Completely untested (rsound dev libs unavailable on my system). Trivial
enough that it's very likely that it'll just work. No port selection,
but could be added by parsing it as part of the device name.
Should fix#4714.
dst was not supposed to be initialized, the mp_audio_ setters (which
initialize dst's fields) assume it is -> shit happens. Regression from
recent changes. Was probably harmless.
This is pretty pointless, but I believe it allows us to claim that the
new code is not affected by the copyright of the old code. This is
needed, because the original mp_audio struct was written by someone who
has disagreed with LGPL relicensing (it was called af_data at the time,
and was defined in af.h).
The "GPL'ed" struct contents that surive are pretty trivial: just the
data pointer, and some metadata like the format, samplerate, etc. - but
at least in this case, any new code would be extremely similar anyway,
and I'm not really sure whether it's OK to claim different copyright. So
what we do is we just use AVFrame (which of course is LGPL with 100%
certainty), and add some accessors around it to adapt it to mpv
conventions.
Also, this gets rid of some annoying conventions of mp_audio, like the
struct fields that require using an accessor to write to them anyway.
For the most part, this change is only dumb replacements of mp_audio
related functions and fields. One minor actual change is that you can't
allocate the new type on the stack anymore.
Some code still uses mp_audio. All audio filter code will be deleted, so
it makes no sense to convert this code. (Audio filters which are LGPL
and which we keep will have to be ported to a new filter infrastructure
anyway.) player/audio.c uses it because it interacts with the old filter
code. push.c has some complex use of mp_audio and mp_audio_buffer, but
this and pull.c will most likely be rewritten to do something else.
This was _always_ called, even if the resampling was static, or the
filter was inserted for format conversion only. This should have been
fine, as I expected the function not to enable resampling when the
compensation is unset, and the source/target rates are the same. But
this is not the case, and it always enables resampling.
So explicitly avoid the call. If we have already called it successfully,
it's better not do avoid it (to overwrite the previous compensation
value), but it will also be cheap/no-op then.
Probably fixes#4716.
Any bad HRESULTs should have been printed already and lots of failure modes
don't have an HRESULT leading to awkward hr = E_FAIL business.
This also checks the exit status of GetBufferSize in the align hack. A final
fatal message is added if either of the retry hacks fail.
This API isn't deprecated (yet?), but it's still inferior and harder to
use than avcodec_free_context().
Leave the call only in 1 case in af_lavcac3enc.c, where we apparently
seriously close and reopen the encoder for whatever reason.
Use avcodec_free_context() unstead of random other calls. Actually it
was already used in the second case, but calling avcodec_close() is
redundant.
Don't crash if allocating a codec context fails.
Previously, the entire convert_buffer was being copied to the desination without
regard to the fact that it may be packed and therefore smaller.
The allocated conversion buffer was also way to big
bytes * (channels * samples) ** 2
instead of
bytes * channels * samples
This shouldn't affect which are chosen, but it should speed up the search by
putting more common configurations earlier so that a working sample format and
sample rates can be found sooner obviating the need to search them for each
iteration of the outer loops.
The loop to select the native wasapi_format for the incoming audio was
not breaking correctly when it found the most desirable format. It
therefore executed completely leaving the least desirable format (u8) as
the choice.
fixes#4582
This is the last sample format that was only in mpv and not in FFmpeg
(except the spdif special formats). It was a huge pain, even if the
removed code in af_lavrresample is pretty small after all.
Note that this drops S24 from the ao_coreaudio AOs too. I'm not sure
about the impact, but I expect it doesn't matter.
af_fmt_change_bytes() was unused as well, so remove that too.
I'd actually be somewhat interested in supporting this, as it could help
testing the S24 conversion code. But then again it's only a pain,
there's no immediate need, and it would require new options to make
ao_pcm.c select this output format at all.
Do conversion directly, using the infrastructure that was added before.
This also rewrites part of format negotation, I guess.
I couldn't test the format that was used for S24 - my hardware does not
report support for it. So I commented it, as it could be buggy. Testing
this with the wasapi_formats[] entry for 24/24 uncommented would be
appreciated.
Instead of the infrastructure added in the previous commit to do the
conversion within the AO.
If this is used, and snd_pcm_status_get_avail() returns more frames than
snd_pcm_write*() actually accepts, you will get some nice audio
corruption.
Also, this mutates the data passed via play(), which is rather fishy,
but sort of doesn't matter for now. Surely this will cause unintended
bugs and WTFs.
I plan to remove the S24 sample formats in mpv. It seems like we should
still support this _somehow_ in AOs though. So the idea is to convert
the data to more obscure representations (that would not be useful for
filtering etc. anyway) within the AO.
This commit adds helper to enable this. ao_convert_fmt is meant to
provide mechanisms for this, rather than a generic audio format
description (as the latter leads only to overly generic misery). The
conversion also supports only cases which we think will be needed at
all.
The main advantage of this approach is that we get S24 out of sight,
and that we could support other crazy formats (like S20). The main
disadvantage is that usually S32 will be selected (if both S32 and S24
are available), and there's no user control to force S24. That doesn't
really matter though, and at worst makes testing harder or will lead
to unpleasant arguments with audiophiles (they'd be wrong anyway).
ao_convert_fmt.pad_lsb is ignored, although if we ever find a case in
which playing S32 with data in the LSBs breaks when playing it as padded
24 bit format. (For example, WAVEFORMATEXTENSIBLE recommends setting the
unused bits to 0 if wValidBitsPerSample implies LSB padding.)
