CN113890951A - Echo cancellation method, device and system - Google Patents
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- G10L21/00—Speech or voice signal processing techniques to produce another audible or non-audible signal, e.g. visual or tactile, in order to modify its quality or its intelligibility
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- G10L21/00—Speech or voice signal processing techniques to produce another audible or non-audible signal, e.g. visual or tactile, in order to modify its quality or its intelligibility
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Abstract
The embodiment of the disclosure provides an echo cancellation method, device and system, wherein the method comprises the following steps: receiving a first audio digital signal from a system-on-chip, wherein the first audio digital signal is a digital signal corresponding to a first audio analog signal to be played by a loudspeaker; determining a frequency multiplication sampling clock of a second audio analog signal according to the original sampling frequency information corresponding to the first audio digital signal, and performing oversampling on the second audio analog signal acquired by the microphone according to the frequency multiplication sampling clock to obtain a second audio digital signal, wherein the frequency multiplication sampling clock comprises: presetting sampling frequency and sampling start time, wherein the second audio analog signal is a first audio analog signal after air propagation; and carrying out echo cancellation processing on the received second audio digital signal according to the first audio digital signal to obtain a processed sound digital signal.
Description
Technical Field
The present disclosure relates to the field of communications, and in particular, to a method, an apparatus, and a system for echo cancellation.
Background
Echo cancellation is increasingly used as an important function of audio equipment, in equipment having both a speaker and a microphone, the speaker outputs sound and the microphone simultaneously picks up sound from the environment, and the part of sound has both the speaker sound and the human sound, and in order to accurately pick up the human sound and cancel the interference of the speaker sound, it is necessary to adopt the technical means of echo cancellation to eliminate or reduce the speaker sound in the sound picked up by the microphone as much as possible.
The conventional technical means is to directly collect the analog sound output by the loudspeaker into a DSP (namely a digital signal processor) for analysis and calculation, extract the sound characteristic of the loudspeaker, and subtract the characteristic sound of the loudspeaker from the signal picked up by the microphone.
The echo cancellation architecture in the mode has the disadvantages that a wire is needed to connect the loudspeaker and the DSP, the wire directly samples the analog signal amplified by the power amplifier, namely the analog signal to be input into the loudspeaker is sampled, the arrangement of the loudspeaker is very unfavorable, the DSP and the loudspeaker are required to be designed as a whole, and the local structure of the equipment is limited; further, since the signal output by the sampling speaker is an analog signal, the DSP generally needs a circuit to adapt the gain to utilize the analog signal, so that the whole echo cancellation system has customization, cannot be applied to other product designs at will, and has poor universality.
Disclosure of Invention
In view of this, the embodiments of the present disclosure provide an echo cancellation method, apparatus, and system to solve the following problems in the prior art: in the echo cancellation scheme in the prior art, sampling is required to be performed on an analog signal to be input into a loudspeaker, so that the condition that a wire is required to be connected between the loudspeaker and a DSP to acquire the analog signal is limited, great difficulty is caused to the deployment problem of the loudspeaker, the local structure of equipment is limited, and the universality of a designed echo cancellation architecture is poor.
In one aspect, an embodiment of the present disclosure provides an echo cancellation method, including: receiving a first audio digital signal from a system-on-chip, wherein the first audio digital signal is a digital signal corresponding to a first audio analog signal to be played by a loudspeaker; determining a frequency multiplication sampling clock of a second audio analog signal according to the original sampling frequency information corresponding to the first audio digital signal, and performing oversampling on the second audio analog signal acquired by the microphone according to the frequency multiplication sampling clock to obtain a second audio digital signal, wherein the frequency multiplication sampling clock comprises: presetting sampling frequency and sampling start time, wherein the second audio analog signal is a first audio analog signal after air propagation; and carrying out echo cancellation processing on the received second audio digital signal according to the first audio digital signal to obtain a processed sound digital signal.
In some embodiments, the performing echo cancellation processing on the received second audio digital signal according to the first audio digital signal includes: normalizing the first audio digital signal and the second audio digital signal to obtain normalized first audio data and normalized second audio data, wherein the normalizing comprises: digital decoding and digital filtering; adjusting the amplitude of the first audio data at each preset sampling time point so that the amplitude of the adjusted first audio data at each preset sampling time point is matched with the amplitude of the second audio data at each preset sampling time point; and the echo signals in the second audio data are counteracted through the first audio data after the amplitude adjustment.
In some embodiments, the adjusting the amplitude of the first audio data at each predetermined sampling time point so that the amplitude of the adjusted first audio data at each predetermined sampling time point matches the amplitude of the second audio data at each predetermined sampling time point includes: adjusting the amplitude of first audio data at a first sampling time point by a first predetermined multiple so that the adjusted amplitude is the same as the amplitude of second audio data at the first sampling time point; adjusting the amplitudes of all the preset sampling time points corresponding to the first audio data according to the first preset multiple; sequentially comparing whether the difference value between the amplitude of the second audio data and the amplitude of the adjusted first audio data at each preset sampling time point is greater than or equal to zero from the first sampling time point; and under the condition that all the difference values are larger than or equal to zero, determining that the amplitude of the first audio data at each preset sampling time point after the amplitude adjustment according to the first preset multiple is matched with the amplitude of the second audio data at each preset sampling time point.
