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avfilter: add arbitrary audio FIR filter

Paul B Mahol committed 9 years ago

Commit 49bbfb91 parent f1a4dd510 files · +764 −1Browse files

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10 files
  1. configure +3 −0
  2. doc/filters.texi +43 −0
  3. libavfilter/Makefile +1 −0
  4. libavfilter/af_afir.c +535 −0
  5. libavfilter/af_afir.h +83 −0
  6. libavfilter/allfilters.c +1 −0
  7. libavfilter/version.h +1 −1
  8. libavfilter/x86/Makefile +2 −0
  9. libavfilter/x86/af_afir.asm +60 −0
  10. libavfilter/x86/af_afir_init.c +35 −0
configure+3−0
@@ -3083,6 +3083,8 @@ unix_protocol_select="network" · 3082 unmodified lines
30833083 # filters
30843084 afftfilt_filter_deps="avcodec"
30853085 afftfilt_filter_select="fft"
3086+afir_filter_deps="avcodec"
3087+afir_filter_select="fft"
30863088 amovie_filter_deps="avcodec avformat"
30873089 aresample_filter_deps="swresample"
30883090 ass_filter_deps="libass"
@@ -6476,6 +6478,7 @@ enabled zlib && add_cppflags -DZLIB_CONST · 3387 unmodified lines
64766478
64776479 # conditional library dependencies, in linking order
64786480 enabled afftfilt_filter && prepend avfilter_deps "avcodec"
6481+enabled afir_filter && prepend avfilter_deps "avcodec"
64796482 enabled amovie_filter && prepend avfilter_deps "avformat avcodec"
64806483 enabled aresample_filter && prepend avfilter_deps "swresample"
64816484 enabled atempo_filter && prepend avfilter_deps "avcodec"
doc/filters.texi+43−0
@@ -878,6 +878,49 @@ afftfilt="1-clip((b/nb)*b,0,1)" · 877 unmodified lines
878878 @end example
879879 @end itemize
880880
881+@section afir
882+
883+Apply an arbitrary Frequency Impulse Response filter.
884+
885+This filter is designed for applying long FIR filters,
886+up to 30 seconds long.
887+
888+It can be used as component for digital crossover filters,
889+room equalization, cross talk cancellation, wavefield synthesis,
890+auralization, ambiophonics and ambisonics.
891+
892+This filter uses second stream as FIR coefficients.
893+If second stream holds single channel, it will be used
894+for all input channels in first stream, otherwise
895+number of channels in second stream must be same as
896+number of channels in first stream.
897+
898+It accepts the following parameters:
899+
900+@table @option
901+@item dry
902+Set dry gain. This sets input gain.
903+
904+@item wet
905+Set wet gain. This sets final output gain.
906+
907+@item length
908+Set Impulse Response filter length. Default is 1, which means whole IR is processed.
909+
910+@item again
911+Enable applying gain measured from power of IR.
912+@end table
913+
914+@subsection Examples
915+
916+@itemize
917+@item
918+Apply reverb to stream using mono IR file as second input, complete command using ffmpeg:
919+@example
920+ffmpeg -i input.wav -i middle_tunnel_1way_mono.wav -lavfi afir output.wav
921+@end example
922+@end itemize
923+
881924 @anchor{aformat}
882925 @section aformat
883926
libavfilter/Makefile+1−0
@@ -37,6 +37,7 @@ OBJS-$(CONFIG_AEMPHASIS_FILTER) += af_aemphasis.o · 36 unmodified lines
