mirror of
https://github.com/love2d/love.git
synced 2026-08-12 08:30:56 +02:00
498 lines
9.8 KiB
C++
498 lines
9.8 KiB
C++
/**
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* Copyright (c) 2006-2011 LOVE Development Team
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*
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* This software is provided 'as-is', without any express or implied
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* warranty. In no event will the authors be held liable for any damages
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* arising from the use of this software.
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*
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* Permission is granted to anyone to use this software for any purpose,
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* including commercial applications, and to alter it and redistribute it
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* freely, subject to the following restrictions:
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*
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* 1. The origin of this software must not be misrepresented; you must not
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* claim that you wrote the original software. If you use this software
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* in a product, an acknowledgment in the product documentation would be
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* appreciated but is not required.
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* 2. Altered source versions must be plainly marked as such, and must not be
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* misrepresented as being the original software.
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* 3. This notice may not be removed or altered from any source distribution.
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**/
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#include "Source.h"
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#include "Pool.h"
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// STD
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#include <iostream>
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namespace love
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{
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namespace audio
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{
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namespace openal
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{
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Source::Source(Pool * pool, love::sound::SoundData * soundData)
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: love::audio::Source(Source::TYPE_STATIC), pool(pool), valid(false),
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pitch(1.0f), volume(1.0f), looping(false), offsetSamples(0), offsetSeconds(0), decoder(0)
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{
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alGenBuffers(1, buffers);
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ALenum fmt = getFormat(soundData->getChannels(), soundData->getBits());
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alBufferData(buffers[0], fmt, soundData->getData(), soundData->getSize(), soundData->getSampleRate());
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static float z[3] = {0, 0, 0};
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setFloatv(position, z);
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setFloatv(velocity, z);
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setFloatv(direction, z);
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}
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Source::Source(Pool * pool, love::sound::Decoder * decoder)
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: love::audio::Source(Source::TYPE_STREAM), pool(pool), valid(false),
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pitch(1.0f), volume(1.0f), looping(false), offsetSamples(0), offsetSeconds(0), decoder(decoder)
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{
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decoder->retain();
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alGenBuffers(MAX_BUFFERS, buffers);
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static float z[3] = {0, 0, 0};
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setFloatv(position, z);
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setFloatv(velocity, z);
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setFloatv(direction, z);
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}
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Source::~Source()
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{
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stop();
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alDeleteBuffers((type == TYPE_STATIC) ? 1 : MAX_BUFFERS, buffers);
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if (decoder)
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decoder->release();
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}
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love::audio::Source * Source::copy()
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{
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return 0;
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}
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void Source::play()
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{
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valid = pool->play(this, source);
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if(valid)
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reset(source);
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}
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void Source::stop()
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{
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if (!isStopped())
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pool->stop(this);
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}
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void Source::pause()
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{
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pool->pause(this);
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}
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void Source::resume()
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{
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pool->resume(this);
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}
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void Source::rewind()
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{
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pool->rewind(this);
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}
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bool Source::isStopped() const
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{
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if(valid)
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{
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ALenum state;
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alGetSourcei(source, AL_SOURCE_STATE, &state);
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return (state == AL_STOPPED);
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}
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return true;
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}
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bool Source::isPaused() const
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{
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if(valid)
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{
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ALenum state;
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alGetSourcei(source, AL_SOURCE_STATE, &state);
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return (state == AL_PAUSED);
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}
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return false;
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}
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bool Source::isFinished() const
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{
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return type == TYPE_STATIC ? isStopped() : isStopped() && !isLooping() && decoder->isFinished();
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}
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void Source::update()
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{
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if(valid && type == TYPE_STATIC)
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{
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// Looping mode could have changed.
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alSourcei(source, AL_LOOPING, isLooping() ? AL_TRUE : AL_FALSE);
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}
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else if(valid && type == TYPE_STREAM && !(!isLooping() && isFinished()))
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{
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// Number of processed buffers.
