| 1 | /* |
| 2 | * Copyright (c) 2006 Rob Sykes <robs@users.sourceforge.net> |
| 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/avstring.h" |
| 22 | #include "libavutil/opt.h" |
| 23 | #include "libavutil/samplefmt.h" |
| 24 | #include "avfilter.h" |
| 25 | #include "audio.h" |
| 26 | #include "internal.h" |
| 27 | #include "generate_wave_table.h" |
| 28 | |
| 29 | #define INTERPOLATION_LINEAR 0 |
| 30 | #define INTERPOLATION_QUADRATIC 1 |
| 31 | |
| 32 | typedef struct FlangerContext { |
| 33 | const AVClass *class; |
| 34 | double delay_min; |
| 35 | double delay_depth; |
| 36 | double feedback_gain; |
| 37 | double delay_gain; |
| 38 | double speed; |
| 39 | int wave_shape; |
| 40 | double channel_phase; |
| 41 | int interpolation; |
| 42 | double in_gain; |
| 43 | int max_samples; |
| 44 | uint8_t **delay_buffer; |
| 45 | int delay_buf_pos; |
| 46 | double *delay_last; |
| 47 | float *lfo; |
| 48 | int lfo_length; |
| 49 | int lfo_pos; |
| 50 | } FlangerContext; |
| 51 | |
| 52 | #define OFFSET(x) offsetof(FlangerContext, x) |
| 53 | #define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM |
| 54 | |
| 55 | static const AVOption flanger_options[] = { |
| 56 | { "delay", "base delay in milliseconds", OFFSET(delay_min), AV_OPT_TYPE_DOUBLE, {.dbl=0}, 0, 30, A }, |
| 57 | { "depth", "added swept delay in milliseconds", OFFSET(delay_depth), AV_OPT_TYPE_DOUBLE, {.dbl=2}, 0, 10, A }, |
| 58 | { "regen", "percentage regeneration (delayed signal feedback)", OFFSET(feedback_gain), AV_OPT_TYPE_DOUBLE, {.dbl=0}, -95, 95, A }, |
| 59 | { "width", "percentage of delayed signal mixed with original", OFFSET(delay_gain), AV_OPT_TYPE_DOUBLE, {.dbl=71}, 0, 100, A }, |
| 60 | { "speed", "sweeps per second (Hz)", OFFSET(speed), AV_OPT_TYPE_DOUBLE, {.dbl=0.5}, 0.1, 10, A }, |
| 61 | { "shape", "swept wave shape", OFFSET(wave_shape), AV_OPT_TYPE_INT, {.i64=WAVE_SIN}, WAVE_SIN, WAVE_NB-1, A, "type" }, |
| 62 | { "triangular", NULL, 0, AV_OPT_TYPE_CONST, {.i64=WAVE_TRI}, 0, 0, A, "type" }, |
| 63 | { "t", NULL, 0, AV_OPT_TYPE_CONST, {.i64=WAVE_TRI}, 0, 0, A, "type" }, |
| 64 | { "sinusoidal", NULL, 0, AV_OPT_TYPE_CONST, {.i64=WAVE_SIN}, 0, 0, A, "type" }, |
| 65 | { "s", NULL, 0, AV_OPT_TYPE_CONST, {.i64=WAVE_SIN}, 0, 0, A, "type" }, |
| 66 | { "phase", "swept wave percentage phase-shift for multi-channel", OFFSET(channel_phase), AV_OPT_TYPE_DOUBLE, {.dbl=25}, 0, 100, A }, |
| 67 | { "interp", "delay-line interpolation", OFFSET(interpolation), AV_OPT_TYPE_INT, {.i64=0}, 0, 1, A, "itype" }, |
| 68 | { "linear", NULL, 0, AV_OPT_TYPE_CONST, {.i64=INTERPOLATION_LINEAR}, 0, 0, A, "itype" }, |
| 69 | { "quadratic", NULL, 0, AV_OPT_TYPE_CONST, {.i64=INTERPOLATION_QUADRATIC}, 0, 0, A, "itype" }, |
| 70 | { NULL } |
| 71 | }; |
| 72 | |
| 73 | AVFILTER_DEFINE_CLASS(flanger); |
| 74 | |
| 75 | static int init(AVFilterContext *ctx) |
| 76 | { |
| 77 | FlangerContext *s = ctx->priv; |
| 78 | |
| 79 | s->feedback_gain /= 100; |
| 80 | s->delay_gain /= 100; |
| 81 | s->channel_phase /= 100; |
| 82 | s->delay_min /= 1000; |
| 83 | s->delay_depth /= 1000; |
| 84 | s->in_gain = 1 / (1 + s->delay_gain); |
| 85 | s->delay_gain /= 1 + s->delay_gain; |
| 86 | s->delay_gain *= 1 - fabs(s->feedback_gain); |
| 87 | |
| 88 | return 0; |
| 89 | } |
| 90 | |
| 91 | static int query_formats(AVFilterContext *ctx) |
| 92 | { |
| 93 | AVFilterChannelLayouts *layouts; |
| 94 | AVFilterFormats *formats; |
| 95 | static const enum AVSampleFormat sample_fmts[] = { |
| 96 | AV_SAMPLE_FMT_DBLP, AV_SAMPLE_FMT_NONE |
| 97 | }; |
| 98 | |
| 99 | layouts = ff_all_channel_layouts(); |
| 100 | if (!layouts) |
| 101 | return AVERROR(ENOMEM); |
| 102 | ff_set_common_channel_layouts(ctx, layouts); |
| 103 | |
| 104 | formats = ff_make_format_list(sample_fmts); |
| 105 | if (!formats) |
| 106 | return AVERROR(ENOMEM); |
| 107 | ff_set_common_formats(ctx, formats); |
| 108 | |
| 109 | formats = ff_all_samplerates(); |
| 110 | if (!formats) |
| 111 | return AVERROR(ENOMEM); |
| 112 | ff_set_common_samplerates(ctx, formats); |
| 113 | |
| 114 | return 0; |
| 115 | } |
| 116 | |
| 117 | static int config_input(AVFilterLink *inlink) |
| 118 | { |
| 119 | AVFilterContext *ctx = inlink->dst; |
| 120 | FlangerContext *s = ctx->priv; |
| 121 | |
| 122 | s->max_samples = (s->delay_min + s->delay_depth) * inlink->sample_rate + 2.5; |
| 123 | s->lfo_length = inlink->sample_rate / s->speed; |
| 124 | s->delay_last = av_calloc(inlink->channels, sizeof(*s->delay_last)); |
| 125 | s->lfo = av_calloc(s->lfo_length, sizeof(*s->lfo)); |
| 126 | if (!s->lfo || !s->delay_last) |
| 127 | return AVERROR(ENOMEM); |
| 128 | |
| 129 | ff_generate_wave_table(s->wave_shape, AV_SAMPLE_FMT_FLT, s->lfo, s->lfo_length, |
| 130 | floor(s->delay_min * inlink->sample_rate + 0.5), |
| 131 | s->max_samples - 2., 3 * M_PI_2); |
| 132 | |
| 133 | return av_samples_alloc_array_and_samples(&s->delay_buffer, NULL, |
| 134 | inlink->channels, s->max_samples, |
| 135 | inlink->format, 0); |
| 136 | } |
| 137 | |
| 138 | static int filter_frame(AVFilterLink *inlink, AVFrame *frame) |
| 139 | { |
| 140 | AVFilterContext *ctx = inlink->dst; |
| 141 | FlangerContext *s = ctx->priv; |
| 142 | AVFrame *out_frame; |
| 143 | int chan, i; |
| 144 | |
| 145 | if (av_frame_is_writable(frame)) { |
| 146 | out_frame = frame; |
| 147 | } else { |
| 148 | out_frame = ff_get_audio_buffer(inlink, frame->nb_samples); |
| 149 | if (!out_frame) |
| 150 | return AVERROR(ENOMEM); |
| 151 | av_frame_copy_props(out_frame, frame); |
| 152 | } |
| 153 | |
| 154 | for (i = 0; i < frame->nb_samples; i++) { |
| 155 | |
| 156 | s->delay_buf_pos = (s->delay_buf_pos + s->max_samples - 1) % s->max_samples; |
| 157 | |
| 158 | for (chan = 0; chan < inlink->channels; chan++) { |
| 159 | double *src = (double *)frame->extended_data[chan]; |
