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Commit Graph

1506 Commits

Author SHA1 Message Date
wm4
8c82555e41 ao_oss: fix a dumb calculation
period_size used the wrong unit, and even if the unit had been correct,
was assigned the wrong value.

Probably fixes #4642.
2017-07-21 19:45:59 +02:00
wm4
ddd068491c Replace remaining avcodec_close() calls
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.
2017-07-16 12:51:48 +02:00
Kevin Mitchell
c5dfd66e14 ao_wasapi: remove redundant / outdated comment
Where this was moved from, it made slightly more sense. Here what the comment is
trying to say is already pretty obvious from the code.
2017-07-10 21:01:39 -07:00
Kevin Mitchell
63b6aa3f57 ao_waspi: use switch for handling fix_format errors 2017-07-10 21:01:39 -07:00
Kevin Mitchell
4389ddcc34 ao_wasapi: don't repeat format negotiation on align hack
Even if it did return a different result, the bufferFrameCount from the align
hack would be wrong anyway.
2017-07-10 21:01:39 -07:00
Kevin Mitchell
71cc28b804 ao_wasapi: fix leak on align hack 2017-07-10 21:01:39 -07:00
wm4
b016760a28 ad_spdif: minor cleanups
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.
2017-07-10 16:40:52 +02:00
Kevin Mitchell
e9f729c17c audio/out: fix comment typo 2017-07-09 13:46:13 -07:00
Kevin Mitchell
6666b25b73 ao_wasapi: enable packed 24 bit output 2017-07-09 13:46:13 -07:00
Kevin Mitchell
a081c8d372 audio/out: correct copy length in ao_read_data_converted
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
2017-07-09 13:46:13 -07:00
Kevin Mitchell
03abd704ec ao_wasapi: reorder channels and samplerates to speed up search
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.
2017-07-09 13:46:13 -07:00
Kevin Mitchell
7568715563 ao_wasapi: minor cosmetic fixes 2017-07-09 13:44:09 -07:00
Kevin Mitchell
2514e542e5 ao_wasapi: try correct initial format
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
2017-07-09 13:43:54 -07:00
wm4
03596ac551 audio: drop AF_FORMAT_S24
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.
2017-07-07 17:56:22 +02:00
wm4
300097536d ao_pcm: drop AF_FORMAT_S24 usage
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.
2017-07-07 17:56:18 +02:00
wm4
2e1eb8b37c ao_oss: drop AF_FORMAT_S24 usage
Can't test / don't care.
2017-07-07 17:56:18 +02:00
wm4
adbb429296 ao_sndio: drop AF_FORMAT_S24 usage
I can't test it, so I'm dropping it without replacement. If anyone is
interested in readding support, it would be done like the ao_alsa.c
change.
2017-07-07 17:56:18 +02:00
wm4
4e11549593 ao_wasapi_utils: be slightly more clever when converting channel map 2017-07-07 17:56:18 +02:00
wm4
951c1a4907 ao_wasapi: drop use of AF_FORMAT_S24
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.
2017-07-07 17:56:18 +02:00
wm4
4cb5e53ada ao_alsa: drop use of AF_FORMAT_S24
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.
2017-07-07 17:56:18 +02:00
wm4
90dd229871 audio/out: add helper code to do 24 bit conversion in AO
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.)
2017-07-07 17:54:05 +02:00
wm4
d5702d3b95 ad_lavc, vd_lavc, sd_lavc: consistently use avcodec_free_context()
Instead of various ad-hoc ways to achieve the same thing. (The API was
added only later.)
2017-07-06 16:25:42 +02:00
wm4
d0e8d6114b ao_coreaudio: insane hack for passing through AC3 as float PCM
This uses the same hack as Kodi uses, and I suspect MPlayer/ancient mpv
also did this (but didn't research that).
2017-06-30 09:06:01 +02:00
wm4
3e9075787f ao_wasapi: UWP wrapper hack support
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.
2017-06-29 10:38:05 +02:00
Pedro Pombeiro
4637b029cd Universal Windows Plaform (UWP) support
libmpv only. Some things are still missing.

