TODO changelog and version bump
Signed-off-by: Nicolas George <geo...@nsup.org>
---
doc/filters.texi | 48 +++
libavfilter/Makefile | 1 +
libavfilter/af_mbfequalizer.c | 567 ++++++++++++++++++++++++++++++++++
libavfilter/allfilters.c | 1 +
4 files changed, 617 insertions(+)
create mode 100644 libavfilter/af_mbfequalizer.c
I think I'll rewrite the bands parser to allow per-channel options and make
the structure of the code clearer. But I can already send the whole patch
for your consideration.
diff --git a/doc/filters.texi b/doc/filters.texi
index 8c5d3a5760..c6917fad38 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -4383,6 +4383,54 @@
lv2=p=http\\\\://www.openavproductions.com/artyfx#bitta:c=crush=0.3
@end example
@end itemize
+@section mbfequalizer
+Multiband fast equalizer.
+
+As an equalizer, it allows to selectively alter the volume of the different
+frequencies in the audio signal. This particular equalizer does not allow a
+very sharp separation of the frequencies, but that gives it other
+advantages. It is rather fast. It does not add latency. It supports all
+sample formats, and for integer formats it uses bit-exact integer
+arithmetic.
+
+It accepts the following parameters:
+
+@table @option
+@item defs
+Definition of bands and gain. The syntax is:
+"gain/freq/gain[/freq/gain...][|gain/freq/gain...]"
+where @var{freq} is the cut frequency between bands and @var{gain} the
+associated gain.
+Gain and bands for several channels can be defined by separating them with
+the pipe "|" character; if only one definition is given, it is used for all
+channels; otherwise, the number of definitions must match the number of
+input channels.
+Frequencies should be in ascending order. Gain greater than 1 or negative
+may result in overflows and ugly noise or other strange results.
+
+@item global_gain
+Gain applied to all channels and all bands. Can be greater than 1 without
+risk; in case of overflow, soft clipping is applied.
+
+@item keep_all_bands
+By default, bands beyond @var{sample_rate}/4 are discarded. If this option
+is enabled, they are kept; it can result in strange effects.
+@end table
+
+A Gnuplot formula mapping the frequency to the gain is printed at verbose
+log level. It can be used to check the result, proably with
+"set logscale x; db(x) = 20*log(x)/log(10); plot [20:2000] db(...)".
+
+@subsection Examples
+
+Assuming 3.1 input, reduce high frequencies on left and right, keep a narrow
+band on center and dampen bass on LFE. All gains were lowered by 5dB, then
+raised again using global gain, and the -2.8dB for the middle band was found
+by looking at the graph.
+@example
+mbfequalizer=-5dB/7040/-15dB|-5dB/7040/-15dB|0/220/-2.8dB/880/0|-10dB/110/-5dB:global_gain=+5dB
+@end example
+
@section mcompand
Multiband Compress or expand the audio's dynamic range.
diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index 37d4eee858..f5690a50db 100644
--- a/libavfilter/Makefile
+++ b/libavfilter/Makefile
@@ -124,6 +124,7 @@ OBJS-$(CONFIG_LOWPASS_FILTER) += af_biquads.o
OBJS-$(CONFIG_LOWSHELF_FILTER) += af_biquads.o
OBJS-$(CONFIG_LV2_FILTER) += af_lv2.o
OBJS-$(CONFIG_MCOMPAND_FILTER) += af_mcompand.o
+OBJS-$(CONFIG_MBFEQUALIZER_FILTER) += af_mbfequalizer.o
OBJS-$(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_mbfequalizer.c b/libavfilter/af_mbfequalizer.c
new file mode 100644
index 0000000000..aadd3cb3da
--- /dev/null
+++ b/libavfilter/af_mbfequalizer.c
@@ -0,0 +1,567 @@
+/*
+ * Copyright (c) 2019 Nicolas George <geo...@nsup.org>
+ *
+ * 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
+ */
+
+/*
+ * Low-pass-high-pass equalizer using a simple sliding weighed average.
