diff --git a/Changelog b/Changelog index a8ff9b1c6e..cbee5fa078 100644 --- a/Changelog +++ b/Changelog @@ -6,6 +6,7 @@ version - aecho filter - perspective filter ported from libmpcodecs - ffprobe -show_programs option +- compand filter version 2.0: diff --git a/doc/filters.texi b/doc/filters.texi index 0c18446d49..3404716802 100644 --- a/doc/filters.texi +++ b/doc/filters.texi @@ -1176,6 +1176,83 @@ front_center.wav -map '[LFE]' lfe.wav -map '[SL]' side_left.wav -map '[SR]' side_right.wav @end example +@section compand + +Compress or expand audio dynamic range. + +A description of the accepted options follows. + +@table @option +@item attacks +@item decays +Set list of times in seconds for each channel over which the instantaneous +level of the input signal is averaged to determine its volume. +@option{attacks} refers to increase of volume and @option{decays} refers +to decrease of volume. +For most situations, the attack time (response to the audio getting louder) +should be shorter than the decay time because the human ear is more sensitive +to sudden loud audio than sudden soft audio. +Typical value for attack is @code{0.3} seconds and for decay @code{0.8} +seconds. + +@item points +Set list of points for transfer function, specified in dB relative to maximum +possible signal amplitude. +Each key points list need to be defined using the following syntax: +@code{x0/y0 x1/y1 x2/y2 ...}. + +The input values must be in strictly increasing order but the transfer +function does not have to me monotonically rising. +The point @code{0/0} is assumed but may be overridden (by @code{0/out-dBn}). +Typical values for the transfer function are @code{-70/-70 -60/-20}. + +@item soft-knee +Set amount for which the points at where adjacent line segments on the +transfer function meet will be rounded. Defaults is @code{0.01}. + +@item gain +Set additional gain in dB to be applied at all points on the transfer function +and allows easy adjustment of the overall gain. +Default is @code{0}. + +@item volume +Set initial volume in dB to be assumed for each channel when filtering starts. +This permits the user to supply a nominal level initially, so that, +for example, a very large gain is not applied to initial signal levels before +the companding has begun to operate. A typical value for audio which is +initially quiet is -90 dB. Default is @code{0}. + +@item delay +Set delay in seconds. Default is @code{0}. The input audio +is analysed immediately, but audio is delayed before being fed to the +volume adjuster. Specifying a delay approximately equal to the attack/decay +times allows the filter to effectively operate in predictive rather than +reactive mode. +@end table + +@subsection Examples +@itemize +@item +Make music with both quiet and loud passages suitable for listening +in a noisy environment: +@example +compand=.3 .3:1 1:-90/-60 -60/-40 -40/-30 -20/-20:6:0:-90:0.2 +@end example + +@item +Noise-gate for when the noise is at a lower level than the signal: +@example +compand=.1 .1:.2 .2:-900/-900 -50.1/-900 -50/-50:.01:0:-90:.1 +@end example + +@item +Here is another noise-gate, this time for when the noise is at a higher level +than the signal (making it, in some ways, similar to squelch): +@example +compand=.1 .1:.1 .1:-45.1/-45.1 -45/-900 0/-900:.01:45:-90:.1 +@end example +@end itemize + @section earwax Make audio easier to listen to on headphones. diff --git a/libavfilter/Makefile b/libavfilter/Makefile index f54e100731..3751d5481e 100644 --- a/libavfilter/Makefile +++ b/libavfilter/Makefile @@ -84,6 +84,7 @@ OBJS-$(CONFIG_BASS_FILTER) += af_biquads.o OBJS-$(CONFIG_BIQUAD_FILTER) += af_biquads.o OBJS-$(CONFIG_CHANNELMAP_FILTER) += af_channelmap.o OBJS-$(CONFIG_CHANNELSPLIT_FILTER) += af_channelsplit.o +OBJS-$(CONFIG_COMPAND_FILTER) += af_compand.o OBJS-$(CONFIG_EARWAX_FILTER) += af_earwax.o OBJS-$(CONFIG_EBUR128_FILTER) += f_ebur128.o OBJS-$(CONFIG_EQUALIZER_FILTER) += af_biquads.o diff --git a/libavfilter/af_compand.c b/libavfilter/af_compand.c new file mode 100644 index 0000000000..8630cdacac --- /dev/null +++ b/libavfilter/af_compand.c @@ -0,0 +1,515 @@ +/* + * Copyright (c) 1999 Chris Bagwell + * Copyright (c) 1999 Nick Bailey + * Copyright (c) 2007 Rob Sykes + * Copyright (c) 2013 Paul B Mahol + * + * 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/avstring.h" +#include "libavutil/opt.h" +#include "libavutil/samplefmt.h" +#include "avfilter.h" +#include "audio.