This allows getting the log at all with --no-terminal and without having
to retrieve log messages manually with the client API. The log level is
hardcoded to -v. A higher log level would lead to too much log output
(huge file sizes and latency issues due to waiting on the disk), and
isn't too useful in general anyway. For debugging, the terminal can be
used instead.
Fixes#1472.
(Maybe these options should have been named --autofit-max and
--autofit-min, but since --autofit-larger already exists, use
--autofit-smaller for symmetry.)
--sub-scale-by-window=no attempts to keep subs always at the same pixel
size.
The implementation is a bit all over the place, because it compensates
already done scaling by an inverse scale factor, but it will probably do
its job.
Fixes#1424. (The semantics and name of --sub-scale-with-window are
kept, and this adds a new option - the name is confusingly similar, but
it's actually analogue to --osd-scale-by-window.)
This attempts to increase user-friendliness by excluding useless tags.
It should be especially helpful with mp4 files, because the FFmpeg mp4
demuxer adds tons of completely useless information to the metadata.
Fixes#1403.
- --lua and --lua-opts change to --script and --script-opts
- 'lua' default script dirs change to 'scripts'
- DOCS updated
- 'lua-settings' dir was _not_ modified
The old lua-based names/dirs still work, but display a warning.
Signed-off-by: wm4 <wm4@nowhere>
Makeshift-solution for working around certain fontconfig issues.
With --use-text-osd=no, libass and fontconfig won't be initialized, and
fontconfig won't block everything with scanning for fonts.
It's passed with the '--format' option to youtube-dl.
If it isn't set, we don't pass '--format best' so that youtube-dl can
use the options from its configuration file.
Signed-off-by: wm4 <wm4@nowhere>
Probably needs to be polished a bit more. Also, might require a key
binding that can set/clear the loop points in a more intuitive way.
For now, something like this can be put into input.conf to use it:
ctrl+y set ab-loop-a ${time-pos} # set A
ctrl+x set ab-loop-b ${time-pos} # set B
ctrl+c set ab-loop-a no # clear (mostly)
Fixes#1241.
Make the changes started in commit c827ae5f more eloborate, and provide
an option to control the amount of data read before the seek-target. To
achieve this, rewrite the loop that finds the lowest still acceptable
target cluster. It is now searched by time instead of file position. The
behavior (both with and without preroll option) may be different from
before this change, although it shouldn't be worse.
The change demux_mkv_read_cues() fixes a bug: when seeking after playing
normally, the code would erroneously assume that durations are set. This
doesn't happen if the first operation after loading was a seek instead
of playback.
The main need I see for this is with libmpv - it would be confusing if
some application showed up as "mpv" on whateverthehell PulseAudio uses
it for (generally it does show up on various PA GUI tools).
Note that you can't pass .cue or .edl files to it, at least not yet.
Requested in context of allowing to specify custom chapters. For that
to work well, we probably need to add some sort of chapter metadata
pseudo-demuxer.
This is probably what libmpv users want; and it also improves error
reporting (or we'd have to add a way to communicate such mid-playback
failures as events).
No development activity (or even any sign of life) for almost a year.
A replacement based on youtube-dl will probably be provided before the
next mpv release. Ask on the IRC channel if you want to test.
Simplify the Lua check too: libquvi linking against a different Lua
version than mpv was a frequent issue, but with libquvi gone, no
direct dependency uses Lua, and such a clash is rather unlikely.
Apparently using the stream index is the best way to refer to the same
streams across multiple FFmpeg-using programs, even if the stream index
itself is rarely meaningful in any way.
For Matroska, there are some possible problems, depending how FFmpeg
actually adds streams. Normally they seem to match though.
This is the first of a series of commits that will change the Cocoa way in a
way that is easily embeddable inside parent views. To reach that point common
code must avoid referencing the parent NSWindow since that could be the host
application's window.
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.
This would play some silence in case video was slower than audio. If
framedropping is already enabled, there's no other way to keep A/V
sync, short of changing audio playback speed (which would give worse
results). The --audiodrop option inserted silence if there was more
than 500ms desync.
This worked somewhat, but I think it was a silly idea after all. Whether
the playback experience is really bad or slightly worse doesn't really
matter. There also was a subtle bug with PTS handling, that apparently
caused A/V desync anyway at ridiculous playback speeds.
Just remove this feature; nobody is going to use it anyway.
