--- /dev/null
--- /dev/null
++/*
++ * Copyright (c) 1999 Chris Bagwell
++ * Copyright (c) 1999 Nick Bailey
++ * Copyright (c) 2007 Rob Sykes <robs@users.sourceforge.net>
++ * Copyright (c) 2013 Paul B Mahol
++ * Copyright (c) 2014 Andrew Kelley
++ *
++ * 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 compand filter
++ */
++
++#include <string.h>
++
++#include "libavutil/channel_layout.h"
++#include "libavutil/common.h"
++#include "libavutil/mathematics.h"
++#include "libavutil/mem.h"
++#include "libavutil/opt.h"
++#include "audio.h"
++#include "avfilter.h"
++#include "formats.h"
++#include "internal.h"
++
++typedef struct ChanParam {
++ float attack;
++ float decay;
++ float volume;
++} ChanParam;
++
++typedef struct CompandSegment {
++ float x, y;
++ float a, b;
++} CompandSegment;
++
++typedef struct CompandContext {
++ const AVClass *class;
++ int nb_channels;
++ int nb_segments;
++ char *attacks, *decays, *points;
++ CompandSegment *segments;
++ ChanParam *channels;
++ float in_min_lin;
++ float out_min_lin;
++ double curve_dB;
++ double gain_dB;
++ double initial_volume;
++ double delay;
++ AVFrame *delay_frame;
++ int delay_samples;
++ int delay_count;
++ int delay_index;
++ int64_t pts;
++
++ int (*compand)(AVFilterContext *ctx, AVFrame *frame);
++} CompandContext;
++
++#define OFFSET(x) offsetof(CompandContext, x)
++#define A AV_OPT_FLAG_AUDIO_PARAM
++
++static const AVOption compand_options[] = {
++ { "attacks", "set time over which increase of volume is determined", OFFSET(attacks), AV_OPT_TYPE_STRING, { .str = "0.3" }, 0, 0, A },
++ { "decays", "set time over which decrease of volume is determined", OFFSET(decays), AV_OPT_TYPE_STRING, { .str = "0.8" }, 0, 0, A },
++ { "points", "set points of transfer function", OFFSET(points), AV_OPT_TYPE_STRING, { .str = "-70/-70|-60/-20" }, 0, 0, A },
++ { "soft-knee", "set soft-knee", OFFSET(curve_dB), AV_OPT_TYPE_DOUBLE, { .dbl = 0.01 }, 0.01, 900, A },
++ { "gain", "set output gain", OFFSET(gain_dB), AV_OPT_TYPE_DOUBLE, { .dbl = 0 }, -900, 900, A },
++ { "volume", "set initial volume", OFFSET(initial_volume), AV_OPT_TYPE_DOUBLE, { .dbl = 0 }, -900, 0, A },
++ { "delay", "set delay for samples before sending them to volume adjuster", OFFSET(delay), AV_OPT_TYPE_DOUBLE, { .dbl = 0 }, 0, 20, A },
++ { NULL }
++};
++
++static const AVClass compand_class = {
++ .class_name = "compand filter",
++ .item_name = av_default_item_name,
++ .option = compand_options,
++ .version = LIBAVUTIL_VERSION_INT,
++};
++
++static av_cold int init(AVFilterContext *ctx)
++{
++ CompandContext *s = ctx->priv;
++ s->pts = AV_NOPTS_VALUE;
++ return 0;
++}
++
++static av_cold void uninit(AVFilterContext *ctx)
++{
++ CompandContext *s = ctx->priv;
++
++ av_freep(&s->channels);
++ av_freep(&s->segments);
++ av_frame_free(&s->delay_frame);
++}
++
++static int query_formats(AVFilterContext *ctx)
++{
++ AVFilterChannelLayouts *layouts;
++ AVFilterFormats *formats;
++ static const enum AVSampleFormat sample_fmts[] = {
++ AV_SAMPLE_FMT_FLTP,
++ AV_SAMPLE_FMT_NONE
++ };
++
++ layouts = ff_all_channel_layouts();
++ if (!layouts)
++ return AVERROR(ENOMEM);
++ ff_set_common_channel_layouts(ctx, layouts);
++
++ formats = ff_make_format_list(sample_fmts);
++ if (!formats)
++ return AVERROR(ENOMEM);
++ ff_set_common_formats(ctx, formats);
++
++ formats = ff_all_samplerates();
++ if (!formats)
++ return AVERROR(ENOMEM);
++ ff_set_common_samplerates(ctx, formats);
++
++ return 0;
++}
++
++static void count_items(char *item_str, int *nb_items)
++{
++ char *p;
++
++ *nb_items = 1;
++ for (p = item_str; *p; p++) {
++ if (*p == '|')
++ (*nb_items)++;
++ }
++}
++
++static void update_volume(ChanParam *cp, float in)
++{
++ float delta = in - cp->volume;
++
++ if (delta > 0.0)
++ cp->volume += delta * cp->attack;
++ else
++ cp->volume += delta * cp->decay;
++}
++
++static float get_volume(CompandContext *s, float in_lin)
++{
++ CompandSegment *cs;
++ float in_log, out_log;
++ int i;
++
++ if (in_lin < s->in_min_lin)
++ return s->out_min_lin;
++
++ in_log = logf(in_lin);
++
++ for (i = 1; i < s->nb_segments; i++)
++ if (in_log <= s->segments[i].x)
++ break;
++ cs = &s->segments[i - 1];
++ in_log -= cs->x;
++ out_log = cs->y + in_log * (cs->a * in_log + cs->b);
++
++ return expf(out_log);
++}
++
++static int compand_nodelay(AVFilterContext *ctx, AVFrame *frame)
++{
++ CompandContext *s = ctx->priv;
++ AVFilterLink *inlink = ctx->inputs[0];
++ const int channels = s->nb_channels;
++ const int nb_samples = frame->nb_samples;
++ AVFrame *out_frame;
++ int chan, i;
++ int err;
++
++ if (av_frame_is_writable(frame)) {
++ out_frame = frame;
++ } else {
++ out_frame = ff_get_audio_buffer(inlink, nb_samples);
++ if (!out_frame) {
++ av_frame_free(&frame);
++ return AVERROR(ENOMEM);
++ }
++ err = av_frame_copy_props(out_frame, frame);
++ if (err < 0) {
++ av_frame_free(&out_frame);
++ av_frame_free(&frame);
++ return err;
++ }
++ }
++
++ for (chan = 0; chan < channels; chan++) {
++ const float *src = (float *)frame->extended_data[chan];
++ float *dst = (float *)out_frame->extended_data[chan];
++ ChanParam *cp = &s->channels[chan];
++
++ for (i = 0; i < nb_samples; i++) {
++ update_volume(cp, fabs(src[i]));
++
++ dst[i] = av_clipf(src[i] * get_volume(s, cp->volume), -1.0f, 1.0f);
++ }
++ }
++
++ if (frame != out_frame)
++ av_frame_free(&frame);
++
++ return ff_filter_frame(ctx->outputs[0], out_frame);
++}
++
++#define MOD(a, b) (((a) >= (b)) ? (a) - (b) : (a))
++
++static int compand_delay(AVFilterContext *ctx, AVFrame *frame)
++{
++ CompandContext *s = ctx->priv;
++ AVFilterLink *inlink = ctx->inputs[0];
++ const int channels = s->nb_channels;
++ const int nb_samples = frame->nb_samples;
++ int chan, i, dindex = 0, oindex, count = 0;
++ AVFrame *out_frame = NULL;
++ int err;
++
++ if (s->pts == AV_NOPTS_VALUE) {
++ s->pts = (frame->pts == AV_NOPTS_VALUE) ? 0 : frame->pts;
++ }
++
++ for (chan = 0; chan < channels; chan++) {
++ AVFrame *delay_frame = s->delay_frame;
++ const float *src = (float *)frame->extended_data[chan];
++ float *dbuf = (float *)delay_frame->extended_data[chan];
