| Line | Branch | Exec | Source |
|---|---|---|---|
| 1 | /* | ||
| 2 | * Copyright (c) 2011 Mina Nagy Zaki | ||
| 3 | * Copyright (c) 2000 Edward Beingessner And Sundry Contributors. | ||
| 4 | * This source code is freely redistributable and may be used for any purpose. | ||
| 5 | * This copyright notice must be maintained. Edward Beingessner And Sundry | ||
| 6 | * Contributors are not responsible for the consequences of using this | ||
| 7 | * software. | ||
| 8 | * | ||
| 9 | * This file is part of FFmpeg. | ||
| 10 | * | ||
| 11 | * FFmpeg is free software; you can redistribute it and/or | ||
| 12 | * modify it under the terms of the GNU Lesser General Public | ||
| 13 | * License as published by the Free Software Foundation; either | ||
| 14 | * version 2.1 of the License, or (at your option) any later version. | ||
| 15 | * | ||
| 16 | * FFmpeg is distributed in the hope that it will be useful, | ||
| 17 | * but WITHOUT ANY WARRANTY; without even the implied warranty of | ||
| 18 | * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU | ||
| 19 | * Lesser General Public License for more details. | ||
| 20 | * | ||
| 21 | * You should have received a copy of the GNU Lesser General Public | ||
| 22 | * License along with FFmpeg; if not, write to the Free Software | ||
| 23 | * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA | ||
| 24 | */ | ||
| 25 | |||
| 26 | /** | ||
| 27 | * @file | ||
| 28 | * Stereo Widening Effect. Adds audio cues to move stereo image in | ||
| 29 | * front of the listener. Adapted from the libsox earwax effect. | ||
| 30 | */ | ||
| 31 | |||
| 32 | #include "libavutil/channel_layout.h" | ||
| 33 | #include "avfilter.h" | ||
| 34 | #include "audio.h" | ||
| 35 | #include "filters.h" | ||
| 36 | #include "formats.h" | ||
| 37 | |||
| 38 | #define NUMTAPS 32 | ||
| 39 | |||
| 40 | static const int8_t filt[NUMTAPS * 2] = { | ||
| 41 | /* 30° 330° */ | ||
| 42 | 4, -6, /* 32 tap stereo FIR filter. */ | ||
| 43 | 4, -11, /* One side filters as if the */ | ||
| 44 | -1, -5, /* signal was from 30 degrees */ | ||
| 45 | 3, 3, /* from the ear, the other as */ | ||
| 46 | -2, 5, /* if 330 degrees. */ | ||
| 47 | -5, 0, | ||
| 48 | 9, 1, | ||
| 49 | 6, 3, /* Input */ | ||
| 50 | -4, -1, /* Left Right */ | ||
| 51 | -5, -3, /* __________ __________ */ | ||
| 52 | -2, -5, /* | | | | */ | ||
| 53 | -7, 1, /* .---| Hh,0(f) | | Hh,0(f) |---. */ | ||
| 54 | 6, -7, /* / |__________| |__________| \ */ | ||
| 55 | 30, -29, /* / \ / \ */ | ||
| 56 | 12, -3, /* / X \ */ | ||
| 57 | -11, 4, /* / / \ \ */ | ||
| 58 | -3, 7, /* ____V_____ __________V V__________ _____V____ */ | ||
| 59 | -20, 23, /* | | | | | | | | */ | ||
| 60 | 2, 0, /* | Hh,30(f) | | Hh,330(f)| | Hh,330(f)| | Hh,30(f) | */ | ||
| 61 | 1, -6, /* |__________| |__________| |__________| |__________| */ | ||
| 62 | -14, -5, /* \ ___ / \ ___ / */ | ||
| 63 | 15, -18, /* \ / \ / _____ \ / \ / */ | ||
| 64 | 6, 7, /* `->| + |<--' / \ `-->| + |<-' */ | ||
| 65 | 15, -10, /* \___/ _/ \_ \___/ */ | ||
| 66 | -14, 22, /* \ / \ / \ / */ | ||
| 67 | -7, -2, /* `--->| | | |<---' */ | ||
| 68 | -4, 9, /* \_/ \_/ */ | ||
| 69 | 6, -12, /* */ | ||
| 70 | 6, -6, /* Headphones */ | ||
| 71 | 0, -11, | ||
| 72 | 0, -5, | ||
| 73 | 4, 0}; | ||
| 74 | |||
| 75 | typedef struct EarwaxContext { | ||
| 76 | int16_t filter[2][NUMTAPS]; | ||
| 77 | int16_t taps[4][NUMTAPS * 2]; | ||
| 78 | |||
| 79 | AVFrame *frame[2]; | ||
| 80 | } EarwaxContext; | ||
