diff options
Diffstat (limited to 'songdbj/org/tritonus/share/sampled/FloatSampleTools.java')
| -rw-r--r-- | songdbj/org/tritonus/share/sampled/FloatSampleTools.java | 696 |
1 files changed, 0 insertions, 696 deletions
diff --git a/songdbj/org/tritonus/share/sampled/FloatSampleTools.java b/songdbj/org/tritonus/share/sampled/FloatSampleTools.java deleted file mode 100644 index 76913ba39e..0000000000 --- a/songdbj/org/tritonus/share/sampled/FloatSampleTools.java +++ /dev/null @@ -1,696 +0,0 @@ -/* - * FloatSampleTools.java - * - * This file is part of Tritonus: http://www.tritonus.org/ - */ - -/* - * Copyright (c) 2000,2004 by Florian Bomers <http://www.bomers.de> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU Library General Public License as published - * by the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU Library General Public License for more details. - * - * You should have received a copy of the GNU Library General Public - * License along with this program; if not, write to the Free Software - * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. - */ - -/* -|<--- this code is formatted to fit into 80 columns --->| -*/ - -package org.tritonus.share.sampled; - -import java.util.*; -import javax.sound.sampled.*; -import org.tritonus.share.TDebug; - -/** - * Utility functions for handling data in normalized float arrays. - * Each sample is linear in the range of [-1.0f, +1.0f]. - * <p> - * Currently, the following bit sizes are supported: - * <ul> - * <li>8-bit - * <li>16-bit - * <li>packed 24-bit (stored in 3 bytes) - * <li>32-bit - * </ul> - * 8-bit data can be unsigned or signed. All other data is only - * supported in signed encoding. - * - * @see FloatSampleBuffer - * @author Florian Bomers - */ - -public class FloatSampleTools { - - /** default number of bits to be dithered: 0.7f */ - public static final float DEFAULT_DITHER_BITS = 0.7f; - - private static Random random = null; - - // sample width (must be in order !) - static final int F_8=1; - static final int F_16=2; - static final int F_24=3; - static final int F_32=4; - static final int F_SAMPLE_WIDTH_MASK=F_8 | F_16 | F_24 | F_32; - - // format bit-flags - static final int F_SIGNED=8; - static final int F_BIGENDIAN=16; - - // supported formats - static final int CT_8S=F_8 | F_SIGNED; - static final int CT_8U=F_8; - static final int CT_16SB=F_16 | F_SIGNED | F_BIGENDIAN; - static final int CT_16SL=F_16 | F_SIGNED; - static final int CT_24SB=F_24 | F_SIGNED | F_BIGENDIAN; - static final int CT_24SL=F_24 | F_SIGNED; - static final int CT_32SB=F_32 | F_SIGNED | F_BIGENDIAN; - static final int CT_32SL=F_32 | F_SIGNED; - - // ////////////////////////////// initialization /////////////////////////////// // - - /** prevent instanciation */ - private FloatSampleTools() { - } - - - // /////////////////// FORMAT / FORMAT TYPE /////////////////////////////////// // - - /** - * only allow "packed" samples -- currently no support for 18, 20, 24_32 bits. - * @throws IllegalArgumentException - */ - static void checkSupportedSampleSize(int ssib, int channels, int frameSize) { - if ((ssib*channels) != frameSize * 8) { - throw new IllegalArgumentException("unsupported sample size: "+ssib - +" stored in "+(frameSize/channels)+" bytes."); - } - } - - - /** - * Get the formatType code from the given format. - * @throws IllegalArgumentException - */ - static int getFormatType(AudioFormat format) { - boolean signed = format.getEncoding().equals(AudioFormat.Encoding.PCM_SIGNED); - if (!signed && - !format.getEncoding().equals(AudioFormat.Encoding.PCM_UNSIGNED)) { - throw new IllegalArgumentException - ("unsupported encoding: only PCM encoding supported."); - } - if (!signed && format.getSampleSizeInBits() != 8) { - throw new IllegalArgumentException - ("unsupported encoding: only 8-bit can be unsigned"); - } - checkSupportedSampleSize(format.getSampleSizeInBits(), - format.getChannels(), - format.getFrameSize()); - - int formatType = getFormatType(format.getSampleSizeInBits(), - signed, format.isBigEndian()); - return