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mpv/player/audio.c

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/*
* This file is part of mpv.
*
player: change license of most core files to LGPL These files have all in common that they were fully or mostly taken from mplayer.c. (mplayer.c was a huge file that contains almost all of the playback core, until it was split into multiple parts.) This was probably the hardest part to relicense, because so much code was moved around all the time. player/audio.c still does not compile. We'll have to redo audio filtering. Once that is done, we can probably actually provide an actual LGPL configure switch. Here is a relatively detailed list of potential issues: 8d190244: author did not reply, parts were made GPL-only in a previous commit. 7882ea9b: author could not be reached, but the code is gone. wscript still has --datadir switch, but I don't think this is relevant to copyright. f197efd5: unclear origin, but I consider the code gone anyway (replaced with generic OSD mechanisms). 8337d9c2: author did not reply, but only the option still exists (under a different name), other code was removed. d8fd7131: did not reply. Disabled in a previous commit. 05258251: same author as above. Both fields actually seem to have vanished (even when tracking renames), so no action taken. d459e644, 268b2c1a: author did not reply, but we reuse only the options (with different names and slightly or fully different semantics, and completely different implementations), so I don't think this is relevant for copyright. 09e742fe, 17c39c4e: same as above. e8a173de, bff4b3ee: author could not be reached. The commands were reworked to properties, and the code outside of the TV code were moved back to the TV code. So I don't think copyright applies to the current command.c parts (mp_property_tv_color, mp_property_tv_freq, mp_property_tv_scan). The TV parts remain GPL. 0810e427: could not be reached. Disabled in a previous commit. 43744a2d: unknown author, but this was replaced by dynamic alloc (if the change is even copyrightable). 116ca0c7: unknown author; reasoning see input.c relicensing commit. e7e4d1d8: these semantics still exist, but as generic code, and this code was fully removed. f1175cd9: the author of the cited patch is unknown, and upon inspection it turns out that I was only using the idea to pause the player on EOF, so I claim it's not copyright relevant. 25affdcc: author could not be reached (yet) - but it's only a function rename, not copyrightable. 5728504c was committed by Arpi (who agreed), but hints that it might be by a different author. In fact it seems to be mostly this patch: http://lists.mplayerhq.hu/pipermail/mplayer-dev-eng/2001-November/002041.html The author did not respond, but it all seems to have been removed later. It's a terrible mess though. Arpi reverted the A-V sync code at first, but left the RTC code for a while. The following commits remove these changes 100%: 14b35442, 7181a091, 31482783, 614f8475, df58e822. cehoyos did explicitly not agree to LGPL, but was involved in the following changes: c99d8fc8: applied a patch and didn't modify it, the original author agreed. 40ac0d31: author could not be reached, but all code is gone anyway. The "af" command has a similar function, but works completely different and actually reuses a mechanism older than this patch. 54350436: applied a patch, but didn't modify it, except for adding a German translation, which was removed later. a2dda036: same situation as above 240b743e: this was made GPL-only in a previous commit 7b25afd7: same as above (for now) kirijua could not be reached, but was a regular patch contributor: c2c997fd: video equalizer code move; probably not copyrightable. Is GPL due to Nick anyway. be54f481: technically, this became the audio track property later. But all what is left is the fact that you pass a track ID to it, so consider the original coypright non-relevant. 2f376d1b: this was rewritten in b7052b43, but for now we can afford to be careful, so this was marked as GPL only in a previous commit. 43844d09: remaining parts in main.c were reverted in a previous commit. anders has mostly disagreed with the LGPL relicensing. Does not want libaf to become LGPL, but made some concessions. In particular, he granted us permission to relicense 4943e9c52c and 242aa6ebd4. We also consider some of his changes remaining in mpv not relevant for copyright (such as 735de602 - we won't remove the this option completely). We will completely remove his other contributions, including the entire audio filter chain. For now, this stuff is marked as GPL only. The remaining question is how much code in player/audio.c (based on the former mplayer.c and dec_audio.c) is under his copyright. I made claims about this in a previous commit. Nick(ols) Kurshev, svn username "nick" and "nickols_k", could not be reached. He had a lot of changes in early MPlayer. It seems all of that was removed, at least in mpv. His main work, like VIDIX or libswscale work, does not exist in mpv anymore, but the changes to mplayer.c and other core parts still deserve attention: a4119f6b, fb927549, ad3529b8, e11b23dc, 5f2178be, 93c371d5: removed in b43d67e0, d1628d12, 24ed01fe, df58e822. 0a83c6ec, 104c125e, 4e067f62, aec5dcc8, b587a3d6, f3de6e6b: DR, VAA, and "tune" stuff was fully removed later on or replaced with other mechanisms. 340183b0: screenshots were redone later (the VOCTRL was even removed, with an independent implementation using the same VOCTRL a few years later), so not relevant anymore. Basically only the 's' shortcut remains (but not its implementation). 92c5c274, bffd4007, 555c6766: for now marked as GPL only in a previous commit. Might contain some trace amounts of "michael"'s copyright, who agreed to LGPL only once the core is relicensed. This will still be respected, but I don't think it matters at this in this case. (Some code touched by him was merged into mplayer.c, and then disappeared after heavy refactoring.) I tried to be as careful and as complete as possible. It can't be excluded that amends to this will be made later. This does not make the player LGPL yet.