3737 OBJS-$(CONFIG_AEVAL_FILTER) += aeval.o
3838 OBJS-$(CONFIG_AFADE_FILTER) += af_afade.o
3939 OBJS-$(CONFIG_AFFTFILT_FILTER) += af_afftfilt.o window_func.o
40+OBJS-$(CONFIG_AFIR_FILTER) += af_afir.o
4041 OBJS-$(CONFIG_AFORMAT_FILTER) += af_aformat.o
4142 OBJS-$(CONFIG_AGATE_FILTER) += af_agate.o
4243 OBJS-$(CONFIG_AINTERLEAVE_FILTER) += f_interleave.o
libavfilter/af_afir.c+535−0
@@ -0,0 +1,535 @@
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 "libavutil/audio_fifo.h"
27+#include "libavutil/common.h"
28+#include "libavutil/float_dsp.h"
29+#include "libavutil/opt.h"
30+#include "libavcodec/avfft.h"
31+
32+#include "audio.h"
33+#include "avfilter.h"
34+#include "formats.h"
35+#include "internal.h"
36+#include "af_afir.h"
37+
38+static void fcmul_add_c(float *sum, const float *t, const float *c, ptrdiff_t len)
39+{
40+ int n;
41+
42+ for (n = 0; n < len; n++) {
43+ const float cre = c[2 * n ];
44+ const float cim = c[2 * n + 1];
45+ const float tre = t[2 * n ];
46+ const float tim = t[2 * n + 1];
47+
48+ sum[2 * n ] += tre * cre - tim * cim;
49+ sum[2 * n + 1] += tre * cim + tim * cre;
50+ }
51+
52+ sum[2 * n] += t[2 * n] * c[2 * n];
53+}
54+
55+static int fir_channel(AVFilterContext *ctx, void *arg, int ch, int nb_jobs)
56+{
57+ AudioFIRContext *s = ctx->priv;
58+ const float *src = (const float *)s->in[0]->extended_data[ch];
59+ int index1 = (s->index + 1) % 3;
60+ int index2 = (s->index + 2) % 3;
61+ float *sum = s->sum[ch];
62+ AVFrame *out = arg;
63+ float *block;
64+ float *dst;
65+ int n, i, j;
66+
67+ memset(sum, 0, sizeof(*sum) * s->fft_length);
68+ block = s->block[ch] + s->part_index * s->block_size;
69+ memset(block, 0, sizeof(*block) * s->fft_length);
70+
71+ s->fdsp->vector_fmul_scalar(block + s->part_size, src, s->dry_gain, s->nb_samples);
72+ emms_c();
73+
74+ av_rdft_calc(s->rdft[ch], block);
75+ block[2 * s->part_size] = block[1];
76+ block[1] = 0;
77+
78+ j = s->part_index;
79+
80+ for (i = 0; i < s->nb_partitions; i++) {
81+ const int coffset = i * s->coeff_size;
82+ const FFTComplex *coeff = s->coeff[ch * !s->one2many] + coffset;
83+
84+ block = s->block[ch] + j * s->block_size;
85+ s->fcmul_add(sum, block, (const float *)coeff, s->part_size);
86+
87+ if (j == 0)
88+ j = s->nb_partitions;
89+ j--;
90+ }
91+
92+ sum[1] = sum[2 * s->part_size];
93+ av_rdft_calc(s->irdft[ch], sum);
94+
95+ dst = (float *)s->buffer->extended_data[ch] + index1 * s->part_size;
96+ for (n = 0; n < s->part_size; n++) {
97+ dst[n] += sum[n];
98+ }
99+
100+ dst = (float *)s->buffer->extended_data[ch] + index2 * s->part_size;
101+
102+ memcpy(dst, sum + s->part_size, s->part_size * sizeof(*dst));
103+
104+ dst = (float *)s->buffer->extended_data[ch] + s->index * s->part_size;
105+
106+ if (out) {
107+ float *ptr = (float *)out->extended_data[ch];
108+ s->fdsp->vector_fmul_scalar(ptr, dst, s->gain * s->wet_gain, out->nb_samples);
109+ emms_c();
110+ }
111+
112+ return 0;
113+}
114+
115+static int fir_frame(AudioFIRContext *s, AVFilterLink *outlink)
116+{
117+ AVFilterContext *ctx = outlink->src;
118+ AVFrame *out = NULL;