formatType; - } - - /** - * @throws IllegalArgumentException - */ - static int getFormatType(int ssib, boolean signed, boolean bigEndian) { - int bytesPerSample=ssib/8; - int res=0; - if (ssib==8) { - res=F_8; - } else if (ssib==16) { - res=F_16; - } else if (ssib==24) { - res=F_24; - } else if (ssib==32) { - res=F_32; - } - if (res==0) { - throw new IllegalArgumentException - ("FloatSampleBuffer: unsupported sample size of " - +ssib+" bits per sample."); - } - if (!signed && bytesPerSample>1) { - throw new IllegalArgumentException - ("FloatSampleBuffer: unsigned samples larger than " - +"8 bit are not supported"); - } - if (signed) { - res|=F_SIGNED; - } - if (bigEndian && (ssib!=8)) { - res|=F_BIGENDIAN; - } - return res; - } - - static int getSampleSize(int formatType) { - switch (formatType & F_SAMPLE_WIDTH_MASK) { - case F_8: return 1; - case F_16: return 2; - case F_24: return 3; - case F_32: return 4; - } - return 0; - } - - /** - * Return a string representation of this format - */ - static String formatType2Str(int formatType) { - String res=""+formatType+": "; - switch (formatType & F_SAMPLE_WIDTH_MASK) { - case F_8: - res+="8bit"; - break; - case F_16: - res+="16bit"; - break; - case F_24: - res+="24bit"; - break; - case F_32: - res+="32bit"; - break; - } - res+=((formatType & F_SIGNED)==F_SIGNED)?" signed":" unsigned"; - if ((formatType & F_SAMPLE_WIDTH_MASK)!=F_8) { - res+=((formatType & F_BIGENDIAN)==F_BIGENDIAN)? - " big endian":" little endian"; - } - return res; - } - - - // /////////////////// BYTE 2 FLOAT /////////////////////////////////// // - - private static final float twoPower7=128.0f; - private static final float twoPower15=32768.0f; - private static final float twoPower23=8388608.0f; - private static final float twoPower31=2147483648.0f; - - private static final float invTwoPower7=1/twoPower7; - private static final float invTwoPower15=1/twoPower15; - private static final float invTwoPower23=1/twoPower23; - private static final float invTwoPower31=1/twoPower31; - - - /** - * Conversion function to convert an interleaved byte array to - * a List of interleaved float arrays. The float arrays will contain normalized - * samples in the range [-1.0f, +1.0f]. The input array - * provides bytes in the format specified in <code>format</code>. - * <p> - * Only PCM formats are accepted. The method will convert all - * byte values from - * <code>input[inByteOffset]</code> to - * <code>input[inByteOffset + (frameCount * format.getFrameSize()) - 1]</code> - * to floats from - * <code>output(n)[outOffset]</code> to - * <code>output(n)[outOffset + frameCount - 1]</code> - * - * @param input the audio data in an byte array - * @param inByteOffset index in input where to start the conversion - * @param output list of float[] arrays which receive the converted audio data. - * if the list does not contain enough elements, or individual float arrays - * are not large enough, they are created. - * @param outOffset the start offset in <code>output</code> - * @param frameCount number of frames to be converted - * @param format the input format. Only packed PCM is allowed - * @throws IllegalArgumentException if one of the parameters is out of bounds - * - * @see #byte2floatInterleaved(byte[],int,float[],int,int,AudioFormat) - */ - public static void byte2float(byte[] input, int inByteOffset, - List<float[]> output, int outOffset, int frameCount, - //List output, int outOffset, int frameCount, - AudioFormat format) { - for (int channel = 0; channel < format.getChannels(); channel++) { - float[] data; - if (output.size() < channel) { - data = new float[frameCount + outOffset]; - output.add(data); - } else { - data = output.get(channel); - if (data.length < frameCount + outOffset) { - data = new float[frameCount + outOffset]; - output.set(channel, data); - } - } - - byte2floatGeneric(input, inByteOffset, format.getFrameSize(), - data, outOffset, - frameCount, format); - inByteOffset += format.getFrameSize() / format.getChannels(); - } - } - - - /** - * Conversion function to