avfilter: add audio pulsator filter

Signed-off-by: Paul B Mahol <onemda@gmail.com>

Changelog

@@ -37,6 +37,7 @@ version <next>:
 - compensationdelay filter
 - acompressor filter
 - support encoding 16-bit RLE SGI images
+- apulsator filter
 
 
 version 2.8:

doc/filters.texi

@@ -1030,6 +1030,63 @@ It accepts the following values:
 @end table
 @end table
 
+@section apulsator
+
+Audio pulsator is something between an autopanner and a tremolo.
+But it can produce funny stereo effects as well. Pulsator changes the volume
+of the left and right channel based on a LFO (low frequency oscillator) with
+different waveforms and shifted phases.
+This filter have the ability to define an offset between left and right
+channel. An offset of 0 means that both LFO shapes match each other.
+The left and right channel are altered equally - a conventional tremolo.
+An offset of 50% means that the shape of the right channel is exactly shifted
+in phase (or moved backwards about half of the frequency) - pulsator acts as
+an autopanner. At 1 both curves match again. Every setting in between moves the
+phase shift gapless between all stages and produces some "bypassing" sounds with
+sine and triangle waveforms. The more you set the offset near 1 (starting from
+the 0.5) the faster the signal passes from the left to the right speaker.
+
+The filter accepts the following options:
+
+@table @option
+@item level_in
+Set input gain. By default it is 1. Range is [0.015625 - 64].
+
+@item level_out
+Set output gain. By default it is 1. Range is [0.015625 - 64].
+
+@item mode
+Set waveform shape the LFO will use. Can be one of: sine, triangle, square,
+sawup or sawdown. Default is sine.
+
+@item amount
+Set modulation. Define how much of original signal is affected by the LFO.
+
+@item offset_l
+Set left channel offset. Default is 0. Allowed range is [0 - 1].
+
+@item offset_r
+Set right channel offset. Default is 0.5. Allowed range is [0 - 1].
+
+@item width
+Set pulse width. Default is 1. Allowed range is [0 - 2].
+
+@item timing
+Set possible timing mode. Can be one of: bpm, ms or hz. Default is hz.
+
+@item bpm
+Set bpm. Default is 120. Allowed range is [30 - 300]. Only used if timing
+is set to bpm.
+
+@item ms
+Set ms. Default is 500. Allowed range is [10 - 2000]. Only used if timing
+is set to ms.
+
+@item hz
+Set frequency in Hz. Default is 2. Allowed range is [0.01 - 100]. Only used
+if timing is set to hz.
+@end table
+
 @anchor{aresample}
 @section aresample
 

libavfilter/Makefile

@@ -40,6 +40,7 @@ OBJS-$(CONFIG_ANULL_FILTER)                  += af_anull.o
 OBJS-$(CONFIG_APAD_FILTER)                   += af_apad.o
 OBJS-$(CONFIG_APERMS_FILTER)                 += f_perms.o
 OBJS-$(CONFIG_APHASER_FILTER)                += af_aphaser.o generate_wave_table.o
+OBJS-$(CONFIG_APULSATOR_FILTER)              += af_apulsator.o
 OBJS-$(CONFIG_AREALTIME_FILTER)              += f_realtime.o
 OBJS-$(CONFIG_ARESAMPLE_FILTER)              += af_aresample.o
 OBJS-$(CONFIG_AREVERSE_FILTER)               += f_reverse.o

