/* * OpenAL audio output driver for MPlayer * * Copyleft 2006 by Reimar Döffinger (Reimar.Doeffinger@stud.uni-karlsruhe.de) * * This file is part of mpv. * * mpv 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. * * mpv 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 mpv. If not, see . */ #include "config.h" #include #include #include #ifdef __APPLE__ #ifndef AL_FORMAT_MONO_FLOAT32 #define AL_FORMAT_MONO_FLOAT32 0x10010 #endif #ifndef AL_FORMAT_STEREO_FLOAT32 #define AL_FORMAT_STEREO_FLOAT32 0x10011 #endif #ifndef AL_FORMAT_MONO_DOUBLE_EXT #define AL_FORMAT_MONO_DOUBLE_EXT 0x10012 #endif #include #else #ifdef OPENAL_AL_H #include #include #include #else #include #include #include #endif #endif // __APPLE__ #include "common/msg.h" #include "ao.h" #include "internal.h" #include "audio/format.h" #include "osdep/timer.h" #include "options/m_option.h" #define MAX_CHANS MP_NUM_CHANNELS #define NUM_BUF 128 #define CHUNK_SAMPLES 256 static ALuint buffers[MAX_CHANS][NUM_BUF]; static ALuint sources[MAX_CHANS]; static int cur_buf[MAX_CHANS]; static int unqueue_buf[MAX_CHANS]; static struct ao *ao_data; struct priv { char *cfg_device; ALenum al_format; int chunk_size; }; static void reset(struct ao *ao); static int control(struct ao *ao, enum aocontrol cmd, void *arg) { switch (cmd) { case AOCONTROL_GET_VOLUME: case AOCONTROL_SET_VOLUME: { ALfloat volume; ao_control_vol_t *vol = (ao_control_vol_t *)arg; if (cmd == AOCONTROL_SET_VOLUME) { volume = (vol->left + vol->right) / 200.0; alListenerf(AL_GAIN, volume); } alGetListenerf(AL_GAIN, &volume); vol->left = vol->right = volume * 100; return CONTROL_TRUE; } case AOCONTROL_HAS_SOFT_VOLUME: return CONTROL_TRUE; } return CONTROL_UNKNOWN; } static int validate_device_opt(struct mp_log *log, const m_option_t *opt, struct bstr name, struct bstr param) { if (bstr_equals0(param, "help")) { if (alcIsExtensionPresent(NULL, "ALC_ENUMERATE_ALL_EXT") != AL_TRUE) { mp_fatal(log, "Device listing not supported.\n"); return M_OPT_EXIT; } const char *list = alcGetString(NULL, ALC_ALL_DEVICES_SPECIFIER); mp_info(log, "OpenAL devices:\n"); while (list && *list) { mp_info(log, " '%s'\n", list); list = list + strlen(list) + 1; } return M_OPT_EXIT; } return 0; } static void list_devs(struct ao *ao, struct ao_device_list *list) { if (alcIsExtensionPresent(NULL, "ALC_ENUMERATE_ALL_EXT") != AL_TRUE) return; const char *devs = alcGetString(NULL, ALC_ALL_DEVICES_SPECIFIER); while (devs && *devs) { ao_device_list_add(list, ao, &(struct ao_device_desc){devs, devs}); devs = devs + strlen(devs) + 1; } } struct speaker { int id; float pos[3]; }; static const struct speaker speaker_pos[] = { {MP_SPEAKER_ID_FL, {-0.500, 0, -0.866}}, // -30 deg {MP_SPEAKER_ID_FR, { 0.500, 0, -0.866}}, // 30 deg {MP_SPEAKER_ID_FC, { 0, 0, -1}}, // 0 deg {MP_SPEAKER_ID_LFE, { 0, -1, 0}}, // below {MP_SPEAKER_ID_BL, {-0.609, 0, 0.793}}, // -142.5 deg {MP_SPEAKER_ID_BR, { 0.609, 0, 0.793}}, // 142.5 deg {MP_SPEAKER_ID_BC, { 0, 0, 1}}, // 180 deg {MP_SPEAKER_ID_SL, {-0.985, 0, 0.174}}, // -100 deg {MP_SPEAKER_ID_SR, { 0.985, 0, 0.174}}, // 100 deg {-1}, }; static ALenum get_al_format(int format) { switch (format) { case AF_FORMAT_U8P: return AL_FORMAT_MONO8; case AF_FORMAT_S16P: return AL_FORMAT_MONO16; case AF_FORMAT_FLOATP: if (alIsExtensionPresent((ALchar*)"AL_EXT_float32") == AL_TRUE) return AL_FORMAT_MONO_FLOAT32; break; case AF_FORMAT_DOUBLEP: if (alIsExtensionPresent((ALchar*)"AL_EXT_double") == AL_TRUE) return AL_FORMAT_MONO_DOUBLE_EXT; break; } return AL_FALSE; } // close audio device static void uninit(struct ao *ao) { ALCcontext *ctx = alcGetCurrentContext(); ALCdevice *dev = alcGetContextsDevice(ctx); reset(ao); alcMakeContextCurrent(NULL); alcDestroyContext(ctx); alcCloseDevice(dev); ao_data = NULL; } static int init(struct ao *ao) { float position[3] = {0, 0, 0}; float direction[6] = {0, 0, -1, 0, 1, 0}; ALCdevice *dev = NULL; ALCcontext *ctx = NULL; ALCint freq = 0; ALCint attribs[] = {ALC_FREQUENCY, ao->samplerate, 0, 0}; int i; struct priv *p = ao->priv; if (ao_data) { MP_FATAL(ao, "Not reentrant!