1
0
mirror of https://github.com/mpv-player/mpv synced 2024-12-28 18:12:22 +00:00
mpv/player/core.h

645 lines
23 KiB
C
Raw Normal View History

/*
* 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/>.
*/
#ifndef MPLAYER_MP_CORE_H
#define MPLAYER_MP_CORE_H
2015-07-06 20:28:28 +00:00
#include <pthread.h>
#include <stdatomic.h>
#include <stdbool.h>
#include "libmpv/client.h"
#include "common/common.h"
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
#include "filters/filter.h"
#include "filters/f_output_chain.h"
#include "options/options.h"
#include "sub/osd.h"
audio: introduce a new type to hold audio frames This is pretty pointless, but I believe it allows us to claim that the new code is not affected by the copyright of the old code. This is needed, because the original mp_audio struct was written by someone who has disagreed with LGPL relicensing (it was called af_data at the time, and was defined in af.h). The "GPL'ed" struct contents that surive are pretty trivial: just the data pointer, and some metadata like the format, samplerate, etc. - but at least in this case, any new code would be extremely similar anyway, and I'm not really sure whether it's OK to claim different copyright. So what we do is we just use AVFrame (which of course is LGPL with 100% certainty), and add some accessors around it to adapt it to mpv conventions. Also, this gets rid of some annoying conventions of mp_audio, like the struct fields that require using an accessor to write to them anyway. For the most part, this change is only dumb replacements of mp_audio related functions and fields. One minor actual change is that you can't allocate the new type on the stack anymore. Some code still uses mp_audio. All audio filter code will be deleted, so it makes no sense to convert this code. (Audio filters which are LGPL and which we keep will have to be ported to a new filter infrastructure anyway.) player/audio.c uses it because it interacts with the old filter code. push.c has some complex use of mp_audio and mp_audio_buffer, but this and pull.c will most likely be rewritten to do something else.
2017-08-16 19:00:20 +00:00
#include "audio/aframe.h"
#include "video/mp_image.h"
#include "video/out/vo.h"
// definitions used internally by the core player code
enum stop_play_reason {
KEEP_PLAYING = 0, // playback of a file is actually going on
// must be 0, numeric values of others do not matter
AT_END_OF_FILE, // file has ended, prepare to play next
// also returned on unrecoverable playback errors
mplayer: turn playtree into a list, and change per-file option handling Summary: - There is no playtree anymore. It's reduced to a simple list. - Options are now always global. You can still have per-file options, but these are optional and require special syntax. - The slave command pt_step has been removed, and playlist_next and playlist_prev added. (See etc/input.conf changes.) This is a user visible incompatible change, and will break slave-mode applications. - The pt_clear slave command is renamed to playlist_clear. - Playtree entries could have multiple files. This is not the case anymore, and playlist entries have always exactly one entry. Whenever something adds more than one file (like ASX playlists or dvd:// or dvdnav:// on the command line), all files are added as separate playlist entries. Note that some of the changes are quite deep and violent. Expect regressions. The playlist parsing code in particular is of low quality. I didn't try to improve it, and merely spent to least effort necessary to keep it somehow working. (Especially ASX playlist handling.) The playtree code was complicated and bloated. It was also barely used. Most users don't even know that mplayer manages the playlist as tree, or how to use it. The most obscure features was probably specifying a tree on command line (with '{' and '}' to create/close tree nodes). It filled the player code with complexity and confused users with weird slave commands like pt_up. Replace the playtree with a simple flat playlist. Playlist parsers that actually return trees are changed to append all files to the playlist pre-order. It used to be the responsibility of the playtree code to change per-file config options. Now this is done by the player core, and the playlist code is free of such details. Options are not per-file by default anymore. This was a very obscure and complicated feature that confused even experienced users. Consider the following command line: mplayer file1.mkv file2.mkv --no-audio file3.mkv This will disable the audio for file2.mkv only, because options are per-file by default. To make the option affect all files, you're supposed to put it before the first file. This is bad, because normally you don't need per-file options. They are very rarely needed, and the only reasonable use cases I can imagine are use of the encode backend (mplayer encode branch), or for debugging. The normal use case is made harder, and the feature is perceived as bug. Even worse, correct usage is hard to explain for users. Make all options global by default. The position of an option isn't significant anymore (except for options that compensate each other, consider --shuffle --no-shuffle). One other important change is that no options are reset anymore if a new file is started. If you change settings with slave mode commands, they will not be changed by playing a new file. (Exceptions include settings that are too file specific, like audio/subtitle stream selection.) There is still some need for per-file options. Debugging and encoding are use cases that profit from per-file options. Per-file profiles (as well as per-protocol and per-VO/AO options) need the implementation related mechanisms to backup and restore options when the playback file changes. Simplify the save-slot stuff, which is possible because there is no hierarchical play tree anymore. Now there's a simple backup field. Add a way to specify per-file options on command line. Example: mplayer f1.mkv -o0 --{ -o1 f2.mkv -o2 f3.mkv --} f4.mkv -o3 will have the following options per file set: f1.mkv, f4.mkv: -o0 -o3 f2.mkv, f3.mkv: -o0 -o3 -o1 -o2 The options --{ and --} start and end per-file options. All files inside the { } will be affected by the options equally (similar to how global options and multiple files are handled). When playback of a file starts, the per-file options are set according to the command line. When playback ends, the per-file options are restored to the values when playback started.
