mirror of https://github.com/mpv-player/mpv
638 lines
21 KiB
C
638 lines
21 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 <stdbool.h>
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#include <inttypes.h>
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#include <limits.h>
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#include <math.h>
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#include <assert.h>
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#include "config.h"
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#include "talloc.h"
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#include "common/msg.h"
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#include "common/encode.h"
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#include "options/options.h"
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#include "common/common.h"
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#include "audio/mixer.h"
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#include "audio/audio.h"
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#include "audio/audio_buffer.h"
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#include "audio/decode/dec_audio.h"
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#include "audio/filter/af.h"
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#include "audio/out/ao.h"
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#include "demux/demux.h"
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#include "video/decode/dec_video.h"
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#include "core.h"
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#include "command.h"
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static int update_playback_speed_filters(struct MPContext *mpctx)
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{
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struct MPOpts *opts = mpctx->opts;
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double speed = mpctx->audio_speed;
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struct af_stream *afs = mpctx->d_audio->afilter;
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// Use pitch correction only for speed adjustments by the user, not minor
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// sync correction ones.
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bool use_pitch_correction = opts->pitch_correction &&
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opts->playback_speed != 1.0;
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// Make sure only exactly one filter changes speed; resetting them all
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// and setting 1 filter is the easiest way to achieve this.
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af_control_all(afs, AF_CONTROL_SET_PLAYBACK_SPEED, &(double){1});
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af_control_all(afs, AF_CONTROL_SET_PLAYBACK_SPEED_RESAMPLE, &(double){1});
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if (speed == 1.0)
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return af_remove_by_label(afs, "playback-speed");
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// Compatibility: if the user uses --af=scaletempo, always use this
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// filter to change speed. Don't insert a second filter (any) either.
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if (!af_find_by_label(afs, "playback-speed") &&
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af_control_any_rev(afs, AF_CONTROL_SET_PLAYBACK_SPEED, &speed))
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return 0;
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int method = AF_CONTROL_SET_PLAYBACK_SPEED_RESAMPLE;
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if (use_pitch_correction)
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method = AF_CONTROL_SET_PLAYBACK_SPEED;
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if (!af_control_any_rev(afs, method, &speed)) {
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if (af_remove_by_label(afs, "playback-speed") < 0)
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return -1;
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char *filter = method == AF_CONTROL_SET_PLAYBACK_SPEED
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? "scaletempo" : "lavrresample";
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if (af_add(afs, filter, "playback-speed", NULL) < 0)
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return -1;
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// Try again.
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if (!af_control_any_rev(afs, method, &speed))
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return -1;
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}
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return 0;
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}
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static int recreate_audio_filters(struct MPContext *mpctx)
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{
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assert(mpctx->d_audio);
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if (update_playback_speed_filters(mpctx) < 0) {
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mpctx->opts->playback_speed = 1.0;
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mpctx->speed_factor_a = 1.0;
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mpctx->audio_speed = 1.0;
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mp_notify(mpctx, MP_EVENT_CHANGE_ALL, NULL);
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}
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struct af_stream *afs = mpctx->d_audio->afilter;
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if (afs->initialized < 1 && af_init(afs) < 0) {
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MP_ERR(mpctx, "Couldn't find matching filter/ao format!\n");
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return -1;
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}
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mixer_reinit_audio(mpctx->mixer, mpctx->ao, afs);
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return 0;
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}
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int reinit_audio_filters(struct MPContext *mpctx)
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{
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struct dec_audio *d_audio = mpctx->d_audio;
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if (!d_audio)
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return 0;
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af_uninit(mpctx->d_audio->afilter);
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if (af_init(mpctx->d_audio->afilter) < 0)
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return -1;
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if (recreate_audio_filters(mpctx) < 0)
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return -1;
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return 1;
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}
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// Call this if opts->playback_speed or mpctx->speed_correction changes.
