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Some of these have only limited use, and some of these have no use at
all. Remove them. They make maintainance harder and nobody needs them.
It's possible that many of the removed drivers were very useful a dozen
of years ago, but now it's 2012.
Note that some of these could be added back, in case they were more
useful than I thought. But right now, they are just a burden.
Reason for removal for each module:
vo_3dfx, vo_dfbmga, vo_dxr3, vo_ivtv, vo_mga, vo_s3fb,
vo_tdfxfb, vo_xmga, vo_tdfx_vid:
All of these are for very specific and outdated hardware. Some
of them require non-standard kernel drivers or do direct HW
access.
vo_dga: the most crappy and ancient way to get fast output on X.
vo_aa: there's vo_caca for the same purpose.
vo_ggi: this never lived, and is entirely useless.
vo_mpegpes: for DVB cards, I can't test this and it's crappy.
vo_fbdev, vo_fbdev2: there's vo_directfb2
vo_bl: what is this even? But it's neither important, nor alive.
vo_svga, vo_vesa: you want to use this? You can't be serious.
vo_wii: I can't test this, and who the hell uses this?
vo_xvr100: some Sun thing.
vo_xover: only useful in connection with xvr100.
ao_nas: still alive, but I doubt it has any meaning today.
ao_sun: Sun.
ao_win32: use ao_dsound or ao_portaudio instead.
ao_ivtv: removed along vo_ivtv.
Also get rid of anything SDL related. SDL 1.x is total crap for video
output, and will be replaced with SDL 2.x soon (perhaps), so if you
want to use SDL, write output drivers for SDL 2.x.
Additionally, I accidentally damaged Sun support, which made me
completely remove Sun/Solaris support. Nobody cares about this anyway.
Some left overs from previous commits removing modules were cleaned up.
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Conflicts:
.gitignore
bstr.c
cfg-mplayer.h
defaultopts.c
libvo/video_out.c
The conflict in bstr.c is due to uau adding a bstr_getline function in
commit 2ba8b91a97e7e8. This function already existed in this branch.
While uau's function is obviously derived from mine, it's incompatible.
His function preserves line breaks, while mine strips them. Add a
bstr_strip_linebreaks function, fix all other uses of bstr_getline, and
pick uau's implementation.
In .gitignore, change vo_gl3_shaders.h to use an absolute path
additional to resolving the merge conflict.
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There was some confusion about the "flags" field in demuxer packets.
Demuxers set it to either 1 or 0x10 to indicate a keyframe (and the
field was not used to indicate anything else). This didn't cause
visible problems because nothing read the value. Replace the "flags"
field with a boolean "keyframe" field. Set AV_PKT_FLAG_KEY based on
this field in packets fed to libavcodec video decoders (looks like PNG
and ZeroCodec are the only ones which depend on values from demuxer;
previously this was hardcoded to true for PNG).
Make demux_mf set the keyframe field in every packet. This matters for
PNG files now that the demuxer flag is forwarded to libavcodec.
Fix logic setting the field in demux_mkv. It had probably not been
updated when adding SimpleBlock support. This probably makes no
difference for any current practical use.
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Add support for using libavcodec decoders that do not have entries in
codecs.conf. This is currently only used with demux_lavf, and the
codec selection is based on codec_id returned by libavformat. Also
modify codec-related terminal output somewhat to make it use
information from libavcodec and avoid excessively long default output.
The new any-lavc-codec support is implemented with codecs.conf entries
that invoke vd_ffmpeg/ad_ffmpeg without directly specifying any
libavcodec codec name. In this mode, the decoders now instead select
the libavcodec codec based on codec_id previously set by demux_lavf
(if any). These new "generic" codecs.conf entries specify "status
buggy", so that they're tried after any specific entries with
higher-priority status.
