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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 22:38:29 +01:00
/*
Update license headers Signed-off-by: wm4 <wm4@nowhere>
2015-04-13 09:36:54 +02:00
* 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 22:38:29 +01:00
*
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 15:53:41 +02:00
* 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 22:38:29 +01:00
*
Update license headers Signed-off-by: wm4 <wm4@nowhere>
2015-04-13 09:36:54 +02:00
* 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 22:38:29 +01:00
* 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 15:53:41 +02:00
* 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 22:38:29 +01:00
*
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 15:53:41 +02:00
* 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 22:38:29 +01:00
*/
#include <stddef.h>
#include <stdbool.h>
#include <inttypes.h>
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 20:40:43 +02:00
#include <limits.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 22:38:29 +01:00
#include <math.h>
#include <assert.h>
mpv_talloc.h: rename from talloc.h This change helps avoiding conflict with talloc.h from libtalloc.
2016-01-11 19:03:40 +01:00
#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 22:38:29 +01:00
Split mpvcore/ into common/, misc/, bstr/
2013-12-17 02:39:45 +01:00
#include "common/msg.h"
encode: don't access ao->pts This field will be moved out of the ao struct. The encoding code was basically using an invalid way of accessing this field. Since the AO will be moved into its own thread too and will do its own buffering, the AO and the playback core might not even agree which sample a PTS timestamp belongs to. Add some extrapolation code to handle this case.
2014-03-07 15:23:03 +01:00
#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 02:02:25 +01:00
#include "options/options.h"
Split mpvcore/ into common/, misc/, bstr/
2013-12-17 02:39:45 +01:00
#include "common/common.h"
audio: add AO deviation logging Pretty dumb (and doesn't handle pausing or other discontinuities), but at least somewhat idiot-proof.
2015-10-08 23:08:20 +02:00
#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 22:38:29 +01:00
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 12:48:30 +02:00
#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 22:38:29 +01:00
#include "audio/out/ao.h"
#include "demux/demux.h"
audio: refactor how data is passed to AO This replaces the two buffers (ao_chain.ao_buffer in the core, and buffer_state.buffers in the AO) with a single queue. Instead of having a byte based buffer, the queue is simply a list of audio frames, as output by the decoder. This should make dataflow simpler and reduce copying. It also attempts to simplify fill_audio_out_buffers(), the function I always hated most, because it's full of subtle and buggy logic. Unfortunately, I got assaulted by corner cases, dumb features (attempt at seamless looping, really?), and other crap, so it got pretty complicated again. fill_audio_out_buffers() is still full of subtle and buggy logic. Maybe it got worse. On the other hand, maybe there really is some progress. Who knows. Originally, the data flow parts was meant to be in f_output_chain, but due to tricky interactions with the playloop code, it's now in the dummy filter in audio.c. At least this improves the way the audio PTS is passed to the encoder in encoding mode. Now it attempts to pass frames directly, along with the pts, which should minimize timestamp problems. But to be honest, encoder mode is one big kludge that shouldn't exist in this way. This commit should be considered pre-alpha code. There are lots of bugs still hiding.
2020-08-28 20:23:54 +02:00
#include "filters/f_async_queue.h"
audio: move to decoder wrapper Use the decoder wrapper that was introduced for video. This removes all code duplication the old audio decoder wrapper had with the video code. (The audio wrapper was copy pasted from the video one over a decade ago, and has been kept in sync ever since by the power of copy&paste. Since the original copy&paste was possibly done by someone who did not answer to the LGPL relicensing, this should also remove all doubts about whether any of this code is left, since we now completely remove any code that could possibly have been based on it.) There is some complication with spdif handling, and a minor behavior change (it will restrict the list of codecs to spdif if spdif is to be used), but there should not be any difference in practice.
2018-01-29 06:18:33 +01:00
#include "filters/f_decoder_wrapper.h"
audio: refactor how data is passed to AO This replaces the two buffers (ao_chain.ao_buffer in the core, and buffer_state.buffers in the AO) with a single queue. Instead of having a byte based buffer, the queue is simply a list of audio frames, as output by the decoder. This should make dataflow simpler and reduce copying. It also attempts to simplify fill_audio_out_buffers(), the function I always hated most, because it's full of subtle and buggy logic. Unfortunately, I got assaulted by corner cases, dumb features (attempt at seamless looping, really?), and other crap, so it got pretty complicated again. fill_audio_out_buffers() is still full of subtle and buggy logic. Maybe it got worse. On the other hand, maybe there really is some progress. Who knows. Originally, the data flow parts was meant to be in f_output_chain, but due to tricky interactions with the playloop code, it's now in the dummy filter in audio.c. At least this improves the way the audio PTS is passed to the encoder in encoding mode. Now it attempts to pass frames directly, along with the pts, which should minimize timestamp problems. But to be honest, encoder mode is one big kludge that shouldn't exist in this way. This commit should be considered pre-alpha code. There are lots of bugs still hiding.
2020-08-28 20:23:54 +02:00
#include "filters/filter_internal.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 22:38:29 +01:00
Rename mp_core.h to core.h Get rid of the mp_ prefix.
2013-12-17 01:08:53 +01:00
#include "core.h"
client API: add events for video and audio reconfig
2014-02-17 02:52:26 +01:00
#include "command.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 22:38:29 +01:00
audio: refactor: work towards unentangling audio decoding and filtering Similar to the video path. dec_audio.c now handles decoding only. It also looks very similar to dec_video.c, and actually contains some of the rewritten code from it. (A further goal might be unifying the decoders, I guess.) High potential for regressions.
2016-01-21 22:10:15 +01:00
enum {
AD_OK = 0,
AD_EOF = -2,
AD_WAIT = -4,
};
audio: refactor how data is passed to AO This replaces the two buffers (ao_chain.ao_buffer in the core, and buffer_state.buffers in the AO) with a single queue. Instead of having a byte based buffer, the queue is simply a list of audio frames, as output by the decoder. This should make dataflow simpler and reduce copying. It also attempts to simplify fill_audio_out_buffers(), the function I always hated most, because it's full of subtle and buggy logic. Unfortunately, I got assaulted by corner cases, dumb features (attempt at seamless looping, really?), and other crap, so it got pretty complicated again. fill_audio_out_buffers() is still full of subtle and buggy logic. Maybe it got worse. On the other hand, maybe there really is some progress. Who knows. Originally, the data flow parts was meant to be in f_output_chain, but due to tricky interactions with the playloop code, it's now in the dummy filter in audio.c. At least this improves the way the audio PTS is passed to the encoder in encoding mode. Now it attempts to pass frames directly, along with the pts, which should minimize timestamp problems. But to be honest, encoder mode is one big kludge that shouldn't exist in this way. This commit should be considered pre-alpha code. There are lots of bugs still hiding.
2020-08-28 20:23:54 +02:00
static void ao_process(struct mp_filter *f);
audio: rewrite filtering glue code Use the new filtering code for audio too.
2018-01-18 14:44:20 +01:00
static void update_speed_filters(struct MPContext *mpctx)
audio: do not require full audio chain reinit for speed changes Actually, it didn't really require that before (most work was avoided), but some bits had to be run anyway. Separate the speed change into a light-weight function, which merely updates already created filters, and a heavy-weight one which messes with filter insertion. This also happens to fix the case where the filters would "forget" the current speed (force resampling, change speed, hit a volume control to force af_volume insertion - it will reset speed and desync). Since we now always run the light-weight function, remove the af_scaletempo verbose message that is printed on speed setting. Other than that, all setters are cheap.
2015-11-04 21:31:53 +01:00
{
audio: rewrite filtering glue code Use the new filtering code for audio too.
2018-01-18 14:44:20 +01:00
struct ao_chain *ao_c = mpctx->ao_chain;
if (!ao_c)
audio: do not require full audio chain reinit for speed changes Actually, it didn't really require that before (most work was avoided), but some bits had to be run anyway. Separate the speed change into a light-weight function, which merely updates already created filters, and a heavy-weight one which messes with filter insertion. This also happens to fix the case where the filters would "forget" the current speed (force resampling, change speed, hit a volume control to force af_volume insertion - it will reset speed and desync). Since we now always run the light-weight function, remove the af_scaletempo verbose message that is printed on speed setting. Other than that, all setters are cheap.
2015-11-04 21:31:53 +01:00
return;
audio: make changing playback speed slightly more robust Always use af_scaletempo if it's inserted, even if the option --audio-pitch-correction=no is set. Make sure all filters are reset on speed change. It's conceivable that dynamic changes to the filter chain at runtime leave filters around without resetting their speed parameters. Also move the code to a separate function.
2015-03-07 20:14:42 +01:00
audio: rewrite filtering glue code Use the new filtering code for audio too.
2018-01-18 14:44:20 +01:00
double speed = mpctx->opts->playback_speed;
double resample = mpctx->speed_factor_a;
audio: redo video-sync=display-adrop This mode drops or repeats audio data to adapt to video speed, instead of resampling it or such. It was added to deal with SPDIF. The implementation was part of fill_audio_out_buffers() - the entire function is something whose complexity exploded in my face, and which I want to clean up, and this is hopefully a first step. Put it in a filter, and mess with the shitty glue code. It's all sort of roundabout and illogical, but that can be rectified later. The important part is that it works much like the resample or scaletempo filters. For PCM audio, this does not work on samples anymore. This makes it much worse. But for PCM you can use saner mechanisms that sound better. Also, something about PTS tracking is wrong. But not wasting more time on this.
2020-05-23 04:04:46 +02:00
double drop = 1.0;
audio: do not require full audio chain reinit for speed changes Actually, it didn't really require that before (most work was avoided), but some bits had to be run anyway. Separate the speed change into a light-weight function, which merely updates already created filters, and a heavy-weight one which messes with filter insertion. This also happens to fix the case where the filters would "forget" the current speed (force resampling, change speed, hit a volume control to force af_volume insertion - it will reset speed and desync). Since we now always run the light-weight function, remove the af_scaletempo verbose message that is printed on speed setting. Other than that, all setters are cheap.
2015-11-04 21:31:53 +01:00
audio: rewrite filtering glue code Use the new filtering code for audio too.
2018-01-18 14:44:20 +01:00
if (!mpctx->opts->pitch_correction) {
resample *= speed;
speed = 1.0;
af_lavrresample: deprecate this filter The future direction might be not having such a user-visible filter at all, similar to how vf_scale went away (or actually, redirects to libavfilter's vf_scale).
2018-01-12 04:16:39 +01:00
}
player: add video-sync=display-tempo So far there was no way to sync video to display and have audio sync to video without changes in pitch. With this option the audio does not get resampled (pitch change) and instead the corrected audio speed is applied to audio filters.
2022-08-03 23:47:07 +02:00
if (mpctx->display_sync_active) {
switch (mpctx->video_out->opts->video_sync) {
case VS_DISP_ADROP:
drop *= speed * resample;
resample = speed = 1.0;
break;
case VS_DISP_TEMPO:
speed = mpctx->audio_speed;
resample = 1.0;
break;
}
audio: redo video-sync=display-adrop This mode drops or repeats audio data to adapt to video speed, instead of resampling it or such. It was added to deal with SPDIF. The implementation was part of fill_audio_out_buffers() - the entire function is something whose complexity exploded in my face, and which I want to clean up, and this is hopefully a first step. Put it in a filter, and mess with the shitty glue code. It's all sort of roundabout and illogical, but that can be rectified later. The important part is that it works much like the resample or scaletempo filters. For PCM audio, this does not work on samples anymore. This makes it much worse. But for PCM you can use saner mechanisms that sound better. Also, something about PTS tracking is wrong. But not wasting more time on this.
2020-05-23 04:04:46 +02:00
}
mp_output_chain_set_audio_speed(ao_c->filter, speed, resample, drop);
audio: make changing playback speed slightly more robust Always use af_scaletempo if it's inserted, even if the option --audio-pitch-correction=no is set. Make sure all filters are reset on speed change. It's conceivable that dynamic changes to the filter chain at runtime leave filters around without resetting their speed parameters. Also move the code to a separate function.
2015-03-07 20:14:42 +01:00
}
audio: add audio softvol processing to AO This does what af_volume used to do. Since we couldn't relicense it, just rewrite it. Since we don't have a new filter mechanism yet, and the libavfilter is too inconvenient, do applying the volume gain in ao.c directly. This is done before handling the audio data to the driver. Since push.c runs a separate thread, and pull.c is called asynchronously from the audio driver's thread, the volume value needs to be synchronized. There's no existing central mutex, so do some shit with atomics. Since there's no atomic_float type predefined (which is at least needed when using the legacy wrapper), do some nonsense about reinterpret casting the float value to an int for the purpose of atomic access. Not sure if using memcpy() is undefined behavior, but for now I don't care. The advantage of not using a filter is lower complexity (no filter auto insertion), and lower latency (gain processing is done after our internal audio buffer of at least 200ms). Disavdantages include inability to use native volume control _before_ other filters with custom filter chains, and the need to add new processing for each new sample type. Since this doesn't reuse any of the old GPL code, nor does indirectly rely on it, volume and replaygain handling now works in LGPL mode. How to process the gain is inspired by libavfilter's af_volume (LGPL). In particular, we use exactly the same rounding, and we quantize processing for integer sample types by 256 steps. Some of libavfilter's copyright may or may not apply, but I think not, and it's the same license anyway.
2017-11-29 21:30:10 +01:00
static int recreate_audio_filters(struct MPContext *mpctx)
{
audio: rewrite filtering glue code Use the new filtering code for audio too.
2018-01-18 14:44:20 +01:00
struct ao_chain *ao_c = mpctx->ao_chain;
assert(ao_c);
audio: add audio softvol processing to AO This does what af_volume used to do. Since we couldn't relicense it, just rewrite it. Since we don't have a new filter mechanism yet, and the libavfilter is too inconvenient, do applying the volume gain in ao.c directly. This is done before handling the audio data to the driver. Since push.c runs a separate thread, and pull.c is called asynchronously from the audio driver's thread, the volume value needs to be synchronized. There's no existing central mutex, so do some shit with atomics. Since there's no atomic_float type predefined (which is at least needed when using the legacy wrapper), do some nonsense about reinterpret casting the float value to an int for the purpose of atomic access. Not sure if using memcpy() is undefined behavior, but for now I don't care. The advantage of not using a filter is lower complexity (no filter auto insertion), and lower latency (gain processing is done after our internal audio buffer of at least 200ms). Disavdantages include inability to use native volume control _before_ other filters with custom filter chains, and the need to add new processing for each new sample type. Since this doesn't reuse any of the old GPL code, nor does indirectly rely on it, volume and replaygain handling now works in LGPL mode. How to process the gain is inspired by libavfilter's af_volume (LGPL). In particular, we use exactly the same rounding, and we quantize processing for integer sample types by 256 steps. Some of libavfilter's copyright may or may not apply, but I think not, and it's the same license anyway.
2017-11-29 21:30:10 +01:00
audio: rewrite filtering glue code Use the new filtering code for audio too.
2018-01-18 14:44:20 +01:00
if (!mp_output_chain_update_filters(ao_c->filter, mpctx->opts->af_settings))
audio: add audio softvol processing to AO This does what af_volume used to do. Since we couldn't relicense it, just rewrite it. Since we don't have a new filter mechanism yet, and the libavfilter is too inconvenient, do applying the volume gain in ao.c directly. This is done before handling the audio data to the driver. Since push.c runs a separate thread, and pull.c is called asynchronously from the audio driver's thread, the volume value needs to be synchronized. There's no existing central mutex, so do some shit with atomics. Since there's no atomic_float type predefined (which is at least needed when using the legacy wrapper), do some nonsense about reinterpret casting the float value to an int for the purpose of atomic access. Not sure if using memcpy() is undefined behavior, but for now I don't care. The advantage of not using a filter is lower complexity (no filter auto insertion), and lower latency (gain processing is done after our internal audio buffer of at least 200ms). Disavdantages include inability to use native volume control _before_ other filters with custom filter chains, and the need to add new processing for each new sample type. Since this doesn't reuse any of the old GPL code, nor does indirectly rely on it, volume and replaygain handling now works in LGPL mode. How to process the gain is inspired by libavfilter's af_volume (LGPL). In particular, we use exactly the same rounding, and we quantize processing for integer sample types by 256 steps. Some of libavfilter's copyright may or may not apply, but I think not, and it's the same license anyway.
2017-11-29 21:30:10 +01:00
goto fail;
audio: rewrite filtering glue code Use the new filtering code for audio too.
2018-01-18 14:44:20 +01:00
update_speed_filters(mpctx);
audio: add audio softvol processing to AO This does what af_volume used to do. Since we couldn't relicense it, just rewrite it. Since we don't have a new filter mechanism yet, and the libavfilter is too inconvenient, do applying the volume gain in ao.c directly. This is done before handling the audio data to the driver. Since push.c runs a separate thread, and pull.c is called asynchronously from the audio driver's thread, the volume value needs to be synchronized. There's no existing central mutex, so do some shit with atomics. Since there's no atomic_float type predefined (which is at least needed when using the legacy wrapper), do some nonsense about reinterpret casting the float value to an int for the purpose of atomic access. Not sure if using memcpy() is undefined behavior, but for now I don't care. The advantage of not using a filter is lower complexity (no filter auto insertion), and lower latency (gain processing is done after our internal audio buffer of at least 200ms). Disavdantages include inability to use native volume control _before_ other filters with custom filter chains, and the need to add new processing for each new sample type. Since this doesn't reuse any of the old GPL code, nor does indirectly rely on it, volume and replaygain handling now works in LGPL mode. How to process the gain is inspired by libavfilter's af_volume (LGPL). In particular, we use exactly the same rounding, and we quantize processing for integer sample types by 256 steps. Some of libavfilter's copyright may or may not apply, but I think not, and it's the same license anyway.
2017-11-29 21:30:10 +01:00
mp_notify(mpctx, MPV_EVENT_AUDIO_RECONFIG, NULL);
return 0;
fail:
audio: rewrite filtering glue code Use the new filtering code for audio too.
2018-01-18 14:44:20 +01:00
MP_ERR(mpctx, "Audio filter initialized failed!\n");
audio: add audio softvol processing to AO This does what af_volume used to do. Since we couldn't relicense it, just rewrite it. Since we don't have a new filter mechanism yet, and the libavfilter is too inconvenient, do applying the volume gain in ao.c directly. This is done before handling the audio data to the driver. Since push.c runs a separate thread, and pull.c is called asynchronously from the audio driver's thread, the volume value needs to be synchronized. There's no existing central mutex, so do some shit with atomics. Since there's no atomic_float type predefined (which is at least needed when using the legacy wrapper), do some nonsense about reinterpret casting the float value to an int for the purpose of atomic access. Not sure if using memcpy() is undefined behavior, but for now I don't care. The advantage of not using a filter is lower complexity (no filter auto insertion), and lower latency (gain processing is done after our internal audio buffer of at least 200ms). Disavdantages include inability to use native volume control _before_ other filters with custom filter chains, and the need to add new processing for each new sample type. Since this doesn't reuse any of the old GPL code, nor does indirectly rely on it, volume and replaygain handling now works in LGPL mode. How to process the gain is inspired by libavfilter's af_volume (LGPL). In particular, we use exactly the same rounding, and we quantize processing for integer sample types by 256 steps. Some of libavfilter's copyright may or may not apply, but I think not, and it's the same license anyway.
2017-11-29 21:30:10 +01:00
return -1;
}
int reinit_audio_filters(struct MPContext *mpctx)
{
struct ao_chain *ao_c = mpctx->ao_chain;
if (!ao_c)
return 0;
audio: rewrite filtering glue code Use the new filtering code for audio too.
2018-01-18 14:44:20 +01:00
double delay = mp_output_get_measured_total_delay(ao_c->filter);
audio: add audio softvol processing to AO This does what af_volume used to do. Since we couldn't relicense it, just rewrite it. Since we don't have a new filter mechanism yet, and the libavfilter is too inconvenient, do applying the volume gain in ao.c directly. This is done before handling the audio data to the driver. Since push.c runs a separate thread, and pull.c is called asynchronously from the audio driver's thread, the volume value needs to be synchronized. There's no existing central mutex, so do some shit with atomics. Since there's no atomic_float type predefined (which is at least needed when using the legacy wrapper), do some nonsense about reinterpret casting the float value to an int for the purpose of atomic access. Not sure if using memcpy() is undefined behavior, but for now I don't care. The advantage of not using a filter is lower complexity (no filter auto insertion), and lower latency (gain processing is done after our internal audio buffer of at least 200ms). Disavdantages include inability to use native volume control _before_ other filters with custom filter chains, and the need to add new processing for each new sample type. Since this doesn't reuse any of the old GPL code, nor does indirectly rely on it, volume and replaygain handling now works in LGPL mode. How to process the gain is inspired by libavfilter's af_volume (LGPL). In particular, we use exactly the same rounding, and we quantize processing for integer sample types by 256 steps. Some of libavfilter's copyright may or may not apply, but I think not, and it's the same license anyway.
2017-11-29 21:30:10 +01:00
if (recreate_audio_filters(mpctx) < 0)
return -1;
audio: rewrite filtering glue code Use the new filtering code for audio too.
2018-01-18 14:44:20 +01:00
double ndelay = mp_output_get_measured_total_delay(ao_c->filter);
audio: add audio softvol processing to AO This does what af_volume used to do. Since we couldn't relicense it, just rewrite it. Since we don't have a new filter mechanism yet, and the libavfilter is too inconvenient, do applying the volume gain in ao.c directly. This is done before handling the audio data to the driver. Since push.c runs a separate thread, and pull.c is called asynchronously from the audio driver's thread, the volume value needs to be synchronized. There's no existing central mutex, so do some shit with atomics. Since there's no atomic_float type predefined (which is at least needed when using the legacy wrapper), do some nonsense about reinterpret casting the float value to an int for the purpose of atomic access. Not sure if using memcpy() is undefined behavior, but for now I don't care. The advantage of not using a filter is lower complexity (no filter auto insertion), and lower latency (gain processing is done after our internal audio buffer of at least 200ms). Disavdantages include inability to use native volume control _before_ other filters with custom filter chains, and the need to add new processing for each new sample type. Since this doesn't reuse any of the old GPL code, nor does indirectly rely on it, volume and replaygain handling now works in LGPL mode. How to process the gain is inspired by libavfilter's af_volume (LGPL). In particular, we use exactly the same rounding, and we quantize processing for integer sample types by 256 steps. Some of libavfilter's copyright may or may not apply, but I think not, and it's the same license anyway.
2017-11-29 21:30:10 +01:00
// Only force refresh if the amount of dropped buffered data is going to
// cause "issues" for the A/V sync logic.
audio: rewrite filtering glue code Use the new filtering code for audio too.
2018-01-18 14:44:20 +01:00
if (mpctx->audio_status == STATUS_PLAYING && delay - ndelay >= 0.2)
audio: add audio softvol processing to AO This does what af_volume used to do. Since we couldn't relicense it, just rewrite it. Since we don't have a new filter mechanism yet, and the libavfilter is too inconvenient, do applying the volume gain in ao.c directly. This is done before handling the audio data to the driver. Since push.c runs a separate thread, and pull.c is called asynchronously from the audio driver's thread, the volume value needs to be synchronized. There's no existing central mutex, so do some shit with atomics. Since there's no atomic_float type predefined (which is at least needed when using the legacy wrapper), do some nonsense about reinterpret casting the float value to an int for the purpose of atomic access. Not sure if using memcpy() is undefined behavior, but for now I don't care. The advantage of not using a filter is lower complexity (no filter auto insertion), and lower latency (gain processing is done after our internal audio buffer of at least 200ms). Disavdantages include inability to use native volume control _before_ other filters with custom filter chains, and the need to add new processing for each new sample type. Since this doesn't reuse any of the old GPL code, nor does indirectly rely on it, volume and replaygain handling now works in LGPL mode. How to process the gain is inspired by libavfilter's af_volume (LGPL). In particular, we use exactly the same rounding, and we quantize processing for integer sample types by 256 steps. Some of libavfilter's copyright may or may not apply, but I think not, and it's the same license anyway.
2017-11-29 21:30:10 +01:00
issue_refresh_seek(mpctx, MPSEEK_EXACT);
return 1;
}
audio: replace from_dB function The author of the old code disagreed with LGPL. The new code is a clean room reimplementation by Niklas Haas. It lacks the clamping, but we decided it doesn't matter. Untested, bugs can be fixed later anyway.
