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 av_cold 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 |
|
|
int ret; |
99 |
|
|
|
100 |
|
|
layouts = ff_all_channel_counts(); |
101 |
|
|
if (!layouts) |
102 |
|
|
return AVERROR(ENOMEM); |
103 |
|
|
ret = ff_set_common_channel_layouts(ctx, layouts); |
104 |
|
|
if (ret < 0) |
105 |
|
|
return ret; |
106 |
|
|
|
107 |
|
|
formats = ff_make_format_list(sample_fmts); |
108 |
|
|
if (!formats) |
109 |
|
|
return AVERROR(ENOMEM); |
110 |
|
|
ret = ff_set_common_formats(ctx, formats); |
111 |
|
|
if (ret < 0) |
112 |
|
|
return ret; |
113 |
|
|
|
114 |
|
|
formats = ff_all_samplerates(); |
115 |
|
|
if (!formats) |
116 |
|
|
return AVERROR(ENOMEM); |
117 |
|
|
return ff_set_common_samplerates(ctx, formats); |
118 |
|
|
} |
119 |
|
|
|
120 |
|
|
static int config_input(AVFilterLink *inlink) |
121 |
|
|
{ |
122 |
|
|
AVFilterContext *ctx = inlink->dst; |
123 |
|
|
FlangerContext *s = ctx->priv; |
124 |
|
|
|
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); |
131 |
|
|
|
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); |
135 |
|
|
|
136 |
|
|
return av_samples_alloc_array_and_samples(&s->delay_buffer, NULL, |
137 |
|
|
inlink->channels, s->max_samples, |
138 |
|
|
inlink->format, 0); |
139 |
|
|
} |
140 |
|
|
|
141 |
|
|
static int filter_frame(AVFilterLink *inlink, AVFrame *frame) |
142 |
|
|
{ |
143 |
|
|
AVFilterContext *ctx = inlink->dst; |
144 |
|
|
FlangerContext *s = ctx->priv; |
145 |
|
|
AVFrame *out_frame; |
146 |
|
|
int chan, i; |
147 |
|
|
|
148 |
|
|
if (av_frame_is_writable(frame)) { |
149 |
|
|
out_frame = frame; |
150 |
|
|
} else { |
151 |
|
|
out_frame = ff_get_audio_buffer(ctx->outputs[0], frame->nb_samples); |
152 |
|
|
if (!out_frame) { |
153 |
|
|
av_frame_free(&frame); |
154 |
|
|
return AVERROR(ENOMEM); |
155 |
|
|
} |
156 |
|
|
av_frame_copy_props(out_frame, frame); |
157 |
|
|
} |
158 |
|
|
|
159 |
|
|
for (i = 0; i < frame->nb_samples; i++) { |
160 |
|
|
|
161 |
|
|
s->delay_buf_pos = (s->delay_buf_pos + s->max_samples - 1) % s->max_samples; |
162 |
|
|
|
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; |
167 |
|
|
double delayed; |
168 |
|
|
double in, out; |
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]; |
174 |
|
|
|
175 |
|
|
in = src[i]; |
176 |
|
|
delay_buffer[s->delay_buf_pos] = in + s->delay_last[chan] * |
177 |
|
|
s->feedback_gain; |
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]; |
180 |
|
|
|
181 |
|
|
if (s->interpolation == INTERPOLATION_LINEAR) { |
182 |
|
|
delayed = delayed_0 + (delayed_1 - delayed_0) * frac_delay; |
183 |
|
|
} else { |
184 |
|
|
double a, b; |
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; |
191 |
|
|
} |
192 |
|
|
|
193 |
|
|
s->delay_last[chan] = delayed; |
194 |
|
|
out = in * s->in_gain + delayed * s->delay_gain; |
195 |
|
|
dst[i] = out; |
196 |
|
|
} |
197 |
|
|
s->lfo_pos = (s->lfo_pos + 1) % s->lfo_length; |
198 |
|
|
} |
199 |
|
|
|
200 |
|
|
if (frame != out_frame) |
201 |
|
|
av_frame_free(&frame); |
202 |
|
|
|
203 |
|
|
return ff_filter_frame(ctx->outputs[0], out_frame); |
204 |
|
|
} |
205 |
|
|
|
206 |
|
|
static av_cold void uninit(AVFilterContext *ctx) |
207 |
|
|
{ |
208 |
|
|
FlangerContext *s = ctx->priv; |
209 |
|
|
|
210 |
|
|
av_freep(&s->lfo); |
211 |
|
|
av_freep(&s->delay_last); |
212 |
|
|
|
213 |
|
|
if (s->delay_buffer) |
214 |
|
|
av_freep(&s->delay_buffer[0]); |
215 |
|
|
av_freep(&s->delay_buffer); |
216 |
|
|
} |
217 |
|
|
|
218 |
|
|
static const AVFilterPad flanger_inputs[] = { |
219 |
|
|
{ |
220 |
|
|
.name = "default", |
221 |
|
|
.type = AVMEDIA_TYPE_AUDIO, |
222 |
|
|
.config_props = config_input, |
223 |
|
|
.filter_frame = filter_frame, |
224 |
|
|
}, |
225 |
|
|
{ NULL } |
226 |
|
|
}; |
227 |
|
|
|
228 |
|
|
static const AVFilterPad flanger_outputs[] = { |
229 |
|
|
{ |
230 |
|
|
.name = "default", |
231 |
|
|
.type = AVMEDIA_TYPE_AUDIO, |
232 |
|
|
}, |
233 |
|
|
{ NULL } |
234 |
|
|
}; |
235 |
|
|
|
236 |
|
|
AVFilter ff_af_flanger = { |
237 |
|
|
.name = "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, |
242 |
|
|
.init = init, |
243 |
|
|
.uninit = uninit, |
244 |
|
|
.inputs = flanger_inputs, |
245 |
|
|
.outputs = flanger_outputs, |
246 |
|
|
}; |