1 /*****************************************************************************
2 * sofalizer.c : SOFAlizer filter for virtual binaural acoustics
3 *****************************************************************************
4 * Copyright (C) 2013-2015 Andreas Fuchs, Wolfgang Hrauda,
5 * Acoustics Research Institute (ARI), Vienna, Austria
7 * Authors: Andreas Fuchs <andi.fuchs.mail@gmail.com>
8 * Wolfgang Hrauda <wolfgang.hrauda@gmx.at>
10 * SOFAlizer project coordinator at ARI, main developer of SOFA:
11 * Piotr Majdak <piotr@majdak.at>
13 * This program is free software; you can redistribute it and/or modify it
14 * under the terms of the GNU Lesser General Public License as published by
15 * the Free Software Foundation; either version 2.1 of the License, or
16 * (at your option) any later version.
18 * This program is distributed in the hope that it will be useful,
19 * but WITHOUT ANY WARRANTY; without even the implied warranty of
20 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
21 * GNU Lesser General Public License for more details.
23 * You should have received a copy of the GNU Lesser General Public License
24 * along with this program; if not, write to the Free Software Foundation,
25 * Inc., 51 Franklin Street, Fifth Floor, Boston MA 02110-1301, USA.
26 *****************************************************************************/
31 #include "libavcodec/avfft.h"
32 #include "libavutil/avstring.h"
33 #include "libavutil/channel_layout.h"
34 #include "libavutil/float_dsp.h"
35 #include "libavutil/intmath.h"
36 #include "libavutil/opt.h"
43 #define FREQUENCY_DOMAIN 1
45 typedef struct MySofa { /* contains data of one SOFA file */
46 struct MYSOFA_HRTF *hrtf;
47 struct MYSOFA_LOOKUP *lookup;
48 struct MYSOFA_NEIGHBORHOOD *neighborhood;
49 int ir_samples; /* length of one impulse response (IR) */
50 int n_samples; /* ir_samples to next power of 2 */
51 float *lir, *rir; /* IRs (time-domain) */
56 typedef struct VirtualSpeaker {
62 typedef struct SOFAlizerContext {
65 char *filename; /* name of SOFA file */
66 MySofa sofa; /* contains data of the SOFA file */
68 int sample_rate; /* sample rate from SOFA file */
69 float *speaker_azim; /* azimuth of the virtual loudspeakers */
70 float *speaker_elev; /* elevation of the virtual loudspeakers */
71 char *speakers_pos; /* custom positions of the virtual loudspeakers */
72 float lfe_gain; /* initial gain for the LFE channel */
73 float gain_lfe; /* gain applied to LFE channel */
74 int lfe_channel; /* LFE channel position in channel layout */
76 int n_conv; /* number of channels to convolute */
78 /* buffer variables (for convolution) */
79 float *ringbuffer[2]; /* buffers input samples, length of one buffer: */
80 /* no. input ch. (incl. LFE) x buffer_length */
81 int write[2]; /* current write position to ringbuffer */
82 int buffer_length; /* is: longest IR plus max. delay in all SOFA files */
83 /* then choose next power of 2 */
84 int n_fft; /* number of samples in one FFT block */
87 /* netCDF variables */
88 int *delay[2]; /* broadband delay for each channel/IR to be convolved */
90 float *data_ir[2]; /* IRs for all channels to be convolved */
91 /* (this excludes the LFE) */
93 FFTComplex *temp_fft[2]; /* Array to hold FFT values */
94 FFTComplex *temp_afft[2]; /* Array to accumulate FFT values prior to IFFT */
96 /* control variables */
97 float gain; /* filter gain (in dB) */
98 float rotation; /* rotation of virtual loudspeakers (in degrees) */
99 float elevation; /* elevation of virtual loudspeakers (in deg.) */
100 float radius; /* distance virtual loudspeakers to listener (in metres) */
101 int type; /* processing type */
102 int framesize; /* size of buffer */
103 int normalize; /* should all IRs be normalized upon import ? */
104 int interpolate; /* should wanted IRs be interpolated from neighbors ? */
105 int minphase; /* should all IRs be minphased upon import ? */
106 float anglestep; /* neighbor search angle step, in agles */
107 float radstep; /* neighbor search radius step, in meters */
109 VirtualSpeaker vspkrpos[64];
111 FFTContext *fft[2], *ifft[2];
112 FFTComplex *data_hrtf[2];
114 AVFloatDSPContext *fdsp;
