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Search found 40 results on 2 pages for 'waveform'.

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  • High quality graph/waveform display component in C#.

    - by dlopeztt
    Hi, I'm looking for a fast, professionally looking and customizable waveform display component in C#. I'm wanting to display mainly real-time audio waveforms (fast!) in both time and frequency domain. I would like the ability to zoom, change axis settings, display multiple channels, customize the feel and colors etc... Anybody knows of anything, whether commercial or not? Thank you! Diego

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  • Drawing a waveform in C#

    - by user488792
    Hi! I want to be able to display a WaveForm in C#, along with some simple features such as zooming and selection. I already have the data as a short[] of amplitude values. However, I am an amateur when it comes to hardcoding GUI. I have already found a possible helper class WaveFormClass that may help me achieve this but as a backup, I want to learn how to manually do it. So may I ask for some methods and possibly some links that will help? Thanks!

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  • Drawing Color Spectrum with Waveform

    - by TheDarkIn1978
    i've come across this ActionScript sample, which demonstrates drawing of the color spectrum, one line at a time via a loop, using waveforms. however, the waveform location of each RGB channel create a color spectrum that is missing colors (pure yellow, cyan and magenta) and therefore the spectrum is incomplete. how can i remedy this problem so that the drawn color spectrum will exhibit all colors? // Loop through all of the pixels from '0' to the specified width. for(var i:int = 0; i < nWidth; i++) { // Calculate the color percentage based on the current pixel. nColorPercent = i / nWidth; // Calculate the radians of the angle to use for rotating color values. nRadians = (-360 * nColorPercent) * (Math.PI / 180); // Calculate the RGB channels based on the angle. nR = Math.cos(nRadians) * 127 + 128 << 16; nG = Math.cos(nRadians + 2 * Math.PI / 3) * 127 + 128 << 8; nB = Math.cos(nRadians + 4 * Math.PI / 3) * 127 + 128; // OR the individual color channels together. nColor = nR | nG | nB; }

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  • Trimming bit of the beginning off a recorder waveform

    - by Lowgain
    I've got a flash 10.1 app that lets me record microphone input to a wav without a media server, which I am saving to an Amazon S3 bucket. I have another process running on a server which gets wavs from this bucket, converts to mp3 using LAME and puts them into another bucket. This all works fine, but in converting wav mp3, about 0.1sec or so of silence is added to my sound. In the application this are being used in, perfect sync is critical, so I need to trim off that little bit. If I have to trim it off the original waveform that is okay, I don't expect anything important to happen in that first fraction of a second. What is the best way to go about this? I am using Adobe's WavWriter to convert by ByteArray into a proper waveform. Is there a way I can easily trim off the first few samples from my ByteArray without invalidating the structure? Alternatively, is there a good server-side tool I can use to trim the wav before running it through LAME, or an argument I can give LAME? Or, could I even trim that sound off the mp3 after it has been converted? Thanks!

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  • On the fly volume compression (waveform capping) for VLC or OS X

    - by Grammar Nazi
    I'm looking to impose a hard limit on a movie. In particular when I have earbuds in and a reasonable volume set, loud or sudden sounds (gun shot, dramatic sound effects, gun fire, etc) are loud enough to hurt. Lowering the volume makes speech and other sounds hard to hear. Is there a third party on-the-fly solution for this, or a plug-in for VLC that I can use?

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  • How to write C++ audio processing applications?

    - by cesko82
    Hi everyone, I'm an Electronics and Telecommunications student, next to my graduation. I'm gonna work on a project that involves my knowledge about DSP, music and audio in general. I allready know all the basic mathematic instruments and all the stuff I need to manage it, such as FFT, circular convolution ecc ecc. I want to learn C++ programming basically for one reason: it's very important in the professional world!!! And I think it's one of the most used to write applications working with audio, especially when it's about real time processing. Ok, after this small introduction I would like to know first, which are the most used libraries to work with audio processing in c++?? I was longer looking on the web but i couldn't find a lo of working stuff. (I work under linux with eclipse CDT enviroment). Then I would like to know if there are good sources to learn how to write some working code, such as for example how to write a simple low pass filter. Basically now i will not write real time applications, I would like to start from the processing of a WAV file, or even better an MP3 file, so basically on vectors of samples. Let's say that basically for now I would like to extract the waveform from an audio file, and save it to a thumbnail or to a PNG image. Ok, for now I think it's all I would need. Any ideas, advices, libraries, books, interesting sources about that? Thanks a lot in advance for any kind of answer. Giovanni.

