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.
Balance controls as used by mixer.c was broken, because af_pan.c stopped
accepting its arguments. We have to allow 0 channels explicitly. Also,
fix null pointer access if the matrix parameter is not used.
Regression from commit 82983970.
Signed-off-by: wm4 <wm4@nowhere>
This merges pull request #496. The problem was that at least the
initialization of the distance[] array accessed af_fmtstr_table[]
entries that were out of bounds. Small cosmetic changes applied to
the original pull request.
Always pass around mp_log contexts in the option parser code. This of
course affects all users of this API as well.
In stream.c, pass a mp_null_log, because we can't do it properly yet.
This will be fixed later.
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.
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.
The tmsg stuff was for the internal gettext() based translation system,
which nobody ever attempted to use and thus was removed. mp_gtext() and
set_osd_tmsg() were also for this.
mp_dbg was once enabled in debug mode only, but since we have log level
for enabling debug messages, it seems utterly useless.
This can be reproduced with:
mpv short.wav -af 'lavfi="aecho=0.8:0.9:5000|6800:0.3|0.25"'
An audio file that is just 1-2 seconds long should play for 8-9 seconds,
which audible echo towards the end.
The code assumes that when playing with AF_FILTER_FLAG_EOF, the filter
will either produce output, or has all remaining data flushed. I'm not
really sure whether this really works if there are multiple filters with
EOF handling in the chain. To handle it correctly, af_lavfi should retry
filtering if 1. EOF flag is set, 2. there were input samples, and 3. no
output samples were produced. But currently it seems to work well enough
anyway.
The new signature is actually closer to how it actually works, and
someone who is not familiar to the API and how it works might make fewer
fatal mistakes with the new signature than the old one. Pretty weird.
Do this to sneak in a flags parameter, which will later be used to flush
remaining data of at least vf_lavfi.
This will make af_channels output a channel layout that is compatible
with any destination layout. Not sure if that's a good idea though,
since the way the AO choses a layout is perhaps less predictable. On the
other hand, using the old MPlayer standard layouts doesn't make much
sense either. We'll see whether this improves or breaks someone's use
case.
Apparently this stopped working after some planar changes (broken format
negotiation). Radically change option parsing in an incompatible way.
Suggest alternatives to this filter, since it barely has any importance
anymore.
There are some use cases for this. For example, you can use it to set
defaults of automatically inserted filters (like af_lavrresample). It's
also useful if you have a non-trivial VO configuration, and want to use
--vo to quickly change between the drivers without repeating the whole
configuration in the --vo argument.
This is needed so that new processes (created with fork+exec) don't
inherit open files, which can be important for a number of reasons.
Since O_CLOEXEC is relatively new (POSIX.1-2008, before that Linux
specific), we #define it to 0 in io.h to prevent compilation errors on
older/crappy systems. At least this is the plan.
input.c creates a pipe. For that, add a mp_set_cloexec() function (which
is based on Weston's code in vo_wayland.c, but more correct). We could
use pipe2() instead, but that is Linux specific. Technically, we have a
race condition, but it won't matter.
The AF control commands used an elaborate and unnecessary organization
for the command constants. Get rid of all that and convert the
definitions to a simple enum. Also remove the control commands that
were not really needed, because they were not used outside of the
filters that implemented them.
And by "cleanup", I mean "remove". Actually, only remove the parts that
are redundant and doxygen noise. Move useful parts to the comment above
the function's implementation in the C source file.
Commit 22b3f522 not only redid major aspects of audio decoding, but also
attempted to fix audio format change handling. Before that commit, data
that was already decoded but not yet filtered was thrown away on a
format change. After that commit, data was supposed to finish playing
before rebuilding filters and so on.
It was still buggy, though: the decoder buffer was initialized to the
new format too early, triggering an assertion failure. Move the reinit
call below filtering to fix this.
ad_mpg123.c needs to be adjusted so that it doesn't decode new data
before the format change is actually executed.
Add some more assertions to af_play() (audio filtering) to make sure
input data and configured format don't mismatch. This will also catch
filters which don't set the format on their output data correctly.
Regression due to planar_audio branch.
