本文整理匯總了Java中javax.sound.sampled.AudioFormat.isBigEndian方法的典型用法代碼示例。如果您正苦於以下問題:Java AudioFormat.isBigEndian方法的具體用法?Java AudioFormat.isBigEndian怎麽用?Java AudioFormat.isBigEndian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.isBigEndian方法的15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: getAudioFormat
import javax.sound.sampled.AudioFormat; //導入方法依賴的package包/類
private AudioFormat getAudioFormat(HashSet<AudioFormat> inputFormats) {
for (AudioFormat audioFormat : inputFormats) {
if((audioFormat.getSampleSizeInBits() == 16 || audioFormat.getSampleSizeInBits() == 8)
// && (audioFormat.getSampleRate() == 16000.0F)
&& (audioFormat.getChannels() == 1)
&& (audioFormat.isBigEndian())
&& (audioFormat.getEncoding().equals(AudioFormat.Encoding.PCM_SIGNED))){
return new AudioFormat(16000.0F, audioFormat.getSampleSizeInBits(), audioFormat.getChannels(), true, true);
}
}
return null;
// float sampleRate = 16000.0F;
// // 8000,11025,16000,22050,44100
// int sampleSizeInBits = 16;
// // 8,16
// int channels = 1;
// // 1,2
// boolean signed = true;
// // true,false
// boolean bigEndian = true;
// // true,false
// return new AudioFormat(sampleRate, sampleSizeInBits, channels, signed, bigEndian);
}
示例2: getSignOrEndianChangedFormat
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;
}
示例3: getAudioInputStream
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);
}
示例4: getAudioInputStream
import javax.sound.sampled.AudioFormat; //導入方法依賴的package包/類
@Override
public AudioInputStream getAudioInputStream(Encoding targetEncoding,
AudioInputStream sourceStream) {
if (!isConversionSupported(targetEncoding, sourceStream.getFormat())) {
throw new IllegalArgumentException(
"Unsupported conversion: " + sourceStream.getFormat()
.toString() + " to " + targetEncoding.toString());
}
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);
}
示例5: AudioFloatLSBFilter
import javax.sound.sampled.AudioFormat; //導入方法依賴的package包/類
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;
}
示例6: getAudioInputStream
import javax.sound.sampled.AudioFormat; //導入方法依賴的package包/類
/**
*/
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 getAudioInputStream( targetFormat, sourceStream );
} else {
throw new IllegalArgumentException("Unsupported conversion: " + sourceStream.getFormat().toString() + " to " + targetEncoding.toString() );
}
}
示例7: getOtherBits
import javax.sound.sampled.AudioFormat; //導入方法依賴的package包/類
public static AudioFormat getOtherBits(AudioFormat format, int newBits) {
boolean isSigned = format.getEncoding().equals(AudioFormat.Encoding.PCM_SIGNED);
return new AudioFormat(format.getSampleRate(),
newBits,
format.getChannels(),
isSigned,
(newBits>8)?format.isBigEndian():false);
}
示例8: 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());
}
示例9: getOtherEndianOrSign
import javax.sound.sampled.AudioFormat; //導入方法依賴的package包/類
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;
}
示例10: 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;
}
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;
}
示例12: getOtherChannels
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());
}
示例13: AlawCodecStream
import javax.sound.sampled.AudioFormat; //導入方法依賴的package包/類
AlawCodecStream(AudioInputStream stream, AudioFormat outputFormat) {
super(stream, outputFormat, -1);
AudioFormat inputFormat = stream.getFormat();
// throw an IllegalArgumentException if not ok
if ( ! (isConversionSupported(outputFormat, inputFormat)) ) {
throw new IllegalArgumentException("Unsupported conversion: " + inputFormat.toString() + " to " + outputFormat.toString());
}
//$$fb 2002-07-18: fix for 4714846: JavaSound ULAW (8-bit) encoder erroneously depends on endian-ness
boolean PCMIsBigEndian;
// determine whether we are encoding or decoding
if (AudioFormat.Encoding.ALAW.equals(inputFormat.getEncoding())) {
encode = false;
encodeFormat = inputFormat;
decodeFormat = outputFormat;
PCMIsBigEndian = outputFormat.isBigEndian();
} else {
encode = true;
encodeFormat = outputFormat;
decodeFormat = inputFormat;
PCMIsBigEndian = inputFormat.isBigEndian();
tempBuffer = new byte[tempBufferSize];
}
if (PCMIsBigEndian) {
tabByte1 = ALAW_TABH;
tabByte2 = ALAW_TABL;
highByte = 0;
lowByte = 1;
} else {
tabByte1 = ALAW_TABL;
tabByte2 = ALAW_TABH;
highByte = 1;
lowByte = 0;
}
// set the AudioInputStream length in frames if we know it
if (stream instanceof AudioInputStream) {
frameLength = ((AudioInputStream)stream).getFrameLength();
}
// set framePos to zero
framePos = 0;
frameSize = inputFormat.getFrameSize();
if( frameSize==AudioSystem.NOT_SPECIFIED ) {
frameSize=1;
}
}
示例14: printFormat
import javax.sound.sampled.AudioFormat; //導入方法依賴的package包/類
public static String printFormat(AudioFormat format) {
return format.toString()+" "+(format.isBigEndian()?"big":"little")+" endian";
}
示例15: UlawCodecStream
import javax.sound.sampled.AudioFormat; //導入方法依賴的package包/類
UlawCodecStream(AudioInputStream stream, AudioFormat outputFormat) {
super(stream, outputFormat, AudioSystem.NOT_SPECIFIED);
AudioFormat inputFormat = stream.getFormat();
// throw an IllegalArgumentException if not ok
if (!(isConversionSupported(outputFormat, inputFormat))) {
throw new IllegalArgumentException("Unsupported conversion: " + inputFormat.toString() + " to " + outputFormat.toString());
}
//$$fb 2002-07-18: fix for 4714846: JavaSound ULAW (8-bit) encoder erroneously depends on endian-ness
boolean PCMIsBigEndian;
// determine whether we are encoding or decoding
if (AudioFormat.Encoding.ULAW.equals(inputFormat.getEncoding())) {
encode = false;
encodeFormat = inputFormat;
decodeFormat = outputFormat;
PCMIsBigEndian = outputFormat.isBigEndian();
} else {
encode = true;
encodeFormat = outputFormat;
decodeFormat = inputFormat;
PCMIsBigEndian = inputFormat.isBigEndian();
tempBuffer = new byte[tempBufferSize];
}
// setup tables according to byte order
if (PCMIsBigEndian) {
tabByte1 = ULAW_TABH;
tabByte2 = ULAW_TABL;
highByte = 0;
lowByte = 1;
} else {
tabByte1 = ULAW_TABL;
tabByte2 = ULAW_TABH;
highByte = 1;
lowByte = 0;
}
// set the AudioInputStream length in frames if we know it
if (stream instanceof AudioInputStream) {
frameLength = ((AudioInputStream)stream).getFrameLength();
}
// set framePos to zero
framePos = 0;
frameSize = inputFormat.getFrameSize();
if (frameSize == AudioSystem.NOT_SPECIFIED) {
frameSize = 1;
}
}