In some embodiments, after sequentially comparing whether the difference between the amplitude of the second audio data and the amplitude of the adjusted first audio data at each predetermined sampling time point is greater than or equal to zero, the method further includes: in the case that a second sampling time point with a difference value smaller than zero exists, adjusting the amplitude of the first audio data at the second sampling time point by a second preset multiple so that the adjusted amplitude is the same as the amplitude of the second audio data at the second sampling time point; adjusting the amplitudes of all the preset sampling time points corresponding to the first audio data according to the second preset multiple; sequentially comparing whether the difference value between the amplitude of the second audio data and the amplitude of the adjusted first audio data at each preset sampling time point is greater than or equal to zero from the second sampling time point; and under the condition that all the difference values are larger than or equal to zero, determining that the amplitude of the first audio data at each preset sampling time point after the amplitude adjustment according to the second preset multiple is matched with the amplitude of the second audio data at each preset sampling time point.
In some embodiments, the adjusting the amplitude of the first audio data at each predetermined sampling time point comprises: splitting the first audio data into a plurality of sub audio data according to a preset time period; and respectively adjusting the amplitude of each preset sampling time point corresponding to each sub-audio data.
On the other hand, an embodiment of the present disclosure provides an echo cancellation device, including: the receiving module is used for receiving a first audio digital signal from a system-on-chip, wherein the first audio digital signal is a digital signal corresponding to a first audio analog signal to be played by a loudspeaker; a determining module, configured to determine a frequency-doubled sampling clock of a second audio analog signal according to original sampling frequency information corresponding to the first audio digital signal, so as to perform oversampling on the second audio analog signal acquired by a microphone according to the frequency-doubled sampling clock to obtain a second audio digital signal, where the frequency-doubled sampling clock includes: presetting sampling frequency and sampling start time, wherein the second audio analog signal is a first audio analog signal after air propagation; and the processing module is used for carrying out echo cancellation processing on the second audio digital signal according to the first audio digital signal so as to obtain a processed sound digital signal.
In some embodiments, the processing module comprises: a first processing unit, configured to perform normalization processing on the first audio digital signal and the second audio digital signal to obtain normalized first audio data and normalized second audio data, where the normalization processing includes: digital decoding and digital filtering; the adjusting unit is used for adjusting the amplitude of the first audio data at each preset sampling time point so that the amplitude of the adjusted first audio data at each preset sampling time point is matched with the amplitude of the second audio data at each preset sampling time point; and the second processing unit is used for canceling the echo signal in the second audio data through the first audio data after the amplitude adjustment.
In some embodiments, the adjusting unit is specifically configured to: adjusting the amplitude of first audio data at a first sampling time point by a first predetermined multiple so that the adjusted amplitude is the same as the amplitude of second audio data at the first sampling time point; adjusting the amplitudes of all the preset sampling time points corresponding to the first audio data according to the first preset multiple; sequentially comparing whether the difference value between the amplitude of the second audio data and the amplitude of the adjusted first audio data at each preset sampling time point is greater than or equal to zero from the first sampling time point; and under the condition that all the difference values are larger than or equal to zero, determining that the amplitude of the first audio data at each preset sampling time point after the amplitude adjustment according to the first preset multiple is matched with the amplitude of the second audio data at each preset sampling time point.
In some embodiments, the adjusting unit is further specifically configured to: in the case that a second sampling time point with a difference value smaller than zero exists, adjusting the amplitude of the first audio data at the second sampling time point by a second preset multiple so that the adjusted amplitude is the same as the amplitude of the second audio data at the second sampling time point; adjusting the amplitudes of all the preset sampling time points corresponding to the first audio data according to the second preset multiple; sequentially comparing whether the difference value between the amplitude of the second audio data and the amplitude of the adjusted first audio data at each preset sampling time point is greater than or equal to zero from the second sampling time point; and under the condition that all the difference values are larger than or equal to zero, determining that the amplitude of the first audio data at each preset sampling time point after the amplitude adjustment according to the second preset multiple is matched with the amplitude of the second audio data at each preset sampling time point.
In some embodiments, the adjusting unit is further specifically configured to: splitting the first audio data into a plurality of sub audio data according to a preset time period; and respectively adjusting the amplitude of each preset sampling time point corresponding to each sub-audio data.
In another aspect, an embodiment of the present disclosure provides an echo cancellation system, including: the system level chip, the digital-to-analog converter, the amplifier, the loudspeaker, the phase-locked loop, the digital signal processor, the analog-to-digital converter and the microphone; the digital-to-analog converter is connected with the system-on-chip and the amplifier and is used for processing a first audio digital signal from the system-on-chip into a first audio analog signal; the loudspeaker is connected with the amplifier and used for playing the amplified first audio analog signal; the phase-locked loop is connected with the system-on-chip and used for determining a frequency multiplication sampling clock of a second audio analog signal according to original sampling frequency information corresponding to the first audio digital signal, wherein the frequency multiplication sampling clock comprises: presetting sampling frequency and sampling start time, wherein the second audio analog signal is a first audio analog signal after air propagation; the analog-to-digital converter is connected with the microphone and the digital signal processor and is used for oversampling a second audio analog signal acquired by the microphone according to the frequency multiplication sampling clock to acquire a second audio digital signal; the digital signal processor is connected with the system-on-chip and used for receiving the first audio digital signal from the system-on-chip, receiving the second audio digital signal from the analog-to-digital converter, and performing echo cancellation processing on the second audio digital signal according to the first audio digital signal to obtain a processed sound digital signal.
