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  • Converting Future Composer FC14 tracker music to MIDI data

    - by okw
    I have an old .FC music file from the Amiga/C64 days. It was made using Future Composer. The first four bytes of the file read as FC14 in ASCII; I'm pretty sure that's the version number. I need to dump the channels and their notes into a standard MIDI file in order to play it through my MIDI devices. Is there a way to do this with existing programs? If not, are there any specifications available on the format of these files? I do not require the samples, and I am aware that they will be lost during the conversion process.

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  • In SSIS Convert European Currency Format to United States Currency Format

    - by Rob
    I have an interesting problem. I have an SSIS package that processes account data. We are now processing files from Europe. These files are in a CSV format using text qualifiers. For an example of the problem: In the United States the currency format is 123456.99 (We purposely leave the thousands separator out). The files sent from Europe are coming in with two formats. One is 123456,99 and the other is 123.456,00. SSIS is attempting to parse the text file and place it into a NUMERIC(20,2) field. This causes a parsing error in SSIS even with the text qualifiers. If I change the field to CURRENCY it sends a conversion error. I would like for SSIS to deal with this directly without requiring the data to be in the United States format. Has anyone had this problem? Any help will be greatly appreciated. Rob

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  • .vob to h.264 MP4 Files - Worth The Effort?

    - by harper89
    When I was doing the converting to digital format a while back I chose .VOB due to no quality loss. However recently I have been informed of this h.264 compression method. Time is not an issue here, I don't mind waiting for conversions etc. I also understand that any sort of compression will reduce quality. To test I converted a 4GB .VOB to a .mp4 using h264 in handbrake and the quality loss was very very very hard to notice. From what I have understood through research Space = .mp4(h.264) Quality = .Vob Playback = Both equally supported? But these concerns have yet to be answered: My comparison was done on a computer monitor, would the quality loss be substantially noticable if I purchased a 50 inch TV in the future? Is this type of file highly supported? (I don't want to experience incompatible players) What other issues could a conversion of files such as this cause in the future?

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  • How do I convert an animated GIF to a YouTube friendly video format?

    - by Dave Webb
    My son has made some animations with Pivot Stickfigure Animator which we'd like to upload to YouTube. The problem is Pivot saves as animated GIFs which I can't upload to YouTube. The Wikipedia article recommends using Windows Movie Maker to convert GIF to WMV, but unfortunatley I'm using Window 7 for which you can get the new Windows Live Movie Maker which doesn't seem to support GIFs. I Googled and found an article which said to use Beneton Movie GIF to convert animated GIF to AVI, but this seemed to rely on a 3rd Party application which wasn't installed and so failed. Installing the missing application - pjBmp2Avi - by hand and adding it to the path still didn't allow Beneton to do the conversion. I hoped FFmpeg might do the trick but this only outputs to animated GIFs, it won't read from then. Further Googling found lots of applications with 30 day trials and so on but I was hoping for something free. So any suggestions on how I can convert an animated GIF to a movie file on Windows using free (as in beer) software?

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  • How do I convert an animated GIF to a YouTube friendly video format?

    - by Dave Webb
    My son has made some animations with Pivot Stickfigure Animator which we'd like to upload to YouTube. The problem is Pivot saves as animated GIFs which I can't upload to YouTube. The Wikipedia article recommends using Windows Movie Maker to convert GIF to WMV, but unfortunately I'm using Windows 7 for which you can get the new Windows Live Movie Maker which doesn't seem to support GIFs. I Googled and found an article which said to use Beneton Movie GIF to convert animated GIF to AVI, but this seemed to rely on a 3rd Party application which wasn't installed and so failed. Installing the missing application - pjBmp2Avi - by hand and adding it to the path still didn't allow Beneton to do the conversion. I hoped FFmpeg might do the trick but this only outputs to animated GIFs, it won't read from then. Further Googling found lots of applications with 30 day trials and so on but I was hoping for something free. So any suggestions on how I can convert an animated GIF to a movie file on Windows using free (as in beer) software?

