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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 "config.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/audio_buffer.h"
#include "audio/format.h"
#include "audio/out/ao.h"
#include "demux/demux.h"
#include "filters/f_decoder_wrapper.h"
#include "core.h"
#include "command.h"
enum {
AD_OK = 0,
AD_EOF = -2,
AD_WAIT = -4,
};
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 && mpctx->opts->video_sync == VS_DISP_ADROP) {
drop *= speed * resample;
resample = speed = 1.0;
}
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 void ao_chain_reset_state(struct ao_chain *ao_c)
{
ao_c->last_out_pts = MP_NOPTS_VALUE;
TA_FREEP(&ao_c->output_frame);
ao_c->out_eof = false;
ao_c->underrun = false;
mp_audio_buffer_clear(ao_c->ao_buffer);
}
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->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);
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->output_frame);
talloc_free(ao_c->ao_buffer);
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);
if (!new_mod)
abort();
// 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 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;
if (!ao_c->filter->ao_needs_update)
return;
TA_FREEP(&ao_c->output_frame); // stale?
// The "ideal" filter output format
struct mp_aframe *out_fmt = mp_aframe_new_ref(ao_c->filter->output_aformat);
if (!out_fmt)
abort();
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))
{
mp_output_chain_set_ao(ao_c->filter, mpctx->ao);
talloc_free(out_fmt);
return;
}
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);
}
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);
ao_c->ao = mpctx->ao;
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;
ao_c->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;
}
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_fmt(ao_c->ao_buffer, ao_format, &ao_channels,
ao_rate);
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;
mp_output_chain_set_ao(ao_c->filter, mpctx->ao);
audio_update_volume(mpctx);
mp_notify(mpctx, MPV_EVENT_AUDIO_RECONFIG, NULL);
return;
init_error:
uninit_audio_chain(mpctx);
uninit_audio_out(mpctx);
error_on_track(mpctx, track);
}
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) {
uninit_audio_out(mpctx);
error_on_track(mpctx, track);
return;
}
reinit_audio_chain_src(mpctx, track);
}
// (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->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->ao_buffer = mp_audio_buffer_create(NULL);
ao_c->ao = mpctx->ao;
ao_c->delay = mpctx->opts->audio_delay;
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) {
int rate;
int format;
struct mp_chmap channels;
ao_get_format(mpctx->ao, &rate, &format, &channels);
mp_audio_buffer_reinit_fmt(ao_c->ao_buffer, format, &channels, rate);
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 end point of audio written to the
// ao so far.
double written_audio_pts(struct MPContext *mpctx)
{
struct ao_chain *ao_c = mpctx->ao_chain;
if (!ao_c)
return MP_NOPTS_VALUE;
// end pts of audio that has been output by filters
double a_pts = ao_c->last_out_pts;
if (a_pts == MP_NOPTS_VALUE)
return MP_NOPTS_VALUE;
// Data that was ready for ao but was buffered because ao didn't fully
// accept everything to internal buffers yet. This also does not correctly
// track playback speed, so we use the current speed.
a_pts -= mp_audio_buffer_seconds(ao_c->ao_buffer) * mpctx->audio_speed;
return a_pts;
}
// 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, uint8_t **planes, int samples,
int flags)
{
if (mpctx->paused)
return 0;
struct ao *ao = mpctx->ao;
int samplerate;
int format;
struct mp_chmap channels;
ao_get_format(ao, &samplerate, &format, &channels);
encode_lavc_set_audio_pts(mpctx->encode_lavc_ctx, playing_audio_pts(mpctx));
if (samples == 0)
return 0;
double real_samplerate = samplerate / mpctx->audio_speed;
int played = ao_play(mpctx->ao, (void **)planes, samples, flags);
assert(played <= samples);
if (played > 0) {
mpctx->shown_aframes += played;
mpctx->delay += played / real_samplerate;
mpctx->written_audio += played / (double)samplerate;
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;
int ao_rate;
int ao_format;
struct mp_chmap ao_channels;
ao_get_format(mpctx->ao, &ao_rate, &ao_format, &ao_channels);
double play_samplerate = ao_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_chain->ao_buffer))
return false; // no audio read yet
bool sync_to_video = mpctx->vo_chain && mpctx->video_status != STATUS_EOF &&
!mpctx->vo_chain->is_sparse;
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_pts != MP_NOPTS_VALUE)
sync_pts = mpctx->video_pts - opts->audio_delay;
} else if (mpctx->hrseek_active) {
sync_pts = mpctx->hrseek_pts;
} else {
// If audio-only is enabled mid-stream during playback, sync accordingly.
