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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/>.
*
* Parts under HAVE_LIBAF are partially licensed under GNU General Public
* License (libaf/af.h glue code only).
*/
#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/aconverter.h"
#include "audio/format.h"
#include "audio/decode/dec_audio.h"
#include "audio/out/ao.h"
#include "demux/demux.h"
#include "video/decode/dec_video.h"
#include "core.h"
#include "command.h"
enum {
AD_OK = 0,
AD_ERR = -1,
AD_EOF = -2,
AD_NEW_FMT = -3,
AD_WAIT = -4,
AD_NO_PROGRESS = -5,
AD_STARVE = -6,
};
#if HAVE_LIBAF
#include "audio/audio.h"
#include "audio/filter/af.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->ao_chain->af;
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->ao_chain->af;
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 double db_gain(double db)
{
return pow(10.0, db/20.0);
}
static float compute_replaygain(struct MPContext *mpctx)
{
struct MPOpts *opts = mpctx->opts;
struct ao_chain *ao_c = mpctx->ao_chain;
float rgain = 1.0;
struct replaygain_data *rg = ao_c->af->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->af->initialized < 1)
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;
if (!af_control_any_rev(ao_c->af, AF_CONTROL_SET_VOLUME, &gain)) {
if (gain == 1.0)
return;
MP_VERBOSE(mpctx, "Inserting volume filter.\n");
char *args[] = {"warn", "no", NULL};
if (!(af_add(ao_c->af, "volume", "softvol", args)
&& af_control_any_rev(ao_c->af, AF_CONTROL_SET_VOLUME, &gain)))
MP_ERR(mpctx, "No volume control available.\n");
}
}
/* NOTE: Currently the balance code is seriously buggy: it always changes
* the af_pan mapping between the first two input channels and first two
* output channels to particular values. These values make sense for an
* af_pan instance that was automatically inserted for balance control
* only and is otherwise an identity transform, but if the filter was
* there for another reason, then ignoring and overriding the original
* values is completely wrong.
*/
void audio_update_balance(struct MPContext *mpctx)
{
struct MPOpts *opts = mpctx->opts;
struct ao_chain *ao_c = mpctx->ao_chain;
if (!ao_c || ao_c->af->initialized < 1)
return;
float val = opts->balance;
if (af_control_any_rev(ao_c->af, AF_CONTROL_SET_PAN_BALANCE, &val))
return;
if (val == 0)
return;
struct af_instance *af_pan_balance;
if (!(af_pan_balance = af_add(ao_c->af, "pan", "autopan", NULL))) {
MP_ERR(mpctx, "No balance control available.\n");
return;
}
/* make all other channels pass through since by default pan blocks all */
for (int i = 2; i < AF_NCH; i++) {
float level[AF_NCH] = {0};
level[i] = 1.f;
af_control_ext_t arg_ext = { .ch = i, .arg = level };
af_pan_balance->control(af_pan_balance, AF_CONTROL_SET_PAN_LEVEL,
&arg_ext);
}
af_pan_balance->control(af_pan_balance, AF_CONTROL_SET_PAN_BALANCE, &val);
}
static int recreate_audio_filters(struct MPContext *mpctx)
{
assert(mpctx->ao_chain);
struct af_stream *afs = mpctx->ao_chain->af;
if (afs->initialized < 1 && af_init(afs) < 0)
goto fail;
recreate_speed_filters(mpctx);
if (afs->initialized < 1 && af_init(afs) < 0)
goto fail;
if (mpctx->opts->softvol == SOFTVOL_NO)
MP_ERR(mpctx, "--softvol=no is not supported anymore.\n");
audio_update_volume(mpctx);
audio_update_balance(mpctx);
mp_notify(mpctx, MPV_EVENT_AUDIO_RECONFIG, NULL);
return 0;
fail:
MP_ERR(mpctx, "Couldn't find matching filter/ao format!\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 = 0;
if (ao_c->af->initialized > 0)
delay = af_calc_delay(ao_c->af);
af_uninit(ao_c->af);
if (recreate_audio_filters(mpctx) < 0)
return -1;
// 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 > 0.2)
