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Description: 该水印算法具有以下特点：1）结合音频自身的特性，不仅能够自适应划分音频数据段，而且能够智能调节水印嵌入强度，增强了数字音频水印的鲁棒性；2）能够同时进行数字音频的版权保护与内容认证，并可大致确定篡改发生区域，仿真实验表明，文中算法不仅具有较好的不可感知性，而且对常规处理具有较好的鲁棒性，同时能够对替换等而已篡改做出报警并确定被篡改位置。-The watermarking algorithm has the following characteristics: 1) combined with the audio of its own characteristics, not only adaptive division of the audio data segment, but also can adjust watermark embedding intelligent strength, enhanced digital audio watermark robustness 2) can at the same time digital audio copyright protection and content authentication, and can generally determine the tampering occurred in the region, simulation experiments show that the algorithm not only has better not be perceived, but also on conventional treatment has good robustness, at the same time be able to tamper with it to make replacement alarm and to determine the location has been tampered with.
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Size: 2048 |
Author: byleimin |
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Description: 采集0-7路输入来自直流源经分压器产生的0～5 V直流电压（可功能扩展，8路可接不同的传感器，采集不同的信号,例如:温度、湿度、图象、声音等）ADC0809将各路模拟信号分别转换成8位二进制数字信号,再对各路数据进行显示。各路通道采集方式为以约5s为周期循环采集（即0路、1路……7路、0路…）约每5ms更新，重新采集一次。显示方式为循环显示和单路显示：开关打开时，循环显示0-7路采集的数据；开关闭合时，显示当前单路采集的数据。3个七段数码管显示结果:最左边的一个数码管DS0显示通道号，其它两个数码管DS1和DS2显示采集此通道的经AD转换后十六进制数据结果，范围为:00H-FFH。开关再打开时则继续循环显示。在键盘上按下任意键则退出DOS。-0-7 Acquisition of DC input source from the voltage divider created by the 0 ~ 5 V DC voltage (which can function expansion, 8-way which can be accessed by different sensors, different signal acquisition, such as: temperature, humidity, image, audio and so on) ADC0809 the various analog signals, respectively, into 8-bit binary digital signal, and then displayed on the various data. Various channels for acquisition of about 5s for the acquisition cycle (ie, 0 Road, 1 Road ... ... 7 Road, 0 path ...) updated every 5ms, re-acquisition time. Display for the display and one-way cycle show: switch is open, the cycle shows that the way data collected 0-7 open to close time, shows the current single-channel data acquisition. Seven-Segment LED 3 shows the results: the far left shows a digital control DS0 channel number, the other two DS1 and DS2 digital tube display this channel collected by the AD conversion of the hexadecimal data, range: 00H-FFH . Switch is opened again to continue the cycl
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Size: 1024 |
Author: 廖婷 |
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Description: 主要采用C语言解复用出视频段、音频段以及数据段的相关复用内容，并将其里面的数据储存在结构体数组里，以供其他模块使用。-Mainly uses the C language demultiplexing a video segment, audio segment, and data re-use the contents of the relevant section, and its inside structure of data stored in an array of years, for use by other modules.
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Size: 2048 |
Author: 吴非 |
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Description: 音频分割代码，实现音频特征的提取，并用几个重要的特征进行分割-Audio segmentation code, the extraction of audio features, and with a few important features to segment
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Size: 90112 |
Author: 景雄 |
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Description: 先将47168bit的宿主音频等步长分段,分为1000段，然后混沌序列选择其中的102段，对每段进行3及小波分解，挑选出绝对值最大的系数，运用嵌入公式。-47168bit first step of the host audio sub-divided into section 1000, and then select one of the 102 chaotic sequence segment, on each piece of a 3 and wavelet decomposition, the largest absolute value coefficients selected using the embedded formula.
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Size: 6144 |
Author: 菁菁 |
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Description: his a simple method for silence removal and segmentation of audio streams that contain speech. The method is based in two simple audio features (signal energy and spectral centroid). As long as the feature sequences are extracted, as thresholding approach is applied on those sequence, in order to detect the speech segment-his is a simple method for silence removal and segmentation of audio streams that contain speech. The method is based in two simple audio features (signal energy and spectral centroid). As long as the feature sequences are extracted, as thresholding approach is applied on those sequence, in order to detect the speech segment
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Size: 980992 |
Author: petr |
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Description: A Novel Efficient Approach for Audio Segmentation
a novel approach to audio segmentation
is presented. The problem of detecting audio segments’
limits is treated as a binary classification task.
Frames are classified as “segment limits” vs “nonsegment
limits”. For each audio frame a spectrogram
is computed and eight feature values are extracted from
respective frequency bands. Final decisions are taken
based on a classifier combination scheme
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Size: 225280 |
Author: kvga |
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Description: a novel algorithm to segment
an audio piece into its structural components.
The boundaries of the homogeneous regions are decided
based on various time and frequency domain
features. The algorithm has been designed in 2 stages.