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  • Video converters don't work anymore after reinstalling Windows

    - by tassiekev
    A few days ago, I decided to reinstall Windows 7 as my HD partition seemed to be nearly full and things were slowing down. I'd been using Handbrake almost exclusively to convert TV recordings and used Freemake on occasion. Following the reinstall, I can't get either to work: Handbrake says it's encoding for about 2 seconds and then says it's finished, but there are no converted files of any size. Freemake just says 'Conversion Error' and won't go any further. As an experiment I tried two programs that I don't normally use, VideoReDo & Any Video Converter. Both worked fine. Anyone got any clues?

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  • PDF to HTML - batch converter - most reliable and accurate free AND paid for software?

    - by Rob
    I'm look for either a free or paid-for (about 50$/40pounds) BATCH PDF to HTML converter to convert several PDF files at once. Needs to be able to handle vectored and bitmap images within the file, outputting both as jpegs referenced by the html pages. I've tried iorigsoft paid-for PDF to HTML - problems it seems to hang or just go idle, and the stuff it actually converts have broken links - the wrong name is used for constituent chapters as html. Also tried application from intrapdf.com but this crashes near the beginning of the conversion, consitently. Looked at opensource tools but they look equally flakey or use old PDF versions. Need it on Windows 7 32bit home. Thoughts?

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  • Problem with waveOutWrite and waveOutGetPosition deadlock

    - by MusiGenesis
    I'm working on an app that plays audio continuously using the waveOut... API from winmm.dll. The app uses "leapfrog" buffers, which are basically a bunch of arrays of samples that you dump into the audio queue. Windows plays them seamlessly in sequence, and as each buffer completes Windows calls a callback function. Inside this function, I load the next set of samples into the buffer, process them however, and then dump the buffer back into the audio queue. In this way, the audio plays indefinitely. For animation purposes, I'm trying to incorporate waveOutGetPosition into the application (since the "buffer done" callbacks are irregular enough to cause jerky animation). waveOutGetPosition returns the current position of playback, so it's hyper-precise. The problem is that in my application, making calls to waveOutGetPosition eventually causes the application to lock up - the sound stops and the call never returns. I've boiled things down to a simple app that demonstrates the problem. You can run the app here: http://www.musigenesis.com/SO/waveOut%20demo.exe If you just hear a tiny bit of piano over and over, it's working. It's just meant to demonstrate the problem. The source code for this project is here: http://www.musigenesis.com/SO/WaveOutDemo.zip The first button runs the app in leapfrog mode without making the calls to waveOutGetPosition. If you click this, the app will play forever without breaking (the X button will close it and shut it off). The second button starts the leapfrogger and also starts a forms timer that calls the waveOutGetPosition and displays the current position. Click this and the app will run for a short while and then lock up. On my laptop, it usually locks up in 15-30 seconds; at most it's taken a minute. I have no idea how to fix this, so any help or suggestions would be most welcome. I've found very few posts on this issue, but it seems that there is a potential deadlock, either from multiple calls to waveOutGetPosition or from calls to that and waveOutWrite that occur at the same time. It's possible that I'm calling this too frequently for the system to handle.

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  • i2s0: transmitter underrun (0)

    - by tbarbe
    were doing some audio stuff and I keep seeing this in the Organizer Console. Sun May 2 20:16:48 unknown kernel[0] : i2s0: transmitter underrun (0) Are these transmitter underruns bad? I think its just when were shutting down audio input...but could a few of these cause some issues later on?

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  • iPhone SDK: How to record voices with ambient noise supression?

    - by Harkonian
    Can anyone point me in the right direction on how I would minimize ambient noise while recording someone speaking using the iPhone SDK Core Audio? I'm guessing a band-pass filter that eliminates any frequencies above and below the human vocal range might work. I have no idea how I would implement band filters on audio in the SDK though. The optimum solution would be one that eliminates the noise from the stream before it is written to memory/disk. Some sample code would be appreciated.

