This is a small oversight. The client name (as set on command line
options or, more importantly, the client API) was not set when listing
devices e.g. via the "audio-device-list" property.
Might or might not fix#1578.
Also adjust the log level for an unrelated message.
This reverts commit acc5e8f574.
As expected, some didn't like this. Others won't like this revert.
Whatever.
See #1561.
This should go into mpv 0.8.0 before it's released.
In my opinion the artifacts created by af_scaletempo on extreme slowdown
(50% or so) are too bothersome - but users disagree. So use
af_scaletempo on any speed changes, not just on speedup.
Filters which merely wrap libavfilter (for user-compatibility) like
vf_gradfun had a "lavfi-enable" suboption, which could disable
libavfilter usage. Since none of these filters has an internal
implementation anymore, this was completely useless.
rubberband_available() can return a negative value, which we assigned to
a size_t variable, leading to the frame allocation to fail. This could
spam "Error filtering frame.". (That it spams this instead of exiting
should probably also be considered a bug.)
At least in the realtime mode and in our case, a negative return value
should not have any different meaning from a 0 return value, in
particular because we call rubberband_get_samples_required() or set the
"final" parameter for rubberband_process() to continue/stop processing.
After some testing, I am fairly convinced that these defaults sound
better than the previous settings. This also eliminates some issue
with random crackling and noise.
Also remove the `stretch` option since it has no effect in
realtime mode.
The previous commit on this filter accidentally removed the
RubberBandOptionProcessRealTime option. Without it, the lib prints a
warning and passes the audio through.
Also add the RubberBandOptionSmoothingOn option back. Though for some
reason the output sounds still very wrong.
Remove the confusing crap that allowed a filter using the libavfilter
bridge to be compiled without libavfilter. Instead, compile the wrappers
only if libavfilter is enabled at compile time.
The only filter which still requires it is vf_stereo3d (unfortunately).
Special-case this one. (The whole filter and how it interacts with lavfi
is pure braindeath anyway.)
librubberband exports a big load of options. Normally, the default
settings (whether they're librubberband defaults or our defaults) should
be sufficient, but since I'm not so sure about this, making it
configurable allows others to figure it out for me.
The problem here is that librubberband can buffer an arbitrary amount
of data, but at the same time doesn't provide a way to query how much
data is buffered. So we keep track of this manually, assuming that
librubberband tries to reach the requested time ratio for input and
output (which is probably true).
The disadvantage is that rounding errors could accumulate over time.
(Maybe it should try to round towards keeping the time ratio.)
In theory it could happen that draining on EOF happens incrementally,
and then the unconditional reset could have dropped the remaining
buffered audio.
If "--af=rubberband" is used, librubberband will be used to speed up or
slow down audio with pitch correction.
This still has some problems: the audio delay is not calculated
correctly, so the audio position jitters around by a few milliseconds.
This will probably ruin video timing.
This avoids potentially dropping some small amount of audio data
buffered in filters.
Reinit can be skipped only if the filter is af_scaletempo (which maps to
AF_CONTROL_SET_PLAYBACK_SPEED). The other case using af_lavrresample is
much more complicated due to filter chain politics.
Also, changing speed between 1.0 and something higher typically inserts
or removes the filter, so this obviously requires reinitialization. It
can be prevented by forcing the filter with --af=scaletempo.
Staring at the code a bit, it turns out that changing speed without
losing state is quite easy. The initialization code is big and
complicated, but most of it is specific only to the configured audio
format, not the speed.
Refactor the code so that changing speed at runtime could work. (It's
not actually used yet - the player code still does a complete reinit.
This will be fixed in the next commit.)
The "if (s->speed_tempo == s->speed_pitch)" looks a bit strange, but
does the same thing as the code did before: speed can be changed only if
exactly one flag is set. If both are set or none, speed can't be
changed.
This code skipped initialization if no speed/pitch change was to be
applied.
It also didn't force conversion of the audio to a supported format,
which is probably the most important case in context of compatibility.
With this change applied, af_scaletempo will always force format
conversion.
To make the change less disruptive, make the filter detach if
unconvertable formats are used. Some users use spdif and also have
"af=scaletempo" in their config, so better not completely break this.
In the case the filter was added with the "speed=both" suboption, the
filter also detached itself in this case; but it's an obscure case, so I
don't care about that.
This reverts commit a33b46194c.
It turns out FFmpeg really considers this a bug, and fixed it by making
the decoder output the correct pixel format.
