From 5d2cc00dd01911a3ffab746230f0a54eea7957e1 Mon Sep 17 00:00:00 2001 From: Paul B Mahol Date: Mon, 30 Nov 2015 13:36:58 +0100 Subject: [PATCH] avfilter: add audio emphasis filter Signed-off-by: Paul B Mahol --- 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 600ffeaa5d..fbe6128c59 100644 --- a/Changelog +++ b/Changelog @@ -40,6 +40,7 @@ version : - apulsator filter - sidechaingate audio filter - mipsdspr1 option has been renamed to mipsdsp +- aemphasis filter version 2.8: diff --git a/configure b/configure index ea2a74605d..bf13613093 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 * I" + disable $func + check_ld "cc" "$@" < +#include +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 bf299cac12..12082de47d 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 +@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: + +@table @option +@item level_in +Set input gain. + +@item level_out +Set output gain. + +@item mode +Set filter mode. For restoring material use @code{reproduction} mode, otherwise +use @code{production} mode. Default is @code{reproduction} mode. + +@item type +Set filter type. Selects medium. Can be one of the following: + +@table @option +@item col +select Columbia. +@item emi +select EMI. +@item bsi +select BSI (78RPM). +@item riaa +select RIAA. +@item cd +select Compact Disc (CD). +@item 50fm +select 50µs (FM). +@item 75fm +select 75µs (FM). +@item 50kf +select 50µs (FM-KF). +@item 75kf +select 75µs (FM-KF). +@end table +@end table + @section aeval Modify an audio signal according to the specified expressions. diff --git a/libavfilter/Makefile b/libavfilter/Makefile index 740a640789..8884d1d2e5 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 0000000000..4501858fb8 --- /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 + +#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 6557612066..0eeef53309 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 893ec52531..a2c9462110 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, \