javax.sound.sampled.AudioFormat#getSampleSizeInBits ( )源码实例Demo

下面列出了javax.sound.sampled.AudioFormat#getSampleSizeInBits ( ) 实例代码,或者点击链接到github查看源代码,也可以在右侧发表评论。

源代码1 项目: openjdk-jdk8u   文件: AudioFloatConverter.java
AudioFloatLSBFilter(AudioFloatConverter converter, AudioFormat format) {
    int bits = format.getSampleSizeInBits();
    boolean bigEndian = format.isBigEndian();
    this.converter = converter;
    stepsize = (bits + 7) / 8;
    offset = bigEndian ? (stepsize - 1) : 0;
    int lsb_bits = bits % 8;
    if (lsb_bits == 0)
        mask = (byte) 0x00;
    else if (lsb_bits == 1)
        mask = (byte) 0x80;
    else if (lsb_bits == 2)
        mask = (byte) 0xC0;
    else if (lsb_bits == 3)
        mask = (byte) 0xE0;
    else if (lsb_bits == 4)
        mask = (byte) 0xF0;
    else if (lsb_bits == 5)
        mask = (byte) 0xF8;
    else if (lsb_bits == 6)
        mask = (byte) 0xFC;
    else if (lsb_bits == 7)
        mask = (byte) 0xFE;
    else
        mask = (byte) 0xFF;
}
 
源代码2 项目: jdk8u60   文件: AudioFloatFormatConverter.java
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);
}
 
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);
}
 
源代码4 项目: jdk8u60   文件: Toolkit.java
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;
}
 
源代码5 项目: openjdk-8   文件: AudioFloatFormatConverter.java
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;
}
 
源代码6 项目: openjdk-jdk9   文件: PCMtoPCMCodec.java
@Override
public AudioFormat.Encoding[] getTargetEncodings(AudioFormat sourceFormat) {

    final int sampleSize = sourceFormat.getSampleSizeInBits();
    AudioFormat.Encoding encoding = sourceFormat.getEncoding();
    if (sampleSize == 8) {
        if (encoding.equals(AudioFormat.Encoding.PCM_SIGNED)) {
            return new AudioFormat.Encoding[]{
                    AudioFormat.Encoding.PCM_UNSIGNED
            };
        }
        if (encoding.equals(AudioFormat.Encoding.PCM_UNSIGNED)) {
            return new AudioFormat.Encoding[]{
                    AudioFormat.Encoding.PCM_SIGNED
            };
        }
    } else if (sampleSize == 16) {
        if (encoding.equals(AudioFormat.Encoding.PCM_SIGNED)
                || encoding.equals(AudioFormat.Encoding.PCM_UNSIGNED)) {
            return new AudioFormat.Encoding[]{
                    AudioFormat.Encoding.PCM_UNSIGNED,
                    AudioFormat.Encoding.PCM_SIGNED
            };
        }
    }
    return new AudioFormat.Encoding[0];
}
 
源代码7 项目: hottub   文件: AuFileFormat.java
AuFileFormat(AudioFileFormat.Type type, int lengthInBytes, AudioFormat format, int lengthInFrames) {

        super(type,lengthInBytes,format,lengthInFrames);

        AudioFormat.Encoding encoding = format.getEncoding();

        auType = -1;

        if( AudioFormat.Encoding.ALAW.equals(encoding) ) {
            if( format.getSampleSizeInBits()==8 ) {
                auType = AU_ALAW_8;
            }
        } else if( AudioFormat.Encoding.ULAW.equals(encoding) ) {
            if( format.getSampleSizeInBits()==8 ) {
                auType = AU_ULAW_8;
            }
        } else if( AudioFormat.Encoding.PCM_SIGNED.equals(encoding) ) {
            if( format.getSampleSizeInBits()==8 ) {
                auType = AU_LINEAR_8;
            } else if( format.getSampleSizeInBits()==16 ) {
                auType = AU_LINEAR_16;
            } else if( format.getSampleSizeInBits()==24 ) {
                auType = AU_LINEAR_24;
            } else if( format.getSampleSizeInBits()==32 ) {
                auType = AU_LINEAR_32;
            }
        }

