Method and equipment for processing digital audio in variable speed

Abstract

The invention relates to the audio signal processing technology and discloses a method and equipment for processing digital audio in variable speed. In the invention, a pair of perfect reconstructing window functions WL and WR with amplitude attenuation and increase characteristics is used to act on original digital audio according to different delays to obtain a pair of windowing data; and an audio waveform is reconstructed by using the windowing data to obtain audio after variable-speed process. The defection on a fundamental tone period and the relativity of the audio and the time-frequency conversion are avoided, so that the calculation amount is extremely small. In addition, the playing time of the contents is prolonged or shortened by using the compaction and the introduction of theself waveform of an audio signal without changing the audio waveform, so that the original tone quality can be better maintained.

Description

technical field [0001] The invention relates to audio signal processing technology, in particular to audio frequency variable speed processing in audio signal processing technology. Background technique [0002] In various multimedia applications, the playback speed adjustment of recorded digital audio has a wide demand. For example, reducing the sound speed of voice playback can help people with hearing or understanding impairments and foreign language beginners improve the intelligibility of listening; while increasing the sound speed can help listeners save time in obtaining information from recordings. In addition, the adjustment of the sound speed of music playback can also change the rhythm of the music to obtain unique effects; for the accompanying sound in the video, the adjustment of the sound speed of the audio playback can make the video speed up or slow down at the same time, so that the audience can Hear synchronous, undistorted audio. [0003] However, direct...

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Problems solved by technology

[0004] However, the inventors of the present invention have found that in the algorithm implemented based on the overlap add technique, it is necessary to determine the time delay of the overlapping window by detecting the similarity of the waveform (waveform similarity), and this type of method can only deal with Speech with a clear pitch period, it uses cross-correlation detection technology in the time domain or frequency domain to obtain the delay of similar waveforms as the delay of overlapping windows, so the calculation consumes a lot and the processing sound quality is average; while based on time-frequency transformation and The spectrum processing algorithm can process general audio including speech and music. It resamples the original digital audio to change the sampling rate, and then converts the digital audio with the changed sampling rate to the frequency domain to obtain the audio spectrum, and then Frequency shift is performed on the spectrum, and the processed spectrum is transformed back to the time domain. This algorithm is generally implemented using a short time Fourier transform that can be perfectly reconstructed. In order to obtain higher processing Sound quality, the short-time Fourier transform needs to process a long audio at a time. Although this method can obtain better processing sound quality, it has a large amount of calculation and storage. On handheld and mobile devices, due to the computing power and Constrained by power consumption, the algorithm is almost impossible to achieve

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