List of usage examples for javax.sound.sampled AudioFormat AudioFormat
public AudioFormat(Encoding encoding, float sampleRate, int sampleSizeInBits, int channels, int frameSize, float frameRate, boolean bigEndian)
From source file:com.andrewkroh.cicso.rtp.AudioFileStreamer.java
private static AudioFormat toAlawFormat(AudioFormat source) { Preconditions.checkNotNull(source, "Source AudioFormat cannot be null."); return new AudioFormat(AudioFormat.Encoding.ALAW, source.getSampleRate(), 8, // sample size in bits source.getChannels(), 1, // frame size in bytes source.getFrameRate(), source.isBigEndian()); }
From source file:com.andrewkroh.cicso.rtp.AudioFileStreamer.java
private static AudioFormat toPcm16Format(AudioFormat source, boolean bigEndianOutput) { Preconditions.checkNotNull(source, "Source AudioFormat cannot be null."); return new AudioFormat(AudioFormat.Encoding.PCM_UNSIGNED, source.getSampleRate(), 16, // sample size in bits source.getChannels(), 2, // frame size in bytes source.getFrameRate(), bigEndianOutput); }
From source file:com.andrewkroh.cicso.rtp.AudioFileStreamer.java
private static AudioFormat toUlawFormat(AudioFormat source) { Preconditions.checkNotNull(source, "Source AudioFormat cannot be null."); return new AudioFormat(AudioFormat.Encoding.ULAW, source.getSampleRate(), 8, // sample size in bits source.getChannels(), 1, // frame size in bytes source.getFrameRate(), source.isBigEndian()); }
From source file:com.player.BasicMP3Player.java
/** * Inits a DateLine.<br>//from w w w .j a va 2 s . c o m * We check if the line supports Gain and Pan controls. From the AudioInputStream, i.e. from the * sound file, we fetch information about the format of the audio data. These information include * the sampling frequency, the number of channels and the size of the samples. There information * are needed to ask JavaSound for a suitable output line for this audio file. Furthermore, we * have to give JavaSound a hint about how big the internal buffer for the line should be. Here, * we say AudioSystem.NOT_SPECIFIED, signaling that we don't care about the exact size. JavaSound * will use some default value for the buffer size. */ private void createLine() throws LineUnavailableException { log.info("Create Line"); if (m_line == null) { AudioFormat sourceFormat = m_audioInputStream.getFormat(); log.info("Create Line : Source format : " + sourceFormat.toString()); AudioFormat targetFormat = new AudioFormat(AudioFormat.Encoding.PCM_SIGNED, sourceFormat.getSampleRate(), 16, sourceFormat.getChannels(), sourceFormat.getChannels() * 2, sourceFormat.getSampleRate(), false); log.info("Create Line : Target format: " + targetFormat); // Keep a reference on encoded stream to progress notification. m_encodedaudioInputStream = m_audioInputStream; try { // Get total length in bytes of the encoded stream. encodedLength = m_encodedaudioInputStream.available(); } catch (IOException e) { log.error("Cannot get m_encodedaudioInputStream.available()", e); } // Create decoded stream. m_audioInputStream = AudioSystem.getAudioInputStream(targetFormat, m_audioInputStream); AudioFormat audioFormat = m_audioInputStream.getFormat(); DataLine.Info info = new DataLine.Info(SourceDataLine.class, audioFormat, AudioSystem.NOT_SPECIFIED); m_line = (SourceDataLine) AudioSystem.getLine(info); /*-- Display supported controls --*/ Control[] c = m_line.getControls(); for (int p = 0; p < c.length; p++) { log.debug("Controls : " + c[p].toString()); } /*-- Is Gain Control supported ? --*/ if (m_line.isControlSupported(FloatControl.Type.MASTER_GAIN)) { m_gainControl = (FloatControl) m_line.getControl(FloatControl.Type.MASTER_GAIN); log.info("Master Gain Control : [" + m_gainControl.getMinimum() + "," + m_gainControl.getMaximum() + "] " + m_gainControl.getPrecision()); } /*-- Is Pan control supported ? --*/ if (m_line.isControlSupported(FloatControl.Type.PAN)) { m_panControl = (FloatControl) m_line.getControl(FloatControl.Type.PAN); log.info("Pan Control : [" + m_panControl.getMinimum() + "," + m_panControl.getMaximum() + "] " + m_panControl.getPrecision()); } } }
From source file:BasicPlayer.java