    }
 
源代码8 项目: jdk8u-jdk   文件: AuFileFormat.java
AuFileFormat(AudioFileFormat.Type type, int lengthInBytes, AudioFormat format, int lengthInFrames) {

        super(type,lengthInBytes,format,lengthInFrames);

        AudioFormat.Encoding encoding = format.getEncoding();

        auType = -1;

        if( AudioFormat.Encoding.ALAW.equals(encoding) ) {
            if( format.getSampleSizeInBits()==8 ) {
                auType = AU_ALAW_8;
            }
        } else if( AudioFormat.Encoding.ULAW.equals(encoding) ) {
            if( format.getSampleSizeInBits()==8 ) {
                auType = AU_ULAW_8;
            }
        } else if( AudioFormat.Encoding.PCM_SIGNED.equals(encoding) ) {
            if( format.getSampleSizeInBits()==8 ) {
                auType = AU_LINEAR_8;
            } else if( format.getSampleSizeInBits()==16 ) {
                auType = AU_LINEAR_16;
            } else if( format.getSampleSizeInBits()==24 ) {
                auType = AU_LINEAR_24;
            } else if( format.getSampleSizeInBits()==32 ) {
                auType = AU_LINEAR_32;
            }
        }

    }
 
源代码9 项目: openjdk-jdk8u   文件: AudioFloatFormatConverter.java
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;
}
 
源代码10 项目: jdk8u-dev-jdk   文件: AudioFloatFormatConverter.java
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;
}
 
源代码11 项目: hottub   文件: WaveFloatFileWriter.java
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);
}
 
源代码12 项目: Bytecoder   文件: AuFileWriter.java
/**
 * 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());
}
 
源代码13 项目: openjdk-jdk8u   文件: SoftMixingDataLine.java
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;
}
 
源代码14 项目: jdk8u-jdk   文件: AuFileFormat.java
AuFileFormat(AudioFileFormat.Type type, int lengthInBytes, AudioFormat format, int lengthInFrames) {

        super(type,lengthInBytes,format,lengthInFrames);

        AudioFormat.Encoding encoding = format.getEncoding();

        auType = -1;

        if( AudioFormat.Encoding.ALAW.equals(encoding) ) {
            if( format.getSampleSizeInBits()==8 ) {
                auType = AU_ALAW_8;
            }
        } else if( AudioFormat.Encoding.ULAW.equals(encoding) ) {
            if( format.getSampleSizeInBits()==8 ) {
                auType = AU_ULAW_8;
            }
        } else if( AudioFormat.Encoding.PCM_SIGNED.equals(encoding) ) {
            if( format.getSampleSizeInBits()==8 ) {
                auType = AU_LINEAR_8;
            } else if( format.getSampleSizeInBits()==16 ) {
                auType = AU_LINEAR_16;
            } else if( format.getSampleSizeInBits()==24 ) {
                auType = AU_LINEAR_24;
            } else if( format.getSampleSizeInBits()==32 ) {
                auType = AU_LINEAR_32;
            }
        }

    }
 
源代码15 项目: openjdk-jdk9   文件: BothEndiansAndSigns.java
public static AudioFormat getOtherEndianOrSign(AudioFormat format) {
    AudioFormat.Encoding newEnc = null;
    boolean newEndian = format.isBigEndian();
    boolean isSigned = format.getEncoding().equals(AudioFormat.Encoding.PCM_SIGNED);
    boolean isUnsigned = format.getEncoding().equals(AudioFormat.Encoding.PCM_UNSIGNED);
    if ((isSigned || isUnsigned) && format.getSampleSizeInBits() > 0) {
        if (format.getSampleSizeInBits() == 8) {
            // return the other signed'ness
            if (isSigned) {
                newEnc = AudioFormat.Encoding.PCM_UNSIGNED;
            } else {
                newEnc = AudioFormat.Encoding.PCM_SIGNED;
            }
        } else {
            newEnc = format.getEncoding();
            newEndian = !newEndian;
        }
        if (newEnc != null) {
            return new AudioFormat(newEnc, format.getSampleRate(),
                                   format.getSampleSizeInBits(),
                                   format.getChannels(),
                                   format.getFrameSize(),
                                   format.getFrameRate(),
                                   newEndian);
        }
    }
    return null;
}
 