libavfilter/af_apulsator.c

+/*
+ * Copyright (c) 2001-2010 Krzysztof Foltman, Markus Schmidt, Thor Harald Johansen and others
+ *
+ * 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/opt.h"
+#include "avfilter.h"
+#include "internal.h"
+#include "audio.h"
+
+enum PulsatorModes { SINE, TRIANGLE, SQUARE, SAWUP, SAWDOWN, NB_MODES };
+enum PulsatorTimings { UNIT_BPM, UNIT_MS, UNIT_HZ, NB_TIMINGS };
+
+typedef struct SimpleLFO {
+    double phase;
+    double freq;
+    double offset;
+    double amount;
+    double pwidth;
+    int mode;
+    int srate;
+} SimpleLFO;
+
+typedef struct AudioPulsatorContext {
+    const AVClass *class;
+    int mode;
+    double level_in;
+    double level_out;
+    double amount;
+    double offset_l;
+    double offset_r;
+    double pwidth;
+    double bpm;
+    double hz;
+    int ms;
+    int timing;
+
+    SimpleLFO lfoL, lfoR;
+} AudioPulsatorContext;
+
+#define OFFSET(x) offsetof(AudioPulsatorContext, x)
+#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
+
+static const AVOption apulsator_options[] = {
+    { "level_in",   "set input gain", OFFSET(level_in),  AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.015625, 64, FLAGS, },
+    { "level_out", "set output gain", OFFSET(level_out), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.015625, 64, FLAGS, },
+    { "mode",             "set mode", OFFSET(mode),      AV_OPT_TYPE_INT,    {.i64=SINE}, SINE,   NB_MODES-1, FLAGS, "mode" },
+    {   "sine",                 NULL, 0,                 AV_OPT_TYPE_CONST,  {.i64=SINE},    0,            0, FLAGS, "mode" },
+    {   "triangle",             NULL, 0,                 AV_OPT_TYPE_CONST,  {.i64=TRIANGLE},0,            0, FLAGS, "mode" },
+    {   "square",               NULL, 0,                 AV_OPT_TYPE_CONST,  {.i64=SQUARE},  0,            0, FLAGS, "mode" },
+    {   "sawup",                NULL, 0,                 AV_OPT_TYPE_CONST,  {.i64=SAWUP},   0,            0, FLAGS, "mode" },
+    {   "sawdown",              NULL, 0,                 AV_OPT_TYPE_CONST,  {.i64=SAWDOWN}, 0,            0, FLAGS, "mode" },
+    { "amount",     "set modulation", OFFSET(amount),    AV_OPT_TYPE_DOUBLE, {.dbl=1},       0,            1, FLAGS },
+    { "offset_l",     "set offset L", OFFSET(offset_l),  AV_OPT_TYPE_DOUBLE, {.dbl=0},       0,            1, FLAGS },
+    { "offset_r",     "set offset R", OFFSET(offset_r),  AV_OPT_TYPE_DOUBLE, {.dbl=.5},      0,            1, FLAGS },
+    { "width",     "set pulse width", OFFSET(pwidth),    AV_OPT_TYPE_DOUBLE, {.dbl=1},       0,            2, FLAGS },
+    { "timing",         "set timing", OFFSET(timing),    AV_OPT_TYPE_INT,    {.i64=2},       0, NB_TIMINGS-1, FLAGS, "timing" },
+    {   "bpm",                  NULL, 0,                 AV_OPT_TYPE_CONST,  {.i64=UNIT_BPM},  0,          0, FLAGS, "timing" },
+    {   "ms",                   NULL, 0,                 AV_OPT_TYPE_CONST,  {.i64=UNIT_MS},   0,          0, FLAGS, "timing" },
+    {   "hz",                   NULL, 0,                 AV_OPT_TYPE_CONST,  {.i64=UNIT_HZ},   0,          0, FLAGS, "timing" },
+    { "bpm",               "set BPM", OFFSET(bpm),       AV_OPT_TYPE_DOUBLE, {.dbl=120},    30,          300, FLAGS },
+    { "ms",                 "set ms", OFFSET(ms),        AV_OPT_TYPE_INT,    {.i64=500},    10,         2000, FLAGS },
+    { "hz",          "set frequency", OFFSET(hz),        AV_OPT_TYPE_DOUBLE, {.dbl=2},    0.01,          100, FLAGS },
+    { NULL }
+};
+
+AVFILTER_DEFINE_CLASS(apulsator);
+
+static void lfo_advance(SimpleLFO *lfo, unsigned count)
+{
+    lfo->phase = fabs(lfo->phase + count * lfo->freq / lfo->srate);
+    if (lfo->phase >= 1)
+        lfo->phase = fmod(lfo->phase, 1);
+}
+
+static double lfo_get_value(SimpleLFO *lfo)
+{
+    double phs = FFMIN(100, lfo->phase / FFMIN(1.99, FFMAX(0.01, lfo->pwidth)) + lfo->offset);
+    double val;
+
+    if (phs > 1)
+        phs = fmod(phs, 1.);
+
+    switch (lfo->mode) {
+    case SINE:
+        val = sin(phs * 2 * M_PI);
+        break;
+    case TRIANGLE:
+        if (phs > 0.75)
+            val = (phs - 0.75) * 4 - 1;
+        else if (phs > 0.25)
+            val = -4 * phs + 2;
+        else
+            val = phs * 4;
+        break;
+    case SQUARE:
+        val = phs < 0.5 ? -1 : +1;
+        break;
+    case SAWUP:
+        val = phs * 2 - 1;
+        break;
+    case SAWDOWN:
+        val = 1 - phs * 2;
+        break;
+    }
+
