mirror of
https://github.com/FFmpeg/FFmpeg.git
synced 2024-12-12 19:18:44 +02:00
384 lines
12 KiB
C
384 lines
12 KiB
C
/*
|
|
* 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 RIAACurve {
|
|
BiquadCoeffs r1;
|
|
BiquadCoeffs brickw;
|
|
int use_brickw;
|
|
} RIAACurve;
|
|
|
|
typedef struct AudioEmphasisContext {
|
|
const AVClass *class;
|
|
int mode, type;
|
|
double level_in, level_out;
|
|
|
|
RIAACurve rc;
|
|
|
|
AVFrame *w;
|
|
} AudioEmphasisContext;
|
|
|
|
#define OFFSET(x) offsetof(AudioEmphasisContext, x)
|
|
#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_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 void biquad_process(BiquadCoeffs *bq, double *dst, const double *src, int nb_samples,
|
|
double *w, double level_in, double level_out)
|
|
{
|
|
const double a0 = bq->a0;
|
|
const double a1 = bq->a1;
|
|
const double a2 = bq->a2;
|
|
const double b1 = bq->b1;
|
|
const double b2 = bq->b2;
|
|
double w1 = w[0];
|
|
double w2 = w[1];
|
|
|
|
for (int i = 0; i < nb_samples; i++) {
|
|
double n = src[i] * level_in;
|
|
double tmp = n - w1 * b1 - w2 * b2;
|
|
double out = tmp * a0 + w1 * a1 + w2 * a2;
|
|
|
|
w2 = w1;
|
|
w1 = tmp;
|
|
|
|
dst[i] = out * level_out;
|
|
}
|
|
|
|
w[0] = w1;
|
|
w[1] = w2;
|
|
}
|
|
|
|
typedef struct ThreadData {
|
|
AVFrame *in, *out;
|
|
} ThreadData;
|
|
|
|
static int filter_channels(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
|
|
{
|
|
AudioEmphasisContext *s = ctx->priv;
|
|
const double level_out = s->level_out;
|
|
const double level_in = s->level_in;
|
|
ThreadData *td = arg;
|
|
AVFrame *out = td->out;
|
|
AVFrame *in = td->in;
|
|
const int start = (in->ch_layout.nb_channels * jobnr) / nb_jobs;
|
|
const int end = (in->ch_layout.nb_channels * (jobnr+1)) / nb_jobs;
|
|
|
|
for (int ch = start; ch < end; ch++) {
|
|
const double *src = (const double *)in->extended_data[ch];
|
|
double *w = (double *)s->w->extended_data[ch];
|
|
double *dst = (double *)out->extended_data[ch];
|
|
|
|
if (s->rc.use_brickw) {
|
|
biquad_process(&s->rc.brickw, dst, src, in->nb_samples, w + 2, level_in, 1.);
|
|
biquad_process(&s->rc.r1, dst, dst, in->nb_samples, w, 1., level_out);
|
|
} else {
|
|
biquad_process(&s->rc.r1, dst, src, in->nb_samples, w, level_in, level_out);
|
|
}
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
|
|
{
|
|
AVFilterContext *ctx = inlink->dst;
|
|
AVFilterLink *outlink = ctx->outputs[0];
|
|
ThreadData td;
|
|
AVFrame *out;
|
|
|
|
if (av_frame_is_writable(in)) {
|
|
out = in;
|
|
} else {
|
|
out = ff_get_audio_buffer(outlink, in->nb_samples);
|
|
if (!out) {
|
|
av_frame_free(&in);
|
|
return AVERROR(ENOMEM);
|
|
}
|
|
av_frame_copy_props(out, in);
|
|
}
|
|
|
|
td.in = in; td.out = out;
|
|
ff_filter_execute(ctx, filter_channels, &td, NULL,
|
|
FFMIN(inlink->ch_layout.nb_channels, ff_filter_get_nb_threads(ctx)));
|
|
|
|
if (in != out)
|
|
av_frame_free(&in);
|
|
return ff_filter_frame(outlink, out);
|
|
}
|
|
|
|
static inline void set_highshelf_rbj(BiquadCoeffs *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(BiquadCoeffs *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;
|
|
|
|
if (!s->w)
|
|
s->w = ff_get_audio_buffer(inlink, 4);
|
|
if (!s->w)
|
|
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.r1, cfreq, q, 1. / gain, sr);
|
|
else
|
|
set_highshelf_rbj(&s->rc.r1, cfreq, q, gain, sr);
|
|
s->rc.use_brickw = 0;
|
|
} else {
|
|
s->rc.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.r1.a0 = coeffs.a0 * gc;
|
|
s->rc.r1.a1 = coeffs.a1 * gc;
|
|
s->rc.r1.a2 = coeffs.a2 * gc;
|
|
s->rc.r1.b1 = coeffs.b1;
|
|
s->rc.r1.b2 = coeffs.b2;
|
|
}
|
|
|
|
cutfreq = FFMIN(0.45 * sr, 21000.);
|
|
set_lp_rbj(&s->rc.brickw, cutfreq, 0.707, sr, 1.);
|
|
|
|
return 0;
|
|
}
|
|
|
|
static int process_command(AVFilterContext *ctx, const char *cmd, const char *args,
|
|
char *res, int res_len, int flags)
|
|
{
|
|
int ret;
|
|
|
|
ret = ff_filter_process_command(ctx, cmd, args, res, res_len, flags);
|
|
if (ret < 0)
|
|
return ret;
|
|
|
|
return config_input(ctx->inputs[0]);
|
|
}
|
|
|
|
static av_cold void uninit(AVFilterContext *ctx)
|
|
{
|
|
AudioEmphasisContext *s = ctx->priv;
|
|
|
|
av_frame_free(&s->w);
|
|
}
|
|
|
|
static const AVFilterPad avfilter_af_aemphasis_inputs[] = {
|
|
{
|
|
.name = "default",
|
|
.type = AVMEDIA_TYPE_AUDIO,
|
|
.config_props = config_input,
|
|
.filter_frame = filter_frame,
|
|
},
|
|
};
|
|
|
|
static const AVFilterPad avfilter_af_aemphasis_outputs[] = {
|
|
{
|
|
.name = "default",
|
|
.type = AVMEDIA_TYPE_AUDIO,
|
|
},
|
|
};
|
|
|
|
const AVFilter ff_af_aemphasis = {
|
|
.name = "aemphasis",
|
|
.description = NULL_IF_CONFIG_SMALL("Audio emphasis."),
|
|
.priv_size = sizeof(AudioEmphasisContext),
|
|
.priv_class = &aemphasis_class,
|
|
.uninit = uninit,
|
|
FILTER_INPUTS(avfilter_af_aemphasis_inputs),
|
|
FILTER_OUTPUTS(avfilter_af_aemphasis_outputs),
|
|
FILTER_SINGLE_SAMPLEFMT(AV_SAMPLE_FMT_DBLP),
|
|
.process_command = process_command,
|
|
.flags = AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC |
|
|
AVFILTER_FLAG_SLICE_THREADS,
|
|
};
|