2017-06-23 13:53:41 +00:00
* 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
player: change license of most core files to LGPL These files have all in common that they were fully or mostly taken from mplayer.c. (mplayer.c was a huge file that contains almost all of the playback core, until it was split into multiple parts.) This was probably the hardest part to relicense, because so much code was moved around all the time. player/audio.c still does not compile. We'll have to redo audio filtering. Once that is done, we can probably actually provide an actual LGPL configure switch. Here is a relatively detailed list of potential issues: 8d190244: author did not reply, parts were made GPL-only in a previous commit. 7882ea9b: author could not be reached, but the code is gone. wscript still has --datadir switch, but I don't think this is relevant to copyright. f197efd5: unclear origin, but I consider the code gone anyway (replaced with generic OSD mechanisms). 8337d9c2: author did not reply, but only the option still exists (under a different name), other code was removed. d8fd7131: did not reply. Disabled in a previous commit. 05258251: same author as above. Both fields actually seem to have vanished (even when tracking renames), so no action taken. d459e644, 268b2c1a: author did not reply, but we reuse only the options (with different names and slightly or fully different semantics, and completely different implementations), so I don't think this is relevant for copyright. 09e742fe, 17c39c4e: same as above. e8a173de, bff4b3ee: author could not be reached. The commands were reworked to properties, and the code outside of the TV code were moved back to the TV code. So I don't think copyright applies to the current command.c parts (mp_property_tv_color, mp_property_tv_freq, mp_property_tv_scan). The TV parts remain GPL. 0810e427: could not be reached. Disabled in a previous commit. 43744a2d: unknown author, but this was replaced by dynamic alloc (if the change is even copyrightable). 116ca0c7: unknown author; reasoning see input.c relicensing commit. e7e4d1d8: these semantics still exist, but as generic code, and this code was fully removed. f1175cd9: the author of the cited patch is unknown, and upon inspection it turns out that I was only using the idea to pause the player on EOF, so I claim it's not copyright relevant. 25affdcc: author could not be reached (yet) - but it's only a function rename, not copyrightable. 5728504c was committed by Arpi (who agreed), but hints that it might be by a different author. In fact it seems to be mostly this patch: http://lists.mplayerhq.hu/pipermail/mplayer-dev-eng/2001-November/002041.html The author did not respond, but it all seems to have been removed later. It's a terrible mess though. Arpi reverted the A-V sync code at first, but left the RTC code for a while. The following commits remove these changes 100%: 14b35442, 7181a091, 31482783, 614f8475, df58e822. cehoyos did explicitly not agree to LGPL, but was involved in the following changes: c99d8fc8: applied a patch and didn't modify it, the original author agreed. 40ac0d31: author could not be reached, but all code is gone anyway. The "af" command has a similar function, but works completely different and actually reuses a mechanism older than this patch. 54350436: applied a patch, but didn't modify it, except for adding a German translation, which was removed later. a2dda036: same situation as above 240b743e: this was made GPL-only in a previous commit 7b25afd7: same as above (for now) kirijua could not be reached, but was a regular patch contributor: c2c997fd: video equalizer code move; probably not copyrightable. Is GPL due to Nick anyway. be54f481: technically, this became the audio track property later. But all what is left is the fact that you pass a track ID to it, so consider the original coypright non-relevant. 2f376d1b: this was rewritten in b7052b43, but for now we can afford to be careful, so this was marked as GPL only in a previous commit. 43844d09: remaining parts in main.c were reverted in a previous commit. anders has mostly disagreed with the LGPL relicensing. Does not want libaf to become LGPL, but made some concessions. In particular, he granted us permission to relicense 4943e9c52c and 242aa6ebd4. We also consider some of his changes remaining in mpv not relevant for copyright (such as 735de602 - we won't remove the this option completely). We will completely remove his other contributions, including the entire audio filter chain. For now, this stuff is marked as GPL only. The remaining question is how much code in player/audio.c (based on the former mplayer.c and dec_audio.c) is under his copyright. I made claims about this in a previous commit. Nick(ols) Kurshev, svn username "nick" and "nickols_k", could not be reached. He had a lot of changes in early MPlayer. It seems all of that was removed, at least in mpv. His main work, like VIDIX or libswscale work, does not exist in mpv anymore, but the changes to mplayer.c and other core parts still deserve attention: a4119f6b, fb927549, ad3529b8, e11b23dc, 5f2178be, 93c371d5: removed in b43d67e0, d1628d12, 24ed01fe, df58e822. 