GCC Code Coverage Report
Directory: ../../../ffmpeg/ Exec Total Coverage
File: src/libavfilter/af_acrossover.c Lines: 0 217 0.0 %
Date: 2021-01-20 23:14:43 Branches: 0 137 0.0 %

Line Branch Exec Source
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/*
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 * This file is part of FFmpeg.
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 *
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 * FFmpeg is free software; you can redistribute it and/or
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 * modify it under the terms of the GNU Lesser General Public
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 * License as published by the Free Software Foundation; either
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 * version 2.1 of the License, or (at your option) any later version.
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 *
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 * FFmpeg is distributed in the hope that it will be useful,
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 * but WITHOUT ANY WARRANTY; without even the implied warranty of
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 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
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 * Lesser General Public License for more details.
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 *
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 * You should have received a copy of the GNU Lesser General Public
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 * License along with FFmpeg; if not, write to the Free Software
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 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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 */
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/**
20
 * @file
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 * Crossover filter
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 *
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 * Split an audio stream into several bands.
24
 */
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#include "libavutil/attributes.h"
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#include "libavutil/avstring.h"
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#include "libavutil/channel_layout.h"
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#include "libavutil/eval.h"
30
#include "libavutil/float_dsp.h"
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#include "libavutil/internal.h"
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#include "libavutil/opt.h"
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34
#include "audio.h"
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#include "avfilter.h"
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#include "formats.h"
37
#include "internal.h"
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#define MAX_SPLITS 16
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#define MAX_BANDS MAX_SPLITS + 1
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#define B0 0
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#define B1 1
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#define B2 2
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#define A1 3
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#define A2 4
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typedef struct BiquadCoeffs {
49
    double cd[5];
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    float cf[5];
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} BiquadCoeffs;
52
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typedef struct AudioCrossoverContext {
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    const AVClass *class;
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    char *splits_str;
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    char *gains_str;
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    int order_opt;
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    float level_in;
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    int order;
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    int filter_count;
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    int first_order;
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    int ap_filter_count;
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    int nb_splits;
66
    float splits[MAX_SPLITS];
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    float gains[MAX_BANDS];
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    BiquadCoeffs lp[MAX_BANDS][20];
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    BiquadCoeffs hp[MAX_BANDS][20];
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    BiquadCoeffs ap[MAX_BANDS][20];
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    AVFrame *xover;