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  • iPhone: CPU power to do DSP/Fourier transform/frequency domain?

    - by mahboudz
    I want to analyze MIC audio on an ongoing basis (not just a snipper or prerecorded sample), and display frequency graph and filter out certain aspects of the audio. Is the iPhone powerful enough for that? I suspect the answer is a yes, given the Google and iPhone voice recognition, Shazaam and other music recognition apps, and guitar tuner apps out there. However, I don't know what limitations I'll have to deal with. Anyone play around with this area?

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  • iPhone SDK audioSession question.

    - by Morion
    Hi to all. In my app i record and play audio at the same time. The app is almost finished. But there is one thing, that annoying me. When audio session is set to PlayAndRecord, sounds become quiet in comparison with the same sounds with the SoloAmbient category. Is there any way to make sound louder using PlayAndRecord?

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  • text to image conversion with JSON response

    - by ruhit
    i have made an application to convert text into image and it working out fine,now i am using JSON for conversion,it also working but except only two fields.....why i dont know, my codes are given below...please help me , is there any better way? // img.html excoflare enter your text here: Font Size:          Color:               Font:                 Height:              Width:                                            img.php require_once 'JSON/JSON.php'; header('Content-type: application/json'); header ("Content-type: image/png"); $text =$_REQUEST['text']; $text=json_encode($text); $path="C:\wamp\www\image"; $height=$_REQUEST['height']; $width=$_REQUEST['width']; define("WIDTH", $width); json_encode(WIDTH); define("HEIGHT", $height); json_encode(HEIGHT); $img = imagecreate(WIDTH, HEIGHT); imagesavealpha($img, true); $trans_colour = imagecolorallocatealpha($img, 0, 0, 0, 127); imagefill($img, 0, 0, $trans_colour); $getcolor=$_REQUEST['color']; switch($getcolor) { case 'red': $red = imagecolorallocate($img, 223, 14, 91); $color=json_encode($red); break; case 'white': $white = imagecolorallocate($img, 255, 255, 255); $color=json_encode($white); break; case 'black': $black = imagecolorallocate($img, 0,0,0); $color=json_encode($black); break; case 'grey': $grey = imagecolorallocate($img, 128, 128, 128); $color=json_encode($grey); break; // default: // break; } //$background_color = imagecolorallocate ($img, 25, 25, 25); $font = $_REQUEST['font']; //$font=json_encode($font); $fontsize =$_REQUEST['size']; //$fontsize=json_encode($fontsize); imagettftext($img, $fontsize, 0, 20, 20, $color, $font, $text); //Create image imagepng($img); imagepng($img,"$path/img.png"); //destroy image ImageDestroy($img); //header ('Content-type: image/png'); ? Thanks in advance..

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  • What is a lightweight cross platform WAV playing library?

    - by Lokkju
    I'm looking for a lightweight way to make my program (written in C) be able to play audio files on either windows or linux. I am currently using windows native calls, which is essentially just a single call that is passed a filename. I would like something similar that works on linux. The audio files are Microsoft PCM, Single channel, 22Khz Any Suggestions?

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  • Identifying voice as male or female

    - by duder
    I'm not much into audio engineering, so please be easy on me. I'm receiving an audio file as input, and need to detect whether the speaker is male or female. Any ideas how to go about doing this? I'm using php, but am open to using other languages, and don't mind learning a little bit of sound theory as long as the time is proportionate to the task.

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  • Flash Media Live Encoder

    - by jeph perro
    I am using Adobe Flash Media Live Encoder to stream live video to a video streaming server. The webcam is in our office pointed out the window. Thankfully, Flash Media Live Encoder has a checkbox to un-include audio. I am wondering how I can push a recorded message to the audio ( or music ). Is there any way I can play a recording and have it behave like a microphone?

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  • How does the Ableton warp algorithm work exactly?

