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Recording_sound_from_microphone
- 允许用麦克风输入语音，记录到一个临时文件，语音时长只允许10s-permits the use of the microphone, voice input, records to a temporary document, voice long time only allowed 10s
sndpeek
- sndpeek is just what it sounds (and looks) like: real-time 3D animated display/playback can use mic-input or wav/aiff/snd/raw/mat file (with playback) time-domain waveform FFT magnitude spectrum 3D waterfall plot lissajous! (inte
SoundCatcher
- This project demonstrates an implementation of the waterfall spectrogram and use of statistical data to trigger events in near real-time. This code is an elaboration of my previous submission (SoundViewer). This demonstration utilizes the Wave classe
AudioCodeVC
- 利用Visual C++实现了音频数据信号采集，可以显示波形、可采集传感器等非音频信号，能够实时显示等。对于数字媒体专业利用VC学习有很大的入门引导作用。-The use of Visual C++ implementation of the audio data signal acquisition, can be displayed waveform can be collected sensors and other non-audio signals, real-time display
Rfc3550
- Abstract This memorandum describes RTP, the real-time transport protocol. RTP provides end-to-end network transport functions suitable for applications transmitting real-time data, such as audio, video or simulation data, over multicas
ADCPUSART
- * THE PRESENT FIRMWARE WHICH IS FOR GUIDANCE ONLY AIMS AT PROVIDING CUSTOMERS * WITH CODING INFORMATION REGARDING THEIR PRODUCTS IN ORDER FOR THEM TO SAVE * TIME. AS A RESULT, STMICROELECTRONICS SHALL NOT BE HELD LIABLE FOR ANY * DIRECT, IN
EXTIPUSART
- * THE PRESENT FIRMWARE WHICH IS FOR GUIDANCE ONLY AIMS AT PROVIDING CUSTOMERS * WITH CODING INFORMATION REGARDING THEIR PRODUCTS IN ORDER FOR THEM TO SAVE * TIME. AS A RESULT, STMICROELECTRONICS SHALL NOT BE HELD LIABLE FOR ANY * DIRECT, IN
AudioRecordB
- 音频的输入与输出Multimedia,及时输入及时输出，利用QAudioInput-The audio input and output of Multimedia, timely input in real time output, the use of QAudioInput
libg7221
- ITU 的G722.1编码器，包含Annex C，是当前复杂度最低的编码器，并且压缩质量非常好，适合高并发voip使用，实测单i5cpu可以支持3000路以上的实时并行处理，采样率16000和32000 码率16000-48000，以800为梯度。-The ITU G722.1 encoder, including Annex C, is currently the least complex of the encoder, and the compression quality is very
aec-master
- webrtc 的回声抵消(aec、aecm)算法主要包括以下几个重要模块：1.回声时延估计 2.NLMS(归一化最小均方自适应算法) 3.NLP（非线性滤波） 4.CNG(舒适噪声产生），一般经典aec算法还应包括双端检测(DT)。考虑到webrtc使用的NLMS、NLP和CNG都属于经典算法范畴，故只做简略介绍，本文重点介绍webrtc的回声时延估计算法,这也是webrtc回声抵消算法区别一般算法(如视频会议中的算法)比较有特色的地方。-webrtc echo canceller (aec,
speech-signal-filter
- 使用matlab语言对一段语音信号进行滤波处理，首先分析时域信号，之后进行傅立叶变换，转换成频域，使用巴特沃斯低通滤波器去除高频部分。-Use matlab language of a voice signal is filtered first time domain signal after the Fourier transform is converted into the frequency domain, using a Butterworth low-pass filter to
libogg
- 首先对输入音频PCM信号进行时频分析，决定MDCT的长度，即加窗，然后进行MDCT变换；同时对原始音频信号要进行FFT分析。两种变换的频谱系输入给心理声学模型单元，MDCT系数用于噪声掩蔽计算，H可结果用于音调掩蔽特性计算，共同构造总的掩蔽曲线。然后根据MDCT系数及掩蔽曲线，对频谱系数进行线性预测分析用LPC(Linear Prediction Coefficience，线性预测系数)表示频谱包络，即基底曲线(Floor Curve)；或通过线性分段逼近方式获得基底曲线。从MDCT系数中去掉
FineAudio 1.0 by bagie
- Small audio player that uses BASS components as a sound engine. BASS components can be found at: http://un4seen.com/. They are also included in the attachment. One hilarious feature of this audio player is the "Funny LED" effect which