2017-05-05 01:21:57 +02:00
static double db_gain(double db)
audio: move replaygain control to top-level options af_volume is deprecated, and so are its replaygain sub-options. To make it possible to use replaygain without deprecated options (and of course to make it available at all after af_volume is dropped), reintroduce them as top-level options. This also means that they are easily changeable at runtime by using them as properties. Change the "volume" property to use the new update mechanism as well. We don't actually bother sharing the implementation between new and deprecated mechanisms, as the deprecated one will simply be deleted. For the from_dB() functions, we mention anders' copyright, although I'm not sure if a mere formula is copyrightable. This will have to be determined later. This whole change is mostly untested. Our distributed human CI will take care of it.
2017-04-26 21:45:50 +02:00
{
audio: replace from_dB function The author of the old code disagreed with LGPL. The new code is a clean room reimplementation by Niklas Haas. It lacks the clamping, but we decided it doesn't matter. Untested, bugs can be fixed later anyway.
2017-05-05 01:21:57 +02:00
return pow(10.0, db/20.0);
audio: move replaygain control to top-level options af_volume is deprecated, and so are its replaygain sub-options. To make it possible to use replaygain without deprecated options (and of course to make it available at all after af_volume is dropped), reintroduce them as top-level options. This also means that they are easily changeable at runtime by using them as properties. Change the "volume" property to use the new update mechanism as well. We don't actually bother sharing the implementation between new and deprecated mechanisms, as the deprecated one will simply be deleted. For the from_dB() functions, we mention anders' copyright, although I'm not sure if a mere formula is copyrightable. This will have to be determined later. This whole change is mostly untested. Our distributed human CI will take care of it.
2017-04-26 21:45:50 +02:00
}
static float compute_replaygain(struct MPContext *mpctx)
{
struct MPOpts *opts = mpctx->opts;
float rgain = 1.0;
audio: add audio softvol processing to AO This does what af_volume used to do. Since we couldn't relicense it, just rewrite it. Since we don't have a new filter mechanism yet, and the libavfilter is too inconvenient, do applying the volume gain in ao.c directly. This is done before handling the audio data to the driver. Since push.c runs a separate thread, and pull.c is called asynchronously from the audio driver's thread, the volume value needs to be synchronized. There's no existing central mutex, so do some shit with atomics. Since there's no atomic_float type predefined (which is at least needed when using the legacy wrapper), do some nonsense about reinterpret casting the float value to an int for the purpose of atomic access. Not sure if using memcpy() is undefined behavior, but for now I don't care. The advantage of not using a filter is lower complexity (no filter auto insertion), and lower latency (gain processing is done after our internal audio buffer of at least 200ms). Disavdantages include inability to use native volume control _before_ other filters with custom filter chains, and the need to add new processing for each new sample type. Since this doesn't reuse any of the old GPL code, nor does indirectly rely on it, volume and replaygain handling now works in LGPL mode. How to process the gain is inspired by libavfilter's af_volume (LGPL). In particular, we use exactly the same rounding, and we quantize processing for integer sample types by 256 steps. Some of libavfilter's copyright may or may not apply, but I think not, and it's the same license anyway.
2017-11-29 21:30:10 +01:00
struct replaygain_data *rg = NULL;
struct track *track = mpctx->current_track[0][STREAM_AUDIO];
if (track)
rg = track->stream->codec->replaygain_data;
audio: merge --replaygain-track and --replaygain-album into one option This is probably better than separate options. For example, the user does not have to guess which one is applied if both options are enabled.
2017-04-27 00:21:17 +02:00
if (opts->rgain_mode && rg) {
audio: move replaygain control to top-level options af_volume is deprecated, and so are its replaygain sub-options. To make it possible to use replaygain without deprecated options (and of course to make it available at all after af_volume is dropped), reintroduce them as top-level options. This also means that they are easily changeable at runtime by using them as properties. Change the "volume" property to use the new update mechanism as well. We don't actually bother sharing the implementation between new and deprecated mechanisms, as the deprecated one will simply be deleted. For the from_dB() functions, we mention anders' copyright, although I'm not sure if a mere formula is copyrightable. This will have to be determined later. This whole change is mostly untested. Our distributed human CI will take care of it.
2017-04-26 21:45:50 +02:00
MP_VERBOSE(mpctx, "Replaygain: Track=%f/%f Album=%f/%f\n",
rg->track_gain, rg->track_peak,
rg->album_gain, rg->album_peak);
float gain, peak;
audio: merge --replaygain-track and --replaygain-album into one option This is probably better than separate options. For example, the user does not have to guess which one is applied if both options are enabled.
2017-04-27 00:21:17 +02:00
if (opts->rgain_mode == 1) {
audio: move replaygain control to top-level options af_volume is deprecated, and so are its replaygain sub-options. To make it possible to use replaygain without deprecated options (and of course to make it available at all after af_volume is dropped), reintroduce them as top-level options. This also means that they are easily changeable at runtime by using them as properties. Change the "volume" property to use the new update mechanism as well. We don't actually bother sharing the implementation between new and deprecated mechanisms, as the deprecated one will simply be deleted. For the from_dB() functions, we mention anders' copyright, although I'm not sure if a mere formula is copyrightable. This will have to be determined later. This whole change is mostly untested. Our distributed human CI will take care of it.
2017-04-26 21:45:50 +02:00
gain = rg->track_gain;
peak = rg->track_peak;
} else {
gain = rg->album_gain;
peak = rg->album_peak;
}
gain += opts->rgain_preamp;
audio: replace from_dB function The author of the old code disagreed with LGPL. The new code is a clean room reimplementation by Niklas Haas. It lacks the clamping, but we decided it doesn't matter. Untested, bugs can be fixed later anyway.
2017-05-05 01:21:57 +02:00
rgain = db_gain(gain);
audio: move replaygain control to top-level options af_volume is deprecated, and so are its replaygain sub-options. To make it possible to use replaygain without deprecated options (and of course to make it available at all after af_volume is dropped), reintroduce them as top-level options. This also means that they are easily changeable at runtime by using them as properties. Change the "volume" property to use the new update mechanism as well. We don't actually bother sharing the implementation between new and deprecated mechanisms, as the deprecated one will simply be deleted. For the from_dB() functions, we mention anders' copyright, although I'm not sure if a mere formula is copyrightable. This will have to be determined later. This whole change is mostly untested. Our distributed human CI will take care of it.
2017-04-26 21:45:50 +02:00
MP_VERBOSE(mpctx, "Applying replay-gain: %f\n", rgain);
if (!opts->rgain_clip) { // clipping prevention
rgain = MPMIN(rgain, 1.0 / peak);
MP_VERBOSE(mpctx, "...with clipping prevention: %f\n", rgain);
}
} else if (opts->rgain_fallback) {
audio: replace from_dB function The author of the old code disagreed with LGPL. The new code is a clean room reimplementation by Niklas Haas. It lacks the clamping, but we decided it doesn't matter. Untested, bugs can be fixed later anyway.
2017-05-05 01:21:57 +02:00
rgain = db_gain(opts->rgain_fallback);
audio: move replaygain control to top-level options af_volume is deprecated, and so are its replaygain sub-options. To make it possible to use replaygain without deprecated options (and of course to make it available at all after af_volume is dropped), reintroduce them as top-level options. This also means that they are easily changeable at runtime by using them as properties. Change the "volume" property to use the new update mechanism as well. We don't actually bother sharing the implementation between new and deprecated mechanisms, as the deprecated one will simply be deleted. For the from_dB() functions, we mention anders' copyright, although I'm not sure if a mere formula is copyrightable. This will have to be determined later. This whole change is mostly untested. Our distributed human CI will take care of it.
2017-04-26 21:45:50 +02:00
MP_VERBOSE(mpctx, "Applying fallback gain: %f\n", rgain);
}
return rgain;
}
audio: refactor mixer code and delete mixer.c mixer.c didn't really deserve to be separate anymore, as half of its contents were unnecessary glue code after recent changes. It also created a weird split between audio.c and af.c due to the fact that mixer.c could insert audio filters. With the code being in audio.c directly, together with other code that unserts filters during runtime, it will be possible to cleanup this code a bit and make it work like the video filter code. As part of this change, make the balance code work like the volume code, and add an option to back the current balance value. Also, since the balance semantics are unexpected for most users (panning between the audio channels, instead of just changing the relative volume), and there are some other volumes, formally deprecate both the old property and the new option.
2016-07-17 19:21:28 +02:00
// Called when opts->softvol_volume or opts->softvol_mute were changed.
void audio_update_volume(struct MPContext *mpctx)
{
struct MPOpts *opts = mpctx->opts;
struct ao_chain *ao_c = mpctx->ao_chain;
audio: add audio softvol processing to AO This does what af_volume used to do. Since we couldn't relicense it, just rewrite it. Since we don't have a new filter mechanism yet, and the libavfilter is too inconvenient, do applying the volume gain in ao.c directly. This is done before handling the audio data to the driver. Since push.c runs a separate thread, and pull.c is called asynchronously from the audio driver's thread, the volume value needs to be synchronized. There's no existing central mutex, so do some shit with atomics. Since there's no atomic_float type predefined (which is at least needed when using the legacy wrapper), do some nonsense about reinterpret casting the float value to an int for the purpose of atomic access. Not sure if using memcpy() is undefined behavior, but for now I don't care. The advantage of not using a filter is lower complexity (no filter auto insertion), and lower latency (gain processing is done after our internal audio buffer of at least 200ms). Disavdantages include inability to use native volume control _before_ other filters with custom filter chains, and the need to add new processing for each new sample type. Since this doesn't reuse any of the old GPL code, nor does indirectly rely on it, volume and replaygain handling now works in LGPL mode. How to process the gain is inspired by libavfilter's af_volume (LGPL). In particular, we use exactly the same rounding, and we quantize processing for integer sample types by 256 steps. Some of libavfilter's copyright may or may not apply, but I think not, and it's the same license anyway.
2017-11-29 21:30:10 +01:00
if (!ao_c || !ao_c->ao)
audio: refactor mixer code and delete mixer.c mixer.c didn't really deserve to be separate anymore, as half of its contents were unnecessary glue code after recent changes. It also created a weird split between audio.c and af.c due to the fact that mixer.c could insert audio filters. With the code being in audio.c directly, together with other code that unserts filters during runtime, it will be possible to cleanup this code a bit and make it work like the video filter code. As part of this change, make the balance code work like the volume code, and add an option to back the current balance value. Also, since the balance semantics are unexpected for most users (panning between the audio channels, instead of just changing the relative volume), and there are some other volumes, formally deprecate both the old property and the new option.
2016-07-17 19:21:28 +02:00
return;
float gain = MPMAX(opts->softvol_volume / 100.0, 0);
audio: fix replaygain volume scale The new replaygain code accidentally applied the linear gain as cubic volume level. Fix this by moving the computation of the volume scale out of the af_volume filter. (Still haven't verified whether the replaygain code works correctly.)
2017-04-27 00:15:22 +02:00
gain = pow(gain, 3);
audio: move replaygain control to top-level options af_volume is deprecated, and so are its replaygain sub-options. To make it possible to use replaygain without deprecated options (and of course to make it available at all after af_volume is dropped), reintroduce them as top-level options. This also means that they are easily changeable at runtime by using them as properties. Change the "volume" property to use the new update mechanism as well. We don't actually bother sharing the implementation between new and deprecated mechanisms, as the deprecated one will simply be deleted. For the from_dB() functions, we mention anders' copyright, although I'm not sure if a mere formula is copyrightable. This will have to be determined later. This whole change is mostly untested. Our distributed human CI will take care of it.
2017-04-26 21:45:50 +02:00
gain *= compute_replaygain(mpctx);
audio: refactor mixer code and delete mixer.c mixer.c didn't really deserve to be separate anymore, as half of its contents were unnecessary glue code after recent changes. It also created a weird split between audio.c and af.c due to the fact that mixer.c could insert audio filters. With the code being in audio.c directly, together with other code that unserts filters during runtime, it will be possible to cleanup this code a bit and make it work like the video filter code. As part of this change, make the balance code work like the volume code, and add an option to back the current balance value. Also, since the balance semantics are unexpected for most users (panning between the audio channels, instead of just changing the relative volume), and there are some other volumes, formally deprecate both the old property and the new option.
2016-07-17 19:21:28 +02:00
if (opts->softvol_mute == 1)
gain = 0.0;
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 22:38:29 +01:00
audio: add audio softvol processing to AO This does what af_volume used to do. Since we couldn't relicense it, just rewrite it. Since we don't have a new filter mechanism yet, and the libavfilter is too inconvenient, do applying the volume gain in ao.c directly. This is done before handling the audio data to the driver. Since push.c runs a separate thread, and pull.c is called asynchronously from the audio driver's thread, the volume value needs to be synchronized. There's no existing central mutex, so do some shit with atomics. Since there's no atomic_float type predefined (which is at least needed when using the legacy wrapper), do some nonsense about reinterpret casting the float value to an int for the purpose of atomic access. Not sure if using memcpy() is undefined behavior, but for now I don't care. The advantage of not using a filter is lower complexity (no filter auto insertion), and lower latency (gain processing is done after our internal audio buffer of at least 200ms). Disavdantages include inability to use native volume control _before_ other filters with custom filter chains, and the need to add new processing for each new sample type. Since this doesn't reuse any of the old GPL code, nor does indirectly rely on it, volume and replaygain handling now works in LGPL mode. How to process the gain is inspired by libavfilter's af_volume (LGPL). In particular, we use exactly the same rounding, and we quantize processing for integer sample types by 256 steps. Some of libavfilter's copyright may or may not apply, but I think not, and it's the same license anyway.
2017-11-29 21:30:10 +01:00
ao_set_gain(ao_c->ao, gain);
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 22:38:29 +01:00
}
audio: do not require full audio chain reinit for speed changes Actually, it didn't really require that before (most work was avoided), but some bits had to be run anyway. Separate the speed change into a light-weight function, which merely updates already created filters, and a heavy-weight one which messes with filter insertion. This also happens to fix the case where the filters would "forget" the current speed (force resampling, change speed, hit a volume control to force af_volume insertion - it will reset speed and desync). Since we now always run the light-weight function, remove the af_scaletempo verbose message that is printed on speed setting. Other than that, all setters are cheap.
2015-11-04 21:31:53 +01:00
// Call this if opts->playback_speed or mpctx->speed_factor_* change.
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 18:40:16 +02:00
void update_playback_speed(struct MPContext *mpctx)
command: move setting playback speed to a separate function
2014-10-02 02:49:05 +02:00
{
audio: do not require full audio chain reinit for speed changes Actually, it didn't really require that before (most work was avoided), but some bits had to be run anyway. Separate the speed change into a light-weight function, which merely updates already created filters, and a heavy-weight one which messes with filter insertion. This also happens to fix the case where the filters would "forget" the current speed (force resampling, change speed, hit a volume control to force af_volume insertion - it will reset speed and desync). Since we now always run the light-weight function, remove the af_scaletempo verbose message that is printed on speed setting. Other than that, all setters are cheap.
2015-11-04 21:31:53 +01:00
mpctx->audio_speed = mpctx->opts->playback_speed * mpctx->speed_factor_a;
mpctx->video_speed = mpctx->opts->playback_speed * mpctx->speed_factor_v;
command: move setting playback speed to a separate function
2014-10-02 02:49:05 +02:00
audio: rewrite filtering glue code Use the new filtering code for audio too.
2018-01-18 14:44:20 +01:00
update_speed_filters(mpctx);
command: move setting playback speed to a separate function
2014-10-02 02:49:05 +02:00
}
audio: add AOCONTROL_UPDATE_MEDIA_ROLE This is used to notify an AO about the type of media that is being played. Either a movie or music.
2022-09-04 11:49:23 +02:00
static bool has_video_track(struct MPContext *mpctx)
{
if (mpctx->vo_chain && mpctx->vo_chain->is_coverart)
return false;
for (int n = 0; n < mpctx->num_tracks; n++) {
struct track *track = mpctx->tracks[n];
if (track->type == STREAM_VIDEO && !track->attached_picture && !track->image)
return true;
}
return false;
}
void audio_update_media_role(struct MPContext *mpctx)
{
if (!mpctx->ao)
return;
enum aocontrol_media_role role = has_video_track(mpctx) ?
AOCONTROL_MEDIA_ROLE_MOVIE : AOCONTROL_MEDIA_ROLE_MUSIC;
ao_control(mpctx->ao, AOCONTROL_UPDATE_MEDIA_ROLE, &role);
}
player: refactor: eliminate MPContext.d_audio
2016-01-21 22:24:20 +01:00
static void ao_chain_reset_state(struct ao_chain *ao_c)
{
audio: rewrite filtering glue code Use the new filtering code for audio too.
2018-01-18 14:44:20 +01:00
ao_c->last_out_pts = MP_NOPTS_VALUE;
audio: move to decoder wrapper Use the decoder wrapper that was introduced for video. This removes all code duplication the old audio decoder wrapper had with the video code. (The audio wrapper was copy pasted from the video one over a decade ago, and has been kept in sync ever since by the power of copy&paste. Since the original copy&paste was possibly done by someone who did not answer to the LGPL relicensing, this should also remove all doubts about whether any of this code is left, since we now completely remove any code that could possibly have been based on it.) There is some complication with spdif handling, and a minor behavior change (it will restrict the list of codecs to spdif if spdif is to be used), but there should not be any difference in practice.
2018-01-29 06:18:33 +01:00
ao_c->out_eof = false;
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 19:34:04 +02:00
ao_c->underrun = false;
audio: refactor how data is passed to AO This replaces the two buffers (ao_chain.ao_buffer in the core, and buffer_state.buffers in the AO) with a single queue. Instead of having a byte based buffer, the queue is simply a list of audio frames, as output by the decoder. This should make dataflow simpler and reduce copying. It also attempts to simplify fill_audio_out_buffers(), the function I always hated most, because it's full of subtle and buggy logic. Unfortunately, I got assaulted by corner cases, dumb features (attempt at seamless looping, really?), and other crap, so it got pretty complicated again. fill_audio_out_buffers() is still full of subtle and buggy logic. Maybe it got worse. On the other hand, maybe there really is some progress. Who knows. Originally, the data flow parts was meant to be in f_output_chain, but due to tricky interactions with the playloop code, it's now in the dummy filter in audio.c. At least this improves the way the audio PTS is passed to the encoder in encoding mode. Now it attempts to pass frames directly, along with the pts, which should minimize timestamp problems. But to be honest, encoder mode is one big kludge that shouldn't exist in this way. This commit should be considered pre-alpha code. There are lots of bugs still hiding.
2020-08-28 20:23:54 +02:00
ao_c->start_pts_known = false;
ao_c->start_pts = MP_NOPTS_VALUE;
ao_c->untimed_throttle = false;
ao_c->underrun = false;
player: refactor: eliminate MPContext.d_audio
2016-01-21 22:24:20 +01:00
}
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 23:01:55 +02:00
void reset_audio_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 02:05:52 +02:00
if (mpctx->ao_chain) {
player: refactor: eliminate MPContext.d_audio
2016-01-21 22:24:20 +01:00
ao_chain_reset_state(mpctx->ao_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 02:05:52 +02:00
struct track *t = mpctx->ao_chain->track;
if (t && t->dec)
f_decoder_wrapper: replace most public fields with setters/getters I may (optionally) move decoding to a separate thread in a future change. It's a bit attractive to move the entire decoder wrapper to there, so if the demuxer has a new packet, it doesn't have to wake up the main thread, and can directly wake up the decoder. (Although that's bullshit, since there's a queue in between, and libavcodec's multi-threaded decoding plays cross-threads ping pong with packets anyway. On the other hand, the main thread would still have to shuffle the packets around, so whatever, just seems like better design.) As preparation, there shouldn't be any mutable state exposed by the wrapper. But there's still a large number of corner-caseish crap, so just use setters/getters for them. This recorder thing will inherently not work, so it'll have to be disabled if threads are used. This is a bit painful, but probably still the right thing. Like speculatively pulling teeth.
2020-02-28 21:17:55 +01:00
mp_decoder_wrapper_set_play_dir(t->dec, mpctx->play_dir);
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 02:05:52 +02:00
}
audio: refactor: work towards unentangling audio decoding and filtering Similar to the video path. dec_audio.c now handles decoding only. It also looks very similar to dec_video.c, and actually contains some of the rewritten code from it. (A further goal might be unifying the decoders, I guess.) High potential for regressions.
2016-01-21 22:10:15 +01:00
mpctx->audio_status = mpctx->ao_chain ? STATUS_SYNCING : STATUS_EOF;
player: minor simplification in A/V-sync related code Just minor things.
2015-01-30 23:49:30 +01:00
mpctx->delay = 0;
audio: reduce excessive logging of delayed audio start Since this is a messy and fragile mechanism, I want it logged (even if it's somewhat in conflict with the verbose logging policy). On the other hand, it's unconditionally logged on every playloop iteration. So add some nonsense to log it only on progress.
2020-09-03 12:18:42 +02:00
mpctx->logged_async_diff = -1;
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 23:01:55 +02:00
}
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 19:57:49 +02:00
void uninit_audio_out(struct MPContext *mpctx)
{
audio: refactor how data is passed to AO This replaces the two buffers (ao_chain.ao_buffer in the core, and buffer_state.buffers in the AO) with a single queue. Instead of having a byte based buffer, the queue is simply a list of audio frames, as output by the decoder. This should make dataflow simpler and reduce copying. It also attempts to simplify fill_audio_out_buffers(), the function I always hated most, because it's full of subtle and buggy logic. Unfortunately, I got assaulted by corner cases, dumb features (attempt at seamless looping, really?), and other crap, so it got pretty complicated again. fill_audio_out_buffers() is still full of subtle and buggy logic. Maybe it got worse. On the other hand, maybe there really is some progress. Who knows. Originally, the data flow parts was meant to be in f_output_chain, but due to tricky interactions with the playloop code, it's now in the dummy filter in audio.c. At least this improves the way the audio PTS is passed to the encoder in encoding mode. Now it attempts to pass frames directly, along with the pts, which should minimize timestamp problems. But to be honest, encoder mode is one big kludge that shouldn't exist in this way. This commit should be considered pre-alpha code. There are lots of bugs still hiding.
2020-08-28 20:23:54 +02:00
struct ao_chain *ao_c = mpctx->ao_chain;
if (ao_c) {
ao_c->ao_queue = NULL;
TA_FREEP(&ao_c->queue_filter);
ao_c->ao = 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 19:57:49 +02:00
if (mpctx->ao) {
// Note: with gapless_audio, stop_play is not correctly set
audio: do not show audio draining message when it does not make sense Just for the redundant message. The function which is called here, ao_drain(), does not care in which state it is called, and already handled this gracefully.
2020-09-01 21:27:51 +02:00
if ((mpctx->opts->gapless_audio || mpctx->stop_play == AT_END_OF_FILE) &&
ao_is_playing(mpctx->ao) && !get_internal_paused(mpctx))
{
audio: refactor how data is passed to AO This replaces the two buffers (ao_chain.ao_buffer in the core, and buffer_state.buffers in the AO) with a single queue. Instead of having a byte based buffer, the queue is simply a list of audio frames, as output by the decoder. This should make dataflow simpler and reduce copying. It also attempts to simplify fill_audio_out_buffers(), the function I always hated most, because it's full of subtle and buggy logic. Unfortunately, I got assaulted by corner cases, dumb features (attempt at seamless looping, really?), and other crap, so it got pretty complicated again. fill_audio_out_buffers() is still full of subtle and buggy logic. Maybe it got worse. On the other hand, maybe there really is some progress. Who knows. Originally, the data flow parts was meant to be in f_output_chain, but due to tricky interactions with the playloop code, it's now in the dummy filter in audio.c. At least this improves the way the audio PTS is passed to the encoder in encoding mode. Now it attempts to pass frames directly, along with the pts, which should minimize timestamp problems. But to be honest, encoder mode is one big kludge that shouldn't exist in this way. This commit should be considered pre-alpha code. There are lots of bugs still hiding.