In the first stage, a vocal/non-vocal/silence classification
is done using multinomial softmax regression. The
second stage uses a hidden Markov model to ‘smooth’
the previous output as well as enforce the time dependent
structuring.
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Size: 156672 |
Author: kvga |
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Description: This project describes the work done on the development of an audio segmentation and classification system. Many existing works on audio classification deal with the problem of classifying known homogeneous audio segments. In this work, audio recordings are divided into acoustically similar regions and classified into basic audio types such as speech, music or silence. Audio features used in this project include Mel Frequency Cepstral Coefficients (MFCC), Zero Crossing Rate and Short Term Energy (STE). These features were extracted from audio files that were stored in a WAV format. Possible use of features, which are extracted directly from MPEG audio files, is also considered. Statistical based methods are used to segment and classify audio signals using these features. The classification methods used include the General Mixture Model (GMM) and the k- Nearest Neighbour (k-NN) algorithms. It is shown that the system implemented achieves an accuracy rate of more than 95 for discrete audio classification.-This project describes the work done on the development of an audio segmentation and classification system. Many existing works on audio classification deal with the problem of classifying known homogeneous audio segments. In this work, audio recordings are divided into acoustically similar regions and classified into basic audio types such as speech, music or silence. Audio features used in this project include Mel Frequency Cepstral Coefficients (MFCC), Zero Crossing Rate and Short Term Energy (STE). These features were extracted from audio files that were stored in a WAV format. Possible use of features, which are extracted directly from MPEG audio files, is also considered. Statistical based methods are used to segment and classify audio signals using these features. The classification methods used include the General Mixture Model (GMM) and the k- Nearest Neighbour (k-NN) algorithms. It is shown that the system implemented achieves an accuracy rate of more than 95 for discrete audio classification.
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Size: 653312 |
Author: kvga |
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Description: 这是一个音频切割文件，音频切割audio segment可以将读入的wav行数据进行语音的切分是实现语音软件的重要函数。-i write this function for a long time .
it s very useful for one who want kown about signal.
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Size: 4058112 |
Author: 张晓坤 |
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Description: 本代码为编码开关代码，编码开关也就是数字音响中的
360度旋转的数字音量以及显示器上用的（单键飞梭开
关）等类似鼠标滚轮的手动计数输入设备。
我使用的编码开关为5个引脚的，其中2个引脚为按下
转轮开关（也就相当于鼠标中键）。另外3个引脚用来
检测旋转方向以及旋转步数的检测端。引脚分别为a,b,c
b接地a，c分别接到P2.0和P2.1口并分别接两个10K上拉
电阻，并且a，c需要分别对地接一个104的电容，否则
因为编码开关的触点抖动会引起轻微误动作。本程序不
使用定时器，不占用中断，不使用延时代码，并对每个
细分步数进行判断，避免一切误动作，性能超级稳定。
我使用的编码器是APLS的EC11B可以参照附件的时序图
编码器控制流水灯最能说明问题，下面是以一段流水
灯来演示。-This code for code switch code, coding switch is the digital audio
360 degrees of rotation digital volume and displays use on the open list key shuttle (
Closed) and similar mouse wheel manual count input device.
I use a code switch for five of the pins, with two pins for press
The runner switch (also is equivalent to the middle mouse button). Three other pins are used to
The rotation direction detection and rotation of the number of detection. Pins are respectively for a, b, c
B, c received a ground respectively P2.0 and P2.1 mouth and take two 10 K respectively on pull
Resistance, and a, c need to pick up a respectively in 104, otherwise the capacitance
Because of the code switch contacts jitter can cause minor wrong action. This procedure does not
Use timer, do not take up interrupt, do not use delay, and each code
Segment for judgment, avoid steps all misoperation, super stable performance.