issue_refresh_seek(mpctx, MPSEEK_EXACT);
return 1;
}
#else /* HAVE_LIBAV */
void audio_update_volume(struct MPContext *mpctx) {}
void audio_update_balance(struct MPContext *mpctx) {}
int reinit_audio_filters(struct MPContext *mpctx) { return 0; }
#endif /* else HAVE_LIBAF */
// 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 HAVE_LIBAF
if (!mpctx->ao_chain || mpctx->ao_chain->af->initialized < 1)
return;
if (!update_speed_filters(mpctx))
recreate_audio_filters(mpctx);
#endif
}
static void ao_chain_reset_state(struct ao_chain *ao_c)
{
ao_c->pts = MP_NOPTS_VALUE;
ao_c->pts_reset = false;
TA_FREEP(&ao_c->input_frame);
TA_FREEP(&ao_c->output_frame);
#if HAVE_LIBAF
af_seek_reset(ao_c->af);
#endif
if (ao_c->conv)
mp_aconverter_flush(ao_c->conv);
mp_audio_buffer_clear(ao_c->ao_buffer);
if (ao_c->audio_src)
audio_reset_decoding(ao_c->audio_src);
}
void reset_audio_state(struct MPContext *mpctx)
{
if (mpctx->ao_chain)
ao_chain_reset_state(mpctx->ao_chain);
mpctx->audio_status = mpctx->ao_chain ? STATUS_SYNCING : STATUS_EOF;
mpctx->delay = 0;
mpctx->audio_drop_throttle = 0;
mpctx->audio_stat_start = 0;
mpctx->audio_allow_second_chance_seek = false;
}
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;
talloc_free(mpctx->ao_decoder_fmt);
mpctx->ao_decoder_fmt = NULL;
}
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;
assert(track->d_audio == ao_c->audio_src);
track->d_audio = NULL;
audio_uninit(ao_c->audio_src);
}
if (ao_c->filter_src)
lavfi_set_connected(ao_c->filter_src, false);
#if HAVE_LIBAF
af_destroy(ao_c->af);
#endif
talloc_free(ao_c->conv);
talloc_free(ao_c->input_frame);
talloc_free(ao_c->input_format);
talloc_free(ao_c->output_frame);
talloc_free(ao_c->filter_input_format);
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;
}
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 (!mp_aframe_config_is_valid(ao_c->input_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.
mp_wakeup_core(mpctx);
return;
}
// Weak gapless audio: drain AO on decoder format changes
if (mpctx->ao_decoder_fmt && mpctx->ao && opts->gapless_audio < 0 &&
!mp_aframe_config_equals(mpctx->ao_decoder_fmt, ao_c->input_format))
{
uninit_audio_out(mpctx);
}
TA_FREEP(&ao_c->output_frame);
int out_rate = 0;
int out_format = 0;
struct mp_chmap out_channels = {0};
if (mpctx->ao) {
ao_get_format(mpctx->ao, &out_rate, &out_format, &out_channels);
} else if (af_fmt_is_pcm(mp_aframe_get_format(ao_c->input_format))) {
out_rate = opts->force_srate;
out_format = opts->audio_output_format;
if (opts->audio_output_channels.num_chmaps == 1)
out_channels = opts->audio_output_channels.chmaps[0];
}
#if HAVE_LIBAF
struct af_stream *afs = ao_c->af;
struct mp_audio in_format;
mp_audio_config_from_aframe(&in_format, ao_c->input_format);
if (mpctx->ao && mp_audio_config_equals(&in_format, &afs->input))
return;
afs->output = (struct mp_audio){0};
afs->output.rate = out_rate;
mp_audio_set_format(&afs->output, out_format);
mp_audio_set_channels(&afs->output, &out_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;
}
out_rate = afs->output.rate;
out_format = afs->output.format;
out_channels = afs->output.channels;
#else
if (mpctx->ao && ao_c->filter_input_format &&
mp_aframe_config_equals(ao_c->filter_input_format, ao_c->input_format))
return;
TA_FREEP(&ao_c->filter_input_format);
if (!out_rate)
out_rate = mp_aframe_get_rate(ao_c->input_format);
if (!out_format)
out_format = mp_aframe_get_format(ao_c->input_format);
if (!out_channels.num)
mp_aframe_get_chmap(ao_c->input_format, &out_channels);
#endif
if (!mpctx->ao) {
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 = 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->audio_src) {
