mirror of https://git.ffmpeg.org/ffmpeg.git
250 lines
8.3 KiB
C
250 lines
8.3 KiB
C
/*
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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/avassert.h"
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#include "libavutil/channel_layout.h"
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#include "libavutil/opt.h"
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#include "avfilter.h"
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#include "formats.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 hertz;
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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(hertz), 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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default: av_assert0(0);
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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 , &(AVChannelLayout)AV_CHANNEL_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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return ff_set_common_all_samplerates(ctx);
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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->hertz; break;
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default: av_assert0(0);
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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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};
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const 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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FILTER_INPUTS(inputs),
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FILTER_OUTPUTS(ff_audio_default_filterpad),
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FILTER_QUERY_FUNC(query_formats),
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};
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