FFmpeg coverage

Line	Branch	Exec	Source
1			/*
2			* Copyright (c) 2017 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
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			/**
22			* @file
23			* An arbitrary audio FIR filter
24			*/
25
26			#include <float.h>
27
28			#include "libavutil/avassert.h"
29			#include "libavutil/cpu.h"
30			#include "libavutil/mem.h"
31			#include "libavutil/tx.h"
32			#include "libavutil/avstring.h"
33			#include "libavutil/channel_layout.h"
34			#include "libavutil/float_dsp.h"
35			#include "libavutil/frame.h"
36			#include "libavutil/log.h"
37			#include "libavutil/opt.h"
38			#include "libavutil/rational.h"
39
40			#include "audio.h"
41			#include "avfilter.h"
42			#include "filters.h"
43			#include "formats.h"
44			#include "af_afirdsp.h"
45
46			#define MAX_IR_STREAMS 32
47
48			typedef struct AudioFIRSegment {
49			int nb_partitions;
50			int part_size;
51			int block_size;
52			int fft_length;
53			int coeff_size;
54			int input_size;
55			int input_offset;
56
57			int *output_offset;
58			int *part_index;
59
60			AVFrame *sumin;
61			AVFrame *sumout;
62			AVFrame *blockout;
63			AVFrame *tempin;
64			AVFrame *tempout;
65			AVFrame *buffer;
66			AVFrame *coeff;
67			AVFrame *input;
68			AVFrame *output;
69
70			AVTXContext **ctx, **tx, **itx;
71			av_tx_fn ctx_fn, tx_fn, itx_fn;
72			} AudioFIRSegment;
73
74			typedef struct AudioFIRContext {
75			const AVClass *class;
76
77			float wet_gain;
78			float dry_gain;
79			float length;
80			int gtype;
81			float ir_norm;
82			float ir_link;
83			float ir_gain;
84			int ir_format;
85			int ir_load;
86			float max_ir_len;
87			int response;
88			int w, h;
89			AVRational frame_rate;
90			int ir_channel;
91			int minp;
92			int maxp;
93			int nb_irs;
94			int prev_selir;
95			int selir;
96			int precision;
97			int format;
98
99			int eof_coeffs[MAX_IR_STREAMS];
100			int have_coeffs[MAX_IR_STREAMS];
101			int nb_taps[MAX_IR_STREAMS];
102			int nb_segments[MAX_IR_STREAMS];
103			int max_offset[MAX_IR_STREAMS];
104			int nb_channels;
105			int one2many;
106			int prev_is_disabled;
107			int *loading;
108			double *ch_gain;
109
110			AudioFIRSegment seg[MAX_IR_STREAMS][1024];
111
112			AVFrame *in;
113			AVFrame *xfade[2];
114			AVFrame *fadein[2];
115			AVFrame *ir[MAX_IR_STREAMS];
116			AVFrame *norm_ir[MAX_IR_STREAMS];
117			int min_part_size;
118			int max_part_size;
119			int64_t pts;
120
121			AudioFIRDSPContext afirdsp;
122			AVFloatDSPContext *fdsp;
123			} AudioFIRContext;
124
125			#define DEPTH 32
126			#include "afir_template.c"
127
128			#undef DEPTH
129			#define DEPTH 64
130			#include "afir_template.c"
131
132		✗	static int fir_channel(AVFilterContext *ctx, AVFrame *out, int ch)
133			{
134		✗	AudioFIRContext *s = ctx->priv;
135		✗	const int min_part_size = s->min_part_size;
136		✗	const int prev_selir = s->prev_selir;
137		✗	const int selir = s->selir;
138
139		✗	for (int offset = 0; offset < out->nb_samples; offset += min_part_size) {
140		✗	switch (s->format) {
141		✗	case AV_SAMPLE_FMT_FLTP:
142		✗	fir_quantums_float(ctx, s, out, min_part_size, ch, offset, prev_selir, selir);
143		✗	break;
144		✗	case AV_SAMPLE_FMT_DBLP:
145		✗	fir_quantums_double(ctx, s, out, min_part_size, ch, offset, prev_selir, selir);
146		✗	break;
147			}
148
149		✗	if (selir != prev_selir && s->loading[ch] != 0)
