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* command: shuffle some crap aroundwm42019-11-251-0/+2
| | | | | | | | | | | | | | | | | | | | | | | | 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.
* options: deprecate --video-sync=display-adropwm42019-11-171-0/+1
| | | | A stupid thing that will probably be in the way.
* player: remove commented declarationwm42019-11-171-1/+0
| | | | It was commented almost 2 years ago in a "rewrite everything" commit.
* video: do not disable display-sync on A/V desyncwm42019-10-171-1/+0
| | | | | | | | | | | | | | | | | | | | | | | | | | On a audio/video desync by more than 0.5 seconds, display-sync mode was disabled, and not enabled again (until playback restart, e.g. a seek). The idea was that it this only happens when this playback mode is broken and can't perform well anyway (A/V desync is a clear indication that something is very wrong). Instead of behaving like a god damn POS, it should revert to the more robust audio-sync mode. Unfortunately, this could happen sporadically due to temporary system performance problems, such as toggling fullscreen. Users didn't like this, and asked for a function to disable it, or to recover in some other way. This mechanism is questionable anyway. If an ignorant user enables display-sync, and encounters problems with it (without being able to determine that display-sync is messing up), the player will still behave like a POS on every playback, and even after every seek. It might actually be helpful to fail more consistently. Also, I've found that it's sill relatively reliable anyway even without this mechanism. So just remove the fallback. Fixes: #7048
* player: partially rework --cache-pausewm42019-10-111-0/+5
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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...
* loadfile: don't always accidentally always prefetchingwm42019-09-291-0/+1
| | | | | | | | | | | | | | demux_start_prefetch() was called unconditionally in two cases. This is completely wrong. I'm not sure what part of my brain died off that something this obviously wrong went in. The prefetch case is a bit more complicated. It's a different thread, so you can't access just access mpctx->opts there. So add an explicit field for this, which is the simplest way to get this done. (Even if it's bad factoring.) Fixes: c1f1a0845e03885eebe63 Fixes: 556e204a112ee286972e5
* demux, command: add a third stream recording mechanismwm42019-09-191-1/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | That's right, and it's probably not the end of it. I'll just claim that I have no idea how to create a proper user interface for this, so I'm creating multiple partially-orthogonal, of which some may work better in each of its special use cases. Until now, there was --record-file. You get relatively good control about what is muxed, and it can use the cache. But it sucks that it's bound to playback. If you pause while it's set, muxing stops. If you seek while it's set, the output will be sort-of trashed, and that's by design. Then --stream-record was added. This is a bit better (especially for live streams), but you can't really control well when muxing stops or ends. In particular, it can't use the cache (it just dumps whatever the underlying demuxer returns). Today, the idea is that the user should just be able to select a time range to dump to a file, and it should not affected by the user seeking around in the cache. In addition, the stream may still be running, so there's some need to continue dumping, even if it's redundant to --stream-record. One notable thing is that it uses the async command shit. Not sure whether this is a good idea. Maybe not, but whatever. Also, a user can always use the "async" prefix to pretend it doesn't. Much of this was barely tested (especially the reinterleaving crap), let's just hope it mostly works. I'm sure you can tolerate the one or other crash?
* player: fix --loop with backward playbackwm42019-09-191-0/+1
| | | | | | | | | | | Obviously should seek back to the end of the file when it loops. Also remove some minor code duplication around start times. This isn't the correct solution by the way. Rather than hoping we know a reasonable start/end time, this stuff should instruct the demuxer to seek to the exact location. It'll work with 99% of all normal files, but add an appropriate comment (that basically says the function is bullshit) to get_start_time() anyway.
* player: modify/simplify AB-loop behaviorwm42019-09-191-1/+3
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This changes the behavior of the --ab-loop-a/b options. In addition, it makes it work with backward playback mode. The most obvious change is that the both the A and B point need to be set now before any looping happens. Unlike before, unset points don't implicitly use the start or end of the file. I think the old behavior was a feature that was explicitly added/wanted. Well, it's gone now. This is because of 2 reasons: 1. I never liked this feature, and it always got in my way (as user). 2. It's inherently annoying with backward playback mode. In backward playback mode, the user wants to set A/B in the wrong order. The ab-loop command will first set A, then B, so if you use this command during backward playback, A will be set to a higher timestamps than B. If you switch back to forward playback mode, the loop would stop working. I want the loop to just continue to work, and the chosen solution conflicts with the removed feature. The order issue above _could_ be fixed by also switching the AB-loop user option values around on direction switch. But there are no other instances of option changes magically affecting other options, and doing this would probably lead to unexpected misery (dying from corner cases and such). Another solution is sorting the A/B points by timestamps after copying them from the user options. Then A/B options set in backward mode will work in forward mode. This is the chosen solution. If you sort the points, you don't know anymore whether the unset point is supposed to signify the end or the start of the file. The AB-loop code is slightly better abstracted now, so it should be easy to restore the removed feature. It would still require coming up with a solution for backwards playback, though. A minor change is that if one point is set and the other is unset, I'm rendering both the chapter markers and the marker for the set point. Why? I don't know. My test file had chapters, and I guess I decided this looked better. This commit also fixes some subtle and obvious issues that I already forgot about when I wrote this commit message. It cleans up some minor code duplication and nonsense too. Regarding backward playback, the code uses an unsanitary mix of internal ("transformed") and user timestamps. So the play_dir variable appears more than usual. To mention one unfixed issue: if you set an AB-loop that is completely past the end of the file, it will get stuck in an infinite seeking loop once playback reaches the end of the file. Fixing this reliably seemed annoying, so the fix is "just don't do this". It's not a hard freeze anyway.