++ ChanParam *cp = &s->channels[chan];
++ float *dst;
++
++ count = s->delay_count;
++ dindex = s->delay_index;
++ for (i = 0, oindex = 0; i < nb_samples; i++) {
++ const float in = src[i];
++ update_volume(cp, fabs(in));
++
++ if (count >= s->delay_samples) {
++ if (!out_frame) {
++ out_frame = ff_get_audio_buffer(inlink, nb_samples - i);
++ if (!out_frame) {
++ av_frame_free(&frame);
++ return AVERROR(ENOMEM);
++ }
++ err = av_frame_copy_props(out_frame, frame);
++ if (err < 0) {
++ av_frame_free(&out_frame);
++ av_frame_free(&frame);
++ return err;
++ }
++ out_frame->pts = s->pts;
++ s->pts += av_rescale_q(nb_samples - i,
++ (AVRational){ 1, inlink->sample_rate },
++ inlink->time_base);
++ }
++
++ dst = (float *)out_frame->extended_data[chan];
++ dst[oindex++] = av_clipf(dbuf[dindex] *
++ get_volume(s, cp->volume), -1.0f, 1.0f);
++ } else {
++ count++;
++ }
++
++ dbuf[dindex] = in;
++ dindex = MOD(dindex + 1, s->delay_samples);
++ }
++ }
++
++ s->delay_count = count;
++ s->delay_index = dindex;
++
++ av_frame_free(&frame);
++ return out_frame ? ff_filter_frame(ctx->outputs[0], out_frame) : 0;
++}
++
++static int compand_drain(AVFilterLink *outlink)
++{
++ AVFilterContext *ctx = outlink->src;
++ CompandContext *s = ctx->priv;
++ const int channels = s->nb_channels;
++ AVFrame *frame = NULL;
++ int chan, i, dindex;
++
++ /* 2048 is to limit output frame size during drain */
++ frame = ff_get_audio_buffer(outlink, FFMIN(2048, s->delay_count));
++ if (!frame)
++ return AVERROR(ENOMEM);
++ frame->pts = s->pts;
++ s->pts += av_rescale_q(frame->nb_samples,
++ (AVRational){ 1, outlink->sample_rate }, outlink->time_base);
++
++ for (chan = 0; chan < channels; chan++) {
++ AVFrame *delay_frame = s->delay_frame;
++ float *dbuf = (float *)delay_frame->extended_data[chan];
++ float *dst = (float *)frame->extended_data[chan];
++ ChanParam *cp = &s->channels[chan];
++
++ dindex = s->delay_index;
++ for (i = 0; i < frame->nb_samples; i++) {
++ dst[i] = av_clipf(dbuf[dindex] * get_volume(s, cp->volume),
++ -1.0f, 1.0f);
++ dindex = MOD(dindex + 1, s->delay_samples);
++ }
++ }
++ s->delay_count -= frame->nb_samples;
++ s->delay_index = dindex;
++
++ return ff_filter_frame(outlink, frame);
++}
++
++static int config_output(AVFilterLink *outlink)
++{
++ AVFilterContext *ctx = outlink->src;
++ CompandContext *s = ctx->priv;
++ const int sample_rate = outlink->sample_rate;
++ double radius = s->curve_dB * M_LN10 / 20.0;
++ char *p, *saveptr = NULL;
++ const int channels =
++ av_get_channel_layout_nb_channels(outlink->channel_layout);
++ int nb_attacks, nb_decays, nb_points;
++ int new_nb_items, num;
++ int i;
++ int err;
++
++
++ count_items(s->attacks, &nb_attacks);
++ count_items(s->decays, &nb_decays);
++ count_items(s->points, &nb_points);
++
++ if (channels <= 0) {
++ av_log(ctx, AV_LOG_ERROR, "Invalid number of channels: %d\n", channels);
++ return AVERROR(EINVAL);
++ }
++
++ if (nb_attacks > channels || nb_decays > channels) {
++ av_log(ctx, AV_LOG_ERROR,