| 160 | double *dst = (double *)out_frame->extended_data[chan]; |
| 161 | double delayed_0, delayed_1; |
| 162 | double delayed; |
| 163 | double in, out; |
| 164 | int channel_phase = chan * s->lfo_length * s->channel_phase + .5; |
| 165 | double delay = s->lfo[(s->lfo_pos + channel_phase) % s->lfo_length]; |
| 166 | int int_delay = (int)delay; |
| 167 | double frac_delay = modf(delay, &delay); |
| 168 | double *delay_buffer = (double *)s->delay_buffer[chan]; |
| 169 | |
| 170 | in = src[i]; |
| 171 | delay_buffer[s->delay_buf_pos] = in + s->delay_last[chan] * |
| 172 | s->feedback_gain; |
| 173 | delayed_0 = delay_buffer[(s->delay_buf_pos + int_delay++) % s->max_samples]; |
| 174 | delayed_1 = delay_buffer[(s->delay_buf_pos + int_delay++) % s->max_samples]; |
| 175 | |
| 176 | if (s->interpolation == INTERPOLATION_LINEAR) { |
| 177 | delayed = delayed_0 + (delayed_1 - delayed_0) * frac_delay; |
| 178 | } else { |
| 179 | double a, b; |
| 180 | double delayed_2 = delay_buffer[(s->delay_buf_pos + int_delay++) % s->max_samples]; |
| 181 | delayed_2 -= delayed_0; |
| 182 | delayed_1 -= delayed_0; |
| 183 | a = delayed_2 * .5 - delayed_1; |
| 184 | b = delayed_1 * 2 - delayed_2 *.5; |
| 185 | delayed = delayed_0 + (a * frac_delay + b) * frac_delay; |
| 186 | } |
| 187 | |
| 188 | s->delay_last[chan] = delayed; |
| 189 | out = in * s->in_gain + delayed * s->delay_gain; |
| 190 | dst[i] = out; |
| 191 | } |
| 192 | s->lfo_pos = (s->lfo_pos + 1) % s->lfo_length; |
| 193 | } |
| 194 | |
| 195 | if (frame != out_frame) |
| 196 | av_frame_free(&frame); |
| 197 | |
| 198 | return ff_filter_frame(ctx->outputs[0], out_frame); |
| 199 | } |
| 200 | |
| 201 | static av_cold void uninit(AVFilterContext *ctx) |
| 202 | { |
| 203 | FlangerContext *s = ctx->priv; |
| 204 | |
| 205 | av_freep(&s->lfo); |
| 206 | av_freep(&s->delay_last); |
| 207 | |
| 208 | if (s->delay_buffer) |
| 209 | av_freep(&s->delay_buffer[0]); |
| 210 | av_freep(&s->delay_buffer); |
| 211 | } |
| 212 | |
| 213 | static const AVFilterPad flanger_inputs[] = { |
| 214 | { |
| 215 | .name = "default", |
| 216 | .type = AVMEDIA_TYPE_AUDIO, |
| 217 | .config_props = config_input, |
| 218 | .filter_frame = filter_frame, |
| 219 | }, |
| 220 | { NULL } |
| 221 | }; |
| 222 | |
| 223 | static const AVFilterPad flanger_outputs[] = { |
| 224 | { |
| 225 | .name = "default", |
| 226 | .type = AVMEDIA_TYPE_AUDIO, |
| 227 | }, |
| 228 | { NULL } |
| 229 | }; |
| 230 | |
| 231 | AVFilter ff_af_flanger = { |
| 232 | .name = "flanger", |
| 233 | .description = NULL_IF_CONFIG_SMALL("Apply a flanging effect to the audio."), |
| 234 | .query_formats = query_formats, |
| 235 | .priv_size = sizeof(FlangerContext), |
| 236 | .priv_class = &flanger_class, |
| 237 | .init = init, |
| 238 | .uninit = uninit, |
| 239 | .inputs = flanger_inputs, |
| 240 | .outputs = flanger_outputs, |
| 241 | }; |