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mpv/audio/out/ao.c

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/*
* This file is part of MPlayer.
*
* MPlayer 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.
*
* MPlayer 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 MPlayer; if not, write to the Free Software Foundation, Inc.,
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
*/
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <assert.h>
#include "talloc.h"
#include "config.h"
#include "ao.h"
#include "internal.h"
#include "audio/format.h"
#include "audio/audio.h"
#include "options/options.h"
#include "options/m_config.h"
#include "osdep/timer.h"
#include "common/msg.h"
#include "common/common.h"
#include "common/global.h"
extern const struct ao_driver audio_out_oss;
extern const struct ao_driver audio_out_coreaudio;
extern const struct ao_driver audio_out_rsound;
extern const struct ao_driver audio_out_sndio;
extern const struct ao_driver audio_out_pulse;
extern const struct ao_driver audio_out_jack;
extern const struct ao_driver audio_out_openal;
extern const struct ao_driver audio_out_null;
extern const struct ao_driver audio_out_alsa;
extern const struct ao_driver audio_out_dsound;
extern const struct ao_driver audio_out_wasapi;
extern const struct ao_driver audio_out_pcm;
extern const struct ao_driver audio_out_lavc;
extern const struct ao_driver audio_out_portaudio;
extern const struct ao_driver audio_out_sdl;
static const struct ao_driver * const audio_out_drivers[] = {
// native:
#if HAVE_COREAUDIO
&audio_out_coreaudio,
#endif
#if HAVE_PULSE
&audio_out_pulse,
#endif
#if HAVE_SNDIO
&audio_out_sndio,
#endif
#if HAVE_ALSA
&audio_out_alsa,
#endif
#if HAVE_WASAPI
&audio_out_wasapi,
#endif
#if HAVE_OSS_AUDIO
&audio_out_oss,
#endif
#if HAVE_DSOUND
&audio_out_dsound,
#endif
#if HAVE_PORTAUDIO
&audio_out_portaudio,
#endif
// wrappers:
#if HAVE_JACK
&audio_out_jack,
#endif
#if HAVE_OPENAL
&audio_out_openal,
#endif
#if HAVE_SDL1 || HAVE_SDL2
&audio_out_sdl,
#endif
&audio_out_null,
// should not be auto-selected:
&audio_out_pcm,
#if HAVE_ENCODING
&audio_out_lavc,
#endif
#if HAVE_RSOUND
&audio_out_rsound,
#endif
NULL
};
static bool get_desc(struct m_obj_desc *dst, int index)
{
if (index >= MP_ARRAY_SIZE(audio_out_drivers) - 1)
return false;
const struct ao_driver *ao = audio_out_drivers[index];
*dst = (struct m_obj_desc) {
.name = ao->name,
.description = ao->description,
.priv_size = ao->priv_size,
.priv_defaults = ao->priv_defaults,
.options = ao->options,
.hidden = ao->encode,
.p = ao,
};
return true;
}
// For the ao option
const struct m_obj_list ao_obj_list = {
.get_desc = get_desc,
.description = "audio outputs",
.allow_unknown_entries = true,
.allow_trailer = true,
};
static struct ao *ao_create(bool probing, struct mpv_global *global,
struct input_ctx *input_ctx,
struct encode_lavc_context *encode_lavc_ctx,
int samplerate, int format, struct mp_chmap channels,
char *name, char **args)
{
struct mp_log *log = mp_log_new(NULL, global->log, "ao");
struct m_obj_desc desc;
if (!m_obj_list_find(&desc, &ao_obj_list, bstr0(name))) {
mp_msg(log, MSGL_ERR, "Audio output %s not found!\n", name);
talloc_free(log);
return NULL;
};
struct ao *ao = talloc_ptrtype(NULL, ao);
talloc_steal(ao, log);
*ao = (struct ao) {
.driver = desc.p,
.probing = probing,
.encode_lavc_ctx = encode_lavc_ctx,
.input_ctx = input_ctx,
.samplerate = samplerate,
.channels = channels,
.format = format,
.log = mp_log_new(ao, log, name),
};
if (ao->driver->encode != !!ao->encode_lavc_ctx)
goto error;
struct m_config *config = m_config_from_obj_desc(ao, ao->log, &desc);
if (m_config_apply_defaults(config, name, global->opts->ao_defs) < 0)
goto error;
if (m_config_set_obj_params(config, args) < 0)
goto error;
ao->priv = config->optstruct;
char *chmap = mp_chmap_to_str(&ao->channels);
MP_VERBOSE(ao, "requested format: %d Hz, %s channels, %s\n",
ao->samplerate, chmap, af_fmt_to_str(ao->format));
talloc_free(chmap);
ao->api = ao->driver->play ? &ao_api_push : &ao_api_pull;
ao->api_priv = talloc_zero_size(ao, ao->api->priv_size);
assert(!ao->api->priv_defaults && !ao->api->options);
if (ao->driver->init(ao) < 0)
goto error;
ao->sstride = af_fmt2bits(ao->format) / 8;
ao->num_planes = 1;
if (af_fmt_is_planar(ao->format)) {
ao->num_planes = ao->channels.num;
} else {
ao->sstride *= ao->channels.num;
}
ao->bps = ao->samplerate * ao->sstride;
if (!ao->device_buffer && ao->driver->get_space) {
ao->device_buffer = ao->driver->get_space(ao);
MP_VERBOSE(ao, "device buffer: %d samples.\n", ao->device_buffer);
}
ao->buffer = MPMAX(ao->device_buffer, MIN_BUFFER * ao->samplerate);
MP_VERBOSE(ao, "using soft-buffer of %d samples.\n", ao->buffer);
if (ao->api->init(ao) < 0)
goto error;
return ao;
error:
talloc_free(ao);
return NULL;
}
struct ao *ao_init_best(struct mpv_global *global,
struct input_ctx *input_ctx,
struct encode_lavc_context *encode_lavc_ctx,
int samplerate, int format, struct mp_chmap channels)
{
struct mp_log *log = mp_log_new(NULL, global->log, "ao");
struct ao *ao = NULL;
struct m_obj_settings *ao_list = global->opts->audio_driver_list;
if (ao_list && ao_list[0].name) {
for (int n = 0; ao_list[n].name; n++) {
if (strlen(ao_list[n].name) == 0)
goto autoprobe;
mp_verbose(log, "Trying preferred audio driver '%s'\n",
ao_list[n].name);
ao = ao_create(false, global, input_ctx, encode_lavc_ctx,
samplerate, format, channels,
ao_list[n].name, ao_list[n].attribs);
if (ao)
goto done;
mp_warn(log, "Failed to initialize audio driver '%s'\n",
ao_list[n].name);
}
goto done;
}
autoprobe:
// now try the rest...