Add new directive "anyinput" to codecs.conf syntax. This means the
entry will always match regardless of fourcc. This is used for the
above new codecs.conf entries (so the driver always gets to decide
whether to accept the input, and will fail init() if it can't find a
suitable codec in libavcodec). Remove parsing support for the obsolete
codecs.conf directive "cpuflags". This directive has not had any
effect and has not been used in default codecs.conf since many years
ago.
Shorten codec-related terminal output. When using libavcodec decoders,
show the libavcodec long_name field rather than codecs.conf "info"
field as the name of the codec. Stop showing the codecs.conf entry
name and "vfm/afm" name by default, as these are rarely needed;
they're now in verbose output only. Show "VIDEO:" line at VO
initialization rather than at demuxer open. This didn't really belong
in demuxer code; the new location may show more accurate values (known
after decoder has been opened) and works right if video track is
changed after initial demuxer open.
The vd.c changes (primarily done for terminal output changes) remove
round-to-even behavior from code setting dimensions based on aspect
ratio. I hope nothing depended on this; at least the even values were
not consistently guaranteed anyway, as the rounding code did not run
if the video file did not specify a nonzero aspect value.
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written_audio_pts() can be called even if no audio track is active (at
least through get_current_time() when there's no known video PTS).
This triggered a crash due to NULL dereference. Add a check to return
MP_NOPTS_VALUE if no audio track exists.
Also remove a questionable update_osd_msg() call from per-file
initialization code. The call was at a point where an audio track
might be selected but not properly initialized, possibly also causing
a crash if update_osd_msg() queries current position. I don't see any
reason why the call would have been needed; it should get called
anyway before OSD contents are actually used for the new file.
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Add an alternate mode for option parser objects (struct m_config)
which is not inherently tied to any particular instance of an option
value struct. Instead, this type or parsers can be used to initialize
defaults in or parse values into a struct given as a parameter. They
do not have the save slot functionality used for main player
configuration. The new functionality will be used to replace the
separate subopt_helper.c parsing code that is currently used to parse
per-object suboptions in VOs etc.
Previously, option default values were handled by initializing them in
external code before creating a parser. This initialization was done
with constants even for dynamically-allocated types like strings.
Because trying to free a pointer to a constant would cause a crash
when trying to replace the default with another value, parser
initialization code then replaced all the original defaults with
dynamically-allocated copies. This replace-with-copy behavior is no
longer supported for new-style options; instead the option definition
itself may contain a default value (new OPTDEF macros), and the new
function m_config_initialize() is used to set all options to their
default values. Convert the existing initialized dynamically allocated
options in main config (the string options --dumpfile, --term-osd-esc,
--input=conf) to use this. Other non-dynamic ones could be later
converted to use this style of initialization too.
There's currently no public call to free all dynamically allocated
options in a given option struct because I intend to use talloc
functionality for that (make them children of the struct and free with
it).
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When using an audio output without a native playback rate (such as
ao_pcm), the code plays audio further when the current write position
is behind video. After support for continuing audio after the end of
video was added, this could cause a deadlock: audio was not played
further, but neither was EOF triggered. Fix the code to properly
handle playback of remaining audio after video ends in the untimed
audio case (audio-only case was not affected, only the case where a
video stream exists but ends before the audio stream).
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Each option type had three separate operations to copy option values
between memory locations: copy between general memory locations
("copy"), copy from general memory to active configuration of the
program ("set"), and in the other direction ("save"). No normal option
depends on this distinction any more. Change everything to define and
use a single "copy" operation only. Change the special options
"include" and "profile", which depended on hacky option types, to be
special-cased directly in option parsing instead. Remove the now
unused option types m_option_type_func and m_option_type_func_param.
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Handle -v flags as a special case in command line preparsing stage,
and change the option entry into a dummy one. Specifying "v" in config
file no longer works (and the dummy entry shows an error in this
case); "msglevel" can still be used for that purpose. Because the flag
is now interpreted at an earlier parsing stage, it now affects the
printing of some early messages that were only affected by the
MPLAYER_VERBOSE environment variable before.
The main motivation for this change is to get rid of the last
CONF_TYPE_FUNC option.
