Description: 本实验旨在了解如何将模拟的声音信号转化为数字信号，重点掌握在数字声音四个重要的技术指标：采样速率、量化比特数、声道数和编码方法。-This experiment aimed at understanding how analog voice signal into a digital signal, with a focus in the hands of digital audio technology to four key indicators: the sampling rate, quantization bits, the number of channels and the encoding method. Platform: |
Size: 6246400 |
Author:贺美琛 |
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Description: These are a couple of m-files to read and write mp3 audio files [i.e. files compressed using MPEG-Audio layer 3 encoding] under Matlab.在matlab中读写mp3，这是2.0版本-These are a couple of m-files to read and write mp3 audio files [i.e. files compressed using MPEG-Audio layer 3 encoding] under Matlab. Platform: |
Size: 475136 |
Author:corner |
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Description: 运用Matlab语言编程，进行信号分析的能力。音频信号是一种连续变化的模拟信号，计算机只能处理和记录二进制的数字信号，由自然音源而得的音频信号必须经过采样，量化和编码，变成二进制数据后才能送到计算机进行再编辑和存贮，通过本实验中了解模拟信号采样和重构的完整过程，加深对采样定理的理解。-Matlab programming language to use for signal analysis. Audio signal is a continuous change of the analog signal, the computer can only handle and record the binary digital signal, derived from natural sources of the audio signal to be sampled, quantification and encoding binary data into a computer in order to re- editing and storage, through this experiment to understand the analog signal sampling and reconstruction of the full process, to deepen understanding of the sampling theorem. Platform: |
Size: 1024 |
Author:tong |
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Description: mp3read和mp3write是直接写在wavread更换和wavwrite访问MPEG音频MP3文件。其特点包括： -旨在重复wavread完整语法和wavwrite -还支持上的动态下采样和渠道mpg123的合并-文件优化的很长的MP3 -只需要解码的部分-使用popen函数进行编码，以避免大量临时文件（可用时-看到我popenw墨西哥） -试图消除写保护回路的时间序列文件/通过阅读“热身”样本-包括辅助二进制软件包在Linux，Windows，Mac的PPC的，苹果，英特尔，和Mac英特尔- 64位-These versions of mp3read and mp3write are direct drop-in replacements for wavread and wavwrite to access MPEG audio mp3 files. Features include:
- aims to duplicate complete syntax of wavread and wavwrite
- also supports on-the-fly downsampling and channel merging of mpg123
- optimized for very long mp3 files- only decodes the needed portion
- uses popen for encoding to avoid a large temporary file (when available- see my popenw mex)
- attempts to preserve time alignment of files through read/write loop by removing "warm up" samples
- package includes helper binaries for Linux, Windows, Mac-PPC, Mac-Intel, and Mac-Intel-64bit
Platform: |
Size: 2195456 |
Author:young |
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Description: 功能列表= mmwrite（文件名，选择... ...）
mmwrite是能写的AVI，WMV，的WMA，ASF文件。对于AVI文件，您可以选择从现有的编解码器压缩的音频和视频流。
对于WMV，WMA和ASF的编码默认为16位的Windows Media 9率为44100Hz立体声为98％质量的音频和Windows媒体视频9 98
质量。质量可以指定音频和视频。
环绕声，似乎只与AVI和多通道编码不支持。编写任何其他文件类型不支持。本使用Windows的DirectX基础设施，以便其他操作系统是出于运气。
输入：
文件名：这一定是第一个参数，并指定文件名写。
影片结构：视频结构相匹配的mmread输出。在最低限度，必须有4个领域的“框架”，“时代”，“高度”和“宽度”。该“框架”字段必须是一个结构与一个字段数组“的CDATA”的原始框架，它包含由彩色编码的数据宽度高度（3）UINT8s。钍-function list = mmwrite(filename,...options...)
mmwrite is able to write AVI,WMV,WMA,ASF files. For AVI files you can choose from the available codecs to compress the audio and video streams.
For WMV,WMA and ASF the encoding defaults to Windows Media 9 44100Hz 16bit stereo 98 quality for the audio and Windows Media 9 Video with 98
quality. The quality can be specified for both audio and video.
Surround sound only seems to work with AVI and multi-pass encoding is not supported. Writing any other file type is not supported. This uses Windows DirectX infrastructure, so other OSs are out of luck.
INPUT:
filename: This must be the first parameter and specifies the filename to write.
video structure: The video structure matches the output of mmread. At a minimum it must have 4 fields "frames", "times", "height" and "width". The "frames" field must be a struct array with a field "cdata" that contains the raw frame data encoded as height by width by color(3) as UINT8s. Th Platform: |
Size: 113664 |
Author:zaaa |
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Description: ：针对MP3 编码方式和帧结构进行分析，利用帧中的比例因子提出了一种压缩域水印算法。算法通过引入帧变化分析步
骤，很好地实现了水印的自适应嵌入。对于不同的系统要求，还可以通过修改门限信噪比的值SNRlim按需进行水印嵌入，使算法
具有更大灵活性。同时在牺牲一定的嵌入空间前提下，实现了水印信息的自同步。实验表明，对于不同风格的音乐，该水印算法
具有很好的不可听性。-In this paper，the encoding method and the frame structure of MP3 are expounded in detail，a watermark algorithm
is proposed based on the scale factor in MP3 frame.By introducing the step of frame altering analysis，the algorithm
can realize that watermarks is embedded adaptively.By modifying threshold SNRlim（the ratio of signal to noise），the algorithm
can meet different requirements of some system，that is what enable the algorithm to have a bigger flexibility.Based on expense
of some embedding positions，the watermark can be embedded with self-synchronization.The experiment results show
this algorithm can get a good audio effect with various types of audio. Platform: |
Size: 1386496 |
Author:郑杨 |
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Description: 用matlab进行双音频电话的频谱分析 。从输入的音频信号中，读出电话的编码-Using matlab for spectral analysis of dual-tone phone. From the input audio signal, the read out encoding telephone Platform: |
Size: 1024 |
Author:卫理 |
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Description: Linear-Predictive-Coder
MATLAB Implementation of LPC algorithm for speech signal
# Why LPC?
In communication systems it is often necessary to transmit audio(speech) signal in compressed or encoded form because of bandwidth limitation of the channel. In this regard, ‘Linear predictive coding(LPC)’ is an effctive method of speech coding at a low bit-rate.
# Features
** Analysis/Encoding phase,Synthesis/Decoding phase.
**Human voice modelled with all-pole filter.
** LPC parameters(filter coefficients, pitch, gain etc) extraction at the decoding phase.
** Non-overlapping frames of 30 milliseconds in duration
# How To Run
** Make sure MATLAB(latest version) is installed
** Put both files(LPC.m with .mp3 file) in the same folder
** Open LPC.m file and run it.
## Comments
Different audio (.mp3) files can be coded/decoded by changing the input file name in the code. Platform: |
Size: 2048 |
Author:japaoli |
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Search - encoding audio matlab