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DSP_dg
- 这是本人用Matlab实现音频噪声滤除的试验，要求分析并选用适当滤波器-using Matlab audio noise filtering tests, requirements analysis and the choice of appropriate filter
DSP2
- DSB-SC信号的生成与解调 1) 用离散（DSP）的方法生成DSB信号 2) 载波频率为150KHz，音频为500Hz和2000Hz的混合音。 3) 加入高斯白噪声 （4） 语音信号 的传输。 改变抽样频率和量化台阶大小，观察重建信号以及量化噪声信号的波形；对于语音信号主观评价声音质量的变化。 -DSB-SC signal generation and demodulation 1) Discrete (DSP) methods to generate
valctr
- PT2258 is a 6-Channel Electronic Volume Controller IC utilizing CMOS Technology specially designed for the new generation of AV Multi-Channel Audio System. PT2258 provides an I2C Control Interface, an attenuation range of 0 to -79dB at 1dB/step,
ISD4004
- ISD4004系列工作电压3V,单片录放时间8至16分钟,音质好,适用于移动电话及其他便携式电子产品中。芯片采用CMOS技术,内含振荡器、防混淆滤波器、平滑滤波器、音频放大器、自动静噪及高密度多电平闪烁存贮陈列。芯片设计是基于所有操作必须由微控制器控制,操作命令可通过串行通信接口(SPI或Microwire)送入。芯片采用多电平直接模拟量存储技术, 每个采样值直接存贮在片内闪烁存贮器中,因此能够非常真实、自然地再现语音、音乐、音调和效果声,避免了一般固体录音电路因量化和压缩造成的量化噪声和"金属
05-music-coding
- Music Coding LPC-based codecs model the sound source to achieve good compression. Works well for voice. Terrible for music. What if you can’t model the source? Model the limitations of the human ear
signal_to_noise
- /* DEscr iptION */ /* 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 */ /* straig
AIC23--chinese-datasheet
- TLV320AIC23（以下简称AIC23）是TI推出的一款高性能的立体声音频Codec芯片，内置耳机输出放大器，支持MIC和LINE IN两种输入方式（二选一），且对输入和输出都具有可编程增益调节。AIC23的模数转换（ADCs）和数模转换（DACs）部件高度集成在芯片内部，采用了先进的Sigma－delta过采样技术，可以在8K到96K的频率范围内提供16bit、20bit、24bit和32bit的采样，ADC和DAC的输出信噪比分别可以达到90dB和100dB。与此同时，AIC23还具有很
zxg
- 对具有周期性的音频进行加噪声后自相关然后输出为wav音频，使用tc编写-A periodic audio add noise then related it and then output as wav audio
yuyinxinhaochuli
- 对一段语音信号进行加噪处理，然后用低通滤波、带通滤波、高通滤波对其进行处理，程序完美运行，但要注意音频文件的路径问题，否则运行会报错！-A voice signal plus noise, then low-pass filter, band pass filter, high pass filter processing, the program works perfect, but to pay attention to the audio path to the file, otherwi
Solution-AX2K-noise-issue
- AX2K噪声问题解决办法,AX2K 是两颗MP3 解码芯片，可广泛地用于车机，BoomBOX 等应用，整体方案具有功耗小，音质好，接口灵活，二次开发方便等特点，产品实测DAC 的音频指标，SNR > 79dB，THD > 79dB ，带宽可以达到20Hz~20KHz。-Solution AX2K noise issue, AX2K two MP3 decoder chip, can be widely used in the car, BoomBOX application over
vhdl_sigmadelta
- Sigma-Delta audio DAC. Connect a RC lowpass filter to the pin to reduce noise and improve the audio quality.
STM32-USB-sound-card
- 基于ARMM3内核由ST公司开发的STM32VET6编写的USB声卡，达到-普通市面上销售的USB声卡的效果，对输入声音的讯号具有降噪，高、低音激励以及混响效果的加入等作用。-Based ARMM3 kernel developed by ST STM32VET6 prepared USB sound card, reach- ordinary commercially available USB sound card' s effect on the input audio signal
signal_detector
- A behavioral descr iption of a noise cancellation based on audio detection or RF detection, written in a synthesizable subset of VHDL.
zybo_zynq_audio
- Zybo xc7z010 uation board,ssm6203音频编码器，PC端给音频输入，HPH输出口输出过滤噪音的音频，软件：xilinx vivado, vivado HLS, SDK-Zybo xc7z010 uation board, ssm6203 audio encoder, PC end to the audio input, HPH output port noise filter audio software: xilinx vivado, vivado HLS, SD
yinpin_display0925
- 实现音频的I2S通信，音频柱的显示，及其噪声的处理等功能-Realization of audio I2S communications, audio column display, and its noise processing, and other functions
DSP-Sound-processor-
- 硕士论文。主要包括：1、升频差值的matlab仿真与实现 2、延时、混响和均衡效果的实现 3、基于心理学模型和噪声整形的matlab仿真与听声测试 4、音频处理算法的DSP实现-Master thesis. Mainly includes: 1, litres, the difference of the frequency of MATLAB simulation and 2, to achieve delay, reverb, and the effect of equalization t
gai-V2.2
- Power System Transient Stability Program, can be transient stability, Realization of 10 digital audio recognition program A noise auxiliary data analysis method.
5537
- ML estimation method can be a good signal to noise ratio, Calculate the multifractal trend fluctuation analysis, LM386 audio signal amplification.
ughrn
- Calculate the multifractal trend fluctuation analysis, Independent component analysis algorithm reduces the raw data noise, Realization of 10 digital audio recognition progra.
ax703
- Add noise processing, Through repeated training fOdXGmXlate have higher recognition rate, LM386 audio signal amplification.