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Conflicts:
bstr.c
bstr.h
libvo/cocoa_common.m
libvo/gl_common.c
libvo/video_out.c
mplayer.c
screenshot.c
sub/subassconvert.c
Merge of cocoa_common.m done by pigoz.
Picking my version of screenshot.c. The fix in commit aadf1002f8a will
be redone in a follow-up commit, as the original commit causes too many
conflicts with the work done locally in this branch, and other work in
progress.
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MSWindows does not have properly working support for detecting events
on file descriptors. As a result the current mplayer2 code does not
support waking up when new input events occur. Make the central
playloop wake up more often to poll for events; otherwise response
would be a lot laggier than on better operating systems during pause
or other cases where the process would not otherwise wake up.
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Make A/V sync at the start of playback with nonzero --delay behave the
same way as it does when seeking to the beginning later, meaning video
plays from the start and audio is truncated or padded with silence to
match timing. This was already the default behavior in case the
streams in the file started at different times, but not if the
mismatch was due to --delay. Trigger similar audio synchronization
when switching to a new video stream. Previously, switching a video
stream on after playing for some time in audio-only mode was buggy and
caused initial desync equal to the duration of prior audio-only
playback.
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Uninitialize video and audio outputs when switching to a file without
a corresponding track (audio-only file / file with no sound), or when
entering --idle mode. Switching track choice to "off" during playback
already did this.
It could be useful to have a mode where the video window stays open
even when no video plays, but implementing that properly would require
more than just leaving the window on screen like the code did before
this commit.
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Conflicts:
command.c
libao2/ao_alsa.c
libao2/ao_dsound.c
libao2/ao_pulse.c
libao2/audio_out.h
mixer.c
mixer.h
mplayer.c
Replace my mixer changes with uau's implementation, which is based on
my code.
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The player tried to disable mute before exiting, so that if mute is
emulated by setting volume to 0 and the volume setting is a
system-global one, we don't leave it at 0. However, the logic doing
this at process exit was flawed, as volume settings are handled by
audio output instances and the audio output that set the mute state
may have been closed earlier. Trying to write reliably working logic
that restores volume at exit only would be tricky, so change the code
to always unmute an audio driver before closing it and restore mute
status if one is opened again later.
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MPlayer volume control was originally implemented with the assumption
that it controls a system-wide volume setting which keeps its value
even if a process closes and reopens the audio device. However, this
is not actually true for --softvol mode or some audio output APIs that
only consider volume as a per-client setting for software mixing. This
could have annoying results, as the volume would be reset to a default
value if the AO was closed and reopened, for example whem moving to a
new file or crossing ordered chapter boundaries. Add code to set the
previous volume again after audio reinitialization if the current
audio chain is known to behave this way (softvol active or the AO
driver is known to not keep persistent volume externally).
This also avoids an inconsistency with the mute flag. The frontend
assumed the mute status is persistent across file changes, but it
could be similarly lost.
The audio drivers that are assumed to not keep persistent volume are:
coreaudio, dsound, esd, nas, openal, sdl. None of these changes have
been tested. I'm guessing that ESD and NAS do per-connection
non-persistent volume settings.
Partially based on code by wm4.
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Current volume was always queried from the the audio output driver (or
filter in case of --softvol). The only case where it was stored on
mixer level was that when turning off mute, volume was set to the
value it had before mute was activated. Change the mixer code to
always store the current target volume internally. It still checks for
significant changes from external sources and resets the internal
value in that case.
The main functionality changes are:
Volume will now be kept separately from mute status. Increasing or
decreasing volume will now change it relative to the original value
before mute, even if mute is implemented by setting AO level volume to
0. Volume changes no longer automatically disable mute. The exception
is relative changes up (like the volume increase key in default
keybindings); that's the only case which still disables mute.