+ *
+ * Input signal: u(t)
+ * Low freqs: a(t)
+ * High freqs: b(t)
+ *
+ * a(t) = (1-k) a(t-1) + k u(t)
+ * b(t) = u(t) - a(t)
+ *
+ * F = sample frequency; f = input frequency; ω = 2πf/F
+ * For u(t) = exp(iωt),
+ * a(t) = k/(1-exp(-iω)(1-k)) exp(iwt)
+ * b(t) = (1-k)(1-exp(-iω))/(1-exp(-iω)(1-k)) exp(iwt)
+ * hi/low = (1-k)(1-exp(-iω)) / k
+ * R(k,w) = (1-k)*(1-exp(-I*w)) / k
+ *
+ * Cut when |hi/low| = 1, i.e. |1-exp(iω)| = k/(1-k)
+ * Triangle 0, 1, exp(iω) has sides 1, 1, k/(1-k)
+ * w = 2 asin(1/2 k/(1-k))
+ * k/(1-k) = 2 sin(ω/2)
+ * k = 2 sin(ω/2) / (1 + 2 sin(ω/2))
+ *
+ * Gnuplot script: response for cut frequency 110 Hz and 440 Hz.
+ * I = {0,1}
+ * F = 48000
+ * w(f) = 2*pi*f/F
+ * db(x) = 20*log(x)/log(10)
+ * set samples 1000
+ * set grid
+ * set logscale x 2
+ * k(f) = 2*sin(w(f)/2) / (1+2*sin(w(f)/2))
+ * d(f, x) = abs(exp(I*w(x))-(1-k(f)))
+ * al(f, x) = k(f) / d(f, x)
+ * ah(f, x) = abs((1-k(f))*(1-exp(I*w(x)))) / d(f, x)
+ * plot [20:16000] [-20:0] db(al(110, x)), db(ah(110, x)), db(al(880, x)),
db(ah(880, x))
+ *
+ * Approximation without sin():
+ * ω - ω³/6 ≤ sin(ω) ≤ ω - ω³/6 + ω⁵/120
+ * ω - ω³/24 ≤ 2sin(ω/2) ≤ ω - ω³/24 + ω⁵/1920
+ * Accurate enough with 0 ≤ ω ≤ π/2.
+ *
+ * Clip to [-A;A] with soft decay on [A(1-k); +∞[:
+ * s(x) = (Ak)² / (x - A(1-2k))
+ */
+
+/*
+
+aevalsrc=sin(13.75*4.35*exp(0.7*t)):d=10
+arealtime,asplit=3[a][b][c];
+[a]showwaves=s=1764x128[ap];
+[b]mbfequalizer=1:0,showwaves=s=1764x128[bp];
+[c]mbfequalizer=0:1,showwaves=s=1764x128[cp];
+[ap][bp][cp]vstack=3
+
+*/
+
+#include "libavutil/avassert.h"
+#include "libavutil/bprint.h"
+#include "libavutil/eval.h"
+#include "libavutil/opt.h"
+#include "avfilter.h"
+#include "audio.h"
+#include "internal.h"
+
+typedef struct EqBand {
+ int k; /* k(2πf) */
+ int g; /* gain of the high band */
+} EqBand;
+
+typedef struct EqContext EqContext;
+struct EqContext {
+ const AVClass *class;
+ char *defs;
+ EqBand *bands;
+ void *vals;
+ void (*process_samples)(EqContext *s, unsigned channels, unsigned samples,
+ AVFrame *frame, AVFrame *frame_out);
+ double global_gain_opt;
+ unsigned channels;
+ int global_gain;
+ int keep_all_bands;
+};
+
+#define FPOINT 15
+#define SOFTCLIP_SHIFT 2 /* k = 1/(1<<SHIFT) */
+
+#if 1&&HAVE_FAST_64BIT
+
+static inline int fpmul_int(int a, int b)
+{
+ return ((int64_t)a * b) >> FPOINT;
+}
+
+#else
+
+static inline int fpmul_int(int a, int b)
+{
+#define MASK ((1 << FPOINT) - 1)
+ return (((a >> FPOINT) * (b >> FPOINT)) << FPOINT)
+ + (a >> FPOINT) * (b & MASK)
+ + (b >> FPOINT) * (a & MASK)
+ + (((a & MASK) * (b & MASK)) >> FPOINT);
