floatsamplebuffer.java

linux下建立JAVA虚拟机的源码KAFFE

/*
 * FloatSampleBuffer.java
 */

/*
 *  Copyright (c) 2000 by Florian Bomers <[e-mail]>
 *
 *   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.
 */

package	org.tritonus.share.sampled;

import java.util.ArrayList;
import java.util.Random;

import javax.sound.sampled.AudioFormat;

/**
 * A class for small buffers of samples in linear, 32-bit
 * floating point format. 
 * <p>
 * It is supposed to be a replacement of the byte[] stream
 * architecture of JavaSound, especially for chains of
 * AudioInputStreams. Ideally, all involved AudioInputStreams
 * handle reading into a FloatSampleBuffer. 
 * <p>
 * Specifications:
 * <ol>
 * <li>Channels are separated, i.e. for stereo there are 2 float arrays
 *     with the samples for the left and right channel
 * <li>All data is handled in samples, where one sample means
 *     one float value in each channel
 * <li>All samples are normalized to the interval [-1.0...1.0]
 * </ol>
 * <p>
 * When a cascade of AudioInputStreams use FloatSampleBuffer for
 * processing, they may implement the interface FloatSampleInput.
 * This signals that this stream may provide float buffers
 * for reading. The data is <i>not</i> converted back to bytes,
 * but stays in a single buffer that is passed from stream to stream.
 * For that serves the read(FloatSampleBuffer) method, which is
 * then used as replacement for the byte-based read functions of
 * AudioInputStream.<br>
 * However, backwards compatibility must always be retained, so
 * even when an AudioInputStream implements FloatSampleInput,
 * it must work the same way when any of the byte-based read methods
 * is called.<br>
 * As an example, consider the following set-up:<br>
 * <ul>
 * <li>auAIS is an AudioInputStream (AIS) that reads from an AU file
 *     in 8bit pcm at 8000Hz. It does not implement FloatSampleInput.
 * <li>pcmAIS1 is an AIS that reads from auAIS and converts the data 
 *     to PCM 16bit. This stream implements FloatSampleInput, i.e. it 
 *     can generate float audio data from the ulaw samples.
 * <li>pcmAIS2 reads from pcmAIS1 and adds a reverb.
 *     It operates entirely on floating point samples.
 * <li>The method that reads from pcmAIS2 (i.e. AudioSystem.write) does 
 *     not handle floating point samples. 
 * </ul>
 * So, what happens when a block of samples is read from pcmAIS2 ?
 * <ol>
 * <li>the read(byte[]) method of pcmAIS2 is called
 * <li>pcmAIS2 always operates on floating point samples, so
 *     it uses an own instance of FloatSampleBuffer and initializes
 *     it with the number of samples requested in the read(byte[])
 *     method.
 * <li>It queries pcmAIS1 for the FloatSampleInput interface. As it
 *     implements it, pcmAIS2 calls the read(FloatSampleBuffer) method
 *     of pcmAIS1.
 * <li>pcmAIS1 notes that its underlying stream does not support floats,
 *     so it instantiates a byte buffer which can hold the number of
 *     samples of the FloatSampleBuffer passed to it. It calls the
 *     read(byte[]) method of auAIS.
 * <li>auAIS fills the buffer with the bytes.
 * <li>pcmAIS1 calls the <code>initFromByteArray</code> method of
 *     the float buffer to initialize it with the 8 bit data.
 * <li>Then pcmAIS1 processes the data: as the float buffer is
 *     normalized, it does nothing with the buffer - and returns
 *     control to pcmAIS2. The SampleSizeInBits field of the
 *     AudioFormat of pcmAIS1 defines that it should be 16 bits.
 * <li>pcmAIS2 receives the filled buffer from pcmAIS1 and does
 *     its processing on the buffer - it adds the reverb.
 * <li>As pcmAIS2's read(byte[]) method had been called, pcmAIS2
 *     calls the <code>convertToByteArray</code> method of
 *     the float buffer to fill the byte buffer with the
 *     resulting samples.
 * </ol>
 * <p>
 * To summarize, here are some advantages when using a FloatSampleBuffer 
 * for streaming:
 * <ul>
 * <li>no conversions from/to bytes need to be done during processing
 * <li>the sample size in bits is irrelevant - normalized range
 * <li>higher quality for processing
 * <li>separated channels (easy process/remove/add channels)
 * <li>potentially less copying of audio data, as processing
 * of the float samples is generally done in-place. The same
 * instance of a FloatSampleBuffer may be used from the data source
 * to the final data sink.
 * </ul>
 * <p>
 * Simple benchmarks showed that the processing needs
 * for the conversion to and from float is about the same as
 * when converting it to shorts or ints without dithering, 
 * and significantly higher with dithering. An own implementation 
 * of a random number generator may improve this.
 * <p>
 * &quot;Lazy&quot; deletion of samples and channels:<br>
 * <ul>
 * <li>When the sample count is reduced, the arrays are not resized, but
 * only the member variable <code>sampleCount</code> is reduced. A subsequent
 * increase of the sample count (which will occur frequently), will check
 * that and eventually reuse the existing array.
 * <li>When a channel is deleted, it is not removed from memory but only
 * hidden. Subsequent insertions of a channel will check whether a hidden channel
 * can be reused.
 * </ul>
 * The lazy mechanism can save many array instantiation (and copy-) operations
 * for the sake of performance. All relevant methods exist in a second
 * version which allows explicitely to disable lazy deletion.
 * <p>
 * Use the <code>reset</code> functions to clear the memory and remove 
 * hidden samples and channels.
 * <p>
 * Note that the lazy mechanism implies that the arrays returned
 * from <code>getChannel(int)</code> may have a greater size
 * than getSampleCount(). Consequently, be sure to never rely on the 
 * length field of the sample arrays.
 * <p>
 * As an example, consider a chain of converters that all act
 * on the same instance of FloatSampleBuffer. Some converters
 * may decrease the sample count (e.g. sample rate converter) and
 * delete channels (e.g. PCM2PCM converter). So, processing of one
 * block will decrease both. For the next block, all starts
 * from the beginning. With the lazy mechanism, all float arrays
 * are only created once for processing all blocks.<br>
 * Having lazy disabled would require for each chunk that is processed
 * <ol>
 * <li>new instantiation of all channel arrays
 * at the converter chain beginning as they have been
 * either deleted or decreased in size during processing of the 
 * previous chunk, and
 * <li>re-instantiation of all channel arrays for
 * the reduction of the sample count.
 * </ol>
 * <p>
 * Dithering:<br>
 * By default, this class uses dithering for reduction 
 * of sample width (e.g. original data was 16bit, target 
 * data is 8bit). As dithering may be needed in other cases 
 * (especially when the float samples are processed using DSP
 * algorithms), or it is preferred to switch it off,
 * dithering can be explicitely switched on or off with
 * the method setDitherMode(int).<br>
 * For a discussion about dithering, see
 * <a href="http://www.iqsoft.com/IQSMagazine/BobsSoapbox/Dithering.htm">
 * here</a> and 
 * <a href="http://www.iqsoft.com/IQSMagazine/BobsSoapbox/Dithering2.htm">
 * here</a>.
 *
 * @author Florian Bomers
 */

