diff options
author | Sven Gothel <[email protected]> | 2013-07-19 05:08:34 +0200 |
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committer | Sven Gothel <[email protected]> | 2013-07-19 05:08:34 +0200 |
commit | da7210c6498b6fcc2fbf829684ea399a6b4c3f65 (patch) | |
tree | 8acf7efc5006cfeabc8e3f453c344ff3b4ddc456 /src/jogl/classes/com/jogamp/opengl/util/av/AudioSink.java | |
parent | 24ba0ac528603ee527f002d350bcfb19754f2457 (diff) |
FFMPEGPlayer Audio Sink Refactoring ..
- AudioSink w/ AudioFrame and formats public
- ALAudioSink uses a circular buffer now, hence relaxes the one-threaded player mode
- FFMPEGMediaPlayer uses multiple audio frames (equal to the ALAudioSink number)
and wraps data to NIO buffer w/o copy.
- FFMPEGMediaPlayer audio threading currently disabled: distorted sound
Seems that the ALAudioSink's circular buffer usage is good enough for now.
- Verbosity only w/ DEBUG flag
- New SyncedRingbuffer for effcient synced buffering
Diffstat (limited to 'src/jogl/classes/com/jogamp/opengl/util/av/AudioSink.java')
-rw-r--r-- | src/jogl/classes/com/jogamp/opengl/util/av/AudioSink.java | 168 |
1 files changed, 168 insertions, 0 deletions
diff --git a/src/jogl/classes/com/jogamp/opengl/util/av/AudioSink.java b/src/jogl/classes/com/jogamp/opengl/util/av/AudioSink.java new file mode 100644 index 000000000..5caeb969a --- /dev/null +++ b/src/jogl/classes/com/jogamp/opengl/util/av/AudioSink.java @@ -0,0 +1,168 @@ +/** + * 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 jogamp.opengl.Debug; + +public interface AudioSink { + public static final boolean DEBUG = Debug.debug("AudioSink"); + + /** Specifies the audio data type. Currently only PCM is supported. */ + public static enum AudioDataType { PCM }; + + /** + * Specifies the audio data format. + */ + public static class AudioDataFormat { + public AudioDataFormat(AudioDataType dataType, int sampleRate, int sampleSize, int channelCount, boolean signed, boolean littleEndian) { + this.dataType = dataType; + this.sampleRate = sampleRate; + this.sampleSize = sampleSize; + this.channelCount = channelCount; + this.signed = signed; + this.littleEndian = littleEndian; + } + /** Audio data type. */ + public final AudioDataType dataType; + /** 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; + public final boolean littleEndian; + + public String toString() { + return "AudioDataFormat[type "+dataType+", sampleRate "+sampleRate+", sampleSize "+sampleSize+", channelCount "+channelCount+ + ", signed "+signed+", "+(littleEndian?"little":"big")+"endian]"; } + } + /** Default {@link AudioDataFormat}, [type PCM, sampleRate 44100, sampleSize 16, channelCount 2, signed, littleEndian]. */ + public static final AudioDataFormat DefaultFormat = new AudioDataFormat(AudioDataType.PCM, 44100, 16, 2, true /* signed */, true /* littleEndian */); + + public static class AudioFrame { + public final ByteBuffer data; + public final int dataSize; + public final int audioPTS; + + public AudioFrame(ByteBuffer data, int dataSize, int audioPTS) { + if( dataSize > data.remaining() ) { + throw new IllegalArgumentException("Give size "+dataSize+" exceeds remaining bytes in ls "+data+". "+this); + } + this.data=data; + this.dataSize=dataSize; + this.audioPTS=audioPTS; + } + + public String toString() { return "AudioFrame[apts "+audioPTS+", data "+data+", payloadSize "+dataSize+"]"; } + } + + /** + * 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 preferred {@link AudioDataFormat} by this sink. + * <p> + * The preferred format shall reflect this sinks most native format, + * i.e. best performance w/o data conversion. + * </p> + * @see #initSink(AudioDataFormat) + */ + public AudioDataFormat getPreferredFormat(); + + /** + * Initializes the sink. + * <p> + * Implementation shall try to match the given <code>requestedFormat</code> {@link AudioDataFormat} + * as close as possible, regarding it's capabilities. + * </p> + * <p> + * A user may consider {@link #getPreferredFormat()} and pass this value + * to utilize best performance and <i>behavior</i>. + * </p> + * The {@link #DefaultFormat} <i>should be</i> supported by all implementations. + * </p> + * @param requestedFormat the requested {@link AudioDataFormat}. + * @param bufferCount number of buffers for sink + * @return if successful the chosen AudioDataFormat based on the <code>requestedFormat</code> and this sinks capabilities, otherwise <code>null</code>. + */ + public AudioDataFormat initSink(AudioDataFormat requestedFormat, int bufferCount); + + + /** Destroys this instance, i.e. closes all streams and devices allocated. */ + public void destroy(); + + /** + * Returns the number of bytes queued for playing. + * <p> + * {@link #initSink(AudioDataFormat)} must be called first. + * </p> + */ + public int getQueuedByteCount(); + + /** + * Returns the queued buffer time in milliseconds for playing. + * <p> + * {@link #initSink(AudioDataFormat)} must be called first. + * </p> + */ + public int getQueuedTime(); + + /** + * Returns the number of buffers in the sink available for writing. + * <p> + * {@link #initSink(AudioDataFormat)} must be called first. + * </p> + */ + public int getWritableBufferCount(); + + /** + * Returns true if data is available to be written in the sink. + * <p> + * {@link #initSink(AudioDataFormat)} must be called first. + * </p> + */ + public boolean isDataAvailable(int data_size); + + /** + * Writes the remaining bytes of the given direct ByteBuffer to this sink. + * <p> + * The data must comply with the chosen {@link AudioDataFormat} as returned by {@link #initSink(AudioDataFormat)}. + * </p> + */ + public void writeData(AudioFrame audioFrame); +} |