+#undef MASK
+}
+
+#endif
+
+static int64_t fpmul_int64_t(int a, int64_t b)
+{
+ uint64_t a0 = a & 0xFFFFFFFF;
+ uint64_t b0 = b & 0xFFFFFFFF;
+ /*int64_t a1 = a >> 32;*/ /* a1 = 0 */
+ int64_t b1 = b >> 32;
+ uint64_t r0 = a0 * b0;
+ int64_t r1a = (int64_t)a0 * b1;
+ /*int64_t r1b = (int64_t)b0 * a1;*/
+ int64_t r2 = 0 /*a1 * b1*/;
+ r2 += (r1a >> 32) /*+ (r1b >> 32)*/;
+ r1a &= 0xFFFFFFFF;
+ /*r1b &= 0xFFFFFFFF;*/
+ r2 += (r1a /*+ r1b*/ + (r0 >> 32)) >> 32;
+ r0 += (r1a /*+ r1b*/) << 32;
+ return (int64_t)(r0 >> FPOINT) + (r2 << (64 - FPOINT));
+}
+
+static inline float fpmul_float(int a, float b)
+{
+ return a * b / (1 << FPOINT);
+}
+
+static inline double fpmul_double(int a, double b)
+{
+ return a * b / (1 << FPOINT);
+}
+
+static int approx_factor(int f, int sf)
+{
+ /* 31089 / 151 =~ 2*Pi*32768 / 1000 */
+ int w = av_rescale(f, 31089, 151 * sf);
+ int s = w - fpmul_int(w, fpmul_int(w, w)) / 24;
+ int r = (s << FPOINT) / ((1 << FPOINT) + s);
+ return r;
+}
+
+static void reorder_bands(EqBand *bands, size_t nb_bands, unsigned sample_rate)
+{
+ size_t i, j;
+
+ if (nb_bands == 1)
+ return;
+ /* bands arrive in ascending order with kf = gain below,
+ and k = 0 on the final band;
+ we want them in descending order with kf = gain above;
+ therefore, reverse the order of k except the final 0,
+ and reverse the order of kf including the final 0 */
+ for (i = 0, j = nb_bands - 2; i < j; i++, j--)
+ FFSWAP(int, bands[i].k, bands[j].k);
+ for (i = 0, j = nb_bands - 1; i < j; i++, j--)
+ FFSWAP(int, bands[i].g, bands[j].g);
+ for (i = 0; i < nb_bands - 1; i++)
+ bands[i].k = approx_factor(bands[i].k, sample_rate);
+}
+
+static void gain_to_bprint(AVBPrint *bp, unsigned rate, EqBand *bands)
+{
+ double k = bands->k / (double)(1 << FPOINT);
+ double g = bands->g / (double)(1 << FPOINT);
+
+ if (bands->k == 0) {
+ av_bprintf(bp, "%.5f", g);
+ return;
+ }
+ av_bprintf(bp, "(%.5f*%.5f*(1-exp(-{0,1}*2*pi*x/%d))+%.5f*", g, 1 - k,
rate, k);
+ gain_to_bprint(bp, rate, bands + 1);
+ av_bprintf(bp, ")");
+ av_bprintf(bp, "/(1-exp(-{0,1}*2*pi*x/%d)*%.5f)", rate, 1 - k);
+}
+
+static void print_gain(AVFilterContext *ctx, unsigned channel, EqBand *bands)
+{
+ EqContext *s = ctx->priv;
+ AVBPrint bp;
+ unsigned rate = ctx->outputs[0]->sample_rate;
+
+ av_bprint_init(&bp, 0, AV_BPRINT_SIZE_AUTOMATIC);
+ if (s->global_gain != 1 << FPOINT)
+ av_bprintf(&bp, "%.5f*", s->global_gain / (double)(1 << FPOINT));
+ av_bprintf(&bp, "abs(");
+ gain_to_bprint(&bp, rate, bands);
+ av_bprintf(&bp, ")");
+ if (av_bprint_is_complete(&bp))
