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Split mplayer.c mplayer.c was a bit too big. Split it into multiple files. I hope the way it's split makes sense. Maybe some things don't make too much sense, or go against intuition. These will fixed as soon as I notice them. Some files are a bit questionable (misc.c, osd.c, configfiles.c), and suggestions how to organize this better are welcome. Regressions are possible due to reorganized include statements. Obviously I didn't just copy mplayer.c's orgy of include statements, but recreated them for each file. It's easily possible that there are oversights and mistakes, which will show up on other platforms. There is one actual change: the public avutil.h include is removed from encode.h, and I tried to replace most FFMIN/FFMAX/av_clip uses. I consider using libavutil too much as dangerous, because the set of include files they recursively pull in is rather arbitrary and is different between FFmpeg and Libav.
2013-10-29 21:38:29 +00:00
/*
Update license headers Signed-off-by: wm4 <wm4@nowhere>
2015-04-13 07:36:54 +00:00
* This file is part of mpv.
Split mplayer.c mplayer.c was a bit too big. Split it into multiple files. I hope the way it's split makes sense. Maybe some things don't make too much sense, or go against intuition. These will fixed as soon as I notice them. Some files are a bit questionable (misc.c, osd.c, configfiles.c), and suggestions how to organize this better are welcome. Regressions are possible due to reorganized include statements. Obviously I didn't just copy mplayer.c's orgy of include statements, but recreated them for each file. It's easily possible that there are oversights and mistakes, which will show up on other platforms. There is one actual change: the public avutil.h include is removed from encode.h, and I tried to replace most FFMIN/FFMAX/av_clip uses. I consider using libavutil too much as dangerous, because the set of include files they recursively pull in is rather arbitrary and is different between FFmpeg and Libav.
2013-10-29 21:38:29 +00:00
*
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.
Split mplayer.c mplayer.c was a bit too big. Split it into multiple files. I hope the way it's split makes sense. Maybe some things don't make too much sense, or go against intuition. These will fixed as soon as I notice them. Some files are a bit questionable (misc.c, osd.c, configfiles.c), and suggestions how to organize this better are welcome. Regressions are possible due to reorganized include statements. Obviously I didn't just copy mplayer.c's orgy of include statements, but recreated them for each file. It's easily possible that there are oversights and mistakes, which will show up on other platforms. There is one actual change: the public avutil.h include is removed from encode.h, and I tried to replace most FFMIN/FFMAX/av_clip uses. I consider using libavutil too much as dangerous, because the set of include files they recursively pull in is rather arbitrary and is different between FFmpeg and Libav.
2013-10-29 21:38:29 +00:00
*
Update license headers Signed-off-by: wm4 <wm4@nowhere>
2015-04-13 07:36:54 +00:00
* mpv is distributed in the hope that it will be useful,
Split mplayer.c mplayer.c was a bit too big. Split it into multiple files. I hope the way it's split makes sense. Maybe some things don't make too much sense, or go against intuition. These will fixed as soon as I notice them. Some files are a bit questionable (misc.c, osd.c, configfiles.c), and suggestions how to organize this better are welcome. Regressions are possible due to reorganized include statements. Obviously I didn't just copy mplayer.c's orgy of include statements, but recreated them for each file. It's easily possible that there are oversights and mistakes, which will show up on other platforms. There is one actual change: the public avutil.h include is removed from encode.h, and I tried to replace most FFMIN/FFMAX/av_clip uses. I consider using libavutil too much as dangerous, because the set of include files they recursively pull in is rather arbitrary and is different between FFmpeg and Libav.
2013-10-29 21:38:29 +00:00
* 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.
Split mplayer.c mplayer.c was a bit too big. Split it into multiple files. I hope the way it's split makes sense. Maybe some things don't make too much sense, or go against intuition. These will fixed as soon as I notice them. Some files are a bit questionable (misc.c, osd.c, configfiles.c), and suggestions how to organize this better are welcome. Regressions are possible due to reorganized include statements. Obviously I didn't just copy mplayer.c's orgy of include statements, but recreated them for each file. It's easily possible that there are oversights and mistakes, which will show up on other platforms. There is one actual change: the public avutil.h include is removed from encode.h, and I tried to replace most FFMIN/FFMAX/av_clip uses. I consider using libavutil too much as dangerous, because the set of include files they recursively pull in is rather arbitrary and is different between FFmpeg and Libav.
2013-10-29 21:38:29 +00:00
*
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/>.
Split mplayer.c mplayer.c was a bit too big. Split it into multiple files. I hope the way it's split makes sense. Maybe some things don't make too much sense, or go against intuition. These will fixed as soon as I notice them. Some files are a bit questionable (misc.c, osd.c, configfiles.c), and suggestions how to organize this better are welcome. Regressions are possible due to reorganized include statements. Obviously I didn't just copy mplayer.c's orgy of include statements, but recreated them for each file. It's easily possible that there are oversights and mistakes, which will show up on other platforms. There is one actual change: the public avutil.h include is removed from encode.h, and I tried to replace most FFMIN/FFMAX/av_clip uses. I consider using libavutil too much as dangerous, because the set of include files they recursively pull in is rather arbitrary and is different between FFmpeg and Libav.
2013-10-29 21:38:29 +00:00
*/
#include <stddef.h>
#include <stdbool.h>
#include <inttypes.h>
#include <math.h>
#include <assert.h>
#include "config.h"
mpv_talloc.h: rename from talloc.h This change helps avoiding conflict with talloc.h from libtalloc.
2016-01-11 18:03:40 +00:00
#include "mpv_talloc.h"
Split mplayer.c mplayer.c was a bit too big. Split it into multiple files. I hope the way it's split makes sense. Maybe some things don't make too much sense, or go against intuition. These will fixed as soon as I notice them. Some files are a bit questionable (misc.c, osd.c, configfiles.c), and suggestions how to organize this better are welcome. Regressions are possible due to reorganized include statements. Obviously I didn't just copy mplayer.c's orgy of include statements, but recreated them for each file. It's easily possible that there are oversights and mistakes, which will show up on other platforms. There is one actual change: the public avutil.h include is removed from encode.h, and I tried to replace most FFMIN/FFMAX/av_clip uses. I consider using libavutil too much as dangerous, because the set of include files they recursively pull in is rather arbitrary and is different between FFmpeg and Libav.
2013-10-29 21:38:29 +00:00
Split mpvcore/ into common/, misc/, bstr/
2013-12-17 01:39:45 +00:00
#include "common/msg.h"
Move options/config related files from mpvcore/ to options/ Since m_option.h and options.h are extremely often included, a lot of files have to be changed. Moving path.c/h to options/ is a bit questionable, but since this is mainly about access to config files (which are also handled in options/), it's probably ok.
2013-12-17 01:02:25 +00:00
#include "options/options.h"
video: auto-insert software rotation filter If the VO can't do rotation, insert a filter to do this. Note that this doesn't reuse the filter insertion code from command.c (used by "vf" input command), because that would end up more complicated: we don't even want to change the user filter option.
2014-04-20 21:54:13 +00:00
#include "options/m_config.h"
#include "options/m_option.h"
Split mpvcore/ into common/, misc/, bstr/
2013-12-17 01:39:45 +00:00
#include "common/common.h"
#include "common/encode.h"
Move options/config related files from mpvcore/ to options/ Since m_option.h and options.h are extremely often included, a lot of files have to be changed. Moving path.c/h to options/ is a bit questionable, but since this is mainly about access to config files (which are also handled in options/), it's probably ok.
2013-12-17 01:02:25 +00:00
#include "options/m_property.h"
player: move video display code out of the playloop Basically move the code from playloop.c to video.c. The new function write_video() now contains the code that was part of run_playloop(). There are no functional changes, except handling "new_frame_shown" slightly differently. This is done so that we don't need new a new MPContext field or a return value for write_video() to signal this condition. Instead, it's handled indirectly.
2014-07-30 21:24:08 +00:00
#include "osdep/timer.h"
Split mplayer.c mplayer.c was a bit too big. Split it into multiple files. I hope the way it's split makes sense. Maybe some things don't make too much sense, or go against intuition. These will fixed as soon as I notice them. Some files are a bit questionable (misc.c, osd.c, configfiles.c), and suggestions how to organize this better are welcome. Regressions are possible due to reorganized include statements. Obviously I didn't just copy mplayer.c's orgy of include statements, but recreated them for each file. It's easily possible that there are oversights and mistakes, which will show up on other platforms. There is one actual change: the public avutil.h include is removed from encode.h, and I tried to replace most FFMIN/FFMAX/av_clip uses. I consider using libavutil too much as dangerous, because the set of include files they recursively pull in is rather arbitrary and is different between FFmpeg and Libav.
2013-10-29 21:38:29 +00:00
#include "audio/out/ao.h"
audio: make libaf derived code optional This code could not be relicensed. The intention was to write new filter code (which could handle both audio and video), but that's a bit of work. Write some code that can do audio conversion (resampling, downmixing, etc.) without the old audio filter chain code in order to speed up the LGPL relicensing. If you build with --disable-libaf, nothing in audio/filter/* is compiled in. It breaks a few features, such as --volume, --af, pitch correction on speed changes, replaygain. Most likely this adds some bugs, even if --disable-libaf is not used. (How the fuck does EOF notification work again anyway?)
2017-09-21 10:48:30 +00:00
#include "audio/format.h"
Split mplayer.c mplayer.c was a bit too big. Split it into multiple files. I hope the way it's split makes sense. Maybe some things don't make too much sense, or go against intuition. These will fixed as soon as I notice them. Some files are a bit questionable (misc.c, osd.c, configfiles.c), and suggestions how to organize this better are welcome. Regressions are possible due to reorganized include statements. Obviously I didn't just copy mplayer.c's orgy of include statements, but recreated them for each file. It's easily possible that there are oversights and mistakes, which will show up on other platforms. There is one actual change: the public avutil.h include is removed from encode.h, and I tried to replace most FFMIN/FFMAX/av_clip uses. I consider using libavutil too much as dangerous, because the set of include files they recursively pull in is rather arbitrary and is different between FFmpeg and Libav.
2013-10-29 21:38:29 +00:00
#include "demux/demux.h"
#include "stream/stream.h"
Rename sub.c/.h to osd.c/.h This was way too misleading. osd.c merely calls the subtitle renderers, instead of actually dealing with subtitles.
2013-11-24 11:58:06 +00:00
#include "sub/osd.h"
video: move struct mp_hwdec_info into its own header file This means most code accessing this struct must now include hwdec.h instead of dec_video.h. I just put it into dec_video.h at first because I thought a separate file would be a waste, but it's more proper to do it this way, as there are too many files which include dec_video.h only to get the mp_hwdec_info definition.
2013-11-23 20:26:31 +00:00
#include "video/hwdec.h"
video: make decoder wrapper a filter Move dec_video.c to filters/f_decoder_wrapper.c. It essentially becomes a source filter. vd.h mostly disappears, because mp_filter takes care of the dataflow, but its remains are in struct mp_decoder_fns. One goal is to simplify dataflow by letting the filter framework handle it (or more accurately, using its conventions). One result is that the decode calls disappear from video.c, because we simply connect the decoder wrapper and the filter chain with mp_pin_connect(). Another goal is to eventually remove the code duplication between the audio and video paths for this. This commit prepares for this by trying to make f_decoder_wrapper.c extensible, so it can be used for audio as well later. Decoder framedropping changes a bit. It doesn't seem to be worse than before, and it's an obscure feature, so I'm content with its new state. Some special code that was apparently meant to avoid dropping too many frames in a row is removed, though. I'm not sure how the source code tree should be organized. For one, video/decode/vd_lavc.c is the only file in its directory, which is a bit annoying.
2018-01-28 09:08:45 +00:00
#include "filters/f_decoder_wrapper.h"
Split mplayer.c mplayer.c was a bit too big. Split it into multiple files. I hope the way it's split makes sense. Maybe some things don't make too much sense, or go against intuition. These will fixed as soon as I notice them. Some files are a bit questionable (misc.c, osd.c, configfiles.c), and suggestions how to organize this better are welcome. Regressions are possible due to reorganized include statements. Obviously I didn't just copy mplayer.c's orgy of include statements, but recreated them for each file. It's easily possible that there are oversights and mistakes, which will show up on other platforms. There is one actual change: the public avutil.h include is removed from encode.h, and I tried to replace most FFMIN/FFMAX/av_clip uses. I consider using libavutil too much as dangerous, because the set of include files they recursively pull in is rather arbitrary and is different between FFmpeg and Libav.
2013-10-29 21:38:29 +00:00
#include "video/out/vo.h"
Rename mp_core.h to core.h Get rid of the mp_ prefix.
2013-12-17 00:08:53 +00:00
#include "core.h"
Split mplayer.c mplayer.c was a bit too big. Split it into multiple files. I hope the way it's split makes sense. Maybe some things don't make too much sense, or go against intuition. These will fixed as soon as I notice them. Some files are a bit questionable (misc.c, osd.c, configfiles.c), and suggestions how to organize this better are welcome. Regressions are possible due to reorganized include statements. Obviously I didn't just copy mplayer.c's orgy of include statements, but recreated them for each file. It's easily possible that there are oversights and mistakes, which will show up on other platforms. There is one actual change: the public avutil.h include is removed from encode.h, and I tried to replace most FFMIN/FFMAX/av_clip uses. I consider using libavutil too much as dangerous, because the set of include files they recursively pull in is rather arbitrary and is different between FFmpeg and Libav.
2013-10-29 21:38:29 +00:00
#include "command.h"
player: move video display code out of the playloop Basically move the code from playloop.c to video.c. The new function write_video() now contains the code that was part of run_playloop(). There are no functional changes, except handling "new_frame_shown" slightly differently. This is done so that we don't need new a new MPContext field or a return value for write_video() to signal this condition. Instead, it's handled indirectly.
2014-07-30 21:24:08 +00:00
#include "screenshot.h"
enum {
// update_video() - code also uses: <0 error, 0 eof, >0 progress
VD_ERROR = -1,
VD_EOF = 0, // end of file - no new output
VD_PROGRESS = 1, // progress, but no output; repeat call with no waiting
VD_NEW_FRAME = 2, // the call produced a new frame
VD_WAIT = 3, // no EOF, but no output; wait until wakeup
};
static const char av_desync_help_text[] =
player: slim down A/V desync warning I don't think most of these suggestions are overly helpful. Just get rid of them.
2015-06-30 17:06:00 +00:00
"\n"
"Audio/Video desynchronisation detected! Possible reasons include too slow\n"
"hardware, temporary CPU spikes, broken drivers, and broken files. Audio\n"
"position will not match to the video (see A-V status field).\n"
"\n";
Split mplayer.c mplayer.c was a bit too big. Split it into multiple files. I hope the way it's split makes sense. Maybe some things don't make too much sense, or go against intuition. These will fixed as soon as I notice them. Some files are a bit questionable (misc.c, osd.c, configfiles.c), and suggestions how to organize this better are welcome. Regressions are possible due to reorganized include statements. Obviously I didn't just copy mplayer.c's orgy of include statements, but recreated them for each file. It's easily possible that there are oversights and mistakes, which will show up on other platforms. There is one actual change: the public avutil.h include is removed from encode.h, and I tried to replace most FFMIN/FFMAX/av_clip uses. I consider using libavutil too much as dangerous, because the set of include files they recursively pull in is rather arbitrary and is different between FFmpeg and Libav.
2013-10-29 21:38:29 +00:00
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
static bool recreate_video_filters(struct MPContext *mpctx)
Split mplayer.c mplayer.c was a bit too big. Split it into multiple files. I hope the way it's split makes sense. Maybe some things don't make too much sense, or go against intuition. These will fixed as soon as I notice them. Some files are a bit questionable (misc.c, osd.c, configfiles.c), and suggestions how to organize this better are welcome. Regressions are possible due to reorganized include statements. Obviously I didn't just copy mplayer.c's orgy of include statements, but recreated them for each file. It's easily possible that there are oversights and mistakes, which will show up on other platforms. There is one actual change: the public avutil.h include is removed from encode.h, and I tried to replace most FFMIN/FFMAX/av_clip uses. I consider using libavutil too much as dangerous, because the set of include files they recursively pull in is rather arbitrary and is different between FFmpeg and Libav.
2013-10-29 21:38:29 +00:00
{
struct MPOpts *opts = mpctx->opts;
video: decouple filtering/decoding slightly more Lots of noise to remove the vfilter/vo fields from dec_video. From now on, video filtering and output will still be done together, summarized under struct vo_chain. There is the question where exactly the vf_chain should go in such a decoupled architecture. The end goal is being able to place a "complex" filter between video decoders and output (which will culminate in natural integration of A->V filters for natural integration of libavfilter audio visualizations). The vf_chain is still useful for "final" processing, such as format conversions and deinterlacing. Also, there's only 1 VO and 1 --vf option. So having 1 vf_chain for a VO seems ideal, since otherwise there would be no natural way to handle all these existing options and mechanisms. There is still some work required to truly decouple decoding.
2016-01-13 23:18:48 +00:00
struct vo_chain *vo_c = mpctx->vo_chain;
player: refactor: eliminate MPContext.d_video Eventually we want the VO be driven by a A->V filter, so a decoder doesn't even have to exist. Some features definitely require a decoder though (like reporting the decoder in use, hardware decoding, etc.), so for each thing which accessed d_video, it has to be redecided if and how it can access decoder state. At least the "framedrop" property slightly changes semantics: you can now always set this property, even if no video is active. Some untested changes in this commit, but our bio-based distributed test suite has to take care of this.
2016-01-17 17:07:50 +00:00
assert(vo_c);
Split mplayer.c mplayer.c was a bit too big. Split it into multiple files. I hope the way it's split makes sense. Maybe some things don't make too much sense, or go against intuition. These will fixed as soon as I notice them. Some files are a bit questionable (misc.c, osd.c, configfiles.c), and suggestions how to organize this better are welcome. Regressions are possible due to reorganized include statements. Obviously I didn't just copy mplayer.c's orgy of include statements, but recreated them for each file. It's easily possible that there are oversights and mistakes, which will show up on other platforms. There is one actual change: the public avutil.h include is removed from encode.h, and I tried to replace most FFMIN/FFMAX/av_clip uses. I consider using libavutil too much as dangerous, because the set of include files they recursively pull in is rather arbitrary and is different between FFmpeg and Libav.
2013-10-29 21:38:29 +00:00
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
return mp_output_chain_update_filters(vo_c->filter, opts->vf_settings);
Split mplayer.c mplayer.c was a bit too big. Split it into multiple files. I hope the way it's split makes sense. Maybe some things don't make too much sense, or go against intuition. These will fixed as soon as I notice them. Some files are a bit questionable (misc.c, osd.c, configfiles.c), and suggestions how to organize this better are welcome. Regressions are possible due to reorganized include statements. Obviously I didn't just copy mplayer.c's orgy of include statements, but recreated them for each file. It's easily possible that there are oversights and mistakes, which will show up on other platforms. There is one actual change: the public avutil.h include is removed from encode.h, and I tried to replace most FFMIN/FFMAX/av_clip uses. I consider using libavutil too much as dangerous, because the set of include files they recursively pull in is rather arbitrary and is different between FFmpeg and Libav.
2013-10-29 21:38:29 +00:00
}
int reinit_video_filters(struct MPContext *mpctx)
{
video: decouple filtering/decoding slightly more Lots of noise to remove the vfilter/vo fields from dec_video. From now on, video filtering and output will still be done together, summarized under struct vo_chain. There is the question where exactly the vf_chain should go in such a decoupled architecture. The end goal is being able to place a "complex" filter between video decoders and output (which will culminate in natural integration of A->V filters for natural integration of libavfilter audio visualizations). The vf_chain is still useful for "final" processing, such as format conversions and deinterlacing. Also, there's only 1 VO and 1 --vf option. So having 1 vf_chain for a VO seems ideal, since otherwise there would be no natural way to handle all these existing options and mechanisms. There is still some work required to truly decouple decoding.
2016-01-13 23:18:48 +00:00
struct vo_chain *vo_c = mpctx->vo_chain;
Split mplayer.c mplayer.c was a bit too big. Split it into multiple files. I hope the way it's split makes sense. Maybe some things don't make too much sense, or go against intuition. These will fixed as soon as I notice them. Some files are a bit questionable (misc.c, osd.c, configfiles.c), and suggestions how to organize this better are welcome. Regressions are possible due to reorganized include statements. Obviously I didn't just copy mplayer.c's orgy of include statements, but recreated them for each file. It's easily possible that there are oversights and mistakes, which will show up on other platforms. There is one actual change: the public avutil.h include is removed from encode.h, and I tried to replace most FFMIN/FFMAX/av_clip uses. I consider using libavutil too much as dangerous, because the set of include files they recursively pull in is rather arbitrary and is different between FFmpeg and Libav.
2013-10-29 21:38:29 +00:00
player: refactor: eliminate MPContext.d_video Eventually we want the VO be driven by a A->V filter, so a decoder doesn't even have to exist. Some features definitely require a decoder though (like reporting the decoder in use, hardware decoding, etc.), so for each thing which accessed d_video, it has to be redecided if and how it can access decoder state. At least the "framedrop" property slightly changes semantics: you can now always set this property, even if no video is active. Some untested changes in this commit, but our bio-based distributed test suite has to take care of this.
2016-01-17 17:07:50 +00:00
if (!vo_c)
command: allow changing filters before video chain initialization Apparently this is more intuitive. Somewhat tricky, because of the odd state after loading a file but before initializing the VO.
2014-03-30 17:21:54 +00:00
return 0;
Split mplayer.c mplayer.c was a bit too big. Split it into multiple files. I hope the way it's split makes sense. Maybe some things don't make too much sense, or go against intuition. These will fixed as soon as I notice them. Some files are a bit questionable (misc.c, osd.c, configfiles.c), and suggestions how to organize this better are welcome. Regressions are possible due to reorganized include statements. Obviously I didn't just copy mplayer.c's orgy of include statements, but recreated them for each file. It's easily possible that there are oversights and mistakes, which will show up on other platforms. There is one actual change: the public avutil.h include is removed from encode.h, and I tried to replace most FFMIN/FFMAX/av_clip uses. I consider using libavutil too much as dangerous, because the set of include files they recursively pull in is rather arbitrary and is different between FFmpeg and Libav.
2013-10-29 21:38:29 +00:00
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
if (!recreate_video_filters(mpctx))
return -1;
video: refactor: slightly disentangle video filtering
2016-01-15 21:23:01 +00:00
player: force refresh seek when changing audio filters Unfortunately I see no better solution. The refresh seek is skipped if the amount of buffered audio is not overly huge. Unfortunately softvol af_volume insertion still can cause this issue, because it's outside of the normal dynamic filter chain changing code. Move the video refresh call to reinit_video_filters() to make it more uniform along with the audio code.
2016-02-09 21:19:01 +00:00
mp_force_video_refresh(mpctx);
video: refactor: slightly disentangle video filtering
2016-01-15 21:23:01 +00:00
mp_notify(mpctx, MPV_EVENT_VIDEO_RECONFIG, NULL);
command: allow changing filters before video chain initialization Apparently this is more intuitive. Somewhat tricky, because of the odd state after loading a file but before initializing the VO.
2014-03-30 17:21:54 +00:00
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
return 0;
video: decouple filtering/decoding slightly more Lots of noise to remove the vfilter/vo fields from dec_video. From now on, video filtering and output will still be done together, summarized under struct vo_chain. There is the question where exactly the vf_chain should go in such a decoupled architecture. The end goal is being able to place a "complex" filter between video decoders and output (which will culminate in natural integration of A->V filters for natural integration of libavfilter audio visualizations). The vf_chain is still useful for "final" processing, such as format conversions and deinterlacing. Also, there's only 1 VO and 1 --vf option. So having 1 vf_chain for a VO seems ideal, since otherwise there would be no natural way to handle all these existing options and mechanisms. There is still some work required to truly decouple decoding.
2016-01-13 23:18:48 +00:00
}
static void vo_chain_reset_state(struct vo_chain *vo_c)
{
vo_seek_reset(vo_c->vo);
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
vo_c->underrun = false;
player: avoid underrun wakeup loop The VO underrun detection (just a weak heuristic) added in commit f26dfb flagged the underrun state every time it was checked, and since the check happened in every playloop iteration, this caused the playloop to wake up itself on every iteration. It burned an entire core while in this state. Fix this by flagging this condition only once (as it should be), and requiring that a frame is displayed to trigger it again. This makes it work similar as the audio underrun check. The bug report referenced below says --demuxer-thread=no avoided this. This is because the demuxer layer doesn't do proper underrun reporting if the reader thread is disabled. Fixes: #7259
2019-12-16 00:15:43 +00:00
vo_c->underrun_signaled = false;
Split mplayer.c mplayer.c was a bit too big. Split it into multiple files. I hope the way it's split makes sense. Maybe some things don't make too much sense, or go against intuition. These will fixed as soon as I notice them. Some files are a bit questionable (misc.c, osd.c, configfiles.c), and suggestions how to organize this better are welcome. Regressions are possible due to reorganized include statements. Obviously I didn't just copy mplayer.c's orgy of include statements, but recreated them for each file. It's easily possible that there are oversights and mistakes, which will show up on other platforms. There is one actual change: the public avutil.h include is removed from encode.h, and I tried to replace most FFMIN/FFMAX/av_clip uses. I consider using libavutil too much as dangerous, because the set of include files they recursively pull in is rather arbitrary and is different between FFmpeg and Libav.
2013-10-29 21:38:29 +00:00
}
player: split seek_reset() This also reduces some code duplication with other parts of the code. The changfe is mostly cosmetic, although there are also some subtle changes in behavior. At least one change is that the big desync message is now printed after every seek.
2014-07-30 21:01:55 +00:00
void reset_video_state(struct MPContext *mpctx)
{
player: ensure backward playback state is propagated on track switching Track switching doesn't run reset_playback_state(), so a track enabled at runtime during backward playback would lead to a messed up state. This commit just does a bad code monkey fix to this. It feels like there needs to be a much better way to propagate this state.
2019-06-03 00:05:52 +00:00
if (mpctx->vo_chain) {
video: decouple filtering/decoding slightly more Lots of noise to remove the vfilter/vo fields from dec_video. From now on, video filtering and output will still be done together, summarized under struct vo_chain. There is the question where exactly the vf_chain should go in such a decoupled architecture. The end goal is being able to place a "complex" filter between video decoders and output (which will culminate in natural integration of A->V filters for natural integration of libavfilter audio visualizations). The vf_chain is still useful for "final" processing, such as format conversions and deinterlacing. Also, there's only 1 VO and 1 --vf option. So having 1 vf_chain for a VO seems ideal, since otherwise there would be no natural way to handle all these existing options and mechanisms. There is still some work required to truly decouple decoding.
2016-01-13 23:18:48 +00:00
vo_chain_reset_state(mpctx->vo_chain);
player: ensure backward playback state is propagated on track switching Track switching doesn't run reset_playback_state(), so a track enabled at runtime during backward playback would lead to a messed up state. This commit just does a bad code monkey fix to this. It feels like there needs to be a much better way to propagate this state.
2019-06-03 00:05:52 +00:00
struct track *t = mpctx->vo_chain->track;
if (t && t->dec)
t->dec->play_dir = mpctx->play_dir;
}
player: split seek_reset() This also reduces some code duplication with other parts of the code. The changfe is mostly cosmetic, although there are also some subtle changes in behavior. At least one change is that the big desync message is now printed after every seek.
2014-07-30 21:01:55 +00:00
video: pass future frames to VO Now the VO can request a number of future frames with the last parameter of vo_set_queue_params(). This will be helpful to fix the interpolation code. Note that the first frame (after playback start or seeking) will usually not have any future frames (to make seeking fast). Near the end of the file, the number of future frames will become lower as well.
2015-07-01 17:22:40 +00:00
for (int n = 0; n < mpctx->num_next_frames; n++)
mp_image_unrefp(&mpctx->next_frames[n]);
mpctx->num_next_frames = 0;
player: when seeking past EOF with --keep-open, seek to last frame It feels strange that seeking past EOF with --keep-open actually leaves the player at a random position. You can't even unpause, because the demuxer is in the EOF state, and what you see on screen is just what was around before the seek. Improve this by attempting to seek to the last video frame if EOF happens. We explicitly don't do this if EOF was reached normally to increase robustness (if the VO got a frame since the last seek, it obviously means we had normal playback before EOF). If an error happens when trying to find the last frame (such as not actually finding a last frame because e.g. the demuxer misbehaves), this will probably turn your CPU into a heater. There is no logic to prevent reinitiating the last-frame search if the last-frame search reached EOF. (Pausing usually prevents that EOF is reached again after a successful last-frame search.) Fixes #819.
2014-12-07 01:47:09 +00:00
mp_image_unrefp(&mpctx->saved_frame);
video: fix and simplify video format changes and last frame display The previous commit broke these things, and fixing them is separate in this commit in order to reduce the volume of changes. Move the image queue from the VO to the playback core. The image queue is a remnant of the old way how vdpau was implemented, and increasingly became more and more an artifact. In the end, it did only one thing: computing the duration of the current frame. This was done by taking the PTS difference between the current and the future frame. We keep this, but by moving it out of the VO, we don't have to special-case format changes anymore. This simplifies the code a lot. Since we need the queue to compute the duration only, a queue size larger than 2 makes no sense, and we can hardcode that. Also change how the last frame is handled. The last frame is a bit of a problem, because video timing works by showing one frame after another, which makes it a special case. Make the VO provide a function to notify us when the frame is done, instead. The frame duration is used for that. This is not perfect. For example, changing playback speed during the last frame doesn't update the end time. Pausing will not stop the clock that times the last frame. But I don't think this matters for such a corner case.
2014-08-12 21:17:35 +00:00
player: split seek_reset() This also reduces some code duplication with other parts of the code. The changfe is mostly cosmetic, although there are also some subtle changes in behavior. At least one change is that the big desync message is now printed after every seek.
2014-07-30 21:01:55 +00:00
mpctx->delay = 0;
mpctx->time_frame = 0;
video: get rid of video_next_pts field Not really needed anymore. Code should be mostly equivalent. Also get rid of some other now-unused or outdated things.
2014-08-22 12:20:00 +00:00
mpctx->video_pts = MP_NOPTS_VALUE;
video: use demuxer-signaled duration for last video frame Helps with gif, probably does unwanted things with other formats. This doesn't handle --end quite correctly, but this could be added later. Fixes #3924.
2016-12-21 17:18:24 +00:00
mpctx->last_frame_duration = 0;
player: refactor display-sync frame duration calculations Get rid of get_past_frame_durations(), which was a bit too messy. Add a past_frames array, which contains the same information in a more reasonable way. This also means that we can get the exact current and past frame durations without going through awful stuff. (The main problem is that vo_pts_history contains future frames as well, which is needed for frame backstepping etc., but gets in the way here.) Also disable the automatic disabling of display-sync if the frame duration changes, and extend the frame durations allowed for display sync. To allow arbitrarily high durations, vo.c needs to be changed to pause and potentially redraw OSD while showing a single frame, so they're still limited. In an attempt to deal with VFR, calculate the overall speed using the average FPS. The frame scheduling itself does not use the average FPS, but the duration of the current frame. This does not work too well, but provides a good base for further improvements. Where this commit actually helps a lot is dealing with rounded timestamps, e.g. if the container framerate is wrong or unknown, or if the muxer wrote incorrectly rounded timestamps. While the rounding errors apparently can't be get rid of completely in the general case, this is still much better than e.g. disabling display-sync completely just because some frame durations go out of bounds.
2015-11-13 21:45:40 +00:00
mpctx->num_past_frames = 0;
player: split seek_reset() This also reduces some code duplication with other parts of the code. The changfe is mostly cosmetic, although there are also some subtle changes in behavior. At least one change is that the big desync message is now printed after every seek.
2014-07-30 21:01:55 +00:00
mpctx->total_avsync_change = 0;
video: do not use MP_NOPTS_VALUE for A/V difference There's no need for this, it just creates more corner cases. Also always reset it on seeks etc..
2015-05-24 18:57:31 +00:00
mpctx->last_av_difference = 0;
command: add mistimed-frame-count property Does what the manpage says. This is a replacement incrementing the dropped frame counter (see previous commit).
2015-10-30 13:05:41 +00:00
mpctx->mistimed_frames_total = 0;
player: split seek_reset() This also reduces some code duplication with other parts of the code. The changfe is mostly cosmetic, although there are also some subtle changes in behavior. At least one change is that the big desync message is now printed after every seek.
2014-07-30 21:01:55 +00:00
mpctx->drop_message_shown = 0;
player: add display sync mode If this mode is enabled, the player tries to strictly synchronize video to display refresh. It will adjust playback speed to match the display, so if you play 23.976 fps video on a 24 Hz screen, playback speed is increased by approximately 1/1000. Audio wll be resampled to keep up with playback. This is different from the default sync mode, which will sync video to audio, with the consequence that video might skip or repeat a frame once in a while to make video keep up with audio. This is still unpolished. There are some major problems as well; in particular, mkv VFR files won't work well. The reason is that Matroska is terrible and rounds timestamps to milliseconds. This makes it rather hard to guess the framerate of a section of video that is playing. We could probably fix this by just accepting jittery timestamps (instead of explicitly disabling the sync code in this case), but I'm not ready to accept such a solution yet. Another issue is that we are extremely reliant on OS video and audio APIs working in an expected manner, which of course is not too often the case. Consequently, the new sync mode is a bit fragile.
2015-08-10 16:43:25 +00:00
mpctx->display_sync_drift_dir = 0;
player: split seek_reset() This also reduces some code duplication with other parts of the code. The changfe is mostly cosmetic, although there are also some subtle changes in behavior. At least one change is that the big desync message is now printed after every seek.
2014-07-30 21:01:55 +00:00
player: refactor: eliminate MPContext.d_video Eventually we want the VO be driven by a A->V filter, so a decoder doesn't even have to exist. Some features definitely require a decoder though (like reporting the decoder in use, hardware decoding, etc.), so for each thing which accessed d_video, it has to be redecided if and how it can access decoder state. At least the "framedrop" property slightly changes semantics: you can now always set this property, even if no video is active. Some untested changes in this commit, but our bio-based distributed test suite has to take care of this.
2016-01-17 17:07:50 +00:00
mpctx->video_status = mpctx->vo_chain ? STATUS_SYNCING : STATUS_EOF;
player: split seek_reset() This also reduces some code duplication with other parts of the code. The changfe is mostly cosmetic, although there are also some subtle changes in behavior. At least one change is that the big desync message is now printed after every seek.
2014-07-30 21:01:55 +00:00
}
player: remove central uninit_player() function and flags mess Each subsystem (or similar thing) had an INITIALIZED_ flag assigned. The main use of this was that you could pass a bitmask of these flags to uninit_player(). Except in some situations where you wanted to uninitialize nearly everything, this wasn't really useful. Moreover, it was quite annoying that subsystems had most of the code in a specific file, but the uninit code in loadfile.c (because that's where uninit_player() was implemented). Simplify all this. Remove the flags; e.g. instead of testing for the INITIALIZED_AO flag, test whether mpctx->ao is set. Move uninit code to separate functions, e.g. uninit_audio_out().
2014-10-03 17:57:49 +00:00
void uninit_video_out(struct MPContext *mpctx)
{
uninit_video_chain(mpctx);
player: fix excessive CPU usage in audio-only mode Caused by one of the --force-window commits. The unconditional uninit_video_out() call (which normally should be idempotent) raised sporadic MPV_EVENT_VIDEO_RECONFIG events. This is ok, except for the fact that clients (like a Lua script or libmpv users) would cause the event loop to run again after receiving it, triggering a feedback loop. Fix it by sending the events really only on a change.
2015-09-22 09:19:39 +00:00
if (mpctx->video_out) {
player: remove central uninit_player() function and flags mess Each subsystem (or similar thing) had an INITIALIZED_ flag assigned. The main use of this was that you could pass a bitmask of these flags to uninit_player(). Except in some situations where you wanted to uninitialize nearly everything, this wasn't really useful. Moreover, it was quite annoying that subsystems had most of the code in a specific file, but the uninit code in loadfile.c (because that's where uninit_player() was implemented). Simplify all this. Remove the flags; e.g. instead of testing for the INITIALIZED_AO flag, test whether mpctx->ao is set. Move uninit code to separate functions, e.g. uninit_audio_out().
2014-10-03 17:57:49 +00:00
vo_destroy(mpctx->video_out);
player: fix excessive CPU usage in audio-only mode Caused by one of the --force-window commits. The unconditional uninit_video_out() call (which normally should be idempotent) raised sporadic MPV_EVENT_VIDEO_RECONFIG events. This is ok, except for the fact that clients (like a Lua script or libmpv users) would cause the event loop to run again after receiving it, triggering a feedback loop. Fix it by sending the events really only on a change.
2015-09-22 09:19:39 +00:00
mp_notify(mpctx, MPV_EVENT_VIDEO_RECONFIG, NULL);
}
player: remove central uninit_player() function and flags mess Each subsystem (or similar thing) had an INITIALIZED_ flag assigned. The main use of this was that you could pass a bitmask of these flags to uninit_player(). Except in some situations where you wanted to uninitialize nearly everything, this wasn't really useful. Moreover, it was quite annoying that subsystems had most of the code in a specific file, but the uninit code in loadfile.c (because that's where uninit_player() was implemented). Simplify all this. Remove the flags; e.g. instead of testing for the INITIALIZED_AO flag, test whether mpctx->ao is set. Move uninit code to separate functions, e.g. uninit_audio_out().
2014-10-03 17:57:49 +00:00
mpctx->video_out = NULL;
}
video: decouple filtering/decoding slightly more Lots of noise to remove the vfilter/vo fields from dec_video. From now on, video filtering and output will still be done together, summarized under struct vo_chain. There is the question where exactly the vf_chain should go in such a decoupled architecture. The end goal is being able to place a "complex" filter between video decoders and output (which will culminate in natural integration of A->V filters for natural integration of libavfilter audio visualizations). The vf_chain is still useful for "final" processing, such as format conversions and deinterlacing. Also, there's only 1 VO and 1 --vf option. So having 1 vf_chain for a VO seems ideal, since otherwise there would be no natural way to handle all these existing options and mechanisms. There is still some work required to truly decouple decoding.
