[FFmpeg-cvslog] avfilter: add multiband compand filter
Paul B Mahol
git at videolan.org
Fri Nov 17 21:34:16 EET 2017
ffmpeg | branch: master | Paul B Mahol <onemda at gmail.com> | Sat May 13 11:57:42 2017 +0200| [5d7c76566cdd0544a4bda59b520be22bd7ad7f30] | committer: Paul B Mahol
avfilter: add multiband compand filter
Signed-off-by: Paul B Mahol <onemda at gmail.com>
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=5d7c76566cdd0544a4bda59b520be22bd7ad7f30
---
Changelog | 1 +
doc/filters.texi | 16 ++
libavfilter/Makefile | 1 +
libavfilter/af_mcompand.c | 689 ++++++++++++++++++++++++++++++++++++++++++++++
libavfilter/allfilters.c | 1 +
libavfilter/version.h | 4 +-
6 files changed, 710 insertions(+), 2 deletions(-)
diff --git a/Changelog b/Changelog
index d2b5530ad7..119ab678e5 100644
--- a/Changelog
+++ b/Changelog
@@ -15,6 +15,7 @@ version <next>:
- Raw aptX muxer and demuxer
- NVIDIA NVDEC-accelerated H.264, HEVC, VC1 and VP9 hwaccel decoding
- Intel QSV-accelerated overlay filter
+- mcompand audio filter
version 3.4:
diff --git a/doc/filters.texi b/doc/filters.texi
index 4a35c44c7b..5d99437871 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -3270,6 +3270,22 @@ lowpass=c=LFE
@end example
@end itemize
+ at section mcompand
+Multiband Compress or expand the audio's dynamic range.
+
+The input audio is divided into bands using 4th order Linkwitz-Riley IIRs.
+This is akin to the crossover of a loudspeaker, and results in flat frequency
+response when absent compander action.
+
+It accepts the following parameters:
+
+ at table @option
+ at item args
+This option syntax is:
+attack,decay,[attack,decay..] soft-knee points crossover_frequency [delay [initial_volume [gain]]] | attack,decay ...
+For explanation of each item refer to compand filter documentation.
+ at end table
+
@anchor{pan}
@section pan
diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index b7ddcd226d..9acae3ff5b 100644
--- a/libavfilter/Makefile
+++ b/libavfilter/Makefile
@@ -101,6 +101,7 @@ OBJS-$(CONFIG_JOIN_FILTER) += af_join.o
OBJS-$(CONFIG_LADSPA_FILTER) += af_ladspa.o
OBJS-$(CONFIG_LOUDNORM_FILTER) += af_loudnorm.o ebur128.o
OBJS-$(CONFIG_LOWPASS_FILTER) += af_biquads.o
+OBJS-$(CONFIG_MCOMPAND_FILTER) += af_mcompand.o
OBJS-$(CONFIG_PAN_FILTER) += af_pan.o
OBJS-$(CONFIG_REPLAYGAIN_FILTER) += af_replaygain.o
OBJS-$(CONFIG_RESAMPLE_FILTER) += af_resample.o
diff --git a/libavfilter/af_mcompand.c b/libavfilter/af_mcompand.c
new file mode 100644
index 0000000000..02f987a6a8
--- /dev/null
+++ b/libavfilter/af_mcompand.c
@@ -0,0 +1,689 @@
+/*
+ * COpyright (c) 2002 Daniel Pouzzner
+ * 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
+ * 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 multiband 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 CompandSegment {
+ double x, y;
+ double a, b;
+} CompandSegment;
+
+typedef struct CompandT {
+ CompandSegment *segments;
+ int nb_segments;
+ double in_min_lin;
+ double out_min_lin;