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());
}
 
源代码17 项目: openjdk-jdk8u-backup   文件: AudioFloatConverter.java
public static AudioFloatConverter getConverter(AudioFormat format) {
    AudioFloatConverter conv = null;
    if (format.getFrameSize() == 0)
        return null;
    if (format.getFrameSize() !=
            ((format.getSampleSizeInBits() + 7) / 8) * format.getChannels()) {
        return null;
    }
    if (format.getEncoding().equals(Encoding.PCM_SIGNED)) {
        if (format.isBigEndian()) {
            if (format.getSampleSizeInBits() <= 8) {
                conv = new AudioFloatConversion8S();
            } else if (format.getSampleSizeInBits() > 8 &&
                  format.getSampleSizeInBits() <= 16) {
                conv = new AudioFloatConversion16SB();
            } else if (format.getSampleSizeInBits() > 16 &&
                  format.getSampleSizeInBits() <= 24) {
                conv = new AudioFloatConversion24SB();
            } else if (format.getSampleSizeInBits() > 24 &&
                  format.getSampleSizeInBits() <= 32) {
                conv = new AudioFloatConversion32SB();
            } else if (format.getSampleSizeInBits() > 32) {
                conv = new AudioFloatConversion32xSB(((format
                        .getSampleSizeInBits() + 7) / 8) - 4);
            }
        } else {
            if (format.getSampleSizeInBits() <= 8) {
                conv = new AudioFloatConversion8S();
            } else if (format.getSampleSizeInBits() > 8 &&
                     format.getSampleSizeInBits() <= 16) {
                conv = new AudioFloatConversion16SL();
            } else if (format.getSampleSizeInBits() > 16 &&
                     format.getSampleSizeInBits() <= 24) {
                conv = new AudioFloatConversion24SL();
            } else if (format.getSampleSizeInBits() > 24 &&
                     format.getSampleSizeInBits() <= 32) {
                conv = new AudioFloatConversion32SL();
            } else if (format.getSampleSizeInBits() > 32) {
                conv = new AudioFloatConversion32xSL(((format
                        .getSampleSizeInBits() + 7) / 8) - 4);
            }
        }
    } else if (format.getEncoding().equals(Encoding.PCM_UNSIGNED)) {
        if (format.isBigEndian()) {
            if (format.getSampleSizeInBits() <= 8) {
                conv = new AudioFloatConversion8U();
            } else if (format.getSampleSizeInBits() > 8 &&
                    format.getSampleSizeInBits() <= 16) {
                conv = new AudioFloatConversion16UB();
            } else if (format.getSampleSizeInBits() > 16 &&
                    format.getSampleSizeInBits() <= 24) {
                conv = new AudioFloatConversion24UB();
            } else if (format.getSampleSizeInBits() > 24 &&
                    format.getSampleSizeInBits() <= 32) {
                conv = new AudioFloatConversion32UB();
            } else if (format.getSampleSizeInBits() > 32) {
                conv = new AudioFloatConversion32xUB(((
                        format.getSampleSizeInBits() + 7) / 8) - 4);
            }
        } else {
            if (format.getSampleSizeInBits() <= 8) {
                conv = new AudioFloatConversion8U();
            } else if (format.getSampleSizeInBits() > 8 &&
                    format.getSampleSizeInBits() <= 16) {
                conv = new AudioFloatConversion16UL();
            } else if (format.getSampleSizeInBits() > 16 &&
                    format.getSampleSizeInBits() <= 24) {
                conv = new AudioFloatConversion24UL();
            } else if (format.getSampleSizeInBits() > 24 &&
                    format.getSampleSizeInBits() <= 32) {
                conv = new AudioFloatConversion32UL();
            } else if (format.getSampleSizeInBits() > 32) {
                conv = new AudioFloatConversion32xUL(((
                        format.getSampleSizeInBits() + 7) / 8) - 4);
            }
        }
    } else if (format.getEncoding().equals(Encoding.PCM_FLOAT)) {
        if (format.getSampleSizeInBits() == 32) {
            if (format.isBigEndian())
                conv = new AudioFloatConversion32B();
            else
                conv = new AudioFloatConversion32L();
        } else if (format.getSampleSizeInBits() == 64) {
            if (format.isBigEndian())
                conv = new AudioFloatConversion64B();
            else
                conv = new AudioFloatConversion64L();
        }

