1 /*
2 * Copyright (c) 2020 Paul B Mahol
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 License
8 * 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
14 * GNU Lesser General Public License for more details.
15 *
16 * You should have received a copy of the GNU Lesser General Public License
17 * along with FFmpeg; if not, write to the Free Software Foundation, Inc.,
18 * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
19 */
20
21 #include "libavutil/eval.h"
22 #include "libavutil/opt.h"
23 #include "libavutil/tx.h"
24 #include "audio.h"
25 #include "avfilter.h"
26 #include "internal.h"
27 #include "window_func.h"
28
29 typedef struct AudioFIRSourceContext {
30 const AVClass *class;
31
32 char *freq_points_str;
33 char *magnitude_str;
34 char *phase_str;
35 int nb_taps;
36 int sample_rate;
37 int nb_samples;
38 int win_func;
39
40 AVComplexFloat *complexf;
41 float *freq;
42 float *magnitude;
43 float *phase;
44 int freq_size;
45 int magnitude_size;
46 int phase_size;
47 int nb_freq;
48 int nb_magnitude;
49 int nb_phase;
50
51 float *taps;
52 float *win;
53 int64_t pts;
54
55 AVTXContext *tx_ctx;
56 av_tx_fn tx_fn;
57 } AudioFIRSourceContext;
58
59 #define OFFSET(x) offsetof(AudioFIRSourceContext, x)
60 #define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
61
62 static const AVOption afirsrc_options[] = {
63 { "taps", "set number of taps", OFFSET(nb_taps), AV_OPT_TYPE_INT, {.i64=1025}, 9, UINT16_MAX, FLAGS },
64 { "t", "set number of taps", OFFSET(nb_taps), AV_OPT_TYPE_INT, {.i64=1025}, 9, UINT16_MAX, FLAGS },
65 { "frequency", "set frequency points", OFFSET(freq_points_str), AV_OPT_TYPE_STRING, {.str="0 1"}, 0, 0, FLAGS },
66 { "f", "set frequency points", OFFSET(freq_points_str), AV_OPT_TYPE_STRING, {.str="0 1"}, 0, 0, FLAGS },
67 { "magnitude", "set magnitude values", OFFSET(magnitude_str), AV_OPT_TYPE_STRING, {.str="1 1"}, 0, 0, FLAGS },
68 { "m", "set magnitude values", OFFSET(magnitude_str), AV_OPT_TYPE_STRING, {.str="1 1"}, 0, 0, FLAGS },
69 { "phase", "set phase values", OFFSET(phase_str), AV_OPT_TYPE_STRING, {.str="0 0"}, 0, 0, FLAGS },
70 { "p", "set phase values", OFFSET(phase_str), AV_OPT_TYPE_STRING, {.str="0 0"}, 0, 0, FLAGS },
71 { "sample_rate", "set sample rate", OFFSET(sample_rate), AV_OPT_TYPE_INT, {.i64=44100}, 1, INT_MAX, FLAGS },
72 { "r", "set sample rate", OFFSET(sample_rate), AV_OPT_TYPE_INT, {.i64=44100}, 1, INT_MAX, FLAGS },
73 { "nb_samples", "set the number of samples per requested frame", OFFSET(nb_samples), AV_OPT_TYPE_INT, {.i64 = 1024}, 1, INT_MAX, FLAGS },
74 { "n", "set the number of samples per requested frame", OFFSET(nb_samples), AV_OPT_TYPE_INT, {.i64 = 1024}, 1, INT_MAX, FLAGS },
75 { "win_func", "set window function", OFFSET(win_func), AV_OPT_TYPE_INT, {.i64=WFUNC_BLACKMAN}, 0, NB_WFUNC-1, FLAGS, "win_func" },
76 { "w", "set window function", OFFSET(win_func), AV_OPT_TYPE_INT, {.i64=WFUNC_BLACKMAN}, 0, NB_WFUNC-1, FLAGS, "win_func" },