117 static int close_sofa(struct MySofa *sofa)
119 if (sofa->neighborhood)
120 mysofa_neighborhood_free(sofa->neighborhood);
121 sofa->neighborhood = NULL;
123 mysofa_lookup_free(sofa->lookup);
126 mysofa_free(sofa->hrtf);
128 av_freep(&sofa->fir);
133 static int preload_sofa(AVFilterContext *ctx, char *filename, int *samplingrate)
135 struct SOFAlizerContext *s = ctx->priv;
136 struct MYSOFA_HRTF *mysofa;
140 mysofa = mysofa_load(filename, &ret);
141 s->sofa.hrtf = mysofa;
142 if (ret || !mysofa) {
143 av_log(ctx, AV_LOG_ERROR, "Can't find SOFA-file '%s'\n", filename);
144 return AVERROR(EINVAL);
147 ret = mysofa_check(mysofa);
148 if (ret != MYSOFA_OK) {
149 av_log(ctx, AV_LOG_ERROR, "Selected SOFA file is invalid. Please select valid SOFA file.\n");
154 mysofa_loudness(s->sofa.hrtf);
157 mysofa_minphase(s->sofa.hrtf, 0.01f);
159 mysofa_tocartesian(s->sofa.hrtf);
161 s->sofa.lookup = mysofa_lookup_init(s->sofa.hrtf);
162 if (s->sofa.lookup == NULL)
163 return AVERROR(EINVAL);
166 s->sofa.neighborhood = mysofa_neighborhood_init_withstepdefine(s->sofa.hrtf,
171 s->sofa.fir = av_calloc(s->sofa.hrtf->N * s->sofa.hrtf->R, sizeof(*s->sofa.fir));
173 return AVERROR(ENOMEM);
175 if (mysofa->DataSamplingRate.elements != 1)
176 return AVERROR(EINVAL);
177 av_log(ctx, AV_LOG_DEBUG, "Original IR length: %d.\n", mysofa->N);
178 *samplingrate = mysofa->DataSamplingRate.values[0];
179 license = mysofa_getAttribute(mysofa->attributes, (char *)"License");
181 av_log(ctx, AV_LOG_INFO, "SOFA license: %s\n", license);
186 static int parse_channel_name(AVFilterContext *ctx, char **arg, int *rchannel)
188 int len, i, channel_id = 0;
189 int64_t layout, layout0;
192 /* try to parse a channel name, e.g. "FL" */
193 if (av_sscanf(*arg, "%7[A-Z]%n", buf, &len)) {
194 layout0 = layout = av_get_channel_layout(buf);
195 /* channel_id <- first set bit in layout */
196 for (i = 32; i > 0; i >>= 1) {
197 if (layout >= 1LL << i) {
202 /* reject layouts that are not a single channel */
203 if (channel_id >= 64 || layout0 != 1LL << channel_id) {
204 av_log(ctx, AV_LOG_WARNING, "Failed to parse \'%s\' as channel name.\n", buf);
205 return AVERROR(EINVAL);
207 *rchannel = channel_id;
210 } else if (av_sscanf(*arg, "%d%n", &channel_id, &len) == 1) {
211 if (channel_id < 0 || channel_id >= 64) {
212 av_log(ctx, AV_LOG_WARNING, "Failed to parse \'%d\' as channel number.\n", channel_id);
213 return AVERROR(EINVAL);
215 *rchannel = channel_id;
219 return AVERROR(EINVAL);
222 static void parse_speaker_pos(AVFilterContext *ctx, int64_t in_channel_layout)
224 SOFAlizerContext *s = ctx->priv;
225 char *arg, *tokenizer, *p, *args = av_strdup(s->speakers_pos);
231 while ((arg = av_strtok(p, "|", &tokenizer))) {
236 if (parse_channel_name(ctx, &arg, &out_ch_id)) {
239 if (av_sscanf(arg, "%f %f", &azim, &elev) == 2) {
240 s->vspkrpos[out_ch_id].set = 1;
241 s->vspkrpos[out_ch_id].azim = azim;
242 s->vspkrpos[out_ch_id].elev = elev;
243 } else if (av_sscanf(arg, "%f", &azim) == 1) {
244 s->vspkrpos[out_ch_id].set = 1;
245 s->vspkrpos[out_ch_id].azim = azim;
246 s->vspkrpos[out_ch_id].elev = 0;
253 static int get_speaker_pos(AVFilterContext *ctx,
254 float *speaker_azim, float *speaker_elev)
256 struct SOFAlizerContext *s = ctx->priv;
257 uint64_t channels_layout = ctx->inputs[0]->channel_layout;
258 float azim[64] = { 0 };
259 float elev[64] = { 0 };
260 int m, ch, n_conv = ctx->inputs[0]->channels; /* get no. input channels */
262 if (n_conv < 0 || n_conv > 64)
263 return AVERROR(EINVAL);
268 parse_speaker_pos(ctx, channels_layout);
270 /* set speaker positions according to input channel configuration: */
271 for (m = 0, ch = 0; ch < n_conv && m < 64; m++) {
272 uint64_t mask = channels_layout & (1ULL << m);
275 case AV_CH_FRONT_LEFT: azim[ch] = 30; break;
276 case AV_CH_FRONT_RIGHT: azim[ch] = 330; break;
277 case AV_CH_FRONT_CENTER: azim[ch] = 0; break;
278 case AV_CH_LOW_FREQUENCY:
279 case AV_CH_LOW_FREQUENCY_2: s->lfe_channel = ch; break;
280 case AV_CH_BACK_LEFT: azim[ch] = 150; break;
281 case AV_CH_BACK_RIGHT: azim[ch] = 210; break;
282 case AV_CH_BACK_CENTER: azim[ch] = 180; break;
283 case AV_CH_SIDE_LEFT: azim[ch] = 90; break;
