EVB Phase Vocoder算法与MaxxBass算法在虚拟低音增强应用中的比较

介绍了虚拟低音增强实现中的MaxxBass算法与EVB Phase Vocoder算法。通过对比分析了基于时频分析的虚拟低音算法相对于传统非线性算法在算法灵活性、低音效果还原和合成信号畸变控制等方面的优势。实验证明．EVB Phase Vocoder算法合成的虚拟低音增强效果明显优于MaxxBass算法。     

小型扬声器感知低音的生成算法优化

虚拟低音技术针对扬声器低频重放问题提供了一种较好的处理方式,其建立在“虚拟音高”理论基础上,将扬声器输入信号在不能重放的低频部分利用其高次谐波成分在听感上虚拟出基频音高.考虑到小型扬声器频带窄、失真大等特性,在虚拟低音的相位声码器算法基础上,用二阶求导的方式对频谱低频峰值进行取点,对不同频段生成不同的谐波个数,简化幅度控制,提出适合于小型扬声器的低音谐波生成模块优化算法并实现.主观实验结果表明改进后的算法在增强低音效果的同时也能保证较好的整体音质效果.     

一种多脉冲激励线性预测声码器的仿真实现

本文是笔者所开发的语音信号处理课程中的一个研究型实验教学案例.文中概述了多脉冲激励线性预测声器的编码方法,并用MATLAB做了仿真实现.在本声码器实现方法中,在发送端对系统参数进行量化时,用量化LPC滤波器的PARCOR系数（部分相关系数,PARt CORrelation）代替传统的对滤波器的线性预测系数的量化;在接收端,将PARCOR系数用文中所给的算法转换为线性滤波器的线性预测系数,进行系统重构合成语音,大大降低了传输比特率,并且提高了合成语音质量.     

语音信号的线性预测编码技术

本文以一个双声道声码器为例,来说明线性预测编码技术在处理语音信号中的应用.一个2400b/s,双声道线性预测编码的声码器,使用高速、可编程序的位片型微处理机,字长可扩充.并使用2K的RAM和1K的ROM的数据存储器.该系统设计成循环时间为208毫微秒的双总线结构.该声码器能同时处理两个声道,用微处理机和线性预测编码技术开发出该系统,并叙述使用该技术设计声码器的要领.     

低速率语音编码技术

本文介绍了语音处理技术，各种基于ＤＳＰ技术的语音压缩算法和各种语音编码标准及领域，介绍了几种语音ＤＳＰ的专用芯片。     

AN IMPROVED 2.4 kb／s MIXED EXCITATION LINEAR PREDICTION VOCODER

This letter presents two improvements on 2.4 kb/s Mixed-Excitation Linear Prediction (MELP) vocoder. The one is a new parameter Redzc named energy to differential zerocrossing rate which is used in adaptation of V/UV decision of transitional segments and low energy level speech segments. The other is a multi-path searching method for Multi-Stage Vector Quantization (MSVQ) of line spectral frequency. Subjective tests show that the intelligiblity and naturallity of improved MELP vocoder are preferable to those of the original one.     

Source detection: Influence of the Doerbecker’s noise reduction algorithm; Simulation in the case of binaural cochlear implant coding

Cochlear implant (CI) provides good auditory performances in quiet, but recognition in noisy environments is still a challenge for CI users. Bilateral hearing can improve speech perception; it has been shown with normal hearing aids, but how source localization is affected is still an open question.In this study source localization is studied in the case of a binaural cochlear implant coding. Signal was received on the two ears and then processed using the classical noise reduction algorithm, suggested by Doerbecker and based on the spectral properties of the signal. Two correction strategies, Ephreim & Malah and Scalart, have been inserted in the processing chain.Tests were conducted, “in simulation” with 20 normal hearing listeners who received a cochlear implant coded signal. An analytic interpretation of the results offers a frame to quantify the different contributions.Results show that the Doerbecker’s algorithm affects the source localization. This effect was lowered when a percentage of the input signal was re-injected into the system. Both correction strategies were equivalent.     

多路G.729A声码器在TMS320C54x上的实现

ITU-UG.729A的ANSIC代码能够从ITU免费下载,但是它只实现了单路的应用,并且由于编解码的计算量巨大,ANSIC代码并不直接适用于数字信号处理器芯片(DSP)。论文描述了采用ITU-UG.729A8-kbit/s算法的多路声码器在TMS320C54xDSP上的实时实现方法。文章首先介绍了G.729A算法标准,然后讨论了如何构建C语言级的多路声码器,最后研究了如何将该C代码转换成TMS320C54x的汇编语言程序,并同时满足实时要求。整个程序在笔者自行开发的目标板上实现,并已成功地用于相应的VOIP系统中。     

变码率分类型实时语音声码器

本文提出的声码器将语音分成静音、清音、浊音和混合音四类。用自适应方法进行分频带清浊音判决和有声／无声判决,提高了分类算法的稳定性、准确性和灵活性、准确性和灵活性,还保持了混合语音的音质,且无须对清浊音判决结果进行编码。对清音和浊音的频谱分别采用不同的ＬＳＰ量化表进行编码,从而用标量量化器替代子矢量量化器,降低了复杂度。声码器的码率最高２．４ｋｂｐｓ,最低为１００ｂｐｓ,平均码率１．４ｋｂｐｓ。实时     

Improved Populating Code-Excited Linear Predictive Coder

1Introduction,TheCode--ExcitedLinearPredictive(CELP)[13coderprovidedgoodqualityspeechatmediumandlowbitrates,butthisqualityspeechwasatthecostofverycomputationalcomplexity.Recently,therealtimeimplementationoftheCELPcodersonalowpricedigitalsignalprocessorchi…