Method and device for improving audio quality, and medium

Abstract

The invention discloses a method and device for improving audio quality and a readable storage medium. The method comprises the following steps: receiving audio data, and storing the audio data in a CPU cache; judging whether the CPU cache has a residual space or not; in response to the fact that the CPU cache does not have the residual space, storing the audio data into an SSD cache; judging whether the total voice flow stored in the SSD cache is smaller than a first threshold value of network link transmission voice or not; in response to the fact that the total voice flow stored in the SSDcache is smaller than a first threshold value of network link transmission voice, judging whether the time delay is smaller than a second threshold value or not; and in response to the fact that the time delay is not smaller than the second threshold, reducing the SSD cache size to reduce the time delay. According to the method and device for improving the audio quality and the medium provided bythe invention, the multi-level cache is designed, and the size of the secondary cache is dynamically adjusted according to the voice time delay, the uplink and downlink voice flow and the voice compression condition, so that the best overall performance of the voice jitter quality and the voice time delay is ensured, and the audio quality is improved to the maximum extent.

Description

technical field [0001] The present invention relates to the field of signal transmission, more specifically, to a method, device and readable medium for improving audio quality. Background technique [0002] In video conferencing systems, audio quality is critical. When voice data is transmitted in the network environment, there may be a delay, and the delay of the network will also change in different periods of time. This change in network delay will cause jitter, and the jitter will cause audio quality distortion and intermittent . Therefore, all video conferencing systems need to perform audio anti-jitter processing. The traditional audio anti-jitter technology is to build a buffer area to save data for a long enough time, so that the slowest data packets can arrive in time before the audio is restored, thereby eliminating the adverse effects of delay. The larger the amount of data in the buffer area , the greater the degree of data jitter the receiving end can withst...

AI Technical Summary

Problems solved by technology

The traditional audio anti-jitter technology is to build a buffer area to save data for a long enough time, so that the slowest data packets can arrive in time before the audio is restored, thereby eliminating the adverse effects of delay. The larger the amount of data in the buffer area , the greater the degree of data jitter the receiving end can withstand, but the greater the corresponding delay

Therefore, although the traditional voice anti-jitter method solves the audio jitter elimination, it increases the fixed processing delay time, and cannot adjust the delay time with the jitter change of the network data.

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