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96d0a3f56c
mpv/player/audio.c
wm4 579c4dac34 audio: change a detail about filter insertion 
The af_add() function has a problem: if the inserted filter returns
AF_DETACH during init, the function will have a dangling pointer. Until
now this was avoided by making sure none of the used filters actually
return AF_DETACH, but it's getting infeasible.

Solve this by requiring passing an unique label to af_add(), which is
then used instead of the pointer.
2015-04-07 21:24:22 +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 update_playback_speed_filters(struct MPContext *mpctx)
{
struct MPOpts *opts = mpctx->opts;
double speed = opts->playback_speed;
struct af_stream *afs = mpctx->d_audio->afilter;
// Make sure only exactly one filter changes speed; resetting them all
// and setting 1 filter is the easiest way to achieve this.
af_control_all(afs, AF_CONTROL_SET_PLAYBACK_SPEED, &(double){1});
af_control_all(afs, AF_CONTROL_SET_PLAYBACK_SPEED_RESAMPLE, &(double){1});
if (speed == 1.0)
return af_remove_by_label(afs, "playback-speed");
// Compatibility: if the user uses --af=scaletempo, always use this
// filter to change speed. Don't insert a second filter (any) either.
if (!af_find_by_label(afs, "playback-speed") &&
af_control_any_rev(afs, AF_CONTROL_SET_PLAYBACK_SPEED, &speed))
return 0;
int method = AF_CONTROL_SET_PLAYBACK_SPEED_RESAMPLE;
if (opts->pitch_correction)
method = AF_CONTROL_SET_PLAYBACK_SPEED;
if (!af_control_any_rev(afs, method, &speed)) {
if (af_remove_by_label(afs, "playback-speed") < 0)
return -1;
char *filter = method == AF_CONTROL_SET_PLAYBACK_SPEED
? "scaletempo" : "lavrresample";
if (af_add(afs, filter, "playback-speed", NULL) < 0)
return -1;
// Try again.
if (!af_control_any_rev(afs, method, &speed))
return -1;
}
return 0;
}
static int recreate_audio_filters(struct MPContext *mpctx)
{
assert(mpctx->d_audio);
if (update_playback_speed_filters(mpctx) < 0) {
mpctx->opts->playback_speed = 1.0;
mp_notify(mpctx, MP_EVENT_CHANGE_ALL, NULL);
}
struct af_stream *afs = mpctx->d_audio->afilter;
if (afs->initialized < 1 && af_init(afs) < 0) {
MP_ERR(mpctx, "Couldn't find matching filter/ao format!\n");
return -1;
}
mixer_reinit_audio(mpctx->mixer, mpctx->ao, afs);
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 set_playback_speed(struct MPContext *mpctx, double new_speed)
{
struct MPOpts *opts = mpctx->opts;
// Adjust time until next frame flip for nosound mode
mpctx->time_frame *= opts->playback_speed / new_speed;
opts->playback_speed = new_speed;
if (!mpctx->d_audio || mpctx->d_audio->afilter->initialized < 1)
return;
recreate_audio_filters(mpctx);
}
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;
mpctx->delay = 0;
}
void uninit_audio_out(struct MPContext *mpctx)
{
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_drain(mpctx->ao);
mixer_uninit_audio(mpctx->mixer);
ao_uninit(mpctx->ao);
}
mpctx->ao = NULL;
talloc_free(mpctx->ao_decoder_fmt);
mpctx->ao_decoder_fmt = NULL;
}
void uninit_audio_chain(struct MPContext *mpctx)
{
if (mpctx->d_audio) {
mixer_uninit_audio(mpctx->mixer);
audio_uninit(mpctx->d_audio);
mpctx->d_audio = NULL;
talloc_free(mpctx->ao_buffer);
mpctx->ao_buffer = NULL;
mpctx->audio_status = STATUS_EOF;
reselect_demux_streams(mpctx);
}
}
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_audio_out(mpctx);
goto no_audio;
}
mp_notify(mpctx, MPV_EVENT_AUDIO_RECONFIG, NULL);
if (!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->pool = mp_audio_pool_create(mpctx->d_audio);
mpctx->d_audio->afilter = af_new(mpctx->global);
mpctx->d_audio->afilter->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 = mpctx->d_audio->decode_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;
}
// Weak gapless audio: drain AO on decoder format changes
if (mpctx->ao_decoder_fmt && mpctx->ao && opts->gapless_audio < 0 &&
!mp_audio_config_equals(mpctx->ao_decoder_fmt, &in_format))
{
uninit_audio_out(mpctx);
