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* Implement backwards playbackwm42019-09-194-13/+139
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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.
* f_decoder_wrapper: move cover art retrievalwm42019-09-191-5/+5
| | | | | This is basically a refactor in preparation for future changes and shouldn't have much influence on actual behavior.
* Merge branch 'master' into pr6360Jan Ekström2019-03-111-1/+1
|\ | | | | | | | | | | Manual changes done: * Merged the interface-changes under the already master'd changes. * Moved the hwdec-related option changes to video/decode/vd_lavc.c.
| * audio: fix segfault caused by incorrect number of planeszc622019-02-231-1/+1
| | | | | | | | | | | | | | Use `mp_aframe_get_planes` to properly get the number of planes, instead of assuming it to be the number of channels. Fixes #6092
* | Merge commit '559a400ac36e75a8d73ba263fd7fa6736df1c2da' into ↵Anton Kindestam2018-12-051-8/+13
|\ \ | |/ |/| | | | | | | wm4-commits--merge-edition This bumps libmpv version to 1.103
| * player: get rid of mpv_global.optswm42018-05-241-8/+13
| | | | | | | | | | | | | | | | This was always a legacy thing. Remove it by applying an orgy of mp_get_config_group() calls, and sometimes m_config_cache_alloc() or mp_read_option_raw(). win32 changes untested.
* | filters: Add cuda/nvdec deinterlacing auto-filter using vf_yadif_cudaPhilip Langdale2018-11-191-0/+4
|/ | | | | | | | | | | Historically, there's been no way to offer deinterlacing with nvdec, and for cuviddec, it required a command line flag, with no way to toggle while playing. Now that we have a cuda deinterlacing filter in ffmpeg, we can hook it up hook it up as the cuda auto-deinterlacer. In practice, this isn't going to be present very often, due to the licensing mess with the cuda sdk, but we can support it when it is there.
* f_lavfi: support setting common filter options like "threads"wm42018-04-291-1/+2
| | | | | | | | | AVFilterContext instances support some additional AVOptions over the actual filter. This includes useful options like "threads". We didn't support setting those for the "direct" wrapper (--vf=yadif:threads=1 failed). Change this. It requires setting options on the AVFilterContext directly, except the code for positional parameters still needs to access the actual filter's AVOptions.
* f_decoder_wrapper: fix a typo in log messagewm42018-04-291-1/+1
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* filter: hide warning when disconnecting pins drops frameswm42018-04-291-2/+2
| | | | | | | | | | Sometimes this hints that there's a bug, but sometimes it's normal. Since the code for --end/--frames puts frames that should not be shown anymore back into the pin, using those options will show this warning when playback ends. This is a minor annoyance. We could change how it's done (e.g. set an explicit flag somewhere), but that seems bothersome, so just change the message from warning to verbose.
* f_output_chain: remove a redundant variablewm42018-04-291-4/+2
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* video: remove internal stereo_out flagwm42018-04-291-2/+1
| | | | | | Also rename stereo3d to stereo_in. The only real change is that the vo_gpu OSD code now uses the actual stereo 3D mode, instead of the --video-steroe-mode value. (Why does this vo_gpu code even exist?)
* f_output_chain: log status of auto filterswm42018-04-294-0/+48
| | | | | Just so users don't think the filters do anything when they don't insert any filters.
* f_output_chain: log input instead of output formatwm42018-04-291-44/+38
| | | | | | | I think this is more intuitive. This requires a dedicated "out" dummy filter. But keep the "in" dummy filter for symmetry, like in the old filter code. (We could remove the "in" dummy filter, because the first actual filter would still show the real input format.)
* video: pass through container fps to filterswm42018-04-194-9/+9
| | | | | | | | | | | | This means vf_vapoursynth doesn't need a hack to work around the filter code, and libavfilter filters now actually get the frame_rate field on input pads set. The libavfilter doxygen says the frame_rate field is only to be set if the frame rate is known to be constant, and uses the word "must" (which probably means they really mean it?) - but ffmpeg.c sets the field to mere guesses anyway, and it looks like this normally won't lead to problems.