2012-07-31 19:33:26 +00:00
PT_NEXT_ENTRY, // prepare to play next entry in playlist
PT_CURRENT_ENTRY, // prepare to play mpctx->playlist->current
player: fix subtle idle mode differences on early program start If the user manages to run a "loadfile x append" command before the loop in mp_play_files() is entered, then the player could start playing these. This isn't expected, because appending files to the playlist in idle mode does not normally start playback. It could happen because there is a short time window where commands are processed before the loop is entered (such as running the command when a script is loaded). The idle mode semantics are pretty weird: if files were provided in advance (on the command line), then these should be played immediately. But if idle mode was already entered, and something is appended to the playlist using "append", i.e. without explicitly triggering playback, then it should remain in idle mode. Try to follow this by redefining PT_STOP to strictly mean idle mode. Remove the playlist->current check from idle_loop(), since only the stop_play field counts now (cf. what mp_set_playlist_entry() does). This actually introduces the possibility that playlist->current, and with it playlist-pos, are set to something, even though playback is not active or being started. Previously, this was only possible during state transitions, such as when changing playlist entries. Very annoyingly, this means the current way MPV_EVENT_IDLE was sent doesn't work anymore. Logically, idle mode can be "active" even if idle_loop() was not entered yet (between the time after mp_initialize() and before the loop in mp_play_files()). Instead of worrying about this, redo the "idle-active" property, and deprecate the event. See: #7543
2020-03-21 13:01:38 +00:00
PT_STOP, // stop playback / idle mode
PT_QUIT, // stop playback, quit player
PT_ERROR, // play next playlist entry (due to an error)
};
enum mp_osd_seek_info {
OSD_SEEK_INFO_BAR = 1,
OSD_SEEK_INFO_TEXT = 2,
OSD_SEEK_INFO_CHAPTER_TEXT = 4,
OSD_SEEK_INFO_CURRENT_FILE = 8,
};
enum {
// other constants
OSD_LEVEL_INVISIBLE = 4,
OSD_BAR_SEEK = 256,
MAX_NUM_VO_PTS = 100,
};
enum seek_type {
MPSEEK_NONE = 0,
MPSEEK_RELATIVE,
MPSEEK_ABSOLUTE,
MPSEEK_FACTOR,
MPSEEK_BACKSTEP,
MPSEEK_CHAPTER,
};
enum seek_precision {
// The following values are numerically sorted by increasing precision
MPSEEK_DEFAULT = 0,
MPSEEK_KEYFRAME,
MPSEEK_EXACT,
MPSEEK_VERY_EXACT,
};
enum seek_flags {
MPSEEK_FLAG_DELAY = 1 << 0, // give player chance to coalesce multiple seeks
MPSEEK_FLAG_NOFLUSH = 1 << 1, // keeping remaining data for seamless loops
};
struct seek_params {
enum seek_type type;
enum seek_precision exact;
double amount;
unsigned flags; // MPSEEK_FLAG_*
};
// Information about past video frames that have been sent to the VO.
struct frame_info {
double pts;
double duration; // PTS difference to next frame
double approx_duration; // possibly fixed/smoothed out duration
double av_diff; // A/V diff at time of scheduling
int num_vsyncs; // scheduled vsyncs, if using display-sync
};
struct track {
enum stream_type type;
// Currently used for decoding.
bool selected;
// The type specific ID, also called aid (audio), sid (subs), vid (video).
// For UI purposes only; this ID doesn't have anything to do with any
// IDs coming from demuxers or container files.
int user_tid;
int demuxer_id; // same as stream->demuxer_id. -1 if not set.
int ff_index; // same as stream->ff_index, or 0.
int hls_bitrate; // same as stream->hls_bitrate. 0 if not set.
int program_id; // same as stream->program_id. -1 if not set.
char *title;
bool default_track, forced_track, dependent_track;
bool visual_impaired_track, hearing_impaired_track;
bool image;
bool attached_picture;
char *lang;
// If this track is from an external file (e.g. subtitle file).
bool is_external;
bool no_default; // pretend it's not external for auto-selection
bool no_auto_select;
char *external_filename;
bool auto_loaded;
struct demuxer *demuxer;
// Invariant: !stream || stream->demuxer == demuxer
struct sh_stream *stream;
// Current subtitle state (or cached state if selected==false).
struct dec_sub *d_sub;
// Current decoding state (NULL if selected==false)
struct mp_decoder_wrapper *dec;
// Where the decoded result goes to (one of them is not NULL if active)
struct vo_chain *vo_c;
struct ao_chain *ao_c;
struct mp_pin *sink;
};
// Summarizes video filtering and output.
struct vo_chain {
struct mp_log *log;
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
struct mp_output_chain *filter;
struct vo *vo;
struct track *track;
struct mp_pin *filter_src;
struct mp_pin *dec_src;
// - video consists of a single picture, which should be shown only once
// - do not sync audio to video in any way
bool is_coverart;
// - video consists of sparse still images
bool is_sparse;
bool sparse_eof_signalled;
player: partially rework --cache-pause The --cache-pause feature (enabled by default) will pause playback for a while if network runs out of data. If this is not done, then playback will go on frame-wise (as packets are slowly read from the network and then instantly decoded and displayed). This feature is actually useless, as you won't get nice playback no matter what if network is too slow, but I guess I still prefer this behavior for some reason. This commit changes this behavior from using the demuxer cache state only, to trying to use underrun information from the AO/VO. This means if you have a very large audio buffer, then cache-pausing will trigger once that buffer is depleted, which will be some time _after_ the demuxer cache has run out. This requires explicit support from the AO. Otherwise, the behavior should be mostly the same as before this commit. This does not care about the AO buffer. In theory, the AO may underrun, then the player will write some data to the AO buffer, then the AO will recover and play this bit of data, then the player will probably trigger the cache-pause behavior. The probability of this happening should be pretty low, so I will hold off fixing this until the next refactor of the AO chain (if ever). The VO underflow detection was devised and tested in 5 minutes, and may not be correct. At least I'm fairly sure that the combination of all the factors should make incorrect behavior relatively unlikely, but problems are possible. Also, the demux_reader_state.underrun field may be inaccurate. It's only the present state at the time demux_get_reader_state() was called, and may exclude past underruns. In theory, this could cause "close" cases to be missed. Then you might get an audio underrun without cache-pausing acting on it. If the stars align, this could happen multiple times in the row, effectively making this feature not work. The most user-visible consequence of this change is that the user will now see an AO underrun warning every time the cache runs out. Maybe this cache-pause feature should just be removed...
