2 * Copyright (c) 2006 Rob Sykes <robs@users.sourceforge.net>
4 * This file is part of FFmpeg.
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.
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.
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
21 #include "libavutil/avstring.h"
22 #include "libavutil/opt.h"
23 #include "libavutil/samplefmt.h"
27 #include "generate_wave_table.h"
29 #define INTERPOLATION_LINEAR 0
30 #define INTERPOLATION_QUADRATIC 1
32 typedef struct FlangerContext {
44 uint8_t **delay_buffer;
52 #define OFFSET(x) offsetof(FlangerContext, x)
53 #define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
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" },
73 AVFILTER_DEFINE_CLASS(flanger);
75 static int init(AVFilterContext *ctx)
77 FlangerContext *s = ctx->priv;
79 s->feedback_gain /= 100;
81 s->channel_phase /= 100;
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);
91 static int query_formats(AVFilterContext *ctx)
93 AVFilterChannelLayouts *layouts;
94 AVFilterFormats *formats;
95 static const enum AVSampleFormat sample_fmts[] = {
96 AV_SAMPLE_FMT_DBLP, AV_SAMPLE_FMT_NONE
99 layouts = ff_all_channel_layouts();
101 return AVERROR(ENOMEM);
102 ff_set_common_channel_layouts(ctx, layouts);
104 formats = ff_make_format_list(sample_fmts);
106 return AVERROR(ENOMEM);
107 ff_set_common_formats(ctx, formats);
109 formats = ff_all_samplerates();
111 return AVERROR(ENOMEM);
112 ff_set_common_samplerates(ctx, formats);
117 static int config_input(AVFilterLink *inlink)
119 AVFilterContext *ctx = inlink->dst;
120 FlangerContext *s = ctx->priv;
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);
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);
133 return av_samples_alloc_array_and_samples(&s->delay_buffer, NULL,
134 inlink->channels, s->max_samples,
138 static int filter_frame(AVFilterLink *inlink, AVFrame *frame)
140 AVFilterContext *ctx = inlink->dst;
141 FlangerContext *s = ctx->priv;
145 if (av_frame_is_writable(frame)) {
148 out_frame = ff_get_audio_buffer(inlink, frame->nb_samples);
150 return AVERROR(ENOMEM);
151 av_frame_copy_props(out_frame, frame);
154 for (i = 0; i < frame->nb_samples; i++) {
156 s->delay_buf_pos = (s->delay_buf_pos + s->max_samples - 1) % s->max_samples;
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;
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];
171 delay_buffer[s->delay_buf_pos] = in + s->delay_last[chan] *
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];
176 if (s->interpolation == INTERPOLATION_LINEAR) {
177 delayed = delayed_0 + (delayed_1 - delayed_0) * frac_delay;
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;
188 s->delay_last[chan] = delayed;
189 out = in * s->in_gain + delayed * s->delay_gain;
192 s->lfo_pos = (s->lfo_pos + 1) % s->lfo_length;
195 if (frame != out_frame)
196 av_frame_free(&frame);
198 return ff_filter_frame(ctx->outputs[0], out_frame);
201 static av_cold void uninit(AVFilterContext *ctx)
203 FlangerContext *s = ctx->priv;
206 av_freep(&s->delay_last);
209 av_freep(&s->delay_buffer[0]);
210 av_freep(&s->delay_buffer);
213 static const AVFilterPad flanger_inputs[] = {
216 .type = AVMEDIA_TYPE_AUDIO,
217 .config_props = config_input,
218 .filter_frame = filter_frame,
223 static const AVFilterPad flanger_outputs[] = {
226 .type = AVMEDIA_TYPE_AUDIO,
231 AVFilter ff_af_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,
239 .inputs = flanger_inputs,
240 .outputs = flanger_outputs,