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vocoder n.〔美國〕 語音編碼機。

vocoid

The author realize the selectable mode vocoder under visual c + + 6 . 0 environment . this paper shows the contrastive results between the original speech signal and the synthesized signal of smv under various modes and environment , it also shows the results of the test of smv ' s performance . at last the direction of the smv ' s studying is introduced on the basis of analysis of speech signal 作者在vc6 . 0環境下模擬實現了smv算法,本文給出了smv在不同模式、對不同環境下的語音信號進行編碼的對比結果,介紹了對其主客觀性能進行測試的結果,并進一步從中得出結論,最后在對語音信號進行分析的基礎上提出了下一步的工作方向。

The paper makes great efforts on the software optimization of evrc vocoder . based on the understanding of tms320c64xx cpu structure , we do deeply - optimization on the loop which appear usually in voice signal processing , and this improve the utility ratio of cpu and the parallelity degree of cpu function cell . at the same time , we utilize the bit - exact test to test the fixed - point evrc vocoder with the test vectors of tia / eia / is - 718 , which improve the robustness of the vocoder 本文圍繞定點evrc聲碼器的軟件優化,做了大量的工作,在充分理解tms320c64xxcpu結構的基礎上,針對語音信號處理中大量出現的循環運算進行了深度優化,大大提高了cpu的利用率以及cpu功能單元的并行程度,同時,我們還用tia / eia / is - 718的測試向量對定點evrc聲碼器進行了嚴格比特對準測試,提高了聲碼器的魯棒性。

Mixed excitation linear prediction ( melp ) vocoder is a kind of speech coding algorithm providing superior speech quality under very low rate even 1 . 2kbps , as well as its capability withstanding strong background noise . melp vocoders base on lpc vocoders . furthermore they add some new features to mimic the natural speech 混合激勵( melp )聲碼器是近年來提出的一種以經典lpc聲碼器為基礎的性能優良的語音編碼方案,對它的研究方興未艾,現已取得了不少的成果,可以在1 . 2kbps的碼率下取得mos分為3 . 0左右的合成語音,并且具有比較強的抗背景噪聲的性能。

Firstly , it introduces the development of speech coding , along with the significance of the low bit rate speech coding . it also compares the model of traditional dualistic excitation lpc vocoder and the multi - band excitation vocoder , and lucubrates the analytical method of frequency domain and time domain in the parameter extraction of multi - band excitation vocoding . secondly , based on the parameter extraction operation of keynote cycle , it adopts time domain in rough estimate operation of keynote and frequency domain in fine estimate operation of keynote , in according to the immediacy required in practice , to minish operation amount 本文闡述了一種基于fpga的多帶激勵語音編碼器的研究與設計,首先介紹語音編碼研究的發展狀況以及低速率語音編碼研究的意義,接著對比分析了傳統二元激勵lpc聲碼器模型和多帶激勵編碼器模型,并深入研究了多帶激勵語音編碼參數提取的頻域和時域分析法,然后根據實際應用的實時性要求,為了減小運算量,在基音周期參數的提取的算法實現上,本文采用在時域進行基音粗估運算,在頻域進行基音精細估計運算。

In this paper , we briefly introduced the performance of wave coding and vocoder , emphasizedly studied the principle and performance of variable rate vocoder q4401 , including the internal construction and pins , qcelp coder & vocoder , pcm interface , cpu interface initialization process , command format and so on . we also designed a application circuit , with the experiment validated its performance . in this design , the pcm interface chip is tp3057 , it was used to finish a / d transform , the compress coding was finished by q4401 , the initialization and control were accomplished by 8051 singlechip 重點是研究變速率語音編解碼芯片q4401的工作原理及性能。其中包括q4401的內部結構及管腳、 qcelp編碼方式、 pcm接口、 cpu接口、初始化過程、命令格式等,并在此基礎上,設計一個實際的應用電路,通過實驗,驗證其性能。在設計中用pcm接口芯片tp3057來完成從模擬信號到數字信號的轉換,再由q4401進行壓縮編碼,對q4401的初始化及控制由8051單片機來完成。

Basing the american federal 2 . 4kbps melp algorithm , the analysis platform was established for analyzing the and testing the performance of melp codec . this article analyzes the capability of the melp vocoders . finally some advice are given to realize the vocoder in hardware or software 本文以美國聯邦標準2 . 4kbps melp算法為基礎,在matlab上建立起了分析melp算法的軟件平臺,對其性能進行了分析并提出了一些改進的建議;另外還針對melp算法的特點對其軟硬件實現進行了探討。

The informal subjective test in real - time processing indicates that the synthetic speech quality of amr vocoder is better than it of rpe - ltp in gsm system , achieving toll quality , that can apply to devices employing the amr vocoder within the 3gpp system in terms of software and hardware 實時處理的非正式主觀測試表明,合成語音質量優于gsm的rpe - ltp的語音質量,達到長途語音質量,完全可以實際應用,為第三代移動通信中聲碼器設備的研制奠定了良好的軟件和硬件基礎。

