/* * 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/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 = "0.3" }, 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 } }; 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 }; 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 == ' ' || *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] = av_clipd(src[i] * get_volume(s, cp->volume), -1, 1); } } 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++] = av_clipd(dbuf[dindex] * get_volume(s, cp->volume), -1, 1); } 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] = av_clipd(dbuf[dindex] * get_volume(s, cp->volume), -1, 1); 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); 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); 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); } #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 (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->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; outlink->flags |= FF_LINK_FLAG_REQUEST_LOOP; 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 = 0; ret = ff_request_frame(ctx->inputs[0]); if (ret == AVERROR_EOF && !ctx->is_disabled && 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 = { .name = "compand", .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, };