mpv/player/loadfile.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/>.
*/
#include <stddef.h>
#include <stdbool.h>
#include <strings.h>
#include <inttypes.h>
#include <assert.h>
#include <libavutil/avutil.h>
#include "config.h"
#include "mpv_talloc.h"
#include "misc/thread_pool.h"
#include "misc/thread_tools.h"
#include "osdep/io.h"
#include "osdep/terminal.h"
#include "osdep/threads.h"
#include "osdep/timer.h"
scripting: change when/how player waits for scripts being loaded Fundamentally, scripts are loaded asynchronously, but as a feature, there was code to wait until a script is loaded (for a certain arbitrary definition of "loaded"). This was done in scripting.c with the wait_loaded() function. This called mp_idle(), and since there are commands to load/unload scripts, it meant the player core loop could be entered recursively. I think this is a major complication and has some problems. For example, if you had a script that does 'os.execute("sleep inf")', then every time you ran a command to load an instance of the script would add a new stack frame of mp_idle(). This would lead to some sort of reentrancy horror that is hard to debug. Also misc/dispatch.c contains a somewhat tricky mess to support such recursive invocations. There were also some bugs due to this and due to unforeseen interactions with other messes. This scripting stuff was the only thing making use of that reentrancy, and future commands that have "logical" waiting for something should be implemented differently. So get rid of it. Change the code to wait only in the player initialization phase: the only place where it really has to wait is before playback is started, because scripts might want to set options or hooks that interact with playback initialization. Unloading of builtin scripts (can happen with e.g. "set osc no") is left asynchronous; the unloading wasn't too robust anyway, and this change won't make a difference if someone is trying to break it intentionally. Note that this is not in mp_initialize(), because mpv_initialize() uses this by locking the core, which would have the same problem. In the future, commands which logically wait should use different mechanisms. Originally I thought the current approach (that is removed with this commit) should be used, but it's too much of a mess and can't even be used in some cases. Examples are: - "loadfile" should be made blocking (needs to run the normal player code and manually unblock the thread issuing the command) - "add-sub" should not freeze the player until the URL is opened (needs to run opening on a separate thread) Possibly the current scripting behavior could be restored once new mechanisms exist, and if it turns out that anyone needs it. With this commit there should be no further instances of recursive playloop invocations (other than the case in the following commit), since all mp_idle()/mp_wait_events() calls are done strictly from the main thread (and not commands/properties or libmpv client API that "lock" the main thread).
2018-04-15 08:14:00 +00:00
#include "client.h"
#include "common/msg.h"
#include "common/global.h"
#include "options/path.h"
#include "options/m_config.h"
#include "options/parse_configfile.h"
#include "common/playlist.h"
#include "options/options.h"
#include "options/m_property.h"
#include "common/common.h"
#include "common/encode.h"
#include "common/recorder.h"
2013-12-17 00:23:09 +00:00
#include "input/input.h"
#include "audio/out/ao.h"
#include "filters/f_decoder_wrapper.h"
#include "filters/f_lavfi.h"
#include "filters/filter_internal.h"
#include "demux/demux.h"
#include "stream/stream.h"
#include "sub/dec_sub.h"
#include "external_files.h"
#include "video/out/vo.h"
#include "core.h"
#include "command.h"
#include "libmpv/client.h"
// Called from the demuxer thread if a new packet is available, or other changes.
static void wakeup_demux(void *pctx)
{
struct MPContext *mpctx = pctx;
mp_wakeup_core(mpctx);
}
// Called by foreign threads when playback should be stopped and such.
void mp_abort_playback_async(struct MPContext *mpctx)
{
mp_cancel_trigger(mpctx->playback_abort);
pthread_mutex_lock(&mpctx->abort_lock);
for (int n = 0; n < mpctx->num_abort_list; n++) {
struct mp_abort_entry *abort = mpctx->abort_list[n];
if (abort->coupled_to_playback)
mp_abort_trigger_locked(mpctx, abort);
}
pthread_mutex_unlock(&mpctx->abort_lock);
}
// Add it to the global list, and allocate required data structures.
void mp_abort_add(struct MPContext *mpctx, struct mp_abort_entry *abort)
{
pthread_mutex_lock(&mpctx->abort_lock);
assert(!abort->cancel);
abort->cancel = mp_cancel_new(NULL);
MP_TARRAY_APPEND(NULL, mpctx->abort_list, mpctx->num_abort_list, abort);
mp_abort_recheck_locked(mpctx, abort);
pthread_mutex_unlock(&mpctx->abort_lock);
}
// Remove Add it to the global list, and free/clear required data structures.
// Does not deallocate the abort value itself.
void mp_abort_remove(struct MPContext *mpctx, struct mp_abort_entry *abort)
{
pthread_mutex_lock(&mpctx->abort_lock);
for (int n = 0; n < mpctx->num_abort_list; n++) {
if (mpctx->abort_list[n] == abort) {
MP_TARRAY_REMOVE_AT(mpctx->abort_list, mpctx->num_abort_list, n);
TA_FREEP(&abort->cancel);
abort = NULL; // it's not free'd, just clear for the assert below
break;
}
}
assert(!abort); // should have been in the list
pthread_mutex_unlock(&mpctx->abort_lock);
}
// Verify whether the abort needs to be signaled after changing certain fields
// in abort.
void mp_abort_recheck_locked(struct MPContext *mpctx,
struct mp_abort_entry *abort)
{
if ((abort->coupled_to_playback && mp_cancel_test(mpctx->playback_abort)) ||
mpctx->abort_all)
{
mp_abort_trigger_locked(mpctx, abort);
}
}
void mp_abort_trigger_locked(struct MPContext *mpctx,
struct mp_abort_entry *abort)
{
mp_cancel_trigger(abort->cancel);
}
player: make playback termination asynchronous Until now, stopping playback aborted the demuxer and I/O layer violently by signaling mp_cancel (bound to libavformat's AVIOInterruptCB mechanism). Change it to try closing them gracefully. The main purpose is to silence those libavformat errors that happen when you request termination. Most of libavformat barely cares about the termination mechanism (AVIOInterruptCB), and essentially it's like the network connection is abruptly severed, or file I/O suddenly returns I/O errors. There were issues with dumb TLS warnings, parsers complaining about incomplete data, and some special protocols that require server communication to gracefully disconnect. We still want to abort it forcefully if it refuses to terminate on its own, so a timeout is required. Users can set the timeout to 0, which should give them the old behavior. This also removes the old mechanism that treats certain commands (like "quit") specially, and tries to terminate the demuxers even if the core is currently frozen. This is for situations where the core synchronized to the demuxer or stream layer while network is unresponsive. This in turn can only happen due to the "program" or "cache-size" properties in the current code (see one of the previous commits). Also, the old mechanism doesn't fit particularly well with the new one. We wouldn't want to abort playback immediately on a "quit" command - the new code is all about giving it a chance to end it gracefully. We'd need some sort of watchdog thread or something equally complicated to handle this. So just remove it. The change in osd.c is to prevent that it clears the status line while waiting for termination. The normal status line code doesn't output anything useful at this point, and the code path taken clears it, both of which is an annoying behavior change, so just let it show the old one.
2018-05-19 16:41:13 +00:00
static void kill_demuxers_reentrant(struct MPContext *mpctx,
struct demuxer **demuxers, int num_demuxers)
{
struct demux_free_async_state **items = NULL;
int num_items = 0;
for (int n = 0; n < num_demuxers; n++) {
struct demuxer *d = demuxers[n];
if (!demux_cancel_test(d)) {
// Make sure it is set if it wasn't yet.
demux_set_wakeup_cb(d, wakeup_demux, mpctx);
struct demux_free_async_state *item = demux_free_async(d);
if (item) {
MP_TARRAY_APPEND(NULL, items, num_items, item);
d = NULL;
}
}
demux_cancel_and_free(d);
}
if (!num_items)
return;
MP_DBG(mpctx, "Terminating demuxers...\n");
double end = mp_time_sec() + mpctx->opts->demux_termination_timeout;
bool force = false;
while (num_items) {
double wait = end - mp_time_sec();
for (int n = 0; n < num_items; n++) {
struct demux_free_async_state *item = items[n];
if (demux_free_async_finish(item)) {
items[n] = items[num_items - 1];
num_items -= 1;
n--;
goto repeat;
} else if (wait < 0) {
demux_free_async_force(item);
if (!force)
MP_VERBOSE(mpctx, "Forcefully terminating demuxers...\n");
force = true;
}
}
if (wait >= 0)
mp_set_timeout(mpctx, wait);
mp_idle(mpctx);
repeat:;
}
talloc_free(items);
MP_DBG(mpctx, "Done terminating demuxers.\n");
}
static void uninit_demuxer(struct MPContext *mpctx)
{
for (int r = 0; r < NUM_PTRACKS; r++) {
for (int t = 0; t < STREAM_TYPE_COUNT; t++)
mpctx->current_track[r][t] = NULL;
}
player: make playback termination asynchronous Until now, stopping playback aborted the demuxer and I/O layer violently by signaling mp_cancel (bound to libavformat's AVIOInterruptCB mechanism). Change it to try closing them gracefully. The main purpose is to silence those libavformat errors that happen when you request termination. Most of libavformat barely cares about the termination mechanism (AVIOInterruptCB), and essentially it's like the network connection is abruptly severed, or file I/O suddenly returns I/O errors. There were issues with dumb TLS warnings, parsers complaining about incomplete data, and some special protocols that require server communication to gracefully disconnect. We still want to abort it forcefully if it refuses to terminate on its own, so a timeout is required. Users can set the timeout to 0, which should give them the old behavior. This also removes the old mechanism that treats certain commands (like "quit") specially, and tries to terminate the demuxers even if the core is currently frozen. This is for situations where the core synchronized to the demuxer or stream layer while network is unresponsive. This in turn can only happen due to the "program" or "cache-size" properties in the current code (see one of the previous commits). Also, the old mechanism doesn't fit particularly well with the new one. We wouldn't want to abort playback immediately on a "quit" command - the new code is all about giving it a chance to end it gracefully. We'd need some sort of watchdog thread or something equally complicated to handle this. So just remove it. The change in osd.c is to prevent that it clears the status line while waiting for termination. The normal status line code doesn't output anything useful at this point, and the code path taken clears it, both of which is an annoying behavior change, so just let it show the old one.
2018-05-19 16:41:13 +00:00
mpctx->seek_slave = NULL;
talloc_free(mpctx->chapters);
mpctx->chapters = NULL;
mpctx->num_chapters = 0;
mp_abort_cache_dumping(mpctx);
player: make playback termination asynchronous Until now, stopping playback aborted the demuxer and I/O layer violently by signaling mp_cancel (bound to libavformat's AVIOInterruptCB mechanism). Change it to try closing them gracefully. The main purpose is to silence those libavformat errors that happen when you request termination. Most of libavformat barely cares about the termination mechanism (AVIOInterruptCB), and essentially it's like the network connection is abruptly severed, or file I/O suddenly returns I/O errors. There were issues with dumb TLS warnings, parsers complaining about incomplete data, and some special protocols that require server communication to gracefully disconnect. We still want to abort it forcefully if it refuses to terminate on its own, so a timeout is required. Users can set the timeout to 0, which should give them the old behavior. This also removes the old mechanism that treats certain commands (like "quit") specially, and tries to terminate the demuxers even if the core is currently frozen. This is for situations where the core synchronized to the demuxer or stream layer while network is unresponsive. This in turn can only happen due to the "program" or "cache-size" properties in the current code (see one of the previous commits). Also, the old mechanism doesn't fit particularly well with the new one. We wouldn't want to abort playback immediately on a "quit" command - the new code is all about giving it a chance to end it gracefully. We'd need some sort of watchdog thread or something equally complicated to handle this. So just remove it. The change in osd.c is to prevent that it clears the status line while waiting for termination. The normal status line code doesn't output anything useful at this point, and the code path taken clears it, both of which is an annoying behavior change, so just let it show the old one.