}
struct af_stream *afs = mpctx->d_audio->afilter;
afs->output = (struct mp_audio){0};
if (mpctx->ao) {
ao_get_format(mpctx->ao, &afs->output);
} else if (!AF_FORMAT_IS_SPECIAL(in_format.format)) {
afs->output.rate = opts->force_srate;
mp_audio_set_format(&afs->output, opts->audio_output_format);
mp_audio_set_channels(&afs->output, &opts->audio_output_channels);
}
// filter input format: same as codec's output format:
afs->input = in_format;
// 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 (af_init(afs) < 0) {
MP_ERR(mpctx, "Error at audio filter chain pre-init!\n");
goto init_error;
}
if (!mpctx->ao) {
mp_chmap_remove_useless_channels(&afs->output.channels,
&opts->audio_output_channels);
mp_audio_set_channels(&afs->output, &afs->output.channels);
mpctx->ao = ao_init_best(mpctx->global, mpctx->input,
mpctx->encode_lavc_ctx, afs->output.rate,
afs->output.format, afs->output.channels);
struct ao *ao = mpctx->ao;
if (!ao) {
MP_ERR(mpctx, "Could not open/initialize audio device -> no sound.\n");
mpctx->error_playing = MPV_ERROR_AO_INIT_FAILED;
goto init_error;
}
struct mp_audio fmt;
ao_get_format(ao, &fmt);
mp_audio_buffer_reinit(mpctx->ao_buffer, &fmt);
afs->output = fmt;
if (!mp_audio_config_equals(&afs->output, &afs->filter_output))
afs->initialized = 0;
mpctx->ao_decoder_fmt = talloc(NULL, struct mp_audio);
*mpctx->ao_decoder_fmt = in_format;
MP_INFO(mpctx, "AO: [%s] %s\n", ao_get_name(ao),
mp_audio_config_to_str(&fmt));
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;
set_playback_speed(mpctx, opts->playback_speed);
return;
init_error:
uninit_audio_chain(mpctx);
uninit_audio_out(mpctx);
no_audio:
if (track)
error_on_track(mpctx, track);
}
// 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 = d_audio->decode_format;
if (!mp_audio_config_valid(&in_format) || d_audio->afilter->initialized < 1)
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
if (d_audio->waiting)
a_pts -= d_audio->waiting->samples / (double)in_format.rate;
// 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;
if (!opts->initial_audio_sync) {
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 &&
mpctx->video_status != STATUS_EOF;
double sync_pts = MP_NOPTS_VALUE;
if (sync_to_video) {
if (mpctx->video_status < STATUS_READY)
return false; // wait until we know a video PTS
if (mpctx->video_next_pts != MP_NOPTS_VALUE)
sync_pts = mpctx->video_next_pts - (opts->audio_delay - mpctx->delay);
} 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
}
double ptsdiff = written_pts - sync_pts;
// Missing timestamp, or PTS reset, or just broken.
if (written_pts == MP_NOPTS_VALUE || fabs(ptsdiff) > 3600) {
MP_WARN(mpctx, "Failed audio resync.\n");
mpctx->audio_status = STATUS_FILLING;
return true;
}
int align = af_format_sample_alignment(out_format.format);
*skip = (-ptsdiff * play_samplerate) / align * align;
return true;
}
static void do_fill_audio_out_buffers(struct MPContext *mpctx, double endpts)
{
struct MPOpts *opts = mpctx->opts;
struct dec_audio *d_audio = mpctx->d_audio;
if (mpctx->ao && ao_query_and_reset_events(mpctx->ao, AO_EVENT_RELOAD)) {
ao_reset(mpctx->ao);
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 / 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;
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;
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;
}
// 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;
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;
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;
}
}
void fill_audio_out_buffers(struct MPContext *mpctx, double endpts)
{
do_fill_audio_out_buffers(mpctx, endpts);
// Run audio playback state machine again to display the actual audio PTS
// as current time on OSD in audio-only mode in most situations.
if (mpctx->audio_status == STATUS_SYNCING)
do_fill_audio_out_buffers(mpctx, endpts);
}
// 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);
}
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