/*
* FloatSampleTools.java
*
* This file is part of Tritonus: http://www.tritonus.org/
*/
/*
* Copyright (c) 2000-2006 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 ddf.minim.javasound;
import java.util.List;
import java.util.Random;
import javax.sound.sampled.AudioFormat;
/**
* 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>unpacked 24-bit (stored in 4 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 = 3;
static final int F_24_4 = 4;
static final int F_32 = 5;
static final int F_SAMPLE_WIDTH_MASK = F_8 | F_16 | F_24_3 | F_24_4 | 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_24_3SB = F_24_3 | F_SIGNED | F_BIGENDIAN;
static final int CT_24_3SL = F_24_3 | F_SIGNED;
static final int CT_24_4SB = F_24_4 | F_SIGNED | F_BIGENDIAN;
static final int CT_24_4SL = F_24_4 | 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 bits --
* except 24 bits stored in 4 bytes.
*
* @throws IllegalArgumentException
*/
static void checkSupportedSampleSize(int ssib, int channels, int frameSize) {
if (ssib == 24 && frameSize == 4 * channels) {
// 24 bits stored in 4 bytes is OK (24_4)
return;
}
if ((ssib * channels) != frameSize * 8) {
throw new IllegalArgumentException("unsupported sample size: "
+ ssib + " bits 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(),
format.getFrameSize() / format.getChannels(), signed,
format.isBigEndian());
return formatType;
}
/**
* @throws IllegalArgumentException
*/
static int getFormatType(int ssib, int bytesPerSample, boolean signed,
boolean bigEndian) {
int res = 0;
if (ssib == 24 || (bytesPerSample == ssib / 8)) {
if (ssib == 8) {
res = F_8;
} else if (ssib == 16) {
res = F_16;
} else if (ssib == 24) {
if (bytesPerSample == 3) {
res = F_24_3;
} else if (bytesPerSample == 4) {
res = F_24_4;
}
} else if (ssib == 32) {
res = F_32;
}
}
if (res == 0) {
throw new IllegalArgumentException(
"ConversionTool: unsupported sample size of " + ssib
+ " bits per sample in " + bytesPerSample
+ " bytes.");
}
if (!signed && bytesPerSample > 1) {
throw new IllegalArgumentException(
"ConversionTool: 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_3:
return 3;
case F_24_4:
return 4;
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_3:
res += "24_3bit";
break;
case F_24_4:
res += "24_4bit";
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;
/**
* @see #byte2float(byte[] input, int inByteOffset, Object[] output, int
* outOffset, int frameCount, AudioFormat format, boolean
* allowAddChannel)
*/
public static void byte2float(byte[] input, int inByteOffset,
List<float[]> output, int outOffset, int frameCount,
AudioFormat format) {
byte2float(input, inByteOffset, output, outOffset, frameCount, format,
true);
}
/**
* @param output an array of float[] arrays
* @throws ArrayIndexOutOfBoundsException if output does not
* format.getChannels() elements
* @see #byte2float(byte[] input, int inByteOffset, Object[] output, int
* outOffset, int frameCount, AudioFormat format, boolean
* allowAddChannel)
*/
public static void byte2float(byte[] input, int inByteOffset,
Object[] output, int outOffset, int frameCount, AudioFormat format) {
byte2float(input, inByteOffset, output, outOffset, frameCount, format,
true);
}
/**
* @param output an array of float[] arrays
* @param allowAddChannel if true, and output has fewer channels than
* format, then only output.length channels are filled
* @throws ArrayIndexOutOfBoundsException if output does not
* format.getChannels() elements
* @see #byte2float(byte[] input, int inByteOffset, Object[] output, int
* outOffset, int frameCount, AudioFormat format, boolean
* allowAddChannel)
*/
public static void byte2float(byte[] input, int inByteOffset,
Object[] output, int outOffset, int frameCount, AudioFormat format,
boolean allowAddChannel) {
int channels = format.getChannels();
if (!allowAddChannel && channels > output.length) {
channels = output.length;
}
if (output.length < channels) {
throw new ArrayIndexOutOfBoundsException(
"too few channel output array");
}
for (int channel = 0; channel < channels; channel++) {
float[] data = (float[]) output[channel];
if (data.length < frameCount + outOffset) {
data = new float[frameCount + outOffset];
output[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 a List of
* interleaved float arrays. The float arrays will contain normalized
* samples in the range [-1.0, +1.0]. 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
* @param allowAddChannel if true, channels may be added to
* <code>output</code> to match the number of input channels,
* otherwise, only the first output.size() channels of input data
* are converted.
