mirror of
https://github.com/mpv-player/mpv
synced 2024-12-22 23:02:37 +00:00
037c37519b
Before this change, AOs could have internal alignment, and play() would not consume the trailing data if the size passed to it is not aligned. Change this to require AOs to report their alignment (via period_size), and make sure to always send aligned data. The buffer reported by get_space() now always has to be correct and reliable. If play() does not consume all data provided (which is bounded by get_space()), an error is printed. This is preparation for potential further AO changes. I casually checked alsa/lavc/null/pcm, the other AOs might or might not work.
248 lines
6.5 KiB
C
248 lines
6.5 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;
|
|
};
|
|
|
|
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 = {.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),
|
|
{0}
|
|
},
|
|
.options_prefix = "ao-null",
|
|
};
|