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mirror of https://github.com/mpv-player/mpv synced 2024-12-28 01:52:19 +00:00
mpv/audio/out/push.c
wm4 040c050f2d audio: fix the exact value that is used for the wait time
The comment says that it wakes up the main thread if 50% has been
played, but in reality the value was 0.74/2 => 37.5%. Correct this. This
probably changes little, because it's a very fuzzy heuristic in the
first place.

Also move down the min_wait calculation to where it's actually used.
2014-05-04 20:41:00 +02:00

351 lines
10 KiB
C

/*
* This file is part of mpv.
*
* mpv 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.
*
* 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 General Public License for more details.
*
* You should have received a copy of the GNU General Public License along
* with mpv. If not, see <http://www.gnu.org/licenses/>.
*/
#include <stddef.h>
#include <pthread.h>
#include <inttypes.h>
#include <limits.h>
#include <assert.h>
#include "ao.h"
#include "internal.h"
#include "audio/format.h"
#include "common/msg.h"
#include "common/common.h"
#include "input/input.h"
#include "osdep/threads.h"
#include "osdep/timer.h"
#include "compat/atomics.h"
#include "audio/audio.h"
#include "audio/audio_buffer.h"
struct ao_push_state {
pthread_t thread;
pthread_mutex_t lock;
// uses a separate lock to avoid lock order issues with ao_need_data()
pthread_mutex_t wakeup_lock;
pthread_cond_t wakeup;
// --- protected by lock
struct mp_audio_buffer *buffer;
bool terminate;
bool playing;
// Whether the current buffer contains the complete audio.
bool final_chunk;
double expected_end_time;
// -- protected by wakeup_lock
bool need_wakeup;
};
static void wakeup_playthread(struct ao *ao)
{
struct ao_push_state *p = ao->api_priv;
pthread_mutex_lock(&p->wakeup_lock);
p->need_wakeup = true;
pthread_cond_signal(&p->wakeup);
pthread_mutex_unlock(&p->wakeup_lock);
}
static int control(struct ao *ao, enum aocontrol cmd, void *arg)
{
int r = CONTROL_UNKNOWN;
if (ao->driver->control) {
struct ao_push_state *p = ao->api_priv;
pthread_mutex_lock(&p->lock);
r = ao->driver->control(ao, cmd, arg);
pthread_mutex_unlock(&p->lock);
}
return r;
}
static float get_delay(struct ao *ao)
{
struct ao_push_state *p = ao->api_priv;
pthread_mutex_lock(&p->lock);
double driver_delay = 0;
if (ao->driver->get_delay)
driver_delay = ao->driver->get_delay(ao);
double delay = driver_delay + mp_audio_buffer_seconds(p->buffer);
pthread_mutex_unlock(&p->lock);
if (delay >= AO_EOF_DELAY && p->expected_end_time) {
if (mp_time_sec() > p->expected_end_time) {
MP_ERR(ao, "Audio device EOF reporting is broken!\n");
MP_ERR(ao, "Please report this problem.\n");
delay = 0;
}
}
return delay;
}
static void reset(struct ao *ao)
{
struct ao_push_state *p = ao->api_priv;
pthread_mutex_lock(&p->lock);
if (ao->driver->reset)
ao->driver->reset(ao);
mp_audio_buffer_clear(p->buffer);
p->playing = false;
wakeup_playthread(ao);
pthread_mutex_unlock(&p->lock);
}
static void pause(struct ao *ao)
{
struct ao_push_state *p = ao->api_priv;
pthread_mutex_lock(&p->lock);
if (ao->driver->pause)
ao->driver->pause(ao);
p->playing = false;
wakeup_playthread(ao);
pthread_mutex_unlock(&p->lock);
}
static void resume(struct ao *ao)
{
struct ao_push_state *p = ao->api_priv;
pthread_mutex_lock(&p->lock);
if (ao->driver->resume)
ao->driver->resume(ao);
p->playing = true; // tentatively
p->expected_end_time = 0;
wakeup_playthread(ao);
pthread_mutex_unlock(&p->lock);
}
static void drain(struct ao *ao)
{
if (ao->driver->drain) {
struct ao_push_state *p = ao->api_priv;
pthread_mutex_lock(&p->lock);
ao->driver->drain(ao);
pthread_mutex_unlock(&p->lock);
} else {
ao_wait_drain(ao);
}
}
static int get_space(struct ao *ao)
{
struct ao_push_state *p = ao->api_priv;
pthread_mutex_lock(&p->lock);
int space = mp_audio_buffer_get_write_available(p->buffer);
if (ao->driver->get_space) {
// The following code attempts to keep the total buffered audio to
// MIN_BUFFER in order to improve latency.
int device_space = ao->driver->get_space(ao);
int device_buffered = ao->device_buffer - device_space;
int soft_buffered = mp_audio_buffer_samples(p->buffer);
int min_buffer = MIN_BUFFER * ao->samplerate;
int missing = min_buffer - device_buffered - soft_buffered;
// But always keep the device's buffer filled as much as we can.
int device_missing = device_space - soft_buffered;
missing = MPMAX(missing, device_missing);
space = MPMIN(space, missing);
space = MPMAX(0, space);
}
pthread_mutex_unlock(&p->lock);
return space;
}
static int play(struct ao *ao, void **data, int samples, int flags)
{
struct ao_push_state *p = ao->api_priv;
pthread_mutex_lock(&p->lock);