/** * Inits a DateLine.<br>// www. j ava2 s . c o m * * We check if the line supports Gain and Pan controls. * * From the AudioInputStream, i.e. from the sound file, we * fetch information about the format of the audio data. These * information include the sampling frequency, the number of * channels and the size of the samples. There information * are needed to ask JavaSound for a suitable output line * for this audio file. * Furthermore, we have to give JavaSound a hint about how * big the internal buffer for the line should be. Here, * we say AudioSystem.NOT_SPECIFIED, signaling that we don't * care about the exact size. JavaSound will use some default * value for the buffer size. */ protected void createLine() throws LineUnavailableException { log.info("Create Line"); if (m_line == null) { AudioFormat sourceFormat = m_audioInputStream.getFormat(); log.info("Create Line : Source format : " + sourceFormat.toString()); int nSampleSizeInBits = sourceFormat.getSampleSizeInBits(); if (nSampleSizeInBits <= 0) nSampleSizeInBits = 16; if ((sourceFormat.getEncoding() == AudioFormat.Encoding.ULAW) || (sourceFormat.getEncoding() == AudioFormat.Encoding.ALAW)) nSampleSizeInBits = 16; if (nSampleSizeInBits != 8) nSampleSizeInBits = 16; AudioFormat targetFormat = new AudioFormat(AudioFormat.Encoding.PCM_SIGNED, sourceFormat.getSampleRate(), nSampleSizeInBits, sourceFormat.getChannels(), sourceFormat.getChannels() * (nSampleSizeInBits / 8), sourceFormat.getSampleRate(), false); log.info("Create Line : Target format: " + targetFormat); // Keep a reference on encoded stream to progress notification. m_encodedaudioInputStream = m_audioInputStream; try { // Get total length in bytes of the encoded stream. encodedLength = m_encodedaudioInputStream.available(); } catch (IOException e) { log.error("Cannot get m_encodedaudioInputStream.available()", e); } // Create decoded stream. m_audioInputStream = AudioSystem.getAudioInputStream(targetFormat, m_audioInputStream); AudioFormat audioFormat = m_audioInputStream.getFormat(); DataLine.Info info = new DataLine.Info(SourceDataLine.class, audioFormat, AudioSystem.NOT_SPECIFIED); Mixer mixer = getMixer(m_mixerName); if (mixer != null) { log.info("Mixer : " + mixer.getMixerInfo().toString()); m_line = (SourceDataLine) mixer.getLine(info); } else { m_line = (SourceDataLine) AudioSystem.getLine(info); m_mixerName = null; } log.info("Line : " + m_line.toString()); log.debug("Line Info : " + m_line.getLineInfo().toString()); log.debug("Line AudioFormat: " + m_line.getFormat().toString()); } }
From source file:com.skratchdot.electribe.model.esx.impl.SampleImpl.java
/** * @param file//from w w w . j a v a 2s .c om * @throws EsxException */ protected SampleImpl(File file) throws EsxException { super(); init(); // Declare our streams and formats AudioFormat audioFormatEncoded; AudioFormat audioFormatDecoded; AudioInputStream audioInputStreamEncoded; AudioInputStream audioInputStreamDecoded; try { // Initialize our streams and formats audioInputStreamEncoded = AudioSystem.getAudioInputStream(file); audioFormatEncoded = audioInputStreamEncoded.getFormat(); audioFormatDecoded = new AudioFormat(AudioFormat.Encoding.PCM_SIGNED, audioFormatEncoded.getSampleRate(), 16, audioFormatEncoded.getChannels(), audioFormatEncoded.getChannels() * 2, audioFormatEncoded.getSampleRate(), true); audioInputStreamDecoded = AudioSystem.getAudioInputStream(audioFormatDecoded, audioInputStreamEncoded); // We have a decoded stereo audio stream // Now we need to get the stream info into a list we can manipulate byte[] audioData = new byte[4096]; int nBytesRead = 0; long nTotalBytesRead = 0; List<Byte> audioDataListChannel1 = new ArrayList<Byte>(); List<Byte> audioDataListChannel2 = new ArrayList<Byte>(); boolean isAudioDataStereo = false; // Set isAudioDataStereo if (audioFormatEncoded.getChannels() == 1) { isAudioDataStereo = false; } else if (audioFormatEncoded.getChannels() == 2) { isAudioDataStereo = true; } else { throw new EsxException("Sample has too many channels: " + file.getAbsolutePath()); } // Convert stream to list. This needs to be optimized. Converting // a byte at a time is probably too slow... while (nBytesRead >= 