It's just confusing; users are encouraged to edit input.conf instead
(changing the argument to the "add" command).
Update input.conf to keep the old behavior.
Until now, you could override only level 3 with --osd-status-msg. Extend
this, add add --osd-msg1 to --osd-msg3 (one for each OSD level). OSD
level 0 always means disable OSD, so that isn't included.
--osd-msg3 corresponds to --osd-status-msg, but they're not exactly the
same. To allow more customization, --osd-msgN do not include the OSD
symbol. The symbol can be manually added with "${osd-sym-cc}". We keep
the "old" option for some short-term compatibility.
--osd-msg1 should be particularly useful; for example you could do:
--osd-msg1='${?pause==yes:${osd-sym-cc}}'
to display a "paused" symbol when paused, and nothing during normal
playback. (Although admittedly, the syntax is quite a bit of work.)
--hls-bitrate=min/max lets you select the min or max bitrate. That's it.
Something more sophisticated might be possible, but is probably not even
worth the effort.
This inserts an automatic conversion filter if a Matroska file is marked
as 3D (StereoMode element). The basic idea is similar to video rotation
and colorspace handling: the 3D mode is added as a property to the video
params. Depending on this property, a video filter can be inserted.
As of this commit, extending mp_image_params is actually completely
unnecessary - but the idea is that it will make it easier to integrate
with VOs supporting stereo 3D mogrification. Although vo_opengl does
support some stereo rendering, it didn't support the mode my sample file
used, so I'll leave that part for later.
Not that most mappings from Matroska mode to vf_stereo3d mode are
probably wrong, and some are missing.
Assuming that Matroska modes, and vf_stereo3d in modes, and out modes
are all the same might be an oversimplification - we'll see.
See issue #1045.
Add the --cache-secs option, which literally overrides the value of
--demuxer-readahead-secs if the stream cache is active. The default
value is very high (10 seconds), which means it can act as network
cache.
Remove the old behavior of trying to pause once the byte cache runs
low. Instead, do something similar wit the demuxer cache. The nice
thing is that we can guess how many seconds of video it has cached,
and we can make better decisions. But for now, apply a relatively
naive heuristic: if the cache is below 0.5 secs, pause, and wait
until at least 2 secs are available.
Note that due to timestamp reordering, the estimated cached duration
of video might be inaccurate, depending on the file format. If the
file format has DTS, it's easy, otherwise the duration will seemingly
jump back and forth.
--demuxer-readahead-secs now controls how much the demuxer should
readahead by an amount of seconds. This is based on the raw packet
timestamps. It's not always very exact. For example, h264 in Matroska
does not store any linear timestamps (only PTS values which are going
to be reordered by the decoder), so this heuristic is usually off by
several hundred milliseconds.
The decision whether to readahead is basically OR-ed with the other
--demuxer-readahead-packets options. Change the manpage descriptions
to subtly convey these semantics.
Since the display FPS is currently detected on X11 only (and even there
it's known to be wrong on certain setups), it seems like a good idea to
make this user-configurable.
The VO is run inside its own thread. It also does most of video timing.
The playloop hands the image data and a realtime timestamp to the VO,
and the VO does the rest.
In particular, this allows the playloop to do other things, instead of
blocking for video redraw. But if anything accesses the VO during video
timing, it will block.
This also fixes vo_sdl.c event handling; but that is only a side-effect,
since reimplementing the broken way would require more effort.
Also drop --softsleep. In theory, this option helps if the kernel's
sleeping mechanism is too inaccurate for video timing. In practice, I
haven't ever encountered a situation where it helps, and it just burns
CPU cycles. On the other hand it's probably actively harmful, because
it prevents the libavcodec decoder threads from doing real work.
Side note:
Originally, I intended that multiple frames can be queued to the VO. But
this is not done, due to problems with OSD and other certain features.
OSD in particular is simply designed in a way that it can be neither
timed nor copied, so you do have to render it into the video frame
before you can draw the next frame. (Subtitles have no such restriction.
sd_lavc was even updated to fix this.) It seems the right solution to
queuing multiple VO frames is rendering on VO-backed framebuffers, like
vo_vdpau.c does. This requires VO driver support, and is out of scope
of this commit.
As consequence, the VO has a queue size of 1. The existing video queue
is just needed to compute frame duration, and will be moved out in the
next commit.
Almost nothing was left of it.