284 case AV_CH_SIDE_RIGHT: azim[ch] = 270; break;
285 case AV_CH_FRONT_LEFT_OF_CENTER: azim[ch] = 15; break;
286 case AV_CH_FRONT_RIGHT_OF_CENTER: azim[ch] = 345; break;
287 case AV_CH_TOP_CENTER: azim[ch] = 0;
288 elev[ch] = 90; break;
289 case AV_CH_TOP_FRONT_LEFT: azim[ch] = 30;
290 elev[ch] = 45; break;
291 case AV_CH_TOP_FRONT_CENTER: azim[ch] = 0;
292 elev[ch] = 45; break;
293 case AV_CH_TOP_FRONT_RIGHT: azim[ch] = 330;
294 elev[ch] = 45; break;
295 case AV_CH_TOP_BACK_LEFT: azim[ch] = 150;
296 elev[ch] = 45; break;
297 case AV_CH_TOP_BACK_RIGHT: azim[ch] = 210;
298 elev[ch] = 45; break;
299 case AV_CH_TOP_BACK_CENTER: azim[ch] = 180;
300 elev[ch] = 45; break;
301 case AV_CH_WIDE_LEFT: azim[ch] = 90; break;
302 case AV_CH_WIDE_RIGHT: azim[ch] = 270; break;
303 case AV_CH_SURROUND_DIRECT_LEFT: azim[ch] = 90; break;
304 case AV_CH_SURROUND_DIRECT_RIGHT: azim[ch] = 270; break;
305 case AV_CH_STEREO_LEFT: azim[ch] = 90; break;
306 case AV_CH_STEREO_RIGHT: azim[ch] = 270; break;
309 return AVERROR(EINVAL);
312 if (s->vspkrpos[m].set) {
313 azim[ch] = s->vspkrpos[m].azim;
314 elev[ch] = s->vspkrpos[m].elev;
321 memcpy(speaker_azim, azim, n_conv * sizeof(float));
322 memcpy(speaker_elev, elev, n_conv * sizeof(float));
328 typedef struct ThreadData {
336 FFTComplex **temp_fft;
337 FFTComplex **temp_afft;
340 static int sofalizer_convolute(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
342 SOFAlizerContext *s = ctx->priv;
343 ThreadData *td = arg;
344 AVFrame *in = td->in, *out = td->out;
346 int *write = &td->write[jobnr];
347 const int *const delay = td->delay[jobnr];
348 const float *const ir = td->ir[jobnr];
349 int *n_clippings = &td->n_clippings[jobnr];
350 float *ringbuffer = td->ringbuffer[jobnr];
351 float *temp_src = td->temp_src[jobnr];
352 const int ir_samples = s->sofa.ir_samples; /* length of one IR */
353 const int n_samples = s->sofa.n_samples;
354 const int planar = in->format == AV_SAMPLE_FMT_FLTP;
355 const int mult = 1 + !planar;
356 const float *src = (const float *)in->extended_data[0]; /* get pointer to audio input buffer */
357 float *dst = (float *)out->extended_data[jobnr * planar]; /* get pointer to audio output buffer */
358 const int in_channels = s->n_conv; /* number of input channels */
359 /* ring buffer length is: longest IR plus max. delay -> next power of 2 */
360 const int buffer_length = s->buffer_length;
361 /* -1 for AND instead of MODULO (applied to powers of 2): */
362 const uint32_t modulo = (uint32_t)buffer_length - 1;
363 float *buffer[64]; /* holds ringbuffer for each input channel */
371 for (l = 0; l < in_channels; l++) {
372 /* get starting address of ringbuffer for each input channel */
373 buffer[l] = ringbuffer + l * buffer_length;
376 for (i = 0; i < in->nb_samples; i++) {
377 const float *temp_ir = ir; /* using same set of IRs for each sample */
381 for (l = 0; l < in_channels; l++) {
382 const float *srcp = (const float *)in->extended_data[l];
384 /* write current input sample to ringbuffer (for each channel) */
385 buffer[l][wr] = srcp[i];
388 for (l = 0; l < in_channels; l++) {
389 /* write current input sample to ringbuffer (for each channel) */
390 buffer[l][wr] = src[l];
394 /* loop goes through all channels to be convolved */
395 for (l = 0; l < in_channels; l++) {
396 const float *const bptr = buffer[l];
398 if (l == s->lfe_channel) {
399 /* LFE is an input channel but requires no convolution */
400 /* apply gain to LFE signal and add to output buffer */
401 dst[0] += *(buffer[s->lfe_channel] + wr) * s->gain_lfe;
402 temp_ir += n_samples;
406 /* current read position in ringbuffer: input sample write position
407 * - delay for l-th ch. + diff. betw. IR length and buffer length
408 * (mod buffer length) */
409 read = (wr - delay[l] - (ir_samples - 1) + buffer_length) & modulo;
411 if (read + ir_samples < buffer_length) {
412 memmove(temp_src, bptr + read, ir_samples * sizeof(*temp_src));
414 int len = FFMIN(n_samples - (read % ir_samples), buffer_length - read);
416 memmove(temp_src, bptr + read, len * sizeof(*temp_src));
417 memmove(temp_src + len, bptr, (n_samples - len) * sizeof(*temp_src));
420 /* multiply signal and IR, and add up the results */
