| Commit message (Collapse) | Author | Age | Files | Lines |
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stream_cue, which provided the cue:// protocol handler, was extremely
hacky and didn't even manage to play some samples I tried.
Remove it, because it's plain unneeded. There is much better support
for .cue files elsewhere:
- libcdio can play pairs of .cue/.bin files:
mplayer cdda:// --cdrom-device=your_cue_file.cue
Note that if the .cue file is not accompanied by a .cue file, but
an encoded file for example, this most likely won't work.
- mplayer can play .cue files directly:
mplayer your_cue_file.cue
This works, even if the .cue file comes with encoded files that are
not .bin . Note that if you play .bin files, mplayer will assume a
specific raw audio format. If the format doesn't match, mplayer will
play noise and destroy your speakers. Note that format mismatches are
extremely common, because the endianness seems to be essentially
random. (libcdio uses a clever algorithm to detect the endian, and
doesn't have this problem.)
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To quote the manpage: "This filter is untested, maybe even unusable."
And it seems they were never touched again after it was added many
years ago (except for cosmetic changes). Just get rid of them.
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This reverts commit c8b0f2115447f2fdd5bb8090d06153a4bf72d9ff.
This was a very bad idea. It caused A/V desync with some crappy AVI
files, and upon inspecting ad_mad.c, it seems all hope is lost.
Go back to the prefering the mpg123 & libav codecs.
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Conflicts:
DOCS/man/en/vo.rst
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Also make some minor cosmetic changes.
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About a year ago, ubitux converted most of the old manpage from the
hard to maintain nroff format to reStructuredText. This was not merged
back into the master repository immediately. The argument was that the
new manpage still required work to be done. However, progress was very
slow. Even worse: the old manpage wasn't updated, because it was
scheduled for deletion, and updating it would have meant useless work.
Now the situation is that the new manpage still isn't finished, and the
old manpage is grossly out of sync with the player. This is not helpful
for users. Additionally, keeping the new manpage in a separate branch,
while the normal development repository for code had the old manpage,
was very inconvenient, because you couldn't just update the
documentation in the same commit as the code.
Even though the new manpage isn't finished yet, merging it now seems to
be the best course of action. Squash-merge the manpage development
branch [1], revision e89f5dd3f2, which branches from the mplayer2
master branch after revision 159102e0cb.
Committers:
* Clément Bœsch <ubitux@gmail.com> (Initial conversion to RST.)
* Uoti Urpala <uau@mplayer2.org> (Many updates.)
* Myself (Minor edits.)
Most text of the manpage has been directly taken from the old manpage,
because this is a conversion, not a complete rewrite.
[1] http://git.mplayer2.org/uau/mplayer2.git/log/?h=man
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These were unused.
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The --title option, which sets the GUI window caption, is now expanded
as slave mode property string (like osd_show_property_text). Make the
default value for --title include the filename. This makes a behavior
similar to --use-filename-title the default.
Remove the --use-filename-title option, as it's redundant now.
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Commit dc2a4863af9b0e introduced a new way of specifying default values
for strings (you're supposed to use OPTDEF_STR() instead of putting it
into the option struct, such as it was done in defaultopts.c). The code
to handle the old way was explicitly disabled, which caused random
crashes when used.
Allow the old way again. With the main option struct in particular, I
see no reason why some option defaults should be specified in
defaultopts.c, and some directly along the options.
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This seems rather pointless considering there are still stupid
bstrdup0()s left, but maybe this is the right direction.
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There is no reason why generated source files shouldn't be part of the
clean target, as opposed to distclean. On the contrary, having them
in distclean only looks dangerous when trying to deal with broken
dependency rules. Move them to clean, except config.h (which would
require configure to be run again).
Also, some recently added generated files were missing from the clean
targets.
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This was a horrible little tool to detect the host CPU at build time.
Forgotten in commit 74df1d8e05aa226c7e8.
Also remove forgotten codec-cfg entry in .gitignore .
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The internal array of default key bindings is removed. Include the
file etc/input.conf at compile time (using the file2header tool), and
parse the default binds from etc/input.conf at startup time.
This lowers maintainance overhead, and makes sure the default bindings
and etc/input.conf don't deviate. Commit f30bf73bf22ed0542 already
made sure etc/input.conf matches the default bindings, so this commit
shouldn't change anything user-visible.
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Builtin (i.e. default) binds are still separately handled, and this
commit shouldn't change any user-visible behavior.
