Literally copy-pasted from the same commit for video filters. (Once new
code for filters is implemented, this will all go away or at least get
unified anyway.)
This was supposed to restrict output to formats supported by us. But we
usually support all FFmpeg sample formats anyway (if not, it will error
out gracefully, and we would add the missing format). Basically, it's
just useless bloat.
Basically, see the example in input.rst.
This is better than the "old" vf-toggle method, because it doesn't
require the user to duplicate the filter string in mpv.conf and
input.conf.
Some aspects of this changes are untested, so enjoy your alpha testing.
Remove low quality drc filter. Anyone whishing to have dynamic range
compression should use the much more powerful acompressor ffmpeg filter:
mpv --af=lavfi=[acompressor] INPUT
Or with parameters:
mpv --af=lavfi=[acompressor=threshold=-25dB:ratio=3:makeup=8dB] INPUT
Refer to https://ffmpeg.org/ffmpeg-filters.html#acompressor for a full
list of supported parameters.
Signed-off-by: wm4 <wm4@nowhere>
In a first pass, we check whether libavcodec is present.
Then we try to compile a snippet and check for FFmpeg vs. Libav. (This
could probably also be done by somehow checking the pkgconfig version.
But pkg-config can't deal with that idiotic FFmpeg idea that a micro
version number >= 100 identifies FFmpeg vs. Libav.)
After that we check the project-specific version numbers. This means it
can no longer happen that we accidentally allow older, unsupported
versions of FFmpeg, just because the Libav version numbers are somehow
this way.
Also drop the resampler checks. We hardcode which resampler to each with
each project. A user can no longer force use of libavresample with
FFmpeg.
The FFmpeg versions we support all have the APIs we were checking for.
Only Libav missed them. Simplify this by explicitly checking for FFmpeg
in the code, instead of trying to detect the presence of the API.
Looks quite like a bug. If you have a filter chain with only the
dynaudnorm filter, and send call av_buffersrc_add_frame(s, NULL), then
subsequent av_buffersink_get_frame() calls will return EAGAIN instead of
EOF.
This was apparently caused by a recent change in FFmpeg.
Some other circumstances (which I didn't fully analyze and which is due
to the playloop's absurd temporary-EOF behavior on seeks) then led the
decoder loop to send data again, but since libavfilter was stuck in the
EOF state now, it could never recover. It kept sending new input (due to
missing output), until the demuxer refused to return more audio packets.
Each time a filter error was printed.
Fortunately, it's pretty easy to workaround. We just mark the p->eof
flag as we send an EOF frame to libavfilter. The p->eof flag is used
only to recover from temporary EOF: it resets the filter if new data is
available again. We don't care much about av_buffersink_get_frame()
returning a broken EAGAIN state in this situation and essentially ignore
it, meaning if we get EAGAIN after sending EOF, we assume effectively
that EOF was fully reached.
This was in the parser code all along. As far as I can tell, *cp was
intended. There is no need to check cp for NULL (nor does it make any
sense to do so every time around the loop) for AF_CONTROL_COMMAND.
However, s->matrixstr can be NULL, so checking for that separately is in
order.
For stereo and typical L/R-first channel arrangements, this avoids
undesirable phasing artifacts, especially obvious when speed is changed
and then reset. Without this, there is a very audible change in the
stereo field even when librubberband is no longer actually making any
speed changes.
Normally I'd prefer a bunch of smaller functions with fewer parameters
over a single function with a lot of parameters. But future changes will
require messing with the parameters in a slightly more complex way, so a
combined function will be needed anyway. The now-unused "global"
parameter is required for later as well.
There are situations where the resampler is destroyed and recreated
during playback. If recreating the resampler unexpectedly fails, the
filter function is supposed to return an error. This wasn't done
correctly, because get_out_samples() accessed the resampler before the
check. Move the check up to fix this.
The code actually kept going out of EOF mode into resync mode back into
EOF mode when the playloop had to wait after an audio EOF caused by the
endpts. This would break seamless looping (as added by the next commit).
Apply endpts earlier, to ensure the filter_audio() function always
returns AD_EOF in this case.
The idiotic ao_buffer makes this an amazing pain in the ass.
The touched code is for seek resets and such - we simply want to reset
the entire resample state. But I noticed after a seek a tiny bit of
audio is missing (mpv's audio sync code inserted silence to compensate).
It turns out swr_drop_output() either does not reset some internal state
as we expect, or it's designed to drop not only buffered samples, but
also future samples.
On the other hand, libavresample's avresample_read(), does not have this
problem. (It is also pretty explicit in what it does - return/skip
buffered data, nothing else.)
Is the libswresample behavior a bug? Or a feature? Does nobody even
know? Who cares - use the hammer to unfuck the situation. Destroy and
deallocate the libswresample context and recreate it. On every seek.
The demuxer layer usually doesn't log per-stream information, and even
the replaygain information was logged only if it came from tags.
So log it in af_volume instead.