0a83c6ec, 104c125e, 4e067f62, aec5dcc8, b587a3d6, f3de6e6b: DR, VAA, and "tune" stuff was fully removed later on or replaced with other mechanisms. 340183b0: screenshots were redone later (the VOCTRL was even removed, with an independent implementation using the same VOCTRL a few years later), so not relevant anymore. Basically only the 's' shortcut remains (but not its implementation). 92c5c274, bffd4007, 555c6766: for now marked as GPL only in a previous commit. Might contain some trace amounts of "michael"'s copyright, who agreed to LGPL only once the core is relicensed. This will still be respected, but I don't think it matters at this in this case. (Some code touched by him was merged into mplayer.c, and then disappeared after heavy refactoring.) I tried to be as careful and as complete as possible. It can't be excluded that amends to this will be made later. This does not make the player LGPL yet.
2017-06-23 13:53:41 +00:00
* GNU Lesser General Public License for more details.
*
player: change license of most core files to LGPL These files have all in common that they were fully or mostly taken from mplayer.c. (mplayer.c was a huge file that contains almost all of the playback core, until it was split into multiple parts.) This was probably the hardest part to relicense, because so much code was moved around all the time. player/audio.c still does not compile. We'll have to redo audio filtering. Once that is done, we can probably actually provide an actual LGPL configure switch. Here is a relatively detailed list of potential issues: 8d190244: author did not reply, parts were made GPL-only in a previous commit. 7882ea9b: author could not be reached, but the code is gone. wscript still has --datadir switch, but I don't think this is relevant to copyright. f197efd5: unclear origin, but I consider the code gone anyway (replaced with generic OSD mechanisms). 8337d9c2: author did not reply, but only the option still exists (under a different name), other code was removed. d8fd7131: did not reply. Disabled in a previous commit. 05258251: same author as above. Both fields actually seem to have vanished (even when tracking renames), so no action taken. d459e644, 268b2c1a: author did not reply, but we reuse only the options (with different names and slightly or fully different semantics, and completely different implementations), so I don't think this is relevant for copyright. 09e742fe, 17c39c4e: same as above. e8a173de, bff4b3ee: author could not be reached. The commands were reworked to properties, and the code outside of the TV code were moved back to the TV code. So I don't think copyright applies to the current command.c parts (mp_property_tv_color, mp_property_tv_freq, mp_property_tv_scan). The TV parts remain GPL. 0810e427: could not be reached. Disabled in a previous commit. 43744a2d: unknown author, but this was replaced by dynamic alloc (if the change is even copyrightable). 116ca0c7: unknown author; reasoning see input.c relicensing commit. e7e4d1d8: these semantics still exist, but as generic code, and this code was fully removed. f1175cd9: the author of the cited patch is unknown, and upon inspection it turns out that I was only using the idea to pause the player on EOF, so I claim it's not copyright relevant. 25affdcc: author could not be reached (yet) - but it's only a function rename, not copyrightable. 5728504c was committed by Arpi (who agreed), but hints that it might be by a different author. In fact it seems to be mostly this patch: http://lists.mplayerhq.hu/pipermail/mplayer-dev-eng/2001-November/002041.html The author did not respond, but it all seems to have been removed later. It's a terrible mess though. Arpi reverted the A-V sync code at first, but left the RTC code for a while. The following commits remove these changes 100%: 14b35442, 7181a091, 31482783, 614f8475, df58e822. cehoyos did explicitly not agree to LGPL, but was involved in the following changes: c99d8fc8: applied a patch and didn't modify it, the original author agreed. 40ac0d31: author could not be reached, but all code is gone anyway. The "af" command has a similar function, but works completely different and actually reuses a mechanism older than this patch. 