[FFmpeg-devel] [PATCH] avfilter: add audio emphasis filter
Paul B Mahol
onemda at gmail.com
Thu Dec 3 23:52:23 CET 2015
Signed-off-by: Paul B Mahol <onemda at gmail.com>
---
Changelog | 1 +
configure | 26 ++++
doc/filters.texi | 46 ++++++
libavfilter/Makefile | 1 +
libavfilter/af_aemphasis.c | 370 +++++++++++++++++++++++++++++++++++++++++++++
libavfilter/allfilters.c | 1 +
libavfilter/version.h | 2 +-
7 files changed, 446 insertions(+), 1 deletion(-)
create mode 100644 libavfilter/af_aemphasis.c
diff --git a/Changelog b/Changelog
index 2d2a92b..552fab1 100644
--- a/Changelog
+++ b/Changelog
@@ -39,6 +39,7 @@ version <next>:
- support encoding 16-bit RLE SGI images
- apulsator filter
- sidechaingate audio filter
+- aemphasis filter
version 2.8:
diff --git a/configure b/configure
index a30d831..10631e8 100755
--- a/configure
+++ b/configure
@@ -1051,6 +1051,21 @@ int main(void){ $func(); }
EOF
}
+check_complexfunc(){
+ log check_complexfunc "$@"
+ func=$1
+ narg=$2
+ shift 2
+ test $narg = 2 && args="f, g" || args="f"
+ disable $func
+ check_ld "cc" "$@" <<EOF && enable $func
+#include <complex.h>
+#include <math.h>
+float foo(complex float f, complex float g) { return $func($args); }
+int main(void){ return (int) foo; }
+EOF
+}
+
check_mathfunc(){
log check_mathfunc "$@"
func=$1
@@ -1768,6 +1783,11 @@ INTRINSICS_LIST="
intrinsics_neon
"
+COMPLEX_FUNCS="
+ cabs
+ cexp
+"
+
MATH_FUNCS="
atanf
atan2f
@@ -1903,6 +1923,7 @@ HAVE_LIST="
$ARCH_FEATURES
$ATOMICS_LIST
$BUILTIN_LIST
+ $COMPLEX_FUNCS
$HAVE_LIST_CMDLINE
$HAVE_LIST_PUB
$HEADERS_LIST
@@ -2785,6 +2806,7 @@ unix_protocol_deps="sys_un_h"
unix_protocol_select="network"
# filters
+aemphasis_filter_deps="cabs cexp"
amovie_filter_deps="avcodec avformat"
aresample_filter_deps="swresample"
ass_filter_deps="libass"
@@ -5324,6 +5346,10 @@ for func in $MATH_FUNCS; do
eval check_mathfunc $func \${${func}_args:-1}
done
+for func in $COMPLEX_FUNCS; do
+ eval check_complexfunc $func \${${func}_args:-1}
+done
+
# these are off by default, so fail if requested and not available
enabled avfoundation_indev && { check_header_oc AVFoundation/AVFoundation.h || disable avfoundation_indev; }
enabled avfoundation_indev && { check_lib2 CoreGraphics/CoreGraphics.h CGGetActiveDisplayList -framework CoreGraphics ||
diff --git a/doc/filters.texi b/doc/filters.texi
index bf299ca..12082de 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -528,6 +528,52 @@ aecho=0.8:0.9:1000|1800:0.3|0.25
@end example
@end itemize
+ at section aemphasis
+Audio emphasis filter creates or restores material directly taken from LPs or
+emphased CDs with different filter curves. E.g. to store music on vinyl the
+signal has to be altered by a filter first to even out the disadvantages of
+this recording medium.
+Once the material is played back the inverse filter has to be applied to
+restore the distortion of the frequency response.
+
+The filter accepts the following options:
+
+ at table @option
+ at item level_in
+Set input gain.
+
+ at item level_out
+Set output gain.
+
+ at item mode
+Set filter mode. For restoring material use @code{reproduction} mode, otherwise
+use @code{production} mode. Default is @code{reproduction} mode.
+
+ at item type
+Set filter type. Selects medium. Can be one of the following:
+
+ at table @option
+ at item col
+select Columbia.
+ at item emi
+select EMI.
+ at item bsi
+select BSI (78RPM).
+ at item riaa
+select RIAA.
+ at item cd
+select Compact Disc (CD).
+ at item 50fm
+select 50µs (FM).
+ at item 75fm
+select 75µs (FM).
