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Design And Implementation, Based On The Mcf5249 The Voip Voice Processing Module

Posted on:2007-04-20Degree:MasterType:Thesis
Country:ChinaCandidate:F TianFull Text:PDF
GTID:2208360185956610Subject:Signal and Information Processing
Abstract/Summary:
VoIP, which stands for Voice over Internet Protocol, is the IP voice and speech technology. Its basic idea is to realize voice and speech business though transmitting IP packet. The most outstanding advantage of VoIP is capable of utilizing the world Internet networking to provide IP telephone service, which provides more and better value-added services than traditional telephone business.Up to now, VoIP has developed rapidly all over the world and implemented a majority part of traditional PSTN functions, based on which people raised additional requirement on speech quality that is not satisfied in VoIP now. But, as the inherent trait of delay and lack of bandwidth on the internet, the speech quality of VoIP is not as good as desired, which once prevent the development of VoIP. To improve the speech quality, it is necessary to adopt such speech processing techniques like speech compression, voice activity detection, echo cancellation, Jitter Buffer and so on.Built upon the Project Designing VoIP Terminal Based on ColdFire Platform, this paper designs speech processing module and successfully applied this solution on target platform. The speech processing module can be divided into two parts, the first part includes speech compression, voice activity detection and echo cancellation module, which improve speech quality; the other part includes DTMF and CPT module, which generate and detect some necessary telephony signal in the communication.This thesis is organized as follows. Section I introduces the main techniques in the VoIP speech module and analyze the relative algorithm. Section II describes the design approach and implementation of speech module on MCF5249 ColdFire core. The speech codec optimizes G.729A codes and added voice activity detection of G.729B to save bandwidth; the implementation of acoustic echo cancellation uses NLMS algorithm and it can reduce echo though designing adaptive FIR filter and speech detector; the DTMF and CPT generate signal using two second order digital sinusoidal oscillators and detect signal by picking up the frequency information. But only get the frequency information is not enough in CPT detector, this thesis introduces a method...
Keywords/Search Tags:VoIP, Speech Compression, Echo Cancellation, DTMF, CPT
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