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void update_playback_speed(struct MPContext *mpctx)
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{
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struct MPOpts *opts = mpctx->opts;
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double old_speed_factor_a = mpctx->speed_factor_a;
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double old_audio_speed = mpctx->audio_speed;
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mpctx->audio_speed = opts->playback_speed * mpctx->speed_factor_a;
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mpctx->video_speed = opts->playback_speed * mpctx->speed_factor_v;
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if (mpctx->speed_factor_a == old_speed_factor_a &&
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mpctx->audio_speed == old_audio_speed)
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return;
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if (!mpctx->d_audio || mpctx->d_audio->afilter->initialized < 1)
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return;
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recreate_audio_filters(mpctx);
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}
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void reset_audio_state(struct MPContext *mpctx)
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{
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if (mpctx->d_audio)
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audio_reset_decoding(mpctx->d_audio);
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if (mpctx->ao_buffer)
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mp_audio_buffer_clear(mpctx->ao_buffer);
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mpctx->audio_status = mpctx->d_audio ? STATUS_SYNCING : STATUS_EOF;
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mpctx->delay = 0;
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}
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void uninit_audio_out(struct MPContext *mpctx)
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{
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if (mpctx->ao) {
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// Note: with gapless_audio, stop_play is not correctly set
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if (mpctx->opts->gapless_audio || mpctx->stop_play == AT_END_OF_FILE)
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ao_drain(mpctx->ao);
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mixer_uninit_audio(mpctx->mixer);
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ao_uninit(mpctx->ao);
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mp_notify(mpctx, MPV_EVENT_AUDIO_RECONFIG, NULL);
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}
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mpctx->ao = NULL;
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talloc_free(mpctx->ao_decoder_fmt);
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mpctx->ao_decoder_fmt = NULL;
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}
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void uninit_audio_chain(struct MPContext *mpctx)
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{
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if (mpctx->d_audio) {
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mixer_uninit_audio(mpctx->mixer);
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audio_uninit(mpctx->d_audio);
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mpctx->d_audio = NULL;
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talloc_free(mpctx->ao_buffer);
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mpctx->ao_buffer = NULL;
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mpctx->audio_status = STATUS_EOF;
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reselect_demux_streams(mpctx);
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mp_notify(mpctx, MPV_EVENT_AUDIO_RECONFIG, NULL);
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}
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}
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void reinit_audio_chain(struct MPContext *mpctx)
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{
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struct MPOpts *opts = mpctx->opts;
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struct track *track = mpctx->current_track[0][STREAM_AUDIO];
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struct sh_stream *sh = track ? track->stream : NULL;
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if (!sh) {
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uninit_audio_out(mpctx);
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goto no_audio;
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}
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mp_notify(mpctx, MPV_EVENT_AUDIO_RECONFIG, NULL);
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if (!mpctx->d_audio) {
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mpctx->d_audio = talloc_zero(NULL, struct dec_audio);
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mpctx->d_audio->log = mp_log_new(mpctx->d_audio, mpctx->log, "!ad");
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mpctx->d_audio->global = mpctx->global;
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mpctx->d_audio->opts = opts;
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mpctx->d_audio->header = sh;
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mpctx->d_audio->pool = mp_audio_pool_create(mpctx->d_audio);
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mpctx->d_audio->afilter = af_new(mpctx->global);
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mpctx->d_audio->afilter->replaygain_data = sh->audio->replaygain_data;
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mpctx->d_audio->spdif_passthrough = true;
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mpctx->ao_buffer = mp_audio_buffer_create(NULL);
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if (!audio_init_best_codec(mpctx->d_audio))
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goto init_error;
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reset_audio_state(mpctx);
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if (mpctx->ao) {
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struct mp_audio fmt;
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ao_get_format(mpctx->ao, &fmt);
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mp_audio_buffer_reinit(mpctx->ao_buffer, &fmt);
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}
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}
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assert(mpctx->d_audio);
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struct mp_audio in_format = mpctx->d_audio->decode_format;
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if (!mp_audio_config_valid(&in_format)) {
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// We don't know the audio format yet - so configure it later as we're
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// resyncing. fill_audio_buffers() will call this function again.