This changes option parsing as well as filter defaults slightly. The
default is now to encode to spdif (this is way more useful than writing
raw AC3 - what was this even useful for, other than writing broken ac3
-in-wav files?). The bitrate parameter is now always in kbps.
Remove the awkward planarization. It had to be done because the AC3
encoder requires planar formats, but now we support them natively.
Try to simplify buffer management with mp_audio_buffer.
Improve checking for buffer overflows and out of bound writes. In
theory, these shouldn't happen due to AC3 fixed frame sizes, but being
paranoid is better.
The format negotiation is the same, except don't confusingly copy the
input format into af->data, just to overwrite it later. af->data should
alwass contain the output format, and the existing code was just a very
misguided use of the af_test_output() helper function.
Just set af->data to the output format immediately, and modify the input
format properly.
Also, if format negotiation fails (and needs another iteration), don't
initialize the libavcodec encoder.
Before this commit, the af_instance->mul/delay values were in bytes.
Using bytes is confusing for non-interleaved audio, so switch mul to
samples, and delay to seconds. For delay, seconds are more intuitive
than bytes or samples, because it's used for the latency calculation.
We also might want to replace the delay mechanism with real PTS
tracking inside the filter chain some time in the future, and PTS
will also require time-adjustments to be done in seconds.
For most filters, we just remove the redundant mul=1 initialization.
(Setting this used to be required, but not anymore.)
Allocate af_instance->data in generic code before filter initialization.
Every filter needs af->data (since it contains the output
configuration), so there's no reason why every filter should allocate
and free it.
Remove RESIZE_LOCAL_BUFFER(), and replace it with mp_audio_realloc_min().
Interestingly, most code becomes simpler, because the new function takes
the size in samples, and not in bytes. There are larger change in
af_scaletempo.c and af_lavcac3enc.c, because these had copied and
modified versions of the RESIZE_LOCAL_BUFFER macro/function.
Based on earlier work by Stefano Pigozzi.
There are 2 changes:
1. Instead of mp_audio.audio, mp_audio.planes[0] must be used.
2. mp_audio.len used to contain the size of the audio in bytes. Now
mp_audio.samples must be used. (Where 1 sample is the smallest unit
of audio that covers all channels.)
Also, some filters need changes to reject non-interleaved formats
properly.
Nothing uses the non-interleaved features yet, but this is needed so
that things don't just break when doing so.
My main problem with this is that the output format will be incorrect.
(This doesn't matter right, because there are no samples output.)
This assumes all audio filters can deal with len==0 passed in for
filtering (though I wouldn't see why not).
A filter can still signal an error by returning NULL.
af_lavrresample has to be fixed, since resampling 0 samples makes
libavresample fail and return a negative error code. (Even though it's
not documented to return an error code!)
In theory, af_volume could use separate volume levels for each channel.
But this was never used anywhere.
MPlayer implemented something similar before (svn r36498), but kept the
old path for some reason.
Also do some cosmetic changes, like merging definition and
initialization of local variables.
Remove an annoying debug mp_msg() from af_open(). It just printed the
command line parameters; if this is really needed, it could be added
to af.c instead (similar as to what vf.c does).
Helps with readability. Also remove the ctx_opt_set_* helper macros and
use av_opt_set_* directly (I think these macros were used because the
lines ended up too long, but this commit removes two indentation levels,
giving more space).
This should allow to make format negotiation much simpler, since it
takes the responsibility to compare actual input and accepted input
formats from the filters. It's also backwards compatible. Filters which
have expensive initialization still can use the old method.
It's probably better if all auto-inserted filters are removed when doing
an af_add operation. If they're really needed, they will be
automatically re-added.
Fix the error message. It used to be for an actual internal error, but
now it happens when format negotiation fails, e.g. when trying to use
spdif and real audio filters.
The configure followed 5 different convetions of defines because the next guy
always wanted to introduce a new better way to uniform it[1]. For an
hypothetic feature 'hurr' you could have had:
* #define HAVE_HURR 1 / #undef HAVE_DURR
* #define HAVE_HURR / #undef HAVE_DURR
* #define CONFIG_HURR 1 / #undef CONFIG_DURR
* #define HAVE_HURR 1 / #define HAVE_DURR 0
* #define CONFIG_HURR 1 / #define CONFIG_DURR 0
All is now uniform and uses:
* #define HAVE_HURR 1
* #define HAVE_DURR 0
We like definining to 0 as opposed to `undef` bcause it can help spot typos
and is very helpful when doing big reorganizations in the code.
