Description: 语音合成程序！psalo频域基音同步叠加方法。它首先对原始语音信号进行短时频域变换，得到短时谱和短时谱包络。短时谱除以短时谱包络得到声源短时谱，对声源短时谱的实部和虚部分别进行线性插值，就可以达到改变语音信号基频的目的，然后再进行频域反变换，可得到变换后的短时语音信号。短时谱包络部分也可以独立改变，以达到改变音色的目的。-speech synthesis procedures! Psalo frequency domain pitch synchronous superposition method. It is first of the original speech signal for short-time frequency domain transform, to be short-term and short-term spectral envelope spectrum. Short-term spectrum divided by the short-time spectral envelope of the sound source to be short-term spectrum of the sound spectrum of short-term real and imaginary parts of linear interpolation, they could change the voice-frequency signals to the end, then we will proceed to frequency-domain transform, Transform available after the short speech signal. Short-term spectral envelope can be independent of changes to achieve the objective of changing colors. Platform: |
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Author:hzh |
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Description:
语音识别模块的性能指标：
高识别率：在一定的硬件环境下，平均准确率可达97％；
端点检测：在识别过程，具备端点检测功能；
实时识别：具有唤醒功能；
环境噪音：一定范围之内，仍可保证高识别率-speech recognition module performance indicators : high recognition rate : some of the hardware environment, average accuracy rate of up to 97%; Endpoint Detection : the identification process, with endpoint detection; Real-time Recognition : The wake-up function; Environmental noise : within a certain range, can ensure high recognition rate Platform: |
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Author:唐永锋 |
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Description: 自相关基音检测算法是语音信号处理的关键技术，算法的效率直接影响语音信号实时处理的质
量。-Autocorrelation pitch detection algorithm is the key to speech signal processing technology, a direct impact on the efficiency of algorithms in real-time speech signal processing quality. Platform: |
Size: 70656 |
Author:jiang1013nan |
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Description: Digital Signal and Image Processing Using MATLAB
The most important theoretical aspects of image and signal processing (ISP) for both deterministic and random signals are covered in this guide to using MATLAB® . The discussion is also supported by exercises and computer simulations relating to real applications such as speech processing and fetal-heart–rhythm tracking, and more than 200 programs and functions for numerical experiments are provided with commentary.-Digital Signal and Image Processing Using MATLABThe most important theoretical aspects of image and signal processing (ISP) for both deterministic and random signals are covered in this guide to using MATLAB Platform: |
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Author:andy |
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Description: 摘要 : 在 MAT LAB环境下利用语音工具箱 Voice Box实现基于连续概率密度隐含马尔科夫模型的汉语语音识别系统。在
实时录音的情况下 , 利用该语音识别系统 , 不同的人对 20条 2～8个字的语音命令进行识别 , 准确率可达到 95 ％ , 识别时间
115～3 s , 实现了小词汇量连续语音的非特定人的实时识别。-Abstract: In the circumstances the use of MAT LAB toolbox Voice Box Voice realize implied probability density based on continuous Markov model of Chinese speech recognition system. In real-time recording, the use of the voice recognition systems, different people on the 20 2 ~ 8-word voice command recognition accuracy rate can reach 95 percent, identify the time 115 ~ 3 s, the realization of a small vocabulary continuous voice of the non-specific real-time identification of people. Platform: |
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Author:tz1985 |
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Description: 一个基于PC声卡和matlab的实时语音信号采集分析系统，带有界面图形，功能较完善-A matlab based on PC sound card and the real-time voice signal acquisition analysis system with graphic interface, function better Platform: |
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Author:zhaoqingtian |
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Description: 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理。-Endpoint detection is the use of digital processing techniques to identify the speech signals in a variety of paragraphs (such as phoneme, syllable, morpheme, word, etc.) the starting point and the end position. Speech endpoint detection is a paragraph beginning and end speech analysis, speech synthesis and speech recognition of a necessary link. Traditional endpoint detection method is to obtain from the wav file voice sampling, its sub-frame and calculate the short-term energy and zero-crossing rate parameters, and then proceed to endpoint detection. This work is called off-line approach can not achieve real-time speech signal processing, analysis of the speech signal has a certain limitations. In this paper, through the development of ActiveX controls, in the MATLAB environment to embedded figure window to use GUI program, the realization of speech signal processing real-time endpoint detection. Platform: |
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Author:cike |
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Description: MATLAB语音识别的一个源代码，不是很成熟，只能识别英文，实时性比较差，但是能运行，可以做为实验代码和待修改的源代码。-Speech Recognition a MATLAB source code is not very mature, and only recognizes English, real-time is rather poor, but it can run, you can be used as experimental code and modify the source code. Platform: |
Size: 870400 |
Author:老蔡 |
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Description: 观测实时模拟信号（语音）的频谱
用音频设备采集一段语音，将语音存为.wav格式。对wav文件作分段傅里叶变换分析。语音是分音节的，应把它分段分析，而且实际运用中的数字信号处理的FFT的点数是有限的，一般只能达到千点。用傅里叶反变换IFFT，从频域恢复信号。画出频谱图和语音波形图。
