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Commit Graph

41 Commits

Author SHA1 Message Date
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
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
Kevin Mitchell
a0884c82a9 audio: add af_select_best_samplerate function
This function chooses the best match to a given samplerate from a provided
list. This can be used, for example, by the ao to decide what samplerate to use
for output.
2016-03-17 02:31:05 -07:00
wm4
617aff6cda audio: fix af_fmt_change_bytes() with spdif formats
This could accidentally change some spdif formats to AAC (because AAC is
the first on the list and will match first). spdif formats are
inherently uninterchangeable, so treat them as their own class of
formats (like int vs. float).

Might fix some issues with ao_wasapi.c.
2015-11-07 15:07:50 +01:00
Kevin Mitchell
b7144ad8bf audio/format: revise af_format_conversion_score
* (de)planarize -1
* pad 1 byte -8
* truncate 1 byte -1024
* float -> int 1048576 * (8 - dst_bytes)
* int -> float -512

Now the score is negative if and only if the conversion is lossy
(e.g. previously s24 -> float was given a negative (lossy) score),
However, int->float is still considered bad
(s16->float is worse than than s16->s32).

This penalizes any loss of precision more than performance / bandwidth hits.
For example, previously s24->s16p was considered equal to s24->u8.

Finally, we penalize padding more than (de)planarizing as this will
increase the output size for example with ao_lavc.
2015-09-10 23:58:09 -07:00
wm4
e721660e6d ao_lavc: use new sample format determination code
This is just a refactor, which makes it use the previously introduced
function, and allows us to make af_format_conversion_score() private.

(We drop 2 unlikely warning messages too... who cares.)
2015-09-10 23:38:42 +02:00
wm4
60a617df31 audio/format: add function for determining sample conversion candidates 2015-09-10 23:30:51 +02:00
wm4
e45f469280 audio/format: fix interlaved vs. non-interleaved conversions
This mixed up the returned score for some interleaved/non-interleaved
comparisons. Changing interleaving subtracted 1 point, while extending
sample size by 1 byte also subtracted 1 point.

(This scoring system is not ideal - it'd be much cleaner to do a 3-way
sample format comparison instead, and sort the formats according to the
comparison instead of the score.)
2015-09-10 23:29:31 +02:00
wm4
dc04541ba8 audio/format: actually prefer float over double sample format
...for int->float conversions. This code accidentally inverted the
condition.
2015-09-10 23:25:27 +02:00
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
wm4
d6737c5fab audio: replace format name table
Having a big switch() is simpler.
2015-06-26 23:06:21 +02:00
wm4
efb50cabe6 audio: remove bitmask format definition mess
Audio formats used a semi-clever schema to encode the properties of the
PCM encoding as bitfields into the format integer value.

The af_fmt_change_bits() implementation becomes a bit weird, but it's
an improvement to the rest of the code.

(I've always disliked it, so why not get rid of it.)
2015-06-26 23:05:16 +02:00
wm4
831d7c3c40 audio: remove S8, U16, U24, U32 formats
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.
2015-06-16 21:11:59 +02:00
Marcin Kurczewski
f43017bfe9 Update license headers
Signed-off-by: wm4 <wm4@nowhere>
2015-04-13 12:10:01 +02:00
Kevin Mitchell
46b9df9f9e audio: make all format query shortcuts macros
af_fmt_is_float and af_fmt_is_planar were previously inconsistent with
AF_FORAMT_IS_SPECIAL/AF_FORMAT_IS_IEC61937
2015-04-03 15:40:01 -07:00
wm4
eb482140d9 audio: fix spdif packet size unit
In commit 5f8b060e I blindly assumed that the packet sizes were in
pseudo-samples, but they were actually in bytes. Oops.

(The effect was that cutting the audio was a bit less precise than it
can be.)

Also remove the packet size from ad_spdif.c; it didn't actually use it,
and simply takes what the spdif "muxer" returns.
2015-03-10 17:11:38 +01:00
wm4
69c61a882d audio: fix spdif DTS packet size
Broken in one of the previous commits.
2015-03-10 15:33:01 +01:00
wm4
5f8b060ec2 ad_spdif: move frame sizes to a general function
Needed for the next commit. This commit should probably be reverted as
soon as we're working with full audio frames internally, instead of
"flat" FIFOs.
2015-03-10 15:12:52 +01:00
Kevin Mitchell
2006069ca2 ao/format: add af_fmt_is_float 2014-12-01 03:40:23 -08:00
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
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
0112143fda Split mpvcore/ into common/, misc/, bstr/ 2013-12-17 02:39:45 +01:00
wm4
3f7e1f0492 audio/format: add heuristic to estimate loss on format conversion
The added function af_format_conversion_score() can be used to select
the best sample format to convert to in order to reduce loss and extra
conversion work.

It calculates a "loss" score when going from one format to another, and
for each conversion that needs to be done a certain score is subtracted.
Thus, if you have to convert from one format to a set of other formats,
you can calculate the score for each conversion, and pick the one with
the highest score.

Conversion between int and float is considered the worst case. One odd
consequence is that when converting from s32 to u8 or float, u8 will be
picked.