\n"); return -1; } ao_data = ao; struct mp_chmap_sel sel = {0}; for (i = 0; speaker_pos[i].id != -1; i++) mp_chmap_sel_add_speaker(&sel, speaker_pos[i].id); if (!ao_chmap_sel_adjust(ao, &sel, &ao->channels)) goto err_out; struct speaker speakers[MAX_CHANS]; for (i = 0; i < ao->channels.num; i++) { speakers[i].id = -1; for (int n = 0; speaker_pos[n].id >= 0; n++) { if (speaker_pos[n].id == ao->channels.speaker[i]) speakers[i] = speaker_pos[n]; } if (speakers[i].id < 0) { MP_FATAL(ao, "Unknown channel layout\n"); goto err_out; } } char *dev_name = p->cfg_device; if (!dev_name || !dev_name[0]) dev_name = ao->device; dev = alcOpenDevice(dev_name && dev_name[0] ? dev_name : NULL); if (!dev) { MP_FATAL(ao, "could not open device\n"); goto err_out; } ctx = alcCreateContext(dev, attribs); alcMakeContextCurrent(ctx); alListenerfv(AL_POSITION, position); alListenerfv(AL_ORIENTATION, direction); alGenSources(ao->channels.num, sources); for (i = 0; i < ao->channels.num; i++) { cur_buf[i] = 0; unqueue_buf[i] = 0; alGenBuffers(NUM_BUF, buffers[i]); alSourcefv(sources[i], AL_POSITION, speakers[i].pos); alSource3f(sources[i], AL_VELOCITY, 0, 0, 0); } alcGetIntegerv(dev, ALC_FREQUENCY, 1, &freq); if (alcGetError(dev) == ALC_NO_ERROR && freq) ao->samplerate = freq; p->al_format = AL_FALSE; int try_formats[AF_FORMAT_COUNT]; af_get_best_sample_formats(ao->format, try_formats); for (int n = 0; try_formats[n]; n++) { p->al_format = get_al_format(try_formats[n]); if (p->al_format != AL_FALSE) { ao->format = try_formats[n]; break; } } if (p->al_format == AL_FALSE) { MP_FATAL(ao, "Can't find appropriate sample format.\n"); uninit(ao); goto err_out; } p->chunk_size = CHUNK_SAMPLES * af_fmt_to_bytes(ao->format); return 0; err_out: ao_data = NULL; return -1; } static void drain(struct ao *ao) { ALint state; alGetSourcei(sources[0], AL_SOURCE_STATE, &state); while (state == AL_PLAYING) { mp_sleep_us(10000); alGetSourcei(sources[0], AL_SOURCE_STATE, &state); } } static void unqueue_buffers(void) { ALint p; int s; for (s = 0; s < ao_data->channels.num; s++) { int till_wrap = NUM_BUF - unqueue_buf[s]; alGetSourcei(sources[s], AL_BUFFERS_PROCESSED, &p); if (p >= till_wrap) { alSourceUnqueueBuffers(sources[s], till_wrap, &buffers[s][unqueue_buf[s]]); unqueue_buf[s] = 0; p -= till_wrap; } if (p) { alSourceUnqueueBuffers(sources[s], p, &buffers[s][unqueue_buf[s]]); unqueue_buf[s] += p; } } } /** * \brief stop playing and empty buffers (for seeking/pause) */ static void reset(struct ao *ao) { alSourceStopv(ao->channels.num, sources); unqueue_buffers(); } /** * \brief stop playing, keep buffers (for pause) */ static void audio_pause(struct ao *ao) { alSourcePausev(ao->channels.num, sources); } /** * \brief resume playing, after audio_pause() */ static void audio_resume(struct ao *ao) { alSourcePlayv(ao->channels.num, sources); } static int get_space(struct ao *ao) { ALint queued; unqueue_buffers(); alGetSourcei(sources[0], AL_BUFFERS_QUEUED, &queued); queued = NUM_BUF - queued - 3; if (queued < 0) return 0; return queued * CHUNK_SAMPLES; } /** * \brief write data into buffer and reset underrun flag */ static int play(struct ao *ao, void **data, int samples, int flags) { struct priv *p = ao->priv; ALint state; int num = samples / CHUNK_SAMPLES; for (int i = 0; i < num; i++) { for (int ch = 0; ch < ao->channels.num; ch++) { char *d = data[ch]; d += i * p->chunk_size; alBufferData(buffers[ch][cur_buf[ch]], p->al_format, d, p->chunk_size, ao->samplerate); alSourceQueueBuffers(sources[ch], 1, &buffers[ch][cur_buf[ch]]); cur_buf[ch] = (cur_buf[ch] + 1) % NUM_BUF; } } alGetSourcei(sources[0], AL_SOURCE_STATE, &state); if (state != AL_PLAYING) // checked here in case of an underrun alSourcePlayv(ao->channels.num, sources); return num * CHUNK_SAMPLES; } static double get_delay(struct ao *ao) { ALint queued; unqueue_buffers(); alGetSourcei(sources[0], AL_BUFFERS_QUEUED, &queued); return queued * CHUNK_SAMPLES / (double)ao->samplerate; } #define OPT_BASE_STRUCT struct priv const struct ao_driver audio_out_openal = { .description = "OpenAL audio output", .name = "openal", .init = init, .uninit = uninit, .control = control, .get_space = get_space, .play = play, .get_delay = get_delay, .pause = audio_pause, .resume = audio_resume, .reset = reset, .drain = drain, .list_devs = list_devs, .priv_size = sizeof(struct priv), .options = (const struct m_option[]) { OPT_STRING_VALIDATE("device", cfg_device, 0, validate_device_opt, DEVICE_OPT_DEPRECATION), {0} }, .legacy_prefix = "ao-openal", };