waveform

问题 I've already searched at Stackoverflow and google, but haven't found what I'm looking for. So far I got the audio raw data(WAV File) and I want to visualize it. private void Form1_Load(object sender, EventArgs e) { FileStream fs = new FileStream("D:\\tada.wav", FileMode.Open); BinaryReader reader = new BinaryReader(fs); char[] data = new char[4]; long fsize; long wfxSize; long dataSize; WaveFormatEx wfx; //RIFF reader.Read(data, 0, 4); fsize = reader.ReadInt32(); //WAVE reader.Read(data, 0, 4

问题 Is there an algorithm that can be used to determine whether a sample of data taken at fixed time intervals approximates a sine wave? 回答1: Take the fourier transform which transforms the data into a frequency table (search for fft, fast fourier transformation, for an implementation. For example, FFTW). If it is a sinus or cosinus, the frequency table will contain one very high value corresponding to the frequency you're searching for and some noise at other frequencies. Alternatively, match

问题 Is it possible? I'd like run over a batch of audio files and produce graphic waveform files from them. Similar to what SoundCloud displays, but perhaps the module(s) im seeking will offer a few more options. Whats out there? whats good? 回答1: See this example, it is exactly what you want in pure Python. You have another options too, but using audiolab . 来源： https://stackoverflow.com/questions/5010941/generating-a-audio-waveform-graphic-within-python

问题 I've looked around Stack Overflow for an answer to this, but nowhere seems to give the correct answer or direction... My project will allow a user to upload a WAV, which ultimately will be converted to a low quality MP3 using FFmpeg on the server and it'll all be stored and served on Amazon S3. The next obstacle is working out how to extract a reliable waveform visualisation from this uploaded sound. I'm using Python and Django on Linux Ubuntu 10 on a VPS for this project... I'm, at the vert

问题 I'm looking for how to draw the sound waves according to music. I want waves like this image here is some discussion about displaying Waves from music WaveForm on IOS rendering a waveform on an iphone audio waveform visualisation with iPhone Github Example Links https://github.com/gyetvan-andras/cocoa-waveform https://github.com/benalavi/waveform https://github.com/JingWZ/WaveView https://github.com/gyetvan-andras/cocoa-waveform https://github.com/ioslovers/ATTabandHoldAudioRecord https:/

问题 Yesterday I finalised the code for detecting the audio energy of a track displayed over time, which I will eventually use as part of my audio thumbnailing project. However I would also like a method that can detect the pitch of a track displayed over time, so I have 2 options from which to base my research upon. [y, fs, nb] = wavread('Three.wav'); %# Load the signal into variable y frameWidth = 441; %# 10 msec numSamples = length(y); %# Number of samples in y numFrames = floor(numSamples

问题 I was wondering if anyone has any suggestions on how to go about rendering a waveform of an audio file. I wold like to enable the user to set an in and out point of an audio track and I need to have a waveform so you can see where to put the points. Are there any libraries available for this or does it need to be a completely custom solution? Is it even called a waveform ? Maybe there's a better word for it so I can do some more searching. TIA ! 回答1: I'm reposting my answer from this question

问题 I'm writing an audio waveform editor in Cocoa with a wide range of zoom options. At its widest, it shows a waveform for an entire song (~10 million samples in view). At its narrowest, it shows a pixel accurate representation of the sound wave (~1 thousand samples in a view). I want to be able to smoothly transition between these zoom levels. Some commercial editors like Ableton Live seem to do this in a very inexpensive fashion. My current implementation satisfies my desired zoom range, but

问题 I am trying to implement a low pass filter in Java. My requirement is very simple,I have to eliminate signals beyond a particular frequency (Single dimension). Looks like Butterworth filter would suit my need. Now the important thing is that CPU time should be as low as possible. There would be close to a million sample the filter would have to process and our users don't like waiting too long. Are there any readymade implementation of Butterworth filters which has optimal algorithms for

问题 I'm writing a c program to generate a sinusoidal wave that slowly ramps up frequency from f1 to f2 for a giving time interval. I have written this c program to ramp the frequency from 0 to 10 Hz but the problem is that the frequency changes after completion of 360 degrees. If I try to change the frequency between 0 and 360 degree that the transition is not smooth and it is abrupt. This is the equation the sin that I have used y = Amplitude*sin(freq*phase) int main(int argc, char *argv[]) {