+ at item 50kf
+select 50µs (FM-KF).
+ at item 75kf
+select 75µs (FM-KF).
+ at end table
+ at end table
+
@section aeval
Modify an audio signal according to the specified expressions.
diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index 740a640..8884d1d 100644
--- a/libavfilter/Makefile
+++ b/libavfilter/Makefile
@@ -27,6 +27,7 @@ OBJS-$(CONFIG_ACOMPRESSOR_FILTER) += af_sidechaincompress.o
OBJS-$(CONFIG_ACROSSFADE_FILTER) += af_afade.o
OBJS-$(CONFIG_ADELAY_FILTER) += af_adelay.o
OBJS-$(CONFIG_AECHO_FILTER) += af_aecho.o
+OBJS-$(CONFIG_AEMPHASIS_FILTER) += af_aemphasis.o
OBJS-$(CONFIG_AEVAL_FILTER) += aeval.o
OBJS-$(CONFIG_AFADE_FILTER) += af_afade.o
OBJS-$(CONFIG_AFORMAT_FILTER) += af_aformat.o
diff --git a/libavfilter/af_aemphasis.c b/libavfilter/af_aemphasis.c
new file mode 100644
index 0000000..4501858
--- /dev/null
+++ b/libavfilter/af_aemphasis.c
@@ -0,0 +1,370 @@
+/*
+ * Copyright (c) 2001-2010 Krzysztof Foltman, Markus Schmidt, Thor Harald Johansen, Damien Zammit and others
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg 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
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#include <complex.h>
+
+#include "libavutil/opt.h"
+#include "avfilter.h"
+#include "internal.h"
+#include "audio.h"
+
+typedef struct BiquadCoeffs {
+ double a0, a1, a2, b1, b2;
+} BiquadCoeffs;
+
+typedef struct BiquadD2 {
+ double a0, a1, a2, b1, b2, w1, w2;
+} BiquadD2;
+
+typedef struct RIAACurve {
+ BiquadD2 r1;
+ BiquadD2 brickw;
+ int use_brickw;
+} RIAACurve;
+
+typedef struct AudioEmphasisContext {
+ const AVClass *class;
+ int mode, type;
+ double level_in, level_out;
+
+ RIAACurve *rc;
+} AudioEmphasisContext;
+
+#define OFFSET(x) offsetof(AudioEmphasisContext, x)
+#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
+
+static const AVOption aemphasis_options[] = {
+ { "level_in", "set input gain", OFFSET(level_in), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0, 64, FLAGS },
+ { "level_out", "set output gain", OFFSET(level_out), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0, 64, FLAGS },
+ { "mode", "set filter mode", OFFSET(mode), AV_OPT_TYPE_INT, {.i64=0}, 0, 1, FLAGS, "mode" },
+ { "reproduction", NULL, 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, FLAGS, "mode" },
+ { "production", NULL, 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, FLAGS, "mode" },
+ { "type", "set filter type", OFFSET(type), AV_OPT_TYPE_INT, {.i64=4}, 0, 8, FLAGS, "type" },
+ { "col", "Columbia", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, FLAGS, "type" },
+ { "emi", "EMI", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, FLAGS, "type" },
+ { "bsi", "BSI (78RPM)", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, FLAGS, "type" },
+ { "riaa", "RIAA", 0, AV_OPT_TYPE_CONST, {.i64=3}, 0, 0, FLAGS, "type" },
+ { "cd", "Compact Disc (CD)", 0, AV_OPT_TYPE_CONST, {.i64=4}, 0, 0, FLAGS, "type" },
+ { "50fm", "50µs (FM)", 0, AV_OPT_TYPE_CONST, {.i64=5}, 0, 0, FLAGS, "type" },
+ { "75fm", "75µs (FM)", 0, AV_OPT_TYPE_CONST, {.i64=6}, 0, 0, FLAGS, "type" },