* f_lavfi: add an option to use old audio PTS handling for af_lavfiwm42018-04-151-0/+31
| | | | | The fix-pts option basically uses the old af_lavfi's (before filter rewrite) timestamp logic. The rest is explained in the manpage.
* audio: do not try to resample spdif datawm42018-04-151-0/+5
| | | | | | Normally we don't even try this, but in corner cases it can happen. For example when inserting lavcac3enc at runtime, and display-sync-resample was active.
* f_output_chain: fix typowm42018-04-151-1/+1
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* audio: change format negotiation, remove channel remix fudgingwm42018-04-153-229/+89
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The audio format neogitation code was pretty complicated, although the idea was simple: when the format changes (or on the first audio frame), filter only the new frame through the entire filter chain, discard the resulting frame, but use the format to initialize the AO. This was useful for "fudging" the channel remix behavior (upmix or downmix), and moving it before other filters. Apparently this was useful for things like DRC filters, which might work better in stereo, and which also can only achieve the desired volume levels by doing it before a downmix, which would modify the volume. This mechanism was introduced in commit 60048b7eb957b (which the commit message also describes as "idiotic heuristic"). Knowing the output format is inherently necessary for this, because otherwise we can't know what the hell the user defined filters will do. There were problems with robustness. Some filters needed more than one frame. Resampling in particular would discard initial audio at high resampling ratios. Some filters might drop audio intentionally (like clipping data on timestamp ranges). There were also allegations that some decoders output 0 length frames (although that is invalid in libavcodec). The state machine was excessively complex and hard to understand too. There are 3 things that could have been done: 1. Fix robustness problems by doing more heuristics, like repeating audio frames or simply decoding several frames. Since filters can behave differently, this would have added lots of complexity. 2. Make use of libavfilter's format negotiation, and add the same to mpv builtin filters. This is sort of annoying, because the format negotiation in libavfilter changes the state of the filters. It also reports only some parameters (mostly all for audio, but a lot of holes for video). It would remove some of the state machine, but not all. 3. Drop the channel remix fudging, and do the same as the video chain. This would not require format negotiation, but instead you can just filter the audio frames, and look what comes out of it. If nothing comes out, simply never create an AO. This commit selects option 3. It removes the remix fudging, which means the loss of a feature. Users can instead add "--af=format=channels=2" before their DRC filter, or something. I'm also considering changing the default for --audio-channels back to stereo, and downmix in the decoder or at the start of the filter chain, which would give the same results, except requiring more configuration. Implementation-wise, this is still a bit different from the video path. The VO always remains the same instance, while the AO might have to be recreated on configuration changes. This still requires explicit format change handling + draining old data, but by putting it into f_autoconvert, not much new code is needed.
* f_autoconvert: be less clever about running specific codepathswm42018-04-151-28/+14
| | | | | | | This tried to avoid running the audio/video functions depending on whether any of the audio or video related format restrictions were called (so the filter would show an error if a mismatching media type was passed in). It was a shit idea anyway, so fuck it.
* f_lavfi: use new libavfilter iteration APIwm42018-04-031-3/+5
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* f_decoder_wrapper: retry decoding if libavcodec returns invalid statewm42018-03-261-2/+7
| | | | | | | | | | | | | | | | | At least the libavcodec MediaCodec wrapper sometimes seems to break the libavcodec API, and does the following sequence: send_packet() -> EAGAIN receive_frame() -> EAGAIN send_packet() -> OK The libavcodec API never allows returning EAGAIN from both functions, so we discarded packets in this case. Change it to retrying decoding, for the sake of MediaCodec. Note that it could also happen due to internal bugs in the vd_lavc.c hw fallback code, but if there are any remaining, they should be fixed properly instead. Requested.