We select fpga of type xc3s200 as hardware to design the coder and display the hardware resources inside , moreover study the method and steps of designing dsp , based on fpga , by using system generator , finally , it emphasizes the design process of multi - band excitation vocoder . we can work out the module of high pass filter and the module of low pass filter , module of divide frame , module of keynote rough estimate , module of keynote fine estimate , module of band - separated v / u judgment / verdict and module of band - separated amplitude estimate , by using simulink , ise and system generator 本文選用型號為xc3s200的fpga作為設計編碼器的核心硬件,介紹了其內部所含的硬件資源,并研究了利用systemgenerator基于fpga設計dsp的方法和步驟,最后,本文把重點放在多帶激勵語音編碼器的設計上,利用simulink , ise和systemgenerator分別設計其中的高通低通濾波器模塊、分幀疊加模塊、基音粗估模塊、基音精細估計模塊、分帶v / u判決模塊、分帶幅度估計模塊。

To improve the voice quality , this thesis start with replacing the present qcelp vocoder with a new one . after analysis and comparison among candidate vocoders that can be used in scdma , itu - t g729a , which is the simplified version of g . 729 algorithm , proved itself the optimum one 本文從更新聲碼器及增強其抗噪聲能力入手解決語音質量問題,經過分析比較,選擇了itu - tg . 729的簡化算法g . 729a實現聲碼器,以代替目前性能不穩定的基于qcelp算法的聲碼器。

It is highly important to enhance the error robustness of g . 729 vocoder . on the subjective sensitivity analysis of g . 729 vocoder bits using pesq algorithm , the problem is concentrated on how to protect those subjectively important bits by using the additional 1 . 6kbps channel bandwidth in scdma 在pesq客觀語音質量評價工具分析g . 729碼元主觀敏感度的基礎上,提出了如何利用scdma系統1 . 6kbps剩余業務信道帶寬對主觀最敏感比特進行保護的問題。

These are mixed - excitation , aperiodic pulses , pulse dispersion , and adaptive spectral enhancement . so the melp vocoder can produce more natural sounding speech . with a analysis to melp coding arithmetic , a speech codec scheme in software is presented in this paper 本論文在對melp編碼算法進行分析的基礎上,提出了用軟件實現語音信號編解碼的方案;然后,基于melp標準,開發了仿真系統,實現對語音信號的編解碼。

Moreover , the paper discusses the principle of software and hardware design , based on the implementation of multi - channel aivir vocoder on the tms320c6203 . the results obtained from emulation are in bit exact agreement with the calculated results provided by 3gpp 本項目實現了基于tms320c6203的靈活高效多通道amr聲碼器,仿真和錄放驗證的結果與3gpp提供的結果滿足比特級精確要求。

And a new pitch extraction algorithm , an active / inactive frame decision algorithm and a voiced / unvoiced frame decision algorithm are developed with the aims to improve the quality of the vocoder and reduce its overall computation load 本文引入了一種新的基音周期計算方法,靜音幀判決算法,清音幀判決算法,清濁音信息接收端重建等新算法,提高了合成語音的質量,降低了算法的總計算量。

At last , our vocoder is compared with the vocoder benchmark of delphi company , with the result that the performance of our vocoder is better than delphi ' s obviously , thus make the vocoder has applied value 最后,將我們開發的聲碼器與delphi公司開發的聲碼器基準程序( benchmark )作了比較,結果表明,我們的聲碼器的性能要明顯高于delphi公司的,這使得evrc聲碼器具有較大的應用價值。

The paper expatiates design of the practice apply system using the ambe arithmetic , and introduces the correlation field and applied range about voice ' s digital signal process . and talks about the superiority of low speed voice ' s vocoder 本論文主要研究了應用ambe算法,進行實際的應用系統設計,介紹了語音數字信號處理涉及到的研究領域,應用范圍,語音高效編解碼低速率的優勢。

In the study of speech coding in present , the mixed excitation linear prediction ( melp ) is a kind of relatively good method . the melp vocoder is the new federal standard speech coder 在現有的語音編碼研究中,混合激勵線性預測編碼( melp )是一種比較好的方法, melp編碼方法已經被確定為美國新的聯邦語音編碼標準。

It was selected by the united states department of defense digital voice processing consortium ( ddvpc ) . the melp vocoder is based on the traditional lpc parametric model , but also includes four additional features 該算法結合了二元激勵、碼激勵和多帶激勵的優點,能在較低的碼率下得到較好的再生語音。

Because our vocoder is mainly used in the base station , where the vocoder needs the excellent performance , so we select the tms320c64xx , the most advanced dsps of ti 由于我們開發的聲碼器主要用于基站,所以對性能的要求較高,選用了ti公司的最先進的dsp芯片tms320c64xx 。

The evrc vocoder is varialbe rate , the maximal encode rate is skbps , its voice quality is closed to qcelp - 13k , and has better ability of anti - disturbance Evrc聲碼器是可變速率的,最大編碼速率為8kbps ,在話音質量上接近于qcelp - 13k的語音編碼器,且具有更好的抗干擾能力。