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  • Any screen capture software that captures webcam, microphone inputs too ?

    - by mohanr
    I am going to conduct a user study. Apart from capturing the screen while the user is interacting with the system, I also want to capture the video/audio of the user. Is there any software that in addition to capturing the screen also overlays it with the webcam/microphone inputs. The goal is to capture the complete experience of the user: key/mouse interactions with the system along with their facial/vocal responses. I know that I can maybe run a screen-capture software and also run a software for capturing webcam audio/video alongside and try to sync/overlay both these streams with timestamps. But I am going to be dealing with probably several hundred hours of data. So I am looking for a tool that can streamline the process for me amap and help me keep my sanity at end of the process. Thanks,

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  • Silverlight 4 - encoding PCM data from the microphone

    - by Richard
    Hi I've written a basic SL4 application to capture audio data from the microphone using CaptureSource. The trouble is, it's raw PCM output - which means huge and uncompressed. Given that I need this application to run purely within a SL4 environment, how can I compress the PCM audio data into something that can be delivered to a remote server more easily? In conversation, people have suggested Speex and WMA for instance, but I haven't found any libraries or examples that work without requiring reference to DLL's that won't work in a SL4 project. Thanks, Richard.

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  • Is it possible to have AVFramework and AudioToolbox framework in one app?

    - by Satyam
    I'm trying to write develop audio related application. In that, I'm using AudioToolBox framework for recording the sound. And I'm using AVFramework to play soudns. When app is stared, it will play some mp3 file using AVFramework. And also initializes Audiotoolbox. In simulator, I'm able to record and play. But when I'm testing it on iPhone, I'm getting following error for initializing AudioToolBox. 2009-12-11 22:25:51.599 StoryBook[807:207] AudioRecorder init AudioSessionInitialize failed with error: 1768843636 Can some one tell me whether we can use both AV as well as Audio Toolbox frame works in one application? Why I'm getting that error?

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  • Changing volume in Java when using JLayer.

    - by Penchant
    I'm using JLayer to play an inputstream of mp3 data from the internet. How do i change the volume of the output? I'm using this code to play it: URL u = new URL(s); URLConnection conn = u.openConnection(); conn.setConnectTimeout(Searcher.timeoutms); conn.setReadTimeout(Searcher.timeoutms); bitstream = new Bitstream(conn.getInputStream()/*new FileInputStream(quick_file)*/); System.out.println(bitstream); decoder = new Decoder(); decoder.setEqualizer(equalizer); audio = FactoryRegistry.systemRegistry().createAudioDevice(); audio.open(decoder); for(int i = quick_positions[0]; i > 0; i--){ Header h = bitstream.readFrame(); if (h == null){ return; } bitstream.closeFrame();

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  • Blackberry Player, custom data source