convert an interleaved byte array to - * an interleaved float array. The float array will contain normalized - * samples in the range [-1.0f, +1.0f]. The input array - * provides bytes in the format specified in <code>format</code>. - * <p> - * Only PCM formats are accepted. The method will convert all - * byte values from - * <code>input[inByteOffset]</code> to - * <code>input[inByteOffset + (frameCount * format.getFrameSize()) - 1]</code> - * to floats from - * <code>output[outOffset]</code> to - * <code>output[outOffset + (frameCount * format.getChannels()) - 1]</code> - * - * @param input the audio data in an byte array - * @param inByteOffset index in input where to start the conversion - * @param output the float array that receives the converted audio data - * @param outOffset the start offset in <code>output</code> - * @param frameCount number of frames to be converted - * @param format the input format. Only packed PCM is allowed - * @throws IllegalArgumentException if one of the parameters is out of bounds - * - * @see #byte2float(byte[],int,List,int,int,AudioFormat) - */ - public static void byte2floatInterleaved(byte[] input, int inByteOffset, - float[] output, int outOffset, int frameCount, - AudioFormat format) { - - byte2floatGeneric(input, inByteOffset, format.getFrameSize() / format.getChannels(), - output, outOffset, frameCount * format.getChannels(), - format); - } - - - - /** - * Generic conversion function to convert a byte array to - * a float array. - * <p> - * Only PCM formats are accepted. The method will convert all - * bytes from - * <code>input[inByteOffset]</code> to - * <code>input[inByteOffset + (sampleCount * (inByteStep - 1)]</code> - * to samples from - * <code>output[outOffset]</code> to - * <code>output[outOffset+sampleCount-1]</code>. - * <p> - * The <code>format</code>'s channel count is ignored. - * <p> - * For mono data, set <code>inByteOffset</code> to <code>format.getFrameSize()</code>.<br> - * For converting interleaved input data, multiply <code>sampleCount</code> - * by the number of channels and set inByteStep to - * <code>format.getFrameSize() / format.getChannels()</code>. - * - * @param sampleCount number of samples to be written to output - * @param inByteStep how many bytes advance for each output sample in <code>output</code>. - * @throws IllegalArgumentException if one of the parameters is out of bounds - * - * @see #byte2floatInterleaved(byte[],int,float[],int,int,AudioFormat) - * @see #byte2float(byte[],int,List,int,int,AudioFormat) - */ - static void byte2floatGeneric(byte[] input, int inByteOffset, int inByteStep, - float[] output, int outOffset, int sampleCount, - AudioFormat format) { - int formatType = getFormatType(format); - - byte2floatGeneric(input, inByteOffset, inByteStep, - output, outOffset, sampleCount, - formatType); - } - - - /** - * Central conversion function from - * a byte array to a normalized float array. In order to accomodate - * interleaved and non-interleaved - * samples, this method takes inByteStep as parameter which - * can be used to flexibly convert the data. - * <p> - * E.g.:<br> - * mono->mono: inByteStep=format.getFrameSize()<br> - * interleaved_stereo->interleaved_stereo: inByteStep=format.getFrameSize()/2, - * sampleCount*2<br> - * stereo->2 mono arrays:<br> - * ---inByteOffset=0, outOffset=0, inByteStep=format.getFrameSize()<br> - * ---inByteOffset=format.getFrameSize()/2, outOffset=1, inByteStep=format.getFrameSize()<br> - */ - static void byte2floatGeneric(byte[] input, int inByteOffset, int inByteStep, - float[] output, int outOffset, int sampleCount, - int formatType) { - //if (TDebug.TraceAudioConverter) { - // TDebug.out("FloatSampleTools.byte2floatGeneric, formatType=" - // +formatType2Str(formatType)); - //} - int endCount = outOffset + sampleCount; - int inIndex = inByteOffset; - for (int outIndex = outOffset; outIndex < endCount; outIndex++, inIndex+=inByteStep) { - // do conversion - switch (formatType) { - case CT_8S: - output[outIndex]= - ((float) input[inIndex])*invTwoPower7; - break; - case