[1]: http://xkcd.com/927/ related
Defining names like min, max etc. in an often used header is not really
a good idea.
Somewhat similar to MPlayer svn commit 36491, but don't use libavutil,
because that typically causes us sorrow.
Roughly follows MPlayer svn commits 36492 and 36493. We also remove
the volume peak reporting. (There are much better libavfilter filters
for this, I think.)
Drop the author and comment fields. They were completely unused - not
even printed in verbose mode, just dead weight.
Also use designated initializers and drop redundant flags.
Set the input/output format in filter init. This doesn't change anything
functionally, but it makes the forced format show up in the filter chain
init verbose output (which sometimes prints the filter chain before all
filters have been configured).
af_format is the old audio conversion filter. It could do all possible
conversions supported by the audio chain. However, ever since the
addition of af_lavrresample, most conversions are done by
libav/swresample, and af_format is used as fallback.
Separate out the fallback cases and remove af_format. af_convert24 does
24 bit <-> 32 bit conversions, while af_convertsignendian does sign and
endian conversions. Maybe the way the conversions are split sounds a bit
odd. But the former changes the size of the audio data, while the latter
is fully in-place, so there's at least different buffer management.
This requires a quite complicated algorithm to make sure all these
"partial" conversion filters can actually get from one format to
another. E.g. s24le->s32be always requires convertsignendian and
convert24, but af.c has no idea what the intermediate format should
be. So I added a graph search (trying every possible format and
filter) to determine required format and filter. When I wrote this,
it seemed this was still better than messing everything into
af_lavrresample, but maybe this is overkill and I'll change my
opinion. For now, it seems nice to get rid of af_format though.
The AC3->IEC61937 conversion isn't supported anymore, but I don't think
this is needed anywhere. Most AOs test all formats explicitly, or use
the AF_FORMAT_IS_IEC61937() macro (which includes AC3).
One positive consequence of this change is that conversions always
include dithering (done by libav/swresample), instead of possibly going
through af_format, which doesn't do anything fancy.
Rename af_force to af_format. It's essentially compatible with command
line uses of af_format. We retain a compatibility alias for af_force.
Calling them separately doesn't really make sense, and all existing
calls to them usually combined them. One subtitle difference was that
af_init() didn't wipe the filter chain if initialization of the chain
itself failed, but that didn't really make sense anyway.
Also remove af_init() from the code for setting balance in mixer.c. The
mixer should be in the initialized state only if audio is fully
initialized, so the af_init() call made no sense.
Note that the filter "editing" code in command.c doesn't really do a
nice job of handling errors in case recreating an _old_ (known to work)
filter chain unexpectedly fails, and this obscure/rare case might be
differently handled after this change.
Softvol always used a linear multiplier for volume control. This was
converted to dB, and then back to linear in af_volume. Remove this non-
sense. We still try to keep the command line argument to af_volume in
dB, though.
It's quite unlikely, but functions like mp_find_user_config_file() can
return NULL, e.g. if $HOME is unset.
Fix all the code that didn't check for this correctly yet.
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.
The --speed option and the speed property used float. Change them to
double.
Change the commands that manipulate the property (speed_mult/add) to
double as well. Since the cycle command shares code with the add
command, we change that as well.
The reason for this change is that this allows better control over
speed, such as stepping by semitones. Using floats is also just plain
unnecessary.
In general, this warning can hint to actual bugs. We don't enable it
yet, because it would conflict with some unmerged code, and we should
check with clang too (this commit was done by testing with gcc).
Make the VF/VO/AO option parser available to audio filters. No audio
filter uses this yet, but it's still a quite intrusive change.
In particular, the commands for manipulating filters at runtime
completely change. We delete the old code, and use the same
infrastructure as for video filters. (This forces complete
reinitialization of the filter chain, which hopefully isn't a problem
for any use cases. The old code forced reinitialization too, but it
could potentially allow a filter to cache things; e.g. consider loaded
ladspa plugins and such.)
