CN113038339B - System for eliminating echo and improving audio quality - Google Patents
System for eliminating echo and improving audio quality Download PDFInfo
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- CN113038339B CN113038339B CN201911250456.2A CN201911250456A CN113038339B CN 113038339 B CN113038339 B CN 113038339B CN 201911250456 A CN201911250456 A CN 201911250456A CN 113038339 B CN113038339 B CN 113038339B
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
- H04R3/00—Circuits for transducers, loudspeakers or microphones
- H04R3/02—Circuits for transducers, loudspeakers or microphones for preventing acoustic reaction, i.e. acoustic oscillatory feedback
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Abstract
The application relates to a system for eliminating echo and improving audio quality, which comprises: the audio processor is used for playing, collecting audio, recording and processing audio signals; the digital-to-analog converter is connected with the audio processor; the power amplifier is connected with the digital-to-analog converter; the microphone comprises a loudspeaker for externally playing audio connected with a power amplifier, and a voltage division and filtering module positioned between the power amplifier and the loudspeaker, wherein the voltage division and filtering module is 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 filtering signals except the frequency response of the microphone according to the frequency response of the microphone; the second analog-to-digital converter is connected with the voltage division and filtering module and is connected with the audio processor; and the first analog-to-digital converter is connected with the audio processor and is connected with the microphone.
Description
Technical Field
The present application relates to the field of audio processing technologies, and in particular, to a system for eliminating echoes and improving audio quality.
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 record 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, the field of the art will often involve DACs: 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 allows 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 playback spectrum, 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 speaker spectrum, thereby suppressing 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 on the frequency spectrum results in low efficiency in echo cancellation processing and loss of effective signals recorded by a microphone.
Technical content
In order to solve the above problems, the present application provides a system for eliminating echo and improving audio quality, which is capable of collecting playback signals that are closest to the actually recorded echo signal spectrum.
The application provides a system for eliminating echo and improving audio quality, which comprises:
the audio processor is used for playing, collecting audio, recording and processing audio signals;
the digital-to-analog converter DAC is connected with the audio processor and is used for converting the audio digital signals sent by the audio processor into analog signals to be output;
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;
a loudspeaker connected with the power amplifier for externally playing audio, and
the voltage division and filtering module is positioned 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, filtering out signals except the frequency response of the microphone;
the second analog-to-digital converter ADC is connected with the voltage division and filtering module, is connected with the audio processor and is used for converting the signals passing through the voltage division and filtering module into digital signals and transmitting the digital signals to the audio processor for processing; and
and the first analog-to-digital converter ADC is connected with the microphone and is used for transmitting signals recorded by the microphone to the audio processor for processing.
The predetermined standard range is a standard range of levels that conforms to the analog-to-digital converter ADC.
The voltage division and filtering module comprises a voltage divider and a filter.
The voltage divider is located before the filter to divide the level of the obtained signal.
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 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.
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 is provided withThird resistance (R)3) And a second capacitance (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 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.
The system related to the application collects the playback signal from the front end of the loudspeaker, and the signal frequency spectrum after the compression and filtering treatment is closest to the echo signal frequency spectrum actually recorded into the microphone, so that the efficiency of eliminating the echo by the audio silencing processor is improved, and the change of the sound frequency spectrum of a speaker and the sound distortion are avoided.
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 further block diagram schematic of an embodiment of the present application.
Fig. 4 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 allows a speaker sound spectrum to pass through according to real-time change of comparison between the two signals, suppresses background noise, i.e., the playback spectrum, reduces energy thereof, and achieves the effect of suppressing echo. When someone speaks, the audio processor analyzes the signal and analyzes the speaker spectrum, thereby suppressing 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 audio processor is used for playing, collecting audio, recording and processing audio signals;
the digital-to-analog converter DAC is connected with the audio processor and is used for converting the audio digital signals sent by the audio processor into analog signals to be output;
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;
a loudspeaker connected with the power amplifier for externally playing audio, and
the voltage division and filtering module is positioned 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, filtering out signals except the frequency response of the microphone;
the second analog-to-digital converter ADC is connected with the voltage division and filtering module, is connected with the audio processor and is used for converting the signals passing through the voltage division and filtering module into digital signals and transmitting the digital signals to the audio processor for processing; and
and the first analog-to-digital converter ADC is connected with the microphone and is used for transmitting signals recorded by the microphone to the audio processor for processing.
The predetermined standard range is a standard range of levels that conforms to the analog-to-digital converter ADC.
As shown in fig. 3, the voltage divider and filter module includes a voltage divider and a filter. The voltage divider is located before the filter to divide the level of the obtained signal.
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 collected playback signal is basically consistent with the actually recorded echo signal frequency spectrum, and the digital filter designed according to the collected 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. 4, 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 is passed through a voltage dividing resistor R2And R4To VO+Partial pressure is carried out, as shown in FIG. 4 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 can not exceed 20 Hz-10 KHz, 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 C6The 6.8K omega resistor and the 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 variable voltage, and the direct current is 0Hz on the frequency spectrum, the sampling is meaningless. 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.
In fact, C5 and R3 form an RC high-pass filter, and the calculation formula of the cut-off frequency can be also usedIt follows that signals below the cut-off frequency of about 21.1Hz can be filtered out.
Finally, the above embodiments are only for illustrating the technical solutions of the present invention and not for limiting, although the present invention has been described in detail with reference to the preferred embodiments, it should be understood by those skilled in the art that modifications or equivalent substitutions may be made to the technical solutions of the present invention without departing from the spirit and scope of the technical solutions of the present invention, and all of them should be covered in the claims of the present invention.
Claims (10)
1. A system for canceling echo to improve audio quality, comprising:
the audio processor is used for playing, collecting audio, recording and processing audio signals;
the digital-to-analog converter DAC is connected with the audio processor and is used for converting the audio digital signals sent by the audio processor into analog signals to be output;
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;
a loudspeaker connected with the power amplifier for externally playing audio, and
the voltage division and filtering module is positioned 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, filtering out signals except the frequency response of the microphone;
the second analog-to-digital converter ADC is connected with the voltage division and filtering module, is connected with the audio processor and is used for converting the signals passing through the voltage division and filtering module into digital signals and transmitting the digital signals to the audio processor for processing; and
and the first analog-to-digital converter ADC is connected with the microphone and is used for transmitting signals recorded by the microphone to the audio processor for processing.
2. The system of claim 1, wherein the predetermined standard range is a standard range of levels for an analog-to-digital converter (ADC).
3. The system of claim 1, wherein the voltage divider and filter module comprises a voltage divider and a filter.
4. The system as claimed in claim 3, wherein the voltage divider is located before the filter to divide the level of the obtained signal.
5. The system as claimed in claim 4, wherein the voltage divider comprises at least two voltage dividing resistors, each of which is a first voltage dividing 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.
6. The system of claim 5 wherein V is the same as VO+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.
7. The system of claim 3, wherein the filter comprises an RC low-pass filter and/or an RC high-pass filter.
8. The system of claim 7, wherein the RC low pass filter is designed to have 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.
9. The system of claim 8, wherein the frequency response range of the 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.
10. The system of claim 9 wherein the formula is calculated based on the cut-off 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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CN105376683A (en) * | 2014-08-06 | 2016-03-02 | 北京卓锐微技术有限公司 | Silicon microphone amplifier capable of eliminating noise of charge pump |
CN108447483A (en) * | 2018-05-18 | 2018-08-24 | 深圳市亿道数码技术有限公司 | Speech recognition system |
CN208538474U (en) * | 2018-05-18 | 2019-02-22 | 深圳市亿道数码技术有限公司 | Speech recognition system |
CN109817238A (en) * | 2019-03-14 | 2019-05-28 | 百度在线网络技术(北京)有限公司 | Audio signal sample device, acoustic signal processing method and device |
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US7774079B2 (en) * | 2005-10-03 | 2010-08-10 | Sigmatel, Inc. | Method and system for receiving and decoding audio signals |
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CN105376683A (en) * | 2014-08-06 | 2016-03-02 | 北京卓锐微技术有限公司 | Silicon microphone amplifier capable of eliminating noise of charge pump |
CN108447483A (en) * | 2018-05-18 | 2018-08-24 | 深圳市亿道数码技术有限公司 | Speech recognition system |
CN208538474U (en) * | 2018-05-18 | 2019-02-22 | 深圳市亿道数码技术有限公司 | Speech recognition system |
CN109817238A (en) * | 2019-03-14 | 2019-05-28 | 百度在线网络技术(北京)有限公司 | Audio signal sample device, acoustic signal processing method and device |
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