avfilter: add firequalizer filter

Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Muhammad Faiz <mfcc64@gmail.com>

Changelog

@@ -6,6 +6,7 @@ version <next>:
 - fieldhint filter
 - loop video filter and aloop audio filter
 - Bob Weaver deinterlacing filter
+- firequalizer filter
 
 
 version 3.0:

MAINTAINERS

@@ -353,6 +353,7 @@ Filters:
   af_biquads.c                          Paul B Mahol
   af_chorus.c                           Paul B Mahol
   af_compand.c                          Paul B Mahol
+  af_firequalizer.c                     Muhammad Faiz
   af_ladspa.c                           Paul B Mahol
   af_pan.c                              Nicolas George
   af_sidechaincompress.c                Paul B Mahol

configure

@@ -2861,6 +2861,8 @@ eq_filter_deps="gpl"
 fftfilt_filter_deps="avcodec"
 fftfilt_filter_select="rdft"
 find_rect_filter_deps="avcodec avformat gpl"
+firequalizer_filter_deps="avcodec"
+firequalizer_filter_select="rdft"
 flite_filter_deps="libflite"
 frei0r_filter_deps="frei0r dlopen"
 frei0r_src_filter_deps="frei0r dlopen"

doc/filters.texi

@@ -2366,6 +2366,115 @@ Sets the difference coefficient (default: 2.5). 0.0 means mono sound
 Enable clipping. By default is enabled.
 @end table
 
+@section firequalizer
+Apply FIR Equalization using arbitrary frequency response.
+
+The filter accepts the following option:
+
+@table @option
+@item gain
+Set gain curve equation (in dB). The expression can contain variables:
+@table @option
+@item f
+the evaluated frequency
+@item sr
+sample rate
+@item ch
+channel number, set to 0 when multichannels evaluation is disabled
+@item chid
+channel id, see libavutil/channel_layout.h, set to the first channel id when
+multichannels evaluation is disabled
+@item chs
+number of channels
+@item chlayout
+channel_layout, see libavutil/channel_layout.h
+
+@end table
+and functions:
+@table @option
+@item gain_interpolate(f)
+interpolate gain on frequency f based on gain_entry
+@end table
+This option is also available as command. Default is @code{gain_interpolate(f)}.
+
+@item gain_entry
+Set gain entry for gain_interpolate function. The expression can
+contain functions:
+@table @option
+@item entry(f, g)
+store gain entry at frequency f with value g
+@end table
+This option is also available as command.
+
+@item delay
+Set filter delay in seconds. Higher value means more accurate.
+Default is @code{0.01}.
+
+@item accuracy
+Set filter accuracy in Hz. Lower value means more accurate.
+Default is @code{5}.
+
+@item wfunc
+Set window function. Acceptable values are:
+@table @option
+@item rectangular
+rectangular window, useful when gain curve is already smooth
+@item hann
+hann window (default)
+@item hamming
+hamming window
+@item blackman
+blackman window
+@item nuttall3
+3-terms continuous 1st derivative nuttall window
+@item mnuttall3
+minimum 3-terms discontinuous nuttall window
+@item nuttall
+4-terms continuous 1st derivative nuttall window
+@item bnuttall
+minimum 4-terms discontinuous nuttall (blackman-nuttall) window
+@item bharris
+blackman-harris window
+@end table
+
+@item fixed
+If enabled, use fixed number of audio samples. This improves speed when
+filtering with large delay. Default is disabled.
+
+@item multi
+Enable multichannels evaluation on gain. Default is disabled.
+@end table
+
+@subsection Examples
+@itemize
+@item
+lowpass at 1000 Hz:
+@example
+firequalizer=gain='if(lt(f,1000), 0, -INF)'
+@end example
+@item
+lowpass at 1000 Hz with gain_entry:
+@example
+firequalizer=gain_entry='entry(1000,0); entry(1001, -INF)'
+@end example
+@item
+custom equalization:
+@example
+firequalizer=gain_entry='entry(100,0); entry(400, -4); entry(1000, -6); entry(2000, 0)'
+@end example
+@item
+higher delay:
+@example
+firequalizer=delay=0.1:fixed=on
+@end example
+@item
+lowpass on left channel, highpass on right channel:
+@example
+firequalizer=gain='if(eq(chid,1), gain_interpolate(f), if(eq(chid,2), gain_interpolate(1e6+f), 0))'
+:gain_entry='entry(1000, 0); entry(1001,-INF); entry(1e6+1000,0)':multi=on
+@end example
+@end itemize
+
 @section flanger
 Apply a flanging effect to the audio.
 

libavfilter/Makefile

@@ -80,6 +80,7 @@ OBJS-$(CONFIG_EARWAX_FILTER)                 += af_earwax.o
 OBJS-$(CONFIG_EBUR128_FILTER)                += f_ebur128.o
 OBJS-$(CONFIG_EQUALIZER_FILTER)              += af_biquads.o
 OBJS-$(CONFIG_EXTRASTEREO_FILTER)            += af_extrastereo.o
+OBJS-$(CONFIG_FIREQUALIZER_FILTER)           += af_firequalizer.o
 OBJS-$(CONFIG_FLANGER_FILTER)                += af_flanger.o generate_wave_table.o
 OBJS-$(CONFIG_HIGHPASS_FILTER)               += af_biquads.o
 OBJS-$(CONFIG_JOIN_FILTER)                   += af_join.o

libavfilter/allfilters.c

@@ -101,6 +101,7 @@ void avfilter_register_all(void)
     REGISTER_FILTER(EBUR128,        ebur128,        af);
     REGISTER_FILTER(EQUALIZER,      equalizer,      af);
     REGISTER_FILTER(EXTRASTEREO,    extrastereo,    af);
+    REGISTER_FILTER(FIREQUALIZER,   firequalizer,   af);
     REGISTER_FILTER(FLANGER,        flanger,        af);
     REGISTER_FILTER(HIGHPASS,       highpass,       af);
     REGISTER_FILTER(JOIN,           join,           af);

libavfilter/version.h

@@ -30,7 +30,7 @@
 #include "libavutil/version.h"
 
 #define LIBAVFILTER_VERSION_MAJOR   6
-#define LIBAVFILTER_VERSION_MINOR  34
+#define LIBAVFILTER_VERSION_MINOR  35
 #define LIBAVFILTER_VERSION_MICRO 100
 
 #define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \