Method for fast code changing in large range to audio information to break virus

A wide range of audio information technology, applied in speech analysis, instruments, electrical components, etc., can solve problems such as undetectable and intranet attacks
CN101394402AInactive Publication Date: 2009-03-25้‚“ๅญฆ้”‹
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Patent Information

Authority / Receiving Office
CN ยท China
Current Assignee / Owner
้‚“ๅญฆ้”‹
Publication Date
2009-03-25
Estimated Expiration
Not applicable ยท inactive patent

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Abstract

The invention discloses a method for rapidly code-variable destroying virus at large range for audio information. The method comprises the following steps: respectively decoding a mp2 audio frame to bit allocation information, scaling factor information and sample code information according to the standard of iso11171-3, subjecting the bit allocation, the scaling factor and the sample code to unified code-variable re-calculation to change all bits of each coded data frame except the initial 20 bits of the mp2 audio information, and coding storage bit allocation information, quantification factor selection information, quantification factor information and sub-band quantification code information after the re-calculation together with an original header and additional information to generate a frame format. The method can achieve large comprehensive variable coding rate (up to 98.57%) of each frame, and the variable coding rate of the audio coding section is up to 100%.
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Description

technical field

[0001] The invention provides a method for rapidly transcoding and destroying viruses in a wide range of audio information. Background technique

[0002] MpegI is a video / audio data compression coding standard formulated by the Motion Picture Experts Group. MpegILayer2 is referred to as mp2, and mp2 compression coding is one of the audio data compression standards (the second layer). mp2 as extension.

[0003] The principle of Mp2 compression coding is:

[0004] In the first step, the audio signal is sampled at uniform time intervals, and the data files obtained after sampling are generally stored as pcm (Pulse Code Modulation) codes, which can generally be played directly by hardware.

[0005] In the second step, the voice code (pcm) is converted from a time-domain signal to a frequency-domain signal using a Fourier time-frequency filter. Among them, the Fourier time-frequency filter is a typical high-low pass filter. The filtered frequency domain signal ...

Claims

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