| Line | Branch | Exec | Source |
|---|---|---|---|
| 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/mem.h" | ||
| 22 | #include "libavutil/opt.h" | ||
| 23 | #include "libavutil/samplefmt.h" | ||
| 24 | #include "avfilter.h" | ||
| 25 | #include "audio.h" | ||
| 26 | #include "filters.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, .unit = "type" }, | ||
| 62 | { "triangular", NULL, 0, AV_OPT_TYPE_CONST, {.i64=WAVE_TRI}, 0, 0, A, .unit = "type" }, | ||
| 63 | { "t", NULL, 0, AV_OPT_TYPE_CONST, {.i64=WAVE_TRI}, 0, 0, A, .unit = "type" }, | ||
| 64 | { "sinusoidal", NULL, 0, AV_OPT_TYPE_CONST, {.i64=WAVE_SIN}, 0, 0, A, .unit = "type" }, | ||
| 65 | { "s", NULL, 0, AV_OPT_TYPE_CONST, {.i64=WAVE_SIN}, 0, 0, A, .unit = "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, .unit = "itype" }, | ||
| 68 | { "linear", NULL, 0, AV_OPT_TYPE_CONST, {.i64=INTERPOLATION_LINEAR}, 0, 0, A, .unit = "itype" }, | ||
| 69 | { "quadratic", NULL, 0, AV_OPT_TYPE_CONST, {.i64=INTERPOLATION_QUADRATIC}, 0, 0, A, .unit = "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 config_input(AVFilterLink *inlink) | |
| 92 | { | ||
| 93 | ✗ | AVFilterContext *ctx = inlink->dst; | |
| 94 | ✗ | FlangerContext *s = ctx->priv; | |
| 95 | |||
| 96 | ✗ | s->max_samples = (s->delay_min + s->delay_depth) * inlink->sample_rate + 2.5; | |
| 97 | ✗ | s->lfo_length = inlink->sample_rate / s->speed; | |
| 98 | ✗ | s->delay_last = av_calloc(inlink->ch_layout.nb_channels, sizeof(*s->delay_last)); | |
| 99 | ✗ | s->lfo = av_calloc(s->lfo_length, sizeof(*s->lfo)); | |
| 100 | ✗ | if (!s->lfo || !s->delay_last) | |
| 101 | ✗ | return AVERROR(ENOMEM); | |
| 102 | |||
| 103 | ✗ | ff_generate_wave_table(s->wave_shape, AV_SAMPLE_FMT_FLT, s->lfo, s->lfo_length, | |
| 104 | ✗ | rint(s->delay_min * inlink->sample_rate), | |
| 105 | ✗ | s->max_samples - 2., 3 * M_PI_2); | |
| 106 | |||
| 107 | ✗ | return av_samples_alloc_array_and_samples(&s->delay_buffer, NULL, | |
| 108 | inlink->ch_layout.nb_channels, s->max_samples, | ||
| 109 | ✗ | inlink->format, 0); | |
| 110 | } | ||
| 111 | |||
| 112 | ✗ | static int filter_frame(AVFilterLink *inlink, AVFrame *frame) | |
| 113 | { | ||
| 114 | ✗ | AVFilterContext *ctx = inlink->dst; | |
| 115 | ✗ | FlangerContext *s = ctx->priv; | |
| 116 | AVFrame *out_frame; | ||
| 117 | int chan, i; | ||
| 118 | |||
| 119 | ✗ | if (av_frame_is_writable(frame)) { | |
| 120 | ✗ | out_frame = frame; | |
| 121 | } else { | ||
| 122 | ✗ | out_frame = ff_get_audio_buffer(ctx->outputs[0], frame->nb_samples); | |
| 123 | ✗ | if (!out_frame) { | |
| 124 | ✗ | av_frame_free(&frame); | |
| 125 | ✗ | return AVERROR(ENOMEM); | |
| 126 | } | ||
| 127 | ✗ | av_frame_copy_props(out_frame, frame); | |
| 128 | } | ||
| 129 | |||
| 130 | ✗ | for (i = 0; i < frame->nb_samples; i++) { | |
| 131 | |||
| 132 | ✗ | s->delay_buf_pos = (s->delay_buf_pos + s->max_samples - 1) % s->max_samples; | |
| 133 | |||
| 134 | ✗ | for (chan = 0; chan < inlink->ch_layout.nb_channels; chan++) { | |
| 135 | ✗ | double *src = (double *)frame->extended_data[chan]; | |
