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authorMax Kellermann <max@duempel.org>2013-04-17 01:04:27 +0200
committerMax Kellermann <max@duempel.org>2013-04-17 01:04:27 +0200
commit750b2ad6a87e4081e68c5e88924c121b9ab6a078 (patch)
tree86af3844cb06eac9fb23267bc6eb9f35dd0d5a32 /src/output/openal_output_plugin.c
parent86c276f5383d5362ae65e58fe5ea522f907ed724 (diff)
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output/openal: convert to C++
Diffstat (limited to 'src/output/openal_output_plugin.c')
-rw-r--r--src/output/openal_output_plugin.c279
1 files changed, 0 insertions, 279 deletions
diff --git a/src/output/openal_output_plugin.c b/src/output/openal_output_plugin.c
deleted file mode 100644
index ebd35ef12..000000000
--- a/src/output/openal_output_plugin.c
+++ /dev/null
@@ -1,279 +0,0 @@
-/*
- * 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,
-};