119+ int ret;
120+
121+ s->nb_samples = FFMIN(s->part_size, av_audio_fifo_size(s->fifo[0]));
122+
123+ if (!s->want_skip) {
124+ out = ff_get_audio_buffer(outlink, s->nb_samples);
125+ if (!out)
126+ return AVERROR(ENOMEM);
127+ }
128+
129+ s->in[0] = ff_get_audio_buffer(ctx->inputs[0], s->nb_samples);
130+ if (!s->in[0]) {
131+ av_frame_free(&out);
132+ return AVERROR(ENOMEM);
133+ }
134+
135+ av_audio_fifo_peek(s->fifo[0], (void **)s->in[0]->extended_data, s->nb_samples);
136+
137+ ctx->internal->execute(ctx, fir_channel, out, NULL, outlink->channels);
138+
139+ s->part_index = (s->part_index + 1) % s->nb_partitions;
140+
141+ av_audio_fifo_drain(s->fifo[0], s->nb_samples);
142+
143+ if (!s->want_skip) {
144+ out->pts = s->pts;
145+ if (s->pts != AV_NOPTS_VALUE)
146+ s->pts += av_rescale_q(out->nb_samples, (AVRational){1, outlink->sample_rate}, outlink->time_base);
147+ }
148+
149+ s->index++;
150+ if (s->index == 3)
151+ s->index = 0;
152+
153+ av_frame_free(&s->in[0]);
154+
155+ if (s->want_skip == 1) {
156+ s->want_skip = 0;
157+ ret = 0;
158+ } else {
159+ ret = ff_filter_frame(outlink, out);
160+ }
161+
162+ return ret;
163+}
164+
165+static int convert_coeffs(AVFilterContext *ctx)
166+{
167+ AudioFIRContext *s = ctx->priv;
168+ int i, ch, n, N;
169+ float power = 0;
170+
171+ s->nb_taps = av_audio_fifo_size(s->fifo[1]);
172+ if (s->nb_taps <= 0)
173+ return AVERROR(EINVAL);
174+
175+ for (n = 4; (1 << n) < s->nb_taps; n++);
176+ N = FFMIN(n, 16);
177+ s->ir_length = 1 << n;
178+ s->fft_length = (1 << (N + 1)) + 1;
179+ s->part_size = 1 << (N - 1);
180+ s->block_size = FFALIGN(s->fft_length, 32);
181+ s->coeff_size = FFALIGN(s->part_size + 1, 32);
182+ s->nb_partitions = (s->nb_taps + s->part_size - 1) / s->part_size;
183+ s->nb_coeffs = s->ir_length + s->nb_partitions;
184+
185+ for (ch = 0; ch < ctx->inputs[0]->channels; ch++) {
186+ s->sum[ch] = av_calloc(s->fft_length, sizeof(**s->sum));
187+ if (!s->sum[ch])
188+ return AVERROR(ENOMEM);
189+ }
190+
191+ for (ch = 0; ch < ctx->inputs[1]->channels; ch++) {
192+ s->coeff[ch] = av_calloc(s->nb_partitions * s->coeff_size, sizeof(**s->coeff));
193+ if (!s->coeff[ch])
194+ return AVERROR(ENOMEM);
195+ }
196+
197+ for (ch = 0; ch < ctx->inputs[0]->channels; ch++) {
198+ s->block[ch] = av_calloc(s->nb_partitions * s->block_size, sizeof(**s->block));
199+ if (!s->block[ch])
200+ return AVERROR(ENOMEM);
201+ }
202+
203+ for (ch = 0; ch < ctx->inputs[0]->channels; ch++) {
204+ s->rdft[ch] = av_rdft_init(N, DFT_R2C);
205+ s->irdft[ch] = av_rdft_init(N, IDFT_C2R);
206+ if (!s->rdft[ch] || !s->irdft[ch])
207+ return AVERROR(ENOMEM);
208+ }
209+
210+ s->in[1] = ff_get_audio_buffer(ctx->inputs[1], s->nb_taps);
211+ if (!s->in[1])
212+ return AVERROR(ENOMEM);
213+
214+ s->buffer = ff_get_audio_buffer(ctx->inputs[0], s->part_size * 3);
215+ if (!s->buffer)
216+ return AVERROR(ENOMEM);
217+
218+ av_audio_fifo_read(s->fifo[1], (void **)s->in[1]->extended_data, s->nb_taps);
219+
220+ for (ch = 0; ch < ctx->inputs[1]->channels; ch++) {
221+ float *time = (float *)s->in[1]->extended_data[!s->one2many * ch];
222+ float *block = s->block[ch];
223+ FFTComplex *coeff = s->coeff[ch];
224+
225+ power += s->fdsp->scalarproduct_float(time, time, s->nb_taps);
226+
227+ for (i = FFMAX(1, s->length * s->nb_taps); i < s->nb_taps; i++)
228+ time[i] = 0;
229+
230+ for (i = 0; i < s->nb_partitions; i++) {
231+ const float scale = 1.f / s->part_size;
232+ const int toffset = i * s->part_size;
233+ const int coffset = i * s->coeff_size;
234+ const int boffset = s->part_size;
235+ const int remaining = s->nb_taps - (i * s->part_size);
236+ const int size = remaining >= s->part_size ? s->part_size : remaining;
237+
238+ memset(block, 0, sizeof(*block) * s->fft_length);
239+ memcpy(block + boffset, time + toffset, size * sizeof(*block));
240+
241+ av_rdft_calc(s->rdft[0], block);
242+
243+ coeff[coffset].re = block[0] * scale;
244+ coeff[coffset].im = 0;
245+ for (n = 1; n < s->part_size; n++) {
246+ coeff[coffset + n].re = block[2 * n] * scale;
247+ coeff[coffset + n].im = block[2 * n + 1] * scale;
248+ }
249+ coeff[coffset + s->part_size].re = block[1] * scale;
250+ coeff[coffset + s->part_size].im = 0;
251+ }
252+ }
253+
254+ av_frame_free(&s->in[1]);
255+ s->gain = s->again ? 1.f / sqrtf(power / ctx->inputs[1]->channels) : 1.f;
256+ av_log(ctx, AV_LOG_DEBUG, "nb_taps: %d\n", s->nb_taps);
257+ av_log(ctx, AV_LOG_DEBUG, "nb_partitions: %d\n", s->nb_partitions);
258+ av_log(ctx, AV_LOG_DEBUG, "partition size: %d\n", s->part_size);
259+ av_log(ctx, AV_LOG_DEBUG, "ir_length: %d\n", s->ir_length);
260+
261+ s->have_coeffs = 1;
262+
263+ return 0;
264+}
265+
266+static int read_ir(AVFilterLink *link, AVFrame *frame)
267+{
268+ AVFilterContext *ctx = link->dst;
269+ AudioFIRContext *s = ctx->priv;
270+ int nb_taps, max_nb_taps;
271+
272+ av_audio_fifo_write(s->fifo[1], (void **)frame->extended_data,
273+ frame->nb_samples);
274+ av_frame_free(&frame);
275+
276+ nb_taps = av_audio_fifo_size(s->fifo[1]);
277+ max_nb_taps = MAX_IR_DURATION * ctx->outputs[0]->sample_rate;
278+ if (nb_taps > max_nb_taps) {
279+ av_log(ctx, AV_LOG_ERROR, "Too big number of coefficients: %d > %d.\n", nb_taps, max_nb_taps);
280+ return AVERROR(EINVAL);
281+ }
282+
283+ return 0;
284+}
285+
286+static int filter_frame(AVFilterLink *link, AVFrame *frame)
287+{
288+ AVFilterContext *ctx = link->dst;
289+ AudioFIRContext *s = ctx->priv;
290+ AVFilterLink *outlink = ctx->outputs[0];
291+ int ret = 0;
292+
293+ av_audio_fifo_write(s->fifo[0], (void **)frame->extended_data,
294+ frame->nb_samples);
295+ if (s->pts == AV_NOPTS_VALUE)
296+ s->pts = frame->pts;
297+
298+ av_frame_free(&frame);
299+
300+ if (!s->have_coeffs && s->eof_coeffs) {
301+ ret = convert_coeffs(ctx);
302+ if (ret < 0)
303+ return ret;
304+ }
305+
306+ if (s->have_coeffs) {
307+ while (av_audio_fifo_size(s->fifo[0]) >= s->part_size) {
308+ ret = fir_frame(s, outlink);
309+ if (ret < 0)
310+ break;
311+ }
312+ }
313+ return ret;
314+}
315+
316+static int request_frame(AVFilterLink *outlink)
317+{
318+ AVFilterContext *ctx = outlink->src;
319+ AudioFIRContext *s = ctx->priv;
320+ int ret;
321+
322+ if (!s->eof_coeffs) {
323+ ret = ff_request_frame(ctx->inputs[1]);
324+ if (ret == AVERROR_EOF) {
325+ s->eof_coeffs = 1;
326+ ret = 0;
327+ }
328+ return ret;
329+ }
330+ ret = ff_request_frame(ctx->inputs[0]);
331+ if (ret == AVERROR_EOF && s->have_coeffs) {
332+ if (s->need_padding) {
333+ AVFrame *silence = ff_get_audio_buffer(outlink, s->part_size);
334+
335+ if (!silence)
336+ return AVERROR(ENOMEM);
337+ av_audio_fifo_write(s->fifo[0], (void **)silence->extended_data,
338+ silence->nb_samples);
339+ av_frame_free(&silence);
340+ s->need_padding = 0;
341+ }
342+
343+ while (av_audio_fifo_size(s->fifo[0]) > 0) {
344+ ret = fir_frame(s, outlink);
345+ if (ret < 0)
346+ return ret;
347+ }
348+ ret = AVERROR_EOF;
349+ }
350+ return ret;
351+}
352+
353+static int query_formats(AVFilterContext *ctx)
354+{
355+ AVFilterFormats *formats;
356+ AVFilterChannelLayouts *layouts;
357+ static const enum AVSampleFormat sample_fmts[] = {
358+ AV_SAMPLE_FMT_FLTP,
359+ AV_SAMPLE_FMT_NONE
360+ };
361+ int ret, i;
362+
363+ layouts = ff_all_channel_counts();
364+ if ((ret = ff_channel_layouts_ref(layouts, &ctx->outputs[0]->in_channel_layouts)) < 0)
365+ return ret;
366+
367+ for (i = 0; i < 2; i++) {
368+ layouts = ff_all_channel_counts();
369+ if ((ret = ff_channel_layouts_ref(layouts, &ctx->inputs[i]->out_channel_layouts)) < 0)
370+ return ret;
371+ }
372+
373+ formats = ff_make_format_list(sample_fmts);
374+ if ((ret = ff_set_common_formats(ctx, formats)) < 0)
375+ return ret;
376+
377+ formats = ff_all_samplerates();
378+ return ff_set_common_samplerates(ctx, formats);
379+}
380+
381+static int config_output(AVFilterLink *outlink)
382+{
383+ AVFilterContext *ctx = outlink->src;
384+ AudioFIRContext *s = ctx->priv;
385+
386+ if (ctx->inputs[0]->channels != ctx->inputs[1]->channels &&
387+ ctx->inputs[1]->channels != 1) {
388+ av_log(ctx, AV_LOG_ERROR,
389+ "Second input must have same number of channels as first input or "
390+ "exactly 1 channel.\n");
391+ return AVERROR(EINVAL);
392+ }
393+
394+ s->one2many = ctx->inputs[1]->channels == 1;
395+ outlink->sample_rate = ctx->inputs[0]->sample_rate;
396+ outlink->time_base = ctx->inputs[0]->time_base;
397+ outlink->channel_layout = ctx->inputs[0]->channel_layout;
398+ outlink->channels = ctx->inputs[0]->channels;
399+
400+ s->fifo[0] = av_audio_fifo_alloc(ctx->inputs[0]->format, ctx->inputs[0]->channels, 1024);
401+ s->fifo[1] = av_audio_fifo_alloc(ctx->inputs[1]->format, ctx->inputs[1]->channels, 1024);
402+ if (!s->fifo[0] || !s->fifo[1])
403+ return AVERROR(ENOMEM);
404+
405+ s->sum = av_calloc(outlink->channels, sizeof(*s->sum));
406+ s->coeff = av_calloc(ctx->inputs[1]->channels, sizeof(*s->coeff));
407+ s->block = av_calloc(ctx->inputs[0]->channels, sizeof(*s->block));
408+ s->rdft = av_calloc(outlink->channels, sizeof(*s->rdft));
409+ s->irdft = av_calloc(outlink->channels, sizeof(*s->irdft));
410+ if (!s->sum || !s->coeff || !s->block || !s->rdft || !s->irdft)
411+ return AVERROR(ENOMEM);
412+
413+ s->nb_channels = outlink->channels;
414+ s->nb_coef_channels = ctx->inputs[1]->channels;
