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specsub_modified11
语音唤醒就是在设备待机状态下，用户说出特定的语音指令（唤醒词）使设备进入工作状态或完成某一操作。设备为了实现语音唤醒功能，就需要设备实时监听，也就是实时的录音并分析有没有唤醒词来唤醒设备。评价一个语音唤醒系统的标准有三个：唤醒正确率、误报率、功耗。一般情况下，唤醒正确率越高 ，误报率也越高。好的系统就需要唤醒率高，误报率低，功耗低。(Speech wakeup is when the device is in standby mode, the user speaks a particular voice command (wake up word), so that the device enters the work state or completes an operation. In order to realize the function of speech wakeup, the device needs real-time monitoring, that is, real-time recording and analyzing whether or not wake words are used to wake up the equipment. There are three criteria for evaluating a voice wakeup system: wakeup accuracy, false positive rate, and power consumption. Generally, the higher the correct rate of arousal, the higher the false positive rate. A good system requires high wake-up rate, low false positive rate and low power consumption.)
- 2017-08-15 16:45:55下载
- 积分：1
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LMS
主要是研究光纤通信系统中的克服PMD效应的电域均衡技术，构建电域均衡器的模型并进行仿真实验。找到补偿PMD效应的较理想的电域均衡器结构。(Is to study, fiber-optic communication systems to overcome the effects of PMD electrical domain equalization technology, building a power domain equalizer model and simulation experiments. PMD compensation effect to find a better balance of power domain structure.)
- 2008-05-28 19:44:28下载
- 积分：1
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MMSE_IC
基于LMS(最小均方误差算法)的自适应滤波 基于LMS(最小均方误差算法)的自适应滤波(based on the LMS (MMSE) algorithm based on the LMS adaptive filtering (minimum mean square error algorithm) the adaptive filtering based on the LMS (minimum mean square error algorithm) Adaptive Filter)
- 2020-06-29 23:00:01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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lunwen
本文主要研究普通计算环境下语音自动识别(ASR)技术的软件实时实现，基于语音识别的基本原理和过程。在VC环境下建立了一套语音识别系统及其演示软件。实验表明，该系统对特定人情况有较高的识别率和一定的实时性(In this paper, under the computing environment on general automatic speech recognition (ASR) technology to achieve real-time software, speech recognition based on the basic principles and processes. VC environment in the establishment of a voice recognition system and its presentation software. The experiments show that the systems are in particular have a high recognition rate and a certain degree of real-time)
- 2009-06-06 15:14:09下载
- 积分：1
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为均衡带限信号所引起失真的横向或格型自适应均衡器
为均衡带限信号所引起失真的横向或格型自适应均衡器(其中横向FIR系统长M=11), 系统输入是取值为±1的随机序列 ，其均值为零；参考信号 ；信道具有脉冲响应：
式中w用来控制信道的幅度失真(w = 2～4，例如，取w = 2.9,3.1,3.3,3.5等),而且信道受到均值为零、方差为 (例如，取 ，相当于信噪比为30dB)的高斯白噪声 的干扰。试比较基于下列五种算法自适应均衡器在不同信道失真、不同噪声干扰下的收敛情况(对应于每一种情况，在同一坐标下画出其学习曲线)：
横向/格-梯型结构LMS算法[1-4]
横向/格-梯型结构RLS算法[1-4]
()
- 2007-09-12 20:18:58下载
- 积分：1
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LMSjiangzao
LMS多麦克风语音降噪的主程序是lmsspdn.m(Multi-microphone noise reduction LMS voice is the main program lmsspdn.m)
- 2020-07-04 19:00:01下载
- 积分：1
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kpca
核主成分分析法,用matlab实现,极为精彩.不可错过(Kernel Principal Component Analysis method, using matlab realize, is extremely exciting. Not to be missed)
- 2008-08-18 19:03:01下载
- 积分：1
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hhspectrum
关于希尔伯特黄频谱的计算程序，提取的频谱信息可用于语音识别、故障检测等。(on Hilbert Huang spectrum calculation procedures to extract information of the spectrum can be used for voice recognition, fault detection.)
- 2007-03-22 16:19:26下载
- 积分：1
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Audio-Watermarking-master
鲁棒数字音频盲水印水印置乱算法实现基于阿诺德变换的离散小波变换，离散余弦变换(Robust Blind Digital Audio Watermarking
Implementation Based on Watermark Scrambling Algorithm-Arnold Transform
Discrete Wavelet Transform,Discrete Cosine Transform,Erro Correcting Code)
- 2018-04-16 12:52:31下载
- 积分：1