Fixes#1565. Reverts the fix#1528, though it should work fine with
a recent git master FFmpeg.
The intention is that we can test vo_opengl with high bit depth PNGs
better. This throws libswscale completely out of the loop, which before
was needed in order to convert from big endian to little endian.
Also apply a minimal cleanup to fmt-conversion.c (unrelated).
Most things stopped using this field for better support of growing
files. Go through the trouble to repalce the remaining uses, so it can
be removed.
Also move the "streaming" field; saves 4 bytes (wow!).
Fix return types and return values to make them more consistent. Some
reformatting and making code more concise.
In stream_reconnect(), avoid the additional mp_cancel_test() call by
moving the "connection lost" message below the mp_cancel_wait() call,
which effectively leads to the same behavior when the stream was already
canceled. (The goal is not to show the message in this case.)
Merge stream_seek_long() into stream_seek(). It was the only caller.
Always clear the eof flag on seeks.
Reduce access to stream internals in cache.c and stream_lavf.c.
Make it accept "," as separator, instead of only ":". Do this by using
the key-value-list parser. Before this, the option was stored as a
string, with the option parser verifying that the option value as
correct. Now it's stored pre-parsed, although the log levels still
require separate verification and parsing-on-use to some degree (which
is why the msg-level option type doesn't go away).
Because the internal type changes, the client API "native" type also
changes. This could be prevented with some more effort, but I don't
think it's worth it - if MPV_FORMAT_STRING is used, it still works the
same, just with a different separator on read accesses.
Before this, unquoted occurrences of ":" lead to parsing errors. There's
no reason to reject it, especially since the traditional MPlayer syntax
uses ":" as separator. (Which is also the reason why ":" was rejected
before: the parser shares this code for handling splitting/quoting, and
we merely checked explicitly whether the option was split on ",".)
Fixes#1445 without failing in the case where file outside.mkv with N
chapters references inside.mkv with N+1 chapters. Since we don't care
about the number of chapters in inner files (yet; nested chapters would
change this, but also make chapters dynamic anyways), don't check there.
Much cleaner. The difference between timeline_info and
inner_timeline_info is that timeline_info is stuff the top-level cares
about and inner_timeline_info is just for the current chapter.
This introduces a new option linear-scaling, which is now implied by
srgb, icc-profile and sigmoid-upscaling.
Notably, this means (sigmoidized) linear upscaling is now enabled by
default in opengl-hq mode. The impact should be negligible, and there
has been no observation of negative side effects of sigmoidized scaling,
so it feels safe to do so.
Autoload external audio files only if there's at least a video track
(which is not coverart pseudo-video).
Enable external audio file autoloading by default. Now that we actively
avoid doing stupid things like loading an external audio file for an
audio-only file, this should be fine.
Additionally, don't autoload subtitles if a subtitle is played.
Although you currently can't play subtitles without audio or video,
it's disturbing and stupid that the player might load subtitle files
with different extension and then fail.
Giving this such a prominent place is not really appropriate anymore.
Most people seeing this would probably expect a release changelog, not
something about MPlayer.
Since the page still could be useful for former MPlayer users (in
particular to avoid confusion with renamed options etc.), still keep
it in the DOCS directory.
Reindent the whole handle_realaudio() function, and make the surrouding
if block return early instead.
Also contains some cosmetics to the sipr swapping, which hopefully does
not change the semantics, but is untested (the kind of cosmetic changes
everyone loves so much). May the person responsible for sipr rot in
hell. (It was probably done to obfuscate the codec?)
Staring at the code, it doesn't look like the extra code for "normal"
audio is needed. Most of it looks like artifacts from the previous code
structure (much of it was added in the initial commit). I couldn't find
a sample that uses this code path to fully confirm this, though.
I suppose it could lead to subtle changes in behavior in presence of
realvideo files that change aspect radio. With the only sample I had
available, the behavior actually improved (azumi.mkv from the MPlayer
samples FTP; when starting playback in the middle it used the wrong
aspect ratio).
Appears to work, so we can drop some code. For some really odd reason,
the descrambling done on the timestamp requires millisecond units (due
to the "algorithm", not the libavcodec API).
Fixes vp9 missing timestamps. This requires a brand new libavcodec (the
patch for this was just applied to FFmpeg git master).
The timestamp mangling is applied to VP9 only. It'd probably work with
other codecs, but it's not needed. It could break in various ways, so
it has to be explicitly checked for every enabled codec.
