Description: 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 Platform: |
Size: 1693696 |
Author:夏天 |
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Description: 语音降噪。从Codec AD50采集话筒语音，通过DSP TMS320vc5402处理，在送到AD50输出降噪后语音，涉及加汉宁窗，切比雪夫滤波器，快速傅立叶变换和反FFT，有声无声判断谱分解，谱合成等功能-Voice noise. Codec AD50 collected from the microphone voice, through the DSP TMS320vc5402 treatment, in AD50 to the output noise after the voice, involve Hanning windows, Chebyshev filter, Fast Fourier Transform and anti-FFT, audio silent judge spectral decomposition, spectral synthesis and other functions Platform: |
Size: 44032 |
Author:黄胜华 |
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Description: 本文介绍了一种基于SPCE061A单片机，采用自适应算法的数字抗噪声系统，实现在高噪声环境中语音信号的清晰识别，并给出了自适应数字抗噪声系统的硬件结构和软件流程图。此抗噪声系统可用于抗噪声送话器和车载通信系统中实现清晰通讯。-This article describes an approach based on single-chip microcomputer SPCE061A adaptive algorithm for digital anti-noise system, realize in high-noise environment of the clear voice signal identification, and gives the adaptive digital anti-noise system hardware structure and software flow chart . This anti-noise system can be used for anti-noise microphone and car communication systems to achieve clear communication. Platform: |
Size: 256000 |
Author:niu xiaolei |
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Description: 传统双话筒降噪应用中，要求采用定向话筒对抗话音串扰问题，并且语音损伤较大，采用抗交叉串绕算法后，仅噪声被抵消，语音信号被保留。分为两个阶段，第一阶段，学习环境，第二阶段，抵消噪声。-Traditional dual-microphone noise reduction applications, the adoption of directional microphone voice against crosstalk, and a larger voice injury, using anti-cross-string around the algorithm, the only noise being offset, voice signals are retained. Divided into two phases, the first stage, the learning environment, the second phase, the offset noise. Platform: |
Size: 1024 |
Author:chenziqiang |
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Description: Commercially available active noise control headphones rely on fixed analog controllers to drive "anti-noise" loudspeakers. Our design uses an adaptive controller to optimally cancel unwanted acoustic noise. This headphone would be particularly useful for workers who operate or work near heavy machinery and engines because the noise is selectively eliminated. Desired sounds, such as speech and warning signals, are left to be heard clearly. The adaptive control algorithm is implemented on a Texas Instruments (TI™ )
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TMS320C30GEL digital signal processor (DSP), which drives a Sony CD550 headphone/microphone system. Our experiments indicate that adaptive noise control results in a dramatic improvement in performance over fixed noise control. This improvement is due to the availability of high-performance programmable DSPs and the self-optimizing and tracking
capabilities of the adaptive controller in response to the surrounding noise. Platform: |
Size: 282624 |
Author:Davi Souza |
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Description: Frequency Analyzer.
Whistle a melody and watch this program graph the pitch in real time. The Frequency Analyzer (download now) technically speaking performs a Fast Fourier Transform of the sound (you need a sound card and a microphone to use this program). What it means is that it analyzes your voice and splits it into its component frequencies. Watch patterns of speech, the harmonics of vowels and the noise of sibilants. The size of the program is only 148k! (zipped 74k) Can you believe that?! There are web pages that make you download more than that just to be able to look at them. Platform: |
Size: 122880 |
Author:kapa |
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Description: /* DESCRIPTION */
/* Measuring the signal to noise ratio of an audio signal using the */
/* TMS320C6416 DSK. Uses a sampling rate of 48000 Hz (48 kHz). */
/* */
/* Takes an input from a microphone or from a CD player and feeds */
/* straight through to the loudspeakers. Displays singal to noise ratio. */-/* DESCRIPTION */
/* Measuring the signal to noise ratio of an audio signal using the */
/* TMS320C6416 DSK. Uses a sampling rate of 48000 Hz (48 kHz). */
/* */
/* Takes an input from a microphone or from a CD player and feeds */
/* straight through to the loudspeakers. Displays singal to noise ratio. */ Platform: |
Size: 836608 |
Author:giorgio |
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Description: 单麦克风和双麦克风语音增强系统的研究，这是基础的双麦克风降噪算法，但是很实用-Single microphone and dual-microphone speech enhancement system, which is based dual-microphone noise reduction algorithm, but very practical Platform: |
Size: 171008 |
Author:liubin |
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Description: 传统双话筒降噪应用中，要求采用定向话筒对抗话音串扰问题，并且语音损伤较大，采用抗交叉串绕算法后，，仅噪声被抵消，语音信号被保留。分为两个阶段，第一阶段，学习环境，第二阶段，抵消噪声。
-The application of the traditional dual-microphone noise reduction, directional microphone to confront the problem of voice crosstalk, and the voice more damage with anti-cross-crosstalk algorithm, the only noise is offset, the voice signal is retained. Divided into two phases, the first phase, the learning environment, the second phase, offset and noise. Platform: |
Size: 1024 |
Author:藐视 |
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Description: FM2010芯片是基于迷你阵列麦克风（SAM）专利技术的，采用空间滤波技术、远/近距离定向拾取声音信号、抑制声学噪声、消除声学回声的低功耗低成本的单芯片。本文将介绍迷你阵列麦克风技术在手持通信产品中的设计要点、FM2010芯片主要功能，及其在GSM手机中的典型应用。
本人提供全套FM2010调试用时的芯片手册，参数调试手册，寄存器手册，及固件程序。-FM2010 chip is based on the mini-array microphone (SAM) patented technology, the use of spatial filtering technique, far/close-directional pick up sound signals, inhibition of acoustic noise, acoustic echo elimination of low-power low-cost single chip. This article will introduce hand-held mini-array microphone technology in the design features of communication products, FM2010 chip main function, and its typical application of GSM mobile phones. I provide a full set of chips when debugging FM2010 manual, parameter commissioning manual, register manuals, and firmware. Platform: |
Size: 1351680 |
Author:feilong |
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Description: 基于rls算法的多麦克风降噪matlab程序代码-Matlab code based on the the rls algorithm of multi-microphone noise reduction Platform: |
Size: 468992 |
Author:戴咪嘟 |
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Description: 采用维纳滤波实现降噪的详细描述，包含单个话筒与多个话筒两种情形-Wiener filtering noise using a detailed description of a single microphone and a plurality of microphones including two cases Platform: |
Size: 1168384 |
Author:祁才君 |
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Description: The material in this work is organized into five chapters, including this one. The
focus is on the time-domain algorithms for both the single and multiple microphone
cases. The work discussed in these chapters is as follows.
In Chap. 2, we study the noise reduction problem with a single microphone by
using a filtering vector for the estimation of the desired signal sample.
Chapter 3 generalizes the ideas of Chap. 2 with a rectangular filtering matrix for
the estimation of the desired signal vector.
In Chap. 4, we study the speech enhancement problem with a microphone array
by using a long filtering vector.
Finally, Chap. 5 extends the results of Chap. 4 with a rectangular filtering matrix Platform: |
Size: 506880 |
Author:infinity |
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