54350436: applied a patch, but didn't modify it, except for adding a German translation, which was removed later. a2dda036: same situation as above 240b743e: this was made GPL-only in a previous commit 7b25afd7: same as above (for now) kirijua could not be reached, but was a regular patch contributor: c2c997fd: video equalizer code move; probably not copyrightable. Is GPL due to Nick anyway. be54f481: technically, this became the audio track property later. But all what is left is the fact that you pass a track ID to it, so consider the original coypright non-relevant. 2f376d1b: this was rewritten in b7052b43, but for now we can afford to be careful, so this was marked as GPL only in a previous commit. 43844d09: remaining parts in main.c were reverted in a previous commit. anders has mostly disagreed with the LGPL relicensing. Does not want libaf to become LGPL, but made some concessions. In particular, he granted us permission to relicense 4943e9c52c and 242aa6ebd4. We also consider some of his changes remaining in mpv not relevant for copyright (such as 735de602 - we won't remove the this option completely). We will completely remove his other contributions, including the entire audio filter chain. For now, this stuff is marked as GPL only. The remaining question is how much code in player/audio.c (based on the former mplayer.c and dec_audio.c) is under his copyright. I made claims about this in a previous commit. Nick(ols) Kurshev, svn username "nick" and "nickols_k", could not be reached. He had a lot of changes in early MPlayer. It seems all of that was removed, at least in mpv. His main work, like VIDIX or libswscale work, does not exist in mpv anymore, but the changes to mplayer.c and other core parts still deserve attention: a4119f6b, fb927549, ad3529b8, e11b23dc, 5f2178be, 93c371d5: removed in b43d67e0, d1628d12, 24ed01fe, df58e822. 0a83c6ec, 104c125e, 4e067f62, aec5dcc8, b587a3d6, f3de6e6b: DR, VAA, and "tune" stuff was fully removed later on or replaced with other mechanisms. 340183b0: screenshots were redone later (the VOCTRL was even removed, with an independent implementation using the same VOCTRL a few years later), so not relevant anymore. Basically only the 's' shortcut remains (but not its implementation). 92c5c274, bffd4007, 555c6766: for now marked as GPL only in a previous commit. Might contain some trace amounts of "michael"'s copyright, who agreed to LGPL only once the core is relicensed. This will still be respected, but I don't think it matters at this in this case. (Some code touched by him was merged into mplayer.c, and then disappeared after heavy refactoring.) I tried to be as careful and as complete as possible. It can't be excluded that amends to this will be made later. This does not make the player LGPL yet.
2017-06-23 13:53:41 +00:00
* 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_GPL are licensed under GNU General Public License.
*/
#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;
player: gross hack to improve non-hr seeking with external audio tracks Relative seeks backwards with external audio tracks does not always work well: it tends to happen that video seek back further than audio, so audio will remain silent until the audio's after-seek position is reached. This happens because we strictly seek both video and audio demuxer to the approximate desirted target PTS, and then start decoding from that. Commit 81358380 removes an older method that was supposed to deal with this. It was sort of bad, because it could lead to the playback core freezing by waiting on network. Ideally, the demuxer layer would probably somehow deal with such seeks, and do them in a way the audio is seeked after video. Currently this is infeasible, because the demuxer layer assumes a single demuxer, and external tracks simply use separate demuxer layers. (MPlayer actually had a pseudo-demuxer that joined external tracks into a single demuxer, but this is not flexible enough - and also, the demuxer layer as it currently exists can't deal with dynamically removing external tracks either. Maybe some time in the future.) Instead, add a gross hack, that essentially reseeks the audio if it detects that it's too far off. The result is actually not too bad, because we can reuse the mechanism that is used for instant track switching. This way we can make sure of the right position, without having to care about certain other issues. It should be noted that if the audio demuxer is used for other tracks too, and the demuxer does not support refresh seeking, audio will probably be off by even a higher amount. But this should be rare.
