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mpv/audio/out/ao_openal.c
Christoph Heinrich 91cc0d8cf6 options: transition options from OPT_FLAG to OPT_BOOL 
c784820454 introduced a bool option type
as a replacement for the flag type, but didn't actually transition and
remove the flag type because it would have been too much mundane work.
2023-02-21 17:15:17 +00:00

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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 <stdlib.h>
#include <stdio.h>
#include <inttypes.h>
#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
#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 MAX_BUF 128
#define MAX_SAMPLES 32768
static ALuint buffers[MAX_BUF];
static ALuint buffer_size[MAX_BUF];
static ALuint source;
static int cur_buf;
static int unqueue_buf;
static struct ao *ao_data;
struct priv {
ALenum al_format;
int num_buffers;
int num_samples;
bool direct_channels;
};
static int control(struct ao *ao, enum aocontrol cmd, void *arg)
{
switch (cmd) {
case AOCONTROL_GET_VOLUME:
case AOCONTROL_SET_VOLUME: {
ALfloat volume;
float *vol = arg;
if (cmd == AOCONTROL_SET_VOLUME) {
volume = *vol / 100.0;
alListenerf(AL_GAIN, volume);
}
alGetListenerf(AL_GAIN, &volume);
*vol = volume * 100;
return CONTROL_TRUE;
}
case AOCONTROL_GET_MUTE:
case AOCONTROL_SET_MUTE: {
bool mute = *(bool *)arg;
// openal has no mute control, only gain.
// Thus reverse the muted state to get required gain
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;
}
}
return CONTROL_UNKNOWN;
}
static enum af_format get_supported_format(int format)
{
switch (format) {
case AF_FORMAT_U8:
if (alGetEnumValue((ALchar*)"AL_FORMAT_MONO8"))
return AF_FORMAT_U8;
break;
case AF_FORMAT_S16:
if (alGetEnumValue((ALchar*)"AL_FORMAT_MONO16"))
return AF_FORMAT_S16;
break;
case AF_FORMAT_S32:
if (strstr(alGetString(AL_RENDERER), "X-Fi") != NULL)
return AF_FORMAT_S32;
break;
case AF_FORMAT_FLOAT:
if (alIsExtensionPresent((ALchar*)"AL_EXT_float32") == AL_TRUE)
return AF_FORMAT_FLOAT;
break;
}
return AL_FALSE;
}
static ALenum get_supported_layout(int format, int channels)
{
const char *channel_str[] = {
[1] = "MONO",
[2] = "STEREO",
[4] = "QUAD",
[6] = "51CHN",
[7] = "61CHN",
[8] = "71CHN",
};
const char *format_str[] = {
[AF_FORMAT_U8] = "8",
[AF_FORMAT_S16] = "16",
[AF_FORMAT_S32] = "32",
[AF_FORMAT_FLOAT] = "_FLOAT32",
};
if (channel_str[channels] == NULL || format_str[format] == NULL)
return AL_FALSE;
char enum_name[32];
// AF_FORMAT_FLOAT uses same enum name as AF_FORMAT_S32 for multichannel
// playback, while it is different for mono and stereo.
// OpenAL Soft does not support AF_FORMAT_S32 and seems to reuse the names.
if (channels > 2 && format == AF_FORMAT_FLOAT)
format = AF_FORMAT_S32;
snprintf(enum_name, sizeof(enum_name), "AL_FORMAT_%s%s", channel_str[channels],
format_str[format]);
if (alGetEnumValue((ALchar*)enum_name)) {
return alGetEnumValue((ALchar*)enum_name);
}
return AL_FALSE;
}
// close audio device
static void uninit(struct ao *ao)
{
struct priv *p = ao->priv;
alSourceStop(source);
alSourcei(source, AL_BUFFER, 0);
alDeleteBuffers(p->num_buffers, 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};
struct priv *p = ao->priv;
if (ao_data) {
MP_FATAL(ao, "Not reentrant!\n");
return -1;
}
ao_data = ao;
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) {
if (alIsExtensionPresent("AL_SOFT_direct_channels_remix")) {
alSourcei(source,
alGetEnumValue((ALchar*)"AL_DIRECT_CHANNELS_SOFT"),
alcGetEnumValue(dev, "AL_REMIX_UNMATCHED_SOFT"));
} else {
MP_WARN(ao, "Direct channels aren't supported by this version of OpenAL\n");
}
}
cur_buf = 0;
unqueue_buf = 0;
for (int i = 0; i < p->num_buffers; ++i) {
buffer_size[i] = 0;
}
alGenBuffers(p->num_buffers, buffers);
alcGetIntegerv(dev, ALC_FREQUENCY, 1, &freq);
if (alcGetError(dev) == ALC_NO_ERROR && freq)
ao->samplerate = freq;
// Check sample format
int try_formats[AF_FORMAT_COUNT + 1];
enum af_format sample_format = 0;
af_get_best_sample_formats(ao->format, try_formats);
for (int n = 0; try_formats[n]; n++) {
sample_format = get_supported_format(try_formats[n]);
if (sample_format != AF_FORMAT_UNKNOWN) {
ao->format = try_formats[n];
break;
}
}
if (sample_format == AF_FORMAT_UNKNOWN) {
MP_FATAL(ao, "Can't find appropriate sample format.\n");
uninit(ao);
goto err_out;
}
// Check if OpenAL driver supports the desired number of channels.
