Files
mpd/src/output/OpenALOutputPlugin.cxx
Lukas Stabe 0ea5f4ac3a output/openal: check __APPLE__ instead of HAVE_OSX
On OSX, the configure-flag --enable-osx is used to enable the
unsupported osx output. It sets the HAVE_OSX preprocessor define.

src/output/OpenALOutputPlugin.cxx uses this define to determine wether
it is building on OSX, and imports different headers (which have
nothing to do with the osx output) depending on wether or not it is
set.
2013-12-02 13:08:52 +01:00

286 lines
6.2 KiB
C++

/*
* Copyright (C) 2003-2013 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 "util/Error.hxx"
#include "util/Domain.hxx"
#include <glib.h>
#ifndef __APPLE__
#include <AL/al.h>
#include <AL/alc.h>
#else
#include <OpenAL/al.h>
#include <OpenAL/alc.h>
#endif
/* should be enough for buffer size = 2048 */
#define NUM_BUFFERS 16
struct OpenALOutput {
struct audio_output base;
const char *device_name;
ALCdevice *device;
ALCcontext *context;
ALuint buffers[NUM_BUFFERS];
unsigned filled;
ALuint source;
ALenum format;
ALuint frequency;
bool Initialize(const config_param &param, Error &error_r) {
return ao_base_init(&base, &openal_output_plugin, param,
error_r);
}
void Deinitialize() {
ao_base_finish(&base);
}
};
static constexpr Domain openal_output_domain("openal_output");
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);
}
}
gcc_pure
static inline ALint
openal_get_source_i(const OpenALOutput *od, ALenum param)
{
ALint value;
alGetSourcei(od->source, param, &value);
return value;
}
gcc_pure
static inline bool
openal_has_processed(const OpenALOutput *od)
{
return openal_get_source_i(od, AL_BUFFERS_PROCESSED) > 0;
}
gcc_pure
static inline ALint
openal_is_playing(const OpenALOutput *od)
{
return openal_get_source_i(od, AL_SOURCE_STATE) == AL_PLAYING;
}
static bool
openal_setup_context(OpenALOutput *od, Error &error)
{
od->device = alcOpenDevice(od->device_name);
if (od->device == nullptr) {
error.Format(openal_output_domain,
"Error opening OpenAL device \"%s\"",
od->device_name);
return false;
}
od->context = alcCreateContext(od->device, nullptr);
if (od->context == nullptr) {
error.Format(openal_output_domain,
"Error creating context for \"%s\"",
od->device_name);
alcCloseDevice(od->device);
return false;
}
return true;
}
static struct audio_output *
openal_init(const config_param &param, Error &error)
{
const char *device_name = param.GetBlockValue("device");
if (device_name == nullptr) {
device_name = alcGetString(nullptr, ALC_DEFAULT_DEVICE_SPECIFIER);
}
OpenALOutput *od = new OpenALOutput();
if (!od->Initialize(param, error)) {
delete od;
return nullptr;
}
od->device_name = device_name;
return &od->base;
}
static void
openal_finish(struct audio_output *ao)
{
OpenALOutput *od = (OpenALOutput *)ao;
od->Deinitialize();
delete od;
}
static bool
openal_open(struct audio_output *ao, AudioFormat &audio_format,
Error &error)
{
OpenALOutput *od = (OpenALOutput *)ao;
od->format = openal_audio_format(audio_format);
if (!openal_setup_context(od, error)) {
return false;
}
alcMakeContextCurrent(od->context);
alGenBuffers(NUM_BUFFERS, od->buffers);
if (alGetError() != AL_NO_ERROR) {
error.Set(openal_output_domain, "Failed to generate buffers");
return false;
}
alGenSources(1, &od->source);
if (alGetError() != AL_NO_ERROR) {
error.Set(openal_output_domain, "Failed to generate source");
alDeleteBuffers(NUM_BUFFERS, od->buffers);
return false;
}
od->filled = 0;
od->frequency = audio_format.sample_rate;
return true;
}
static void
openal_close(struct audio_output *ao)
{
OpenALOutput *od = (OpenALOutput *)ao;
alcMakeContextCurrent(od->context);
alDeleteSources(1, &od->source);
alDeleteBuffers(NUM_BUFFERS, od->buffers);
alcDestroyContext(od->context);
alcCloseDevice(od->device);
}
static unsigned
openal_delay(struct audio_output *ao)
{
OpenALOutput *od = (OpenALOutput *)ao;
return od->filled < NUM_BUFFERS || openal_has_processed(od)
? 0
/* we don't know exactly how long we must wait for the
next buffer to finish, so this is a random
guess: */
: 50;
}
static size_t
openal_play(struct audio_output *ao, const void *chunk, size_t size,
gcc_unused Error &error)
{
OpenALOutput *od = (OpenALOutput *)ao;
ALuint buffer;
if (alcGetCurrentContext() != od->context) {
alcMakeContextCurrent(od->context);
}
if (od->filled < NUM_BUFFERS) {
/* fill all buffers */
buffer = od->buffers[od->filled];
od->filled++;
} else {
/* wait for processed buffer */
while (!openal_has_processed(od))
g_usleep(10);
alSourceUnqueueBuffers(od->source, 1, &buffer);
}
alBufferData(buffer, od->format, chunk, size, od->frequency);
alSourceQueueBuffers(od->source, 1, &buffer);
if (!openal_is_playing(od))
alSourcePlay(od->source);
return size;
}
static void
openal_cancel(struct audio_output *ao)
{
OpenALOutput *od = (OpenALOutput *)ao;
od->filled = 0;
alcMakeContextCurrent(od->context);
alSourceStop(od->source);
/* force-unqueue all buffers */
alSourcei(od->source, AL_BUFFER, 0);
od->filled = 0;
}
const struct audio_output_plugin openal_output_plugin = {
"openal",
nullptr,
openal_init,
openal_finish,
nullptr,
nullptr,
openal_open,
openal_close,
openal_delay,
nullptr,
openal_play,
nullptr,
openal_cancel,
nullptr,
nullptr,
};