Heavily reworked.

Signed-off-by: wm4 <wm4@nowhere>
2017-06-29 10:36:16 +02:00
Pedro Pombeiro
f22d12ac51 ao_wasapi: do not use deprecated wchar functions
These break on UWP. Based on a patch by Pedro Pombeiro.
2017-06-29 10:35:25 +02:00
wm4
cd25d98bfa Avoid calling close(-1)
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.
2017-06-29 10:31:13 +02:00
wm4
3a3a0aced2 ao_wasapi: remove subtly duplicated code
Seems like this can be slightly simplified.
2017-06-28 18:43:19 +02:00
wm4
3b7e292844 ao_wasapi: remove duplicate code for creating IAudioClient
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.
2017-06-28 18:43:18 +02:00
wm4
c5a82f729b audio/out/pull: detect and log underflows
Mostly for debugging, I guess.
2017-06-28 13:18:59 +02:00
wm4
037c37519b audio/out: require AO drivers to report period size and correct buffer
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.
2017-06-25 15:57:43 +02:00
wm4
4abd5683d5 ao_openal: change license to LGPL
All authors have agreed.
2017-06-24 14:10:14 +02:00
wm4
8922c7b84a chmap: remove misleading "downmix" channel layout name
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.
2017-06-24 11:36:10 +02:00
Niklas Haas
bbe8bb0ae9
ao_pulse: reorder format choice
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)
2017-06-23 21:12:44 +02:00
wm4
5c038e6999 build: simplify OSS checks and remove changes by "bugmen0t"
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.
2017-06-22 13:17:14 +02:00
wm4
eec7f61b5f audio/format: change license to LGPL
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.
2017-06-20 15:37:28 +02:00
wm4
6489b112ad dec_audio, ad_lavc: change license to LGPL
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.
2017-06-14 21:08:59 +02:00
Rudolf Polzer
e2573e5b8d encode_lavc: move from GPL 2+ to LGPL 2.1+. 2017-06-13 14:22:15 -04:00
wm4
cc69650e76 af, vf: improvements to libavfilter bridge
Add the "lavfi-" prefix (details see manpage additons).

Tag the filter name as "(lavfi)" in the verbose filter list output.
2017-05-31 17:42:55 +02:00
wm4
e77ed53459 ad_spdif: change license to LGPL
All authors have agreed. (Even the main author, if you wonder about the
entry in the Copyright file.)
2017-05-21 12:35:53 +02:00
wm4
43aaba4f73 ao_pcm: change license to LGPL
All relevant authors have agreed to the relicensing.

Problem cases:

eca47b1a5e: someone else gets credited for the "idea" of this change,
but it doesn't seem like it was a patch (otherwise reimar would have
said "patch"). Also, the associated code got essentially removed again
anyway. (The option parsing was rewritten fully.)

ffb529e4eb: anonymous/unknown author, but the code was fully removed
anyway. The struct was removed, and the modern code does explicit
read/write calls.

40789473d2: author was not contacted, but this code was removed
anyway. The magic number (0x7ffff000) is still in the new code, but I
don't think that is copyright relevant.

c750b8ab2d: the message was entirely removed.
2017-05-20 12:46:08 +02:00
wm4
7840125e22 audio/out: change license of some core files to LGPL
All contributors of the current code have agreed. ao.c requires a
"driver" entry for each audio output - we assume that if someone who
didn't agree to LGPL added a line, it's fine for ao.c to be LGPL
anyway. If the affected audio output is not disabled at compilation
time, the resulting binary will be GPL anyway, and ootherwise the
code is not included.

The audio output code itself was inspired or partially copied from
libao in 7a2eec4b59 (thus why MPlayer's audio code is named libao2).
Just to be sure we got permission from Aaron Holtzman, Jack Moffitt, and
Stan Seibert, who according to libao's SVN history and README are the
initial author. (Something similar was done for libvo, although the
commit relicensing it forgot to mention it.)

242aa6ebd4: anders mostly disagreed with the LGPL relicensing, but we
got permission for this particular commit.

