/* * 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 "config.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 { 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; if (device_name == NULL) { device_name = alcGetString(NULL, ALC_DEFAULT_DEVICE_SPECIFIER); } od = g_new(struct openal_data, 1); od->device_name = device_name; 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, };