Codec selection for audio and video decoding had a "dynamic plugin"
feature that tried to load a shared library for any codec that had not
been enabled at compilation (disabled by default, but could be enabled
with --enable-dynamic-plugins configure switch; for unknown reasons
some distro packages have enabled it). The implementation was buggy
and could cause normal codec selection fallback to fail if the feature
was enabled. I'm not aware of any real uses of such dynamic plugins
and the feature seems questionable anyway (there are no ABI guarantees
that would make it safe to use). Remove the buggy feature.
Move the buffer storing audio data ready to be fed to the audio output
driver from the audio decoder object to the AO object. This will help
encoding code deal with end of input, and may also be useful to
improve other general gapless audio behavior (as AOs which do not
accept chunks smaller than a certain size may keep them in the buffer
while the decoder changes).
Less data may be dropped now when changing audio filters or switching
timeline parts.
dec_audio.c init_audio_codec() would in one case print
"ADecoder init failed :(\n" and return failure. Its only caller
init_best_audio_codec() printed exactly the same message if the
returned result was failure. Change the latter message to say
"Could not open audio decoder %s.\n" instead. Some of the
per-open-attempt messages are kind of value about their context; this
new message should make it more clear where the attempt to open one
specific codec ends.
Add code to enforce matching pts with video when (re)starting the
audio stream, by either cutting away the first samples or inserting
silence at the beginning. New option -noinitial-audio-sync can be used
to disable this and return to old behavior.
Add support for parameter changes (e.g. channel count) during playback.
This makes decoding AC3 files that switch between 2 and 6 channels
work reasonably well even with -channels 6 and ffac3 decoder.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@31737 b3059339-0415-0410-9bf9-f77b7e298cf2
Fix typo in error message: ACC -> AAC
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32473 b3059339-0415-0410-9bf9-f77b7e298cf2
Avoid printing AAC with SBR warning on every decode call, instead print
it only after every decoder reconfiguration.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32476 b3059339-0415-0410-9bf9-f77b7e298cf2
The ad_functions structs are in rodata, mark some pointers to them
const.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@31606 b3059339-0415-0410-9bf9-f77b7e298cf2
Note that r30455 is wrong, that commit does not in fact change the
default behavior as claimed in the commit message. It only breaks
"-af-adv force=0", which was already pretty much useless though.
scattered all over the place with half of it forgotten in some places.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@30420 b3059339-0415-0410-9bf9-f77b7e298cf2
A couple of months ago MPlayer's ALSA driver started rounding the
amount of input data it was willing to accept in one call down to an
integer multiple of the value it set in ao_data.outburst. In some
configurations it was possible for this value to exceed the 64 KiB
limit on the amount MPlayer was willing to write in a single call to
the AO. As a result ao_alsa accepted 0 bytes in each play() call and
audio playback failed. Fix this by removing the fixed 64 KiB limit on
the amount of audio sent to AO at once; the limit was mostly a remnant
of older code anyway.
Revert 3 old code uglification changes that were done with the excuse
of gcc-2.95 support. The last reverted change was a fix to a bug
introduced in the middle change.
Revert "10l, len may change after initialization time"
This reverts commit ae9db277c7.
Revert "fix declaration after statement, take 2"
This reverts commit 4bceedee93.
Revert "fix declaration after statement"
This reverts commit aef0374c1c.
Replace all USE_ prefixes by CONFIG_ prefixes to indicate
options which are configurable.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@27373 b3059339-0415-0410-9bf9-f77b7e298cf2
Seems to be enough to avoid crashes (due to unaligned SSE2) with FFmpeg vorbis decoding for now.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@27281 b3059339-0415-0410-9bf9-f77b7e298cf2
an almost-trivial implementation.
This allows making the builtin codec structs const, and it also makes
clearer that this "selected" status is not used outside the init functions.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@25689 b3059339-0415-0410-9bf9-f77b7e298cf2
the most likely result is a NULL dereference which isn't much worse.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@24946 b3059339-0415-0410-9bf9-f77b7e298cf2
Remove the following arguments as redundant: in_channels, in_format,
out_minsize, out_maxsize. The first two always equal fields of the
sh_audio_t struct given as the first argument to the function. The
last two are unused after the allocation of sh_audio->a_out_buffer
was changed to be done on demand.
After the out_minsize and out_maxsize arguments are removed the
function preinit_audio_filters() is identical to init_audio_filters(),
so remove it and use the latter instead.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@24922 b3059339-0415-0410-9bf9-f77b7e298cf2
Remove the code allocating sh_audio->a_out_buffer from
init_audio_filters() and let the buffer be allocated by the new
dynamic allocation code.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@24921 b3059339-0415-0410-9bf9-f77b7e298cf2
Rewrite decode_audio to better deal with filters that handle input in
large blocks. It now always places output in sh_audio->a_out_buffer
(which was always given as a parameter before) and reallocates the
buffer if needed. After the changes filters can return arbitrarily
large blocks of data without some of it being lost. The new version
also allows simplifying some code.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@24920 b3059339-0415-0410-9bf9-f77b7e298cf2
Remove code that set sh_audio->a_out_buffer to equal
sh_audio->a_buffer between the calls to init_best_audio_codec and
init_audio_filters. Nothing uses the buffer between those calls.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@24912 b3059339-0415-0410-9bf9-f77b7e298cf2
against instead of directly #including the C file and replace the many extern
declarations by a proper header file.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@24262 b3059339-0415-0410-9bf9-f77b7e298cf2
It incorrectly used the channel count and sample size values from the
decoder even though the filters can change those.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@20768 b3059339-0415-0410-9bf9-f77b7e298cf2
