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Description: This paper deals with the problem of speech enhancement when a
corrupted speech signal with an additive colored noise is the only
information available for processing. Kalman filtering is known as
an effective speech enhancement technique, in which speech signal
is usually modeled as autoregressive (AR) process and represented
in the state-space domain.-This paper deals with the problem of speech enhancement when a corrupted speech signal wit h an additive colored noise is the only informat ion available for processing. Kalman filterin g is known as an effective speech enhancement te chnique. in which speech signal is usually modeled as aut oregressive (AR) process and represented in th e state-space domain.
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Size: 103119 |
Author: rifer |
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Description: This paper deals with the problem of speech enhancement when
only a corrupted speech signal is available for processing. Kalman
filtering is known as an effective speech enhancement technique,
in which speech signal is usually modeled as autoregressive (AR)
model and represented in the state-space domain.-This paper deals with the problem of speech enhancement when only a corrupted speech signa l is available for processing. Kalman filterin g is known as an effective speech enhancement te chnique. in which speech signal is usually modeled as aut oregressive (AR) model and represented in the s tate-space domain.
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Size: 245868 |
Author: rifer |
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Description: This paper deals with the problem of speech enhancement when a
corrupted speech signal with an additive colored noise is the only
information available for processing. Kalman filtering is known as
an effective speech enhancement technique, in which speech signal
is usually modeled as autoregressive (AR) process and represented
in the state-space domain.-This paper deals with the problem of speech enhancement when a corrupted speech signal wit h an additive colored noise is the only informat ion available for processing. Kalman filterin g is known as an effective speech enhancement te chnique. in which speech signal is usually modeled as aut oregressive (AR) process and represented in th e state-space domain.
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Size: 102400 |
Author: rifer |
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Description: This paper deals with the problem of speech enhancement when
only a corrupted speech signal is available for processing. Kalman
filtering is known as an effective speech enhancement technique,
in which speech signal is usually modeled as autoregressive (AR)
model and represented in the state-space domain.-This paper deals with the problem of speech enhancement when only a corrupted speech signa l is available for processing. Kalman filterin g is known as an effective speech enhancement te chnique. in which speech signal is usually modeled as aut oregressive (AR) model and represented in the s tate-space domain.
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Size: 245760 |
Author: rifer |
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Description: 研究了用 C 语言对孤立词语音识别算法进行了仿真，对每个算法的效率、时间复杂性、
空间复杂性进行深入的讨论和分析，-Studied using C language on an isolated word speech recognition algorithm simulation, the efficiency of each algorithm, time complexity, space complexity of an in-depth discussion and analysis,
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Size: 620544 |
Author: 在式朝 |
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Description: 语音端点 检测的一种方法.叫空间信号子算法-Speech Endpoint Detection methods. Called space signal subspace algorithm
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Size: 1024 |
Author: 小孙 |
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Description: 自动语音识别源码对非特定人的孤立词语音识别算法
进行研究。考虑到语音识别技术在嵌入式系统处理能力较弱、存储空间有限的问
题上进行算法的选择和优化，以提高系统的鲁棒性，减少系统的
-Automatic speech recognition source code for non-specific human isolated word speech recognition algorithm research. Taking into account the voice recognition technology in the embedded systems processing capacity is weak, with limited storage space on the issue of algorithm selection and optimization, to improve system robustness, reduce system
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Size: 156672 |
Author: cq |
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Description: 该程序是基于子空间的语音去噪声,该方法使用广泛,能取得很好的语音增强效果!-The program is based on the sub-space of the voice to noise, the method widely used and can achieve very good speech enhancement effect!
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Size: 1024 |
Author: zhjuna |
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Description: 包含了词条及其词性标记，频度信息的词典 练语料的格式要求：
每个词以 / 分隔， / 后是该词的词性标记。词性标记后至少要有一个空格。一个句子的所有词必须在同一行中。击“开始词性标注”选取文本文件（一次可以选择多个）进行标注处理-Includes a term and its part of speech marks, the frequency of information and training Corpus dictionary format requirements: Each word in/separated/post is part of speech marking term. POS tag at least one space after. A sentence of all the words must be in the same line. Hit Start-of-speech tagging select text file (one can select multiple) to mark deal
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Size: 174080 |
Author: 张耀 |
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Description: 该程序是非特定人语音识别程序，原来在SPCE500A下运行，能识别并应答16个命令，存储空间为128K字节。-The procedures were non-specific speech recognition procedure, the original run in SPCE500A can identify and response to 16 orders for the 128K bytes of storage space.