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  • Creating simple waveforms with CoreAudio

    - by Koning Baard
    I am new to CoreAudio, and I would like to output a simple sine wave and square wave with a given frequency and amplitude through the speakers using CA. I don't want to use sound files as I want to synthesize the sound. What do I need to do this? And can you give me an example or tutorial? Thanks.

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  • Create a Sin wave line with Processing

    - by Nintari
    Hey everybody, first post here, and probably an easy one. I've got the code from Processing's reference site: float a = 0.0; float inc = TWO_PI/25.0; for(int i=0; i<100; i=i+4) { line(i, 50, i, 50+sin(a)*40.0); a = a + inc; } http://processing.org/reference/sin_.html However, what I need is a line that follows the curve of a Sin wave, not lines representing points along the curve and ending at the 0 axis. So basically I need to draw an "S" shape with a sin wave equation. Can someone run me through how to do this? Thank you in advance, -Askee

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  • How to implement low pass filter using java

    - by chai
    Hello Everyone, 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 filtering. Regards, Chaitannya

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  • How Do I Convert text to a WAV file With Inaudible Waveform?

    - by Scott
    I am trying to create an audio watermarking system. I figure the best solution is to create an audio file (WAV) based on a unique string of text and then combine this with the original wav. The part that makes this tricky (for me anyway) is: How do I convert the text string to a wav? How do I ensure that the resulting WAV form is inaudible (or at least barely noticeable to the listener). I would prefer this be done server side (via PHP, etc) but if the processing load isn't too much then would be ok with something in Flash or Javascript. I'd be willing to pay someone to create me a workable solution (complete source code that functions as described). Thanks, Scott!

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  • How to repaint a form so that it does not disappear on minimizing?

    - by gn
    I have a form in which i paint a waveform on a button click that is as soon as i click button, the waveform displays. Now when i minimize the form and maximize it again, the waveform disappears.How to repaint it? I have seen people using paint event but i dont know how to use it after/inside the button click event. Please help.

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  • What Qt container class to use for a sorted list?

    - by Dave
    Part of my application involves rendering audio waveforms. The user will be able to zoom in/out of the waveform. Starting at fully zoomed-out, I only want to sample the audio at the necessary internals to draw the waveform at the given resolution. Then, when they zoom in, asynchronously resample the "missing points" and provide a clearer waveform. (Think Google Maps.) I'm not sure the best data structure to use in Qt world. Ideally, I would like to store data samples sorted by time, but with the ability to fill-in points as needed. So, for example, the data points might initially look like: data[0 ms] = 10 data[10 ms] = 32 data[20 ms] = 21 ... But when they zoom in, I would get more points as necessary, perhaps: data[0 ms] = 10 data[2 ms] = 11 data[4 ms] = 18 data[6 ms] = 30 data[10 ms] = 32 data[20 ms] = 21 ... Note that the values in brackets are lookup values (milliseconds), not array indices. In .Net I might have used a SortedList<int, int>. What would be the best class to use in Qt? Or should I use a STL container?

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  • How to speed up drawing of scaled image? Audio playback chokes during window resize.

    - by Paperflyer
    I am writing an audio player for OSX. One view is a custom view that displays a waveform. The waveform is stored as a instance variable of type NSImage with an NSBitmapImageRep. The view also displays a progress indicator (a thick red line). Therefore, it is updated/redrawn every 30 milliseconds. Since it takes a rather long time to recalculate the image, I do that in a background thread after every window resize and update the displayed image once the new image is ready. In the meantime, the original image is scaled to fit the view like this: // The drawing rectangle is slightly smaller than the view, defined by // the two margins. NSRect drawingRect; drawingRect.origin = NSMakePoint(sideEdgeMarginWidth, topEdgeMarginHeight); drawingRect.size = NSMakeSize([self bounds].size.width-2*sideEdgeMarginWidth, [self bounds].size.height-2*topEdgeMarginHeight); [waveform drawInRect:drawingRect fromRect:NSZeroRect operation:NSCompositeSourceOver fraction:1]; The view makes up the biggest part of the window. During live resize, audio starts choking. Selecting the "big" graphic card on my Macbook Pro makes it less bad, but not by much. CPU utilization is somewhere around 20-40% during live resizes. Instruments suggests that rescaling/redrawing of the image is the problem. Once I stop resizing the window, CPU utilization goes down and audio stops glitching. I already tried to disable image interpolation to speed up the drawing like this: [[NSGraphicsContext currentContext] setImageInterpolation:NSImageInterpolationNone]; That helps, but audio still chokes during live resizes. Do you have an idea how to improve this? The main thing is to prevent the audio from choking.

