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ffmpeg/libavfilter/af_aemphasis.c
Ganesh Ajjanagadde 520a5d33f0 lavfi/af_aemphasis: remove unnecessary complex number usage 
complex is not available on all platforms. Furthermore, it is trivial to
rewrite complex number expressions to real arithmetic, and in fact
sometimes advantageous for performance reasons: by wrapping as a complex,
one forces a particular Cartesian representation that is not necessarily optimal for the purpose.

Configure dependencies also removed, and aemphasis is now available across
all platforms.

Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Ganesh Ajjanagadde <gajjanagadde@gmail.com>
2015-12-23 09:22:59 -08:00

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/*
* 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,
};
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