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语音信号混合与分离源代码
最经典的基于ICA实现的语音信号的采集、随机混合，再通过盲分离将混合后的语音信号分离(The most classical ICA-based speech signal acquisition, random mixing, and then the mixed speech signal is separated by blind separation.)
- 2020-06-25 12:00:02下载
- 积分：1
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最终代码说话人识别
实现了基于特定话语的聚类LGB和VQ的说话人识别(Speaker Recognition Based on Clustering LGB and VQ)
- 2019-05-20 15:04:44下载
- 积分：1
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chapter2-low-bit-rate-speech-coding
Chapter 2 Low Bit-Rate Speech Coding
In this chapter an overview is given of speech coding techniques at several bit rates. Most
of them use Linear Prediction. This overview is not meant to be complete its purpose is
to make the reader somewhat familiar with Linear Predictive Coding which is necessary for
a proper understanding of later chapters. Section 2.1 treats the subject of quantisation and
coding. In section 2.2 a description of speech production and speech sounds is given. Coders
based on linear prediction can be considered as being based on a simple speech production
model. This model is explained in section 2.3. Section 2.4 describes various speech coding
algorithms and techniques. Section 2.5 briefly describes some measures for the quality of
coded speech.(In this chapter an overview is given of speech coding techniques at several bit rates. Most
of them use Linear Prediction. This overview is not meant to be complete its purpose is
to make the reader somewhat familiar with Linear Predictive Coding which is necessary for
a proper understanding of later chapters. Section 2.1 treats the subject of quantisation and
coding. In section 2.2 a description of speech production and speech sounds is given. Coders
based on linear prediction can be considered as being based on a simple speech production
model. This model is explained in section 2.3. Section 2.4 describes various speech coding
algorithms and techniques. Section 2.5 briefly describes some measures for the quality of
coded speech.)
- 2010-07-02 19:42:42下载
- 积分：1
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Roomsim
根据房间配置信息生成房间传函，进而生成模拟语音数据(Depending on the room configuration generates a room transfer function, thereby generating an analog voice data)
- 2016-06-02 18:26:54下载
- 积分：1
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GMM-MFCC
基于GMM的MFCC算法的说话人识别，Maltab的高斯混合模型，12维。(MFCC speaker recognition algorithm based on GMM)
- 2015-03-26 19:59:19下载
- 积分：1
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Dereverberation
通常语音信号在增强时会出现混响现象，演讲者为了消除背景混响，不得不频繁地偏转头部的方向，这样会造成脉冲响应的不断改变。我们结合盲解卷法和频谱消去法来提高逆滤波器的滤波效果。我们利用输入语音信号间的相关系数矩阵计算出稳定、精确的室内脉冲响应的逆滤波器，而这些输入信号无需测量室内的脉冲响应就能被观测到。逆滤波能够消除早期的反射，这些反射包含混响中的绝大部分能量。之后，用频谱消去法来抑制逆滤波后的信号的尾部混响。本方法在实际适应性方面的表现通过具体的实验进行了验证，结果表明盲解卷法和频谱消去法的结合相较于单独使用一种方法，能够提供一个更优越的演说环境。(Usually voice signal will be enhanced when the reverberation phenomenon, the speaker in order to eliminate the background reverberation, and had to frequently deflect the direction of the head, this will cause the impulse response of changing. We combine the blind deconvolution and spectral elimination method to improve the filtering effect of the inverse filter. We use the correlation coefficient matrix between the input speech signal to calculate the stable, accurate inverse filter of the indoor impulse response, and these input signals do not need to measure the room impulse response can be observed. Inverse filtering to eliminate early reflections, these reflections contains most of the energy in the reverberation. After the spectrum elimination method to suppress the tail reverberation of the signal after the inverse filtering. Performance in the actual adaptation of the method by specific experimental validation results show that the blind deconvolution and spectrum to eliminate )
- 2012-04-17 22:18:00下载
- 积分：1
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HMM
HMM语音识别,包含主要算法,进行语音识别(HMM speech recognition, contains the main algorithm for speech recognition)
- 2021-04-12 21:18:56下载
- 积分：1
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EM_init_kmeans
高斯混合模型参数初始化程序，在对高斯混合模型的建立之前采用KMEANS算法进行初始化(Gaussian mixture model parameter initialization procedure, in the Gaussian mixture model is initialized before the algorithm used KMEANS)
- 2010-11-18 20:28:33下载
- 积分：1
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GSM_CHANNLE_ENCODE
本文讲述了GSM系统空中接口的信道编码,包括语音信道和控制信道的编码方式和编码实现(This paper describes the GSM system, air interface channel coding, including the voice channel and control channel coding and Coding)
- 2010-07-25 14:18:57下载
- 积分：1
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Digital-Voice-Processing-
本书系统地阐述了语音信号处理的原理、方法、技术和应用，同时给出了部分内容对应的MATLAB仿真源程序。全书共12章，第1章至第7章是基本理论部分，包括语音信号的数字模型、语音信号的短时时域分析和频域分析、语音信号的同态处理、语音信号线性预测分析和矢量量化；第8章至第12章是应用部分，包括语音编码、语音合成、语音识别、语音增强和语音处理的实时实现。本书内容全面，重点突出，原理阐述深入浅出，注重理论与实际应用的结合，可读性强。(This book describes the speech signal processing principles, methods, techniques and applications, and gives the corresponding part of the contents of the MATLAB simulation source. The book is 12 chapters, Chapter 1 to Chapter 7 is the basic theoretical part, including voice signal digital model, speech signal analysis in time domain and frequency domain analysis, speech signal homomorphic processing, speech signal analysis and linear prediction vector quantified Chapter 8 to Chapter 12 is the application of parts, including speech coding, speech synthesis, speech recognition, speech enhancement and voice processing, real-time implementation. The book is comprehensive, focused and Rationale layman, focusing on the combination of theory and practical application, readable.)
- 2021-05-16 00:30:03下载
- 积分：1