Test program used to develop this follows:

#define MAX_FMT 200
struct entry {
    const char *name;
    int score;
};

static int compentry(const void *px1, const void *px2)
{
    const struct entry *x1 = px1;
    const struct entry *x2 = px2;
    if (x1->score > x2->score)
        return 1;
    if (x1->score < x2->score)
        return -1;
    return 0;
}

int main(int argc, char *argv[])
{
    for (int n = 0; af_fmtstr_table[n].name; n++) {
        struct entry entry[MAX_FMT];
        int entries = 0;
        for (int i = 0; af_fmtstr_table[i].name; i++) {
            assert(i < MAX_FMT);
            entry[entries].name = af_fmtstr_table[i].name;
            entry[entries].score =
                af_format_conversion_score(af_fmtstr_table[i].format,
                                           af_fmtstr_table[n].format);
            entries++;
        }
        qsort(&entry[0], entries, sizeof(entry[0]), compentry);
        for (int i = 0; i < entries; i++) {
            printf("%s -> %s: %d \n", af_fmtstr_table[n].name,
                   entry[i].name, entry[i].score);
        }
    }
}
2013-11-16 21:46:17 +01:00
wm4
514c454770 audio: drop "_NE"/"ne" suffix from audio formats
You get the native format by not appending any suffix to the format.

This change includes user-facing names, e.g. for the --format option.
2013-11-15 21:25:05 +01:00
wm4
b2d4b5ee43 audio/format: add non-interleaved audio formats 2013-11-12 23:16:27 +01:00
wm4
3cb4116243 ao: add ao_play_silence, use for ao_alsa and ao_oss
Also add a corresponding function to audio/format.c, which fills an
audio block with silence.
2013-11-10 23:05:59 +01:00
wm4
d74bac22b9 audio/format: convert format macros to enum, drop NE suffix
Turn the sample format definitions into an enum. (The format bits are
still macros.) The native endian versions of the new definitions don't
have a NE suffix anymore, although there are still compatibility defines
since too much code uses the NE variants.

Rename the format bits for special formats to help to distinguish them
from the actual definitions, e.g. AF_FORMAT_AC3 to AF_FORMAT_S_AC3.
2013-11-07 22:13:20 +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
aa48eeac97 audio/format: reformat 2013-11-07 22:12:26 +01:00
wm4
33707c6d63 audio/format: add some helper functions 2013-10-22 01:01:41 +02:00
wm4
ddc9733446 audio: don't allow setting unknown formats from command line
af_str2fmt_short(), which is used by the command line option parser,
allowed passing a hex number. The user could set arbitrary integers as
internal audio formats, even formats which don't exist or make no sense.
This is not very useful, so get rid of it.
2013-08-26 10:09:44 +02:00
wm4
53b5227270 audio: make internal audio format 0 an invalid format
Having to use -1 for that is generally quite annoying.

Audio formats are created from bitmasks, and it can't be excluded that
0 is not a valid format. Fix this by adjusting AF_FORMAT_I so that it
is never 0. Along with AF_FORMAT_F and the special formats, all valid
formats are covered and guaranteed to be non-0.

It's possible that this commit will cause some regressions, as the
check for invalid audio formats changes a bit.
2013-08-26 10:09:41 +02:00
Stefano Pigozzi
c8c70dce57 audio: fix af_fmt_seconds_to_bytes
Was missing samplerate
2013-06-16 19:28:04 +02:00
Stefano Pigozzi
bff03a181f core: add a spsc ringbuffer implementation
Currently every single AO was implementing it's own ringbuffer, many times
with slightly different semantics. This is an attempt to fix the problem.

I stole some good ideas from ao_portaudio's ringbuffer and went from there.
The main difference is this one stores wpos and rpos which are absolute
positions in an "infinite" buffer. To find the actual position for writing /
reading just apply modulo size.

The producer only modifies wpos while the consumer only modifies rpos. This
makes it pretty easy to reason about and make the operations thread safe by
using barriers (thread safety is guaranteed only in the Single-Producer/Single-
Consumer case).

Also adapted ao_coreaudio to use this ringbuffer.
2013-06-16 18:20:39 +02:00
wm4
9dd9ccbd8d audio: add double sample format
To make this easier, get rid of the direct mapping of the
AF_FORMAT_BITS_MASK bit field to number of bytes. This way we can throw
away the unused AF_FORMAT_48BIT and don't have to add ..._56BIT.
2013-05-12 21:24:57 +02:00
wm4
f7a427676c audio: add some setters for mp_audio, and require filters to use them
mp_audio has some redundant fields. Setters like mp_audio_set_format()
initialize these properly.

Also move the mp_audio struct to a the file audio.c.

We can remove a mysterious line of code from af.c:

    in.format |= af_bits2fmt(in.bps * 8);

I'm not sure if this was ever actually needed, or if it was some kind of
"make it work" quick-fix that works against the way things were supposed
to work. All filters etc. now set the format correctly, so if there ever
was a need for this code, it's definitely gone.
2013-05-12 21:24:54 +02:00
wm4
74ab902dea audio: remove support for native alaw/mulaw/adpcm output
This is considered a worthless feature. Note that alaw/mulaw/adpcm input
is unaffected: such data is handed to libavcodec and "decoded" to linear
PCM.
2012-12-11 00:37:54 +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