Sub-band adaptive feedback elimination method and system based on step length control
An adaptive, step-size technology, applied in frequency response correction, transducer acoustic response prevention, speech analysis, etc., can solve problems such as insufficient sound transmission gain, poor convergence effect of adaptive feedback elimination, easy to generate howling, etc. , to achieve the effect of improving processing efficiency, reducing calculation amount, and reducing strong correlation
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Embodiment 1
[0057] Compared with the existing adaptive feedback elimination technology, this embodiment proposes a sub-band adaptive feedback elimination method based on step size control. By improving the compensation control, the step size can be reduced in the local speech process, and the When the local speech stops, the step size will be increased in the short reverberation period to improve the convergence performance of the adaptation.
[0058] Specific as figure 1 As shown, the method of the present embodiment includes the following steps:
[0059] Step 1, acquire the microphone pickup signal d(n) and the speaker output delay signal x(n).
[0060]In this embodiment, the signal collected by the microphone is converted into a digital audio signal, that is, the sound pickup signal d(n) by the microphone, where d(n)=s(n)+r(n). Among them, s(n) is the local language, and r(n) is the feedback signal.
[0061] In this embodiment, the speaker output signal y(n) is delayed by one frame ...
Embodiment 2
[0090] This embodiment proposes a sub-band adaptive feedback elimination system based on step size control, specifically as figure 2 As shown, the system of this embodiment includes a signal acquisition module, a signal pre-emphasis processing module, a signal subband filtering module, an FFT module, a frequency domain filtering module, an updating module, an IFFT module, a subband synthesis filtering module, a signal deemphasis module and frequency shifting module;
[0091] The signal acquisition module is used to acquire the microphone pick-up signal d(n) and the speaker output delay signal x(n); the specific processing process is the same as the above-mentioned embodiment 1.
[0092] The signal d(n) passes through the signal pre-emphasis processing module, the signal sub-band filter module and the FFT module in turn to obtain the frequency domain D(f, k) after pre-emphasis processing, sub-band filter processing and FFT conversion; the signal x(n) in turn After pre-emphas...
Embodiment 3
[0098] The present embodiment is the specific application of embodiment 1 and embodiment 2, and its hardware system consists of a microphone, a loudspeaker, and a DSP audio processing board, according to the attached image 3 Explain that the system is in a room, and the speaker is reflected to the microphone through the room, forming a loop.
[0099] The microphone picks up the local voice and converts it into a digital signal through the AD circuit to obtain d(n), and the speaker output signal y(n) is delayed to obtain x(n).
[0100] The audio system samples at 48K, and then the software downsamples to 16K processing algorithm. The number of data buffer points in one frame is 64 points, the number of FFT transformation points is 256 points, the trailing processing takes 256ms, and the system is divided into 4 sub-bands for calculation.
[0101] The specific implementation process is as follows:
[0102] 1. Perform pre-emphasis processing on x(n) and d(n) respectively, and t...
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