/*
* 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 "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 zr, zi;
freq *= 2.0 * M_PI / sr;
zr = cos(freq);
zi = -sin(freq);
/* |(a0 + a1*z + a2*z^2)/(1 + b1*z + b2*z^2)| */
return hypot(c->a0 + c->a1*zr + c->a2*(zr*zr-zi*zi), c->a1*zi + 2*c->a2*zr*zi) /
hypot(1 + c->b1*zr + c->b2*(zr*zr-zi*zi), c->b1*zi + 2*c->b2*zr*zi);
}
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) {
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);
s->rc[0].use_brickw = 0;
} 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,
};