2016-08-07 14:29:13 +00:00
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);
audio: introduce a new type to hold audio frames This is pretty pointless, but I believe it allows us to claim that the new code is not affected by the copyright of the old code. This is needed, because the original mp_audio struct was written by someone who has disagreed with LGPL relicensing (it was called af_data at the time, and was defined in af.h). The "GPL'ed" struct contents that surive are pretty trivial: just the data pointer, and some metadata like the format, samplerate, etc. - but at least in this case, any new code would be extremely similar anyway, and I'm not really sure whether it's OK to claim different copyright. So what we do is we just use AVFrame (which of course is LGPL with 100% certainty), and add some accessors around it to adapt it to mpv conventions. Also, this gets rid of some annoying conventions of mp_audio, like the struct fields that require using an accessor to write to them anyway. For the most part, this change is only dumb replacements of mp_audio related functions and fields. One minor actual change is that you can't allocate the new type on the stack anymore. Some code still uses mp_audio. All audio filter code will be deleted, so it makes no sense to convert this code. (Audio filters which are LGPL and which we keep will have to be ported to a new filter infrastructure anyway.) player/audio.c uses it because it interacts with the old filter code. push.c has some complex use of mp_audio and mp_audio_buffer, but this and pull.c will most likely be rewritten to do something else.
2017-08-16 19:00:20 +00:00
talloc_free(ao_c->input_format);
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)
audio: introduce a new type to hold audio frames This is pretty pointless, but I believe it allows us to claim that the new code is not affected by the copyright of the old code. This is needed, because the original mp_audio struct was written by someone who has disagreed with LGPL relicensing (it was called af_data at the time, and was defined in af.h). The "GPL'ed" struct contents that surive are pretty trivial: just the data pointer, and some metadata like the format, samplerate, etc. - but at least in this case, any new code would be extremely similar anyway, and I'm not really sure whether it's OK to claim different copyright. So what we do is we just use AVFrame (which of course is LGPL with 100% certainty), and add some accessors around it to adapt it to mpv conventions. Also, this gets rid of some annoying conventions of mp_audio, like the struct fields that require using an accessor to write to them anyway. For the most part, this change is only dumb replacements of mp_audio related functions and fields. One minor actual change is that you can't allocate the new type on the stack anymore. Some code still uses mp_audio. All audio filter code will be deleted, so it makes no sense to convert this code. (Audio filters which are LGPL and which we keep will have to be ported to a new filter infrastructure anyway.) player/audio.c uses it because it interacts with the old filter code. push.c has some complex use of mp_audio and mp_audio_buffer, but this and pull.c will most likely be rewritten to do something else.
2017-08-16 19:00:20 +00:00
{
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;
audio: introduce a new type to hold audio frames This is pretty pointless, but I believe it allows us to claim that the new code is not affected by the copyright of the old code. This is needed, because the original mp_audio struct was written by someone who has disagreed with LGPL relicensing (it was called af_data at the time, and was defined in af.h). The "GPL'ed" struct contents that surive are pretty trivial: just the data pointer, and some metadata like the format, samplerate, etc. - but at least in this case, any new code would be extremely similar anyway, and I'm not really sure whether it's OK to claim different copyright. So what we do is we just use AVFrame (which of course is LGPL with 100% certainty), and add some accessors around it to adapt it to mpv conventions. Also, this gets rid of some annoying conventions of mp_audio, like the struct fields that require using an accessor to write to them anyway. For the most part, this change is only dumb replacements of mp_audio related functions and fields. One minor actual change is that you can't allocate the new type on the stack anymore. Some code still uses mp_audio. All audio filter code will be deleted, so it makes no sense to convert this code. (Audio filters which are LGPL and which we keep will have to be ported to a new filter infrastructure anyway.) player/audio.c uses it because it interacts with the old filter code. push.c has some complex use of mp_audio and mp_audio_buffer, but this and pull.c will most likely be rewritten to do something else.