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; + 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; + uint8_t **delayptrs; + 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=NULL}, 0, 0, A }, + { "decays", "set time over which decrease of volume is determined", OFFSET(decays), AV_OPT_TYPE_STRING, {.str=NULL}, 0, 0, A }, + { "points", "set points of transfer function", OFFSET(points), AV_OPT_TYPE_STRING, {.str=NULL}, 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; + + if (!s->attacks || !s->decays || !s->points) { + av_log(ctx, AV_LOG_ERROR, "Missing attacks and/or decays and/or points.\n"); + return AVERROR(EINVAL); + } + + return 0; +} + +static av_cold void uninit(AVFilterContext *ctx) +{ + CompandContext *s = ctx->priv; + + av_freep(&s->channels); + av_freep(&s->segments); + if (s->delayptrs) + av_freep(&s->delayptrs[0]); + av_freep(&s->delayptrs); +} + +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 == ' ') + (*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++) + if (in_log <= s->segments[i + 1].x) + break; + + cs = &s->segments[i]; + 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; + + if (av_frame_is_writable(frame)) { + out_frame = frame; + } else { + out_frame = ff_get_audio_buffer(inlink, nb_samples); + if (!out_frame) + return AVERROR(ENOMEM); + av_frame_copy_props(out_frame, frame); + } + + for (chan = 0; chan < channels; chan++) { + const double *src = (double *)frame->data[chan]; + double *dst = (double *)out_frame->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, dindex, oindex, count; + AVFrame *out_frame = NULL; + + for (chan = 0; chan < channels; chan++) { + const double *src = (double *)frame->data[chan]; + double *dbuf = (double *)s->delayptrs[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) + return AVERROR(ENOMEM); + av_frame_copy_props(out_frame, frame); + 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->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); + 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 = outlink->channels; + int chan, i, dindex; + AVFrame *frame = NULL; + + 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++) { + double *dbuf = (double *)s->delayptrs[chan]; + double *dst = (double *)frame->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; + int nb_attacks, nb_decays, nb_points; + char *p, *saveptr = NULL; + int new_nb_items, num; + int i; + + count_items(s->attacks, &nb_attacks); + count_items(s->decays, &nb_decays); + count_items(s->points, &nb_points); + + if ((nb_attacks > outlink->channels) || (nb_decays > outlink->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(outlink->channels, sizeof(*s->channels)); + s->segments = av_mallocz_array((nb_points + 4) * 2, sizeof(*s->segments)); + + if (!s->channels || !s->segments) + 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) + 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) + 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); + 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"); + 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"); + return AVERROR(EINVAL); + } + S(i).y -= S(i).x; + av_log(ctx, AV_LOG_DEBUG, "%d: x=%lf y=%lf\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 < outlink->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) { + int ret; + if ((ret = av_samples_alloc_array_and_samples(&s->delayptrs, NULL, + outlink->channels, + s->delay_samples, + outlink->format, 0)) < 0) + return ret; + s->compand = compand_delay; + outlink->flags |= FF_LINK_FLAG_REQUEST_LOOP; + } else { + s->compand = compand_nodelay; + } + 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 && !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 avfilter_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, +}; diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c index bda6e3cc7f..bcebcfcb5b 100644 --- a/libavfilter/allfilters.c +++ b/libavfilter/allfilters.c @@ -80,6 +80,7 @@ void avfilter_register_all(void) REGISTER_FILTER(BIQUAD, biquad, af); REGISTER_FILTER(CHANNELMAP, channelmap, af); REGISTER_FILTER(CHANNELSPLIT, channelsplit, af); + REGISTER_FILTER(COMPAND, compand, af); REGISTER_FILTER(EARWAX, earwax, af); REGISTER_FILTER(EBUR128, ebur128, af); REGISTER_FILTER(EQUALIZER, equalizer, af); diff --git a/libavfilter/version.h b/libavfilter/version.h index 71708bf625..190ea2fdba 100644 --- a/libavfilter/version.h +++ b/libavfilter/version.h @@ -30,8 +30,8 @@ #include "libavutil/avutil.h" #define LIBAVFILTER_VERSION_MAJOR 3 -#define LIBAVFILTER_VERSION_MINOR 81 -#define LIBAVFILTER_VERSION_MICRO 103 +#define LIBAVFILTER_VERSION_MINOR 82 +#define LIBAVFILTER_VERSION_MICRO 100 #define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \ LIBAVFILTER_VERSION_MINOR, \