| 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/opt.h" |
| 28 | #include "audio.h" |
| 29 | #include "avfilter.h" |
| 30 | #include "internal.h" |
| 31 | #include "generate_wave_table.h" |
| 32 | |
| 33 | typedef struct AudioPhaserContext { |
| 34 | const AVClass *class; |
| 35 | double in_gain, out_gain; |
| 36 | double delay; |
| 37 | double decay; |
| 38 | double speed; |
| 39 | |
| 40 | enum WaveType type; |
| 41 | |
| 42 | int delay_buffer_length; |
| 43 | double *delay_buffer; |
| 44 | |
| 45 | int modulation_buffer_length; |
| 46 | int32_t *modulation_buffer; |
| 47 | |
| 48 | int delay_pos, modulation_pos; |
| 49 | |
| 50 | void (*phaser)(struct AudioPhaserContext *p, |
| 51 | uint8_t * const *src, uint8_t **dst, |
| 52 | int nb_samples, int channels); |
| 53 | } AudioPhaserContext; |
| 54 | |
| 55 | #define OFFSET(x) offsetof(AudioPhaserContext, x) |
| 56 | #define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM |
| 57 | |
| 58 | static const AVOption aphaser_options[] = { |
| 59 | { "in_gain", "set input gain", OFFSET(in_gain), AV_OPT_TYPE_DOUBLE, {.dbl=.4}, 0, 1, FLAGS }, |
| 60 | { "out_gain", "set output gain", OFFSET(out_gain), AV_OPT_TYPE_DOUBLE, {.dbl=.74}, 0, 1e9, FLAGS }, |
| 61 | { "delay", "set delay in milliseconds", OFFSET(delay), AV_OPT_TYPE_DOUBLE, {.dbl=3.}, 0, 5, FLAGS }, |
| 62 | { "decay", "set decay", OFFSET(decay), AV_OPT_TYPE_DOUBLE, {.dbl=.4}, 0, .99, FLAGS }, |
| 63 | { "speed", "set modulation speed", OFFSET(speed), AV_OPT_TYPE_DOUBLE, {.dbl=.5}, .1, 2, FLAGS }, |
| 64 | { "type", "set modulation type", OFFSET(type), AV_OPT_TYPE_INT, {.i64=WAVE_TRI}, 0, WAVE_NB-1, FLAGS, "type" }, |
| 65 | { "triangular", NULL, 0, AV_OPT_TYPE_CONST, {.i64=WAVE_TRI}, 0, 0, FLAGS, "type" }, |
| 66 | { "t", NULL, 0, AV_OPT_TYPE_CONST, {.i64=WAVE_TRI}, 0, 0, FLAGS, "type" }, |
| 67 | { "sinusoidal", NULL, 0, AV_OPT_TYPE_CONST, {.i64=WAVE_SIN}, 0, 0, FLAGS, "type" }, |
| 68 | { "s", NULL, 0, AV_OPT_TYPE_CONST, {.i64=WAVE_SIN}, 0, 0, FLAGS, "type" }, |
| 69 | { NULL } |
| 70 | }; |
| 71 | |
| 72 | AVFILTER_DEFINE_CLASS(aphaser); |
| 73 | |
| 74 | static av_cold int init(AVFilterContext *ctx) |
| 75 | { |
| 76 | AudioPhaserContext *p = ctx->priv; |
| 77 | |
| 78 | if (p->in_gain > (1 - p->decay * p->decay)) |
| 79 | av_log(ctx, AV_LOG_WARNING, "in_gain may cause clipping\n"); |
| 80 | if (p->in_gain / (1 - p->decay) > 1 / p->out_gain) |
| 81 | av_log(ctx, AV_LOG_WARNING, "out_gain may cause clipping\n"); |
| 82 | |
| 83 | return 0; |
| 84 | } |
| 85 | |
| 86 | static int query_formats(AVFilterContext *ctx) |
| 87 | { |
| 88 | AVFilterFormats *formats; |
| 89 | AVFilterChannelLayouts *layouts; |
| 90 | static const enum AVSampleFormat sample_fmts[] = { |
| 91 | AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_DBLP, |
| 92 | AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLTP, |
| 93 | AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S32P, |
| 94 | AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16P, |
| 95 | AV_SAMPLE_FMT_NONE |
| 96 | }; |
| 97 | |
| 98 | layouts = ff_all_channel_layouts(); |
| 99 | if (!layouts) |
| 100 | return AVERROR(ENOMEM); |
| 101 | ff_set_common_channel_layouts(ctx, layouts); |
| 102 | |
| 103 | formats = ff_make_format_list(sample_fmts); |
| 104 | if (!formats) |
| 105 | return AVERROR(ENOMEM); |
| 106 | ff_set_common_formats(ctx, formats); |
| 107 | |
| 108 | formats = ff_all_samplerates(); |
| 109 | if (!formats) |
| 110 | return AVERROR(ENOMEM); |
| 111 | ff_set_common_samplerates(ctx, formats); |
| 112 | |
| 113 | return 0; |
| 114 | } |
| 115 | |
| 116 | #define MOD(a, b) (((a) >= (b)) ? (a) - (b) : (a)) |
| 117 | |
| 118 | #define PHASER_PLANAR(name, type) \ |
| 119 | static void phaser_## name ##p(AudioPhaserContext *p, \ |
| 120 | uint8_t * const *src, uint8_t **dst, \ |
| 121 | int nb_samples, int channels) \ |
| 122 | { \ |
| 123 | int i, c, delay_pos, modulation_pos; \ |
| 124 | \ |
| 125 | av_assert0(channels > 0); \ |
| 126 | for (c = 0; c < channels; c++) { \ |
| 127 | type *s = (type *)src[c]; \ |
| 128 | type *d = (type *)dst[c]; \ |
| 129 | double *buffer = p->delay_buffer + \ |
| 130 | c * p->delay_buffer_length; \ |
| 131 | \ |
| 132 | delay_pos = p->delay_pos; \ |
| 133 | modulation_pos = p->modulation_pos; \ |
| 134 | \ |
| 135 | for (i = 0; i < nb_samples; i++, s++, d++) { \ |
| 136 | double v = *s * p->in_gain + buffer[ \ |
| 137 | MOD(delay_pos + p->modulation_buffer[ \ |
| 138 | modulation_pos], \ |
| 139 | p->delay_buffer_length)] * p->decay; \ |
| 140 | \ |
| 141 | modulation_pos = MOD(modulation_pos + 1, \ |
| 142 | p->modulation_buffer_length); \ |
| 143 | delay_pos = MOD(delay_pos + 1, p->delay_buffer_length); \ |
| 144 | buffer[delay_pos] = v; \ |
| 145 | \ |
| 146 | *d = v * p->out_gain; \ |
| 147 | } \ |
| 148 | } \ |
| 149 | \ |
| 150 | p->delay_pos = delay_pos; \ |
| 151 | p->modulation_pos = modulation_pos; \ |
| 152 | } |
| 153 | |
| 154 | #define PHASER(name, type) \ |
| 155 | static void phaser_## name (AudioPhaserContext *p, \ |
| 156 | uint8_t * const *src, uint8_t **dst, \ |
| 157 | int nb_samples, int channels) \ |
| 158 | { \ |
| 159 | int i, c, delay_pos, modulation_pos; \ |
| 160 | type *s = (type *)src[0]; \ |
| 161 | type *d = (type *)dst[0]; \ |
| 162 | double *buffer = p->delay_buffer; \ |
| 163 | \ |
| 164 | delay_pos = p->delay_pos; \ |
| 165 | modulation_pos = p->modulation_pos; \ |
| 166 | \ |
| 167 | for (i = 0; i < nb_samples; i++) { \ |
| 168 | int pos = MOD(delay_pos + p->modulation_buffer[modulation_pos], \ |
| 169 | p->delay_buffer_length) * channels; \ |
| 170 | int npos; \ |
| 171 | \ |
| 172 | delay_pos = MOD(delay_pos + 1, p->delay_buffer_length); \ |
| 173 | npos = delay_pos * channels; \ |
| 174 | for (c = 0; c < channels; c++, s++, d++) { \ |
| 175 | double v = *s * p->in_gain + buffer[pos + c] * p->decay; \ |
| 176 | \ |
| 177 | buffer[npos + c] = v; \ |
| 178 | \ |
| 179 | *d = v * p->out_gain; \ |
| 180 | } \ |
| 181 | \ |
| 182 | modulation_pos = MOD(modulation_pos + 1, \ |
| 183 | p->modulation_buffer_length); \ |
| 184 | } \ |
| 185 | \ |
| 186 | p->delay_pos = delay_pos; \ |
| 187 | p->modulation_pos = modulation_pos; \ |
| 188 | } |
| 189 | |
| 190 | PHASER_PLANAR(dbl, double) |
| 191 | PHASER_PLANAR(flt, float) |
| 192 | PHASER_PLANAR(s16, int16_t) |
| 193 | PHASER_PLANAR(s32, int32_t) |
| 194 | |
| 195 | PHASER(dbl, double) |
| 196 | PHASER(flt, float) |
| 197 | PHASER(s16, int16_t) |
| 198 | PHASER(s32, int32_t) |
| 199 | |
| 200 | static int config_output(AVFilterLink *outlink) |
| 201 | { |
| 202 | AudioPhaserContext *p = outlink->src->priv; |
| 203 | AVFilterLink *inlink = outlink->src->inputs[0]; |
