| Line | Branch | Exec | Source |
|---|---|---|---|
| 1 | /* | ||
| 2 | * Copyright (c) 2013 Paul B Mahol | ||
| 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 | /** | ||
| 22 | * @file | ||
| 23 | * phaser audio filter | ||
| 24 | */ | ||
| 25 | |||
| 26 | #include "libavutil/avassert.h" | ||
| 27 | #include "libavutil/mem.h" | ||
| 28 | #include "libavutil/opt.h" | ||
| 29 | #include "audio.h" | ||
| 30 | #include "avfilter.h" | ||
| 31 | #include "internal.h" | ||
| 32 | #include "generate_wave_table.h" | ||
| 33 | |||
| 34 | typedef struct AudioPhaserContext { | ||
| 35 | const AVClass *class; | ||
| 36 | double in_gain, out_gain; | ||
| 37 | double delay; | ||
| 38 | double decay; | ||
| 39 | double speed; | ||
| 40 | |||
| 41 | int type; | ||
| 42 | |||
| 43 | int delay_buffer_length; | ||
| 44 | double *delay_buffer; | ||
| 45 | |||
| 46 | int modulation_buffer_length; | ||
| 47 | int32_t *modulation_buffer; | ||
| 48 | |||
| 49 | int delay_pos, modulation_pos; | ||
| 50 | |||
| 51 | void (*phaser)(struct AudioPhaserContext *s, | ||
| 52 | uint8_t * const *src, uint8_t **dst, | ||
| 53 | int nb_samples, int channels); | ||
| 54 | } AudioPhaserContext; | ||
| 55 | |||
| 56 | #define OFFSET(x) offsetof(AudioPhaserContext, x) | ||
| 57 | #define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM | ||
| 58 | |||
| 59 | static const AVOption aphaser_options[] = { | ||
| 60 | { "in_gain", "set input gain", OFFSET(in_gain), AV_OPT_TYPE_DOUBLE, {.dbl=.4}, 0, 1, FLAGS }, | ||
| 61 | { "out_gain", "set output gain", OFFSET(out_gain), AV_OPT_TYPE_DOUBLE, {.dbl=.74}, 0, 1e9, FLAGS }, | ||
| 62 | { "delay", "set delay in milliseconds", OFFSET(delay), AV_OPT_TYPE_DOUBLE, {.dbl=3.}, 0, 5, FLAGS }, | ||
| 63 | { "decay", "set decay", OFFSET(decay), AV_OPT_TYPE_DOUBLE, {.dbl=.4}, 0, .99, FLAGS }, | ||
| 64 | { "speed", "set modulation speed", OFFSET(speed), AV_OPT_TYPE_DOUBLE, {.dbl=.5}, .1, 2, FLAGS }, | ||
| 65 | { "type", "set modulation type", OFFSET(type), AV_OPT_TYPE_INT, {.i64=WAVE_TRI}, 0, WAVE_NB-1, FLAGS, .unit = "type" }, | ||
| 66 | { "triangular", NULL, 0, AV_OPT_TYPE_CONST, {.i64=WAVE_TRI}, 0, 0, FLAGS, .unit = "type" }, | ||
| 67 | { "t", NULL, 0, AV_OPT_TYPE_CONST, {.i64=WAVE_TRI}, 0, 0, FLAGS, .unit = "type" }, | ||
| 68 | { "sinusoidal", NULL, 0, AV_OPT_TYPE_CONST, {.i64=WAVE_SIN}, 0, 0, FLAGS, .unit = "type" }, | ||
| 69 | { "s", NULL, 0, AV_OPT_TYPE_CONST, {.i64=WAVE_SIN}, 0, 0, FLAGS, .unit = "type" }, | ||
| 70 | { NULL } | ||
| 71 | }; | ||
| 72 | |||
| 73 | AVFILTER_DEFINE_CLASS(aphaser); | ||
| 74 | |||
| 75 | ✗ | static av_cold int init(AVFilterContext *ctx) | |
| 76 | { | ||