2018-05-19 16:41:13 +00:00
struct demuxer **demuxers = NULL;
int num_demuxers = 0;
if (mpctx->demuxer)
MP_TARRAY_APPEND(NULL, demuxers, num_demuxers, mpctx->demuxer);
mpctx->demuxer = NULL;
for (int i = 0; i < mpctx->num_tracks; i++) {
player: make playback termination asynchronous Until now, stopping playback aborted the demuxer and I/O layer violently by signaling mp_cancel (bound to libavformat's AVIOInterruptCB mechanism). Change it to try closing them gracefully. The main purpose is to silence those libavformat errors that happen when you request termination. Most of libavformat barely cares about the termination mechanism (AVIOInterruptCB), and essentially it's like the network connection is abruptly severed, or file I/O suddenly returns I/O errors. There were issues with dumb TLS warnings, parsers complaining about incomplete data, and some special protocols that require server communication to gracefully disconnect. We still want to abort it forcefully if it refuses to terminate on its own, so a timeout is required. Users can set the timeout to 0, which should give them the old behavior. This also removes the old mechanism that treats certain commands (like "quit") specially, and tries to terminate the demuxers even if the core is currently frozen. This is for situations where the core synchronized to the demuxer or stream layer while network is unresponsive. This in turn can only happen due to the "program" or "cache-size" properties in the current code (see one of the previous commits). Also, the old mechanism doesn't fit particularly well with the new one. We wouldn't want to abort playback immediately on a "quit" command - the new code is all about giving it a chance to end it gracefully. We'd need some sort of watchdog thread or something equally complicated to handle this. So just remove it. The change in osd.c is to prevent that it clears the status line while waiting for termination. The normal status line code doesn't output anything useful at this point, and the code path taken clears it, both of which is an annoying behavior change, so just let it show the old one.
2018-05-19 16:41:13 +00:00
struct track *track = mpctx->tracks[i];
assert(!track->dec && !track->d_sub);
player: make playback termination asynchronous Until now, stopping playback aborted the demuxer and I/O layer violently by signaling mp_cancel (bound to libavformat's AVIOInterruptCB mechanism). Change it to try closing them gracefully. The main purpose is to silence those libavformat errors that happen when you request termination. Most of libavformat barely cares about the termination mechanism (AVIOInterruptCB), and essentially it's like the network connection is abruptly severed, or file I/O suddenly returns I/O errors. There were issues with dumb TLS warnings, parsers complaining about incomplete data, and some special protocols that require server communication to gracefully disconnect. We still want to abort it forcefully if it refuses to terminate on its own, so a timeout is required. Users can set the timeout to 0, which should give them the old behavior. This also removes the old mechanism that treats certain commands (like "quit") specially, and tries to terminate the demuxers even if the core is currently frozen. This is for situations where the core synchronized to the demuxer or stream layer while network is unresponsive. This in turn can only happen due to the "program" or "cache-size" properties in the current code (see one of the previous commits). Also, the old mechanism doesn't fit particularly well with the new one. We wouldn't want to abort playback immediately on a "quit" command - the new code is all about giving it a chance to end it gracefully. We'd need some sort of watchdog thread or something equally complicated to handle this. So just remove it. The change in osd.c is to prevent that it clears the status line while waiting for termination. The normal status line code doesn't output anything useful at this point, and the code path taken clears it, both of which is an annoying behavior change, so just let it show the old one.
2018-05-19 16:41:13 +00:00
assert(!track->vo_c && !track->ao_c);
assert(!track->sink);
assert(!track->remux_sink);
// Demuxers can be added in any order (if they appear mid-stream), and
// we can't know which tracks uses which, so here's some O(n^2) trash.
for (int n = 0; n < num_demuxers; n++) {
if (demuxers[n] == track->demuxer) {
track->demuxer = NULL;
break;
}
}
if (track->demuxer)
MP_TARRAY_APPEND(NULL, demuxers, num_demuxers, track->demuxer);
talloc_free(track);
}
mpctx->num_tracks = 0;
player: make playback termination asynchronous Until now, stopping playback aborted the demuxer and I/O layer violently by signaling mp_cancel (bound to libavformat's AVIOInterruptCB mechanism). Change it to try closing them gracefully. The main purpose is to silence those libavformat errors that happen when you request termination. Most of libavformat barely cares about the termination mechanism (AVIOInterruptCB), and essentially it's like the network connection is abruptly severed, or file I/O suddenly returns I/O errors. There were issues with dumb TLS warnings, parsers complaining about incomplete data, and some special protocols that require server communication to gracefully disconnect. We still want to abort it forcefully if it refuses to terminate on its own, so a timeout is required. Users can set the timeout to 0, which should give them the old behavior. This also removes the old mechanism that treats certain commands (like "quit") specially, and tries to terminate the demuxers even if the core is currently frozen. This is for situations where the core synchronized to the demuxer or stream layer while network is unresponsive. This in turn can only happen due to the "program" or "cache-size" properties in the current code (see one of the previous commits). Also, the old mechanism doesn't fit particularly well with the new one. We wouldn't want to abort playback immediately on a "quit" command - the new code is all about giving it a chance to end it gracefully. We'd need some sort of watchdog thread or something equally complicated to handle this. So just remove it. The change in osd.c is to prevent that it clears the status line while waiting for termination. The normal status line code doesn't output anything useful at this point, and the code path taken clears it, both of which is an annoying behavior change, so just let it show the old one.
2018-05-19 16:41:13 +00:00
kill_demuxers_reentrant(mpctx, demuxers, num_demuxers);
talloc_free(demuxers);
}
#define APPEND(s, ...) mp_snprintf_cat(s, sizeof(s), __VA_ARGS__)
static void print_stream(struct MPContext *mpctx, struct track *t)
{
struct sh_stream *s = t->stream;
const char *tname = "?";
const char *selopt = "?";
const char *langopt = "?";
switch (t->type) {
case STREAM_VIDEO:
tname = "Video"; selopt = "vid"; langopt = NULL;
break;
case STREAM_AUDIO:
tname = "Audio"; selopt = "aid"; langopt = "alang";
break;
case STREAM_SUB:
tname = "Subs"; selopt = "sid"; langopt = "slang";
break;
}
char b[2048] = {0};
APPEND(b, " %3s %-5s", t->selected ? "(+)" : "", tname);
APPEND(b, " --%s=%d", selopt, t->user_tid);
if (t->lang && langopt)
APPEND(b, " --%s=%s", langopt, t->lang);
if (t->default_track)
APPEND(b, " (*)");
if (t->forced_track)
APPEND(b, " (f)");
if (t->attached_picture)
APPEND(b, " [P]");
if (t->title)
APPEND(b, " '%s'", t->title);
const char *codec = s ? s->codec->codec : NULL;
APPEND(b, " (%s", codec ? codec : "<unknown>");
if (t->type == STREAM_VIDEO) {
if (s && s->codec->disp_w)
APPEND(b, " %dx%d", s->codec->disp_w, s->codec->disp_h);
if (s && s->codec->fps)
APPEND(b, " %.3ffps", s->codec->fps);
} else if (t->type == STREAM_AUDIO) {
if (s && s->codec->channels.num)
APPEND(b, " %dch", s->codec->channels.num);
if (s && s->codec->samplerate)
APPEND(b, " %dHz", s->codec->samplerate);
}
APPEND(b, ")");
if (t->is_external)
APPEND(b, " (external)");
MP_INFO(mpctx, "%s\n", b);
}
void print_track_list(struct MPContext *mpctx, const char *msg)
{
if (msg)
MP_INFO(mpctx, "%s\n", msg);
for (int t = 0; t < STREAM_TYPE_COUNT; t++) {
for (int n = 0; n < mpctx->num_tracks; n++)
if (mpctx->tracks[n]->type == t)
print_stream(mpctx, mpctx->tracks[n]);
}
}
void update_demuxer_properties(struct MPContext *mpctx)
{
struct demuxer *demuxer = mpctx->demuxer;
if (!demuxer)
return;
demux: redo timed metadata The old implementation didn't work for the OGG case. Discard the old shit code (instead of fixing it), and write new shit code. The old code was already over a year old, so it's about time to rewrite it for no reason anyway. While it's true that the old code appears to be broken, the main reason to rewrite this is to make it simpler. While the amount of code seems to be about the same, both the concept and the actual tag handling are simpler. The result is probably a bit more correct. The packet struct shrinks by 8 byte. That fact that it wasted 8 bytes per packet for a rather obscure use case was the reason I started this at all (and when I found that OGG updates didn't work). While these 8 bytes aren't going to hurt, the packet struct was getting too bloated. If you buffer a lot of data, these extra fields will add up. Still quite some effort for 8 bytes. Fortunately, it's not like there are any managers that need to be convinced whether it's worth doing. The freedom to waste time on dumb shit. The old implementation attached the current metadata to each packet. When the decoder read the packet, the packet's metadata was made current. The new implementation stores metadata as separate list, and requires that the player frontend tells it the current playback time, which will be used to find the currently valid metadata. In both cases, the objective was to correctly update metadata even if a lot of data is buffered ahead (and to update them correctly when seeking within the demuxer cache). The new implementation is actually slightly more correct, because it uses the playback time for the metadata lookup. Consider if you have an audio filter which buffers 15 seconds (unfortunately such a filter exists), then the old code would update the current title 15 seconds too early, while the new one does it correctly. The new code also simplifies mixing the 3 metadata sources (global, per stream, ICY). We assume these aren't mixed in a meaningful way. The old code tried to be a bit more "exact". I didn't bother to look how the old code did this, but the new code simply always "merges" with the previous metadata, so if a newer tag removes a field, it's going to stick around anyway. I tried to keep it simple. Other approaches include making metadata a special sh_stream with metadata packets. This would have been conceptually clean, but the implementation would probably have been unnatural (and doesn't match well with libavformat's API anyway). It would have been nice to make the metadata updates chapter points (makes a lot of sense for the intended use case, web radio current song information), but I don't think it would have been a good idea to make chapters suddenly so dynamic. (Still an idea to keep in mind; the new code actually makes it easier to work towards this.) You could mention how subtitles are timed metadata, and actually are implemented as sparse packet streams in some formats. mp4 implements chapters as special subtitle stream, AFAIK. (Ironically, this is very not-ideal for files. It would be useful for streaming like web radio, but mp4 is extremely bad for streaming by design for other reasons.) bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla
2019-06-10 00:18:20 +00:00
demux_update(demuxer, get_current_time(mpctx));
int events = demuxer->events;
if ((events & DEMUX_EVENT_INIT) && demuxer->num_editions > 1) {
for (int n = 0; n < demuxer->num_editions; n++) {
struct demux_edition *edition = &demuxer->editions[n];
char b[128] = {0};
APPEND(b, " %3s --edition=%d",
n == demuxer->edition ? "(+)" : "", n);
char *name = mp_tags_get_str(edition->metadata, "title");
if (name)
APPEND(b, " '%s'", name);
if (edition->default_edition)
APPEND(b, " (*)");
MP_INFO(mpctx, "%s\n", b);
}
}
struct demuxer *tracks = mpctx->demuxer;
if (tracks->events & DEMUX_EVENT_STREAMS) {
add_demuxer_tracks(mpctx, tracks);
print_track_list(mpctx, NULL);
tracks->events &= ~DEMUX_EVENT_STREAMS;
}
if (events & DEMUX_EVENT_METADATA) {
struct mp_tags *info =
mp_tags_filtered(mpctx, demuxer->metadata, mpctx->opts->display_tags);
// prev is used to attempt to print changed tags only (to some degree)
struct mp_tags *prev = mpctx->filtered_tags;
int n_prev = 0;
bool had_output = false;
for (int n = 0; n < info->num_keys; n++) {
if (prev && n_prev < prev->num_keys) {
if (strcmp(prev->keys[n_prev], info->keys[n]) == 0) {
n_prev++;
if (strcmp(prev->values[n_prev - 1], info->values[n]) == 0)
continue;
}
}
struct mp_log *log = mp_log_new(NULL, mpctx->log, "!display-tags");
if (!had_output)
mp_info(log, "File tags:\n");
mp_info(log, " %s: %s\n", info->keys[n], info->values[n]);
had_output = true;
talloc_free(log);
}
talloc_free(mpctx->filtered_tags);
mpctx->filtered_tags = info;
mp_notify(mpctx, MPV_EVENT_METADATA_UPDATE, NULL);
}
if (events & DEMUX_EVENT_DURATION)
mp_notify(mpctx, MP_EVENT_DURATION_UPDATE, NULL);
demuxer->events = 0;
}
// Enables or disables the stream for the given track, according to
// track->selected.