+ double curve_dB;
+ double gain_dB;
+} CompandT;
+
+#define N 4
+
+typedef struct PrevCrossover {
+ double in;
+ double out_low;
+ double out_high;
+} PrevCrossover[N * 2];
+
+typedef struct Crossover {
+ PrevCrossover *previous;
+ size_t pos;
+ double coefs[3 *(N+1)];
+} Crossover;
+
+typedef struct CompBand {
+ CompandT transfer_fn;
+ double *attack_rate;
+ double *decay_rate;
+ double *volume;
+ double delay;
+ double topfreq;
+ Crossover filter;
+ AVFrame *delay_buf;
+ size_t delay_size;
+ ptrdiff_t delay_buf_ptr;
+ size_t delay_buf_cnt;
+} CompBand;
+
+typedef struct MCompandContext {
+ const AVClass *class;
+
+ char *args;
+
+ int nb_bands;
+ CompBand *bands;
+ AVFrame *band_buf1, *band_buf2, *band_buf3;
+ int band_samples;
+ size_t delay_buf_size;
+} MCompandContext;
+
+#define OFFSET(x) offsetof(MCompandContext, x)
+#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
+
+static const AVOption mcompand_options[] = {
+ { "args", "set parameters for each band", OFFSET(args), AV_OPT_TYPE_STRING, { .str = "0.005,0.1 6 -47/-40,-34/-34,-17/-33 100 | 0.003,0.05 6 -47/-40,-34/-34,-17/-33 400 | 0.000625,0.0125 6 -47/-40,-34/-34,-15/-33 1600 | 0.0001,0.025 6 -47/-40,-34/-34,-31/-31,-0/-30 6400 | 0,0.025 6 -38/-31,-28/-28,-0/-25 22000" }, 0, 0, A },
+ { NULL }
+};
+
+AVFILTER_DEFINE_CLASS(mcompand);
+
+static av_cold void uninit(AVFilterContext *ctx)
+{
+ MCompandContext *s = ctx->priv;
+ int i;
+
+ av_frame_free(&s->band_buf1);
+ av_frame_free(&s->band_buf2);
+ av_frame_free(&s->band_buf3);
+
+ if (s->bands) {
+ for (i = 0; i < s->nb_bands; i++) {
+ av_freep(&s->bands[i].attack_rate);
+ av_freep(&s->bands[i].decay_rate);
+ av_freep(&s->bands[i].volume);
+ av_freep(&s->bands[i].transfer_fn.segments);
+ av_freep(&s->bands[i].filter.previous);
+ av_frame_free(&s->bands[i].delay_buf);
+ }
+ }
+ av_freep(&s->bands);
+}
+
+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 delimiter)
+{
+ char *p;
+
+ *nb_items = 1;
+ for (p = item_str; *p; p++) {
+ if (*p == delimiter)
+ (*nb_items)++;
+ }
+}
+
+static void update_volume(CompBand *cb, double in, int ch)
+{
+ double delta = in - cb->volume[ch];
+
+ if (delta > 0.0)
+ cb->volume[ch] += delta * cb->attack_rate[ch];
+ else
+ cb->volume[ch] += delta * cb->decay_rate[ch];
+}
+
+static double get_volume(CompandT *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 parse_points(char *points, int nb_points, double radius,
+ CompandT *s, AVFilterContext *ctx)
+{
+ int new_nb_items, num;
+ char *saveptr = NULL;
+ char *p = points;
+ int i;
+
+#define S(x) s->segments[2 * ((x) + 1)]
+ 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");
+ 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=%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;
+ }
+ 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);
+
+ return 0;
+}
+
+static void square_quadratic(double const *x, double *y)
+{
+ y[0] = x[0] * x[0];
+ y[1] = 2 * x[0] * x[1];
