mirror of
https://github.com/mpv-player/mpv
synced 2024-12-28 01:52:19 +00:00
040c050f2d
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.
351 lines
10 KiB
C
351 lines
10 KiB
C
/*
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* This file is part of mpv.
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*
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* mpv is free software; you can redistribute it and/or modify
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* it under the terms of the GNU General Public License as published by
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* the Free Software Foundation; either version 2 of the License, or
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* (at your option) any later version.
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*
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* mpv is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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* GNU General Public License for more details.
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*
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* You should have received a copy of the GNU General Public License along
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* with mpv. If not, see <http://www.gnu.org/licenses/>.
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*/
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#include <stddef.h>
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#include <pthread.h>
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#include <inttypes.h>
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#include <limits.h>
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#include <assert.h>
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#include "ao.h"
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#include "internal.h"
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#include "audio/format.h"
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#include "common/msg.h"
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#include "common/common.h"
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#include "input/input.h"
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#include "osdep/threads.h"
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#include "osdep/timer.h"
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#include "compat/atomics.h"
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#include "audio/audio.h"
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#include "audio/audio_buffer.h"
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struct ao_push_state {
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pthread_t thread;
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pthread_mutex_t lock;
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// uses a separate lock to avoid lock order issues with ao_need_data()
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pthread_mutex_t wakeup_lock;
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pthread_cond_t wakeup;
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// --- protected by lock
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struct mp_audio_buffer *buffer;
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bool terminate;
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bool playing;
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// Whether the current buffer contains the complete audio.
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bool final_chunk;
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double expected_end_time;
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// -- protected by wakeup_lock
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bool need_wakeup;
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};
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static void wakeup_playthread(struct ao *ao)
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{
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struct ao_push_state *p = ao->api_priv;
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pthread_mutex_lock(&p->wakeup_lock);
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p->need_wakeup = true;
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pthread_cond_signal(&p->wakeup);
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pthread_mutex_unlock(&p->wakeup_lock);
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}
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static int control(struct ao *ao, enum aocontrol cmd, void *arg)
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{
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int r = CONTROL_UNKNOWN;
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if (ao->driver->control) {
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struct ao_push_state *p = ao->api_priv;
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pthread_mutex_lock(&p->lock);
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r = ao->driver->control(ao, cmd, arg);
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pthread_mutex_unlock(&p->lock);
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}
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return r;
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}
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static float get_delay(struct ao *ao)
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{
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struct ao_push_state *p = ao->api_priv;
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pthread_mutex_lock(&p->lock);
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double driver_delay = 0;
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if (ao->driver->get_delay)
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driver_delay = ao->driver->get_delay(ao);
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double delay = driver_delay + mp_audio_buffer_seconds(p->buffer);
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pthread_mutex_unlock(&p->lock);
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if (delay >= AO_EOF_DELAY && p->expected_end_time) {
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if (mp_time_sec() > p->expected_end_time) {
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MP_ERR(ao, "Audio device EOF reporting is broken!\n");
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MP_ERR(ao, "Please report this problem.\n");
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delay = 0;
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}
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}
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return delay;
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}
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static void reset(struct ao *ao)
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{
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struct ao_push_state *p = ao->api_priv;
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pthread_mutex_lock(&p->lock);
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if (ao->driver->reset)
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ao->driver->reset(ao);
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mp_audio_buffer_clear(p->buffer);
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p->playing = false;
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wakeup_playthread(ao);
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pthread_mutex_unlock(&p->lock);
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}
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static void pause(struct ao *ao)
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{
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struct ao_push_state *p = ao->api_priv;
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pthread_mutex_lock(&p->lock);
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if (ao->driver->pause)
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ao->driver->pause(ao);
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p->playing = false;
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wakeup_playthread(ao);
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pthread_mutex_unlock(&p->lock);
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}
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static void resume(struct ao *ao)
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{
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struct ao_push_state *p = ao->api_priv;
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pthread_mutex_lock(&p->lock);
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if (ao->driver->resume)
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ao->driver->resume(ao);
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p->playing = true; // tentatively
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p->expected_end_time = 0;
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wakeup_playthread(ao);
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pthread_mutex_unlock(&p->lock);
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}
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static void drain(struct ao *ao)
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{
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if (ao->driver->drain) {
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struct ao_push_state *p = ao->api_priv;
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pthread_mutex_lock(&p->lock);
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ao->driver->drain(ao);
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pthread_mutex_unlock(&p->lock);
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} else {
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ao_wait_drain(ao);
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}
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}
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static int get_space(struct ao *ao)
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{
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struct ao_push_state *p = ao->api_priv;
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pthread_mutex_lock(&p->lock);
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int space = mp_audio_buffer_get_write_available(p->buffer);
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if (ao->driver->get_space) {
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// The following code attempts to keep the total buffered audio to
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// MIN_BUFFER in order to improve latency.