2016-01-13 23:18:48 +00:00
static void vo_chain_uninit(struct vo_chain *vo_c)
{
player: add complex filter graph support See --lavfi-complex option. This is still quite rough. There's no support for dynamic configuration of any kind. There are probably corner cases where playback might freeze or burn 100% CPU (due to dataflow problems when interaction with libavfilter). Future possible plans might include: - freely switch tracks by providing some sort of default track graph label - automatically enabling audio visualization - automatically mix audio or stack video when multiple tracks are selected at once (similar to how multiple sub tracks can be selected)
2016-02-05 22:19:56 +00:00
struct track *track = vo_c->track;
if (track) {
assert(track->vo_c == vo_c);
track->vo_c = NULL;
video: make decoder wrapper a filter Move dec_video.c to filters/f_decoder_wrapper.c. It essentially becomes a source filter. vd.h mostly disappears, because mp_filter takes care of the dataflow, but its remains are in struct mp_decoder_fns. One goal is to simplify dataflow by letting the filter framework handle it (or more accurately, using its conventions). One result is that the decode calls disappear from video.c, because we simply connect the decoder wrapper and the filter chain with mp_pin_connect(). Another goal is to eventually remove the code duplication between the audio and video paths for this. This commit prepares for this by trying to make f_decoder_wrapper.c extensible, so it can be used for audio as well later. Decoder framedropping changes a bit. It doesn't seem to be worse than before, and it's an obscure feature, so I'm content with its new state. Some special code that was apparently meant to avoid dropping too many frames in a row is removed, though. I'm not sure how the source code tree should be organized. For one, video/decode/vd_lavc.c is the only file in its directory, which is a bit annoying.
2018-01-28 09:08:45 +00:00
if (vo_c->dec_src)
assert(track->dec->f->pins[0] == vo_c->dec_src);
talloc_free(track->dec->f);
track->dec = NULL;
player: add complex filter graph support See --lavfi-complex option. This is still quite rough. There's no support for dynamic configuration of any kind. There are probably corner cases where playback might freeze or burn 100% CPU (due to dataflow problems when interaction with libavfilter). Future possible plans might include: - freely switch tracks by providing some sort of default track graph label - automatically enabling audio visualization - automatically mix audio or stack video when multiple tracks are selected at once (similar to how multiple sub tracks can be selected)
2016-02-05 22:19:56 +00:00
}
if (vo_c->filter_src)
player: replace old lavfi wrapper with new filter code lavfi.c is not necessary anymore, because f_lavfi.c (which was actually converted from it) can be used now.
2018-01-26 03:36:47 +00:00
mp_pin_disconnect(vo_c->filter_src);
player: add complex filter graph support See --lavfi-complex option. This is still quite rough. There's no support for dynamic configuration of any kind. There are probably corner cases where playback might freeze or burn 100% CPU (due to dataflow problems when interaction with libavfilter). Future possible plans might include: - freely switch tracks by providing some sort of default track graph label - automatically enabling audio visualization - automatically mix audio or stack video when multiple tracks are selected at once (similar to how multiple sub tracks can be selected)
2016-02-05 22:19:56 +00:00
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
talloc_free(vo_c->filter->f);
video: decouple filtering/decoding slightly more Lots of noise to remove the vfilter/vo fields from dec_video. From now on, video filtering and output will still be done together, summarized under struct vo_chain. There is the question where exactly the vf_chain should go in such a decoupled architecture. The end goal is being able to place a "complex" filter between video decoders and output (which will culminate in natural integration of A->V filters for natural integration of libavfilter audio visualizations). The vf_chain is still useful for "final" processing, such as format conversions and deinterlacing. Also, there's only 1 VO and 1 --vf option. So having 1 vf_chain for a VO seems ideal, since otherwise there would be no natural way to handle all these existing options and mechanisms. There is still some work required to truly decouple decoding.
2016-01-13 23:18:48 +00:00
talloc_free(vo_c);
video: refactor: disentangle decoding/filtering some more This moves some code related to decoding from video.c to dec_video.c, and also removes some accesses to dec_video.c from the filtering code. dec_video.ch is starting to make sense, and simply returns video frames from a demuxer stream. The API exposed is also somewhat intended to be easily changeable to move decoding to a separate thread, if we ever want this (due to libavcodec already being threaded, I don't see much of a reason, but it might still be helpful).
2016-01-16 20:19:52 +00:00
// this does not free the VO
video: decouple filtering/decoding slightly more Lots of noise to remove the vfilter/vo fields from dec_video. From now on, video filtering and output will still be done together, summarized under struct vo_chain. There is the question where exactly the vf_chain should go in such a decoupled architecture. The end goal is being able to place a "complex" filter between video decoders and output (which will culminate in natural integration of A->V filters for natural integration of libavfilter audio visualizations). The vf_chain is still useful for "final" processing, such as format conversions and deinterlacing. Also, there's only 1 VO and 1 --vf option. So having 1 vf_chain for a VO seems ideal, since otherwise there would be no natural way to handle all these existing options and mechanisms. There is still some work required to truly decouple decoding.
2016-01-13 23:18:48 +00:00
}
player: remove central uninit_player() function and flags mess Each subsystem (or similar thing) had an INITIALIZED_ flag assigned. The main use of this was that you could pass a bitmask of these flags to uninit_player(). Except in some situations where you wanted to uninitialize nearly everything, this wasn't really useful. Moreover, it was quite annoying that subsystems had most of the code in a specific file, but the uninit code in loadfile.c (because that's where uninit_player() was implemented). Simplify all this. Remove the flags; e.g. instead of testing for the INITIALIZED_AO flag, test whether mpctx->ao is set. Move uninit code to separate functions, e.g. uninit_audio_out().
2014-10-03 17:57:49 +00:00
void uninit_video_chain(struct MPContext *mpctx)
{
player: refactor: eliminate MPContext.d_video Eventually we want the VO be driven by a A->V filter, so a decoder doesn't even have to exist. Some features definitely require a decoder though (like reporting the decoder in use, hardware decoding, etc.), so for each thing which accessed d_video, it has to be redecided if and how it can access decoder state. At least the "framedrop" property slightly changes semantics: you can now always set this property, even if no video is active. Some untested changes in this commit, but our bio-based distributed test suite has to take care of this.
2016-01-17 17:07:50 +00:00
if (mpctx->vo_chain) {
player: remove central uninit_player() function and flags mess Each subsystem (or similar thing) had an INITIALIZED_ flag assigned. The main use of this was that you could pass a bitmask of these flags to uninit_player(). Except in some situations where you wanted to uninitialize nearly everything, this wasn't really useful. Moreover, it was quite annoying that subsystems had most of the code in a specific file, but the uninit code in loadfile.c (because that's where uninit_player() was implemented). Simplify all this. Remove the flags; e.g. instead of testing for the INITIALIZED_AO flag, test whether mpctx->ao is set. Move uninit code to separate functions, e.g. uninit_audio_out().
2014-10-03 17:57:49 +00:00
reset_video_state(mpctx);
video: decouple filtering/decoding slightly more Lots of noise to remove the vfilter/vo fields from dec_video. From now on, video filtering and output will still be done together, summarized under struct vo_chain. There is the question where exactly the vf_chain should go in such a decoupled architecture. The end goal is being able to place a "complex" filter between video decoders and output (which will culminate in natural integration of A->V filters for natural integration of libavfilter audio visualizations). The vf_chain is still useful for "final" processing, such as format conversions and deinterlacing. Also, there's only 1 VO and 1 --vf option. So having 1 vf_chain for a VO seems ideal, since otherwise there would be no natural way to handle all these existing options and mechanisms. There is still some work required to truly decouple decoding.
2016-01-13 23:18:48 +00:00
vo_chain_uninit(mpctx->vo_chain);
mpctx->vo_chain = NULL;
player: add complex filter graph support See --lavfi-complex option. This is still quite rough. There's no support for dynamic configuration of any kind. There are probably corner cases where playback might freeze or burn 100% CPU (due to dataflow problems when interaction with libavfilter). Future possible plans might include: - freely switch tracks by providing some sort of default track graph label - automatically enabling audio visualization - automatically mix audio or stack video when multiple tracks are selected at once (similar to how multiple sub tracks can be selected)
2016-02-05 22:19:56 +00:00
player: remove central uninit_player() function and flags mess Each subsystem (or similar thing) had an INITIALIZED_ flag assigned. The main use of this was that you could pass a bitmask of these flags to uninit_player(). Except in some situations where you wanted to uninitialize nearly everything, this wasn't really useful. Moreover, it was quite annoying that subsystems had most of the code in a specific file, but the uninit code in loadfile.c (because that's where uninit_player() was implemented). Simplify all this. Remove the flags; e.g. instead of testing for the INITIALIZED_AO flag, test whether mpctx->ao is set. Move uninit code to separate functions, e.g. uninit_audio_out().
2014-10-03 17:57:49 +00:00
mpctx->video_status = STATUS_EOF;
player: slightly simplify how demuxer streams are enabled/disabled Instead of having reselect_demux_streams() look at all streams, make it look at the current stream that is being enabled/disabled.
2016-02-25 21:44:50 +00:00
player: fix excessive CPU usage in audio-only mode Caused by one of the --force-window commits. The unconditional uninit_video_out() call (which normally should be idempotent) raised sporadic MPV_EVENT_VIDEO_RECONFIG events. This is ok, except for the fact that clients (like a Lua script or libmpv users) would cause the event loop to run again after receiving it, triggering a feedback loop. Fix it by sending the events really only on a change.
2015-09-22 09:19:39 +00:00
mp_notify(mpctx, MPV_EVENT_VIDEO_RECONFIG, NULL);
player: remove central uninit_player() function and flags mess Each subsystem (or similar thing) had an INITIALIZED_ flag assigned. The main use of this was that you could pass a bitmask of these flags to uninit_player(). Except in some situations where you wanted to uninitialize nearly everything, this wasn't really useful. Moreover, it was quite annoying that subsystems had most of the code in a specific file, but the uninit code in loadfile.c (because that's where uninit_player() was implemented). Simplify all this. Remove the flags; e.g. instead of testing for the INITIALIZED_AO flag, test whether mpctx->ao is set. Move uninit code to separate functions, e.g. uninit_audio_out().
2014-10-03 17:57:49 +00:00
}
}
player: move audio and video decoder init to separate functions Preparation.
2016-02-05 20:50:37 +00:00
int init_video_decoder(struct MPContext *mpctx, struct track *track)
{
video: make decoder wrapper a filter Move dec_video.c to filters/f_decoder_wrapper.c. It essentially becomes a source filter. vd.h mostly disappears, because mp_filter takes care of the dataflow, but its remains are in struct mp_decoder_fns. One goal is to simplify dataflow by letting the filter framework handle it (or more accurately, using its conventions). One result is that the decode calls disappear from video.c, because we simply connect the decoder wrapper and the filter chain with mp_pin_connect(). Another goal is to eventually remove the code duplication between the audio and video paths for this. This commit prepares for this by trying to make f_decoder_wrapper.c extensible, so it can be used for audio as well later. Decoder framedropping changes a bit. It doesn't seem to be worse than before, and it's an obscure feature, so I'm content with its new state. Some special code that was apparently meant to avoid dropping too many frames in a row is removed, though. I'm not sure how the source code tree should be organized. For one, video/decode/vd_lavc.c is the only file in its directory, which is a bit annoying.
2018-01-28 09:08:45 +00:00
assert(!track->dec);
player: move audio and video decoder init to separate functions Preparation.
2016-02-05 20:50:37 +00:00
if (!track->stream)
goto err_out;
video: make decoder wrapper a filter Move dec_video.c to filters/f_decoder_wrapper.c. It essentially becomes a source filter. vd.h mostly disappears, because mp_filter takes care of the dataflow, but its remains are in struct mp_decoder_fns. One goal is to simplify dataflow by letting the filter framework handle it (or more accurately, using its conventions). One result is that the decode calls disappear from video.c, because we simply connect the decoder wrapper and the filter chain with mp_pin_connect(). Another goal is to eventually remove the code duplication between the audio and video paths for this. This commit prepares for this by trying to make f_decoder_wrapper.c extensible, so it can be used for audio as well later. Decoder framedropping changes a bit. It doesn't seem to be worse than before, and it's an obscure feature, so I'm content with its new state. Some special code that was apparently meant to avoid dropping too many frames in a row is removed, though. I'm not sure how the source code tree should be organized. For one, video/decode/vd_lavc.c is the only file in its directory, which is a bit annoying.
2018-01-28 09:08:45 +00:00
struct mp_filter *parent = mpctx->filter_root;
client API: make mpv_opengl_cb_uninit_gl() behavior slightly nicer Instead of deselecting the video stream plainly, use the slightly more robust error_on_track() function. Also give it an error code (although I'm not sure if this one is confusing, it's better than the one before).
2016-09-09 08:59:09 +00:00
video: make decoder wrapper a filter Move dec_video.c to filters/f_decoder_wrapper.c. It essentially becomes a source filter. vd.h mostly disappears, because mp_filter takes care of the dataflow, but its remains are in struct mp_decoder_fns. One goal is to simplify dataflow by letting the filter framework handle it (or more accurately, using its conventions). One result is that the decode calls disappear from video.c, because we simply connect the decoder wrapper and the filter chain with mp_pin_connect(). Another goal is to eventually remove the code duplication between the audio and video paths for this. This commit prepares for this by trying to make f_decoder_wrapper.c extensible, so it can be used for audio as well later. Decoder framedropping changes a bit. It doesn't seem to be worse than before, and it's an obscure feature, so I'm content with its new state. Some special code that was apparently meant to avoid dropping too many frames in a row is removed, though. I'm not sure how the source code tree should be organized. For one, video/decode/vd_lavc.c is the only file in its directory, which is a bit annoying.
2018-01-28 09:08:45 +00:00
// If possible, set this as parent so the decoder gets the hwdec and DR
// interfaces.
client API: make mpv_opengl_cb_uninit_gl() behavior slightly nicer Instead of deselecting the video stream plainly, use the slightly more robust error_on_track() function. Also give it an error code (although I'm not sure if this one is confusing, it's better than the one before).
2016-09-09 08:59:09 +00:00
// Note: at least mpv_opengl_cb_uninit_gl() relies on being able to get
// rid of all references to the VO by destroying the VO chain. Thus,
// decoders not linked to vo_chain must not use the hwdec context.
video: make decoder wrapper a filter Move dec_video.c to filters/f_decoder_wrapper.c. It essentially becomes a source filter. vd.h mostly disappears, because mp_filter takes care of the dataflow, but its remains are in struct mp_decoder_fns. One goal is to simplify dataflow by letting the filter framework handle it (or more accurately, using its conventions). One result is that the decode calls disappear from video.c, because we simply connect the decoder wrapper and the filter chain with mp_pin_connect(). Another goal is to eventually remove the code duplication between the audio and video paths for this. This commit prepares for this by trying to make f_decoder_wrapper.c extensible, so it can be used for audio as well later. Decoder framedropping changes a bit. It doesn't seem to be worse than before, and it's an obscure feature, so I'm content with its new state. Some special code that was apparently meant to avoid dropping too many frames in a row is removed, though. I'm not sure how the source code tree should be organized. For one, video/decode/vd_lavc.c is the only file in its directory, which is a bit annoying.
2018-01-28 09:08:45 +00:00
if (track->vo_c)
parent = track->vo_c->filter->f;
player: move audio and video decoder init to separate functions Preparation.
2016-02-05 20:50:37 +00:00
video: make decoder wrapper a filter Move dec_video.c to filters/f_decoder_wrapper.c. It essentially becomes a source filter. vd.h mostly disappears, because mp_filter takes care of the dataflow, but its remains are in struct mp_decoder_fns. One goal is to simplify dataflow by letting the filter framework handle it (or more accurately, using its conventions). One result is that the decode calls disappear from video.c, because we simply connect the decoder wrapper and the filter chain with mp_pin_connect(). Another goal is to eventually remove the code duplication between the audio and video paths for this. This commit prepares for this by trying to make f_decoder_wrapper.c extensible, so it can be used for audio as well later. Decoder framedropping changes a bit. It doesn't seem to be worse than before, and it's an obscure feature, so I'm content with its new state. Some special code that was apparently meant to avoid dropping too many frames in a row is removed, though. I'm not sure how the source code tree should be organized. For one, video/decode/vd_lavc.c is the only file in its directory, which is a bit annoying.
2018-01-28 09:08:45 +00:00
track->dec = mp_decoder_wrapper_create(parent, track->stream);
if (!track->dec)
goto err_out;
player: move audio and video decoder init to separate functions Preparation.
2016-02-05 20:50:37 +00:00
video: make decoder wrapper a filter Move dec_video.c to filters/f_decoder_wrapper.c. It essentially becomes a source filter. vd.h mostly disappears, because mp_filter takes care of the dataflow, but its remains are in struct mp_decoder_fns. One goal is to simplify dataflow by letting the filter framework handle it (or more accurately, using its conventions). One result is that the decode calls disappear from video.c, because we simply connect the decoder wrapper and the filter chain with mp_pin_connect(). Another goal is to eventually remove the code duplication between the audio and video paths for this. This commit prepares for this by trying to make f_decoder_wrapper.c extensible, so it can be used for audio as well later. Decoder framedropping changes a bit. It doesn't seem to be worse than before, and it's an obscure feature, so I'm content with its new state. Some special code that was apparently meant to avoid dropping too many frames in a row is removed, though. I'm not sure how the source code tree should be organized. For one, video/decode/vd_lavc.c is the only file in its directory, which is a bit annoying.
2018-01-28 09:08:45 +00:00
if (!mp_decoder_wrapper_reinit(track->dec))
player: move audio and video decoder init to separate functions Preparation.
2016-02-05 20:50:37 +00:00
goto err_out;
return 1;
err_out:
player: add complex filter graph support See --lavfi-complex option. This is still quite rough. There's no support for dynamic configuration of any kind. There are probably corner cases where playback might freeze or burn 100% CPU (due to dataflow problems when interaction with libavfilter). Future possible plans might include: - freely switch tracks by providing some sort of default track graph label - automatically enabling audio visualization - automatically mix audio or stack video when multiple tracks are selected at once (similar to how multiple sub tracks can be selected)
2016-02-05 22:19:56 +00:00
if (track->sink)
player: replace old lavfi wrapper with new filter code lavfi.c is not necessary anymore, because f_lavfi.c (which was actually converted from it) can be used now.
2018-01-26 03:36:47 +00:00
mp_pin_disconnect(track->sink);
player: add complex filter graph support See --lavfi-complex option. This is still quite rough. There's no support for dynamic configuration of any kind. There are probably corner cases where playback might freeze or burn 100% CPU (due to dataflow problems when interaction with libavfilter). Future possible plans might include: - freely switch tracks by providing some sort of default track graph label - automatically enabling audio visualization - automatically mix audio or stack video when multiple tracks are selected at once (similar to how multiple sub tracks can be selected)
2016-02-05 22:19:56 +00:00
track->sink = NULL;
player: move audio and video decoder init to separate functions Preparation.
2016-02-05 20:50:37 +00:00
error_on_track(mpctx, track);
return 0;
}
player: make --lavfi-complex changeable at runtime Tends to be somewhat glitchy if subtitles are enabled, and you enable and disable tracks. On error, this will disable --lavfi-complex, which will result in whatever behavior.
2017-08-12 21:08:48 +00:00
void reinit_video_chain(struct MPContext *mpctx)
player: add complex filter graph support See --lavfi-complex option. This is still quite rough. There's no support for dynamic configuration of any kind. There are probably corner cases where playback might freeze or burn 100% CPU (due to dataflow problems when interaction with libavfilter). Future possible plans might include: - freely switch tracks by providing some sort of default track graph label - automatically enabling audio visualization - automatically mix audio or stack video when multiple tracks are selected at once (similar to how multiple sub tracks can be selected)
2016-02-05 22:19:56 +00:00
{
player: make --lavfi-complex changeable at runtime Tends to be somewhat glitchy if subtitles are enabled, and you enable and disable tracks. On error, this will disable --lavfi-complex, which will result in whatever behavior.
2017-08-12 21:08:48 +00:00
struct track *track = mpctx->current_track[0][STREAM_VIDEO];
if (!track || !track->stream) {
error_on_track(mpctx, track);
return;
}
reinit_video_chain_src(mpctx, track);
player: add complex filter graph support See --lavfi-complex option. This is still quite rough. There's no support for dynamic configuration of any kind. There are probably corner cases where playback might freeze or burn 100% CPU (due to dataflow problems when interaction with libavfilter). Future possible plans might include: - freely switch tracks by providing some sort of default track graph label - automatically enabling audio visualization - automatically mix audio or stack video when multiple tracks are selected at once (similar to how multiple sub tracks can be selected)
2016-02-05 22:19:56 +00:00
}
video: make decoder wrapper a filter Move dec_video.c to filters/f_decoder_wrapper.c. It essentially becomes a source filter. vd.h mostly disappears, because mp_filter takes care of the dataflow, but its remains are in struct mp_decoder_fns. One goal is to simplify dataflow by letting the filter framework handle it (or more accurately, using its conventions). One result is that the decode calls disappear from video.c, because we simply connect the decoder wrapper and the filter chain with mp_pin_connect(). Another goal is to eventually remove the code duplication between the audio and video paths for this. This commit prepares for this by trying to make f_decoder_wrapper.c extensible, so it can be used for audio as well later. Decoder framedropping changes a bit. It doesn't seem to be worse than before, and it's an obscure feature, so I'm content with its new state. Some special code that was apparently meant to avoid dropping too many frames in a row is removed, though. I'm not sure how the source code tree should be organized. For one, video/decode/vd_lavc.c is the only file in its directory, which is a bit annoying.
2018-01-28 09:08:45 +00:00
static void filter_update_subtitles(void *ctx, double pts)
{
struct MPContext *mpctx = ctx;
if (osd_get_render_subs_in_filter(mpctx->osd))
update_subtitles(mpctx, pts);
}
player: make --lavfi-complex changeable at runtime Tends to be somewhat glitchy if subtitles are enabled, and you enable and disable tracks. On error, this will disable --lavfi-complex, which will result in whatever behavior.
2017-08-12 21:08:48 +00:00
// (track=NULL creates a blank chain, used for lavfi-complex)
void reinit_video_chain_src(struct MPContext *mpctx, struct track *track)
Split mplayer.c mplayer.c was a bit too big. Split it into multiple files. I hope the way it's split makes sense. Maybe some things don't make too much sense, or go against intuition. These will fixed as soon as I notice them. Some files are a bit questionable (misc.c, osd.c, configfiles.c), and suggestions how to organize this better are welcome. Regressions are possible due to reorganized include statements. Obviously I didn't just copy mplayer.c's orgy of include statements, but recreated them for each file. It's easily possible that there are oversights and mistakes, which will show up on other platforms. There is one actual change: the public avutil.h include is removed from encode.h, and I tried to replace most FFMIN/FFMAX/av_clip uses. I consider using libavutil too much as dangerous, because the set of include files they recursively pull in is rather arbitrary and is different between FFmpeg and Libav.
2013-10-29 21:38:29 +00:00
{
video: decouple filtering/decoding slightly more Lots of noise to remove the vfilter/vo fields from dec_video. From now on, video filtering and output will still be done together, summarized under struct vo_chain. There is the question where exactly the vf_chain should go in such a decoupled architecture. The end goal is being able to place a "complex" filter between video decoders and output (which will culminate in natural integration of A->V filters for natural integration of libavfilter audio visualizations). The vf_chain is still useful for "final" processing, such as format conversions and deinterlacing. Also, there's only 1 VO and 1 --vf option. So having 1 vf_chain for a VO seems ideal, since otherwise there would be no natural way to handle all these existing options and mechanisms. There is still some work required to truly decouple decoding.
2016-01-13 23:18:48 +00:00
assert(!mpctx->vo_chain);
Split mplayer.c mplayer.c was a bit too big. Split it into multiple files. I hope the way it's split makes sense. Maybe some things don't make too much sense, or go against intuition. These will fixed as soon as I notice them. Some files are a bit questionable (misc.c, osd.c, configfiles.c), and suggestions how to organize this better are welcome. Regressions are possible due to reorganized include statements. Obviously I didn't just copy mplayer.c's orgy of include statements, but recreated them for each file. It's easily possible that there are oversights and mistakes, which will show up on other platforms. There is one actual change: the public avutil.h include is removed from encode.h, and I tried to replace most FFMIN/FFMAX/av_clip uses. I consider using libavutil too much as dangerous, because the set of include files they recursively pull in is rather arbitrary and is different between FFmpeg and Libav.
2013-10-29 21:38:29 +00:00
player: remove --fixed-vo In ancient times, this was needed because it was not default, and many VOs had problems with it. But it was always default in mpv, and all VOs are required to deal with it. Also, running --fixed-vo=no is not useful and just creates weird corner cases. Get rid of it.
2015-02-03 22:09:02 +00:00
if (!mpctx->video_out) {
video: pass some VO params as struct Not particularly elegant, but better than adding more and more stuff to the relevant function signatures.
2014-12-31 18:01:28 +00:00
struct vo_extra ex = {
.input_ctx = mpctx->input,
.osd = mpctx->osd,
.encode_lavc_ctx = mpctx->encode_lavc_ctx,
player, ao, vo: don't call mp_input_wakeup() directly Currently, calling mp_input_wakeup() will wake up the core thread (also called the playloop). This seems odd, but currently the core indeed calls mp_input_wait() when it has nothing more to do. It's done this way because MPlayer used input_ctx as central "mainloop". This is probably going to change. Remove direct calls to this function, and replace it with mp_wakeup_core() calls. ao and vo are changed to use opaque callbacks and not use input_ctx for this purpose. Other code already uses opaque callbacks, or has legitimate reasons to use input_ctx directly (such as sending actual user input).
2016-09-16 12:23:54 +00:00
.wakeup_cb = mp_wakeup_core_cb,
.wakeup_ctx = mpctx,
video: pass some VO params as struct Not particularly elegant, but better than adding more and more stuff to the relevant function signatures.
2014-12-31 18:01:28 +00:00
};
mpctx->video_out = init_best_video_out(mpctx->global, &ex);
Split mplayer.c mplayer.c was a bit too big. Split it into multiple files. I hope the way it's split makes sense. Maybe some things don't make too much sense, or go against intuition. These will fixed as soon as I notice them. Some files are a bit questionable (misc.c, osd.c, configfiles.c), and suggestions how to organize this better are welcome. Regressions are possible due to reorganized include statements. Obviously I didn't just copy mplayer.c's orgy of include statements, but recreated them for each file. It's easily possible that there are oversights and mistakes, which will show up on other platforms. There is one actual change: the public avutil.h include is removed from encode.h, and I tried to replace most FFMIN/FFMAX/av_clip uses. I consider using libavutil too much as dangerous, because the set of include files they recursively pull in is rather arbitrary and is different between FFmpeg and Libav.
2013-10-29 21:38:29 +00:00
if (!mpctx->video_out) {
MP_FATAL(mpctx, "Error opening/initializing "
manpage: replace `-vo` with `--vo`
2016-12-08 17:08:04 +00:00
"the selected video_out (--vo) device.\n");
client API: better error reporting Give somewhat more information on playback failure.
2014-10-28 15:19:07 +00:00
mpctx->error_playing = MPV_ERROR_VO_INIT_FAILED;
Split mplayer.c mplayer.c was a bit too big. Split it into multiple files. I hope the way it's split makes sense. Maybe some things don't make too much sense, or go against intuition. These will fixed as soon as I notice them. Some files are a bit questionable (misc.c, osd.c, configfiles.c), and suggestions how to organize this better are welcome. Regressions are possible due to reorganized include statements. Obviously I didn't just copy mplayer.c's orgy of include statements, but recreated them for each file. It's easily possible that there are oversights and mistakes, which will show up on other platforms. There is one actual change: the public avutil.h include is removed from encode.h, and I tried to replace most FFMIN/FFMAX/av_clip uses. I consider using libavutil too much as dangerous, because the set of include files they recursively pull in is rather arbitrary and is different between FFmpeg and Libav.
2013-10-29 21:38:29 +00:00
goto err_out;
}
mpctx->mouse_cursor_visible = true;
}
player: set PulseAudio stream title to window title Set the PulseAudio stream title, just like the VO window title is set. Refactor update_vo_window_title() so that we can use it for AOs too. The ao_pulse.c bit is stolen from MPlayer.
2013-11-09 23:49:13 +00:00
update_window_title(mpctx, true);
Split mplayer.c mplayer.c was a bit too big. Split it into multiple files. I hope the way it's split makes sense. Maybe some things don't make too much sense, or go against intuition. These will fixed as soon as I notice them. Some files are a bit questionable (misc.c, osd.c, configfiles.c), and suggestions how to organize this better are welcome. Regressions are possible due to reorganized include statements. Obviously I didn't just copy mplayer.c's orgy of include statements, but recreated them for each file. It's easily possible that there are oversights and mistakes, which will show up on other platforms. There is one actual change: the public avutil.h include is removed from encode.h, and I tried to replace most FFMIN/FFMAX/av_clip uses. I consider using libavutil too much as dangerous, because the set of include files they recursively pull in is rather arbitrary and is different between FFmpeg and Libav.
2013-10-29 21:38:29 +00:00
player: refactor: eliminate MPContext.d_video Eventually we want the VO be driven by a A->V filter, so a decoder doesn't even have to exist. Some features definitely require a decoder though (like reporting the decoder in use, hardware decoding, etc.), so for each thing which accessed d_video, it has to be redecided if and how it can access decoder state. At least the "framedrop" property slightly changes semantics: you can now always set this property, even if no video is active. Some untested changes in this commit, but our bio-based distributed test suite has to take care of this.
2016-01-17 17:07:50 +00:00
struct vo_chain *vo_c = talloc_zero(NULL, struct vo_chain);
mpctx->vo_chain = vo_c;
vo_c->log = mpctx->log;
vo_c->vo = mpctx->video_out;
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
vo_c->filter =
mp_output_chain_create(mpctx->filter_root, MP_OUTPUT_CHAIN_VIDEO);
mp_output_chain_set_vo(vo_c->filter, vo_c->vo);
video: make decoder wrapper a filter Move dec_video.c to filters/f_decoder_wrapper.c. It essentially becomes a source filter. vd.h mostly disappears, because mp_filter takes care of the dataflow, but its remains are in struct mp_decoder_fns. One goal is to simplify dataflow by letting the filter framework handle it (or more accurately, using its conventions). One result is that the decode calls disappear from video.c, because we simply connect the decoder wrapper and the filter chain with mp_pin_connect(). Another goal is to eventually remove the code duplication between the audio and video paths for this. This commit prepares for this by trying to make f_decoder_wrapper.c extensible, so it can be used for audio as well later. Decoder framedropping changes a bit. It doesn't seem to be worse than before, and it's an obscure feature, so I'm content with its new state. Some special code that was apparently meant to avoid dropping too many frames in a row is removed, though. I'm not sure how the source code tree should be organized. For one, video/decode/vd_lavc.c is the only file in its directory, which is a bit annoying.
2018-01-28 09:08:45 +00:00
vo_c->filter->update_subtitles = filter_update_subtitles;
vo_c->filter->update_subtitles_ctx = mpctx;
player: refactor: eliminate MPContext.d_video Eventually we want the VO be driven by a A->V filter, so a decoder doesn't even have to exist. Some features definitely require a decoder though (like reporting the decoder in use, hardware decoding, etc.), so for each thing which accessed d_video, it has to be redecided if and how it can access decoder state. At least the "framedrop" property slightly changes semantics: you can now always set this property, even if no video is active. Some untested changes in this commit, but our bio-based distributed test suite has to take care of this.
2016-01-17 17:07:50 +00:00
player: make --lavfi-complex changeable at runtime Tends to be somewhat glitchy if subtitles are enabled, and you enable and disable tracks. On error, this will disable --lavfi-complex, which will result in whatever behavior.
2017-08-12 21:08:48 +00:00
if (track) {
player: add complex filter graph support See --lavfi-complex option. This is still quite rough. There's no support for dynamic configuration of any kind. There are probably corner cases where playback might freeze or burn 100% CPU (due to dataflow problems when interaction with libavfilter). Future possible plans might include: - freely switch tracks by providing some sort of default track graph label - automatically enabling audio visualization - automatically mix audio or stack video when multiple tracks are selected at once (similar to how multiple sub tracks can be selected)
2016-02-05 22:19:56 +00:00
vo_c->track = track;
track->vo_c = vo_c;
if (!init_video_decoder(mpctx, track))
goto err_out;
video: make decoder wrapper a filter Move dec_video.c to filters/f_decoder_wrapper.c. It essentially becomes a source filter. vd.h mostly disappears, because mp_filter takes care of the dataflow, but its remains are in struct mp_decoder_fns. One goal is to simplify dataflow by letting the filter framework handle it (or more accurately, using its conventions). One result is that the decode calls disappear from video.c, because we simply connect the decoder wrapper and the filter chain with mp_pin_connect(). Another goal is to eventually remove the code duplication between the audio and video paths for this. This commit prepares for this by trying to make f_decoder_wrapper.c extensible, so it can be used for audio as well later. Decoder framedropping changes a bit. It doesn't seem to be worse than before, and it's an obscure feature, so I'm content with its new state. Some special code that was apparently meant to avoid dropping too many frames in a row is removed, though. I'm not sure how the source code tree should be organized. For one, video/decode/vd_lavc.c is the only file in its directory, which is a bit annoying.
2018-01-28 09:08:45 +00:00
vo_c->dec_src = track->dec->f->pins[0];
video: fix passing down FPS to vf_vapoursynth To make this less of a mess, remove one of the redundant container_fps fields. Part of #5470.
2018-02-01 08:20:25 +00:00
vo_c->filter->container_fps = track->dec->fps;
player: make --lavfi-complex changeable at runtime Tends to be somewhat glitchy if subtitles are enabled, and you enable and disable tracks. On error, this will disable --lavfi-complex, which will result in whatever behavior.
2017-08-12 21:08:48 +00:00
vo_c->is_coverart = !!track->stream->attached_picture;
demux, player: fix playback of sparse video streams (w/ still images) Fixes several issues playing back mpegts with video streams marked as having "still images". For example, see this video which has frames only every 6s: https://s3.amazonaws.com/tmm1/music-choice.ts Changes include: - start playback right away, without waiting for first video frame - do not consider the sparse video stream in demuxer underrun detection - do not require multiple video frames for the VO - use audio as the master stream for demuxer metadata events - use audio stream for playback time Signed-off-by: Aman Gupta <aman@tmm1.net>
2018-05-03 02:29:11 +00:00
vo_c->is_sparse = track->stream->still_image;
player: cleanup update_fps() function It was called only in 2 places, one of them redundant (the container FPS can not change).
2015-04-20 21:21:38 +00:00
player: add complex filter graph support See --lavfi-complex option. This is still quite rough. There's no support for dynamic configuration of any kind. There are probably corner cases where playback might freeze or burn 100% CPU (due to dataflow problems when interaction with libavfilter). Future possible plans might include: - freely switch tracks by providing some sort of default track graph label - automatically enabling audio visualization - automatically mix audio or stack video when multiple tracks are selected at once (similar to how multiple sub tracks can be selected)
2016-02-05 22:19:56 +00:00
track->vo_c = vo_c;
vo_c->track = track;
video: make decoder wrapper a filter Move dec_video.c to filters/f_decoder_wrapper.c. It essentially becomes a source filter. vd.h mostly disappears, because mp_filter takes care of the dataflow, but its remains are in struct mp_decoder_fns. One goal is to simplify dataflow by letting the filter framework handle it (or more accurately, using its conventions). One result is that the decode calls disappear from video.c, because we simply connect the decoder wrapper and the filter chain with mp_pin_connect(). Another goal is to eventually remove the code duplication between the audio and video paths for this. This commit prepares for this by trying to make f_decoder_wrapper.c extensible, so it can be used for audio as well later. Decoder framedropping changes a bit. It doesn't seem to be worse than before, and it's an obscure feature, so I'm content with its new state. Some special code that was apparently meant to avoid dropping too many frames in a row is removed, though. I'm not sure how the source code tree should be organized. For one, video/decode/vd_lavc.c is the only file in its directory, which is a bit annoying.
2018-01-28 09:08:45 +00:00
mp_pin_connect(vo_c->filter->f->pins[0], vo_c->dec_src);
player: add complex filter graph support See --lavfi-complex option. This is still quite rough. There's no support for dynamic configuration of any kind. There are probably corner cases where playback might freeze or burn 100% CPU (due to dataflow problems when interaction with libavfilter). Future possible plans might include: - freely switch tracks by providing some sort of default track graph label - automatically enabling audio visualization - automatically mix audio or stack video when multiple tracks are selected at once (similar to how multiple sub tracks can be selected)
2016-02-05 22:19:56 +00:00
}
player: refactor: eliminate MPContext.d_video Eventually we want the VO be driven by a A->V filter, so a decoder doesn't even have to exist. Some features definitely require a decoder though (like reporting the decoder in use, hardware decoding, etc.), so for each thing which accessed d_video, it has to be redecided if and how it can access decoder state. At least the "framedrop" property slightly changes semantics: you can now always set this property, even if no video is active. Some untested changes in this commit, but our bio-based distributed test suite has to take care of this.
2016-01-17 17:07:50 +00:00
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
if (!recreate_video_filters(mpctx))
goto err_out;
Split mplayer.c mplayer.c was a bit too big. Split it into multiple files. I hope the way it's split makes sense. Maybe some things don't make too much sense, or go against intuition. These will fixed as soon as I notice them. Some files are a bit questionable (misc.c, osd.c, configfiles.c), and suggestions how to organize this better are welcome. Regressions are possible due to reorganized include statements. Obviously I didn't just copy mplayer.c's orgy of include statements, but recreated them for each file. It's easily possible that there are oversights and mistakes, which will show up on other platforms. There is one actual change: the public avutil.h include is removed from encode.h, and I tried to replace most FFMIN/FFMAX/av_clip uses. I consider using libavutil too much as dangerous, because the set of include files they recursively pull in is rather arbitrary and is different between FFmpeg and Libav.
2013-10-29 21:38:29 +00:00
player: make --stop-screensaver runtime-changeable Move the screensaver enable/disable determination to a central place, and call it if the stop-screensaver property is changed. Also, do not stop the screensaver when in idle mode (i.e. no file is loaded). Fixes #3615.
2016-10-02 10:33:34 +00:00
update_screensaver_state(mpctx);
Split mplayer.c mplayer.c was a bit too big. Split it into multiple files. I hope the way it's split makes sense. Maybe some things don't make too much sense, or go against intuition. These will fixed as soon as I notice them. Some files are a bit questionable (misc.c, osd.c, configfiles.c), and suggestions how to organize this better are welcome. Regressions are possible due to reorganized include statements. Obviously I didn't just copy mplayer.c's orgy of include statements, but recreated them for each file. It's easily possible that there are oversights and mistakes, which will show up on other platforms. There is one actual change: the public avutil.h include is removed from encode.h, and I tried to replace most FFMIN/FFMAX/av_clip uses. I consider using libavutil too much as dangerous, because the set of include files they recursively pull in is rather arbitrary and is different between FFmpeg and Libav.
2013-10-29 21:38:29 +00:00
player: refactor: eliminate MPContext.d_video Eventually we want the VO be driven by a A->V filter, so a decoder doesn't even have to exist. Some features definitely require a decoder though (like reporting the decoder in use, hardware decoding, etc.), so for each thing which accessed d_video, it has to be redecided if and how it can access decoder state. At least the "framedrop" property slightly changes semantics: you can now always set this property, even if no video is active. Some untested changes in this commit, but our bio-based distributed test suite has to take care of this.
2016-01-17 17:07:50 +00:00
vo_set_paused(vo_c->vo, mpctx->paused);
Split mplayer.c mplayer.c was a bit too big. Split it into multiple files. I hope the way it's split makes sense. Maybe some things don't make too much sense, or go against intuition. These will fixed as soon as I notice them. Some files are a bit questionable (misc.c, osd.c, configfiles.c), and suggestions how to organize this better are welcome. Regressions are possible due to reorganized include statements. Obviously I didn't just copy mplayer.c's orgy of include statements, but recreated them for each file. It's easily possible that there are oversights and mistakes, which will show up on other platforms. There is one actual change: the public avutil.h include is removed from encode.h, and I tried to replace most FFMIN/FFMAX/av_clip uses. I consider using libavutil too much as dangerous, because the set of include files they recursively pull in is rather arbitrary and is different between FFmpeg and Libav.
2013-10-29 21:38:29 +00:00
audio: change playback restart and resyncing This commit makes audio decoding non-blocking. If e.g. the network is too slow the playloop will just go to sleep, instead of blocking until enough data is available. For video, this was already done with commit 7083f88c. For audio, it's unfortunately much more complicated, because the audio decoder was used in a blocking manner. Large changes are required to get around this. The whole playback restart mechanism must be turned into a statemachine, especially since it has close interactions with video restart. Lots of video code is thus also changed. (For the record, I don't think switching this code to threads would make this conceptually easier: the code would still have to deal with external input while blocked, so these in-between states do get visible [and thus need to be handled] anyway. On the other hand, it certainly should be possible to modularize this code a bit better.) This will probably cause a bunch of regressions.
2014-07-28 18:40:43 +00:00
// If we switch on video again, ensure audio position matches up.
player: refactor: eliminate MPContext.d_audio
2016-01-21 21:24:20 +00:00
if (mpctx->ao_chain)
audio: change playback restart and resyncing This commit makes audio decoding non-blocking. If e.g. the network is too slow the playloop will just go to sleep, instead of blocking until enough data is available. For video, this was already done with commit 7083f88c. For audio, it's unfortunately much more complicated, because the audio decoder was used in a blocking manner. Large changes are required to get around this. The whole playback restart mechanism must be turned into a statemachine, especially since it has close interactions with video restart. Lots of video code is thus also changed. (For the record, I don't think switching this code to threads would make this conceptually easier: the code would still have to deal with external input while blocked, so these in-between states do get visible [and thus need to be handled] anyway. On the other hand, it certainly should be possible to modularize this code a bit better.) This will probably cause a bunch of regressions.
2014-07-28 18:40:43 +00:00
mpctx->audio_status = STATUS_SYNCING;
player: split seek_reset() This also reduces some code duplication with other parts of the code. The changfe is mostly cosmetic, although there are also some subtle changes in behavior. At least one change is that the big desync message is now printed after every seek.
2014-07-30 21:01:55 +00:00
reset_video_state(mpctx);
reset_subtitle_state(mpctx);
Split mplayer.c mplayer.c was a bit too big. Split it into multiple files. I hope the way it's split makes sense. Maybe some things don't make too much sense, or go against intuition. These will fixed as soon as I notice them. Some files are a bit questionable (misc.c, osd.c, configfiles.c), and suggestions how to organize this better are welcome. Regressions are possible due to reorganized include statements. Obviously I didn't just copy mplayer.c's orgy of include statements, but recreated them for each file. It's easily possible that there are oversights and mistakes, which will show up on other platforms. There is one actual change: the public avutil.h include is removed from encode.h, and I tried to replace most FFMIN/FFMAX/av_clip uses. I consider using libavutil too much as dangerous, because the set of include files they recursively pull in is rather arbitrary and is different between FFmpeg and Libav.
2013-10-29 21:38:29 +00:00
player: make --lavfi-complex changeable at runtime Tends to be somewhat glitchy if subtitles are enabled, and you enable and disable tracks. On error, this will disable --lavfi-complex, which will result in whatever behavior.
2017-08-12 21:08:48 +00:00
return;
Split mplayer.c mplayer.c was a bit too big. Split it into multiple files. I hope the way it's split makes sense. Maybe some things don't make too much sense, or go against intuition. These will fixed as soon as I notice them. Some files are a bit questionable (misc.c, osd.c, configfiles.c), and suggestions how to organize this better are welcome. Regressions are possible due to reorganized include statements. Obviously I didn't just copy mplayer.c's orgy of include statements, but recreated them for each file. It's easily possible that there are oversights and mistakes, which will show up on other platforms. There is one actual change: the public avutil.h include is removed from encode.h, and I tried to replace most FFMIN/FFMAX/av_clip uses. I consider using libavutil too much as dangerous, because the set of include files they recursively pull in is rather arbitrary and is different between FFmpeg and Libav.
2013-10-29 21:38:29 +00:00
err_out:
player: remove central uninit_player() function and flags mess Each subsystem (or similar thing) had an INITIALIZED_ flag assigned. The main use of this was that you could pass a bitmask of these flags to uninit_player(). Except in some situations where you wanted to uninitialize nearly everything, this wasn't really useful. Moreover, it was quite annoying that subsystems had most of the code in a specific file, but the uninit code in loadfile.c (because that's where uninit_player() was implemented). Simplify all this. Remove the flags; e.g. instead of testing for the INITIALIZED_AO flag, test whether mpctx->ao is set. Move uninit code to separate functions, e.g. uninit_audio_out().
2014-10-03 17:57:49 +00:00
uninit_video_chain(mpctx);
player: remove some further current_track dependencies Now it's used for initialization only for audio and video.
2016-02-05 22:41:44 +00:00
error_on_track(mpctx, track);
Split mplayer.c mplayer.c was a bit too big. Split it into multiple files. I hope the way it's split makes sense. Maybe some things don't make too much sense, or go against intuition. These will fixed as soon as I notice them. Some files are a bit questionable (misc.c, osd.c, configfiles.c), and suggestions how to organize this better are welcome. Regressions are possible due to reorganized include statements. Obviously I didn't just copy mplayer.c's orgy of include statements, but recreated them for each file. It's easily possible that there are oversights and mistakes, which will show up on other platforms. There is one actual change: the public avutil.h include is removed from encode.h, and I tried to replace most FFMIN/FFMAX/av_clip uses. I consider using libavutil too much as dangerous, because the set of include files they recursively pull in is rather arbitrary and is different between FFmpeg and Libav.
2013-10-29 21:38:29 +00:00
handle_force_window(mpctx, true);
}
// Try to refresh the video by doing a precise seek to the currently displayed
// frame. This can go wrong in all sorts of ways, so use sparingly.
void mp_force_video_refresh(struct MPContext *mpctx)
{
struct MPOpts *opts = mpctx->opts;
video: refactor: disentangle decoding/filtering some more This moves some code related to decoding from video.c to dec_video.c, and also removes some accesses to dec_video.c from the filtering code. dec_video.ch is starting to make sense, and simply returns video frames from a demuxer stream. The API exposed is also somewhat intended to be easily changeable to move decoding to a separate thread, if we ever want this (due to libavcodec already being threaded, I don't see much of a reason, but it might still be helpful).
2016-01-16 20:19:52 +00:00
struct vo_chain *vo_c = mpctx->vo_chain;
command: allow changing filters before video chain initialization Apparently this is more intuitive. Somewhat tricky, because of the odd state after loading a file but before initializing the VO.
2014-03-30 17:21:54 +00:00
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
if (!vo_c)
command: allow changing filters before video chain initialization Apparently this is more intuitive. Somewhat tricky, because of the odd state after loading a file but before initializing the VO.
2014-03-30 17:21:54 +00:00
return;
Split mplayer.c mplayer.c was a bit too big. Split it into multiple files. I hope the way it's split makes sense. Maybe some things don't make too much sense, or go against intuition. These will fixed as soon as I notice them. Some files are a bit questionable (misc.c, osd.c, configfiles.c), and suggestions how to organize this better are welcome. Regressions are possible due to reorganized include statements. Obviously I didn't just copy mplayer.c's orgy of include statements, but recreated them for each file. It's easily possible that there are oversights and mistakes, which will show up on other platforms. There is one actual change: the public avutil.h include is removed from encode.h, and I tried to replace most FFMIN/FFMAX/av_clip uses. I consider using libavutil too much as dangerous, because the set of include files they recursively pull in is rather arbitrary and is different between FFmpeg and Libav.
2013-10-29 21:38:29 +00:00
// If not paused, the next frame should come soon enough.
player: make refresh seeks slightly more robust Refresh seeks are automatically issued when changing filters, which improves user experience if these filters change buffering or such. The refresh seek could actually overwrite a previously ongoing seek: set pause yes set time-pos 10 set vf "" Here, the video code issued a refresh seek to the previous video position, which could be different from the previously triggered (and still ongoing) seek, this overwriting the seek. Factor all refresh seek handling into a new function, and make it handle ongoing seeks correctly. Remove the weird new canonical_pts field, which actually had no use. Fixes #4757.
2017-08-14 12:02:13 +00:00
if (opts->pause || mpctx->time_frame >= 0.5 ||
mpctx->video_status == STATUS_EOF)
video: don't force video refresh if video is restarting
2015-07-10 16:07:39 +00:00
{
player: make refresh seeks slightly more robust Refresh seeks are automatically issued when changing filters, which improves user experience if these filters change buffering or such. The refresh seek could actually overwrite a previously ongoing seek: set pause yes set time-pos 10 set vf "" Here, the video code issued a refresh seek to the previous video position, which could be different from the previously triggered (and still ongoing) seek, this overwriting the seek. Factor all refresh seek handling into a new function, and make it handle ongoing seeks correctly. Remove the weird new canonical_pts field, which actually had no use. Fixes #4757.
2017-08-14 12:02:13 +00:00
issue_refresh_seek(mpctx, MPSEEK_VERY_EXACT);
player: use symbolic constant for seek precision Meh.
2015-03-04 16:21:05 +00:00
}
Split mplayer.c mplayer.c was a bit too big. Split it into multiple files. I hope the way it's split makes sense. Maybe some things don't make too much sense, or go against intuition. These will fixed as soon as I notice them. Some files are a bit questionable (misc.c, osd.c, configfiles.c), and suggestions how to organize this better are welcome. Regressions are possible due to reorganized include statements. Obviously I didn't just copy mplayer.c's orgy of include statements, but recreated them for each file. It's easily possible that there are oversights and mistakes, which will show up on other platforms. There is one actual change: the public avutil.h include is removed from encode.h, and I tried to replace most FFMIN/FFMAX/av_clip uses. I consider using libavutil too much as dangerous, because the set of include files they recursively pull in is rather arbitrary and is different between FFmpeg and Libav.
2013-10-29 21:38:29 +00:00
}
video: make decoder wrapper a filter Move dec_video.c to filters/f_decoder_wrapper.c. It essentially becomes a source filter. vd.h mostly disappears, because mp_filter takes care of the dataflow, but its remains are in struct mp_decoder_fns. One goal is to simplify dataflow by letting the filter framework handle it (or more accurately, using its conventions). One result is that the decode calls disappear from video.c, because we simply connect the decoder wrapper and the filter chain with mp_pin_connect(). Another goal is to eventually remove the code duplication between the audio and video paths for this. This commit prepares for this by trying to make f_decoder_wrapper.c extensible, so it can be used for audio as well later. Decoder framedropping changes a bit. It doesn't seem to be worse than before, and it's an obscure feature, so I'm content with its new state. Some special code that was apparently meant to avoid dropping too many frames in a row is removed, though. I'm not sure how the source code tree should be organized. For one, video/decode/vd_lavc.c is the only file in its directory, which is a bit annoying.
2018-01-28 09:08:45 +00:00
static void check_framedrop(struct MPContext *mpctx, struct vo_chain *vo_c)
video: change everything Change how the video decoding loop works. The structure should now be a bit easier to follow. The interactions on format changes are (probably) simpler. This also aligns the decoding loop with future planned changes, such as moving various things to separate threads.
2014-05-01 21:53:18 +00:00
{
struct MPOpts *opts = mpctx->opts;
// check for frame-drop:
video: minor simplification of the old framedrop code No changes in functionality, other than being slightly more correct at stream EOF.
2014-08-13 19:50:23 +00:00
if (mpctx->video_status == STATUS_PLAYING && !mpctx->paused &&
player: refactor: some more minor decoder/output decoupling These changes don't make too much sense without context, but are preparation for later. Then the audio_src/video_src fields will be actually be NULL under circumstances.
2016-01-29 21:46:28 +00:00
mpctx->audio_status == STATUS_PLAYING && !ao_untimed(mpctx->ao) &&
video: make decoder wrapper a filter Move dec_video.c to filters/f_decoder_wrapper.c. It essentially becomes a source filter. vd.h mostly disappears, because mp_filter takes care of the dataflow, but its remains are in struct mp_decoder_fns. One goal is to simplify dataflow by letting the filter framework handle it (or more accurately, using its conventions). One result is that the decode calls disappear from video.c, because we simply connect the decoder wrapper and the filter chain with mp_pin_connect(). Another goal is to eventually remove the code duplication between the audio and video paths for this. This commit prepares for this by trying to make f_decoder_wrapper.c extensible, so it can be used for audio as well later. Decoder framedropping changes a bit. It doesn't seem to be worse than before, and it's an obscure feature, so I'm content with its new state. Some special code that was apparently meant to avoid dropping too many frames in a row is removed, though. I'm not sure how the source code tree should be organized. For one, video/decode/vd_lavc.c is the only file in its directory, which is a bit annoying.
2018-01-28 09:08:45 +00:00
vo_c->track && vo_c->track->dec && (opts->frame_dropping & 2))
video: change everything Change how the video decoding loop works. The structure should now be a bit easier to follow. The interactions on format changes are (probably) simpler. This also aligns the decoding loop with future planned changes, such as moving various things to separate threads.
2014-05-01 21:53:18 +00:00
{
video: fix passing down FPS to vf_vapoursynth To make this less of a mess, remove one of the redundant container_fps fields. Part of #5470.
2018-02-01 08:20:25 +00:00
float fps = vo_c->filter->container_fps;
video: make decoder wrapper a filter Move dec_video.c to filters/f_decoder_wrapper.c. It essentially becomes a source filter. vd.h mostly disappears, because mp_filter takes care of the dataflow, but its remains are in struct mp_decoder_fns. One goal is to simplify dataflow by letting the filter framework handle it (or more accurately, using its conventions). One result is that the decode calls disappear from video.c, because we simply connect the decoder wrapper and the filter chain with mp_pin_connect(). Another goal is to eventually remove the code duplication between the audio and video paths for this. This commit prepares for this by trying to make f_decoder_wrapper.c extensible, so it can be used for audio as well later. Decoder framedropping changes a bit. It doesn't seem to be worse than before, and it's an obscure feature, so I'm content with its new state. Some special code that was apparently meant to avoid dropping too many frames in a row is removed, though. I'm not sure how the source code tree should be organized. For one, video/decode/vd_lavc.c is the only file in its directory, which is a bit annoying.
2018-01-28 09:08:45 +00:00
// it's a crappy heuristic; avoid getting upset by incorrect fps
if (fps <= 20 || fps >= 500)
return;
double frame_time = 1.0 / fps;
// try to drop as many frames as we appear to be behind
vo_c->track->dec->attempt_framedrops =
MPCLAMP((mpctx->last_av_difference - 0.010) / frame_time, 0, 100);
player: change aspects of cover art handling Cover art handling is a disgusting hack that causes a mess in all components. And this will stay this way. This is the Xth time I've changed cover art handling, and that will probably also continue. But change the code such that cover art is injected into the demux packet stream, instead of having an explicit special case it in the decoder glue code. (This is somewhat more similar to the cover art hack in libavformat.) To avoid that the over art picture is decoded again on each seek, we need some additional "caching" in player/video.c. Decoding it after each seek would work as well, but since cover art pictures can be pretty huge, it's probably ok to invest some lines of code into caching it. One weird thing is that the cover art packet will remain queued after seeks, but that is probably not an issue. In exchange, we can drop the dec_video.c code, which is pretty convenient for one of the following commits. This code duplicates a bunch of lower-level decode calls and does icky messing with this weird state stuff, so I'm glad it goes away.
2017-01-10 13:03:41 +00:00
}
video: display last frame, drain frames on video reconfig Until now, the player didn't care to drain frames on video reconfig. Instead, the VO was reconfigured (i.e. resized) before the queued frames finished displaying. This can for example be observed by passing multiple images with different size as mf:// filename. Then the window would resize one frame before image with the new size is displayed. With --vo=vdpau, the effect is worse, because this VO queues more than 1 frame internally. Fix this by explicitly draining buffered frames before video reconfig. Raise the display time of the last frame. Otherwise, the last frame would be shown for a very short time only. This usually doesn't matter, but helps when playing image files. This is a byproduct of frame draining, because normally, video timing is based on the frames queued to the VO, and we can't do that with frames of different size or format. So we pretend that the frame before the change is the last frame in order to time it. This code is incorrect though: it tries to use the framerate, which often doesn't make sense. But it's good enough to test this code with mf://.
2013-12-10 18:33:11 +00:00
}
video: move some code around No functional changes. init_vo() is now needed a bit further down, and moving it keeps definition and use close. adjust_sync() will be used by a function further up in one of the following commits.
2014-08-22 12:19:01 +00:00
/* Modify video timing to match the audio timeline. There are two main
* reasons this is needed. First, video and audio can start from different
* positions at beginning of file or after a seek (MPlayer starts both
* immediately even if they have different pts). Second, the file can have
* audio timestamps that are inconsistent with the duration of the audio
* packets, for example two consecutive timestamp values differing by
* one second but only a packet with enough samples for half a second
* of playback between them.
*/
static void adjust_sync(struct MPContext *mpctx, double v_pts, double frame_time)
Split mplayer.c mplayer.c was a bit too big. Split it into multiple files. I hope the way it's split makes sense. Maybe some things don't make too much sense, or go against intuition. These will fixed as soon as I notice them. Some files are a bit questionable (misc.c, osd.c, configfiles.c), and suggestions how to organize this better are welcome. Regressions are possible due to reorganized include statements. Obviously I didn't just copy mplayer.c's orgy of include statements, but recreated them for each file. It's easily possible that there are oversights and mistakes, which will show up on other platforms. There is one actual change: the public avutil.h include is removed from encode.h, and I tried to replace most FFMIN/FFMAX/av_clip uses. I consider using libavutil too much as dangerous, because the set of include files they recursively pull in is rather arbitrary and is different between FFmpeg and Libav.
2013-10-29 21:38:29 +00:00
{
struct MPOpts *opts = mpctx->opts;
video: change everything Change how the video decoding loop works. The structure should now be a bit easier to follow. The interactions on format changes are (probably) simpler. This also aligns the decoding loop with future planned changes, such as moving various things to separate threads.
2014-05-01 21:53:18 +00:00
video: move some code around No functional changes. init_vo() is now needed a bit further down, and moving it keeps definition and use close. adjust_sync() will be used by a function further up in one of the following commits.
2014-08-22 12:19:01 +00:00
if (mpctx->audio_status != STATUS_PLAYING)
return;
client API, video: signal reconfig at the right time Filter reconfig can now happen a few frames before VO reconfig.
2014-05-01 23:28:52 +00:00
player: remove redundant variable mpctx->audio_delay always has the same value as opts->audio_delay. (This was not the case a long time ago, when the audio-delay property didn't actually write to opts->audio_delay. I think.)
2015-01-29 11:10:07 +00:00
double a_pts = written_audio_pts(mpctx) + opts->audio_delay - mpctx->delay;
video: move some code around No functional changes. init_vo() is now needed a bit further down, and moving it keeps definition and use close. adjust_sync() will be used by a function further up in one of the following commits.
2014-08-22 12:19:01 +00:00
double av_delay = a_pts - v_pts;
double change = av_delay * 0.1;
player: speed up audio/video re-sync when there is a huge delay when there is a huge delay between audio/video sync, it can take a really long time to converge back. this speeds up the resync time by increasing the max_change allowed per iteration. Signed-off-by: wm4 <wm4@nowhere>
2016-10-20 18:20:23 +00:00
double factor = fabs(av_delay) < 0.3 ? 0.1 : 0.4;
video: move some code around No functional changes. init_vo() is now needed a bit further down, and moving it keeps definition and use close. adjust_sync() will be used by a function further up in one of the following commits.
2014-08-22 12:19:01 +00:00
double max_change = opts->default_max_pts_correction >= 0 ?
player: speed up audio/video re-sync when there is a huge delay when there is a huge delay between audio/video sync, it can take a really long time to converge back. this speeds up the resync time by increasing the max_change allowed per iteration. Signed-off-by: wm4 <wm4@nowhere>
2016-10-20 18:20:23 +00:00
opts->default_max_pts_correction : frame_time * factor;
video: move some code around No functional changes. init_vo() is now needed a bit further down, and moving it keeps definition and use close. adjust_sync() will be used by a function further up in one of the following commits.
2014-08-22 12:19:01 +00:00
if (change < -max_change)
change = -max_change;
else if (change > max_change)
change = max_change;
mpctx->delay += change;
mpctx->total_avsync_change += change;
player: disable total-avsync-change update in display-sync mode The total-avsync-change property made no sense in display-sync mode (in addition to making not all that much sense in general).
2015-10-27 19:56:58 +00:00
if (mpctx->display_sync_active)
mpctx->total_avsync_change = 0;
Split mplayer.c mplayer.c was a bit too big. Split it into multiple files. I hope the way it's split makes sense. Maybe some things don't make too much sense, or go against intuition. These will fixed as soon as I notice them. Some files are a bit questionable (misc.c, osd.c, configfiles.c), and suggestions how to organize this better are welcome. Regressions are possible due to reorganized include statements. Obviously I didn't just copy mplayer.c's orgy of include statements, but recreated them for each file. It's easily possible that there are oversights and mistakes, which will show up on other platforms. There is one actual change: the public avutil.h include is removed from encode.h, and I tried to replace most FFMIN/FFMAX/av_clip uses. I consider using libavutil too much as dangerous, because the set of include files they recursively pull in is rather arbitrary and is different between FFmpeg and Libav.
2013-10-29 21:38:29 +00:00
}
video: pass future frames to VO Now the VO can request a number of future frames with the last parameter of vo_set_queue_params(). This will be helpful to fix the interpolation code. Note that the first frame (after playback start or seeking) will usually not have any future frames (to make seeking fast). Near the end of the file, the number of future frames will become lower as well.
2015-07-01 17:22:40 +00:00
// Make the frame at position 0 "known" to the playback logic. This must happen
// only once for each frame, so this function has to be called carefully.
// Generally, if position 0 gets a new frame, this must be called.
static void handle_new_frame(struct MPContext *mpctx)
video: refactor aspects of queue and EOF handling Instead of touching the 2-entry queue in mpctx->next_frame directly, move some of it to functions.
2015-03-25 12:33:34 +00:00
{
video: pass future frames to VO Now the VO can request a number of future frames with the last parameter of vo_set_queue_params(). This will be helpful to fix the interpolation code. Note that the first frame (after playback start or seeking) will usually not have any future frames (to make seeking fast). Near the end of the file, the number of future frames will become lower as well.
2015-07-01 17:22:40 +00:00
assert(mpctx->num_next_frames >= 1);
video: refactor aspects of queue and EOF handling Instead of touching the 2-entry queue in mpctx->next_frame directly, move some of it to functions.
2015-03-25 12:33:34 +00:00
double frame_time = 0;
video: pass future frames to VO Now the VO can request a number of future frames with the last parameter of vo_set_queue_params(). This will be helpful to fix the interpolation code. Note that the first frame (after playback start or seeking) will usually not have any future frames (to make seeking fast). Near the end of the file, the number of future frames will become lower as well.
2015-07-01 17:22:40 +00:00
double pts = mpctx->next_frames[0]->pts;
demux, player: fix playback of sparse video streams (w/ still images) Fixes several issues playing back mpegts with video streams marked as having "still images". For example, see this video which has frames only every 6s: https://s3.amazonaws.com/tmm1/music-choice.ts Changes include: - start playback right away, without waiting for first video frame - do not consider the sparse video stream in demuxer underrun detection - do not require multiple video frames for the VO - use audio as the master stream for demuxer metadata events - use audio stream for playback time Signed-off-by: Aman Gupta <aman@tmm1.net>
2018-05-03 02:29:11 +00:00
bool is_sparse = mpctx->vo_chain && mpctx->vo_chain->is_sparse;
video: refactor aspects of queue and EOF handling Instead of touching the 2-entry queue in mpctx->next_frame directly, move some of it to functions.
2015-03-25 12:33:34 +00:00
if (mpctx->video_pts != MP_NOPTS_VALUE) {
frame_time = pts - mpctx->video_pts;
demux, player: fix playback of sparse video streams (w/ still images) Fixes several issues playing back mpegts with video streams marked as having "still images". For example, see this video which has frames only every 6s: https://s3.amazonaws.com/tmm1/music-choice.ts Changes include: - start playback right away, without waiting for first video frame - do not consider the sparse video stream in demuxer underrun detection - do not require multiple video frames for the VO - use audio as the master stream for demuxer metadata events - use audio stream for playback time Signed-off-by: Aman Gupta <aman@tmm1.net>
2018-05-03 02:29:11 +00:00
double tolerance = mpctx->demuxer->ts_resets_possible &&
!is_sparse ? 5 : 1e4;
video: better heuristic for timestamp resets Reduce the default tolerance for timestamp jumps from 60 to 15 seconds. For .ts files, where ts_resets_possible coming from AVFMT_TS_DISCONT is set, apply a more sophisticated heuristic. It's clear that such a file wouldn't have a framerate below, say, 23hz. If the demuxer reports a lower fps, we allow longer PTS jumps. This should replace long pauses on discontinuities with .ts files with at most a short stutter. Of course, all kinds of things could go wrong anyway if the source is VFR, or FFmpeg's frame rate detection fails in some other way. I haven't found such a file yet, though.
2015-05-20 12:24:55 +00:00
if (frame_time <= 0 || frame_time >= tolerance) {
// Assume a discontinuity.
video: refactor aspects of queue and EOF handling Instead of touching the 2-entry queue in mpctx->next_frame directly, move some of it to functions.
2015-03-25 12:33:34 +00:00
MP_WARN(mpctx, "Invalid video timestamp: %f -> %f\n",
mpctx->video_pts, pts);
player: resync audio only on larger timestamp discontinuities Helps with files that have occasional broken timestamps. For larger discontinuities, e.g. caused by actual timestamp resets, we still want to realign audio. (I guess in general, this should be removed and replaced by a more general resync-on-desync logic, but not now.)
2015-12-04 15:55:55 +00:00
frame_time = 0;
video: refactor aspects of queue and EOF handling Instead of touching the 2-entry queue in mpctx->next_frame directly, move some of it to functions.
2015-03-25 12:33:34 +00:00
}
}
mpctx->delay -= frame_time;
video: take first frame into account in audio-sync mode It appears commit 4ad68d94523c3d101a broke handling the first video frame duration through roundabout ways (I think because the duration of the first frame was now available at all in the normal case). The first frame was cut short, which showed up especially with looping, or if the file had a low FPS. This questionable change seems to fix it without breaking any other known cases => push and call it a day. The display-sync mode did not have this problem. Fixes: #7150
2019-11-16 12:47:23 +00:00
mpctx->time_frame += frame_time / mpctx->video_speed;
if (mpctx->video_status >= STATUS_PLAYING)
video: refactor aspects of queue and EOF handling Instead of touching the 2-entry queue in mpctx->next_frame directly, move some of it to functions.
2015-03-25 12:33:34 +00:00
adjust_sync(mpctx, pts, frame_time);
MP_TRACE(mpctx, "frametime=%5.3f\n", frame_time);
}
vo: change internal API for drawing frames draw_image_timed is renamed to draw_frame. struct frame_timing is renamed to vo_frame. flip_page_timed is merged into draw_frame (the additional parameters are part of struct vo_frame). draw_frame also deprecates VOCTRL_REDRAW_FRAME, and replaces it with a method that works for both VOs which can cache the current frame, and VOs which need to redraw it anyway. This is preparation to making the interpolation and (work in progress) display sync code saner. Lots of other refactoring, and also some simplifications.
2015-07-01 17:24:28 +00:00
// Remove the first frame in mpctx->next_frames
static void shift_frames(struct MPContext *mpctx)
{
if (mpctx->num_next_frames < 1)
return;
talloc_free(mpctx->next_frames[0]);
for (int n = 0; n < mpctx->num_next_frames - 1; n++)
mpctx->next_frames[n] = mpctx->next_frames[n + 1];
mpctx->num_next_frames -= 1;
}
video: pass future frames to VO Now the VO can request a number of future frames with the last parameter of vo_set_queue_params(). This will be helpful to fix the interpolation code. Note that the first frame (after playback start or seeking) will usually not have any future frames (to make seeking fast). Near the end of the file, the number of future frames will become lower as well.
2015-07-01 17:22:40 +00:00
static int get_req_frames(struct MPContext *mpctx, bool eof)
{
// On EOF, drain all frames.
player: always require a future frame with display-sync enabled We need a frame duration even on start, because the number of vsyncs the frame is shown is predetermined. (vo_opengl actually makes use of this property in certain cases.)
2015-11-13 21:42:42 +00:00
if (eof)
video: pass future frames to VO Now the VO can request a number of future frames with the last parameter of vo_set_queue_params(). This will be helpful to fix the interpolation code. Note that the first frame (after playback start or seeking) will usually not have any future frames (to make seeking fast). Near the end of the file, the number of future frames will become lower as well.
2015-07-01 17:22:40 +00:00
return 1;
video: do not buffer extra frames with VO_CAP_NORETAIN outputs This fixes playback stalls on some mediacodec hardware decoders, which expect that frame buffers will be rendered and returned back to the decoder as soon as possible. Specifically, the issue was observed on an NVidia SHIELD Android TV, only when playing an H264 sample which switched between interlaced and non-interlaced frames. On an interlacing change, the decoder expects all outstanding frames would be returned to it before it would emit any new frames. Since a single extra frame always remained buffered by mpv, playback would stall. After this commit, no extra frames are buffered by mpv when using vo_mediacodec_embed.
2018-02-16 03:57:34 +00:00
if (mpctx->video_out->driver->caps & VO_CAP_NORETAIN)
return 1;
demux, player: fix playback of sparse video streams (w/ still images) Fixes several issues playing back mpegts with video streams marked as having "still images". For example, see this video which has frames only every 6s: https://s3.amazonaws.com/tmm1/music-choice.ts Changes include: - start playback right away, without waiting for first video frame - do not consider the sparse video stream in demuxer underrun detection - do not require multiple video frames for the VO - use audio as the master stream for demuxer metadata events - use audio stream for playback time Signed-off-by: Aman Gupta <aman@tmm1.net>
2018-05-03 02:29:11 +00:00
if (mpctx->vo_chain && mpctx->vo_chain->is_sparse)
return 1;
video: don't read ahead a frame in --untimed mode The extra frame is used to compute the exact frame duration. But frame drop is disabvled with --untimed.
2018-03-02 13:03:41 +00:00
if (mpctx->opts->untimed || mpctx->video_out->driver->untimed)
return 1;
video: always decode 2 frames on playback restart Unless --video-latency-hacks, always decode 2 frames on playback restart. This in turn will always compute the correct frame duration (even for the first frame), which in turn happens to fix that playback with an image at the beginning breaks display. If a still image precedes video, and the size/format of the frame is different from that of the video following it, the incorrect frame duration caused vo_reconfig2() to be called early, causing the window to resize, and the renderer to clear the image to black. Specifically, it hit the default value of 1 second duration (for still images), so the image was displayed for 1 second, and changed to black until the next proper video frame was displayed. Normally this does not happen. Even if a video file displays still images, it normally repeats the still image at the video's FPS (which is sane). But you can construct such files, or use EDL to construct something similarly behaving. This change may increase seek latency a bit in audio video-sync mode (the default). It needs to wait until 2 frames are decoded, before it bothers to display the first frame. This is done even when seeking. In theory it might be good to introduce a "seek preview" mode, which shows the target image without all the preparations needed for starting playback. (For example, it could not decode audio.) But since I'm using video-sync=display-resample, which already needed to always decode 2 frames, I don't think this is a terribly high priority, nor do I consider the slightly slower seeking a regression. Fixes: #6765
2019-10-06 20:20:10 +00:00
// Normally require at least 2 frames, so we can compute a frame duration.
video: add option to reduce latency by 1 or 2 frames The playback start logic explicitly waits until the first frame has been displayed. Usually this will introduce a wait of 1 vsync. For normal playback this doesn't matter, but with respect to low latency needs, this only leads to additional data getting queued up in the demuxer or network buffers. Another thing is that the timing logic decodes 1 frame ahead (= 1 frame extra latency) to determine the exact duration of a frame. To be fair, there doesn't really seem to be a hard reason why this is needed. With the current code, enabling the option does lead to A/V desync sometimes (if the demuxer FPS is too inaccurate), and also frame drops at playback start in some situations. But this all seems to be avoidable, if the timing logic were to be rewritten completely, which should probably happen in the future. Thus the new option comes with the warning that it can be removed any time. This is also why the option has "hack" in the name.
2018-03-02 14:37:34 +00:00
int min = mpctx->opts->video_latency_hacks ? 1 : 2;
player: always require a future frame with display-sync enabled We need a frame duration even on start, because the number of vsyncs the frame is shown is predetermined. (vo_opengl actually makes use of this property in certain cases.)
2015-11-13 21:42:42 +00:00
// On the first frame, output a new frame as quickly as possible.
if (mpctx->video_pts == MP_NOPTS_VALUE)
video: always decode 2 frames on playback restart Unless --video-latency-hacks, always decode 2 frames on playback restart. This in turn will always compute the correct frame duration (even for the first frame), which in turn happens to fix that playback with an image at the beginning breaks display. If a still image precedes video, and the size/format of the frame is different from that of the video following it, the incorrect frame duration caused vo_reconfig2() to be called early, causing the window to resize, and the renderer to clear the image to black. Specifically, it hit the default value of 1 second duration (for still images), so the image was displayed for 1 second, and changed to black until the next proper video frame was displayed. Normally this does not happen. Even if a video file displays still images, it normally repeats the still image at the video's FPS (which is sane). But you can construct such files, or use EDL to construct something similarly behaving. This change may increase seek latency a bit in audio video-sync mode (the default). It needs to wait until 2 frames are decoded, before it bothers to display the first frame. This is done even when seeking. In theory it might be good to introduce a "seek preview" mode, which shows the target image without all the preparations needed for starting playback. (For example, it could not decode audio.) But since I'm using video-sync=display-resample, which already needed to always decode 2 frames, I don't think this is a terribly high priority, nor do I consider the slightly slower seeking a regression. Fixes: #6765
2019-10-06 20:20:10 +00:00
return min;
player: always require a future frame with display-sync enabled We need a frame duration even on start, because the number of vsyncs the frame is shown is predetermined. (vo_opengl actually makes use of this property in certain cases.)
2015-11-13 21:42:42 +00:00
vo: minor simplification for queue size handling Instead of calling it "future frames" and adding or subtracting 1 from it, always call it "requested frames". This simplifies it a bit. MPContext.next_frames had 2 added to it; this was mainly to ensure a minimum size of 2. Drop it and assume VO_MAX_REQ_FRAMES is at least 2; together with the other changes, this can be the exact size of the array.
2015-07-20 19:12:46 +00:00
int req = vo_get_num_req_frames(mpctx->video_out);
video: add option to reduce latency by 1 or 2 frames The playback start logic explicitly waits until the first frame has been displayed. Usually this will introduce a wait of 1 vsync. For normal playback this doesn't matter, but with respect to low latency needs, this only leads to additional data getting queued up in the demuxer or network buffers. Another thing is that the timing logic decodes 1 frame ahead (= 1 frame extra latency) to determine the exact duration of a frame. To be fair, there doesn't really seem to be a hard reason why this is needed. With the current code, enabling the option does lead to A/V desync sometimes (if the demuxer FPS is too inaccurate), and also frame drops at playback start in some situations. But this all seems to be avoidable, if the timing logic were to be rewritten completely, which should probably happen in the future. Thus the new option comes with the warning that it can be removed any time. This is also why the option has "hack" in the name.
2018-03-02 14:37:34 +00:00
return MPCLAMP(req, min, MP_ARRAY_SIZE(mpctx->next_frames) - 1);
video: pass future frames to VO Now the VO can request a number of future frames with the last parameter of vo_set_queue_params(). This will be helpful to fix the interpolation code. Note that the first frame (after playback start or seeking) will usually not have any future frames (to make seeking fast). Near the end of the file, the number of future frames will become lower as well.
2015-07-01 17:22:40 +00:00
}
video: refactor aspects of queue and EOF handling Instead of touching the 2-entry queue in mpctx->next_frame directly, move some of it to functions.
2015-03-25 12:33:34 +00:00
// Whether it's fine to call add_new_frame() now.
static bool needs_new_frame(struct MPContext *mpctx)
{
video: pass future frames to VO Now the VO can request a number of future frames with the last parameter of vo_set_queue_params(). This will be helpful to fix the interpolation code. Note that the first frame (after playback start or seeking) will usually not have any future frames (to make seeking fast). Near the end of the file, the number of future frames will become lower as well.
2015-07-01 17:22:40 +00:00
return mpctx->num_next_frames < get_req_frames(mpctx, false);
video: refactor aspects of queue and EOF handling Instead of touching the 2-entry queue in mpctx->next_frame directly, move some of it to functions.
2015-03-25 12:33:34 +00:00
}
video: pass future frames to VO Now the VO can request a number of future frames with the last parameter of vo_set_queue_params(). This will be helpful to fix the interpolation code. Note that the first frame (after playback start or seeking) will usually not have any future frames (to make seeking fast). Near the end of the file, the number of future frames will become lower as well.
2015-07-01 17:22:40 +00:00
// Queue a frame to mpctx->next_frames[]. Call only if needs_new_frame() signals ok.
video: refactor aspects of queue and EOF handling Instead of touching the 2-entry queue in mpctx->next_frame directly, move some of it to functions.
2015-03-25 12:33:34 +00:00
static void add_new_frame(struct MPContext *mpctx, struct mp_image *frame)
{
player: simplify backstepping Basically reimplement it. The old implementation was quite stupid, and was probably done this way because video filtering and output used to be way less decoupled. Now we can reimplement it in a very simple way: when backstepping, seek to current time, but keep the last frame that was supposed to be discarded when reaching the target time. When the seek finishes, prepend the saved frame to the video frame queue. A disadvantage is that the new implementation fails to skip over timeline boundaries (ordered chapters etc.), but this never worked properly anyway. It's possible that this will be fixed some time in the future.
