/**
* Copyright 2013 JogAmp Community. All rights reserved.
*
* Redistribution and use in source and binary forms, with or without modification, are
* permitted provided that the following conditions are met:
*
* 1. Redistributions of source code must retain the above copyright notice, this list of
* conditions and the following disclaimer.
*
* 2. Redistributions in binary form must reproduce the above copyright notice, this list
* of conditions and the following disclaimer in the documentation and/or other materials
* provided with the distribution.
*
* THIS SOFTWARE IS PROVIDED BY JogAmp Community ``AS IS'' AND ANY EXPRESS OR IMPLIED
* WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND
* FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL JogAmp Community OR
* CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
* CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR
* SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON
* ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
* NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*
* The views and conclusions contained in the software and documentation are those of the
* authors and should not be interpreted as representing official policies, either expressed
* or implied, of JogAmp Community.
*/
package com.jogamp.opengl.util.av;
import java.nio.ByteBuffer;
import com.jogamp.opengl.util.TimeFrameI;
import jogamp.opengl.Debug;
public interface AudioSink {
public static final boolean DEBUG = Debug.debug("AudioSink");
/** Default frame duration in millisecond, i.e. 1 frame per {@value} ms. */
public static final int DefaultFrameDuration = 32;
/** Initial audio queue size in milliseconds. {@value} ms, i.e. 16 frames per 32 ms. See {@link #init(AudioFormat, float, int, int, int)}.*/
public static final int DefaultInitialQueueSize = 16 * 32; // 512 ms
/** Audio queue grow size in milliseconds. {@value} ms, i.e. 16 frames per 32 ms. See {@link #init(AudioFormat, float, int, int, int)}.*/
public static final int DefaultQueueGrowAmount = 16 * 32; // 512 ms
/** Audio queue limit w/ video in milliseconds. {@value} ms, i.e. 96 frames per 32 ms. See {@link #init(AudioFormat, float, int, int, int)}.*/
public static final int DefaultQueueLimitWithVideo = 96 * 32; // 3072 ms
/** Audio queue limit w/o video in milliseconds. {@value} ms, i.e. 32 frames per 32 ms. See {@link #init(AudioFormat, float, int, int, int)}.*/
public static final int DefaultQueueLimitAudioOnly = 32 * 32; // 1024 ms
/**
* Specifies the linear audio PCM format.
*/
public static class AudioFormat {
/**
* @param sampleRate sample rate in Hz (1/s)
* @param sampleSize sample size in bits
* @param channelCount number of channels
* @param signed true if signed number, false for unsigned
* @param fixedP true for fixed point value, false for unsigned floating point value with a sampleSize of 32 (float) or 64 (double)
* @param planar true for planar data package (each channel in own data buffer), false for packed data channels interleaved in one buffer.
* @param littleEndian true for little-endian, false for big endian
*/
public AudioFormat(final int sampleRate, final int sampleSize, final int channelCount, final boolean signed, final boolean fixedP, final boolean planar, final boolean littleEndian) {
this.sampleRate = sampleRate;
this.sampleSize = sampleSize;
this.channelCount = channelCount;
this.signed = signed;
this.fixedP = fixedP;
this.planar = planar;
this.littleEndian = littleEndian;
if( !fixedP ) {
if( sampleSize != 32 && sampleSize != 64 ) {
throw new IllegalArgumentException("Floating point: sampleSize "+sampleSize+" bits");
}
if( !signed ) {
throw new IllegalArgumentException("Floating point: unsigned");
}
}
}
/** Sample rate in Hz (1/s). */
public final int sampleRate;
/** Sample size in bits. */
public final int sampleSize;
/** Number of channels. */
public final int channelCount;
public final boolean signed;
/** Fixed or floating point values. Floating point 'float' has {@link #sampleSize} 32, 'double' has {@link #sampleSize} 64. */
public final boolean fixedP;
/** Planar or packed samples. If planar, each channel has their own data buffer. If packed, channel data is interleaved in one buffer. */
public final boolean planar;
public final boolean littleEndian;
//
// Time <-> Bytes
//
/**
* Returns the byte size of the given milliseconds
* according to {@link #sampleSize}, {@link #channelCount} and {@link #sampleRate}.
* <p>
* Time -> Byte Count
* </p>
*/
public final int getDurationsByteSize(final int millisecs) {
final int bytesPerSample = sampleSize >>> 3; // /8
return millisecs * ( channelCount * bytesPerSample * ( sampleRate / 1000 ) );
}
/**
* Returns the duration in milliseconds of the given byte count
* according to {@link #sampleSize}, {@link #channelCount} and {@link #sampleRate}.
* <p>
* Byte Count -> Time
* </p>
*/
public final int getBytesDuration(final int byteCount) {
final int bytesPerSample = sampleSize >>> 3; // /8
return byteCount / ( channelCount * bytesPerSample * ( sampleRate / 1000 ) );
}
/**
* Returns the duration in milliseconds of the given sample count per frame and channel
* according to the {@link #sampleRate}, i.e.
