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3694af6076
mpv/player/audio.c
Dudemanguy fe6d9b6962 player: rearrange video sync opts/enums/defines 
The video sync logic for mpv lies completely within its core at
essentially the highest layer of abstraction. The problem with this is
that it is impossible for VOs to know what video sync mode mpv is
currently using since it has no access to the opts. Because different
video sync modes completely changes how mpv's render loop operates, it's
reasonable that a VO may want to change how it renders based on the
current mode (see the next commit for an example).

Let's just move the video sync option to mp_vo_opts. MPContext, of
course, can still access the value of the option so it only requires
minor changes in player/video.c. Additionally, move the VS_IS_DISP
define from to player/core.h to common/common.h. All VOs already have
access to common/common.h, and there's no need for them to gain access
to everything that's in player/core.h.
2022-04-11 18:14:22 +00:00

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/*
* 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/>.
*/
#include <stddef.h>
#include <stdbool.h>
#include <inttypes.h>
#include <limits.h>
#include <math.h>
#include <assert.h>
#include "config.h"
#include "mpv_talloc.h"
#include "common/msg.h"
#include "common/encode.h"
#include "options/options.h"
#include "common/common.h"
#include "osdep/timer.h"
#include "audio/format.h"
#include "audio/out/ao.h"
#include "demux/demux.h"
#include "filters/f_async_queue.h"
#include "filters/f_decoder_wrapper.h"
#include "filters/filter_internal.h"
#include "core.h"
#include "command.h"
enum {
AD_OK = 0,
AD_EOF = -2,
AD_WAIT = -4,
};
static void ao_process(struct mp_filter *f);
static void update_speed_filters(struct MPContext *mpctx)
{
struct ao_chain *ao_c = mpctx->ao_chain;
if (!ao_c)
return;
double speed = mpctx->opts->playback_speed;
double resample = mpctx->speed_factor_a;
double drop = 1.0;
if (!mpctx->opts->pitch_correction) {
resample *= speed;
speed = 1.0;
}
if (mpctx->display_sync_active && mpctx->video_out->opts->video_sync == VS_DISP_ADROP) {
drop *= speed * resample;
resample = speed = 1.0;
}
mp_output_chain_set_audio_speed(ao_c->filter, speed, resample, drop);
}
static int recreate_audio_filters(struct MPContext *mpctx)
{
struct ao_chain *ao_c = mpctx->ao_chain;
assert(ao_c);
if (!mp_output_chain_update_filters(ao_c->filter, mpctx->opts->af_settings))
goto fail;
update_speed_filters(mpctx);
mp_notify(mpctx, MPV_EVENT_AUDIO_RECONFIG, NULL);
return 0;
fail:
MP_ERR(mpctx, "Audio filter initialized failed!\n");
return -1;
}
int reinit_audio_filters(struct MPContext *mpctx)
{
struct ao_chain *ao_c = mpctx->ao_chain;
if (!ao_c)
return 0;
double delay = mp_output_get_measured_total_delay(ao_c->filter);
if (recreate_audio_filters(mpctx) < 0)
return -1;
double ndelay = mp_output_get_measured_total_delay(ao_c->filter);
// Only force refresh if the amount of dropped buffered data is going to
// cause "issues" for the A/V sync logic.
if (mpctx->audio_status == STATUS_PLAYING && delay - ndelay >= 0.2)
issue_refresh_seek(mpctx, MPSEEK_EXACT);
return 1;
}
static double db_gain(double db)
{
return pow(10.0, db/20.0);
}
static float compute_replaygain(struct MPContext *mpctx)
{
struct MPOpts *opts = mpctx->opts;
float rgain = 1.0;
struct replaygain_data *rg = NULL;
struct track *track = mpctx->current_track[0][STREAM_AUDIO];
if (track)
rg = track->stream->codec->replaygain_data;
if (opts->rgain_mode && rg) {
MP_VERBOSE(mpctx, "Replaygain: Track=%f/%f Album=%f/%f\n",
rg->track_gain, rg->track_peak,
rg->album_gain, rg->album_peak);
float gain, peak;
if (opts->rgain_mode == 1) {
gain = rg->track_gain;
peak = rg->track_peak;
} else {
gain = rg->album_gain;
peak = rg->album_peak;
}
gain += opts->rgain_preamp;
rgain = db_gain(gain);
MP_VERBOSE(mpctx, "Applying replay-gain: %f\n", rgain);
if (!opts->rgain_clip) { // clipping prevention
rgain = MPMIN(rgain, 1.0 / peak);
MP_VERBOSE(mpctx, "...with clipping prevention: %f\n", rgain);
}
} else if (opts->rgain_fallback) {
rgain = db_gain(opts->rgain_fallback);
MP_VERBOSE(mpctx, "Applying fallback gain: %f\n", rgain);
}
return rgain;
}
// Called when opts->softvol_volume or opts->softvol_mute were changed.
void audio_update_volume(struct MPContext *mpctx)
{
struct MPOpts *opts = mpctx->opts;
struct ao_chain *ao_c = mpctx->ao_chain;
if (!ao_c || !ao_c->ao)
return;
float gain = MPMAX(opts->softvol_volume / 100.0, 0);
gain = pow(gain, 3);
gain *= compute_replaygain(mpctx);
if (opts->softvol_mute == 1)
gain = 0.0;
ao_set_gain(ao_c->ao, gain);
}
// Call this if opts->playback_speed or mpctx->speed_factor_* change.
void update_playback_speed(struct MPContext *mpctx)
{
mpctx->audio_speed = mpctx->opts->playback_speed * mpctx->speed_factor_a;
mpctx->video_speed = mpctx->opts->playback_speed * mpctx->speed_factor_v;
update_speed_filters(mpctx);
}
static void ao_chain_reset_state(struct ao_chain *ao_c)
{
ao_c->last_out_pts = MP_NOPTS_VALUE;
ao_c->out_eof = false;
ao_c->underrun = false;
ao_c->start_pts_known = false;
ao_c->start_pts = MP_NOPTS_VALUE;
ao_c->untimed_throttle = false;
ao_c->underrun = false;
}
void reset_audio_state(struct MPContext *mpctx)
{
if (mpctx->ao_chain) {
ao_chain_reset_state(mpctx->ao_chain);
struct track *t = mpctx->ao_chain->track;
if (t && t->dec)
mp_decoder_wrapper_set_play_dir(t->dec, mpctx->play_dir);
}
mpctx->audio_status = mpctx->ao_chain ? STATUS_SYNCING : STATUS_EOF;
mpctx->delay = 0;
mpctx->logged_async_diff = -1;
}
void uninit_audio_out(struct MPContext *mpctx)
{
struct ao_chain *ao_c = mpctx->ao_chain;
if (ao_c) {
ao_c->ao_queue = NULL;
TA_FREEP(&ao_c->queue_filter);
ao_c->ao = NULL;
}
if (mpctx->ao) {
// Note: with gapless_audio, stop_play is not correctly set
if ((mpctx->opts->gapless_audio || mpctx->stop_play == AT_END_OF_FILE) &&
ao_is_playing(mpctx->ao) && !get_internal_paused(mpctx))
{
MP_VERBOSE(mpctx, "draining left over audio\n");
ao_drain(mpctx->ao);
}
ao_uninit(mpctx->ao);
mp_notify(mpctx, MPV_EVENT_AUDIO_RECONFIG, NULL);
}
mpctx->ao = NULL;
TA_FREEP(&mpctx->ao_filter_fmt);
}
static void ao_chain_uninit(struct ao_chain *ao_c)
{
struct track *track = ao_c->track;
if (track) {
assert(track->ao_c == ao_c);
track->ao_c = NULL;
if (ao_c->dec_src)
assert(track->dec->f->pins[0] == ao_c->dec_src);
talloc_free(track->dec->f);
track->dec = NULL;
}
if (ao_c->filter_src)
mp_pin_disconnect(ao_c->filter_src);
talloc_free(ao_c->filter->f);
talloc_free(ao_c->ao_filter);
talloc_free(ao_c);
}
void uninit_audio_chain(struct MPContext *mpctx)
{
if (mpctx->ao_chain) {
ao_chain_uninit(mpctx->ao_chain);
mpctx->ao_chain = NULL;
mpctx->audio_status = STATUS_EOF;
mp_notify(mpctx, MPV_EVENT_AUDIO_RECONFIG, NULL);
}
}
static char *audio_config_to_str_buf(char *buf, size_t buf_sz, int rate,
int format, struct mp_chmap channels)
{
char ch[128];
mp_chmap_to_str_buf(ch, sizeof(ch), &channels);
char *hr_ch = mp_chmap_to_str_hr(&channels);
if (strcmp(hr_ch, ch) != 0)
mp_snprintf_cat(ch, sizeof(ch), " (%s)", hr_ch);
snprintf(buf, buf_sz, "%dHz %s %dch %s", rate,
ch, channels.num, af_fmt_to_str(format));
return buf;
}
// Decide whether on a format change, we should reinit the AO.
