Commit 8b6a5d19 authored by Serge Ziryukin's avatar Serge Ziryukin Committed by Max Kellermann

openal output plugin

parent 129920e8
...@@ -590,6 +590,10 @@ OUTPUT_SRC += src/output/oss_plugin.c ...@@ -590,6 +590,10 @@ OUTPUT_SRC += src/output/oss_plugin.c
MIXER_SRC += src/mixer/oss_mixer.c MIXER_SRC += src/mixer/oss_mixer.c
endif endif
if HAVE_OPENAL
OUTPUT_SRC += src/output/openal_plugin.c
endif
if HAVE_OSX if HAVE_OSX
OUTPUT_SRC += src/output/osx_plugin.c OUTPUT_SRC += src/output/osx_plugin.c
endif endif
......
...@@ -711,6 +711,11 @@ AC_ARG_ENABLE(oss, ...@@ -711,6 +711,11 @@ AC_ARG_ENABLE(oss,
[disable OSS support (default: enable)]),, [disable OSS support (default: enable)]),,
enable_oss=yes) enable_oss=yes)
AC_ARG_ENABLE(openal,
AS_HELP_STRING([--enable-openal],
[enable OpenAL support (default: disable)]),,
enable_openal=no)
AC_ARG_ENABLE(pulse, AC_ARG_ENABLE(pulse,
AS_HELP_STRING([--enable-pulse], AS_HELP_STRING([--enable-pulse],
[enable support for the PulseAudio sound server]),, [enable support for the PulseAudio sound server]),,
...@@ -779,6 +784,19 @@ fi ...@@ -779,6 +784,19 @@ fi
AM_CONDITIONAL(HAVE_OSS, test x$enable_oss = xyes) AM_CONDITIONAL(HAVE_OSS, test x$enable_oss = xyes)
if test x$enable_openal = xyes; then
PKG_CHECK_MODULES([OPENAL], [openal],
AC_DEFINE(HAVE_OPENAL, 1, [Define for OpenAL support]),
enable_openal=no)
fi
if test x$enable_openal = xyes; then
MPD_CFLAGS="$MPD_CFLAGS $OPENAL_CFLAGS"
MPD_LIBS="$MPD_LIBS $OPENAL_LIBS"
fi
AM_CONDITIONAL(HAVE_OPENAL, test x$enable_openal = xyes)
if test x$enable_fifo = xyes; then if test x$enable_fifo = xyes; then
AC_CHECK_FUNC([mkfifo], AC_CHECK_FUNC([mkfifo],
[enable_fifo=yes;AC_DEFINE([HAVE_FIFO], 1, [enable_fifo=yes;AC_DEFINE([HAVE_FIFO], 1,
...@@ -1297,6 +1315,12 @@ else ...@@ -1297,6 +1315,12 @@ else
echo " OSS support ...................disabled" echo " OSS support ...................disabled"
fi fi
if test x$enable_openal = xyes; then
echo " OpenAL support ................enabled"
else
echo " OpenAL support ................disabled"
fi
if test x$enable_osx = xyes; then if test x$enable_osx = xyes; then
echo " OS X support ..................enabled" echo " OS X support ..................enabled"
else else
...@@ -1338,6 +1362,7 @@ echo "" ...@@ -1338,6 +1362,7 @@ echo ""
if if
test x$enable_ao = xno && test x$enable_ao = xno &&
test x$enable_oss = xno && test x$enable_oss = xno &&
test x$enable_openal = xno &&
test x$enable_shout = xno && test x$enable_shout = xno &&
test x$enable_recorder_output = xno && test x$enable_recorder_output = xno &&
test x$enable_httpd_output = xno && test x$enable_httpd_output = xno &&
......
/*
* Copyright (C) 2003-2009 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 "../output_api.h"
#include "../timer.h"
#include <glib.h>
#include <AL/al.h>
#include <AL/alc.h>
#undef G_LOG_DOMAIN
#define G_LOG_DOMAIN "openal"
/* should be enough for buffer size = 2048 */
#define NUM_BUFFERS 16
struct openal_data {
const char *device_name;
ALCdevice *device;
ALCcontext *context;
Timer *timer;
ALuint buffers[NUM_BUFFERS];
int filled;
ALuint source;
ALenum format;
ALuint frequency;
};
static inline GQuark
openal_output_quark(void)
{
return g_quark_from_static_string("openal_output");
}
static ALenum
openal_audio_format(struct audio_format *audio_format)
{
/* Only 8 and 16 bit samples are supported */
if (audio_format->bits != 16 && audio_format->bits != 8)
audio_format->bits = 16;
switch (audio_format->bits)
{
case 16:
if (audio_format->channels == 2)
return AL_FORMAT_STEREO16;
if (audio_format->channels == 1)
return AL_FORMAT_MONO16;
break;
case 8:
if (audio_format->channels == 2)
return AL_FORMAT_STEREO8;
if (audio_format->channels == 1)
return AL_FORMAT_MONO8;
break;
}
return 0;
}
static bool
openal_setup_context(struct openal_data *od,
GError **error)
{
od->device = alcOpenDevice(od->device_name);
if (od->device == NULL) {
g_set_error(error, openal_output_quark(), 0,
"Error opening OpenAL device \"%s\"\n",
od->device_name);
return false;
}
od->context = alcCreateContext(od->device, NULL);
if (od->context == NULL) {
g_set_error(error, openal_output_quark(), 0,
"Error creating context for \"%s\"\n",
od->device_name);
alcCloseDevice(od->device);
return false;
}
return true;
}
static void
openal_unqueue_buffers(struct openal_data *od)
{
ALint num;
ALuint buffer;
alGetSourcei(od->source, AL_BUFFERS_QUEUED, &num);