+ y[2] = 2 * x[0] * x[2] + x[1] * x[1];
+ y[3] = 2 * x[1] * x[2];
+ y[4] = x[2] * x[2];
+}
+
+static int crossover_setup(AVFilterLink *outlink, Crossover *p, double frequency)
+{
+ double w0 = 2 * M_PI * frequency / outlink->sample_rate;
+ double Q = sqrt(.5), alpha = sin(w0) / (2*Q);
+ double x[9], norm;
+ int i;
+
+ if (w0 > M_PI)
+ return AVERROR(EINVAL);
+
+ x[0] = (1 - cos(w0))/2; /* Cf. filter_LPF in biquads.c */
+ x[1] = 1 - cos(w0);
+ x[2] = (1 - cos(w0))/2;
+ x[3] = (1 + cos(w0))/2; /* Cf. filter_HPF in biquads.c */
+ x[4] = -(1 + cos(w0));
+ x[5] = (1 + cos(w0))/2;
+ x[6] = 1 + alpha;
+ x[7] = -2*cos(w0);
+ x[8] = 1 - alpha;
+
+ for (norm = x[6], i = 0; i < 9; ++i)
+ x[i] /= norm;
+
+ square_quadratic(x , p->coefs);
+ square_quadratic(x + 3, p->coefs + 5);
+ square_quadratic(x + 6, p->coefs + 10);
+
+ p->previous = av_calloc(outlink->channels, sizeof(*p->previous));
+ if (!p->previous)
+ return AVERROR(ENOMEM);
+
+ return 0;
+}
+
+static int config_output(AVFilterLink *outlink)
+{
+ AVFilterContext *ctx = outlink->src;
+ MCompandContext *s = ctx->priv;
+ int ret, ch, i, k, new_nb_items, nb_bands;
+ char *p = s->args, *saveptr = NULL;
+ int max_delay_size = 0;
+
+ count_items(s->args, &nb_bands, '|');
+ s->nb_bands = FFMAX(1, nb_bands);
+
+ s->bands = av_calloc(nb_bands, sizeof(*s->bands));
+ if (!s->bands)
+ return AVERROR(ENOMEM);
+
+ for (i = 0, new_nb_items = 0; i < nb_bands; i++) {
+ int nb_points, nb_attacks, nb_items = 0;
+ char *tstr2, *tstr = av_strtok(p, "|", &saveptr);
+ char *p2, *p3, *saveptr2 = NULL, *saveptr3 = NULL;
+ double radius;
+
+ if (!tstr) {
+ uninit(ctx);
+ return AVERROR(EINVAL);
+ }
+ p = NULL;
+
+ p2 = tstr;
+ count_items(tstr, &nb_items, ' ');
+ tstr2 = av_strtok(p2, " ", &saveptr2);
+ if (!tstr2) {
+ av_log(ctx, AV_LOG_ERROR, "at least one attacks/decays rate is mandatory\n");
+ uninit(ctx);
+ return AVERROR(EINVAL);
+ }
+ p2 = NULL;
+ p3 = tstr2;
+
+ count_items(tstr2, &nb_attacks, ',');
+ if (!nb_attacks || nb_attacks & 1) {
+ av_log(ctx, AV_LOG_ERROR, "number of attacks rate plus decays rate must be even\n");
+ uninit(ctx);
+ return AVERROR(EINVAL);
+ }
+
+ s->bands[i].attack_rate = av_calloc(outlink->channels, sizeof(double));
+ s->bands[i].decay_rate = av_calloc(outlink->channels, sizeof(double));
+ s->bands[i].volume = av_calloc(outlink->channels, sizeof(double));
+ for (k = 0; k < FFMIN(nb_attacks / 2, outlink->channels); k++) {
+ char *tstr3 = av_strtok(p3, ",", &saveptr3);
+
+ p3 = NULL;
+ sscanf(tstr3, "%lf", &s->bands[i].attack_rate[k]);
+ tstr3 = av_strtok(p3, ",", &saveptr3);
+ sscanf(tstr3, "%lf", &s->bands[i].decay_rate[k]);
+
+ if (s->bands[i].attack_rate[k] > 1.0 / outlink->sample_rate) {
+ s->bands[i].attack_rate[k] = 1.0 - exp(-1.0 / (outlink->sample_rate * s->bands[i].attack_rate[k]));
+ } else {
+ s->bands[i].attack_rate[k] = 1.0;
+ }
+
+ if (s->bands[i].decay_rate[k] > 1.0 / outlink->sample_rate) {
+ s->bands[i].decay_rate[k] = 1.0 - exp(-1.0 / (outlink->sample_rate * s->bands[i].decay_rate[k]));
