diff options
Diffstat (limited to 'src/jogl/classes')
4 files changed, 247 insertions, 92 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 index 7f477a57d..8751fc816 100644 --- a/src/jogl/classes/com/jogamp/opengl/util/av/AudioSink.java +++ b/src/jogl/classes/com/jogamp/opengl/util/av/AudioSink.java @@ -39,33 +39,46 @@ public interface 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(AudioDataFormat, float, int, int, int)}.*/ + /** 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(AudioDataFormat, float, int, int, int)}.*/ + /** 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(AudioDataFormat, float, int, int, int)}.*/ + /** 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(AudioDataFormat, float, int, int, int)}.*/ + /** 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 audio data type. Currently only PCM is supported. */ - public static enum AudioDataType { PCM }; - /** - * Specifies the audio data format. + * Specifies the linear audio PCM format. */ - public static class AudioDataFormat { - public AudioDataFormat(AudioDataType dataType, int sampleRate, int sampleSize, int channelCount, boolean signed, boolean fixedP, boolean littleEndian) { - this.dataType = dataType; + 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(int sampleRate, int sampleSize, int channelCount, boolean signed, boolean fixedP, boolean planar, 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"); + } + } } - /** Audio data type. */ - public final AudioDataType dataType; + /** Sample rate in Hz (1/s). */ public final int sampleRate; /** Sample size in bits. */ @@ -73,15 +86,25 @@ public interface AudioSink { /** 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, */ + /** 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}. + * according to {@link #sampleSize}, {@link #channelCount} and {@link #sampleRate}. + * <p> + * Time -> Byte Count + * </p> */ - public final int getByteSize(int millisecs) { + public final int getDurationsByteSize(int millisecs) { final int bytesPerSample = sampleSize >>> 3; // /8 return millisecs * ( channelCount * bytesPerSample * ( sampleRate / 1000 ) ); } @@ -89,6 +112,9 @@ public interface AudioSink { /** * 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(int byteCount) { final int bytesPerSample = sampleSize >>> 3; // /8 @@ -96,11 +122,14 @@ public interface AudioSink { } /** - * Returns the duration in milliseconds of the given and sample count per frame and channel + * 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(int sampleCount) { @@ -116,6 +145,9 @@ public interface AudioSink { * 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. */ @@ -130,21 +162,44 @@ public interface AudioSink { * 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}. + * 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 getSamplesByteSize(int sampleCount) { + public final int getSamplesByteCount(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 sampleCount sample count + */ + public final int getBytesSampleCount(int byteCount) { + return ( byteCount << 3 ) / sampleSize; + } + public String toString() { - return "AudioDataFormat[type "+dataType+", sampleRate "+sampleRate+", sampleSize "+sampleSize+", channelCount "+channelCount+ - ", signed "+signed+", fixedP "+fixedP+", "+(littleEndian?"little":"big")+"endian]"; } + return "AudioDataFormat[sampleRate "+sampleRate+", sampleSize "+sampleSize+", channelCount "+channelCount+ + ", signed "+signed+", fixedP "+fixedP+", "+(planar?"planar":"packed")+", "+(littleEndian?"little":"big")+"-endian]"; } } - /** Default {@link AudioDataFormat}, [type PCM, sampleRate 44100, sampleSize 16, channelCount 2, signed, fixedP, littleEndian]. */ - public static final AudioDataFormat DefaultFormat = new AudioDataFormat(AudioDataType.PCM, 44100, 16, 2, true /* signed */, true /* fixed point */, true /* littleEndian */); + /** 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; @@ -227,38 +282,54 @@ public interface AudioSink { public boolean setVolume(float v); /** - * Returns the preferred {@link AudioDataFormat} by this sink. + * Returns the preferred {@link