+ av_log(ctx, AV_LOG_VERBOSE, "Channel %d: gain: %s\n", channel, bp.str);
+ else
+ av_log(ctx, AV_LOG_WARNING, "Not enough memory to build gain formula;
no consequence on output.\n");
+ av_bprint_finalize(&bp, NULL);
+}
+
+static inline int unpack_sample_uint8_t(uint8_t s)
+{
+ return ((int)s - 0x80) << FPOINT;
+}
+
+static inline int unpack_sample_int16_t(int16_t s)
+{
+ return ((int)s) << FPOINT;
+}
+
+static inline int64_t unpack_sample_int32_t(int32_t s)
+{
+ return ((int64_t)s) << FPOINT;
+}
+
+static inline float unpack_sample_float(float s)
+{
+ return s;
+}
+
+static inline double unpack_sample_double(double s)
+{
+ return s;
+}
+
+#define SOFTCLIP_PLUS(A, D, x) (A - D * D / ((x) - (A - 2 * D)))
+#define SOFTCLIP_INT(A, D, x) ((x) < -A + D ? -SOFTCLIP_PLUS(A, D, -(x)) : \
+ (x) > +A - D ? +SOFTCLIP_PLUS(A, D, +(x)) : (x))
+#define SOFTCLIP(A, x) SOFTCLIP_INT(A, (A / 8), x)
+
+static inline uint8_t pack_sample_uint8_t(int gain, int v)
+{
+ v >>= FPOINT;
+ v = fpmul_int(gain, v);
+ return 0x80 + SOFTCLIP(((int64_t)0x80), v);
+}
+
+static inline int16_t pack_sample_int16_t(int gain, int v)
+{
+ v >>= FPOINT;
+ v = fpmul_int(gain, v);
+ return SOFTCLIP(0x8000, v);
+}
+
+static inline int32_t pack_sample_int32_t(int gain, int64_t v)
+{
+ v >>= FPOINT;
+ v = fpmul_int64_t(gain, v);
+ return SOFTCLIP(((int64_t)0x80000000), v);
+}
+
+static inline float pack_sample_float(int gain, float v)
+{
+ v = fpmul_float(gain, v);
+ return SOFTCLIP(1.0, v);
+}
+
+static inline double pack_sample_double(int gain, double v)
+{
+ v = fpmul_double(gain, v);
+ return SOFTCLIP(1.0, v);
+}
+
+#define PROCESS_SAMPLE(type_smp, type_calc) do { \
+ type_calc iv = unpack_sample_##type_smp(*(samples_in++)); \
+ type_calc outval = 0; \
+ for (; band->k; band++) { \
+ type_calc ov1 = *val += fpmul_##type_calc(band->k, iv - *val); \
+ type_calc ov2 = iv - ov1; \
+ val++; \
+ outval += fpmul_##type_calc(band->g, ov2); \
+ iv = ov1; \
+ } \
+ outval += fpmul_##type_calc(band->g, iv); \
+ band++; \
+ *(samples_out++) = pack_sample_##type_smp(s->global_gain, outval); \
+} while (0);
+
+#define DEFINE_PROCESS_SAMPLES(fmt, type_smp, type_calc) \
+static void process_samples_##fmt(EqContext *s, unsigned channels, unsigned
samples, \
+ AVFrame *frame, AVFrame *frame_out) \
+{ \
+ type_smp *samples_in = (type_smp *)frame->data[0]; \
+ type_smp *samples_out = (type_smp *)frame_out->data[0]; \
+ unsigned sample, ch; \
+ for (sample = 0; sample < samples; sample++) { \
+ type_calc *val = s->vals; \
+ EqBand *band = s->bands; \
+ for (ch = 0; ch < channels; ch++) { \