    - by Alex
    Hello I must create a custom media player within the application with support for mp3 and wav files. I read in the documentation i cant seek or get the media file duration without a custom datasoruce. I checked the demo in the JDE 4.6 but i have still problems... I cant get the duration, it return much more then the expected so i`m sure i screwed up something while i modified the code to read the mp3 file locally from the filesystem. Somebody can help me what i did wrong ? (I can hear the mp3, so the player plays it correctly from start to end) I must support OSs = 4.6. Thank You Here is my modified datasource LimitedRateStreaminSource.java * Copyright © 1998-2009 Research In Motion Ltd. Note: For the sake of simplicity, this sample application may not leverage resource bundles and resource strings. However, it is STRONGLY recommended that application developers make use of the localization features available within the BlackBerry development platform to ensure a seamless application experience across a variety of languages and geographies. For more information on localizing your application, please refer to the BlackBerry Java Development Environment Development Guide associated with this release. */ package com.halcyon.tawkwidget.model; import java.io.IOException; import java.io.InputStream; import java.io.OutputStream; import javax.microedition.io.Connector; import javax.microedition.io.file.FileConnection; import javax.microedition.media.Control; import javax.microedition.media.protocol.ContentDescriptor; import javax.microedition.media.protocol.DataSource; import javax.microedition.media.protocol.SourceStream; import net.rim.device.api.io.SharedInputStream; /** * The data source used by the BufferedPlayback's media player. / public final class LimitedRateStreamingSource extends DataSource { /* The max size to be read from the stream at one time. */ private static final int READ_CHUNK = 512; // bytes /** A reference to the field which displays the load status. */ //private TextField _loadStatusField; /** A reference to the field which displays the player status. */ //private TextField _playStatusField; /** * The minimum number of bytes that must be buffered before the media file * will begin playing. */ private int _startBuffer = 200000; /** The maximum size (in bytes) of a single read. */ private int _readLimit = 32000; /** * The minimum forward byte buffer which must be maintained in order for * the video to keep playing. If the forward buffer falls below this * number, the playback will pause until the buffer increases. */ private int _pauseBytes = 64000; /** * The minimum forward byte buffer required to resume * playback after a pause. */ private int _resumeBytes = 128000; /** The stream connection over which media content is passed. */ //private ContentConnection _contentConnection; private FileConnection _fileConnection; /** An input stream shared between several readers. */ private SharedInputStream _readAhead; /** A stream to the buffered resource. */ private LimitedRateSourceStream _feedToPlayer; /** The MIME type of the remote media file. */ private String _forcedContentType; /** A counter for the total number of buffered bytes */ private volatile int _totalRead; /** A flag used to tell the connection thread to stop */ private volatile boolean _stop; /** * A flag used to indicate that the initial buffering is complete. In * other words, that the current buffer is larger than the defined start * buffer size. */ private volatile boolean _bufferingComplete; /** A flag used to indicate that the remote file download is complete. */ private volatile boolean _downloadComplete; /** The thread which retrieves the remote media file. */ private ConnectionThread _loaderThread; /** The local save file into which the remote file is written. */ private FileConnection _saveFile; /** A stream for the local save file. */ private OutputStream _saveStream; /** * Constructor. * @param locator The locator that describes the DataSource. */ public LimitedRateStreamingSource(String locator) { super(locator); } /** * Open a connection to the locator. * @throws IOException */ public void connect() throws IOException { //Open the connection to the remote file. _fileConnection = (FileConnection)Connector.open(getLocator(), Connector.READ); //Cache a reference to the locator. String locator = getLocator(); //Report status. System.out.println("Loading: " + locator); //System.out.println("Size: " + _contentConnection.getLength()); System.out.println("Size: " + _fileConnection.totalSize()); //The name of the remote file begins after the last forward slash. int filenameStart = locator.lastIndexOf('/'); //The file name ends at the first instance of a semicolon. int paramStart = locator.indexOf(';'); //If there is no semicolon, the file name ends at the end of the line. if (paramStart < 0) { paramStart = locator.length(); } //Extract the file name. String filename = locator.substring(filenameStart, paramStart); System.out.println("Filename: " + filename); //Open a local save file with the same name as the remote file. _saveFile = (FileConnection) Connector.open("file:///SDCard/blackberry/music" + filename, Connector.READ_WRITE); //If the file doesn't already exist, create it. if (!