+ { "50kf", "50µs (FM-KF)", 0, AV_OPT_TYPE_CONST, {.i64=7}, 0, 0, FLAGS, "type" },
+ { "75kf", "75µs (FM-KF)", 0, AV_OPT_TYPE_CONST, {.i64=8}, 0, 0, FLAGS, "type" },
+ { NULL }
+};
+
+AVFILTER_DEFINE_CLASS(aemphasis);
+
+static inline double biquad(BiquadD2 *bq, double in)
+{
+ double n = in;
+ double tmp = n - bq->w1 * bq->b1 - bq->w2 * bq->b2;
+ double out = tmp * bq->a0 + bq->w1 * bq->a1 + bq->w2 * bq->a2;
+
+ bq->w2 = bq->w1;
+ bq->w1 = tmp;
+
+ return out;
+}
+
+static int filter_frame(AVFilterLink *inlink, AVFrame *in)
+{
+ AVFilterContext *ctx = inlink->dst;
+ AVFilterLink *outlink = ctx->outputs[0];
+ AudioEmphasisContext *s = ctx->priv;
+ const double *src = (const double *)in->data[0];
+ const double level_out = s->level_out;
+ const double level_in = s->level_in;
+ AVFrame *out;
+ double *dst;
+ int n, c;
+
+ if (av_frame_is_writable(in)) {
+ out = in;
+ } else {
+ out = ff_get_audio_buffer(inlink, in->nb_samples);
+ if (!out) {
+ av_frame_free(&in);
+ return AVERROR(ENOMEM);
+ }
+ av_frame_copy_props(out, in);
+ }
+ dst = (double *)out->data[0];
+
+ for (n = 0; n < in->nb_samples; n++) {
+ for (c = 0; c < inlink->channels; c++)
+ 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);
+ dst += inlink->channels;
+ src += inlink->channels;
+ }
+
+ if (in != out)
+ av_frame_free(&in);
+ return ff_filter_frame(outlink, out);
+}
+
+static int query_formats(AVFilterContext *ctx)
+{
+ AVFilterChannelLayouts *layouts;
+ AVFilterFormats *formats;
+ static const enum AVSampleFormat sample_fmts[] = {
+ AV_SAMPLE_FMT_DBL,
+ AV_SAMPLE_FMT_NONE
+ };
+ int ret;
+
+ layouts = ff_all_channel_counts();
+ if (!layouts)
+ return AVERROR(ENOMEM);
+ ret = ff_set_common_channel_layouts(ctx, layouts);
+ if (ret < 0)
+ return ret;
+
+ formats = ff_make_format_list(sample_fmts);
+ if (!formats)
+ return AVERROR(ENOMEM);
+ ret = ff_set_common_formats(ctx, formats);
+ if (ret < 0)
+ return ret;
+
+ formats = ff_all_samplerates();
+ if (!formats)
+ return AVERROR(ENOMEM);
+ return ff_set_common_samplerates(ctx, formats);
+}
+
+static inline void set_highshelf_rbj(BiquadD2 *bq, double freq, double q, double peak, double sr)
+{
+ double A = sqrt(peak);
+ double w0 = freq * 2 * M_PI / sr;
+ double alpha = sin(w0) / (2 * q);
+ double cw0 = cos(w0);
+ double tmp = 2 * sqrt(A) * alpha;
+ double b0 = 0, ib0 = 0;
+
+ bq->a0 = A*( (A+1) + (A-1)*cw0 + tmp);
+ bq->a1 = -2*A*( (A-1) + (A+1)*cw0);
+ bq->a2 = A*( (A+1) + (A-1)*cw0 - tmp);
+ b0 = (A+1) - (A-1)*cw0 + tmp;
+ bq->b1 = 2*( (A-1) - (A+1)*cw0);
+ bq->b2 = (A+1) - (A-1)*cw0 - tmp;
+
+ ib0 = 1 / b0;
+ bq->b1 *= ib0;
+ bq->b2 *= ib0;
+ bq->a0 *= ib0;
+ bq->a1 *= ib0;
+ bq->a2 *= ib0;
+}
+
+static inline void set_lp_rbj(BiquadD2 *bq, double fc, double q, double sr, double gain)
+{
+ double omega = 2.0 * M_PI * fc / sr;
+ double sn = sin(omega);
+ double cs = cos(omega);
+ double alpha = sn/(2 * q);
+ double inv = 1.0/(1.0 + alpha);
+
+ bq->a2 = bq->a0 = gain * inv * (1.0 - cs) * 0.5;
+ bq->a1 = bq->a0 + bq->a0;
+ bq->b1 = (-2.0 * cs * inv);
+ bq->b2 = ((1.0 - alpha) * inv);
+}
+