* <pre>
* ( 1000f * sampleCount ) / sampleRate
* </pre>
* <p>
* Sample Count -> Time
* </p>
* @param sampleCount sample count per frame and channel
*/
public final float getSamplesDuration(final int sampleCount) {
return ( 1000f * sampleCount ) / sampleRate;
}
/**
* Returns the rounded frame count of the given milliseconds and frame duration.
* <pre>
* Math.max( 1, millisecs / frameDuration + 0.5f )
* </pre>
* <p>
* Note: <code>frameDuration</code> can be derived by <i>sample count per frame and channel</i>
* via {@link #getSamplesDuration(int)}.
* </p>
* <p>
* Frame Time -> Frame Count
* </p>
* @param millisecs time in milliseconds
* @param frameDuration duration per frame in milliseconds.
*/
public final int getFrameCount(final int millisecs, final float frameDuration) {
return Math.max(1, (int) ( millisecs / frameDuration + 0.5f ));
}
/**
* Returns the byte size of given sample count
* according to the {@link #sampleSize}, i.e.:
* <pre>
* sampleCount * ( sampleSize / 8 )
* </pre>
* <p>
* Note: To retrieve the byte size for all channels,
* you need to pre-multiply <code>sampleCount</code> with {@link #channelCount}.
* </p>
* <p>
* Sample Count -> Byte Count
* </p>
* @param sampleCount sample count
*/
public final int getSamplesByteCount(final int sampleCount) {
return sampleCount * ( sampleSize >>> 3 );
}
/**
* Returns the sample count of given byte count
* according to the {@link #sampleSize}, i.e.:
* <pre>
* ( byteCount * 8 ) / sampleSize
* </pre>
* <p>
* Note: If <code>byteCount</code> covers all channels and you request the sample size per channel,
* you need to divide the result by <code>sampleCount</code> by {@link #channelCount}.
* </p>
* <p>
* Byte Count -> Sample Count
* </p>
* @param byteCount number of bytes
*/
public final int getBytesSampleCount(final int byteCount) {
return ( byteCount << 3 ) / sampleSize;
}
@Override
public String toString() {
return "AudioDataFormat[sampleRate "+sampleRate+", sampleSize "+sampleSize+", channelCount "+channelCount+
", signed "+signed+", fixedP "+fixedP+", "+(planar?"planar":"packed")+", "+(littleEndian?"little":"big")+"-endian]"; }
}
/** Default {@link AudioFormat}, [type PCM, sampleRate 44100, sampleSize 16, channelCount 2, signed, fixedP, !planar, littleEndian]. */
public static final AudioFormat DefaultFormat = new AudioFormat(44100, 16, 2, true /* signed */,
true /* fixed point */, false /* planar */, true /* littleEndian */);
public static abstract class AudioFrame extends TimeFrameI {
protected int byteSize;
public AudioFrame() {
this.byteSize = 0;
}
public AudioFrame(final int pts, final int duration, final int byteCount) {
super(pts, duration);
this.byteSize=byteCount;
}
/** Get this frame's size in bytes. */
public final int getByteSize() { return byteSize; }
/** Set this frame's size in bytes. */
public final void setByteSize(final int size) { this.byteSize=size; }
@Override
public String toString() {
return "AudioFrame[pts " + pts + " ms, l " + duration + " ms, "+byteSize + " bytes]";
}
}
public static class AudioDataFrame extends AudioFrame {
protected final ByteBuffer data;
public AudioDataFrame(final int pts, final int duration, final ByteBuffer bytes, final int byteCount) {
super(pts, duration, byteCount);
if( byteCount > bytes.remaining() ) {
throw new IllegalArgumentException("Give size "+byteCount+" exceeds remaining bytes in ls "+bytes+". "+this);
}
this.data=bytes;
}
/** Get this frame's data. */
public final ByteBuffer getData() { return data; }
@Override
public String toString() {
return "AudioDataFrame[pts " + pts + " ms, l " + duration + " ms, "+byteSize + " bytes, " + data + "]";
}
}
/**
* Returns the <code>initialized state</code> of this instance.
* <p>
* The <code>initialized state</code> is affected by this instance
* overall availability, i.e. after instantiation,
* as well as by {@link #destroy()}.
* </p>
*/
public boolean isInitialized();
/** Returns the playback speed. */
public float getPlaySpeed();
/**
* Sets the playback speed.
* <p>
* To simplify test, play speed is <i>normalized</i>, i.e.
* <ul>
* <li><code>1.0f</code>: if <code> Math.abs(1.0f - rate) < 0.01f </code></li>
* </ul>
* </p>
* @return true if successful, otherwise false, i.e. due to unsupported value range of implementation.
*/
public boolean setPlaySpeed(float s);
/** Returns the volume. */
public float getVolume();
/**
* Sets the volume [0f..1f].
* <p>
* To simplify test, volume is <i>normalized</i>, i.e.
* <ul>
* <li><code>0.0f</code>: if <code> Math.abs(v) < 0.01f </code></li>
* <li><code>1.0f</code>: if <code> Math.abs(1.0f - v) < 0.01f </code></li>
* </ul>
* </p>
* @return true if successful, otherwise false, i.e. due to unsupported value range of implementation.