2019-10-11 17:34:04 +00:00
bool underrun;
bool underrun_signaled;
};
// Like vo_chain, for audio.
struct ao_chain {
struct mp_log *log;
struct MPContext *mpctx;
bool spdif_passthrough, spdif_failed;
struct mp_output_chain *filter;
struct ao *ao;
struct mp_async_queue *ao_queue;
struct mp_filter *queue_filter;
struct mp_filter *ao_filter;
double ao_resume_time;
bool out_eof;
double last_out_pts;
double start_pts;
bool start_pts_known;
struct track *track;
struct mp_pin *filter_src;
struct mp_pin *dec_src;
player: partially rework --cache-pause The --cache-pause feature (enabled by default) will pause playback for a while if network runs out of data. If this is not done, then playback will go on frame-wise (as packets are slowly read from the network and then instantly decoded and displayed). This feature is actually useless, as you won't get nice playback no matter what if network is too slow, but I guess I still prefer this behavior for some reason. This commit changes this behavior from using the demuxer cache state only, to trying to use underrun information from the AO/VO. This means if you have a very large audio buffer, then cache-pausing will trigger once that buffer is depleted, which will be some time _after_ the demuxer cache has run out. This requires explicit support from the AO. Otherwise, the behavior should be mostly the same as before this commit. This does not care about the AO buffer. In theory, the AO may underrun, then the player will write some data to the AO buffer, then the AO will recover and play this bit of data, then the player will probably trigger the cache-pause behavior. The probability of this happening should be pretty low, so I will hold off fixing this until the next refactor of the AO chain (if ever). The VO underflow detection was devised and tested in 5 minutes, and may not be correct. At least I'm fairly sure that the combination of all the factors should make incorrect behavior relatively unlikely, but problems are possible. Also, the demux_reader_state.underrun field may be inaccurate. It's only the present state at the time demux_get_reader_state() was called, and may exclude past underruns. In theory, this could cause "close" cases to be missed. Then you might get an audio underrun without cache-pausing acting on it. If the stars align, this could happen multiple times in the row, effectively making this feature not work. The most user-visible consequence of this change is that the user will now see an AO underrun warning every time the cache runs out. Maybe this cache-pause feature should just be removed...
2019-10-11 17:34:04 +00:00
double delay;
bool untimed_throttle;
bool ao_underrun; // last known AO state
bool underrun; // for cache pause logic
};
/* Note that playback can be paused, stopped, etc. at any time. While paused,
* playback restart is still active, because you want seeking to work even
* if paused.
* The main purpose of distinguishing these states is proper reinitialization
* of A/V sync.
*/
enum playback_status {
// code may compare status values numerically
STATUS_SYNCING, // seeking for a position to resume
STATUS_READY, // buffers full, playback can be started any time
STATUS_PLAYING, // normal playback
STATUS_DRAINING, // decoding has ended; still playing out queued buffers
STATUS_EOF, // playback has ended, or is disabled
};
const char *mp_status_str(enum playback_status st);
extern const int num_ptracks[STREAM_TYPE_COUNT];
// Maximum of all num_ptracks[] values.
#define MAX_PTRACKS 2
typedef struct MPContext {
bool initialized;
bool is_cli;
struct mpv_global *global;
struct MPOpts *opts;
struct mp_log *log;
struct stats_ctx *stats;
struct m_config *mconfig;
struct input_ctx *input;
struct mp_client_api *clients;
struct mp_dispatch_queue *dispatch;
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
struct mp_cancel *playback_abort;
// Number of asynchronous tasks that still need to finish until MPContext
// destruction is ok. It's implied that the async tasks call
// mp_wakeup_core() each time this is decremented.
// As using an atomic+wakeup would be racy, this is a normal integer, and
// mp_dispatch_lock must be called to change it.
int64_t outstanding_async;
struct mp_thread_pool *thread_pool; // for coarse I/O, often during loading
struct mp_log *statusline;
struct osd_state *osd;
player: redo terminal OSD and status line handling The terminal OSD code includes the handling of the terminal status line, showing player OSD messages on the terminal, and showing subtitles on terminal (the latter two only if there is no video window, or if terminal OSD is forced). This didn't handle some corner cases correctly. For example, showing an OSD message on the terminal always cleared the previous line, even if the line was an important message (or even just the command prompt, if most other messages were silenced). Attempt to handle this correctly by keeping track of how many lines the terminal OSD currently consists of. Since there could be race conditions with other messages being printed, implement this in msg.c. Now msg.c expects that MSGL_STATUS messages rewrite the status line, so the caller is forced to use a single mp_msg() call to set the status line. Instead of littering print_status() all over the place, update the status only once per playloop iteration in update_osd_msg(). In audio- only mode, the status line might now be a little bit off, but it's perhaps ok. Print the status line only if it has changed, or if another message was printed. This might help with extremely slow terminals, although in audio+video mode, it'll still be updated very often (A-V sync display changes on every frame). Instead of hardcoding the terminal sequences, use terminfo/termcap to get the sequences. Remove the --term-osd-esc option, which allowed to override the hardcoded escapes - it's useless now. The fallback for terminals with no escape sequences for moving the cursor and clearing a line is removed. This somewhat breaks status line display on these terminals, including the MS Windows console: instead of querying the terminal size and clearing the line manually by padding the output with spaces, the line is simply not cleared. I don't expect this to be a problem on UNIX, and on MS Windows we could emulate escape sequences. Note that terminal OSD (other than the status line) was broken anyway on these terminals. In osd.c, the function get_term_width() is not used anymore, so remove it. To remind us that the MS Windows console apparently adds a line break when writint the last column, adjust screen_width in terminal- win.c accordingly.