0ef8e55573: nick could not be reached, but the include statement was
removed again anyway.

879e05a7c1: iive agreed to LGPL v3+ only, but this line of code was
removed anyway, so ao_null.c can be LGPL v2.1+.

9dd8f241ac: patch author could not be reached, but the corresponding
code (old slave mode interface) was completely removed later.
2017-05-20 11:43:57 +02:00
James Ross-Gowan
3a7b4df4bf ao_wasapi: set name of event thread 2017-05-18 00:11:14 +10:00
wm4
faefbbaaa5 af_format: change license to LGPL
This case is a bit weird, because MPlayer certainly also has a file
named af_format.c. Both appear to have the function of converting audio
data between sample formats.

However, mpv's af_format.c is a rewrite, and doesn't actually do
conversion by itself. It's similar to vf_format.c, and forces the
generic filter chain code to insert conversion filters, instead of doing
conversion explicitly.

mpv's current af_format.c started out as af_force.c in d9582ad0a4. It
was renamed to af_format.c in e60b8f181d, while the old af_format.c was
split into two new filters. In 943c785619 the filename was changed to
af_format.c as well.

The new af_format.c does not contain any libaf code, except for some
potentially copy & pasted skeleton and boilerplate code. (We don't
account for this in per-filter file licenses, as the old libaf code
has to be removed fully, at which point the filters will have to be
ported to another framework, which will removed that boilerplate code.)

The old filters based on af_format.c were progressively replaced and
removed. Support for non-native endian and formats with signedness
different from native FFmpeg was completely removed in 831d7c3c40.
The old 24 bit conversion code was removed in 552dc0d564 (made
unnecessary by 5a9f817bfd).

Also list hwdec_vaglx.c as GPL-only, which doesn't have anything to do
with this commit.
2017-05-11 11:25:45 +02:00
wm4
bda25e17b6 af_scaletempo: change license to LGPL
All authors have agreed.

The initial commit d33703496c as well as the current code contain this
line:

  * inspired by SoundTouch library by Olli Parviainen

We assume this is about the algorithm (not the code), and the author of
the original patch actually wrote all code himself.
2017-05-09 12:53:37 +02:00
wm4
5eec3d08d5 af_lavcac3enc: change license to LGPL
All authors have agreed.

As usual with these things, this probably does not include residues from
the libaf framework.
2017-05-09 12:46:40 +02:00
wm4
04df16bfd3 ao_pulse, ao_rsound: change license to LGPL
All authors have agreed.

One exception is 71247a97b3, whose author was not asked, but we deem
the change as trivial. (And technically it was replaced when the audio
chain dropped non-native endian sample formats.)
2017-05-08 14:09:49 +02:00
wm4
c87224bf1b ao_coreaudio: change license to LGPL
All authors have agreed to the relicensing.

The code was pretty much rewritten by Stefano Pigozzi. Since the rewrite
happened incrementally, and seems to include refactored portions of
older code, this relicensing was done on the pre-refactor code do.

The original commit adding this AO (as ao_macosx.c) credits Timothy J.
Wood as original author. He was asked and agreed to LGPL. It's not
entirely sure from which project this code came from, but it's probably
libao. In that project, Stanley Seibert made some changes to it (who as
a major developer of libao was asked just to be sure), and also Ralph
Giles and Ben Hines made two small changes. The latter were not asked,
but none of their code survived anyway.
2017-05-08 13:57:40 +02:00
wm4
380bc03823 ad.h: change license to LGPL
All authors have agreed.

Commit 94d3170bd0 is a bit murky: Nick could not be reached, and arpi's
changes were obviously inspired or copied from Nick's. However, the
changed symbols were removed and do not exist anymore.
2017-05-05 07:32:35 +02:00
wm4
1db603efc3 audio/fmt-conversion: change license to LGPL
Although pretty similar to the probably unrelicensable
video/fmt-conversion.c/h (basically using the same idea, but for audio),
it was written by someone else. The format mapping was first added in
commit ad95e046c2.
2017-05-05 07:25:55 +02:00