Java AudioFormat.getSampleRate方法代碼示例

import javax.sound.sampled.AudioFormat; //導入方法依賴的package包/類
public static AudioInputStream getPCMConvertedAudioInputStream(AudioInputStream ais) {
    // we can't open the device for non-PCM playback, so we have
    // convert any other encodings to PCM here (at least we try!)
    AudioFormat af = ais.getFormat();

    if( (!af.getEncoding().equals(AudioFormat.Encoding.PCM_SIGNED)) &&
        (!af.getEncoding().equals(AudioFormat.Encoding.PCM_UNSIGNED))) {

        try {
            AudioFormat newFormat =
                new AudioFormat( AudioFormat.Encoding.PCM_SIGNED,
                                 af.getSampleRate(),
                                 16,
                                 af.getChannels(),
                                 af.getChannels() * 2,
                                 af.getSampleRate(),
                                 Platform.isBigEndian());
            ais = AudioSystem.getAudioInputStream(newFormat, ais);
        } catch (Exception e) {
            if (Printer.err) e.printStackTrace();
            ais = null;
        }
    }

    return ais;
}
 

import javax.sound.sampled.AudioFormat; //導入方法依賴的package包/類
public AudioInputStream getAudioInputStream(Encoding targetEncoding,
        AudioInputStream sourceStream) {
    if (sourceStream.getFormat().getEncoding().equals(targetEncoding))
        return sourceStream;
    AudioFormat format = sourceStream.getFormat();
    int channels = format.getChannels();
    Encoding encoding = targetEncoding;
    float samplerate = format.getSampleRate();
    int bits = format.getSampleSizeInBits();
    boolean bigendian = format.isBigEndian();
    if (targetEncoding.equals(Encoding.PCM_FLOAT))
        bits = 32;
    AudioFormat targetFormat = new AudioFormat(encoding, samplerate, bits,
            channels, channels * bits / 8, samplerate, bigendian);
    return getAudioInputStream(targetFormat, sourceStream);
}
 

import javax.sound.sampled.AudioFormat; //導入方法依賴的package包/類
protected static AudioFormat getSignOrEndianChangedFormat(AudioFormat format) {
    boolean isSigned = format.getEncoding().equals(AudioFormat.Encoding.PCM_SIGNED);
    boolean isUnsigned = format.getEncoding().equals(AudioFormat.Encoding.PCM_UNSIGNED);
    if (format.getSampleSizeInBits() > 8 && isSigned) {
        // if this is PCM_SIGNED and 16-bit or higher, then try with endian-ness magic
        return new AudioFormat(format.getEncoding(),
                               format.getSampleRate(), format.getSampleSizeInBits(), format.getChannels(),
                               format.getFrameSize(), format.getFrameRate(), !format.isBigEndian());
    }
    else if (format.getSampleSizeInBits() == 8 && (isSigned || isUnsigned)) {
        // if this is PCM and 8-bit, then try with signed-ness magic
        return new AudioFormat(isSigned?AudioFormat.Encoding.PCM_UNSIGNED:AudioFormat.Encoding.PCM_SIGNED,
                               format.getSampleRate(), format.getSampleSizeInBits(), format.getChannels(),
                               format.getFrameSize(), format.getFrameRate(), format.isBigEndian());
    }
    return null;
}
 

import javax.sound.sampled.AudioFormat; //導入方法依賴的package包/類
@Override
public AudioInputStream getAudioInputStream(AudioFormat.Encoding targetEncoding, AudioInputStream sourceStream) {

    if( isConversionSupported(targetEncoding, sourceStream.getFormat()) ) {

        AudioFormat sourceFormat = sourceStream.getFormat();
        AudioFormat targetFormat = new AudioFormat( targetEncoding,
                                                    sourceFormat.getSampleRate(),
                                                    sourceFormat.getSampleSizeInBits(),
                                                    sourceFormat.getChannels(),
                                                    sourceFormat.getFrameSize(),
                                                    sourceFormat.getFrameRate(),
                                                    sourceFormat.isBigEndian() );

        return getConvertedStream(targetFormat, sourceStream);

    } else {
        throw new IllegalArgumentException("Unsupported conversion: " + sourceStream.getFormat().toString() + " to " + targetEncoding.toString() );
    }