| 77 | ✗ | AudioPhaserContext *s = ctx->priv; | |
| 78 | |||
| 79 | ✗ | if (s->in_gain > (1 - s->decay * s->decay)) | |
| 80 | ✗ | av_log(ctx, AV_LOG_WARNING, "in_gain may cause clipping\n"); | |
| 81 | ✗ | if (s->in_gain / (1 - s->decay) > 1 / s->out_gain) | |
| 82 | ✗ | av_log(ctx, AV_LOG_WARNING, "out_gain may cause clipping\n"); | |
| 83 | |||
| 84 | ✗ | return 0; | |
| 85 | } | ||
| 86 | |||
| 87 | #define MOD(a, b) (((a) >= (b)) ? (a) - (b) : (a)) | ||
| 88 | |||
| 89 | #define PHASER_PLANAR(name, type) \ | ||
| 90 | static void phaser_## name ##p(AudioPhaserContext *s, \ | ||
| 91 | uint8_t * const *ssrc, uint8_t **ddst, \ | ||
| 92 | int nb_samples, int channels) \ | ||
| 93 | { \ | ||
| 94 | int i, c, delay_pos, modulation_pos; \ | ||
| 95 | \ | ||
| 96 | av_assert0(channels > 0); \ | ||
| 97 | for (c = 0; c < channels; c++) { \ | ||
| 98 | type *src = (type *)ssrc[c]; \ | ||
| 99 | type *dst = (type *)ddst[c]; \ | ||
| 100 | double *buffer = s->delay_buffer + \ | ||
| 101 | c * s->delay_buffer_length; \ | ||
| 102 | \ | ||
| 103 | delay_pos = s->delay_pos; \ | ||
| 104 | modulation_pos = s->modulation_pos; \ | ||
| 105 | \ | ||
| 106 | for (i = 0; i < nb_samples; i++, src++, dst++) { \ | ||
| 107 | double v = *src * s->in_gain + buffer[ \ | ||
| 108 | MOD(delay_pos + s->modulation_buffer[ \ | ||
| 109 | modulation_pos], \ | ||
| 110 | s->delay_buffer_length)] * s->decay; \ | ||
| 111 | \ | ||
| 112 | modulation_pos = MOD(modulation_pos + 1, \ | ||
| 113 | s->modulation_buffer_length); \ | ||
| 114 | delay_pos = MOD(delay_pos + 1, s->delay_buffer_length); \ | ||
| 115 | buffer[delay_pos] = v; \ | ||
| 116 | \ | ||
| 117 | *dst = v * s->out_gain; \ | ||
| 118 | } \ | ||
| 119 | } \ | ||
| 120 | \ | ||
| 121 | s->delay_pos = delay_pos; \ | ||
| 122 | s->modulation_pos = modulation_pos; \ | ||
| 123 | } | ||
| 124 | |||
| 125 | #define PHASER(name, type) \ | ||
| 126 | static void phaser_## name (AudioPhaserContext *s, \ | ||
| 127 | uint8_t * const *ssrc, uint8_t **ddst, \ | ||
| 128 | int nb_samples, int channels) \ | ||
| 129 | { \ | ||
| 130 | int i, c, delay_pos, modulation_pos; \ | ||
| 131 | type *src = (type *)ssrc[0]; \ | ||
| 132 | type *dst = (type *)ddst[0]; \ | ||
| 133 | double *buffer = s->delay_buffer; \ | ||
| 134 | \ | ||
| 135 | delay_pos = s->delay_pos; \ | ||
| 136 | modulation_pos = s->modulation_pos; \ | ||
| 137 | \ | ||
| 138 | for (i = 0; i < nb_samples; i++) { \ | ||
| 139 | int pos = MOD(delay_pos + s->modulation_buffer[modulation_pos], \ | ||
| 140 | s->delay_buffer_length) * channels; \ | ||
| 141 | int npos; \ | ||
| 142 | \ | ||
| 143 | delay_pos = MOD(delay_pos + 1, s->delay_buffer_length); \ | ||
