libavfilter/af_apulsator.c

   1 /*
   2  * Copyright (c) 2001-2010 Krzysztof Foltman, Markus Schmidt, Thor Harald Johansen and others
   3  *
   4  * This file is part of FFmpeg.
   5  *
   6  * FFmpeg is free software; you can redistribute it and/or
   7  * modify it under the terms of the GNU Lesser General Public
   8  * License as published by the Free Software Foundation; either
   9  * version 2.1 of the License, or (at your option) any later version.
  10  *
  11  * FFmpeg is distributed in the hope that it will be useful,
  12  * but WITHOUT ANY WARRANTY; without even the implied warranty of
  13  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
  14  * Lesser General Public License for more details.
  15  *
  16  * You should have received a copy of the GNU Lesser General Public
  17  * License along with FFmpeg; if not, write to the Free Software
  18  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  19  */
  20
  21 #include "libavutil/avassert.h"
  22 #include "libavutil/channel_layout.h"
  23 #include "libavutil/opt.h"
  24 #include "avfilter.h"
  25 #include "filters.h"
  26 #include "formats.h"
  27 #include "audio.h"
  28
  29 enum PulsatorModes { SINE, TRIANGLE, SQUARE, SAWUP, SAWDOWN, NB_MODES };
  30 enum PulsatorTimings { UNIT_BPM, UNIT_MS, UNIT_HZ, NB_TIMINGS };
  31
  32 typedef struct SimpleLFO {
  33     double phase;
  34     double freq;
  35     double offset;
  36     double amount;
  37     double pwidth;
  38     int mode;
  39     int srate;
  40 } SimpleLFO;
  41
  42 typedef struct AudioPulsatorContext {
  43     const AVClass *class;
  44     int mode;
  45     double level_in;
  46     double level_out;
  47     double amount;
  48     double offset_l;
  49     double offset_r;
  50     double pwidth;
  51     double bpm;
  52     double hertz;
  53     int ms;
  54     int timing;
  55
  56     SimpleLFO lfoL, lfoR;
  57 } AudioPulsatorContext;
  58
  59 #define OFFSET(x) offsetof(AudioPulsatorContext, x)
  60 #define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
  61
  62 static const AVOption apulsator_options[] = {
  63     { "level_in",   "set input gain", OFFSET(level_in),  AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.015625, 64, FLAGS, },
  64     { "level_out", "set output gain", OFFSET(level_out), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.015625, 64, FLAGS, },
  65     { "mode",             "set mode", OFFSET(mode),      AV_OPT_TYPE_INT,    {.i64=SINE}, SINE,   NB_MODES-1, FLAGS, .unit = "mode" },
  66     {   "sine",                 NULL, 0,                 AV_OPT_TYPE_CONST,  {.i64=SINE},    0,            0, FLAGS, .unit = "mode" },
  67     {   "triangle",             NULL, 0,                 AV_OPT_TYPE_CONST,  {.i64=TRIANGLE},0,            0, FLAGS, .unit = "mode" },
  68     {   "square",               NULL, 0,                 AV_OPT_TYPE_CONST,  {.i64=SQUARE},  0,            0, FLAGS, .unit = "mode" },
  69     {   "sawup",                NULL, 0,                 AV_OPT_TYPE_CONST,  {.i64=SAWUP},   0,            0, FLAGS, .unit = "mode" },
  70     {   "sawdown",              NULL, 0,                 AV_OPT_TYPE_CONST,  {.i64=SAWDOWN}, 0,            0, FLAGS, .unit = "mode" },
  71     { "amount",     "set modulation", OFFSET(amount),    AV_OPT_TYPE_DOUBLE, {.dbl=1},       0,            1, FLAGS },