CT_8U: - output[outIndex]= - ((float) ((input[inIndex] & 0xFF)-128))*invTwoPower7; - break; - case CT_16SB: - output[outIndex]= - ((float) ((input[inIndex]<<8) - | (input[inIndex+1] & 0xFF)))*invTwoPower15; - break; - case CT_16SL: - output[outIndex]= - ((float) ((input[inIndex+1]<<8) - | (input[inIndex] & 0xFF)))*invTwoPower15; - break; - case CT_24SB: - output[outIndex]= - ((float) ((input[inIndex]<<16) - | ((input[inIndex+1] & 0xFF)<<8) - | (input[inIndex+2] & 0xFF)))*invTwoPower23; - break; - case CT_24SL: - output[outIndex]= - ((float) ((input[inIndex+2]<<16) - | ((input[inIndex+1] & 0xFF)<<8) - | (input[inIndex] & 0xFF)))*invTwoPower23; - break; - case CT_32SB: - output[outIndex]= - ((float) ((input[inIndex]<<24) - | ((input[inIndex+1] & 0xFF)<<16) - | ((input[inIndex+2] & 0xFF)<<8) - | (input[inIndex+3] & 0xFF)))*invTwoPower31; - break; - case CT_32SL: - output[outIndex]= - ((float) ((input[inIndex+3]<<24) - | ((input[inIndex+2] & 0xFF)<<16) - | ((input[inIndex+1] & 0xFF)<<8) - | (input[inIndex] & 0xFF)))*invTwoPower31; - break; - default: - throw new IllegalArgumentException - ("unsupported format="+formatType2Str(formatType)); - } - } - } - - // /////////////////// FLOAT 2 BYTE /////////////////////////////////// // - - private static byte quantize8(float sample, float ditherBits) { - if (ditherBits!=0) { - sample+=random.nextFloat()*ditherBits; - } - if (sample>=127.0f) { - return (byte) 127; - } else if (sample<=-128.0f) { - return (byte) -128; - } else { - return (byte) (sample<0?(sample-0.5f):(sample+0.5f)); - } - } - - private static int quantize16(float sample, float ditherBits) { - if (ditherBits!=0) { - sample+=random.nextFloat()*ditherBits; - } - if (sample>=32767.0f) { - return 32767; - } else if (sample<=-32768.0f) { - return -32768; - } else { - return (int) (sample<0?(sample-0.5f):(sample+0.5f)); - } - } - - private static int quantize24(float sample, float ditherBits) { - if (ditherBits!=0) { - sample+=random.nextFloat()*ditherBits; - } - if (sample>=8388607.0f) { - return 8388607; - } else if (sample<=-8388608.0f) { - return -8388608; - } else { - return (int) (sample<0?(sample-0.5f):(sample+0.5f)); - } - } - - private static int quantize32(float sample, float ditherBits) { - if (ditherBits!=0) { - sample+=random.nextFloat()*ditherBits; - } - if (sample>=2147483647.0f) { - return 2147483647; - } else if (sample<=-2147483648.0f) { - return -2147483648; - } else { - return (int) (sample<0?(sample-0.5f):(sample+0.5f)); - } - } - - - /** - * Conversion function to convert a non-interleaved float audio data to - * an interleaved byte array. The float arrays contains normalized - * samples in the range [-1.0f, +1.0f]. The output array - * will receive bytes in the format specified in <code>format</code>. - * Exactly <code>format.getChannels()</code> channels are converted - * regardless of the number of elements in <code>input</code>. If <code>input</code> - * does not provide enough channels, an </code>IllegalArgumentException<code> is thrown. - * <p> - * Only PCM formats are accepted. The method will convert all - * samples from <code>input(n)[inOffset]</code> to - * <code>input(n)[inOffset + frameCount - 1]</code> - * to byte values from <code>output[outByteOffset]</code> to - * <code>output[outByteOffset + (frameCount * format.getFrameSize()) - 1]</code> - * <p> - * Dithering should be used when the output resolution is significantly - * lower than the original resolution. This includes if the original - * data was 16-bit and it is now converted to 8-bit, or if the - * data was generated in the float domain. No dithering need to be used - * if the original sample data was in e.g. 8-bit and the resulting output - * data has a higher resolution. If dithering is used, a sensitive value - * is DEFAULT_DITHER_BITS. - * - * @param input a List of float arrays with the input audio data - * @param inOffset index in the input arrays where to start the conversion - * @param output the byte array that receives the converted audio data - * @param