2017-08-16 19:00:20 +00:00
}
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;
audio: introduce a new type to hold audio frames This is pretty pointless, but I believe it allows us to claim that the new code is not affected by the copyright of the old code. This is needed, because the original mp_audio struct was written by someone who has disagreed with LGPL relicensing (it was called af_data at the time, and was defined in af.h). The "GPL'ed" struct contents that surive are pretty trivial: just the data pointer, and some metadata like the format, samplerate, etc. - but at least in this case, any new code would be extremely similar anyway, and I'm not really sure whether it's OK to claim different copyright. So what we do is we just use AVFrame (which of course is LGPL with 100% certainty), and add some accessors around it to adapt it to mpv conventions. Also, this gets rid of some annoying conventions of mp_audio, like the struct fields that require using an accessor to write to them anyway. For the most part, this change is only dumb replacements of mp_audio related functions and fields. One minor actual change is that you can't allocate the new type on the stack anymore. Some code still uses mp_audio. All audio filter code will be deleted, so it makes no sense to convert this code. (Audio filters which are LGPL and which we keep will have to be ported to a new filter infrastructure anyway.) player/audio.c uses it because it interacts with the old filter code. push.c has some complex use of mp_audio and mp_audio_buffer, but this and pull.c will most likely be rewritten to do something else.
2017-08-16 19:00:20 +00:00
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 &&
audio: introduce a new type to hold audio frames This is pretty pointless, but I believe it allows us to claim that the new code is not affected by the copyright of the old code. This is needed, because the original mp_audio struct was written by someone who has disagreed with LGPL relicensing (it was called af_data at the time, and was defined in af.h). The "GPL'ed" struct contents that surive are pretty trivial: just the data pointer, and some metadata like the format, samplerate, etc. - but at least in this case, any new code would be extremely similar anyway, and I'm not really sure whether it's OK to claim different copyright. So what we do is we just use AVFrame (which of course is LGPL with 100% certainty), and add some accessors around it to adapt it to mpv conventions. Also, this gets rid of some annoying conventions of mp_audio, like the struct fields that require using an accessor to write to them anyway. For the most part, this change is only dumb replacements of mp_audio related functions and fields. One minor actual change is that you can't allocate the new type on the stack anymore. Some code still uses mp_audio. All audio filter code will be deleted, so it makes no sense to convert this code. (Audio filters which are LGPL and which we keep will have to be ported to a new filter infrastructure anyway.) player/audio.c uses it because it interacts with the old filter code. push.c has some complex use of mp_audio and mp_audio_buffer, but this and pull.c will most likely be rewritten to do something else.
2017-08-16 19:00:20 +00:00
!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;
audio: introduce a new type to hold audio frames This is pretty pointless, but I believe it allows us to claim that the new code is not affected by the copyright of the old code. This is needed, because the original mp_audio struct was written by someone who has disagreed with LGPL relicensing (it was called af_data at the time, and was defined in af.h). The "GPL'ed" struct contents that surive are pretty trivial: just the data pointer, and some metadata like the format, samplerate, etc. - but at least in this case, any new code would be extremely similar anyway, and I'm not really sure whether it's OK to claim different copyright. So what we do is we just use AVFrame (which of course is LGPL with 100% certainty), and add some accessors around it to adapt it to mpv conventions. Also, this gets rid of some annoying conventions of mp_audio, like the struct fields that require using an accessor to write to them anyway. For the most part, this change is only dumb replacements of mp_audio related functions and fields. One minor actual change is that you can't allocate the new type on the stack anymore. Some code still uses mp_audio. All audio filter code will be deleted, so it makes no sense to convert this code. (Audio filters which are LGPL and which we keep will have to be ported to a new filter infrastructure anyway.) player/audio.c uses it because it interacts with the old filter code. push.c has some complex use of mp_audio and mp_audio_buffer, but this and pull.c will most likely be rewritten to do something else.
2017-08-16 19:00:20 +00:00
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);
audio: introduce a new type to hold audio frames This is pretty pointless, but I believe it allows us to claim that the new code is not affected by the copyright of the old code. This is needed, because the original mp_audio struct was written by someone who has disagreed with LGPL relicensing (it was called af_data at the time, and was defined in af.h). The "GPL'ed" struct contents that surive are pretty trivial: just the data pointer, and some metadata like the format, samplerate, etc. - but at least in this case, any new code would be extremely similar anyway, and I'm not really sure whether it's OK to claim different copyright. So what we do is we just use AVFrame (which of course is LGPL with 100% certainty), and add some accessors around it to adapt it to mpv conventions. Also, this gets rid of some annoying conventions of mp_audio, like the struct fields that require using an accessor to write to them anyway. For the most part, this change is only dumb replacements of mp_audio related functions and fields. One minor actual change is that you can't allocate the new type on the stack anymore. Some code still uses mp_audio. All audio filter code will be deleted, so it makes no sense to convert this code. (Audio filters which are LGPL and which we keep will have to be ported to a new filter infrastructure anyway.) player/audio.c uses it because it interacts with the old filter code. push.c has some complex use of mp_audio and mp_audio_buffer, but this and pull.c will most likely be rewritten to do something else.
