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7791e25fa0
mpv/player/audio.c
wm4 7791e25fa0 audio: fix initial sync with huge AO buffer 
With e.g --start=-3 --audio-buffer=10 the decoder entered EOF state
before the initial sync was finished, entered STATUS_EOF, and just
started playing audio from a random position.

This doesn't handle seeking outside of the file, which is a different
case. E.g. --start=30:00 with audio and video enabled in a file shorter
than 30:00 will play a random last part of audio. This could perhaps be
fixed by using the hr-seek target for cutting audio, instead of the
video PTS, but that would be kind of intrusive, so don't do it for now.
The simpler solution, assuming audio EOF on video EOF, wouldn't work,
because we allow audio to start before video, or to last after video.
2014-09-06 13:33:29 +02:00

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/*
* This file is part of MPlayer.
*
* MPlayer is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*
* MPlayer is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License along
* with MPlayer; if not, write to the Free Software Foundation, Inc.,
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
*/
#include <stddef.h>
#include <stdbool.h>
#include <inttypes.h>
#include <limits.h>
#include <math.h>
#include <assert.h>
#include "config.h"
#include "talloc.h"
#include "common/msg.h"
#include "common/encode.h"
#include "options/options.h"
#include "common/common.h"
#include "audio/mixer.h"
#include "audio/audio.h"
#include "audio/audio_buffer.h"
#include "audio/decode/dec_audio.h"
#include "audio/filter/af.h"
#include "audio/out/ao.h"
#include "demux/demux.h"
#include "video/decode/dec_video.h"
#include "core.h"
#include "command.h"
static int recreate_audio_filters(struct MPContext *mpctx)
{
struct dec_audio *d_audio = mpctx->d_audio;
struct MPOpts *opts = mpctx->opts;
assert(d_audio);
struct mp_audio in_format;
mp_audio_buffer_get_format(d_audio->decode_buffer, &in_format);
struct mp_audio out_format;
ao_get_format(mpctx->ao, &out_format);
int new_srate;
if (af_control_any_rev(d_audio->afilter, AF_CONTROL_SET_PLAYBACK_SPEED,
&opts->playback_speed))
new_srate = in_format.rate;
else {
new_srate = in_format.rate * opts->playback_speed;
if (new_srate != out_format.rate)
opts->playback_speed = new_srate / (double)in_format.rate;
}
if (!audio_init_filters(d_audio, new_srate,
&out_format.rate, &out_format.channels, &out_format.format))
{
MP_ERR(mpctx, "Couldn't find matching filter/ao format!\n");
return -1;
}
mixer_reinit_audio(mpctx->mixer, mpctx->ao, mpctx->d_audio->afilter);
return 0;
}
int reinit_audio_filters(struct MPContext *mpctx)
{
struct dec_audio *d_audio = mpctx->d_audio;
if (!d_audio)
return 0;
af_uninit(mpctx->d_audio->afilter);
if (af_init(mpctx->d_audio->afilter) < 0)
return -1;
if (recreate_audio_filters(mpctx) < 0)
return -1;
return 1;
}
void reset_audio_state(struct MPContext *mpctx)
{
if (mpctx->d_audio)
audio_reset_decoding(mpctx->d_audio);
if (mpctx->ao_buffer)
mp_audio_buffer_clear(mpctx->ao_buffer);
mpctx->audio_status = mpctx->d_audio ? STATUS_SYNCING : STATUS_EOF;
}
void reinit_audio_chain(struct MPContext *mpctx)
{
struct MPOpts *opts = mpctx->opts;
struct track *track = mpctx->current_track[0][STREAM_AUDIO];
struct sh_stream *sh = track ? track->stream : NULL;
if (!sh) {
uninit_player(mpctx, INITIALIZED_AO);
goto no_audio;
}
mp_notify(mpctx, MPV_EVENT_AUDIO_RECONFIG, NULL);
if (!(mpctx->initialized_flags & INITIALIZED_ACODEC)) {
mpctx->initialized_flags |= INITIALIZED_ACODEC;
assert(!mpctx->d_audio);
mpctx->d_audio = talloc_zero(NULL, struct dec_audio);
mpctx->d_audio->log = mp_log_new(mpctx->d_audio, mpctx->log, "!ad");
mpctx->d_audio->global = mpctx->global;
mpctx->d_audio->opts = opts;
mpctx->d_audio->header = sh;
mpctx->d_audio->replaygain_data = sh->audio->replaygain_data;
mpctx->ao_buffer = mp_audio_buffer_create(NULL);
if (!audio_init_best_codec(mpctx->d_audio, opts->audio_decoders))
goto init_error;
reset_audio_state(mpctx);
if (mpctx->ao) {
struct mp_audio fmt;
ao_get_format(mpctx->ao, &fmt);
mp_audio_buffer_reinit(mpctx->ao_buffer, &fmt);
}
}
assert(mpctx->d_audio);
struct mp_audio in_format;
mp_audio_buffer_get_format(mpctx->d_audio->decode_buffer, &in_format);
if (!mp_audio_config_valid(&in_format)) {
// We don't know the audio format yet - so configure it later as we're
// resyncing. fill_audio_buffers() will call this function again.