AudioFormat} by this sink. * <p> - * The preferred format shall reflect this sinks most native format, + * 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> - * @see #initSink(AudioDataFormat) + * <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 AudioDataFormat getPreferredFormat(); + public boolean isSupported(AudioFormat format); /** * Initializes the sink. * <p> - * Implementation shall try to match the given <code>requestedFormat</code> {@link AudioDataFormat} - * as close as possible, regarding it's capabilities. + * Implementation must match the given <code>requestedFormat</code> {@link AudioFormat}. * </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. + * 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 AudioDataFormat}. + * @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 if successful the chosen AudioDataFormat based on the <code>requestedFormat</code> and this sinks capabilities, otherwise <code>null</code>. + * @return true if successful, otherwise false */ - public AudioDataFormat init(AudioDataFormat requestedFormat, float frameDuration, - int initialQueueSize, int queueGrowAmount, int queueLimit); + public boolean init(AudioFormat requestedFormat, float frameDuration, + int initialQueueSize, int queueGrowAmount, int queueLimit); /** * Returns true, if {@link #play()} has been requested <i>and</i> the sink is still playing, @@ -285,7 +356,7 @@ public interface AudioSink { /** * Flush all queued buffers, implies {@link #pause()}. * <p> - * {@link #init(AudioDataFormat, float, int, int, int)} must be called first. + * {@link #init(AudioFormat, float, int, int, int)} must be called first. * </p> * @see #play() * @see #pause() @@ -298,17 +369,17 @@ public interface AudioSink { /** * Returns the number of allocated buffers as requested by - * {@link #init(AudioDataFormat, float, int, int, int)}. + * {@link #init(AudioFormat, float, int, int, int)}. */ public int getFrameCount(); - /** @return the current enqueued frames count since {@link #init(AudioDataFormat, float, int, int, int)}. */ + /** @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(AudioDataFormat, float, int, int, int)} must be called first. + * {@link #init(AudioFormat, float, int, int, int)} must be called first. * </p> */ public int getQueuedFrameCount(); @@ -316,7 +387,7 @@ public interface AudioSink { /** * Returns the current number of bytes queued for playing. * <p> - * {@link #init(AudioDataFormat, float, int, int, int)} must be called first. + * {@link #init(AudioFormat, float, int, int, int)} must be called first. * </p> */ public int getQueuedByteCount(); @@ -324,7 +395,7 @@ public interface AudioSink { /** * Returns the current queued frame time in milliseconds for playing. * <p> - * {@link #init(AudioDataFormat, float, int, int, int)} must be called first. + * {@link #init(AudioFormat, float, int, int, int)} must be called first. * </p> */ public int getQueuedTime(); @@ -337,7 +408,7 @@ public interface AudioSink { /** * Returns the current number of frames in the sink available for writing. * <p> - * {@link #init(AudioDataFormat, float, int, int, int)} must be called first. + * {@link #init(AudioFormat, float, int, int, int)} must be called first. * </p> */ public int getFreeFrameCount(); @@ -345,10 +416,10 @@ public interface AudioSink { /** * Enqueue the remaining bytes of the given {@link AudioDataFrame}'s direct ByteBuffer to this sink. * <p> - * The data must comply with the chosen {@link AudioDataFormat} as returned by {@link #initSink(AudioDataFormat)}. + * The data must comply with the chosen {@link AudioFormat} as returned by {@link #initSink(AudioFormat)}. * </p> * <p> - * {@link #init(AudioDataFormat, float, int, int, int)} must be called first. + * {@link #init(AudioFormat, float, int, int, int)} must be called first. * </p> * @returns the enqueued internal {@link AudioFrame}, which may differ from the input <code>audioDataFrame</code>. * @deprecated User shall use {@link #enqueueData(int, ByteBuffer, int)}, which allows implementation @@ -359,10 +430,10 @@ public interface AudioSink { /** * 