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mpctx->sleeptime = 0;
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return;
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}
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// Weak gapless audio: drain AO on decoder format changes
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if (mpctx->ao_decoder_fmt && mpctx->ao && opts->gapless_audio < 0 &&
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!mp_audio_config_equals(mpctx->ao_decoder_fmt, &in_format))
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{
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uninit_audio_out(mpctx);
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}
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struct af_stream *afs = mpctx->d_audio->afilter;
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afs->output = (struct mp_audio){0};
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if (mpctx->ao) {
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ao_get_format(mpctx->ao, &afs->output);
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} else if (af_fmt_is_pcm(in_format.format)) {
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afs->output.rate = opts->force_srate;
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mp_audio_set_format(&afs->output, opts->audio_output_format);
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mp_audio_set_channels(&afs->output, &opts->audio_output_channels);
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}
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// filter input format: same as codec's output format:
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afs->input = in_format;
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// Determine what the filter chain outputs. recreate_audio_filters() also
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// needs this for testing whether playback speed is changed by resampling
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// or using a special filter.
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if (af_init(afs) < 0) {
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MP_ERR(mpctx, "Error at audio filter chain pre-init!\n");
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goto init_error;
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}
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if (!mpctx->ao) {
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mp_chmap_remove_useless_channels(&afs->output.channels,
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&opts->audio_output_channels);
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mp_audio_set_channels(&afs->output, &afs->output.channels);
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mpctx->ao = ao_init_best(mpctx->global, mpctx->input,
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mpctx->encode_lavc_ctx, afs->output.rate,
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afs->output.format, afs->output.channels);
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struct mp_audio fmt = {0};
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if (mpctx->ao)
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ao_get_format(mpctx->ao, &fmt);
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// Verify passthrough format was not changed.
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if (mpctx->ao && af_fmt_is_spdif(afs->output.format)) {
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if (!mp_audio_config_equals(&afs->output, &fmt)) {
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MP_ERR(mpctx, "Passthrough format unsupported.\n");
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ao_uninit(mpctx->ao);
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mpctx->ao = NULL;
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}
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}
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if (!mpctx->ao) {
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// If spdif was used, try to fallback to PCM.
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if (af_fmt_is_spdif(afs->output.format) &&
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mpctx->d_audio->spdif_passthrough)
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{
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mpctx->d_audio->spdif_passthrough = false;
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if (!audio_init_best_codec(mpctx->d_audio))
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goto init_error;
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reset_audio_state(mpctx);
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reinit_audio_chain(mpctx);
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return;
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}
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MP_ERR(mpctx, "Could not open/initialize audio device -> no sound.\n");
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mpctx->error_playing = MPV_ERROR_AO_INIT_FAILED;
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goto init_error;
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}
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mp_audio_buffer_reinit(mpctx->ao_buffer, &fmt);
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afs->output = fmt;
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if (!mp_audio_config_equals(&afs->output, &afs->filter_output))
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afs->initialized = 0;
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mpctx->ao_decoder_fmt = talloc(NULL, struct mp_audio);
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*mpctx->ao_decoder_fmt = in_format;
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MP_INFO(mpctx, "AO: [%s] %s\n", ao_get_name(mpctx->ao),
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mp_audio_config_to_str(&fmt));
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MP_VERBOSE(mpctx, "AO: Description: %s\n", ao_get_description(mpctx->ao));
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update_window_title(mpctx, true);
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}
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if (recreate_audio_filters(mpctx) < 0)
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goto init_error;
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update_playback_speed(mpctx);
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return;
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init_error:
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uninit_audio_chain(mpctx);
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uninit_audio_out(mpctx);
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no_audio:
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if (track)
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error_on_track(mpctx, track);
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}
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// Return pts value corresponding to the end point of audio written to the
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// ao so far.
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double written_audio_pts(struct MPContext *mpctx)
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{
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struct dec_audio *d_audio = mpctx->d_audio;
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if (!d_audio)
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return MP_NOPTS_VALUE;
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struct mp_audio in_format = d_audio->decode_format;
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if (!mp_audio_config_valid(&in_format) || d_audio->afilter->initialized < 1)
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return MP_NOPTS_VALUE;
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// first calculate the end pts of audio that has been output by decoder
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double a_pts = d_audio->pts;
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if (a_pts == MP_NOPTS_VALUE)
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return MP_NOPTS_VALUE;
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// d_audio->pts is the timestamp of the latest input packet with
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// known pts that the decoder has decoded. d_audio->pts_bytes is
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// the amount of bytes the decoder has written after that timestamp.