void reselect_demux_stream(struct MPContext *mpctx, struct track *track)
{
if (!track->stream)
return;
double pts = get_current_time(mpctx);
if (pts != MP_NOPTS_VALUE) {
pts += get_track_seek_offset(mpctx, track);
if (track->type == STREAM_SUB)
pts -= 10.0;
}
demuxer_select_track(track->demuxer, track->stream, pts, track->selected);
if (track == mpctx->seek_slave)
mpctx->seek_slave = NULL;
}
static void enable_demux_thread(struct MPContext *mpctx, struct demuxer *demux)
{
if (mpctx->opts->demuxer_thread && !demux->fully_read) {
demux_set_wakeup_cb(demux, wakeup_demux, mpctx);
demux_start_thread(demux);
}
}
static int find_new_tid(struct MPContext *mpctx, enum stream_type t)
{
int new_id = 0;
for (int i = 0; i < mpctx->num_tracks; i++) {
struct track *track = mpctx->tracks[i];
if (track->type == t)
new_id = MPMAX(new_id, track->user_tid);
}
return new_id + 1;
}
static struct track *add_stream_track(struct MPContext *mpctx,
struct demuxer *demuxer,
struct sh_stream *stream)
{
for (int i = 0; i < mpctx->num_tracks; i++) {
struct track *track = mpctx->tracks[i];
if (track->stream == stream)
return track;
}
struct track *track = talloc_ptrtype(NULL, track);
*track = (struct track) {
.type = stream->type,
.user_tid = find_new_tid(mpctx, stream->type),
.demuxer_id = stream->demuxer_id,
.ff_index = stream->ff_index,
.title = stream->title,
.default_track = stream->default_track,
.forced_track = stream->forced_track,
.dependent_track = stream->dependent_track,
.visual_impaired_track = stream->visual_impaired_track,
.hearing_impaired_track = stream->hearing_impaired_track,
.attached_picture = stream->attached_picture != NULL,
.lang = stream->lang,
.demuxer = demuxer,
.stream = stream,
};
MP_TARRAY_APPEND(mpctx, mpctx->tracks, mpctx->num_tracks, track);
mp_notify(mpctx, MPV_EVENT_TRACKS_CHANGED, NULL);
return track;
}
void add_demuxer_tracks(struct MPContext *mpctx, struct demuxer *demuxer)
{
for (int n = 0; n < demux_get_num_stream(demuxer); n++)
add_stream_track(mpctx, demuxer, demux_get_stream(demuxer, n));
}
// Result numerically higher => better match. 0 == no match.
static int match_lang(char **langs, char *lang)
{
for (int idx = 0; langs && langs[idx]; idx++) {
if (lang && strcasecmp(langs[idx], lang) == 0)
return INT_MAX - idx;
}
return 0;
}
/* Get the track wanted by the user.
* tid is the track ID requested by the user (-2: deselect, -1: default)
* lang is a string list, NULL is same as empty list
* Sort tracks based on the following criteria, and pick the first:
* 0a) track matches ff-index (always wins)
* 0b) track matches tid (almost always wins)
* 0c) track is not from --external-file
* 1) track is external (no_default cancels this)
* 1b) track was passed explicitly (is not an auto-loaded subtitle)
* 2) earlier match in lang list
* 3a) track is marked forced
* 3b) track is marked default
* 4) attached picture, HLS bitrate
* 5) lower track number
* If select_fallback is not set, 5) is only used to determine whether a
* matching track is preferred over another track. Otherwise, always pick a
* track (if nothing else matches, return the track with lowest ID).
*/
// Return whether t1 is preferred over t2
static bool compare_track(struct track *t1, struct track *t2, char **langs,
struct MPOpts *opts)
{
if (!opts->autoload_files && t1->is_external != t2->is_external)
return !t1->is_external;
bool ext1 = t1->is_external && !t1->no_default;
bool ext2 = t2->is_external && !t2->no_default;
if (ext1 != ext2)
return ext1;
if (t1->auto_loaded != t2->auto_loaded)
return !t1->auto_loaded;
int l1 = match_lang(langs, t1->lang), l2 = match_lang(langs, t2->lang);
if (l1 != l2)
return l1 > l2;
if (t1->forced_track != t2->forced_track)
return t1->forced_track;
if (t1->default_track != t2->default_track)
return t1->default_track;
if (t1->attached_picture != t2->attached_picture)
return !t1->attached_picture;
if (t1->stream && t2->stream && opts->hls_bitrate >= 0 &&
t1->stream->hls_bitrate != t2->stream->hls_bitrate)
{
bool t1_ok = t1->stream->hls_bitrate <= opts->hls_bitrate;
bool t2_ok = t2->stream->hls_bitrate <= opts->hls_bitrate;
if (t1_ok != t2_ok)
return t1_ok;
if (t1_ok && t2_ok)
return t1->stream->hls_bitrate > t2->stream->hls_bitrate;
return t1->stream->hls_bitrate < t2->stream->hls_bitrate;
}
return t1->user_tid <= t2->user_tid;
}
static bool duplicate_track(struct MPContext *mpctx, int order,
enum stream_type type, struct track *track)
{
for (int i = 0; i < order; i++) {
if (mpctx->current_track[i][type] == track)
return true;
}
return false;
}
struct track *select_default_track(struct MPContext *mpctx, int order,
enum stream_type type)
{
struct MPOpts *opts = mpctx->opts;
int tid = opts->stream_id[order][type];
char **langs = opts->stream_lang[type];
if (tid == -2)
return NULL;
bool select_fallback = type == STREAM_VIDEO || type == STREAM_AUDIO;
struct track *pick = NULL;
for (int n = 0; n < mpctx->num_tracks; n++) {
struct track *track = mpctx->tracks[n];
if (track->type != type)
continue;
if (track->user_tid == tid)
return track;
if (track->no_auto_select)
continue;
if (duplicate_track(mpctx, order, type, track))
continue;
if (!pick || compare_track(track, pick, langs, mpctx->opts))
pick = track;
}
if (pick && !select_fallback && !(pick->is_external && !pick->no_default)
&& !match_lang(langs, pick->lang) && !pick->default_track
&& !pick->forced_track)
pick = NULL;
if (pick && pick->attached_picture && !mpctx->opts->audio_display)
pick = NULL;
if (pick && !opts->autoload_files && pick->is_external)
pick = NULL;
return pick;
}
static char *track_layout_hash(struct MPContext *mpctx)
{
char *h = talloc_strdup(NULL, "");
for (int type = 0; type < STREAM_TYPE_COUNT; type++) {
for (int n = 0; n < mpctx->num_tracks; n++) {
struct track *track = mpctx->tracks[n];
if (track->type != type)
continue;
h = talloc_asprintf_append_buffer(h, "%d-%d-%d-%d-%s\n", type,
track->user_tid, track->default_track, track->is_external,
track->lang ? track->lang : "");
}
}
return h;
}
// Normally, video/audio/sub track selection is persistent across files. This
// code resets track selection if the new file has a different track layout.
static void check_previous_track_selection(struct MPContext *mpctx)
{
struct MPOpts *opts = mpctx->opts;
if (!mpctx->track_layout_hash)
return;
char *h = track_layout_hash(mpctx);
if (strcmp(h, mpctx->track_layout_hash) != 0) {
// Reset selection, but only if they're not "auto" or "off". The
// defaults are -1 (default selection), or -2 (off) for secondary tracks.
for (int t = 0; t < STREAM_TYPE_COUNT; t++) {
for (int i = 0; i < NUM_PTRACKS; i++) {
if (opts->stream_id[i][t] >= 0)
opts->stream_id[i][t] = i == 0 ? -1 : -2;
}
}
talloc_free(mpctx->track_layout_hash);
mpctx->track_layout_hash = NULL;
}
talloc_free(h);
}
void mp_switch_track_n(struct MPContext *mpctx, int order, enum stream_type type,
struct track *track, int flags)
{
assert(!track || track->type == type);
assert(order >= 0 && order < NUM_PTRACKS);
// Mark the current track selection as explicitly user-requested. (This is
// different from auto-selection or disabling a track due to errors.)
if (flags & FLAG_MARK_SELECTION)
mpctx->opts->stream_id[order][type] = track ? track->user_tid : -2;
// No decoder should be initialized yet.
if (!mpctx->demuxer)
return;
struct track *current = mpctx->current_track[order][type];
if (track == current)
return;
if (current && current->sink) {
MP_ERR(mpctx, "Can't disable input to complex filter.\n");
return;
}
if ((type == STREAM_VIDEO && mpctx->vo_chain && !mpctx->vo_chain->track) ||
(type == STREAM_AUDIO && mpctx->ao_chain && !mpctx->ao_chain->track))
{
MP_ERR(mpctx, "Can't switch away from complex filter output.\n");
return;
}
if (track && track->selected) {
// Track has been selected in a different order parameter.
MP_ERR(mpctx, "Track %d is already selected.\n", track->user_tid);
return;
}
if (order == 0) {
if (type == STREAM_VIDEO) {
uninit_video_chain(mpctx);
if (!track)
handle_force_window(mpctx, true);
} else if (type == STREAM_AUDIO) {
clear_audio_output_buffers(mpctx);
uninit_audio_chain(mpctx);
uninit_audio_out(mpctx);
}
}
if (type == STREAM_SUB)
uninit_sub(mpctx, current);
if (current) {
if (current->remux_sink)
close_recorder_and_error(mpctx);
current->selected = false;
reselect_demux_stream(mpctx, current);
}
mpctx->current_track[order][type] = track;
if (track) {
track->selected = true;
reselect_demux_stream(mpctx, track);
}
2014-03-03 22:58:19 +00:00
if (type == STREAM_VIDEO && order == 0) {
reinit_video_chain(mpctx);
} else if (type == STREAM_AUDIO && order == 0) {
reinit_audio_chain(mpctx);
} else if (type == STREAM_SUB && order >= 0 && order <= 2) {
reinit_sub(mpctx, track);
}
mp_notify(mpctx, MPV_EVENT_TRACK_SWITCHED, NULL);
mp_wakeup_core(mpctx);
talloc_free(mpctx->track_layout_hash);
mpctx->track_layout_hash = talloc_steal(mpctx, track_layout_hash(mpctx));
}
void mp_switch_track(struct MPContext *mpctx, enum stream_type type,
struct track *track, int flags)
{
mp_switch_track_n(mpctx, 0, type, track, flags);
}
void mp_deselect_track(struct MPContext *mpctx, struct track *track)
{
if (track && track->selected) {
for (int t = 0; t < NUM_PTRACKS; t++)
mp_switch_track_n(mpctx, t, track->type, NULL, 0);
}
}
struct track *mp_track_by_tid(struct MPContext *mpctx, enum stream_type type,
int tid)
{
if (tid == -1)
return mpctx->current_track[0][type];
for (int n = 0; n < mpctx->num_tracks; n++) {
struct track *track = mpctx->tracks[n];
if (track->type == type && track->user_tid == tid)
return track;
}
return NULL;
}
bool mp_remove_track(struct MPContext *mpctx, struct track *track)
{
if (!track->is_external)
return false;
mp_deselect_track(mpctx, track);
if (track->selected)
return false;
struct demuxer *d = track->demuxer;
if (mpctx->seek_slave == track)
mpctx->seek_slave = NULL;
int index = 0;
while (index < mpctx->num_tracks && mpctx->tracks[index] != track)
index++;
MP_TARRAY_REMOVE_AT(mpctx->tracks, mpctx->num_tracks, index);
talloc_free(track);
// Close the demuxer, unless there is still a track using it. These are
// all external tracks.
bool in_use = false;
for (int n = mpctx->num_tracks - 1; n >= 0 && !in_use; n--)
in_use |= mpctx->tracks[n]->demuxer == d;
if (!in_use)
demux_cancel_and_free(d);
mp_notify(mpctx, MPV_EVENT_TRACKS_CHANGED, NULL);
return true;
}
// Add the given file as additional track. The filter argument controls how or
// if tracks are auto-selected at any point.
// To be run on a worker thread, locked (temporarily unlocks core).