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Now input.conf is loaded into memory at once, instead of streaming the
file into the parser.
The real reason for this change is that I want to be able to read the
config file from memory. (Using fmemopen() would have been simpler, but
that is available on sane platforms only.)
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vo_osd_changed() was a weird function: it was used both to query and
mutate state, which is a bad combination. The VOs used it to query
and reset the state, and the mplayer frontend mostly used it to set
the state. In some cases, the frontend did both (that code used a
variable "int hack" to backup the state and set it again).
Simplify it and make the VOs use a vo_osd_has_changed() function to
query whether the OSD bitmaps have to be recreated. vo_osd_changed()
on the other hand is now used to update state only. The OSD change
state is reset when osd_draw_text() is called.
Update vo_corevideo.m to use vo_osd_resized() as well (forgotten change
from libass-OSD merge).
Simplify osd_set_text() and its usages.
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There are still various other RTSP implementations available, such as
libnemesi, live555, and libav. The mplayer native version was a huge
chunk of old unmaintained code.
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This was done with the help of callcatcher [1]. Only functions which
are statically known to be unused are removed.
Some unused functions are not removed yet, because they might be needed
in the near future (such as open_output_stream for the encode branch).
There is one user visible change: the --subcc option did nothing, and is
removed with this commit.
[1] http://www.skynet.ie/~caolan/Packages/callcatcher.html
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__DARWIN is still not defined, use __APPLE__ instead
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This was probably forgotten in the commit that removed the dependency on
freetype.
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The code will be simpler. Also slower, because strlen is removed, but
it's very unlikely this matters at all.
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The code to format the playback time was duplicated a few times. There
were also minor differences in how the time is formatted. Remove most
of these differences. This also fixes a bug in the output of the
osd_show_progression command, introduced in 74e7a1e937c10d9f4.
There was some logic to display the percent position in the OSD status
for a short while after seeking. Remove that logic and always display
the percent position.
Make --osd-fractions a flag option. This removes the ability to show
the number of frames played since the start of the current second
(i.e. the fraction of the time was turned into a frame number). This
features wasn't so great anyway, because modern video file formats
don't always have a (valid) FPS set, and could lead to inaccurate
display.
Still to sort out:
Unfortunately, the terminal status is still formatted differently from
the OSD, and even worse, it has a completely different time source.
Not sure if I like how the status line looks now (it's a bit "full"?).
Maybe it will be changed again later.
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This seems to be more portable. Should fix compilation on OSX and
FreeBSD. Apparently also works on MinGW-w64.
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OSX mangles symbols with "_".
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Summary:
- There is no playtree anymore. It's reduced to a simple list.
- Options are now always global. You can still have per-file options,
but these are optional and require special syntax.
- The slave command pt_step has been removed, and playlist_next
and playlist_prev added. (See etc/input.conf changes.)
This is a user visible incompatible change, and will break slave-mode
applications.
- The pt_clear slave command is renamed to playlist_clear.
- Playtree entries could have multiple files. This is not the case
anymore, and playlist entries have always exactly one entry. Whenever
something adds more than one file (like ASX playlists or dvd:// or
dvdnav:// on the command line), all files are added as separate
playlist entries.
Note that some of the changes are quite deep and violent. Expect
regressions.
The playlist parsing code in particular is of low quality. I didn't try
to improve it, and merely spent to least effort necessary to keep it
somehow working. (Especially ASX playlist handling.)
The playtree code was complicated and bloated. It was also barely used.
Most users don't even know that mplayer manages the playlist as tree,
or how to use it. The most obscure features was probably specifying a
tree on command line (with '{' and '}' to create/close tree nodes). It
filled the player code with complexity and confused users with weird
slave commands like pt_up.
Replace the playtree with a simple flat playlist. Playlist parsers that
actually return trees are changed to append all files to the playlist
pre-order.
It used to be the responsibility of the playtree code to change per-file
config options. Now this is done by the player core, and the playlist
code is free of such details.
Options are not per-file by default anymore. This was a very obscure and
complicated feature that confused even experienced users. Consider the
following command line:
mplayer file1.mkv file2.mkv --no-audio file3.mkv
This will disable the audio for file2.mkv only, because options are
per-file by default. To make the option affect all files, you're
supposed to put it before the first file.
This is bad, because normally you don't need per-file options. They are
very rarely needed, and the only reasonable use cases I can imagine are
use of the encode backend (mplayer encode branch), or for debugging. The
normal use case is made harder, and the feature is perceived as bug.