2020-08-28 20:23:54 +02:00
MP_VERBOSE(mpctx, "draining left over audio\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 19:57:49 +02:00
ao_drain(mpctx->ao);
audio: refactor how data is passed to AO This replaces the two buffers (ao_chain.ao_buffer in the core, and buffer_state.buffers in the AO) with a single queue. Instead of having a byte based buffer, the queue is simply a list of audio frames, as output by the decoder. This should make dataflow simpler and reduce copying. It also attempts to simplify fill_audio_out_buffers(), the function I always hated most, because it's full of subtle and buggy logic. Unfortunately, I got assaulted by corner cases, dumb features (attempt at seamless looping, really?), and other crap, so it got pretty complicated again. fill_audio_out_buffers() is still full of subtle and buggy logic. Maybe it got worse. On the other hand, maybe there really is some progress. Who knows. Originally, the data flow parts was meant to be in f_output_chain, but due to tricky interactions with the playloop code, it's now in the dummy filter in audio.c. At least this improves the way the audio PTS is passed to the encoder in encoding mode. Now it attempts to pass frames directly, along with the pts, which should minimize timestamp problems. But to be honest, encoder mode is one big kludge that shouldn't exist in this way. This commit should be considered pre-alpha code. There are lots of bugs still hiding.
2020-08-28 20:23:54 +02:00
}
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 19:57:49 +02:00
ao_uninit(mpctx->ao);
audio: add some change notifications We must be sure that every change comes with a notification. Otherwise, some property changes could possibly be missed.
2015-06-09 18:29:11 +02:00
mp_notify(mpctx, MPV_EVENT_AUDIO_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 19:57:49 +02:00
}
mpctx->ao = NULL;
audio: rewrite filtering glue code Use the new filtering code for audio too.
2018-01-18 14:44:20 +01:00
TA_FREEP(&mpctx->ao_filter_fmt);
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 19:57:49 +02:00
}
player: refactor: eliminate MPContext.d_audio
2016-01-21 22:24:20 +01:00
static void ao_chain_uninit(struct ao_chain *ao_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 23:19:56 +01:00
struct track *track = ao_c->track;
if (track) {
assert(track->ao_c == ao_c);
track->ao_c = NULL;
audio: move to decoder wrapper Use the decoder wrapper that was introduced for video. This removes all code duplication the old audio decoder wrapper had with the video code. (The audio wrapper was copy pasted from the video one over a decade ago, and has been kept in sync ever since by the power of copy&paste. Since the original copy&paste was possibly done by someone who did not answer to the LGPL relicensing, this should also remove all doubts about whether any of this code is left, since we now completely remove any code that could possibly have been based on it.) There is some complication with spdif handling, and a minor behavior change (it will restrict the list of codecs to spdif if spdif is to be used), but there should not be any difference in practice.
2018-01-29 06:18:33 +01:00
if (ao_c->dec_src)
assert(track->dec->f->pins[0] == ao_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 23:19:56 +01:00
}
if (ao_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 04:36:47 +01:00
mp_pin_disconnect(ao_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 23:19:56 +01:00
audio: rewrite filtering glue code Use the new filtering code for audio too.
2018-01-18 14:44:20 +01:00
talloc_free(ao_c->filter->f);
audio: refactor how data is passed to AO This replaces the two buffers (ao_chain.ao_buffer in the core, and buffer_state.buffers in the AO) with a single queue. Instead of having a byte based buffer, the queue is simply a list of audio frames, as output by the decoder. This should make dataflow simpler and reduce copying. It also attempts to simplify fill_audio_out_buffers(), the function I always hated most, because it's full of subtle and buggy logic. Unfortunately, I got assaulted by corner cases, dumb features (attempt at seamless looping, really?), and other crap, so it got pretty complicated again. fill_audio_out_buffers() is still full of subtle and buggy logic. Maybe it got worse. On the other hand, maybe there really is some progress. Who knows. Originally, the data flow parts was meant to be in f_output_chain, but due to tricky interactions with the playloop code, it's now in the dummy filter in audio.c. At least this improves the way the audio PTS is passed to the encoder in encoding mode. Now it attempts to pass frames directly, along with the pts, which should minimize timestamp problems. But to be honest, encoder mode is one big kludge that shouldn't exist in this way. This commit should be considered pre-alpha code. There are lots of bugs still hiding.
2020-08-28 20:23:54 +02:00
talloc_free(ao_c->ao_filter);
player: refactor: eliminate MPContext.d_audio
2016-01-21 22:24:20 +01:00
talloc_free(ao_c);
}
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 19:57:49 +02:00
void uninit_audio_chain(struct MPContext *mpctx)
{
audio: refactor: work towards unentangling audio decoding and filtering Similar to the video path. dec_audio.c now handles decoding only. It also looks very similar to dec_video.c, and actually contains some of the rewritten code from it. (A further goal might be unifying the decoders, I guess.) High potential for regressions.
2016-01-21 22:10:15 +01:00
if (mpctx->ao_chain) {
player: refactor: eliminate MPContext.d_audio
2016-01-21 22:24:20 +01:00
ao_chain_uninit(mpctx->ao_chain);
audio: refactor: work towards unentangling audio decoding and filtering Similar to the video path. dec_audio.c now handles decoding only. It also looks very similar to dec_video.c, and actually contains some of the rewritten code from it. (A further goal might be unifying the decoders, I guess.) High potential for regressions.
2016-01-21 22:10:15 +01:00
mpctx->ao_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 23:19:56 +01:00
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 19:57:49 +02:00
mpctx->audio_status = STATUS_EOF;
audio: add some change notifications We must be sure that every change comes with a notification. Otherwise, some property changes could possibly be missed.
2015-06-09 18:29:11 +02:00
mp_notify(mpctx, MPV_EVENT_AUDIO_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 19:57:49 +02:00
}
}
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 12:48:30 +02:00
static char *audio_config_to_str_buf(char *buf, size_t buf_sz, int rate,
int format, struct mp_chmap channels)
audio: introduce a new type to hold audio frames This is pretty pointless, but I believe it allows us to claim that the new code is not affected by the copyright of the old code. This is needed, because the original mp_audio struct was written by someone who has disagreed with LGPL relicensing (it was called af_data at the time, and was defined in af.h). The "GPL'ed" struct contents that surive are pretty trivial: just the data pointer, and some metadata like the format, samplerate, etc. - but at least in this case, any new code would be extremely similar anyway, and I'm not really sure whether it's OK to claim different copyright. So what we do is we just use AVFrame (which of course is LGPL with 100% certainty), and add some accessors around it to adapt it to mpv conventions. Also, this gets rid of some annoying conventions of mp_audio, like the struct fields that require using an accessor to write to them anyway. For the most part, this change is only dumb replacements of mp_audio related functions and fields. One minor actual change is that you can't allocate the new type on the stack anymore. Some code still uses mp_audio. All audio filter code will be deleted, so it makes no sense to convert this code. (Audio filters which are LGPL and which we keep will have to be ported to a new filter infrastructure anyway.) player/audio.c uses it because it interacts with the old filter code. push.c has some complex use of mp_audio and mp_audio_buffer, but this and pull.c will most likely be rewritten to do something else.
2017-08-16 21:00:20 +02:00
{
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 12:48:30 +02:00
char ch[128];
mp_chmap_to_str_buf(ch, sizeof(ch), &channels);
char *hr_ch = mp_chmap_to_str_hr(&channels);
if (strcmp(hr_ch, ch) != 0)
mp_snprintf_cat(ch, sizeof(ch), " (%s)", hr_ch);
snprintf(buf, buf_sz, "%dHz %s %dch %s", rate,
ch, channels.num, af_fmt_to_str(format));
return buf;
audio: introduce a new type to hold audio frames This is pretty pointless, but I believe it allows us to claim that the new code is not affected by the copyright of the old code. This is needed, because the original mp_audio struct was written by someone who has disagreed with LGPL relicensing (it was called af_data at the time, and was defined in af.h). The "GPL'ed" struct contents that surive are pretty trivial: just the data pointer, and some metadata like the format, samplerate, etc. - but at least in this case, any new code would be extremely similar anyway, and I'm not really sure whether it's OK to claim different copyright. So what we do is we just use AVFrame (which of course is LGPL with 100% certainty), and add some accessors around it to adapt it to mpv conventions. Also, this gets rid of some annoying conventions of mp_audio, like the struct fields that require using an accessor to write to them anyway. For the most part, this change is only dumb replacements of mp_audio related functions and fields. One minor actual change is that you can't allocate the new type on the stack anymore. Some code still uses mp_audio. All audio filter code will be deleted, so it makes no sense to convert this code. (Audio filters which are LGPL and which we keep will have to be ported to a new filter infrastructure anyway.) player/audio.c uses it because it interacts with the old filter code. push.c has some complex use of mp_audio and mp_audio_buffer, but this and pull.c will most likely be rewritten to do something else.
2017-08-16 21:00:20 +02:00
}
audio: don't recreate AO if a filter changes the output format Until recently, the AO was reinitialized strictly only on decoder format changes. But the commit for simplifying audio format negotiation removed this. Now the AO is recreated for any format change. This is sort of annoying if you change playback speed. The insertion/removal of af_scaletempo can change the sample format. For example, the acompressor filter will convert output to double, so toggling scaletempo will force the format back to float. This recreates the AO under the --gapless-audio=weak default. This likely affects a lot of other filters too. Work this around by allowing sample format changes, and keeping the current AO format in these cases. This is probably not a big problem. Most audio APIs force the output format to float anyway. This means you actually have to worry about what the default gapless mode does to your audio. If you start with a file that uses 8 bit per sample, and then continue playing a 24 bit FLAC, it will be converted down to 8 bit per sample. (Assuming they are played in a way that uses the gapless logic.)
2018-04-12 18:47:25 +02:00
// Decide whether on a format change, we should reinit the AO.
static bool keep_weak_gapless_format(struct mp_aframe *old, struct mp_aframe* new)
{
bool res = false;
struct mp_aframe *new_mod = mp_aframe_new_ref(new);
various: replace if + abort() with MP_HANDLE_OOM() MP_HANDLE_OOM also aborts but calls assert() first, which will result in an useful message if compiled in debug mode.
2023-01-10 18:59:21 +01:00
MP_HANDLE_OOM(new_mod);
audio: don't recreate AO if a filter changes the output format Until recently, the AO was reinitialized strictly only on decoder format changes. But the commit for simplifying audio format negotiation removed this. Now the AO is recreated for any format change. This is sort of annoying if you change playback speed. The insertion/removal of af_scaletempo can change the sample format. For example, the acompressor filter will convert output to double, so toggling scaletempo will force the format back to float. This recreates the AO under the --gapless-audio=weak default. This likely affects a lot of other filters too. Work this around by allowing sample format changes, and keeping the current AO format in these cases. This is probably not a big problem. Most audio APIs force the output format to float anyway. This means you actually have to worry about what the default gapless mode does to your audio. If you start with a file that uses 8 bit per sample, and then continue playing a 24 bit FLAC, it will be converted down to 8 bit per sample. (Assuming they are played in a way that uses the gapless logic.)
2018-04-12 18:47:25 +02:00
// If the sample formats are compatible (== libswresample generally can
// convert them), keep the AO. On other changes, recreate it.
int old_fmt = mp_aframe_get_format(old);
int new_fmt = mp_aframe_get_format(new);
if (af_format_conversion_score(old_fmt, new_fmt) == INT_MIN)
goto done; // completely incompatible formats
if (!mp_aframe_set_format(new_mod, old_fmt))
goto done;
res = mp_aframe_config_equals(old, new_mod);
done:
talloc_free(new_mod);
return res;
}
audio: refactor how data is passed to AO This replaces the two buffers (ao_chain.ao_buffer in the core, and buffer_state.buffers in the AO) with a single queue. Instead of having a byte based buffer, the queue is simply a list of audio frames, as output by the decoder. This should make dataflow simpler and reduce copying. It also attempts to simplify fill_audio_out_buffers(), the function I always hated most, because it's full of subtle and buggy logic. Unfortunately, I got assaulted by corner cases, dumb features (attempt at seamless looping, really?), and other crap, so it got pretty complicated again. fill_audio_out_buffers() is still full of subtle and buggy logic. Maybe it got worse. On the other hand, maybe there really is some progress. Who knows. Originally, the data flow parts was meant to be in f_output_chain, but due to tricky interactions with the playloop code, it's now in the dummy filter in audio.c. At least this improves the way the audio PTS is passed to the encoder in encoding mode. Now it attempts to pass frames directly, along with the pts, which should minimize timestamp problems. But to be honest, encoder mode is one big kludge that shouldn't exist in this way. This commit should be considered pre-alpha code. There are lots of bugs still hiding.
2020-08-28 20:23:54 +02:00
static void ao_chain_set_ao(struct ao_chain *ao_c, struct ao *ao)
{
if (ao_c->ao != ao) {
assert(!ao_c->ao);
ao_c->ao = ao;
ao_c->ao_queue = ao_get_queue(ao_c->ao);
ao_c->queue_filter = mp_async_queue_create_filter(ao_c->ao_filter,
MP_PIN_IN, ao_c->ao_queue);
mp_async_queue_set_notifier(ao_c->queue_filter, ao_c->ao_filter);
// Make sure filtering never stops with frames stuck in access filter.
mp_filter_set_high_priority(ao_c->queue_filter, true);
audio: fix replaygain being completely broken Switching to a new file while keeping the AO didn't update the volume. While there's an explicit audio_update_volume() call in reinit_audio_chain_src(), it doesn't work, because at that point ao_chain->ao is still NULL, and it does nothing. That's pretty weird and might cause other problems (what happens if you try to mute while the AO is "floating"?). Regarding gapless, trying to use the AO gain for replaygain is also gross nonsense, because the new replaygain computed gain would affect audio from the previous file. It looks like replaygain should be in an af_volume filter maybe. On the other hand, I enjoy setting ridiculous replaygain-preamp values and compensating with a low volume setting, which would not work well if both gains were applied to the audio independently. For now, just add the missing call. This is orthogonal to fixing replaygain "properly".
2021-05-07 14:49:36 +02:00
audio_update_volume(ao_c->mpctx);
audio: refactor how data is passed to AO This replaces the two buffers (ao_chain.ao_buffer in the core, and buffer_state.buffers in the AO) with a single queue. Instead of having a byte based buffer, the queue is simply a list of audio frames, as output by the decoder. This should make dataflow simpler and reduce copying. It also attempts to simplify fill_audio_out_buffers(), the function I always hated most, because it's full of subtle and buggy logic. Unfortunately, I got assaulted by corner cases, dumb features (attempt at seamless looping, really?), and other crap, so it got pretty complicated again. fill_audio_out_buffers() is still full of subtle and buggy logic. Maybe it got worse. On the other hand, maybe there really is some progress. Who knows. Originally, the data flow parts was meant to be in f_output_chain, but due to tricky interactions with the playloop code, it's now in the dummy filter in audio.c. At least this improves the way the audio PTS is passed to the encoder in encoding mode. Now it attempts to pass frames directly, along with the pts, which should minimize timestamp problems. But to be honest, encoder mode is one big kludge that shouldn't exist in this way. This commit should be considered pre-alpha code. There are lots of bugs still hiding.
2020-08-28 20:23:54 +02:00
}
if (ao_c->filter->ao_needs_update)
mp_output_chain_set_ao(ao_c->filter, ao_c->ao);
mp_filter_wakeup(ao_c->ao_filter);
}
audio: check ao driver init failure to avoid use after free reinit_audio_filters_and_output function will free mpctx->ao_chain when there is a failure during audio initialization. So modify it to return -1 in case of init failure. The return value is checked to avoid use after free. Reported by Address Sanitizer when manually specifying --ao which triggers "Failed to initialize audio driver" error.
2021-07-13 16:06:52 +02:00
static int reinit_audio_filters_and_output(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 22:38:29 +01:00
{
struct MPOpts *opts = mpctx->opts;
audio: refactor: work towards unentangling audio decoding and filtering Similar to the video path. dec_audio.c now handles decoding only. It also looks very similar to dec_video.c, and actually contains some of the rewritten code from it. (A further goal might be unifying the decoders, I guess.) High potential for regressions.
2016-01-21 22:10:15 +01:00
struct ao_chain *ao_c = mpctx->ao_chain;
audio: refactor: separate audio init and filter/output init Before this commit, reinit_audio_chain() did 2 things: create all the management data structures and initialize the decoder, and handling lazy filter/output init (as well as dealing with format changes). For the second purpose, it could be called multiple times (even though it wasn't really idempotent). This was pretty weird, so make them separate functions. The new function is actually idempotent too. It also turns out the reinit functions don't have to call themselves recursively for the spdif PCM fallback.
2016-01-29 22:44:35 +01:00
assert(ao_c);
player: remove some further current_track dependencies Now it's used for initialization only for audio and video.
2016-02-05 23:41:44 +01:00
struct track *track = ao_c->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 22:38:29 +01:00
audio: refactor how data is passed to AO This replaces the two buffers (ao_chain.ao_buffer in the core, and buffer_state.buffers in the AO) with a single queue. Instead of having a byte based buffer, the queue is simply a list of audio frames, as output by the decoder. This should make dataflow simpler and reduce copying. It also attempts to simplify fill_audio_out_buffers(), the function I always hated most, because it's full of subtle and buggy logic. Unfortunately, I got assaulted by corner cases, dumb features (attempt at seamless looping, really?), and other crap, so it got pretty complicated again. fill_audio_out_buffers() is still full of subtle and buggy logic. Maybe it got worse. On the other hand, maybe there really is some progress. Who knows. Originally, the data flow parts was meant to be in f_output_chain, but due to tricky interactions with the playloop code, it's now in the dummy filter in audio.c. At least this improves the way the audio PTS is passed to the encoder in encoding mode. Now it attempts to pass frames directly, along with the pts, which should minimize timestamp problems. But to be honest, encoder mode is one big kludge that shouldn't exist in this way. This commit should be considered pre-alpha code. There are lots of bugs still hiding.
2020-08-28 20:23:54 +02:00
assert(ao_c->filter->ao_needs_update);
audio: add a "weak" gapless mode, and make it default Basically, this allows gapless playback with similar files (including the ordered chapter case), while still being robust in general. The implementation is quite simplistic on purpose, in order to avoid all the weird corner cases that can occur when creating the filter chain. The consequence is that it might do not-gapless playback in more cases when needed, but if that bothers you, you still can use the normal gapless mode. Just using "--gapless-audio" or "--gapless-audio=yes" selects the old mode.
2014-06-08 23:54:05 +02:00
audio: rewrite filtering glue code Use the new filtering code for audio too.
2018-01-18 14:44:20 +01:00
// The "ideal" filter output format
struct mp_aframe *out_fmt = mp_aframe_new_ref(ao_c->filter->output_aformat);
various: replace if + abort() with MP_HANDLE_OOM() MP_HANDLE_OOM also aborts but calls assert() first, which will result in an useful message if compiled in debug mode.
2023-01-10 18:59:21 +01:00
MP_HANDLE_OOM(out_fmt);
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 12:48:30 +02:00
audio: improve behavior if filters output nothing during probing Just bail out immediately (and disable audio) if format probing has no result, instead of doing nothing and then apparently freezing. This can happen with bogus filters, cases where the first audio frame is essentially dropped by filters (can happen with large resampling factors), and if the audio track contains no packets at all, or all packets fail to decode.
2018-02-21 15:43:24 +01:00
if (!mp_aframe_config_is_valid(out_fmt)) {
talloc_free(out_fmt);
goto init_error;
}
audio: rewrite filtering glue code Use the new filtering code for audio too.
2018-01-18 14:44:20 +01:00
if (af_fmt_is_pcm(mp_aframe_get_format(out_fmt))) {
if (opts->force_srate)
mp_aframe_set_rate(out_fmt, opts->force_srate);
if (opts->audio_output_format)
mp_aframe_set_format(out_fmt, opts->audio_output_format);
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 12:48:30 +02:00
if (opts->audio_output_channels.num_chmaps == 1)
audio: rewrite filtering glue code Use the new filtering code for audio too.
2018-01-18 14:44:20 +01:00
mp_aframe_set_chmap(out_fmt, &opts->audio_output_channels.chmaps[0]);
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 12:48:30 +02:00
}
audio: rewrite filtering glue code Use the new filtering code for audio too.
2018-01-18 14:44:20 +01:00
// Weak gapless audio: if the filter output format is the same as the
// previous one, keep the AO and don't reinit anything.
// Strong gapless: always keep the AO
if ((mpctx->ao_filter_fmt && mpctx->ao && opts->gapless_audio < 0 &&
audio: don't recreate AO if a filter changes the output format Until recently, the AO was reinitialized strictly only on decoder format changes. But the commit for simplifying audio format negotiation removed this. Now the AO is recreated for any format change. This is sort of annoying if you change playback speed. The insertion/removal of af_scaletempo can change the sample format. For example, the acompressor filter will convert output to double, so toggling scaletempo will force the format back to float. This recreates the AO under the --gapless-audio=weak default. This likely affects a lot of other filters too. Work this around by allowing sample format changes, and keeping the current AO format in these cases. This is probably not a big problem. Most audio APIs force the output format to float anyway. This means you actually have to worry about what the default gapless mode does to your audio. If you start with a file that uses 8 bit per sample, and then continue playing a 24 bit FLAC, it will be converted down to 8 bit per sample. (Assuming they are played in a way that uses the gapless logic.)
2018-04-12 18:47:25 +02:00
keep_weak_gapless_format(mpctx->ao_filter_fmt, out_fmt)) ||
audio: rewrite filtering glue code Use the new filtering code for audio too.
2018-01-18 14:44:20 +01:00
(mpctx->ao && opts->gapless_audio > 0))
{
audio: refactor how data is passed to AO This replaces the two buffers (ao_chain.ao_buffer in the core, and buffer_state.buffers in the AO) with a single queue. Instead of having a byte based buffer, the queue is simply a list of audio frames, as output by the decoder. This should make dataflow simpler and reduce copying. It also attempts to simplify fill_audio_out_buffers(), the function I always hated most, because it's full of subtle and buggy logic. Unfortunately, I got assaulted by corner cases, dumb features (attempt at seamless looping, really?), and other crap, so it got pretty complicated again. fill_audio_out_buffers() is still full of subtle and buggy logic. Maybe it got worse. On the other hand, maybe there really is some progress. Who knows. Originally, the data flow parts was meant to be in f_output_chain, but due to tricky interactions with the playloop code, it's now in the dummy filter in audio.c. At least this improves the way the audio PTS is passed to the encoder in encoding mode. Now it attempts to pass frames directly, along with the pts, which should minimize timestamp problems. But to be honest, encoder mode is one big kludge that shouldn't exist in this way. This commit should be considered pre-alpha code. There are lots of bugs still hiding.
2020-08-28 20:23:54 +02:00
ao_chain_set_ao(ao_c, mpctx->ao);
talloc_free(out_fmt);
audio: check ao driver init failure to avoid use after free reinit_audio_filters_and_output function will free mpctx->ao_chain when there is a failure during audio initialization. So modify it to return -1 in case of init failure. The return value is checked to avoid use after free. Reported by Address Sanitizer when manually specifying --ao which triggers "Failed to initialize audio driver" error.
2021-07-13 16:06:52 +02:00
return 0;
audio: refactor how data is passed to AO This replaces the two buffers (ao_chain.ao_buffer in the core, and buffer_state.buffers in the AO) with a single queue. Instead of having a byte based buffer, the queue is simply a list of audio frames, as output by the decoder. This should make dataflow simpler and reduce copying. It also attempts to simplify fill_audio_out_buffers(), the function I always hated most, because it's full of subtle and buggy logic. Unfortunately, I got assaulted by corner cases, dumb features (attempt at seamless looping, really?), and other crap, so it got pretty complicated again. fill_audio_out_buffers() is still full of subtle and buggy logic. Maybe it got worse. On the other hand, maybe there really is some progress. Who knows. Originally, the data flow parts was meant to be in f_output_chain, but due to tricky interactions with the playloop code, it's now in the dummy filter in audio.c. At least this improves the way the audio PTS is passed to the encoder in encoding mode. Now it attempts to pass frames directly, along with the pts, which should minimize timestamp problems. But to be honest, encoder mode is one big kludge that shouldn't exist in this way. This commit should be considered pre-alpha code. There are lots of bugs still hiding.
