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mpv/player/audio.c
wm4 fedaad8250 player: separate controls for user and video controlled speed 
For video sync, we want separate playback speed controls for user-
requested speed and the "correction" speed for video timing. Further, we
use this separation to make sure only a resampler is inserted if
playback speed is only changed for video sync correction.

As of this commit, this is basically inactive code. It's just
preparation for the video sync code (the following commit).
2015-08-10 18:40:16 +02:00

638 lines
21 KiB
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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 General Public License as published by
* the Free Software Foundation; either version 2 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 General Public License for more details.
*
* You should have received a copy of the GNU 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 "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 = mpctx->audio_speed;
struct af_stream *afs = mpctx->d_audio->afilter;
// Use pitch correction only for speed adjustments by the user, not minor
// sync correction ones.
bool use_pitch_correction = opts->pitch_correction &&
opts->playback_speed != 1.0;
// 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 (use_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;
mpctx->speed_factor_a = 1.0;
mpctx->audio_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;
}
// Call this if opts->playback_speed or mpctx->speed_correction changes.
void update_playback_speed(struct MPContext *mpctx)
{
struct MPOpts *opts = mpctx->opts;
double old_speed_factor_a = mpctx->speed_factor_a;
double old_audio_speed = mpctx->audio_speed;
mpctx->audio_speed = opts->playback_speed * mpctx->speed_factor_a;
mpctx->video_speed = opts->playback_speed * mpctx->speed_factor_v;
if (mpctx->speed_factor_a == old_speed_factor_a &&
mpctx->audio_speed == old_audio_speed)
return;
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);
mp_notify(mpctx, MPV_EVENT_AUDIO_RECONFIG, NULL);
}
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);
mp_notify(mpctx, MPV_EVENT_AUDIO_RECONFIG, NULL);
}
}
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->d_audio->spdif_passthrough = true;
mpctx->ao_buffer = mp_audio_buffer_create(NULL);
if (!audio_init_best_codec(mpctx->d_audio))
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_fmt_is_pcm(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 mp_audio fmt = {0};
if (mpctx->ao)
ao_get_format(mpctx->ao, &fmt);
// Verify passthrough format was not changed.
if (mpctx->ao && af_fmt_is_spdif(afs->output.format)) {
if (!mp_audio_config_equals(&afs->output, &fmt)) {
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 (af_fmt_is_spdif(afs->output.format) &&
mpctx->d_audio->spdif_passthrough)
{
mpctx->d_audio->spdif_passthrough = false;
if (!audio_init_best_codec(mpctx->d_audio))
goto init_error;
reset_audio_state(mpctx);
reinit_audio_chain(mpctx);
return;
}
MP_ERR(mpctx, "Could not open/initialize audio device -> no sound.\n");
mpctx->error_playing = MPV_ERROR_AO_INIT_FAILED;
goto init_error;
}
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(mpctx->ao),
mp_audio_config_to_str(&fmt));
MP_VERBOSE(mpctx, "AO: Description: %s\n", ao_get_description(mpctx->ao));
update_window_title(mpctx, true);
}
if (recreate_audio_filters(mpctx) < 0)
goto init_error;
update_playback_speed(mpctx);
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 audio_speed, so multiply by it
// to get the length in original units without speedup or slowdown
a_pts -= buffered_output * mpctx->audio_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->audio_speed * ao_get_delay(mpctx->ao);
}
static int write_to_ao(struct MPContext *mpctx, struct mp_audio *data, int flags)
{
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->audio_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 / mpctx->audio_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;
}
void 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 / 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);
}
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