+    return val * lfo->amount;
+}
+
+static int filter_frame(AVFilterLink *inlink, AVFrame *in)
+{
+    AVFilterContext *ctx = inlink->dst;
+    AVFilterLink *outlink = ctx->outputs[0];
+    AudioPulsatorContext *s = ctx->priv;
+    const double *src = (const double *)in->data[0];
+    const int nb_samples = in->nb_samples;
+    const double level_out = s->level_out;
+    const double level_in = s->level_in;
+    const double amount = s->amount;
+    AVFrame *out;
+    double *dst;
+    int n;
+
+    if (av_frame_is_writable(in)) {
+        out = in;
+    } else {
+        out = ff_get_audio_buffer(inlink, in->nb_samples);
+        if (!out) {
+            av_frame_free(&in);
+            return AVERROR(ENOMEM);
+        }
+        av_frame_copy_props(out, in);
+    }
+    dst = (double *)out->data[0];
+
+    for (n = 0; n < nb_samples; n++) {
+        double outL;
+        double outR;
+        double inL = src[0] * level_in;
+        double inR = src[1] * level_in;
+        double procL = inL;
+        double procR = inR;
+
+        procL *= lfo_get_value(&s->lfoL) * 0.5 + amount / 2;
+        procR *= lfo_get_value(&s->lfoR) * 0.5 + amount / 2;
+
+        outL = procL + inL * (1 - amount);
+        outR = procR + inR * (1 - amount);
+
+        outL *= level_out;
+        outR *= level_out;
+
+        dst[0] = outL;
+        dst[1] = outR;
+
+        lfo_advance(&s->lfoL, 1);
+        lfo_advance(&s->lfoR, 1);
+
+        dst += 2;
+        src += 2;
+    }
+
+    if (in != out)
+        av_frame_free(&in);
+
+    return ff_filter_frame(outlink, out);
+}
+
+static int query_formats(AVFilterContext *ctx)
+{
+    AVFilterChannelLayouts *layout = NULL;
+    AVFilterFormats *formats = NULL;
+    int ret;
+
+    if ((ret = ff_add_format                 (&formats, AV_SAMPLE_FMT_DBL  )) < 0 ||
+        (ret = ff_set_common_formats         (ctx     , formats            )) < 0 ||
+        (ret = ff_add_channel_layout         (&layout , AV_CH_LAYOUT_STEREO)) < 0 ||
+        (ret = ff_set_common_channel_layouts (ctx     , layout             )) < 0)
+        return ret;
+
+    formats = ff_all_samplerates();
+    return ff_set_common_samplerates(ctx, formats);
+}
+
+static int config_input(AVFilterLink *inlink)
+{
+    AVFilterContext *ctx = inlink->dst;
+    AudioPulsatorContext *s = ctx->priv;
+    double freq;
+
+    switch (s->timing) {
+    case UNIT_BPM:  freq = s->bpm / 60;         break;
+    case UNIT_MS:   freq = 1 / (s->ms / 1000.); break;
+    case UNIT_HZ:   freq = s->hz;               break;
+    }
+
+    s->lfoL.freq   = freq;
+    s->lfoR.freq   = freq;
+    s->lfoL.mode   = s->mode;
+    s->lfoR.mode   = s->mode;
+    s->lfoL.offset = s->offset_l;
+    s->lfoR.offset = s->offset_r;
+    s->lfoL.srate  = inlink->sample_rate;
+    s->lfoR.srate  = inlink->sample_rate;
+    s->lfoL.amount = s->amount;
+    s->lfoR.amount = s->amount;
+    s->lfoL.pwidth = s->pwidth;
+    s->lfoR.pwidth = s->pwidth;
+
+    return 0;
+}
+
+static const AVFilterPad inputs[] = {
+    {
+        .name         = "default",
+        .type         = AVMEDIA_TYPE_AUDIO,
+        .config_props = config_input,
+        .filter_frame = filter_frame,
+    },
+    { NULL }
+};
+
+static const AVFilterPad outputs[] = {
+    {
+        .name = "default",
+        .type = AVMEDIA_TYPE_AUDIO,
+    },
+    { NULL }
+};
+
+AVFilter ff_af_apulsator = {
+    .name          = "apulsator",
+    .description   = NULL_IF_CONFIG_SMALL("Audio pulsator."),
+    .priv_size     = sizeof(AudioPulsatorContext),
+    .priv_class    = &apulsator_class,
+    .query_formats = query_formats,
+    .inputs        = inputs,
+    .outputs       = outputs,
+};

libavfilter/allfilters.c

@@ -62,6 +62,7 @@ void avfilter_register_all(void)
     REGISTER_FILTER(APAD,           apad,           af);
     REGISTER_FILTER(APERMS,         aperms,         af);
     REGISTER_FILTER(APHASER,        aphaser,        af);
+    REGISTER_FILTER(APULSATOR,      apulsator,      af);
     REGISTER_FILTER(AREALTIME,      arealtime,      af);
     REGISTER_FILTER(ARESAMPLE,      aresample,      af);
     REGISTER_FILTER(AREVERSE,       areverse,       af);

libavfilter/version.h

@@ -30,7 +30,7 @@
 #include "libavutil/version.h"
 
 #define LIBAVFILTER_VERSION_MAJOR   6
-#define LIBAVFILTER_VERSION_MINOR  17
+#define LIBAVFILTER_VERSION_MINOR  18
 #define LIBAVFILTER_VERSION_MICRO 100
 
 #define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \