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mpv/player/audio.c

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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 "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) {
switch (mpctx->video_out->opts->video_sync) {
case VS_DISP_ADROP:
drop *= speed * resample;
resample = speed = 1.0;
break;
case VS_DISP_TEMPO:
speed = mpctx->audio_speed;
resample = 1.0;
break;
}
}
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 bool has_video_track(struct MPContext *mpctx)
{
if (mpctx->vo_chain && mpctx->vo_chain->is_coverart)
return false;
for (int n = 0; n < mpctx->num_tracks; n++) {
struct track *track = mpctx->tracks[n];
if (track->type == STREAM_VIDEO && !track->attached_picture && !track->image)
return true;
}
return false;
}
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->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);
MP_HANDLE_OOM(new_mod);
// 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);
MP_HANDLE_OOM(out_fmt);
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);
}
if (!has_video_track(mpctx))
ao_flags |= AO_INIT_MEDIA_ROLE_MUSIC;
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;
bool eof = mpctx->audio_status == STATUS_EOF;
ao_set_paused(mpctx->ao, get_internal_paused(mpctx), eof);
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;
if (mpctx->video_status != STATUS_EOF)
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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