* player: make a function staticwm42019-09-191-1/+0
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* Implement backwards playbackwm42019-09-191-0/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | See manpage additions. This is a huge hack. You can bet there are shit tons of bugs. It's literally forcing square pegs into round holes. Hopefully, the manpage wall of text makes it clear enough that the whole shit can easily crash and burn. (Although it shouldn't literally crash. That would be a bug. It possibly _could_ start a fire by entering some sort of endless loop, not a literal one, just something where it tries to do work without making progress.) (Some obvious bugs I simply ignored for this initial version, but there's a number of potential bugs I can't even imagine. Normal playback should remain completely unaffected, though.) How this works is also described in the manpage. Basically, we demux in reverse, then we decode in reverse, then we render in reverse. The decoding part is the simplest: just reorder the decoder output. This weirdly integrates with the timeline/ordered chapter code, which also has special requirements on feeding the packets to the decoder in a non-straightforward way (it doesn't conflict, although a bugmessmass breaks correct slicing of segments, so EDL/ordered chapter playback is broken in backward direction). Backward demuxing is pretty involved. In theory, it could be much easier: simply iterating the usual demuxer output backward. But this just doesn't fit into our code, so there's a cthulhu nightmare of shit. To be specific, each stream (audio, video) is reversed separately. At least this means we can do backward playback within cached content (for example, you could play backwards in a live stream; on that note, it disables prefetching, which would lead to losing new live video, but this could be avoided). The fuckmess also meant that I didn't bother trying to support subtitles. Subtitles are a problem because they're "sparse" streams. They need to be "passively" demuxed: you don't try to read a subtitle packet, you demux audio and video, and then look whether there was a subtitle packet. This means to get subtitles for a time range, you need to know that you demuxed video and audio over this range, which becomes pretty messy when you demux audio and video backwards separately. Backward display is the most weird (and potentially buggy) part. To avoid that we need to touch a LOT of timing code, we negate all timestamps. The basic idea is that due to the navigation, all comparisons and subtractions of timestamps keep working, and you don't need to touch every single of them to "reverse" them. E.g.: bool before = pts_a < pts_b; would need to be: bool before = forward ? pts_a < pts_b : pts_a > pts_b; or: bool before = pts_a * dir < pts_b * dir; or if you, as it's implemented now, just do this after decoding: pts_a *= dir; pts_b *= dir; and then in the normal timing/renderer code: bool before = pts_a < pts_b; Consequently, we don't need many changes in the latter code. But some assumptions inhererently true for forward playback may have been broken anyway. What is mainly needed is fixing places where values are passed between positive and negative "domains". For example, seeking and timestamp user display always uses positive timestamps. The main mess is that it's not obvious which domain a given variable should or does use. Well, in my tests with a single file, it suddenly started to work when I did this. I'm honestly surprised that it did, and that I didn't have to change a single line in the timing code past decoder (just something minor to make external/cached text subtitles display). I committed it immediately while avoiding thinking about it. But there really likely are subtle problems of all sorts. As far as I'm aware, gstreamer also supports backward playback. When I looked at this years ago, I couldn't find a way to actually try this, and I didn't revisit it now. Back then I also read talk slides from the person who implemented it, and I'm not sure if and which ideas I might have taken from it. It's possible that the timestamp reversal is inspired by it, but I didn't check. (I think it claimed that it could avoid large changes by changing a sign?) VapourSynth has some sort of reverse function, which provides a backward view on a video. The function itself is trivial to implement, as VapourSynth aims to provide random access to video by frame numbers (so you just request decreasing frame numbers). From what I remember, it wasn't exactly fluid, but it worked. It's implemented by creating an index, and seeking to the target on demand, and a bunch of caching. mpv could use it, but it would either require using VapourSynth as demuxer and decoder for everything, or replacing the current file every time something is supposed to be played backwards. FFmpeg's libavfilter has reversal filters for audio and video. These require buffering the entire media data of the file, and don't really fit into mpv's architecture. It could be used by playing a libavfilter graph that also demuxes, but that's like VapourSynth but worse.
