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Audio resampling method

A resampling and audio technology, applied in the direction of digital recording/reproduction, etc., can solve the problem of large amount of calculation, etc., achieve the effect of occupying less CPU resources, concise code, and increasing redundant spectrum

Active Publication Date: 2010-05-26
ANYKA (GUANGZHOU) MICROELECTRONICS TECH CO LTD
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Problems solved by technology

[0006] In addition to performing FIR filtering on the input signal above 7th order, the above implementation method also needs to transform the time-domain signal to the frequency domain for processing, and then convert it back to the time-domain output, which requires a relatively large amount of calculation.

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Embodiment Construction

[0039] An audio resampling method first generates a FIR filter coefficient table.

[0040] The filter bank coefficients are pre-processed, and these coefficient tables are generated based on the Blackman-Nuttall windowed sampling function of the Blackman-Nuttall window. The coefficient table is a one-dimensional array, but it is actually grouped, and the size of each group is the order of the filter. For example, in the upsampling filter coefficient table, each group of 8 array elements represents a group of coefficients of an 8-order filter. When resampling, the obtained coefficients are not read from the coefficient table sequentially, but the coefficients are skipped and selected according to the resampling output-to-input ratio. The filter coefficients are calculated in advance, and the filter bank is determined. There is no need to dynamically calculate the coefficients during the resampling process, which increases the amount of calculation.

[0041] Such as figure 1A...

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Abstract

The invention discloses an audio resampling method, comprising the following steps: (A) generating finite impulse response filter coefficients and constituting a coefficient table to be stored in an external memory; (B) initializing parameters; (C) pre-treating the input audio sample; and (D) performing interpolation treatment correspondingly in an upward resampling process, and performing extraction treatment correspondingly in a downward resampling process; and outputting the audio sample after resampling. The invention has the advantages that each parameter of a filter is calculated in advance and constitutes a parameter table written in the external memory like a flash memory, thus reducing calculation intensity in algorithm, simplifying code implementation correspondingly, improving calculation efficiency and lowering occupied CPU resource; and only about 10MHz basic frequency is needed even under the ARM7 processor; the processed audio has complete signals, no loss of frequency spectrum and no increased redundant frequency spectrum; and the audio data with different sampling frequencies can be converted into audio data with identical sampling frequency by resampling, thus achieving the sound mixing effect.

Description

technical field [0001] The invention relates to a data resampling method, in particular to an audio frequency resampling method. Background technique [0002] In the field of digital signal processing, the problem of sampling rate conversion is often encountered, such as enlarging or reducing the sampling rate, that is, the problem of resampling. The re-sampling technology is to process one set of digital signals into another set of digital signals, which can be realized by special hardware or by software. At present, the signal acquisition instruments put into use in China all have the resampling function realized by hardware method. The vibration signal is obtained by the sensor, and after low-pass filtering by the filter circuit, the digital extraction is performed by the hardware circuit. This method, which relies on hardware filtering and decimation, is inflexible and expensive. Therefore, software methods are often used to achieve resampling. [0003] The software ...

Claims

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Application Information

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IPC IPC(8): G11B20/10
Inventor 李军胡胜发
Owner ANYKA (GUANGZHOU) MICROELECTRONICS TECH CO LTD
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