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    AVFrame *input_frame;
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    AVFrame *frames[MAX_BANDS];
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    int (*filter_channels)(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs);
80
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    AVFloatDSPContext *fdsp;
82
} AudioCrossoverContext;
83
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#define OFFSET(x) offsetof(AudioCrossoverContext, x)
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#define AF AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_FILTERING_PARAM
86
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static const AVOption acrossover_options[] = {
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    { "split", "set split frequencies", OFFSET(splits_str), AV_OPT_TYPE_STRING, {.str="500"}, 0, 0, AF },
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    { "order", "set filter order",      OFFSET(order_opt),  AV_OPT_TYPE_INT,    {.i64=1},     0, 9, AF, "m" },
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    { "2nd",   "2nd order (12 dB/8ve)", 0,                  AV_OPT_TYPE_CONST,  {.i64=0},     0, 0, AF, "m" },
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    { "4th",   "4th order (24 dB/8ve)", 0,                  AV_OPT_TYPE_CONST,  {.i64=1},     0, 0, AF, "m" },
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    { "6th",   "6th order (36 dB/8ve)", 0,                  AV_OPT_TYPE_CONST,  {.i64=2},     0, 0, AF, "m" },
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    { "8th",   "8th order (48 dB/8ve)", 0,                  AV_OPT_TYPE_CONST,  {.i64=3},     0, 0, AF, "m" },
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    { "10th",  "10th order (60 dB/8ve)",0,                  AV_OPT_TYPE_CONST,  {.i64=4},     0, 0, AF, "m" },
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    { "12th",  "12th order (72 dB/8ve)",0,                  AV_OPT_TYPE_CONST,  {.i64=5},     0, 0, AF, "m" },
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    { "14th",  "14th order (84 dB/8ve)",0,                  AV_OPT_TYPE_CONST,  {.i64=6},     0, 0, AF, "m" },
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    { "16th",  "16th order (96 dB/8ve)",0,                  AV_OPT_TYPE_CONST,  {.i64=7},     0, 0, AF, "m" },
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    { "18th",  "18th order (108 dB/8ve)",0,                 AV_OPT_TYPE_CONST,  {.i64=8},     0, 0, AF, "m" },
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    { "20th",  "20th order (120 dB/8ve)",0,                 AV_OPT_TYPE_CONST,  {.i64=9},     0, 0, AF, "m" },
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    { "level", "set input gain",        OFFSET(level_in),   AV_OPT_TYPE_FLOAT,  {.dbl=1},     0, 1, AF },
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    { "gain",  "set output bands gain", OFFSET(gains_str),  AV_OPT_TYPE_STRING, {.str="1.f"}, 0, 0, AF },
102
    { NULL }
103
};
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AVFILTER_DEFINE_CLASS(acrossover);
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static int parse_gains(AVFilterContext *ctx)
108
{
109
    AudioCrossoverContext *s = ctx->priv;
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    char *p, *arg, *saveptr = NULL;
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    int i, ret = 0;
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    saveptr = NULL;
114
    p = s->gains_str;
115
    for (i = 0; i < MAX_BANDS; i++) {
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        float gain;
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        char c[3] = { 0 };
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        if (!(arg = av_strtok(p, " |", &saveptr)))
120
            break;
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        p = NULL;
123
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        if (av_sscanf(arg, "%f%2s", &gain, c) < 1) {
125
            av_log(ctx, AV_LOG_ERROR, "Invalid syntax for gain[%d].\n", i);
126
            ret = AVERROR(EINVAL);
127
            break;
128
        }
129
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        if (c[0] == 'd' && c[1] == 'B')
131
            s->gains[i] = expf(gain * M_LN10 / 20.f);
132
        else
133
            s->gains[i] = gain;
134
    }
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    for (; i < MAX_BANDS; i++)
137
        s->gains[i] = 1.f;
138
139
    return ret;
140
}
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static av_cold int init(AVFilterContext *ctx)
143
{
144
    AudioCrossoverContext *s = ctx->priv;
145
    char *p, *arg, *saveptr = NULL;
146
    int i, ret = 0;
147
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    s->fdsp = avpriv_float_dsp_alloc(0);