| 136 | ✗ | double *dst = (double *)out_frame->extended_data[chan]; | |
| 137 | double delayed_0, delayed_1; | ||
| 138 | double delayed; | ||
| 139 | double in, out; | ||
| 140 | ✗ | int channel_phase = chan * s->lfo_length * s->channel_phase + .5; | |
| 141 | ✗ | double delay = s->lfo[(s->lfo_pos + channel_phase) % s->lfo_length]; | |
| 142 | ✗ | int int_delay = (int)delay; | |
| 143 | ✗ | double frac_delay = modf(delay, &delay); | |
| 144 | ✗ | double *delay_buffer = (double *)s->delay_buffer[chan]; | |
| 145 | |||
| 146 | ✗ | in = src[i]; | |
| 147 | ✗ | delay_buffer[s->delay_buf_pos] = in + s->delay_last[chan] * | |
| 148 | ✗ | s->feedback_gain; | |
| 149 | ✗ | delayed_0 = delay_buffer[(s->delay_buf_pos + int_delay++) % s->max_samples]; | |
| 150 | ✗ | delayed_1 = delay_buffer[(s->delay_buf_pos + int_delay++) % s->max_samples]; | |
| 151 | |||
| 152 | ✗ | if (s->interpolation == INTERPOLATION_LINEAR) { | |
| 153 | ✗ | delayed = delayed_0 + (delayed_1 - delayed_0) * frac_delay; | |
| 154 | } else { | ||
| 155 | double a, b; | ||
| 156 | ✗ | double delayed_2 = delay_buffer[(s->delay_buf_pos + int_delay++) % s->max_samples]; | |
| 157 | ✗ | delayed_2 -= delayed_0; | |
| 158 | ✗ | delayed_1 -= delayed_0; | |
| 159 | ✗ | a = delayed_2 * .5 - delayed_1; | |
| 160 | ✗ | b = delayed_1 * 2 - delayed_2 *.5; | |
| 161 | ✗ | delayed = delayed_0 + (a * frac_delay + b) * frac_delay; | |
| 162 | } | ||
| 163 | |||
| 164 | ✗ | s->delay_last[chan] = delayed; | |
| 165 | ✗ | out = in * s->in_gain + delayed * s->delay_gain; | |
| 166 | ✗ | dst[i] = out; | |
| 167 | } | ||
| 168 | ✗ | s->lfo_pos = (s->lfo_pos + 1) % s->lfo_length; | |
| 169 | } | ||
| 170 | |||
| 171 | ✗ | if (frame != out_frame) | |
| 172 | ✗ | av_frame_free(&frame); | |
| 173 | |||
| 174 | ✗ | return ff_filter_frame(ctx->outputs[0], out_frame); | |
| 175 | } | ||
| 176 | |||
| 177 | ✗ | static av_cold void uninit(AVFilterContext *ctx) | |
| 178 | { | ||
| 179 | ✗ | FlangerContext *s = ctx->priv; | |
| 180 | |||
| 181 | ✗ | av_freep(&s->lfo); | |
| 182 | ✗ | av_freep(&s->delay_last); | |
| 183 | |||
| 184 | ✗ | if (s->delay_buffer) | |
| 185 | ✗ | av_freep(&s->delay_buffer[0]); | |
| 186 | ✗ | av_freep(&s->delay_buffer); | |
| 187 | ✗ | } | |
| 188 | |||
| 189 | static const AVFilterPad flanger_inputs[] = { | ||
| 190 | { | ||
| 191 | .name = "default", | ||
| 192 | .type = AVMEDIA_TYPE_AUDIO, | ||
| 193 | .config_props = config_input, | ||
| 194 | .filter_frame = filter_frame, | ||
| 195 | }, | ||
| 196 | }; | ||
| 197 | |||
| 198 | const AVFilter ff_af_flanger = { | ||
| 199 | .name = "flanger", | ||
| 200 | .description = NULL_IF_CONFIG_SMALL("Apply a flanging effect to the audio."), | ||
| 201 | .priv_size = sizeof(FlangerContext), | ||
| 202 | .priv_class = &flanger_class, | ||
| 203 | .init = init, | ||
| 204 | .uninit = uninit, | ||
| 205 | FILTER_INPUTS(flanger_inputs), | ||
| 206 | FILTER_OUTPUTS(ff_audio_default_filterpad), | ||
| 207 | FILTER_SINGLE_SAMPLEFMT(AV_SAMPLE_FMT_DBLP), | ||
| 208 | }; | ||
| 209 |