for (int i = 0; audio_out_drivers[i]; i++) {
ao = ao_create(true, global, input_ctx, encode_lavc_ctx,
samplerate, format, channels,
(char *)audio_out_drivers[i]->name, NULL);
if (ao)
goto done;
}
done:
talloc_free(log);
return ao;
}
// Uninitialize and destroy the AO. Remaining audio must be dropped.
void ao_uninit(struct ao *ao)
{
ao->api->uninit(ao);
talloc_free(ao);
}
// Queue the given audio data. Start playback if it hasn't started yet. Return
// the number of samples that was accepted (the core will try to queue the rest
// again later). Should never block.
// data: start pointer for each plane. If the audio data is packed, only
// data[0] is valid, otherwise there is a plane for each channel.
// samples: size of the audio data (see ao->sstride)
// flags: currently AOPLAY_FINAL_CHUNK can be set
int ao_play(struct ao *ao, void **data, int samples, int flags)
{
return ao->api->play(ao, data, samples, flags);
}
int ao_control(struct ao *ao, enum aocontrol cmd, void *arg)
{
switch (cmd) {
case AOCONTROL_HAS_TEMP_VOLUME:
return !ao->no_persistent_volume;
case AOCONTROL_HAS_PER_APP_VOLUME:
return !!ao->per_application_mixer;
default:
if (ao->api->control)
return ao->api->control(ao, cmd, arg);
}
return CONTROL_UNKNOWN;
}
// Return size of the buffered data in seconds. Can include the device latency.
// Basically, this returns how much data there is still to play, and how long
// it takes until the last sample in the buffer reaches the speakers. This is
// used for audio/video synchronization, so it's very important to implement
// this correctly.
double ao_get_delay(struct ao *ao)
{
return ao->api->get_delay(ao);
}
// Return free size of the internal audio buffer. This controls how much audio
// the core should decode and try to queue with ao_play().
int ao_get_space(struct ao *ao)
{
return ao->api->get_space(ao);
}
// Stop playback and empty buffers. Essentially go back to the state after
// ao->init().
void ao_reset(struct ao *ao)
{
if (ao->api->reset)
ao->api->reset(ao);
}
// Pause playback. Keep the current buffer. ao_get_delay() must return the
// same value as before pausing.
void ao_pause(struct ao *ao)
{
if (ao->api->pause)
ao->api->pause(ao);
}
// Resume playback. Play the remaining buffer. If the driver doesn't support
// pausing, it has to work around this and e.g. use ao_play_silence() to fill
// the lost audio.
void ao_resume(struct ao *ao)
{
if (ao->api->resume)
ao->api->resume(ao);
}
// Be careful with locking
void ao_wait_drain(struct ao *ao)
{
// This is probably not entirely accurate, but good enough.
mp_sleep_us(ao_get_delay(ao) * 1000000);
ao_reset(ao);
}
// Block until the current audio buffer has played completely.
void ao_drain(struct ao *ao)
{
if (ao->api->drain) {
ao->api->drain(ao);
} else {
ao_wait_drain(ao);
}
}
bool ao_eof_reached(struct ao *ao)
{
return ao_get_delay(ao) < AO_EOF_DELAY;
}
bool ao_chmap_sel_adjust(struct ao *ao, const struct mp_chmap_sel *s,
struct mp_chmap *map)
{
return mp_chmap_sel_adjust(s, map);
}
bool ao_chmap_sel_get_def(struct ao *ao, const struct mp_chmap_sel *s,
struct mp_chmap *map, int num)
{
return mp_chmap_sel_get_def(s, map, num);
}
// --- The following functions just return immutable information.
void ao_get_format(struct ao *ao, struct mp_audio *format)
{
*format = (struct mp_audio){0};
mp_audio_set_format(format, ao->format);
mp_audio_set_channels(format, &ao->channels);
format->rate = ao->samplerate;
}
const char *ao_get_name(struct ao *ao)
{
return ao->driver->name;
}
const char *ao_get_description(struct ao *ao)
{
return ao->driver->description;
}
bool ao_untimed(struct ao *ao)
{
return ao->untimed;
}
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