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* video: switch from using display aspect to sample aspectwm42015-12-194-21/+19
| | | | | | | | | | | | | | | | MPlayer traditionally always used the display aspect ratio, e.g. 16:9, while FFmpeg uses the sample (aka pixel) aspect ratio. Both have a bunch of advantages and disadvantages. Actually, it seems using sample aspect ratio is generally nicer. The main reason for the change is making mpv closer to how FFmpeg works in order to make life easier. It's also nice that everything uses integer fractions instead of floats now (except --video-aspect option/property). Note that there is at least 1 user-visible change: vf_dsize now does not set the display size, only the display aspect ratio. This is because the image_params d_w/d_h fields did not just set the display aspect, but also the size (except in encoding mode).
* sub: remove unused video width/height headerswm42015-12-181-3/+0
| | | | | | | Apparently, this was replaced by the SD_CTRL_SET_VIDEO_PARAMS set dimensions. But I can't find out when this happened - possibly, these fields were never used by sd_lavc.c, and only by the (long removed) MPlayer dvdsub decoder.
* player: init playback speed correctlywm42015-12-101-0/+2
| | | | | Usually not a problem, but could not be initialized early enough in some corner cases.
* sub: allow feeding bitmap subs in advancewm42015-12-051-2/+2
| | | | | | | | | | | | | | | | | | | | | | | Until now, feeding packets to the decoder in advance was done for text subtitles only. This was possible because libass buffers all subtitle data anyway (in ASS_Track). sd_lavc, responsible for bitmap subs, does not do this. But it can buffer a small number of subtitle frames ahead. Enable this. Repurpose the sub_accept_packets_in_advance(). Instead of "can take all packets" it means "can take 1 packet" now. (The old meaning is still needed locally in dec_sub.c; keep it there.) It asks the decoder whether there is place for at least 1 subtitle packet. sd_lavc implements it and returns true if its internal fixed-size subtitle queue still has a free slot. (The implementation of this in dec_sub.c isn't entirely clean. For one, decode_chain() ignores this mechanism, so it's implied that bitmap subtitles do not use the subtitle filter chain in any advanced way.) Also fix 2 bugs in the sd_lavc queue handling. Subtitles must be checked in reverse, because the first entry will often have endpts==NOPTS, which would always match. alloc_sub() must cycle the queue buffer, because it reuses memory allocations (like sub.imgs) by design.
* player: remove redundant checkwm42015-12-051-1/+1
| | | | Found by Coverity.
* player: don't make display-sync panic on timestamp discontinuitieswm42015-12-041-2/+2
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* player: resync audio only on larger timestamp discontinuitieswm42015-12-041-2/+2
| | | | | | | | | Helps with files that have occasional broken timestamps. For larger discontinuities, e.g. caused by actual timestamp resets, we still want to realign audio. (I guess in general, this should be removed and replaced by a more general resync-on-desync logic, but not now.)
* client API: disallow masking MPV_EVENT_SHUTDOWNwm42015-12-021-0/+2
| | | | | | | | | This makes no sense, because the client is obligated to react to this event. This also happens to fix a deadlock with JSON IPC clients sending "disable_event all", because MPV_EVENT_SHUTDOWN was used to stop the thread driving the socket connection (fixes #2558).
* osd: do not let OSD messages overwrite --osd-msgN textwm42015-11-291-14/+8
| | | | | | | | | Requested. Don't overwrite permanent OSD text set with e.g. --osd-msg1. Instead, append the OSD message to it (on the next line). Note that with --osd-msg1, seeking will still overwrite the OSD with the playback status for a while. If you do not want this, use --osd-msg3 --osd-level=3 instead.
* vo_opengl: fix interpolation with display-syncwm42015-11-281-1/+2
| | | | | | | | | | | | | | | | | | | | At least I hope so. Deriving the duration from the pts was not really correct. It doesn't include speed adjustments, and becomes completely wrong of the user e.g. changes the playback speed by a huge amount. Pass through the accurate duration value by adding a new vo_frame field. The value for vsync_offset was not correct either. We don't need the error for the next frame, but the error for the current one. This wasn't noticed because it makes no difference in symmetric cases, like 24 fps on 60 Hz. I'm still not entirely confident in the correctness of this, but it sure is an improvement. Also, remove the MP_STATS() calls - they're not really useful to debug anything anymore.
* player: fix commit 50bb209awm42015-11-281-1/+1
| | | | Well, this was stupid.
* vo: change vo_frame field unitswm42015-11-271-1/+2
| | | | | | | This was just converting back and forth between int64_t/microseconds and double/seconds. Remove this stupidity. The pts/duration fields are still in microseconds, but they have no meaning in the display-sync case (also drop printing the pts field from opengl/video.c - it's always 0).