2014-01-13 19:05:41 +00:00
char *term_osd_text;
char *term_osd_status;
char *term_osd_subs;
player: redo terminal OSD and status line handling The terminal OSD code includes the handling of the terminal status line, showing player OSD messages on the terminal, and showing subtitles on terminal (the latter two only if there is no video window, or if terminal OSD is forced). This didn't handle some corner cases correctly. For example, showing an OSD message on the terminal always cleared the previous line, even if the line was an important message (or even just the command prompt, if most other messages were silenced). Attempt to handle this correctly by keeping track of how many lines the terminal OSD currently consists of. Since there could be race conditions with other messages being printed, implement this in msg.c. Now msg.c expects that MSGL_STATUS messages rewrite the status line, so the caller is forced to use a single mp_msg() call to set the status line. Instead of littering print_status() all over the place, update the status only once per playloop iteration in update_osd_msg(). In audio- only mode, the status line might now be a little bit off, but it's perhaps ok. Print the status line only if it has changed, or if another message was printed. This might help with extremely slow terminals, although in audio+video mode, it'll still be updated very often (A-V sync display changes on every frame). Instead of hardcoding the terminal sequences, use terminfo/termcap to get the sequences. Remove the --term-osd-esc option, which allowed to override the hardcoded escapes - it's useless now. The fallback for terminals with no escape sequences for moving the cursor and clearing a line is removed. This somewhat breaks status line display on these terminals, including the MS Windows console: instead of querying the terminal size and clearing the line manually by padding the output with spaces, the line is simply not cleared. I don't expect this to be a problem on UNIX, and on MS Windows we could emulate escape sequences. Note that terminal OSD (other than the status line) was broken anyway on these terminals. In osd.c, the function get_term_width() is not used anymore, so remove it. To remind us that the MS Windows console apparently adds a line break when writint the last column, adjust screen_width in terminal- win.c accordingly.
2014-01-13 19:05:41 +00:00
char *term_osd_contents;
char *term_osd_title;
char *last_window_title;
struct voctrl_playback_state vo_playback_state;
int add_osd_seek_info; // bitfield of enum mp_osd_seek_info
double osd_visible; // for the osd bar only
int osd_function;
double osd_function_visible;
double osd_msg_visible;
double osd_msg_next_duration;
double osd_last_update;
bool osd_force_update, osd_idle_update;
char *osd_msg_text;
bool osd_show_pos;
struct osd_progbar_state osd_progbar;
mplayer: turn playtree into a list, and change per-file option handling Summary: - There is no playtree anymore. It's reduced to a simple list. - Options are now always global. You can still have per-file options, but these are optional and require special syntax. - The slave command pt_step has been removed, and playlist_next and playlist_prev added. (See etc/input.conf changes.) This is a user visible incompatible change, and will break slave-mode applications. - The pt_clear slave command is renamed to playlist_clear. - Playtree entries could have multiple files. This is not the case anymore, and playlist entries have always exactly one entry. Whenever something adds more than one file (like ASX playlists or dvd:// or dvdnav:// on the command line), all files are added as separate playlist entries. Note that some of the changes are quite deep and violent. Expect regressions. The playlist parsing code in particular is of low quality. I didn't try to improve it, and merely spent to least effort necessary to keep it somehow working. (Especially ASX playlist handling.) The playtree code was complicated and bloated. It was also barely used. Most users don't even know that mplayer manages the playlist as tree, or how to use it. The most obscure features was probably specifying a tree on command line (with '{' and '}' to create/close tree nodes). It filled the player code with complexity and confused users with weird slave commands like pt_up. Replace the playtree with a simple flat playlist. Playlist parsers that actually return trees are changed to append all files to the playlist pre-order. It used to be the responsibility of the playtree code to change per-file config options. Now this is done by the player core, and the playlist code is free of such details. Options are not per-file by default anymore. This was a very obscure and complicated feature that confused even experienced users. Consider the following command line: mplayer file1.mkv file2.mkv --no-audio file3.mkv This will disable the audio for file2.mkv only, because options are per-file by default. To make the option affect all files, you're supposed to put it before the first file. This is bad, because normally you don't need per-file options. They are very rarely needed, and the only reasonable use cases I can imagine are use of the encode backend (mplayer encode branch), or for debugging. The normal use case is made harder, and the feature is perceived as bug. Even worse, correct usage is hard to explain for users. Make all options global by default. The position of an option isn't significant anymore (except for options that compensate each other, consider --shuffle --no-shuffle). One other important change is that no options are reset anymore if a new file is started. If you change settings with slave mode commands, they will not be changed by playing a new file. (Exceptions include settings that are too file specific, like audio/subtitle stream selection.) There is still some need for per-file options. Debugging and encoding are use cases that profit from per-file options. Per-file profiles (as well as per-protocol and per-VO/AO options) need the implementation related mechanisms to backup and restore options when the playback file changes. Simplify the save-slot stuff, which is possible because there is no hierarchical play tree anymore. Now there's a simple backup field. Add a way to specify per-file options on command line. Example: mplayer f1.mkv -o0 --{ -o1 f2.mkv -o2 f3.mkv --} f4.mkv -o3 will have the following options per file set: f1.mkv, f4.mkv: -o0 -o3 f2.mkv, f3.mkv: -o0 -o3 -o1 -o2 The options --{ and --} start and end per-file options. All files inside the { } will be affected by the options equally (similar to how global options and multiple files are handled). When playback of a file starts, the per-file options are set according to the command line. When playback ends, the per-file options are restored to the values when playback started.