2016-01-12 22:49:00 +00:00
assert(mpctx->num_next_frames < MP_ARRAY_SIZE(mpctx->next_frames));
video: refactor aspects of queue and EOF handling Instead of touching the 2-entry queue in mpctx->next_frame directly, move some of it to functions.
2015-03-25 12:33:34 +00:00
assert(frame);
video: pass future frames to VO Now the VO can request a number of future frames with the last parameter of vo_set_queue_params(). This will be helpful to fix the interpolation code. Note that the first frame (after playback start or seeking) will usually not have any future frames (to make seeking fast). Near the end of the file, the number of future frames will become lower as well.
2015-07-01 17:22:40 +00:00
mpctx->next_frames[mpctx->num_next_frames++] = frame;
if (mpctx->num_next_frames == 1)
handle_new_frame(mpctx);
video: refactor aspects of queue and EOF handling Instead of touching the 2-entry queue in mpctx->next_frame directly, move some of it to functions.
2015-03-25 12:33:34 +00:00
}
video: check whether there are enough frames after filtering Move the check to a function. Run the check a second time after decoding/filtering. This second check is strictly speaking redundant (which is why it wasn't done until now), but it avoids a useless playloop iteration.
2014-09-18 17:20:22 +00:00
// Enough video filtered already to push one frame to the VO?
video: refactor aspects of queue and EOF handling Instead of touching the 2-entry queue in mpctx->next_frame directly, move some of it to functions.
2015-03-25 12:33:34 +00:00
// Set eof to true if no new frames are to be expected.
static bool have_new_frame(struct MPContext *mpctx, bool eof)
video: check whether there are enough frames after filtering Move the check to a function. Run the check a second time after decoding/filtering. This second check is strictly speaking redundant (which is why it wasn't done until now), but it avoids a useless playloop iteration.
2014-09-18 17:20:22 +00:00
{
video: pass future frames to VO Now the VO can request a number of future frames with the last parameter of vo_set_queue_params(). This will be helpful to fix the interpolation code. Note that the first frame (after playback start or seeking) will usually not have any future frames (to make seeking fast). Near the end of the file, the number of future frames will become lower as well.
2015-07-01 17:22:40 +00:00
return mpctx->num_next_frames >= get_req_frames(mpctx, eof);
video: check whether there are enough frames after filtering Move the check to a function. Run the check a second time after decoding/filtering. This second check is strictly speaking redundant (which is why it wasn't done until now), but it avoids a useless playloop iteration.
2014-09-18 17:20:22 +00:00
}
video: pass future frames to VO Now the VO can request a number of future frames with the last parameter of vo_set_queue_params(). This will be helpful to fix the interpolation code. Note that the first frame (after playback start or seeking) will usually not have any future frames (to make seeking fast). Near the end of the file, the number of future frames will become lower as well.
2015-07-01 17:22:40 +00:00
// Fill mpctx->next_frames[] with a newly filtered or decoded image.
video: use symbolic constants instead of magic integers In my opinion this is not really necessary, since there's only a single user of update_video(), but others reading this code would probably hate me for using magic integer values instead of symbolic constants. This should be a purely cosmetic commit; any changes in behavior are bugs.
2014-07-18 13:11:21 +00:00
// returns VD_* code
player: make repeated hr-seeks past EOF trigger EOF as expected If you have a normal file with audio and video, and keep "spamming" forward hr-seeks, the player just kept showing the last video frame instead of exiting or playing the next file. This started happening since commit 6bcda94cb. Although not a bug per se, it was odd, and very user-noticable. The main problem was that the pending seek command was processed before the EOF was "noticed". Processing the command reset everything, so the player did not terminate playback, but repeated the seek. This commit restores the old behavior. For one, it makes video return the correct status (video.c). The parameter is a bit ugly, but better than duplicating the logic or having another MPContext field. (As a minor detail, setting r=VD_EOF makes sure have_new_frame() returns true, rather than going through another iteration or whatever the hell will happen instead, which would clobber logical_eof.) Another thing is making the seek logic actually wait until the seek outcome has been determined if audio is also active. Audio needs to wait for video in order to get the video seek target position. (Which in turn is because hr-seek still "snaps" to video frames. You can't seek in between two frames, so audio can't just use the seek target, but always has to wait on the timestamp of the video frame. This has other disadvantages and is a misdesign, but not something I'll fix today.) In theory, this might make hr-seeks less responsive, because it needs to fully decode/filter the audio too, but in practice most time is spent on video, which had to be fully decoded before this change. (In general, hr-seek could probably just show a random frame when a queued hr-seek overrides the current hr-seek, which would probably lead to a better user experience, but that's out of scope.) Fixes: #7206
2019-12-14 13:17:16 +00:00
static int video_output_image(struct MPContext *mpctx, bool *logical_eof)
Split mplayer.c mplayer.c was a bit too big. Split it into multiple files. I hope the way it's split makes sense. Maybe some things don't make too much sense, or go against intuition. These will fixed as soon as I notice them. Some files are a bit questionable (misc.c, osd.c, configfiles.c), and suggestions how to organize this better are welcome. Regressions are possible due to reorganized include statements. Obviously I didn't just copy mplayer.c's orgy of include statements, but recreated them for each file. It's easily possible that there are oversights and mistakes, which will show up on other platforms. There is one actual change: the public avutil.h include is removed from encode.h, and I tried to replace most FFMIN/FFMAX/av_clip uses. I consider using libavutil too much as dangerous, because the set of include files they recursively pull in is rather arbitrary and is different between FFmpeg and Libav.
2013-10-29 21:38:29 +00:00
{
player: use different variable to indicate coverart Slightly better.
2016-02-01 21:14:32 +00:00
struct vo_chain *vo_c = mpctx->vo_chain;
video: make track switching work for external images Until now, this didn't work, since the external image had pts 0; so enabling video at a later time did nothing, because the image was discarded. Since hrseek now ends on the last frame (instead of nothing), reusing the hrseek mechanism solves this, and we don't even need to treat the cursed coverart case separately.
2019-11-17 00:37:59 +00:00
bool hrseek = false;
double hrseek_pts = mpctx->hrseek_pts;
player: don't apply weird timestamp tolerance on backstep Hr-seek has some sort of tolerance against timestamps, where it allows for up to 5ms deviation. This means it can work only for videos with up to 200 FPS framerate. There were complains about how it doesn't work with videos beyond some high fps. (1000 was mentioned, although that sounds more like it's about the limit that .mkv has.) I suspect this is because otherwise, it might be hard to hit a timestamp with --start, which specifies timestamps as integer, and thus will most likely never represent a timestamp exactly. Another part of the problem is that mpv uses 64 bit floats for timestamps, so fractional parts are never represented exactly. (Both the "tolerance" and using floats for timestamps were things introduced before my time.) Anyway, in the backstep case, we can be relatively sure that the timestamp will be exact (as in, the same unmodified value that was returned by the filter chain), so we can make an exception for that, in order to fix backstep. Untested. (For that you have users.) May help with #7208.
2019-12-03 20:43:32 +00:00
double tolerance = mpctx->hrseek_backstep ? 0 : .005;
video: make track switching work for external images Until now, this didn't work, since the external image had pts 0; so enabling video at a later time did nothing, because the image was discarded. Since hrseek now ends on the last frame (instead of nothing), reusing the hrseek mechanism solves this, and we don't even need to treat the cursed coverart case separately.
2019-11-17 00:37:59 +00:00
if (mpctx->video_status == STATUS_SYNCING) {
hrseek = mpctx->hrseek_active;
// playback_pts is normally only set when audio and video have started
// playing normally. If video is in syncing mode, then this must mean
// video was just enabled via track switching - skip to current time.
if (!hrseek && mpctx->playback_pts != MP_NOPTS_VALUE) {
hrseek = true;
hrseek_pts = mpctx->playback_pts;
}
}
player: when seeking past EOF with --keep-open, seek to last frame It feels strange that seeking past EOF with --keep-open actually leaves the player at a random position. You can't even unpause, because the demuxer is in the EOF state, and what you see on screen is just what was around before the seek. Improve this by attempting to seek to the last video frame if EOF happens. We explicitly don't do this if EOF was reached normally to increase robustness (if the VO got a frame since the last seek, it obviously means we had normal playback before EOF). If an error happens when trying to find the last frame (such as not actually finding a last frame because e.g. the demuxer misbehaves), this will probably turn your CPU into a heater. There is no logic to prevent reinitiating the last-frame search if the last-frame search reached EOF. (Pausing usually prevents that EOF is reached again after a successful last-frame search.) Fixes #819.
2014-12-07 01:47:09 +00:00
player: use different variable to indicate coverart Slightly better.
2016-02-01 21:14:32 +00:00
if (vo_c->is_coverart) {
video: refactor queue handling This simplifies the code, and fixes an odd bug: the second-last frame was displayed for a very short duration if framedrop was enabled. The reason was that basically the time difference between second-last and last frame were skipped, because at this point EOF was already signaled. Also see commit b0959488 for a similar issue in the same code. This removes the messiness of the next_frame 2-frame queue, and strictly runs the "new frame" code when a frame is moved to the first position of the queue, instead of somehow messing with return codes. This also merges update_video() into video_output_image().
2014-08-22 12:20:30 +00:00
if (vo_has_frame(mpctx->video_out))
video: use symbolic constants instead of magic integers In my opinion this is not really necessary, since there's only a single user of update_video(), but others reading this code would probably hate me for using magic integer values instead of symbolic constants. This should be a purely cosmetic commit; any changes in behavior are bugs.
2014-07-18 13:11:21 +00:00
return VD_EOF;
video: refactor: disentangle decoding/filtering some more This moves some code related to decoding from video.c to dec_video.c, and also removes some accesses to dec_video.c from the filtering code. dec_video.ch is starting to make sense, and simply returns video frames from a demuxer stream. The API exposed is also somewhat intended to be easily changeable to move decoding to a separate thread, if we ever want this (due to libavcodec already being threaded, I don't see much of a reason, but it might still be helpful).
2016-01-16 20:19:52 +00:00
hrseek = false;
video: reduce non-sense messages when playing coverart Don't print PTS warnings by skipping the normal video path.
2014-08-13 19:55:46 +00:00
}
video: refactor aspects of queue and EOF handling Instead of touching the 2-entry queue in mpctx->next_frame directly, move some of it to functions.
2015-03-25 12:33:34 +00:00
if (have_new_frame(mpctx, false))
video: filter new frames at a better time (2) We generally want 2 things: 1. minimal wakeups for decoding each frame 2. minimal number of frames decoded on continuous seeking Commit 35810cb8 changed this a bit, and fixed 1. But it broke 2., and now it decodes 2 frames instead of 1 when you keep seeking (arrow key held down or such). This made seeking appear slower. Fix this by making the logic more explicit. In particular, call the filters only if we actually try to get a new frame. When playing with --no-audio and all other distractions disabled (like OSC), it still wakes up 2 times per frame - but the second time is merely because the VO didn't accept the new frame yet.
2014-09-22 16:06:59 +00:00
return VD_NEW_FRAME;
video: filter new frames at a better time Move this code below the code that "shifts" the newly filtered frame. This allows us to skip a useless playloop iteration later, because obviously we need to filter a new frame after the previous frame has been "shifted", and not before that.
2014-09-18 17:17:38 +00:00
// Get a new frame if we need one.
video: filter new frames at a better time (2) We generally want 2 things: 1. minimal wakeups for decoding each frame 2. minimal number of frames decoded on continuous seeking Commit 35810cb8 changed this a bit, and fixed 1. But it broke 2., and now it decodes 2 frames instead of 1 when you keep seeking (arrow key held down or such). This made seeking appear slower. Fix this by making the logic more explicit. In particular, call the filters only if we actually try to get a new frame. When playing with --no-audio and all other distractions disabled (like OSC), it still wakes up 2 times per frame - but the second time is merely because the VO didn't accept the new frame yet.
2014-09-22 16:06:59 +00:00
int r = VD_PROGRESS;
video: refactor aspects of queue and EOF handling Instead of touching the 2-entry queue in mpctx->next_frame directly, move some of it to functions.
2015-03-25 12:33:34 +00:00
if (needs_new_frame(mpctx)) {
video: filter new frames at a better time (2) We generally want 2 things: 1. minimal wakeups for decoding each frame 2. minimal number of frames decoded on continuous seeking Commit 35810cb8 changed this a bit, and fixed 1. But it broke 2., and now it decodes 2 frames instead of 1 when you keep seeking (arrow key held down or such). This made seeking appear slower. Fix this by making the logic more explicit. In particular, call the filters only if we actually try to get a new frame. When playing with --no-audio and all other distractions disabled (like OSC), it still wakes up 2 times per frame - but the second time is merely because the VO didn't accept the new frame yet.
2014-09-22 16:06:59 +00:00
// Filter a new frame.
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_image *img = NULL;
struct mp_frame frame = mp_pin_out_read(vo_c->filter->f->pins[1]);
video: make decoder wrapper a filter Move dec_video.c to filters/f_decoder_wrapper.c. It essentially becomes a source filter. vd.h mostly disappears, because mp_filter takes care of the dataflow, but its remains are in struct mp_decoder_fns. One goal is to simplify dataflow by letting the filter framework handle it (or more accurately, using its conventions). One result is that the decode calls disappear from video.c, because we simply connect the decoder wrapper and the filter chain with mp_pin_connect(). Another goal is to eventually remove the code duplication between the audio and video paths for this. This commit prepares for this by trying to make f_decoder_wrapper.c extensible, so it can be used for audio as well later. Decoder framedropping changes a bit. It doesn't seem to be worse than before, and it's an obscure feature, so I'm content with its new state. Some special code that was apparently meant to avoid dropping too many frames in a row is removed, though. I'm not sure how the source code tree should be organized. For one, video/decode/vd_lavc.c is the only file in its directory, which is a bit annoying.
2018-01-28 09:08:45 +00:00
if (frame.type == MP_FRAME_NONE) {
r = vo_c->filter->got_output_eof ? VD_EOF : VD_WAIT;
} else if (frame.type == MP_FRAME_EOF) {
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
r = VD_EOF;
} else if (frame.type == MP_FRAME_VIDEO) {
img = frame.data;
} else {
MP_ERR(mpctx, "unexpected frame type %s\n",
mp_frame_type_str(frame.type));
mp_frame_unref(&frame);
return VD_ERROR;
}
video: filter new frames at a better time Move this code below the code that "shifts" the newly filtered frame. This allows us to skip a useless playloop iteration later, because obviously we need to filter a new frame after the previous frame has been "shifted", and not before that.
2014-09-18 17:17:38 +00:00
if (img) {
player: minor simplification No need to pass endpts down in such a dumb way. Also remove an outdated comment somewhere.
2016-02-27 19:02:51 +00:00
double endpts = get_play_end_pts(mpctx);
player: fix --end for backwards playback We need to transform the timestamp returned by get_play_end_pts(). I considered making it return the transformed timestamp directly. There are 4 callers; 2 need a transformed timestamps, 2 don't. So I guess it doesn't matter.
2019-05-25 21:04:26 +00:00
if (endpts != MP_NOPTS_VALUE)
endpts *= mpctx->play_dir;
video: don't discard video frames after endpts Instead of letting it keep decoding by trying to find a new frame, "plug" the frame queue by not removing it. (Or actually, by putting it back instead of discarding it.) Matters for seamless looping (following commits), and possibly some other corner cases. The added function vf_unread_output_frame() is a bit of a sin, but still reasonable, since its implementation is trivial.
2016-08-18 18:37:25 +00:00
if ((endpts != MP_NOPTS_VALUE && img->pts >= endpts) ||
mpctx->max_frames == 0)
{
video: rewrite filtering glue code Get rid of the old vf.c code. Replace it with a generic filtering framework, which can potentially handle more than just --vf. At least reimplementing --af with this code is planned. This changes some --vf semantics (including runtime behavior and the "vf" command). The most important ones are listed in interface-changes. vf_convert.c is renamed to f_swscale.c. It is now an internal filter that can not be inserted by the user manually. f_lavfi.c is a refactor of player/lavfi.c. The latter will be removed once --lavfi-complex is reimplemented on top of f_lavfi.c. (which is conceptually easy, but a big mess due to the data flow changes). The existing filters are all changed heavily. The data flow of the new filter framework is different. Especially EOF handling changes - EOF is now a "frame" rather than a state, and must be passed through exactly once. Another major thing is that all filters must support dynamic format changes. The filter reconfig() function goes away. (This sounds complex, but since all filters need to handle EOF draining anyway, they can use the same code, and it removes the mess with reconfig() having to predict the output format, which completely breaks with libavfilter anyway.) In addition, there is no automatic format negotiation or conversion. libavfilter's primitive and insufficient API simply doesn't allow us to do this in a reasonable way. Instead, filters can use f_autoconvert as sub-filter, and tell it which formats they support. This filter will in turn add actual conversion filters, such as f_swscale, to perform necessary format changes. vf_vapoursynth.c uses the same basic principle of operation as before, but with worryingly different details in data flow. Still appears to work. The hardware deint filters (vf_vavpp.c, vf_d3d11vpp.c, vf_vdpaupp.c) are heavily changed. Fortunately, they all used refqueue.c, which is for sharing the data flow logic (especially for managing future/past surfaces and such). It turns out it can be used to factor out most of the data flow. Some of these filters accepted software input. Instead of having ad-hoc upload code in each filter, surface upload is now delegated to f_autoconvert, which can use f_hwupload to perform this. Exporting VO capabilities is still a big mess (mp_stream_info stuff). The D3D11 code drops the redundant image formats, and all code uses the hw_subfmt (sw_format in FFmpeg) instead. Although that too seems to be a big mess for now. f_async_queue is unused.
2018-01-16 10:53:44 +00:00
mp_pin_out_unread(vo_c->filter->f->pins[1], frame);
video: don't discard video frames after endpts Instead of letting it keep decoding by trying to find a new frame, "plug" the frame queue by not removing it. (Or actually, by putting it back instead of discarding it.) Matters for seamless looping (following commits), and possibly some other corner cases. The added function vf_unread_output_frame() is a bit of a sin, but still reasonable, since its implementation is trivial.
2016-08-18 18:37:25 +00:00
img = NULL;
video: make frame skipping code slightly more readable
2015-03-25 12:58:14 +00:00
r = VD_EOF;
player: don't apply weird timestamp tolerance on backstep Hr-seek has some sort of tolerance against timestamps, where it allows for up to 5ms deviation. This means it can work only for videos with up to 200 FPS framerate. There were complains about how it doesn't work with videos beyond some high fps. (1000 was mentioned, although that sounds more like it's about the limit that .mkv has.) I suspect this is because otherwise, it might be hard to hit a timestamp with --start, which specifies timestamps as integer, and thus will most likely never represent a timestamp exactly. Another part of the problem is that mpv uses 64 bit floats for timestamps, so fractional parts are never represented exactly. (Both the "tolerance" and using floats for timestamps were things introduced before my time.) Anyway, in the backstep case, we can be relatively sure that the timestamp will be exact (as in, the same unmodified value that was returned by the filter chain), so we can make an exception for that, in order to fix backstep. Untested. (For that you have users.) May help with #7208.
2019-12-03 20:43:32 +00:00
} else if (hrseek && (img->pts < hrseek_pts - tolerance ||
video: if hr-seek goes past last frame, seek to last frame This gives much better behavior in general, and is what we want if video somehow ends earlier than audio. A common special is using an audio file with an external image file. This commit makes things like switching aspect ratio work (provided the demuxer for the image behaves correctly, which currently isn't the case with demux_mf.c). Since the image file had timestamp 0, it was usually skipped by hr-seek, and changed properties weren't applied to it at the start of the filter chain.
2019-11-16 22:54:43 +00:00
mpctx->hrseek_lastframe))
{
/* just skip - but save in case it was the last frame */
player: when seeking past EOF with --keep-open, seek to last frame It feels strange that seeking past EOF with --keep-open actually leaves the player at a random position. You can't even unpause, because the demuxer is in the EOF state, and what you see on screen is just what was around before the seek. Improve this by attempting to seek to the last video frame if EOF happens. We explicitly don't do this if EOF was reached normally to increase robustness (if the VO got a frame since the last seek, it obviously means we had normal playback before EOF). If an error happens when trying to find the last frame (such as not actually finding a last frame because e.g. the demuxer misbehaves), this will probably turn your CPU into a heater. There is no logic to prevent reinitiating the last-frame search if the last-frame search reached EOF. (Pausing usually prevents that EOF is reached again after a successful last-frame search.) Fixes #819.
2014-12-07 01:47:09 +00:00
mp_image_setrefp(&mpctx->saved_frame, img);
video: filter new frames at a better time Move this code below the code that "shifts" the newly filtered frame. This allows us to skip a useless playloop iteration later, because obviously we need to filter a new frame after the previous frame has been "shifted", and not before that.
2014-09-18 17:17:38 +00:00
} else {
video: refactor: disentangle decoding/filtering some more This moves some code related to decoding from video.c to dec_video.c, and also removes some accesses to dec_video.c from the filtering code. dec_video.ch is starting to make sense, and simply returns video frames from a demuxer stream. The API exposed is also somewhat intended to be easily changeable to move decoding to a separate thread, if we ever want this (due to libavcodec already being threaded, I don't see much of a reason, but it might still be helpful).
2016-01-16 20:19:52 +00:00
if (hrseek && mpctx->hrseek_backstep) {
player: simplify backstepping Basically reimplement it. The old implementation was quite stupid, and was probably done this way because video filtering and output used to be way less decoupled. Now we can reimplement it in a very simple way: when backstepping, seek to current time, but keep the last frame that was supposed to be discarded when reaching the target time. When the seek finishes, prepend the saved frame to the video frame queue. A disadvantage is that the new implementation fails to skip over timeline boundaries (ordered chapters etc.), but this never worked properly anyway. It's possible that this will be fixed some time in the future.
2016-01-12 22:49:00 +00:00
if (mpctx->saved_frame) {
add_new_frame(mpctx, mpctx->saved_frame);
mpctx->saved_frame = NULL;
} else {
MP_WARN(mpctx, "Backstep failed.\n");
}
mpctx->hrseek_backstep = false;
}
video: refactor aspects of queue and EOF handling Instead of touching the 2-entry queue in mpctx->next_frame directly, move some of it to functions.
2015-03-25 12:33:34 +00:00
add_new_frame(mpctx, img);
video: make frame skipping code slightly more readable
2015-03-25 12:58:14 +00:00
img = NULL;
video: filter new frames at a better time Move this code below the code that "shifts" the newly filtered frame. This allows us to skip a useless playloop iteration later, because obviously we need to filter a new frame after the previous frame has been "shifted", and not before that.
2014-09-18 17:17:38 +00:00
}
video: make frame skipping code slightly more readable
2015-03-25 12:58:14 +00:00
talloc_free(img);
video: filter new frames at a better time Move this code below the code that "shifts" the newly filtered frame. This allows us to skip a useless playloop iteration later, because obviously we need to filter a new frame after the previous frame has been "shifted", and not before that.
2014-09-18 17:17:38 +00:00
}
}
video: if hr-seek goes past last frame, seek to last frame This gives much better behavior in general, and is what we want if video somehow ends earlier than audio. A common special is using an audio file with an external image file. This commit makes things like switching aspect ratio work (provided the demuxer for the image behaves correctly, which currently isn't the case with demux_mf.c). Since the image file had timestamp 0, it was usually skipped by hr-seek, and changed properties weren't applied to it at the start of the filter chain.
2019-11-16 22:54:43 +00:00
// If hr-seek went past EOF, use the last frame.
if (r <= 0 && hrseek && mpctx->saved_frame && r == VD_EOF) {
video: refactor aspects of queue and EOF handling Instead of touching the 2-entry queue in mpctx->next_frame directly, move some of it to functions.
2015-03-25 12:33:34 +00:00
add_new_frame(mpctx, mpctx->saved_frame);
player: when seeking past EOF with --keep-open, seek to last frame It feels strange that seeking past EOF with --keep-open actually leaves the player at a random position. You can't even unpause, because the demuxer is in the EOF state, and what you see on screen is just what was around before the seek. Improve this by attempting to seek to the last video frame if EOF happens. We explicitly don't do this if EOF was reached normally to increase robustness (if the VO got a frame since the last seek, it obviously means we had normal playback before EOF). If an error happens when trying to find the last frame (such as not actually finding a last frame because e.g. the demuxer misbehaves), this will probably turn your CPU into a heater. There is no logic to prevent reinitiating the last-frame search if the last-frame search reached EOF. (Pausing usually prevents that EOF is reached again after a successful last-frame search.) Fixes #819.
2014-12-07 01:47:09 +00:00
mpctx->saved_frame = NULL;
player: make repeated hr-seeks past EOF trigger EOF as expected If you have a normal file with audio and video, and keep "spamming" forward hr-seeks, the player just kept showing the last video frame instead of exiting or playing the next file. This started happening since commit 6bcda94cb. Although not a bug per se, it was odd, and very user-noticable. The main problem was that the pending seek command was processed before the EOF was "noticed". Processing the command reset everything, so the player did not terminate playback, but repeated the seek. This commit restores the old behavior. For one, it makes video return the correct status (video.c). The parameter is a bit ugly, but better than duplicating the logic or having another MPContext field. (As a minor detail, setting r=VD_EOF makes sure have_new_frame() returns true, rather than going through another iteration or whatever the hell will happen instead, which would clobber logical_eof.) Another thing is making the seek logic actually wait until the seek outcome has been determined if audio is also active. Audio needs to wait for video in order to get the video seek target position. (Which in turn is because hr-seek still "snaps" to video frames. You can't seek in between two frames, so audio can't just use the seek target, but always has to wait on the timestamp of the video frame. This has other disadvantages and is a misdesign, but not something I'll fix today.) In theory, this might make hr-seeks less responsive, because it needs to fully decode/filter the audio too, but in practice most time is spent on video, which had to be fully decoded before this change. (In general, hr-seek could probably just show a random frame when a queued hr-seek overrides the current hr-seek, which would probably lead to a better user experience, but that's out of scope.) Fixes: #7206
2019-12-14 13:17:16 +00:00
r = VD_EOF;
*logical_eof = true;
player: when seeking past EOF with --keep-open, seek to last frame It feels strange that seeking past EOF with --keep-open actually leaves the player at a random position. You can't even unpause, because the demuxer is in the EOF state, and what you see on screen is just what was around before the seek. Improve this by attempting to seek to the last video frame if EOF happens. We explicitly don't do this if EOF was reached normally to increase robustness (if the VO got a frame since the last seek, it obviously means we had normal playback before EOF). If an error happens when trying to find the last frame (such as not actually finding a last frame because e.g. the demuxer misbehaves), this will probably turn your CPU into a heater. There is no logic to prevent reinitiating the last-frame search if the last-frame search reached EOF. (Pausing usually prevents that EOF is reached again after a successful last-frame search.) Fixes #819.
2014-12-07 01:47:09 +00:00
}
video: refactor aspects of queue and EOF handling Instead of touching the 2-entry queue in mpctx->next_frame directly, move some of it to functions.
2015-03-25 12:33:34 +00:00
return have_new_frame(mpctx, r <= 0) ? VD_NEW_FRAME : r;
Split mplayer.c mplayer.c was a bit too big. Split it into multiple files. I hope the way it's split makes sense. Maybe some things don't make too much sense, or go against intuition. These will fixed as soon as I notice them. Some files are a bit questionable (misc.c, osd.c, configfiles.c), and suggestions how to organize this better are welcome. Regressions are possible due to reorganized include statements. Obviously I didn't just copy mplayer.c's orgy of include statements, but recreated them for each file. It's easily possible that there are oversights and mistakes, which will show up on other platforms. There is one actual change: the public avutil.h include is removed from encode.h, and I tried to replace most FFMIN/FFMAX/av_clip uses. I consider using libavutil too much as dangerous, because the set of include files they recursively pull in is rather arbitrary and is different between FFmpeg and Libav.
2013-10-29 21:38:29 +00:00
}
player: move video display code out of the playloop Basically move the code from playloop.c to video.c. The new function write_video() now contains the code that was part of run_playloop(). There are no functional changes, except handling "new_frame_shown" slightly differently. This is done so that we don't need new a new MPContext field or a return value for write_video() to signal this condition. Instead, it's handled indirectly.
2014-07-30 21:24:08 +00:00
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
static bool check_for_hwdec_fallback(struct MPContext *mpctx)
{
struct vo_chain *vo_c = mpctx->vo_chain;
video: make decoder wrapper a filter Move dec_video.c to filters/f_decoder_wrapper.c. It essentially becomes a source filter. vd.h mostly disappears, because mp_filter takes care of the dataflow, but its remains are in struct mp_decoder_fns. One goal is to simplify dataflow by letting the filter framework handle it (or more accurately, using its conventions). One result is that the decode calls disappear from video.c, because we simply connect the decoder wrapper and the filter chain with mp_pin_connect(). Another goal is to eventually remove the code duplication between the audio and video paths for this. This commit prepares for this by trying to make f_decoder_wrapper.c extensible, so it can be used for audio as well later. Decoder framedropping changes a bit. It doesn't seem to be worse than before, and it's an obscure feature, so I'm content with its new state. Some special code that was apparently meant to avoid dropping too many frames in a row is removed, though. I'm not sure how the source code tree should be organized. For one, video/decode/vd_lavc.c is the only file in its directory, which is a bit annoying.
2018-01-28 09:08:45 +00:00
if (!vo_c->filter->failed_output_conversion || !vo_c->track)
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
return false;
video: make decoder wrapper a filter Move dec_video.c to filters/f_decoder_wrapper.c. It essentially becomes a source filter. vd.h mostly disappears, because mp_filter takes care of the dataflow, but its remains are in struct mp_decoder_fns. One goal is to simplify dataflow by letting the filter framework handle it (or more accurately, using its conventions). One result is that the decode calls disappear from video.c, because we simply connect the decoder wrapper and the filter chain with mp_pin_connect(). Another goal is to eventually remove the code duplication between the audio and video paths for this. This commit prepares for this by trying to make f_decoder_wrapper.c extensible, so it can be used for audio as well later. Decoder framedropping changes a bit. It doesn't seem to be worse than before, and it's an obscure feature, so I'm content with its new state. Some special code that was apparently meant to avoid dropping too many frames in a row is removed, though. I'm not sure how the source code tree should be organized. For one, video/decode/vd_lavc.c is the only file in its directory, which is a bit annoying.
2018-01-28 09:08:45 +00:00
if (mp_decoder_wrapper_control(vo_c->track->dec,
VDCTRL_FORCE_HWDEC_FALLBACK, NULL) != CONTROL_OK)
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
return false;
mp_output_chain_reset_harder(vo_c->filter);
return true;
}
video: move some more code around No functional changes.
2014-08-13 19:54:13 +00:00
/* Update avsync before a new video frame is displayed. Actually, this can be
* called arbitrarily often before the actual display.
* This adjusts the time of the next video frame */
static void update_avsync_before_frame(struct MPContext *mpctx)
{
struct MPOpts *opts = mpctx->opts;
struct vo *vo = mpctx->video_out;
player: remove some unnecessary coverart special cases These should not be needed, since video is in EOF mode in this case anyway. Not too sure about the video.c case to be honest, well, here goes nothing.
2019-11-17 00:39:54 +00:00
if (mpctx->video_status < STATUS_READY) {
video: move some more code around No functional changes.
2014-08-13 19:54:13 +00:00
mpctx->time_frame = 0;
player: add display sync mode If this mode is enabled, the player tries to strictly synchronize video to display refresh. It will adjust playback speed to match the display, so if you play 23.976 fps video on a 24 Hz screen, playback speed is increased by approximately 1/1000. Audio wll be resampled to keep up with playback. This is different from the default sync mode, which will sync video to audio, with the consequence that video might skip or repeat a frame once in a while to make video keep up with audio. This is still unpolished. There are some major problems as well; in particular, mkv VFR files won't work well. The reason is that Matroska is terrible and rounds timestamps to milliseconds. This makes it rather hard to guess the framerate of a section of video that is playing. We could probably fix this by just accepting jittery timestamps (instead of explicitly disabling the sync code in this case), but I'm not ready to accept such a solution yet. Another issue is that we are extremely reliant on OS video and audio APIs working in an expected manner, which of course is not too often the case. Consequently, the new sync mode is a bit fragile.
2015-08-10 16:43:25 +00:00
} else if (mpctx->display_sync_active || opts->video_sync == VS_NONE) {
// don't touch the timing
video: move some more code around No functional changes.
2014-08-13 19:54:13 +00:00
} else if (mpctx->audio_status == STATUS_PLAYING &&
mpctx->video_status == STATUS_PLAYING &&
!ao_untimed(mpctx->ao))
{
double buffered_audio = ao_get_delay(mpctx->ao);
player: dump audio jitter to stats file This allows us to plot the difference between video timestamps, and the adjusted video timestamps due to syncing video to audio speed.
2015-02-01 17:32:26 +00:00
player: separate controls for user and video controlled speed For video sync, we want separate playback speed controls for user- requested speed and the "correction" speed for video timing. Further, we use this separation to make sure only a resampler is inserted if playback speed is only changed for video sync correction. As of this commit, this is basically inactive code. It's just preparation for the video sync code (the following commit).
2015-08-10 16:40:16 +00:00
double predicted = mpctx->delay / mpctx->video_speed +
player: dump audio jitter to stats file This allows us to plot the difference between video timestamps, and the adjusted video timestamps due to syncing video to audio speed.
2015-02-01 17:32:26 +00:00
mpctx->time_frame;
double difference = buffered_audio - predicted;
MP_STATS(mpctx, "value %f audio-diff", difference);
video: move some more code around No functional changes.
2014-08-13 19:54:13 +00:00
if (opts->autosync) {
/* Smooth reported playback position from AO by averaging
* it with the value expected based on previus value and
* time elapsed since then. May help smooth video timing
* with audio output that have inaccurate position reporting.
* This is badly implemented; the behavior of the smoothing
* now undesirably depends on how often this code runs
* (mainly depends on video frame rate). */
buffered_audio = predicted + difference / opts->autosync;
}
player: separate controls for user and video controlled speed For video sync, we want separate playback speed controls for user- requested speed and the "correction" speed for video timing. Further, we use this separation to make sure only a resampler is inserted if playback speed is only changed for video sync correction. As of this commit, this is basically inactive code. It's just preparation for the video sync code (the following commit).
2015-08-10 16:40:16 +00:00
mpctx->time_frame = buffered_audio - mpctx->delay / mpctx->video_speed;
video: move some more code around No functional changes.
2014-08-13 19:54:13 +00:00
} else {
/* If we're more than 200 ms behind the right playback
* position, don't try to speed up display of following
* frames to catch up; continue with default speed from
* the current frame instead.
* If untimed is set always output frames immediately
* without sleeping.
*/
if (mpctx->time_frame < -0.2 || opts->untimed || vo->driver->untimed)
mpctx->time_frame = 0;
}
}
player: minor refactor for A/V diff computation Calculate the A/V difference directly in the display sync code, instead of the awkward current way, which reuses the fields for audio sync. We still set time_frame, because it makes falling back to audio sync somewhat smoother.
2015-10-28 22:50:54 +00:00
// Update the A/V sync difference when a new video frame is being shown.
static void update_av_diff(struct MPContext *mpctx, double offset)
player: move video display code out of the playloop Basically move the code from playloop.c to video.c. The new function write_video() now contains the code that was part of run_playloop(). There are no functional changes, except handling "new_frame_shown" slightly differently. This is done so that we don't need new a new MPContext field or a return value for write_video() to signal this condition. Instead, it's handled indirectly.
2014-07-30 21:24:08 +00:00
{
player: remove redundant variable mpctx->audio_delay always has the same value as opts->audio_delay. (This was not the case a long time ago, when the audio-delay property didn't actually write to opts->audio_delay. I think.)
2015-01-29 11:10:07 +00:00
struct MPOpts *opts = mpctx->opts;
player: reset last_av_difference if not applicable Don't let stale values linger around. Also fix a slightly related case in audio.c.
2014-09-19 20:38:00 +00:00
mpctx->last_av_difference = 0;
video: refactor queue handling This simplifies the code, and fixes an odd bug: the second-last frame was displayed for a very short duration if framedrop was enabled. The reason was that basically the time difference between second-last and last frame were skipped, because at this point EOF was already signaled. Also see commit b0959488 for a similar issue in the same code. This removes the messiness of the next_frame 2-frame queue, and strictly runs the "new frame" code when a frame is moved to the first position of the queue, instead of somehow messing with return codes. This also merges update_video() into video_output_image().
2014-08-22 12:20:30 +00:00
player: move video display code out of the playloop Basically move the code from playloop.c to video.c. The new function write_video() now contains the code that was part of run_playloop(). There are no functional changes, except handling "new_frame_shown" slightly differently. This is done so that we don't need new a new MPContext field or a return value for write_video() to signal this condition. Instead, it's handled indirectly.
2014-07-30 21:24:08 +00:00
if (mpctx->audio_status != STATUS_PLAYING ||
mpctx->video_status != STATUS_PLAYING)
return;
demux, player: fix playback of sparse video streams (w/ still images) Fixes several issues playing back mpegts with video streams marked as having "still images". For example, see this video which has frames only every 6s: https://s3.amazonaws.com/tmm1/music-choice.ts Changes include: - start playback right away, without waiting for first video frame - do not consider the sparse video stream in demuxer underrun detection - do not require multiple video frames for the VO - use audio as the master stream for demuxer metadata events - use audio stream for playback time Signed-off-by: Aman Gupta <aman@tmm1.net>
2018-05-03 02:29:11 +00:00
if (mpctx->vo_chain && mpctx->vo_chain->is_sparse)
return;
player: move video display code out of the playloop Basically move the code from playloop.c to video.c. The new function write_video() now contains the code that was part of run_playloop(). There are no functional changes, except handling "new_frame_shown" slightly differently. This is done so that we don't need new a new MPContext field or a return value for write_video() to signal this condition. Instead, it's handled indirectly.
2014-07-30 21:24:08 +00:00
double a_pos = playing_audio_pts(mpctx);
player: minor refactor for A/V diff computation Calculate the A/V difference directly in the display sync code, instead of the awkward current way, which reuses the fields for audio sync. We still set time_frame, because it makes falling back to audio sync somewhat smoother.
2015-10-28 22:50:54 +00:00
if (a_pos != MP_NOPTS_VALUE && mpctx->video_pts != MP_NOPTS_VALUE) {
mpctx->last_av_difference = a_pos - mpctx->video_pts
+ opts->audio_delay + offset;
}
player: move video display code out of the playloop Basically move the code from playloop.c to video.c. The new function write_video() now contains the code that was part of run_playloop(). There are no functional changes, except handling "new_frame_shown" slightly differently. This is done so that we don't need new a new MPContext field or a return value for write_video() to signal this condition. Instead, it's handled indirectly.