149
    if (!s->fdsp)
150
        return AVERROR(ENOMEM);
151
152
    p = s->splits_str;
153
    for (i = 0; i < MAX_SPLITS; i++) {
154
        float freq;
155
156
        if (!(arg = av_strtok(p, " |", &saveptr)))
157
            break;
158
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        p = NULL;
160
161
        if (av_sscanf(arg, "%f", &freq) != 1) {
162
            av_log(ctx, AV_LOG_ERROR, "Invalid syntax for frequency[%d].\n", i);
163
            return AVERROR(EINVAL);
164
        }
165
        if (freq <= 0) {
166
            av_log(ctx, AV_LOG_ERROR, "Frequency %f must be positive number.\n", freq);
167
            return AVERROR(EINVAL);
168
        }
169
170
        if (i > 0 && freq <= s->splits[i-1]) {
171
            av_log(ctx, AV_LOG_ERROR, "Frequency %f must be in increasing order.\n", freq);
172
            return AVERROR(EINVAL);
173
        }
174
175
        s->splits[i] = freq;
176
    }
177
178
    s->nb_splits = i;
179
180
    ret = parse_gains(ctx);
181
    if (ret < 0)
182
        return ret;
183
184
    for (i = 0; i <= s->nb_splits; i++) {
185
        AVFilterPad pad  = { 0 };
186
        char *name;
187
188
        pad.type = AVMEDIA_TYPE_AUDIO;
189
        name = av_asprintf("out%d", ctx->nb_outputs);
190
        if (!name)
191
            return AVERROR(ENOMEM);
192
        pad.name = name;
193
194
        if ((ret = ff_insert_outpad(ctx, i, &pad)) < 0) {
195
            av_freep(&pad.name);
196
            return ret;
197
        }
198
    }
199
200
    return ret;
201
}
202
203
static void set_lp(BiquadCoeffs *b, double fc, double q, double sr)
204
{
205
    double omega = 2. * M_PI * fc / sr;
206
    double cosine = cos(omega);
207
    double alpha = sin(omega) / (2. * q);
208
209
    double b0 = (1. - cosine) / 2.;
210
    double b1 = 1. - cosine;
211
    double b2 = (1. - cosine) / 2.;
212
    double a0 = 1. + alpha;
213
    double a1 = -2. * cosine;
214
    double a2 = 1. - alpha;
215
216
    b->cd[B0] =  b0 / a0;
217
    b->cd[B1] =  b1 / a0;
218
    b->cd[B2] =  b2 / a0;
219
    b->cd[A1] = -a1 / a0;
220
    b->cd[A2] = -a2 / a0;
221
222
    b->cf[B0] = b->cd[B0];
223
    b->cf[B1] = b->cd[B1];
224
    b->cf[B2] = b->cd[B2];
225
    b->cf[A1] = b->cd[A1];
226
    b->cf[A2] = b->cd[A2];
227
}
228
229
static void set_hp(BiquadCoeffs *b, double fc, double q, double sr)
230
{
231
    double omega = 2. * M_PI * fc / sr;
232
    double cosine = cos(omega);
233
    double alpha = sin(omega) / (2. * q);
234
235
    double b0 = (1. + cosine) / 2.;
236
    double b1 = -1. - cosine;
237
    double b2 = (1. + cosine) / 2.;
238
    double a0 = 1. + alpha;
239
    double a1 = -2. * cosine;
240
    double a2 = 1. - alpha;
241
242
    b->cd[B0] =  b0 / a0;
243
    b->cd[B1] =  b1 / a0;
244
    b->cd[B2] =  b2 / a0;
245
    b->cd[A1] = -a1 / a0;
246
    b->cd[A2] = -a2 / a0;
247
248
    b->cf[B0] = b->cd[B0];
249
    b->cf[B1] = b->cd[B1];
250
    b->cf[B2] = b->cd[B2];
251
    b->cf[A1] = b->cd[A1];
252
    b->cf[A2] = b->cd[A2];
253
}
254
255
static void set_ap(BiquadCoeffs *b, double fc, double q, double sr)
256
{
257
    double omega = 2. * M_PI * fc / sr;
258
    double cosine = cos(omega);
259
    double alpha = sin(omega) / (2. * q);
260
261
    double a0 = 1. + alpha;
262
    double a1 = -2. * cosine;
263
    double a2 = 1. - alpha;
264
    double b0 = a2;
265
    double b1 = a1;
266
    double b2 = a0;
267
268
    b->cd[B0] =  b0 / a0;
269
    b->cd[B1] =  b1 / a0;
270
    b->cd[B2] =  b2 / a0;
271
    b->cd[A1] = -a1 / a0;
272
    b->cd[A2] = -a2 / a0;
273
274
    b->cf[B0] = b->cd[B0];
275
    b->cf[B1] = b->cd[B1];
276
    b->cf[B2] = b->cd[B2];
277
    b->cf[A1] = b->cd[A1];
278
    b->cf[A2] = b->cd[A2];
279
}
280
281
static void set_ap1(BiquadCoeffs *b, double fc, double sr)
282
{
283
    double omega = 2. * M_PI * fc / sr;
284
285
    b->cd[A1] = exp(-omega);
286
    b->cd[A2] = 0.;
287
    b->cd[B0] = -b->cd[A1];
288
    b->cd[B1] = 1.;
289
    b->cd[B2] = 0.;
290
291
    b->cf[B0] = b->cd[B0];
292
    b->cf[B1] = b->cd[B1];
293
    b->cf[B2] = b->cd[B2];
294
    b->cf[A1] = b->cd[A1];
295
    b->cf[A2] = b->cd[A2];
296
}
297
298
static void calc_q_factors(int order, double *q)
299
{
300
    double n = order / 2.;
301
302
    for (int i = 0; i < n / 2; i++)
303
        q[i] = 1. / (-2. * cos(M_PI * (2. * (i + 1) + n - 1.) / (2. * n)));
304
}
305
306
static int query_formats(AVFilterContext *ctx)
307
{
308
    AVFilterFormats *formats;
309
    AVFilterChannelLayouts *layouts;
310
    static const enum AVSampleFormat sample_fmts[] = {
311
        AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_DBLP,
312
        AV_SAMPLE_FMT_NONE
313
    };
314
    int ret;
315
316
    layouts = ff_all_channel_counts();
317
    if (!layouts)
318
        return AVERROR(ENOMEM);
319
    ret = ff_set_common_channel_layouts(ctx, layouts);
320
    if (ret < 0)
