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Commit Graph

24 Commits

Author SHA1 Message Date
wm4
1a1e631ccd build: deal with endian mess
There is no standard mechanism for detecting endianess. Doing it at
compile time in a portable way is probably hard. Doing it properly
with a configure check is probably hard too. Using the endian
definitions in <sys/types.h> (usually includes <endian.h>, which is
not available everywhere) works under circumstances, but the previous
commit broke it on OSX.

Ideally all code should be endian dependent, but that is not possible
due to the dependencies (such as FFmpeg, some video output APIs, some
audio output APIs).

Create a header osdep/endian.h, which contains various fallbacks.
Note that the last fallback uses libavutil; however, it's not clear
whether AV_HAVE_BIGENDIAN is a public symbol, or whether including
<libavutil/bswap.h> really makes it visible. And in fact we don't want
to pollute the namespace with libavutil definitions either. Thus it's
only the last fallback.
2014-07-10 00:58:56 +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
fdd5d00be3 audio: fix signedness of AF_FORMAT_S32P
This was marked as unsigned, but it's signed. Found by xylosper.
2014-02-05 18:53:00 +01:00
wm4
2c08bf1bd7 Reduce recursive config.h inclusions in headers
In my opinion, config.h inclusions should be kept to a minimum. MPlayer
code really liked including config.h everywhere, though, even in often
used header files. Try to reduce this.
2013-12-18 17:12:21 +01:00
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
0ed0f4d33a audio/format: fix doublep sample format
This was accidentally equivalent to floatp.
2013-11-16 21:46:16 +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
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
406241005e core: move contents to mpvcore (2/2)
Followup commit. Fixes all the files references.
2013-08-06 22:52:31 +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
c5ee7740c4 ao_portaudio: use mp_ring 2013-06-16 18:20:39 +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
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
reimar
a4177fd581 audio: make AC3 pass-through with ad_spdif work
Do not fall back to 0 for samplerate when parser is not initialized.

Might fix some issues with using -ac spdifenc with audio in MKV
or MP4.

git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35517 b3059339-0415-0410-9bf9-f77b7e298cf2

Replace outdated list of unsupported formats by list of supported formats.

git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35534 b3059339-0415-0410-9bf9-f77b7e298cf2

Do not call af_fmt2str on the same data over and over.

git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35535 b3059339-0415-0410-9bf9-f77b7e298cf2

ad_spdif: use the more specific AF_FORMAT_AC3_LE when
we handle AC3.

git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35536 b3059339-0415-0410-9bf9-f77b7e298cf2

Make AF_FORMAT_IS_IEC61937 include AF_FORMAT_IS_AC3.

Our AC3 "sample format" is also iec61937.

git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35537 b3059339-0415-0410-9bf9-f77b7e298cf2

af_format: support endianness conversion also for iec61937
formats in general, not just AC3.

git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35538 b3059339-0415-0410-9bf9-f77b7e298cf2

Conflicts:
	audio/filter/af_format.c

af_format: Fix check_format, non-special formats are of course supported.

git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35545 b3059339-0415-0410-9bf9-f77b7e298cf2

Note: see mplayer bug #2110
2012-12-03 21:08:52 +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