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.
This tried to use AF_FORMAT_DOUBLE as KSDATAFORMAT_SUBTYPE_IEEE_FLOAT,
with wBitsPerSample==64. This is probably not allowed, and drivers
appear to react inconsistently to it. (With one user, the format was
accepted during format negotiation, but then rejected on actual init.)
Remove it, which essentially forces it to fall back to some other
format. (Looks like it'll use af_select_best_samplerate(), which would
probably make it try S32 next.)
The af_fmt_from_planar() is so that we don't have to care about
AF_FORMAT_FLOATP. Wasapi always requires packed data anyway.
This should actually handle other potentially unknown sample formats
better.
This changes that set_waveformat() always set the exact format. Now it
might set a "close" format instead. But all callers seem to deal with
this well. Although in theory, callers should probably handle the
fallback. The next cleanup (if ever) can take care of this.
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>
The buffer_size may be updated before the process callback is called for
the first time. Or, the connection graph could change, which changes the
latency of the pipeline after mpv's output. Ensure we keep on top of
these changes by registering callbacks to update our latency estimation.
It appears some device can be missing if we filter too many. In
particular, I've seen devices starting with "front" and "sysdefault"
being mapped to different hardware. I conclude that it's not sane trying
to present a nice device list to users in ALSA. It's fucked. (Although
kodi appears to attempt some intense "beautification" of the device
list, which includes parsing parameters from the device name and such.
Well, let's not.)
No other audio API requires such ridiculous acrobatics.
Apparently POLLERR can be set if poll is called while the device is in
the SND_PCM_STATE_PREPARED state. So assume that we can simply call
snd_pcm_status() to check whether the error is because the device went
away (i.e. we expect it to return ENODEV if this happened).
This avoids sporadic device lost warnings and AO reloads. The actual
device lost case is untested.
For example, previously, --audio-device='alsa/' would provide ao->device="" to
the alsa driver in spite of the fact that this is an already parsed option. To
avoid requiring a check of ao->device[0] in every driver, make sure this never
happens.
Fall back on PATH_DEV_DSP if nothing is set.
This mirrors the behaviour of --audio-device / --alsa-device.
There doesn't appear to be a general way to list devices with oss, so
--audio-device=help doesn't list oss devices except for the default one if the
file exists.
Previously --audio-device was ignored entirely by ao_oss.
fixes#4122
See: https://msdn.microsoft.com/en-us/library/windows/desktop/dd743946.aspx
Microsoft example code often uses a SAFE_RELEASE macro like the one in
the above link. This makes it easier to avoid errors when releasing COM
interfaces. It also reduces noise in COM-heavy code.
ao_wasapi.h also had a macro called SAFE_RELEASE, though unlike the
version above, its SAFE_RELEASE macro accepted a second parameter which
allowed it to destroy arbitrary objects other than just COM interfaces.
This renames ao_wasapi's SAFE_RELEASE to SAFE_DESTROY, which should more
accurately reflect what it does and prevent confusion with the Microsoft
version.
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.
This can be useful in other contexts.
Note that we end up setting AVCodecContext.width/height instead of
coded_width/coded_height now. AVCodecParameters can't set coded_width,
but this is probably more correct anyway.
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.
Since we set "skip_manual", we can actually get frames with this set.
Currently, only AV_PKT_FLAG_DISCARD will trigger this flag, and only
mov.c sets the latter flags, so this is related to FFmpeg's half-broken
mp4 edit list support.
Apparently you set the native sample rate when passing through AC3.
This fixes passthrough with 44100 Hz AC3.
Avoid opening a decoder for this and only open the parser. (Hopefully
DTS will also support this some time in the future or so - having to
open a decoder just to get the profile is dumb.)
Same deal as with video. Including the EOF handling.
(It would be nice if this code were not duplicated, but right now we're
not even close to unifying the audio and video code paths.)
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.
Remove ad_spdif from the normal codec list, and select it explicitly.
One goal was to decouple this from the normal codec selection, so
they're less entangled and the decoder selection code can be simplified
in the far future. This means spdif codec selection is now done
explicitly via select_spdif_codec(). We can also remove the weird
requirements on "dts" and "dts-hd" for the --audio-spdif option, and it
can just do the right thing.
Now both video and audio codecs consist of a single codec family each,
vd_lavc and ad_lavc.
Currently, if init_filter fails after lavf_ctx is allocated, uninit is called
which frees lavf_ctx, but doesn't clear the pointer in spdif_ctx. So, on the
next call of decode_packet, it thinks it is already initialized and uses it,
resulting in a crash on my system.
We log a large number of formats, but we rarely log the result of the
probing. Change this.
The logic in try_format_exclusive() changes slightly, but should be
equivalent. EXIT_ON_ERROR() checks for FAILED(), which should be
exclusive to SUCCEEDED().
Long planned. Leads to some sanity.
There still are some rather gross things. Especially g_groups is ugly,
and a hack that can hopefully be removed. (There is a plan for it, but
whether it's implemented depends on how much energy is left.)
Until now, this was only implemented for ao_alsa and AOs not using
push.c. ao_alsa.c relied on enabling funny underrun semantics for
avoiding resets on lower levels, while other AOs using push.c didn't do
anything.
Change this and at least make push.c copy silent data to the AO. This
still isn't perfect as keeping track of how much silence was played when
seems complex, so we don't do it. The consequence is that frame-stepping
will essentially randomize the A/V offset (it'll recover immediately
when unpausing, but still ugly). Also, in order to empty the currently
buffered audio on seeks etc., we still call ao_driver->reset and so on,
so the AO driver will still need to handle this specially.
The intent is to make behavior with ALSA less weird (for one we can
remove the code in ao_alsa.c that tries to trigger an initial
underflow). Also might help with #3754.
The "default" entry (which is and always was mpv/mplayer's default) does
not have a description set in the ALSA API. (While "sysdefault"
strangely has.)
Instead of an empty description, this should show something nice, so
reuse the ao.c code for naming default devices (see previous commit).
It's still a bit ugly that audio-device-list will have a default entry
for "Autoselect device" and "Default (alsa)", but then again we probably
want to allow the user to force ALSA (i.e. prevent fallbacks to other
AOs) just because ALSA is so flaky and makes this a legitimate feature.
This will make it easier for AOs to add explicit default device entries.
(See next commit.)
Hopefully this change doesn't lead accidentally to bogus "Default"
entries to appear, but then it can only happen if the device ID is
empty, which would mean the underlying audio API returned bogus entries.
Use the device name as fallback. This is ugly, but still better than
skipping the description entirely. This can be an issue on ALSA, where
the API can return entries without proper description.
This happens when ALSA gives us more channels than we asked for, for
whatever reasons. It looks like this wasn't handled correctly. The mpv
and ALSA channel counts could mismatch, which would lead to UB.
I couldn't actually trigger this case, though. I'm fairly sure that
drivers or plugins exist that do it anyway. (Inofficial ALSA motto: if
it can be broken, then why not break it?)
If the input is already mono or stereo, or if channel map selection
results in mono or stereo, then disable further use of the champ ALSA
API (or rather, stop trusting its results). Then we behave like a simple
application that only wants to output mono or stereo.
See #3045 and #2905. I couldn't actually test these cases, but this
commit is supposed to fix them.
set_chmap() skipped _setting_ the ALSA chmap if chmap use was requested
to be disabled by setting dev_chmap.num=0 by the caller, but it still
queried the current ALSA channel map. We don't trust it that much, so
disable that as well.
But we still query and log it, because that could be helpful for
debugging. Otherwise we could skip the entire set_chmap() call in these
cases.