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Size: 121856 |
Author: 微风 |
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Description: 从x86寻址方式开始讲了Linux的分段分页方式、内核空间和用户空间、Linux汇编等。-X86 mode from the beginning of a speech addressing the sub-paging mode Linux, the kernel space and user space, Linux and other compilation.
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Size: 1332224 |
Author: 杨阳 |
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Description: 动态时间规整 算法是一种运用动态规划原理计算时间矢量相似度的方法 在语音识
别等领域有着广泛的应用 窗口法是针对 算法的特点 采用了动态高效的空间管理办法 解
-
!"#
决了通常的 算法需要存储一个较大的矩阵 因而在存储空间有限的硬件系统中难以实现的问题 窗口法已经在实际系统中得到了应用
-Dynamic time warping algorithm is a calculation using the principle of dynamic programming time for vector similarity methods in areas such as in speech recognition has a wide application window for the algorithm is characterized by the use of a dynamic and efficient approach to space management solution- ! "# Summary algorithms usually need to store a large matrix and therefore of limited storage space in the hardware system of the problem difficult to achieve in the window system has been in practice has been applied
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Size: 90112 |
Author: sdc |
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Description: 提出一种有效的语音增强算法，该算法使用重叠相加FF-r滤波器进行谱幅度估计，给出噪声估计的两个均值参数，此参数适用于任何噪声．而且进行噪声估计无需检测语音间隙，无需VAD的运算过程，克服了单信道噪声
压缩算法的缺点．-An effective speech enhancement algorithm using the overlap-add FF-r filter in spectral amplitude estimation, given the noise mean estimate two parameters, this parameter applies to any noise. And no detection of speech to noise estimation space, without VAD computing process, overcome the single-channel noise suppression algorithm shortcomings.
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Size: 193536 |
Author: Decheng Yu |
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Description: 一种改进的语音增强算法的
子空间上的先进。它可以减少残留噪声引起的 -An Improved Algorithm for Speech Enhancement
Advanced sub-space. It can reduce the residual noise caused by
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Size: 263168 |
Author: 刘翔 |
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Description: 背单词程序：
程序使用词汇表文件，其中每个词条由单词和解释两部分组成，编程完成：用户输入一个单词，程序将给出单词的词性和中文意思。
建立一个词汇表测试用词汇表名为新东方红宝书.txt，用c语言文件操作打开该文件，并将文件读入命名为dictionary[DICTSIZE]的数组内。读入数据存储于dictionary[DICTSIZE]所分配的地址空间内，以8字节表示一个信息。-Program uses the vocabulary file, where each entry and explain the words of two parts, programming to complete: user enters a word, the program will give part of speech and the Chinese words mean. Test to establish a glossary of new vocabulary called Eastern Red Book. Txt, file operations with the c language to open the file and read the file named dictionary [DICTSIZE] within the array. Read the data stored in the dictionary [DICTSIZE] within the allocated address space, a message of 8 bytes.
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Size: 414720 |
Author: 张雯 |
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Description: LBG分类算法
用初始室心随机法和扰动因子分裂法两种方法，比较不同方法不同参数设置时的分类性能。
-LBG classification algorithm vector quantization: vector normalization within a certain range for a particular type, consists of two steps: first generate a codebook, which is the speech feature vector space by the first process- also known as clustering speech parameter sequence as a vector, the reference code for classified- also known as quantization. Clustering algorithm: it is relatively simple and commonly used K-means clustering algorithm. LBG is a clustering algorithm, which is generally assumed that the codebook size is fixed, and for a power of 2. Codebook is small, then expanding until it reaches the requirements. It is often an existing classification split into two subclasses, and initial value with the new code word to each subclass. LBG algorithm on random data and a certain regularity (and meet certain Gaussian distribution) data classification, and look at the performance of the LBG algorithm, the initial chamber heart random disturbance factor-secession law are two
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Size: 86016 |
Author: zzc |
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Description: WINCE 5.0 捷通语音合成引擎测试版，成熟稳定，提供例子程序和DLL以及LIB和头文件，提供技术支持。
