OpenAL output plugin
---
Makefile.am | 4 +
configure.ac | 25 +++++
src/output/openal_plugin.c | 248 ++++++++++++++++++++++++++++++++++++++++++++
src/output_list.c | 4 +
4 files changed, 281 insertions(+), 0 deletions(-)
create mode 100644 src/output/openal_plugin.c
diff --git a/Makefile.am b/Makefile.am
index b017d97..5a6a8da 100644
--- a/Makefile.am
+++ b/Makefile.am
@@ -590,6 +590,10 @@ OUTPUT_SRC += src/output/oss_plugin.c
MIXER_SRC += src/mixer/oss_mixer.c
endif
+if HAVE_OPENAL
+OUTPUT_SRC += src/output/openal_plugin.c
+endif
+
if HAVE_OSX
OUTPUT_SRC += src/output/osx_plugin.c
endif
diff --git a/configure.ac b/configure.ac
index 681feae..5497f3c 100644
--- a/configure.ac
+++ b/configure.ac
@@ -706,6 +706,11 @@ AC_ARG_ENABLE(oss,
[disable OSS support (default: enable)]),,
enable_oss=yes)
+AC_ARG_ENABLE(openal,
+ AS_HELP_STRING([--enable-openal],
+ [enable OpenAL support (default: disable)]),,
+ enable_openal=no)
+
AC_ARG_ENABLE(pulse,
AS_HELP_STRING([--enable-pulse],
[enable support for the PulseAudio sound server]),,
@@ -774,6 +779,19 @@ fi
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$enable_fifo = xyes; then
AC_CHECK_FUNC([mkfifo],
[enable_fifo=yes;AC_DEFINE([HAVE_FIFO], 1,
@@ -1292,6 +1310,12 @@ else
echo " OSS support ...................disabled"
fi
+if test x$enable_openal = xyes; then
+ echo " OpenAL support ................enabled"
+else
+ echo " OpenAL support ................disabled"
+fi
+
if test x$enable_osx = xyes; then
echo " OS X support ..................enabled"
else
@@ -1333,6 +1357,7 @@ echo ""
if
test x$enable_ao = xno &&
test x$enable_oss = xno &&
+ test x$enable_openal = xno &&
test x$enable_shout = xno &&
test x$enable_recorder_output = xno &&
test x$enable_httpd_output = xno &&
diff --git a/src/output/openal_plugin.c b/src/output/openal_plugin.c
new file mode 100644
index 0000000..c1c8af1
--- /dev/null
+++ b/src/output/openal_plugin.c
@@ -0,0 +1,248 @@
+/*
+ * 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 32
+
+struct openal_data {
+ 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 void *
+openal_init(G_GNUC_UNUSED const struct audio_format *audio_format,
+ const struct config_param *param,
+ GError **error)
+{
+ const char *device_name = config_get_block_string(param, "device",
NULL);
+ ALCdevice *device = alcOpenDevice(device_name);
+ ALCcontext *context;
+ struct openal_data *od;
+
+ if (device_name == NULL) {
+ device_name = "default device";
+ }
+
+ if (device == NULL) {
+ g_set_error(error, openal_output_quark(), 0,
+ "failed to open %s\n",
+ device_name);
+ return NULL;
+ }
+
+ context = alcCreateContext(device, NULL);
+
+ if (context == NULL) {
+ g_set_error(error, openal_output_quark(), 0,
+ "failed to create context on %s\n",
+ device_name);
+ alcCloseDevice(device);
+ return NULL;
+ }
+
+ od = g_new(struct openal_data, 1);
+ od->context = NULL;
+ od->device = device;
+ od->context = context;
+
+ return od;
+}
+
+static void
+openal_finish(void *data)
+{
+ struct openal_data *od = data;
+
+ alcDestroyContext(od->context);
+ alcCloseDevice(od->device);
+
+ g_free(od);
+}
+
+static bool
+openal_open(void *data, struct audio_format *audio_format,
+ GError **error)
+{
+ struct openal_data *od = data;
+
+ alcMakeContextCurrent(od->context);
+
+ 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;
+ }
+
+ alGenBuffers(NUM_BUFFERS, od->buffers);
+
+ if (alGetError() != AL_NO_ERROR) {
+ g_set_error(error, openal_output_quark(), 0,
+ "faled 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);
+}
+
+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;
+
+ if (alcGetCurrentContext() != od->context) {
+ alcMakeContextCurrent(od->context);
+ }
+
+ if (od->filled < NUM_BUFFERS) {
+ /* fill all buffers */
+ buffer = od->buffers[od->filled];
+ od->filled++;
+ } else {
+ alGetSourcei(od->source, AL_BUFFERS_PROCESSED, &num);
+
+ /* 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);
+
+ /* start playback when first buffer filled */
+ if (od->filled == 1) {
+ alSourcePlay(od->source);
+ }
+
+ return size;
+}
+
+static void
+openal_cancel(void *data)
+{
+ struct openal_data *od = data;
+ ALint num;
+ ALuint buffers[NUM_BUFFERS];
+
+ alcMakeContextCurrent(od->context);
+ alSourceStop(od->source);
+ alGetSourcei(od->source, AL_BUFFERS_QUEUED, &num);
+ alSourceUnqueueBuffers(od->source, num, buffers);
+ od->filled = 0;
+}
+
+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,
+};
diff --git a/src/output_list.c b/src/output_list.c
index 74a9be8..476701a 100644
--- a/src/output_list.c
+++ b/src/output_list.c
@@ -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 ao_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 solaris_output_plugin;
extern const struct audio_output_plugin pulse_plugin;
@@ -56,6 +57,9 @@ const struct audio_output_plugin *audio_output_plugins[] = {
#ifdef HAVE_OSS
&oss_output_plugin,
#endif
+#ifdef HAVE_OPENAL
+ &openal_output_plugin,
+#endif
#ifdef HAVE_OSX
&osxPlugin,
#endif
--
1.6.3.2
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