2012-07-31 19:33:26 +00:00
struct playlist *playlist;
struct playlist_entry *playing; // currently playing file
char *filename; // immutable copy of playing->filename (or NULL)
char *stream_open_filename;
char **playlist_paths; // used strictly for playlist validation
int playlist_paths_len;
enum stop_play_reason stop_play;
bool playback_initialized; // playloop can be run/is running
int error_playing;
// Return code to use with PT_QUIT
2013-08-02 08:32:38 +00:00
int quit_custom_rc;
bool has_quit_custom_rc;
// Global file statistics
int files_played; // played without issues (even if stopped by user)
int files_errored; // played, but errors happened at one point
int files_broken; // couldn't be played at all
// Current file statistics
int64_t shown_vframes, shown_aframes;
struct demux_chapter *chapters;
int num_chapters;
struct demuxer *demuxer;
struct mp_tags *filtered_tags;
struct track **tracks;
int num_tracks;
char *track_layout_hash;
// Selected tracks. NULL if no track selected.
// There can be num_ptracks[type] of the same STREAM_TYPE selected at once.
// Currently, this is used for the secondary subtitle track only.
struct track *current_track[MAX_PTRACKS][STREAM_TYPE_COUNT];
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
struct mp_filter *filter_root;
struct mp_filter *lavfi;
char *lavfi_graph;
struct ao *ao;
struct mp_aframe *ao_filter_fmt; // for weak gapless audio check
struct ao_chain *ao_chain;
struct vo_chain *vo_chain;
struct vo *video_out;
// next_frame[0] is the next frame, next_frame[1] the one after that.
// The +1 is for adding 1 additional frame in backstep mode.
struct mp_image *next_frames[VO_MAX_REQ_FRAMES + 1];
int num_next_frames;
struct mp_image *saved_frame; // for hrseek_lastframe and hrseek_backstep
enum playback_status video_status, audio_status;
bool restart_complete;
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
int play_dir;
// Factors to multiply with opts->playback_speed to get the total audio or
// video speed (usually 1.0, but can be set to by the sync code).
double speed_factor_v, speed_factor_a;
// Redundant values set from opts->playback_speed and speed_factor_*.
// update_playback_speed() updates them from the other fields.
double audio_speed, video_speed;
bool display_sync_active;
int display_sync_drift_dir;
// Timing error (in seconds) due to rounding on vsync boundaries
double display_sync_error;
// Number of mistimed frames.
int mistimed_frames_total;
bool hrseek_active; // skip all data until hrseek_pts
bool hrseek_lastframe; // drop everything until last frame reached
bool hrseek_backstep; // go to frame before seek target
double hrseek_pts;
struct seek_params current_seek;
bool ab_loop_clip; // clip to the "b" part of an A-B loop if available
// AV sync: the next frame should be shown when the audio out has this
// much (in seconds) buffered data left. Increased when more data is
// written to the ao, decreased when moving to the next video frame.
double delay;
// AV sync: time in seconds until next frame should be shown
double time_frame;
// How much video timing has been changed to make it match the audio
// timeline. Used for status line information only.
double total_avsync_change;
// A-V sync difference when last frame was displayed. Kept to display
// the same value if the status line is updated at a time where no new
// video frame is shown.
double last_av_difference;
/* timestamp of video frame currently visible on screen
* (or at least queued to be flipped by VO) */
double video_pts;
// Last seek target.
double last_seek_pts;
// Frame duration field from demuxer. Only used for duration of the last
// video frame.
double last_frame_duration;
// Video PTS, or audio PTS if video has ended.
double playback_pts;
// For logging only.
double logged_async_diff;
int last_chapter;
// Past timestamps etc.
// The newest frame is at index 0.
struct frame_info *past_frames;
int num_past_frames;
core: add backstep support Allows stepping back one frame via the frame_back_step inout command, bound to "," by default. This uses the precise seeking facility, and a perfect frame index built on the fly. The index is built during playback and precise seeking, and contains (as of this commit) the last 100 displayed or skipped frames. This index is used to find the PTS of the previous frame, which is then used as target for a precise seek. If no PTS is found, the core attempts to do a seek before the current frame, and skip decoded frames until the current frame is reached; this will create a sufficient index and the normal backstep algorithm can be applied. This can be rather slow. The worst case for backstepping is about the same as the worst case for precise seeking if the previous frame can be deduced from the index. If not, the worst case will be twice as slow. There's also some minor danger that the index is incorrect in case framedropping is involved. For framedropping due to --framedrop, this problem is ignored (use of --framedrop is discouraged anyway). For framedropping during precise seeking (done to make it faster), we try to not add frames to the index that are produced when this can happen. I'm not sure how well that works (or if the logic is sane), and it's sure to break with some video filters. In the worst case, backstepping might silently skip frames if you backstep after a user-initiated precise seek. (Precise seeks to do indexing are not affected.) Likewise, video filters that somehow change timing of frames and do not do this in a deterministic way (i.e. if you seek to a position, frames with different timings are produced than when the position is reached during normal playback) will make backstepping silently jump to the wrong frame. Enabling/disabling filters during playback (like for example deinterlacing) will have similar bad effects.
2013-04-24 17:31:48 +00:00
double last_idle_tick;
double next_cache_update;
double sleeptime; // number of seconds to sleep before next iteration
double mouse_timer;
unsigned int mouse_event_ts;
bool mouse_cursor_visible;
// used to prevent hanging in some error cases
double start_timestamp;
// Timestamp from the last time some timing functions read the
// current time, in nanoseconds.