}
 

import javax.sound.sampled.AudioFormat; //導入方法依賴的package包/類
public AudioFloatInputStreamResampler(AudioFloatInputStream ais,
        AudioFormat format) {
    this.ais = ais;
    AudioFormat sourceFormat = ais.getFormat();
    targetFormat = new AudioFormat(sourceFormat.getEncoding(), format
            .getSampleRate(), sourceFormat.getSampleSizeInBits(),
            sourceFormat.getChannels(), sourceFormat.getFrameSize(),
            format.getSampleRate(), sourceFormat.isBigEndian());
    nrofchannels = targetFormat.getChannels();
    Object interpolation = format.getProperty("interpolation");
    if (interpolation != null && (interpolation instanceof String)) {
        String resamplerType = (String) interpolation;
        if (resamplerType.equalsIgnoreCase("point"))
            this.resampler = new SoftPointResampler();
        if (resamplerType.equalsIgnoreCase("linear"))
            this.resampler = new SoftLinearResampler2();
        if (resamplerType.equalsIgnoreCase("linear1"))
            this.resampler = new SoftLinearResampler();
        if (resamplerType.equalsIgnoreCase("linear2"))
            this.resampler = new SoftLinearResampler2();
        if (resamplerType.equalsIgnoreCase("cubic"))
            this.resampler = new SoftCubicResampler();
        if (resamplerType.equalsIgnoreCase("lanczos"))
            this.resampler = new SoftLanczosResampler();
        if (resamplerType.equalsIgnoreCase("sinc"))
            this.resampler = new SoftSincResampler();
    }
    if (resampler == null)
        resampler = new SoftLinearResampler2(); // new
    // SoftLinearResampler2();
    pitch[0] = sourceFormat.getSampleRate() / format.getSampleRate();
    pad = resampler.getPadding();
    pad2 = pad * 2;
    ibuffer = new float[nrofchannels][buffer_len + pad2];
    ibuffer2 = new float[nrofchannels * buffer_len];
    ibuffer_index = buffer_len + pad;
    ibuffer_len = buffer_len;
}
 

import javax.sound.sampled.AudioFormat; //導入方法依賴的package包/類
static boolean isFullySpecifiedPCMFormat(AudioFormat format) {
    if (!format.getEncoding().equals(AudioFormat.Encoding.PCM_SIGNED)
        && !format.getEncoding().equals(AudioFormat.Encoding.PCM_UNSIGNED)) {
        return false;
    }
    if ((format.getFrameRate() <= 0)
        || (format.getSampleRate() <= 0)
        || (format.getSampleSizeInBits() <= 0)
        || (format.getFrameSize() <= 0)
        || (format.getChannels() <= 0)) {
        return false;
    }
    return true;
}
 

import javax.sound.sampled.AudioFormat; //導入方法依賴的package包/類
public static AudioFormat getOtherChannels(AudioFormat format, int newChannels) {
    int newFrameSize;
    if (newChannels <= 0 || format.getChannels() <= 0 || format.getFrameSize() <= 0) {
        newFrameSize = -1;
    } else {
        newFrameSize = format.getFrameSize() / format.getChannels() * newChannels;
    }
    return new AudioFormat(format.getEncoding(),
                           format.getSampleRate(),
                           format.getSampleSizeInBits(),
                           newChannels,
                           newFrameSize,
                           format.getFrameRate(),
                           format.isBigEndian());
}
 

import javax.sound.sampled.AudioFormat; //導入方法依賴的package包/類
public AudioSample getAudioSample(float[] left, float[] right, AudioFormat format, int bufferSize)
{
  FloatSampleBuffer sample = new FloatSampleBuffer(2, left.length, format.getSampleRate());
  System.arraycopy(left, 0, sample.getChannel(0), 0, left.length);
  System.arraycopy(right, 0, sample.getChannel(1), 0, right.length);
  return getAudioSampleImp(sample, format, bufferSize);
}
 

