[FFmpeg-devel] [PATCH] lavfi: add compand filter
Paul B Mahol
onemda at gmail.com
Thu Jul 25 15:24:58 CEST 2013
Signed-off-by: Paul B Mahol <onemda at gmail.com>
---
doc/filters.texi | 48 +++++
libavfilter/Makefile | 1 +
libavfilter/af_compand.c | 514 +++++++++++++++++++++++++++++++++++++++++++++++
libavfilter/allfilters.c | 1 +
4 files changed, 564 insertions(+)
create mode 100644 libavfilter/af_compand.c
diff --git a/doc/filters.texi b/doc/filters.texi
index 0c18446..080c598 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -1176,6 +1176,54 @@ front_center.wav -map '[LFE]' lfe.wav -map '[SL]' side_left.wav -map '[SR]'
side_right.wav
@end example
+ at section compand
+
+Compress or expand the dynamic range of the audio.
+
+A description of the accepted parameters follows.
+
+ at table @option
+ at item attacks
+ at 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.
+ at option{attacks} refer to increase of volume and @option{decays} to decrease of
+volume.
+For most situations, tha attack time (response to the music getting louder)
+should be shorter than the decay time because the human ear is more sensitive
+to sudden loud music than sudden soft music.
+Typical value for attack is 0.3 seconds and for decay 0.8 seconds.
+
+ at item points
+Set list of points for tranfer function, specified in dB relative to maximum possible
+signal amplitued.
+The input values must be in strictly increasing order but the transfer function does
+not have to me monotonically rising. The point 0/0 is assumed but may be overriden
+(by 0\out-dBn). Typical values for the transfer function are @code{-70\-60|-20\0}.
+
+ at item soft-knee
+Set amount for which the points at where adjacent line segments on the transfer function meet will be rounded.
+
+ at 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}.
+
+ at 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 operates. A typical value, for audio which is initially quiet is -90 dB.
+Default is @code{0}.
+
+ at item delay
+Set delay in seconds. Default is @code{0}. The input audio
+is analysed immediately, but is delayed before being fed to the
+volume adjuster. Specifying a delay approximately equal to the attack/decay
+times allow the filter to effectively operate in predictive rather than
+reactive mode.
+ at end table
+
@section earwax
Make audio easier to listen to on headphones.
diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index f54e100..3751d54 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 0000000..e76b3fd
--- /dev/null
+++ b/libavfilter/af_compand.c
@@ -0,0 +1,514 @@
+/*
+ * Copyright (c) 1999 Chris Bagwell
+ * Copyright (c) 1999 Nick Bailey
+ * Copyright (c) 2007 Rob Sykes <robs at users.sourceforge.net>
+ * 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];
+ double volume = get_volume(s, cp->volume);
+
+ for (i = 0; i < nb_samples; i++) {
+ update_volume(cp, fabs(src[i]));
+
+ dst[i] = av_clipd(src[i] * 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];
+ double *dst;
+ ChanParam *cp = &s->channels[chan];
+
+ 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))
+ 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 the dynamic range of the audio."),
+ .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 bda6e3c..bcebcfc 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);
--
1.7.11.2
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