MP_VERBOSE(mpctx, "Falling back to PCM output.\n");
ao_c->spdif_passthrough = false;
ao_c->spdif_failed = true;
ao_c->audio_src->try_spdif = false;
if (!audio_init_best_codec(ao_c->audio_src))
goto init_error;
reset_audio_state(mpctx);
mp_aframe_reset(ao_c->input_format);
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);
#if HAVE_LIBAF
afs->output = (struct mp_audio){0};
afs->output.rate = ao_rate;
mp_audio_set_format(&afs->output, ao_format);
mp_audio_set_channels(&afs->output, &ao_channels);
if (!mp_audio_config_equals(&afs->output, &afs->filter_output))
afs->initialized = 0;
#else
int in_rate = mp_aframe_get_rate(ao_c->input_format);
int in_format = mp_aframe_get_format(ao_c->input_format);
struct mp_chmap in_chmap = {0};
mp_aframe_get_chmap(ao_c->input_format, &in_chmap);
if (!mp_aconverter_reconfig(ao_c->conv, in_rate, in_format, in_chmap,
ao_rate, ao_format, ao_channels))
{
MP_ERR(mpctx, "Cannot convert audio data for output.\n");
goto init_error;
}
ao_c->filter_input_format = mp_aframe_new_ref(ao_c->input_format);
#endif
mpctx->ao_decoder_fmt = mp_aframe_new_ref(ao_c->input_format);
char tmp[80];
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;
}
#if HAVE_LIBAF
if (recreate_audio_filters(mpctx) < 0)
goto init_error;
#endif
update_playback_speed(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->d_audio);
if (!track->stream)
goto init_error;
track->d_audio = talloc_zero(NULL, struct dec_audio);
struct dec_audio *d_audio = track->d_audio;
d_audio->log = mp_log_new(d_audio, mpctx->log, "!ad");
d_audio->global = mpctx->global;
d_audio->opts = mpctx->opts;
d_audio->header = track->stream;
d_audio->codec = track->stream->codec;
d_audio->try_spdif = true;
if (!audio_init_best_codec(d_audio))
goto init_error;
return 1;
init_error:
if (track->sink)
lavfi_set_connected(track->sink, false);
track->sink = NULL;
audio_uninit(track->d_audio);
track->d_audio = 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;
#if HAVE_LIBAF
ao_c->af = af_new(mpctx->global);
if (track && track->stream)
ao_c->af->replaygain_data = track->stream->codec->replaygain_data;
#else
ao_c->conv = mp_aconverter_create(mpctx->global, mpctx->log, NULL);
#endif
ao_c->spdif_passthrough = true;
ao_c->pts = MP_NOPTS_VALUE;
ao_c->ao_buffer = mp_audio_buffer_create(NULL);
ao_c->ao = mpctx->ao;
ao_c->input_format = mp_aframe_create();
if (track) {
ao_c->track = track;
track->ao_c = ao_c;
if (!init_audio_decoder(mpctx, track))
goto init_error;
ao_c->audio_src = track->d_audio;
}
reset_audio_state(mpctx);
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);
}
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;
// first calculate the end pts of audio that has been output by decoder
double a_pts = ao_c->pts;
if (a_pts == MP_NOPTS_VALUE)
return MP_NOPTS_VALUE;
// Data buffered in audio filters, measured in seconds of "missing" output
double buffered_output = 0;
#if HAVE_LIBAF
if (ao_c->af->initialized < 1)
return MP_NOPTS_VALUE;
buffered_output += af_calc_delay(ao_c->af);
#endif
if (ao_c->conv)
buffered_output += mp_aconverter_get_latency(ao_c->conv);
if (ao_c->output_frame)
buffered_output += mp_aframe_duration(ao_c->output_frame);
// 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(ao_c->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;
}
// 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);
#if HAVE_ENCODING
encode_lavc_set_audio_pts(mpctx->encode_lavc_ctx, playing_audio_pts(mpctx));
#endif
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);
// Heuristic: if audio is "too far" ahead, and one of them is a separate
// track, allow a refresh seek to the correct position to fix it.