+static double freq_gain(BiquadCoeffs *c, double freq, double sr)
+{
+ double complex z, w;
+
+ freq *= 2.0 * M_PI / sr;
+ w = 0 + I * freq;
+ z = 1.0 / cexp(w);
+
+ return cabs(((double complex)c->a0 + c->a1 * z + c->a2 * z*z) /
+ ((double complex)1.0 + c->b1 * z + c->b2 * z*z));
+}
+
+static int config_input(AVFilterLink *inlink)
+{
+ double i, j, k, g, t, a0, a1, a2, b1, b2, tau1, tau2, tau3;
+ double cutfreq, gain1kHz, gc, sr = inlink->sample_rate;
+ AVFilterContext *ctx = inlink->dst;
+ AudioEmphasisContext *s = ctx->priv;
+ BiquadCoeffs coeffs;
+ int ch;
+
+ s->rc = av_calloc(inlink->channels, sizeof(*s->rc));
+ if (!s->rc)
+ return AVERROR(ENOMEM);
+
+ switch (s->type) {
+ case 0: //"Columbia"
+ i = 100.;
+ j = 500.;
+ k = 1590.;
+ break;
+ case 1: //"EMI"
+ i = 70.;
+ j = 500.;
+ k = 2500.;
+ break;
+ case 2: //"BSI(78rpm)"
+ i = 50.;
+ j = 353.;
+ k = 3180.;
+ break;
+ case 3: //"RIAA"
+ default:
+ tau1 = 0.003180;
+ tau2 = 0.000318;
+ tau3 = 0.000075;
+ i = 1. / (2. * M_PI * tau1);
+ j = 1. / (2. * M_PI * tau2);
+ k = 1. / (2. * M_PI * tau3);
+ break;
+ case 4: //"CD Mastering"
+ tau1 = 0.000050;
+ tau2 = 0.000015;
+ tau3 = 0.0000001;// 1.6MHz out of audible range for null impact
+ i = 1. / (2. * M_PI * tau1);
+ j = 1. / (2. * M_PI * tau2);
+ k = 1. / (2. * M_PI * tau3);
+ break;
+ case 5: //"50µs FM (Europe)"
+ tau1 = 0.000050;
+ tau2 = tau1 / 20;// not used
+ tau3 = tau1 / 50;//
+ i = 1. / (2. * M_PI * tau1);
+ j = 1. / (2. * M_PI * tau2);
+ k = 1. / (2. * M_PI * tau3);
+ break;
+ case 6: //"75µs FM (US)"
+ tau1 = 0.000075;
+ tau2 = tau1 / 20;// not used
+ tau3 = tau1 / 50;//
+ i = 1. / (2. * M_PI * tau1);
+ j = 1. / (2. * M_PI * tau2);
+ k = 1. / (2. * M_PI * tau3);
+ break;
+ }
+
+ i *= 2 * M_PI;
+ j *= 2 * M_PI;
+ k *= 2 * M_PI;
+
+ t = 1. / sr;
+
+ //swap a1 b1, a2 b2
+ if (s->type == 7 || s->type == 8) {
+ s->rc[0].use_brickw = 0;
+ double tau = (s->type == 7 ? 0.000050 : 0.000075);
+ double f = 1.0 / (2 * M_PI * tau);
+ double nyq = sr * 0.5;
+ double gain = sqrt(1.0 + nyq * nyq / (f * f)); // gain at Nyquist
+ double cfreq = sqrt((gain - 1.0) * f * f); // frequency
+ double q = 1.0;
+
+ if (s->type == 8)
+ q = pow((sr / 3269.0) + 19.5, -0.25); // somewhat poor curve-fit
+ if (s->type == 7)
+ q = pow((sr / 4750.0) + 19.5, -0.25);
+ if (s->mode == 0)
+ set_highshelf_rbj(&s->rc[0].r1, cfreq, q, 1. / gain, sr);
+ else
+ set_highshelf_rbj(&s->rc[0].r1, cfreq, q, gain, sr);
+ } else {
+ s->rc[0].use_brickw = 1;
+ if (s->mode == 0) { // Reproduction
+ g = 1. / (4.+2.*i*t+2.*k*t+i*k*t*t);
+ a0 = (2.*t+j*t*t)*g;
+ 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,
+};
diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
index 6557612..0eeef53 100644
--- a/libavfilter/allfilters.c
+++ b/libavfilter/allfilters.c
@@ -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);
diff --git a/libavfilter/version.h b/libavfilter/version.h
index 893ec52..a2c9462 100644
--- a/libavfilter/version.h
+++ b/libavfilter/version.h
@@ -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, \
--
1.9.1
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