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ALint processed;
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alGetSourcei(source, AL_BUFFERS_PROCESSED, &processed);
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while(processed--)
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{
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ALuint buffer;
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float curOffsetSamples, curOffsetSecs;
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alGetSourcef(source, AL_SAMPLE_OFFSET, &curOffsetSamples);
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ALint b;
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alGetSourcei(source, AL_BUFFER, &b);
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int freq;
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alGetBufferi(b, AL_FREQUENCY, &freq);
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curOffsetSecs = curOffsetSamples / freq;
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// Get a free buffer.
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alSourceUnqueueBuffers(source, 1, &buffer);
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float newOffsetSamples, newOffsetSecs;
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alGetSourcef(source, AL_SAMPLE_OFFSET, &newOffsetSamples);
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newOffsetSecs = newOffsetSamples / freq;
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offsetSamples += (curOffsetSamples - newOffsetSamples);
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offsetSeconds += (curOffsetSecs - newOffsetSecs);
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streamAtomic(buffer, decoder);
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alSourceQueueBuffers(source, 1, &buffer);
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}
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}
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}
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void Source::setPitch(float pitch)
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{
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if(valid)
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alSourcef(source, AL_PITCH, pitch);
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this->pitch = pitch;
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}
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float Source::getPitch() const
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{
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if(valid)
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{
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ALfloat f;
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alGetSourcef(source, AL_PITCH, &f);
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return f;
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}
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// In case the Source isn't playing.
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return pitch;
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}
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void Source::setVolume(float volume)
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{
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if(valid)
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{
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alSourcef(source, AL_GAIN, volume);
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}
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this->volume = volume;
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}
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float Source::getVolume() const
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{
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if(valid)
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{
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ALfloat f;
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alGetSourcef(source, AL_GAIN, &f);
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return f;
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}
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// In case the Source isn't playing.
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return volume;
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}
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void Source::seekAtomic(float offset, void * unit)
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{
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if (valid)
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{
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switch (*((Source::Unit*) unit)) {
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case Source::UNIT_SAMPLES:
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if (type == TYPE_STREAM) {
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ALint buffer;
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alGetSourcei(source, AL_BUFFER, &buffer);
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int freq;
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alGetBufferi(buffer, AL_FREQUENCY, &freq);
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offset /= freq;
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decoder->seek(offset);
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} else {
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alSourcef(source, AL_SAMPLE_OFFSET, offset);
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}
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break;
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case Source::UNIT_SECONDS:
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default:
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if (type == TYPE_STREAM) {
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decoder->seek(offset);
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} else {
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alSourcef(source, AL_SEC_OFFSET, offset);
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}
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break;
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}
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}
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}
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void Source::seek(float offset, Source::Unit unit)
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{
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return pool->seek(this, offset, &unit);
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}
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float Source::tellAtomic(void * unit) const
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{
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if (valid)
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{
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float offset;
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switch (*((Source::Unit*) unit)) {
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case Source::UNIT_SAMPLES:
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alGetSourcef(source, AL_SAMPLE_OFFSET, &offset);
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if (type == TYPE_STREAM) offset += offsetSamples;
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break;
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case Source::UNIT_SECONDS:
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default:
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alGetSourcef(source, AL_SAMPLE_OFFSET, &offset);
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ALint buffer;
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alGetSourcei(source, AL_BUFFER, &buffer);
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int freq;
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alGetBufferi(buffer, AL_FREQUENCY, &freq);
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offset /= freq;
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if (type == TYPE_STREAM) offset += offsetSeconds;
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break;
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}
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return offset;
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}
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return 0.0f;
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}
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float Source::tell(Source::Unit unit)
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{
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return pool->tell(this, &unit);
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}
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void Source::setPosition(float * v)
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{
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if(valid)
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alSourcefv(source, AL_POSITION, v);
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setFloatv(position, v);
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}
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void Source::getPosition(float * v) const
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{
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if(valid)
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alGetSourcefv(source, AL_POSITION, v);
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else
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setFloatv(v, position);
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}
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void Source::setVelocity(float * v)
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{
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if(valid)
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alSourcefv(source, AL_VELOCITY, v);
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setFloatv(velocity, v);
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}
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void Source::getVelocity(float * v) const
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{
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if(valid)
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alGetSourcefv(source, AL_VELOCITY, v);
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else
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setFloatv(v, velocity);
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}
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void Source::setDirection(float * v)
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{
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if(valid)
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alSourcefv(source, AL_DIRECTION, v);
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else
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setFloatv(direction, v);
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}
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void Source::getDirection(float * v) const
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{
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if(valid)
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alGetSourcefv(source, AL_DIRECTION, v);
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else
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setFloatv(v, direction);
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}
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void Source::setLooping(bool looping)
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{
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if(valid && type == TYPE_STATIC)
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alSourcei(source, AL_LOOPING, looping ? AL_TRUE : AL_FALSE);
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this->looping = looping;
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}
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bool Source::isLooping() const
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{
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return looping;
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}
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void Source::playAtomic()
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{
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if(type == TYPE_STATIC)
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{
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alSourcei(source, AL_BUFFER, buffers[0]);
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}
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else if(type == TYPE_STREAM)
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{
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int usedBuffers = 0;
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for(unsigned int i = 0; i < MAX_BUFFERS; i++)
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{
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int decoded = streamAtomic(buffers[i], decoder);
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++usedBuffers;
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if(decoder->isFinished())
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break;
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}
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if(usedBuffers > 0)
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alSourceQueueBuffers(source, usedBuffers, buffers);
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}
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// Set these properties. These may have changed while we've
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// been without an AL source.