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int device_space = ao->driver->get_space(ao);
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int device_buffered = ao->device_buffer - device_space;
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int soft_buffered = mp_audio_buffer_samples(p->buffer);
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int min_buffer = MIN_BUFFER * ao->samplerate;
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int missing = min_buffer - device_buffered - soft_buffered;
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// But always keep the device's buffer filled as much as we can.
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int device_missing = device_space - soft_buffered;
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missing = MPMAX(missing, device_missing);
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space = MPMIN(space, missing);
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space = MPMAX(0, space);
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}
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pthread_mutex_unlock(&p->lock);
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return space;
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}
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static int play(struct ao *ao, void **data, int samples, int flags)
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{
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struct ao_push_state *p = ao->api_priv;
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pthread_mutex_lock(&p->lock);
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int write_samples = mp_audio_buffer_get_write_available(p->buffer);
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write_samples = MPMIN(write_samples, samples);
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struct mp_audio audio;
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mp_audio_buffer_get_format(p->buffer, &audio);
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for (int n = 0; n < ao->num_planes; n++)
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audio.planes[n] = data[n];
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audio.samples = write_samples;
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mp_audio_buffer_append(p->buffer, &audio);
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p->final_chunk = !!(flags & AOPLAY_FINAL_CHUNK);
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p->playing = true;
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p->expected_end_time = 0;
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wakeup_playthread(ao);
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pthread_mutex_unlock(&p->lock);
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return write_samples;
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}
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// called locked
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static int ao_play_data(struct ao *ao)
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{
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struct ao_push_state *p = ao->api_priv;
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struct mp_audio data;
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mp_audio_buffer_peek(p->buffer, &data);
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int max = data.samples;
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int space = ao->driver->get_space ? ao->driver->get_space(ao) : INT_MAX;
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if (data.samples > space)
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data.samples = space;
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if (data.samples <= 0)
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return 0;
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int flags = 0;
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if (p->final_chunk && data.samples == max)
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flags |= AOPLAY_FINAL_CHUNK;
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int r = ao->driver->play(ao, data.planes, data.samples, flags);
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if (r > data.samples) {
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MP_WARN(ao, "Audio device returned non-sense value.");
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r = data.samples;
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}
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if (r > 0)
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mp_audio_buffer_skip(p->buffer, r);
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if (p->final_chunk && mp_audio_buffer_samples(p->buffer) == 0) {
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p->playing = false;
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p->expected_end_time = mp_time_sec() + AO_EOF_DELAY + 0.25; // + margin
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if (ao->driver->get_delay)
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p->expected_end_time += ao->driver->get_delay(ao);
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}
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return r;
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}
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static void *playthread(void *arg)
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{
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struct ao *ao = arg;
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struct ao_push_state *p = ao->api_priv;
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while (1) {
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pthread_mutex_lock(&p->lock);
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if (p->terminate) {
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pthread_mutex_unlock(&p->lock);
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return NULL;
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}
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double timeout = 2.0;
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if (p->playing) {
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int r = ao_play_data(ao);
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// The device buffers are not necessarily full, but writing to the
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// AO buffer will wake up this thread anyway.