+ if (s->channels == 1) \
+ band = s->bands; \
+ PROCESS_SAMPLE(type_smp, type_calc); \
+ } \
+ } \
+} \
+static void process_samples_##fmt##p(EqContext *s, unsigned channels, unsigned
samples, \
+ AVFrame *frame, AVFrame *frame_out) \
+{ \
+ EqBand *band0, *band; \
+ unsigned sample, ch; \
+ type_calc *val0, *val; \
+ band0 = s->bands; \
+ val0 = s->vals; \
+ for (ch = 0; ch < channels; ch++) { \
+ type_smp *samples_in = (type_smp *)frame->data[ch]; \
+ type_smp *samples_out = (type_smp *)frame_out->data[ch]; \
+ for (sample = 0; sample < samples; sample++) { \
+ band = band0; \
+ val = val0; \
+ PROCESS_SAMPLE(type_smp, type_calc); \
+ } \
+ if (s->channels > 1) \
+ band0 = band; \
+ val0 = val; \
+ } \
+}
+
+DEFINE_PROCESS_SAMPLES(u8, uint8_t, int)
+DEFINE_PROCESS_SAMPLES(s16, int16_t, int)
+DEFINE_PROCESS_SAMPLES(s32, int32_t, int64_t)
+DEFINE_PROCESS_SAMPLES(flt, float, float)
+DEFINE_PROCESS_SAMPLES(dbl, double, double)
+
+static int filter_frame(AVFilterLink *link, AVFrame *frame)
+{
+ AVFilterContext *ctx = link->dst;
+ EqContext *s = ctx->priv;
+ AVFrame *frame_out;
+
+ if (av_frame_is_writable(frame)) {
+ frame_out = frame;
+ } else {
+ frame_out = ff_get_audio_buffer(ctx->outputs[0], frame->nb_samples);
+ if (!frame_out) {
+ av_frame_free(&frame);
+ return AVERROR(ENOMEM);
+ }
+ }
+ s->process_samples(s, link->channels, frame->nb_samples, frame, frame_out);
+ if (frame_out != frame)
+ av_frame_free(&frame);
+ return ff_filter_frame(ctx->outputs[0], frame_out);
+}
+
+static int init(AVFilterContext *ctx)
+{
+ EqContext *s = ctx->priv;
+ EqBand *bands = NULL, *band;
+ unsigned nb_bands = 0, nb_channels = 1;
+ const char *defs = s->defs ? s->defs : "1";
+ char *tail;
+ double gain, freq;
+ int ret = AVERROR(EINVAL);
+
+ s->global_gain = round(s->global_gain_opt * (1 << FPOINT));
+ while (1) {
+ gain = av_strtod(defs, &tail);
+ if (tail == defs) {
+ av_log(ctx, AV_LOG_ERROR, "Error parsing defs: expected gain, got
\"%s\"\n", defs);
+ goto error;
+ }
+ defs = tail;
+ band = av_dynarray2_add((void **)&bands, &nb_bands, sizeof(*bands),
NULL);
+ if (!band)
+ return AVERROR(ENOMEM);
+ band->g = round(gain * (1 << FPOINT));
+ band->k = 0;
+ if (*defs == 0)
+ break;
+ if (*defs == '/') {
+ defs++;
+ freq = av_strtod(defs, &tail);
+ if (tail == defs || *tail != '/') {
+ av_log(ctx, AV_LOG_ERROR, "Error parsing defs: expected
/freq/, got \"%s\"\n", defs - 1);
+ goto error;
+ }
+ defs = tail + 1;
+ band->k = round(1000 * freq);
+ } else if (*defs == '|') {
+ defs++;
+ nb_channels++;
+ }
+ }
+ av_assert0(nb_bands > 0);
