Commit Graph

31 Commits

Author SHA1 Message Date
wm4 6147bcce35 audio: fix format function consistency issues
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".
2015-06-26 23:06:37 +02:00
Marcin Kurczewski f43017bfe9 Update license headers
Signed-off-by: wm4 <wm4@nowhere>
2015-04-13 12:10:01 +02:00
wm4 c8ecb66269 ao_pcm: add append mode
Pretty useful for debugging, although a bit useless or possibly
misleading too (see comments in the manpage).
2015-01-14 22:14:56 +01:00
wm4 df43e2d22a ao_pcm: simplify
Also shuts up Coverity.
2014-11-21 10:09:38 +01:00
wm4 81bf9a1963 audio: cleanup spdif format definitions
Before this commit, there was AF_FORMAT_AC3 (the original spdif format,
used for AC3 and DTS core), and AF_FORMAT_IEC61937 (used for AC3, DTS
and DTS-HD), which was handled as some sort of superset for
AF_FORMAT_AC3. There also was AF_FORMAT_MPEG2, which used
IEC61937-framing, but still was handled as something "separate".

Technically, all of them are pretty similar, but may use different
bitrates. Since digital passthrough pretends to be PCM (just with
special headers that wrap digital packets), this is easily detectable by
the higher samplerate or higher number of channels, so I don't know why
you'd need a separate "class" of sample formats (AF_FORMAT_AC3 vs.
AF_FORMAT_IEC61937) to distinguish them. Actually, this whole thing is
just a mess.

Simplify this by handling all these formats the same way.
AF_FORMAT_IS_IEC61937() now returns 1 for all spdif formats (even MP3).
All AOs just accept all spdif formats now - whether that works or not is
not really clear (seems inconsistent due to earlier attempts to make
DTS-HD work). But on the other hand, enabling spdif requires manual user
interaction, so it doesn't matter much if initialization fails in
slightly less graceful ways if it can't work at all.

At a later point, we will support passthrough with ao_pulse. It seems
the PulseAudio API wants to know the codec type (or maybe not - feeding
it DTS while telling it it's AC3 works), add separate formats for each
codecs. While this reminds of the earlier chaos, it's stricter, and most
code just uses AF_FORMAT_IS_IEC61937().

Also, modify AF_FORMAT_TYPE_MASK (renamed from AF_FORMAT_POINT_MASK) to
include special formats, so that it always describes the fundamental
sample format type. This also ensures valid AF formats are never 0 (this
was probably broken in one of the earlier commits from today).
2014-09-23 23:11:54 +02:00
wm4 b745c2d005 audio: drop swapped-endian audio formats
Until now, the audio chain could handle both little endian and big
endian formats. This actually doesn't make much sense, since the audio
API and the HW will most likely prefer native formats. Or at the very
least, it should be trivial for audio drivers to do the byte swapping
themselves.

From now on, the audio chain contains native-endian formats only. All
AOs and some filters are adjusted. af_convertsignendian.c is now wrongly
named, but the filter name is adjusted. In some cases, the audio
infrastructure was reused on the demuxer side, but that is relatively
easy to rectify.

This is a quite intrusive and radical change. It's possible that it will
break some things (especially if they're obscure or not Linux), so watch
out for regressions. It's probably still better to do it the bulldozer
way, since slow transition and researching foreign platforms would take
a lot of time and effort.
2014-09-23 23:09:25 +02:00
wm4 5b5a3d0c46 audio: remove swapped-endian spdif formats
IEC 61937 frames should always be little endian (little endian 16 bit
words). I don't see any apparent need why the audio chain should handle
swapped-endian formats.

It could be that some audio outputs might want them (especially on big
endian architectures). On the other hand, it's not clear how that works
on these architectures, and it's not even known whether the current code
works on big endian at all. If something should break, and it should
turn out that swapped-endian spdif is needed on any platform/AO,
swapping still could be done in-place within the affected AO, and
there's no need for the additional complexity in the rest of the player.

Note that af_lavcac3enc outputs big endian spdif frames for unknown
reasons. Normally, the resulting data is just pulled through an auto-
inserted conversion filter and turned into little endian. Maybe this was
done as a trick so that the code didn't have to byte-swap the actual
audio frame. In any case, just make it output little endian frames.

All of this is untested, because I have no receiver hardware.
2014-09-23 19:34:14 +02:00
wm4 6c9ce5bee2 ao_pcm: minor simplification 2014-09-06 12:58:54 +02:00
Amos Onn 8593c4f70b ao_pcm: fix message strings
Signed-off-by: wm4 <wm4@nowhere>
2014-06-15 09:25:15 +02:00
Marcoen Hirschberg 31a10f7c38 af_fmt2bits: change to af_fmt2bps (bytes/sample) where appropriate
In most places where af_fmt2bits is called to get the bits/sample, the
result is immediately converted to bytes/sample. Avoid this by getting
bytes/sample directly by introducing af_fmt2bps.
2014-05-28 21:38:00 +02:00
wm4 c7e620df96 af_lavrresample: remove avresample_set_channel_mapping() fallbacks
This function is now always available.

Also remove includes of reorder_ch.h from some AOs (these are just old
relicts).
2014-03-16 13:19:28 +01:00
wm4 e16c91d07a audio/out: make draining a separate operation
Until now, this was always conflated with uninit. This was ugly, and
also many AOs emulated this manually (or just ignored it). Make draining
an explicit operation, so AOs which support it can provide it, and for
all others generic code will emulate it.

For ao_wasapi, we keep it simple and basically disable the internal
draining implementation (maybe it should be restored later).

Tested on Linux only.
2014-03-09 01:27:41 +01:00
wm4 41f2b26d11 audio/out: make ao struct opaque
We want to move the AO to its own thread. There's no technical reason
for making the ao struct opaque to do this. But it helps us sleep at
night, because we can control access to shared state better.
2014-03-09 00:19:31 +01:00
wm4 0112143fda Split mpvcore/ into common/, misc/, bstr/ 2013-12-17 02:39:45 +01:00
wm4 eb15151705 Move options/config related files from mpvcore/ to options/
Since m_option.h and options.h are extremely often included, a lot of
files have to be changed.

Moving path.c/h to options/ is a bit questionable, but since this is
mainly about access to config files (which are also handled in
options/), it's probably ok.
2013-12-17 02:07:57 +01:00
wm4 380fc765e4 audio/out: prepare for non-interleaved audio
This comes with two internal AO API changes:

1. ao_driver.play now can take non-interleaved audio. For this purpose,
the data pointer is changed to void **data, where data[0] corresponds to
the pointer in the old API. Also, the len argument as well as the return
value are now in samples, not bytes. "Sample" in this context means the
unit of the smallest possible audio frame, i.e. sample_size * channels.

2. ao_driver.get_space now returns samples instead of bytes. (Similar to
the play function.)

Change all AOs to use the new API.

The AO API as exposed to the rest of the player still uses the old API.
It's emulated in ao.c. This is purely to split the commits changing all
AOs and the commits adding actual support for outputting N-I audio.
2013-11-12 23:27:51 +01:00
wm4 bf60281ffb audio/out: reject non-interleaved formats
No AO can handle these, so it would be a problem if they get added
later, and non-interleaved formats get accepted erroneously. Let them
gracefully fall back to other formats.

Most AOs actually would fall back, but to an unrelated formats. This is
covered by this commit too, and if possible they should pick the
interleaved variant if a non-interleaved format is requested.
2013-11-12 23:16:31 +01:00
wm4 91626b1c06 audio: replace af_fmt2str_short -> af_fmt_to_str
Also, remove all af_fmt2str usages.
2013-11-07 22:12:36 +01:00
wm4 d58d4ec93c audio/out: remove useless info struct and redundant fields 2013-10-23 19:30:02 +02:00
wm4 bb5fe4d874 ao_pcm: big endian AC3 in wav doesn't work
At least not with ffmpeg.

Honestly, I have no idea how little endian AC3 works at all, since
ao_pcm doesn't do anything special about it, and treats it like s16le.
Maybe it's broken and ffmpeg has special logic to detect it.
2013-10-22 01:01:07 +02:00
wm4 edd36a3afc audio/out: do some mp_msg conversions
Use the new MP_ macros for some AOs instead of mp_msg.

