audio/filter: remove some useless filters

All of these filters are considered not useful anymore by us. Some have
replacements in libavfilter (useable through af_lavfi).

af_center, af_extrastereo, af_karaoke, af_sinesuppress, af_sub,
af_surround, af_sweep: pretty simple and useless filters which probably
nobody ever wants.

af_ladspa: has a replacement in libavfilter.

af_hrtf: the algorithm doesn't work properly on most sources, and the
implementation was buggy and complicated. (The filter was inherited from
MPlayer; but even in mpv times we had to apply fixes that fixed major
issues with added noise.) There is a ladspa filter if you still want to
use it.

af_export: I'm not even sure what this is supposed to do. Possibly it
was meant for GUIs rendering audio visualizations, but it couldn't
really work well. For example, the size of the audio depended on the
samplerate (fixed number of samples only), and it couldn't retrieve the
complete audio, only fragments. If this is really needed for GUIs, mpv
should add native visualization, or a proper API for it.

1 files changed, 0 insertions, 74 deletions

diff --git a/audio/filter/filter.h b/audio/filter/filter.h
deleted file mode 100644
index 8a1864a940..0000000000
--- a/audio/filter/filter.h
+++ /dev/null
@@ -1,74 +0,0 @@
-/*
- * Copyright (C) 2001 Anders Johansson ajh@atri.curtin.edu.au
- *
- * This file is part of mpv.
- *
- * mpv is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
- * (at your option) any later version.
- *
- * mpv is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
- * GNU General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License along
- * with mpv.  If not, see <http://www.gnu.org/licenses/>.
- */
-
-#if !defined MPLAYER_DSP_H
-# error Never use filter.h directly; include dsp.h instead.
-#endif
-
-#ifndef MPLAYER_FILTER_H
-#define MPLAYER_FILTER_H
-
-
-// Design and implementation of different types of digital filters
-
-
-// Flags used for filter design
-
-// Filter characteristics
-#define LP          0x00010000 // Low pass
-#define HP          0x00020000 // High pass
-#define BP          0x00040000 // Band pass
-#define BS          0x00080000 // Band stop
-#define TYPE_MASK   0x000F0000
-
-// Window types
-#define BOXCAR      0x00000001
-#define TRIANG      0x00000002
-#define HAMMING     0x00000004
-#define HANNING     0x00000008
-#define BLACKMAN    0x00000010
-#define FLATTOP     0x00000011
-#define KAISER      0x00000012
-#define WINDOW_MASK 0x0000001F
-
-// Parallel filter design
-#define FWD         0x00000001 // Forward indexing of polyphase filter
-#define REW         0x00000002 // Reverse indexing of polyphase filter
-#define ODD         0x00000010 // Make filter HP
-
-// Exported functions
-FLOAT_TYPE af_filter_fir(unsigned int n, const FLOAT_TYPE* w, const FLOAT_TYPE* x);
-
-int af_filter_design_fir(unsigned int n, FLOAT_TYPE* w, const FLOAT_TYPE* fc,
-                         unsigned int flags, FLOAT_TYPE opt);
-
-int af_filter_szxform(const FLOAT_TYPE* a, const FLOAT_TYPE* b, FLOAT_TYPE Q,
-                      FLOAT_TYPE fc, FLOAT_TYPE fs, FLOAT_TYPE *k,
-                      FLOAT_TYPE *coef);
-
-/* Add new data to circular queue designed to be used with a FIR
-   filter. xq is the circular queue, in pointing at the new sample, xi
-   current index for xq and n the length of the filter. xq must be n*2
-   long.
-*/
-#define af_filter_updateq(n,xi,xq,in)\
-  xq[xi]=(xq)[(xi)+(n)]=*(in);\
-  xi=(++(xi))&((n)-1);
-
-#endif /* MPLAYER_FILTER_H */