Audio processing method and device, equipment and storage medium
An audio processing and audio signal technology, applied in voice analysis, instruments, etc., can solve the problems of occupation, large audio signal data volume, large transmission bandwidth, etc., and achieve the effect of ensuring call quality, reducing transmission bandwidth, and reducing quantity
Pending Publication Date: 2022-04-05
TENCENT TECH (SHENZHEN) CO LTD
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[0003] However, when the audio signal is encoded in the above manner, the obtained encoded audio signal has a large amount of data, so when it is transmitted to the receiving end, it will occupy a relatively large transmission bandwidth
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[0122] Optionally, in the above step S503, the embodiment of the present application provides an implementation manner of performing interpolation processing on multi-frame sampled audio signals in the frequency domain, specifically as follows:
[0123] For the multi-frame sampled audio signal, perform the following steps respectively: respectively perform Fourier transform on a frame sampled audio signal and the next frame sampled audio signal of the frame sampled audio signal, and obtain the first real power corresponding to the frame sampled audio signal spectrum and the second real power spectrum corresponding to the sampled audio signal of the next frame. Then, based on the first real power spectrum and the second real power spectrum, a predicted power spectrum corresponding to at least one frame of the compensated audio signal between the sampled audio signal of the frame and the sampled audio signal of the next frame is determined. Inverse Fourier transform is performed...
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The embodiment of the invention provides an audio processing method and device, equipment and a storage medium, and relates to the technical field of artificial intelligence, in the method, a sending end extracts a frame of sampling audio signals every N frames of original audio signals, and extracts the original audio signals with audio features meeting feature change conditions as the sampling audio signals, and the characteristics of the original audio are reserved under the condition of realizing interval frame extraction. The sending end encodes the sampled audio signals and then sends the encoded sampled audio signals to the receiving end, compared with a continuous encoding and transmission mode, the number of the encoded and transmitted audio signals is reduced, and the transmission bandwidth of the audio signals is effectively reduced. Secondly, a receiving end performs interpolation processing on the multi-frame sampling audio signals to obtain multi-frame interpolation audio signals; according to the invention, each predicted audio compensation value is adopted to carry out audio compensation on the corresponding interpolation audio signal, so that the obtained target audio signal is close to the original audio signal as much as possible, and the call quality under the condition of interval frame extraction is ensured.
Description
technical field [0001] The embodiments of the present invention relate to the technical field of artificial intelligence, and in particular, to an audio processing method, device, equipment, and storage medium. Background technique [0002] In the voice call application, the sending end collects the analog audio signal through the microphone, then converts the analog audio signal into a digital audio signal through the analog-to-digital conversion circuit, and encodes the digital audio signal through the encoder. Finally, the encoded audio signal is transmitted through the communication network The audio signal is sent to the receiver. Correspondingly, the receiving end decodes the received coded audio signal through the decoder to regain the digital audio signal, and finally plays the digital audio signal through the loudspeaker. [0003] However, when the audio signal is encoded in the above manner, the obtained encoded audio signal has a relatively large amount of data, ...
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IPC IPC(8): G10L21/045
Inventor 梁俊斌
Owner TENCENT TECH (SHENZHEN) CO LTD