This helps passing the channel layout correctly from decoder to audio
filter chain. (Because that part "reuses" the demuxer level codec
parameters, which is very disgusting.)
Note that ffmpeg stuff already passed the channel layout via
mp_copy_lav_codec_headers(). So other than easier dealing with the
demuxer/decoder parameters mess, there's no real advantage to doing
this.
Make the --channels option accept a channel map. Since simple numbers
map to standard layouts with the given number of channels, this is
downwards compatible. Likewise for demux_rawaudio.
This actually breaks audio for 5/6/8 channels. There's no reordering
done yet. The actual reordering will be done inside of af_lavrresample
and has to be made part of the format negotiation.
Anything this option did has been removed in the preceding 3 commits.
Note that even though these options sounded like a good idea (like
setting accuracy vs. speed tradeoffs), they were not really properly
implemented.
Switch the internal channel order to libavcodec's. If the channel number
mismatches at some point, use libavresample for up- or downmixing.
Remove the old af_pan automatic downmixing.
The libavcodec channel order should be equivalent to WAVEFORMATEX order,
at least nowadays. reorder_ch.h assumes that WAVEFORMATEX and libavcodec
might be different, but all defined channels have the same mappings.
Remove the downmixing with af_pan as well as the channel conversion with
af_channels from af.c, and prefer af_lavrresample for this. The
automatic downmixing behavior should be the same as before (if the
--channels option is set to 2, which is the default, the audio output
is forced to 2 channels, and libavresample does all downmixing).
Note that mpv still can't do channel layouts. It will pick the default
channel layout according to the channel count. This will be fixed later
by passing down the channel layout as well.
af_hrtf depends on the order of the input channels, so reorder to ALSA
(for which this code was written). This is better than changing the
filter code, which is more risky.
ao_pulse can accept waveext order directly, so set that as channel
mapping.
Make sure automatically inserted filters are removed on full reinit
(they are re-added later if they are really needed). Automatically
inserted filters were never explicitly removed, instead, it was
expected that redundant conversion filters detach themselves. This
didn't work if there were several chained format conversion filters,
e.g. s16le->floatle->s16le, which could result from repeated filter
insertion and removal. (format filters detach only if input format and
output format are the same.)
Further, the dummy filter (which exists only because af.c can't handle
an empty filter chain for some reason) could introduce bad conversions
due to how the format negotiation works. Change the code so that the
dummy filter never takes part on format negotiation. (It would be better
to fix format negotiation, but that would be much more complicated and
would involving fixing all filters.)
Simplify af_reinit() and remove the start audio filter parameter. This
means format negotiation and filter initialization is run more often,
but should be harmless.
This option can be used to selectively reset settings when playing the
next file in the playlist (i.e. restore mplayer and mplayer2 behavior).
Might remove this option again should it turn out that nobody uses it.
Consider:
mpv --volume 10 file1.mkv file2.mkv
Before this commit, the volume was reset to 10 when playing file2.mkv.
This was inconsistent to most other options. E.g. --brightness is a
rather similar case.
In general, settings should never be reset when playing the next file,
unless the option was explicitly marked file-local. This commit
corrects the behavior of the --volume and --mute options.
File local --volume still works as expected:
mpv --{ --volume 10 file1.mkv file2.mkv --}
This sets the volume always to 10 on playback start.
Move the m_config_leave_file_local() call down so that the mixer code
in uninit_player() can set the option volume and mute variables without
overwriting the global option values.
Another subtle issue is that we don't want to set volume if there's no
need to, which is why the user_set_volume/mute fields are introduced.
This is important because setting the volume might change the system
volume depending on other options.
Commit bc20f2c moved the variable default_max_pts_correction (which
backs the --mc option) to the MPOpts struct. The initializer was
forgotten when doing this, so it was left at 0. This disabled part of
the A/V sync mechanism. This was apparent when using ad_spdif (this
decoder has other A/V sync related problems, and thus triggers this
issue easily).
Closes#59.
Makes sure that seeking to a given time position shows the subtitle at
that position. This can fail if the subtitle packet is not close enough
to the seek target. Always enabled for hr-seeks, and can be manually
enabled for normal seeks with --mkv-subtitle-preroll.
This helps displaying subtitles correctly with ordered chapters. When
switching ordered chapter segments, a seek is performed. If the subtitle
is timed slightly before the start of the segment, it normally won't be
demuxed. This is a problem with all seeks, but in this case normal
playback is affected. Since switching segments always uses hr-seeks,
the code added by this commit is always active in this situation.
If no subtitles are selected or the subtitles come from an external
file, the demuxer should behave exactly as before this commit.
In the last cleanup round, this was accidentally changed from a store
option to int, and the option value was passed as flag value.
(Not that anyone needs/uses/cares about this option...)
The main problem with video PTS was that it wasn't very useful when
playing audio files with cover art. Using the audio time instead was an
obvious solution. Unfortunately, this leads to "inexact" reporting of
the playback time in paused mode, and audio is always ahead by small,
essentially random amounts of time ahead. This is possibly because the
times reported by AOs are not entirely accurate when paused (see commit
9b3bf76).
Switch back to video PTS, and use a simpler way to deal with the cover
art case: if the video has ended, use the audio PTS.
Also see commit f9a259e (and the commits referenced from there).
Trying to step over a segment boundary didn't work, and the video was
stuck at the end of the current chapter. At this point, both video and
audio of the segment has ended, and the segment switching code is going
to call seek() to go to the next segment (the part of the code in
run_playloop that uses end_is_chapter). However, this seek() is not
called if playback is paused, and the framestepping code always paused
before this code is run.