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 AudioDataFormat} as returned by {@link #initSink(AudioDataFormat)}. + * The data must comply with the chosen {@link AudioFormat} as returned by {@link #initSink(AudioFormat)}. * </p> * <p> - * {@link #init(AudioDataFormat, float, int, int, int)} must be called first. + * {@link #init(AudioFormat, float, int, int, int)} must be called first. * </p> * @returns the enqueued internal {@link AudioFrame}. */ diff --git a/src/jogl/classes/jogamp/opengl/openal/av/ALAudioSink.java b/src/jogl/classes/jogamp/opengl/openal/av/ALAudioSink.java index d3964a2cf..3daedd52e 100644 --- a/src/jogl/classes/jogamp/opengl/openal/av/ALAudioSink.java +++ b/src/jogl/classes/jogamp/opengl/openal/av/ALAudioSink.java @@ -39,7 +39,9 @@ import com.jogamp.openal.AL; import com.jogamp.openal.ALC; import com.jogamp.openal.ALCcontext; import com.jogamp.openal.ALCdevice; +import com.jogamp.openal.ALExt; import com.jogamp.openal.ALFactory; +import com.jogamp.openal.util.ALHelpers; import com.jogamp.opengl.util.av.AudioSink; /*** @@ -47,12 +49,16 @@ import com.jogamp.opengl.util.av.AudioSink; */ public class ALAudioSink implements AudioSink { + private static final String AL_SOFT_buffer_samples = "AL_SOFT_buffer_samples"; private static final ALC alc; private static final AL al; + private static final ALExt alExt; private static final boolean staticAvailable; private String deviceSpecifier; private ALCdevice device; + private boolean hasSOFTBufferSamples; + private AudioFormat preferredAudioFormat; private ALCcontext context; private final RecursiveLock lock = LockFactory.createRecursiveLock(); @@ -91,8 +97,10 @@ public class ALAudioSink implements AudioSink { private volatile int enqueuedFrameCount; private int[] alSource = null; - private AudioDataFormat chosenFormat; - private int alFormat; + private AudioFormat chosenFormat; + private int alChannelLayout; + private int alSampleType; + private int alFormat; private boolean initialized; private volatile boolean playRequested = false; @@ -100,9 +108,11 @@ public class ALAudioSink implements AudioSink { static { ALC _alc = null; AL _al = null; + ALExt _alExt = null; try { _alc = ALFactory.getALC(); _al = ALFactory.getAL(); + _alExt = ALFactory.getALExt(); } catch(Throwable t) { if( DEBUG ) { System.err.println("ALAudioSink: Catched "+t.getClass().getName()+": "+t.getMessage()); @@ -111,7 +121,8 @@ public class ALAudioSink implements AudioSink { } alc = _alc; al = _al; - staticAvailable = null != alc && null != al; + alExt = _alExt; + staticAvailable = null != alc && null != al && null != alExt; } public ALAudioSink() { @@ -138,7 +149,7 @@ public class ALAudioSink implements AudioSink { // Create audio context. context = alc.alcCreateContext(device, null); if (context == null) { - throw new RuntimeException("ALAudioSink: Error creating OpenAL context"); + throw new RuntimeException("ALAudioSink: Error creating OpenAL context for "+deviceSpecifier); } lockContext(); @@ -148,6 +159,16 @@ public class ALAudioSink implements AudioSink { throw new RuntimeException("ALAudioSink: Error making OpenAL context current"); } + hasSOFTBufferSamples = al.alIsExtensionPresent(AL_SOFT_buffer_samples); + preferredAudioFormat = queryPreferredAudioFormat(); + if( DEBUG | true ) { + System.out.println("ALAudioSink: OpenAL Extensions:"+al.alGetString(AL.AL_EXTENSIONS)); + System.out.println("ALAudioSink: Null device OpenAL Extensions:"+alc.alcGetString(null, ALC.ALC_EXTENSIONS)); + System.out.println("ALAudioSink: Device "+deviceSpecifier+" OpenAL Extensions:"+alc.alcGetString(device, ALC.ALC_EXTENSIONS)); + System.out.println("ALAudioSink: hasSOFTBufferSamples "+hasSOFTBufferSamples); + System.out.println("ALAudioSink: preferredAudioFormat "+preferredAudioFormat); + } + // Create source { alSource = new int[1]; @@ -175,6 +196,16 @@ public class ALAudioSink implements AudioSink { } } + private final AudioFormat queryPreferredAudioFormat() { + int sampleRate = DefaultFormat.sampleRate; + final int[] value = new int[1]; + alc.alcGetIntegerv(device, ALC.ALC_FREQUENCY, 1, value, 0); + if ( alc.alcGetError(device) == ALC.ALC_NO_ERROR ) { + sampleRate = value[0]; + } + return new AudioFormat(sampleRate, DefaultFormat.sampleSize, DefaultFormat.channelCount, DefaultFormat.signed, DefaultFormat.fixedP, DefaultFormat.planar, DefaultFormat.littleEndian); + } + private final void lockContext() { lock.lock(); alc.alcMakeContextCurrent(context); @@ -212,53 +243,78 @@ public class ALAudioSink implements AudioSink { final int alBuffersLen = null != alBufferNames ? alBufferNames.length : 0; final int ctxHash = context != null ? context.hashCode() : 0; return "ALAudioSink[init "+initialized+", playRequested "+playRequested+", device "+deviceSpecifier+", ctx "+toHexString(ctxHash)+", alSource "+alSrcName+ - ", chosen "+chosenFormat+", alFormat "+toHexString(alFormat)+ - ", playSpeed "+playSpeed+", buffers[total "+alBuffersLen+", avail "+alFramesAvail.size()+", "+ + ", chosen "+chosenFormat+ + ", al[chan "+ALHelpers.alChannelLayoutName(alChannelLayout)+", type "+ALHelpers.alSampleTypeName(alSampleType)+ + ", fmt "+toHexString(alFormat)+", soft "+hasSOFTBufferSamples+ + "], playSpeed "+playSpeed+", buffers[total "+alBuffersLen+", avail "+alFramesAvail.size()+", "+ "queued["+alFramesPlaying.size()+", apts "+getPTS()+", "+getQueuedTime() + " ms, " + alBufferBytesQueued+" bytes], "+ "queue[g "+frameGrowAmount+", l "+frameLimit+"]"; } + public final String getPerfString() { final int alBuffersLen = null != alBufferNames ? alBufferNames.length : 0; return "Play [buffer "+alFramesPlaying.size()+"/"+alBuffersLen+", apts "+getPTS()+", "+getQueuedTime() + " ms, " + alBufferBytesQueued+" bytes]"; } @Override - public final AudioDataFormat getPreferredFormat() { - return DefaultFormat; + public final AudioFormat getPreferredFormat() { + if( !staticAvailable ) { + return null; + } + return preferredAudioFormat; + } + + @Override + public final int getMaxSupportedChannels() { + if( !staticAvailable ) { + return 0; + } + return hasSOFTBufferSamples ? 8 : 2; } @Override - public final AudioDataFormat init(AudioDataFormat requestedFormat, float frameDuration, int initialQueueSize, int queueGrowAmount, int queueLimit) { + public final boolean isSupported(AudioFormat format) { if( !staticAvailable ) { - return null; + return false; } - if( !requestedFormat.fixedP || - !requestedFormat.littleEndian || - ( 1 != requestedFormat.channelCount && requestedFormat.channelCount != 2 ) || - ( 8 != requestedFormat.sampleSize && requestedFormat.sampleSize != 16 ) - ) { - return null; // not supported w/ OpenAL - } - // final float samplePeriod = 1.0f / requestedFormat.sampleRate; - switch( requestedFormat.channelCount ) { - case 1: { - switch ( requestedFormat.sampleSize ) { - case 8: - alFormat = AL.AL_FORMAT_MONO8; break; - case 16: - alFormat = AL.AL_FORMAT_MONO16; break; - } - } break; - case 2: - switch ( requestedFormat.sampleSize ) { - case 8: - alFormat = AL.AL_FORMAT_STEREO8; break; - case 16: - alFormat = AL.AL_FORMAT_STEREO16; break; + if( format.planar || !format.littleEndian ) { + // FIXME big-endian supported w/ SOFT where it's native format! + return false; + } + final int alChannelLayout = ALHelpers.getDefaultALChannelLayout(format.channelCount); + if( AL.AL_NONE != alChannelLayout ) { + final int alSampleType = ALHelpers.getALSampleType(format.sampleSize, format.signed, format.fixedP); + if( AL.AL_NONE != alSampleType ) { + lockContext(); + try { + final int alFormat = ALHelpers.getALFormat(alChannelLayout, alSampleType, hasSOFTBufferSamples, al, alExt); + return AL.AL_NONE != alFormat; + } finally { + unlockContext(); } + } } + return false; + } + + @Override + public final boolean init(AudioFormat requestedFormat, float frameDuration, int initialQueueSize, int queueGrowAmount, int queueLimit) { + if( !staticAvailable ) { + return false; + } + alChannelLayout = ALHelpers.getDefaultALChannelLayout(requestedFormat.channelCount); + alSampleType = ALHelpers.getALSampleType(requestedFormat.sampleSize, requestedFormat.signed, requestedFormat.fixedP); lockContext(); try { + if( AL.AL_NONE != alChannelLayout && AL.AL_NONE != alSampleType ) { + alFormat = ALHelpers.getALFormat(alChannelLayout, alSampleType, hasSOFTBufferSamples, al, alExt); + } else { + alFormat = AL.AL_NONE; + } + if( AL.AL_NONE == alFormat ) { + // not supported + return false; + } // Allocate buffers destroyBuffers(); { @@ -290,7 +346,7 @@ public class ALAudioSink implements AudioSink { } chosenFormat = requestedFormat; - return chosenFormat; + return true; } private static int[] concat(int[] first, int[] second) { @@ -548,7 +604,15 @@ public class ALAudioSink implements AudioSink { if( !alFramesPlaying.put( alFrame ) ) { throw new InternalError("Internal Error: "+this); } - al.alBufferData(alFrame.alBuffer, alFormat, bytes, byteCount, chosenFormat.sampleRate); + if( hasSOFTBufferSamples ) { + final int samplesPerChannel = chosenFormat.getBytesSampleCount(byteCount) / chosenFormat.channelCount; + // final int samplesPerChannel = ALHelpers.bytesToSampleCount(byteCount, alChannelLayout, alSampleType); + alExt.alBufferSamplesSOFT(alFrame.alBuffer, chosenFormat.sampleRate, alFormat, + samplesPerChannel, alChannelLayout, alSampleType, bytes); + } else { + al.alBufferData(alFrame.alBuffer, alFormat, bytes, byteCount, chosenFormat.sampleRate); + } + final int[] alBufferNames = new int[] { alFrame.alBuffer }; al.alSourceQueueBuffers(alSource[0], 1, alBufferNames, 0); alErr = al.alGetError(); diff --git a/src/jogl/classes/jogamp/opengl/util/av/JavaSoundAudioSink.java b/src/jogl/classes/jogamp/opengl/util/av/JavaSoundAudioSink.java index b1b9c1b6c..85fab96a4 100644 --- a/src/jogl/classes/jogamp/opengl/util/av/JavaSoundAudioSink.java +++ b/src/jogl/classes/jogamp/opengl/util/av/JavaSoundAudioSink.java @@ -33,7 +33,7 @@ public class JavaSoundAudioSink implements AudioSink { private int bufferCount; private byte [] sampleData = new byte[BUFFER_SIZE]; private boolean initialized = false; - private AudioDataFormat chosenFormat = null; + private AudioSink.AudioFormat chosenFormat = null; private volatile boolean playRequested = false; private float volume = 1.0f; @@ -77,14 +77,24 @@ public class JavaSoundAudioSink implements AudioSink { } @Override - public AudioDataFormat getPreferredFormat() { + public AudioSink.AudioFormat getPreferredFormat() { return DefaultFormat; } @Override - public AudioDataFormat init(AudioDataFormat requestedFormat, float frameDuration, int initialQueueSize, int queueGrowAmount, int queueLimit) { + public final int getMaxSupportedChannels() { + return 2; + } + + @Override + public final boolean isSupported(AudioSink.AudioFormat format) { + return true; + } + + @Override + public boolean init(AudioSink.AudioFormat requestedFormat, float frameDuration, int initialQueueSize, int queueGrowAmount, int queueLimit) { if( !staticAvailable ) { - return null; + return false; } // Create the audio format we wish to use format = new javax.sound.sampled.AudioFormat(requestedFormat.sampleRate, requestedFormat.sampleSize, requestedFormat.channelCount, requestedFormat.signed, !requestedFormat.littleEndian); @@ -105,7 +115,7 @@ public class JavaSoundAudioSink implements AudioSink { } catch (Exception e) { initialized=false; } - return chosenFormat; + return true; } @Override diff --git a/src/jogl/classes/jogamp/opengl/util/av/NullAudioSink.java b/src/jogl/classes/jogamp/opengl/util/av/NullAudioSink.java index 83799b074..723bb9dd1 100644 --- a/src/jogl/classes/jogamp/opengl/util/av/NullAudioSink.java +++ b/src/jogl/classes/jogamp/opengl/util/av/NullAudioSink.java @@ -42,13 +42,23 @@ public class NullAudioSink implements AudioSink { } @Override - public AudioDataFormat getPreferredFormat() { + public AudioFormat getPreferredFormat() { return DefaultFormat; } @Override - public AudioDataFormat init(AudioDataFormat requestedFormat, float frameDuration, int initialQueueSize, int queueGrowAmount, int queueLimit) { - return requestedFormat; + public final int getMaxSupportedChannels() { + return 8; + } + + @Override + public final boolean isSupported(AudioFormat format) { + return true; + } + + @Override + public boolean init(AudioFormat requestedFormat, float frameDuration, int initialQueueSize, int queueGrowAmount, int queueLimit) { + return true; } @Override |