The only thing this commit actually removes is support for reading
input commands from stdin. But you can emulate this via:
--input-file=/dev/stdin --input-terminal=no
However, this won't work on Windows. Just use a named pipe.
This adds a thread to the demuxer which reads packets asynchronously.
It will do so until a configurable minimum packet queue size is
reached. (See options.rst additions.)
For now, the thread is disabled by default. There are some corner cases
that have to be fixed, such as fixing cache behavior with webradios.
Note that most interaction with the demuxer is still blocking, so if
e.g. network dies, the player will still freeze. But this change will
make it possible to remove most causes for freezing.
Most of the new code in demux.c actually consists of weird caches to
compensate for thread-safety issues (with the previously single-threaded
design), or to avoid blocking by having to wait on the demuxer thread.
Most of the changes in the player are due to the fact that we must not
access the source stream directly. the demuxer thread already accesses
it, and the stream stuff is not thread-safe.
For timeline stuff (like ordered chapters), we enable the thread for the
current segment only. We also clear its packet queue on seek, so that
the remaining (unconsumed) readahead buffer doesn't waste memory.
Keep in mind that insane subtitles (such as ASS typesetting muxed into
mkv files) will practically disable the readahead, because the total
queue size is considered when checking whether the minimum queue size
was reached.
For remarks, pretty much see the manpage additions. Could help with
network streams that require too much seeking (maybe), or might be
extended to help with the use case of watching and downloading a file
at the same time.
In general, it might be a useless feature and could be removed again.
Also clarify the semantics.
It seems --idx didn't do anything. Possibly it used to change how the
now removed legacy demuxers like demux_avi used to behave. Or maybe
it was accidental.
--forceidx basically becomes --index=force. It's possible that new
index modes will be added in the future, so I'm keeping it
extensible, instead of e.g. creating --force-index.
Does anyone actually use this?
For now, update it, because it's the only case left where an option
points to a global variable (and not a struct offset).
Similar to previous commits.
This also renames --doubleclick-time to --input-doubleclick-time, and
--key-fifo-size to --input-key-fifo-size. We could keep the old names,
but these options are very obscure, and renaming them seems better for
consistency.
Additionally to removing the global variables, this makes the options
more uniform. --ssf-... becomes --sws-..., and --sws becomes --sws-
scaler. For --sws-scaler, use choices instead of magic integer values.
Pretty much nothing changes, but using -tv-scan with suboptions doesn't
work anymore (instead of "-tv-scan x" it's "-tv scan-x" now). Flat
options ("-tv-scan-x") stay compatible.
This simply writes the file name as a comment to the top of the watch later
config file.
It can be useful to the user for determining whether a watch later config file
can be manually removed (e.g. in case the corresponding media file has been
deleted) or not.
stream.start_pos was needed for optical media only, and (apparently) not
for very good reasons. Just get rid of it.
For stream_dvd, we don't need to do anything. Byte seeking was already
removed from it earlier.
For stream_cdda and stream_vcd, emulate the start_pos by offsetting the
stream pos as seen by the rest of mpv.
The bits in discnav.c and loadfile.c were for dealing with the code
seeking back to the start in demux.c. Handle this differently by
assuming the demuxer is always initialized with the stream at start
position, and instead seek back if initializing the demuxer fails.
Remove the --sb option, which worked by modifying stream.start_pos. If
someone really wants this option, it could be added back by creating a
"slice" stream (actually ffmpeg already has such a thing).
Some options change from percentages to number of kilobytes; there are
no cache options using percentages anymore.
Raise the default values. The cache is now 25000 kilobytes, although if
your connection is slow enough, the maximum is probably never reached.
(Although all the memory will still be used as seekback-cache.)
Remove the separate --audio-file-cache option, and use the cache default
settings for it.
This allows disabling of decoder framedrop during hr-seek.
It's basically another useless option, but it will help exploring
whether this framedropping really makes seeking faster, or whether
disabling it helps with precise seeking (especially frame backstepping).
Also remove MSGL_SMODE and friends.
Note: The indent in options.rst was added to work around a bug in
ReportLab that causes the PDF manual build to fail.
This collects statistics and other things. The option dumps raw data
into a file. A script to visualize this data is included too.
Litter some of the player code with calls that generate these
statistics.