_saveFile.exists()) { _saveFile.create(); } System.out.println("---------- 1"); //Open the file for writing. _saveFile.setReadable(true); //Open a shared input stream to the local save file to //allow many simultaneous readers. SharedInputStream fileStream = SharedInputStream.getSharedInputStream(_saveFile.openInputStream()); //Begin reading at the beginning of the file. fileStream.setCurrentPosition(0); System.out.println("---------- 2"); //If the local file is smaller than the remote file... if (_saveFile.fileSize() < _fileConnection.totalSize()) { System.out.println("---------- 3"); //Did not get the entire file, set the system to try again. _saveFile.setWritable(true); System.out.println("---------- 4"); //A non-null save stream is used as a flag later to indicate that //the file download was incomplete. _saveStream = _saveFile.openOutputStream(); System.out.println("---------- 5"); //Use a new shared input stream for buffered reading. _readAhead = SharedInputStream.getSharedInputStream(_fileConnection.openInputStream()); System.out.println("---------- 6"); } else { //The download is complete. System.out.println("---------- 7"); _downloadComplete = true; //We can use the initial input stream to read the buffered media. _readAhead = fileStream; System.out.println("---------- 8"); //We can close the remote connection. _fileConnection.close(); System.out.println("---------- 9"); } if (_forcedContentType != null) { //Use the user-defined content type if it is set. System.out.println("---------- 10"); _feedToPlayer = new LimitedRateSourceStream(_readAhead, _forcedContentType); System.out.println("---------- 11"); } else { System.out.println("---------- 12"); //Otherwise, use the MIME types of the remote file. // _feedToPlayer = new LimitedRateSourceStream(_readAhead, _fileConnection)); } System.out.println("---------- 13"); } /** * Destroy and close all existing connections. */ public void disconnect() { try { if (_saveStream != null) { //Destroy the stream to the local save file. _saveStream.close(); _saveStream = null; } //Close the local save file. _saveFile.close(); if (_readAhead != null) { //Close the reader stream. _readAhead.close(); _readAhead = null; } //Close the remote file connection. _fileConnection.close(); //Close the stream to the player. _feedToPlayer.close(); } catch (Exception e) { System.err.println(e.getMessage()); } } /** * Returns the content type of the remote file. * @return The content type of the remote file. */ public String getContentType() { return _feedToPlayer.getContentDescriptor().getContentType(); } /** * Returns a stream to the buffered resource. * @return A stream to the buffered resource. */ public SourceStream[] getStreams() { return new SourceStream[] { _feedToPlayer }; } /** * Starts the connection thread used to download the remote file. */ public void start() throws IOException { //If the save stream is null, we have already completely downloaded //the file. if (_saveStream != null) { //Open the connection thread to finish downloading the file. _loaderThread = new ConnectionThread(); _loaderThread.start(); } } /** * Stop the connection thread. */ public void stop() throws IOException { //Set the boolean flag to stop the thread. _stop = true; } /** * @see javax.microedition.media.Controllable#getControl(String) */ public Control getControl(String controlType) { // No implemented Controls. return null; } /** * @see javax.microedition.media.Controllable#getControls() */ public Control[] getControls() { // No implemented Controls. return null; } /** * Force the lower level stream to a given content type. Must be called * before the connect function in order to work. * @param contentType The content type to use. */ public void setContentType(String contentType) { _forcedContentType = contentType; } /** * A stream to the buffered media resource. */ private final class LimitedRateSourceStream implements SourceStream { /** A stream to the local copy of the remote resource. */ private SharedInputStream _baseSharedStream; /** Describes the content type of the media file. */ private ContentDescriptor _contentDescriptor; /** * Constructor. Creates a LimitedRateSourceStream from * the given InputStream. * @param inputStream The input stream used to create a new reader. * @param contentType The content type of the remote file. */ LimitedRateSourceStream(InputStream inputStream, String contentType) { System.out.println("[LimitedRateSoruceStream]---------- 1"); _baseSharedStream = SharedInputStream.getSharedInputStream(inputStream); System.out.println("[LimitedRateSoruceStream]---------- 2"); _contentDescriptor = new ContentDescriptor(contentType); System.out.println("[LimitedRateSoruceStream]---------- 3"); } /** * Returns the content descriptor for this stream. * @return The content descriptor for this stream. */ public ContentDescriptor getContentDescriptor() { return _contentDescriptor; } /** * Returns the length provided by the connection. * @return long The length provided by the connection. */ public long getContentLength() { return _fileConnection.totalSize(); } /** * Returns the seek type of the stream. */ public int getSeekType() { return RANDOM_ACCESSIBLE; //return SEEKABLE_TO_START; } /** * Returns the maximum size (in bytes) of a single read. */ public int getTransferSize() { return _readLimit; } /** * Writes bytes from the buffer into a byte array for playback. * @param bytes The buffer into which the data is read. * @param off The start offset in array b at which the data is written. * @param len The maximum number of bytes to read. * @return the total number of bytes read into the buffer, or -1 if * there is no more data because the end of the stream has been reached. * @throws IOException */ public int read(byte[] bytes, int off, int len) throws IOException { System.out.println("[LimitedRateSoruceStream]---------- 5"); System.out.println("Read Request for: " + len + " bytes"); //Limit bytes read to our readLimit. int readLength = len; System.out.println("[LimitedRateSoruceStream]---------- 6"); if (readLength > getReadLimit()) { readLength = getReadLimit(); } //The number of available byes in the buffer. int available; //A boolean flag indicating that the thread should pause //until the buffer has increased sufficiently. boolean paused = false; System.out.println("[LimitedRateSoruceStream]---------- 7"); for (;;) { available = _baseSharedStream.available(); System.out.println("[LimitedRateSoruceStream]---------- 8"); if (_downloadComplete) { //Ignore all restrictions if downloading is complete. System.out.println("Complete, Reading: " + len + " - Available: " + available); return _baseSharedStream.read(bytes, off, len); } else if(_bufferingComplete) { if (paused && available > getResumeBytes()) { //If the video is paused due to buffering, but the //number of available byes is sufficiently high, //resume playback of the media. System.out.println("Resuming - Available: " + available); paused = false; return _baseSharedStream.read(bytes, off, readLength); } else if(!paused && (available > getPauseBytes() || available > readLength)) { //We have enough information for this media playback. if (available < getPauseBytes()) { //If the buffer is now insufficient, set the //pause flag. paused = true; } System.out.println("Reading: " + readLength + " - Available: " + available); return _baseSharedStream.read(bytes, off, readLength); } else if(!paused) { //Set pause until loaded enough to resume. paused = true; } } else { //We are not ready to start yet, try sleeping to allow the //buffer to increase. try { Thread.sleep(500); } catch (Exception e) { System.err.println(e.getMessage()); } } } } /** * @see javax.microedition.media.protocol.SourceStream#seek(long) */ public long seek(long where) throws IOException { _baseSharedStream.setCurrentPosition((int) where); return _baseSharedStream.getCurrentPosition(); } /** * @see javax.microedition.media.protocol.SourceStream#tell() */ public long tell() { return _baseSharedStream.getCurrentPosition(); } /** * Close the stream. * @throws IOException */ void close() throws IOException { _baseSharedStream.close(); } /** * @see javax.microedition.media.Controllable#getControl(String) */ public Control getControl(String controlType) { // No implemented controls. return null; } /** * @see javax.microedition.media.Controllable#getControls() */ public Control[] getControls() { // No implemented controls. return null; } } /** * A thread which downloads the remote file and writes it to the local file. */ private final class ConnectionThread extends Thread { /** * Download the remote media file, then write it to the local * file. * @see java.lang.Thread#run() */ public void run() { try { byte[] data = new byte[READ_CHUNK]; int len = 0; //Until we reach the end of the file. while (-1 != (len = _readAhead.read(data))) { _totalRead += len; if (!_bufferingComplete && _totalRead > getStartBuffer()) { //We have enough of a buffer to begin playback. _bufferingComplete = true; System.out.println("Initial Buffering Complete"); } if (_stop) { //Stop reading. return; } } System.out.println("Downloading Complete"); System.out.println("Total Read: " + _totalRead); //If the downloaded data is not the same size //as the remote file, something is wrong. if (_totalRead != _fileConnection.totalSize()) { System.err.println("* Unable to Download entire file *"); } _downloadComplete = true; _readAhead.setCurrentPosition(0); //Write downloaded data to the local file. while (-1 != (len = _readAhead.read(data))) { _saveStream.write(data); } } catch (Exception e) { System.err.println(e.toString()); } } } /** * Gets the minimum forward byte buffer which must be maintained in * order for the video to keep playing. * @return The pause byte buffer. */ int getPauseBytes() { return _pauseBytes; } /** * Sets the minimum forward buffer which must be maintained in order * for the video to keep playing. * @param pauseBytes The new pause byte buffer. */ void setPauseBytes(int pauseBytes) { _pauseBytes = pauseBytes; } /** * Gets the maximum size (in bytes) of a single read. * @return The maximum size (in bytes) of a single read. */ int getReadLimit() { return _readLimit; } /** * Sets the maximum size (in bytes) of a single read. * @param readLimit The new maximum size (in bytes) of a single read. */ void setReadLimit(int readLimit) { _readLimit = readLimit; } /** * Gets the minimum forward byte buffer required to resume * playback after a pause. * @return The resume byte buffer. */ int getResumeBytes() { return _resumeBytes; } /** * Sets the minimum forward byte buffer required to resume * playback after a pause. * @param resumeBytes The new resume byte buffer. */ void setResumeBytes(int resumeBytes) { _resumeBytes = resumeBytes; } /** * Gets the minimum number of bytes that must be buffered before the * media file will begin playing. * @return The start byte buffer. */ int getStartBuffer() { return _startBuffer; } /** * Sets the minimum number of bytes that must be buffered before the * media file will begin playing. * @param startBuffer The new start byte buffer. */ void setStartBuffer(int startBuffer) { _startBuffer = startBuffer; } } And in this way i use it: LimitedRateStreamingSource source = new LimitedRateStreamingSource("file:///SDCard/music3.mp3"); source.setContentType("audio/mpeg"); mediaPlayer = javax.microedition.media.Manager.createPlayer(source); mediaPlayer.addPlayerListener(this); mediaPlayer.realize(); mediaPlayer.prefetch(); After start i use mediaPlayer.getDuration it returns lets say around 24:22 (the inbuild media player in the blackberry say the file length is 4:05) I tried to get the duration in the listener and there unfortunatly returned around 64 minutes, so im sure something is not good inside the datasoruce....