| 81 | |||
| 82 | 1 | static int query_formats(const AVFilterContext *ctx, | |
| 83 | AVFilterFormatsConfig **cfg_in, | ||
| 84 | AVFilterFormatsConfig **cfg_out) | ||
| 85 | { | ||
| 86 | static const enum AVSampleFormat formats[] = { | ||
| 87 | AV_SAMPLE_FMT_S16P, | ||
| 88 | AV_SAMPLE_FMT_NONE, | ||
| 89 | }; | ||
| 90 | static const AVChannelLayout layouts[] = { | ||
| 91 | AV_CHANNEL_LAYOUT_STEREO, | ||
| 92 | { .nb_channels = 0 }, | ||
| 93 | }; | ||
| 94 | static const int sample_rates[] = { 44100, -1 }; | ||
| 95 | |||
| 96 | int ret; | ||
| 97 | |||
| 98 | 1 | ret = ff_set_common_formats_from_list2(ctx, cfg_in, cfg_out, formats); | |
| 99 |
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1 | if (ret < 0) |
| 100 | ✗ | return ret; | |
| 101 | |||
| 102 | 1 | ret = ff_set_common_samplerates_from_list2(ctx, cfg_in, cfg_out, sample_rates); | |
| 103 |
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1 | if (ret < 0) |
| 104 | ✗ | return ret; | |
| 105 | |||
| 106 | 1 | ret = ff_set_common_channel_layouts_from_list2(ctx, cfg_in, cfg_out, layouts); | |
| 107 |
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1 | if (ret < 0) |
| 108 | ✗ | return ret; | |
| 109 | |||
| 110 | 1 | return 0; | |
| 111 | } | ||
| 112 | |||
| 113 | //FIXME: replace with DSPContext.scalarproduct_int16 | ||
| 114 | 160 | static inline int16_t *scalarproduct(const int16_t *in, const int16_t *endin, | |
| 115 | const int16_t *filt, int16_t *out) | ||
| 116 | { | ||
| 117 | int32_t sample; | ||
| 118 | int16_t j; | ||
| 119 | |||
| 120 |
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327840 | while (in < endin) { |
| 121 | 327680 | sample = 0; | |
| 122 |
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10813440 | for (j = 0; j < NUMTAPS; j++) |
| 123 | 10485760 | sample += in[j] * filt[j]; | |
| 124 | 327680 | *out = av_clip_int16(sample >> 7); | |
| 125 | 327680 | out++; | |
| 126 | 327680 | in++; | |
| 127 | } | ||
| 128 | |||
| 129 | 160 | return out; | |
| 130 | } | ||
| 131 | |||
| 132 | 1 | static int config_input(AVFilterLink *inlink) | |
| 133 | { | ||
| 134 | 1 | EarwaxContext *s = inlink->dst->priv; | |
| 135 | |||
| 136 |
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33 | for (int i = 0; i < NUMTAPS; i++) { |
| 137 | 32 | s->filter[0][i] = filt[i * 2]; | |
| 138 | 32 | s->filter[1][i] = filt[i * 2 + 1]; | |
| 139 | } | ||
| 140 | |||
| 141 | 1 | return 0; | |
| 142 | } | ||
| 143 | |||
| 144 | 80 | static void convolve(AVFilterContext *ctx, AVFrame *in, | |
| 145 | int input_ch, int output_ch, | ||
| 146 | int filter_ch, int tap_ch) | ||
| 147 | { | ||
| 148 | 80 | EarwaxContext *s = ctx->priv; | |
| 149 | int16_t *taps, *endin, *dst, *src; | ||
| 150 | int len; | ||
| 151 | |||
| 152 | 80 | taps = s->taps[tap_ch]; | |
| 153 | 80 | dst = (int16_t *)s->frame[input_ch]->data[output_ch]; | |
| 154 | 80 | src = (int16_t *)in->data[input_ch]; | |
| 155 | |||
| 156 | 80 | len = FFMIN(NUMTAPS, in->nb_samples); | |
| 157 | // copy part of new input and process with saved input | ||
| 158 | 80 | memcpy(taps+NUMTAPS, src, len * sizeof(*taps)); | |
| 159 | 80 | dst = scalarproduct(taps, taps + len, s->filter[filter_ch], dst); | |
| 160 | |||
| 161 | // process current input | ||
| 162 |
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80 | if (in->nb_samples >= NUMTAPS) { |
| 163 | 80 | endin = src + in->nb_samples - NUMTAPS; | |
| 164 | 80 | scalarproduct(src, endin, s->filter[filter_ch], dst); | |
