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Voice signal processing method, device and system

A voice signal processing and voice signal technology, applied in the field of wireless communication, can solve problems affecting voice quality, incomplete segmentation, limited number of sub-bands, etc., and achieve the effect of improving the quality of voice communication, increasing the length and quantity, and increasing the number of combinations

Inactive Publication Date: 2018-07-13
SHISHI FTGMDC COMM EQUIP
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  • Summary
  • Abstract
  • Description
  • Claims
  • Application Information

AI Technical Summary

Problems solved by technology

[0004] However, due to the very limited bandwidth of analog voice, the number of divided subbands is limited, which affects the possibility of out-of-order combination, resulting in very limited practical and effective encryption methods, generally limited to less than 16, which cannot meet actual needs
In addition, due to the limitation of filter performance, the segmentation between each sub-band is not complete, which affects the restored speech quality
[0005] To sum up, it can be seen that the means of voice scrambling processing in the existing technology is very limited, and the effect is not satisfactory, which cannot meet the use requirements in production practice.

Method used

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  • Voice signal processing method, device and system
  • Voice signal processing method, device and system
  • Voice signal processing method, device and system

Examples

Experimental program
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Embodiment 1

[0062] figure 1 It is a schematic flow chart of the first voice signal processing method provided by the embodiment of the present invention. The method is applied to the transmitting end to perform scrambling processing on the voice signal to be transmitted. Such as figure 1 As shown, the method includes the following steps:

[0063] Step S102, acquiring a voice signal, and converting the voice signal into a digital coded stream.

[0064] In some possible embodiments, at the transmitting end, an analog voice signal input by a user is received, and the voice signal is isochronously sampled, quantized, and encoded to obtain a PCM (Pulse Code Modulation, pulse code modulation) code stream. Specifically, the speech signal is sampled according to a certain sampling frequency, and then the sample amplitude is quantized and coded, that is, the time-continuous and value-continuous analog speech signal is transformed into a time-discrete and value-discrete digital coded signal to ob...

Embodiment 2

[0092] Corresponding to the scrambling method in the first embodiment above, this embodiment further provides a descrambling method. Use the pilot code (synchronization code / Pilot) inserted in the voice stream to realize the synchronization and definition of the super frame, and then realize the inverse operation of the out-of-sequence process on this basis, so as to realize the descrambling of the scrambled voice.

[0093] Figure 7 It is a schematic flow chart of the second voice signal processing method provided by the embodiment of the present invention. The method is applied to the receiving end to perform descrambling processing on the received scrambled coded signal. Such as Figure 7 As shown, the method includes the following steps:

[0094] Step S702, acquiring the scrambled coded signal, and performing superframe synchronous scanning on the scrambled coded signal to determine the position of the synchronization code of the scrambled coded signal.

[0095] In orde...

Embodiment 3

[0121] Figure 11 The schematic structural diagram of the first voice signal processing device provided by the embodiment of the present invention, the device is applied to the transmitting end, such as Figure 11 As shown, the device includes:

[0122] A code conversion module 1102, configured to acquire a voice signal, and convert the voice signal into a digital code stream;

[0123] A time-domain segmentation module 1104, configured to perform time-domain segmentation on the digital encoded stream to obtain multiple subframes, and divide a set number of consecutive subframes into a superframe;

[0124] The out-of-sequence scrambling module 1106 is configured to perform superframe synchronization and out-of-order processing on each superframe in units of subframes to obtain scrambled encoded signals after scrambling.

[0125] Figure 12 A schematic structural diagram of the second speech signal processing device provided by the embodiment of the present invention, the dev...

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Abstract

The invention provides a voice signal processing method, device and system, and relates to the technical field of wireless communication. The method comprises the following steps: acquiring a voice signal, and converting the voice signal into a digital coding stream; then performing time domain segmentation on the digital coding stream to obtain a plurality of sub-frames, and dividing a continuousset quantity of sub-frames into a super-frame; and performing super-frame synchronization and out-of-order processing on each super-frame by taking the sub-frames as units to obtain a scrambled coding signal. The voice signal is segmented in a time domain, and the length and number of disorders can be increased by increasing or decreasing the length of each sub-frame and increasing or reducing the number of sub-frames included in the super frame, so that the number of disorder combinations is greatly increased, and the actual need can be met. In addition, being different from frequency domainsegmentation, the time domain segmentation is easy to implement, and sub-slots (sub-frames) can be completely independent without affecting each other, thus improving the quality of voice communication.

Description

technical field [0001] The present invention relates to the technical field of wireless communication, in particular to a voice signal processing method, device and system. Background technique [0002] During the use of traditional analog radio stations, sometimes voice communication needs to be scrambled to ensure that voice information is not leaked. [0003] In the prior art, the scrambling processing of the analog voice signal is performed in the frequency domain, that is, at the transmitting end, the voice signal is divided and disordered in the frequency domain, for example, the voice signal of 300Hz-3000Hz is divided into several subbands, and then shuffle these subbands in the frequency domain. Correspondingly, at the receiving end, according to the opposite process, the reverse frequency domain scrambling process is performed on the received scrambled speech signal, so that the speech signal can be restored. [0004] However, due to the very limited bandwidth of ...

Claims

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Application Information

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Patent Type & Authority Applications(China)
IPC IPC(8): H04L1/00
CPCH04L1/0007
Inventor 林英狮
Owner SHISHI FTGMDC COMM EQUIP