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,
+};