mirror of https://git.ffmpeg.org/ffmpeg.git
avfilter: add audio emphasis filter
Signed-off-by: Paul B Mahol <onemda@gmail.com>
This commit is contained in:
Paul B Mahol
parent
7234e04e35
commit
5d2cc00dd0
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@ -40,6 +40,7 @@ version <next>:
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- apulsator filter
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- sidechaingate audio filter
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- mipsdspr1 option has been renamed to mipsdsp
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- aemphasis filter
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version 2.8:
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@ -1051,6 +1051,21 @@ int main(void){ $func(); }
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EOF
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}
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check_complexfunc(){
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log check_complexfunc "$@"
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func=$1
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narg=$2
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shift 2
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test $narg = 2 && args="f, g" || args="f * I"
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disable $func
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check_ld "cc" "$@" <<EOF && enable $func
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#include <complex.h>
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#include <math.h>
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float foo(complex float f, complex float g) { return $func($args); }
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int main(void){ return (int) foo; }
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EOF
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}
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check_mathfunc(){
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log check_mathfunc "$@"
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func=$1
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@ -1768,6 +1783,11 @@ INTRINSICS_LIST="
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intrinsics_neon
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"
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COMPLEX_FUNCS="
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cabs
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cexp
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"
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MATH_FUNCS="
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atanf
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atan2f
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@ -1903,6 +1923,7 @@ HAVE_LIST="
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$ARCH_FEATURES
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$ATOMICS_LIST
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$BUILTIN_LIST
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$COMPLEX_FUNCS
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$HAVE_LIST_CMDLINE
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$HAVE_LIST_PUB
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$HEADERS_LIST
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@ -2785,6 +2806,7 @@ unix_protocol_deps="sys_un_h"
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unix_protocol_select="network"
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# filters
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aemphasis_filter_deps="cabs cexp"
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amovie_filter_deps="avcodec avformat"
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aresample_filter_deps="swresample"
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ass_filter_deps="libass"
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@ -5324,6 +5346,10 @@ for func in $MATH_FUNCS; do
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eval check_mathfunc $func \${${func}_args:-1}
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done
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for func in $COMPLEX_FUNCS; do
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eval check_complexfunc $func \${${func}_args:-1}
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done
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# these are off by default, so fail if requested and not available
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enabled avfoundation_indev && { check_header_oc AVFoundation/AVFoundation.h || disable avfoundation_indev; }
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enabled avfoundation_indev && { check_lib2 CoreGraphics/CoreGraphics.h CGGetActiveDisplayList -framework CoreGraphics ||
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@ -528,6 +528,52 @@ aecho=0.8:0.9:1000|1800:0.3|0.25
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@end example
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@end itemize
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@section aemphasis
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Audio emphasis filter creates or restores material directly taken from LPs or
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emphased CDs with different filter curves. E.g. to store music on vinyl the
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signal has to be altered by a filter first to even out the disadvantages of
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this recording medium.
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Once the material is played back the inverse filter has to be applied to
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restore the distortion of the frequency response.
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The filter accepts the following options:
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@table @option
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@item level_in
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Set input gain.
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@item level_out
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Set output gain.
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@item mode
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Set filter mode. For restoring material use @code{reproduction} mode, otherwise
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use @code{production} mode. Default is @code{reproduction} mode.
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@item type
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Set filter type. Selects medium. Can be one of the following:
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@table @option
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@item col
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select Columbia.
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@item emi
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select EMI.
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@item bsi
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select BSI (78RPM).
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@item riaa
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select RIAA.
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@item cd
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select Compact Disc (CD).
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@item 50fm
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select 50µs (FM).
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@item 75fm
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select 75µs (FM).
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@item 50kf
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select 50µs (FM-KF).
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@item 75kf
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select 75µs (FM-KF).
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@end table
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@end table
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@section aeval
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Modify an audio signal according to the specified expressions.