+ s->bands = av_realloc_f(bands, nb_bands, sizeof(*bands));
+ if (!s->bands)
+ return AVERROR(ENOMEM);
+ s->channels = nb_channels;
+ return 0;
+
+error:
+ av_free(bands);
+ return ret;
+}
+
+static void uninit(AVFilterContext *ctx)
+{
+ EqContext *s = ctx->priv;
+ av_freep(&s->vals);
+ av_freep(&s->bands);
+}
+
+static int query_formats(AVFilterContext *ctx)
+{
+ EqContext *s = ctx->priv;
+ static const enum AVSampleFormat sample_fmts[] = {
+ AV_SAMPLE_FMT_U8,
+ AV_SAMPLE_FMT_U8P,
+ AV_SAMPLE_FMT_S16,
+ AV_SAMPLE_FMT_S16P,
+ AV_SAMPLE_FMT_S32,
+ AV_SAMPLE_FMT_S32P,
+ AV_SAMPLE_FMT_FLT,
+ AV_SAMPLE_FMT_FLTP,
+ AV_SAMPLE_FMT_DBL,
+ AV_SAMPLE_FMT_DBLP,
+ AV_SAMPLE_FMT_NONE
+
+ };
+ AVFilterFormats *formats;
+ AVFilterChannelLayouts *layouts;
+ int 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;
+ if (s->channels == 1) {
+ layouts = ff_all_channel_counts();
+ } else {
+ ret = ff_add_channel_layout(&layouts, FF_COUNT2LAYOUT(s->channels));
+ if (ret < 0)
+ return ret;
+ }
+ if (!layouts)
+ return AVERROR(ENOMEM);
+ ret = ff_set_common_channel_layouts(ctx, layouts);
+ if (ret < 0)
+ return ret;
+ return 0;
+}
+
+static int config_props(AVFilterLink *link)
+{
+ AVFilterContext *ctx = link->dst;
+ EqContext *s = ctx->priv;
+ EqBand *band0, *band;
+ unsigned nb_vals, ch, val_size;
+
+ av_assert0(ctx->outputs[0]->channels == ctx->inputs[0]->channels);
+ av_assert0(s->channels == 1 || link->channels == s->channels);
+ nb_vals = 0;
+ if (!s->keep_all_bands) {
+ band0 = band = s->bands;
+ for (ch = 0; ch < s->channels; ch++) {
+ while (band->k && band->k <= link->sample_rate * 250)
+ *(band0++) = *(band++);
+ while (band->k)
+ av_log(ctx, AV_LOG_VERBOSE, "Channel %d: skipping band %f
Hz\n",
+ ch, (band++)->k / 1000.0);
+ (band0++)->k = 0;
+ band++;
+ }
+ }
+ band0 = s->bands;
+ for (ch = 0; ch < s->channels; ch++) {
+ band = band0;
+ while ((band++)->k)
+ nb_vals++;
+ reorder_bands(band0, band - band0, link->sample_rate);
+ print_gain(ctx, ch, band0);
+ band0 = band;
+ }
+ switch (link->format) {
+ case AV_SAMPLE_FMT_U8: s->process_samples = process_samples_u8; break;
+ case AV_SAMPLE_FMT_U8P: s->process_samples = process_samples_u8p; break;
+ case AV_SAMPLE_FMT_S16: s->process_samples = process_samples_s16; break;
+ case AV_SAMPLE_FMT_S16P: s->process_samples = process_samples_s16p; break;
+ case AV_SAMPLE_FMT_S32: s->process_samples = process_samples_s32; break;
+ case AV_SAMPLE_FMT_S32P: s->process_samples = process_samples_s32p; break;
+ case AV_SAMPLE_FMT_FLT: s->process_samples = process_samples_flt; break;
