/* * Copyright (C) 2001-2010 Krzysztof Foltman, Markus Schmidt, Thor Harald Johansen, Damien Zammit * * 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 */ /** * @file * Audio (Sidechain) Gate filter */ #include "libavutil/audio_fifo.h"#include "libavutil/avassert.h"#include "libavutil/channel_layout.h"#include "libavutil/opt.h"#include "avfilter.h"#include "audio.h"#include "formats.h"#include "hermite.h" typedef struct AudioGateContext {    const AVClass *class;     double level_in;    double level_sc;    double attack;    double release;    double threshold;    double ratio;    double knee;    double makeup;    double range;    int link;    int detection;     double thres;    double knee_start;    double lin_knee_stop;    double knee_stop;    double lin_slope;    double attack_coeff;    double release_coeff;     AVAudioFifo *fifo[2];    int64_t pts;} AudioGateContext; #define OFFSET(x) offsetof(AudioGateContext, x)#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM static const AVOption options[] = {    { "level_in",  "set input level",        OFFSET(level_in),  AV_OPT_TYPE_DOUBLE, {.dbl=1},           0.015625,   64, A },    { "range",     "set max gain reduction", OFFSET(range),     AV_OPT_TYPE_DOUBLE, {.dbl=0.06125},     0, 1, A },    { "threshold", "set threshold",          OFFSET(threshold), AV_OPT_TYPE_DOUBLE, {.dbl=0.125},       0, 1, A },    { "ratio",     "set ratio",              OFFSET(ratio),     AV_OPT_TYPE_DOUBLE, {.dbl=2},           1,  9000, A },    { "attack",    "set attack",             OFFSET(attack),    AV_OPT_TYPE_DOUBLE, {.dbl=20},          0.01, 9000, A },    { "release",   "set release",            OFFSET(release),   AV_OPT_TYPE_DOUBLE, {.dbl=250},         0.01, 9000, A },    { "makeup",    "set makeup gain",        OFFSET(makeup),    AV_OPT_TYPE_DOUBLE, {.dbl=1},           1,   64, A },    { "knee",      "set knee",               OFFSET(knee),      AV_OPT_TYPE_DOUBLE, {.dbl=2.828427125}, 1,    8, A },    { "detection", "set detection",          OFFSET(detection), AV_OPT_TYPE_INT,    {.i64=1},           0,    1, A, "detection" },    {   "peak",    0,                        0,                 AV_OPT_TYPE_CONST,  {.i64=0},           0,    0, A, "detection" },    {   "rms",     0,                        0,                 AV_OPT_TYPE_CONST,  {.i64=1},           0,    0, A, "detection" },    { "link",      "set link",               OFFSET(link),      AV_OPT_TYPE_INT,    {.i64=0},           0,    1, A, "link" },    {   "average", 0,                        0,                 AV_OPT_TYPE_CONST,  {.i64=0},           0,    0, A, "link" },    {   "maximum", 0,                        0,                 AV_OPT_TYPE_CONST,  {.i64=1},           0,    0, A, "link" },    { "level_sc",  "set sidechain gain",     OFFSET(level_sc),  AV_OPT_TYPE_DOUBLE, {.dbl=1},           0.015625,   64, A },    { NULL }}; static int agate_config_input(AVFilterLink *inlink){    AVFilterContext *ctx = inlink->dst;    AudioGateContext *s = ctx->priv;    double lin_threshold = s->threshold;    double lin_knee_sqrt = sqrt(s->knee);    double lin_knee_start;     if (s->detection)        lin_threshold *= lin_threshold;     s->attack_coeff  = FFMIN(1., 1. / (s->attack * inlink->sample_rate / 4000.));    s->release_coeff = FFMIN(1., 1. / (s->release * inlink->sample_rate / 4000.));    s->lin_knee_stop = lin_threshold * lin_knee_sqrt;    lin_knee_start = lin_threshold / lin_knee_sqrt;    s->thres = log(lin_threshold);    s->knee_start = log(lin_knee_start);    s->knee_stop = log(s->lin_knee_stop);     return 0;} // A fake infinity value (because real infinity may break some hosts)#define FAKE_INFINITY (65536.0 * 65536.0) // Check for infinity (with appropriate-ish tolerance)#define IS_FAKE_INFINITY(value) (fabs(value-FAKE_INFINITY) < 1.0) static double output_gain(double lin_slope, double ratio, double thres,                          double knee, double knee_start, double knee_stop,                          double lin_knee_stop, double range){    if (lin_slope < lin_knee_stop) {        double slope = log(lin_slope);        double tratio = ratio;        double gain = 0.;        double delta = 0.;         if (IS_FAKE_INFINITY(ratio))            tratio = 1000.;        gain = (slope - thres) * tratio + thres;        delta = tratio;         if (knee > 1. && slope > knee_start) {            gain = hermite_interpolation(slope, knee_start, knee_stop, ((knee_start - thres) * tratio  + thres), knee_stop, delta, 1.);        }        return FFMAX(range, exp(gain - slope));    }     return 1.;} static void gate(AudioGateContext *s,                 const double *src, double *dst, const double *scsrc,                 int nb_samples, double level_in, double level_sc,                 AVFilterLink *inlink, AVFilterLink *sclink){    const double makeup = s->makeup;    const double attack_coeff = s->attack_coeff;    const double release_coeff = s->release_coeff;    int n, c;     for (n = 0; n < nb_samples; n++, src += inlink->channels, dst += inlink->channels, scsrc += sclink->channels) {        double abs_sample = fabs(scsrc[0] * level_sc), gain = 1.0;         if (s->link == 1) {            for (c = 1; c < sclink->channels; c++)                abs_sample = FFMAX(fabs(scsrc[c] * level_sc), abs_sample);        } else {            for (c = 1; c < sclink->channels; c++)                abs_sample += fabs(scsrc[c] * level_sc);             abs_sample /= sclink->channels;        }         if (s->detection)            abs_sample *= abs_sample;         s->lin_slope += (abs_sample - s->lin_slope) * (abs_sample > s->lin_slope ? attack_coeff : release_coeff);        if (s->lin_slope > 0.0)            gain = output_gain(s->lin_slope, s->ratio, s->thres,                               s->knee, s->knee_start, s->knee_stop,                               s->lin_knee_stop, s->range);         for (c = 0; c < inlink->channels; c++)            dst[c] = src[c] * level_in * gain * makeup;    }} #if CONFIG_AGATE_FILTER #define agate_options optionsAVFILTER_DEFINE_CLASS(agate); static int query_formats(AVFilterContext *ctx){    AVFilterFormats *formats = NULL;    AVFilterChannelLayouts *layouts;    int ret;     if ((ret = ff_add_format(&formats, AV_SAMPLE_FMT_DBL)) < 0)        return ret;    ret = ff_set_common_formats(ctx, formats);    if (ret < 0)        return 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_all_samplerates();    if (!formats)        return AVERROR(ENOMEM);     return ff_set_common_samplerates(ctx, formats);} static int filter_frame(AVFilterLink *inlink, AVFrame *in){    const double *src = (const double *)in->data[0];    AVFilterContext *ctx = inlink->dst;    AVFilterLink *outlink = ctx->outputs[0];    AudioGateContext *s = ctx->priv;    AVFrame *out;    double *dst;     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];     gate(s, src, dst, src, in->nb_samples,         s->level_in, s->level_in, inlink, inlink);     if (out != in)        av_frame_free(&in);    return ff_filter_frame(outlink, out);} static const AVFilterPad inputs[] = {    {        .name         = "default",        .type         = AVMEDIA_TYPE_AUDIO,        .filter_frame = filter_frame,        .config_props = agate_config_input,    },    { NULL }}; static const AVFilterPad outputs[] = {    {        .name = "default",        .type = AVMEDIA_TYPE_AUDIO,    },    { NULL }}; AVFilter ff_af_agate = {    .name           = "agate",    .description    = NULL_IF_CONFIG_SMALL("Audio gate."),    .query_formats  = query_formats,    .priv_size      = sizeof(AudioGateContext),    .priv_class     = &agate_class,    .inputs         = inputs,    .outputs        = outputs,}; #endif /* CONFIG_AGATE_FILTER */ #if CONFIG_SIDECHAINGATE_FILTER #define sidechaingate_options optionsAVFILTER_DEFINE_CLASS(sidechaingate); static int scfilter_frame(AVFilterLink *link, AVFrame *frame){    AVFilterContext *ctx = link->dst;    AudioGateContext *s = ctx->priv;    AVFilterLink *outlink = ctx->outputs[0];    AVFrame *out, *in[2] = { NULL };    double *dst;    int nb_samples;    int i;     for (i = 0; i < 2; i++)        if (link == ctx->inputs[i])            break;    av_assert0(i < 2);    av_audio_fifo_write(s->fifo[i], (void **)frame->extended_data,                        frame->nb_samples);    av_frame_free(&frame);     nb_samples = FFMIN(av_audio_fifo_size(s->fifo[0]), av_audio_fifo_size(s->fifo[1]));    if (!nb_samples)        return 0;     out = ff_get_audio_buffer(outlink, nb_samples);    if (!out)        return AVERROR(ENOMEM);    