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FFmpeg/libavfilter/af_afir.c

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  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/avstring.h"

  29. #include "libavutil/common.h"

  30. #include "libavutil/float_dsp.h"

  31. #include "libavutil/intreadwrite.h"

  32. #include "libavutil/opt.h"

  33. #include "libavutil/xga_font_data.h"

  34. #include "libavcodec/avfft.h"

  35. 

  36. #include "audio.h"

  37. #include "avfilter.h"

  38. #include "filters.h"

  39. #include "formats.h"

  40. #include "internal.h"

  41. #include "af_afir.h"

  42. 

  43. static void fcmul_add_c(float *sum, const float *t, const float *c, ptrdiff_t len)

  44. {

  45.     int n;

  46. 

  47.     for (n = 0; n < len; n++) {

  48.         const float cre = c[2 * n    ];

  49.         const float cim = c[2 * n + 1];

  50.         const float tre = t[2 * n    ];

  51.         const float tim = t[2 * n + 1];

  52. 

  53.         sum[2 * n    ] += tre * cre - tim * cim;

  54.         sum[2 * n + 1] += tre * cim + tim * cre;

  55.     }

  56. 

  57.     sum[2 * n] += t[2 * n] * c[2 * n];

  58. }

  59. 

  60. static void direct(const float *in, const FFTComplex *ir, int len, float *out)

  61. {

  62.     for (int n = 0; n < len; n++)

  63.         for (int m = 0; m <= n; m++)

  64.             out[n] += ir[m].re * in[n - m];

  65. }

  66. 

  67. static void fir_fadd(AudioFIRContext *s, float *dst, const float *src, int nb_samples)

  68. {

  69.     if ((nb_samples & 15) == 0 && nb_samples >= 16) {

  70.         s->fdsp->vector_fmac_scalar(dst, src, 1.f, nb_samples);

  71.     } else {

  72.         for (int n = 0; n < nb_samples; n++)

  73.             dst[n] += src[n];

  74.     }

  75. }

  76. 

  77. static int fir_quantum(AVFilterContext *ctx, AVFrame *out, int ch, int offset)

  78. {

  79.     AudioFIRContext *s = ctx->priv;

  80.     const float *in = (const float *)s->in->extended_data[ch] + offset;

  81.     float *block, *buf, *ptr = (float *)out->extended_data[ch] + offset;

  82.     const int nb_samples = FFMIN(s->min_part_size, out->nb_samples - offset);

  83.     int n, i, j;

  84. 

  85.     for (int segment = 0; segment < s->nb_segments; segment++) {

  86.         AudioFIRSegment *seg = &s->seg[segment];

  87.         float *src = (float *)seg->input->extended_data[ch];

  88.         float *dst = (float *)seg->output->extended_data[ch];

  89.         float *sum = (float *)seg->sum->extended_data[ch];

  90. 

  91.         if (s->min_part_size >= 8) {

  92.             s->fdsp->vector_fmul_scalar(src + seg->input_offset, in, s->dry_gain, FFALIGN(nb_samples, 4));

  93.             emms_c();

  94.         } else {

  95.             for (n = 0; n < nb_samples; n++)

  96.                 src[seg->input_offset + n] = in[n] * s->dry_gain;

  97.         }

  98. 

  99.         seg->output_offset[ch] += s->min_part_size;

  100.         if (seg->output_offset[ch] == seg->part_size) {

  101.             seg->output_offset[ch] = 0;

  102.         } else {

  103.             memmove(src, src + s->min_part_size, (seg->input_size - s->min_part_size) * sizeof(*src));

  104. 

  105.             dst += seg->output_offset[ch];

  106.             fir_fadd(s, ptr, dst, nb_samples);

  107.             continue;

  108.         }

  109. 

  110.         if (seg->part_size < 8) {

  111.             memset(dst, 0, sizeof(*dst) * seg->part_size * seg->nb_partitions);

  112. 

  113.             j = seg->part_index[ch];

  114. 

  115.             for (i = 0; i < seg->nb_partitions; i++) {

  116.                 const int coffset = j * seg->coeff_size;

  117.                 const FFTComplex *coeff = (const FFTComplex *)seg->coeff->extended_data[ch * !s->one2many] + coffset;

  118. 

  119.                 direct(src, coeff, nb_samples, dst);

  120. 

  121.                 if (j == 0)

  122.                     j = seg->nb_partitions;

  123.                 j--;

  124.             }

  125. 

  126.             seg->part_index[ch] = (seg->part_index[ch] + 1) % seg->nb_partitions;

  127. 

  128.             memmove(src, src + s->min_part_size, (seg->input_size - s->min_part_size) * sizeof(*src));

  129. 

  130.             for (n = 0; n < nb_samples; n++) {

  131.                 ptr[n] += dst[n];

  132.             }

  133.             continue;

  134.         }

  135. 

  136.         memset(sum, 0, sizeof(*sum) * seg->fft_length);

  137.         block = (float *)seg->block->extended_data[ch] + seg->part_index[ch] * seg->block_size;

  138.         memset(block + seg->part_size, 0, sizeof(*block) * (seg->fft_length - seg->part_size));

  139. 

  140.         memcpy(block, src, sizeof(*src) * seg->part_size);

  141. 

  142.         av_rdft_calc(seg->rdft[ch], block);

  143.         block[2 * seg->part_size] = block[1];

  144.         block[1] = 0;

  145. 

  146.         j = seg->part_index[ch];

  147. 