This code is supposed to run if dynamic filter insertion (such as when
inserting a volume filter in mixer.c) fails. Then it removes all filters
and recreates the default list of filters. But the code just blew up and
entered an endless loop, because it removed even the sentinel in/out
filters. This could happen when trying to use softvol controls while
using spdif, but also other situations. Fix it by calling the correct
code.
Also remove these obnoxious yoda-conditions.
Mostly copied from vf_lavfi. The parts that could be shared are minor,
because most code is about setting up audio and video, which are too
different.
This won't work with Libav. I used ffplay.c as guide, and noticed too
late that their setup methods are incompatible with Libav's. Trying to
make it work with both would be too much effort. The configure test for
av_opt_set_int_list() should disable af_lavfi gracefully when compiling
with Libav.
Due to option parser chaos, you currently can't have a "," as part of
the filter graph string - not even with quoting or escaping. This will
probably be fixed later.
The audio filter chain is not PTS aware. So we have to do some hacks
to make up a fake PTS, and we have to map the output PTS back to the
filter chain's method of tracking PTS changes and buffering, by
adjusting af->delay.
The libavresample version of the current Libav stable release lacks the
avresample_set_channel_mapping() function. (FFmpeg's libswresample seems
to be fine, because they added swr_set_channel_mapping() first.)
Add a cheap/slow workaround to do channel reordering on our own. We
don't use the recently removed MPlayer code (see commit 586b75a),
because that is not generic enough.
The functionality should be the same as with full-featured
libavresample, and any differences are bugs. It's probably slower,
though.
af_reinit() is responsible for inserting automatic conversion filters
for channel remixing, format conversion, and resampling. We don't
require that a single filter can do all these (even though
af_lavrresample does nearly all of this, sometimes af_format has to be
used instead for format conversions). This makes setting up the chain
more complicated, and a way is needed to prevent endless appending of
conversion filters if a conversion is not possible.
Until now, this used a stupidly simple yet robust static retry limit to
detect failure. This is perfectly fine, and the limit (20) was good
enough to handle about ~5 filters. But with more filters, and if each
filter requires 3 additional conversion filters, this would fail. So
raise the limit to 4 retries per filter. This is still stupidly simple
and robust, but won't arbitrarily fail if the filter count is too large.
If one of the input or output is an unknown layout, but the other is
known, it can still happen that channels are remixed randomly. Avoid
this by forcing default layouts in this case. (Doesn't work if the
channel counts are different.)
This is done in af_lavrresample now, and as part of format negotiation.
Also remove the remaining reorder_channel calls. They were redundant
and did nothing.
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.
Add dummy input and output filters to remove special cases in the format
negotiation code (af_fix_format_conversion() etc.). The output of the
filter chain is now negotiated in exactly the same way as normal
filters.
Negotiate setting the sample rate in the same way as other audio
parameters. As a side effect, the resampler is inserted at the start of
the filter chain instead of the end, but that shouldn't matter much,
especially since conversion and channel mixing are conflated into the
same filter (due to libavresample's API).
Anything this option did has been removed in the preceding 3 commits.
Note that even though these options sounded like a good idea (like
setting accuracy vs. speed tradeoffs), they were not really properly
implemented.
All this option did was deciding whether the resample filter was to be
insert at the beginning or end of the filter chain. Always do what the
option set for accuracy did. I doubt it makes much of a difference.
libavresample does most things in just one go anyway, so it won't
matter.
Dangerous and misleading. If it turns out that this is actually needed
to make certain setups work right, it should be added back in a better
way (in a way it doesn't cause random crashes).
The only thing this option did was changing the behavior of af_volume.
The option decided what sample format af_volume would use, but only if
the sample format was not already float. If the option was set, it would
default to float, otherwise to S16.
Remove use of the option and all associated code, and make af_volume
always use float (unless a af_volume specific sub-option is set).
Silence maximum value tracking. This message is printed when the filter
is destroyed, and it's slightly annoying. Was enabled due to enabling
float by default.
Switch the internal channel order to libavcodec's. If the channel number
mismatches at some point, use libavresample for up- or downmixing.
Remove the old af_pan automatic downmixing.