| 144 | npos = delay_pos * channels; \ | ||
| 145 | for (c = 0; c < channels; c++, src++, dst++) { \ | ||
| 146 | double v = *src * s->in_gain + buffer[pos + c] * s->decay; \ | ||
| 147 | \ | ||
| 148 | buffer[npos + c] = v; \ | ||
| 149 | \ | ||
| 150 | *dst = v * s->out_gain; \ | ||
| 151 | } \ | ||
| 152 | \ | ||
| 153 | modulation_pos = MOD(modulation_pos + 1, \ | ||
| 154 | s->modulation_buffer_length); \ | ||
| 155 | } \ | ||
| 156 | \ | ||
| 157 | s->delay_pos = delay_pos; \ | ||
| 158 | s->modulation_pos = modulation_pos; \ | ||
| 159 | } | ||
| 160 | |||
| 161 | ✗ | PHASER_PLANAR(dbl, double) | |
| 162 | ✗ | PHASER_PLANAR(flt, float) | |
| 163 | ✗ | PHASER_PLANAR(s16, int16_t) | |
| 164 | ✗ | PHASER_PLANAR(s32, int32_t) | |
| 165 | |||
| 166 | ✗ | PHASER(dbl, double) | |
| 167 | ✗ | PHASER(flt, float) | |
| 168 | ✗ | PHASER(s16, int16_t) | |
| 169 | ✗ | PHASER(s32, int32_t) | |
| 170 | |||
| 171 | ✗ | static int config_output(AVFilterLink *outlink) | |
| 172 | { | ||
| 173 | ✗ | AudioPhaserContext *s = outlink->src->priv; | |
| 174 | ✗ | AVFilterLink *inlink = outlink->src->inputs[0]; | |
| 175 | |||
| 176 | ✗ | s->delay_buffer_length = s->delay * 0.001 * inlink->sample_rate + 0.5; | |
| 177 | ✗ | if (s->delay_buffer_length <= 0) { | |
| 178 | ✗ | av_log(outlink->src, AV_LOG_ERROR, "delay is too small\n"); | |
| 179 | ✗ | return AVERROR(EINVAL); | |
| 180 | } | ||
| 181 | ✗ | s->delay_buffer = av_calloc(s->delay_buffer_length, sizeof(*s->delay_buffer) * inlink->ch_layout.nb_channels); | |
| 182 | ✗ | s->modulation_buffer_length = inlink->sample_rate / s->speed + 0.5; | |
| 183 | ✗ | s->modulation_buffer = av_malloc_array(s->modulation_buffer_length, sizeof(*s->modulation_buffer)); | |
| 184 | |||
| 185 | ✗ | if (!s->modulation_buffer || !s->delay_buffer) | |
| 186 | ✗ | return AVERROR(ENOMEM); | |
| 187 | |||
| 188 | ✗ | ff_generate_wave_table(s->type, AV_SAMPLE_FMT_S32, | |
| 189 | ✗ | s->modulation_buffer, s->modulation_buffer_length, | |
| 190 | ✗ | 1., s->delay_buffer_length, M_PI / 2.0); | |
| 191 | |||
| 192 | ✗ | s->delay_pos = s->modulation_pos = 0; | |
| 193 | |||
| 194 | ✗ | switch (inlink->format) { | |
| 195 | ✗ | case AV_SAMPLE_FMT_DBL: s->phaser = phaser_dbl; break; | |
| 196 | ✗ | case AV_SAMPLE_FMT_DBLP: s->phaser = phaser_dblp; break; | |
| 197 | ✗ | case AV_SAMPLE_FMT_FLT: s->phaser = phaser_flt; break; | |
| 198 | ✗ | case AV_SAMPLE_FMT_FLTP: s->phaser = phaser_fltp; break; | |
| 199 | ✗ | case AV_SAMPLE_FMT_S16: s->phaser = phaser_s16; break; | |
| 200 | ✗ | case AV_SAMPLE_FMT_S16P: s->phaser = phaser_s16p; break; | |
| 201 | ✗ | case AV_SAMPLE_FMT_S32: s->phaser = phaser_s32; break; | |