  148.         for (i = 0; i < seg->nb_partitions; i++) {

  149.             const int coffset = j * seg->coeff_size;

  150.             const float *block = (const float *)seg->block->extended_data[ch] + i * seg->block_size;

  151.             const FFTComplex *coeff = (const FFTComplex *)seg->coeff->extended_data[ch * !s->one2many] + coffset;

  152. 

  153.             s->afirdsp.fcmul_add(sum, block, (const float *)coeff, seg->part_size);

  154. 

  155.             if (j == 0)

  156.                 j = seg->nb_partitions;

  157.             j--;

  158.         }

  159. 

  160.         sum[1] = sum[2 * seg->part_size];

  161.         av_rdft_calc(seg->irdft[ch], sum);

  162. 

  163.         buf = (float *)seg->buffer->extended_data[ch];

  164.         fir_fadd(s, buf, sum, seg->part_size);

  165. 

  166.         memcpy(dst, buf, seg->part_size * sizeof(*dst));

  167. 

  168.         buf = (float *)seg->buffer->extended_data[ch];

  169.         memcpy(buf, sum + seg->part_size, seg->part_size * sizeof(*buf));

  170. 

  171.         seg->part_index[ch] = (seg->part_index[ch] + 1) % seg->nb_partitions;

  172. 

  173.         memmove(src, src + s->min_part_size, (seg->input_size - s->min_part_size) * sizeof(*src));

  174. 

  175.         fir_fadd(s, ptr, dst, nb_samples);

  176.     }

  177. 

  178.     if (s->min_part_size >= 8) {

  179.         s->fdsp->vector_fmul_scalar(ptr, ptr, s->wet_gain, FFALIGN(nb_samples, 4));

  180.         emms_c();

  181.     } else {

  182.         for (n = 0; n < nb_samples; n++)

  183.             ptr[n] *= s->wet_gain;

  184.     }

  185. 

  186.     return 0;

  187. }

  188. 

  189. static int fir_channel(AVFilterContext *ctx, AVFrame *out, int ch)

  190. {

  191.     AudioFIRContext *s = ctx->priv;

  192. 

  193.     for (int offset = 0; offset < out->nb_samples; offset += s->min_part_size) {

  194.         fir_quantum(ctx, out, ch, offset);

  195.     }

  196. 

  197.     return 0;

  198. }

  199. 

  200. static int fir_channels(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)

  201. {

  202.     AVFrame *out = arg;

  203.     const int start = (out->channels * jobnr) / nb_jobs;

  204.     const int end = (out->channels * (jobnr+1)) / nb_jobs;

  205. 

  206.     for (int ch = start; ch < end; ch++) {

  207.         fir_channel(ctx, out, ch);

  208.     }

  209. 

  210.     return 0;

  211. }

  212. 

  213. static int fir_frame(AudioFIRContext *s, AVFrame *in, AVFilterLink *outlink)

  214. {

  215.     AVFilterContext *ctx = outlink->src;

  216.     AVFrame *out = NULL;

  217. 

  218.     out = ff_get_audio_buffer(outlink, in->nb_samples);

  219.     if (!out) {

  220.         av_frame_free(&in);

  221.         return AVERROR(ENOMEM);

  222.     }

  223. 

  224.     if (s->pts == AV_NOPTS_VALUE)

  225.         s->pts = in->pts;

  226.     s->in = in;

  227.     ctx->internal->execute(ctx, fir_channels, out, NULL, FFMIN(outlink->channels,

  228.                                                                ff_filter_get_nb_threads(ctx)));

  229. 

  230.     out->pts = s->pts;

  231.     if (s->pts != AV_NOPTS_VALUE)

  232.         s->pts += av_rescale_q(out->nb_samples, (AVRational){1, outlink->sample_rate}, outlink->time_base);

  233. 

  234.     av_frame_free(&in);

  235.     s->in = NULL;

  236. 

  237.     return ff_filter_frame(outlink, out);

  238. }

  239. 

  240. static void drawtext(AVFrame *pic, int x, int y, const char *txt, uint32_t color)

  241. {

  242.     const uint8_t *font;

  243.     int font_height;

  244.     int i;

  245. 

  246.     font = avpriv_cga_font, font_height = 8;

  247. 

  248.     for (i = 0; txt[i]; i++) {

  249.         int char_y, mask;

  250. 

  251.         uint8_t *p = pic->data[0] + y * pic->linesize[0] + (x + i * 8) * 4;

  252.         for (char_y = 0; char_y < font_height; char_y++) {

  253.             for (mask = 0x80; mask; mask >>= 1) {

  254.                 if (font[txt[i] * font_height + char_y] & mask)

  255.                     AV_WL32(p, color);

  256.                 p += 4;

  257.             }

  258.             p += pic->linesize[0] - 8 * 4;

  259.         }

  260.     }

  261. }

  262. 

  263. static void draw_line(AVFrame *out, int x0, int y0, int x1, int y1, uint32_t color)

  264. {

  265.     int dx = FFABS(x1-x0);

  266.     int dy = FFABS(y1-y0), sy = y0 < y1 ? 1 : -1;

  267.     int err = (dx>dy ? dx : -dy) / 2, e2;

  268. 

  269.     for (;;) {

  270.         AV_WL32(out->data[0] + y0 * out->linesize[0] + x0 * 4, color);

  271. 

  272.         if (x0 == x1 && y0 == y1)

  273.             break;

  274. 

  275.         e2 = err;

  276. 

  277.         if (e2 >-dx) {

  278.             err -= dy;

  279.             x0--;

  280.         }

  281. 

  282.         if (e2 < dy) {

  283.             err += dx;

  284.             y0 += sy;

  285.         }

  286.     }

  287. }

  288. 

  289. static void draw_response(AVFilterContext *ctx, AVFrame *out)

  290. {

  291.     AudioFIRContext *s = ctx->priv;

  292.     float *mag, *phase, *delay, min = FLT_MAX, max = FLT_MIN;

  293.     float min_delay = FLT_MAX, max_delay = FLT_MIN;

  294.     int prev_ymag = -1, prev_yphase = -1, prev_ydelay = -1;

  295.     char text[32];

  296.     int channel, i, x;

  297. 

  298.     memset(out->data[0], 0, s->h * out->linesize[0]);

  299. 

  300.     phase = av_malloc_array(s->w, sizeof(*phase));

  301.     mag = av_malloc_array(s->w, sizeof(*mag));

  302.     delay = av_malloc_array(s->w, sizeof(*delay));

  303.     if (!mag || !phase || !delay)

  304.         goto end;

  305. 

  306.     channel = av_clip(s->ir_channel, 0, s->ir[s->selir]->channels - 1);

  307.     for (i = 0; i < s->w; i++) {

  308.         const float *src = (const float *)s->ir[s->selir]->extended_data[channel];

  309.         double w = i * M_PI / (s->w - 1);

  310.         double div, real_num = 0., imag_num = 0., real = 0., imag = 0.;

  311. 

  312.         for (x = 0; x < s->nb_taps; x++) {

  313.             real += cos(-x * w) * src[x];

  314.             imag += sin(-x * w) * src[x];

  315.             real_num += cos(-x * w) * src[x] * x;

  316.             imag_num += sin(-x * w) * src[x] * x;

  317.         }

  318. 

  319.         mag[i] = hypot(real, imag);

  320.         phase[i] = atan2(imag, real);

  321.         div = real * real + imag * imag;

  322.         delay[i] = (real_num * real + imag_num * imag) / div;

  323.         min = fminf(min, mag[i]);

  324.         max = fmaxf(max, mag[i]);

  325.         min_delay = fminf(min_delay, delay[i]);

  326.         max_delay = fmaxf(max_delay, delay[i]);

  327.     }

  328. 

  329.     for (i = 0; i < s->w; i++) {

  330.         int ymag = mag[i] / max * (s->h - 1);

  331.         int ydelay = (delay[i] - min_delay) / (max_delay - min_delay) * (s->h - 1);

  332.         int yphase = (0.5 * (1. + phase[i] / M_PI)) * (s->h - 1);

  333. 

  334.         ymag = s->h - 1 - av_clip(ymag, 0, s->h - 1);

  335.         yphase = s->h - 1 - av_clip(yphase, 0, s->h - 1);

  336.         ydelay = s->h - 1 - av_clip(ydelay, 0, s->h - 1);

  337. 

  338.         if (prev_ymag < 0)

  339.             prev_ymag = ymag;

  340.         if (prev_yphase < 0)

  341.             prev_yphase = yphase;

  342.         if (prev_ydelay < 0)

  343.             prev_ydelay = ydelay;

  344. 

  345.         draw_line(out, i,   ymag, FFMAX(i - 1, 0),   prev_ymag, 0xFFFF00FF);

  346.         draw_line(out, i, yphase, FFMAX(i - 1, 0), prev_yphase, 0xFF00FF00);

  347.         draw_line(out, i, ydelay, FFMAX(i - 1, 0), prev_ydelay, 0xFF00FFFF);

  348. 

  349.         prev_ymag   = ymag;

  350.         prev_yphase = yphase;

  351.         prev_ydelay = ydelay;

  352.     }

  353. 

  354.     if (s->w > 400 && s->h > 100) {

  355.         drawtext(out, 2, 2, "Max Magnitude:", 0xDDDDDDDD);

  356.         snprintf(text, sizeof(text), "%.2f", max);

  357.         drawtext(out, 15 * 8 + 2, 2, text, 0xDDDDDDDD);

  358. 

  359.         drawtext(out, 2, 12, "Min Magnitude:", 0xDDDDDDDD);

  360.         snprintf(text, sizeof(text), "%.2f", min);

  361.         drawtext(out, 15 * 8 + 2, 12, text, 0xDDDDDDDD);

  362. 

  363.         drawtext(out, 2, 22, "Max Delay:", 0xDDDDDDDD);

  364.         snprintf(text, sizeof(text), "%.2f", max_delay);

  365.         drawtext(out, 11 * 8 + 2, 22, text, 0xDDDDDDDD);

  366. 

  367.         drawtext(out, 2, 32, "Min Delay:", 0xDDDDDDDD);

  368.         snprintf(text, sizeof(text), "%.2f", min_delay);

  369.         drawtext(out, 11 * 8 + 2, 32, text, 0xDDDDDDDD);

  370.     }

  371. 

  372. end:

  373.     av_free(delay);

  374.     av_free(phase);

  375.     av_free(mag);

  376. }

  377. 

  378. static int init_segment(AVFilterContext *ctx, AudioFIRSegment *seg,

  379.                         int offset, int nb_partitions, int part_size)

  380. {

  381.     AudioFIRContext *s = ctx->priv;

  382. 

  383.     seg->rdft  = av_calloc(ctx->inputs[0]->channels, sizeof(*seg->rdft));

  384.     seg->irdft = av_calloc(ctx->inputs[0]->channels, sizeof(*seg->irdft));

  385.     if (!seg->rdft || !seg->irdft)

  386.         return AVERROR(ENOMEM);

  387. 

  388.     seg->fft_length    = part_size * 2 + 1;

  389.     seg->part_size     = part_size;

  390.     seg->block_size    = FFALIGN(seg->fft_length, 32);

  391.     seg->coeff_size    = FFALIGN(seg->part_size + 1, 32);

  392.     seg->nb_partitions = nb_partitions;

  393.     seg->input_size    = offset + s->min_part_size;

  394.     seg->input_offset  = offset;

  395. 

