/*******************************************************************************
* Copyright 2011 See AUTHORS file.
*
* Licensed under the Apache License, Version 2.0 (the "License");
* you may not use this file except in compliance with the License.
* You may obtain a copy of the License at
*
* http://www.apache.org/licenses/LICENSE-2.0
*
* Unless required by applicable law or agreed to in writing, software
* distributed under the License is distributed on an "AS IS" BASIS,
* WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
* See the License for the specific language governing permissions and
* limitations under the License.
******************************************************************************/
package com.badlogic.gdx.backends.lwjgl.audio;
import java.nio.ByteBuffer;
import java.nio.IntBuffer;
import org.lwjgl.BufferUtils;
import org.lwjgl.openal.AL11;
import com.badlogic.gdx.audio.AudioDevice;
import com.badlogic.gdx.math.MathUtils;
import com.badlogic.gdx.utils.GdxRuntimeException;
import static org.lwjgl.openal.AL10.*;
/** @author Nathan Sweet */
public class OpenALAudioDevice implements AudioDevice {
static private final int bytesPerSample = 2;
private final OpenALAudio audio;
private final int channels;
private IntBuffer buffers;
private int sourceID = -1;
private int format, sampleRate;
private boolean isPlaying;
private float volume = 1;
private float renderedSeconds, secondsPerBuffer;
private byte[] bytes;
private final int bufferSize;
private final int bufferCount;
private final ByteBuffer tempBuffer;
public OpenALAudioDevice (OpenALAudio audio, int sampleRate, boolean isMono, int bufferSize, int bufferCount) {
this.audio = audio;
channels = isMono ? 1 : 2;
this.bufferSize = bufferSize;
this.bufferCount = bufferCount;
this.format = channels > 1 ? AL_FORMAT_STEREO16 : AL_FORMAT_MONO16;
this.sampleRate = sampleRate;
secondsPerBuffer = (float)bufferSize / bytesPerSample / channels / sampleRate;
tempBuffer = BufferUtils.createByteBuffer(bufferSize);
}
public void writeSamples (short[] samples, int offset, int numSamples) {
if (bytes == null || bytes.length < numSamples * 2) bytes = new byte[numSamples * 2];
int end = Math.min(offset + numSamples, samples.length);
for (int i = offset, ii = 0; i < end; i++) {
short sample = samples[i];
bytes[ii++] = (byte)(sample & 0xFF);
bytes[ii++] = (byte)((sample >> 8) & 0xFF);
}
writeSamples(bytes, 0, numSamples * 2);
}
public void writeSamples (float[] samples, int offset, int numSamples) {
if (bytes == null || bytes.length < numSamples * 2) bytes = new byte[numSamples * 2];
int end = Math.min(offset + numSamples, samples.length);
for (int i = offset, ii = 0; i < end; i++) {
float floatSample = samples[i];
floatSample = MathUtils.clamp(floatSample, -1f, 1f);
int intSample = (int)(floatSample * 32767);
bytes[ii++] = (byte)(intSample & 0xFF);
bytes[ii++] = (byte)((intSample >> 8) & 0xFF);
}
writeSamples(bytes, 0, numSamples * 2);
}
public void writeSamples (byte[] data, int offset, int length) {
if (length < 0) throw new IllegalArgumentException("length cannot be < 0.");
if (sourceID == -1) {
sourceID = audio.obtainSource(true);
if (sourceID == -1) return;
if (buffers == null) {
buffers = BufferUtils.createIntBuffer(bufferCount);
alGenBuffers(buffers);
if (alGetError() != AL_NO_ERROR) throw new GdxRuntimeException("Unabe to allocate audio buffers.");
}
alSourcei(sourceID, AL_LOOPING, AL_FALSE);
alSourcef(sourceID, AL_GAIN, volume);
// Fill initial buffers.
int queuedBuffers = 0;
for (int i = 0; i < bufferCount; i++) {
int bufferID = buffers.get(i);
int written = Math.min(bufferSize, length);
tempBuffer.clear();
tempBuffer.put(data, offset, written).flip();
alBufferData(bufferID, format, tempBuffer, sampleRate);
alSourceQueueBuffers(sourceID, bufferID);
length -= written;
offset += written;
queuedBuffers++;
}
// Queue rest of buffers, empty.
tempBuffer.clear().flip();
for (int i = queuedBuffers; i < bufferCount; i++) {
int bufferID = buffers.get(i);
alBufferData(bufferID, format, tempBuffer, sampleRate);
alSourceQueueBuffers(sourceID, bufferID);
}
alSourcePlay(sourceID);
isPlaying = true;
}
while (length > 0) {
int written = fillBuffer(data, offset, length);
length -= written;
offset += written;
}
}
/** Blocks until some of the data could be buffered. */
private int fillBuffer (byte[] data, int offset, int length) {
int written = Math.min(bufferSize, length);
outer:
while (true) {
int buffers = alGetSourcei(sourceID, AL_BUFFERS_PROCESSED);
while (buffers-- > 0) {
int bufferID = alSourceUnqueueBuffers(sourceID);
if (bufferID == AL_INVALID_VALUE) break;
renderedSeconds += secondsPerBuffer;
tempBuffer.clear();
tempBuffer.put(data, offset, written).flip();
alBufferData(bufferID, format, tempBuffer, sampleRate);
alSourceQueueBuffers(sourceID, bufferID);
break outer;
}
// Wait for buffer to be free.
try {
Thread.sleep((long)(1000 * secondsPerBuffer / bufferCount));
} catch (InterruptedException ignored) {
}
}
// A buffer underflow will cause the source to stop.
if (!isPlaying || alGetSourcei(sourceID, AL_SOURCE_STATE) != AL_PLAYING) {
alSourcePlay(sourceID);
isPlaying = true;
}
return written;
}
public void stop () {
if (sourceID == -1) return;
audio.freeSource(sourceID);
sourceID = -1;
renderedSeconds = 0;
isPlaying = false;
}
public boolean isPlaying () {
if (sourceID == -1) return false;
return isPlaying;
}
public void setVolume (float volume) {
this.volume = volume;
if (sourceID != -1) alSourcef(sourceID, AL_GAIN, volume);
}
public float getPosition () {
if (sourceID == -1) return 0;
return renderedSeconds + alGetSourcef(sourceID, AL11.AL_SEC_OFFSET);
}
public void setPosition (float position) {
renderedSeconds = position;
}
public int getChannels () {
return format == AL_FORMAT_STEREO16 ? 2 : 1;
}
public int getRate () {
return sampleRate;
}
public void dispose () {
if (buffers == null) return;
if (sourceID != -1) {
audio.freeSource(sourceID);
sourceID = -1;
}
alDeleteBuffers(buffers);
buffers = null;
}
public boolean isMono () {
return channels == 1;
}
public int getLatency () {
return (int)(secondsPerBuffer * bufferCount * 1000);
}
}