import javax.sound.sampled.AudioFormat; //導入方法依賴的package包/類
AudioFloatInputStreamResampler(AudioFloatInputStream ais,
        AudioFormat format) {
    this.ais = ais;
    AudioFormat sourceFormat = ais.getFormat();
    targetFormat = new AudioFormat(sourceFormat.getEncoding(), format
            .getSampleRate(), sourceFormat.getSampleSizeInBits(),
            sourceFormat.getChannels(), sourceFormat.getFrameSize(),
            format.getSampleRate(), sourceFormat.isBigEndian());
    nrofchannels = targetFormat.getChannels();
    Object interpolation = format.getProperty("interpolation");
    if (interpolation != null && (interpolation instanceof String)) {
        String resamplerType = (String) interpolation;
        if (resamplerType.equalsIgnoreCase("point"))
            this.resampler = new SoftPointResampler();
        if (resamplerType.equalsIgnoreCase("linear"))
            this.resampler = new SoftLinearResampler2();
        if (resamplerType.equalsIgnoreCase("linear1"))
            this.resampler = new SoftLinearResampler();
        if (resamplerType.equalsIgnoreCase("linear2"))
            this.resampler = new SoftLinearResampler2();
        if (resamplerType.equalsIgnoreCase("cubic"))
            this.resampler = new SoftCubicResampler();
        if (resamplerType.equalsIgnoreCase("lanczos"))
            this.resampler = new SoftLanczosResampler();
        if (resamplerType.equalsIgnoreCase("sinc"))
            this.resampler = new SoftSincResampler();
    }
    if (resampler == null)
        resampler = new SoftLinearResampler2(); // new
                                                // SoftLinearResampler2();
    pitch[0] = sourceFormat.getSampleRate() / format.getSampleRate();
    pad = resampler.getPadding();
    pad2 = pad * 2;
    ibuffer = new float[nrofchannels][buffer_len + pad2];
    ibuffer2 = new float[nrofchannels * buffer_len];
    ibuffer_index = buffer_len + pad;
    ibuffer_len = buffer_len;
}
 

import javax.sound.sampled.AudioFormat; //導入方法依賴的package包/類
AudioFloatInputStreamChannelMixer(AudioFloatInputStream ais,
        int targetChannels) {
    this.sourceChannels = ais.getFormat().getChannels();
    this.targetChannels = targetChannels;
    this.ais = ais;
    AudioFormat format = ais.getFormat();
    targetFormat = new AudioFormat(format.getEncoding(), format
            .getSampleRate(), format.getSampleSizeInBits(),
            targetChannels, (format.getFrameSize() / sourceChannels)
                    * targetChannels, format.getFrameRate(), format
                    .isBigEndian());
}
 

import javax.sound.sampled.AudioFormat; //導入方法依賴的package包/類
public static SF2Sample newSimpleFFTSample_dist(SF2Soundbank sf2,
        String name, double[] data, double base, double preamp) {

    int fftsize = data.length / 2;
    AudioFormat format = new AudioFormat(44100, 16, 1, true, false);
    double basefreq = (base / fftsize) * format.getSampleRate() * 0.5;

    randomPhase(data);
    ifft(data);
    data = realPart(data);

    for (int i = 0; i < data.length; i++) {
        data[i] = (1 - Math.exp(-Math.abs(data[i] * preamp)))
                * Math.signum(data[i]);
    }

    normalize(data, 0.9);
    float[] fdata = toFloat(data);
    fdata = loopExtend(fdata, fdata.length + 512);
    fadeUp(fdata, 80);
    byte[] bdata = toBytes(fdata, format);

    /*
     * Create SoundFont2 sample.
     */
    SF2Sample sample = new SF2Sample(sf2);
    sample.setName(name);
    sample.setData(bdata);
    sample.setStartLoop(256);
    sample.setEndLoop(fftsize + 256);
    sample.setSampleRate((long) format.getSampleRate());
    double orgnote = (69 + 12)
            + (12 * Math.log(basefreq / 440.0) / Math.log(2));
    sample.setOriginalPitch((int) orgnote);
    sample.setPitchCorrection((byte) (-(orgnote - (int) orgnote) * 100.0));
    sf2.addResource(sample);