public class FloatSampleBuffer {

	/** Whether the functions without lazy parameter are lazy or not. */
	private static final boolean LAZY_DEFAULT=true;

	private ArrayList channels=new ArrayList(); // contains for each channel a float array
	private int sampleCount=0;
	private int channelCount=0;
	private float sampleRate=0;
	private int originalFormatType=0;

	/** Constant for setDitherMode: dithering will be enabled if sample size is decreased */
	public static final int DITHER_MODE_AUTOMATIC=0;
	/** Constant for setDitherMode: dithering will be done */
	public static final int DITHER_MODE_ON=1;
	/** Constant for setDitherMode: dithering will not be done */
	public static final int DITHER_MODE_OFF=2;

	private static Random random=null;
	private float ditherBits=0.8f;
	private boolean doDither=false; // set in convertFloatToBytes
	// e.g. the sample rate converter may want to force dithering
	private int ditherMode=DITHER_MODE_AUTOMATIC;

	// sample width (must be in order !)
	private static final int F_8=1;
	private static final int F_16=2;
	private static final int F_24=3;
	private static final int F_32=4;
	private static final int F_SAMPLE_WIDTH_MASK=F_8 | F_16 | F_24 | F_32;
	// format bit-flags
	private static final int F_SIGNED=8;
	private static final int F_BIGENDIAN=16;

	// supported formats
	private static final int CT_8S=F_8 | F_SIGNED;
	private static final int CT_8U=F_8;
	private static final int CT_16SB=F_16 | F_SIGNED | F_BIGENDIAN;
	private static final int CT_16SL=F_16 | F_SIGNED;
	private static final int CT_24SB=F_24 | F_SIGNED | F_BIGENDIAN;
	private static final int CT_24SL=F_24 | F_SIGNED;
	private static final int CT_32SB=F_32 | F_SIGNED | F_BIGENDIAN;
	private static final int CT_32SL=F_32 | F_SIGNED;