Keeping the value internally avoids problems with granularity of
possible volume values supported by AO. Increase/decrease keys could
work unsymmetrically, or when specifying a smaller than default
--volstep, even fail completely. In one case occurring in practice, if
the AO only supports changing volume in steps of about 2 and rounds
down the requested volume, then volume down key would decrease by 4
but volume up would increase by 2 (previous volume plus or minus the
default change of 3, rounded down to a multiple of 2). Now, the
internal value will keep full precision.
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Conflicts:
libvo/vo_kva.c
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Stop trying to read terminal input if a read attempt returns EOF. The
most important case where this matters is when someone runs the player
with stdin redirected from /dev/null and without specifying
--no-consolecontrols. This used to cause 100% CPU load while paused,
as select() would continuously trigger on stdin (the need for
--no-consolecontrols was not apparent to people with older mplayer
versions, as input reading was less efficient and latencies like
hardcoded sleeps kept CPU use well below 100%). Now this will only
cause a "Dead key input" error message.
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Make the code read current real time again after drawing OSD. This
ensures time taken in OSD drawing is properly deducted from the
duration of the following sleep. The main practical effect is to avoid
the A-V field on the status line staying at a value a couple of
milliseconds above 0 (depending on VO).
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Fix a missing check that could sometimes result in video frames being
shown after specified end pts (end of timeline segment or --endpos).
Fix mistaken video EOF detection after aspect change in video stream,
when there is no current valid visible frame but the next frame is
already buffered in VO.
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Conflicts:
bstr.c
bstr.h
etc/input.conf
input/input.c
input/input.h
libao2/ao_pulse.c
libmpcodecs/vf_ass.c
libmpcodecs/vf_vo.c
libvo/gl_common.c
libvo/x11_common.c
mixer.c
mixer.h
mplayer.c
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For ao_pulse, the current latency is not a good indicator of how soon
the AO requires new data to avoid underflow. Add an internal pipe that
can be used to wake up the input loop from select(), and make the
pulseaudio main loop (which runs in a separate thread) use this
mechanism when pulse requests more data. The wakeup signal currently
contains no information about the reason for the wakup, but audio
buffers are always filled when the event loop wakes up.
Also, request a latency of 1 second from the Pulseaudio server. The
default is normally significantly higher. We don't need low latency,
while higher latency helps prevent underflows reduces need for
wakeups.
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Timeline handling converted the pts values from demuxed subtitles to
timeline scale. Change the code to do most subtitle handling in
original subtitle source pts, and instead convert current playback
timeline pts to those units when deciding which subtitle to show.
The main functionality changes are that now demuxed subtitles which
overlap chapter boundaries are handled correctly (at least for libass
subtitles), and external subtitles are assumed to use same pts scale
as current source (this needs improvements later).
Before, a video subtitle that had a duration continuing past the end
of the chapter would continue to be shown for the original duration,
even if the chapter ended and playback switched to a position in the
source where the subtitle shouldn't exist. Now, the subtitle will
correctly end.
Before, external subtitle files were interpreted as specifying pts
values in timeline scale. Now, they're interpreted as specifying pts
values in source file time scale, for _every_ source file. This is
probably more likely to be what the user wants for the "main" source
file in case there is one, but almost certainly not quite right for
multiple source files where the same subs could be shown over
different scenes. If the user wants them to match some main source
file, it's probably still better to have incorrect extra subs for
video from some files than to have every subtitle appearing at the
wrong time. The new code makes it easier to change the interpretation
of the subtitle times, and some configurability should be added in
the future.
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When switching to a timeline part from another file, decoders were
reinitialized after doing the demuxer-level seek. This is necessary
for audio because some decoders read from the demuxer stream during
initialization and the previous stream position before seek could have
been at EOF. However, this initialization sequence could lose first
subtitles or first part of audio.
The problem for subtitles was that the seek itself or audio
initialization could already have buffered subtitle packets from the
new position, and the way subtitles are reinitialized flushes packet
buffers. Thus early subtitles could be lost (even if they were demuxed
- unfortunately demuxers may not know about still active subtitles
earlier in the file, but that's another issue). Fix this by moving
subtitle and |
