/* * 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 "libavutil/avassert.h"#include "libavutil/avstring.h"#include "libavutil/ffmath.h"#include "libavutil/opt.h"#include "libavutil/samplefmt.h"#include "audio.h"#include "avfilter.h"#include "internal.h" typedef struct ChanParam {    double attack;    double decay;    double volume;} ChanParam; typedef struct CompandSegment {    double x, y;    double a, b;} CompandSegment; typedef struct CompandContext {    const AVClass *class;    int nb_segments;    char *attacks, *decays, *points;    CompandSegment *segments;    ChanParam *channels;    double in_min_lin;    double 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|AV_OPT_FLAG_FILTERING_PARAM static const AVOption compand_options[] = {    { "attacks", "set time over which increase of volume is determined", OFFSET(attacks), AV_OPT_TYPE_STRING, { .str = "0" }, 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|1/0" }, 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 }}; AVFILTER_DEFINE_CLASS(compand); 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_DBLP,        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 void count_items(char *item_str, int *nb_items){    char *p;     *nb_items = 1;    for (p = item_str; *p; p++) {        if (*p == ' ' || *p == '|')            (*nb_items)++;    }} static void update_volume(ChanParam *cp, double in){    double delta = in - cp->volume;     if (delta > 0.0)        cp->volume += delta * cp->attack;    else        cp->volume += delta * cp->decay;} static double get_volume(CompandContext *s, double in_lin){    CompandSegment *cs;    double in_log, out_log;    int i;     if (in_lin < s->in_min_lin)        return s->out_min_lin;     in_log = log(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 exp(out_log);} static int compand_nodelay(AVFilterContext *ctx, AVFrame *frame){    CompandContext *s    = ctx->priv;    AVFilterLink *inlink = ctx->inputs[0];    const int channels   = inlink->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 double *src = (double *)frame->extended_data[chan];        double *dst = (double *)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] = src[i] * get_volume(s, cp->volume);        }    }     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 = inlink->channels;    const int nb_samples = frame->nb_samples;    int chan, i, av_uninit(dindex), oindex, av_uninit(count);    AVFrame *out_frame   = NULL;    int err;     if (s->pts == AV_NOPTS_VALUE) {        s->pts = (frame->pts == AV_NOPTS_VALUE) ? 0 : frame->pts;    }     av_assert1(channels > 0); /* would corrupt delay_count and delay_index */     for (chan = 0; chan < channels; chan++) {        AVFrame *delay_frame = s->delay_frame;        const double *src    = (double *)frame->extended_data[chan];        double *dbuf         = (double *)delay_frame->extended_data[chan];        ChanParam *cp        = &s->channels[chan];        double *dst;         count  = s->delay_count;        dindex = s->delay_index;        for (i = 0, oindex = 0; i < nb_samples; i++) {            const double 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 = (double *)out_frame->extended_data[chan];                dst[oindex++] = dbuf[dindex] * get_volume(s, cp->volume);            } else {                count++;            }             dbuf[dindex] = in;            dindex = MOD(dindex + 1, s->delay_samples);        }    }     s->delay_count = count;    s->delay_index = dindex;     av_frame_free(&frame);     if (out_frame) {        err = ff_filter_frame(ctx->outputs[0], out_frame);        return err;    }     return 0;} static int compand_drain(AVFilterLink *outlink){    AVFilterContext *ctx = outlink->src;    CompandContext *s    = ctx->priv;    const int channels   = outlink->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);     av_assert0(channels > 0);    for (chan = 0; chan < channels; chan++) {        AVFrame *delay_frame = s->delay_frame;        double *dbuf = (double *)delay_frame->extended_data[chan];        double *dst = (double *)frame->extended_data[chan];        ChanParam *cp = &s->channels[chan];         dindex = s->delay_index;        for (i = 0; i < frame->nb_samples; i++) {            dst[i] = dbuf[dindex] * get_volume(s, cp->volume);            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    = outlink->channels;    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->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 = av_strtok(p, " |", &saveptr);        if (!tstr) {            uninit(ctx);            return AVERROR(EINVAL);        }        p = NULL;        new_nb_items += sscanf(tstr, "%lf", &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 = av_strtok(p, " |", &saveptr);        if (!tstr) {            uninit(ctx);            return AVERROR(EINVAL);        }        p = NULL;        new_nb_items += sscanf(tstr, "%lf", &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);    }     for (i = nb_decays; i < channels; i++) {        s->channels[i].attack = s->channels[nb_decays - 1].attack;        s->channels[i].decay = s->channels[nb_decays - 1].decay;    } #define S(x) s->segments[2 * ((x) + 1)]    p = s->points;    for (i = 0, new_nb_items = 0; i < nb_points; i++) {        char *tstr = av_strtok(p, " |", &saveptr);        p = NULL;        if (!tstr || sscanf(tstr, "%lf/%lf", &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;         if (fabs(g1 - g2))            continue;        num--;        for (j = --i; j < num; j++)            S(j) = S(j + 1);    }     for (i = 0; i < s->nb_segments; 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; i < s->nb_segments; 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 = hypot(L(2).x - L(4).x, L(2).y - L(4).y);        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 = hypot(L(0).x - L(2).x, L(0).y - L(2).y);        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;    }

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