// cancel will generally be used to abort the loading process, but on success
// the demuxer is changed to be slaved to mpctx->playback_abort instead.
int mp_add_external_file(struct MPContext *mpctx, char *filename,
enum stream_type filter, struct mp_cancel *cancel)
{
struct MPOpts *opts = mpctx->opts;
if (!filename || mp_cancel_test(cancel))
return -1;
char *disp_filename = filename;
if (strncmp(disp_filename, "memory://", 9) == 0)
disp_filename = "memory://"; // avoid noise
struct demuxer_params params = {
.is_top_level = true,
};
switch (filter) {
case STREAM_SUB:
params.force_format = opts->sub_demuxer_name;
break;
case STREAM_AUDIO:
params.force_format = opts->audio_demuxer_name;
break;
}
mp_core_unlock(mpctx);
struct demuxer *demuxer =
demux_open_url(filename, &params, cancel, mpctx->global);
if (demuxer)
enable_demux_thread(mpctx, demuxer);
mp_core_lock(mpctx);
// The command could have overlapped with playback exiting. (We don't care
// if playback has started again meanwhile - weird, but not a problem.)
if (mpctx->stop_play)
goto err_out;
if (!demuxer)
goto err_out;
if (filter != STREAM_SUB && opts->rebase_start_time)
demux_set_ts_offset(demuxer, -demuxer->start_time);
bool has_any = false;
for (int n = 0; n < demux_get_num_stream(demuxer); n++) {
struct sh_stream *sh = demux_get_stream(demuxer, n);
if (sh->type == filter || filter == STREAM_TYPE_COUNT) {
has_any = true;
break;
}
}
if (!has_any) {
char *tname = mp_tprintf(20, "%s ", stream_type_name(filter));
if (filter == STREAM_TYPE_COUNT)
tname = "";
MP_ERR(mpctx, "No %sstreams in file %s.\n", tname, disp_filename);
goto err_out;
}
int first_num = -1;
for (int n = 0; n < demux_get_num_stream(demuxer); n++) {
struct sh_stream *sh = demux_get_stream(demuxer, n);
struct track *t = add_stream_track(mpctx, demuxer, sh);
t->is_external = true;
if (sh->title && sh->title[0]) {
t->title = talloc_strdup(t, sh->title);
} else {
t->title = talloc_strdup(t, mp_basename(disp_filename));
}
t->external_filename = talloc_strdup(t, filename);
t->no_default = sh->type != filter;
t->no_auto_select = t->no_default;
if (first_num < 0 && (filter == STREAM_TYPE_COUNT || sh->type == filter))
first_num = mpctx->num_tracks - 1;
}
mp_cancel_set_parent(demuxer->cancel, mpctx->playback_abort);
return first_num;
err_out:
demux_cancel_and_free(demuxer);
if (!mp_cancel_test(cancel))
MP_ERR(mpctx, "Can not open external file %s.\n", disp_filename);
return -1;
}
// to be run on a worker thread, locked (temporarily unlocks core)
static void open_external_files(struct MPContext *mpctx, char **files,
enum stream_type filter)
{
// Need a copy, because the option value could be mutated during iteration.
void *tmp = talloc_new(NULL);
files = mp_dup_str_array(tmp, files);
for (int n = 0; files && files[n]; n++)
mp_add_external_file(mpctx, files[n], filter, mpctx->playback_abort);
talloc_free(tmp);
}
// See mp_add_external_file() for meaning of cancel parameter.
void autoload_external_files(struct MPContext *mpctx, struct mp_cancel *cancel)
{
if (mpctx->opts->sub_auto < 0 && mpctx->opts->audiofile_auto < 0)
return;
if (!mpctx->opts->autoload_files || strcmp(mpctx->filename, "-") == 0)
return;
void *tmp = talloc_new(NULL);
struct subfn *list = find_external_files(mpctx->global, mpctx->filename,
mpctx->opts);
talloc_steal(tmp, list);
int sc[STREAM_TYPE_COUNT] = {0};
for (int n = 0; n < mpctx->num_tracks; n++) {
if (!mpctx->tracks[n]->attached_picture)
sc[mpctx->tracks[n]->type]++;
}
for (int i = 0; list && list[i].fname; i++) {
char *filename = list[i].fname;
char *lang = list[i].lang;
for (int n = 0; n < mpctx->num_tracks; n++) {
struct track *t = mpctx->tracks[n];
if (t->demuxer && strcmp(t->demuxer->filename, filename) == 0)
goto skip;
}
if (list[i].type == STREAM_SUB && !sc[STREAM_VIDEO] && !sc[STREAM_AUDIO])
goto skip;
if (list[i].type == STREAM_AUDIO && !sc[STREAM_VIDEO])
goto skip;
int first = mp_add_external_file(mpctx, filename, list[i].type, cancel);
if (first < 0)
goto skip;
for (int n = first; n < mpctx->num_tracks; n++) {
struct track *t = mpctx->tracks[n];
t->auto_loaded = true;
if (!t->lang)
t->lang = talloc_strdup(t, lang);
}
skip:;
}
talloc_free(tmp);
}
// Do stuff to a newly loaded playlist. This includes any processing that may
// be required after loading a playlist.
void prepare_playlist(struct MPContext *mpctx, struct playlist *pl)
{
struct MPOpts *opts = mpctx->opts;
pl->current = NULL;
if (opts->playlist_pos >= 0)
pl->current = playlist_entry_from_index(pl, opts->playlist_pos);
if (opts->shuffle)
playlist_shuffle(pl);
if (opts->merge_files)
merge_playlist_files(pl);
if (!pl->current)
pl->current = mp_check_playlist_resume(mpctx, pl);
if (!pl->current)
pl->current = pl->first;
}
// Replace the current playlist entry with playlist contents. Moves the entries
// from the given playlist pl, so the entries don't actually need to be copied.
static void transfer_playlist(struct MPContext *mpctx, struct playlist *pl)
{
if (pl->first) {
prepare_playlist(mpctx, pl);
struct playlist_entry *new = pl->current;
if (mpctx->playlist->current)
playlist_add_redirect(pl, mpctx->playlist->current->filename);
playlist_transfer_entries(mpctx->playlist, pl);
// current entry is replaced
if (mpctx->playlist->current)
playlist_remove(mpctx->playlist, mpctx->playlist->current);
if (new)
mpctx->playlist->current = new;
} else {
MP_WARN(mpctx, "Empty playlist!\n");
}
}
static void process_hooks(struct MPContext *mpctx, char *name)
{
mp_hook_start(mpctx, name);
player: make playback termination asynchronous Until now, stopping playback aborted the demuxer and I/O layer violently by signaling mp_cancel (bound to libavformat's AVIOInterruptCB mechanism). Change it to try closing them gracefully. The main purpose is to silence those libavformat errors that happen when you request termination. Most of libavformat barely cares about the termination mechanism (AVIOInterruptCB), and essentially it's like the network connection is abruptly severed, or file I/O suddenly returns I/O errors. There were issues with dumb TLS warnings, parsers complaining about incomplete data, and some special protocols that require server communication to gracefully disconnect. We still want to abort it forcefully if it refuses to terminate on its own, so a timeout is required. Users can set the timeout to 0, which should give them the old behavior. This also removes the old mechanism that treats certain commands (like "quit") specially, and tries to terminate the demuxers even if the core is currently frozen. This is for situations where the core synchronized to the demuxer or stream layer while network is unresponsive. This in turn can only happen due to the "program" or "cache-size" properties in the current code (see one of the previous commits). Also, the old mechanism doesn't fit particularly well with the new one. We wouldn't want to abort playback immediately on a "quit" command - the new code is all about giving it a chance to end it gracefully. We'd need some sort of watchdog thread or something equally complicated to handle this. So just remove it. The change in osd.c is to prevent that it clears the status line while waiting for termination. The normal status line code doesn't output anything useful at this point, and the code path taken clears it, both of which is an annoying behavior change, so just let it show the old one.
2018-05-19 16:41:13 +00:00
while (!mp_hook_test_completion(mpctx, name)) {
mp_idle(mpctx);
player: make playback termination asynchronous Until now, stopping playback aborted the demuxer and I/O layer violently by signaling mp_cancel (bound to libavformat's AVIOInterruptCB mechanism). Change it to try closing them gracefully. The main purpose is to silence those libavformat errors that happen when you request termination. Most of libavformat barely cares about the termination mechanism (AVIOInterruptCB), and essentially it's like the network connection is abruptly severed, or file I/O suddenly returns I/O errors. There were issues with dumb TLS warnings, parsers complaining about incomplete data, and some special protocols that require server communication to gracefully disconnect. We still want to abort it forcefully if it refuses to terminate on its own, so a timeout is required. Users can set the timeout to 0, which should give them the old behavior. This also removes the old mechanism that treats certain commands (like "quit") specially, and tries to terminate the demuxers even if the core is currently frozen. This is for situations where the core synchronized to the demuxer or stream layer while network is unresponsive. This in turn can only happen due to the "program" or "cache-size" properties in the current code (see one of the previous commits). Also, the old mechanism doesn't fit particularly well with the new one. We wouldn't want to abort playback immediately on a "quit" command - the new code is all about giving it a chance to end it gracefully. We'd need some sort of watchdog thread or something equally complicated to handle this. So just remove it. The change in osd.c is to prevent that it clears the status line while waiting for termination. The normal status line code doesn't output anything useful at this point, and the code path taken clears it, both of which is an annoying behavior change, so just let it show the old one.
2018-05-19 16:41:13 +00:00
// We have no idea what blocks a hook, so just do a full abort.
if (mpctx->stop_play)
mp_abort_playback_async(mpctx);
}
}
// to be run on a worker thread, locked (temporarily unlocks core)
static void load_chapters(struct MPContext *mpctx)
{
struct demuxer *src = mpctx->demuxer;
bool free_src = false;
char *chapter_file = mpctx->opts->chapter_file;
if (chapter_file && chapter_file[0]) {
chapter_file = talloc_strdup(NULL, chapter_file);
mp_core_unlock(mpctx);
struct demuxer *demux = demux_open_url(chapter_file, NULL,
mpctx->playback_abort,
mpctx->global);
mp_core_lock(mpctx);
if (demux) {
src = demux;
free_src = true;
}
talloc_free(mpctx->chapters);
mpctx->chapters = NULL;
talloc_free(chapter_file);
}
if (src && !mpctx->chapters) {
talloc_free(mpctx->chapters);
mpctx->num_chapters = src->num_chapters;
mpctx->chapters = demux_copy_chapter_data(src->chapters, src->num_chapters);
if (mpctx->opts->rebase_start_time) {
for (int n = 0; n < mpctx->num_chapters; n++)
mpctx->chapters[n].pts -= src->start_time;
}
}
if (free_src)
demux_cancel_and_free(src);
}
static void load_per_file_options(m_config_t *conf,
struct playlist_param *params,
int params_count)
{
for (int n = 0; n < params_count; n++) {
m_config_set_option_cli(conf, params[n].name, params[n].value,
M_SETOPT_RUNTIME | M_SETOPT_BACKUP);
}
}
static void *open_demux_thread(void *ctx)
{
struct MPContext *mpctx = ctx;
mpthread_set_name("opener");
struct demuxer_params p = {
.force_format = mpctx->open_format,
.stream_flags = mpctx->open_url_flags,
.stream_record = true,
.is_top_level = true,
};
struct demuxer *demux =
demux_open_url(mpctx->open_url, &p, mpctx->open_cancel, mpctx->global);
mpctx->open_res_demuxer = demux;
if (demux) {
MP_VERBOSE(mpctx, "Opening done: %s\n", mpctx->open_url);
if (mpctx->open_for_prefetch && !demux->fully_read) {
int num_streams = demux_get_num_stream(demux);
for (int n = 0; n < num_streams; n++) {
struct sh_stream *sh = demux_get_stream(demux, n);
demuxer_select_track(demux, sh, MP_NOPTS_VALUE, true);
}
demux_set_wakeup_cb(demux, wakeup_demux, mpctx);
demux_start_thread(demux);
demux_start_prefetch(demux);
}
} else {
MP_VERBOSE(mpctx, "Opening failed or was aborted: %s\n", mpctx->open_url);
if (p.demuxer_failed) {
mpctx->open_res_error = MPV_ERROR_UNKNOWN_FORMAT;
} else {
mpctx->open_res_error = MPV_ERROR_LOADING_FAILED;
}
}
atomic_store(&mpctx->open_done, true);
mp_wakeup_core(mpctx);
return NULL;
}
static void cancel_open(struct MPContext *mpctx)
{
if (mpctx->open_cancel)
mp_cancel_trigger(mpctx->open_cancel);
if (mpctx->open_active)
pthread_join(mpctx->open_thread, NULL);
mpctx->open_active = false;
if (mpctx->open_res_demuxer)
demux_cancel_and_free(mpctx->open_res_demuxer);
mpctx->open_res_demuxer = NULL;
TA_FREEP(&mpctx->open_cancel);
TA_FREEP(&mpctx->open_url);
TA_FREEP(&mpctx->open_format);
atomic_store(&mpctx->open_done, false);
}
// Setup all the field to open this url, and make sure a thread is running.