321
        return ret;
322
323
    formats = ff_make_format_list(sample_fmts);
324
    if (!formats)
325
        return AVERROR(ENOMEM);
326
    ret = ff_set_common_formats(ctx, formats);
327
    if (ret < 0)
328
        return ret;
329
330
    formats = ff_all_samplerates();
331
    if (!formats)
332
        return AVERROR(ENOMEM);
333
    return ff_set_common_samplerates(ctx, formats);
334
}
335
336
#define BIQUAD_PROCESS(name, type)                             \
337
static void biquad_process_## name(const type *const c,        \
338
                                   type *b,                    \
339
                                   type *dst, const type *src, \
340
                                   int nb_samples)             \
341
{                                                              \
342
    const type b0 = c[B0];                                     \
343
    const type b1 = c[B1];                                     \
344
    const type b2 = c[B2];                                     \
345
    const type a1 = c[A1];                                     \
346
    const type a2 = c[A2];                                     \
347
    type z1 = b[0];                                            \
348
    type z2 = b[1];                                            \
349
                                                               \
350
    for (int n = 0; n + 1 < nb_samples; n++) {                 \
351
        type in = src[n];                                      \
352
        type out;                                              \
353
                                                               \
354
        out = in * b0 + z1;                                    \
355
        z1 = b1 * in + z2 + a1 * out;                          \
356
        z2 = b2 * in + a2 * out;                               \
357
        dst[n] = out;                                          \
358
                                                               \
359
        n++;                                                   \
360
        in = src[n];                                           \
361
        out = in * b0 + z1;                                    \
362
        z1 = b1 * in + z2 + a1 * out;                          \
363
        z2 = b2 * in + a2 * out;                               \
364
        dst[n] = out;                                          \
365
    }                                                          \
366
                                                               \
367
    if (nb_samples & 1) {                                      \
368
        const int n = nb_samples - 1;                          \
369
        const type in = src[n];                                \
370
        type out;                                              \
371
                                                               \
372
        out = in * b0 + z1;                                    \
373
        z1 = b1 * in + z2 + a1 * out;                          \
374
        z2 = b2 * in + a2 * out;                               \
375
        dst[n] = out;                                          \
376
    }                                                          \
377
                                                               \
378
    b[0] = z1;                                                 \
379
    b[1] = z2;                                                 \
380
}
381
382
BIQUAD_PROCESS(fltp, float)
383
BIQUAD_PROCESS(dblp, double)
384
385
#define XOVER_PROCESS(name, type, one, ff)                                                  \
386
static int filter_channels_## name(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs) \
387
{                                                                                           \
388
    AudioCrossoverContext *s = ctx->priv;                                                   \
389
    AVFrame *in = s->input_frame;                                                           \
390
    AVFrame **frames = s->frames;                                                           \
391
    const int start = (in->channels * jobnr) / nb_jobs;                                     \
392
    const int end = (in->channels * (jobnr+1)) / nb_jobs;                                   \
393
    const int nb_samples = in->nb_samples;                                                  \
394
    const int nb_outs = ctx->nb_outputs;                                                    \
395
    const int first_order = s->first_order;                                                 \
396
                                                                                            \
397
    for (int ch = start; ch < end; ch++) {                                                  \
398
        const type *src = (const type *)in->extended_data[ch];                              \
399