while (num--) {
alSourceUnqueueBuffers(od->source, 1, &buffer);
}
}
static void *
openal_init(G_GNUC_UNUSED const struct audio_format *audio_format,
const struct config_param *param,
G_GNUC_UNUSED GError **error)
{
const char *device_name = config_get_block_string(param, "device", NULL);
struct openal_data *od;
od = g_new(struct openal_data, 1);
od->device_name = device_name;
if (device_name == NULL) {
device_name = alcGetString(NULL, ALC_DEFAULT_DEVICE_SPECIFIER);
}
return od;
}
static void
openal_finish(void *data)
{
struct openal_data *od = data;
g_free(od);
}
static bool
openal_open(void *data, struct audio_format *audio_format,
GError **error)
{
struct openal_data *od = data;
od->format = openal_audio_format(audio_format);
if (!od->format) {
g_set_error(error, openal_output_quark(), 0,
"Unsupported audio format (%i channels, %i bps)",
audio_format->channels,
audio_format->bits);
return false;
}
if (!openal_setup_context(od, error)) {
return false;
}
alcMakeContextCurrent(od->context);
alGenBuffers(NUM_BUFFERS, od->buffers);
if (alGetError() != AL_NO_ERROR) {
g_set_error(error, openal_output_quark(), 0,
"Failed to generate buffers");
return false;
}
alGenSources(1, &od->source);
if (alGetError() != AL_NO_ERROR) {
g_set_error(error, openal_output_quark(), 0,
"Failed to generate source");
alDeleteBuffers(NUM_BUFFERS, od->buffers);
return false;
}
od->filled = 0;
od->timer = timer_new(audio_format);
od->frequency = audio_format->sample_rate;
return true;
}
static void
openal_close(void *data)
{
struct openal_data *od = data;
timer_free(od->timer);
alcMakeContextCurrent(od->context);
alDeleteSources(1, &od->source);
alDeleteBuffers(NUM_BUFFERS, od->buffers);
alcDestroyContext(od->context);
alcCloseDevice(od->device);
}
static size_t
openal_play(void *data, const void *chunk, size_t size,
G_GNUC_UNUSED GError **error)
{
struct openal_data *od = data;
ALuint buffer;
ALint num, state;
if (alcGetCurrentContext() != od->context) {
alcMakeContextCurrent(od->context);
}
alGetSourcei(od->source, AL_BUFFERS_PROCESSED, &num);
if (od->filled < NUM_BUFFERS) {
/* fill all buffers */
buffer = od->buffers[od->filled];
od->filled++;
} else {
/* wait for processed buffer */
while (num < 1) {
if (!od->timer->started) {
timer_start(od->timer);
} else {
timer_sync(od->timer);
}
timer_add(od->timer, size);
alGetSourcei(od->source, AL_BUFFERS_PROCESSED, &num);
}
alSourceUnqueueBuffers(od->source, 1, &buffer);
}
alBufferData(buffer, od->format, chunk, size, od->frequency);
alSourceQueueBuffers(od->source, 1, &buffer);
alGetSourcei(od->source, AL_SOURCE_STATE, &state);
if (state != AL_PLAYING) {
alSourcePlay(od->source);
}
return size;
}
static void
openal_cancel(void *data)
{
struct openal_data *od = data;
od->filled = 0;
alcMakeContextCurrent(od->context);
alSourceStop(od->source);
openal_unqueue_buffers(od);
}
const struct audio_output_plugin openal_output_plugin = {
.name = "openal",
.init = openal_init,
.finish = openal_finish,
.open = openal_open,
.close = openal_close,
.play = openal_play,
.cancel = openal_cancel,
};
...@@ -28,6 +28,7 @@ extern const struct audio_output_plugin pipe_output_plugin; ...@@ -28,6 +28,7 @@ extern const struct audio_output_plugin pipe_output_plugin;
extern const struct audio_output_plugin alsaPlugin; extern const struct audio_output_plugin alsaPlugin;
extern const struct audio_output_plugin ao_output_plugin; extern const struct audio_output_plugin ao_output_plugin;
extern const struct audio_output_plugin oss_output_plugin; extern const struct audio_output_plugin oss_output_plugin;
extern const struct audio_output_plugin openal_output_plugin;
extern const struct audio_output_plugin osxPlugin; extern const struct audio_output_plugin osxPlugin;
extern const struct audio_output_plugin solaris_output_plugin; extern const struct audio_output_plugin solaris_output_plugin;
extern const struct audio_output_plugin pulse_plugin; extern const struct audio_output_plugin pulse_plugin;
...@@ -56,6 +57,9 @@ const struct audio_output_plugin *audio_output_plugins[] = { ...@@ -56,6 +57,9 @@ const struct audio_output_plugin *audio_output_plugins[] = {
#ifdef HAVE_OSS #ifdef HAVE_OSS
&oss_output_plugin, &oss_output_plugin,
#endif #endif
#ifdef HAVE_OPENAL
&openal_output_plugin,
#endif
#ifdef HAVE_OSX #ifdef HAVE_OSX
&osxPlugin, &osxPlugin,
#endif #endif
......
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