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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 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 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;
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;
// 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;
}
case AOCONTROL_HAS_SOFT_VOLUME:
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 && alGetEnumValue((ALchar*)"AL_DIRECT_CHANNELS_SOFT")) {
alSourcei(source, alGetEnumValue((ALchar*)"AL_DIRECT_CHANNELS_SOFT"), AL_TRUE);
}
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->period_size = p->num_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(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;
}
}
/**
* \brief stop playing and empty buffers (for seeking/pause)
*/
static void reset(struct ao *ao)
{
alSourceStop(source);
unqueue_buffers(ao);
}
/**
* \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)
{
struct priv *p = ao->priv;
ALint queued;
unqueue_buffers(ao);
alGetSourcei(source, AL_BUFFERS_QUEUED, &queued);
queued = p->num_buffers - queued;
if (queued < 0)
return 0;
return p->num_samples * queued;
}
/**
* \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;
int buffered_samples = 0;
int num = 0;
if (flags & AOPLAY_FINAL_CHUNK) {
num = 1;
buffered_samples = samples;
} else {
num = samples / p->num_samples;
buffered_samples = num * p->num_samples;
}
for (int i = 0; i < num; i++) {
char *d = *data;
if (flags & AOPLAY_FINAL_CHUNK) {
buffer_size[cur_buf] = samples;
} else {
buffer_size[cur_buf] = 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;
}
ALint state;
alGetSourcei(source, AL_SOURCE_STATE, &state);
if (state != AL_PLAYING) // checked here in case of an underrun
alSourcePlay(source);
return buffered_samples;
}
static double get_delay(struct ao *ao)
{
struct priv *p = ao->priv;
ALint queued;
unqueue_buffers(ao);
alGetSourcei(source, AL_BUFFERS_QUEUED, &queued);
double soft_source_latency = 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)
soft_source_latency = offsets[1] - offsets[0];
} else {
float offset = 0;
alGetSourcef(source, AL_SEC_OFFSET, &offset);
soft_source_latency = -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;
}
return (queued_samples / (double)ao->samplerate) + soft_source_latency;
}
#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),
.priv_defaults = &(const struct priv) {
.num_buffers = 4,
.num_samples = 8192,
.direct_channels = 0,
},
.options = (const struct m_option[]) {
OPT_INTRANGE("num-buffers", num_buffers, 0, 2, MAX_BUF),
OPT_INTRANGE("num-samples", num_samples, 0, 256, MAX_SAMPLES),
OPT_FLAG("direct-channels", direct_channels, 0),
{0}
},
.options_prefix = "openal",
};
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