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a_pts += d_audio->pts_offset / (double)in_format.rate;
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// Now a_pts hopefully holds the pts for end of audio from decoder.
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// Subtract data in buffers between decoder and audio out.
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// Decoded but not filtered
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if (d_audio->waiting)
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a_pts -= d_audio->waiting->samples / (double)in_format.rate;
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// Data buffered in audio filters, measured in seconds of "missing" output
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double buffered_output = af_calc_delay(d_audio->afilter);
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// Data that was ready for ao but was buffered because ao didn't fully
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// accept everything to internal buffers yet
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buffered_output += mp_audio_buffer_seconds(mpctx->ao_buffer);
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// Filters divide audio length by audio_speed, so multiply by it
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// to get the length in original units without speedup or slowdown
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a_pts -= buffered_output * mpctx->audio_speed;
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return a_pts +
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get_track_video_offset(mpctx, mpctx->current_track[0][STREAM_AUDIO]);
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}
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// Return pts value corresponding to currently playing audio.
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double playing_audio_pts(struct MPContext *mpctx)
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{
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double pts = written_audio_pts(mpctx);
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if (pts == MP_NOPTS_VALUE || !mpctx->ao)
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return pts;
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return pts - mpctx->audio_speed * ao_get_delay(mpctx->ao);
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}
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static int write_to_ao(struct MPContext *mpctx, struct mp_audio *data, int flags)
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{
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if (mpctx->paused)
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return 0;
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struct ao *ao = mpctx->ao;
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struct mp_audio out_format;
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ao_get_format(ao, &out_format);
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#if HAVE_ENCODING
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encode_lavc_set_audio_pts(mpctx->encode_lavc_ctx, playing_audio_pts(mpctx));
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#endif
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if (data->samples == 0)
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return 0;
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double real_samplerate = out_format.rate / mpctx->audio_speed;
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int played = ao_play(mpctx->ao, data->planes, data->samples, flags);
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assert(played <= data->samples);
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if (played > 0) {
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mpctx->shown_aframes += played;
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mpctx->delay += played / real_samplerate;
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return played;
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}
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return 0;
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}
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// Return the number of samples that must be skipped or prepended to reach the
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// target audio pts after a seek (for A/V sync or hr-seek).
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// Return value (*skip):
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// >0: skip this many samples
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// =0: don't do anything
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// <0: prepend this many samples of silence
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// Returns false if PTS is not known yet.
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static bool get_sync_samples(struct MPContext *mpctx, int *skip)
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{
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struct MPOpts *opts = mpctx->opts;
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*skip = 0;
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if (mpctx->audio_status != STATUS_SYNCING)
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return true;
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struct mp_audio out_format = {0};
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ao_get_format(mpctx->ao, &out_format);
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double play_samplerate = out_format.rate / mpctx->audio_speed;
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if (!opts->initial_audio_sync) {
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mpctx->audio_status = STATUS_FILLING;
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return true;
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}
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double written_pts = written_audio_pts(mpctx);
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if (written_pts == MP_NOPTS_VALUE && !mp_audio_buffer_samples(mpctx->ao_buffer))
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return false; // no audio read yet
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bool sync_to_video = mpctx->d_video && mpctx->sync_audio_to_video &&
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mpctx->video_status != STATUS_EOF;
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double sync_pts = MP_NOPTS_VALUE;
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if (sync_to_video) {
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if (mpctx->video_status < STATUS_READY)
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return false; // wait until we know a video PTS
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if (mpctx->video_next_pts != MP_NOPTS_VALUE)
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sync_pts = mpctx->video_next_pts - (opts->audio_delay - mpctx->delay);
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} else if (mpctx->hrseek_active) {
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sync_pts = mpctx->hrseek_pts;
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}
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if (sync_pts == MP_NOPTS_VALUE) {
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mpctx->audio_status = STATUS_FILLING;
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return true; // syncing disabled
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}
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double ptsdiff = written_pts - sync_pts;
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// Missing timestamp, or PTS reset, or just broken.