+ } else {
+ s->bands[i].decay_rate[k] = 1.0;
+ }
+ }
+
+ for (ch = k; ch < outlink->channels; ch++) {
+ s->bands[i].attack_rate[ch] = s->bands[i].attack_rate[k - 1];
+ s->bands[i].decay_rate[ch] = s->bands[i].decay_rate[k - 1];
+ }
+
+ tstr2 = av_strtok(p2, " ", &saveptr2);
+ if (!tstr2) {
+ av_log(ctx, AV_LOG_ERROR, "transfer function curve in dB must be set\n");
+ uninit(ctx);
+ return AVERROR(EINVAL);
+ }
+ sscanf(tstr2, "%lf", &s->bands[i].transfer_fn.curve_dB);
+
+ radius = s->bands[i].transfer_fn.curve_dB * M_LN10 / 20.0;
+
+ tstr2 = av_strtok(p2, " ", &saveptr2);
+ if (!tstr2) {
+ av_log(ctx, AV_LOG_ERROR, "transfer points missing\n");
+ uninit(ctx);
+ return AVERROR(EINVAL);
+ }
+
+ count_items(tstr2, &nb_points, ',');
+ s->bands[i].transfer_fn.nb_segments = (nb_points + 4) * 2;
+ s->bands[i].transfer_fn.segments = av_calloc(s->bands[i].transfer_fn.nb_segments,
+ sizeof(CompandSegment));
+ if (!s->bands[i].transfer_fn.segments) {
+ uninit(ctx);
+ return AVERROR(ENOMEM);
+ }
+
+ ret = parse_points(tstr2, nb_points, radius, &s->bands[i].transfer_fn, ctx);
+ if (ret < 0) {
+ av_log(ctx, AV_LOG_ERROR, "transfer points parsing failed\n");
+ uninit(ctx);
+ return ret;
+ }
+
+ tstr2 = av_strtok(p2, " ", &saveptr2);
+ if (!tstr2) {
+ av_log(ctx, AV_LOG_ERROR, "crossover_frequency is missing\n");
+ uninit(ctx);
+ return AVERROR(EINVAL);
+ }
+
+ new_nb_items += sscanf(tstr2, "%lf", &s->bands[i].topfreq) == 1;
+ if (s->bands[i].topfreq < 0 || s->bands[i].topfreq >= outlink->sample_rate / 2) {
+ av_log(ctx, AV_LOG_ERROR, "crossover_frequency should be >=0 and lower than half of sample rate\n");
+ uninit(ctx);
+ return AVERROR(EINVAL);
+ }
+
+ if (s->bands[i].topfreq != 0) {
+ ret = crossover_setup(outlink, &s->bands[i].filter, s->bands[i].topfreq);
+ if (ret < 0) {
+ uninit(ctx);
+ return ret;
+ }
+ }
+
+ tstr2 = av_strtok(p2, " ", &saveptr2);
+ if (tstr2) {
+ sscanf(tstr2, "%lf", &s->bands[i].delay);
+ max_delay_size = FFMAX(max_delay_size, s->bands[i].delay * outlink->sample_rate);
+
+ tstr2 = av_strtok(p2, " ", &saveptr2);
+ if (tstr2) {
+ double initial_volume;
+
+ sscanf(tstr2, "%lf", &initial_volume);
+ initial_volume = pow(10.0, initial_volume / 20);
+
+ for (k = 0; k < outlink->channels; k++) {
+ s->bands[i].volume[k] = initial_volume;
+ }
+
+ tstr2 = av_strtok(p2, " ", &saveptr2);
+ if (tstr2) {
+ sscanf(tstr2, "%lf", &s->bands[i].transfer_fn.gain_dB);
+ }
+ }
+ }
+ }
+ s->nb_bands = new_nb_items;
+
+ for (i = 0; max_delay_size > 0 && i < s->nb_bands; i++) {
+ s->bands[i].delay_buf = ff_get_audio_buffer(outlink, max_delay_size);
+ if (!s->bands[i].delay_buf)
+ return AVERROR(ENOMEM);
+ }
+ s->delay_buf_size = max_delay_size;
+
+ return 0;
+}
+
+#define CONVOLVE _ _ _ _
+
+static void crossover(int ch, Crossover *p,
+ double *ibuf, double *obuf_low,
+ double *obuf_high, size_t len)
+{
+ double out_low, out_high;
+
+ while (len--) {
+ p->pos = p->pos ? p->pos - 1 : N - 1;