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@ -27,6 +27,7 @@ OBJS-$(CONFIG_ACOMPRESSOR_FILTER) += af_sidechaincompress.o
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OBJS-$(CONFIG_ACROSSFADE_FILTER) += af_afade.o
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OBJS-$(CONFIG_ADELAY_FILTER) += af_adelay.o
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OBJS-$(CONFIG_AECHO_FILTER) += af_aecho.o
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OBJS-$(CONFIG_AEMPHASIS_FILTER) += af_aemphasis.o
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OBJS-$(CONFIG_AEVAL_FILTER) += aeval.o
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OBJS-$(CONFIG_AFADE_FILTER) += af_afade.o
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OBJS-$(CONFIG_AFORMAT_FILTER) += af_aformat.o
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@ -0,0 +1,370 @@
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/*
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* Copyright (c) 2001-2010 Krzysztof Foltman, Markus Schmidt, Thor Harald Johansen, Damien Zammit and others
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#include <complex.h>
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#include "libavutil/opt.h"
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#include "avfilter.h"
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#include "internal.h"
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#include "audio.h"
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typedef struct BiquadCoeffs {
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double a0, a1, a2, b1, b2;
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} BiquadCoeffs;
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typedef struct BiquadD2 {
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double a0, a1, a2, b1, b2, w1, w2;
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} BiquadD2;
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typedef struct RIAACurve {
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BiquadD2 r1;
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BiquadD2 brickw;
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int use_brickw;
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} RIAACurve;
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typedef struct AudioEmphasisContext {
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const AVClass *class;
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int mode, type;
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double level_in, level_out;
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RIAACurve *rc;
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} AudioEmphasisContext;
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#define OFFSET(x) offsetof(AudioEmphasisContext, x)
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#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
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static const AVOption aemphasis_options[] = {
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{ "level_in", "set input gain", OFFSET(level_in), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0, 64, FLAGS },
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{ "level_out", "set output gain", OFFSET(level_out), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0, 64, FLAGS },
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{ "mode", "set filter mode", OFFSET(mode), AV_OPT_TYPE_INT, {.i64=0}, 0, 1, FLAGS, "mode" },
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{ "reproduction", NULL, 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, FLAGS, "mode" },
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{ "production", NULL, 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, FLAGS, "mode" },
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{ "type", "set filter type", OFFSET(type), AV_OPT_TYPE_INT, {.i64=4}, 0, 8, FLAGS, "type" },
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{ "col", "Columbia", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, FLAGS, "type" },
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{ "emi", "EMI", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, FLAGS, "type" },
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{ "bsi", "BSI (78RPM)", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, FLAGS, "type" },
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{ "riaa", "RIAA", 0, AV_OPT_TYPE_CONST, {.i64=3}, 0, 0, FLAGS, "type" },
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{ "cd", "Compact Disc (CD)", 0, AV_OPT_TYPE_CONST, {.i64=4}, 0, 0, FLAGS, "type" },
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{ "50fm", "50µs (FM)", 0, AV_OPT_TYPE_CONST, {.i64=5}, 0, 0, FLAGS, "type" },
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{ "75fm", "75µs (FM)", 0, AV_OPT_TYPE_CONST, {.i64=6}, 0, 0, FLAGS, "type" },
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{ "50kf", "50µs (FM-KF)", 0, AV_OPT_TYPE_CONST, {.i64=7}, 0, 0, FLAGS, "type" },
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{ "75kf", "75µs (FM-KF)", 0, AV_OPT_TYPE_CONST, {.i64=8}, 0, 0, FLAGS, "type" },
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{ NULL }
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};
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AVFILTER_DEFINE_CLASS(aemphasis);
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static inline double biquad(BiquadD2 *bq, double in)
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{
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double n = in;
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double tmp = n - bq->w1 * bq->b1 - bq->w2 * bq->b2;
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double out = tmp * bq->a0 + bq->w1 * bq->a1 + bq->w2 * bq->a2;
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bq->w2 = bq->w1;
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bq->w1 = tmp;
