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authorSerge Ziryukin <ftrvxmtrx@gmail.com>2009-08-31 10:26:22 +0300
committerMax Kellermann <max@duempel.org>2009-09-06 17:34:56 +0200
commit8b6a5d19d01ddf0657a64634ccd66b6fde66b735 (patch)
tree7dc7aee402dc42411d45efe3e8310ab2ffc9311b /src
parent129920e8f418d029350ce3c463948eba55d6775c (diff)
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openal output plugin
Diffstat (limited to 'src')
-rw-r--r--src/output/openal_plugin.c267
-rw-r--r--src/output_list.c4
2 files changed, 271 insertions, 0 deletions
diff --git a/src/output/openal_plugin.c b/src/output/openal_plugin.c
new file mode 100644
index 000000000..2d581ebbf
--- /dev/null
+++ b/src/output/openal_plugin.c
@@ -0,0 +1,267 @@
+/*
+ * 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,
+};
diff --git a/src/output_list.c b/src/output_list.c
index 74a9be81c..476701a1a 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