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mpv/player/audio.c
wm4 fee45c0170 player: silence sporadic error messages on audio init 
When the audio format is not known yet and the audio chain is still
initializing, filter reinit will fail. Normally, attempts to
reinitialize filters at this stage should be rare (e.g. user commands
editing the filter chain). But it sometimes happened with track
switching in combination with the video code calling
update_playback_speed() at arbitrary times.

Get rid of the message by not trying to change the filters for the sake
of playback speed update while decoding is still being initialized.
2015-11-10 17:53:05 +01:00

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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 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 "osdep/timer.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"
// Use pitch correction only for speed adjustments by the user, not minor sync
// correction ones.
static int get_speed_method(struct MPContext *mpctx)
{
return mpctx->opts->pitch_correction && mpctx->opts->playback_speed != 1.0
? AF_CONTROL_SET_PLAYBACK_SPEED : AF_CONTROL_SET_PLAYBACK_SPEED_RESAMPLE;
}
// Try to reuse the existing filters to change playback speed. If it works,
// return true; if filter recreation is needed, return false.
static bool update_speed_filters(struct MPContext *mpctx)
{
struct af_stream *afs = mpctx->d_audio->afilter;
double speed = mpctx->audio_speed;
if (afs->initialized < 1)
return false;
// 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_find_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 true;
return !!af_control_any_rev(afs, get_speed_method(mpctx), &speed);
}
// Update speed, and insert/remove filters if necessary.
static void recreate_speed_filters(struct MPContext *mpctx)
{
struct af_stream *afs = mpctx->d_audio->afilter;
if (update_speed_filters(mpctx))
return;
if (af_remove_by_label(afs, "playback-speed") < 0)
goto fail;
if (mpctx->audio_speed == 1.0)
return;
int method = get_speed_method(mpctx);
char *filter = method == AF_CONTROL_SET_PLAYBACK_SPEED
? "scaletempo" : "lavrresample";
if (!af_add(afs, filter, "playback-speed", NULL))
goto fail;
if (!update_speed_filters(mpctx))
goto fail;
return;
fail:
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);
}
static int recreate_audio_filters(struct MPContext *mpctx)
{
assert(mpctx->d_audio);
struct af_stream *afs = mpctx->d_audio->afilter;
if (afs->initialized < 1 && af_init(afs) < 0)
goto fail;
recreate_speed_filters(mpctx);
if (afs->initialized < 1 && af_init(afs) < 0)
goto fail;
mixer_reinit_audio(mpctx->mixer, mpctx->ao, afs);
return 0;
fail:
MP_ERR(mpctx, "Couldn't find matching filter/ao format!\n");
return -1;
}
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);
return recreate_audio_filters(mpctx) < 0 ? -1 : 1;
}
// 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;
if (!mpctx->d_audio || mpctx->d_audio->afilter->initialized < 1)
return;
if (!update_speed_filters(mpctx))
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;
mpctx->audio_drop_throttle = 0;
mpctx->audio_stat_start = 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) {
bool spdif_fallback = af_fmt_is_spdif(afs->output.format) &&
mpctx->d_audio->spdif_passthrough;
bool ao_null_fallback = opts->ao_null_fallback && !spdif_fallback;
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, ao_null_fallback, 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 (spdif_fallback) {
mpctx->d_audio->spdif_passthrough = false;
mpctx->d_audio->spdif_failed = true;
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;
mpctx->written_audio += played / (double)out_format.rate;
return played;
}
return 0;
}
static void dump_audio_stats(struct MPContext *mpctx)
{
if (!mp_msg_test(mpctx->log, MSGL_STATS))
return;
if (mpctx->audio_status != STATUS_PLAYING || !mpctx->ao || mpctx->paused) {
mpctx->audio_stat_start = 0;
return;
}
double delay = ao_get_delay(mpctx->ao);
if (!mpctx->audio_stat_start) {
mpctx->audio_stat_start = mp_time_us();
mpctx->written_audio = delay;
}
double current_audio = mpctx->written_audio - delay;
double current_time = (mp_time_us() - mpctx->audio_stat_start) / 1e6;
MP_STATS(mpctx, "value %f ao-dev", current_audio - current_time);
}
// 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;
} 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 = (int)(-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;
dump_audio_stats(mpctx);
if (mpctx->ao && ao_query_and_reset_events(mpctx->ao, AO_EVENT_RELOAD)) {
ao_reset(mpctx->ao);
uninit_audio_out(mpctx);
if (d_audio) {
if (mpctx->d_audio->spdif_failed) {
mpctx->d_audio->spdif_failed = false;
mpctx->d_audio->spdif_passthrough = true;
if (!audio_init_best_codec(mpctx->d_audio)) {
MP_ERR(mpctx, "Error reinitializing audio.\n");
error_on_track(mpctx, mpctx->current_track[0][STREAM_AUDIO]);
return;
}
}
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;
int align = af_format_sample_alignment(out_format.format);
// 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 skip_duplicate = 0; // >0: skip, <0: duplicate
double drop_limit =
(opts->sync_max_audio_change + opts->sync_max_video_change) / 100;
if (mpctx->display_sync_active && opts->video_sync == VS_DISP_ADROP &&
fabs(mpctx->last_av_difference) >= opts->sync_audio_drop_size &&
mpctx->audio_drop_throttle < drop_limit &&
mpctx->audio_status == STATUS_PLAYING)
{
int samples = ceil(opts->sync_audio_drop_size * play_samplerate);
samples = (samples + align / 2) / align * align;
skip_duplicate = mpctx->last_av_difference >= 0 ? -samples : samples;
playsize = MPMAX(playsize, samples);
mpctx->audio_drop_throttle += 1 - drop_limit - samples / play_samplerate;
}
playsize = playsize / align * align;
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 (skip_duplicate) {
int max = mp_audio_buffer_samples(mpctx->ao_buffer);
if (abs(skip_duplicate) > max)
skip_duplicate = skip_duplicate >= 0 ? max : -max;
mpctx->last_av_difference += skip_duplicate / play_samplerate;
if (skip_duplicate >= 0) {
mp_audio_buffer_skip(mpctx->ao_buffer, skip_duplicate);
MP_STATS(mpctx, "drop-audio");
} else {
mp_audio_buffer_duplicate(mpctx->ao_buffer, -skip_duplicate);
MP_STATS(mpctx, "duplicate-audio");
}
MP_VERBOSE(mpctx, "audio skip_duplicate=%d\n", skip_duplicate);
}
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 || end_sync)
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((int)samples / align * align, 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);
if (audio_eof || data.samples >= align)
data.samples = data.samples / align * align;
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_drop_throttle =
MPMAX(0, mpctx->audio_drop_throttle - played / play_samplerate);
dump_audio_stats(mpctx);
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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