2020-08-28 20:23:54 +02:00
}
// Wait until all played.
if (mpctx->ao && ao_is_playing(mpctx->ao)) {
talloc_free(out_fmt);
audio: check ao driver init failure to avoid use after free reinit_audio_filters_and_output function will free mpctx->ao_chain when there is a failure during audio initialization. So modify it to return -1 in case of init failure. The return value is checked to avoid use after free. Reported by Address Sanitizer when manually specifying --ao which triggers "Failed to initialize audio driver" error.
2021-07-13 16:06:52 +02:00
return 0;
audio: refactor how data is passed to AO This replaces the two buffers (ao_chain.ao_buffer in the core, and buffer_state.buffers in the AO) with a single queue. Instead of having a byte based buffer, the queue is simply a list of audio frames, as output by the decoder. This should make dataflow simpler and reduce copying. It also attempts to simplify fill_audio_out_buffers(), the function I always hated most, because it's full of subtle and buggy logic. Unfortunately, I got assaulted by corner cases, dumb features (attempt at seamless looping, really?), and other crap, so it got pretty complicated again. fill_audio_out_buffers() is still full of subtle and buggy logic. Maybe it got worse. On the other hand, maybe there really is some progress. Who knows. Originally, the data flow parts was meant to be in f_output_chain, but due to tricky interactions with the playloop code, it's now in the dummy filter in audio.c. At least this improves the way the audio PTS is passed to the encoder in encoding mode. Now it attempts to pass frames directly, along with the pts, which should minimize timestamp problems. But to be honest, encoder mode is one big kludge that shouldn't exist in this way. This commit should be considered pre-alpha code. There are lots of bugs still hiding.
2020-08-28 20:23:54 +02:00
}
// Format change during syncing. Force playback start early, then wait.
if (ao_c->ao_queue && mp_async_queue_get_frames(ao_c->ao_queue) &&
mpctx->audio_status == STATUS_SYNCING)
{
mpctx->audio_status = STATUS_READY;
mp_wakeup_core(mpctx);
talloc_free(out_fmt);
audio: check ao driver init failure to avoid use after free reinit_audio_filters_and_output function will free mpctx->ao_chain when there is a failure during audio initialization. So modify it to return -1 in case of init failure. The return value is checked to avoid use after free. Reported by Address Sanitizer when manually specifying --ao which triggers "Failed to initialize audio driver" error.
2021-07-13 16:06:52 +02:00
return 0;
audio: refactor how data is passed to AO This replaces the two buffers (ao_chain.ao_buffer in the core, and buffer_state.buffers in the AO) with a single queue. Instead of having a byte based buffer, the queue is simply a list of audio frames, as output by the decoder. This should make dataflow simpler and reduce copying. It also attempts to simplify fill_audio_out_buffers(), the function I always hated most, because it's full of subtle and buggy logic. Unfortunately, I got assaulted by corner cases, dumb features (attempt at seamless looping, really?), and other crap, so it got pretty complicated again. fill_audio_out_buffers() is still full of subtle and buggy logic. Maybe it got worse. On the other hand, maybe there really is some progress. Who knows. Originally, the data flow parts was meant to be in f_output_chain, but due to tricky interactions with the playloop code, it's now in the dummy filter in audio.c. At least this improves the way the audio PTS is passed to the encoder in encoding mode. Now it attempts to pass frames directly, along with the pts, which should minimize timestamp problems. But to be honest, encoder mode is one big kludge that shouldn't exist in this way. This commit should be considered pre-alpha code. There are lots of bugs still hiding.
2020-08-28 20:23:54 +02:00
}
if (mpctx->audio_status == STATUS_READY) {
audio: rewrite filtering glue code Use the new filtering code for audio too.
2018-01-18 14:44:20 +01:00
talloc_free(out_fmt);
audio: check ao driver init failure to avoid use after free reinit_audio_filters_and_output function will free mpctx->ao_chain when there is a failure during audio initialization. So modify it to return -1 in case of init failure. The return value is checked to avoid use after free. Reported by Address Sanitizer when manually specifying --ao which triggers "Failed to initialize audio driver" error.
2021-07-13 16:06:52 +02:00
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 22:38:29 +01:00
}
audio: rewrite filtering glue code Use the new filtering code for audio too.
2018-01-18 14:44:20 +01:00
uninit_audio_out(mpctx);
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 12:48:30 +02:00
audio: rewrite filtering glue code Use the new filtering code for audio too.
2018-01-18 14:44:20 +01:00
int out_rate = mp_aframe_get_rate(out_fmt);
int out_format = mp_aframe_get_format(out_fmt);
struct mp_chmap out_channels = {0};
mp_aframe_get_chmap(out_fmt, &out_channels);
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 12:48:30 +02:00
audio: rewrite filtering glue code Use the new filtering code for audio too.
2018-01-18 14:44:20 +01:00
int ao_flags = 0;
bool spdif_fallback = af_fmt_is_spdif(out_format) &&
ao_c->spdif_passthrough;
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 12:48:30 +02:00
audio: rewrite filtering glue code Use the new filtering code for audio too.
2018-01-18 14:44:20 +01:00
if (opts->ao_null_fallback && !spdif_fallback)
ao_flags |= AO_INIT_NULL_FALLBACK;
audio: use --audio-channels=auto behavior, except on ALSA This commit adds an --audio-channel=auto-safe mode, and makes it the default. This mode behaves like "auto" with most AOs, except with ao_alsa. The intention is to allow multichannel output by default on sane APIs. ALSA is not sane as in it's so low level that it will e.g. configure any layout over HDMI, even if the connected A/V receiver does not support it. The HDMI fuckup is of course not ALSA's fault, but other audio APIs normally isolate applications from dealing with this and require the user to globally configure the correct output layout. This will help with other AOs too. ao_lavc (encoding) is changed to the new semantics as well, because it used to force stereo (perhaps because encoding mode is supposed to produce safe files for crap devices?). Exclusive mode output on Windows might need to be adjusted accordingly, as it grants the same kind of low level access as ALSA (requires more research). In addition to the things mentioned above, the --audio-channels option is extended to accept a set of channel layouts. This is supposed to be the correct way to configure mpv ALSA multichannel output. You need to put a list of channel layouts that your A/V receiver supports.
2016-08-04 20:49:20 +02:00
audio: rewrite filtering glue code Use the new filtering code for audio too.
2018-01-18 14:44:20 +01:00
if (opts->audio_stream_silence)
ao_flags |= AO_INIT_STREAM_SILENCE;
audio: use --audio-channels=auto behavior, except on ALSA This commit adds an --audio-channel=auto-safe mode, and makes it the default. This mode behaves like "auto" with most AOs, except with ao_alsa. The intention is to allow multichannel output by default on sane APIs. ALSA is not sane as in it's so low level that it will e.g. configure any layout over HDMI, even if the connected A/V receiver does not support it. The HDMI fuckup is of course not ALSA's fault, but other audio APIs normally isolate applications from dealing with this and require the user to globally configure the correct output layout. This will help with other AOs too. ao_lavc (encoding) is changed to the new semantics as well, because it used to force stereo (perhaps because encoding mode is supposed to produce safe files for crap devices?). Exclusive mode output on Windows might need to be adjusted accordingly, as it grants the same kind of low level access as ALSA (requires more research). In addition to the things mentioned above, the --audio-channels option is extended to accept a set of channel layouts. This is supposed to be the correct way to configure mpv ALSA multichannel output. You need to put a list of channel layouts that your A/V receiver supports.
2016-08-04 20:49:20 +02:00
audio: rewrite filtering glue code Use the new filtering code for audio too.
2018-01-18 14:44:20 +01:00
if (opts->audio_exclusive)
ao_flags |= AO_INIT_EXCLUSIVE;
player: add --audio-stream-silence Completely insane that this has to be done. Crap for compensating HDMI crap.
2016-08-09 16:22:06 +02:00
audio: rewrite filtering glue code Use the new filtering code for audio too.
2018-01-18 14:44:20 +01:00
if (af_fmt_is_pcm(out_format)) {
if (!opts->audio_output_channels.set ||
opts->audio_output_channels.auto_safe)
ao_flags |= AO_INIT_SAFE_MULTICHANNEL_ONLY;
audio/out: deprecate "exclusive" sub-options And introduce a global option which does this. Or more precisely, this deprecates the global wasapi and coreaudio options, and adds a new one that merges their functionality. (Due to the way the sub-option deprecation mechanism works, this is simpler.)
2016-09-05 21:07:34 +02:00
audio: rewrite filtering glue code Use the new filtering code for audio too.
2018-01-18 14:44:20 +01:00
mp_chmap_sel_list(&out_channels,
opts->audio_output_channels.chmaps,
opts->audio_output_channels.num_chmaps);
}
audio: do not apply --audio-channels if spdif passthrough is in use If spdif is enabled, the channel layout has no meaning other than setting the number of channels. The number of channels must be fixed to achieve the exact bitrate required. Fixes #3445.
2016-08-22 12:12:10 +02:00
audio: rewrite filtering glue code Use the new filtering code for audio too.
2018-01-18 14:44:20 +01:00
mpctx->ao_filter_fmt = out_fmt;
audio: add option for falling back to ao_null The manpage entry explains this. (Maybe this option could be always enabled and removed. I don't quite remember what valid use-cases there are for just disabling audio entirely, other than that this is also needed for audio decoder init failure.)
2015-10-05 18:53:02 +02:00
audio: rewrite filtering glue code Use the new filtering code for audio too.
2018-01-18 14:44:20 +01:00
mpctx->ao = ao_init_best(mpctx->global, ao_flags, mp_wakeup_core_cb,
mpctx, mpctx->encode_lavc_ctx, out_rate,
out_format, out_channels);
audio: do not allow AO to change passthrough formats This makes no sense, because the format can't be converted anyway. It just sets up the filter chain init code, which will vomit a bunch of useless and confusing messages. So uninit and fail explicitly when this happens.
2015-06-05 22:34:22 +02:00
audio: rewrite filtering glue code Use the new filtering code for audio too.
2018-01-18 14:44:20 +01:00
int ao_rate = 0;
int ao_format = 0;
struct mp_chmap ao_channels = {0};
if (mpctx->ao)
ao_get_format(mpctx->ao, &ao_rate, &ao_format, &ao_channels);
audio: add --audio-spdif as new method for enabling passthrough This provides a new method for enabling spdif passthrough. The old method via --ad (--ad=spdif:ac3 etc.) is deprecated. The deprecated method will probably stop working at some point. This also supports PCM fallback. One caveat is that it will lose at least 1 audio packet in doing so. (I don't care enough to prevent this.) (This is named after the old S/PDIF connector, because it uses the same underlying technology as far as the higher level protoco is concerned. Also, the user should be renamed that passthrough is backwards.)
2015-06-05 22:35:43 +02:00
audio: rewrite filtering glue code Use the new filtering code for audio too.
2018-01-18 14:44:20 +01:00
// Verify passthrough format was not changed.
if (mpctx->ao && af_fmt_is_spdif(out_format)) {
if (out_rate != ao_rate || out_format != ao_format ||
!mp_chmap_equals(&out_channels, &ao_channels))
{
MP_ERR(mpctx, "Passthrough format unsupported.\n");
ao_uninit(mpctx->ao);
mpctx->ao = NULL;
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 22:38:29 +01:00
}
audio: rewrite filtering glue code Use the new filtering code for audio too.
2018-01-18 14:44:20 +01:00
}
audio: switch output to mp_audio_buffer Replace the code that used a single buffer with mp_audio_buffer. This also enables non-interleaved output operation, although it's still disabled, and no AO supports it yet.
2013-11-10 23:38:18 +01:00
audio: rewrite filtering glue code Use the new filtering code for audio too.
2018-01-18 14:44:20 +01:00
if (!mpctx->ao) {
// If spdif was used, try to fallback to PCM.
audio: move to decoder wrapper Use the decoder wrapper that was introduced for video. This removes all code duplication the old audio decoder wrapper had with the video code. (The audio wrapper was copy pasted from the video one over a decade ago, and has been kept in sync ever since by the power of copy&paste. Since the original copy&paste was possibly done by someone who did not answer to the LGPL relicensing, this should also remove all doubts about whether any of this code is left, since we now completely remove any code that could possibly have been based on it.) There is some complication with spdif handling, and a minor behavior change (it will restrict the list of codecs to spdif if spdif is to be used), but there should not be any difference in practice.
2018-01-29 06:18:33 +01:00
if (spdif_fallback && ao_c->track && ao_c->track->dec) {
audio: rewrite filtering glue code Use the new filtering code for audio too.
2018-01-18 14:44:20 +01:00
MP_VERBOSE(mpctx, "Falling back to PCM output.\n");
ao_c->spdif_passthrough = false;
ao_c->spdif_failed = true;
f_decoder_wrapper: replace most public fields with setters/getters I may (optionally) move decoding to a separate thread in a future change. It's a bit attractive to move the entire decoder wrapper to there, so if the demuxer has a new packet, it doesn't have to wake up the main thread, and can directly wake up the decoder. (Although that's bullshit, since there's a queue in between, and libavcodec's multi-threaded decoding plays cross-threads ping pong with packets anyway. On the other hand, the main thread would still have to shuffle the packets around, so whatever, just seems like better design.) As preparation, there shouldn't be any mutable state exposed by the wrapper. But there's still a large number of corner-caseish crap, so just use setters/getters for them. This recorder thing will inherently not work, so it'll have to be disabled if threads are used. This is a bit painful, but probably still the right thing. Like speculatively pulling teeth.
2020-02-28 21:17:55 +01:00
mp_decoder_wrapper_set_spdif_flag(ao_c->track->dec, false);
audio: move to decoder wrapper Use the decoder wrapper that was introduced for video. This removes all code duplication the old audio decoder wrapper had with the video code. (The audio wrapper was copy pasted from the video one over a decade ago, and has been kept in sync ever since by the power of copy&paste. Since the original copy&paste was possibly done by someone who did not answer to the LGPL relicensing, this should also remove all doubts about whether any of this code is left, since we now completely remove any code that could possibly have been based on it.) There is some complication with spdif handling, and a minor behavior change (it will restrict the list of codecs to spdif if spdif is to be used), but there should not be any difference in practice.
2018-01-29 06:18:33 +01:00
if (!mp_decoder_wrapper_reinit(ao_c->track->dec))
audio: rewrite filtering glue code Use the new filtering code for audio too.
2018-01-18 14:44:20 +01:00
goto init_error;
reset_audio_state(mpctx);
mp_output_chain_reset_harder(ao_c->filter);
mp_wakeup_core(mpctx); // reinit with new format next time
audio: check ao driver init failure to avoid use after free reinit_audio_filters_and_output function will free mpctx->ao_chain when there is a failure during audio initialization. So modify it to return -1 in case of init failure. The return value is checked to avoid use after free. Reported by Address Sanitizer when manually specifying --ao which triggers "Failed to initialize audio driver" error.
2021-07-13 16:06:52 +02:00
return 0;
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 12:48:30 +02:00
}
audio/out: make ao struct opaque We want to move the AO to its own thread. There's no technical reason for making the ao struct opaque to do this. But it helps us sleep at night, because we can control access to shared state better.
2014-03-07 15:24:32 +01:00
audio: rewrite filtering glue code Use the new filtering code for audio too.
2018-01-18 14:44:20 +01:00
MP_ERR(mpctx, "Could not open/initialize audio device -> no sound.\n");
mpctx->error_playing = MPV_ERROR_AO_INIT_FAILED;
goto init_error;
}
audio: add a "weak" gapless mode, and make it default Basically, this allows gapless playback with similar files (including the ordered chapter case), while still being robust in general. The implementation is quite simplistic on purpose, in order to avoid all the weird corner cases that can occur when creating the filter chain. The consequence is that it might do not-gapless playback in more cases when needed, but if that bothers you, you still can use the normal gapless mode. Just using "--gapless-audio" or "--gapless-audio=yes" selects the old mode.
2014-06-08 23:54:05 +02:00
audio: increase a buffer size The buffer is written to in `audio_config_to_str_buf` using `snprintf` with a `%s` formatting of a 128-byte buffer. This can overflow the target buffer of 80 causing a truncated output.
2018-10-31 02:04:39 +01:00
char tmp[192];
audio: rewrite filtering glue code Use the new filtering code for audio too.
2018-01-18 14:44:20 +01:00
MP_INFO(mpctx, "AO: [%s] %s\n", ao_get_name(mpctx->ao),
audio_config_to_str_buf(tmp, sizeof(tmp), ao_rate, ao_format,
ao_channels));
MP_VERBOSE(mpctx, "AO: Description: %s\n", ao_get_description(mpctx->ao));
update_window_title(mpctx, true);
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 22:38:29 +01:00
audio: rewrite filtering glue code Use the new filtering code for audio too.
2018-01-18 14:44:20 +01:00
ao_c->ao_resume_time =
opts->audio_wait_open > 0 ? mp_time_sec() + opts->audio_wait_open : 0;
audio: refactor how data is passed to AO This replaces the two buffers (ao_chain.ao_buffer in the core, and buffer_state.buffers in the AO) with a single queue. Instead of having a byte based buffer, the queue is simply a list of audio frames, as output by the decoder. This should make dataflow simpler and reduce copying. It also attempts to simplify fill_audio_out_buffers(), the function I always hated most, because it's full of subtle and buggy logic. Unfortunately, I got assaulted by corner cases, dumb features (attempt at seamless looping, really?), and other crap, so it got pretty complicated again. fill_audio_out_buffers() is still full of subtle and buggy logic. Maybe it got worse. On the other hand, maybe there really is some progress. Who knows. Originally, the data flow parts was meant to be in f_output_chain, but due to tricky interactions with the playloop code, it's now in the dummy filter in audio.c. At least this improves the way the audio PTS is passed to the encoder in encoding mode. Now it attempts to pass frames directly, along with the pts, which should minimize timestamp problems. But to be honest, encoder mode is one big kludge that shouldn't exist in this way. This commit should be considered pre-alpha code. There are lots of bugs still hiding.
2020-08-28 20:23:54 +02:00
ao_set_paused(mpctx->ao, get_internal_paused(mpctx));
ao_chain_set_ao(ao_c, mpctx->ao);
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 22:38:29 +01:00
audio: fix missing volume update on init and reinit This is never updated after the AO inits, so there are several cases where the volume would stay at 100%, even if it shouldn't. This affects initial volume as well as track switching or switching between files.
2017-12-01 21:16:59 +01:00
audio_update_volume(mpctx);
audio: enable pitch correction by default when playing fast Apparently this is what users want. When playing with normal speed, nothing is done. When playing slower than normal, resampling is used instead, because scaletempo (which does the pitch correction) adds too many artifacts.
2014-10-02 02:58:52 +02:00
audio: add AOCONTROL_UPDATE_MEDIA_ROLE This is used to notify an AO about the type of media that is being played. Either a movie or music.
2022-09-04 11:49:23 +02:00
audio_update_media_role(mpctx);
audio: refactor how data is passed to AO This replaces the two buffers (ao_chain.ao_buffer in the core, and buffer_state.buffers in the AO) with a single queue. Instead of having a byte based buffer, the queue is simply a list of audio frames, as output by the decoder. This should make dataflow simpler and reduce copying. It also attempts to simplify fill_audio_out_buffers(), the function I always hated most, because it's full of subtle and buggy logic. Unfortunately, I got assaulted by corner cases, dumb features (attempt at seamless looping, really?), and other crap, so it got pretty complicated again. fill_audio_out_buffers() is still full of subtle and buggy logic. Maybe it got worse. On the other hand, maybe there really is some progress. Who knows. Originally, the data flow parts was meant to be in f_output_chain, but due to tricky interactions with the playloop code, it's now in the dummy filter in audio.c. At least this improves the way the audio PTS is passed to the encoder in encoding mode. Now it attempts to pass frames directly, along with the pts, which should minimize timestamp problems. But to be honest, encoder mode is one big kludge that shouldn't exist in this way. This commit should be considered pre-alpha code. There are lots of bugs still hiding.
2020-08-28 20:23:54 +02:00
// Almost nonsensical hack to deal with certain format change scenarios.
if (mpctx->audio_status == STATUS_PLAYING)
ao_start(mpctx->ao);
mp_wakeup_core(mpctx);
player: add missing audio reconfig events This also takes care of sending the required property change notifications. Fixes #2929 and maybe fixes #2920.
2016-03-13 15:52:17 +01:00
mp_notify(mpctx, MPV_EVENT_AUDIO_RECONFIG, NULL);
audio: check ao driver init failure to avoid use after free reinit_audio_filters_and_output function will free mpctx->ao_chain when there is a failure during audio initialization. So modify it to return -1 in case of init failure. The return value is checked to avoid use after free. Reported by Address Sanitizer when manually specifying --ao which triggers "Failed to initialize audio driver" error.
2021-07-13 16:06:52 +02:00
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 22:38:29 +01:00
audio: refactor: separate audio init and filter/output init Before this commit, reinit_audio_chain() did 2 things: create all the management data structures and initialize the decoder, and handling lazy filter/output init (as well as dealing with format changes). For the second purpose, it could be called multiple times (even though it wasn't really idempotent). This was pretty weird, so make them separate functions. The new function is actually idempotent too. It also turns out the reinit functions don't have to call themselves recursively for the spdif PCM fallback.
2016-01-29 22:44:35 +01:00
init_error:
uninit_audio_chain(mpctx);
uninit_audio_out(mpctx);
player: remove some further current_track dependencies Now it's used for initialization only for audio and video.
2016-02-05 23:41:44 +01:00
error_on_track(mpctx, track);
audio: check ao driver init failure to avoid use after free reinit_audio_filters_and_output function will free mpctx->ao_chain when there is a failure during audio initialization. So modify it to return -1 in case of init failure. The return value is checked to avoid use after free. Reported by Address Sanitizer when manually specifying --ao which triggers "Failed to initialize audio driver" error.
2021-07-13 16:06:52 +02:00
return -1;
audio: refactor: separate audio init and filter/output init Before this commit, reinit_audio_chain() did 2 things: create all the management data structures and initialize the decoder, and handling lazy filter/output init (as well as dealing with format changes). For the second purpose, it could be called multiple times (even though it wasn't really idempotent). This was pretty weird, so make them separate functions. The new function is actually idempotent too. It also turns out the reinit functions don't have to call themselves recursively for the spdif PCM fallback.
2016-01-29 22:44:35 +01:00
}
player: move audio and video decoder init to separate functions Preparation.
2016-02-05 21:50:37 +01:00
int init_audio_decoder(struct MPContext *mpctx, struct track *track)
{
audio: move to decoder wrapper Use the decoder wrapper that was introduced for video. This removes all code duplication the old audio decoder wrapper had with the video code. (The audio wrapper was copy pasted from the video one over a decade ago, and has been kept in sync ever since by the power of copy&paste. Since the original copy&paste was possibly done by someone who did not answer to the LGPL relicensing, this should also remove all doubts about whether any of this code is left, since we now completely remove any code that could possibly have been based on it.) There is some complication with spdif handling, and a minor behavior change (it will restrict the list of codecs to spdif if spdif is to be used), but there should not be any difference in practice.
2018-01-29 06:18:33 +01:00
assert(!track->dec);
player: move audio and video decoder init to separate functions Preparation.
2016-02-05 21:50:37 +01:00
if (!track->stream)
goto init_error;
audio: move to decoder wrapper Use the decoder wrapper that was introduced for video. This removes all code duplication the old audio decoder wrapper had with the video code. (The audio wrapper was copy pasted from the video one over a decade ago, and has been kept in sync ever since by the power of copy&paste. Since the original copy&paste was possibly done by someone who did not answer to the LGPL relicensing, this should also remove all doubts about whether any of this code is left, since we now completely remove any code that could possibly have been based on it.) There is some complication with spdif handling, and a minor behavior change (it will restrict the list of codecs to spdif if spdif is to be used), but there should not be any difference in practice.
2018-01-29 06:18:33 +01:00
track->dec = mp_decoder_wrapper_create(mpctx->filter_root, track->stream);
if (!track->dec)
goto init_error;
player: move audio and video decoder init to separate functions Preparation.
2016-02-05 21:50:37 +01:00
audio: move to decoder wrapper Use the decoder wrapper that was introduced for video. This removes all code duplication the old audio decoder wrapper had with the video code. (The audio wrapper was copy pasted from the video one over a decade ago, and has been kept in sync ever since by the power of copy&paste. Since the original copy&paste was possibly done by someone who did not answer to the LGPL relicensing, this should also remove all doubts about whether any of this code is left, since we now completely remove any code that could possibly have been based on it.) There is some complication with spdif handling, and a minor behavior change (it will restrict the list of codecs to spdif if spdif is to be used), but there should not be any difference in practice.