// Used to turn a new time value to a delta from last time.
int64_t last_time;
struct seek_params seek;
/* Heuristic for relative chapter seeks: keep track which chapter
* the user wanted to go to, even if we aren't exactly within the
* boundaries of that chapter due to an inaccurate seek. */
int last_chapter_seek;
bool last_chapter_flag;
bool paused; // internal pause state
bool playback_active; // not paused, restarting, loading, unloading
bool in_playloop;
// step this many frames, then pause
int step_frames;
// Counted down each frame, stop playback if 0 is reached. (-1 = disable)
int max_frames;
bool playing_msg_shown;
bool paused_for_cache;
player: partially rework --cache-pause The --cache-pause feature (enabled by default) will pause playback for a while if network runs out of data. If this is not done, then playback will go on frame-wise (as packets are slowly read from the network and then instantly decoded and displayed). This feature is actually useless, as you won't get nice playback no matter what if network is too slow, but I guess I still prefer this behavior for some reason. This commit changes this behavior from using the demuxer cache state only, to trying to use underrun information from the AO/VO. This means if you have a very large audio buffer, then cache-pausing will trigger once that buffer is depleted, which will be some time _after_ the demuxer cache has run out. This requires explicit support from the AO. Otherwise, the behavior should be mostly the same as before this commit. This does not care about the AO buffer. In theory, the AO may underrun, then the player will write some data to the AO buffer, then the AO will recover and play this bit of data, then the player will probably trigger the cache-pause behavior. The probability of this happening should be pretty low, so I will hold off fixing this until the next refactor of the AO chain (if ever). The VO underflow detection was devised and tested in 5 minutes, and may not be correct. At least I'm fairly sure that the combination of all the factors should make incorrect behavior relatively unlikely, but problems are possible. Also, the demux_reader_state.underrun field may be inaccurate. It's only the present state at the time demux_get_reader_state() was called, and may exclude past underruns. In theory, this could cause "close" cases to be missed. Then you might get an audio underrun without cache-pausing acting on it. If the stars align, this could happen multiple times in the row, effectively making this feature not work. The most user-visible consequence of this change is that the user will now see an AO underrun warning every time the cache runs out. Maybe this cache-pause feature should just be removed...
2019-10-11 17:34:04 +00:00
bool demux_underrun;
double cache_stop_time;
int cache_buffer;
double cache_update_pts;
// Set after showing warning about decoding being too slow for realtime
// playback rate. Used to avoid showing it multiple times.
bool drop_message_shown;
struct screenshot_ctx *screenshot_ctx;
struct command_ctx *command_ctx;
struct encode_lavc_context *encode_lavc_ctx;
struct mp_ipc_ctx *ipc_ctx;
int64_t builtin_script_ids[5];
pthread_mutex_t abort_lock;
// --- The following fields are protected by abort_lock
struct mp_abort_entry **abort_list;
int num_abort_list;
bool abort_all; // during final termination
// --- Owned by MPContext
pthread_t open_thread;
bool open_active; // open_thread is a valid thread handle, all setup
atomic_bool open_done;
// --- All fields below are immutable while open_active is true.
// Otherwise, they're owned by MPContext.
struct mp_cancel *open_cancel;
char *open_url;
char *open_format;
int open_url_flags;
bool open_for_prefetch;
// --- All fields below are owned by open_thread, unless open_done was set
// to true.
struct demuxer *open_res_demuxer;
int open_res_error;
} MPContext;
// Contains information about an asynchronous work item, how it can be aborted,
// and when. All fields are protected by MPContext.abort_lock.
struct mp_abort_entry {
// General conditions.
bool coupled_to_playback; // trigger when playback is terminated
// Actual trigger to abort the work. Pointer immutable, owner may access
// without holding the abort_lock.
struct mp_cancel *cancel;
// For client API.
struct mpv_handle *client; // non-NULL if done by a client API user
int client_work_type; // client API type, e.h. MPV_EVENT_COMMAND_REPLY
uint64_t client_work_id; // client API user reply_userdata value
// (only valid if client_work_type set)
};
// audio.c
void reset_audio_state(struct MPContext *mpctx);
void reinit_audio_chain(struct MPContext *mpctx);
int init_audio_decoder(struct MPContext *mpctx, struct track *track);
int reinit_audio_filters(struct MPContext *mpctx);
double playing_audio_pts(struct MPContext *mpctx);
void fill_audio_out_buffers(struct MPContext *mpctx);
double written_audio_pts(struct MPContext *mpctx);
void clear_audio_output_buffers(struct MPContext *mpctx);
void update_playback_speed(struct MPContext *mpctx);
void uninit_audio_out(struct MPContext *mpctx);
void uninit_audio_chain(struct MPContext *mpctx);
void reinit_audio_chain_src(struct MPContext *mpctx, struct track *track);
void audio_update_volume(struct MPContext *mpctx);
void reload_audio_output(struct MPContext *mpctx);
void audio_start_ao(struct MPContext *mpctx);
// configfiles.c
void mp_parse_cfgfiles(struct MPContext *mpctx);
void mp_load_auto_profiles(struct MPContext *mpctx);
bool mp_load_playback_resume(struct MPContext *mpctx, const char *file);
void mp_write_watch_later_conf(struct MPContext *mpctx);
command: add delete-watch-later-config This introduces the delete-watch-later-config command, to complement write-watch-later-config. This is an alternative to #8141. The general problem that this change is attempting to help solve has been described in #336, #3169 and #6574. Though persistent playback position of a single file is generally a solved problem, this is not the case for playlists, as described in #8138. The motivation is facilitating intermittent playback of very large playlists, consisting of hundreds of entries each many hours long. Though the current "watch later" mechanism works well - provided that the files each occur only once in that playlist, and are played only via that playlist - the biggest issue is that the position is lost completely should mpv exit uncleanly (e.g. due to a power failure). Existing workarounds (in the form of Lua scripts which call write-watch-later-config periodically) fail in the playlist case, due to the mechanism used by mpv to determine where within a playlist to resume playback from. The missing puzzle piece needed to allow scripts to implement a complete solution to this problem is simply a way to clean up the watch-later configuration that the script asked mpv to write using write-watch-later-config. With that in place, scripts can then register an end-file event listener, check the stop playback reason, and in the "eof" and "stop" case, invoke delete-watch-later-config to delete any saved positions written by write-watch-later-config. The script can then proceed to immediately write a new one when the next file is loaded, which altogether allows mpv to resume from the correct playlist and file position upon next startup. Because events are delivered and executed asynchronously, delete-watch-later-config takes an optional filename argument, to allow scripts to clear watch-later configuration for files after mpv had already moved on from playing them and proceeded to another file. A Lua script which makes use of this change can be found here: https://gist.github.com/CyberShadow/2f71a97fb85ed42146f6d9f522bc34ef (A modification of the one written by @Hakkin, in that this one takes advantage of the new command, and also saves the state immediately when a new file is loaded.)
