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https://gitee.com/openharmony/third_party_ffmpeg
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avfilter: add audio pulsator filter
Signed-off-by: Paul B Mahol <onemda@gmail.com>
This commit is contained in:
Paul B Mahol
parent
ca203e9985
commit
c4f7b8f0db
@ -37,6 +37,7 @@ version <next>:
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- compensationdelay filter
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- acompressor filter
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- support encoding 16-bit RLE SGI images
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- apulsator filter
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version 2.8:
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@ -1030,6 +1030,63 @@ It accepts the following values:
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@end table
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@end table
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@section apulsator
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Audio pulsator is something between an autopanner and a tremolo.
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But it can produce funny stereo effects as well. Pulsator changes the volume
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of the left and right channel based on a LFO (low frequency oscillator) with
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different waveforms and shifted phases.
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This filter have the ability to define an offset between left and right
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channel. An offset of 0 means that both LFO shapes match each other.
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The left and right channel are altered equally - a conventional tremolo.
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An offset of 50% means that the shape of the right channel is exactly shifted
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in phase (or moved backwards about half of the frequency) - pulsator acts as
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an autopanner. At 1 both curves match again. Every setting in between moves the
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phase shift gapless between all stages and produces some "bypassing" sounds with
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sine and triangle waveforms. The more you set the offset near 1 (starting from
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the 0.5) the faster the signal passes from the left to the right speaker.
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The filter accepts the following options:
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@table @option
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@item level_in
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Set input gain. By default it is 1. Range is [0.015625 - 64].
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@item level_out
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Set output gain. By default it is 1. Range is [0.015625 - 64].
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@item mode
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Set waveform shape the LFO will use. Can be one of: sine, triangle, square,
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sawup or sawdown. Default is sine.
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@item amount
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Set modulation. Define how much of original signal is affected by the LFO.
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@item offset_l
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Set left channel offset. Default is 0. Allowed range is [0 - 1].
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@item offset_r
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Set right channel offset. Default is 0.5. Allowed range is [0 - 1].
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@item width
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Set pulse width. Default is 1. Allowed range is [0 - 2].
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@item timing
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Set possible timing mode. Can be one of: bpm, ms or hz. Default is hz.
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@item bpm
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Set bpm. Default is 120. Allowed range is [30 - 300]. Only used if timing
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is set to bpm.
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@item ms
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Set ms. Default is 500. Allowed range is [10 - 2000]. Only used if timing
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is set to ms.
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@item hz
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Set frequency in Hz. Default is 2. Allowed range is [0.01 - 100]. Only used
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if timing is set to hz.
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@end table
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@anchor{aresample}
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@section aresample
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@ -40,6 +40,7 @@ OBJS-$(CONFIG_ANULL_FILTER) += af_anull.o
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OBJS-$(CONFIG_APAD_FILTER) += af_apad.o
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OBJS-$(CONFIG_APERMS_FILTER) += f_perms.o
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OBJS-$(CONFIG_APHASER_FILTER) += af_aphaser.o generate_wave_table.o
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OBJS-$(CONFIG_APULSATOR_FILTER) += af_apulsator.o
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OBJS-$(CONFIG_AREALTIME_FILTER) += f_realtime.o
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OBJS-$(CONFIG_ARESAMPLE_FILTER) += af_aresample.o
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OBJS-$(CONFIG_AREVERSE_FILTER) += f_reverse.o
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254
libavfilter/af_apulsator.c
Normal file
254
libavfilter/af_apulsator.c
Normal file
@ -0,0 +1,254 @@
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/*
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* Copyright (c) 2001-2010 Krzysztof Foltman, Markus Schmidt, Thor Harald Johansen and others
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#include "libavutil/opt.h"
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#include "avfilter.h"
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#include "internal.h"
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#include "audio.h"
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enum PulsatorModes { SINE, TRIANGLE, SQUARE, SAWUP, SAWDOWN, NB_MODES };
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enum PulsatorTimings { UNIT_BPM, UNIT_MS, UNIT_HZ, NB_TIMINGS };
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typedef struct SimpleLFO {
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double phase;
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double freq;
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double offset;
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double amount;
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double pwidth;
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int mode;
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int srate;
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} SimpleLFO;
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typedef struct AudioPulsatorContext {
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const AVClass *class;
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int mode;
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double level_in;
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double level_out;
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double amount;
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double offset_l;