150		✗	s->loading[ch] += min_part_size;
151			}
152
153		✗	return 0;
154			}
155
156		✗	static int fir_channels(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
157			{
158		✗	AVFrame *out = arg;
159		✗	const int start = (out->ch_layout.nb_channels * jobnr) / nb_jobs;
160		✗	const int end = (out->ch_layout.nb_channels * (jobnr+1)) / nb_jobs;
161
162		✗	for (int ch = start; ch < end; ch++)
163		✗	fir_channel(ctx, out, ch);
164
165		✗	return 0;
166			}
167
168		✗	static int fir_frame(AudioFIRContext *s, AVFrame *in, AVFilterLink *outlink)
169			{
170		✗	AVFilterContext *ctx = outlink->src;
171			AVFrame *out;
172
173		✗	out = ff_get_audio_buffer(outlink, in->nb_samples);
174		✗	if (!out) {
175		✗	av_frame_free(&in);
176		✗	return AVERROR(ENOMEM);
177			}
178		✗	av_frame_copy_props(out, in);
179		✗	out->pts = s->pts = in->pts;
180
181		✗	s->in = in;
182		✗	ff_filter_execute(ctx, fir_channels, out, NULL,
183		✗	FFMIN(outlink->ch_layout.nb_channels, ff_filter_get_nb_threads(ctx)));
184		✗	s->prev_is_disabled = ctx->is_disabled;
185
186		✗	av_frame_free(&in);
187		✗	s->in = NULL;
188
189		✗	return ff_filter_frame(outlink, out);
190			}
191
192		✗	static int init_segment(AVFilterContext *ctx, AudioFIRSegment *seg, int selir,
193			int offset, int nb_partitions, int part_size, int index)
194			{
195		✗	AudioFIRContext *s = ctx->priv;
196		✗	const size_t cpu_align = av_cpu_max_align();
197			union { double d; float f; } cscale, scale, iscale;
198			enum AVTXType tx_type;
199			int ret;
200
201		✗	seg->tx = av_calloc(ctx->inputs[0]->ch_layout.nb_channels, sizeof(*seg->tx));
202		✗	seg->ctx = av_calloc(ctx->inputs[0]->ch_layout.nb_channels, sizeof(*seg->ctx));
203		✗	seg->itx = av_calloc(ctx->inputs[0]->ch_layout.nb_channels, sizeof(*seg->itx));
204		✗	if (!seg->tx || !seg->ctx || !seg->itx)
205		✗	return AVERROR(ENOMEM);
206
207		✗	seg->fft_length = (part_size + 1) * 2;
208		✗	seg->part_size = part_size;
209		✗	seg->coeff_size = FFALIGN(seg->part_size + 1, cpu_align);
210		✗	seg->block_size = FFMAX(seg->coeff_size * 2, FFALIGN(seg->fft_length, cpu_align));
211		✗	seg->nb_partitions = nb_partitions;
212		✗	seg->input_size = offset + s->min_part_size;
213		✗	seg->input_offset = offset;
214
215		✗	seg->part_index = av_calloc(ctx->inputs[0]->ch_layout.nb_channels, sizeof(*seg->part_index));
216		✗	seg->output_offset = av_calloc(ctx->inputs[0]->ch_layout.nb_channels, sizeof(*seg->output_offset));
217		✗	if (!seg->part_index || !seg->output_offset)
218		✗	return AVERROR(ENOMEM);
219
220		✗	switch (s->format) {
221		✗	case AV_SAMPLE_FMT_FLTP:
222		✗	cscale.f = 1.f;
223		✗	scale.f = 1.f / sqrtf(2.f * part_size);
224		✗	iscale.f = 1.f / sqrtf(2.f * part_size);
225		✗	tx_type = AV_TX_FLOAT_RDFT;
226		✗	break;
227		✗	case AV_SAMPLE_FMT_DBLP:
228		✗	cscale.d = 1.0;
229		✗	scale.d = 1.0 / sqrt(2.0 * part_size);
230		✗	iscale.d = 1.0 / sqrt(2.0 * part_size);
231		✗	tx_type = AV_TX_DOUBLE_RDFT;
232		✗	break;
233		✗	default:
234			av_assert1(0);
235			}
236
237		✗	for (int ch = 0; ch < ctx->inputs[0]->ch_layout.nb_channels && part_size >= 1; ch++) {
238		✗	ret = av_tx_init(&seg->ctx[ch], &seg->ctx_fn, tx_type,
239			0, 2 * part_size, &cscale, 0);
240		✗	if (ret < 0)
241		✗	return ret;
242
243		✗	ret = av_tx_init(&seg->tx[ch], &seg->tx_fn, tx_type,
244			0, 2 * part_size, &scale, 0);
