| Commit message (Collapse) | Author | Age | Files | Lines |
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Maybe this should be default. On the other hand, this filter does
something even if the volume is neutral: it clips samples against the
allowed range, should the decoder or a previous filter output garbage.
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You had to use "no-replaygain-noclip" to set this option. Rename it, so
that only one negation is needed.
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This adds the options replaygain-track and replaygain-album. If either is set,
the replaygain track or album gain will be automatically read from the track
metadata and the volume adjusted accordingly.
This only supports reading REPLAYGAIN_(TRACK|ALBUM)_GAIN tags. Other formats
like LAME's info header would probably require support from libav.
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This doesn't need quoting, for some reason.
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Since mp_find_user_config_file() is going to get a context argument,
which would be annoying to do in the audio chain (actually I'm just
lazy).
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Similar to af_channels etc...
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Similar situation to af_channels.
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Apparently this stopped working after some planar changes (broken format
negotiation). Radically change option parsing in an incompatible way.
Suggest alternatives to this filter, since it barely has any importance
anymore.
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Don't bother explaining the sample format naming schema. The "ne" bit is
outdated anyway, and anyone who has to use this option will be able to
understand the naming schema just by looking at the names too.
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There are some use cases for this. For example, you can use it to set
defaults of automatically inserted filters (like af_lavrresample). It's
also useful if you have a non-trivial VO configuration, and want to use
--vo to quickly change between the drivers without repeating the whole
configuration in the --vo argument.
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You get the native format by not appending any suffix to the format.
This change includes user-facing names, e.g. for the --format option.
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This changes option parsing as well as filter defaults slightly. The
default is now to encode to spdif (this is way more useful than writing
raw AC3 - what was this even useful for, other than writing broken ac3
-in-wav files?). The bitrate parameter is now always in kbps.
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Roughly follows MPlayer svn commits 36492 and 36493. We also remove
the volume peak reporting. (There are much better libavfilter filters
for this, I think.)
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af_format is the old audio conversion filter. It could do all possible
conversions supported by the audio chain. However, ever since the
addition of af_lavrresample, most conversions are done by
libav/swresample, and af_format is used as fallback.
Separate out the fallback cases and remove af_format. af_convert24 does
24 bit <-> 32 bit conversions, while af_convertsignendian does sign and
endian conversions. Maybe the way the conversions are split sounds a bit
odd. But the former changes the size of the audio data, while the latter
is fully in-place, so there's at least different buffer management.
This requires a quite complicated algorithm to make sure all these
"partial" conversion filters can actually get from one format to
another. E.g. s24le->s32be always requires convertsignendian and
convert24, but af.c has no idea what the intermediate format should
be. So I added a graph search (trying every possible format and
filter) to determine required format and filter. When I wrote this,
it seemed this was still better than messing everything into
af_lavrresample, but maybe this is overkill and I'll change my
opinion. For now, it seems nice to get rid of af_format though.
The AC3->IEC61937 conversion isn't supported anymore, but I don't think
this is needed anywhere. Most AOs test all formats explicitly, or use
the AF_FORMAT_IS_IEC61937() macro (which includes AC3).
One positive consequence of this change is that conversions always
include dithering (done by libav/swresample), instead of possibly going
through af_format, which doesn't do anything fancy.
Rename af_force to af_format. It's essentially compatible with command
line uses of af_format. We retain a compatibility alias for af_force.
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This makes it actually possible to use the filter with more complicated
filter graphs (such as graphs containing the "," character).
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Also add a "o" suboption, which should allow fine control over
libavresample.
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Mostly copied from vf_lavfi. The parts that could be shared are minor,
because most code is about setting up audio and video, which are too
different.
This won't work with Libav. I used ffplay.c as guide, and noticed too
late that their setup methods are incompatible with Libav's. Trying to
make it work with both would be too much effort. The configure test for
av_opt_set_int_list() should disable af_lavfi gracefully when compiling
with Libav.
Due to option parser chaos, you currently can't have a "," as part of
the filter graph string - not even with quoting or escaping. This will
probably be fixed later.
The audio filter chain is not PTS aware. So we have to do some hacks
to make up a fake PTS, and we have to map the output PTS back to the
filter chain's method of tracking PTS changes and buffering, by
adjusting af->delay.
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Its main purpose is for testing in case channel layout stuff breaks, in
particular in connection with old audio filters.
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Reflects the changes over the last few commits.
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Normally, af_lavrresample detaches itself immediately if the input and
output audio parameters are the same. no-detach prevents this.
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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.
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The only thing this option did was changing the behavior of af_volume.
The option decided what sample format af_volume would use, but only if
the sample format was not already float. If the option was set, it would
default to float, otherwise to S16.
Remove use of the option and all associated code, and make af_volume
always use float (unless a af_volume specific sub-option is set).
Silence maximum value tracking. This message is printed when the filter
is destroyed, and it's slightly annoying. Was enabled due to enabling
float by default.
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Remove `af_resample` and `af_lavcresample`. The former is a mess while the
latter uses an API that was long deprecated in libavcodec and is now removed.
`af_lavrresample` rougly has the same features and structure of
`af_lavcresample`.
libswresample fallback by wm4.
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The previous name of this filter was misleading, because it doesn’t actually
normalize volume levels. What it does is closer to performing low-quality
dynamic range compression, hence it is now called af_drc.
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This is considered a worthless feature. Note that alaw/mulaw/adpcm input
is unaffected: such data is handed to libavcodec and "decoded" to linear
PCM.
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This changes the name of this project to mpv. Most user-visible mentions
of "MPlayer" and "mplayer" are changed to "mpv". The binary name and the
default config file location are changed as well.
The new default config file location is: ~/.mpv/
Remove etc/mplayer.desktop. Apparently this was for the MPlayer GUI,
which has been removed from mplayer2 ages ago.
We don't have a logo, and the MS Windows resource files sort-of require
one, so leave etc/mplayer.ico/.xpm as-is.
Remove the debian and rpm packaging scripts. These contained outdated
dependencies and likely were more harmful than useful. (Patches which
add working and well-tested packaging are welcome.)
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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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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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