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Search found 37 results on 2 pages for 'audioformat'.

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  • How to easily Generate Synth Chords Sounds in Android?

    - by barata7
    How to easily Generate Synth Chords Sounds in Android? I wanna be able to generate dynamically an in game Music using 8bit. Tried with AudioTrack, but did not get good results of nice sounds yet. Any examples out there? I have tried the following code without success: public class BitLoose { private final int duration = 1; // seconds private final int sampleRate = 4200; private final int numSamples = duration * sampleRate; private final double sample[] = new double[numSamples]; final AudioTrack audioTrack; public BitLoose() { audioTrack = new AudioTrack(AudioManager.STREAM_MUSIC, sampleRate, AudioFormat.CHANNEL_CONFIGURATION_MONO, AudioFormat.ENCODING_PCM_8BIT, numSamples, AudioTrack.MODE_STREAM); audioTrack.play(); } public void addTone(final int freqOfTone) { // fill out the array for (int i = 0; i < numSamples; ++i) { sample[i] = Math.sin(2 * Math.PI * i / (sampleRate / freqOfTone)); } // convert to 16 bit pcm sound array // assumes the sample buffer is normalised. final byte generatedSnd[] = new byte[numSamples]; int idx = 0; for (final double dVal : sample) { // scale to maximum amplitude final short val = (short) ((((dVal * 255))) % 255); // in 16 bit wav PCM, first byte is the low order byte generatedSnd[idx++] = (byte) (val); } audioTrack.write(generatedSnd, 0, sampleRate); } public void stop() { audioTrack.stop(); }

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  • Stop method not working

    - by avoq
    Hi everyone , can anybody tell me why the following code doesn't work properly? I want to play and stop an audio file. I can do the playback but whenever I click the stop button nothing happens. Here's the code : Thank you. .................. import java.io.*; import javax.sound.sampled.*; import javax.swing.*; import java.awt.event.*; public class SoundClipTest extends JFrame { final JButton button1 = new JButton("Play"); final JButton button2 = new JButton("Stop"); int stopPlayback = 0; // Constructor public SoundClipTest() { button1.setEnabled(true); button2.setEnabled(false); // button play button1.addActionListener( new ActionListener(){ public void actionPerformed(ActionEvent e){ button1.setEnabled(false); button2.setEnabled(true); play(); }// end actionPerformed }// end ActionListener );// end addActionListener() // button stop button2.addActionListener( new ActionListener(){ public void actionPerformed( ActionEvent e){ //Terminate playback before EOF stopPlayback = 1; }//end actionPerformed }//end ActionListener );//end addActionListener() this.setDefaultCloseOperation(JFrame.EXIT_ON_CLOSE); this.setTitle("Test Sound Clip"); this.setSize(300, 200); JToolBar bar = new JToolBar(); bar.add(button1); bar.add(button2); bar.setOrientation(JToolBar.VERTICAL); add("North", bar); add("West", bar); setVisible(true); } void play() { try { final File inputAudio = new File("first.wav"); // First, we get the format of the input file final AudioFileFormat.Type fileType = AudioSystem.getAudioFileFormat(inputAudio).getType(); // Then, we get a clip for playing the audio. final Clip c = AudioSystem.getClip(); // We get a stream for playing the input file. AudioInputStream ais = AudioSystem.getAudioInputStream(inputAudio); // We use the clip to open (but not start) the input stream c.open(ais); // We get the format of the audio codec (not the file format we got above) final AudioFormat audioFormat = ais.getFormat(); c.start(); if (stopPlayback == 1 ) {c.stop();} } catch (UnsupportedAudioFileException e) { e.printStackTrace(); } catch (IOException e) { e.printStackTrace(); } catch (LineUnavailableException e) { e.printStackTrace(); } }// end play public static void main(String[] args) { //new SoundClipTest().play(); new SoundClipTest(); } }

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  • I write bad wave files using Java

    - by Cliff
    I'm writing out wave files in Java using AudioInputStream output = new AudioInputStream(new ByteArrayInputStream(rawPCMSamples), new AudioFormat(22000,16,1,true,false), rawPCMSamples.length) AudioSystem.write(output, AudioFileFormat.Type.WAVE, new FileOutputStream('somefile.wav')) And I get what appears to be corrupt wave files on OSX. They won't play from Finder however using the same code behind a servlet writing directly to the response stream and setting the Content-Type to audio/wave seems to play fine in quicktime. What gives?