Not all AOs are converted, and some only partially. In some cases, some
additional cosmetic changes are made.
2013-08-22 23:12:35 +02:00
Stefano Pigozzi 406241005e core: move contents to mpvcore (2/2)
Followup commit. Fixes all the files references.
2013-08-06 22:52:31 +02:00
wm4 f32a90a839 audio/out: remove options argument from init()
Same as with VOs in the previous commit.
2013-07-22 22:58:09 +02:00
wm4 38f81c618e ao_pcm: use new option API 2013-07-21 23:27:31 +02:00
wm4 4d3a2c7e0d audio/out: remove ao->outburst/buffersize fields
The core didn't use these fields, and use of them was inconsistent
accross AOs. Some didn't use them at all. Some only set them; the values
were completely unused by the core. Some made full use of them.

Remove these fields. In places where they are still needed, make them
private AO state.

Remove the --abs option. It set the buffer size for ao_oss and ao_dsound
(being ignored by all other AOs), and was already marked as obsolete. If
it turns out that it's still needed for ao_oss or ao_dsound, their
default buffer sizes could be adjusted, and if even that doesn't help,
AO suboptions could be added in these cases.
2013-06-16 19:36:56 +02:00
wm4 ecc6e379b2 audio/out: channel map selection
Make all AOs use what has been introduced in the previous commit.

Note that even AOs which can handle all possible layouts (like ao_null)
use the new functions. This might be important if in the future
ao_select_champ() possibly honors global user options about downmixing
and so on.
2013-05-12 21:24:57 +02:00
wm4 ce2515ddb8 ao: remove ao_driver.is_new field
Is unused, is completely pointless.
2013-05-12 21:24:56 +02:00
wm4 aea2328906 audio/out: switch to channel map
This actually breaks audio for 5/6/8 channels. There's no reordering
done yet. The actual reordering will be done inside of af_lavrresample
and has to be made part of the format negotiation.
2013-05-12 21:24:54 +02:00
wm4 358dc47314 ao_pcm: fix references to -novideo
The option is -no-video. Remove the deprecated "fast" suboption, which
did nothing and instructed the user to use "-novideo" instead.

Fix a reference to -novideo in encoding.rst.

Add a "generic" entry about -no-* to the list of renamed options. The
change is already explicitly mentioned in the text above the table, but
even if it's redundant, it makes it harder to overlook.
2012-12-03 21:08:48 +01:00
wm4 4873b32c59 Rename directories, move files (step 2 of 2)
Finish renaming directories and moving files. Adjust all include
statements to make the previous commit compile.

The two commits are separate, because git is bad at tracking renames
and content changes at the same time.

Also take this as an opportunity to remove the separation between
"common" and "mplayer" sources in the Makefile. ("common" used to be
shared between mplayer and mencoder.)
2012-11-12 20:08:18 +01:00
wm4 d4bdd0473d Rename directories, move files (step 1 of 2) (does not compile)
Tis drops the silly lib prefixes, and attempts to organize the tree in
a more logical way. Make the top-level directory less cluttered as
well.

Renames the following directories:
    libaf -> audio/filter
    libao2 -> audio/out
    libvo -> video/out
    libmpdemux -> demux

Split libmpcodecs:
    vf* -> video/filter
    vd*, dec_video.* -> video/decode
    mp_image*, img_format*, ... -> video/
    ad*, dec_audio.* -> audio/decode

libaf/format.* is moved to audio/ - this is similar to how mp_image.*
is located in video/.

Move most top-level .c/.h files to core. (talloc.c/.h is left on top-
level, because it's external.) Park some of the more annoying files
in compat/. Some of these are relicts from the time mplayer used
ffmpeg internals.

sub/ is not split, because it's too much of a mess (subtitle code is
mixed with OSD display and rendering).

Maybe the organization of core is not ideal: it mixes playback core
(like mplayer.c) and utility helpers (like bstr.c/h). Should the need
arise, the playback core will be moved somewhere else, while core
contains all helper and common code.
2012-11-12 20:06:14 +01:00