77 { "rect", "Rectangular", 0, AV_OPT_TYPE_CONST, {.i64=WFUNC_RECT}, 0, 0, FLAGS, "win_func" },
78 { "bartlett", "Bartlett", 0, AV_OPT_TYPE_CONST, {.i64=WFUNC_BARTLETT}, 0, 0, FLAGS, "win_func" },
79 { "hanning", "Hanning", 0, AV_OPT_TYPE_CONST, {.i64=WFUNC_HANNING}, 0, 0, FLAGS, "win_func" },
80 { "hamming", "Hamming", 0, AV_OPT_TYPE_CONST, {.i64=WFUNC_HAMMING}, 0, 0, FLAGS, "win_func" },
81 { "blackman", "Blackman", 0, AV_OPT_TYPE_CONST, {.i64=WFUNC_BLACKMAN}, 0, 0, FLAGS, "win_func" },
82 { "welch", "Welch", 0, AV_OPT_TYPE_CONST, {.i64=WFUNC_WELCH}, 0, 0, FLAGS, "win_func" },
83 { "flattop", "Flat-top", 0, AV_OPT_TYPE_CONST, {.i64=WFUNC_FLATTOP}, 0, 0, FLAGS, "win_func" },
84 { "bharris", "Blackman-Harris", 0, AV_OPT_TYPE_CONST, {.i64=WFUNC_BHARRIS}, 0, 0, FLAGS, "win_func" },
85 { "bnuttall", "Blackman-Nuttall", 0, AV_OPT_TYPE_CONST, {.i64=WFUNC_BNUTTALL}, 0, 0, FLAGS, "win_func" },
86 { "bhann", "Bartlett-Hann", 0, AV_OPT_TYPE_CONST, {.i64=WFUNC_BHANN}, 0, 0, FLAGS, "win_func" },
87 { "sine", "Sine", 0, AV_OPT_TYPE_CONST, {.i64=WFUNC_SINE}, 0, 0, FLAGS, "win_func" },
88 { "nuttall", "Nuttall", 0, AV_OPT_TYPE_CONST, {.i64=WFUNC_NUTTALL}, 0, 0, FLAGS, "win_func" },
89 { "lanczos", "Lanczos", 0, AV_OPT_TYPE_CONST, {.i64=WFUNC_LANCZOS}, 0, 0, FLAGS, "win_func" },
90 { "gauss", "Gauss", 0, AV_OPT_TYPE_CONST, {.i64=WFUNC_GAUSS}, 0, 0, FLAGS, "win_func" },
91 { "tukey", "Tukey", 0, AV_OPT_TYPE_CONST, {.i64=WFUNC_TUKEY}, 0, 0, FLAGS, "win_func" },
92 { "dolph", "Dolph-Chebyshev", 0, AV_OPT_TYPE_CONST, {.i64=WFUNC_DOLPH}, 0, 0, FLAGS, "win_func" },
93 { "cauchy", "Cauchy", 0, AV_OPT_TYPE_CONST, {.i64=WFUNC_CAUCHY}, 0, 0, FLAGS, "win_func" },
94 { "parzen", "Parzen", 0, AV_OPT_TYPE_CONST, {.i64=WFUNC_PARZEN}, 0, 0, FLAGS, "win_func" },
95 { "poisson", "Poisson", 0, AV_OPT_TYPE_CONST, {.i64=WFUNC_POISSON}, 0, 0, FLAGS, "win_func" },
96 { "bohman" , "Bohman", 0, AV_OPT_TYPE_CONST, {.i64=WFUNC_BOHMAN}, 0, 0, FLAGS, "win_func" },
97 {NULL}
98 };
99
100 AVFILTER_DEFINE_CLASS(afirsrc);
101
init(AVFilterContext * ctx)102 static av_cold int init(AVFilterContext *ctx)
103 {
104 AudioFIRSourceContext *s = ctx->priv;
105
106 if (!(s->nb_taps & 1)) {
107 av_log(s, AV_LOG_WARNING, "Number of taps %d must be odd length.\n", s->nb_taps);
108 s->nb_taps |= 1;
109 }
110
111 return 0;
112 }
113
uninit(AVFilterContext * ctx)114 static av_cold void uninit(AVFilterContext *ctx)
115 {
116 AudioFIRSourceContext *s = ctx->priv;
117
118 av_freep(&s->win);
119 av_freep(&s->taps);
120 av_freep(&s->freq);
121 av_freep(&s->magnitude);
122 av_freep(&s->phase);
123 av_freep(&s->complexf);
124 av_tx_uninit(&s->tx_ctx);
125 }
126
query_formats(AVFilterContext * ctx)127 static av_cold int query_formats(AVFilterContext *ctx)
128 {
129 AudioFIRSourceContext *s = ctx->priv;
130 static const int64_t chlayouts[] = { AV_CH_LAYOUT_MONO, -1 };
131 int sample_rates[] = { s->sample_rate, -1 };