In general, this will be helpful to debug timing dependent issues, such
as A/V sync problems. Normally, one could argue that this is the task of
a real profiler, but then we'd have a hard time to include extra
information like audio/video PTS differences. We could also just
hardcode all statistics collection and processing in the player code,
but then we'd end up with something like mplayer's status line, which
was cluttered and required a centralized approach (i.e. getting the data
to the status line; so it was all in mplayer.c). Some players can
visualize such statistics on OSD, but that sounds even more complicated.
So the approach added with this commit sounds sensible.
The stats-conv.py script is rather primitive at the moment and its
output is semi-ugly. It uses matplotlib, so it could probably be
extended to do a lot, so it's not a dead-end.
This re-allows the previous behaviour of being able to reencode with
metadata removed, which is useful when encoding "inconsistently" tagged
data for a device/player that shows file names when tags are not
present.
Will be helpful to track down strange wait times and such issues, as
well when you have develop something timing related. (Then you may print
timestamps in your debug output, and the --msgtime timestamps will help
giving context.)
The values set by this new option can be queried by Lua scripts using
the mp.getopt() function. The function takes a string parameter, and
returns the value of the first key that matches. If no key matches, nil
is returned.
The terminal OSD code includes the handling of the terminal status line,
showing player OSD messages on the terminal, and showing subtitles on
terminal (the latter two only if there is no video window, or if
terminal OSD is forced).
This didn't handle some corner cases correctly. For example, showing an
OSD message on the terminal always cleared the previous line, even if
the line was an important message (or even just the command prompt, if
most other messages were silenced).
Attempt to handle this correctly by keeping track of how many lines the
terminal OSD currently consists of. Since there could be race conditions
with other messages being printed, implement this in msg.c. Now msg.c
expects that MSGL_STATUS messages rewrite the status line, so the caller
is forced to use a single mp_msg() call to set the status line.
Instead of littering print_status() all over the place, update the
status only once per playloop iteration in update_osd_msg(). In audio-
only mode, the status line might now be a little bit off, but it's
perhaps ok.
Print the status line only if it has changed, or if another message was
printed. This might help with extremely slow terminals, although in
audio+video mode, it'll still be updated very often (A-V sync display
changes on every frame).
Instead of hardcoding the terminal sequences, use
terminfo/termcap to get the sequences. Remove the --term-osd-esc option,
which allowed to override the hardcoded escapes - it's useless now.
The fallback for terminals with no escape sequences for moving the
cursor and clearing a line is removed. This somewhat breaks status line
display on these terminals, including the MS Windows console: instead of
querying the terminal size and clearing the line manually by padding the
output with spaces, the line is simply not cleared. I don't expect this
to be a problem on UNIX, and on MS Windows we could emulate escape
sequences. Note that terminal OSD (other than the status line) was
broken anyway on these terminals.
In osd.c, the function get_term_width() is not used anymore, so remove
it. To remind us that the MS Windows console apparently adds a line
break when writint the last column, adjust screen_width in terminal-
win.c accordingly.
Set the flag CODEC_FLAG_OUTPUT_CORRUPT by default. Note that there is
also CODEC_FLAG2_SHOW_ALL, which is older, but this seems to be ffmpeg
only.
Note that whether you want this enabled depends on the user. Some might
prefer that only good frames are output, while others want the decoder
to try as hard as possible to output _anything_. Since mplayer/mpv is
rather the kind of player that tries hard instead of being "clever", set
the new default to override libavcodec's default.
A nice way to test this is switching video tracks. Since mpv doesn't
wait for the next key frame, it'll start feeding the decoder with a
packet from the middle of the stream.
This is relatively hacky, but it's Christmas, so it's ok. This does two
things: 1. allow selecting two subtitle tracks, and 2. include a hack
that renders the second subtitle always as toptitle. See manpage
additions how to use this.
Until now, there were two functions to add input sources (stuff like
stdin input, slave mode, lirc, joystick). Unify them to a single
function (mp_input_add_fd()), and make sure the associated callbacks
always have a context parameter.
Change the lirc and joystick code such that they take store their state
in a context struct (probably worthless), and use the new mp_msg
replacements (the point of this refactoring).
Additionally, get rid of the ugly USE_FD0_CMD_SELECT etc. ifdeffery in
the terminal handling code.
Basically, reimplement --msglevel. Instead of making the new msg code
use the legacy code, make the legacy code use the reimplemented
functionality.
The handling of the deprecated --identify switch changes. It temporarily
stops working; this will be fixed in later commits.
The actual sub-options syntax (like --msglevel-vo=...) goes away, but I
bet nobody knew about this or used this anyway.
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.
