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-rw-r--r--src/output/openal_output_plugin.c279
1 files changed, 279 insertions, 0 deletions
diff --git a/src/output/openal_output_plugin.c b/src/output/openal_output_plugin.c
new file mode 100644
index 000000000..ebd35ef12
--- /dev/null
+++ b/src/output/openal_output_plugin.c
@@ -0,0 +1,279 @@
+/*
+ * Copyright (C) 2003-2011 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 "openal_output_plugin.h"
+#include "output_api.h"
+
+#include <glib.h>
+
+#ifndef HAVE_OSX
+#include <AL/al.h>
+#include <AL/alc.h>
+#else
+#include <OpenAL/al.h>
+#include <OpenAL/alc.h>
+#endif
+
+#undef G_LOG_DOMAIN
+#define G_LOG_DOMAIN "openal"
+
+/* should be enough for buffer size = 2048 */
+#define NUM_BUFFERS 16
+
+struct openal_data {
+ struct audio_output base;
+
+ const char *device_name;
+ ALCdevice *device;
+ ALCcontext *context;
+ ALuint buffers[NUM_BUFFERS];
+ unsigned 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)
+{
+ /* note: cannot map SAMPLE_FORMAT_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 SAMPLE_FORMAT_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 = SAMPLE_FORMAT_S16;
+ return openal_audio_format(audio_format);
+ }
+}
+
+G_GNUC_PURE
+static inline ALint
+openal_get_source_i(const struct openal_data *od, ALenum param)
+{
+ ALint value;
+ alGetSourcei(od->source, param, &value);
+ return value;
+}
+
+G_GNUC_PURE
+static inline bool
+openal_has_processed(const struct openal_data *od)
+{
+ return openal_get_source_i(od, AL_BUFFERS_PROCESSED) > 0;
+}
+
+G_GNUC_PURE
+static inline ALint
+openal_is_playing(const struct openal_data *od)
+{
+ return openal_get_source_i(od, AL_SOURCE_STATE) == AL_PLAYING;
+}
+
+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 struct audio_output *
+openal_init(const struct config_param *param, GError **error_r)
+{
+ const char *device_name = config_get_block_string(param, "device", NULL);
+ struct openal_data *od;
+
+ if (device_name == NULL) {
+ device_name = alcGetString(NULL, ALC_DEFAULT_DEVICE_SPECIFIER);
+ }
+
+ od = g_new(struct openal_data, 1);
+ if (!ao_base_init(&od->base, &openal_output_plugin, param, error_r)) {
+ g_free(od);
+ return NULL;
+ }
+
+ od->device_name = device_name;
+
+ return &od->base;
+}
+
+static void
+openal_finish(struct audio_output *ao)
+{
+ struct openal_data *od = (struct openal_data *)ao;
+
+ ao_base_finish(&od->base);
+ g_free(od);
+}
+
+static bool
+openal_open(struct audio_output *ao, struct audio_format *audio_format,
+ GError **error)
+{
+ struct openal_data *od = (struct openal_data *)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) {
+ 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->frequency = audio_format->sample_rate;
+
+ return true;
+}
+
+static void
+openal_close(struct audio_output *ao)
+{
+ struct openal_data *od = (struct openal_data *)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)
+{
+ struct openal_data *od = (struct openal_data *)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,
+ G_GNUC_UNUSED GError **error)
+{
+ struct openal_data *od = (struct openal_data *)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)
+{
+ struct openal_data *od = (struct openal_data *)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 = {
+ .name = "openal",
+ .init = openal_init,
+ .finish = openal_finish,
+ .open = openal_open,
+ .close = openal_close,
+ .delay = openal_delay,
+ .play = openal_play,
+ .cancel = openal_cancel,
+};