* player: always disable display-sync on desyncswm42015-11-272-22/+13
| | | | | | | | | | | | | | | Instead of periodically trying to enable it again. There are two cases that can happen: 1. A random discontinuity messed everything up, 2. Things are just broken and will desync all the time Until now, it tried to deal with case 1 - but maybe this is really rare, and we don't really need to care about it. On the other hand, case 2 is kind of hard to diagnose if the user doesn't use the terminal. Seeking will reenable display-sync, so you can fix playback if case 1 happens, but still get predictable behavior in case 2.
* player: make display-vdrop mode do what the manpage claimswm42015-11-261-4/+7
| | | | | Don't change video speed in this mode, which is closer to the claim on the manpage that it's close to the behavior of the "audio" mode.
* command, vo: add estimated-display-fps propertywm42015-11-251-1/+30
| | | | | | | | | | This is simply the average refresh rate. Including "bad" samples is actually an advantage, because the property exists only for informational purposes, and will reflect problems such as the driver skipping a vsync. Also export the standard deviation of the vsync frame duration (normalized to the range 0-1) as vsync-jitter property.
* player: log some more display-sync informationwm42015-11-251-3/+6
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* win32: don't show progress indicator in idle modeJames Ross-Gowan2015-11-231-1/+3
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* command: export some per-video-frame informationwm42015-11-221-0/+25
| | | | Utterly useless, but requested. Fixes #2444.
* player: replace mistimed-frame-count with vsync-ratio on status linewm42015-11-181-1/+3
| | | | I think this is much more informative. Maybe.
* player: make timeline switching slightly nicerwm42015-11-183-30/+27
| | | | But not much.
* player: remove OSD subtitle render pathwm42015-11-174-63/+12
| | | | | | | | | | | | | | | | | | | This was used with --no-sub-ass (aka --no-ass). This option (which is not yet removed) strips all styling from the subtitles, and renders them as plaintext only. For some reason, it originally seemed convenient to reuse all the OSD text rendering code (osd_libass.c). While this was indeed simple, it had a bad influence on the rest of the code. For example, it had to decide whether to go through the OSD code path, or the proper subtitle renderer in sd_ass.c. Kill the OSD subtitle renderer. Reimplement --no-sub-ass and also "secondary" subtitles in sd_ass.c. fill_plaintext() contains some rather minor code duplication with osd_libass.c for setting up a dummy ASS_Event and escaping the stripped text. Since sd_ass.c already has to handle "normal" text subtitles, and has code for stripping ASS tags, this remains all relatively simple. Remove all the unnecessary crap from the rest of the code.
* player: use demuxer ts offset to simplify timeline ts handlingwm42015-11-168-45/+15
| | | | | | | | | Use the demux_set_ts_offset() added in the previous commit to base each timeline segment to use timestamps according to its relative position within the overall timeline. As a consequence we don't need to care about these timestamps anymore, and everything becomes simpler. (Another minor but delicious nugget of sanity.)
* player: handle rebasing start time differentlywm42015-11-166-37/+30
| | | | | | | | | | | | | | | | Most of this is explained in the DOCS additions. This gives us slightly more sanity, because there is less interaction between the various parts. The goal is getting rid of the video_offset entirely. The simplification extends to the user API. In particular, we don't need to fix missing parts in the API, such as the lack for a seek command that seeks relatively to the start time. All these things are now transparent. (If someone really wants to know the real timestamps/start time, new properties would have to be added.)
* win32: support taskbar button progress indicatorMartin Herkt2015-11-153-0/+23
| | | | | | | | | | | This adds support for the progress indicator taskbar extension that was introduced with Windows 7 and Windows Server 2008 R2. I don’t like this solution because it keeps its own state and introduces another VOCTRL, but I couldn’t come up with anything less messy. closes #2399
* player: account for minor VO underrunswm42015-11-141-2/+2
| | | | | | | | | If the player sends a frame with duration==0 to the VO, it can trivially underrun. Don't panic, but keep the correct time. Also, returning the absolute time from vo_get_next_frame_start_time() just to turn it into a float with relative time was silly. Rename it and make it return what the caller needs.
* player: remove unused fieldwm42015-11-142-2/+0
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* player: fix audio drift computation at different playback speedswm42015-11-141-8/+9
| | | | | This computed nonsense if the user set a playback speed other than 1 (in addition to the display-sync speed change).
* player: stricter framedrop thresholdwm42015-11-131-3/+2
| | | | | | 80ms allowable desync was a bit too much. It'd allow for a range of 160ms, which everyone can notice. It might also be a bother to apply compensation resampling speed for that long.
