Description: 通过对语音信号的采集和AD50数模转换实现语音信号的处理，并通过C54试验台可以很好的实现语音再现。FIR数据滤波数据需要通过MATLAB来实现。-Through the acquisition and AD50 voice signal digital-analog conversion of voice signal processing, and through the C54 test bed can be a very good voice reproduction. FIR data filtering data to achieve through MATLAB. Platform: |
Size: 2048 |
Author:Faicy |
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Description: 基于 MATLAB 的语音信号分析与处理的课程设计.录制一段自己的语音信号，并对录制的信号进行采样；画出采样后语音信号的时域波形和频谱图；给定滤波器的性能指标，采用窗函数法或双线性变换设计滤波器，并画出滤波器的频率响应；然后用自己设计的滤波器对采集的语音信号进行滤波，画出滤波后信号的时域波形和频谱，并对滤波前后的信号进行对比，分析信号的变化；回放语音信号-MATLAB-based voice signal analysis and processing of the curriculum. Record a voice signal itself, and the recorded signal is sampled draw sampled speech signal time-domain waveform and frequency spectrum filter performance given by using window method or the bilinear transformation to design a filter, and draw the filter frequency response then filter of their own design collection of the speech signal is filtered, the filtered signal to draw the time domain waveform and frequency spectrum, and filter the signal before and after comparison, analysis of signal changes playback voice signal Platform: |
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Author:姚梅 |
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Description: 线性预测分析是现代语音信号处理中最核心的技术之一，为现代语音信号处理的飞速发展立下了赫赫功劳，在语音分析、合成、编码、识别等方面都有着广泛的应用，至今仍是最有效的语音分析技术之一。-Linear prediction analysis is the most modern speech signal processing one of the core technology for the modern rapid development of voice signal processing and established illustrious contribution of the speech analysis, synthesis, coding, recognition and so on are widely used, still is the most effective voice analysis technologies. Platform: |
Size: 1024 |
Author:刘鑫龙 |
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Description: 语音处理系统DSP，介绍了核心处理器TMS320VC5416，编码器TLC320AD50C和TMS320VC5416芯片McBSP口。设计了DSP语音处理系统，分析了各个模块，在Matlab上完成了对语音信号处理的仿真。-Voice processing system DSP, introduced the core processor TMS320VC5416, TMS320VC5416 chip encoder TLC320AD50C and McBSP port. DSP voice processing system designed to analyze the various modules in the Matlab completed the simulation of the voice signal processing. Platform: |
Size: 32768 |
Author:孙洋 |
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Description: 这是我收集来的有关语音信号非线性处理方面的小论文，主要有关语音非线性分析、研究、预测编码以及识别方面，希望对大家有帮助！-This is what I gathered the voice signal processing, nonlinear small papers, mainly the voice of nonlinear analysis, research, and to identify aspects of predictive coding, we want to help! Platform: |
Size: 8979456 |
Author:高菲悦 |
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Description: 这个1999年国际会议的电子版，主要是关于语音信号处理方面的权威会议和著作，希望对语音信号处理的朋友有帮助。-The 1999 International Conference on e-version, the main voice signal processing on the authority of meetings and work, I hope the speech signal processing of friends to help. Platform: |
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Author:zhaohong |
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Description: 对录音信号集 中的某一语音，利用BATLAB设计一美尔例谱算法，并实现。
取信号集 中的一个语音信号：“xxxxxx”，将它作为输入的语音信号来为设计一个美尔倒谱算法，在该算法中，主要设计了以下环节：
1.读入一个语音信号；2.对这个信号归一化；3.对归一化的信号进行加窗处理（这里的矩形窗长度必须为257，重帧长64）；4.进行预加重处理，即通过一个高通滤波器： ；5进行512点的FFT；6.分别取模平方得到功率谱；7.在设计的mel滤波器组中，我采用了25个带通滤波器；8.将得到的功率谱信号通过mel滤波器组，得到相应的25个功率值；9.对这些功率值取自然对数；10.对这些值取离散作弦变换；11.将得到的值去掉直流分量，取其它值作为MFCC参数。
-Concentrated on recording a voice signal, using BATLAB designed Mayer cases spectral algorithm, and implementation.