outByteOffset the start offset in <code>output</code> - * @param frameCount number of frames to be converted. - * @param format the output format. Only packed PCM is allowed - * @param ditherBits if 0, do not dither. Otherwise the number of bits to be dithered - * @throws IllegalArgumentException if one of the parameters is out of bounds - * - * @see #DEFAULT_DITHER_BITS - * @see #float2byteInterleaved(float[],int,byte[],int,int,AudioFormat,float) - */ - //public static void float2byte(List<float[]> input, int inOffset, - public static void float2byte(List input, int inOffset, - byte[] output, int outByteOffset, - int frameCount, - AudioFormat format, float ditherBits) { - for (int channel = 0; channel < format.getChannels(); channel++) { - float[] data = (float[]) input.get(channel); - float2byteGeneric(data, inOffset, - output, outByteOffset, format.getFrameSize(), - frameCount, format, ditherBits); - outByteOffset += format.getFrameSize() / format.getChannels(); - } - } - - /** - * Conversion function to convert an interleaved float array to - * an interleaved byte array. The float array contains normalized - * samples in the range [-1.0f, +1.0f]. The output array - * will receive bytes in the format specified in <code>format</code>. - * <p> - * Only PCM formats are accepted. The method will convert all - * samples from <code>input[inOffset]</code> to - * <code>input[inOffset + (frameCount * format.getChannels()) - 1]</code> - * to byte values from <code>output[outByteOffset]</code> to - * <code>output[outByteOffset + (frameCount * format.getFrameSize()) - 1]</code> - * <p> - * Dithering should be used when the output resolution is significantly - * lower than the original resolution. This includes if the original - * data was 16-bit and it is now converted to 8-bit, or if the - * data was generated in the float domain. No dithering need to be used - * if the original sample data was in e.g. 8-bit and the resulting output - * data has a higher resolution. If dithering is used, a sensitive value - * is DEFAULT_DITHER_BITS. - * - * @param input the audio data in normalized samples - * @param inOffset index in input where to start the conversion - * @param output the byte array that receives the converted audio data - * @param outByteOffset the start offset in <code>output</code> - * @param frameCount number of frames to be converted. - * @param format the output format. Only packed PCM is allowed - * @param ditherBits if 0, do not dither. Otherwise the number of bits to be dithered - * @throws IllegalArgumentException if one of the parameters is out of bounds - * - * @see #DEFAULT_DITHER_BITS - * @see #float2byte(List,int,byte[],int,int,AudioFormat,float) - */ - public static void float2byteInterleaved(float[] input, int inOffset, - byte[] output, int outByteOffset, - int frameCount, - AudioFormat format, float ditherBits) { - float2byteGeneric(input, inOffset, - output, outByteOffset, format.getFrameSize() / format.getChannels(), - frameCount * format.getChannels(), - format, ditherBits); - } - - - - /** - * Generic conversion function to convert a float array to - * a byte array. - * <p> - * Only PCM formats are accepted. The method will convert all - * samples from <code>input[inOffset]</code> to - * <code>input[inOffset+sampleCount-1]</code> - * to byte values from <code>output[outByteOffset]</code> to - * <code>output[outByteOffset + (sampleCount * (outByteStep - 1)]</code>. - * <p> - * The <code>format</code>'s channel count is ignored. - * <p> - * For mono data, set <code>outByteOffset</code> to <code>format.getFrameSize()</code>.