mpctx->sleeptime = 0;
return;
}
if (mpctx->ao_decoder_fmt && (mpctx->initialized_flags & INITIALIZED_AO) &&
!mp_audio_config_equals(mpctx->ao_decoder_fmt, &in_format) &&
opts->gapless_audio < 0)
{
uninit_player(mpctx, INITIALIZED_AO);
}
int ao_srate = opts->force_srate;
int ao_format = opts->audio_output_format;
struct mp_chmap ao_channels = {0};
if (mpctx->initialized_flags & INITIALIZED_AO) {
struct mp_audio out_format;
ao_get_format(mpctx->ao, &out_format);
ao_srate = out_format.rate;
ao_format = out_format.format;
ao_channels = out_format.channels;
} else {
if (!AF_FORMAT_IS_SPECIAL(in_format.format))
ao_channels = opts->audio_output_channels; // automatic downmix
}
// Determine what the filter chain outputs. recreate_audio_filters() also
// needs this for testing whether playback speed is changed by resampling
// or using a special filter.
if (!audio_init_filters(mpctx->d_audio, // preliminary init
// input:
in_format.rate,
// output:
&ao_srate, &ao_channels, &ao_format)) {
MP_ERR(mpctx, "Error at audio filter chain pre-init!\n");
goto init_error;
}
if (!(mpctx->initialized_flags & INITIALIZED_AO)) {
mpctx->initialized_flags |= INITIALIZED_AO;
mp_chmap_remove_useless_channels(&ao_channels,
&opts->audio_output_channels);
mpctx->ao = ao_init_best(mpctx->global, mpctx->input,
mpctx->encode_lavc_ctx, ao_srate, ao_format,
ao_channels);
struct ao *ao = mpctx->ao;
if (!ao) {
MP_ERR(mpctx, "Could not open/initialize audio device -> no sound.\n");
goto init_error;
}
struct mp_audio fmt;
ao_get_format(ao, &fmt);
mp_audio_buffer_reinit(mpctx->ao_buffer, &fmt);
mpctx->ao_decoder_fmt = talloc(NULL, struct mp_audio);
*mpctx->ao_decoder_fmt = in_format;
char *s = mp_audio_config_to_str(&fmt);
MP_INFO(mpctx, "AO: [%s] %s\n", ao_get_name(ao), s);
talloc_free(s);
MP_VERBOSE(mpctx, "AO: Description: %s\n", ao_get_description(ao));
update_window_title(mpctx, true);
}
if (recreate_audio_filters(mpctx) < 0)
goto init_error;
return;
init_error:
uninit_player(mpctx, INITIALIZED_ACODEC | INITIALIZED_AO);
no_audio:
mp_deselect_track(mpctx, track);
MP_INFO(mpctx, "Audio: no audio\n");
}
// Return pts value corresponding to the end point of audio written to the
// ao so far.
double written_audio_pts(struct MPContext *mpctx)
{
struct dec_audio *d_audio = mpctx->d_audio;
if (!d_audio)
return MP_NOPTS_VALUE;
struct mp_audio in_format;
mp_audio_buffer_get_format(d_audio->decode_buffer, &in_format);
if (!mp_audio_config_valid(&in_format) || !d_audio->afilter)
return MP_NOPTS_VALUE;
// first calculate the end pts of audio that has been output by decoder
double a_pts = d_audio->pts;
if (a_pts == MP_NOPTS_VALUE)
return MP_NOPTS_VALUE;
// d_audio->pts is the timestamp of the latest input packet with
// known pts that the decoder has decoded. d_audio->pts_bytes is
// the amount of bytes the decoder has written after that timestamp.
a_pts += d_audio->pts_offset / (double)in_format.rate;
// Now a_pts hopefully holds the pts for end of audio from decoder.
// Subtract data in buffers between decoder and audio out.
// Decoded but not filtered
a_pts -= mp_audio_buffer_seconds(d_audio->decode_buffer);
// Data buffered in audio filters, measured in seconds of "missing" output
double buffered_output = af_calc_delay(d_audio->afilter);
// Data that was ready for ao but was buffered because ao didn't fully
// accept everything to internal buffers yet
buffered_output += mp_audio_buffer_seconds(mpctx->ao_buffer);
// Filters divide audio length by playback_speed, so multiply by it
// to get the length in original units without speedup or slowdown
a_pts -= buffered_output * mpctx->opts->playback_speed;
return a_pts +
get_track_video_offset(mpctx, mpctx->current_track[0][STREAM_AUDIO]);
}
// 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->opts->playback_speed * ao_get_delay(mpctx->ao);
}
static int write_to_ao(struct MPContext *mpctx, struct mp_audio *data, int flags,
double pts)
{
if (mpctx->paused)
return 0;
struct ao *ao = mpctx->ao;
struct mp_audio out_format;
ao_get_format(ao, &out_format);
#if HAVE_ENCODING
encode_lavc_set_audio_pts(mpctx->encode_lavc_ctx, playing_audio_pts(mpctx));
#endif
if (data->samples == 0)
return 0;
double real_samplerate = out_format.rate / mpctx->opts->playback_speed;
int played = ao_play(mpctx->ao, data->planes, data->samples, flags);
assert(played <= data->samples);
if (played > 0) {
mpctx->shown_aframes += played;
mpctx->delay += played / real_samplerate;
return played;
}
return 0;
}
// Return the number of samples that must be skipped or prepended to reach the
// target audio pts after a seek (for A/V sync or hr-seek).