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  • Play a beep that loop and change the frequency/speed

    - by Bono
    Hi all, I am creating an iphone application that use audio. I want to play a beep sound that loop indefinitely. I found an easy way to do that using the upper layer AVAudioPlayer and the numberOfLoops set to "-1". It works fine. But now I want to play this audio and be able to change the rate / speed. It may works like the sound played by a car when approaching an obstacle. At the beginning the beep has a low frequency and this frequency accelerate till reaching a continuous sound biiiiiiiiiiiip ... It seems this is not feasible using the high layer AVAudioPlayer, but even looking at AudioToolBox I found no solution. Does anybody have informations about how to do that? Thanks a lot for helping me!

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  • Creating a music catalog in C# and extracting first 30 seconds as soon as the first words are sung

    - by Rad
    I already read a question: Separation of singing voice from music. I don’t need this complex audio processing. I only need some detection mechanism that would detect that there is some voice/vocal playing while the music is playing (or not playing) I need to extract first 30 seconds when a vocalist starts singing along with full band music. See question 2 below. I want to create a music catalog using ASP.NET MVC 2 and Silverlight clients and C#.NET 4.0 programming language that would be front store. On the backend I would also like to create a desktop WPF/Windows application to create the music catalog from already existing music files, most of which have metadata in them ID3v1, ID3v2.3, ID3v2.4, iTunes MP4, WMA, Vorbis Comments and APE Tags etc. I would possibly like to create a web service that would allow catalog contributors to upload a zipped album and trigger metadata extraction of music data and extraction of music segments as described below. I would be happy if I achieve no. 1 below. Let's say I have 1000ths of songs in mp3 (or other formats) grouped in subfolders using some classification (Genre, Artists, Albums, Composers or other groupings). I want to create tables in DB that would organize songs so they can be searched based on different criteria (year, length, above classification or by song title, description etc) like what iTune store allows to their customers. I want to extract metadata from various formats (I will try to get songs in mp3 format, but there may be other popular formats) and allow music Catalog manager person to add missing data from either desktop or web applications. He or other contributors can upload zipped music via an HTML or Silverlight upload or WPF. Can anybody suggest open source libraries, articles, code snippets that can do that in an automatic way using .NET and possibly SQL Server DB? My main questions are these. This is an audio processing challenge. I want to extract 2 segments of music (questions 1 and 2): 1. How to extract a music segment: 1-2 seconds before a vocal starts singing and up to 30 seconds from that point in time and 2. Much more challenging is to find repeating segments (One would usually find or recognize the names of the songs and songs are usually known by these refrains. How would I go about creating a list of songs that go great together like what Genius from iTune does? Is there any characteristics of music that can be used to match songs? The goal is for people quickly scan and recognize songs i.e. associate melody, words with a title/album so they can make intelligent decisions like buying a song, create similar mood songs. Thanks, Rad

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  • Novocaine - How to loop file playback? (iOS)

    - by lppier
    I'm using Novocaine by alexbw Novocaine for my audio project. I'm playing around with the example code here for file reading. The file plays back with no problem. I would like to loop this recording with the gap between the loops - any suggestion as to how I can do so? Thanks. Pier. // AUDIO FILE READING OHHH YEAHHHH // ======================================== NSArray *pathComponents = [NSArray arrayWithObjects: [NSSearchPathForDirectoriesInDomains(NSDocumentDirectory, NSUserDomainMask, YES) lastObject], @"testrecording.wav", nil]; NSURL *inputFileURL = [NSURL fileURLWithPathComponents:pathComponents]; NSLog(@"URL: %@", inputFileURL); fileReader = [[AudioFileReader alloc] initWithAudioFileURL:inputFileURL samplingRate:audioManager.samplingRate numChannels:audioManager.numOutputChannels]; [fileReader play]; [fileReader setCurrentTime:0.0]; //float duration = fileReader.getDuration; [audioManager setOutputBlock:^(float *data, UInt32 numFrames, UInt32 numChannels) { [fileReader retrieveFreshAudio:data numFrames:numFrames numChannels:numChannels]; NSLog(@"Time: %f", [fileReader getCurrentTime]); }];

