When -channels 2 [default] is specified and the audio decoder used
does not support internal downmixing, automatically add a pan filter
after the decoder to downmix to stereo.
Patch by Clément Bœsch, ubitux gmail com
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32356 b3059339-0415-0410-9bf9-f77b7e298cf2
Enable all of libavcodec, libavformat, libswscale, and libpostproc
together (libavutil is always required).
based on svn commit by diego:
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32226 b3059339-0415-0410-9bf9-f77b7e298cf2
a separate function.
Call this function also from af_add, fixes audio corruption with e.g.
-softvol -af format=s16be (bug #1561).
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@30659 b3059339-0415-0410-9bf9-f77b7e298cf2
not be used without a declaration, causing issues on 64 bit systems.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@30355 b3059339-0415-0410-9bf9-f77b7e298cf2
Add configure option --ffmpeg-source-dir=PATH. If the user specifies
this option then building code that depends on FFmpeg internals is
enabled and the files files which use internal lavf headers will get
them from this path. The FFmpeg libraries linked with must export
needed internal symbols.
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
And if set first parameter of this filter to 1, it will do ac3 passthrough
like hwac3 did.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@25385 b3059339-0415-0410-9bf9-f77b7e298cf2
Substract the delay caused by filter buffering when calculating
currently playing audio position. This matters for af_scaletempo which
buffers significant and varying amounts of data. For other current
filters the effect is normally insignificant.
Instead of the old time-based filter delay field (which was ignored)
this version stores the per-filter delay in units of bytes input read
without corresponding output. This allows the current scaletempo
behavior where other filters before and after it can see the same
nominal samplerate even though the real duration of the data varies;
in this case the other filters can not know the delay they're causing
in terms of real time.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@24928 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
Most of these functions aren't even used in the same translation unit.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@24918 b3059339-0415-0410-9bf9-f77b7e298cf2
Remove the mul/cancel/gcd functions and some related code. Use ff_gcd
instead of the removed af_gcd in af_resample.c.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@24917 b3059339-0415-0410-9bf9-f77b7e298cf2
Change the audio filters to use a double instead of rationals for the
ratio of output to input size. The rationals could overflow when
calculating the overall ratio of a filter chain and gave no real
advantage compared to doubles.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@24916 b3059339-0415-0410-9bf9-f77b7e298cf2
overflow. Negative values do not seem to be used so just remove the
failing test.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@19889 b3059339-0415-0410-9bf9-f77b7e298cf2