-Observing real-time analog signal (voice) of the spectrum collected with the audio devices section of voice, voice saved as. Wav format. Wav file of the sub-Fourier transform analysis to make. Speech is divided into syllables, it should be sub-analysis, and practical application of digital signal processing of the FFT points is limited, generally only reach 1000 points. Fourier inverse transform IFFT, to restore the signal from the frequency domain. Draw the spectrum map and voice waveform. Platform: |
Size: 63488 |
Author:benny |
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Description: 信号频谱分析
1. 观测已知方波信号、正弦波信号的频谱
分析频率为4KHz、8KHz的占空比为50 的方波信号、正弦波信号的频谱。记录所得频谱图。
2. 观测实时模拟信号（语音）的频谱
用音频设备采集一段语音，将语音存为.wav格式。对wav文件作分段傅里叶变换分析。语音是分音节的，应把它分段分析，而且实际运用中的数字信号处理的FFT的点数是有限的，一般只能达到千点。用傅里叶反变换IFFT，从频域恢复信号。画出频谱图和语音波形图-Signal spectrum analysis 1. Observing a known square-wave signal, sine wave signal frequency spectrum analysis 4KHz, 8KHz for the 50 duty cycle square wave signal, sine wave signal frequency spectrum. The recorded frequency spectrum. 2. Observing real-time analog signal (voice) of the spectrum collected with the audio devices section of voice, voice saved as. Wav format. Wav file of the sub-Fourier transform analysis to make. Speech is divided into syllables, it should be sub-analysis, and practical application of digital signal processing of the FFT points is limited, generally only reach 1000 points. Fourier inverse transform IFFT, to restore the signal from the frequency domain. Draw maps and voice waveform spectrum Platform: |
Size: 64512 |
Author:benny |
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Description: 信号频谱分析
1. 观测已知方波信号、正弦波信号的频谱
分析频率为4KHz、8KHz的占空比为50 的方波信号、正弦波信号的频谱。记录所得频谱图。
观测实时模拟信号（语音）的频谱
用音频设备采集一段语音，将语音存为.wav格式。对wav文件作分段傅里叶变换分析。语音是分音节的，应把它分段分析，而且实际运用中的数字信号处理的FFT的点数是有限的，一般只能达到千点。用傅里叶反变换IFFT，从频域恢复信号。画出频谱图和语音波形图-Signal spectrum analysis 1. Observing a known square-wave signal, sine wave signal frequency spectrum analysis 4KHz, 8KHz for the 50 duty cycle square wave signal, sine wave signal frequency spectrum. The recorded frequency spectrum. Observing real-time analog signal (voice) of the spectrum collected with the audio devices section of voice, voice saved as. Wav format. Wav file of the sub-Fourier transform analysis to make. Speech is divided into syllables, it should be sub-analysis, and practical application of digital signal processing of the FFT points is limited, generally only reach 1000 points. Fourier inverse transform IFFT, to restore the signal from the frequency domain. Draw maps and voice waveform spectrum Platform: |
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Author:benny |
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Description: 语音识别技术的最终目标是要让计算机能与人自由交谈。目前，连续语音识别技术正趋于成熟，语音识别也延伸出了诸多实用化的研究方向。今后，语音识别的重点将集中在自然话语识别与理解、实时语音识别和语音识别鲁棒性等方面。作为一门交叉学科，语音识别所涉及到的技术有信号处理、模式识别、概率论和信息论、发声机理、听觉机理和人工智能等。-The ultimate goal of speech recognition technology is to make computers and allowing others to speak freely. At present, continuous speech recognition technology is maturing, voice recognition is also extended out a lot of practical research directions. In the future, speech recognition will focus on Spontaneous Speech Recognition and Understanding, real-time speech recognition and speech recognition robustness and so on. As an interdisciplinary subject, speech recognition technology involved in a signal processing, pattern recognition, probability theory and information theory, vocal mechanism, hearing mechanism and artificial intelligence. Platform: |
Size: 5483520 |
Author:吴斌 |
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Description: MATLAB結合HTK的特徵擷取應用SVM函式 的實際範例 並且可達到即時錄音辨識 輸出 前三個語音辨識的機率
-MATLAB with HTK feature extraction practical examples of application of SVM function and can reach the probability of the first three speech recognition in the real-time recording identification output Platform: |
Size: 972800 |
Author:ou-yang yen |
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Description: Preprocessing and trainning code for a speech recognition in real time in matlab. Look for extra.zip to have the GUI in matlab Platform: |
Size: 8192 |
Author:martal88 |
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Description: 这是我的adsp的project 维纳滤波在语音增强上的应用，是基于维纳滤波的语音增强的matlab实现，其中包括matlab源码，word文档，以及PPT。并设计了简单的
GUI 对语音信号进行加噪处理，然后再其进行维纳滤波。通过gui可以实时改变加入噪声的参数和维纳滤波器的参数，进行分析。并
有频谱显示和语谱对比显示。-This is my project adsp the Wiener filter applied on speech enhancement is based on the Wiener filter speech enhancement matlab implementation, including matlab source code, word documents, and PPT. And designed a simple GUI for adding noise speech signal processing, then it Wiener filtering. You can join the real-time changes in the parameters and noise parameters Wiener filter through gui, for analysis. And a spectrum display and spectral contrast display language. Platform: |
Size: 766976 |
Author:小毛孩 |
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Description: 通过matlab设计的GUI界面实现语音信号的实时录入和频谱分析和滤波等处理，并能够显示各阶段时域波形及频谱图，滤波器种类及参数可调(Through the GUI interface designed by matlab, the real-time input of speech signal, spectrum analysis and filtering are realized, and the time-domain waveforms and spectrograms of each stage can be displayed, and the types and parameters of filters can be adjusted.) Platform: |
Size: 100352 |
Author:电子信息小菜鸡 |
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Search - real time speech matlab