2014-07-30 21:24:08 +00:00
player: minor refactor for A/V diff computation Calculate the A/V difference directly in the display sync code, instead of the awkward current way, which reuses the fields for audio sync. We still set time_frame, because it makes falling back to audio sync somewhat smoother.
2015-10-28 22:50:54 +00:00
if (fabs(mpctx->last_av_difference) > 0.5 && !mpctx->drop_message_shown) {
player: move video display code out of the playloop Basically move the code from playloop.c to video.c. The new function write_video() now contains the code that was part of run_playloop(). There are no functional changes, except handling "new_frame_shown" slightly differently. This is done so that we don't need new a new MPContext field or a return value for write_video() to signal this condition. Instead, it's handled indirectly.
2014-07-30 21:24:08 +00:00
MP_WARN(mpctx, "%s", av_desync_help_text);
mpctx->drop_message_shown = true;
}
}
player: refactor display-sync frame duration calculations Get rid of get_past_frame_durations(), which was a bit too messy. Add a past_frames array, which contains the same information in a more reasonable way. This also means that we can get the exact current and past frame durations without going through awful stuff. (The main problem is that vo_pts_history contains future frames as well, which is needed for frame backstepping etc., but gets in the way here.) Also disable the automatic disabling of display-sync if the frame duration changes, and extend the frame durations allowed for display sync. To allow arbitrarily high durations, vo.c needs to be changed to pause and potentially redraw OSD while showing a single frame, so they're still limited. In an attempt to deal with VFR, calculate the overall speed using the average FPS. The frame scheduling itself does not use the average FPS, but the duration of the current frame. This does not work too well, but provides a good base for further improvements. Where this commit actually helps a lot is dealing with rounded timestamps, e.g. if the container framerate is wrong or unknown, or if the muxer wrote incorrectly rounded timestamps. While the rounding errors apparently can't be get rid of completely in the general case, this is still much better than e.g. disabling display-sync completely just because some frame durations go out of bounds.
2015-11-13 21:45:40 +00:00
double calc_average_frame_duration(struct MPContext *mpctx)
player: redo estimated-vf-fps calculation Additionally to taking the average, this tries to use the demuxer FPS to eliminate jitter, and applies some other heuristics to check if the result is sane. This code will also be used for the display sync code (it will actually make use of the require_exact parameter). (The value of doing this over keeping the simpler demux_mkv hack is somewhat questionable. But at least it allows us to deal with other container formats that use jittery timestamps, such as mp4 remuxed from mkv.)
2015-08-10 16:38:57 +00:00
{
player: refactor display-sync frame duration calculations Get rid of get_past_frame_durations(), which was a bit too messy. Add a past_frames array, which contains the same information in a more reasonable way. This also means that we can get the exact current and past frame durations without going through awful stuff. (The main problem is that vo_pts_history contains future frames as well, which is needed for frame backstepping etc., but gets in the way here.) Also disable the automatic disabling of display-sync if the frame duration changes, and extend the frame durations allowed for display sync. To allow arbitrarily high durations, vo.c needs to be changed to pause and potentially redraw OSD while showing a single frame, so they're still limited. In an attempt to deal with VFR, calculate the overall speed using the average FPS. The frame scheduling itself does not use the average FPS, but the duration of the current frame. This does not work too well, but provides a good base for further improvements. Where this commit actually helps a lot is dealing with rounded timestamps, e.g. if the container framerate is wrong or unknown, or if the muxer wrote incorrectly rounded timestamps. While the rounding errors apparently can't be get rid of completely in the general case, this is still much better than e.g. disabling display-sync completely just because some frame durations go out of bounds.
2015-11-13 21:45:40 +00:00
double total = 0;
int num = 0;
for (int n = 0; n < mpctx->num_past_frames; n++) {
player: fix average frame duration calculation
2016-09-09 16:56:45 +00:00
double dur = mpctx->past_frames[n].approx_duration;
player: refactor display-sync frame duration calculations Get rid of get_past_frame_durations(), which was a bit too messy. Add a past_frames array, which contains the same information in a more reasonable way. This also means that we can get the exact current and past frame durations without going through awful stuff. (The main problem is that vo_pts_history contains future frames as well, which is needed for frame backstepping etc., but gets in the way here.) Also disable the automatic disabling of display-sync if the frame duration changes, and extend the frame durations allowed for display sync. To allow arbitrarily high durations, vo.c needs to be changed to pause and potentially redraw OSD while showing a single frame, so they're still limited. In an attempt to deal with VFR, calculate the overall speed using the average FPS. The frame scheduling itself does not use the average FPS, but the duration of the current frame. This does not work too well, but provides a good base for further improvements. Where this commit actually helps a lot is dealing with rounded timestamps, e.g. if the container framerate is wrong or unknown, or if the muxer wrote incorrectly rounded timestamps. While the rounding errors apparently can't be get rid of completely in the general case, this is still much better than e.g. disabling display-sync completely just because some frame durations go out of bounds.
2015-11-13 21:45:40 +00:00
if (dur <= 0)
continue;
total += dur;
num += 1;
player: redo estimated-vf-fps calculation Additionally to taking the average, this tries to use the demuxer FPS to eliminate jitter, and applies some other heuristics to check if the result is sane. This code will also be used for the display sync code (it will actually make use of the require_exact parameter). (The value of doing this over keeping the simpler demux_mkv hack is somewhat questionable. But at least it allows us to deal with other container formats that use jittery timestamps, such as mp4 remuxed from mkv.)
2015-08-10 16:38:57 +00:00
}
player: refactor display-sync frame duration calculations Get rid of get_past_frame_durations(), which was a bit too messy. Add a past_frames array, which contains the same information in a more reasonable way. This also means that we can get the exact current and past frame durations without going through awful stuff. (The main problem is that vo_pts_history contains future frames as well, which is needed for frame backstepping etc., but gets in the way here.) Also disable the automatic disabling of display-sync if the frame duration changes, and extend the frame durations allowed for display sync. To allow arbitrarily high durations, vo.c needs to be changed to pause and potentially redraw OSD while showing a single frame, so they're still limited. In an attempt to deal with VFR, calculate the overall speed using the average FPS. The frame scheduling itself does not use the average FPS, but the duration of the current frame. This does not work too well, but provides a good base for further improvements. Where this commit actually helps a lot is dealing with rounded timestamps, e.g. if the container framerate is wrong or unknown, or if the muxer wrote incorrectly rounded timestamps. While the rounding errors apparently can't be get rid of completely in the general case, this is still much better than e.g. disabling display-sync completely just because some frame durations go out of bounds.
2015-11-13 21:45:40 +00:00
return num > 0 ? total / num : 0;
player: redo estimated-vf-fps calculation Additionally to taking the average, this tries to use the demuxer FPS to eliminate jitter, and applies some other heuristics to check if the result is sane. This code will also be used for the display sync code (it will actually make use of the require_exact parameter). (The value of doing this over keeping the simpler demux_mkv hack is somewhat questionable. But at least it allows us to deal with other container formats that use jittery timestamps, such as mp4 remuxed from mkv.)
2015-08-10 16:38:57 +00:00
}
player: compute required display-sync speed change differently Instead of looking at the current frame duration for the intended speedup, look at all past frames, and find a good average speed. This ties in with not wanting to average _all_ frame durations, which doesn't make sense in VFR situations. This is currently done in the most naive way possible, but already sort of works for VFR which switches between frame durations that are integer multiples of a base rate. Certainly more improvements could be made, such as trying to adjust directly on FPS changes, instead of averaging everything, but for now this is not needed at all.
2015-11-13 21:47:14 +00:00
// Find a speed factor such that the display FPS is an integer multiple of the
// effective video FPS. If this is not possible, try to do it for multiples,
// which still leads to an improved end result.
// Both parameters are durations in seconds.
static double calc_best_speed(double vsync, double frame)
{
double ratio = frame / vsync;
double best_scale = -1;
double best_dev = INFINITY;
for (int factor = 1; factor <= 5; factor++) {
double scale = ratio * factor / rint(ratio * factor);
double dev = fabs(scale - 1);
if (dev < best_dev) {
best_scale = scale;
best_dev = dev;
}
}
return best_scale;
}
static double find_best_speed(struct MPContext *mpctx, double vsync)
{
double total = 0;
int num = 0;
for (int n = 0; n < mpctx->num_past_frames; n++) {
double dur = mpctx->past_frames[n].approx_duration;
if (dur <= 0)
continue;
total += calc_best_speed(vsync, dur / mpctx->opts->playback_speed);
num++;
}
return num > 0 ? total / num : 1;
}
player: add display sync mode If this mode is enabled, the player tries to strictly synchronize video to display refresh. It will adjust playback speed to match the display, so if you play 23.976 fps video on a 24 Hz screen, playback speed is increased by approximately 1/1000. Audio wll be resampled to keep up with playback. This is different from the default sync mode, which will sync video to audio, with the consequence that video might skip or repeat a frame once in a while to make video keep up with audio. This is still unpolished. There are some major problems as well; in particular, mkv VFR files won't work well. The reason is that Matroska is terrible and rounds timestamps to milliseconds. This makes it rather hard to guess the framerate of a section of video that is playing. We could probably fix this by just accepting jittery timestamps (instead of explicitly disabling the sync code in this case), but I'm not ready to accept such a solution yet. Another issue is that we are extremely reliant on OS video and audio APIs working in an expected manner, which of course is not too often the case. Consequently, the new sync mode is a bit fragile.
2015-08-10 16:43:25 +00:00
static bool using_spdif_passthrough(struct MPContext *mpctx)
{
player: disable DS with spdif transcoding too Otherwise it behaves dumb. (Although you could argue it shouldn't try to guess whether speed changes work, but instead simply disable DS if they don't work.)
2016-07-24 17:19:24 +00:00
if (mpctx->ao_chain && mpctx->ao_chain->ao) {
audio: introduce a new type to hold audio frames This is pretty pointless, but I believe it allows us to claim that the new code is not affected by the copyright of the old code. This is needed, because the original mp_audio struct was written by someone who has disagreed with LGPL relicensing (it was called af_data at the time, and was defined in af.h). The "GPL'ed" struct contents that surive are pretty trivial: just the data pointer, and some metadata like the format, samplerate, etc. - but at least in this case, any new code would be extremely similar anyway, and I'm not really sure whether it's OK to claim different copyright. So what we do is we just use AVFrame (which of course is LGPL with 100% certainty), and add some accessors around it to adapt it to mpv conventions. Also, this gets rid of some annoying conventions of mp_audio, like the struct fields that require using an accessor to write to them anyway. For the most part, this change is only dumb replacements of mp_audio related functions and fields. One minor actual change is that you can't allocate the new type on the stack anymore. Some code still uses mp_audio. All audio filter code will be deleted, so it makes no sense to convert this code. (Audio filters which are LGPL and which we keep will have to be ported to a new filter infrastructure anyway.) player/audio.c uses it because it interacts with the old filter code. push.c has some complex use of mp_audio and mp_audio_buffer, but this and pull.c will most likely be rewritten to do something else.
2017-08-16 19:00:20 +00:00
int samplerate;
int format;
struct mp_chmap channels;
ao_get_format(mpctx->ao_chain->ao, &samplerate, &format, &channels);
return !af_fmt_is_pcm(format);
player: disable DS with spdif transcoding too Otherwise it behaves dumb. (Although you could argue it shouldn't try to guess whether speed changes work, but instead simply disable DS if they don't work.)
2016-07-24 17:19:24 +00:00
}
player: add display sync mode If this mode is enabled, the player tries to strictly synchronize video to display refresh. It will adjust playback speed to match the display, so if you play 23.976 fps video on a 24 Hz screen, playback speed is increased by approximately 1/1000. Audio wll be resampled to keep up with playback. This is different from the default sync mode, which will sync video to audio, with the consequence that video might skip or repeat a frame once in a while to make video keep up with audio. This is still unpolished. There are some major problems as well; in particular, mkv VFR files won't work well. The reason is that Matroska is terrible and rounds timestamps to milliseconds. This makes it rather hard to guess the framerate of a section of video that is playing. We could probably fix this by just accepting jittery timestamps (instead of explicitly disabling the sync code in this case), but I'm not ready to accept such a solution yet. Another issue is that we are extremely reliant on OS video and audio APIs working in an expected manner, which of course is not too often the case. Consequently, the new sync mode is a bit fragile.
2015-08-10 16:43:25 +00:00
return false;
}
player: fix audio drift computation at different playback speeds This computed nonsense if the user set a playback speed other than 1 (in addition to the display-sync speed change).
2015-11-14 20:42:18 +00:00
// Compute the relative audio speed difference by taking A/V dsync into account.
static double compute_audio_drift(struct MPContext *mpctx, double vsync)
player: try to compensate actual audio drift We always let audio slowly desync until a threshold is reached, and then pushed it back by applying a maximum compensation speed. Refine what comes afterwards: instead of playing with the nominal video speed, use the actual required audio speed for keeping sync as measured by the A/V difference. (The "actual" speed is the ideal speed with A/V differences added.) Although this works in theory, it's somewhat questionable how much this works in practice. The ideal time value is actually not exact, but is the time at which the frame is scheduled (could be compensated by using the time_left calculations in handle_display_sync_frame()). It doesn't account for speed changes or catastrophic discontinuities. It uses only 10 past frames.
2015-11-13 21:51:39 +00:00
{
player: fix audio drift computation at different playback speeds This computed nonsense if the user set a playback speed other than 1 (in addition to the display-sync speed change).
2015-11-14 20:42:18 +00:00
// Least-squares linear regression, using relative real time for x, and
// audio desync for y. Assume speed didn't change for the frames we're
player: try to compensate actual audio drift We always let audio slowly desync until a threshold is reached, and then pushed it back by applying a maximum compensation speed. Refine what comes afterwards: instead of playing with the nominal video speed, use the actual required audio speed for keeping sync as measured by the A/V difference. (The "actual" speed is the ideal speed with A/V differences added.) Although this works in theory, it's somewhat questionable how much this works in practice. The ideal time value is actually not exact, but is the time at which the frame is scheduled (could be compensated by using the time_left calculations in handle_display_sync_frame()). It doesn't account for speed changes or catastrophic discontinuities. It uses only 10 past frames.
2015-11-13 21:51:39 +00:00
// looking at for simplicity. This also should actually use the realtime
// (minus paused time) for x, but use vsync scheduling points instead.
if (mpctx->num_past_frames <= 10)
return NAN;
int num = mpctx->num_past_frames - 1;
double sum_x = 0, sum_y = 0, sum_xy = 0, sum_xx = 0;
double x = 0;
for (int n = 0; n < num; n++) {
struct frame_info *frame = &mpctx->past_frames[n + 1];
if (frame->num_vsyncs < 0)
return NAN;
player: fix audio drift computation at different playback speeds This computed nonsense if the user set a playback speed other than 1 (in addition to the display-sync speed change).
2015-11-14 20:42:18 +00:00
double y = frame->av_diff;
player: try to compensate actual audio drift We always let audio slowly desync until a threshold is reached, and then pushed it back by applying a maximum compensation speed. Refine what comes afterwards: instead of playing with the nominal video speed, use the actual required audio speed for keeping sync as measured by the A/V difference. (The "actual" speed is the ideal speed with A/V differences added.) Although this works in theory, it's somewhat questionable how much this works in practice. The ideal time value is actually not exact, but is the time at which the frame is scheduled (could be compensated by using the time_left calculations in handle_display_sync_frame()). It doesn't account for speed changes or catastrophic discontinuities. It uses only 10 past frames.
2015-11-13 21:51:39 +00:00
sum_x += x;
sum_y += y;
sum_xy += x * y;
sum_xx += x * x;
x -= frame->num_vsyncs * vsync;
}
return (sum_x * sum_y - num * sum_xy) / (sum_x * sum_x - num * sum_xx);
}
player: change display-sync audio speed only if needed As long as it's within the desync tolerance, do not change the audio speed at all for resampling. This reduces speed changes which might be caused by jittering timestamps and similar cases. (While in theory you could just not care and change speed every single frame, I'm afraid that such changes could possibly cause audio artifacts. So better just avoid it in the first place.)
2015-11-13 21:50:58 +00:00
static void adjust_audio_resample_speed(struct MPContext *mpctx, double vsync)
player: move audio speed adjustment code Move it (in a cosmetic sense), and also move its invocation to below all the video handling. All other changes remain cosmetic, including moving the framedrop calculation code, and getting rid of the video_speed_correction variable.
2015-11-04 18:37:15 +00:00
{
struct MPOpts *opts = mpctx->opts;
int mode = opts->video_sync;
player: change display-sync audio speed only if needed As long as it's within the desync tolerance, do not change the audio speed at all for resampling. This reduces speed changes which might be caused by jittering timestamps and similar cases. (While in theory you could just not care and change speed every single frame, I'm afraid that such changes could possibly cause audio artifacts. So better just avoid it in the first place.)
2015-11-13 21:50:58 +00:00
if (mode != VS_DISP_RESAMPLE || mpctx->audio_status != STATUS_PLAYING) {
mpctx->speed_factor_a = mpctx->speed_factor_v;
return;
player: move audio speed adjustment code Move it (in a cosmetic sense), and also move its invocation to below all the video handling. All other changes remain cosmetic, including moving the framedrop calculation code, and getting rid of the video_speed_correction variable.
2015-11-04 18:37:15 +00:00
}
player: change display-sync audio speed only if needed As long as it's within the desync tolerance, do not change the audio speed at all for resampling. This reduces speed changes which might be caused by jittering timestamps and similar cases. (While in theory you could just not care and change speed every single frame, I'm afraid that such changes could possibly cause audio artifacts. So better just avoid it in the first place.)
2015-11-13 21:50:58 +00:00
// Try to smooth out audio timing drifts. This can happen if either
// video isn't playing at expected speed, or audio is not playing at
// the requested speed. Both are unavoidable.
// The audio desync is made up of 2 parts: 1. drift due to rounding
// errors and imperfect information, and 2. an offset, due to
// unaligned audio/video start, or disruptive events halting audio
// or video for a small time.
// Instead of trying to be clever, just apply an awfully dumb drift
// compensation with a constant factor, which does what we want. In
// theory we could calculate the exact drift compensation needed,
// but it likely would be wrong anyway, and we'd run into the same
// issues again, except with more complex code.
// 1 means drifts to positive, -1 means drifts to negative
double max_drift = vsync / 2;
double av_diff = mpctx->last_av_difference;
int new = mpctx->display_sync_drift_dir;
if (av_diff * -mpctx->display_sync_drift_dir >= 0)
new = 0;
if (fabs(av_diff) > max_drift)
new = av_diff >= 0 ? 1 : -1;
bool change = mpctx->display_sync_drift_dir != new;
if (new || change) {
if (change)
MP_VERBOSE(mpctx, "Change display sync audio drift: %d\n", new);
mpctx->display_sync_drift_dir = new;
player: move audio speed adjustment code Move it (in a cosmetic sense), and also move its invocation to below all the video handling. All other changes remain cosmetic, including moving the framedrop calculation code, and getting rid of the video_speed_correction variable.
2015-11-04 18:37:15 +00:00
player: change display-sync audio speed only if needed As long as it's within the desync tolerance, do not change the audio speed at all for resampling. This reduces speed changes which might be caused by jittering timestamps and similar cases. (While in theory you could just not care and change speed every single frame, I'm afraid that such changes could possibly cause audio artifacts. So better just avoid it in the first place.)
2015-11-13 21:50:58 +00:00
double max_correct = opts->sync_max_audio_change / 100;
double audio_factor = 1 + max_correct * -mpctx->display_sync_drift_dir;
player: try to compensate actual audio drift We always let audio slowly desync until a threshold is reached, and then pushed it back by applying a maximum compensation speed. Refine what comes afterwards: instead of playing with the nominal video speed, use the actual required audio speed for keeping sync as measured by the A/V difference. (The "actual" speed is the ideal speed with A/V differences added.) Although this works in theory, it's somewhat questionable how much this works in practice. The ideal time value is actually not exact, but is the time at which the frame is scheduled (could be compensated by using the time_left calculations in handle_display_sync_frame()). It doesn't account for speed changes or catastrophic discontinuities. It uses only 10 past frames.
2015-11-13 21:51:39 +00:00
if (new == 0) {
// If we're resetting, actually try to be clever and pick a speed
// which compensates the general drift we're getting.
player: fix audio drift computation at different playback speeds This computed nonsense if the user set a playback speed other than 1 (in addition to the display-sync speed change).
2015-11-14 20:42:18 +00:00
double drift = compute_audio_drift(mpctx, vsync);
player: try to compensate actual audio drift We always let audio slowly desync until a threshold is reached, and then pushed it back by applying a maximum compensation speed. Refine what comes afterwards: instead of playing with the nominal video speed, use the actual required audio speed for keeping sync as measured by the A/V difference. (The "actual" speed is the ideal speed with A/V differences added.) Although this works in theory, it's somewhat questionable how much this works in practice. The ideal time value is actually not exact, but is the time at which the frame is scheduled (could be compensated by using the time_left calculations in handle_display_sync_frame()). It doesn't account for speed changes or catastrophic discontinuities. It uses only 10 past frames.
2015-11-13 21:51:39 +00:00
if (isnormal(drift)) {
player: fix audio drift computation at different playback speeds This computed nonsense if the user set a playback speed other than 1 (in addition to the display-sync speed change).
2015-11-14 20:42:18 +00:00
// other = will be multiplied with audio_factor for final speed
double other = mpctx->opts->playback_speed * mpctx->speed_factor_v;
audio_factor = (mpctx->audio_speed - drift) / other;
player: try to compensate actual audio drift We always let audio slowly desync until a threshold is reached, and then pushed it back by applying a maximum compensation speed. Refine what comes afterwards: instead of playing with the nominal video speed, use the actual required audio speed for keeping sync as measured by the A/V difference. (The "actual" speed is the ideal speed with A/V differences added.) Although this works in theory, it's somewhat questionable how much this works in practice. The ideal time value is actually not exact, but is the time at which the frame is scheduled (could be compensated by using the time_left calculations in handle_display_sync_frame()). It doesn't account for speed changes or catastrophic discontinuities. It uses only 10 past frames.
2015-11-13 21:51:39 +00:00
MP_VERBOSE(mpctx, "Compensation factor: %f\n", audio_factor);
}
}
audio_factor = MPCLAMP(audio_factor, 1 - max_correct, 1 + max_correct);
player: change display-sync audio speed only if needed As long as it's within the desync tolerance, do not change the audio speed at all for resampling. This reduces speed changes which might be caused by jittering timestamps and similar cases. (While in theory you could just not care and change speed every single frame, I'm afraid that such changes could possibly cause audio artifacts. So better just avoid it in the first place.)
2015-11-13 21:50:58 +00:00
mpctx->speed_factor_a = audio_factor * mpctx->speed_factor_v;
}
player: move audio speed adjustment code Move it (in a cosmetic sense), and also move its invocation to below all the video handling. All other changes remain cosmetic, including moving the framedrop calculation code, and getting rid of the video_speed_correction variable.
2015-11-04 18:37:15 +00:00
}
player: add display sync mode If this mode is enabled, the player tries to strictly synchronize video to display refresh. It will adjust playback speed to match the display, so if you play 23.976 fps video on a 24 Hz screen, playback speed is increased by approximately 1/1000. Audio wll be resampled to keep up with playback. This is different from the default sync mode, which will sync video to audio, with the consequence that video might skip or repeat a frame once in a while to make video keep up with audio. This is still unpolished. There are some major problems as well; in particular, mkv VFR files won't work well. The reason is that Matroska is terrible and rounds timestamps to milliseconds. This makes it rather hard to guess the framerate of a section of video that is playing. We could probably fix this by just accepting jittery timestamps (instead of explicitly disabling the sync code in this case), but I'm not ready to accept such a solution yet. Another issue is that we are extremely reliant on OS video and audio APIs working in an expected manner, which of course is not too often the case. Consequently, the new sync mode is a bit fragile.
2015-08-10 16:43:25 +00:00
// Manipulate frame timing for display sync, or do nothing for normal timing.
static void handle_display_sync_frame(struct MPContext *mpctx,
struct vo_frame *frame)
{
struct MPOpts *opts = mpctx->opts;
struct vo *vo = mpctx->video_out;
int mode = opts->video_sync;
if (!mpctx->display_sync_active) {
mpctx->display_sync_error = 0.0;
mpctx->display_sync_drift_dir = 0;
}
mpctx->display_sync_active = false;
options: deprecate --video-sync=display-adrop A stupid thing that will probably be in the way.
2019-11-17 01:11:28 +00:00
if (mode == VS_DISP_ADROP && !mpctx->audio_drop_deprecated_msg) {
MP_WARN(mpctx, "video-sync=display-adrop mode is deprecated and will "
"be removed in the future.\n");
mpctx->audio_drop_deprecated_msg = true;
}
video: do not disable display-sync on A/V desync On a audio/video desync by more than 0.5 seconds, display-sync mode was disabled, and not enabled again (until playback restart, e.g. a seek). The idea was that it this only happens when this playback mode is broken and can't perform well anyway (A/V desync is a clear indication that something is very wrong). Instead of behaving like a god damn POS, it should revert to the more robust audio-sync mode. Unfortunately, this could happen sporadically due to temporary system performance problems, such as toggling fullscreen. Users didn't like this, and asked for a function to disable it, or to recover in some other way. This mechanism is questionable anyway. If an ignorant user enables display-sync, and encounters problems with it (without being able to determine that display-sync is messing up), the player will still behave like a POS on every playback, and even after every seek. It might actually be helpful to fail more consistently. Also, I've found that it's sill relatively reliable anyway even without this mechanism. So just remove the fallback. Fixes: #7048
2019-10-17 17:23:35 +00:00
if (!VS_IS_DISP(mode))
player: always disable display-sync on desyncs Instead of periodically trying to enable it again. There are two cases that can happen: 1. A random discontinuity messed everything up, 2. Things are just broken and will desync all the time Until now, it tried to deal with case 1 - but maybe this is really rare, and we don't really need to care about it. On the other hand, case 2 is kind of hard to diagnose if the user doesn't use the terminal. Seeking will reenable display-sync, so you can fix playback if case 1 happens, but still get predictable behavior in case 2.
2015-11-27 13:40:52 +00:00
return;
player: add display sync mode If this mode is enabled, the player tries to strictly synchronize video to display refresh. It will adjust playback speed to match the display, so if you play 23.976 fps video on a 24 Hz screen, playback speed is increased by approximately 1/1000. Audio wll be resampled to keep up with playback. This is different from the default sync mode, which will sync video to audio, with the consequence that video might skip or repeat a frame once in a while to make video keep up with audio. This is still unpolished. There are some major problems as well; in particular, mkv VFR files won't work well. The reason is that Matroska is terrible and rounds timestamps to milliseconds. This makes it rather hard to guess the framerate of a section of video that is playing. We could probably fix this by just accepting jittery timestamps (instead of explicitly disabling the sync code in this case), but I'm not ready to accept such a solution yet. Another issue is that we are extremely reliant on OS video and audio APIs working in an expected manner, which of course is not too often the case. Consequently, the new sync mode is a bit fragile.
2015-08-10 16:43:25 +00:00
bool resample = mode == VS_DISP_RESAMPLE || mode == VS_DISP_RESAMPLE_VDROP ||
mode == VS_DISP_RESAMPLE_NONE;
bool drop = mode == VS_DISP_VDROP || mode == VS_DISP_RESAMPLE ||
player: add audio drop/duplicate mode Not very robust in the moment.
2015-10-27 19:56:46 +00:00
mode == VS_DISP_ADROP || mode == VS_DISP_RESAMPLE_VDROP;
vo: pass through framedrop flag differently There is some sort-of awkwardness here, because option access needs to happen in a synchronized manner, and the framedrop flag is not in the VO option struct. Remove the mp_read_option_raw() call and the awkward change notification via VO_EVENT_WIN_STATE from command.c, and pass it through as new vo_frame flag.
2018-03-13 11:54:48 +00:00
drop &= frame->can_drop;
player: add display sync mode If this mode is enabled, the player tries to strictly synchronize video to display refresh. It will adjust playback speed to match the display, so if you play 23.976 fps video on a 24 Hz screen, playback speed is increased by approximately 1/1000. Audio wll be resampled to keep up with playback. This is different from the default sync mode, which will sync video to audio, with the consequence that video might skip or repeat a frame once in a while to make video keep up with audio. This is still unpolished. There are some major problems as well; in particular, mkv VFR files won't work well. The reason is that Matroska is terrible and rounds timestamps to milliseconds. This makes it rather hard to guess the framerate of a section of video that is playing. We could probably fix this by just accepting jittery timestamps (instead of explicitly disabling the sync code in this case), but I'm not ready to accept such a solution yet. Another issue is that we are extremely reliant on OS video and audio APIs working in an expected manner, which of course is not too often the case. Consequently, the new sync mode is a bit fragile.
2015-08-10 16:43:25 +00:00
if (resample && using_spdif_passthrough(mpctx))
player: always disable display-sync on desyncs Instead of periodically trying to enable it again. There are two cases that can happen: 1. A random discontinuity messed everything up, 2. Things are just broken and will desync all the time Until now, it tried to deal with case 1 - but maybe this is really rare, and we don't really need to care about it. On the other hand, case 2 is kind of hard to diagnose if the user doesn't use the terminal. Seeking will reenable display-sync, so you can fix playback if case 1 happens, but still get predictable behavior in case 2.
2015-11-27 13:40:52 +00:00
return;
player: add display sync mode If this mode is enabled, the player tries to strictly synchronize video to display refresh. It will adjust playback speed to match the display, so if you play 23.976 fps video on a 24 Hz screen, playback speed is increased by approximately 1/1000. Audio wll be resampled to keep up with playback. This is different from the default sync mode, which will sync video to audio, with the consequence that video might skip or repeat a frame once in a while to make video keep up with audio. This is still unpolished. There are some major problems as well; in particular, mkv VFR files won't work well. The reason is that Matroska is terrible and rounds timestamps to milliseconds. This makes it rather hard to guess the framerate of a section of video that is playing. We could probably fix this by just accepting jittery timestamps (instead of explicitly disabling the sync code in this case), but I'm not ready to accept such a solution yet. Another issue is that we are extremely reliant on OS video and audio APIs working in an expected manner, which of course is not too often the case. Consequently, the new sync mode is a bit fragile.
2015-08-10 16:43:25 +00:00
double vsync = vo_get_vsync_interval(vo) / 1e6;
if (vsync <= 0)
player: always disable display-sync on desyncs Instead of periodically trying to enable it again. There are two cases that can happen: 1. A random discontinuity messed everything up, 2. Things are just broken and will desync all the time Until now, it tried to deal with case 1 - but maybe this is really rare, and we don't really need to care about it. On the other hand, case 2 is kind of hard to diagnose if the user doesn't use the terminal. Seeking will reenable display-sync, so you can fix playback if case 1 happens, but still get predictable behavior in case 2.
2015-11-27 13:40:52 +00:00
return;
player: add display sync mode If this mode is enabled, the player tries to strictly synchronize video to display refresh. It will adjust playback speed to match the display, so if you play 23.976 fps video on a 24 Hz screen, playback speed is increased by approximately 1/1000. Audio wll be resampled to keep up with playback. This is different from the default sync mode, which will sync video to audio, with the consequence that video might skip or repeat a frame once in a while to make video keep up with audio. This is still unpolished. There are some major problems as well; in particular, mkv VFR files won't work well. The reason is that Matroska is terrible and rounds timestamps to milliseconds. This makes it rather hard to guess the framerate of a section of video that is playing. We could probably fix this by just accepting jittery timestamps (instead of explicitly disabling the sync code in this case), but I'm not ready to accept such a solution yet. Another issue is that we are extremely reliant on OS video and audio APIs working in an expected manner, which of course is not too often the case. Consequently, the new sync mode is a bit fragile.
2015-08-10 16:43:25 +00:00
player: don't make display-sync panic on timestamp discontinuities
2015-12-04 16:07:50 +00:00
double adjusted_duration = MPMAX(0, mpctx->past_frames[0].approx_duration);
player: refactor display-sync frame duration calculations Get rid of get_past_frame_durations(), which was a bit too messy. Add a past_frames array, which contains the same information in a more reasonable way. This also means that we can get the exact current and past frame durations without going through awful stuff. (The main problem is that vo_pts_history contains future frames as well, which is needed for frame backstepping etc., but gets in the way here.) Also disable the automatic disabling of display-sync if the frame duration changes, and extend the frame durations allowed for display sync. To allow arbitrarily high durations, vo.c needs to be changed to pause and potentially redraw OSD while showing a single frame, so they're still limited. In an attempt to deal with VFR, calculate the overall speed using the average FPS. The frame scheduling itself does not use the average FPS, but the duration of the current frame. This does not work too well, but provides a good base for further improvements. Where this commit actually helps a lot is dealing with rounded timestamps, e.g. if the container framerate is wrong or unknown, or if the muxer wrote incorrectly rounded timestamps. While the rounding errors apparently can't be get rid of completely in the general case, this is still much better than e.g. disabling display-sync completely just because some frame durations go out of bounds.
2015-11-13 21:45:40 +00:00
adjusted_duration /= opts->playback_speed;
player: don't make display-sync panic on timestamp discontinuities
2015-12-04 16:07:50 +00:00
if (adjusted_duration > 0.5)
player: always disable display-sync on desyncs Instead of periodically trying to enable it again. There are two cases that can happen: 1. A random discontinuity messed everything up, 2. Things are just broken and will desync all the time Until now, it tried to deal with case 1 - but maybe this is really rare, and we don't really need to care about it. On the other hand, case 2 is kind of hard to diagnose if the user doesn't use the terminal. Seeking will reenable display-sync, so you can fix playback if case 1 happens, but still get predictable behavior in case 2.
2015-11-27 13:40:52 +00:00
return;
player: add display sync mode If this mode is enabled, the player tries to strictly synchronize video to display refresh. It will adjust playback speed to match the display, so if you play 23.976 fps video on a 24 Hz screen, playback speed is increased by approximately 1/1000. Audio wll be resampled to keep up with playback. This is different from the default sync mode, which will sync video to audio, with the consequence that video might skip or repeat a frame once in a while to make video keep up with audio. This is still unpolished. There are some major problems as well; in particular, mkv VFR files won't work well. The reason is that Matroska is terrible and rounds timestamps to milliseconds. This makes it rather hard to guess the framerate of a section of video that is playing. We could probably fix this by just accepting jittery timestamps (instead of explicitly disabling the sync code in this case), but I'm not ready to accept such a solution yet. Another issue is that we are extremely reliant on OS video and audio APIs working in an expected manner, which of course is not too often the case. Consequently, the new sync mode is a bit fragile.
2015-08-10 16:43:25 +00:00
player: make display-vdrop mode do what the manpage claims Don't change video speed in this mode, which is closer to the claim on the manpage that it's close to the behavior of the "audio" mode.
2015-11-26 17:53:32 +00:00
mpctx->speed_factor_v = 1.0;
if (mode != VS_DISP_VDROP) {
double best = find_best_speed(mpctx, vsync);
// If it doesn't work, play at normal speed.
if (fabs(best - 1.0) <= opts->sync_max_video_change / 100)
mpctx->speed_factor_v = best;
}
player: add display sync mode If this mode is enabled, the player tries to strictly synchronize video to display refresh. It will adjust playback speed to match the display, so if you play 23.976 fps video on a 24 Hz screen, playback speed is increased by approximately 1/1000. Audio wll be resampled to keep up with playback. This is different from the default sync mode, which will sync video to audio, with the consequence that video might skip or repeat a frame once in a while to make video keep up with audio. This is still unpolished. There are some major problems as well; in particular, mkv VFR files won't work well. The reason is that Matroska is terrible and rounds timestamps to milliseconds. This makes it rather hard to guess the framerate of a section of video that is playing. We could probably fix this by just accepting jittery timestamps (instead of explicitly disabling the sync code in this case), but I'm not ready to accept such a solution yet. Another issue is that we are extremely reliant on OS video and audio APIs working in an expected manner, which of course is not too often the case. Consequently, the new sync mode is a bit fragile.
2015-08-10 16:43:25 +00:00
// Determine for how many vsyncs a frame should be displayed. This can be
// e.g. 2 for 30hz on a 60hz display. It can also be 0 if the video
// framerate is higher than the display framerate.
// We use the speed-adjusted (i.e. real) frame duration for this.
player: move audio speed adjustment code Move it (in a cosmetic sense), and also move its invocation to below all the video handling. All other changes remain cosmetic, including moving the framedrop calculation code, and getting rid of the video_speed_correction variable.
2015-11-04 18:37:15 +00:00
double frame_duration = adjusted_duration / mpctx->speed_factor_v;
player: add display sync mode If this mode is enabled, the player tries to strictly synchronize video to display refresh. It will adjust playback speed to match the display, so if you play 23.976 fps video on a 24 Hz screen, playback speed is increased by approximately 1/1000. Audio wll be resampled to keep up with playback. This is different from the default sync mode, which will sync video to audio, with the consequence that video might skip or repeat a frame once in a while to make video keep up with audio. This is still unpolished. There are some major problems as well; in particular, mkv VFR files won't work well. The reason is that Matroska is terrible and rounds timestamps to milliseconds. This makes it rather hard to guess the framerate of a section of video that is playing. We could probably fix this by just accepting jittery timestamps (instead of explicitly disabling the sync code in this case), but I'm not ready to accept such a solution yet. Another issue is that we are extremely reliant on OS video and audio APIs working in an expected manner, which of course is not too often the case. Consequently, the new sync mode is a bit fragile.
2015-08-10 16:43:25 +00:00
double ratio = (frame_duration + mpctx->display_sync_error) / vsync;
player: less naive rounding
2015-11-11 18:52:35 +00:00
int num_vsyncs = MPMAX(lrint(ratio), 0);
video: fix another A/V difference bug in display-sync mode This didn't show up with cases where the frame pattern has a cycle of 1 or 2 like it is the case with 24-on-24 fps, or 24-on-60 fps. It did show up with 25-on-60 fps. (We don't slow down 25 fps video to 24 on default settings.) In this case, we must not add the timing error of the next frame to the A/V difference estimation of the current frame. Use the previous timing error instead. This is another bug resulting from the confusion about whether we calculate parameters for the currently playing frame, or the one we're about to queue.
2015-10-31 12:19:38 +00:00
double prev_error = mpctx->display_sync_error;
player: add display sync mode If this mode is enabled, the player tries to strictly synchronize video to display refresh. It will adjust playback speed to match the display, so if you play 23.976 fps video on a 24 Hz screen, playback speed is increased by approximately 1/1000. Audio wll be resampled to keep up with playback. This is different from the default sync mode, which will sync video to audio, with the consequence that video might skip or repeat a frame once in a while to make video keep up with audio. This is still unpolished. There are some major problems as well; in particular, mkv VFR files won't work well. The reason is that Matroska is terrible and rounds timestamps to milliseconds. This makes it rather hard to guess the framerate of a section of video that is playing. We could probably fix this by just accepting jittery timestamps (instead of explicitly disabling the sync code in this case), but I'm not ready to accept such a solution yet. Another issue is that we are extremely reliant on OS video and audio APIs working in an expected manner, which of course is not too often the case. Consequently, the new sync mode is a bit fragile.