2020-10-22 16:25:20 +00:00
void mp_delete_watch_later_conf(struct MPContext *mpctx, const char *file);
struct playlist_entry *mp_check_playlist_resume(struct MPContext *mpctx,
struct playlist *playlist);
// loadfile.c
void mp_abort_playback_async(struct MPContext *mpctx);
void mp_abort_add(struct MPContext *mpctx, struct mp_abort_entry *abort);
void mp_abort_remove(struct MPContext *mpctx, struct mp_abort_entry *abort);
void mp_abort_recheck_locked(struct MPContext *mpctx,
struct mp_abort_entry *abort);
void mp_abort_trigger_locked(struct MPContext *mpctx,
struct mp_abort_entry *abort);
int mp_add_external_file(struct MPContext *mpctx, char *filename,
enum stream_type filter, struct mp_cancel *cancel,
bool cover_art);
void mark_track_selection(struct MPContext *mpctx, int order,
enum stream_type type, int value);
#define FLAG_MARK_SELECTION 1
void mp_switch_track(struct MPContext *mpctx, enum stream_type type,
struct track *track, int flags);
void mp_switch_track_n(struct MPContext *mpctx, int order,
enum stream_type type, struct track *track, int flags);
void mp_deselect_track(struct MPContext *mpctx, struct track *track);
struct track *mp_track_by_tid(struct MPContext *mpctx, enum stream_type type,
int tid);
void add_demuxer_tracks(struct MPContext *mpctx, struct demuxer *demuxer);
bool mp_remove_track(struct MPContext *mpctx, struct track *track);
struct playlist_entry *mp_next_file(struct MPContext *mpctx, int direction,
bool force);
void mp_set_playlist_entry(struct MPContext *mpctx, struct playlist_entry *e);
void mp_play_files(struct MPContext *mpctx);
void update_demuxer_properties(struct MPContext *mpctx);
void print_track_list(struct MPContext *mpctx, const char *msg);
void reselect_demux_stream(struct MPContext *mpctx, struct track *track,
bool refresh_only);
void prepare_playlist(struct MPContext *mpctx, struct playlist *pl);
void autoload_external_files(struct MPContext *mpctx, struct mp_cancel *cancel);
struct track *select_default_track(struct MPContext *mpctx, int order,
enum stream_type type);
void prefetch_next(struct MPContext *mpctx);
void update_lavfi_complex(struct MPContext *mpctx);
// main.c
int mp_initialize(struct MPContext *mpctx, char **argv);
struct MPContext *mp_create(void);
void mp_destroy(struct MPContext *mpctx);
void mp_print_version(struct mp_log *log, int always);
void mp_update_logging(struct MPContext *mpctx, bool preinit);
void issue_refresh_seek(struct MPContext *mpctx, enum seek_precision min_prec);
// misc.c
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
double rel_time_to_abs(struct MPContext *mpctx, struct m_rel_time t);
double get_play_end_pts(struct MPContext *mpctx);
double get_play_start_pts(struct MPContext *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
bool get_ab_loop_times(struct MPContext *mpctx, double t[2]);
2013-11-19 21:36:33 +00:00
void merge_playlist_files(struct playlist *pl);
void update_content_type(struct MPContext *mpctx, struct track *track);
void update_vo_playback_state(struct MPContext *mpctx);
void update_window_title(struct MPContext *mpctx, bool force);
void error_on_track(struct MPContext *mpctx, struct track *track);
int stream_dump(struct MPContext *mpctx, const char *source_filename);
double get_track_seek_offset(struct MPContext *mpctx, struct track *track);
// osd.c
void set_osd_bar(struct MPContext *mpctx, int type,
double min, double max, double neutral, double val);
bool set_osd_msg(struct MPContext *mpctx, int level, int time,
const char* fmt, ...) PRINTF_ATTRIBUTE(4,5);
void set_osd_function(struct MPContext *mpctx, int osd_function);
void term_osd_set_subs(struct MPContext *mpctx, const char *text);
void get_current_osd_sym(struct MPContext *mpctx, char *buf, size_t buf_size);
void set_osd_bar_chapters(struct MPContext *mpctx, int type);
// playloop.c
void mp_wait_events(struct MPContext *mpctx);
void mp_set_timeout(struct MPContext *mpctx, double sleeptime);
void mp_wakeup_core(struct MPContext *mpctx);
void mp_wakeup_core_cb(void *ctx);
void mp_core_lock(struct MPContext *mpctx);
void mp_core_unlock(struct MPContext *mpctx);
Relicense some non-MPlayer source files to LGPL 2.1 or later This covers source files which were added in mplayer2 and mpv times only, and where all code is covered by LGPL relicensing agreements. There are probably more files to which this applies, but I'm being conservative here. A file named ao_sdl.c exists in MPlayer too, but the mpv one is a complete rewrite, and was added some time after the original ao_sdl.c was removed. The same applies to vo_sdl.c, for which the SDL2 API is radically different in addition (MPlayer supports SDL 1.2 only). common.c contains only code written by me. But common.h is a strange case: although it originally was named mp_common.h and exists in MPlayer too, by now it contains only definitions written by uau and me. The exceptions are the CONTROL_ defines - thus not changing the license of common.h yet. codec_tags.c contained once large tables generated from MPlayer's codecs.conf, but all of these tables were removed. From demux_playlist.c I'm removing a code fragment from someone who was not asked; this probably could be done later (see commit 15dccc37). misc.c is a bit complicated to reason about (it was split off mplayer.c and thus contains random functions out of this file), but actually all functions have been added post-MPlayer. Except get_relative_time(), which was written by uau, but looks similar to 3 different versions of something similar in each of the Unix/win32/OSX timer source files. I'm not sure what that means in regards to copyright, so I've just moved it into another still-GPL source file for now. screenshot.c once had some minor parts of MPlayer's vf_screenshot.c, but they're all gone.