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return out;
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}
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static int filter_frame(AVFilterLink *inlink, AVFrame *in)
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{
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AVFilterContext *ctx = inlink->dst;
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AVFilterLink *outlink = ctx->outputs[0];
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AudioEmphasisContext *s = ctx->priv;
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const double *src = (const double *)in->data[0];
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const double level_out = s->level_out;
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const double level_in = s->level_in;
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AVFrame *out;
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double *dst;
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int n, c;
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if (av_frame_is_writable(in)) {
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out = in;
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} else {
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out = ff_get_audio_buffer(inlink, in->nb_samples);
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if (!out) {
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av_frame_free(&in);
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return AVERROR(ENOMEM);
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}
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av_frame_copy_props(out, in);
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}
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dst = (double *)out->data[0];
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for (n = 0; n < in->nb_samples; n++) {
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for (c = 0; c < inlink->channels; c++)
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dst[c] = level_out * biquad(&s->rc[c].r1, s->rc[c].use_brickw ? biquad(&s->rc[c].brickw, src[c] * level_in) : src[c] * level_in);
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dst += inlink->channels;
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src += inlink->channels;
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}
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if (in != out)
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av_frame_free(&in);
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return ff_filter_frame(outlink, out);
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}
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static int query_formats(AVFilterContext *ctx)
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{
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AVFilterChannelLayouts *layouts;
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AVFilterFormats *formats;
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static const enum AVSampleFormat sample_fmts[] = {
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AV_SAMPLE_FMT_DBL,
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AV_SAMPLE_FMT_NONE
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};
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int ret;
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layouts = ff_all_channel_counts();
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if (!layouts)
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return AVERROR(ENOMEM);
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ret = ff_set_common_channel_layouts(ctx, layouts);
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if (ret < 0)
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return ret;
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formats = ff_make_format_list(sample_fmts);
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if (!formats)
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return AVERROR(ENOMEM);
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ret = ff_set_common_formats(ctx, formats);
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if (ret < 0)
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return ret;
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formats = ff_all_samplerates();
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if (!formats)
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return AVERROR(ENOMEM);
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return ff_set_common_samplerates(ctx, formats);
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}
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static inline void set_highshelf_rbj(BiquadD2 *bq, double freq, double q, double peak, double sr)
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{
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double A = sqrt(peak);
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double w0 = freq * 2 * M_PI / sr;
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double alpha = sin(w0) / (2 * q);
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double cw0 = cos(w0);
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double tmp = 2 * sqrt(A) * alpha;
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double b0 = 0, ib0 = 0;
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bq->a0 = A*( (A+1) + (A-1)*cw0 + tmp);
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bq->a1 = -2*A*( (A-1) + (A+1)*cw0);
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bq->a2 = A*( (A+1) + (A-1)*cw0 - tmp);
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b0 = (A+1) - (A-1)*cw0 + tmp;
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bq->b1 = 2*( (A-1) - (A+1)*cw0);
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bq->b2 = (A+1) - (A-1)*cw0 - tmp;