Get a voice signal signal focus: "xxxxxx", will it as input for the design of a voice signal to Mel Cepstrum algorithm in the algorithm, the main design of the following links:
1. Read a speech signal 2. On the signal normalized 3. Of the normalized windowed signal processing (here, rectangular window length must be 257, re-frame size 64) 4. For Pre-emphasis processing through a high-pass filter: 5 for 512 points of FFT 6. modulo square were obtained spectrum 7. in the design of mel filter banks, I use a 25 band-pass filter 8 . will be the power spectrum signal by mel filter bank, corresponding to the 25 power values 9. on the power to take the natural logarithm value 10. to take these values as strings of discrete transform 11. will be of value to out DC component, to take other values as MFCC parameters. Platform: |
Size: 13312 |
Author:赵欣 |
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Description: 学习期间，跟着导师做的一个project，主要讲的是语音信号处理在双耳助听器中的应用。-During the study, along with instructors to do a project, mainly about the voice signal processing in binaural hearing aids in the application. Platform: |
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Author:朱启明 |
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Description: 此代码包含了语音信息处理的基本上所有要用到的函数。这是语音信号处理的工具箱，对与研究语音信号处理的人来说，这个工具箱是必不可少的-This code contains the voice information processing essentially all use to the function. This is the speech signal processing toolbox, and research on human speech signal processing, this kit is essential Platform: |
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Author:luzhiqiang |
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Description: 语音识别技术的最终目标是要让计算机能与人自由交谈。目前，连续语音识别技术正趋于成熟，语音识别也延伸出了诸多实用化的研究方向。今后，语音识别的重点将集中在自然话语识别与理解、实时语音识别和语音识别鲁棒性等方面。作为一门交叉学科，语音识别所涉及到的技术有信号处理、模式识别、概率论和信息论、发声机理、听觉机理和人工智能等。-The ultimate goal of speech recognition technology is to make computers and allowing others to speak freely. At present, continuous speech recognition technology is maturing, voice recognition is also extended out a lot of practical research directions. In the future, speech recognition will focus on Spontaneous Speech Recognition and Understanding, real-time speech recognition and speech recognition robustness and so on. As an interdisciplinary subject, speech recognition technology involved in a signal processing, pattern recognition, probability theory and information theory, vocal mechanism, hearing mechanism and artificial intelligence. Platform: |
Size: 5483520 |
Author:吴斌 |
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Description: (A律和 律压缩解压算法实现，其中的C语言文本，DSP，CCS)本试验要求完成的内容是使用AD将外部语音信号采集，DSP对语音信号保存处理，经过DA将处理后的语音信号输出；DSP对语音信号的处理包括A律和 律压缩解压、声音信号的音效处理等。-(A law and legal compression decompression algorithms, including the C language text, DSP, CCS) content required to complete this study is the use of AD to an external audio signal acquisition, DSP save the speech signal processing, through the DA' s voice will be treated signal output DSP of the voice signal processing, including law and law A compression decompression, audio processing sound signals. Platform: |
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Author:RAINY |
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Description: matlab语音信号处理工具箱
本工具箱可配合语音数据库使用，用于计算线性预测语音模型的参数，语音声调转换，语音自动解析分解，语音语速变换及更改发音的重音，音量，清晰度等。配套图书《Speech Processing and Synthesis Toolboxes》已经出版，翻译进行中-matlab signal processing toolbox with the voice of the Toolbox database can be used for calculating linear predictive speech model parameters, voice tone conversion, automatic speech analysis decomposition, transformation and change of voice speech rate accented pronunciation, volume, clarity, etc. . Matching book " Speech Processing and Synthesis Toolboxes" has been published in translations Platform: |
Size: 9350144 |
Author:wujingjing |
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Description: 数字信号处理在电话语音拨号系统中的应用（自动拨号）-Digital signal processing in a telephone voice dialing system (automatic dialing) Platform: |
Size: 77824 |
Author:辰枫 |
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Description: 基语音信号处理中的一些算法代码，比较齐全，希望对各位有些帮助- voice signal processing of some computation code, relatively complete, hope to you some help
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Size: 3035136 |
Author:胡 |
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