Move the framestepping code below the chapter switching code. The added
restart_playback condition makes sure the code is called only after at
least one video frame has been shown. Also don't reset the framestep
counter after seek. It's not needed, and removing it prevents full
unpausing when stepping over a segment boundary.
This also terminates playback when frame stepping at the end of the
file. The --keep-open option can be used to get the old behavior.
The OSX part of the Apple Remote was unmaintained for a long time and was not
working anymore. I tried to update the cookies to what the current versions of
OS X expect without much luck. I decided to remove it since Apple is not
including the IR receiver anymore in new hardware and it's clear that wifi
based remotes are the way to go.
A third party iOS app should be used in it's place. In the future we could look
into having a dedicated iOS Remote Control app like VLC and XBMC do.
The Linux side (`appleir.c`) was relatively tidy but it looks like LIRC can be
configured to work with any version of Apple Remote [1] and is more maintained.
[1] LIRC Apple Remote configs: http://lirc.sourceforge.net/remotes/apple/
Drawing the bar with vector drawings (instead with characters from the
OSD font) offers more flexibility and looks better. This also adds
chapter marks to the OSD bar, which are visible as small triangles on
the top and bottom inner border of the bar.
Change the default position of the OSD bar below the center of the
screen. This is less annoying than putting the bar directly into the
center of the view, where it obscures the video. The new position is
not quite on the bottom of the screen to avoid collisions with
subtitles.
The old centered position can be forced with ``--osd-bar-align-y=0``.
Also make it possible to change the OSD bar width/height with the new
--osd-bar-w and --osd-bar-h options.
It's possible that the new OSD bar renders much slower than the old
one. There are two reasons for this: 1. the character based bar
allowed libass to cache each character, while the vector drawing forces
it to redraw every time the bar position changes. 2., the bar position
is updated at a much higher granularity (the bar position is passed
along as float instead of as integer in the range 0-100, so the bar
will be updated on every single video frame).
gl_video.c contains all rendering code, gl_lcms.c the .icc loader and
creation of 3D LUT (and all LittleCMS specific code). vo_opengl.c is
reduced to interfacing between the various parts.
Makes the code a bit simpler to follow, at least in the "modern"
decoding path (update_video_nocorrect_pts() is used with old demuxers,
which don't return proper packets and need further parsing, so this code
looks less simple now).
There were complaints that ${fps} was printed as e.g. "23.98" instead of
"23.976". Since there's no way to format floats exactly _and_ in a user-
friendly way, just change the default precision for printing floats.
This way it's possible to retrieve correct information about video, like
actual width/height, which in general are available only after at least
one frame has been sent to the video output, such as dwidth/dheight.
mpv_identify.sh becomes a bit slower, because we let it decode enough
audio and video to fill the audio buffers and to send one frame to the
video output. Also, --playing-msg isn't shown anymore with --frames=0
(could be fixed by special-casing it, should this break any use cases).
Note that in some corner cases, like when the demuxer for some reason
returns lots of audio packets but no video packets at the start, but
video actually starts later, the --playing-msg will still be output
before video starts.
This has the same (useless) definition as frame stepping in audio-only
mode: one frame means one playloop iteration. (It's relatively useless,
because one playloop iteration has a random duration. But it makes
--frames=1 work, which is useful again.)
Add new properties "dwidth" and "dheight", which contain the video
size as known by the VO (not necessarily what the VO makes out of them,
i.e. without window scaling and panscan).
Some time ago, all old special-cased commands (like "volume 1" to change
volume by one) have been removed. These commands are still emulated
using simple text replacement. This emulation is done to not break
everyone's input.conf, especially because the input.conf provided by
standard mplayer* still uses the old commands.
Every use of a deprecated command prints a replacement warning, which
was visible only with -v. Make these warnings visible by default.
There's actually not much reason to do this, but since commands like
"volume 5 1" don't work anymore, it's better to be verbose about this.
Also simplify the replacement for "vo_fullscreen".
Normal text was set to gray foreground color. This didn't work for
terminals with white background.
Instead of setting a color for normal text, reset the color attributes.
This way, only errors and warnings are formatted differently.
Also change the default color for MSGL_HINT from bold white to yellow.
A recent change accidentally set the flags options to -1 (probably
confusing it with the defasult value?), which mistakenly set all flags
and rejected all option values (except 0).
We consider FFmpeg 1.x and Libav 0.9.x releases compatible. Support
for FFmpeg 0.9.x and Libav 0.8.x is considered infeasible and has been
dropped in the previous commits. The bits that break compatibility are
mainly the CodecID renaming (trivial, but would require nasty hacks
everywhere), the avcodec_encode_video2() function (missing in older
releases, mandatory in newer ones), and the resampler changes (older
releases miss lib{av,sw}resample, newer versions removed the
libavcodec resampler).
Remove some other compatibility bits that were needed to for releases
for which we drop support.
The comment about Libav 0.9 in compat/libav.h is incorrect and should
have been 0.8 (the symbol is present in Libav 0.9).
The old names have been deprecated a while ago, but were needed for
supporting older ffmpeg/libav versions. The deprecated identifiers
have been removed from recent Libav and FFmpeg git.
This change breaks compatibility with Libav 0.8.x and equivalent
FFmpeg releases.
Move them into per-instance structs. This should get rid of all global
variables in mplayer.c (not counting those referenced by cfg-mplayer.h).
In core/input/ar.c, just remove checking the slave_mode variable. I'm
not sure what this code was supposed to achieve, but slave mode is
broken, slave mode is actually infeasible on OSX (ar.c is completely OSX
specific), and the correct way of doing this would be to disable this
input device per command line switch.