415+ s->want_skip = 1;
416+ s->need_padding = 1;
417+ s->pts = AV_NOPTS_VALUE;
418+
419+ return 0;
420+}
421+
422+static av_cold void uninit(AVFilterContext *ctx)
423+{
424+ AudioFIRContext *s = ctx->priv;
425+ int ch;
426+
427+ if (s->sum) {
428+ for (ch = 0; ch < s->nb_channels; ch++) {
429+ av_freep(&s->sum[ch]);
430+ }
431+ }
432+ av_freep(&s->sum);
433+
434+ if (s->coeff) {
435+ for (ch = 0; ch < s->nb_coef_channels; ch++) {
436+ av_freep(&s->coeff[ch]);
437+ }
438+ }
439+ av_freep(&s->coeff);
440+
441+ if (s->block) {
442+ for (ch = 0; ch < s->nb_channels; ch++) {
443+ av_freep(&s->block[ch]);
444+ }
445+ }
446+ av_freep(&s->block);
447+
448+ if (s->rdft) {
449+ for (ch = 0; ch < s->nb_channels; ch++) {
450+ av_rdft_end(s->rdft[ch]);
451+ }
452+ }
453+ av_freep(&s->rdft);
454+
455+ if (s->irdft) {
456+ for (ch = 0; ch < s->nb_channels; ch++) {
457+ av_rdft_end(s->irdft[ch]);
458+ }
459+ }
460+ av_freep(&s->irdft);
461+
462+ av_frame_free(&s->in[0]);
463+ av_frame_free(&s->in[1]);
464+ av_frame_free(&s->buffer);
465+
466+ av_audio_fifo_free(s->fifo[0]);
467+ av_audio_fifo_free(s->fifo[1]);
468+
469+ av_freep(&s->fdsp);
470+}
471+
472+static av_cold int init(AVFilterContext *ctx)
473+{
474+ AudioFIRContext *s = ctx->priv;
475+
476+ s->fcmul_add = fcmul_add_c;
477+
478+ s->fdsp = avpriv_float_dsp_alloc(0);
479+ if (!s->fdsp)
480+ return AVERROR(ENOMEM);
481+
482+ if (ARCH_X86)
483+ ff_afir_init_x86(s);
484+
485+ return 0;
486+}
487+
488+static const AVFilterPad afir_inputs[] = {
489+ {
490+ .name = "main",
491+ .type = AVMEDIA_TYPE_AUDIO,
492+ .filter_frame = filter_frame,
493+ },{
494+ .name = "ir",
495+ .type = AVMEDIA_TYPE_AUDIO,
496+ .filter_frame = read_ir,
497+ },
498+ { NULL }
499+};
500+
501+static const AVFilterPad afir_outputs[] = {
502+ {
503+ .name = "default",
504+ .type = AVMEDIA_TYPE_AUDIO,
505+ .config_props = config_output,
506+ .request_frame = request_frame,
507+ },
508+ { NULL }
509+};
510+
511+#define AF AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
512+#define OFFSET(x) offsetof(AudioFIRContext, x)
513+
514+static const AVOption afir_options[] = {
515+ { "dry", "set dry gain", OFFSET(dry_gain), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 1, AF },
516+ { "wet", "set wet gain", OFFSET(wet_gain), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 1, AF },
517+ { "length", "set IR length", OFFSET(length), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 1, AF },
518+ { "again", "enable auto gain", OFFSET(again), AV_OPT_TYPE_BOOL, {.i64=1}, 0, 1, AF },
519+ { NULL }
520+};
521+
522+AVFILTER_DEFINE_CLASS(afir);
523+
524+AVFilter ff_af_afir = {
525+ .name = "afir",
526+ .description = NULL_IF_CONFIG_SMALL("Apply Finite Impulse Response filter with supplied coefficients in 2nd stream."),
527+ .priv_size = sizeof(AudioFIRContext),
528+ .priv_class = &afir_class,
529+ .query_formats = query_formats,
530+ .init = init,
531+ .uninit = uninit,
532+ .inputs = afir_inputs,
533+ .outputs = afir_outputs,
534+ .flags = AVFILTER_FLAG_SLICE_THREADS,
535+};
libavfilter/af_afir.h+83−0
@@ -0,0 +1,83 @@
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+#ifndef AVFILTER_AFIR_H
22+#define AVFILTER_AFIR_H
23+
24+#include "libavutil/audio_fifo.h"
25+#include "libavutil/common.h"
26+#include "libavutil/float_dsp.h"
27+#include "libavutil/opt.h"
28+#include "libavcodec/avfft.h"
29+
30+#include "audio.h"
31+#include "avfilter.h"
32+#include "formats.h"
33+#include "internal.h"
34+
35+#define MAX_IR_DURATION 30
36+
37+typedef struct AudioFIRContext {
38+ const AVClass *class;
39+
40+ float wet_gain;
41+ float dry_gain;
42+ float length;
43+ int again;
44+
45+ float gain;
46+
47+ int eof_coeffs;
48+ int have_coeffs;
49+ int nb_coeffs;
50+ int nb_taps;
51+ int part_size;
52+ int part_index;
53+ int coeff_size;
54+ int block_size;
55+ int nb_partitions;
56+ int nb_channels;
57+ int ir_length;
58+ int fft_length;
59+ int nb_coef_channels;
60+ int one2many;
61+ int nb_samples;
62+ int want_skip;
63+ int need_padding;
64+
65+ RDFTContext **rdft, **irdft;
66+ float **sum;
67+ float **block;
68+ FFTComplex **coeff;
69+
70+ AVAudioFifo *fifo[2];
71+ AVFrame *in[2];
72+ AVFrame *buffer;
73+ int64_t pts;
74+ int index;
75+
76+ AVFloatDSPContext *fdsp;
77+ void (*fcmul_add)(float *sum, const float *t, const float *c,
78+ ptrdiff_t len);
79+} AudioFIRContext;
80+
81+void ff_afir_init_x86(AudioFIRContext *s);
82+
83+#endif /* AVFILTER_AFIR_H */
libavfilter/allfilters.c+1−0
@@ -50,6 +50,7 @@ static void register_all(void) · 49 unmodified lines
5050 REGISTER_FILTER(AEVAL, aeval, af);
5151 REGISTER_FILTER(AFADE, afade, af);
5252 REGISTER_FILTER(AFFTFILT, afftfilt, af);
53+ REGISTER_FILTER(AFIR, afir, af);
5354 REGISTER_FILTER(AFORMAT, aformat, af);
5455 REGISTER_FILTER(AGATE, agate, af);
5556 REGISTER_FILTER(AINTERLEAVE, ainterleave, af);
libavfilter/version.h+1−1
@@ -30,7 +30,7 @@ · 29 unmodified lines
3030 #include "libavutil/version.h"
3131
3232 #define LIBAVFILTER_VERSION_MAJOR 6
33−#define LIBAVFILTER_VERSION_MINOR 88
33+#define LIBAVFILTER_VERSION_MINOR 89
3434 #define LIBAVFILTER_VERSION_MICRO 100
3535
3636 #define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \
libavfilter/x86/Makefile+2−0
@@ -1,3 +1,4 @@
1+OBJS-$(CONFIG_AFIR_FILTER) += x86/af_afir_init.o
12 OBJS-$(CONFIG_BLEND_FILTER) += x86/vf_blend_init.o
23 OBJS-$(CONFIG_BWDIF_FILTER) += x86/vf_bwdif_init.o
34 OBJS-$(CONFIG_COLORSPACE_FILTER) += x86/colorspacedsp_init.o
@@ -23,6 +24,7 @@ OBJS-$(CONFIG_VOLUME_FILTER) += x86/af_volume_init.o · 19 unmodified lines
2324 OBJS-$(CONFIG_W3FDIF_FILTER) += x86/vf_w3fdif_init.o
2425 OBJS-$(CONFIG_YADIF_FILTER) += x86/vf_yadif_init.o
2526
27+YASM-OBJS-$(CONFIG_AFIR_FILTER) += x86/af_afir.o
2628 YASM-OBJS-$(CONFIG_BLEND_FILTER) += x86/vf_blend.o
2729 YASM-OBJS-$(CONFIG_BWDIF_FILTER) += x86/vf_bwdif.o
2830 YASM-OBJS-$(CONFIG_COLORSPACE_FILTER) += x86/colorspacedsp.o
libavfilter/x86/af_afir.asm+60−0
@@ -0,0 +1,60 @@
1+;*****************************************************************************
2+;* x86-optimized functions for afir filter
3+;* Copyright (c) 2017 Paul B Mahol
4+;*
5+;* This file is part of FFmpeg.
