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Description: 用DSP实现的一个简单的语音识别系统,只要实现单个词的识别即可,采样率8k,帧长30ms,帧移10ms,系统采样后分帧--端点检测,将检测到的原始语音信号保存下来,基本上一个字在30帧左右,然后提取每帧的LPC参数--将LPC参数转换为LPC倒谱系数,然后利用DTW方法和模板比较.-a brief speech recognition system, as long as the realization of a single word can be identified. 8 k sampling rate, frame size of 30 ms and 10 ms frame shift, hours after sampling frame -- Endpoint Detection, Detection of the original speech signal preserved, basically a word in 30% Then each frame from the LPC parameters -- will be converted to LPC parameters LPC cepstrum coefficients. Then DTW methods and template comparison.
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Size: 1793 |
Author: yautou |
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Description: 本文讨论了AR模型及线性预测的原理，在浮点型DSP TMS320C6713B上实现了语音信号线性预测系数(LPC)的提取，并利用LPC系数用Verilog语言实现了AR模型的Lattice结构。
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Size: 14313 |
Author: 万金油 |
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Description: 用AD的16位定点DSP作音频压缩器-With AD 16-bit fixed-point DSP for audio compressor
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Size: 23552 |
Author: |
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Description: 用DSP实现的一个简单的语音识别系统,只要实现单个词的识别即可,采样率8k,帧长30ms,帧移10ms,系统采样后分帧--端点检测,将检测到的原始语音信号保存下来,基本上一个字在30帧左右,然后提取每帧的LPC参数--将LPC参数转换为LPC倒谱系数,然后利用DTW方法和模板比较.-a brief speech recognition system, as long as the realization of a single word can be identified. 8 k sampling rate, frame size of 30 ms and 10 ms frame shift, hours after sampling frame-- Endpoint Detection, Detection of the original speech signal preserved, basically a word in 30% Then each frame from the LPC parameters-- will be converted to LPC parameters LPC cepstrum coefficients. Then DTW methods and template comparison.
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Size: 1024 |
Author: yautou |
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Description: 是一个ARM214系连的最小系统原理图,希望能多爱好嵌入是开发的人员有帮助-ARM214 is a system even the smallest system diagram is the hope of many enthusiasts are embedded in the development of the staff to help
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Size: 109568 |
Author: 好伟 |
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Description: 本文讨论了AR模型及线性预测的原理，在浮点型DSP TMS320C6713B上实现了语音信号线性预测系数(LPC)的提取，并利用LPC系数用Verilog语言实现了AR模型的Lattice结构。-This article discusses the AR model and the principle of linear prediction, in the floating-point DSP TMS320C6713B realize the voice signal on the linear prediction coefficient (LPC) of the extract, and the use of LPC coefficients using Verilog languages realize the AR model Lattice structure.
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Size: 14336 |
Author: 万金油 |
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Description: can网络参数计算工具合集,包含一个SJA1000通讯参数计算器,LPC系列Arm通讯参数计算器,还有一个能计算Atmel-89C51CC01,英飞凌全系列Can控制器,MicroChip MCP25XX,Motorola 68HC05,飞利浦全系列Can控制器,TI TMS320系列DSP内置Can控制器等芯片的通讯参数和Can网络最大通讯距离的超级强大的计算工具,实在是Can网络开发的必备工具-Collection computational tools can network parameters, including a SJA1000 communication parameter calculator, LPC series of Arm communication parameters calculator, there is an energy calculations Atmel-89C51CC01, Infineon Can a full range of controllers, MicroChip MCP25XX, Motorola 68HC05, Philips Pu Can a full range of controllers, TI TMS320 series DSP chip built-Can controller communication parameters and the largest telecommunications network distance Can super powerful computing tools, it is an indispensable tool for network development Can
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Size: 339968 |
Author: 柯岚 |
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Description: musicdsp source code archive-Analysis
Beat Detector Class
Coefficients for Daubechies wavelets 1-38
DFT
Envelope detector
Envelope Detector class (C++)
Envelope follower with different attack and release
Fast in-place Walsh-Hadamard Transform
FFT
FFT classes in C++ and Object Pascal
Frequency response from biquad coefficients
Java FFT
Look ahead limiting
LPC analysis (autocorrelation+ Levinson-Durbin recursion)
Magnitude and phase plot of arbitrary IIR function, up to 5th order
Measuring interpollation noise
QFT and DQFT (double precision) classes
Simple peak follower
tone detection with Goertzel
Tone detection with Goertzel (x86 ASM)
Effects
2 Wave shaping things
Alien Wah
Band Limited PWM Generator
Bit quantization/reduction effect
Class for waveguide/delay effects
Compressor
Decimator
Delay time calculation for reverberation
DIRAC- Free C/C++ Library for Time and Pitch Manipulation of Audio Based on Time-Frequency Transforms
dynamic convolution
Early echo s with image-mirror techn
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Size: 826368 |
Author: Alan Tang |
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Description: 基于DSP简单语音识别系统程序。。写得不好大家别见怪-Simple Speech Recognition System Based on DSP program. . Well written we Bie Jianguai
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Size: 1024 |
Author: 陈星智 |
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Description: 实现将lpc转换为lpcc系数 用于dsp语音识别-Implementation will be converted to lpcc lpc coefficients for speech recognition dsp
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Size: 1024 |
Author: lxdong |
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Description: 基于LPMCC的语音识别系统实现
语音识别可实现人机交互和语音控制，在X-业控制、消费电子等领域都有广泛应用。结合人发音的生理
结构的特点，使用LPMCC(LPC倒谱美尔变换)作为特征向量，采用动态规划算法作为核心识别算法，在TMS320VC5402芯片上实现了特定人、孤立词的高性能实时识别系统。-Speech Recognition System Based On LPMCC
Speech recognition can achieve human—computer interaction and voice control， and it is used in indUStrial control，consumer electronics and many other fields．Combined with the mechanism of the production of
human s voice． a high perform speech recognition system is built for specific people on TMSVC5402 DSP by using the LPMCC as the feature vector and using dynamic time warping as the matching algorithm．
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Size: 280576 |
Author: edwin |
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Description: LPC-10的标准编解码程序，文件.dsp是其工程描述文件；LPC-10编码处理采样率为8kHz,16bit量化的语音样本，处理帧长为180个样本，压缩输出为54个bit；LPC-10解码为编码的反过程。-Standard LPC-10 codec program files. Dsp project description file 8kHz sampling rate LPC-10 coding processing, 16bit quantify voice sample processing frame length 180 samples compressed output is 54 bits. The LPC-10 decode encoded anti process.
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Size: 20480 |
Author: 王成 |
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Description: 利用MATLAB实现LPC的分析和合成，用于DSP的实现-MATLAB LPC analysis and synthesis for DSP implementations
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Size: 1024 |
Author: zhuojun chen |
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Description: 语音分析合成Matlab工具箱，用于语音编码，语音解码，语音DSP数字化处理，LPC，MELP，PCM，CEP-VOICEBOX is a MATLAB toolbox for speech processing.
Speech synthesis analysis Matlab Toolbox for speech coding, speech decoding, digital voice processing DSP, LPC, MELP, PCM, CEP ...
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Size: 602112 |
Author: lcd |
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Description: DSP实现基于LPC的语音基频提取算法，包含采样、fir滤波器，LPC算法等的编写
-The realization of the speech pitch extraction algorithm based on LPC
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Size: 4192256 |
Author: gaogao |
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Description: LPC vocoder simulation for digital speech processing
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Size: 2275328 |
Author: ALBERTKATEREGA
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