mirror of https://github.com/mpv-player/mpv
305 lines
8.3 KiB
C
305 lines
8.3 KiB
C
/*
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* OpenAL audio output driver for MPlayer
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*
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* Copyleft 2006 by Reimar Döffinger (Reimar.Doeffinger@stud.uni-karlsruhe.de)
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*
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* This file is part of MPlayer.
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*
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* MPlayer is free software; you can redistribute it and/or modify
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* it under the terms of the GNU General Public License as published by
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* the Free Software Foundation; either version 2 of the License, or
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* (at your option) any later version.
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*
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* MPlayer is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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* GNU General Public License for more details.
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*
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* You should have received a copy of the GNU General Public License along
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* along with MPlayer; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#include "config.h"
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#include <stdlib.h>
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#include <stdio.h>
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#include <inttypes.h>
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#ifdef OPENAL_AL_H
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#include <OpenAL/alc.h>
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#include <OpenAL/al.h>
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#include <OpenAL/alext.h>
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#else
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#include <AL/alc.h>
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#include <AL/al.h>
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#include <AL/alext.h>
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#endif
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#include "core/mp_msg.h"
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#include "ao.h"
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#include "audio_out_internal.h"
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#include "audio/format.h"
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#include "osdep/timer.h"
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#include "core/subopt-helper.h"
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static const ao_info_t info =
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{
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"OpenAL audio output",
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"openal",
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"Reimar Döffinger <Reimar.Doeffinger@stud.uni-karlsruhe.de>",
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""
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};
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LIBAO_EXTERN(openal)
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#define MAX_CHANS MP_NUM_CHANNELS
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#define NUM_BUF 128
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#define CHUNK_SIZE 512
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static ALuint buffers[MAX_CHANS][NUM_BUF];
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static ALuint sources[MAX_CHANS];
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static int cur_buf[MAX_CHANS];
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static int unqueue_buf[MAX_CHANS];
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static int16_t *tmpbuf;
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static int control(int cmd, void *arg) {
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switch (cmd) {
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case AOCONTROL_GET_VOLUME:
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case AOCONTROL_SET_VOLUME: {
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ALfloat volume;
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ao_control_vol_t *vol = (ao_control_vol_t *)arg;
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if (cmd == AOCONTROL_SET_VOLUME) {
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volume = (vol->left + vol->right) / 200.0;
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alListenerf(AL_GAIN, volume);
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}
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alGetListenerf(AL_GAIN, &volume);
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vol->left = vol->right = volume * 100;
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return CONTROL_TRUE;
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}
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}
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return CONTROL_UNKNOWN;
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}
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/**
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* \brief print suboption usage help
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*/
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static void print_help(void) {
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mp_msg(MSGT_AO, MSGL_FATAL,
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"\n-ao openal commandline help:\n"
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"Example: mpv -ao openal:device=subdevice\n"
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"\nOptions:\n"
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" device=subdevice\n"
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" Audio device OpenAL should use. Devices can be listed\n"
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" with -ao openal:device=help\n"
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);
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}
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static void list_devices(void) {
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if (alcIsExtensionPresent(NULL, "ALC_ENUMERATE_ALL_EXT") != AL_TRUE) {
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mp_msg(MSGT_AO, MSGL_FATAL, "Device listing not supported.\n");
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return;
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}
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const char *list = alcGetString(NULL, ALC_ALL_DEVICES_SPECIFIER);
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mp_msg(MSGT_AO, MSGL_FATAL, "OpenAL devices:\n");
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while (list && *list) {
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mp_msg(MSGT_AO, MSGL_FATAL, " '%s'\n", list);
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list = list + strlen(list) + 1;
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}
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}
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struct speaker {
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int id;
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float pos[3];
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};
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static const struct speaker speaker_pos[] = {
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{MP_SPEAKER_ID_FL, {-1, 0, 0.5}},
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{MP_SPEAKER_ID_FR, { 1, 0, 0.5}},
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{MP_SPEAKER_ID_FC, { 0, 0, 1}},
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{MP_SPEAKER_ID_LFE, { 0, 0, 0.1}},
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{MP_SPEAKER_ID_BL, {-1, 0, -1}},
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{MP_SPEAKER_ID_BR, { 1, 0, -1}},
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{MP_SPEAKER_ID_BC, { 0, 0, -1}},
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{MP_SPEAKER_ID_SL, {-1, 0, 0}},
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{MP_SPEAKER_ID_SR, { 1, 0, 0}},
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{-1},
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};
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static int init(int rate, const struct mp_chmap *channels, int format,
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int flags)
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{
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float position[3] = {0, 0, 0};
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float direction[6] = {0, 0, 1, 0, -1, 0};
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ALCdevice *dev = NULL;
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ALCcontext *ctx = NULL;
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ALCint freq = 0;
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ALCint attribs[] = {ALC_FREQUENCY, rate, 0, 0};
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int i;
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char *device = NULL;
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const opt_t subopts[] = {
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{"device", OPT_ARG_MSTRZ, &device, NULL},
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{NULL}
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};
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global_ao->no_persistent_volume = true;
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if (subopt_parse(ao_subdevice, subopts) != 0) {
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print_help();
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return 0;
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}
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if (device && strcmp(device, "help") == 0) {
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list_devices();
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goto err_out;
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}
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struct mp_chmap_sel sel = {0};
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for (i = 0; speaker_pos[i].id != -1; i++)
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mp_chmap_sel_add_speaker(&sel, speaker_pos[i].id);
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if (!ao_chmap_sel_adjust(&ao_data, &sel, &ao_data.channels))
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goto err_out;