6+;*
7+;* FFmpeg is free software; you can redistribute it and/or
8+;* modify it under the terms of the GNU Lesser General Public
9+;* License as published by the Free Software Foundation; either
10+;* version 2.1 of the License, or (at your option) any later version.
11+;*
12+;* FFmpeg is distributed in the hope that it will be useful,
13+;* but WITHOUT ANY WARRANTY; without even the implied warranty of
14+;* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15+;* Lesser General Public License for more details.
16+;*
17+;* You should have received a copy of the GNU Lesser General Public
18+;* License along with FFmpeg; if not, write to the Free Software
19+;* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
20+;******************************************************************************
21+
22+%include "libavutil/x86/x86util.asm"
23+
24+SECTION .text
25+
26+;------------------------------------------------------------------------------
27+; void ff_fcmul_add(float *sum, const float *t, const float *c, int len)
28+;------------------------------------------------------------------------------
29+
30+INIT_XMM sse3
31+cglobal fcmul_add, 4,4,6, sum, t, c, len
32+ shl lend, 3
33+ add lend, mmsize*2
34+ add tq, lenq
35+ add cq, lenq
36+ add sumq, lenq
37+ neg lenq
38+ALIGN 16
39+.loop:
40+ movsldup m0, [tq + lenq]
41+ movsldup m3, [tq + lenq+mmsize]
42+ movaps m1, [cq + lenq]
43+ movaps m4, [cq + lenq+mmsize]
44+ mulps m0, m1
45+ mulps m3, m4
46+ shufps m1, m1, 0xb1
47+ shufps m4, m4, 0xb1
48+ movshdup m2, [tq + lenq]
49+ movshdup m5, [tq + lenq+mmsize]
50+ mulps m2, m1
51+ mulps m5, m4
52+ addsubps m0, m2
53+ addsubps m3, m5
54+ addps m0, [sumq + lenq]
55+ addps m3, [sumq + lenq+mmsize]
56+ movaps [sumq + lenq], m0
57+ movaps [sumq + lenq+mmsize], m3
58+ add lenq, mmsize*2
59+ jl .loop
60+ REP_RET
libavfilter/x86/af_afir_init.c+35−0
@@ -0,0 +1,35 @@
1+/*
2+ * This file is part of FFmpeg.
3+ *
4+ * FFmpeg is free software; you can redistribute it and/or
5+ * modify it under the terms of the GNU Lesser General Public
6+ * License as published by the Free Software Foundation; either
7+ * version 2.1 of the License, or (at your option) any later version.
8+ *
9+ * FFmpeg is distributed in the hope that it will be useful,
10+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
11+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
12+ * Lesser General Public License for more details.
13+ *
14+ * You should have received a copy of the GNU Lesser General Public
15+ * License along with FFmpeg; if not, write to the Free Software
16+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
17+ */
18+
19+#include "config.h"
20+#include "libavutil/attributes.h"
21+#include "libavutil/cpu.h"
22+#include "libavutil/x86/cpu.h"
23+#include "libavfilter/af_afir.h"
24+
25+void ff_fcmul_add_sse3(float *sum, const float *t, const float *c,
26+ ptrdiff_t len);
27+
28+av_cold void ff_afir_init_x86(AudioFIRContext *s)
29+{
30+ int cpu_flags = av_get_cpu_flags();
31+
32+ if (EXTERNAL_SSE3(cpu_flags)) {
33+ s->fcmul_add = ff_fcmul_add_sse3;
34+ }
35+}

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