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mpv/audio/out/ao_openal.c

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/*
* 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 <http://www.gnu.org/licenses/>.
*/
#include "config.h"
#include <stdlib.h>
#include <stdio.h>
#include <inttypes.h>
#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 <OpenAL/MacOSX_OALExtensions.h>
#else
#ifdef OPENAL_AL_H
#include <OpenAL/alc.h>
#include <OpenAL/al.h>
#include <OpenAL/alext.h>
#else
#include <AL/alc.h>
#include <AL/al.h>
#include <AL/alext.h>
#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 {
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;
}
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 = 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,
.priv_size = sizeof(struct priv),
};
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