132 static const enum AVSampleFormat sample_fmts[] = {
133 AV_SAMPLE_FMT_FLT,
134 AV_SAMPLE_FMT_NONE
135 };
136
137 AVFilterFormats *formats;
138 AVFilterChannelLayouts *layouts;
139 int ret;
140
141 formats = ff_make_format_list(sample_fmts);
142 if (!formats)
143 return AVERROR(ENOMEM);
144 ret = ff_set_common_formats (ctx, formats);
145 if (ret < 0)
146 return ret;
147
148 layouts = ff_make_format64_list(chlayouts);
149 if (!layouts)
150 return AVERROR(ENOMEM);
151 ret = ff_set_common_channel_layouts(ctx, layouts);
152 if (ret < 0)
153 return ret;
154
155 formats = ff_make_format_list(sample_rates);
156 if (!formats)
157 return AVERROR(ENOMEM);
158 return ff_set_common_samplerates(ctx, formats);
159 }
160
parse_string(char * str,float ** items,int * nb_items,int * items_size)161 static int parse_string(char *str, float **items, int *nb_items, int *items_size)
162 {
163 float *new_items;
164 char *tail;
165
166 new_items = av_fast_realloc(NULL, items_size, 1 * sizeof(float));
167 if (!new_items)
168 return AVERROR(ENOMEM);
169 *items = new_items;
170
171 tail = str;
172 if (!tail)
173 return AVERROR(EINVAL);
174
175 do {
176 (*items)[(*nb_items)++] = av_strtod(tail, &tail);
177 new_items = av_fast_realloc(*items, items_size, (*nb_items + 1) * sizeof(float));
178 if (!new_items)
179 return AVERROR(ENOMEM);
180 *items = new_items;
181 if (tail && *tail)
182 tail++;
183 } while (tail && *tail);
184
185 return 0;
186 }
187
lininterp(AVComplexFloat * complexf,const float * freq,const float * magnitude,const float * phase,int m,int minterp)188 static void lininterp(AVComplexFloat *complexf,
189 const float *freq,
190 const float *magnitude,
191 const float *phase,
192 int m, int minterp)
193 {
194 for (int i = 0; i < minterp; i++) {
195 for (int j = 1; j < m; j++) {
196 const float x = i / (float)minterp;
197
198 if (x <= freq[j]) {
199 const float mg = (x - freq[j-1]) / (freq[j] - freq[j-1]) * (magnitude[j] - magnitude[j-1]) + magnitude[j-1];
200 const float ph = (x - freq[j-1]) / (freq[j] - freq[j-1]) * (phase[j] - phase[j-1]) + phase[j-1];
201
202 complexf[i].re = mg * cosf(ph);
203 complexf[i].im = mg * sinf(ph);
204 break;
205 }
206 }
207 }
208 }
209
config_output(AVFilterLink * outlink)210 static av_cold int config_output(AVFilterLink *outlink)
211 {
212 AVFilterContext *ctx = outlink->src;
213 AudioFIRSourceContext *s = ctx->priv;
214 float overlap, scale = 1.f, compensation;
215 int fft_size, middle, ret;
216
217 s->nb_freq = s->nb_magnitude = s->nb_phase = 0;
218
219 ret = parse_string(s->freq_points_str, &s->freq, &s->nb_freq, &s->freq_size);
220 if (ret < 0)
221 return ret;
222
223 ret = parse_string(s->magnitude_str, &s->magnitude, &s->nb_magnitude, &s->magnitude_size);
224 if (ret < 0)
225 return ret;
226
227 ret = parse_string(s->phase_str, &s->phase, &s->nb_phase, &s->phase_size);
228 if (ret < 0)
229 return ret;
230
231 if (s->nb_freq != s->nb_magnitude && s->nb_freq != s->nb_phase && s->nb_freq >= 2) {