        type *xover = (type *)s->xover->extended_data[ch];                                  \
400
                                                                                            \
401
        s->fdsp->vector_## ff ##mul_scalar((type *)frames[0]->extended_data[ch], src,       \
402
                                    s->level_in, FFALIGN(nb_samples, sizeof(type)));        \
403
                                                                                            \
404
        for (int band = 0; band < nb_outs; band++) {                                        \
405
            for (int f = 0; band + 1 < nb_outs && f < s->filter_count; f++) {               \
406
                const type *prv = (const type *)frames[band]->extended_data[ch];            \
407
                type *dst = (type *)frames[band + 1]->extended_data[ch];                    \
408
                const type *hsrc = f == 0 ? prv : dst;                                      \
409
                type *hp = xover + nb_outs * 20 + band * 20 + f * 2;                        \
410
                const type *const hpc = (type *)&s->hp[band][f].c ## ff;                    \
411
                                                                                            \
412
                biquad_process_## name(hpc, hp, dst, hsrc, nb_samples);                     \
413
            }                                                                               \
414
                                                                                            \
415
            for (int f = 0; band + 1 < nb_outs && f < s->filter_count; f++) {               \
416
                type *dst = (type *)frames[band]->extended_data[ch];                        \
417
                const type *lsrc = dst;                                                     \
418
                type *lp = xover + band * 20 + f * 2;                                       \
419
                const type *const lpc = (type *)&s->lp[band][f].c ## ff;                    \
420
                                                                                            \
421
                biquad_process_## name(lpc, lp, dst, lsrc, nb_samples);                     \
422
            }                                                                               \
423
                                                                                            \
424
            for (int aband = band + 1; aband + 1 < nb_outs; aband++) {                      \
425
                if (first_order) {                                                          \
426
                    const type *asrc = (const type *)frames[band]->extended_data[ch];       \
427
                    type *dst = (type *)frames[band]->extended_data[ch];                    \
428
                    type *ap = xover + nb_outs * 40 + (aband * nb_outs + band) * 20;        \
429
                    const type *const apc = (type *)&s->ap[aband][0].c ## ff;               \
430
                                                                                            \
431
                    biquad_process_## name(apc, ap, dst, asrc, nb_samples);                 \
432
                }                                                                           \
433
                                                                                            \
434
                for (int f = first_order; f < s->ap_filter_count; f++) {                    \
435
                    const type *asrc = (const type *)frames[band]->extended_data[ch];       \
436
                    type *dst = (type *)frames[band]->extended_data[ch];                    \
437
                    type *ap = xover + nb_outs * 40 + (aband * nb_outs + band) * 20 + f * 2;\
438
                    const type *const apc = (type *)&s->ap[aband][f].c ## ff;               \
439
                                                                                            \
440
                    biquad_process_## name(apc, ap, dst, asrc, nb_samples);                 \
441
                }                                                                           \
442
            }                                                                               \
443
        }                                                                                   \
444
                                                                                            \
445
        for (int band = 0; band < nb_outs; band++) {                                        \
446
            const type gain = s->gains[band] * ((band & 1 && first_order) ? -one : one);    \
447
            type *dst = (type *)frames[band]->extended_data[ch];                            \
448
                                                                                            \
449