0) { nBytesRead = audioInputStreamDecoded.read(audioData, 0, audioData.length); // If we aren't at the end of the stream if (nBytesRead > 0) { for (int i = 0; i < nBytesRead; i++) { // MONO if (!isAudioDataStereo) { audioDataListChannel1.add(audioData[i]); audioDataListChannel2.add(audioData[i]); } // STEREO (LEFT) else if (nTotalBytesRead % 4 < 2) { audioDataListChannel1.add(audioData[i]); } // STEREO (RIGHT) else { audioDataListChannel2.add(audioData[i]); } // Update the total amount of bytes we've read nTotalBytesRead++; } } // Throw Exception if sample is too big if (nTotalBytesRead > EsxUtil.MAX_SAMPLE_MEM_IN_BYTES) { throw new EsxException("Sample is too big: " + file.getAbsolutePath()); } } // Set member variables int frameLength = audioDataListChannel1.size() / 2; this.setNumberOfSampleFrames(frameLength); this.setEnd(frameLength - 1); this.setLoopStart(frameLength - 1); this.setSampleRate((int) audioFormatEncoded.getSampleRate()); this.setAudioDataChannel1(EsxUtil.listToByteArray(audioDataListChannel1)); this.setAudioDataChannel2(EsxUtil.listToByteArray(audioDataListChannel2)); this.setStereoOriginal(isAudioDataStereo); // Set calculated Sample Tune (from Sample Rate) SampleTune newSampleTune = EsxFactory.eINSTANCE.createSampleTune(); float newFloat = newSampleTune.calculateSampleTuneFromSampleRate(this.getSampleRate()); newSampleTune.setValue(newFloat); this.setSampleTune(newSampleTune); // Set name String newSampleName = new String(); newSampleName = StringUtils.left(StringUtils.trim(file.getName()), 8); this.setName(newSampleName); // Attempt to set loopStart and End from .wav smpl chunk if (file.getAbsolutePath().toLowerCase().endsWith(".wav")) { try { RIFFWave riffWave = WavFactory.eINSTANCE.createRIFFWave(file); ChunkSampler chunkSampler = (ChunkSampler) riffWave .getFirstChunkByEClass(WavPackage.Literals.CHUNK_SAMPLER); if (chunkSampler != null && chunkSampler.getSampleLoops().size() > 0) { SampleLoop sampleLoop = chunkSampler.getSampleLoops().get(0); Long tempLoopStart = sampleLoop.getStart(); Long tempLoopEnd = sampleLoop.getEnd(); if (tempLoopStart < this.getEnd() && tempLoopStart >= 0) { this.setLoopStart(tempLoopStart.intValue()); } if (tempLoopEnd < this.getEnd() && tempLoopEnd > this.getLoopStart()) { this.setEnd(tempLoopEnd.intValue()); } } } catch (Exception e) { e.printStackTrace(); } } } catch (UnsupportedAudioFileException e) { e.printStackTrace(); throw new EsxException("Invalid audio file: " + file.getAbsolutePath()); } catch (IOException e) { e.printStackTrace(); throw new EsxException("Invalid audio file: " + file.getAbsolutePath()); } catch (Exception e) { e.printStackTrace(); throw new EsxException("Invalid audio file: " + file.getAbsolutePath()); } }
From source file:sx.blah.discord.api.internal.DiscordUtils.java
/** * Converts an {@link AudioInputStream} to 48000Hz 16 bit stereo signed Big Endian PCM format. * * @param stream The original stream./*from ww w .j av a 2 s .co m*/ * @return The PCM encoded stream. */ public static AudioInputStream getPCMStream(AudioInputStream stream) { AudioFormat baseFormat = stream.getFormat(); //Converts first to PCM data. If the data is already PCM data, this will not change anything. AudioFormat toPCM = new AudioFormat(AudioFormat.Encoding.PCM_SIGNED, baseFormat.getSampleRate(), //AudioConnection.OPUS_SAMPLE_RATE, baseFormat.getSampleSizeInBits() != -1 ? baseFormat.getSampleSizeInBits() : 16, baseFormat.getChannels(), //If we are given a frame size, use it. Otherwise, assume 16 bits (2 8bit shorts) per channel. baseFormat.getFrameSize() != -1 ? baseFormat.getFrameSize() : 2 * baseFormat.getChannels(), baseFormat.getFrameRate() != -1 ? baseFormat.getFrameRate() : baseFormat.getSampleRate(), baseFormat.isBigEndian()); AudioInputStream pcmStream = AudioSystem.getAudioInputStream(toPCM, stream); //Then resamples to a sample rate of 48000hz and ensures that data is Big Endian. AudioFormat audioFormat = new AudioFormat(toPCM.getEncoding(), OpusUtil.OPUS_SAMPLE_RATE, toPCM.getSampleSizeInBits(), toPCM.getChannels(), toPCM.getFrameSize(), toPCM.getFrameRate(), true); return AudioSystem.getAudioInputStream(audioFormat, pcmStream); }