According to the embodiment of the invention, the characteristics of the output signal of the loudspeaker are analyzed through the digital signal, the digital signal processor directly obtains the first audio digital signal as the basis of the digital processing process during the subsequent echo cancellation, and then determines the preset sampling frequency and the sampling start time through the original sampling frequency information corresponding to the first audio digital signal so as to realize clock synchronous sampling, and an oversampling mode is adopted, so that the authenticity of the second audio digital signal is ensured, and the accuracy of performing echo cancellation on the second audio digital signal by using the first audio digital signal is also ensured; the whole scheme directly carries out echo processing through first audio digital signal, and then removes analog signal sampling wire, knows and ties up digital signal processor and the structure that the speaker must be correlated with for power amplifier system and microphone pickup system work as two independent systems, and other products of two systems can nimble adaptations, and the limitation is lower.
Drawings
In order to more clearly illustrate the embodiments of the present disclosure or the technical solutions in the prior art, the drawings needed to be used in the description of the embodiments or the prior art will be briefly introduced below, it is obvious that the drawings in the following description are only some embodiments described in the present disclosure, and for those skilled in the art, other drawings can be obtained according to the drawings without creative efforts.
Fig. 1 is a schematic structural diagram of an echo cancellation system provided in the related art;
fig. 2 is a flowchart of an echo cancellation method according to a first embodiment of the disclosure;
fig. 3 is a first schematic structural diagram of an echo cancellation system according to a second embodiment of the present disclosure;
fig. 4 is a schematic structural diagram of an echo cancellation system according to a second embodiment of the present disclosure;
fig. 5 is a first schematic diagram of a sound spectrum provided by a second embodiment of the present disclosure;
fig. 6 is a time domain schematic diagram of a sound spectrum according to a second embodiment of the present disclosure;
fig. 7 is a time domain diagram of a sound spectrum according to a second embodiment of the present disclosure;
fig. 8 is a third time domain schematic diagram of a sound spectrum provided by a second embodiment of the present disclosure;
fig. 9 is a second schematic diagram of a sound spectrum provided by a second embodiment of the present disclosure;
fig. 10 is a schematic structural diagram of an echo cancellation device according to a third embodiment of the present disclosure.
Detailed Description
In order to make the objects, technical solutions and advantages of the embodiments of the present disclosure more clear, the technical solutions of the embodiments of the present disclosure will be described below clearly and completely with reference to the accompanying drawings of the embodiments of the present disclosure. It is to be understood that the described embodiments are only a few embodiments of the present disclosure, and not all embodiments. All other embodiments, which can be derived by a person skilled in the art from the described embodiments of the disclosure without any inventive step, are within the scope of protection of the disclosure.
Unless otherwise defined, technical or scientific terms used herein shall have the ordinary meaning as understood by one of ordinary skill in the art to which this disclosure belongs. The use of "first," "second," and similar terms in this disclosure is not intended to indicate any order, quantity, or importance, but rather is used to distinguish one element from another. The word "comprising" or "comprises", and the like, means that the element or item listed before the word covers the element or item listed after the word and its equivalents, but does not exclude other elements or items. The terms "connected" or "coupled" and the like are not restricted to physical or mechanical connections, but may include electrical connections, whether direct or indirect. "upper", "lower", "left", "right", and the like are used merely to indicate relative positional relationships, and when the absolute position of the object being described is changed, the relative positional relationships may also be changed accordingly.
To maintain the following description of the embodiments of the present disclosure clear and concise, a detailed description of known functions and known components have been omitted from the present disclosure.
As shown in fig. 1, in the echo canceling system of the related art, an SoC (system on chip) or other processor outputs a digital audio signal to a Decodec (decoding chip, i.e., digital-to-analog converter) through a 101 output digital audio data path, a common data bus includes I2C, PCM, etc., and the digital audio signal is decoded into an analog signal by a DAC inside the Decodec, and then the analog signal is output to a PA (power amplifier) through a 102 output analog audio path, amplified by the PA, amplified in power, and amplified by a 103 output analog audio path to drive a speaker to emit sound. 106MIC (microphone array) picks up sounds in the environment, not only sounds of a person speaking, but also speaker sounds introduced through 105 air paths, and this part of mixed sound is sent to Encodec (input encoding chip, i.e., analog-to-digital converter) through 107 input analog audio path, Encodec converts analog signals into digital signals through ADC, and encoded and then sent to DSP through 108 input digital audio data path. Meanwhile, the 111 loudspeaker output signal sampling wire can collect analog signals input into the loudspeaker in real time, digital signals are output after being processed by another Encodec and are also sent to the DSP, the DSP subtracts the signals collected by the 111 loudspeaker output signal sampling wire from the signals collected by the 106MIC to complete echo cancellation processing, namely, the environmental sounds required by loudspeaker sound reservation are eliminated.
Therefore, in the echo cancellation scheme of the related art, a wire is required to be connected between the loudspeaker and the DSP to acquire an analog signal, great difficulty is caused to the problem of deployment of the loudspeaker, the local structure of the equipment is limited, the designed echo cancellation architecture is poor in universality, and other equipment cannot be reused.
If the DSP needs to sample the loudspeaker output signal, the characteristics can be analyzed through the digital signal before power amplification, then the power amplification system (the upper branch in figure 1) and the microphone pickup system (the lower branch in figure 1) can communicate through a digital channel, so that an analog signal sampling lead is removed, the power amplification system and the microphone pickup system work as two independent systems, the two systems can be flexibly adapted to other products, and the design is not limited to be carried out as a whole.
