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author | Max Kellermann <max@duempel.org> | 2013-04-17 01:04:27 +0200 |
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committer | Max Kellermann <max@duempel.org> | 2013-04-17 01:04:27 +0200 |
commit | 750b2ad6a87e4081e68c5e88924c121b9ab6a078 (patch) | |
tree | 86af3844cb06eac9fb23267bc6eb9f35dd0d5a32 /src/output/openal_output_plugin.c | |
parent | 86c276f5383d5362ae65e58fe5ea522f907ed724 (diff) | |
download | mpd-750b2ad6a87e4081e68c5e88924c121b9ab6a078.tar.gz mpd-750b2ad6a87e4081e68c5e88924c121b9ab6a078.tar.xz mpd-750b2ad6a87e4081e68c5e88924c121b9ab6a078.zip |
output/openal: convert to C++
Diffstat (limited to 'src/output/openal_output_plugin.c')
-rw-r--r-- | src/output/openal_output_plugin.c | 279 |
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, -}; |