++ "Number of attacks/decays bigger than number of channels.\n");
++ return AVERROR(EINVAL);
++ }
++
++ uninit(ctx);
++
++ s->nb_channels = channels;
++ s->channels = av_mallocz_array(channels, sizeof(*s->channels));
++ s->nb_segments = (nb_points + 4) * 2;
++ s->segments = av_mallocz_array(s->nb_segments, sizeof(*s->segments));
++
++ if (!s->channels || !s->segments) {
++ uninit(ctx);
++ return AVERROR(ENOMEM);
++ }
++
++ p = s->attacks;
++ for (i = 0, new_nb_items = 0; i < nb_attacks; i++) {
++ char *tstr = strtok_r(p, "|", &saveptr);
++ p = NULL;
++ new_nb_items += sscanf(tstr, "%f", &s->channels[i].attack) == 1;
++ if (s->channels[i].attack < 0) {
++ uninit(ctx);
++ return AVERROR(EINVAL);
++ }
++ }
++ nb_attacks = new_nb_items;
++
++ p = s->decays;
++ for (i = 0, new_nb_items = 0; i < nb_decays; i++) {
++ char *tstr = strtok_r(p, "|", &saveptr);
++ p = NULL;
++ new_nb_items += sscanf(tstr, "%f", &s->channels[i].decay) == 1;
++ if (s->channels[i].decay < 0) {
++ uninit(ctx);
++ return AVERROR(EINVAL);
++ }
++ }
++ nb_decays = new_nb_items;
++
++ if (nb_attacks != nb_decays) {
++ av_log(ctx, AV_LOG_ERROR,
++ "Number of attacks %d differs from number of decays %d.\n",
++ nb_attacks, nb_decays);
++ uninit(ctx);
++ return AVERROR(EINVAL);
++ }
++
++#define S(x) s->segments[2 * ((x) + 1)]
++ p = s->points;
++ for (i = 0, new_nb_items = 0; i < nb_points; i++) {
++ char *tstr = strtok_r(p, "|", &saveptr);
++ p = NULL;
++ if (sscanf(tstr, "%f/%f", &S(i).x, &S(i).y) != 2) {
++ av_log(ctx, AV_LOG_ERROR,
++ "Invalid and/or missing input/output value.\n");
++ uninit(ctx);
++ return AVERROR(EINVAL);
++ }
++ if (i && S(i - 1).x > S(i).x) {
++ av_log(ctx, AV_LOG_ERROR,
++ "Transfer function input values must be increasing.\n");
++ uninit(ctx);
++ return AVERROR(EINVAL);
++ }
++ S(i).y -= S(i).x;
++ av_log(ctx, AV_LOG_DEBUG, "%d: x=%f y=%f\n", i, S(i).x, S(i).y);
++ new_nb_items++;
++ }
++ num = new_nb_items;
++
++ /* Add 0,0 if necessary */
++ if (num == 0 || S(num - 1).x)
++ num++;
++
++#undef S
++#define S(x) s->segments[2 * (x)]
++ /* Add a tail off segment at the start */
++ S(0).x = S(1).x - 2 * s->curve_dB;
++ S(0).y = S(1).y;
++ num++;
++
++ /* Join adjacent colinear segments */
++ for (i = 2; i < num; i++) {
++ double g1 = (S(i - 1).y - S(i - 2).y) * (S(i - 0).x - S(i - 1).x);
++ double g2 = (S(i - 0).y - S(i - 1).y) * (S(i - 1).x - S(i - 2).x);
++ int j;
++
++ /* here we purposefully lose precision so that we can compare floats */
++ if (fabs(g1 - g2))
++ continue;
++ num--;
++ for (j = --i; j < num; j++)
++ S(j) = S(j + 1);
++ }
++
++ for (i = 0; !i || s->segments[i - 2].x; i += 2) {
++ s->segments[i].y += s->gain_dB;
++ s->segments[i].x *= M_LN10 / 20;
++ s->segments[i].y *= M_LN10 / 20;
++ }
++
++#define L(x) s->segments[i - (x)]
++ for (i = 4; s->segments[i - 2].x; i += 2) {
++ double x, y, cx, cy, in1, in2, out1, out2, theta, len, r;
++
++ L(4).a = 0;
++ L(4).b = (L(2).y - L(4).y) / (L(2).x - L(4).x);
++
++ L(2).a = 0;
++ L(2).b = (L(0).y - L(2).y) / (L(0).x - L(2).x);
++