    - by pepperdreamteam
    I'm looking for any documentation or definitive information on Ableton's warp feature. I understand that it has something to do with finding transients, aligning them with an even rhythm and shifting audio samples accordingly. I'm hoping to find ways to approximate warping with more basic audio editing tools. I understand that this is ableton's unique device, really any information about how it works would be helpful. So...does anyone have any 411?

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  • Any tips for how to build a LED sytem thet will light up to music?

    - by daniels
    So basically I would like somehow that given an audio file as input (most likely mp3 or I can use some audio engine that will handle other types too) from my computer to control some LED lights so they will be something like an oscilloscope, like the one in winamp. What would I need to be able to do this? I'm interested in building thing up all by myself, coding, hardware, etc.. I'm going with C++ on Windows.

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  • AudioTrack lag: obtainBuffer timed out

    - by BTR
    I'm playing WAVs on my Android phone by loading the file and feeding the bytes into AudioTrack.write() via the FileInputStream BufferedInputStream DataInputStream method. The audio plays fine and when it is, I can easily adjust sample rate, volume, etc on the fly with nice performance. However, it's taking about two full seconds for a track to start playing. I know AudioTrack has an inescapable delay, but this is ridiculous. Every time I play a track, I get this: 03-13 14:55:57.100: WARN/AudioTrack(3454): obtainBuffer timed out (is the CPU pegged?) 0x2e9348 user=00000960, server=00000000 03-13 14:55:57.340: WARN/AudioFlinger(72): write blocked for 233 msecs, 9 delayed writes, thread 0xba28 I've noticed that the delayed write count increases by one every time I play a track -- even across multiple sessions -- from the time the phone has been turned on. The block time is always 230 - 240ms, which makes sense considering a minimum buffer size of 9600 on this device (9600 / 44100). I've seen this message in countless searches on the Internet, but it usually seems to be related to not playing audio at all or skipping audio. In my case, it's just a delayed start. I'm running all my code in a high priority thread. Here's a truncated-yet-functional version of what I'm doing. This is the thread callback in my playback class. Again, this works (only playing 16-bit, 44.1kHz, stereo files right now), it just takes forever to start and has that obtainBuffer/delayed write message every time. public void run() { // Load file FileInputStream mFileInputStream; try { // mFile is instance of custom file class -- this is correct, // so don't sweat this line mFileInputStream = new FileInputStream(mFile.path()); } catch (FileNotFoundException e) {} BufferedInputStream mBufferedInputStream = new BufferedInputStream(mFileInputStream, mBufferLength); DataInputStream mDataInputStream = new DataInputStream(mBufferedInputStream); // Skip header try { if (mDataInputStream.available() > 44) mDataInputStream.skipBytes(44); } catch (IOException e) {} // Initialize device mAudioTrack = new AudioTrack(AudioManager.STREAM_MUSIC, ConfigManager.SAMPLE_RATE, AudioFormat.CHANNEL_CONFIGURATION_STEREO, AudioFormat.ENCODING_PCM_16BIT, ConfigManager.AUDIO_BUFFER_LENGTH, AudioTrack.MODE_STREAM); mAudioTrack.play(); // Initialize buffer byte[] mByteArray = new byte[mBufferLength]; int mBytesToWrite = 0; int mBytesWritten = 0; // Loop to keep thread running while (mRun) { // This flag is turned on when the user presses "play" while (mPlaying) { try { // Check if data is available if (mDataInputStream.available() > 0) { // Read data from file and write to audio device mBytesToWrite = mDataInputStream.read(mByteArray, 0, mBufferLength); mBytesWritten += mAudioTrack.write(mByteArray, 0, mBytesToWrite); } } catch (IOException e) { } } } } If I can get past the artificially long lag, I can easily deal with the inherit latency by starting my write at a later, predictable position (ie, skip past the minimum buffer length when I start playing a file).

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  • How to set a native object property

    - by theunilife
    ok so im creating a jquery plugin that will allow me to use the new html5 Audio interface and im trying to create an option that is an object that you will be able to set the various listeners but i dont seem to be able to set those options to the listener property of the Audio object.

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