    return sample;
}
 

import javax.sound.sampled.AudioFormat; //導入方法依賴的package包/類
static void isFullySpecifiedAudioFormat(AudioFormat format) {
    if (!format.getEncoding().equals(AudioFormat.Encoding.PCM_SIGNED)
        && !format.getEncoding().equals(AudioFormat.Encoding.PCM_UNSIGNED)
        && !format.getEncoding().equals(AudioFormat.Encoding.ULAW)
        && !format.getEncoding().equals(AudioFormat.Encoding.ALAW)) {
        // we don't know how to verify possibly non-linear encodings
        return;
    }
    if (format.getFrameRate() <= 0) {
        throw new IllegalArgumentException("invalid frame rate: "
                                           +((format.getFrameRate()==-1)?
                                             "NOT_SPECIFIED":String.valueOf(format.getFrameRate())));
    }
    if (format.getSampleRate() <= 0) {
        throw new IllegalArgumentException("invalid sample rate: "
                                           +((format.getSampleRate()==-1)?
                                             "NOT_SPECIFIED":String.valueOf(format.getSampleRate())));
    }
    if (format.getSampleSizeInBits() <= 0) {
        throw new IllegalArgumentException("invalid sample size in bits: "
                                           +((format.getSampleSizeInBits()==-1)?
                                             "NOT_SPECIFIED":String.valueOf(format.getSampleSizeInBits())));
    }
    if (format.getFrameSize() <= 0) {
        throw new IllegalArgumentException("invalid frame size: "
                                           +((format.getFrameSize()==-1)?
                                             "NOT_SPECIFIED":String.valueOf(format.getFrameSize())));
    }
    if (format.getChannels() <= 0) {
        throw new IllegalArgumentException("invalid number of channels: "
                                           +((format.getChannels()==-1)?
                                             "NOT_SPECIFIED":String.valueOf(format.getChannels())));
    }
}
 

import javax.sound.sampled.AudioFormat; //導入方法依賴的package包/類
/**
 * Returns the AudioFileFormat describing the file that will be written from this AudioInputStream.
 * Throws IllegalArgumentException if not supported.
 */
private AudioFileFormat getAudioFileFormat(Type type, AudioInputStream stream) {
    if (!isFileTypeSupported(type, stream)) {
        throw new IllegalArgumentException("File type " + type + " not supported.");
    }

    AudioFormat streamFormat = stream.getFormat();
    AudioFormat.Encoding encoding = streamFormat.getEncoding();

    if (AudioFormat.Encoding.PCM_UNSIGNED.equals(encoding)) {
        encoding = AudioFormat.Encoding.PCM_SIGNED;
    }

    // We always write big endian au files, this is by far the standard
    AudioFormat format = new AudioFormat(encoding,
                                         streamFormat.getSampleRate(),
                                         streamFormat.getSampleSizeInBits(),
                                         streamFormat.getChannels(),
                                         streamFormat.getFrameSize(),
                                         streamFormat.getFrameRate(), true);

    int fileSize;
    if (stream.getFrameLength() != AudioSystem.NOT_SPECIFIED) {
        fileSize = (int)stream.getFrameLength()*streamFormat.getFrameSize() + AuFileFormat.AU_HEADERSIZE;
    } else {
        fileSize = AudioSystem.NOT_SPECIFIED;
    }

    return new AuFileFormat(Type.AU, fileSize, format,
                            (int) stream.getFrameLength());
}
 

import javax.sound.sampled.AudioFormat; //導入方法依賴的package包/類
private AudioInputStream toLittleEndian(AudioInputStream ais) {
    AudioFormat format = ais.getFormat();
    AudioFormat targetFormat = new AudioFormat(format.getEncoding(), format
            .getSampleRate(), format.getSampleSizeInBits(), format
            .getChannels(), format.getFrameSize(), format.getFrameRate(),
            false);
    return AudioSystem.getAudioInputStream(targetFormat, ais);
}
 

import javax.sound.sampled.AudioFormat; //導入方法依賴的package包/類
public static SF2Sample newSimpleFFTSample(SF2Soundbank sf2, String name,
        double[] data, double base, int fadeuptime) {

    int fftsize = data.length / 2;
    AudioFormat format = new AudioFormat(44100, 16, 1, true, false);
    double basefreq = (base / fftsize) * format.getSampleRate() * 0.5;

    randomPhase(data);
    ifft(data);
    data = realPart(data);
    normalize(data, 0.9);
    float[] fdata = toFloat(data);
    fdata = loopExtend(fdata, fdata.length + 512);
    fadeUp(fdata, fadeuptime);
    byte[] bdata = toBytes(fdata, format);

    /*
     * Create SoundFont2 sample.
     */
    SF2Sample sample = new SF2Sample(sf2);
    sample.setName(name);
    sample.setData(bdata);
    sample.setStartLoop(256);
    sample.setEndLoop(fftsize + 256);
    sample.setSampleRate((long) format.getSampleRate());
    double orgnote = (69 + 12)
            + (12 * Math.log(basefreq / 440.0) / Math.log(2));
    sample.setOriginalPitch((int) orgnote);
    sample.setPitchCorrection((byte) (-(orgnote - (int) orgnote) * 100.0));
    sf2.addResource(sample);

    return sample;
}