++ theta = atan2(L(2).y - L(4).y, L(2).x - L(4).x);
++ len = sqrt(pow(L(2).x - L(4).x, 2.) + pow(L(2).y - L(4).y, 2.));
++ r = FFMIN(radius, len);
++ L(3).x = L(2).x - r * cos(theta);
++ L(3).y = L(2).y - r * sin(theta);
++
++ theta = atan2(L(0).y - L(2).y, L(0).x - L(2).x);
++ len = sqrt(pow(L(0).x - L(2).x, 2.) + pow(L(0).y - L(2).y, 2.));
++ r = FFMIN(radius, len / 2);
++ x = L(2).x + r * cos(theta);
++ y = L(2).y + r * sin(theta);
++
++ cx = (L(3).x + L(2).x + x) / 3;
++ cy = (L(3).y + L(2).y + y) / 3;
++
++ L(2).x = x;
++ L(2).y = y;
++
++ in1 = cx - L(3).x;
++ out1 = cy - L(3).y;
++ in2 = L(2).x - L(3).x;
++ out2 = L(2).y - L(3).y;
++ L(3).a = (out2 / in2 - out1 / in1) / (in2 - in1);
++ L(3).b = out1 / in1 - L(3).a * in1;
++ }
++ L(3).x = 0;
++ L(3).y = L(2).y;
++
++ s->in_min_lin = exp(s->segments[1].x);
++ s->out_min_lin = exp(s->segments[1].y);
++
++ for (i = 0; i < channels; i++) {
++ ChanParam *cp = &s->channels[i];
++
++ if (cp->attack > 1.0 / sample_rate)
++ cp->attack = 1.0 - exp(-1.0 / (sample_rate * cp->attack));
++ else
++ cp->attack = 1.0;
++ if (cp->decay > 1.0 / sample_rate)
++ cp->decay = 1.0 - exp(-1.0 / (sample_rate * cp->decay));
++ else
++ cp->decay = 1.0;
++ cp->volume = pow(10.0, s->initial_volume / 20);
++ }
++
++ s->delay_samples = s->delay * sample_rate;
++ if (s->delay_samples <= 0) {
++ s->compand = compand_nodelay;
++ return 0;
++ }
++
++ s->delay_frame = av_frame_alloc();
++ if (!s->delay_frame) {
++ uninit(ctx);
++ return AVERROR(ENOMEM);
++ }
++
++ s->delay_frame->format = outlink->format;
++ s->delay_frame->nb_samples = s->delay_samples;
++ s->delay_frame->channel_layout = outlink->channel_layout;
++
++ err = av_frame_get_buffer(s->delay_frame, 32);
++ if (err)
++ return err;
++
++ s->compand = compand_delay;
++ return 0;
++}
++
++static int filter_frame(AVFilterLink *inlink, AVFrame *frame)
++{
++ AVFilterContext *ctx = inlink->dst;
++ CompandContext *s = ctx->priv;
++
++ return s->compand(ctx, frame);
++}
++
++static int request_frame(AVFilterLink *outlink)
++{
++ AVFilterContext *ctx = outlink->src;
++ CompandContext *s = ctx->priv;
++ int ret;
++
++ ret = ff_request_frame(ctx->inputs[0]);
++
++ if (ret == AVERROR_EOF && s->delay_count)
++ ret = compand_drain(outlink);
++
++ return ret;
++}
++
++static const AVFilterPad compand_inputs[] = {
++ {
++ .name = "default",
++ .type = AVMEDIA_TYPE_AUDIO,
++ .filter_frame = filter_frame,
++ },
++ { NULL }
++};
++
++static const AVFilterPad compand_outputs[] = {
++ {
++ .name = "default",
++ .request_frame = request_frame,
++ .config_props = config_output,
++ .type = AVMEDIA_TYPE_AUDIO,
++ },
++ { NULL }
++};
++
++
++AVFilter ff_af_compand_fork = {
++ .name = "compand_fork",
++ .description = NULL_IF_CONFIG_SMALL(
++ "Compress or expand audio dynamic range."),
++ .query_formats = query_formats,
++ .priv_size = sizeof(CompandContext),
++ .priv_class = &compand_class,
++ .init = init,
++ .uninit = uninit,
++ .inputs = compand_inputs,
++ .outputs = compand_outputs,
++};
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