421 dst[0] += s->fdsp->scalarproduct_float(temp_ir, temp_src, FFALIGN(ir_samples, 32));
422 temp_ir += n_samples;
425 /* clippings counter */
426 if (fabsf(dst[0]) > 1)
429 /* move output buffer pointer by +2 to get to next sample of processed channel: */
432 wr = (wr + 1) & modulo; /* update ringbuffer write position */
435 *write = wr; /* remember write position in ringbuffer for next call */
440 static int sofalizer_fast_convolute(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
442 SOFAlizerContext *s = ctx->priv;
443 ThreadData *td = arg;
444 AVFrame *in = td->in, *out = td->out;
446 int *write = &td->write[jobnr];
447 FFTComplex *hrtf = s->data_hrtf[jobnr]; /* get pointers to current HRTF data */
448 int *n_clippings = &td->n_clippings[jobnr];
449 float *ringbuffer = td->ringbuffer[jobnr];
450 const int ir_samples = s->sofa.ir_samples; /* length of one IR */
451 const int planar = in->format == AV_SAMPLE_FMT_FLTP;
452 const int mult = 1 + !planar;
453 float *dst = (float *)out->extended_data[jobnr * planar]; /* get pointer to audio output buffer */
454 const int in_channels = s->n_conv; /* number of input channels */
455 /* ring buffer length is: longest IR plus max. delay -> next power of 2 */
456 const int buffer_length = s->buffer_length;
457 /* -1 for AND instead of MODULO (applied to powers of 2): */
458 const uint32_t modulo = (uint32_t)buffer_length - 1;
459 FFTComplex *fft_in = s->temp_fft[jobnr]; /* temporary array for FFT input/output data */
460 FFTComplex *fft_acc = s->temp_afft[jobnr];
461 FFTContext *ifft = s->ifft[jobnr];
462 FFTContext *fft = s->fft[jobnr];
463 const int n_conv = s->n_conv;
464 const int n_fft = s->n_fft;
465 const float fft_scale = 1.0f / s->n_fft;
466 FFTComplex *hrtf_offset;
474 /* find minimum between number of samples and output buffer length:
475 * (important, if one IR is longer than the output buffer) */
476 n_read = FFMIN(ir_samples, in->nb_samples);
477 for (j = 0; j < n_read; j++) {
478 /* initialize output buf with saved signal from overflow buf */
479 dst[mult * j] = ringbuffer[wr];
480 ringbuffer[wr] = 0.0f; /* re-set read samples to zero */
481 /* update ringbuffer read/write position */
482 wr = (wr + 1) & modulo;
485 /* initialize rest of output buffer with 0 */
486 for (j = n_read; j < in->nb_samples; j++) {
490 /* fill FFT accumulation with 0 */
491 memset(fft_acc, 0, sizeof(FFTComplex) * n_fft);
493 for (i = 0; i < n_conv; i++) {
494 const float *src = (const float *)in->extended_data[i * planar]; /* get pointer to audio input buffer */
496 if (i == s->lfe_channel) { /* LFE */
497 if (in->format == AV_SAMPLE_FMT_FLT) {
498 for (j = 0; j < in->nb_samples; j++) {
499 /* apply gain to LFE signal and add to output buffer */
500 dst[2 * j] += src[i + j * in_channels] * s->gain_lfe;
503 for (j = 0; j < in->nb_samples; j++) {
504 /* apply gain to LFE signal and add to output buffer */
505 dst[j] += src[j] * s->gain_lfe;
511 /* outer loop: go through all input channels to be convolved */
512 offset = i * n_fft; /* no. samples already processed */
513 hrtf_offset = hrtf + offset;
515 /* fill FFT input with 0 (we want to zero-pad) */
516 memset(fft_in, 0, sizeof(FFTComplex) * n_fft);
518 if (in->format == AV_SAMPLE_FMT_FLT) {
519 for (j = 0; j < in->nb_samples; j++) {
520 /* prepare input for FFT */
521 /* write all samples of current input channel to FFT input array */
522 fft_in[j].re = src[j * in_channels + i];
525 for (j = 0; j < in->nb_samples; j++) {
526 /* prepare input for FFT */
527 /* write all samples of current input channel to FFT input array */
528 fft_in[j].re = src[j];
532 /* transform input signal of current channel to frequency domain */
533 av_fft_permute(fft, fft_in);
534 av_fft_calc(fft, fft_in);
535 for (j = 0; j < n_fft; j++) {
536 const FFTComplex *hcomplex = hrtf_offset + j;
537 const float re = fft_in[j].re;
538 const float im = fft_in[j].im;
540 /* complex multiplication of input signal and HRTFs */
541 /* output channel (real): */
542 fft_acc[j].re += re * hcomplex->re - im * hcomplex->im;
543 /* output channel (imag): */
544 fft_acc[j].im += re * hcomplex->im + im * hcomplex->re;
548 /* transform output signal of current channel back to time domain */
