Based on patch by Yuriy Kaminskiy [yumkam gmail].
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@37330 b3059339-0415-0410-9bf9-f77b7e298cf2
Signed-off-by: wm4 <wm4@nowhere>
From what I understand the division is to align the dimension of the
value from seconds to milliseconds. Hard to tell whether the "rounding"
was intentional or not; I'm tipping on "not".
Found by Coverity.
This rewrites the audio decode loop to some degree. Audio filters don't
do refcounted frames yet, so af.c contains a hacky "emulation".
Remove some of the weird heuristic-heavy code in dec_audio.c. Instead of
estimating how much audio we need to filter, we always filter full
frames. Maybe this should be adjusted later: in case filtering increases
the volume of the audio data, we should try not to buffer too much
filter output by reducing the input that is fed at once.
For ad_spdif.c and ad_mpg123.c, we don't avoid extra copying yet - it
doesn't seem worth the trouble.
Use a pseudo-filter when changing speed with resampling, instead of
somehow changing a samplerate somewhere. This uses the same underlying
mechanism, but is a bit more structured and cleaner. It also makes some
of the following changes easier.
Since we now always use filters to change audio speed, move most of the
work set_playback_speed() does to recreate_audio_filters().
Although the "af" command already could do this, it seems it's better
to introduce a lower level mechanism for now. This avoids some messy
issues, since that code would recursive call reinit_audio_chain().
To be used by the next commit.
There's no real reason why audio_init_filter() should exist. Just use
af_init or af_reinit directly. (We lose a useless message; the same
information is printed in a quite close place with more details.)
Requires less code, and the way the filter chain is marked as having
failed to initialize allows just switching off audio instead of
crashing if trying to insert a volume filter in mixer.c fails, and
recreating the old filter chain fails too.
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.
Sometimes, --af=hrtf produces heavy artifacts or silence. It's possible
that this commit fixes these issues. My theory is that usually, the
uninitialized coefficients quickly converge to sane values as more audio
is filtered, which would explain why there are often artifacts on init,
with normal playback after that. It's also possible that sometimes, the
uninitialized values were NaN or inf, so that the artifacts (or silence)
would never go away.
Fix this by initializing the coefficients to 0. I'm not sure if this is
correct, but certainly better than before.
See issue #1104.
The filter output size can be 0. Due to how filtering works, this is
nothing unusual, but avresample_convert() will return 0. The same case
is already handling with "normal" resampling (this commit fixes the
reordering code).
Additionally, don't use an assert(). avresample_convert() failing is
unusual, but might also happen due to e.g. internal out of memory
conditions, so we shouldn't just crash on it.
Curiously observed with --ao=oss --audio-channels=5.1 when changing
speed.
It seems hrtf works in 48khz only - and if that wasn't the input, the
filter just exited with an error. Make it request the 48khz instead. The
player will insert a resampling filter.
Not sure why it wasn't done like this in the first place.
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.
Use OPT_KEYVALUELIST() for all places where AVOptions are directly set
from mpv command line options. This allows escaping values, better
diagnostics (also no more "pal"), and somehow reduces code size.
Remove the old crappy option parser (av_opts.c).
The final goal is all mp_msg calls produce complete lines. We want this
because otherwise, race conditions could corrupt the terminal output,
and it's inconvenient for the client API too. This commit works towards
this goal. There's still code that has this not fixed yet, though.
rgain is not an additive value. It's a multiplier/gain.
Previous behaviour produced negative level values in some cases
(when rgain < 1.0) which caused volume to be louder when its value
was lowered.
CC: @mpv-player/stable
Signed-off-by: Mohammad Alsaleh <CE.Mohammad.AlSaleh@gmail.com>
Signed-off-by: wm4 <wm4@nowhere>
While I'm not very fond of "const", it's important for declarations
(it decides whether a symbol is emitted in a read-only or read/write
section). Fix all these cases, so we have writeable global data only
when we really need.
I hate tabs.
This replaces all tabs in all source files with spaces. The only
exception is old-makefile. The replacement was made by running the
GNU coreutils "expand" command on every file. Since the replacement was
automatic, it's possible that some formatting was destroyed (but perhaps
only if it was assuming that the end of a tab does not correspond to
aligning the end to multiples of 8 spaces).
This was accidentally broken in commit b72ba3f7. I somehow made the
wild assumption that replaygain adjusted the volume relative to 0%
instead of 100%.
The detach suboption was similarly broken.
Maybe this should be default. On the other hand, this filter does
something even if the volume is neutral: it clips samples against the
allowed range, should the decoder or a previous filter output garbage.
Currently, both replaygain adjustment and user volume control (if
softvol is enabled) share the same variable. Sharing the variable would
cause especially if --volume is used; then the replaygain volume would
always be overwritten.
Now both gain values are simple added right before doing filtering.
This adds the options replaygain-track and replaygain-album. If either is set,
the replaygain track or album gain will be automatically read from the track
metadata and the volume adjusted accordingly.
This only supports reading REPLAYGAIN_(TRACK|ALBUM)_GAIN tags. Other formats
like LAME's info header would probably require support from libav.
The main incompatibility was that Libav didn't have av_opt_set_int_list.
But since that function is excessively ugly and idiotic (look how it
handles types), I'm not missing it much. Use an aformat filter instead
to handle the functionality that was indirectly provided by it. This is
similar to how vf_lavfi works.
The other incompatibility was channel handling. Libav consistently uses
channel layouts only, why ffmpeg still requires messing with channel
counts to some degree. Get rid of most channel count uses (and hope
channel layouts are "exact" enough). Only in one case FFmpeg fails with
a runtime check if we feed it AVFrames with channel count unset.
Another issue were AVFrame accessor functions. FFmpeg introduced these
for ABI compatibility with Libav. I refuse to use them, and it's not my
problem if FFmpeg doesn't manage to provide a stable ABI for fields
provided both by FFmpeg and Libav.
