Method for encoding and decoding audio

Abstract

The invention discloses a method for coding and decoding an audio, which mainly solves the problems that the prior method for coding and decoding the audio has low compression ratio and poor reconstruction audio quality. The method comprises the following steps: adopting a time frequency translation and frequency domain filtering method or a time domain filtering and time frequency translation method to analyze an audio signal so as to obtain a frequency domain residual error signal; dividing the frequency domain residual error signal into a low frequency residual error signal and a high frequency residual error signal, and performing direct coding on the low frequency residual error signal and performing parameter coding on the high frequency residual error signal respectively; reconstructing the high frequency residual error signal through the decoded low frequency residual error signal and the decoded high frequency residual error; recombining the decoded low frequency residual error signal and the reconstructed high frequency residual error signal to obtain a reconstructed frequency domain residual error signal; and finally adopting a frequency domain inverse filtering and time frequency inverse transform method or a time frequency inverse transform and time domain inverse filtering method to obtain a reconstructed audio signal. The method eliminates the redundancy in the frequency domain residual error signal and improves the compression ratio of audio coding, the channel utilization rate and the audio transmission quality, and is applied to multimedia communication and consumer electronics.

Description

technical field [0001] The invention relates to the technical field of audio coding and decoding, in particular to a signal reconstruction method based on high-frequency residual reconstruction, which is used in the fields of multimedia communication and consumer electronics. Background technique [0002] In the field of multimedia communication, audio including voice, especially broadband audio, has gradually become one of the main communication services. However, the audio signal has a wide frequency band and a large amount of encoded data, which brings great difficulties to the real-time transmission and effective storage of the audio signal. Although audio coding algorithms such as MP3, AAC, EAAC, and EAAC+ have been able to compress and encode audio signals well and meet the requirements of certain applications, they are still not well qualified for mobile multimedia communications and various mobile platforms that are currently developing. business. So it is necessar...

AI Technical Summary

Problems solved by technology

[0008] In the above two existing technologies, regardless of whether the low-frequency sub-band is periodically shifted or folded as a whole block according to the prior art 1, or the frequency multiplication is reproduced according to the prior art 2, the harmonics are mechanically restored , without considering the diversity and variability of audio and speech signals. In addition, when copying, the sub-bands are extracted and copied in sequence. Since the waveforms of the low-frequency sub-bands and high-frequency sub-bands are inherently different, the copied high-frequency sub-bands are the same as the original Compared with the high-frequency sub-band, there will be a large waveform difference or peak difference, so the accuracy of the reconstructed high-frequency signal is not too high, which will affect the quality of the reconstructed audio signal

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