<br> - * For converting interleaved input data, multiply <code>sampleCount</code> - * by the number of channels and set outByteStep to - * <code>format.getFrameSize() / format.getChannels()</code>. - * - * @param sampleCount number of samples in input to be converted. - * @param outByteStep how many bytes advance for each input sample in <code>input</code>. - * @throws IllegalArgumentException if one of the parameters is out of bounds - * - * @see #float2byteInterleaved(float[],int,byte[],int,int,AudioFormat,float) - * @see #float2byte(List,int,byte[],int,int,AudioFormat,float) - */ - static void float2byteGeneric(float[] input, int inOffset, - byte[] output, int outByteOffset, int outByteStep, - int sampleCount, - AudioFormat format, float ditherBits) { - int formatType = getFormatType(format); - - float2byteGeneric(input, inOffset, - output, outByteOffset, outByteStep, - sampleCount, - formatType, ditherBits); - } - - - /** - * Central conversion function from normalized float array to - * a byte array. In order to accomodate interleaved and non-interleaved - * samples, this method takes outByteStep as parameter which - * can be used to flexibly convert the data. - * <p> - * E.g.:<br> - * mono->mono: outByteStep=format.getFrameSize()<br> - * interleaved stereo->interleaved stereo: outByteStep=format.getFrameSize()/2, sampleCount*2<br> - * 2 mono arrays->stereo:<br> - * ---inOffset=0, outByteOffset=0, outByteStep=format.getFrameSize()<br> - * ---inOffset=1, outByteOffset=format.getFrameSize()/2, outByteStep=format.getFrameSize()<br> - */ - static void float2byteGeneric(float[] input, int inOffset, - byte[] output, int outByteOffset, int outByteStep, - int sampleCount, int formatType, float ditherBits) { - //if (TDebug.TraceAudioConverter) { - // TDebug.out("FloatSampleBuffer.float2byteGeneric, formatType=" - // +"formatType2Str(formatType)); - //} - - if (inOffset < 0 - || inOffset + sampleCount > input.length - || sampleCount < 0) { - throw new IllegalArgumentException("invalid input index: " - +"input.length="+input.length - +" inOffset="+inOffset - +" sampleCount="+sampleCount); - } - if (outByteOffset < 0 - || outByteOffset + (sampleCount * outByteStep) > output.length - || outByteStep < getSampleSize(formatType)) { - throw new IllegalArgumentException("invalid output index: " - +"output.length="+output.length - +" outByteOffset="+outByteOffset - +" sampleCount="+sampleCount - +" format="+formatType2Str(formatType)); - } - - if (ditherBits!=0.0f && random==null) { - // create the random number generator for dithering - random=new Random(); - } - int endSample = inOffset + sampleCount; - int iSample; - int outIndex = outByteOffset; - for (int inIndex = inOffset; - inIndex < endSample; - inIndex++, outIndex+=outByteStep) { - // do conversion - switch (formatType) { - case CT_8S: - output[outIndex]=quantize8(input[inIndex]*twoPower7, ditherBits); - break; - case CT_8U: - output[outIndex]=(byte) (quantize8((input[inIndex]*twoPower7), ditherBits)+128); - break; - case CT_16SB: - iSample=quantize16(input[inIndex]*twoPower15, ditherBits); - output[outIndex]=(byte) (iSample >> 8); - output[outIndex+1]=(byte) (iSample & 0xFF); - break; - case CT_16SL: - iSample=quantize16(input[inIndex]*twoPower15, ditherBits); - output[outIndex+1]=(byte) (iSample >> 8); - output[outIndex]=(byte) (iSample & 0xFF); - break; - case CT_24SB: - iSample=quantize24(input[inIndex]*twoPower23, ditherBits); - output[outIndex]=(byte) (iSample >> 16); - output[outIndex+1]=(byte) ((iSample >>> 8) & 0xFF); - output[outIndex+2]=(byte) (iSample & 0xFF); - break; - case CT_24SL: - iSample=quantize24(input[inIndex]*twoPower23, ditherBits); - output[outIndex+2]=(byte) (iSample >> 16); - output[outIndex+1]=(byte) ((iSample >>> 8) & 0xFF); - output[outIndex]=(byte) (iSample & 0xFF); - break; - case CT_32SB: - iSample=quantize32(input[inIndex]*twoPower31, ditherBits); - output[outIndex]=(byte) (iSample >> 24); - output[outIndex+1]=(byte) ((iSample >>> 16) & 0xFF); - output[outIndex+2]=(byte) ((iSample >>> 8) & 0xFF); - output[outIndex+3]=(byte) (iSample & 0xFF); - break; - case CT_32SL: - iSample=quantize32(input[inIndex]*twoPower31, ditherBits); - output[outIndex+3]=(byte) (iSample >> 24); - output[outIndex+2]=(byte) ((iSample >>> 16) & 0xFF); - output[outIndex+1]=(byte) ((iSample >>> 8) & 0xFF); - output[outIndex]=(byte) (iSample & 0xFF); - break; - default: - throw new IllegalArgumentException - ("unsupported format="+formatType2Str(formatType)); - } - } - } -} |