Even worse, correct usage is hard to explain for users.
Make all options global by default. The position of an option isn't
significant anymore (except for options that compensate each other,
consider --shuffle --no-shuffle).
One other important change is that no options are reset anymore if a
new file is started. If you change settings with slave mode commands,
they will not be changed by playing a new file. (Exceptions include
settings that are too file specific, like audio/subtitle stream
selection.)
There is still some need for per-file options. Debugging and encoding
are use cases that profit from per-file options. Per-file profiles (as
well as per-protocol and per-VO/AO options) need the implementation
related mechanisms to backup and restore options when the playback file
changes.
Simplify the save-slot stuff, which is possible because there is no
hierarchical play tree anymore. Now there's a simple backup field.
Add a way to specify per-file options on command line. Example:
mplayer f1.mkv -o0 --{ -o1 f2.mkv -o2 f3.mkv --} f4.mkv -o3
will have the following options per file set:
f1.mkv, f4.mkv: -o0 -o3
f2.mkv, f3.mkv: -o0 -o3 -o1 -o2
The options --{ and --} start and end per-file options. All files inside
the { } will be affected by the options equally (similar to how global
options and multiple files are handled). When playback of a file starts,
the per-file options are set according to the command line. When
playback ends, the per-file options are restored to the values when
playback started.
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This was intended for translating filenames from filesystem charset to
the terminal charset. Modern sane platforms use UTF-8 for everything,
and on Windows we use unicode APIs, so this is not needed anymore.
Remove filename_recode, all uses of it, options and configure checks
related to terminal output charset, and code that tries to determine
the same.
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This had very limited usefulness, and you're much better off using
ffmpeg directly. Even if that should not be sufficient, the mplayer
encoding branch might provide a better way out.
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Including <malloc.h>, especially if all you want is malloc(), has no
legitimate uses (on sane platforms at least). Remove the check for it,
and remove all uses in the code.
Remove unused check for alloca().
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Also, replace the only use of memalign: use av_malloc instead in sub.c.
(av_malloc allocates with the required alignment restrictions.)
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Pausing the player used to print the message "===== PAUSE =====". It
also inserted a newline for some reason. When pausing and unpausing a
lot, the terminal would be clobbered with "old" useless status lines.
Remove the pause message, and display the status message instead. This
looks better, doesn't fill up the terminal with crap, and needs less
code.
Side note: when cache is enabled, the status line is reprinted on every
idle iteration to reflect possible cache changes. If the platform's
WAKEUP_PERIOD is very small (like on Windows) and terminal output is
slow (like on Windows), it's possible that this leads to a minor
performance degradation. This is probably not a problem (and I don't
care anyway), but maybe something that should be kept in mind.
Disabling the status line with --quiet will help.
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Most of these demuxers and decoders are provided in better form by
libav, while the mplayer builtin ones are essentially unmaintained. The
only legimitate use case for not using the libav ones was working around
libav bugs or bugs related to the way mplayer uses libav. Instead of
trying to keep dead code alive, development effort should go into
improving libav or the mplayer libav glue code.
Note that the libav demuxer have been preferred over the mplayer builtin
ones for a while in mplayer2. There were some exceptions: playing DVDs
with dvdnav or playing network sources. (That's because some stream
modules and network.c requested explicit file formats, such as
DEMUXER_TYPE_MPEG_PS, which mapped to builtin demuxers.) With this
commit, they are switched to use libav. One caveat is that the requested
format is not passed to libavformat, instead we rely on the auto probing
to select the correct libav demuxer (see code in demux_open_stream()).
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For some reason, these 3 VOs basically call exit() if something
went wrong.
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Work around PulseAudio bugs more effectively. In particular, this
should avoid two issues: playback never finishing at end of file /
segment due to PulseAudio always claiming there's still time before
audio playback reaches the end, and jerky playback especially after
seeking due to bogus output from PulseAudio's timing interpolation
code.
This time, I looked into the PulseAudio code itself and analyzed the
bugs causing problems. Fortunately, two of the serious ones can be
worked around in client code. Write a new get_delay() implementation
doing that, and remove some of the previous workarounds which are now
unnecessary. Also add a pa_stream_trigger() call to ensure playback of
files shorter than prebuf value starts (btw doing that by setting a
low prebuf hits yet another PulseAudio bug, even if you then write the
whole file in one call).