245		✗	if (ret < 0)
246		✗	return ret;
247		✗	ret = av_tx_init(&seg->itx[ch], &seg->itx_fn, tx_type,
248			1, 2 * part_size, &iscale, 0);
249		✗	if (ret < 0)
250		✗	return ret;
251			}
252
253		✗	seg->sumin = ff_get_audio_buffer(ctx->inputs[0], seg->fft_length);
254		✗	seg->sumout = ff_get_audio_buffer(ctx->inputs[0], seg->fft_length);
255		✗	seg->blockout = ff_get_audio_buffer(ctx->inputs[0], seg->block_size * seg->nb_partitions);
256		✗	seg->tempin = ff_get_audio_buffer(ctx->inputs[0], seg->block_size);
257		✗	seg->tempout = ff_get_audio_buffer(ctx->inputs[0], seg->block_size);
258		✗	seg->buffer = ff_get_audio_buffer(ctx->inputs[0], seg->part_size);
259		✗	seg->input = ff_get_audio_buffer(ctx->inputs[0], seg->input_size);
260		✗	seg->output = ff_get_audio_buffer(ctx->inputs[0], seg->part_size * 5);
261		✗	if (!seg->buffer || !seg->sumin || !seg->sumout || !seg->blockout ||
262		✗	!seg->input || !seg->output || !seg->tempin || !seg->tempout)
263		✗	return AVERROR(ENOMEM);
264
265		✗	return 0;
266			}
267
268		✗	static void uninit_segment(AVFilterContext *ctx, AudioFIRSegment *seg)
269			{
270		✗	AudioFIRContext *s = ctx->priv;
271
272		✗	if (seg->ctx) {
273		✗	for (int ch = 0; ch < s->nb_channels; ch++)
274		✗	av_tx_uninit(&seg->ctx[ch]);
275			}
276		✗	av_freep(&seg->ctx);
277
278		✗	if (seg->tx) {
279		✗	for (int ch = 0; ch < s->nb_channels; ch++)
280		✗	av_tx_uninit(&seg->tx[ch]);
281			}
282		✗	av_freep(&seg->tx);
283
284		✗	if (seg->itx) {
285		✗	for (int ch = 0; ch < s->nb_channels; ch++)
286		✗	av_tx_uninit(&seg->itx[ch]);
287			}
288		✗	av_freep(&seg->itx);
289
290		✗	av_freep(&seg->output_offset);
291		✗	av_freep(&seg->part_index);
292
293		✗	av_frame_free(&seg->tempin);
294		✗	av_frame_free(&seg->tempout);
295		✗	av_frame_free(&seg->blockout);
296		✗	av_frame_free(&seg->sumin);
297		✗	av_frame_free(&seg->sumout);
298		✗	av_frame_free(&seg->buffer);
299		✗	av_frame_free(&seg->input);
300		✗	av_frame_free(&seg->output);
301		✗	seg->input_size = 0;
302
303		✗	for (int i = 0; i < MAX_IR_STREAMS; i++)
304		✗	av_frame_free(&seg->coeff);
305		✗	}
306
307		✗	static int convert_coeffs(AVFilterContext *ctx, int selir)
308			{
309		✗	AudioFIRContext *s = ctx->priv;
310			int ret, nb_taps, cur_nb_taps;
311
312		✗	if (!s->nb_taps[selir]) {
313			int part_size, max_part_size;
314		✗	int left, offset = 0;
315
316		✗	s->nb_taps[selir] = ff_inlink_queued_samples(ctx->inputs[1 + selir]);
317		✗	if (s->nb_taps[selir] <= 0)
318		✗	return AVERROR(EINVAL);
319
320		✗	if (s->minp > s->maxp)
321		✗	s->maxp = s->minp;
322
323		✗	if (s->nb_segments[selir])
324		✗	goto skip;
325
326		✗	left = s->nb_taps[selir];
327		✗	part_size = 1 << av_log2(s->minp);
328		✗	max_part_size = 1 << av_log2(s->maxp);
329
330		✗	for (int i = 0; left > 0; i++) {
331		✗	int step = (part_size == max_part_size) ? INT_MAX : 1 + (i == 0);
332		✗	int nb_partitions = FFMIN(step, (left + part_size - 1) / part_size);
333
334		✗	s->nb_segments[selir] = i + 1;
335		✗	ret = init_segment(ctx, &s->seg[selir][i], selir, offset, nb_partitions, part_size, i);
336		✗	if (ret < 0)
337		✗	return ret;
338		✗	offset += nb_partitions * part_size;
339		✗	s->max_offset[selir] = offset;
340		✗	left -= nb_partitions * part_size;
341		✗	part_size *= 2;
342		✗	part_size = FFMIN(part_size, max_part_size);
343			}
344			}
345
346		✗	skip:
347		✗	if (!s->ir[selir]) {
348		✗	ret = ff_inlink_consume_samples(ctx->inputs[1 + selir], s->nb_taps[selir], s->nb_taps[selir], &s->ir[selir]);
349		✗	if (ret < 0)
350		✗	return ret;
351		✗	if (ret == 0)
352		✗	return AVERROR_BUG;
353			}
354
355		✗	cur_nb_taps = s->ir[selir]->nb_samples;
356		✗	nb_taps = cur_nb_taps;
357
358		✗	if (!s->norm_ir[selir] || s->norm_ir[selir]->nb_samples < nb_taps) {
359		✗	av_frame_free(&s->norm_ir[selir]);
360		✗	s->norm_ir[selir] = ff_get_audio_buffer(ctx->inputs[0], FFALIGN(nb_taps, 8));
361		✗	if (!s->norm_ir[selir])
362		✗	return AVERROR(ENOMEM);
363			}
364
365		✗	av_log(ctx, AV_LOG_DEBUG, "nb_taps: %d\n", cur_nb_taps);
366		✗	av_log(ctx, AV_LOG_DEBUG, "nb_segments: %d\n", s->nb_segments[selir]);
367
368		✗	switch (s->format) {
369		✗	case AV_SAMPLE_FMT_FLTP:
370		✗	for (int ch = 0; ch < s->nb_channels; ch++) {
371		✗	const float *tsrc = (const float *)s->ir[selir]->extended_data[!s->one2many * ch];
372
373		✗	s->ch_gain[ch] = ir_gain_float(ctx, s, nb_taps, tsrc);
374			}
375
376		✗	if (s->ir_link) {
377		✗	float gain = +INFINITY;
378
379		✗	for (int ch = 0; ch < s->nb_channels; ch++)
380		✗	gain = fminf(gain, s->ch_gain[ch]);
381
382		✗	for (int ch = 0; ch < s->nb_channels; ch++)
383		✗	s->ch_gain[ch] = gain;
384			}
385
386		✗	for (int ch = 0; ch < s->nb_channels; ch++) {
387		✗	const float *tsrc = (const float *)s->ir[selir]->extended_data[!s->one2many * ch];
388		✗	float *time = (float *)s->norm_ir[selir]->extended_data[ch];
389
390		✗	memcpy(time, tsrc, sizeof(*time) * nb_taps);
391		✗	for (int i = FFMAX(1, s->length * nb_taps); i < nb_taps; i++)
392		✗	time[i] = 0;
393
394		✗	ir_scale_float(ctx, s, nb_taps, ch, time, s->ch_gain[ch]);
395
396		✗	for (int n = 0; n < s->nb_segments[selir]; n++) {
397		✗	AudioFIRSegment *seg = &s->seg[selir][n];
398
399		✗	if (!seg->coeff)
400		✗	seg->coeff = ff_get_audio_buffer(ctx->inputs[0], seg->nb_partitions * seg->coeff_size * 2);
401		✗	if (!seg->coeff)
402		✗	return AVERROR(ENOMEM);
403
404		✗	for (int i = 0; i < seg->nb_partitions; i++)
405		✗	convert_channel_float(ctx, s, ch, seg, i, selir);
406			}
407			}
408		✗	break;
409		✗	case AV_SAMPLE_FMT_DBLP:
410		✗	for (int ch = 0; ch < s->nb_channels; ch++) {
411		✗	const double *tsrc = (const double *)s->ir[selir]->extended_data[!s->one2many * ch];
412
413		✗	s->ch_gain[ch] = ir_gain_double(ctx, s, nb_taps, tsrc);
414			}
415
416		✗	if (s->ir_link) {
417		✗	double gain = +INFINITY;
418
419		✗	for (int ch = 0; ch < s->nb_channels; ch++)
420		✗	gain = fmin(gain, s->ch_gain[ch]);
421
422		✗	for (int ch = 0; ch < s->nb_channels; ch++)
423		✗	s->ch_gain[ch] = gain;
424			}
425
426		✗	for (int ch = 0; ch < s->nb_channels; ch++) {
427		✗	const double *tsrc = (const double *)s->ir[selir]->extended_data[!s->one2many * ch];
428		✗	double *time = (double *)s->norm_ir[selir]->extended_data[ch];
429
430		✗	memcpy(time, tsrc, sizeof(*time) * nb_taps);
431		✗	for (int i = FFMAX(1, s->length * nb_taps); i < nb_taps; i++)
432		✗	time[i] = 0;
433
434		✗	ir_scale_double(ctx, s, nb_taps, ch, time, s->ch_gain[ch]);
435
436		✗	for (int n = 0; n < s->nb_segments[selir]; n++) {
437		✗	AudioFIRSegment *seg = &s->seg[selir][n];
438
439		✗	if (!seg->coeff)
440		✗	seg->coeff = ff_get_audio_buffer(ctx->inputs[0], seg->nb_partitions * seg->coeff_size * 2);
441		✗	if (!seg->coeff)