Based on the above thought, the first embodiment of the present disclosure provides an echo cancellation method, the flow of which is shown in fig. 2, and the method includes steps S201 to S203:
s201, receiving a first audio digital signal from a system-on-chip, wherein the first audio digital signal is a digital signal corresponding to a first audio analog signal to be played by a loudspeaker;
s202, determining a frequency-doubling sampling clock of the second audio analog signal according to the original sampling frequency information corresponding to the first audio digital signal, and performing oversampling on the second audio analog signal acquired by the microphone according to the frequency-doubling sampling clock to obtain the second audio digital signal, where the frequency-doubling sampling clock includes: presetting sampling frequency and sampling start time, wherein the second audio analog signal is a first audio analog signal after air propagation;
and S203, performing echo cancellation processing on the received second audio digital signal according to the first audio digital signal to obtain a processed sound digital signal.
According to the embodiment of the invention, the characteristics of the output signal of the loudspeaker are analyzed through the digital signal, the digital signal processor directly obtains the first audio digital signal as the basis of the digital processing process during the subsequent echo cancellation, and then determines the preset sampling frequency and the sampling start time through the original sampling frequency information corresponding to the first audio digital signal so as to realize clock synchronous sampling, and an oversampling mode is adopted, so that the authenticity of the second audio digital signal is ensured, and the accuracy of performing echo cancellation on the second audio digital signal by using the first audio digital signal is also ensured; the whole scheme directly carries out echo processing through first audio digital signal, and then removes analog signal sampling wire, knows and ties up digital signal processor and the structure that the speaker must be correlated with for power amplifier system and microphone pickup system work as two independent systems, and other products of two systems can nimble adaptations, and the limitation is lower.
In the process of performing echo cancellation processing on a received second audio digital signal according to a first audio digital signal, firstly, normalization processing needs to be performed on the first audio digital signal and the second audio digital signal to obtain normalized first audio data and normalized second audio data, wherein the normalization processing generally includes processing processes such as digital decoding and digital filtering; secondly, the amplitude of the first audio data at each preset sampling time point needs to be adjusted, so that the amplitude of the adjusted first audio data at each preset sampling time point is matched with the amplitude of the second audio data at each preset sampling time point; finally, the echo signal in the second audio data is counteracted through the first audio data after the amplitude adjustment.
In the foregoing process, the process of adjusting the amplitude of the first audio data at each predetermined sampling time point may be a process of adjusting multiple times, and in a specific implementation, the basic adjusting process is as follows:
adjusting the amplitude of the first audio data at the first sampling time point by a first predetermined multiple so that the adjusted amplitude is the same as the amplitude of the second audio data at the first sampling time point; adjusting the amplitudes of all preset sampling time points corresponding to the first audio data according to a first preset multiple; sequentially comparing whether the difference value between the amplitude of the second audio data and the amplitude of the adjusted first audio data at each preset sampling time point is greater than or equal to zero from the first sampling time point; and under the condition that all the difference values are larger than or equal to zero, determining that the amplitude of the first audio data at each preset sampling time point after the amplitude adjustment according to the first preset multiple is matched with the amplitude of the second audio data at each preset sampling time point.
If the difference value of each predetermined sampling time point is greater than or equal to zero, it can be determined that the amplitude of the first audio data is adjusted to be very close to the acquired second audio data, and the echo in the second audio data can be cancelled by using the first audio data corresponding to the current amplitude, so that a clearer sound digital signal is obtained.
When the difference between the amplitude of the second audio data at each predetermined sampling time point and the amplitude of the adjusted first audio data is sequentially compared to be greater than or equal to zero, there may be a case where the difference is smaller than zero. This adjustment is to ensure that the first audio data more closely resembles the actual second audio digital signal. Subsequently, adjusting the amplitudes of all the preset sampling time points corresponding to the first audio data according to a second preset multiple; sequentially comparing whether the difference value between the amplitude of the second audio data and the amplitude of the adjusted first audio data at each preset sampling time point is greater than or equal to zero or not from the second sampling time point; under the condition that all the difference values are larger than or equal to zero, determining that the amplitude value of the first audio data at each preset sampling time point after amplitude value adjustment according to a second preset multiple is matched with the amplitude value of the second audio data at each preset sampling time point; finally, the echo in the second audio data is counteracted through the first audio data after the amplitude adjustment is carried out according to the second preset multiple.
In the actual implementation process, since the second audio analog signal is propagated through air, the amplitude is variable, and the adjustment process may be performed tens of times or hundreds of times, but the predetermined multiple of the adjustment of the predetermined sampling time point with the difference value of the last time being less than zero is used as the adjustment multiple of the whole first audio data no matter how many times the adjustment is performed.
The above process of the embodiment of the present disclosure is described by taking amplitude amplification of the first audio data as an example, and in a specific implementation, the amplitude of the second audio data may be reduced by a predetermined multiple, and no matter the amplitude of the first audio data is amplified or the amplitude of the second audio data is reduced, the amplitude of each predetermined sampling time point of the whole first audio data should not exceed the amplitude of each predetermined sampling time point of the second audio data, so as to ensure a better echo cancellation effect.
In order to reduce data processing amount and ensure a better echo cancellation effect, in the process of adjusting the amplitude of the first audio data at each predetermined sampling time point, the first audio data may be split into a plurality of sub audio data according to a predetermined time period, and then the amplitudes of the predetermined sampling time points corresponding to the sub audio data are respectively adjusted, that is, the amplitudes of the first audio data at the predetermined sampling time points are all performed in a segmented manner, and after the amplitude adjustment of the first sub audio data is completed, the adjustment multiple of the amplitude of the second sub audio data is unrelated to the first sub audio data.
