Multi-speed audio encoding method

A voice and audio, multi-rate technology, applied in the field of coding, can solve the problems that affect the quality of the output signal, the quality of voice or audio signal has not been significantly improved, and achieve the effect of improving quality

Inactive Publication Date: 2011-08-10
HUAWEI TECH CO LTD
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Problems solved by technology

Therefore, when the number of bits is insufficient, only part of the code stream can be decoded at the decoding end, and because the parameters of the transform domain encoding at the encoding end are written in the code stream in the order of low frequency to high frequency, only a small amount can be recovered at this time. low-frequency signal, resulting in that when the bit rate increases, the quality of the output voice or audio signal does not improve significantly
In addition, when calculating the index value of the grid point, the importance of the grid point is determined according to the size of the spectral energy. When the number of bits is not enough

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[0022] In order to make the object, technical solution and advantages of the present invention more clearly, the present invention will be further described in detail below in conjunction with the accompanying drawings and specific embodiments.

[0023] figure 2 It is a flow chart of the multi-rate speech and audio coding method in the embodiment of the present invention. Such as figure 2 As shown, the flow process of the multi-rate speech and audio coding method in the embodiment of the present invention includes the steps as follows:

[0024] In step 201, the encoding end receives input data.

[0025] In this step, the nearest grid point C can be found in the RE8 grid according to the principle of proximity to the input spectrum vector x ; Move grid point C x The size of the corresponding spectrum capability and the total number of bits are used as input data, and the encoding end receives the input data and performs the following processing on it.

[0026] Step 202, ...

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Abstract

The embodiment of the invention discloses a multi-rate voice frequency encoding method. The method comprises the following steps: computing frequency spectrum of a perceptual weighting filter and a first ratio of the frequency spectrum of two previous layers of synthetic voice which are firstly encoded and then decoded by an input signal for each adjacent lattice point; and encoding index values of the corresponding lattice point of the first ratio into a code stream according to a descending order of the first ratio. The method can help improve the quality of output voice or sound signals when lacking bits during transform domain encoding.

Description

technical field [0001] The present invention relates to encoding technology, in particular to a multi-rate voice and audio encoding method. Background technique [0002] In the current multi-rate speech and audio coding, when the input signal is more in line with the music characteristics or the code rate is high (for example, in AMR-WB+, G.729.1 and G.VBR), transform domain coding is often used, that is The time-domain signal is transformed into the frequency domain by a transform method, such as Modified Discrete Cosine Transform (MDCT) or Fast Fourier Transform (FFT), etc. When quantizing the parameters of the transform domain coding, the lattice vector quantization technology is mostly used. figure 1 It is a flowchart of a lattice vector quantization method in the prior art. Such as figure 1 As shown, in general, lattice vector quantization includes the following steps: [0003] Step 101, find the corresponding grid point for the frequency domain signal according to ...

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Application Information

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IPC IPC(8): G10L19/00G10L19/24G10L19/02
CPCG10L19/24
Inventor 刘泽新马付伟肖玮
Owner HUAWEI TECH CO LTD
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