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ib0 = 1 / b0;
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bq->b1 *= ib0;
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bq->b2 *= ib0;
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bq->a0 *= ib0;
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bq->a1 *= ib0;
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bq->a2 *= ib0;
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}
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static inline void set_lp_rbj(BiquadD2 *bq, double fc, double q, double sr, double gain)
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{
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double omega = 2.0 * M_PI * fc / sr;
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double sn = sin(omega);
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double cs = cos(omega);
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double alpha = sn/(2 * q);
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double inv = 1.0/(1.0 + alpha);
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bq->a2 = bq->a0 = gain * inv * (1.0 - cs) * 0.5;
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bq->a1 = bq->a0 + bq->a0;
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bq->b1 = (-2.0 * cs * inv);
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bq->b2 = ((1.0 - alpha) * inv);
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}
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static double freq_gain(BiquadCoeffs *c, double freq, double sr)
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{
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double complex z, w;
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freq *= 2.0 * M_PI / sr;
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w = 0 + I * freq;
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z = 1.0 / cexp(w);
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return cabs(((double complex)c->a0 + c->a1 * z + c->a2 * z*z) /
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((double complex)1.0 + c->b1 * z + c->b2 * z*z));
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}
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static int config_input(AVFilterLink *inlink)
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{
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double i, j, k, g, t, a0, a1, a2, b1, b2, tau1, tau2, tau3;
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double cutfreq, gain1kHz, gc, sr = inlink->sample_rate;
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AVFilterContext *ctx = inlink->dst;
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AudioEmphasisContext *s = ctx->priv;
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BiquadCoeffs coeffs;
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int ch;
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s->rc = av_calloc(inlink->channels, sizeof(*s->rc));
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if (!s->rc)
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return AVERROR(ENOMEM);
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switch (s->type) {
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case 0: //"Columbia"
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i = 100.;
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j = 500.;
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k = 1590.;
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break;
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case 1: //"EMI"
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i = 70.;
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j = 500.;
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k = 2500.;
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break;
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case 2: //"BSI(78rpm)"
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i = 50.;
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j = 353.;
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k = 3180.;
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break;
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case 3: //"RIAA"
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default:
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tau1 = 0.003180;
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tau2 = 0.000318;
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tau3 = 0.000075;
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i = 1. / (2. * M_PI * tau1);
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j = 1. / (2. * M_PI * tau2);
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k = 1. / (2. * M_PI * tau3);
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break;
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case 4: //"CD Mastering"
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tau1 = 0.000050;
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tau2 = 0.000015;
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tau3 = 0.0000001;// 1.6MHz out of audible range for null impact
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i = 1. / (2. * M_PI * tau1);
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j = 1. / (2. * M_PI * tau2);
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k = 1. / (2. * M_PI * tau3);
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break;
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case 5: //"50µs FM (Europe)"
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tau1 = 0.000050;
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tau2 = tau1 / 20;// not used
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tau3 = tau1 / 50;//
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i = 1. / (2. * M_PI * tau1);
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j = 1. / (2. * M_PI * tau2);
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k = 1. / (2. * M_PI * tau3);