    }

    if ((format.getEncoding().equals(Encoding.PCM_SIGNED) ||
            format.getEncoding().equals(Encoding.PCM_UNSIGNED)) &&
            (format.getSampleSizeInBits() % 8 != 0)) {
        conv = new AudioFloatLSBFilter(conv, format);
    }

    if (conv != null)
        conv.format = format;
    return conv;
}
 
源代码18 项目: opsu-dance   文件: SoundController.java
/**
 * Loads and returns a Clip from an audio input stream.
 * @param ref the resource name
 * @param audioIn the audio input stream
 * @param isMP3 true if MP3, false if WAV
 * @return the loaded and opened clip
 */
private static MultiClip loadClip(String ref, AudioInputStream audioIn, boolean isMP3)
		throws IOException, LineUnavailableException {
	AudioFormat format = audioIn.getFormat();
	if (isMP3) {
		AudioFormat decodedFormat = new AudioFormat(
				AudioFormat.Encoding.PCM_SIGNED, format.getSampleRate(), 16,
				format.getChannels(), format.getChannels() * 2, format.getSampleRate(), false);
		AudioInputStream decodedAudioIn = AudioSystem.getAudioInputStream(decodedFormat, audioIn);
		format = decodedFormat;
		audioIn = decodedAudioIn;
	}
	DataLine.Info info = new DataLine.Info(Clip.class, format);
	if (AudioSystem.isLineSupported(info))
		return new MultiClip(ref, audioIn);

	// try to find closest matching line
	Clip clip = AudioSystem.getClip();
	AudioFormat[] formats = ((DataLine.Info) clip.getLineInfo()).getFormats();
	int bestIndex = -1;
	float bestScore = 0;
	float sampleRate = format.getSampleRate();
	if (sampleRate < 0)
		sampleRate = clip.getFormat().getSampleRate();
	float oldSampleRate = sampleRate;
	while (true) {
		for (int i = 0; i < formats.length; i++) {
			AudioFormat curFormat = formats[i];
			AudioFormat newFormat = new AudioFormat(
					sampleRate, curFormat.getSampleSizeInBits(),
					curFormat.getChannels(), true, curFormat.isBigEndian());
			formats[i] = newFormat;
			DataLine.Info newLine = new DataLine.Info(Clip.class, newFormat);
			if (AudioSystem.isLineSupported(newLine) &&
			    AudioSystem.isConversionSupported(newFormat, format)) {
				float score = 1
						+ (newFormat.getSampleRate() == sampleRate ? 5 : 0)
						+ (newFormat.getSampleSizeInBits() == format.getSampleSizeInBits() ? 5 : 0)
						+ (newFormat.getChannels() == format.getChannels() ? 5 : 0)
						+ (newFormat.isBigEndian() == format.isBigEndian() ? 1 : 0)
						+ newFormat.getSampleRate() / 11025
						+ newFormat.getChannels()
						+ newFormat.getSampleSizeInBits() / 8;
				if (score > bestScore) {
					bestIndex = i;
					bestScore = score;
				}
			}
		}
		if (bestIndex < 0) {
			if (oldSampleRate < 44100) {
				if (sampleRate > 44100)
					break;
				sampleRate *= 2;
			} else {
				if (sampleRate < 44100)
					break;
				sampleRate /= 2;
			}
		} else
			break;
	}
	if (bestIndex >= 0)
		return new MultiClip(ref, AudioSystem.getAudioInputStream(formats[bestIndex], audioIn));