  396.     seg->part_index    = av_calloc(ctx->inputs[0]->channels, sizeof(*seg->part_index));

  397.     seg->output_offset = av_calloc(ctx->inputs[0]->channels, sizeof(*seg->output_offset));

  398.     if (!seg->part_index || !seg->output_offset)

  399.         return AVERROR(ENOMEM);

  400. 

  401.     for (int ch = 0; ch < ctx->inputs[0]->channels && part_size >= 8; ch++) {

  402.         seg->rdft[ch]  = av_rdft_init(av_log2(2 * part_size), DFT_R2C);

  403.         seg->irdft[ch] = av_rdft_init(av_log2(2 * part_size), IDFT_C2R);

  404.         if (!seg->rdft[ch] || !seg->irdft[ch])

  405.             return AVERROR(ENOMEM);

  406.     }

  407. 

  408.     seg->sum    = ff_get_audio_buffer(ctx->inputs[0], seg->fft_length);

  409.     seg->block  = ff_get_audio_buffer(ctx->inputs[0], seg->nb_partitions * seg->block_size);

  410.     seg->buffer = ff_get_audio_buffer(ctx->inputs[0], seg->part_size);

  411.     seg->coeff  = ff_get_audio_buffer(ctx->inputs[1 + s->selir], seg->nb_partitions * seg->coeff_size * 2);

  412.     seg->input  = ff_get_audio_buffer(ctx->inputs[0], seg->input_size);

  413.     seg->output = ff_get_audio_buffer(ctx->inputs[0], seg->part_size);

  414.     if (!seg->buffer || !seg->sum || !seg->block || !seg->coeff || !seg->input || !seg->output)

  415.         return AVERROR(ENOMEM);

  416. 

  417.     return 0;

  418. }

  419. 

  420. static void uninit_segment(AVFilterContext *ctx, AudioFIRSegment *seg)

  421. {

  422.     AudioFIRContext *s = ctx->priv;

  423. 

  424.     if (seg->rdft) {

  425.         for (int ch = 0; ch < s->nb_channels; ch++) {

  426.             av_rdft_end(seg->rdft[ch]);

  427.         }

  428.     }

  429.     av_freep(&seg->rdft);

  430. 

  431.     if (seg->irdft) {

  432.         for (int ch = 0; ch < s->nb_channels; ch++) {

  433.             av_rdft_end(seg->irdft[ch]);

  434.         }

  435.     }

  436.     av_freep(&seg->irdft);

  437. 

  438.     av_freep(&seg->output_offset);

  439.     av_freep(&seg->part_index);

  440. 

  441.     av_frame_free(&seg->block);

  442.     av_frame_free(&seg->sum);

  443.     av_frame_free(&seg->buffer);

  444.     av_frame_free(&seg->coeff);

  445.     av_frame_free(&seg->input);

  446.     av_frame_free(&seg->output);

  447.     seg->input_size = 0;

  448. }

  449. 

  450. static int convert_coeffs(AVFilterContext *ctx)

  451. {

  452.     AudioFIRContext *s = ctx->priv;

  453.     int ret, i, ch, n, cur_nb_taps;

  454.     float power = 0;

  455. 

  456.     if (!s->nb_taps) {

  457.         int part_size, max_part_size;

  458.         int left, offset = 0;

  459. 

  460.         s->nb_taps = ff_inlink_queued_samples(ctx->inputs[1 + s->selir]);

  461.         if (s->nb_taps <= 0)

  462.             return AVERROR(EINVAL);

  463. 

  464.         if (s->minp > s->maxp) {

  465.             s->maxp = s->minp;

  466.         }

  467. 

  468.         left = s->nb_taps;

  469.         part_size = 1 << av_log2(s->minp);

  470.         max_part_size = 1 << av_log2(s->maxp);

  471. 

  472.         s->min_part_size = part_size;

  473. 

  474.         for (i = 0; left > 0; i++) {

  475.             int step = part_size == max_part_size ? INT_MAX : 1 + (i == 0);

  476.             int nb_partitions = FFMIN(step, (left + part_size - 1) / part_size);

  477. 

  478.             s->nb_segments = i + 1;

  479.             ret = init_segment(ctx, &s->seg[i], offset, nb_partitions, part_size);

  480.             if (ret < 0)

  481.                 return ret;

  482.             offset += nb_partitions * part_size;

  483.             left -= nb_partitions * part_size;

  484.             part_size *= 2;

  485.             part_size = FFMIN(part_size, max_part_size);

  486.         }

  487.     }

  488. 

  489.     if (!s->ir[s->selir]) {

  490.         ret = ff_inlink_consume_samples(ctx->inputs[1 + s->selir], s->nb_taps, s->nb_taps, &s->ir[s->selir]);

  491.         if (ret < 0)

  492.             return ret;

  493.         if (ret == 0)

  494.             return AVERROR_BUG;

  495.     }

  496. 

  497.     if (s->response)

  498.         draw_response(ctx, s->video);

  499. 

  500.     s->gain = 1;

  501.     cur_nb_taps = s->ir[s->selir]->nb_samples;

  502. 

  503.     switch (s->gtype) {

  504.     case -1:

  505.         /* nothing to do */

  506.         break;

  507.     case 0:

  508.         for (ch = 0; ch < ctx->inputs[1 + s->selir]->channels; ch++) {

  509.             float *time = (float *)s->ir[s->selir]->extended_data[!s->one2many * ch];

  510. 

