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mpv/audio/out/ao_null.c
wm4 769ac6fb7b audio/out: always round get_space on period size
Round get_space() results in the same way play() rounds the input size.
Some audio APIs do this for various reasons.

This affects only "push" based AOs. Some of these need no change,
because they either do it already right (like ao_openal), or they seem
not to have any such requirements (like ao_pulse).

Needed for the following commit.
2014-09-06 12:59:00 +02:00

227 lines
5.9 KiB
C

/*
* null audio output driver
*
* 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.
*/
/*
* 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 "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;
// 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
};
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;
ao->untimed = priv->untimed;
struct mp_chmap_sel sel = {0};
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();
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 float 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;
return delay / (double)ao->samplerate;
}
#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),
{0}
},
};