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/*
* Copyright 2003-2016 The Music Player Daemon Project
* http://www.musicpd.org
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License along
* with this program; if not, write to the Free Software Foundation, Inc.,
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
*/
#include "config.h"
#include "OpenALOutputPlugin.hxx"
#include "../OutputAPI.hxx"
#include "../Wrapper.hxx"
#include "util/RuntimeError.hxx"
#include <unistd.h>
#ifndef __APPLE__
#include <AL/al.h>
#include <AL/alc.h>
#else
#include <OpenAL/al.h>
#include <OpenAL/alc.h>
#endif
class OpenALOutput {
friend struct AudioOutputWrapper<OpenALOutput>;
/* should be enough for buffer size = 2048 */
static constexpr unsigned NUM_BUFFERS = 16;
AudioOutput base;
const char *device_name;
ALCdevice *device;
ALCcontext *context;
ALuint buffers[NUM_BUFFERS];
unsigned filled;
ALuint source;
ALenum format;
ALuint frequency;
OpenALOutput(const ConfigBlock &block);
static OpenALOutput *Create(const ConfigBlock &block);
void Open(AudioFormat &audio_format);
void Close();
gcc_pure
std::chrono::steady_clock::duration Delay() const {
return filled < NUM_BUFFERS || HasProcessed()
? std::chrono::steady_clock::duration::zero()
/* we don't know exactly how long we must wait
for the next buffer to finish, so this is a
random guess: */
: std::chrono::milliseconds(50);
}
size_t Play(const void *chunk, size_t size);
void Cancel();
private:
gcc_pure
ALint GetSourceI(ALenum param) const {
ALint value;
alGetSourcei(source, param, &value);
return value;
}
gcc_pure
bool HasProcessed() const {
return GetSourceI(AL_BUFFERS_PROCESSED) > 0;
}
gcc_pure
bool IsPlaying() const {
return GetSourceI(AL_SOURCE_STATE) == AL_PLAYING;
}
/**
* Throws #std::runtime_error on error.
*/
void SetupContext();
};
static ALenum
openal_audio_format(AudioFormat &audio_format)
{
/* note: cannot map SampleFormat::S8 to AL_FORMAT_STEREO8 or
AL_FORMAT_MONO8 since OpenAL expects unsigned 8 bit
samples, while MPD uses signed samples */
switch (audio_format.format) {
case SampleFormat::S16:
if (audio_format.channels == 2)
return AL_FORMAT_STEREO16;
if (audio_format.channels == 1)
return AL_FORMAT_MONO16;
/* fall back to mono */
audio_format.channels = 1;
return openal_audio_format(audio_format);
default:
/* fall back to 16 bit */
audio_format.format = SampleFormat::S16;
return openal_audio_format(audio_format);
}
}
inline void
OpenALOutput::SetupContext()
{
device = alcOpenDevice(device_name);
if (device == nullptr)
throw FormatRuntimeError("Error opening OpenAL device \"%s\"",
device_name);
context = alcCreateContext(device, nullptr);
if (context == nullptr) {
alcCloseDevice(device);
throw FormatRuntimeError("Error creating context for \"%s\"",
device_name);
}
}
OpenALOutput::OpenALOutput(const ConfigBlock &block)
:base(openal_output_plugin, block),
device_name(block.GetBlockValue("device"))
{
if (device_name == nullptr)
device_name = alcGetString(nullptr,
ALC_DEFAULT_DEVICE_SPECIFIER);
}
inline OpenALOutput *
OpenALOutput::Create(const ConfigBlock &block)
{
return new OpenALOutput(block);
}
inline void
OpenALOutput::Open(AudioFormat &audio_format)
{
format = openal_audio_format(audio_format);
SetupContext();
alcMakeContextCurrent(context);
alGenBuffers(NUM_BUFFERS, buffers);
if (alGetError() != AL_NO_ERROR)
throw std::runtime_error("Failed to generate buffers");
alGenSources(1, &source);
if (alGetError() != AL_NO_ERROR) {
alDeleteBuffers(NUM_BUFFERS, buffers);
throw std::runtime_error("Failed to generate source");
}
filled = 0;
frequency = audio_format.sample_rate;
}
inline void
OpenALOutput::Close()
{
alcMakeContextCurrent(context);
alDeleteSources(1, &source);
alDeleteBuffers(NUM_BUFFERS, buffers);
alcDestroyContext(context);
alcCloseDevice(device);
}
inline size_t
OpenALOutput::Play(const void *chunk, size_t size)
{
if (alcGetCurrentContext() != context)
alcMakeContextCurrent(context);
ALuint buffer;
if (filled < NUM_BUFFERS) {
/* fill all buffers */
buffer = buffers[filled];
filled++;
} else {
/* wait for processed buffer */
while (!HasProcessed())
usleep(10);
alSourceUnqueueBuffers(source, 1, &buffer);
}
alBufferData(buffer, format, chunk, size, frequency);
alSourceQueueBuffers(source, 1, &buffer);
if (!IsPlaying())
alSourcePlay(source);
return size;
}
inline void
OpenALOutput::Cancel()
{
filled = 0;
alcMakeContextCurrent(context);
alSourceStop(source);
/* force-unqueue all buffers */
alSourcei(source, AL_BUFFER, 0);
filled = 0;
}
typedef AudioOutputWrapper<OpenALOutput> Wrapper;
const struct AudioOutputPlugin openal_output_plugin = {
"openal",
nullptr,
&Wrapper::Init,
&Wrapper::Finish,
nullptr,
nullptr,
&Wrapper::Open,
&Wrapper::Close,
&Wrapper::Delay,
nullptr,
&Wrapper::Play,
nullptr,
&Wrapper::Cancel,
nullptr,
nullptr,
};
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