int num_channels = ao->channels.num;
do {
p->al_format = get_supported_layout(sample_format, num_channels);
if (p->al_format == AL_FALSE) {
num_channels = num_channels - 1;
}
} while (p->al_format == AL_FALSE && num_channels > 1);
// Request number of speakers for output from ao.
const struct mp_chmap possible_layouts[] = {
{0}, // empty
MP_CHMAP_INIT_MONO, // mono
MP_CHMAP_INIT_STEREO, // stereo
{0}, // 2.1
MP_CHMAP4(FL, FR, BL, BR), // 4.0
{0}, // 5.0
MP_CHMAP6(FL, FR, FC, LFE, BL, BR), // 5.1
MP_CHMAP7(FL, FR, FC, LFE, SL, SR, BC), // 6.1
MP_CHMAP8(FL, FR, FC, LFE, BL, BR, SL, SR), // 7.1
};
ao->channels = possible_layouts[num_channels];
if (!ao->channels.num)
mp_chmap_set_unknown(&ao->channels, num_channels);
if (p->al_format == AL_FALSE || !mp_chmap_is_valid(&ao->channels)) {
MP_FATAL(ao, "Can't find appropriate channel layout.\n");
uninit(ao);
goto err_out;
}
ao->device_buffer = p->num_buffers * p->num_samples;
return 0;
err_out:
ao_data = NULL;
return -1;
}
static void unqueue_buffers(struct ao *ao)
{
struct priv *q = ao->priv;
ALint p;
int till_wrap = q->num_buffers - 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;
}
}
static void reset(struct ao *ao)
{
alSourceStop(source);
unqueue_buffers(ao);
}
static bool audio_set_pause(struct ao *ao, bool pause)
{
if (pause) {
alSourcePause(source);
} else {
alSourcePlay(source);
}
return true;
}
static bool audio_write(struct ao *ao, void **data, int samples)
{
struct priv *p = ao->priv;
int num = (samples + p->num_samples - 1) / p->num_samples;
for (int i = 0; i < num; i++) {
char *d = *data;
buffer_size[cur_buf] =
MPMIN(samples - i * p->num_samples, p->num_samples);
d += i * buffer_size[cur_buf] * ao->sstride;
alBufferData(buffers[cur_buf], p->al_format, d,
buffer_size[cur_buf] * ao->sstride, ao->samplerate);
alSourceQueueBuffers(source, 1, &buffers[cur_buf]);
cur_buf = (cur_buf + 1) % p->num_buffers;
}
return true;
}
static void audio_start(struct ao *ao)
{
alSourcePlay(source);
}
static void get_state(struct ao *ao, struct mp_pcm_state *state)
{
struct priv *p = ao->priv;
ALint queued;
unqueue_buffers(ao);
alGetSourcei(source, AL_BUFFERS_QUEUED, &queued);
double source_offset = 0;
if(alIsExtensionPresent("AL_SOFT_source_latency")) {
ALdouble offsets[2];
LPALGETSOURCEDVSOFT alGetSourcedvSOFT = alGetProcAddress("alGetSourcedvSOFT");
alGetSourcedvSOFT(source, AL_SEC_OFFSET_LATENCY_SOFT, offsets);
// Additional latency to the play buffer, the remaining seconds to be
// played minus the offset (seconds already played)
source_offset = offsets[1] - offsets[0];
} else {
float offset = 0;
alGetSourcef(source, AL_SEC_OFFSET, &offset);
source_offset = -offset;
}
int queued_samples = 0;
for (int i = 0, index = cur_buf; i < queued; ++i) {
queued_samples += buffer_size[index];
index = (index + 1) % p->num_buffers;
}
state->delay = queued_samples / (double)ao->samplerate + source_offset;
state->queued_samples = queued_samples;
state->free_samples = MPMAX(p->num_buffers - queued, 0) * p->num_samples;
ALint source_state = 0;
alGetSourcei(source, AL_SOURCE_STATE, &source_state);
state->playing = source_state == AL_PLAYING;
}
#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_state = get_state,
.write = audio_write,
.start = audio_start,
.set_pause = audio_set_pause,
.reset = reset,
.priv_size = sizeof(struct priv),
.priv_defaults = &(const struct priv) {
.num_buffers = 4,
.num_samples = 8192,
.direct_channels = true,
},
.options = (const struct m_option[]) {
{"num-buffers", OPT_INT(num_buffers), M_RANGE(2, MAX_BUF)},
{"num-samples", OPT_INT(num_samples), M_RANGE(256, MAX_SAMPLES)},
{"direct-channels", OPT_BOOL(direct_channels)},
{0}
},
.options_prefix = "openal",
};
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