  511.             for (i = 0; i < cur_nb_taps; i++)

  512.                 power += FFABS(time[i]);

  513.         }

  514.         s->gain = ctx->inputs[1 + s->selir]->channels / power;

  515.         break;

  516.     case 1:

  517.         for (ch = 0; ch < ctx->inputs[1 + s->selir]->channels; ch++) {

  518.             float *time = (float *)s->ir[s->selir]->extended_data[!s->one2many * ch];

  519. 

  520.             for (i = 0; i < cur_nb_taps; i++)

  521.                 power += time[i];

  522.         }

  523.         s->gain = ctx->inputs[1 + s->selir]->channels / power;

  524.         break;

  525.     case 2:

  526.         for (ch = 0; ch < ctx->inputs[1 + s->selir]->channels; ch++) {

  527.             float *time = (float *)s->ir[s->selir]->extended_data[!s->one2many * ch];

  528. 

  529.             for (i = 0; i < cur_nb_taps; i++)

  530.                 power += time[i] * time[i];

  531.         }

  532.         s->gain = sqrtf(ch / power);

  533.         break;

  534.     default:

  535.         return AVERROR_BUG;

  536.     }

  537. 

  538.     s->gain = FFMIN(s->gain * s->ir_gain, 1.f);

  539.     av_log(ctx, AV_LOG_DEBUG, "power %f, gain %f\n", power, s->gain);

  540.     for (ch = 0; ch < ctx->inputs[1 + s->selir]->channels; ch++) {

  541.         float *time = (float *)s->ir[s->selir]->extended_data[!s->one2many * ch];

  542. 

  543.         s->fdsp->vector_fmul_scalar(time, time, s->gain, FFALIGN(cur_nb_taps, 4));

  544.     }

  545. 

  546.     av_log(ctx, AV_LOG_DEBUG, "nb_taps: %d\n", cur_nb_taps);

  547.     av_log(ctx, AV_LOG_DEBUG, "nb_segments: %d\n", s->nb_segments);

  548. 

  549.     for (ch = 0; ch < ctx->inputs[1 + s->selir]->channels; ch++) {

  550.         float *time = (float *)s->ir[s->selir]->extended_data[!s->one2many * ch];

  551.         int toffset = 0;

  552. 

  553.         for (i = FFMAX(1, s->length * s->nb_taps); i < s->nb_taps; i++)

  554.             time[i] = 0;

  555. 

  556.         av_log(ctx, AV_LOG_DEBUG, "channel: %d\n", ch);

  557. 

  558.         for (int segment = 0; segment < s->nb_segments; segment++) {

  559.             AudioFIRSegment *seg = &s->seg[segment];

  560.             float *block = (float *)seg->block->extended_data[ch];

  561.             FFTComplex *coeff = (FFTComplex *)seg->coeff->extended_data[ch];

  562. 

  563.             av_log(ctx, AV_LOG_DEBUG, "segment: %d\n", segment);

  564. 

  565.             for (i = 0; i < seg->nb_partitions; i++) {

  566.                 const float scale = 1.f / seg->part_size;

  567.                 const int coffset = i * seg->coeff_size;

  568.                 const int remaining = s->nb_taps - toffset;

  569.                 const int size = remaining >= seg->part_size ? seg->part_size : remaining;

  570. 

  571.                 if (size < 8) {

  572.                     for (n = 0; n < size; n++)

  573.                         coeff[coffset + n].re = time[toffset + n];

  574. 

  575.                     toffset += size;

  576.                     continue;

  577.                 }

  578. 

  579.                 memset(block, 0, sizeof(*block) * seg->fft_length);

  580.                 memcpy(block, time + toffset, size * sizeof(*block));

  581. 

  582.                 av_rdft_calc(seg->rdft[0], block);

  583. 

  584.                 coeff[coffset].re = block[0] * scale;

  585.                 coeff[coffset].im = 0;

  586.                 for (n = 1; n < seg->part_size; n++) {

  587.                     coeff[coffset + n].re = block[2 * n] * scale;

  588.                     coeff[coffset + n].im = block[2 * n + 1] * scale;

  589.                 }

  590.                 coeff[coffset + seg->part_size].re = block[1] * scale;

  591.                 coeff[coffset + seg->part_size].im = 0;

  592. 

  593.                 toffset += size;

  594.             }

  595. 

  596.             av_log(ctx, AV_LOG_DEBUG, "nb_partitions: %d\n", seg->nb_partitions);

  597.             av_log(ctx, AV_LOG_DEBUG, "partition size: %d\n", seg->part_size);

  598.             av_log(ctx, AV_LOG_DEBUG, "block size: %d\n", seg->block_size);

  599.             av_log(ctx, AV_LOG_DEBUG, "fft_length: %d\n", seg->fft_length);

  600.             av_log(ctx, AV_LOG_DEBUG, "coeff_size: %d\n", seg->coeff_size);

  601.             av_log(ctx, AV_LOG_DEBUG, "input_size: %d\n", seg->input_size);

  602.             av_log(ctx, AV_LOG_DEBUG, "input_offset: %d\n", seg->input_offset);

  603.         }

  604.     }

  605. 

  606.     s->have_coeffs = 1;

  607. 

  608.     return 0;

  609. }

  610. 

  611. static int check_ir(AVFilterLink *link)

  612. {

  613.     AVFilterContext *ctx = link->dst;

  614.     AudioFIRContext *s = ctx->priv;

  615.     int nb_taps, max_nb_taps;

  616. 