* Merge commit '559a400ac36e75a8d73ba263fd7fa6736df1c2da' into ↵Anton Kindestam2018-12-051-9/+36
|\ | | | | | | | | | | wm4-commits--merge-edition This bumps libmpv version to 1.103
| * demux, stream: rip out the classic stream cachewm42018-08-311-2/+0
| | | | | | | | | | | | The demuxer cache is the only cache now. Might need another change to combat seeking failures in mp4 etc. The only bad thing is the loss of cache-speed, which was sort of nice to have.
| * player: change the role of the "stop_play" and "playing" variablewm42018-05-241-2/+3
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Before this, mpctx->playing was often used to determine whether certain new state could be added to the playback state. In particular this affected external files (which added tracks and demuxers). The variable was checked to prevent that they were added before the corresponding uninit code. We want to make a small part of uninit asynchronous, but mpctx->playing needs to stay in the place where it is. It can't be used for this purpose anymore. Use mpctx->stop_play instead. Make it never have the value 0 outside of loading/playback. On unloading, it obviously has to be non-0. Change some other code in playloop.c to use this, because it seems slightly more correct. But mostly this is preparation for the following commit.
| * player: simplify edition switchingwm42018-05-241-1/+0
| | | | | | | | | | | | | | | | | | | | | | | | | | | | The player fully restarts playback when the edition or disk title is changed. Before this, the player tried to reinitialized playback partially. For example, it did not print a new "Playing: <file>" message, and did not send playback end to libmpv users (scripts or applications). This playback restart code was a bit messy and could have unforeseen interactions with various state. There have been bugs before. Since it's a mostly cosmetic thing for an obscure feature, just change it to a full restart. This works well, though since it may have consequences for scripts or client API users, mention it in interface-changes.rst.
| * player: make various commands for managing external tracks abortablewm42018-05-241-3/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Until now, they could be aborted only by ending playback, and calling mpv_abort_async_command didn't do anything. This requires furthering the mess how playback abort is done. The main reason why mp_cancel exists at all is to avoid that a "frozen" demuxer (blocked on network I/O or whatever) cannot freeze the core. The core should always get its way. Previously, there was a single mp_cancel handle, that could be signaled, and all demuxers would unfreeze. With external files, we might want to abort loading of a certain external file, which automatically means they need a separate mp_cancel. So give every demuxer its own mp_cancel, and "slave" it to whatever parent mp_cancel handles aborting. Since the mpv demuxer API conflates creating the demuxer and reading the file headers, mp_cancel strictly need to be created before the demuxer is created (or we couldn't abort loading). Although we give every demuxer its own mp_cancel (as "enforced" by cancel_and_free_demuxer), it's still rather messy to create/destroy it along with the demuxer.
| * client API: kill async commands on terminationwm42018-05-241-0/+1
| | | | | | | | | | | | | | | | | | | | | | This affects async commands started by client API, commands with async capability run in a sync way by client API (think mpv_command_node() with "subprocess"), and detached async work. Since scripts might want to do some cleanup work (that might involve launching processes, don't ask), we don't unconditionally kill everything on exit, but apply an arbitrary timeout of 2 seconds until async commands are aborted.
| * command: add a way to abort asynchronous commandswm42018-05-241-0/+22
| | | | | | | | | | | | | | | | | | | | | | | | Many asynchronous commands are potentially long running operations, such as loading something from network or running a foreign process. Obviously it shouldn't just be possible for them to freeze the player if they don't terminate as expected. Also, there will be situations where you want to explicitly stop some of those operations explicitly. So add an infrastructure for this. Commands have to support this explicitly. The next commit uses this to actually add support to a command.
| * player: rename "lock" to "abort_lock"wm42018-05-241-2/+2
| | | | | | | | | | | | If a struct as large as MPContext contains a field named "lock", it creates the impression that it is the primary lock for MPContext. This is wrong, the lock just protects a single field.
| * player: make all external file loading actions asyncwm42018-05-241-1/+3
| | | | | | | | | | Still missing: not freezing when removing a track (i.e. closing demuxer) with the sub-remove/audio-remove/rescan-external-files commands.