技术咨询QQ：1005974096
WINCE 语音合成有下列技术特点：
1，声音清晰（清脆）
2，语速可以调节
3，占用系统资源少，对系统要求少
4，代码占用空间少
5，编程接口方便
6，声音甜美
7，技术成熟可靠
8，支持C++、C#、VB.NET语言编程-WINCE 5.0 Jietong speech synthesis engine beta, mature and stable, providing examples of programs and DLL' s and LIB and header files to provide technical support. Technical Advisory QQ: 1005974096 WINCE speech synthesis following technical features: a clear, sound (crisp), speed can be adjusted, occupying less system resources, system requirements, the code space, programming interface to facilitate sweet voice 7, mature and reliable technology 8, support C++, C#, VB.NET programming language
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Size: 1088512 |
Author: softn |
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Description: WINCE 6.0 捷通语音合成引擎测试程序，成熟稳定，提供例子程序和DLL以及LIB和头文件，提供技术支持。
技术咨询QQ：1005974096
WINCE 语音合成有下列技术特点：
1，声音清晰（清脆）
2，语速可以调节
3，占用系统资源少，对系统要求少
4，代码占用空间少
5，编程接口方便
6，声音甜美
7，技术成熟可靠
8，支持C++、C#、VB.NET语言编程-WINCE 6.0 Jietong speech synthesis engine test program, mature and stable, and provides examples of programs and DLL' s and LIB and header files, and provide technical support. Technical Advisory QQ: 1005974096 WINCE speech synthesis following technical features: a clear, sound (crisp), speed can be adjusted, occupying less system resources, system requirements, the code space, programming interface to facilitate sweet voice 7, mature and reliable technology 8, support C++, C#, VB.NET programming language
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Size: 1085440 |
Author: softn |
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Description: 于市售 STM32 开发板上实现特定人语音识别处理项目。识别流程是：预滤波、ADC、分帧、端点检测、预加重、加窗、特征提取、特征匹配。端点检测(VAD)采用短时幅度和短时过零率相结合。检测出有效语音后，根据人耳听觉感知特性,计算每帧语音的 Mel 频率倒谱系数(MFCC)。然后采用动态时间弯折(DTW)算法与特征模板相匹配,最终输出识别结果。先用Matlab对上述算法进行仿真，经数次试验求得算法内所需各系数的最优值。而后将算法移植到 STM32 开发板上，移植过程中根据 STM32 上存储空间相对较小、计算能力也相对较弱的实际情况，对算法进行优化。最终完成于 STM32 微处理器上的特定人语音识别系统。-Implement speech recognition processing project in commercially available STM32 development board. Identification is the process: pre-filter, ADC, framing, endpoint detection, pre-emphasis, windowing, feature extraction, feature matching. Endpoint detection (VAD) short-time amplitude and short-term zero rate combined. After detecting an effective voice, according to the characteristics of human auditory perception, calculated for each frame of speech Mel Frequency Cepstral Coefficients (MFCC). Then dynamic time warping (DTW) algorithm and feature template matches the final output recognition result. First with Matlab simulation algorithm described above, after several trials to get the optimal value of each coefficient within the desired algorithm. The algorithm will migrate to STM32 development board, the porting process according to the STM32 relatively small storage space, computing power is relatively weak situation of the optimization algorithm. Finally completed on the STM32 micr
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Size: 325632 |
Author: Chenkly |
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Description: Sam是一个用C编写的非常小的文本到语音（TTS）程序，运行在大多数流行的平台上。它是由Don't Ask software（现为SoftVoice，Inc.）于1982年出版的Commodore C64语音软件SAM（软件自动嘴）的C语言改编而成。它包括一个称为背诵器的文本到音素转换器和一个用于最终输出的音素到语音例程。它是如此的小，以至于它也可以在嵌入式计算机上工作。在我的电脑上，它占用的磁盘空间不到39KB（在嵌入式设备上要小得多，因为不需要执行开销），而且是一个完全独立的程序。为了立即输出，它使用SDL库，否则可以保存.wav文件。(Sam is a very small Text-To-Speech (TTS) program written in C, that runs on most popular platforms. It is an adaption to C of the speech software SAM (Software Automatic Mouth) for the Commodore C64 published in the year 1982 by Don't Ask Software (now SoftVoice, Inc.). It includes a Text-To-Phoneme converter called reciter and a Phoneme-To-Speech routine for the final output. It is so small that it will work also on embedded computers. On my computer it takes less than 39KB (much smaller on embedded devices as the executable-overhead is not necessary) of disk space and is a fully stand alone program. For immediate output it uses the SDL-library, otherwise it can save .wav files.)
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Size: 55296 |
Author: fa_Q |
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