  617.     nb_taps = ff_inlink_queued_samples(link);

  618.     max_nb_taps = s->max_ir_len * ctx->outputs[0]->sample_rate;

  619.     if (nb_taps > max_nb_taps) {

  620.         av_log(ctx, AV_LOG_ERROR, "Too big number of coefficients: %d > %d.\n", nb_taps, max_nb_taps);

  621.         return AVERROR(EINVAL);

  622.     }

  623. 

  624.     return 0;

  625. }

  626. 

  627. static int activate(AVFilterContext *ctx)

  628. {

  629.     AudioFIRContext *s = ctx->priv;

  630.     AVFilterLink *outlink = ctx->outputs[0];

  631.     int ret, status, available, wanted;

  632.     AVFrame *in = NULL;

  633.     int64_t pts;

  634. 

  635.     FF_FILTER_FORWARD_STATUS_BACK_ALL(ctx->outputs[0], ctx);

  636.     if (s->response)

  637.         FF_FILTER_FORWARD_STATUS_BACK_ALL(ctx->outputs[1], ctx);

  638.     if (!s->eof_coeffs[s->selir]) {

  639.         ret = check_ir(ctx->inputs[1 + s->selir]);

  640.         if (ret < 0)

  641.             return ret;

  642. 

  643.         if (ff_outlink_get_status(ctx->inputs[1 + s->selir]) == AVERROR_EOF)

  644.             s->eof_coeffs[s->selir] = 1;

  645. 

  646.         if (!s->eof_coeffs[s->selir]) {

  647.             if (ff_outlink_frame_wanted(ctx->outputs[0]))

  648.                 ff_inlink_request_frame(ctx->inputs[1 + s->selir]);

  649.             else if (s->response && ff_outlink_frame_wanted(ctx->outputs[1]))

  650.                 ff_inlink_request_frame(ctx->inputs[1 + s->selir]);

  651.             return 0;

  652.         }

  653.     }

  654. 

  655.     if (!s->have_coeffs && s->eof_coeffs[s->selir]) {

  656.         ret = convert_coeffs(ctx);

  657.         if (ret < 0)

  658.             return ret;

  659.     }

  660. 

  661.     available = ff_inlink_queued_samples(ctx->inputs[0]);

  662.     wanted = FFMAX(s->min_part_size, (available / s->min_part_size) * s->min_part_size);

  663.     ret = ff_inlink_consume_samples(ctx->inputs[0], wanted, wanted, &in);

  664.     if (ret > 0)

  665.         ret = fir_frame(s, in, outlink);

  666. 

  667.     if (ret < 0)

  668.         return ret;

  669. 

  670.     if (s->response && s->have_coeffs) {

  671.         int64_t old_pts = s->video->pts;

  672.         int64_t new_pts = av_rescale_q(s->pts, ctx->inputs[0]->time_base, ctx->outputs[1]->time_base);

  673. 

  674.         if (ff_outlink_frame_wanted(ctx->outputs[1]) && old_pts < new_pts) {

  675.             AVFrame *clone;

  676.             s->video->pts = new_pts;

  677.             clone = av_frame_clone(s->video);

  678.             if (!clone)

  679.                 return AVERROR(ENOMEM);

  680.             return ff_filter_frame(ctx->outputs[1], clone);

  681.         }

  682.     }

  683. 

  684.     if (ff_inlink_queued_samples(ctx->inputs[0]) >= s->min_part_size) {

  685.         ff_filter_set_ready(ctx, 10);

  686.         return 0;

  687.     }

  688. 

  689.     if (ff_inlink_acknowledge_status(ctx->inputs[0], &status, &pts)) {

  690.         if (status == AVERROR_EOF) {

  691.             ff_outlink_set_status(ctx->outputs[0], status, pts);

  692.             if (s->response)

  693.                 ff_outlink_set_status(ctx->outputs[1], status, pts);

  694.             return 0;

  695.         }

  696.     }

  697. 

  698.     if (ff_outlink_frame_wanted(ctx->outputs[0]) &&

  699.         !ff_outlink_get_status(ctx->inputs[0])) {

  700.         ff_inlink_request_frame(ctx->inputs[0]);

  701.         return 0;

  702.     }

  703. 

  704.     if (s->response &&

  705.         ff_outlink_frame_wanted(ctx->outputs[1]) &&

  706.         !ff_outlink_get_status(ctx->inputs[0])) {

  707.         ff_inlink_request_frame(ctx->inputs[0]);

  708.         return 0;

  709.     }

  710. 

  711.     return FFERROR_NOT_READY;

  712. }

  713. 

  714. static int query_formats(AVFilterContext *ctx)

  715. {

  716.     AudioFIRContext *s = ctx->priv;

  717.     AVFilterFormats *formats;

  718.     AVFilterChannelLayouts *layouts;

  719.     static const enum AVSampleFormat sample_fmts[] = {

  720.         AV_SAMPLE_FMT_FLTP,

  721.         AV_SAMPLE_FMT_NONE

  722.     };

  723.     static const enum AVPixelFormat pix_fmts[] = {

  724.         AV_PIX_FMT_RGB0,

  725.         AV_PIX_FMT_NONE

  726.     };

  727.     int ret;

  728. 

  729.     if (s->response) {

  730.         AVFilterLink *videolink = ctx->outputs[1];

  731.         formats = ff_make_format_list(pix_fmts);

  732.         if ((ret = ff_formats_ref(formats, &videolink->incfg.formats)) < 0)

  733.             return ret;

  734.     }

  735. 

  736.     layouts = ff_all_channel_counts();

  737.     if (!layouts)

  738.         return AVERROR(ENOMEM);

  739. 

  740.     if (s->ir_format) {

  741.         ret = ff_set_common_channel_layouts(ctx, layouts);

  742.         if (ret < 0)

  743.             return ret;

  744.     } else {

  745.         AVFilterChannelLayouts *mono = NULL;

  746. 