static void start_open(struct MPContext *mpctx, char *url, int url_flags,
bool for_prefetch)
{
cancel_open(mpctx);
assert(!mpctx->open_active);
assert(!mpctx->open_cancel);
assert(!mpctx->open_res_demuxer);
assert(!atomic_load(&mpctx->open_done));
mpctx->open_cancel = mp_cancel_new(NULL);
mpctx->open_url = talloc_strdup(NULL, url);
mpctx->open_format = talloc_strdup(NULL, mpctx->opts->demuxer_name);
mpctx->open_url_flags = url_flags;
mpctx->open_for_prefetch = for_prefetch && mpctx->opts->demuxer_thread;
if (mpctx->opts->load_unsafe_playlists)
mpctx->open_url_flags = 0;
if (pthread_create(&mpctx->open_thread, NULL, open_demux_thread, mpctx)) {
cancel_open(mpctx);
return;
}
mpctx->open_active = true;
}
static void open_demux_reentrant(struct MPContext *mpctx)
{
char *url = mpctx->stream_open_filename;
if (mpctx->open_active) {
bool done = atomic_load(&mpctx->open_done);
bool failed = done && !mpctx->open_res_demuxer;
bool correct_url = strcmp(mpctx->open_url, url) == 0;
if (correct_url && !failed) {
MP_VERBOSE(mpctx, "Using prefetched/prefetching URL.\n");
} else if (correct_url && failed) {
MP_VERBOSE(mpctx, "Prefetched URL failed, retrying.\n");
cancel_open(mpctx);
} else {
if (done) {
MP_VERBOSE(mpctx, "Dropping finished prefetch of wrong URL.\n");
} else {
2017-10-23 08:53:28 +00:00
MP_VERBOSE(mpctx, "Aborting ongoing prefetch of wrong URL.\n");
}
cancel_open(mpctx);
}
}
if (!mpctx->open_active)
start_open(mpctx, url, mpctx->playing->stream_flags, false);
// User abort should cancel the opener now.
mp_cancel_set_parent(mpctx->open_cancel, mpctx->playback_abort);
while (!atomic_load(&mpctx->open_done)) {
mp_idle(mpctx);
if (mpctx->stop_play)
mp_abort_playback_async(mpctx);
}
if (mpctx->open_res_demuxer) {
mpctx->demuxer = mpctx->open_res_demuxer;
mpctx->open_res_demuxer = NULL;
mp_cancel_set_parent(mpctx->demuxer->cancel, mpctx->playback_abort);
} else {
mpctx->error_playing = mpctx->open_res_error;
}
cancel_open(mpctx); // cleanup
}
void prefetch_next(struct MPContext *mpctx)
{
if (!mpctx->opts->prefetch_open)
return;
struct playlist_entry *new_entry = mp_next_file(mpctx, +1, false, false);
if (new_entry && !mpctx->open_active && new_entry->filename) {
MP_VERBOSE(mpctx, "Prefetching: %s\n", new_entry->filename);
start_open(mpctx, new_entry->filename, new_entry->stream_flags, true);
}
}
// Destroy the complex filter, and remove the references to the filter pads.
// (Call cleanup_deassociated_complex_filters() to close decoders/VO/AO
// that are not connected anymore due to this.)
static void deassociate_complex_filters(struct MPContext *mpctx)
{
for (int n = 0; n < mpctx->num_tracks; n++)
mpctx->tracks[n]->sink = NULL;
if (mpctx->vo_chain)
mpctx->vo_chain->filter_src = NULL;
if (mpctx->ao_chain)
mpctx->ao_chain->filter_src = NULL;
TA_FREEP(&mpctx->lavfi);
TA_FREEP(&mpctx->lavfi_graph);
}
// Close all decoders and sinks (AO/VO) that are not connected to either
// a track or a filter pad.
static void cleanup_deassociated_complex_filters(struct MPContext *mpctx)
{
for (int n = 0; n < mpctx->num_tracks; n++) {
struct track *track = mpctx->tracks[n];
if (!(track->sink || track->vo_c || track->ao_c)) {
if (track->dec && !track->vo_c && !track->ao_c) {
talloc_free(track->dec->f);
track->dec = NULL;
}
track->selected = false;
}
}
if (mpctx->vo_chain && !mpctx->vo_chain->dec_src &&
!mpctx->vo_chain->filter_src)
{
uninit_video_chain(mpctx);
}
if (mpctx->ao_chain && !mpctx->ao_chain->dec_src &&
!mpctx->ao_chain->filter_src)
{
uninit_audio_chain(mpctx);
}
}
static void kill_outputs(struct MPContext *mpctx, struct track *track)
{
if (track->vo_c || track->ao_c) {
MP_VERBOSE(mpctx, "deselecting track %d for lavfi-complex option\n",
track->user_tid);
mp_switch_track(mpctx, track->type, NULL, 0);
}
assert(!(track->vo_c || track->ao_c));
}
// >0: changed, 0: no change, -1: error
static int reinit_complex_filters(struct MPContext *mpctx, bool force_uninit)
{
char *graph = mpctx->opts->lavfi_complex;
bool have_graph = graph && graph[0] && !force_uninit;
if (have_graph && mpctx->lavfi &&
strcmp(graph, mpctx->lavfi_graph) == 0 &&
!mp_filter_has_failed(mpctx->lavfi))
return 0;
if (!mpctx->lavfi && !have_graph)
return 0;
// Deassociate the old filter pads. We leave both sources (tracks) and
// sinks (AO/VO) "dangling", connected to neither track or filter pad.
// Later, we either reassociate them with new pads, or uninit them if
// they are still dangling. This avoids too interruptive actions like
// recreating the VO.
deassociate_complex_filters(mpctx);
bool success = false;
if (!have_graph) {
success = true; // normal full removal of graph
goto done;
}
struct mp_lavfi *l =
mp_lavfi_create_graph(mpctx->filter_root, 0, false, NULL, graph);
if (!l)
goto done;
mpctx->lavfi = l->f;
mpctx->lavfi_graph = talloc_strdup(NULL, graph);
mp_filter_set_error_handler(mpctx->lavfi, mpctx->filter_root);
for (int n = 0; n < mpctx->lavfi->num_pins; n++)
mp_pin_disconnect(mpctx->lavfi->pins[n]);
struct mp_pin *pad = mp_filter_get_named_pin(mpctx->lavfi, "vo");
if (pad && mp_pin_get_dir(pad) == MP_PIN_OUT) {
if (mpctx->vo_chain && mpctx->vo_chain->track)
kill_outputs(mpctx, mpctx->vo_chain->track);
if (!mpctx->vo_chain) {
reinit_video_chain_src(mpctx, NULL);
if (!mpctx->vo_chain)
goto done;
}
struct vo_chain *vo_c = mpctx->vo_chain;
assert(!vo_c->track);
vo_c->filter_src = pad;
mp_pin_connect(vo_c->filter->f->pins[0], vo_c->filter_src);
}
pad = mp_filter_get_named_pin(mpctx->lavfi, "ao");
if (pad && mp_pin_get_dir(pad) == MP_PIN_OUT) {
if (mpctx->ao_chain && mpctx->ao_chain->track)
kill_outputs(mpctx, mpctx->ao_chain->track);
if (!mpctx->ao_chain) {
reinit_audio_chain_src(mpctx, NULL);
if (!mpctx->ao_chain)
goto done;
}
struct ao_chain *ao_c = mpctx->ao_chain;
assert(!ao_c->track);
ao_c->filter_src = pad;
mp_pin_connect(ao_c->filter->f->pins[0], ao_c->filter_src);
}
for (int n = 0; n < mpctx->num_tracks; n++) {
struct track *track = mpctx->tracks[n];
char label[32];
char prefix;
switch (track->type) {
case STREAM_VIDEO: prefix = 'v'; break;
case STREAM_AUDIO: prefix = 'a'; break;
default: continue;
}
snprintf(label, sizeof(label), "%cid%d", prefix, track->user_tid);
pad = mp_filter_get_named_pin(mpctx->lavfi, label);
if (!pad)
continue;
if (mp_pin_get_dir(pad) != MP_PIN_IN)
continue;
assert(!mp_pin_is_connected(pad));
assert(!track->sink);
kill_outputs(mpctx, track);
track->sink = pad;
track->selected = true;
if (!track->dec) {
if (track->type == STREAM_VIDEO && !init_video_decoder(mpctx, track))
goto done;
if (track->type == STREAM_AUDIO && !init_audio_decoder(mpctx, track))
goto done;
}
mp_pin_connect(track->sink, track->dec->f->pins[0]);
}
// Don't allow unconnected pins. Libavfilter would make the data flow a
// real pain anyway.
for (int n = 0; n < mpctx->lavfi->num_pins; n++) {
struct mp_pin *pin = mpctx->lavfi->pins[n];
if (!mp_pin_is_connected(pin)) {
MP_ERR(mpctx, "Pad %s is not connected to anything.\n",
mp_pin_get_name(pin));
goto done;
}
}
success = true;
done:
if (!success)
deassociate_complex_filters(mpctx);
cleanup_deassociated_complex_filters(mpctx);
if (mpctx->playback_initialized) {
for (int n = 0; n < mpctx->num_tracks; n++)
reselect_demux_stream(mpctx, mpctx->tracks[n]);
}
mp_notify(mpctx, MPV_EVENT_TRACKS_CHANGED, NULL);
return success ? 1 : -1;
}
void update_lavfi_complex(struct MPContext *mpctx)
{
if (mpctx->playback_initialized) {
if (reinit_complex_filters(mpctx, false) != 0)
issue_refresh_seek(mpctx, MPSEEK_EXACT);
}
}
// Worker thread for loading external files and such. This is needed to avoid
// freezing the core when waiting for network while loading these.
static void load_external_opts_thread(void *p)
{
void **a = p;
struct MPContext *mpctx = a[0];
struct mp_waiter *waiter = a[1];
mp_core_lock(mpctx);
load_chapters(mpctx);
open_external_files(mpctx, mpctx->opts->audio_files, STREAM_AUDIO);
open_external_files(mpctx, mpctx->opts->sub_name, STREAM_SUB);
open_external_files(mpctx, mpctx->opts->external_files, STREAM_TYPE_COUNT);
autoload_external_files(mpctx, mpctx->playback_abort);
mp_waiter_wakeup(waiter, 0);
mp_wakeup_core(mpctx);
mp_core_unlock(mpctx);
}
static void load_external_opts(struct MPContext *mpctx)
{
struct mp_waiter wait = MP_WAITER_INITIALIZER;
void *a[] = {mpctx, &wait};
if (!mp_thread_pool_queue(mpctx->thread_pool, load_external_opts_thread, a)) {
mpctx->stop_play = PT_ERROR;
return;
}
player: make playback termination asynchronous Until now, stopping playback aborted the demuxer and I/O layer violently by signaling mp_cancel (bound to libavformat's AVIOInterruptCB mechanism). Change it to try closing them gracefully. The main purpose is to silence those libavformat errors that happen when you request termination. Most of libavformat barely cares about the termination mechanism (AVIOInterruptCB), and essentially it's like the network connection is abruptly severed, or file I/O suddenly returns I/O errors. There were issues with dumb TLS warnings, parsers complaining about incomplete data, and some special protocols that require server communication to gracefully disconnect. We still want to abort it forcefully if it refuses to terminate on its own, so a timeout is required. Users can set the timeout to 0, which should give them the old behavior. This also removes the old mechanism that treats certain commands (like "quit") specially, and tries to terminate the demuxers even if the core is currently frozen. This is for situations where the core synchronized to the demuxer or stream layer while network is unresponsive. This in turn can only happen due to the "program" or "cache-size" properties in the current code (see one of the previous commits). Also, the old mechanism doesn't fit particularly well with the new one. We wouldn't want to abort playback immediately on a "quit" command - the new code is all about giving it a chance to end it gracefully. We'd need some sort of watchdog thread or something equally complicated to handle this. So just remove it. The change in osd.c is to prevent that it clears the status line while waiting for termination. The normal status line code doesn't output anything useful at this point, and the code path taken clears it, both of which is an annoying behavior change, so just let it show the old one.