549 av_fft_permute(ifft, fft_acc);
550 av_fft_calc(ifft, fft_acc);
552 for (j = 0; j < in->nb_samples; j++) {
553 /* write output signal of current channel to output buffer */
554 dst[mult * j] += fft_acc[j].re * fft_scale;
557 for (j = 0; j < ir_samples - 1; j++) { /* overflow length is IR length - 1 */
558 /* write the rest of output signal to overflow buffer */
559 int write_pos = (wr + j) & modulo;
561 *(ringbuffer + write_pos) += fft_acc[in->nb_samples + j].re * fft_scale;
564 /* go through all samples of current output buffer: count clippings */
565 for (i = 0; i < out->nb_samples; i++) {
566 /* clippings counter */
567 if (fabsf(dst[i * mult]) > 1) { /* if current output sample > 1 */
572 /* remember read/write position in ringbuffer for next call */
578 static int filter_frame(AVFilterLink *inlink, AVFrame *in)
580 AVFilterContext *ctx = inlink->dst;
581 SOFAlizerContext *s = ctx->priv;
582 AVFilterLink *outlink = ctx->outputs[0];
583 int n_clippings[2] = { 0 };
587 out = ff_get_audio_buffer(outlink, in->nb_samples);
590 return AVERROR(ENOMEM);
592 av_frame_copy_props(out, in);
594 td.in = in; td.out = out; td.write = s->write;
595 td.delay = s->delay; td.ir = s->data_ir; td.n_clippings = n_clippings;
596 td.ringbuffer = s->ringbuffer; td.temp_src = s->temp_src;
597 td.temp_fft = s->temp_fft;
598 td.temp_afft = s->temp_afft;
600 if (s->type == TIME_DOMAIN) {
601 ctx->internal->execute(ctx, sofalizer_convolute, &td, NULL, 2);
602 } else if (s->type == FREQUENCY_DOMAIN) {
603 ctx->internal->execute(ctx, sofalizer_fast_convolute, &td, NULL, 2);
607 /* display error message if clipping occurred */
608 if (n_clippings[0] + n_clippings[1] > 0) {
609 av_log(ctx, AV_LOG_WARNING, "%d of %d samples clipped. Please reduce gain.\n",
610 n_clippings[0] + n_clippings[1], out->nb_samples * 2);
614 return ff_filter_frame(outlink, out);
617 static int activate(AVFilterContext *ctx)
619 AVFilterLink *inlink = ctx->inputs[0];
620 AVFilterLink *outlink = ctx->outputs[0];
621 SOFAlizerContext *s = ctx->priv;
625 FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink);
628 ret = ff_inlink_consume_samples(inlink, s->nb_samples, s->nb_samples, &in);
630 ret = ff_inlink_consume_frame(inlink, &in);
634 return filter_frame(inlink, in);
636 FF_FILTER_FORWARD_STATUS(inlink, outlink);
637 FF_FILTER_FORWARD_WANTED(outlink, inlink);
639 return FFERROR_NOT_READY;
642 static int query_formats(AVFilterContext *ctx)
644 struct SOFAlizerContext *s = ctx->priv;
645 AVFilterFormats *formats = NULL;
646 AVFilterChannelLayouts *layouts = NULL;
647 int ret, sample_rates[] = { 48000, -1 };
648 static const enum AVSampleFormat sample_fmts[] = {
649 AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLTP,
653 formats = ff_make_format_list(sample_fmts);
655 return AVERROR(ENOMEM);
656 ret = ff_set_common_formats(ctx, formats);
660 layouts = ff_all_channel_layouts();
662 return AVERROR(ENOMEM);
664 ret = ff_channel_layouts_ref(layouts, &ctx->inputs[0]->outcfg.channel_layouts);
669 ret = ff_add_channel_layout(&layouts, AV_CH_LAYOUT_STEREO);
673 ret = ff_channel_layouts_ref(layouts, &ctx->outputs[0]->incfg.channel_layouts);
677 sample_rates[0] = s->sample_rate;
678 formats = ff_make_format_list(sample_rates);
680 return AVERROR(ENOMEM);
681 return ff_set_common_samplerates(ctx, formats);
684 static int getfilter_float(AVFilterContext *ctx, float x, float y, float z,
685 float *left, float *right,
686 float *delay_left, float *delay_right)
688 struct SOFAlizerContext *s = ctx->priv;
689 float c[3], delays[2];
695 c[0] = x, c[1] = y, c[2] = z;
696 nearest = mysofa_lookup(s->sofa.lookup, c);
698 return AVERROR(EINVAL);
700 if (s->interpolate) {
701 neighbors = mysofa_neighborhood(s->sofa.neighborhood, nearest);
702 res = mysofa_interpolate(s->sofa.hrtf, c,
704 s->sofa.fir, delays);
706 if (s->sofa.hrtf->DataDelay.elements > s->sofa.hrtf->R) {
707 delays[0] = s->sofa.hrtf->DataDelay.values[nearest * s->sofa.hrtf->R];
708 delays[1] = s->sofa.hrtf->DataDelay.values[nearest * s->sofa.hrtf->R + 1];
710 delays[0] = s->sofa.hrtf->DataDelay.values[0];
711 delays[1] = s->sofa.hrtf->DataDelay.values[1];