442		✗	return AVERROR(ENOMEM);
443
444		✗	for (int i = 0; i < seg->nb_partitions; i++)
445		✗	convert_channel_double(ctx, s, ch, seg, i, selir);
446			}
447			}
448		✗	break;
449			}
450
451		✗	s->have_coeffs[selir] = 1;
452
453		✗	return 0;
454			}
455
456		✗	static int check_ir(AVFilterLink *link, int selir)
457			{
458		✗	AVFilterContext *ctx = link->dst;
459		✗	AudioFIRContext *s = ctx->priv;
460			int nb_taps, max_nb_taps;
461
462		✗	nb_taps = ff_inlink_queued_samples(link);
463		✗	max_nb_taps = s->max_ir_len * ctx->outputs[0]->sample_rate;
464		✗	if (nb_taps > max_nb_taps) {
465		✗	av_log(ctx, AV_LOG_ERROR, "Too big number of coefficients: %d > %d.\n", nb_taps, max_nb_taps);
466		✗	return AVERROR(EINVAL);
467			}
468
469		✗	if (ff_inlink_check_available_samples(link, nb_taps + 1) == 1)
470		✗	s->eof_coeffs[selir] = 1;
471
472		✗	return 0;
473			}
474
475		✗	static int activate(AVFilterContext *ctx)
476			{
477		✗	AudioFIRContext *s = ctx->priv;
478		✗	AVFilterLink *outlink = ctx->outputs[0];
479			int ret, status, available, wanted;
480		✗	AVFrame *in = NULL;
481			int64_t pts;
482
483		✗	FF_FILTER_FORWARD_STATUS_BACK_ALL(ctx->outputs[0], ctx);
484
485		✗	for (int i = 0; i < s->nb_irs; i++) {
486		✗	const int selir = i;
487
488		✗	if (s->ir_load && selir != s->selir)
489		✗	continue;
490
491		✗	if (!s->eof_coeffs[selir]) {
492		✗	ret = check_ir(ctx->inputs[1 + selir], selir);
493		✗	if (ret < 0)
494		✗	return ret;
495
496		✗	if (!s->eof_coeffs[selir]) {
497		✗	if (ff_outlink_frame_wanted(ctx->outputs[0]))
498		✗	ff_inlink_request_frame(ctx->inputs[1 + selir]);
499		✗	return 0;
500			}
501			}
502
503		✗	if (!s->have_coeffs[selir] && s->eof_coeffs[selir]) {
504		✗	ret = convert_coeffs(ctx, selir);
505		✗	if (ret < 0)
506		✗	return ret;
507			}
508			}
509
510		✗	available = ff_inlink_queued_samples(ctx->inputs[0]);
511		✗	wanted = FFMAX(s->min_part_size, (available / s->min_part_size) * s->min_part_size);
512		✗	ret = ff_inlink_consume_samples(ctx->inputs[0], wanted, wanted, &in);
513		✗	if (ret > 0)
514		✗	ret = fir_frame(s, in, outlink);
515
516		✗	if (s->selir != s->prev_selir && s->loading[0] == 0)
517		✗	s->prev_selir = s->selir;
518
519		✗	if (ret < 0)
520		✗	return ret;
521
522		✗	if (ff_inlink_queued_samples(ctx->inputs[0]) >= s->min_part_size) {
523		✗	ff_filter_set_ready(ctx, 10);
524		✗	return 0;
525			}
526
527		✗	if (ff_inlink_acknowledge_status(ctx->inputs[0], &status, &pts)) {
528		✗	if (status == AVERROR_EOF) {
529		✗	ff_outlink_set_status(ctx->outputs[0], status, pts);
530		✗	return 0;
531			}
532			}
533
534		✗	if (ff_outlink_frame_wanted(ctx->outputs[0])) {
535		✗	ff_inlink_request_frame(ctx->inputs[0]);
536		✗	return 0;
537			}
538
539		✗	return FFERROR_NOT_READY;
540			}
541
542		✗	static int query_formats(const AVFilterContext *ctx,
543			AVFilterFormatsConfig **cfg_in,
544			AVFilterFormatsConfig **cfg_out)
545			{
546		✗	const AudioFIRContext *s = ctx->priv;
547			static const enum AVSampleFormat sample_fmts[3][3] = {
548			{ AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_DBLP, AV_SAMPLE_FMT_NONE },
549			{ AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_NONE },
550			{ AV_SAMPLE_FMT_DBLP, AV_SAMPLE_FMT_NONE },
551			};
552			int ret;
553
554		✗	if (!s->ir_format) {
555		✗	AVFilterChannelLayouts *mono = NULL;