for (i = 0; i < 2; i++) {        in[i] = ff_get_audio_buffer(ctx->inputs[i], nb_samples);        if (!in[i]) {            av_frame_free(&in[0]);            av_frame_free(&in[1]);            av_frame_free(&out);            return AVERROR(ENOMEM);        }        av_audio_fifo_read(s->fifo[i], (void **)in[i]->data, nb_samples);    }     dst = (double *)out->data[0];    out->pts = s->pts;    s->pts += nb_samples;     gate(s, (double *)in[0]->data[0], dst,         (double *)in[1]->data[0], nb_samples,         s->level_in, s->level_sc,         ctx->inputs[0], ctx->inputs[1]);     av_frame_free(&in[0]);    av_frame_free(&in[1]);     return ff_filter_frame(outlink, out);} static int screquest_frame(AVFilterLink *outlink){    AVFilterContext *ctx = outlink->src;    AudioGateContext *s = ctx->priv;    int i;     /* get a frame on each input */    for (i = 0; i < 2; i++) {        AVFilterLink *inlink = ctx->inputs[i];        if (!av_audio_fifo_size(s->fifo[i]))            return ff_request_frame(inlink);    }     return 0;} static int scquery_formats(AVFilterContext *ctx){    AVFilterFormats *formats;    AVFilterChannelLayouts *layouts = NULL;    static const enum AVSampleFormat sample_fmts[] = {        AV_SAMPLE_FMT_DBL,        AV_SAMPLE_FMT_NONE    };    int ret, i;     if (!ctx->inputs[0]->in_channel_layouts ||        !ctx->inputs[0]->in_channel_layouts->nb_channel_layouts) {        av_log(ctx, AV_LOG_WARNING,               "No channel layout for input 1\n");            return AVERROR(EAGAIN);    }     if ((ret = ff_add_channel_layout(&layouts, ctx->inputs[0]->in_channel_layouts->channel_layouts[0])) < 0 ||        (ret = ff_channel_layouts_ref(layouts, &ctx->outputs[0]->in_channel_layouts)) < 0)        return ret;     for (i = 0; i < 2; i++) {        layouts = ff_all_channel_counts();        if ((ret = ff_channel_layouts_ref(layouts, &ctx->inputs[i]->out_channel_layouts)) < 0)            return ret;    }     formats = ff_make_format_list(sample_fmts);    if ((ret = ff_set_common_formats(ctx, formats)) < 0)        return ret;     formats = ff_all_samplerates();    return ff_set_common_samplerates(ctx, formats);} static int scconfig_output(AVFilterLink *outlink){    AVFilterContext *ctx = outlink->src;    AudioGateContext *s = ctx->priv;     if (ctx->inputs[0]->sample_rate != ctx->inputs[1]->sample_rate) {        av_log(ctx, AV_LOG_ERROR,               "Inputs must have the same sample rate "               "%d for in0 vs %d for in1\n",               ctx->inputs[0]->sample_rate, ctx->inputs[1]->sample_rate);        return AVERROR(EINVAL);    }     outlink->sample_rate = ctx->inputs[0]->sample_rate;    outlink->time_base   = ctx->inputs[0]->time_base;    outlink->channel_layout = ctx->inputs[0]->channel_layout;    outlink->channels = ctx->inputs[0]->channels;     s->fifo[0] = av_audio_fifo_alloc(ctx->inputs[0]->format, ctx->inputs[0]->channels, 1024);    s->fifo[1] = av_audio_fifo_alloc(ctx->inputs[1]->format, ctx->inputs[1]->channels, 1024);    if (!s->fifo[0] || !s->fifo[1])        return AVERROR(ENOMEM);      agate_config_input(ctx->inputs[0]);     return 0;} static av_cold void uninit(AVFilterContext *ctx){    AudioGateContext *s = ctx->priv;     av_audio_fifo_free(s->fifo[0]);    av_audio_fifo_free(s->fifo[1]);} static const AVFilterPad sidechaingate_inputs[] = {    {        .name           = "main",        .type           = AVMEDIA_TYPE_AUDIO,        .filter_frame   = scfilter_frame,    },{        .name           = "sidechain",        .type           = AVMEDIA_TYPE_AUDIO,        .filter_frame   = scfilter_frame,    },    { NULL }}; static const AVFilterPad sidechaingate_outputs[] = {    {        .name          = "default",        .type          = AVMEDIA_TYPE_AUDIO,        .config_props  = scconfig_output,        .request_frame = screquest_frame,    },    { NULL }}; AVFilter ff_af_sidechaingate = {    .name           = "sidechaingate",    .description    = NULL_IF_CONFIG_SMALL("Audio sidechain gate."),    .priv_size      = sizeof(AudioGateContext),    .priv_class     = &sidechaingate_class,    .query_formats  = scquery_formats,    .uninit         = uninit,    .inputs         = sidechaingate_inputs,    .outputs        = sidechaingate_outputs,};#endif  /* CONFIG_SIDECHAINGATE_FILTER */