[FFmpeg-devel] [PATCH] avfilter: add audio pulsator filter
Paul B Mahol
onemda at gmail.com
Tue Dec 1 13:04:35 CET 2015
Signed-off-by: Paul B Mahol <onemda at gmail.com>
---
doc/filters.texi | 57 +++++++++
libavfilter/Makefile | 1 +
libavfilter/af_apulsator.c | 279 +++++++++++++++++++++++++++++++++++++++++++++
libavfilter/allfilters.c | 1 +
4 files changed, 338 insertions(+)
create mode 100644 libavfilter/af_apulsator.c
diff --git a/doc/filters.texi b/doc/filters.texi
index fc71a99..a4afcac 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -1030,6 +1030,63 @@ It accepts the following values:
@end table
@end table
+ at 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:
+
+ at table @option
+ at item level_in
+Set input gain. By default it is 1. Range is [0.015625 - 64].
+
+ at item level_out
+Set output gain. By default it is 1. Range is [0.015625 - 64].
+
+ at item mode
+Set waveform shape the LFO will use. Can be one of: sine, triangle, square,
+sawup or sawdown. Default is sine.
+
+ at item amount
+Set modulation. Define how much of original signal is affected by the LFO.
+
+ at item offset_l
+Set left channel offset. Default is 0. Allowed range is [0 - 1].
+
+ at item offset_r
+Set right channel offset. Default is 0.5. Allowed range is [0 - 1].
+
+ at item width
+Set pulse width.
+
+ at item timing
+Set possible timing mode. Can be one of: bpm, ms or hz. Default is hz.
+
+ at item bpm
+Set bpm. Default is 120. Allowed range is [30 - 300]. Only used if timing
+is set to bpm.
+
+ at item ms
+Set ms. Default is 500. Allowed range is [10 - 2000]. Only used if timing
+is set to ms.
+
+ at item hz
+Set frequency in Hz. Default is 2. Allowed range is [0.01 - 100]. Only used
+if timing is set to hz.
+ at end table
+
@anchor{aresample}
@section aresample
diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index e31bdaa..b6c0d7b 100644
--- a/libavfilter/Makefile
+++ b/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
diff --git a/libavfilter/af_apulsator.c b/libavfilter/af_apulsator.c
new file mode 100644
index 0000000..9100eff
--- /dev/null
+++ b/libavfilter/af_apulsator.c
@@ -0,0 +1,279 @@
+/*
+ * 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 > 0.5)
+ val = (phs - 0.5) * 4 * -1;
+ else if (phs > 0.25)
+ val = 1 - (phs - 0.25) * 4;
+ 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;
+ static const enum AVSampleFormat sample_fmts[] = {
+ AV_SAMPLE_FMT_DBL,
+ AV_SAMPLE_FMT_NONE
+ };
+ int ret;
+
+ ret = ff_add_channel_layout(&layout, AV_CH_LAYOUT_STEREO);
+ if (ret < 0)
+ return ret;
+ ret = ff_set_common_channel_layouts(ctx, layout);
+ if (ret < 0)
+ goto fail;
+
+ formats = ff_make_format_list(sample_fmts);
+ if (!formats) {
+ ret = AVERROR(ENOMEM);
+ goto fail;
+ }
+ ret = ff_set_common_formats(ctx, formats);
+ if (ret < 0)
+ goto fail;
+
+ formats = ff_all_samplerates();
+ if (!formats) {
+ ret = AVERROR(ENOMEM);
+ goto fail;
+ }
+ ret = ff_set_common_samplerates(ctx, formats);
+ if (!ret)
+ return ret;
+fail:
+ ff_channel_layouts_unref(&layout);
+ return ret;
+}
+
+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,
+};
diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
index ccd3f35..9502ebf 100644
--- a/libavfilter/allfilters.c
+++ b/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);
--
1.9.1
More information about the ffmpeg-devel
mailing list