345B (en) | System for optimizing audio signal in audio acquisition process | |
CN113038339B (en) | System for eliminating echo and improving audio quality | |
CN113035218B (en) | Method for optimizing audio signal in audio acquisition process | |
CN113035224B (en) | Equipment for eliminating echo and improving audio quality | |
US8831236B2 (en) | Generator and generation method of pseudo-bass | |
CN113035219B (en) | Method for eliminating echo and improving audio quality | |
CN114286253B (en) | Audio processing method and device and audio playing equipment | |
TWI413111B (en) | Method and apparatus for elimination noise background noise (2) | |
CN106231502A (en) | The frequency response method of reduction treatment of a kind of phase-shift circuit and circuit | |
JP4658924B2 (en) | Filter circuit and reproducing apparatus using the same | |
TWI651970B (en) | Crossover device | |
CN106023998A (en) | Camera audio input device, denoising method and camera | |
CN109427345B (en) | Wind noise detection method, device and system | |
US6628794B1 (en) | Method and apparatus for level limitation in a digital hearing aid | |
CN110932685A (en) | Digital amplitude-frequency balance power amplifier based on MATLAB design | |
JPH07506677A (en) | Apparatus and method for voice band reduction | |
CN216930302U (en) | Circuit for eliminating loudspeaker noise | |
KR102443510B1 (en) | Apparatus for reducing noise from voice signal of low-impedance microphone in intercom system | |
CN106658306B (en) | Partial device |
Legal Events
Date | Code | Title | Description |
---|---|---|---|
PB01 | Publication | ||
PB01 | Publication | ||
SE01 | Entry into force of request for substantive examination | ||
SE01 | Entry into force of request for substantive examination | ||
GR01 | Patent grant | ||
GR01 | Patent grant |