static bool keep_weak_gapless_format(struct mp_aframe *old, struct mp_aframe* new)
{
bool res = false;
struct mp_aframe *new_mod = mp_aframe_new_ref(new);
if (!new_mod)
abort();
// If the sample formats are compatible (== libswresample generally can
// convert them), keep the AO. On other changes, recreate it.
int old_fmt = mp_aframe_get_format(old);
int new_fmt = mp_aframe_get_format(new);
if (af_format_conversion_score(old_fmt, new_fmt) == INT_MIN)
goto done; // completely incompatible formats
if (!mp_aframe_set_format(new_mod, old_fmt))
goto done;
res = mp_aframe_config_equals(old, new_mod);
done:
talloc_free(new_mod);
return res;
}
static void ao_chain_set_ao(struct ao_chain *ao_c, struct ao *ao)
{
if (ao_c->ao != ao) {
assert(!ao_c->ao);
ao_c->ao = ao;
ao_c->ao_queue = ao_get_queue(ao_c->ao);
ao_c->queue_filter = mp_async_queue_create_filter(ao_c->ao_filter,
MP_PIN_IN, ao_c->ao_queue);
mp_async_queue_set_notifier(ao_c->queue_filter, ao_c->ao_filter);
// Make sure filtering never stops with frames stuck in access filter.
mp_filter_set_high_priority(ao_c->queue_filter, true);
audio_update_volume(ao_c->mpctx);
}
if (ao_c->filter->ao_needs_update)
mp_output_chain_set_ao(ao_c->filter, ao_c->ao);
mp_filter_wakeup(ao_c->ao_filter);
}
static int reinit_audio_filters_and_output(struct MPContext *mpctx)
{
struct MPOpts *opts = mpctx->opts;
struct ao_chain *ao_c = mpctx->ao_chain;
assert(ao_c);
struct track *track = ao_c->track;
assert(ao_c->filter->ao_needs_update);
// The "ideal" filter output format
struct mp_aframe *out_fmt = mp_aframe_new_ref(ao_c->filter->output_aformat);
if (!out_fmt)
abort();
if (!mp_aframe_config_is_valid(out_fmt)) {
talloc_free(out_fmt);
goto init_error;
}
if (af_fmt_is_pcm(mp_aframe_get_format(out_fmt))) {
if (opts->force_srate)
mp_aframe_set_rate(out_fmt, opts->force_srate);
if (opts->audio_output_format)
mp_aframe_set_format(out_fmt, opts->audio_output_format);
if (opts->audio_output_channels.num_chmaps == 1)
mp_aframe_set_chmap(out_fmt, &opts->audio_output_channels.chmaps[0]);
}
// Weak gapless audio: if the filter output format is the same as the
// previous one, keep the AO and don't reinit anything.
// Strong gapless: always keep the AO
if ((mpctx->ao_filter_fmt && mpctx->ao && opts->gapless_audio < 0 &&
keep_weak_gapless_format(mpctx->ao_filter_fmt, out_fmt)) ||
(mpctx->ao && opts->gapless_audio > 0))
{
ao_chain_set_ao(ao_c, mpctx->ao);
talloc_free(out_fmt);
return 0;
}
// Wait until all played.
if (mpctx->ao && ao_is_playing(mpctx->ao)) {
talloc_free(out_fmt);
return 0;
}
// Format change during syncing. Force playback start early, then wait.
if (ao_c->ao_queue && mp_async_queue_get_frames(ao_c->ao_queue) &&
mpctx->audio_status == STATUS_SYNCING)
{
mpctx->audio_status = STATUS_READY;
mp_wakeup_core(mpctx);
talloc_free(out_fmt);
return 0;
}
if (mpctx->audio_status == STATUS_READY) {
talloc_free(out_fmt);
return 0;
}
uninit_audio_out(mpctx);
int out_rate = mp_aframe_get_rate(out_fmt);
int out_format = mp_aframe_get_format(out_fmt);
struct mp_chmap out_channels = {0};
mp_aframe_get_chmap(out_fmt, &out_channels);
int ao_flags = 0;
bool spdif_fallback = af_fmt_is_spdif(out_format) &&
ao_c->spdif_passthrough;
if (opts->ao_null_fallback && !spdif_fallback)
ao_flags |= AO_INIT_NULL_FALLBACK;
if (opts->audio_stream_silence)
ao_flags |= AO_INIT_STREAM_SILENCE;
if (opts->audio_exclusive)
ao_flags |= AO_INIT_EXCLUSIVE;
if (af_fmt_is_pcm(out_format)) {
if (!opts->audio_output_channels.set ||
opts->audio_output_channels.auto_safe)
ao_flags |= AO_INIT_SAFE_MULTICHANNEL_ONLY;
mp_chmap_sel_list(&out_channels,
opts->audio_output_channels.chmaps,
opts->audio_output_channels.num_chmaps);
}
mpctx->ao_filter_fmt = out_fmt;
mpctx->ao = ao_init_best(mpctx->global, ao_flags, mp_wakeup_core_cb,
mpctx, mpctx->encode_lavc_ctx, out_rate,
out_format, out_channels);
int ao_rate = 0;
int ao_format = 0;
struct mp_chmap ao_channels = {0};
if (mpctx->ao)
ao_get_format(mpctx->ao, &ao_rate, &ao_format, &ao_channels);
// Verify passthrough format was not changed.
if (mpctx->ao && af_fmt_is_spdif(out_format)) {
if (out_rate != ao_rate || out_format != ao_format ||
!mp_chmap_equals(&out_channels, &ao_channels))
{
MP_ERR(mpctx, "Passthrough format unsupported.\n");
ao_uninit(mpctx->ao);
mpctx->ao = NULL;
}
}
if (!mpctx->ao) {
// If spdif was used, try to fallback to PCM.
if (spdif_fallback && ao_c->track && ao_c->track->dec) {
MP_VERBOSE(mpctx, "Falling back to PCM output.\n");
ao_c->spdif_passthrough = false;
ao_c->spdif_failed = true;
mp_decoder_wrapper_set_spdif_flag(ao_c->track->dec, false);
if (!mp_decoder_wrapper_reinit(ao_c->track->dec))
goto init_error;
reset_audio_state(mpctx);
mp_output_chain_reset_harder(ao_c->filter);
mp_wakeup_core(mpctx); // reinit with new format next time
return 0;
}
MP_ERR(mpctx, "Could not open/initialize audio device -> no sound.\n");
mpctx->error_playing = MPV_ERROR_AO_INIT_FAILED;
goto init_error;
}
char tmp[192];
MP_INFO(mpctx, "AO: [%s] %s\n", ao_get_name(mpctx->ao),
audio_config_to_str_buf(tmp, sizeof(tmp), ao_rate, ao_format,
ao_channels));
MP_VERBOSE(mpctx, "AO: Description: %s\n", ao_get_description(mpctx->ao));
update_window_title(mpctx, true);
ao_c->ao_resume_time =
opts->audio_wait_open > 0 ? mp_time_sec() + opts->audio_wait_open : 0;
ao_set_paused(mpctx->ao, get_internal_paused(mpctx));
ao_chain_set_ao(ao_c, mpctx->ao);
audio_update_volume(mpctx);
// Almost nonsensical hack to deal with certain format change scenarios.
if (mpctx->audio_status == STATUS_PLAYING)
ao_start(mpctx->ao);
mp_wakeup_core(mpctx);
mp_notify(mpctx, MPV_EVENT_AUDIO_RECONFIG, NULL);
return 0;
init_error:
uninit_audio_chain(mpctx);
uninit_audio_out(mpctx);
error_on_track(mpctx, track);
return -1;
}
int init_audio_decoder(struct MPContext *mpctx, struct track *track)
{
assert(!track->dec);
if (!track->stream)
goto init_error;
track->dec = mp_decoder_wrapper_create(mpctx->filter_root, track->stream);
if (!track->dec)
goto init_error;
if (track->ao_c)
mp_decoder_wrapper_set_spdif_flag(track->dec, true);
if (!mp_decoder_wrapper_reinit(track->dec))
goto init_error;
return 1;
init_error:
if (track->sink)
mp_pin_disconnect(track->sink);
track->sink = NULL;
error_on_track(mpctx, track);
return 0;
}
void reinit_audio_chain(struct MPContext *mpctx)
{
struct track *track = NULL;
track = mpctx->current_track[0][STREAM_AUDIO];
if (!track || !track->stream) {
if (!mpctx->encode_lavc_ctx)
uninit_audio_out(mpctx);
error_on_track(mpctx, track);
return;
}
reinit_audio_chain_src(mpctx, track);
}
static const struct mp_filter_info ao_filter = {
.name = "ao",
.process = ao_process,
};
// (track=NULL creates a blank chain, used for lavfi-complex)
void reinit_audio_chain_src(struct MPContext *mpctx, struct track *track)
{
assert(!mpctx->ao_chain);
mp_notify(mpctx, MPV_EVENT_AUDIO_RECONFIG, NULL);
struct ao_chain *ao_c = talloc_zero(NULL, struct ao_chain);
mpctx->ao_chain = ao_c;
ao_c->mpctx = mpctx;
ao_c->log = mpctx->log;
ao_c->filter =
mp_output_chain_create(mpctx->filter_root, MP_OUTPUT_CHAIN_AUDIO);
ao_c->spdif_passthrough = true;
ao_c->last_out_pts = MP_NOPTS_VALUE;
ao_c->delay = mpctx->opts->audio_delay;
ao_c->ao_filter = mp_filter_create(mpctx->filter_root, &ao_filter);
if (!ao_c->filter || !ao_c->ao_filter)
goto init_error;
ao_c->ao_filter->priv = ao_c;
mp_filter_add_pin(ao_c->ao_filter, MP_PIN_IN, "in");
mp_pin_connect(ao_c->ao_filter->pins[0], ao_c->filter->f->pins[1]);
if (track) {
ao_c->track = track;
track->ao_c = ao_c;
if (!init_audio_decoder(mpctx, track))
goto init_error;
ao_c->dec_src = track->dec->f->pins[0];
mp_pin_connect(ao_c->filter->f->pins[0], ao_c->dec_src);
}
reset_audio_state(mpctx);
if (recreate_audio_filters(mpctx) < 0)
goto init_error;
if (mpctx->ao)
audio_update_volume(mpctx);
mp_wakeup_core(mpctx);
return;
init_error:
uninit_audio_chain(mpctx);
uninit_audio_out(mpctx);
error_on_track(mpctx, track);
}
// Return pts value corresponding to the start point of audio written to the
// ao queue so far.
double written_audio_pts(struct MPContext *mpctx)
{
return mpctx->ao_chain ? mpctx->ao_chain->last_out_pts : MP_NOPTS_VALUE;
}
// Return pts value corresponding to currently playing audio.
double playing_audio_pts(struct MPContext *mpctx)
{
double pts = written_audio_pts(mpctx);
if (pts == MP_NOPTS_VALUE || !mpctx->ao)
return pts;
return pts - mpctx->audio_speed * ao_get_delay(mpctx->ao);
}
// This garbage is needed for untimed AOs. These consume audio infinitely fast,
// so try keeping approximate A/V sync by blocking audio transfer as needed.
static void update_throttle(struct MPContext *mpctx)
{
struct ao_chain *ao_c = mpctx->ao_chain;
bool new_throttle = mpctx->audio_status == STATUS_PLAYING &&
mpctx->delay > 0 && ao_c && ao_c->ao &&
ao_untimed(ao_c->ao) &&
mpctx->video_status != STATUS_EOF;
if (ao_c && new_throttle != ao_c->untimed_throttle) {
ao_c->untimed_throttle = new_throttle;
mp_wakeup_core(mpctx);
mp_filter_wakeup(ao_c->ao_filter);
}
}
static void ao_process(struct mp_filter *f)
{
struct ao_chain *ao_c = f->priv;
struct MPContext *mpctx = ao_c->mpctx;
if (!ao_c->queue_filter) {
// This will eventually lead to the creation of the AO + queue, due
// to how f_output_chain and AO management works.
mp_pin_out_request_data(f->ppins[0]);
// Check for EOF with no data case, which is a mess because everything
// hates us.
struct mp_frame frame = mp_pin_out_read(f->ppins[0]);
if (frame.type == MP_FRAME_EOF) {
MP_VERBOSE(mpctx, "got EOF with no data before it\n");
ao_c->out_eof = true;
mpctx->audio_status = STATUS_DRAINING;
mp_wakeup_core(mpctx);
} else if (frame.type) {
mp_pin_out_unread(f->ppins[0], frame);
}
return;
}
// Due to mp_async_queue_set_notifier() this function is called when the
// queue becomes full. This affects state changes in the normal playloop,
// so wake it up. But avoid redundant wakeups during normal playback.
if (mpctx->audio_status != STATUS_PLAYING &&
mp_async_queue_is_full(ao_c->ao_queue))
mp_wakeup_core(mpctx);
if (mpctx->audio_status == STATUS_SYNCING && !ao_c->start_pts_known)
return;
if (ao_c->untimed_throttle)
return;
if (!mp_pin_can_transfer_data(ao_c->queue_filter->pins[0], f->ppins[0]))
return;
struct mp_frame frame = mp_pin_out_read(f->ppins[0]);
if (frame.type == MP_FRAME_AUDIO) {
struct mp_aframe *af = frame.data;
double endpts = get_play_end_pts(mpctx);
if (endpts != MP_NOPTS_VALUE) {
endpts *= mpctx->play_dir;
// Avoid decoding and discarding the entire rest of the file.
if (mp_aframe_get_pts(af) >= endpts) {
mp_pin_out_unread(f->ppins[0], frame);
if (!ao_c->out_eof) {
ao_c->out_eof = true;
mp_pin_in_write(ao_c->queue_filter->pins[0], MP_EOF_FRAME);
}
return;
}
}
double startpts = mpctx->audio_status == STATUS_SYNCING ?
ao_c->start_pts : MP_NOPTS_VALUE;
mp_aframe_clip_timestamps(af, startpts, endpts);
int samples = mp_aframe_get_size(af);
if (!samples) {
mp_filter_internal_mark_progress(f);
mp_frame_unref(&frame);
return;
}
ao_c->out_eof = false;
if (mpctx->audio_status == STATUS_DRAINING ||
mpctx->audio_status == STATUS_EOF)
{
// If a new frame comes decoder/filter EOF, we should preferably
// call get_sync_pts() again, which (at least in obscure situations)
// may require us to wait a while until the sync PTS is known. Our
// code sucks and can't deal with that, so jump through a hoop to
// get things done in the correct order.
mp_pin_out_unread(f->ppins[0], frame);
ao_c->start_pts_known = false;
mpctx->audio_status = STATUS_SYNCING;
mp_wakeup_core(mpctx);
MP_VERBOSE(mpctx, "new audio frame after EOF\n");
return;
}
mpctx->shown_aframes += samples;
double real_samplerate = mp_aframe_get_rate(af) / mpctx->audio_speed;
mpctx->delay += samples / real_samplerate;
ao_c->last_out_pts = mp_aframe_end_pts(af);
update_throttle(mpctx);
// Gapless case: the AO is still playing from previous file. It makes
// no sense to wait, and in fact the "full queue" event we're waiting
// for may never happen, so start immediately.