+#define _ out_low += p->coefs[j] * p->previous[ch][p->pos + j].in \
+ - p->coefs[2*N+2 + j] * p->previous[ch][p->pos + j].out_low, j++;
+ {
+ int j = 1;
+ out_low = p->coefs[0] * *ibuf;
+ CONVOLVE
+ *obuf_low++ = out_low;
+ }
+#undef _
+#define _ out_high += p->coefs[j+N+1] * p->previous[ch][p->pos + j].in \
+ - p->coefs[2*N+2 + j] * p->previous[ch][p->pos + j].out_high, j++;
+ {
+ int j = 1;
+ out_high = p->coefs[N+1] * *ibuf;
+ CONVOLVE
+ *obuf_high++ = out_high;
+ }
+ p->previous[ch][p->pos + N].in = p->previous[ch][p->pos].in = *ibuf++;
+ p->previous[ch][p->pos + N].out_low = p->previous[ch][p->pos].out_low = out_low;
+ p->previous[ch][p->pos + N].out_high = p->previous[ch][p->pos].out_high = out_high;
+ }
+}
+
+static int mcompand_channel(MCompandContext *c, CompBand *l, double *ibuf, double *obuf, int len, int ch)
+{
+ int i;
+
+ for (i = 0; i < len; i++) {
+ double level_in_lin, level_out_lin, checkbuf;
+ /* Maintain the volume fields by simulating a leaky pump circuit */
+ update_volume(l, fabs(ibuf[i]), ch);
+
+ /* Volume memory is updated: perform compand */
+ level_in_lin = l->volume[ch];
+ level_out_lin = get_volume(&l->transfer_fn, level_in_lin);
+
+ if (c->delay_buf_size <= 0) {
+ checkbuf = ibuf[i] * level_out_lin;
+ obuf[i] = checkbuf;
+ } else {
+ double *delay_buf = (double *)l->delay_buf->extended_data[ch];
+
+ /* FIXME: note that this lookahead algorithm is really lame:
+ the response to a peak is released before the peak
+ arrives. */
+
+ /* because volume application delays differ band to band, but
+ total delay doesn't, the volume is applied in an iteration
+ preceding that in which the sample goes to obuf, except in
+ the band(s) with the longest vol app delay.
+
+ the offset between delay_buf_ptr and the sample to apply
+ vol to, is a constant equal to the difference between this
+ band's delay and the longest delay of all the bands. */
+
+ if (l->delay_buf_cnt >= l->delay_size) {
+ checkbuf =
+ delay_buf[(l->delay_buf_ptr +
+ c->delay_buf_size -
+ l->delay_size) % c->delay_buf_size] * level_out_lin;
+ delay_buf[(l->delay_buf_ptr + c->delay_buf_size -
+ l->delay_size) % c->delay_buf_size] = checkbuf;
+ }
+ if (l->delay_buf_cnt >= c->delay_buf_size) {
+ obuf[i] = delay_buf[l->delay_buf_ptr];
+ } else {
+ l->delay_buf_cnt++;
+ }
+ delay_buf[l->delay_buf_ptr++] = ibuf[i];
+ l->delay_buf_ptr %= c->delay_buf_size;
+ }
+ }
+
+ return 0;
+}
+
+static int filter_frame(AVFilterLink *inlink, AVFrame *in)
+{
+ AVFilterContext *ctx = inlink->dst;
+ AVFilterLink *outlink = ctx->outputs[0];
+ MCompandContext *s = ctx->priv;
+ AVFrame *out, *abuf, *bbuf, *cbuf;
+ int ch, band, i;
+
+ out = ff_get_audio_buffer(outlink, in->nb_samples);
+ if (!out) {
+ av_frame_free(&in);
+ return AVERROR(ENOMEM);
+ }
+
+ if (s->band_samples < in->nb_samples) {
+ av_frame_free(&s->band_buf1);
+ av_frame_free(&s->band_buf2);
+ av_frame_free(&s->band_buf3);
+
+ s->band_buf1 = ff_get_audio_buffer(outlink, in->nb_samples);