2018-05-19 16:41:13 +00:00
while (!mp_waiter_poll(&wait)) {
mp_idle(mpctx);
player: make playback termination asynchronous Until now, stopping playback aborted the demuxer and I/O layer violently by signaling mp_cancel (bound to libavformat's AVIOInterruptCB mechanism). Change it to try closing them gracefully. The main purpose is to silence those libavformat errors that happen when you request termination. Most of libavformat barely cares about the termination mechanism (AVIOInterruptCB), and essentially it's like the network connection is abruptly severed, or file I/O suddenly returns I/O errors. There were issues with dumb TLS warnings, parsers complaining about incomplete data, and some special protocols that require server communication to gracefully disconnect. We still want to abort it forcefully if it refuses to terminate on its own, so a timeout is required. Users can set the timeout to 0, which should give them the old behavior. This also removes the old mechanism that treats certain commands (like "quit") specially, and tries to terminate the demuxers even if the core is currently frozen. This is for situations where the core synchronized to the demuxer or stream layer while network is unresponsive. This in turn can only happen due to the "program" or "cache-size" properties in the current code (see one of the previous commits). Also, the old mechanism doesn't fit particularly well with the new one. We wouldn't want to abort playback immediately on a "quit" command - the new code is all about giving it a chance to end it gracefully. We'd need some sort of watchdog thread or something equally complicated to handle this. So just remove it. The change in osd.c is to prevent that it clears the status line while waiting for termination. The normal status line code doesn't output anything useful at this point, and the code path taken clears it, both of which is an annoying behavior change, so just let it show the old one.
2018-05-19 16:41:13 +00:00
if (mpctx->stop_play)
mp_abort_playback_async(mpctx);
}
mp_waiter_wait(&wait);
}
// Start playing the current playlist entry.
// Handle initialization and deinitialization.
static void play_current_file(struct MPContext *mpctx)
{
struct MPOpts *opts = mpctx->opts;
double playback_start = -1e100;
assert(mpctx->stop_play);
mp_notify(mpctx, MPV_EVENT_START_FILE, NULL);
stream: redo playback abort handling This mechanism originates from MPlayer's way of dealing with blocking network, but it's still useful. On opening and closing, mpv waits for network synchronously, and also some obscure commands and use-cases can lead to such blocking. In these situations, the stream is asynchronously forced to stop by "interrupting" it. The old design interrupting I/O was a bit broken: polling with a callback, instead of actively interrupting it. Change the direction of this. There is no callback anymore, and the player calls mp_cancel_trigger() to force the stream to return. libavformat (via stream_lavf.c) has the old broken design, and fixing it would require fixing libavformat, which won't happen so quickly. So we have to keep that part. But everything above the stream layer is prepared for a better design, and more sophisticated methods than mp_cancel_test() could be easily introduced. There's still one problem: commands are still run in the central playback loop, which we assume can block on I/O in the worst case. That's not a problem yet, because we simply mark some commands as being able to stop playback of the current file ("quit" etc.), so input.c could abort playback as soon as such a command is queued. But there are also commands abort playback only conditionally, and the logic for that is in the playback core and thus "unreachable". For example, "playlist_next" aborts playback only if there's a next file. We don't want it to always abort playback. As a quite ugly hack, abort playback only if at least 2 abort commands are queued - this pretty much happens only if the core is frozen and doesn't react to input.
2014-09-13 12:23:08 +00:00
mp_cancel_reset(mpctx->playback_abort);
mpctx->error_playing = MPV_ERROR_LOADING_FAILED;
mpctx->stop_play = 0;
mpctx->filename = NULL;
mpctx->shown_aframes = 0;
mpctx->shown_vframes = 0;
mpctx->last_vo_pts = MP_NOPTS_VALUE;
mpctx->last_chapter_seek = -2;
mpctx->last_chapter_pts = MP_NOPTS_VALUE;
mpctx->last_chapter = -2;
mpctx->paused = false;
mpctx->playing_msg_shown = false;
mpctx->max_frames = -1;
mpctx->video_speed = mpctx->audio_speed = opts->playback_speed;
mpctx->speed_factor_a = mpctx->speed_factor_v = 1.0;
mpctx->display_sync_error = 0.0;
mpctx->display_sync_active = false;
// let get_current_time() show 0 as start time (before playback_pts is set)
mpctx->last_seek_pts = 0.0;
mpctx->seek = (struct seek_params){ 0 };
video: rewrite filtering glue code Get rid of the old vf.c code. Replace it with a generic filtering framework, which can potentially handle more than just --vf. At least reimplementing --af with this code is planned. This changes some --vf semantics (including runtime behavior and the "vf" command). The most important ones are listed in interface-changes. vf_convert.c is renamed to f_swscale.c. It is now an internal filter that can not be inserted by the user manually. f_lavfi.c is a refactor of player/lavfi.c. The latter will be removed once --lavfi-complex is reimplemented on top of f_lavfi.c. (which is conceptually easy, but a big mess due to the data flow changes). The existing filters are all changed heavily. The data flow of the new filter framework is different. Especially EOF handling changes - EOF is now a "frame" rather than a state, and must be passed through exactly once. Another major thing is that all filters must support dynamic format changes. The filter reconfig() function goes away. (This sounds complex, but since all filters need to handle EOF draining anyway, they can use the same code, and it removes the mess with reconfig() having to predict the output format, which completely breaks with libavfilter anyway.) In addition, there is no automatic format negotiation or conversion. libavfilter's primitive and insufficient API simply doesn't allow us to do this in a reasonable way. Instead, filters can use f_autoconvert as sub-filter, and tell it which formats they support. This filter will in turn add actual conversion filters, such as f_swscale, to perform necessary format changes. vf_vapoursynth.c uses the same basic principle of operation as before, but with worryingly different details in data flow. Still appears to work. The hardware deint filters (vf_vavpp.c, vf_d3d11vpp.c, vf_vdpaupp.c) are heavily changed. Fortunately, they all used refqueue.c, which is for sharing the data flow logic (especially for managing future/past surfaces and such). It turns out it can be used to factor out most of the data flow. Some of these filters accepted software input. Instead of having ad-hoc upload code in each filter, surface upload is now delegated to f_autoconvert, which can use f_hwupload to perform this. Exporting VO capabilities is still a big mess (mp_stream_info stuff). The D3D11 code drops the redundant image formats, and all code uses the hw_subfmt (sw_format in FFmpeg) instead. Although that too seems to be a big mess for now. f_async_queue is unused.
2018-01-16 10:53:44 +00:00
mpctx->filter_root = mp_filter_create_root(mpctx->global);
mp_filter_root_set_wakeup_cb(mpctx->filter_root, mp_wakeup_core_cb, mpctx);
reset_playback_state(mpctx);
mpctx->playing = mpctx->playlist->current;
if (!mpctx->playing || !mpctx->playing->filename)
goto terminate_playback;
mpctx->playing->reserved += 1;
mpctx->filename = talloc_strdup(NULL, mpctx->playing->filename);
mpctx->stream_open_filename = mpctx->filename;
mpctx->add_osd_seek_info &= OSD_SEEK_INFO_CURRENT_FILE;
if (opts->reset_options) {
for (int n = 0; opts->reset_options[n]; n++) {
const char *opt = opts->reset_options[n];
if (opt[0]) {
if (strcmp(opt, "all") == 0) {
m_config_backup_all_opts(mpctx->mconfig);
} else {
m_config_backup_opt(mpctx->mconfig, opt);
}
}
}
}
mp_load_auto_profiles(mpctx);
mp_load_playback_resume(mpctx, mpctx->filename);
load_per_file_options(mpctx->mconfig, mpctx->playing->params,
mpctx->playing->num_params);
mpctx->max_frames = opts->play_frames;
handle_force_window(mpctx, false);
if (mpctx->playlist->first != mpctx->playing ||
mpctx->playlist->last != mpctx->playing ||
mpctx->playing->num_redirects)
MP_INFO(mpctx, "Playing: %s\n", mpctx->filename);
assert(mpctx->demuxer == NULL);
process_hooks(mpctx, "on_load");
if (mpctx->stop_play)
goto terminate_playback;
if (opts->stream_dump && opts->stream_dump[0]) {
if (stream_dump(mpctx, mpctx->stream_open_filename) >= 0)
mpctx->error_playing = 1;
goto terminate_playback;
}
open_demux_reentrant(mpctx);
if (!mpctx->stop_play && !mpctx->demuxer) {
process_hooks(mpctx, "on_load_fail");
if (strcmp(mpctx->stream_open_filename, mpctx->filename) != 0 &&
!mpctx->stop_play)
{
mpctx->error_playing = MPV_ERROR_LOADING_FAILED;
open_demux_reentrant(mpctx);
}
2018-01-02 14:20:53 +00:00
}
if (!mpctx->demuxer || mpctx->stop_play)
goto terminate_playback;
if (mpctx->demuxer->playlist) {
struct playlist *pl = mpctx->demuxer->playlist;
int entry_stream_flags = 0;
loadfile, ytdl_hook: don't reject EDL-resolved URLs through playlist The ytdl wrapper can resolve web links to playlists. This playlist is passed as big memory:// blob, and will contain further quite normal web links. When playback of one of these playlist entries starts, ytdl is called again and will resolve the web link to a media URL again. This didn't work if playlist entries resolved to EDL URLs. Playback was rejected with a "potentially unsafe URL from playlist" error. This was completely weird and unexpected: using the playlist entry directly on the command line worked fine, and there isn't a reason why it should be different for a playlist entry (both are resolved by the ytdl wrapper anyway). Also, if the only EDL URL was added via audio-add or sub-add, the URL was accessed successfully. The reason this happened is because the playlist entries were marked as STREAM_SAFE_ONLY, and edl:// is not marked as "safe". Playlist entries passed via command line directly are not marked, so resolving them to EDL worked. Fix this by making the ytdl hook set load-unsafe-playlists while the playlist is parsed. (After the playlist is parsed, and before the first playlist entry is played, file-local options are reset again.) Further, extend the load-unsafe-playlists option so that the playlist entries are not marked while the playlist is loaded. Since playlist entries are already verified, this should change nothing about the actual security situation. There are now 2 locations which check load_unsafe_playlists. The old one is a bit redundant now. In theory, the playlist loading code might not be the only code which sets these flags, so keeping the old code is somewhat justified (and in any case it doesn't hurt to keep it). In general, the security concept sucks (and always did). I can for example not answer the question whether you can "break" this mechanism with various combinations of archives, EDL files, playlists files, compromised sites, and so on. You probably can, and I'm fully aware that it's probably possible, so don't blame me.
