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
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
simplify resampling factor if possible, so more then one resampler can be used, libaf will still die if there are too many like it does with the default resampler (2 with sampling rates which are relative prime are too many ...)
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@13731 b3059339-0415-0410-9bf9-f77b7e298cf2
libaf doesnt seem to support planar audio, so we need to convert it :(
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@13714 b3059339-0415-0410-9bf9-f77b7e298cf2