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break;
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case 6: //"75µs FM (US)"
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tau1 = 0.000075;
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tau2 = tau1 / 20;// not used
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tau3 = tau1 / 50;//
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i = 1. / (2. * M_PI * tau1);
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j = 1. / (2. * M_PI * tau2);
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k = 1. / (2. * M_PI * tau3);
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break;
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}
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i *= 2 * M_PI;
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j *= 2 * M_PI;
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k *= 2 * M_PI;
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t = 1. / sr;
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//swap a1 b1, a2 b2
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if (s->type == 7 || s->type == 8) {
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s->rc[0].use_brickw = 0;
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double tau = (s->type == 7 ? 0.000050 : 0.000075);
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double f = 1.0 / (2 * M_PI * tau);
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double nyq = sr * 0.5;
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double gain = sqrt(1.0 + nyq * nyq / (f * f)); // gain at Nyquist
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double cfreq = sqrt((gain - 1.0) * f * f); // frequency
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double q = 1.0;
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if (s->type == 8)
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q = pow((sr / 3269.0) + 19.5, -0.25); // somewhat poor curve-fit
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if (s->type == 7)
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q = pow((sr / 4750.0) + 19.5, -0.25);
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if (s->mode == 0)
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set_highshelf_rbj(&s->rc[0].r1, cfreq, q, 1. / gain, sr);
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else
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set_highshelf_rbj(&s->rc[0].r1, cfreq, q, gain, sr);
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} else {
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s->rc[0].use_brickw = 1;
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if (s->mode == 0) { // Reproduction
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g = 1. / (4.+2.*i*t+2.*k*t+i*k*t*t);
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a0 = (2.*t+j*t*t)*g;
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a1 = (2.*j*t*t)*g;
|
||||
a2 = (-2.*t+j*t*t)*g;
|
||||
b1 = (-8.+2.*i*k*t*t)*g;
|
||||
b2 = (4.-2.*i*t-2.*k*t+i*k*t*t)*g;
|
||||
} else { // Production
|
||||
g = 1. / (2.*t+j*t*t);
|
||||
a0 = (4.+2.*i*t+2.*k*t+i*k*t*t)*g;
|
||||
a1 = (-8.+2.*i*k*t*t)*g;
|
||||
a2 = (4.-2.*i*t-2.*k*t+i*k*t*t)*g;
|
||||
b1 = (2.*j*t*t)*g;
|
||||
b2 = (-2.*t+j*t*t)*g;
|
||||
}
|
||||
|
||||
coeffs.a0 = a0;
|
||||
coeffs.a1 = a1;
|
||||
coeffs.a2 = a2;
|
||||
coeffs.b1 = b1;
|
||||
coeffs.b2 = b2;
|
||||
|
||||
// the coeffs above give non-normalized value, so it should be normalized to produce 0dB at 1 kHz
|
||||
// find actual gain
|
||||
// Note: for FM emphasis, use 100 Hz for normalization instead
|
||||
gain1kHz = freq_gain(&coeffs, 1000.0, sr);
|
||||
// divide one filter's x[n-m] coefficients by that value
|
||||
gc = 1.0 / gain1kHz;
|
||||
s->rc[0].r1.a0 = coeffs.a0 * gc;
|
||||
s->rc[0].r1.a1 = coeffs.a1 * gc;
|
||||
s->rc[0].r1.a2 = coeffs.a2 * gc;
|
||||
s->rc[0].r1.b1 = coeffs.b1;
|
||||
s->rc[0].r1.b2 = coeffs.b2;
|
||||
}
|
||||
|
||||
cutfreq = FFMIN(0.45 * sr, 21000.);
|
||||
set_lp_rbj(&s->rc[0].brickw, cutfreq, 0.707, sr, 1.);
|
||||
|
||||
for (ch = 1; ch < inlink->channels; ch++) {
|
||||
memcpy(&s->rc[ch], &s->rc[0], sizeof(RIAACurve));
|
||||
}
|
||||
|
||||
return 0;
|
||||
}
|
||||
|
||||
static av_cold void uninit(AVFilterContext *ctx)
|
||||
{
|
||||
AudioEmphasisContext *s = ctx->priv;
|
||||
av_freep(&s->rc);
|
||||
}
|
||||
|
||||
static const AVFilterPad avfilter_af_aemphasis_inputs[] = {
|
||||
{
|
||||
.name = "default",
|
||||
.type = AVMEDIA_TYPE_AUDIO,
|
||||
.config_props = config_input,
|
||||
.filter_frame = filter_frame,
|
||||
},
|
||||
{ NULL }
|
||||
};
|
||||
|
||||
static const AVFilterPad avfilter_af_aemphasis_outputs[] = {
|
||||
{
|
||||
.name = "default",
|
||||
.type = AVMEDIA_TYPE_AUDIO,
|
||||
},
|
||||
{ NULL }
|
||||
};
|
||||
|
||||
AVFilter ff_af_aemphasis = {
|
||||
.name = "aemphasis",
|
||||
.description = NULL_IF_CONFIG_SMALL("Audio emphasis."),
|
||||
.priv_size = sizeof(AudioEmphasisContext),
|
||||
.priv_class = &aemphasis_class,
|
||||
.uninit = uninit,
|
||||
.query_formats = query_formats,
|
||||
.inputs = avfilter_af_aemphasis_inputs,
|
||||
.outputs = avfilter_af_aemphasis_outputs,
|
||||
};
|
||||
|
|
@ -49,6 +49,7 @@ void avfilter_register_all(void)
|
|||
REGISTER_FILTER(ACROSSFADE, acrossfade, af);
|
||||
REGISTER_FILTER(ADELAY, adelay, af);
|
||||
REGISTER_FILTER(AECHO, aecho, af);
|
||||
REGISTER_FILTER(AEMPHASIS, aemphasis, af);
|
||||
REGISTER_FILTER(AEVAL, aeval, af);
|
||||
REGISTER_FILTER(AFADE, afade, af);
|
||||
REGISTER_FILTER(AFORMAT, aformat, af);
|
||||
|
|
|
|||
|
|
@ -30,7 +30,7 @@
|
|||
#include "libavutil/version.h"
|
||||
|
||||
#define LIBAVFILTER_VERSION_MAJOR 6
|
||||
#define LIBAVFILTER_VERSION_MINOR 19
|
||||
#define LIBAVFILTER_VERSION_MINOR 20
|
||||
#define LIBAVFILTER_VERSION_MICRO 100
|
||||
|
||||
#define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \
|
||||
|
|
|
|||
Loading…
Reference in New Issue