556		✗	AVFilterChannelLayouts *layouts = ff_all_channel_counts();
557
558		✗	if ((ret = ff_channel_layouts_ref(layouts, &cfg_in[0]->channel_layouts)) < 0)
559		✗	return ret;
560		✗	if ((ret = ff_channel_layouts_ref(layouts, &cfg_out[0]->channel_layouts)) < 0)
561		✗	return ret;
562
563		✗	ret = ff_add_channel_layout(&mono, &(AVChannelLayout)AV_CHANNEL_LAYOUT_MONO);
564		✗	if (ret)
565		✗	return ret;
566		✗	for (int i = 1; i < ctx->nb_inputs; i++) {
567		✗	if ((ret = ff_channel_layouts_ref(mono, &cfg_in[i]->channel_layouts)) < 0)
568		✗	return ret;
569			}
570			}
571
572		✗	if ((ret = ff_set_sample_formats_from_list2(ctx, cfg_in, cfg_out,
573		✗	sample_fmts[s->precision])) < 0)
574		✗	return ret;
575
576		✗	return 0;
577			}
578
579		✗	static int config_output(AVFilterLink *outlink)
580			{
581		✗	AVFilterContext *ctx = outlink->src;
582		✗	AudioFIRContext *s = ctx->priv;
583			int ret;
584
585		✗	s->one2many = ctx->inputs[1 + s->selir]->ch_layout.nb_channels == 1;
586		✗	outlink->sample_rate = ctx->inputs[0]->sample_rate;
587		✗	outlink->time_base = ctx->inputs[0]->time_base;
588		✗	if ((ret = av_channel_layout_copy(&outlink->ch_layout, &ctx->inputs[0]->ch_layout)) < 0)
589		✗	return ret;
590		✗	outlink->ch_layout.nb_channels = ctx->inputs[0]->ch_layout.nb_channels;
591
592		✗	s->format = outlink->format;
593		✗	s->nb_channels = outlink->ch_layout.nb_channels;
594		✗	s->ch_gain = av_calloc(ctx->inputs[0]->ch_layout.nb_channels, sizeof(*s->ch_gain));
595		✗	s->loading = av_calloc(ctx->inputs[0]->ch_layout.nb_channels, sizeof(*s->loading));
596		✗	if (!s->loading || !s->ch_gain)
597		✗	return AVERROR(ENOMEM);
598
599		✗	s->fadein[0] = ff_get_audio_buffer(outlink, s->min_part_size);
600		✗	s->fadein[1] = ff_get_audio_buffer(outlink, s->min_part_size);
601		✗	if (!s->fadein[0] || !s->fadein[1])
602		✗	return AVERROR(ENOMEM);
603
604		✗	s->xfade[0] = ff_get_audio_buffer(outlink, s->min_part_size);
605		✗	s->xfade[1] = ff_get_audio_buffer(outlink, s->min_part_size);
606		✗	if (!s->xfade[0] || !s->xfade[1])
607		✗	return AVERROR(ENOMEM);
608
609		✗	switch (s->format) {
610		✗	case AV_SAMPLE_FMT_FLTP:
611		✗	for (int ch = 0; ch < s->nb_channels; ch++) {
612		✗	float *dst0 = (float *)s->xfade[0]->extended_data[ch];
613		✗	float *dst1 = (float *)s->xfade[1]->extended_data[ch];
614
615		✗	for (int n = 0; n < s->min_part_size; n++) {
616		✗	dst0[n] = (n + 1.f) / s->min_part_size;
617		✗	dst1[n] = 1.f - dst0[n];
618			}
619			}
620		✗	break;
621		✗	case AV_SAMPLE_FMT_DBLP:
622		✗	for (int ch = 0; ch < s->nb_channels; ch++) {
623		✗	double *dst0 = (double *)s->xfade[0]->extended_data[ch];
624		✗	double *dst1 = (double *)s->xfade[1]->extended_data[ch];
625
626		✗	for (int n = 0; n < s->min_part_size; n++) {
627		✗	dst0[n] = (n + 1.0) / s->min_part_size;
628		✗	dst1[n] = 1.0 - dst0[n];
629			}
630			}
631		✗	break;
632			}
633
634		✗	return 0;
635			}
636
637		✗	static av_cold void uninit(AVFilterContext *ctx)
638			{
639		✗	AudioFIRContext *s = ctx->priv;
640
641		✗	av_freep(&s->fdsp);
642		✗	av_freep(&s->ch_gain);
643		✗	av_freep(&s->loading);
644
645		✗	for (int i = 0; i < s->nb_irs; i++) {
646		✗	for (int j = 0; j < s->nb_segments[i]; j++)
647		✗	uninit_segment(ctx, &s->seg[i][j]);
648
649		✗	av_frame_free(&s->ir[i]);
650		✗	av_frame_free(&s->norm_ir[i]);
651			}
652
653		✗	av_frame_free(&s->fadein[0]);