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double offset_r;
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double pwidth;
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double bpm;
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double hz;
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int ms;
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int timing;
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SimpleLFO lfoL, lfoR;
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} AudioPulsatorContext;
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#define OFFSET(x) offsetof(AudioPulsatorContext, x)
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#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
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static const AVOption apulsator_options[] = {
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{ "level_in", "set input gain", OFFSET(level_in), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.015625, 64, FLAGS, },
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{ "level_out", "set output gain", OFFSET(level_out), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.015625, 64, FLAGS, },
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{ "mode", "set mode", OFFSET(mode), AV_OPT_TYPE_INT, {.i64=SINE}, SINE, NB_MODES-1, FLAGS, "mode" },
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{ "sine", NULL, 0, AV_OPT_TYPE_CONST, {.i64=SINE}, 0, 0, FLAGS, "mode" },
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{ "triangle", NULL, 0, AV_OPT_TYPE_CONST, {.i64=TRIANGLE},0, 0, FLAGS, "mode" },
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{ "square", NULL, 0, AV_OPT_TYPE_CONST, {.i64=SQUARE}, 0, 0, FLAGS, "mode" },
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{ "sawup", NULL, 0, AV_OPT_TYPE_CONST, {.i64=SAWUP}, 0, 0, FLAGS, "mode" },
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{ "sawdown", NULL, 0, AV_OPT_TYPE_CONST, {.i64=SAWDOWN}, 0, 0, FLAGS, "mode" },
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{ "amount", "set modulation", OFFSET(amount), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0, 1, FLAGS },
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{ "offset_l", "set offset L", OFFSET(offset_l), AV_OPT_TYPE_DOUBLE, {.dbl=0}, 0, 1, FLAGS },
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{ "offset_r", "set offset R", OFFSET(offset_r), AV_OPT_TYPE_DOUBLE, {.dbl=.5}, 0, 1, FLAGS },
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{ "width", "set pulse width", OFFSET(pwidth), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0, 2, FLAGS },
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{ "timing", "set timing", OFFSET(timing), AV_OPT_TYPE_INT, {.i64=2}, 0, NB_TIMINGS-1, FLAGS, "timing" },
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{ "bpm", NULL, 0, AV_OPT_TYPE_CONST, {.i64=UNIT_BPM}, 0, 0, FLAGS, "timing" },
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{ "ms", NULL, 0, AV_OPT_TYPE_CONST, {.i64=UNIT_MS}, 0, 0, FLAGS, "timing" },
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{ "hz", NULL, 0, AV_OPT_TYPE_CONST, {.i64=UNIT_HZ}, 0, 0, FLAGS, "timing" },
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{ "bpm", "set BPM", OFFSET(bpm), AV_OPT_TYPE_DOUBLE, {.dbl=120}, 30, 300, FLAGS },
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{ "ms", "set ms", OFFSET(ms), AV_OPT_TYPE_INT, {.i64=500}, 10, 2000, FLAGS },
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{ "hz", "set frequency", OFFSET(hz), AV_OPT_TYPE_DOUBLE, {.dbl=2}, 0.01, 100, FLAGS },
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{ NULL }
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};
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AVFILTER_DEFINE_CLASS(apulsator);
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static void lfo_advance(SimpleLFO *lfo, unsigned count)
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{
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lfo->phase = fabs(lfo->phase + count * lfo->freq / lfo->srate);
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if (lfo->phase >= 1)
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lfo->phase = fmod(lfo->phase, 1);
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}
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static double lfo_get_value(SimpleLFO *lfo)
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{
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double phs = FFMIN(100, lfo->phase / FFMIN(1.99, FFMAX(0.01, lfo->pwidth)) + lfo->offset);
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double val;
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if (phs > 1)
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phs = fmod(phs, 1.);
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switch (lfo->mode) {
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case SINE:
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val = sin(phs * 2 * M_PI);
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break;
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case TRIANGLE:
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if (phs > 0.75)
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val = (phs - 0.75) * 4 - 1;
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else if (phs > 0.25)
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val = -4 * phs + 2;
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else
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val = phs * 4;
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break;
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case SQUARE:
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val = phs < 0.5 ? -1 : +1;
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break;
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case SAWUP:
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val = phs * 2 - 1;
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break;
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case SAWDOWN:
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val = 1 - phs * 2;
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break;
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}
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return val * lfo->amount;
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}
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static int filter_frame(AVFilterLink *inlink, AVFrame *in)
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{
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AVFilterContext *ctx = inlink->dst;
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AVFilterLink *outlink = ctx->outputs[0];
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AudioPulsatorContext *s = ctx->priv;
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const double *src = (const double *)in->data[0];
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const int nb_samples = in->nb_samples;
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const double level_out = s->level_out;
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const double level_in = s->level_in;
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const double amount = s->amount;
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AVFrame *out;
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double *dst;
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int n;
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if (av_frame_is_writable(in)) {
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out = in;
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} else {
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out = ff_get_audio_buffer(inlink, in->nb_samples);
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if (!out) {
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av_frame_free(&in);
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return AVERROR(ENOMEM);
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}
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av_frame_copy_props(out, in);
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}