2018-01-29 06:18:33 +01:00
if (track->ao_c)
f_decoder_wrapper: replace most public fields with setters/getters I may (optionally) move decoding to a separate thread in a future change. It's a bit attractive to move the entire decoder wrapper to there, so if the demuxer has a new packet, it doesn't have to wake up the main thread, and can directly wake up the decoder. (Although that's bullshit, since there's a queue in between, and libavcodec's multi-threaded decoding plays cross-threads ping pong with packets anyway. On the other hand, the main thread would still have to shuffle the packets around, so whatever, just seems like better design.) As preparation, there shouldn't be any mutable state exposed by the wrapper. But there's still a large number of corner-caseish crap, so just use setters/getters for them. This recorder thing will inherently not work, so it'll have to be disabled if threads are used. This is a bit painful, but probably still the right thing. Like speculatively pulling teeth.
2020-02-28 21:17:55 +01:00
mp_decoder_wrapper_set_spdif_flag(track->dec, true);
player: move audio and video decoder init to separate functions Preparation.
2016-02-05 21:50:37 +01:00
audio: move to decoder wrapper Use the decoder wrapper that was introduced for video. This removes all code duplication the old audio decoder wrapper had with the video code. (The audio wrapper was copy pasted from the video one over a decade ago, and has been kept in sync ever since by the power of copy&paste. Since the original copy&paste was possibly done by someone who did not answer to the LGPL relicensing, this should also remove all doubts about whether any of this code is left, since we now completely remove any code that could possibly have been based on it.) There is some complication with spdif handling, and a minor behavior change (it will restrict the list of codecs to spdif if spdif is to be used), but there should not be any difference in practice.
2018-01-29 06:18:33 +01:00
if (!mp_decoder_wrapper_reinit(track->dec))
player: move audio and video decoder init to separate functions Preparation.
2016-02-05 21:50:37 +01:00
goto init_error;
return 1;
init_error:
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 23:19:56 +01:00
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 04:36:47 +01:00
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 23:19:56 +01:00
track->sink = NULL;
player: move audio and video decoder init to separate functions Preparation.
2016-02-05 21:50:37 +01:00
error_on_track(mpctx, track);
return 0;
}
audio: refactor: separate audio init and filter/output init Before this commit, reinit_audio_chain() did 2 things: create all the management data structures and initialize the decoder, and handling lazy filter/output init (as well as dealing with format changes). For the second purpose, it could be called multiple times (even though it wasn't really idempotent). This was pretty weird, so make them separate functions. The new function is actually idempotent too. It also turns out the reinit functions don't have to call themselves recursively for the spdif PCM fallback.
2016-01-29 22:44:35 +01:00
void reinit_audio_chain(struct MPContext *mpctx)
{
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 23:08:48 +02:00
struct track *track = NULL;
track = mpctx->current_track[0][STREAM_AUDIO];
if (!track || !track->stream) {
audio: prevent uninit_audio_out during encoding There was a simple oversight that meant audio outputs were uninitialized during an encoding, which is not allowed, the encoding would stop with numerous errors. I added a single line to prevent the call of uninit_audio_out in reinit_audio_chain if the encoder was active and this appears to have fixed the problem without breaking anything else. Fixes #8568
2021-02-17 12:04:07 +01:00
if (!mpctx->encode_lavc_ctx)
uninit_audio_out(mpctx);
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 23:08:48 +02:00
error_on_track(mpctx, track);
return;
}
reinit_audio_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 23:19:56 +01:00
}
audio: refactor: separate audio init and filter/output init Before this commit, reinit_audio_chain() did 2 things: create all the management data structures and initialize the decoder, and handling lazy filter/output init (as well as dealing with format changes). For the second purpose, it could be called multiple times (even though it wasn't really idempotent). This was pretty weird, so make them separate functions. The new function is actually idempotent too. It also turns out the reinit functions don't have to call themselves recursively for the spdif PCM fallback.
2016-01-29 22:44:35 +01:00
audio: refactor how data is passed to AO This replaces the two buffers (ao_chain.ao_buffer in the core, and buffer_state.buffers in the AO) with a single queue. Instead of having a byte based buffer, the queue is simply a list of audio frames, as output by the decoder. This should make dataflow simpler and reduce copying. It also attempts to simplify fill_audio_out_buffers(), the function I always hated most, because it's full of subtle and buggy logic. Unfortunately, I got assaulted by corner cases, dumb features (attempt at seamless looping, really?), and other crap, so it got pretty complicated again. fill_audio_out_buffers() is still full of subtle and buggy logic. Maybe it got worse. On the other hand, maybe there really is some progress. Who knows. Originally, the data flow parts was meant to be in f_output_chain, but due to tricky interactions with the playloop code, it's now in the dummy filter in audio.c. At least this improves the way the audio PTS is passed to the encoder in encoding mode. Now it attempts to pass frames directly, along with the pts, which should minimize timestamp problems. But to be honest, encoder mode is one big kludge that shouldn't exist in this way. This commit should be considered pre-alpha code. There are lots of bugs still hiding.
2020-08-28 20:23:54 +02:00
static const struct mp_filter_info ao_filter = {
.name = "ao",
.process = ao_process,
};
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 23:08:48 +02:00
// (track=NULL creates a blank chain, used for lavfi-complex)
void reinit_audio_chain_src(struct MPContext *mpctx, struct track *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 23:19:56 +01:00
{
assert(!mpctx->ao_chain);
audio: refactor: separate audio init and filter/output init Before this commit, reinit_audio_chain() did 2 things: create all the management data structures and initialize the decoder, and handling lazy filter/output init (as well as dealing with format changes). For the second purpose, it could be called multiple times (even though it wasn't really idempotent). This was pretty weird, so make them separate functions. The new function is actually idempotent too. It also turns out the reinit functions don't have to call themselves recursively for the spdif PCM fallback.
2016-01-29 22:44:35 +01:00
mp_notify(mpctx, MPV_EVENT_AUDIO_RECONFIG, NULL);
player: remove double assignment in declaration Fixes CID 1350058. (Still looks like this might be valid C, in some fucked up way.)
2016-02-12 16:00:14 +01:00
struct ao_chain *ao_c = talloc_zero(NULL, struct ao_chain);
audio: refactor: separate audio init and filter/output init Before this commit, reinit_audio_chain() did 2 things: create all the management data structures and initialize the decoder, and handling lazy filter/output init (as well as dealing with format changes). For the second purpose, it could be called multiple times (even though it wasn't really idempotent). This was pretty weird, so make them separate functions. The new function is actually idempotent too. It also turns out the reinit functions don't have to call themselves recursively for the spdif PCM fallback.
2016-01-29 22:44:35 +01:00
mpctx->ao_chain = ao_c;
audio: refactor how data is passed to AO This replaces the two buffers (ao_chain.ao_buffer in the core, and buffer_state.buffers in the AO) with a single queue. Instead of having a byte based buffer, the queue is simply a list of audio frames, as output by the decoder. This should make dataflow simpler and reduce copying. It also attempts to simplify fill_audio_out_buffers(), the function I always hated most, because it's full of subtle and buggy logic. Unfortunately, I got assaulted by corner cases, dumb features (attempt at seamless looping, really?), and other crap, so it got pretty complicated again. fill_audio_out_buffers() is still full of subtle and buggy logic. Maybe it got worse. On the other hand, maybe there really is some progress. Who knows. Originally, the data flow parts was meant to be in f_output_chain, but due to tricky interactions with the playloop code, it's now in the dummy filter in audio.c. At least this improves the way the audio PTS is passed to the encoder in encoding mode. Now it attempts to pass frames directly, along with the pts, which should minimize timestamp problems. But to be honest, encoder mode is one big kludge that shouldn't exist in this way. This commit should be considered pre-alpha code. There are lots of bugs still hiding.
2020-08-28 20:23:54 +02:00
ao_c->mpctx = mpctx;
audio: refactor: separate audio init and filter/output init Before this commit, reinit_audio_chain() did 2 things: create all the management data structures and initialize the decoder, and handling lazy filter/output init (as well as dealing with format changes). For the second purpose, it could be called multiple times (even though it wasn't really idempotent). This was pretty weird, so make them separate functions. The new function is actually idempotent too. It also turns out the reinit functions don't have to call themselves recursively for the spdif PCM fallback.
2016-01-29 22:44:35 +01:00
ao_c->log = mpctx->log;
audio: rewrite filtering glue code Use the new filtering code for audio too.
2018-01-18 14:44:20 +01:00
ao_c->filter =
mp_output_chain_create(mpctx->filter_root, MP_OUTPUT_CHAIN_AUDIO);
audio: refactor: separate audio init and filter/output init Before this commit, reinit_audio_chain() did 2 things: create all the management data structures and initialize the decoder, and handling lazy filter/output init (as well as dealing with format changes). For the second purpose, it could be called multiple times (even though it wasn't really idempotent). This was pretty weird, so make them separate functions. The new function is actually idempotent too. It also turns out the reinit functions don't have to call themselves recursively for the spdif PCM fallback.
2016-01-29 22:44:35 +01:00
ao_c->spdif_passthrough = true;
audio: move to decoder wrapper Use the decoder wrapper that was introduced for video. This removes all code duplication the old audio decoder wrapper had with the video code. (The audio wrapper was copy pasted from the video one over a decade ago, and has been kept in sync ever since by the power of copy&paste. Since the original copy&paste was possibly done by someone who did not answer to the LGPL relicensing, this should also remove all doubts about whether any of this code is left, since we now completely remove any code that could possibly have been based on it.) There is some complication with spdif handling, and a minor behavior change (it will restrict the list of codecs to spdif if spdif is to be used), but there should not be any difference in practice.
2018-01-29 06:18:33 +01:00
ao_c->last_out_pts = MP_NOPTS_VALUE;
command: shuffle some crap around This is preparation to get rid of the option-to-property bridge (mp_on_set_option). This is a pretty insane thing that redirects accesses to options to properties. It was needed in the ever ongoing transition from something to... something else. A good example for the need of this bridge is applying profiles at runtime. This obviously goes through the config parser, but should also make all changes effective, for which traditionally the property layer is used. There isn't much left that needs this bridge. This commit changes a bunch of options (which also have a property implementation) to use option change notifications instead. Many of the properties are still left, but perform unrelated functions like OSD formatting. This should be mostly compatible. There may be some subtle behavior changes. For example, "hwdec" and "record-file" do not check for changes anymore before applying them, so writing the current value to them suddenly does something, while it was ignored before. DVB changes untested, but should work.
2019-11-25 00:26:36 +01:00
ao_c->delay = mpctx->opts->audio_delay;
audio: refactor: separate audio init and filter/output init Before this commit, reinit_audio_chain() did 2 things: create all the management data structures and initialize the decoder, and handling lazy filter/output init (as well as dealing with format changes). For the second purpose, it could be called multiple times (even though it wasn't really idempotent). This was pretty weird, so make them separate functions. The new function is actually idempotent too. It also turns out the reinit functions don't have to call themselves recursively for the spdif PCM fallback.
2016-01-29 22:44:35 +01:00
audio: refactor how data is passed to AO This replaces the two buffers (ao_chain.ao_buffer in the core, and buffer_state.buffers in the AO) with a single queue. Instead of having a byte based buffer, the queue is simply a list of audio frames, as output by the decoder. This should make dataflow simpler and reduce copying. It also attempts to simplify fill_audio_out_buffers(), the function I always hated most, because it's full of subtle and buggy logic. Unfortunately, I got assaulted by corner cases, dumb features (attempt at seamless looping, really?), and other crap, so it got pretty complicated again. fill_audio_out_buffers() is still full of subtle and buggy logic. Maybe it got worse. On the other hand, maybe there really is some progress. Who knows. Originally, the data flow parts was meant to be in f_output_chain, but due to tricky interactions with the playloop code, it's now in the dummy filter in audio.c. At least this improves the way the audio PTS is passed to the encoder in encoding mode. Now it attempts to pass frames directly, along with the pts, which should minimize timestamp problems. But to be honest, encoder mode is one big kludge that shouldn't exist in this way. This commit should be considered pre-alpha code. There are lots of bugs still hiding.
2020-08-28 20:23:54 +02:00
ao_c->ao_filter = mp_filter_create(mpctx->filter_root, &ao_filter);
if (!ao_c->filter || !ao_c->ao_filter)
goto init_error;
ao_c->ao_filter->priv = ao_c;
mp_filter_add_pin(ao_c->ao_filter, MP_PIN_IN, "in");
mp_pin_connect(ao_c->ao_filter->pins[0], ao_c->filter->f->pins[1]);
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 23:08:48 +02:00
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 23:19:56 +01:00
ao_c->track = track;
track->ao_c = ao_c;
if (!init_audio_decoder(mpctx, track))
goto init_error;
audio: move to decoder wrapper Use the decoder wrapper that was introduced for video. This removes all code duplication the old audio decoder wrapper had with the video code. (The audio wrapper was copy pasted from the video one over a decade ago, and has been kept in sync ever since by the power of copy&paste. Since the original copy&paste was possibly done by someone who did not answer to the LGPL relicensing, this should also remove all doubts about whether any of this code is left, since we now completely remove any code that could possibly have been based on it.) There is some complication with spdif handling, and a minor behavior change (it will restrict the list of codecs to spdif if spdif is to be used), but there should not be any difference in practice.
2018-01-29 06:18:33 +01:00
ao_c->dec_src = track->dec->f->pins[0];
mp_pin_connect(ao_c->filter->f->pins[0], ao_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 23:19:56 +01:00
}
audio: refactor: separate audio init and filter/output init Before this commit, reinit_audio_chain() did 2 things: create all the management data structures and initialize the decoder, and handling lazy filter/output init (as well as dealing with format changes). For the second purpose, it could be called multiple times (even though it wasn't really idempotent). This was pretty weird, so make them separate functions. The new function is actually idempotent too. It also turns out the reinit functions don't have to call themselves recursively for the spdif PCM fallback.
2016-01-29 22:44:35 +01:00
reset_audio_state(mpctx);
audio: rewrite filtering glue code Use the new filtering code for audio too.
2018-01-18 14:44:20 +01:00
if (recreate_audio_filters(mpctx) < 0)
goto init_error;
Revert "audio: set audio chain ao on reinit" This reverts commit 3239e41277173bace5ecc2a22910c4660642429c. I'm fairly sure this is wrong, and my next commit should fix it properly. I'm not really sure, though. Normally, the AO is set again by reinit_audio_filters_and_output() after the new audio chain has decoded a frame and knows the new format. The reason replaygain (and apparently the thing the reverted commit tried to fix) didn't work is because they work asynchronously to the audio played by the AO (i.e. buggy and hard to fix).
2021-05-07 14:48:52 +02:00
if (mpctx->ao)
audio: fix missing volume update on init and reinit This is never updated after the AO inits, so there are several cases where the volume would stay at 100%, even if it shouldn't. This affects initial volume as well as track switching or switching between files.
2017-12-01 21:16:59 +01:00
audio_update_volume(mpctx);
audio: refactor: separate audio init and filter/output init Before this commit, reinit_audio_chain() did 2 things: create all the management data structures and initialize the decoder, and handling lazy filter/output init (as well as dealing with format changes). For the second purpose, it could be called multiple times (even though it wasn't really idempotent). This was pretty weird, so make them separate functions. The new function is actually idempotent too. It also turns out the reinit functions don't have to call themselves recursively for the spdif PCM fallback.
2016-01-29 22:44:35 +01:00
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 14:24:15 +02:00
mp_wakeup_core(mpctx);
audio: refactor: separate audio init and filter/output init Before this commit, reinit_audio_chain() did 2 things: create all the management data structures and initialize the decoder, and handling lazy filter/output init (as well as dealing with format changes). For the second purpose, it could be called multiple times (even though it wasn't really idempotent). This was pretty weird, so make them separate functions. The new function is actually idempotent too. It also turns out the reinit functions don't have to call themselves recursively for the spdif PCM fallback.
2016-01-29 22:44:35 +01:00
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 22:38:29 +01:00
init_error:
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 19:57:49 +02:00
uninit_audio_chain(mpctx);
uninit_audio_out(mpctx);
player: remove some further current_track dependencies Now it's used for initialization only for audio and video.
2016-02-05 23:41:44 +01:00
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 22:38:29 +01:00
}
audio: refactor how data is passed to AO This replaces the two buffers (ao_chain.ao_buffer in the core, and buffer_state.buffers in the AO) with a single queue. Instead of having a byte based buffer, the queue is simply a list of audio frames, as output by the decoder. This should make dataflow simpler and reduce copying. It also attempts to simplify fill_audio_out_buffers(), the function I always hated most, because it's full of subtle and buggy logic. Unfortunately, I got assaulted by corner cases, dumb features (attempt at seamless looping, really?), and other crap, so it got pretty complicated again. fill_audio_out_buffers() is still full of subtle and buggy logic. Maybe it got worse. On the other hand, maybe there really is some progress. Who knows. Originally, the data flow parts was meant to be in f_output_chain, but due to tricky interactions with the playloop code, it's now in the dummy filter in audio.c. At least this improves the way the audio PTS is passed to the encoder in encoding mode. Now it attempts to pass frames directly, along with the pts, which should minimize timestamp problems. But to be honest, encoder mode is one big kludge that shouldn't exist in this way. This commit should be considered pre-alpha code. There are lots of bugs still hiding.
2020-08-28 20:23:54 +02:00
// Return pts value corresponding to the start point of audio written to the
// ao queue so far.
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 22:38:29 +01:00
double written_audio_pts(struct MPContext *mpctx)
{
audio: refactor how data is passed to AO This replaces the two buffers (ao_chain.ao_buffer in the core, and buffer_state.buffers in the AO) with a single queue. Instead of having a byte based buffer, the queue is simply a list of audio frames, as output by the decoder. This should make dataflow simpler and reduce copying. It also attempts to simplify fill_audio_out_buffers(), the function I always hated most, because it's full of subtle and buggy logic. Unfortunately, I got assaulted by corner cases, dumb features (attempt at seamless looping, really?), and other crap, so it got pretty complicated again. fill_audio_out_buffers() is still full of subtle and buggy logic. Maybe it got worse. On the other hand, maybe there really is some progress. Who knows. Originally, the data flow parts was meant to be in f_output_chain, but due to tricky interactions with the playloop code, it's now in the dummy filter in audio.c. At least this improves the way the audio PTS is passed to the encoder in encoding mode. Now it attempts to pass frames directly, along with the pts, which should minimize timestamp problems. But to be honest, encoder mode is one big kludge that shouldn't exist in this way. This commit should be considered pre-alpha code. There are lots of bugs still hiding.
2020-08-28 20:23:54 +02:00
return mpctx->ao_chain ? mpctx->ao_chain->last_out_pts : MP_NOPTS_VALUE;
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 22:38:29 +01:00
}
// Return pts value corresponding to currently playing audio.
double playing_audio_pts(struct MPContext *mpctx)
{
double pts = written_audio_pts(mpctx);
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 20:40:43 +02:00
if (pts == MP_NOPTS_VALUE || !mpctx->ao)
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 22:38:29 +01:00
return pts;
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 18:40:16 +02:00
return pts - mpctx->audio_speed * ao_get_delay(mpctx->ao);
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 22:38:29 +01:00
}
audio: refactor how data is passed to AO This replaces the two buffers (ao_chain.ao_buffer in the core, and buffer_state.buffers in the AO) with a single queue. Instead of having a byte based buffer, the queue is simply a list of audio frames, as output by the decoder. This should make dataflow simpler and reduce copying. It also attempts to simplify fill_audio_out_buffers(), the function I always hated most, because it's full of subtle and buggy logic. Unfortunately, I got assaulted by corner cases, dumb features (attempt at seamless looping, really?), and other crap, so it got pretty complicated again. fill_audio_out_buffers() is still full of subtle and buggy logic. Maybe it got worse. On the other hand, maybe there really is some progress. Who knows. Originally, the data flow parts was meant to be in f_output_chain, but due to tricky interactions with the playloop code, it's now in the dummy filter in audio.c. At least this improves the way the audio PTS is passed to the encoder in encoding mode. Now it attempts to pass frames directly, along with the pts, which should minimize timestamp problems. But to be honest, encoder mode is one big kludge that shouldn't exist in this way. This commit should be considered pre-alpha code. There are lots of bugs still hiding.
2020-08-28 20:23:54 +02:00
// This garbage is needed for untimed AOs. These consume audio infinitely fast,
// so try keeping approximate A/V sync by blocking audio transfer as needed.
static void update_throttle(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 22:38:29 +01:00
{
audio: refactor how data is passed to AO This replaces the two buffers (ao_chain.ao_buffer in the core, and buffer_state.buffers in the AO) with a single queue. Instead of having a byte based buffer, the queue is simply a list of audio frames, as output by the decoder. This should make dataflow simpler and reduce copying. It also attempts to simplify fill_audio_out_buffers(), the function I always hated most, because it's full of subtle and buggy logic. Unfortunately, I got assaulted by corner cases, dumb features (attempt at seamless looping, really?), and other crap, so it got pretty complicated again. fill_audio_out_buffers() is still full of subtle and buggy logic. Maybe it got worse. On the other hand, maybe there really is some progress. Who knows. Originally, the data flow parts was meant to be in f_output_chain, but due to tricky interactions with the playloop code, it's now in the dummy filter in audio.c. At least this improves the way the audio PTS is passed to the encoder in encoding mode. Now it attempts to pass frames directly, along with the pts, which should minimize timestamp problems. But to be honest, encoder mode is one big kludge that shouldn't exist in this way. This commit should be considered pre-alpha code. There are lots of bugs still hiding.
2020-08-28 20:23:54 +02:00
struct ao_chain *ao_c = mpctx->ao_chain;
bool new_throttle = mpctx->audio_status == STATUS_PLAYING &&
mpctx->delay > 0 && ao_c && ao_c->ao &&
ao_untimed(ao_c->ao) &&
mpctx->video_status != STATUS_EOF;
if (ao_c && new_throttle != ao_c->untimed_throttle) {
ao_c->untimed_throttle = new_throttle;
mp_wakeup_core(mpctx);
mp_filter_wakeup(ao_c->ao_filter);
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 22:38:29 +01:00
}
}
audio: refactor how data is passed to AO This replaces the two buffers (ao_chain.ao_buffer in the core, and buffer_state.buffers in the AO) with a single queue. Instead of having a byte based buffer, the queue is simply a list of audio frames, as output by the decoder. This should make dataflow simpler and reduce copying. It also attempts to simplify fill_audio_out_buffers(), the function I always hated most, because it's full of subtle and buggy logic. Unfortunately, I got assaulted by corner cases, dumb features (attempt at seamless looping, really?), and other crap, so it got pretty complicated again. fill_audio_out_buffers() is still full of subtle and buggy logic. Maybe it got worse. On the other hand, maybe there really is some progress. Who knows. Originally, the data flow parts was meant to be in f_output_chain, but due to tricky interactions with the playloop code, it's now in the dummy filter in audio.c. At least this improves the way the audio PTS is passed to the encoder in encoding mode. Now it attempts to pass frames directly, along with the pts, which should minimize timestamp problems. But to be honest, encoder mode is one big kludge that shouldn't exist in this way. This commit should be considered pre-alpha code. There are lots of bugs still hiding.
2020-08-28 20:23:54 +02:00
static void ao_process(struct mp_filter *f)
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 22:38:29 +01:00
{
audio: refactor how data is passed to AO This replaces the two buffers (ao_chain.ao_buffer in the core, and buffer_state.buffers in the AO) with a single queue. Instead of having a byte based buffer, the queue is simply a list of audio frames, as output by the decoder. This should make dataflow simpler and reduce copying. It also attempts to simplify fill_audio_out_buffers(), the function I always hated most, because it's full of subtle and buggy logic. Unfortunately, I got assaulted by corner cases, dumb features (attempt at seamless looping, really?), and other crap, so it got pretty complicated again. fill_audio_out_buffers() is still full of subtle and buggy logic. Maybe it got worse. On the other hand, maybe there really is some progress. Who knows. Originally, the data flow parts was meant to be in f_output_chain, but due to tricky interactions with the playloop code, it's now in the dummy filter in audio.c. At least this improves the way the audio PTS is passed to the encoder in encoding mode. Now it attempts to pass frames directly, along with the pts, which should minimize timestamp problems. But to be honest, encoder mode is one big kludge that shouldn't exist in this way. This commit should be considered pre-alpha code. There are lots of bugs still hiding.