The libavcodec channel order should be equivalent to WAVEFORMATEX order,
at least nowadays. reorder_ch.h assumes that WAVEFORMATEX and libavcodec
might be different, but all defined channels have the same mappings.
Remove the downmixing with af_pan as well as the channel conversion with
af_channels from af.c, and prefer af_lavrresample for this. The
automatic downmixing behavior should be the same as before (if the
--channels option is set to 2, which is the default, the audio output
is forced to 2 channels, and libavresample does all downmixing).
Note that mpv still can't do channel layouts. It will pick the default
channel layout according to the channel count. This will be fixed later
by passing down the channel layout as well.
af_hrtf depends on the order of the input channels, so reorder to ALSA
(for which this code was written). This is better than changing the
filter code, which is more risky.
ao_pulse can accept waveext order directly, so set that as channel
mapping.
If format negotiation fails, and additional filters are inserted to fix
this, don't try to reinitialize the filter immediately. Instead, correct
the audio format, and let the caller retry.
Add a retry counter to af_reinit() to ensure that misbehaving filters
can't put the format negotiation into an endless loop.
Refactor to remove the duplicated format filter insertion code. Allow
other format converting filters to be inserted on format mismatches.
af_info.test_conversion checks whether conversion between two formats
would work with the given filter; do this to avoid having to insert
multiple conversion filters at once and such things. (Although this
isn't ideal: what if we want to avoid af_format for some conversions?
What if we want to split af_format in endian-swapping filters etc.?)
Prefer af_lavrresample for conversions that it supports natively,
otherwise let af_format handle the full conversion.
Make sure automatically inserted filters are removed on full reinit
(they are re-added later if they are really needed). Automatically
inserted filters were never explicitly removed, instead, it was
expected that redundant conversion filters detach themselves. This
didn't work if there were several chained format conversion filters,
e.g. s16le->floatle->s16le, which could result from repeated filter
insertion and removal. (format filters detach only if input format and
output format are the same.)
Further, the dummy filter (which exists only because af.c can't handle
an empty filter chain for some reason) could introduce bad conversions
due to how the format negotiation works. Change the code so that the
dummy filter never takes part on format negotiation. (It would be better
to fix format negotiation, but that would be much more complicated and
would involving fixing all filters.)
Simplify af_reinit() and remove the start audio filter parameter. This
means format negotiation and filter initialization is run more often,
but should be harmless.
The format was locked to s16. Extend it to accept all other ffmpeg
sample formats, and even allow different in- and output formats. The
generic filter code will still insert af_format on format mismatches,
though.
The change in af_scaletempo actually fixes a memory leak. af->data
contained a pointer to an allocated buffer, which was overwritten
during format negotiation. Set the format explicitly instead.
Remove `af_resample` and `af_lavcresample`. The former is a mess while the
latter uses an API that was long deprecated in libavcodec and is now removed.
`af_lavrresample` rougly has the same features and structure of
`af_lavcresample`.
libswresample fallback by wm4.
avcodec_encode_audio() was deprecated, and was finally removed from
Libav and FFmpeg git.
This appears to work. I get heavy A/V desync with -ao alsa and -ao pcm,
but this was already so before this change.
The previous name of this filter was misleading, because it doesn’t actually
normalize volume levels. What it does is closer to performing low-quality
dynamic range compression, hence it is now called af_drc.
This function sucks and apparently is not very portable (at least on
mingw, the configure check fails). Also remove the emulation of that
function from osdep/strsep*, and remove the configure check.
af_volnorm can process either int16_t or float audio data. The float
version used 0 to INT_MAX as full value range, when it should be 0 to
1. This effectively disabled the filter (due to all input being
considered to fall in the silence range). Fix.
Reported by Tobias Jacobi <liquid.acid@gmx.net>.
Add `mp_find_config_file` to search different known paths and use that in
ass_mp to look for the fontconfig configuration file.
Some incidental changes spawned by this feature where:
* Buffer allocation for the strings containing the paths is now performed
with talloc. All of the allocations are done on a NULL context, but it still
improves readability of the code.
* Move the OSX function for lookup inside of a bundle: this code path was
currently not used by the bundle generated with `make osxbundle`. The plan
is to use it again in a future commit to get a fontconfig config file.
