| Commit message (Collapse) | Author | Age | Files | Lines |
|---|
| |\ |
|
| | |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| | |
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).
|
| | |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| | |
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.
|
| | |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| | |
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.
|
| |\|
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| | |
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.
|
| | |
| |
| |
| |
| |
| |
| |
| |
| | |
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.
|
| | |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| | |
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.
|
| | |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| | |
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.
|
| |\|
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| | |
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.
|
| | |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| | |
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.
|
| | |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| | |
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.
|
| | |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| | |
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.
|
| |\|
| |
| |
| |
| | |
Conflicts:
libvo/vo_kva.c
|
| | | |
|
| | |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| | |
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.
|
| | |
| |
| |
| |
| |
| |
| |
| | |
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).
|
| | |
| |
| |
| |
| |
| |
| |
| |
| | |
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.
|
| |\|
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| | |
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
|
| | |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| | |
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.
|
| | |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| | |
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.
|
| | |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| | |
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 video reinitialization before the demuxer seek; they
don't have the problems which prevent that for audio.
Audio initialization can already decode and buffer some output.
However, the seek_reset() call done last would then throw away this
buffered output. Work around this by adding an extra flag to
seek_reset().
|
| | | |
|
| | |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| | |
Restructure parts of the code in the main play loop. The main
functionality difference is that if a video track ends first, now
audio will continue to be played until it ends too.
Now the process also wakes up less often if there's no need to update
video or audio. This will reduce unnecessary wakeups especially when
paused, but may make handling of input events laggier when fd-based
notifications are not supported (like most input on Windows).
|
| | |
| |
| |
| |
| |
| | |
Change the terminal status line to show "???" instead of a huge
negative number if audio or video pts is missing (there was a partial
workaround for audio before, but not video or A-V difference).
|
| |\|
| |
| |
| |
| |
| |
| |
| | |
Conflicts:
command.c
mp_core.h
mplayer.c
screenshot.c
|
| | |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| | |
Remove the old EDL implementation that was activated with the --edl
option. It is mostly redundant and inferior compared to the newer
demux_edl support, though currently there's no support for using the
same EDL files with the new implementation and the mute functionality
of the old implementation is not supported. The main reason to remove
the old implementation at this point is that the mute functionality
would conflict with following audio volume handling changes, and
working on the old code would be a wasted effort in the long run as at
some point it would be removed anyway.
The --edlout functionality is kept for now, even though after this
commit there is no code that could directly read its output.
|
| | |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| | |
Windows uses a legacy codepage for char* / runtime functions accepting
char *. Using UTF-8 as the codepage with setlocale() is explicitly
forbidden.
Work this around by overriding the MSVCRT functions with wrapper
macros, that assume UTF-8 and use "proper" API calls like _wopen etc.
to deal with unicode filenames. All code that uses standard functions
that take or return filenames must now include osdep/io.h. stat()
can't be overridden, because MinGW-w64 itself defines "stat" as a
macro. Change code to use use mp_stat() instead.
This is not perfectly clean, but still somewhat sane, and much better
than littering the rest of the mplayer code with MinGW specific hacks.
It's also a bit fragile, but that's actually little different from the
previous situation. Also, MinGW is unlikely to ever include a nice way
of dealing with this.
|
| | |
| |
| |
| |
| |
| |
| | |
The _UWIN define causes the mingw headers not to declare deprecated (on
Windows) function names such as open and mkdir. But the code uses these. I
have no idea why this used to work (if it even did), but the original
reason why it was defined seems to have vanished.
|
| | |
| |
| |
| |
| |
| |
| |
| | |
The name "MPlayer2" isn't used anywhere. It's either "MPlayer" or
"mplayer2". Make it more consistent by using "mplayer2" instead.
Note that the version string passed as network user-agent changes from
"MPlayer" to "mplayer2" as well.
|
| | |
| |
| |
| |
| |
| | |
The current code tried to print -1000 as unsigned integer if the
chapter time was unknown. Print -1 instead. This affects only the
-identify output used for slave mode, such as ID_CHAPTER_0_START.
|
| |\|
| |
| |
| |
| |
| | |
Conflicts:
mplayer.c
screenshot.c
|
| | |
| |
| |
| |
| |
| |
| |
| | |
The terminal OSD line was written with mp_msg(MSGT_CPLAYER, ...) but
erased with printf(). This meant that disabling MSGT_CPLAYER messages
would prevent the terminal line from being printed, but a line
(probably unrelated) would still be cleared. Change the clearing code
to use mp_msg(MSGT_CPLAYER, ...) too.
|
| | |
| |
| |
| |
| |
| |
| |
| |
| | |
The vd_ffmpeg decode() function returned without doing anything if the
input packet had size 0. This meant that flushing buffered frames at
EOF did not work. Remove this test. Have the core code skip such
packets coming from the file being played instead (Libav treats
0-sized packets as flush signals anyway, so better assume such packets
do not represent real frames with any codec).
|
| | |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| | |
Remove the private bswap and intreadwrite.h implementations and use
libavutil headers instead.
Originally these headers weren't publicly installed by libavutil at
all. That already changed in 2010, but the pure C bswap version in
installed headers was very inefficient. That was recently (2011-12)
improved and now using the public bswap version probably shouldn't
cause noticeable performance problems, at least if using a new enough
compiler.
|
| | |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
