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mirror of https://github.com/mpv-player/mpv synced 2024-11-14 22:48:35 +01:00
mpv/audio/out/ao_null.c
wm4 054c02ad64 ao_null: add --ao-null-format option for debugging
Helpful especially to test spdif fallback and so on.
2018-01-30 03:10:27 -08:00

253 lines
6.7 KiB
C

/*
* null audio output driver
*
* 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/>.
*/
/*
* Note: this does much more than just ignoring audio output. It simulates
* (to some degree) an ideal AO.
*/
#include <stdio.h>
#include <stdlib.h>
#include <math.h>
#include "mpv_talloc.h"
#include "config.h"
#include "osdep/timer.h"
#include "options/m_option.h"
#include "common/common.h"
#include "common/msg.h"
#include "audio/format.h"
#include "ao.h"
#include "internal.h"
struct priv {
bool paused;
double last_time;
bool playing_final;
float buffered; // samples
int buffersize; // samples
int untimed;
float bufferlen; // seconds
float speed; // multiplier
float latency_sec; // seconds
float latency; // samples
int broken_eof;
int broken_delay;
// Minimal unit of audio samples that can be written at once. If play() is
// called with sizes not aligned to this, a rounded size will be returned.
// (This is not needed by the AO API, but many AOs behave this way.)
int outburst; // samples
struct m_channels channel_layouts;
int format;
};
static void drain(struct ao *ao)
{
struct priv *priv = ao->priv;
if (ao->untimed) {
priv->buffered = 0;
return;
}
if (priv->paused)
return;
double now = mp_time_sec();
if (priv->buffered > 0) {
priv->buffered -= (now - priv->last_time) * ao->samplerate * priv->speed;
if (priv->buffered < 0) {
if (!priv->playing_final)
MP_ERR(ao, "buffer underrun\n");
priv->buffered = 0;
}
}
priv->last_time = now;
}
static int init(struct ao *ao)
{
struct priv *priv = ao->priv;
if (priv->format)
ao->format = priv->format;
ao->untimed = priv->untimed;
struct mp_chmap_sel sel = {.tmp = ao};
if (priv->channel_layouts.num_chmaps) {
for (int n = 0; n < priv->channel_layouts.num_chmaps; n++)
mp_chmap_sel_add_map(&sel, &priv->channel_layouts.chmaps[n]);
} else {
mp_chmap_sel_add_any(&sel);
}
if (!ao_chmap_sel_adjust(ao, &sel, &ao->channels))
mp_chmap_from_channels(&ao->channels, 2);
priv->latency = priv->latency_sec * ao->samplerate;
// A "buffer" for this many seconds of audio
int bursts = (int)(ao->samplerate * priv->bufferlen + 1) / priv->outburst;
priv->buffersize = priv->outburst * bursts + priv->latency;
priv->last_time = mp_time_sec();
ao->period_size = priv->outburst;
return 0;
}
// close audio device
static void uninit(struct ao *ao)
{
}
static void wait_drain(struct ao *ao)
{
struct priv *priv = ao->priv;
drain(ao);
if (!priv->paused)
mp_sleep_us(1000000.0 * priv->buffered / ao->samplerate / priv->speed);
}
// stop playing and empty buffers (for seeking/pause)
static void reset(struct ao *ao)
{
struct priv *priv = ao->priv;
priv->buffered = 0;
priv->playing_final = false;
}
// stop playing, keep buffers (for pause)
static void pause(struct ao *ao)
{
struct priv *priv = ao->priv;
drain(ao);
priv->paused = true;
}
// resume playing, after pause()
static void resume(struct ao *ao)
{
struct priv *priv = ao->priv;
drain(ao);
priv->paused = false;
priv->last_time = mp_time_sec();
}
static int get_space(struct ao *ao)
{
struct priv *priv = ao->priv;
drain(ao);
int samples = priv->buffersize - priv->latency - priv->buffered;
return samples / priv->outburst * priv->outburst;
}
static int play(struct ao *ao, void **data, int samples, int flags)
{
struct priv *priv = ao->priv;
int accepted;
resume(ao);
if (priv->buffered <= 0)
priv->buffered = priv->latency; // emulate fixed latency
priv->playing_final = flags & AOPLAY_FINAL_CHUNK;
if (priv->playing_final) {
// Last audio chunk - don't round to outburst.
accepted = MPMIN(priv->buffersize - priv->buffered, samples);
} else {
int maxbursts = (priv->buffersize - priv->buffered) / priv->outburst;
int playbursts = samples / priv->outburst;
int bursts = playbursts > maxbursts ? maxbursts : playbursts;
accepted = bursts * priv->outburst;
}
priv->buffered += accepted;
return accepted;
}
static double get_delay(struct ao *ao)
{
struct priv *priv = ao->priv;
drain(ao);
// Note how get_delay returns the delay in audio device time (instead of
// adjusting for speed), since most AOs seem to also do that.
double delay = priv->buffered;
// Drivers with broken EOF handling usually always report the same device-
// level delay that is additional to the buffer time.
if (priv->broken_eof && priv->buffered < priv->latency)
delay = priv->latency;
delay /= ao->samplerate;
if (priv->broken_delay) { // Report only multiples of outburst
double q = priv->outburst / (double)ao->samplerate;
if (delay > 0)
delay = (int)(delay / q) * q;
}
return delay;
}
#define OPT_BASE_STRUCT struct priv
const struct ao_driver audio_out_null = {
.description = "Null audio output",
.name = "null",
.init = init,
.uninit = uninit,
.reset = reset,
.get_space = get_space,
.play = play,
.get_delay = get_delay,
.pause = pause,
.resume = resume,
.drain = wait_drain,
.priv_size = sizeof(struct priv),
.priv_defaults = &(const struct priv) {
.bufferlen = 0.2,
.outburst = 256,
.speed = 1,
},
.options = (const struct m_option[]) {
OPT_FLAG("untimed", untimed, 0),
OPT_FLOATRANGE("buffer", bufferlen, 0, 0, 100),
OPT_INTRANGE("outburst", outburst, 0, 1, 100000),
OPT_FLOATRANGE("speed", speed, 0, 0, 10000),
OPT_FLOATRANGE("latency", latency_sec, 0, 0, 100),
OPT_FLAG("broken-eof", broken_eof, 0),
OPT_FLAG("broken-delay", broken_delay, 0),
OPT_CHANNELS("channel-layouts", channel_layouts, 0),
OPT_AUDIOFORMAT("format", format, 0),
{0}
},
.options_prefix = "ao-null",
};