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.
audio filtering the default.
This mostly means lavcresample being the default instead of plain "resample".
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@30455 b3059339-0415-0410-9bf9-f77b7e298cf2
Where 8 channel support is non-trivial (e.g. ao_dsound), at least ensure we
fail gracefully.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@29868 b3059339-0415-0410-9bf9-f77b7e298cf2
The libdvdread4 and libdvdnav directories, which are externals in the
svn repository, are at least for now not included in any form. I added
configure checks to automatically disable internal libdvdread and
libdvdnav if the corresponding directories are not present; if they're
added manually then things work the same as in svn.
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
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
- Improved runtime control system
- 3 New filter panning, compressor/limiter and a noise gate
- The compressor/limiter and the noise gate are not yet finished
- The panning filter does combined mixing and channel routing and
can be used to down-mix from stereo to mono (for example)
- Improvements to volume and channel
- volume now has a very good soft clipping using sin()
- channel can handle generic routing of audio data
- Conversion of all filters to handle floating point data
- Cleanup of message printing
- Fix for the sig 11 bug reported by Denes
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@8608 b3059339-0415-0410-9bf9-f77b7e298cf2
-- Support for runtime cpu detection
-- Stand alone compile of libaf
-- Unlimited number of channels (compiletime switch)
-- Sample format defined by bit-fields
-- New formats: float, A-Law and mu-law
-- Format conversion set in human readable format
i.e. format=4:us_be to set 32 bit unsigned big endian output
-- Format reporting in human readable format
-- Volume control has only one parameter for setting the volume
i.e. volume=-10.0:1:0:1 to set atenuation = -10dB
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@8168 b3059339-0415-0410-9bf9-f77b7e298cf2