* player: try to compensate actual audio driftwm42015-11-132-0/+41
| | | | | | | | | | | | | | | | We always let audio slowly desync until a threshold is reached, and then pushed it back by applying a maximum compensation speed. Refine what comes afterwards: instead of playing with the nominal video speed, use the actual required audio speed for keeping sync as measured by the A/V difference. (The "actual" speed is the ideal speed with A/V differences added.) Although this works in theory, it's somewhat questionable how much this works in practice. The ideal time value is actually not exact, but is the time at which the frame is scheduled (could be compensated by using the time_left calculations in handle_display_sync_frame()). It doesn't account for speed changes or catastrophic discontinuities. It uses only 10 past frames.
* player: change display-sync audio speed only if neededwm42015-11-131-38/+48
| | | | | | | | | | As long as it's within the desync tolerance, do not change the audio speed at all for resampling. This reduces speed changes which might be caused by jittering timestamps and similar cases. (While in theory you could just not care and change speed every single frame, I'm afraid that such changes could possibly cause audio artifacts. So better just avoid it in the first place.)
* player: remove display_sync_disable_counterwm42015-11-132-11/+8
| | | | We can implement it differently and drop a tiny bit of state.
* command: add vsync-ratio propertywm42015-11-133-6/+33
| | | | | | | | This is very "illustrative", unlike the video-speed-correction property, and thus useful. It can also be used to observe scheduling errors, which are not detected by the core. (These happen due to rounding errors; possibly not evne our fault, but coming from files with rounded timestamps and so on.)
* player: compute required display-sync speed change differentlywm42015-11-131-22/+36
| | | | | | | | | | | | | Instead of looking at the current frame duration for the intended speedup, look at all past frames, and find a good average speed. This ties in with not wanting to average _all_ frame durations, which doesn't make sense in VFR situations. This is currently done in the most naive way possible, but already sort of works for VFR which switches between frame durations that are integer multiples of a base rate. Certainly more improvements could be made, such as trying to adjust directly on FPS changes, instead of averaging everything, but for now this is not needed at all.
* player: smooth out frame durations by averaging themwm42015-11-131-1/+1
| | | | | | | | | | | | | | | | | | Helps somewhat with muxer-rounded timestamps. There is some danger that this introduces a timestamp drift. But since they are averaged values (unlike as when using an incorrect container framerate hint), any potential drift shouldn't be too brutal, or compensate itself soon. So I won't bother yet with comparing the results with the real timestamp, unless we run into actual problems. Of course we still prefer potentially real timestamps over the approximated ones. But unless the timestamps match the container FPS, we can't know whether they are (no, checking whether the they have microsecond components would be cheating). Perhaps in future, we could let the demuxer export the timebase - if the timebase is not 1000 (or divisible by it), we know that millisecond-rounded timestamps won't happen.
* player: refactor display-sync frame duration calculationswm42015-11-135-134/+98
| | | | | | | | | | | | | | | | | | | | | | | | | | | Get rid of get_past_frame_durations(), which was a bit too messy. Add a past_frames array, which contains the same information in a more reasonable way. This also means that we can get the exact current and past frame durations without going through awful stuff. (The main problem is that vo_pts_history contains future frames as well, which is needed for frame backstepping etc., but gets in the way here.) Also disable the automatic disabling of display-sync if the frame duration changes, and extend the frame durations allowed for display sync. To allow arbitrarily high durations, vo.c needs to be changed to pause and potentially redraw OSD while showing a single frame, so they're still limited. In an attempt to deal with VFR, calculate the overall speed using the average FPS. The frame scheduling itself does not use the average FPS, but the duration of the current frame. This does not work too well, but provides a good base for further improvements. Where this commit actually helps a lot is dealing with rounded timestamps, e.g. if the container framerate is wrong or unknown, or if the muxer wrote incorrectly rounded timestamps. While the rounding errors apparently can't be get rid of completely in the general case, this is still much better than e.g. disabling display-sync completely just because some frame durations go out of bounds.
* player: always require a future frame with display-sync enabledwm42015-11-131-2/+6
| | | | | | We need a frame duration even on start, because the number of vsyncs the frame is shown is predetermined. (vo_opengl actually makes use of this property in certain cases.)
* command: rename vo-missed-frame-count propertywm42015-11-132-13/+7
| | | | | | | | | "Missed" implies the frame was dropped, but what really happens is that the following frame will be shown later than intended (due to the current frame skipping a vsync). (As of this commit, this property is still inactive and always returns 0. See git blame for details.)
* player: less naive roundingwm42015-11-111-1/+1
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* player: silence sporadic error messages on audio initwm42015-11-101-1/+1
| | | | | | | | | | | | When the audio format is not known yet and the audio chain is still initializing, filter reinit will fail. Normally, attempts to reinitialize filters at this stage should be rare (e.g. user commands editing the filter chain). But it sometimes happened with track switching in combination with the video code calling update_playback_speed() at arbitrary times. Get rid of the message by not trying to change the filters for the sake of playback speed update while decoding is still being initialized.