There are still a couple of known PulseAudio bugs that can not be
worked around in client code. Especially, bug 4 below can cause issues
when pausing.
Below is a copy of a message I sent to the pulseaudio-discuss mailing
list, describing some of the PulseAudio bugs:
==================================================
A lot of mplayer2 users with PulseAudio have experienced problems. I
investigated some of those and confirmed that they are caused by
PulseAudio. There are quite a few distinct PulseAudio bugs; some are
analyzed below. Overall, however, I wonder why there are so many fairly
obvious bugs in a widely used piece of software. Is there no
maintenance? Or do people not test it? Some of the bugs are probably
less obvious if you request low latency (though they're not specific to
higher-latency case); do people test the low-latency case only?
1. The timing interpolation functionality can return completely bogus
values for playback position and latency, especially after seeking
(mplayer2 does cork / flush / uncork, as flushing alone does not seem to
remove data already in sink). I've seen quickly repeated seeks report
over 10 second latency, when there aren't any buffers anywhere that big.
I have not investigated the exact cause. Instead I disabled
interpolation and added code to always call
pa_stream_update_timing_info(). (I assume that always waiting for this
to complete, instead of doing custom interpolation, may give bad
performance if it queries a remote server. But at least it works better
locally.)
2. Position/latency reporting is wrong at the end of a stream (after the
lack of more data triggers underflow status). As a result mplayer2 never
ends the playback of a file, as it's waiting forever for audio to finish
playing. The reason for this is that the calculations in PulseAudio add
the whole length of data in the sink to the current latency (subtract
from position), even if the sink does not contain that much data *from
this stream* in underflow conditions. I was able to work around this bug
by calculating latency from pa_timing_info data myself as follows
(ti=pa_timing_info):
int64_t latency = pa_bytes_to_usec(ti->write_index - ti->read_index, ss);
latency -= ti->transport_usec;
int64_t sink_latency = ti->sink_usec;
if (!ti->playing)
// this part is missing from PulseAudio itself
sink_latency -= pa_bytes_to_usec(ti->since_underrun, ss);
if (sink_latency > 0)
latency += sink_latency;
if (latency < 0)
latency = 0;
However, this still doesn't always work due to the next bug.
3. The since_underrun field in pa_timing_info is wrong if PulseAudio is
resampling the stream. As a result, the above code indicated that the
playback of a 0.1 second 8-bit mono file would take about 0.5 seconds.
This bug is in pa_sink_input_peek(). The problematic parts are:
ilength = pa_resampler_request(i->thread_info.resampler, slength);
...
if (ilength > block_size_max_sink_input)
ilength = block_size_max_sink_input;
...
pa_memblockq_seek(i->thread_info.render_memblockq, (int64_t) slength, PA_SEEK_RELATIVE, TRUE);
...
i->thread_info.underrun_for += ilength;
This is measuring audio in two different units, bytes for
resampled-to-sink (slength) and original stream (ilength). However, the
block_size_max_sink_input test only adjusts ilength; after that the
values may be out of sync. Thus underrun_for is incremented by less than
it should be to match the slength value used in pa_memblockq_seek.
4. Stream rewind functionality breaks if the sink is suspended (while
the stream is corked). Thus, if you pause for more than 5 seconds
without other audio playing, things are broken after that. The most
obvious symptom is that playback can continue for a significant time
after corking. This is caused by sink_input and sink getting out of
sync. First, after uncorking a stream on a suspended sink,
pa_sink_input_request_rewind() is called while the sink is still in
suspended state. This sets sink_input->thread_info.rewrite_nbytes to -1
and calls pa_sink_request_rewind(). However, the sink ignores rewind
requests while suspended. Thus this particular rewind does nothing. The
problem is that rewrite_nbytes is left at -1. Further calls to
pa_sink_input_request_rewind() do nothing because "nbytes =
PA_MAX(i->thread_info.rewrite_nbytes, nbytes);" sets nbytes to -1, and
the call to pa_sink_request_rewind() is under "if (nbytes != (size_t)
-1) {". Usually, after a sink responds to a rewind request,
rewrite_bytes is reset in pa_sink_input_process_rewind(), but this
doesn't happen if the sink ever ignores one request. This broken state
can be resolved if pa_sink_input_process_rewind() is called due to a
rewind triggered by _another_ stream.
There were more bugs, but I'll leave those for later.
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XMMS has been dead since 2007.
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