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 32typedef 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 55static 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 73AVFILTER_DEFINE_CLASS(flanger); 74 75static 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 91static 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 117static 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 138static 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 201static 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 213static 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 223static const AVFilterPad flanger_outputs[] = { 224 { 225 .name = "default", 226 .type = AVMEDIA_TYPE_AUDIO, 227 }, 228 { NULL } 229}; 230 231AVFilter 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}; 242