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Description: 对WAV音频文件进行PCM量化及编码，最终生成二进制向量-PCM WAV audio files on the quantization and coding, ultimately generating a binary vector
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Size: 69632 |
Author: gymsun |
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Description: 在MATLAB环境下实现量化后的PCM编码-In the MATLAB environment to achieve quantized PCM coding
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Size: 2048 |
Author: 宋贤明 |
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Description: 低通信号抽样定理、13折线A律逐次比较型PCM编译码-Low-pass signal sampling theorem, 13 successive line A comparative law PCM codec
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Size: 72704 |
Author: shadow |
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Description: C语言读取WAV PCM音频文件的头文件-C language to read WAV PCM audio file of the header files
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Size: 1024 |
Author: hjin |
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Description: matlab实验报告 学习A律PCM量化器的matlab代码-Programmable test report: Learning A law PCM quantizer matlab code
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Size: 1024 |
Author: hanhan |
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Description: PCM pulse code modulatioon matlab
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Size: 4096 |
Author: Alberto |
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Description: Implementation of a speech codec based on coding of speech at 8 kbit/s
using conjugate-structure algebraic-code-excited linear-prediction (CS-ACELP)
- We took .wav files that is sampled at 8000 Hz using 16-bit linear PCM. The encoding process is done every 10ms frame or 80 samples. For the preprocessing stage, the samples are high passed with cut-off frequency of 140 Hz and scaled down by 2. A total of 240 samples are buffer for windowing and autocorrelation computation. The autocorrelation coefficients are used to calculate the LP filter coefficients using the Levinson-Durbin algorithm. The LP filter coefficients are converted to Line Spectral Pair (LSP) coefficients. LSP coefficients are converted back to the LP filter coefficients, which is just the reverse process of the conversion from LP to LSP. This module is exactly what the decoder will need in order to convert the LSP coefficients to LP coefficients. We decided not to implement the LSF quantization module because we did not have the codebook information when we designed our system.
The open-loop pitch delay is calculated first for each frame. Then the closed-loop pitch
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Size: 40960 |
Author: coco |
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Description: 是关于通信系统仿真的一些程序，有RS码在CT2中的系统仿真的模型和PCM的一些在不同参数下的图形分析-RS码和PCM码
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Size: 1220608 |
Author: fuwayu |
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Description: 运用仿真抽样信号的频谱, 分析满足不同条件下的抽样情况、验证奈奎斯特抽样定理不同量化电平情况下的信噪-The use of simulation sampling frequency spectrum, analyzing a sample under different conditions to meet the situation, verify the Nyquist sampling theorem under different quantization level of the signal to noise
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Size: 415744 |
Author: 至尊宝 |
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Description: 降采样程序，从16k16bit数据 降采样到8k16bit-resample program，it resample the 16k16bit pcm to 8k16bit pcm
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Size: 8192 |
Author: 王旭 |
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Description: 讲述了语音编码的基本概念。
主要包括PCM、ΔM、CVSD、ADPCM、DCME-ADPCM、E-ADPCM、LPC等概念-Describes the basic concepts of speech coding. Include PCM, ΔM, CVSD, ADPCM, DCME-ADPCM, E-ADPCM, LPC concepts
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Size: 7168 |
Author: 睿铂 |
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Description: In Matlab
模拟与分析PCM不同量化级别的量化噪声
可以直接运行-In Matlab， simulation and analysis of different quantization levels of PCM quantization noise ，can be run directly
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Size: 2048 |
Author: luocheng |
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Description: 基于matlab的数字频带通信系统仿真，第三步：PCM编码-Matlab digital band-based communication system simulation, the third step: PCM coding
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Size: 1024 |
Author: 艾丝凡 |
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Description: 播放WAV文件,支持PCM和ADPCM文件-play wav file
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Size: 1976320 |
Author: 何柳青 |
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Description: 有关PCM编码的，并有误码率分析，无密码-The PCM coding, no password
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Size: 2048 |
Author: wang |
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Description: 语音信号处理：This package encloses a small application dedicated to speech signal processing
at 8 kHz or 16 kHz. This application reads standard *.wav or *.pcm files through UI dialog box, plots and plays the signal with vertical dynamic crosshairs.-This package encloses a small application dedicated to speech signal processing at 8 kHz or 16 kHz. This application reads standard*.wav or*.pcm files through UI dialog box, plots and plays the signal with vertical dynamic crosshairs.
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Size: 327680 |
Author: 银杏 |
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Description:
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Size: 13312 |
Author: jack |
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Description: 是一个可以将MP3转化成PCM流的一个软件程序源代码，对制作MP3格式转化器相当有用！大家都看下吧-Is an MP3 into PCM stream to a software source code, on the production of MP3 format converter very useful! Let everyone see next
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Size: 95232 |
Author: bruce.yu |
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Description: MP3解码原理：首选将MP3数据帧解包，解出帧头信息和边带信息；然后采用霍夫曼解码解出比特分配信息；接着在逆变换中利用频谱系数，在综合滤波器中将32位子带合并成一个宽带信号。18个频谱值执行32位IMDCT（逆改进型离散余弦变换），将生成的576个频谱值变换成长度为32的18个连续的频谱。通过18次运算，多相位综合滤波器将这些频谱转换到时域，完成波形重构，生成立体声PCM音频数据。-jiema and bianma
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Size: 58368 |
Author: wang |
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Description: 用C++编写的一个软件，完成对一个语音信号的内插和抽取。语音数据以wav格式、单声道存储，编码方式为PCM。可完成的功能为：
（1）读wav文件；
（2）写wav文件；
（3）对语音数据进行内插和抽取。
-Written in C++, a software, to complete a voice signal decimation and interpolation. Wav format, voice data, single-channel storage, encoding for the PCM. The function to be completed: (1) read wav files (2) to write wav file (3)complete decimation and interpolation to the voice data.
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Size: 1024 |
Author: 梦游 |
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