            s->fdsp->vector_## ff ##mul_scalar(dst, dst, gain,                              \
450
                                               FFALIGN(nb_samples, sizeof(type)));          \
451
        }                                                                                   \
452
    }                                                                                       \
453
                                                                                            \
454
    return 0;                                                                               \
455
}
456
457
XOVER_PROCESS(fltp, float, 1.f, f)
458
XOVER_PROCESS(dblp, double, 1.0, d)
459
460
static int config_input(AVFilterLink *inlink)
461
{
462
    AVFilterContext *ctx = inlink->dst;
463
    AudioCrossoverContext *s = ctx->priv;
464
    int sample_rate = inlink->sample_rate;
465
    double q[16];
466
467
    s->order = (s->order_opt + 1) * 2;
468
    s->filter_count = s->order / 2;
469
    s->first_order = s->filter_count & 1;
470
    s->ap_filter_count = s->filter_count / 2 + s->first_order;
471
    calc_q_factors(s->order, q);
472
473
    for (int band = 0; band <= s->nb_splits; band++) {
474
        if (s->first_order) {
475
            set_lp(&s->lp[band][0], s->splits[band], 0.5, sample_rate);
476
            set_hp(&s->hp[band][0], s->splits[band], 0.5, sample_rate);
477
        }
478
479
        for (int n = s->first_order; n < s->filter_count; n++) {
480
            const int idx = s->filter_count / 2 - ((n + s->first_order) / 2 - s->first_order) - 1;
481
482
            set_lp(&s->lp[band][n], s->splits[band], q[idx], sample_rate);
483
            set_hp(&s->hp[band][n], s->splits[band], q[idx], sample_rate);
484
        }
485
486
        if (s->first_order)
487
            set_ap1(&s->ap[band][0], s->splits[band], sample_rate);
488
489
        for (int n = s->first_order; n < s->ap_filter_count; n++) {
490
            const int idx = (s->filter_count / 2 - ((n * 2 + s->first_order) / 2 - s->first_order) - 1);
491
492
            set_ap(&s->ap[band][n], s->splits[band], q[idx], sample_rate);
493
        }
494
    }
495
496
    switch (inlink->format) {
497
    case AV_SAMPLE_FMT_FLTP: s->filter_channels = filter_channels_fltp; break;
498
    case AV_SAMPLE_FMT_DBLP: s->filter_channels = filter_channels_dblp; break;
499
    }
500
501
    s->xover = ff_get_audio_buffer(inlink, 2 * (ctx->nb_outputs * 10 + ctx->nb_outputs * 10 +
502
                                                ctx->nb_outputs * ctx->nb_outputs * 10));
503
    if (!s->xover)
504
        return AVERROR(ENOMEM);
505
506
    return 0;
507
}
508
509
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
510
{
511
    AVFilterContext *ctx = inlink->dst;
512
    AudioCrossoverContext *s = ctx->priv;
513
    AVFrame **frames = s->frames;
514
    int i, ret = 0;
515
516
    for (i = 0; i < ctx->nb_outputs; i++) {
517
        frames[i] = ff_get_audio_buffer(ctx->outputs[i], in->nb_samples);
518
519
        if (!frames[i]) {
520
            ret = AVERROR(ENOMEM);
521
            break;
522
        }
523
524
        frames[i]->pts = in->pts;
525
    }
526
527
    if (ret < 0)
528
        goto fail;
529
530
    s->input_frame = in;
531
    ctx->internal->execute(ctx, s->filter_channels, NULL, NULL, FFMIN(inlink->channels,
532
                                                                      ff_filter_get_nb_threads(ctx)));
533
534
    for (i = 0; i < ctx->nb_outputs; i++) {
535
        ret = ff_filter_frame(ctx->outputs[i], frames[i]);
536
        frames[i] = NULL;
537
        if (ret < 0)
538
            break;
539
    }
540
541
fail:
542
    for (i = 0; i < ctx->nb_outputs; i++)
543
        av_frame_free(&frames[i]);
544
    av_frame_free(&in);
545
    s->input_frame = NULL;
546
547
    return ret;
548
}
549
550
static av_cold void uninit(AVFilterContext *ctx)
551
{
552
    AudioCrossoverContext *s = ctx->priv;
553
    int i;
554
555
    av_freep(&s->fdsp);
556
    av_frame_free(&s->xover);
557
558
    for (i = 0; i < ctx->nb_outputs; i++)
559
        av_freep(&ctx->output_pads[i].name);
560
}
561
562
static const AVFilterPad inputs[] = {
563
    {
564
        .name         = "default",
565
        .type         = AVMEDIA_TYPE_AUDIO,
566
        .filter_frame = filter_frame,
567
        .config_props = config_input,
568
    },
569
    { NULL }
570
};
571
572
AVFilter ff_af_acrossover = {
573
    .name           = "acrossover",
574
    .description    = NULL_IF_CONFIG_SMALL("Split audio into per-bands streams."),
575
    .priv_size      = sizeof(AudioCrossoverContext),
576
    .priv_class     = &acrossover_class,
577
    .init           = init,
578
    .uninit         = uninit,
579
    .query_formats  = query_formats,
580
    .inputs         = inputs,
581
    .outputs        = NULL,
582
    .flags          = AVFILTER_FLAG_DYNAMIC_OUTPUTS |
583
                      AVFILTER_FLAG_SLICE_THREADS,
584
};