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bool buffers_full = r <= 0;
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// We have to estimate when the AO needs data again.
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if (buffers_full && ao->driver->get_delay) {
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float buffered_audio = ao->driver->get_delay(ao);
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timeout = buffered_audio - 0.050;
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// Keep extra safety margin if the buffers are large
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if (timeout > 0.100)
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timeout = MPMAX(timeout - 0.200, 0.100);
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} else {
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timeout = 0;
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}
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// Half of the buffer played -> wakeup playback thread to get more.
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double min_wait = ao->device_buffer / (double)ao->samplerate;
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if (timeout <= min_wait / 2 + 0.001)
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mp_input_wakeup(ao->input_ctx);
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// Avoid wasting CPU - this assumes ao_play_data() usually fills the
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// audio buffer as far as possible, so even if the device buffer
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// is not full, we can only wait for the core.
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timeout = MPMAX(timeout, min_wait * 0.75);
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}
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pthread_mutex_unlock(&p->lock);
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pthread_mutex_lock(&p->wakeup_lock);
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if (!p->need_wakeup)
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mpthread_cond_timedwait(&p->wakeup, &p->wakeup_lock, timeout);
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p->need_wakeup = false;
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pthread_mutex_unlock(&p->wakeup_lock);
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}
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return NULL;
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}
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static void uninit(struct ao *ao)
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{
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struct ao_push_state *p = ao->api_priv;
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pthread_mutex_lock(&p->lock);
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p->terminate = true;
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wakeup_playthread(ao);
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pthread_mutex_unlock(&p->lock);
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pthread_join(p->thread, NULL);
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ao->driver->uninit(ao);
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pthread_cond_destroy(&p->wakeup);
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pthread_mutex_destroy(&p->lock);
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pthread_mutex_destroy(&p->wakeup_lock);
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}
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static int init(struct ao *ao)
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{
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struct ao_push_state *p = ao->api_priv;
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pthread_mutex_init(&p->lock, NULL);
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pthread_mutex_init(&p->wakeup_lock, NULL);
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pthread_cond_init(&p->wakeup, NULL);
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p->buffer = mp_audio_buffer_create(ao);
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mp_audio_buffer_reinit_fmt(p->buffer, ao->format,
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&ao->channels, ao->samplerate);
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mp_audio_buffer_preallocate_min(p->buffer, ao->buffer);
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if (pthread_create(&p->thread, NULL, playthread, ao)) {
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ao->driver->uninit(ao);
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return -1;
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}
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return 0;
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}
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const struct ao_driver ao_api_push = {
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.init = init,
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.control = control,
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.uninit = uninit,
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.reset = reset,
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.get_space = get_space,
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.play = play,
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.get_delay = get_delay,
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.pause = pause,
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.resume = resume,
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.drain = drain,
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.priv_size = sizeof(struct ao_push_state),
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};
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// Must be called locked.
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int ao_play_silence(struct ao *ao, int samples)
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{
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assert(ao->api == &ao_api_push);
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if (samples <= 0 || AF_FORMAT_IS_SPECIAL(ao->format) || !ao->driver->play)
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return 0;
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char *p = talloc_size(NULL, samples * ao->sstride);
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af_fill_silence(p, samples * ao->sstride, ao->format);
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void *tmp[MP_NUM_CHANNELS];
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for (int n = 0; n < MP_NUM_CHANNELS; n++)
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tmp[n] = p;
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int r = ao->driver->play(ao, tmp, samples, 0);
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talloc_free(p);
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return r;
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}
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// Notify the core that new data should be sent to the AO. Normally, the core
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// uses a heuristic based on ao_delay() when to refill the buffers, but this
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// can be used to reduce wait times. Can be called from any thread.
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void ao_need_data(struct ao *ao)
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{
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assert(ao->api == &ao_api_push);
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// wakeup the play thread at least once
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wakeup_playthread(ao);
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}
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