+ case AV_SAMPLE_FMT_FLTP: s->process_samples = process_samples_fltp; break;
+ case AV_SAMPLE_FMT_DBL: s->process_samples = process_samples_dbl; break;
+ case AV_SAMPLE_FMT_DBLP: s->process_samples = process_samples_dblp; break;
+ default: av_assert0(0);
+ }
+ switch (link->format) {
+ case AV_SAMPLE_FMT_U8:
+ case AV_SAMPLE_FMT_U8P:
+ case AV_SAMPLE_FMT_S16:
+ case AV_SAMPLE_FMT_S16P: val_size = sizeof(int); break;
+ case AV_SAMPLE_FMT_S32:
+ case AV_SAMPLE_FMT_S32P: val_size = sizeof(int64_t); break;
+ case AV_SAMPLE_FMT_FLT:
+ case AV_SAMPLE_FMT_FLTP: val_size = sizeof(float); break;
+ case AV_SAMPLE_FMT_DBL:
+ case AV_SAMPLE_FMT_DBLP: val_size = sizeof(double); break;
+ default: av_assert0(0);
+ }
+ s->vals = av_calloc(nb_vals, val_size);
+ if (!s->vals)
+ return AVERROR(ENOMEM);
+ return 0;
+}
+
+#define OFFSET(x) offsetof(EqContext, x)
+#define FLAGS AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_FILTERING_PARAM
+static const AVOption mbfequalizer_options[] = {
+ { "defs", "definitions of the bands", OFFSET(defs), AV_OPT_TYPE_STRING, {
.str = NULL }, 0, 0, FLAGS },
+ { "global_gain", "gain applied to all channels", OFFSET(global_gain_opt),
AV_OPT_TYPE_DOUBLE,
+ { .dbl = 1 }, 0, INT_MAX
>> FPOINT, FLAGS },
+ { "keep_all_bands", "keep bands beyond sample_rate/4",
OFFSET(keep_all_bands), AV_OPT_TYPE_BOOL,
+ { .i64 = 0 }, 0, 1,
FLAGS },
+ { NULL }
+};
+
+AVFILTER_DEFINE_CLASS(mbfequalizer);
+
+static const AVFilterPad mbfequalizer_inputs[] = {
+ {
+ .name = "default",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .config_props = config_props,
+ .filter_frame = filter_frame,
+ },
+ { NULL }
+};
+
+static const AVFilterPad mbfequalizer_outputs[] = {
+ {
+ .name = "default",
+ .type = AVMEDIA_TYPE_AUDIO,
+ },
+ { NULL }
+};
+
+AVFilter ff_af_mbfequalizer = {
+ .name = "mbfequalizer",
+ .description = NULL_IF_CONFIG_SMALL("Multiband fast equalizer"),
+ .init = init,
+ .uninit = uninit,
+ .query_formats = query_formats,
+ .priv_size = sizeof(EqContext),
+ .priv_class = &mbfequalizer_class,
+ .inputs = mbfequalizer_inputs,
+ .outputs = mbfequalizer_outputs,
+};
diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
index c295f8e403..1f3a1a7827 100644
--- a/libavfilter/allfilters.c
+++ b/libavfilter/allfilters.c
@@ -117,6 +117,7 @@ extern AVFilter ff_af_lowpass;
extern AVFilter ff_af_lowshelf;
extern AVFilter ff_af_lv2;
extern AVFilter ff_af_mcompand;
+extern AVFilter ff_af_mbfequalizer;
extern AVFilter ff_af_pan;
extern AVFilter ff_af_replaygain;
extern AVFilter ff_af_resample;
--
2.24.0
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