| 202 | ✗ | case AV_SAMPLE_FMT_S32P: s->phaser = phaser_s32p; break; | |
| 203 | ✗ | default: av_assert0(0); | |
| 204 | } | ||
| 205 | |||
| 206 | ✗ | return 0; | |
| 207 | } | ||
| 208 | |||
| 209 | ✗ | static int filter_frame(AVFilterLink *inlink, AVFrame *inbuf) | |
| 210 | { | ||
| 211 | ✗ | AudioPhaserContext *s = inlink->dst->priv; | |
| 212 | ✗ | AVFilterLink *outlink = inlink->dst->outputs[0]; | |
| 213 | AVFrame *outbuf; | ||
| 214 | |||
| 215 | ✗ | if (av_frame_is_writable(inbuf)) { | |
| 216 | ✗ | outbuf = inbuf; | |
| 217 | } else { | ||
| 218 | ✗ | outbuf = ff_get_audio_buffer(outlink, inbuf->nb_samples); | |
| 219 | ✗ | if (!outbuf) { | |
| 220 | ✗ | av_frame_free(&inbuf); | |
| 221 | ✗ | return AVERROR(ENOMEM); | |
| 222 | } | ||
| 223 | ✗ | av_frame_copy_props(outbuf, inbuf); | |
| 224 | } | ||
| 225 | |||
| 226 | ✗ | s->phaser(s, inbuf->extended_data, outbuf->extended_data, | |
| 227 | outbuf->nb_samples, outbuf->ch_layout.nb_channels); | ||
| 228 | |||
| 229 | ✗ | if (inbuf != outbuf) | |
| 230 | ✗ | av_frame_free(&inbuf); | |
| 231 | |||
| 232 | ✗ | return ff_filter_frame(outlink, outbuf); | |
| 233 | } | ||
| 234 | |||
| 235 | ✗ | static av_cold void uninit(AVFilterContext *ctx) | |
| 236 | { | ||
| 237 | ✗ | AudioPhaserContext *s = ctx->priv; | |
| 238 | |||
| 239 | ✗ | av_freep(&s->delay_buffer); | |
| 240 | ✗ | av_freep(&s->modulation_buffer); | |
| 241 | ✗ | } | |
| 242 | |||
| 243 | static const AVFilterPad aphaser_inputs[] = { | ||
| 244 | { | ||
| 245 | .name = "default", | ||
| 246 | .type = AVMEDIA_TYPE_AUDIO, | ||
| 247 | .filter_frame = filter_frame, | ||
| 248 | }, | ||
| 249 | }; | ||
| 250 | |||
| 251 | static const AVFilterPad aphaser_outputs[] = { | ||
| 252 | { | ||
| 253 | .name = "default", | ||
| 254 | .type = AVMEDIA_TYPE_AUDIO, | ||
| 255 | .config_props = config_output, | ||
| 256 | }, | ||
| 257 | }; | ||
| 258 | |||
| 259 | const AVFilter ff_af_aphaser = { | ||
| 260 | .name = "aphaser", | ||
| 261 | .description = NULL_IF_CONFIG_SMALL("Add a phasing effect to the audio."), | ||
| 262 | .priv_size = sizeof(AudioPhaserContext), | ||
| 263 | .init = init, | ||
| 264 | .uninit = uninit, | ||
| 265 | FILTER_INPUTS(aphaser_inputs), | ||
| 266 | FILTER_OUTPUTS(aphaser_outputs), | ||
| 267 | FILTER_SAMPLEFMTS(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_DBLP, | ||
| 268 | AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLTP, | ||
| 269 | AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S32P, | ||
| 270 | AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16P), | ||
| 271 | .priv_class = &aphaser_class, | ||
| 272 | }; | ||
| 273 |