2015-08-10 16:43:25 +00:00
mpctx->display_sync_error += frame_duration - num_vsyncs * vsync;
msg: reinterpret a bunch of message levels I've decided that MP_TRACE means “noisy spam per frame”, whereas MP_DBG just means “more verbose debugging messages than MSGL_V”. Basically, MSGL_DBG shouldn't create spam per frame like it currently does, and MSGL_V should make sense to the end-user and provide mostly additional informational output. MP_DBG is basically what I want to make the new default for --log-file, so the cut-off point for MP_DBG is if we probably want to know if for debugging purposes but the user most likely doesn't care about on the terminal. Also, the debug callbacks for libass and ffmpeg got bumped in their verbosity levels slightly, because being external components they're a bit less relevant to mpv debugging, and a bit too over-eager in what they consider to be relevant information. I exclusively used the "try it on my machine and remove messages from MSGL_* until it does what I want it to" approach of refactoring, so YMMV.
2017-09-28 09:53:57 +00:00
MP_TRACE(mpctx, "s=%f vsyncs=%d dur=%f ratio=%f err=%.20f (%f/%f)\n",
mpctx->speed_factor_v, num_vsyncs, adjusted_duration, ratio,
mpctx->display_sync_error, mpctx->display_sync_error / vsync,
mpctx->display_sync_error / frame_duration);
player: add display sync mode If this mode is enabled, the player tries to strictly synchronize video to display refresh. It will adjust playback speed to match the display, so if you play 23.976 fps video on a 24 Hz screen, playback speed is increased by approximately 1/1000. Audio wll be resampled to keep up with playback. This is different from the default sync mode, which will sync video to audio, with the consequence that video might skip or repeat a frame once in a while to make video keep up with audio. This is still unpolished. There are some major problems as well; in particular, mkv VFR files won't work well. The reason is that Matroska is terrible and rounds timestamps to milliseconds. This makes it rather hard to guess the framerate of a section of video that is playing. We could probably fix this by just accepting jittery timestamps (instead of explicitly disabling the sync code in this case), but I'm not ready to accept such a solution yet. Another issue is that we are extremely reliant on OS video and audio APIs working in an expected manner, which of course is not too often the case. Consequently, the new sync mode is a bit fragile.
2015-08-10 16:43:25 +00:00
video: do not disable display-sync on A/V desync On a audio/video desync by more than 0.5 seconds, display-sync mode was disabled, and not enabled again (until playback restart, e.g. a seek). The idea was that it this only happens when this playback mode is broken and can't perform well anyway (A/V desync is a clear indication that something is very wrong). Instead of behaving like a god damn POS, it should revert to the more robust audio-sync mode. Unfortunately, this could happen sporadically due to temporary system performance problems, such as toggling fullscreen. Users didn't like this, and asked for a function to disable it, or to recover in some other way. This mechanism is questionable anyway. If an ignorant user enables display-sync, and encounters problems with it (without being able to determine that display-sync is messing up), the player will still behave like a POS on every playback, and even after every seek. It might actually be helpful to fail more consistently. Also, I've found that it's sill relatively reliable anyway even without this mechanism. So just remove the fallback. Fixes: #7048
2019-10-17 17:23:35 +00:00
double av_diff = mpctx->last_av_difference;
player: move audio speed adjustment code Move it (in a cosmetic sense), and also move its invocation to below all the video handling. All other changes remain cosmetic, including moving the framedrop calculation code, and getting rid of the video_speed_correction variable.
2015-11-04 18:37:15 +00:00
MP_STATS(mpctx, "value %f avdiff", av_diff);
// Intended number of additional display frames to drop (<0) or repeat (>0)
int drop_repeat = 0;
player: stricter framedrop threshold 80ms allowable desync was a bit too much. It'd allow for a range of 160ms, which everyone can notice. It might also be a bother to apply compensation resampling speed for that long.
2015-11-13 21:53:38 +00:00
// If we are too far ahead/behind, attempt to drop/repeat frames.
player: move audio speed adjustment code Move it (in a cosmetic sense), and also move its invocation to below all the video handling. All other changes remain cosmetic, including moving the framedrop calculation code, and getting rid of the video_speed_correction variable.
2015-11-04 18:37:15 +00:00
// Tolerate some desync to avoid frame dropping due to jitter.
player: stricter framedrop threshold 80ms allowable desync was a bit too much. It'd allow for a range of 160ms, which everyone can notice. It might also be a bother to apply compensation resampling speed for that long.
2015-11-13 21:53:38 +00:00
if (drop && fabs(av_diff) >= 0.020 && fabs(av_diff) / vsync >= 1)
player: move audio speed adjustment code Move it (in a cosmetic sense), and also move its invocation to below all the video handling. All other changes remain cosmetic, including moving the framedrop calculation code, and getting rid of the video_speed_correction variable.
2015-11-04 18:37:15 +00:00
drop_repeat = -av_diff / vsync; // round towards 0
player: add display sync mode If this mode is enabled, the player tries to strictly synchronize video to display refresh. It will adjust playback speed to match the display, so if you play 23.976 fps video on a 24 Hz screen, playback speed is increased by approximately 1/1000. Audio wll be resampled to keep up with playback. This is different from the default sync mode, which will sync video to audio, with the consequence that video might skip or repeat a frame once in a while to make video keep up with audio. This is still unpolished. There are some major problems as well; in particular, mkv VFR files won't work well. The reason is that Matroska is terrible and rounds timestamps to milliseconds. This makes it rather hard to guess the framerate of a section of video that is playing. We could probably fix this by just accepting jittery timestamps (instead of explicitly disabling the sync code in this case), but I'm not ready to accept such a solution yet. Another issue is that we are extremely reliant on OS video and audio APIs working in an expected manner, which of course is not too often the case. Consequently, the new sync mode is a bit fragile.
2015-08-10 16:43:25 +00:00
// We can only drop all frames at most. We can repeat much more frames,
// but we still limit it to 10 times the original frames to avoid that
// corner cases or exceptional situations cause too much havoc.
drop_repeat = MPCLAMP(drop_repeat, -num_vsyncs, num_vsyncs * 10);
num_vsyncs += drop_repeat;
player: always show the first frame in DS mode Fixes bogus frame drop counter in cover art mode.
2016-04-24 18:14:51 +00:00
// Always show the first frame.
if (mpctx->num_past_frames <= 1 && num_vsyncs < 1)
num_vsyncs = 1;
player: add display sync mode If this mode is enabled, the player tries to strictly synchronize video to display refresh. It will adjust playback speed to match the display, so if you play 23.976 fps video on a 24 Hz screen, playback speed is increased by approximately 1/1000. Audio wll be resampled to keep up with playback. This is different from the default sync mode, which will sync video to audio, with the consequence that video might skip or repeat a frame once in a while to make video keep up with audio. This is still unpolished. There are some major problems as well; in particular, mkv VFR files won't work well. The reason is that Matroska is terrible and rounds timestamps to milliseconds. This makes it rather hard to guess the framerate of a section of video that is playing. We could probably fix this by just accepting jittery timestamps (instead of explicitly disabling the sync code in this case), but I'm not ready to accept such a solution yet. Another issue is that we are extremely reliant on OS video and audio APIs working in an expected manner, which of course is not too often the case. Consequently, the new sync mode is a bit fragile.
2015-08-10 16:43:25 +00:00
// Estimate the video position, so we can calculate a good A/V difference
player: minor refactor for A/V diff computation Calculate the A/V difference directly in the display sync code, instead of the awkward current way, which reuses the fields for audio sync. We still set time_frame, because it makes falling back to audio sync somewhat smoother.
2015-10-28 22:50:54 +00:00
// value below. This is used to estimate A/V drift.
player: account for minor VO underruns If the player sends a frame with duration==0 to the VO, it can trivially underrun. Don't panic, but keep the correct time. Also, returning the absolute time from vo_get_next_frame_start_time() just to turn it into a float with relative time was silly. Rename it and make it return what the caller needs.
2015-11-14 20:44:59 +00:00
double time_left = vo_get_delay(vo);
player: add display sync mode If this mode is enabled, the player tries to strictly synchronize video to display refresh. It will adjust playback speed to match the display, so if you play 23.976 fps video on a 24 Hz screen, playback speed is increased by approximately 1/1000. Audio wll be resampled to keep up with playback. This is different from the default sync mode, which will sync video to audio, with the consequence that video might skip or repeat a frame once in a while to make video keep up with audio. This is still unpolished. There are some major problems as well; in particular, mkv VFR files won't work well. The reason is that Matroska is terrible and rounds timestamps to milliseconds. This makes it rather hard to guess the framerate of a section of video that is playing. We could probably fix this by just accepting jittery timestamps (instead of explicitly disabling the sync code in this case), but I'm not ready to accept such a solution yet. Another issue is that we are extremely reliant on OS video and audio APIs working in an expected manner, which of course is not too often the case. Consequently, the new sync mode is a bit fragile.
2015-08-10 16:43:25 +00:00
// We also know that the timing is (necessarily) off, because we have to
// align frame timings on the vsync boundaries. This is unavoidable, and
video: fix another A/V difference bug in display-sync mode This didn't show up with cases where the frame pattern has a cycle of 1 or 2 like it is the case with 24-on-24 fps, or 24-on-60 fps. It did show up with 25-on-60 fps. (We don't slow down 25 fps video to 24 on default settings.) In this case, we must not add the timing error of the next frame to the A/V difference estimation of the current frame. Use the previous timing error instead. This is another bug resulting from the confusion about whether we calculate parameters for the currently playing frame, or the one we're about to queue.
2015-10-31 12:19:38 +00:00
// for the sake of the A/V sync calculations we pretend it's perfect.
time_left += prev_error;
player: fix display sync A/V difference estimation on drops When dropping or repeating frames, we essentially influence when the frame after the next frame will be shown, not the next frame. This led to dropping/repeating frames 2 times, because the A/V difference had a delay of one frame. Compensate it with the expected value.
2015-10-28 22:50:37 +00:00
// Likewise, we know sync is off, but is going to be compensated.
player: minor refactor for A/V diff computation Calculate the A/V difference directly in the display sync code, instead of the awkward current way, which reuses the fields for audio sync. We still set time_frame, because it makes falling back to audio sync somewhat smoother.
2015-10-28 22:50:54 +00:00
time_left += drop_repeat * vsync;
player: fix display-sync timing if audio take long on resume In display-sync mode, the very first video frame is idiotically fully timed, even though audio has not been synced yet at this point, and the video frame is more like a "preview" frame. But since it's fully timed, an underflow is detected if audio takes longer than the display time of the frame (we send the second frame only after audio is done). The timing code will try to compensate for the determined desync, but it really shouldn't. So explicitly discard the timing info in this specific case. On the other hand, if the first frame still hasn't finished display, we can pretend everything is ok. This is a hack - ideally, we either would send a frame without timing info (and then send it again or so when playback starts properly), or we would add real pause support to the VO, and pause it during syncing.
2016-08-07 12:06:54 +00:00
// If syncing took too long, disregard timing of the first frame.
if (mpctx->num_past_frames == 2 && time_left < 0) {
vo_discard_timing_info(vo);
time_left = 0;
}
player: log some more display-sync information
2015-11-25 21:07:47 +00:00
if (drop_repeat) {
command: add mistimed-frame-count property Does what the manpage says. This is a replacement incrementing the dropped frame counter (see previous commit).
2015-10-30 13:05:41 +00:00
mpctx->mistimed_frames_total += 1;
player: log some more display-sync information
2015-11-25 21:07:47 +00:00
MP_STATS(mpctx, "mistimed");
}
command: add mistimed-frame-count property Does what the manpage says. This is a replacement incrementing the dropped frame counter (see previous commit).
2015-10-30 13:05:41 +00:00
player: disable total-avsync-change update in display-sync mode The total-avsync-change property made no sense in display-sync mode (in addition to making not all that much sense in general).
2015-10-27 19:56:58 +00:00
mpctx->total_avsync_change = 0;
player: another fix to A/V difference calculation in display-sync mode update_av_diff() works on the timestamps, while time_left is in real time. When playing at not-1 speed, these are very different, and cause the A/V difference to jitter. Fix this by scaling the expected A/V desync to the correct range.
2015-11-01 18:53:48 +00:00
update_av_diff(mpctx, time_left * opts->playback_speed);
player: minor refactor for A/V diff computation Calculate the A/V difference directly in the display sync code, instead of the awkward current way, which reuses the fields for audio sync. We still set time_frame, because it makes falling back to audio sync somewhat smoother.
2015-10-28 22:50:54 +00:00
command: add vsync-ratio property This is very "illustrative", unlike the video-speed-correction property, and thus useful. It can also be used to observe scheduling errors, which are not detected by the core. (These happen due to rounding errors; possibly not evne our fault, but coming from files with rounded timestamps and so on.)
2015-11-13 21:48:32 +00:00
mpctx->past_frames[0].num_vsyncs = num_vsyncs;
player: try to compensate actual audio drift We always let audio slowly desync until a threshold is reached, and then pushed it back by applying a maximum compensation speed. Refine what comes afterwards: instead of playing with the nominal video speed, use the actual required audio speed for keeping sync as measured by the A/V difference. (The "actual" speed is the ideal speed with A/V differences added.) Although this works in theory, it's somewhat questionable how much this works in practice. The ideal time value is actually not exact, but is the time at which the frame is scheduled (could be compensated by using the time_left calculations in handle_display_sync_frame()). It doesn't account for speed changes or catastrophic discontinuities. It uses only 10 past frames.
2015-11-13 21:51:39 +00:00
mpctx->past_frames[0].av_diff = mpctx->last_av_difference;
command: add vsync-ratio property This is very "illustrative", unlike the video-speed-correction property, and thus useful. It can also be used to observe scheduling errors, which are not detected by the core. (These happen due to rounding errors; possibly not evne our fault, but coming from files with rounded timestamps and so on.)
2015-11-13 21:48:32 +00:00
player: change display-sync audio speed only if needed As long as it's within the desync tolerance, do not change the audio speed at all for resampling. This reduces speed changes which might be caused by jittering timestamps and similar cases. (While in theory you could just not care and change speed every single frame, I'm afraid that such changes could possibly cause audio artifacts. So better just avoid it in the first place.)
2015-11-13 21:50:58 +00:00
if (resample) {
adjust_audio_resample_speed(mpctx, vsync);
} else {
mpctx->speed_factor_a = 1.0;
}
player: move audio speed adjustment code Move it (in a cosmetic sense), and also move its invocation to below all the video handling. All other changes remain cosmetic, including moving the framedrop calculation code, and getting rid of the video_speed_correction variable.
2015-11-04 18:37:15 +00:00
player: minor refactor for A/V diff computation Calculate the A/V difference directly in the display sync code, instead of the awkward current way, which reuses the fields for audio sync. We still set time_frame, because it makes falling back to audio sync somewhat smoother.
2015-10-28 22:50:54 +00:00
// A bad guess, only needed when reverting to audio sync.
mpctx->time_frame = time_left;
player: add display sync mode If this mode is enabled, the player tries to strictly synchronize video to display refresh. It will adjust playback speed to match the display, so if you play 23.976 fps video on a 24 Hz screen, playback speed is increased by approximately 1/1000. Audio wll be resampled to keep up with playback. This is different from the default sync mode, which will sync video to audio, with the consequence that video might skip or repeat a frame once in a while to make video keep up with audio. This is still unpolished. There are some major problems as well; in particular, mkv VFR files won't work well. The reason is that Matroska is terrible and rounds timestamps to milliseconds. This makes it rather hard to guess the framerate of a section of video that is playing. We could probably fix this by just accepting jittery timestamps (instead of explicitly disabling the sync code in this case), but I'm not ready to accept such a solution yet. Another issue is that we are extremely reliant on OS video and audio APIs working in an expected manner, which of course is not too often the case. Consequently, the new sync mode is a bit fragile.
2015-08-10 16:43:25 +00:00
vo: change vo_frame field units This was just converting back and forth between int64_t/microseconds and double/seconds. Remove this stupidity. The pts/duration fields are still in microseconds, but they have no meaning in the display-sync case (also drop printing the pts field from opengl/video.c - it's always 0).
2015-11-27 21:04:44 +00:00
frame->vsync_interval = vsync;
vo_opengl: fix interpolation with display-sync At least I hope so. Deriving the duration from the pts was not really correct. It doesn't include speed adjustments, and becomes completely wrong of the user e.g. changes the playback speed by a huge amount. Pass through the accurate duration value by adding a new vo_frame field. The value for vsync_offset was not correct either. We don't need the error for the next frame, but the error for the current one. This wasn't noticed because it makes no difference in symmetric cases, like 24 fps on 60 Hz. I'm still not entirely confident in the correctness of this, but it sure is an improvement. Also, remove the MP_STATS() calls - they're not really useful to debug anything anymore.
2015-11-28 14:45:35 +00:00
frame->vsync_offset = -prev_error;
frame->ideal_frame_duration = frame_duration;
player: add display sync mode If this mode is enabled, the player tries to strictly synchronize video to display refresh. It will adjust playback speed to match the display, so if you play 23.976 fps video on a 24 Hz screen, playback speed is increased by approximately 1/1000. Audio wll be resampled to keep up with playback. This is different from the default sync mode, which will sync video to audio, with the consequence that video might skip or repeat a frame once in a while to make video keep up with audio. This is still unpolished. There are some major problems as well; in particular, mkv VFR files won't work well. The reason is that Matroska is terrible and rounds timestamps to milliseconds. This makes it rather hard to guess the framerate of a section of video that is playing. We could probably fix this by just accepting jittery timestamps (instead of explicitly disabling the sync code in this case), but I'm not ready to accept such a solution yet. Another issue is that we are extremely reliant on OS video and audio APIs working in an expected manner, which of course is not too often the case. Consequently, the new sync mode is a bit fragile.
2015-08-10 16:43:25 +00:00
frame->num_vsyncs = num_vsyncs;
frame->display_synced = true;
mpctx->display_sync_active = true;
player: change display-sync audio speed only if needed As long as it's within the desync tolerance, do not change the audio speed at all for resampling. This reduces speed changes which might be caused by jittering timestamps and similar cases. (While in theory you could just not care and change speed every single frame, I'm afraid that such changes could possibly cause audio artifacts. So better just avoid it in the first place.)
2015-11-13 21:50:58 +00:00
update_playback_speed(mpctx);
player: add display sync mode If this mode is enabled, the player tries to strictly synchronize video to display refresh. It will adjust playback speed to match the display, so if you play 23.976 fps video on a 24 Hz screen, playback speed is increased by approximately 1/1000. Audio wll be resampled to keep up with playback. This is different from the default sync mode, which will sync video to audio, with the consequence that video might skip or repeat a frame once in a while to make video keep up with audio. This is still unpolished. There are some major problems as well; in particular, mkv VFR files won't work well. The reason is that Matroska is terrible and rounds timestamps to milliseconds. This makes it rather hard to guess the framerate of a section of video that is playing. We could probably fix this by just accepting jittery timestamps (instead of explicitly disabling the sync code in this case), but I'm not ready to accept such a solution yet. Another issue is that we are extremely reliant on OS video and audio APIs working in an expected manner, which of course is not too often the case. Consequently, the new sync mode is a bit fragile.
2015-08-10 16:43:25 +00:00
player: change display-sync audio speed only if needed As long as it's within the desync tolerance, do not change the audio speed at all for resampling. This reduces speed changes which might be caused by jittering timestamps and similar cases. (While in theory you could just not care and change speed every single frame, I'm afraid that such changes could possibly cause audio artifacts. So better just avoid it in the first place.)
2015-11-13 21:50:58 +00:00
MP_STATS(mpctx, "value %f aspeed", mpctx->speed_factor_a - 1);
MP_STATS(mpctx, "value %f vspeed", mpctx->speed_factor_v - 1);
player: always disable display-sync on desyncs Instead of periodically trying to enable it again. There are two cases that can happen: 1. A random discontinuity messed everything up, 2. Things are just broken and will desync all the time Until now, it tried to deal with case 1 - but maybe this is really rare, and we don't really need to care about it. On the other hand, case 2 is kind of hard to diagnose if the user doesn't use the terminal. Seeking will reenable display-sync, so you can fix playback if case 1 happens, but still get predictable behavior in case 2.
2015-11-27 13:40:52 +00:00
}
player: add display sync mode If this mode is enabled, the player tries to strictly synchronize video to display refresh. It will adjust playback speed to match the display, so if you play 23.976 fps video on a 24 Hz screen, playback speed is increased by approximately 1/1000. Audio wll be resampled to keep up with playback. This is different from the default sync mode, which will sync video to audio, with the consequence that video might skip or repeat a frame once in a while to make video keep up with audio. This is still unpolished. There are some major problems as well; in particular, mkv VFR files won't work well. The reason is that Matroska is terrible and rounds timestamps to milliseconds. This makes it rather hard to guess the framerate of a section of video that is playing. We could probably fix this by just accepting jittery timestamps (instead of explicitly disabling the sync code in this case), but I'm not ready to accept such a solution yet. Another issue is that we are extremely reliant on OS video and audio APIs working in an expected manner, which of course is not too often the case. Consequently, the new sync mode is a bit fragile.
2015-08-10 16:43:25 +00:00
player: always disable display-sync on desyncs Instead of periodically trying to enable it again. There are two cases that can happen: 1. A random discontinuity messed everything up, 2. Things are just broken and will desync all the time Until now, it tried to deal with case 1 - but maybe this is really rare, and we don't really need to care about it. On the other hand, case 2 is kind of hard to diagnose if the user doesn't use the terminal. Seeking will reenable display-sync, so you can fix playback if case 1 happens, but still get predictable behavior in case 2.
2015-11-27 13:40:52 +00:00
static void schedule_frame(struct MPContext *mpctx, struct vo_frame *frame)
{
handle_display_sync_frame(mpctx, frame);
player: add display sync mode If this mode is enabled, the player tries to strictly synchronize video to display refresh. It will adjust playback speed to match the display, so if you play 23.976 fps video on a 24 Hz screen, playback speed is increased by approximately 1/1000. Audio wll be resampled to keep up with playback. This is different from the default sync mode, which will sync video to audio, with the consequence that video might skip or repeat a frame once in a while to make video keep up with audio. This is still unpolished. There are some major problems as well; in particular, mkv VFR files won't work well. The reason is that Matroska is terrible and rounds timestamps to milliseconds. This makes it rather hard to guess the framerate of a section of video that is playing. We could probably fix this by just accepting jittery timestamps (instead of explicitly disabling the sync code in this case), but I'm not ready to accept such a solution yet. Another issue is that we are extremely reliant on OS video and audio APIs working in an expected manner, which of course is not too often the case. Consequently, the new sync mode is a bit fragile.
2015-08-10 16:43:25 +00:00
player: remove display_sync_disable_counter We can implement it differently and drop a tiny bit of state.
2015-11-13 21:49:50 +00:00
if (mpctx->num_past_frames > 1 &&
((mpctx->past_frames[1].num_vsyncs >= 0) != mpctx->display_sync_active))
{
player: add display sync mode If this mode is enabled, the player tries to strictly synchronize video to display refresh. It will adjust playback speed to match the display, so if you play 23.976 fps video on a 24 Hz screen, playback speed is increased by approximately 1/1000. Audio wll be resampled to keep up with playback. This is different from the default sync mode, which will sync video to audio, with the consequence that video might skip or repeat a frame once in a while to make video keep up with audio. This is still unpolished. There are some major problems as well; in particular, mkv VFR files won't work well. The reason is that Matroska is terrible and rounds timestamps to milliseconds. This makes it rather hard to guess the framerate of a section of video that is playing. We could probably fix this by just accepting jittery timestamps (instead of explicitly disabling the sync code in this case), but I'm not ready to accept such a solution yet. Another issue is that we are extremely reliant on OS video and audio APIs working in an expected manner, which of course is not too often the case. Consequently, the new sync mode is a bit fragile.
2015-08-10 16:43:25 +00:00
MP_VERBOSE(mpctx, "Video sync mode %s.\n",
mpctx->display_sync_active ? "enabled" : "disabled");
}
player: minor refactor for A/V diff computation Calculate the A/V difference directly in the display sync code, instead of the awkward current way, which reuses the fields for audio sync. We still set time_frame, because it makes falling back to audio sync somewhat smoother.
2015-10-28 22:50:54 +00:00
if (!mpctx->display_sync_active) {
player: change display-sync audio speed only if needed As long as it's within the desync tolerance, do not change the audio speed at all for resampling. This reduces speed changes which might be caused by jittering timestamps and similar cases. (While in theory you could just not care and change speed every single frame, I'm afraid that such changes could possibly cause audio artifacts. So better just avoid it in the first place.)
2015-11-13 21:50:58 +00:00
mpctx->speed_factor_a = 1.0;
mpctx->speed_factor_v = 1.0;
update_playback_speed(mpctx);
player: minor refactor for A/V diff computation Calculate the A/V difference directly in the display sync code, instead of the awkward current way, which reuses the fields for audio sync. We still set time_frame, because it makes falling back to audio sync somewhat smoother.
2015-10-28 22:50:54 +00:00
update_av_diff(mpctx, mpctx->time_frame > 0 ?
mpctx->time_frame * mpctx->video_speed : 0);
}
}
player: refactor display-sync frame duration calculations Get rid of get_past_frame_durations(), which was a bit too messy. Add a past_frames array, which contains the same information in a more reasonable way. This also means that we can get the exact current and past frame durations without going through awful stuff. (The main problem is that vo_pts_history contains future frames as well, which is needed for frame backstepping etc., but gets in the way here.) Also disable the automatic disabling of display-sync if the frame duration changes, and extend the frame durations allowed for display sync. To allow arbitrarily high durations, vo.c needs to be changed to pause and potentially redraw OSD while showing a single frame, so they're still limited. In an attempt to deal with VFR, calculate the overall speed using the average FPS. The frame scheduling itself does not use the average FPS, but the duration of the current frame. This does not work too well, but provides a good base for further improvements. Where this commit actually helps a lot is dealing with rounded timestamps, e.g. if the container framerate is wrong or unknown, or if the muxer wrote incorrectly rounded timestamps. While the rounding errors apparently can't be get rid of completely in the general case, this is still much better than e.g. disabling display-sync completely just because some frame durations go out of bounds.
2015-11-13 21:45:40 +00:00
// Determine the mpctx->past_frames[0] frame duration.
static void calculate_frame_duration(struct MPContext *mpctx)
video: move frame duration code to a separate function Minor preparation for something else.
2015-08-01 20:17:09 +00:00
{
video: fix passing down FPS to vf_vapoursynth To make this less of a mess, remove one of the redundant container_fps fields. Part of #5470.
2018-02-01 08:20:25 +00:00
struct vo_chain *vo_c = mpctx->vo_chain;
player: refactor display-sync frame duration calculations Get rid of get_past_frame_durations(), which was a bit too messy. Add a past_frames array, which contains the same information in a more reasonable way. This also means that we can get the exact current and past frame durations without going through awful stuff. (The main problem is that vo_pts_history contains future frames as well, which is needed for frame backstepping etc., but gets in the way here.) Also disable the automatic disabling of display-sync if the frame duration changes, and extend the frame durations allowed for display sync. To allow arbitrarily high durations, vo.c needs to be changed to pause and potentially redraw OSD while showing a single frame, so they're still limited. In an attempt to deal with VFR, calculate the overall speed using the average FPS. The frame scheduling itself does not use the average FPS, but the duration of the current frame. This does not work too well, but provides a good base for further improvements. Where this commit actually helps a lot is dealing with rounded timestamps, e.g. if the container framerate is wrong or unknown, or if the muxer wrote incorrectly rounded timestamps. While the rounding errors apparently can't be get rid of completely in the general case, this is still much better than e.g. disabling display-sync completely just because some frame durations go out of bounds.
2015-11-13 21:45:40 +00:00
assert(mpctx->num_past_frames >= 1 && mpctx->num_next_frames >= 1);
video: move frame duration code to a separate function Minor preparation for something else.
2015-08-01 20:17:09 +00:00
video: fix passing down FPS to vf_vapoursynth To make this less of a mess, remove one of the redundant container_fps fields. Part of #5470.
2018-02-01 08:20:25 +00:00
double demux_duration = vo_c->filter->container_fps > 0
? 1.0 / vo_c->filter->container_fps : -1;
video: trust demuxer framerate on invalid timestamps If the PTS goes backwards (whether it's a timestamp reset or some other problem) would just use 0 as frame duration. (At least until the logic for detecting divergence with the timestamps gets active.) Trust the demuxer framerate in these cases instead, if it's available. I think this improves behavior slightly with some broken files.
2016-09-26 15:55:00 +00:00
double duration = demux_duration;
player: refactor display-sync frame duration calculations Get rid of get_past_frame_durations(), which was a bit too messy. Add a past_frames array, which contains the same information in a more reasonable way. This also means that we can get the exact current and past frame durations without going through awful stuff. (The main problem is that vo_pts_history contains future frames as well, which is needed for frame backstepping etc., but gets in the way here.) Also disable the automatic disabling of display-sync if the frame duration changes, and extend the frame durations allowed for display sync. To allow arbitrarily high durations, vo.c needs to be changed to pause and potentially redraw OSD while showing a single frame, so they're still limited. In an attempt to deal with VFR, calculate the overall speed using the average FPS. The frame scheduling itself does not use the average FPS, but the duration of the current frame. This does not work too well, but provides a good base for further improvements. Where this commit actually helps a lot is dealing with rounded timestamps, e.g. if the container framerate is wrong or unknown, or if the muxer wrote incorrectly rounded timestamps. While the rounding errors apparently can't be get rid of completely in the general case, this is still much better than e.g. disabling display-sync completely just because some frame durations go out of bounds.
2015-11-13 21:45:40 +00:00
if (mpctx->num_next_frames >= 2) {
double pts0 = mpctx->next_frames[0]->pts;
double pts1 = mpctx->next_frames[1]->pts;
if (pts0 != MP_NOPTS_VALUE && pts1 != MP_NOPTS_VALUE && pts1 >= pts0)
duration = pts1 - pts0;
video: move frame duration code to a separate function Minor preparation for something else.
2015-08-01 20:17:09 +00:00
}
player: refactor display-sync frame duration calculations Get rid of get_past_frame_durations(), which was a bit too messy. Add a past_frames array, which contains the same information in a more reasonable way. This also means that we can get the exact current and past frame durations without going through awful stuff. (The main problem is that vo_pts_history contains future frames as well, which is needed for frame backstepping etc., but gets in the way here.) Also disable the automatic disabling of display-sync if the frame duration changes, and extend the frame durations allowed for display sync. To allow arbitrarily high durations, vo.c needs to be changed to pause and potentially redraw OSD while showing a single frame, so they're still limited. In an attempt to deal with VFR, calculate the overall speed using the average FPS. The frame scheduling itself does not use the average FPS, but the duration of the current frame. This does not work too well, but provides a good base for further improvements. Where this commit actually helps a lot is dealing with rounded timestamps, e.g. if the container framerate is wrong or unknown, or if the muxer wrote incorrectly rounded timestamps. While the rounding errors apparently can't be get rid of completely in the general case, this is still much better than e.g. disabling display-sync completely just because some frame durations go out of bounds.
2015-11-13 21:45:40 +00:00
// The following code tries to compensate for rounded Matroska timestamps
// by "unrounding" frame durations, or if not possible, approximating them.
player: slightly adjust framerate guessing heuristic Some files not only use rounded timestamps, but they also do it incorrectly. They may jitter between up to 4 specific frame durations. In this case, I found a file that mostly used 41ms and 42ms, but also had 40ms and 43ms outliers (often but not always following each other). This breaks the assumption of the framerate estimation code that the frame duration can deviate up to 1ms. If it jitters around 4 possible frame durations, the maximum deviation is 3ms. Increase it accordingly. The change might make playback of "true VFR" video via display-sync mode worse, but it's not like it was particularly good in the first place. Also, the check whether to usen the container FPS needs to be stricter. In the worst case, num_dur is 1, which doesn't really indicate any evidence that the framerate is correct. Only if there are "enough" frames the deviation check will become meaningful. 16 is an arbitrary value that has been designated "enough" by myself. Also otuput the frame duration values for --dump-stats.
2016-08-29 15:15:06 +00:00
// These formats usually round on 1ms. Some muxers do this incorrectly,
// and might go off by 1ms more, and compensate for it later by an equal
// rounding error into the opposite direction.
double tolerance = 0.001 * 3 + 0.0001;
player: refactor display-sync frame duration calculations Get rid of get_past_frame_durations(), which was a bit too messy. Add a past_frames array, which contains the same information in a more reasonable way. This also means that we can get the exact current and past frame durations without going through awful stuff. (The main problem is that vo_pts_history contains future frames as well, which is needed for frame backstepping etc., but gets in the way here.) Also disable the automatic disabling of display-sync if the frame duration changes, and extend the frame durations allowed for display sync. To allow arbitrarily high durations, vo.c needs to be changed to pause and potentially redraw OSD while showing a single frame, so they're still limited. In an attempt to deal with VFR, calculate the overall speed using the average FPS. The frame scheduling itself does not use the average FPS, but the duration of the current frame. This does not work too well, but provides a good base for further improvements. Where this commit actually helps a lot is dealing with rounded timestamps, e.g. if the container framerate is wrong or unknown, or if the muxer wrote incorrectly rounded timestamps. While the rounding errors apparently can't be get rid of completely in the general case, this is still much better than e.g. disabling display-sync completely just because some frame durations go out of bounds.
2015-11-13 21:45:40 +00:00
double total = 0;
int num_dur = 0;
for (int n = 1; n < mpctx->num_past_frames; n++) {
// Eliminate likely outliers using a really dumb heuristic.
double dur = mpctx->past_frames[n].duration;
if (dur <= 0 || fabs(dur - duration) >= tolerance)
break;
total += dur;
num_dur += 1;
}
player: smooth out frame durations by averaging them Helps somewhat with muxer-rounded timestamps. There is some danger that this introduces a timestamp drift. But since they are averaged values (unlike as when using an incorrect container framerate hint), any potential drift shouldn't be too brutal, or compensate itself soon. So I won't bother yet with comparing the results with the real timestamp, unless we run into actual problems. Of course we still prefer potentially real timestamps over the approximated ones. But unless the timestamps match the container FPS, we can't know whether they are (no, checking whether the they have microsecond components would be cheating). Perhaps in future, we could let the demuxer export the timebase - if the timebase is not 1000 (or divisible by it), we know that millisecond-rounded timestamps won't happen.
2015-11-13 21:46:55 +00:00
double approx_duration = num_dur > 0 ? total / num_dur : duration;
player: refactor display-sync frame duration calculations Get rid of get_past_frame_durations(), which was a bit too messy. Add a past_frames array, which contains the same information in a more reasonable way. This also means that we can get the exact current and past frame durations without going through awful stuff. (The main problem is that vo_pts_history contains future frames as well, which is needed for frame backstepping etc., but gets in the way here.) Also disable the automatic disabling of display-sync if the frame duration changes, and extend the frame durations allowed for display sync. To allow arbitrarily high durations, vo.c needs to be changed to pause and potentially redraw OSD while showing a single frame, so they're still limited. In an attempt to deal with VFR, calculate the overall speed using the average FPS. The frame scheduling itself does not use the average FPS, but the duration of the current frame. This does not work too well, but provides a good base for further improvements. Where this commit actually helps a lot is dealing with rounded timestamps, e.g. if the container framerate is wrong or unknown, or if the muxer wrote incorrectly rounded timestamps. While the rounding errors apparently can't be get rid of completely in the general case, this is still much better than e.g. disabling display-sync completely just because some frame durations go out of bounds.
2015-11-13 21:45:40 +00:00
// Try if the demuxer frame rate fits - if so, just take it.
if (demux_duration > 0) {
// Note that even if each timestamp is within rounding tolerance, it
// could literally not add up (e.g. if demuxer FPS is rounded itself).
if (fabs(duration - demux_duration) < tolerance &&
video: trust container FPS early on if possible If the container FPS is correct, this can help getting ideal mix factors for vo_gpu interpolation mode. Otherwise, it doesn't matter.
2018-05-10 14:49:50 +00:00
fabs(total - demux_duration * num_dur) < tolerance &&
(num_dur >= 16 || num_dur >= mpctx->num_past_frames - 4))
player: refactor display-sync frame duration calculations Get rid of get_past_frame_durations(), which was a bit too messy. Add a past_frames array, which contains the same information in a more reasonable way. This also means that we can get the exact current and past frame durations without going through awful stuff. (The main problem is that vo_pts_history contains future frames as well, which is needed for frame backstepping etc., but gets in the way here.) Also disable the automatic disabling of display-sync if the frame duration changes, and extend the frame durations allowed for display sync. To allow arbitrarily high durations, vo.c needs to be changed to pause and potentially redraw OSD while showing a single frame, so they're still limited. In an attempt to deal with VFR, calculate the overall speed using the average FPS. The frame scheduling itself does not use the average FPS, but the duration of the current frame. This does not work too well, but provides a good base for further improvements. Where this commit actually helps a lot is dealing with rounded timestamps, e.g. if the container framerate is wrong or unknown, or if the muxer wrote incorrectly rounded timestamps. While the rounding errors apparently can't be get rid of completely in the general case, this is still much better than e.g. disabling display-sync completely just because some frame durations go out of bounds.
2015-11-13 21:45:40 +00:00
{
approx_duration = demux_duration;
}
}
mpctx->past_frames[0].duration = duration;
mpctx->past_frames[0].approx_duration = approx_duration;
player: slightly adjust framerate guessing heuristic Some files not only use rounded timestamps, but they also do it incorrectly. They may jitter between up to 4 specific frame durations. In this case, I found a file that mostly used 41ms and 42ms, but also had 40ms and 43ms outliers (often but not always following each other). This breaks the assumption of the framerate estimation code that the frame duration can deviate up to 1ms. If it jitters around 4 possible frame durations, the maximum deviation is 3ms. Increase it accordingly. The change might make playback of "true VFR" video via display-sync mode worse, but it's not like it was particularly good in the first place. Also, the check whether to usen the container FPS needs to be stricter. In the worst case, num_dur is 1, which doesn't really indicate any evidence that the framerate is correct. Only if there are "enough" frames the deviation check will become meaningful. 16 is an arbitrary value that has been designated "enough" by myself. Also otuput the frame duration values for --dump-stats.
2016-08-29 15:15:06 +00:00
MP_STATS(mpctx, "value %f frame-duration", MPMAX(0, duration));
MP_STATS(mpctx, "value %f frame-duration-approx", MPMAX(0, approx_duration));
video: move frame duration code to a separate function Minor preparation for something else.