713 res = s->sofa.hrtf->DataIR.values + nearest * s->sofa.hrtf->N * s->sofa.hrtf->R;
716 *delay_left = delays[0];
717 *delay_right = delays[1];
720 fr = res + s->sofa.hrtf->N;
722 memcpy(left, fl, sizeof(float) * s->sofa.hrtf->N);
723 memcpy(right, fr, sizeof(float) * s->sofa.hrtf->N);
728 static int load_data(AVFilterContext *ctx, int azim, int elev, float radius, int sample_rate)
730 struct SOFAlizerContext *s = ctx->priv;
733 int n_conv = s->n_conv; /* no. channels to convolve */
735 float delay_l; /* broadband delay for each IR */
737 int nb_input_channels = ctx->inputs[0]->channels; /* no. input channels */
738 float gain_lin = expf((s->gain - 3 * nb_input_channels) / 20 * M_LN10); /* gain - 3dB/channel */
739 FFTComplex *data_hrtf_l = NULL;
740 FFTComplex *data_hrtf_r = NULL;
741 FFTComplex *fft_in_l = NULL;
742 FFTComplex *fft_in_r = NULL;
743 float *data_ir_l = NULL;
744 float *data_ir_r = NULL;
745 int offset = 0; /* used for faster pointer arithmetics in for-loop */
746 int i, j, azim_orig = azim, elev_orig = elev;
751 av_log(ctx, AV_LOG_DEBUG, "IR length: %d.\n", s->sofa.hrtf->N);
752 s->sofa.ir_samples = s->sofa.hrtf->N;
753 s->sofa.n_samples = 1 << (32 - ff_clz(s->sofa.ir_samples));
755 n_samples = s->sofa.n_samples;
756 ir_samples = s->sofa.ir_samples;
758 if (s->type == TIME_DOMAIN) {
759 s->data_ir[0] = av_calloc(n_samples, sizeof(float) * s->n_conv);
760 s->data_ir[1] = av_calloc(n_samples, sizeof(float) * s->n_conv);
762 if (!s->data_ir[0] || !s->data_ir[1]) {
763 ret = AVERROR(ENOMEM);
768 s->delay[0] = av_calloc(s->n_conv, sizeof(int));
769 s->delay[1] = av_calloc(s->n_conv, sizeof(int));
771 if (!s->delay[0] || !s->delay[1]) {
772 ret = AVERROR(ENOMEM);
776 /* get temporary IR for L and R channel */
777 data_ir_l = av_calloc(n_conv * n_samples, sizeof(*data_ir_l));
778 data_ir_r = av_calloc(n_conv * n_samples, sizeof(*data_ir_r));
779 if (!data_ir_r || !data_ir_l) {
780 ret = AVERROR(ENOMEM);
784 if (s->type == TIME_DOMAIN) {
785 s->temp_src[0] = av_calloc(n_samples, sizeof(float));
786 s->temp_src[1] = av_calloc(n_samples, sizeof(float));
787 if (!s->temp_src[0] || !s->temp_src[1]) {
788 ret = AVERROR(ENOMEM);
793 s->speaker_azim = av_calloc(s->n_conv, sizeof(*s->speaker_azim));
794 s->speaker_elev = av_calloc(s->n_conv, sizeof(*s->speaker_elev));
795 if (!s->speaker_azim || !s->speaker_elev) {
796 ret = AVERROR(ENOMEM);
800 /* get speaker positions */
801 if ((ret = get_speaker_pos(ctx, s->speaker_azim, s->speaker_elev)) < 0) {
802 av_log(ctx, AV_LOG_ERROR, "Couldn't get speaker positions. Input channel configuration not supported.\n");
806 for (i = 0; i < s->n_conv; i++) {
807 float coordinates[3];
809 /* load and store IRs and corresponding delays */
810 azim = (int)(s->speaker_azim[i] + azim_orig) % 360;
811 elev = (int)(s->speaker_elev[i] + elev_orig) % 90;
813 coordinates[0] = azim;
814 coordinates[1] = elev;
815 coordinates[2] = radius;
817 mysofa_s2c(coordinates);
819 /* get id of IR closest to desired position */
820 ret = getfilter_float(ctx, coordinates[0], coordinates[1], coordinates[2],
821 data_ir_l + n_samples * i,
822 data_ir_r + n_samples * i,
827 s->delay[0][i] = delay_l * sample_rate;
828 s->delay[1][i] = delay_r * sample_rate;
830 s->sofa.max_delay = FFMAX3(s->sofa.max_delay, s->delay[0][i], s->delay[1][i]);
833 /* get size of ringbuffer (longest IR plus max. delay) */
834 /* then choose next power of 2 for performance optimization */
835 n_current = n_samples + s->sofa.max_delay;
836 /* length of longest IR plus max. delay */
837 n_max = FFMAX(n_max, n_current);
839 /* buffer length is longest IR plus max. delay -> next power of 2
840 (32 - count leading zeros gives required exponent) */
841 s->buffer_length = 1 << (32 - ff_clz(n_max));
842 s->n_fft = n_fft = 1 << (32 - ff_clz(n_max + s->framesize));
844 if (s->type == FREQUENCY_DOMAIN) {
845 av_fft_end(s->fft[0]);
846 av_fft_end(s->fft[1]);
847 s->fft[0] = av_fft_init(av_log2(s->n_fft), 0);
848 s->fft[1] = av_fft_init(av_log2(s->n_fft), 0);
849 av_fft_end(s->ifft[0]);
850 av_fft_end(s->ifft[1]);
851 s->ifft[0] = av_fft_init(av_log2(s->n_fft), 1);
852 s->ifft[1] = av_fft_init(av_log2(s->n_fft), 1);
854 if (!s->fft[0] || !s->fft[1] || !s->ifft[0] || !s->ifft[1]) {