Based on the foregoing embodiments, a second embodiment of the present disclosure further provides an echo cancellation system, a schematic structure of which is shown in fig. 3, and includes:
the system level chip comprises a system level chip 1, a digital-to-analog converter 2, an amplifier 3, a loudspeaker 4, a phase-locked loop 5, a digital signal processor 6, an analog-to-digital converter 7 and a microphone 8; wherein,
the digital-to-analog converter is connected with the system-on-chip and the amplifier and is used for processing the first audio digital signal from the system-on-chip into a first audio analog signal;
the loudspeaker is connected with the amplifier and used for playing the amplified first audio analog signal;
the phase-locked loop is connected with the system-on-chip and used for determining a frequency multiplication sampling clock of the second audio analog signal according to original sampling frequency information corresponding to the first audio digital signal, wherein the frequency multiplication sampling clock comprises: presetting sampling frequency and sampling start time, wherein the second audio analog signal is a first audio analog signal after air propagation;
the analog-to-digital converter is connected with the microphone and the digital signal processor and is used for oversampling a second audio analog signal acquired by the microphone according to a frequency multiplication sampling clock to obtain a second audio digital signal;
and the digital signal processor is connected with the system-on-chip and used for receiving the first audio digital signal from the system-on-chip, receiving the second audio digital signal from the analog-to-digital converter and carrying out echo cancellation processing on the second audio digital signal according to the first audio digital signal so as to obtain a processed sound digital signal.
The echo cancellation system shown in fig. 3 of the present disclosure realizes the functions that can be realized in the DSP in the first embodiment by using a phase-locked loop, but it is only an implementation manner, and the structure of the echo cancellation system in the second embodiment may also be as shown in fig. 4, that is, the phase-locked loop is removed, and the functions of the phase-locked loop are integrated in the digital signal processor, which is not limited herein.
The above-described embodiments are exemplified below with reference to the structure of fig. 3 and other drawings.
As shown in fig. 3, the embodiment of the present disclosure eliminates 111 a speaker output signal sampling wire in fig. 1, simultaneously sends digital audio to be output by a speaker to a DSP through a digital interface, adds a PLL (clock phase locked loop) circuit, generates 101 a frequency-multiplied clock (i.e., determines a predetermined sampling frequency and a sampling start time) of an output digital audio data clock, and synchronously oversamples sounds picked up by a MIC (microphone).
In fig. 3, 101 output digital audio data (i.e. the first audio digital signal) is a digital signal, and the represented sound spectrum is a discretization characteristic, as shown by a solid line in fig. 5, it is a single straight line, and after decoding by the digital-to-analog converter, it becomes 102 output analog audio, which is an analog signal, because of low-pass filtering in the decoding process, its time domain signal is as shown in fig. 6, and in fig. 6, time t 1-time t6 represents the sampling time of the original digital audio data, so only at the sampling time, it is the real original signal, the signal of the sampling interval is the signal recovered by the low-pass filtering system, and this part of the recovered signal is relatively smooth and is also the maximum difference point with the real environmental sound.
The real world signal has rich frequency, the frequency spectrum is continuous and is a continuous curve, the time domain characteristic is shown in fig. 7, and a sawtooth waveform exists in the sampling interval and contains any frequency component.
And the digital audio data required to be output by the loudspeaker is sent to a PLL (phase locked loop), and because the digital audio data has original sampling frequency information inside, a frequency multiplication clock synchronous with the original adopted clock is generated after frequency multiplication processing, and the frequency multiplication clock is used as a new sampling clock to oversample the sound picked up by the MIC. As shown in fig. 8, the number of samples is increased in the original sampling interval, T1, T2, T3, and the like are the sampling times at which 101 outputs digital audio data, and T1, T2, T3, and the like are the sampling times of the mixed signal picked up for the MIC.
The spectrum interval of the samples of the 101 output digital audio data is the interval shown by the solid line in fig. 5, the interval is relatively large, and the data on the sampling points is real data, the 108 input digital audio data (i.e., the second audio digital signal) is the interval shown by the dotted line in fig. 5 because of oversampling, the spectrum interval is the interval shown by the dotted line in fig. 5, the frequency points not only include the solid line frequency points, but also the frequency points between the solid lines, the difference between the frequency points is small, and the data on the frequency points is real sound data.
Since the clock for inputting the digital audio data 108 is generated by the clock for outputting the digital audio data 101, the two data have the same polarity, and therefore, they can be put together at the same time for comparison. After digital decoding and digital filtering, 101 output digital audio data received by the DSP is shown by a thick solid line in fig. 9, and 108 input digital audio data received by the DSP is shown by a thick dotted line in fig. 9, because sound picked up by the MIC is both speaker sound and ambient sound, but speaker sound is output after amplification, the amplitude of the 108 input digital audio data spectrum is larger than that of 101 output digital audio data, and 101 output digital audio data is only represented by normalized energy after being processed by the DSP, that is, the energy amplitudes in the two spectra represent different meanings, and before subtraction, the two spectra have to be unified in amplitude meaning, that is, amplitude alignment.
The frequency characteristics of the spectrum curve of the analog signal without special filtering amplification treatment are consistent with the input signal, but the amplification factor of the amplitude is different, and the processes of adjusting the amplitude of the 101 output digital audio data and the 108 input digital audio data to match are as follows (1) - (10):
(1) the first 101 frequency point for outputting digital audio data is taken as a comparison frequency point, i.e. the first solid line frequency point in fig. 5, and the amplitudes of the frequency points at the same positions of the input digital audio data are compared 108.
(2) The 101 output digital audio data amplitude is scaled to be equal to the 108 input digital audio data amplitude at the frequency point, and the scaling factor X1 is recorded.