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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 configure the framesize using AudioUnit.framework on iOS

    - by Piperoman
    I have an audio app i need to capture mic samples to encode into mp3 with ffmpeg First configure the audio: /** * We need to specifie our format on which we want to work. * We use Linear PCM cause its uncompressed and we work on raw data. * for more informations check. * * We want 16 bits, 2 bytes (short bytes) per packet/frames at 8khz */ AudioStreamBasicDescription audioFormat; audioFormat.mSampleRate = SAMPLE_RATE; audioFormat.mFormatID = kAudioFormatLinearPCM; audioFormat.mFormatFlags = kAudioFormatFlagIsPacked | kAudioFormatFlagIsSignedInteger; audioFormat.mFramesPerPacket = 1; audioFormat.mChannelsPerFrame = 1; audioFormat.mBitsPerChannel = audioFormat.mChannelsPerFrame*sizeof(SInt16)*8; audioFormat.mBytesPerPacket = audioFormat.mChannelsPerFrame*sizeof(SInt16); audioFormat.mBytesPerFrame = audioFormat.mChannelsPerFrame*sizeof(SInt16); The recording callback is: static OSStatus recordingCallback(void *inRefCon, AudioUnitRenderActionFlags *ioActionFlags, const AudioTimeStamp *inTimeStamp, UInt32 inBusNumber, UInt32 inNumberFrames, AudioBufferList *ioData) { NSLog(@"Log record: %lu", inBusNumber); NSLog(@"Log record: %lu", inNumberFrames); NSLog(@"Log record: %lu", (UInt32)inTimeStamp); // the data gets rendered here AudioBuffer buffer; // a variable where we check the status OSStatus status; /** This is the reference to the object who owns the callback. */ AudioProcessor *audioProcessor = (__bridge AudioProcessor*) inRefCon; /** on this point we define the number of channels, which is mono for the iphone. the number of frames is usally 512 or 1024. */ buffer.mDataByteSize = inNumberFrames * sizeof(SInt16); // sample size buffer.mNumberChannels = 1; // one channel buffer.mData = malloc( inNumberFrames * sizeof(SInt16) ); // buffer size // we put our buffer into a bufferlist array for rendering AudioBufferList bufferList; bufferList.mNumberBuffers = 1; bufferList.mBuffers[0] = buffer; // render input and check for error status = AudioUnitRender([audioProcessor audioUnit], ioActionFlags, inTimeStamp, inBusNumber, inNumberFrames, &bufferList); [audioProcessor hasError:status:__FILE__:__LINE__]; // process the bufferlist in the audio processor [audioProcessor processBuffer:&bufferList]; // clean up the buffer free(bufferList.mBuffers[0].mData); //NSLog(@"RECORD"); return noErr; } With data: inBusNumber = 1 inNumberFrames = 1024 inTimeStamp = 80444304 // All the time same inTimeStamp, this is strange However, the framesize that i need to encode mp3 is 1152. How can i configure it? If i do buffering, that implies a delay, but i would like to avoid this because is a real time app. If i use this configuration, each buffer i get trash trailing samples, 1152 - 1024 = 128 bad samples. All samples are SInt16.

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  • Prefered method for looping sound flash as3

    - by Brian Heylin
    Hi there, I'm having some issues with looping a sound in flash AS3, in that when I tell the sound to loop I get a slight delay at the end/beginning of the audio. The audio is clipped correctly and will play without a gap on garage band. I know that there are issues with sound in general in flash, bugs with encodings and the inaccuracies with the SOUND_COMPLETE event (And Adobe should be embarrassed with their handling of these issues) I have tried to use the built in loop argument in the play method on the Sound class and also react on the SOUND_COMPLETE event, but both cause a delay. But has anyone come up with a technique for looping a sound without any noticeable gap?