* f_hwtransfer: more detailed loggingwm42018-03-151-3/+4
| | | | This also switches the order, because that makes more sense.
* f_hwtransfer: fix a logic errorwm42018-03-151-2/+2
| | | | | Jesus Christ, how did I get this wrong, or never verified proper function. This fixes --vf=vdpaupp not working with yuv420p input.
* audio: improve behavior if filters output nothing during probingwm42018-02-211-1/+4
| | | | | | | | | | 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.
* video: fix --video-rotate in some caseswm42018-02-181-1/+1
| | | | Which idiot wrote this code? [Yeah, me.]
* filter: fix potential NULL pointer derefwm42018-02-161-1/+1
| | | | | | The rest of the function should be executed only if both are set. It seems like in practice this didn't happen yet with only one of them unset, but in theory it's possible. Found by Coverity.
* f_lavfi: extend filter help outputwm42018-02-131-1/+41
| | | | | | Also print type and help string. Also print AV_OPT_TYPE_CONST, which are like the mpv choice option type, except different. Print them as separate lines because FFmpeg usually has help strings for them too.
* filter: extend documentation commentswm42018-02-131-6/+43
| | | | Add more explanations, and also fix some blatantly wrong things.
* filter: simplify/fix external filter graph usagewm42018-02-131-34/+37
| | | | | | | | | | | | | | | | There was the following problem: if a filter graph had asynchronous filters, and the filter graph user did not call mp_filter_run() (and only accessed the mp_pins), then filtering could stall, because using mp_pin_out_request_data() only recursively invoked filtering if the data_requested flag wasn't already set. The latter can happen if a request was tried earlier but failed, and then an asynchronous filter actually produced output that would satisfy the request. Obviously, it has to invoke filtering again to get the requested frame. Fix this by organizing the code differently, and making sure to invoke mp_filter_run() on every request (if there's nothing to do, it doesn't do anything anyway). Simplify it a bit by removing things which are not needed, like connecting filter graphs with different root filters.
* f_lavfi: fix typo in commentwm42018-02-131-1/+1
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* filter: adjust root log prefixwm42018-02-131-2/+3
| | | | | | | | | | Avoids that the audio decoder shows up with a "[root/ad]" log prefix. This is an annoying consequence of mp_log_new(): if a log parent doesn't have a prefix with "!", it'll add the prefix to all mp_logs created from it. This should probably be fixed in the mp_log code itself, but doing so would be a big deal as we'd have to make sure all the other log prefixes are what we want. So work it around for now.
* video: make --deinterlace and HW deinterlace filters always deinterlacewm42018-02-131-1/+1
| | | | | | | | | | | | | | | | Before this, we made deinterlacing dependent on the video codec metadata (AVFrame.interlaced_frame for libavcodec). So even if --deinterlace=yes was set, we skipped deinterlacing if the flag wasn't set. This is very unreliable and there are many streams with flags incorrectly set. The potential problem is that this might upset people who alwase enabled deinterlace and hoped it worked. But it's likely these people were screwed by this setting anyway. The new behavior is less tricky and easier to understand, and this preferable. Maybe one day we could introduce a --deinterlace=auto, which does the right thing, but of course this would be hard to implement (esecially with hwdec). Fixes #5219.
* audio: move back PTS jump detection to before filter chainwm42018-02-132-1/+16
| | | | | | | | | | | 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.
* f_decoder_wrapper: fix log message incorrect for audiowm42018-02-051-1/+1
| | | | | This code is used by both video and audio, so the text should not talk about video.
* f_demux_in: give it a slightly better filter namewm42018-02-051-1/+1
| | | | Matters for logging.
* filter: don't randomly lose async wakeup notificationswm42018-02-051-5/+4
| | | | | | | | | | | | Another "what was I thinking" thing - destroying filters explicitly skipped async wakeups for no reason. These were notifications for filters that are not going to be destroyed too, and so their wakeup will be lost, leading to stalled playback. This is completely unnecessary and the special code can be removed. Fixes #5488. (This case destroyed all audio filters due to AO init failure, which could make clear out the f_demux_in.c wakeup for video, and "freeze" playback.)