2015-08-01 20:17:09 +00:00
}
player: minor simplification No need to pass endpts down in such a dumb way. Also remove an outdated comment somewhere.
2016-02-27 19:02:51 +00:00
void write_video(struct MPContext *mpctx)
player: move video display code out of the playloop Basically move the code from playloop.c to video.c. The new function write_video() now contains the code that was part of run_playloop(). There are no functional changes, except handling "new_frame_shown" slightly differently. This is done so that we don't need new a new MPContext field or a return value for write_video() to signal this condition. Instead, it's handled indirectly.
2014-07-30 21:24:08 +00:00
{
struct MPOpts *opts = mpctx->opts;
player: refactor: eliminate MPContext.d_video Eventually we want the VO be driven by a A->V filter, so a decoder doesn't even have to exist. Some features definitely require a decoder though (like reporting the decoder in use, hardware decoding, etc.), so for each thing which accessed d_video, it has to be redecided if and how it can access decoder state. At least the "framedrop" property slightly changes semantics: you can now always set this property, even if no video is active. Some untested changes in this commit, but our bio-based distributed test suite has to take care of this.
2016-01-17 17:07:50 +00:00
if (!mpctx->vo_chain)
player: move video display code out of the playloop Basically move the code from playloop.c to video.c. The new function write_video() now contains the code that was part of run_playloop(). There are no functional changes, except handling "new_frame_shown" slightly differently. This is done so that we don't need new a new MPContext field or a return value for write_video() to signal this condition. Instead, it's handled indirectly.
2014-07-30 21:24:08 +00:00
return;
player: remove some further current_track dependencies Now it's used for initialization only for audio and video.
2016-02-05 22:41:44 +00:00
struct track *track = mpctx->vo_chain->track;
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 vo_chain *vo_c = mpctx->vo_chain;
struct vo *vo = vo_c->vo;
if (vo_c->filter->reconfig_happened) {
mp_notify(mpctx, MPV_EVENT_VIDEO_RECONFIG, NULL);
vo_c->filter->reconfig_happened = false;
}
player: move video display code out of the playloop Basically move the code from playloop.c to video.c. The new function write_video() now contains the code that was part of run_playloop(). There are no functional changes, except handling "new_frame_shown" slightly differently. This is done so that we don't need new a new MPContext field or a return value for write_video() to signal this condition. Instead, it's handled indirectly.
2014-07-30 21:24:08 +00:00
video: exit early when nothing to do These cases were probably confusing. Exit early, which makes it much clearer what's going on. Should not change anything functionally.
2014-08-13 19:51:26 +00:00
// Actual playback starts when both audio and video are ready.
if (mpctx->video_status == STATUS_READY)
return;
if (mpctx->paused && mpctx->video_status >= STATUS_READY)
return;
player: make repeated hr-seeks past EOF trigger EOF as expected If you have a normal file with audio and video, and keep "spamming" forward hr-seeks, the player just kept showing the last video frame instead of exiting or playing the next file. This started happening since commit 6bcda94cb. Although not a bug per se, it was odd, and very user-noticable. The main problem was that the pending seek command was processed before the EOF was "noticed". Processing the command reset everything, so the player did not terminate playback, but repeated the seek. This commit restores the old behavior. For one, it makes video return the correct status (video.c). The parameter is a bit ugly, but better than duplicating the logic or having another MPContext field. (As a minor detail, setting r=VD_EOF makes sure have_new_frame() returns true, rather than going through another iteration or whatever the hell will happen instead, which would clobber logical_eof.) Another thing is making the seek logic actually wait until the seek outcome has been determined if audio is also active. Audio needs to wait for video in order to get the video seek target position. (Which in turn is because hr-seek still "snaps" to video frames. You can't seek in between two frames, so audio can't just use the seek target, but always has to wait on the timestamp of the video frame. This has other disadvantages and is a misdesign, but not something I'll fix today.) In theory, this might make hr-seeks less responsive, because it needs to fully decode/filter the audio too, but in practice most time is spent on video, which had to be fully decoded before this change. (In general, hr-seek could probably just show a random frame when a queued hr-seek overrides the current hr-seek, which would probably lead to a better user experience, but that's out of scope.) Fixes: #7206
2019-12-14 13:17:16 +00:00
bool logical_eof = false;
int r = video_output_image(mpctx, &logical_eof);
video: refactor queue handling This simplifies the code, and fixes an odd bug: the second-last frame was displayed for a very short duration if framedrop was enabled. The reason was that basically the time difference between second-last and last frame were skipped, because at this point EOF was already signaled. Also see commit b0959488 for a similar issue in the same code. This removes the messiness of the next_frame 2-frame queue, and strictly runs the "new frame" code when a frame is moved to the first position of the queue, instead of somehow messing with return codes. This also merges update_video() into video_output_image().
2014-08-22 12:20:30 +00:00
MP_TRACE(mpctx, "video_output_image: %d\n", r);
player: move video display code out of the playloop Basically move the code from playloop.c to video.c. The new function write_video() now contains the code that was part of run_playloop(). There are no functional changes, except handling "new_frame_shown" slightly differently. This is done so that we don't need new a new MPContext field or a return value for write_video() to signal this condition. Instead, it's handled indirectly.
2014-07-30 21:24:08 +00:00
video: fix and simplify video format changes and last frame display The previous commit broke these things, and fixing them is separate in this commit in order to reduce the volume of changes. Move the image queue from the VO to the playback core. The image queue is a remnant of the old way how vdpau was implemented, and increasingly became more and more an artifact. In the end, it did only one thing: computing the duration of the current frame. This was done by taking the PTS difference between the current and the future frame. We keep this, but by moving it out of the VO, we don't have to special-case format changes anymore. This simplifies the code a lot. Since we need the queue to compute the duration only, a queue size larger than 2 makes no sense, and we can hardcode that. Also change how the last frame is handled. The last frame is a bit of a problem, because video timing works by showing one frame after another, which makes it a special case. Make the VO provide a function to notify us when the frame is done, instead. The frame duration is used for that. This is not perfect. For example, changing playback speed during the last frame doesn't update the end time. Pausing will not stop the clock that times the last frame. But I don't think this matters for such a corner case.
2014-08-12 21:17:35 +00:00
if (r < 0)
goto error;
player: move video display code out of the playloop Basically move the code from playloop.c to video.c. The new function write_video() now contains the code that was part of run_playloop(). There are no functional changes, except handling "new_frame_shown" slightly differently. This is done so that we don't need new a new MPContext field or a return value for write_video() to signal this condition. Instead, it's handled indirectly.
2014-07-30 21:24:08 +00:00
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
if (r == VD_WAIT) {
// Heuristic to detect underruns.
player: avoid underrun wakeup loop The VO underrun detection (just a weak heuristic) added in commit f26dfb flagged the underrun state every time it was checked, and since the check happened in every playloop iteration, this caused the playloop to wake up itself on every iteration. It burned an entire core while in this state. Fix this by flagging this condition only once (as it should be), and requiring that a frame is displayed to trigger it again. This makes it work similar as the audio underrun check. The bug report referenced below says --demuxer-thread=no avoided this. This is because the demuxer layer doesn't do proper underrun reporting if the reader thread is disabled. Fixes: #7259
2019-12-16 00:15:43 +00:00
if (mpctx->video_status == STATUS_PLAYING && !vo_still_displaying(vo) &&
!vo_c->underrun_signaled)
{
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
vo_c->underrun = true;
player: avoid underrun wakeup loop The VO underrun detection (just a weak heuristic) added in commit f26dfb flagged the underrun state every time it was checked, and since the check happened in every playloop iteration, this caused the playloop to wake up itself on every iteration. It burned an entire core while in this state. Fix this by flagging this condition only once (as it should be), and requiring that a frame is displayed to trigger it again. This makes it work similar as the audio underrun check. The bug report referenced below says --demuxer-thread=no avoided this. This is because the demuxer layer doesn't do proper underrun reporting if the reader thread is disabled. Fixes: #7259
2019-12-16 00:15:43 +00:00
vo_c->underrun_signaled = true;
}
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
// Demuxer will wake us up for more packets to decode.
video: fix and simplify video format changes and last frame display The previous commit broke these things, and fixing them is separate in this commit in order to reduce the volume of changes. Move the image queue from the VO to the playback core. The image queue is a remnant of the old way how vdpau was implemented, and increasingly became more and more an artifact. In the end, it did only one thing: computing the duration of the current frame. This was done by taking the PTS difference between the current and the future frame. We keep this, but by moving it out of the VO, we don't have to special-case format changes anymore. This simplifies the code a lot. Since we need the queue to compute the duration only, a queue size larger than 2 makes no sense, and we can hardcode that. Also change how the last frame is handled. The last frame is a bit of a problem, because video timing works by showing one frame after another, which makes it a special case. Make the VO provide a function to notify us when the frame is done, instead. The frame duration is used for that. This is not perfect. For example, changing playback speed during the last frame doesn't update the end time. Pausing will not stop the clock that times the last frame. But I don't think this matters for such a corner case.
2014-08-12 21:17:35 +00:00
return;
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
}
player: move video display code out of the playloop Basically move the code from playloop.c to video.c. The new function write_video() now contains the code that was part of run_playloop(). There are no functional changes, except handling "new_frame_shown" slightly differently. This is done so that we don't need new a new MPContext field or a return value for write_video() to signal this condition. Instead, it's handled indirectly.
2014-07-30 21:24:08 +00:00
video: minor simplification This is mostly equivalent, but simpler, and reduces some duplication.
2014-08-22 12:18:40 +00:00
if (r == VD_EOF) {
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
if (check_for_hwdec_fallback(mpctx))
return;
if (vo_c->filter->failed_output_conversion)
goto error;
player: move video display code out of the playloop Basically move the code from playloop.c to video.c. The new function write_video() now contains the code that was part of run_playloop(). There are no functional changes, except handling "new_frame_shown" slightly differently. This is done so that we don't need new a new MPContext field or a return value for write_video() to signal this condition. Instead, it's handled indirectly.
2014-07-30 21:24:08 +00:00
mpctx->delay = 0;
mpctx->last_av_difference = 0;
player: add option to control duration of image display The --image-display-duration option controls how long an image is displayed. It's also possible to display the image forever (until manual user interaction stops playback). With this, the core drops the old method to "drain" video (i.e. waiting for the last frame duration on end of playback). Instead, we reuse MPContext.time_frame. The old mechanism was disabled for non-images anyway. Fixes #3425.
2016-08-17 20:45:44 +00:00
if (mpctx->video_status <= STATUS_PLAYING) {
mpctx->video_status = STATUS_DRAINING;
get_relative_time(mpctx);
if (mpctx->num_past_frames == 1 && mpctx->past_frames[0].pts == 0 &&
!mpctx->ao_chain)
{
player: log if video is considered an image It's a heristic that can fail, so better log it.
2016-08-21 10:27:48 +00:00
MP_VERBOSE(mpctx, "assuming this is an image\n");
player: add option to control duration of image display The --image-display-duration option controls how long an image is displayed. It's also possible to display the image forever (until manual user interaction stops playback). With this, the core drops the old method to "drain" video (i.e. waiting for the last frame duration on end of playback). Instead, we reuse MPContext.time_frame. The old mechanism was disabled for non-images anyway. Fixes #3425.
2016-08-17 20:45:44 +00:00
mpctx->time_frame += opts->image_display_duration;
video: use demuxer-signaled duration for last video frame Helps with gif, probably does unwanted things with other formats. This doesn't handle --end quite correctly, but this could be added later. Fixes #3924.
2016-12-21 17:18:24 +00:00
} else if (mpctx->last_frame_duration > 0) {
MP_VERBOSE(mpctx, "using demuxer frame duration for last frame\n");
mpctx->time_frame += mpctx->last_frame_duration;
player: add option to control duration of image display The --image-display-duration option controls how long an image is displayed. It's also possible to display the image forever (until manual user interaction stops playback). With this, the core drops the old method to "drain" video (i.e. waiting for the last frame duration on end of playback). Instead, we reuse MPContext.time_frame. The old mechanism was disabled for non-images anyway. Fixes #3425.
2016-08-17 20:45:44 +00:00
} else {
mpctx->time_frame = 0;
}
player: don't wait for last video frame in encode mode This code makes the player wait using real time, which makes sense for normal playback, but not encode mode.
2018-04-26 16:46:18 +00:00
// Encode mode can't honor this; it'll only delay finishing.
if (mpctx->encode_lavc_ctx)
mpctx->time_frame = 0;
player: add option to control duration of image display The --image-display-duration option controls how long an image is displayed. It's also possible to display the image forever (until manual user interaction stops playback). With this, the core drops the old method to "drain" video (i.e. waiting for the last frame duration on end of playback). Instead, we reuse MPContext.time_frame. The old mechanism was disabled for non-images anyway. Fixes #3425.
2016-08-17 20:45:44 +00:00
}
video: actually wait for last frame being rendered on EOF The video timing code could just decide that EOF was reached before it was displayed. This is not really a problem for normal playback (if you use something like --keep-open it'd show the last frame anyway, otherwise it'd at best flash it on screen before destroying the window). But in encode mode, it really matters, and makes the difference between having one frame more or less in the output file. Fix this by waiting for the VO before starting the real EOF. vo_is_ready_for_frame() is normally used to determine when the VO frame queue has enough space to send a new frame. Since the VO frame queue is currently at most 1 frame, it being signaled means the remaining frame was consumed and thus sent to the VO driver. If it returns false, it will wake up the playloop as soon as the state changes. I also considered using vo_still_displaying(), but it's not reliable, because it checks the realtime of the frame end display time.
2018-04-29 14:04:50 +00:00
// Wait for the VO to signal actual EOF, then exit if the frame timer
// has expired.
video: fix player not exiting if no video frame was rendered E.g. "mpv null:// --demuxer=rawvideo" will "hang" by waiting for video EOF forever. It's not signalled correctly because of the last-frame corner case, which attempts to wait until the current frame is finally displayed (which is signalled by whether a new frame can be queued, see commit 1a339fa09d for some details). If no frame was ever queued, the VO is not configured, and vo_is_ready_for_frame() never returns true. Fix this by using vo_has_frame(), which seems to be exactly the correct thing we need.
2019-01-11 10:02:23 +00:00
bool has_frame = vo_has_frame(vo); // maybe not configured
video: actually wait for last frame being rendered on EOF The video timing code could just decide that EOF was reached before it was displayed. This is not really a problem for normal playback (if you use something like --keep-open it'd show the last frame anyway, otherwise it'd at best flash it on screen before destroying the window). But in encode mode, it really matters, and makes the difference between having one frame more or less in the output file. Fix this by waiting for the VO before starting the real EOF. vo_is_ready_for_frame() is normally used to determine when the VO frame queue has enough space to send a new frame. Since the VO frame queue is currently at most 1 frame, it being signaled means the remaining frame was consumed and thus sent to the VO driver. If it returns false, it will wake up the playloop as soon as the state changes. I also considered using vo_still_displaying(), but it's not reliable, because it checks the realtime of the frame end display time.
2018-04-29 14:04:50 +00:00
if (mpctx->video_status == STATUS_DRAINING &&
video: fix player not exiting if no video frame was rendered E.g. "mpv null:// --demuxer=rawvideo" will "hang" by waiting for video EOF forever. It's not signalled correctly because of the last-frame corner case, which attempts to wait until the current frame is finally displayed (which is signalled by whether a new frame can be queued, see commit 1a339fa09d for some details). If no frame was ever queued, the VO is not configured, and vo_is_ready_for_frame() never returns true. Fix this by using vo_has_frame(), which seems to be exactly the correct thing we need.
2019-01-11 10:02:23 +00:00
(vo_is_ready_for_frame(vo, -1) || !has_frame))
video: actually wait for last frame being rendered on EOF The video timing code could just decide that EOF was reached before it was displayed. This is not really a problem for normal playback (if you use something like --keep-open it'd show the last frame anyway, otherwise it'd at best flash it on screen before destroying the window). But in encode mode, it really matters, and makes the difference between having one frame more or less in the output file. Fix this by waiting for the VO before starting the real EOF. vo_is_ready_for_frame() is normally used to determine when the VO frame queue has enough space to send a new frame. Since the VO frame queue is currently at most 1 frame, it being signaled means the remaining frame was consumed and thus sent to the VO driver. If it returns false, it will wake up the playloop as soon as the state changes. I also considered using vo_still_displaying(), but it's not reliable, because it checks the realtime of the frame end display time.
2018-04-29 14:04:50 +00:00
{
player: add option to control duration of image display The --image-display-duration option controls how long an image is displayed. It's also possible to display the image forever (until manual user interaction stops playback). With this, the core drops the old method to "drain" video (i.e. waiting for the last frame duration on end of playback). Instead, we reuse MPContext.time_frame. The old mechanism was disabled for non-images anyway. Fixes #3425.
2016-08-17 20:45:44 +00:00
mpctx->time_frame -= get_relative_time(mpctx);
player: litter code with explicit wakeup calls This does 3 kinds of changes: - change sleeptime=x to mp_set_timeout() - change sleeptime=0 to mp_wakeup_core() calls (to be more explicit) - change commands etc. to call mp_wakeup_core() if they do changes that require the playloop to be rerun This is preparation for the following changes. The goal is to process client API requests without having to rerun the playloop every time. As of this commit, the changes should not change behavior. In particular, the playloop is still implicitly woken up on every command.
2016-09-16 12:24:15 +00:00
mp_set_timeout(mpctx, mpctx->time_frame);
video: fix player not exiting if no video frame was rendered E.g. "mpv null:// --demuxer=rawvideo" will "hang" by waiting for video EOF forever. It's not signalled correctly because of the last-frame corner case, which attempts to wait until the current frame is finally displayed (which is signalled by whether a new frame can be queued, see commit 1a339fa09d for some details). If no frame was ever queued, the VO is not configured, and vo_is_ready_for_frame() never returns true. Fix this by using vo_has_frame(), which seems to be exactly the correct thing we need.
2019-01-11 10:02:23 +00:00
if (mpctx->time_frame <= 0 || !has_frame) {
player: add option to control duration of image display The --image-display-duration option controls how long an image is displayed. It's also possible to display the image forever (until manual user interaction stops playback). With this, the core drops the old method to "drain" video (i.e. waiting for the last frame duration on end of playback). Instead, we reuse MPContext.time_frame. The old mechanism was disabled for non-images anyway. Fixes #3425.
2016-08-17 20:45:44 +00:00
MP_VERBOSE(mpctx, "video EOF reached\n");
mpctx->video_status = STATUS_EOF;
encode: get rid of the output packet queue Until recently, ao_lavc and vo_lavc started encoding whenever the core happened to send them data. Since audio and video are not initialized at the same time, and the muxer was not necessarily opened when the first encoder started to produce data, the resulting packets were put into a queue. As soon as the muxer was opened, the queue was flushed. Change this to make the core wait with sending data until all encoders are initialized. This has the advantage that we don't need to queue up the packets.
2018-04-29 17:42:18 +00:00
encode_lavc_stream_eof(mpctx->encode_lavc_ctx, STREAM_VIDEO);
player: add option to control duration of image display The --image-display-duration option controls how long an image is displayed. It's also possible to display the image forever (until manual user interaction stops playback). With this, the core drops the old method to "drain" video (i.e. waiting for the last frame duration on end of playback). Instead, we reuse MPContext.time_frame. The old mechanism was disabled for non-images anyway. Fixes #3425.
2016-08-17 20:45:44 +00:00
}
}
player: silence spam in verbose mode when playing audio with cover art When playing cover art, it conceptually reaches EOF as soon as the image was put on the VO, causing the EOF message to be repeated every time new audio was decoded. Just silence the message.
2015-04-14 12:36:15 +00:00
MP_DBG(mpctx, "video EOF (status=%d)\n", mpctx->video_status);
video: fix attached picture mode Playing audio files with embedded cover art broke due to some of the recent changes. Treat video EOF properly, and don't burn the CPU. Disable hrseek for video in attached picture mode, since the decoder will always produce a new image, which makes hrseek never terminate. Fixes #970.
2014-07-31 19:57:05 +00:00
return;
player: move video display code out of the playloop Basically move the code from playloop.c to video.c. The new function write_video() now contains the code that was part of run_playloop(). There are no functional changes, except handling "new_frame_shown" slightly differently. This is done so that we don't need new a new MPContext field or a return value for write_video() to signal this condition. Instead, it's handled indirectly.
2014-07-30 21:24:08 +00:00
}
video: minor simplification This is mostly equivalent, but simpler, and reduces some duplication.
2014-08-22 12:18:40 +00:00
if (mpctx->video_status > STATUS_PLAYING)
mpctx->video_status = STATUS_PLAYING;
if (r != VD_NEW_FRAME) {
player: litter code with explicit wakeup calls This does 3 kinds of changes: - change sleeptime=x to mp_set_timeout() - change sleeptime=0 to mp_wakeup_core() calls (to be more explicit) - change commands etc. to call mp_wakeup_core() if they do changes that require the playloop to be rerun This is preparation for the following changes. The goal is to process client API requests without having to rerun the playloop every time. As of this commit, the changes should not change behavior. In particular, the playloop is still implicitly woken up on every command.
2016-09-16 12:24:15 +00:00
mp_wakeup_core(mpctx); // Decode more in next iteration.
player: move video display code out of the playloop Basically move the code from playloop.c to video.c. The new function write_video() now contains the code that was part of run_playloop(). There are no functional changes, except handling "new_frame_shown" slightly differently. This is done so that we don't need new a new MPContext field or a return value for write_video() to signal this condition. Instead, it's handled indirectly.
2014-07-30 21:24:08 +00:00
return;
video: minor simplification This is mostly equivalent, but simpler, and reduces some duplication.
2014-08-22 12:18:40 +00:00
}
player: move video display code out of the playloop Basically move the code from playloop.c to video.c. The new function write_video() now contains the code that was part of run_playloop(). There are no functional changes, except handling "new_frame_shown" slightly differently. This is done so that we don't need new a new MPContext field or a return value for write_video() to signal this condition. Instead, it's handled indirectly.
2014-07-30 21:24:08 +00:00
video: fix and simplify video format changes and last frame display The previous commit broke these things, and fixing them is separate in this commit in order to reduce the volume of changes. Move the image queue from the VO to the playback core. The image queue is a remnant of the old way how vdpau was implemented, and increasingly became more and more an artifact. In the end, it did only one thing: computing the duration of the current frame. This was done by taking the PTS difference between the current and the future frame. We keep this, but by moving it out of the VO, we don't have to special-case format changes anymore. This simplifies the code a lot. Since we need the queue to compute the duration only, a queue size larger than 2 makes no sense, and we can hardcode that. Also change how the last frame is handled. The last frame is a bit of a problem, because video timing works by showing one frame after another, which makes it a special case. Make the VO provide a function to notify us when the frame is done, instead. The frame duration is used for that. This is not perfect. For example, changing playback speed during the last frame doesn't update the end time. Pausing will not stop the clock that times the last frame. But I don't think this matters for such a corner case.
2014-08-12 21:17:35 +00:00
// Filter output is different from VO input?
video: pass future frames to VO Now the VO can request a number of future frames with the last parameter of vo_set_queue_params(). This will be helpful to fix the interpolation code. Note that the first frame (after playback start or seeking) will usually not have any future frames (to make seeking fast). Near the end of the file, the number of future frames will become lower as well.
2015-07-01 17:22:40 +00:00
struct mp_image_params p = mpctx->next_frames[0]->params;
video: fix and simplify video format changes and last frame display The previous commit broke these things, and fixing them is separate in this commit in order to reduce the volume of changes. Move the image queue from the VO to the playback core. The image queue is a remnant of the old way how vdpau was implemented, and increasingly became more and more an artifact. In the end, it did only one thing: computing the duration of the current frame. This was done by taking the PTS difference between the current and the future frame. We keep this, but by moving it out of the VO, we don't have to special-case format changes anymore. This simplifies the code a lot. Since we need the queue to compute the duration only, a queue size larger than 2 makes no sense, and we can hardcode that. Also change how the last frame is handled. The last frame is a bit of a problem, because video timing works by showing one frame after another, which makes it a special case. Make the VO provide a function to notify us when the frame is done, instead. The frame duration is used for that. This is not perfect. For example, changing playback speed during the last frame doesn't update the end time. Pausing will not stop the clock that times the last frame. But I don't think this matters for such a corner case.
2014-08-12 21:17:35 +00:00
if (!vo->params || !mp_image_params_equal(&p, vo->params)) {
// Changing config deletes the current frame; wait until it's finished.
player: respect --untimed on last frame The last video frame is another case that has a separate code path, although it's pretty similar to the one in commit 73e5aa87. Fix this in a different way, which also takes care of the last frame case, although without context the code becomes slightly more tricky. As further cleanup, move the decision about framedropping itself to the same place, so the check in vo.c becomes much simpler. The check for the vo->driver->encode flag, which is remvoed completely, was redundant too. Fixes #1480.
2015-01-16 12:45:43 +00:00
if (vo_still_displaying(vo))
video: fix and simplify video format changes and last frame display The previous commit broke these things, and fixing them is separate in this commit in order to reduce the volume of changes. Move the image queue from the VO to the playback core. The image queue is a remnant of the old way how vdpau was implemented, and increasingly became more and more an artifact. In the end, it did only one thing: computing the duration of the current frame. This was done by taking the PTS difference between the current and the future frame. We keep this, but by moving it out of the VO, we don't have to special-case format changes anymore. This simplifies the code a lot. Since we need the queue to compute the duration only, a queue size larger than 2 makes no sense, and we can hardcode that. Also change how the last frame is handled. The last frame is a bit of a problem, because video timing works by showing one frame after another, which makes it a special case. Make the VO provide a function to notify us when the frame is done, instead. The frame duration is used for that. This is not perfect. For example, changing playback speed during the last frame doesn't update the end time. Pausing will not stop the clock that times the last frame. But I don't think this matters for such a corner case.
2014-08-12 21:17:35 +00:00
return;
const struct vo_driver *info = mpctx->video_out->driver;
player: print anamorphic size only if video is anamorphic Has been annoying me since forever.
2014-11-02 16:34:02 +00:00
char extra[20] = {0};
video: switch from using display aspect to sample aspect MPlayer traditionally always used the display aspect ratio, e.g. 16:9, while FFmpeg uses the sample (aka pixel) aspect ratio. Both have a bunch of advantages and disadvantages. Actually, it seems using sample aspect ratio is generally nicer. The main reason for the change is making mpv closer to how FFmpeg works in order to make life easier. It's also nice that everything uses integer fractions instead of floats now (except --video-aspect option/property). Note that there is at least 1 user-visible change: vf_dsize now does not set the display size, only the display aspect ratio. This is because the image_params d_w/d_h fields did not just set the display aspect, but also the size (except in encoding mode).
2015-12-19 19:04:31 +00:00
if (p.p_w != p.p_h) {
int d_w, d_h;
mp_image_params_get_dsize(&p, &d_w, &d_h);
snprintf(extra, sizeof(extra), " => %dx%d", d_w, d_h);
}
player: print hw format on "VO: " line too Useful for distinguishing bit depth when hardware decoding. (To the degree it's useful to show it at all. This just brings the hardware decoding case on the same level of showing information as the software decode call.)
2017-01-29 11:53:58 +00:00
char sfmt[20] = {0};
if (p.hw_subfmt)
snprintf(sfmt, sizeof(sfmt), "[%s]", mp_imgfmt_to_name(p.hw_subfmt));
MP_INFO(mpctx, "VO: [%s] %dx%d%s %s%s\n",
info->name, p.w, p.h, extra, mp_imgfmt_to_name(p.imgfmt), sfmt);
video: fix and simplify video format changes and last frame display The previous commit broke these things, and fixing them is separate in this commit in order to reduce the volume of changes. Move the image queue from the VO to the playback core. The image queue is a remnant of the old way how vdpau was implemented, and increasingly became more and more an artifact. In the end, it did only one thing: computing the duration of the current frame. This was done by taking the PTS difference between the current and the future frame. We keep this, but by moving it out of the VO, we don't have to special-case format changes anymore. This simplifies the code a lot. Since we need the queue to compute the duration only, a queue size larger than 2 makes no sense, and we can hardcode that. Also change how the last frame is handled. The last frame is a bit of a problem, because video timing works by showing one frame after another, which makes it a special case. Make the VO provide a function to notify us when the frame is done, instead. The frame duration is used for that. This is not perfect. For example, changing playback speed during the last frame doesn't update the end time. Pausing will not stop the clock that times the last frame. But I don't think this matters for such a corner case.
2014-08-12 21:17:35 +00:00
MP_VERBOSE(mpctx, "VO: Description: %s\n", info->description);
vo: add vo_reconfig2() 1. I want to get away from mp_image_params (maybe). 2. For encoding mode, it's convenient to get the nominal_fps, which is a mp_image field, and not in mp_image_params.
2018-04-22 17:36:16 +00:00
int vo_r = vo_reconfig2(vo, mpctx->next_frames[0]);
client API: better error reporting Give somewhat more information on playback failure.
2014-10-28 15:19:07 +00:00
if (vo_r < 0) {
mpctx->error_playing = MPV_ERROR_VO_INIT_FAILED;
video: fix and simplify video format changes and last frame display The previous commit broke these things, and fixing them is separate in this commit in order to reduce the volume of changes. Move the image queue from the VO to the playback core. The image queue is a remnant of the old way how vdpau was implemented, and increasingly became more and more an artifact. In the end, it did only one thing: computing the duration of the current frame. This was done by taking the PTS difference between the current and the future frame. We keep this, but by moving it out of the VO, we don't have to special-case format changes anymore. This simplifies the code a lot. Since we need the queue to compute the duration only, a queue size larger than 2 makes no sense, and we can hardcode that. Also change how the last frame is handled. The last frame is a bit of a problem, because video timing works by showing one frame after another, which makes it a special case. Make the VO provide a function to notify us when the frame is done, instead. The frame duration is used for that. This is not perfect. For example, changing playback speed during the last frame doesn't update the end time. Pausing will not stop the clock that times the last frame. But I don't think this matters for such a corner case.
2014-08-12 21:17:35 +00:00
goto error;
client API: better error reporting Give somewhat more information on playback failure.
2014-10-28 15:19:07 +00:00
}
video: redo video equalizer option handling I really wouldn't care much about this, but some parts of the core code are under HAVE_GPL, so there's some need to get rid of it. Simply turn the video equalizer from its current fine-grained handling with vf/vo fallbacks into global options. This makes updating them much simpler. This removes any possibility of applying video equalizers in filters, which affects vf_scale, and the previously removed vf_eq. Not a big loss, since the preferred VOs have this builtin. Remove video equalizer handling from vo_direct3d, vo_sdl, vo_vaapi, and vo_xv. I'm not going to waste my time on these legacy VOs. vo.eq_opts_cache exists _only_ to send a VOCTRL_SET_EQUALIZER, which exists _only_ to trigger a redraw. This seems silly, but for now I feel like this is less of a pain. The rest of the equalizer using code is self-updating. See commit 96b906a51d5 for how some video equalizer code was GPL only. Some command line option names and ranges can probably be traced back to a GPL only committer, but we don't consider these copyrightable.
2017-08-22 15:01:35 +00:00
mp_notify(mpctx, MPV_EVENT_VIDEO_RECONFIG, NULL);
video: fix and simplify video format changes and last frame display The previous commit broke these things, and fixing them is separate in this commit in order to reduce the volume of changes. Move the image queue from the VO to the playback core. The image queue is a remnant of the old way how vdpau was implemented, and increasingly became more and more an artifact. In the end, it did only one thing: computing the duration of the current frame. This was done by taking the PTS difference between the current and the future frame. We keep this, but by moving it out of the VO, we don't have to special-case format changes anymore. This simplifies the code a lot. Since we need the queue to compute the duration only, a queue size larger than 2 makes no sense, and we can hardcode that. Also change how the last frame is handled. The last frame is a bit of a problem, because video timing works by showing one frame after another, which makes it a special case. Make the VO provide a function to notify us when the frame is done, instead. The frame duration is used for that. This is not perfect. For example, changing playback speed during the last frame doesn't update the end time. Pausing will not stop the clock that times the last frame. But I don't think this matters for such a corner case.
2014-08-12 21:17:35 +00:00
}
player: rearrange some A/V-sync related code Move the update_avsync_before_frame() call further down. Moving it closer to where the time_frame value is used (and which the function updates) should make the code more readable. With this change, there's no need anymore to reset the time_frame value on the video reconfig path. Move the update_avsync_after_frame() up. Now no meaningful amount of time passes since the previous get_relative_time() call anymore, and the second one can be removed.
2015-01-30 20:18:30 +00:00
mpctx->time_frame -= get_relative_time(mpctx);
update_avsync_before_frame(mpctx);
player: make sure non-video subtitle rendering is reset if video resumes If video reaches EOF, subtitle timing will be switched to timing without video frames. This means it calls osd_set_force_video_pts() and overrides the PTS of whatever video frame is current (since the video frame's PTS has nothing to do with the current playback position anymore). This was not reset when seeking back into video. Subtitles wouldn't show up, or if there was a subtitle displayed, it would get stuck with it. In particular, this could happen even if EOF was only temporary (such as with --keep-open). Fix this by clearing the override PTS whenever a video frame is shown. Fixes #3770.
2016-11-18 12:01:46 +00:00
// Enforce timing subtitles to video frames.
osd_set_force_video_pts(mpctx->osd, MP_NOPTS_VALUE);
sub: change how subtitles are read Slightly change how it is decided when a new packet should be read. Switch to demux_read_packet_async(), and let the player "wait properly" until required subtitle packets arrive, instead of blocking everything. Move distinguishing the cases of passive and active reading into the demuxer, where it belongs.
2015-12-29 00:35:52 +00:00
if (!update_subtitles(mpctx, mpctx->next_frames[0]->pts)) {
Fix various typos in log messages
2017-10-23 08:53:28 +00:00
MP_VERBOSE(mpctx, "Video frame delayed due to waiting on subtitles.\n");
sub: change how subtitles are read Slightly change how it is decided when a new packet should be read. Switch to demux_read_packet_async(), and let the player "wait properly" until required subtitle packets arrive, instead of blocking everything. Move distinguishing the cases of passive and active reading into the demuxer, where it belongs.
2015-12-29 00:35:52 +00:00
return;
}
video: move display and timing to a separate thread The VO is run inside its own thread. It also does most of video timing. The playloop hands the image data and a realtime timestamp to the VO, and the VO does the rest. In particular, this allows the playloop to do other things, instead of blocking for video redraw. But if anything accesses the VO during video timing, it will block. This also fixes vo_sdl.c event handling; but that is only a side-effect, since reimplementing the broken way would require more effort. Also drop --softsleep. In theory, this option helps if the kernel's sleeping mechanism is too inaccurate for video timing. In practice, I haven't ever encountered a situation where it helps, and it just burns CPU cycles. On the other hand it's probably actively harmful, because it prevents the libavcodec decoder threads from doing real work. Side note: Originally, I intended that multiple frames can be queued to the VO. But this is not done, due to problems with OSD and other certain features. OSD in particular is simply designed in a way that it can be neither timed nor copied, so you do have to render it into the video frame before you can draw the next frame. (Subtitles have no such restriction. sd_lavc was even updated to fix this.) It seems the right solution to queuing multiple VO frames is rendering on VO-backed framebuffers, like vo_vdpau.c does. This requires VO driver support, and is out of scope of this commit. As consequence, the VO has a queue size of 1. The existing video queue is just needed to compute frame duration, and will be moved out in the next commit.
2014-08-12 21:02:08 +00:00
double time_frame = MPMAX(mpctx->time_frame, -1);
int64_t pts = mp_time_us() + (int64_t)(time_frame * 1e6);
player: move video display code out of the playloop Basically move the code from playloop.c to video.c. The new function write_video() now contains the code that was part of run_playloop(). There are no functional changes, except handling "new_frame_shown" slightly differently. This is done so that we don't need new a new MPContext field or a return value for write_video() to signal this condition. Instead, it's handled indirectly.
2014-07-30 21:24:08 +00:00
video: better pipelining with vf_vapoursynth Most of this is explained in the code comments. This change should improve performance with vapoursynth, especially if concurrent requests are used. This should change nothing if vf_vapoursynth is not in the filter chain, since non-threaded filters obviously can not asynchronously finish filtering of frames.
2015-01-03 02:01:58 +00:00
// wait until VO wakes us up to get more frames
player: add display sync mode If this mode is enabled, the player tries to strictly synchronize video to display refresh. It will adjust playback speed to match the display, so if you play 23.976 fps video on a 24 Hz screen, playback speed is increased by approximately 1/1000. Audio wll be resampled to keep up with playback. This is different from the default sync mode, which will sync video to audio, with the consequence that video might skip or repeat a frame once in a while to make video keep up with audio. This is still unpolished. There are some major problems as well; in particular, mkv VFR files won't work well. The reason is that Matroska is terrible and rounds timestamps to milliseconds. This makes it rather hard to guess the framerate of a section of video that is playing. We could probably fix this by just accepting jittery timestamps (instead of explicitly disabling the sync code in this case), but I'm not ready to accept such a solution yet. Another issue is that we are extremely reliant on OS video and audio APIs working in an expected manner, which of course is not too often the case. Consequently, the new sync mode is a bit fragile.
2015-08-10 16:43:25 +00:00
// (NB: in theory, the 1st frame after display sync mode change uses the
// wrong waiting mode)
video: make decoder wrapper a filter Move dec_video.c to filters/f_decoder_wrapper.c. It essentially becomes a source filter. vd.h mostly disappears, because mp_filter takes care of the dataflow, but its remains are in struct mp_decoder_fns. One goal is to simplify dataflow by letting the filter framework handle it (or more accurately, using its conventions). One result is that the decode calls disappear from video.c, because we simply connect the decoder wrapper and the filter chain with mp_pin_connect(). Another goal is to eventually remove the code duplication between the audio and video paths for this. This commit prepares for this by trying to make f_decoder_wrapper.c extensible, so it can be used for audio as well later. Decoder framedropping changes a bit. It doesn't seem to be worse than before, and it's an obscure feature, so I'm content with its new state. Some special code that was apparently meant to avoid dropping too many frames in a row is removed, though. I'm not sure how the source code tree should be organized. For one, video/decode/vd_lavc.c is the only file in its directory, which is a bit annoying.
2018-01-28 09:08:45 +00:00
if (!vo_is_ready_for_frame(vo, mpctx->display_sync_active ? -1 : pts))
video: better pipelining with vf_vapoursynth Most of this is explained in the code comments. This change should improve performance with vapoursynth, especially if concurrent requests are used. This should change nothing if vf_vapoursynth is not in the filter chain, since non-threaded filters obviously can not asynchronously finish filtering of frames.