855 av_log(ctx, AV_LOG_ERROR, "Unable to create FFT contexts of size %d.\n", s->n_fft);
856 ret = AVERROR(ENOMEM);
861 if (s->type == TIME_DOMAIN) {
862 s->ringbuffer[0] = av_calloc(s->buffer_length, sizeof(float) * nb_input_channels);
863 s->ringbuffer[1] = av_calloc(s->buffer_length, sizeof(float) * nb_input_channels);
864 } else if (s->type == FREQUENCY_DOMAIN) {
865 /* get temporary HRTF memory for L and R channel */
866 data_hrtf_l = av_malloc_array(n_fft, sizeof(*data_hrtf_l) * n_conv);
867 data_hrtf_r = av_malloc_array(n_fft, sizeof(*data_hrtf_r) * n_conv);
868 if (!data_hrtf_r || !data_hrtf_l) {
869 ret = AVERROR(ENOMEM);
873 s->ringbuffer[0] = av_calloc(s->buffer_length, sizeof(float));
874 s->ringbuffer[1] = av_calloc(s->buffer_length, sizeof(float));
875 s->temp_fft[0] = av_malloc_array(s->n_fft, sizeof(FFTComplex));
876 s->temp_fft[1] = av_malloc_array(s->n_fft, sizeof(FFTComplex));
877 s->temp_afft[0] = av_malloc_array(s->n_fft, sizeof(FFTComplex));
878 s->temp_afft[1] = av_malloc_array(s->n_fft, sizeof(FFTComplex));
879 if (!s->temp_fft[0] || !s->temp_fft[1] ||
880 !s->temp_afft[0] || !s->temp_afft[1]) {
881 ret = AVERROR(ENOMEM);
886 if (!s->ringbuffer[0] || !s->ringbuffer[1]) {
887 ret = AVERROR(ENOMEM);
891 if (s->type == FREQUENCY_DOMAIN) {
892 fft_in_l = av_calloc(n_fft, sizeof(*fft_in_l));
893 fft_in_r = av_calloc(n_fft, sizeof(*fft_in_r));
894 if (!fft_in_l || !fft_in_r) {
895 ret = AVERROR(ENOMEM);
900 for (i = 0; i < s->n_conv; i++) {
903 offset = i * n_samples; /* no. samples already written */
905 lir = data_ir_l + offset;
906 rir = data_ir_r + offset;
908 if (s->type == TIME_DOMAIN) {
909 for (j = 0; j < ir_samples; j++) {
910 /* load reversed IRs of the specified source position
911 * sample-by-sample for left and right ear; and apply gain */
912 s->data_ir[0][offset + j] = lir[ir_samples - 1 - j] * gain_lin;
913 s->data_ir[1][offset + j] = rir[ir_samples - 1 - j] * gain_lin;
915 } else if (s->type == FREQUENCY_DOMAIN) {
916 memset(fft_in_l, 0, n_fft * sizeof(*fft_in_l));
917 memset(fft_in_r, 0, n_fft * sizeof(*fft_in_r));
919 offset = i * n_fft; /* no. samples already written */
920 for (j = 0; j < ir_samples; j++) {
921 /* load non-reversed IRs of the specified source position
922 * sample-by-sample and apply gain,
923 * L channel is loaded to real part, R channel to imag part,
924 * IRs are shifted by L and R delay */
925 fft_in_l[s->delay[0][i] + j].re = lir[j] * gain_lin;
926 fft_in_r[s->delay[1][i] + j].re = rir[j] * gain_lin;
929 /* actually transform to frequency domain (IRs -> HRTFs) */
930 av_fft_permute(s->fft[0], fft_in_l);
931 av_fft_calc(s->fft[0], fft_in_l);
932 memcpy(data_hrtf_l + offset, fft_in_l, n_fft * sizeof(*fft_in_l));
933 av_fft_permute(s->fft[0], fft_in_r);
934 av_fft_calc(s->fft[0], fft_in_r);
935 memcpy(data_hrtf_r + offset, fft_in_r, n_fft * sizeof(*fft_in_r));
939 if (s->type == FREQUENCY_DOMAIN) {
940 s->data_hrtf[0] = av_malloc_array(n_fft * s->n_conv, sizeof(FFTComplex));
941 s->data_hrtf[1] = av_malloc_array(n_fft * s->n_conv, sizeof(FFTComplex));
942 if (!s->data_hrtf[0] || !s->data_hrtf[1]) {
943 ret = AVERROR(ENOMEM);
947 memcpy(s->data_hrtf[0], data_hrtf_l, /* copy HRTF data to */
948 sizeof(FFTComplex) * n_conv * n_fft); /* filter struct */
949 memcpy(s->data_hrtf[1], data_hrtf_r,
950 sizeof(FFTComplex) * n_conv * n_fft);
954 av_freep(&data_hrtf_l); /* free temporary HRTF memory */
955 av_freep(&data_hrtf_r);
957 av_freep(&data_ir_l); /* free temprary IR memory */
958 av_freep(&data_ir_r);
960 av_freep(&fft_in_l); /* free temporary FFT memory */
966 static av_cold int init(AVFilterContext *ctx)
968 SOFAlizerContext *s = ctx->priv;
972 av_log(ctx, AV_LOG_ERROR, "Valid SOFA filename must be set.\n");
973 return AVERROR(EINVAL);
976 /* preload SOFA file, */
977 ret = preload_sofa(ctx, s->filename, &s->sample_rate);
979 /* file loading error */
980 av_log(ctx, AV_LOG_ERROR, "Error while loading SOFA file: '%s'\n", s->filename);
981 } else { /* no file loading error, resampling not required */
982 av_log(ctx, AV_LOG_DEBUG, "File '%s' loaded.\n", s->filename);
986 av_log(ctx, AV_LOG_ERROR, "No valid SOFA file could be loaded. Please specify valid SOFA file.\n");
