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语音编解码
相关语句
  speech codec
    Design and Implementation of Algorithm for Speech Codec Based on DSP
    语音编解码算法在DSP芯片上的研究与实现
短句来源
    IP Core Design of RPE-LTP Speech Codec Based on FPGA
    基于FPGA的RPE-LTP语音编解码的IP核设计
短句来源
    729 speech codec and the optimization of IMDCT in algorithm and program based on Leadtek VFAST DSP chip, the optimization and implementation of IMDCT is the key element of the implementation of MP3 decoder. Firstly, the architecture of VFAST Chip is introduced in the paper.
    729语音编解码系统在丽台公司VFAST DSP芯片上的优化实现,描述了IMDCT变换在此芯片上实现的算法和程序上的优化,IMDCT的优化实现对MP3解码器在VFAST DSP上的实现起到了重要的作用。
短句来源
    After testing the IP core which we have designed to implement the RPE-LTP speech codec algorithm, the result is satisfying.
    由本课题设计实现的RPE-LTP语音编解码算法的IP Core,取得了很好的试验结果。
短句来源
    RPE-LTP(Regular Pulse Excited — Long Term Prediction) speech codec algorithm is one of the standards for GSM(Global System for Mobile Communication).
    RPE-LTP(规则脉冲激励-长期预测)语音编解码算法是GSM(全球移动通信系统)所使用的全速率语音编解码标准。
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  “语音编解码”译为未确定词的双语例句
    A DSP Implementation of Digital Speech Coder
    数字语音编解码一种DSP实现方法
短句来源
    723.1. Many design of G.
    723.1是低速率语音编解码的解决方案之一。
短句来源
    This is for the state of the hardware technology's development in the GSM mobile communication setting-up period.
    目前对RPE-LTP语音编解码算法的实现方案设计中,大多采用DSP(数字信号处理器)作为首选,这与GSM移动通信系统建立时期的硬件技术发展状况有关。
短句来源
    The subjective listening test results show that the WI speech quality using the proposed scheme at four bits per frame for REW quantization exceeds the reconstructed one using the matrix quantized REW magnitude spectra based on DCT with ten bits per frame.
    主观听力测试结果表明该量化方案在每帧4比特时的WI语音编解码质量要优于用基于DCT的REW幅度矩阵量化方案在每帧10比特时的重建语音质量。
短句来源
    The speech coding algorithm,the hardware architecture and work flow of the vocoder system,software implementation and code optimization are introduced.
    介绍了语音编解码算法原理、声码器系统的硬件结构、工作流程以及软件实现与代码优化。
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  speech codec
729 is the primarily recommended speech codec by H.
      
Due to adopting these optimizing methods and programming skills, we have implemented a high-speed speech codec that can process concurrently 20 voice channels with single TMS320C6201 chip in IP telephony gateway.
      
This paper gives a brief description of the Skyphone and the factors which influence the speech codec design.
      
This paper describes methods for mode selection in multirate speech codecs, such as the AMR (Adaptive Multi-Rate), that is the mandatory speech codec selected in 3GPP (3rd Generation Partnership Project) defined mobile networks.
      
A speech codec used in a VoIP environment must be able to deal with lost packets.
      
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How to efficiently quantize magnitude spectra with limited bits is a key issue in low bit rate parameter speech coding. The quantization of rapidly evolving waveform (REW) magnitude spectra in waveform interpolation speech coding is discussed. This paper proposes a quantization scheme with the perceptual weighting for REW magnitude spectra based on variable dimension (VD) of vector and discrete cosine transform (DCT). This kind of method decreases the bit numbers for encoding REW magnitude spectra, reduces storage...

How to efficiently quantize magnitude spectra with limited bits is a key issue in low bit rate parameter speech coding. The quantization of rapidly evolving waveform (REW) magnitude spectra in waveform interpolation speech coding is discussed. This paper proposes a quantization scheme with the perceptual weighting for REW magnitude spectra based on variable dimension (VD) of vector and discrete cosine transform (DCT). This kind of method decreases the bit numbers for encoding REW magnitude spectra, reduces storage requirement and computational complexity and enhances the perceptual quality of coded speech. The subjective listening test results show that the WI speech quality using the proposed scheme at four bits per frame for REW quantization exceeds the reconstructed one using the matrix quantized REW magnitude spectra based on DCT with ten bits per frame.

在低速率参数语音编码算法中,如何用有限的比特数有效地量化幅度谱是一个关键问题。本文对波形内插语音编码模型中快渐变波形幅度的量化问题进行了深入研究和分析,提出了一种基于矢量变维和DCT的REW幅度感觉加权量化方案,该方法降低了编码比特数,减少了存储和计算复杂度,增强了编码语音的感性质量。主观听力测试结果表明该量化方案在每帧4比特时的WI语音编解码质量要优于用基于DCT的REW幅度矩阵量化方案在每帧10比特时的重建语音质量。

The algorithm and hardware implementation of a high quality 600 bps vocoder based on TMS320VC55x DSP is presented in this paper.The speech coding algorithm,the hardware architecture and work flow of the vocoder system,software implementation and code optimization are introduced.Through C and assembly language mixing programming,assembly instruction optimization,memory cost and computation complexity are greatly reduced and meet the target requirement for real-time.

给出了一种编码速率为600b/s的高质量声码器算法及基于DSP芯片的硬件实现。介绍了语音编解码算法原理、声码器系统的硬件结构、工作流程以及软件实现与代码优化。针对C55xDSP芯片的结构特点,采用C与汇编混合编程,汇编指令优化等方法,大大降低了算法的存储复杂度和运算复杂度,达到了实时性要求。

A/D converter AD73311 was used for AMBE speech coding device AMBE-2000 to realize 2.0~2.4kbps speech coding.This paper introduces characteristic and performance analysis,and then presents a schematic design and test results.

选择AD73311模数/数模转换器和基于AMBE语音压缩算法的一款较新的声码器AMBE-2000作为核心部件,设计一种语音编解码模块.实现了模数/数模转换,以使其能够在2.0~2.4 kbps的低比特速率情况下产生比较好的话音质量.

 
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