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dst = (double *)out->data[0];
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for (n = 0; n < nb_samples; n++) {
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double outL;
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double outR;
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double inL = src[0] * level_in;
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double inR = src[1] * level_in;
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double procL = inL;
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double procR = inR;
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procL *= lfo_get_value(&s->lfoL) * 0.5 + amount / 2;
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procR *= lfo_get_value(&s->lfoR) * 0.5 + amount / 2;
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outL = procL + inL * (1 - amount);
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outR = procR + inR * (1 - amount);
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outL *= level_out;
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outR *= level_out;
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dst[0] = outL;
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dst[1] = outR;
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lfo_advance(&s->lfoL, 1);
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lfo_advance(&s->lfoR, 1);
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dst += 2;
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src += 2;
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}
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if (in != out)
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av_frame_free(&in);
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return ff_filter_frame(outlink, out);
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}
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static int query_formats(AVFilterContext *ctx)
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{
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AVFilterChannelLayouts *layout = NULL;
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AVFilterFormats *formats = NULL;
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int ret;
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if ((ret = ff_add_format (&formats, AV_SAMPLE_FMT_DBL )) < 0 ||
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(ret = ff_set_common_formats (ctx , formats )) < 0 ||
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(ret = ff_add_channel_layout (&layout , AV_CH_LAYOUT_STEREO)) < 0 ||
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(ret = ff_set_common_channel_layouts (ctx , layout )) < 0)
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return ret;
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formats = ff_all_samplerates();
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return ff_set_common_samplerates(ctx, formats);
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}
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static int config_input(AVFilterLink *inlink)
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{
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AVFilterContext *ctx = inlink->dst;
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AudioPulsatorContext *s = ctx->priv;
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double freq;
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switch (s->timing) {
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case UNIT_BPM: freq = s->bpm / 60; break;
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case UNIT_MS: freq = 1 / (s->ms / 1000.); break;
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case UNIT_HZ: freq = s->hz; break;
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}
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s->lfoL.freq = freq;
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s->lfoR.freq = freq;
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s->lfoL.mode = s->mode;
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s->lfoR.mode = s->mode;
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s->lfoL.offset = s->offset_l;
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s->lfoR.offset = s->offset_r;
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s->lfoL.srate = inlink->sample_rate;
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s->lfoR.srate = inlink->sample_rate;
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s->lfoL.amount = s->amount;
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s->lfoR.amount = s->amount;
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s->lfoL.pwidth = s->pwidth;
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s->lfoR.pwidth = s->pwidth;
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return 0;
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}
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static const AVFilterPad inputs[] = {
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{
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.name = "default",
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.type = AVMEDIA_TYPE_AUDIO,
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.config_props = config_input,
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.filter_frame = filter_frame,
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},
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{ NULL }
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};
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static const AVFilterPad outputs[] = {
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{
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.name = "default",
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.type = AVMEDIA_TYPE_AUDIO,
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},
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{ NULL }
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};
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AVFilter ff_af_apulsator = {
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.name = "apulsator",
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.description = NULL_IF_CONFIG_SMALL("Audio pulsator."),
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.priv_size = sizeof(AudioPulsatorContext),
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.priv_class = &apulsator_class,
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.query_formats = query_formats,
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.inputs = inputs,
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.outputs = outputs,
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};
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@ -62,6 +62,7 @@ void avfilter_register_all(void)
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REGISTER_FILTER(APAD, apad, af);
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REGISTER_FILTER(APERMS, aperms, af);
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REGISTER_FILTER(APHASER, aphaser, af);
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REGISTER_FILTER(APULSATOR, apulsator, af);
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REGISTER_FILTER(AREALTIME, arealtime, af);
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REGISTER_FILTER(ARESAMPLE, aresample, af);
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REGISTER_FILTER(AREVERSE, areverse, af);
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@ -30,7 +30,7 @@
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#include "libavutil/version.h"
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#define LIBAVFILTER_VERSION_MAJOR 6
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#define LIBAVFILTER_VERSION_MINOR 17
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#define LIBAVFILTER_VERSION_MINOR 18
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#define LIBAVFILTER_VERSION_MICRO 100
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#define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \
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