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af_flanger.c
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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;
40  double channel_phase;
42  double in_gain;
46  double *delay_last;
47  float *lfo;
49  int lfo_pos;
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 
92 {
95  static const enum AVSampleFormat sample_fmts[] = {
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 
130  floor(s->delay_min * inlink->sample_rate + 0.5),
131  s->max_samples - 2., 3 * M_PI_2);
132 
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 
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 
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 };