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.
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.
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.
This is probably the 3rd time the user-visible behavior changes. This
time, switch back because not normalizing seems to be the more expected
behavior from users.
This code removes filters which can not take spdif inout. This was made
so that PCM filters are transparently dropped in spdif mode.
This entered an endless loop with:
--af=lavcac3enc:::2 --audio-channels=5.1
The forced number of output channels is incompatible with spdif. It's
trying to insert af_lavrresample as conversion filter to compensate for
it. Of course this doesn't work, which triggers the PCM filter removal.
Then it goes on normally - since the new state is exactly as before, it
will try the same thing again, forever.
Fix by reusing the retry counter, which is a very dumb but very
effective measure against these cases of filter negotiation failure. We
could try to be more clever (for example, if the removed filter is a
conversion filter, we can be sure this won't work, and error out
immediately). But better keep it simple and robust.
This flag was used by some filters and made sure none of these filters
were inserted twice. This triggers only if the user explicitly tries to
add multiple filters (and not e.g. due to auto-insertion), so at best
this warned the user from doing something potentially pointless. At
worst, it blocked some (mildly) legitimate use-cases. Get rid of it.
Also see #2322.
Some users still use this filter, so the filter was going to be kept.
But I overlooked that libavfilter provides this filter. Remove the
redundant wrapper from mpv. Something like --af=lavfi=bs2b should work
and give exactly the same results.
All of these filters are considered not useful anymore by us. Some have
replacements in libavfilter (useable through af_lavfi).
af_center, af_extrastereo, af_karaoke, af_sinesuppress, af_sub,
af_surround, af_sweep: pretty simple and useless filters which probably
nobody ever wants.
af_ladspa: has a replacement in libavfilter.
af_hrtf: the algorithm doesn't work properly on most sources, and the
implementation was buggy and complicated. (The filter was inherited from
MPlayer; but even in mpv times we had to apply fixes that fixed major
issues with added noise.) There is a ladspa filter if you still want to
use it.
af_export: I'm not even sure what this is supposed to do. Possibly it
was meant for GUIs rendering audio visualizations, but it couldn't
really work well. For example, the size of the audio depended on the
samplerate (fixed number of samples only), and it couldn't retrieve the
complete audio, only fragments. If this is really needed for GUIs, mpv
should add native visualization, or a proper API for it.
Some filter chains require a huge number of auto-inserted conversion
filters. There is an overly stupid safeguard against infinite filter
insertions, which counts the number of conversion filters inserted. This
triggered accidentally in this case. Fix by resetting this counter after
a non-conversion filter was successfully configured.
Replace all the check macros with function calls. Give them all the
same case and naming schema.
Drop af_fmt2bits(). Only af_fmt2bps() survives as af_fmt_to_bytes().
Introduce af_fmt_is_pcm(), and use it in situations that used
!AF_FORMAT_IS_SPECIAL. Nobody really knew what a "special" format
was. It simply meant "not PCM".
Basically, af_fix_format_conversion() behaves stupid you insert a
conversion filter that won't work, and adding back the conversion test
function is the simplest fix to it.
This attempted to find a minimal filter graph for a format conversion
involving multiple conversion filters. With the last 2 commits it
becomes dead code - remove it.
They are useless. Not only are they actually rarely in use; but
libavcodec doesn't even output them, as libavcodec has no such sample
formats for decoded audio.
Even if it should happen that we actually still need them (e.g. if doing
direct hardware output), there are better solutions. Swapping the sign
is a fast and lossless operation and can be done inplace, so AO actually
needing it could do this directly.
If you wonder why we keep U8 instead of S8: because libavcodec does it.
Some time ago, a mechanism was added for automatically removing PCM-only
filters if the input format is spdif.
This could cause an infinite loop if the AO did not support spdif, but
was falling back to some PCM format. Then this code tried to remove the
last filter, which is a dummy filter for receiving and queuing filter
output. af_remove() simply fails gracefully in this case, so this
happens over and over again.
Fix by explicitly checking whether the filter to remove is a dummy
filter. (af_remove() also fails only if the dummy filters are attempted
to be removed - checking this directly is simpler.)
It could happen that a lavrresample filter would keep its old output
format when the decoder changed its output format. This simply happened
because the output format was never reset.
Normally, this was not an issue, because lavrresample filters only
inserted for format conversion were removed on format changes. But if
--no-audio-pitch-correction is set and playback speed is changed, then
there is a "permanent" lavrresample filter in the filter chain, which
shows this behavior.