I use of APLS is encoder EC11B can refer to the attachment sequence chart
Encode
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Size: 824320 |
Author: 赵博伟 |
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Description: select a audio segment from a audio file
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Size: 468992 |
Author: zheng |
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Description: 语音芯片，一线串口通信方式；可通过按键、红外遥控设定播报的音频段数及声音大小-Voice chip, first-line serial communication through the buttons, IR remote control is set to broadcast the audio segment number and size of the sound
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Size: 41984 |
Author: 禾丫 |
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Description: 研究中我么采用分段的方法对数字音频信号嵌入水印。为便于研
究，以灰度图像为数字水印原始信号经二值化处理后得到水印再嵌入
音频信号的离散变换域中，从而得到嵌入水印的音频信号，其实现过
程原理框图如图1 所示。在水印嵌入的过程主要实现四个阶段的任
务：(1)数据准备与预处理，主要是准备好水印原始图片和待嵌入水印
的音频信号 (2)音频信号分段处理，用于嵌入水印的音频数据部分等
分成(m1*m2)个音频数据段 (3)水印图像降维处理，在此过程灰度图像
水印转化为二值图象, 要将其嵌入到一维的数字音频信号中, 必须将
二维的图像转化成一维的序列 (4)将水印嵌入到数字音频信号中,从而
最终获得带数字水印的音频信号。-The study I Mody, segmented digital audio signal embedded watermark. Discrete transform domain to facilitate the study of the grayscale image for the digital watermark the original signal is obtained after the binarization processing watermark embedded in the audio signal, thereby obtaining the watermark embedded in the audio signal, and its implementation process schematic block diagram shown in Figure 1. Mainly in the watermark embedding process to achieve the task of four phases: (1) The data preparation and pretreatment, is ready to watermark the original picture and to be embedded in the audio signal of the watermark (2) audio signal segment processing for embedding watermark The audio data portion is divided into (m1* m2) audio data segment (3) The watermark image dimension reduction processing, binary image is converted to grayscale image watermark in this process, and to be embedded into a digital audio signal of a one-dimensional , two-dimensional images must be converted in
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Size: 1024 |
Author: 吴吟骋 |
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Description: 利用 SPCE061A单片机，扩展 SD卡设备，实现录放音功能。要求具有下述功能：
1. SPCE061A对 SD卡的读写操作
a) SPCE061A对 SD卡的操作方式为 SPI模式；
b) 提供 SD卡插入检测功能，并进行语音提示；
c) 提供 SD卡写保护检测功能，并进行语音提示；
d) 可以对 SD卡进行初始化、扇区写、扇区读、扇区擦除、得到 SD卡的容量信息等操作；
2. 录放音功能
a) 录音数据存储在 SD卡中；
b) 支持多段录音；
c) 放音支持“上一曲”、“下一曲”；
d) 可以删除所有的录音片断，重新开始录音； -Using SPCE061A, expansion SD card equipment, audio recording and playback function. Required to have the following functions: 1. Of SPCE061A SD card read and write operations a) the operation of SPCE061A SD card in SPI mode b) provide the SD card into the detection function, and voice prompt c) the SD card write protection detection and voice prompts d) the SD card can be initialized sector write sector Reading sector erase SD card capacity information and other operating 2 audio recording and playback functions a) recording data stored in the SD card b) support for multi-segment recording c) playback support " on a" next one " d) can delete all the audio clips, and re-start recording
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Size: 8656896 |
Author: 纪春龙 |
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Description: matlab编程，画出一段语音的语谱图，在不同时间声音的不同频率，可以比较出有声段和无声段的能量不同。-Matlab programming, draw the spectrogram of a voice, the sound of different frequencies in different time, you can compare audio segment and the energy of the silent segment.
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Size: 1024 |
Author: 刘星 |
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Description: 语音特征信号识别是语音识别研究领域中的一个重要方面,一般采用模式匹配的原理解
决。语音识别的运算过程为:首先,待识别语音转化为电信号后输入识别系统,经过预处理后用数学方法提取语音特征信号,提取出的语音特征信号可以看成该段语音的模式。然后将该段语音模型同已知参考模式相比较,获得最佳匹配的参考模式为该段语音的识别结果.-Speech characteristic signal recognition is an important aspect in the field of speech recognition research, generally USES pattern matching the original understanding
Definitely. Operation process of speech recognition are as follows: first, after waiting for voice recognition into electrical signal input and recognition system, after pretreatment
Mathematical method was used to extract speech signal, and extracts the phonetic characteristics of signals can be as the segment of speech patterns. And then the
Audio model compared to a known reference model, get the best match the reference pattern for this speech recognition results.
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Size: 375808 |
Author: 吴军 |
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Description: 对给定的音频列表文件，将给定的音频插入到音频列表文件中。-insert audio segment to given audio file list
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Size: 26624 |
Author: treeson |
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Description: 设计一个简单的GUI界面，对导入的音频文件进行时域特性分析。计算其时域波形、平均幅度、短时能量、短时过零率。对该音频文件进行加噪处理，分析其时域波形变化。设定一个固定时长，对该音频添加回声并保存为“*.wav”格式文件。
对音频文件的时域分析是对音频文件进行传输、降噪及其他再编辑的基础。其中短时能量可用作清浊音区分的标准和语音起始点的判断。短时过零率可用作判断有声段与无声段和从背景噪声中提取语音信号。
-Design a simple GUI interface that time domain characteristics analysis on import audio files.Calculate the average time domain waveform, amplitude, short-time energy and short-time zero crossing ratio.Add to deal with the noise of the audio files, analyze the changes in the time domain waveform.To set a fixed time, to add the audio echo and saved as a *. Wav format file.
The time domain analysis of the audio file was transmitted to audio files, noise reduction and other editor again.The short-time energy voiced distinguish between standard and can be used as a clear voice starting point judgement.Short-time zero crossing ratio can be used as a judgment audio segment and silent segment and extract speech signal background noise.
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Size: 52224 |
Author: liwan |
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Description: vb编写的8位音频提取程序，适合将英语教材多段截取到数据库中去，附源代码。-vb prepared eight audio extraction program, adapted to English textbooks more interception to the to segment, with source code.
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Size: 874496 |
Author: 胡苦力 |
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