2019-01-04 12:48:27 +00:00
if (!pl->disable_safety && !mpctx->opts->load_unsafe_playlists) {
entry_stream_flags = STREAM_SAFE_ONLY;
if (mpctx->demuxer->is_network)
entry_stream_flags |= STREAM_NETWORK_ONLY;
}
for (struct playlist_entry *e = pl->first; e; e = e->next)
e->stream_flags |= entry_stream_flags;
transfer_playlist(mpctx, pl);
mp_notify_property(mpctx, "playlist");
mpctx->error_playing = 2;
goto terminate_playback;
}
if (mpctx->opts->rebase_start_time)
demux_set_ts_offset(mpctx->demuxer, -mpctx->demuxer->start_time);
enable_demux_thread(mpctx, mpctx->demuxer);
add_demuxer_tracks(mpctx, mpctx->demuxer);
load_external_opts(mpctx);
if (mpctx->stop_play)
goto terminate_playback;
check_previous_track_selection(mpctx);
process_hooks(mpctx, "on_preloaded");
if (mpctx->stop_play)
goto terminate_playback;
if (reinit_complex_filters(mpctx, false) < 0)
goto terminate_playback;
assert(NUM_PTRACKS == 2); // opts->stream_id is hardcoded to 2
for (int t = 0; t < STREAM_TYPE_COUNT; t++) {
for (int i = 0; i < NUM_PTRACKS; i++) {
struct track *sel = NULL;
bool taken = (t == STREAM_VIDEO && mpctx->vo_chain) ||
(t == STREAM_AUDIO && mpctx->ao_chain);
if (!taken && opts->stream_auto_sel)
sel = select_default_track(mpctx, i, t);
mpctx->current_track[i][t] = sel;
}
}
for (int t = 0; t < STREAM_TYPE_COUNT; t++) {
for (int i = 0; i < NUM_PTRACKS; i++) {
struct track *track = mpctx->current_track[i][t];
if (track) {
if (track->selected) {
MP_ERR(mpctx, "Track %d can't be selected twice.\n",
track->user_tid);
mpctx->current_track[i][t] = NULL;
} else {
track->selected = true;
}
}
}
}
for (int n = 0; n < mpctx->num_tracks; n++)
reselect_demux_stream(mpctx, mpctx->tracks[n]);
update_demuxer_properties(mpctx);
update_playback_speed(mpctx);
reinit_video_chain(mpctx);
reinit_audio_chain(mpctx);
reinit_sub_all(mpctx);
if (mpctx->encode_lavc_ctx) {
if (mpctx->vo_chain)
encode_lavc_expect_stream(mpctx->encode_lavc_ctx, STREAM_VIDEO);
if (mpctx->ao_chain)
encode_lavc_expect_stream(mpctx->encode_lavc_ctx, STREAM_AUDIO);
encode_lavc_set_metadata(mpctx->encode_lavc_ctx,
mpctx->demuxer->metadata);
}
if (!mpctx->vo_chain && !mpctx->ao_chain && opts->stream_auto_sel) {
MP_FATAL(mpctx, "No video or audio streams selected.\n");
mpctx->error_playing = MPV_ERROR_NOTHING_TO_PLAY;
goto terminate_playback;
}
if (mpctx->vo_chain && mpctx->vo_chain->is_coverart) {
MP_INFO(mpctx,
"Displaying attached picture. Use --no-audio-display to prevent this.\n");
}
if (!mpctx->vo_chain)
handle_force_window(mpctx, true);
MP_VERBOSE(mpctx, "Starting playback...\n");
mpctx->playback_initialized = true;
mp_notify(mpctx, MPV_EVENT_FILE_LOADED, NULL);
update_screensaver_state(mpctx);
if (mpctx->max_frames == 0) {
if (!mpctx->stop_play)
mpctx->stop_play = PT_NEXT_ENTRY;
mpctx->error_playing = 0;
goto terminate_playback;
}
if (opts->demuxer_cache_wait) {
demux_start_prefetch(mpctx->demuxer);
while (!mpctx->stop_play) {
struct demux_reader_state s;
demux_get_reader_state(mpctx->demuxer, &s);
if (s.idle)
break;
mp_idle(mpctx);
}
}
player: modify/simplify AB-loop behavior This changes the behavior of the --ab-loop-a/b options. In addition, it makes it work with backward playback mode. The most obvious change is that the both the A and B point need to be set now before any looping happens. Unlike before, unset points don't implicitly use the start or end of the file. I think the old behavior was a feature that was explicitly added/wanted. Well, it's gone now. This is because of 2 reasons: 1. I never liked this feature, and it always got in my way (as user). 2. It's inherently annoying with backward playback mode. In backward playback mode, the user wants to set A/B in the wrong order. The ab-loop command will first set A, then B, so if you use this command during backward playback, A will be set to a higher timestamps than B. If you switch back to forward playback mode, the loop would stop working. I want the loop to just continue to work, and the chosen solution conflicts with the removed feature. The order issue above _could_ be fixed by also switching the AB-loop user option values around on direction switch. But there are no other instances of option changes magically affecting other options, and doing this would probably lead to unexpected misery (dying from corner cases and such). Another solution is sorting the A/B points by timestamps after copying them from the user options. Then A/B options set in backward mode will work in forward mode. This is the chosen solution. If you sort the points, you don't know anymore whether the unset point is supposed to signify the end or the start of the file. The AB-loop code is slightly better abstracted now, so it should be easy to restore the removed feature. It would still require coming up with a solution for backwards playback, though. A minor change is that if one point is set and the other is unset, I'm rendering both the chapter markers and the marker for the set point. Why? I don't know. My test file had chapters, and I guess I decided this looked better. This commit also fixes some subtle and obvious issues that I already forgot about when I wrote this commit message. It cleans up some minor code duplication and nonsense too. Regarding backward playback, the code uses an unsanitary mix of internal ("transformed") and user timestamps. So the play_dir variable appears more than usual. To mention one unfixed issue: if you set an AB-loop that is completely past the end of the file, it will get stuck in an infinite seeking loop once playback reaches the end of the file. Fixing this reliably seemed annoying, so the fix is "just don't do this". It's not a hard freeze anyway.
2019-05-26 23:24:22 +00:00
// (Not get_play_start_pts(), which would always trigger a seek.)
double play_start_pts = rel_time_to_abs(mpctx, opts->play_start);
Implement backwards playback See manpage additions. This is a huge hack. You can bet there are shit tons of bugs. It's literally forcing square pegs into round holes. Hopefully, the manpage wall of text makes it clear enough that the whole shit can easily crash and burn. (Although it shouldn't literally crash. That would be a bug. It possibly _could_ start a fire by entering some sort of endless loop, not a literal one, just something where it tries to do work without making progress.) (Some obvious bugs I simply ignored for this initial version, but there's a number of potential bugs I can't even imagine. Normal playback should remain completely unaffected, though.) How this works is also described in the manpage. Basically, we demux in reverse, then we decode in reverse, then we render in reverse. The decoding part is the simplest: just reorder the decoder output. This weirdly integrates with the timeline/ordered chapter code, which also has special requirements on feeding the packets to the decoder in a non-straightforward way (it doesn't conflict, although a bugmessmass breaks correct slicing of segments, so EDL/ordered chapter playback is broken in backward direction). Backward demuxing is pretty involved. In theory, it could be much easier: simply iterating the usual demuxer output backward. But this just doesn't fit into our code, so there's a cthulhu nightmare of shit. To be specific, each stream (audio, video) is reversed separately. At least this means we can do backward playback within cached content (for example, you could play backwards in a live stream; on that note, it disables prefetching, which would lead to losing new live video, but this could be avoided). The fuckmess also meant that I didn't bother trying to support subtitles. Subtitles are a problem because they're "sparse" streams. They need to be "passively" demuxed: you don't try to read a subtitle packet, you demux audio and video, and then look whether there was a subtitle packet. This means to get subtitles for a time range, you need to know that you demuxed video and audio over this range, which becomes pretty messy when you demux audio and video backwards separately. Backward display is the most weird (and potentially buggy) part. To avoid that we need to touch a LOT of timing code, we negate all timestamps. The basic idea is that due to the navigation, all comparisons and subtractions of timestamps keep working, and you don't need to touch every single of them to "reverse" them. E.g.: bool before = pts_a < pts_b; would need to be: bool before = forward ? pts_a < pts_b : pts_a > pts_b; or: bool before = pts_a * dir < pts_b * dir; or if you, as it's implemented now, just do this after decoding: pts_a *= dir; pts_b *= dir; and then in the normal timing/renderer code: bool before = pts_a < pts_b; Consequently, we don't need many changes in the latter code. But some assumptions inhererently true for forward playback may have been broken anyway. What is mainly needed is fixing places where values are passed between positive and negative "domains". For example, seeking and timestamp user display always uses positive timestamps. The main mess is that it's not obvious which domain a given variable should or does use. Well, in my tests with a single file, it suddenly started to work when I did this. I'm honestly surprised that it did, and that I didn't have to change a single line in the timing code past decoder (just something minor to make external/cached text subtitles display). I committed it immediately while avoiding thinking about it. But there really likely are subtle problems of all sorts. As far as I'm aware, gstreamer also supports backward playback. When I looked at this years ago, I couldn't find a way to actually try this, and I didn't revisit it now. Back then I also read talk slides from the person who implemented it, and I'm not sure if and which ideas I might have taken from it. It's possible that the timestamp reversal is inspired by it, but I didn't check. (I think it claimed that it could avoid large changes by changing a sign?) VapourSynth has some sort of reverse function, which provides a backward view on a video. The function itself is trivial to implement, as VapourSynth aims to provide random access to video by frame numbers (so you just request decreasing frame numbers). From what I remember, it wasn't exactly fluid, but it worked. It's implemented by creating an index, and seeking to the target on demand, and a bunch of caching. mpv could use it, but it would either require using VapourSynth as demuxer and decoder for everything, or replacing the current file every time something is supposed to be played backwards. FFmpeg's libavfilter has reversal filters for audio and video. These require buffering the entire media data of the file, and don't really fit into mpv's architecture. It could be used by playing a libavfilter graph that also demuxes, but that's like VapourSynth but worse.
2019-05-18 00:10:51 +00:00
// Backward playback -> start from end by default.
if (play_start_pts == MP_NOPTS_VALUE && opts->play_dir < 0)
play_start_pts = get_start_time(mpctx, -1);
Implement backwards playback See manpage additions. This is a huge hack. You can bet there are shit tons of bugs. It's literally forcing square pegs into round holes. Hopefully, the manpage wall of text makes it clear enough that the whole shit can easily crash and burn. (Although it shouldn't literally crash. That would be a bug. It possibly _could_ start a fire by entering some sort of endless loop, not a literal one, just something where it tries to do work without making progress.) (Some obvious bugs I simply ignored for this initial version, but there's a number of potential bugs I can't even imagine. Normal playback should remain completely unaffected, though.) How this works is also described in the manpage. Basically, we demux in reverse, then we decode in reverse, then we render in reverse. The decoding part is the simplest: just reorder the decoder output. This weirdly integrates with the timeline/ordered chapter code, which also has special requirements on feeding the packets to the decoder in a non-straightforward way (it doesn't conflict, although a bugmessmass breaks correct slicing of segments, so EDL/ordered chapter playback is broken in backward direction). Backward demuxing is pretty involved. In theory, it could be much easier: simply iterating the usual demuxer output backward. But this just doesn't fit into our code, so there's a cthulhu nightmare of shit. To be specific, each stream (audio, video) is reversed separately. At least this means we can do backward playback within cached content (for example, you could play backwards in a live stream; on that note, it disables prefetching, which would lead to losing new live video, but this could be avoided). The fuckmess also meant that I didn't bother trying to support subtitles. Subtitles are a problem because they're "sparse" streams. They need to be "passively" demuxed: you don't try to read a subtitle packet, you demux audio and video, and then look whether there was a subtitle packet. This means to get subtitles for a time range, you need to know that you demuxed video and audio over this range, which becomes pretty messy when you demux audio and video backwards separately. Backward display is the most weird (and potentially buggy) part. To avoid that we need to touch a LOT of timing code, we negate all timestamps. The basic idea is that due to the navigation, all comparisons and subtractions of timestamps keep working, and you don't need to touch every single of them to "reverse" them. E.g.: bool before = pts_a < pts_b; would need to be: bool before = forward ? pts_a < pts_b : pts_a > pts_b; or: bool before = pts_a * dir < pts_b * dir; or if you, as it's implemented now, just do this after decoding: pts_a *= dir; pts_b *= dir; and then in the normal timing/renderer code: bool before = pts_a < pts_b; Consequently, we don't need many changes in the latter code. But some assumptions inhererently true for forward playback may have been broken anyway. What is mainly needed is fixing places where values are passed between positive and negative "domains". For example, seeking and timestamp user display always uses positive timestamps. The main mess is that it's not obvious which domain a given variable should or does use. Well, in my tests with a single file, it suddenly started to work when I did this. I'm honestly surprised that it did, and that I didn't have to change a single line in the timing code past decoder (just something minor to make external/cached text subtitles display). I committed it immediately while avoiding thinking about it. But there really likely are subtle problems of all sorts. As far as I'm aware, gstreamer also supports backward playback. When I looked at this years ago, I couldn't find a way to actually try this, and I didn't revisit it now. Back then I also read talk slides from the person who implemented it, and I'm not sure if and which ideas I might have taken from it. It's possible that the timestamp reversal is inspired by it, but I didn't check. (I think it claimed that it could avoid large changes by changing a sign?) VapourSynth has some sort of reverse function, which provides a backward view on a video. The function itself is trivial to implement, as VapourSynth aims to provide random access to video by frame numbers (so you just request decreasing frame numbers). From what I remember, it wasn't exactly fluid, but it worked. It's implemented by creating an index, and seeking to the target on demand, and a bunch of caching. mpv could use it, but it would either require using VapourSynth as demuxer and decoder for everything, or replacing the current file every time something is supposed to be played backwards. FFmpeg's libavfilter has reversal filters for audio and video. These require buffering the entire media data of the file, and don't really fit into mpv's architecture. It could be used by playing a libavfilter graph that also demuxes, but that's like VapourSynth but worse.