Fix by explicitly resetting output formats for all filters which support
it.
Note: this can crash with libswresample in some cases. I'm not sure if
this is mpv's fault or libswresample's, but since it works with
libavresample, I'm going to assume it's not our's.
The af_add() function has a problem: if the inserted filter returns
AF_DETACH during init, the function will have a dangling pointer. Until
now this was avoided by making sure none of the used filters actually
return AF_DETACH, but it's getting infeasible.
Solve this by requiring passing an unique label to af_add(), which is
then used instead of the pointer.
Although the libraries we use for resampling (libavresample and
libswresample) do not support changing sampelrate on the fly, this makes
it easier to make sure no audio buffers are implicitly dropped. In fact,
this commit adds additional code to drain the resampler explicitly.
Changing speed twice without feeding audio in-between made it crash
with libavresample inc ertain cases (libswresample is fine). This is
probably a libavresample bug. Hopefully this will be fixed, and also I
attempted to workaround the situation that crashes it. (It seems to
point in direction of random memory corruption, though.)
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.
Just to make sure all filters get the correct format. Together wih the
check in af_add_output_frame(), this asserts that
af->prev->fmt_out == af->fmt_in
This also requires setting the "in" pseudo-filter (s->first) formats
correctly. Before this commit, the fmt_in/fmt_out fields weren't used
for this filter.
The goal is switching the whole audio chain to using refcounted frames.
This brings the architecture closer to FFmpeg, enables better
integration with libavfilter, will reduce useless copying somewhat, and
will probably allow better timestamp tracking.
For now, every filter goes through a semi-awful wrapper in
af_do_filter(), though. This will be fixed step by step, and the wrapper
should eventually be removed. Another thing that will have to be done is
improving the timestamp handling and avoiding extra copies for the AO.
Some of the new code is rather similar to the video filter code (the
core filter code basically just has types replaced). Such code
duplication is normally very unwanted, but in this case there's probably
no other choice. On the other hand, this code is pretty simple (even if
somewhat tricky). Maybe there will be unified filter code in the future,
but this is still far away.
The purpose of this function was to filter only as much audio input as
needed to produce a certain amount of audio output. This could (in
theory) avoid excessive buffering when e.g. changing playback speed with
resampling.
Use of this was already removed in commit 5fd8a1e0. No problems were
experienced, so let's assume this feature is practically worthless.
(Though it's possible that it was quite useful over a decade ago, or in
some cornercases with evil files.)
Makes all of overlay_add work on windows/mingw.
Since we now don't explicitly check for mmap() anymore (it's always
present), this also requires us to make af_export.c compile, but I
haven't tested it.
This rewrites the audio decode loop to some degree. Audio filters don't
do refcounted frames yet, so af.c contains a hacky "emulation".
Remove some of the weird heuristic-heavy code in dec_audio.c. Instead of
estimating how much audio we need to filter, we always filter full
frames. Maybe this should be adjusted later: in case filtering increases
the volume of the audio data, we should try not to buffer too much
filter output by reducing the input that is fed at once.
For ad_spdif.c and ad_mpg123.c, we don't avoid extra copying yet - it
doesn't seem worth the trouble.
Use a pseudo-filter when changing speed with resampling, instead of
somehow changing a samplerate somewhere. This uses the same underlying
mechanism, but is a bit more structured and cleaner. It also makes some
of the following changes easier.
Since we now always use filters to change audio speed, move most of the
work set_playback_speed() does to recreate_audio_filters().
Although the "af" command already could do this, it seems it's better
to introduce a lower level mechanism for now. This avoids some messy
issues, since that code would recursive call reinit_audio_chain().
To be used by the next commit.
There's no real reason why audio_init_filter() should exist. Just use
af_init or af_reinit directly. (We lose a useless message; the same
information is printed in a quite close place with more details.)
Requires less code, and the way the filter chain is marked as having
failed to initialize allows just switching off audio instead of
crashing if trying to insert a volume filter in mixer.c fails, and
recreating the old filter chain fails too.
The final goal is all mp_msg calls produce complete lines. We want this
because otherwise, race conditions could corrupt the terminal output,
and it's inconvenient for the client API too. This commit works towards
this goal. There's still code that has this not fixed yet, though.
While I'm not very fond of "const", it's important for declarations
(it decides whether a symbol is emitted in a read-only or read/write
section). Fix all these cases, so we have writeable global data only
when we really need.
wm4
stepshal
Marcin Kurczewski
Alessandro Ghedini