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dd357a7d53
mpv/audio/out/ao_openal.c
LAGonauta dd357a7d53 ao/openal: Add support for direct channels output 
Uses OpenAL Soft's AL_DIRECT_CHANNELS_SOFT extension and can be controlled through
a new CLI option, --openal-direct-channels.
This allows one to send the audio data direrctly to the desired channel without
effects applied.
2018-04-15 00:57:01 +03:00

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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 Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 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 Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser 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[NUM_BUF];
static ALuint source;
static int cur_buf;
static int unqueue_buf;
static struct ao *ao_data;
struct priv {
ALenum al_format;
int chunk_size;
int direct_channels;
};
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_GET_MUTE:
case AOCONTROL_SET_MUTE: {
bool mute = *(bool *)arg;
ALfloat al_mute = (ALfloat)(!mute);
if (cmd == AOCONTROL_SET_MUTE) {
alSourcef(source, AL_GAIN, al_mute);
}
alGetSourcef(source, AL_GAIN, &al_mute);
*(bool *)arg = !((bool)al_mute);
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 enum af_format get_af_format(int format)
{
switch (format) {
case AF_FORMAT_U8:
if (alGetEnumValue("AL_FORMAT_MONO8"))
return AL_TRUE;
break;
case AF_FORMAT_S16:
if (alGetEnumValue("AL_FORMAT_MONO16"))
return AL_TRUE;
break;
case AF_FORMAT_S32:
if (strstr(alGetString(AL_RENDERER), "X-Fi") != NULL)
return AL_TRUE;
break;
case AF_FORMAT_FLOAT:
if (alIsExtensionPresent((ALchar*)"AL_EXT_float32") == AL_TRUE)
return AL_TRUE;
break;
case AF_FORMAT_DOUBLE:
if (alIsExtensionPresent((ALchar*)"AL_EXT_double") == AL_TRUE)
return AL_TRUE;
break;
}
return AL_FALSE;
}
static ALenum get_al_format(struct ao *ao, int format)
{
switch (format) {
case AF_FORMAT_U8:
switch (ao->channels.num) {
case 8:
if (alGetEnumValue("AL_FORMAT_71CHN8")) {
return alGetEnumValue("AL_FORMAT_71CHN8");
}
case 7:
if (alGetEnumValue("AL_FORMAT_61CHN8")) {
return alGetEnumValue("AL_FORMAT_61CHN8");
}
case 6:
if (alGetEnumValue("AL_FORMAT_51CHN8")) {
return alGetEnumValue("AL_FORMAT_51CHN8");
}
case 4:
if (alGetEnumValue("AL_FORMAT_QUAD8")) {
return alGetEnumValue("AL_FORMAT_QUAD8");
}
case 2:
if (alGetEnumValue("AL_FORMAT_STEREO8")) {
return alGetEnumValue("AL_FORMAT_STEREO8");
}
default:
return alGetEnumValue("AL_FORMAT_MONO8");
}
case AF_FORMAT_S16:
switch (ao->channels.num) {
case 8:
if (alGetEnumValue("AL_FORMAT_71CHN16")) {
return alGetEnumValue("AL_FORMAT_71CHN16");
}
case 7:
if (alGetEnumValue("AL_FORMAT_61CHN16")) {
return alGetEnumValue("AL_FORMAT_61CHN16");
}
case 6:
if (alGetEnumValue("AL_FORMAT_51CHN16")) {
return alGetEnumValue("AL_FORMAT_51CHN16");
}
case 4:
if (alGetEnumValue("AL_FORMAT_QUAD16")) {
return alGetEnumValue("AL_FORMAT_QUAD16");
}
case 2:
if (alGetEnumValue("AL_FORMAT_STEREO16")) {
return alGetEnumValue("AL_FORMAT_STEREO16");
}
default:
return alGetEnumValue("AL_FORMAT_MONO16");
}
case AF_FORMAT_S32:
if (strstr(alGetString(AL_RENDERER), "X-Fi") != NULL) {
switch (ao->channels.num) {
case 8:
if (alGetEnumValue("AL_FORMAT_71CHN32")) {