2015-01-03 02:01:58 +00:00
return;
video: move display and timing to a separate thread The VO is run inside its own thread. It also does most of video timing. The playloop hands the image data and a realtime timestamp to the VO, and the VO does the rest. In particular, this allows the playloop to do other things, instead of blocking for video redraw. But if anything accesses the VO during video timing, it will block. This also fixes vo_sdl.c event handling; but that is only a side-effect, since reimplementing the broken way would require more effort. Also drop --softsleep. In theory, this option helps if the kernel's sleeping mechanism is too inaccurate for video timing. In practice, I haven't ever encountered a situation where it helps, and it just burns CPU cycles. On the other hand it's probably actively harmful, because it prevents the libavcodec decoder threads from doing real work. Side note: Originally, I intended that multiple frames can be queued to the VO. But this is not done, due to problems with OSD and other certain features. OSD in particular is simply designed in a way that it can be neither timed nor copied, so you do have to render it into the video frame before you can draw the next frame. (Subtitles have no such restriction. sd_lavc was even updated to fix this.) It seems the right solution to queuing multiple VO frames is rendering on VO-backed framebuffers, like vo_vdpau.c does. This requires VO driver support, and is out of scope of this commit. As consequence, the VO has a queue size of 1. The existing video queue is just needed to compute frame duration, and will be moved out in the next commit.
2014-08-12 21:02:08 +00:00
video: pass future frames to VO Now the VO can request a number of future frames with the last parameter of vo_set_queue_params(). This will be helpful to fix the interpolation code. Note that the first frame (after playback start or seeking) will usually not have any future frames (to make seeking fast). Near the end of the file, the number of future frames will become lower as well.
2015-07-01 17:22:40 +00:00
assert(mpctx->num_next_frames >= 1);
player: refactor display-sync frame duration calculations Get rid of get_past_frame_durations(), which was a bit too messy. Add a past_frames array, which contains the same information in a more reasonable way. This also means that we can get the exact current and past frame durations without going through awful stuff. (The main problem is that vo_pts_history contains future frames as well, which is needed for frame backstepping etc., but gets in the way here.) Also disable the automatic disabling of display-sync if the frame duration changes, and extend the frame durations allowed for display sync. To allow arbitrarily high durations, vo.c needs to be changed to pause and potentially redraw OSD while showing a single frame, so they're still limited. In an attempt to deal with VFR, calculate the overall speed using the average FPS. The frame scheduling itself does not use the average FPS, but the duration of the current frame. This does not work too well, but provides a good base for further improvements. Where this commit actually helps a lot is dealing with rounded timestamps, e.g. if the container framerate is wrong or unknown, or if the muxer wrote incorrectly rounded timestamps. While the rounding errors apparently can't be get rid of completely in the general case, this is still much better than e.g. disabling display-sync completely just because some frame durations go out of bounds.
2015-11-13 21:45:40 +00:00
if (mpctx->num_past_frames >= MAX_NUM_VO_PTS)
mpctx->num_past_frames--;
MP_TARRAY_INSERT_AT(mpctx, mpctx->past_frames, mpctx->num_past_frames, 0,
(struct frame_info){0});
command: add vsync-ratio property This is very "illustrative", unlike the video-speed-correction property, and thus useful. It can also be used to observe scheduling errors, which are not detected by the core. (These happen due to rounding errors; possibly not evne our fault, but coming from files with rounded timestamps and so on.)
2015-11-13 21:48:32 +00:00
mpctx->past_frames[0] = (struct frame_info){
.pts = mpctx->next_frames[0]->pts,
.num_vsyncs = -1,
};
player: refactor display-sync frame duration calculations Get rid of get_past_frame_durations(), which was a bit too messy. Add a past_frames array, which contains the same information in a more reasonable way. This also means that we can get the exact current and past frame durations without going through awful stuff. (The main problem is that vo_pts_history contains future frames as well, which is needed for frame backstepping etc., but gets in the way here.) Also disable the automatic disabling of display-sync if the frame duration changes, and extend the frame durations allowed for display sync. To allow arbitrarily high durations, vo.c needs to be changed to pause and potentially redraw OSD while showing a single frame, so they're still limited. In an attempt to deal with VFR, calculate the overall speed using the average FPS. The frame scheduling itself does not use the average FPS, but the duration of the current frame. This does not work too well, but provides a good base for further improvements. Where this commit actually helps a lot is dealing with rounded timestamps, e.g. if the container framerate is wrong or unknown, or if the muxer wrote incorrectly rounded timestamps. While the rounding errors apparently can't be get rid of completely in the general case, this is still much better than e.g. disabling display-sync completely just because some frame durations go out of bounds.
2015-11-13 21:45:40 +00:00
calculate_frame_duration(mpctx);
video: limit number of frames sent to VO to the VO requested amount vo_frame can have more than 1 frame - the extra frames are future references, which are sometimes useful for filtering (vo_opengl interpolation). There's no harm in reducing the number of frames sent to the VO requested amount of future frames, so do that. Doesn't actually reduce the number of concurrently in use frames in practice.
2016-07-07 14:10:13 +00:00
int req = vo_get_num_req_frames(mpctx->video_out);
assert(req >= 1 && req <= VO_MAX_REQ_FRAMES);
video: move up vo_frame setup
2015-07-28 21:54:39 +00:00
struct vo_frame dummy = {
.pts = pts,
.duration = -1,
video: disable interpolation during framestepping It just causes annoying artifacts. Interestingly, this means keeping down the frame stepping key (".") will play video with interpolation disabled.
2015-08-25 19:48:11 +00:00
.still = mpctx->step_frames > 0,
vo: pass through framedrop flag differently There is some sort-of awkwardness here, because option access needs to happen in a synchronized manner, and the framedrop flag is not in the VO option struct. Remove the mp_read_option_raw() call and the awkward change notification via VO_EVENT_WIN_STATE from command.c, and pass it through as new vo_frame flag.
2018-03-13 11:54:48 +00:00
.can_drop = opts->frame_dropping & 1,
video: limit number of frames sent to VO to the VO requested amount vo_frame can have more than 1 frame - the extra frames are future references, which are sometimes useful for filtering (vo_opengl interpolation). There's no harm in reducing the number of frames sent to the VO requested amount of future frames, so do that. Doesn't actually reduce the number of concurrently in use frames in practice.
2016-07-07 14:10:13 +00:00
.num_frames = MPMIN(mpctx->num_next_frames, req),
player: add display sync mode If this mode is enabled, the player tries to strictly synchronize video to display refresh. It will adjust playback speed to match the display, so if you play 23.976 fps video on a 24 Hz screen, playback speed is increased by approximately 1/1000. Audio wll be resampled to keep up with playback. This is different from the default sync mode, which will sync video to audio, with the consequence that video might skip or repeat a frame once in a while to make video keep up with audio. This is still unpolished. There are some major problems as well; in particular, mkv VFR files won't work well. The reason is that Matroska is terrible and rounds timestamps to milliseconds. This makes it rather hard to guess the framerate of a section of video that is playing. We could probably fix this by just accepting jittery timestamps (instead of explicitly disabling the sync code in this case), but I'm not ready to accept such a solution yet. Another issue is that we are extremely reliant on OS video and audio APIs working in an expected manner, which of course is not too often the case. Consequently, the new sync mode is a bit fragile.
2015-08-10 16:43:25 +00:00
.num_vsyncs = 1,
video: move up vo_frame setup
2015-07-28 21:54:39 +00:00
};
for (int n = 0; n < dummy.num_frames; n++)
dummy.frames[n] = mpctx->next_frames[n];
struct vo_frame *frame = vo_frame_ref(&dummy);
command: add vsync-ratio property This is very "illustrative", unlike the video-speed-correction property, and thus useful. It can also be used to observe scheduling errors, which are not detected by the core. (These happen due to rounding errors; possibly not evne our fault, but coming from files with rounded timestamps and so on.)
2015-11-13 21:48:32 +00:00
double diff = mpctx->past_frames[0].approx_duration;
player: refactor display-sync frame duration calculations Get rid of get_past_frame_durations(), which was a bit too messy. Add a past_frames array, which contains the same information in a more reasonable way. This also means that we can get the exact current and past frame durations without going through awful stuff. (The main problem is that vo_pts_history contains future frames as well, which is needed for frame backstepping etc., but gets in the way here.) Also disable the automatic disabling of display-sync if the frame duration changes, and extend the frame durations allowed for display sync. To allow arbitrarily high durations, vo.c needs to be changed to pause and potentially redraw OSD while showing a single frame, so they're still limited. In an attempt to deal with VFR, calculate the overall speed using the average FPS. The frame scheduling itself does not use the average FPS, but the duration of the current frame. This does not work too well, but provides a good base for further improvements. Where this commit actually helps a lot is dealing with rounded timestamps, e.g. if the container framerate is wrong or unknown, or if the muxer wrote incorrectly rounded timestamps. While the rounding errors apparently can't be get rid of completely in the general case, this is still much better than e.g. disabling display-sync completely just because some frame durations go out of bounds.
2015-11-13 21:45:40 +00:00
if (opts->untimed || vo->driver->untimed)
diff = -1; // disable frame dropping and aspects of frame timing
video: fix and simplify video format changes and last frame display The previous commit broke these things, and fixing them is separate in this commit in order to reduce the volume of changes. Move the image queue from the VO to the playback core. The image queue is a remnant of the old way how vdpau was implemented, and increasingly became more and more an artifact. In the end, it did only one thing: computing the duration of the current frame. This was done by taking the PTS difference between the current and the future frame. We keep this, but by moving it out of the VO, we don't have to special-case format changes anymore. This simplifies the code a lot. Since we need the queue to compute the duration only, a queue size larger than 2 makes no sense, and we can hardcode that. Also change how the last frame is handled. The last frame is a bit of a problem, because video timing works by showing one frame after another, which makes it a special case. Make the VO provide a function to notify us when the frame is done, instead. The frame duration is used for that. This is not perfect. For example, changing playback speed during the last frame doesn't update the end time. Pausing will not stop the clock that times the last frame. But I don't think this matters for such a corner case.
2014-08-12 21:17:35 +00:00
if (diff >= 0) {
video: move display and timing to a separate thread The VO is run inside its own thread. It also does most of video timing. The playloop hands the image data and a realtime timestamp to the VO, and the VO does the rest. In particular, this allows the playloop to do other things, instead of blocking for video redraw. But if anything accesses the VO during video timing, it will block. This also fixes vo_sdl.c event handling; but that is only a side-effect, since reimplementing the broken way would require more effort. Also drop --softsleep. In theory, this option helps if the kernel's sleeping mechanism is too inaccurate for video timing. In practice, I haven't ever encountered a situation where it helps, and it just burns CPU cycles. On the other hand it's probably actively harmful, because it prevents the libavcodec decoder threads from doing real work. Side note: Originally, I intended that multiple frames can be queued to the VO. But this is not done, due to problems with OSD and other certain features. OSD in particular is simply designed in a way that it can be neither timed nor copied, so you do have to render it into the video frame before you can draw the next frame. (Subtitles have no such restriction. sd_lavc was even updated to fix this.) It seems the right solution to queuing multiple VO frames is rendering on VO-backed framebuffers, like vo_vdpau.c does. This requires VO driver support, and is out of scope of this commit. As consequence, the VO has a queue size of 1. The existing video queue is just needed to compute frame duration, and will be moved out in the next commit.
2014-08-12 21:02:08 +00:00
// expected A/V sync correction is ignored
player: separate controls for user and video controlled speed For video sync, we want separate playback speed controls for user- requested speed and the "correction" speed for video timing. Further, we use this separation to make sure only a resampler is inserted if playback speed is only changed for video sync correction. As of this commit, this is basically inactive code. It's just preparation for the video sync code (the following commit).
2015-08-10 16:40:16 +00:00
diff /= mpctx->video_speed;
video: move display and timing to a separate thread The VO is run inside its own thread. It also does most of video timing. The playloop hands the image data and a realtime timestamp to the VO, and the VO does the rest. In particular, this allows the playloop to do other things, instead of blocking for video redraw. But if anything accesses the VO during video timing, it will block. This also fixes vo_sdl.c event handling; but that is only a side-effect, since reimplementing the broken way would require more effort. Also drop --softsleep. In theory, this option helps if the kernel's sleeping mechanism is too inaccurate for video timing. In practice, I haven't ever encountered a situation where it helps, and it just burns CPU cycles. On the other hand it's probably actively harmful, because it prevents the libavcodec decoder threads from doing real work. Side note: Originally, I intended that multiple frames can be queued to the VO. But this is not done, due to problems with OSD and other certain features. OSD in particular is simply designed in a way that it can be neither timed nor copied, so you do have to render it into the video frame before you can draw the next frame. (Subtitles have no such restriction. sd_lavc was even updated to fix this.) It seems the right solution to queuing multiple VO frames is rendering on VO-backed framebuffers, like vo_vdpau.c does. This requires VO driver support, and is out of scope of this commit. As consequence, the VO has a queue size of 1. The existing video queue is just needed to compute frame duration, and will be moved out in the next commit.
2014-08-12 21:02:08 +00:00
if (mpctx->time_frame < 0)
diff += mpctx->time_frame;
video: move up vo_frame setup
2015-07-28 21:54:39 +00:00
frame->duration = MPCLAMP(diff, 0, 10) * 1e6;
video: move display and timing to a separate thread The VO is run inside its own thread. It also does most of video timing. The playloop hands the image data and a realtime timestamp to the VO, and the VO does the rest. In particular, this allows the playloop to do other things, instead of blocking for video redraw. But if anything accesses the VO during video timing, it will block. This also fixes vo_sdl.c event handling; but that is only a side-effect, since reimplementing the broken way would require more effort. Also drop --softsleep. In theory, this option helps if the kernel's sleeping mechanism is too inaccurate for video timing. In practice, I haven't ever encountered a situation where it helps, and it just burns CPU cycles. On the other hand it's probably actively harmful, because it prevents the libavcodec decoder threads from doing real work. Side note: Originally, I intended that multiple frames can be queued to the VO. But this is not done, due to problems with OSD and other certain features. OSD in particular is simply designed in a way that it can be neither timed nor copied, so you do have to render it into the video frame before you can draw the next frame. (Subtitles have no such restriction. sd_lavc was even updated to fix this.) It seems the right solution to queuing multiple VO frames is rendering on VO-backed framebuffers, like vo_vdpau.c does. This requires VO driver support, and is out of scope of this commit. As consequence, the VO has a queue size of 1. The existing video queue is just needed to compute frame duration, and will be moved out in the next commit.
2014-08-12 21:02:08 +00:00
}
player: move video display code out of the playloop Basically move the code from playloop.c to video.c. The new function write_video() now contains the code that was part of run_playloop(). There are no functional changes, except handling "new_frame_shown" slightly differently. This is done so that we don't need new a new MPContext field or a return value for write_video() to signal this condition. Instead, it's handled indirectly.
2014-07-30 21:24:08 +00:00
video: pass future frames to VO Now the VO can request a number of future frames with the last parameter of vo_set_queue_params(). This will be helpful to fix the interpolation code. Note that the first frame (after playback start or seeking) will usually not have any future frames (to make seeking fast). Near the end of the file, the number of future frames will become lower as well.
2015-07-01 17:22:40 +00:00
mpctx->video_pts = mpctx->next_frames[0]->pts;
player: move video display code out of the playloop Basically move the code from playloop.c to video.c. The new function write_video() now contains the code that was part of run_playloop(). There are no functional changes, except handling "new_frame_shown" slightly differently. This is done so that we don't need new a new MPContext field or a return value for write_video() to signal this condition. Instead, it's handled indirectly.
2014-07-30 21:24:08 +00:00
mpctx->last_vo_pts = mpctx->video_pts;
video: use demuxer-signaled duration for last video frame Helps with gif, probably does unwanted things with other formats. This doesn't handle --end quite correctly, but this could be added later. Fixes #3924.
2016-12-21 17:18:24 +00:00
mpctx->last_frame_duration =
mpctx->next_frames[0]->pkt_duration / mpctx->video_speed;
player: move video display code out of the playloop Basically move the code from playloop.c to video.c. The new function write_video() now contains the code that was part of run_playloop(). There are no functional changes, except handling "new_frame_shown" slightly differently. This is done so that we don't need new a new MPContext field or a return value for write_video() to signal this condition. Instead, it's handled indirectly.
2014-07-30 21:24:08 +00:00
player: refactor display-sync frame duration calculations Get rid of get_past_frame_durations(), which was a bit too messy. Add a past_frames array, which contains the same information in a more reasonable way. This also means that we can get the exact current and past frame durations without going through awful stuff. (The main problem is that vo_pts_history contains future frames as well, which is needed for frame backstepping etc., but gets in the way here.) Also disable the automatic disabling of display-sync if the frame duration changes, and extend the frame durations allowed for display sync. To allow arbitrarily high durations, vo.c needs to be changed to pause and potentially redraw OSD while showing a single frame, so they're still limited. In an attempt to deal with VFR, calculate the overall speed using the average FPS. The frame scheduling itself does not use the average FPS, but the duration of the current frame. This does not work too well, but provides a good base for further improvements. Where this commit actually helps a lot is dealing with rounded timestamps, e.g. if the container framerate is wrong or unknown, or if the muxer wrote incorrectly rounded timestamps. While the rounding errors apparently can't be get rid of completely in the general case, this is still much better than e.g. disabling display-sync completely just because some frame durations go out of bounds.
2015-11-13 21:45:40 +00:00
shift_frames(mpctx);
player: minor refactor for A/V diff computation Calculate the A/V difference directly in the display sync code, instead of the awkward current way, which reuses the fields for audio sync. We still set time_frame, because it makes falling back to audio sync somewhat smoother.
2015-10-28 22:50:54 +00:00
schedule_frame(mpctx, frame);
player: rearrange some A/V-sync related code Move the update_avsync_before_frame() call further down. Moving it closer to where the time_frame value is used (and which the function updates) should make the code more readable. With this change, there's no need anymore to reset the time_frame value on the video reconfig path. Move the update_avsync_after_frame() up. Now no meaningful amount of time passes since the previous get_relative_time() call anymore, and the second one can be removed.
2015-01-30 20:18:30 +00:00
player: rate-limit OSD text update There's no need to update OSD messages and the terminal status if nobody is going to see it. Since the player doesn't block on video display anymore, this update happens to often and probably burns slightly more CPU than necessary. (OSD redrawing is handled separately, so it's just mostly useless text processing and such.) Change it so that it's updated only on every video frame or all 50ms (whatever comes first). For VO OSD, we could in theory try to lock to the OSD redraw heuristic or the display refresh rate, but that's more complicated and doesn't work for the terminal status.
2014-09-25 18:25:24 +00:00
mpctx->osd_force_update = true;
player: move video display code out of the playloop Basically move the code from playloop.c to video.c. The new function write_video() now contains the code that was part of run_playloop(). There are no functional changes, except handling "new_frame_shown" slightly differently. This is done so that we don't need new a new MPContext field or a return value for write_video() to signal this condition. Instead, it's handled indirectly.
2014-07-30 21:24:08 +00:00
update_osd_msg(mpctx);
video: move up vo_frame setup
2015-07-28 21:54:39 +00:00
vo_queue_frame(vo, frame);
vo: change internal API for drawing frames draw_image_timed is renamed to draw_frame. struct frame_timing is renamed to vo_frame. flip_page_timed is merged into draw_frame (the additional parameters are part of struct vo_frame). draw_frame also deprecates VOCTRL_REDRAW_FRAME, and replaces it with a method that works for both VOs which can cache the current frame, and VOs which need to redraw it anyway. This is preparation to making the interpolation and (work in progress) display sync code saner. Lots of other refactoring, and also some simplifications.
2015-07-01 17:24:28 +00:00
video: make decoder wrapper a filter Move dec_video.c to filters/f_decoder_wrapper.c. It essentially becomes a source filter. vd.h mostly disappears, because mp_filter takes care of the dataflow, but its remains are in struct mp_decoder_fns. One goal is to simplify dataflow by letting the filter framework handle it (or more accurately, using its conventions). One result is that the decode calls disappear from video.c, because we simply connect the decoder wrapper and the filter chain with mp_pin_connect(). Another goal is to eventually remove the code duplication between the audio and video paths for this. This commit prepares for this by trying to make f_decoder_wrapper.c extensible, so it can be used for audio as well later. Decoder framedropping changes a bit. It doesn't seem to be worse than before, and it's an obscure feature, so I'm content with its new state. Some special code that was apparently meant to avoid dropping too many frames in a row is removed, though. I'm not sure how the source code tree should be organized. For one, video/decode/vd_lavc.c is the only file in its directory, which is a bit annoying.
2018-01-28 09:08:45 +00:00
check_framedrop(mpctx, vo_c);
video: pass future frames to VO Now the VO can request a number of future frames with the last parameter of vo_set_queue_params(). This will be helpful to fix the interpolation code. Note that the first frame (after playback start or seeking) will usually not have any future frames (to make seeking fast). Near the end of the file, the number of future frames will become lower as well.
2015-07-01 17:22:40 +00:00
// The frames were shifted down; "initialize" the new first entry.
if (mpctx->num_next_frames >= 1)
handle_new_frame(mpctx);
video: refactor aspects of queue and EOF handling Instead of touching the 2-entry queue in mpctx->next_frame directly, move some of it to functions.
2015-03-25 12:33:34 +00:00
player: move video display code out of the playloop Basically move the code from playloop.c to video.c. The new function write_video() now contains the code that was part of run_playloop(). There are no functional changes, except handling "new_frame_shown" slightly differently. This is done so that we don't need new a new MPContext field or a return value for write_video() to signal this condition. Instead, it's handled indirectly.
2014-07-30 21:24:08 +00:00
mpctx->shown_vframes++;
video: move display and timing to a separate thread The VO is run inside its own thread. It also does most of video timing. The playloop hands the image data and a realtime timestamp to the VO, and the VO does the rest. In particular, this allows the playloop to do other things, instead of blocking for video redraw. But if anything accesses the VO during video timing, it will block. This also fixes vo_sdl.c event handling; but that is only a side-effect, since reimplementing the broken way would require more effort. Also drop --softsleep. In theory, this option helps if the kernel's sleeping mechanism is too inaccurate for video timing. In practice, I haven't ever encountered a situation where it helps, and it just burns CPU cycles. On the other hand it's probably actively harmful, because it prevents the libavcodec decoder threads from doing real work. Side note: Originally, I intended that multiple frames can be queued to the VO. But this is not done, due to problems with OSD and other certain features. OSD in particular is simply designed in a way that it can be neither timed nor copied, so you do have to render it into the video frame before you can draw the next frame. (Subtitles have no such restriction. sd_lavc was even updated to fix this.) It seems the right solution to queuing multiple VO frames is rendering on VO-backed framebuffers, like vo_vdpau.c does. This requires VO driver support, and is out of scope of this commit. As consequence, the VO has a queue size of 1. The existing video queue is just needed to compute frame duration, and will be moved out in the next commit.
2014-08-12 21:02:08 +00:00
if (mpctx->video_status < STATUS_PLAYING) {
player: move video display code out of the playloop Basically move the code from playloop.c to video.c. The new function write_video() now contains the code that was part of run_playloop(). There are no functional changes, except handling "new_frame_shown" slightly differently. This is done so that we don't need new a new MPContext field or a return value for write_video() to signal this condition. Instead, it's handled indirectly.
2014-07-30 21:24:08 +00:00
mpctx->video_status = STATUS_READY;
video: move display and timing to a separate thread The VO is run inside its own thread. It also does most of video timing. The playloop hands the image data and a realtime timestamp to the VO, and the VO does the rest. In particular, this allows the playloop to do other things, instead of blocking for video redraw. But if anything accesses the VO during video timing, it will block. This also fixes vo_sdl.c event handling; but that is only a side-effect, since reimplementing the broken way would require more effort. Also drop --softsleep. In theory, this option helps if the kernel's sleeping mechanism is too inaccurate for video timing. In practice, I haven't ever encountered a situation where it helps, and it just burns CPU cycles. On the other hand it's probably actively harmful, because it prevents the libavcodec decoder threads from doing real work. Side note: Originally, I intended that multiple frames can be queued to the VO. But this is not done, due to problems with OSD and other certain features. OSD in particular is simply designed in a way that it can be neither timed nor copied, so you do have to render it into the video frame before you can draw the next frame. (Subtitles have no such restriction. sd_lavc was even updated to fix this.) It seems the right solution to queuing multiple VO frames is rendering on VO-backed framebuffers, like vo_vdpau.c does. This requires VO driver support, and is out of scope of this commit. As consequence, the VO has a queue size of 1. The existing video queue is just needed to compute frame duration, and will be moved out in the next commit.
2014-08-12 21:02:08 +00:00
// After a seek, make sure to wait until the first frame is visible.
video: add option to reduce latency by 1 or 2 frames The playback start logic explicitly waits until the first frame has been displayed. Usually this will introduce a wait of 1 vsync. For normal playback this doesn't matter, but with respect to low latency needs, this only leads to additional data getting queued up in the demuxer or network buffers. Another thing is that the timing logic decodes 1 frame ahead (= 1 frame extra latency) to determine the exact duration of a frame. To be fair, there doesn't really seem to be a hard reason why this is needed. With the current code, enabling the option does lead to A/V desync sometimes (if the demuxer FPS is too inaccurate), and also frame drops at playback start in some situations. But this all seems to be avoidable, if the timing logic were to be rewritten completely, which should probably happen in the future. Thus the new option comes with the warning that it can be removed any time. This is also why the option has "hack" in the name.
2018-03-02 14:37:34 +00:00
if (!opts->video_latency_hacks) {
vo_wait_frame(vo);
MP_VERBOSE(mpctx, "first video frame after restart shown\n");
}
video: move display and timing to a separate thread The VO is run inside its own thread. It also does most of video timing. The playloop hands the image data and a realtime timestamp to the VO, and the VO does the rest. In particular, this allows the playloop to do other things, instead of blocking for video redraw. But if anything accesses the VO during video timing, it will block. This also fixes vo_sdl.c event handling; but that is only a side-effect, since reimplementing the broken way would require more effort. Also drop --softsleep. In theory, this option helps if the kernel's sleeping mechanism is too inaccurate for video timing. In practice, I haven't ever encountered a situation where it helps, and it just burns CPU cycles. On the other hand it's probably actively harmful, because it prevents the libavcodec decoder threads from doing real work. Side note: Originally, I intended that multiple frames can be queued to the VO. But this is not done, due to problems with OSD and other certain features. OSD in particular is simply designed in a way that it can be neither timed nor copied, so you do have to render it into the video frame before you can draw the next frame. (Subtitles have no such restriction. sd_lavc was even updated to fix this.) It seems the right solution to queuing multiple VO frames is rendering on VO-backed framebuffers, like vo_vdpau.c does. This requires VO driver support, and is out of scope of this commit. As consequence, the VO has a queue size of 1. The existing video queue is just needed to compute frame duration, and will be moved out in the next commit.
2014-08-12 21:02:08 +00:00
}
player: move video display code out of the playloop Basically move the code from playloop.c to video.c. The new function write_video() now contains the code that was part of run_playloop(). There are no functional changes, except handling "new_frame_shown" slightly differently. This is done so that we don't need new a new MPContext field or a return value for write_video() to signal this condition. Instead, it's handled indirectly.
2014-07-30 21:24:08 +00:00
mp_notify(mpctx, MPV_EVENT_TICK, NULL);
video: set EOF status as soon as possible See what the added code comment says. Normally when this is needed, it's the cover art case. But this flag is not set when using an external image. This gives weird seek behavior, because the frame will be "normally" displayed for its determined duration, and during normal video playback, the video pts will be used - which is always 0 here. This should happen only if audio is active. Otherwise, we're more or less in image viewer mode, where the image should be displayed for a configured duration.
2019-11-16 23:24:19 +00:00
if (vo_c->filter->got_output_eof && !mpctx->num_next_frames &&
mpctx->ao_chain)
{
MP_VERBOSE(mpctx, "assuming this was the last video frame\n");
// The main point of doing this is to prevent use of this for the
// playback_pts if audio is still running (=> seek behavior).
player: move video display code out of the playloop Basically move the code from playloop.c to video.c. The new function write_video() now contains the code that was part of run_playloop(). There are no functional changes, except handling "new_frame_shown" slightly differently. This is done so that we don't need new a new MPContext field or a return value for write_video() to signal this condition. Instead, it's handled indirectly.
2014-07-30 21:24:08 +00:00
mpctx->video_status = STATUS_EOF;
video: set EOF status as soon as possible See what the added code comment says. Normally when this is needed, it's the cover art case. But this flag is not set when using an external image. This gives weird seek behavior, because the frame will be "normally" displayed for its determined duration, and during normal video playback, the video pts will be used - which is always 0 here. This should happen only if audio is active. Otherwise, we're more or less in image viewer mode, where the image should be displayed for a configured duration.
2019-11-16 23:24:19 +00:00
}
player: some further playloop cleanups Handle --term-playing-msg at a better place. Move MPV_EVENT_TICK hack into a separate function. Also add some words to the client API that you shouldn't use it. (But better leave breaking it for later.) Handle --frames and frame_step differently. Remove the mess from the playloop, and do it after frame display. Give up on the weird semantics for audio-only mode (they didn't make sense anyway), and adjust the manpage accordingly.
2014-08-03 18:25:03 +00:00
player: make repeated hr-seeks past EOF trigger EOF as expected If you have a normal file with audio and video, and keep "spamming" forward hr-seeks, the player just kept showing the last video frame instead of exiting or playing the next file. This started happening since commit 6bcda94cb. Although not a bug per se, it was odd, and very user-noticable. The main problem was that the pending seek command was processed before the EOF was "noticed". Processing the command reset everything, so the player did not terminate playback, but repeated the seek. This commit restores the old behavior. For one, it makes video return the correct status (video.c). The parameter is a bit ugly, but better than duplicating the logic or having another MPContext field. (As a minor detail, setting r=VD_EOF makes sure have_new_frame() returns true, rather than going through another iteration or whatever the hell will happen instead, which would clobber logical_eof.) Another thing is making the seek logic actually wait until the seek outcome has been determined if audio is also active. Audio needs to wait for video in order to get the video seek target position. (Which in turn is because hr-seek still "snaps" to video frames. You can't seek in between two frames, so audio can't just use the seek target, but always has to wait on the timestamp of the video frame. This has other disadvantages and is a misdesign, but not something I'll fix today.) In theory, this might make hr-seeks less responsive, because it needs to fully decode/filter the audio too, but in practice most time is spent on video, which had to be fully decoded before this change. (In general, hr-seek could probably just show a random frame when a queued hr-seek overrides the current hr-seek, which would probably lead to a better user experience, but that's out of scope.) Fixes: #7206
2019-12-14 13:17:16 +00:00
// hr-seek past EOF -> returns last frame, but terminates playback.
if (logical_eof)
mpctx->video_status = STATUS_EOF;
player: some further playloop cleanups Handle --term-playing-msg at a better place. Move MPV_EVENT_TICK hack into a separate function. Also add some words to the client API that you shouldn't use it. (But better leave breaking it for later.) Handle --frames and frame_step differently. Remove the mess from the playloop, and do it after frame display. Give up on the weird semantics for audio-only mode (they didn't make sense anyway), and adjust the manpage accordingly.
2014-08-03 18:25:03 +00:00
if (mpctx->video_status != STATUS_EOF) {
if (mpctx->step_frames > 0) {
mpctx->step_frames--;
player: unmess pause state handling Merge the pause_player() and unpause_player() functions. Make sure the pause events are emitted properly. We can now set the internal pause state based on a predicate, instead of e.g. handle_pause_on_low_cache() making a mess to trigger the internal pause state as wanted. Preparation for some more changes.
2017-04-14 16:22:45 +00:00
if (!mpctx->step_frames)
set_pause_state(mpctx, true);
player: some further playloop cleanups Handle --term-playing-msg at a better place. Move MPV_EVENT_TICK hack into a separate function. Also add some words to the client API that you shouldn't use it. (But better leave breaking it for later.) Handle --frames and frame_step differently. Remove the mess from the playloop, and do it after frame display. Give up on the weird semantics for audio-only mode (they didn't make sense anyway), and adjust the manpage accordingly.
2014-08-03 18:25:03 +00:00
}
player: never overwrite stop_play field This is a real pain: if a quit command is received, it's set to PT_QUIT. And then other code could overwrite it, making it not quit. The annoying bit is that stop_play is written and read in many places. Just not overwriting it unconditionally seems to be the best course of action.
2015-07-08 19:31:31 +00:00
if (mpctx->max_frames == 0 && !mpctx->stop_play)
player: signal EOF when using --frames
2014-10-10 13:14:11 +00:00
mpctx->stop_play = AT_END_OF_FILE;
player: some further playloop cleanups Handle --term-playing-msg at a better place. Move MPV_EVENT_TICK hack into a separate function. Also add some words to the client API that you shouldn't use it. (But better leave breaking it for later.) Handle --frames and frame_step differently. Remove the mess from the playloop, and do it after frame display. Give up on the weird semantics for audio-only mode (they didn't make sense anyway), and adjust the manpage accordingly.
2014-08-03 18:25:03 +00:00
if (mpctx->max_frames > 0)
mpctx->max_frames--;
}
video: fix and simplify video format changes and last frame display The previous commit broke these things, and fixing them is separate in this commit in order to reduce the volume of changes. Move the image queue from the VO to the playback core. The image queue is a remnant of the old way how vdpau was implemented, and increasingly became more and more an artifact. In the end, it did only one thing: computing the duration of the current frame. This was done by taking the PTS difference between the current and the future frame. We keep this, but by moving it out of the VO, we don't have to special-case format changes anymore. This simplifies the code a lot. Since we need the queue to compute the duration only, a queue size larger than 2 makes no sense, and we can hardcode that. Also change how the last frame is handled. The last frame is a bit of a problem, because video timing works by showing one frame after another, which makes it a special case. Make the VO provide a function to notify us when the frame is done, instead. The frame duration is used for that. This is not perfect. For example, changing playback speed during the last frame doesn't update the end time. Pausing will not stop the clock that times the last frame. But I don't think this matters for such a corner case.
2014-08-12 21:17:35 +00:00
player: avoid underrun wakeup loop The VO underrun detection (just a weak heuristic) added in commit f26dfb flagged the underrun state every time it was checked, and since the check happened in every playloop iteration, this caused the playloop to wake up itself on every iteration. It burned an entire core while in this state. Fix this by flagging this condition only once (as it should be), and requiring that a frame is displayed to trigger it again. This makes it work similar as the audio underrun check. The bug report referenced below says --demuxer-thread=no avoided this. This is because the demuxer layer doesn't do proper underrun reporting if the reader thread is disabled. Fixes: #7259
2019-12-16 00:15:43 +00:00
vo_c->underrun_signaled = false;
player: litter code with explicit wakeup calls This does 3 kinds of changes: - change sleeptime=x to mp_set_timeout() - change sleeptime=0 to mp_wakeup_core() calls (to be more explicit) - change commands etc. to call mp_wakeup_core() if they do changes that require the playloop to be rerun This is preparation for the following changes. The goal is to process client API requests without having to rerun the playloop every time. As of this commit, the changes should not change behavior. In particular, the playloop is still implicitly woken up on every command.
2016-09-16 12:24:15 +00:00
mp_wakeup_core(mpctx);
video: fix and simplify video format changes and last frame display The previous commit broke these things, and fixing them is separate in this commit in order to reduce the volume of changes. Move the image queue from the VO to the playback core. The image queue is a remnant of the old way how vdpau was implemented, and increasingly became more and more an artifact. In the end, it did only one thing: computing the duration of the current frame. This was done by taking the PTS difference between the current and the future frame. We keep this, but by moving it out of the VO, we don't have to special-case format changes anymore. This simplifies the code a lot. Since we need the queue to compute the duration only, a queue size larger than 2 makes no sense, and we can hardcode that. Also change how the last frame is handled. The last frame is a bit of a problem, because video timing works by showing one frame after another, which makes it a special case. Make the VO provide a function to notify us when the frame is done, instead. The frame duration is used for that. This is not perfect. For example, changing playback speed during the last frame doesn't update the end time. Pausing will not stop the clock that times the last frame. But I don't think this matters for such a corner case.
2014-08-12 21:17:35 +00:00
return;
error:
MP_FATAL(mpctx, "Could not initialize video chain.\n");
player: remove central uninit_player() function and flags mess Each subsystem (or similar thing) had an INITIALIZED_ flag assigned. The main use of this was that you could pass a bitmask of these flags to uninit_player(). Except in some situations where you wanted to uninitialize nearly everything, this wasn't really useful. Moreover, it was quite annoying that subsystems had most of the code in a specific file, but the uninit code in loadfile.c (because that's where uninit_player() was implemented). Simplify all this. Remove the flags; e.g. instead of testing for the INITIALIZED_AO flag, test whether mpctx->ao is set. Move uninit code to separate functions, e.g. uninit_audio_out().
2014-10-03 17:57:49 +00:00
uninit_video_chain(mpctx);
player: remove some further current_track dependencies Now it's used for initialization only for audio and video.
2016-02-05 22:41:44 +00:00
error_on_track(mpctx, track);
video: fix and simplify video format changes and last frame display The previous commit broke these things, and fixing them is separate in this commit in order to reduce the volume of changes. Move the image queue from the VO to the playback core. The image queue is a remnant of the old way how vdpau was implemented, and increasingly became more and more an artifact. In the end, it did only one thing: computing the duration of the current frame. This was done by taking the PTS difference between the current and the future frame. We keep this, but by moving it out of the VO, we don't have to special-case format changes anymore. This simplifies the code a lot. Since we need the queue to compute the duration only, a queue size larger than 2 makes no sense, and we can hardcode that. Also change how the last frame is handled. The last frame is a bit of a problem, because video timing works by showing one frame after another, which makes it a special case. Make the VO provide a function to notify us when the frame is done, instead. The frame duration is used for that. This is not perfect. For example, changing playback speed during the last frame doesn't update the end time. Pausing will not stop the clock that times the last frame. But I don't think this matters for such a corner case.
2014-08-12 21:17:35 +00:00
handle_force_window(mpctx, true);
player: litter code with explicit wakeup calls This does 3 kinds of changes: - change sleeptime=x to mp_set_timeout() - change sleeptime=0 to mp_wakeup_core() calls (to be more explicit) - change commands etc. to call mp_wakeup_core() if they do changes that require the playloop to be rerun This is preparation for the following changes. The goal is to process client API requests without having to rerun the playloop every time. As of this commit, the changes should not change behavior. In particular, the playloop is still implicitly woken up on every command.
2016-09-16 12:24:15 +00:00
mp_wakeup_core(mpctx);
player: move video display code out of the playloop Basically move the code from playloop.c to video.c. The new function write_video() now contains the code that was part of run_playloop(). There are no functional changes, except handling "new_frame_shown" slightly differently. This is done so that we don't need new a new MPContext field or a return value for write_video() to signal this condition. Instead, it's handled indirectly.
2014-07-30 21:24:08 +00:00
}