2020-08-28 20:23:54 +02:00
struct ao_chain *ao_c = f->priv;
struct MPContext *mpctx = ao_c->mpctx;
audio: rewrite filtering glue code Use the new filtering code for audio too.
2018-01-18 14:44:20 +01:00
audio: refactor how data is passed to AO This replaces the two buffers (ao_chain.ao_buffer in the core, and buffer_state.buffers in the AO) with a single queue. Instead of having a byte based buffer, the queue is simply a list of audio frames, as output by the decoder. This should make dataflow simpler and reduce copying. It also attempts to simplify fill_audio_out_buffers(), the function I always hated most, because it's full of subtle and buggy logic. Unfortunately, I got assaulted by corner cases, dumb features (attempt at seamless looping, really?), and other crap, so it got pretty complicated again. fill_audio_out_buffers() is still full of subtle and buggy logic. Maybe it got worse. On the other hand, maybe there really is some progress. Who knows. Originally, the data flow parts was meant to be in f_output_chain, but due to tricky interactions with the playloop code, it's now in the dummy filter in audio.c. At least this improves the way the audio PTS is passed to the encoder in encoding mode. Now it attempts to pass frames directly, along with the pts, which should minimize timestamp problems. But to be honest, encoder mode is one big kludge that shouldn't exist in this way. This commit should be considered pre-alpha code. There are lots of bugs still hiding.
2020-08-28 20:23:54 +02:00
if (!ao_c->queue_filter) {
// This will eventually lead to the creation of the AO + queue, due
// to how f_output_chain and AO management works.
mp_pin_out_request_data(f->ppins[0]);
player: fix another nightmarish corner case Pretty much fuck this shit.
2020-10-08 00:36:41 +02:00
// Check for EOF with no data case, which is a mess because everything
// hates us.
struct mp_frame frame = mp_pin_out_read(f->ppins[0]);
if (frame.type == MP_FRAME_EOF) {
MP_VERBOSE(mpctx, "got EOF with no data before it\n");
ao_c->out_eof = true;
mpctx->audio_status = STATUS_DRAINING;
mp_wakeup_core(mpctx);
} else if (frame.type) {
mp_pin_out_unread(f->ppins[0], frame);
}
audio: refactor how data is passed to AO This replaces the two buffers (ao_chain.ao_buffer in the core, and buffer_state.buffers in the AO) with a single queue. Instead of having a byte based buffer, the queue is simply a list of audio frames, as output by the decoder. This should make dataflow simpler and reduce copying. It also attempts to simplify fill_audio_out_buffers(), the function I always hated most, because it's full of subtle and buggy logic. Unfortunately, I got assaulted by corner cases, dumb features (attempt at seamless looping, really?), and other crap, so it got pretty complicated again. fill_audio_out_buffers() is still full of subtle and buggy logic. Maybe it got worse. On the other hand, maybe there really is some progress. Who knows. Originally, the data flow parts was meant to be in f_output_chain, but due to tricky interactions with the playloop code, it's now in the dummy filter in audio.c. At least this improves the way the audio PTS is passed to the encoder in encoding mode. Now it attempts to pass frames directly, along with the pts, which should minimize timestamp problems. But to be honest, encoder mode is one big kludge that shouldn't exist in this way. This commit should be considered pre-alpha code. There are lots of bugs still hiding.
2020-08-28 20:23:54 +02:00
return;
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 20:40:43 +02:00
}
audio: fix typo
2021-12-03 23:04:16 +01:00
// Due to mp_async_queue_set_notifier() this function is called when the
audio: refactor how data is passed to AO This replaces the two buffers (ao_chain.ao_buffer in the core, and buffer_state.buffers in the AO) with a single queue. Instead of having a byte based buffer, the queue is simply a list of audio frames, as output by the decoder. This should make dataflow simpler and reduce copying. It also attempts to simplify fill_audio_out_buffers(), the function I always hated most, because it's full of subtle and buggy logic. Unfortunately, I got assaulted by corner cases, dumb features (attempt at seamless looping, really?), and other crap, so it got pretty complicated again. fill_audio_out_buffers() is still full of subtle and buggy logic. Maybe it got worse. On the other hand, maybe there really is some progress. Who knows. Originally, the data flow parts was meant to be in f_output_chain, but due to tricky interactions with the playloop code, it's now in the dummy filter in audio.c. At least this improves the way the audio PTS is passed to the encoder in encoding mode. Now it attempts to pass frames directly, along with the pts, which should minimize timestamp problems. But to be honest, encoder mode is one big kludge that shouldn't exist in this way. This commit should be considered pre-alpha code. There are lots of bugs still hiding.
2020-08-28 20:23:54 +02:00
// queue becomes full. This affects state changes in the normal playloop,
// so wake it up. But avoid redundant wakeups during normal playback.
if (mpctx->audio_status != STATUS_PLAYING &&
mp_async_queue_is_full(ao_c->ao_queue))
mp_wakeup_core(mpctx);
audio: log A/V initial sync status
2019-11-17 01:17:34 +01:00
audio: refactor how data is passed to AO This replaces the two buffers (ao_chain.ao_buffer in the core, and buffer_state.buffers in the AO) with a single queue. Instead of having a byte based buffer, the queue is simply a list of audio frames, as output by the decoder. This should make dataflow simpler and reduce copying. It also attempts to simplify fill_audio_out_buffers(), the function I always hated most, because it's full of subtle and buggy logic. Unfortunately, I got assaulted by corner cases, dumb features (attempt at seamless looping, really?), and other crap, so it got pretty complicated again. fill_audio_out_buffers() is still full of subtle and buggy logic. Maybe it got worse. On the other hand, maybe there really is some progress. Who knows. Originally, the data flow parts was meant to be in f_output_chain, but due to tricky interactions with the playloop code, it's now in the dummy filter in audio.c. At least this improves the way the audio PTS is passed to the encoder in encoding mode. Now it attempts to pass frames directly, along with the pts, which should minimize timestamp problems. But to be honest, encoder mode is one big kludge that shouldn't exist in this way. This commit should be considered pre-alpha code. There are lots of bugs still hiding.
2020-08-28 20:23:54 +02:00
if (mpctx->audio_status == STATUS_SYNCING && !ao_c->start_pts_known)
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 22:38:29 +01:00
audio: refactor how data is passed to AO This replaces the two buffers (ao_chain.ao_buffer in the core, and buffer_state.buffers in the AO) with a single queue. Instead of having a byte based buffer, the queue is simply a list of audio frames, as output by the decoder. This should make dataflow simpler and reduce copying. It also attempts to simplify fill_audio_out_buffers(), the function I always hated most, because it's full of subtle and buggy logic. Unfortunately, I got assaulted by corner cases, dumb features (attempt at seamless looping, really?), and other crap, so it got pretty complicated again. fill_audio_out_buffers() is still full of subtle and buggy logic. Maybe it got worse. On the other hand, maybe there really is some progress. Who knows. Originally, the data flow parts was meant to be in f_output_chain, but due to tricky interactions with the playloop code, it's now in the dummy filter in audio.c. At least this improves the way the audio PTS is passed to the encoder in encoding mode. Now it attempts to pass frames directly, along with the pts, which should minimize timestamp problems. But to be honest, encoder mode is one big kludge that shouldn't exist in this way. This commit should be considered pre-alpha code. There are lots of bugs still hiding.
2020-08-28 20:23:54 +02:00
if (ao_c->untimed_throttle)
return;
audio: refactor: work towards unentangling audio decoding and filtering Similar to the video path. dec_audio.c now handles decoding only. It also looks very similar to dec_video.c, and actually contains some of the rewritten code from it. (A further goal might be unifying the decoders, I guess.) High potential for regressions.
2016-01-21 22:10:15 +01:00
audio: refactor how data is passed to AO This replaces the two buffers (ao_chain.ao_buffer in the core, and buffer_state.buffers in the AO) with a single queue. Instead of having a byte based buffer, the queue is simply a list of audio frames, as output by the decoder. This should make dataflow simpler and reduce copying. It also attempts to simplify fill_audio_out_buffers(), the function I always hated most, because it's full of subtle and buggy logic. Unfortunately, I got assaulted by corner cases, dumb features (attempt at seamless looping, really?), and other crap, so it got pretty complicated again. fill_audio_out_buffers() is still full of subtle and buggy logic. Maybe it got worse. On the other hand, maybe there really is some progress. Who knows. Originally, the data flow parts was meant to be in f_output_chain, but due to tricky interactions with the playloop code, it's now in the dummy filter in audio.c. At least this improves the way the audio PTS is passed to the encoder in encoding mode. Now it attempts to pass frames directly, along with the pts, which should minimize timestamp problems. But to be honest, encoder mode is one big kludge that shouldn't exist in this way. This commit should be considered pre-alpha code. There are lots of bugs still hiding.
2020-08-28 20:23:54 +02:00
if (!mp_pin_can_transfer_data(ao_c->queue_filter->pins[0], f->ppins[0]))
return;
audio: improve aspects of EOF handling The code actually kept going out of EOF mode into resync mode back into EOF mode when the playloop had to wait after an audio EOF caused by the endpts. This would break seamless looping (as added by the next commit). Apply endpts earlier, to ensure the filter_audio() function always returns AD_EOF in this case. The idiotic ao_buffer makes this an amazing pain in the ass.
2016-08-18 20:38:09 +02:00
audio: refactor how data is passed to AO This replaces the two buffers (ao_chain.ao_buffer in the core, and buffer_state.buffers in the AO) with a single queue. Instead of having a byte based buffer, the queue is simply a list of audio frames, as output by the decoder. This should make dataflow simpler and reduce copying. It also attempts to simplify fill_audio_out_buffers(), the function I always hated most, because it's full of subtle and buggy logic. Unfortunately, I got assaulted by corner cases, dumb features (attempt at seamless looping, really?), and other crap, so it got pretty complicated again. fill_audio_out_buffers() is still full of subtle and buggy logic. Maybe it got worse. On the other hand, maybe there really is some progress. Who knows. Originally, the data flow parts was meant to be in f_output_chain, but due to tricky interactions with the playloop code, it's now in the dummy filter in audio.c. At least this improves the way the audio PTS is passed to the encoder in encoding mode. Now it attempts to pass frames directly, along with the pts, which should minimize timestamp problems. But to be honest, encoder mode is one big kludge that shouldn't exist in this way. This commit should be considered pre-alpha code. There are lots of bugs still hiding.
2020-08-28 20:23:54 +02:00
struct mp_frame frame = mp_pin_out_read(f->ppins[0]);
if (frame.type == MP_FRAME_AUDIO) {
struct mp_aframe *af = frame.data;
audio: improve aspects of EOF handling The code actually kept going out of EOF mode into resync mode back into EOF mode when the playloop had to wait after an audio EOF caused by the endpts. This would break seamless looping (as added by the next commit). Apply endpts earlier, to ensure the filter_audio() function always returns AD_EOF in this case. The idiotic ao_buffer makes this an amazing pain in the ass.
2016-08-18 20:38:09 +02:00
audio: refactor how data is passed to AO This replaces the two buffers (ao_chain.ao_buffer in the core, and buffer_state.buffers in the AO) with a single queue. Instead of having a byte based buffer, the queue is simply a list of audio frames, as output by the decoder. This should make dataflow simpler and reduce copying. It also attempts to simplify fill_audio_out_buffers(), the function I always hated most, because it's full of subtle and buggy logic. Unfortunately, I got assaulted by corner cases, dumb features (attempt at seamless looping, really?), and other crap, so it got pretty complicated again. fill_audio_out_buffers() is still full of subtle and buggy logic. Maybe it got worse. On the other hand, maybe there really is some progress. Who knows. Originally, the data flow parts was meant to be in f_output_chain, but due to tricky interactions with the playloop code, it's now in the dummy filter in audio.c. At least this improves the way the audio PTS is passed to the encoder in encoding mode. Now it attempts to pass frames directly, along with the pts, which should minimize timestamp problems. But to be honest, encoder mode is one big kludge that shouldn't exist in this way. This commit should be considered pre-alpha code. There are lots of bugs still hiding.
2020-08-28 20:23:54 +02:00
double endpts = get_play_end_pts(mpctx);
audio: improve aspects of EOF handling The code actually kept going out of EOF mode into resync mode back into EOF mode when the playloop had to wait after an audio EOF caused by the endpts. This would break seamless looping (as added by the next commit). Apply endpts earlier, to ensure the filter_audio() function always returns AD_EOF in this case. The idiotic ao_buffer makes this an amazing pain in the ass.
2016-08-18 20:38:09 +02:00
if (endpts != MP_NOPTS_VALUE) {
audio: refactor how data is passed to AO This replaces the two buffers (ao_chain.ao_buffer in the core, and buffer_state.buffers in the AO) with a single queue. Instead of having a byte based buffer, the queue is simply a list of audio frames, as output by the decoder. This should make dataflow simpler and reduce copying. It also attempts to simplify fill_audio_out_buffers(), the function I always hated most, because it's full of subtle and buggy logic. Unfortunately, I got assaulted by corner cases, dumb features (attempt at seamless looping, really?), and other crap, so it got pretty complicated again. fill_audio_out_buffers() is still full of subtle and buggy logic. Maybe it got worse. On the other hand, maybe there really is some progress. Who knows. Originally, the data flow parts was meant to be in f_output_chain, but due to tricky interactions with the playloop code, it's now in the dummy filter in audio.c. At least this improves the way the audio PTS is passed to the encoder in encoding mode. Now it attempts to pass frames directly, along with the pts, which should minimize timestamp problems. But to be honest, encoder mode is one big kludge that shouldn't exist in this way. This commit should be considered pre-alpha code. There are lots of bugs still hiding.
2020-08-28 20:23:54 +02:00
endpts *= mpctx->play_dir;
// Avoid decoding and discarding the entire rest of the file.
if (mp_aframe_get_pts(af) >= endpts) {
mp_pin_out_unread(f->ppins[0], frame);
if (!ao_c->out_eof) {
ao_c->out_eof = true;
mp_pin_in_write(ao_c->queue_filter->pins[0], MP_EOF_FRAME);
}
return;
player: fix --end with large values Causea a simple integer overflow. Fixes #4650.
2017-08-08 14:11:58 +02:00
}
audio: improve aspects of EOF handling The code actually kept going out of EOF mode into resync mode back into EOF mode when the playloop had to wait after an audio EOF caused by the endpts. This would break seamless looping (as added by the next commit). Apply endpts earlier, to ensure the filter_audio() function always returns AD_EOF in this case. The idiotic ao_buffer makes this an amazing pain in the ass.
2016-08-18 20:38:09 +02:00
}
audio: refactor how data is passed to AO This replaces the two buffers (ao_chain.ao_buffer in the core, and buffer_state.buffers in the AO) with a single queue. Instead of having a byte based buffer, the queue is simply a list of audio frames, as output by the decoder. This should make dataflow simpler and reduce copying. It also attempts to simplify fill_audio_out_buffers(), the function I always hated most, because it's full of subtle and buggy logic. Unfortunately, I got assaulted by corner cases, dumb features (attempt at seamless looping, really?), and other crap, so it got pretty complicated again. fill_audio_out_buffers() is still full of subtle and buggy logic. Maybe it got worse. On the other hand, maybe there really is some progress. Who knows. Originally, the data flow parts was meant to be in f_output_chain, but due to tricky interactions with the playloop code, it's now in the dummy filter in audio.c. At least this improves the way the audio PTS is passed to the encoder in encoding mode. Now it attempts to pass frames directly, along with the pts, which should minimize timestamp problems. But to be honest, encoder mode is one big kludge that shouldn't exist in this way. This commit should be considered pre-alpha code. There are lots of bugs still hiding.
2020-08-28 20:23:54 +02:00
double startpts = mpctx->audio_status == STATUS_SYNCING ?
ao_c->start_pts : MP_NOPTS_VALUE;
mp_aframe_clip_timestamps(af, startpts, endpts);
int samples = mp_aframe_get_size(af);
if (!samples) {
mp_filter_internal_mark_progress(f);
mp_frame_unref(&frame);
return;
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 12:48:30 +02:00
}
audio: refactor how data is passed to AO This replaces the two buffers (ao_chain.ao_buffer in the core, and buffer_state.buffers in the AO) with a single queue. Instead of having a byte based buffer, the queue is simply a list of audio frames, as output by the decoder. This should make dataflow simpler and reduce copying. It also attempts to simplify fill_audio_out_buffers(), the function I always hated most, because it's full of subtle and buggy logic. Unfortunately, I got assaulted by corner cases, dumb features (attempt at seamless looping, really?), and other crap, so it got pretty complicated again. fill_audio_out_buffers() is still full of subtle and buggy logic. Maybe it got worse. On the other hand, maybe there really is some progress. Who knows. Originally, the data flow parts was meant to be in f_output_chain, but due to tricky interactions with the playloop code, it's now in the dummy filter in audio.c. At least this improves the way the audio PTS is passed to the encoder in encoding mode. Now it attempts to pass frames directly, along with the pts, which should minimize timestamp problems. But to be honest, encoder mode is one big kludge that shouldn't exist in this way. This commit should be considered pre-alpha code. There are lots of bugs still hiding.
2020-08-28 20:23:54 +02:00
ao_c->out_eof = false;
audio: improve aspects of EOF handling The code actually kept going out of EOF mode into resync mode back into EOF mode when the playloop had to wait after an audio EOF caused by the endpts. This would break seamless looping (as added by the next commit). Apply endpts earlier, to ensure the filter_audio() function always returns AD_EOF in this case. The idiotic ao_buffer makes this an amazing pain in the ass.
2016-08-18 20:38:09 +02:00
audio: refactor how data is passed to AO This replaces the two buffers (ao_chain.ao_buffer in the core, and buffer_state.buffers in the AO) with a single queue. Instead of having a byte based buffer, the queue is simply a list of audio frames, as output by the decoder. This should make dataflow simpler and reduce copying. It also attempts to simplify fill_audio_out_buffers(), the function I always hated most, because it's full of subtle and buggy logic. Unfortunately, I got assaulted by corner cases, dumb features (attempt at seamless looping, really?), and other crap, so it got pretty complicated again. fill_audio_out_buffers() is still full of subtle and buggy logic. Maybe it got worse. On the other hand, maybe there really is some progress. Who knows. Originally, the data flow parts was meant to be in f_output_chain, but due to tricky interactions with the playloop code, it's now in the dummy filter in audio.c. At least this improves the way the audio PTS is passed to the encoder in encoding mode. Now it attempts to pass frames directly, along with the pts, which should minimize timestamp problems. But to be honest, encoder mode is one big kludge that shouldn't exist in this way. This commit should be considered pre-alpha code. There are lots of bugs still hiding.
2020-08-28 20:23:54 +02:00
if (mpctx->audio_status == STATUS_DRAINING ||
mpctx->audio_status == STATUS_EOF)
{
// If a new frame comes decoder/filter EOF, we should preferably
// call get_sync_pts() again, which (at least in obscure situations)
// may require us to wait a while until the sync PTS is known. Our
// code sucks and can't deal with that, so jump through a hoop to
// get things done in the correct order.
mp_pin_out_unread(f->ppins[0], frame);
ao_c->start_pts_known = false;
mpctx->audio_status = STATUS_SYNCING;
mp_wakeup_core(mpctx);
MP_VERBOSE(mpctx, "new audio frame after EOF\n");
return;
audio: improve aspects of EOF handling The code actually kept going out of EOF mode into resync mode back into EOF mode when the playloop had to wait after an audio EOF caused by the endpts. This would break seamless looping (as added by the next commit). Apply endpts earlier, to ensure the filter_audio() function always returns AD_EOF in this case. The idiotic ao_buffer makes this an amazing pain in the ass.
2016-08-18 20:38:09 +02:00
}
audio: refactor how data is passed to AO This replaces the two buffers (ao_chain.ao_buffer in the core, and buffer_state.buffers in the AO) with a single queue. Instead of having a byte based buffer, the queue is simply a list of audio frames, as output by the decoder. This should make dataflow simpler and reduce copying. It also attempts to simplify fill_audio_out_buffers(), the function I always hated most, because it's full of subtle and buggy logic. Unfortunately, I got assaulted by corner cases, dumb features (attempt at seamless looping, really?), and other crap, so it got pretty complicated again. fill_audio_out_buffers() is still full of subtle and buggy logic. Maybe it got worse. On the other hand, maybe there really is some progress. Who knows. Originally, the data flow parts was meant to be in f_output_chain, but due to tricky interactions with the playloop code, it's now in the dummy filter in audio.c. At least this improves the way the audio PTS is passed to the encoder in encoding mode. Now it attempts to pass frames directly, along with the pts, which should minimize timestamp problems. But to be honest, encoder mode is one big kludge that shouldn't exist in this way. This commit should be considered pre-alpha code. There are lots of bugs still hiding.
2020-08-28 20:23:54 +02:00
mpctx->shown_aframes += samples;
double real_samplerate = mp_aframe_get_rate(af) / mpctx->audio_speed;
mpctx->delay += samples / real_samplerate;
ao_c->last_out_pts = mp_aframe_end_pts(af);
update_throttle(mpctx);
// Gapless case: the AO is still playing from previous file. It makes
// no sense to wait, and in fact the "full queue" event we're waiting
// for may never happen, so start immediately.
// If the new audio starts "later" (big video sync offset), transfer
// of data is stopped somewhere else.
if (mpctx->audio_status == STATUS_SYNCING && ao_is_playing(ao_c->ao)) {
mpctx->audio_status = STATUS_READY;
mp_wakeup_core(mpctx);
MP_VERBOSE(mpctx, "previous audio still playing; continuing\n");
}
audio: refactor: work towards unentangling audio decoding and filtering Similar to the video path. dec_audio.c now handles decoding only. It also looks very similar to dec_video.c, and actually contains some of the rewritten code from it. (A further goal might be unifying the decoders, I guess.) High potential for regressions.
2016-01-21 22:10:15 +01:00
audio: refactor how data is passed to AO This replaces the two buffers (ao_chain.ao_buffer in the core, and buffer_state.buffers in the AO) with a single queue. Instead of having a byte based buffer, the queue is simply a list of audio frames, as output by the decoder. This should make dataflow simpler and reduce copying. It also attempts to simplify fill_audio_out_buffers(), the function I always hated most, because it's full of subtle and buggy logic. Unfortunately, I got assaulted by corner cases, dumb features (attempt at seamless looping, really?), and other crap, so it got pretty complicated again. fill_audio_out_buffers() is still full of subtle and buggy logic. Maybe it got worse. On the other hand, maybe there really is some progress. Who knows. Originally, the data flow parts was meant to be in f_output_chain, but due to tricky interactions with the playloop code, it's now in the dummy filter in audio.c. At least this improves the way the audio PTS is passed to the encoder in encoding mode. Now it attempts to pass frames directly, along with the pts, which should minimize timestamp problems. But to be honest, encoder mode is one big kludge that shouldn't exist in this way. This commit should be considered pre-alpha code. There are lots of bugs still hiding.
2020-08-28 20:23:54 +02:00
mp_pin_in_write(ao_c->queue_filter->pins[0], frame);
} else if (frame.type == MP_FRAME_EOF) {
MP_VERBOSE(mpctx, "audio filter EOF\n");
audio: refactor: work towards unentangling audio decoding and filtering Similar to the video path. dec_audio.c now handles decoding only. It also looks very similar to dec_video.c, and actually contains some of the rewritten code from it. (A further goal might be unifying the decoders, I guess.) High potential for regressions.