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if (written_pts == MP_NOPTS_VALUE || fabs(ptsdiff) > 3600) {
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MP_WARN(mpctx, "Failed audio resync.\n");
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mpctx->audio_status = STATUS_FILLING;
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return true;
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}
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int align = af_format_sample_alignment(out_format.format);
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*skip = (-ptsdiff * play_samplerate) / align * align;
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return true;
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}
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void fill_audio_out_buffers(struct MPContext *mpctx, double endpts)
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{
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struct MPOpts *opts = mpctx->opts;
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struct dec_audio *d_audio = mpctx->d_audio;
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if (mpctx->ao && ao_query_and_reset_events(mpctx->ao, AO_EVENT_RELOAD)) {
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ao_reset(mpctx->ao);
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uninit_audio_out(mpctx);
|
|
if (d_audio)
|
|
mpctx->audio_status = STATUS_SYNCING;
|
|
}
|
|
|
|
if (!d_audio)
|
|
return;
|
|
|
|
if (d_audio->afilter->initialized < 1 || !mpctx->ao) {
|
|
// Probe the initial audio format. Returns AD_OK (and does nothing) if
|
|
// the format is already known.
|
|
int r = initial_audio_decode(mpctx->d_audio);
|
|
if (r == AD_WAIT)
|
|
return; // continue later when new data is available
|
|
if (r != AD_OK) {
|
|
mpctx->d_audio->init_retries += 1;
|
|
if (mpctx->d_audio->init_retries >= 50) {
|
|
MP_ERR(mpctx, "Error initializing audio.\n");
|
|
error_on_track(mpctx, mpctx->current_track[0][STREAM_AUDIO]);
|
|
return;
|
|
}
|
|
}
|
|
reinit_audio_chain(mpctx);
|
|
mpctx->sleeptime = 0;
|
|
return; // try again next iteration
|
|
}
|
|
|
|
struct mp_audio out_format = {0};
|
|
ao_get_format(mpctx->ao, &out_format);
|
|
double play_samplerate = out_format.rate / mpctx->audio_speed;
|
|
|
|
// If audio is infinitely fast, somehow try keeping approximate A/V sync.
|
|
if (mpctx->audio_status == STATUS_PLAYING && ao_untimed(mpctx->ao) &&
|
|
mpctx->video_status != STATUS_EOF && mpctx->delay > 0)
|
|
return;
|
|
|
|
int playsize = ao_get_space(mpctx->ao);
|
|
|
|
int skip = 0;
|
|
bool sync_known = get_sync_samples(mpctx, &skip);
|
|
if (skip > 0) {
|
|
playsize = MPMIN(skip + 1, MPMAX(playsize, 2500)); // buffer extra data
|
|
} else if (skip < 0) {
|
|
playsize = MPMAX(1, playsize + skip); // silence will be prepended
|
|
}
|
|
|
|
int status = AD_OK;
|
|
bool working = false;
|
|
if (playsize > mp_audio_buffer_samples(mpctx->ao_buffer)) {
|
|
status = audio_decode(d_audio, mpctx->ao_buffer, playsize);
|
|
if (status == AD_WAIT)
|
|
return;
|
|
if (status == AD_NEW_FMT) {
|
|
/* The format change isn't handled too gracefully. A more precise
|
|
* implementation would require draining buffered old-format audio
|
|
* while displaying video, then doing the output format switch.
|
|
*/
|
|
if (mpctx->opts->gapless_audio < 1)
|
|
uninit_audio_out(mpctx);
|
|
reinit_audio_chain(mpctx);
|
|
mpctx->sleeptime = 0;
|
|
return; // retry on next iteration
|
|
}
|
|
if (status == AD_ERR)
|
|
mpctx->sleeptime = 0;
|
|
working = true;
|
|
}
|
|
|
|
// If EOF was reached before, but now something can be decoded, try to
|
|
// restart audio properly. This helps with video files where audio starts
|
|
// later. Retrying is needed to get the correct sync PTS.
|
|
if (mpctx->audio_status >= STATUS_DRAINING && status == AD_OK) {
|
|
mpctx->audio_status = STATUS_SYNCING;
|
|
return; // retry on next iteration
|
|
}
|
|
|
|
bool end_sync = false;
|
|
if (skip >= 0) {
|
|
int max = mp_audio_buffer_samples(mpctx->ao_buffer);
|
|
mp_audio_buffer_skip(mpctx->ao_buffer, MPMIN(skip, max));
|
|
// If something is left, we definitely reached the target time.