2019-05-18 00:10:51 +00:00
if (play_start_pts != MP_NOPTS_VALUE) {
queue_seek(mpctx, MPSEEK_ABSOLUTE, play_start_pts, MPSEEK_DEFAULT, 0);
execute_queued_seek(mpctx);
}
update_internal_pause_state(mpctx);
open_recorder(mpctx, true);
playback_start = mp_time_sec();
mpctx->error_playing = 0;
mpctx->in_playloop = true;
while (!mpctx->stop_play)
run_playloop(mpctx);
mpctx->in_playloop = false;
MP_VERBOSE(mpctx, "EOF code: %d \n", mpctx->stop_play);
terminate_playback:
if (!mpctx->stop_play)
mpctx->stop_play = PT_ERROR;
if (mpctx->stop_play != AT_END_OF_FILE)
clear_audio_output_buffers(mpctx);
update_core_idle_state(mpctx);
process_hooks(mpctx, "on_unload");
if (mpctx->step_frames)
opts->pause = 1;
close_recorder(mpctx);
// time to uninit all, except global stuff:
reinit_complex_filters(mpctx, true);
uninit_audio_chain(mpctx);
uninit_video_chain(mpctx);
uninit_sub_all(mpctx);
if (!opts->gapless_audio && !mpctx->encode_lavc_ctx)
uninit_audio_out(mpctx);
mpctx->playback_initialized = false;
player: make playback termination asynchronous Until now, stopping playback aborted the demuxer and I/O layer violently by signaling mp_cancel (bound to libavformat's AVIOInterruptCB mechanism). Change it to try closing them gracefully. The main purpose is to silence those libavformat errors that happen when you request termination. Most of libavformat barely cares about the termination mechanism (AVIOInterruptCB), and essentially it's like the network connection is abruptly severed, or file I/O suddenly returns I/O errors. There were issues with dumb TLS warnings, parsers complaining about incomplete data, and some special protocols that require server communication to gracefully disconnect. We still want to abort it forcefully if it refuses to terminate on its own, so a timeout is required. Users can set the timeout to 0, which should give them the old behavior. This also removes the old mechanism that treats certain commands (like "quit") specially, and tries to terminate the demuxers even if the core is currently frozen. This is for situations where the core synchronized to the demuxer or stream layer while network is unresponsive. This in turn can only happen due to the "program" or "cache-size" properties in the current code (see one of the previous commits). Also, the old mechanism doesn't fit particularly well with the new one. We wouldn't want to abort playback immediately on a "quit" command - the new code is all about giving it a chance to end it gracefully. We'd need some sort of watchdog thread or something equally complicated to handle this. So just remove it. The change in osd.c is to prevent that it clears the status line while waiting for termination. The normal status line code doesn't output anything useful at this point, and the code path taken clears it, both of which is an annoying behavior change, so just let it show the old one.
2018-05-19 16:41:13 +00:00
uninit_demuxer(mpctx);
// Possibly stop ongoing async commands.
mp_abort_playback_async(mpctx);
m_config_restore_backups(mpctx->mconfig);
TA_FREEP(&mpctx->filter_root);
talloc_free(mpctx->filtered_tags);
mpctx->filtered_tags = NULL;
mp_notify(mpctx, MPV_EVENT_TRACKS_CHANGED, NULL);
bool nothing_played = !mpctx->shown_aframes && !mpctx->shown_vframes &&
mpctx->error_playing <= 0;
struct mpv_event_end_file end_event = {0};
switch (mpctx->stop_play) {
case PT_ERROR:
case AT_END_OF_FILE:
{
if (mpctx->error_playing == 0 && nothing_played)
mpctx->error_playing = MPV_ERROR_NOTHING_TO_PLAY;
if (mpctx->error_playing < 0) {
end_event.error = mpctx->error_playing;
end_event.reason = MPV_END_FILE_REASON_ERROR;
} else if (mpctx->error_playing == 2) {
end_event.reason = MPV_END_FILE_REASON_REDIRECT;
} else {
end_event.reason = MPV_END_FILE_REASON_EOF;
}
if (mpctx->playing) {
// Played/paused for longer than 1 second -> ok
mpctx->playing->playback_short =
playback_start < 0 || mp_time_sec() - playback_start < 1.0;
mpctx->playing->init_failed = nothing_played;
}
break;
}
// Note that error_playing is meaningless in these cases.
case PT_NEXT_ENTRY:
case PT_CURRENT_ENTRY:
case PT_STOP: end_event.reason = MPV_END_FILE_REASON_STOP; break;
case PT_QUIT: end_event.reason = MPV_END_FILE_REASON_QUIT; break;
};
mp_notify(mpctx, MPV_EVENT_END_FILE, &end_event);
MP_VERBOSE(mpctx, "finished playback, %s (reason %d)\n",
mpv_error_string(end_event.error), end_event.reason);
if (end_event.error == MPV_ERROR_UNKNOWN_FORMAT)
MP_ERR(mpctx, "Failed to recognize file format.\n");
MP_INFO(mpctx, "\n");
if (mpctx->playing)
playlist_entry_unref(mpctx->playing);
mpctx->playing = NULL;
talloc_free(mpctx->filename);
mpctx->filename = NULL;
mpctx->stream_open_filename = NULL;
if (end_event.error < 0 && nothing_played) {
mpctx->files_broken++;
} else if (end_event.error < 0) {
mpctx->files_errored++;
} else {
mpctx->files_played++;
}
assert(mpctx->stop_play);
}
// Determine the next file to play. Note that if this function returns non-NULL,
// it can have side-effects and mutate mpctx.
// direction: -1 (previous) or +1 (next)
// force: if true, don't skip playlist entries marked as failed
// mutate: if true, change loop counters
struct playlist_entry *mp_next_file(struct MPContext *mpctx, int direction,
bool force, bool mutate)
{
struct playlist_entry *next = playlist_get_next(mpctx->playlist, direction);
if (next && direction < 0 && !force) {
// Don't jump to files that would immediately go to next file anyway
while (next && next->playback_short)
next = next->prev;
// Always allow jumping to first file
if (!next && mpctx->opts->loop_times == 1)
next = mpctx->playlist->first;
}
if (!next && mpctx->opts->loop_times != 1) {
if (direction > 0) {
if (mpctx->opts->shuffle)
playlist_shuffle(mpctx->playlist);
next = mpctx->playlist->first;
if (next && mpctx->opts->loop_times > 1)
mpctx->opts->loop_times--;
} else {
next = mpctx->playlist->last;
// Don't jump to files that would immediately go to next file anyway
while (next && next->playback_short)
next = next->prev;
}
bool ignore_failures = mpctx->opts->loop_times == -2;
if (!force && next && next->init_failed && !ignore_failures) {
// Don't endless loop if no file in playlist is playable
bool all_failed = true;
struct playlist_entry *cur;
for (cur = mpctx->playlist->first; cur; cur = cur->next) {
all_failed &= cur->init_failed;
if (!all_failed)
break;
}
if (all_failed)
next = NULL;
}
}
return next;
}
// Play all entries on the playlist, starting from the current entry.
// Return if all done.
void mp_play_files(struct MPContext *mpctx)
{
scripting: change when/how player waits for scripts being loaded Fundamentally, scripts are loaded asynchronously, but as a feature, there was code to wait until a script is loaded (for a certain arbitrary definition of "loaded"). This was done in scripting.c with the wait_loaded() function. This called mp_idle(), and since there are commands to load/unload scripts, it meant the player core loop could be entered recursively. I think this is a major complication and has some problems. For example, if you had a script that does 'os.execute("sleep inf")', then every time you ran a command to load an instance of the script would add a new stack frame of mp_idle(). This would lead to some sort of reentrancy horror that is hard to debug. Also misc/dispatch.c contains a somewhat tricky mess to support such recursive invocations. There were also some bugs due to this and due to unforeseen interactions with other messes. This scripting stuff was the only thing making use of that reentrancy, and future commands that have "logical" waiting for something should be implemented differently. So get rid of it. Change the code to wait only in the player initialization phase: the only place where it really has to wait is before playback is started, because scripts might want to set options or hooks that interact with playback initialization. Unloading of builtin scripts (can happen with e.g. "set osc no") is left asynchronous; the unloading wasn't too robust anyway, and this change won't make a difference if someone is trying to break it intentionally. Note that this is not in mp_initialize(), because mpv_initialize() uses this by locking the core, which would have the same problem. In the future, commands which logically wait should use different mechanisms. Originally I thought the current approach (that is removed with this commit) should be used, but it's too much of a mess and can't even be used in some cases. Examples are: - "loadfile" should be made blocking (needs to run the normal player code and manually unblock the thread issuing the command) - "add-sub" should not freeze the player until the URL is opened (needs to run opening on a separate thread) Possibly the current scripting behavior could be restored once new mechanisms exist, and if it turns out that anyone needs it. With this commit there should be no further instances of recursive playloop invocations (other than the case in the following commit), since all mp_idle()/mp_wait_events() calls are done strictly from the main thread (and not commands/properties or libmpv client API that "lock" the main thread).
2018-04-15 08:14:00 +00:00
// Wait for all scripts to load before possibly starting playback.
if (!mp_clients_all_initialized(mpctx)) {
MP_VERBOSE(mpctx, "Waiting for scripts...\n");
while (!mp_clients_all_initialized(mpctx))
mp_idle(mpctx);
mp_wakeup_core(mpctx); // avoid lost wakeups during waiting
MP_VERBOSE(mpctx, "Done loading scripts.\n");
}
prepare_playlist(mpctx, mpctx->playlist);
for (;;) {
assert(mpctx->stop_play);
idle_loop(mpctx);
if (mpctx->stop_play == PT_QUIT)
break;
play_current_file(mpctx);
if (mpctx->stop_play == PT_QUIT)
break;
struct playlist_entry *new_entry = mpctx->playlist->current;
if (mpctx->stop_play == PT_NEXT_ENTRY || mpctx->stop_play == PT_ERROR ||
mpctx->stop_play == AT_END_OF_FILE || mpctx->stop_play == PT_STOP)
{
new_entry = mp_next_file(mpctx, +1, false, true);
}
mpctx->playlist->current = new_entry;
mpctx->playlist->current_was_replaced = false;
mpctx->stop_play = PT_STOP;
if (!mpctx->playlist->current && mpctx->opts->player_idle_mode < 2)
break;
}
cancel_open(mpctx);
if (mpctx->encode_lavc_ctx) {
// Make sure all streams get finished.
uninit_audio_out(mpctx);
uninit_video_out(mpctx);
if (!encode_lavc_free(mpctx->encode_lavc_ctx))
mpctx->stop_play = PT_ERROR;
mpctx->encode_lavc_ctx = NULL;
}
}
// Abort current playback and set the given entry to play next.
// e must be on the mpctx->playlist.
void mp_set_playlist_entry(struct MPContext *mpctx, struct playlist_entry *e)
{
assert(!e || playlist_entry_to_index(mpctx->playlist, e) >= 0);
mpctx->playlist->current = e;
mpctx->playlist->current_was_replaced = false;
// Make it pick up the new entry.
if (!mpctx->stop_play)
mpctx->stop_play = PT_CURRENT_ENTRY;
mp_wakeup_core(mpctx);
}
static void set_track_recorder_sink(struct track *track,
struct mp_recorder_sink *sink)
{
if (track->d_sub)
sub_set_recorder_sink(track->d_sub, sink);
if (track->dec)
track->dec->recorder_sink = sink;
track->remux_sink = sink;
}
void close_recorder(struct MPContext *mpctx)
{
if (!mpctx->recorder)
return;
for (int n = 0; n < mpctx->num_tracks; n++)
set_track_recorder_sink(mpctx->tracks[n], NULL);
mp_recorder_destroy(mpctx->recorder);
mpctx->recorder = NULL;
}
// Like close_recorder(), but also unset the option. Intended for use on errors.
void close_recorder_and_error(struct MPContext *mpctx)
{
close_recorder(mpctx);
talloc_free(mpctx->opts->record_file);
mpctx->opts->record_file = NULL;
mp_notify_property(mpctx, "record-file");
MP_ERR(mpctx, "Disabling stream recording.\n");
}
void open_recorder(struct MPContext *mpctx, bool on_init)
{
if (!mpctx->playback_initialized)
return;
close_recorder(mpctx);
char *target = mpctx->opts->record_file;
if (!target || !target[0])
return;
struct sh_stream **streams = NULL;
int num_streams = 0;
for (int n = 0; n < mpctx->num_tracks; n++) {
struct track *track = mpctx->tracks[n];
if (track->stream && track->selected && (track->d_sub || track->dec))
MP_TARRAY_APPEND(NULL, streams, num_streams, track->stream);
}
mpctx->recorder = mp_recorder_create(mpctx->global, mpctx->opts->record_file,
streams, num_streams);
if (!mpctx->recorder) {
talloc_free(streams);
close_recorder_and_error(mpctx);
return;
}
if (!on_init)
mp_recorder_mark_discontinuity(mpctx->recorder);
int n_stream = 0;
for (int n = 0; n < mpctx->num_tracks; n++) {
struct track *track = mpctx->tracks[n];
if (n_stream >= num_streams)
break;
// (We expect track->stream not to be reused on other tracks.)
if (track->stream == streams[n_stream]) {
struct mp_recorder_sink * sink =
mp_recorder_get_sink(mpctx->recorder, streams[n_stream]);
assert(sink);
set_track_recorder_sink(track, sink);
n_stream++;
}
}
talloc_free(streams);
}