2016-01-21 22:10:15 +01:00
audio: refactor how data is passed to AO This replaces the two buffers (ao_chain.ao_buffer in the core, and buffer_state.buffers in the AO) with a single queue. Instead of having a byte based buffer, the queue is simply a list of audio frames, as output by the decoder. This should make dataflow simpler and reduce copying. It also attempts to simplify fill_audio_out_buffers(), the function I always hated most, because it's full of subtle and buggy logic. Unfortunately, I got assaulted by corner cases, dumb features (attempt at seamless looping, really?), and other crap, so it got pretty complicated again. fill_audio_out_buffers() is still full of subtle and buggy logic. Maybe it got worse. On the other hand, maybe there really is some progress. Who knows. Originally, the data flow parts was meant to be in f_output_chain, but due to tricky interactions with the playloop code, it's now in the dummy filter in audio.c. At least this improves the way the audio PTS is passed to the encoder in encoding mode. Now it attempts to pass frames directly, along with the pts, which should minimize timestamp problems. But to be honest, encoder mode is one big kludge that shouldn't exist in this way. This commit should be considered pre-alpha code. There are lots of bugs still hiding.
2020-08-28 20:23:54 +02:00
ao_c->out_eof = true;
mp_wakeup_core(mpctx);
audio: improve aspects of EOF handling The code actually kept going out of EOF mode into resync mode back into EOF mode when the playloop had to wait after an audio EOF caused by the endpts. This would break seamless looping (as added by the next commit). Apply endpts earlier, to ensure the filter_audio() function always returns AD_EOF in this case. The idiotic ao_buffer makes this an amazing pain in the ass.
2016-08-18 20:38:09 +02:00
audio: refactor how data is passed to AO This replaces the two buffers (ao_chain.ao_buffer in the core, and buffer_state.buffers in the AO) with a single queue. Instead of having a byte based buffer, the queue is simply a list of audio frames, as output by the decoder. This should make dataflow simpler and reduce copying. It also attempts to simplify fill_audio_out_buffers(), the function I always hated most, because it's full of subtle and buggy logic. Unfortunately, I got assaulted by corner cases, dumb features (attempt at seamless looping, really?), and other crap, so it got pretty complicated again. fill_audio_out_buffers() is still full of subtle and buggy logic. Maybe it got worse. On the other hand, maybe there really is some progress. Who knows. Originally, the data flow parts was meant to be in f_output_chain, but due to tricky interactions with the playloop code, it's now in the dummy filter in audio.c. At least this improves the way the audio PTS is passed to the encoder in encoding mode. Now it attempts to pass frames directly, along with the pts, which should minimize timestamp problems. But to be honest, encoder mode is one big kludge that shouldn't exist in this way. This commit should be considered pre-alpha code. There are lots of bugs still hiding.
2020-08-28 20:23:54 +02:00
mp_pin_in_write(ao_c->queue_filter->pins[0], frame);
mp_filter_internal_mark_progress(f);
} else {
mp_frame_unref(&frame);
}
audio: refactor: work towards unentangling audio decoding and filtering Similar to the video path. dec_audio.c now handles decoding only. It also looks very similar to dec_video.c, and actually contains some of the rewritten code from it. (A further goal might be unifying the decoders, I guess.) High potential for regressions.
2016-01-21 22:10:15 +01:00
}
audio: move some fallback handling to common AO reload function Now a reload requested by an AO behaves in exactly the same way as changing an AO-related options (like --audio-channels or --audio-exclusive). This is good for testing and uniform behavior. (You could go as far as saying it's a necessity, because the spotty and obscure AO reload behavior is hard to reproduce and thus hard to test at all.)
2016-10-05 16:42:43 +02:00
void reload_audio_output(struct MPContext *mpctx)
{
if (!mpctx->ao)
return;
ao_reset(mpctx->ao);
uninit_audio_out(mpctx);
reinit_audio_filters(mpctx); // mostly to issue refresh seek
audio: move to decoder wrapper Use the decoder wrapper that was introduced for video. This removes all code duplication the old audio decoder wrapper had with the video code. (The audio wrapper was copy pasted from the video one over a decade ago, and has been kept in sync ever since by the power of copy&paste. Since the original copy&paste was possibly done by someone who did not answer to the LGPL relicensing, this should also remove all doubts about whether any of this code is left, since we now completely remove any code that could possibly have been based on it.) There is some complication with spdif handling, and a minor behavior change (it will restrict the list of codecs to spdif if spdif is to be used), but there should not be any difference in practice.
2018-01-29 06:18:33 +01:00
struct ao_chain *ao_c = mpctx->ao_chain;
if (ao_c) {
reset_audio_state(mpctx);
mp_output_chain_reset_harder(ao_c->filter);
}
audio: move some fallback handling to common AO reload function Now a reload requested by an AO behaves in exactly the same way as changing an AO-related options (like --audio-channels or --audio-exclusive). This is good for testing and uniform behavior. (You could go as far as saying it's a necessity, because the spotty and obscure AO reload behavior is hard to reproduce and thus hard to test at all.)
2016-10-05 16:42:43 +02:00
// Whether we can use spdif might have changed. If we failed to use spdif
// in the previous initialization, try it with spdif again (we'll fallback
// to PCM again if necessary).
audio: move to decoder wrapper Use the decoder wrapper that was introduced for video. This removes all code duplication the old audio decoder wrapper had with the video code. (The audio wrapper was copy pasted from the video one over a decade ago, and has been kept in sync ever since by the power of copy&paste. Since the original copy&paste was possibly done by someone who did not answer to the LGPL relicensing, this should also remove all doubts about whether any of this code is left, since we now completely remove any code that could possibly have been based on it.) There is some complication with spdif handling, and a minor behavior change (it will restrict the list of codecs to spdif if spdif is to be used), but there should not be any difference in practice.
2018-01-29 06:18:33 +01:00
if (ao_c && ao_c->track) {
struct mp_decoder_wrapper *dec = ao_c->track->dec;
if (dec && ao_c->spdif_failed) {
audio: move some fallback handling to common AO reload function Now a reload requested by an AO behaves in exactly the same way as changing an AO-related options (like --audio-channels or --audio-exclusive). This is good for testing and uniform behavior. (You could go as far as saying it's a necessity, because the spotty and obscure AO reload behavior is hard to reproduce and thus hard to test at all.)
2016-10-05 16:42:43 +02:00
ao_c->spdif_passthrough = true;
ao_c->spdif_failed = false;
f_decoder_wrapper: replace most public fields with setters/getters I may (optionally) move decoding to a separate thread in a future change. It's a bit attractive to move the entire decoder wrapper to there, so if the demuxer has a new packet, it doesn't have to wake up the main thread, and can directly wake up the decoder. (Although that's bullshit, since there's a queue in between, and libavcodec's multi-threaded decoding plays cross-threads ping pong with packets anyway. On the other hand, the main thread would still have to shuffle the packets around, so whatever, just seems like better design.) As preparation, there shouldn't be any mutable state exposed by the wrapper. But there's still a large number of corner-caseish crap, so just use setters/getters for them. This recorder thing will inherently not work, so it'll have to be disabled if threads are used. This is a bit painful, but probably still the right thing. Like speculatively pulling teeth.
2020-02-28 21:17:55 +01:00
mp_decoder_wrapper_set_spdif_flag(ao_c->track->dec, true);
audio: move to decoder wrapper Use the decoder wrapper that was introduced for video. This removes all code duplication the old audio decoder wrapper had with the video code. (The audio wrapper was copy pasted from the video one over a decade ago, and has been kept in sync ever since by the power of copy&paste. Since the original copy&paste was possibly done by someone who did not answer to the LGPL relicensing, this should also remove all doubts about whether any of this code is left, since we now completely remove any code that could possibly have been based on it.) There is some complication with spdif handling, and a minor behavior change (it will restrict the list of codecs to spdif if spdif is to be used), but there should not be any difference in practice.
2018-01-29 06:18:33 +01:00
if (!mp_decoder_wrapper_reinit(dec)) {
audio: move some fallback handling to common AO reload function Now a reload requested by an AO behaves in exactly the same way as changing an AO-related options (like --audio-channels or --audio-exclusive). This is good for testing and uniform behavior. (You could go as far as saying it's a necessity, because the spotty and obscure AO reload behavior is hard to reproduce and thus hard to test at all.)
2016-10-05 16:42:43 +02:00
MP_ERR(mpctx, "Error reinitializing audio.\n");
error_on_track(mpctx, ao_c->track);
}
}
}
mp_wakeup_core(mpctx);
}
audio: refactor how data is passed to AO This replaces the two buffers (ao_chain.ao_buffer in the core, and buffer_state.buffers in the AO) with a single queue. Instead of having a byte based buffer, the queue is simply a list of audio frames, as output by the decoder. This should make dataflow simpler and reduce copying. It also attempts to simplify fill_audio_out_buffers(), the function I always hated most, because it's full of subtle and buggy logic. Unfortunately, I got assaulted by corner cases, dumb features (attempt at seamless looping, really?), and other crap, so it got pretty complicated again. fill_audio_out_buffers() is still full of subtle and buggy logic. Maybe it got worse. On the other hand, maybe there really is some progress. Who knows. Originally, the data flow parts was meant to be in f_output_chain, but due to tricky interactions with the playloop code, it's now in the dummy filter in audio.c. At least this improves the way the audio PTS is passed to the encoder in encoding mode. Now it attempts to pass frames directly, along with the pts, which should minimize timestamp problems. But to be honest, encoder mode is one big kludge that shouldn't exist in this way. This commit should be considered pre-alpha code. There are lots of bugs still hiding.
2020-08-28 20:23:54 +02:00
// Returns audio start pts for seeking or video sync.
// Returns false if PTS is not known yet.
static bool get_sync_pts(struct MPContext *mpctx, double *pts)
{
struct MPOpts *opts = mpctx->opts;
*pts = MP_NOPTS_VALUE;
if (!opts->initial_audio_sync)
return true;
bool sync_to_video = mpctx->vo_chain && mpctx->video_status != STATUS_EOF &&
!mpctx->vo_chain->is_sparse;
if (sync_to_video) {
if (mpctx->video_status < STATUS_READY)
return false; // wait until we know a video PTS
if (mpctx->video_pts != MP_NOPTS_VALUE)
*pts = mpctx->video_pts - opts->audio_delay;
} else if (mpctx->hrseek_active) {
*pts = mpctx->hrseek_pts;
} else {
// If audio-only is enabled mid-stream during playback, sync accordingly.
*pts = mpctx->playback_pts;
}
return true;
}
// Look whether audio can be started yet - if audio has to start some time
// after video.
audio: slightly simplify audio_start_ao() Get rid of an indirection; no behavior change.
2020-09-03 12:22:20 +02:00
// Caller needs to ensure mpctx->restart_complete is OK
void audio_start_ao(struct MPContext *mpctx)
audio: refactor how data is passed to AO This replaces the two buffers (ao_chain.ao_buffer in the core, and buffer_state.buffers in the AO) with a single queue. Instead of having a byte based buffer, the queue is simply a list of audio frames, as output by the decoder. This should make dataflow simpler and reduce copying. It also attempts to simplify fill_audio_out_buffers(), the function I always hated most, because it's full of subtle and buggy logic. Unfortunately, I got assaulted by corner cases, dumb features (attempt at seamless looping, really?), and other crap, so it got pretty complicated again. fill_audio_out_buffers() is still full of subtle and buggy logic. Maybe it got worse. On the other hand, maybe there really is some progress. Who knows. Originally, the data flow parts was meant to be in f_output_chain, but due to tricky interactions with the playloop code, it's now in the dummy filter in audio.c. At least this improves the way the audio PTS is passed to the encoder in encoding mode. Now it attempts to pass frames directly, along with the pts, which should minimize timestamp problems. But to be honest, encoder mode is one big kludge that shouldn't exist in this way. This commit should be considered pre-alpha code. There are lots of bugs still hiding.
2020-08-28 20:23:54 +02:00
{
struct ao_chain *ao_c = mpctx->ao_chain;
if (!ao_c || !ao_c->ao || mpctx->audio_status != STATUS_READY)
return;
double pts = MP_NOPTS_VALUE;
if (!get_sync_pts(mpctx, &pts))
return;
double apts = playing_audio_pts(mpctx); // (basically including mpctx->delay)
if (pts != MP_NOPTS_VALUE && apts != MP_NOPTS_VALUE && pts < apts &&
mpctx->video_status != STATUS_EOF)
{
double diff = (apts - pts) / mpctx->opts->playback_speed;
audio: do not wake up player when waiting for audio state and paused Bullshit.
2020-09-01 21:26:19 +02:00
if (!get_internal_paused(mpctx))
mp_set_timeout(mpctx, diff);
audio: reduce excessive logging of delayed audio start Since this is a messy and fragile mechanism, I want it logged (even if it's somewhat in conflict with the verbose logging policy). On the other hand, it's unconditionally logged on every playloop iteration. So add some nonsense to log it only on progress.
2020-09-03 12:18:42 +02:00
if (mpctx->logged_async_diff != diff) {
MP_VERBOSE(mpctx, "delaying audio start %f vs. %f, diff=%f\n",
apts, pts, diff);
mpctx->logged_async_diff = diff;
}
audio: refactor how data is passed to AO This replaces the two buffers (ao_chain.ao_buffer in the core, and buffer_state.buffers in the AO) with a single queue. Instead of having a byte based buffer, the queue is simply a list of audio frames, as output by the decoder. This should make dataflow simpler and reduce copying. It also attempts to simplify fill_audio_out_buffers(), the function I always hated most, because it's full of subtle and buggy logic. Unfortunately, I got assaulted by corner cases, dumb features (attempt at seamless looping, really?), and other crap, so it got pretty complicated again. fill_audio_out_buffers() is still full of subtle and buggy logic. Maybe it got worse. On the other hand, maybe there really is some progress. Who knows. Originally, the data flow parts was meant to be in f_output_chain, but due to tricky interactions with the playloop code, it's now in the dummy filter in audio.c. At least this improves the way the audio PTS is passed to the encoder in encoding mode. Now it attempts to pass frames directly, along with the pts, which should minimize timestamp problems. But to be honest, encoder mode is one big kludge that shouldn't exist in this way. This commit should be considered pre-alpha code. There are lots of bugs still hiding.
2020-08-28 20:23:54 +02:00
return;
}
MP_VERBOSE(mpctx, "starting audio playback\n");
ao_start(ao_c->ao);
mpctx->audio_status = STATUS_PLAYING;
audio: add two minor log messages This would have made the problem fixed in the previous commit a bit more obvious from the log output.
2021-04-29 17:12:44 +02:00
if (ao_c->out_eof) {
audio: refactor how data is passed to AO This replaces the two buffers (ao_chain.ao_buffer in the core, and buffer_state.buffers in the AO) with a single queue. Instead of having a byte based buffer, the queue is simply a list of audio frames, as output by the decoder. This should make dataflow simpler and reduce copying. It also attempts to simplify fill_audio_out_buffers(), the function I always hated most, because it's full of subtle and buggy logic. Unfortunately, I got assaulted by corner cases, dumb features (attempt at seamless looping, really?), and other crap, so it got pretty complicated again. fill_audio_out_buffers() is still full of subtle and buggy logic. Maybe it got worse. On the other hand, maybe there really is some progress. Who knows. Originally, the data flow parts was meant to be in f_output_chain, but due to tricky interactions with the playloop code, it's now in the dummy filter in audio.c. At least this improves the way the audio PTS is passed to the encoder in encoding mode. Now it attempts to pass frames directly, along with the pts, which should minimize timestamp problems. But to be honest, encoder mode is one big kludge that shouldn't exist in this way. This commit should be considered pre-alpha code. There are lots of bugs still hiding.
2020-08-28 20:23:54 +02:00
mpctx->audio_status = STATUS_DRAINING;
audio: add two minor log messages This would have made the problem fixed in the previous commit a bit more obvious from the log output.
2021-04-29 17:12:44 +02:00
MP_VERBOSE(mpctx, "audio draining\n");
}
audio: refactor how data is passed to AO This replaces the two buffers (ao_chain.ao_buffer in the core, and buffer_state.buffers in the AO) with a single queue. Instead of having a byte based buffer, the queue is simply a list of audio frames, as output by the decoder. This should make dataflow simpler and reduce copying. It also attempts to simplify fill_audio_out_buffers(), the function I always hated most, because it's full of subtle and buggy logic. Unfortunately, I got assaulted by corner cases, dumb features (attempt at seamless looping, really?), and other crap, so it got pretty complicated again. fill_audio_out_buffers() is still full of subtle and buggy logic. Maybe it got worse. On the other hand, maybe there really is some progress. Who knows. Originally, the data flow parts was meant to be in f_output_chain, but due to tricky interactions with the playloop code, it's now in the dummy filter in audio.c. At least this improves the way the audio PTS is passed to the encoder in encoding mode. Now it attempts to pass frames directly, along with the pts, which should minimize timestamp problems. But to be honest, encoder mode is one big kludge that shouldn't exist in this way. This commit should be considered pre-alpha code. There are lots of bugs still hiding.
2020-08-28 20:23:54 +02:00
ao_c->underrun = false;
audio: reduce excessive logging of delayed audio start Since this is a messy and fragile mechanism, I want it logged (even if it's somewhat in conflict with the verbose logging policy). On the other hand, it's unconditionally logged on every playloop iteration. So add some nonsense to log it only on progress.
2020-09-03 12:18:42 +02:00
mpctx->logged_async_diff = -1;
audio: refactor how data is passed to AO This replaces the two buffers (ao_chain.ao_buffer in the core, and buffer_state.buffers in the AO) with a single queue. Instead of having a byte based buffer, the queue is simply a list of audio frames, as output by the decoder. This should make dataflow simpler and reduce copying. It also attempts to simplify fill_audio_out_buffers(), the function I always hated most, because it's full of subtle and buggy logic. Unfortunately, I got assaulted by corner cases, dumb features (attempt at seamless looping, really?), and other crap, so it got pretty complicated again. fill_audio_out_buffers() is still full of subtle and buggy logic. Maybe it got worse. On the other hand, maybe there really is some progress. Who knows. Originally, the data flow parts was meant to be in f_output_chain, but due to tricky interactions with the playloop code, it's now in the dummy filter in audio.c. At least this improves the way the audio PTS is passed to the encoder in encoding mode. Now it attempts to pass frames directly, along with the pts, which should minimize timestamp problems. But to be honest, encoder mode is one big kludge that shouldn't exist in this way. This commit should be considered pre-alpha code. There are lots of bugs still hiding.
2020-08-28 20:23:54 +02:00
mp_wakeup_core(mpctx);
}
player: minor simplification No need to pass endpts down in such a dumb way. Also remove an outdated comment somewhere.
2016-02-27 20:02:51 +01:00
void fill_audio_out_buffers(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 22:38:29 +01:00
{
struct MPOpts *opts = mpctx->opts;
audio: move some fallback handling to common AO reload function Now a reload requested by an AO behaves in exactly the same way as changing an AO-related options (like --audio-channels or --audio-exclusive). This is good for testing and uniform behavior. (You could go as far as saying it's a necessity, because the spotty and obscure AO reload behavior is hard to reproduce and thus hard to test at all.)
2016-10-05 16:42:43 +02:00
if (mpctx->ao && ao_query_and_reset_events(mpctx->ao, AO_EVENT_RELOAD))
reload_audio_output(mpctx);
audio/out: make ao_request_reload() idempotent This is what you would expect. Before this commit, each ao_request_reload() call would just queue a reload command, and then recreate the AO for the number of times the function was called. Instead of sending a command, introduce some sort of event retrieval mechanism. At least for the reload case, use atomics, because we're too lazy to setup an extra mutex.
2014-11-09 09:58:44 +01:00
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 19:42:18 +02:00
if (mpctx->ao && ao_query_and_reset_events(mpctx->ao,
AO_EVENT_INITIAL_UNBLOCK))
ao_unblock(mpctx->ao);
audio: refactor how data is passed to AO This replaces the two buffers (ao_chain.ao_buffer in the core, and buffer_state.buffers in the AO) with a single queue. Instead of having a byte based buffer, the queue is simply a list of audio frames, as output by the decoder. This should make dataflow simpler and reduce copying. It also attempts to simplify fill_audio_out_buffers(), the function I always hated most, because it's full of subtle and buggy logic. Unfortunately, I got assaulted by corner cases, dumb features (attempt at seamless looping, really?), and other crap, so it got pretty complicated again. fill_audio_out_buffers() is still full of subtle and buggy logic. Maybe it got worse. On the other hand, maybe there really is some progress. Who knows. Originally, the data flow parts was meant to be in f_output_chain, but due to tricky interactions with the playloop code, it's now in the dummy filter in audio.c. At least this improves the way the audio PTS is passed to the encoder in encoding mode. Now it attempts to pass frames directly, along with the pts, which should minimize timestamp problems. But to be honest, encoder mode is one big kludge that shouldn't exist in this way. This commit should be considered pre-alpha code. There are lots of bugs still hiding.
2020-08-28 20:23:54 +02:00
update_throttle(mpctx);
audio: move some fallback handling to common AO reload function Now a reload requested by an AO behaves in exactly the same way as changing an AO-related options (like --audio-channels or --audio-exclusive). This is good for testing and uniform behavior. (You could go as far as saying it's a necessity, because the spotty and obscure AO reload behavior is hard to reproduce and thus hard to test at all.)
2016-10-05 16:42:43 +02:00
struct ao_chain *ao_c = mpctx->ao_chain;
audio: refactor: work towards unentangling audio decoding and filtering Similar to the video path. dec_audio.c now handles decoding only. It also looks very similar to dec_video.c, and actually contains some of the rewritten code from it. (A further goal might be unifying the decoders, I guess.) High potential for regressions.
2016-01-21 22:10:15 +01:00
if (!ao_c)
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 23:24:08 +02:00
return;
audio: move decoder context from sh_audio into new struct Move all state that basically changes during decoding or is needed in order to manage decoding itself into a new struct (dec_audio). sh_audio (defined in stheader.h) is supposed to be the audio stream header. This should reflect the file headers for the stream. Putting the decoder context there is strange design, to say the least.
2013-11-23 21:22:17 +01:00
audio: rewrite filtering glue code Use the new filtering code for audio too.
2018-01-18 14:44:20 +01:00
if (ao_c->filter->failed_output_conversion) {
error_on_track(mpctx, ao_c->track);
return;
}
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 12:48:30 +02:00
audio: check ao driver init failure to avoid use after free reinit_audio_filters_and_output function will free mpctx->ao_chain when there is a failure during audio initialization. So modify it to return -1 in case of init failure. The return value is checked to avoid use after free. Reported by Address Sanitizer when manually specifying --ao which triggers "Failed to initialize audio driver" error.
2021-07-13 16:06:52 +02:00
if (ao_c->filter->ao_needs_update) {
if (reinit_audio_filters_and_output(mpctx) < 0)
return;
}
player: add --audio-wait-open options Complements the option added in the previous commit.
2016-08-09 16:26:44 +02:00
audio: move back PTS jump detection to before filter chain The recent changes to player/audio.c moved PTS jump detection to after audio filtering. This was mostly done for convenience, because dataflow between decoder and filters was made "automatic", and jump detection would have to be done as filter. Now move it back to after decoders, again out of convenience. The future direction is to make the dataflow between filters and AO automatic, so this is a bit in the way. Another reason is that speed changes tend to cause jumps - these are legitimate, but get annoying quickly.
2018-02-12 18:00:23 +01:00
if (mpctx->vo_chain && ao_c->track && ao_c->track->dec &&
f_decoder_wrapper: replace most public fields with setters/getters I may (optionally) move decoding to a separate thread in a future change. It's a bit attractive to move the entire decoder wrapper to there, so if the demuxer has a new packet, it doesn't have to wake up the main thread, and can directly wake up the decoder. (Although that's bullshit, since there's a queue in between, and libavcodec's multi-threaded decoding plays cross-threads ping pong with packets anyway. On the other hand, the main thread would still have to shuffle the packets around, so whatever, just seems like better design.) As preparation, there shouldn't be any mutable state exposed by the wrapper. But there's still a large number of corner-caseish crap, so just use setters/getters for them. This recorder thing will inherently not work, so it'll have to be disabled if threads are used. This is a bit painful, but probably still the right thing. Like speculatively pulling teeth.
2020-02-28 21:17:55 +01:00
mp_decoder_wrapper_get_pts_reset(ao_c->track->dec))
audio: move back PTS jump detection to before filter chain The recent changes to player/audio.c moved PTS jump detection to after audio filtering. This was mostly done for convenience, because dataflow between decoder and filters was made "automatic", and jump detection would have to be done as filter. Now move it back to after decoders, again out of convenience. The future direction is to make the dataflow between filters and AO automatic, so this is a bit in the way. Another reason is that speed changes tend to cause jumps - these are legitimate, but get annoying quickly.