* swresample: minor simplificationwm42018-02-031-7/+6
| | | | Cosmetic and no change in behavior. At least I think this looks simpler.
* swresample: remove unnecessary request for new inputwm42018-02-031-1/+2
| | | | | | | | | | | We called mp_pin_out_request_data() if there was input _and_ output. This is not how it should be: we should request new input only if output was requested, but we could not produce any output. On the other hand, the upper half of the process() function will request new input if output is required, but all input was consumed. But this requires calling mp_filter_internal_mark_progress(), as otherwise the general filter logic would not know that we can continue.
* swresample: actually reinit resampler on large speed changeswm42018-02-031-5/+13
| | | | | | | | | | | | | | | | | | | If the speed is changed by a large amount, we need to effectively change the output rate by a large amount, and swr_set_compensation() is apparently not designed to handle such large changes well. So it's better to reinitialize the resampler on all large changes. Also, strictly reinitialize the resampler if the rate changes, otherwise it could happen that libavresample (which does not automatically initialize resampling if avresample_set_compensation() is used) would never apply speed changes properly. Also document some conditions better that handle corner cases (remove the inline condition from the if gating the compensation code). It also appears that we crashed with very large compensation ratios (when raising audio speed quickly by keeping the "[" key down), and this commit accidentally mitigates it by not allowing large compensation.
* f_output_chain: remove unused got_input_eof fieldwm42018-02-032-4/+1
| | | | Was used by the player code before decoders were moved to filters.
* f_utils: fix leak in frame duration filterwm42018-02-031-0/+1
| | | | | vf_vapoursynth used this. Could cause a crash at VO uninit, if the leaked frame was allocated via VO DR.
* swresample: limit output size of audio frameswm42018-02-032-35/+58
| | | | | | | | | | | | | | Similar to the previous commit, and for the same reasons. Unlike with af_scaletempo, resampling does not have a natural frame size, so we set an arbitrary size limit on output frames. We add a new option to control this size, although I'm not sure whether anyone will use it, so mark it for testing only. Note that we go through some effort to avoid buffering data in libswresample itself. One reason is that we might have to reinitialize the resampler completely when changing speed, which drops the buffered data. Another is that I'm not sure whether the resampler will do the right thing when applying dynamic speed changes.
* filter: add/use a convenience functionwm42018-02-033-2/+12
| | | | | I guess this is generally useful for filters which buffer data internally.
* options: slightly improve filter help output for lavfi bridgewm42018-02-033-4/+79
| | | | | | | | | | | | | | --vf=help will now list libavfilter filters, and e.g. --vf=yadif=help will list libavfilter filter options. The latter is rather bare, because the AVOption API is really awful (holy shit how is it so bad), and would require us to handle _every_ option type manually. Alternatively we could call av_opt_show2(), which ffmpeg uses for help output in its CLI tools and which is much more detailed. But it's rather foreign and forces output through av_log(), so I don't really want to use it.
* audio: move to decoder wrapperwm42018-01-303-8/+92
| | | | | | | | | | | | | | | | 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.
* video: make decoder wrapper a filterwm42018-01-3010-6/+863
| | | | | | | | | | | | | | | | | | | | | | | | | Move dec_video.c to filters/f_decoder_wrapper.c. It essentially becomes a source filter. vd.h mostly disappears, because mp_filter takes care of the dataflow, but its remains are in struct mp_decoder_fns. One goal is to simplify dataflow by letting the filter framework handle it (or more accurately, using its conventions). One result is that the decode calls disappear from video.c, because we simply connect the decoder wrapper and the filter chain with mp_pin_connect(). Another goal is to eventually remove the code duplication between the audio and video path