|
|
end_sync |= sync_known && skip < max;
|
|
} else if (skip < 0) {
|
|
if (-skip > playsize) { // heuristic against making the buffer too large
|
|
ao_reset(mpctx->ao); // some AOs repeat data on underflow
|
|
mpctx->audio_status = STATUS_DRAINING;
|
|
mpctx->delay = 0;
|
|
return;
|
|
}
|
|
mp_audio_buffer_prepend_silence(mpctx->ao_buffer, -skip);
|
|
end_sync = true;
|
|
}
|
|
|
|
if (mpctx->audio_status == STATUS_SYNCING) {
|
|
if (end_sync)
|
|
mpctx->audio_status = STATUS_FILLING;
|
|
if (status != AD_OK && !mp_audio_buffer_samples(mpctx->ao_buffer))
|
|
mpctx->audio_status = STATUS_EOF;
|
|
if (working)
|
|
mpctx->sleeptime = 0;
|
|
return; // continue on next iteration
|
|
}
|
|
|
|
assert(mpctx->audio_status >= STATUS_FILLING);
|
|
|
|
// Even if we're done decoding and syncing, let video start first - this is
|
|
// required, because sending audio to the AO already starts playback.
|
|
if (mpctx->audio_status == STATUS_FILLING && mpctx->sync_audio_to_video &&
|
|
mpctx->video_status <= STATUS_READY)
|
|
{
|
|
mpctx->audio_status = STATUS_READY;
|
|
return;
|
|
}
|
|
|
|
bool audio_eof = status == AD_EOF;
|
|
bool partial_fill = false;
|
|
int playflags = 0;
|
|
|
|
if (endpts != MP_NOPTS_VALUE) {
|
|
double samples = (endpts - written_audio_pts(mpctx) - opts->audio_delay)
|
|
* play_samplerate;
|
|
if (playsize > samples) {
|
|
playsize = MPMAX(samples, 0);
|
|
audio_eof = true;
|
|
partial_fill = true;
|
|
}
|
|
}
|
|
|
|
if (playsize > mp_audio_buffer_samples(mpctx->ao_buffer)) {
|
|
playsize = mp_audio_buffer_samples(mpctx->ao_buffer);
|
|
partial_fill = true;
|
|
}
|
|
|
|
audio_eof &= partial_fill;
|
|
|
|
// With gapless audio, delay this to ao_uninit. There must be only
|
|
// 1 final chunk, and that is handled when calling ao_uninit().
|
|
if (audio_eof && !opts->gapless_audio)
|
|
playflags |= AOPLAY_FINAL_CHUNK;
|
|
|
|
struct mp_audio data;
|
|
mp_audio_buffer_peek(mpctx->ao_buffer, &data);
|
|
data.samples = MPMIN(data.samples, mpctx->paused ? 0 : playsize);
|
|
int played = write_to_ao(mpctx, &data, playflags);
|
|
assert(played >= 0 && played <= data.samples);
|
|
mp_audio_buffer_skip(mpctx->ao_buffer, played);
|
|
|
|
mpctx->audio_status = STATUS_PLAYING;
|
|
if (audio_eof && !playsize) {
|
|
mpctx->audio_status = STATUS_DRAINING;
|
|
// Wait until the AO has played all queued data. In the gapless case,
|
|
// we trigger EOF immediately, and let it play asynchronously.
|
|
if (ao_eof_reached(mpctx->ao) || opts->gapless_audio)
|
|
mpctx->audio_status = STATUS_EOF;
|
|
}
|
|
}
|
|
|
|
// Drop data queued for output, or which the AO is currently outputting.
|
|
void clear_audio_output_buffers(struct MPContext *mpctx)
|
|
{
|
|
if (mpctx->ao)
|
|
ao_reset(mpctx->ao);
|
|
if (mpctx->ao_buffer)
|
|
mp_audio_buffer_clear(mpctx->ao_buffer);
|
|
}
|