2018-02-12 18:00:23 +01:00
{
audio: raise log level of playback reset on audio timestamp Make it a warning. This is such an intrusive and shitty hack (but of course my fault) that it should not be hidden.
2019-10-06 20:30:17 +02:00
MP_WARN(mpctx, "Reset playback due to audio timestamp reset.\n");
player: detect audio PTS jumps, make video PTS heuristic less aggressive This is another attempt at making files with sparse video frames work better. The problem is that you generally can't know whether a jump in video timestamps is just a (very) long video frame, or a timestamp reset. Due to the existence of files with sparse video frames (new frame only every few seconds or longer), every heuristic will be arbitrary (in general, at least). But we can use the fact that if video is continuous, audio should also be continuous. Audio discontinuities can be easily detected, and if that happens, reset some of the playback state. The way the playback state is reset is rather radical (resets decoders as well), but it's just better not to cause too much obscure stuff to happen here. If the A/V sync code were to be rewritten, it should probably strictly use PTS values (not this strange time_frame/delay stuff), which would make it much easier to detect such situations and to react to them.
2016-01-09 20:27:03 +01:00
reset_playback_state(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 14:24:15 +02:00
mp_wakeup_core(mpctx);
player: detect audio PTS jumps, make video PTS heuristic less aggressive This is another attempt at making files with sparse video frames work better. The problem is that you generally can't know whether a jump in video timestamps is just a (very) long video frame, or a timestamp reset. Due to the existence of files with sparse video frames (new frame only every few seconds or longer), every heuristic will be arbitrary (in general, at least). But we can use the fact that if video is continuous, audio should also be continuous. Audio discontinuities can be easily detected, and if that happens, reset some of the playback state. The way the playback state is reset is rather radical (resets decoders as well), but it's just better not to cause too much obscure stuff to happen here. If the A/V sync code were to be rewritten, it should probably strictly use PTS values (not this strange time_frame/delay stuff), which would make it much easier to detect such situations and to react to them.
2016-01-09 20:27:03 +01:00
}
audio: refactor how data is passed to AO This replaces the two buffers (ao_chain.ao_buffer in the core, and buffer_state.buffers in the AO) with a single queue. Instead of having a byte based buffer, the queue is simply a list of audio frames, as output by the decoder. This should make dataflow simpler and reduce copying. It also attempts to simplify fill_audio_out_buffers(), the function I always hated most, because it's full of subtle and buggy logic. Unfortunately, I got assaulted by corner cases, dumb features (attempt at seamless looping, really?), and other crap, so it got pretty complicated again. fill_audio_out_buffers() is still full of subtle and buggy logic. Maybe it got worse. On the other hand, maybe there really is some progress. Who knows. Originally, the data flow parts was meant to be in f_output_chain, but due to tricky interactions with the playloop code, it's now in the dummy filter in audio.c. At least this improves the way the audio PTS is passed to the encoder in encoding mode. Now it attempts to pass frames directly, along with the pts, which should minimize timestamp problems. But to be honest, encoder mode is one big kludge that shouldn't exist in this way. This commit should be considered pre-alpha code. There are lots of bugs still hiding.
2020-08-28 20:23:54 +02:00
if (mpctx->audio_status == STATUS_SYNCING) {
double pts;
bool ok = get_sync_pts(mpctx, &pts);
// If the AO is still playing from the previous file (due to gapless),
// but if video is active, this may not work if audio starts later than
// video, and gapless has no advantages anyway. So block doing anything
// until the old audio is fully played.
// (Buggy if AO underruns.)
if (mpctx->ao && ao_is_playing(mpctx->ao) &&
audio: add two minor log messages This would have made the problem fixed in the previous commit a bit more obvious from the log output.
2021-04-29 17:12:44 +02:00
mpctx->video_status != STATUS_EOF) {
MP_VERBOSE(mpctx, "blocked, waiting for old audio to play\n");
audio: refactor how data is passed to AO This replaces the two buffers (ao_chain.ao_buffer in the core, and buffer_state.buffers in the AO) with a single queue. Instead of having a byte based buffer, the queue is simply a list of audio frames, as output by the decoder. This should make dataflow simpler and reduce copying. It also attempts to simplify fill_audio_out_buffers(), the function I always hated most, because it's full of subtle and buggy logic. Unfortunately, I got assaulted by corner cases, dumb features (attempt at seamless looping, really?), and other crap, so it got pretty complicated again. fill_audio_out_buffers() is still full of subtle and buggy logic. Maybe it got worse. On the other hand, maybe there really is some progress. Who knows. Originally, the data flow parts was meant to be in f_output_chain, but due to tricky interactions with the playloop code, it's now in the dummy filter in audio.c. At least this improves the way the audio PTS is passed to the encoder in encoding mode. Now it attempts to pass frames directly, along with the pts, which should minimize timestamp problems. But to be honest, encoder mode is one big kludge that shouldn't exist in this way. This commit should be considered pre-alpha code. There are lots of bugs still hiding.
2020-08-28 20:23:54 +02:00
ok = false;
audio: add two minor log messages This would have made the problem fixed in the previous commit a bit more obvious from the log output.
2021-04-29 17:12:44 +02:00
}
audio: refactor how data is passed to AO This replaces the two buffers (ao_chain.ao_buffer in the core, and buffer_state.buffers in the AO) with a single queue. Instead of having a byte based buffer, the queue is simply a list of audio frames, as output by the decoder. This should make dataflow simpler and reduce copying. It also attempts to simplify fill_audio_out_buffers(), the function I always hated most, because it's full of subtle and buggy logic. Unfortunately, I got assaulted by corner cases, dumb features (attempt at seamless looping, really?), and other crap, so it got pretty complicated again. fill_audio_out_buffers() is still full of subtle and buggy logic. Maybe it got worse. On the other hand, maybe there really is some progress. Who knows. Originally, the data flow parts was meant to be in f_output_chain, but due to tricky interactions with the playloop code, it's now in the dummy filter in audio.c. At least this improves the way the audio PTS is passed to the encoder in encoding mode. Now it attempts to pass frames directly, along with the pts, which should minimize timestamp problems. But to be honest, encoder mode is one big kludge that shouldn't exist in this way. This commit should be considered pre-alpha code. There are lots of bugs still hiding.
2020-08-28 20:23:54 +02:00
if (ao_c->start_pts_known != ok || ao_c->start_pts != pts) {
ao_c->start_pts_known = ok;
ao_c->start_pts = pts;
mp_filter_wakeup(ao_c->ao_filter);
}
audio: strictly align audio on spdif frames We still have a sample-based buffer between filters and audio outputs. In order to avoid cutting frames into half (which can upset receivers), we strictly need to align the boundaries on which we cut the audio.
2015-11-04 16:47:11 +01:00
audio: refactor how data is passed to AO This replaces the two buffers (ao_chain.ao_buffer in the core, and buffer_state.buffers in the AO) with a single queue. Instead of having a byte based buffer, the queue is simply a list of audio frames, as output by the decoder. This should make dataflow simpler and reduce copying. It also attempts to simplify fill_audio_out_buffers(), the function I always hated most, because it's full of subtle and buggy logic. Unfortunately, I got assaulted by corner cases, dumb features (attempt at seamless looping, really?), and other crap, so it got pretty complicated again. fill_audio_out_buffers() is still full of subtle and buggy logic. Maybe it got worse. On the other hand, maybe there really is some progress. Who knows. Originally, the data flow parts was meant to be in f_output_chain, but due to tricky interactions with the playloop code, it's now in the dummy filter in audio.c. At least this improves the way the audio PTS is passed to the encoder in encoding mode. Now it attempts to pass frames directly, along with the pts, which should minimize timestamp problems. But to be honest, encoder mode is one big kludge that shouldn't exist in this way. This commit should be considered pre-alpha code. There are lots of bugs still hiding.
2020-08-28 20:23:54 +02:00
if (ao_c->ao && mp_async_queue_is_full(ao_c->ao_queue)) {
mpctx->audio_status = STATUS_READY;
mp_wakeup_core(mpctx);
MP_VERBOSE(mpctx, "audio ready\n");
} else if (ao_c->out_eof) {
// Force playback start early.
mpctx->audio_status = STATUS_READY;
audio: rewrite filtering glue code Use the new filtering code for audio too.
2018-01-18 14:44:20 +01:00
mp_wakeup_core(mpctx);
audio: refactor how data is passed to AO This replaces the two buffers (ao_chain.ao_buffer in the core, and buffer_state.buffers in the AO) with a single queue. Instead of having a byte based buffer, the queue is simply a list of audio frames, as output by the decoder. This should make dataflow simpler and reduce copying. It also attempts to simplify fill_audio_out_buffers(), the function I always hated most, because it's full of subtle and buggy logic. Unfortunately, I got assaulted by corner cases, dumb features (attempt at seamless looping, really?), and other crap, so it got pretty complicated again. fill_audio_out_buffers() is still full of subtle and buggy logic. Maybe it got worse. On the other hand, maybe there really is some progress. Who knows. Originally, the data flow parts was meant to be in f_output_chain, but due to tricky interactions with the playloop code, it's now in the dummy filter in audio.c. At least this improves the way the audio PTS is passed to the encoder in encoding mode. Now it attempts to pass frames directly, along with the pts, which should minimize timestamp problems. But to be honest, encoder mode is one big kludge that shouldn't exist in this way. This commit should be considered pre-alpha code. There are lots of bugs still hiding.
2020-08-28 20:23:54 +02:00
MP_VERBOSE(mpctx, "audio ready (and EOF)\n");
audio: rewrite filtering glue code Use the new filtering code for audio too.
2018-01-18 14:44:20 +01:00
}
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 20:40:43 +02:00
}
audio: refactor how data is passed to AO This replaces the two buffers (ao_chain.ao_buffer in the core, and buffer_state.buffers in the AO) with a single queue. Instead of having a byte based buffer, the queue is simply a list of audio frames, as output by the decoder. This should make dataflow simpler and reduce copying. It also attempts to simplify fill_audio_out_buffers(), the function I always hated most, because it's full of subtle and buggy logic. Unfortunately, I got assaulted by corner cases, dumb features (attempt at seamless looping, really?), and other crap, so it got pretty complicated again. fill_audio_out_buffers() is still full of subtle and buggy logic. Maybe it got worse. On the other hand, maybe there really is some progress. Who knows. Originally, the data flow parts was meant to be in f_output_chain, but due to tricky interactions with the playloop code, it's now in the dummy filter in audio.c. At least this improves the way the audio PTS is passed to the encoder in encoding mode. Now it attempts to pass frames directly, along with the pts, which should minimize timestamp problems. But to be honest, encoder mode is one big kludge that shouldn't exist in this way. This commit should be considered pre-alpha code. There are lots of bugs still hiding.
2020-08-28 20:23:54 +02:00
if (ao_c->ao && !ao_is_playing(ao_c->ao) && !ao_c->underrun &&
(mpctx->audio_status == STATUS_PLAYING ||
mpctx->audio_status == STATUS_DRAINING))
audio: fix late audio start Regression since commit bbcd0b6a. This code is just cursed, because it's a fragile state machine with no proper tests, and which could be done in a much simpler way. Without doubt this change will cause a regression in some ridiculous corner case as well. Fixes #3610 (the cause of it, not the behavior it resulted in).
2016-10-02 13:47:19 +02:00
{
audio: refactor how data is passed to AO This replaces the two buffers (ao_chain.ao_buffer in the core, and buffer_state.buffers in the AO) with a single queue. Instead of having a byte based buffer, the queue is simply a list of audio frames, as output by the decoder. This should make dataflow simpler and reduce copying. It also attempts to simplify fill_audio_out_buffers(), the function I always hated most, because it's full of subtle and buggy logic. Unfortunately, I got assaulted by corner cases, dumb features (attempt at seamless looping, really?), and other crap, so it got pretty complicated again. fill_audio_out_buffers() is still full of subtle and buggy logic. Maybe it got worse. On the other hand, maybe there really is some progress. Who knows. Originally, the data flow parts was meant to be in f_output_chain, but due to tricky interactions with the playloop code, it's now in the dummy filter in audio.c. At least this improves the way the audio PTS is passed to the encoder in encoding mode. Now it attempts to pass frames directly, along with the pts, which should minimize timestamp problems. But to be honest, encoder mode is one big kludge that shouldn't exist in this way. This commit should be considered pre-alpha code. There are lots of bugs still hiding.
2020-08-28 20:23:54 +02:00
// Should be playing, but somehow isn't.
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 20:40:43 +02:00
audio: refactor how data is passed to AO This replaces the two buffers (ao_chain.ao_buffer in the core, and buffer_state.buffers in the AO) with a single queue. Instead of having a byte based buffer, the queue is simply a list of audio frames, as output by the decoder. This should make dataflow simpler and reduce copying. It also attempts to simplify fill_audio_out_buffers(), the function I always hated most, because it's full of subtle and buggy logic. Unfortunately, I got assaulted by corner cases, dumb features (attempt at seamless looping, really?), and other crap, so it got pretty complicated again. fill_audio_out_buffers() is still full of subtle and buggy logic. Maybe it got worse. On the other hand, maybe there really is some progress. Who knows. Originally, the data flow parts was meant to be in f_output_chain, but due to tricky interactions with the playloop code, it's now in the dummy filter in audio.c. At least this improves the way the audio PTS is passed to the encoder in encoding mode. Now it attempts to pass frames directly, along with the pts, which should minimize timestamp problems. But to be honest, encoder mode is one big kludge that shouldn't exist in this way. This commit should be considered pre-alpha code. There are lots of bugs still hiding.
2020-08-28 20:23:54 +02:00
if (ao_c->out_eof && !mp_async_queue_get_frames(ao_c->ao_queue)) {
MP_VERBOSE(mpctx, "AO signaled EOF (while in state %s)\n",
mp_status_str(mpctx->audio_status));
audio: better sync behavior on bogus EOF In situations when the demuxer reports EOF, but immediately "recovers" after that and returns new data, it could happen that audio sync was skipped. Deal with this by actually entering the EOF state, instead of assuming this will happen later.
2014-07-30 22:29:24 +02:00
mpctx->audio_status = STATUS_EOF;
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 14:24:15 +02:00
mp_wakeup_core(mpctx);
audio: refactor how data is passed to AO This replaces the two buffers (ao_chain.ao_buffer in the core, and buffer_state.buffers in the AO) with a single queue. Instead of having a byte based buffer, the queue is simply a list of audio frames, as output by the decoder. This should make dataflow simpler and reduce copying. It also attempts to simplify fill_audio_out_buffers(), the function I always hated most, because it's full of subtle and buggy logic. Unfortunately, I got assaulted by corner cases, dumb features (attempt at seamless looping, really?), and other crap, so it got pretty complicated again. fill_audio_out_buffers() is still full of subtle and buggy logic. Maybe it got worse. On the other hand, maybe there really is some progress. Who knows. Originally, the data flow parts was meant to be in f_output_chain, but due to tricky interactions with the playloop code, it's now in the dummy filter in audio.c. At least this improves the way the audio PTS is passed to the encoder in encoding mode. Now it attempts to pass frames directly, along with the pts, which should minimize timestamp problems. But to be honest, encoder mode is one big kludge that shouldn't exist in this way. This commit should be considered pre-alpha code. There are lots of bugs still hiding.
2020-08-28 20:23:54 +02:00
// stops untimed AOs, stops pull AOs from streaming silence
ao_reset(ao_c->ao);
} else {
if (!ao_c->ao_underrun) {
MP_WARN(mpctx, "Audio device underrun detected.\n");
ao_c->ao_underrun = true;
mp_wakeup_core(mpctx);
ao_c->underrun = true;
}
player: really start audio only once video is ready The check whether video is ready yet was done only in STATUS_FILLING. But it also switched to STATUS_READY, which means the next time fill_audio_out_buffers() was called, audio would actually be started before video. In most situations, this bug didn't show up, because it was only triggered if the demuxer didn't provide video packets quickly enough, but did for audio packets. Also log when audio is started. (I hate fill_audio_out_buffers(), why did I write it?)
2016-04-20 13:05:29 +02:00
audio: refactor how data is passed to AO This replaces the two buffers (ao_chain.ao_buffer in the core, and buffer_state.buffers in the AO) with a single queue. Instead of having a byte based buffer, the queue is simply a list of audio frames, as output by the decoder. This should make dataflow simpler and reduce copying. It also attempts to simplify fill_audio_out_buffers(), the function I always hated most, because it's full of subtle and buggy logic. Unfortunately, I got assaulted by corner cases, dumb features (attempt at seamless looping, really?), and other crap, so it got pretty complicated again. fill_audio_out_buffers() is still full of subtle and buggy logic. Maybe it got worse. On the other hand, maybe there really is some progress. Who knows. Originally, the data flow parts was meant to be in f_output_chain, but due to tricky interactions with the playloop code, it's now in the dummy filter in audio.c. At least this improves the way the audio PTS is passed to the encoder in encoding mode. Now it attempts to pass frames directly, along with the pts, which should minimize timestamp problems. But to be honest, encoder mode is one big kludge that shouldn't exist in this way. This commit should be considered pre-alpha code. There are lots of bugs still hiding.
2020-08-28 20:23:54 +02:00
// Wait until buffers are filled before recovering underrun.
if (ao_c->out_eof || mp_async_queue_is_full(ao_c->ao_queue)) {
MP_VERBOSE(mpctx, "restarting audio after underrun\n");
ao_start(mpctx->ao_chain->ao);
ao_c->ao_underrun = false;
ao_c->underrun = false;
mp_wakeup_core(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 22:38:29 +01:00
}
audio: refactor how data is passed to AO This replaces the two buffers (ao_chain.ao_buffer in the core, and buffer_state.buffers in the AO) with a single queue. Instead of having a byte based buffer, the queue is simply a list of audio frames, as output by the decoder. This should make dataflow simpler and reduce copying. It also attempts to simplify fill_audio_out_buffers(), the function I always hated most, because it's full of subtle and buggy logic. Unfortunately, I got assaulted by corner cases, dumb features (attempt at seamless looping, really?), and other crap, so it got pretty complicated again. fill_audio_out_buffers() is still full of subtle and buggy logic. Maybe it got worse. On the other hand, maybe there really is some progress. Who knows. Originally, the data flow parts was meant to be in f_output_chain, but due to tricky interactions with the playloop code, it's now in the dummy filter in audio.c. At least this improves the way the audio PTS is passed to the encoder in encoding mode. Now it attempts to pass frames directly, along with the pts, which should minimize timestamp problems. But to be honest, encoder mode is one big kludge that shouldn't exist in this way. This commit should be considered pre-alpha code. There are lots of bugs still hiding.
2020-08-28 20:23:54 +02:00
if (mpctx->audio_status == STATUS_PLAYING && ao_c->out_eof) {
mpctx->audio_status = STATUS_DRAINING;
MP_VERBOSE(mpctx, "audio draining\n");
mp_wakeup_core(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 22:38:29 +01:00
}
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 20:40:43 +02:00
player: fix another nightmarish corner case Pretty much fuck this shit.
2020-10-08 00:36:41 +02:00
if (mpctx->audio_status == STATUS_DRAINING) {
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 20:40:43 +02:00
// Wait until the AO has played all queued data. In the gapless case,
// we trigger EOF immediately, and let it play asynchronously.
player: fix another nightmarish corner case Pretty much fuck this shit.
2020-10-08 00:36:41 +02:00
if (!ao_c->ao || (!ao_is_playing(ao_c->ao) ||
(opts->gapless_audio && !ao_untimed(ao_c->ao))))
audio: refactor how data is passed to AO This replaces the two buffers (ao_chain.ao_buffer in the core, and buffer_state.buffers in the AO) with a single queue. Instead of having a byte based buffer, the queue is simply a list of audio frames, as output by the decoder. This should make dataflow simpler and reduce copying. It also attempts to simplify fill_audio_out_buffers(), the function I always hated most, because it's full of subtle and buggy logic. Unfortunately, I got assaulted by corner cases, dumb features (attempt at seamless looping, really?), and other crap, so it got pretty complicated again. fill_audio_out_buffers() is still full of subtle and buggy logic. Maybe it got worse. On the other hand, maybe there really is some progress. Who knows. Originally, the data flow parts was meant to be in f_output_chain, but due to tricky interactions with the playloop code, it's now in the dummy filter in audio.c. At least this improves the way the audio PTS is passed to the encoder in encoding mode. Now it attempts to pass frames directly, along with the pts, which should minimize timestamp problems. But to be honest, encoder mode is one big kludge that shouldn't exist in this way. This commit should be considered pre-alpha code. There are lots of bugs still hiding.
2020-08-28 20:23:54 +02:00
{
MP_VERBOSE(mpctx, "audio EOF reached\n");
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 20:40:43 +02:00
mpctx->audio_status = STATUS_EOF;
audio: refactor how data is passed to AO This replaces the two buffers (ao_chain.ao_buffer in the core, and buffer_state.buffers in the AO) with a single queue. Instead of having a byte based buffer, the queue is simply a list of audio frames, as output by the decoder. This should make dataflow simpler and reduce copying. It also attempts to simplify fill_audio_out_buffers(), the function I always hated most, because it's full of subtle and buggy logic. Unfortunately, I got assaulted by corner cases, dumb features (attempt at seamless looping, really?), and other crap, so it got pretty complicated again. fill_audio_out_buffers() is still full of subtle and buggy logic. Maybe it got worse. On the other hand, maybe there really is some progress. Who knows. Originally, the data flow parts was meant to be in f_output_chain, but due to tricky interactions with the playloop code, it's now in the dummy filter in audio.c. At least this improves the way the audio PTS is passed to the encoder in encoding mode. Now it attempts to pass frames directly, along with the pts, which should minimize timestamp problems. But to be honest, encoder mode is one big kludge that shouldn't exist in this way. This commit should be considered pre-alpha code. There are lots of bugs still hiding.
2020-08-28 20:23:54 +02:00
mp_wakeup_core(mpctx);
audio: avoid missed wakeups with ab-loops Could get "stuck".
2016-08-24 12:14:48 +02:00
}
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 20:40:43 +02:00
}
audio: refactor how data is passed to AO This replaces the two buffers (ao_chain.ao_buffer in the core, and buffer_state.buffers in the AO) with a single queue. Instead of having a byte based buffer, the queue is simply a list of audio frames, as output by the decoder. This should make dataflow simpler and reduce copying. It also attempts to simplify fill_audio_out_buffers(), the function I always hated most, because it's full of subtle and buggy logic. Unfortunately, I got assaulted by corner cases, dumb features (attempt at seamless looping, really?), and other crap, so it got pretty complicated again. fill_audio_out_buffers() is still full of subtle and buggy logic. Maybe it got worse. On the other hand, maybe there really is some progress. Who knows. Originally, the data flow parts was meant to be in f_output_chain, but due to tricky interactions with the playloop code, it's now in the dummy filter in audio.c. At least this improves the way the audio PTS is passed to the encoder in encoding mode. Now it attempts to pass frames directly, along with the pts, which should minimize timestamp problems. But to be honest, encoder mode is one big kludge that shouldn't exist in this way. This commit should be considered pre-alpha code. There are lots of bugs still hiding.
2020-08-28 20:23:54 +02:00
audio: slightly simplify audio_start_ao() Get rid of an indirection; no behavior change.
2020-09-03 12:22:20 +02:00
if (mpctx->restart_complete)
audio_start_ao(mpctx); // in case it got delayed
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 22:38:29 +01:00
}
player: factor audio buffer clearing code Note that the change in seek_reset is not entirely equivalent: we even drop the remainder of buffered audio when seeking. This should be more correct, because the whole point of the reset_ao parameter is to control whether audio queued for output should be dropped or not.
2013-11-08 20:02:09 +01:00
// Drop data queued for output, or which the AO is currently outputting.
void clear_audio_output_buffers(struct MPContext *mpctx)
{
player: fix crash on early audio uninit Could crash when exiting playback in very early stages of initialization. CC: @mpv-player/stable
2014-10-16 01:03:02 +02:00
if (mpctx->ao)
player: factor audio buffer clearing code Note that the change in seek_reset is not entirely equivalent: we even drop the remainder of buffered audio when seeking. This should be more correct, because the whole point of the reset_ao parameter is to control whether audio queued for output should be dropped or not.
2013-11-08 20:02:09 +01:00
ao_reset(mpctx->ao);
}