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struct speaker speakers[MAX_CHANS];
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for (i = 0; i < ao_data.channels.num; i++) {
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speakers[i].id = -1;
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for (int n = 0; speaker_pos[n].id >= 0; n++) {
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if (speaker_pos[n].id == ao_data.channels.speaker[i])
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speakers[i] = speaker_pos[n];
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}
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if (speakers[i].id < 0) {
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mp_msg(MSGT_AO, MSGL_FATAL, "[OpenAL] Unknown channel layout\n");
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goto err_out;
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}
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}
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dev = alcOpenDevice(device);
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if (!dev) {
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mp_msg(MSGT_AO, MSGL_FATAL, "[OpenAL] could not open device\n");
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goto err_out;
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}
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ctx = alcCreateContext(dev, attribs);
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alcMakeContextCurrent(ctx);
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alListenerfv(AL_POSITION, position);
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alListenerfv(AL_ORIENTATION, direction);
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alGenSources(ao_data.channels.num, sources);
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for (i = 0; i < ao_data.channels.num; i++) {
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cur_buf[i] = 0;
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unqueue_buf[i] = 0;
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alGenBuffers(NUM_BUF, buffers[i]);
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alSourcefv(sources[i], AL_POSITION, speakers[i].pos);
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alSource3f(sources[i], AL_VELOCITY, 0, 0, 0);
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}
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alcGetIntegerv(dev, ALC_FREQUENCY, 1, &freq);
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if (alcGetError(dev) == ALC_NO_ERROR && freq)
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rate = freq;
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ao_data.samplerate = rate;
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ao_data.format = AF_FORMAT_S16_NE;
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ao_data.bps = ao_data.channels.num * rate * 2;
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ao_data.buffersize = CHUNK_SIZE * NUM_BUF;
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ao_data.outburst = ao_data.channels.num * CHUNK_SIZE;
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tmpbuf = malloc(CHUNK_SIZE);
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free(device);
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return 1;
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err_out:
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free(device);
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return 0;
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}
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// close audio device
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static void uninit(int immed) {
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ALCcontext *ctx = alcGetCurrentContext();
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ALCdevice *dev = alcGetContextsDevice(ctx);
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free(tmpbuf);
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if (!immed) {
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ALint state;
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alGetSourcei(sources[0], AL_SOURCE_STATE, &state);
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while (state == AL_PLAYING) {
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mp_sleep_us(10000);
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alGetSourcei(sources[0], AL_SOURCE_STATE, &state);
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}
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}
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reset();
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alcMakeContextCurrent(NULL);
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alcDestroyContext(ctx);
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alcCloseDevice(dev);
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}
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static void unqueue_buffers(void) {
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ALint p;
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int s;
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for (s = 0; s < ao_data.channels.num; s++) {
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int till_wrap = NUM_BUF - unqueue_buf[s];
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alGetSourcei(sources[s], AL_BUFFERS_PROCESSED, &p);
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if (p >= till_wrap) {
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alSourceUnqueueBuffers(sources[s], till_wrap, &buffers[s][unqueue_buf[s]]);
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unqueue_buf[s] = 0;
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p -= till_wrap;
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}
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if (p) {
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alSourceUnqueueBuffers(sources[s], p, &buffers[s][unqueue_buf[s]]);
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unqueue_buf[s] += p;
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}
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}
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}
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/**
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* \brief stop playing and empty buffers (for seeking/pause)
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*/
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static void reset(void) {
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alSourceStopv(ao_data.channels.num, sources);
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unqueue_buffers();
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}
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/**
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* \brief stop playing, keep buffers (for pause)
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*/
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static void audio_pause(void) {
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alSourcePausev(ao_data.channels.num, sources);
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}
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/**
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* \brief resume playing, after audio_pause()
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*/
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static void audio_resume(void) {
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alSourcePlayv(ao_data.channels.num, sources);
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}
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static int get_space(void) {
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ALint queued;
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unqueue_buffers();
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alGetSourcei(sources[0], AL_BUFFERS_QUEUED, &queued);
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queued = NUM_BUF - queued - 3;
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if (queued < 0) return 0;
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return queued * CHUNK_SIZE * ao_data.channels.num;
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}
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/**
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* \brief write data into buffer and reset underrun flag
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*/
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static int play(void *data, int len, int flags) {
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ALint state;
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int i, j, k;
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int ch;
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int16_t *d = data;
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len /= ao_data.channels.num * CHUNK_SIZE;
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for (i = 0; i < len; i++) {
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for (ch = 0; ch < ao_data.channels.num; ch++) {
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for (j = 0, k = ch; j < CHUNK_SIZE / 2; j++, k += ao_data.channels.num)
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tmpbuf[j] = d[k];
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alBufferData(buffers[ch][cur_buf[ch]], AL_FORMAT_MONO16, tmpbuf,
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CHUNK_SIZE, ao_data.samplerate);
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alSourceQueueBuffers(sources[ch], 1, &buffers[ch][cur_buf[ch]]);
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cur_buf[ch] = (cur_buf[ch] + 1) % NUM_BUF;
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}
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d += ao_data.channels.num * CHUNK_SIZE / 2;
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}
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alGetSourcei(sources[0], AL_SOURCE_STATE, &state);
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if (state != AL_PLAYING) // checked here in case of an underrun
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alSourcePlayv(ao_data.channels.num, sources);
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return len * ao_data.channels.num * CHUNK_SIZE;
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}
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static float get_delay(void) {
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ALint queued;
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unqueue_buffers();
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alGetSourcei(sources[0], AL_BUFFERS_QUEUED, &queued);
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return queued * CHUNK_SIZE / 2 / (float)ao_data.samplerate;
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}
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