(3) The remaining bin magnitudes of the 101 output digital audio data are all scaled by X1 times.
(4) Comparing the amplitudes of all frequency points of the 108 input digital audio data one by one, and subtracting the amplitudes of the frequency points corresponding to the 101 output digital audio data, wherein the amplitudes of the corresponding frequency points are obtained by digital filtering fitting between the 101 output digital audio data frequency points.
(5) If the subtraction value is greater than or equal to zero, then the scaling factor X1 is correct; if a negative number occurs, it is determined that the scaling factor is erroneous.
(6) And selecting the second frequency point 101 for outputting the digital audio data as a comparison frequency point, repeating the comparison and scaling operation, recording the scaling multiple X2, and judging whether the frequency point is correct or not.
(7) And sequentially comparing the frequency points to the last 101 output digital audio data frequency point or the nth frequency point.
(8) And taking the latest Xn as the final scaling multiple.
(9) All frequency points of 101 output digital audio data are scaled by Xn.
(10) The curve of the 101 output digital audio data scaled by Xn is subtracted from the curve of the 108 input digital audio data spectrum, as shown in fig. 9, and the difference is the ambient sound for eliminating the influence of the speaker sound, thereby completing the echo cancellation operation.
The frequency characteristics of the frequency spectrum curve of the analog signal subjected to special filtering amplification processing of sound in a certain frequency band are changed with input, and echo cancellation can be completed only by dividing the frequency spectrum of the 101 output digital audio data according to special filtering frequency points, dividing the frequency spectrum into corresponding frequency bands and executing the operation in each frequency band.
The third embodiment of the present disclosure also provides an echo cancellation device, a structural schematic of which is shown in fig. 10, including:
the receiving module 10 is configured to receive a first audio digital signal from a system-on-chip, where the first audio digital signal is a digital signal corresponding to a first audio analog signal to be played by a speaker; a determining module 20, coupled to the receiving module 10, configured to determine a frequency-doubled sampling clock of the second audio analog signal according to original sampling frequency information corresponding to the first audio digital signal, so as to perform oversampling on the second audio analog signal acquired by the microphone according to the frequency-doubled sampling clock to obtain the second audio digital signal, where the frequency-doubled sampling clock includes: presetting sampling frequency and sampling start time, wherein the second audio analog signal is a first audio analog signal after air propagation; and the processing module 30 is coupled to the determining module 20, and configured to perform echo cancellation processing on the second audio digital signal according to the first audio digital signal to obtain a processed sound digital signal.
The processing module may include: the first processing unit is configured to perform normalization processing on the first audio digital signal and the second audio digital signal to obtain normalized first audio data and normalized second audio data, where the normalization processing includes: digital decoding and digital filtering; the adjusting unit is used for adjusting the amplitude of the first audio data at each preset sampling time point so that the amplitude of the adjusted first audio data at each preset sampling time point is matched with the amplitude of the second audio data at each preset sampling time point; and the second processing unit is used for canceling the echo signal in the second audio data through the first audio data after the amplitude adjustment.
In a specific implementation, the process of adjusting the amplitude of the first audio data at each predetermined sampling time point may be a process of adjusting multiple times, and in a specific implementation, the adjusting unit may specifically be configured to: adjusting the amplitude of the first audio data at the first sampling time point by a first predetermined multiple so that the adjusted amplitude is the same as the amplitude of the second audio data at the first sampling time point; adjusting the amplitudes of all preset sampling time points corresponding to the first audio data according to a first preset multiple; sequentially comparing whether the difference value between the amplitude of the second audio data and the amplitude of the adjusted first audio data at each preset sampling time point is greater than or equal to zero from the first sampling time point; and under the condition that all the difference values are larger than or equal to zero, determining that the amplitude of the first audio data at each preset sampling time point after the amplitude adjustment according to the first preset multiple is matched with the amplitude of the second audio data at each preset sampling time point.
If the difference value of each predetermined sampling time point is greater than or equal to zero, it can be determined that the amplitude of the first audio data is adjusted to be very close to the acquired second audio data, and the echo in the second audio data can be cancelled by using the first audio data corresponding to the current amplitude, so that a clearer sound digital signal is obtained.
When the difference between the amplitude of the second audio data at each predetermined sampling time point and the amplitude of the adjusted first audio data is sequentially compared to be greater than or equal to zero, there may be a case where the difference is smaller than zero, and at this time, because each predetermined sampling time point is sequentially compared, the adjusting unit may be further configured to:
in the case that there is a second sampling time point whose difference is less than zero, adjusting the amplitude of the first audio data at the second sampling time point by a second predetermined multiple so that the adjusted amplitude is the same as the amplitude of the second audio data at the second sampling time point; adjusting the amplitudes of all the preset sampling time points corresponding to the first audio data according to a second preset multiple; sequentially comparing whether the difference value between the amplitude of the second audio data and the amplitude of the adjusted first audio data at each preset sampling time point is greater than or equal to zero from the second sampling time point; and under the condition that all the difference values are larger than or equal to zero, determining that the amplitude of the first audio data at each preset sampling time point after the amplitude adjustment according to the second preset multiple is matched with the amplitude of the second audio data at each preset sampling time point.
In the actual implementation process, since the second audio analog signal is propagated through air, the amplitude is variable, and the adjustment process may be performed tens of times or hundreds of times, but the predetermined multiple of the adjustment of the predetermined sampling time point with the difference value of the last time being less than zero is used as the adjustment multiple of the whole first audio data no matter how many times the adjustment is performed.