  72     { "offset_l",     "set offset L", OFFSET(offset_l),  AV_OPT_TYPE_DOUBLE, {.dbl=0},       0,            1, FLAGS },
  73     { "offset_r",     "set offset R", OFFSET(offset_r),  AV_OPT_TYPE_DOUBLE, {.dbl=.5},      0,            1, FLAGS },
  74     { "width",     "set pulse width", OFFSET(pwidth),    AV_OPT_TYPE_DOUBLE, {.dbl=1},       0,            2, FLAGS },
  75     { "timing",         "set timing", OFFSET(timing),    AV_OPT_TYPE_INT,    {.i64=2},       0, NB_TIMINGS-1, FLAGS, .unit = "timing" },
  76     {   "bpm",                  NULL, 0,                 AV_OPT_TYPE_CONST,  {.i64=UNIT_BPM},  0,          0, FLAGS, .unit = "timing" },
  77     {   "ms",                   NULL, 0,                 AV_OPT_TYPE_CONST,  {.i64=UNIT_MS},   0,          0, FLAGS, .unit = "timing" },
  78     {   "hz",                   NULL, 0,                 AV_OPT_TYPE_CONST,  {.i64=UNIT_HZ},   0,          0, FLAGS, .unit = "timing" },
  79     { "bpm",               "set BPM", OFFSET(bpm),       AV_OPT_TYPE_DOUBLE, {.dbl=120},    30,          300, FLAGS },
  80     { "ms",                 "set ms", OFFSET(ms),        AV_OPT_TYPE_INT,    {.i64=500},    10,         2000, FLAGS },
  81     { "hz",          "set frequency", OFFSET(hertz),     AV_OPT_TYPE_DOUBLE, {.dbl=2},    0.01,          100, FLAGS },
  82     { NULL }
  83 };
  84
  85 AVFILTER_DEFINE_CLASS(apulsator);
  86
  87 static void lfo_advance(SimpleLFO *lfo, unsigned count)
  88 {
  89     lfo->phase = fabs(lfo->phase + count * lfo->freq / lfo->srate);
  90     if (lfo->phase >= 1)
  91         lfo->phase = fmod(lfo->phase, 1);
  92 }
  93
  94 static double lfo_get_value(SimpleLFO *lfo)
  95 {
  96     double phs = FFMIN(100, lfo->phase / FFMIN(1.99, FFMAX(0.01, lfo->pwidth)) + lfo->offset);
  97     double val;
  98
  99     if (phs > 1)
 100         phs = fmod(phs, 1.);
 101
 102     switch (lfo->mode) {
 103     case SINE:
 104         val = sin(phs * 2 * M_PI);
 105         break;
 106     case TRIANGLE:
 107         if (phs > 0.75)
 108             val = (phs - 0.75) * 4 - 1;
 109         else if (phs > 0.25)
 110             val = -4 * phs + 2;
 111         else
 112             val = phs * 4;
 113         break;
 114     case SQUARE:
 115         val = phs < 0.5 ? -1 : +1;
 116         break;
 117     case SAWUP:
 118         val = phs * 2 - 1;
 119         break;
 120     case SAWDOWN:
 121         val = 1 - phs * 2;
 122         break;
 123     default: av_assert0(0);
 124     }
 125
 126     return val * lfo->amount;
 127 }
 128
 129 static int filter_frame(AVFilterLink *inlink, AVFrame *in)
 130 {
 131     AVFilterContext *ctx = inlink->dst;
 132     AVFilterLink *outlink = ctx->outputs[0];
 133     AudioPulsatorContext *s = ctx->priv;
 134     const double *src = (const double *)in->data[0];
 135     const int nb_samples = in->nb_samples;
 136     const double level_out = s->level_out;
 137     const double level_in = s->level_in;
 138     const double amount = s->amount;
 139     AVFrame *out;
 140     double *dst;
 141     int n;
 142
 143     if (av_frame_is_writable(in)) {
 144         out = in;
 145     } else {
 146         out = ff_get_audio_buffer(inlink, in->nb_samples);
 147         if (!out) {
 148             av_frame_free(&in);
 149             return AVERROR(ENOMEM);
 150         }