232 av_log(ctx, AV_LOG_ERROR, "Number of frequencies, magnitudes and phases must be same and >= 2.\n");
233 return AVERROR(EINVAL);
234 }
235
236 for (int i = 0; i < s->nb_freq; i++) {
237 if (i == 0 && s->freq[i] != 0.f) {
238 av_log(ctx, AV_LOG_ERROR, "First frequency must be 0.\n");
239 return AVERROR(EINVAL);
240 }
241
242 if (i == s->nb_freq - 1 && s->freq[i] != 1.f) {
243 av_log(ctx, AV_LOG_ERROR, "Last frequency must be 1.\n");
244 return AVERROR(EINVAL);
245 }
246
247 if (i && s->freq[i] < s->freq[i-1]) {
248 av_log(ctx, AV_LOG_ERROR, "Frequencies must be in increasing order.\n");
249 return AVERROR(EINVAL);
250 }
251 }
252
253 fft_size = 1 << (av_log2(s->nb_taps) + 1);
254 s->complexf = av_calloc(fft_size * 2, sizeof(*s->complexf));
255 if (!s->complexf)
256 return AVERROR(ENOMEM);
257
258 ret = av_tx_init(&s->tx_ctx, &s->tx_fn, AV_TX_FLOAT_FFT, 1, fft_size, &scale, 0);
259 if (ret < 0)
260 return ret;
261
262 s->taps = av_calloc(s->nb_taps, sizeof(*s->taps));
263 if (!s->taps)
264 return AVERROR(ENOMEM);
265
266 s->win = av_calloc(s->nb_taps, sizeof(*s->win));
267 if (!s->win)
268 return AVERROR(ENOMEM);
269
270 generate_window_func(s->win, s->nb_taps, s->win_func, &overlap);
271
272 lininterp(s->complexf, s->freq, s->magnitude, s->phase, s->nb_freq, fft_size / 2);
273
274 s->tx_fn(s->tx_ctx, s->complexf + fft_size, s->complexf, sizeof(float));
275
276 compensation = 2.f / fft_size;
277 middle = s->nb_taps / 2;
278
279 for (int i = 0; i <= middle; i++) {
280 s->taps[ i] = s->complexf[fft_size + middle - i].re * compensation * s->win[i];
281 s->taps[middle + i] = s->complexf[fft_size + i].re * compensation * s->win[middle + i];
282 }
283
284 s->pts = 0;
285
286 return 0;
287 }
288
request_frame(AVFilterLink * outlink)289 static int request_frame(AVFilterLink *outlink)
290 {
291 AVFilterContext *ctx = outlink->src;
292 AudioFIRSourceContext *s = ctx->priv;
293 AVFrame *frame;
294 int nb_samples;
295
296 nb_samples = FFMIN(s->nb_samples, s->nb_taps - s->pts);
297 if (!nb_samples)
298 return AVERROR_EOF;
299
300 if (!(frame = ff_get_audio_buffer(outlink, nb_samples)))
301 return AVERROR(ENOMEM);
302
303 memcpy(frame->data[0], s->taps + s->pts, nb_samples * sizeof(float));
304
305 frame->pts = s->pts;
306 s->pts += nb_samples;
307 return ff_filter_frame(outlink, frame);
308 }
309
310 static const AVFilterPad afirsrc_outputs[] = {
311 {
312 .name = "default",
313 .type = AVMEDIA_TYPE_AUDIO,
314 .request_frame = request_frame,
315 .config_props = config_output,
316 },
317 { NULL }
318 };
319
320 AVFilter ff_asrc_afirsrc = {
321 .name = "afirsrc",
322 .description = NULL_IF_CONFIG_SMALL("Generate a FIR coefficients audio stream."),
323 .query_formats = query_formats,
324 .init = init,
325 .uninit = uninit,
326 .priv_size = sizeof(AudioFIRSourceContext),
327 .inputs = NULL,
328 .outputs = afirsrc_outputs,
329 .priv_class = &afirsrc_class,
330 };
331