| * command: make sub-add and audio-add commands asyncwm42018-05-241-1/+1
| | | | | | | | | | | | | | | | | | | | Pretty trivial, since commands can be async now, and the common code even provides convenience like running commands on a worker thread. The only ugly thing is that mp_add_external_file() needs an extra flag for locking. This is because there's still some code which calls this synchronously from the main thread, and unlocking the core makes no sense there.
| * command: add infrastructure for async commandswm42018-05-241-0/+4
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This enables two types of command behavior: 1. Plain async behavior, like "loadfile" not completing until the file is fully loaded. 2. Running parts of the command on worker threads, e.g. for I/O, such as "sub-add" doing network accesses on a thread while the core continues. Both have no implementation yet, and most new code is actually inactive. The plan is to implement a number of useful cases in the following commits. The most tricky part is handling internal keybindings (input.conf) and the multi-command feature (concatenating commands with ";"). It requires a bunch of roundabout code to make it do the expected thing in combination with async commands. There is the question how commands should be handled that come in at a higher rate than what can be handled by the core. Currently, it will simply queue up input.conf commands as long as memory lasts. The client API is limited by the size of the reply queue per client. For commands which require a worker thread, the thread pool is limited to 30 threads, and then will queue up work in memory. The number is completely arbitrary.
* | player: expose hearing/visual impaired flags on audio tracksAman Gupta2018-08-131-0/+1
| | | | | | | | Signed-off-by: Aman Gupta <aman@tmm1.net>
* | player: simplify edition switchingwm42018-05-311-1/+0
|/ | | | | | | | | | | | | | The player fully restarts playback when the edition or disk title is changed. Before this, the player tried to reinitialized playback partially. For example, it did not print a new "Playing: <file>" message, and did not send playback end to libmpv users (scripts or applications). This playback restart code was a bit messy and could have unforeseen interactions with various state. There have been bugs before. Since it's a mostly cosmetic thing for an obscure feature, just change it to a full restart. This works well, though since it may have consequences for scripts or client API users, mention it in interface-changes.rst.
* demux, player: fix playback of sparse video streams (w/ still images)Aman Gupta2018-05-241-0/+2
| | | | | | | | | | | | | | | Fixes several issues playing back mpegts with video streams marked as having "still images". For example, see this video which has frames only every 6s: https://s3.amazonaws.com/tmm1/music-choice.ts Changes include: - start playback right away, without waiting for first video frame - do not consider the sparse video stream in demuxer underrun detection - do not require multiple video frames for the VO - use audio as the master stream for demuxer metadata events - use audio stream for playback time Signed-off-by: Aman Gupta <aman@tmm1.net>
* player: remove in_dispatch fieldwm42018-04-181-1/+0
| | | | (Not sure if worth the trouble, but it does seem less awkward.)
* demux, player: mark dependent tracksAman Gupta2018-04-171-1/+1
| | | | | | | ffmpeg marks audio tracks which are not meant to be played standalone as DEPENDENT. these are typically used in DVB broadcasts for audio descriptions, and are meant to be mixed into the main audio track during playback.
* client API: cleanup mpv_handle terminationwm42018-03-151-1/+1
| | | | | | | | | | | | | | | This changes how mpv_terminate_destroy() and mpv_detach_destroy() behave. The doxygen in client.h tries to point out the differences. The goal is to make this more useful to the API user (making it behave like refcounting). This will be refined in follow up commits. Initialization is unfortunately closely tied to termination, so that changes as well. This also removes earlier hacks that make sure that some parts of FFmpeg initialization are run in the playback thread (instead of the user's thread). This does not matter with standard FFmpeg, and I have no reason to care about this anymore.
* scripting: make a function staticwm42018-03-081-1/+0
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* client API: deprecate opengl-cb API and introduce a replacement APIwm42018-02-281-2/+0
| | | | | | | | | | | | | | | | | | | | | | | | | The purpose of the new API is to make it useable with other APIs than OpenGL, especially D3D11 and vulkan. In theory it's now possible to support other vo_gpu backends, as well as backends that don't use the vo_gpu code at all. This also aims to get rid of the dumb mpv_get_sub_api() function. The life cycle of the new mpv_render_context is a bit different from mpv_opengl_cb_context, and you explicitly create/destroy the new context, instead of calling init/uninit on an object returned by mpv_get_sub_api(). In other to make the render API generic, it's annoyingly EGL style, and requires you to pass in API-specific objects to generic functions. This is to avoid explicit objects like the internal ra API has, because that sounds more complicated and annoying for an API that's supposed to never change. The opengl_cb API will continue to exist for a bit longer, but internally there are already a few tradeoffs, like reduced thread-safety. Mostly untested. Seems to work fine with mpc-qt.
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