int write_samples = mp_audio_buffer_get_write_available(p->buffer);
write_samples = MPMIN(write_samples, samples);
struct mp_audio audio;
mp_audio_buffer_get_format(p->buffer, &audio);
for (int n = 0; n < ao->num_planes; n++)
audio.planes[n] = data[n];
audio.samples = write_samples;
mp_audio_buffer_append(p->buffer, &audio);
p->final_chunk = !!(flags & AOPLAY_FINAL_CHUNK);
p->playing = true;
p->expected_end_time = 0;
wakeup_playthread(ao);
pthread_mutex_unlock(&p->lock);
return write_samples;
}
// called locked
static int ao_play_data(struct ao *ao)
{
struct ao_push_state *p = ao->api_priv;
struct mp_audio data;
mp_audio_buffer_peek(p->buffer, &data);
int max = data.samples;
int space = ao->driver->get_space ? ao->driver->get_space(ao) : INT_MAX;
if (data.samples > space)
data.samples = space;
if (data.samples <= 0)
return 0;
int flags = 0;
if (p->final_chunk && data.samples == max)
flags |= AOPLAY_FINAL_CHUNK;
int r = ao->driver->play(ao, data.planes, data.samples, flags);
if (r > data.samples) {
MP_WARN(ao, "Audio device returned non-sense value.");
r = data.samples;
}
if (r > 0)
mp_audio_buffer_skip(p->buffer, r);
if (p->final_chunk && mp_audio_buffer_samples(p->buffer) == 0) {
p->playing = false;
p->expected_end_time = mp_time_sec() + AO_EOF_DELAY + 0.25; // + margin
if (ao->driver->get_delay)
p->expected_end_time += ao->driver->get_delay(ao);
}
return r;
}
static void *playthread(void *arg)
{
struct ao *ao = arg;
struct ao_push_state *p = ao->api_priv;
while (1) {
pthread_mutex_lock(&p->lock);
if (p->terminate) {
pthread_mutex_unlock(&p->lock);
return NULL;
}
double timeout = 2.0;
if (p->playing) {
int r = ao_play_data(ao);
// The device buffers are not necessarily full, but writing to the
// AO buffer will wake up this thread anyway.
bool buffers_full = r <= 0;
// We have to estimate when the AO needs data again.
if (buffers_full && ao->driver->get_delay) {
float buffered_audio = ao->driver->get_delay(ao);
timeout = buffered_audio - 0.050;
// Keep extra safety margin if the buffers are large
if (timeout > 0.100)
timeout = MPMAX(timeout - 0.200, 0.100);
} else {
timeout = 0;
}
// Half of the buffer played -> wakeup playback thread to get more.
double min_wait = ao->device_buffer / (double)ao->samplerate;
if (timeout <= min_wait / 2 + 0.001)
mp_input_wakeup(ao->input_ctx);
// Avoid wasting CPU - this assumes ao_play_data() usually fills the
// audio buffer as far as possible, so even if the device buffer
// is not full, we can only wait for the core.
timeout = MPMAX(timeout, min_wait * 0.75);
}
pthread_mutex_unlock(&p->lock);
pthread_mutex_lock(&p->wakeup_lock);
if (!p->need_wakeup)
mpthread_cond_timedwait(&p->wakeup, &p->wakeup_lock, timeout);
p->need_wakeup = false;
pthread_mutex_unlock(&p->wakeup_lock);
}
return NULL;
}
static void uninit(struct ao *ao)
{
struct ao_push_state *p = ao->api_priv;
pthread_mutex_lock(&p->lock);
p->terminate = true;
wakeup_playthread(ao);
pthread_mutex_unlock(&p->lock);
pthread_join(p->thread, NULL);
ao->driver->uninit(ao);
pthread_cond_destroy(&p->wakeup);
pthread_mutex_destroy(&p->lock);
pthread_mutex_destroy(&p->wakeup_lock);
}
static int init(struct ao *ao)
{
struct ao_push_state *p = ao->api_priv;
pthread_mutex_init(&p->lock, NULL);
pthread_mutex_init(&p->wakeup_lock, NULL);
pthread_cond_init(&p->wakeup, NULL);
p->buffer = mp_audio_buffer_create(ao);
mp_audio_buffer_reinit_fmt(p->buffer, ao->format,
&ao->channels, ao->samplerate);
mp_audio_buffer_preallocate_min(p->buffer, ao->buffer);
if (pthread_create(&p->thread, NULL, playthread, ao)) {
ao->driver->uninit(ao);
return -1;
}
return 0;
}
const struct ao_driver ao_api_push = {
.init = init,
.control = control,
.uninit = uninit,
.reset = reset,
.get_space = get_space,
.play = play,
.get_delay = get_delay,
.pause = pause,
.resume = resume,
.drain = drain,
.priv_size = sizeof(struct ao_push_state),
};
// Must be called locked.
int ao_play_silence(struct ao *ao, int samples)
{
assert(ao->api == &ao_api_push);
if (samples <= 0 || AF_FORMAT_IS_SPECIAL(ao->format) || !ao->driver->play)
return 0;
char *p = talloc_size(NULL, samples * ao->sstride);
af_fill_silence(p, samples * ao->sstride, ao->format);
void *tmp[MP_NUM_CHANNELS];
for (int n = 0; n < MP_NUM_CHANNELS; n++)
tmp[n] = p;
int r = ao->driver->play(ao, tmp, samples, 0);
talloc_free(p);
return r;
}
// Notify the core that new data should be sent to the AO. Normally, the core
// uses a heuristic based on ao_delay() when to refill the buffers, but this
// can be used to reduce wait times. Can be called from any thread.
void ao_need_data(struct ao *ao)
{
assert(ao->api == &ao_api_push);
// wakeup the play thread at least once
wakeup_playthread(ao);
}