UWP does not support the whole IMMDevice API. Instead, you need to use a
new API (available starting from Windows 8), which is in addition not in
MinGW, and extremely unpleasant to use.
The wasapiuwp2.dll wrapper is a small custom MSVC DLL, which does this
instead, and returns a normal IAudioClient.
Before this, ao_wasapi did not initialize on UWP.
While this is perfectly OK on Unix, it causes annoying valgrind
warnings, and might be otherwise confusing to others.
On Windows, the runtime can actually abort the process if this is
called.
push.c part taken from a patch by Pedro Pombeiro.
The code accounting for the terrible AUDCLNT_E_BUFFER_SIZE_NOT_ALIGNED
semantics (which MSDN claims can happen "starting with Windows 7" - so
probably on Windows 10 too) duplicated the call for creating the
IAudioClient. That's not great, so get rid of it.
Let wasapi_thread_init() handle this. It has a retry loop anyway. This
redoes device lookup and format negotiation, but potential failures due
to race conditions (what if the driver decides to change behavior)
shouldn't be worse than before.
Before this change, AOs could have internal alignment, and play() would
not consume the trailing data if the size passed to it is not aligned.
Change this to require AOs to report their alignment (via period_size),
and make sure to always send aligned data.
The buffer reported by get_space() now always has to be correct and
reliable. If play() does not consume all data provided (which is bounded
by get_space()), an error is printed.
This is preparation for potential further AO changes.
I casually checked alsa/lavc/null/pcm, the other AOs might or might not
work.
I'm not even sure when/if FFmpeg produces those. It's just confusing. If
you really need this, you can still use dl-dr. I expect that most use is
unintentional.
Probably fixes#4545.
Right now, the current order pretty much means that pulse defaults to
S16 for arbitrary unsupported formats, but fallback to float would make
more sense since it's the easiest to convert everything to without
requiring dithering, and PA will probably just internally convert things
to float anyway.
Also move S32 above S16, which essentially means format_maps is sorted
by preference. (Although ao_pulse currently ignores this and always
picks the first as a fallback)
The user bugmen0t was apparently a shared github account with publicly
available login. Thus, we can't get LGPL relicensing permission from the
people who used this account. To relicense successfully, we have to
remove all their changes.
This commit should remove 20d1fc13, f26fb009, defbe48d. It also should
remove whatever test fragments were copied from the ancient configure,
as well as some configure logic (potentially that device path stuff).
I think this change still preserves the most important use-cases of OSS:
BSDs, and the Linux OSS emulation (the latter for testing only).
According to an OSS user, the 4front checks were probably broken anyway.
The SunAudio stuff was probably for (Open)Solaris, which is dead.
ao_oss.c itself will remain GPL, and still contains bugmen0t changes.
Although the origins lie somewhere in libaf, which was written by
"anders" and who explicitly disagreed with the LGPL relicensing, we can
change the license of these files, because all code was written by
"alex", who agreed with the relicensing.
The only things that remain from anders' code is the AF_FORMAT_ and af_
prefixes (see e.g. 66f4e563). It was alex who redid this file and added
the format identifiers we have today (507121f7). It's also nice to see
that alex actually claimed copyright on format.c (221a599f). In commit
efb50cab even the bitmask concept (which anders introduced with his
early af_format.c code) was removed, and essentially all lines and
symbols by anders were dropped.
To put it into perspective: the original af_format code was for
converting actual sample data and relied on OSS sample format
identifiers, mpv's format.c/h provides its own sample formats, but
does not do any data conversion.
Remove an now inaccurate comment from format.c (it somehow even survived
the typo that was present in the original commit). Also remove most of
the format.c include statements - most of them are technically anders'
code. We keep limits.h though.
All relevant authors of the current code have agreed.
As always, there are the usual historical artifacts that could be
mentioned. For example, there used to be a large number of decoders
by various authors who were not asked, but whose code was all 100%
removed. (Mostly due to FFmpeg providing all codecs.)
One point of contention is that Nick Kurshev might have refactored the
old audio decoder code in 2001. Basically, there are hints that it might
have been done by him, such as Arpi's commit message stating that the
code was imported from MPlayerXP (Nick's fork), or all the files having
his name in the "maintainer" field. On the other hand, the murky history
of ad.h weakens this - it could be that Arpi started this work, and Nick
took it (and possibly finished it).
In any case, Nick could not be reached, so there is no agreement for
LGPL relicensing from him. We're changing the license anyway, and assume
that his change in itself is not copyrightable. He only moved code, and
in addition used the equivalent video decoder framework (done by Arpi,
who agreed) as template. For example, ad_functions_s was basically
vd_functions_s, which the signature of the decode callback changed to
the same as audio_decode(). ad_functions_s also had a comment that said
it interfaces with "video decoder drivers" (I'm fixing this comment in
this commit).
I verified that no additional code was added that is copyright-relevant,
still in today's code, and not copied from the existing code at the time
(either from the previous audio decoder code or the video framework
code). What apparently matters here is that none of the old code was not
written by Nick, and the authors of the old code have given his
agreement, and (probably) that Nick didn't add actual new code (none
that would have survived), that was not trivially based on the old one
(i.e. no new copyrightable "work").
A copyright expert told me that this kind of change can be considered
not relevant for copyright, so here we go.
Rewriting this would end with the same code anyway, and the naming
conventions can't be copyrighted.
