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.
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.)
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".
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.)
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.
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.
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).
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.
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.
bstr.c doesn't really deserve its own directory, and compat had just
a few files, most of which may as well be in osdep. There isn't really
any justification for these extra directories, so get rid of them.
The compat/libav.h was empty - just delete it. We changed our approach
to API compatibility, and will likely not need it anymore.
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.
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.
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.
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);
}
}
}
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.
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.
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.
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.
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
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.)
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.