 151         av_frame_copy_props(out, in);
 152     }
 153     dst = (double *)out->data[0];
 154
 155     for (n = 0; n < nb_samples; n++) {
 156         double outL;
 157         double outR;
 158         double inL = src[0] * level_in;
 159         double inR = src[1] * level_in;
 160         double procL = inL;
 161         double procR = inR;
 162
 163         procL *= lfo_get_value(&s->lfoL) * 0.5 + amount / 2;
 164         procR *= lfo_get_value(&s->lfoR) * 0.5 + amount / 2;
 165
 166         outL = procL + inL * (1 - amount);
 167         outR = procR + inR * (1 - amount);
 168
 169         outL *= level_out;
 170         outR *= level_out;
 171
 172         dst[0] = outL;
 173         dst[1] = outR;
 174
 175         lfo_advance(&s->lfoL, 1);
 176         lfo_advance(&s->lfoR, 1);
 177
 178         dst += 2;
 179         src += 2;
 180     }
 181
 182     if (in != out)
 183         av_frame_free(&in);
 184
 185     return ff_filter_frame(outlink, out);
 186 }
 187
 188 static int query_formats(const AVFilterContext *ctx,
 189                          AVFilterFormatsConfig **cfg_in,
 190                          AVFilterFormatsConfig **cfg_out)
 191 {
 192     static const enum AVSampleFormat formats[] = {
 193         AV_SAMPLE_FMT_DBL,
 194         AV_SAMPLE_FMT_NONE,
 195     };
 196     static const AVChannelLayout layouts[] = {
 197         AV_CHANNEL_LAYOUT_STEREO,
 198         { .nb_channels = 0 },
 199     };
 200
 201     int ret;
 202
 203     ret = ff_set_common_formats_from_list2(ctx, cfg_in, cfg_out, formats);
 204     if (ret < 0)
 205         return ret;
 206
 207     ret = ff_set_common_channel_layouts_from_list2(ctx, cfg_in, cfg_out, layouts);
 208     if (ret < 0)
 209         return ret;
 210
 211     return 0;
 212 }
 213
 214 static int config_input(AVFilterLink *inlink)
 215 {
 216     AVFilterContext *ctx = inlink->dst;
 217     AudioPulsatorContext *s = ctx->priv;
 218     double freq;
 219
 220     switch (s->timing) {
 221     case UNIT_BPM:  freq = s->bpm / 60;         break;
 222     case UNIT_MS:   freq = 1 / (s->ms / 1000.); break;
 223     case UNIT_HZ:   freq = s->hertz;            break;
 224     default: av_assert0(0);
 225     }
 226
 227     s->lfoL.freq   = freq;
 228     s->lfoR.freq   = freq;
 229     s->lfoL.mode   = s->mode;
 230     s->lfoR.mode   = s->mode;
 231     s->lfoL.offset = s->offset_l;
 232     s->lfoR.offset = s->offset_r;
 233     s->lfoL.srate  = inlink->sample_rate;
 234     s->lfoR.srate  = inlink->sample_rate;
 235     s->lfoL.amount = s->amount;
 236     s->lfoR.amount = s->amount;
 237     s->lfoL.pwidth = s->pwidth;
 238     s->lfoR.pwidth = s->pwidth;
 239
 240     return 0;
 241 }
 242
 243 static const AVFilterPad inputs[] = {
 244     {
 245         .name         = "default",
 246         .type         = AVMEDIA_TYPE_AUDIO,
 247         .config_props = config_input,
 248         .filter_frame = filter_frame,
 249     },
 250 };
 251
 252 const FFFilter ff_af_apulsator = {
 253     .p.name        = "apulsator",
 254     .p.description = NULL_IF_CONFIG_SMALL("Audio pulsator."),
 255     .p.priv_class  = &apulsator_class,
 256     .priv_size     = sizeof(AudioPulsatorContext),
 257     FILTER_INPUTS(inputs),
 258     FILTER_OUTPUTS(ff_audio_default_filterpad),
 259     FILTER_QUERY_FUNC2(query_formats),
 260 };