// If the new audio starts "later" (big video sync offset), transfer
// of data is stopped somewhere else.
if (mpctx->audio_status == STATUS_SYNCING && ao_is_playing(ao_c->ao)) {
mpctx->audio_status = STATUS_READY;
mp_wakeup_core(mpctx);
MP_VERBOSE(mpctx, "previous audio still playing; continuing\n");
}
mp_pin_in_write(ao_c->queue_filter->pins[0], frame);
} else if (frame.type == MP_FRAME_EOF) {
MP_VERBOSE(mpctx, "audio filter EOF\n");
ao_c->out_eof = true;
mp_wakeup_core(mpctx);
mp_pin_in_write(ao_c->queue_filter->pins[0], frame);
mp_filter_internal_mark_progress(f);
} else {
mp_frame_unref(&frame);
}
}
void reload_audio_output(struct MPContext *mpctx)
{
if (!mpctx->ao)
return;
ao_reset(mpctx->ao);
uninit_audio_out(mpctx);
reinit_audio_filters(mpctx); // mostly to issue refresh seek
struct ao_chain *ao_c = mpctx->ao_chain;
if (ao_c) {
reset_audio_state(mpctx);
mp_output_chain_reset_harder(ao_c->filter);
}
// Whether we can use spdif might have changed. If we failed to use spdif
// in the previous initialization, try it with spdif again (we'll fallback
// to PCM again if necessary).
if (ao_c && ao_c->track) {
struct mp_decoder_wrapper *dec = ao_c->track->dec;
if (dec && ao_c->spdif_failed) {
ao_c->spdif_passthrough = true;
ao_c->spdif_failed = false;
mp_decoder_wrapper_set_spdif_flag(ao_c->track->dec, true);
if (!mp_decoder_wrapper_reinit(dec)) {
MP_ERR(mpctx, "Error reinitializing audio.\n");
error_on_track(mpctx, ao_c->track);
}
}
}
mp_wakeup_core(mpctx);
}
// Returns audio start pts for seeking or video sync.
// Returns false if PTS is not known yet.
static bool get_sync_pts(struct MPContext *mpctx, double *pts)
{
struct MPOpts *opts = mpctx->opts;
*pts = MP_NOPTS_VALUE;
if (!opts->initial_audio_sync)
return true;
bool sync_to_video = mpctx->vo_chain && mpctx->video_status != STATUS_EOF &&
!mpctx->vo_chain->is_sparse;
if (sync_to_video) {
if (mpctx->video_status < STATUS_READY)
return false; // wait until we know a video PTS
if (mpctx->video_pts != MP_NOPTS_VALUE)
*pts = mpctx->video_pts - opts->audio_delay;
} else if (mpctx->hrseek_active) {
*pts = mpctx->hrseek_pts;
} else {
// If audio-only is enabled mid-stream during playback, sync accordingly.
*pts = mpctx->playback_pts;
}
return true;
}
// Look whether audio can be started yet - if audio has to start some time
// after video.
// Caller needs to ensure mpctx->restart_complete is OK
void audio_start_ao(struct MPContext *mpctx)
{
struct ao_chain *ao_c = mpctx->ao_chain;
if (!ao_c || !ao_c->ao || mpctx->audio_status != STATUS_READY)
return;
double pts = MP_NOPTS_VALUE;
if (!get_sync_pts(mpctx, &pts))
return;
double apts = playing_audio_pts(mpctx); // (basically including mpctx->delay)
if (pts != MP_NOPTS_VALUE && apts != MP_NOPTS_VALUE && pts < apts &&
mpctx->video_status != STATUS_EOF)
{
double diff = (apts - pts) / mpctx->opts->playback_speed;
if (!get_internal_paused(mpctx))
mp_set_timeout(mpctx, diff);
if (mpctx->logged_async_diff != diff) {
MP_VERBOSE(mpctx, "delaying audio start %f vs. %f, diff=%f\n",
apts, pts, diff);
mpctx->logged_async_diff = diff;
}
return;
}
MP_VERBOSE(mpctx, "starting audio playback\n");
ao_start(ao_c->ao);
mpctx->audio_status = STATUS_PLAYING;
if (ao_c->out_eof) {
mpctx->audio_status = STATUS_DRAINING;
MP_VERBOSE(mpctx, "audio draining\n");
}
ao_c->underrun = false;
mpctx->logged_async_diff = -1;
mp_wakeup_core(mpctx);
}
void fill_audio_out_buffers(struct MPContext *mpctx)
{
struct MPOpts *opts = mpctx->opts;
if (mpctx->ao && ao_query_and_reset_events(mpctx->ao, AO_EVENT_RELOAD))
reload_audio_output(mpctx);
if (mpctx->ao && ao_query_and_reset_events(mpctx->ao,
AO_EVENT_INITIAL_UNBLOCK))
ao_unblock(mpctx->ao);
update_throttle(mpctx);
struct ao_chain *ao_c = mpctx->ao_chain;
if (!ao_c)
return;
if (ao_c->filter->failed_output_conversion) {
error_on_track(mpctx, ao_c->track);
return;
}
if (ao_c->filter->ao_needs_update) {
if (reinit_audio_filters_and_output(mpctx) < 0)
return;
}
if (mpctx->vo_chain && ao_c->track && ao_c->track->dec &&
mp_decoder_wrapper_get_pts_reset(ao_c->track->dec))
{
MP_WARN(mpctx, "Reset playback due to audio timestamp reset.\n");
reset_playback_state(mpctx);
mp_wakeup_core(mpctx);
}
if (mpctx->audio_status == STATUS_SYNCING) {
double pts;
bool ok = get_sync_pts(mpctx, &pts);
// If the AO is still playing from the previous file (due to gapless),
// but if video is active, this may not work if audio starts later than
// video, and gapless has no advantages anyway. So block doing anything
// until the old audio is fully played.