// Return value (*skip):
// >0: skip this many samples
// =0: don't do anything
// <0: prepend this many samples of silence
// Returns false if PTS is not known yet.
static bool get_sync_samples(struct MPContext *mpctx, int *skip)
{
struct MPOpts *opts = mpctx->opts;
*skip = 0;
if (mpctx->audio_status != STATUS_SYNCING)
return true;
struct mp_audio out_format = {0};
ao_get_format(mpctx->ao, &out_format);
double play_samplerate = out_format.rate / opts->playback_speed;
bool is_pcm = !(out_format.format & AF_FORMAT_SPECIAL_MASK); // no spdif
if (!opts->initial_audio_sync || !is_pcm) {
mpctx->audio_status = STATUS_FILLING;
return true;
}
double written_pts = written_audio_pts(mpctx);
if (written_pts == MP_NOPTS_VALUE && !mp_audio_buffer_samples(mpctx->ao_buffer))
return false; // no audio read yet
bool sync_to_video = mpctx->d_video && mpctx->sync_audio_to_video;
double sync_pts = MP_NOPTS_VALUE;
if (sync_to_video) {
if (mpctx->video_next_pts != MP_NOPTS_VALUE) {
sync_pts = mpctx->video_next_pts;
} else if (mpctx->video_status < STATUS_READY) {
return false; // wait until we know a video PTS
}
} else if (mpctx->hrseek_active) {
sync_pts = mpctx->hrseek_pts;
}
if (sync_pts == MP_NOPTS_VALUE) {
mpctx->audio_status = STATUS_FILLING;
return true; // syncing disabled
}
if (sync_to_video)
sync_pts -= mpctx->audio_delay - mpctx->delay;
double ptsdiff = written_pts - sync_pts;
// Missing timestamp, or PTS reset, or just broken.
if (written_pts == MP_NOPTS_VALUE || fabs(ptsdiff) > 300) {
MP_WARN(mpctx, "Failed audio resync.\n");
mpctx->audio_status = STATUS_FILLING;
return true;
}
*skip = -ptsdiff * play_samplerate;
return true;
}
void fill_audio_out_buffers(struct MPContext *mpctx, double endpts)
{
struct MPOpts *opts = mpctx->opts;
struct dec_audio *d_audio = mpctx->d_audio;
if (!d_audio)
return;
if (!d_audio->afilter || !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
mpctx->d_audio->init_retries += 1;
MP_VERBOSE(mpctx, "Initial audio packets read: %d\n",
mpctx->d_audio->init_retries);
if (r != AD_OK && mpctx->d_audio->init_retries >= 50) {
MP_ERR(mpctx, "Error initializing audio.\n");
struct track *track = mpctx->current_track[0][STREAM_AUDIO];
mp_deselect_track(mpctx, track);
return;
}
reinit_audio_chain(mpctx);
return; // try again next iteration
}
struct mp_audio out_format = {0};
ao_get_format(mpctx->ao, &out_format);
double play_samplerate = out_format.rate / opts->playback_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;
// if paused, just initialize things (audio format & pts)
int playsize = 1;
if (!mpctx->paused)
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
}
if (opts->insert_silence) {
float S = 0.5;
if (!mpctx->paused && mpctx->audio_status == STATUS_PLAYING &&
mpctx->last_av_difference - mpctx->insert_silence > S)
mpctx->insert_silence += S;
if (mpctx->insert_silence > 0) {
int samples = MPMIN(playsize, play_samplerate * mpctx->insert_silence);
mp_audio_buffer_prepend_silence(mpctx->ao_buffer, samples);
mpctx->insert_silence -= samples / play_samplerate;
}
}
int status = AD_OK;
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_player(mpctx, INITIALIZED_AO);
reinit_audio_chain(mpctx);
mpctx->sleeptime = 0;
return; // retry on next iteration
}
}
// 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;
mpctx->sleeptime = 0;
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;
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) - mpctx->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;
if (mpctx->paused)
playsize = 0;
struct mp_audio data;
mp_audio_buffer_peek(mpctx->ao_buffer, &data);
data.samples = MPMIN(data.samples, playsize);
int played = write_to_ao(mpctx, &data, playflags, written_audio_pts(mpctx));
assert(played >= 0 && played <= data.samples);
mp_audio_buffer_skip(mpctx->ao_buffer, played);
mpctx->audio_status = STATUS_PLAYING;
if (audio_eof) {
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);
mp_audio_buffer_clear(mpctx->ao_buffer);
}
}
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