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raw-wave-lib-v0.1
- raw-wave-lib：一个非常好的wav文件读写的库，包含测试文件。在Linux/windows下运行良好。使用C语音书写-raw-wave-lib : a very good wav file reader of the library, including test paper. Linux / windows running good. Speech writing C
jiyushilianglianghuadeshuohrshibe
- 基于矢量量化的说话人识别本文从语音信号的预处理开始分析， 主要研究了特征参数的选择、提取、及识别算法，应用全极点模型，提取了语 音信号的线性预测倒谱系数和美尔倒谱系数，并进一步获得其一阶差分，将倒 谱系数与其一阶差分结合在一起形成新的特征参数。在识别算法方面，本文对 矢量量化的方法进行了研究，用Matlab语言实现了说话人识别系统的仿真与验 证。实验证明这种参数与单纯的线性预测倒谱系数和美尔倒谱系数相比更为有 效。- speech paper，help you study
jiyuneirongdeyinpinjiansuoyan
- 本文根据上述的研究，采用基于Mel倒谱系数特征的隐马尔可夫模型对音 乐进行分类。在音乐特征提取方面，以感知特征和Mel倒谱系数组成特征向量 在音乐分类方面，以隐马尔可夫模型作为分类器，对音乐进行聚类和分类。通过 有监督的学习方式进行聚类，分类时将测试样本归入似然值最大的类别，对同一 音频抽取若干样本，对样本识别结果采用投票法，判定该音频的音乐类别，使分 类的准确率得到进一步的提高。根据上述方法进行了仿真实验，并对实验结果进 行了分析。本文将音频数据分为5类，对4种分类器
F2_6764
- 端点检测是指用数字处理技术来找出语音信号中的各种段落（如音素、音节、词素、词等）的始点和终点的位置。语音段起止端点检测是语音分析、语音合成和语音识别中的一个必要环节。传统的端点检测方法是从wav文件中获取语音采样，将其分帧并计算短时能量和过零率参数，然后进行端点检测。这种工作方式被称为离线处理方法 ，无法实现语音信号的实时处理，对于语音信号分析具有一定的局限性。本文通过开发ActiveX控件，在MATLAB环境下将其嵌入到figure窗口中，以GUI程序的方式使用，实现语音信号端点检测的实时处
ACOMPARISONOFCLASSIFIERSFORDETECTINGEMOTIONFROMSP
- paper for speech emotion recognition
speechsegmentation
- speech segmentation paper
Method-of-Converting-Speech-Codec-Formats-between-
- Abstract - In this paper, proposed an efficient method of converting speech codec formats between G.723.1 and G.729А. The proposed method is composed of four steps: LSP conversion, pitch conversion, adaptive codebook search and fixed cod
mirexport
- understand the triangul [voiceEmotionMP3.rar] - for Emotional Speech Recognition Software for the voice paper for the MP3 format, including sadness, anger, happiness, Panic, calm voice, and other in [speakerprotected1.rar] - Matlab procedure
mirfluctuation
- understand the triangul [voiceEmotionMP3.rar] - for Emotional Speech Recognition Software for the voice paper for the MP3 format, including sadness, anger, happiness, Panic, calm voice, and other in [speakerprotected1.rar] - Matlab procedure
Matlab-spectrumananalysis
- 该文给出了一种利用matlab系统实现频谱分析与显示的方法。该方法对语音信号进行基于FFT的短时频谱分析，频谱图的伪彩色映射及显示。-This paper presents a system implementation using matlab spectrum analysis and display methods. The method is based on the speech signal short-time FFT spectrum analysis, spectrum and
my-paper
- effecient speech recognition using wavelet and g-effecient speech recognition using wavelet and gmm
19_M-W-K-D-D_SPA2010_final
- This paper presents results of speaker recognition experiments using short Polish sentences. We developed and analyzed various vector quantization representations in order to first maximize identification effectiveness and second to compare VQ (vecto
progress-in-speech-compression
- 为了满足数字通信及其它商业应用的需求，语音压缩编码技术得到了迅速发展。介绍了目前语音压缩编码技术 的研究进展，主要包括连续可变斜率增量调制(CVSD)、小波分析、多脉冲激励线性预测编码、散布脉冲码激励线性预 测(DP-CELP)、多重脉冲散布非均匀代数码本激励线性预测(MPD-USACELP)、波形内插(Ⅷ)、线谱对(频率)(LSP)的量化-In order to satisfy deman凼of the digital communication and other commercia
The-research-of-anti-niose-speech
- 论文首先介绍了传统的语音特征参数MFCC，它是基于人耳听觉 特性设计的一种特征参数，在静音环境下能得到较高的识别率，但在 信噪比较低时识别率急剧下降，不利于实用化。本文通过对MFCC算 法的分析和研究，发现其中的FFT和DCT在整个时频空间使用固定的 。分析窗，这不符合语音信号特性，而小波变换具有多分辨率特性，更 符合人耳的听觉特性。因此，本文将小波变换和MFCC算法相结合， 提出了三种新的语音识别特-Speech recognition has wide use in
visual-speech-recognition
- this paper descibe about speech recognition. you will be helpful about learing this field.
Digital-Speech-Input-System
- Abstract HTK is a speech signal processing toolkit developed by CUED based on C language. It is widely used for speech signal recognition， speech synthesis，character recognition and so on. This paper describes the process and principles of speech r
Speech
- 文本朗读系统是TTS(语音合成)的一个应用，具有文本获取、语音合成和语音播放等基本功能。本文利用Visual C++进行程序设计，实现英文文本的朗读等功能。 -Text to speech system TTS (speech synthesis), an application has access to text, speech synthesis and voice playback function. In this paper, Visual C++ programming, th
Speech-Recognition-System
- 本文介绍了基于MATLAB的语音识别系统，包括对语音信号的特征提取，包括语音信号的特征提取，快速傅立叶变换，离散余弦转换，线性预测分析，梅尔频率倒谱系数以及高斯混合模型。-This paper aims at development and performance analysis of a speaker dependent speech recognition system using MATLAB® . The issues that were considered are 1
speech reconstruction+SLP
- This paper proposes a new variant of the least square autoregressive (LSAR) method for speech reconstruction, which can estimate via least squares a segment of missing samples by applying the linear prediction (LP) model of speech. First, we show t
speech coding based on SLP-2009
- This paper describes a novel speech coding concept created by introducing sparsity constraints in a linear prediction scheme both on the residual and on the prediction vector. The residual is efficiently encoded using well known multi-pulse excitat