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  • Signal amplitude against time in Java

    - by wsr74ws84
    I'm racking my brain in order to solve a knotty problem (at least for me). While playing an audio file (using Java) I want the signal amplitude to be displayed against time. I mean I'd like to implement a small panel showing a sort of oscilloscope (spectrum analyzer). The audio signal should be viewed in the time domain (vertical axis is amplitude and the horizontal axis is time). Does anyone know how to do it? Is there a good tutorial I can rely on? Since I know very little about Java, I hope someone can help me.

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  • Html5 - Callback when media is ready on iPad wont work

    - by Kap
    I'm trying to add a callback to a HTML5 audio element on an iPad. I added an eventlistener to the element, the myOtherThing() starts but there is no sound. If I pause and the play the sound again the audio starts. This works in chrome. Does anyone have an idea how I can do this? myAudioElement.src = "path_to_file"; addEventListener("canplay", function(){ myAudioElement.play(); myOtherThing.start(); }); SOLVED Just wanted to share my solution here, just in case someone else needs it. As far as I understand the iPad does not trigger any events without user interactions. So to be able to use "canply", "playing" and all the other events you need to use the built in media controller. Once you press play in that controller, the events gets triggered. After that you can use your custom interface.

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  • Question on ExtAudioFileRead and AudioBuffer for iPhone SDK

    - by backspacer
    I'm developing an iPhone app that uses the Extended Audio File Services. I try to use ExtAudioFileRead to read the audio file, and store the data in an AudioBufferList structure. AudioBufferList is defined as: struct AudioBufferList { UInt32 mNumberBuffers; AudioBuffer mBuffers[1]; }; typedef struct AudioBufferList AudioBufferList; and AudioBuffer is defined as struct AudioBuffer { UInt32 mNumberChannels; UInt32 mDataByteSize; void* mData; }; typedef struct AudioBuffer AudioBuffer; I want to manipulate the mData but I wonder what does the void* mean. Why is it void*? How can I decide what data type is actually stored in mData?

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  • Can FLV AAC stream be played in Android

    - by HariKJ
    Hi, I'm trying to build a radio player and the client is providing a stream which is a FLV container with the audio being AAC When I read the headers it shows up as audio/aacp. I have tried all possible ways such as using the 1) Streaming through mediaplayer (Does not work) 2) Use the NPR mode of using a proxy stream (I get a broken pipe exception) 3) Play it in chunks ( Plays but I need the SDCard and the playback is not very great) 4) Use the GPL'd FAAD2 Library but I would have to pay the royalty fee Can some one help me out on figuring this issue out. The last option that I have is to have my client change the stream to mp3 container (which I know that it works) Regards, Hari

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  • android spectrum analysis of streaming input

    - by TheBeeKeeper
    for a school project I am trying to make an android application that, once started, will perform a spectrum analysis of live audio received from the microphone or a bluetooth headset. I know I should be using FFT, and have been looking at moonblink's open source audio analyzer ( http://code.google.com/p/moonblink/wiki/Audalyzer ) but am not familiar with android development, and his code is turning out to be too difficult for me to work with. So I suppose my questions are, are there any easier java based, or open source android apps that do spectrum analysis I can reference? Or is there any helpful information that can be given, such as; steps that need be taken to get the microphone input, put it into an fft algorithm, then display a graph of frequency and pitch over time from its output? Any help would be appreciated, thanks.

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  • iPhone SDK SDL_openAudio with Multitasking Support

    - by brokedid
    Hello, I'm playing audio from a Online Live RTPS Stream with ffmpeg(because Apple doesn't support rtsp live streaming). Now I would play my Stream in the background. I started a thread in the background and registered the music for Background support. When the Application is entering in Background the NSThread is paused, and then Resuming after returning from background. If I start playing a Music (MP3-Stream) in the Application which use official Apple Frameworks then when the App is entering Background both Streams are played. What can I do to fix this?

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  • Preinitialize BackgroundAudioPlayer in WP7?

    - by kgrevehagen
    When I am using the BackgroundAudioPlayer in my Windows Phone 7 application, it takes a lot of time to load the first time I want to play a song. Is there any way of preinitializing the BackgroundAudioPlayer before playing the first track, so that when I start playing, it starts right along? I have googled it, but no luck. I am just using BackgroundAudioPlayer.Instance when I e.g. want to play, pause, stop etc an audiotrack. Is there something other i could do to fix this? Thanks in advance!

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