// (Buggy if AO underruns.)
if (mpctx->ao && ao_is_playing(mpctx->ao) &&
mpctx->video_status != STATUS_EOF) {
MP_VERBOSE(mpctx, "blocked, waiting for old audio to play\n");
ok = false;
}
if (ao_c->start_pts_known != ok || ao_c->start_pts != pts) {
ao_c->start_pts_known = ok;
ao_c->start_pts = pts;
mp_filter_wakeup(ao_c->ao_filter);
}
if (ao_c->ao && mp_async_queue_is_full(ao_c->ao_queue)) {
mpctx->audio_status = STATUS_READY;
mp_wakeup_core(mpctx);
MP_VERBOSE(mpctx, "audio ready\n");
} else if (ao_c->out_eof) {
// Force playback start early.
mpctx->audio_status = STATUS_READY;
mp_wakeup_core(mpctx);
MP_VERBOSE(mpctx, "audio ready (and EOF)\n");
}
}
if (ao_c->ao && !ao_is_playing(ao_c->ao) && !ao_c->underrun &&
(mpctx->audio_status == STATUS_PLAYING ||
mpctx->audio_status == STATUS_DRAINING))
{
// Should be playing, but somehow isn't.
if (ao_c->out_eof && !mp_async_queue_get_frames(ao_c->ao_queue)) {
MP_VERBOSE(mpctx, "AO signaled EOF (while in state %s)\n",
mp_status_str(mpctx->audio_status));
mpctx->audio_status = STATUS_EOF;
mp_wakeup_core(mpctx);
// stops untimed AOs, stops pull AOs from streaming silence
ao_reset(ao_c->ao);
} else {
if (!ao_c->ao_underrun) {
MP_WARN(mpctx, "Audio device underrun detected.\n");
ao_c->ao_underrun = true;
mp_wakeup_core(mpctx);
ao_c->underrun = true;
}
// Wait until buffers are filled before recovering underrun.
if (ao_c->out_eof || mp_async_queue_is_full(ao_c->ao_queue)) {
MP_VERBOSE(mpctx, "restarting audio after underrun\n");
ao_start(mpctx->ao_chain->ao);
ao_c->ao_underrun = false;
ao_c->underrun = false;
mp_wakeup_core(mpctx);
}
}
}
if (mpctx->audio_status == STATUS_PLAYING && ao_c->out_eof) {
mpctx->audio_status = STATUS_DRAINING;
MP_VERBOSE(mpctx, "audio draining\n");
mp_wakeup_core(mpctx);
}
if (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_c->ao || (!ao_is_playing(ao_c->ao) ||
(opts->gapless_audio && !ao_untimed(ao_c->ao))))
{
MP_VERBOSE(mpctx, "audio EOF reached\n");
mpctx->audio_status = STATUS_EOF;
mp_wakeup_core(mpctx);
}
}
if (mpctx->restart_complete)
audio_start_ao(mpctx); // in case it got delayed
}
// 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);
}
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