990 s->fdsp = avpriv_float_dsp_alloc(0);
992 return AVERROR(ENOMEM);
997 static int config_input(AVFilterLink *inlink)
999 AVFilterContext *ctx = inlink->dst;
1000 SOFAlizerContext *s = ctx->priv;
1003 if (s->type == FREQUENCY_DOMAIN)
1004 s->nb_samples = s->framesize;
1006 /* gain -3 dB per channel */
1007 s->gain_lfe = expf((s->gain - 3 * inlink->channels + s->lfe_gain) / 20 * M_LN10);
1009 s->n_conv = inlink->channels;
1011 /* load IRs to data_ir[0] and data_ir[1] for required directions */
1012 if ((ret = load_data(ctx, s->rotation, s->elevation, s->radius, inlink->sample_rate)) < 0)
1015 av_log(ctx, AV_LOG_DEBUG, "Samplerate: %d Channels to convolute: %d, Length of ringbuffer: %d x %d\n",
1016 inlink->sample_rate, s->n_conv, inlink->channels, s->buffer_length);
1021 static av_cold void uninit(AVFilterContext *ctx)
1023 SOFAlizerContext *s = ctx->priv;
1025 close_sofa(&s->sofa);
1026 av_fft_end(s->ifft[0]);
1027 av_fft_end(s->ifft[1]);
1028 av_fft_end(s->fft[0]);
1029 av_fft_end(s->fft[1]);
1034 av_freep(&s->delay[0]);
1035 av_freep(&s->delay[1]);
1036 av_freep(&s->data_ir[0]);
1037 av_freep(&s->data_ir[1]);
1038 av_freep(&s->ringbuffer[0]);
1039 av_freep(&s->ringbuffer[1]);
1040 av_freep(&s->speaker_azim);
1041 av_freep(&s->speaker_elev);
1042 av_freep(&s->temp_src[0]);
1043 av_freep(&s->temp_src[1]);
1044 av_freep(&s->temp_afft[0]);
1045 av_freep(&s->temp_afft[1]);
1046 av_freep(&s->temp_fft[0]);
1047 av_freep(&s->temp_fft[1]);
1048 av_freep(&s->data_hrtf[0]);
1049 av_freep(&s->data_hrtf[1]);
1053 #define OFFSET(x) offsetof(SOFAlizerContext, x)
1054 #define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
1056 static const AVOption sofalizer_options[] = {
1057 { "sofa", "sofa filename", OFFSET(filename), AV_OPT_TYPE_STRING, {.str=NULL}, .flags = FLAGS },
1058 { "gain", "set gain in dB", OFFSET(gain), AV_OPT_TYPE_FLOAT, {.dbl=0}, -20, 40, .flags = FLAGS },
1059 { "rotation", "set rotation" , OFFSET(rotation), AV_OPT_TYPE_FLOAT, {.dbl=0}, -360, 360, .flags = FLAGS },
1060 { "elevation", "set elevation", OFFSET(elevation), AV_OPT_TYPE_FLOAT, {.dbl=0}, -90, 90, .flags = FLAGS },
1061 { "radius", "set radius", OFFSET(radius), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 5, .flags = FLAGS },
1062 { "type", "set processing", OFFSET(type), AV_OPT_TYPE_INT, {.i64=1}, 0, 1, .flags = FLAGS, "type" },
1063 { "time", "time domain", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, .flags = FLAGS, "type" },
1064 { "freq", "frequency domain", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, .flags = FLAGS, "type" },
1065 { "speakers", "set speaker custom positions", OFFSET(speakers_pos), AV_OPT_TYPE_STRING, {.str=0}, 0, 0, .flags = FLAGS },
1066 { "lfegain", "set lfe gain", OFFSET(lfe_gain), AV_OPT_TYPE_FLOAT, {.dbl=0}, -20,40, .flags = FLAGS },
1067 { "framesize", "set frame size", OFFSET(framesize), AV_OPT_TYPE_INT, {.i64=1024},1024,96000, .flags = FLAGS },
1068 { "normalize", "normalize IRs", OFFSET(normalize), AV_OPT_TYPE_BOOL, {.i64=1}, 0, 1, .flags = FLAGS },
1069 { "interpolate","interpolate IRs from neighbors", OFFSET(interpolate),AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, .flags = FLAGS },
1070 { "minphase", "minphase IRs", OFFSET(minphase), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, .flags = FLAGS },
1071 { "anglestep", "set neighbor search angle step", OFFSET(anglestep), AV_OPT_TYPE_FLOAT, {.dbl=.5}, 0.01, 10, .flags = FLAGS },
1072 { "radstep", "set neighbor search radius step", OFFSET(radstep), AV_OPT_TYPE_FLOAT, {.dbl=.01}, 0.01, 1, .flags = FLAGS },
1076 AVFILTER_DEFINE_CLASS(sofalizer);
1078 static const AVFilterPad inputs[] = {
1081 .type = AVMEDIA_TYPE_AUDIO,
1082 .config_props = config_input,
1087 static const AVFilterPad outputs[] = {
1090 .type = AVMEDIA_TYPE_AUDIO,
1095 AVFilter ff_af_sofalizer = {
1096 .name = "sofalizer",
1097 .description = NULL_IF_CONFIG_SMALL("SOFAlizer (Spatially Oriented Format for Acoustics)."),
1098 .priv_size = sizeof(SOFAlizerContext),
1099 .priv_class = &sofalizer_class,
1101 .activate = activate,
1103 .query_formats = query_formats,
1106 .flags = AVFILTER_FLAG_SLICE_THREADS,