sync_pts = mpctx->playback_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) {
MP_WARN(mpctx, "Failed audio resync.\n");
mpctx->audio_status = STATUS_FILLING;
return true;
}
ptsdiff = MPCLAMP(ptsdiff, -3600, 3600);
MP_VERBOSE(mpctx, "audio sync: sync_to_video=%d, offset=%f\n",
sync_to_video, ptsdiff);
int align = af_format_sample_alignment(ao_format);
*skip = (int)(-ptsdiff * play_samplerate) / align * align;
return true;
}
static bool copy_output(struct MPContext *mpctx, struct ao_chain *ao_c,
int minsamples, double endpts, bool *seteof)
{
struct mp_audio_buffer *outbuf = ao_c->ao_buffer;
int ao_rate;
int ao_format;
struct mp_chmap ao_channels;
ao_get_format(ao_c->ao, &ao_rate, &ao_format, &ao_channels);
while (mp_audio_buffer_samples(outbuf) < minsamples) {
int cursamples = mp_audio_buffer_samples(outbuf);
int maxsamples = INT_MAX;
if (endpts != MP_NOPTS_VALUE) {
double rate = ao_rate / mpctx->audio_speed;
double curpts = written_audio_pts(mpctx);
if (curpts != MP_NOPTS_VALUE) {
double remaining =
(endpts - curpts - mpctx->opts->audio_delay) * rate;
maxsamples = MPCLAMP(remaining, 0, INT_MAX);
}
}
if (!ao_c->output_frame || !mp_aframe_get_size(ao_c->output_frame)) {
TA_FREEP(&ao_c->output_frame);
struct mp_frame frame = mp_pin_out_read(ao_c->filter->f->pins[1]);
if (frame.type == MP_FRAME_AUDIO) {
ao_c->output_frame = frame.data;
ao_c->out_eof = false;
ao_c->last_out_pts = mp_aframe_end_pts(ao_c->output_frame);
} else if (frame.type == MP_FRAME_EOF) {
ao_c->out_eof = true;
} else if (frame.type) {
MP_ERR(mpctx, "unknown frame type\n");
mp_frame_unref(&frame);
}
}
// out of data
if (!ao_c->output_frame) {
if (ao_c->out_eof) {
*seteof = true;
return true;
}
return false;
}
if (cursamples + mp_aframe_get_size(ao_c->output_frame) > maxsamples) {
if (cursamples < maxsamples) {
uint8_t **data = mp_aframe_get_data_ro(ao_c->output_frame);
mp_audio_buffer_append(outbuf, (void **)data,
maxsamples - cursamples);
mp_aframe_skip_samples(ao_c->output_frame,
maxsamples - cursamples);
}
*seteof = true;
return true;
}
uint8_t **data = mp_aframe_get_data_ro(ao_c->output_frame);
mp_audio_buffer_append(outbuf, (void **)data,
mp_aframe_get_size(ao_c->output_frame));
TA_FREEP(&ao_c->output_frame);
}
return true;
}
/* Try to get at least minsamples decoded+filtered samples in outbuf
* (total length including possible existing data).
* Return 0 on success, or negative AD_* error code.
* In the former case outbuf has at least minsamples buffered on return.
* In case of EOF/error it might or might not be. */
static int filter_audio(struct MPContext *mpctx, struct mp_audio_buffer *outbuf,
int minsamples)
{
struct ao_chain *ao_c = mpctx->ao_chain;
double endpts = get_play_end_pts(mpctx);
if (endpts != MP_NOPTS_VALUE)
endpts *= mpctx->play_dir;
bool eof = false;
if (!copy_output(mpctx, ao_c, minsamples, endpts, &eof))
return AD_WAIT;
return eof ? AD_EOF : AD_OK;
}
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);
}
void fill_audio_out_buffers(struct MPContext *mpctx)
{
struct MPOpts *opts = mpctx->opts;
bool was_eof = mpctx->audio_status == STATUS_EOF;
dump_audio_stats(mpctx);
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);
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 is set due to gapless from previous file, then we can try to
// filter normally until the filter tells us to change the AO)
if (!mpctx->ao) {
// Probe the initial audio format.