if (ptsdiff > 0.2 && mpctx->audio_allow_second_chance_seek && sync_to_video) {
struct ao_chain *ao_c = mpctx->ao_chain;
if (ao_c && ao_c->track && mpctx->vo_chain && mpctx->vo_chain->track &&
ao_c->track->demuxer != mpctx->vo_chain->track->demuxer)
{
struct track *track = ao_c->track;
double pts = mpctx->video_pts;
if (pts != MP_NOPTS_VALUE)
pts += get_track_seek_offset(mpctx, track);
// (disable it first to make it take any effect)
demuxer_select_track(track->demuxer, track->stream, pts, false);
demuxer_select_track(track->demuxer, track->stream, pts, true);
reset_audio_state(mpctx);
MP_VERBOSE(mpctx, "retrying audio seek\n");
return false;
}
}
mpctx->audio_allow_second_chance_seek = false;
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 eof, 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);
#if HAVE_LIBAF
struct af_stream *afs = mpctx->ao_chain->af;
if (af_output_frame(afs, eof) < 0)
return true; // error, stop doing stuff
struct mp_audio *mpa = af_read_output_frame(afs);
ao_c->output_frame = mp_audio_to_aframe(mpa);
talloc_free(mpa);
#else
if (eof)
mp_aconverter_write_input(ao_c->conv, NULL);
mp_aconverter_set_speed(ao_c->conv, mpctx->audio_speed);
bool got_eof;
ao_c->output_frame = mp_aconverter_read_output(ao_c->conv, &got_eof);
#endif
}
if (!ao_c->output_frame)
return false; // out of data
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;
}
static int decode_new_frame(struct ao_chain *ao_c)
{
if (ao_c->input_frame)
return AD_OK;
int res = DATA_EOF;
if (ao_c->filter_src) {
res = lavfi_request_frame_a(ao_c->filter_src, &ao_c->input_frame);
} else if (ao_c->audio_src) {
audio_work(ao_c->audio_src);
res = audio_get_frame(ao_c->audio_src, &ao_c->input_frame);
}
if (ao_c->input_frame)
mp_aframe_config_copy(ao_c->input_format, ao_c->input_frame);
switch (res) {
case DATA_OK: return AD_OK;
case DATA_WAIT: return AD_WAIT;
case DATA_AGAIN: return AD_NO_PROGRESS;
case DATA_STARVE: return AD_STARVE;
case DATA_EOF: return AD_EOF;
default: abort();
}
}
/* 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;
#if HAVE_LIBAF
struct af_stream *afs = ao_c->af;
if (afs->initialized < 1)
return AD_ERR;
#else
if (!ao_c->filter_input_format)
return AD_ERR;
#endif
MP_STATS(ao_c, "start audio");
double endpts = get_play_end_pts(mpctx);
bool eof = false;
int res;
while (1) {
res = 0;
if (copy_output(mpctx, ao_c, minsamples, endpts, false, &eof))
break;
res = decode_new_frame(ao_c);
if (res == AD_NO_PROGRESS)
continue;
if (res == AD_WAIT || res == AD_STARVE)
break;
if (res < 0) {
// drain filters first (especially for true EOF case)
copy_output(mpctx, ao_c, minsamples, endpts, true, &eof);
break;
}
// On format change, make sure to drain the filter chain.
