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 av_cold 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
100 layouts = ff_all_channel_counts();
102 return AVERROR(ENOMEM);
103 ret = ff_set_common_channel_layouts(ctx, layouts);
107 formats = ff_make_format_list(sample_fmts);
109 return AVERROR(ENOMEM);
110 ret = ff_set_common_formats(ctx, formats);
114 formats = ff_all_samplerates();
116 return AVERROR(ENOMEM);
117 return ff_set_common_samplerates(ctx, formats);
120 static int config_input(AVFilterLink *inlink)
122 AVFilterContext *ctx = inlink->dst;
123 FlangerContext *s = ctx->priv;
125 s->max_samples = (s->delay_min + s->delay_depth) * inlink->sample_rate + 2.5;
126 s->lfo_length = inlink->sample_rate / s->speed;
127 s->delay_last = av_calloc(inlink->channels, sizeof(*s->delay_last));
128 s->lfo = av_calloc(s->lfo_length, sizeof(*s->lfo));
129 if (!s->lfo || !s->delay_last)
130 return AVERROR(ENOMEM);
132 ff_generate_wave_table(s->wave_shape, AV_SAMPLE_FMT_FLT, s->lfo, s->lfo_length,
133 rint(s->delay_min * inlink->sample_rate),
134 s->max_samples - 2., 3 * M_PI_2);
136 return av_samples_alloc_array_and_samples(&s->delay_buffer, NULL,
137 inlink->channels, s->max_samples,
141 static int filter_frame(AVFilterLink *inlink, AVFrame *frame)
143 AVFilterContext *ctx = inlink->dst;
144 FlangerContext *s = ctx->priv;
148 if (av_frame_is_writable(frame)) {
151 out_frame = ff_get_audio_buffer(ctx->outputs[0], frame->nb_samples);
153 av_frame_free(&frame);
154 return AVERROR(ENOMEM);
156 av_frame_copy_props(out_frame, frame);
159 for (i = 0; i < frame->nb_samples; i++) {
161 s->delay_buf_pos = (s->delay_buf_pos + s->max_samples - 1) % s->max_samples;
163 for (chan = 0; chan < inlink->channels; chan++) {
164 double *src = (double *)frame->extended_data[chan];
165 double *dst = (double *)out_frame->extended_data[chan];
166 double delayed_0, delayed_1;
169 int channel_phase = chan * s->lfo_length * s->channel_phase + .5;
170 double delay = s->lfo[(s->lfo_pos + channel_phase) % s->lfo_length];
171 int int_delay = (int)delay;
172 double frac_delay = modf(delay, &delay);
173 double *delay_buffer = (double *)s->delay_buffer[chan];
176 delay_buffer[s->delay_buf_pos] = in + s->delay_last[chan] *
178 delayed_0 = delay_buffer[(s->delay_buf_pos + int_delay++) % s->max_samples];
179 delayed_1 = delay_buffer[(s->delay_buf_pos + int_delay++) % s->max_samples];
181 if (s->interpolation == INTERPOLATION_LINEAR) {
182 delayed = delayed_0 + (delayed_1 - delayed_0) * frac_delay;
185 double delayed_2 = delay_buffer[(s->delay_buf_pos + int_delay++) % s->max_samples];
186 delayed_2 -= delayed_0;
187 delayed_1 -= delayed_0;
188 a = delayed_2 * .5 - delayed_1;
189 b = delayed_1 * 2 - delayed_2 *.5;
190 delayed = delayed_0 + (a * frac_delay + b) * frac_delay;
193 s->delay_last[chan] = delayed;
194 out = in * s->in_gain + delayed * s->delay_gain;
197 s->lfo_pos = (s->lfo_pos + 1) % s->lfo_length;
200 if (frame != out_frame)
201 av_frame_free(&frame);
203 return ff_filter_frame(ctx->outputs[0], out_frame);
206 static av_cold void uninit(AVFilterContext *ctx)
208 FlangerContext *s = ctx->priv;
211 av_freep(&s->delay_last);
214 av_freep(&s->delay_buffer[0]);
215 av_freep(&s->delay_buffer);
218 static const AVFilterPad flanger_inputs[] = {
221 .type = AVMEDIA_TYPE_AUDIO,
222 .config_props = config_input,
223 .filter_frame = filter_frame,
228 static const AVFilterPad flanger_outputs[] = {
231 .type = AVMEDIA_TYPE_AUDIO,
236 const AVFilter ff_af_flanger = {
238 .description = NULL_IF_CONFIG_SMALL("Apply a flanging effect to the audio."),
239 .query_formats = query_formats,
240 .priv_size = sizeof(FlangerContext),
241 .priv_class = &flanger_class,
244 .inputs = flanger_inputs,
245 .outputs = flanger_outputs,