* external_files: deduplicate bstr functionsKevin Mitchell2015-11-091-20/+4
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* command: make display-fps property writablewm42015-11-091-7/+9
| | | | | | | Has the same function as setting the option. This commit changes the property in a bunch of other ways. For example if the VO is not created, it will return the option value.
* player: use input instead of output format for spdif checkwm42015-11-041-1/+1
| | | | | | This check disables the display-sync resample method. If the filters convert PCM to AC3, we can still insert a filter to change speed. This is because filters are inserted at the beginning of the filter chain.
* audio: do not require full audio chain reinit for speed changeswm42015-11-041-57/+66
| | | | | | | | | | | | | | | 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.
* player: move audio speed adjustment codewm42015-11-041-54/+60
| | | | | | | | | Move it (in a cosmetic sense), and also move its invocation to below all the video handling. All other changes remain cosmetic, including moving the framedrop calculation code, and getting rid of the video_speed_correction variable.
* audio: strictly align audio on spdif frameswm42015-11-041-3/+7
| | | | | | 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.
* options: handle terminal/logging settings eagerlywm42015-11-041-0/+1
| | | | | | | | | | | Update msg.c state immediately if a terminal or logging setting is set. Until now, this was delayed until mp[v]_initialize() was called. When using the client API, you could easily miss logged error messages, even when logging was initialized early on by calling mpv_request_log_messages(). (Properties can't be used for this either, because properties do not work before mpv_initialize().)
* player: fix display-sync adrop speed limitingwm42015-11-041-1/+2
| | | | Commit 49d94853 worked only at the start of playback.
* player: limit speed change in display-sync adrop modewm42015-11-032-0/+9
| | | | | | | | | | Discontinuities (like toggling fullscreen) can cause multiple frames to be dropped in succession, which sounds very weird. It's better to drop some video frames instead to compensate for larger desyncs. We roughly base it on the maximum allowed speed changes (audio change is "additional" to the video change to account for deviations when playing at max. video speed change).
* player: another fix to A/V difference calculation in display-sync modewm42015-11-011-1/+1
| | | | | | | update_av_diff() works on the timestamps, while time_left is in real time. When playing at not-1 speed, these are very different, and cause the A/V difference to jitter. Fix this by scaling the expected A/V desync to the correct range.
* video: fix another A/V difference bug in display-sync modewm42015-10-311-2/+3
| | | | | | | | | | | | | | | This didn't show up with cases where the frame pattern has a cycle of 1 or 2 like it is the case with 24-on-24 fps, or 24-on-60 fps. It did show up with 25-on-60 fps. (We don't slow down 25 fps video to 24 on default settings.) In this case, we must not add the timing error of the next frame to the A/V difference estimation of the current frame. Use the previous timing error instead. This is another bug resulting from the confusion about whether we calculate parameters for the currently playing frame, or the one we're about to queue.
* command: add mistimed-frame-count propertywm42015-10-303-3/+18
| | | | | Does what the manpage says. This is a replacement incrementing the dropped frame counter (see previous commit).
* video: fix framedrop accounting in display-sync modewm42015-10-301-2/+0
| | | | | | | | | | | | | | Commit a1315c76 broke this slightly. Frame drops got counted multiple times, and also vo.c was actually trying to "render" the dropped frame over and over again (normally not a problem, since frames are always queued "tightly" in display-sync mode, but could have caused 100% CPU usage in some rare corner cases). Do not repeat already dropped frames, but still treat new frames with num_vsyncs==0 as dropped frames. Also, strictly count dropped frames in the VO. This means we don't count "soft" dropped frames anymore (frames that are shown, but for fewer vsyncs than intended). This will be adjusted in the next commit.
* player: raise display sync desync tolerancewm42015-10-281-5/+2
| | | | | | | Bump it to 80, and 2 vsyncs. This is another measure against vsync jitter. Admittedly this is a bit simplistic (and we should probably estimate a stable estimated vsync phase instead), but for now this will do.
* player: reset AO stats on pause and other discontinuitieswm42015-10-281-1/+3
| | | | It's annoying.
* player: simplify display-adrop mode safeguardwm42015-10-281-8/+1
| | | | | | It's not needed, because the additional data is not appended, but is the total size of the audio buffer. The maximum size is the static audio drop size (or twice, if the audio is duplicated).
* player: minor refactor for A/V diff computationwm42015-10-281-19/+27
| | | | | | | | Calculate the A/V difference directly in the display sync code, instead of the awkward current way, which reuses the fields for audio sync. We still set time_frame, because it makes falling back to audio sync somewhat smoother.
* player: fix display sync A/V difference estimation on dropswm42015-10-281-0/