| 204 | |
| 205 | p->delay_buffer_length = p->delay * 0.001 * inlink->sample_rate + 0.5; |
| 206 | p->delay_buffer = av_calloc(p->delay_buffer_length, sizeof(*p->delay_buffer) * inlink->channels); |
| 207 | p->modulation_buffer_length = inlink->sample_rate / p->speed + 0.5; |
| 208 | p->modulation_buffer = av_malloc_array(p->modulation_buffer_length, sizeof(*p->modulation_buffer)); |
| 209 | |
| 210 | if (!p->modulation_buffer || !p->delay_buffer) |
| 211 | return AVERROR(ENOMEM); |
| 212 | |
| 213 | ff_generate_wave_table(p->type, AV_SAMPLE_FMT_S32, |
| 214 | p->modulation_buffer, p->modulation_buffer_length, |
| 215 | 1., p->delay_buffer_length, M_PI / 2.0); |
| 216 | |
| 217 | p->delay_pos = p->modulation_pos = 0; |
| 218 | |
| 219 | switch (inlink->format) { |
| 220 | case AV_SAMPLE_FMT_DBL: p->phaser = phaser_dbl; break; |
| 221 | case AV_SAMPLE_FMT_DBLP: p->phaser = phaser_dblp; break; |
| 222 | case AV_SAMPLE_FMT_FLT: p->phaser = phaser_flt; break; |
| 223 | case AV_SAMPLE_FMT_FLTP: p->phaser = phaser_fltp; break; |
| 224 | case AV_SAMPLE_FMT_S16: p->phaser = phaser_s16; break; |
| 225 | case AV_SAMPLE_FMT_S16P: p->phaser = phaser_s16p; break; |
| 226 | case AV_SAMPLE_FMT_S32: p->phaser = phaser_s32; break; |
| 227 | case AV_SAMPLE_FMT_S32P: p->phaser = phaser_s32p; break; |
| 228 | default: av_assert0(0); |
| 229 | } |
| 230 | |
| 231 | return 0; |
| 232 | } |
| 233 | |
| 234 | static int filter_frame(AVFilterLink *inlink, AVFrame *inbuf) |
| 235 | { |
| 236 | AudioPhaserContext *p = inlink->dst->priv; |
| 237 | AVFilterLink *outlink = inlink->dst->outputs[0]; |
| 238 | AVFrame *outbuf; |
| 239 | |
| 240 | if (av_frame_is_writable(inbuf)) { |
| 241 | outbuf = inbuf; |
| 242 | } else { |
| 243 | outbuf = ff_get_audio_buffer(inlink, inbuf->nb_samples); |
| 244 | if (!outbuf) |
| 245 | return AVERROR(ENOMEM); |
| 246 | av_frame_copy_props(outbuf, inbuf); |
| 247 | } |
| 248 | |
| 249 | p->phaser(p, inbuf->extended_data, outbuf->extended_data, |
| 250 | outbuf->nb_samples, av_frame_get_channels(outbuf)); |
| 251 | |
| 252 | if (inbuf != outbuf) |
| 253 | av_frame_free(&inbuf); |
| 254 | |
| 255 | return ff_filter_frame(outlink, outbuf); |
| 256 | } |
| 257 | |
| 258 | static av_cold void uninit(AVFilterContext *ctx) |
| 259 | { |
| 260 | AudioPhaserContext *p = ctx->priv; |
| 261 | |
| 262 | av_freep(&p->delay_buffer); |
| 263 | av_freep(&p->modulation_buffer); |
| 264 | } |
| 265 | |
| 266 | static const AVFilterPad aphaser_inputs[] = { |
| 267 | { |
| 268 | .name = "default", |
| 269 | .type = AVMEDIA_TYPE_AUDIO, |
| 270 | .filter_frame = filter_frame, |
| 271 | }, |
| 272 | { NULL } |
| 273 | }; |
| 274 | |
| 275 | static const AVFilterPad aphaser_outputs[] = { |
| 276 | { |
| 277 | .name = "default", |
| 278 | .type = AVMEDIA_TYPE_AUDIO, |
| 279 | .config_props = config_output, |
| 280 | }, |
| 281 | { NULL } |
| 282 | }; |
| 283 | |
| 284 | AVFilter ff_af_aphaser = { |
| 285 | .name = "aphaser", |
| 286 | .description = NULL_IF_CONFIG_SMALL("Add a phasing effect to the audio."), |
| 287 | .query_formats = query_formats, |
| 288 | .priv_size = sizeof(AudioPhaserContext), |
| 289 | .init = init, |
| 290 | .uninit = uninit, |
| 291 | .inputs = aphaser_inputs, |
| 292 | .outputs = aphaser_outputs, |
| 293 | .priv_class = &aphaser_class, |
| 294 | }; |