#if HAVE_LIBAF
struct mp_audio in_format;
mp_audio_config_from_aframe(&in_format, ao_c->input_format);
if (!mp_audio_config_equals(&afs->input, &in_format)) {
copy_output(mpctx, ao_c, minsamples, endpts, true, &eof);
res = AD_NEW_FMT;
break;
}
#else
if (!mp_aframe_config_equals(ao_c->filter_input_format,
ao_c->input_format))
{
copy_output(mpctx, ao_c, minsamples, endpts, true, &eof);
res = AD_NEW_FMT;
break;
}
#endif
double pts = mp_aframe_get_pts(ao_c->input_frame);
if (pts == MP_NOPTS_VALUE) {
ao_c->pts = MP_NOPTS_VALUE;
} else {
// Attempt to detect jumps in PTS. Even for the lowest sample rates
// and with worst container rounded timestamp, this should be a
// margin more than enough.
double desync = pts - ao_c->pts;
if (ao_c->pts != MP_NOPTS_VALUE && fabs(desync) > 0.1) {
MP_WARN(ao_c, "Invalid audio PTS: %f -> %f\n",
ao_c->pts, pts);
if (desync >= 5)
ao_c->pts_reset = true;
}
ao_c->pts = mp_aframe_end_pts(ao_c->input_frame);
}
#if HAVE_LIBAF
struct mp_audio *mpa = mp_audio_from_aframe(ao_c->input_frame);
talloc_free(ao_c->input_frame);
ao_c->input_frame = NULL;
if (!mpa)
abort();
if (af_filter_frame(afs, mpa) < 0)
return AD_ERR;
#else
if (mp_aconverter_write_input(ao_c->conv, ao_c->input_frame))
ao_c->input_frame = NULL;
#endif
}
if (res == 0 && mp_audio_buffer_samples(outbuf) < minsamples && eof)
res = AD_EOF;
MP_STATS(ao_c, "end audio");
return res;
}
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
// 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).
struct ao_chain *ao_c = mpctx->ao_chain;
if (ao_c) {
struct dec_audio *d_audio = ao_c->audio_src;
if (d_audio && ao_c->spdif_failed) {
ao_c->spdif_passthrough = true;
ao_c->spdif_failed = false;
d_audio->try_spdif = true;
#if HAVE_LIBAF
ao_c->af->initialized = 0;
#endif
TA_FREEP(&ao_c->filter_input_format);
if (!audio_init_best_codec(d_audio)) {
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);
struct ao_chain *ao_c = mpctx->ao_chain;
if (!ao_c)
return;
bool is_initialized = !!ao_c->filter_input_format;
#if HAVE_LIBAF
is_initialized = ao_c->af->initialized == 1;
#endif
if (!is_initialized || !mpctx->ao) {
// Probe the initial audio format. Returns AD_OK (and does nothing) if
// the format is already known.
int r = AD_NO_PROGRESS;
while (r == AD_NO_PROGRESS)
r = decode_new_frame(mpctx->ao_chain);
if (r == AD_WAIT)
return; // continue later when new data is available
if (r == AD_EOF) {
mpctx->audio_status = STATUS_EOF;
return;
}
reinit_audio_filters_and_output(mpctx);
mp_wakeup_core(mpctx);
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->pts_reset) {
MP_VERBOSE(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);
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 (status == AD_WAIT)
return;
if (status == AD_NO_PROGRESS || status == AD_STARVE) {
mp_wakeup_core(mpctx);
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_filters_and_output(mpctx);
mp_wakeup_core(mpctx);
return; // retry on next iteration
}
if (status == AD_ERR)
mp_wakeup_core(mpctx);
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) {
if (mpctx->vo_chain && !mpctx->vo_chain->is_coverart &&
mpctx->video_status <= STATUS_READY)
return;
MP_VERBOSE(mpctx, "starting audio playback\n");
}
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;
// 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;
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);
}
}
}
}
// 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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