RepoMirrors/mpv
1
0
Fork 0
mirror of https://github.com/mpv-player/mpv synced 2024-12-30 02:52:10 +00:00
Code Issues Releases Wiki Activity
a6c8b4efa5
mpv/player/audio.c
wm4 f26dfb6e4d player: partially rework --cache-pause 
The --cache-pause feature (enabled by default) will pause playback for a
while if network runs out of data. If this is not done, then playback
will go on frame-wise (as packets are slowly read from the network and
then instantly decoded and displayed). This feature is actually useless,
as you won't get nice playback no matter what if network is too slow,
but I guess I still prefer this behavior for some reason.

This commit changes this behavior from using the demuxer cache state
only, to trying to use underrun information from the AO/VO. This means
if you have a very large audio buffer, then cache-pausing will trigger
once that buffer is depleted, which will be some time _after_ the
demuxer cache has run out.

This requires explicit support from the AO. Otherwise, the behavior
should be mostly the same as before this commit.

This does not care about the AO buffer. In theory, the AO may underrun,
then the player will write some data to the AO buffer, then the AO will
recover and play this bit of data, then the player will probably trigger
the cache-pause behavior. The probability of this happening should be
pretty low, so I will hold off fixing this until the next refactor of
the AO chain (if ever).

The VO underflow detection was devised and tested in 5 minutes, and may
not be correct. At least I'm fairly sure that the combination of all the
factors should make incorrect behavior relatively unlikely, but problems
are possible.

Also, the demux_reader_state.underrun field may be inaccurate. It's only
the present state at the time demux_get_reader_state() was called, and
may exclude past underruns. In theory, this could cause "close" cases to
be missed. Then you might get an audio underrun without cache-pausing
acting on it. If the stars align, this could happen multiple times in
the row, effectively making this feature not work.

The most user-visible consequence of this change is that the user
will now see an AO underrun warning every time the cache runs out.

Maybe this cache-pause feature should just be removed...
2019-10-11 20:01:51 +02:00

1037 lines
32 KiB
C
Raw Blame History

/*
* 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,
};
// Try to reuse the existing filters to change playback speed. If it works,
// return true; if filter recreation is needed, return false.
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;
if (!mpctx->opts->pitch_correction) {
resample *= speed;
speed = 1.0;
}
mp_output_chain_set_audio_speed(ao_c->filter, speed, resample);
}
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)
t->dec->play_dir = 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;
ao_c->track->dec->try_spdif = 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)
track->dec->try_spdif = 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;
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->vo_chain->is_coverart &&
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_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);
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;
dec->try_spdif = 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 &&
ao_c->track->dec->pts_reset)
{
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))
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
}
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 = 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 (skip_duplicate) {
int max = mp_audio_buffer_samples(ao_c->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(ao_c->ao_buffer, skip_duplicate);
MP_STATS(mpctx, "drop-audio");
} else {
mp_audio_buffer_duplicate(ao_c->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(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);
}
Reference in New Issue View Git Blame Copy Permalink