654		✗	av_frame_free(&s->fadein[1]);
655
656		✗	av_frame_free(&s->xfade[0]);
657		✗	av_frame_free(&s->xfade[1]);
658		✗	}
659
660		✗	static av_cold int init(AVFilterContext *ctx)
661			{
662		✗	AudioFIRContext *s = ctx->priv;
663			AVFilterPad pad;
664			int ret;
665
666		✗	s->prev_selir = FFMIN(s->nb_irs - 1, s->selir);
667
668		✗	pad = (AVFilterPad) {
669			.name = "main",
670			.type = AVMEDIA_TYPE_AUDIO,
671			};
672
673		✗	ret = ff_append_inpad(ctx, &pad);
674		✗	if (ret < 0)
675		✗	return ret;
676
677		✗	for (int n = 0; n < s->nb_irs; n++) {
678		✗	pad = (AVFilterPad) {
679		✗	.name = av_asprintf("ir%d", n),
680			.type = AVMEDIA_TYPE_AUDIO,
681			};
682
683		✗	if (!pad.name)
684		✗	return AVERROR(ENOMEM);
685
686		✗	ret = ff_append_inpad_free_name(ctx, &pad);
687		✗	if (ret < 0)
688		✗	return ret;
689			}
690
691		✗	s->fdsp = avpriv_float_dsp_alloc(0);
692		✗	if (!s->fdsp)
693		✗	return AVERROR(ENOMEM);
694
695		✗	ff_afir_init(&s->afirdsp);
696
697		✗	s->min_part_size = 1 << av_log2(s->minp);
698		✗	s->max_part_size = 1 << av_log2(s->maxp);
699
700		✗	return 0;
701			}
702
703		✗	static int process_command(AVFilterContext *ctx,
704			const char *cmd,
705			const char *arg,
706			char *res,
707			int res_len,
708			int flags)
709			{
710		✗	AudioFIRContext *s = ctx->priv;
711			int prev_selir, ret;
712
713		✗	prev_selir = s->selir;
714		✗	ret = ff_filter_process_command(ctx, cmd, arg, res, res_len, flags);
715		✗	if (ret < 0)
716		✗	return ret;
717
718		✗	s->selir = FFMIN(s->nb_irs - 1, s->selir);
719		✗	if (s->selir != prev_selir) {
720		✗	s->prev_selir = prev_selir;
721
722		✗	for (int ch = 0; ch < s->nb_channels; ch++)
723		✗	s->loading[ch] = 1;
724			}
725
726		✗	return 0;
727			}
728
729			#define AF AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
730			#define AFR AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM
731			#define VF AV_OPT_FLAG_VIDEO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
732			#define OFFSET(x) offsetof(AudioFIRContext, x)
733
734			static const AVOption afir_options[] = {
735			{ "dry", "set dry gain", OFFSET(dry_gain), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, AFR },
736			{ "wet", "set wet gain", OFFSET(wet_gain), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, AFR },
737			{ "length", "set IR length", OFFSET(length), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 1, AF },
738			{ "gtype", "set IR auto gain type",OFFSET(gtype), AV_OPT_TYPE_INT, {.i64=0}, -1, 4, AF|AV_OPT_FLAG_DEPRECATED, .unit = "gtype" },
739			{ "none", "without auto gain", 0, AV_OPT_TYPE_CONST, {.i64=-1}, 0, 0, AF|AV_OPT_FLAG_DEPRECATED, .unit = "gtype" },
740			{ "peak", "peak gain", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF|AV_OPT_FLAG_DEPRECATED, .unit = "gtype" },
741			{ "dc", "DC gain", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF|AV_OPT_FLAG_DEPRECATED, .unit = "gtype" },
742			{ "gn", "gain to noise", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, AF|AV_OPT_FLAG_DEPRECATED, .unit = "gtype" },
743			{ "ac", "AC gain", 0, AV_OPT_TYPE_CONST, {.i64=3}, 0, 0, AF|AV_OPT_FLAG_DEPRECATED, .unit = "gtype" },
744			{ "rms", "RMS gain", 0, AV_OPT_TYPE_CONST, {.i64=4}, 0, 0, AF|AV_OPT_FLAG_DEPRECATED, .unit = "gtype" },
745			{ "irnorm", "set IR norm", OFFSET(ir_norm), AV_OPT_TYPE_FLOAT, {.dbl=1}, -1, 2, AF },
