/* * 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 "filters.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, .unit = "mode" },    { "reproduction",              NULL,            0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, FLAGS, .unit = "mode" },    { "production",                NULL,            0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, FLAGS, .unit = "mode" },    { "type",         "set filter type", OFFSET(type), AV_OPT_TYPE_INT,   {.i64=4}, 0, 8, FLAGS, .unit = "type" },    { "col",                 "Columbia",            0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, FLAGS, .unit = "type" },    { "emi",                      "EMI",            0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, FLAGS, .unit = "type" },    { "bsi",              "BSI (78RPM)",            0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, FLAGS, .unit = "type" },    { "riaa",                    "RIAA",            0, AV_OPT_TYPE_CONST, {.i64=3}, 0, 0, FLAGS, .unit = "type" },    { "cd",         "Compact Disc (CD)",            0, AV_OPT_TYPE_CONST, {.i64=4}, 0, 0, FLAGS, .unit = "type" },    { "50fm",               "50µs (FM)",            0, AV_OPT_TYPE_CONST, {.i64=5}, 0, 0, FLAGS, .unit = "type" },    { "75fm",               "75µs (FM)",            0, AV_OPT_TYPE_CONST, {.i64=6}, 0, 0, FLAGS, .unit = "type" },    { "50kf",            "50µs (FM-KF)",            0, AV_OPT_TYPE_CONST, {.i64=7}, 0, 0, FLAGS, .unit = "type" },    { "75kf",            "75µs (FM-KF)",            0, AV_OPT_TYPE_CONST, {.i64=8}, 0, 0, FLAGS, .unit = "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,    },}; const FFFilter ff_af_aemphasis = {    .p.name        = "aemphasis",    .p.description = NULL_IF_CONFIG_SMALL("Audio emphasis."),    .p.priv_class  = &aemphasis_class,    .p.flags       = AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC |                     AVFILTER_FLAG_SLICE_THREADS,    .priv_size     = sizeof(AudioEmphasisContext),    .uninit        = uninit,    FILTER_INPUTS(avfilter_af_aemphasis_inputs),    FILTER_OUTPUTS(ff_audio_default_filterpad),    FILTER_SINGLE_SAMPLEFMT(AV_SAMPLE_FMT_DBLP),    .process_command = process_command,};