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
This fixes operation with current ffmpeg releases.
Note that this planarization is slow and should be reverted once proper
planar audio support is there in mpv.
Most of these are reimar fixing issues found by Coverity static
analyzer, and possibly some more cleanup commits independent from
this.
Since these commits are rather noisy, squash them all together.
Try to make code a bit clearer.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35294 b3059339-0415-0410-9bf9-f77b7e298cf2
Conflicts:
audio/out/ao_alsa.c
Check the correct variable for NULL.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35323 b3059339-0415-0410-9bf9-f77b7e298cf2
Remove pointless unreachable code (the loop condition already checks
the 0xff case).
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35325 b3059339-0415-0410-9bf9-f77b7e298cf2
Fix typo that might have caused reading beyond the string end.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35326 b3059339-0415-0410-9bf9-f77b7e298cf2
Do not needlessly use "long" types.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35331 b3059339-0415-0410-9bf9-f77b7e298cf2
Use AV_RB32 to avoid sign extension issues and validate offset before using it.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35332 b3059339-0415-0410-9bf9-f77b7e298cf2
Remove nonsense casts.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35343 b3059339-0415-0410-9bf9-f77b7e298cf2
Fix crash in case sh_audio allocation failed.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35348 b3059339-0415-0410-9bf9-f77b7e298cf2
Fix potential NULL dereference.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35351 b3059339-0415-0410-9bf9-f77b7e298cf2
Conflicts:
libmpcodecs/ad_ffmpeg.c
Note: Slightly modified.
Fix malloc failure check to check the correct variable.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35353 b3059339-0415-0410-9bf9-f77b7e298cf2
Avoid code duplication and pointless casts.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35363 b3059339-0415-0410-9bf9-f77b7e298cf2
Conflicts:
stream/tv.c
Error out if an invalid channel list name was specified
instead of continuing and reading outside array bounds
all over the place.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35364 b3059339-0415-0410-9bf9-f77b7e298cf2
Conflicts:
stream/tv.c
Make array "static const".
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35365 b3059339-0415-0410-9bf9-f77b7e298cf2
Properly free resources even when encountering many
parse errors.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35367 b3059339-0415-0410-9bf9-f77b7e298cf2
Conflicts:
parser-cfg.c
Avoid leaks in error handling.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35380 b3059339-0415-0410-9bf9-f77b7e298cf2
Do not do sign comparisons on "char" type which can be both signed or unsigned.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35381 b3059339-0415-0410-9bf9-f77b7e298cf2
Free cookies file data after parsing it.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35382 b3059339-0415-0410-9bf9-f77b7e298cf2
http_set_field only makes a copy of the string, so we still need to
free it.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35383 b3059339-0415-0410-9bf9-f77b7e298cf2
check4proxies does not modify input URL, so mark it const.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35390 b3059339-0415-0410-9bf9-f77b7e298cf2
Remove proxy "support" from stream_rtp and stream_upd, trying
to use a http proxy for UDP connections makes no sense.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35394 b3059339-0415-0410-9bf9-f77b7e298cf2
Conflicts:
stream/stream_rtp.c
stream/stream_udp.c
Add url_new_with_proxy function to reduce code duplication and memleaks.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35395 b3059339-0415-0410-9bf9-f77b7e298cf2
Conflicts:
stream/pnm.c
stream/stream_live555.c
stream/stream_nemesi.c
stream/stream_rtsp.c
Fix off-by-one errors in file descriptor validity checks.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35402 b3059339-0415-0410-9bf9-f77b7e298cf2
Remove pointless cast.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35403 b3059339-0415-0410-9bf9-f77b7e298cf2
Abort when opening the file failed instead of calling
"write" with an invalid descriptor.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35404 b3059339-0415-0410-9bf9-f77b7e298cf2
Remove pointless local variable.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35411 b3059339-0415-0410-9bf9-f77b7e298cf2
Conflicts:
stream/http.c
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.
wm4
Mohammad Alsaleh
Kevin Mitchell
Alessandro Ghedini
11rcombs
Stefano Pigozzi
eng
Martin
Uoti Urpala
reimar
Rudolf Polzer