746			{ "irlink", "set IR link", OFFSET(ir_link), AV_OPT_TYPE_BOOL, {.i64=1}, 0, 1, AF },
747			{ "irgain", "set IR gain", OFFSET(ir_gain), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 1, AF },
748			{ "irfmt", "set IR format", OFFSET(ir_format), AV_OPT_TYPE_INT, {.i64=1}, 0, 1, AF, .unit = "irfmt" },
749			{ "mono", "single channel", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, .unit = "irfmt" },
750			{ "input", "same as input", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, .unit = "irfmt" },
751			{ "maxir", "set max IR length", OFFSET(max_ir_len), AV_OPT_TYPE_FLOAT, {.dbl=30}, 0.1, 60, AF },
752			{ "response", "show IR frequency response", OFFSET(response), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, VF|AV_OPT_FLAG_DEPRECATED },
753			{ "channel", "set IR channel to display frequency response", OFFSET(ir_channel), AV_OPT_TYPE_INT, {.i64=0}, 0, 1024, VF|AV_OPT_FLAG_DEPRECATED },
754			{ "size", "set video size", OFFSET(w), AV_OPT_TYPE_IMAGE_SIZE, {.str = "hd720"}, 0, 0, VF|AV_OPT_FLAG_DEPRECATED },
755			{ "rate", "set video rate", OFFSET(frame_rate), AV_OPT_TYPE_VIDEO_RATE, {.str = "25"}, 0, INT32_MAX, VF|AV_OPT_FLAG_DEPRECATED },
756			{ "minp", "set min partition size", OFFSET(minp), AV_OPT_TYPE_INT, {.i64=8192}, 1, 65536, AF },
757			{ "maxp", "set max partition size", OFFSET(maxp), AV_OPT_TYPE_INT, {.i64=8192}, 8, 65536, AF },
758			{ "nbirs", "set number of input IRs",OFFSET(nb_irs),AV_OPT_TYPE_INT, {.i64=1}, 1, 32, AF },
759			{ "ir", "select IR", OFFSET(selir), AV_OPT_TYPE_INT, {.i64=0}, 0, 31, AFR },
760			{ "precision", "set processing precision", OFFSET(precision), AV_OPT_TYPE_INT, {.i64=0}, 0, 2, AF, .unit = "precision" },
761			{ "auto", "set auto processing precision", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, .unit = "precision" },
762			{ "float", "set single-floating point processing precision", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, .unit = "precision" },
763			{ "double","set double-floating point processing precision", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, AF, .unit = "precision" },
764			{ "irload", "set IR loading type", OFFSET(ir_load), AV_OPT_TYPE_INT, {.i64=0}, 0, 1, AF, .unit = "irload" },
765			{ "init", "load all IRs on init", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, .unit = "irload" },
766			{ "access", "load IR on access", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, .unit = "irload" },
767			{ NULL }
768			};
769
770			AVFILTER_DEFINE_CLASS(afir);
771
772			static const AVFilterPad outputs[] = {
773			{
774			.name = "default",
775			.type = AVMEDIA_TYPE_AUDIO,
776			.config_props = config_output,
777			},
778			};
779
780			const FFFilter ff_af_afir = {
781			.p.name = "afir",
782			.p.description = NULL_IF_CONFIG_SMALL("Apply Finite Impulse Response filter with supplied coefficients in additional stream(s)."),
783			.p.priv_class = &afir_class,
784			.p.flags = AVFILTER_FLAG_DYNAMIC_INPUTS |
785			AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL |
786			AVFILTER_FLAG_SLICE_THREADS,
787			.priv_size = sizeof(AudioFIRContext),
788			FILTER_QUERY_FUNC2(query_formats),
789			FILTER_OUTPUTS(outputs),
790			.init = init,
791			.activate = activate,
792			.uninit = uninit,
793			.process_command = process_command,
794			};
795