2016-01-19 17:36:06 +00:00
double get_relative_time(struct MPContext *mpctx);
void reset_playback_state(struct MPContext *mpctx);
void set_pause_state(struct MPContext *mpctx, bool user_pause);
void update_internal_pause_state(struct MPContext *mpctx);
void update_core_idle_state(struct MPContext *mpctx);
core: add backstep support Allows stepping back one frame via the frame_back_step inout command, bound to "," by default. This uses the precise seeking facility, and a perfect frame index built on the fly. The index is built during playback and precise seeking, and contains (as of this commit) the last 100 displayed or skipped frames. This index is used to find the PTS of the previous frame, which is then used as target for a precise seek. If no PTS is found, the core attempts to do a seek before the current frame, and skip decoded frames until the current frame is reached; this will create a sufficient index and the normal backstep algorithm can be applied. This can be rather slow. The worst case for backstepping is about the same as the worst case for precise seeking if the previous frame can be deduced from the index. If not, the worst case will be twice as slow. There's also some minor danger that the index is incorrect in case framedropping is involved. For framedropping due to --framedrop, this problem is ignored (use of --framedrop is discouraged anyway). For framedropping during precise seeking (done to make it faster), we try to not add frames to the index that are produced when this can happen. I'm not sure how well that works (or if the logic is sane), and it's sure to break with some video filters. In the worst case, backstepping might silently skip frames if you backstep after a user-initiated precise seek. (Precise seeks to do indexing are not affected.) Likewise, video filters that somehow change timing of frames and do not do this in a deterministic way (i.e. if you seek to a position, frames with different timings are produced than when the position is reached during normal playback) will make backstepping silently jump to the wrong frame. Enabling/disabling filters during playback (like for example deinterlacing) will have similar bad effects.
2013-04-24 17:31:48 +00:00
void add_step_frame(struct MPContext *mpctx, int dir);
void queue_seek(struct MPContext *mpctx, enum seek_type type, double amount,
enum seek_precision exact, int flags);
double get_time_length(struct MPContext *mpctx);
double get_start_time(struct MPContext *mpctx, int dir);
double get_current_time(struct MPContext *mpctx);
double get_playback_time(struct MPContext *mpctx);
int get_percent_pos(struct MPContext *mpctx);
double get_current_pos_ratio(struct MPContext *mpctx, bool use_range);
int get_current_chapter(struct MPContext *mpctx);
char *chapter_display_name(struct MPContext *mpctx, int chapter);
char *chapter_name(struct MPContext *mpctx, int chapter);
double chapter_start_time(struct MPContext *mpctx, int chapter);
int get_chapter_count(struct MPContext *mpctx);
int get_cache_buffering_percentage(struct MPContext *mpctx);
void execute_queued_seek(struct MPContext *mpctx);
void run_playloop(struct MPContext *mpctx);
void mp_idle(struct MPContext *mpctx);
void idle_loop(struct MPContext *mpctx);
int handle_force_window(struct MPContext *mpctx, bool force);
void seek_to_last_frame(struct MPContext *mpctx);
void update_screensaver_state(struct MPContext *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
void update_ab_loop_clip(struct MPContext *mpctx);
bool get_internal_paused(struct MPContext *mpctx);
// scripting.c
struct mp_script_args {
const struct mp_scripting *backend;
struct MPContext *mpctx;
struct mp_log *log;
struct mpv_handle *client;
const char *filename;
const char *path;
};
struct mp_scripting {
const char *name; // e.g. "lua script"
const char *file_ext; // e.g. "lua"
bool no_thread; // don't run load() on dedicated thread
int (*load)(struct mp_script_args *args);
};
bool mp_load_scripts(struct MPContext *mpctx);
void mp_load_builtin_scripts(struct MPContext *mpctx);
int64_t mp_load_user_script(struct MPContext *mpctx, const char *fname);
// sub.c
void reset_subtitle_state(struct MPContext *mpctx);
void reinit_sub(struct MPContext *mpctx, struct track *track);
void reinit_sub_all(struct MPContext *mpctx);
void uninit_sub(struct MPContext *mpctx, struct track *track);
void uninit_sub_all(struct MPContext *mpctx);
void update_osd_msg(struct MPContext *mpctx);
bool update_subtitles(struct MPContext *mpctx, double video_pts);
// video.c
void reset_video_state(struct MPContext *mpctx);
int init_video_decoder(struct MPContext *mpctx, struct track *track);
void reinit_video_chain(struct MPContext *mpctx);
void reinit_video_chain_src(struct MPContext *mpctx, struct track *track);
int reinit_video_filters(struct MPContext *mpctx);
void write_video(struct MPContext *mpctx);
void mp_force_video_refresh(struct MPContext *mpctx);
void uninit_video_out(struct MPContext *mpctx);
void uninit_video_chain(struct MPContext *mpctx);
double calc_average_frame_duration(struct MPContext *mpctx);
#endif /* MPLAYER_MP_CORE_H */