	// still couldn't find anything, try the default clip format
	return new MultiClip(ref, AudioSystem.getAudioInputStream(clip.getFormat(), audioIn));
}
 
源代码19 项目: haxademic   文件: NormalizeMonoWav.java
/**
 * If this is a mono sound, obtains the single sample contained
 * within this frame, else obtains the first (left) sample
 * contained in the specified frame.
 *
 * @param frameNum the index of the frame to access
 * @return an integer representation of the bytes contained within
 * the specified frame
 * @throws SoundException if the frame number is invalid.
 */
public int getSampleValue(int frameNum) throws SoundException {
	//Before we get started, lets make sure that frame exists
	if (frameNum >= getAudioFileFormat().getFrameLength()) {
		printError("You are trying to access the sample at index: "
				+ (frameNum) + ", but the last valid index is at " +
				(getAudioFileFormat().getFrameLength() - 1));
	} else if (frameNum < 0) {
		printError("You asked for the sample at index: " + (frameNum) +
				".  This number is less than zero.  Please try" +
				"again using an index in the range [0," +
				(getAudioFileFormat().getFrameLength() - 1) + "]");
	}

	AudioFormat format = getAudioFileFormat().getFormat();
	int sampleSizeInBits = format.getSampleSizeInBits();
	boolean isBigEndian = format.isBigEndian();

	byte[] theFrame = getFrame(frameNum);

	if (format.getEncoding().equals(AudioFormat.Encoding.PCM_SIGNED)) {
		//since we're always returning the left sample,
		//we don't care if we're mono or stereo, left is
		//always first in the frame
		if (sampleSizeInBits == 8) { //8 bits == 1 byte
			return theFrame[0];
		} else if (sampleSizeInBits == 16)
			return TConversionTool.bytesToInt16(theFrame, 0,
					isBigEndian);
		else if (sampleSizeInBits == 24)
			return TConversionTool.bytesToInt24(theFrame, 0,
					isBigEndian);
		else if (sampleSizeInBits == 32)
			return TConversionTool.bytesToInt32(theFrame, 0,
					isBigEndian);
		else {
			printError("Unsupported audio encoding.  The sample " +
					"size is not recognized as a standard " +
					"format.");
			return -1;
		}
	} else if (format.getEncoding().equals(AudioFormat.Encoding.PCM_UNSIGNED)) {
		if (sampleSizeInBits == 8)
			return TTTConversionTool.unsignedByteToInt(theFrame[0]) -
					(int)Math.pow(2, 7);
		else if (sampleSizeInBits == 16)
			return TTTConversionTool.unsignedByteToInt16(theFrame, 0,
					isBigEndian) -
					(int)Math.pow(2, 15);
		else if (sampleSizeInBits == 24)
			return TTTConversionTool.unsignedByteToInt24(theFrame, 0,
					isBigEndian) -
					(int)Math.pow(2, 23);
		else if (sampleSizeInBits == 32)
			return TTTConversionTool.unsignedByteToInt32(theFrame, 0,
					isBigEndian) -
					(int)Math.pow(2, 31);
		else {
			printError("Unsupported audio encoding.  The sample " +
					"size is not recognized as a standard " +
					"format.");
			return -1;
		}
	} else if (format.getEncoding().equals(AudioFormat.Encoding.ALAW)) {
		return TTTConversionTool.alaw2linear(buffer[0]);
	} else if (format.getEncoding().equals(AudioFormat.Encoding.ULAW)) {
		return TTTConversionTool.ulaw2linear(buffer[0]);
	} else {
		printError("unsupported audio encoding: " +
				format.getEncoding() + ".  Currently only PCM, " +
				"ALAW and ULAW are supported.  Please try again" +
				"with a different file.");
		return -1;
	}
}
 
源代码20 项目: jdk8u60   文件: AudioFloatFormatConverter.java
AudioFloatFormatConverterInputStream(AudioFormat targetFormat,
        AudioFloatInputStream stream) {
    this.stream = stream;
    converter = AudioFloatConverter.getConverter(targetFormat);
    fsize = ((targetFormat.getSampleSizeInBits() + 7) / 8);
}