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Java AudioFormat.getEncoding方法代碼示例

本文整理匯總了Java中javax.sound.sampled.AudioFormat.getEncoding方法的典型用法代碼示例。如果您正苦於以下問題：Java AudioFormat.getEncoding方法的具體用法？Java AudioFormat.getEncoding怎麽用？Java AudioFormat.getEncoding使用的例子？那麽, 這裏精選的方法代碼示例或許可以為您提供幫助。您也可以進一步了解該方法所在類javax.sound.sampled.AudioFormat的用法示例。


在下文中一共展示了AudioFormat.getEncoding方法的14個代碼示例，這些例子默認根據受歡迎程度排序。您可以為喜歡或者感覺有用的代碼點讚，您的評價將有助於係統推薦出更棒的Java代碼示例。

示例1: getAudioFileTypes

import javax.sound.sampled.AudioFormat; //導入方法依賴的package包/類
@Override
public Type[] getAudioFileTypes(AudioInputStream stream) {

    Type[] filetypes = new Type[types.length];
    System.arraycopy(types, 0, filetypes, 0, types.length);

    // make sure we can write this stream
    AudioFormat format = stream.getFormat();
    AudioFormat.Encoding encoding = format.getEncoding();

    if (AudioFormat.Encoding.ALAW.equals(encoding)
            || AudioFormat.Encoding.ULAW.equals(encoding)
            || AudioFormat.Encoding.PCM_SIGNED.equals(encoding)
            || AudioFormat.Encoding.PCM_UNSIGNED.equals(encoding)
            || AudioFormat.Encoding.PCM_FLOAT.equals(encoding)) {
        return filetypes;
    }

    return new Type[0];
}
開發者ID:AdoptOpenJDK，項目名稱:openjdk-jdk10，代碼行數:21，代碼來源:AuFileWriter.java

示例2: convertAudioBytes

import javax.sound.sampled.AudioFormat; //導入方法依賴的package包/類
/**
 * Convert the audio bytes into the stream
 * 
 * @param format The audio format being decoded
 * @param audio_bytes The audio byts
 * @param two_bytes_data True if we using double byte data
 * @return The byte bufer of data
 */
private static ByteBuffer convertAudioBytes(AudioFormat format, byte[] audio_bytes, boolean two_bytes_data) {
	ByteBuffer dest = ByteBuffer.allocateDirect(audio_bytes.length);
	dest.order(ByteOrder.nativeOrder());
	ByteBuffer src = ByteBuffer.wrap(audio_bytes);
	src.order(ByteOrder.BIG_ENDIAN);
	if (two_bytes_data) {
		ShortBuffer dest_short = dest.asShortBuffer();
		ShortBuffer src_short = src.asShortBuffer();
		while (src_short.hasRemaining())
			dest_short.put(src_short.get());
	} else {
		while (src.hasRemaining()) {
			byte b = src.get();
			if (format.getEncoding() == Encoding.PCM_SIGNED) {
				b = (byte) (b + 127);
			}
			dest.put(b);
		}
	}
	dest.rewind();
	return dest;
}
開發者ID:SkidJava，項目名稱:BaseClient，代碼行數:31，代碼來源:AiffData.java

示例3: WaveFileFormat

import javax.sound.sampled.AudioFormat; //導入方法依賴的package包/類
WaveFileFormat(AudioFileFormat.Type type, int lengthInBytes, AudioFormat format, int lengthInFrames) {

        super(type,lengthInBytes,format,lengthInFrames);

        AudioFormat.Encoding encoding = format.getEncoding();

        if( encoding.equals(AudioFormat.Encoding.ALAW) ) {
            waveType = WAVE_FORMAT_ALAW;
        } else if( encoding.equals(AudioFormat.Encoding.ULAW) ) {
            waveType = WAVE_FORMAT_MULAW;
        } else if( encoding.equals(AudioFormat.Encoding.PCM_SIGNED) ||
                   encoding.equals(AudioFormat.Encoding.PCM_UNSIGNED) ) {
            waveType = WAVE_FORMAT_PCM;
        } else {
            waveType = WAVE_FORMAT_UNKNOWN;
        }
    }
開發者ID:SunburstApps，項目名稱:OpenJSharp，代碼行數:18，代碼來源:WaveFileFormat.java

示例4: getAudioFileTypes

import javax.sound.sampled.AudioFormat; //導入方法依賴的package包/類
public AudioFileFormat.Type[] getAudioFileTypes(AudioInputStream stream) {

        AudioFileFormat.Type[] filetypes = new AudioFileFormat.Type[types.length];
        System.arraycopy(types, 0, filetypes, 0, types.length);

        // make sure we can write this stream
        AudioFormat format = stream.getFormat();
        AudioFormat.Encoding encoding = format.getEncoding();

        if( (AudioFormat.Encoding.ALAW.equals(encoding)) ||
            (AudioFormat.Encoding.ULAW.equals(encoding)) ||
            (AudioFormat.Encoding.PCM_SIGNED.equals(encoding)) ||
            (AudioFormat.Encoding.PCM_UNSIGNED.equals(encoding)) ) {

            return filetypes;
        }

        return new AudioFileFormat.Type[0];
    }
開發者ID:SunburstApps，項目名稱:OpenJSharp，代碼行數:20，代碼來源:AuFileWriter.java