mp_pin_out_request_data(ao_c->filter->f->pins[1]);
reinit_audio_filters_and_output(mpctx);
if (!mpctx->ao_chain)
return;
if (ao_c->filter->got_output_eof &&
mpctx->audio_status != STATUS_EOF)
{
mpctx->audio_status = STATUS_EOF;
MP_VERBOSE(mpctx, "audio EOF without any data\n");
mp_filter_reset(ao_c->filter->f);
encode_lavc_stream_eof(mpctx->encode_lavc_ctx, STREAM_AUDIO);
}
return; // try again next iteration
}
if (ao_c->ao_resume_time > mp_time_sec()) {
double remaining = ao_c->ao_resume_time - mp_time_sec();
mp_set_timeout(mpctx, remaining);
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);
return;
}
int ao_rate;
int ao_format;
struct mp_chmap ao_channels;
ao_get_format(mpctx->ao, &ao_rate, &ao_format, &ao_channels);
double play_samplerate = ao_rate / mpctx->audio_speed;
int align = af_format_sample_alignment(ao_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);
if (ao_query_and_reset_events(mpctx->ao, AO_EVENT_UNDERRUN)) {
if (!ao_c->underrun)
MP_WARN(mpctx, "Audio device underrun detected.\n");
ao_c->underrun = true;
}
// Stop feeding data if an underrun happened. Something else needs to
// "unblock" audio after underrun. handle_update_cache() does this and can
// take the network state into account.
if (ao_c->underrun)
return;
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
}
playsize = playsize / align * align;
int status = mpctx->audio_status >= STATUS_DRAINING ? AD_EOF : AD_OK;
bool working = false;
if (playsize > mp_audio_buffer_samples(ao_c->ao_buffer)) {
status = filter_audio(mpctx, ao_c->ao_buffer, playsize);
if (ao_c->filter->ao_needs_update) {
reinit_audio_filters_and_output(mpctx);
mp_wakeup_core(mpctx);
return; // retry on next iteration
}
if (status == AD_WAIT)
return;
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 &&
mp_audio_buffer_samples(ao_c->ao_buffer) > 0)
{
mpctx->audio_status = STATUS_SYNCING;
return; // retry on next iteration
}
bool end_sync = false;
if (skip >= 0) {
int max = mp_audio_buffer_samples(ao_c->ao_buffer);
mp_audio_buffer_skip(ao_c->ao_buffer, MPMIN(skip, max));
// If something is left, we definitely reached the target time.
end_sync |= sync_known && skip < max;
working |= skip > 0;
} 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(ao_c->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(ao_c->ao_buffer))
mpctx->audio_status = STATUS_EOF;
if (working || end_sync)
mp_wakeup_core(mpctx);
return; // continue on next iteration
}
assert(mpctx->audio_status >= STATUS_FILLING);
// We already have as much data as the audio device wants, and can start
// writing it any time.
if (mpctx->audio_status == STATUS_FILLING)
mpctx->audio_status = STATUS_READY;
// 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_READY) {
// Warning: relies on handle_playback_restart() being called afterwards.
return;
}
bool audio_eof = status == AD_EOF;
bool partial_fill = false;
int playflags = 0;
if (playsize > mp_audio_buffer_samples(ao_c->ao_buffer)) {
playsize = mp_audio_buffer_samples(ao_c->ao_buffer);
partial_fill = true;
}
audio_eof &= partial_fill;
if (audio_eof && playsize < align)
playsize = 0;
// With gapless audio, delay this to ao_uninit. There must be only
// 1 final chunk, and that is handled when calling ao_uninit().
// If video is still on-going, trying to do gapless is pointless, as video
// will have to continue for a while with audio stopped (but still try to
// do it if gapless is forced, mostly for testing).
if (audio_eof && (!opts->gapless_audio ||
(opts->gapless_audio <= 0 && mpctx->video_status != STATUS_EOF)))
playflags |= AOPLAY_FINAL_CHUNK;
uint8_t **planes;
int samples;
mp_audio_buffer_peek(ao_c->ao_buffer, &planes, &samples);
if (audio_eof || samples >= align)
samples = samples / align * align;
samples = MPMIN(samples, mpctx->paused ? 0 : playsize);
int played = write_to_ao(mpctx, planes, samples, playflags);
assert(played >= 0 && played <= samples);
mp_audio_buffer_skip(ao_c->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;
if (!was_eof) {
MP_VERBOSE(mpctx, "audio EOF reached\n");
mp_wakeup_core(mpctx);
encode_lavc_stream_eof(mpctx->encode_lavc_ctx, STREAM_AUDIO);
}
}
}
}
// 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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