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Audio grouping, sampling, transmitting and recomposing method

An audio and sampling point technology, applied in the field of data transmission, to avoid data blocking, stabilize transmission quality, and improve user experience

Inactive Publication Date: 2016-12-07
深圳羚羊极速科技有限公司
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  • Abstract
  • Description
  • Claims
  • Application Information

AI Technical Summary

Problems solved by technology

[0004] The purpose of the present invention is to provide a method for recombining audio packet sampling transmission to solve many problems in the prior art

Method used

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  • Audio grouping, sampling, transmitting and recomposing method

Examples

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example 1

[0030] Such as figure 2 As shown, it is assumed that the application scenario is a web conference that requires high real-time transmission, and according to the requirements of step S101, the data collected by the collection end is cached in the CaptureBuffer. Assume each acquisition contains 256 sample points.

[0031] According to step S102 , the unit sampling CaptureBuffer is split, the sampling data of odd bits are stored in CaptureBuffer1, and the sampling data of even bits are stored in CaptureBuffer2. Both split buffers contain 128 sampling points.

[0032] According to step S103, prepare two encoders (sampling rate 8K, number of sampling bits 16, mono), and separately compress and encode the audio data of the two CaptureBuffer1 and CaptureBuffer2 into EncoderBuffer1 and EncoderBuffer2, one of which adopts the way of TCP transmission, and the other way uses UDP transmission.

[0033] According to step S104, the receiving end prepares two decoders (sampling rate 8K,...

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PUM

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Abstract

The invention discloses an audio grouping, sampling, transmitting and recomposing method. The audio grouping, sampling, transmitting and recomposing method comprises the steps of, at a collecting end, separating each audio pcm data unit according to unit sampling points at uniform intervals to form two groups (or multiple groups) of data units; respectively carrying out coding compression and transmission for the data units; decoding the data at a receiving end; if the two groups (or multiple groups) of data units are received, recomposing a complete audio data unit; and if only one group is received, expanding the sampling point data to enable the sampling point to have the same sampling precision as the collecting end. Such grouping way provided in the invention can restore most of the original voice effect after only one frame of grouped audio data is received, so that the probability of blockage or loss is reduced. Furthermore, after the complete grouped data are received, the tone quality also can be restored completely.

Description

technical field [0001] The invention belongs to the technical field of data transmission, in particular to a system with high demand for audio transmission in a scene with interactive applications. Background technique [0002] In today's Internet + society, it has become very frequent and common for people to watch videos, live broadcasts, etc. through the Internet. Audio transmission on the Internet has high requirements for real-time and fluency, especially in interactive application scenarios (such as video calls, video conferences, etc.), audio stuttering will be very sensitive to people. The traditional implementation method is to collect the pcm unit data, compress, encode and transmit it, and then decode and play it at the playback end. When a frame of data is blocked or lost, there will be obvious lag, showing a poor user experience. [0003] In view of this, there is an urgent need for a method for audio packet sampling transmission and reassembly to solve the pr...

Claims

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Application Information

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IPC IPC(8): H04L29/06H04L29/08G10L19/002
CPCH04L65/60H04L69/161G10L19/002H04L67/5651
Inventor 张磊
Owner 深圳羚羊极速科技有限公司
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