搜索资源列表
g729-annE-1998
- 嵌入式开发中，语音编码G729的算法。 8 kbit_s CS-ACELP语音编码算法-Embedded development, speech coding algorithm G729. 8 kbit_s CS-ACELP speech coding algorithm
DSP_DESIGN
- 基于C54X的DSP载波数字解调系统的设计，该硬件系统工作稳定，可以实现信号的实时处理，如可以实现语音信号的实时处理，通信信号的实时调制解调灯。在此基础上，为了验证DSP芯片的高性能，在此硬件平台上实现了两个具体应用：一个是高性能的FIR数字滤波器；一个是2FSK数字解调算法的实现。-C54X DSP-based carrier digital demodulation system design, the hardware system stability, can achieve the r
DLSSNBA
- 一种新的基于麦克风阵列的近场声源定位和语音分离算法，它结合双波束二维定位和近场最小方差波束形成技术在阵列近场范围内实现声源定位和语音分离。-based on a microphone array of near-field acoustic source localization and speech separation algorithm, It combines two-beam two-dimensional positioning and near-field minimum vari
ALGORITHM
- A SOFT MODEL-ORDER SUBSPACE BASED SPEECH ENHANCEMENT ALGORITHM
DTW
- 特定人语音识别算法_DTW算法，语音识别论文-Dependent Speech Recognition Algorithm _DTW algorithm, speech recognition thesis
052520
- 提出了一种用于矢量量化的改进的聚类算法,该算法在MKM(Modified K-Means)算法的框架的基础上,对初始码本的生成、失真测度的选择、*型胞腔的处理等方面进行了改进,从而减少了原算法在能量和增益上对聚类结果的影响.并将该算法应用于波形编辑孤立字识别器,这种识别器直接对语音样本的时域波形进行训练和聚类,不需要提取语音参数,算法复杂度较低,加上提出的聚类算法失真测度简单易实现,对芯片的运算能力要求不高,非常适用于有低成本要求的语音识别器场合.通过中文元音字识别的实验证明,在相同码本尺寸下
Burg2
- 线性预测是语音信号分析中用得比较多的一种方法#&’()算法在线性预测中占有重要地位#格型滤波 器在语音信号分析中起了重要作用$本文采用&’()算法的逆运算对语音信号进行合成#得到了较好的合成效果$ -Linear prediction analysis of speech signal is used more than one way#& ' () linear prediction algorithm plays an important role in cell-typ
modelbasedonspectrumprediction
- 文章展示了基于高斯混合模型的语音频谱预测方法。频谱预测可能在传包过程中预防丢包这方面起到大作用。期望最大化算法用两倍或三倍的连续语音因素来测试模型。模型被用来设计第一，儿等指令预测量。预测表用频谱分配状态来估计并和一个简单的参考模型对比。最好的预测表得到一个平均频率扭曲值是0.46dB小于参考模型-This paper presents methods for speech spectrum prediction based on Gaussian mixture models. Spec
micphonearray
- 几篇关于麦克风阵列语音增强算法的论文，对语音处理有一定的帮助，-Several microphone array speech enhancement algorithm on paper, on the voice processing have some help,
Efficient-LSP-quantization-algorithm
- 为在极低速率下实现高质量的语音编码,提出了一 种新的有效的线谱对( LSP)参数量化算法——P-RS- MSM Q 算法。此算法以多帧联合矩阵量化作为基本框架,引入了基 于超级帧模式的均值去除和帧间预测策略、 矩阵分裂和子矩 阵多级量化策略 同时提出了基于语音帧短时谱能量的帧 内加权和基于超级帧中各子帧重要性的帧间加权策略等。-Very low rates in high-quality speech coding, a new and effective LSP (LSP)
an-adaptive-algorithm-for-mel-cepstral-analysis-o
- hidden markov model based text to speech
speech-enhancement
- 本文介绍了一种基于自适应滤波算法的语音信号增强处理的方法。文中在扼要介绍了目前常用的语音增强方法的基础上，重点介绍了采用LMS算法的自适应语音增强系统。对三种LMS算法（基本LMS算法,符号误差LMS算法及NLMS算法）进行仔细研究，并用Matlab仿真实现，验证比较了三种算法的语音增强效果。-In this paper, a system of speech enhancement is discussed based on self-adaptive algorithm. On the ba
shuangmaikefengjiangzao
- 单麦克风和双麦克风语音增强系统的研究，这是基础的双麦克风降噪算法，但是很实用-Single microphone and dual-microphone speech enhancement system, which is based dual-microphone noise reduction algorithm, but very practical
jiyuzhend-epujiafa
- 基于帧间重叠谱减法的语音增强算法及实现，可以进作为基础的谱减法算法学习-Overlapping spectral subtraction speech enhancement algorithm based on interframe achieve, you can enter as spectral subtraction algorithm based on learning
LMS
- LMS算法多麦克风语音降噪，语音降噪处理，基于matlab-LMS algorithm for multi-microphone speech noise reduction, voice, noise reduction, based on matlab
speech-coummunication-
- In this paper we present a robust voice activity detection algorithm which enables further bit rate reduction for practical speech communication systems. -In this paper we present a robust voice activity detection algorithm which e
Speech-recognition-in-MATLAB-
- 介绍了一种基于MATLAB的多个特定人连接词语音识别的方法，并提出了在进行端点检测时，引入平均的概念能进一步提高识别率。此设计是以LPCC系数、DTW算法为核心的基于图形界面的设计。-A Based on MATLAB more specific conjunctions speech recognition, and during endpoint detection, the introduction of the concept of average to further improve
cs-speech-enhancement
- 文利用带噪语音经特征基函数矩阵转换后所具有的稀疏特性，用最大似然估计方法对转换后得到的稀疏 分量进行非线性压缩去噪，然后再经过反变换和重构恢复出原始语音信号的估计。特征基函数矩阵反映了语音数据本 身的统计特性，因此具有很好的合理性和可取性。仿真结果表明利用稀疏编码方法能极大程度地抑制背景噪卢，与小波消噪法相比优势明显。-a speech enhancement algorithm based Compressed Sensing.
A-Speech-Analysis-Algorithm.-Model-Reference-Adap
- A Speech Analysis Algorithm Which Eliminates the Influence of Pitch Using the Model Reference Adaptive System
A-New-Word-Separation-Algorithm-for-Continuous.ra
- A New Word Separation Algorithm for Continuous Bangia Speech Recognition