  747.         if ((ret = ff_channel_layouts_ref(layouts, &ctx->inputs[0]->outcfg.channel_layouts)) < 0)

  748.             return ret;

  749.         if ((ret = ff_channel_layouts_ref(layouts, &ctx->outputs[0]->incfg.channel_layouts)) < 0)

  750.             return ret;

  751. 

  752.         ret = ff_add_channel_layout(&mono, AV_CH_LAYOUT_MONO);

  753.         if (ret)

  754.             return ret;

  755.         for (int i = 1; i < ctx->nb_inputs; i++) {

  756.             if ((ret = ff_channel_layouts_ref(mono, &ctx->inputs[i]->outcfg.channel_layouts)) < 0)

  757.                 return ret;

  758.         }

  759.     }

  760. 

  761.     formats = ff_make_format_list(sample_fmts);

  762.     if ((ret = ff_set_common_formats(ctx, formats)) < 0)

  763.         return ret;

  764. 

  765.     formats = ff_all_samplerates();

  766.     return ff_set_common_samplerates(ctx, formats);

  767. }

  768. 

  769. static int config_output(AVFilterLink *outlink)

  770. {

  771.     AVFilterContext *ctx = outlink->src;

  772.     AudioFIRContext *s = ctx->priv;

  773. 

  774.     s->one2many = ctx->inputs[1 + s->selir]->channels == 1;

  775.     outlink->sample_rate = ctx->inputs[0]->sample_rate;

  776.     outlink->time_base   = ctx->inputs[0]->time_base;

  777.     outlink->channel_layout = ctx->inputs[0]->channel_layout;

  778.     outlink->channels = ctx->inputs[0]->channels;

  779. 

  780.     s->nb_channels = outlink->channels;

  781.     s->nb_coef_channels = ctx->inputs[1 + s->selir]->channels;

  782.     s->pts = AV_NOPTS_VALUE;

  783. 

  784.     return 0;

  785. }

  786. 

  787. static av_cold void uninit(AVFilterContext *ctx)

  788. {

  789.     AudioFIRContext *s = ctx->priv;

  790. 

  791.     for (int i = 0; i < s->nb_segments; i++) {

  792.         uninit_segment(ctx, &s->seg[i]);

  793.     }

  794. 

  795.     av_freep(&s->fdsp);

  796. 

  797.     for (int i = 0; i < s->nb_irs; i++) {

  798.         av_frame_free(&s->ir[i]);

  799.     }

  800. 

  801.     for (unsigned i = 1; i < ctx->nb_inputs; i++)

  802.         av_freep(&ctx->input_pads[i].name);

  803. 

  804.     av_frame_free(&s->video);

  805. }

  806. 

  807. static int config_video(AVFilterLink *outlink)

  808. {

  809.     AVFilterContext *ctx = outlink->src;

  810.     AudioFIRContext *s = ctx->priv;

  811. 

  812.     outlink->sample_aspect_ratio = (AVRational){1,1};

  813.     outlink->w = s->w;

  814.     outlink->h = s->h;

  815.     outlink->frame_rate = s->frame_rate;

  816.     outlink->time_base = av_inv_q(outlink->frame_rate);

  817. 

  818.     av_frame_free(&s->video);

  819.     s->video = ff_get_video_buffer(outlink, outlink->w, outlink->h);

  820.     if (!s->video)

  821.         return AVERROR(ENOMEM);

  822. 

  823.     return 0;

  824. }

  825. 

  826. void ff_afir_init(AudioFIRDSPContext *dsp)

  827. {

  828.     dsp->fcmul_add = fcmul_add_c;

  829. 

  830.     if (ARCH_X86)

  831.         ff_afir_init_x86(dsp);

  832. }

  833. 

  834. static av_cold int init(AVFilterContext *ctx)

  835. {

  836.     AudioFIRContext *s = ctx->priv;

  837.     AVFilterPad pad, vpad;

  838.     int ret;

  839. 

  840.     pad = (AVFilterPad) {

  841.         .name = "main",

  842.         .type = AVMEDIA_TYPE_AUDIO,

  843.     };

  844. 

  845.     ret = ff_insert_inpad(ctx, 0, &pad);

  846.     if (ret < 0)

  847.         return ret;

  848. 

  849.     for (int n = 0; n < s->nb_irs; n++) {

  850.         pad = (AVFilterPad) {

  851.             .name = av_asprintf("ir%d", n),

  852.             .type = AVMEDIA_TYPE_AUDIO,

  853.         };

  854. 

  855.         if (!pad.name)

  856.             return AVERROR(ENOMEM);

  857. 

  858.         ret = ff_insert_inpad(ctx, n + 1, &pad);

  859.         if (ret < 0) {

  860.             av_freep(&pad.name);

  861.             return ret;

  862.         }

  863.     }

  864. 

  865.     pad = (AVFilterPad) {

  866.         .name          = "default",

  867.         .type          = AVMEDIA_TYPE_AUDIO,

  868.         .config_props  = config_output,

  869.     };

  870. 

  871.     ret = ff_insert_outpad(ctx, 0, &pad);

  872.     if (ret < 0)

  873.         return ret;

  874. 

  875.     if (s->response) {

  876.         vpad = (AVFilterPad){

  877.             .name         = "filter_response",

  878.             .type         = AVMEDIA_TYPE_VIDEO,

  879.             .config_props = config_video,

  880.         };

  881. 

  882.         ret = ff_insert_outpad(ctx, 1, &vpad);

  883.         if (ret < 0)

  884.             return ret;

  885.     }

  886. 