	//////////////////////////////// initialization /////////////////////////////////

	public FloatSampleBuffer() {
		this(0,0,1);
	}

	public FloatSampleBuffer(int channelCount, int sampleCount, float sampleRate) {
		init(channelCount, sampleCount, sampleRate, LAZY_DEFAULT);
	}

	public FloatSampleBuffer(byte[] buffer, int offset, int byteCount,
	                         AudioFormat format) {
		this(format.getChannels(),
		     byteCount/(format.getSampleSizeInBits()/8*format.getChannels()),
		     format.getSampleRate());
		initFromByteArray(buffer, offset, byteCount, format);
	}

	protected void init(int channelCount, int sampleCount, float sampleRate) {
		init(channelCount, sampleCount, sampleRate, LAZY_DEFAULT);
	}

	protected void init(int channelCount, int sampleCount, float sampleRate, boolean lazy) {
		if (channelCount<0 || sampleCount<0) {
			throw new IllegalArgumentException(
			    "Invalid parameters in initialization of FloatSampleBuffer.");
		}
		setSampleRate(sampleRate);
		if (getSampleCount()!=sampleCount || getChannelCount()!=channelCount) {
			createChannels(channelCount, sampleCount, lazy);
		}
	}

	private void createChannels(int channelCount, int sampleCount, boolean lazy) {
		this.sampleCount=sampleCount;
		// lazy delete of all channels. Intentionally lazy !
		this.channelCount=0;
		for (int ch=0; ch<channelCount; ch++) {
			insertChannel(ch, false, lazy);
		}
		if (!lazy) {
			// remove hidden channels
			while (channels.size()>channelCount) {
				channels.remove(channels.size()-1);
			}
		}
	}


	public void initFromByteArray(byte[] buffer, int offset, int byteCount,
	                              AudioFormat format) {
		initFromByteArray(buffer, offset, byteCount, format, LAZY_DEFAULT);
	}

	public void initFromByteArray(byte[] buffer, int offset, int byteCount,
	                              AudioFormat format, boolean lazy) {
		if (offset+byteCount>buffer.length) {
			throw new IllegalArgumentException
			("FloatSampleBuffer.initFromByteArray: buffer too small.");
		}
		boolean signed=format.getEncoding().equals(AudioFormat.Encoding.PCM_SIGNED);
		if (!signed &&
		        !format.getEncoding().equals(AudioFormat.Encoding.PCM_UNSIGNED)) {
			throw new IllegalArgumentException
			("FloatSampleBuffer: only PCM samples are possible.");
		}
		int bytesPerSample=format.getSampleSizeInBits()/8;
		int bytesPerFrame=bytesPerSample*format.getChannels();
		int thisSampleCount=byteCount/bytesPerFrame;
		init(format.getChannels(), thisSampleCount, format.getSampleRate(), lazy);
		int formatType=getFormatType(format.getSampleSizeInBits(),
		                             signed, format.isBigEndian());
		// save format for automatic dithering mode
		originalFormatType=formatType;
		for (int ch=0; ch<format.getChannels(); ch++) {
			convertByteToFloat(buffer, offset, bytesPerFrame, formatType, 
			                   getChannel(ch), 0, sampleCount);
			offset+=bytesPerSample; // next channel
		}
	}

	public void initFromFloatSampleBuffer(FloatSampleBuffer source) {
		init(source.getChannelCount(), source.getSampleCount(), source.getSampleRate());
		for (int ch=0; ch<getChannelCount(); ch++) {
			System.arraycopy(source.getChannel(ch), 0, getChannel(ch), 0, sampleCount);
		}
	}

	/**
	 * deletes all channels, frees memory...
	 * This also removes hidden channels by lazy remove.
	 */
	public void reset() {
		init(0,0,1, false);
	}

	/**
	 * destroys any existing data and creates new channels.
	 * It also destroys lazy removed channels and samples.
	 */
	public void reset(int channels, int sampleCount, float sampleRate) {
		init(channels, sampleCount, sampleRate, false);
	}

	//////////////////////////////// conversion back to bytes /////////////////////////////////

	/**
	 * returns the required size of the buffer
	 * when convertToByteArray(..) is called