This commit adds an --audio-channel=auto-safe mode, and makes it the
default. This mode behaves like "auto" with most AOs, except with
ao_alsa. The intention is to allow multichannel output by default on
sane APIs. ALSA is not sane as in it's so low level that it will e.g.
configure any layout over HDMI, even if the connected A/V receiver does
not support it. The HDMI fuckup is of course not ALSA's fault, but other
audio APIs normally isolate applications from dealing with this and
require the user to globally configure the correct output layout.
This will help with other AOs too. ao_lavc (encoding) is changed to the
new semantics as well, because it used to force stereo (perhaps because
encoding mode is supposed to produce safe files for crap devices?).
Exclusive mode output on Windows might need to be adjusted accordingly,
as it grants the same kind of low level access as ALSA (requires more
research).
In addition to the things mentioned above, the --audio-channels option
is extended to accept a set of channel layouts. This is supposed to be
the correct way to configure mpv ALSA multichannel output. You need to
put a list of channel layouts that your A/V receiver supports.
We send a refcounted frame to the encoder, but then disrespect
refcounting rules and write to the frame data without making sure the
buffer is really writeable.
In theory this can lead to reallocation on every frame is the encoder
really keeps a reference. If we really cared, we could fix this by
providing a buffer pool. But then again, we don't care.
This code was supposed to adjust existing conversion filters (to make
them output a different format). But the code was just broken,
apparently a refactoring accident. It accessed af instead of af->prev.
The bug tended to add new conversion filters, even if an existing one
could have been used. (Can be tested by inserting a dummy lavrresample
filter followed by a format filter which forces conversion.)
In addition, it's probably better to return the actual error code if
reinitializing the filter fails. It would then respect an AF_FALSE
return value, which means format negotiation failed, instead of a
generic error.
af_volume has a volumedb sub-option, which allows the user to set an
explicit volume. Until recently, the player read back this value and
used it as initial softvol volume. But now it just overwrites it.
Instead of overwriting it, multiply the different gain values. Above
all, this will do the right thing if only softvol is used, or if the
user only adds the af_volume filter manually.
Normally, you want downmixing to happen first thing in the filter chain.
This is reflected in codec downmixing, which feeds the filter chain
downmixed audio in the first place. Doing this has the advantage of
needing less data to process. But the main motivation is that if there
is a drc filter in the chain, you want to process it the downmixed
audio.
Add an idiotic heuristic to achieve this. It tries to detect whether the
audio was indeed automatically downmixed (or upmixed). To detect what
the output format is going to be, it builds the filter chain normally,
and then retries with the heuristic applied (and for extra paranoia,
retries without the heuristic again if it fails to successfully rebuild
the filter chain for unknown reasons). This is simple and will work in
almost all cases.
Doing it in a more complete way is rather hard, because filters are so
generic. For example, we know absolutely nothing about the behavior of
af_lavfi, which creates an opaque filter graph with libavfilter. We
don't know why a filter would e.g. change the channel layout on its
output. (Our heuristic bails out in this case.) We're also slave to the
lowest common denominator of how our format negotiation works, and how
libavfilter's works.
In theory, we could make this mechanism explicit by introducing a
special dummy filter. The filter chain would then try to convert between
input and output formats at the dummy filter, which would give the user
more control over how downmix happens. On the other hand, the user could
just insert explicit conversion filters instead, so this would probably
have questionable value.
The algorithm and functionality is the same, but the code becomes much
simpler and easier to follow.
The assumption that there is only 1 conversion filter (lavrresample)
helps with the simplification, but the main change is to use the same
code for format/channels/rate. Get rid of the different AF_CONTROL_SET_*
controls, and change the af->data parameters directly. (af->data is
badly named, but essentially is a placeholder for the output format.)
Also, instead of trying to use the af_reinit() loop to init inserted
conversion filters or filters with changed output formats, do it inline,
and move the common code to a filter_reinit() function. This gets rid of
the awful retry variable.
In general, this should not change any runtime behavior.
This happens to be better for the af_volume filter (for softvol), and
saves some code too. It's "better" because you want to affect the
final filtered audio, such as after a manually inserted drc filter.
Instead of hardcoding what the libavcodec ac3 encoder expects, configure
it based on the AVCodec fields.
Unfortunately, it doesn't export the list of sample rates, so that is
done manually. This commit actually fixes the rate always to 48Khz. I
don't even know whether the other rates worked. (Possibly did, but
they'd still change the spdif parameters, and would work differently
from ad_spdif.c.)
The complex filter support that will be added makes much more complex
use of libavfilter, and I'm not going to bother with adding hacks to
keep libavfilter optional.
Remove flc-frc <-> sl<->sr. This was just plain wrong, and a mistaken
change to make 7.1 work properly on CoreAudio with 7.1(rear) layout.
Also see the following commit.
Add br-br <-> sl<->sr, because we decided that it makes sense.
Note that this "fudging" is applied only if the channel pairs are
replaced, i.e. they would get dropped and be replaced with silence. This
is done to compensate for libswresample's default rematrixing (which
takes care of some more common cases).
This also makes it refcounted, i.e. the new AVFrame will reference the
mp_audio buffers, instead of potentially forcing the consumer of the
AVFrame to copy the data.
All the extra code is for handling the >8 channels case, which requires
very messy dealing with the extended_ fields (not our fault).