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  • Java writes bad wave files

    - by Cliff
    I'm writing out wave files in Java using AudioInputStream output = new AudioInputStream(new ByteArrayInputStream(rawPCMSamples), new AudioFormat(22000,16,1,true,false), rawPCMSamples.length) AudioSystem.write(output, AudioFileFormat.Type.WAVE, new FileOutputStream('somefile.wav')) And I get what appears to be corrupt wave files on OSX. They won't play from Finder however using the same code behind a servlet writing directly to the response stream and setting the Content-Type to audio/wave seems to play fine in quicktime. What gives?

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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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  • iPhone SDK Smaller CAF files: lower recording quality with Audio Queues?

    - by Nick
    In my iPhone app I have voice recording functionality the utilizes Audio Queue voice recording functions of the SDK. I'm saving directly to CAF format and using the following settings for the AudioStreamBasicDescription reference: audioFormat.mFormatID = kAudioFormatLinearPCM; I can see that there are other format ids I could use like: kAudioFormatLinearPCM kAudioFormatAppleLossless kAudioFormatAppleIMA4 kAudioFormatiLBC kAudioFormatULaw kAudioFormatALaw My knowledge of sound formats is very limited so my question is... which of these should I use to create the lowest compressed audio recording files? Plus, are there other settings I should apply to lower the quality and filesize even further?

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  • Write wave files to memory in Java

    - by Cliff
    I'm trying to figure out why my servlet code creates wave files with improper headers. I use: AudioSystem.write( new AudioInputStream( new ByteArrayInputStream(memoryBytes), new AudioFormat(22000, 16, 1, true,false), memoryBytes.length ), AudioFileFormat.Type.WAVE, servletOutputStream ); taking a byte array from memory containing raw PCM samples and a servlet output stream that gets returned to the client. In the result I get a normal wave file but with zeros in the chunk size fields. Is the API broken? I would think that the size could be filled in using the size passed in the audio input stream. But now, after typing this out I'm thinking its not making this info available to the outer write() method on AudioSystem. It seems like the AudioSystem.write call needs a size parameter unless it is able to pull the size from the stream... which wouldn't work with an arbitrary sized stream. Does anyone know how to make this example work?

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  • Could not realize media player

    - by user556894
    I am using this code to run avi file using jmf but the error come like "Could not realize media player" and how to open all video format using jmf import javax.media.*; import javax.media.format.*; import java.io.*; import java.util.*; public class Test{ public static void main(String a[]) throws Exception{ CaptureDeviceInfo di = null; Player p = null; Vector deviceList = CaptureDeviceManager.getDeviceList(new AudioFormat("linear", 44100, 16, 2)); if (deviceList.size() > 0){ di = (CaptureDeviceInfo)deviceList.firstElement(); System.out.println((di.getLocator()).toExternalForm()); }else{ System.out.println("Exiting"); System.exit(-1); } try{ p = Manager.createPlayer(di.getLocator()); }catch (IOException e){ System.out.println(e); }catch (NoPlayerException e) { System.out.println(e); } System.out.println("Playing Started"); p.start(); } }

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  • what (clip) and DataLine.Info represents...?

    - by user528050
    I got this code from one of my friend. import java.io.*; import javax.sound.sampled.*; public class xx { public static void main(String args[]) { try { File f=new File("mm.wav"); AudioInputStream a=AudioSystem.getAudioInputStream(f); AudioFormat au=a.getFormat(); DataLine.Info di=new DataLine.Info(Clip.class,au); Clip c=(Clip)AudioSystem.getLine(di); c.open(a); c.start(); } catch(Exception e) { System.out.println("Exception caught "); } } } But i didn't understand what this line means Cilp c=(Clip)AudioSystem.getLine(di); what (clip) represents....? And my 2nd problem is what is the DataLine is it an interface and what is the meaning of this statement DataLine.Info....?