* @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,
AudioFormat format, boolean allowAddChannel) {
int channels = format.getChannels();
if (!allowAddChannel && channels > output.size()) {
channels = output.size();
}
for (int channel = 0; channel < channels; 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 one audio channel in an interleaved byte
* array to a float array. The float array will contain normalized samples
* in the range [-1.0, +1.0]. 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 channel the channel number to extract from the input audio data
* @param input the audio data in an byte array
* @param inByteOffset index in input where to start the conversion
* @param output the of float array which 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
*/
public static void byte2float(int channel, byte[] input, int inByteOffset,
float[] output, int outOffset, int frameCount, AudioFormat format) {
if (channel >= format.getChannels()) {
throw new IllegalArgumentException("channel out of bounds");
}
if (output.length < frameCount + outOffset) {
throw new IllegalArgumentException("data is too small");
}
// "select" the channel
inByteOffset += format.getFrameSize() / format.getChannels() * channel;
byte2floatGeneric(input, inByteOffset, format.getFrameSize(), output,
outOffset, frameCount, format);
}
/**
* 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] = input[inIndex] * invTwoPower7;
break;
case CT_8U:
output[outIndex] = ((input[inIndex] & 0xFF) - 128) * invTwoPower7;
break;
case CT_16SB:
output[outIndex] = ((input[inIndex] << 8)
| (input[inIndex + 1] & 0xFF))
* invTwoPower15;
break;
case CT_16SL:
output[outIndex] = ((input[inIndex + 1] << 8)
| (input[inIndex] & 0xFF))
* invTwoPower15;
break;
case CT_24_3SB:
output[outIndex] = ((input[inIndex] << 16)
| ((input[inIndex + 1] & 0xFF) << 8)
| (input[inIndex + 2] & 0xFF))
* invTwoPower23;
break;
case CT_24_3SL:
output[outIndex] = ((input[inIndex + 2] << 16)
| ((input[inIndex + 1] & 0xFF) << 8)
| (input[inIndex] & 0xFF))
* invTwoPower23;
break;
case CT_24_4SB:
output[outIndex] = ((input[inIndex + 1] << 16)
| ((input[inIndex + 2] & 0xFF) << 8)
| (input[inIndex + 3] & 0xFF))
* invTwoPower23;
break;
case CT_24_4SL:
// TODO: verify the indexes
output[outIndex] = ((input[inIndex + 3] << 16)
| ((input[inIndex + 2] & 0xFF) << 8)
| (input[inIndex + 1] & 0xFF))
* invTwoPower23;
break;
case CT_32SB:
output[outIndex] = ((input[inIndex] << 24)
| ((input[inIndex + 1] & 0xFF) << 16)
| ((input[inIndex + 2] & 0xFF) << 8)
| (input[inIndex + 3] & 0xFF))
* invTwoPower31;
break;
case CT_32SL:
output[outIndex] = ((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,
byte[] output, int outByteOffset, int frameCount,
AudioFormat format, float ditherBits) {
for (int channel = 0; channel < format.getChannels(); channel++) {
float[] data = input.get(channel);
float2byteGeneric(data, inOffset, output, outByteOffset,
format.getFrameSize(), frameCount, format, ditherBits);
outByteOffset += format.getFrameSize() / format.getChannels();
}
}
/**
* @param input an array of float[] arrays
* @throws ArrayIndexOutOfBoundsException if one of the parameters is out of
* bounds
* @see #float2byte(Object[], int, byte[], int, int, AudioFormat, float)
*/
public static void float2byte(Object[] input, int inOffset, byte[] output,
int outByteOffset, int frameCount, AudioFormat format,
float ditherBits) {
int channels = format.getChannels();
for (int channel = 0; channel < channels; channel++) {
float[] data = (float[]) input[channel];
float2byteGeneric(data, inOffset, output, outByteOffset,
format.getFrameSize(), frameCount, format, ditherBits);
outByteOffset += format.getFrameSize() / format.getChannels();
}
}
/**
* @param input an array of float[] arrays
* @param channels how many channels to use from the input array
* @param frameSize only as optimization, the number of bytes per sample
* frame
* @throws ArrayIndexOutOfBoundsException if one of the parameters is out of
* bounds
* @see #float2byte(Object[], int, byte[], int, int, AudioFormat, float)
*/
static void float2byte(Object[] input, int inOffset, byte[] output,
int outByteOffset, int frameCount, int formatCode, int channels,
int frameSize, float ditherBits) {
int sampleSize = frameSize / channels;
for (int channel = 0; channel < channels; channel++) {
float[] data = (float[]) input[channel];
float2byteGeneric(data, inOffset, output, outByteOffset, frameSize,
frameCount, formatCode, ditherBits);
outByteOffset += sampleSize;
}
}
/**
* 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)
|| outByteStep < getSampleSize(formatType)) {
throw new IllegalArgumentException("invalid output index: "
+ "output.length=" + output.length + " outByteOffset="
+ outByteOffset + " outByteStep=" + outByteStep
+ " 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_24_3SB:
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_24_3SL:
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_24_4SB:
// TODO: verify
iSample = quantize24(input[inIndex] * twoPower23, ditherBits);
output[outIndex + 0] = 0;
output[outIndex + 1] = (byte) (iSample >> 16);
output[outIndex + 2] = (byte) ((iSample >>> 8) & 0xFF);
output[outIndex + 3] = (byte) (iSample & 0xFF);
break;
case CT_24_4SL:
// TODO: verify
iSample = quantize24(input[inIndex] * twoPower23, ditherBits);
output[outIndex + 3] = (byte) (iSample >> 16);
output[outIndex + 2] = (byte) ((iSample >>> 8) & 0xFF);
output[outIndex + 1] = (byte) (iSample & 0xFF);
output[outIndex + 0] = 0;
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));
}
}
}
}