2017-08-16 19:00:20 +00:00
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;
audio: introduce a new type to hold audio frames This is pretty pointless, but I believe it allows us to claim that the new code is not affected by the copyright of the old code. This is needed, because the original mp_audio struct was written by someone who has disagreed with LGPL relicensing (it was called af_data at the time, and was defined in af.h). The "GPL'ed" struct contents that surive are pretty trivial: just the data pointer, and some metadata like the format, samplerate, etc. - but at least in this case, any new code would be extremely similar anyway, and I'm not really sure whether it's OK to claim different copyright. So what we do is we just use AVFrame (which of course is LGPL with 100% certainty), and add some accessors around it to adapt it to mpv conventions. Also, this gets rid of some annoying conventions of mp_audio, like the struct fields that require using an accessor to write to them anyway. For the most part, this change is only dumb replacements of mp_audio related functions and fields. One minor actual change is that you can't allocate the new type on the stack anymore. Some code still uses mp_audio. All audio filter code will be deleted, so it makes no sense to convert this code. (Audio filters which are LGPL and which we keep will have to be ported to a new filter infrastructure anyway.) player/audio.c uses it because it interacts with the old filter code. push.c has some complex use of mp_audio and mp_audio_buffer, but this and pull.c will most likely be rewritten to do something else.
2017-08-16 19:00:20 +00:00
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);
player: gross hack to improve non-hr seeking with external audio tracks Relative seeks backwards with external audio tracks does not always work well: it tends to happen that video seek back further than audio, so audio will remain silent until the audio's after-seek position is reached. This happens because we strictly seek both video and audio demuxer to the approximate desirted target PTS, and then start decoding from that. Commit 81358380 removes an older method that was supposed to deal with this. It was sort of bad, because it could lead to the playback core freezing by waiting on network. Ideally, the demuxer layer would probably somehow deal with such seeks, and do them in a way the audio is seeked after video. Currently this is infeasible, because the demuxer layer assumes a single demuxer, and external tracks simply use separate demuxer layers. (MPlayer actually had a pseudo-demuxer that joined external tracks into a single demuxer, but this is not flexible enough - and also, the demuxer layer as it currently exists can't deal with dynamically removing external tracks either. Maybe some time in the future.) Instead, add a gross hack, that essentially reseeks the audio if it detects that it's too far off. The result is actually not too bad, because we can reuse the mechanism that is used for instant track switching. This way we can make sure of the right position, without having to care about certain other issues. It should be noted that if the audio demuxer is used for other tracks too, and the demuxer does not support refresh seeking, audio will probably be off by even a higher amount. But this should be rare.
2016-08-07 14:29:13 +00:00
// 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
audio: introduce a new type to hold audio frames This is pretty pointless, but I believe it allows us to claim that the new code is not affected by the copyright of the old code. This is needed, because the original mp_audio struct was written by someone who has disagreed with LGPL relicensing (it was called af_data at the time, and was defined in af.h). The "GPL'ed" struct contents that surive are pretty trivial: just the data pointer, and some metadata like the format, samplerate, etc. - but at least in this case, any new code would be extremely similar anyway, and I'm not really sure whether it's OK to claim different copyright. So what we do is we just use AVFrame (which of course is LGPL with 100% certainty), and add some accessors around it to adapt it to mpv conventions. Also, this gets rid of some annoying conventions of mp_audio, like the struct fields that require using an accessor to write to them anyway. For the most part, this change is only dumb replacements of mp_audio related functions and fields. One minor actual change is that you can't allocate the new type on the stack anymore. Some code still uses mp_audio. All audio filter code will be deleted, so it makes no sense to convert this code. (Audio filters which are LGPL and which we keep will have to be ported to a new filter infrastructure anyway.) player/audio.c uses it because it interacts with the old filter code. push.c has some complex use of mp_audio and mp_audio_buffer, but this and pull.c will most likely be rewritten to do something else.
2017-08-16 19:00:20 +00:00
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;
2017-03-14 14:55:42 +00:00
if (!was_eof) {
MP_VERBOSE(mpctx, "audio EOF reached\n");
mp_wakeup_core(mpctx);
2017-03-14 14:55:42 +00:00
}
}
}
}
// 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);
}