The above process of the embodiment of the present disclosure is described by taking amplitude amplification of the first audio data as an example, and in a specific implementation, the amplitude of the second audio data may be reduced by a predetermined multiple, and no matter the amplitude of the first audio data is amplified or the amplitude of the second audio data is reduced, the amplitude of each predetermined sampling time point of the whole first audio data should not exceed the amplitude of each predetermined sampling time point of the second audio data, so as to ensure a better echo cancellation effect.
In order to reduce the data processing amount and ensure a better echo cancellation effect, the adjusting unit may be further configured to: splitting the first audio data into a plurality of sub audio data according to a preset time period; and respectively adjusting the amplitude of each preset sampling time point corresponding to each sub-audio data. That is, the adjustment of the amplitude of the first audio data at each predetermined sampling time point is performed in a segmented manner, and after the adjustment of the amplitude of the first sub audio data is completed, the adjustment multiple of the amplitude of the second sub audio data is independent of the first sub audio data.
According to the embodiment of the invention, the characteristics of the output signal of the loudspeaker are analyzed through the digital signal, the digital signal processor directly obtains the first audio digital signal as the basis of the digital processing process during the subsequent echo cancellation, and then determines the preset sampling frequency and the sampling start time through the original sampling frequency information corresponding to the first audio digital signal so as to realize clock synchronous sampling, and an oversampling mode is adopted, so that the authenticity of the second audio digital signal is ensured, and the accuracy of performing echo cancellation on the second audio digital signal by using the first audio digital signal is also ensured; the whole scheme directly carries out echo processing through first audio digital signal, and then removes analog signal sampling wire, knows and ties up digital signal processor and the structure that the speaker must be correlated with for power amplifier system and microphone pickup system work as two independent systems, and other products of two systems can nimble adaptations, and the limitation is lower.
Moreover, although exemplary embodiments have been described herein, the scope thereof includes any and all embodiments based on the disclosure with equivalent elements, modifications, omissions, combinations (e.g., of various embodiments across), adaptations or alterations. The elements of the claims are to be interpreted broadly based on the language employed in the claims and not limited to examples described in the present specification or during the prosecution of the application, which examples are to be construed as non-exclusive. It is intended, therefore, that the specification and examples be considered as exemplary only, with a true scope and spirit being indicated by the following claims and their full scope of equivalents.
The above description is intended to be illustrative and not restrictive. For example, the above-described examples (or one or more versions thereof) may be used in combination with each other. For example, other embodiments may be used by those of ordinary skill in the art upon reading the above description. In addition, in the foregoing detailed description, various features may be grouped together to streamline the disclosure. This should not be interpreted as an intention that a disclosed feature not claimed is essential to any claim. Rather, the subject matter of the present disclosure may lie in less than all features of a particular disclosed embodiment. Thus, the following claims are hereby incorporated into the detailed description as examples or embodiments, with each claim standing on its own as a separate embodiment, and it is contemplated that these embodiments may be combined with each other in various combinations or permutations. The scope of the disclosure should be determined with reference to the appended claims, along with the full scope of equivalents to which such claims are entitled.
While the present disclosure has been described in detail with reference to the embodiments, the present disclosure is not limited to the specific embodiments, and those skilled in the art can make various modifications and alterations based on the concept of the present disclosure, and the modifications and alterations should fall within the scope of the present disclosure as claimed.
Claims (10)
1. An echo cancellation method, comprising:
receiving a first audio digital signal from a system-on-chip, wherein the first audio digital signal is a digital signal corresponding to a first audio analog signal to be played by a loudspeaker;
determining a frequency multiplication sampling clock of a second audio analog signal according to the original sampling frequency information corresponding to the first audio digital signal, and performing oversampling on the second audio analog signal acquired by a microphone according to the frequency multiplication sampling clock to obtain a second audio digital signal, wherein the frequency multiplication sampling clock comprises: presetting sampling frequency and sampling start time, wherein the second audio analog signal is a first audio analog signal after air propagation;
and performing echo cancellation processing on the received second audio digital signal according to the first audio digital signal to obtain a processed sound digital signal.
2. The echo cancellation method of claim 1, wherein said performing echo cancellation processing on said received second audio digital signal based on said first audio digital signal comprises:
normalizing the first audio digital signal and the second audio digital signal to obtain normalized first audio data and normalized second audio data, wherein the normalizing comprises: digital decoding and digital filtering;
adjusting the amplitude of the first audio data at each preset sampling time point so that the amplitude of the adjusted first audio data at each preset sampling time point is matched with the amplitude of the adjusted second audio data at each preset sampling time point;
and cancelling the echo signal in the second audio data through the first audio data after the amplitude adjustment.
3. The echo cancellation method of claim 2, wherein said adjusting the amplitude of the first audio data at each of the predetermined sampling time points such that the amplitude of the adjusted first audio data at each of the predetermined sampling time points matches the amplitude of the adjusted second audio data at each of the predetermined sampling time points comprises:
adjusting the amplitude of the first audio data at a first sampling time point by a first predetermined multiple so that the adjusted amplitude is the same as the amplitude of the second audio data at the first sampling time point;
adjusting the amplitudes of all the preset sampling time points corresponding to the first audio data according to the first preset multiple;
sequentially comparing whether the difference value between the amplitude of the second audio data and the amplitude of the adjusted first audio data at each preset sampling time point is greater than or equal to zero from the first sampling time point;
and under the condition that all the difference values are larger than or equal to zero, determining that the amplitude of the first audio data at each preset sampling time point after the amplitude adjustment according to the first preset multiple is matched with the amplitude of the second audio data at each preset sampling time point.