+ s->band_buf2 = ff_get_audio_buffer(outlink, in->nb_samples);
+ s->band_buf3 = ff_get_audio_buffer(outlink, in->nb_samples);
+ s->band_samples = in->nb_samples;
+ }
+
+ for (ch = 0; ch < outlink->channels; ch++) {
+ double *a, *dst = (double *)out->extended_data[ch];
+
+ for (band = 0, abuf = in, bbuf = s->band_buf2, cbuf = s->band_buf1; band < s->nb_bands; band++) {
+ CompBand *b = &s->bands[band];
+
+ if (b->topfreq) {
+ crossover(ch, &b->filter, (double *)abuf->extended_data[ch],
+ (double *)bbuf->extended_data[ch], (double *)cbuf->extended_data[ch], in->nb_samples);
+ } else {
+ bbuf = abuf;
+ abuf = cbuf;
+ }
+
+ if (abuf == in)
+ abuf = s->band_buf3;
+ mcompand_channel(s, b, (double *)bbuf->extended_data[ch], (double *)abuf->extended_data[ch], out->nb_samples, ch);
+ a = (double *)abuf->extended_data[ch];
+ for (i = 0; i < out->nb_samples; i++) {
+ dst[i] += a[i];
+ }
+
+ FFSWAP(AVFrame *, abuf, cbuf);
+ }
+ }
+
+ out->pts = in->pts;
+ av_frame_free(&in);
+ return ff_filter_frame(outlink, out);
+}
+
+static int request_frame(AVFilterLink *outlink)
+{
+ AVFilterContext *ctx = outlink->src;
+ int ret;
+
+ ret = ff_request_frame(ctx->inputs[0]);
+
+ return ret;
+}
+
+static const AVFilterPad mcompand_inputs[] = {
+ {
+ .name = "default",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .filter_frame = filter_frame,
+ },
+ { NULL }
+};
+
+static const AVFilterPad mcompand_outputs[] = {
+ {
+ .name = "default",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .request_frame = request_frame,
+ .config_props = config_output,
+ },
+ { NULL }
+};
+
+
+AVFilter ff_af_mcompand = {
+ .name = "mcompand",
+ .description = NULL_IF_CONFIG_SMALL(
+ "Multiband Compress or expand audio dynamic range."),
+ .query_formats = query_formats,
+ .priv_size = sizeof(MCompandContext),
+ .priv_class = &mcompand_class,
+ .uninit = uninit,
+ .inputs = mcompand_inputs,
+ .outputs = mcompand_outputs,
+};
diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
index 3647a111ec..a838309569 100644
--- a/libavfilter/allfilters.c
+++ b/libavfilter/allfilters.c
@@ -112,6 +112,7 @@ static void register_all(void)
REGISTER_FILTER(LADSPA, ladspa, af);
REGISTER_FILTER(LOUDNORM, loudnorm, af);
REGISTER_FILTER(LOWPASS, lowpass, af);
+ REGISTER_FILTER(MCOMPAND, mcompand, af);
REGISTER_FILTER(PAN, pan, af);
REGISTER_FILTER(REPLAYGAIN, replaygain, af);
REGISTER_FILTER(RESAMPLE, resample, af);
diff --git a/libavfilter/version.h b/libavfilter/version.h
index 908dc4938a..33d9ad7bc7 100644
--- a/libavfilter/version.h
+++ b/libavfilter/version.h
@@ -30,8 +30,8 @@
#include "libavutil/version.h"
#define LIBAVFILTER_VERSION_MAJOR 7
-#define LIBAVFILTER_VERSION_MINOR 0
-#define LIBAVFILTER_VERSION_MICRO 101
+#define LIBAVFILTER_VERSION_MINOR 1
+#define LIBAVFILTER_VERSION_MICRO 100
#define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \
LIBAVFILTER_VERSION_MINOR, \
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