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[
Speech/Voice recognition/combine
]
voice464
DL : 0
基于dtw算法的语音识别 1 ENFRAME.M和MELBANKM.M取自voicebox工具箱 2 vad.m实现端点检测 3 mfcc.m计算mfcc参数 4 dtw.m实现DTW算法[训练] dtw2.m实现DTW高效算法 5 testdtw.m测试程序-based speech recognition algorithm and a ENFRAME.M MELBANKM.M from v oicebox Toolbox 2 vad.m achieve endpoint detection three mfcc.m mfcc calculation parameters four dt w.m achieve DTW [training] dtw2.m DTW achieve efficient algorithm for measuring 5 testdtw.m test procedures
Date
: 2008-10-13
Size
: 751.36kb
User
:
q024100404
[
Speech/Voice recognition/combine
]
voice464
DL : 0
基于dtw算法的语音识别 1 ENFRAME.M和MELBANKM.M取自voicebox工具箱 2 vad.m实现端点检测 3 mfcc.m计算mfcc参数 4 dtw.m实现DTW算法[训练] dtw2.m实现DTW高效算法 5 testdtw.m测试程序-based speech recognition algorithm and a ENFRAME.M MELBANKM.M from v oicebox Toolbox 2 vad.m achieve endpoint detection three mfcc.m mfcc calculation parameters four dt w.m achieve DTW [training] dtw2.m DTW achieve efficient algorithm for measuring 5 testdtw.m test procedures
Date
: 2025-12-29
Size
: 751kb
User
:
q024100404
[
Speech/Voice recognition/combine
]
TTS1
DL : 0
用VC实现的语音库的建立的程序. 首先建立单音节语音库,然后将所有的单音节语音文件合成一个大的.wav文件,并且制定索引机制,便于快捷查找-VC realize voice with the establishment of library procedures. First of all, the establishment of monosyllabic speech database, and then all of the monosyllabic voice synthesis of a large document. Wav files, and to develop indexing mechanisms to facilitate the quick search
Date
: 2025-12-29
Size
: 1.92mb
User
:
powerpl
[
Speech/Voice recognition/combine
]
newsounds
DL : 0
包括1千多个中文字音的朗读,和英文字母和数字的朗读,是一个很好的,齐全的音屏文件包,提供给大家分享.希望大家如果有语音库的资料也别望与我联系.-Including a more than 1000 Chinese reading a divergence in pronunciation, and English letters and numbers read aloud, is a good, full-screen sound package, available to share with you all. Hope that if there is voice database and I do not look Contact.
Date
: 2025-12-29
Size
: 1.76mb
User
:
杨加昌
[
Speech/Voice recognition/combine
]
SpeechSynthesisTechnologybasedonFreeTTS
DL : 0
Java 提供了语音API技术用于支持应用程序或W eb 页面中的人机语音交互, 其核心技术包括语音合成和语音识别。FreeTTS 是Java 语音合成技术的实现手段之一, 以创建电话号码朗读程序为例, 探讨了基于该技术的Java 语音API实现方法及相关问题。-API provides a Java technology used to support voice applications, or W eb page man-machine voice interaction, and its core technologies, including speech synthesis and speech recognition. FreeTTS speech synthesis technology is a Java implementation of one of the means to create a reading program phone numbers for example, to explore the technology based on the Java Speech API implementation methods and related issues.
Date
: 2025-12-29
Size
: 268kb
User
:
吕鹏
[
Speech/Voice recognition/combine
]
tts
DL : 0
This function requires the Mirosoft Win32 Speech API (SAPI). Examples: Speak the text tts( I can speak. ) List availble voices tts( I can speak. , List ) Do not speak out, store the speech in a variable w = tts( I can speak. ,[],-4,44100) wavplay(w,44100) - This function requires the Mirosoft Win32 Speech API (SAPI). Examples: Speak the text tts( I can speak. ) List availble voices tts( I can speak. , List ) Do not speak out, store the speech in a variable w = tts( I can speak. ,[],-4,44100) wavplay(w,44100)
Date
: 2025-12-29
Size
: 1kb
User
:
G眉ven
[
Speech/Voice recognition/combine
]
mic1
DL : 0
There are four major types of adaptive filtering configurations adaptive system identification, adaptive noise cancellation, adaptive linear prediction, and adaptive inverse system. All of the above systems are similar in the implementation of the algorithm, but different in system configuration. All 4 systems have the same general parts an input x(n), a desired result d(n), an output y(n), an adaptive transfer function w(n), and an error signal e(n) which is the difference between the desired output d(n) and the actual output y(n). In addition to these parts, the system identification and the inverse system configurations have an unknown linear system u(n) that can receive an input and give a linear output to the given input [2].-There are four major types of adaptive filtering configurations adaptive system identification, adaptive noise cancellation, adaptive linear prediction, and adaptive inverse system. All of the above systems are similar in the implementation of the algorithm, but different in system configuration. All 4 systems have the same general parts an input x(n), a desired result d(n), an output y(n), an adaptive transfer function w(n), and an error signal e(n) which is the difference between the desired output d(n) and the actual output y(n). In addition to these parts, the system identification and the inverse system configurations have an unknown linear system u(n) that can receive an input and give a linear output to the given input [2].
Date
: 2025-12-29
Size
: 123kb
User
:
maidenfreak
[
Speech/Voice recognition/combine
]
speech-recognition-system
DL : 0
A robust speech recognit ion system based on discriminative learning of environmental features is proposed for recognit ion of environmental features in high noise background, and a g radient descent algo rithm was adopted for parameters opt imization. Ex periments w ere carried out at different levels of back g round noise for SNR, basic accuracy, noise- resistance and system environment adaptability. T he experi mental result s show that the system has good recog nition performance in high noisy environments. The sys tem can meet different needs of applicat ion.
Date
: 2025-12-29
Size
: 164kb
User
:
yanyeguo
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