4. The echo cancellation method of claim 3, wherein said sequentially comparing whether the difference between the amplitude of the second audio data and the amplitude of the adjusted first audio data at each predetermined sampling time point is greater than or equal to zero further comprises:
in the case that there is a second sampling time point whose difference is smaller than zero, adjusting the amplitude of the first audio data at the second sampling time point by a second predetermined multiple so that the adjusted amplitude is the same as the amplitude of the second audio data at the second sampling time point;
adjusting the amplitudes of all the preset sampling time points corresponding to the first audio data according to the second preset multiple;
sequentially comparing whether the difference value between the amplitude of the second audio data and the amplitude of the adjusted first audio data at each preset sampling time point is greater than or equal to zero from the second sampling time point;
and under the condition that all the difference values are larger than or equal to zero, determining that the amplitude of the first audio data at each preset sampling time point after the amplitude adjustment according to the second preset multiple is matched with the amplitude of the second audio data at each preset sampling time point.
5. The echo cancellation method of any one of claims 2 through 4, wherein said adjusting the amplitude of the first audio data at each predetermined sampling time point comprises:
splitting the first audio data into a plurality of sub audio data according to a preset time period;
and respectively adjusting the amplitude of each preset sampling time point corresponding to each sub-audio data.
6. An echo cancellation device, comprising:
the receiving module is used for receiving a first audio digital signal from a system-on-chip, wherein the first audio digital signal is a digital signal corresponding to a first audio analog signal to be played by a loudspeaker;
a determining module, configured to determine a frequency-doubled sampling clock of a second audio analog signal according to original sampling frequency information corresponding to the first audio digital signal, so as to perform oversampling on the second audio analog signal acquired by a microphone according to the frequency-doubled sampling clock to obtain a second audio digital signal, where the frequency-doubled sampling clock includes: presetting sampling frequency and sampling start time, wherein the second audio analog signal is a first audio analog signal after air propagation;
and the processing module is used for carrying out echo cancellation processing on the received second audio digital signal according to the first audio digital signal so as to obtain a processed sound digital signal.
7. The echo cancellation device of claim 6, wherein the processing module comprises:
a first processing unit, configured to perform normalization processing on the first audio digital signal and the second audio digital signal to obtain normalized first audio data and normalized second audio data, where the normalization processing includes: digital decoding and digital filtering;
the adjusting unit is used for adjusting the amplitude of the first audio data at each preset sampling time point so that the amplitude of the adjusted first audio data at each preset sampling time point is matched with the amplitude of the adjusted second audio data at each preset sampling time point;
and the second processing unit is used for canceling the echo signal in the second audio data through the first audio data after the amplitude adjustment.
8. The echo cancellation device of claim 7, wherein the adjustment unit is specifically configured to:
adjusting the amplitude of the first audio data at a first sampling time point by a first predetermined multiple so that the adjusted amplitude is the same as the amplitude of the second audio data at the first sampling time point;
adjusting the amplitudes of all the preset sampling time points corresponding to the first audio data according to the first preset multiple;
sequentially comparing whether the difference value between the amplitude of the second audio data and the amplitude of the adjusted first audio data at each preset sampling time point is greater than or equal to zero from the first sampling time point;
and under the condition that all the difference values are larger than or equal to zero, determining that the amplitude of the first audio data at each preset sampling time point after the amplitude adjustment according to the first preset multiple is matched with the amplitude of the second audio data at each preset sampling time point.
9. The echo cancellation device of claim 8, wherein the adjustment unit is further specifically configured to:
in the case that there is a second sampling time point whose difference is smaller than zero, adjusting the amplitude of the first audio data at the second sampling time point by a second predetermined multiple so that the adjusted amplitude is the same as the amplitude of the second audio data at the second sampling time point;
adjusting the amplitudes of all the preset sampling time points corresponding to the first audio data according to the second preset multiple;
sequentially comparing whether the difference value between the amplitude of the second audio data and the amplitude of the adjusted first audio data at each preset sampling time point is greater than or equal to zero from the second sampling time point;
and under the condition that all the difference values are larger than or equal to zero, determining that the amplitude of the first audio data at each preset sampling time point after the amplitude adjustment according to the second preset multiple is matched with the amplitude of the second audio data at each preset sampling time point.
10. An echo cancellation system, comprising:
the system level chip, the digital-to-analog converter, the amplifier, the loudspeaker, the phase-locked loop, the digital signal processor, the analog-to-digital converter and the microphone; wherein,
the digital-to-analog converter is connected with the system-on-chip and the amplifier and is used for processing a first audio digital signal from the system-on-chip into a first audio analog signal;
the loudspeaker is connected with the amplifier and used for playing the amplified first audio analog signal;
the phase-locked loop is connected with the system-on-chip and used for determining a frequency multiplication sampling clock of a second audio analog signal according to original sampling frequency information corresponding to the first audio digital signal, wherein the frequency multiplication sampling clock comprises: presetting sampling frequency and sampling start time, wherein the second audio analog signal is a first audio analog signal after air propagation;
the analog-to-digital converter is connected with the microphone and the digital signal processor and is used for oversampling a second audio analog signal acquired by the microphone according to the frequency multiplication sampling clock to acquire a second audio digital signal;
the digital signal processor is connected with the system-on-chip and used for receiving the first audio digital signal from the system-on-chip, receiving the second audio digital signal from the analog-to-digital converter, and performing echo cancellation processing on the second audio digital signal according to the first audio digital signal to obtain a processed sound digital signal.
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