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  • Remove a child of root of XML using java

    - by Sachin Mhetre
    This is my xml file. - <deviceparameters> - <parameter componenttype="TextBox"> <name>Operating Type</name> <oid>1.3.6.1.4.1.31163.5.1.1</oid> <writable>true</writable> <description>The operating type defines which waveform type is used. This configuration takes several seconds to execute</description> - <paramvalues type="Integer"> <value default="No">123</value> </paramvalues> </parameter> - <parameter componenttype="TextBox"> <name>Active Waveform Status</name> <oid>1.3.6.1.4.1.31163.5.1.2</oid> <writable>false</writable> <description>Show the status of the waveform configured by operatingType</description> - <paramvalues type="String"> <value default="yes">Active</value> </paramvalues> </parameter> </deviceparameters> I want to remove node with name 'Active wavwform Status'. How can I reomve that particular node from xml. The code I have written is below. rootElement = doc.getDocumentElement(); NodeList nList = doc.getElementsByTagName("parameter"); String nodeName = TF_name.getText(); System.out.println(""+nList.getLength()); for (int temp = 0; temp < nList.getLength();temp++) { Node nNode = nList.item(temp); Element eElement = (Element) nNode; String upname1 = getTagValue("name", eElement); if(upname1.equals(nodeName)) { System.out.println("Parent: "+nNode.getParentNode().getNodeName()); System.out.println("nodename: "+nNode.getNodeName()); System.out.println("rmoving ...."); rootElement.removeChild(nNode); System.out.println("removed..."); } }

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  • Looking for audio-sample-managing/librarian software

    - by Fuxi
    hi all, i'm looking for a good software for managing audio samples (.wav) it should be capable of: quickly browsing through directory/file-structure (eg. cursorkeys) + autoplay waveform-vizualisation + possibility to set cursor position (for playing) possibility to create own favourite-folder-structure + quickly sort in by sample-type any recommendations? thanks

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  • FMOD on non-playing audio

    - by coldrising
    Hey, is there any way to get the audio spectrum of a section of a song using FMOD if it is not playing? Can I render a full song waveform using FMOD (+opengl/openframeworks/etc.) before the song is playing?

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  • Silverlight audio components

    - by Shurup
    Do you know any good commertial or free audio / sound components that can be used in Silverlight (better on the client)? Desired features: visualizing (waveform, peakform etc., zoom, editing volume, pan, start and end markers etc.) editing (channels: volume, pan, mute, fading, etc., merge, cut, copy, paste, etc., effects) mixing different sound tracks, creating a downmix exporting result in some in Silverlight usable (compressed) formats Some useful projects that I've founded: Saluse MediaKit for Silverlight (MP3 decoder, AudioPreProcessor, visualizers, effects) Microsoft Design Toolbox sound mixer (incl. source code and video) and Demo

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  • Simple wave generator with SDL in c++

    - by Vlad Popescu
    i am having problems understanding how the audio part of the sdl library works now, i know that when you initialize it, you have to specify the frequency and a callback<< function, which i think is then called automatically at the given frequency. can anyone who worked with the sdl library write a simple example that would use sdl_audio to generate a 440 hz square wave (since it is the simplest waveform) at a sampling frequency of 44000 hz? thanks in advance

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  • Real-time spectrum analyzer with API

    - by bobobobo
    I'm looking for a C or C++ API that will give me real-time spectrum analysis of a waveform on Windows. I'm not entirely sure how large a sample window it should need to determine frequency content, but the smaller the better. For example, if it can work with a 0.5 second long sample and determine frequency content to the Hz, that would be wicked-awesome.

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