语音信号分析在数字语音处理中起着十分重要的作用。
Voice analysis is an important subject of the digital sound processing.
介绍了一个以高速多DSP为核心的数字语音处理卡的设计方法。
The design method of a digital voice processing card based on high speed DSPs is described.
主要说明了数字语音处理芯片FM1073在蓝牙车载免提系统中进行回声消除处理的应用方法。
In this paper, we describe an application of digital speech processing chip FM1073 to improve the performance of acoustic echo cancellation in the Bluetooth hands-free system.
随着现代科学技术的蓬勃发展，数字语音信号处理作为一个跨学科、综合性的研究领域已成为当今的一个研究热点。
With the development of modern science and technology, as a multidisciplinary and comprehensive subject, digital speech signal processing has become a hot research area nowadays.
该成果的硬件设备可适应不同的实时数字信号处理要求，应用于雷达、通讯、遥控、仪器、语音、智能等各种场合。
The hardware equipments are suitable for different real-time digital processing and apply to radar, communication, remote, instruments, voice and intelligence systems.
此外，一些典型情况的嘶哑语音信号被数字信号处理方法分析，这些方法包括FFT、LPC、倒谱技术、伪彩色编码等。
Moreover, hoarse speech signals of some typical cases are analyzed by the methods of digital signal processing, including FFT, LPC, Cepstrum.
该仪器基于PC机结构并融合了高速DSP技术和语音数字处理的研究成果。
The equipment is PC-based and integrates the high speed DSP technology and our newest research results in digital speech processing.
在语音信号数字处理过程中，语音编码技术是至关重要的。
In the digital processing of speech signals, speech coding technique is most important.
数字滤波在语音和图像处理、HDTV、模式识别、谱分析等应用中经常用到。
It is useful of Digital filter in voice processing, image processing, HDTV, Pattern Recognition and spectrum analysis.
理论上详细介绍了HMM模型及与语音识别相关的语音数字信号处理。
This paper also introduces HMM model and speech digital signal processing associated with speech recognition .
本文利用新一代可编制逻辑器件在数字信号处理领域的优势，对语音信号低码率编码的FPGA实现进行了深入研究。
Use the predominance of new programmable logic chip on digital signal domain lucubrate on the implement with FPGA of speech signal low bit rate coding.
介绍一种可用于语音处理的低功耗、高速率的抽取和内插数字滤波器的集成电路设计方法。
A low power, high speed design method of decimation and interpolation digital filters for audio signal processing is described.
频移处理是数字语音(音乐)信号处理中一项十分重要的技术。
It is an important technology to shift the frequency in digital vocal (musical) signal processing.
通过增强处理可以改善原系统在外界环境干扰条件下的性能，提高语音数字通信质量。
The performance of the speech process system and the quality of the digital speech communication can also be improved in the disturbed environment by enhancement implement.
本设计基于软件无线电的思想，使用软件的方法直接对短波信号进行处理，实现AM语音调制语音信号全数字解调。
In my design, signal is processed directly with software method based on software radio idea, which realizes the AM signal demodulation.
与此同时，结合数字信号处理知识，分别用谐波修正和时变数字滤波器的方法，完成基于骨导信号的语音重构。
At the same time, based on the knowledge of digital signal processing, reconstruction by harmonic correction and time variant digital filter was proposed.
在过去的几十年中，多速率数字滤波器组在许多领域已得到广泛的应用，如语音、图像处理和通信领域等。
In the past decades, multirate digital filter Banks attract many attentions in various fields, such as speech, image and communication applications.
为克服以往滤波器组参数调整复杂问题和实现电子耳蜗语音处理的快速数字化计算，提出了将小波变换应用于电子耳蜗的语音处理。
A new method on the basis of wavelet transform (WT) was presented to easily adjust the parameters of filter bank and realize fast algorithm of speech processing for cochlear implants.
本课题从现代通信实验系统的要求出发，设计并实现了基于DSP(数字信号处理器)的语音编解码系统。
Starting from the request of the modern communication experimental system, the subject have designed and implemented a voice codec system based on DSP (digital signal processor).
本文首先总结了语音信号数字化处理过程，分析了常用的几种端点检测方法，并给出了其实验结果与一些相应的改进。
Firstly, the digital speech signal processing and some common speech endpoint detection methods are summarized and analysed. Some experiment results and improvements are also shown in the paper.
线性预测分析技术是对语音信号处理各个领域都具有很高价值的一种基本的语音信号数字处理技术，它是语音信号数字处理技术在很多方面得以实现的基础。
Linear prediction analysis is a basic technique for digital processing of speech signal and the base for the technique of digital processing of speech signal to be implemented in many applications.
本文介绍了一个基于定点数字信号处理器ADSP2181的语音控制器，着重讨论了其语音特征参数计算方法。
Introduced in this paper is a speech controller based on fixed-point DSP ADSP2181, which adopts mel-scale frequency cepstrum coefficient as principal speech recogniti0n parameter.
本文介绍了一个基于定点数字信号处理器ADSP2181的语音控制器，着重讨论了其语音特征参数计算方法。
Introduced in this paper is a speech controller based on fixed-point DSP ADSP2181, which adopts mel-scale frequency cepstrum coefficient as principal speech recogniti0n parameter.
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