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alSourcef(source, AL_PITCH, pitch);
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alSourcef(source, AL_GAIN, volume);
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alSourcePlay(source);
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valid = true; //if it fails it will be set to false again
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//but this prevents a horrible, horrible bug
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}
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void Source::stopAtomic()
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{
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if(valid)
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{
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if(type == TYPE_STATIC)
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{
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alSourceStop(source);
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}
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else if(type == TYPE_STREAM)
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{
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alSourceStop(source);
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int queued = 0;
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alGetSourcei(source, AL_BUFFERS_QUEUED, &queued);
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while(queued--)
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{
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ALuint buffer;
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alSourceUnqueueBuffers(source, 1, &buffer);
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}
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}
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alSourcei(source, AL_BUFFER, AL_NONE);
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}
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rewindAtomic();
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valid = false;
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}
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void Source::pauseAtomic()
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{
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if(valid)
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{
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alSourcePause(source);
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}
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}
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void Source::resumeAtomic()
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{
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if(valid)
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{
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alSourcePlay(source);
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}
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}
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void Source::rewindAtomic()
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{
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if(valid && type == TYPE_STATIC)
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{
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alSourceRewind(source);
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}
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else if(valid && type == TYPE_STREAM)
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{
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decoder->rewind();
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offsetSamples = 0;
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offsetSeconds = 0;
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}
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}
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void Source::reset(ALenum source)
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{
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alSourcefv(source, AL_POSITION, position);
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alSourcefv(source, AL_VELOCITY, velocity);
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alSourcefv(source, AL_DIRECTION, direction);
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alSourcef(source, AL_PITCH, pitch);
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alSourcef(source, AL_GAIN, volume);
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}
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void Source::setFloatv(float * dst, const float * src) const
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{
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dst[0] = src[0];
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dst[1] = src[1];
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dst[2] = src[2];
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}
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ALenum Source::getFormat(int channels, int bits) const
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{
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if(channels == 1 && bits == 8)
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return AL_FORMAT_MONO8;
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else if(channels == 1 && bits == 16)
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return AL_FORMAT_MONO16;
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else if(channels == 2 && bits == 8)
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return AL_FORMAT_STEREO8;
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else if(channels == 2 && bits == 16)
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return AL_FORMAT_STEREO16;
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else
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return 0;
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}
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int Source::streamAtomic(ALuint buffer, love::sound::Decoder * d)
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{
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// Get more sound data.
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int decoded = d->decode();
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int fmt = getFormat(d->getChannels(), d->getBits());
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if(fmt != 0)
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alBufferData(buffer, fmt, d->getBuffer(), decoded, d->getSampleRate());
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if(decoder->isFinished() && isLooping()) {
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offsetSamples = 0;
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offsetSeconds = 0;
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d->rewind();
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}
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return decoded;
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}
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bool Source::isStatic() const
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{
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return (type == TYPE_STATIC);
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}
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} // openal
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} // audio
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} // love
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