| 165 | |||
| 166 | // save part of input for next round | ||
| 167 | 80 | memcpy(taps, endin, NUMTAPS * sizeof(*taps)); | |
| 168 | } else { | ||
| 169 | ✗ | memmove(taps, taps + in->nb_samples, NUMTAPS * sizeof(*taps)); | |
| 170 | } | ||
| 171 | 80 | } | |
| 172 | |||
| 173 | 40 | static void mix(AVFilterContext *ctx, AVFrame *out, | |
| 174 | int output_ch, int f0, int f1, int i0, int i1) | ||
| 175 | { | ||
| 176 | 40 | EarwaxContext *s = ctx->priv; | |
| 177 | 40 | const int16_t *srcl = (const int16_t *)s->frame[f0]->data[i0]; | |
| 178 | 40 | const int16_t *srcr = (const int16_t *)s->frame[f1]->data[i1]; | |
| 179 | 40 | int16_t *dst = (int16_t *)out->data[output_ch]; | |
| 180 | |||
| 181 |
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163880 | for (int n = 0; n < out->nb_samples; n++) |
| 182 | 163840 | dst[n] = av_clip_int16(srcl[n] + srcr[n]); | |
| 183 | 40 | } | |
| 184 | |||
| 185 | 20 | static int filter_frame(AVFilterLink *inlink, AVFrame *in) | |
| 186 | { | ||
| 187 | 20 | AVFilterContext *ctx = inlink->dst; | |
| 188 | 20 | EarwaxContext *s = ctx->priv; | |
| 189 | 20 | AVFilterLink *outlink = ctx->outputs[0]; | |
| 190 | 20 | AVFrame *out = ff_get_audio_buffer(outlink, in->nb_samples); | |
| 191 | |||
| 192 |
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60 | for (int ch = 0; ch < 2; ch++) { |
| 193 |
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40 | if (!s->frame[ch] || s->frame[ch]->nb_samples < in->nb_samples) { |
| 194 | 2 | av_frame_free(&s->frame[ch]); | |
| 195 | 2 | s->frame[ch] = ff_get_audio_buffer(outlink, in->nb_samples); | |
| 196 |
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2 | if (!s->frame[ch]) { |
| 197 | ✗ | av_frame_free(&in); | |
| 198 | ✗ | av_frame_free(&out); | |
| 199 | ✗ | return AVERROR(ENOMEM); | |
| 200 | } | ||
| 201 | } | ||
| 202 | } | ||
| 203 | |||
| 204 |
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20 | if (!out) { |
| 205 | ✗ | av_frame_free(&in); | |
| 206 | ✗ | return AVERROR(ENOMEM); | |
| 207 | } | ||
| 208 | 20 | av_frame_copy_props(out, in); | |
| 209 | |||
| 210 | 20 | convolve(ctx, in, 0, 0, 0, 0); | |
| 211 | 20 | convolve(ctx, in, 0, 1, 1, 1); | |
| 212 | 20 | convolve(ctx, in, 1, 0, 0, 2); | |
| 213 | 20 | convolve(ctx, in, 1, 1, 1, 3); | |
| 214 | |||
| 215 | 20 | mix(ctx, out, 0, 0, 1, 1, 0); | |
| 216 | 20 | mix(ctx, out, 1, 0, 1, 0, 1); | |
| 217 | |||
| 218 | 20 | av_frame_free(&in); | |
| 219 | 20 | return ff_filter_frame(outlink, out); | |
| 220 | } | ||
| 221 | |||
| 222 | 2 | static av_cold void uninit(AVFilterContext *ctx) | |
| 223 | { | ||
| 224 | 2 | EarwaxContext *s = ctx->priv; | |
| 225 | |||
| 226 | 2 | av_frame_free(&s->frame[0]); | |
| 227 | 2 | av_frame_free(&s->frame[1]); | |
| 228 | 2 | } | |
| 229 | |||
| 230 | static const AVFilterPad earwax_inputs[] = { | ||
| 231 | { | ||
| 232 | .name = "default", | ||
| 233 | .type = AVMEDIA_TYPE_AUDIO, | ||
| 234 | .filter_frame = filter_frame, | ||
| 235 | .config_props = config_input, | ||
| 236 | }, | ||
| 237 | }; | ||
| 238 | |||
| 239 | const FFFilter ff_af_earwax = { | ||
| 240 | .p.name = "earwax", | ||
| 241 | .p.description = NULL_IF_CONFIG_SMALL("Widen the stereo image."), | ||
| 242 | .priv_size = sizeof(EarwaxContext), | ||
| 243 | .uninit = uninit, | ||
| 244 | FILTER_INPUTS(earwax_inputs), | ||
| 245 | FILTER_OUTPUTS(ff_audio_default_filterpad), | ||
| 246 | FILTER_QUERY_FUNC2(query_formats), | ||
| 247 | }; | ||
| 248 |