A method of audio compression based on ip intercom system

An audio compression and intercom system technology, applied in voice analysis, instruments, calculations, etc., can solve problems such as high equipment and network requirements, low bandwidth utilization, low transmission efficiency, etc., and achieve high engineering significance and market value, transmission The effect of reduced efficiency and minimized overall bandwidth consumption

Active Publication Date: 2022-04-08
ZHEJIANG UNIV OF TECH
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Problems solved by technology

[0009] In order to overcome the deficiencies of the prior art, such as low transmission efficiency, low bandwidth utilization, and high equipment and network requirements, the present invention provides an audio compression method based on an IP intercom system, which has the following characteristics: ① High transmission efficiency ② High bandwidth utilization ③Consider both call delay and voice quality

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  • A method of audio compression based on ip intercom system
  • A method of audio compression based on ip intercom system
  • A method of audio compression based on ip intercom system

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Embodiment Construction

[0059] The present invention will be described in further detail below in conjunction with the accompanying drawings.

[0060] refer to Figure 1 ~ Figure 3 , a method based on audio compression in an IP intercom system, a two-dimensional K-u algorithm, that is, a method using two-bit K-means and uniform quantization in audio compression, including the following steps:

[0061] 1) Normalization of audio signals and establishment of two-dimensional signals

[0062] The sending end samples the audio signal, A 1 , A 2 , A 3 , A 4 ,...,A 2N , a total of 2N sampling points are obtained, and the maximum value A is obtained max and minimum A min , and then normalize the amplitude S of each sampling point i for:

[0063]

[0064] denoted as S 1 , S 2 , S 3 , S 4 ,...,S 2N . In two-dimensional quantization, two adjacent sampling points form a two-dimensional signal point, the original odd-numbered sampling point is the abscissa, and the even-numbered sampling point is...

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Abstract

A method based on audio compression in an IP intercom system, comprising the following steps: 1) normalization of the audio signal and establishment of a two-dimensional signal; 2) finding the Euclidean distance from the sampling point to the far point; 3) determining the critical radius R; 4) classification of audio signals; 5) classification and quantization; 6) encoding and sending; 7) decoding and playing. The invention provides an audio compression method based on an IP intercom system, which has high transmission efficiency, high bandwidth utilization rate, call delay and voice quality. The invention applies two-dimensional K-means and uniform The quantitative algorithm, here we name it as the two-dimensional K‑u algorithm. In IP intercom, the collection, quantization and coding of voice data are completed, and the transmission cost, transmission bandwidth and voice quality are jointly considered to realize the optimization of transmission efficiency and voice quality.

Description

technical field [0001] The invention relates to the field of audio processing, in particular to an audio compression method in an IP-based intercom system. Background technique [0002] In recent years, domestic emergencies have emerged one after another, and the social demand for efficient handling of emergencies and information security has continued to increase, and IP intercom has played an increasingly important role in it. However, in emergencies, the network status is often not well guaranteed. Therefore, an IP intercom system with low network requirements and high voice quality is particularly important. A crucial part of this is the compression of the audio. By compressing audio, transmission cost and bandwidth can be reduced, and the efficiency of voice transmission can be improved. [0003] The audio compression methods used in traditional IP intercom mainly include G.711, G.723, G.728, G.729, etc. [0004] G.711 is a non-linear quantization of voice analog sig...

Claims

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Application Information

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Patent Type & Authority Patents(China)
IPC IPC(8): G10L19/00G06K9/62
CPCG10L19/00G06F18/23213
Inventor 孟利民柴荣灿蒋维应颂翔林梦嫚
Owner ZHEJIANG UNIV OF TECH
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