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mirror of https://github.com/mpv-player/mpv.git synced 2024-09-20 03:52:22 +02:00
mpv/audio/out/ao_null.c
wm4 478d39c574 audio: fix inefficient behavior with ao_alsa, remove period_size field
It is now the AO's responsibility to handle period size alignment. The
ao->period_size alignment field is unused as of the recent audio
refactor commit. Remove it.

It turns out that ao_alsa shows extremely inefficient behavior as a
consequence of the removal of period size aligned writes in the
mentioned refactor commit. This is because it could get into a state
where it repeatedly wrote single samples (as small as 1 sample), and
starved the rest of the player as a consequence. Too bad. Explicitly
align the size in ao_alsa. Other AOs, which need this, should do the
same.

One reason why it broke so badly with ao_alsa was that it retried the
write() even if all reported space could be written. So stop doing that
too. Retry the write only if we somehow wrote less.

I'm not sure about ao_pulse.
2020-08-29 16:27:56 +02:00

232 lines
6.1 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;
float buffered; // samples
int buffersize; // samples
bool playing;
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)
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;
ao->device_buffer = priv->outburst * bursts + priv->latency;
priv->last_time = mp_time_sec();
return 0;
}
// close audio device
static void uninit(struct ao *ao)
{
}
// stop playing and empty buffers (for seeking/pause)
static void reset(struct ao *ao)
{
struct priv *priv = ao->priv;
priv->buffered = 0;
priv->playing = false;
}
static void start(struct ao *ao)
{
struct priv *priv = ao->priv;
if (priv->paused)
MP_ERR(ao, "illegal state: start() while paused\n");
drain(ao);
priv->paused = false;
priv->last_time = mp_time_sec();
priv->playing = true;
}
static bool set_pause(struct ao *ao, bool paused)
{
struct priv *priv = ao->priv;
if (!priv->playing)
MP_ERR(ao, "illegal state: set_pause() while not playing\n");
if (priv->paused != paused) {
drain(ao);
priv->paused = paused;
if (!priv->paused)
priv->last_time = mp_time_sec();
}
return true;
}
static bool audio_write(struct ao *ao, void **data, int samples)
{
struct priv *priv = ao->priv;
if (priv->buffered <= 0)
priv->buffered = priv->latency; // emulate fixed latency
priv->buffered += samples;
return true;
}
static void get_state(struct ao *ao, struct mp_pcm_state *state)
{
struct priv *priv = ao->priv;
drain(ao);
state->free_samples = ao->device_buffer - priv->latency - priv->buffered;
state->free_samples = state->free_samples / priv->outburst * priv->outburst;
state->queued_samples = priv->buffered;
// Note how get_state returns the delay in audio device time (instead of
// adjusting for speed), since most AOs seem to also do that.
state->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)
state->delay = priv->latency;
state->delay /= ao->samplerate;
if (priv->broken_delay) { // Report only multiples of outburst
double q = priv->outburst / (double)ao->samplerate;
if (state->delay > 0)
state->delay = (int)(state->delay / q) * q;
}
state->playing = priv->playing && priv->buffered > 0;
}
#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_state = get_state,
.set_pause = set_pause,
.write = audio_write,
.start = start,
.priv_size = sizeof(struct priv),
.priv_defaults = &(const struct priv) {
.bufferlen = 0.2,
.outburst = 256,
.speed = 1,
},
.options = (const struct m_option[]) {
{"untimed", OPT_FLAG(untimed)},
{"buffer", OPT_FLOAT(bufferlen), M_RANGE(0, 100)},
{"outburst", OPT_INT(outburst), M_RANGE(1, 100000)},
{"speed", OPT_FLOAT(speed), M_RANGE(0, 10000)},
{"latency", OPT_FLOAT(latency_sec), M_RANGE(0, 100)},
{"broken-eof", OPT_FLAG(broken_eof)},
{"broken-delay", OPT_FLAG(broken_delay)},
{"channel-layouts", OPT_CHANNELS(channel_layouts)},
{"format", OPT_AUDIOFORMAT(format)},
{0}
},
.options_prefix = "ao-null",
};