*/
public boolean setVolume(float v);
/**
* Returns the preferred {@link AudioFormat} by this sink.
* <p>
* The preferred format is guaranteed to be supported
* and shall reflect this sinks most native format,
* i.e. best performance w/o data conversion.
* </p>
* <p>
* Known {@link #AudioFormat} attributes considered by implementations:
* <ul>
* <li>ALAudioSink: {@link AudioFormat#sampleRate}.
* </ul>
* </p>
* @see #initSink(AudioFormat)
* @see #isSupported(AudioFormat)
*/
public AudioFormat getPreferredFormat();
/** Return the maximum number of supported channels. */
public int getMaxSupportedChannels();
/**
* Returns true if the given format is supported by the sink, otherwise false.
* @see #initSink(AudioFormat)
* @see #getPreferredFormat()
*/
public boolean isSupported(AudioFormat format);
/**
* Initializes the sink.
* <p>
* Implementation must match the given <code>requestedFormat</code> {@link AudioFormat}.
* </p>
* <p>
* Caller shall validate <code>requestedFormat</code> via {@link #isSupported(AudioFormat)}
* beforehand and try to find a suitable supported one.
* {@link #getPreferredFormat()} and {@link #getMaxSupportedChannels()} may help.
* </p>
* @param requestedFormat the requested {@link AudioFormat}.
* @param frameDuration average or fixed frame duration in milliseconds
* helping a caching {@link AudioFrame} based implementation to determine the frame count in the queue.
* See {@link #DefaultFrameDuration}.
* @param initialQueueSize initial time in milliseconds to queue in this sink, see {@link #DefaultInitialQueueSize}.
* @param queueGrowAmount time in milliseconds to grow queue if full, see {@link #DefaultQueueGrowAmount}.
* @param queueLimit maximum time in milliseconds the queue can hold (and grow), see {@link #DefaultQueueLimitWithVideo} and {@link #DefaultQueueLimitAudioOnly}.
* @return true if successful, otherwise false
*/
public boolean init(AudioFormat requestedFormat, float frameDuration,
int initialQueueSize, int queueGrowAmount, int queueLimit);
/**
* Returns the {@link AudioFormat} as chosen by {@link #init(AudioFormat, float, int, int, int)},
* i.e. it shall match the <i>requestedFormat</i>.
*/
public AudioFormat getChosenFormat();
/**
* Returns true, if {@link #play()} has been requested <i>and</i> the sink is still playing,
* otherwise false.
*/
public boolean isPlaying();
/**
* Play buffers queued via {@link #enqueueData(AudioFrame)} from current internal position.
* If no buffers are yet queued or the queue runs empty, playback is being continued when buffers are enqueued later on.
* @see #enqueueData(AudioFrame)
* @see #pause()
*/
public void play();
/**
* Pause playing buffers while keeping enqueued data incl. it's internal position.
* @see #play()
* @see #flush()
* @see #enqueueData(AudioFrame)
*/
public void pause();
/**
* Flush all queued buffers, implies {@link #pause()}.
* <p>
* {@link #init(AudioFormat, float, int, int, int)} must be called first.
* </p>
* @see #play()
* @see #pause()
* @see #enqueueData(AudioFrame)
*/
public void flush();
/** Destroys this instance, i.e. closes all streams and devices allocated. */
public void destroy();
/**
* Returns the number of allocated buffers as requested by
* {@link #init(AudioFormat, float, int, int, int)}.
*/
public int getFrameCount();
/** @return the current enqueued frames count since {@link #init(AudioFormat, float, int, int, int)}. */
public int getEnqueuedFrameCount();
/**
* Returns the current number of frames queued for playing.
* <p>
* {@link #init(AudioFormat, float, int, int, int)} must be called first.
* </p>
*/
public int getQueuedFrameCount();
/**
* Returns the current number of bytes queued for playing.
* <p>
* {@link #init(AudioFormat, float, int, int, int)} must be called first.
* </p>
*/
public int getQueuedByteCount();
/**
* Returns the current queued frame time in milliseconds for playing.
* <p>
* {@link #init(AudioFormat, float, int, int, int)} must be called first.
* </p>
*/
public int getQueuedTime();
/**
* Return the current audio presentation timestamp (PTS) in milliseconds.
*/
public int getPTS();
/**
* Returns the current number of frames in the sink available for writing.
* <p>
* {@link #init(AudioFormat, float, int, int, int)} must be called first.
* </p>
*/
public int getFreeFrameCount();
/**
* Enqueue <code>byteCount</code> bytes of the remaining bytes of the given NIO {@link ByteBuffer} to this sink.
* <p>
* The data must comply with the chosen {@link AudioFormat} as returned by {@link #initSink(AudioFormat)}.
* </p>
* <p>
* {@link #init(AudioFormat, float, int, int, int)} must be called first.
* </p>
* @returns the enqueued internal {@link AudioFrame}.
*/
public AudioFrame enqueueData(int pts, ByteBuffer bytes, int byteCount);
}