return alGetEnumValue("AL_FORMAT_71CHN32");
}
break;
case 7:
if (alGetEnumValue("AL_FORMAT_61CHN32")) {
return alGetEnumValue("AL_FORMAT_61CHN32");
}
break;
case 6:
if (alGetEnumValue("AL_FORMAT_51CHN32")) {
return alGetEnumValue("AL_FORMAT_51CHN32");
}
break;
case 4:
if (alGetEnumValue("AL_FORMAT_QUAD32")) {
return alGetEnumValue("AL_FORMAT_QUAD32");
}
break;
case 2:
if (alGetEnumValue("AL_FORMAT_STEREO32")) {
return alGetEnumValue("AL_FORMAT_STEREO32");
}
default:
return alGetEnumValue("AL_FORMAT_MONO32");
}
}
}
return AL_FALSE;
}
// close audio device
static void uninit(struct ao *ao)
{
alSourceStop(source);
alSourcei(source, AL_BUFFER, 0);
alDeleteBuffers(NUM_BUF, buffers);
alDeleteSources(1, &source);
ALCcontext *ctx = alcGetCurrentContext();
ALCdevice *dev = alcGetContextsDevice(ctx);
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(1, &source);
if (p->direct_channels && alGetEnumValue((ALchar*)"AL_DIRECT_CHANNELS_SOFT")) {
alSourcei(source, alGetEnumValue((ALchar*)"AL_DIRECT_CHANNELS_SOFT"), AL_TRUE);
}
cur_buf = 0;
unqueue_buf = 0;
alGenBuffers(NUM_BUF, buffers);
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 + 1];
af_get_best_sample_formats(ao->format, try_formats);
for (int n = 0; try_formats[n]; n++) {
p->al_format = get_al_format(ao, 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);
ao->period_size = CHUNK_SAMPLES;
return 0;
err_out:
ao_data = NULL;
return -1;
}
static void drain(struct ao *ao)
{
ALint state;
alGetSourcei(source, AL_SOURCE_STATE, &state);
while (state == AL_PLAYING) {
mp_sleep_us(10000);
alGetSourcei(source, AL_SOURCE_STATE, &state);
}
}
static void unqueue_buffers(void)
{
ALint p;
int till_wrap = NUM_BUF - unqueue_buf;
alGetSourcei(source, AL_BUFFERS_PROCESSED, &p);
if (p >= till_wrap) {
alSourceUnqueueBuffers(source, till_wrap, &buffers[unqueue_buf]);
unqueue_buf = 0;
p -= till_wrap;
}
if (p) {
alSourceUnqueueBuffers(source, p, &buffers[unqueue_buf]);
unqueue_buf += p;
}
}
/**
* \brief stop playing and empty buffers (for seeking/pause)
*/
static void reset(struct ao *ao)
{
alSourceStop(source);
unqueue_buffers();
}
/**
* \brief stop playing, keep buffers (for pause)
*/
static void audio_pause(struct ao *ao)
{
alSourcePause(source);
}
/**
* \brief resume playing, after audio_pause()
*/
static void audio_resume(struct ao *ao)
{
alSourcePlay(source);
}
static int get_space(struct ao *ao)
{
ALint queued;
unqueue_buffers();
alGetSourcei(source, 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++) {
char *d = data[0];
d += i * p->chunk_size * ao->channels.num;
alBufferData(buffers[cur_buf], p->al_format, d, p->chunk_size * ao->channels.num, ao->samplerate);
alSourceQueueBuffers(source, 1, &buffers[cur_buf]);
cur_buf = (cur_buf + 1) % NUM_BUF;
}
alGetSourcei(source, AL_SOURCE_STATE, &state);
if (state != AL_PLAYING) // checked here in case of an underrun
alSourcePlay(source);
return num * CHUNK_SAMPLES;
}
static double get_delay(struct ao *ao)
{
ALint queued;
unqueue_buffers();
alGetSourcei(source, 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),
.options = (const struct m_option[]) {
OPT_FLAG("direct-channels", direct_channels, 0),
{0}
},
.options_prefix = "openal",
};
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