示例5: getAudioFileTypes

import javax.sound.sampled.AudioFormat; //導入方法依賴的package包/類
public AudioFileFormat.Type[] getAudioFileTypes(AudioInputStream stream) {

        AudioFileFormat.Type[] filetypes = new AudioFileFormat.Type[types.length];
        System.arraycopy(types, 0, filetypes, 0, types.length);

        // make sure we can write this stream
        AudioFormat format = stream.getFormat();
        AudioFormat.Encoding encoding = format.getEncoding();

        if( AudioFormat.Encoding.ALAW.equals(encoding) ||
            AudioFormat.Encoding.ULAW.equals(encoding) ||
            AudioFormat.Encoding.PCM_SIGNED.equals(encoding) ||
            AudioFormat.Encoding.PCM_UNSIGNED.equals(encoding) ) {

            return filetypes;
        }

        return new AudioFileFormat.Type[0];
    }
開發者ID:lambdalab-mirror，項目名稱:jdk8u-jdk，代碼行數:20，代碼來源:WaveFileWriter.java

示例6: AudioFloatInputStreamChannelMixer

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());
}
開發者ID:AdoptOpenJDK，項目名稱:openjdk-jdk10，代碼行數:13，代碼來源:AudioFloatFormatConverter.java

示例7: convertAudioInputStreamToPcmSigned

import javax.sound.sampled.AudioFormat; //導入方法依賴的package包/類
private void convertAudioInputStreamToPcmSigned() {
    AudioFormat format = audioInputStream.getFormat();
    if (format.getEncoding() != AudioFormat.Encoding.PCM_SIGNED) {
        audioInputStream = AudioSystem.getAudioInputStream(
                new AudioFormat(AudioFormat.Encoding.PCM_SIGNED,
                        format.getSampleRate(),
                        16,
                        1,
                        2,
                        format.getSampleRate(),
                        false), audioInputStream);
    }
}
開發者ID:eightbitjim，項目名稱:cassette-nibbler，代碼行數:14，代碼來源:AudioFile.java

示例8: AuFileFormat

import javax.sound.sampled.AudioFormat; //導入方法依賴的package包/類
AuFileFormat(final AudioFileFormat.Type type, final long byteLength,
             final AudioFormat format, final long frameLength) {
    super(type, byteLength, format, frameLength);

    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;
        }
    } else if (AudioFormat.Encoding.PCM_FLOAT.equals(encoding)) {
        if (format.getSampleSizeInBits() == 32) {
            auType = AU_FLOAT;
        }
    }
}
開發者ID:AdoptOpenJDK，項目名稱:openjdk-jdk10，代碼行數:33，代碼來源:AuFileFormat.java

示例9: AudioFloatInputStreamResampler

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;
}
開發者ID:AdoptOpenJDK，項目名稱:openjdk-jdk10，代碼行數:39，代碼來源:SoftMixingDataLine.java

示例10: toLittleEndian

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);
}
開發者ID:lambdalab-mirror，項目名稱:jdk8u-jdk，代碼行數:9，代碼來源:WaveFloatFileWriter.java

示例11: AudioFloatInputStreamResampler

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;
}
開發者ID:AdoptOpenJDK，項目名稱:openjdk-jdk10，代碼行數:39，代碼來源:AudioFloatFormatConverter.java

示例12: AuFileFormat

import javax.sound.sampled.AudioFormat; //導入方法依賴的package包/類
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;
            }
        }

    }
開發者ID:lambdalab-mirror，項目名稱:jdk8u-jdk，代碼行數:30，代碼來源:AuFileFormat.java

示例13: getAudioFileFormat

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());
}
開發者ID:AdoptOpenJDK，項目名稱:openjdk-jdk10，代碼行數:35，代碼來源:AuFileWriter.java

示例14: isConversionSupported

import javax.sound.sampled.AudioFormat; //導入方法依賴的package包/類
/**
 * Indicates whether the format converter supports conversion to one
 * particular format from another.
 *
 * @param  targetFormat desired format of outgoing data
 * @param  sourceFormat format of the incoming data
 * @return {@code true} if the conversion is supported, otherwise
 *         {@code false}
 * @throws NullPointerException if {@code targetFormat} or
 *         {@code sourceFormat} are {@code null}
 */
public boolean isConversionSupported(final AudioFormat targetFormat,
                                     final AudioFormat sourceFormat) {
    final Encoding targetEncoding = targetFormat.getEncoding();
    return Stream.of(getTargetFormats(targetEncoding, sourceFormat))
                 .anyMatch(targetFormat::matches);
}
開發者ID:AdoptOpenJDK，項目名稱:openjdk-jdk10，代碼行數:18，代碼來源:FormatConversionProvider.java


注：本文中的javax.sound.sampled.AudioFormat.getEncoding方法示例由純淨天空整理自Github/MSDocs等開源代碼及文檔管理平台，相關代碼片段篩選自各路編程大神貢獻的開源項目，源碼版權歸原作者所有，傳播和使用請參考對應項目的License；未經允許，請勿轉載。