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  • isAlive problem..Help to understand how it works

    - by max
    I get this error: "non-static method isAlive() cannot be referenced from a static context" what's wrong with this code..please. I'd like to detect if the thread is alive... Any help in terms of code will be highly appreciated..thanks max class RecThread extends Thread { public void run() { recFile = new File("recorded_track.wav"); // Output file type AudioFileFormat.Type fileType = null; fileType = AudioFileFormat.Type.WAVE; // if rcOn =1 thread is alive int rcOn; try { // starts recording targetDataLine.open(audioFormat); targetDataLine.start(); AudioSystem.write(new AudioInputStream(targetDataLine), fileType, recFile); if (RecThread.isAlive() == true) { rcOn =1; } else { rcOn =0; } } catch (Exception e) { showException(e); } // update actions recAction.setEnabled(true); stopRecAction.setEnabled(false); } }

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  • How to initialize audio with Vala/SDL

    - by ioev
    I've been trying to figure this out for a few hours now. In order to start up the audio, I need to create an SDL.AudioSpec object and pass it to SDL.Audio.Open. The problem is, AudioSpec is a class with a private constructor, so when I try to create one I get: sdl.vala:18.25-18.43: error: `SDL.AudioSpec' does not have a default constructor AudioSpec audiospec = new SDL.AudioSpec(); ^^^^^^^^^^^^^^^^^^^ And if I try to just assign values to it's member vars like a struct (it's a struct in normal sdl) I get: sdl.vala:20.3-20.25: error: use of possibly unassigned local variable `audiospec' audiospec.freq = 22050; ^^^^^^^^^^^^^^^^^^^^^^^ I found the valac doc here: http://valadoc.org/sdl/SDL.AudioSpec.html But it isn't much help at all. The offending code block looks like this: // setup the audio configuration AudioSpec audiospec; AudioSpec specback; audiospec.freq = 22050; audiospec.format = SDL.AudioFormat.S16LSB; audiospec.channels = 2; audiospec.samples = 512; // try to initialize sound with these values if (SDL.Audio.open(audiospec, specback) < 0) { stdout.printf("ERROR! Check audio settings!\n"); return 1; } Any help would be greatly appreciated!

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  • How can i use a commandlinetool (ie. sox) via subprocess.Popen with mod_wsgi?

    - by marue
    I have a custom django filefield that makes use of sox, a commandline audiotool. This works pretty well as long as i use the django development server. But as soon as i switch to the production server, using apache2 and mod_wsgi, mod_wsgi catches every output to stdout. This makes it impossible to use the commandline tool to evaluate the file, for example use it to check if the uploaded file really is an audio file like this: filetype=subprocess.Popen([sox,'--i','-t','%s'%self.path], shell=False,\ stdout=subprocess.PIPE, stderr=subprocess.PIPE) (filetype,error)=filetype.communicate() if error: raise EnvironmentError((1,'AudioFile error while determining audioformat: %s'%error)) Is there a way to workaround for this? edit the error i get is "missing filename". I am using mod_wsgi 2.5, standard with ubuntu 8.04. edit2 What exactly happens, when i call subprocess.Popen from within django in mod_wsgi? Shouldn't subprocess stdin/stdout be independent from django stdin/stdout? In that case mod_wsgi should not affect programms called via subprocess... I'm really confused right now, because the file i am trying to access is a temporary file, created via a filenamevariable that i pass to the file creation and the subprocess command. That file is being written to /tmp, where the rights are 777, so it can't be a rights issue. And the error message is not "file does not exist", but "missing filename", which suggests i am not passing a filename as parameter to the commandlinetool.

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