  887.     s->fdsp = avpriv_float_dsp_alloc(0);

  888.     if (!s->fdsp)

  889.         return AVERROR(ENOMEM);

  890. 

  891.     ff_afir_init(&s->afirdsp);

  892. 

  893.     return 0;

  894. }

  895. 

  896. static int process_command(AVFilterContext *ctx,

  897.                            const char *cmd,

  898.                            const char *arg,

  899.                            char *res,

  900.                            int res_len,

  901.                            int flags)

  902. {

  903.     AudioFIRContext *s = ctx->priv;

  904.     int prev_ir = s->selir;

  905.     int ret = ff_filter_process_command(ctx, cmd, arg, res, res_len, flags);

  906. 

  907.     if (ret < 0)

  908.         return ret;

  909. 

  910.     s->selir = FFMIN(s->nb_irs - 1, s->selir);

  911. 

  912.     if (prev_ir != s->selir) {

  913.         s->have_coeffs = 0;

  914.     }

  915. 

  916.     return 0;

  917. }

  918. 

  919. #define AF AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM

  920. #define AFR AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM

  921. #define VF AV_OPT_FLAG_VIDEO_PARAM|AV_OPT_FLAG_FILTERING_PARAM

  922. #define OFFSET(x) offsetof(AudioFIRContext, x)

  923. 

  924. static const AVOption afir_options[] = {

  925.     { "dry",    "set dry gain",      OFFSET(dry_gain),   AV_OPT_TYPE_FLOAT, {.dbl=1},    0, 10, AF },

  926.     { "wet",    "set wet gain",      OFFSET(wet_gain),   AV_OPT_TYPE_FLOAT, {.dbl=1},    0, 10, AF },

  927.     { "length", "set IR length",     OFFSET(length),     AV_OPT_TYPE_FLOAT, {.dbl=1},    0,  1, AF },

  928.     { "gtype",  "set IR auto gain type",OFFSET(gtype),   AV_OPT_TYPE_INT,   {.i64=0},   -1,  2, AF, "gtype" },

  929.     {  "none",  "without auto gain", 0,                  AV_OPT_TYPE_CONST, {.i64=-1},   0,  0, AF, "gtype" },

  930.     {  "peak",  "peak gain",         0,                  AV_OPT_TYPE_CONST, {.i64=0},    0,  0, AF, "gtype" },

  931.     {  "dc",    "DC gain",           0,                  AV_OPT_TYPE_CONST, {.i64=1},    0,  0, AF, "gtype" },

  932.     {  "gn",    "gain to noise",     0,                  AV_OPT_TYPE_CONST, {.i64=2},    0,  0, AF, "gtype" },

  933.     { "irgain", "set IR gain",       OFFSET(ir_gain),    AV_OPT_TYPE_FLOAT, {.dbl=1},    0,  1, AF },

  934.     { "irfmt",  "set IR format",     OFFSET(ir_format),  AV_OPT_TYPE_INT,   {.i64=1},    0,  1, AF, "irfmt" },

  935.     {  "mono",  "single channel",    0,                  AV_OPT_TYPE_CONST, {.i64=0},    0,  0, AF, "irfmt" },

  936.     {  "input", "same as input",     0,                  AV_OPT_TYPE_CONST, {.i64=1},    0,  0, AF, "irfmt" },

  937.     { "maxir",  "set max IR length", OFFSET(max_ir_len), AV_OPT_TYPE_FLOAT, {.dbl=30}, 0.1, 60, AF },

  938.     { "response", "show IR frequency response", OFFSET(response), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, VF },

  939.     { "channel", "set IR channel to display frequency response", OFFSET(ir_channel), AV_OPT_TYPE_INT, {.i64=0}, 0, 1024, VF },

  940.     { "size",   "set video size",    OFFSET(w),          AV_OPT_TYPE_IMAGE_SIZE, {.str = "hd720"}, 0, 0, VF },

  941.     { "rate",   "set video rate",    OFFSET(frame_rate), AV_OPT_TYPE_VIDEO_RATE, {.str = "25"}, 0, INT32_MAX, VF },

  942.     { "minp",   "set min partition size", OFFSET(minp),  AV_OPT_TYPE_INT,   {.i64=8192}, 1, 32768, AF },

  943.     { "maxp",   "set max partition size", OFFSET(maxp),  AV_OPT_TYPE_INT,   {.i64=8192}, 8, 32768, AF },

  944.     { "nbirs",  "set number of input IRs",OFFSET(nb_irs),AV_OPT_TYPE_INT,   {.i64=1},    1,    32, AF },

  945.     { "ir",     "select IR",              OFFSET(selir), AV_OPT_TYPE_INT,   {.i64=0},    0,    31, AFR },

  946.     { NULL }

  947. };

  948. 

  949. AVFILTER_DEFINE_CLASS(afir);

  950. 

  951. AVFilter ff_af_afir = {

  952.     .name          = "afir",

  953.     .description   = NULL_IF_CONFIG_SMALL("Apply Finite Impulse Response filter with supplied coefficients in additional stream(s)."),

  954.     .priv_size     = sizeof(AudioFIRContext),

  955.     .priv_class    = &afir_class,

  956.     .query_formats = query_formats,

  957.     .init          = init,

  958.     .activate      = activate,

  959.     .uninit        = uninit,

  960.     .process_command = process_command,

  961.     .flags         = AVFILTER_FLAG_DYNAMIC_INPUTS  |

  962.                      AVFILTER_FLAG_DYNAMIC_OUTPUTS |

  963.                      AVFILTER_FLAG_SLICE_THREADS,

  964. };